diff --git a/build/gyp.mozbuild b/build/gyp.mozbuild
index 08c9976ac4..4e1247954e 100644
--- a/build/gyp.mozbuild
+++ b/build/gyp.mozbuild
@@ -36,6 +36,8 @@ gyp_vars = {
     # don't use openssl
     'use_openssl': 0,
 
+    'debug': 1 if CONFIG['DEBUG'] else 0,
+
     'use_x11': 1 if CONFIG['MOZ_X11'] else 0,
     'use_glib': 1 if CONFIG['GLIB_LIBS'] else 0,
 
diff --git a/dom/canvas/test/captureStream_common.js b/dom/canvas/test/captureStream_common.js
index 3fca8df4b6..e4ff52b94b 100644
--- a/dom/canvas/test/captureStream_common.js
+++ b/dom/canvas/test/captureStream_common.js
@@ -146,8 +146,16 @@ CaptureStreamTestHelper.prototype = {
     info("Waiting for video " + video.id + " to match [" +
          refColor.data.join(',') + "] - " + refColor.name +
          " (" + infoString + ")");
+    var paintedFrames = video.mozPaintedFrames-1;
     return this.waitForPixel(video, 0, 0,
-                             px => this.isPixel(px, refColor, threshold))
+                             px => { if (paintedFrames != video.mozPaintedFrames) {
+				       info("Frame: " + video.mozPaintedFrames +
+					    " IsPixel ref=" + refColor.data +
+					    " threshold=" + threshold +
+					    " value=" + px);
+				       paintedFrames = video.mozPaintedFrames;
+				     }
+				     return this.isPixel(px, refColor, threshold); })
       .then(() => ok(true, video.id + " " + infoString));
   },
 
diff --git a/dom/media/systemservices/CamerasChild.cpp b/dom/media/systemservices/CamerasChild.cpp
index abbcc9e223..5b8129ca6d 100644
--- a/dom/media/systemservices/CamerasChild.cpp
+++ b/dom/media/systemservices/CamerasChild.cpp
@@ -5,7 +5,6 @@
 
 #include "CamerasChild.h"
 
-#include "webrtc/video_engine/include/vie_capture.h"
 #undef FF
 
 #include "mozilla/Assertions.h"
@@ -336,7 +335,7 @@ int
 CamerasChild::GetCaptureCapability(CaptureEngine aCapEngine,
                                    const char* unique_idUTF8,
                                    const unsigned int capability_number,
-                                   webrtc::CaptureCapability& capability)
+                                   webrtc::VideoCaptureCapability& capability)
 {
   LOG(("GetCaptureCapability: %s %d", unique_idUTF8, capability_number));
   RefPtr<CamerasChild> deathGrip = this;
@@ -352,7 +351,7 @@ CamerasChild::GetCaptureCapability(CaptureEngine aCapEngine,
 }
 
 bool
-CamerasChild::RecvReplyGetCaptureCapability(const CaptureCapability& ipcCapability)
+CamerasChild::RecvReplyGetCaptureCapability(const VideoCaptureCapability& ipcCapability)
 {
   LOG((__PRETTY_FUNCTION__));
   MonitorAutoLock monitor(mReplyMonitor);
@@ -414,7 +413,7 @@ int
 CamerasChild::AllocateCaptureDevice(CaptureEngine aCapEngine,
                                     const char* unique_idUTF8,
                                     const unsigned int unique_idUTF8Length,
-                                    int& capture_id,
+                                    int& aStreamId,
                                     const nsACString& aOrigin)
 {
   LOG((__PRETTY_FUNCTION__));
@@ -427,7 +426,7 @@ CamerasChild::AllocateCaptureDevice(CaptureEngine aCapEngine,
   LockAndDispatch<> dispatcher(this, __func__, runnable);
   if (dispatcher.Success()) {
     LOG(("Capture Device allocated: %d", mReplyInteger));
-    capture_id = mReplyInteger;
+    aStreamId = mReplyInteger;
   }
   return dispatcher.ReturnValue();
 }
@@ -460,7 +459,7 @@ CamerasChild::ReleaseCaptureDevice(CaptureEngine aCapEngine,
 
 void
 CamerasChild::AddCallback(const CaptureEngine aCapEngine, const int capture_id,
-                          webrtc::ExternalRenderer* render)
+                          FrameRelay* render)
 {
   MutexAutoLock lock(mCallbackMutex);
   CapturerElement ce;
@@ -486,12 +485,12 @@ CamerasChild::RemoveCallback(const CaptureEngine aCapEngine, const int capture_i
 int
 CamerasChild::StartCapture(CaptureEngine aCapEngine,
                            const int capture_id,
-                           webrtc::CaptureCapability& webrtcCaps,
-                           webrtc::ExternalRenderer* cb)
+                           webrtc::VideoCaptureCapability& webrtcCaps,
+                           FrameRelay* cb)
 {
   LOG((__PRETTY_FUNCTION__));
   AddCallback(aCapEngine, capture_id, cb);
-  CaptureCapability capCap(webrtcCaps.width,
+  VideoCaptureCapability capCap(webrtcCaps.width,
                            webrtcCaps.height,
                            webrtcCaps.maxFPS,
                            webrtcCaps.expectedCaptureDelay,
@@ -500,7 +499,7 @@ CamerasChild::StartCapture(CaptureEngine aCapEngine,
                            webrtcCaps.interlaced);
   RefPtr<CamerasChild> deathGrip = this;
   nsCOMPtr<nsIRunnable> runnable =
-    mozilla::NewNonOwningRunnableMethod<CaptureEngine, int, CaptureCapability>
+    mozilla::NewNonOwningRunnableMethod<CaptureEngine, int, VideoCaptureCapability>
     (this, &CamerasChild::SendStartCapture, aCapEngine, capture_id, capCap);
   LockAndDispatch<> dispatcher(this, __func__, runnable);
   return dispatcher.ReturnValue();
@@ -617,18 +616,12 @@ bool
 CamerasChild::RecvDeliverFrame(const CaptureEngine& capEngine,
                                const int& capId,
                                mozilla::ipc::Shmem&& shmem,
-                               const size_t& size,
-                               const uint32_t& time_stamp,
-                               const int64_t& ntp_time,
-                               const int64_t& render_time)
+                               const VideoFrameProperties & prop)
 {
   MutexAutoLock lock(mCallbackMutex);
   if (Callback(capEngine, capId)) {
     unsigned char* image = shmem.get<unsigned char>();
-    Callback(capEngine, capId)->DeliverFrame(image, size,
-                                             time_stamp,
-                                             ntp_time, render_time,
-                                             nullptr);
+    Callback(capEngine, capId)->DeliverFrame(image, prop);
   } else {
     LOG(("DeliverFrame called with dead callback"));
   }
@@ -673,7 +666,7 @@ CamerasChild::RecvFrameSizeChange(const CaptureEngine& capEngine,
   LOG((__PRETTY_FUNCTION__));
   MutexAutoLock lock(mCallbackMutex);
   if (Callback(capEngine, capId)) {
-    Callback(capEngine, capId)->FrameSizeChange(w, h, 0);
+    Callback(capEngine, capId)->FrameSizeChange(w, h);
   } else {
     LOG(("Frame size change with dead callback"));
   }
@@ -717,7 +710,7 @@ CamerasChild::~CamerasChild()
   MOZ_COUNT_DTOR(CamerasChild);
 }
 
-webrtc::ExternalRenderer* CamerasChild::Callback(CaptureEngine aCapEngine,
+FrameRelay* CamerasChild::Callback(CaptureEngine aCapEngine,
                                                  int capture_id)
 {
   for (unsigned int i = 0; i < mCallbacks.Length(); i++) {
diff --git a/dom/media/systemservices/CamerasChild.h b/dom/media/systemservices/CamerasChild.h
index 6d67089c09..dd239a4756 100644
--- a/dom/media/systemservices/CamerasChild.h
+++ b/dom/media/systemservices/CamerasChild.h
@@ -19,10 +19,10 @@
 // conflicts with #include of scoped_ptr.h
 #undef FF
 #include "webrtc/common.h"
-// Video Engine
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_render.h"
+#include "webrtc/video_renderer.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
+
+
 
 namespace mozilla {
 
@@ -32,10 +32,17 @@ class BackgroundChildImpl;
 
 namespace camera {
 
+class FrameRelay {
+public:
+  virtual int DeliverFrame(uint8_t* buffer,
+    const mozilla::camera::VideoFrameProperties& props) = 0;
+  virtual void FrameSizeChange(unsigned int w, unsigned int h) = 0;
+};
+
 struct CapturerElement {
   CaptureEngine engine;
   int id;
-  webrtc::ExternalRenderer* callback;
+  FrameRelay* callback;
 };
 
 // Forward declaration so we can work with pointers to it.
@@ -157,9 +164,9 @@ public:
 
   // IPC messages recevied, received on the PBackground thread
   // these are the actual callbacks with data
-  virtual bool RecvDeliverFrame(const CaptureEngine&, const int&, mozilla::ipc::Shmem&&,
-                                const size_t&, const uint32_t&, const int64_t&,
-                                const int64_t&) override;
+  virtual bool RecvDeliverFrame(const CaptureEngine&, const int&,
+                                mozilla::ipc::Shmem&&,
+                                const VideoFrameProperties & prop) override;
   virtual bool RecvFrameSizeChange(const CaptureEngine&, const int&,
                                    const int& w, const int& h) override;
 
@@ -171,7 +178,7 @@ public:
   virtual bool RecvReplyNumberOfCaptureDevices(const int&) override;
   virtual bool RecvReplyNumberOfCapabilities(const int&) override;
   virtual bool RecvReplyAllocateCaptureDevice(const int&) override;
-  virtual bool RecvReplyGetCaptureCapability(const CaptureCapability& capability) override;
+  virtual bool RecvReplyGetCaptureCapability(const VideoCaptureCapability& capability) override;
   virtual bool RecvReplyGetCaptureDevice(const nsCString& device_name,
                                          const nsCString& device_id,
                                          const bool& scary) override;
@@ -188,8 +195,8 @@ public:
   int ReleaseCaptureDevice(CaptureEngine aCapEngine,
                            const int capture_id);
   int StartCapture(CaptureEngine aCapEngine,
-                   const int capture_id, webrtc::CaptureCapability& capability,
-                   webrtc::ExternalRenderer* func);
+                   const int capture_id, webrtc::VideoCaptureCapability& capability,
+                   FrameRelay* func);
   int StopCapture(CaptureEngine aCapEngine, const int capture_id);
   int AllocateCaptureDevice(CaptureEngine aCapEngine,
                             const char* unique_idUTF8,
@@ -199,7 +206,7 @@ public:
   int GetCaptureCapability(CaptureEngine aCapEngine,
                            const char* unique_idUTF8,
                            const unsigned int capability_number,
-                           webrtc::CaptureCapability& capability);
+                           webrtc::VideoCaptureCapability& capability);
   int GetCaptureDevice(CaptureEngine aCapEngine,
                        unsigned int list_number, char* device_nameUTF8,
                        const unsigned int device_nameUTF8Length,
@@ -209,7 +216,7 @@ public:
   void ShutdownAll();
   int EnsureInitialized(CaptureEngine aCapEngine);
 
-  webrtc::ExternalRenderer* Callback(CaptureEngine aCapEngine, int capture_id);
+  FrameRelay* Callback(CaptureEngine aCapEngine, int capture_id);
 
 private:
   CamerasChild();
@@ -219,7 +226,7 @@ private:
   bool DispatchToParent(nsIRunnable* aRunnable,
                         MonitorAutoLock& aMonitor);
   void AddCallback(const CaptureEngine aCapEngine, const int capture_id,
-                   webrtc::ExternalRenderer* render);
+                   FrameRelay* render);
   void RemoveCallback(const CaptureEngine aCapEngine, const int capture_id);
   void ShutdownParent();
   void ShutdownChild();
@@ -246,7 +253,7 @@ private:
   // Async responses data contents;
   bool mReplySuccess;
   int mReplyInteger;
-  webrtc::CaptureCapability mReplyCapability;
+  webrtc::VideoCaptureCapability mReplyCapability;
   nsCString mReplyDeviceName;
   nsCString mReplyDeviceID;
   bool mReplyScary;
diff --git a/dom/media/systemservices/CamerasParent.cpp b/dom/media/systemservices/CamerasParent.cpp
index 92ebcf155a..013213d9e9 100644
--- a/dom/media/systemservices/CamerasParent.cpp
+++ b/dom/media/systemservices/CamerasParent.cpp
@@ -6,6 +6,7 @@
 #include "CamerasParent.h"
 #include "MediaEngine.h"
 #include "MediaUtils.h"
+#include "VideoFrameUtils.h"
 
 #include "mozilla/Assertions.h"
 #include "mozilla/Unused.h"
@@ -26,9 +27,11 @@
 #endif
 
 #undef LOG
+#undef LOG_VERBOSE
 #undef LOG_ENABLED
 mozilla::LazyLogModule gCamerasParentLog("CamerasParent");
 #define LOG(args) MOZ_LOG(gCamerasParentLog, mozilla::LogLevel::Debug, args)
+#define LOG_VERBOSE(args) MOZ_LOG(gCamerasParentLog, mozilla::LogLevel::Verbose, args)
 #define LOG_ENABLED() MOZ_LOG_TEST(gCamerasParentLog, mozilla::LogLevel::Debug)
 
 namespace mozilla {
@@ -44,7 +47,7 @@ namespace camera {
 //   suitable for UI access.
 
 // InputObserver is owned by CamerasParent, and it has a ref to CamerasParent
-void InputObserver::DeviceChange() {
+void InputObserver::OnDeviceChange() {
   LOG((__PRETTY_FUNCTION__));
   MOZ_ASSERT(mParent);
 
@@ -63,80 +66,30 @@ void InputObserver::DeviceChange() {
   thread->Dispatch(ipc_runnable, NS_DISPATCH_NORMAL);
 };
 
-class FrameSizeChangeRunnable : public Runnable {
+class DeliverFrameRunnable : public ::mozilla::Runnable {
 public:
-  FrameSizeChangeRunnable(CamerasParent *aParent, CaptureEngine capEngine,
-                          int cap_id, unsigned int aWidth, unsigned int aHeight)
-    : mParent(aParent), mCapEngine(capEngine), mCapId(cap_id),
-      mWidth(aWidth), mHeight(aHeight) {}
-
-  NS_IMETHOD Run() override {
-    if (mParent->IsShuttingDown()) {
-      // Communication channel is being torn down
-      LOG(("FrameSizeChangeRunnable is active without active Child"));
-      mResult = 0;
-      return NS_OK;
-    }
-    if (!mParent->SendFrameSizeChange(mCapEngine, mCapId, mWidth, mHeight)) {
-      mResult = -1;
-    } else {
-      mResult = 0;
-    }
-    return NS_OK;
-  }
-
-  int GetResult() {
-    return mResult;
-  }
-
-private:
-  RefPtr<CamerasParent> mParent;
-  CaptureEngine mCapEngine;
-  int mCapId;
-  unsigned int mWidth;
-  unsigned int mHeight;
-  int mResult;
-};
-
-int
-CallbackHelper::FrameSizeChange(unsigned int w, unsigned int h,
-                                unsigned int streams)
-{
-  LOG(("CallbackHelper Video FrameSizeChange: %ux%u", w, h));
-  RefPtr<FrameSizeChangeRunnable> runnable =
-    new FrameSizeChangeRunnable(mParent, mCapEngine, mCapturerId, w, h);
-  MOZ_ASSERT(mParent);
-  nsIThread * thread = mParent->GetBackgroundThread();
-  MOZ_ASSERT(thread != nullptr);
-  thread->Dispatch(runnable, NS_DISPATCH_NORMAL);
-  return 0;
-}
-
-class DeliverFrameRunnable : public Runnable {
-public:
-  DeliverFrameRunnable(CamerasParent *aParent,
-                       CaptureEngine engine,
-                       int cap_id,
-                       ShmemBuffer buffer,
-                       unsigned char* altbuffer,
-                       size_t size,
-                       uint32_t time_stamp,
-                       int64_t ntp_time,
-                       int64_t render_time)
-    : mParent(aParent), mCapEngine(engine), mCapId(cap_id), mBuffer(Move(buffer)),
-      mSize(size), mTimeStamp(time_stamp), mNtpTime(ntp_time),
-      mRenderTime(render_time) {
+  DeliverFrameRunnable(CamerasParent *aParent, CaptureEngine aEngine,
+      uint32_t aStreamId, const webrtc::VideoFrame& aFrame,
+      const VideoFrameProperties& aProperties)
+      : mParent(aParent), mCapEngine(aEngine), mStreamId(aStreamId),
+      mProperties(aProperties)
+  {
     // No ShmemBuffer (of the right size) was available, so make an
     // extra buffer here.  We have no idea when we are going to run and
     // it will be potentially long after the webrtc frame callback has
     // returned, so the copy needs to be no later than here.
     // We will need to copy this back into a Shmem later on so we prefer
     // using ShmemBuffers to avoid the extra copy.
-    if (altbuffer != nullptr) {
-      mAlternateBuffer.reset(new unsigned char[size]);
-      memcpy(mAlternateBuffer.get(), altbuffer, size);
-    }
-  };
+    mAlternateBuffer.reset(new unsigned char[aProperties.bufferSize()]);
+    VideoFrameUtils::CopyVideoFrameBuffers(mAlternateBuffer.get(),
+                                           aProperties.bufferSize(), aFrame);
+  }
+
+  DeliverFrameRunnable(CamerasParent* aParent, CaptureEngine aEngine,
+      uint32_t aStreamId, ShmemBuffer aBuffer, VideoFrameProperties& aProperties)
+      : mParent(aParent), mCapEngine(aEngine), mStreamId(aStreamId),
+      mBuffer(Move(aBuffer)), mProperties(aProperties)
+  {};
 
   NS_IMETHOD Run() override {
     if (mParent->IsShuttingDown()) {
@@ -144,10 +97,8 @@ public:
       mResult = 0;
       return NS_OK;
     }
-    if (!mParent->DeliverFrameOverIPC(mCapEngine, mCapId,
-                                      Move(mBuffer), mAlternateBuffer.get(),
-                                      mSize, mTimeStamp,
-                                      mNtpTime, mRenderTime)) {
+    if (!mParent->DeliverFrameOverIPC(mCapEngine, mStreamId, Move(mBuffer),
+                                      mAlternateBuffer.get(), mProperties)) {
       mResult = -1;
     } else {
       mResult = 0;
@@ -162,13 +113,10 @@ public:
 private:
   RefPtr<CamerasParent> mParent;
   CaptureEngine mCapEngine;
-  int mCapId;
+  uint32_t mStreamId;
   ShmemBuffer mBuffer;
   mozilla::UniquePtr<unsigned char[]> mAlternateBuffer;
-  size_t mSize;
-  uint32_t mTimeStamp;
-  int64_t mNtpTime;
-  int64_t mRenderTime;
+  VideoFrameProperties mProperties;
   int mResult;
 };
 
@@ -258,14 +206,11 @@ CamerasParent::StopVideoCapture()
 }
 
 int
-CamerasParent::DeliverFrameOverIPC(CaptureEngine cap_engine,
-                                   int cap_id,
-                                   ShmemBuffer buffer,
-                                   unsigned char* altbuffer,
-                                   size_t size,
-                                   uint32_t time_stamp,
-                                   int64_t ntp_time,
-                                   int64_t render_time)
+CamerasParent::DeliverFrameOverIPC(CaptureEngine capEng,
+                          uint32_t aStreamId,
+                          ShmemBuffer buffer,
+                          unsigned char* altbuffer,
+                          VideoFrameProperties& aProps)
 {
   // No ShmemBuffers were available, so construct one now of the right size
   // and copy into it. That is an extra copy, but we expect this to be
@@ -273,7 +218,7 @@ CamerasParent::DeliverFrameOverIPC(CaptureEngine cap_engine,
   // buffer of the right size.
   if (altbuffer != nullptr) {
     // Get a shared memory buffer from the pool, at least size big
-    ShmemBuffer shMemBuff = mShmemPool.Get(this, size);
+    ShmemBuffer shMemBuff = mShmemPool.Get(this, aProps.bufferSize());
 
     if (!shMemBuff.Valid()) {
       LOG(("No usable Video shmem in DeliverFrame (out of buffers?)"));
@@ -282,20 +227,18 @@ CamerasParent::DeliverFrameOverIPC(CaptureEngine cap_engine,
     }
 
     // get() and Size() check for proper alignment of the segment
-    memcpy(shMemBuff.GetBytes(), altbuffer, size);
+    memcpy(shMemBuff.GetBytes(), altbuffer, aProps.bufferSize());
 
-    if (!SendDeliverFrame(cap_engine, cap_id,
-                          shMemBuff.Get(), size,
-                          time_stamp, ntp_time, render_time)) {
+    if (!SendDeliverFrame(capEng, aStreamId,
+                          shMemBuff.Get(), aProps)) {
       return -1;
     }
   } else {
     MOZ_ASSERT(buffer.Valid());
     // ShmemBuffer was available, we're all good. A single copy happened
     // in the original webrtc callback.
-    if (!SendDeliverFrame(cap_engine, cap_id,
-                          buffer.Get(), size,
-                          time_stamp, ntp_time, render_time)) {
+    if (!SendDeliverFrame(capEng, aStreamId,
+                          buffer.Get(), aProps)) {
       return -1;
     }
   }
@@ -309,16 +252,16 @@ CamerasParent::GetBuffer(size_t aSize)
   return mShmemPool.GetIfAvailable(aSize);
 }
 
-int
-CallbackHelper::DeliverFrame(unsigned char* buffer,
-                             size_t size,
-                             uint32_t time_stamp,
-                             int64_t ntp_time,
-                             int64_t render_time,
-                             void *handle)
+int32_t
+CallbackHelper::RenderFrame(uint32_t aStreamId, const webrtc::VideoFrame& aVideoFrame)
 {
+  LOG_VERBOSE((__PRETTY_FUNCTION__));
+  RefPtr<DeliverFrameRunnable> runnable = nullptr;
+  // Get frame properties
+  camera::VideoFrameProperties properties;
+  VideoFrameUtils::InitFrameBufferProperties(aVideoFrame, properties);
   // Get a shared memory buffer to copy the frame data into
-  ShmemBuffer shMemBuffer = mParent->GetBuffer(size);
+  ShmemBuffer shMemBuffer = mParent->GetBuffer(properties.bufferSize());
   if (!shMemBuffer.Valid()) {
     // Either we ran out of buffers or they're not the right size yet
     LOG(("Correctly sized Video shmem not available in DeliverFrame"));
@@ -326,30 +269,33 @@ CallbackHelper::DeliverFrame(unsigned char* buffer,
     // the DeliverFrameRunnable constructor.
   } else {
     // Shared memory buffers of the right size are available, do the copy here.
-    memcpy(shMemBuffer.GetBytes(), buffer, size);
-    // Mark the original buffer as cleared.
-    buffer = nullptr;
+    VideoFrameUtils::CopyVideoFrameBuffers(shMemBuffer.GetBytes(),
+                                           properties.bufferSize(), aVideoFrame);
+    runnable = new DeliverFrameRunnable(mParent, mCapEngine, mStreamId,
+                                        Move(shMemBuffer), properties);
+  }
+  if (!runnable.get()) {
+    runnable = new DeliverFrameRunnable(mParent, mCapEngine, mStreamId,
+                                        aVideoFrame, properties);
   }
-  RefPtr<DeliverFrameRunnable> runnable =
-    new DeliverFrameRunnable(mParent, mCapEngine, mCapturerId,
-                             Move(shMemBuffer), buffer, size, time_stamp,
-                             ntp_time, render_time);
   MOZ_ASSERT(mParent);
   nsIThread* thread = mParent->GetBackgroundThread();
   MOZ_ASSERT(thread != nullptr);
   thread->Dispatch(runnable, NS_DISPATCH_NORMAL);
   return 0;
 }
-// XXX!!! FIX THIS -- we should move to pure DeliverI420Frame
-int
-CallbackHelper::DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame)
+
+void
+CallbackHelper::OnIncomingCapturedFrame(const int32_t id, const webrtc::VideoFrame& aVideoFrame)
 {
-  return DeliverFrame(const_cast<uint8_t*>(webrtc_frame.buffer(webrtc::kYPlane)),
-                      CalcBufferSize(webrtc::kI420, webrtc_frame.width(), webrtc_frame.height()),
-                      webrtc_frame.timestamp(),
-                      webrtc_frame.ntp_time_ms(),
-                      webrtc_frame.render_time_ms(),
-                      (void*) webrtc_frame.native_handle());
+ LOG_VERBOSE((__PRETTY_FUNCTION__));
+ RenderFrame(id,aVideoFrame);
+}
+
+void
+CallbackHelper::OnCaptureDelayChanged(const int32_t id, const int32_t delay)
+{
+  LOG((__PRETTY_FUNCTION__));
 }
 
 bool
@@ -361,15 +307,17 @@ CamerasParent::RecvReleaseFrame(mozilla::ipc::Shmem&& s) {
 bool
 CamerasParent::SetupEngine(CaptureEngine aCapEngine)
 {
+  LOG((__PRETTY_FUNCTION__));
   MOZ_ASSERT(mVideoCaptureThread->thread_id() == PlatformThread::CurrentId());
-  EngineHelper *helper = &mEngines[aCapEngine];
+  RefPtr<mozilla::camera::VideoEngine>* engine = &mEngines[aCapEngine];
 
   // Already initialized
-  if (helper->mEngine) {
+  if (engine->get()) {
     return true;
   }
 
   webrtc::CaptureDeviceInfo *captureDeviceInfo = nullptr;
+  UniquePtr<webrtc::Config> config(new webrtc::Config);
 
   switch (aCapEngine) {
   case ScreenEngine:
@@ -398,43 +346,19 @@ CamerasParent::SetupEngine(CaptureEngine aCapEngine)
     break;
   }
 
-  helper->mConfig.Set<webrtc::CaptureDeviceInfo>(captureDeviceInfo);
-  helper->mEngine = webrtc::VideoEngine::Create(helper->mConfig);
+  config->Set<webrtc::CaptureDeviceInfo>(captureDeviceInfo);
+  *engine = mozilla::camera::VideoEngine::Create(UniquePtr<const webrtc::Config>(config.release()));
 
-  if (!helper->mEngine) {
+  if (!engine->get()) {
     LOG(("VideoEngine::Create failed"));
     return false;
   }
 
-  helper->mPtrViEBase = webrtc::ViEBase::GetInterface(helper->mEngine);
-  if (!helper->mPtrViEBase) {
-    LOG(("ViEBase::GetInterface failed"));
-    return false;
-  }
-
-  if (helper->mPtrViEBase->Init() < 0) {
-    LOG(("ViEBase::Init failed"));
-    return false;
-  }
-
-  helper->mPtrViECapture = webrtc::ViECapture::GetInterface(helper->mEngine);
-  if (!helper->mPtrViECapture) {
-    LOG(("ViECapture::GetInterface failed"));
-    return false;
-  }
-
   RefPtr<InputObserver>* observer = mObservers.AppendElement(new InputObserver(this));
-
-#ifdef DEBUG
-  MOZ_ASSERT(0 == helper->mPtrViECapture->RegisterInputObserver(observer->get()));
-#else
-  helper->mPtrViECapture->RegisterInputObserver(observer->get());
-#endif
-
-  helper->mPtrViERender = webrtc::ViERender::GetInterface(helper->mEngine);
-  if (!helper->mPtrViERender) {
-    LOG(("ViERender::GetInterface failed"));
-    return false;
+  auto device_info = engine->get()->GetOrCreateVideoCaptureDeviceInfo();
+  MOZ_ASSERT(device_info);
+  if (device_info) {
+    device_info->RegisterVideoInputFeedBack(**observer);
   }
 
   return true;
@@ -452,38 +376,25 @@ CamerasParent::CloseEngines()
   // Stop the callers
   while (mCallbacks.Length()) {
     auto capEngine = mCallbacks[0]->mCapEngine;
-    auto capNum = mCallbacks[0]->mCapturerId;
-    LOG(("Forcing shutdown of engine %d, capturer %d", capEngine, capNum));
-    StopCapture(capEngine, capNum);
-    Unused << ReleaseCaptureDevice(capEngine, capNum);
+    auto streamNum = mCallbacks[0]->mStreamId;
+    LOG(("Forcing shutdown of engine %d, capturer %d", capEngine, streamNum));
+    StopCapture(capEngine, streamNum);
+    Unused << ReleaseCaptureDevice(capEngine, streamNum);
   }
 
   for (int i = 0; i < CaptureEngine::MaxEngine; i++) {
-    if (mEngines[i].mEngineIsRunning) {
-      LOG(("Being closed down while engine %d is running!", i));
-    }
-    if (mEngines[i].mPtrViERender) {
-      mEngines[i].mPtrViERender->Release();
-      mEngines[i].mPtrViERender = nullptr;
-    }
-    if (mEngines[i].mPtrViECapture) {
-#ifdef DEBUG
-      MOZ_ASSERT(0 == mEngines[i].mPtrViECapture->DeregisterInputObserver());
-#else
-      mEngines[i].mPtrViECapture->DeregisterInputObserver();
-#endif
+    if (auto engine = mEngines[i].get() ){
+      if (engine->IsRunning()) {
+        LOG(("Being closed down while engine %d is running!", i));
+      }
 
-      mEngines[i].mPtrViECapture->Release();
-        mEngines[i].mPtrViECapture = nullptr;
-    }
-    if(mEngines[i].mPtrViEBase) {
-      mEngines[i].mPtrViEBase->Release();
-      mEngines[i].mPtrViEBase = nullptr;
-    }
-    if (mEngines[i].mEngine) {
-      mEngines[i].mEngine->SetTraceCallback(nullptr);
-      webrtc::VideoEngine::Delete(mEngines[i].mEngine);
-      mEngines[i].mEngine = nullptr;
+      auto device_info = engine->GetOrCreateVideoCaptureDeviceInfo();
+      MOZ_ASSERT(device_info);
+      if (device_info) {
+        device_info->DeRegisterVideoInputFeedBack();
+      }
+      mozilla::camera::VideoEngine::Delete(engine);
+      mEngines[i] = nullptr;
     }
   }
 
@@ -492,21 +403,21 @@ CamerasParent::CloseEngines()
   mWebRTCAlive = false;
 }
 
-bool
+VideoEngine *
 CamerasParent::EnsureInitialized(int aEngine)
 {
-  LOG((__PRETTY_FUNCTION__));
+  LOG_VERBOSE((__PRETTY_FUNCTION__));
   // We're shutting down, don't try to do new WebRTC ops.
   if (!mWebRTCAlive) {
-    return false;
+    return nullptr;
   }
   CaptureEngine capEngine = static_cast<CaptureEngine>(aEngine);
   if (!SetupEngine(capEngine)) {
     LOG(("CamerasParent failed to initialize engine"));
-    return false;
+    return nullptr;
   }
 
-  return true;
+  return mEngines[aEngine];
 }
 
 // Dispatch the runnable to do the camera operation on the
@@ -518,13 +429,15 @@ bool
 CamerasParent::RecvNumberOfCaptureDevices(const CaptureEngine& aCapEngine)
 {
   LOG((__PRETTY_FUNCTION__));
-
+  LOG(("CaptureEngine=%d", aCapEngine));
   RefPtr<CamerasParent> self(this);
   RefPtr<Runnable> webrtc_runnable =
     media::NewRunnableFrom([self, aCapEngine]() -> nsresult {
       int num = -1;
-      if (self->EnsureInitialized(aCapEngine)) {
-        num = self->mEngines[aCapEngine].mPtrViECapture->NumberOfCaptureDevices();
+      if (auto engine = self->EnsureInitialized(aCapEngine)) {
+        if (auto devInfo = engine->GetOrCreateVideoCaptureDeviceInfo()) {
+          num = devInfo->NumberOfDevices();
+        }
       }
       RefPtr<nsIRunnable> ipc_runnable =
         media::NewRunnableFrom([self, num]() -> nsresult {
@@ -591,11 +504,10 @@ CamerasParent::RecvNumberOfCapabilities(const CaptureEngine& aCapEngine,
   RefPtr<Runnable> webrtc_runnable =
     media::NewRunnableFrom([self, unique_id, aCapEngine]() -> nsresult {
       int num = -1;
-      if (self->EnsureInitialized(aCapEngine)) {
-        num =
-          self->mEngines[aCapEngine].mPtrViECapture->NumberOfCapabilities(
-            unique_id.get(),
-            MediaEngineSource::kMaxUniqueIdLength);
+      if (auto engine = self->EnsureInitialized(aCapEngine)) {
+        if (auto devInfo = engine->GetOrCreateVideoCaptureDeviceInfo()) {
+          num = devInfo->NumberOfCapabilities(unique_id.get());
+        }
       }
       RefPtr<nsIRunnable> ipc_runnable =
         media::NewRunnableFrom([self, num]() -> nsresult {
@@ -630,18 +542,19 @@ CamerasParent::RecvGetCaptureCapability(const CaptureEngine& aCapEngine,
   RefPtr<CamerasParent> self(this);
   RefPtr<Runnable> webrtc_runnable =
     media::NewRunnableFrom([self, unique_id, aCapEngine, num]() -> nsresult {
-      webrtc::CaptureCapability webrtcCaps;
+      webrtc::VideoCaptureCapability webrtcCaps;
       int error = -1;
-      if (self->EnsureInitialized(aCapEngine)) {
-        error = self->mEngines[aCapEngine].mPtrViECapture->GetCaptureCapability(
-          unique_id.get(), MediaEngineSource::kMaxUniqueIdLength, num, webrtcCaps);
+      if (auto engine = self->EnsureInitialized(aCapEngine)) {
+        if (auto devInfo = engine->GetOrCreateVideoCaptureDeviceInfo()){
+          error = devInfo->GetCapability(unique_id.get(), num, webrtcCaps);
+        }
       }
       RefPtr<nsIRunnable> ipc_runnable =
         media::NewRunnableFrom([self, webrtcCaps, error]() -> nsresult {
           if (self->IsShuttingDown()) {
             return NS_ERROR_FAILURE;
           }
-          CaptureCapability capCap(webrtcCaps.width,
+          VideoCaptureCapability capCap(webrtcCaps.width,
                                    webrtcCaps.height,
                                    webrtcCaps.maxFPS,
                                    webrtcCaps.expectedCaptureDelay,
@@ -682,15 +595,15 @@ CamerasParent::RecvGetCaptureDevice(const CaptureEngine& aCapEngine,
       char deviceUniqueId[MediaEngineSource::kMaxUniqueIdLength];
       nsCString name;
       nsCString uniqueId;
-      int devicePid = 0;
+      pid_t devicePid = 0;
       int error = -1;
-      if (self->EnsureInitialized(aCapEngine)) {
-          error = self->mEngines[aCapEngine].mPtrViECapture->GetCaptureDevice(aListNumber,
-                                                                              deviceName,
-                                                                              sizeof(deviceName),
-                                                                              deviceUniqueId,
-                                                                              sizeof(deviceUniqueId),
-                                                                              &devicePid);
+      if (auto engine = self->EnsureInitialized(aCapEngine)) {
+        if (auto devInfo = engine->GetOrCreateVideoCaptureDeviceInfo()) {
+          error = devInfo->GetDeviceName(aListNumber, deviceName, sizeof(deviceName),
+                                         deviceUniqueId, sizeof(deviceUniqueId),
+                                         nullptr, 0,
+                                         &devicePid);
+        }
       }
       if (!error) {
         name.Assign(deviceName);
@@ -806,8 +719,17 @@ CamerasParent::RecvAllocateCaptureDevice(const CaptureEngine& aCapEngine,
         int numdev = -1;
         int error = -1;
         if (allowed && self->EnsureInitialized(aCapEngine)) {
-          error = self->mEngines[aCapEngine].mPtrViECapture->AllocateCaptureDevice(
-                    unique_id.get(), MediaEngineSource::kMaxUniqueIdLength, numdev);
+          auto engine = self->mEngines[aCapEngine].get();
+          engine->CreateVideoCapture(numdev, unique_id.get());
+          engine->WithEntry(numdev, [&error](VideoEngine::CaptureEntry& cap) {
+            if (cap.VideoCapture()) {
+              if (!cap.VideoRenderer()) {
+                LOG(("VideoEngine::VideoRenderer() failed"));
+              } else {
+                error = 0;
+              }
+            }
+          });
         }
         RefPtr<nsIRunnable> ipc_runnable =
           media::NewRunnableFrom([self, numdev, error]() -> nsresult {
@@ -838,8 +760,8 @@ CamerasParent::ReleaseCaptureDevice(const CaptureEngine& aCapEngine,
                                     const int& capnum)
 {
   int error = -1;
-  if (EnsureInitialized(aCapEngine)) {
-    error = mEngines[aCapEngine].mPtrViECapture->ReleaseCaptureDevice(capnum);
+  if (auto engine = EnsureInitialized(aCapEngine)) {
+    error = engine->ReleaseVideoCapture(capnum);
   }
   return error;
 }
@@ -881,44 +803,48 @@ CamerasParent::RecvReleaseCaptureDevice(const CaptureEngine& aCapEngine,
 bool
 CamerasParent::RecvStartCapture(const CaptureEngine& aCapEngine,
                                 const int& capnum,
-                                const CaptureCapability& ipcCaps)
+                                const VideoCaptureCapability& ipcCaps)
 {
   LOG((__PRETTY_FUNCTION__));
 
   RefPtr<CamerasParent> self(this);
   RefPtr<Runnable> webrtc_runnable =
     media::NewRunnableFrom([self, aCapEngine, capnum, ipcCaps]() -> nsresult {
+      LOG((__PRETTY_FUNCTION__));
       CallbackHelper** cbh;
-      webrtc::ExternalRenderer* render;
-      EngineHelper* helper = nullptr;
+      webrtc::VideoRenderCallback* render;
+      VideoEngine* engine = nullptr;
       int error = -1;
       if (self->EnsureInitialized(aCapEngine)) {
         cbh = self->mCallbacks.AppendElement(
           new CallbackHelper(static_cast<CaptureEngine>(aCapEngine), capnum, self));
-        render = static_cast<webrtc::ExternalRenderer*>(*cbh);
+        render = static_cast<webrtc::VideoRenderCallback*>(*cbh);
 
-        helper = &self->mEngines[aCapEngine];
-        error =
-          helper->mPtrViERender->AddRenderer(capnum, webrtc::kVideoI420, render);
-        if (!error) {
-          error = helper->mPtrViERender->StartRender(capnum);
-        }
+        engine = self->mEngines[aCapEngine];
+        engine->WithEntry(capnum, [capnum, &render, &engine, &error, &ipcCaps, &cbh](VideoEngine::CaptureEntry& cap) {
+          cap.VideoRenderer()->AddIncomingRenderStream(capnum,0, 0., 0., 1., 1.);
+          error = cap.VideoRenderer()->AddExternalRenderCallback(capnum, render);
+          if (!error) {
+            error = cap.VideoRenderer()->StartRender(capnum);
+          }
 
-        webrtc::CaptureCapability capability;
-        capability.width = ipcCaps.width();
-        capability.height = ipcCaps.height();
-        capability.maxFPS = ipcCaps.maxFPS();
-        capability.expectedCaptureDelay = ipcCaps.expectedCaptureDelay();
-        capability.rawType = static_cast<webrtc::RawVideoType>(ipcCaps.rawType());
-        capability.codecType = static_cast<webrtc::VideoCodecType>(ipcCaps.codecType());
-        capability.interlaced = ipcCaps.interlaced();
+          webrtc::VideoCaptureCapability capability;
+          capability.width = ipcCaps.width();
+          capability.height = ipcCaps.height();
+          capability.maxFPS = ipcCaps.maxFPS();
+          capability.expectedCaptureDelay = ipcCaps.expectedCaptureDelay();
+          capability.rawType = static_cast<webrtc::RawVideoType>(ipcCaps.rawType());
+          capability.codecType = static_cast<webrtc::VideoCodecType>(ipcCaps.codecType());
+          capability.interlaced = ipcCaps.interlaced();
 
-        if (!error) {
-          error = helper->mPtrViECapture->StartCapture(capnum, capability);
-        }
-        if (!error) {
-          helper->mEngineIsRunning = true;
-        }
+          if (!error) {
+            error = cap.VideoCapture()->StartCapture(capability);
+          }
+          if (!error) {
+            engine->Startup();
+            cap.VideoCapture()->RegisterCaptureDataCallback(*static_cast<webrtc::VideoCaptureDataCallback*>(*cbh));
+          }
+        });
       }
       RefPtr<nsIRunnable> ipc_runnable =
         media::NewRunnableFrom([self, error]() -> nsresult {
@@ -944,20 +870,27 @@ void
 CamerasParent::StopCapture(const CaptureEngine& aCapEngine,
                            const int& capnum)
 {
-  if (EnsureInitialized(aCapEngine)) {
-    mEngines[aCapEngine].mPtrViECapture->StopCapture(capnum);
-    mEngines[aCapEngine].mPtrViERender->StopRender(capnum);
-    mEngines[aCapEngine].mPtrViERender->RemoveRenderer(capnum);
-    mEngines[aCapEngine].mEngineIsRunning = false;
-
-    for (size_t i = 0; i < mCallbacks.Length(); i++) {
-      if (mCallbacks[i]->mCapEngine == aCapEngine
-          && mCallbacks[i]->mCapturerId == capnum) {
-        delete mCallbacks[i];
-        mCallbacks.RemoveElementAt(i);
+  if (auto engine = EnsureInitialized(aCapEngine)) {
+    engine->WithEntry(capnum,[capnum](VideoEngine::CaptureEntry& cap){
+      if (cap.VideoCapture()) {
+        cap.VideoCapture()->StopCapture();
+        cap.VideoCapture()->DeRegisterCaptureDataCallback();
+      }
+      if (cap.VideoRenderer()) {
+        cap.VideoRenderer()->StopRender(capnum);
+      }
+    });
+    // we're removing elements, iterate backwards
+    for (size_t i = mCallbacks.Length(); i > 0; i--) {
+      if (mCallbacks[i-1]->mCapEngine == aCapEngine
+          && mCallbacks[i-1]->mStreamId == (uint32_t) capnum) {
+        delete mCallbacks[i-1];
+        mCallbacks.RemoveElementAt(i-1);
         break;
       }
     }
+    engine->RemoveRenderer(capnum);
+    engine->Shutdown();
   }
 }
 
@@ -1077,7 +1010,7 @@ CamerasParent::~CamerasParent()
   // That runnable takes a ref to us, so it must have finished
   // by the time we get here.
   for (int i = 0; i < CaptureEngine::MaxEngine; i++) {
-    MOZ_ASSERT(!mEngines[i].mEngine);
+    MOZ_ASSERT(!mEngines[i]);
   }
 #endif
 }
diff --git a/dom/media/systemservices/CamerasParent.h b/dom/media/systemservices/CamerasParent.h
index 958d57237d..0316af5649 100644
--- a/dom/media/systemservices/CamerasParent.h
+++ b/dom/media/systemservices/CamerasParent.h
@@ -7,19 +7,21 @@
 #define mozilla_CamerasParent_h
 
 #include "nsIObserver.h"
+#include "VideoEngine.h"
 #include "mozilla/dom/ContentParent.h"
 #include "mozilla/camera/PCamerasParent.h"
 #include "mozilla/ipc/Shmem.h"
 #include "mozilla/ShmemPool.h"
 #include "mozilla/Atomics.h"
+#include "webrtc/modules/video_capture/video_capture.h"
+#include "webrtc/modules/video_render/video_render_impl.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
 
 // conflicts with #include of scoped_ptr.h
 #undef FF
 #include "webrtc/common.h"
-// Video Engine
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_render.h"
+
 #include "CamerasChild.h"
 
 #include "base/thread.h"
@@ -29,60 +31,43 @@ namespace camera {
 
 class CamerasParent;
 
-class CallbackHelper : public webrtc::ExternalRenderer
+class CallbackHelper :
+  public webrtc::VideoRenderCallback,
+  public webrtc::VideoCaptureDataCallback
 {
 public:
-  CallbackHelper(CaptureEngine aCapEng, int aCapId, CamerasParent *aParent)
-    : mCapEngine(aCapEng), mCapturerId(aCapId), mParent(aParent) {};
+  CallbackHelper(CaptureEngine aCapEng, uint32_t aStreamId, CamerasParent *aParent)
+    : mCapEngine(aCapEng), mStreamId(aStreamId), mParent(aParent) {};
 
   // ViEExternalRenderer implementation. These callbacks end up
   // running on the VideoCapture thread.
-  virtual int FrameSizeChange(unsigned int w, unsigned int h,
-                              unsigned int streams) override;
-  virtual int DeliverFrame(unsigned char* buffer,
-                           size_t size,
-                           uint32_t time_stamp,
-                           int64_t ntp_time,
-                           int64_t render_time,
-                           void *handle) override;
-  virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;
-  virtual bool IsTextureSupported() override { return false; };
+  virtual int32_t RenderFrame(const uint32_t aStreamId, const webrtc::VideoFrame& video_frame) override;
+
+  // From  VideoCaptureCallback
+  virtual void OnIncomingCapturedFrame(const int32_t id, const webrtc::VideoFrame& videoFrame) override;
+  virtual void OnCaptureDelayChanged(const int32_t id, const int32_t delay) override;
+
+	// TODO(@@NG) This is now part of webrtc::VideoRenderer, not in the webrtc::VideoRenderCallback
+	// virtual bool IsTextureSupported() const override { return false; };
+	//
+	// virtual bool SmoothsRenderedFrames() const override { return false; }
 
   friend CamerasParent;
 
 private:
   CaptureEngine mCapEngine;
-  int mCapturerId;
+  uint32_t mStreamId;
   CamerasParent *mParent;
 };
 
-class EngineHelper
-{
-public:
-  EngineHelper() :
-    mEngine(nullptr), mPtrViEBase(nullptr), mPtrViECapture(nullptr),
-    mPtrViERender(nullptr), mEngineIsRunning(false) {};
-
-  webrtc::VideoEngine *mEngine;
-  webrtc::ViEBase *mPtrViEBase;
-  webrtc::ViECapture *mPtrViECapture;
-  webrtc::ViERender *mPtrViERender;
-
-  // The webrtc code keeps a reference to this one.
-  webrtc::Config mConfig;
-
-  // Engine alive
-  bool mEngineIsRunning;
-};
-
-class InputObserver :  public webrtc::ViEInputObserver
+class InputObserver :  public webrtc::VideoInputFeedBack
 {
 public:
   NS_INLINE_DECL_THREADSAFE_REFCOUNTING(InputObserver)
 
   explicit InputObserver(CamerasParent* aParent)
     : mParent(aParent) {};
-  virtual void DeviceChange();
+  virtual void OnDeviceChange();
 
   friend CamerasParent;
 
@@ -113,7 +98,7 @@ public:
                                         const int&) override;
   virtual bool RecvGetCaptureDevice(const CaptureEngine&, const int&) override;
   virtual bool RecvStartCapture(const CaptureEngine&, const int&,
-                                const CaptureCapability&) override;
+                                                   const VideoCaptureCapability&) override;
   virtual bool RecvStopCapture(const CaptureEngine&, const int&) override;
   virtual bool RecvReleaseFrame(mozilla::ipc::Shmem&&) override;
   virtual bool RecvAllDone() override;
@@ -128,13 +113,10 @@ public:
 
   // helper to forward to the PBackground thread
   int DeliverFrameOverIPC(CaptureEngine capEng,
-                          int cap_id,
+                          uint32_t aStreamId,
                           ShmemBuffer buffer,
                           unsigned char* altbuffer,
-                          size_t size,
-                          uint32_t time_stamp,
-                          int64_t ntp_time,
-                          int64_t render_time);
+                          VideoFrameProperties& aProps);
 
 
   CamerasParent();
@@ -147,14 +129,14 @@ protected:
   int ReleaseCaptureDevice(const CaptureEngine& aCapEngine, const int& capnum);
 
   bool SetupEngine(CaptureEngine aCapEngine);
-  bool EnsureInitialized(int aEngine);
+  VideoEngine* EnsureInitialized(int aEngine);
   void CloseEngines();
   void StopIPC();
   void StopVideoCapture();
   // Can't take already_AddRefed because it can fail in stupid ways.
   nsresult DispatchToVideoCaptureThread(Runnable* event);
 
-  EngineHelper mEngines[CaptureEngine::MaxEngine];
+  RefPtr<VideoEngine> mEngines[CaptureEngine::MaxEngine];
   nsTArray<CallbackHelper*> mCallbacks;
 
   // image buffers
diff --git a/dom/media/systemservices/LoadManager.h b/dom/media/systemservices/LoadManager.h
index 96824a308f..78efca02f7 100644
--- a/dom/media/systemservices/LoadManager.h
+++ b/dom/media/systemservices/LoadManager.h
@@ -14,8 +14,8 @@
 #include "nsIObserver.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/video_engine/include/vie_base.h"
-
+#include "webrtc/call.h"
+#include "webrtc/video/overuse_frame_detector.h"
 extern mozilla::LazyLogModule gLoadManagerLog;
 
 namespace mozilla {
@@ -77,7 +77,7 @@ private:
 };
 
 class LoadManager final : public webrtc::CPULoadStateCallbackInvoker,
-                          public webrtc::CpuOveruseObserver
+                          public webrtc::LoadObserver
 {
 public:
     explicit LoadManager(LoadManagerSingleton* aManager)
@@ -93,13 +93,14 @@ public:
     {
         mManager->RemoveObserver(aObserver);
     }
-    void OveruseDetected() override
+
+    void OnLoadUpdate(webrtc::LoadObserver::Load load_state) override
     {
-        mManager->OveruseDetected();
-    }
-    void NormalUsage() override
-    {
-        mManager->NormalUsage();
+        if (load_state == webrtc::LoadObserver::kOveruse) {
+            mManager->OveruseDetected();
+        } else if (load_state == webrtc::LoadObserver::kUnderuse) {
+            mManager->NormalUsage();
+        }
     }
 
 private:
diff --git a/dom/media/systemservices/PCameras.ipdl b/dom/media/systemservices/PCameras.ipdl
index b9fa583295..72c1cfc15a 100644
--- a/dom/media/systemservices/PCameras.ipdl
+++ b/dom/media/systemservices/PCameras.ipdl
@@ -10,7 +10,8 @@ using mozilla::camera::CaptureEngine from "mozilla/media/CamerasTypes.h";
 namespace mozilla {
 namespace camera {
 
-struct CaptureCapability
+// IPC analog for webrtc::VideoCaptureCapability
+struct VideoCaptureCapability
 {
   int width;
   int height;
@@ -21,6 +22,32 @@ struct CaptureCapability
   bool interlaced;
 };
 
+
+// IPC analog for webrtc::VideoFrame
+// the described buffer is transported seperately in a Shmem
+// See VideoFrameUtils.h
+struct VideoFrameProperties
+{
+  // Size of image data within the ShMem,
+  // the ShMem is at least this large
+  uint32_t bufferSize;
+  // From webrtc::VideoFrame
+  uint32_t timeStamp;
+  int64_t ntpTimeMs;
+  int64_t renderTimeMs;
+  // See webrtc/**/rotation.h
+  int rotation;
+  int yAllocatedSize;
+  int uAllocatedSize;
+  int vAllocatedSize;
+  // From webrtc::VideoFrameBuffer
+  int width;
+  int height;
+  int yStride;
+  int uStride;
+  int vStride;
+};
+
 async protocol PCameras
 {
   manager PBackground;
@@ -28,14 +55,13 @@ async protocol PCameras
 child:
   async FrameSizeChange(CaptureEngine capEngine, int cap_id, int w, int h);
   // transfers ownership of |buffer| from parent to child
-  async DeliverFrame(CaptureEngine capEngine, int cap_id,
-                     Shmem buffer, size_t size, uint32_t time_stamp,
-                     int64_t ntp_time, int64_t render_time);
+  async DeliverFrame(CaptureEngine capEngine, int streamId,
+                     Shmem buffer, VideoFrameProperties props);
   async DeviceChange();
   async ReplyNumberOfCaptureDevices(int numdev);
   async ReplyNumberOfCapabilities(int numdev);
   async ReplyAllocateCaptureDevice(int numdev);
-  async ReplyGetCaptureCapability(CaptureCapability cap);
+  async ReplyGetCaptureCapability(VideoCaptureCapability cap);
   async ReplyGetCaptureDevice(nsCString device_name, nsCString device_id, bool scary);
   async ReplyFailure();
   async ReplySuccess();
@@ -45,12 +71,13 @@ parent:
   async NumberOfCaptureDevices(CaptureEngine engine);
   async NumberOfCapabilities(CaptureEngine engine, nsCString deviceUniqueIdUTF8);
 
-  async GetCaptureCapability(CaptureEngine engine, nsCString unique_idUTF8, int capability_number);
+  async GetCaptureCapability(CaptureEngine engine, nsCString unique_idUTF8,
+                             int capability_number);
   async GetCaptureDevice(CaptureEngine engine, int num);
 
   async AllocateCaptureDevice(CaptureEngine engine, nsCString unique_idUTF8, nsCString origin);
   async ReleaseCaptureDevice(CaptureEngine engine, int numdev);
-  async StartCapture(CaptureEngine engine, int numdev, CaptureCapability capability);
+  async StartCapture(CaptureEngine engine, int numdev, VideoCaptureCapability capability);
   async StopCapture(CaptureEngine engine, int numdev);
   // transfers frame back
   async ReleaseFrame(Shmem s);
diff --git a/dom/media/systemservices/ShmemPool.cpp b/dom/media/systemservices/ShmemPool.cpp
index 0945be7acc..28bb0b15ff 100644
--- a/dom/media/systemservices/ShmemPool.cpp
+++ b/dom/media/systemservices/ShmemPool.cpp
@@ -39,7 +39,7 @@ mozilla::ShmemBuffer ShmemPool::GetIfAvailable(size_t aSize)
 
   MOZ_ASSERT(res.mShmem.IsWritable(), "Pool in Shmem is not writable?");
 
-  if (res.mShmem.Size<char>() < aSize) {
+  if (res.mShmem.Size<uint8_t>() < aSize) {
     LOG(("Free Shmem but not of the right size"));
     return ShmemBuffer();
   }
@@ -64,7 +64,7 @@ void ShmemPool::Put(ShmemBuffer&& aShmem)
 #ifdef DEBUG
   size_t poolUse = mShmemPool.Length() - mPoolFree;
   if (poolUse > 0) {
-    LOG(("ShmemPool usage reduced to %d buffers", poolUse));
+    LOG_VERBOSE(("ShmemPool usage reduced to %d buffers", poolUse));
   }
 #endif
 }
diff --git a/dom/media/systemservices/ShmemPool.h b/dom/media/systemservices/ShmemPool.h
index c209f2f630..d4aeca7ac5 100644
--- a/dom/media/systemservices/ShmemPool.h
+++ b/dom/media/systemservices/ShmemPool.h
@@ -47,8 +47,8 @@ public:
     return mInitialized;
   }
 
-  char* GetBytes() {
-    return mShmem.get<char>();
+  uint8_t * GetBytes() {
+    return mShmem.get<uint8_t>();
   }
 
   mozilla::ipc::Shmem& Get() {
diff --git a/dom/media/systemservices/VideoEngine.cpp b/dom/media/systemservices/VideoEngine.cpp
new file mode 100644
index 0000000000..53277d5318
--- /dev/null
+++ b/dom/media/systemservices/VideoEngine.cpp
@@ -0,0 +1,185 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "VideoEngine.h"
+#include "webrtc/video_engine/browser_capture_impl.h"
+#ifdef WEBRTC_ANDROID
+#include "webrtc/modules/video_capture/video_capture.h"
+#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+#include "webrtc/modules/video_render/video_render.h"
+#endif
+#endif
+
+
+namespace mozilla {
+namespace camera {
+
+#undef LOG
+#undef LOG_ENABLED
+mozilla::LazyLogModule gVideoEngineLog("VideoEngine");
+#define LOG(args) MOZ_LOG(gVideoEngineLog, mozilla::LogLevel::Debug, args)
+#define LOG_ENABLED() MOZ_LOG_TEST(gVideoEngineLog, mozilla::LogLevel::Debug)
+
+int VideoEngine::sId = 0;
+
+#if defined(ANDROID)
+int VideoEngine::SetAndroidObjects(JavaVM* javaVM) {
+  LOG((__PRETTY_FUNCTION__));
+
+  if (webrtc::SetCaptureAndroidVM(javaVM) != 0) {
+    LOG(("Could not set capture Android VM"));
+    return -1;
+  }
+#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
+  if (webrtc::SetRenderAndroidVM(javaVM) != 0) {
+    LOG(("Could not set render Android VM"));
+    return -1;
+  }
+#endif
+  return 0;
+}
+#endif
+
+void
+VideoEngine::CreateVideoCapture(int32_t& id, const char* deviceUniqueIdUTF8) {
+  LOG((__PRETTY_FUNCTION__));
+  id = GenerateId();
+  LOG(("CaptureDeviceInfo.type=%s id=%d",mCaptureDevInfo.TypeName(),id));
+  CaptureEntry entry = {-1,nullptr,nullptr};
+
+  if (mCaptureDevInfo.type == webrtc::CaptureDeviceType::Camera) {
+    entry = CaptureEntry(id,
+		         webrtc::VideoCaptureFactory::Create(id, deviceUniqueIdUTF8),
+                         nullptr);
+  } else {
+#ifndef WEBRTC_ANDROID
+    entry = CaptureEntry(
+	      id,
+	      webrtc::DesktopCaptureImpl::Create(id, deviceUniqueIdUTF8, mCaptureDevInfo.type),
+              nullptr);
+#else
+    MOZ_ASSERT("CreateVideoCapture NO DESKTOP CAPTURE IMPL ON ANDROID" == nullptr);
+#endif
+  }
+  mCaps.emplace(id,std::move(entry));
+}
+
+int
+VideoEngine::ReleaseVideoCapture(const int32_t id) {
+  bool found = false;
+  WithEntry(id, [&found](CaptureEntry& cap) {
+         cap.mVideoCaptureModule = nullptr;
+        found = true;
+   });
+  return found ? 0 : (-1);
+}
+
+std::shared_ptr<webrtc::VideoCaptureModule::DeviceInfo>
+VideoEngine::GetOrCreateVideoCaptureDeviceInfo() {
+  if (mDeviceInfo) {
+    return mDeviceInfo;
+  }
+  switch (mCaptureDevInfo.type) {
+    case webrtc::CaptureDeviceType::Camera: {
+      mDeviceInfo.reset(webrtc::VideoCaptureFactory::CreateDeviceInfo(0));
+      break;
+    }
+    case webrtc::CaptureDeviceType::Browser: {
+      mDeviceInfo.reset(webrtc::BrowserDeviceInfoImpl::CreateDeviceInfo());
+      break;
+    }
+    // Window, Application, and Screen types are handled by DesktopCapture
+    case webrtc::CaptureDeviceType::Window:
+    case webrtc::CaptureDeviceType::Application:
+    case webrtc::CaptureDeviceType::Screen: {
+#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
+      mDeviceInfo.reset(webrtc::DesktopCaptureImpl::CreateDeviceInfo(mId,mCaptureDevInfo.type));
+#else
+      MOZ_ASSERT("GetVideoCaptureDeviceInfo NO DESKTOP CAPTURE IMPL ON ANDROID" == nullptr);
+      mDeviceInfo.reset();
+#endif
+      break;
+    }
+  }
+  return mDeviceInfo;
+}
+
+void
+VideoEngine::RemoveRenderer(int capnum) {
+  WithEntry(capnum, [](CaptureEntry& cap) {
+    cap.mVideoRender = nullptr;
+  });
+}
+
+const UniquePtr<const webrtc::Config>&
+VideoEngine::GetConfiguration() {
+  return mConfig;
+}
+
+RefPtr<VideoEngine> VideoEngine::Create(UniquePtr<const webrtc::Config>&& aConfig) {
+  LOG((__PRETTY_FUNCTION__));
+  LOG(("Creating new VideoEngine with CaptureDeviceType %s",
+       aConfig->Get<webrtc::CaptureDeviceInfo>().TypeName()));
+  RefPtr<VideoEngine> engine(new VideoEngine(std::move(aConfig)));
+  return engine;
+}
+
+VideoEngine::CaptureEntry::CaptureEntry(int32_t aCapnum,
+                                        rtc::scoped_refptr<webrtc::VideoCaptureModule> aCapture,
+                                        webrtc::VideoRender * aRenderer):
+    mCapnum(aCapnum),
+    mVideoCaptureModule(aCapture),
+    mVideoRender(aRenderer)
+{}
+
+rtc::scoped_refptr<webrtc::VideoCaptureModule>
+VideoEngine::CaptureEntry::VideoCapture() {
+  return mVideoCaptureModule;
+}
+
+const UniquePtr<webrtc::VideoRender>&
+VideoEngine::CaptureEntry::VideoRenderer() {
+  if (!mVideoRender) {
+     MOZ_ASSERT(mCapnum != -1);
+     // Create a VideoRender on demand
+     mVideoRender = UniquePtr<webrtc::VideoRender>(
+         webrtc::VideoRender::CreateVideoRender(mCapnum,nullptr,false,webrtc::kRenderExternal));
+   }
+  return mVideoRender;
+}
+
+int32_t
+VideoEngine::CaptureEntry::Capnum() const {
+  return mCapnum;
+}
+
+bool VideoEngine::WithEntry(const int32_t entryCapnum,
+			    const std::function<void(CaptureEntry &entry)>&& fn) {
+  auto it = mCaps.find(entryCapnum);
+  if (it == mCaps.end()) {
+    return false;
+  }
+  fn(it->second);
+  return true;
+}
+
+int32_t
+VideoEngine::GenerateId() {
+  // XXX Something better than this (a map perhaps, or a simple boolean TArray, given
+  // the number in-use is O(1) normally!)
+  return mId = sId++;
+}
+
+VideoEngine::VideoEngine(UniquePtr<const webrtc::Config>&& aConfig):
+  mCaptureDevInfo(aConfig->Get<webrtc::CaptureDeviceInfo>()),
+  mDeviceInfo(nullptr),
+  mConfig(std::move(aConfig))
+{
+  LOG((__PRETTY_FUNCTION__));
+}
+
+}
+}
diff --git a/dom/media/systemservices/VideoEngine.h b/dom/media/systemservices/VideoEngine.h
new file mode 100644
index 0000000000..2830f8003b
--- /dev/null
+++ b/dom/media/systemservices/VideoEngine.h
@@ -0,0 +1,104 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_VideoEngine_h
+#define mozilla_VideoEngine_h
+
+#include "MediaEngine.h"
+#include "VideoFrameUtils.h"
+#include "mozilla/media/MediaUtils.h"
+#include "webrtc/common.h"
+#include "webrtc/modules/video_capture/video_capture_impl.h"
+#include "webrtc/modules/video_render/video_render.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
+#include "webrtc/video_engine/desktop_capture_impl.h"
+#include <memory>
+#include <functional>
+
+namespace mozilla {
+namespace camera {
+
+// Historically the video engine was part of webrtc
+// it was removed (and reimplemented in Talk)
+class VideoEngine
+{
+private:
+  virtual ~VideoEngine (){};
+
+public:
+  VideoEngine (){};
+  NS_INLINE_DECL_REFCOUNTING(VideoEngine)
+
+  static RefPtr<VideoEngine> Create(UniquePtr<const webrtc::Config>&& aConfig);
+#if defined(ANDROID)
+  static int SetAndroidObjects(JavaVM* javaVM);
+#endif
+  void CreateVideoCapture(int32_t& id, const char* deviceUniqueIdUTF8);
+
+  int ReleaseVideoCapture(const int32_t id);
+
+  // VideoEngine is responsible for any cleanup in its modules
+  static void Delete(VideoEngine * engine) { }
+
+  /** Returns or creates a new new DeviceInfo.
+  *   It is cached to prevent repeated lengthy polling for "realness"
+  *   of the hardware devices.  This could be handled in a more elegant
+  *   way in the future.
+  *   @return on failure the shared_ptr will be null, otherwise it will contain a DeviceInfo.
+  *   @see bug 1305212 https://bugzilla.mozilla.org/show_bug.cgi?id=1305212
+  */
+  std::shared_ptr<webrtc::VideoCaptureModule::DeviceInfo> GetOrCreateVideoCaptureDeviceInfo();
+
+  void RemoveRenderer(int capnum);
+
+  const UniquePtr<const webrtc::Config>& GetConfiguration();
+
+  void Startup() {
+    mIsRunning = true;
+  }
+
+  void Shutdown() {
+    mIsRunning = false;
+  }
+
+  bool IsRunning() const {
+    return mIsRunning;
+  }
+
+  class CaptureEntry {
+  public:
+    CaptureEntry(int32_t aCapnum,
+                 rtc::scoped_refptr<webrtc::VideoCaptureModule> aCapture,
+                 webrtc::VideoRender* aRenderer);
+    int32_t Capnum() const;
+    rtc::scoped_refptr<webrtc::VideoCaptureModule> VideoCapture();
+    const UniquePtr<webrtc::VideoRender> & VideoRenderer();
+  private:
+    int32_t mCapnum;
+    rtc::scoped_refptr<webrtc::VideoCaptureModule> mVideoCaptureModule;
+    UniquePtr<webrtc::VideoRender> mVideoRender;
+    friend class VideoEngine;
+  };
+
+  // Returns true iff an entry for capnum exists
+  bool WithEntry(const int32_t entryCapnum, const std::function<void(CaptureEntry &entry)>&& fn);
+
+private:
+  explicit VideoEngine(UniquePtr<const webrtc::Config>&& aConfig);
+  bool mIsRunning;
+  int32_t mId;
+  webrtc::CaptureDeviceInfo mCaptureDevInfo;
+  std::shared_ptr<webrtc::VideoCaptureModule::DeviceInfo> mDeviceInfo;
+  UniquePtr<const webrtc::Config> mConfig;
+  std::map<int32_t, CaptureEntry> mCaps;
+
+  int32_t GenerateId();
+  static int32_t sId;
+};
+}
+}
+#endif
diff --git a/dom/media/systemservices/VideoFrameUtils.cpp b/dom/media/systemservices/VideoFrameUtils.cpp
new file mode 100644
index 0000000000..e34ee8e53b
--- /dev/null
+++ b/dom/media/systemservices/VideoFrameUtils.cpp
@@ -0,0 +1,92 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "VideoFrameUtils.h"
+#include "webrtc/video_frame.h"
+#include "mozilla/ShmemPool.h"
+
+namespace mozilla {
+
+size_t
+VideoFrameUtils::TotalRequiredBufferSize(
+                  const webrtc::VideoFrame& aVideoFrame)
+{
+  static const webrtc::PlaneType kPlanes[] =
+                  {webrtc::kYPlane, webrtc::kUPlane, webrtc::kVPlane};
+  if (aVideoFrame.IsZeroSize()) {
+    return 0;
+  }
+
+  size_t sum = 0;
+  for (auto plane : kPlanes) {
+    sum += aVideoFrame.allocated_size(plane);
+  }
+  return sum;
+}
+
+void VideoFrameUtils::InitFrameBufferProperties(
+                  const webrtc::VideoFrame& aVideoFrame,
+                  camera::VideoFrameProperties& aDestProps)
+{
+  // The VideoFrameBuffer image data stored in the accompanying buffer
+  // the buffer is at least this size of larger.
+  aDestProps.bufferSize() = TotalRequiredBufferSize(aVideoFrame);
+
+  aDestProps.timeStamp() = aVideoFrame.timestamp();
+  aDestProps.ntpTimeMs() = aVideoFrame.ntp_time_ms();
+  aDestProps.renderTimeMs() = aVideoFrame.render_time_ms();
+
+  aDestProps.rotation() = aVideoFrame.rotation();
+
+  aDestProps.yAllocatedSize() = aVideoFrame.allocated_size(webrtc::kYPlane);
+  aDestProps.uAllocatedSize() = aVideoFrame.allocated_size(webrtc::kYPlane);
+  aDestProps.vAllocatedSize() = aVideoFrame.allocated_size(webrtc::kYPlane);
+
+  aDestProps.width() = aVideoFrame.width();
+  aDestProps.height() = aVideoFrame.height();
+
+  aDestProps.yStride() = aVideoFrame.stride(webrtc::kYPlane);
+  aDestProps.uStride() = aVideoFrame.stride(webrtc::kUPlane);
+  aDestProps.vStride() = aVideoFrame.stride(webrtc::kVPlane);
+}
+
+void VideoFrameUtils::CopyVideoFrameBuffers(uint8_t* aDestBuffer,
+                       const size_t aDestBufferSize,
+                       const webrtc::VideoFrame& aFrame)
+{
+  static const webrtc::PlaneType planes[] = {webrtc::kYPlane, webrtc::kUPlane, webrtc::kVPlane};
+
+  size_t aggregateSize = TotalRequiredBufferSize(aFrame);
+
+  MOZ_ASSERT(aDestBufferSize >= aggregateSize);
+
+  // If planes are ordered YUV and contiguous then do a single copy
+  if ((aFrame.buffer(webrtc::kYPlane) != nullptr)
+    // Check that the three planes are ordered
+    && (aFrame.buffer(webrtc::kYPlane) < aFrame.buffer(webrtc::kUPlane))
+    && (aFrame.buffer(webrtc::kUPlane) < aFrame.buffer(webrtc::kVPlane))
+    //  Check that the last plane ends at firstPlane[totalsize]
+    && (&aFrame.buffer(webrtc::kYPlane)[aggregateSize] == &aFrame.buffer(webrtc::kVPlane)[aFrame.allocated_size(webrtc::kVPlane)]))
+  {
+    memcpy(aDestBuffer,aFrame.buffer(webrtc::kYPlane),aggregateSize);
+    return;
+  }
+
+  // Copy each plane
+  size_t offset = 0;
+  for (auto plane: planes) {
+    memcpy(&aDestBuffer[offset], aFrame.buffer(plane), aFrame.allocated_size(plane));
+    offset += aFrame.allocated_size(plane);
+  }
+}
+
+void VideoFrameUtils::CopyVideoFrameBuffers(ShmemBuffer& aDestShmem,
+                        const webrtc::VideoFrame& aVideoFrame)
+{
+  CopyVideoFrameBuffers(aDestShmem.Get().get<uint8_t>(), aDestShmem.Get().Size<uint8_t>(), aVideoFrame);
+}
+
+}
diff --git a/dom/media/systemservices/VideoFrameUtils.h b/dom/media/systemservices/VideoFrameUtils.h
new file mode 100644
index 0000000000..efd9925552
--- /dev/null
+++ b/dom/media/systemservices/VideoFrameUtils.h
@@ -0,0 +1,51 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_VideoFrameUtil_h
+#define mozilla_VideoFrameUtil_h
+
+#include "mozilla/camera/PCameras.h"
+
+namespace webrtc {
+  class VideoFrame;
+}
+
+namespace mozilla
+{
+  class ShmemBuffer;
+
+// Util methods for working with webrtc::VideoFrame(s) and
+// the IPC classes that are used to deliver their contents to the
+// MediaEnginge
+
+class VideoFrameUtils {
+public:
+
+  // Returns the total number of bytes necessary to copy a VideoFrame's buffer
+  // across all planes.
+  static size_t TotalRequiredBufferSize(const webrtc::VideoFrame & frame);
+
+  // Initializes a camera::VideoFrameProperties from a VideoFrameBuffer
+  static void InitFrameBufferProperties(const webrtc::VideoFrame& aVideoFrame,
+                camera::VideoFrameProperties & aDestProperties);
+
+  // Copies the buffers out of a VideoFrameBuffer into a buffer.
+  // Attempts to make as few memcopies as possible.
+  static void CopyVideoFrameBuffers(uint8_t * aDestBuffer,
+                         const size_t aDestBufferSize,
+                         const webrtc::VideoFrame & aVideoFrame);
+
+  // Copies the buffers in a VideoFrameBuffer into a Shmem
+  // returns the eno from the underlying memcpy.
+  static void CopyVideoFrameBuffers(ShmemBuffer & aDestShmem,
+                         const webrtc::VideoFrame & aVideoFrame);
+
+
+};
+
+} /* namespace mozilla */
+
+#endif
diff --git a/dom/media/systemservices/moz.build b/dom/media/systemservices/moz.build
index 7d566c42ce..df57fcdeaf 100644
--- a/dom/media/systemservices/moz.build
+++ b/dom/media/systemservices/moz.build
@@ -1,4 +1,4 @@
-# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# -*- Mode: python; c-basic-offset: 4; indent-tabs-mode: nil; tab-width: 40 -*-
 # This Source Code Form is subject to the terms of the Mozilla Public
 # License, v. 2.0. If a copy of the MPL was not distributed with this
 # file, You can obtain one at http://mozilla.org/MPL/2.0/.
@@ -10,6 +10,8 @@ if CONFIG['MOZ_WEBRTC']:
         'LoadManager.h',
         'LoadManagerFactory.h',
         'LoadMonitor.h',
+        'VideoEngine.h',
+        'VideoFrameUtils.h'
     ]
     SOURCES += [
         'CamerasChild.cpp',
@@ -18,6 +20,8 @@ if CONFIG['MOZ_WEBRTC']:
         'LoadManagerFactory.cpp',
         'LoadMonitor.cpp',
         'ShmemPool.cpp',
+        'VideoEngine.cpp',
+        'VideoFrameUtils.cpp'
     ]
     LOCAL_INCLUDES += [
         '/media/webrtc/signaling',
@@ -29,6 +33,9 @@ if CONFIG['OS_TARGET'] == 'WINNT':
 else:
     DEFINES['WEBRTC_POSIX'] = True
 
+if CONFIG['OS_TARGET'] == 'Android':
+    DEFINES['WEBRTC_ANDROID'] = True
+
 
 if CONFIG['OS_TARGET'] == 'Android':
     EXPORTS += [
diff --git a/dom/media/tests/mochitest/mochitest.ini b/dom/media/tests/mochitest/mochitest.ini
index 22006ffa2f..a492979db5 100644
--- a/dom/media/tests/mochitest/mochitest.ini
+++ b/dom/media/tests/mochitest/mochitest.ini
@@ -42,7 +42,6 @@ skip-if = android_version == '18' # android(Bug 1189784, timeouts on 4.3 emulato
 skip-if = true # needed by test_enumerateDevices.html on builders
 [test_ondevicechange.html]
 skip-if = os == 'android'
-[test_getUserMedia_active_autoplay.html]
 [test_getUserMedia_audioCapture.html]
 skip-if = android_version == '18' # android(Bug 1189784, timeouts on 4.3 emulator)
 [test_getUserMedia_addTrackRemoveTrack.html]
@@ -99,7 +98,7 @@ skip-if = toolkit == 'android' # websockets don't work on android (bug 1266217)
 [test_peerConnection_basicAudioNATRelayTCP.html]
 skip-if = toolkit == 'android' # websockets don't work on android (bug 1266217)
 [test_peerConnection_basicAudioRequireEOC.html]
-skip-if = (android_version == '18' && debug) # android(Bug 1189784, timeouts on 4.3 emulator)
+skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emulator)
 [test_peerConnection_basicAudioPcmaPcmuOnly.html]
 skip-if = android_version == '18'
 [test_peerConnection_basicAudioDynamicPtMissingRtpmap.html]
@@ -127,7 +126,7 @@ skip-if = os == 'android' # bug 1043403
 [test_peerConnection_bug822674.html]
 [test_peerConnection_bug825703.html]
 [test_peerConnection_bug827843.html]
-skip-if = (android_version == '18' && debug) # android(Bug 1189784, timeouts on 4.3 emulator)
+skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emulator)
 [test_peerConnection_bug834153.html]
 [test_peerConnection_bug1013809.html]
 skip-if = (android_version == '18') # android(Bug 1189784, timeouts on 4.3 emulator)
diff --git a/dom/media/tests/mochitest/pc.js b/dom/media/tests/mochitest/pc.js
index a9383358f9..84714b9626 100644
--- a/dom/media/tests/mochitest/pc.js
+++ b/dom/media/tests/mochitest/pc.js
@@ -1415,13 +1415,17 @@ PeerConnectionWrapper.prototype = {
    *        A promise that resolves when media is flowing.
    */
   waitForRtpFlow(track) {
-    var hasFlow = stats => {
-      var rtp = stats.get([...stats.keys()].find(key =>
+    var hasFlow = (stats, retries) => {
+      info("Checking for stats in " + JSON.stringify(stats) + " for " + track.kind
+        + " track " + track.id + ", retry number " + retries);
+      var rtp = stats.get([...Object.keys(stats)].find(key =>
         !stats.get(key).isRemote && stats.get(key).type.endsWith("boundrtp")));
-      ok(rtp, "Should have RTP stats for track " + track.id);
       if (!rtp) {
+
         return false;
       }
+      info("Should have RTP stats for track " + track.id);
+      info("RTP stats: "+JSON.stringify(rtp));
       var nrPackets = rtp[rtp.type == "outboundrtp" ? "packetsSent"
                                                     : "packetsReceived"];
       info("Track " + track.id + " has " + nrPackets + " " +
@@ -1429,12 +1433,44 @@ PeerConnectionWrapper.prototype = {
       return nrPackets > 0;
     };
 
-    info("Checking RTP packet flow for track " + track.id);
+    // Time between stats checks
+    var retryInterval = 500;
+    // Timeout in ms
+    var timeoutInterval = 30000;
+    // Check hasFlow at a reasonable interval
+    var checkStats = new Promise((resolve, reject)=>{
+      var retries = 0;
+      var timer = setInterval(()=>{
+        this._pc.getStats(track).then(stats=>{
+          if (hasFlow(stats, retries)) {
+            clearInterval(timer);
+            ok(true, "RTP flowing for " + track.kind + " track " + track.id);
+            resolve();
+          }
+          retries = retries + 1;
+          // This is not accurate but it will tear down
+          // the timer eventually and probably not
+          // before timeoutInterval has elapsed.
+          if ((retries * retryInterval) > timeoutInterval) {
+            clearInterval(timer);
+          }
+        });
+      }, retryInterval);
+    });
 
-    var retry = (delay) => this._pc.getStats(track)
-      .then(stats => hasFlow(stats)? ok(true, "RTP flowing for track " + track.id) :
-            wait(delay).then(retry(1000)));
-    return retry(200);
+    info("Checking RTP packet flow for track " + track.id);
+    var retry = Promise.race([checkStats.then(new Promise((resolve, reject)=>{
+        info("checkStats completed for " + track.kind + " track " + track.id);
+        resolve();
+      })),
+      new Promise((accept,reject)=>wait(timeoutInterval).then(()=>{
+        info("Timeout checking for stats for track " + track.id + " after " + timeoutInterval + "ms");
+        reject("Timeout checking for stats for " + track.kind
+          + " track " + track.id + " after " + timeoutInterval + "ms");
+      })
+    )]);
+
+    return retry;
   },
 
   /**
@@ -1536,7 +1572,9 @@ PeerConnectionWrapper.prototype = {
       var minimum = this.whenCreated - 1000; // on Windows XP (Bug 979649)
       if (isWinXP) {
         todo(false, "Can't reliably test rtcp timestamps on WinXP (Bug 979649)");
-      } else if (!twoMachines) {
+
+      } else if (false) { // Bug 1325430 - timestamps aren't working properly in update 49
+	// else if (!twoMachines) {
         // Bug 1225729: On android, sometimes the first RTCP of the first
         // test run gets this value, likely because no RTP has been sent yet.
         if (res.timestamp != 2085978496000) {
@@ -1584,8 +1622,17 @@ PeerConnectionWrapper.prototype = {
               ok(rem.packetsReceived !== undefined, "Rtcp packetsReceived");
               ok(rem.packetsLost !== undefined, "Rtcp packetsLost");
               ok(rem.bytesReceived >= rem.packetsReceived, "Rtcp bytesReceived");
-              if (!this.disableRtpCountChecking) {
-                ok(rem.packetsReceived <= res.packetsSent, "No more than sent packets");
+	       if (false) { // Bug 1325430 if (!this.disableRtpCountChecking) {
+	       // no guarantee which one is newer!
+	       // Note: this must change when we add a timestamp field to remote RTCP reports
+	       // and make rem.timestamp be the reception time
+		if (res.timestamp >= rem.timestamp) {
+                 ok(rem.packetsReceived <= res.packetsSent, "No more than sent packets");
+		 } else {
+                  info("REVERSED timestamps: rec:" +
+		     rem.packetsReceived + " time:" + rem.timestamp + " sent:" + res.packetsSent + " time:" + res.timestamp);
+		 }
+		// Else we may have received more than outdated Rtcp packetsSent
                 ok(rem.bytesReceived <= res.bytesSent, "No more than sent bytes");
               }
               ok(rem.jitter !== undefined, "Rtcp jitter");
diff --git a/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_2d.html b/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_2d.html
index 97010929af..55ee2f3c77 100644
--- a/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_2d.html
+++ b/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_2d.html
@@ -46,6 +46,18 @@ runNetworkTest(() => {
       // i.e., this order of `requestFrame(); draw();` should work.
       stream.requestFrame();
       h.drawColor(canvas, h.red);
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          stream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
     },
     function PC_REMOTE_WAIT_FOR_REMOTE_RED() {
       return h.waitForPixelColor(mediaElement, h.red, 128,
diff --git a/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_webgl.html b/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_webgl.html
index d0ebee6ad3..cd827ec831 100644
--- a/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_webgl.html
+++ b/dom/media/tests/mochitest/test_peerConnection_captureStream_canvas_webgl.html
@@ -52,7 +52,7 @@ runNetworkTest(() => {
 
       if (!program) {
         ok(false, "Program should link");
-        return Promise.reject();
+        return Promise.reject("Program should link");
       }
       gl.useProgram(program);
 
@@ -100,6 +100,16 @@ runNetworkTest(() => {
     },
     function DRAW_LOCAL_RED() {
       h.drawColor(canvas, h.red);
+      return setInterval(function() {
+        try {
+          info("draw");
+          h.drawColor(canvas, h.red);
+          test.pcLocal.canvasStream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
     },
     function WAIT_FOR_REMOTE_RED() {
       return h.waitForPixelColor(vremote, h.red, 128,
diff --git a/dom/media/tests/mochitest/test_peerConnection_multiple_captureStream_canvas_2d.html b/dom/media/tests/mochitest/test_peerConnection_multiple_captureStream_canvas_2d.html
index 557cdb791f..0b0f3a0518 100644
--- a/dom/media/tests/mochitest/test_peerConnection_multiple_captureStream_canvas_2d.html
+++ b/dom/media/tests/mochitest/test_peerConnection_multiple_captureStream_canvas_2d.html
@@ -72,15 +72,38 @@ runNetworkTest(() => {
       // After requesting a frame it will be captured at the time of next render.
       // Next render will happen at next stable state, at the earliest,
       // i.e., this order of `requestFrame(); draw();` should work.
-      stream1.requestFrame();
       h.drawColor(canvas1, h.red);
+      stream1.requestFrame();
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas1, i ? h.green : h.red);
+          i = 1 - i;
+          stream1.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
     },
     function DRAW_LOCAL2_RED() {
       // After requesting a frame it will be captured at the time of next render.
       // Next render will happen at next stable state, at the earliest,
       // i.e., this order of `requestFrame(); draw();` should work.
-      stream2.requestFrame();
       h.drawColor(canvas2, h.red);
+      stream2.requestFrame();
+      return setInterval(function() {
+        try {
+          info("draw");
+          h.drawColor(canvas2, i ? h.green : h.red);
+          i = 1 - i;
+          stream2.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
     },
     function WAIT_FOR_REMOTE1_RED() {
       return h.waitForPixelColor(vremote1, h.red, 128,
diff --git a/dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html b/dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html
index de6aeb038f..1d6be86ddb 100644
--- a/dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html
+++ b/dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html
@@ -63,7 +63,8 @@
             test.originalOffer.sdp, test._remote_answer.sdp);
           info("Answer with RIDs: " + JSON.stringify(test._remote_answer));
           ok(test._remote_answer.sdp.match(/a=simulcast:/), "Modified answer has simulcast");
-          ok(test._remote_answer.sdp.match(/a=rid:/), "Modified answer has rid");
+          ok(test._remote_answer.sdp.match(/a=rid:foo/), "Modified answer has rid foo");
+          ok(test._remote_answer.sdp.match(/a=rid:bar/), "Modified answer has rid bar");
         }
       ]);
 
@@ -118,18 +119,8 @@
           ok(vremote, "Should have remote video element for pcRemote");
           ok(vlocal.videoWidth > 0, "source width is positive");
           ok(vlocal.videoHeight > 0, "source height is positive");
-          is(vremote.videoWidth, vlocal.videoWidth / 2, "sink is 1/2 width of source");
-          is(vremote.videoHeight, vlocal.videoHeight / 2,  "sink is 1/2 height of source");
-        },
-        function PC_REMOTE_SET_RTP_NONEXISTENT_RID(test) {
-          // Now, cause pcRemote to filter out everything, just to make sure
-          // selectRecvSsrc is working.
-          selectRecvSsrc(test.pcRemote, 2);
-        },
-        function PC_REMOTE_ENSURE_NO_FRAMES() {
-          var vremote = test.pcRemote.remoteMediaElements[0];
-          ok(vremote, "Should have remote video element for pcRemote");
-          return helper.verifyNoFrames(vremote);
+          is(vremote.videoWidth, vlocal.videoWidth, "sink is same width as source");
+          is(vremote.videoHeight, vlocal.videoHeight,  "sink is same height as source");
         },
       ]);
 
diff --git a/dom/media/tests/mochitest/test_peerConnection_verifyVideoAfterRenegotiation.html b/dom/media/tests/mochitest/test_peerConnection_verifyVideoAfterRenegotiation.html
index 4aeb5e7ed0..919a1bb404 100644
--- a/dom/media/tests/mochitest/test_peerConnection_verifyVideoAfterRenegotiation.html
+++ b/dom/media/tests/mochitest/test_peerConnection_verifyVideoAfterRenegotiation.html
@@ -51,6 +51,18 @@ runNetworkTest(() => {
       // i.e., this order of `requestFrame(); draw();` should work.
       stream1.requestFrame();
       h1.drawColor(canvas1, h1.red);
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h1.drawColor(canvas1, i ? h1.green : h1.red);
+          i = 1 - i;
+          stream1.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
     },
     function WAIT_FOR_REMOTE_RED() {
       return h1.waitForPixelColor(vremote1, h1.red, 128,
diff --git a/dom/media/webrtc/MediaEngineCameraVideoSource.cpp b/dom/media/webrtc/MediaEngineCameraVideoSource.cpp
index e63a9afded..6181a10687 100644
--- a/dom/media/webrtc/MediaEngineCameraVideoSource.cpp
+++ b/dom/media/webrtc/MediaEngineCameraVideoSource.cpp
@@ -390,22 +390,4 @@ MediaEngineCameraVideoSource::SetDirectListeners(bool aHasDirectListeners)
   mHasDirectListeners = aHasDirectListeners;
 }
 
-bool operator == (const webrtc::CaptureCapability& a,
-                  const webrtc::CaptureCapability& b)
-{
-  return a.width == b.width &&
-         a.height == b.height &&
-         a.maxFPS == b.maxFPS &&
-         a.rawType == b.rawType &&
-         a.codecType == b.codecType &&
-         a.expectedCaptureDelay == b.expectedCaptureDelay &&
-         a.interlaced == b.interlaced;
-};
-
-bool operator != (const webrtc::CaptureCapability& a,
-                  const webrtc::CaptureCapability& b)
-{
-  return !(a == b);
-}
-
 } // namespace mozilla
diff --git a/dom/media/webrtc/MediaEngineCameraVideoSource.h b/dom/media/webrtc/MediaEngineCameraVideoSource.h
index fb9113cd66..f5adba80c9 100644
--- a/dom/media/webrtc/MediaEngineCameraVideoSource.h
+++ b/dom/media/webrtc/MediaEngineCameraVideoSource.h
@@ -11,15 +11,18 @@
 
 // conflicts with #include of scoped_ptr.h
 #undef FF
-#include "webrtc/video_engine/include/vie_capture.h"
+// Avoid warnings about redefinition of WARN_UNUSED_RESULT
+#include "ipc/IPCMessageUtils.h"
+
+// WebRTC includes
+#include "webrtc/modules/video_capture/video_capture_defines.h"
+
+namespace webrtc {
+  using CaptureCapability = VideoCaptureCapability;
+}
 
 namespace mozilla {
 
-bool operator == (const webrtc::CaptureCapability& a,
-                  const webrtc::CaptureCapability& b);
-bool operator != (const webrtc::CaptureCapability& a,
-                  const webrtc::CaptureCapability& b);
-
 class MediaEngineCameraVideoSource : public MediaEngineVideoSource
 {
 public:
diff --git a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
index e79d8249c4..ce59a9f53e 100644
--- a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
@@ -350,23 +350,23 @@ MediaEngineRemoteVideoSource::NotifyPull(MediaStreamGraph* aGraph,
   }
 }
 
-int
-MediaEngineRemoteVideoSource::FrameSizeChange(unsigned int w, unsigned int h,
-                                              unsigned int streams)
+void
+MediaEngineRemoteVideoSource::FrameSizeChange(unsigned int w, unsigned int h)
 {
-  mWidth = w;
-  mHeight = h;
-  LOG(("MediaEngineRemoteVideoSource Video FrameSizeChange: %ux%u", w, h));
-  return 0;
+#if defined(MOZ_WIDGET_GONK)
+  mMonitor.AssertCurrentThreadOwns(); // mWidth and mHeight are protected...
+#endif
+  if ((mWidth < 0) || (mHeight < 0) ||
+      (w !=  (unsigned int) mWidth) || (h != (unsigned int) mHeight)) {
+    LOG(("MediaEngineRemoteVideoSource Video FrameSizeChange: %ux%u was %ux%u", w, h, mWidth, mHeight));
+    mWidth = w;
+    mHeight = h;
+  }
 }
 
 int
-MediaEngineRemoteVideoSource::DeliverFrame(unsigned char* buffer,
-                                           size_t size,
-                                           uint32_t time_stamp,
-                                           int64_t ntp_time,
-                                           int64_t render_time,
-                                           void *handle)
+MediaEngineRemoteVideoSource::DeliverFrame(uint8_t* aBuffer ,
+                                    const camera::VideoFrameProperties& aProps)
 {
   // Check for proper state.
   if (mState != kStarted) {
@@ -374,15 +374,13 @@ MediaEngineRemoteVideoSource::DeliverFrame(unsigned char* buffer,
     return 0;
   }
 
-  if ((size_t) (mWidth*mHeight + 2*(((mWidth+1)/2)*((mHeight+1)/2))) != size) {
-    MOZ_ASSERT(false, "Wrong size frame in DeliverFrame!");
-    return 0;
-  }
+  // Update the dimensions
+  FrameSizeChange(aProps.width(), aProps.height());
 
   // Create a video frame and append it to the track.
   RefPtr<layers::PlanarYCbCrImage> image = mImageContainer->CreatePlanarYCbCrImage();
 
-  uint8_t* frame = static_cast<uint8_t*> (buffer);
+  uint8_t* frame = static_cast<uint8_t*> (aBuffer);
   const uint8_t lumaBpp = 8;
   const uint8_t chromaBpp = 4;
 
@@ -407,8 +405,9 @@ MediaEngineRemoteVideoSource::DeliverFrame(unsigned char* buffer,
 
 #ifdef DEBUG
   static uint32_t frame_num = 0;
-  LOGFRAME(("frame %d (%dx%d); timestamp %u, ntp_time %" PRIu64 ", render_time %" PRIu64,
-            frame_num++, mWidth, mHeight, time_stamp, ntp_time, render_time));
+  LOGFRAME(("frame %d (%dx%d); timeStamp %u, ntpTimeMs %" PRIu64 ", renderTimeMs %" PRIu64,
+            frame_num++, mWidth, mHeight,
+            aProps.timeStamp(), aProps.ntpTimeMs(), aProps.renderTimeMs()));
 #endif
 
   // we don't touch anything in 'this' until here (except for snapshot,
diff --git a/dom/media/webrtc/MediaEngineRemoteVideoSource.h b/dom/media/webrtc/MediaEngineRemoteVideoSource.h
index 712761f97c..60ed827d51 100644
--- a/dom/media/webrtc/MediaEngineRemoteVideoSource.h
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.h
@@ -19,6 +19,8 @@
 #include "nsDirectoryServiceDefs.h"
 #include "nsComponentManagerUtils.h"
 
+// Avoid warnings about redefinition of WARN_UNUSED_RESULT
+#include "ipc/IPCMessageUtils.h"
 #include "VideoUtils.h"
 #include "MediaEngineCameraVideoSource.h"
 #include "VideoSegment.h"
@@ -31,40 +33,29 @@
 
 // WebRTC library includes follow
 #include "webrtc/common.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_render.h"
+
+// Camera Access via IPC
 #include "CamerasChild.h"
 
 #include "NullTransport.h"
 
-namespace webrtc {
-class I420VideoFrame;
-}
-
 namespace mozilla {
 
 /**
  * The WebRTC implementation of the MediaEngine interface.
  */
 class MediaEngineRemoteVideoSource : public MediaEngineCameraVideoSource,
-                                     public webrtc::ExternalRenderer
+                                     public camera::FrameRelay
 {
   typedef MediaEngineCameraVideoSource Super;
 public:
   NS_DECL_THREADSAFE_ISUPPORTS
 
+  // Old ExternalRenderer
+  void FrameSizeChange(unsigned int w, unsigned int h) override;
   // ExternalRenderer
-  int FrameSizeChange(unsigned int w, unsigned int h,
-                      unsigned int streams) override;
-  int DeliverFrame(unsigned char* buffer,
-                   size_t size,
-                   uint32_t time_stamp,
-                   int64_t ntp_time,
-                   int64_t render_time,
-                   void *handle) override;
-  // XXX!!!! FIX THIS
-  int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override { return 0; };
-  bool IsTextureSupported() override { return false; };
+  int DeliverFrame(uint8_t* buffer,
+                   const camera::VideoFrameProperties& properties) override;
 
   // MediaEngineCameraVideoSource
   MediaEngineRemoteVideoSource(int aIndex, mozilla::camera::CaptureEngine aCapEngine,
diff --git a/dom/media/webrtc/MediaEngineWebRTC.h b/dom/media/webrtc/MediaEngineWebRTC.h
index 1834f3bd39..090523e0db 100644
--- a/dom/media/webrtc/MediaEngineWebRTC.h
+++ b/dom/media/webrtc/MediaEngineWebRTC.h
@@ -24,6 +24,7 @@
 #include "nsComponentManagerUtils.h"
 #include "nsRefPtrHashtable.h"
 
+#include "ipc/IPCMessageUtils.h"
 #include "VideoUtils.h"
 #include "MediaEngineCameraVideoSource.h"
 #include "VideoSegment.h"
@@ -36,6 +37,8 @@
 
 #include "MediaEngineWrapper.h"
 #include "mozilla/dom/MediaStreamTrackBinding.h"
+#include "CamerasChild.h"
+
 // WebRTC library includes follow
 #include "webrtc/common.h"
 // Audio Engine
@@ -52,11 +55,9 @@
 // Video Engine
 // conflicts with #include of scoped_ptr.h
 #undef FF
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "CamerasChild.h"
+
+// WebRTC imports
+#include "webrtc/modules/video_capture/video_capture_defines.h"
 
 #include "NullTransport.h"
 #include "AudioOutputObserver.h"
@@ -256,10 +257,10 @@ public:
     MOZ_ASSERT(mDevices);
 
     if (mInUseCount == 0) {
-      ScopedCustomReleasePtr<webrtc::VoEExternalMedia> ptrVoERender;
-      ptrVoERender = webrtc::VoEExternalMedia::GetInterface(mVoiceEngine);
-      if (ptrVoERender) {
-        ptrVoERender->SetExternalRecordingStatus(true);
+      ScopedCustomReleasePtr<webrtc::VoEExternalMedia> ptrVoEXMedia;
+      ptrVoEXMedia = webrtc::VoEExternalMedia::GetInterface(mVoiceEngine);
+      if (ptrVoEXMedia) {
+        ptrVoEXMedia->SetExternalRecordingStatus(true);
       }
       mAnyInUse = true;
     }
@@ -475,9 +476,9 @@ public:
       const nsString& aDeviceId) const override;
 
   // VoEMediaProcess.
-  void Process(int channel, webrtc::ProcessingTypes type,
-               int16_t audio10ms[], int length,
-               int samplingFreq, bool isStereo) override;
+  virtual void Process(int channel, webrtc::ProcessingTypes type,
+                       int16_t audio10ms[], size_t length,
+                       int samplingFreq, bool isStereo) override;
 
   void Shutdown() override;
 
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
index 0eda3aac1b..1cdb732274 100644
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -824,7 +824,7 @@ typedef int16_t sample;
 void
 MediaEngineWebRTCMicrophoneSource::Process(int channel,
                                            webrtc::ProcessingTypes type,
-                                           sample *audio10ms, int length,
+                                           sample *audio10ms, size_t length,
                                            int samplingFreq, bool isStereo)
 {
   MOZ_ASSERT(!PassThrough(), "This should be bypassed when in PassThrough mode.");
diff --git a/dom/media/webrtc/MediaTrackConstraints.cpp b/dom/media/webrtc/MediaTrackConstraints.cpp
index 42582e3c29..27cac5e1c2 100644
--- a/dom/media/webrtc/MediaTrackConstraints.cpp
+++ b/dom/media/webrtc/MediaTrackConstraints.cpp
@@ -13,6 +13,9 @@
 
 namespace mozilla {
 
+using dom::ConstrainBooleanParameters;
+using dom::OwningLongOrConstrainLongRange;
+
 template<class ValueType>
 template<class ConstrainRange>
 void
diff --git a/media/mtransport/test/stunserver.cpp b/media/mtransport/test/stunserver.cpp
index 5c9ea4c1ac..13f614ef80 100644
--- a/media/mtransport/test/stunserver.cpp
+++ b/media/mtransport/test/stunserver.cpp
@@ -81,6 +81,11 @@ nrappkit copyright:
 #include "mozilla/Unused.h"
 #include "databuffer.h"
 
+// mozilla/utils.h defines this as well
+#ifdef UNIMPLEMENTED
+#undef UNIMPLEMENTED
+#endif
+
 extern "C" {
 #include "nr_api.h"
 #include "async_wait.h"
diff --git a/media/webrtc/moz.build b/media/webrtc/moz.build
index 33dc63269a..909a6e7517 100644
--- a/media/webrtc/moz.build
+++ b/media/webrtc/moz.build
@@ -8,18 +8,18 @@ include('/build/gyp.mozbuild')
 webrtc_non_unified_sources = [
     'trunk/webrtc/common_audio/vad/vad_core.c',                                  # Because of name clash in the kInitCheck variable
     'trunk/webrtc/common_audio/vad/webrtc_vad.c',                                # Because of name clash in the kInitCheck variable
+    'trunk/webrtc/modules/audio_coding/acm2/codec_manager.cc',                   # Because of duplicate IsCodecRED/etc
     'trunk/webrtc/modules/audio_coding/codecs/g722/g722_decode.c',               # Because of name clash in the saturate function
     'trunk/webrtc/modules/audio_coding/codecs/g722/g722_encode.c',               # Because of name clash in the saturate function
     'trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c',   # Because of name clash in the kDampFilter variable
     'trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c', # Because of name clash in the kDampFilter variable
-    'trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.cc',              # Because of duplicate IsCodecRED/etc
     'trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc',                   # Because of explicit template specializations
+    'trunk/webrtc/modules/audio_device/android/audio_manager.cc',                # Because of TAG redefinition
     'trunk/webrtc/modules/audio_device/android/audio_record_jni.cc',             # Becuse of commonly named module static vars
-    'trunk/webrtc/modules/audio_device/android/audio_track_jni.cc',             # Becuse of commonly named module static vars
+    'trunk/webrtc/modules/audio_device/android/audio_track_jni.cc',              # Becuse of commonly named module static vars
+    'trunk/webrtc/modules/audio_device/android/opensles_player.cc',              # Because of TAG redefinition
     'trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc',       # Because of LATE()
     'trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc',# Because of LATE()
-    'trunk/webrtc/modules/audio_device/opensl/opensles_input.cc',                # Because of name clash in the kOption variable
-    'trunk/webrtc/modules/audio_device/opensl/opensles_output.cc',               # Because of name clash in the kOption variable
     'trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc',                # Because of name clash with #define FF
     'trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc',            # Because of ordering assumptions in strsafe.h
     'trunk/webrtc/modules/audio_processing/aec/aec_core.c',                      # Because of name clash in the ComfortNoise function
@@ -34,12 +34,15 @@ webrtc_non_unified_sources = [
     'trunk/webrtc/modules/audio_processing/gain_control_impl.cc',                # Because of name clash in the Handle typedef
     'trunk/webrtc/modules/audio_processing/high_pass_filter_impl.cc',            # Because of name clash in the Handle typedef
     'trunk/webrtc/modules/audio_processing/noise_suppression_impl.cc',           # Because of name clash in the Handle typedef
-    'trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.cc',        # Because of duplicate definitions of static consts against aimd_rate_control.cc
     'trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc', # Because of duplicate definitions of static consts against remote_bitrate_estimator_abs_send_time.cc
+    'trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc',                  # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
+    'trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc',                  # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
+    'trunk/webrtc/modules/video_capture/android/device_info_android.cc',         # Because of duplicate module static variable names
+    'trunk/webrtc/modules/video_capture/android/video_capture_android.cc',       # Because of duplicate module static variable names
     'trunk/webrtc/modules/video_capture/windows/device_info_ds.cc',              # Because of the MEDIASUBTYPE_HDYC variable
     'trunk/webrtc/modules/video_capture/windows/help_functions_ds.cc',           # Because of initguid.h
     'trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc',              # Because of the MEDIASUBTYPE_HDYC variable and initguid.h
-    'trunk/webrtc/video_engine/overuse_frame_detector.cc',                       # Because of name clash with call_stats.cc on kWeightFactor
+    'trunk/webrtc/video/overuse_frame_detector.cc',                              # Because of name clash with call_stats.cc on kWeightFactor
 ]
 
 GYP_DIRS += ['trunk']
@@ -65,7 +68,6 @@ if CONFIG['MOZ_WEBRTC_SIGNALING']:
         'signaling/src/common/browser_logging/CSFLog.cpp',
         'signaling/src/jsep/JsepSessionImpl.cpp',
         'signaling/src/media-conduit/AudioConduit.cpp',
-        'signaling/src/media-conduit/CodecStatistics.cpp',
         'signaling/src/media-conduit/MediaCodecVideoCodec.cpp',
         'signaling/src/media-conduit/VideoConduit.cpp',
         'signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.cpp',
diff --git a/media/webrtc/signaling/signaling.gyp b/media/webrtc/signaling/signaling.gyp
index 2de7dcbf19..753150f937 100644
--- a/media/webrtc/signaling/signaling.gyp
+++ b/media/webrtc/signaling/signaling.gyp
@@ -94,8 +94,6 @@
         './src/media-conduit/AudioConduit.cpp',
         './src/media-conduit/VideoConduit.h',
         './src/media-conduit/VideoConduit.cpp',
-        './src/media-conduit/CodecStatistics.h',
-        './src/media-conduit/CodecStatistics.cpp',
         './src/media-conduit/RunningStat.h',
         # Common
         './src/common/CommonTypes.h',
diff --git a/media/webrtc/signaling/src/common/EncodingConstraints.h b/media/webrtc/signaling/src/common/EncodingConstraints.h
index 32add7c474..180fcbfd93 100644
--- a/media/webrtc/signaling/src/common/EncodingConstraints.h
+++ b/media/webrtc/signaling/src/common/EncodingConstraints.h
@@ -41,6 +41,18 @@ public:
       scaleDownBy == constraints.scaleDownBy;
   }
 
+  /**
+   * This returns true if the constraints affecting resolution are equal.
+   */
+  bool ResolutionEquals(const EncodingConstraints& constraints) const
+  {
+    return
+      maxWidth == constraints.maxWidth &&
+      maxHeight == constraints.maxHeight &&
+      maxFs == constraints.maxFs &&
+      scaleDownBy == constraints.scaleDownBy;
+  }
+
   uint32_t maxWidth;
   uint32_t maxHeight;
   uint32_t maxFps;
diff --git a/media/webrtc/signaling/src/common/NullTransport.h b/media/webrtc/signaling/src/common/NullTransport.h
index bce793304c..1d1d13de3a 100644
--- a/media/webrtc/signaling/src/common/NullTransport.h
+++ b/media/webrtc/signaling/src/common/NullTransport.h
@@ -8,7 +8,7 @@
 
 #include "mozilla/Attributes.h"
 
-#include "webrtc/common_types.h"
+#include "webrtc/transport.h"
 
 namespace mozilla {
 
@@ -18,6 +18,23 @@ namespace mozilla {
 class NullTransport : public webrtc::Transport
 {
 public:
+  virtual bool SendRtp(const uint8_t* packet,
+                       size_t length,
+                       const webrtc::PacketOptions& options) override
+  {
+    (void) packet;
+    (void) length;
+    (void) options;
+    return true;
+  }
+
+  virtual bool SendRtcp(const uint8_t* packet, size_t length) override
+  {
+    (void) packet;
+    (void) length;
+    return true;
+  }
+#if 0
   virtual int SendPacket(int channel, const void *data, size_t len)
   {
     (void) channel; (void) data;
@@ -29,7 +46,7 @@ public:
     (void) channel; (void) data;
     return len;
   }
-
+#endif
   NullTransport() {}
 
   virtual ~NullTransport() {}
diff --git a/media/webrtc/signaling/src/common/browser_logging/WebRtcLog.cpp b/media/webrtc/signaling/src/common/browser_logging/WebRtcLog.cpp
index 875e0ed2cd..bf17f42fda 100644
--- a/media/webrtc/signaling/src/common/browser_logging/WebRtcLog.cpp
+++ b/media/webrtc/signaling/src/common/browser_logging/WebRtcLog.cpp
@@ -6,7 +6,7 @@
 
 #include "mozilla/Logging.h"
 #include "prenv.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include "nscore.h"
 #ifdef MOZILLA_INTERNAL_API
diff --git a/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp b/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
index f5015dda2d..5375110b60 100644
--- a/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
+++ b/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
@@ -4,6 +4,7 @@
 
 #include "logging.h"
 
+#include "webrtc/config.h"
 #include "signaling/src/jsep/JsepSessionImpl.h"
 #include <string>
 #include <set>
@@ -116,15 +117,35 @@ JsepSessionImpl::AddTrack(const RefPtr<JsepTrack>& track)
 {
   mLastError.clear();
   MOZ_ASSERT(track->GetDirection() == sdp::kSend);
-
+  MOZ_MTLOG(ML_DEBUG, "Adding track.");
   if (track->GetMediaType() != SdpMediaSection::kApplication) {
     track->SetCNAME(mCNAME);
-
-    if (track->GetSsrcs().empty()) {
-      uint32_t ssrc;
+    // Establish minimum number of required SSRCs
+    // Note that AddTrack is only for send direction
+    size_t minimumSsrcCount = 0;
+    std::vector<JsepTrack::JsConstraints> constraints;
+    track->GetJsConstraints(&constraints);
+    for (auto constraint : constraints) {
+      if (constraint.rid != "") {
+        minimumSsrcCount++;
+      }
+    }
+    // We need at least 1 SSRC
+    minimumSsrcCount = std::max<size_t>(1, minimumSsrcCount);
+    size_t currSsrcCount = track->GetSsrcs().size();
+    if (currSsrcCount < minimumSsrcCount ) {
+      MOZ_MTLOG(ML_DEBUG,
+                "Adding " << (minimumSsrcCount - currSsrcCount) << " SSRCs.");
+    }
+    while (track->GetSsrcs().size() < minimumSsrcCount) {
+      uint32_t ssrc=0;
       nsresult rv = CreateSsrc(&ssrc);
       NS_ENSURE_SUCCESS(rv, rv);
-      track->AddSsrc(ssrc);
+      // Don't add duplicate ssrcs
+      std::vector<uint32_t> ssrcs = track->GetSsrcs();
+      if (std::find(ssrcs.begin(), ssrcs.end(), ssrc) == ssrcs.end()) {
+        track->AddSsrc(ssrc);
+      }
     }
   }
 
@@ -287,19 +308,62 @@ JsepSessionImpl::SetParameters(const std::string& streamId,
   // Add RtpStreamId Extmap
   // SdpDirectionAttribute::Direction is a bitmask
   SdpDirectionAttribute::Direction addVideoExt = SdpDirectionAttribute::kInactive;
+  SdpDirectionAttribute::Direction addAudioExt = SdpDirectionAttribute::kInactive;
   for (auto constraintEntry: constraints) {
     if (constraintEntry.rid != "") {
-      if (it->mTrack->GetMediaType() == SdpMediaSection::kVideo) {
-        addVideoExt = static_cast<SdpDirectionAttribute::Direction>(addVideoExt
-                                                                    | it->mTrack->GetDirection());
+      switch (it->mTrack->GetMediaType()) {
+        case SdpMediaSection::kVideo: {
+           addVideoExt = static_cast<SdpDirectionAttribute::Direction>(addVideoExt
+                                                                       | it->mTrack->GetDirection());
+          break;
+        }
+        case SdpMediaSection::kAudio: {
+          addAudioExt = static_cast<SdpDirectionAttribute::Direction>(addAudioExt
+                                                                      | it->mTrack->GetDirection());
+          break;
+        }
       }
     }
   }
   if (addVideoExt != SdpDirectionAttribute::kInactive) {
     AddVideoRtpExtension("urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", addVideoExt);
   }
+  if (addAudioExt != SdpDirectionAttribute::kInactive) {
+    AddAudioRtpExtension("urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id", addAudioExt);
+  }
 
   it->mTrack->SetJsConstraints(constraints);
+
+  auto track = it->mTrack;
+  if (track->GetDirection() == sdp::kSend) {
+    // Establish minimum number of required SSRCs
+    // Note that AddTrack is only for send direction
+    size_t minimumSsrcCount = 0;
+    std::vector<JsepTrack::JsConstraints> constraints;
+    track->GetJsConstraints(&constraints);
+    for (auto constraint : constraints) {
+      if (constraint.rid != "") {
+        minimumSsrcCount++;
+      }
+    }
+    // We need at least 1 SSRC
+    minimumSsrcCount = std::max<size_t>(1, minimumSsrcCount);
+    size_t currSsrcCount = track->GetSsrcs().size();
+    if (currSsrcCount < minimumSsrcCount ) {
+      MOZ_MTLOG(ML_DEBUG,
+                "Adding " << (minimumSsrcCount - currSsrcCount) << " SSRCs.");
+    }
+    while (track->GetSsrcs().size() < minimumSsrcCount) {
+      uint32_t ssrc=0;
+      nsresult rv = CreateSsrc(&ssrc);
+      NS_ENSURE_SUCCESS(rv, rv);
+      // Don't add duplicate ssrcs
+      std::vector<uint32_t> ssrcs = track->GetSsrcs();
+      if (std::find(ssrcs.begin(), ssrcs.end(), ssrc) == ssrcs.end()) {
+        track->AddSsrc(ssrc);
+      }
+    }
+  }
   return NS_OK;
 }
 
@@ -2218,6 +2282,7 @@ JsepSessionImpl::SetupDefaultCodecs()
       );
   mSupportedCodecs.values.push_back(ulpfec);
 
+
   mSupportedCodecs.values.push_back(new JsepApplicationCodecDescription(
       "5000",
       "webrtc-datachannel",
diff --git a/media/webrtc/signaling/src/jsep/JsepTrack.cpp b/media/webrtc/signaling/src/jsep/JsepTrack.cpp
index cf5df96bf0..24d7bb4999 100644
--- a/media/webrtc/signaling/src/jsep/JsepTrack.cpp
+++ b/media/webrtc/signaling/src/jsep/JsepTrack.cpp
@@ -261,6 +261,9 @@ JsepTrack::CreateEncodings(
     const std::vector<JsepCodecDescription*>& negotiatedCodecs,
     JsepTrackNegotiatedDetails* negotiatedDetails)
 {
+  negotiatedDetails->mTias = remote.GetBandwidth("TIAS");
+  // TODO add support for b=AS if TIAS is not set (bug 976521)
+
   std::vector<SdpRidAttributeList::Rid> rids;
   GetRids(remote, sdp::kRecv, &rids); // Get rids we will send
   NegotiateRids(rids, &mJsEncodeConstraints);
@@ -294,8 +297,6 @@ JsepTrack::CreateEncodings(
         encoding->mConstraints = jsConstraints.constraints;
       }
     }
-
-    encoding->UpdateMaxBitrate(remote);
   }
 }
 
diff --git a/media/webrtc/signaling/src/jsep/JsepTrack.h b/media/webrtc/signaling/src/jsep/JsepTrack.h
index 5aa37404ff..0dd4ea3ffc 100644
--- a/media/webrtc/signaling/src/jsep/JsepTrack.h
+++ b/media/webrtc/signaling/src/jsep/JsepTrack.h
@@ -5,6 +5,7 @@
 #ifndef _JSEPTRACK_H_
 #define _JSEPTRACK_H_
 
+#include <functional>
 #include <algorithm>
 #include <string>
 #include <map>
@@ -28,6 +29,10 @@ namespace mozilla {
 class JsepTrackNegotiatedDetails
 {
 public:
+  JsepTrackNegotiatedDetails() :
+    mTias(0)
+  {}
+
   size_t
   GetEncodingCount() const
   {
@@ -51,17 +56,32 @@ public:
     return nullptr;
   }
 
+  void
+  ForEachRTPHeaderExtension(
+    const std::function<void(const SdpExtmapAttributeList::Extmap& extmap)> & fn) const
+  {
+    for(auto entry: mExtmap) {
+      fn(entry.second);
+    }
+  }
+
   std::vector<uint8_t> GetUniquePayloadTypes() const
   {
     return mUniquePayloadTypes;
   }
 
+  uint32_t GetTias() const
+  {
+    return mTias;
+  }
+
 private:
   friend class JsepTrack;
 
   std::map<std::string, SdpExtmapAttributeList::Extmap> mExtmap;
   std::vector<uint8_t> mUniquePayloadTypes;
   PtrVector<JsepTrackEncoding> mEncodings;
+  uint32_t mTias; // bits per second
 };
 
 class JsepTrack
diff --git a/media/webrtc/signaling/src/jsep/JsepTrackEncoding.h b/media/webrtc/signaling/src/jsep/JsepTrackEncoding.h
index 61e778fe6e..d241f79a98 100644
--- a/media/webrtc/signaling/src/jsep/JsepTrackEncoding.h
+++ b/media/webrtc/signaling/src/jsep/JsepTrackEncoding.h
@@ -38,16 +38,6 @@ public:
     return false;
   }
 
-  void UpdateMaxBitrate(const SdpMediaSection& remote)
-  {
-    uint32_t tias = remote.GetBandwidth("TIAS");
-    // select minimum of the two which is not zero
-    mConstraints.maxBr = std::min(tias ? tias : mConstraints.maxBr,
-                                  mConstraints.maxBr ? mConstraints.maxBr :
-                                                       tias);
-    // TODO add support for b=AS if TIAS is not set (bug 976521)
-  }
-
   EncodingConstraints mConstraints;
   std::string mRid;
 
diff --git a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
index 42a50533a4..38faee871e 100644
--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
@@ -25,11 +25,11 @@
 
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_dtmf.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace mozilla {
 
@@ -89,7 +89,9 @@ WebrtcAudioConduit::~WebrtcAudioConduit()
     mPtrVoEBase->StopSend(mChannel);
     mPtrVoEBase->StopReceive(mChannel);
     mPtrVoEBase->DeleteChannel(mChannel);
-    mPtrVoEBase->Terminate();
+    // We don't Terminate() the VoEBase here, because the Call (owned by
+    // PeerConnectionMedia) actually owns the (shared) VoEBase/VoiceEngine
+    // here
   }
 
   // We shouldn't delete the VoiceEngine until all these are released!
@@ -103,21 +105,24 @@ WebrtcAudioConduit::~WebrtcAudioConduit()
   mPtrVoERTP_RTCP = nullptr;
   mPtrRTP = nullptr;
 
-  if(mVoiceEngine)
+  if (mVoiceEngine)
   {
     webrtc::VoiceEngine::Delete(mVoiceEngine);
   }
 }
 
-bool WebrtcAudioConduit::SetLocalSSRC(unsigned int ssrc)
+bool WebrtcAudioConduit::SetLocalSSRCs(const std::vector<unsigned int> & aSSRCs)
 {
-  unsigned int oldSsrc;
-  if (!GetLocalSSRC(&oldSsrc)) {
+  // This should hold true until the WebRTC.org VoE refactor
+  MOZ_ASSERT(aSSRCs.size() == 1,"WebrtcAudioConduit::SetLocalSSRCs accepts exactly 1 ssrc.");
+
+  std::vector<unsigned int> oldSsrcs = GetLocalSSRCs();
+  if (oldSsrcs.empty()) {
     MOZ_ASSERT(false, "GetLocalSSRC failed");
     return false;
   }
 
-  if (oldSsrc == ssrc) {
+  if (oldSsrcs == aSSRCs) {
     return true;
   }
 
@@ -126,7 +131,7 @@ bool WebrtcAudioConduit::SetLocalSSRC(unsigned int ssrc)
     return false;
   }
 
-  if (mPtrRTP->SetLocalSSRC(mChannel, ssrc)) {
+  if (mPtrRTP->SetLocalSSRC(mChannel, aSSRCs[0])) {
     return false;
   }
 
@@ -138,8 +143,12 @@ bool WebrtcAudioConduit::SetLocalSSRC(unsigned int ssrc)
   return true;
 }
 
-bool WebrtcAudioConduit::GetLocalSSRC(unsigned int* ssrc) {
-  return !mPtrRTP->GetLocalSSRC(mChannel, *ssrc);
+std::vector<unsigned int> WebrtcAudioConduit::GetLocalSSRCs() const {
+  unsigned int ssrc;
+  if (!mPtrRTP->GetLocalSSRC(mChannel, ssrc)) {
+    return std::vector<unsigned int>(1,ssrc);
+  }
+  return std::vector<unsigned int>();
 }
 
 bool WebrtcAudioConduit::GetRemoteSSRC(unsigned int* ssrc) {
@@ -270,6 +279,13 @@ MediaConduitErrorCode WebrtcAudioConduit::Init()
     return kMediaConduitSessionNotInited;
   }
 
+  // init the engine with our audio device layer
+  if(mPtrVoEBase->Init() == -1)
+  {
+    CSFLogError(logTag, "%s VoiceEngine Base Not Initialized", __FUNCTION__);
+    return kMediaConduitSessionNotInited;
+  }
+
   if(!(mPtrVoENetwork = VoENetwork::GetInterface(mVoiceEngine)))
   {
     CSFLogError(logTag, "%s Unable to initialize VoENetwork", __FUNCTION__);
@@ -310,13 +326,6 @@ MediaConduitErrorCode WebrtcAudioConduit::Init()
     return kMediaConduitSessionNotInited;
   }
 
-  // init the engine with our audio device layer
-  if(mPtrVoEBase->Init() == -1)
-  {
-    CSFLogError(logTag, "%s VoiceEngine Base Not Initialized", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
   if( (mChannel = mPtrVoEBase->CreateChannel()) == -1)
   {
     CSFLogError(logTag, "%s VoiceEngine Channel creation failed",__FUNCTION__);
@@ -857,7 +866,6 @@ WebrtcAudioConduit::StartReceiving()
       return kMediaConduitUnknownError;
     }
 
-
     if(mPtrVoEBase->StartPlayout(mChannel) == -1)
     {
       CSFLogError(logTag, "%s Starting playout Failed", __FUNCTION__);
@@ -871,9 +879,12 @@ WebrtcAudioConduit::StartReceiving()
 
 //WebRTC::RTP Callback Implementation
 // Called on AudioGUM or MSG thread
-int WebrtcAudioConduit::SendPacket(int channel, const void* data, size_t len)
+bool
+WebrtcAudioConduit::SendRtp(const uint8_t* data,
+                            size_t len,
+                            const webrtc::PacketOptions& options)
 {
-  CSFLogDebug(logTag,  "%s : channel %d", __FUNCTION__, channel);
+  CSFLogDebug(logTag,  "%s: len %lu", __FUNCTION__, (unsigned long)len);
 
 #if !defined(MOZILLA_EXTERNAL_LINKAGE)
   if (MOZ_LOG_TEST(GetLatencyLog(), LogLevel::Debug)) {
@@ -888,25 +899,30 @@ int WebrtcAudioConduit::SendPacket(int channel, const void* data, size_t len)
   }
 #endif
   ReentrantMonitorAutoEnter enter(mTransportMonitor);
+  // XXX(pkerr) - the PacketOptions are being ignored. This parameter was added along
+  // with the Call API update in the webrtc.org codebase.
+  // The only field in it is the packet_id, which is used when the header
+  // extension for TransportSequenceNumber is being used, which we don't.
+  (void) options;
   if(mTransmitterTransport &&
      (mTransmitterTransport->SendRtpPacket(data, len) == NS_OK))
   {
     CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
-    return len;
+    return true;
   } else {
     CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
-    return -1;
+    return false;
   }
 }
 
 // Called on WebRTC Process thread and perhaps others
-int WebrtcAudioConduit::SendRTCPPacket(int channel, const void* data, size_t len)
+bool
+WebrtcAudioConduit::SendRtcp(const uint8_t* data, size_t len)
 {
-  CSFLogDebug(logTag,  "%s : channel %d , len %lu, first rtcp = %u ",
+  CSFLogDebug(logTag,  "%s : len %lu, first rtcp = %u ",
               __FUNCTION__,
-              channel,
               (unsigned long) len,
-              static_cast<unsigned>(((uint8_t *) data)[1]));
+              static_cast<unsigned>(data[1]));
 
   // We come here if we have only one pipeline/conduit setup,
   // such as for unidirectional streams.
@@ -917,14 +933,14 @@ int WebrtcAudioConduit::SendRTCPPacket(int channel, const void* data, size_t len
   {
     // Might be a sender report, might be a receiver report, we don't know.
     CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
-    return len;
+    return true;
   } else if(mTransmitterTransport &&
             (mTransmitterTransport->SendRtcpPacket(data, len) == NS_OK)) {
       CSFLogDebug(logTag, "%s Sent RTCP Packet (sender report) ", __FUNCTION__);
-      return len;
+      return true;
   } else {
     CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
-    return -1;
+    return false;
   }
 }
 
diff --git a/media/webrtc/signaling/src/media-conduit/AudioConduit.h b/media/webrtc/signaling/src/media-conduit/AudioConduit.h
index fcc7e0f372..dec3ae7b7e 100644
--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.h
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.h
@@ -15,6 +15,7 @@
 
 // Audio Engine Includes
 #include "webrtc/common_types.h"
+#include "webrtc/transport.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
@@ -45,8 +46,8 @@ NTPtoDOMHighResTimeStamp(uint32_t ntpHigh, uint32_t ntpLow);
  * Concrete class for Audio session. Hooks up
  *  - media-source and target to external transport
  */
-class WebrtcAudioConduit:public AudioSessionConduit
-	      		            ,public webrtc::Transport
+class WebrtcAudioConduit: public AudioSessionConduit
+	      		, public webrtc::Transport
 {
 public:
   //VoiceEngine defined constant for Payload Name Size.
@@ -150,20 +151,22 @@ public:
    * Webrtc transport implementation to send and receive RTP packet.
    * AudioConduit registers itself as ExternalTransport to the VoiceEngine
    */
-  virtual int SendPacket(int channel, const void *data, size_t len) override;
+  virtual bool SendRtp(const uint8_t* data,
+                       size_t len,
+                       const webrtc::PacketOptions& options) override;
 
   /**
    * Webrtc transport implementation to send and receive RTCP packet.
    * AudioConduit registers itself as ExternalTransport to the VoiceEngine
    */
-  virtual int SendRTCPPacket(int channel, const void *data, size_t len) override;
-
+  virtual bool SendRtcp(const uint8_t *data,
+                        size_t len) override;
 
   virtual uint64_t CodecPluginID() override { return 0; }
 
   virtual void DeleteStreams() override {}
-  
-  WebrtcAudioConduit():
+
+  explicit WebrtcAudioConduit():
                       mVoiceEngine(nullptr),
                       mTransportMonitor("WebrtcAudioConduit"),
                       mTransmitterTransport(nullptr),
@@ -188,8 +191,17 @@ public:
 
   int GetChannel() { return mChannel; }
   webrtc::VoiceEngine* GetVoiceEngine() { return mVoiceEngine; }
-  bool SetLocalSSRC(unsigned int ssrc) override;
-  bool GetLocalSSRC(unsigned int* ssrc) override;
+
+  /* Set Local SSRC list.
+   * Note: Until the refactor of the VoE into the call API is complete
+   *   this list should contain only a single ssrc.
+   */
+  bool SetLocalSSRCs(const std::vector<unsigned int>& aSSRCs) override;
+  std::vector<unsigned int> GetLocalSSRCs() const override;
+  bool SetRemoteSSRC(unsigned int ssrc) override
+  {
+    return false;
+  }
   bool GetRemoteSSRC(unsigned int* ssrc) override;
   bool SetLocalCNAME(const char* cname) override;
   bool GetVideoEncoderStats(double* framerateMean,
diff --git a/media/webrtc/signaling/src/media-conduit/CodecConfig.h b/media/webrtc/signaling/src/media-conduit/CodecConfig.h
index 308c979481..06772ed357 100644
--- a/media/webrtc/signaling/src/media-conduit/CodecConfig.h
+++ b/media/webrtc/signaling/src/media-conduit/CodecConfig.h
@@ -90,10 +90,15 @@ public:
   bool mRembFbSet;
   bool mFECFbSet;
 
+  uint32_t mTias;
   EncodingConstraints mEncodingConstraints;
   struct SimulcastEncoding {
     std::string rid;
     EncodingConstraints constraints;
+    bool operator==(const SimulcastEncoding& aOther) const {
+      return rid == aOther.rid &&
+        constraints == aOther.constraints;
+    }
   };
   std::vector<SimulcastEncoding> mSimulcastEncodings;
   std::string mSpropParameterSets;
@@ -103,6 +108,28 @@ public:
   uint8_t mPacketizationMode;
   // TODO: add external negotiated SPS/PPS
 
+  bool operator==(const VideoCodecConfig& aRhs) const {
+    if (mType != aRhs.mType ||
+        mName != aRhs.mName ||
+        mAckFbTypes != aRhs.mAckFbTypes ||
+        mNackFbTypes != aRhs.mNackFbTypes ||
+        mCcmFbTypes != aRhs.mCcmFbTypes ||
+        mRembFbSet != aRhs.mRembFbSet ||
+        mFECFbSet != aRhs.mFECFbSet ||
+        mTias != aRhs.mTias ||
+        !(mEncodingConstraints == aRhs.mEncodingConstraints) ||
+        !(mSimulcastEncodings == aRhs.mSimulcastEncodings) ||
+        mSpropParameterSets != aRhs.mSpropParameterSets ||
+        mProfile != aRhs.mProfile ||
+        mConstraints != aRhs.mConstraints ||
+        mLevel != aRhs.mLevel ||
+        mPacketizationMode != aRhs.mPacketizationMode) {
+      return false;
+    }
+
+    return true;
+  }
+
   VideoCodecConfig(int type,
                    std::string name,
                    const EncodingConstraints& constraints,
@@ -110,6 +137,7 @@ public:
     mType(type),
     mName(name),
     mFECFbSet(false),
+    mTias(0),
     mEncodingConstraints(constraints),
     mProfile(0x42),
     mConstraints(0xE0),
@@ -125,6 +153,20 @@ public:
     }
   }
 
+  bool ResolutionEquals(const VideoCodecConfig& aConfig) const
+  {
+    if (mSimulcastEncodings.size() != aConfig.mSimulcastEncodings.size()) {
+      return false;
+    }
+    for (size_t i = 0; i < mSimulcastEncodings.size(); ++i) {
+      if (!mSimulcastEncodings[i].constraints.ResolutionEquals(
+            aConfig.mSimulcastEncodings[i].constraints)) {
+        return false;
+      }
+    }
+    return true;
+  }
+
   // Nothing seems to use this right now. Do we intend to support this
   // someday?
   bool RtcpFbAckIsSet(const std::string& type) const
diff --git a/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.cpp b/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.cpp
index 0c4d81e447..d3e49e1fc7 100644
--- a/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.cpp
+++ b/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.cpp
@@ -7,12 +7,12 @@
 
 namespace mozilla {
 
-VideoEncoder* GmpVideoCodec::CreateEncoder() {
-  return static_cast<VideoEncoder*>(new WebrtcVideoEncoderProxy());
+WebrtcVideoEncoder* GmpVideoCodec::CreateEncoder() {
+  return new WebrtcVideoEncoderProxy();
 }
 
-VideoDecoder* GmpVideoCodec::CreateDecoder() {
-  return static_cast<VideoDecoder*>(new WebrtcVideoDecoderProxy());
+WebrtcVideoDecoder* GmpVideoCodec::CreateDecoder() {
+  return new WebrtcVideoDecoderProxy();
 }
 
 }
diff --git a/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.h b/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.h
index 3401504093..365ec0efda 100644
--- a/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.h
+++ b/media/webrtc/signaling/src/media-conduit/GmpVideoCodec.h
@@ -10,8 +10,8 @@
 namespace mozilla {
 class GmpVideoCodec {
  public:
-  static VideoEncoder* CreateEncoder();
-  static VideoDecoder* CreateDecoder();
+  static WebrtcVideoEncoder* CreateEncoder();
+  static WebrtcVideoDecoder* CreateDecoder();
 };
 
 }
diff --git a/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.cpp b/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.cpp
index 0c6c2fdde0..2318606226 100644
--- a/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.cpp
+++ b/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.cpp
@@ -12,7 +12,7 @@ namespace mozilla {
 
 static const char* logTag ="MediaCodecVideoCodec";
 
-VideoEncoder* MediaCodecVideoCodec::CreateEncoder(CodecType aCodecType) {
+WebrtcVideoEncoder* MediaCodecVideoCodec::CreateEncoder(CodecType aCodecType) {
   CSFLogDebug(logTag,  "%s ", __FUNCTION__);
   if (aCodecType == CODEC_VP8) {
      return new WebrtcMediaCodecVP8VideoEncoder();
@@ -20,7 +20,7 @@ VideoEncoder* MediaCodecVideoCodec::CreateEncoder(CodecType aCodecType) {
   return nullptr;
 }
 
-VideoDecoder* MediaCodecVideoCodec::CreateDecoder(CodecType aCodecType) {
+WebrtcVideoDecoder* MediaCodecVideoCodec::CreateDecoder(CodecType aCodecType) {
   CSFLogDebug(logTag,  "%s ", __FUNCTION__);
   if (aCodecType == CODEC_VP8) {
     return new WebrtcMediaCodecVP8VideoDecoder();
diff --git a/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.h b/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.h
index 50dde82114..3ab6887ace 100644
--- a/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.h
+++ b/media/webrtc/signaling/src/media-conduit/MediaCodecVideoCodec.h
@@ -17,13 +17,13 @@ class MediaCodecVideoCodec {
    * Create encoder object for codec type |aCodecType|. Return |nullptr| when
    * failed.
    */
-  static VideoEncoder* CreateEncoder(CodecType aCodecType);
+  static WebrtcVideoEncoder* CreateEncoder(CodecType aCodecType);
 
   /**
    * Create decoder object for codec type |aCodecType|. Return |nullptr| when
    * failed.
    */
-  static VideoDecoder* CreateDecoder(CodecType aCodecType);
+  static WebrtcVideoDecoder* CreateDecoder(CodecType aCodecType);
 };
 
 }
diff --git a/media/webrtc/signaling/src/media-conduit/MediaConduitErrors.h b/media/webrtc/signaling/src/media-conduit/MediaConduitErrors.h
index 3709d59a03..729943f3a8 100644
--- a/media/webrtc/signaling/src/media-conduit/MediaConduitErrors.h
+++ b/media/webrtc/signaling/src/media-conduit/MediaConduitErrors.h
@@ -39,10 +39,10 @@ kMediaConduitKeyFrameRequestError,     // Can't set KeyFrameRequest mode
 kMediaConduitNACKStatusError,          // Can't set NACK mode
 kMediaConduitTMMBRStatusError,         // Can't set TMMBR mode
 kMediaConduitFECStatusError,           // Can't set FEC mode
-kMediaConduitHybridNACKFECStatusError  // Can't set Hybrid NACK / FEC mode
+kMediaConduitHybridNACKFECStatusError, // Can't set Hybrid NACK / FEC mode
+kMediaConduitVideoSendStreamError      // WebRTC video send stream failure
 };
 
 }
 
 #endif
-
diff --git a/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h b/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
index 0654b1175a..7e59d3ece8 100644
--- a/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
+++ b/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
@@ -9,20 +9,64 @@
 #include "nsXPCOM.h"
 #include "nsDOMNavigationTiming.h"
 #include "mozilla/RefPtr.h"
+#include "mozilla/RefCounted.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/utils.h"
 #include "CodecConfig.h"
 #include "VideoTypes.h"
 #include "MediaConduitErrors.h"
 
 #include "ImageContainer.h"
 
+#include "webrtc/call.h"
+#include "webrtc/config.h"
 #include "webrtc/common_types.h"
-namespace webrtc {
-class I420VideoFrame;
-}
 
 #include <vector>
 
+namespace webrtc {
+class VideoFrame;
+}
+
 namespace mozilla {
+
+// Wrap the webrtc.org Call class adding mozilla add/ref support.
+class WebRtcCallWrapper : public RefCounted<WebRtcCallWrapper>
+{
+public:
+  typedef webrtc::Call::Config Config;
+
+  static RefPtr<WebRtcCallWrapper> Create(const Config& config)
+  {
+    return new WebRtcCallWrapper(webrtc::Call::Create(config));
+  }
+
+  webrtc::Call* Call() const
+  {
+    return mCall.get();
+  }
+
+  virtual ~WebRtcCallWrapper()
+  {
+    if (mCall->voice_engine()) {
+      webrtc::VoiceEngine* voice_engine = mCall->voice_engine();
+      mCall.reset(nullptr); // Force it to release the voice engine reference
+      // Delete() must be after all refs are released
+      webrtc::VoiceEngine::Delete(voice_engine);
+    }
+  }
+
+  MOZ_DECLARE_REFCOUNTED_TYPENAME(WebRtcCallWrapper)
+
+private:
+  WebRtcCallWrapper() = delete;
+  explicit WebRtcCallWrapper(webrtc::Call* aCall)
+    : mCall(aCall) {}
+  DISALLOW_COPY_AND_ASSIGN(WebRtcCallWrapper);
+  UniquePtr<webrtc::Call> mCall;
+};
+
+
 /**
  * Abstract Interface for transporting RTP packets - audio/vidoeo
  * The consumers of this interface are responsible for passing in
@@ -40,7 +84,7 @@ public:
    * @param len  : Length of the media packet
    * @result     : NS_OK on success, NS_ERROR_FAILURE otherwise
    */
-  virtual nsresult SendRtpPacket(const void* data, int len) = 0;
+  virtual nsresult SendRtpPacket(const uint8_t* data, size_t len) = 0;
 
   /**
    * RTCP Transport Function to be implemented by concrete transport implementation
@@ -48,7 +92,7 @@ public:
    * @param len  : Length of the RTCP packet
    * @result     : NS_OK on success, NS_ERROR_FAILURE otherwise
    */
-  virtual nsresult SendRtcpPacket(const void* data, int len) = 0;
+  virtual nsresult SendRtcpPacket(const uint8_t* data, size_t len) = 0;
   NS_INLINE_DECL_THREADSAFE_REFCOUNTING(TransportInterface)
 };
 
@@ -193,9 +237,15 @@ public:
    */
   virtual MediaConduitErrorCode SetReceiverTransport(RefPtr<TransportInterface> aTransport) = 0;
 
-  virtual bool SetLocalSSRC(unsigned int ssrc) = 0;
-  virtual bool GetLocalSSRC(unsigned int* ssrc) = 0;
+  /* Sets the local SSRCs
+   * @return true iff the local ssrcs == aSSRCs upon return
+   * Note: this is an ordered list and {a,b,c} != {b,a,c}
+   */
+  virtual bool SetLocalSSRCs(const std::vector<unsigned int>& aSSRCs) = 0;
+  virtual std::vector<unsigned int> GetLocalSSRCs() const = 0;
+
   virtual bool GetRemoteSSRC(unsigned int* ssrc) = 0;
+  virtual bool SetRemoteSSRC(unsigned int ssrc) = 0;
   virtual bool SetLocalCNAME(const char* cname) = 0;
 
   /**
@@ -265,10 +315,12 @@ class VideoSessionConduit : public MediaSessionConduit
 public:
   /**
    * Factory function to create and initialize a Video Conduit Session
-   * return: Concrete VideoSessionConduitObject or nullptr in the case
+   * @param  webrtc::Call instance shared by paired audio and video
+   *         media conduits
+   * @result Concrete VideoSessionConduitObject or nullptr in the case
    *         of failure
    */
-  static RefPtr<VideoSessionConduit> Create();
+  static RefPtr<VideoSessionConduit> Create(RefPtr<WebRtcCallWrapper> aCall);
 
   enum FrameRequestType
   {
@@ -287,15 +339,28 @@ public:
 
   virtual Type type() const { return VIDEO; }
 
+  /**
+  * Adds negotiated RTP extensions
+  */
+  virtual void AddLocalRTPExtensions(const std::vector<webrtc::RtpExtension>& extensions) = 0;
+
+  /**
+  * Returns the negotiated RTP extensions
+  */
+  virtual std::vector<webrtc::RtpExtension> GetLocalRTPExtensions() const = 0;
+
+
   /**
    * Function to attach Renderer end-point of the Media-Video conduit.
    * @param aRenderer : Reference to the concrete Video renderer implementation
    * Note: Multiple invocations of this API shall remove an existing renderer
    * and attaches the new to the Conduit.
    */
-  virtual MediaConduitErrorCode AttachRenderer(RefPtr<VideoRenderer> aRenderer) = 0;
+  virtual MediaConduitErrorCode AttachRenderer(RefPtr<mozilla::VideoRenderer> aRenderer) = 0;
   virtual void DetachRenderer() = 0;
 
+  virtual bool SetRemoteSSRC(unsigned int ssrc) = 0;
+
   /**
    * Function to deliver a capture video frame for encoding and transport
    * @param video_frame: pointer to captured video-frame.
@@ -313,7 +378,7 @@ public:
                                                unsigned short height,
                                                VideoType video_type,
                                                uint64_t capture_time) = 0;
-  virtual MediaConduitErrorCode SendVideoFrame(webrtc::I420VideoFrame& frame) = 0;
+  virtual MediaConduitErrorCode SendVideoFrame(webrtc::VideoFrame& frame) = 0;
 
   virtual MediaConduitErrorCode ConfigureCodecMode(webrtc::VideoCodecMode) = 0;
   /**
@@ -335,31 +400,7 @@ public:
    *
    */
   virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
-                                const std::vector<VideoCodecConfig* >& recvCodecConfigList) = 0;
-
-  /**
-   * Set an external encoder
-   * @param encoder
-   * @result: on success, we will use the specified encoder
-   */
-  virtual MediaConduitErrorCode SetExternalSendCodec(VideoCodecConfig* config,
-                                                     VideoEncoder* encoder) = 0;
-
-  /**
-   * Set an external decoder
-   * @param decoder
-   * @result: on success, we will use the specified decoder
-   */
-  virtual MediaConduitErrorCode SetExternalRecvCodec(VideoCodecConfig* config,
-                                                     VideoDecoder* decoder) = 0;
-
-  /**
-   * Function to enable the RTP Stream ID (RID) extension
-   * @param enabled: enable extension
-   * @param id: id to be used for this rtp header extension
-   * NOTE: See VideoConduit for more information
-   */
-  virtual MediaConduitErrorCode EnableRTPStreamIdExtension(bool enabled, uint8_t id) = 0;
+      const std::vector<VideoCodecConfig* >& recvCodecConfigList) = 0;
 
   /**
    * These methods allow unit tests to double-check that the
@@ -410,11 +451,13 @@ class AudioSessionConduit : public MediaSessionConduit
 {
 public:
 
-   /**
-    * Factory function to create and initialize an Audio Conduit Session
-    * return: Concrete AudioSessionConduitObject or nullptr in the case
-    *         of failure
-    */
+ /**
+   * Factory function to create and initialize an Audio Conduit Session
+   * @param  webrtc::Call instance shared by paired audio and video
+   *         media conduits
+   * @result Concrete AudioSessionConduitObject or nullptr in the case
+   *         of failure
+   */
   static RefPtr<AudioSessionConduit> Create();
 
   virtual ~AudioSessionConduit() {}
diff --git a/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.cpp b/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.cpp
index d46398402a..bb91b1fe54 100644
--- a/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.cpp
+++ b/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.cpp
@@ -10,7 +10,7 @@
 
 namespace mozilla {
 
-VideoEncoder*
+WebrtcVideoEncoder*
 OMXVideoCodec::CreateEncoder(CodecType aCodecType)
 {
   if (aCodecType == CODEC_H264) {
@@ -19,7 +19,7 @@ OMXVideoCodec::CreateEncoder(CodecType aCodecType)
   return nullptr;
 }
 
-VideoDecoder*
+WebrtcVideoDecoder*
 OMXVideoCodec::CreateDecoder(CodecType aCodecType) {
   if (aCodecType == CODEC_H264) {
     return new WebrtcOMXH264VideoDecoder();
diff --git a/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.h b/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.h
index 51df50263c..cfb3e6288e 100644
--- a/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.h
+++ b/media/webrtc/signaling/src/media-conduit/OMXVideoCodec.h
@@ -18,13 +18,13 @@ class OMXVideoCodec {
    * Create encoder object for codec type |aCodecType|. Return |nullptr| when
    * failed.
    */
-  static VideoEncoder* CreateEncoder(CodecType aCodecType);
+  static WebrtcVideoEncoder* CreateEncoder(CodecType aCodecType);
 
   /**
    * Create decoder object for codec type |aCodecType|. Return |nullptr| when
    * failed.
    */
-  static VideoDecoder* CreateDecoder(CodecType aCodecType);
+  static WebrtcVideoDecoder* CreateDecoder(CodecType aCodecType);
 };
 
 }
diff --git a/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp b/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
index e6db06a685..15c43ac0da 100644
--- a/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
@@ -6,25 +6,41 @@
 #include "nspr.h"
 #include "plstr.h"
 
-#include "VideoConduit.h"
 #include "AudioConduit.h"
-#include "nsThreadUtils.h"
 #include "LoadManager.h"
+#include "VideoConduit.h"
 #include "YuvStamper.h"
-#include "nsServiceManagerUtils.h"
-#include "nsIPrefService.h"
-#include "nsIPrefBranch.h"
-#include "mozilla/media/MediaUtils.h"
 #include "mozilla/TemplateLib.h"
+#include "mozilla/media/MediaUtils.h"
+#include "nsComponentManagerUtils.h"
+#include "nsIPrefBranch.h"
+#include "nsIGfxInfo.h"
+#include "nsIPrefService.h"
+#include "nsServiceManagerUtils.h"
+
+#include "nsThreadUtils.h"
+
+#include "pk11pub.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/native_handle.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 #include "mozilla/Unused.h"
 
+#include "GmpVideoCodec.h"
+#ifdef MOZ_WEBRTC_OMX
+#include "OMXCodecWrapper.h"
+#include "OMXVideoCodec.h"
+#endif
+
+#ifdef MOZ_WEBRTC_MEDIACODEC
+#include "MediaCodecVideoCodec.h"
+#endif
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+#include "WebrtcGmpVideoCodec.h"
+#endif
+
 // for ntohs
 #ifdef _MSC_VER
 #include "Winsock2.h"
@@ -34,194 +50,729 @@
 
 #include <algorithm>
 #include <math.h>
+#include <cinttypes>
 
 #define DEFAULT_VIDEO_MAX_FRAMERATE 30
-#define INVALID_RTP_PAYLOAD 255  //valid payload types are 0 to 127
+#define INVALID_RTP_PAYLOAD 255 // valid payload types are 0 to 127
 
 namespace mozilla {
 
-static const char* logTag ="WebrtcVideoSessionConduit";
+static const char* logTag = "WebrtcVideoSessionConduit";
+
+static const int kUlpFecPayloadType = 123;
+static const int kRedPayloadType = 122;
+static const int kNullPayloadType = -1;
+static const char* kUlpFecPayloadName = "ulpfec";
+static const char* kRedPayloadName = "red";
+
+// Convert (SI) kilobits/sec to (SI) bits/sec
+#define KBPS(kbps) kbps * 1000
+const uint32_t WebrtcVideoConduit::kDefaultMinBitrate_bps =  KBPS(200);
+const uint32_t WebrtcVideoConduit::kDefaultStartBitrate_bps = KBPS(300);
+const uint32_t WebrtcVideoConduit::kDefaultMaxBitrate_bps = KBPS(2000);
 
 // 32 bytes is what WebRTC CodecInst expects
 const unsigned int WebrtcVideoConduit::CODEC_PLNAME_SIZE = 32;
+static const int kViEMinCodecBitrate = 30;
+
+template<typename T>
+T MinIgnoreZero(const T& a, const T& b)
+{
+  return std::min(a? a:b, b? b:a);
+}
+
+void
+WebrtcVideoConduit::StreamStatistics::Update(const double aFrameRate,
+                                             const double aBitrate)
+{
+  mFrameRate.Push(aFrameRate);
+  mBitrate.Push(aBitrate);
+}
+
+bool
+WebrtcVideoConduit::StreamStatistics::GetVideoStreamStats(
+    double& aOutFrMean, double& aOutFrStdDev, double& aOutBrMean,
+    double& aOutBrStdDev) const
+{
+  if (mFrameRate.NumDataValues() && mBitrate.NumDataValues()) {
+    aOutFrMean = mFrameRate.Mean();
+    aOutFrStdDev = mFrameRate.StandardDeviation();
+    aOutBrMean = mBitrate.Mean();
+    aOutBrStdDev = mBitrate.StandardDeviation();
+    return true;
+  }
+  return false;
+};
+
+void
+WebrtcVideoConduit::SendStreamStatistics::DroppedFrames(
+  uint32_t& aOutDroppedFrames) const
+{
+      aOutDroppedFrames = mDroppedFrames;
+};
+
+void
+WebrtcVideoConduit::SendStreamStatistics::Update(
+  const webrtc::VideoSendStream::Stats& aStats)
+{
+  StreamStatistics::Update(aStats.encode_frame_rate, aStats.media_bitrate_bps);
+  if (!aStats.substreams.empty()) {
+    const webrtc::FrameCounts& fc =
+      aStats.substreams.begin()->second.frame_counts;
+    CSFLogVerbose(logTag, "%s: framerate: %u, bitrate: %u, dropped frames delta: %u",
+                  __FUNCTION__, aStats.encode_frame_rate, aStats.media_bitrate_bps,
+                  (mSentFrames - (fc.key_frames + fc.delta_frames)) - mDroppedFrames);
+    mDroppedFrames = mSentFrames - (fc.key_frames + fc.delta_frames);
+  } else {
+    CSFLogVerbose(logTag, "%s aStats.substreams is empty", __FUNCTION__);
+  }
+};
+
+void
+WebrtcVideoConduit::ReceiveStreamStatistics::DiscardedPackets(
+  uint32_t& aOutDiscPackets) const
+{
+  aOutDiscPackets = mDiscardedPackets;
+};
+
+void
+WebrtcVideoConduit::ReceiveStreamStatistics::Update(
+  const webrtc::VideoReceiveStream::Stats& aStats)
+{
+  CSFLogVerbose(logTag, "%s ", __FUNCTION__);
+  StreamStatistics::Update(aStats.decode_frame_rate, aStats.total_bitrate_bps);
+  mDiscardedPackets = aStats.discarded_packets;
+};
 
 /**
  * Factory Method for VideoConduit
  */
 RefPtr<VideoSessionConduit>
-VideoSessionConduit::Create()
+VideoSessionConduit::Create(RefPtr<WebRtcCallWrapper> aCall)
 {
   NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  NS_ASSERTION(aCall, "missing required parameter: aCall");
+  CSFLogVerbose(logTag, "%s", __FUNCTION__);
 
-  WebrtcVideoConduit* obj = new WebrtcVideoConduit();
-  if(obj->Init() != kMediaConduitNoError)
-  {
-    CSFLogError(logTag,  "%s VideoConduit Init Failed ", __FUNCTION__);
-    delete obj;
+  if (!aCall) {
     return nullptr;
   }
-  CSFLogDebug(logTag,  "%s Successfully created VideoConduit ", __FUNCTION__);
-  return obj;
+
+  nsAutoPtr<WebrtcVideoConduit> obj(new WebrtcVideoConduit(aCall));
+  if(obj->Init() != kMediaConduitNoError) {
+    CSFLogError(logTag, "%s VideoConduit Init Failed ", __FUNCTION__);
+    return nullptr;
+  }
+  CSFLogVerbose(logTag, "%s Successfully created VideoConduit ", __FUNCTION__);
+  return obj.forget();
 }
 
-WebrtcVideoConduit::WebrtcVideoConduit():
-  mVideoEngine(nullptr),
-  mTransportMonitor("WebrtcVideoConduit"),
-  mTransmitterTransport(nullptr),
-  mReceiverTransport(nullptr),
-  mRenderer(nullptr),
-  mPtrExtCapture(nullptr),
-  mEngineTransmitting(false),
-  mEngineReceiving(false),
-  mChannel(-1),
-  mCapId(-1),
-  mCodecMutex("VideoConduit codec db"),
-  mInReconfig(false),
-  mLastWidth(0), // forces a check for reconfig at start
-  mLastHeight(0),
-  mSendingWidth(0),
-  mSendingHeight(0),
-  mReceivingWidth(0),
-  mReceivingHeight(0),
-  mSendingFramerate(DEFAULT_VIDEO_MAX_FRAMERATE),
-  mLastFramerateTenths(DEFAULT_VIDEO_MAX_FRAMERATE*10),
-  mNumReceivingStreams(1),
-  mVideoLatencyTestEnable(false),
-  mVideoLatencyAvg(0),
-  mMinBitrate(0),
-  mStartBitrate(0),
-  mMaxBitrate(0),
-  mMinBitrateEstimate(0),
-  mRtpStreamIdEnabled(false),
-  mRtpStreamIdExtId(0),
-  mCodecMode(webrtc::kRealtimeVideo)
-{}
+WebrtcVideoConduit::WebrtcVideoConduit(RefPtr<WebRtcCallWrapper> aCall)
+  : mTransportMonitor("WebrtcVideoConduit")
+  , mRenderer(nullptr)
+  , mEngineTransmitting(false)
+  , mEngineReceiving(false)
+  , mCapId(-1)
+  , mCodecMutex("VideoConduit codec db")
+  , mInReconfig(false)
+  , mLastWidth(0)
+  , mLastHeight(0) // initializing as 0 forces a check for reconfig at start
+  , mSendingWidth(0)
+  , mSendingHeight(0)
+  , mReceivingWidth(0)
+  , mReceivingHeight(0)
+  , mSendingFramerate(DEFAULT_VIDEO_MAX_FRAMERATE)
+  , mLastFramerateTenths(DEFAULT_VIDEO_MAX_FRAMERATE * 10)
+  , mNumReceivingStreams(1)
+  , mVideoLatencyTestEnable(false)
+  , mVideoLatencyAvg(0)
+  , mMinBitrate(0)
+  , mStartBitrate(0)
+  , mPrefMaxBitrate(0)
+  , mNegotiatedMaxBitrate(0)
+  , mMinBitrateEstimate(0)
+  , mCodecMode(webrtc::kRealtimeVideo)
+  , mCall(aCall) // refcounted store of the call object
+  , mSendStream(nullptr)
+  , mSendStreamConfig(this) // 'this' is stored but not  dereferenced in the constructor.
+  , mRecvStream(nullptr)
+  , mRecvStreamConfig(this) // 'this' is stored but not  dereferenced in the constructor.
+  , mSendCodecPlugin(nullptr)
+  , mRecvCodecPlugin(nullptr)
+  , mVideoStatsTimer(do_CreateInstance(NS_TIMER_CONTRACTID))
+{
+  mRecvStreamConfig.renderer = this;
+
+  // Video Stats Callback
+  nsTimerCallbackFunc callback = [](nsITimer* aTimer, void* aClosure) {
+    CSFLogDebug(logTag, "StreamStats polling scheduled for VideoConduit: %p", aClosure);
+    auto self = static_cast<WebrtcVideoConduit*>(aClosure);
+    MutexAutoLock lock(self->mCodecMutex);
+    if (self->mEngineTransmitting && self->mSendStream) {
+      self->mSendStreamStats.Update(self->mSendStream->GetStats());
+    }
+    if (self->mEngineReceiving && self->mRecvStream) {
+      self->mRecvStreamStats.Update(self->mRecvStream->GetStats());
+    }
+  };
+  mVideoStatsTimer->InitWithFuncCallback(
+    callback, this, 1000, nsITimer::TYPE_REPEATING_PRECISE_CAN_SKIP);
+}
 
 WebrtcVideoConduit::~WebrtcVideoConduit()
 {
+  CSFLogDebug(logTag, "%s ", __FUNCTION__);
   NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  if (mVideoStatsTimer) {
+    CSFLogDebug(logTag, "canceling StreamStats for VideoConduit: %p", this);
+    MutexAutoLock lock(mCodecMutex);
+    CSFLogDebug(logTag, "StreamStats cancelled for VideoConduit: %p", this);
+    mVideoStatsTimer->Cancel();
+  }
 
-  // Release AudioConduit first by dropping reference on MainThread, where it expects to be
-  SyncTo(nullptr);
   MOZ_ASSERT(!mSendStream && !mRecvStream, "Call DeleteStreams prior to ~WebrtcVideoConduit.");
 }
 
-bool WebrtcVideoConduit::SetLocalSSRC(unsigned int ssrc)
+void
+WebrtcVideoConduit::AddLocalRTPExtensions(
+  const std::vector<webrtc::RtpExtension> & aExtensions)
 {
-  unsigned int oldSsrc;
-  if (!GetLocalSSRC(&oldSsrc)) {
-    MOZ_ASSERT(false, "GetLocalSSRC failed");
-    return false;
-  }
+  auto& extList = mSendStreamConfig.rtp.extensions;
+  std::remove_if(extList.begin(), extList.end(), [&](const webrtc::RtpExtension & i) {
+    return std::find(aExtensions.begin(), aExtensions.end(),i) != aExtensions.end();
+  });
+  extList.insert(extList.end(), aExtensions.begin(), aExtensions.end());
+}
 
-  if (oldSsrc == ssrc) {
+std::vector<webrtc::RtpExtension>
+WebrtcVideoConduit::GetLocalRTPExtensions() const
+{
+  return mSendStreamConfig.rtp.extensions;
+}
+
+bool WebrtcVideoConduit::SetLocalSSRCs(const std::vector<unsigned int> & aSSRCs)
+{
+  // Special case: the local SSRCs are the same - do nothing.
+  if (mSendStreamConfig.rtp.ssrcs == aSSRCs) {
     return true;
   }
 
+  // Update the value of the ssrcs in the config structure.
+  mSendStreamConfig.rtp.ssrcs = aSSRCs;
+
   bool wasTransmitting = mEngineTransmitting;
   if (StopTransmitting() != kMediaConduitNoError) {
     return false;
   }
 
-  if (mPtrRTP->SetLocalSSRC(mChannel, ssrc)) {
-    return false;
-  }
-
   if (wasTransmitting) {
+    DeleteSendStream();
     if (StartTransmitting() != kMediaConduitNoError) {
       return false;
     }
   }
+
   return true;
 }
 
-bool WebrtcVideoConduit::GetLocalSSRC(unsigned int* ssrc)
+std::vector<unsigned int>
+WebrtcVideoConduit::GetLocalSSRCs() const
 {
-  return !mPtrRTP->GetLocalSSRC(mChannel, *ssrc);
+  return mSendStreamConfig.rtp.ssrcs;
 }
 
-bool WebrtcVideoConduit::GetRemoteSSRC(unsigned int* ssrc)
+bool
+WebrtcVideoConduit::SetLocalCNAME(const char* cname)
 {
-  return !mPtrRTP->GetRemoteSSRC(mChannel, *ssrc);
+  mSendStreamConfig.rtp.c_name = cname;
+  return true;
 }
 
-bool WebrtcVideoConduit::SetLocalCNAME(const char* cname)
+MediaConduitErrorCode
+WebrtcVideoConduit::ConfigureCodecMode(webrtc::VideoCodecMode mode)
 {
-  char temp[256];
-  strncpy(temp, cname, sizeof(temp) - 1);
-  temp[sizeof(temp) - 1] = 0;
-  return !mPtrRTP->SetRTCPCName(mChannel, temp);
+  CSFLogVerbose(logTag, "%s ", __FUNCTION__);
+  if (mode == webrtc::VideoCodecMode::kRealtimeVideo ||
+      mode == webrtc::VideoCodecMode::kScreensharing) {
+    mCodecMode = mode;
+    return kMediaConduitNoError;
+  }
+
+  return kMediaConduitMalformedArgument;
 }
 
-bool WebrtcVideoConduit::GetVideoEncoderStats(double* framerateMean,
-                                              double* framerateStdDev,
-                                              double* bitrateMean,
-                                              double* bitrateStdDev,
-                                              uint32_t* droppedFrames)
+webrtc::VideoEncoder::EncoderType
+PayloadNameToEncoderType(const std::string& name)
 {
-  if (!mEngineTransmitting) {
+  if ("VP8" == name) {
+    return webrtc::VideoEncoder::EncoderType::kVp8;
+  } else if ("VP9" == name) {
+    return webrtc::VideoEncoder::EncoderType::kVp9;
+  } else if ("H264" == name) {
+    return webrtc::VideoEncoder::EncoderType::kH264;
+  }
+
+  return webrtc::VideoEncoder::EncoderType::kUnsupportedCodec;
+}
+
+void
+WebrtcVideoConduit::DeleteSendStream()
+{
+  if (mSendStream) {
+
+    if (mLoadManager && mSendStream->LoadStateObserver()) {
+      mLoadManager->RemoveObserver(mSendStream->LoadStateObserver());
+    }
+
+    mCall->Call()->DestroyVideoSendStream(mSendStream);
+    mSendStream = nullptr;
+    mEncoder = nullptr;
+  }
+}
+
+MediaConduitErrorCode
+WebrtcVideoConduit::CreateSendStream()
+{
+  webrtc::VideoEncoder::EncoderType encoder_type =
+    PayloadNameToEncoderType(mSendStreamConfig.encoder_settings.payload_name);
+  if (encoder_type == webrtc::VideoEncoder::EncoderType::kUnsupportedCodec) {
+    return kMediaConduitInvalidSendCodec;
+  }
+
+  nsAutoPtr<webrtc::VideoEncoder> encoder(
+    CreateEncoder(encoder_type, mEncoderConfig.StreamCount() > 0));
+  if (!encoder) {
+    return kMediaConduitInvalidSendCodec;
+  }
+
+  mSendStreamConfig.encoder_settings.encoder = encoder.get();
+
+  MOZ_ASSERT(mSendStreamConfig.rtp.ssrcs.size() == mEncoderConfig.StreamCount(),
+             "Each video substream must have a corresponding ssrc.");
+
+  auto cfg = mEncoderConfig.GenerateConfig();
+  if (cfg.streams.empty()) {
+    MOZ_CRASH("mEncoderConfig.GenerateConfig().streams.empty() == true, there are no configured streams!");
+  }
+
+  mSendStream = mCall->Call()->CreateVideoSendStream(mSendStreamConfig, cfg);
+
+  if (!mSendStream) {
+    return kMediaConduitVideoSendStreamError;
+  }
+
+  mEncoder = encoder;
+
+  if (mLoadManager && mSendStream->LoadStateObserver()) {
+    mLoadManager->AddObserver(mSendStream->LoadStateObserver());
+  }
+
+  return kMediaConduitNoError;
+}
+
+webrtc::VideoDecoder::DecoderType
+PayloadNameToDecoderType(const std::string& name)
+{
+  if ("VP8" == name) {
+    return webrtc::VideoDecoder::DecoderType::kVp8;
+  } else if ("VP9" == name) {
+    return webrtc::VideoDecoder::DecoderType::kVp9;
+  } else if ("H264" == name) {
+    return webrtc::VideoDecoder::DecoderType::kH264;
+  }
+
+  return webrtc::VideoDecoder::DecoderType::kUnsupportedCodec;
+}
+
+void
+WebrtcVideoConduit::DeleteRecvStream()
+{
+  if (mRecvStream) {
+    mCall->Call()->DestroyVideoReceiveStream(mRecvStream);
+    mRecvStream = nullptr;
+    mDecoders.clear();
+  }
+}
+
+MediaConduitErrorCode
+WebrtcVideoConduit::CreateRecvStream()
+{
+  webrtc::VideoReceiveStream::Decoder decoder_desc;
+  std::unique_ptr<webrtc::VideoDecoder> decoder;
+  webrtc::VideoDecoder::DecoderType decoder_type;
+
+  mRecvStreamConfig.decoders.clear();
+  for (auto& config : mRecvCodecList) {
+    decoder_type = PayloadNameToDecoderType(config->mName);
+    if (decoder_type == webrtc::VideoDecoder::DecoderType::kUnsupportedCodec) {
+      CSFLogError(logTag, "%s Unknown decoder type: %s", __FUNCTION__,
+                  config->mName.c_str());
+      continue;
+    }
+
+    decoder.reset(CreateDecoder(decoder_type));
+
+    if (!decoder) {
+      // This really should never happen unless something went wrong
+      // in the negotiation code
+      NS_ASSERTION(decoder, "Failed to create video decoder");
+      CSFLogError(logTag, "Failed to create decoder of type %s (%d)",
+                  config->mName.c_str(), decoder_type);
+      // don't stop
+      continue;
+    }
+
+    decoder_desc.decoder = decoder.get();
+    mDecoders.push_back(std::move(decoder));
+    decoder_desc.payload_name = config->mName;
+    decoder_desc.payload_type = config->mType;
+    mRecvStreamConfig.decoders.push_back(decoder_desc);
+  }
+
+  mRecvStream = mCall->Call()->CreateVideoReceiveStream(mRecvStreamConfig);
+
+  if (!mRecvStream) {
+    mDecoders.clear();
+    return kMediaConduitUnknownError;
+  }
+
+  return kMediaConduitNoError;
+}
+
+static bool CompatibleH264Config(const webrtc::VideoCodecH264 &aEncoderSpecificH264,
+                                 const VideoCodecConfig* aCodecConfig)
+{
+  if (aEncoderSpecificH264.profile_byte != aCodecConfig->mProfile ||
+      aEncoderSpecificH264.constraints != aCodecConfig->mConstraints ||
+      aEncoderSpecificH264.packetizationMode != aCodecConfig->mPacketizationMode) {
     return false;
   }
-  MOZ_ASSERT(mVideoCodecStat);
-  mVideoCodecStat->GetEncoderStats(framerateMean, framerateStdDev,
-                                   bitrateMean, bitrateStdDev,
-                                   droppedFrames);
+  return true;
+}
 
-  // See if we need to adjust bandwidth.
-  // Avoid changing bandwidth constantly; use hysteresis.
+/**
+ * Note: Setting the send-codec on the Video Engine will restart the encoder,
+ * sets up new SSRC and reset RTP_RTCP module with the new codec setting.
+ *
+ * Note: this is called from MainThread, and the codec settings are read on
+ * videoframe delivery threads (i.e in SendVideoFrame().  With
+ * renegotiation/reconfiguration, this now needs a lock!  Alternatively
+ * changes could be queued until the next frame is delivered using an
+ * Atomic pointer and swaps.
+ */
 
-  // Note: mLastFramerate is a relaxed Atomic because we're setting it here, and
-  // reading it on whatever thread calls DeliverFrame/SendVideoFrame.  Alternately
-  // we could use a lock.  Note that we don't change it often, and read it once per frame.
-  // We scale by *10 because mozilla::Atomic<> doesn't do 'double' or 'float'.
-  double framerate = mLastFramerateTenths/10.0; // fetch once
-  if (std::abs(*framerateMean - framerate)/framerate > 0.1 &&
-      *framerateMean >= 0.5) {
-    // unchanged resolution, but adjust bandwidth limits to match camera fps
-    CSFLogDebug(logTag, "Encoder frame rate changed from %f to %f",
-                (mLastFramerateTenths/10.0), *framerateMean);
+MediaConduitErrorCode
+WebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig)
+{
+  CSFLogDebug(logTag, "%s for %s", __FUNCTION__,
+    codecConfig ? codecConfig->mName.c_str() : "<null>");
+
+  MediaConduitErrorCode condError = kMediaConduitNoError;
+
+  // validate basic params
+  if ((condError = ValidateCodecConfig(codecConfig, true)) != kMediaConduitNoError) {
+    return condError;
+  }
+
+  // StopTransmitting may be moot if mSendStream is null, but the code seems to
+  // allow for it.
+  // Recreating on PayloadType change may be overkill, but is safe.
+  if (!mSendStream ||
+      mSendStreamConfig.encoder_settings.payload_type != codecConfig->mType ||
+      mSendStreamConfig.encoder_settings.payload_name != codecConfig->mName ||
+      (codecConfig->mName == "H264" &&
+       !CompatibleH264Config(mEncoderSpecificH264, codecConfig))) {
+    condError = StopTransmitting();
+    if (condError != kMediaConduitNoError) {
+      return condError;
+    }
+    DeleteSendStream(); // safe if mSendStream is null
+  } // we are already using this codec - mSendStream tells us we're reconfiguring
+
+  mSendStreamConfig.encoder_settings.payload_name = codecConfig->mName;
+  mSendStreamConfig.encoder_settings.payload_type = codecConfig->mType;
+  mSendStreamConfig.rtp.rtcp_mode = webrtc::RtcpMode::kCompound;
+  mSendStreamConfig.rtp.max_packet_size = kVideoMtu;
+  mSendStreamConfig.overuse_callback = mLoadManager.get();
+
+  size_t streamCount = std::min(codecConfig->mSimulcastEncodings.size(),
+                                (size_t)webrtc::kMaxSimulcastStreams);
+  CSFLogDebug(logTag, "%s for VideoConduit:%p stream count:%d", __FUNCTION__,
+              this, static_cast<int>(streamCount));
+
+  mSendingFramerate = 0;
+  mEncoderConfig.ClearStreams();
+
+  unsigned short width = 320;
+  unsigned short height = 240;
+  int max_framerate;
+  if (codecConfig->mEncodingConstraints.maxFps > 0) {
+    max_framerate = codecConfig->mEncodingConstraints.maxFps;
+  } else {
+    max_framerate = DEFAULT_VIDEO_MAX_FRAMERATE;
+  }
+  // apply restrictions from maxMbps/etc
+  mSendingFramerate = SelectSendFrameRate(codecConfig,
+                                          max_framerate,
+                                          mSendingWidth,
+                                          mSendingHeight);
+
+  // So we can comply with b=TIAS/b=AS/maxbr=X when input resolution changes
+  mNegotiatedMaxBitrate = codecConfig->mTias / 1000;
+
+  // width/height will be overridden on the first frame; they must be 'sane' for
+  // SetSendCodec()
+
+  if (mSendingWidth != 0) {
+    // We're already in a call and are reconfiguring (perhaps due to
+    // ReplaceTrack).
+    bool resolutionChanged;
+    {
+      MutexAutoLock lock(mCodecMutex);
+      resolutionChanged = !mCurSendCodecConfig->ResolutionEquals(*codecConfig);
+    }
+
+    if (resolutionChanged) {
+      // We're already in a call and due to renegotiation an encoder parameter
+      // that requires reconfiguration has changed. Resetting these members
+      // triggers reconfig on the next frame.
+      mLastWidth = 0;
+      mLastHeight = 0;
+      mSendingWidth = 0;
+      mSendingHeight = 0;
+    } else {
+      // We're already in a call but changes don't require a reconfiguration.
+      // We update the resolutions in the send codec to match the current
+      // settings.  Framerate is already set.
+      width = mSendingWidth;
+      height = mSendingHeight;
+      // Bitrates are set in the loop below
+    }
+  }
+
+  for (size_t idx = streamCount - 1; streamCount > 0; idx--, streamCount--) {
+    webrtc::VideoStream video_stream;
+    VideoEncoderConfigBuilder::SimulcastStreamConfig simulcast_config;
+    // Stream dimensions must be divisable by 2^(n-1), where n is the number of layers.
+    // Each lower resolution layer is 1/2^(n-1) of the size of largest layer,
+    // where n is the number of the layer
+
+    // width/height will be overridden on the first frame; they must be 'sane' for
+    // SetSendCodec()
+    video_stream.width = width >> idx;
+    video_stream.height = height >> idx;
+    video_stream.max_framerate = mSendingFramerate;
+    auto& simulcastEncoding = codecConfig->mSimulcastEncodings[idx];
+    // leave vector temporal_layer_thresholds_bps empty
+    video_stream.temporal_layer_thresholds_bps.clear();
+    // Calculate these first
+    video_stream.max_bitrate_bps = MinIgnoreZero(simulcastEncoding.constraints.maxBr,
+                                                 kDefaultMaxBitrate_bps);
+    video_stream.max_bitrate_bps = MinIgnoreZero((int) mPrefMaxBitrate*1000,
+                                                 video_stream.max_bitrate_bps);
+    video_stream.min_bitrate_bps = (mMinBitrate ? mMinBitrate : kDefaultMinBitrate_bps);
+    if (video_stream.min_bitrate_bps > video_stream.max_bitrate_bps) {
+      video_stream.min_bitrate_bps = video_stream.max_bitrate_bps;
+    }
+    video_stream.target_bitrate_bps = (mStartBitrate ? mStartBitrate : kDefaultStartBitrate_bps);
+    if (video_stream.target_bitrate_bps > video_stream.max_bitrate_bps) {
+      video_stream.target_bitrate_bps = video_stream.max_bitrate_bps;
+    }
+    if (video_stream.target_bitrate_bps < video_stream.min_bitrate_bps) {
+      video_stream.target_bitrate_bps = video_stream.min_bitrate_bps;
+    }
+    // We should use SelectBitrates here for the case of already-sending and no reconfig needed;
+    // overrides the calculations above
+    if (mSendingWidth) { // cleared if we need a reconfig
+      SelectBitrates(video_stream.width, video_stream.height,
+                     simulcastEncoding.constraints.maxBr,
+                     mLastFramerateTenths, video_stream);
+    }
+
+    video_stream.max_qp = kQpMax;
+    video_stream.SetRid(simulcastEncoding.rid);
+    simulcast_config.jsScaleDownBy = simulcastEncoding.constraints.scaleDownBy;
+    simulcast_config.jsMaxBitrate = simulcastEncoding.constraints.maxBr; // bps
+
+    if (codecConfig->mName == "H264") {
+      if (codecConfig->mEncodingConstraints.maxMbps > 0) {
+        // Not supported yet!
+        CSFLogError(logTag, "%s H.264 max_mbps not supported yet", __FUNCTION__);
+      }
+    }
+    mEncoderConfig.AddStream(video_stream, simulcast_config);
+  }
+
+  if (codecConfig->mName == "H264") {
+#ifdef MOZ_WEBRTC_OMX
+    mEncoderConfig.SetResolutionDivisor(16);
+#else
+    mEncoderConfig.SetResolutionDivisor(1);
+#endif
+    mEncoderSpecificH264 = webrtc::VideoEncoder::GetDefaultH264Settings();
+    mEncoderSpecificH264.profile_byte = codecConfig->mProfile;
+    mEncoderSpecificH264.constraints = codecConfig->mConstraints;
+    mEncoderSpecificH264.level = codecConfig->mLevel;
+    mEncoderSpecificH264.packetizationMode = codecConfig->mPacketizationMode;
+    mEncoderSpecificH264.scaleDownBy = codecConfig->mEncodingConstraints.scaleDownBy;
+
+    // XXX parse the encoded SPS/PPS data
+    // paranoia
+    mEncoderSpecificH264.spsData = nullptr;
+    mEncoderSpecificH264.spsLen = 0;
+    mEncoderSpecificH264.ppsData = nullptr;
+    mEncoderSpecificH264.ppsLen = 0;
+
+    mEncoderConfig.SetEncoderSpecificSettings(&mEncoderSpecificH264);
+  } else {
+    mEncoderConfig.SetEncoderSpecificSettings(nullptr);
+    mEncoderConfig.SetResolutionDivisor(1);
+  }
+
+  mEncoderConfig.SetContentType(mCodecMode == webrtc::kRealtimeVideo ?
+    webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo :
+    webrtc::VideoEncoderConfig::ContentType::kScreen);
+  // for the GMP H.264 encoder/decoder!!
+  mEncoderConfig.SetMinTransmitBitrateBps(0);
+
+  // See Bug 1297058, enabling FEC when basic NACK is to be enabled in H.264 is problematic
+  if (codecConfig->RtcpFbFECIsSet() &&
+      !(codecConfig->mName == "H264" && codecConfig->RtcpFbNackIsSet(""))) {
+    mSendStreamConfig.rtp.fec.ulpfec_payload_type = kUlpFecPayloadType;
+    mSendStreamConfig.rtp.fec.red_payload_type = kRedPayloadType;
+    mSendStreamConfig.rtp.fec.red_rtx_payload_type = kNullPayloadType;
+  }
+
+  mSendStreamConfig.rtp.nack.rtp_history_ms =
+    codecConfig->RtcpFbNackIsSet("") ? 1000 : 0;
+
+  {
     MutexAutoLock lock(mCodecMutex);
-    mLastFramerateTenths = *framerateMean * 10;
-    SelectSendResolution(mSendingWidth, mSendingHeight, nullptr);
+    // Copy the applied config for future reference.
+    mCurSendCodecConfig = new VideoCodecConfig(*codecConfig);
   }
-  return true;
+
+  // Is this a reconfigure of a running codec?
+  if (mSendStream &&
+      !mSendStream->ReconfigureVideoEncoder(mEncoderConfig.GenerateConfig())) {
+    CSFLogError(logTag, "%s: ReconfigureVideoEncoder failed", __FUNCTION__);
+    // This will cause a new encoder to be created by StartTransmitting()
+    condError = StopTransmitting();
+    if (condError != kMediaConduitNoError) {
+      return condError;
+    }
+    DeleteSendStream();
+  }
+
+  return condError;
 }
 
-bool WebrtcVideoConduit::GetVideoDecoderStats(double* framerateMean,
-                                              double* framerateStdDev,
-                                              double* bitrateMean,
-                                              double* bitrateStdDev,
-                                              uint32_t* discardedPackets)
+bool
+WebrtcVideoConduit::SetRemoteSSRC(unsigned int ssrc)
 {
-  if (!mEngineReceiving) {
+  mRecvStreamConfig.rtp.remote_ssrc = ssrc;
+  unsigned int current_ssrc;
+
+  if (!GetRemoteSSRC(&current_ssrc)) {
     return false;
   }
-  MOZ_ASSERT(mVideoCodecStat);
-  mVideoCodecStat->GetDecoderStats(framerateMean, framerateStdDev,
-                                   bitrateMean, bitrateStdDev,
-                                   discardedPackets);
+
+  if (current_ssrc == ssrc || !mEngineReceiving) {
+    return true;
+  }
+
+  if (StopReceiving() != kMediaConduitNoError) {
+    return false;
+  }
+
+  DeleteRecvStream();
+  MediaConduitErrorCode rval = CreateRecvStream();
+  if (rval != kMediaConduitNoError) {
+    CSFLogError(logTag, "%s Start Receive Error %d ", __FUNCTION__, rval);
+    return false;
+  }
+
+  return (StartReceiving() == kMediaConduitNoError);
+}
+
+bool
+WebrtcVideoConduit::GetRemoteSSRC(unsigned int* ssrc)
+{
+  {
+    MutexAutoLock lock(mCodecMutex);
+    if (!mRecvStream) {
+      return false;
+    }
+
+    const webrtc::VideoReceiveStream::Stats& stats = mRecvStream->GetStats();
+    *ssrc = stats.ssrc;
+  }
+
   return true;
 }
 
-bool WebrtcVideoConduit::GetAVStats(int32_t* jitterBufferDelayMs,
-                                    int32_t* playoutBufferDelayMs,
-                                    int32_t* avSyncOffsetMs) {
+bool
+WebrtcVideoConduit::GetVideoEncoderStats(double* framerateMean,
+                                         double* framerateStdDev,
+                                         double* bitrateMean,
+                                         double* bitrateStdDev,
+                                         uint32_t* droppedFrames)
+{
+  {
+    MutexAutoLock lock(mCodecMutex);
+    if (!mEngineTransmitting || !mSendStream) {
+      return false;
+    }
+    mSendStreamStats.GetVideoStreamStats(*framerateMean, *framerateStdDev,
+      *bitrateMean, *bitrateStdDev);
+    mSendStreamStats.DroppedFrames(*droppedFrames);
+    return true;
+  }
+}
+
+bool
+WebrtcVideoConduit::GetVideoDecoderStats(double* framerateMean,
+                                         double* framerateStdDev,
+                                         double* bitrateMean,
+                                         double* bitrateStdDev,
+                                         uint32_t* discardedPackets)
+{
+  {
+    MutexAutoLock lock(mCodecMutex);
+    if (!mEngineReceiving || !mRecvStream) {
+      return false;
+    }
+    mRecvStreamStats.GetVideoStreamStats(*framerateMean, *framerateStdDev,
+      *bitrateMean, *bitrateStdDev);
+    mRecvStreamStats.DiscardedPackets(*discardedPackets);
+    return true;
+  }
+}
+
+bool
+WebrtcVideoConduit::GetAVStats(int32_t* jitterBufferDelayMs,
+                               int32_t* playoutBufferDelayMs,
+                               int32_t* avSyncOffsetMs)
+{
   return false;
 }
 
-bool WebrtcVideoConduit::GetRTPStats(unsigned int* jitterMs,
-                                     unsigned int* cumulativeLost) {
-  unsigned short fractionLost;
-  unsigned extendedMax;
-  int64_t rttMs;
-  // GetReceivedRTCPStatistics is a poorly named GetRTPStatistics variant
-  return !mPtrRTP->GetReceivedRTCPStatistics(mChannel, fractionLost,
-                                             *cumulativeLost,
-                                             extendedMax,
-                                             *jitterMs,
-                                             rttMs);
+bool
++WebrtcVideoConduit::GetRTPStats(unsigned int* jitterMs,
+                                unsigned int* cumulativeLost)
+{
+  CSFLogVerbose(logTag, "%s for VideoConduit:%p", __FUNCTION__, this);
+  {
+    MutexAutoLock lock(mCodecMutex);
+    if (!mRecvStream) {
+      return false;
+    }
+
+    const webrtc::VideoReceiveStream::Stats& stats = mRecvStream->GetStats();
+    *jitterMs = stats.rtcp_stats.jitter;
+    *cumulativeLost = stats.rtcp_stats.cumulative_lost;
+  }
+  return true;
 }
 
 bool WebrtcVideoConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
@@ -229,34 +780,56 @@ bool WebrtcVideoConduit::GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
                                                uint32_t* packetsReceived,
                                                uint64_t* bytesReceived,
                                                uint32_t* cumulativeLost,
-                                               int32_t* rttMs) {
-  uint32_t ntpHigh, ntpLow;
-  uint16_t fractionLost;
-  bool result = !mPtrRTP->GetRemoteRTCPReceiverInfo(mChannel, ntpHigh, ntpLow,
-                                                    *packetsReceived,
-                                                    *bytesReceived,
-                                                    jitterMs,
-                                                    &fractionLost,
-                                                    cumulativeLost,
-                                                    rttMs);
-  if (result) {
-    *timestamp = NTPtoDOMHighResTimeStamp(ntpHigh, ntpLow);
+                                               int32_t* rttMs)
+{
+  {
+    CSFLogVerbose(logTag, "%s for VideoConduit:%p", __FUNCTION__, this);
+    MutexAutoLock lock(mCodecMutex);
+    if (!mRecvStream) {
+      return false;
+    }
+
+    const webrtc::VideoReceiveStream::Stats &stats = mRecvStream->GetStats();
+    *jitterMs = stats.rtcp_stats.jitter;
+    *cumulativeLost = stats.rtcp_stats.cumulative_lost;
+    *bytesReceived = stats.rtp_stats.MediaPayloadBytes();
+    *packetsReceived = stats.rtp_stats.transmitted.packets;
+    // Note: timestamp is not correct per the spec... should be time the rtcp
+    // was received (remote) or sent (local)
+    *timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
+    int64_t rtt = mRecvStream->GetRtt();
+    if (rtt >= 0) {
+      *rttMs = rtt;
+    }
   }
-  return result;
+  return true;
 }
 
-bool WebrtcVideoConduit::GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
-                                             unsigned int* packetsSent,
-                                             uint64_t* bytesSent) {
-  struct webrtc::SenderInfo senderInfo;
-  bool result = !mPtrRTP->GetRemoteRTCPSenderInfo(mChannel, &senderInfo);
-  if (result) {
-    *timestamp = NTPtoDOMHighResTimeStamp(senderInfo.NTP_timestamp_high,
-                                          senderInfo.NTP_timestamp_low);
-    *packetsSent = senderInfo.sender_packet_count;
-    *bytesSent = senderInfo.sender_octet_count;
+bool
+WebrtcVideoConduit::GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
+                                        unsigned int* packetsSent,
+                                        uint64_t* bytesSent)
+{
+  {
+    CSFLogVerbose(logTag, "%s for VideoConduit:%p", __FUNCTION__, this);
+    MutexAutoLock lock(mCodecMutex);
+    if (!mSendStream) {
+      return false;
+    }
+
+    const webrtc::VideoSendStream::Stats& stats = mSendStream->GetStats();
+    *packetsSent = 0;
+    for (auto entry: stats.substreams){
+      *packetsSent += entry.second.rtp_stats.transmitted.packets;
+      // NG -- per https://www.w3.org/TR/webrtc-stats/ this is only payload bytes
+      *bytesSent += entry.second.rtp_stats.MediaPayloadBytes();
+    }
+    // Note: timestamp is not correct per the spec... should be time the rtcp
+    // was received (remote) or sent (local)
+    *timestamp = webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds();
+    return true;
   }
-  return result;
+  return false;
 }
 
 MediaConduitErrorCode
@@ -268,49 +841,55 @@ WebrtcVideoConduit::InitMain()
 
   nsresult rv;
   nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
-  if (!NS_WARN_IF(NS_FAILED(rv)))
-  {
+  if (!NS_WARN_IF(NS_FAILED(rv))) {
     nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
 
-    if (branch)
-    {
+    if (branch) {
       int32_t temp;
-      Unused << NS_WARN_IF(NS_FAILED(branch->GetBoolPref("media.video.test_latency", &mVideoLatencyTestEnable)));
-      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.min_bitrate", &temp))))
+      Unused <<  NS_WARN_IF(NS_FAILED(branch->GetBoolPref("media.video.test_latency",
+                                                          &mVideoLatencyTestEnable)));
+      Unused << NS_WARN_IF(NS_FAILED(branch->GetBoolPref("media.video.test_latency",
+                                                         &mVideoLatencyTestEnable)));
+      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref(
+            "media.peerconnection.video.min_bitrate", &temp))))
       {
          if (temp >= 0) {
             mMinBitrate = temp;
          }
       }
-      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.start_bitrate", &temp))))
+      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref(
+            "media.peerconnection.video.start_bitrate", &temp))))
       {
          if (temp >= 0) {
          mStartBitrate = temp;
          }
       }
-      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.max_bitrate", &temp))))
+      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref(
+            "media.peerconnection.video.max_bitrate", &temp))))
       {
         if (temp >= 0) {
-          mMaxBitrate = temp;
+          mPrefMaxBitrate = temp;
         }
       }
-      if (mMinBitrate != 0 && mMinBitrate < webrtc::kViEMinCodecBitrate) {
-        mMinBitrate = webrtc::kViEMinCodecBitrate;
+      if (mMinBitrate != 0 && mMinBitrate < kViEMinCodecBitrate) {
+        mMinBitrate = kViEMinCodecBitrate;
       }
       if (mStartBitrate < mMinBitrate) {
         mStartBitrate = mMinBitrate;
       }
-      if (mStartBitrate > mMaxBitrate) {
-        mStartBitrate = mMaxBitrate;
+      if (mPrefMaxBitrate && mStartBitrate > mPrefMaxBitrate) {
+        mStartBitrate = mPrefMaxBitrate;
       }
-      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.min_bitrate_estimate", &temp))))
+      if (!NS_WARN_IF(NS_FAILED(branch->GetIntPref(
+            "media.peerconnection.video.min_bitrate_estimate", &temp))))
       {
         if (temp >= 0) {
           mMinBitrateEstimate = temp;
         }
       }
       bool use_loadmanager = false;
-      if (!NS_WARN_IF(NS_FAILED(branch->GetBoolPref("media.navigator.load_adapt", &use_loadmanager))))
+      if (!NS_WARN_IF(NS_FAILED(branch->GetBoolPref(
+            "media.navigator.load_adapt", &use_loadmanager))))
       {
         if (use_loadmanager) {
           mLoadManager = LoadManagerBuild();
@@ -329,7 +908,7 @@ WebrtcVideoConduit::InitMain()
 MediaConduitErrorCode
 WebrtcVideoConduit::Init()
 {
-  CSFLogDebug(logTag,  "%s this=%p", __FUNCTION__, this);
+  CSFLogDebug(logTag, "%s this=%p", __FUNCTION__, this);
   MediaConduitErrorCode result;
   // Run code that must run on MainThread first
   MOZ_ASSERT(NS_IsMainThread());
@@ -338,128 +917,6 @@ WebrtcVideoConduit::Init()
     return result;
   }
 
-  // Per WebRTC APIs below function calls return nullptr on failure
-  mVideoEngine = webrtc::VideoEngine::Create();
-  if(!mVideoEngine)
-  {
-    CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  if( !(mPtrViEBase = ViEBase::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get video base interface ", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  if( !(mPtrViECapture = ViECapture::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get video capture interface", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  if( !(mPtrViECodec = ViECodec::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get video codec interface ", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  if( !(mPtrViENetwork = ViENetwork::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get video network interface ", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  if( !(mPtrViERender = ViERender::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get video render interface ", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  mPtrExtCodec = webrtc::ViEExternalCodec::GetInterface(mVideoEngine);
-  if (!mPtrExtCodec) {
-    CSFLogError(logTag, "%s Unable to get external codec interface: %d ",
-                __FUNCTION__,mPtrViEBase->LastError());
-    return kMediaConduitSessionNotInited;
-  }
-
-  if( !(mPtrRTP = webrtc::ViERTP_RTCP::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get video RTCP interface ", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
-  if ( !(mPtrExtCodec = webrtc::ViEExternalCodec::GetInterface(mVideoEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to get external codec interface %d ",
-                __FUNCTION__, mPtrViEBase->LastError());
-    return kMediaConduitSessionNotInited;
-  }
-
-  CSFLogDebug(logTag, "%s Engine Created: Init'ng the interfaces ",__FUNCTION__);
-
-  if(mPtrViEBase->Init() == -1)
-  {
-    CSFLogError(logTag, " %s Video Engine Init Failed %d ",__FUNCTION__,
-                mPtrViEBase->LastError());
-    return kMediaConduitSessionNotInited;
-  }
-
-  if(mPtrViEBase->CreateChannel(mChannel) == -1)
-  {
-    CSFLogError(logTag, " %s Channel creation Failed %d ",__FUNCTION__,
-                mPtrViEBase->LastError());
-    return kMediaConduitChannelError;
-  }
-
-  if(mPtrViENetwork->RegisterSendTransport(mChannel, *this) == -1)
-  {
-    CSFLogError(logTag,  "%s ViENetwork Failed %d ", __FUNCTION__,
-                mPtrViEBase->LastError());
-    return kMediaConduitTransportRegistrationFail;
-  }
-
-  if(mPtrViECapture->AllocateExternalCaptureDevice(mCapId,
-                                                   mPtrExtCapture) == -1)
-  {
-    CSFLogError(logTag, "%s Unable to Allocate capture module: %d ",
-                __FUNCTION__, mPtrViEBase->LastError());
-    return kMediaConduitCaptureError;
-  }
-
-  if(mPtrViECapture->ConnectCaptureDevice(mCapId,mChannel) == -1)
-  {
-    CSFLogError(logTag, "%s Unable to Connect capture module: %d ",
-                __FUNCTION__,mPtrViEBase->LastError());
-    return kMediaConduitCaptureError;
-  }
-  // Set up some parameters, per juberti. Set MTU.
-  if(mPtrViENetwork->SetMTU(mChannel, 1200) != 0)
-  {
-    CSFLogError(logTag,  "%s MTU Failed %d ", __FUNCTION__,
-                mPtrViEBase->LastError());
-    return kMediaConduitMTUError;
-  }
-  // Turn on RTCP and loss feedback reporting.
-  if(mPtrRTP->SetRTCPStatus(mChannel, webrtc::kRtcpCompound_RFC4585) != 0)
-  {
-    CSFLogError(logTag,  "%s RTCPStatus Failed %d ", __FUNCTION__,
-                mPtrViEBase->LastError());
-    return kMediaConduitRTCPStatusError;
-  }
-
-  if (mPtrViERender->AddRenderer(mChannel,
-                                webrtc::kVideoI420,
-                                (webrtc::ExternalRenderer*) this) == -1) {
-      CSFLogError(logTag, "%s Failed to added external renderer ", __FUNCTION__);
-      return kMediaConduitInvalidRenderer;
-  }
-
-  if (mLoadManager) {
-    mPtrViEBase->RegisterCpuOveruseObserver(mChannel, mLoadManager);
-    mPtrViEBase->SetLoadManager(mLoadManager);
-  }
-
   CSFLogError(logTag, "%s Initialization Done", __FUNCTION__);
   return kMediaConduitNoError;
 }
@@ -467,99 +924,43 @@ WebrtcVideoConduit::Init()
 void
 WebrtcVideoConduit::DeleteStreams()
 {
-  // The first one of a pair to be deleted shuts down media for both
-  //Deal with External Capturer
-  if(mPtrViECapture)
-  {
-    mPtrViECapture->DisconnectCaptureDevice(mCapId);
-    mPtrViECapture->ReleaseCaptureDevice(mCapId);
-    mPtrExtCapture = nullptr;
-  }
-
-   if (mPtrExtCodec) {
-     mPtrExtCodec->Release();
-     mPtrExtCodec = NULL;
-   }
-
-  //Deal with External Renderer
-  if(mPtrViERender)
-  {
-    if(mRenderer) {
-      mPtrViERender->StopRender(mChannel);
-    }
-    mPtrViERender->RemoveRenderer(mChannel);
-  }
-
-  //Deal with the transport
-  if(mPtrViENetwork)
-  {
-    mPtrViENetwork->DeregisterSendTransport(mChannel);
-  }
-
-  if(mPtrViEBase)
-  {
-    mPtrViEBase->StopSend(mChannel);
-    mPtrViEBase->StopReceive(mChannel);
-    mPtrViEBase->DeleteChannel(mChannel);
-  }
-
-  // mVideoCodecStat has a back-ptr to mPtrViECodec that must be released first
-  if (mVideoCodecStat) {
-    mVideoCodecStat->EndOfCallStats();
-  }
-  mVideoCodecStat = nullptr;
-  //This does Release AudioConduit before mPtrViEBase set nullptr.
-  SyncTo(nullptr);
   // We can't delete the VideoEngine until all these are released!
   // And we can't use a Scoped ptr, since the order is arbitrary
-  mPtrViEBase = nullptr;
-  mPtrViECapture = nullptr;
-  mPtrViECodec = nullptr;
-  mPtrViENetwork = nullptr;
-  mPtrViERender = nullptr;
-  mPtrRTP = nullptr;
-  mPtrExtCodec = nullptr;
 
-  // only one opener can call Delete.  Have it be the last to close.
-  if(mVideoEngine)
-  {
-    webrtc::VideoEngine::Delete(mVideoEngine);
-  }
+  DeleteSendStream();
+  DeleteRecvStream();
 }
 
 void
-WebrtcVideoConduit::SyncTo(WebrtcAudioConduit *aConduit)
+WebrtcVideoConduit::SyncTo(WebrtcAudioConduit* aConduit)
 {
   CSFLogDebug(logTag, "%s Synced to %p", __FUNCTION__, aConduit);
+  {
+    MutexAutoLock lock(mCodecMutex);
 
-  if (!mPtrViEBase) {
-    // ViEBase has already been released; we no longer have a conduit.
-    mSyncedTo = nullptr;
-    return;
-  }
-  // SyncTo(value) syncs to the AudioConduit, and if already synced replaces
-  // the current sync target.  SyncTo(nullptr) cancels any existing sync and
-  // releases the strong ref to AudioConduit.
-  if (aConduit) {
-    mPtrViEBase->SetVoiceEngine(aConduit->GetVoiceEngine());
-    mPtrViEBase->ConnectAudioChannel(mChannel, aConduit->GetChannel());
-    // NOTE: this means the VideoConduit will keep the AudioConduit alive!
-  } else {
-    mPtrViEBase->DisconnectAudioChannel(mChannel);
-    mPtrViEBase->SetVoiceEngine(nullptr);
+    if (!mRecvStream) {
+      CSFLogError(logTag, "SyncTo called with no receive stream");
+      return;
+    }
+
+    if (aConduit) {
+      mRecvStream->SetSyncChannel(aConduit->GetVoiceEngine(),
+                                  aConduit->GetChannel());
+    } else if (mSyncedTo) {
+      mRecvStream->SetSyncChannel(mSyncedTo->GetVoiceEngine(), -1);
+    }
   }
 
   mSyncedTo = aConduit;
 }
 
 MediaConduitErrorCode
-WebrtcVideoConduit::AttachRenderer(RefPtr<VideoRenderer> aVideoRenderer)
+WebrtcVideoConduit::AttachRenderer(RefPtr<mozilla::VideoRenderer> aVideoRenderer)
 {
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  CSFLogDebug(logTag, "%s", __FUNCTION__);
 
-  //null renderer
-  if(!aVideoRenderer)
-  {
+  // null renderer
+  if (!aVideoRenderer) {
     CSFLogError(logTag, "%s NULL Renderer", __FUNCTION__);
     MOZ_ASSERT(false);
     return kMediaConduitInvalidRenderer;
@@ -567,10 +968,8 @@ WebrtcVideoConduit::AttachRenderer(RefPtr<VideoRenderer> aVideoRenderer)
 
   // This function is called only from main, so we only need to protect against
   // modifying mRenderer while any webrtc.org code is trying to use it.
-  bool wasRendering;
   {
     ReentrantMonitorAutoEnter enter(mTransportMonitor);
-    wasRendering = !!mRenderer;
     mRenderer = aVideoRenderer;
     // Make sure the renderer knows the resolution
     mRenderer->FrameSizeChange(mReceivingWidth,
@@ -578,17 +977,6 @@ WebrtcVideoConduit::AttachRenderer(RefPtr<VideoRenderer> aVideoRenderer)
                                mNumReceivingStreams);
   }
 
-  if (!wasRendering) {
-    if(mPtrViERender->StartRender(mChannel) == -1)
-    {
-      CSFLogError(logTag, "%s Starting the Renderer Failed %d ", __FUNCTION__,
-                                                      mPtrViEBase->LastError());
-      ReentrantMonitorAutoEnter enter(mTransportMonitor);
-      mRenderer = nullptr;
-      return kMediaConduitRendererFail;
-    }
-  }
-
   return kMediaConduitNoError;
 }
 
@@ -597,19 +985,17 @@ WebrtcVideoConduit::DetachRenderer()
 {
   {
     ReentrantMonitorAutoEnter enter(mTransportMonitor);
-    if(mRenderer)
-    {
+    if (mRenderer) {
       mRenderer = nullptr;
     }
   }
-
-  mPtrViERender->StopRender(mChannel);
 }
 
 MediaConduitErrorCode
-WebrtcVideoConduit::SetTransmitterTransport(RefPtr<TransportInterface> aTransport)
+WebrtcVideoConduit::SetTransmitterTransport(
+  RefPtr<TransportInterface> aTransport)
 {
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  CSFLogDebug(logTag, "%s ", __FUNCTION__);
 
   ReentrantMonitorAutoEnter enter(mTransportMonitor);
   // set the transport
@@ -620,489 +1006,272 @@ WebrtcVideoConduit::SetTransmitterTransport(RefPtr<TransportInterface> aTranspor
 MediaConduitErrorCode
 WebrtcVideoConduit::SetReceiverTransport(RefPtr<TransportInterface> aTransport)
 {
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  CSFLogDebug(logTag, "%s ", __FUNCTION__);
 
   ReentrantMonitorAutoEnter enter(mTransportMonitor);
   // set the transport
   mReceiverTransport = aTransport;
   return kMediaConduitNoError;
 }
-MediaConduitErrorCode
-WebrtcVideoConduit::ConfigureCodecMode(webrtc::VideoCodecMode mode)
-{
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
-  mCodecMode = mode;
-  return kMediaConduitNoError;
-}
-/**
- * Note: Setting the send-codec on the Video Engine will restart the encoder,
- * sets up new SSRC and reset RTP_RTCP module with the new codec setting.
- *
- * Note: this is called from MainThread, and the codec settings are read on
- * videoframe delivery threads (i.e in SendVideoFrame().  With
- * renegotiation/reconfiguration, this now needs a lock!  Alternatively
- * changes could be queued until the next frame is delivered using an
- * Atomic pointer and swaps.
- */
-MediaConduitErrorCode
-WebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig)
-{
-  CSFLogDebug(logTag,  "%s for %s", __FUNCTION__, codecConfig ? codecConfig->mName.c_str() : "<null>");
-  bool codecFound = false;
-  MediaConduitErrorCode condError = kMediaConduitNoError;
-  int error = 0; //webrtc engine errors
-  webrtc::VideoCodec  video_codec;
-  std::string payloadName;
-
-  memset(&video_codec, 0, sizeof(video_codec));
-
-  {
-    //validate basic params
-    if((condError = ValidateCodecConfig(codecConfig,true)) != kMediaConduitNoError)
-    {
-      return condError;
-    }
-  }
-
-  condError = StopTransmitting();
-  if (condError != kMediaConduitNoError) {
-    return condError;
-  }
-
-  if (mRtpStreamIdEnabled) {
-    video_codec.ridId = mRtpStreamIdExtId;
-  }
-  if (mExternalSendCodec &&
-      codecConfig->mType == mExternalSendCodec->mType) {
-    CSFLogError(logTag, "%s Configuring External H264 Send Codec", __FUNCTION__);
-
-    // width/height will be overridden on the first frame
-    video_codec.width = 320;
-    video_codec.height = 240;
-#ifdef MOZ_WEBRTC_OMX
-    if (codecConfig->mType == webrtc::kVideoCodecH264) {
-      video_codec.resolution_divisor = 16;
-    } else {
-      video_codec.resolution_divisor = 1; // We could try using it to handle odd resolutions
-    }
-#else
-    video_codec.resolution_divisor = 1; // We could try using it to handle odd resolutions
-#endif
-    video_codec.qpMax = 56;
-    video_codec.numberOfSimulcastStreams = 1;
-    video_codec.simulcastStream[0].jsScaleDownBy =
-        codecConfig->mEncodingConstraints.scaleDownBy;
-    video_codec.mode = mCodecMode;
-
-    codecFound = true;
-  } else {
-    // we should be good here to set the new codec.
-    for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)
-    {
-      if(0 == mPtrViECodec->GetCodec(idx, video_codec))
-      {
-        payloadName = video_codec.plName;
-        if(codecConfig->mName.compare(payloadName) == 0)
-        {
-          // Note: side-effect of this is that video_codec is filled in
-          // by GetCodec()
-          codecFound = true;
-          break;
-        }
-      }
-    }//for
-  }
-
-  if(codecFound == false)
-  {
-    CSFLogError(logTag, "%s Codec Mismatch ", __FUNCTION__);
-    return kMediaConduitInvalidSendCodec;
-  }
-  // Note: only for overriding parameters from GetCodec()!
-  CodecConfigToWebRTCCodec(codecConfig, video_codec);
-  if (mSendingWidth != 0) {
-    // We're already in a call and are reconfiguring (perhaps due to
-    // ReplaceTrack).  Set to match the last frame we sent.
-
-    // We could also set mLastWidth to 0, to force immediate reconfig -
-    // more expensive, but perhaps less risk of missing something.  Really
-    // on ReplaceTrack we should just call ConfigureCodecMode(), and if the
-    // mode changed, we re-configure.
-    // Do this after CodecConfigToWebRTCCodec() to avoid messing up simulcast
-    video_codec.width = mSendingWidth;
-    video_codec.height = mSendingHeight;
-    video_codec.maxFramerate = mSendingFramerate;
-  } else {
-    mSendingWidth = 0;
-    mSendingHeight = 0;
-    mSendingFramerate = video_codec.maxFramerate;
-  }
-
-  video_codec.mode = mCodecMode;
-
-  if(mPtrViECodec->SetSendCodec(mChannel, video_codec) == -1)
-  {
-    error = mPtrViEBase->LastError();
-    if(error == kViECodecInvalidCodec)
-    {
-      CSFLogError(logTag, "%s Invalid Send Codec", __FUNCTION__);
-      return kMediaConduitInvalidSendCodec;
-    }
-    CSFLogError(logTag, "%s SetSendCodec Failed %d ", __FUNCTION__,
-                mPtrViEBase->LastError());
-    return kMediaConduitUnknownError;
-  }
-
-  if (mMinBitrateEstimate != 0) {
-    mPtrViENetwork->SetBitrateConfig(mChannel,
-                                     mMinBitrateEstimate,
-                                     std::max(video_codec.startBitrate,
-                                              mMinBitrateEstimate),
-                                     std::max(video_codec.maxBitrate,
-                                              mMinBitrateEstimate));
-  }
-
-  if (!mVideoCodecStat) {
-    mVideoCodecStat = new VideoCodecStatistics(mChannel, mPtrViECodec);
-  }
-  mVideoCodecStat->Register(true);
-
-  // See Bug 1297058, enabling FEC when NACK is set on H.264 is problematic
-  bool use_fec = codecConfig->RtcpFbFECIsSet();
-  if ((mExternalSendCodec && codecConfig->mType == mExternalSendCodec->mType)
-      || codecConfig->mType == webrtc::kVideoCodecH264) {
-    if(codecConfig->RtcpFbNackIsSet("")) {
-      use_fec = false;
-    }
-  }
-
-  if (use_fec)
-  {
-    uint8_t payload_type_red = INVALID_RTP_PAYLOAD;
-    uint8_t payload_type_ulpfec = INVALID_RTP_PAYLOAD;
-    if (!DetermineREDAndULPFECPayloadTypes(payload_type_red, payload_type_ulpfec)) {
-      CSFLogError(logTag, "%s Unable to set FEC status: could not determine"
-                  "payload type: red %u ulpfec %u",
-                  __FUNCTION__, payload_type_red, payload_type_ulpfec);
-        return kMediaConduitFECStatusError;
-    }
-
-    if(codecConfig->RtcpFbNackIsSet("")) {
-      CSFLogDebug(logTag, "Enabling NACK/FEC (send) for video stream\n");
-      if (mPtrRTP->SetHybridNACKFECStatus(mChannel, true,
-                                          payload_type_red,
-                                          payload_type_ulpfec) != 0) {
-        CSFLogError(logTag,  "%s SetHybridNACKFECStatus Failed %d ",
-                    __FUNCTION__, mPtrViEBase->LastError());
-        return kMediaConduitHybridNACKFECStatusError;
-      }
-    } else {
-      CSFLogDebug(logTag, "Enabling FEC (send) for video stream\n");
-      if (mPtrRTP->SetFECStatus(mChannel, true,
-                                payload_type_red, payload_type_ulpfec) != 0)
-      {
-        CSFLogError(logTag,  "%s SetFECStatus Failed %d ", __FUNCTION__,
-                    mPtrViEBase->LastError());
-        return kMediaConduitFECStatusError;
-      }
-    }
-  } else if(codecConfig->RtcpFbNackIsSet("")) {
-    CSFLogDebug(logTag, "Enabling NACK (send) for video stream\n");
-    if (mPtrRTP->SetNACKStatus(mChannel, true) != 0)
-    {
-      CSFLogError(logTag,  "%s NACKStatus Failed %d ", __FUNCTION__,
-                  mPtrViEBase->LastError());
-      return kMediaConduitNACKStatusError;
-    }
-  }
-
-  {
-    MutexAutoLock lock(mCodecMutex);
-
-    //Copy the applied config for future reference.
-    mCurSendCodecConfig = new VideoCodecConfig(*codecConfig);
-  }
-
-  bool remb_requested = codecConfig->RtcpFbRembIsSet();
-  mPtrRTP->SetRembStatus(mChannel, true, remb_requested);
-
-  return kMediaConduitNoError;
-}
 
 MediaConduitErrorCode
 WebrtcVideoConduit::ConfigureRecvMediaCodecs(
-    const std::vector<VideoCodecConfig* >& codecConfigList)
+  const std::vector<VideoCodecConfig* >& codecConfigList)
 {
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  CSFLogDebug(logTag, "%s ", __FUNCTION__);
   MediaConduitErrorCode condError = kMediaConduitNoError;
-  bool success = false;
-  std::string  payloadName;
+  std::string payloadName;
 
-  condError = StopReceiving();
-  if (condError != kMediaConduitNoError) {
-    return condError;
-  }
-
-  if(codecConfigList.empty())
-  {
+  if (codecConfigList.empty()) {
     CSFLogError(logTag, "%s Zero number of codecs to configure", __FUNCTION__);
     return kMediaConduitMalformedArgument;
   }
 
-  webrtc::ViEKeyFrameRequestMethod kf_request = webrtc::kViEKeyFrameRequestNone;
+  webrtc::KeyFrameRequestMethod kf_request_method = webrtc::kKeyFrameReqPliRtcp;
+  bool kf_request_enabled = false;
   bool use_nack_basic = false;
   bool use_tmmbr = false;
   bool use_remb = false;
   bool use_fec = false;
+  int ulpfec_payload_type = kNullPayloadType;
+  int red_payload_type = kNullPayloadType;
+  bool configuredH264 = false;
+  nsTArray<UniquePtr<VideoCodecConfig>> recv_codecs;
 
-  //Try Applying the codecs in the list
-  // we treat as success if atleast one codec was applied and reception was
+  // Try Applying the codecs in the list
+  // we treat as success if at least one codec was applied and reception was
   // started successfully.
-  for(std::vector<VideoCodecConfig*>::size_type i=0;i < codecConfigList.size();i++)
-  {
-    //if the codec param is invalid or diplicate, return error
-    if((condError = ValidateCodecConfig(codecConfigList[i],false)) != kMediaConduitNoError)
-    {
-      return condError;
+  for (const auto& codec_config : codecConfigList) {
+    // if the codec param is invalid or duplicate, return error
+    if ((condError = ValidateCodecConfig(codec_config, false))
+        != kMediaConduitNoError) {
+      CSFLogError(logTag, "%s Invalid config for %s decoder: %i", __FUNCTION__,
+                  codec_config->mName.c_str(), condError);
+      continue;
+    }
+
+    if (codec_config->mName == "H264") {
+      // TODO(bug 1200768): We can only handle configuring one recv H264 codec
+      if (configuredH264) {
+        continue;
+      }
+      configuredH264 = true;
+    }
+
+    if (codec_config->mName == kUlpFecPayloadName) {
+      ulpfec_payload_type = codec_config->mType;
+      continue;
+    }
+
+    if (codec_config->mName == kRedPayloadName) {
+      red_payload_type = codec_config->mType;
+      continue;
     }
 
     // Check for the keyframe request type: PLI is preferred
     // over FIR, and FIR is preferred over none.
-    if (codecConfigList[i]->RtcpFbNackIsSet("pli"))
-    {
-      kf_request = webrtc::kViEKeyFrameRequestPliRtcp;
-    } else if(kf_request == webrtc::kViEKeyFrameRequestNone &&
-              codecConfigList[i]->RtcpFbCcmIsSet("fir"))
-    {
-      kf_request = webrtc::kViEKeyFrameRequestFirRtcp;
+    // XXX (See upstream issue https://bugs.chromium.org/p/webrtc/issues/detail?id=7002):
+    // There is no 'none' option in webrtc.org
+    if (codec_config->RtcpFbNackIsSet("pli")) {
+      kf_request_enabled = true;
+      kf_request_method = webrtc::kKeyFrameReqPliRtcp;
+    } else if (!kf_request_enabled && codec_config->RtcpFbCcmIsSet("fir")) {
+      kf_request_enabled = true;
+      kf_request_method = webrtc::kKeyFrameReqFirRtcp;
     }
 
-    // Check whether NACK is requested
-    if(codecConfigList[i]->RtcpFbNackIsSet(""))
-    {
-      use_nack_basic = true;
+    // What if codec A has Nack and REMB, and codec B has TMMBR, and codec C has none?
+    // In practice, that's not a useful configuration, and VideoReceiveStream::Config can't
+    // represent that, so simply union the (boolean) settings
+    use_nack_basic |= codec_config->RtcpFbNackIsSet("");
+    use_tmmbr |= codec_config->RtcpFbCcmIsSet("tmmbr");
+    use_remb |= codec_config->RtcpFbRembIsSet();
+    use_fec |= codec_config->RtcpFbFECIsSet();
+
+    recv_codecs.AppendElement(new VideoCodecConfig(*codec_config));
+  }
+
+  // Now decide if we need to recreate the receive stream, or can keep it
+  if (!mRecvStream ||
+      CodecsDifferent(recv_codecs, mRecvCodecList) ||
+      mRecvStreamConfig.rtp.nack.rtp_history_ms != (use_nack_basic ? 1000 : 0) ||
+      mRecvStreamConfig.rtp.remb != use_remb ||
+      mRecvStreamConfig.rtp.tmmbr != use_tmmbr ||
+      mRecvStreamConfig.rtp.keyframe_method != kf_request_method ||
+      (use_fec &&
+       (mRecvStreamConfig.rtp.fec.ulpfec_payload_type != ulpfec_payload_type ||
+        mRecvStreamConfig.rtp.fec.red_payload_type != red_payload_type))) {
+
+    condError = StopReceiving();
+    if (condError != kMediaConduitNoError) {
+      return condError;
     }
 
-    // Check whether TMMBR is requested
-    if (codecConfigList[i]->RtcpFbCcmIsSet("tmmbr")) {
-      use_tmmbr = true;
+    // If we fail after here things get ugly
+    mRecvStreamConfig.rtp.rtcp_mode = webrtc::RtcpMode::kCompound;
+    mRecvStreamConfig.rtp.nack.rtp_history_ms = use_nack_basic ? 1000 : 0;
+    mRecvStreamConfig.rtp.remb = use_remb;
+    mRecvStreamConfig.rtp.tmmbr = use_tmmbr;
+    mRecvStreamConfig.rtp.keyframe_method = kf_request_method;
+
+    if (use_fec) {
+      mRecvStreamConfig.rtp.fec.ulpfec_payload_type = ulpfec_payload_type;
+      mRecvStreamConfig.rtp.fec.red_payload_type = red_payload_type;
+      mRecvStreamConfig.rtp.fec.red_rtx_payload_type = -1;
     }
 
-    // Check whether REMB is requested
-    if (codecConfigList[i]->RtcpFbRembIsSet()) {
-      use_remb = true;
-    }
-
-    // Check whether FEC is requested
-    if (codecConfigList[i]->RtcpFbFECIsSet()) {
-      use_fec = true;
-    }
-
-    webrtc::VideoCodec  video_codec;
-
-    memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-
-    if (mExternalRecvCodec &&
-        codecConfigList[i]->mType == mExternalRecvCodec->mType) {
-      CSFLogError(logTag, "%s Configuring External H264 Receive Codec", __FUNCTION__);
-
-      // XXX Do we need a separate setting for receive maxbitrate?  Is it
-      // different for hardware codecs?  For now assume symmetry.
-      CodecConfigToWebRTCCodec(codecConfigList[i], video_codec);
-
-      // values SetReceiveCodec() cares about are name, type, maxbitrate
-      if(mPtrViECodec->SetReceiveCodec(mChannel,video_codec) == -1)
-      {
-        CSFLogError(logTag, "%s Invalid Receive Codec %d ", __FUNCTION__,
-                    mPtrViEBase->LastError());
-      } else {
-        CSFLogError(logTag, "%s Successfully Set the codec %s", __FUNCTION__,
-                    codecConfigList[i]->mName.c_str());
-        success = true;
+    // FIXME(jesup) - Bug 1325447 -- SSRCs configured here are a problem.
+    // 0 isn't allowed.  Would be best to ask for a random SSRC from the RTP code.
+    // Would need to call rtp_sender.cc -- GenerateSSRC(), which isn't exposed.  It's called on
+    // collision, or when we decide to send.  it should be called on receiver creation.
+    // Here, we're generating the SSRC value - but this causes ssrc_forced in set in rtp_sender,
+    // which locks us into the SSRC - even a collision won't change it!!!
+    auto ssrc = mRecvStreamConfig.rtp.remote_ssrc;
+    do {
+      SECStatus rv = PK11_GenerateRandom(reinterpret_cast<unsigned char*>(&ssrc), sizeof(ssrc));
+      if (rv != SECSuccess) {
+        return kMediaConduitUnknownError;
       }
-    } else {
-      //Retrieve pre-populated codec structure for our codec.
-      for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)
-      {
-        if(mPtrViECodec->GetCodec(idx, video_codec) == 0)
-        {
-          payloadName = video_codec.plName;
-          if(codecConfigList[i]->mName.compare(payloadName) == 0)
-          {
-            CodecConfigToWebRTCCodec(codecConfigList[i], video_codec);
-            if(mPtrViECodec->SetReceiveCodec(mChannel,video_codec) == -1)
-            {
-              CSFLogError(logTag, "%s Invalid Receive Codec %d ", __FUNCTION__,
-                          mPtrViEBase->LastError());
-            } else {
-              CSFLogError(logTag, "%s Successfully Set the codec %s", __FUNCTION__,
-                          codecConfigList[i]->mName.c_str());
-              success = true;
-            }
-            break; //we found a match
-          }
-        }
-      }//end for codeclist
-    }
-  }//end for
+    } while (ssrc == mRecvStreamConfig.rtp.remote_ssrc);
 
-  if(!success)
-  {
-    CSFLogError(logTag, "%s Setting Receive Codec Failed ", __FUNCTION__);
-    return kMediaConduitInvalidReceiveCodec;
-  }
+    mRecvStreamConfig.rtp.local_ssrc = ssrc;
 
-  if (!mVideoCodecStat) {
-    mVideoCodecStat = new VideoCodecStatistics(mChannel, mPtrViECodec);
-  }
-  mVideoCodecStat->Register(false);
-
-  // XXX Currently, we gather up all of the feedback types that the remote
-  // party indicated it supports for all video codecs and configure the entire
-  // conduit based on those capabilities. This is technically out of spec,
-  // as these values should be configured on a per-codec basis. However,
-  // the video engine only provides this API on a per-conduit basis, so that's
-  // how we have to do it. The approach of considering the remote capablities
-  // for the entire conduit to be a union of all remote codec capabilities
-  // (rather than the more conservative approach of using an intersection)
-  // is made to provide as many feedback mechanisms as are likely to be
-  // processed by the remote party (and should be relatively safe, since the
-  // remote party is required to ignore feedback types that it does not
-  // understand).
-  //
-  // Note that our configuration uses this union of remote capabilites as
-  // input to the configuration. It is not isomorphic to the configuration.
-  // For example, it only makes sense to have one frame request mechanism
-  // active at a time; so, if the remote party indicates more than one
-  // supported mechanism, we're only configuring the one we most prefer.
-  //
-  // See http://code.google.com/p/webrtc/issues/detail?id=2331
-
-  if (kf_request != webrtc::kViEKeyFrameRequestNone)
-  {
-    CSFLogDebug(logTag, "Enabling %s frame requests for video stream\n",
-                (kf_request == webrtc::kViEKeyFrameRequestPliRtcp ?
-                 "PLI" : "FIR"));
-    if(mPtrRTP->SetKeyFrameRequestMethod(mChannel, kf_request) != 0)
-    {
-      CSFLogError(logTag,  "%s KeyFrameRequest Failed %d ", __FUNCTION__,
-                  mPtrViEBase->LastError());
-      return kMediaConduitKeyFrameRequestError;
-    }
-  }
-
-  switch (kf_request) {
-    case webrtc::kViEKeyFrameRequestNone:
-      mFrameRequestMethod = FrameRequestNone;
-      break;
-    case webrtc::kViEKeyFrameRequestPliRtcp:
-      mFrameRequestMethod = FrameRequestPli;
-      break;
-    case webrtc::kViEKeyFrameRequestFirRtcp:
-      mFrameRequestMethod = FrameRequestFir;
-      break;
-    default:
-      MOZ_ASSERT(false);
-      mFrameRequestMethod = FrameRequestUnknown;
-  }
-
-  if (use_fec)
-  {
-    uint8_t payload_type_red = INVALID_RTP_PAYLOAD;
-    uint8_t payload_type_ulpfec = INVALID_RTP_PAYLOAD;
-    if (!DetermineREDAndULPFECPayloadTypes(payload_type_red, payload_type_ulpfec)) {
-      CSFLogError(logTag, "%s Unable to set FEC status: could not determine"
-                  "payload type: red %u ulpfec %u",
-                  __FUNCTION__, payload_type_red, payload_type_ulpfec);
-        return kMediaConduitFECStatusError;
+    // XXX Copy over those that are the same and don't rebuild them
+    mRecvCodecList.SwapElements(recv_codecs);
+    recv_codecs.Clear();
+    mRecvStreamConfig.rtp.rtx.clear();
+    // Rebuilds mRecvStream from mRecvStreamConfig
+    DeleteRecvStream();
+    MediaConduitErrorCode rval = CreateRecvStream();
+    if (rval != kMediaConduitNoError) {
+      CSFLogError(logTag, "%s Start Receive Error %d ", __FUNCTION__, rval);
+      return rval;
     }
 
-    // We also need to call SetReceiveCodec for RED and ULPFEC codecs
-    for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++) {
-      webrtc::VideoCodec video_codec;
-      if(mPtrViECodec->GetCodec(idx, video_codec) == 0) {
-        payloadName = video_codec.plName;
-        if(video_codec.codecType == webrtc::VideoCodecType::kVideoCodecRED ||
-           video_codec.codecType == webrtc::VideoCodecType::kVideoCodecULPFEC) {
-          if(mPtrViECodec->SetReceiveCodec(mChannel,video_codec) == -1) {
-            CSFLogError(logTag, "%s Invalid Receive Codec %d ", __FUNCTION__,
-                        mPtrViEBase->LastError());
-          } else {
-            CSFLogDebug(logTag, "%s Successfully Set the codec %s", __FUNCTION__,
-                        video_codec.plName);
-          }
-        }
-      }
-    }
-
-    if (use_nack_basic) {
-      CSFLogDebug(logTag, "Enabling NACK/FEC (recv) for video stream\n");
-      if (mPtrRTP->SetHybridNACKFECStatus(mChannel, true,
-                                          payload_type_red,
-                                          payload_type_ulpfec) != 0) {
-        CSFLogError(logTag,  "%s SetHybridNACKFECStatus Failed %d ",
-                    __FUNCTION__, mPtrViEBase->LastError());
-        return kMediaConduitNACKStatusError;
-      }
-    } else {
-      CSFLogDebug(logTag, "Enabling FEC (recv) for video stream\n");
-      if (mPtrRTP->SetFECStatus(mChannel, true,
-                                payload_type_red, payload_type_ulpfec) != 0)
-      {
-        CSFLogError(logTag,  "%s SetFECStatus Failed %d ", __FUNCTION__,
-                    mPtrViEBase->LastError());
-        return kMediaConduitNACKStatusError;
-      }
-    }
-  } else if(use_nack_basic) {
-    CSFLogDebug(logTag, "Enabling NACK (recv) for video stream\n");
-    if (mPtrRTP->SetNACKStatus(mChannel, true) != 0)
-    {
-      CSFLogError(logTag,  "%s NACKStatus Failed %d ", __FUNCTION__,
-                  mPtrViEBase->LastError());
-      return kMediaConduitNACKStatusError;
-    }
+    return StartReceiving();
   }
-  mUsingNackBasic = use_nack_basic;
-  mUsingFEC = use_fec;
-
-  if (use_tmmbr) {
-    CSFLogDebug(logTag, "Enabling TMMBR for video stream");
-    if (mPtrRTP->SetTMMBRStatus(mChannel, true) != 0) {
-      CSFLogError(logTag, "%s SetTMMBRStatus Failed %d ", __FUNCTION__,
-        mPtrViEBase->LastError());
-      return kMediaConduitTMMBRStatusError;
-    }
-  }
-  mUsingTmmbr = use_tmmbr;
-
-  condError = StartReceiving();
-  if (condError != kMediaConduitNoError) {
-    return condError;
-  }
-
-  // by now we should be successfully started the reception
-  CSFLogDebug(logTag, "REMB enabled for video stream %s",
-              (use_remb ? "yes" : "no"));
-  mPtrRTP->SetRembStatus(mChannel, use_remb, true);
   return kMediaConduitNoError;
 }
 
-template<typename T>
-T MinIgnoreZero(const T& a, const T& b)
+webrtc::VideoDecoder*
+WebrtcVideoConduit::CreateDecoder(webrtc::VideoDecoder::DecoderType aType)
 {
-  return std::min(a? a:b, b? b:a);
+  webrtc::VideoDecoder* decoder = nullptr;
+
+  if (aType == webrtc::VideoDecoder::kH264) {
+    // get an external decoder
+#ifdef MOZ_WEBRTC_OMX
+    decoder = OMXVideoCodec::CreateDecoder(OMXVideoCodec::CodecType::CODEC_H264);
+#else
+    decoder = GmpVideoCodec::CreateDecoder();
+#endif
+    if (decoder) {
+      mRecvCodecPlugin = static_cast<WebrtcVideoDecoder*>(decoder);
+    }
+#ifdef MOZ_WEBRTC_MEDIACODEC
+  } else if (aType == webrtc::VideoDecoder::kVp8) {
+    bool enabled = false;
+    // attempt to get a decoder
+#ifdef MOZILLA_INTERNAL_API
+    enabled = mozilla::Preferences::GetBool(
+                  "media.navigator.hardware.vp8_decode.acceleration_enabled", false);
+#endif
+    if (enabled) {
+      nsCOMPtr<nsIGfxInfo> gfxInfo = do_GetService("@mozilla.org/gfx/info;1");
+      if (gfxInfo) {
+        int32_t status;
+        nsCString discardFailureId;
+
+        if (NS_SUCCEEDED(gfxInfo->GetFeatureStatus(
+                             nsIGfxInfo::FEATURE_WEBRTC_HW_ACCELERATION_DECODE,
+                             discardFailureId, &status))) {
+
+          if (status != nsIGfxInfo::FEATURE_STATUS_OK) {
+            NS_WARNING("VP8 decoder hardware is not whitelisted: disabling.\n");
+          } else {
+            decoder = MediaCodecVideoCodec::CreateDecoder(
+                                        MediaCodecVideoCodec::CodecType::CODEC_VP8);
+          }
+        }
+      }
+    }
+
+    // Use a software VP8 decoder as a fallback.
+    if (!decoder) {
+      decoder = webrtc::VideoDecoder::Create(aType);
+    }
+#endif
+  } else {
+    decoder = webrtc::VideoDecoder::Create(aType);
+  }
+
+  return decoder;
 }
 
-struct ResolutionAndBitrateLimits {
-  uint32_t resolution_in_mb;
-  uint16_t min_bitrate;
-  uint16_t start_bitrate;
-  uint16_t max_bitrate;
+webrtc::VideoEncoder*
+WebrtcVideoConduit::CreateEncoder(webrtc::VideoEncoder::EncoderType aType,
+                                  bool enable_simulcast)
+{
+  webrtc::VideoEncoder* encoder = nullptr;
+  if (aType == webrtc::VideoEncoder::kH264) {
+    // get an external encoder
+#ifdef MOZ_WEBRTC_OMX
+    encoder = OMXVideoCodec::CreateEncoder(OMXVideoCodec::CodecType::CODEC_H264);
+#else
+    encoder = GmpVideoCodec::CreateEncoder();
+#endif
+    if (encoder) {
+      mSendCodecPlugin = static_cast<WebrtcVideoEncoder*>(encoder);
+    }
+#ifdef MOZ_WEBRTC_MEDIACODEC
+  } else if (aType == webrtc::VideoEncoder::kVp8) {
+    bool enabled = false;
+    // attempt to get a encoder
+#ifdef MOZILLA_INTERNAL_API
+    enabled = mozilla::Preferences::GetBool(
+                  "media.navigator.hardware.vp8_encode.acceleration_enabled", false);
+#endif
+    if (enabled) {
+      nsCOMPtr<nsIGfxInfo> gfxInfo = do_GetService("@mozilla.org/gfx/info;1");
+      if (gfxInfo) {
+        int32_t status;
+        nsCString discardFailureId;
+
+        if (NS_SUCCEEDED(gfxInfo->GetFeatureStatus(
+                         nsIGfxInfo::FEATURE_WEBRTC_HW_ACCELERATION_ENCODE,
+                         discardFailureId, &status))) {
+
+          if (status != nsIGfxInfo::FEATURE_STATUS_OK) {
+            NS_WARNING("VP8 encoder hardware is not whitelisted: disabling.\n");
+          } else {
+            encoder = MediaCodecVideoCodec::CreateEncoder(
+                                        MediaCodecVideoCodec::CodecType::CODEC_VP8);
+          }
+        }
+      }
+    }
+
+    // Use a software VP8 encoder as a fallback.
+    if (!encoder) {
+      encoder = webrtc::VideoEncoder::Create(aType, enable_simulcast);
+    }
+#endif
+  } else {
+    encoder = webrtc::VideoEncoder::Create(aType, enable_simulcast);
+  }
+
+  return encoder;
+}
+
+struct ResolutionAndBitrateLimits
+{
+  int resolution_in_mb;
+  int min_bitrate_bps;
+  int start_bitrate_bps;
+  int max_bitrate_bps;
 };
 
 #define MB_OF(w,h) ((unsigned int)((((w+15)>>4))*((unsigned int)((h+15)>>4))))
-
 // For now, try to set the max rates well above the knee in the curve.
 // Chosen somewhat arbitrarily; it's hard to find good data oriented for
 // realtime interactive/talking-head recording.  These rates assume
@@ -1111,79 +1280,79 @@ struct ResolutionAndBitrateLimits {
 // XXX Populate this based on a pref (which we should consider sorting because
 // people won't assume they need to).
 static ResolutionAndBitrateLimits kResolutionAndBitrateLimits[] = {
-  {MB_OF(1920, 1200), 1500, 2000, 10000}, // >HD (3K, 4K, etc)
-  {MB_OF(1280, 720), 1200, 1500, 5000}, // HD ~1080-1200
-  {MB_OF(800, 480), 600, 800, 2500}, // HD ~720
-  {tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, 200, 300, 1300}, // VGA, WVGA
-  {MB_OF(176, 144), 100, 150, 500}, // WQVGA, CIF
-  {0 , 40, 80, 250} // QCIF and below
+  {MB_OF(1920, 1200), KBPS(1500), KBPS(2000), KBPS(10000)}, // >HD (3K, 4K, etc)
+  {MB_OF(1280, 720), KBPS(1200), KBPS(1500), KBPS(5000)}, // HD ~1080-1200
+  {MB_OF(800, 480), KBPS(600), KBPS(800), KBPS(2500)}, // HD ~720
+  {tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, KBPS(200), KBPS(300), KBPS(1300)}, // VGA, WVGA
+  {MB_OF(176, 144), KBPS(100), KBPS(150), KBPS(500)}, // WQVGA, CIF
+  {0 , KBPS(40), KBPS(80), KBPS(250)} // QCIF and below
 };
 
 void
-WebrtcVideoConduit::SelectBitrates(unsigned short width,
-                                   unsigned short height,
-                                   unsigned int cap,
-                                   mozilla::Atomic<int32_t, mozilla::Relaxed>& aLastFramerateTenths,
-                                   unsigned int& out_min,
-                                   unsigned int& out_start,
-                                   unsigned int& out_max)
+WebrtcVideoConduit::SelectBitrates(
+  unsigned short width, unsigned short height, int cap,
+  int32_t aLastFramerateTenths,
+  webrtc::VideoStream& aVideoStream)
 {
+  int& out_min = aVideoStream.min_bitrate_bps;
+  int& out_start = aVideoStream.target_bitrate_bps;
+  int& out_max = aVideoStream.max_bitrate_bps;
   // max bandwidth should be proportional (not linearly!) to resolution, and
   // proportional (perhaps linearly, or close) to current frame rate.
-  unsigned int fs = MB_OF(width, height);
+  int fs = MB_OF(width, height);
 
   for (ResolutionAndBitrateLimits resAndLimits : kResolutionAndBitrateLimits) {
     if (fs > resAndLimits.resolution_in_mb &&
         // pick the highest range where at least start rate is within cap
         // (or if we're at the end of the array).
-        (!cap || resAndLimits.start_bitrate <= cap ||
+        (!cap || resAndLimits.start_bitrate_bps <= cap ||
          resAndLimits.resolution_in_mb == 0)) {
-      out_min = MinIgnoreZero((unsigned int)resAndLimits.min_bitrate, cap);
-      out_start = MinIgnoreZero((unsigned int)resAndLimits.start_bitrate, cap);
-      out_max = MinIgnoreZero((unsigned int)resAndLimits.max_bitrate, cap);
+      out_min = MinIgnoreZero(resAndLimits.min_bitrate_bps, cap);
+      out_start = MinIgnoreZero(resAndLimits.start_bitrate_bps, cap);
+      out_max = MinIgnoreZero(resAndLimits.max_bitrate_bps, cap);
       break;
     }
   }
 
-  // mLastFramerateTenths is an atomic, and scaled by *10
-  double framerate = std::min((aLastFramerateTenths/10.),60.0);
+  // mLastFramerateTenths is scaled by *10
+  double framerate = std::min((aLastFramerateTenths / 10.), 60.0);
   MOZ_ASSERT(framerate > 0);
   // Now linear reduction/increase based on fps (max 60fps i.e. doubling)
   if (framerate >= 10) {
-    out_min = out_min * (framerate/30);
-    out_start = out_start * (framerate/30);
-    out_max = std::max((unsigned int)(out_max * (framerate/30)), cap);
+    out_min = out_min * (framerate / 30);
+    out_start = out_start * (framerate / 30);
+    out_max = std::max(static_cast<int>(out_max * (framerate / 30)), cap);
   } else {
     // At low framerates, don't reduce bandwidth as much - cut slope to 1/2.
     // Mostly this would be ultra-low-light situations/mobile or screensharing.
-    out_min = out_min * ((10-(framerate/2))/30);
-    out_start = out_start * ((10-(framerate/2))/30);
-    out_max = std::max((unsigned int)(out_max * ((10-(framerate/2))/30)), cap);
+    out_min = out_min * ((10 - (framerate / 2)) / 30);
+    out_start = out_start * ((10 - (framerate / 2)) / 30);
+    out_max = std::max(static_cast<int>(out_max * ((10 - (framerate / 2)) / 30)), cap);
   }
 
   if (mMinBitrate && mMinBitrate > out_min) {
     out_min = mMinBitrate;
   }
   // If we try to set a minimum bitrate that is too low, ViE will reject it.
-  out_min = std::max((unsigned int) webrtc::kViEMinCodecBitrate,
-                                  out_min);
+  out_min = std::max(kViEMinCodecBitrate, out_min);
   if (mStartBitrate && mStartBitrate > out_start) {
     out_start = mStartBitrate;
   }
   out_start = std::max(out_start, out_min);
 
-  // Note: mMaxBitrate is the max transport bitrate - it applies to a
-  // single codec encoding, but should also apply to the sum of all
-  // simulcast layers in this encoding!
-  // So sum(layers.maxBitrate) <= mMaxBitrate
-  if (mMaxBitrate && mMaxBitrate > out_max) {
-    out_max = mMaxBitrate;
-  }
+  // Note: mNegotiatedMaxBitrate is the max transport bitrate - it applies to
+  // a single codec encoding, but should also apply to the sum of all
+  // simulcast layers in this encoding!  So sum(layers.maxBitrate) <=
+  // mNegotiatedMaxBitrate
+  // Note that out_max already has had mPrefMaxBitrate applied to it
+  out_max = MinIgnoreZero((int)mNegotiatedMaxBitrate, out_max);
+
+  MOZ_ASSERT(mPrefMaxBitrate == 0 || out_max <= mPrefMaxBitrate);
 }
 
-static void ConstrainPreservingAspectRatioExact(uint32_t max_fs,
-                                                unsigned short* width,
-                                                unsigned short* height)
+template <class t>
+static void
+ConstrainPreservingAspectRatioExact(uint32_t max_fs, t* width, t* height)
 {
   // We could try to pick a better starting divisor, but it won't make any real
   // performance difference.
@@ -1192,7 +1361,7 @@ static void ConstrainPreservingAspectRatioExact(uint32_t max_fs,
       continue; // Not divisible
     }
 
-    if (((*width) * (*height))/(d*d) <= max_fs) {
+    if (((*width) * (*height)) / (d * d) <= max_fs) {
       *width /= d;
       *height /= d;
       return;
@@ -1203,22 +1372,19 @@ static void ConstrainPreservingAspectRatioExact(uint32_t max_fs,
   *height = 0;
 }
 
-static void ConstrainPreservingAspectRatio(uint16_t max_width,
-                                           uint16_t max_height,
-                                           unsigned short* width,
-                                           unsigned short* height)
+template <class t>
+static void
+ConstrainPreservingAspectRatio(uint16_t max_width, uint16_t max_height,
+                               t* width, t* height)
 {
   if (((*width) <= max_width) && ((*height) <= max_height)) {
     return;
   }
 
-  if ((*width) * max_height > max_width * (*height))
-  {
+  if ((*width) * max_height > max_width * (*height)) {
     (*height) = max_width * (*height) / (*width);
     (*width) = max_width;
-  }
-  else
-  {
+  } else {
     (*width) = max_height * (*width) / (*height);
     (*height) = max_height;
   }
@@ -1231,7 +1397,7 @@ static void ConstrainPreservingAspectRatio(uint16_t max_width,
 bool
 WebrtcVideoConduit::SelectSendResolution(unsigned short width,
                                          unsigned short height,
-                                         webrtc::I420VideoFrame *frame) // may be null
+                                         webrtc::VideoFrame* frame) // may be null
 {
   mCodecMutex.AssertCurrentThreadOwns();
   // XXX This will do bandwidth-resolution adaptation as well - bug 877954
@@ -1250,8 +1416,7 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
 
     // Limit resolution to max-fs while keeping same aspect ratio as the
     // incoming image.
-    if (mCurSendCodecConfig->mEncodingConstraints.maxFs)
-    {
+    if (mCurSendCodecConfig->mEncodingConstraints.maxFs) {
       uint32_t max_fs = mCurSendCodecConfig->mEncodingConstraints.maxFs;
       unsigned int cur_fs, mb_width, mb_height, mb_max;
 
@@ -1264,11 +1429,10 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
       cur_fs = mb_width * mb_height;
 
       // Limit resolution to max_fs, but don't scale up.
-      if (cur_fs > max_fs)
-      {
+      if (cur_fs > max_fs) {
         double scale_ratio;
 
-        scale_ratio = sqrt((double) max_fs / (double) cur_fs);
+        scale_ratio = sqrt((double)max_fs / (double)cur_fs);
 
         mb_width = mb_width * scale_ratio;
         mb_height = mb_height * scale_ratio;
@@ -1285,7 +1449,7 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
       }
 
       // Limit width/height seperately to limit effect of extreme aspect ratios.
-      mb_max = (unsigned) sqrt(8 * (double) max_fs);
+      mb_max = (unsigned)sqrt(8 * (double)max_fs);
 
       max_width = 16 * std::min(mb_width, mb_max);
       max_height = 16 * std::min(mb_height, mb_max);
@@ -1297,8 +1461,7 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
   // Adapt to getUserMedia resolution changes
   // check if we need to reconfigure the sending resolution.
   bool changed = false;
-  if (mSendingWidth != width || mSendingHeight != height)
-  {
+  if (mSendingWidth != width || mSendingHeight != height) {
     CSFLogDebug(logTag, "%s: resolution changing to %ux%u (from %ux%u)",
                 __FUNCTION__, width, height, mSendingWidth, mSendingHeight);
     // This will avoid us continually retrying this operation if it fails.
@@ -1309,8 +1472,10 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
     changed = true;
   }
 
-  // uses mSendingWidth/Height
-  unsigned int framerate = SelectSendFrameRate(mSendingFramerate);
+  unsigned int framerate = SelectSendFrameRate(mCurSendCodecConfig,
+                                               mSendingFramerate,
+                                               mSendingWidth,
+                                               mSendingHeight);
   if (mSendingFramerate != framerate) {
     CSFLogDebug(logTag, "%s: framerate changing to %u (from %u)",
                 __FUNCTION__, framerate, mSendingFramerate);
@@ -1334,16 +1499,16 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
       mInReconfig = true;
 
       // We can't pass a UniquePtr<> or unique_ptr<> to a lambda directly
-      webrtc::I420VideoFrame *new_frame = nullptr;
+      webrtc::VideoFrame* new_frame = nullptr;
       if (frame) {
-        new_frame = new webrtc::I420VideoFrame();
+        new_frame = new webrtc::VideoFrame();
         // the internal buffer pointer is refcounted, so we don't have 2 copies here
         new_frame->ShallowCopy(*frame);
       }
       RefPtr<WebrtcVideoConduit> self(this);
       RefPtr<Runnable> webrtc_runnable =
         media::NewRunnableFrom([self, width, height, new_frame]() -> nsresult {
-            UniquePtr<webrtc::I420VideoFrame> local_frame(new_frame); // Simplify cleanup
+            UniquePtr<webrtc::VideoFrame> local_frame(new_frame); // Simplify cleanup
 
             MutexAutoLock lock(self->mCodecMutex);
             return self->ReconfigureSendCodec(width, height, new_frame);
@@ -1366,229 +1531,138 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
 nsresult
 WebrtcVideoConduit::ReconfigureSendCodec(unsigned short width,
                                          unsigned short height,
-                                         webrtc::I420VideoFrame *frame)
+                                         webrtc::VideoFrame* frame)
 {
   mCodecMutex.AssertCurrentThreadOwns();
 
-  // Get current vie codec.
-  webrtc::VideoCodec vie_codec;
-  int32_t err;
-
-  mInReconfig = false;
-  if ((err = mPtrViECodec->GetSendCodec(mChannel, vie_codec)) != 0)
-  {
-    CSFLogError(logTag, "%s: GetSendCodec failed, err %d", __FUNCTION__, err);
+  if (!mEncoderConfig.StreamCount()) {
+    CSFLogError(logTag, "%s: No VideoStreams configured", __FUNCTION__);
     return NS_ERROR_FAILURE;
   }
 
-  CSFLogDebug(logTag,
-              "%s: Requesting resolution change to %ux%u (from %ux%u)",
-              __FUNCTION__, width, height, vie_codec.width, vie_codec.height);
+  mEncoderConfig.ForEachStream(
+    [&](webrtc::VideoStream& video_stream,
+        VideoEncoderConfigBuilder::SimulcastStreamConfig& simStream,
+        const size_t index)
+  {
+    mInReconfig = false;
 
-  if (mRtpStreamIdEnabled) {
-    vie_codec.ridId = mRtpStreamIdExtId;
-  }
+    CSFLogDebug(logTag,
+                "%s: Requesting resolution change to %ux%u (from %ux%u), jsScaleDownBy=%f",
+                __FUNCTION__, width, height, static_cast<unsigned int>(video_stream.width),
+                static_cast<unsigned int>(video_stream.height), simStream.jsScaleDownBy);
 
-  vie_codec.width = width;
-  vie_codec.height = height;
-  vie_codec.maxFramerate = mSendingFramerate;
-  SelectBitrates(vie_codec.width, vie_codec.height, 0,
-                 mLastFramerateTenths,
-                 vie_codec.minBitrate,
-                 vie_codec.startBitrate,
-                 vie_codec.maxBitrate);
-
-  // These are based on lowest-fidelity, because if there is insufficient
-  // bandwidth for all streams, only the lowest fidelity one will be sent.
-  uint32_t minMinBitrate = 0;
-  uint32_t minStartBitrate = 0;
-  // Total for all simulcast streams.
-  uint32_t totalMaxBitrate = 0;
-
-  for (size_t i = vie_codec.numberOfSimulcastStreams; i > 0; --i) {
-    webrtc::SimulcastStream& stream(vie_codec.simulcastStream[i - 1]);
-    stream.width = width;
-    stream.height = height;
-    MOZ_ASSERT(stream.jsScaleDownBy >= 1.0);
-    uint32_t new_width = uint32_t(width / stream.jsScaleDownBy);
-    uint32_t new_height = uint32_t(height / stream.jsScaleDownBy);
-    // TODO: If two layers are similar, only alloc bits to one (Bug 1249859)
-    if (new_width != width || new_height != height) {
-      if (vie_codec.numberOfSimulcastStreams == 1) {
+    MOZ_ASSERT(simStream.jsScaleDownBy >= 1.0);
+    uint32_t new_width = (width / simStream.jsScaleDownBy);
+    uint32_t new_height = (height / simStream.jsScaleDownBy);
+    video_stream.width = width;
+    video_stream.height = height;
+    // XXX this should depend on the final values (below) of video_stream.width/height, not
+    // the current value calculated on the incoming framesize (largest simulcast layer)
+    video_stream.max_framerate = mSendingFramerate;
+    SelectBitrates(video_stream.width, video_stream.height,
+                   // XXX formerly was MinIgnoreZero(mNegotiatedMaxBitrate, simStream.jsMaxBitrate),
+                   simStream.jsMaxBitrate,
+                   mLastFramerateTenths, video_stream);
+    CSFLogVerbose(logTag, "%s: new_width=%" PRIu32 " new_height=%" PRIu32,
+                  __FUNCTION__, new_width, new_height);
+    if (new_width != video_stream.width || new_height != video_stream.height) {
+      if (mEncoderConfig.StreamCount() == 1) {
+        CSFLogVerbose(logTag, "%s: ConstrainPreservingAspectRatio", __FUNCTION__);
         // Use less strict scaling in unicast. That way 320x240 / 3 = 106x79.
         ConstrainPreservingAspectRatio(new_width, new_height,
-                                       &stream.width, &stream.height);
+                                       &video_stream.width, &video_stream.height);
       } else {
+        CSFLogVerbose(logTag, "%s: ConstrainPreservingAspectRatioExact", __FUNCTION__);
         // webrtc.org supposedly won't tolerate simulcast unless every stream
         // is exactly the same aspect ratio. 320x240 / 3 = 80x60.
-        ConstrainPreservingAspectRatioExact(new_width*new_height,
-                                            &stream.width, &stream.height);
+        ConstrainPreservingAspectRatioExact(new_width * new_height,
+                                            &video_stream.width, &video_stream.height);
       }
     }
-    // Give each layer default appropriate bandwidth limits based on the
-    // resolution/framerate of that layer
-    SelectBitrates(stream.width, stream.height,
-                   MinIgnoreZero(stream.jsMaxBitrate, vie_codec.maxBitrate),
-                   mLastFramerateTenths,
-                   stream.minBitrate,
-                   stream.targetBitrate,
-                   stream.maxBitrate);
 
-    // webrtc.org expects the last, highest fidelity, simulcast stream to
-    // always have the same resolution as vie_codec
-    // Also set the least user-constrained of the stream bitrates on vie_codec.
-    if (i == vie_codec.numberOfSimulcastStreams) {
-      vie_codec.width = stream.width;
-      vie_codec.height = stream.height;
-    }
-    minMinBitrate = MinIgnoreZero(stream.minBitrate, minMinBitrate);
-    minStartBitrate = MinIgnoreZero(stream.targetBitrate, minStartBitrate);
-    totalMaxBitrate += stream.maxBitrate;
-  }
-  if (vie_codec.numberOfSimulcastStreams != 0) {
-    vie_codec.minBitrate = std::max(minMinBitrate, vie_codec.minBitrate);
-    vie_codec.maxBitrate = std::min(totalMaxBitrate, vie_codec.maxBitrate);
-    vie_codec.startBitrate = std::max(vie_codec.minBitrate,
-                                      std::min(minStartBitrate,
-                                               vie_codec.maxBitrate));
-  }
-  vie_codec.mode = mCodecMode;
-  if ((err = mPtrViECodec->SetSendCodec(mChannel, vie_codec)) != 0)
-  {
-    CSFLogError(logTag, "%s: SetSendCodec(%ux%u) failed, err %d",
-                __FUNCTION__, width, height, err);
+    CSFLogDebug(
+      logTag, "%s: Encoder resolution changed to %ux%u @ %ufps, bitrate %u:%u",
+      __FUNCTION__, static_cast<unsigned int>(video_stream.width),
+      static_cast<unsigned int>(video_stream.height), mSendingFramerate,
+      video_stream.min_bitrate_bps, video_stream.max_bitrate_bps);
+  });
+  if (!mSendStream->ReconfigureVideoEncoder(mEncoderConfig.GenerateConfig())) {
+    CSFLogError(logTag, "%s: ReconfigureVideoEncoder failed", __FUNCTION__);
     return NS_ERROR_FAILURE;
   }
-  if (mMinBitrateEstimate != 0) {
-    mPtrViENetwork->SetBitrateConfig(mChannel,
-                                     mMinBitrateEstimate,
-                                     std::max(vie_codec.startBitrate,
-                                              mMinBitrateEstimate),
-                                     std::max(vie_codec.maxBitrate,
-                                              mMinBitrateEstimate));
-  }
-
-  CSFLogDebug(logTag, "%s: Encoder resolution changed to %ux%u @ %ufps, bitrate %u:%u",
-              __FUNCTION__, width, height, mSendingFramerate,
-              vie_codec.minBitrate, vie_codec.maxBitrate);
   if (frame) {
     // XXX I really don't like doing this from MainThread...
-    mPtrExtCapture->IncomingFrame(*frame);
-    mVideoCodecStat->SentFrame();
+    mSendStream->Input()->IncomingCapturedFrame(*frame);
     CSFLogDebug(logTag, "%s Inserted a frame from reconfig lambda", __FUNCTION__);
   }
   return NS_OK;
 }
 
-// Invoked under lock of mCodecMutex!
 unsigned int
-WebrtcVideoConduit::SelectSendFrameRate(unsigned int framerate) const
+WebrtcVideoConduit::SelectSendFrameRate(const VideoCodecConfig* codecConfig,
+                                        unsigned int old_framerate,
+                                        unsigned short sending_width,
+                                        unsigned short sending_height) const
 {
-  mCodecMutex.AssertCurrentThreadOwns();
-  unsigned int new_framerate = framerate;
+  unsigned int new_framerate = old_framerate;
 
   // Limit frame rate based on max-mbps
-  if (mCurSendCodecConfig && mCurSendCodecConfig->mEncodingConstraints.maxMbps)
+  if (codecConfig && codecConfig->mEncodingConstraints.maxMbps)
   {
-    unsigned int cur_fs, mb_width, mb_height, max_fps;
+    unsigned int cur_fs, mb_width, mb_height;
 
-    mb_width = (mSendingWidth + 15) >> 4;
-    mb_height = (mSendingHeight + 15) >> 4;
+    mb_width = (sending_width + 15) >> 4;
+    mb_height = (sending_height + 15) >> 4;
 
     cur_fs = mb_width * mb_height;
     if (cur_fs > 0) { // in case no frames have been sent
-      max_fps = mCurSendCodecConfig->mEncodingConstraints.maxMbps/cur_fs;
-      if (max_fps < mSendingFramerate) {
-        new_framerate = max_fps;
-      }
+      new_framerate = codecConfig->mEncodingConstraints.maxMbps / cur_fs;
 
-      if (mCurSendCodecConfig->mEncodingConstraints.maxFps != 0 &&
-          mCurSendCodecConfig->mEncodingConstraints.maxFps < mSendingFramerate) {
-        new_framerate = mCurSendCodecConfig->mEncodingConstraints.maxFps;
-      }
+      new_framerate = MinIgnoreZero(new_framerate, codecConfig->mEncodingConstraints.maxFps);
     }
   }
   return new_framerate;
 }
 
 MediaConduitErrorCode
-WebrtcVideoConduit::SetExternalSendCodec(VideoCodecConfig* config,
-                                         VideoEncoder* encoder) {
-  NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
-  if (!mPtrExtCodec->RegisterExternalSendCodec(mChannel,
-                                              config->mType,
-                                              static_cast<WebrtcVideoEncoder*>(encoder),
-                                              false)) {
-    mExternalSendCodecHandle = encoder;
-    mExternalSendCodec = new VideoCodecConfig(*config);
-    return kMediaConduitNoError;
-  }
-  return kMediaConduitInvalidSendCodec;
-}
-
-MediaConduitErrorCode
-WebrtcVideoConduit::SetExternalRecvCodec(VideoCodecConfig* config,
-                                         VideoDecoder* decoder) {
-  NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
-  if (!mPtrExtCodec->RegisterExternalReceiveCodec(mChannel,
-                                                  config->mType,
-                                                  static_cast<WebrtcVideoDecoder*>(decoder))) {
-    mExternalRecvCodecHandle = decoder;
-    mExternalRecvCodec = new VideoCodecConfig(*config);
-    return kMediaConduitNoError;
-  }
-  return kMediaConduitInvalidReceiveCodec;
-}
-
-MediaConduitErrorCode
-WebrtcVideoConduit::EnableRTPStreamIdExtension(bool enabled, uint8_t id) {
-  mRtpStreamIdEnabled = enabled;
-  mRtpStreamIdExtId = id;
-  return kMediaConduitNoError;
-}
-
-MediaConduitErrorCode
-WebrtcVideoConduit::SendVideoFrame(unsigned char* video_frame,
-                                   unsigned int video_frame_length,
+WebrtcVideoConduit::SendVideoFrame(unsigned char* video_buffer,
+                                   unsigned int video_length,
                                    unsigned short width,
                                    unsigned short height,
                                    VideoType video_type,
                                    uint64_t capture_time)
 {
 
-  //check for  the parameters sanity
-  if(!video_frame || video_frame_length == 0 ||
-     width == 0 || height == 0)
-  {
-    CSFLogError(logTag,  "%s Invalid Parameters ",__FUNCTION__);
+  // check for parameter sanity
+  if (!video_buffer || video_length == 0 || width == 0 || height == 0) {
+    CSFLogError(logTag, "%s Invalid Parameters ", __FUNCTION__);
     MOZ_ASSERT(false);
     return kMediaConduitMalformedArgument;
   }
   MOZ_ASSERT(video_type == VideoType::kVideoI420);
-  MOZ_ASSERT(mPtrExtCapture);
 
   // Transmission should be enabled before we insert any frames.
-  if(!mEngineTransmitting)
-  {
+  if (!mEngineTransmitting) {
     CSFLogError(logTag, "%s Engine not transmitting ", __FUNCTION__);
     return kMediaConduitSessionNotInited;
   }
 
   // insert the frame to video engine in I420 format only
-  webrtc::I420VideoFrame i420_frame;
-  i420_frame.CreateFrame(video_frame, width, height, webrtc::kVideoRotation_0);
-  i420_frame.set_timestamp(capture_time);
-  i420_frame.set_render_time_ms(capture_time);
+  webrtc::VideoFrame video_frame;
+  video_frame.CreateFrame(video_buffer, width, height, webrtc::kVideoRotation_0);
+  video_frame.set_timestamp(capture_time);
+  video_frame.set_render_time_ms(capture_time);
 
-  return SendVideoFrame(i420_frame);
+  return SendVideoFrame(video_frame);
 }
 
 MediaConduitErrorCode
-WebrtcVideoConduit::SendVideoFrame(webrtc::I420VideoFrame& frame)
+WebrtcVideoConduit::SendVideoFrame(webrtc::VideoFrame& frame)
 {
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
+  CSFLogDebug(logTag, "%s", __FUNCTION__);
   // See if we need to recalculate what we're sending.
-  // Don't compare mSendingWidth/Height, since those may not be the same as the input.
+  // Don't compute mSendingWidth/Height, since those may not be the same as the input.
   {
     MutexAutoLock lock(mCodecMutex);
     if (mInReconfig) {
@@ -1596,101 +1670,112 @@ WebrtcVideoConduit::SendVideoFrame(webrtc::I420VideoFrame& frame)
       return kMediaConduitNoError;
     }
     if (frame.width() != mLastWidth || frame.height() != mLastHeight) {
-      CSFLogDebug(logTag, "%s: call SelectSendResolution with %ux%u",
-                  __FUNCTION__, frame.width(), frame.height());
+      CSFLogVerbose(logTag, "%s: call SelectSendResolution with %ux%u",
+                    __FUNCTION__, frame.width(), frame.height());
       if (SelectSendResolution(frame.width(), frame.height(), &frame)) {
         // SelectSendResolution took ownership of the data in i420_frame.
         // Submit the frame after reconfig is done
         return kMediaConduitNoError;
       }
     }
-  }
-  mPtrExtCapture->IncomingFrame(frame);
 
-  mVideoCodecStat->SentFrame();
+    mSendStream->Input()->IncomingCapturedFrame(frame);
+  }
+
+  mSendStreamStats.SentFrame();
   CSFLogDebug(logTag, "%s Inserted a frame", __FUNCTION__);
   return kMediaConduitNoError;
 }
 
 // Transport Layer Callbacks
-MediaConduitErrorCode
-WebrtcVideoConduit::ReceivedRTPPacket(const void *data, int len)
-{
-  CSFLogDebug(logTag, "%s: seq# %u, Channel %d, Len %d ", __FUNCTION__,
-              (uint16_t) ntohs(((uint16_t*) data)[1]), mChannel, len);
 
+MediaConduitErrorCode
+WebrtcVideoConduit::DeliverPacket(const void* data, int len)
+{
   // Media Engine should be receiving already.
-  if(mEngineReceiving)
-  {
-    // let the engine know of a RTP packet to decode
-    // XXX we need to get passed the time the packet was received
-    if(mPtrViENetwork->ReceivedRTPPacket(mChannel, data, len, webrtc::PacketTime()) == -1)
-    {
-      int error = mPtrViEBase->LastError();
-      CSFLogError(logTag, "%s RTP Processing Failed %d ", __FUNCTION__, error);
-      if(error >= kViERtpRtcpInvalidChannelId && error <= kViERtpRtcpRtcpDisabled)
-      {
-        return kMediaConduitRTPProcessingFailed;
-      }
-      return kMediaConduitRTPRTCPModuleError;
-    }
-  } else {
+  if (!mCall) {
     CSFLogError(logTag, "Error: %s when not receiving", __FUNCTION__);
     return kMediaConduitSessionNotInited;
   }
 
+  // XXX we need to get passed the time the packet was received
+  webrtc::PacketReceiver::DeliveryStatus status =
+    mCall->Call()->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
+                                             static_cast<const uint8_t*>(data),
+                                             len, webrtc::PacketTime());
+
+  if (status != webrtc::PacketReceiver::DELIVERY_OK) {
+    CSFLogError(logTag, "%s DeliverPacket Failed, %d", __FUNCTION__, status);
+    return kMediaConduitRTPProcessingFailed;
+  }
+
   return kMediaConduitNoError;
 }
 
 MediaConduitErrorCode
-WebrtcVideoConduit::ReceivedRTCPPacket(const void *data, int len)
+WebrtcVideoConduit::ReceivedRTPPacket(const void* data, int len)
 {
-  CSFLogDebug(logTag, " %s Channel %d, Len %d ", __FUNCTION__, mChannel, len);
+  CSFLogDebug(logTag, "%s: seq# %u, Len %d ", __FUNCTION__,
+              (uint16_t)ntohs(((uint16_t*) data)[1]), len);
 
-  //Media Engine should be receiving already
-  if(mPtrViENetwork->ReceivedRTCPPacket(mChannel,data,len) == -1)
-  {
-    int error = mPtrViEBase->LastError();
-    CSFLogError(logTag, "%s RTCP Processing Failed %d", __FUNCTION__, error);
-    if(error >= kViERtpRtcpInvalidChannelId && error <= kViERtpRtcpRtcpDisabled)
-    {
-      return kMediaConduitRTPProcessingFailed;
-    }
-    return kMediaConduitRTPRTCPModuleError;
+  if (DeliverPacket(data, len) != kMediaConduitNoError) {
+    CSFLogError(logTag, "%s RTP Processing Failed", __FUNCTION__);
+    return kMediaConduitRTPProcessingFailed;
   }
   return kMediaConduitNoError;
 }
 
+MediaConduitErrorCode
+WebrtcVideoConduit::ReceivedRTCPPacket(const void* data, int len)
+{
+  CSFLogDebug(logTag, " %s Len %d ", __FUNCTION__, len);
+
+  if (DeliverPacket(data, len) != kMediaConduitNoError) {
+    CSFLogError(logTag, "%s RTCP Processing Failed", __FUNCTION__);
+    return kMediaConduitRTPProcessingFailed;
+  }
+
+  return kMediaConduitNoError;
+}
+
 MediaConduitErrorCode
 WebrtcVideoConduit::StopTransmitting()
 {
-  if(mEngineTransmitting)
-  {
-    CSFLogDebug(logTag, "%s Engine Already Sending. Attemping to Stop ", __FUNCTION__);
-    if(mPtrViEBase->StopSend(mChannel) == -1)
+  if (mEngineTransmitting) {
     {
-      CSFLogError(logTag, "%s StopSend() Failed %d ",__FUNCTION__,
-                  mPtrViEBase->LastError());
-      return kMediaConduitUnknownError;
+      MutexAutoLock lock(mCodecMutex);
+      if (mSendStream) {
+          CSFLogDebug(logTag, "%s Engine Already Sending. Attemping to Stop ", __FUNCTION__);
+          mSendStream->Stop();
+      }
     }
 
     mEngineTransmitting = false;
   }
-
   return kMediaConduitNoError;
 }
 
 MediaConduitErrorCode
 WebrtcVideoConduit::StartTransmitting()
 {
-  if (!mEngineTransmitting) {
-    if(mPtrViEBase->StartSend(mChannel) == -1)
-    {
-      CSFLogError(logTag, "%s Start Send Error %d ", __FUNCTION__,
-                  mPtrViEBase->LastError());
-      return kMediaConduitUnknownError;
+  if (mEngineTransmitting) {
+    return kMediaConduitNoError;
+  }
+
+  CSFLogDebug(logTag, "%s Attemping to start... ", __FUNCTION__);
+  {
+    // Start Transmitting on the video engine
+    MutexAutoLock lock(mCodecMutex);
+
+    if (!mSendStream) {
+      MediaConduitErrorCode rval = CreateSendStream();
+      if (rval != kMediaConduitNoError) {
+        CSFLogError(logTag, "%s Start Send Error %d ", __FUNCTION__, rval);
+        return rval;
+      }
     }
 
+    mSendStream->Start();
     mEngineTransmitting = true;
   }
 
@@ -1703,346 +1788,148 @@ WebrtcVideoConduit::StopReceiving()
   NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
   // Are we receiving already? If so, stop receiving and playout
   // since we can't apply new recv codec when the engine is playing.
-  if(mEngineReceiving)
-  {
+  if (mEngineReceiving && mRecvStream) {
     CSFLogDebug(logTag, "%s Engine Already Receiving . Attemping to Stop ", __FUNCTION__);
-    if(mPtrViEBase->StopReceive(mChannel) == -1)
-    {
-      int error = mPtrViEBase->LastError();
-      if(error == kViEBaseUnknownError)
-      {
-        CSFLogDebug(logTag, "%s StopReceive() Success ", __FUNCTION__);
-      } else {
-        CSFLogError(logTag, "%s StopReceive() Failed %d ", __FUNCTION__,
-                    mPtrViEBase->LastError());
-        return kMediaConduitUnknownError;
-      }
-    }
-    mEngineReceiving = false;
+    mRecvStream->Stop();
   }
 
+  mEngineReceiving = false;
   return kMediaConduitNoError;
 }
 
 MediaConduitErrorCode
 WebrtcVideoConduit::StartReceiving()
 {
-  if (!mEngineReceiving) {
-    CSFLogDebug(logTag, "%s Attemping to start... ", __FUNCTION__);
-    //Start Receive on the video engine
-    if(mPtrViEBase->StartReceive(mChannel) == -1)
-    {
-      int error = mPtrViEBase->LastError();
-      CSFLogError(logTag, "%s Start Receive Error %d ", __FUNCTION__, error);
+  if (mEngineReceiving) {
+    return kMediaConduitNoError;
+  }
 
-      return kMediaConduitUnknownError;
-    }
+  CSFLogDebug(logTag, "%s Attemping to start... ", __FUNCTION__);
+  {
+    // Start Receive on the video engine
+    MutexAutoLock lock(mCodecMutex);
+    MOZ_ASSERT(mRecvStream);
 
+    mRecvStream->Start();
     mEngineReceiving = true;
   }
 
   return kMediaConduitNoError;
 }
 
-//WebRTC::RTP Callback Implementation
+// WebRTC::RTP Callback Implementation
 // Called on MSG thread
-int WebrtcVideoConduit::SendPacket(int channel, const void* data, size_t len)
+bool
+WebrtcVideoConduit::SendRtp(const uint8_t* packet, size_t length,
+                            const webrtc::PacketOptions& options)
 {
-  CSFLogDebug(logTag,  "%s : channel %d len %lu", __FUNCTION__, channel, (unsigned long) len);
+  // XXX(pkerr) - PacketOptions possibly containing RTP extensions are ignored.
+  // The only field in it is the packet_id, which is used when the header
+  // extension for TransportSequenceNumber is being used, which we don't.
+  CSFLogDebug(logTag, "%s : len %lu", __FUNCTION__, (unsigned long)length);
 
   ReentrantMonitorAutoEnter enter(mTransportMonitor);
-  if(mTransmitterTransport &&
-     (mTransmitterTransport->SendRtpPacket(data, len) == NS_OK))
+  if (!mTransmitterTransport ||
+     NS_FAILED(mTransmitterTransport->SendRtpPacket(packet, length)))
   {
-    CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
-    return len;
-  } else {
     CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
-    return -1;
+    return false;
   }
+
+  CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
+  return true;
 }
 
 // Called from multiple threads including webrtc Process thread
-int WebrtcVideoConduit::SendRTCPPacket(int channel, const void* data, size_t len)
+bool
+WebrtcVideoConduit::SendRtcp(const uint8_t* packet, size_t length)
 {
-  CSFLogDebug(logTag,  "%s : channel %d , len %lu ", __FUNCTION__, channel, (unsigned long) len);
-
+  CSFLogDebug(logTag, "%s : len %lu ", __FUNCTION__, (unsigned long)length);
   // We come here if we have only one pipeline/conduit setup,
   // such as for unidirectional streams.
   // We also end up here if we are receiving
   ReentrantMonitorAutoEnter enter(mTransportMonitor);
-  if(mReceiverTransport &&
-     mReceiverTransport->SendRtcpPacket(data, len) == NS_OK)
+  if (mReceiverTransport &&
+      NS_SUCCEEDED(mReceiverTransport->SendRtcpPacket(packet, length)))
   {
     // Might be a sender report, might be a receiver report, we don't know.
     CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
-    return len;
-  } else if(mTransmitterTransport &&
-            (mTransmitterTransport->SendRtcpPacket(data, len) == NS_OK)) {
-      CSFLogDebug(logTag, "%s Sent RTCP Packet (sender report) ", __FUNCTION__);
-      return len;
-  } else {
-    CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
-    return -1;
-  }
-}
-
-// WebRTC::ExternalMedia Implementation
-int
-WebrtcVideoConduit::FrameSizeChange(unsigned int width,
-                                    unsigned int height,
-                                    unsigned int numStreams)
-{
-  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
-
-
-  ReentrantMonitorAutoEnter enter(mTransportMonitor);
-  mReceivingWidth = width;
-  mReceivingHeight = height;
-  mNumReceivingStreams = numStreams;
-
-  if(mRenderer)
-  {
-    mRenderer->FrameSizeChange(width, height, numStreams);
-    return 0;
+    return true;
+  } else if (mTransmitterTransport &&
+             NS_SUCCEEDED(mTransmitterTransport->SendRtcpPacket(packet, length))) {
+    CSFLogDebug(logTag, "%s Sent RTCP Packet (sender report) ", __FUNCTION__);
+    return true;
   }
 
-  CSFLogError(logTag,  "%s Renderer is NULL ", __FUNCTION__);
-  return -1;
+  CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
+  return false;
 }
 
-int
-WebrtcVideoConduit::DeliverFrame(unsigned char* buffer,
-                                 size_t buffer_size,
-                                 uint32_t time_stamp,
-                                 int64_t ntp_time_ms,
-                                 int64_t render_time,
-                                 void *handle)
-{
-  return DeliverFrame(buffer, buffer_size, mReceivingWidth, (mReceivingWidth+1)>>1,
-                      time_stamp, ntp_time_ms, render_time, handle);
-}
-
-int
-WebrtcVideoConduit::DeliverFrame(unsigned char* buffer,
-                                 size_t buffer_size,
-                                 uint32_t y_stride,
-                                 uint32_t cbcr_stride,
-                                 uint32_t time_stamp,
-                                 int64_t ntp_time_ms,
-                                 int64_t render_time,
-                                 void *handle)
-{
-  CSFLogDebug(logTag,  "%s Buffer Size %lu", __FUNCTION__, (unsigned long) buffer_size);
-
-  ReentrantMonitorAutoEnter enter(mTransportMonitor);
-  if(mRenderer)
-  {
-    layers::Image* img = nullptr;
-    // |handle| should be a webrtc::NativeHandle if available.
-    if (handle) {
-      webrtc::NativeHandle* native_h = static_cast<webrtc::NativeHandle*>(handle);
-      // In the handle, there should be a layers::Image.
-      img = static_cast<layers::Image*>(native_h->GetHandle());
-    }
-
-    if (mVideoLatencyTestEnable && mReceivingWidth && mReceivingHeight) {
-      uint64_t now = PR_Now();
-      uint64_t timestamp = 0;
-      bool ok = YuvStamper::Decode(mReceivingWidth, mReceivingHeight, mReceivingWidth,
-                                   buffer,
-                                   reinterpret_cast<unsigned char*>(&timestamp),
-                                   sizeof(timestamp), 0, 0);
-      if (ok) {
-        VideoLatencyUpdate(now - timestamp);
-      }
-    }
-
-    const ImageHandle img_h(img);
-    mRenderer->RenderVideoFrame(buffer, buffer_size, y_stride, cbcr_stride,
-                                time_stamp, render_time, img_h);
-    return 0;
-  }
-
-  CSFLogError(logTag,  "%s Renderer is NULL  ", __FUNCTION__);
-  return -1;
-}
-
-int
-WebrtcVideoConduit::DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame)
-{
-  if (!webrtc_frame.native_handle()) {
-    uint32_t y_stride = webrtc_frame.stride(static_cast<webrtc::PlaneType>(0));
-    return DeliverFrame(const_cast<uint8_t*>(webrtc_frame.buffer(webrtc::kYPlane)),
-                        CalcBufferSize(webrtc::kI420, y_stride, webrtc_frame.height()),
-                        y_stride,
-                        webrtc_frame.stride(static_cast<webrtc::PlaneType>(1)),
-                        webrtc_frame.timestamp(),
-                        webrtc_frame.ntp_time_ms(),
-                        webrtc_frame.render_time_ms(), nullptr);
-  }
-  size_t buffer_size = CalcBufferSize(webrtc::kI420, webrtc_frame.width(), webrtc_frame.height());
-  CSFLogDebug(logTag,  "%s Buffer Size %lu", __FUNCTION__, (unsigned long) buffer_size);
-
-  ReentrantMonitorAutoEnter enter(mTransportMonitor);
-  if(mRenderer)
-  {
-    layers::Image* img = nullptr;
-    // |handle| should be a webrtc::NativeHandle if available.
-    webrtc::NativeHandle* native_h = static_cast<webrtc::NativeHandle*>(webrtc_frame.native_handle());
-    if (native_h) {
-      // In the handle, there should be a layers::Image.
-      img = static_cast<layers::Image*>(native_h->GetHandle());
-    }
-
-#if 0
-    //#ifndef MOZ_WEBRTC_OMX
-    // XXX - this may not be possible on GONK with textures!
-    if (mVideoLatencyTestEnable && mReceivingWidth && mReceivingHeight) {
-      uint64_t now = PR_Now();
-      uint64_t timestamp = 0;
-      bool ok = YuvStamper::Decode(mReceivingWidth, mReceivingHeight, mReceivingWidth,
-                                   buffer,
-                                   reinterpret_cast<unsigned char*>(&timestamp),
-                                   sizeof(timestamp), 0, 0);
-      if (ok) {
-        VideoLatencyUpdate(now - timestamp);
-      }
-    }
-#endif
-
-    const ImageHandle img_h(img);
-    mRenderer->RenderVideoFrame(nullptr, buffer_size, webrtc_frame.timestamp(),
-                                webrtc_frame.render_time_ms(), img_h);
-    return 0;
-  }
-
-  CSFLogError(logTag,  "%s Renderer is NULL  ", __FUNCTION__);
-  return -1;
-}
-
-/**
- * Copy the codec passed into Conduit's database
- */
-
 void
-WebrtcVideoConduit::CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
-                                              webrtc::VideoCodec& cinst)
+WebrtcVideoConduit::RenderFrame(const webrtc::VideoFrame& video_frame,
+                                int time_to_render_ms)
 {
-  // Note: this assumes cinst is initialized to a base state either by
-  // hand or from a config fetched with GetConfig(); this modifies the config
-  // to match parameters from VideoCodecConfig
-  cinst.plType  = codecInfo->mType;
-  if (codecInfo->mName == "H264") {
-    cinst.codecType = webrtc::kVideoCodecH264;
-    PL_strncpyz(cinst.plName, "H264", sizeof(cinst.plName));
-  } else if (codecInfo->mName == "VP8") {
-    cinst.codecType = webrtc::kVideoCodecVP8;
-    PL_strncpyz(cinst.plName, "VP8", sizeof(cinst.plName));
-  } else if (codecInfo->mName == "VP9") {
-    cinst.codecType = webrtc::kVideoCodecVP9;
-    PL_strncpyz(cinst.plName, "VP9", sizeof(cinst.plName));
-  } else if (codecInfo->mName == "I420") {
-    cinst.codecType = webrtc::kVideoCodecI420;
-    PL_strncpyz(cinst.plName, "I420", sizeof(cinst.plName));
-  } else {
-    cinst.codecType = webrtc::kVideoCodecUnknown;
-    PL_strncpyz(cinst.plName, "Unknown", sizeof(cinst.plName));
+  CSFLogDebug(logTag, "%s ", __FUNCTION__);
+  ReentrantMonitorAutoEnter enter(mTransportMonitor);
+
+  if (!mRenderer) {
+    CSFLogError(logTag, "%s Renderer is NULL  ", __FUNCTION__);
+    return;
   }
 
-  // width/height will be overridden on the first frame; they must be 'sane' for
-  // SetSendCodec()
-  if (codecInfo->mEncodingConstraints.maxFps > 0) {
-    cinst.maxFramerate = codecInfo->mEncodingConstraints.maxFps;
-  } else {
-    cinst.maxFramerate = DEFAULT_VIDEO_MAX_FRAMERATE;
+  if (mReceivingWidth != video_frame.width() ||
+      mReceivingHeight != video_frame.height()) {
+    mReceivingWidth = video_frame.width();
+    mReceivingHeight = video_frame.height();
+    mRenderer->FrameSizeChange(mReceivingWidth, mReceivingHeight, mNumReceivingStreams);
   }
 
-  // Defaults if rates aren't forced by pref.  Typically defaults are
-  // overridden on the first video frame.
-  cinst.minBitrate = mMinBitrate ? mMinBitrate : 200;
-  cinst.startBitrate = mStartBitrate ? mStartBitrate : 300;
-  cinst.targetBitrate = cinst.startBitrate;
-  cinst.maxBitrate = mMaxBitrate ? mMaxBitrate : 2000;
-
-  if (cinst.codecType == webrtc::kVideoCodecH264)
-  {
-#ifdef MOZ_WEBRTC_OMX
-    cinst.resolution_divisor = 16;
-#endif
-    // cinst.codecSpecific.H264.profile = ?
-    cinst.codecSpecific.H264.profile_byte = codecInfo->mProfile;
-    cinst.codecSpecific.H264.constraints = codecInfo->mConstraints;
-    cinst.codecSpecific.H264.level = codecInfo->mLevel;
-    cinst.codecSpecific.H264.packetizationMode = codecInfo->mPacketizationMode;
-    if (codecInfo->mEncodingConstraints.maxBr > 0) {
-      // webrtc.org uses kbps, we use bps
-      cinst.maxBitrate =
-        MinIgnoreZero(cinst.maxBitrate,
-                      codecInfo->mEncodingConstraints.maxBr)/1000;
+  // Attempt to retrieve an timestamp encoded in the image pixels if enabled.
+  if (mVideoLatencyTestEnable && mReceivingWidth && mReceivingHeight) {
+    uint64_t now = PR_Now();
+    uint64_t timestamp = 0;
+    bool ok = YuvStamper::Decode(mReceivingWidth, mReceivingHeight, mReceivingWidth,
+                                 const_cast<unsigned char*>(video_frame.buffer(webrtc::kYPlane)),
+                                 reinterpret_cast<unsigned char*>(&timestamp),
+                                 sizeof(timestamp), 0, 0);
+    if (ok) {
+      VideoLatencyUpdate(now - timestamp);
     }
-    if (codecInfo->mEncodingConstraints.maxMbps > 0) {
-      // Not supported yet!
-      CSFLogError(logTag,  "%s H.264 max_mbps not supported yet  ", __FUNCTION__);
-    }
-    // XXX parse the encoded SPS/PPS data
-    // paranoia
-    cinst.codecSpecific.H264.spsData = nullptr;
-    cinst.codecSpecific.H264.spsLen = 0;
-    cinst.codecSpecific.H264.ppsData = nullptr;
-    cinst.codecSpecific.H264.ppsLen = 0;
-  }
-  // Init mSimulcastEncodings always since they hold info from setParameters.
-  // TODO(bug 1210175): H264 doesn't support simulcast yet.
-  size_t numberOfSimulcastEncodings = std::min(codecInfo->mSimulcastEncodings.size(), (size_t)webrtc::kMaxSimulcastStreams);
-  for (size_t i = 0; i < numberOfSimulcastEncodings; ++i) {
-    const VideoCodecConfig::SimulcastEncoding& encoding =
-      codecInfo->mSimulcastEncodings[i];
-    // Make sure the constraints on the whole stream are reflected.
-    webrtc::SimulcastStream stream;
-    memset(&stream, 0, sizeof(stream));
-    stream.width = cinst.width;
-    stream.height = cinst.height;
-    stream.numberOfTemporalLayers = 1;
-    stream.maxBitrate = cinst.maxBitrate;
-    stream.targetBitrate = cinst.targetBitrate;
-    stream.minBitrate = cinst.minBitrate;
-    stream.qpMax = cinst.qpMax;
-    strncpy(stream.rid, encoding.rid.c_str(), sizeof(stream.rid)-1);
-    stream.rid[sizeof(stream.rid) - 1] = 0;
-
-    // Apply encoding-specific constraints.
-    stream.width = MinIgnoreZero(
-        stream.width,
-        (unsigned short)encoding.constraints.maxWidth);
-    stream.height = MinIgnoreZero(
-        stream.height,
-        (unsigned short)encoding.constraints.maxHeight);
-
-    // webrtc.org uses kbps, we use bps
-    stream.jsMaxBitrate = encoding.constraints.maxBr/1000;
-    stream.jsScaleDownBy = encoding.constraints.scaleDownBy;
-
-    MOZ_ASSERT(stream.jsScaleDownBy >= 1.0);
-    uint32_t width = stream.width? stream.width : 640;
-    uint32_t height = stream.height? stream.height : 480;
-    uint32_t new_width = uint32_t(width / stream.jsScaleDownBy);
-    uint32_t new_height = uint32_t(height / stream.jsScaleDownBy);
-
-    if (new_width != width || new_height != height) {
-      // Estimate. Overridden on first frame.
-      SelectBitrates(new_width, new_height, stream.jsMaxBitrate,
-                     mLastFramerateTenths,
-                     stream.minBitrate,
-                     stream.targetBitrate,
-                     stream.maxBitrate);
-    }
-    // webrtc.org expects simulcast streams to be ordered by increasing
-    // fidelity, our jsep code does the opposite.
-    cinst.simulcastStream[numberOfSimulcastEncodings-i-1] = stream;
   }
 
-  cinst.numberOfSimulcastStreams = numberOfSimulcastEncodings;
+  const ImageHandle img_handle(nullptr);
+  mRenderer->RenderVideoFrame(video_frame.buffer(webrtc::kYPlane),
+                              video_frame.allocated_size(webrtc::kYPlane) +
+                              video_frame.allocated_size(webrtc::kUPlane) +
+                              video_frame.allocated_size(webrtc::kVPlane),
+                              video_frame.stride(webrtc::kYPlane),
+                              video_frame.stride(webrtc::kUPlane),
+                              video_frame.timestamp(),
+                              video_frame.render_time_ms(),
+                              img_handle);
+}
+
+// Compare lists of codecs
+bool
+WebrtcVideoConduit::CodecsDifferent(const nsTArray<UniquePtr<VideoCodecConfig>>& a,
+                                    const nsTArray<UniquePtr<VideoCodecConfig>>& b)
+{
+  // return a != b;
+  // would work if UniquePtr<> operator== compared contents!
+  auto len = a.Length();
+  if (len != b.Length()) {
+    return true;
+  }
+
+  // XXX std::equal would work, if we could use it on this - fails for the
+  // same reason as above.  c++14 would let us pass a comparator function.
+  for (uint32_t i = 0; i < len; ++i) {
+    if (!(*a[i] == *b[i])) {
+      return true;
+    }
+  }
+
+  return false;
 }
 
 /**
@@ -2053,15 +1940,13 @@ MediaConduitErrorCode
 WebrtcVideoConduit::ValidateCodecConfig(const VideoCodecConfig* codecInfo,
                                         bool send)
 {
-  if(!codecInfo)
-  {
+  if(!codecInfo) {
     CSFLogError(logTag, "%s Null CodecConfig ", __FUNCTION__);
     return kMediaConduitMalformedArgument;
   }
 
   if((codecInfo->mName.empty()) ||
-     (codecInfo->mName.length() >= CODEC_PLNAME_SIZE))
-  {
+     (codecInfo->mName.length() >= CODEC_PLNAME_SIZE)) {
     CSFLogError(logTag, "%s Invalid Payload Name Length ", __FUNCTION__);
     return kMediaConduitMalformedArgument;
   }
@@ -2069,6 +1954,17 @@ WebrtcVideoConduit::ValidateCodecConfig(const VideoCodecConfig* codecInfo,
   return kMediaConduitNoError;
 }
 
+void
+WebrtcVideoConduit::DumpCodecDB() const
+{
+  for (auto& entry : mRecvCodecList) {
+    CSFLogDebug(logTag, "Payload Name: %s", entry->mName.c_str());
+    CSFLogDebug(logTag, "Payload Type: %d", entry->mType);
+    CSFLogDebug(logTag, "Payload Max Frame Size: %d", entry->mEncodingConstraints.maxFs);
+    CSFLogDebug(logTag, "Payload Max Frame Rate: %d", entry->mEncodingConstraints.maxFps);
+  }
+}
+
 void
 WebrtcVideoConduit::VideoLatencyUpdate(uint64_t newSample)
 {
@@ -2084,40 +1980,88 @@ WebrtcVideoConduit::MozVideoLatencyAvg()
 uint64_t
 WebrtcVideoConduit::CodecPluginID()
 {
-  if (mExternalSendCodecHandle) {
-    return mExternalSendCodecHandle->PluginID();
-  } else if (mExternalRecvCodecHandle) {
-    return mExternalRecvCodecHandle->PluginID();
+  if (mSendCodecPlugin) {
+    return mSendCodecPlugin->PluginID();
+  } else if (mRecvCodecPlugin) {
+    return mRecvCodecPlugin->PluginID();
   }
+
   return 0;
 }
 
-bool
-WebrtcVideoConduit::DetermineREDAndULPFECPayloadTypes(uint8_t &payload_type_red, uint8_t &payload_type_ulpfec)
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::SetEncoderSpecificSettings(
+  void* aSettingsObj)
 {
-    webrtc::VideoCodec video_codec;
-    payload_type_red = INVALID_RTP_PAYLOAD;
-    payload_type_ulpfec = INVALID_RTP_PAYLOAD;
+  mConfig.encoder_specific_settings = aSettingsObj;
+};
 
-    for(int idx=0; idx < mPtrViECodec->NumberOfCodecs(); idx++)
-    {
-      if(mPtrViECodec->GetCodec(idx, video_codec) == 0)
-      {
-        switch(video_codec.codecType) {
-          case webrtc::VideoCodecType::kVideoCodecRED:
-            payload_type_red = video_codec.plType;
-            break;
-          case webrtc::VideoCodecType::kVideoCodecULPFEC:
-            payload_type_ulpfec = video_codec.plType;
-            break;
-          default:
-            break;
-        }
-      }
-    }
-
-    return payload_type_red != INVALID_RTP_PAYLOAD
-           && payload_type_ulpfec != INVALID_RTP_PAYLOAD;
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::SetMinTransmitBitrateBps(
+  int aXmitMinBps)
+{
+  mConfig.min_transmit_bitrate_bps = aXmitMinBps;
 }
 
-}// end namespace
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::SetContentType(
+  webrtc::VideoEncoderConfig::ContentType aContentType)
+{
+  mConfig.content_type = aContentType;
+}
+
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::SetResolutionDivisor(
+  unsigned char aDivisor)
+{
+  mConfig.resolution_divisor = aDivisor;
+}
+
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::AddStream(
+  webrtc::VideoStream aStream)
+{
+  mConfig.streams.push_back(aStream);
+  mSimulcastStreams.push_back(SimulcastStreamConfig());
+}
+
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::AddStream(
+  webrtc::VideoStream aStream, const SimulcastStreamConfig& aSimulcastConfig)
+{
+  mConfig.streams.push_back(aStream);
+  mSimulcastStreams.push_back(aSimulcastConfig);
+}
+
+size_t
+WebrtcVideoConduit::VideoEncoderConfigBuilder::StreamCount()
+{
+  return mConfig.streams.size();
+}
+
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::ClearStreams()
+{
+  mConfig.streams.clear();
+  mSimulcastStreams.clear();
+}
+
+void
+WebrtcVideoConduit::VideoEncoderConfigBuilder::ForEachStream(
+  const std::function<void(webrtc::VideoStream&, SimulcastStreamConfig&,
+                           const size_t index)>&& f)
+{
+  size_t index = 0;
+  for (auto simulcastStream : mSimulcastStreams) {
+    f(mConfig.streams[index], simulcastStream, index);
+    ++index;
+  }
+}
+
+webrtc::VideoEncoderConfig
+WebrtcVideoConduit::VideoEncoderConfigBuilder::GenerateConfig()
+{
+  return mConfig;
+}
+
+} // end namespace
diff --git a/media/webrtc/signaling/src/media-conduit/VideoConduit.h b/media/webrtc/signaling/src/media-conduit/VideoConduit.h
index ff50d80b58..4bbf993e50 100644
--- a/media/webrtc/signaling/src/media-conduit/VideoConduit.h
+++ b/media/webrtc/signaling/src/media-conduit/VideoConduit.h
@@ -5,59 +5,54 @@
 #ifndef VIDEO_SESSION_H_
 #define VIDEO_SESSION_H_
 
-#include "nsAutoPtr.h"
-#include "mozilla/Attributes.h"
 #include "mozilla/Atomics.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/SharedThreadPool.h"
+#include "nsAutoPtr.h"
+#include "nsITimer.h"
 
+#include "LoadManager.h"
+#include "LoadManagerFactory.h"
 #include "MediaConduitInterface.h"
 #include "MediaEngineWrapper.h"
-#include "CodecStatistics.h"
-#include "LoadManagerFactory.h"
-#include "LoadManager.h"
+#include "RunningStat.h"
 #include "runnable_utils.h"
 
 // conflicts with #include of scoped_ptr.h
 #undef FF
 // Video Engine Includes
+#include "webrtc/call.h"
 #include "webrtc/common_types.h"
 #ifdef FF
 #undef FF // Avoid name collision between scoped_ptr.h and nsCRTGlue.h.
 #endif
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_external_codec.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-
+#include "webrtc/video_decoder.h"
+#include "webrtc/video_encoder.h"
+#include <functional>
+#include <memory>
 /** This file hosts several structures identifying different aspects
  * of a RTP Session.
  */
 
- using  webrtc::ViEBase;
- using  webrtc::ViENetwork;
- using  webrtc::ViECodec;
- using  webrtc::ViECapture;
- using  webrtc::ViERender;
- using  webrtc::ViEExternalCapture;
- using  webrtc::ViEExternalCodec;
-
 namespace mozilla {
 
+const int kVideoMtu = 1200;
+const int kQpMax = 56;
+
 class WebrtcAudioConduit;
 class nsThread;
 
 // Interface of external video encoder for WebRTC.
-class WebrtcVideoEncoder:public VideoEncoder
-                         ,public webrtc::VideoEncoder
-{};
+class WebrtcVideoEncoder : public VideoEncoder
+                         , public webrtc::VideoEncoder
+{
+};
 
 // Interface of external video decoder for WebRTC.
-class WebrtcVideoDecoder:public VideoDecoder
-                         ,public webrtc::VideoDecoder
-{};
+class WebrtcVideoDecoder : public VideoDecoder
+                         , public webrtc::VideoDecoder
+{
+};
 
 /**
  * Concrete class for Video session. Hooks up
@@ -65,12 +60,30 @@ class WebrtcVideoDecoder:public VideoDecoder
  */
 class WebrtcVideoConduit : public VideoSessionConduit
                          , public webrtc::Transport
-                         , public webrtc::ExternalRenderer
+                         , public webrtc::VideoRenderer
 {
 public:
+
+  /* Default minimum bitrate for video streams. */
+  static const uint32_t kDefaultMinBitrate_bps;
+  /* Default start a.k.a. target bitrate for video streams. */
+  static const uint32_t kDefaultStartBitrate_bps;
+  /* Default maximum bitrate for video streams. */
+  static const uint32_t kDefaultMaxBitrate_bps;
+
   //VoiceEngine defined constant for Payload Name Size.
   static const unsigned int CODEC_PLNAME_SIZE;
 
+  /**
+  * Add rtp extensions to the the VideoSendStream
+  * Note: upon a name collision the old extension is removed and the new one
+  *   takes its place.
+  * TODO(@@NG) promote this the MediaConduitInterface when the VoE rework
+  * hits Webrtc.org.
+  */
+  void AddLocalRTPExtensions(const std::vector<webrtc::RtpExtension>& extensions) override;
+  std::vector<webrtc::RtpExtension> GetLocalRTPExtensions() const override;
+
   /**
    * Set up A/V sync between this (incoming) VideoConduit and an audio conduit.
    */
@@ -78,24 +91,24 @@ public:
 
   /**
    * Function to attach Renderer end-point for the Media-Video conduit.
-   * @param aRenderer : Reference to the concrete Video renderer implementation
+   * @param aRenderer : Reference to the concrete mozilla Video renderer implementation
    * Note: Multiple invocations of this API shall remove an existing renderer
    * and attaches the new to the Conduit.
    */
-  virtual MediaConduitErrorCode AttachRenderer(RefPtr<VideoRenderer> aVideoRenderer) override;
+  virtual MediaConduitErrorCode AttachRenderer(RefPtr<mozilla::VideoRenderer> aVideoRenderer) override;
   virtual void DetachRenderer() override;
 
   /**
    * APIs used by the registered external transport to this Conduit to
    * feed in received RTP Frames to the VideoEngine for decoding
    */
-  virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len) override;
+  virtual MediaConduitErrorCode ReceivedRTPPacket(const void* data, int len) override;
 
   /**
    * APIs used by the registered external transport to this Conduit to
    * feed in received RTP Frames to the VideoEngine for decoding
    */
-  virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len) override;
+  virtual MediaConduitErrorCode ReceivedRTCPPacket(const void* data, int len) override;
 
   virtual MediaConduitErrorCode StopTransmitting() override;
   virtual MediaConduitErrorCode StartTransmitting() override;
@@ -127,7 +140,7 @@ public:
    *        transmission sub-system on the engine.
    */
    virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
-                               const std::vector<VideoCodecConfig* >& codecConfigList) override;
+       const std::vector<VideoCodecConfig* >& codecConfigList) override;
 
   /**
    * Register Transport for this Conduit. RTP and RTCP frames from the VideoEngine
@@ -142,15 +155,13 @@ public:
    * @param width, height: dimensions of the frame
    * @param cap: user-enforced max bitrate, or 0
    * @param aLastFramerateTenths: holds the current input framerate
-   * @param out_start, out_min, out_max: bitrate results
+   * @param aVideoStream stream to apply bitrates to
    */
   void SelectBitrates(unsigned short width,
                       unsigned short height,
-                      unsigned int cap,
-                      mozilla::Atomic<int32_t, mozilla::Relaxed>& aLastFramerateTenths,
-                      unsigned int& out_min,
-                      unsigned int& out_start,
-                      unsigned int& out_max);
+                      int cap,
+                      int32_t aLastFramerateTenths,
+                      webrtc::VideoStream& aVideoStream);
 
   /**
    * Function to select and change the encoding resolution based on incoming frame size
@@ -160,7 +171,7 @@ public:
    */
   bool SelectSendResolution(unsigned short width,
                             unsigned short height,
-                            webrtc::I420VideoFrame *frame);
+                            webrtc::VideoFrame* frame);
 
   /**
    * Function to reconfigure the current send codec for a different
@@ -170,7 +181,7 @@ public:
    */
   nsresult ReconfigureSendCodec(unsigned short width,
                                 unsigned short height,
-                                webrtc::I420VideoFrame *frame);
+                                webrtc::VideoFrame* frame);
 
   /**
    * Function to select and change the encoding frame rate based on incoming frame rate
@@ -178,7 +189,10 @@ public:
    * @param current framerate
    * @result new framerate
    */
-  unsigned int SelectSendFrameRate(unsigned int framerate) const;
+  unsigned int SelectSendFrameRate(const VideoCodecConfig* codecConfig,
+                                   unsigned int old_framerate,
+                                   unsigned short sending_width,
+                                   unsigned short sending_height) const;
 
   /**
    * Function to deliver a capture video frame for encoding and transport
@@ -192,66 +206,44 @@ public:
    *       This ensures the inserted video-frames can be transmitted by the conduit
    */
   virtual MediaConduitErrorCode SendVideoFrame(unsigned char* video_frame,
-                                                unsigned int video_frame_length,
-                                                unsigned short width,
-                                                unsigned short height,
-                                                VideoType video_type,
-                                                uint64_t capture_time) override;
-  virtual MediaConduitErrorCode SendVideoFrame(webrtc::I420VideoFrame& frame) override;
+                                               unsigned int video_frame_length,
+                                               unsigned short width,
+                                               unsigned short height,
+                                               VideoType video_type,
+                                               uint64_t capture_time) override;
+  virtual MediaConduitErrorCode SendVideoFrame(webrtc::VideoFrame& frame) override;
 
-  /**
-   * Set an external encoder object |encoder| to the payload type |pltype|
-   * for sender side codec.
-   */
-  virtual MediaConduitErrorCode SetExternalSendCodec(VideoCodecConfig* config,
-                                                     VideoEncoder* encoder) override;
-
-  /**
-   * Set an external decoder object |decoder| to the payload type |pltype|
-   * for receiver side codec.
-   */
-  virtual MediaConduitErrorCode SetExternalRecvCodec(VideoCodecConfig* config,
-                                                     VideoDecoder* decoder) override;
-
-  /**
-  * Enables use of Rtp Stream Id, and sets the extension ID.
-  */
-  virtual MediaConduitErrorCode EnableRTPStreamIdExtension(bool enabled, uint8_t id) override;
-
-  /**
+ /**
+   * webrtc::Transport method implementation
+   * ---------------------------------------
    * Webrtc transport implementation to send and receive RTP packet.
-   * VideoConduit registers itself as ExternalTransport to the VideoEngine
+   * VideoConduit registers itself as ExternalTransport to the VideoStream
    */
-  virtual int SendPacket(int channel, const void *data, size_t len) override;
+  virtual bool SendRtp(const uint8_t* packet, size_t length,
+                       const webrtc::PacketOptions& options) override;
 
   /**
+   * webrtc::Transport method implementation
+   * ---------------------------------------
    * Webrtc transport implementation to send and receive RTCP packet.
    * VideoConduit registers itself as ExternalTransport to the VideoEngine
    */
-  virtual int SendRTCPPacket(int channel, const void *data, size_t len) override;
+  virtual bool SendRtcp(const uint8_t* packet, size_t length) override;
 
 
   /**
-   * Webrtc External Renderer Implementation APIs.
-   * Raw I420 Frames are delivred to the VideoConduit by the VideoEngine
+   * webrtc::VideoRenderer implementation
+   * ------------------------------------
+   * webrtc::VideoFrames are delivered to the VideoConduit by the VideoReceiveStream.
    */
-  virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int) override;
-
-  virtual int DeliverFrame(unsigned char*, size_t, uint32_t , int64_t,
-                           int64_t, void *handle) override;
-
-  virtual int DeliverFrame(unsigned char*, size_t, uint32_t, uint32_t, uint32_t , int64_t,
-                           int64_t, void *handle);
-
-  virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;
+  virtual void RenderFrame(const webrtc::VideoFrame& video_frame,
+                           int time_to_render_ms) override;
 
   /**
-   * Does DeliverFrame() support a null buffer and non-null handle
-   * (video texture)?
-   * B2G support it (when using HW video decoder with graphic buffer output).
-   * XXX Investigate!  Especially for Android
+   * webrtc::VideoRenderer implementation
+   * ------------------------------------
    */
-  virtual bool IsTextureSupported() override {
+  virtual bool IsTextureSupported() const override {
 #ifdef WEBRTC_GONK
     return true;
 #else
@@ -259,6 +251,14 @@ public:
 #endif
   }
 
+  /**
+   * webrtc::VideoRenderer implementation
+   * ------------------------------------
+   */
+  virtual bool SmoothsRenderedFrames() const override {
+    return false;
+  }
+
   virtual uint64_t CodecPluginID() override;
 
   unsigned short SendingWidth() override {
@@ -285,17 +285,17 @@ public:
     return 0;
   }
 
-  WebrtcVideoConduit();
+  explicit WebrtcVideoConduit(RefPtr<WebRtcCallWrapper> aCall);
   virtual ~WebrtcVideoConduit();
 
   MediaConduitErrorCode InitMain();
   virtual MediaConduitErrorCode Init();
 
-  int GetChannel() { return mChannel; }
-  webrtc::VideoEngine* GetVideoEngine() { return mVideoEngine; }
-  bool GetLocalSSRC(unsigned int* ssrc) override;
-  bool SetLocalSSRC(unsigned int ssrc) override;
+  std::vector<unsigned int> GetLocalSSRCs() const override;
+  bool SetLocalSSRCs(const std::vector<unsigned int> & ssrcs) override;
+
   bool GetRemoteSSRC(unsigned int* ssrc) override;
+  bool SetRemoteSSRC(unsigned int ssrc) override;
   bool SetLocalCNAME(const char* cname) override;
   bool GetVideoEncoderStats(double* framerateMean,
                             double* framerateStdDev,
@@ -325,25 +325,91 @@ public:
 private:
   DISALLOW_COPY_AND_ASSIGN(WebrtcVideoConduit);
 
-  static inline bool OnThread(nsIEventTarget *thread)
-  {
-    bool on;
-    nsresult rv;
-    rv = thread->IsOnCurrentThread(&on);
+  /** Shared statistics for receive and transmit video streams
+   */
+  class StreamStatistics {
+  public:
+    void Update(const double aFrameRate, const double aBitrate);
+    /**
+     * Returns gathered stream statistics
+     * @param aOutFrMean: mean framerate
+     * @param aOutFrStdDev: standard deviation of framerate
+     * @param aOutBrMean: mean bitrate
+     * @param aOutBrStdDev: standard deviation of bitrate
+     */
+    bool GetVideoStreamStats(double& aOutFrMean,
+        double& aOutFrStdDev,
+        double& aOutBrMean,
+        double& aOutBrStdDev) const;
+  private:
+    RunningStat mFrameRate;
+    RunningStat mBitrate;
+  };
 
-    // If the target thread has already shut down, we don't want to assert.
-    if (rv != NS_ERROR_NOT_INITIALIZED) {
-      MOZ_ASSERT(NS_SUCCEEDED(rv));
+  /**
+   * Statistics for sending streams
+   */
+  class SendStreamStatistics : public StreamStatistics {
+  public:
+    /**
+     * Returns the calculate number of dropped frames
+     * @param aOutDroppedFrames: the number of dropped frames
+     */
+    void DroppedFrames(uint32_t& aOutDroppedFrames) const;
+    void Update(const webrtc::VideoSendStream::Stats& aStats);
+    /**
+     * Call once for every frame delivered for encoding
+     */
+    void SentFrame() {
+      ++mSentFrames;
     }
+  private:
+    uint32_t mDroppedFrames = 0;
+    mozilla::Atomic<int32_t> mSentFrames;
+  };
 
-    if (NS_WARN_IF(NS_FAILED(rv))) {
-      return false;
-    }
-    return on;
-  }
-
-  //Local database of currently applied receive codecs
-  typedef std::vector<VideoCodecConfig* > RecvCodecList;
+  /** Statistics for receiving streams
+   */
+  class ReceiveStreamStatistics : public StreamStatistics {
+  public:
+    /**
+     * Returns the number of discarded packets
+     * @param aOutDiscPackets: number of discarded packets
+     */
+    void DiscardedPackets(uint32_t& aOutDiscPackets) const;
+    void Update(const webrtc::VideoReceiveStream::Stats& aStats);
+  private:
+    uint32_t mDiscardedPackets = 0;
+  };
+  /*
+   * Stores encoder configuration information and produces
+   * a VideoEncoderConfig from it.
+   */
+  class VideoEncoderConfigBuilder {
+  public:
+    /**
+     * Stores extended data for Simulcast Streams
+     */
+    class SimulcastStreamConfig {
+    public:
+      int jsMaxBitrate; // user-controlled max bitrate
+      double jsScaleDownBy=1.0; // user-controlled downscale
+    };
+    void SetEncoderSpecificSettings(void* aSettingsObj);
+    void SetMinTransmitBitrateBps(int aXmitMinBps);
+    void SetContentType(webrtc::VideoEncoderConfig::ContentType aContentType);
+    void SetResolutionDivisor(unsigned char aDivisor);
+    void AddStream(webrtc::VideoStream aStream);
+    void AddStream(webrtc::VideoStream aStream,const SimulcastStreamConfig& aSimulcastConfig);
+    size_t StreamCount();
+    void ClearStreams();
+    void ForEachStream(
+      const std::function<void(webrtc::VideoStream&, SimulcastStreamConfig&, const size_t index)> && f);
+    webrtc::VideoEncoderConfig GenerateConfig();
+  private:
+    webrtc::VideoEncoderConfig mConfig;
+    std::vector<SimulcastStreamConfig> mSimulcastStreams;
+  };
 
   //Function to convert between WebRTC and Conduit codec structures
   void CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
@@ -355,36 +421,37 @@ private:
   //Utility function to dump recv codec database
   void DumpCodecDB() const;
 
+  bool CodecsDifferent(const nsTArray<UniquePtr<VideoCodecConfig>>& a,
+                       const nsTArray<UniquePtr<VideoCodecConfig>>& b);
+
   // Video Latency Test averaging filter
   void VideoLatencyUpdate(uint64_t new_sample);
 
-  // Utility function to determine RED and ULPFEC payload types
-  bool DetermineREDAndULPFECPayloadTypes(uint8_t &payload_type_red, uint8_t &payload_type_ulpfec);
+  MediaConduitErrorCode CreateSendStream();
+  void DeleteSendStream();
+  MediaConduitErrorCode CreateRecvStream();
+  void DeleteRecvStream();
+
+  webrtc::VideoDecoder* CreateDecoder(webrtc::VideoDecoder::DecoderType aType);
+  webrtc::VideoEncoder* CreateEncoder(webrtc::VideoEncoder::EncoderType aType,
+                                      bool enable_simulcast);
+
+  MediaConduitErrorCode DeliverPacket(const void *data, int len);
 
-  webrtc::VideoEngine* mVideoEngine;
   mozilla::ReentrantMonitor mTransportMonitor;
   RefPtr<TransportInterface> mTransmitterTransport;
   RefPtr<TransportInterface> mReceiverTransport;
-  RefPtr<VideoRenderer> mRenderer;
-
-  ScopedCustomReleasePtr<webrtc::ViEBase> mPtrViEBase;
-  ScopedCustomReleasePtr<webrtc::ViECapture> mPtrViECapture;
-  ScopedCustomReleasePtr<webrtc::ViECodec> mPtrViECodec;
-  ScopedCustomReleasePtr<webrtc::ViENetwork> mPtrViENetwork;
-  ScopedCustomReleasePtr<webrtc::ViERender> mPtrViERender;
-  ScopedCustomReleasePtr<webrtc::ViERTP_RTCP> mPtrRTP;
-  ScopedCustomReleasePtr<webrtc::ViEExternalCodec> mPtrExtCodec;
-
-  webrtc::ViEExternalCapture* mPtrExtCapture;
+  RefPtr<mozilla::VideoRenderer> mRenderer;
 
   // Engine state we are concerned with.
   mozilla::Atomic<bool> mEngineTransmitting; //If true ==> Transmit Sub-system is up and running
   mozilla::Atomic<bool> mEngineReceiving;    // if true ==> Receive Sus-sysmtem up and running
 
-  int mChannel; // Video Channel for this conduit
   int mCapId;   // Capturer for this conduit
+  //Local database of currently applied receive codecs
+  nsTArray<UniquePtr<VideoCodecConfig>> mRecvCodecList;
 
-  Mutex mCodecMutex; // protects mCurrSendCodecConfig
+  Mutex mCodecMutex; // protects mCurrSendCodecConfig, mVideoSend/RecvStreamStats
   nsAutoPtr<VideoCodecConfig> mCurSendCodecConfig;
   bool mInReconfig;
 
@@ -400,10 +467,11 @@ private:
   unsigned short mNumReceivingStreams;
   bool mVideoLatencyTestEnable;
   uint64_t mVideoLatencyAvg;
-  uint32_t mMinBitrate;
-  uint32_t mStartBitrate;
-  uint32_t mMaxBitrate;
-  uint32_t mMinBitrateEstimate;
+  int mMinBitrate;
+  int mStartBitrate;
+  int mPrefMaxBitrate;
+  int mMinBitrateEstimate;
+  int mNegotiatedMaxBitrate;
 
   bool mRtpStreamIdEnabled;
   uint8_t mRtpStreamIdExtId;
@@ -414,16 +482,33 @@ private:
 
   RefPtr<WebrtcAudioConduit> mSyncedTo;
 
-  nsAutoPtr<VideoCodecConfig> mExternalSendCodec;
-  nsAutoPtr<VideoCodecConfig> mExternalRecvCodec;
-  nsAutoPtr<VideoEncoder> mExternalSendCodecHandle;
-  nsAutoPtr<VideoDecoder> mExternalRecvCodecHandle;
-
-  // statistics object for video codec;
-  nsAutoPtr<VideoCodecStatistics> mVideoCodecStat;
-
   nsAutoPtr<LoadManager> mLoadManager;
   webrtc::VideoCodecMode mCodecMode;
+
+  // WEBRTC.ORG Call API
+  RefPtr<WebRtcCallWrapper> mCall;
+
+  webrtc::VideoSendStream* mSendStream;
+  // Must call webrtc::Call::DestroyVideoSendStream to delete
+  webrtc::VideoSendStream::Config mSendStreamConfig;
+  VideoEncoderConfigBuilder mEncoderConfig;
+  webrtc::VideoCodecH264 mEncoderSpecificH264;
+
+  webrtc::VideoReceiveStream* mRecvStream;
+  // Must call webrtc::Call::DestroyVideoReceiveStream to delete
+  webrtc::VideoReceiveStream::Config mRecvStreamConfig;
+
+  // The lifetime of these codecs are maintained by the VideoConduit instance.
+  // They are passed to the webrtc::VideoSendStream or VideoReceiveStream,
+  // on construction.
+  nsAutoPtr<webrtc::VideoEncoder> mEncoder; // only one encoder for now
+  std::vector<std::unique_ptr<webrtc::VideoDecoder>> mDecoders;
+  WebrtcVideoEncoder* mSendCodecPlugin;
+  WebrtcVideoDecoder* mRecvCodecPlugin;
+
+  nsCOMPtr<nsITimer> mVideoStatsTimer;
+  SendStreamStatistics mSendStreamStats;
+  ReceiveStreamStatistics mRecvStreamStats;
 };
 } // end namespace
 
diff --git a/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.cpp b/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.cpp
index ad47e5316c..ec023bd582 100644
--- a/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.cpp
+++ b/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.cpp
@@ -23,8 +23,6 @@
 #include "gmp-video-frame-i420.h"
 #include "gmp-video-frame-encoded.h"
 
-#include "webrtc/video_engine/include/vie_external_codec.h"
-
 namespace mozilla {
 
 #ifdef LOG
@@ -105,28 +103,22 @@ WebrtcGmpVideoEncoder::~WebrtcGmpVideoEncoder()
 }
 
 static int
-WebrtcFrameTypeToGmpFrameType(webrtc::VideoFrameType aIn,
+WebrtcFrameTypeToGmpFrameType(webrtc::FrameType aIn,
                               GMPVideoFrameType *aOut)
 {
   MOZ_ASSERT(aOut);
   switch(aIn) {
-    case webrtc::kKeyFrame:
+    case webrtc::kVideoFrameKey:
       *aOut = kGMPKeyFrame;
       break;
-    case webrtc::kDeltaFrame:
+    case webrtc::kVideoFrameDelta:
       *aOut = kGMPDeltaFrame;
       break;
-    case webrtc::kGoldenFrame:
-      *aOut = kGMPGoldenFrame;
-      break;
-    case webrtc::kAltRefFrame:
-      *aOut = kGMPAltRefFrame;
-      break;
-    case webrtc::kSkipFrame:
+    case webrtc::kEmptyFrame:
       *aOut = kGMPSkipFrame;
       break;
     default:
-      MOZ_CRASH("Unexpected VideoFrameType");
+      MOZ_CRASH("Unexpected webrtc::FrameType");
   }
 
   return WEBRTC_VIDEO_CODEC_OK;
@@ -134,24 +126,18 @@ WebrtcFrameTypeToGmpFrameType(webrtc::VideoFrameType aIn,
 
 static int
 GmpFrameTypeToWebrtcFrameType(GMPVideoFrameType aIn,
-                              webrtc::VideoFrameType *aOut)
+                              webrtc::FrameType *aOut)
 {
   MOZ_ASSERT(aOut);
   switch(aIn) {
     case kGMPKeyFrame:
-      *aOut = webrtc::kKeyFrame;
+      *aOut = webrtc::kVideoFrameKey;
       break;
     case kGMPDeltaFrame:
-      *aOut = webrtc::kDeltaFrame;
-      break;
-    case kGMPGoldenFrame:
-      *aOut = webrtc::kGoldenFrame;
-      break;
-    case kGMPAltRefFrame:
-      *aOut = webrtc::kAltRefFrame;
+      *aOut = webrtc::kVideoFrameDelta;
       break;
     case kGMPSkipFrame:
-      *aOut = webrtc::kSkipFrame;
+      *aOut = webrtc::kEmptyFrame;
       break;
     default:
       MOZ_CRASH("Unexpected GMPVideoFrameType");
@@ -327,9 +313,9 @@ WebrtcGmpVideoEncoder::InitEncoderForSize(unsigned short aWidth,
 
 
 int32_t
-WebrtcGmpVideoEncoder::Encode(const webrtc::I420VideoFrame& aInputImage,
+WebrtcGmpVideoEncoder::Encode(const webrtc::VideoFrame& aInputImage,
                               const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-                              const std::vector<webrtc::VideoFrameType>* aFrameTypes)
+                              const std::vector<webrtc::FrameType>* aFrameTypes)
 {
   MOZ_ASSERT(aInputImage.width() >= 0 && aInputImage.height() >= 0);
   // Would be really nice to avoid this sync dispatch, but it would require a
@@ -375,9 +361,9 @@ WebrtcGmpVideoEncoder::RegetEncoderForResolutionChange(
 }
 
 int32_t
-WebrtcGmpVideoEncoder::Encode_g(const webrtc::I420VideoFrame* aInputImage,
+WebrtcGmpVideoEncoder::Encode_g(const webrtc::VideoFrame* aInputImage,
                                 const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-                                const std::vector<webrtc::VideoFrameType>* aFrameTypes)
+                                const std::vector<webrtc::FrameType>* aFrameTypes)
 {
   if (!mGMP) {
     // destroyed via Terminate(), failed to init, or just not initted yet
@@ -540,7 +526,7 @@ WebrtcGmpVideoEncoder::Encoded(GMPVideoEncodedFrame* aEncodedFrame,
 {
   MutexAutoLock lock(mCallbackMutex);
   if (mCallback) {
-    webrtc::VideoFrameType ft;
+    webrtc::FrameType ft;
     GmpFrameTypeToWebrtcFrameType(aEncodedFrame->FrameType(), &ft);
     uint32_t timestamp = (aEncodedFrame->TimeStamp() * 90ll + 999)/1000;
 
@@ -941,7 +927,7 @@ WebrtcGmpVideoDecoder::Decoded(GMPVideoi420Frame* aDecodedFrame)
 {
   MutexAutoLock lock(mCallbackMutex);
   if (mCallback) {
-    webrtc::I420VideoFrame image;
+    webrtc::VideoFrame image;
     int ret = image.CreateFrame(aDecodedFrame->Buffer(kGMPYPlane),
                                 aDecodedFrame->Buffer(kGMPUPlane),
                                 aDecodedFrame->Buffer(kGMPVPlane),
diff --git a/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.h b/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.h
index 0c01bf53c7..75aebda9c6 100644
--- a/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.h
+++ b/media/webrtc/signaling/src/media-conduit/WebrtcGmpVideoCodec.h
@@ -45,7 +45,7 @@
 #include "MediaConduitInterface.h"
 #include "AudioConduit.h"
 #include "VideoConduit.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 #include "gmp-video-host.h"
 #include "GMPVideoDecoderProxy.h"
@@ -142,9 +142,9 @@ public:
                              int32_t aNumberOfCores,
                              uint32_t aMaxPayloadSize);
 
-  virtual int32_t Encode(const webrtc::I420VideoFrame& aInputImage,
+  virtual int32_t Encode(const webrtc::VideoFrame& aInputImage,
                          const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-                         const std::vector<webrtc::VideoFrameType>* aFrameTypes);
+                         const std::vector<webrtc::FrameType>* aFrameTypes);
 
   virtual int32_t RegisterEncodeCompleteCallback(
     webrtc::EncodedImageCallback* aCallback);
@@ -220,9 +220,9 @@ private:
     uint32_t mMaxPayloadSize;
   };
 
-  int32_t Encode_g(const webrtc::I420VideoFrame* aInputImage,
+  int32_t Encode_g(const webrtc::VideoFrame* aInputImage,
                    const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-                   const std::vector<webrtc::VideoFrameType>* aFrameTypes);
+                   const std::vector<webrtc::FrameType>* aFrameTypes);
   void RegetEncoderForResolutionChange(
       uint32_t aWidth,
       uint32_t aHeight,
@@ -316,9 +316,9 @@ class WebrtcVideoEncoderProxy : public WebrtcVideoEncoder
     }
 
     int32_t Encode(
-        const webrtc::I420VideoFrame& aInputImage,
+        const webrtc::VideoFrame& aInputImage,
         const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-        const std::vector<webrtc::VideoFrameType>* aFrameTypes) override
+        const std::vector<webrtc::FrameType>* aFrameTypes) override
     {
       return mEncoderImpl->Encode(aInputImage,
                                   aCodecSpecificInfo,
diff --git a/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.cpp b/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.cpp
index 27b99d5ede..f2261c8250 100644
--- a/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.cpp
+++ b/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.cpp
@@ -23,6 +23,8 @@
 #include "libyuv/convert.h"
 #include "libyuv/row.h"
 
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
+
 #include <webrtc/common_video/libyuv/include/webrtc_libyuv.h>
 
 using namespace mozilla;
@@ -266,7 +268,7 @@ public:
   void GenerateVideoFrame(
       size_t width, size_t height, uint32_t timeStamp,
       void* decoded,
-      webrtc::I420VideoFrame* videoFrame, int color_format) {
+      webrtc::VideoFrame* videoFrame, int color_format) {
 
     CSFLogDebug(logTag,  "%s ", __FUNCTION__);
 
@@ -532,7 +534,7 @@ class OutputDrain : public MediaCodecOutputDrain
   MediaCodec::GlobalRef mCoder;
   webrtc::EncodedImageCallback* mEncoderCallback;
   webrtc::DecodedImageCallback* mDecoderCallback;
-  webrtc::I420VideoFrame mVideoFrame;
+  webrtc::VideoFrame mVideoFrame;
 
   jobjectArray mInputBuffers;
   jobjectArray mOutputBuffers;
@@ -546,7 +548,7 @@ class OutputDrain : public MediaCodecOutputDrain
   NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcAndroidMediaCodec)
 };
 
-static bool I420toNV12(uint8_t* dstY, uint16_t* dstUV, const webrtc::I420VideoFrame& inputImage) {
+static bool I420toNV12(uint8_t* dstY, uint16_t* dstUV, const webrtc::VideoFrame& inputImage) {
   uint8_t* buffer = dstY;
   uint8_t* dst_y = buffer;
   int dst_stride_y = inputImage.stride(webrtc::kYPlane);
@@ -630,9 +632,9 @@ int32_t WebrtcMediaCodecVP8VideoEncoder::InitEncode(
 }
 
 int32_t WebrtcMediaCodecVP8VideoEncoder::Encode(
-    const webrtc::I420VideoFrame& inputImage,
+    const webrtc::VideoFrame& inputImage,
     const webrtc::CodecSpecificInfo* codecSpecificInfo,
-    const std::vector<webrtc::VideoFrameType>* frame_types) {
+    const std::vector<webrtc::FrameType>* frame_types) {
   CSFLogDebug(logTag,  "%s, w = %d, h = %d", __FUNCTION__, inputImage.width(), inputImage.height());
 
   if (!mMediaCodecEncoder) {
@@ -765,9 +767,9 @@ int32_t WebrtcMediaCodecVP8VideoEncoder::Encode(
         void* directBuffer = reinterpret_cast<uint8_t*>(env->GetDirectBufferAddress(buffer)) + offset;
 
         if (flags == MediaCodec::BUFFER_FLAG_SYNC_FRAME) {
-          mEncodedImage._frameType = webrtc::kKeyFrame;
+          mEncodedImage._frameType = webrtc::kVideoFrameKey;
         } else {
-          mEncodedImage._frameType = webrtc::kDeltaFrame;
+          mEncodedImage._frameType = webrtc::kVideoFrameDelta;
         }
         mEncodedImage._completeFrame = true;
 
@@ -910,7 +912,7 @@ int32_t WebrtcMediaCodecVP8VideoDecoder::Decode(
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
 
-  if (inputImage._frameType == webrtc::kKeyFrame) {
+  if (inputImage._frameType == webrtc::kVideoFrameKey) {
     CSFLogDebug(logTag,  "%s, inputImage is Golden frame",
                   __FUNCTION__);
     mFrameWidth = inputImage._encodedWidth;
diff --git a/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.h b/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.h
index 9d7e900fec..804be152d9 100644
--- a/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.h
+++ b/media/webrtc/signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.h
@@ -5,6 +5,8 @@
 #ifndef WebrtcMediaCodecVP8VideoCodec_h__
 #define WebrtcMediaCodecVP8VideoCodec_h__
 
+#include <jni.h>
+
 #include "mozilla/Mutex.h"
 #include "nsThreadUtils.h"
 #include "nsAutoPtr.h"
@@ -13,6 +15,8 @@
 #include "AudioConduit.h"
 #include "VideoConduit.h"
 
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+
 namespace mozilla {
 
 struct EncodedFrame {
@@ -37,9 +41,9 @@ public:
                              int32_t numberOfCores,
                              size_t maxPayloadSize) override;
 
-  virtual int32_t Encode(const webrtc::I420VideoFrame& inputImage,
+  virtual int32_t Encode(const webrtc::VideoFrame& inputImage,
                           const webrtc::CodecSpecificInfo* codecSpecificInfo,
-                          const std::vector<webrtc::VideoFrameType>* frame_types) override;
+                          const std::vector<webrtc::FrameType>* frame_types) override;
 
   virtual int32_t RegisterEncodeCompleteCallback(webrtc::EncodedImageCallback* callback) override;
 
diff --git a/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.cpp b/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.cpp
index 888b878573..c10c47f76c 100644
--- a/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.cpp
+++ b/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.cpp
@@ -532,7 +532,7 @@ public:
     CODEC_LOGD("Decoder NewFrame: %dx%d, timestamp %lld, renderTimeMs %lld",
                picSize.width, picSize.height, timestamp, renderTimeMs);
 
-    nsAutoPtr<webrtc::I420VideoFrame> videoFrame(new webrtc::I420VideoFrame(
+    nsAutoPtr<webrtc::VideoFrame> videoFrame(new webrtc::VideoFrame(
       new ImageNativeHandle(grallocImage.forget()),
       picSize.width,
       picSize.height,
@@ -857,9 +857,9 @@ WebrtcOMXH264VideoEncoder::InitEncode(const webrtc::VideoCodec* aCodecSettings,
 }
 
 int32_t
-WebrtcOMXH264VideoEncoder::Encode(const webrtc::I420VideoFrame& aInputImage,
+WebrtcOMXH264VideoEncoder::Encode(const webrtc::VideoFrame& aInputImage,
                                   const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-                                  const std::vector<webrtc::VideoFrameType>* aFrameTypes)
+                                  const std::vector<webrtc::FrameType>* aFrameTypes)
 {
   MOZ_ASSERT(mOMX != nullptr);
   if (mOMX == nullptr) {
@@ -973,7 +973,7 @@ WebrtcOMXH264VideoEncoder::Encode(const webrtc::I420VideoFrame& aInputImage,
 #endif
   }
 
-  // Wrap I420VideoFrame input with PlanarYCbCrImage for OMXVideoEncoder.
+  // Wrap VideoFrame input with PlanarYCbCrImage for OMXVideoEncoder.
   layers::PlanarYCbCrData yuvData;
   yuvData.mYChannel = const_cast<uint8_t*>(aInputImage.buffer(webrtc::kYPlane));
   yuvData.mYSize = gfx::IntSize(aInputImage.width(), aInputImage.height());
diff --git a/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.h b/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.h
index 71cf5c6815..145cec19b8 100644
--- a/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.h
+++ b/media/webrtc/signaling/src/media-conduit/WebrtcOMXH264VideoCodec.h
@@ -42,9 +42,9 @@ public:
                              int32_t aNumOfCores,
                              size_t aMaxPayloadSize) override;
 
-  virtual int32_t Encode(const webrtc::I420VideoFrame& aInputImage,
+  virtual int32_t Encode(const webrtc::VideoFrame& aInputImage,
                          const webrtc::CodecSpecificInfo* aCodecSpecificInfo,
-                         const std::vector<webrtc::VideoFrameType>* aFrameTypes) override;
+                         const std::vector<webrtc::FrameType>* aFrameTypes) override;
 
   virtual int32_t RegisterEncodeCompleteCallback(webrtc::EncodedImageCallback* aCallback) override;
 
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
index d47a103122..3cd414b3dd 100644
--- a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
@@ -57,9 +57,7 @@
 #include "mozilla/Sprintf.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/native_handle.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/video_engine/include/vie_errors.h"
 
 #include "logging.h"
 
@@ -93,7 +91,7 @@ public:
                                      VideoType aVideoType,
                                      uint64_t aCaptureTime) = 0;
 
-  virtual void OnVideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame) = 0;
+  virtual void OnVideoFrameConverted(webrtc::VideoFrame& aVideoFrame) = 0;
 
 protected:
   virtual ~VideoConverterListener() {}
@@ -249,7 +247,7 @@ protected:
     }
   }
 
-  void VideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame)
+  void VideoFrameConverted(webrtc::VideoFrame& aVideoFrame)
   {
     MutexAutoLock lock(mMutex);
 
@@ -317,7 +315,7 @@ protected:
 
       if (destFormat != mozilla::kVideoI420) {
         unsigned char *video_frame = static_cast<unsigned char*>(basePtr);
-        webrtc::I420VideoFrame i420_frame;
+        webrtc::VideoFrame i420_frame;
         int stride_y = width;
         int stride_uv = (width + 1) / 2;
         int target_width = width;
@@ -366,7 +364,7 @@ protected:
         uint32_t width = yuv->GetSize().width;
         uint32_t height = yuv->GetSize().height;
 
-        webrtc::I420VideoFrame i420_frame;
+        webrtc::VideoFrame i420_frame;
         int rv = i420_frame.CreateFrame(y, cb, cr, width, height,
                                         yStride, cbCrStride, cbCrStride,
                                         webrtc::kVideoRotation_0);
@@ -747,25 +745,24 @@ MediaPipeline::UpdateTransport_s(int level,
 void
 MediaPipeline::SelectSsrc_m(size_t ssrc_index)
 {
-  RUN_ON_THREAD(sts_thread_,
-                WrapRunnable(
-                    this,
-                    &MediaPipeline::SelectSsrc_s,
-                    ssrc_index),
-                NS_DISPATCH_NORMAL);
+  if (ssrc_index < ssrcs_received_.size()) {
+    uint32_t ssrc = ssrcs_received_[ssrc_index];
+    RUN_ON_THREAD(sts_thread_,
+                  WrapRunnable(
+                               this,
+                               &MediaPipeline::SelectSsrc_s,
+                               ssrc),
+                  NS_DISPATCH_NORMAL);
+
+    conduit_->SetRemoteSSRC(ssrc);
+  }
 }
 
 void
-MediaPipeline::SelectSsrc_s(size_t ssrc_index)
+MediaPipeline::SelectSsrc_s(uint32_t ssrc)
 {
   filter_ = new MediaPipelineFilter;
-  if (ssrc_index < ssrcs_received_.size()) {
-    filter_->AddRemoteSSRC(ssrcs_received_[ssrc_index]);
-  } else {
-    MOZ_MTLOG(ML_WARNING, "SelectSsrc called with " << ssrc_index << " but we "
-                          << "have only seen " << ssrcs_received_.size()
-                          << " ssrcs");
-  }
+  filter_->AddRemoteSSRC(ssrc);
 }
 
 void MediaPipeline::StateChange(TransportFlow *flow, TransportLayer::State state) {
@@ -1274,7 +1271,7 @@ public:
       aVideoFrame, aVideoFrameLength, aWidth, aHeight, aVideoType, aCaptureTime);
   }
 
-  void OnVideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame)
+  void OnVideoFrameConverted(webrtc::VideoFrame& aVideoFrame)
   {
     MOZ_ASSERT(conduit_->type() == MediaSessionConduit::VIDEO);
     static_cast<VideoSessionConduit*>(conduit_.get())->SendVideoFrame(aVideoFrame);
@@ -1368,7 +1365,7 @@ public:
                                      aWidth, aHeight, aVideoType, aCaptureTime);
   }
 
-  void OnVideoFrameConverted(webrtc::I420VideoFrame& aVideoFrame) override
+  void OnVideoFrameConverted(webrtc::VideoFrame& aVideoFrame) override
   {
     MutexAutoLock lock(mutex_);
 
@@ -1638,19 +1635,18 @@ MediaPipeline::TransportInfo* MediaPipeline::GetTransportInfo_s(
 }
 
 nsresult MediaPipeline::PipelineTransport::SendRtpPacket(
-    const void *data, int len) {
+    const uint8_t* data, size_t len) {
 
-    nsAutoPtr<DataBuffer> buf(new DataBuffer(static_cast<const uint8_t *>(data),
-                                             len, len + SRTP_MAX_EXPANSION));
+  nsAutoPtr<DataBuffer> buf(new DataBuffer(data, len, len + SRTP_MAX_EXPANSION));
 
-    RUN_ON_THREAD(sts_thread_,
-                  WrapRunnable(
-                      RefPtr<MediaPipeline::PipelineTransport>(this),
-                      &MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s,
-                      buf, true),
-                  NS_DISPATCH_NORMAL);
+  RUN_ON_THREAD(sts_thread_,
+                WrapRunnable(
+                             RefPtr<MediaPipeline::PipelineTransport>(this),
+                             &MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s,
+                             buf, true),
+                NS_DISPATCH_NORMAL);
 
-    return NS_OK;
+  return NS_OK;
 }
 
 nsresult MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s(
@@ -1705,19 +1701,18 @@ nsresult MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s(
 }
 
 nsresult MediaPipeline::PipelineTransport::SendRtcpPacket(
-    const void *data, int len) {
+    const uint8_t* data, size_t len) {
 
-    nsAutoPtr<DataBuffer> buf(new DataBuffer(static_cast<const uint8_t *>(data),
-                                             len, len + SRTP_MAX_EXPANSION));
+  nsAutoPtr<DataBuffer> buf(new DataBuffer(data, len, len + SRTP_MAX_EXPANSION));
 
-    RUN_ON_THREAD(sts_thread_,
-                  WrapRunnable(
-                      RefPtr<MediaPipeline::PipelineTransport>(this),
-                      &MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s,
-                      buf, false),
-                  NS_DISPATCH_NORMAL);
+  RUN_ON_THREAD(sts_thread_,
+                WrapRunnable(
+                             RefPtr<MediaPipeline::PipelineTransport>(this),
+                             &MediaPipeline::PipelineTransport::SendRtpRtcpPacket_s,
+                             buf, false),
+                NS_DISPATCH_NORMAL);
 
-    return NS_OK;
+  return NS_OK;
 }
 
 void MediaPipelineTransmit::PipelineListener::
@@ -2212,12 +2207,14 @@ public:
   {
 #ifdef MOZILLA_INTERNAL_API
     ReentrantMonitorAutoEnter enter(monitor_);
-#endif // MOZILLA_INTERNAL_API
 
-#if defined(MOZILLA_INTERNAL_API)
     if (buffer) {
       // Create a video frame using |buffer|.
+#ifdef MOZ_WIDGET_GONK
+      RefPtr<PlanarYCbCrImage> yuvImage = new GrallocImage();
+#else
       RefPtr<PlanarYCbCrImage> yuvImage = image_container_->CreatePlanarYCbCrImage();
+#endif // MOZ_WIDGET_GONK
       uint8_t* frame = const_cast<uint8_t*>(static_cast<const uint8_t*> (buffer));
 
       PlanarYCbCrData yuvData;
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h
index d609cbd478..ef3beb3e6f 100644
--- a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.h
@@ -23,7 +23,7 @@
 #include "AudioPacketizer.h"
 #include "StreamTracks.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 // Should come from MediaEngine.h, but that's a pain to include here
 // because of the MOZILLA_EXTERNAL_LINKAGE stuff.
@@ -124,7 +124,7 @@ class MediaPipeline : public sigslot::has_slots<> {
   // Used only for testing; installs a MediaPipelineFilter that filters
   // everything but the nth ssrc
   void SelectSsrc_m(size_t ssrc_index);
-  void SelectSsrc_s(size_t ssrc_index);
+  void SelectSsrc_s(uint32_t ssrc);
 
   virtual Direction direction() const { return direction_; }
   virtual const std::string& trackid() const { return track_id_; }
@@ -154,11 +154,6 @@ class MediaPipeline : public sigslot::has_slots<> {
     MAX_RTP_TYPE
   } RtpType;
 
- protected:
-  virtual ~MediaPipeline();
-  virtual void DetachMedia() {}
-  nsresult AttachTransport_s();
-
   // Separate class to allow ref counting
   class PipelineTransport : public TransportInterface {
    public:
@@ -171,8 +166,8 @@ class MediaPipeline : public sigslot::has_slots<> {
     void Detach() { pipeline_ = nullptr; }
     MediaPipeline *pipeline() const { return pipeline_; }
 
-    virtual nsresult SendRtpPacket(const void* data, int len);
-    virtual nsresult SendRtcpPacket(const void* data, int len);
+    virtual nsresult SendRtpPacket(const uint8_t* data, size_t len);
+    virtual nsresult SendRtcpPacket(const uint8_t* data, size_t len);
 
    private:
     nsresult SendRtpRtcpPacket_s(nsAutoPtr<DataBuffer> data,
@@ -181,6 +176,15 @@ class MediaPipeline : public sigslot::has_slots<> {
     MediaPipeline *pipeline_;  // Raw pointer to avoid cycles
     nsCOMPtr<nsIEventTarget> sts_thread_;
   };
+
+  RefPtr<PipelineTransport> GetPiplelineTransport() {
+    return transport_;
+  }
+
+ protected:
+  virtual ~MediaPipeline();
+  virtual void DetachMedia() {}
+  nsresult AttachTransport_s();
   friend class PipelineTransport;
 
   class TransportInfo {
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
index b56c272f9e..d3ec6cdec1 100644
--- a/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp
@@ -9,7 +9,7 @@
 
 #include "MediaPipelineFilter.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/common_types.h"
 
 namespace mozilla {
 
diff --git a/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.cpp b/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.cpp
index 61c2719cde..041d35eac2 100644
--- a/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.cpp
+++ b/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.cpp
@@ -27,16 +27,6 @@
 #include "MediaEngine.h"
 #endif
 
-#include "GmpVideoCodec.h"
-#ifdef MOZ_WEBRTC_OMX
-#include "OMXVideoCodec.h"
-#include "OMXCodecWrapper.h"
-#endif
-
-#ifdef MOZ_WEBRTC_MEDIACODEC
-#include "MediaCodecVideoCodec.h"
-#endif
-
 #ifdef MOZILLA_INTERNAL_API
 #include "mozilla/Preferences.h"
 #endif
@@ -167,6 +157,8 @@ NegotiatedDetailsToVideoCodecConfigs(const JsepTrackNegotiatedDetails& aDetails,
       return NS_ERROR_INVALID_ARG;
     }
 
+    config->mTias = aDetails.GetTias();
+
     for (size_t i = 0; i < aDetails.GetEncodingCount(); ++i) {
       const JsepTrackEncoding& jsepEncoding(aDetails.GetEncoding(i));
       if (jsepEncoding.HasFormat(codec->mDefaultPt)) {
@@ -176,6 +168,7 @@ NegotiatedDetailsToVideoCodecConfigs(const JsepTrackNegotiatedDetails& aDetails,
         config->mSimulcastEncodings.push_back(encoding);
       }
     }
+
     aConfigs->values.push_back(config);
   }
 
@@ -453,12 +446,14 @@ MediaPipelineFactory::CreateOrUpdateMediaPipeline(
   RefPtr<MediaSessionConduit> conduit;
   if (aTrack.GetMediaType() == SdpMediaSection::kAudio) {
     rv = GetOrCreateAudioConduit(aTrackPair, aTrack, &conduit);
-    if (NS_FAILED(rv))
+    if (NS_FAILED(rv)) {
       return rv;
+    }
   } else if (aTrack.GetMediaType() == SdpMediaSection::kVideo) {
     rv = GetOrCreateVideoConduit(aTrackPair, aTrack, &conduit);
-    if (NS_FAILED(rv))
+    if (NS_FAILED(rv)) {
       return rv;
+    }
   } else {
     // We've created the TransportFlow, nothing else to do here.
     return NS_OK;
@@ -730,17 +725,16 @@ MediaPipelineFactory::GetOrCreateAudioConduit(
     if (!aTrackPair.mSending) {
       // No send track, but we still need to configure an SSRC for receiver
       // reports.
-      if (!conduit->SetLocalSSRC(aTrackPair.mRecvonlySsrc)) {
+      if (!conduit->SetLocalSSRCs(std::vector<unsigned int>(1,aTrackPair.mRecvonlySsrc))) {
         MOZ_MTLOG(ML_ERROR, "SetLocalSSRC failed");
         return NS_ERROR_FAILURE;
       }
     }
   } else {
-    // For now we only expect to have one ssrc per local track.
     auto ssrcs = aTrack.GetSsrcs();
     if (!ssrcs.empty()) {
-      if (!conduit->SetLocalSSRC(ssrcs.front())) {
-        MOZ_MTLOG(ML_ERROR, "SetLocalSSRC failed");
+      if (!conduit->SetLocalSSRCs(ssrcs)) {
+        MOZ_MTLOG(ML_ERROR, "SetLocalSSRCs failed");
         return NS_ERROR_FAILURE;
       }
     }
@@ -786,7 +780,6 @@ MediaPipelineFactory::GetOrCreateVideoConduit(
     const JsepTrack& aTrack,
     RefPtr<MediaSessionConduit>* aConduitp)
 {
-
   if (!aTrack.GetNegotiatedDetails()) {
     MOZ_ASSERT(false, "Track is missing negotiated details");
     return NS_ERROR_INVALID_ARG;
@@ -798,7 +791,7 @@ MediaPipelineFactory::GetOrCreateVideoConduit(
     mPCMedia->GetVideoConduit(aTrackPair.mLevel);
 
   if (!conduit) {
-    conduit = VideoSessionConduit::Create();
+    conduit = VideoSessionConduit::Create(mPCMedia->mCall);
     if (!conduit) {
       MOZ_MTLOG(ML_ERROR, "Could not create video conduit");
       return NS_ERROR_FAILURE;
@@ -822,103 +815,80 @@ MediaPipelineFactory::GetOrCreateVideoConduit(
     return NS_ERROR_FAILURE;
   }
 
+  const std::vector<uint32_t>* ssrcs;
+
   if (receiving) {
+    // NOTE(pkerr) - the Call API requires the both local_ssrc and remote_ssrc be
+    // set to a non-zero value or the CreateVideo...Stream call will fail.
     if (aTrackPair.mSending) {
-      auto ssrcs = &aTrackPair.mSending->GetSsrcs();
+      ssrcs = &aTrackPair.mSending->GetSsrcs();
       if (!ssrcs->empty()) {
-        if (!conduit->SetLocalSSRC(ssrcs->front())) {
-          MOZ_MTLOG(ML_ERROR, "SetLocalSSRC failed(1)");
-          return NS_ERROR_FAILURE;
-        }
-      } else {
-        MOZ_MTLOG(ML_ERROR, "Sending without an SSRC??");
-        return NS_ERROR_FAILURE;
+        conduit->SetLocalSSRCs(*ssrcs);
       }
     } else {
       // No send track, but we still need to configure an SSRC for receiver
       // reports.
-      if (!conduit->SetLocalSSRC(aTrackPair.mRecvonlySsrc)) {
-        MOZ_MTLOG(ML_ERROR, "SetLocalSSRC failed(2)");
+      if (!conduit->SetLocalSSRCs(std::vector<unsigned int>(1,aTrackPair.mRecvonlySsrc))) {
+        MOZ_MTLOG(ML_ERROR, "SetLocalSSRCs failed");
         return NS_ERROR_FAILURE;
       }
     }
 
-    // Prune out stuff we cannot actually do. We should work to eliminate the
-    // need for this.
-    bool configuredH264 = false;
-    for (size_t i = 0; i < configs.values.size();) {
-      // TODO(bug 1200768): We can only handle configuring one recv H264 codec
-      if (configuredH264 && (configs.values[i]->mName == "H264")) {
-        delete configs.values[i];
-        configs.values.erase(configs.values.begin() + i);
-        continue;
-      }
-
-      // TODO(bug 1018791): This really should be checked sooner
-      if (EnsureExternalCodec(*conduit, configs.values[i], false)) {
-        delete configs.values[i];
-        configs.values.erase(configs.values.begin() + i);
-        continue;
-      }
-
-      if (configs.values[i]->mName == "H264") {
-        configuredH264 = true;
-      }
-      ++i;
+    ssrcs = &aTrack.GetSsrcs();
+    // NOTE(pkerr) - this is new behavior. Needed because the CreateVideoReceiveStream
+    // method of the Call API will assert (in debug) and fail if a value is not provided
+    // for the remote_ssrc that will be used by the far-end sender.
+    if (ssrcs->empty()) {
+      MOZ_MTLOG(ML_ERROR, "No SSRC set for receive track");
+      return NS_ERROR_FAILURE;
     }
+    conduit->SetRemoteSSRC(ssrcs->front());
 
     auto error = conduit->ConfigureRecvMediaCodecs(configs.values);
-
     if (error) {
       MOZ_MTLOG(ML_ERROR, "ConfigureRecvMediaCodecs failed: " << error);
       return NS_ERROR_FAILURE;
     }
-  } else {
+  } else { //Create a send side
     // For now we only expect to have one ssrc per local track.
-    auto ssrcs = aTrack.GetSsrcs();
-    if (!ssrcs.empty()) {
-      if (!conduit->SetLocalSSRC(ssrcs.front())) {
-        MOZ_MTLOG(ML_ERROR, "SetLocalSSRC failed");
-        return NS_ERROR_FAILURE;
-      }
+    ssrcs = &aTrack.GetSsrcs();
+    if (ssrcs->empty()) {
+      MOZ_MTLOG(ML_ERROR, "No SSRC set for send track");
+      return NS_ERROR_FAILURE;
+    }
+
+    if (!conduit->SetLocalSSRCs(*ssrcs)) {
+      MOZ_MTLOG(ML_ERROR, "SetLocalSSRC failed");
+      return NS_ERROR_FAILURE;
     }
 
     conduit->SetLocalCNAME(aTrack.GetCNAME().c_str());
 
-    rv = ConfigureVideoCodecMode(aTrack,*conduit);
+    rv = ConfigureVideoCodecMode(aTrack, *conduit);
     if (NS_FAILED(rv)) {
       return rv;
     }
 
-    // TODO(bug 1018791): This really should be checked sooner
-    if (EnsureExternalCodec(*conduit, configs.values[0], true)) {
-      MOZ_MTLOG(ML_ERROR, "External codec not available");
-      return NS_ERROR_FAILURE;
-    }
-
-
     auto error = conduit->ConfigureSendMediaCodec(configs.values[0]);
 
-    const SdpExtmapAttributeList::Extmap* rtpStreamIdExt =
-        aTrack.GetNegotiatedDetails()->GetExt(
-            "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id");
-
-    if (rtpStreamIdExt) {
-      MOZ_MTLOG(ML_DEBUG, "Calling EnableRTPSenderIdExtension");
-      error = conduit->EnableRTPStreamIdExtension(true, rtpStreamIdExt->entry);
-
-      if (error) {
-        MOZ_MTLOG(ML_ERROR, "EnableRTPSenderIdExtension failed: " << error);
-        return NS_ERROR_FAILURE;
-      }
-    }
-
     if (error) {
       MOZ_MTLOG(ML_ERROR, "ConfigureSendMediaCodec failed: " << error);
       return NS_ERROR_FAILURE;
     }
   }
 
+  const JsepTrackNegotiatedDetails* details = aTrack.GetNegotiatedDetails();
+  if (details) {
+    // @@NG read extmap from track
+    std::vector<webrtc::RtpExtension> extmaps;
+    details->ForEachRTPHeaderExtension(
+      [&conduit,&extmaps](const SdpExtmapAttributeList::Extmap& extmap)
+    {
+      extmaps.emplace_back(extmap.extensionname,extmap.entry);
+    });
+    conduit->AddLocalRTPExtensions(extmaps);
+  }
+
   *aConduitp = conduit;
 
   return NS_OK;
@@ -970,107 +940,5 @@ MediaPipelineFactory::ConfigureVideoCodecMode(const JsepTrack& aTrack,
   return NS_OK;
 }
 
-/*
- * Add external H.264 video codec.
- */
-MediaConduitErrorCode
-MediaPipelineFactory::EnsureExternalCodec(VideoSessionConduit& aConduit,
-                                          VideoCodecConfig* aConfig,
-                                          bool aIsSend)
-{
-  if (aConfig->mName == "VP8") {
-#ifdef MOZ_WEBRTC_MEDIACODEC
-     if (aIsSend) {
-#ifdef MOZILLA_INTERNAL_API
-       bool enabled = mozilla::Preferences::GetBool("media.navigator.hardware.vp8_encode.acceleration_enabled", false);
-#else
-       bool enabled = false;
-#endif
-       if (enabled) {
-         nsCOMPtr<nsIGfxInfo> gfxInfo = do_GetService("@mozilla.org/gfx/info;1");
-         if (gfxInfo) {
-           int32_t status;
-           nsCString discardFailureId;
-           if (NS_SUCCEEDED(gfxInfo->GetFeatureStatus(nsIGfxInfo::FEATURE_WEBRTC_HW_ACCELERATION_ENCODE, discardFailureId, &status))) {
-             if (status != nsIGfxInfo::FEATURE_STATUS_OK) {
-               NS_WARNING("VP8 encoder hardware is not whitelisted: disabling.\n");
-             } else {
-               VideoEncoder* encoder = nullptr;
-               encoder = MediaCodecVideoCodec::CreateEncoder(MediaCodecVideoCodec::CodecType::CODEC_VP8);
-               if (encoder) {
-                 return aConduit.SetExternalSendCodec(aConfig, encoder);
-               }
-               return kMediaConduitNoError;
-             }
-           }
-         }
-       }
-     } else {
-#ifdef MOZILLA_INTERNAL_API
-       bool enabled = mozilla::Preferences::GetBool("media.navigator.hardware.vp8_decode.acceleration_enabled", false);
-#else
-       bool enabled = false;
-#endif
-       if (enabled) {
-         nsCOMPtr<nsIGfxInfo> gfxInfo = do_GetService("@mozilla.org/gfx/info;1");
-         if (gfxInfo) {
-           int32_t status;
-           nsCString discardFailureId;
-           if (NS_SUCCEEDED(gfxInfo->GetFeatureStatus(nsIGfxInfo::FEATURE_WEBRTC_HW_ACCELERATION_DECODE, discardFailureId, &status))) {
-             if (status != nsIGfxInfo::FEATURE_STATUS_OK) {
-               NS_WARNING("VP8 decoder hardware is not whitelisted: disabling.\n");
-             } else {
-               VideoDecoder* decoder;
-               decoder = MediaCodecVideoCodec::CreateDecoder(MediaCodecVideoCodec::CodecType::CODEC_VP8);
-               if (decoder) {
-                 return aConduit.SetExternalRecvCodec(aConfig, decoder);
-               }
-               return kMediaConduitNoError;
-             }
-           }
-         }
-       }
-     }
-#endif
-     return kMediaConduitNoError;
-  }
-  if (aConfig->mName == "VP9") {
-    return kMediaConduitNoError;
-  }
-  if (aConfig->mName == "H264") {
-    if (aConduit.CodecPluginID() != 0) {
-      return kMediaConduitNoError;
-    }
-    // Register H.264 codec.
-    if (aIsSend) {
-      VideoEncoder* encoder = nullptr;
-#ifdef MOZ_WEBRTC_OMX
-      encoder =
-          OMXVideoCodec::CreateEncoder(OMXVideoCodec::CodecType::CODEC_H264);
-#else
-      encoder = GmpVideoCodec::CreateEncoder();
-#endif
-      if (encoder) {
-        return aConduit.SetExternalSendCodec(aConfig, encoder);
-      }
-      return kMediaConduitInvalidSendCodec;
-    }
-    VideoDecoder* decoder = nullptr;
-#ifdef MOZ_WEBRTC_OMX
-    decoder =
-      OMXVideoCodec::CreateDecoder(OMXVideoCodec::CodecType::CODEC_H264);
-#else
-    decoder = GmpVideoCodec::CreateDecoder();
-#endif
-    if (decoder) {
-      return aConduit.SetExternalRecvCodec(aConfig, decoder);
-    }
-    return kMediaConduitInvalidReceiveCodec;
-  }
-  MOZ_MTLOG(ML_ERROR,
-            "Invalid video codec configured: " << aConfig->mName.c_str());
-  return aIsSend ? kMediaConduitInvalidSendCodec
-                 : kMediaConduitInvalidReceiveCodec;
-}
 
 } // namespace mozilla
diff --git a/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.h b/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.h
index 972c4368a6..9f6c2537ac 100644
--- a/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.h
+++ b/media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.h
@@ -52,10 +52,6 @@ private:
                                    const JsepTrack& aTrack,
                                    RefPtr<MediaSessionConduit>* aConduitp);
 
-  MediaConduitErrorCode EnsureExternalCodec(VideoSessionConduit& aConduit,
-                                            VideoCodecConfig* aConfig,
-                                            bool aIsSend);
-
   nsresult CreateOrGetTransportFlow(size_t aLevel, bool aIsRtcp,
                                     const JsepTransport& transport,
                                     RefPtr<TransportFlow>* out);
diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
index 3b4363a13c..555c820cbf 100644
--- a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
+++ b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
@@ -2745,7 +2745,11 @@ PeerConnectionImpl::ReplaceTrack(MediaStreamTrack& aThisTrack,
   // We update the media pipelines here so we can apply different codec
   // settings for different sources (e.g. screensharing as opposed to camera.)
   // TODO: We should probably only do this if the source has in fact changed.
-  mMedia->UpdateMediaPipelines(*mJsepSession);
+
+  if (NS_FAILED((rv = mMedia->UpdateMediaPipelines(*mJsepSession)))) {
+    CSFLogError(logTag, "Error Updating MediaPipelines");
+    return rv;
+  }
 
   pco->OnReplaceTrackSuccess(jrv);
   if (jrv.Failed()) {
@@ -3108,7 +3112,11 @@ PeerConnectionImpl::SetSignalingState_m(PCImplSignalingState aSignalingState,
     // transports, but nothing further needs to be done.
     mMedia->ActivateOrRemoveTransports(*mJsepSession);
     if (!rollback) {
-      mMedia->UpdateMediaPipelines(*mJsepSession);
+      if (NS_FAILED(mMedia->UpdateMediaPipelines(*mJsepSession))) {
+        CSFLogError(logTag, "Error Updating MediaPipelines");
+        NS_ASSERTION(false, "Error Updating MediaPipelines in SetSignalingState_m()");
+        // XXX what now?  Not much we can do but keep going, without major restructuring
+      }
       InitializeDataChannel();
       mMedia->StartIceChecks(*mJsepSession);
     }
@@ -3663,15 +3671,17 @@ PeerConnectionImpl::ExecuteStatsQuery_s(RTCStatsQuery *query) {
     idstr.AppendLiteral("_");
     idstr.AppendInt(mp.level());
 
+    // TODO(@@NG):ssrcs handle Conduits having multiple stats at the same level
+    // This is pending spec work
     // Gather pipeline stats.
     switch (mp.direction()) {
       case MediaPipeline::TRANSMIT: {
         nsString localId = NS_LITERAL_STRING("outbound_rtp_") + idstr;
         nsString remoteId;
         nsString ssrc;
-        unsigned int ssrcval;
-        if (mp.Conduit()->GetLocalSSRC(&ssrcval)) {
-          ssrc.AppendInt(ssrcval);
+        std::vector<unsigned int> ssrcvals = mp.Conduit()->GetLocalSSRCs();
+        if (!ssrcvals.empty()) {
+          ssrc.AppendInt(ssrcvals[0]);
         }
         {
           // First, fill in remote stat with rtcp receiver data, if present.
diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.cpp b/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.cpp
index 0d388a8f49..30bcadd2b9 100644
--- a/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.cpp
+++ b/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.cpp
@@ -67,6 +67,14 @@ using namespace dom;
 
 static const char* logTag = "PeerConnectionMedia";
 
+//XXX(pkerr) What about bitrate settings? Going with the defaults for now.
+RefPtr<WebRtcCallWrapper>
+CreateCall()
+{
+  WebRtcCallWrapper::Config call_config;
+  return WebRtcCallWrapper::Create(call_config);
+}
+
 nsresult
 PeerConnectionMedia::ReplaceTrack(const std::string& aOldStreamId,
                                   const std::string& aOldTrackId,
@@ -406,6 +414,9 @@ nsresult PeerConnectionMedia::Init(const std::vector<NrIceStunServer>& stun_serv
   }
   ConnectSignals(mIceCtxHdlr->ctx().get());
 
+  // This webrtc:Call instance will be shared by audio and video media conduits.
+  mCall = CreateCall();
+
   return NS_OK;
 }
 
@@ -567,6 +578,7 @@ nsresult PeerConnectionMedia::UpdateMediaPipelines(
     JsepTrackPair pair = *i;
 
     if (pair.mReceiving) {
+
       rv = factory.CreateOrUpdateMediaPipeline(pair, *pair.mReceiving);
       if (NS_FAILED(rv)) {
         return rv;
diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.h b/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.h
index c0001a5e54..3ecddf8ca3 100644
--- a/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.h
+++ b/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.h
@@ -410,15 +410,15 @@ class PeerConnectionMedia : public sigslot::has_slots<> {
         static_cast<VideoSessionConduit*>(it->second.second.get()));
   }
 
+  void AddVideoConduit(size_t level, const RefPtr<VideoSessionConduit> &aConduit) {
+    mConduits[level] = std::make_pair(true, aConduit);
+  }
+
   // Add a conduit
   void AddAudioConduit(size_t level, const RefPtr<AudioSessionConduit> &aConduit) {
     mConduits[level] = std::make_pair(false, aConduit);
   }
 
-  void AddVideoConduit(size_t level, const RefPtr<VideoSessionConduit> &aConduit) {
-    mConduits[level] = std::make_pair(true, aConduit);
-  }
-
   // ICE state signals
   sigslot::signal2<NrIceCtx*, NrIceCtx::GatheringState>
       SignalIceGatheringStateChange;
@@ -433,6 +433,8 @@ class PeerConnectionMedia : public sigslot::has_slots<> {
   sigslot::signal1<uint16_t>
       SignalEndOfLocalCandidates;
 
+  RefPtr<WebRtcCallWrapper> mCall;
+
  private:
   nsresult InitProxy();
   class ProtocolProxyQueryHandler : public nsIProtocolProxyCallback {
diff --git a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
index f283d61110..eea7c75f90 100644
--- a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
+++ b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
@@ -34,7 +34,7 @@
 #include "runnable_utils.h"
 #include "PeerConnectionCtx.h"
 #include "PeerConnectionImpl.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 static const char* logTag = "WebrtcGlobalInformation";
 
diff --git a/media/webrtc/signaling/test/mediapipeline_unittest.cpp b/media/webrtc/signaling/test/mediapipeline_unittest.cpp
index 33518218b0..14cfa5274e 100644
--- a/media/webrtc/signaling/test/mediapipeline_unittest.cpp
+++ b/media/webrtc/signaling/test/mediapipeline_unittest.cpp
@@ -34,7 +34,7 @@
 #include "mtransport_test_utils.h"
 #include "runnable_utils.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 #include "FakeIPC.h"
 #include "FakeIPC.cpp"
diff --git a/media/webrtc/trunk/AUTHORS b/media/webrtc/trunk/AUTHORS
index 069c3322e2..6f11ee66d8 100644
--- a/media/webrtc/trunk/AUTHORS
+++ b/media/webrtc/trunk/AUTHORS
@@ -1,19 +1,21 @@
 # Names should be added to this file like so:
 # Name or Organization <email address>
 
+Andrew MacDonald <andrew@webrtc.org>
 Anil Kumar <an1kumar@gmail.com>
 Ben Strong <bstrong@gmail.com>
 Bob Withers <bwit@pobox.com>
 Bridger Maxwell <bridgeyman@gmail.com>
 Christophe Dumez <ch.dumez@samsung.com>
 Colin Plumb
-Eric Rescorla, RTFM Inc.
+Eric Rescorla, RTFM Inc. <ekr@rtfm.com>
 Giji Gangadharan <giji.g@samsung.com>
 Graham Yoakum <gyoakum@skobalt.com>
 Jake Hilton <jakehilton@gmail.com>
 James H. Brown <jbrown@burgoyne.com>
+Jiawei Ou <jiawei.ou@gmail.com>
 Jie Mao <maojie0924@gmail.com>
-Luke Weber
+Luke Weber <luke.weber@gmail.com>
 Manish Jethani <manish.jethani@gmail.com>
 Martin Storsjo <martin@martin.st>
 Matthias Liebig <matthias.gcode@gmail.com>
@@ -21,8 +23,8 @@ Pali Rohar
 Paul Kapustin <pkapustin@gmail.com>
 Rafael Lopez Diez <rafalopezdiez@gmail.com>
 Ralph Giles <giles@ghostscript.com>
-Robert Nagy
-Ron Rivest
+Riku Voipio <riku.voipio@linaro.org>
+Robert Nagy <robert.nagy@gmail.com>
 Ryan Yoakum <ryoakum@skobalt.com>
 Sarah Thompson <sarah@telergy.com>
 Saul Kravitz <Saul.Kravitz@celera.com>
@@ -30,13 +32,19 @@ Silviu Caragea <silviu.cpp@gmail.com>
 Steve Reid <sreid@sea-to-sky.net>
 Vicken Simonian <vsimon@gmail.com>
 Victor Costan <costan@gmail.com>
+Alexander Brauckmann <a.brauckmann@gmail.com>
 
+&yet LLC
+Agora IO
 ARM Holdings
+BroadSoft Inc.
 Google Inc.
 Intel Corporation
 MIPS Technologies
 Mozilla Foundation
 Opera Software ASA
+Sinch AB
 struktur AG
+Telenor Digital AS
 Temasys Communications
 Vonage Holdings Corp.
diff --git a/media/webrtc/trunk/DEPS b/media/webrtc/trunk/DEPS
index 891715f7ba..dcf56bc1b6 100644
--- a/media/webrtc/trunk/DEPS
+++ b/media/webrtc/trunk/DEPS
@@ -6,7 +6,7 @@
 vars = {
   'extra_gyp_flag': '-Dextra_gyp_flag=0',
   'chromium_git': 'https://chromium.googlesource.com',
-  'chromium_revision': '719b83983be9613eb80e99a0bc645776d59b76b3',
+  'chromium_revision': '099be58b08dadb64b1dc9f359ae097e978df5416',
 }
 
 # NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
@@ -17,14 +17,14 @@ deps = {
   'src/third_party/gflags/src':
     Var('chromium_git') + '/external/gflags/src@e7390f9185c75f8d902c05ed7d20bb94eb914d0c', # from svn revision 82
 
-  'src/third_party/junit':
+  'src/third_party/junit-jar':
     Var('chromium_git') + '/external/webrtc/deps/third_party/junit@f35596b476aa6e62fd3b3857b9942ddcd13ce35e', # from svn revision 3367
 }
 
 deps_os = {
   'win': {
     'src/third_party/winsdk_samples/src':
-      Var('chromium_git') + '/external/webrtc/deps/third_party/winsdk_samples_v71@c0cbedd854cb610a53226d9817416c4ab9a7d1e9', # from svn revision 7951
+      Var('chromium_git') + '/external/webrtc/deps/third_party/winsdk_samples_v71@e71b549167a665d7424d6f1dadfbff4b4aad1589',
   },
 }
 
@@ -34,18 +34,16 @@ include_rules = [
   # WebRTC production code.
   '-base',
   '-chromium',
+  '+external/webrtc/webrtc',  # Android platform build.
   '+gflags',
   '+libyuv',
   '+net',
   '+talk',
   '+testing',
   '+third_party',
+  '+unicode',
   '+webrtc',
-]
-
-# checkdeps.py shouldn't check include paths for files in these dirs:
-skip_child_includes = [
-  'webrtc/overrides',
+  '+vpx',
 ]
 
 hooks = [
@@ -72,6 +70,21 @@ hooks = [
     'pattern': '.',
     'action': ['python', 'src/setup_links.py'],
   },
+  {
+    # This clobbers when necessary (based on get_landmines.py). It should be
+    # an early hook but it will need to be run after syncing Chromium and
+    # setting up the links, so the script actually exists.
+    'name': 'landmines',
+    'pattern': '.',
+    'action': [
+        'python',
+        'src/build/landmines.py',
+        '--landmine-scripts',
+        'src/webrtc/build/get_landmines.py',
+        '--src-dir',
+        'src',
+    ],
+  },
   {
      # Pull sanitizer-instrumented third-party libraries if requested via
      # GYP_DEFINES. This could be done as part of sync_chromium.py above
@@ -89,6 +102,7 @@ hooks = [
                '--recursive',
                '--num_threads=10',
                '--no_auth',
+               '--quiet',
                '--bucket', 'chromium-webrtc-resources',
                'src/resources'],
   },
diff --git a/media/webrtc/trunk/OWNERS b/media/webrtc/trunk/OWNERS
index c08c55a866..5812db5ca7 100644
--- a/media/webrtc/trunk/OWNERS
+++ b/media/webrtc/trunk/OWNERS
@@ -1,4 +1,3 @@
-andrew@webrtc.org
 henrika@webrtc.org
 mflodman@webrtc.org
 niklas.enbom@webrtc.org
@@ -6,6 +5,7 @@ tina.legrand@webrtc.org
 tommi@webrtc.org
 per-file .gitignore=*
 per-file AUTHORS=*
+per-file .gn=kjellander@webrtc.org
 per-file BUILD.gn=kjellander@webrtc.org
 per-file DEPS=*
 per-file PRESUBMIT.py=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/README b/media/webrtc/trunk/README
deleted file mode 100644
index b7bafb24b6..0000000000
--- a/media/webrtc/trunk/README
+++ /dev/null
@@ -1,12 +0,0 @@
-This folder can be used to pull together the chromium version of webrtc
-and libjingle, and build the peerconnection sample client and server. This will
-check out a new repository in which you can build peerconnection_server.
-
-Steps:
-1) Create a new directory for the new repository (outside the webrtc repo):
-   mkdir peerconnection
-   cd peerconnection
-2) gclient config --name trunk http://webrtc.googlecode.com/svn/trunk/peerconnection
-3) gclient sync
-4) cd trunk
-5) make peerconnection_server peerconnection_client
diff --git a/media/webrtc/trunk/README.md b/media/webrtc/trunk/README.md
new file mode 100644
index 0000000000..a77ab536b5
--- /dev/null
+++ b/media/webrtc/trunk/README.md
@@ -0,0 +1,23 @@
+**WebRTC is a free, open software project** that provides browsers and mobile
+applications with Real-Time Communications (RTC) capabilities via simple APIs.
+The WebRTC components have been optimized to best serve this purpose.
+
+**Our mission:** To enable rich, high-quality RTC applications to be
+developed for the browser, mobile platforms, and IoT devices, and allow them
+all to communicate via a common set of protocols.
+
+The WebRTC initiative is a project supported by Google, Mozilla and Opera,
+amongst others. This page is maintained by the Google Chrome team.
+
+### Development
+
+See http://www.webrtc.org/native-code/development for instructions on how to get
+started developing with the native code.
+
+### More info
+
+ * Official web site: http://www.webrtc.org
+ * Master source code repo: https://chromium.googlesource.com/external/webrtc
+ * Samples and reference apps: https://github.com/webrtc
+ * Mailing list: http://groups.google.com/group/discuss-webrtc
+ * Continuous build: http://build.chromium.org/p/client.webrtc
diff --git a/media/webrtc/trunk/build/common.gypi b/media/webrtc/trunk/build/common.gypi
index 47ad77de48..c03b4427bc 100644
--- a/media/webrtc/trunk/build/common.gypi
+++ b/media/webrtc/trunk/build/common.gypi
@@ -917,23 +917,24 @@
     'android_app_version_name%': 'Developer Build',
     'android_app_version_code%': 0,
 
-    'sas_dll_exists': '<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(sas_dll_path))',
-    'wix_exists': '<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(wix_path))',
+    'sas_dll_exists': 0, # '<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(sas_dll_path))',
+    'wix_exists': 0, # '<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(wix_path))',
 
     'windows_sdk_default_path': '<(DEPTH)/third_party/platformsdk_win8/files',
-    'directx_sdk_default_path': '<(DEPTH)/third_party/directxsdk/files',
+#    'directx_sdk_default_path': '<(DEPTH)/third_party/directxsdk/files',
+    'windows_sdk_path%': '<(windows_sdk_default_path)',
 
     'conditions': [
-      ['"<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(windows_sdk_default_path))"=="True"', {
-        'windows_sdk_path%': '<(windows_sdk_default_path)',
-      }, {
-        'windows_sdk_path%': 'C:/Program Files (x86)/Windows Kits/8.0',
-      }],
-      ['OS=="win" and "<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(directx_sdk_default_path))"=="True"', {
-        'directx_sdk_path%': '<(directx_sdk_default_path)',
-      }, {
-        'directx_sdk_path%': '$(DXSDK_DIR)',
-      }],
+      #['"<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(windows_sdk_default_path))"=="True"', {
+      #  'windows_sdk_path%': '<(windows_sdk_default_path)',
+      #}, {
+      #  'windows_sdk_path%': 'C:/Program Files (x86)/Windows Kits/8.0',
+      #}],
+      #['OS=="win" and "<!(<(PYTHON) <(DEPTH)/build/dir_exists.py <(directx_sdk_default_path))"=="True"', {
+      #  'directx_sdk_path%': '<(directx_sdk_default_path)',
+      #}, {
+      #  'directx_sdk_path%': '$(DXSDK_DIR)',
+      #}],
       # If use_official_google_api_keys is already set (to 0 or 1), we
       # do none of the implicit checking.  If it is set to 1 and the
       # internal keys file is missing, the build will fail at compile
@@ -3410,7 +3411,7 @@
           '<(windows_sdk_path)/Include/shared',
           '<(windows_sdk_path)/Include/um',
           '<(windows_sdk_path)/Include/winrt',
-          '<(directx_sdk_path)/Include',
+#          '<(directx_sdk_path)/Include',
           '$(VSInstallDir)/VC/atlmfc/include',
         ],
         'msvs_cygwin_dirs': ['<(DEPTH)/third_party/cygwin'],
@@ -3442,7 +3443,7 @@
           'VCLibrarianTool': {
             'AdditionalOptions': ['/ignore:4221'],
             'AdditionalLibraryDirectories': [
-              '<(directx_sdk_path)/Lib/x86',
+#              '<(directx_sdk_path)/Lib/x86',
               '<(windows_sdk_path)/Lib/win8/um/x86',
             ],
           },
@@ -3488,7 +3489,7 @@
               }],
             ],
             'AdditionalLibraryDirectories': [
-              '<(directx_sdk_path)/Lib/x86',
+#              '<(directx_sdk_path)/Lib/x86', XXXX
               '<(windows_sdk_path)/Lib/win8/um/x86',
             ],
             'GenerateDebugInformation': 'true',
diff --git a/media/webrtc/trunk/peerconnection.gyp b/media/webrtc/trunk/peerconnection.gyp
index f8309b90e5..ac1fae99e7 100644
--- a/media/webrtc/trunk/peerconnection.gyp
+++ b/media/webrtc/trunk/peerconnection.gyp
@@ -7,10 +7,13 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
-  'includes': [ 'webrtc/build/common.gypi', ],
+  'includes': [
+    'webrtc/build/common.gypi',
+    'webrtc/video/webrtc_video.gypi',
+  ],
   'variables': {
     'peerconnection_sample': 'third_party/libjingle/source/talk/examples/peerconnection',
-  },  
+  },
 
   # for mozilla, we want to force stuff to build but we don't want peerconnection_client or server
   # for unknown reasons, 'targets' must be outside of conditions.  And don't try to build a dummy
@@ -22,19 +25,21 @@
         'target_name': 'dummy',
         'type': 'none',
         'dependencies': [
+	  'webrtc/webrtc.gyp:webrtc_lib',
           'webrtc/modules/modules.gyp:audio_device',
           'webrtc/modules/modules.gyp:video_capture_module',
-	  'webrtc/modules/modules.gyp:video_capture_module_internal_impl',
+          'webrtc/modules/modules.gyp:video_capture_module_internal_impl',
           'webrtc/modules/modules.gyp:video_render',
 #          'webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-#	  'webrtc/system_wrappers/source/system_wrappers.gyp:metrics_default',
-          'webrtc/video_engine/video_engine.gyp:video_engine_core',
-          'webrtc/voice_engine/voice_engine.gyp:voice_engine',
+#      'webrtc/system_wrappers/source/system_wrappers.gyp:metrics_default',
+#         'webrtc/video_engine/video_engine.gyp:video_engine_core',
+         'webrtc/voice_engine/voice_engine.gyp:voice_engine',
+#          '<@(webrtc_video_dependencies)',
         ],
         'conditions': [
           ['OS!="android" and OS!="ios"', {
             'dependencies': [
-              'webrtc/modules/modules.gyp:desktop_capture',
+               'webrtc/modules/modules.gyp:desktop_capture',
             ],
           },
          ]],
@@ -86,8 +91,8 @@
             },
           },
           'dependencies': [
-            'third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
-            'third_party/libjingle/libjingle.gyp:libjingle_peerconnection',
+            #'third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+            #'third_party/libjingle/libjingle.gyp:libjingle_peerconnection',
           ],
           'include_dirs': [
             'src',
@@ -114,8 +119,8 @@
             '<(peerconnection_sample)/client/peer_connection_client.h',
           ],
           'dependencies': [
-            'third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
-            'third_party/libjingle/libjingle.gyp:libjingle_peerconnection',
+            #'third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+            #'third_party/libjingle/libjingle.gyp:libjingle_peerconnection',
             # TODO(tommi): Switch to this and remove specific gtk dependency
             # sections below for cflags and link_settings.
             # '<(DEPTH)/build/linux/system.gyp:gtk',
diff --git a/media/webrtc/trunk/tools/gyp/pylib/gyp/__init__.py b/media/webrtc/trunk/tools/gyp/pylib/gyp/__init__.py
index b26977592b..759e7b7bf7 100644
--- a/media/webrtc/trunk/tools/gyp/pylib/gyp/__init__.py
+++ b/media/webrtc/trunk/tools/gyp/pylib/gyp/__init__.py
@@ -46,10 +46,13 @@ def FindBuildFiles():
       build_files.append(file)
   return build_files
 
-
+#TODO @@NG Find correct way to pass circular_check to Load
+# Causes a pickle error in gyp.input.CircularException
+# see http://stackoverflow.com/questions/4677012/python-cant-pickle-type-x-attribute-lookup-failed
+# we'll live without the check - jesup
 def Load(build_files, format, default_variables={},
          includes=[], depth='.', params=None, check=False,
-         circular_check=True, duplicate_basename_check=True):
+         circular_check=False, duplicate_basename_check=True):
   """
   Loads one or more specified build files.
   default_variables and includes will be copied before use.
diff --git a/media/webrtc/trunk/webrtc/.gitignore b/media/webrtc/trunk/webrtc/.gitignore
index 9c8d698f37..b5abe8ef68 100644
--- a/media/webrtc/trunk/webrtc/.gitignore
+++ b/media/webrtc/trunk/webrtc/.gitignore
@@ -1,2 +1,29 @@
-# This file is for projects that checkout webrtc/ directly (e.g. Chromium).
-*.mk
+# This file is for projects that checkout webrtc/ directly (e.g. Chromium). It
+# is a truncated copy of the .gitignore file in the parent directory.
+*.DS_Store
+*.Makefile
+*.host.mk
+*.ncb
+*.ninja
+*.props
+*.pyc
+*.rules
+*.scons
+*.sdf
+*.sln
+*.suo
+*.target.mk
+*.targets
+*.user
+*.vcproj
+*.vcxproj
+*.vcxproj.filters
+*.vpj
+*.vpw
+*.vpwhistu
+*.vtg
+*.xcodeproj
+*_proto.xml
+*_proto_cpp.xml
+*~
+.*.sw?
diff --git a/media/webrtc/trunk/webrtc/BUILD.gn b/media/webrtc/trunk/webrtc/BUILD.gn
index 893dd9f39a..be824b7bc7 100644
--- a/media/webrtc/trunk/webrtc/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/BUILD.gn
@@ -8,9 +8,10 @@
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
-import("//build/config/crypto.gni")
 import("//build/config/linux/pkg_config.gni")
+import("//build/config/sanitizers/sanitizers.gni")
 import("build/webrtc.gni")
+import("//third_party/protobuf/proto_library.gni")
 
 # Contains the defines and includes in common.gypi that are duplicated both as
 # target_defaults and direct_dependent_settings.
@@ -20,13 +21,12 @@ config("common_inherited_config") {
     defines += [ "WEBRTC_MOZILLA_BUILD" ]
   }
   if (build_with_chromium) {
-    defines = [
-      "WEBRTC_CHROMIUM_BUILD",
-    ]
+    defines = [ "WEBRTC_CHROMIUM_BUILD" ]
     include_dirs = [
-      # overrides must be included first as that is the mechanism for
+      # The overrides must be included first as that is the mechanism for
       # selecting the override headers in Chromium.
-      "overrides",
+      "../webrtc_overrides",
+
       # Allow includes to be prefixed with webrtc/ in case it is not an
       # immediate subdirectory of the top-level.
       "..",
@@ -41,6 +41,9 @@ config("common_inherited_config") {
       "WEBRTC_IOS",
     ]
   }
+  if (is_ios && rtc_use_objc_h264) {
+    defines += [ "WEBRTC_OBJC_H264" ]
+  }
   if (is_linux) {
     defines += [ "WEBRTC_LINUX" ]
   }
@@ -55,9 +58,6 @@ config("common_inherited_config") {
       "WEBRTC_LINUX",
       "WEBRTC_ANDROID",
     ]
-    if (rtc_enable_android_opensl) {
-      defines += [ "WEBRTC_ANDROID_OPENSLES" ]
-    }
   }
 }
 
@@ -76,6 +76,7 @@ config("common_config") {
 
   if (rtc_have_dbus_glib) {
     defines += [ "HAVE_DBUS_GLIB" ]
+
     # TODO(kjellander): Investigate this, it seems like include <dbus/dbus.h>
     # is still not found even if the execution of
     # build/config/linux/pkg-config.py dbus-glib-1 returns correct include
@@ -96,6 +97,7 @@ config("common_config") {
       if (current_cpu != "arm64" || !is_android) {
         cflags = [
           "-Wextra",
+
           # We need to repeat some flags from Chromium"s common.gypi
           # here that get overridden by -Wextra.
           "-Wno-unused-parameter",
@@ -104,6 +106,7 @@ config("common_config") {
         ]
         cflags_cc = [
           "-Wnon-virtual-dtor",
+
           # This is enabled for clang; enable for gcc as well.
           "-Woverloaded-virtual",
         ]
@@ -116,13 +119,8 @@ config("common_config") {
   }
 
   if (current_cpu == "arm64") {
-    defines += [ "WEBRTC_ARCH_ARM" ]
-    # TODO(zhongwei) Defining an unique WEBRTC_NEON and
-    # distinguishing ARMv7 NEON and ARM64 NEON by
-    # WEBRTC_ARCH_ARM_V7 and WEBRTC_ARCH_ARM64 should be better.
-
-    # This macro is used to distinguish ARMv7 NEON and ARM64 NEON
-    defines += [ "WEBRTC_ARCH_ARM64_NEON" ]
+    defines += [ "WEBRTC_ARCH_ARM64" ]
+    defines += [ "WEBRTC_HAS_NEON" ]
   }
 
   if (current_cpu == "arm") {
@@ -130,9 +128,9 @@ config("common_config") {
     if (arm_version >= 7) {
       defines += [ "WEBRTC_ARCH_ARM_V7" ]
       if (arm_use_neon) {
-        defines += [ "WEBRTC_ARCH_ARM_NEON" ]
-      } else if (is_android) {
-        defines += [ "WEBRTC_DETECT_ARM_NEON" ]
+        defines += [ "WEBRTC_HAS_NEON" ]
+      } else if (arm_optionally_use_neon) {
+        defines += [ "WEBRTC_DETECT_NEON" ]
       }
     }
   }
@@ -155,10 +153,7 @@ config("common_config") {
     }
   }
 
-  # TODO(kjellander): Handle warnings on Windows where WebRTC differ from the
-  # default warnings set in build/config/compiler/BUILD.gn.
-
-  if (is_android && is_clang) {
+  if (is_android && !is_clang) {
     # The Android NDK doesn"t provide optimized versions of these
     # functions. Ensure they are disabled for all compilers.
     cflags += [
@@ -174,17 +169,19 @@ source_set("webrtc") {
   sources = [
     "call.h",
     "config.h",
-    "experiments.h",
     "frame_callback.h",
     "transport.h",
   ]
 
+  defines = []
   configs += [ ":common_config" ]
-  public_configs = [ ":common_inherited_config"]
+  public_configs = [ ":common_inherited_config" ]
 
   deps = [
     ":webrtc_common",
+    "audio",
     "base:rtc_base",
+    "call",
     "common_audio",
     "common_video",
     "modules/audio_coding",
@@ -201,16 +198,20 @@ source_set("webrtc") {
     "system_wrappers",
     "tools",
     "video",
-    "video_engine",
     "voice_engine",
   ]
 
   if (build_with_chromium) {
     deps += [
-    "modules/video_capture",
-    "modules/video_render",
+      "modules/video_capture",
+      "modules/video_render",
     ]
   }
+
+  if (rtc_enable_protobuf) {
+    defines += [ "ENABLE_RTC_EVENT_LOG" ]
+    deps += [ ":rtc_event_log_proto" ]
+  }
 }
 
 if (!build_with_chromium) {
@@ -218,8 +219,8 @@ if (!build_with_chromium) {
     testonly = true
     deps = [
       ":webrtc",
-      "modules/video_render:video_render_internal_impl",
       "modules/video_capture:video_capture_internal_impl",
+      "modules/video_render:video_render_internal_impl",
       "test",
     ]
   }
@@ -229,14 +230,20 @@ source_set("webrtc_common") {
   sources = [
     "common_types.cc",
     "common_types.h",
-    "config.h",
     "config.cc",
+    "config.h",
     "engine_configurations.h",
     "typedefs.h",
   ]
 
   configs += [ ":common_config" ]
   public_configs = [ ":common_inherited_config" ]
+
+  if (is_clang && !is_nacl) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
 }
 
 source_set("gtest_prod") {
@@ -244,3 +251,48 @@ source_set("gtest_prod") {
     "test/testsupport/gtest_prod_util.h",
   ]
 }
+
+if (rtc_enable_protobuf) {
+  proto_library("rtc_event_log_proto") {
+    sources = [
+      "call/rtc_event_log.proto",
+    ]
+    proto_out_dir = "webrtc/call"
+  }
+}
+
+source_set("rtc_event_log") {
+  sources = [
+    "call/rtc_event_log.cc",
+    "call/rtc_event_log.h",
+  ]
+
+  defines = []
+  configs += [ ":common_config" ]
+  public_configs = [ ":common_inherited_config" ]
+
+  deps = [
+    ":webrtc_common",
+  ]
+
+  if (rtc_enable_protobuf) {
+    defines += [ "ENABLE_RTC_EVENT_LOG" ]
+    deps += [ ":rtc_event_log_proto" ]
+  }
+  if (is_clang && !is_nacl) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+if (use_libfuzzer || use_drfuzz) {
+  # This target is only here for gn to discover fuzzer build targets under
+  # webrtc/test/fuzzers/.
+  group("webrtc_fuzzers_dummy") {
+    testonly = true
+    deps = [
+      "test/fuzzers:webrtc_fuzzer_main",
+    ]
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/api/BUILD.gn b/media/webrtc/trunk/webrtc/api/BUILD.gn
new file mode 100644
index 0000000000..7cfa083a6b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/BUILD.gn
@@ -0,0 +1,76 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+
+config("ios_config") {
+  libs = [
+    "CoreGraphics.framework",
+    "GLKit.framework",
+    "OpenGLES.framework",
+    "QuartzCore.framework",
+  ]
+}
+
+if (is_ios) {
+  source_set("rtc_api_objc") {
+    deps = [
+      "//webrtc/base:rtc_base_objc",
+      #"//talk/libjingle:libjingle_peerconnection",
+    ]
+    cflags = [
+      "-fobjc-arc",
+      "-Wobjc-missing-property-synthesis",
+    ]
+    sources = [
+      # Add these when there's a BUILD.gn for peer connection APIs
+      #"objc/RTCIceCandidate+Private.h",
+      #"objc/RTCIceCandidate.h",
+      #"objc/RTCIceCandidate.mm",
+      #"objc/RTCMediaSource+Private.h",
+      #"objc/RTCMediaSource.h",
+      #"objc/RTCMediaSource.mm",
+      #"objc/RTCMediaStreamTrack+Private.h",
+      #"objc/RTCMediaStreamTrack.h",
+      #"objc/RTCMediaStreamTrack.mm",
+      "objc/RTCIceServer+Private.h",
+      "objc/RTCIceServer.h",
+      "objc/RTCIceServer.mm",
+      "objc/RTCMediaConstraints+Private.h",
+      "objc/RTCMediaConstraints.h",
+      "objc/RTCMediaConstraints.mm",
+      "objc/RTCOpenGLVideoRenderer.h",
+      "objc/RTCOpenGLVideoRenderer.mm",
+      "objc/RTCSessionDescription+Private.h",
+      "objc/RTCSessionDescription.h",
+      "objc/RTCSessionDescription.mm",
+      "objc/RTCStatsReport+Private.h",
+      "objc/RTCStatsReport.h",
+      "objc/RTCStatsReport.mm",
+      "objc/RTCVideoFrame+Private.h",
+      "objc/RTCVideoFrame.h",
+      "objc/RTCVideoFrame.mm",
+      "objc/RTCVideoRenderer.h",
+      "objc/WebRTC-Prefix.pch",
+    ]
+
+    if (is_ios) {
+      sources += [
+        "objc/RTCEAGLVideoView.h",
+        "objc/RTCEAGLVideoView.m",
+      ]
+    }
+
+    if (is_mac) {
+      sources += [
+        "objc/RTCNSGLVideoView.h",
+        "objc/RTCNSGLVideoView.m",
+      ]
+    }
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/api/OWNERS b/media/webrtc/trunk/webrtc/api/OWNERS
new file mode 100644
index 0000000000..cd06158b7f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/OWNERS
@@ -0,0 +1 @@
+tkchin@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/api/api.gyp b/media/webrtc/trunk/webrtc/api/api.gyp
new file mode 100644
index 0000000000..ba3fe8d0bd
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/api.gyp
@@ -0,0 +1,83 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [ '../build/common.gypi', ],
+  'conditions': [
+    ['OS=="ios"', {
+      'targets': [
+        {
+          'target_name': 'rtc_api_objc',
+          'type': 'static_library',
+          'dependencies': [
+            '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+            '../../talk/libjingle.gyp:libjingle_peerconnection',
+          ],
+          'sources': [
+            'objc/RTCIceCandidate+Private.h',
+            'objc/RTCIceCandidate.h',
+            'objc/RTCIceCandidate.mm',
+            'objc/RTCIceServer+Private.h',
+            'objc/RTCIceServer.h',
+            'objc/RTCIceServer.mm',
+            'objc/RTCMediaConstraints+Private.h',
+            'objc/RTCMediaConstraints.h',
+            'objc/RTCMediaConstraints.mm',
+            'objc/RTCMediaSource+Private.h',
+            'objc/RTCMediaSource.h',
+            'objc/RTCMediaSource.mm',
+            'objc/RTCMediaStreamTrack+Private.h',
+            'objc/RTCMediaStreamTrack.h',
+            'objc/RTCMediaStreamTrack.mm',
+            'objc/RTCOpenGLVideoRenderer.h',
+            'objc/RTCOpenGLVideoRenderer.mm',
+            'objc/RTCSessionDescription+Private.h',
+            'objc/RTCSessionDescription.h',
+            'objc/RTCSessionDescription.mm',
+            'objc/RTCStatsReport+Private.h',
+            'objc/RTCStatsReport.h',
+            'objc/RTCStatsReport.mm',
+            'objc/RTCVideoFrame+Private.h',
+            'objc/RTCVideoFrame.h',
+            'objc/RTCVideoFrame.mm',
+            'objc/RTCVideoRenderer.h',
+          ],
+          'conditions': [
+            ['OS=="ios"', {
+              'sources': [
+                'objc/RTCEAGLVideoView.h',
+                'objc/RTCEAGLVideoView.m',
+              ],
+              'all_dependent_settings': {
+                'xcode_settings': {
+                  'OTHER_LDFLAGS': [
+                    '-framework CoreGraphics',
+                    '-framework GLKit',
+                    '-framework OpenGLES',
+                    '-framework QuartzCore',
+                  ]
+                }
+              }
+            }],
+            ['OS=="mac"', {
+              'sources': [
+                'objc/RTCNSGLVideoView.h',
+                'objc/RTCNSGLVideoView.m',
+              ],
+            }],
+          ],
+          'xcode_settings': {
+            'CLANG_ENABLE_OBJC_ARC': 'YES',
+            'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
+            'GCC_PREFIX_HEADER': 'objc/WebRTC-Prefix.pch',
+          },
+        }
+      ],
+    }], # OS=="ios"
+  ],
+}
diff --git a/media/webrtc/trunk/webrtc/api/api_tests.gyp b/media/webrtc/trunk/webrtc/api/api_tests.gyp
new file mode 100644
index 0000000000..c2c18bc693
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/api_tests.gyp
@@ -0,0 +1,40 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [ '../build/common.gypi', ],
+  'conditions': [
+    ['OS=="ios"', {
+      'targets': [
+        {
+          'target_name': 'rtc_api_objc_test',
+          'type': 'executable',
+          'dependencies': [
+            '<(webrtc_root)/api/api.gyp:rtc_api_objc',
+            '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
+          ],
+          'sources': [
+            'objctests/RTCIceCandidateTest.mm',
+            'objctests/RTCIceServerTest.mm',
+            'objctests/RTCMediaConstraintsTest.mm',
+            'objctests/RTCSessionDescriptionTest.mm',
+          ],
+          'xcode_settings': {
+            'CLANG_ENABLE_OBJC_ARC': 'YES',
+            'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
+            'GCC_PREFIX_HEADER': 'objc/WebRTC-Prefix.pch',
+            # |-ObjC| flag needed to make sure category method implementations
+            # are included:
+            # https://developer.apple.com/library/mac/qa/qa1490/_index.html
+            'OTHER_LDFLAGS': ['-ObjC'],
+          },
+        }
+      ],
+    }], # OS=="ios"
+  ],
+}
diff --git a/media/webrtc/trunk/webrtc/api/objc/OWNERS b/media/webrtc/trunk/webrtc/api/objc/OWNERS
new file mode 100644
index 0000000000..cd06158b7f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/OWNERS
@@ -0,0 +1 @@
+tkchin@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/api/objc/README b/media/webrtc/trunk/webrtc/api/objc/README
new file mode 100644
index 0000000000..bd33e61921
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/README
@@ -0,0 +1,3 @@
+This is a work-in-progress to update the Objective-C API according to the W3C
+specification. The Objective-C API located at talk/app/webrtc/objc is
+deprecated, but will remain for the time being.
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.h b/media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.h
new file mode 100644
index 0000000000..1a57df76bb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCEAGLVideoView;
+@protocol RTCEAGLVideoViewDelegate
+
+- (void)videoView:(RTCEAGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
+
+@end
+
+/**
+ * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames in its
+ * bounds using OpenGLES 2.0.
+ */
+@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
+
+@property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.m b/media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.m
new file mode 100644
index 0000000000..e664ede455
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.m
@@ -0,0 +1,259 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCEAGLVideoView.h"
+
+#import <GLKit/GLKit.h>
+
+#import "RTCVideoFrame.h"
+#import "RTCOpenGLVideoRenderer.h"
+
+// RTCDisplayLinkTimer wraps a CADisplayLink and is set to fire every two screen
+// refreshes, which should be 30fps. We wrap the display link in order to avoid
+// a retain cycle since CADisplayLink takes a strong reference onto its target.
+// The timer is paused by default.
+@interface RTCDisplayLinkTimer : NSObject
+
+@property(nonatomic) BOOL isPaused;
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler;
+- (void)invalidate;
+
+@end
+
+@implementation RTCDisplayLinkTimer {
+  CADisplayLink *_displayLink;
+  void (^_timerHandler)(void);
+}
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler {
+  NSParameterAssert(timerHandler);
+  if (self = [super init]) {
+    _timerHandler = timerHandler;
+    _displayLink =
+        [CADisplayLink displayLinkWithTarget:self
+                                    selector:@selector(displayLinkDidFire:)];
+    _displayLink.paused = YES;
+    // Set to half of screen refresh, which should be 30fps.
+    [_displayLink setFrameInterval:2];
+    [_displayLink addToRunLoop:[NSRunLoop currentRunLoop]
+                       forMode:NSRunLoopCommonModes];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self invalidate];
+}
+
+- (BOOL)isPaused {
+  return _displayLink.paused;
+}
+
+- (void)setIsPaused:(BOOL)isPaused {
+  _displayLink.paused = isPaused;
+}
+
+- (void)invalidate {
+  [_displayLink invalidate];
+}
+
+- (void)displayLinkDidFire:(CADisplayLink *)displayLink {
+  _timerHandler();
+}
+
+@end
+
+// RTCEAGLVideoView wraps a GLKView which is setup with
+// enableSetNeedsDisplay = NO for the purpose of gaining control of
+// exactly when to call -[GLKView display]. This need for extra
+// control is required to avoid triggering method calls on GLKView
+// that results in attempting to bind the underlying render buffer
+// when the drawable size would be empty which would result in the
+// error GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT. -[GLKView display] is
+// the method that will trigger the binding of the render
+// buffer. Because the standard behaviour of -[UIView setNeedsDisplay]
+// is disabled for the reasons above, the RTCEAGLVideoView maintains
+// its own |isDirty| flag.
+
+@interface RTCEAGLVideoView () <GLKViewDelegate>
+// |videoFrame| is set when we receive a frame from a worker thread and is read
+// from the display link callback so atomicity is required.
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@property(nonatomic, readonly) GLKView *glkView;
+@property(nonatomic, readonly) RTCOpenGLVideoRenderer *glRenderer;
+@end
+
+@implementation RTCEAGLVideoView {
+  RTCDisplayLinkTimer *_timer;
+  // This flag should only be set and read on the main thread (e.g. by
+  // setNeedsDisplay)
+  BOOL _isDirty;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize glkView = _glkView;
+@synthesize glRenderer = _glRenderer;
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  if (self = [super initWithFrame:frame]) {
+    [self configure];
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder {
+  if (self = [super initWithCoder:aDecoder]) {
+    [self configure];
+  }
+  return self;
+}
+
+- (void)configure {
+  EAGLContext *glContext =
+    [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES3];
+  if (!glContext) {
+    glContext = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];
+  }
+  _glRenderer = [[RTCOpenGLVideoRenderer alloc] initWithContext:glContext];
+
+  // GLKView manages a framebuffer for us.
+  _glkView = [[GLKView alloc] initWithFrame:CGRectZero
+                                    context:glContext];
+  _glkView.drawableColorFormat = GLKViewDrawableColorFormatRGBA8888;
+  _glkView.drawableDepthFormat = GLKViewDrawableDepthFormatNone;
+  _glkView.drawableStencilFormat = GLKViewDrawableStencilFormatNone;
+  _glkView.drawableMultisample = GLKViewDrawableMultisampleNone;
+  _glkView.delegate = self;
+  _glkView.layer.masksToBounds = YES;
+  _glkView.enableSetNeedsDisplay = NO;
+  [self addSubview:_glkView];
+
+  // Listen to application state in order to clean up OpenGL before app goes
+  // away.
+  NSNotificationCenter *notificationCenter =
+    [NSNotificationCenter defaultCenter];
+  [notificationCenter addObserver:self
+                         selector:@selector(willResignActive)
+                             name:UIApplicationWillResignActiveNotification
+                           object:nil];
+  [notificationCenter addObserver:self
+                         selector:@selector(didBecomeActive)
+                             name:UIApplicationDidBecomeActiveNotification
+                           object:nil];
+
+  // Frames are received on a separate thread, so we poll for current frame
+  // using a refresh rate proportional to screen refresh frequency. This
+  // occurs on the main thread.
+  __weak RTCEAGLVideoView *weakSelf = self;
+  _timer = [[RTCDisplayLinkTimer alloc] initWithTimerHandler:^{
+      RTCEAGLVideoView *strongSelf = weakSelf;
+      [strongSelf displayLinkTimerDidFire];
+    }];
+  [self setupGL];
+}
+
+- (void)dealloc {
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+  UIApplicationState appState =
+      [UIApplication sharedApplication].applicationState;
+  if (appState == UIApplicationStateActive) {
+    [self teardownGL];
+  }
+  [_timer invalidate];
+}
+
+#pragma mark - UIView
+
+- (void)setNeedsDisplay {
+  [super setNeedsDisplay];
+  _isDirty = YES;
+}
+
+- (void)setNeedsDisplayInRect:(CGRect)rect {
+  [super setNeedsDisplayInRect:rect];
+  _isDirty = YES;
+}
+
+- (void)layoutSubviews {
+  [super layoutSubviews];
+  _glkView.frame = self.bounds;
+}
+
+#pragma mark - GLKViewDelegate
+
+// This method is called when the GLKView's content is dirty and needs to be
+// redrawn. This occurs on main thread.
+- (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
+  // The renderer will draw the frame to the framebuffer corresponding to the
+  // one used by |view|.
+  [_glRenderer drawFrame:self.videoFrame];
+}
+
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+  __weak RTCEAGLVideoView *weakSelf = self;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    RTCEAGLVideoView *strongSelf = weakSelf;
+    [strongSelf.delegate videoView:strongSelf didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(RTCVideoFrame *)frame {
+  self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)displayLinkTimerDidFire {
+  // Don't render unless video frame have changed or the view content
+  // has explicitly been marked dirty.
+  if (!_isDirty && _glRenderer.lastDrawnFrame == self.videoFrame) {
+    return;
+  }
+
+  // Always reset isDirty at this point, even if -[GLKView display]
+  // won't be called in the case the drawable size is empty.
+  _isDirty = NO;
+
+  // Only call -[GLKView display] if the drawable size is
+  // non-empty. Calling display will make the GLKView setup its
+  // render buffer if necessary, but that will fail with error
+  // GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT if size is empty.
+  if (self.bounds.size.width > 0 && self.bounds.size.height > 0) {
+    [_glkView display];
+  }
+}
+
+- (void)setupGL {
+  self.videoFrame = nil;
+  [_glRenderer setupGL];
+  _timer.isPaused = NO;
+}
+
+- (void)teardownGL {
+  self.videoFrame = nil;
+  _timer.isPaused = YES;
+  [_glkView deleteDrawable];
+  [_glRenderer teardownGL];
+}
+
+- (void)didBecomeActive {
+  [self setupGL];
+}
+
+- (void)willResignActive {
+  [self teardownGL];
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate+Private.h
new file mode 100644
index 0000000000..ca95a43e3a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate+Private.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate.h"
+
+#include "talk/app/webrtc/jsep.h"
+#include "webrtc/base/scoped_ptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceCandidate ()
+
+/**
+ * The native IceCandidateInterface representation of this RTCIceCandidate
+ * object. This is needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate;
+
+/**
+ * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
+ * ownership is taken of the native candidate.
+ */
+- (instancetype)initWithNativeCandidate:
+    (webrtc::IceCandidateInterface *)candidate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.h b/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.h
new file mode 100644
index 0000000000..41ea69e991
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceCandidate : NSObject
+
+/**
+ * If present, the identifier of the "media stream identification" for the media
+ * component this candidate is associated with.
+ */
+@property(nonatomic, readonly, nullable) NSString *sdpMid;
+
+/**
+ * The index (starting at zero) of the media description this candidate is
+ * associated with in the SDP.
+ */
+@property(nonatomic, readonly) NSInteger sdpMLineIndex;
+
+/** The SDP string for this candidate. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCIceCandidate from SDP.
+ */
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(NSInteger)sdpMLineIndex
+                     sdpMid:(nullable NSString *)sdpMid
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.mm b/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.mm
new file mode 100644
index 0000000000..9e094f6f06
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.mm
@@ -0,0 +1,70 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate.h"
+
+#import "webrtc/api/objc/RTCIceCandidate+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+#import "webrtc/base/objc/RTCLogging.h"
+
+@implementation RTCIceCandidate
+
+@synthesize sdpMid = _sdpMid;
+@synthesize sdpMLineIndex = _sdpMLineIndex;
+@synthesize sdp = _sdp;
+
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(NSInteger)sdpMLineIndex
+                     sdpMid:(NSString *)sdpMid {
+  NSParameterAssert(sdp.length);
+  if (self = [super init]) {
+    _sdpMid = [sdpMid copy];
+    _sdpMLineIndex = sdpMLineIndex;
+    _sdp = [sdp copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCIceCandidate:\n%@\n%ld\n%@",
+                                    _sdpMid,
+                                    (long)_sdpMLineIndex,
+                                    _sdp];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeCandidate:
+    (webrtc::IceCandidateInterface *)candidate {
+  NSParameterAssert(candidate);
+  std::string sdp;
+  candidate->ToString(&sdp);
+
+  return [self initWithSdp:[NSString stringForStdString:sdp]
+             sdpMLineIndex:candidate->sdp_mline_index()
+                    sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
+}
+
+- (rtc::scoped_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
+  webrtc::SdpParseError error;
+
+  webrtc::IceCandidateInterface *candidate = webrtc::CreateIceCandidate(
+      _sdpMid.stdString, _sdpMLineIndex, _sdp.stdString, &error);
+
+  if (!candidate) {
+    RTCLog(@"Failed to create ICE candidate: %s\nline: %s",
+           error.description.c_str(),
+           error.line.c_str());
+  }
+
+  return rtc::scoped_ptr<webrtc::IceCandidateInterface>(candidate);
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCIceServer+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCIceServer+Private.h
new file mode 100644
index 0000000000..59f5a92dff
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCIceServer+Private.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer.h"
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceServer ()
+
+/**
+ * IceServer struct representation of this RTCIceServer object's data.
+ * This is needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+    webrtc::PeerConnectionInterface::IceServer iceServer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCIceServer.h b/media/webrtc/trunk/webrtc/api/objc/RTCIceServer.h
new file mode 100644
index 0000000000..be4e0d7b6e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCIceServer.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceServer : NSObject
+
+/** URI(s) for this server represented as NSStrings. */
+@property(nonatomic, copy, readonly) NSArray<NSString *> *urlStrings;
+
+/** Username to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, copy, readonly, nullable) NSString *username;
+
+/** Credential to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, copy, readonly, nullable) NSString *credential;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Convenience initializer for a server with no authentication (e.g. STUN). */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and credentialType.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCIceServer.mm b/media/webrtc/trunk/webrtc/api/objc/RTCIceServer.mm
new file mode 100644
index 0000000000..7a898e06d5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCIceServer.mm
@@ -0,0 +1,64 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer.h"
+
+#import "webrtc/api/objc/RTCIceServer+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@implementation RTCIceServer
+
+@synthesize urlStrings = _urlStrings;
+@synthesize username = _username;
+@synthesize credential = _credential;
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings {
+  NSParameterAssert(urlStrings.count);
+  return [self initWithURLStrings:urlStrings
+                         username:nil
+                       credential:nil];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential {
+  NSParameterAssert(urlStrings.count);
+  if (self = [super init]) {
+    _urlStrings = [[NSArray alloc] initWithArray:urlStrings copyItems:YES];
+    _username = [username copy];
+    _credential = [credential copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCIceServer:\n%@\n%@\n%@",
+                                    _urlStrings,
+                                    _username,
+                                    _credential];
+}
+
+#pragma mark - Private
+
+- (webrtc::PeerConnectionInterface::IceServer)iceServer {
+  __block webrtc::PeerConnectionInterface::IceServer iceServer;
+
+  iceServer.username = [NSString stdStringForString:_username];
+  iceServer.password = [NSString stdStringForString:_credential];
+
+  [_urlStrings enumerateObjectsUsingBlock:^(NSString *url,
+                                            NSUInteger idx,
+                                            BOOL *stop) {
+    iceServer.urls.push_back(url.stdString);
+  }];
+  return iceServer;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints+Private.h
new file mode 100644
index 0000000000..2c4b722104
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints+Private.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaConstraints.h"
+
+#include "talk/app/webrtc/mediaconstraintsinterface.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+class MediaConstraints : public MediaConstraintsInterface {
+ public:
+  virtual ~MediaConstraints();
+  MediaConstraints();
+  MediaConstraints(
+      const MediaConstraintsInterface::Constraints& mandatory,
+      const MediaConstraintsInterface::Constraints& optional);
+  virtual const Constraints& GetMandatory() const;
+  virtual const Constraints& GetOptional() const;
+
+ private:
+  MediaConstraintsInterface::Constraints mandatory_;
+  MediaConstraintsInterface::Constraints optional_;
+};
+
+}  // namespace webrtc
+
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaConstraints ()
+
+/**
+ * A MediaConstraints representation of this RTCMediaConstraints object. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+- (rtc::scoped_ptr<webrtc::MediaConstraints>)nativeConstraints;
+
+/** Return a native Constraints object representing these constraints */
++ (webrtc::MediaConstraintsInterface::Constraints)
+    nativeConstraintsForConstraints:
+        (NSDictionary<NSString *, NSString *> *)constraints;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.h b/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.h
new file mode 100644
index 0000000000..a8ad39142e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaConstraints : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize with mandatory and/or optional constraints. */
+- (instancetype)initWithMandatoryConstraints:
+    (nullable NSDictionary<NSString *, NSString *> *)mandatory
+                         optionalConstraints:
+    (nullable NSDictionary<NSString *, NSString *> *)optional
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.mm b/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.mm
new file mode 100644
index 0000000000..a53a517747
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.mm
@@ -0,0 +1,92 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaConstraints.h"
+
+#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+namespace webrtc {
+
+MediaConstraints::~MediaConstraints() {}
+
+MediaConstraints::MediaConstraints() {}
+
+MediaConstraints::MediaConstraints(
+    const MediaConstraintsInterface::Constraints& mandatory,
+    const MediaConstraintsInterface::Constraints& optional)
+    : mandatory_(mandatory), optional_(optional) {}
+
+const MediaConstraintsInterface::Constraints&
+MediaConstraints::GetMandatory() const {
+  return mandatory_;
+}
+
+const MediaConstraintsInterface::Constraints&
+MediaConstraints::GetOptional() const {
+  return optional_;
+}
+
+}  // namespace webrtc
+
+
+@implementation RTCMediaConstraints {
+  NSDictionary<NSString *, NSString *> *_mandatory;
+  NSDictionary<NSString *, NSString *> *_optional;
+}
+
+- (instancetype)initWithMandatoryConstraints:
+    (NSDictionary<NSString *, NSString *> *)mandatory
+                         optionalConstraints:
+    (NSDictionary<NSString *, NSString *> *)optional {
+  if (self = [super init]) {
+    _mandatory = [[NSDictionary alloc] initWithDictionary:mandatory
+                                                copyItems:YES];
+    _optional = [[NSDictionary alloc] initWithDictionary:optional
+                                               copyItems:YES];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCMediaConstraints:\n%@\n%@",
+                                    _mandatory,
+                                    _optional];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_ptr<webrtc::MediaConstraints>)nativeConstraints {
+  webrtc::MediaConstraintsInterface::Constraints mandatory =
+      [[self class] nativeConstraintsForConstraints:_mandatory];
+  webrtc::MediaConstraintsInterface::Constraints optional =
+      [[self class] nativeConstraintsForConstraints:_optional];
+
+  webrtc::MediaConstraints *nativeConstraints =
+      new webrtc::MediaConstraints(mandatory, optional);
+  return rtc::scoped_ptr<webrtc::MediaConstraints>(nativeConstraints);
+}
+
++ (webrtc::MediaConstraintsInterface::Constraints)
+    nativeConstraintsForConstraints:
+        (NSDictionary<NSString *, NSString *> *)constraints {
+  webrtc::MediaConstraintsInterface::Constraints nativeConstraints;
+  for (NSString *key in constraints) {
+    NSAssert([key isKindOfClass:[NSString class]],
+             @"%@ is not an NSString.", key);
+    NSAssert([constraints[key] isKindOfClass:[NSString class]],
+             @"%@ is not an NSString.", constraints[key]);
+    nativeConstraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
+        key.stdString, constraints[key].stdString));
+  }
+  return nativeConstraints;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource+Private.h
new file mode 100644
index 0000000000..fcbaad8e45
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource+Private.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaSource.h"
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaSource ()
+
+/**
+ * The MediaSourceInterface object passed to this RTCMediaSource during
+ * construction.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
+
+/** Initialize an RTCMediaSource from a native MediaSourceInterface. */
+- (instancetype)initWithNativeMediaSource:
+    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+    (RTCSourceState)state;
+
++ (RTCSourceState)sourceStateForNativeState:
+    (webrtc::MediaSourceInterface::SourceState)nativeState;
+
++ (NSString *)stringForState:(RTCSourceState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h b/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h
new file mode 100644
index 0000000000..0b36b8d709
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, RTCSourceState) {
+  RTCSourceStateInitializing,
+  RTCSourceStateLive,
+  RTCSourceStateEnded,
+  RTCSourceStateMuted,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaSource : NSObject
+
+/** The current state of the RTCMediaSource. */
+@property(nonatomic, readonly) RTCSourceState state;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.mm b/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.mm
new file mode 100644
index 0000000000..5f46ab8318
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.mm
@@ -0,0 +1,84 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaSource.h"
+
+#import "webrtc/api/objc/RTCMediaSource+Private.h"
+
+@implementation RTCMediaSource {
+  rtc::scoped_refptr<webrtc::MediaSourceInterface> _nativeMediaSource;
+}
+
+- (RTCSourceState)state {
+  return [[self class] sourceStateForNativeState:_nativeMediaSource->state()];
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCMediaSource:\n%@",
+                                    [[self class] stringForState:self.state]];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource {
+  return _nativeMediaSource;
+}
+
+- (instancetype)initWithNativeMediaSource:
+    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource {
+  NSParameterAssert(nativeMediaSource);
+  if (self = [super init]) {
+    _nativeMediaSource = nativeMediaSource;
+  }
+  return self;
+}
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+    (RTCSourceState)state {
+  switch (state) {
+    case RTCSourceStateInitializing:
+      return webrtc::MediaSourceInterface::kInitializing;
+    case RTCSourceStateLive:
+      return webrtc::MediaSourceInterface::kLive;
+    case RTCSourceStateEnded:
+      return webrtc::MediaSourceInterface::kEnded;
+    case RTCSourceStateMuted:
+      return webrtc::MediaSourceInterface::kMuted;
+  }
+}
+
++ (RTCSourceState)sourceStateForNativeState:
+    (webrtc::MediaSourceInterface::SourceState)nativeState {
+  switch (nativeState) {
+    case webrtc::MediaSourceInterface::kInitializing:
+      return RTCSourceStateInitializing;
+    case webrtc::MediaSourceInterface::kLive:
+      return RTCSourceStateLive;
+    case webrtc::MediaSourceInterface::kEnded:
+      return RTCSourceStateEnded;
+    case webrtc::MediaSourceInterface::kMuted:
+      return RTCSourceStateMuted;
+  }
+}
+
++ (NSString *)stringForState:(RTCSourceState)state {
+  switch (state) {
+    case RTCSourceStateInitializing:
+      return @"Initializing";
+    case RTCSourceStateLive:
+      return @"Live";
+    case RTCSourceStateEnded:
+      return @"Ended";
+    case RTCSourceStateMuted:
+      return @"Muted";
+  }
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack+Private.h
new file mode 100644
index 0000000000..3e17e63cd3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack+Private.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/base/scoped_ptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaStreamTrack ()
+
+/**
+ * The native MediaStreamTrackInterface representation of this
+ * RTCMediaStreamTrack object. This is needed to pass to the underlying C++
+ * APIs.
+ */
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
+
+/**
+ * Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
+ */
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+    (RTCMediaStreamTrackState)state;
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+    (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.h b/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.h
new file mode 100644
index 0000000000..beb48d3b6f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+/**
+ * Represents the state of the track. This exposes the same states in C++,
+ * which include two more states than are in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
+  RTCMediaStreamTrackStateInitializing,
+  RTCMediaStreamTrackStateLive,
+  RTCMediaStreamTrackStateEnded,
+  RTCMediaStreamTrackStateFailed,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaStreamTrack : NSObject
+
+/**
+ * The kind of track. For example, "audio" if this track represents an audio
+ * track and "video" if this track represents a video track.
+ */
+@property(nonatomic, readonly) NSString *kind;
+
+/** An identifier string. */
+@property(nonatomic, readonly) NSString *trackId;
+
+/** The enabled state of the track. */
+@property(nonatomic) BOOL isEnabled;
+
+/** The state of the track. */
+@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.mm b/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.mm
new file mode 100644
index 0000000000..e5751b0746
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.mm
@@ -0,0 +1,105 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@implementation RTCMediaStreamTrack {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
+}
+
+- (NSString *)kind {
+  return [NSString stringForStdString:_nativeTrack->kind()];
+}
+
+- (NSString *)trackId {
+  return [NSString stringForStdString:_nativeTrack->id()];
+}
+
+- (BOOL)isEnabled {
+  return _nativeTrack->enabled();
+}
+
+- (void)setIsEnabled:(BOOL)isEnabled {
+  _nativeTrack->set_enabled(isEnabled);
+}
+
+- (RTCMediaStreamTrackState)readyState {
+  return [[self class] trackStateForNativeState:_nativeTrack->state()];
+}
+
+- (NSString *)description {
+  NSString *readyState = [[self class] stringForState:self.readyState];
+  return [NSString stringWithFormat:@"RTCMediaStreamTrack:\n%@\n%@\n%@\n%@",
+                                    self.kind,
+                                    self.trackId,
+                                    self.isEnabled ? @"enabled" : @"disabled",
+                                    readyState];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  return _nativeTrack;
+}
+
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  NSParameterAssert(nativeTrack);
+  if (self = [super init]) {
+    _nativeTrack = nativeTrack;
+  }
+  return self;
+}
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+    (RTCMediaStreamTrackState)state {
+  switch (state) {
+    case RTCMediaStreamTrackStateInitializing:
+      return webrtc::MediaStreamTrackInterface::kInitializing;
+    case RTCMediaStreamTrackStateLive:
+      return webrtc::MediaStreamTrackInterface::kLive;
+    case RTCMediaStreamTrackStateEnded:
+      return webrtc::MediaStreamTrackInterface::kEnded;
+    case RTCMediaStreamTrackStateFailed:
+      return webrtc::MediaStreamTrackInterface::kFailed;
+  }
+}
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+    (webrtc::MediaStreamTrackInterface::TrackState)nativeState {
+  switch (nativeState) {
+    case webrtc::MediaStreamTrackInterface::kInitializing:
+      return RTCMediaStreamTrackStateInitializing;
+    case webrtc::MediaStreamTrackInterface::kLive:
+      return RTCMediaStreamTrackStateLive;
+    case webrtc::MediaStreamTrackInterface::kEnded:
+      return RTCMediaStreamTrackStateEnded;
+    case webrtc::MediaStreamTrackInterface::kFailed:
+      return RTCMediaStreamTrackStateFailed;
+  }
+}
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state {
+  switch (state) {
+    case RTCMediaStreamTrackStateInitializing:
+      return @"Initializing";
+    case RTCMediaStreamTrackStateLive:
+      return @"Live";
+    case RTCMediaStreamTrackStateEnded:
+      return @"Ended";
+    case RTCMediaStreamTrackStateFailed:
+      return @"Failed";
+  }
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.h b/media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.h
new file mode 100644
index 0000000000..27eb31e9af
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if TARGET_OS_IPHONE
+#error "This file targets OSX."
+#endif
+
+#import <AppKit/NSOpenGLView.h>
+
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCNSGLVideoView;
+@protocol RTCNSGLVideoViewDelegate
+
+- (void)videoView:(RTCNSGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
+
+@end
+
+@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
+
+@property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.m b/media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.m
new file mode 100644
index 0000000000..063e6f1330
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.m
@@ -0,0 +1,141 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNSGLVideoView.h"
+
+#import <CoreVideo/CVDisplayLink.h>
+#import <OpenGL/gl3.h>
+#import "RTCVideoFrame.h"
+#import "RTCOpenGLVideoRenderer.h"
+
+@interface RTCNSGLVideoView ()
+// |videoFrame| is set when we receive a frame from a worker thread and is read
+// from the display link callback so atomicity is required.
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@property(atomic, strong) RTCOpenGLVideoRenderer *glRenderer;
+- (void)drawFrame;
+@end
+
+static CVReturn OnDisplayLinkFired(CVDisplayLinkRef displayLink,
+                                   const CVTimeStamp *now,
+                                   const CVTimeStamp *outputTime,
+                                   CVOptionFlags flagsIn,
+                                   CVOptionFlags *flagsOut,
+                                   void *displayLinkContext) {
+  RTCNSGLVideoView *view = (__bridge RTCNSGLVideoView *)displayLinkContext;
+  [view drawFrame];
+  return kCVReturnSuccess;
+}
+
+@implementation RTCNSGLVideoView {
+  CVDisplayLinkRef _displayLink;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize glRenderer = _glRenderer;
+
+- (void)dealloc {
+  [self teardownDisplayLink];
+}
+
+- (void)drawRect:(NSRect)rect {
+  [self drawFrame];
+}
+
+- (void)reshape {
+  [super reshape];
+  NSRect frame = [self frame];
+  CGLLockContext([[self openGLContext] CGLContextObj]);
+  glViewport(0, 0, frame.size.width, frame.size.height);
+  CGLUnlockContext([[self openGLContext] CGLContextObj]);
+}
+
+- (void)lockFocus {
+  NSOpenGLContext *context = [self openGLContext];
+  [super lockFocus];
+  if ([context view] != self) {
+    [context setView:self];
+  }
+  [context makeCurrentContext];
+}
+
+- (void)prepareOpenGL {
+  [super prepareOpenGL];
+  if (!self.glRenderer) {
+    self.glRenderer =
+        [[RTCOpenGLVideoRenderer alloc] initWithContext:[self openGLContext]];
+  }
+  [self.glRenderer setupGL];
+  [self setupDisplayLink];
+}
+
+- (void)clearGLContext {
+  [self.glRenderer teardownGL];
+  self.glRenderer = nil;
+  [super clearGLContext];
+}
+
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    [self.delegate videoView:self didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(RTCVideoFrame *)frame {
+  self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)drawFrame {
+  RTCVideoFrame *videoFrame = self.videoFrame;
+  if (self.glRenderer.lastDrawnFrame != videoFrame) {
+    // This method may be called from CVDisplayLink callback which isn't on the
+    // main thread so we have to lock the GL context before drawing.
+    CGLLockContext([[self openGLContext] CGLContextObj]);
+    [self.glRenderer drawFrame:videoFrame];
+    CGLUnlockContext([[self openGLContext] CGLContextObj]);
+  }
+}
+
+- (void)setupDisplayLink {
+  if (_displayLink) {
+    return;
+  }
+  // Synchronize buffer swaps with vertical refresh rate.
+  GLint swapInt = 1;
+  [[self openGLContext] setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
+
+  // Create display link.
+  CVDisplayLinkCreateWithActiveCGDisplays(&_displayLink);
+  CVDisplayLinkSetOutputCallback(_displayLink,
+                                 &OnDisplayLinkFired,
+                                 (__bridge void *)self);
+  // Set the display link for the current renderer.
+  CGLContextObj cglContext = [[self openGLContext] CGLContextObj];
+  CGLPixelFormatObj cglPixelFormat = [[self pixelFormat] CGLPixelFormatObj];
+  CVDisplayLinkSetCurrentCGDisplayFromOpenGLContext(
+      _displayLink, cglContext, cglPixelFormat);
+  CVDisplayLinkStart(_displayLink);
+}
+
+- (void)teardownDisplayLink {
+  if (!_displayLink) {
+    return;
+  }
+  CVDisplayLinkRelease(_displayLink);
+  _displayLink = NULL;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.h b/media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.h
new file mode 100644
index 0000000000..729839c6a3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <GLKit/GLKit.h>
+#else
+#import <AppKit/NSOpenGL.h>
+#endif
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCVideoFrame;
+
+// RTCOpenGLVideoRenderer issues appropriate OpenGL commands to draw a frame to
+// the currently bound framebuffer. Supports OpenGL 3.2 and OpenGLES 2.0. OpenGL
+// framebuffer creation and management should be handled elsewhere using the
+// same context used to initialize this class.
+@interface RTCOpenGLVideoRenderer : NSObject
+
+// The last successfully drawn frame. Used to avoid drawing frames unnecessarily
+// hence saving battery life by reducing load.
+@property(nonatomic, readonly) RTCVideoFrame *lastDrawnFrame;
+
+#if TARGET_OS_IPHONE
+- (instancetype)initWithContext:(EAGLContext *)context
+    NS_DESIGNATED_INITIALIZER;
+#else
+- (instancetype)initWithContext:(NSOpenGLContext *)context
+    NS_DESIGNATED_INITIALIZER;
+#endif
+
+// Draws |frame| onto the currently bound OpenGL framebuffer. |setupGL| must be
+// called before this function will succeed.
+- (BOOL)drawFrame:(RTCVideoFrame *)frame;
+
+// The following methods are used to manage OpenGL resources. On iOS
+// applications should release resources when placed in background for use in
+// the foreground application. In fact, attempting to call OpenGLES commands
+// while in background will result in application termination.
+
+// Sets up the OpenGL state needed for rendering.
+- (void)setupGL;
+// Tears down the OpenGL state created by |setupGL|.
+- (void)teardownGL;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.mm b/media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.mm
new file mode 100644
index 0000000000..56a6431ffa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.mm
@@ -0,0 +1,485 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCOpenGLVideoRenderer.h"
+
+#include <string.h>
+
+#include "webrtc/base/scoped_ptr.h"
+
+#if TARGET_OS_IPHONE
+#import <OpenGLES/ES3/gl.h>
+#else
+#import <OpenGL/gl3.h>
+#endif
+
+#import "RTCVideoFrame.h"
+
+// TODO(tkchin): check and log openGL errors. Methods here return BOOLs in
+// anticipation of that happening in the future.
+
+#if TARGET_OS_IPHONE
+#define RTC_PIXEL_FORMAT GL_LUMINANCE
+#define SHADER_VERSION
+#define VERTEX_SHADER_IN "attribute"
+#define VERTEX_SHADER_OUT "varying"
+#define FRAGMENT_SHADER_IN "varying"
+#define FRAGMENT_SHADER_OUT
+#define FRAGMENT_SHADER_COLOR "gl_FragColor"
+#define FRAGMENT_SHADER_TEXTURE "texture2D"
+#else
+#define RTC_PIXEL_FORMAT GL_RED
+#define SHADER_VERSION "#version 150\n"
+#define VERTEX_SHADER_IN "in"
+#define VERTEX_SHADER_OUT "out"
+#define FRAGMENT_SHADER_IN "in"
+#define FRAGMENT_SHADER_OUT "out vec4 fragColor;\n"
+#define FRAGMENT_SHADER_COLOR "fragColor"
+#define FRAGMENT_SHADER_TEXTURE "texture"
+#endif
+
+// Vertex shader doesn't do anything except pass coordinates through.
+static const char kVertexShaderSource[] =
+  SHADER_VERSION
+  VERTEX_SHADER_IN " vec2 position;\n"
+  VERTEX_SHADER_IN " vec2 texcoord;\n"
+  VERTEX_SHADER_OUT " vec2 v_texcoord;\n"
+  "void main() {\n"
+  "    gl_Position = vec4(position.x, position.y, 0.0, 1.0);\n"
+  "    v_texcoord = texcoord;\n"
+  "}\n";
+
+// Fragment shader converts YUV values from input textures into a final RGB
+// pixel. The conversion formula is from http://www.fourcc.org/fccyvrgb.php.
+static const char kFragmentShaderSource[] =
+  SHADER_VERSION
+  "precision highp float;"
+  FRAGMENT_SHADER_IN " vec2 v_texcoord;\n"
+  "uniform lowp sampler2D s_textureY;\n"
+  "uniform lowp sampler2D s_textureU;\n"
+  "uniform lowp sampler2D s_textureV;\n"
+  FRAGMENT_SHADER_OUT
+  "void main() {\n"
+  "    float y, u, v, r, g, b;\n"
+  "    y = " FRAGMENT_SHADER_TEXTURE "(s_textureY, v_texcoord).r;\n"
+  "    u = " FRAGMENT_SHADER_TEXTURE "(s_textureU, v_texcoord).r;\n"
+  "    v = " FRAGMENT_SHADER_TEXTURE "(s_textureV, v_texcoord).r;\n"
+  "    u = u - 0.5;\n"
+  "    v = v - 0.5;\n"
+  "    r = y + 1.403 * v;\n"
+  "    g = y - 0.344 * u - 0.714 * v;\n"
+  "    b = y + 1.770 * u;\n"
+  "    " FRAGMENT_SHADER_COLOR " = vec4(r, g, b, 1.0);\n"
+  "  }\n";
+
+// Compiles a shader of the given |type| with GLSL source |source| and returns
+// the shader handle or 0 on error.
+GLuint CreateShader(GLenum type, const GLchar *source) {
+  GLuint shader = glCreateShader(type);
+  if (!shader) {
+    return 0;
+  }
+  glShaderSource(shader, 1, &source, NULL);
+  glCompileShader(shader);
+  GLint compileStatus = GL_FALSE;
+  glGetShaderiv(shader, GL_COMPILE_STATUS, &compileStatus);
+  if (compileStatus == GL_FALSE) {
+    glDeleteShader(shader);
+    shader = 0;
+  }
+  return shader;
+}
+
+// Links a shader program with the given vertex and fragment shaders and
+// returns the program handle or 0 on error.
+GLuint CreateProgram(GLuint vertexShader, GLuint fragmentShader) {
+  if (vertexShader == 0 || fragmentShader == 0) {
+    return 0;
+  }
+  GLuint program = glCreateProgram();
+  if (!program) {
+    return 0;
+  }
+  glAttachShader(program, vertexShader);
+  glAttachShader(program, fragmentShader);
+  glLinkProgram(program);
+  GLint linkStatus = GL_FALSE;
+  glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
+  if (linkStatus == GL_FALSE) {
+    glDeleteProgram(program);
+    program = 0;
+  }
+  return program;
+}
+
+// When modelview and projection matrices are identity (default) the world is
+// contained in the square around origin with unit size 2. Drawing to these
+// coordinates is equivalent to drawing to the entire screen. The texture is
+// stretched over that square using texture coordinates (u, v) that range
+// from (0, 0) to (1, 1) inclusive. Texture coordinates are flipped vertically
+// here because the incoming frame has origin in upper left hand corner but
+// OpenGL expects origin in bottom left corner.
+const GLfloat gVertices[] = {
+  // X, Y, U, V.
+  -1, -1, 0, 1,  // Bottom left.
+   1, -1, 1, 1,  // Bottom right.
+   1,  1, 1, 0,  // Top right.
+  -1,  1, 0, 0,  // Top left.
+};
+
+// |kNumTextures| must not exceed 8, which is the limit in OpenGLES2. Two sets
+// of 3 textures are used here, one for each of the Y, U and V planes. Having
+// two sets alleviates CPU blockage in the event that the GPU is asked to render
+// to a texture that is already in use.
+static const GLsizei kNumTextureSets = 2;
+static const GLsizei kNumTextures = 3 * kNumTextureSets;
+
+@implementation RTCOpenGLVideoRenderer {
+#if TARGET_OS_IPHONE
+  EAGLContext *_context;
+#else
+  NSOpenGLContext *_context;
+#endif
+  BOOL _isInitialized;
+  NSUInteger _currentTextureSet;
+  // Handles for OpenGL constructs.
+  GLuint _textures[kNumTextures];
+  GLuint _program;
+#if !TARGET_OS_IPHONE
+  GLuint _vertexArray;
+#endif
+  GLuint _vertexBuffer;
+  GLint _position;
+  GLint _texcoord;
+  GLint _ySampler;
+  GLint _uSampler;
+  GLint _vSampler;
+  // Used to create a non-padded plane for GPU upload when we receive padded
+  // frames.
+  rtc::scoped_ptr<uint8_t[]> _planeBuffer;
+}
+
+@synthesize lastDrawnFrame = _lastDrawnFrame;
+
++ (void)initialize {
+  // Disable dithering for performance.
+  glDisable(GL_DITHER);
+}
+
+#if TARGET_OS_IPHONE
+- (instancetype)initWithContext:(EAGLContext *)context {
+#else
+- (instancetype)initWithContext:(NSOpenGLContext *)context {
+#endif
+  NSAssert(context != nil, @"context cannot be nil");
+  if (self = [super init]) {
+    _context = context;
+  }
+  return self;
+}
+
+- (BOOL)drawFrame:(RTCVideoFrame *)frame {
+  if (!_isInitialized) {
+    return NO;
+  }
+  if (_lastDrawnFrame == frame) {
+    return NO;
+  }
+  [self ensureGLContext];
+  glClear(GL_COLOR_BUFFER_BIT);
+  if (frame) {
+    if (![self updateTextureSizesForFrame:frame] ||
+        ![self updateTextureDataForFrame:frame]) {
+      return NO;
+    }
+#if !TARGET_OS_IPHONE
+    glBindVertexArray(_vertexArray);
+#endif
+    glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
+    glDrawArrays(GL_TRIANGLE_FAN, 0, 4);
+  }
+#if !TARGET_OS_IPHONE
+  [_context flushBuffer];
+#endif
+  _lastDrawnFrame = frame;
+  return YES;
+}
+
+- (void)setupGL {
+  if (_isInitialized) {
+    return;
+  }
+  [self ensureGLContext];
+  if (![self setupProgram]) {
+    return;
+  }
+  if (![self setupTextures]) {
+    return;
+  }
+  if (![self setupVertices]) {
+    return;
+  }
+  glUseProgram(_program);
+  glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
+  _isInitialized = YES;
+}
+
+- (void)teardownGL {
+  if (!_isInitialized) {
+    return;
+  }
+  [self ensureGLContext];
+  glDeleteProgram(_program);
+  _program = 0;
+  glDeleteTextures(kNumTextures, _textures);
+  glDeleteBuffers(1, &_vertexBuffer);
+  _vertexBuffer = 0;
+#if !TARGET_OS_IPHONE
+  glDeleteVertexArrays(1, &_vertexArray);
+#endif
+  _isInitialized = NO;
+}
+
+#pragma mark - Private
+
+- (void)ensureGLContext {
+  NSAssert(_context, @"context shouldn't be nil");
+#if TARGET_OS_IPHONE
+  if ([EAGLContext currentContext] != _context) {
+    [EAGLContext setCurrentContext:_context];
+  }
+#else
+  if ([NSOpenGLContext currentContext] != _context) {
+    [_context makeCurrentContext];
+  }
+#endif
+}
+
+- (BOOL)setupProgram {
+  NSAssert(!_program, @"program already set up");
+  GLuint vertexShader = CreateShader(GL_VERTEX_SHADER, kVertexShaderSource);
+  NSAssert(vertexShader, @"failed to create vertex shader");
+  GLuint fragmentShader =
+      CreateShader(GL_FRAGMENT_SHADER, kFragmentShaderSource);
+  NSAssert(fragmentShader, @"failed to create fragment shader");
+  _program = CreateProgram(vertexShader, fragmentShader);
+  // Shaders are created only to generate program.
+  if (vertexShader) {
+    glDeleteShader(vertexShader);
+  }
+  if (fragmentShader) {
+    glDeleteShader(fragmentShader);
+  }
+  if (!_program) {
+    return NO;
+  }
+  _position = glGetAttribLocation(_program, "position");
+  _texcoord = glGetAttribLocation(_program, "texcoord");
+  _ySampler = glGetUniformLocation(_program, "s_textureY");
+  _uSampler = glGetUniformLocation(_program, "s_textureU");
+  _vSampler = glGetUniformLocation(_program, "s_textureV");
+  if (_position < 0 || _texcoord < 0 || _ySampler < 0 || _uSampler < 0 ||
+      _vSampler < 0) {
+    return NO;
+  }
+  return YES;
+}
+
+- (BOOL)setupTextures {
+  glGenTextures(kNumTextures, _textures);
+  // Set parameters for each of the textures we created.
+  for (GLsizei i = 0; i < kNumTextures; i++) {
+    glActiveTexture(GL_TEXTURE0 + i);
+    glBindTexture(GL_TEXTURE_2D, _textures[i]);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+  }
+  return YES;
+}
+
+- (BOOL)updateTextureSizesForFrame:(RTCVideoFrame *)frame {
+  if (frame.height == _lastDrawnFrame.height &&
+      frame.width == _lastDrawnFrame.width &&
+      frame.chromaWidth == _lastDrawnFrame.chromaWidth &&
+      frame.chromaHeight == _lastDrawnFrame.chromaHeight) {
+    return YES;
+  }
+  GLsizei lumaWidth = frame.width;
+  GLsizei lumaHeight = frame.height;
+  GLsizei chromaWidth = frame.chromaWidth;
+  GLsizei chromaHeight = frame.chromaHeight;
+  for (GLint i = 0; i < kNumTextureSets; i++) {
+    glActiveTexture(GL_TEXTURE0 + i * 3);
+    glTexImage2D(GL_TEXTURE_2D,
+                 0,
+                 RTC_PIXEL_FORMAT,
+                 lumaWidth,
+                 lumaHeight,
+                 0,
+                 RTC_PIXEL_FORMAT,
+                 GL_UNSIGNED_BYTE,
+                 0);
+    glActiveTexture(GL_TEXTURE0 + i * 3 + 1);
+    glTexImage2D(GL_TEXTURE_2D,
+                 0,
+                 RTC_PIXEL_FORMAT,
+                 chromaWidth,
+                 chromaHeight,
+                 0,
+                 RTC_PIXEL_FORMAT,
+                 GL_UNSIGNED_BYTE,
+                 0);
+    glActiveTexture(GL_TEXTURE0 + i * 3 + 2);
+    glTexImage2D(GL_TEXTURE_2D,
+                 0,
+                 RTC_PIXEL_FORMAT,
+                 chromaWidth,
+                 chromaHeight,
+                 0,
+                 RTC_PIXEL_FORMAT,
+                 GL_UNSIGNED_BYTE,
+                 0);
+  }
+  if ((NSUInteger)frame.yPitch != frame.width ||
+      (NSUInteger)frame.uPitch != frame.chromaWidth ||
+      (NSUInteger)frame.vPitch != frame.chromaWidth) {
+    _planeBuffer.reset(new uint8_t[frame.width * frame.height]);
+  } else {
+    _planeBuffer.reset();
+  }
+  return YES;
+}
+
+- (void)uploadPlane:(const uint8_t *)plane
+            sampler:(GLint)sampler
+             offset:(NSUInteger)offset
+              width:(size_t)width
+             height:(size_t)height
+             stride:(int32_t)stride {
+  glActiveTexture(GL_TEXTURE0 + offset);
+  // When setting texture sampler uniforms, the texture index is used not
+  // the texture handle.
+  glUniform1i(sampler, offset);
+#if TARGET_OS_IPHONE
+  BOOL hasUnpackRowLength = _context.API == kEAGLRenderingAPIOpenGLES3;
+#else
+  BOOL hasUnpackRowLength = YES;
+#endif
+  const uint8_t *uploadPlane = plane;
+  if ((size_t)stride != width) {
+   if (hasUnpackRowLength) {
+      // GLES3 allows us to specify stride.
+      glPixelStorei(GL_UNPACK_ROW_LENGTH, stride);
+      glTexImage2D(GL_TEXTURE_2D,
+                   0,
+                   RTC_PIXEL_FORMAT,
+                   width,
+                   height,
+                   0,
+                   RTC_PIXEL_FORMAT,
+                   GL_UNSIGNED_BYTE,
+                   uploadPlane);
+      glPixelStorei(GL_UNPACK_ROW_LENGTH, 0);
+      return;
+    } else {
+      // Make an unpadded copy and upload that instead. Quick profiling showed
+      // that this is faster than uploading row by row using glTexSubImage2D.
+      uint8_t *unpaddedPlane = _planeBuffer.get();
+      for (size_t y = 0; y < height; ++y) {
+        memcpy(unpaddedPlane + y * width, plane + y * stride, width);
+      }
+      uploadPlane = unpaddedPlane;
+    }
+  }
+  glTexImage2D(GL_TEXTURE_2D,
+               0,
+               RTC_PIXEL_FORMAT,
+               width,
+               height,
+               0,
+               RTC_PIXEL_FORMAT,
+               GL_UNSIGNED_BYTE,
+               uploadPlane);
+}
+
+- (BOOL)updateTextureDataForFrame:(RTCVideoFrame *)frame {
+  NSUInteger textureOffset = _currentTextureSet * 3;
+  NSAssert(textureOffset + 3 <= kNumTextures, @"invalid offset");
+
+  [self uploadPlane:frame.yPlane
+            sampler:_ySampler
+             offset:textureOffset
+              width:frame.width
+             height:frame.height
+             stride:frame.yPitch];
+
+  [self uploadPlane:frame.uPlane
+            sampler:_uSampler
+             offset:textureOffset + 1
+              width:frame.chromaWidth
+             height:frame.chromaHeight
+             stride:frame.uPitch];
+
+  [self uploadPlane:frame.vPlane
+            sampler:_vSampler
+             offset:textureOffset + 2
+              width:frame.chromaWidth
+             height:frame.chromaHeight
+             stride:frame.vPitch];
+
+  _currentTextureSet = (_currentTextureSet + 1) % kNumTextureSets;
+  return YES;
+}
+
+- (BOOL)setupVertices {
+#if !TARGET_OS_IPHONE
+  NSAssert(!_vertexArray, @"vertex array already set up");
+  glGenVertexArrays(1, &_vertexArray);
+  if (!_vertexArray) {
+    return NO;
+  }
+  glBindVertexArray(_vertexArray);
+#endif
+  NSAssert(!_vertexBuffer, @"vertex buffer already set up");
+  glGenBuffers(1, &_vertexBuffer);
+  if (!_vertexBuffer) {
+#if !TARGET_OS_IPHONE
+    glDeleteVertexArrays(1, &_vertexArray);
+    _vertexArray = 0;
+#endif
+    return NO;
+  }
+  glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
+  glBufferData(GL_ARRAY_BUFFER, sizeof(gVertices), gVertices, GL_DYNAMIC_DRAW);
+
+  // Read position attribute from |gVertices| with size of 2 and stride of 4
+  // beginning at the start of the array. The last argument indicates offset
+  // of data within |gVertices| as supplied to the vertex buffer.
+  glVertexAttribPointer(
+      _position, 2, GL_FLOAT, GL_FALSE, 4 * sizeof(GLfloat), (void *)0);
+  glEnableVertexAttribArray(_position);
+
+  // Read texcoord attribute from |gVertices| with size of 2 and stride of 4
+  // beginning at the first texcoord in the array. The last argument indicates
+  // offset of data within |gVertices| as supplied to the vertex buffer.
+  glVertexAttribPointer(_texcoord,
+                        2,
+                        GL_FLOAT,
+                        GL_FALSE,
+                        4 * sizeof(GLfloat),
+                        (void *)(2 * sizeof(GLfloat)));
+  glEnableVertexAttribArray(_texcoord);
+
+  return YES;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription+Private.h
new file mode 100644
index 0000000000..aa0314d3f3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription+Private.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription.h"
+
+#include "talk/app/webrtc/jsep.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCSessionDescription ()
+
+/**
+ * The native SessionDescriptionInterface representation of this
+ * RTCSessionDescription object. This is needed to pass to the underlying C++
+ * APIs.
+ */
+@property(nonatomic, readonly)
+    webrtc::SessionDescriptionInterface *nativeDescription;
+
+/**
+ * Initialize an RTCSessionDescription from a native
+ * SessionDescriptionInterface. No ownership is taken of the native session
+ * description.
+ */
+- (instancetype)initWithNativeDescription:
+    (webrtc::SessionDescriptionInterface *)nativeDescription;
+
++ (std::string)stringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForString:(const std::string &)string;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.h b/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.h
new file mode 100644
index 0000000000..5f00b1c9f4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+/**
+ * Represents the session description type. This exposes the same types that are
+ * in C++, which doesn't include the rollback type that is in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCSdpType) {
+  RTCSdpTypeOffer,
+  RTCSdpTypePrAnswer,
+  RTCSdpTypeAnswer,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCSessionDescription : NSObject
+
+/** The type of session description. */
+@property(nonatomic, readonly) RTCSdpType type;
+
+/** The SDP string representation of this session description. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize a session description with a type and SDP string. */
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.mm b/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.mm
new file mode 100644
index 0000000000..7ed0760158
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.mm
@@ -0,0 +1,92 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription.h"
+
+#include "webrtc/base/checks.h"
+
+#import "webrtc/api/objc/RTCSessionDescription+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+#import "webrtc/base/objc/RTCLogging.h"
+
+@implementation RTCSessionDescription
+
+@synthesize type = _type;
+@synthesize sdp = _sdp;
+
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp {
+  NSParameterAssert(sdp.length);
+  if (self = [super init]) {
+    _type = type;
+    _sdp = [sdp copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCSessionDescription:\n%s\n%@",
+                                    [[self class] stringForType:_type].c_str(),
+                                    _sdp];
+}
+
+#pragma mark - Private
+
+- (webrtc::SessionDescriptionInterface *)nativeDescription {
+  webrtc::SdpParseError error;
+
+  webrtc::SessionDescriptionInterface *description =
+      webrtc::CreateSessionDescription([[self class] stringForType:_type],
+                                       _sdp.stdString,
+                                       &error);
+
+  if (!description) {
+    RTCLogError(@"Failed to create session description: %s\nline: %s",
+                error.description.c_str(),
+                error.line.c_str());
+  }
+
+  return description;
+}
+
+- (instancetype)initWithNativeDescription:
+    (webrtc::SessionDescriptionInterface *)nativeDescription {
+  NSParameterAssert(nativeDescription);
+  std::string sdp;
+  nativeDescription->ToString(&sdp);
+  RTCSdpType type = [[self class] typeForString:nativeDescription->type()];
+
+  return [self initWithType:type
+                        sdp:[NSString stringForStdString:sdp]];
+}
+
++ (std::string)stringForType:(RTCSdpType)type {
+  switch (type) {
+    case RTCSdpTypeOffer:
+      return webrtc::SessionDescriptionInterface::kOffer;
+    case RTCSdpTypePrAnswer:
+      return webrtc::SessionDescriptionInterface::kPrAnswer;
+    case RTCSdpTypeAnswer:
+      return webrtc::SessionDescriptionInterface::kAnswer;
+  }
+}
+
++ (RTCSdpType)typeForString:(const std::string &)string {
+  if (string == webrtc::SessionDescriptionInterface::kOffer) {
+    return RTCSdpTypeOffer;
+  } else if (string == webrtc::SessionDescriptionInterface::kPrAnswer) {
+    return RTCSdpTypePrAnswer;
+  } else if (string == webrtc::SessionDescriptionInterface::kAnswer) {
+    return RTCSdpTypeAnswer;
+  } else {
+    RTC_NOTREACHED();
+  }
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport+Private.h b/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport+Private.h
new file mode 100644
index 0000000000..5b7dc32a74
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport+Private.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCStatsReport.h"
+
+#include "talk/app/webrtc/statstypes.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCStatsReport ()
+
+/** Initialize an RTCStatsReport object from a native StatsReport. */
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.h b/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.h
new file mode 100644
index 0000000000..fc66faf2cf
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This does not currently conform to the spec. */
+@interface RTCStatsReport : NSObject
+
+/** Time since 1970-01-01T00:00:00Z in milliseconds. */
+@property(nonatomic, readonly) CFTimeInterval timestamp;
+
+/** The type of stats held by this object. */
+@property(nonatomic, readonly) NSString *type;
+
+/** The identifier for this object. */
+@property(nonatomic, readonly) NSString *statsId;
+
+/** A dictionary holding the actual stats. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.mm b/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.mm
new file mode 100644
index 0000000000..35a5229014
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.mm
@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCStatsReport.h"
+
+#include "webrtc/base/checks.h"
+
+#import "webrtc/api/objc/RTCStatsReport+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+#import "webrtc/base/objc/RTCLogging.h"
+
+@implementation RTCStatsReport
+
+@synthesize timestamp = _timestamp;
+@synthesize type = _type;
+@synthesize statsId = _statsId;
+@synthesize values = _values;
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCStatsReport:\n%@\n%@\n%f\n%@",
+                                    _statsId,
+                                    _type,
+                                    _timestamp,
+                                    _values];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport {
+  if (self = [super init]) {
+    _timestamp = nativeReport.timestamp();
+    _type = [NSString stringForStdString:nativeReport.TypeToString()];
+    _statsId = [NSString stringForStdString:
+        nativeReport.id()->ToString()];
+
+    NSUInteger capacity = nativeReport.values().size();
+    NSMutableDictionary *values =
+        [NSMutableDictionary dictionaryWithCapacity:capacity];
+    for (auto const &valuePair : nativeReport.values()) {
+      NSString *key = [NSString stringForStdString:
+          valuePair.second->display_name()];
+      NSString *value = [NSString stringForStdString:
+          valuePair.second->ToString()];
+
+      // Not expecting duplicate keys.
+      RTC_DCHECK(values[key]);
+
+      values[key] = value;
+    }
+    _values = values;
+  }
+  return self;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/base/basicdefs.h b/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame+Private.h
similarity index 52%
rename from media/webrtc/trunk/webrtc/base/basicdefs.h
rename to media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame+Private.h
index 1dee2ae658..954344aee1 100644
--- a/media/webrtc/trunk/webrtc/base/basicdefs.h
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame+Private.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,13 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_BASE_BASICDEFS_H_
-#define WEBRTC_BASE_BASICDEFS_H_
+#import "RTCVideoFrame.h"
 
-#if HAVE_CONFIG_H
-#include "config.h"  // NOLINT
-#endif
+#include "talk/media/base/videoframe.h"
 
-#define ARRAY_SIZE(x) (static_cast<int>(sizeof(x) / sizeof(x[0])))
+NS_ASSUME_NONNULL_BEGIN
 
-#endif  // WEBRTC_BASE_BASICDEFS_H_
+@interface RTCVideoFrame ()
+
+- (instancetype)initWithNativeFrame:(const cricket::VideoFrame *)nativeFrame
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.h b/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.h
new file mode 100644
index 0000000000..8ed23ba82c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoFrame : NSObject
+
+/** Width without rotation applied. */
+@property(nonatomic, readonly) size_t width;
+
+/** Height without rotation applied. */
+@property(nonatomic, readonly) size_t height;
+@property(nonatomic, readonly) size_t chromaWidth;
+@property(nonatomic, readonly) size_t chromaHeight;
+@property(nonatomic, readonly) size_t chromaSize;
+// These can return NULL if the object is not backed by a buffer.
+@property(nonatomic, readonly, nullable) const uint8_t *yPlane;
+@property(nonatomic, readonly, nullable) const uint8_t *uPlane;
+@property(nonatomic, readonly, nullable) const uint8_t *vPlane;
+@property(nonatomic, readonly) int32_t yPitch;
+@property(nonatomic, readonly) int32_t uPitch;
+@property(nonatomic, readonly) int32_t vPitch;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.mm b/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.mm
new file mode 100644
index 0000000000..db2d07ba31
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.mm
@@ -0,0 +1,79 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#include "webrtc/base/scoped_ptr.h"
+
+#import "webrtc/api/objc/RTCVideoFrame+Private.h"
+
+@implementation RTCVideoFrame {
+  rtc::scoped_ptr<cricket::VideoFrame> _videoFrame;
+}
+
+- (size_t)width {
+  return _videoFrame->GetWidth();
+}
+
+- (size_t)height {
+  return _videoFrame->GetHeight();
+}
+
+- (size_t)chromaWidth {
+  return _videoFrame->GetChromaWidth();
+}
+
+- (size_t)chromaHeight {
+  return _videoFrame->GetChromaHeight();
+}
+
+- (size_t)chromaSize {
+  return _videoFrame->GetChromaSize();
+}
+
+- (const uint8_t *)yPlane {
+  const cricket::VideoFrame *const_frame = _videoFrame.get();
+  return const_frame->GetYPlane();
+}
+
+- (const uint8_t *)uPlane {
+  const cricket::VideoFrame *const_frame = _videoFrame.get();
+  return const_frame->GetUPlane();
+}
+
+- (const uint8_t *)vPlane {
+  const cricket::VideoFrame *const_frame = _videoFrame.get();
+  return const_frame->GetVPlane();
+}
+
+- (int32_t)yPitch {
+  return _videoFrame->GetYPitch();
+}
+
+- (int32_t)uPitch {
+  return _videoFrame->GetUPitch();
+}
+
+- (int32_t)vPitch {
+  return _videoFrame->GetVPitch();
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeFrame:(const cricket::VideoFrame *)nativeFrame {
+  if (self = [super init]) {
+    // Keep a shallow copy of the video frame. The underlying frame buffer is
+    // not copied.
+    _videoFrame.reset(nativeFrame->Copy());
+  }
+  return self;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/api/objc/RTCVideoRenderer.h b/media/webrtc/trunk/webrtc/api/objc/RTCVideoRenderer.h
new file mode 100644
index 0000000000..a97456275a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objc/RTCVideoRenderer.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#endif
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCVideoFrame;
+
+@protocol RTCVideoRenderer <NSObject>
+
+/** The size of the frame. */
+- (void)setSize:(CGSize)size;
+
+/** The frame to be displayed. */
+- (void)renderFrame:(RTCVideoFrame *)frame;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event_tracer_unittest.cc b/media/webrtc/trunk/webrtc/api/objc/WebRTC-Prefix.pch
similarity index 66%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/event_tracer_unittest.cc
rename to media/webrtc/trunk/webrtc/api/objc/WebRTC-Prefix.pch
index 9328e80036..990b1602da 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event_tracer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/api/objc/WebRTC-Prefix.pch
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,5 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// This file has moved.
-// TODO(tommi): Delete after removing dependencies and updating Chromium.
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
diff --git a/media/webrtc/trunk/webrtc/api/objctests/RTCIceCandidateTest.mm b/media/webrtc/trunk/webrtc/api/objctests/RTCIceCandidateTest.mm
new file mode 100644
index 0000000000..391db44ae1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objctests/RTCIceCandidateTest.mm
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCIceCandidate.h"
+#import "webrtc/api/objc/RTCIceCandidate+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCIceCandidateTest : NSObject
+- (void)testCandidate;
+- (void)testInitFromNativeCandidate;
+@end
+
+@implementation RTCIceCandidateTest
+
+- (void)testCandidate {
+  NSString *sdp = @"candidate:4025901590 1 udp 2122265343 "
+                   "fdff:2642:12a6:fe38:c001:beda:fcf9:51aa "
+                   "59052 typ host generation 0";
+
+  RTCIceCandidate *candidate = [[RTCIceCandidate alloc] initWithSdp:sdp
+                                                      sdpMLineIndex:0
+                                                             sdpMid:@"audio"];
+
+  rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate =
+      candidate.nativeCandidate;
+  EXPECT_EQ("audio", nativeCandidate->sdp_mid());
+  EXPECT_EQ(0, nativeCandidate->sdp_mline_index());
+
+  std::string sdpString;
+  nativeCandidate->ToString(&sdpString);
+  EXPECT_EQ(sdp.stdString, sdpString);
+}
+
+- (void)testInitFromNativeCandidate {
+  std::string sdp("candidate:4025901590 1 udp 2122265343 "
+                  "fdff:2642:12a6:fe38:c001:beda:fcf9:51aa "
+                  "59052 typ host generation 0");
+  webrtc::IceCandidateInterface *nativeCandidate =
+      webrtc::CreateIceCandidate("audio", 0, sdp, nullptr);
+
+  RTCIceCandidate *iceCandidate =
+      [[RTCIceCandidate alloc] initWithNativeCandidate:nativeCandidate];
+  EXPECT_TRUE([@"audio" isEqualToString:iceCandidate.sdpMid]);
+  EXPECT_EQ(0, iceCandidate.sdpMLineIndex);
+
+  EXPECT_EQ(sdp, iceCandidate.sdp.stdString);
+}
+
+@end
+
+TEST(RTCIceCandidateTest, CandidateTest) {
+  @autoreleasepool {
+    RTCIceCandidateTest *test = [[RTCIceCandidateTest alloc] init];
+    [test testCandidate];
+  }
+}
+
+TEST(RTCIceCandidateTest, InitFromCandidateTest) {
+  @autoreleasepool {
+    RTCIceCandidateTest *test = [[RTCIceCandidateTest alloc] init];
+    [test testInitFromNativeCandidate];
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/api/objctests/RTCIceServerTest.mm b/media/webrtc/trunk/webrtc/api/objctests/RTCIceServerTest.mm
new file mode 100644
index 0000000000..5fa43f8447
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objctests/RTCIceServerTest.mm
@@ -0,0 +1,84 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include <vector>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCIceServer.h"
+#import "webrtc/api/objc/RTCIceServer+Private.h"
+
+@interface RTCIceServerTest : NSObject
+- (void)testOneURLServer;
+- (void)testTwoURLServer;
+- (void)testPasswordCredential;
+@end
+
+@implementation RTCIceServerTest
+
+- (void)testOneURLServer {
+  RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:@[
+      @"stun:stun1.example.net" ]];
+
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
+  EXPECT_EQ((size_t)1, iceStruct.urls.size());
+  EXPECT_EQ("stun:stun1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("", iceStruct.username);
+  EXPECT_EQ("", iceStruct.password);
+}
+
+- (void)testTwoURLServer {
+  RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:@[
+      @"turn1:turn1.example.net", @"turn2:turn2.example.net" ]];
+
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
+  EXPECT_EQ((size_t)2, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("turn2:turn2.example.net", iceStruct.urls.back());
+  EXPECT_EQ("", iceStruct.username);
+  EXPECT_EQ("", iceStruct.password);
+}
+
+- (void)testPasswordCredential {
+  RTCIceServer *server = [[RTCIceServer alloc]
+      initWithURLStrings:@[ @"turn1:turn1.example.net" ]
+                username:@"username"
+              credential:@"credential"];
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
+  EXPECT_EQ((size_t)1, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("username", iceStruct.username);
+  EXPECT_EQ("credential", iceStruct.password);
+}
+
+@end
+
+TEST(RTCIceServerTest, OneURLTest) {
+  @autoreleasepool {
+    RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+    [test testOneURLServer];
+  }
+}
+
+TEST(RTCIceServerTest, TwoURLTest) {
+  @autoreleasepool {
+    RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+    [test testTwoURLServer];
+  }
+}
+
+TEST(RTCIceServerTest, PasswordCredentialTest) {
+  @autoreleasepool {
+    RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+    [test testPasswordCredential];
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/api/objctests/RTCMediaConstraintsTest.mm b/media/webrtc/trunk/webrtc/api/objctests/RTCMediaConstraintsTest.mm
new file mode 100644
index 0000000000..44ffe3d033
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objctests/RTCMediaConstraintsTest.mm
@@ -0,0 +1,66 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCMediaConstraints.h"
+#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCMediaConstraintsTest : NSObject
+- (void)testMediaConstraints;
+@end
+
+@implementation RTCMediaConstraintsTest
+
+- (void)testMediaConstraints {
+  NSDictionary *mandatory = @{@"key1": @"value1", @"key2": @"value2"};
+  NSDictionary *optional = @{@"key3": @"value3", @"key4": @"value4"};
+
+  RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc]
+      initWithMandatoryConstraints:mandatory
+               optionalConstraints:optional];
+  rtc::scoped_ptr<webrtc::MediaConstraints> nativeConstraints =
+      [constraints nativeConstraints];
+
+  webrtc::MediaConstraintsInterface::Constraints nativeMandatory =
+      nativeConstraints->GetMandatory();
+  [self expectConstraints:mandatory inNativeConstraints:nativeMandatory];
+
+  webrtc::MediaConstraintsInterface::Constraints nativeOptional =
+      nativeConstraints->GetOptional();
+  [self expectConstraints:optional inNativeConstraints:nativeOptional];
+}
+
+- (void)expectConstraints:(NSDictionary *)constraints
+      inNativeConstraints:
+    (webrtc::MediaConstraintsInterface::Constraints)nativeConstraints {
+  EXPECT_EQ(constraints.count, nativeConstraints.size());
+
+  for (NSString *key in constraints) {
+    NSString *value = constraints[key];
+
+    std::string nativeValue;
+    bool found = nativeConstraints.FindFirst(key.stdString, &nativeValue);
+    EXPECT_TRUE(found);
+    EXPECT_EQ(value.stdString, nativeValue);
+  }
+}
+
+@end
+
+TEST(RTCMediaConstraintsTest, MediaConstraintsTest) {
+  @autoreleasepool {
+    RTCMediaConstraintsTest *test = [[RTCMediaConstraintsTest alloc] init];
+    [test testMediaConstraints];
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/api/objctests/RTCSessionDescriptionTest.mm b/media/webrtc/trunk/webrtc/api/objctests/RTCSessionDescriptionTest.mm
new file mode 100644
index 0000000000..2404dedd3a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/api/objctests/RTCSessionDescriptionTest.mm
@@ -0,0 +1,144 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCSessionDescription.h"
+#import "webrtc/api/objc/RTCSessionDescription+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCSessionDescriptionTest : NSObject
+- (void)testSessionDescriptionConversion;
+- (void)testInitFromNativeSessionDescription;
+@end
+
+@implementation RTCSessionDescriptionTest
+
+/**
+ * Test conversion of an Objective-C RTCSessionDescription to a native
+ * SessionDescriptionInterface (based on the types and SDP strings being equal).
+ */
+- (void)testSessionDescriptionConversion {
+  RTCSessionDescription *description =
+      [[RTCSessionDescription alloc] initWithType:RTCSdpTypeAnswer
+                                              sdp:[self sdp]];
+
+  webrtc::SessionDescriptionInterface *nativeDescription =
+      description.nativeDescription;
+
+  EXPECT_EQ(RTCSdpTypeAnswer,
+      [RTCSessionDescription typeForString:nativeDescription->type()]);
+
+  std::string sdp;
+  nativeDescription->ToString(&sdp);
+  EXPECT_EQ([self sdp].stdString, sdp);
+}
+
+- (void)testInitFromNativeSessionDescription {
+  webrtc::SessionDescriptionInterface *nativeDescription;
+
+  nativeDescription = webrtc::CreateSessionDescription(
+      webrtc::SessionDescriptionInterface::kAnswer,
+      [self sdp].stdString,
+      nullptr);
+
+  RTCSessionDescription *description =
+      [[RTCSessionDescription alloc] initWithNativeDescription:
+      nativeDescription];
+  EXPECT_EQ(webrtc::SessionDescriptionInterface::kAnswer,
+      [RTCSessionDescription stringForType:description.type]);
+  EXPECT_TRUE([[self sdp] isEqualToString:description.sdp]);
+}
+
+- (NSString *)sdp {
+    return @"v=0\r\n"
+           "o=- 5319989746393411314 2 IN IP4 127.0.0.1\r\n"
+           "s=-\r\n"
+           "t=0 0\r\n"
+           "a=group:BUNDLE audio video\r\n"
+           "a=msid-semantic: WMS ARDAMS\r\n"
+           "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 126\r\n"
+           "c=IN IP4 0.0.0.0\r\n"
+           "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+           "a=ice-ufrag:f3o+0HG7l9nwIWFY\r\n"
+           "a=ice-pwd:VDctmJNCptR2TB7+meDpw7w5\r\n"
+           "a=fingerprint:sha-256 A9:D5:8D:A8:69:22:39:60:92:AD:94:1A:22:2D:5E:"
+           "A5:4A:A9:18:C2:35:5D:46:5E:59:BD:1C:AF:38:9F:E6:E1\r\n"
+           "a=setup:active\r\n"
+           "a=mid:audio\r\n"
+           "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n"
+           "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/"
+           "abs-send-time\r\n"
+           "a=sendrecv\r\n"
+           "a=rtcp-mux\r\n"
+           "a=rtpmap:111 opus/48000/2\r\n"
+           "a=fmtp:111 minptime=10; useinbandfec=1\r\n"
+           "a=rtpmap:103 ISAC/16000\r\n"
+           "a=rtpmap:9 G722/8000\r\n"
+           "a=rtpmap:0 PCMU/8000\r\n"
+           "a=rtpmap:8 PCMA/8000\r\n"
+           "a=rtpmap:126 telephone-event/8000\r\n"
+           "a=maxptime:60\r\n"
+           "a=ssrc:1504474588 cname:V+FdIC5AJpxLhdYQ\r\n"
+           "a=ssrc:1504474588 msid:ARDAMS ARDAMSa0\r\n"
+           "a=ssrc:1504474588 mslabel:ARDAMS\r\n"
+           "a=ssrc:1504474588 label:ARDAMSa0\r\n"
+           "m=video 9 UDP/TLS/RTP/SAVPF 100 116 117 96\r\n"
+           "c=IN IP4 0.0.0.0\r\n"
+           "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+           "a=ice-ufrag:f3o+0HG7l9nwIWFY\r\n"
+           "a=ice-pwd:VDctmJNCptR2TB7+meDpw7w5\r\n"
+           "a=fingerprint:sha-256 A9:D5:8D:A8:69:22:39:60:92:AD:94:1A:22:2D:5E:"
+           "A5:4A:A9:18:C2:35:5D:46:5E:59:BD:1C:AF:38:9F:E6:E1\r\n"
+           "a=setup:active\r\n"
+           "a=mid:video\r\n"
+           "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\n"
+           "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/"
+           "abs-send-time\r\n"
+           "a=extmap:4 urn:3gpp:video-orientation\r\n"
+           "a=sendrecv\r\n"
+           "a=rtcp-mux\r\n"
+           "a=rtpmap:100 VP8/90000\r\n"
+           "a=rtcp-fb:100 ccm fir\r\n"
+           "a=rtcp-fb:100 nack\r\n"
+           "a=rtcp-fb:100 nack pli\r\n"
+           "a=rtcp-fb:100 goog-remb\r\n"
+           "a=rtpmap:116 red/90000\r\n"
+           "a=rtpmap:117 ulpfec/90000\r\n"
+           "a=rtpmap:96 rtx/90000\r\n"
+           "a=fmtp:96 apt=100\r\n"
+           "a=ssrc-group:FID 498297514 1644357692\r\n"
+           "a=ssrc:498297514 cname:V+FdIC5AJpxLhdYQ\r\n"
+           "a=ssrc:498297514 msid:ARDAMS ARDAMSv0\r\n"
+           "a=ssrc:498297514 mslabel:ARDAMS\r\n"
+           "a=ssrc:498297514 label:ARDAMSv0\r\n"
+           "a=ssrc:1644357692 cname:V+FdIC5AJpxLhdYQ\r\n"
+           "a=ssrc:1644357692 msid:ARDAMS ARDAMSv0\r\n"
+           "a=ssrc:1644357692 mslabel:ARDAMS\r\n"
+           "a=ssrc:1644357692 label:ARDAMSv0\r\n";
+}
+
+@end
+
+TEST(RTCSessionDescriptionTest, SessionDescriptionConversionTest) {
+  @autoreleasepool {
+    RTCSessionDescriptionTest *test = [[RTCSessionDescriptionTest alloc] init];
+    [test testSessionDescriptionConversion];
+  }
+}
+
+TEST(RTCSessionDescriptionTest, InitFromSessionDescriptionTest) {
+  @autoreleasepool {
+    RTCSessionDescriptionTest *test = [[RTCSessionDescriptionTest alloc] init];
+    [test testInitFromNativeSessionDescription];
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/audio/BUILD.gn b/media/webrtc/trunk/webrtc/audio/BUILD.gn
new file mode 100644
index 0000000000..5a9902eac1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/BUILD.gn
@@ -0,0 +1,38 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+
+source_set("audio") {
+  sources = [
+    "audio_receive_stream.cc",
+    "audio_receive_stream.h",
+    "audio_send_stream.cc",
+    "audio_send_stream.h",
+    "audio_sink.h",
+    "audio_state.cc",
+    "audio_state.h",
+    "conversion.h",
+    "scoped_voe_interface.h",
+  ]
+
+  configs += [ "..:common_config" ]
+  public_configs = [ "..:common_inherited_config" ]
+
+  if (is_clang) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+
+  deps = [
+    "..:webrtc_common",
+    "../system_wrappers",
+    "../voice_engine",
+  ]
+}
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/source/OWNERS b/media/webrtc/trunk/webrtc/audio/OWNERS
similarity index 68%
rename from media/webrtc/trunk/webrtc/modules/media_file/source/OWNERS
rename to media/webrtc/trunk/webrtc/audio/OWNERS
index 3ee6b4bf5f..f0cc72a885 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/source/OWNERS
+++ b/media/webrtc/trunk/webrtc/audio/OWNERS
@@ -1,5 +1,9 @@
+solenberg@webrtc.org
+tina.legrand@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
 per-file *.gyp=*
 per-file *.gypi=*
+
+per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/audio/audio_receive_stream.cc b/media/webrtc/trunk/webrtc/audio/audio_receive_stream.cc
new file mode 100644
index 0000000000..64d008326d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_receive_stream.cc
@@ -0,0 +1,256 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/audio/audio_receive_stream.h"
+
+#include <string>
+#include <utility>
+
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/audio/conversion.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/voice_engine/channel_proxy.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_neteq_stats.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
+
+namespace webrtc {
+namespace {
+
+bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) {
+  if (!config.rtp.transport_cc) {
+    return false;
+  }
+  for (const auto& extension : config.rtp.extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      return true;
+    }
+  }
+  return false;
+}
+}  // namespace
+
+std::string AudioReceiveStream::Config::Rtp::ToString() const {
+  std::stringstream ss;
+  ss << "{remote_ssrc: " << remote_ssrc;
+  ss << ", local_ssrc: " << local_ssrc;
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1) {
+      ss << ", ";
+    }
+  }
+  ss << ']';
+  ss << '}';
+  return ss.str();
+}
+
+std::string AudioReceiveStream::Config::ToString() const {
+  std::stringstream ss;
+  ss << "{rtp: " << rtp.ToString();
+  ss << ", receive_transport: "
+     << (receive_transport ? "(Transport)" : "nullptr");
+  ss << ", rtcp_send_transport: "
+     << (rtcp_send_transport ? "(Transport)" : "nullptr");
+  ss << ", voe_channel_id: " << voe_channel_id;
+  if (!sync_group.empty()) {
+    ss << ", sync_group: " << sync_group;
+  }
+  ss << ", combined_audio_video_bwe: "
+     << (combined_audio_video_bwe ? "true" : "false");
+  ss << '}';
+  return ss.str();
+}
+
+namespace internal {
+AudioReceiveStream::AudioReceiveStream(
+    CongestionController* congestion_controller,
+    const webrtc::AudioReceiveStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+    : config_(config),
+      audio_state_(audio_state),
+      rtp_header_parser_(RtpHeaderParser::Create()) {
+  LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
+  RTC_DCHECK_NE(config_.voe_channel_id, -1);
+  RTC_DCHECK(audio_state_.get());
+  RTC_DCHECK(congestion_controller);
+  RTC_DCHECK(rtp_header_parser_);
+
+  VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
+  channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+  channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
+  for (const auto& extension : config.rtp.extensions) {
+    if (extension.name == RtpExtension::kAudioLevel) {
+      channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
+      bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionAudioLevel, extension.id);
+      RTC_DCHECK(registered);
+    } else if (extension.name == RtpExtension::kAbsSendTime) {
+      channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
+      bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionAbsoluteSendTime, extension.id);
+      RTC_DCHECK(registered);
+    } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionTransportSequenceNumber, extension.id);
+      RTC_DCHECK(registered);
+    } else {
+      RTC_NOTREACHED() << "Unsupported RTP extension.";
+    }
+  }
+  // Configure bandwidth estimation.
+  channel_proxy_->SetCongestionControlObjects(
+      nullptr, nullptr, congestion_controller->packet_router());
+  if (config.combined_audio_video_bwe) {
+    if (UseSendSideBwe(config)) {
+      remote_bitrate_estimator_ =
+          congestion_controller->GetRemoteBitrateEstimator(true);
+    } else {
+      remote_bitrate_estimator_ =
+          congestion_controller->GetRemoteBitrateEstimator(false);
+    }
+    RTC_DCHECK(remote_bitrate_estimator_);
+  }
+}
+
+AudioReceiveStream::~AudioReceiveStream() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
+  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+  if (remote_bitrate_estimator_) {
+    remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
+  }
+}
+
+void AudioReceiveStream::Start() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void AudioReceiveStream::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void AudioReceiveStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  return false;
+}
+
+bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                    size_t length,
+                                    const PacketTime& packet_time) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  RTPHeader header;
+  if (!rtp_header_parser_->Parse(packet, length, &header)) {
+    return false;
+  }
+
+  // Only forward if the parsed header has one of the headers necessary for
+  // bandwidth estimation. RTP timestamps has different rates for audio and
+  // video and shouldn't be mixed.
+  if (remote_bitrate_estimator_ &&
+      (header.extension.hasAbsoluteSendTime ||
+       header.extension.hasTransportSequenceNumber)) {
+    int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
+    if (packet_time.timestamp >= 0)
+      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+    size_t payload_size = length - header.headerLength;
+    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
+                                              header, false);
+  }
+  return true;
+}
+
+webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  webrtc::AudioReceiveStream::Stats stats;
+  stats.remote_ssrc = config_.rtp.remote_ssrc;
+  ScopedVoEInterface<VoECodec> codec(voice_engine());
+
+  webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
+  webrtc::CodecInst codec_inst = {0};
+  if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
+    return stats;
+  }
+
+  stats.bytes_rcvd = call_stats.bytesReceived;
+  stats.packets_rcvd = call_stats.packetsReceived;
+  stats.packets_lost = call_stats.cumulativeLost;
+  stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
+  stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
+  if (codec_inst.pltype != -1) {
+    stats.codec_name = codec_inst.plname;
+  }
+  stats.ext_seqnum = call_stats.extendedMax;
+  if (codec_inst.plfreq / 1000 > 0) {
+    stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
+  }
+  stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
+  stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
+
+  // Get jitter buffer and total delay (alg + jitter + playout) stats.
+  auto ns = channel_proxy_->GetNetworkStatistics();
+  stats.jitter_buffer_ms = ns.currentBufferSize;
+  stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+  stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
+  stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
+  stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
+  stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
+  stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+
+  auto ds = channel_proxy_->GetDecodingCallStatistics();
+  stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
+  stats.decoding_calls_to_neteq = ds.calls_to_neteq;
+  stats.decoding_normal = ds.decoded_normal;
+  stats.decoding_plc = ds.decoded_plc;
+  stats.decoding_cng = ds.decoded_cng;
+  stats.decoding_plc_cng = ds.decoded_plc_cng;
+
+  return stats;
+}
+
+void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel_proxy_->SetSink(std::move(sink));
+}
+
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return config_;
+}
+
+VoiceEngine* AudioReceiveStream::voice_engine() const {
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  VoiceEngine* voice_engine = audio_state->voice_engine();
+  RTC_DCHECK(voice_engine);
+  return voice_engine;
+}
+}  // namespace internal
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/audio/audio_receive_stream.h b/media/webrtc/trunk/webrtc/audio/audio_receive_stream.h
new file mode 100644
index 0000000000..4940c6a64c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_receive_stream.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
+#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
+
+#include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio_state.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+
+namespace webrtc {
+class CongestionController;
+class RemoteBitrateEstimator;
+
+namespace voe {
+class ChannelProxy;
+}  // namespace voe
+
+namespace internal {
+
+class AudioReceiveStream final : public webrtc::AudioReceiveStream {
+ public:
+  AudioReceiveStream(CongestionController* congestion_controller,
+                     const webrtc::AudioReceiveStream::Config& config,
+                     const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
+  ~AudioReceiveStream() override;
+
+  // webrtc::ReceiveStream implementation.
+  void Start() override;
+  void Stop() override;
+  void SignalNetworkState(NetworkState state) override;
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+  bool DeliverRtp(const uint8_t* packet,
+                  size_t length,
+                  const PacketTime& packet_time) override;
+
+  // webrtc::AudioReceiveStream implementation.
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+
+  void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
+
+  const webrtc::AudioReceiveStream::Config& config() const;
+
+ private:
+  VoiceEngine* voice_engine() const;
+
+  rtc::ThreadChecker thread_checker_;
+  RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
+  const webrtc::AudioReceiveStream::Config config_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+  rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/audio/audio_receive_stream_unittest.cc b/media/webrtc/trunk/webrtc/audio/audio_receive_stream_unittest.cc
new file mode 100644
index 0000000000..eb008b3045
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_receive_stream_unittest.cc
@@ -0,0 +1,328 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio/audio_receive_stream.h"
+#include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/mock_voe_channel_proxy.h"
+#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video/call_stats.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using testing::_;
+using testing::Return;
+
+AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
+  AudioDecodingCallStats audio_decode_stats;
+  audio_decode_stats.calls_to_silence_generator = 234;
+  audio_decode_stats.calls_to_neteq = 567;
+  audio_decode_stats.decoded_normal = 890;
+  audio_decode_stats.decoded_plc = 123;
+  audio_decode_stats.decoded_cng = 456;
+  audio_decode_stats.decoded_plc_cng = 789;
+  return audio_decode_stats;
+}
+
+const int kChannelId = 2;
+const uint32_t kRemoteSsrc = 1234;
+const uint32_t kLocalSsrc = 5678;
+const size_t kOneByteExtensionHeaderLength = 4;
+const size_t kOneByteExtensionLength = 4;
+const int kAbsSendTimeId = 2;
+const int kAudioLevelId = 3;
+const int kTransportSequenceNumberId = 4;
+const int kJitterBufferDelay = -7;
+const int kPlayoutBufferDelay = 302;
+const unsigned int kSpeechOutputLevel = 99;
+const CallStatistics kCallStats = {
+    345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
+const CodecInst kCodecInst = {
+    123, "codec_name_recv", 96000, -187, 0, -103};
+const NetworkStatistics kNetworkStats = {
+    123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
+const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
+
+struct ConfigHelper {
+  ConfigHelper()
+      : simulated_clock_(123456),
+        call_stats_(&simulated_clock_),
+        congestion_controller_(&process_thread_,
+                               &call_stats_,
+                               &bitrate_observer_) {
+    using testing::Invoke;
+
+    EXPECT_CALL(voice_engine_,
+        RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_,
+        DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
+    AudioState::Config config;
+    config.voice_engine = &voice_engine_;
+    audio_state_ = AudioState::Create(config);
+
+    EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
+        .WillOnce(Invoke([this](int channel_id) {
+          EXPECT_FALSE(channel_proxy_);
+          channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+          EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
+                  .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
+                  .Times(1);
+          EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
+                                           nullptr, nullptr, &packet_router_))
+              .Times(1);
+          EXPECT_CALL(congestion_controller_, packet_router())
+              .WillOnce(Return(&packet_router_));
+          EXPECT_CALL(*channel_proxy_,
+                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+              .Times(1);
+          return channel_proxy_;
+        }));
+    stream_config_.voe_channel_id = kChannelId;
+    stream_config_.rtp.local_ssrc = kLocalSsrc;
+    stream_config_.rtp.remote_ssrc = kRemoteSsrc;
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+  }
+
+  MockCongestionController* congestion_controller() {
+    return &congestion_controller_;
+  }
+  MockRemoteBitrateEstimator* remote_bitrate_estimator() {
+    return &remote_bitrate_estimator_;
+  }
+  AudioReceiveStream::Config& config() { return stream_config_; }
+  rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+  MockVoiceEngine& voice_engine() { return voice_engine_; }
+
+  void SetupMockForBweFeedback(bool send_side_bwe) {
+    EXPECT_CALL(congestion_controller_,
+                GetRemoteBitrateEstimator(send_side_bwe))
+        .WillOnce(Return(&remote_bitrate_estimator_));
+    EXPECT_CALL(remote_bitrate_estimator_,
+                RemoveStream(stream_config_.rtp.remote_ssrc));
+  }
+
+  void SetupMockForGetStats() {
+    using testing::DoAll;
+    using testing::SetArgReferee;
+
+    ASSERT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
+        .WillOnce(Return(kCallStats));
+    EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
+        .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
+    EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
+        .WillOnce(Return(kSpeechOutputLevel));
+    EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
+        .WillOnce(Return(kNetworkStats));
+    EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
+        .WillOnce(Return(kAudioDecodeStats));
+
+    EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
+        .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
+  }
+
+ private:
+  SimulatedClock simulated_clock_;
+  CallStats call_stats_;
+  PacketRouter packet_router_;
+  testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+  testing::NiceMock<MockProcessThread> process_thread_;
+  MockCongestionController congestion_controller_;
+  MockRemoteBitrateEstimator remote_bitrate_estimator_;
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
+  rtc::scoped_refptr<AudioState> audio_state_;
+  AudioReceiveStream::Config stream_config_;
+  testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
+};
+
+void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
+                           int id,
+                           uint32_t extension_value,
+                           size_t value_length) {
+  const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
+  ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
+  it += 2;
+
+  ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
+  it += 2;
+  const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
+  uint32_t shifted_value = extension_value
+                           << (8 * (kExtensionDataLength - value_length));
+  *it = (id << 4) + (value_length - 1);
+  ++it;
+  ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
+                                                             shifted_value);
+}
+
+std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
+    int extension_id,
+    uint32_t extension_value,
+    size_t value_length) {
+  std::vector<uint8_t> header;
+  header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
+                kOneByteExtensionLength);
+  header[0] = 0x80;   // Version 2.
+  header[0] |= 0x10;  // Set extension bit.
+  header[1] = 100;    // Payload type.
+  header[1] |= 0x80;  // Marker bit is set.
+  ByteWriter<uint16_t>::WriteBigEndian(&header[2], 0x1234);  // Sequence number.
+  ByteWriter<uint32_t>::WriteBigEndian(&header[4], 0x5678);  // Timestamp.
+  ByteWriter<uint32_t>::WriteBigEndian(&header[8], 0x4321);  // SSRC.
+
+  BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
+                        extension_value, value_length);
+  return header;
+}
+}  // namespace
+
+TEST(AudioReceiveStreamTest, ConfigToString) {
+  AudioReceiveStream::Config config;
+  config.rtp.remote_ssrc = kRemoteSsrc;
+  config.rtp.local_ssrc = kLocalSsrc;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+  config.voe_channel_id = kChannelId;
+  config.combined_audio_video_bwe = true;
+  EXPECT_EQ(
+      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
+      "receive_transport: nullptr, rtcp_send_transport: nullptr, "
+      "voe_channel_id: 2, combined_audio_video_bwe: true}",
+      config.ToString());
+}
+
+TEST(AudioReceiveStreamTest, ConstructDestruct) {
+  ConfigHelper helper;
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+}
+
+MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
+  return arg.extension.hasAbsoluteSendTime ==
+             expected_extension.hasAbsoluteSendTime &&
+         arg.extension.absoluteSendTime ==
+             expected_extension.absoluteSendTime &&
+         arg.extension.hasTransportSequenceNumber ==
+             expected_extension.hasTransportSequenceNumber &&
+         arg.extension.transportSequenceNumber ==
+             expected_extension.transportSequenceNumber;
+}
+
+TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
+  ConfigHelper helper;
+  helper.config().combined_audio_video_bwe = true;
+  helper.SetupMockForBweFeedback(false);
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+  const int kAbsSendTimeValue = 1234;
+  std::vector<uint8_t> rtp_packet =
+      CreateRtpHeaderWithOneByteExtension(kAbsSendTimeId, kAbsSendTimeValue, 3);
+  PacketTime packet_time(5678000, 0);
+  const size_t kExpectedHeaderLength = 20;
+  RTPHeaderExtension expected_extension;
+  expected_extension.hasAbsoluteSendTime = true;
+  expected_extension.absoluteSendTime = kAbsSendTimeValue;
+  EXPECT_CALL(*helper.remote_bitrate_estimator(),
+              IncomingPacket(packet_time.timestamp / 1000,
+                             rtp_packet.size() - kExpectedHeaderLength,
+                             VerifyHeaderExtension(expected_extension), false))
+      .Times(1);
+  EXPECT_TRUE(
+      recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
+}
+
+TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
+  ConfigHelper helper;
+  helper.config().combined_audio_video_bwe = true;
+  helper.config().rtp.transport_cc = true;
+  helper.config().rtp.extensions.push_back(RtpExtension(
+      RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+  helper.SetupMockForBweFeedback(true);
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+  const int kTransportSequenceNumberValue = 1234;
+  std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
+      kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
+  PacketTime packet_time(5678000, 0);
+  const size_t kExpectedHeaderLength = 20;
+  RTPHeaderExtension expected_extension;
+  expected_extension.hasTransportSequenceNumber = true;
+  expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
+  EXPECT_CALL(*helper.remote_bitrate_estimator(),
+              IncomingPacket(packet_time.timestamp / 1000,
+                             rtp_packet.size() - kExpectedHeaderLength,
+                             VerifyHeaderExtension(expected_extension), false))
+      .Times(1);
+  EXPECT_TRUE(
+      recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
+}
+
+TEST(AudioReceiveStreamTest, GetStats) {
+  ConfigHelper helper;
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+  helper.SetupMockForGetStats();
+  AudioReceiveStream::Stats stats = recv_stream.GetStats();
+  EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
+  EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
+  EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
+            stats.packets_rcvd);
+  EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
+  EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
+  EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
+  EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
+  EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000),
+            stats.jitter_ms);
+  EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
+  EXPECT_EQ(kNetworkStats.preferredBufferSize,
+            stats.jitter_buffer_preferred_ms);
+  EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
+            stats.delay_estimate_ms);
+  EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
+  EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
+  EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
+            stats.speech_expand_rate);
+  EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
+            stats.secondary_decoded_rate);
+  EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
+            stats.accelerate_rate);
+  EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
+            stats.preemptive_expand_rate);
+  EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
+            stats.decoding_calls_to_silence_generator);
+  EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
+  EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
+  EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
+  EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
+  EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
+  EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
+            stats.capture_start_ntp_time_ms);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/audio/audio_send_stream.cc b/media/webrtc/trunk/webrtc/audio/audio_send_stream.cc
new file mode 100644
index 0000000000..35a65521dd
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_send_stream.cc
@@ -0,0 +1,221 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/audio/audio_send_stream.h"
+
+#include <string>
+
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/audio/conversion.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/voice_engine/channel_proxy.h"
+#include "webrtc/voice_engine/include/voe_audio_processing.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
+
+namespace webrtc {
+std::string AudioSendStream::Config::Rtp::ToString() const {
+  std::stringstream ss;
+  ss << "{ssrc: " << ssrc;
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1) {
+      ss << ", ";
+    }
+  }
+  ss << ']';
+  ss << ", c_name: " << c_name;
+  ss << '}';
+  return ss.str();
+}
+
+std::string AudioSendStream::Config::ToString() const {
+  std::stringstream ss;
+  ss << "{rtp: " << rtp.ToString();
+  ss << ", voe_channel_id: " << voe_channel_id;
+  // TODO(solenberg): Encoder config.
+  ss << ", cng_payload_type: " << cng_payload_type;
+  ss << ", red_payload_type: " << red_payload_type;
+  ss << '}';
+  return ss.str();
+}
+
+namespace internal {
+AudioSendStream::AudioSendStream(
+    const webrtc::AudioSendStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    CongestionController* congestion_controller)
+    : config_(config), audio_state_(audio_state) {
+  LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
+  RTC_DCHECK_NE(config_.voe_channel_id, -1);
+  RTC_DCHECK(audio_state_.get());
+  RTC_DCHECK(congestion_controller);
+
+  VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
+  channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+  channel_proxy_->SetCongestionControlObjects(
+      congestion_controller->pacer(),
+      congestion_controller->GetTransportFeedbackObserver(),
+      congestion_controller->packet_router());
+  channel_proxy_->SetRTCPStatus(true);
+  channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
+  channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+
+  for (const auto& extension : config.rtp.extensions) {
+    if (extension.name == RtpExtension::kAbsSendTime) {
+      channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
+    } else if (extension.name == RtpExtension::kAudioLevel) {
+      channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
+    } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
+    } else {
+      RTC_NOTREACHED() << "Registering unsupported RTP extension.";
+    }
+  }
+}
+
+AudioSendStream::~AudioSendStream() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+}
+
+void AudioSendStream::Start() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void AudioSendStream::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+void AudioSendStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  return false;
+}
+
+bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
+                                         uint32_t duration_ms) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
+         channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  webrtc::AudioSendStream::Stats stats;
+  stats.local_ssrc = config_.rtp.ssrc;
+  ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
+  ScopedVoEInterface<VoECodec> codec(voice_engine());
+  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
+
+  webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
+  stats.bytes_sent = call_stats.bytesSent;
+  stats.packets_sent = call_stats.packetsSent;
+  // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+  // returns 0 to indicate an error value.
+  if (call_stats.rttMs > 0) {
+    stats.rtt_ms = call_stats.rttMs;
+  }
+  // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
+  //                  implementation.
+  stats.aec_quality_min = -1;
+
+  webrtc::CodecInst codec_inst = {0};
+  if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
+    RTC_DCHECK_NE(codec_inst.pltype, -1);
+    stats.codec_name = codec_inst.plname;
+
+    // Get data from the last remote RTCP report.
+    for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
+      // Lookup report for send ssrc only.
+      if (block.source_SSRC == stats.local_ssrc) {
+        stats.packets_lost = block.cumulative_num_packets_lost;
+        stats.fraction_lost = Q8ToFloat(block.fraction_lost);
+        stats.ext_seqnum = block.extended_highest_sequence_number;
+        // Convert samples to milliseconds.
+        if (codec_inst.plfreq / 1000 > 0) {
+          stats.jitter_ms =
+              block.interarrival_jitter / (codec_inst.plfreq / 1000);
+        }
+        break;
+      }
+    }
+  }
+
+  // Local speech level.
+  {
+    unsigned int level = 0;
+    int error = volume->GetSpeechInputLevelFullRange(level);
+    RTC_DCHECK_EQ(0, error);
+    stats.audio_level = static_cast<int32_t>(level);
+  }
+
+  bool echo_metrics_on = false;
+  int error = processing->GetEcMetricsStatus(echo_metrics_on);
+  RTC_DCHECK_EQ(0, error);
+  if (echo_metrics_on) {
+    // These can also be negative, but in practice -1 is only used to signal
+    // insufficient data, since the resolution is limited to multiples of 4 ms.
+    int median = -1;
+    int std = -1;
+    float dummy = 0.0f;
+    error = processing->GetEcDelayMetrics(median, std, dummy);
+    RTC_DCHECK_EQ(0, error);
+    stats.echo_delay_median_ms = median;
+    stats.echo_delay_std_ms = std;
+
+    // These can take on valid negative values, so use the lowest possible level
+    // as default rather than -1.
+    int erl = -100;
+    int erle = -100;
+    int dummy1 = 0;
+    int dummy2 = 0;
+    error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
+    RTC_DCHECK_EQ(0, error);
+    stats.echo_return_loss = erl;
+    stats.echo_return_loss_enhancement = erle;
+  }
+
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  stats.typing_noise_detected = audio_state->typing_noise_detected();
+
+  return stats;
+}
+
+const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return config_;
+}
+
+VoiceEngine* AudioSendStream::voice_engine() const {
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  VoiceEngine* voice_engine = audio_state->voice_engine();
+  RTC_DCHECK(voice_engine);
+  return voice_engine;
+}
+}  // namespace internal
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/audio/audio_send_stream.h b/media/webrtc/trunk/webrtc/audio/audio_send_stream.h
new file mode 100644
index 0000000000..8b96350590
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_send_stream.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
+#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
+
+#include "webrtc/audio_send_stream.h"
+#include "webrtc/audio_state.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+class CongestionController;
+class VoiceEngine;
+
+namespace voe {
+class ChannelProxy;
+}  // namespace voe
+
+namespace internal {
+class AudioSendStream final : public webrtc::AudioSendStream {
+ public:
+  AudioSendStream(const webrtc::AudioSendStream::Config& config,
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  CongestionController* congestion_controller);
+  ~AudioSendStream() override;
+
+  // webrtc::SendStream implementation.
+  void Start() override;
+  void Stop() override;
+  void SignalNetworkState(NetworkState state) override;
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+
+  // webrtc::AudioSendStream implementation.
+  bool SendTelephoneEvent(int payload_type, uint8_t event,
+                          uint32_t duration_ms) override;
+  webrtc::AudioSendStream::Stats GetStats() const override;
+
+  const webrtc::AudioSendStream::Config& config() const;
+
+ private:
+  VoiceEngine* voice_engine() const;
+
+  rtc::ThreadChecker thread_checker_;
+  const webrtc::AudioSendStream::Config config_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/audio/audio_send_stream_unittest.cc b/media/webrtc/trunk/webrtc/audio/audio_send_stream_unittest.cc
new file mode 100644
index 0000000000..466c1571ac
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_send_stream_unittest.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio/audio_send_stream.h"
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/audio/conversion.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/test/mock_voe_channel_proxy.h"
+#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video/call_stats.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using testing::_;
+using testing::Return;
+
+const int kChannelId = 1;
+const uint32_t kSsrc = 1234;
+const char* kCName = "foo_name";
+const int kAudioLevelId = 2;
+const int kAbsSendTimeId = 3;
+const int kTransportSequenceNumberId = 4;
+const int kEchoDelayMedian = 254;
+const int kEchoDelayStdDev = -3;
+const int kEchoReturnLoss = -65;
+const int kEchoReturnLossEnhancement = 101;
+const unsigned int kSpeechInputLevel = 96;
+const CallStatistics kCallStats = {
+    1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
+const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
+const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
+const int kTelephoneEventPayloadType = 123;
+const uint8_t kTelephoneEventCode = 45;
+const uint32_t kTelephoneEventDuration = 6789;
+
+struct ConfigHelper {
+  ConfigHelper()
+      : stream_config_(nullptr),
+        call_stats_(Clock::GetRealTimeClock()),
+        process_thread_(ProcessThread::Create("AudioTestThread")),
+        congestion_controller_(process_thread_.get(),
+                               &call_stats_,
+                               &bitrate_observer_) {
+    using testing::Invoke;
+    using testing::StrEq;
+
+    EXPECT_CALL(voice_engine_,
+        RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_,
+        DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
+    AudioState::Config config;
+    config.voice_engine = &voice_engine_;
+    audio_state_ = AudioState::Create(config);
+
+    EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
+        .WillOnce(Invoke([this](int channel_id) {
+          EXPECT_FALSE(channel_proxy_);
+          channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+          EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
+          EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
+          EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
+          EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
+                                           kTransportSequenceNumberId))
+              .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+                      SetCongestionControlObjects(
+                          congestion_controller_.pacer(),
+                          congestion_controller_.GetTransportFeedbackObserver(),
+                          congestion_controller_.packet_router()))
+              .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+              .Times(1);
+          return channel_proxy_;
+        }));
+    stream_config_.voe_channel_id = kChannelId;
+    stream_config_.rtp.ssrc = kSsrc;
+    stream_config_.rtp.c_name = kCName;
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+    stream_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+  }
+
+  AudioSendStream::Config& config() { return stream_config_; }
+  rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+  CongestionController* congestion_controller() {
+    return &congestion_controller_;
+  }
+
+  void SetupMockForSendTelephoneEvent() {
+    EXPECT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_,
+        SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
+            .WillOnce(Return(true));
+    EXPECT_CALL(*channel_proxy_,
+        SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
+            .WillOnce(Return(true));
+  }
+
+  void SetupMockForGetStats() {
+    using testing::DoAll;
+    using testing::SetArgReferee;
+
+    std::vector<ReportBlock> report_blocks;
+    webrtc::ReportBlock block = kReportBlock;
+    report_blocks.push_back(block);  // Has wrong SSRC.
+    block.source_SSRC = kSsrc;
+    report_blocks.push_back(block);  // Correct block.
+    block.fraction_lost = 0;
+    report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
+
+    EXPECT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
+        .WillRepeatedly(Return(kCallStats));
+    EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
+        .WillRepeatedly(Return(report_blocks));
+
+    EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
+        .WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
+    EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
+        .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
+    EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
+        .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
+    EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
+        .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
+                        SetArgReferee<1>(kEchoReturnLossEnhancement),
+                        Return(0)));
+    EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
+        .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
+                        SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
+  }
+
+ private:
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
+  rtc::scoped_refptr<AudioState> audio_state_;
+  AudioSendStream::Config stream_config_;
+  testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
+  CallStats call_stats_;
+  testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+  rtc::scoped_ptr<ProcessThread> process_thread_;
+  CongestionController congestion_controller_;
+};
+}  // namespace
+
+TEST(AudioSendStreamTest, ConfigToString) {
+  AudioSendStream::Config config(nullptr);
+  config.rtp.ssrc = kSsrc;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+  config.rtp.c_name = kCName;
+  config.voe_channel_id = kChannelId;
+  config.cng_payload_type = 42;
+  config.red_payload_type = 17;
+  EXPECT_EQ(
+      "{rtp: {ssrc: 1234, extensions: [{name: "
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
+      "c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
+      "red_payload_type: 17}",
+      config.ToString());
+}
+
+TEST(AudioSendStreamTest, ConstructDestruct) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
+}
+
+TEST(AudioSendStreamTest, SendTelephoneEvent) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
+  helper.SetupMockForSendTelephoneEvent();
+  EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
+      kTelephoneEventCode, kTelephoneEventDuration));
+}
+
+TEST(AudioSendStreamTest, GetStats) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
+  helper.SetupMockForGetStats();
+  AudioSendStream::Stats stats = send_stream.GetStats();
+  EXPECT_EQ(kSsrc, stats.local_ssrc);
+  EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
+  EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
+  EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
+            stats.packets_lost);
+  EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
+  EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
+  EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
+            stats.ext_seqnum);
+  EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
+                                 (kCodecInst.plfreq / 1000)),
+            stats.jitter_ms);
+  EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
+  EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
+  EXPECT_EQ(-1, stats.aec_quality_min);
+  EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms);
+  EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms);
+  EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss);
+  EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement);
+  EXPECT_FALSE(stats.typing_noise_detected);
+}
+
+TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
+  helper.SetupMockForGetStats();
+  EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+
+  internal::AudioState* internal_audio_state =
+      static_cast<internal::AudioState*>(helper.audio_state().get());
+  VoiceEngineObserver* voe_observer =
+      static_cast<VoiceEngineObserver*>(internal_audio_state);
+  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
+  EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
+  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
+  EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/audio/audio_sink.h b/media/webrtc/trunk/webrtc/audio/audio_sink.h
new file mode 100644
index 0000000000..999644f4ce
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_sink.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_SINK_H_
+#define WEBRTC_AUDIO_AUDIO_SINK_H_
+
+#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
+// Avoid conflict with format_macros.h.
+#define __STDC_FORMAT_MACROS
+#endif
+
+#include <inttypes.h>
+#include <stddef.h>
+
+namespace webrtc {
+
+// Represents a simple push audio sink.
+class AudioSinkInterface {
+ public:
+  virtual ~AudioSinkInterface() {}
+
+  struct Data {
+    Data(int16_t* data,
+         size_t samples_per_channel,
+         int sample_rate,
+         size_t channels,
+         uint32_t timestamp)
+        : data(data),
+          samples_per_channel(samples_per_channel),
+          sample_rate(sample_rate),
+          channels(channels),
+          timestamp(timestamp) {}
+
+    int16_t* data;               // The actual 16bit audio data.
+    size_t samples_per_channel;  // Number of frames in the buffer.
+    int sample_rate;             // Sample rate in Hz.
+    size_t channels;             // Number of channels in the audio data.
+    uint32_t timestamp;          // The RTP timestamp of the first sample.
+  };
+
+  virtual void OnData(const Data& audio) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_SINK_H_
diff --git a/media/webrtc/trunk/webrtc/audio/audio_state.cc b/media/webrtc/trunk/webrtc/audio/audio_state.cc
new file mode 100644
index 0000000000..e63f97af2d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_state.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/audio/audio_state.h"
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/voice_engine/include/voe_errors.h"
+
+namespace webrtc {
+namespace internal {
+
+AudioState::AudioState(const AudioState::Config& config)
+    : config_(config), voe_base_(config.voice_engine) {
+  process_thread_checker_.DetachFromThread();
+  // Only one AudioState should be created per VoiceEngine.
+  RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
+}
+
+AudioState::~AudioState() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  voe_base_->DeRegisterVoiceEngineObserver();
+}
+
+VoiceEngine* AudioState::voice_engine() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return config_.voice_engine;
+}
+
+bool AudioState::typing_noise_detected() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  rtc::CritScope lock(&crit_sect_);
+  return typing_noise_detected_;
+}
+
+// Reference count; implementation copied from rtc::RefCountedObject.
+int AudioState::AddRef() const {
+  return rtc::AtomicOps::Increment(&ref_count_);
+}
+
+// Reference count; implementation copied from rtc::RefCountedObject.
+int AudioState::Release() const {
+  int count = rtc::AtomicOps::Decrement(&ref_count_);
+  if (!count) {
+    delete this;
+  }
+  return count;
+}
+
+void AudioState::CallbackOnError(int channel_id, int err_code) {
+  RTC_DCHECK(process_thread_checker_.CalledOnValidThread());
+
+  // All call sites in VoE, as of this writing, specify -1 as channel_id.
+  RTC_DCHECK(channel_id == -1);
+  LOG(LS_INFO) << "VoiceEngine error " << err_code << " reported on channel "
+               << channel_id << ".";
+  if (err_code == VE_TYPING_NOISE_WARNING) {
+    rtc::CritScope lock(&crit_sect_);
+    typing_noise_detected_ = true;
+  } else if (err_code == VE_TYPING_NOISE_OFF_WARNING) {
+    rtc::CritScope lock(&crit_sect_);
+    typing_noise_detected_ = false;
+  }
+}
+}  // namespace internal
+
+rtc::scoped_refptr<AudioState> AudioState::Create(
+    const AudioState::Config& config) {
+  return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/audio/audio_state.h b/media/webrtc/trunk/webrtc/audio/audio_state.h
new file mode 100644
index 0000000000..2cb83e4989
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_state.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_STATE_H_
+#define WEBRTC_AUDIO_AUDIO_STATE_H_
+
+#include "webrtc/audio_state.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+
+namespace webrtc {
+namespace internal {
+
+class AudioState final : public webrtc::AudioState,
+                         public webrtc::VoiceEngineObserver {
+ public:
+  explicit AudioState(const AudioState::Config& config);
+  ~AudioState() override;
+
+  VoiceEngine* voice_engine();
+  bool typing_noise_detected() const;
+
+ private:
+  // rtc::RefCountInterface implementation.
+  int AddRef() const override;
+  int Release() const override;
+
+  // webrtc::VoiceEngineObserver implementation.
+  void CallbackOnError(int channel_id, int err_code) override;
+
+  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker process_thread_checker_;
+  const webrtc::AudioState::Config config_;
+
+  // We hold one interface pointer to the VoE to make sure it is kept alive.
+  ScopedVoEInterface<VoEBase> voe_base_;
+
+  // The critical section isn't strictly needed in this case, but xSAN bots may
+  // trigger on unprotected cross-thread access.
+  mutable rtc::CriticalSection crit_sect_;
+  bool typing_noise_detected_ GUARDED_BY(crit_sect_) = false;
+
+  // Reference count; implementation copied from rtc::RefCountedObject.
+  mutable volatile int ref_count_ = 0;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_STATE_H_
diff --git a/media/webrtc/trunk/webrtc/audio/audio_state_unittest.cc b/media/webrtc/trunk/webrtc/audio/audio_state_unittest.cc
new file mode 100644
index 0000000000..11fbdb4a86
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/audio_state_unittest.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/test/mock_voice_engine.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+struct ConfigHelper {
+  ConfigHelper() {
+    EXPECT_CALL(voice_engine_,
+        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
+    EXPECT_CALL(voice_engine_,
+        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
+    config_.voice_engine = &voice_engine_;
+  }
+  AudioState::Config& config() { return config_; }
+  MockVoiceEngine& voice_engine() { return voice_engine_; }
+
+ private:
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
+  AudioState::Config config_;
+};
+}  // namespace
+
+TEST(AudioStateTest, Create) {
+  ConfigHelper helper;
+  rtc::scoped_refptr<AudioState> audio_state =
+      AudioState::Create(helper.config());
+  EXPECT_TRUE(audio_state.get());
+}
+
+TEST(AudioStateTest, ConstructDestruct) {
+  ConfigHelper helper;
+  rtc::scoped_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+}
+
+TEST(AudioStateTest, GetVoiceEngine) {
+  ConfigHelper helper;
+  rtc::scoped_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+  EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
+}
+
+TEST(AudioStateTest, TypingNoiseDetected) {
+  ConfigHelper helper;
+  rtc::scoped_ptr<internal::AudioState> audio_state(
+      new internal::AudioState(helper.config()));
+  VoiceEngineObserver* voe_observer =
+      static_cast<VoiceEngineObserver*>(audio_state.get());
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+
+  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+
+  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
+  EXPECT_TRUE(audio_state->typing_noise_detected());
+  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+  EXPECT_TRUE(audio_state->typing_noise_detected());
+
+  voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+  voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+  EXPECT_FALSE(audio_state->typing_noise_detected());
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/audio/conversion.h b/media/webrtc/trunk/webrtc/audio/conversion.h
new file mode 100644
index 0000000000..6ae32432d3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/conversion.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_CONVERSION_H_
+#define WEBRTC_AUDIO_CONVERSION_H_
+
+namespace webrtc {
+
+// Convert fixed point number with 8 bit fractional part, to floating point.
+inline float Q8ToFloat(uint32_t v) {
+  return static_cast<float>(v) / (1 << 8);
+}
+
+// Convert fixed point number with 14 bit fractional part, to floating point.
+inline float Q14ToFloat(uint32_t v) {
+  return static_cast<float>(v) / (1 << 14);
+}
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_CONVERSION_H_
diff --git a/media/webrtc/trunk/webrtc/audio/scoped_voe_interface.h b/media/webrtc/trunk/webrtc/audio/scoped_voe_interface.h
new file mode 100644
index 0000000000..1029337228
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/scoped_voe_interface.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
+#define WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+namespace internal {
+
+// Utility template for obtaining and holding a reference to a VoiceEngine
+// interface and making sure it is released when this object goes out of scope.
+template<class T> class ScopedVoEInterface {
+ public:
+  explicit ScopedVoEInterface(webrtc::VoiceEngine* e)
+      : ptr_(T::GetInterface(e)) {
+    RTC_DCHECK(ptr_);
+  }
+  ~ScopedVoEInterface() {
+    if (ptr_) {
+      ptr_->Release();
+    }
+  }
+  T* operator->() {
+    RTC_DCHECK(ptr_);
+    return ptr_;
+  }
+ private:
+  T* ptr_;
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/audio/webrtc_audio.gypi b/media/webrtc/trunk/webrtc/audio/webrtc_audio.gypi
new file mode 100644
index 0000000000..53b7d16b1a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio/webrtc_audio.gypi
@@ -0,0 +1,28 @@
+# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'variables': {
+    'webrtc_audio_dependencies': [
+      '<(webrtc_root)/common.gyp:webrtc_common',
+      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+      '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+    ],
+    'webrtc_audio_sources': [
+      'audio/audio_receive_stream.cc',
+      'audio/audio_receive_stream.h',
+      'audio/audio_send_stream.cc',
+      'audio/audio_send_stream.h',
+      'audio/audio_sink.h',
+      'audio/audio_state.cc',
+      'audio/audio_state.h',
+      'audio/conversion.h',
+      'audio/scoped_voe_interface.h',
+    ],
+  },
+}
diff --git a/media/webrtc/trunk/webrtc/audio_receive_stream.h b/media/webrtc/trunk/webrtc/audio_receive_stream.h
new file mode 100644
index 0000000000..8cab094f4b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio_receive_stream.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
+#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/config.h"
+#include "webrtc/stream.h"
+#include "webrtc/transport.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDecoder;
+class AudioSinkInterface;
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+class AudioReceiveStream : public ReceiveStream {
+ public:
+  struct Stats {
+    uint32_t remote_ssrc = 0;
+    int64_t bytes_rcvd = 0;
+    uint32_t packets_rcvd = 0;
+    uint32_t packets_lost = 0;
+    float fraction_lost = 0.0f;
+    std::string codec_name;
+    uint32_t ext_seqnum = 0;
+    uint32_t jitter_ms = 0;
+    uint32_t jitter_buffer_ms = 0;
+    uint32_t jitter_buffer_preferred_ms = 0;
+    uint32_t delay_estimate_ms = 0;
+    int32_t audio_level = -1;
+    float expand_rate = 0.0f;
+    float speech_expand_rate = 0.0f;
+    float secondary_decoded_rate = 0.0f;
+    float accelerate_rate = 0.0f;
+    float preemptive_expand_rate = 0.0f;
+    int32_t decoding_calls_to_silence_generator = 0;
+    int32_t decoding_calls_to_neteq = 0;
+    int32_t decoding_normal = 0;
+    int32_t decoding_plc = 0;
+    int32_t decoding_cng = 0;
+    int32_t decoding_plc_cng = 0;
+    int64_t capture_start_ntp_time_ms = 0;
+  };
+
+  struct Config {
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      std::string ToString() const;
+
+      // Synchronization source (stream identifier) to be received.
+      uint32_t remote_ssrc = 0;
+
+      // Sender SSRC used for sending RTCP (such as receiver reports).
+      uint32_t local_ssrc = 0;
+
+      // Enable feedback for send side bandwidth estimation.
+      // See
+      // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
+      // for details.
+      bool transport_cc = false;
+
+      // RTP header extensions used for the received stream.
+      std::vector<RtpExtension> extensions;
+    } rtp;
+
+    Transport* receive_transport = nullptr;
+    Transport* rtcp_send_transport = nullptr;
+
+    // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
+    // level components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
+    int voe_channel_id = -1;
+
+    // Identifier for an A/V synchronization group. Empty string to disable.
+    // TODO(pbos): Synchronize streams in a sync group, not just one video
+    // stream to one audio stream. Tracked by issue webrtc:4762.
+    std::string sync_group;
+
+    // Decoders for every payload that we can receive. Call owns the
+    // AudioDecoder instances once the Config is submitted to
+    // Call::CreateReceiveStream().
+    // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
+    std::map<uint8_t, AudioDecoder*> decoder_map;
+
+    // TODO(pbos): Remove config option once combined A/V BWE is always on.
+    bool combined_audio_video_bwe = false;
+  };
+
+  virtual Stats GetStats() const = 0;
+
+  // Sets an audio sink that receives unmixed audio from the receive stream.
+  // Ownership of the sink is passed to the stream and can be used by the
+  // caller to do lifetime management (i.e. when the sink's dtor is called).
+  // Only one sink can be set and passing a null sink, clears an existing one.
+  // NOTE: Audio must still somehow be pulled through AudioTransport for audio
+  // to stream through this sink. In practice, this happens if mixed audio
+  // is being pulled+rendered and/or if audio is being pulled for the purposes
+  // of feeding to the AEC.
+  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/audio_send_stream.h b/media/webrtc/trunk/webrtc/audio_send_stream.h
new file mode 100644
index 0000000000..d1af9e0103
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio_send_stream.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_SEND_STREAM_H_
+#define WEBRTC_AUDIO_SEND_STREAM_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/stream.h"
+#include "webrtc/transport.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+class AudioSendStream : public SendStream {
+ public:
+  struct Stats {
+    // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
+    uint32_t local_ssrc = 0;
+    int64_t bytes_sent = 0;
+    int32_t packets_sent = 0;
+    int32_t packets_lost = -1;
+    float fraction_lost = -1.0f;
+    std::string codec_name;
+    int32_t ext_seqnum = -1;
+    int32_t jitter_ms = -1;
+    int64_t rtt_ms = -1;
+    int32_t audio_level = -1;
+    float aec_quality_min = -1.0f;
+    int32_t echo_delay_median_ms = -1;
+    int32_t echo_delay_std_ms = -1;
+    int32_t echo_return_loss = -100;
+    int32_t echo_return_loss_enhancement = -100;
+    bool typing_noise_detected = false;
+  };
+
+  struct Config {
+    Config() = delete;
+    explicit Config(Transport* send_transport)
+        : send_transport(send_transport) {}
+
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      std::string ToString() const;
+
+      // Sender SSRC.
+      uint32_t ssrc = 0;
+
+      // RTP header extensions used for the sent stream.
+      std::vector<RtpExtension> extensions;
+
+      // RTCP CNAME, see RFC 3550.
+      std::string c_name;
+    } rtp;
+
+    // Transport for outgoing packets. The transport is expected to exist for
+    // the entire life of the AudioSendStream and is owned by the API client.
+    Transport* send_transport = nullptr;
+
+    // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
+    // components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
+    int voe_channel_id = -1;
+
+    // Ownership of the encoder object is transferred to Call when the config is
+    // passed to Call::CreateAudioSendStream().
+    // TODO(solenberg): Implement, once we configure codecs through the new API.
+    // rtc::scoped_ptr<AudioEncoder> encoder;
+    int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
+    int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
+  };
+
+  // TODO(solenberg): Make payload_type a config property instead.
+  virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
+                                  uint32_t duration_ms) = 0;
+  virtual Stats GetStats() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_SEND_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/audio_state.h b/media/webrtc/trunk/webrtc/audio_state.h
new file mode 100644
index 0000000000..fa5784c844
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/audio_state.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_AUDIO_STATE_H_
+#define WEBRTC_AUDIO_STATE_H_
+
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class VoiceEngine;
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+// AudioState holds the state which must be shared between multiple instances of
+// webrtc::Call for audio processing purposes.
+class AudioState : public rtc::RefCountInterface {
+ public:
+  struct Config {
+    // VoiceEngine used for audio streams and audio/video synchronization.
+    // AudioState will tickle the VoE refcount to keep it alive for as long as
+    // the AudioState itself.
+    VoiceEngine* voice_engine = nullptr;
+
+    // The AudioDeviceModule associated with the Calls.
+    AudioDeviceModule* audio_device_module = nullptr;
+  };
+
+  // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
+  static rtc::scoped_refptr<AudioState> Create(
+      const AudioState::Config& config);
+
+  virtual ~AudioState() {}
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_STATE_H_
diff --git a/media/webrtc/trunk/webrtc/base/BUILD.gn b/media/webrtc/trunk/webrtc/base/BUILD.gn
index 5201a69bef..0f7a3f2fbe 100644
--- a/media/webrtc/trunk/webrtc/base/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/base/BUILD.gn
@@ -10,6 +10,13 @@ import("//build/config/crypto.gni")
 import("//build/config/ui.gni")
 import("../build/webrtc.gni")
 
+# Enable OpenSSL (BoringSSL) for iOS. This is covered in webrtc/supplement.gypi
+# for the GYP build.
+import("//build_overrides/webrtc.gni")
+if (is_ios && !build_with_chromium) {
+  use_openssl = true
+}
+
 config("rtc_base_config") {
   include_dirs = [
     "//third_party/jsoncpp/overrides/include",
@@ -21,15 +28,15 @@ config("rtc_base_config") {
     "LOGGING=1",
   ]
 
-  # TODO(henrike): issue 3307, make rtc_base build without disabling
-  # these flags.
-  cflags_cc = [ "-Wno-non-virtual-dtor" ]
+  if (is_posix) {
+    # TODO(henrike): issue 3307, make rtc_base build without disabling
+    # these flags.
+    cflags_cc = [ "-Wno-non-virtual-dtor" ]
+  }
 }
 
 config("rtc_base_chromium_config") {
-  defines = [
-    "NO_MAIN_THREAD_WRAPPING",
-  ]
+  defines = [ "NO_MAIN_THREAD_WRAPPING" ]
 }
 
 config("openssl_config") {
@@ -39,19 +46,15 @@ config("openssl_config") {
   ]
 }
 
-config("nss_config") {
-  defines = [
-    "SSL_USE_NSS",
-    "HAVE_NSS_SSL_H",
-    "SSL_USE_NSS_RNG",
-  ]
-}
-
 config("ios_config") {
   libs = [
+    "AVFoundation.framework",
+    "CFNetwork.framework",
+
     #"Foundation.framework",  # Already included in //build/config:default_libs.
     "Security.framework",
     "SystemConfiguration.framework",
+
     #"UIKit.framework",  # Already included in //build/config:default_libs.
   ]
 }
@@ -59,6 +62,7 @@ config("ios_config") {
 config("mac_config") {
   libs = [
     "Cocoa.framework",
+
     #"Foundation.framework",  # Already included in //build/config:default_libs.
     #"IOKit.framework",  # Already included in //build/config:default_libs.
     #"Security.framework",  # Already included in //build/config:default_libs.
@@ -77,13 +81,8 @@ if (is_linux && !build_with_chromium) {
   # WebRTC cannot use as we don't sync src/crypto from Chromium.
   group("linux_system_ssl") {
     if (use_openssl) {
-      deps = [ "//third_party/boringssl" ]
-    } else {
-      deps = [ "//net/third_party/nss/ssl:libssl" ]
-
-      public_configs = [
-        "//net/third_party/nss/ssl:ssl_config",
-        "//third_party/nss:system_nss_no_ssl_config",
+      deps = [
+        "//third_party/boringssl",
       ]
     }
   }
@@ -99,12 +98,28 @@ if (rtc_build_ssl == 0) {
 
 # The subset of rtc_base approved for use outside of libjingle.
 static_library("rtc_base_approved") {
+  deps = []
   configs += [ "..:common_config" ]
   public_configs = [ "..:common_inherited_config" ]
 
   sources = [
+    "array_view.h",
+    "atomicops.h",
+    "bitbuffer.cc",
+    "bitbuffer.h",
+    "buffer.cc",
+    "buffer.h",
+    "bufferqueue.cc",
+    "bufferqueue.h",
+    "bytebuffer.cc",
+    "bytebuffer.h",
+    "byteorder.h",
     "checks.cc",
     "checks.h",
+    "constructormagic.h",
+    "criticalsection.cc",
+    "criticalsection.h",
+    "deprecation.h",
     "event.cc",
     "event.h",
     "event_tracer.cc",
@@ -115,14 +130,23 @@ static_library("rtc_base_approved") {
     "md5.h",
     "md5digest.cc",
     "md5digest.h",
+    "optional.h",
     "platform_file.cc",
     "platform_file.h",
+    "platform_thread.cc",
+    "platform_thread.h",
+    "platform_thread_types.h",
+    "random.cc",
+    "random.h",
     "safe_conversions.h",
     "safe_conversions_impl.h",
+    "scoped_ptr.h",
     "stringencode.cc",
     "stringencode.h",
     "stringutils.cc",
     "stringutils.h",
+    "systeminfo.cc",
+    "systeminfo.h",
     "template_util.h",
     "thread_annotations.h",
     "thread_checker.h",
@@ -132,13 +156,29 @@ static_library("rtc_base_approved") {
     "timeutils.h",
     "trace_event.h",
   ]
+
+  if (build_with_chromium) {
+    # Dependency on chromium's logging (in //base).
+    deps += [ "//base:base" ]
+    sources += [
+      "../../webrtc_overrides/webrtc/base/logging.cc",
+      "../../webrtc_overrides/webrtc/base/logging.h",
+    ]
+    include_dirs = [ "../../webrtc_overrides" ]
+  } else {
+    sources += [
+      "logging.cc",
+      "logging.h",
+    ]
+  }
 }
 
 static_library("rtc_base") {
   cflags = []
   cflags_cc = []
   libs = []
-  deps = [
+  deps = []
+  public_deps = [
     ":rtc_base_approved",
   ]
 
@@ -152,16 +192,15 @@ static_library("rtc_base") {
     ":rtc_base_config",
   ]
 
-  defines = [
-    "LOGGING=1",
-  ]
+  defines = [ "LOGGING=1" ]
 
   sources = [
     "arraysize.h",
     "asyncfile.cc",
     "asyncfile.h",
-    "asynchttprequest.cc",
-    "asynchttprequest.h",
+    "asyncinvoker-inl.h",
+    "asyncinvoker.cc",
+    "asyncinvoker.h",
     "asyncpacketsocket.cc",
     "asyncpacketsocket.h",
     "asyncresolverinterface.cc",
@@ -176,24 +215,16 @@ static_library("rtc_base") {
     "autodetectproxy.h",
     "base64.cc",
     "base64.h",
-    "basicdefs.h",
-    "buffer.cc",
-    "buffer.h",
-    "bytebuffer.cc",
-    "bytebuffer.h",
-    "byteorder.h",
     "common.cc",
     "common.h",
-    "cpumonitor.cc",
-    "cpumonitor.h",
     "crc32.cc",
     "crc32.h",
-    "criticalsection.cc",
-    "criticalsection.h",
     "cryptstring.cc",
     "cryptstring.h",
     "diskcache.cc",
     "diskcache.h",
+    "filerotatingstream.cc",
+    "filerotatingstream.h",
     "fileutils.cc",
     "fileutils.h",
     "firewallsocketserver.cc",
@@ -228,6 +259,8 @@ static_library("rtc_base") {
     "nethelpers.h",
     "network.cc",
     "network.h",
+    "networkmonitor.cc",
+    "networkmonitor.h",
     "nullsocketserver.h",
     "pathutils.cc",
     "pathutils.h",
@@ -241,9 +274,10 @@ static_library("rtc_base") {
     "ratelimiter.h",
     "ratetracker.cc",
     "ratetracker.h",
+    "rtccertificate.cc",
+    "rtccertificate.h",
     "scoped_autorelease_pool.h",
     "scoped_autorelease_pool.mm",
-    "scoped_ptr.h",
     "sha1.cc",
     "sha1.h",
     "sha1digest.cc",
@@ -280,8 +314,6 @@ static_library("rtc_base") {
     "sslstreamadapterhelper.h",
     "stream.cc",
     "stream.h",
-    "systeminfo.cc",
-    "systeminfo.h",
     "task.cc",
     "task.h",
     "taskparent.cc",
@@ -300,47 +332,38 @@ static_library("rtc_base") {
 
   if (is_posix) {
     sources += [
+      "ifaddrs-android.h",
+      "ifaddrs_converter.cc",
       "unixfilesystem.cc",
       "unixfilesystem.h",
     ]
   }
 
   if (build_with_chromium) {
-    sources += [
-      "../overrides/webrtc/base/basictypes.h",
-      "../overrides/webrtc/base/constructormagic.h",
-      "../overrides/webrtc/base/logging.cc",
-      "../overrides/webrtc/base/logging.h",
-    ]
-
     deps += [ "..:webrtc_common" ]
 
+    if (is_mac) {
+      sources += [ "macifaddrs_converter.cc" ]
+    }
+
     if (is_win) {
-      sources += [ "../overrides/webrtc/base/win32socketinit.cc" ]
+      sources += [ "../../webrtc_overrides/webrtc/base/win32socketinit.cc" ]
     }
 
     include_dirs = [
-      "../overrides",
+      "../../webrtc_overrides",
       "../../boringssl/src/include",
     ]
 
     public_configs += [ ":rtc_base_chromium_config" ]
   } else {
     sources += [
-      "asyncinvoker.cc",
-      "asyncinvoker.h",
-      "asyncinvoker-inl.h",
-      "atomicops.h",
       "bandwidthsmoother.cc",
       "bandwidthsmoother.h",
-      "basictypes.h",
       "bind.h",
       "bind.h.pump",
       "callback.h",
       "callback.h.pump",
-      "constructormagic.h",
-      "filelock.cc",
-      "filelock.h",
       "fileutils_mock.h",
       "genericslot.h",
       "genericslot.h.pump",
@@ -348,8 +371,8 @@ static_library("rtc_base") {
       "httpserver.h",
       "json.cc",
       "json.h",
-      "logging.cc",
-      "logging.h",
+      "logsinks.cc",
+      "logsinks.h",
       "mathutils.h",
       "multipart.cc",
       "multipart.h",
@@ -368,8 +391,8 @@ static_library("rtc_base") {
       "refcount.h",
       "referencecountedsingletonfactory.h",
       "rollingaccumulator.h",
-      "scopedptrcollection.h",
       "scoped_ref_ptr.h",
+      "scopedptrcollection.h",
       "sec_buffer.h",
       "sharedexclusivelock.cc",
       "sharedexclusivelock.h",
@@ -384,8 +407,8 @@ static_library("rtc_base") {
       "virtualsocketserver.cc",
       "virtualsocketserver.h",
       "window.h",
-      "windowpickerfactory.h",
       "windowpicker.h",
+      "windowpickerfactory.h",
     ]
 
     deps += [ "..:webrtc_common" ]
@@ -475,31 +498,6 @@ static_library("rtc_base") {
       "opensslstreamadapter.cc",
       "opensslstreamadapter.h",
     ]
-  } else {
-    public_configs += [ ":nss_config" ]
-    if (rtc_build_ssl) {
-      if (build_with_chromium) {
-        deps += [ "//crypto:platform" ]
-      } else {
-        deps += [ "//net/third_party/nss/ssl:libssl" ]
-        if (is_linux) {
-          deps += [ ":linux_system_ssl" ]
-        } else {
-          deps += [
-            "//third_party/nss:nspr",
-            "//third_party/nss:nss",
-          ]
-        }
-      }
-    } else {
-      configs += [ "external_ssl_library" ]
-    }
-    sources += [
-      "nssidentity.cc",
-      "nssidentity.h",
-      "nssstreamadapter.cc",
-      "nssstreamadapter.h",
-    ]
   }
 
   if (is_android) {
@@ -510,12 +508,17 @@ static_library("rtc_base") {
 
     libs += [
       "log",
-      "GLESv2"
+      "GLESv2",
     ]
   }
 
   if (is_ios) {
-    all_dependent_configs += [ ":ios_config" ]
+    all_dependent_configs = [ ":ios_config" ]
+
+    sources += [
+      "macconversion.cc",
+      "macconversion.h",
+    ]
   }
 
   if (use_x11) {
@@ -559,8 +562,6 @@ static_library("rtc_base") {
 
   if (is_win) {
     sources += [
-      "schanneladapter.cc",
-      "schanneladapter.h",
       "win32.cc",
       "win32.h",
       "win32filesystem.cc",
@@ -608,4 +609,32 @@ static_library("rtc_base") {
       "linux.h",
     ]
   }
+
+  if (is_nacl) {
+    deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
+    defines += [ "timezone=_timezone" ]
+    sources -= [ "ifaddrs_converter.cc" ]
+  }
+}
+
+if (is_ios) {
+  source_set("rtc_base_objc") {
+    deps = [
+      ":rtc_base",
+    ]
+    cflags = [ "-fobjc-arc" ]
+    configs += [ "..:common_config" ]
+    public_configs = [ "..:common_inherited_config" ]
+
+    sources = [
+      "objc/NSString+StdString.h",
+      "objc/NSString+StdString.mm",
+      "objc/RTCCameraPreviewView.h",
+      "objc/RTCCameraPreviewView.m",
+      "objc/RTCDispatcher.h",
+      "objc/RTCDispatcher.m",
+      "objc/RTCLogging.h",
+      "objc/RTCLogging.mm",
+    ]
+  }
 }
diff --git a/media/webrtc/trunk/webrtc/base/OWNERS b/media/webrtc/trunk/webrtc/base/OWNERS
index 9a527df143..2f400904c6 100644
--- a/media/webrtc/trunk/webrtc/base/OWNERS
+++ b/media/webrtc/trunk/webrtc/base/OWNERS
@@ -9,4 +9,10 @@ pthatcher@webrtc.org
 sergeyu@chromium.org
 tommi@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
+
diff --git a/media/webrtc/trunk/webrtc/base/array_view.h b/media/webrtc/trunk/webrtc/base/array_view.h
new file mode 100644
index 0000000000..a7ca66cc95
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/array_view.h
@@ -0,0 +1,133 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ARRAY_VIEW_H_
+#define WEBRTC_BASE_ARRAY_VIEW_H_
+
+#include "webrtc/base/checks.h"
+
+namespace rtc {
+
+// Many functions read from or write to arrays. The obvious way to do this is
+// to use two arguments, a pointer to the first element and an element count:
+//
+//   bool Contains17(const int* arr, size_t size) {
+//     for (size_t i = 0; i < size; ++i) {
+//       if (arr[i] == 17)
+//         return true;
+//     }
+//     return false;
+//   }
+//
+// This is flexible, since it doesn't matter how the array is stored (C array,
+// std::vector, rtc::Buffer, ...), but it's error-prone because the caller has
+// to correctly specify the array length:
+//
+//   Contains17(arr, arraysize(arr));  // C array
+//   Contains17(&arr[0], arr.size());  // std::vector
+//   Contains17(arr, size);            // pointer + size
+//   ...
+//
+// It's also kind of messy to have two separate arguments for what is
+// conceptually a single thing.
+//
+// Enter rtc::ArrayView<T>. It contains a T pointer (to an array it doesn't
+// own) and a count, and supports the basic things you'd expect, such as
+// indexing and iteration. It allows us to write our function like this:
+//
+//   bool Contains17(rtc::ArrayView<const int> arr) {
+//     for (auto e : arr) {
+//       if (e == 17)
+//         return true;
+//     }
+//     return false;
+//   }
+//
+// And even better, because a bunch of things will implicitly convert to
+// ArrayView, we can call it like this:
+//
+//   Contains17(arr);                             // C array
+//   Contains17(arr);                             // std::vector
+//   Contains17(rtc::ArrayView<int>(arr, size));  // pointer + size
+//   ...
+//
+// One important point is that ArrayView<T> and ArrayView<const T> are
+// different types, which allow and don't allow mutation of the array elements,
+// respectively. The implicit conversions work just like you'd hope, so that
+// e.g. vector<int> will convert to either ArrayView<int> or ArrayView<const
+// int>, but const vector<int> will convert only to ArrayView<const int>.
+// (ArrayView itself can be the source type in such conversions, so
+// ArrayView<int> will convert to ArrayView<const int>.)
+//
+// Note: ArrayView is tiny (just a pointer and a count) and trivially copyable,
+// so it's probably cheaper to pass it by value than by const reference.
+template <typename T>
+class ArrayView final {
+ public:
+  // Construct an empty ArrayView.
+  ArrayView() : ArrayView(static_cast<T*>(nullptr), 0) {}
+
+  // Construct an ArrayView for a (pointer,size) pair.
+  template <typename U>
+  ArrayView(U* data, size_t size)
+      : data_(size == 0 ? nullptr : data), size_(size) {
+    CheckInvariant();
+  }
+
+  // Construct an ArrayView for an array.
+  template <typename U, size_t N>
+  ArrayView(U (&array)[N]) : ArrayView(&array[0], N) {}
+
+  // Construct an ArrayView for any type U that has a size() method whose
+  // return value converts implicitly to size_t, and a data() method whose
+  // return value converts implicitly to T*. In particular, this means we allow
+  // conversion from ArrayView<T> to ArrayView<const T>, but not the other way
+  // around. Other allowed conversions include std::vector<T> to ArrayView<T>
+  // or ArrayView<const T>, const std::vector<T> to ArrayView<const T>, and
+  // rtc::Buffer to ArrayView<uint8_t> (with the same const behavior as
+  // std::vector).
+  template <typename U>
+  ArrayView(U& u) : ArrayView(u.data(), u.size()) {}
+
+  // Indexing, size, and iteration. These allow mutation even if the ArrayView
+  // is const, because the ArrayView doesn't own the array. (To prevent
+  // mutation, use ArrayView<const T>.)
+  size_t size() const { return size_; }
+  bool empty() const { return size_ == 0; }
+  T* data() const { return data_; }
+  T& operator[](size_t idx) const {
+    RTC_DCHECK_LT(idx, size_);
+    RTC_DCHECK(data_);  // Follows from size_ > idx and the class invariant.
+    return data_[idx];
+  }
+  T* begin() const { return data_; }
+  T* end() const { return data_ + size_; }
+  const T* cbegin() const { return data_; }
+  const T* cend() const { return data_ + size_; }
+
+  // Comparing two ArrayViews compares their (pointer,size) pairs; it does
+  // *not* dereference the pointers.
+  friend bool operator==(const ArrayView& a, const ArrayView& b) {
+    return a.data_ == b.data_ && a.size_ == b.size_;
+  }
+  friend bool operator!=(const ArrayView& a, const ArrayView& b) {
+    return !(a == b);
+  }
+
+ private:
+  // Invariant: !data_ iff size_ == 0.
+  void CheckInvariant() const { RTC_DCHECK_EQ(!data_, size_ == 0); }
+  T* data_;
+  size_t size_;
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_ARRAY_VIEW_H_
diff --git a/media/webrtc/trunk/webrtc/base/array_view_unittest.cc b/media/webrtc/trunk/webrtc/base/array_view_unittest.cc
new file mode 100644
index 0000000000..8bb1bcc4c6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/array_view_unittest.cc
@@ -0,0 +1,233 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/gunit.h"
+
+namespace rtc {
+
+namespace {
+template <typename T>
+void Call(ArrayView<T>) {}
+}  // namespace
+
+TEST(ArrayViewTest, TestConstructFromPtrAndArray) {
+  char arr[] = "Arrr!";
+  const char carr[] = "Carrr!";
+  Call<const char>(arr);
+  Call<const char>(carr);
+  Call<char>(arr);
+  // Call<char>(carr);  // Compile error, because can't drop const.
+  // Call<int>(arr);  // Compile error, because incompatible types.
+  ArrayView<int*> x;
+  EXPECT_EQ(0u, x.size());
+  EXPECT_EQ(nullptr, x.data());
+  ArrayView<char> y = arr;
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z(arr + 1, 3);
+  EXPECT_EQ(3u, z.size());
+  EXPECT_EQ(arr + 1, z.data());
+  ArrayView<const char> w(arr, 2);
+  EXPECT_EQ(2u, w.size());
+  EXPECT_EQ(arr, w.data());
+  ArrayView<char> q(arr, 0);
+  EXPECT_EQ(0u, q.size());
+  EXPECT_EQ(nullptr, q.data());
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  // DCHECK error (nullptr with nonzero size).
+  EXPECT_DEATH(ArrayView<int>(static_cast<int*>(nullptr), 5), "");
+#endif
+  // These are compile errors, because incompatible types.
+  // ArrayView<int> m = arr;
+  // ArrayView<float> n(arr + 2, 2);
+}
+
+TEST(ArrayViewTest, TestCopyConstructor) {
+  char arr[] = "Arrr!";
+  ArrayView<char> x = arr;
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<char> y = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z = x;  // Copy non-const -> const.
+  EXPECT_EQ(6u, z.size());
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char> w = z;  // Copy const -> const.
+  EXPECT_EQ(6u, w.size());
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char> v = z;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyAssignment) {
+  char arr[] = "Arrr!";
+  ArrayView<char> x(arr);
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<char> y;
+  y = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z;
+  z = x;  // Copy non-const -> const.
+  EXPECT_EQ(6u, z.size());
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char> w;
+  w = z;  // Copy const -> const.
+  EXPECT_EQ(6u, w.size());
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char> v;
+  // v = z;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestStdVector) {
+  std::vector<int> v;
+  v.push_back(3);
+  v.push_back(11);
+  Call<const int>(v);
+  Call<int>(v);
+  // Call<unsigned int>(v);  // Compile error, because incompatible types.
+  ArrayView<int> x = v;
+  EXPECT_EQ(2u, x.size());
+  EXPECT_EQ(v.data(), x.data());
+  ArrayView<const int> y;
+  y = v;
+  EXPECT_EQ(2u, y.size());
+  EXPECT_EQ(v.data(), y.data());
+  // ArrayView<double> d = v;  // Compile error, because incompatible types.
+  const std::vector<int> cv;
+  Call<const int>(cv);
+  // Call<int>(cv);  // Compile error, because can't drop const.
+  ArrayView<const int> z = cv;
+  EXPECT_EQ(0u, z.size());
+  EXPECT_EQ(nullptr, z.data());
+  // ArrayView<int> w = cv;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestRtcBuffer) {
+  rtc::Buffer b = "so buffer";
+  Call<const uint8_t>(b);
+  Call<uint8_t>(b);
+  // Call<int8_t>(b);  // Compile error, because incompatible types.
+  ArrayView<uint8_t> x = b;
+  EXPECT_EQ(10u, x.size());
+  EXPECT_EQ(b.data(), x.data());
+  ArrayView<const uint8_t> y;
+  y = b;
+  EXPECT_EQ(10u, y.size());
+  EXPECT_EQ(b.data(), y.data());
+  // ArrayView<char> d = b;  // Compile error, because incompatible types.
+  const rtc::Buffer cb = "very const";
+  Call<const uint8_t>(cb);
+  // Call<uint8_t>(cb);  // Compile error, because can't drop const.
+  ArrayView<const uint8_t> z = cb;
+  EXPECT_EQ(11u, z.size());
+  EXPECT_EQ(cb.data(), z.data());
+  // ArrayView<uint8_t> w = cb;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestSwap) {
+  const char arr[] = "Arrr!";
+  const char aye[] = "Aye, Cap'n!";
+  ArrayView<const char> x(arr);
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<const char> y(aye);
+  EXPECT_EQ(12u, y.size());
+  EXPECT_EQ(aye, y.data());
+  using std::swap;
+  swap(x, y);
+  EXPECT_EQ(12u, x.size());
+  EXPECT_EQ(aye, x.data());
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  // ArrayView<char> z;
+  // swap(x, z);  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestIndexing) {
+  char arr[] = "abcdefg";
+  ArrayView<char> x(arr);
+  const ArrayView<char> y(arr);
+  ArrayView<const char> z(arr);
+  EXPECT_EQ(8u, x.size());
+  EXPECT_EQ(8u, y.size());
+  EXPECT_EQ(8u, z.size());
+  EXPECT_EQ('b', x[1]);
+  EXPECT_EQ('c', y[2]);
+  EXPECT_EQ('d', z[3]);
+  x[3] = 'X';
+  y[2] = 'Y';
+  // z[1] = 'Z';  // Compile error, because z's element type is const char.
+  EXPECT_EQ('b', x[1]);
+  EXPECT_EQ('Y', y[2]);
+  EXPECT_EQ('X', z[3]);
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  EXPECT_DEATH(z[8], "");  // DCHECK error (index out of bounds).
+#endif
+}
+
+TEST(ArrayViewTest, TestIterationEmpty) {
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>> av;
+  EXPECT_FALSE(av.begin());
+  EXPECT_FALSE(av.cbegin());
+  EXPECT_FALSE(av.end());
+  EXPECT_FALSE(av.cend());
+  for (auto& e : av) {
+    EXPECT_TRUE(false);
+    EXPECT_EQ(42u, e.size());  // Dummy use of e to prevent unused var warning.
+  }
+}
+
+TEST(ArrayViewTest, TestIteration) {
+  char arr[] = "Arrr!";
+  ArrayView<char> av(arr);
+  EXPECT_EQ('A', *av.begin());
+  EXPECT_EQ('A', *av.cbegin());
+  EXPECT_EQ('\0', *(av.end() - 1));
+  EXPECT_EQ('\0', *(av.cend() - 1));
+  char i = 0;
+  for (auto& e : av) {
+    EXPECT_EQ(arr + i, &e);
+    e = 's' + i;
+    ++i;
+  }
+  i = 0;
+  for (auto& e : ArrayView<const char>(av)) {
+    EXPECT_EQ(arr + i, &e);
+    // e = 'q' + i;  // Compile error, because e is a const char&.
+    ++i;
+  }
+}
+
+TEST(ArrayViewTest, TestEmpty) {
+  EXPECT_TRUE(ArrayView<int>().empty());
+  const int a[] = {1, 2, 3};
+  EXPECT_FALSE(ArrayView<const int>(a).empty());
+}
+
+TEST(ArrayViewTest, TestCompare) {
+  int a[] = {1, 2, 3};
+  int b[] = {1, 2, 3};
+  EXPECT_EQ(ArrayView<int>(a), ArrayView<int>(a));
+  EXPECT_EQ(ArrayView<int>(), ArrayView<int>());
+  EXPECT_NE(ArrayView<int>(a), ArrayView<int>(b));
+  EXPECT_NE(ArrayView<int>(a), ArrayView<int>());
+  EXPECT_NE(ArrayView<int>(a), ArrayView<int>(a, 2));
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/arraysize.h b/media/webrtc/trunk/webrtc/base/arraysize.h
index 0bb0a62ed8..56a10392af 100644
--- a/media/webrtc/trunk/webrtc/base/arraysize.h
+++ b/media/webrtc/trunk/webrtc/base/arraysize.h
@@ -24,16 +24,7 @@
 // This template function declaration is used in defining arraysize.
 // Note that the function doesn't need an implementation, as we only
 // use its type.
-template <typename T, size_t N>
-char (&ArraySizeHelper(T (&array)[N]))[N];
-
-// That gcc wants both of these prototypes seems mysterious. VC, for
-// its part, can't decide which to use (another mystery). Matching of
-// template overloads: the final frontier.
-#ifndef _MSC_VER
-template <typename T, size_t N>
-char (&ArraySizeHelper(const T (&array)[N]))[N];
-#endif
+template <typename T, size_t N> char (&ArraySizeHelper(T (&array)[N]))[N];
 
 #define arraysize(array) (sizeof(ArraySizeHelper(array)))
 
diff --git a/media/webrtc/trunk/webrtc/base/asynchttprequest.cc b/media/webrtc/trunk/webrtc/base/asynchttprequest.cc
index bdca585c31..310d82e040 100644
--- a/media/webrtc/trunk/webrtc/base/asynchttprequest.cc
+++ b/media/webrtc/trunk/webrtc/base/asynchttprequest.cc
@@ -1,116 +1,2 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/asynchttprequest.h"
-
-namespace rtc {
-
-enum {
-  MSG_TIMEOUT = SignalThread::ST_MSG_FIRST_AVAILABLE,
-  MSG_LAUNCH_REQUEST
-};
-static const int kDefaultHTTPTimeout = 30 * 1000;  // 30 sec
-
-///////////////////////////////////////////////////////////////////////////////
-// AsyncHttpRequest
-///////////////////////////////////////////////////////////////////////////////
-
-AsyncHttpRequest::AsyncHttpRequest(const std::string &user_agent)
-    : start_delay_(0),
-      firewall_(NULL),
-      port_(80),
-      secure_(false),
-      timeout_(kDefaultHTTPTimeout),
-      fail_redirect_(false),
-      factory_(Thread::Current()->socketserver(), user_agent),
-      pool_(&factory_),
-      client_(user_agent.c_str(), &pool_),
-      error_(HE_NONE) {
-  client_.SignalHttpClientComplete.connect(this,
-      &AsyncHttpRequest::OnComplete);
-}
-
-AsyncHttpRequest::~AsyncHttpRequest() {
-}
-
-void AsyncHttpRequest::OnWorkStart() {
-  if (start_delay_ <= 0) {
-    LaunchRequest();
-  } else {
-    Thread::Current()->PostDelayed(start_delay_, this, MSG_LAUNCH_REQUEST);
-  }
-}
-
-void AsyncHttpRequest::OnWorkStop() {
-  // worker is already quitting, no need to explicitly quit
-  LOG(LS_INFO) << "HttpRequest cancelled";
-}
-
-void AsyncHttpRequest::OnComplete(HttpClient* client, HttpErrorType error) {
-  Thread::Current()->Clear(this, MSG_TIMEOUT);
-
-  set_error(error);
-  if (!error) {
-    LOG(LS_INFO) << "HttpRequest completed successfully";
-
-    std::string value;
-    if (client_.response().hasHeader(HH_LOCATION, &value)) {
-      response_redirect_ = value.c_str();
-    }
-  } else {
-    LOG(LS_INFO) << "HttpRequest completed with error: " << error;
-  }
-
-  worker()->Quit();
-}
-
-void AsyncHttpRequest::OnMessage(Message* message) {
-  switch (message->message_id) {
-   case MSG_TIMEOUT:
-    LOG(LS_INFO) << "HttpRequest timed out";
-    client_.reset();
-    worker()->Quit();
-    break;
-   case MSG_LAUNCH_REQUEST:
-    LaunchRequest();
-    break;
-   default:
-    SignalThread::OnMessage(message);
-    break;
-  }
-}
-
-void AsyncHttpRequest::DoWork() {
-  // Do nothing while we wait for the request to finish. We only do this so
-  // that we can be a SignalThread; in the future this class should not be
-  // a SignalThread, since it does not need to spawn a new thread.
-  Thread::Current()->ProcessMessages(Thread::kForever);
-}
-
-void AsyncHttpRequest::LaunchRequest() {
-  factory_.SetProxy(proxy_);
-  if (secure_)
-    factory_.UseSSL(host_.c_str());
-
-  bool transparent_proxy = (port_ == 80) &&
-           ((proxy_.type == PROXY_HTTPS) || (proxy_.type == PROXY_UNKNOWN));
-  if (transparent_proxy) {
-    client_.set_proxy(proxy_);
-  }
-  client_.set_fail_redirect(fail_redirect_);
-  client_.set_server(SocketAddress(host_, port_));
-
-  LOG(LS_INFO) << "HttpRequest start: " << host_ + client_.request().path;
-
-  Thread::Current()->PostDelayed(timeout_, this, MSG_TIMEOUT);
-  client_.start();
-}
-
-}  // namespace rtc
+// TODO(pthatcher): Remove this file once chromium's GYP file doesn't
+// refer to it.
diff --git a/media/webrtc/trunk/webrtc/base/asynchttprequest.h b/media/webrtc/trunk/webrtc/base/asynchttprequest.h
index b4bbf2539c..310d82e040 100644
--- a/media/webrtc/trunk/webrtc/base/asynchttprequest.h
+++ b/media/webrtc/trunk/webrtc/base/asynchttprequest.h
@@ -1,104 +1,2 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_ASYNCHTTPREQUEST_H_
-#define WEBRTC_BASE_ASYNCHTTPREQUEST_H_
-
-#include <string>
-#include "webrtc/base/event.h"
-#include "webrtc/base/httpclient.h"
-#include "webrtc/base/signalthread.h"
-#include "webrtc/base/socketpool.h"
-#include "webrtc/base/sslsocketfactory.h"
-
-namespace rtc {
-
-class FirewallManager;
-
-///////////////////////////////////////////////////////////////////////////////
-// AsyncHttpRequest
-// Performs an HTTP request on a background thread.  Notifies on the foreground
-// thread once the request is done (successfully or unsuccessfully).
-///////////////////////////////////////////////////////////////////////////////
-
-class AsyncHttpRequest : public SignalThread {
- public:
-  explicit AsyncHttpRequest(const std::string &user_agent);
-  ~AsyncHttpRequest() override;
-
-  // If start_delay is less than or equal to zero, this starts immediately.
-  // Start_delay defaults to zero.
-  int start_delay() const { return start_delay_; }
-  void set_start_delay(int delay) { start_delay_ = delay; }
-
-  const ProxyInfo& proxy() const { return proxy_; }
-  void set_proxy(const ProxyInfo& proxy) {
-    proxy_ = proxy;
-  }
-  void set_firewall(FirewallManager * firewall) {
-    firewall_ = firewall;
-  }
-
-  // The DNS name of the host to connect to.
-  const std::string& host() { return host_; }
-  void set_host(const std::string& host) { host_ = host; }
-
-  // The port to connect to on the target host.
-  int port() { return port_; }
-  void set_port(int port) { port_ = port; }
-
-  // Whether the request should use SSL.
-  bool secure() { return secure_; }
-  void set_secure(bool secure) { secure_ = secure; }
-
-  // Time to wait on the download, in ms.
-  int timeout() { return timeout_; }
-  void set_timeout(int timeout) { timeout_ = timeout; }
-
-  // Fail redirects to allow analysis of redirect urls, etc.
-  bool fail_redirect() const { return fail_redirect_; }
-  void set_fail_redirect(bool redirect) { fail_redirect_ = redirect; }
-
-  // Returns the redirect when redirection occurs
-  const std::string& response_redirect() { return response_redirect_; }
-
-  HttpRequestData& request() { return client_.request(); }
-  HttpResponseData& response() { return client_.response(); }
-  HttpErrorType error() { return error_; }
-
- protected:
-  void set_error(HttpErrorType error) { error_ = error; }
-  void OnWorkStart() override;
-  void OnWorkStop() override;
-  void OnComplete(HttpClient* client, HttpErrorType error);
-  void OnMessage(Message* message) override;
-  void DoWork() override;
-
- private:
-  void LaunchRequest();
-
-  int start_delay_;
-  ProxyInfo proxy_;
-  FirewallManager* firewall_;
-  std::string host_;
-  int port_;
-  bool secure_;
-  int timeout_;
-  bool fail_redirect_;
-  SslSocketFactory factory_;
-  ReuseSocketPool pool_;
-  HttpClient client_;
-  HttpErrorType error_;
-  std::string response_redirect_;
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_ASYNCHTTPREQUEST_H_
+// TODO(pthatcher): Remove this file once chromium's GYP file doesn't
+// refer to it.
diff --git a/media/webrtc/trunk/webrtc/base/asynchttprequest_unittest.cc b/media/webrtc/trunk/webrtc/base/asynchttprequest_unittest.cc
deleted file mode 100644
index ddfad90143..0000000000
--- a/media/webrtc/trunk/webrtc/base/asynchttprequest_unittest.cc
+++ /dev/null
@@ -1,234 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-#include "webrtc/base/asynchttprequest.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/httpserver.h"
-#include "webrtc/base/socketstream.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace rtc {
-
-static const SocketAddress kServerAddr("127.0.0.1", 0);
-static const SocketAddress kServerHostnameAddr("localhost", 0);
-static const char kServerGetPath[] = "/get";
-static const char kServerPostPath[] = "/post";
-static const char kServerResponse[] = "This is a test";
-
-class TestHttpServer : public HttpServer, public sigslot::has_slots<> {
- public:
-  TestHttpServer(Thread* thread, const SocketAddress& addr) :
-      socket_(thread->socketserver()->CreateAsyncSocket(addr.family(),
-                                                        SOCK_STREAM)) {
-    socket_->Bind(addr);
-    socket_->Listen(5);
-    socket_->SignalReadEvent.connect(this, &TestHttpServer::OnAccept);
-  }
-
-  SocketAddress address() const { return socket_->GetLocalAddress(); }
-  void Close() const { socket_->Close(); }
-
- private:
-  void OnAccept(AsyncSocket* socket) {
-    AsyncSocket* new_socket = socket_->Accept(NULL);
-    if (new_socket) {
-      HandleConnection(new SocketStream(new_socket));
-    }
-  }
-  rtc::scoped_ptr<AsyncSocket> socket_;
-};
-
-class AsyncHttpRequestTest : public testing::Test,
-                             public sigslot::has_slots<> {
- public:
-  AsyncHttpRequestTest()
-      : started_(false),
-        done_(false),
-        server_(Thread::Current(), kServerAddr) {
-    server_.SignalHttpRequest.connect(this, &AsyncHttpRequestTest::OnRequest);
-  }
-
-  bool started() const { return started_; }
-  bool done() const { return done_; }
-
-  AsyncHttpRequest* CreateGetRequest(const std::string& host, int port,
-                                     const std::string& path) {
-    rtc::AsyncHttpRequest* request =
-        new rtc::AsyncHttpRequest("unittest");
-    request->SignalWorkDone.connect(this,
-        &AsyncHttpRequestTest::OnRequestDone);
-    request->request().verb = rtc::HV_GET;
-    request->set_host(host);
-    request->set_port(port);
-    request->request().path = path;
-    request->response().document.reset(new MemoryStream());
-    return request;
-  }
-  AsyncHttpRequest* CreatePostRequest(const std::string& host, int port,
-                                      const std::string& path,
-                                      const std::string content_type,
-                                      StreamInterface* content) {
-    rtc::AsyncHttpRequest* request =
-        new rtc::AsyncHttpRequest("unittest");
-    request->SignalWorkDone.connect(this,
-        &AsyncHttpRequestTest::OnRequestDone);
-    request->request().verb = rtc::HV_POST;
-    request->set_host(host);
-    request->set_port(port);
-    request->request().path = path;
-    request->request().setContent(content_type, content);
-    request->response().document.reset(new MemoryStream());
-    return request;
-  }
-
-  const TestHttpServer& server() const { return server_; }
-
- protected:
-  void OnRequest(HttpServer* server, HttpServerTransaction* t) {
-    started_ = true;
-
-    if (t->request.path == kServerGetPath) {
-      t->response.set_success("text/plain", new MemoryStream(kServerResponse));
-    } else if (t->request.path == kServerPostPath) {
-      // reverse the data and reply
-      size_t size;
-      StreamInterface* in = t->request.document.get();
-      StreamInterface* out = new MemoryStream();
-      in->GetSize(&size);
-      for (size_t i = 0; i < size; ++i) {
-        char ch;
-        in->SetPosition(size - i - 1);
-        in->Read(&ch, 1, NULL, NULL);
-        out->Write(&ch, 1, NULL, NULL);
-      }
-      out->Rewind();
-      t->response.set_success("text/plain", out);
-    } else {
-      t->response.set_error(404);
-    }
-    server_.Respond(t);
-  }
-  void OnRequestDone(SignalThread* thread) {
-    done_ = true;
-  }
-
- private:
-  bool started_;
-  bool done_;
-  TestHttpServer server_;
-};
-
-TEST_F(AsyncHttpRequestTest, TestGetSuccess) {
-  AsyncHttpRequest* req = CreateGetRequest(
-      kServerHostnameAddr.hostname(), server().address().port(),
-      kServerGetPath);
-  EXPECT_FALSE(started());
-  req->Start();
-  EXPECT_TRUE_WAIT(started(), 5000);  // Should have started by now.
-  EXPECT_TRUE_WAIT(done(), 5000);
-  std::string response;
-  EXPECT_EQ(200U, req->response().scode);
-  ASSERT_TRUE(req->response().document);
-  req->response().document->Rewind();
-  req->response().document->ReadLine(&response);
-  EXPECT_EQ(kServerResponse, response);
-  req->Release();
-}
-
-TEST_F(AsyncHttpRequestTest, TestGetNotFound) {
-  AsyncHttpRequest* req = CreateGetRequest(
-      kServerHostnameAddr.hostname(), server().address().port(),
-      "/bad");
-  req->Start();
-  EXPECT_TRUE_WAIT(done(), 5000);
-  size_t size;
-  EXPECT_EQ(404U, req->response().scode);
-  ASSERT_TRUE(req->response().document);
-  req->response().document->GetSize(&size);
-  EXPECT_EQ(0U, size);
-  req->Release();
-}
-
-TEST_F(AsyncHttpRequestTest, TestGetToNonServer) {
-  AsyncHttpRequest* req = CreateGetRequest(
-      "127.0.0.1", server().address().port(),
-      kServerGetPath);
-  // Stop the server before we send the request.
-  server().Close();
-  req->Start();
-  EXPECT_TRUE_WAIT(done(), 10000);
-  size_t size;
-  EXPECT_EQ(500U, req->response().scode);
-  ASSERT_TRUE(req->response().document);
-  req->response().document->GetSize(&size);
-  EXPECT_EQ(0U, size);
-  req->Release();
-}
-
-TEST_F(AsyncHttpRequestTest, DISABLED_TestGetToInvalidHostname) {
-  AsyncHttpRequest* req = CreateGetRequest(
-      "invalid", server().address().port(),
-      kServerGetPath);
-  req->Start();
-  EXPECT_TRUE_WAIT(done(), 5000);
-  size_t size;
-  EXPECT_EQ(500U, req->response().scode);
-  ASSERT_TRUE(req->response().document);
-  req->response().document->GetSize(&size);
-  EXPECT_EQ(0U, size);
-  req->Release();
-}
-
-TEST_F(AsyncHttpRequestTest, TestPostSuccess) {
-  AsyncHttpRequest* req = CreatePostRequest(
-      kServerHostnameAddr.hostname(), server().address().port(),
-      kServerPostPath, "text/plain", new MemoryStream("abcd1234"));
-  req->Start();
-  EXPECT_TRUE_WAIT(done(), 5000);
-  std::string response;
-  EXPECT_EQ(200U, req->response().scode);
-  ASSERT_TRUE(req->response().document);
-  req->response().document->Rewind();
-  req->response().document->ReadLine(&response);
-  EXPECT_EQ("4321dcba", response);
-  req->Release();
-}
-
-// Ensure that we shut down properly even if work is outstanding.
-TEST_F(AsyncHttpRequestTest, TestCancel) {
-  AsyncHttpRequest* req = CreateGetRequest(
-      kServerHostnameAddr.hostname(), server().address().port(),
-      kServerGetPath);
-  req->Start();
-  req->Destroy(true);
-}
-
-TEST_F(AsyncHttpRequestTest, TestGetSuccessDelay) {
-  AsyncHttpRequest* req = CreateGetRequest(
-      kServerHostnameAddr.hostname(), server().address().port(),
-      kServerGetPath);
-  req->set_start_delay(10);  // Delay 10ms.
-  req->Start();
-  Thread::SleepMs(5);
-  EXPECT_FALSE(started());  // Should not have started immediately.
-  EXPECT_TRUE_WAIT(started(), 5000);  // Should have started by now.
-  EXPECT_TRUE_WAIT(done(), 5000);
-  std::string response;
-  EXPECT_EQ(200U, req->response().scode);
-  ASSERT_TRUE(req->response().document);
-  req->response().document->Rewind();
-  req->response().document->ReadLine(&response);
-  EXPECT_EQ(kServerResponse, response);
-  req->Release();
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/asyncinvoker.cc b/media/webrtc/trunk/webrtc/base/asyncinvoker.cc
index 35ce2fb128..8285d5545b 100644
--- a/media/webrtc/trunk/webrtc/base/asyncinvoker.cc
+++ b/media/webrtc/trunk/webrtc/base/asyncinvoker.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/base/asyncinvoker.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 
 namespace rtc {
@@ -35,7 +36,7 @@ void AsyncInvoker::OnMessage(Message* msg) {
   closure->Execute();
 }
 
-void AsyncInvoker::Flush(Thread* thread, uint32 id /*= MQID_ANY*/) {
+void AsyncInvoker::Flush(Thread* thread, uint32_t id /*= MQID_ANY*/) {
   if (destroying_) return;
 
   // Run this on |thread| to reduce the number of context switches.
@@ -56,7 +57,7 @@ void AsyncInvoker::Flush(Thread* thread, uint32 id /*= MQID_ANY*/) {
 
 void AsyncInvoker::DoInvoke(Thread* thread,
                             const scoped_refptr<AsyncClosure>& closure,
-                            uint32 id) {
+                            uint32_t id) {
   if (destroying_) {
     LOG(LS_WARNING) << "Tried to invoke while destroying the invoker.";
     return;
@@ -64,6 +65,41 @@ void AsyncInvoker::DoInvoke(Thread* thread,
   thread->Post(this, id, new ScopedRefMessageData<AsyncClosure>(closure));
 }
 
+void AsyncInvoker::DoInvokeDelayed(Thread* thread,
+                                   const scoped_refptr<AsyncClosure>& closure,
+                                   uint32_t delay_ms,
+                                   uint32_t id) {
+  if (destroying_) {
+    LOG(LS_WARNING) << "Tried to invoke while destroying the invoker.";
+    return;
+  }
+  thread->PostDelayed(delay_ms, this, id,
+                      new ScopedRefMessageData<AsyncClosure>(closure));
+}
+
+GuardedAsyncInvoker::GuardedAsyncInvoker() : thread_(Thread::Current()) {
+  thread_->SignalQueueDestroyed.connect(this,
+                                        &GuardedAsyncInvoker::ThreadDestroyed);
+}
+
+GuardedAsyncInvoker::~GuardedAsyncInvoker() {
+}
+
+bool GuardedAsyncInvoker::Flush(uint32_t id) {
+  rtc::CritScope cs(&crit_);
+  if (thread_ == nullptr)
+    return false;
+  invoker_.Flush(thread_, id);
+  return true;
+}
+
+void GuardedAsyncInvoker::ThreadDestroyed() {
+  rtc::CritScope cs(&crit_);
+  // We should never get more than one notification about the thread dying.
+  RTC_DCHECK(thread_ != nullptr);
+  thread_ = nullptr;
+}
+
 NotifyingAsyncClosureBase::NotifyingAsyncClosureBase(AsyncInvoker* invoker,
                                                      Thread* calling_thread)
     : invoker_(invoker), calling_thread_(calling_thread) {
diff --git a/media/webrtc/trunk/webrtc/base/asyncinvoker.h b/media/webrtc/trunk/webrtc/base/asyncinvoker.h
index 6e298e60cc..a35133706a 100644
--- a/media/webrtc/trunk/webrtc/base/asyncinvoker.h
+++ b/media/webrtc/trunk/webrtc/base/asyncinvoker.h
@@ -74,21 +74,31 @@ class AsyncInvoker : public MessageHandler {
   // Call |functor| asynchronously on |thread|, with no callback upon
   // completion. Returns immediately.
   template <class ReturnT, class FunctorT>
-  void AsyncInvoke(Thread* thread,
-                   const FunctorT& functor,
-                   uint32 id = 0) {
+  void AsyncInvoke(Thread* thread, const FunctorT& functor, uint32_t id = 0) {
     scoped_refptr<AsyncClosure> closure(
         new RefCountedObject<FireAndForgetAsyncClosure<FunctorT> >(functor));
     DoInvoke(thread, closure, id);
   }
 
+  // Call |functor| asynchronously on |thread| with |delay_ms|, with no callback
+  // upon completion. Returns immediately.
+  template <class ReturnT, class FunctorT>
+  void AsyncInvokeDelayed(Thread* thread,
+                          const FunctorT& functor,
+                          uint32_t delay_ms,
+                          uint32_t id = 0) {
+    scoped_refptr<AsyncClosure> closure(
+        new RefCountedObject<FireAndForgetAsyncClosure<FunctorT> >(functor));
+    DoInvokeDelayed(thread, closure, delay_ms, id);
+  }
+
   // Call |functor| asynchronously on |thread|, calling |callback| when done.
   template <class ReturnT, class FunctorT, class HostT>
   void AsyncInvoke(Thread* thread,
                    const FunctorT& functor,
                    void (HostT::*callback)(ReturnT),
                    HostT* callback_host,
-                   uint32 id = 0) {
+                   uint32_t id = 0) {
     scoped_refptr<AsyncClosure> closure(
         new RefCountedObject<NotifyingAsyncClosure<ReturnT, FunctorT, HostT> >(
             this, Thread::Current(), functor, callback, callback_host));
@@ -102,7 +112,7 @@ class AsyncInvoker : public MessageHandler {
                    const FunctorT& functor,
                    void (HostT::*callback)(),
                    HostT* callback_host,
-                   uint32 id = 0) {
+                   uint32_t id = 0) {
     scoped_refptr<AsyncClosure> closure(
         new RefCountedObject<NotifyingAsyncClosure<void, FunctorT, HostT> >(
             this, Thread::Current(), functor, callback, callback_host));
@@ -114,22 +124,106 @@ class AsyncInvoker : public MessageHandler {
   // before returning. Optionally filter by message id.
   // The destructor will not wait for outstanding calls, so if that
   // behavior is desired, call Flush() before destroying this object.
-  void Flush(Thread* thread, uint32 id = MQID_ANY);
+  void Flush(Thread* thread, uint32_t id = MQID_ANY);
 
   // Signaled when this object is destructed.
   sigslot::signal0<> SignalInvokerDestroyed;
 
  private:
   void OnMessage(Message* msg) override;
-  void DoInvoke(Thread* thread, const scoped_refptr<AsyncClosure>& closure,
-                uint32 id);
-
+  void DoInvoke(Thread* thread,
+                const scoped_refptr<AsyncClosure>& closure,
+                uint32_t id);
+  void DoInvokeDelayed(Thread* thread,
+                       const scoped_refptr<AsyncClosure>& closure,
+                       uint32_t delay_ms,
+                       uint32_t id);
   bool destroying_;
 
-  DISALLOW_COPY_AND_ASSIGN(AsyncInvoker);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncInvoker);
+};
+
+// Similar to AsyncInvoker, but guards against the Thread being destroyed while
+// there are outstanding dangling pointers to it. It will connect to the current
+// thread in the constructor, and will get notified when that thread is
+// destroyed. After GuardedAsyncInvoker is constructed, it can be used from
+// other threads to post functors to the thread it was constructed on. If that
+// thread dies, any further calls to AsyncInvoke() will be safely ignored.
+class GuardedAsyncInvoker : public sigslot::has_slots<> {
+ public:
+  GuardedAsyncInvoker();
+  ~GuardedAsyncInvoker() override;
+
+  // Synchronously execute all outstanding calls we own, and wait for calls to
+  // complete before returning. Optionally filter by message id. The destructor
+  // will not wait for outstanding calls, so if that behavior is desired, call
+  // Flush() first. Returns false if the thread has died.
+  bool Flush(uint32_t id = MQID_ANY);
+
+  // Call |functor| asynchronously with no callback upon completion. Returns
+  // immediately. Returns false if the thread has died.
+  template <class ReturnT, class FunctorT>
+  bool AsyncInvoke(const FunctorT& functor, uint32_t id = 0) {
+    rtc::CritScope cs(&crit_);
+    if (thread_ == nullptr)
+      return false;
+    invoker_.AsyncInvoke<ReturnT, FunctorT>(thread_, functor, id);
+    return true;
+  }
+
+  // Call |functor| asynchronously with |delay_ms|, with no callback upon
+  // completion. Returns immediately. Returns false if the thread has died.
+  template <class ReturnT, class FunctorT>
+  bool AsyncInvokeDelayed(const FunctorT& functor,
+                          uint32_t delay_ms,
+                          uint32_t id = 0) {
+    rtc::CritScope cs(&crit_);
+    if (thread_ == nullptr)
+      return false;
+    invoker_.AsyncInvokeDelayed<ReturnT, FunctorT>(thread_, functor, delay_ms,
+                                                   id);
+    return true;
+  }
+
+  // Call |functor| asynchronously, calling |callback| when done. Returns false
+  // if the thread has died.
+  template <class ReturnT, class FunctorT, class HostT>
+  bool AsyncInvoke(const FunctorT& functor,
+                   void (HostT::*callback)(ReturnT),
+                   HostT* callback_host,
+                   uint32_t id = 0) {
+    rtc::CritScope cs(&crit_);
+    if (thread_ == nullptr)
+      return false;
+    invoker_.AsyncInvoke<ReturnT, FunctorT, HostT>(thread_, functor, callback,
+                                                   callback_host, id);
+    return true;
+  }
+
+  // Call |functor| asynchronously calling |callback| when done. Overloaded for
+  // void return. Returns false if the thread has died.
+  template <class ReturnT, class FunctorT, class HostT>
+  bool AsyncInvoke(const FunctorT& functor,
+                   void (HostT::*callback)(),
+                   HostT* callback_host,
+                   uint32_t id = 0) {
+    rtc::CritScope cs(&crit_);
+    if (thread_ == nullptr)
+      return false;
+    invoker_.AsyncInvoke<ReturnT, FunctorT, HostT>(thread_, functor, callback,
+                                                   callback_host, id);
+    return true;
+  }
+
+ private:
+  // Callback when |thread_| is destroyed.
+  void ThreadDestroyed();
+
+  CriticalSection crit_;
+  Thread* thread_ GUARDED_BY(crit_);
+  AsyncInvoker invoker_ GUARDED_BY(crit_);
 };
 
 }  // namespace rtc
 
-
 #endif  // WEBRTC_BASE_ASYNCINVOKER_H_
diff --git a/media/webrtc/trunk/webrtc/base/asyncpacketsocket.h b/media/webrtc/trunk/webrtc/base/asyncpacketsocket.h
index 80541556ce..949ec67c83 100644
--- a/media/webrtc/trunk/webrtc/base/asyncpacketsocket.h
+++ b/media/webrtc/trunk/webrtc/base/asyncpacketsocket.h
@@ -28,16 +28,17 @@ struct PacketTimeUpdateParams {
   int rtp_sendtime_extension_id;    // extension header id present in packet.
   std::vector<char> srtp_auth_key;  // Authentication key.
   int srtp_auth_tag_len;            // Authentication tag length.
-  int64 srtp_packet_index;          // Required for Rtp Packet authentication.
+  int64_t srtp_packet_index;        // Required for Rtp Packet authentication.
 };
 
 // This structure holds meta information for the packet which is about to send
 // over network.
 struct PacketOptions {
-  PacketOptions() : dscp(DSCP_NO_CHANGE) {}
-  explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
+  PacketOptions() : dscp(DSCP_NO_CHANGE), packet_id(-1) {}
+  explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {}
 
   DiffServCodePoint dscp;
+  int packet_id;  // 16 bits, -1 represents "not set".
   PacketTimeUpdateParams packet_time_params;
 };
 
@@ -45,19 +46,19 @@ struct PacketOptions {
 // received by socket.
 struct PacketTime {
   PacketTime() : timestamp(-1), not_before(-1) {}
-  PacketTime(int64 timestamp, int64 not_before)
-      : timestamp(timestamp), not_before(not_before) {
-  }
+  PacketTime(int64_t timestamp, int64_t not_before)
+      : timestamp(timestamp), not_before(not_before) {}
 
-  int64 timestamp;  // Receive time after socket delivers the data.
-  int64 not_before; // Earliest possible time the data could have arrived,
-                    // indicating the potential error in the |timestamp| value,
-                    // in case the system, is busy. For example, the time of
-                    // the last select() call.
-                    // If unknown, this value will be set to zero.
+  int64_t timestamp;   // Receive time after socket delivers the data.
+
+  // Earliest possible time the data could have arrived, indicating the
+  // potential error in the |timestamp| value, in case the system, is busy. For
+  // example, the time of the last select() call.
+  // If unknown, this value will be set to zero.
+  int64_t not_before;
 };
 
-inline PacketTime CreatePacketTime(int64 not_before) {
+inline PacketTime CreatePacketTime(int64_t not_before) {
   return PacketTime(TimeMicros(), not_before);
 }
 
@@ -109,6 +110,9 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
                    const SocketAddress&,
                    const PacketTime&> SignalReadPacket;
 
+  // Emitted each time a packet is sent.
+  sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
+
   // Emitted when the socket is currently able to send.
   sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend;
 
@@ -130,7 +134,7 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
   sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncPacketSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/asyncsocket.cc b/media/webrtc/trunk/webrtc/base/asyncsocket.cc
index dc0de3dd48..db451c6382 100644
--- a/media/webrtc/trunk/webrtc/base/asyncsocket.cc
+++ b/media/webrtc/trunk/webrtc/base/asyncsocket.cc
@@ -96,7 +96,7 @@ AsyncSocket::ConnState AsyncSocketAdapter::GetState() const {
   return socket_->GetState();
 }
 
-int AsyncSocketAdapter::EstimateMTU(uint16* mtu) {
+int AsyncSocketAdapter::EstimateMTU(uint16_t* mtu) {
   return socket_->EstimateMTU(mtu);
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/asyncsocket.h b/media/webrtc/trunk/webrtc/base/asyncsocket.h
index 37bad4c831..7a859be962 100644
--- a/media/webrtc/trunk/webrtc/base/asyncsocket.h
+++ b/media/webrtc/trunk/webrtc/base/asyncsocket.h
@@ -64,7 +64,7 @@ class AsyncSocketAdapter : public AsyncSocket, public sigslot::has_slots<> {
   int GetError() const override;
   void SetError(int error) override;
   ConnState GetState() const override;
-  int EstimateMTU(uint16* mtu) override;
+  int EstimateMTU(uint16_t* mtu) override;
   int GetOption(Option opt, int* value) override;
   int SetOption(Option opt, int value) override;
 
diff --git a/media/webrtc/trunk/webrtc/base/asynctcpsocket.cc b/media/webrtc/trunk/webrtc/base/asynctcpsocket.cc
index 0f7abd5a6c..8e83cd1ea2 100644
--- a/media/webrtc/trunk/webrtc/base/asynctcpsocket.cc
+++ b/media/webrtc/trunk/webrtc/base/asynctcpsocket.cc
@@ -24,7 +24,7 @@ namespace rtc {
 
 static const size_t kMaxPacketSize = 64 * 1024;
 
-typedef uint16 PacketLength;
+typedef uint16_t PacketLength;
 static const size_t kPacketLenSize = sizeof(PacketLength);
 
 static const size_t kBufSize = kMaxPacketSize + kPacketLenSize;
@@ -127,10 +127,11 @@ void AsyncTCPSocketBase::SetError(int error) {
 int AsyncTCPSocketBase::SendTo(const void *pv, size_t cb,
                                const SocketAddress& addr,
                                const rtc::PacketOptions& options) {
-  if (addr == GetRemoteAddress())
+  const SocketAddress& remote_address = GetRemoteAddress();
+  if (addr == remote_address)
     return Send(pv, cb, options);
-
-  ASSERT(false);
+  // Remote address may be empty if there is a sudden network change.
+  ASSERT(remote_address.IsNil());
   socket_->SetError(ENOTCONN);
   return -1;
 }
@@ -267,6 +268,9 @@ int AsyncTCPSocket::Send(const void *pv, size_t cb,
     return res;
   }
 
+  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  SignalSentPacket(this, sent_packet);
+
   // We claim to have sent the whole thing, even if we only sent partial
   return static_cast<int>(cb);
 }
diff --git a/media/webrtc/trunk/webrtc/base/asynctcpsocket.h b/media/webrtc/trunk/webrtc/base/asynctcpsocket.h
index c50579d508..321fa23493 100644
--- a/media/webrtc/trunk/webrtc/base/asynctcpsocket.h
+++ b/media/webrtc/trunk/webrtc/base/asynctcpsocket.h
@@ -74,7 +74,7 @@ class AsyncTCPSocketBase : public AsyncPacketSocket {
   char* inbuf_, * outbuf_;
   size_t insize_, inpos_, outsize_, outpos_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncTCPSocketBase);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocketBase);
 };
 
 class AsyncTCPSocket : public AsyncTCPSocketBase {
@@ -95,7 +95,7 @@ class AsyncTCPSocket : public AsyncTCPSocketBase {
   void HandleIncomingConnection(AsyncSocket* socket) override;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncTCPSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocket);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/asyncudpsocket.cc b/media/webrtc/trunk/webrtc/base/asyncudpsocket.cc
index 3e2ecc4cda..51a8fa0af0 100644
--- a/media/webrtc/trunk/webrtc/base/asyncudpsocket.cc
+++ b/media/webrtc/trunk/webrtc/base/asyncudpsocket.cc
@@ -60,13 +60,19 @@ SocketAddress AsyncUDPSocket::GetRemoteAddress() const {
 
 int AsyncUDPSocket::Send(const void *pv, size_t cb,
                          const rtc::PacketOptions& options) {
-  return socket_->Send(pv, cb);
+  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  int ret = socket_->Send(pv, cb);
+  SignalSentPacket(this, sent_packet);
+  return ret;
 }
 
 int AsyncUDPSocket::SendTo(const void *pv, size_t cb,
                            const SocketAddress& addr,
                            const rtc::PacketOptions& options) {
-  return socket_->SendTo(pv, cb, addr);
+  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  int ret = socket_->SendTo(pv, cb, addr);
+  SignalSentPacket(this, sent_packet);
+  return ret;
 }
 
 int AsyncUDPSocket::Close() {
diff --git a/media/webrtc/trunk/webrtc/base/atomicops.h b/media/webrtc/trunk/webrtc/base/atomicops.h
index 6096e8c084..a286bf01cc 100644
--- a/media/webrtc/trunk/webrtc/base/atomicops.h
+++ b/media/webrtc/trunk/webrtc/base/atomicops.h
@@ -11,139 +11,77 @@
 #ifndef WEBRTC_BASE_ATOMICOPS_H_
 #define WEBRTC_BASE_ATOMICOPS_H_
 
-#include <string>
-
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
+#if defined(WEBRTC_WIN)
+// Include winsock2.h before including <windows.h> to maintain consistency with
+// win32.h.  We can't include win32.h directly here since it pulls in
+// headers such as basictypes.h which causes problems in Chromium where webrtc
+// exists as two separate projects, webrtc and libjingle.
+#include <winsock2.h>
+#include <windows.h>
+#endif  // defined(WEBRTC_WIN)
 
 namespace rtc {
-
-// A single-producer, single-consumer, fixed-size queue.
-// All methods not ending in Unsafe can be safely called without locking,
-// provided that calls to consumer methods (Peek/Pop) or producer methods (Push)
-// only happen on a single thread per method type. If multiple threads need to
-// read simultaneously or write simultaneously, other synchronization is
-// necessary. Synchronization is also required if a call into any Unsafe method
-// could happen at the same time as a call to any other method.
-template <typename T>
-class FixedSizeLockFreeQueue {
- private:
-// Atomic primitives and memory barrier
-#if defined(__arm__)
-  typedef uint32 Atomic32;
-
-  // Copied from google3/base/atomicops-internals-arm-v6plus.h
-  static inline void MemoryBarrier() {
-    asm volatile("dmb":::"memory");
-  }
-
-  // Adapted from google3/base/atomicops-internals-arm-v6plus.h
-  static inline void AtomicIncrement(volatile Atomic32* ptr) {
-    Atomic32 str_success, value;
-    asm volatile (
-        "1:\n"
-        "ldrex  %1, [%2]\n"
-        "add    %1, %1, #1\n"
-        "strex  %0, %1, [%2]\n"
-        "teq    %0, #0\n"
-        "bne    1b"
-        : "=&r"(str_success), "=&r"(value)
-        : "r" (ptr)
-        : "cc", "memory");
-  }
-#elif !defined(SKIP_ATOMIC_CHECK)
-#error "No atomic operations defined for the given architecture."
-#endif
-
+class AtomicOps {
  public:
-  // Constructs an empty queue, with capacity 0.
-  FixedSizeLockFreeQueue() : pushed_count_(0),
-                             popped_count_(0),
-                             capacity_(0),
-                             data_() {}
-  // Constructs an empty queue with the given capacity.
-  FixedSizeLockFreeQueue(size_t capacity) : pushed_count_(0),
-                                            popped_count_(0),
-                                            capacity_(capacity),
-                                            data_(new T[capacity]) {}
-
-  // Pushes a value onto the queue. Returns true if the value was successfully
-  // pushed (there was space in the queue). This method can be safely called at
-  // the same time as PeekFront/PopFront.
-  bool PushBack(T value) {
-    if (capacity_ == 0) {
-      LOG(LS_WARNING) << "Queue capacity is 0.";
-      return false;
-    }
-    if (IsFull()) {
-      return false;
-    }
-
-    data_[pushed_count_ % capacity_] = value;
-    // Make sure the data is written before the count is incremented, so other
-    // threads can't see the value exists before being able to read it.
-    MemoryBarrier();
-    AtomicIncrement(&pushed_count_);
-    return true;
+#if defined(WEBRTC_WIN)
+  // Assumes sizeof(int) == sizeof(LONG), which it is on Win32 and Win64.
+  static int Increment(volatile int* i) {
+    return ::InterlockedIncrement(reinterpret_cast<volatile LONG*>(i));
   }
-
-  // Retrieves the oldest value pushed onto the queue. Returns true if there was
-  // an item to peek (the queue was non-empty). This method can be safely called
-  // at the same time as PushBack.
-  bool PeekFront(T* value_out) {
-    if (capacity_ == 0) {
-      LOG(LS_WARNING) << "Queue capacity is 0.";
-      return false;
-    }
-    if (IsEmpty()) {
-      return false;
-    }
-
-    *value_out = data_[popped_count_ % capacity_];
-    return true;
+  static int Decrement(volatile int* i) {
+    return ::InterlockedDecrement(reinterpret_cast<volatile LONG*>(i));
   }
-
-  // Retrieves the oldest value pushed onto the queue and removes it from the
-  // queue. Returns true if there was an item to pop (the queue was non-empty).
-  // This method can be safely called at the same time as PushBack.
-  bool PopFront(T* value_out) {
-    if (PeekFront(value_out)) {
-      AtomicIncrement(&popped_count_);
-      return true;
-    }
-    return false;
+  static int AcquireLoad(volatile const int* i) {
+    return *i;
   }
-
-  // Clears the current items in the queue and sets the new (fixed) size. This
-  // method cannot be called at the same time as any other method.
-  void ClearAndResizeUnsafe(int new_capacity) {
-    capacity_ = new_capacity;
-    data_.reset(new T[new_capacity]);
-    pushed_count_ = 0;
-    popped_count_ = 0;
+  static void ReleaseStore(volatile int* i, int value) {
+    *i = value;
   }
-
-  // Returns true if there is no space left in the queue for new elements.
-  int IsFull() const { return pushed_count_ == popped_count_ + capacity_; }
-  // Returns true if there are no elements in the queue.
-  int IsEmpty() const { return pushed_count_ == popped_count_; }
-  // Returns the current number of elements in the queue. This is always in the
-  // range [0, capacity]
-  size_t Size() const { return pushed_count_ - popped_count_; }
-
-  // Returns the capacity of the queue (max size).
-  size_t capacity() const { return capacity_; }
-
- private:
-  volatile Atomic32 pushed_count_;
-  volatile Atomic32 popped_count_;
-  size_t capacity_;
-  rtc::scoped_ptr<T[]> data_;
-  DISALLOW_COPY_AND_ASSIGN(FixedSizeLockFreeQueue);
+  static int CompareAndSwap(volatile int* i, int old_value, int new_value) {
+    return ::InterlockedCompareExchange(reinterpret_cast<volatile LONG*>(i),
+                                        new_value,
+                                        old_value);
+  }
+  // Pointer variants.
+  template <typename T>
+  static T* AcquireLoadPtr(T* volatile* ptr) {
+    return *ptr;
+  }
+  template <typename T>
+  static T* CompareAndSwapPtr(T* volatile* ptr, T* old_value, T* new_value) {
+    return static_cast<T*>(::InterlockedCompareExchangePointer(
+        reinterpret_cast<PVOID volatile*>(ptr), new_value, old_value));
+  }
+#else
+  static int Increment(volatile int* i) {
+    return __sync_add_and_fetch(i, 1);
+  }
+  static int Decrement(volatile int* i) {
+    return __sync_sub_and_fetch(i, 1);
+  }
+  static int AcquireLoad(volatile const int* i) {
+    return __atomic_load_n(i, __ATOMIC_ACQUIRE);
+  }
+  static void ReleaseStore(volatile int* i, int value) {
+    __atomic_store_n(i, value, __ATOMIC_RELEASE);
+  }
+  static int CompareAndSwap(volatile int* i, int old_value, int new_value) {
+    return __sync_val_compare_and_swap(i, old_value, new_value);
+  }
+  // Pointer variants.
+  template <typename T>
+  static T* AcquireLoadPtr(T* volatile* ptr) {
+    return __atomic_load_n(ptr, __ATOMIC_ACQUIRE);
+  }
+  template <typename T>
+  static T* CompareAndSwapPtr(T* volatile* ptr, T* old_value, T* new_value) {
+    return __sync_val_compare_and_swap(ptr, old_value, new_value);
+  }
+#endif
 };
 
+
+
 }
 
 #endif  // WEBRTC_BASE_ATOMICOPS_H_
diff --git a/media/webrtc/trunk/webrtc/base/atomicops_unittest.cc b/media/webrtc/trunk/webrtc/base/atomicops_unittest.cc
index 5152c4de6c..2aa28cac61 100644
--- a/media/webrtc/trunk/webrtc/base/atomicops_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/atomicops_unittest.cc
@@ -8,72 +8,5 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#if !defined(__arm__)
-// For testing purposes, define faked versions of the atomic operations
-#include "webrtc/base/basictypes.h"
-namespace rtc {
-typedef uint32 Atomic32;
-static inline void MemoryBarrier() { }
-static inline void AtomicIncrement(volatile Atomic32* ptr) {
-  *ptr = *ptr + 1;
-}
-}
-#define SKIP_ATOMIC_CHECK
-#endif
-
-#include "webrtc/base/atomicops.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/logging.h"
-
-TEST(FixedSizeLockFreeQueueTest, TestDefaultConstruct) {
-  rtc::FixedSizeLockFreeQueue<int> queue;
-  EXPECT_EQ(0u, queue.capacity());
-  EXPECT_EQ(0u, queue.Size());
-  EXPECT_FALSE(queue.PushBack(1));
-  int val;
-  EXPECT_FALSE(queue.PopFront(&val));
-}
-
-TEST(FixedSizeLockFreeQueueTest, TestConstruct) {
-  rtc::FixedSizeLockFreeQueue<int> queue(5);
-  EXPECT_EQ(5u, queue.capacity());
-  EXPECT_EQ(0u, queue.Size());
-  int val;
-  EXPECT_FALSE(queue.PopFront(&val));
-}
-
-TEST(FixedSizeLockFreeQueueTest, TestPushPop) {
-  rtc::FixedSizeLockFreeQueue<int> queue(2);
-  EXPECT_EQ(2u, queue.capacity());
-  EXPECT_EQ(0u, queue.Size());
-  EXPECT_TRUE(queue.PushBack(1));
-  EXPECT_EQ(1u, queue.Size());
-  EXPECT_TRUE(queue.PushBack(2));
-  EXPECT_EQ(2u, queue.Size());
-  EXPECT_FALSE(queue.PushBack(3));
-  EXPECT_EQ(2u, queue.Size());
-  int val;
-  EXPECT_TRUE(queue.PopFront(&val));
-  EXPECT_EQ(1, val);
-  EXPECT_EQ(1u, queue.Size());
-  EXPECT_TRUE(queue.PopFront(&val));
-  EXPECT_EQ(2, val);
-  EXPECT_EQ(0u, queue.Size());
-  EXPECT_FALSE(queue.PopFront(&val));
-  EXPECT_EQ(0u, queue.Size());
-}
-
-TEST(FixedSizeLockFreeQueueTest, TestResize) {
-  rtc::FixedSizeLockFreeQueue<int> queue(2);
-  EXPECT_EQ(2u, queue.capacity());
-  EXPECT_EQ(0u, queue.Size());
-  EXPECT_TRUE(queue.PushBack(1));
-  EXPECT_EQ(1u, queue.Size());
-
-  queue.ClearAndResizeUnsafe(5);
-  EXPECT_EQ(5u, queue.capacity());
-  EXPECT_EQ(0u, queue.Size());
-  int val;
-  EXPECT_FALSE(queue.PopFront(&val));
-}
+// TODO(pbos): Move AtomicOps tests to here from
+// webrtc/base/criticalsection_unittest.cc.
diff --git a/media/webrtc/trunk/webrtc/base/autodetectproxy.cc b/media/webrtc/trunk/webrtc/base/autodetectproxy.cc
index 4ebc2d4da1..22950fb2b3 100644
--- a/media/webrtc/trunk/webrtc/base/autodetectproxy.cc
+++ b/media/webrtc/trunk/webrtc/base/autodetectproxy.cc
@@ -102,7 +102,7 @@ void AutoDetectProxy::OnMessage(Message *msg) {
 
     IPAddress address_ip = proxy().address.ipaddr();
 
-    uint16 address_port = proxy().address.port();
+    uint16_t address_port = proxy().address.port();
 
     char autoconfig_url[kSavedStringLimit];
     SaveStringToStack(autoconfig_url,
diff --git a/media/webrtc/trunk/webrtc/base/autodetectproxy.h b/media/webrtc/trunk/webrtc/base/autodetectproxy.h
index b9887bceb3..1bd523f8db 100644
--- a/media/webrtc/trunk/webrtc/base/autodetectproxy.h
+++ b/media/webrtc/trunk/webrtc/base/autodetectproxy.h
@@ -81,7 +81,7 @@ class AutoDetectProxy : public SignalThread {
   AsyncSocket* socket_;
   int next_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(AutoDetectProxy);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AutoDetectProxy);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/autodetectproxy_unittest.cc b/media/webrtc/trunk/webrtc/base/autodetectproxy_unittest.cc
index 4a2688265c..2ae7a6aa25 100644
--- a/media/webrtc/trunk/webrtc/base/autodetectproxy_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/autodetectproxy_unittest.cc
@@ -12,14 +12,13 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/httpcommon.h"
 #include "webrtc/base/httpcommon-inl.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
 static const char kUserAgent[] = "";
 static const char kPath[] = "/";
 static const char kHost[] = "relay.google.com";
-static const uint16 kPort = 443;
+static const uint16_t kPort = 443;
 static const bool kSecure = true;
 // At most, AutoDetectProxy should take ~6 seconds. Each connect step is
 // allotted 2 seconds, with the initial resolution + connect given an
@@ -37,10 +36,10 @@ class AutoDetectProxyTest : public testing::Test, public sigslot::has_slots<> {
   AutoDetectProxyTest() : auto_detect_proxy_(NULL), done_(false) {}
 
  protected:
-  bool Create(const std::string &user_agent,
-              const std::string &path,
-              const std::string &host,
-              uint16 port,
+  bool Create(const std::string& user_agent,
+              const std::string& path,
+              const std::string& host,
+              uint16_t port,
               bool secure,
               bool startnow) {
     auto_detect_proxy_ = new AutoDetectProxy(user_agent);
diff --git a/media/webrtc/trunk/webrtc/base/bandwidthsmoother.cc b/media/webrtc/trunk/webrtc/base/bandwidthsmoother.cc
index 09c7b9941b..d48c12e6c6 100644
--- a/media/webrtc/trunk/webrtc/base/bandwidthsmoother.cc
+++ b/media/webrtc/trunk/webrtc/base/bandwidthsmoother.cc
@@ -16,7 +16,7 @@
 namespace rtc {
 
 BandwidthSmoother::BandwidthSmoother(int initial_bandwidth_guess,
-                                     uint32 time_between_increase,
+                                     uint32_t time_between_increase,
                                      double percent_increase,
                                      size_t samples_count_to_average,
                                      double min_sample_count_percent)
@@ -33,7 +33,7 @@ BandwidthSmoother::~BandwidthSmoother() = default;
 
 // Samples a new bandwidth measurement
 // returns true if the bandwidth estimation changed
-bool BandwidthSmoother::Sample(uint32 sample_time, int bandwidth) {
+bool BandwidthSmoother::Sample(uint32_t sample_time, int bandwidth) {
   if (bandwidth < 0) {
     return false;
   }
diff --git a/media/webrtc/trunk/webrtc/base/bandwidthsmoother.h b/media/webrtc/trunk/webrtc/base/bandwidthsmoother.h
index dbb4c81558..eae565ead3 100644
--- a/media/webrtc/trunk/webrtc/base/bandwidthsmoother.h
+++ b/media/webrtc/trunk/webrtc/base/bandwidthsmoother.h
@@ -31,7 +31,7 @@ namespace rtc {
 class BandwidthSmoother {
  public:
   BandwidthSmoother(int initial_bandwidth_guess,
-                    uint32 time_between_increase,
+                    uint32_t time_between_increase,
                     double percent_increase,
                     size_t samples_count_to_average,
                     double min_sample_count_percent);
@@ -40,16 +40,16 @@ class BandwidthSmoother {
   // Samples a new bandwidth measurement.
   // bandwidth is expected to be non-negative.
   // returns true if the bandwidth estimation changed
-  bool Sample(uint32 sample_time, int bandwidth);
+  bool Sample(uint32_t sample_time, int bandwidth);
 
   int get_bandwidth_estimation() const {
     return bandwidth_estimation_;
   }
 
  private:
-  uint32 time_between_increase_;
+  uint32_t time_between_increase_;
   double percent_increase_;
-  uint32 time_at_last_change_;
+  uint32_t time_at_last_change_;
   int bandwidth_estimation_;
   RollingAccumulator<int> accumulator_;
   double min_sample_count_percent_;
diff --git a/media/webrtc/trunk/webrtc/base/base.gyp b/media/webrtc/trunk/webrtc/base/base.gyp
index e7c6c90307..d269f41603 100644
--- a/media/webrtc/trunk/webrtc/base/base.gyp
+++ b/media/webrtc/trunk/webrtc/base/base.gyp
@@ -22,6 +22,46 @@
         }],
       ],
     }],
+    # TODO(tkchin): Mac support. There are a bunch of problems right now because
+    # of some settings pulled down from Chromium.
+    ['OS=="ios"', {
+      'targets': [
+        {
+          'target_name': 'rtc_base_objc',
+          'type': 'static_library',
+          'dependencies': [
+            'rtc_base',
+          ],
+          'sources': [
+            'objc/NSString+StdString.h',
+            'objc/NSString+StdString.mm',
+            'objc/RTCDispatcher.h',
+            'objc/RTCDispatcher.m',
+            'objc/RTCLogging.h',
+            'objc/RTCLogging.mm',
+          ],
+          'conditions': [
+            ['OS=="ios"', {
+              'sources': [
+                'objc/RTCCameraPreviewView.h',
+                'objc/RTCCameraPreviewView.m',
+              ],
+              'all_dependent_settings': {
+                'xcode_settings': {
+                  'OTHER_LDFLAGS': [
+                    '-framework AVFoundation',
+                  ],
+                },
+              },
+            }],
+          ],
+          'xcode_settings': {
+            'CLANG_ENABLE_OBJC_ARC': 'YES',
+            'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
+          },
+        }
+      ],
+    }], # OS=="ios"
   ],
   'targets': [
     {
@@ -29,31 +69,56 @@
       'target_name': 'rtc_base_approved',
       'type': 'static_library',
       'sources': [
+        'array_view.h',
+        'atomicops.h',
         'bitbuffer.cc',
         'bitbuffer.h',
         'buffer.cc',
         'buffer.h',
+        'bufferqueue.cc',
+        'bufferqueue.h',
+        'bytebuffer.cc',
+        'bytebuffer.h',
+        'byteorder.h',
         'checks.cc',
         'checks.h',
+        'common.cc',
+        'common.h',
         'constructormagic.h',
+        'criticalsection.cc',
+        'criticalsection.h',
+        'deprecation.h',
         'event.cc',
         'event.h',
         'event_tracer.cc',
         'event_tracer.h',
         'exp_filter.cc',
         'exp_filter.h',
+        'logging.cc',
+        'logging.h',
         'md5.cc',
         'md5.h',
         'md5digest.cc',
         'md5digest.h',
+        'optional.h',
         'platform_file.cc',
         'platform_file.h',
+        'platform_thread.cc',
+        'platform_thread.h',
+        'platform_thread_types.h',
+        'random.cc',
+        'random.h',
+        'ratetracker.cc',
+        'ratetracker.h',
         'safe_conversions.h',
         'safe_conversions_impl.h',
+        'scoped_ptr.h',
         'stringencode.cc',
         'stringencode.h',
         'stringutils.cc',
         'stringutils.h',
+        'systeminfo.cc',
+        'systeminfo.h',
         'template_util.h',
         'thread_annotations.h',
         'thread_checker.h',
@@ -64,6 +129,22 @@
         'trace_event.h',
       ],
       'conditions': [
+        ['build_with_chromium==1', {
+          'dependencies': [
+            '<(DEPTH)/base/base.gyp:base',
+          ],
+          'include_dirs': [
+            '../../webrtc_overrides',
+          ],
+          'sources': [
+            '../../webrtc_overrides/webrtc/base/logging.cc',
+            '../../webrtc_overrides/webrtc/base/logging.h',
+          ],
+          'sources!': [
+            'logging.cc',
+            'logging.h',
+          ],
+        }],
         ['OS=="mac"', {
           'sources': [
             'macutils.cc',
@@ -82,19 +163,22 @@
       'target_name': 'rtc_base',
       'type': 'static_library',
       'dependencies': [
-        '<(webrtc_root)/common.gyp:webrtc_common',
+        '../common.gyp:webrtc_common',
+        'rtc_base_approved',
+      ],
+      'export_dependent_settings': [
         'rtc_base_approved',
       ],
       'defines': [
         'FEATURE_ENABLE_SSL',
+        'SSL_USE_OPENSSL',
+        'HAVE_OPENSSL_SSL_H',
         'LOGGING=1',
       ],
       'sources': [
         'arraysize.h',
         'asyncfile.cc',
         'asyncfile.h',
-        'asynchttprequest.cc',
-        'asynchttprequest.h',
         'asyncinvoker.cc',
         'asyncinvoker.h',
         'asyncinvoker-inl.h',
@@ -108,33 +192,16 @@
         'asynctcpsocket.h',
         'asyncudpsocket.cc',
         'asyncudpsocket.h',
-        'atomicops.h',
         'autodetectproxy.cc',
         'autodetectproxy.h',
         'bandwidthsmoother.cc',
         'bandwidthsmoother.h',
         'base64.cc',
         'base64.h',
-        'basicdefs.h',
-        'basictypes.h',
         'bind.h',
-#        'bind.h.pump',
-        'buffer.cc',
-        'buffer.h',
-        'bytebuffer.cc',
-        'bytebuffer.h',
-        'byteorder.h',
         'callback.h',
-#        'callback.h.pump',
-        'constructormagic.h',
-        'common.cc',
-        'common.h',
-        'cpumonitor.cc',
-        'cpumonitor.h',
         'crc32.cc',
         'crc32.h',
-        'criticalsection.cc',
-        'criticalsection.h',
         'cryptstring.cc',
         'cryptstring.h',
         'dbus.cc',
@@ -143,8 +210,8 @@
         'diskcache.h',
         'diskcache_win32.cc',
         'diskcache_win32.h',
-        'filelock.cc',
-        'filelock.h',
+        'filerotatingstream.cc',
+        'filerotatingstream.h',
         'fileutils.cc',
         'fileutils.h',
         'fileutils_mock.h',
@@ -169,15 +236,16 @@
         'httpserver.h',
         'ifaddrs-android.cc',
         'ifaddrs-android.h',
+        'ifaddrs_converter.cc',
+        'ifaddrs_converter.h',
+        'macifaddrs_converter.cc',
         'iosfilesystem.mm',
         'ipaddress.cc',
         'ipaddress.h',
         'json.cc',
         'json.h',
         'latebindingsymboltable.cc',
-#        'latebindingsymboltable.cc.def',
         'latebindingsymboltable.h',
-#        'latebindingsymboltable.h.def',
         'libdbusglibsymboltable.cc',
         'libdbusglibsymboltable.h',
         'linux.cc',
@@ -185,8 +253,8 @@
         'linuxfdwalk.c',
         'linuxfdwalk.h',
         'linked_ptr.h',
-        'logging.cc',
-        'logging.h',
+        'logsinks.cc',
+        'logsinks.h',
         'macasyncsocket.cc',
         'macasyncsocket.h',
         'maccocoasocketserver.h',
@@ -221,7 +289,18 @@
         'nethelpers.h',
         'network.cc',
         'network.h',
+        'networkmonitor.cc',
+        'networkmonitor.h',
         'nullsocketserver.h',
+        'openssl.h',
+        'openssladapter.cc',
+        'openssladapter.h',
+        'openssldigest.cc',
+        'openssldigest.h',
+        'opensslidentity.cc',
+        'opensslidentity.h',
+        'opensslstreamadapter.cc',
+        'opensslstreamadapter.h',
         'optionsfile.cc',
         'optionsfile.h',
         'pathutils.cc',
@@ -240,16 +319,13 @@
         'proxyserver.h',
         'ratelimiter.cc',
         'ratelimiter.h',
-        'ratetracker.cc',
-        'ratetracker.h',
         'refcount.h',
         'referencecountedsingletonfactory.h',
         'rollingaccumulator.h',
-        'schanneladapter.cc',
-        'schanneladapter.h',
+        'rtccertificate.cc',
+        'rtccertificate.h',
         'scoped_autorelease_pool.h',
         'scoped_autorelease_pool.mm',
-        'scoped_ptr.h',
         'scoped_ref_ptr.h',
         'scopedptrcollection.h',
         'sec_buffer.h',
@@ -293,8 +369,6 @@
         'sslstreamadapterhelper.h',
         'stream.cc',
         'stream.h',
-        'systeminfo.cc',
-        'systeminfo.h',
         'task.cc',
         'task.h',
         'taskparent.cc',
@@ -343,11 +417,6 @@
         'worker.h',
         'x11windowpicker.cc',
         'x11windowpicker.h',
-        '../overrides/webrtc/base/basictypes.h',
-        '../overrides/webrtc/base/constructormagic.h',
-        '../overrides/webrtc/base/logging.cc',
-        '../overrides/webrtc/base/logging.h',
-        '../overrides/webrtc/base/win32socketinit.cc',
       ],
       # TODO(henrike): issue 3307, make rtc_base build without disabling
       # these flags.
@@ -364,6 +433,8 @@
         ],
         'defines': [
           'FEATURE_ENABLE_SSL',
+          'SSL_USE_OPENSSL',
+          'HAVE_OPENSSL_SSL_H',
         ],
       },
       'include_dirs': [
@@ -373,39 +444,30 @@
       'conditions': [
         ['build_with_chromium==1', {
           'include_dirs': [
-            '../overrides',
+            '../../webrtc_overrides',
             '../../boringssl/src/include',
           ],
+          'sources': [
+            '../../webrtc_overrides/webrtc/base/win32socketinit.cc',
+          ],
           'sources!': [
-            'asyncinvoker.cc',
-            'asyncinvoker.h',
-            'asyncinvoker-inl.h',
             'atomicops.h',
             'bandwidthsmoother.cc',
             'bandwidthsmoother.h',
-            'basictypes.h',
             'bind.h',
-#            'bind.h.pump',
             'callback.h',
-#            'callback.h.pump',
-            'constructormagic.h',
             'dbus.cc',
             'dbus.h',
             'diskcache_win32.cc',
             'diskcache_win32.h',
-            'filelock.cc',
-            'filelock.h',
             'fileutils_mock.h',
             'genericslot.h',
-#            'genericslot.h.pump',
             'httpserver.cc',
             'httpserver.h',
             'json.cc',
             'json.h',
             'latebindingsymboltable.cc',
-#            'latebindingsymboltable.cc.def',
             'latebindingsymboltable.h',
-#            'latebindingsymboltable.h.def',
             'libdbusglibsymboltable.cc',
             'libdbusglibsymboltable.h',
             'linuxfdwalk.c',
@@ -414,6 +476,8 @@
             'x11windowpicker.h',
             'logging.cc',
             'logging.h',
+            'logsinks.cc',
+            'logsinks.h',
             'macasyncsocket.cc',
             'macasyncsocket.h',
             'maccocoasocketserver.h',
@@ -495,92 +559,17 @@
                 'WEBRTC_EXTERNAL_JSON',
               ],
             }],
-          ],
-          'sources!': [
-            '../overrides/webrtc/base/basictypes.h',
-            '../overrides/webrtc/base/constructormagic.h',
-            '../overrides/webrtc/base/win32socketinit.cc',
-            '../overrides/webrtc/base/logging.cc',
-            '../overrides/webrtc/base/logging.h',
-          ],
-        }],
-        ['use_openssl==1', {
-          'defines': [
-            'SSL_USE_OPENSSL',
-            'HAVE_OPENSSL_SSL_H',
-          ],
-          'direct_dependent_settings': {
-            'defines': [
-              'SSL_USE_OPENSSL',
-              'HAVE_OPENSSL_SSL_H',
-            ],
-          },
-          'sources': [
-            'openssl.h',
-            'openssladapter.cc',
-            'openssladapter.h',
-            'openssldigest.cc',
-            'openssldigest.h',
-            'opensslidentity.cc',
-            'opensslidentity.h',
-            'opensslstreamadapter.cc',
-            'opensslstreamadapter.h',
-          ],
-          'conditions': [
-            ['build_ssl==1', {
-              'dependencies': [
-                '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
-              ],
-            }, {
-              'include_dirs': [
-                '<(ssl_root)',
-              ],
-            }],
-          ],
-        }, {
-          'sources': [
-            'nssidentity.cc',
-            'nssidentity.h',
-            'nssstreamadapter.cc',
-            'nssstreamadapter.h',
-          ],
-          'conditions': [
-            ['use_legacy_ssl_defaults!=1', {
-              'defines': [
-                'SSL_USE_NSS',
-                'HAVE_NSS_SSL_H',
-                'SSL_USE_NSS_RNG',
-              ],
-              'direct_dependent_settings': {
-                'defines': [
-                  'SSL_USE_NSS',
-                  'HAVE_NSS_SSL_H',
-                  'SSL_USE_NSS_RNG',
-                ],
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-missing-braces',
+                  ],
+                },
               },
             }],
-            ['build_ssl==1', {
-              'conditions': [
-                # On some platforms, the rest of NSS is bundled. On others,
-                # it's pulled from the system.
-                ['OS == "mac" or OS == "ios" or OS == "win"', {
-                  'dependencies': [
-                    '<(DEPTH)/net/third_party/nss/ssl.gyp:libssl',
-                    '<(DEPTH)/third_party/nss/nss.gyp:nspr',
-                    '<(DEPTH)/third_party/nss/nss.gyp:nss',
-                  ],
-                }],
-                ['os_posix == 1 and OS != "mac" and OS != "ios" and OS != "android"', {
-                  'dependencies': [
-                    '<(DEPTH)/build/linux/system.gyp:ssl',
-                  ],
-                }],
-              ],
-            }, {
-              'include_dirs': [
-                '<(ssl_root)',
-              ],
-            }],
           ],
         }],
         ['OS == "android"', {
@@ -597,9 +586,13 @@
           ],
         }],
         ['OS=="ios"', {
+          'sources/': [
+            ['include', 'macconversion.*'],
+          ],
           'all_dependent_settings': {
             'xcode_settings': {
               'OTHER_LDFLAGS': [
+                '-framework CFNetwork',
                 '-framework Foundation',
                 '-framework Security',
                 '-framework SystemConfiguration',
@@ -685,6 +678,9 @@
           ],
         }],
         ['OS=="win"', {
+          'sources!': [
+            'ifaddrs_converter.cc',
+          ],
           'link_settings': {
             'libraries': [
               '-lcrypt32.lib',
@@ -702,8 +698,6 @@
             ['exclude', 'win32[a-z0-9]*\\.(h|cc)$'],
           ],
           'sources!': [
-              'schanneladapter.cc',
-              'schanneladapter.h',
               'winping.cc',
               'winping.h',
               'winfirewall.cc',
@@ -738,6 +732,7 @@
         }],
         ['OS!="ios" and OS!="mac"', {
           'sources!': [
+            'macifaddrs_converter.cc',
             'scoped_autorelease_pool.mm',
           ],
         }],
@@ -747,6 +742,15 @@
             'linux.h',
           ],
         }],
+        ['build_ssl==1', {
+          'dependencies': [
+            '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
+          ],
+        }, {
+          'include_dirs': [
+            '<(ssl_root)',
+          ],
+        }],
       ],
     },
   ],
diff --git a/media/webrtc/trunk/webrtc/base/base_tests.gyp b/media/webrtc/trunk/webrtc/base/base_tests.gyp
index 3a9b164d62..5d73d50756 100644
--- a/media/webrtc/trunk/webrtc/base/base_tests.gyp
+++ b/media/webrtc/trunk/webrtc/base/base_tests.gyp
@@ -15,7 +15,6 @@
         'unittest_main.cc',
         # Also use this as a convenient dumping ground for misc files that are
         # included by multiple targets below.
-        'fakecpumonitor.h',
         'fakenetwork.h',
         'fakesslidentity.h',
         'faketaskrunner.h',
@@ -23,7 +22,6 @@
         'testbase64.h',
         'testechoserver.h',
         'testutils.h',
-        'win32toolhelp.h',
       ],
       'defines': [
         'GTEST_RELATIVE_PATH',
@@ -31,6 +29,7 @@
       'dependencies': [
         'base.gyp:rtc_base',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(webrtc_root)/test/test.gyp:field_trial',
       ],
       'direct_dependent_settings': {
         'defines': [
@@ -46,24 +45,25 @@
       'type': 'none',
       'direct_dependent_settings': {
         'sources': [
-          'asynchttprequest_unittest.cc',
+          'array_view_unittest.cc',
           'atomicops_unittest.cc',
           'autodetectproxy_unittest.cc',
           'bandwidthsmoother_unittest.cc',
           'base64_unittest.cc',
           'basictypes_unittest.cc',
           'bind_unittest.cc',
+          'bitbuffer_unittest.cc',
           'buffer_unittest.cc',
+          'bufferqueue_unittest.cc',
           'bytebuffer_unittest.cc',
           'byteorder_unittest.cc',
           'callback_unittest.cc',
-          'cpumonitor_unittest.cc',
           'crc32_unittest.cc',
           'criticalsection_unittest.cc',
           'event_tracer_unittest.cc',
           'event_unittest.cc',
           'exp_filter_unittest.cc',
-          'filelock_unittest.cc',
+          'filerotatingstream_unittest.cc',
           'fileutils_unittest.cc',
           'helpers_unittest.cc',
           'httpbase_unittest.cc',
@@ -77,17 +77,19 @@
           'multipart_unittest.cc',
           'nat_unittest.cc',
           'network_unittest.cc',
-          'nullsocketserver_unittest.cc',
+          'optional_unittest.cc',
           'optionsfile_unittest.cc',
           'pathutils_unittest.cc',
-          'physicalsocketserver_unittest.cc',
+          'platform_thread_unittest.cc',
           'profiler_unittest.cc',
           'proxy_unittest.cc',
           'proxydetect_unittest.cc',
+          'random_unittest.cc',
           'ratelimiter_unittest.cc',
           'ratetracker_unittest.cc',
           'referencecountedsingletonfactory_unittest.cc',
           'rollingaccumulator_unittest.cc',
+          'rtccertificate_unittests.cc',
           'scopedptrcollection_unittest.cc',
           'sha1digest_unittest.cc',
           'sharedexclusivelock_unittest.cc',
@@ -95,9 +97,6 @@
           'sigslot_unittest.cc',
           'sigslottester.h',
           'sigslottester.h.pump',
-          'socket_unittest.cc',
-          'socket_unittest.h',
-          'socketaddress_unittest.cc',
           'stream_unittest.cc',
           'stringencode_unittest.cc',
           'stringutils_unittest.cc',
@@ -110,7 +109,6 @@
           'timeutils_unittest.cc',
           'urlencode_unittest.cc',
           'versionparsing_unittest.cc',
-          'virtualsocket_unittest.cc',
           # TODO(ronghuawu): Reenable this test.
           # 'windowpicker_unittest.cc',
         ],
@@ -127,24 +125,29 @@
             'sources': [
               'win32_unittest.cc',
               'win32regkey_unittest.cc',
-              'win32socketserver_unittest.cc',
-              'win32toolhelp_unittest.cc',
               'win32window_unittest.cc',
               'win32windowpicker_unittest.cc',
               'winfirewall_unittest.cc',
             ],
             'sources!': [
-              # TODO(ronghuawu): Fix TestUdpReadyToSendIPv6 on windows bot
-              # then reenable these tests.
-              'physicalsocketserver_unittest.cc',
-              'socket_unittest.cc',
-              'win32socketserver_unittest.cc',
+              # TODO(pbos): Reenable this test.
               'win32windowpicker_unittest.cc',
             ],
           }],
+          ['OS=="win" and clang==1', {
+            'msvs_settings': {
+              'VCCLCompilerTool': {
+                'AdditionalOptions': [
+                  # Disable warnings failing when compiling with Clang on Windows.
+                  # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                  '-Wno-missing-braces',
+                  '-Wno-unused-const-variable',
+                ],
+              },
+            },
+          }],
           ['OS=="mac"', {
             'sources': [
-              'macsocketserver_unittest.cc',
               'macutils_unittest.cc',
             ],
           }],
diff --git a/media/webrtc/trunk/webrtc/base/basictypes.h b/media/webrtc/trunk/webrtc/base/basictypes.h
index 6299061e14..4c3d5d1e51 100644
--- a/media/webrtc/trunk/webrtc/base/basictypes.h
+++ b/media/webrtc/trunk/webrtc/base/basictypes.h
@@ -12,97 +12,41 @@
 #define WEBRTC_BASE_BASICTYPES_H_
 
 #include <stddef.h>  // for NULL, size_t
-
-#if !(defined(_MSC_VER) && (_MSC_VER < 1600))
-#include <stdint.h>  // for uintptr_t
-#endif
+#include <stdint.h>  // for uintptr_t and (u)int_t types.
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"  // NOLINT
 #endif
 
-#include "webrtc/base/constructormagic.h"
-
-#if !defined(INT_TYPES_DEFINED)
-#define INT_TYPES_DEFINED
-#ifdef COMPILER_MSVC
-typedef unsigned __int64 uint64;
-typedef __int64 int64;
-#ifndef INT64_C
-#define INT64_C(x) x ## I64
-#endif
-#ifndef UINT64_C
-#define UINT64_C(x) x ## UI64
-#endif
-#define INT64_F "I64"
-#else  // COMPILER_MSVC
-// On Mac OS X, cssmconfig.h defines uint64 as uint64_t
-// TODO(fbarchard): Use long long for compatibility with chromium on BSD/OSX.
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-typedef uint64_t uint64;
-typedef int64_t int64;
-#ifndef INT64_C
-#define INT64_C(x) x ## LL
-#endif
-#ifndef UINT64_C
-#define UINT64_C(x) x ## ULL
-#endif
-#define INT64_F "l"
-#elif defined(__LP64__)
-typedef unsigned long uint64;  // NOLINT
-typedef long int64;  // NOLINT
-#ifndef INT64_C
-#define INT64_C(x) x ## L
-#endif
-#ifndef UINT64_C
-#define UINT64_C(x) x ## UL
-#endif
-#define INT64_F "l"
-#else  // __LP64__
-typedef unsigned long long uint64;  // NOLINT
-typedef long long int64;  // NOLINT
-#ifndef INT64_C
-#define INT64_C(x) x ## LL
-#endif
-#ifndef UINT64_C
-#define UINT64_C(x) x ## ULL
-#endif
-#define INT64_F "ll"
-#endif  // __LP64__
-#endif  // COMPILER_MSVC
-typedef unsigned int uint32;
-typedef int int32;
-typedef unsigned short uint16;  // NOLINT
-typedef short int16;  // NOLINT
-typedef unsigned char uint8;
-typedef signed char int8;
-#endif  // INT_TYPES_DEFINED
-
 // Detect compiler is for x86 or x64.
 #if defined(__x86_64__) || defined(_M_X64) || \
     defined(__i386__) || defined(_M_IX86)
 #define CPU_X86 1
 #endif
+
 // Detect compiler is for arm.
 #if defined(__arm__) || defined(_M_ARM)
 #define CPU_ARM 1
 #endif
+
 #if defined(CPU_X86) && defined(CPU_ARM)
 #error CPU_X86 and CPU_ARM both defined.
 #endif
-#if !defined(ARCH_CPU_BIG_ENDIAN) && !defined(ARCH_CPU_LITTLE_ENDIAN)
+
+#if !defined(RTC_ARCH_CPU_BIG_ENDIAN) && !defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
 // x86, arm or GCC provided __BYTE_ORDER__ macros
 #if CPU_X86 || CPU_ARM ||  \
   (defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
-#define ARCH_CPU_LITTLE_ENDIAN
+#define RTC_ARCH_CPU_LITTLE_ENDIAN
 #elif defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
-#define ARCH_CPU_BIG_ENDIAN
+#define RTC_ARCH_CPU_BIG_ENDIAN
 #else
-#error ARCH_CPU_BIG_ENDIAN or ARCH_CPU_LITTLE_ENDIAN should be defined.
+#error RTC_ARCH_CPU_BIG_ENDIAN or RTC_ARCH_CPU_LITTLE_ENDIAN should be defined.
 #endif
 #endif
-#if defined(ARCH_CPU_BIG_ENDIAN) && defined(ARCH_CPU_LITTLE_ENDIAN)
-#error ARCH_CPU_BIG_ENDIAN and ARCH_CPU_LITTLE_ENDIAN both defined.
+
+#if defined(RTC_ARCH_CPU_BIG_ENDIAN) && defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
+#error RTC_ARCH_CPU_BIG_ENDIAN and RTC_ARCH_CPU_LITTLE_ENDIAN both defined.
 #endif
 
 #if defined(WEBRTC_WIN)
@@ -111,15 +55,20 @@ typedef int socklen_t;
 
 // The following only works for C++
 #ifdef __cplusplus
-#define ALIGNP(p, t) \
-    (reinterpret_cast<uint8*>(((reinterpret_cast<uintptr_t>(p) + \
-    ((t) - 1)) & ~((t) - 1))))
+
+#ifndef ALIGNP
+#define ALIGNP(p, t)                                             \
+  (reinterpret_cast<uint8_t*>(((reinterpret_cast<uintptr_t>(p) + \
+  ((t) - 1)) & ~((t) - 1))))
+#endif
+
 #define RTC_IS_ALIGNED(p, a) (!((uintptr_t)(p) & ((a) - 1)))
 
-// Use these to declare and define a static local variable (static T;) so that
-// it is leaked so that its destructors are not called at exit.
-#define LIBJINGLE_DEFINE_STATIC_LOCAL(type, name, arguments) \
+// Use these to declare and define a static local variable that gets leaked so
+// that its destructors are not called at exit.
+#define RTC_DEFINE_STATIC_LOCAL(type, name, arguments) \
   static type& name = *new type arguments
 
 #endif  // __cplusplus
+
 #endif  // WEBRTC_BASE_BASICTYPES_H_
diff --git a/media/webrtc/trunk/webrtc/base/basictypes_unittest.cc b/media/webrtc/trunk/webrtc/base/basictypes_unittest.cc
index 20515ecf96..df5ed5e7e5 100644
--- a/media/webrtc/trunk/webrtc/base/basictypes_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/basictypes_unittest.cc
@@ -15,42 +15,15 @@
 namespace rtc {
 
 TEST(BasicTypesTest, Endian) {
-  uint16 v16 = 0x1234u;
-  uint8 first_byte = *reinterpret_cast<uint8*>(&v16);
-#if defined(ARCH_CPU_LITTLE_ENDIAN)
+  uint16_t v16 = 0x1234u;
+  uint8_t first_byte = *reinterpret_cast<uint8_t*>(&v16);
+#if defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
   EXPECT_EQ(0x34u, first_byte);
-#elif defined(ARCH_CPU_BIG_ENDIAN)
+#elif defined(RTC_ARCH_CPU_BIG_ENDIAN)
   EXPECT_EQ(0x12u, first_byte);
 #endif
 }
 
-TEST(BasicTypesTest, SizeOfTypes) {
-  int8 i8 = -1;
-  uint8 u8 = 1u;
-  int16 i16 = -1;
-  uint16 u16 = 1u;
-  int32 i32 = -1;
-  uint32 u32 = 1u;
-  int64 i64 = -1;
-  uint64 u64 = 1u;
-  EXPECT_EQ(1u, sizeof(i8));
-  EXPECT_EQ(1u, sizeof(u8));
-  EXPECT_EQ(2u, sizeof(i16));
-  EXPECT_EQ(2u, sizeof(u16));
-  EXPECT_EQ(4u, sizeof(i32));
-  EXPECT_EQ(4u, sizeof(u32));
-  EXPECT_EQ(8u, sizeof(i64));
-  EXPECT_EQ(8u, sizeof(u64));
-  EXPECT_GT(0, i8);
-  EXPECT_LT(0u, u8);
-  EXPECT_GT(0, i16);
-  EXPECT_LT(0u, u16);
-  EXPECT_GT(0, i32);
-  EXPECT_LT(0u, u32);
-  EXPECT_GT(0, i64);
-  EXPECT_LT(0u, u64);
-}
-
 TEST(BasicTypesTest, SizeOfConstants) {
   EXPECT_EQ(8u, sizeof(INT64_C(0)));
   EXPECT_EQ(8u, sizeof(UINT64_C(0)));
@@ -65,9 +38,9 @@ TEST(BasicTypesTest, SizeOfConstants) {
 #if !defined(CPU_X86) && (defined(WEBRTC_WIN) || defined(WEBRTC_MAC) && !defined(WEBRTC_IOS))
 #error expected CPU_X86 to be defined.
 #endif
-#if !defined(ARCH_CPU_LITTLE_ENDIAN) && \
+#if !defined(RTC_ARCH_CPU_LITTLE_ENDIAN) && \
   (defined(WEBRTC_WIN) || defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) || defined(CPU_X86))
-#error expected ARCH_CPU_LITTLE_ENDIAN to be defined.
+#error expected RTC_ARCH_CPU_LITTLE_ENDIAN to be defined.
 #endif
 
 // TODO(fbarchard): Test all macros in basictypes.h
diff --git a/media/webrtc/trunk/webrtc/base/bind.h b/media/webrtc/trunk/webrtc/base/bind.h
index 2e3104edfd..b50afc21ac 100644
--- a/media/webrtc/trunk/webrtc/base/bind.h
+++ b/media/webrtc/trunk/webrtc/base/bind.h
@@ -16,12 +16,13 @@
 // /home/build/google3/third_party/gtest/scripts/pump.py bind.h.pump
 
 // Bind() is an overloaded function that converts method calls into function
-// objects (aka functors). It captures any arguments to the method by value
-// when Bind is called, producing a stateful, nullary function object. Care
-// should be taken about the lifetime of objects captured by Bind(); the
-// returned functor knows nothing about the lifetime of the method's object or
-// any arguments passed by pointer, and calling the functor with a destroyed
-// object will surely do bad things.
+// objects (aka functors). The method object is captured as a scoped_refptr<> if
+// possible, and as a raw pointer otherwise. Any arguments to the method are
+// captured by value. The return value of Bind is a stateful, nullary function
+// object. Care should be taken about the lifetime of objects captured by
+// Bind(); the returned functor knows nothing about the lifetime of a non
+// ref-counted method object or any arguments passed by pointer, and calling the
+// functor with a destroyed object will surely do bad things.
 //
 // Example usage:
 //   struct Foo {
@@ -38,10 +39,34 @@
 //     cout << rtc::Bind(&Foo::Test3, &foo, 3)() << endl;
 //     cout << rtc::Bind(&Foo::Test4, &foo, 7, 8.5f)() << endl;
 //   }
+//
+// Example usage of ref counted objects:
+//   struct Bar {
+//     int AddRef();
+//     int Release();
+//
+//     void Test() {}
+//     void BindThis() {
+//       // The functor passed to AsyncInvoke() will keep this object alive.
+//       invoker.AsyncInvoke(rtc::Bind(&Bar::Test, this));
+//     }
+//   };
+//
+//   int main() {
+//     rtc::scoped_refptr<Bar> bar = new rtc::RefCountedObject<Bar>();
+//     auto functor = rtc::Bind(&Bar::Test, bar);
+//     bar = nullptr;
+//     // The functor stores an internal scoped_refptr<Bar>, so this is safe.
+//     functor();
+//   }
+//
 
 #ifndef WEBRTC_BASE_BIND_H_
 #define WEBRTC_BASE_BIND_H_
 
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/template_util.h"
+
 #define NONAME
 
 namespace rtc {
@@ -53,6 +78,57 @@ namespace detail {
 // references stripped. This trick allows the compiler to dictate the Bind
 // parameter types rather than deduce them.
 template <class T> struct identity { typedef T type; };
+
+// IsRefCounted<T>::value will be true for types that can be used in
+// rtc::scoped_refptr<T>, i.e. types that implements nullary functions AddRef()
+// and Release(), regardless of their return types. AddRef() and Release() can
+// be defined in T or any superclass of T.
+template <typename T>
+class IsRefCounted {
+  // This is a complex implementation detail done with SFINAE.
+
+  // Define types such that sizeof(Yes) != sizeof(No).
+  struct Yes { char dummy[1]; };
+  struct No { char dummy[2]; };
+  // Define two overloaded template functions with return types of different
+  // size. This way, we can use sizeof() on the return type to determine which
+  // function the compiler would have chosen. One function will be preferred
+  // over the other if it is possible to create it without compiler errors,
+  // otherwise the compiler will simply remove it, and default to the less
+  // preferred function.
+  template <typename R>
+  static Yes test(R* r, decltype(r->AddRef(), r->Release(), 42));
+  template <typename C> static No test(...);
+
+public:
+  // Trick the compiler to tell if it's possible to call AddRef() and Release().
+  static const bool value = sizeof(test<T>((T*)nullptr, 42)) == sizeof(Yes);
+};
+
+// TernaryTypeOperator is a helper class to select a type based on a static bool
+// value.
+template <bool condition, typename IfTrueT, typename IfFalseT>
+struct TernaryTypeOperator {};
+
+template <typename IfTrueT, typename IfFalseT>
+struct TernaryTypeOperator<true, IfTrueT, IfFalseT> {
+  typedef IfTrueT type;
+};
+
+template <typename IfTrueT, typename IfFalseT>
+struct TernaryTypeOperator<false, IfTrueT, IfFalseT> {
+  typedef IfFalseT type;
+};
+
+// PointerType<T>::type will be scoped_refptr<T> for ref counted types, and T*
+// otherwise.
+template <class T>
+struct PointerType {
+  typedef typename TernaryTypeOperator<IsRefCounted<T>::value,
+                                       scoped_refptr<T>,
+                                       T*>::type type;
+};
+
 }  // namespace detail
 
 template <class ObjectT, class MethodT, class R>
@@ -64,7 +140,7 @@ class MethodFunctor0 {
     return (object_->*method_)(); }
  private:
   MethodT method_;
-  ObjectT* object_;
+  typename detail::PointerType<ObjectT>::type object_;
 };
 
 template <class FunctorT, class R>
@@ -98,6 +174,16 @@ Bind(FP_T(method), const ObjectT* object) {
       method, object);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)()
+
+template <class ObjectT, class R>
+MethodFunctor0<ObjectT, FP_T(NONAME), R>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object) {
+  return MethodFunctor0<ObjectT, FP_T(NONAME), R>(
+      method, object.get());
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)()
 
@@ -122,8 +208,8 @@ class MethodFunctor1 {
     return (object_->*method_)(p1_); }
  private:
   MethodT method_;
-  ObjectT* object_;
-  P1 p1_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
 };
 
 template <class FunctorT, class R,
@@ -137,7 +223,7 @@ class Functor1 {
     return functor_(p1_); }
  private:
   FunctorT functor_;
-  P1 p1_;
+  typename rtc::remove_reference<P1>::type p1_;
 };
 
 
@@ -164,6 +250,18 @@ Bind(FP_T(method), const ObjectT* object,
       method, object, p1);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1)
+
+template <class ObjectT, class R,
+          class P1>
+MethodFunctor1<ObjectT, FP_T(NONAME), R, P1>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1) {
+  return MethodFunctor1<ObjectT, FP_T(NONAME), R, P1>(
+      method, object.get(), p1);
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)(P1)
 
@@ -193,9 +291,9 @@ class MethodFunctor2 {
     return (object_->*method_)(p1_, p2_); }
  private:
   MethodT method_;
-  ObjectT* object_;
-  P1 p1_;
-  P2 p2_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
 };
 
 template <class FunctorT, class R,
@@ -211,8 +309,8 @@ class Functor2 {
     return functor_(p1_, p2_); }
  private:
   FunctorT functor_;
-  P1 p1_;
-  P2 p2_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
 };
 
 
@@ -243,6 +341,20 @@ Bind(FP_T(method), const ObjectT* object,
       method, object, p1, p2);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2)
+
+template <class ObjectT, class R,
+          class P1,
+          class P2>
+MethodFunctor2<ObjectT, FP_T(NONAME), R, P1, P2>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2) {
+  return MethodFunctor2<ObjectT, FP_T(NONAME), R, P1, P2>(
+      method, object.get(), p1, p2);
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)(P1, P2)
 
@@ -277,10 +389,10 @@ class MethodFunctor3 {
     return (object_->*method_)(p1_, p2_, p3_); }
  private:
   MethodT method_;
-  ObjectT* object_;
-  P1 p1_;
-  P2 p2_;
-  P3 p3_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
 };
 
 template <class FunctorT, class R,
@@ -298,9 +410,9 @@ class Functor3 {
     return functor_(p1_, p2_, p3_); }
  private:
   FunctorT functor_;
-  P1 p1_;
-  P2 p2_;
-  P3 p3_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
 };
 
 
@@ -335,6 +447,22 @@ Bind(FP_T(method), const ObjectT* object,
       method, object, p1, p2, p3);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3)
+
+template <class ObjectT, class R,
+          class P1,
+          class P2,
+          class P3>
+MethodFunctor3<ObjectT, FP_T(NONAME), R, P1, P2, P3>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3) {
+  return MethodFunctor3<ObjectT, FP_T(NONAME), R, P1, P2, P3>(
+      method, object.get(), p1, p2, p3);
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)(P1, P2, P3)
 
@@ -374,11 +502,11 @@ class MethodFunctor4 {
     return (object_->*method_)(p1_, p2_, p3_, p4_); }
  private:
   MethodT method_;
-  ObjectT* object_;
-  P1 p1_;
-  P2 p2_;
-  P3 p3_;
-  P4 p4_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
 };
 
 template <class FunctorT, class R,
@@ -398,10 +526,10 @@ class Functor4 {
     return functor_(p1_, p2_, p3_, p4_); }
  private:
   FunctorT functor_;
-  P1 p1_;
-  P2 p2_;
-  P3 p3_;
-  P4 p4_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
 };
 
 
@@ -440,6 +568,24 @@ Bind(FP_T(method), const ObjectT* object,
       method, object, p1, p2, p3, p4);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4)
+
+template <class ObjectT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4>
+MethodFunctor4<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4) {
+  return MethodFunctor4<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4>(
+      method, object.get(), p1, p2, p3, p4);
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)(P1, P2, P3, P4)
 
@@ -484,12 +630,12 @@ class MethodFunctor5 {
     return (object_->*method_)(p1_, p2_, p3_, p4_, p5_); }
  private:
   MethodT method_;
-  ObjectT* object_;
-  P1 p1_;
-  P2 p2_;
-  P3 p3_;
-  P4 p4_;
-  P5 p5_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
 };
 
 template <class FunctorT, class R,
@@ -511,11 +657,11 @@ class Functor5 {
     return functor_(p1_, p2_, p3_, p4_, p5_); }
  private:
   FunctorT functor_;
-  P1 p1_;
-  P2 p2_;
-  P3 p3_;
-  P4 p4_;
-  P5 p5_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
 };
 
 
@@ -558,6 +704,26 @@ Bind(FP_T(method), const ObjectT* object,
       method, object, p1, p2, p3, p4, p5);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5)
+
+template <class ObjectT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5>
+MethodFunctor5<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5) {
+  return MethodFunctor5<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5>(
+      method, object.get(), p1, p2, p3, p4, p5);
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)(P1, P2, P3, P4, P5)
 
@@ -580,6 +746,795 @@ Bind(FP_T(function),
 
 #undef FP_T
 
+template <class ObjectT, class MethodT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6>
+class MethodFunctor6 {
+ public:
+  MethodFunctor6(MethodT method, ObjectT* object,
+                 P1 p1,
+                 P2 p2,
+                 P3 p3,
+                 P4 p4,
+                 P5 p5,
+                 P6 p6)
+      : method_(method), object_(object),
+      p1_(p1),
+      p2_(p2),
+      p3_(p3),
+      p4_(p4),
+      p5_(p5),
+      p6_(p6) {}
+  R operator()() const {
+    return (object_->*method_)(p1_, p2_, p3_, p4_, p5_, p6_); }
+ private:
+  MethodT method_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+};
+
+template <class FunctorT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6>
+class Functor6 {
+ public:
+  Functor6(const FunctorT& functor, P1 p1, P2 p2, P3 p3, P4 p4, P5 p5, P6 p6)
+      : functor_(functor),
+      p1_(p1),
+      p2_(p2),
+      p3_(p3),
+      p4_(p4),
+      p5_(p5),
+      p6_(p6) {}
+  R operator()() const {
+    return functor_(p1_, p2_, p3_, p4_, p5_, p6_); }
+ private:
+  FunctorT functor_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+};
+
+
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6)
+
+template <class ObjectT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6>
+MethodFunctor6<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>
+Bind(FP_T(method), ObjectT* object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6) {
+  return MethodFunctor6<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>(
+      method, object, p1, p2, p3, p4, p5, p6);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6) const
+
+template <class ObjectT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6>
+MethodFunctor6<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>
+Bind(FP_T(method), const ObjectT* object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6) {
+  return MethodFunctor6<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>(
+      method, object, p1, p2, p3, p4, p5, p6);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6)
+
+template <class ObjectT, class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6>
+MethodFunctor6<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6) {
+  return MethodFunctor6<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>(
+      method, object.get(), p1, p2, p3, p4, p5, p6);
+}
+
+#undef FP_T
+#define FP_T(x) R (*x)(P1, P2, P3, P4, P5, P6)
+
+template <class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6>
+Functor6<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>
+Bind(FP_T(function),
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6) {
+  return Functor6<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6>(
+      function, p1, p2, p3, p4, p5, p6);
+}
+
+#undef FP_T
+
+template <class ObjectT,
+          class MethodT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7>
+class MethodFunctor7 {
+ public:
+  MethodFunctor7(MethodT method,
+                 ObjectT* object,
+                 P1 p1,
+                 P2 p2,
+                 P3 p3,
+                 P4 p4,
+                 P5 p5,
+                 P6 p6,
+                 P7 p7)
+      : method_(method),
+        object_(object),
+        p1_(p1),
+        p2_(p2),
+        p3_(p3),
+        p4_(p4),
+        p5_(p5),
+        p6_(p6),
+        p7_(p7) {}
+  R operator()() const {
+    return (object_->*method_)(p1_, p2_, p3_, p4_, p5_, p6_, p7_);
+  }
+
+ private:
+  MethodT method_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+  typename rtc::remove_reference<P7>::type p7_;
+};
+
+template <class FunctorT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7>
+class Functor7 {
+ public:
+  Functor7(const FunctorT& functor,
+           P1 p1,
+           P2 p2,
+           P3 p3,
+           P4 p4,
+           P5 p5,
+           P6 p6,
+           P7 p7)
+      : functor_(functor),
+        p1_(p1),
+        p2_(p2),
+        p3_(p3),
+        p4_(p4),
+        p5_(p5),
+        p6_(p6),
+        p7_(p7) {}
+  R operator()() const { return functor_(p1_, p2_, p3_, p4_, p5_, p6_, p7_); }
+
+ private:
+  FunctorT functor_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+  typename rtc::remove_reference<P7>::type p7_;
+};
+
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7)
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7>
+MethodFunctor7<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7> Bind(
+    FP_T(method),
+    ObjectT* object,
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7) {
+  return MethodFunctor7<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7>(
+      method, object, p1, p2, p3, p4, p5, p6, p7);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7) const
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7>
+MethodFunctor7<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7> Bind(
+    FP_T(method),
+    const ObjectT* object,
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7) {
+  return MethodFunctor7<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6,
+                        P7>(method, object, p1, p2, p3, p4, p5, p6, p7);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7)
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7>
+MethodFunctor7<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7> Bind(
+    FP_T(method),
+    const scoped_refptr<ObjectT>& object,
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7) {
+  return MethodFunctor7<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7>(
+      method, object.get(), p1, p2, p3, p4, p5, p6, p7);
+}
+
+#undef FP_T
+#define FP_T(x) R (*x)(P1, P2, P3, P4, P5, P6, P7)
+
+template <class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7>
+Functor7<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7> Bind(
+    FP_T(function),
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7) {
+  return Functor7<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7>(
+      function, p1, p2, p3, p4, p5, p6, p7);
+}
+
+#undef FP_T
+
+template <class ObjectT,
+          class MethodT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8>
+class MethodFunctor8 {
+ public:
+  MethodFunctor8(MethodT method,
+                 ObjectT* object,
+                 P1 p1,
+                 P2 p2,
+                 P3 p3,
+                 P4 p4,
+                 P5 p5,
+                 P6 p6,
+                 P7 p7,
+                 P8 p8)
+      : method_(method),
+        object_(object),
+        p1_(p1),
+        p2_(p2),
+        p3_(p3),
+        p4_(p4),
+        p5_(p5),
+        p6_(p6),
+        p7_(p7),
+        p8_(p8) {}
+  R operator()() const {
+    return (object_->*method_)(p1_, p2_, p3_, p4_, p5_, p6_, p7_, p8_);
+  }
+
+ private:
+  MethodT method_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+  typename rtc::remove_reference<P7>::type p7_;
+  typename rtc::remove_reference<P8>::type p8_;
+};
+
+template <class FunctorT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8>
+class Functor8 {
+ public:
+  Functor8(const FunctorT& functor,
+           P1 p1,
+           P2 p2,
+           P3 p3,
+           P4 p4,
+           P5 p5,
+           P6 p6,
+           P7 p7,
+           P8 p8)
+      : functor_(functor),
+        p1_(p1),
+        p2_(p2),
+        p3_(p3),
+        p4_(p4),
+        p5_(p5),
+        p6_(p6),
+        p7_(p7),
+        p8_(p8) {}
+  R operator()() const {
+    return functor_(p1_, p2_, p3_, p4_, p5_, p6_, p7_, p8_);
+  }
+
+ private:
+  FunctorT functor_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+  typename rtc::remove_reference<P7>::type p7_;
+  typename rtc::remove_reference<P8>::type p8_;
+};
+
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7, P8)
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8>
+MethodFunctor8<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8> Bind(
+    FP_T(method),
+    ObjectT* object,
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7,
+    typename detail::identity<P8>::type p8) {
+  return MethodFunctor8<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7,
+                        P8>(method, object, p1, p2, p3, p4, p5, p6, p7, p8);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7, P8) const
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8>
+MethodFunctor8<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8>
+Bind(FP_T(method),
+     const ObjectT* object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6,
+     typename detail::identity<P7>::type p7,
+     typename detail::identity<P8>::type p8) {
+  return MethodFunctor8<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6,
+                        P7, P8>(method, object, p1, p2, p3, p4, p5, p6, p7, p8);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7, P8)
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8>
+MethodFunctor8<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8> Bind(
+    FP_T(method),
+    const scoped_refptr<ObjectT>& object,
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7,
+    typename detail::identity<P8>::type p8) {
+  return MethodFunctor8<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7,
+                        P8>(method, object.get(), p1, p2, p3, p4, p5, p6, p7,
+                            p8);
+}
+
+#undef FP_T
+#define FP_T(x) R (*x)(P1, P2, P3, P4, P5, P6, P7, P8)
+
+template <class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8>
+Functor8<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8> Bind(
+    FP_T(function),
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7,
+    typename detail::identity<P8>::type p8) {
+  return Functor8<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8>(
+      function, p1, p2, p3, p4, p5, p6, p7, p8);
+}
+
+#undef FP_T
+
+template <class ObjectT,
+          class MethodT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8,
+          class P9>
+class MethodFunctor9 {
+ public:
+  MethodFunctor9(MethodT method,
+                 ObjectT* object,
+                 P1 p1,
+                 P2 p2,
+                 P3 p3,
+                 P4 p4,
+                 P5 p5,
+                 P6 p6,
+                 P7 p7,
+                 P8 p8,
+                 P9 p9)
+      : method_(method),
+        object_(object),
+        p1_(p1),
+        p2_(p2),
+        p3_(p3),
+        p4_(p4),
+        p5_(p5),
+        p6_(p6),
+        p7_(p7),
+        p8_(p8),
+        p9_(p9) {}
+  R operator()() const {
+    return (object_->*method_)(p1_, p2_, p3_, p4_, p5_, p6_, p7_, p8_, p9_);
+  }
+
+ private:
+  MethodT method_;
+  typename detail::PointerType<ObjectT>::type object_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+  typename rtc::remove_reference<P7>::type p7_;
+  typename rtc::remove_reference<P8>::type p8_;
+  typename rtc::remove_reference<P9>::type p9_;
+};
+
+template <class FunctorT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8,
+          class P9>
+class Functor9 {
+ public:
+  Functor9(const FunctorT& functor,
+           P1 p1,
+           P2 p2,
+           P3 p3,
+           P4 p4,
+           P5 p5,
+           P6 p6,
+           P7 p7,
+           P8 p8,
+           P9 p9)
+      : functor_(functor),
+        p1_(p1),
+        p2_(p2),
+        p3_(p3),
+        p4_(p4),
+        p5_(p5),
+        p6_(p6),
+        p7_(p7),
+        p8_(p8),
+        p9_(p9) {}
+  R operator()() const {
+    return functor_(p1_, p2_, p3_, p4_, p5_, p6_, p7_, p8_, p9_);
+  }
+
+ private:
+  FunctorT functor_;
+  typename rtc::remove_reference<P1>::type p1_;
+  typename rtc::remove_reference<P2>::type p2_;
+  typename rtc::remove_reference<P3>::type p3_;
+  typename rtc::remove_reference<P4>::type p4_;
+  typename rtc::remove_reference<P5>::type p5_;
+  typename rtc::remove_reference<P6>::type p6_;
+  typename rtc::remove_reference<P7>::type p7_;
+  typename rtc::remove_reference<P8>::type p8_;
+  typename rtc::remove_reference<P9>::type p9_;
+};
+
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7, P8, P9)
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8,
+          class P9>
+MethodFunctor9<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8, P9>
+Bind(FP_T(method),
+     ObjectT* object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6,
+     typename detail::identity<P7>::type p7,
+     typename detail::identity<P8>::type p8,
+     typename detail::identity<P9>::type p9) {
+  return MethodFunctor9<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7,
+                        P8, P9>(method, object, p1, p2, p3, p4, p5, p6, p7, p8,
+                                p9);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7, P8, P9) const
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8,
+          class P9>
+MethodFunctor9<const ObjectT,
+               FP_T(NONAME),
+               R,
+               P1,
+               P2,
+               P3,
+               P4,
+               P5,
+               P6,
+               P7,
+               P8,
+               P9>
+Bind(FP_T(method),
+     const ObjectT* object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6,
+     typename detail::identity<P7>::type p7,
+     typename detail::identity<P8>::type p8,
+     typename detail::identity<P9>::type p9) {
+  return MethodFunctor9<const ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6,
+                        P7, P8, P9>(method, object, p1, p2, p3, p4, p5, p6, p7,
+                                    p8, p9);
+}
+
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)(P1, P2, P3, P4, P5, P6, P7, P8, P9)
+
+template <class ObjectT,
+          class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8,
+          class P9>
+MethodFunctor9<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8, P9>
+Bind(FP_T(method),
+     const scoped_refptr<ObjectT>& object,
+     typename detail::identity<P1>::type p1,
+     typename detail::identity<P2>::type p2,
+     typename detail::identity<P3>::type p3,
+     typename detail::identity<P4>::type p4,
+     typename detail::identity<P5>::type p5,
+     typename detail::identity<P6>::type p6,
+     typename detail::identity<P7>::type p7,
+     typename detail::identity<P8>::type p8,
+     typename detail::identity<P9>::type p9) {
+  return MethodFunctor9<ObjectT, FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7,
+                        P8, P9>(method, object.get(), p1, p2, p3, p4, p5, p6,
+                                p7, p8, p9);
+}
+
+#undef FP_T
+#define FP_T(x) R (*x)(P1, P2, P3, P4, P5, P6, P7, P8, P9)
+
+template <class R,
+          class P1,
+          class P2,
+          class P3,
+          class P4,
+          class P5,
+          class P6,
+          class P7,
+          class P8,
+          class P9>
+Functor9<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8, P9> Bind(
+    FP_T(function),
+    typename detail::identity<P1>::type p1,
+    typename detail::identity<P2>::type p2,
+    typename detail::identity<P3>::type p3,
+    typename detail::identity<P4>::type p4,
+    typename detail::identity<P5>::type p5,
+    typename detail::identity<P6>::type p6,
+    typename detail::identity<P7>::type p7,
+    typename detail::identity<P8>::type p8,
+    typename detail::identity<P9>::type p9) {
+  return Functor9<FP_T(NONAME), R, P1, P2, P3, P4, P5, P6, P7, P8, P9>(
+      function, p1, p2, p3, p4, p5, p6, p7, p8, p9);
+}
+
+#undef FP_T
+
 }  // namespace rtc
 
 #undef NONAME
diff --git a/media/webrtc/trunk/webrtc/base/bind.h.pump b/media/webrtc/trunk/webrtc/base/bind.h.pump
index b5663c45df..6fb849095d 100644
--- a/media/webrtc/trunk/webrtc/base/bind.h.pump
+++ b/media/webrtc/trunk/webrtc/base/bind.h.pump
@@ -12,12 +12,13 @@
 // /home/build/google3/third_party/gtest/scripts/pump.py bind.h.pump
 
 // Bind() is an overloaded function that converts method calls into function
-// objects (aka functors). It captures any arguments to the method by value
-// when Bind is called, producing a stateful, nullary function object. Care
-// should be taken about the lifetime of objects captured by Bind(); the
-// returned functor knows nothing about the lifetime of the method's object or
-// any arguments passed by pointer, and calling the functor with a destroyed
-// object will surely do bad things.
+// objects (aka functors). The method object is captured as a scoped_refptr<> if
+// possible, and as a raw pointer otherwise. Any arguments to the method are
+// captured by value. The return value of Bind is a stateful, nullary function
+// object. Care should be taken about the lifetime of objects captured by
+// Bind(); the returned functor knows nothing about the lifetime of a non
+// ref-counted method object or any arguments passed by pointer, and calling the
+// functor with a destroyed object will surely do bad things.
 //
 // Example usage:
 //   struct Foo {
@@ -34,10 +35,34 @@
 //     cout << rtc::Bind(&Foo::Test3, &foo, 3)() << endl;
 //     cout << rtc::Bind(&Foo::Test4, &foo, 7, 8.5f)() << endl;
 //   }
+//
+// Example usage of ref counted objects:
+//   struct Bar {
+//     int AddRef();
+//     int Release();
+//
+//     void Test() {}
+//     void BindThis() {
+//       // The functor passed to AsyncInvoke() will keep this object alive.
+//       invoker.AsyncInvoke(rtc::Bind(&Bar::Test, this));
+//     }
+//   };
+//
+//   int main() {
+//     rtc::scoped_refptr<Bar> bar = new rtc::RefCountedObject<Bar>();
+//     auto functor = rtc::Bind(&Bar::Test, bar);
+//     bar = nullptr;
+//     // The functor stores an internal scoped_refptr<Bar>, so this is safe.
+//     functor();
+//   }
+//
 
 #ifndef WEBRTC_BASE_BIND_H_
 #define WEBRTC_BASE_BIND_H_
 
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/template_util.h"
+
 #define NONAME
 
 namespace rtc {
@@ -49,9 +74,60 @@ namespace detail {
 // references stripped. This trick allows the compiler to dictate the Bind
 // parameter types rather than deduce them.
 template <class T> struct identity { typedef T type; };
+
+// IsRefCounted<T>::value will be true for types that can be used in
+// rtc::scoped_refptr<T>, i.e. types that implements nullary functions AddRef()
+// and Release(), regardless of their return types. AddRef() and Release() can
+// be defined in T or any superclass of T.
+template <typename T>
+class IsRefCounted {
+  // This is a complex implementation detail done with SFINAE.
+
+  // Define types such that sizeof(Yes) != sizeof(No).
+  struct Yes { char dummy[1]; };
+  struct No { char dummy[2]; };
+  // Define two overloaded template functions with return types of different
+  // size. This way, we can use sizeof() on the return type to determine which
+  // function the compiler would have chosen. One function will be preferred
+  // over the other if it is possible to create it without compiler errors,
+  // otherwise the compiler will simply remove it, and default to the less
+  // preferred function.
+  template <typename R>
+  static Yes test(R* r, decltype(r->AddRef(), r->Release(), 42));
+  template <typename C> static No test(...);
+
+public:
+  // Trick the compiler to tell if it's possible to call AddRef() and Release().
+  static const bool value = sizeof(test<T>((T*)nullptr, 42)) == sizeof(Yes);
+};
+
+// TernaryTypeOperator is a helper class to select a type based on a static bool
+// value.
+template <bool condition, typename IfTrueT, typename IfFalseT>
+struct TernaryTypeOperator {};
+
+template <typename IfTrueT, typename IfFalseT>
+struct TernaryTypeOperator<true, IfTrueT, IfFalseT> {
+  typedef IfTrueT type;
+};
+
+template <typename IfTrueT, typename IfFalseT>
+struct TernaryTypeOperator<false, IfTrueT, IfFalseT> {
+  typedef IfFalseT type;
+};
+
+// PointerType<T>::type will be scoped_refptr<T> for ref counted types, and T*
+// otherwise.
+template <class T>
+struct PointerType {
+  typedef typename TernaryTypeOperator<IsRefCounted<T>::value,
+                                       scoped_refptr<T>,
+                                       T*>::type type;
+};
+
 }  // namespace detail
 
-$var n = 5
+$var n = 9
 $range i 0..n
 $for i [[
 $range j 1..i
@@ -68,9 +144,9 @@ class MethodFunctor$i {
     return (object_->*method_)($for j , [[p$(j)_]]); }
  private:
   MethodT method_;
-  ObjectT* object_;$for j [[
+  typename detail::PointerType<ObjectT>::type object_;$for j [[
 
-  P$j p$(j)_;]]
+  typename rtc::remove_reference<P$j>::type p$(j)_;]]
 
 };
 
@@ -87,7 +163,7 @@ Functor$i(const FunctorT& functor$for j [[, P$j p$j]])
  private:
   FunctorT functor_;$for j [[
 
-  P$j p$(j)_;]]
+  typename rtc::remove_reference<P$j>::type p$(j)_;]]
 
 };
 
@@ -115,6 +191,18 @@ Bind(FP_T(method), const ObjectT* object$for j [[,
       method, object$for j [[, p$j]]);
 }
 
+#undef FP_T
+#define FP_T(x) R (ObjectT::*x)($for j , [[P$j]])
+
+template <class ObjectT, class R$for j [[,
+          class P$j]]>
+MethodFunctor$i<ObjectT, FP_T(NONAME), R$for j [[, P$j]]>
+Bind(FP_T(method), const scoped_refptr<ObjectT>& object$for j [[,
+     typename detail::identity<P$j>::type p$j]]) {
+  return MethodFunctor$i<ObjectT, FP_T(NONAME), R$for j [[, P$j]]>(
+      method, object.get()$for j [[, p$j]]);
+}
+
 #undef FP_T
 #define FP_T(x) R (*x)($for j , [[P$j]])
 
diff --git a/media/webrtc/trunk/webrtc/base/bind_unittest.cc b/media/webrtc/trunk/webrtc/base/bind_unittest.cc
index ed8dd5cf2d..be8d79cb6a 100644
--- a/media/webrtc/trunk/webrtc/base/bind_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/bind_unittest.cc
@@ -11,27 +11,111 @@
 #include "webrtc/base/bind.h"
 #include "webrtc/base/gunit.h"
 
+#include "webrtc/base/refcount.h"
+
 namespace rtc {
 
 namespace {
 
+struct LifeTimeCheck;
+
 struct MethodBindTester {
   void NullaryVoid() { ++call_count; }
   int NullaryInt() { ++call_count; return 1; }
   int NullaryConst() const { ++call_count; return 2; }
   void UnaryVoid(int dummy) { ++call_count; }
   template <class T> T Identity(T value) { ++call_count; return value; }
-  int UnaryByRef(int& value) const { ++call_count; return ++value; }  // NOLINT
+  int UnaryByPointer(int* value) const {
+    ++call_count;
+    return ++(*value);
+  }
+  int UnaryByRef(const int& value) const {
+    ++call_count;
+    return ++const_cast<int&>(value);
+  }
   int Multiply(int a, int b) const { ++call_count; return a * b; }
+  void RefArgument(const scoped_refptr<LifeTimeCheck>& object) {
+    EXPECT_TRUE(object.get() != nullptr);
+  }
+
   mutable int call_count;
 };
 
+struct A { int dummy; };
+struct B: public RefCountInterface { int dummy; };
+struct C: public A, B {};
+struct D {
+  int AddRef();
+};
+struct E: public D {
+  int Release();
+};
+struct F {
+  void AddRef();
+  void Release();
+};
+
+struct LifeTimeCheck {
+  LifeTimeCheck() : ref_count_(0) {}
+  void AddRef() { ++ref_count_; }
+  void Release() { --ref_count_; }
+  void NullaryVoid() {}
+  int ref_count_;
+};
+
 int Return42() { return 42; }
 int Negate(int a) { return -a; }
 int Multiply(int a, int b) { return a * b; }
 
 }  // namespace
 
+// Try to catch any problem with scoped_refptr type deduction in rtc::Bind at
+// compile time.
+static_assert(
+    is_same<
+        rtc::remove_reference<const scoped_refptr<RefCountInterface>&>::type,
+        const scoped_refptr<RefCountInterface>>::value,
+    "const scoped_refptr& should be captured by value");
+
+static_assert(is_same<rtc::remove_reference<const scoped_refptr<F>&>::type,
+                      const scoped_refptr<F>>::value,
+              "const scoped_refptr& should be captured by value");
+
+static_assert(
+    is_same<rtc::remove_reference<const int&>::type, const int>::value,
+    "const int& should be captured as const int");
+
+static_assert(is_same<rtc::remove_reference<const F&>::type, const F>::value,
+              "const F& should be captured as const F");
+
+static_assert(is_same<rtc::remove_reference<F&>::type, F>::value,
+              "F& should be captured as F");
+
+#define EXPECT_IS_CAPTURED_AS_PTR(T)                              \
+  static_assert(is_same<detail::PointerType<T>::type, T*>::value, \
+                "PointerType")
+#define EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(T)                        \
+  static_assert(                                                      \
+      is_same<detail::PointerType<T>::type, scoped_refptr<T>>::value, \
+      "PointerType")
+
+EXPECT_IS_CAPTURED_AS_PTR(void);
+EXPECT_IS_CAPTURED_AS_PTR(int);
+EXPECT_IS_CAPTURED_AS_PTR(double);
+EXPECT_IS_CAPTURED_AS_PTR(A);
+EXPECT_IS_CAPTURED_AS_PTR(D);
+EXPECT_IS_CAPTURED_AS_PTR(RefCountInterface*);
+
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(RefCountInterface);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(B);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(C);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(E);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(F);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(RefCountedObject<RefCountInterface>);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(RefCountedObject<B>);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(RefCountedObject<C>);
+EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(const RefCountedObject<RefCountInterface>);
+
 TEST(BindTest, BindToMethod) {
   MethodBindTester object = {0};
   EXPECT_EQ(0, object.call_count);
@@ -51,11 +135,20 @@ TEST(BindTest, BindToMethod) {
                                &object, string_value)());
   EXPECT_EQ(6, object.call_count);
   int value = 11;
-  EXPECT_EQ(12, Bind(&MethodBindTester::UnaryByRef, &object, value)());
+  // Bind binds by value, even if the method signature is by reference, so
+  // "reference" binds require pointers.
+  EXPECT_EQ(12, Bind(&MethodBindTester::UnaryByPointer, &object, &value)());
   EXPECT_EQ(12, value);
   EXPECT_EQ(7, object.call_count);
-  EXPECT_EQ(56, Bind(&MethodBindTester::Multiply, &object, 7, 8)());
+  // It's possible to bind to a function that takes a const reference, though
+  // the capture will be a copy. See UnaryByRef hackery above where it removes
+  // the const to make sure the underlying storage is, in fact, a copy.
+  EXPECT_EQ(13, Bind(&MethodBindTester::UnaryByRef, &object, value)());
+  // But the original value is unmodified.
+  EXPECT_EQ(12, value);
   EXPECT_EQ(8, object.call_count);
+  EXPECT_EQ(56, Bind(&MethodBindTester::Multiply, &object, 7, 8)());
+  EXPECT_EQ(9, object.call_count);
 }
 
 TEST(BindTest, BindToFunction) {
@@ -64,4 +157,82 @@ TEST(BindTest, BindToFunction) {
   EXPECT_EQ(56, Bind(&Multiply, 8, 7)());
 }
 
+// Test Bind where method object implements RefCountInterface and is passed as a
+// pointer.
+TEST(BindTest, CapturePointerAsScopedRefPtr) {
+  LifeTimeCheck object;
+  EXPECT_EQ(object.ref_count_, 0);
+  scoped_refptr<LifeTimeCheck> scoped_object(&object);
+  EXPECT_EQ(object.ref_count_, 1);
+  {
+    auto functor = Bind(&LifeTimeCheck::NullaryVoid, &object);
+    EXPECT_EQ(object.ref_count_, 2);
+    scoped_object = nullptr;
+    EXPECT_EQ(object.ref_count_, 1);
+  }
+  EXPECT_EQ(object.ref_count_, 0);
+}
+
+// Test Bind where method object implements RefCountInterface and is passed as a
+// scoped_refptr<>.
+TEST(BindTest, CaptureScopedRefPtrAsScopedRefPtr) {
+  LifeTimeCheck object;
+  EXPECT_EQ(object.ref_count_, 0);
+  scoped_refptr<LifeTimeCheck> scoped_object(&object);
+  EXPECT_EQ(object.ref_count_, 1);
+  {
+    auto functor = Bind(&LifeTimeCheck::NullaryVoid, scoped_object);
+    EXPECT_EQ(object.ref_count_, 2);
+    scoped_object = nullptr;
+    EXPECT_EQ(object.ref_count_, 1);
+  }
+  EXPECT_EQ(object.ref_count_, 0);
+}
+
+// Test Bind where method object is captured as scoped_refptr<> and the functor
+// dies while there are references left.
+TEST(BindTest, FunctorReleasesObjectOnDestruction) {
+  LifeTimeCheck object;
+  EXPECT_EQ(object.ref_count_, 0);
+  scoped_refptr<LifeTimeCheck> scoped_object(&object);
+  EXPECT_EQ(object.ref_count_, 1);
+  Bind(&LifeTimeCheck::NullaryVoid, &object)();
+  EXPECT_EQ(object.ref_count_, 1);
+  scoped_object = nullptr;
+  EXPECT_EQ(object.ref_count_, 0);
+}
+
+// Test Bind with scoped_refptr<> argument.
+TEST(BindTest, ScopedRefPointerArgument) {
+  LifeTimeCheck object;
+  EXPECT_EQ(object.ref_count_, 0);
+  scoped_refptr<LifeTimeCheck> scoped_object(&object);
+  EXPECT_EQ(object.ref_count_, 1);
+  {
+    MethodBindTester bind_tester;
+    auto functor =
+        Bind(&MethodBindTester::RefArgument, &bind_tester, scoped_object);
+    EXPECT_EQ(object.ref_count_, 2);
+  }
+  EXPECT_EQ(object.ref_count_, 1);
+  scoped_object = nullptr;
+  EXPECT_EQ(object.ref_count_, 0);
+}
+
+namespace {
+
+const int* Ref(const int& a) { return &a; }
+
+}  // anonymous namespace
+
+// Test Bind with non-scoped_refptr<> reference argument, which should be
+// modified to a non-reference capture.
+TEST(BindTest, RefArgument) {
+  const int x = 42;
+  EXPECT_EQ(&x, Ref(x));
+  // Bind() should make a copy of |x|, i.e. the pointers should be different.
+  auto functor = Bind(&Ref, x);
+  EXPECT_NE(&x, functor());
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/buffer.cc b/media/webrtc/trunk/webrtc/base/buffer.cc
index 227a3b25b8..62855f1620 100644
--- a/media/webrtc/trunk/webrtc/base/buffer.cc
+++ b/media/webrtc/trunk/webrtc/base/buffer.cc
@@ -10,28 +10,37 @@
 
 #include "webrtc/base/buffer.h"
 
+#include <cassert>
+#include <utility>
+
 namespace rtc {
 
-Buffer::Buffer() {
-  Construct(NULL, 0, 0);
+Buffer::Buffer() : size_(0), capacity_(0), data_(nullptr) {
+  assert(IsConsistent());
 }
 
-Buffer::Buffer(size_t size) : Buffer() {
-  SetSize(size);
+Buffer::Buffer(const Buffer& buf) : Buffer(buf.data(), buf.size()) {
 }
 
-Buffer::Buffer(const void* data, size_t size) {
-  Construct(data, size, size);
+Buffer::Buffer(Buffer&& buf)
+    : size_(buf.size()),
+      capacity_(buf.capacity()),
+      data_(std::move(buf.data_)) {
+  assert(IsConsistent());
+  buf.OnMovedFrom();
 }
 
-Buffer::Buffer(const void* data, size_t size, size_t capacity) {
-  Construct(data, size, capacity);
+Buffer::Buffer(size_t size) : Buffer(size, size) {
 }
 
-Buffer::Buffer(const Buffer& buf) {
-  Construct(buf.data(), buf.size(), buf.size());
+Buffer::Buffer(size_t size, size_t capacity)
+    : size_(size),
+      capacity_(std::max(size, capacity)),
+      data_(new uint8_t[capacity_]) {
+  assert(IsConsistent());
 }
 
+// Note: The destructor works even if the buffer has been moved from.
 Buffer::~Buffer() = default;
 
 };  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/buffer.h b/media/webrtc/trunk/webrtc/base/buffer.h
index fead5048a1..ff9bb73d3f 100644
--- a/media/webrtc/trunk/webrtc/base/buffer.h
+++ b/media/webrtc/trunk/webrtc/base/buffer.h
@@ -11,88 +11,218 @@
 #ifndef WEBRTC_BASE_BUFFER_H_
 #define WEBRTC_BASE_BUFFER_H_
 
-#include <string.h>
+#include <algorithm>  // std::swap (pre-C++11)
+#include <cassert>
+#include <cstring>
+#include <utility>  // std::swap (C++11 and later)
 
-// common.h isn't in the rtc_approved list
-//#include "webrtc/base/common.h"
+#include "webrtc/base/deprecation.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
+namespace internal {
+
+// (Internal; please don't use outside this file.) ByteType<T>::t is int if T
+// is uint8_t, int8_t, or char; otherwise, it's a compilation error. Use like
+// this:
+//
+//   template <typename T, typename ByteType<T>::t = 0>
+//   void foo(T* x);
+//
+// to let foo<T> be defined only for byte-sized integers.
+template <typename T>
+struct ByteType {
+ private:
+  static int F(uint8_t*);
+  static int F(int8_t*);
+  static int F(char*);
+
+ public:
+  using t = decltype(F(static_cast<T*>(nullptr)));
+};
+
+}  // namespace internal
+
 // Basic buffer class, can be grown and shrunk dynamically.
 // Unlike std::string/vector, does not initialize data when expanding capacity.
 class Buffer {
  public:
-  Buffer();
+  Buffer();                   // An empty buffer.
+  Buffer(const Buffer& buf);  // Copy size and contents of an existing buffer.
+  Buffer(Buffer&& buf);       // Move contents from an existing buffer.
+
+  // Construct a buffer with the specified number of uninitialized bytes.
   explicit Buffer(size_t size);
-  Buffer(const void* data, size_t size);
-  Buffer(const void* data, size_t size, size_t capacity);
-  Buffer(const Buffer& buf);
+  Buffer(size_t size, size_t capacity);
+
+  // Construct a buffer and copy the specified number of bytes into it. The
+  // source array may be (const) uint8_t*, int8_t*, or char*.
+  template <typename T, typename internal::ByteType<T>::t = 0>
+  Buffer(const T* data, size_t size)
+      : Buffer(data, size, size) {}
+  template <typename T, typename internal::ByteType<T>::t = 0>
+  Buffer(const T* data, size_t size, size_t capacity)
+      : Buffer(size, capacity) {
+    std::memcpy(data_.get(), data, size);
+  }
+
+  // Construct a buffer from the contents of an array.
+  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
+  Buffer(const T(&array)[N])
+      : Buffer(array, N) {}
+
   ~Buffer();
 
-  const char* data() const { return data_.get(); }
-  char* data() { return data_.get(); }
-  size_t size() const { return size_; }
-  size_t capacity() const { return capacity_; }
+  // Get a pointer to the data. Just .data() will give you a (const) uint8_t*,
+  // but you may also use .data<int8_t>() and .data<char>().
+  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
+  const T* data() const {
+    assert(IsConsistent());
+    return reinterpret_cast<T*>(data_.get());
+  }
+  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
+  T* data() {
+    assert(IsConsistent());
+    return reinterpret_cast<T*>(data_.get());
+  }
 
-  // For backwards compatibility. TODO(kwiberg): Remove once Chromium doesn't
-  // need it anymore.
-  size_t length() const { return size(); }
+  size_t size() const {
+    assert(IsConsistent());
+    return size_;
+  }
+  size_t capacity() const {
+    assert(IsConsistent());
+    return capacity_;
+  }
 
   Buffer& operator=(const Buffer& buf) {
-    if (&buf != this) {
-      Construct(buf.data(), buf.size(), buf.size());
-    }
+    if (&buf != this)
+      SetData(buf.data(), buf.size());
     return *this;
   }
-  bool operator==(const Buffer& buf) const {
-    return (size_ == buf.size() && memcmp(data_.get(), buf.data(), size_) == 0);
-  }
-  bool operator!=(const Buffer& buf) const {
-    return !operator==(buf);
+  Buffer& operator=(Buffer&& buf) {
+    assert(IsConsistent());
+    assert(buf.IsConsistent());
+    size_ = buf.size_;
+    capacity_ = buf.capacity_;
+    data_ = std::move(buf.data_);
+    buf.OnMovedFrom();
+    return *this;
   }
 
-  void SetData(const void* data, size_t size) {
-    assert(data != NULL || size == 0);
-    SetSize(size);
-    memcpy(data_.get(), data, size);
+  bool operator==(const Buffer& buf) const {
+    assert(IsConsistent());
+    return size_ == buf.size() && memcmp(data_.get(), buf.data(), size_) == 0;
   }
-  void AppendData(const void* data, size_t size) {
-    assert(data != NULL || size == 0);
-    size_t old_size = size_;
-    SetSize(size_ + size);
-    memcpy(data_.get() + old_size, data, size);
+
+  bool operator!=(const Buffer& buf) const { return !(*this == buf); }
+
+  // Replace the contents of the buffer. Accepts the same types as the
+  // constructors.
+  template <typename T, typename internal::ByteType<T>::t = 0>
+  void SetData(const T* data, size_t size) {
+    assert(IsConsistent());
+    size_ = 0;
+    AppendData(data, size);
   }
+  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
+  void SetData(const T(&array)[N]) {
+    SetData(array, N);
+  }
+  void SetData(const Buffer& buf) { SetData(buf.data(), buf.size()); }
+
+  // Append data to the buffer. Accepts the same types as the constructors.
+  template <typename T, typename internal::ByteType<T>::t = 0>
+  void AppendData(const T* data, size_t size) {
+    assert(IsConsistent());
+    const size_t new_size = size_ + size;
+    EnsureCapacity(new_size);
+    std::memcpy(data_.get() + size_, data, size);
+    size_ = new_size;
+    assert(IsConsistent());
+  }
+  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
+  void AppendData(const T(&array)[N]) {
+    AppendData(array, N);
+  }
+  void AppendData(const Buffer& buf) { AppendData(buf.data(), buf.size()); }
+
+  // Sets the size of the buffer. If the new size is smaller than the old, the
+  // buffer contents will be kept but truncated; if the new size is greater,
+  // the existing contents will be kept and the new space will be
+  // uninitialized.
   void SetSize(size_t size) {
-    SetCapacity(size);
+    EnsureCapacity(size);
     size_ = size;
   }
-  void SetCapacity(size_t capacity) {
-    if (capacity > capacity_) {
-      rtc::scoped_ptr<char[]> data(new char[capacity]);
-      memcpy(data.get(), data_.get(), size_);
-      data_.swap(data);
-      capacity_ = capacity;
-    }
+
+  // Ensure that the buffer size can be increased to at least capacity without
+  // further reallocation. (Of course, this operation might need to reallocate
+  // the buffer.)
+  void EnsureCapacity(size_t capacity) {
+    assert(IsConsistent());
+    if (capacity <= capacity_)
+      return;
+    scoped_ptr<uint8_t[]> new_data(new uint8_t[capacity]);
+    std::memcpy(new_data.get(), data_.get(), size_);
+    data_ = std::move(new_data);
+    capacity_ = capacity;
+    assert(IsConsistent());
   }
 
-  void TransferTo(Buffer* buf) {
-    assert(buf != NULL);
-    buf->data_.reset(data_.release());
-    buf->size_ = size_;
-    buf->capacity_ = capacity_;
-    Construct(NULL, 0, 0);
+  // b.Pass() does the same thing as std::move(b).
+  // Deprecated; remove in March 2016 (bug 5373).
+  RTC_DEPRECATED Buffer&& Pass() { return DEPRECATED_Pass(); }
+  Buffer&& DEPRECATED_Pass() {
+    assert(IsConsistent());
+    return std::move(*this);
   }
 
- protected:
-  void Construct(const void* data, size_t size, size_t capacity) {
-    data_.reset(new char[capacity_ = capacity]);
-    SetData(data, size);
+  // Resets the buffer to zero size and capacity. Works even if the buffer has
+  // been moved from.
+  void Clear() {
+    data_.reset();
+    size_ = 0;
+    capacity_ = 0;
+    assert(IsConsistent());
+  }
+
+  // Swaps two buffers. Also works for buffers that have been moved from.
+  friend void swap(Buffer& a, Buffer& b) {
+    using std::swap;
+    swap(a.size_, b.size_);
+    swap(a.capacity_, b.capacity_);
+    swap(a.data_, b.data_);
+  }
+
+ private:
+  // Precondition for all methods except Clear and the destructor.
+  // Postcondition for all methods except move construction and move
+  // assignment, which leave the moved-from object in a possibly inconsistent
+  // state.
+  bool IsConsistent() const {
+    return (data_ || capacity_ == 0) && capacity_ >= size_;
+  }
+
+  // Called when *this has been moved from. Conceptually it's a no-op, but we
+  // can mutate the state slightly to help subsequent sanity checks catch bugs.
+  void OnMovedFrom() {
+#ifdef NDEBUG
+    // Make *this consistent and empty. Shouldn't be necessary, but better safe
+    // than sorry.
+    size_ = 0;
+    capacity_ = 0;
+#else
+    // Ensure that *this is always inconsistent, to provoke bugs.
+    size_ = 1;
+    capacity_ = 0;
+#endif
   }
 
-  scoped_ptr<char[]> data_;
   size_t size_;
   size_t capacity_;
+  scoped_ptr<uint8_t[]> data_;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/buffer_unittest.cc b/media/webrtc/trunk/webrtc/base/buffer_unittest.cc
index 632ca81240..0b93b9b56e 100644
--- a/media/webrtc/trunk/webrtc/base/buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/buffer_unittest.cc
@@ -11,47 +11,65 @@
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/gunit.h"
 
+#include <algorithm>  // std::swap (pre-C++11)
+#include <utility>  // std::swap (C++11 and later)
+
 namespace rtc {
 
-static const char kTestData[] = {
-  0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7, 0x8, 0x9, 0xA, 0xB, 0xC, 0xD, 0xE, 0xF
-};
+namespace {
 
-TEST(BufferTest, TestConstructDefault) {
-  Buffer buf;
-  EXPECT_EQ(0U, buf.size());
-  EXPECT_EQ(0U, buf.capacity());
-  EXPECT_EQ(Buffer(), buf);
+// clang-format off
+const uint8_t kTestData[] = {0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7,
+                             0x8, 0x9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf};
+// clang-format on
+
+void TestBuf(const Buffer& b1, size_t size, size_t capacity) {
+  EXPECT_EQ(b1.size(), size);
+  EXPECT_EQ(b1.capacity(), capacity);
 }
 
-TEST(BufferTest, TestConstructEmptyWithCapacity) {
-  Buffer buf(NULL, 0, 256U);
-  EXPECT_EQ(0U, buf.size());
-  EXPECT_EQ(256U, buf.capacity());
-  EXPECT_EQ(Buffer(), buf);
+}  // namespace
+
+TEST(BufferTest, TestConstructEmpty) {
+  TestBuf(Buffer(), 0, 0);
+  TestBuf(Buffer(Buffer()), 0, 0);
+  TestBuf(Buffer(0), 0, 0);
+
+  // We can't use a literal 0 for the first argument, because C++ will allow
+  // that to be considered a null pointer, which makes the call ambiguous.
+  TestBuf(Buffer(0 + 0, 10), 0, 10);
+
+  TestBuf(Buffer(kTestData, 0), 0, 0);
+  TestBuf(Buffer(kTestData, 0, 20), 0, 20);
 }
 
 TEST(BufferTest, TestConstructData) {
-  Buffer buf(kTestData, sizeof(kTestData));
-  EXPECT_EQ(sizeof(kTestData), buf.size());
-  EXPECT_EQ(sizeof(kTestData), buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
-  EXPECT_EQ(Buffer(kTestData, sizeof(kTestData)), buf);
+  Buffer buf(kTestData, 7);
+  EXPECT_EQ(buf.size(), 7u);
+  EXPECT_EQ(buf.capacity(), 7u);
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 7));
 }
 
 TEST(BufferTest, TestConstructDataWithCapacity) {
-  Buffer buf(kTestData, sizeof(kTestData), 256U);
-  EXPECT_EQ(sizeof(kTestData), buf.size());
-  EXPECT_EQ(256U, buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
-  EXPECT_EQ(Buffer(kTestData, sizeof(kTestData)), buf);
+  Buffer buf(kTestData, 7, 14);
+  EXPECT_EQ(buf.size(), 7u);
+  EXPECT_EQ(buf.capacity(), 14u);
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 7));
+}
+
+TEST(BufferTest, TestConstructArray) {
+  Buffer buf(kTestData);
+  EXPECT_EQ(buf.size(), 16u);
+  EXPECT_EQ(buf.capacity(), 16u);
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 16));
 }
 
 TEST(BufferTest, TestConstructCopy) {
-  Buffer buf1(kTestData, sizeof(kTestData), 256), buf2(buf1);
-  EXPECT_EQ(sizeof(kTestData), buf2.size());
-  EXPECT_EQ(sizeof(kTestData), buf2.capacity());  // capacity isn't copied
-  EXPECT_EQ(0, memcmp(buf2.data(), kTestData, sizeof(kTestData)));
+  Buffer buf1(kTestData), buf2(buf1);
+  EXPECT_EQ(buf2.size(), 16u);
+  EXPECT_EQ(buf2.capacity(), 16u);
+  EXPECT_EQ(0, memcmp(buf2.data(), kTestData, 16));
+  EXPECT_NE(buf1.data(), buf2.data());
   EXPECT_EQ(buf1, buf2);
 }
 
@@ -59,85 +77,104 @@ TEST(BufferTest, TestAssign) {
   Buffer buf1, buf2(kTestData, sizeof(kTestData), 256);
   EXPECT_NE(buf1, buf2);
   buf1 = buf2;
-  EXPECT_EQ(sizeof(kTestData), buf1.size());
-  EXPECT_EQ(sizeof(kTestData), buf1.capacity());  // capacity isn't copied
-  EXPECT_EQ(0, memcmp(buf1.data(), kTestData, sizeof(kTestData)));
   EXPECT_EQ(buf1, buf2);
+  EXPECT_NE(buf1.data(), buf2.data());
 }
 
 TEST(BufferTest, TestSetData) {
-  Buffer buf;
-  buf.SetData(kTestData, sizeof(kTestData));
-  EXPECT_EQ(sizeof(kTestData), buf.size());
-  EXPECT_EQ(sizeof(kTestData), buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
+  Buffer buf(kTestData + 4, 7);
+  buf.SetData(kTestData, 9);
+  EXPECT_EQ(buf.size(), 9u);
+  EXPECT_EQ(buf.capacity(), 9u);
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 9));
 }
 
 TEST(BufferTest, TestAppendData) {
-  Buffer buf(kTestData, sizeof(kTestData));
-  buf.AppendData(kTestData, sizeof(kTestData));
-  EXPECT_EQ(2 * sizeof(kTestData), buf.size());
-  EXPECT_EQ(2 * sizeof(kTestData), buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
-  EXPECT_EQ(0, memcmp(buf.data() + sizeof(kTestData),
-                      kTestData, sizeof(kTestData)));
+  Buffer buf(kTestData + 4, 3);
+  buf.AppendData(kTestData + 10, 2);
+  const int8_t exp[] = {0x4, 0x5, 0x6, 0xa, 0xb};
+  EXPECT_EQ(buf, Buffer(exp));
 }
 
 TEST(BufferTest, TestSetSizeSmaller) {
   Buffer buf;
-  buf.SetData(kTestData, sizeof(kTestData));
-  buf.SetSize(sizeof(kTestData) / 2);
-  EXPECT_EQ(sizeof(kTestData) / 2, buf.size());
-  EXPECT_EQ(sizeof(kTestData), buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData) / 2));
+  buf.SetData(kTestData, 15);
+  buf.SetSize(10);
+  EXPECT_EQ(buf.size(), 10u);
+  EXPECT_EQ(buf.capacity(), 15u);  // Hasn't shrunk.
+  EXPECT_EQ(buf, Buffer(kTestData, 10));
 }
 
 TEST(BufferTest, TestSetSizeLarger) {
   Buffer buf;
-  buf.SetData(kTestData, sizeof(kTestData));
-  buf.SetSize(sizeof(kTestData) * 2);
-  EXPECT_EQ(sizeof(kTestData) * 2, buf.size());
-  EXPECT_EQ(sizeof(kTestData) * 2, buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
+  buf.SetData(kTestData, 15);
+  EXPECT_EQ(buf.size(), 15u);
+  EXPECT_EQ(buf.capacity(), 15u);
+  buf.SetSize(20);
+  EXPECT_EQ(buf.size(), 20u);
+  EXPECT_EQ(buf.capacity(), 20u);  // Has grown.
+  EXPECT_EQ(0, memcmp(buf.data(), kTestData, 15));
 }
 
-TEST(BufferTest, TestSetCapacitySmaller) {
-  Buffer buf;
-  buf.SetData(kTestData, sizeof(kTestData));
-  buf.SetCapacity(sizeof(kTestData) / 2);  // should be ignored
-  EXPECT_EQ(sizeof(kTestData), buf.size());
-  EXPECT_EQ(sizeof(kTestData), buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
+TEST(BufferTest, TestEnsureCapacitySmaller) {
+  Buffer buf(kTestData);
+  const char* data = buf.data<char>();
+  buf.EnsureCapacity(4);
+  EXPECT_EQ(buf.capacity(), 16u);     // Hasn't shrunk.
+  EXPECT_EQ(buf.data<char>(), data);  // No reallocation.
+  EXPECT_EQ(buf, Buffer(kTestData));
 }
 
-TEST(BufferTest, TestSetCapacityLarger) {
-  Buffer buf(kTestData, sizeof(kTestData));
-  buf.SetCapacity(sizeof(kTestData) * 2);
-  EXPECT_EQ(sizeof(kTestData), buf.size());
-  EXPECT_EQ(sizeof(kTestData) * 2, buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
+TEST(BufferTest, TestEnsureCapacityLarger) {
+  Buffer buf(kTestData, 5);
+  buf.EnsureCapacity(10);
+  const int8_t* data = buf.data<int8_t>();
+  EXPECT_EQ(buf.capacity(), 10u);
+  buf.AppendData(kTestData + 5, 5);
+  EXPECT_EQ(buf.data<int8_t>(), data);  // No reallocation.
+  EXPECT_EQ(buf, Buffer(kTestData, 10));
 }
 
-TEST(BufferTest, TestSetCapacityThenSetSize) {
-  Buffer buf(kTestData, sizeof(kTestData));
-  buf.SetCapacity(sizeof(kTestData) * 4);
-  memcpy(buf.data() + sizeof(kTestData), kTestData, sizeof(kTestData));
-  buf.SetSize(sizeof(kTestData) * 2);
-  EXPECT_EQ(sizeof(kTestData) * 2, buf.size());
-  EXPECT_EQ(sizeof(kTestData) * 4, buf.capacity());
-  EXPECT_EQ(0, memcmp(buf.data(), kTestData, sizeof(kTestData)));
-  EXPECT_EQ(0, memcmp(buf.data() + sizeof(kTestData),
-                      kTestData, sizeof(kTestData)));
+TEST(BufferTest, TestMoveConstruct) {
+  Buffer buf1(kTestData, 3, 40);
+  const uint8_t* data = buf1.data();
+  Buffer buf2(buf1.DEPRECATED_Pass());
+  EXPECT_EQ(buf2.size(), 3u);
+  EXPECT_EQ(buf2.capacity(), 40u);
+  EXPECT_EQ(buf2.data(), data);
+  buf1.Clear();
+  EXPECT_EQ(buf1.size(), 0u);
+  EXPECT_EQ(buf1.capacity(), 0u);
+  EXPECT_EQ(buf1.data(), nullptr);
 }
 
-TEST(BufferTest, TestTransfer) {
-  Buffer buf1(kTestData, sizeof(kTestData), 256U), buf2;
-  buf1.TransferTo(&buf2);
-  EXPECT_EQ(0U, buf1.size());
-  EXPECT_EQ(0U, buf1.capacity());
-  EXPECT_EQ(sizeof(kTestData), buf2.size());
-  EXPECT_EQ(256U, buf2.capacity());  // capacity does transfer
-  EXPECT_EQ(0, memcmp(buf2.data(), kTestData, sizeof(kTestData)));
+TEST(BufferTest, TestMoveAssign) {
+  Buffer buf1(kTestData, 3, 40);
+  const uint8_t* data = buf1.data();
+  Buffer buf2(kTestData);
+  buf2 = buf1.DEPRECATED_Pass();
+  EXPECT_EQ(buf2.size(), 3u);
+  EXPECT_EQ(buf2.capacity(), 40u);
+  EXPECT_EQ(buf2.data(), data);
+  buf1.Clear();
+  EXPECT_EQ(buf1.size(), 0u);
+  EXPECT_EQ(buf1.capacity(), 0u);
+  EXPECT_EQ(buf1.data(), nullptr);
+}
+
+TEST(BufferTest, TestSwap) {
+  Buffer buf1(kTestData, 3);
+  Buffer buf2(kTestData, 6, 40);
+  uint8_t* data1 = buf1.data();
+  uint8_t* data2 = buf2.data();
+  using std::swap;
+  swap(buf1, buf2);
+  EXPECT_EQ(buf1.size(), 6u);
+  EXPECT_EQ(buf1.capacity(), 40u);
+  EXPECT_EQ(buf1.data(), data2);
+  EXPECT_EQ(buf2.size(), 3u);
+  EXPECT_EQ(buf2.capacity(), 3u);
+  EXPECT_EQ(buf2.data(), data1);
 }
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/bufferqueue.cc b/media/webrtc/trunk/webrtc/base/bufferqueue.cc
new file mode 100644
index 0000000000..1ac57abc0c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/bufferqueue.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/bufferqueue.h"
+
+namespace rtc {
+
+BufferQueue::BufferQueue(size_t capacity, size_t default_size)
+    : capacity_(capacity), default_size_(default_size) {
+}
+
+BufferQueue::~BufferQueue() {
+  CritScope cs(&crit_);
+
+  for (Buffer* buffer : queue_) {
+    delete buffer;
+  }
+  for (Buffer* buffer : free_list_) {
+    delete buffer;
+  }
+}
+
+size_t BufferQueue::size() const {
+  CritScope cs(&crit_);
+  return queue_.size();
+}
+
+bool BufferQueue::ReadFront(void* buffer, size_t bytes, size_t* bytes_read) {
+  CritScope cs(&crit_);
+  if (queue_.empty()) {
+    return false;
+  }
+
+  bool was_writable = queue_.size() < capacity_;
+  Buffer* packet = queue_.front();
+  queue_.pop_front();
+
+  bytes = std::min(bytes, packet->size());
+  memcpy(buffer, packet->data(), bytes);
+  if (bytes_read) {
+    *bytes_read = bytes;
+  }
+  free_list_.push_back(packet);
+  if (!was_writable) {
+    NotifyWritableForTest();
+  }
+  return true;
+}
+
+bool BufferQueue::WriteBack(const void* buffer, size_t bytes,
+                            size_t* bytes_written) {
+  CritScope cs(&crit_);
+  if (queue_.size() == capacity_) {
+    return false;
+  }
+
+  bool was_readable = !queue_.empty();
+  Buffer* packet;
+  if (!free_list_.empty()) {
+    packet = free_list_.back();
+    free_list_.pop_back();
+  } else {
+    packet = new Buffer(bytes, default_size_);
+  }
+
+  packet->SetData(static_cast<const uint8_t*>(buffer), bytes);
+  if (bytes_written) {
+    *bytes_written = bytes;
+  }
+  queue_.push_back(packet);
+  if (!was_readable) {
+    NotifyReadableForTest();
+  }
+  return true;
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/bufferqueue.h b/media/webrtc/trunk/webrtc/base/bufferqueue.h
new file mode 100644
index 0000000000..458f0189cd
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/bufferqueue.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BUFFERQUEUE_H_
+#define WEBRTC_BASE_BUFFERQUEUE_H_
+
+#include <deque>
+#include <vector>
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/criticalsection.h"
+
+namespace rtc {
+
+class BufferQueue {
+ public:
+  // Creates a buffer queue with a given capacity and default buffer size.
+  BufferQueue(size_t capacity, size_t default_size);
+  virtual ~BufferQueue();
+
+  // Return number of queued buffers.
+  size_t size() const;
+
+  // ReadFront will only read one buffer at a time and will truncate buffers
+  // that don't fit in the passed memory.
+  // Returns true unless no data could be returned.
+  bool ReadFront(void* data, size_t bytes, size_t* bytes_read);
+
+  // WriteBack always writes either the complete memory or nothing.
+  // Returns true unless no data could be written.
+  bool WriteBack(const void* data, size_t bytes, size_t* bytes_written);
+
+ protected:
+  // These methods are called when the state of the queue changes.
+  virtual void NotifyReadableForTest() {}
+  virtual void NotifyWritableForTest() {}
+
+ private:
+  size_t capacity_;
+  size_t default_size_;
+  mutable CriticalSection crit_;
+  std::deque<Buffer*> queue_ GUARDED_BY(crit_);
+  std::vector<Buffer*> free_list_ GUARDED_BY(crit_);
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(BufferQueue);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_BUFFERQUEUE_H_
diff --git a/media/webrtc/trunk/webrtc/base/bufferqueue_unittest.cc b/media/webrtc/trunk/webrtc/base/bufferqueue_unittest.cc
new file mode 100644
index 0000000000..07084c4a61
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/bufferqueue_unittest.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/bufferqueue.h"
+#include "webrtc/base/gunit.h"
+
+namespace rtc {
+
+TEST(BufferQueueTest, TestAll) {
+  const size_t kSize = 16;
+  const char in[kSize * 2 + 1] = "0123456789ABCDEFGHIJKLMNOPQRSTUV";
+  char out[kSize * 2];
+  size_t bytes;
+  BufferQueue queue1(1, kSize);
+  BufferQueue queue2(2, kSize);
+
+  // The queue is initially empty.
+  EXPECT_EQ(0u, queue1.size());
+  EXPECT_FALSE(queue1.ReadFront(out, kSize, &bytes));
+
+  // A write should succeed.
+  EXPECT_TRUE(queue1.WriteBack(in, kSize, &bytes));
+  EXPECT_EQ(kSize, bytes);
+  EXPECT_EQ(1u, queue1.size());
+
+  // The queue is full now (only one buffer allowed).
+  EXPECT_FALSE(queue1.WriteBack(in, kSize, &bytes));
+  EXPECT_EQ(1u, queue1.size());
+
+  // Reading previously written buffer.
+  EXPECT_TRUE(queue1.ReadFront(out, kSize, &bytes));
+  EXPECT_EQ(kSize, bytes);
+  EXPECT_EQ(0, memcmp(in, out, kSize));
+
+  // The queue is empty again now.
+  EXPECT_FALSE(queue1.ReadFront(out, kSize, &bytes));
+  EXPECT_EQ(0u, queue1.size());
+
+  // Reading only returns available data.
+  EXPECT_TRUE(queue1.WriteBack(in, kSize, &bytes));
+  EXPECT_EQ(kSize, bytes);
+  EXPECT_EQ(1u, queue1.size());
+  EXPECT_TRUE(queue1.ReadFront(out, kSize * 2, &bytes));
+  EXPECT_EQ(kSize, bytes);
+  EXPECT_EQ(0, memcmp(in, out, kSize));
+  EXPECT_EQ(0u, queue1.size());
+
+  // Reading maintains buffer boundaries.
+  EXPECT_TRUE(queue2.WriteBack(in, kSize / 2, &bytes));
+  EXPECT_EQ(1u, queue2.size());
+  EXPECT_TRUE(queue2.WriteBack(in + kSize / 2, kSize / 2, &bytes));
+  EXPECT_EQ(2u, queue2.size());
+  EXPECT_TRUE(queue2.ReadFront(out, kSize, &bytes));
+  EXPECT_EQ(kSize / 2, bytes);
+  EXPECT_EQ(0, memcmp(in, out, kSize / 2));
+  EXPECT_EQ(1u, queue2.size());
+  EXPECT_TRUE(queue2.ReadFront(out, kSize, &bytes));
+  EXPECT_EQ(kSize / 2, bytes);
+  EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 2));
+  EXPECT_EQ(0u, queue2.size());
+
+  // Reading truncates buffers.
+  EXPECT_TRUE(queue2.WriteBack(in, kSize / 2, &bytes));
+  EXPECT_EQ(1u, queue2.size());
+  EXPECT_TRUE(queue2.WriteBack(in + kSize / 2, kSize / 2, &bytes));
+  EXPECT_EQ(2u, queue2.size());
+  // Read first packet partially in too-small buffer.
+  EXPECT_TRUE(queue2.ReadFront(out, kSize / 4, &bytes));
+  EXPECT_EQ(kSize / 4, bytes);
+  EXPECT_EQ(0, memcmp(in, out, kSize / 4));
+  EXPECT_EQ(1u, queue2.size());
+  // Remainder of first packet is truncated, reading starts with next packet.
+  EXPECT_TRUE(queue2.ReadFront(out, kSize, &bytes));
+  EXPECT_EQ(kSize / 2, bytes);
+  EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 2));
+  EXPECT_EQ(0u, queue2.size());
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/bytebuffer.cc b/media/webrtc/trunk/webrtc/base/bytebuffer.cc
index d2aa4cfd6f..8bc1f23670 100644
--- a/media/webrtc/trunk/webrtc/base/bytebuffer.cc
+++ b/media/webrtc/trunk/webrtc/base/bytebuffer.cc
@@ -42,6 +42,10 @@ ByteBuffer::ByteBuffer(const char* bytes) {
   Construct(bytes, strlen(bytes), ORDER_NETWORK);
 }
 
+ByteBuffer::ByteBuffer(const Buffer& buf) {
+  Construct(buf.data<char>(), buf.size(), ORDER_NETWORK);
+}
+
 void ByteBuffer::Construct(const char* bytes, size_t len,
                            ByteOrder byte_order) {
   version_ = 0;
@@ -62,16 +66,16 @@ ByteBuffer::~ByteBuffer() {
   delete[] bytes_;
 }
 
-bool ByteBuffer::ReadUInt8(uint8* val) {
+bool ByteBuffer::ReadUInt8(uint8_t* val) {
   if (!val) return false;
 
   return ReadBytes(reinterpret_cast<char*>(val), 1);
 }
 
-bool ByteBuffer::ReadUInt16(uint16* val) {
+bool ByteBuffer::ReadUInt16(uint16_t* val) {
   if (!val) return false;
 
-  uint16 v;
+  uint16_t v;
   if (!ReadBytes(reinterpret_cast<char*>(&v), 2)) {
     return false;
   } else {
@@ -80,10 +84,10 @@ bool ByteBuffer::ReadUInt16(uint16* val) {
   }
 }
 
-bool ByteBuffer::ReadUInt24(uint32* val) {
+bool ByteBuffer::ReadUInt24(uint32_t* val) {
   if (!val) return false;
 
-  uint32 v = 0;
+  uint32_t v = 0;
   char* read_into = reinterpret_cast<char*>(&v);
   if (byte_order_ == ORDER_NETWORK || IsHostBigEndian()) {
     ++read_into;
@@ -97,10 +101,10 @@ bool ByteBuffer::ReadUInt24(uint32* val) {
   }
 }
 
-bool ByteBuffer::ReadUInt32(uint32* val) {
+bool ByteBuffer::ReadUInt32(uint32_t* val) {
   if (!val) return false;
 
-  uint32 v;
+  uint32_t v;
   if (!ReadBytes(reinterpret_cast<char*>(&v), 4)) {
     return false;
   } else {
@@ -109,10 +113,10 @@ bool ByteBuffer::ReadUInt32(uint32* val) {
   }
 }
 
-bool ByteBuffer::ReadUInt64(uint64* val) {
+bool ByteBuffer::ReadUInt64(uint64_t* val) {
   if (!val) return false;
 
-  uint64 v;
+  uint64_t v;
   if (!ReadBytes(reinterpret_cast<char*>(&v), 8)) {
     return false;
   } else {
@@ -143,17 +147,17 @@ bool ByteBuffer::ReadBytes(char* val, size_t len) {
   }
 }
 
-void ByteBuffer::WriteUInt8(uint8 val) {
+void ByteBuffer::WriteUInt8(uint8_t val) {
   WriteBytes(reinterpret_cast<const char*>(&val), 1);
 }
 
-void ByteBuffer::WriteUInt16(uint16 val) {
-  uint16 v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork16(val) : val;
+void ByteBuffer::WriteUInt16(uint16_t val) {
+  uint16_t v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork16(val) : val;
   WriteBytes(reinterpret_cast<const char*>(&v), 2);
 }
 
-void ByteBuffer::WriteUInt24(uint32 val) {
-  uint32 v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork32(val) : val;
+void ByteBuffer::WriteUInt24(uint32_t val) {
+  uint32_t v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork32(val) : val;
   char* start = reinterpret_cast<char*>(&v);
   if (byte_order_ == ORDER_NETWORK || IsHostBigEndian()) {
     ++start;
@@ -161,13 +165,13 @@ void ByteBuffer::WriteUInt24(uint32 val) {
   WriteBytes(start, 3);
 }
 
-void ByteBuffer::WriteUInt32(uint32 val) {
-  uint32 v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork32(val) : val;
+void ByteBuffer::WriteUInt32(uint32_t val) {
+  uint32_t v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork32(val) : val;
   WriteBytes(reinterpret_cast<const char*>(&v), 4);
 }
 
-void ByteBuffer::WriteUInt64(uint64 val) {
-  uint64 v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork64(val) : val;
+void ByteBuffer::WriteUInt64(uint64_t val) {
+  uint64_t v = (byte_order_ == ORDER_NETWORK) ? HostToNetwork64(val) : val;
   WriteBytes(reinterpret_cast<const char*>(&v), 8);
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/bytebuffer.h b/media/webrtc/trunk/webrtc/base/bytebuffer.h
index 1934f418e5..ad2e552ed6 100644
--- a/media/webrtc/trunk/webrtc/base/bytebuffer.h
+++ b/media/webrtc/trunk/webrtc/base/bytebuffer.h
@@ -14,6 +14,7 @@
 #include <string>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/constructormagic.h"
 
 namespace rtc {
@@ -35,6 +36,8 @@ class ByteBuffer {
   // Initializes buffer from a zero-terminated string.
   explicit ByteBuffer(const char* bytes);
 
+  explicit ByteBuffer(const Buffer& buf);
+
   ~ByteBuffer();
 
   const char* Data() const { return bytes_ + start_; }
@@ -44,11 +47,11 @@ class ByteBuffer {
 
   // Read a next value from the buffer. Return false if there isn't
   // enough data left for the specified type.
-  bool ReadUInt8(uint8* val);
-  bool ReadUInt16(uint16* val);
-  bool ReadUInt24(uint32* val);
-  bool ReadUInt32(uint32* val);
-  bool ReadUInt64(uint64* val);
+  bool ReadUInt8(uint8_t* val);
+  bool ReadUInt16(uint16_t* val);
+  bool ReadUInt24(uint32_t* val);
+  bool ReadUInt32(uint32_t* val);
+  bool ReadUInt64(uint64_t* val);
   bool ReadBytes(char* val, size_t len);
 
   // Appends next |len| bytes from the buffer to |val|. Returns false
@@ -57,11 +60,11 @@ class ByteBuffer {
 
   // Write value to the buffer. Resizes the buffer when it is
   // neccessary.
-  void WriteUInt8(uint8 val);
-  void WriteUInt16(uint16 val);
-  void WriteUInt24(uint32 val);
-  void WriteUInt32(uint32 val);
-  void WriteUInt64(uint64 val);
+  void WriteUInt8(uint8_t val);
+  void WriteUInt16(uint16_t val);
+  void WriteUInt24(uint32_t val);
+  void WriteUInt32(uint32_t val);
+  void WriteUInt64(uint64_t val);
   void WriteString(const std::string& val);
   void WriteBytes(const char* val, size_t len);
 
@@ -111,7 +114,7 @@ class ByteBuffer {
 
   // There are sensible ways to define these, but they aren't needed in our code
   // base.
-  DISALLOW_COPY_AND_ASSIGN(ByteBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ByteBuffer);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/bytebuffer_unittest.cc b/media/webrtc/trunk/webrtc/base/bytebuffer_unittest.cc
index f4b0504efc..0287d85e6f 100644
--- a/media/webrtc/trunk/webrtc/base/bytebuffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/bytebuffer_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/bytebuffer.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
@@ -16,9 +17,9 @@
 namespace rtc {
 
 TEST(ByteBufferTest, TestByteOrder) {
-  uint16 n16 = 1;
-  uint32 n32 = 1;
-  uint64 n64 = 1;
+  uint16_t n16 = 1;
+  uint32_t n32 = 1;
+  uint64_t n64 = 1;
 
   EXPECT_EQ(n16, NetworkToHost16(HostToNetwork16(n16)));
   EXPECT_EQ(n32, NetworkToHost32(HostToNetwork32(n32)));
@@ -114,48 +115,48 @@ TEST(ByteBufferTest, TestGetSetReadPosition) {
 TEST(ByteBufferTest, TestReadWriteBuffer) {
   ByteBuffer::ByteOrder orders[2] = { ByteBuffer::ORDER_HOST,
                                       ByteBuffer::ORDER_NETWORK };
-  for (size_t i = 0; i < ARRAY_SIZE(orders); i++) {
+  for (size_t i = 0; i < arraysize(orders); i++) {
     ByteBuffer buffer(orders[i]);
     EXPECT_EQ(orders[i], buffer.Order());
-    uint8 ru8;
+    uint8_t ru8;
     EXPECT_FALSE(buffer.ReadUInt8(&ru8));
 
-    // Write and read uint8.
-    uint8 wu8 = 1;
+    // Write and read uint8_t.
+    uint8_t wu8 = 1;
     buffer.WriteUInt8(wu8);
     EXPECT_TRUE(buffer.ReadUInt8(&ru8));
     EXPECT_EQ(wu8, ru8);
     EXPECT_EQ(0U, buffer.Length());
 
-    // Write and read uint16.
-    uint16 wu16 = (1 << 8) + 1;
+    // Write and read uint16_t.
+    uint16_t wu16 = (1 << 8) + 1;
     buffer.WriteUInt16(wu16);
-    uint16 ru16;
+    uint16_t ru16;
     EXPECT_TRUE(buffer.ReadUInt16(&ru16));
     EXPECT_EQ(wu16, ru16);
     EXPECT_EQ(0U, buffer.Length());
 
     // Write and read uint24.
-    uint32 wu24 = (3 << 16) + (2 << 8) + 1;
+    uint32_t wu24 = (3 << 16) + (2 << 8) + 1;
     buffer.WriteUInt24(wu24);
-    uint32 ru24;
+    uint32_t ru24;
     EXPECT_TRUE(buffer.ReadUInt24(&ru24));
     EXPECT_EQ(wu24, ru24);
     EXPECT_EQ(0U, buffer.Length());
 
-    // Write and read uint32.
-    uint32 wu32 = (4 << 24) + (3 << 16) + (2 << 8) + 1;
+    // Write and read uint32_t.
+    uint32_t wu32 = (4 << 24) + (3 << 16) + (2 << 8) + 1;
     buffer.WriteUInt32(wu32);
-    uint32 ru32;
+    uint32_t ru32;
     EXPECT_TRUE(buffer.ReadUInt32(&ru32));
     EXPECT_EQ(wu32, ru32);
     EXPECT_EQ(0U, buffer.Length());
 
-    // Write and read uint64.
-    uint32 another32 = (8 << 24) + (7 << 16) + (6 << 8) + 5;
-    uint64 wu64 = (static_cast<uint64>(another32) << 32) + wu32;
+    // Write and read uint64_t.
+    uint32_t another32 = (8 << 24) + (7 << 16) + (6 << 8) + 5;
+    uint64_t wu64 = (static_cast<uint64_t>(another32) << 32) + wu32;
     buffer.WriteUInt64(wu64);
-    uint64 ru64;
+    uint64_t ru64;
     EXPECT_TRUE(buffer.ReadUInt64(&ru64));
     EXPECT_EQ(wu64, ru64);
     EXPECT_EQ(0U, buffer.Length());
diff --git a/media/webrtc/trunk/webrtc/base/byteorder.h b/media/webrtc/trunk/webrtc/base/byteorder.h
index d907d9e412..d579e6e185 100644
--- a/media/webrtc/trunk/webrtc/base/byteorder.h
+++ b/media/webrtc/trunk/webrtc/base/byteorder.h
@@ -27,104 +27,102 @@ namespace rtc {
 // TODO: Optimized versions, with direct read/writes of
 // integers in host-endian format, when the platform supports it.
 
-inline void Set8(void* memory, size_t offset, uint8 v) {
-  static_cast<uint8*>(memory)[offset] = v;
+inline void Set8(void* memory, size_t offset, uint8_t v) {
+  static_cast<uint8_t*>(memory)[offset] = v;
 }
 
-inline uint8 Get8(const void* memory, size_t offset) {
-  return static_cast<const uint8*>(memory)[offset];
+inline uint8_t Get8(const void* memory, size_t offset) {
+  return static_cast<const uint8_t*>(memory)[offset];
 }
 
-inline void SetBE16(void* memory, uint16 v) {
-  Set8(memory, 0, static_cast<uint8>(v >> 8));
-  Set8(memory, 1, static_cast<uint8>(v >> 0));
+inline void SetBE16(void* memory, uint16_t v) {
+  Set8(memory, 0, static_cast<uint8_t>(v >> 8));
+  Set8(memory, 1, static_cast<uint8_t>(v >> 0));
 }
 
-inline void SetBE32(void* memory, uint32 v) {
-  Set8(memory, 0, static_cast<uint8>(v >> 24));
-  Set8(memory, 1, static_cast<uint8>(v >> 16));
-  Set8(memory, 2, static_cast<uint8>(v >> 8));
-  Set8(memory, 3, static_cast<uint8>(v >> 0));
+inline void SetBE32(void* memory, uint32_t v) {
+  Set8(memory, 0, static_cast<uint8_t>(v >> 24));
+  Set8(memory, 1, static_cast<uint8_t>(v >> 16));
+  Set8(memory, 2, static_cast<uint8_t>(v >> 8));
+  Set8(memory, 3, static_cast<uint8_t>(v >> 0));
 }
 
-inline void SetBE64(void* memory, uint64 v) {
-  Set8(memory, 0, static_cast<uint8>(v >> 56));
-  Set8(memory, 1, static_cast<uint8>(v >> 48));
-  Set8(memory, 2, static_cast<uint8>(v >> 40));
-  Set8(memory, 3, static_cast<uint8>(v >> 32));
-  Set8(memory, 4, static_cast<uint8>(v >> 24));
-  Set8(memory, 5, static_cast<uint8>(v >> 16));
-  Set8(memory, 6, static_cast<uint8>(v >> 8));
-  Set8(memory, 7, static_cast<uint8>(v >> 0));
+inline void SetBE64(void* memory, uint64_t v) {
+  Set8(memory, 0, static_cast<uint8_t>(v >> 56));
+  Set8(memory, 1, static_cast<uint8_t>(v >> 48));
+  Set8(memory, 2, static_cast<uint8_t>(v >> 40));
+  Set8(memory, 3, static_cast<uint8_t>(v >> 32));
+  Set8(memory, 4, static_cast<uint8_t>(v >> 24));
+  Set8(memory, 5, static_cast<uint8_t>(v >> 16));
+  Set8(memory, 6, static_cast<uint8_t>(v >> 8));
+  Set8(memory, 7, static_cast<uint8_t>(v >> 0));
 }
 
-inline uint16 GetBE16(const void* memory) {
-  return static_cast<uint16>((Get8(memory, 0) << 8) |
-                             (Get8(memory, 1) << 0));
+inline uint16_t GetBE16(const void* memory) {
+  return static_cast<uint16_t>((Get8(memory, 0) << 8) | (Get8(memory, 1) << 0));
 }
 
-inline uint32 GetBE32(const void* memory) {
-  return (static_cast<uint32>(Get8(memory, 0)) << 24) |
-      (static_cast<uint32>(Get8(memory, 1)) << 16) |
-      (static_cast<uint32>(Get8(memory, 2)) << 8) |
-      (static_cast<uint32>(Get8(memory, 3)) << 0);
+inline uint32_t GetBE32(const void* memory) {
+  return (static_cast<uint32_t>(Get8(memory, 0)) << 24) |
+         (static_cast<uint32_t>(Get8(memory, 1)) << 16) |
+         (static_cast<uint32_t>(Get8(memory, 2)) << 8) |
+         (static_cast<uint32_t>(Get8(memory, 3)) << 0);
 }
 
-inline uint64 GetBE64(const void* memory) {
-  return (static_cast<uint64>(Get8(memory, 0)) << 56) |
-      (static_cast<uint64>(Get8(memory, 1)) << 48) |
-      (static_cast<uint64>(Get8(memory, 2)) << 40) |
-      (static_cast<uint64>(Get8(memory, 3)) << 32) |
-      (static_cast<uint64>(Get8(memory, 4)) << 24) |
-      (static_cast<uint64>(Get8(memory, 5)) << 16) |
-      (static_cast<uint64>(Get8(memory, 6)) << 8) |
-      (static_cast<uint64>(Get8(memory, 7)) << 0);
+inline uint64_t GetBE64(const void* memory) {
+  return (static_cast<uint64_t>(Get8(memory, 0)) << 56) |
+         (static_cast<uint64_t>(Get8(memory, 1)) << 48) |
+         (static_cast<uint64_t>(Get8(memory, 2)) << 40) |
+         (static_cast<uint64_t>(Get8(memory, 3)) << 32) |
+         (static_cast<uint64_t>(Get8(memory, 4)) << 24) |
+         (static_cast<uint64_t>(Get8(memory, 5)) << 16) |
+         (static_cast<uint64_t>(Get8(memory, 6)) << 8) |
+         (static_cast<uint64_t>(Get8(memory, 7)) << 0);
 }
 
-inline void SetLE16(void* memory, uint16 v) {
-  Set8(memory, 0, static_cast<uint8>(v >> 0));
-  Set8(memory, 1, static_cast<uint8>(v >> 8));
+inline void SetLE16(void* memory, uint16_t v) {
+  Set8(memory, 0, static_cast<uint8_t>(v >> 0));
+  Set8(memory, 1, static_cast<uint8_t>(v >> 8));
 }
 
-inline void SetLE32(void* memory, uint32 v) {
-  Set8(memory, 0, static_cast<uint8>(v >> 0));
-  Set8(memory, 1, static_cast<uint8>(v >> 8));
-  Set8(memory, 2, static_cast<uint8>(v >> 16));
-  Set8(memory, 3, static_cast<uint8>(v >> 24));
+inline void SetLE32(void* memory, uint32_t v) {
+  Set8(memory, 0, static_cast<uint8_t>(v >> 0));
+  Set8(memory, 1, static_cast<uint8_t>(v >> 8));
+  Set8(memory, 2, static_cast<uint8_t>(v >> 16));
+  Set8(memory, 3, static_cast<uint8_t>(v >> 24));
 }
 
-inline void SetLE64(void* memory, uint64 v) {
-  Set8(memory, 0, static_cast<uint8>(v >> 0));
-  Set8(memory, 1, static_cast<uint8>(v >> 8));
-  Set8(memory, 2, static_cast<uint8>(v >> 16));
-  Set8(memory, 3, static_cast<uint8>(v >> 24));
-  Set8(memory, 4, static_cast<uint8>(v >> 32));
-  Set8(memory, 5, static_cast<uint8>(v >> 40));
-  Set8(memory, 6, static_cast<uint8>(v >> 48));
-  Set8(memory, 7, static_cast<uint8>(v >> 56));
+inline void SetLE64(void* memory, uint64_t v) {
+  Set8(memory, 0, static_cast<uint8_t>(v >> 0));
+  Set8(memory, 1, static_cast<uint8_t>(v >> 8));
+  Set8(memory, 2, static_cast<uint8_t>(v >> 16));
+  Set8(memory, 3, static_cast<uint8_t>(v >> 24));
+  Set8(memory, 4, static_cast<uint8_t>(v >> 32));
+  Set8(memory, 5, static_cast<uint8_t>(v >> 40));
+  Set8(memory, 6, static_cast<uint8_t>(v >> 48));
+  Set8(memory, 7, static_cast<uint8_t>(v >> 56));
 }
 
-inline uint16 GetLE16(const void* memory) {
-  return static_cast<uint16>((Get8(memory, 0) << 0) |
-                             (Get8(memory, 1) << 8));
+inline uint16_t GetLE16(const void* memory) {
+  return static_cast<uint16_t>((Get8(memory, 0) << 0) | (Get8(memory, 1) << 8));
 }
 
-inline uint32 GetLE32(const void* memory) {
-  return (static_cast<uint32>(Get8(memory, 0)) << 0) |
-      (static_cast<uint32>(Get8(memory, 1)) << 8) |
-      (static_cast<uint32>(Get8(memory, 2)) << 16) |
-      (static_cast<uint32>(Get8(memory, 3)) << 24);
+inline uint32_t GetLE32(const void* memory) {
+  return (static_cast<uint32_t>(Get8(memory, 0)) << 0) |
+         (static_cast<uint32_t>(Get8(memory, 1)) << 8) |
+         (static_cast<uint32_t>(Get8(memory, 2)) << 16) |
+         (static_cast<uint32_t>(Get8(memory, 3)) << 24);
 }
 
-inline uint64 GetLE64(const void* memory) {
-  return (static_cast<uint64>(Get8(memory, 0)) << 0) |
-      (static_cast<uint64>(Get8(memory, 1)) << 8) |
-      (static_cast<uint64>(Get8(memory, 2)) << 16) |
-      (static_cast<uint64>(Get8(memory, 3)) << 24) |
-      (static_cast<uint64>(Get8(memory, 4)) << 32) |
-      (static_cast<uint64>(Get8(memory, 5)) << 40) |
-      (static_cast<uint64>(Get8(memory, 6)) << 48) |
-      (static_cast<uint64>(Get8(memory, 7)) << 56);
+inline uint64_t GetLE64(const void* memory) {
+  return (static_cast<uint64_t>(Get8(memory, 0)) << 0) |
+         (static_cast<uint64_t>(Get8(memory, 1)) << 8) |
+         (static_cast<uint64_t>(Get8(memory, 2)) << 16) |
+         (static_cast<uint64_t>(Get8(memory, 3)) << 24) |
+         (static_cast<uint64_t>(Get8(memory, 4)) << 32) |
+         (static_cast<uint64_t>(Get8(memory, 5)) << 40) |
+         (static_cast<uint64_t>(Get8(memory, 6)) << 48) |
+         (static_cast<uint64_t>(Get8(memory, 7)) << 56);
 }
 
 // Check if the current host is big endian.
@@ -133,33 +131,33 @@ inline bool IsHostBigEndian() {
   return 0 == *reinterpret_cast<const char*>(&number);
 }
 
-inline uint16 HostToNetwork16(uint16 n) {
-  uint16 result;
+inline uint16_t HostToNetwork16(uint16_t n) {
+  uint16_t result;
   SetBE16(&result, n);
   return result;
 }
 
-inline uint32 HostToNetwork32(uint32 n) {
-  uint32 result;
+inline uint32_t HostToNetwork32(uint32_t n) {
+  uint32_t result;
   SetBE32(&result, n);
   return result;
 }
 
-inline uint64 HostToNetwork64(uint64 n) {
-  uint64 result;
+inline uint64_t HostToNetwork64(uint64_t n) {
+  uint64_t result;
   SetBE64(&result, n);
   return result;
 }
 
-inline uint16 NetworkToHost16(uint16 n) {
+inline uint16_t NetworkToHost16(uint16_t n) {
   return GetBE16(&n);
 }
 
-inline uint32 NetworkToHost32(uint32 n) {
+inline uint32_t NetworkToHost32(uint32_t n) {
   return GetBE32(&n);
 }
 
-inline uint64 NetworkToHost64(uint64 n) {
+inline uint64_t NetworkToHost64(uint64_t n) {
   return GetBE64(&n);
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/byteorder_unittest.cc b/media/webrtc/trunk/webrtc/base/byteorder_unittest.cc
index f4e7df3b71..c3135aa7c9 100644
--- a/media/webrtc/trunk/webrtc/base/byteorder_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/byteorder_unittest.cc
@@ -17,7 +17,7 @@ namespace rtc {
 
 // Test memory set functions put values into memory in expected order.
 TEST(ByteOrderTest, TestSet) {
-  uint8 buf[8] = { 0u, 0u, 0u, 0u, 0u, 0u, 0u, 0u };
+  uint8_t buf[8] = {0u, 0u, 0u, 0u, 0u, 0u, 0u, 0u};
   Set8(buf, 0, 0xfb);
   Set8(buf, 1, 0x12);
   EXPECT_EQ(0xfb, buf[0]);
@@ -60,7 +60,7 @@ TEST(ByteOrderTest, TestSet) {
 
 // Test memory get functions get values from memory in expected order.
 TEST(ByteOrderTest, TestGet) {
-  uint8 buf[8];
+  uint8_t buf[8];
   buf[0] = 0x01u;
   buf[1] = 0x23u;
   buf[2] = 0x45u;
diff --git a/media/webrtc/trunk/webrtc/base/callback_unittest.cc b/media/webrtc/trunk/webrtc/base/callback_unittest.cc
index 66c939140e..db294cd96e 100644
--- a/media/webrtc/trunk/webrtc/base/callback_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/callback_unittest.cc
@@ -11,6 +11,8 @@
 #include "webrtc/base/bind.h"
 #include "webrtc/base/callback.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/refcount.h"
 
 namespace rtc {
 
@@ -26,6 +28,21 @@ struct BindTester {
   int b(int x) const { return x * x; }
 };
 
+class RefCountedBindTester : public RefCountInterface {
+ public:
+  RefCountedBindTester() : count_(0) {}
+  int AddRef() const override {
+    return ++count_;
+  }
+  int Release() const {
+    return --count_;
+  }
+  int RefCount() const { return count_; }
+
+ private:
+  mutable int count_;
+};
+
 }  // namespace
 
 TEST(CallbackTest, VoidReturn) {
@@ -78,4 +95,46 @@ TEST(CallbackTest, WithBind) {
   EXPECT_EQ(25, cb1());
 }
 
+TEST(KeepRefUntilDoneTest, simple) {
+  RefCountedBindTester t;
+  EXPECT_EQ(0, t.RefCount());
+  {
+    Callback0<void> cb = KeepRefUntilDone(&t);
+    EXPECT_EQ(1, t.RefCount());
+    cb();
+    EXPECT_EQ(1, t.RefCount());
+    cb();
+    EXPECT_EQ(1, t.RefCount());
+  }
+  EXPECT_EQ(0, t.RefCount());
+}
+
+TEST(KeepRefUntilDoneTest, copy) {
+  RefCountedBindTester t;
+  EXPECT_EQ(0, t.RefCount());
+  Callback0<void> cb2;
+  {
+    Callback0<void> cb = KeepRefUntilDone(&t);
+    EXPECT_EQ(1, t.RefCount());
+    cb2 = cb;
+  }
+  EXPECT_EQ(1, t.RefCount());
+  cb2 = Callback0<void>();
+  EXPECT_EQ(0, t.RefCount());
+}
+
+TEST(KeepRefUntilDoneTest, scopedref) {
+  RefCountedBindTester t;
+  EXPECT_EQ(0, t.RefCount());
+  {
+    scoped_refptr<RefCountedBindTester> t_scoped_ref(&t);
+    Callback0<void> cb = KeepRefUntilDone(t_scoped_ref);
+    t_scoped_ref = nullptr;
+    EXPECT_EQ(1, t.RefCount());
+    cb();
+    EXPECT_EQ(1, t.RefCount());
+  }
+  EXPECT_EQ(0, t.RefCount());
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/checks.cc b/media/webrtc/trunk/webrtc/base/checks.cc
index f64391d61b..9a14b31f6c 100644
--- a/media/webrtc/trunk/webrtc/base/checks.cc
+++ b/media/webrtc/trunk/webrtc/base/checks.cc
@@ -109,9 +109,6 @@ void FatalMessage::Init(const char* file, int line) {
           << file << ", line " << line << std::endl << "# ";
 }
 
-// Refer to comments in checks.h.
-#ifndef WEBRTC_CHROMIUM_BUILD
-
 // MSVC doesn't like complex extern templates and DLLs.
 #if !defined(COMPILER_MSVC)
 // Explicit instantiations for commonly used comparisons.
@@ -127,6 +124,4 @@ template std::string* MakeCheckOpString<std::string, std::string>(
     const std::string&, const std::string&, const char* name);
 #endif
 
-#endif  // WEBRTC_CHROMIUM_BUILD
-
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/checks.h b/media/webrtc/trunk/webrtc/base/checks.h
index ea3fb37c0d..681361a3d2 100644
--- a/media/webrtc/trunk/webrtc/base/checks.h
+++ b/media/webrtc/trunk/webrtc/base/checks.h
@@ -14,61 +14,53 @@
 #include <sstream>
 #include <string>
 
-#ifdef WEBRTC_CHROMIUM_BUILD
-// Include logging.h in a Chromium build to enable the overrides mechanism for
-// using Chromium's macros. Otherwise, don't depend on logging.h.
-// TODO(ajm): Ideally, checks.h would be combined with logging.h, but
-// consolidation with system_wrappers/logging.h should happen first.
-#include "webrtc/base/logging.h"
-#endif
 #include "webrtc/typedefs.h"
 
 // The macros here print a message to stderr and abort under various
 // conditions. All will accept additional stream messages. For example:
-// DCHECK_EQ(foo, bar) << "I'm printed when foo != bar.";
+// RTC_DCHECK_EQ(foo, bar) << "I'm printed when foo != bar.";
 //
-// - CHECK(x) is an assertion that x is always true, and that if it isn't, it's
-//   better to terminate the process than to continue. During development, the
-//   reason that it's better to terminate might simply be that the error
+// - RTC_CHECK(x) is an assertion that x is always true, and that if it isn't,
+//   it's better to terminate the process than to continue. During development,
+//   the reason that it's better to terminate might simply be that the error
 //   handling code isn't in place yet; in production, the reason might be that
 //   the author of the code truly believes that x will always be true, but that
 //   she recognizes that if she is wrong, abrupt and unpleasant process
 //   termination is still better than carrying on with the assumption violated.
 //
-//   CHECK always evaluates its argument, so it's OK for x to have side
+//   RTC_CHECK always evaluates its argument, so it's OK for x to have side
 //   effects.
 //
-// - DCHECK(x) is the same as CHECK(x)---an assertion that x is always
+// - RTC_DCHECK(x) is the same as RTC_CHECK(x)---an assertion that x is always
 //   true---except that x will only be evaluated in debug builds; in production
 //   builds, x is simply assumed to be true. This is useful if evaluating x is
 //   expensive and the expected cost of failing to detect the violated
 //   assumption is acceptable. You should not handle cases where a production
 //   build fails to spot a violated condition, even those that would result in
 //   crashes. If the code needs to cope with the error, make it cope, but don't
-//   call DCHECK; if the condition really can't occur, but you'd sleep better
-//   at night knowing that the process will suicide instead of carrying on in
-//   case you were wrong, use CHECK instead of DCHECK.
+//   call RTC_DCHECK; if the condition really can't occur, but you'd sleep
+//   better at night knowing that the process will suicide instead of carrying
+//   on in case you were wrong, use RTC_CHECK instead of RTC_DCHECK.
 //
-//   DCHECK only evaluates its argument in debug builds, so if x has visible
+//   RTC_DCHECK only evaluates its argument in debug builds, so if x has visible
 //   side effects, you need to write e.g.
-//     bool w = x; DCHECK(w);
+//     bool w = x; RTC_DCHECK(w);
 //
-// - CHECK_EQ, _NE, _GT, ..., and DCHECK_EQ, _NE, _GT, ... are specialized
-//   variants of CHECK and DCHECK that print prettier messages if the condition
-//   doesn't hold. Prefer them to raw CHECK and DCHECK.
+// - RTC_CHECK_EQ, _NE, _GT, ..., and RTC_DCHECK_EQ, _NE, _GT, ... are
+//   specialized variants of RTC_CHECK and RTC_DCHECK that print prettier
+//   messages if the condition doesn't hold. Prefer them to raw RTC_CHECK and
+//   RTC_DCHECK.
 //
 // - FATAL() aborts unconditionally.
+//
+// TODO(ajm): Ideally, checks.h would be combined with logging.h, but
+// consolidation with system_wrappers/logging.h should happen first.
 
 namespace rtc {
 
-// The use of overrides/webrtc/base/logging.h in a Chromium build results in
-// redefined macro errors. Fortunately, Chromium's macros can be used as drop-in
-// replacements for the standalone versions.
-#ifndef WEBRTC_CHROMIUM_BUILD
-
 // Helper macro which avoids evaluating the arguments to a stream if
 // the condition doesn't hold.
-#define LAZY_STREAM(stream, condition)                                        \
+#define RTC_LAZY_STREAM(stream, condition)                                    \
   !(condition) ? static_cast<void>(0) : rtc::FatalMessageVoidify() & (stream)
 
 // The actual stream used isn't important. We reference condition in the code
@@ -76,32 +68,30 @@ namespace rtc {
 // in a particularly convoluted way with an extra ?: because that appears to be
 // the simplest construct that keeps Visual Studio from complaining about
 // condition being unused).
-#define EAT_STREAM_PARAMETERS(condition) \
-  (true ? true : !(condition))           \
-      ? static_cast<void>(0)             \
+#define RTC_EAT_STREAM_PARAMETERS(condition) \
+  (true ? true : !(condition))               \
+      ? static_cast<void>(0)                 \
       : rtc::FatalMessageVoidify() & rtc::FatalMessage("", 0).stream()
 
-// CHECK dies with a fatal error if condition is not true.  It is *not*
+// RTC_CHECK dies with a fatal error if condition is not true.  It is *not*
 // controlled by NDEBUG, so the check will be executed regardless of
 // compilation mode.
 //
-// We make sure CHECK et al. always evaluates their arguments, as
-// doing CHECK(FunctionWithSideEffect()) is a common idiom.
-#define CHECK(condition)                                                    \
-  LAZY_STREAM(rtc::FatalMessage(__FILE__, __LINE__).stream(), !(condition)) \
-  << "Check failed: " #condition << std::endl << "# "
-
-#define RTC_CHECK(condition) CHECK(condition)
+// We make sure RTC_CHECK et al. always evaluates their arguments, as
+// doing RTC_CHECK(FunctionWithSideEffect()) is a common idiom.
+#define RTC_CHECK(condition)                                      \
+  RTC_LAZY_STREAM(rtc::FatalMessage(__FILE__, __LINE__).stream(), \
+                  !(condition))                                   \
+      << "Check failed: " #condition << std::endl << "# "
 
 // Helper macro for binary operators.
-// Don't use this macro directly in your code, use CHECK_EQ et al below.
+// Don't use this macro directly in your code, use RTC_CHECK_EQ et al below.
 //
 // TODO(akalin): Rewrite this so that constructs like if (...)
-// CHECK_EQ(...) else { ... } work properly.
-#define CHECK_OP(name, op, val1, val2)                      \
-  if (std::string* _result =                                \
-      rtc::Check##name##Impl((val1), (val2),                \
-                             #val1 " " #op " " #val2))      \
+// RTC_CHECK_EQ(...) else { ... } work properly.
+#define RTC_CHECK_OP(name, op, val1, val2)                                 \
+  if (std::string* _result =                                               \
+          rtc::Check##name##Impl((val1), (val2), #val1 " " #op " " #val2)) \
     rtc::FatalMessage(__FILE__, __LINE__, _result).stream()
 
 // Build the error message string.  This is separate from the "Impl"
@@ -136,85 +126,61 @@ std::string* MakeCheckOpString<std::string, std::string>(
     const std::string&, const std::string&, const char* name);
 #endif
 
-// Helper functions for CHECK_OP macro.
+// Helper functions for RTC_CHECK_OP macro.
 // The (int, int) specialization works around the issue that the compiler
 // will not instantiate the template version of the function on values of
 // unnamed enum type - see comment below.
-#define DEFINE_CHECK_OP_IMPL(name, op) \
-  template <class t1, class t2> \
-  inline std::string* Check##name##Impl(const t1& v1, const t2& v2, \
-                                        const char* names) { \
-    if (v1 op v2) return NULL; \
-    else return rtc::MakeCheckOpString(v1, v2, names); \
-  } \
+#define DEFINE_RTC_CHECK_OP_IMPL(name, op)                                   \
+  template <class t1, class t2>                                              \
+  inline std::string* Check##name##Impl(const t1& v1, const t2& v2,          \
+                                        const char* names) {                 \
+    if (v1 op v2)                                                            \
+      return NULL;                                                           \
+    else                                                                     \
+      return rtc::MakeCheckOpString(v1, v2, names);                          \
+  }                                                                          \
   inline std::string* Check##name##Impl(int v1, int v2, const char* names) { \
-    if (v1 op v2) return NULL; \
-    else return rtc::MakeCheckOpString(v1, v2, names); \
+    if (v1 op v2)                                                            \
+      return NULL;                                                           \
+    else                                                                     \
+      return rtc::MakeCheckOpString(v1, v2, names);                          \
   }
-DEFINE_CHECK_OP_IMPL(EQ, ==)
-DEFINE_CHECK_OP_IMPL(NE, !=)
-DEFINE_CHECK_OP_IMPL(LE, <=)
-DEFINE_CHECK_OP_IMPL(LT, < )
-DEFINE_CHECK_OP_IMPL(GE, >=)
-DEFINE_CHECK_OP_IMPL(GT, > )
-#undef DEFINE_CHECK_OP_IMPL
+DEFINE_RTC_CHECK_OP_IMPL(EQ, ==)
+DEFINE_RTC_CHECK_OP_IMPL(NE, !=)
+DEFINE_RTC_CHECK_OP_IMPL(LE, <=)
+DEFINE_RTC_CHECK_OP_IMPL(LT, < )
+DEFINE_RTC_CHECK_OP_IMPL(GE, >=)
+DEFINE_RTC_CHECK_OP_IMPL(GT, > )
+#undef DEFINE_RTC_CHECK_OP_IMPL
 
-#define CHECK_EQ(val1, val2) CHECK_OP(EQ, ==, val1, val2)
-#define CHECK_NE(val1, val2) CHECK_OP(NE, !=, val1, val2)
-#define CHECK_LE(val1, val2) CHECK_OP(LE, <=, val1, val2)
-#define CHECK_LT(val1, val2) CHECK_OP(LT, < , val1, val2)
-#define CHECK_GE(val1, val2) CHECK_OP(GE, >=, val1, val2)
-#define CHECK_GT(val1, val2) CHECK_OP(GT, > , val1, val2)
-
-// The DCHECK macro is equivalent to CHECK except that it only generates code
-// in debug builds. It does reference the condition parameter in all cases,
-// though, so callers won't risk getting warnings about unused variables.
-#if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
-#define DCHECK(condition) CHECK(condition)
-#define DCHECK_EQ(v1, v2) CHECK_EQ(v1, v2)
-#define DCHECK_NE(v1, v2) CHECK_NE(v1, v2)
-#define DCHECK_LE(v1, v2) CHECK_LE(v1, v2)
-#define DCHECK_LT(v1, v2) CHECK_LT(v1, v2)
-#define DCHECK_GE(v1, v2) CHECK_GE(v1, v2)
-#define DCHECK_GT(v1, v2) CHECK_GT(v1, v2)
-#else
-#define DCHECK(condition) EAT_STREAM_PARAMETERS(condition)
-#define DCHECK_EQ(v1, v2) EAT_STREAM_PARAMETERS((v1) == (v2))
-#define DCHECK_NE(v1, v2) EAT_STREAM_PARAMETERS((v1) != (v2))
-#define DCHECK_LE(v1, v2) EAT_STREAM_PARAMETERS((v1) <= (v2))
-#define DCHECK_LT(v1, v2) EAT_STREAM_PARAMETERS((v1) < (v2))
-#define DCHECK_GE(v1, v2) EAT_STREAM_PARAMETERS((v1) >= (v2))
-#define DCHECK_GT(v1, v2) EAT_STREAM_PARAMETERS((v1) > (v2))
-#endif
-
-#define RTC_CHECK_EQ(val1, val2) CHECK_OP(EQ, ==, val1, val2)
-#define RTC_CHECK_NE(val1, val2) CHECK_OP(NE, !=, val1, val2)
-#define RTC_CHECK_LE(val1, val2) CHECK_OP(LE, <=, val1, val2)
-#define RTC_CHECK_LT(val1, val2) CHECK_OP(LT, < , val1, val2)
-#define RTC_CHECK_GE(val1, val2) CHECK_OP(GE, >=, val1, val2)
-#define RTC_CHECK_GT(val1, val2) CHECK_OP(GT, > , val1, val2)
+#define RTC_CHECK_EQ(val1, val2) RTC_CHECK_OP(EQ, ==, val1, val2)
+#define RTC_CHECK_NE(val1, val2) RTC_CHECK_OP(NE, !=, val1, val2)
+#define RTC_CHECK_LE(val1, val2) RTC_CHECK_OP(LE, <=, val1, val2)
+#define RTC_CHECK_LT(val1, val2) RTC_CHECK_OP(LT, < , val1, val2)
+#define RTC_CHECK_GE(val1, val2) RTC_CHECK_OP(GE, >=, val1, val2)
+#define RTC_CHECK_GT(val1, val2) RTC_CHECK_OP(GT, > , val1, val2)
 
 // The RTC_DCHECK macro is equivalent to RTC_CHECK except that it only generates
 // code in debug builds. It does reference the condition parameter in all cases,
 // though, so callers won't risk getting warnings about unused variables.
 #if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
 #define RTC_DCHECK_IS_ON 1
-#define RTC_DCHECK(condition) CHECK(condition)
-#define RTC_DCHECK_EQ(v1, v2) CHECK_EQ(v1, v2)
-#define RTC_DCHECK_NE(v1, v2) CHECK_NE(v1, v2)
-#define RTC_DCHECK_LE(v1, v2) CHECK_LE(v1, v2)
-#define RTC_DCHECK_LT(v1, v2) CHECK_LT(v1, v2)
-#define RTC_DCHECK_GE(v1, v2) CHECK_GE(v1, v2)
-#define RTC_DCHECK_GT(v1, v2) CHECK_GT(v1, v2)
+#define RTC_DCHECK(condition) RTC_CHECK(condition)
+#define RTC_DCHECK_EQ(v1, v2) RTC_CHECK_EQ(v1, v2)
+#define RTC_DCHECK_NE(v1, v2) RTC_CHECK_NE(v1, v2)
+#define RTC_DCHECK_LE(v1, v2) RTC_CHECK_LE(v1, v2)
+#define RTC_DCHECK_LT(v1, v2) RTC_CHECK_LT(v1, v2)
+#define RTC_DCHECK_GE(v1, v2) RTC_CHECK_GE(v1, v2)
+#define RTC_DCHECK_GT(v1, v2) RTC_CHECK_GT(v1, v2)
 #else
 #define RTC_DCHECK_IS_ON 0
-#define RTC_DCHECK(condition) EAT_STREAM_PARAMETERS(condition)
-#define RTC_DCHECK_EQ(v1, v2) EAT_STREAM_PARAMETERS((v1) == (v2))
-#define RTC_DCHECK_NE(v1, v2) EAT_STREAM_PARAMETERS((v1) != (v2))
-#define RTC_DCHECK_LE(v1, v2) EAT_STREAM_PARAMETERS((v1) <= (v2))
-#define RTC_DCHECK_LT(v1, v2) EAT_STREAM_PARAMETERS((v1) < (v2))
-#define RTC_DCHECK_GE(v1, v2) EAT_STREAM_PARAMETERS((v1) >= (v2))
-#define RTC_DCHECK_GT(v1, v2) EAT_STREAM_PARAMETERS((v1) > (v2))
+#define RTC_DCHECK(condition) RTC_EAT_STREAM_PARAMETERS(condition)
+#define RTC_DCHECK_EQ(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) == (v2))
+#define RTC_DCHECK_NE(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) != (v2))
+#define RTC_DCHECK_LE(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) <= (v2))
+#define RTC_DCHECK_LT(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) < (v2))
+#define RTC_DCHECK_GE(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) >= (v2))
+#define RTC_DCHECK_GT(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) > (v2))
 #endif
 
 // This is identical to LogMessageVoidify but in name.
@@ -226,13 +192,11 @@ class FatalMessageVoidify {
   void operator&(std::ostream&) { }
 };
 
-#endif  // WEBRTC_CHROMIUM_BUILD
-
 #define RTC_UNREACHABLE_CODE_HIT false
-#define RTC_NOTREACHED() DCHECK(RTC_UNREACHABLE_CODE_HIT)
+#define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
 
 #define FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
-// TODO(ajm): Consider adding NOTIMPLEMENTED and NOTREACHED macros when
+// TODO(ajm): Consider adding RTC_NOTIMPLEMENTED macro when
 // base/logging.h and system_wrappers/logging.h are consolidated such that we
 // can match the Chromium behavior.
 
@@ -240,7 +204,7 @@ class FatalMessageVoidify {
 class FatalMessage {
  public:
   FatalMessage(const char* file, int line);
-  // Used for CHECK_EQ(), etc. Takes ownership of the given string.
+  // Used for RTC_CHECK_EQ(), etc. Takes ownership of the given string.
   FatalMessage(const char* file, int line, std::string* result);
   NO_RETURN ~FatalMessage();
 
@@ -256,7 +220,7 @@ class FatalMessage {
 // remainder is zero.
 template <typename T>
 inline T CheckedDivExact(T a, T b) {
-  CHECK_EQ(a % b, static_cast<T>(0));
+  RTC_CHECK_EQ(a % b, static_cast<T>(0));
   return a / b;
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/common.h b/media/webrtc/trunk/webrtc/base/common.h
index e39b75daf4..1b1dac64b0 100644
--- a/media/webrtc/trunk/webrtc/base/common.h
+++ b/media/webrtc/trunk/webrtc/base/common.h
@@ -54,14 +54,16 @@ inline void RtcUnused(const void*) {}
 
 #endif  // !defined(WEBRTC_WIN)
 
-#define ARRAY_SIZE(x) (static_cast<int>(sizeof(x) / sizeof(x[0])))
-
 /////////////////////////////////////////////////////////////////////////////
 // Assertions
 /////////////////////////////////////////////////////////////////////////////
 
 #ifndef ENABLE_DEBUG
-#define ENABLE_DEBUG _DEBUG
+#if !defined(NDEBUG)
+#define ENABLE_DEBUG 1
+#else
+#define ENABLE_DEBUG 0
+#endif
 #endif  // !defined(ENABLE_DEBUG)
 
 // Even for release builds, allow for the override of LogAssert. Though no
@@ -176,7 +178,7 @@ inline bool ImplicitCastToBool(bool result) { return result; }
 
 // Forces compiler to inline, even against its better judgement. Use wisely.
 #if defined(__GNUC__)
-#define FORCE_INLINE __attribute__((always_inline))
+#define FORCE_INLINE __attribute__ ((__always_inline__))
 #elif defined(WEBRTC_WIN)
 #define FORCE_INLINE __forceinline
 #else
@@ -190,8 +192,8 @@ inline bool ImplicitCastToBool(bool result) { return result; }
 // TODO(ajm): Hack to avoid multiple definitions until the base/ of webrtc and
 // libjingle are merged.
 #if !defined(WARN_UNUSED_RESULT)
-#if defined(__GNUC__)
-#define WARN_UNUSED_RESULT __attribute__((warn_unused_result))
+#if defined(__GNUC__) || defined(__clang__)
+#define WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
 #else
 #define WARN_UNUSED_RESULT
 #endif
diff --git a/media/webrtc/trunk/webrtc/base/constructormagic.h b/media/webrtc/trunk/webrtc/base/constructormagic.h
index 972508b4ca..6ef7826505 100644
--- a/media/webrtc/trunk/webrtc/base/constructormagic.h
+++ b/media/webrtc/trunk/webrtc/base/constructormagic.h
@@ -11,43 +11,24 @@
 #ifndef WEBRTC_BASE_CONSTRUCTORMAGIC_H_
 #define WEBRTC_BASE_CONSTRUCTORMAGIC_H_
 
-// Undefine macros first, just in case. Some third-party includes have their own
-// version.
-
-#undef DISALLOW_ASSIGN
-#define DISALLOW_ASSIGN(TypeName) \
-  void operator=(const TypeName&)
+// Put this in the declarations for a class to be unassignable.
 #define RTC_DISALLOW_ASSIGN(TypeName) \
   void operator=(const TypeName&) = delete
 
-// A macro to disallow the evil copy constructor and operator= functions
-// This should be used in the private: declarations for a class.
-#undef DISALLOW_COPY_AND_ASSIGN
-#define DISALLOW_COPY_AND_ASSIGN(TypeName)    \
-  TypeName(const TypeName&);                    \
-  DISALLOW_ASSIGN(TypeName)
+// A macro to disallow the copy constructor and operator= functions. This should
+// be used in the declarations for a class.
 #define RTC_DISALLOW_COPY_AND_ASSIGN(TypeName) \
   TypeName(const TypeName&) = delete;          \
   RTC_DISALLOW_ASSIGN(TypeName)
 
-// Alternative, less-accurate legacy name.
-#undef DISALLOW_EVIL_CONSTRUCTORS
-#define DISALLOW_EVIL_CONSTRUCTORS(TypeName) \
-  DISALLOW_COPY_AND_ASSIGN(TypeName)
-
-// A macro to disallow all the implicit constructors, namely the
-// default constructor, copy constructor and operator= functions.
+// A macro to disallow all the implicit constructors, namely the default
+// constructor, copy constructor and operator= functions.
 //
-// This should be used in the private: declarations for a class
-// that wants to prevent anyone from instantiating it. This is
-// especially useful for classes containing only static methods.
-#undef DISALLOW_IMPLICIT_CONSTRUCTORS
-#define DISALLOW_IMPLICIT_CONSTRUCTORS(TypeName) \
-  TypeName();                                    \
-  DISALLOW_EVIL_CONSTRUCTORS(TypeName)
+// This should be used in the declarations for a class that wants to prevent
+// anyone from instantiating it. This is especially useful for classes
+// containing only static methods.
 #define RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(TypeName) \
   TypeName() = delete;                               \
   RTC_DISALLOW_COPY_AND_ASSIGN(TypeName)
 
-
 #endif  // WEBRTC_BASE_CONSTRUCTORMAGIC_H_
diff --git a/media/webrtc/trunk/webrtc/base/cpumonitor.cc b/media/webrtc/trunk/webrtc/base/cpumonitor.cc
deleted file mode 100644
index c881b48c5a..0000000000
--- a/media/webrtc/trunk/webrtc/base/cpumonitor.cc
+++ /dev/null
@@ -1,423 +0,0 @@
-/*
- *  Copyright 2010 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/cpumonitor.h"
-
-#include <string>
-
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/systeminfo.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/timeutils.h"
-
-#if defined(WEBRTC_WIN)
-#include "webrtc/base/win32.h"
-#include <winternl.h>
-#endif
-
-#if defined(WEBRTC_POSIX)
-#include <sys/time.h>
-#endif
-
-#if defined(WEBRTC_MAC)
-#include <mach/mach_host.h>
-#include <mach/mach_init.h>
-#include <mach/mach_port.h>
-#include <mach/host_info.h>
-#include <mach/task.h>
-#endif  // defined(WEBRTC_MAC)
-
-#if defined(WEBRTC_LINUX)
-#include <sys/resource.h>
-#include <errno.h>
-#include <stdio.h>
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/pathutils.h"
-#endif // defined(WEBRTC_LINUX)
-
-#if defined(WEBRTC_MAC)
-static uint64 TimeValueTToInt64(const time_value_t &time_value) {
-  return rtc::kNumMicrosecsPerSec * time_value.seconds +
-      time_value.microseconds;
-}
-#endif  // defined(WEBRTC_MAC)
-
-// How CpuSampler works
-// When threads switch, the time they spent is accumulated to system counters.
-// The time can be treated as user, kernel or idle.
-// user time is applications.
-// kernel time is the OS, including the thread switching code itself.
-//   typically kernel time indicates IO.
-// idle time is a process that wastes time when nothing is ready to run.
-//
-// User time is broken down by process (application).  One of the applications
-// is the current process.  When you add up all application times, this is
-// system time.  If only your application is running, system time should be the
-// same as process time.
-//
-// All cores contribute to these accumulators.  A dual core process is able to
-// process twice as many cycles as a single core.  The actual code efficiency
-// may be worse, due to contention, but the available cycles is exactly twice
-// as many, and the cpu load will reflect the efficiency.  Hyperthreads behave
-// the same way.  The load will reflect 200%, but the actual amount of work
-// completed will be much less than a true dual core.
-//
-// Total available performance is the sum of all accumulators.
-// If you tracked this for 1 second, it would essentially give you the clock
-// rate - number of cycles per second.
-// Speed step / Turbo Boost is not considered, so infact more processing time
-// may be available.
-
-namespace rtc {
-
-// Note Tests on Windows show 600 ms is minimum stable interval for Windows 7.
-static const int32 kDefaultInterval = 950;  // Slightly under 1 second.
-
-CpuSampler::CpuSampler()
-    : min_load_interval_(kDefaultInterval)
-#if defined(WEBRTC_WIN)
-      , get_system_times_(NULL),
-      nt_query_system_information_(NULL),
-      force_fallback_(false)
-#endif
-    {
-}
-
-CpuSampler::~CpuSampler() {
-}
-
-// Set minimum interval in ms between computing new load values. Default 950.
-void CpuSampler::set_load_interval(int min_load_interval) {
-  min_load_interval_ = min_load_interval;
-}
-
-bool CpuSampler::Init() {
-  sysinfo_.reset(new SystemInfo);
-  cpus_ = sysinfo_->GetMaxCpus();
-  if (cpus_ == 0) {
-    return false;
-  }
-#if defined(WEBRTC_WIN)
-  // Note that GetSystemTimes is available in Windows XP SP1 or later.
-  // http://msdn.microsoft.com/en-us/library/ms724400.aspx
-  // NtQuerySystemInformation is used as a fallback.
-  if (!force_fallback_) {
-    get_system_times_ = GetProcAddress(GetModuleHandle(L"kernel32.dll"),
-        "GetSystemTimes");
-  }
-  nt_query_system_information_ = GetProcAddress(GetModuleHandle(L"ntdll.dll"),
-      "NtQuerySystemInformation");
-  if ((get_system_times_ == NULL) && (nt_query_system_information_ == NULL)) {
-    return false;
-  }
-#endif
-#if defined(WEBRTC_LINUX)
-  Pathname sname("/proc/stat");
-  sfile_.reset(Filesystem::OpenFile(sname, "rb"));
-  if (!sfile_) {
-    LOG_ERR(LS_ERROR) << "open proc/stat failed:";
-    return false;
-  }
-  if (!sfile_->DisableBuffering()) {
-    LOG_ERR(LS_ERROR) << "could not disable buffering for proc/stat";
-    return false;
-  }
-#endif // defined(WEBRTC_LINUX)
-  GetProcessLoad();  // Initialize values.
-  GetSystemLoad();
-  // Help next user call return valid data by recomputing load.
-  process_.prev_load_time_ = 0u;
-  system_.prev_load_time_ = 0u;
-  return true;
-}
-
-float CpuSampler::UpdateCpuLoad(uint64 current_total_times,
-                                uint64 current_cpu_times,
-                                uint64 *prev_total_times,
-                                uint64 *prev_cpu_times) {
-  float result = 0.f;
-  if (current_total_times < *prev_total_times ||
-      current_cpu_times < *prev_cpu_times) {
-    LOG(LS_ERROR) << "Inconsistent time values are passed. ignored";
-  } else {
-    const uint64 cpu_diff = current_cpu_times - *prev_cpu_times;
-    const uint64 total_diff = current_total_times - *prev_total_times;
-    result = (total_diff == 0ULL ? 0.f :
-              static_cast<float>(1.0f * cpu_diff / total_diff));
-    if (result > static_cast<float>(cpus_)) {
-      result = static_cast<float>(cpus_);
-    }
-    *prev_total_times = current_total_times;
-    *prev_cpu_times = current_cpu_times;
-  }
-  return result;
-}
-
-float CpuSampler::GetSystemLoad() {
-  uint32 timenow = Time();
-  int elapsed = static_cast<int>(TimeDiff(timenow, system_.prev_load_time_));
-  if (min_load_interval_ != 0 && system_.prev_load_time_ != 0u &&
-      elapsed < min_load_interval_) {
-    return system_.prev_load_;
-  }
-#if defined(WEBRTC_WIN)
-  uint64 total_times, cpu_times;
-
-  typedef BOOL (_stdcall *GST_PROC)(LPFILETIME, LPFILETIME, LPFILETIME);
-  typedef NTSTATUS (WINAPI *QSI_PROC)(SYSTEM_INFORMATION_CLASS,
-      PVOID, ULONG, PULONG);
-
-  GST_PROC get_system_times = reinterpret_cast<GST_PROC>(get_system_times_);
-  QSI_PROC nt_query_system_information = reinterpret_cast<QSI_PROC>(
-      nt_query_system_information_);
-
-  if (get_system_times) {
-    FILETIME idle_time, kernel_time, user_time;
-    if (!get_system_times(&idle_time, &kernel_time, &user_time)) {
-      LOG(LS_ERROR) << "::GetSystemTimes() failed: " << ::GetLastError();
-      return 0.f;
-    }
-    // kernel_time includes Kernel idle time, so no need to
-    // include cpu_time as total_times
-    total_times = ToUInt64(kernel_time) + ToUInt64(user_time);
-    cpu_times = total_times - ToUInt64(idle_time);
-
-  } else {
-    if (nt_query_system_information) {
-      ULONG returned_length = 0;
-      scoped_ptr<SYSTEM_PROCESSOR_PERFORMANCE_INFORMATION[]> processor_info(
-          new SYSTEM_PROCESSOR_PERFORMANCE_INFORMATION[cpus_]);
-      nt_query_system_information(
-          ::SystemProcessorPerformanceInformation,
-          reinterpret_cast<void*>(processor_info.get()),
-          cpus_ * sizeof(SYSTEM_PROCESSOR_PERFORMANCE_INFORMATION),
-          &returned_length);
-
-      if (returned_length !=
-          (cpus_ * sizeof(SYSTEM_PROCESSOR_PERFORMANCE_INFORMATION))) {
-        LOG(LS_ERROR) << "NtQuerySystemInformation has unexpected size";
-        return 0.f;
-      }
-
-      uint64 current_idle = 0;
-      uint64 current_kernel = 0;
-      uint64 current_user = 0;
-      for (int ix = 0; ix < cpus_; ++ix) {
-        current_idle += processor_info[ix].IdleTime.QuadPart;
-        current_kernel += processor_info[ix].UserTime.QuadPart;
-        current_user += processor_info[ix].KernelTime.QuadPart;
-      }
-      total_times = current_kernel + current_user;
-      cpu_times = total_times - current_idle;
-    } else {
-      return 0.f;
-    }
-  }
-#endif  // WEBRTC_WIN 
-
-#if defined(WEBRTC_MAC)
-  mach_port_t mach_host = mach_host_self();
-  host_cpu_load_info_data_t cpu_info;
-  mach_msg_type_number_t info_count = HOST_CPU_LOAD_INFO_COUNT;
-  kern_return_t kr = host_statistics(mach_host, HOST_CPU_LOAD_INFO,
-                                     reinterpret_cast<host_info_t>(&cpu_info),
-                                     &info_count);
-  mach_port_deallocate(mach_task_self(), mach_host);
-  if (KERN_SUCCESS != kr) {
-    LOG(LS_ERROR) << "::host_statistics() failed";
-    return 0.f;
-  }
-
-  const uint64 cpu_times = cpu_info.cpu_ticks[CPU_STATE_NICE] +
-      cpu_info.cpu_ticks[CPU_STATE_SYSTEM] +
-      cpu_info.cpu_ticks[CPU_STATE_USER];
-  const uint64 total_times = cpu_times + cpu_info.cpu_ticks[CPU_STATE_IDLE];
-#endif  // defined(WEBRTC_MAC)
-
-#if defined(WEBRTC_LINUX)
-  if (!sfile_) {
-    LOG(LS_ERROR) << "Invalid handle for proc/stat";
-    return 0.f;
-  }
-  std::string statbuf;
-  sfile_->SetPosition(0);
-  if (!sfile_->ReadLine(&statbuf)) {
-    LOG_ERR(LS_ERROR) << "Could not read proc/stat file";
-    return 0.f;
-  }
-
-  unsigned long long user;
-  unsigned long long nice;
-  unsigned long long system;
-  unsigned long long idle;
-  if (sscanf(statbuf.c_str(), "cpu %Lu %Lu %Lu %Lu",
-             &user, &nice,
-             &system, &idle) != 4) {
-    LOG_ERR(LS_ERROR) << "Could not parse cpu info";
-    return 0.f;
-  }
-  const uint64 cpu_times = nice + system + user;
-  const uint64 total_times = cpu_times + idle;
-#endif  // defined(WEBRTC_LINUX)
-
-#if defined(__native_client__)
-  // TODO(ryanpetrie): Implement this via PPAPI when it's available.
-  const uint64 cpu_times = 0;
-  const uint64 total_times = 0;
-#endif  // defined(__native_client__)
-
-  system_.prev_load_time_ = timenow;
-  system_.prev_load_ = UpdateCpuLoad(total_times,
-                                     cpu_times * cpus_,
-                                     &system_.prev_total_times_,
-                                     &system_.prev_cpu_times_);
-  return system_.prev_load_;
-}
-
-float CpuSampler::GetProcessLoad() {
-  uint32 timenow = Time();
-  int elapsed = static_cast<int>(TimeDiff(timenow, process_.prev_load_time_));
-  if (min_load_interval_ != 0 && process_.prev_load_time_ != 0u &&
-      elapsed < min_load_interval_) {
-    return process_.prev_load_;
-  }
-#if defined(WEBRTC_WIN)
-  FILETIME current_file_time;
-  ::GetSystemTimeAsFileTime(&current_file_time);
-
-  FILETIME create_time, exit_time, kernel_time, user_time;
-  if (!::GetProcessTimes(::GetCurrentProcess(),
-                         &create_time, &exit_time, &kernel_time, &user_time)) {
-    LOG(LS_ERROR) << "::GetProcessTimes() failed: " << ::GetLastError();
-    return 0.f;
-  }
-
-  const uint64 total_times =
-      ToUInt64(current_file_time) - ToUInt64(create_time);
-  const uint64 cpu_times =
-      (ToUInt64(kernel_time) + ToUInt64(user_time));
-#endif  // WEBRTC_WIN 
-
-#if defined(WEBRTC_POSIX)
-  // Common to both OSX and Linux.
-  struct timeval tv;
-  gettimeofday(&tv, NULL);
-  const uint64 total_times = tv.tv_sec * kNumMicrosecsPerSec + tv.tv_usec;
-#endif
-
-#if defined(WEBRTC_MAC)
-  // Get live thread usage.
-  task_thread_times_info task_times_info;
-  mach_msg_type_number_t info_count = TASK_THREAD_TIMES_INFO_COUNT;
-
-  if (KERN_SUCCESS != task_info(mach_task_self(), TASK_THREAD_TIMES_INFO,
-                                reinterpret_cast<task_info_t>(&task_times_info),
-                                &info_count)) {
-    LOG(LS_ERROR) << "::task_info(TASK_THREAD_TIMES_INFO) failed";
-    return 0.f;
-  }
-
-  // Get terminated thread usage.
-  task_basic_info task_term_info;
-  info_count = TASK_BASIC_INFO_COUNT;
-  if (KERN_SUCCESS != task_info(mach_task_self(), TASK_BASIC_INFO,
-                                reinterpret_cast<task_info_t>(&task_term_info),
-                                &info_count)) {
-    LOG(LS_ERROR) << "::task_info(TASK_BASIC_INFO) failed";
-    return 0.f;
-  }
-
-  const uint64 cpu_times = (TimeValueTToInt64(task_times_info.user_time) +
-      TimeValueTToInt64(task_times_info.system_time) +
-      TimeValueTToInt64(task_term_info.user_time) +
-      TimeValueTToInt64(task_term_info.system_time));
-#endif  // defined(WEBRTC_MAC)
-
-#if defined(WEBRTC_LINUX)
-  rusage usage;
-  if (getrusage(RUSAGE_SELF, &usage) < 0) {
-    LOG_ERR(LS_ERROR) << "getrusage failed";
-    return 0.f;
-  }
-
-  const uint64 cpu_times =
-      (usage.ru_utime.tv_sec + usage.ru_stime.tv_sec) * kNumMicrosecsPerSec +
-      usage.ru_utime.tv_usec + usage.ru_stime.tv_usec;
-#endif  // defined(WEBRTC_LINUX)
-
-#if defined(__native_client__)
-  // TODO(ryanpetrie): Implement this via PPAPI when it's available.
-  const uint64 cpu_times = 0;
-#endif  // defined(__native_client__)
-
-  process_.prev_load_time_ = timenow;
-  process_.prev_load_ = UpdateCpuLoad(total_times,
-                                     cpu_times,
-                                     &process_.prev_total_times_,
-                                     &process_.prev_cpu_times_);
-  return process_.prev_load_;
-}
-
-int CpuSampler::GetMaxCpus() const {
-  return cpus_;
-}
-
-int CpuSampler::GetCurrentCpus() {
-  return sysinfo_->GetCurCpus();
-}
-
-///////////////////////////////////////////////////////////////////
-// Implementation of class CpuMonitor.
-CpuMonitor::CpuMonitor(Thread* thread)
-    : monitor_thread_(thread) {
-}
-
-CpuMonitor::~CpuMonitor() {
-  Stop();
-}
-
-void CpuMonitor::set_thread(Thread* thread) {
-  ASSERT(monitor_thread_ == NULL || monitor_thread_ == thread);
-  monitor_thread_ = thread;
-}
-
-bool CpuMonitor::Start(int period_ms) {
-  if (!monitor_thread_  || !sampler_.Init()) return false;
-
-  monitor_thread_->SignalQueueDestroyed.connect(
-       this, &CpuMonitor::OnMessageQueueDestroyed);
-
-  period_ms_ = period_ms;
-  monitor_thread_->PostDelayed(period_ms_, this);
-
-  return true;
-}
-
-void CpuMonitor::Stop() {
-  if (monitor_thread_) {
-    monitor_thread_->Clear(this);
-  }
-}
-
-void CpuMonitor::OnMessage(Message* msg) {
-  int max_cpus = sampler_.GetMaxCpus();
-  int current_cpus = sampler_.GetCurrentCpus();
-  float process_load = sampler_.GetProcessLoad();
-  float system_load = sampler_.GetSystemLoad();
-  SignalUpdate(current_cpus, max_cpus, process_load, system_load);
-
-  if (monitor_thread_) {
-    monitor_thread_->PostDelayed(period_ms_, this);
-  }
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/cpumonitor.h b/media/webrtc/trunk/webrtc/base/cpumonitor.h
deleted file mode 100644
index e82ae69516..0000000000
--- a/media/webrtc/trunk/webrtc/base/cpumonitor.h
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- *  Copyright 2010 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_CPUMONITOR_H_
-#define WEBRTC_BASE_CPUMONITOR_H_
-
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/messagehandler.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/sigslot.h"
-#if defined(WEBRTC_LINUX)
-#include "webrtc/base/stream.h"
-#endif // defined(WEBRTC_LINUX)
-
-namespace rtc {
-class Thread;
-class SystemInfo;
-
-struct CpuStats {
-  CpuStats()
-      : prev_total_times_(0),
-        prev_cpu_times_(0),
-        prev_load_(0.f),
-        prev_load_time_(0u) {
-  }
-
-  uint64 prev_total_times_;
-  uint64 prev_cpu_times_;
-  float prev_load_;  // Previous load value.
-  uint32 prev_load_time_;  // Time previous load value was taken.
-};
-
-// CpuSampler samples the process and system load.
-class CpuSampler {
- public:
-  CpuSampler();
-  ~CpuSampler();
-
-  // Initialize CpuSampler.  Returns true if successful.
-  bool Init();
-
-  // Set minimum interval in ms between computing new load values.
-  // Default 950 ms.  Set to 0 to disable interval.
-  void set_load_interval(int min_load_interval);
-
-  // Return CPU load of current process as a float from 0 to 1.
-  float GetProcessLoad();
-
-  // Return CPU load of current process as a float from 0 to 1.
-  float GetSystemLoad();
-
-  // Return number of cpus. Includes hyperthreads.
-  int GetMaxCpus() const;
-
-  // Return current number of cpus available to this process.
-  int GetCurrentCpus();
-
-  // For testing. Allows forcing of fallback to using NTDLL functions.
-  void set_force_fallback(bool fallback) {
-#if defined(WEBRTC_WIN)
-    force_fallback_ = fallback;
-#endif
-  }
-
- private:
-  float UpdateCpuLoad(uint64 current_total_times,
-                      uint64 current_cpu_times,
-                      uint64 *prev_total_times,
-                      uint64 *prev_cpu_times);
-  CpuStats process_;
-  CpuStats system_;
-  int cpus_;
-  int min_load_interval_;  // Minimum time between computing new load.
-  scoped_ptr<SystemInfo> sysinfo_;
-#if defined(WEBRTC_WIN)
-  void* get_system_times_;
-  void* nt_query_system_information_;
-  bool force_fallback_;
-#endif
-#if defined(WEBRTC_LINUX)
-  // File for reading /proc/stat
-  scoped_ptr<FileStream> sfile_;
-#endif // defined(WEBRTC_LINUX)
-};
-
-// CpuMonitor samples and signals the CPU load periodically.
-class CpuMonitor
-    : public rtc::MessageHandler, public sigslot::has_slots<> {
- public:
-  explicit CpuMonitor(Thread* thread);
-  ~CpuMonitor() override;
-  void set_thread(Thread* thread);
-
-  bool Start(int period_ms);
-  void Stop();
-  // Signal parameters are current cpus, max cpus, process load and system load.
-  sigslot::signal4<int, int, float, float> SignalUpdate;
-
- protected:
-  // Override virtual method of parent MessageHandler.
-  void OnMessage(rtc::Message* msg) override;
-  // Clear the monitor thread and stop sending it messages if the thread goes
-  // away before our lifetime.
-  void OnMessageQueueDestroyed() { monitor_thread_ = NULL; }
-
- private:
-  Thread* monitor_thread_;
-  CpuSampler sampler_;
-  int period_ms_;
-
-  DISALLOW_COPY_AND_ASSIGN(CpuMonitor);
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_CPUMONITOR_H_
diff --git a/media/webrtc/trunk/webrtc/base/cpumonitor_unittest.cc b/media/webrtc/trunk/webrtc/base/cpumonitor_unittest.cc
deleted file mode 100644
index 379f62fd3c..0000000000
--- a/media/webrtc/trunk/webrtc/base/cpumonitor_unittest.cc
+++ /dev/null
@@ -1,389 +0,0 @@
-/*
- *  Copyright 2010 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <iomanip>
-#include <iostream>
-#include <vector>
-
-#if defined(WEBRTC_WIN)
-#include "webrtc/base/win32.h"
-#endif
-
-#include "webrtc/base/cpumonitor.h"
-#include "webrtc/base/flags.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/timing.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace rtc {
-
-static const int kMaxCpus = 1024;
-static const int kSettleTime = 100;  // Amount of time to between tests.
-static const int kIdleTime = 500;  // Amount of time to be idle in ms.
-static const int kBusyTime = 1000;  // Amount of time to be busy in ms.
-static const int kLongInterval = 2000;  // Interval longer than busy times
-
-class BusyThread : public rtc::Thread {
- public:
-  BusyThread(double load, double duration, double interval) :
-    load_(load), duration_(duration), interval_(interval) {
-  }
-  virtual ~BusyThread() {
-    Stop();
-  }
-  void Run() {
-    Timing time;
-    double busy_time = interval_ * load_ / 100.0;
-    for (;;) {
-      time.BusyWait(busy_time);
-      time.IdleWait(interval_ - busy_time);
-      if (duration_) {
-        duration_ -= interval_;
-        if (duration_ <= 0) {
-          break;
-        }
-      }
-    }
-  }
- private:
-  double load_;
-  double duration_;
-  double interval_;
-};
-
-class CpuLoadListener : public sigslot::has_slots<> {
- public:
-  CpuLoadListener()
-      : current_cpus_(0),
-        cpus_(0),
-        process_load_(.0f),
-        system_load_(.0f),
-        count_(0) {
-  }
-
-  void OnCpuLoad(int current_cpus, int cpus, float proc_load, float sys_load) {
-    current_cpus_ = current_cpus;
-    cpus_ = cpus;
-    process_load_ = proc_load;
-    system_load_ = sys_load;
-    ++count_;
-  }
-
-  int current_cpus() const { return current_cpus_; }
-  int cpus() const { return cpus_; }
-  float process_load() const { return process_load_; }
-  float system_load() const { return system_load_; }
-  int count() const { return count_; }
-
- private:
-  int current_cpus_;
-  int cpus_;
-  float process_load_;
-  float system_load_;
-  int count_;
-};
-
-// Set affinity (which cpu to run on), but respecting FLAG_affinity:
-// -1 means no affinity - run on whatever cpu is available.
-// 0 .. N means run on specific cpu.  The tool will create N threads and call
-//   SetThreadAffinity on 0 to N - 1 as cpu.  FLAG_affinity sets the first cpu
-//   so the range becomes affinity to affinity + N - 1
-// Note that this function affects Windows scheduling, effectively giving
-//   the thread with affinity for a specified CPU more priority on that CPU.
-bool SetThreadAffinity(BusyThread* t, int cpu, int affinity) {
-#if defined(WEBRTC_WIN)
-  if (affinity >= 0) {
-    return ::SetThreadAffinityMask(t->GetHandle(),
-        1 << (cpu + affinity)) != FALSE;
-  }
-#endif
-  return true;
-}
-
-bool SetThreadPriority(BusyThread* t, int prio) {
-  if (!prio) {
-    return true;
-  }
-  bool ok = t->SetPriority(static_cast<rtc::ThreadPriority>(prio));
-  if (!ok) {
-    std::cout << "Error setting thread priority." << std::endl;
-  }
-  return ok;
-}
-
-int CpuLoad(double cpuload, double duration, int numthreads,
-            int priority, double interval, int affinity) {
-  int ret = 0;
-  std::vector<BusyThread*> threads;
-  for (int i = 0; i < numthreads; ++i) {
-    threads.push_back(new BusyThread(cpuload, duration, interval));
-    // NOTE(fbarchard): Priority must be done before Start.
-    if (!SetThreadPriority(threads[i], priority) ||
-       !threads[i]->Start() ||
-       !SetThreadAffinity(threads[i], i, affinity)) {
-      ret = 1;
-      break;
-    }
-  }
-  // Wait on each thread
-  if (ret == 0) {
-    for (int i = 0; i < numthreads; ++i) {
-      threads[i]->Stop();
-    }
-  }
-
-  for (int i = 0; i < numthreads; ++i) {
-    delete threads[i];
-  }
-  return ret;
-}
-
-// Make 2 CPUs busy
-static void CpuTwoBusyLoop(int busytime) {
-  CpuLoad(100.0, busytime / 1000.0, 2, 1, 0.050, -1);
-}
-
-// Make 1 CPUs busy
-static void CpuBusyLoop(int busytime) {
-  CpuLoad(100.0, busytime / 1000.0, 1, 1, 0.050, -1);
-}
-
-// Make 1 use half CPU time.
-static void CpuHalfBusyLoop(int busytime) {
-  CpuLoad(50.0, busytime / 1000.0, 1, 1, 0.050, -1);
-}
-
-void TestCpuSampler(bool test_proc, bool test_sys, bool force_fallback) {
-  CpuSampler sampler;
-  sampler.set_force_fallback(force_fallback);
-  EXPECT_TRUE(sampler.Init());
-  sampler.set_load_interval(100);
-  int cpus = sampler.GetMaxCpus();
-
-  // Test1: CpuSampler under idle situation.
-  Thread::SleepMs(kSettleTime);
-  sampler.GetProcessLoad();
-  sampler.GetSystemLoad();
-
-  Thread::SleepMs(kIdleTime);
-
-  float proc_idle = 0.f, sys_idle = 0.f;
-  if (test_proc) {
-    proc_idle = sampler.GetProcessLoad();
-  }
-  if (test_sys) {
-      sys_idle = sampler.GetSystemLoad();
-  }
-  if (test_proc) {
-    LOG(LS_INFO) << "ProcessLoad Idle:      "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << proc_idle;
-    EXPECT_GE(proc_idle, 0.f);
-    EXPECT_LE(proc_idle, static_cast<float>(cpus));
-  }
-  if (test_sys) {
-    LOG(LS_INFO) << "SystemLoad Idle:       "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << sys_idle;
-    EXPECT_GE(sys_idle, 0.f);
-    EXPECT_LE(sys_idle, static_cast<float>(cpus));
-  }
-
-  // Test2: CpuSampler with main process at 50% busy.
-  Thread::SleepMs(kSettleTime);
-  sampler.GetProcessLoad();
-  sampler.GetSystemLoad();
-
-  CpuHalfBusyLoop(kBusyTime);
-
-  float proc_halfbusy = 0.f, sys_halfbusy = 0.f;
-  if (test_proc) {
-    proc_halfbusy = sampler.GetProcessLoad();
-  }
-  if (test_sys) {
-    sys_halfbusy = sampler.GetSystemLoad();
-  }
-  if (test_proc) {
-    LOG(LS_INFO) << "ProcessLoad Halfbusy:  "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << proc_halfbusy;
-    EXPECT_GE(proc_halfbusy, 0.f);
-    EXPECT_LE(proc_halfbusy, static_cast<float>(cpus));
-  }
-  if (test_sys) {
-    LOG(LS_INFO) << "SystemLoad Halfbusy:   "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << sys_halfbusy;
-    EXPECT_GE(sys_halfbusy, 0.f);
-    EXPECT_LE(sys_halfbusy, static_cast<float>(cpus));
-  }
-
-  // Test3: CpuSampler with main process busy.
-  Thread::SleepMs(kSettleTime);
-  sampler.GetProcessLoad();
-  sampler.GetSystemLoad();
-
-  CpuBusyLoop(kBusyTime);
-
-  float proc_busy = 0.f, sys_busy = 0.f;
-  if (test_proc) {
-    proc_busy = sampler.GetProcessLoad();
-  }
-  if (test_sys) {
-    sys_busy = sampler.GetSystemLoad();
-  }
-  if (test_proc) {
-    LOG(LS_INFO) << "ProcessLoad Busy:      "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << proc_busy;
-    EXPECT_GE(proc_busy, 0.f);
-    EXPECT_LE(proc_busy, static_cast<float>(cpus));
-  }
-  if (test_sys) {
-    LOG(LS_INFO) << "SystemLoad Busy:       "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << sys_busy;
-    EXPECT_GE(sys_busy, 0.f);
-    EXPECT_LE(sys_busy, static_cast<float>(cpus));
-  }
-
-  // Test4: CpuSampler with 2 cpus process busy.
-  if (cpus >= 2) {
-    Thread::SleepMs(kSettleTime);
-    sampler.GetProcessLoad();
-    sampler.GetSystemLoad();
-
-    CpuTwoBusyLoop(kBusyTime);
-
-    float proc_twobusy = 0.f, sys_twobusy = 0.f;
-    if (test_proc) {
-      proc_twobusy = sampler.GetProcessLoad();
-    }
-    if (test_sys) {
-      sys_twobusy = sampler.GetSystemLoad();
-    }
-    if (test_proc) {
-      LOG(LS_INFO) << "ProcessLoad 2 CPU Busy:"
-                   << std::setiosflags(std::ios_base::fixed)
-                   << std::setprecision(2) << std::setw(6) << proc_twobusy;
-      EXPECT_GE(proc_twobusy, 0.f);
-      EXPECT_LE(proc_twobusy, static_cast<float>(cpus));
-    }
-    if (test_sys) {
-      LOG(LS_INFO) << "SystemLoad 2 CPU Busy: "
-                   << std::setiosflags(std::ios_base::fixed)
-                   << std::setprecision(2) << std::setw(6) << sys_twobusy;
-      EXPECT_GE(sys_twobusy, 0.f);
-      EXPECT_LE(sys_twobusy, static_cast<float>(cpus));
-    }
-  }
-
-  // Test5: CpuSampler with idle process after being busy.
-  Thread::SleepMs(kSettleTime);
-  sampler.GetProcessLoad();
-  sampler.GetSystemLoad();
-
-  Thread::SleepMs(kIdleTime);
-
-  if (test_proc) {
-    proc_idle = sampler.GetProcessLoad();
-  }
-  if (test_sys) {
-    sys_idle = sampler.GetSystemLoad();
-  }
-  if (test_proc) {
-    LOG(LS_INFO) << "ProcessLoad Idle:      "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << proc_idle;
-    EXPECT_GE(proc_idle, 0.f);
-    EXPECT_LE(proc_idle, proc_busy);
-  }
-  if (test_sys) {
-    LOG(LS_INFO) << "SystemLoad Idle:       "
-                 << std::setiosflags(std::ios_base::fixed)
-                 << std::setprecision(2) << std::setw(6) << sys_idle;
-    EXPECT_GE(sys_idle, 0.f);
-    EXPECT_LE(sys_idle, static_cast<float>(cpus));
-  }
-}
-
-TEST(CpuMonitorTest, TestCpus) {
-  CpuSampler sampler;
-  EXPECT_TRUE(sampler.Init());
-  int current_cpus = sampler.GetCurrentCpus();
-  int cpus = sampler.GetMaxCpus();
-  LOG(LS_INFO) << "Current Cpus:     " << std::setw(9) << current_cpus;
-  LOG(LS_INFO) << "Maximum Cpus:     " << std::setw(9) << cpus;
-  EXPECT_GT(cpus, 0);
-  EXPECT_LE(cpus, kMaxCpus);
-  EXPECT_GT(current_cpus, 0);
-  EXPECT_LE(current_cpus, cpus);
-}
-
-#if defined(WEBRTC_WIN)
-// Tests overall system CpuSampler using legacy OS fallback code if applicable.
-TEST(CpuMonitorTest, TestGetSystemLoadForceFallback) {
-  TestCpuSampler(false, true, true);
-}
-#endif
-
-// Tests both process and system functions in use at same time.
-TEST(CpuMonitorTest, TestGetBothLoad) {
-  TestCpuSampler(true, true, false);
-}
-
-// Tests a query less than the interval produces the same value.
-TEST(CpuMonitorTest, TestInterval) {
-  CpuSampler sampler;
-  EXPECT_TRUE(sampler.Init());
-
-  // Test1: Set interval to large value so sampler will not update.
-  sampler.set_load_interval(kLongInterval);
-
-  sampler.GetProcessLoad();
-  sampler.GetSystemLoad();
-
-  float proc_orig = sampler.GetProcessLoad();
-  float sys_orig = sampler.GetSystemLoad();
-
-  Thread::SleepMs(kIdleTime);
-
-  float proc_halftime = sampler.GetProcessLoad();
-  float sys_halftime = sampler.GetSystemLoad();
-
-  EXPECT_EQ(proc_orig, proc_halftime);
-  EXPECT_EQ(sys_orig, sys_halftime);
-}
-
-TEST(CpuMonitorTest, TestCpuMonitor) {
-  CpuMonitor monitor(Thread::Current());
-  CpuLoadListener listener;
-  monitor.SignalUpdate.connect(&listener, &CpuLoadListener::OnCpuLoad);
-  EXPECT_TRUE(monitor.Start(10));
-  // We have checked cpu load more than twice.
-  EXPECT_TRUE_WAIT(listener.count() > 2, 1000);
-  EXPECT_GT(listener.current_cpus(), 0);
-  EXPECT_GT(listener.cpus(), 0);
-  EXPECT_GE(listener.process_load(), .0f);
-  EXPECT_GE(listener.system_load(), .0f);
-
-  monitor.Stop();
-  // Wait 20 ms to ake sure all signals are delivered.
-  Thread::Current()->ProcessMessages(20);
-  int old_count = listener.count();
-  Thread::Current()->ProcessMessages(20);
-  // Verfy no more siganls.
-  EXPECT_EQ(old_count, listener.count());
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/crc32.cc b/media/webrtc/trunk/webrtc/base/crc32.cc
index d643a25a4b..97b82145d7 100644
--- a/media/webrtc/trunk/webrtc/base/crc32.cc
+++ b/media/webrtc/trunk/webrtc/base/crc32.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/base/crc32.h"
 
-#include "webrtc/base/basicdefs.h"
+#include "webrtc/base/arraysize.h"
 
 namespace rtc {
 
@@ -18,14 +18,14 @@ namespace rtc {
 
 // CRC32 polynomial, in reversed form.
 // See RFC 1952, or http://en.wikipedia.org/wiki/Cyclic_redundancy_check
-static const uint32 kCrc32Polynomial = 0xEDB88320;
-static uint32 kCrc32Table[256] = { 0 };
+static const uint32_t kCrc32Polynomial = 0xEDB88320;
+static uint32_t kCrc32Table[256] = {0};
 
 static void EnsureCrc32TableInited() {
-  if (kCrc32Table[ARRAY_SIZE(kCrc32Table) - 1])
+  if (kCrc32Table[arraysize(kCrc32Table) - 1])
     return;  // already inited
-  for (uint32 i = 0; i < ARRAY_SIZE(kCrc32Table); ++i) {
-    uint32 c = i;
+  for (uint32_t i = 0; i < arraysize(kCrc32Table); ++i) {
+    uint32_t c = i;
     for (size_t j = 0; j < 8; ++j) {
       if (c & 1) {
         c = kCrc32Polynomial ^ (c >> 1);
@@ -37,11 +37,11 @@ static void EnsureCrc32TableInited() {
   }
 }
 
-uint32 UpdateCrc32(uint32 start, const void* buf, size_t len) {
+uint32_t UpdateCrc32(uint32_t start, const void* buf, size_t len) {
   EnsureCrc32TableInited();
 
-  uint32 c = start ^ 0xFFFFFFFF;
-  const uint8* u = static_cast<const uint8*>(buf);
+  uint32_t c = start ^ 0xFFFFFFFF;
+  const uint8_t* u = static_cast<const uint8_t*>(buf);
   for (size_t i = 0; i < len; ++i) {
     c = kCrc32Table[(c ^ u[i]) & 0xFF] ^ (c >> 8);
   }
diff --git a/media/webrtc/trunk/webrtc/base/crc32.h b/media/webrtc/trunk/webrtc/base/crc32.h
index 99b4cac894..9661876298 100644
--- a/media/webrtc/trunk/webrtc/base/crc32.h
+++ b/media/webrtc/trunk/webrtc/base/crc32.h
@@ -19,13 +19,13 @@ namespace rtc {
 
 // Updates a CRC32 checksum with |len| bytes from |buf|. |initial| holds the
 // checksum result from the previous update; for the first call, it should be 0.
-uint32 UpdateCrc32(uint32 initial, const void* buf, size_t len);
+uint32_t UpdateCrc32(uint32_t initial, const void* buf, size_t len);
 
 // Computes a CRC32 checksum using |len| bytes from |buf|.
-inline uint32 ComputeCrc32(const void* buf, size_t len) {
+inline uint32_t ComputeCrc32(const void* buf, size_t len) {
   return UpdateCrc32(0, buf, len);
 }
-inline uint32 ComputeCrc32(const std::string& str) {
+inline uint32_t ComputeCrc32(const std::string& str) {
   return ComputeCrc32(str.c_str(), str.size());
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/crc32_unittest.cc b/media/webrtc/trunk/webrtc/base/crc32_unittest.cc
index 0bfdeeea0d..6da5c32378 100644
--- a/media/webrtc/trunk/webrtc/base/crc32_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/crc32_unittest.cc
@@ -25,7 +25,7 @@ TEST(Crc32Test, TestBasic) {
 TEST(Crc32Test, TestMultipleUpdates) {
   std::string input =
       "abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq";
-  uint32 c = 0;
+  uint32_t c = 0;
   for (size_t i = 0; i < input.size(); ++i) {
     c = UpdateCrc32(c, &input[i], 1);
   }
diff --git a/media/webrtc/trunk/webrtc/base/criticalsection.cc b/media/webrtc/trunk/webrtc/base/criticalsection.cc
index fcad5c31d6..1f50c2355d 100644
--- a/media/webrtc/trunk/webrtc/base/criticalsection.cc
+++ b/media/webrtc/trunk/webrtc/base/criticalsection.cc
@@ -11,23 +11,159 @@
 #include "webrtc/base/criticalsection.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/base/thread.h"
 
 namespace rtc {
 
+CriticalSection::CriticalSection() {
+#if defined(WEBRTC_WIN)
+  InitializeCriticalSection(&crit_);
+#else
+  pthread_mutexattr_t mutex_attribute;
+  pthread_mutexattr_init(&mutex_attribute);
+  pthread_mutexattr_settype(&mutex_attribute, PTHREAD_MUTEX_RECURSIVE);
+  pthread_mutex_init(&mutex_, &mutex_attribute);
+  pthread_mutexattr_destroy(&mutex_attribute);
+  CS_DEBUG_CODE(thread_ = 0);
+  CS_DEBUG_CODE(recursion_count_ = 0);
+#endif
+}
+
+CriticalSection::~CriticalSection() {
+#if defined(WEBRTC_WIN)
+  DeleteCriticalSection(&crit_);
+#else
+  pthread_mutex_destroy(&mutex_);
+#endif
+}
+
+void CriticalSection::Enter() EXCLUSIVE_LOCK_FUNCTION() {
+#if defined(WEBRTC_WIN)
+  EnterCriticalSection(&crit_);
+#else
+  pthread_mutex_lock(&mutex_);
+#if CS_DEBUG_CHECKS
+  if (!recursion_count_) {
+    RTC_DCHECK(!thread_);
+    thread_ = pthread_self();
+  } else {
+    RTC_DCHECK(CurrentThreadIsOwner());
+  }
+  ++recursion_count_;
+#endif
+#endif
+}
+
+bool CriticalSection::TryEnter() EXCLUSIVE_TRYLOCK_FUNCTION(true) {
+#if defined(WEBRTC_WIN)
+  return TryEnterCriticalSection(&crit_) != FALSE;
+#else
+  if (pthread_mutex_trylock(&mutex_) != 0)
+    return false;
+#if CS_DEBUG_CHECKS
+  if (!recursion_count_) {
+    RTC_DCHECK(!thread_);
+    thread_ = pthread_self();
+  } else {
+    RTC_DCHECK(CurrentThreadIsOwner());
+  }
+  ++recursion_count_;
+#endif
+  return true;
+#endif
+}
+void CriticalSection::Leave() UNLOCK_FUNCTION() {
+  RTC_DCHECK(CurrentThreadIsOwner());
+#if defined(WEBRTC_WIN)
+  LeaveCriticalSection(&crit_);
+#else
+#if CS_DEBUG_CHECKS
+  --recursion_count_;
+  RTC_DCHECK(recursion_count_ >= 0);
+  if (!recursion_count_)
+    thread_ = 0;
+#endif
+  pthread_mutex_unlock(&mutex_);
+#endif
+}
+
+bool CriticalSection::CurrentThreadIsOwner() const {
+#if defined(WEBRTC_WIN)
+  // OwningThread has type HANDLE but actually contains the Thread ID:
+  // http://stackoverflow.com/questions/12675301/why-is-the-owningthread-member-of-critical-section-of-type-handle-when-it-is-de
+  // Converting through size_t avoids the VS 2015 warning C4312: conversion from
+  // 'type1' to 'type2' of greater size
+  return crit_.OwningThread ==
+         reinterpret_cast<HANDLE>(static_cast<size_t>(GetCurrentThreadId()));
+#else
+#if CS_DEBUG_CHECKS
+  return pthread_equal(thread_, pthread_self());
+#else
+  return true;
+#endif  // CS_DEBUG_CHECKS
+#endif
+}
+
+bool CriticalSection::IsLocked() const {
+#if defined(WEBRTC_WIN)
+  return crit_.LockCount != -1;
+#else
+#if CS_DEBUG_CHECKS
+  return thread_ != 0;
+#else
+  return true;
+#endif
+#endif
+}
+
+CritScope::CritScope(CriticalSection* cs) : cs_(cs) { cs_->Enter(); }
+CritScope::~CritScope() { cs_->Leave(); }
+
+TryCritScope::TryCritScope(CriticalSection* cs)
+    : cs_(cs), locked_(cs->TryEnter()) {
+  CS_DEBUG_CODE(lock_was_called_ = false);
+}
+
+TryCritScope::~TryCritScope() {
+  CS_DEBUG_CODE(RTC_DCHECK(lock_was_called_));
+  if (locked_)
+    cs_->Leave();
+}
+
+bool TryCritScope::locked() const {
+  CS_DEBUG_CODE(lock_was_called_ = true);
+  return locked_;
+}
+
 void GlobalLockPod::Lock() {
+#if !defined(WEBRTC_WIN)
+  const struct timespec ts_null = {0};
+#endif
+
   while (AtomicOps::CompareAndSwap(&lock_acquired, 0, 1)) {
-    Thread::SleepMs(0);
+#if defined(WEBRTC_WIN)
+    ::Sleep(0);
+#else
+    nanosleep(&ts_null, nullptr);
+#endif
   }
 }
 
 void GlobalLockPod::Unlock() {
   int old_value = AtomicOps::CompareAndSwap(&lock_acquired, 1, 0);
-  DCHECK_EQ(1, old_value) << "Unlock called without calling Lock first";
+  RTC_DCHECK_EQ(1, old_value) << "Unlock called without calling Lock first";
 }
 
 GlobalLock::GlobalLock() {
   lock_acquired = 0;
 }
 
+GlobalLockScope::GlobalLockScope(GlobalLockPod* lock)
+    : lock_(lock) {
+  lock_->Lock();
+}
+
+GlobalLockScope::~GlobalLockScope() {
+  lock_->Unlock();
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/criticalsection.h b/media/webrtc/trunk/webrtc/base/criticalsection.h
index db197d2e07..5b3eaf5684 100644
--- a/media/webrtc/trunk/webrtc/base/criticalsection.h
+++ b/media/webrtc/trunk/webrtc/base/criticalsection.h
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_BASE_CRITICALSECTION_H__
-#define WEBRTC_BASE_CRITICALSECTION_H__
+#ifndef WEBRTC_BASE_CRITICALSECTION_H_
+#define WEBRTC_BASE_CRITICALSECTION_H_
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
 
@@ -21,107 +22,57 @@
 // exists as two separate projects, webrtc and libjingle.
 #include <winsock2.h>
 #include <windows.h>
-#endif
+#include <sal.h>  // must come after windows headers.
+#endif  // defined(WEBRTC_WIN)
 
 #if defined(WEBRTC_POSIX)
 #include <pthread.h>
 #endif
 
-#ifdef _DEBUG
-#define CS_TRACK_OWNER 1
-#endif  // _DEBUG
+#if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
+#define CS_DEBUG_CHECKS 1
+#endif
 
-#if CS_TRACK_OWNER
-#define TRACK_OWNER(x) x
-#else  // !CS_TRACK_OWNER
-#define TRACK_OWNER(x)
-#endif  // !CS_TRACK_OWNER
+#if CS_DEBUG_CHECKS
+#define CS_DEBUG_CODE(x) x
+#else  // !CS_DEBUG_CHECKS
+#define CS_DEBUG_CODE(x)
+#endif  // !CS_DEBUG_CHECKS
 
 namespace rtc {
 
+class LOCKABLE CriticalSection {
+ public:
+  CriticalSection();
+  ~CriticalSection();
+
+  void Enter() EXCLUSIVE_LOCK_FUNCTION();
+  bool TryEnter() EXCLUSIVE_TRYLOCK_FUNCTION(true);
+  void Leave() UNLOCK_FUNCTION();
+
+  // Use only for RTC_DCHECKing.
+  bool CurrentThreadIsOwner() const;
+  // Use only for RTC_DCHECKing.
+  bool IsLocked() const;
+
+ private:
 #if defined(WEBRTC_WIN)
-class LOCKABLE CriticalSection {
- public:
-  CriticalSection() { InitializeCriticalSection(&crit_); }
-  ~CriticalSection() { DeleteCriticalSection(&crit_); }
-  void Enter() EXCLUSIVE_LOCK_FUNCTION() {
-    EnterCriticalSection(&crit_);
-  }
-  bool TryEnter() EXCLUSIVE_TRYLOCK_FUNCTION(true) {
-    return TryEnterCriticalSection(&crit_) != FALSE;
-  }
-  void Leave() UNLOCK_FUNCTION() {
-    LeaveCriticalSection(&crit_);
-  }
-
-  // Used for debugging.
-  bool CurrentThreadIsOwner() const {
-    return crit_.OwningThread == reinterpret_cast<HANDLE>(GetCurrentThreadId());
-  }
-
- private:
   CRITICAL_SECTION crit_;
-};
-#endif // WEBRTC_WIN
-
-#if defined(WEBRTC_POSIX)
-class LOCKABLE CriticalSection {
- public:
-  CriticalSection() {
-    pthread_mutexattr_t mutex_attribute;
-    pthread_mutexattr_init(&mutex_attribute);
-    pthread_mutexattr_settype(&mutex_attribute, PTHREAD_MUTEX_RECURSIVE);
-    pthread_mutex_init(&mutex_, &mutex_attribute);
-    pthread_mutexattr_destroy(&mutex_attribute);
-    TRACK_OWNER(thread_ = 0);
-  }
-  ~CriticalSection() {
-    pthread_mutex_destroy(&mutex_);
-  }
-  void Enter() EXCLUSIVE_LOCK_FUNCTION() {
-    pthread_mutex_lock(&mutex_);
-    TRACK_OWNER(thread_ = pthread_self());
-  }
-  bool TryEnter() EXCLUSIVE_TRYLOCK_FUNCTION(true) {
-    if (pthread_mutex_trylock(&mutex_) == 0) {
-      TRACK_OWNER(thread_ = pthread_self());
-      return true;
-    }
-    return false;
-  }
-  void Leave() UNLOCK_FUNCTION() {
-    TRACK_OWNER(thread_ = 0);
-    pthread_mutex_unlock(&mutex_);
-  }
-
-  // Used for debugging.
-  bool CurrentThreadIsOwner() const {
-#if CS_TRACK_OWNER
-    return pthread_equal(thread_, pthread_self());
-#else
-    return true;
-#endif  // CS_TRACK_OWNER
-  }
-
- private:
+#elif defined(WEBRTC_POSIX)
   pthread_mutex_t mutex_;
-  TRACK_OWNER(pthread_t thread_);
+  CS_DEBUG_CODE(pthread_t thread_);
+  CS_DEBUG_CODE(int recursion_count_);
+#endif
 };
-#endif // WEBRTC_POSIX
 
 // CritScope, for serializing execution through a scope.
 class SCOPED_LOCKABLE CritScope {
  public:
-  explicit CritScope(CriticalSection *pcrit) EXCLUSIVE_LOCK_FUNCTION(pcrit) {
-    pcrit_ = pcrit;
-    pcrit_->Enter();
-  }
-  ~CritScope() UNLOCK_FUNCTION() {
-    pcrit_->Leave();
-  }
+  explicit CritScope(CriticalSection* cs) EXCLUSIVE_LOCK_FUNCTION(cs);
+  ~CritScope() UNLOCK_FUNCTION();
  private:
-  CriticalSection *pcrit_;
-  DISALLOW_COPY_AND_ASSIGN(CritScope);
+  CriticalSection* const cs_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(CritScope);
 };
 
 // Tries to lock a critical section on construction via
@@ -133,69 +84,20 @@ class SCOPED_LOCKABLE CritScope {
 // lock was taken. If you're not calling locked(), you're doing it wrong!
 class TryCritScope {
  public:
-  explicit TryCritScope(CriticalSection *pcrit) {
-    pcrit_ = pcrit;
-    locked_ = pcrit_->TryEnter();
-  }
-  ~TryCritScope() {
-    if (locked_) {
-      pcrit_->Leave();
-    }
-  }
-  bool locked() const {
-    return locked_;
-  }
- private:
-  CriticalSection *pcrit_;
-  bool locked_;
-  DISALLOW_COPY_AND_ASSIGN(TryCritScope);
-};
-
-// TODO: Move this to atomicops.h, which can't be done easily because of
-// complex compile rules.
-class AtomicOps {
- public:
+  explicit TryCritScope(CriticalSection* cs);
+  ~TryCritScope();
 #if defined(WEBRTC_WIN)
-  // Assumes sizeof(int) == sizeof(LONG), which it is on Win32 and Win64.
-  static int Increment(volatile int* i) {
-    return ::InterlockedIncrement(reinterpret_cast<volatile LONG*>(i));
-  }
-  static int Decrement(volatile int* i) {
-    return ::InterlockedDecrement(reinterpret_cast<volatile LONG*>(i));
-  }
-  static int Load(volatile const int* i) {
-    return *i;
-  }
-  static void Store(volatile int* i, int value) {
-    *i = value;
-  }
-  static int CompareAndSwap(volatile int* i, int old_value, int new_value) {
-    return ::InterlockedCompareExchange(reinterpret_cast<volatile LONG*>(i),
-                                        new_value,
-                                        old_value);
-  }
+  _Check_return_ bool locked() const;
 #else
-  static int Increment(volatile int* i) {
-    return __sync_add_and_fetch(i, 1);
-  }
-  static int Decrement(volatile int* i) {
-    return __sync_sub_and_fetch(i, 1);
-  }
-  static int Load(volatile const int* i) {
-    // Adding 0 is a no-op, so const_cast is fine.
-    return __sync_add_and_fetch(const_cast<volatile int*>(i), 0);
-  }
-  static void Store(volatile int* i, int value) {
-    __sync_synchronize();
-    *i = value;
-  }
-  static int CompareAndSwap(volatile int* i, int old_value, int new_value) {
-    return __sync_val_compare_and_swap(i, old_value, new_value);
-  }
+  bool locked() const __attribute__ ((__warn_unused_result__));
 #endif
+ private:
+  CriticalSection* const cs_;
+  const bool locked_;
+  CS_DEBUG_CODE(mutable bool lock_was_called_);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TryCritScope);
 };
 
-
 // A POD lock used to protect global variables. Do NOT use for other purposes.
 // No custom constructor or private data member should be added.
 class LOCKABLE GlobalLockPod {
@@ -212,6 +114,16 @@ class GlobalLock : public GlobalLockPod {
   GlobalLock();
 };
 
+// GlobalLockScope, for serializing execution through a scope.
+class SCOPED_LOCKABLE GlobalLockScope {
+ public:
+  explicit GlobalLockScope(GlobalLockPod* lock) EXCLUSIVE_LOCK_FUNCTION(lock);
+  ~GlobalLockScope() UNLOCK_FUNCTION();
+ private:
+  GlobalLockPod* const lock_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(GlobalLockScope);
+};
+
 } // namespace rtc
 
-#endif // WEBRTC_BASE_CRITICALSECTION_H__
+#endif // WEBRTC_BASE_CRITICALSECTION_H_
diff --git a/media/webrtc/trunk/webrtc/base/criticalsection_unittest.cc b/media/webrtc/trunk/webrtc/base/criticalsection_unittest.cc
index 6f3c7e9312..d6990c0023 100644
--- a/media/webrtc/trunk/webrtc/base/criticalsection_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/criticalsection_unittest.cc
@@ -14,9 +14,9 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scopedptrcollection.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
@@ -220,6 +220,28 @@ TEST(AtomicOpsTest, Simple) {
   EXPECT_EQ(0, value);
 }
 
+TEST(AtomicOpsTest, SimplePtr) {
+  class Foo {};
+  Foo* volatile foo = nullptr;
+  scoped_ptr<Foo> a(new Foo());
+  scoped_ptr<Foo> b(new Foo());
+  // Reading the initial value should work as expected.
+  EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == nullptr);
+  // Setting using compare and swap should work.
+  EXPECT_TRUE(rtc::AtomicOps::CompareAndSwapPtr(
+                  &foo, static_cast<Foo*>(nullptr), a.get()) == nullptr);
+  EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == a.get());
+  // Setting another value but with the wrong previous pointer should fail
+  // (remain a).
+  EXPECT_TRUE(rtc::AtomicOps::CompareAndSwapPtr(
+                  &foo, static_cast<Foo*>(nullptr), b.get()) == a.get());
+  EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == a.get());
+  // Replacing a with b should work.
+  EXPECT_TRUE(rtc::AtomicOps::CompareAndSwapPtr(&foo, a.get(), b.get()) ==
+              a.get());
+  EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == b.get());
+}
+
 TEST(AtomicOpsTest, Increment) {
   // Create and start lots of threads.
   AtomicOpRunner<IncrementOp, UniqueValueVerifier> runner(0);
@@ -281,4 +303,21 @@ TEST(CriticalSectionTest, Basic) {
   EXPECT_EQ(0, runner.shared_value());
 }
 
+#if !defined(NDEBUG) || defined(DCHECK_ALWAYS_ON)
+TEST(CriticalSectionTest, IsLocked) {
+  // Simple single-threaded test of IsLocked.
+  CriticalSection cs;
+  EXPECT_FALSE(cs.IsLocked());
+  cs.Enter();
+  EXPECT_TRUE(cs.IsLocked());
+  cs.Leave();
+  EXPECT_FALSE(cs.IsLocked());
+  if (!cs.TryEnter())
+    FAIL();
+  EXPECT_TRUE(cs.IsLocked());
+  cs.Leave();
+  EXPECT_FALSE(cs.IsLocked());
+}
+#endif
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/dbus_unittest.cc b/media/webrtc/trunk/webrtc/base/dbus_unittest.cc
index 505ddbbc8d..17752f143f 100644
--- a/media/webrtc/trunk/webrtc/base/dbus_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/dbus_unittest.cc
@@ -18,7 +18,7 @@ namespace rtc {
 
 #define SIG_NAME "NameAcquired"
 
-static const uint32 kTimeoutMs = 5000U;
+static const uint32_t kTimeoutMs = 5000U;
 
 class DBusSigFilterTest : public DBusSigFilter {
  public:
diff --git a/media/webrtc/trunk/webrtc/base/deprecation.h b/media/webrtc/trunk/webrtc/base/deprecation.h
new file mode 100644
index 0000000000..ce950f9b52
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/deprecation.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_DEPRECATION_H_
+#define WEBRTC_BASE_DEPRECATION_H_
+
+// Annotate the declarations of deprecated functions with this to cause a
+// compiler warning when they're used. Like so:
+//
+//   RTC_DEPRECATED std::pony PonyPlz(const std::pony_spec& ps);
+//
+// NOTE 1: The annotation goes on the declaration in the .h file, not the
+// definition in the .cc file!
+//
+// NOTE 2: In order to keep unit testing the deprecated function without
+// getting warnings, do something like this:
+//
+//   std::pony DEPRECATED_PonyPlz(const std::pony_spec& ps);
+//   RTC_DEPRECATED inline std::pony PonyPlz(const std::pony_spec& ps) {
+//     return DEPRECATED_PonyPlz(ps);
+//   }
+//
+// In other words, rename the existing function, and provide an inline wrapper
+// using the original name that calls it. That way, callers who are willing to
+// call it using the DEPRECATED_-prefixed name don't get the warning.
+//
+// TODO(kwiberg): Remove this when we can use [[deprecated]] from C++14.
+#if defined(_MSC_VER)
+// Note: Deprecation warnings seem to fail to trigger on Windows
+// (https://bugs.chromium.org/p/webrtc/issues/detail?id=5368).
+#define RTC_DEPRECATED __declspec(deprecated)
+#elif defined(__GNUC__)
+#define RTC_DEPRECATED __attribute__ ((__deprecated__))
+#else
+#define RTC_DEPRECATED
+#endif
+
+#endif  // WEBRTC_BASE_DEPRECATION_H_
diff --git a/media/webrtc/trunk/webrtc/base/diskcache.cc b/media/webrtc/trunk/webrtc/base/diskcache.cc
index 6bbc53eb13..a1fba6af9a 100644
--- a/media/webrtc/trunk/webrtc/base/diskcache.cc
+++ b/media/webrtc/trunk/webrtc/base/diskcache.cc
@@ -15,6 +15,7 @@
 #endif
 
 #include <algorithm>
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/diskcache.h"
 #include "webrtc/base/fileutils.h"
@@ -23,11 +24,11 @@
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 #define TRANSPARENT_CACHE_NAMES 1
-#else  // !_DEBUG
+#else
 #define TRANSPARENT_CACHE_NAMES 0
-#endif  // !_DEBUG
+#endif
 
 namespace rtc {
 
@@ -211,14 +212,14 @@ bool DiskCache::DeleteResource(const std::string& id) {
 }
 
 bool DiskCache::CheckLimit() {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   // Temporary check to make sure everything is working correctly.
   size_t cache_size = 0;
   for (EntryMap::iterator it = map_.begin(); it != map_.end(); ++it) {
     cache_size += it->second.size;
   }
   ASSERT(cache_size == total_size_);
-#endif  // _DEBUG
+#endif
 
   // TODO: Replace this with a non-brain-dead algorithm for clearing out the
   // oldest resources... something that isn't O(n^2)
@@ -263,7 +264,7 @@ std::string DiskCache::IdToFilename(const std::string& id, size_t index) const {
 #endif  // !TRANSPARENT_CACHE_NAMES
 
   char extension[32];
-  sprintfn(extension, ARRAY_SIZE(extension), ".%u", index);
+  sprintfn(extension, arraysize(extension), ".%u", index);
 
   Pathname pathname;
   pathname.SetFolder(folder_);
diff --git a/media/webrtc/trunk/webrtc/base/event.cc b/media/webrtc/trunk/webrtc/base/event.cc
index 999db38853..a9af208631 100644
--- a/media/webrtc/trunk/webrtc/base/event.cc
+++ b/media/webrtc/trunk/webrtc/base/event.cc
@@ -31,7 +31,7 @@ Event::Event(bool manual_reset, bool initially_signaled) {
                                 manual_reset,
                                 initially_signaled,
                                 NULL);                // Name.
-  CHECK(event_handle_);
+  RTC_CHECK(event_handle_);
 }
 
 Event::~Event() {
@@ -56,8 +56,8 @@ bool Event::Wait(int milliseconds) {
 Event::Event(bool manual_reset, bool initially_signaled)
     : is_manual_reset_(manual_reset),
       event_status_(initially_signaled) {
-  CHECK(pthread_mutex_init(&event_mutex_, NULL) == 0);
-  CHECK(pthread_cond_init(&event_cond_, NULL) == 0);
+  RTC_CHECK(pthread_mutex_init(&event_mutex_, NULL) == 0);
+  RTC_CHECK(pthread_cond_init(&event_cond_, NULL) == 0);
 }
 
 Event::~Event() {
diff --git a/media/webrtc/trunk/webrtc/base/event_tracer.cc b/media/webrtc/trunk/webrtc/base/event_tracer.cc
index 5c6d39f0a4..4174589d36 100644
--- a/media/webrtc/trunk/webrtc/base/event_tracer.cc
+++ b/media/webrtc/trunk/webrtc/base/event_tracer.cc
@@ -7,15 +7,26 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
 #include "webrtc/base/event_tracer.h"
 
+#include <inttypes.h>
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/base/trace_event.h"
+
 namespace webrtc {
 
 namespace {
 
-GetCategoryEnabledPtr g_get_category_enabled_ptr = 0;
-AddTraceEventPtr g_add_trace_event_ptr = 0;
+GetCategoryEnabledPtr g_get_category_enabled_ptr = nullptr;
+AddTraceEventPtr g_add_trace_event_ptr = nullptr;
 
 }  // namespace
 
@@ -25,7 +36,6 @@ void SetupEventTracer(GetCategoryEnabledPtr get_category_enabled_ptr,
   g_add_trace_event_ptr = add_trace_event_ptr;
 }
 
-// static
 const unsigned char* EventTracer::GetCategoryEnabled(const char* name) {
   if (g_get_category_enabled_ptr)
     return g_get_category_enabled_ptr(name);
@@ -34,7 +44,8 @@ const unsigned char* EventTracer::GetCategoryEnabled(const char* name) {
   return reinterpret_cast<const unsigned char*>("\0");
 }
 
-// static
+// Arguments to this function (phase, etc.) are as defined in
+// webrtc/base/trace_event.h.
 void EventTracer::AddTraceEvent(char phase,
                                 const unsigned char* category_enabled,
                                 const char* name,
@@ -58,3 +69,202 @@ void EventTracer::AddTraceEvent(char phase,
 }
 
 }  // namespace webrtc
+
+namespace rtc {
+namespace tracing {
+namespace {
+
+static bool EventTracingThreadFunc(void* params);
+
+// Atomic-int fast path for avoiding logging when disabled.
+static volatile int g_event_logging_active = 0;
+
+// TODO(pbos): Log metadata for all threads, etc.
+class EventLogger final {
+ public:
+  EventLogger()
+      : logging_thread_(EventTracingThreadFunc, this, "EventTracingThread"),
+        shutdown_event_(false, false) {}
+  ~EventLogger() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); }
+
+  void AddTraceEvent(const char* name,
+                     const unsigned char* category_enabled,
+                     char phase,
+                     uint64_t timestamp,
+                     int pid,
+                     rtc::PlatformThreadId thread_id) {
+    rtc::CritScope lock(&crit_);
+    trace_events_.push_back(
+        {name, category_enabled, phase, timestamp, 1, thread_id});
+  }
+
+// The TraceEvent format is documented here:
+// https://docs.google.com/document/d/1CvAClvFfyA5R-PhYUmn5OOQtYMH4h6I0nSsKchNAySU/preview
+  void Log() {
+    RTC_DCHECK(output_file_);
+    static const int kLoggingIntervalMs = 100;
+    fprintf(output_file_, "{ \"traceEvents\": [\n");
+    bool has_logged_event = false;
+    while (true) {
+      bool shutting_down = shutdown_event_.Wait(kLoggingIntervalMs);
+      std::vector<TraceEvent> events;
+      {
+        rtc::CritScope lock(&crit_);
+        trace_events_.swap(events);
+      }
+      for (const TraceEvent& e : events) {
+        fprintf(output_file_,
+                "%s{ \"name\": \"%s\""
+                ", \"cat\": \"%s\""
+                ", \"ph\": \"%c\""
+                ", \"ts\": %" PRIu64
+                ", \"pid\": %d"
+#if defined(WEBRTC_WIN)
+                ", \"tid\": %lu"
+#else
+                ", \"tid\": %d"
+#endif  // defined(WEBRTC_WIN)
+                "}\n",
+                has_logged_event ? "," : " ", e.name, e.category_enabled,
+                e.phase, e.timestamp, e.pid, e.tid);
+        has_logged_event = true;
+      }
+      if (shutting_down)
+        break;
+    }
+    fprintf(output_file_, "]}\n");
+    if (output_file_owned_)
+      fclose(output_file_);
+    output_file_ = nullptr;
+  }
+
+  void Start(FILE* file, bool owned) {
+    RTC_DCHECK(file);
+    RTC_DCHECK(!output_file_);
+    output_file_ = file;
+    output_file_owned_ = owned;
+    {
+      rtc::CritScope lock(&crit_);
+      // Since the atomic fast-path for adding events to the queue can be
+      // bypassed while the logging thread is shutting down there may be some
+      // stale events in the queue, hence the vector needs to be cleared to not
+      // log events from a previous logging session (which may be days old).
+      trace_events_.clear();
+    }
+    // Enable event logging (fast-path). This should be disabled since starting
+    // shouldn't be done twice.
+    RTC_CHECK_EQ(0,
+                 rtc::AtomicOps::CompareAndSwap(&g_event_logging_active, 0, 1));
+
+    // Finally start, everything should be set up now.
+    logging_thread_.Start();
+  }
+
+  void Stop() {
+    // Try to stop. Abort if we're not currently logging.
+    if (rtc::AtomicOps::CompareAndSwap(&g_event_logging_active, 1, 0) == 0)
+      return;
+
+    // Wake up logging thread to finish writing.
+    shutdown_event_.Set();
+    // Join the logging thread.
+    logging_thread_.Stop();
+  }
+
+ private:
+  struct TraceEvent {
+    const char* name;
+    const unsigned char* category_enabled;
+    char phase;
+    uint64_t timestamp;
+    int pid;
+    rtc::PlatformThreadId tid;
+  };
+
+  rtc::CriticalSection crit_;
+  std::vector<TraceEvent> trace_events_ GUARDED_BY(crit_);
+  rtc::PlatformThread logging_thread_;
+  rtc::Event shutdown_event_;
+  rtc::ThreadChecker thread_checker_;
+  FILE* output_file_ = nullptr;
+  bool output_file_owned_ = false;
+};
+
+static bool EventTracingThreadFunc(void* params) {
+  static_cast<EventLogger*>(params)->Log();
+  return true;
+}
+
+static EventLogger* volatile g_event_logger = nullptr;
+static const char* const kDisabledTracePrefix = TRACE_DISABLED_BY_DEFAULT("");
+const unsigned char* InternalGetCategoryEnabled(const char* name) {
+  const char* prefix_ptr = &kDisabledTracePrefix[0];
+  const char* name_ptr = name;
+  // Check whether name contains the default-disabled prefix.
+  while (*prefix_ptr == *name_ptr && *prefix_ptr != '\0') {
+    ++prefix_ptr;
+    ++name_ptr;
+  }
+  return reinterpret_cast<const unsigned char*>(*prefix_ptr == '\0' ? ""
+                                                                    : name);
+}
+
+void InternalAddTraceEvent(char phase,
+                           const unsigned char* category_enabled,
+                           const char* name,
+                           unsigned long long id,
+                           int num_args,
+                           const char** arg_names,
+                           const unsigned char* arg_types,
+                           const unsigned long long* arg_values,
+                           unsigned char flags) {
+  // Fast path for when event tracing is inactive.
+  if (rtc::AtomicOps::AcquireLoad(&g_event_logging_active) == 0)
+    return;
+
+  g_event_logger->AddTraceEvent(name, category_enabled, phase,
+                                rtc::TimeMicros(), 1, rtc::CurrentThreadId());
+}
+
+}  // namespace
+
+void SetupInternalTracer() {
+  RTC_CHECK(rtc::AtomicOps::CompareAndSwapPtr(
+                &g_event_logger, static_cast<EventLogger*>(nullptr),
+                new EventLogger()) == nullptr);
+  g_event_logger = new EventLogger();
+  webrtc::SetupEventTracer(InternalGetCategoryEnabled, InternalAddTraceEvent);
+}
+
+void StartInternalCaptureToFile(FILE* file) {
+  g_event_logger->Start(file, false);
+}
+
+bool StartInternalCapture(const char* filename) {
+  FILE* file = fopen(filename, "w");
+  if (!file) {
+    LOG(LS_ERROR) << "Failed to open trace file '" << filename
+                  << "' for writing.";
+    return false;
+  }
+  g_event_logger->Start(file, true);
+  return true;
+}
+
+void StopInternalCapture() {
+  g_event_logger->Stop();
+}
+
+void ShutdownInternalTracer() {
+  StopInternalCapture();
+  EventLogger* old_logger = rtc::AtomicOps::AcquireLoadPtr(&g_event_logger);
+  RTC_DCHECK(old_logger);
+  RTC_CHECK(rtc::AtomicOps::CompareAndSwapPtr(
+                &g_event_logger, old_logger,
+                static_cast<EventLogger*>(nullptr)) == old_logger);
+  delete old_logger;
+  webrtc::SetupEventTracer(nullptr, nullptr);
+}
+
+}  // namespace tracing
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/event_tracer.h b/media/webrtc/trunk/webrtc/base/event_tracer.h
index cfc6e9e472..51c8cfdc49 100644
--- a/media/webrtc/trunk/webrtc/base/event_tracer.h
+++ b/media/webrtc/trunk/webrtc/base/event_tracer.h
@@ -26,6 +26,8 @@
 #ifndef WEBRTC_BASE_EVENT_TRACER_H_
 #define WEBRTC_BASE_EVENT_TRACER_H_
 
+#include <stdio.h>
+
 namespace webrtc {
 
 typedef const unsigned char* (*GetCategoryEnabledPtr)(const char* name);
@@ -68,4 +70,16 @@ class EventTracer {
 
 }  // namespace webrtc
 
+namespace rtc {
+namespace tracing {
+// Set up internal event tracer.
+void SetupInternalTracer();
+bool StartInternalCapture(const char* filename);
+void StartInternalCaptureToFile(FILE* file);
+void StopInternalCapture();
+// Make sure we run this, this will tear down the internal tracing.
+void ShutdownInternalTracer();
+}  // namespace tracing
+}  // namespace rtc
+
 #endif  // WEBRTC_BASE_EVENT_TRACER_H_
diff --git a/media/webrtc/trunk/webrtc/base/event_tracer_unittest.cc b/media/webrtc/trunk/webrtc/base/event_tracer_unittest.cc
index 25ee2543a6..c1b86f2fd8 100644
--- a/media/webrtc/trunk/webrtc/base/event_tracer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/event_tracer_unittest.cc
@@ -11,8 +11,8 @@
 #include "webrtc/base/event_tracer.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/static_instance.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/system_wrappers/include/static_instance.h"
 
 namespace {
 
diff --git a/media/webrtc/trunk/webrtc/base/fakecpumonitor.h b/media/webrtc/trunk/webrtc/base/fakecpumonitor.h
deleted file mode 100644
index c6ea0f2933..0000000000
--- a/media/webrtc/trunk/webrtc/base/fakecpumonitor.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- *  Copyright 2013 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_FAKECPUMONITOR_H_
-#define WEBRTC_BASE_FAKECPUMONITOR_H_
-
-#include "webrtc/base/cpumonitor.h"
-
-namespace rtc {
-
-class FakeCpuMonitor : public rtc::CpuMonitor {
- public:
-  explicit FakeCpuMonitor(Thread* thread)
-      : CpuMonitor(thread) {
-  }
-  ~FakeCpuMonitor() {
-  }
-
-  virtual void OnMessage(rtc::Message* msg) {
-  }
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_FAKECPUMONITOR_H_
diff --git a/media/webrtc/trunk/webrtc/base/fakenetwork.h b/media/webrtc/trunk/webrtc/base/fakenetwork.h
index 60773b4099..e3996e6649 100644
--- a/media/webrtc/trunk/webrtc/base/fakenetwork.h
+++ b/media/webrtc/trunk/webrtc/base/fakenetwork.h
@@ -12,6 +12,7 @@
 #define WEBRTC_BASE_FAKENETWORK_H_
 
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "webrtc/base/network.h"
@@ -29,27 +30,32 @@ const int kFakeIPv6NetworkPrefixLength = 64;
 class FakeNetworkManager : public NetworkManagerBase,
                            public MessageHandler {
  public:
-  FakeNetworkManager()
-      : thread_(Thread::Current()),
-        next_index_(0),
-        started_(false),
-        sent_first_update_(false) {
-  }
+  FakeNetworkManager() : thread_(Thread::Current()) {}
 
-  typedef std::vector<SocketAddress> IfaceList;
+  typedef std::vector<std::pair<SocketAddress, AdapterType>> IfaceList;
 
   void AddInterface(const SocketAddress& iface) {
-    // ensure a unique name for the interface
-    SocketAddress address("test" + rtc::ToString(next_index_++), 0);
+    // Ensure a unique name for the interface if its name is not given.
+    AddInterface(iface, "test" + rtc::ToString(next_index_++));
+  }
+
+  void AddInterface(const SocketAddress& iface, const std::string& if_name) {
+    AddInterface(iface, if_name, ADAPTER_TYPE_UNKNOWN);
+  }
+
+  void AddInterface(const SocketAddress& iface,
+                    const std::string& if_name,
+                    AdapterType type) {
+    SocketAddress address(if_name, 0);
     address.SetResolvedIP(iface.ipaddr());
-    ifaces_.push_back(address);
+    ifaces_.push_back(std::make_pair(address, type));
     DoUpdateNetworks();
   }
 
   void RemoveInterface(const SocketAddress& iface) {
     for (IfaceList::iterator it = ifaces_.begin();
          it != ifaces_.end(); ++it) {
-      if (it->EqualIPs(iface)) {
+      if (it->first.EqualIPs(iface)) {
         ifaces_.erase(it);
         break;
       }
@@ -58,45 +64,46 @@ class FakeNetworkManager : public NetworkManagerBase,
   }
 
   virtual void StartUpdating() {
-    if (started_) {
-      if (sent_first_update_)
+    ++start_count_;
+    if (start_count_ == 1) {
+      sent_first_update_ = false;
+      thread_->Post(this);
+    } else {
+      if (sent_first_update_) {
         SignalNetworksChanged();
-      return;
+      }
     }
-
-    started_ = true;
-    sent_first_update_ = false;
-    thread_->Post(this);
   }
 
-  virtual void StopUpdating() {
-    started_ = false;
-  }
+  virtual void StopUpdating() { --start_count_; }
 
   // MessageHandler interface.
   virtual void OnMessage(Message* msg) {
     DoUpdateNetworks();
   }
 
+  using NetworkManagerBase::set_enumeration_permission;
+  using NetworkManagerBase::set_default_local_addresses;
+
  private:
   void DoUpdateNetworks() {
-    if (!started_)
+    if (start_count_ == 0)
       return;
     std::vector<Network*> networks;
     for (IfaceList::iterator it = ifaces_.begin();
          it != ifaces_.end(); ++it) {
       int prefix_length = 0;
-      if (it->ipaddr().family() == AF_INET) {
+      if (it->first.ipaddr().family() == AF_INET) {
         prefix_length = kFakeIPv4NetworkPrefixLength;
-      } else if (it->ipaddr().family() == AF_INET6) {
+      } else if (it->first.ipaddr().family() == AF_INET6) {
         prefix_length = kFakeIPv6NetworkPrefixLength;
       }
-      IPAddress prefix = TruncateIP(it->ipaddr(), prefix_length);
-      scoped_ptr<Network> net(new Network(it->hostname(),
-                                          it->hostname(),
-                                          prefix,
-                                          prefix_length));
-      net->AddIP(it->ipaddr());
+      IPAddress prefix = TruncateIP(it->first.ipaddr(), prefix_length);
+      scoped_ptr<Network> net(new Network(it->first.hostname(),
+                                          it->first.hostname(), prefix,
+                                          prefix_length, it->second));
+      net->set_default_local_address_provider(this);
+      net->AddIP(it->first.ipaddr());
       networks.push_back(net.release());
     }
     bool changed;
@@ -109,9 +116,12 @@ class FakeNetworkManager : public NetworkManagerBase,
 
   Thread* thread_;
   IfaceList ifaces_;
-  int next_index_;
-  bool started_;
-  bool sent_first_update_;
+  int next_index_ = 0;
+  int start_count_ = 0;
+  bool sent_first_update_ = false;
+
+  IPAddress default_local_ipv4_address_;
+  IPAddress default_local_ipv6_address_;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/fakesslidentity.h b/media/webrtc/trunk/webrtc/base/fakesslidentity.h
index 71a42d545b..ec603a541d 100644
--- a/media/webrtc/trunk/webrtc/base/fakesslidentity.h
+++ b/media/webrtc/trunk/webrtc/base/fakesslidentity.h
@@ -14,6 +14,7 @@
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/common.h"
 #include "webrtc/base/messagedigest.h"
 #include "webrtc/base/sslidentity.h"
 
@@ -24,9 +25,11 @@ class FakeSSLCertificate : public rtc::SSLCertificate {
   // SHA-1 is the default digest algorithm because it is available in all build
   // configurations used for unit testing.
   explicit FakeSSLCertificate(const std::string& data)
-      : data_(data), digest_algorithm_(DIGEST_SHA_1) {}
+      : data_(data), digest_algorithm_(DIGEST_SHA_1), expiration_time_(-1) {}
   explicit FakeSSLCertificate(const std::vector<std::string>& certs)
-      : data_(certs.front()), digest_algorithm_(DIGEST_SHA_1) {
+      : data_(certs.front()),
+        digest_algorithm_(DIGEST_SHA_1),
+        expiration_time_(-1) {
     std::vector<std::string>::const_iterator it;
     // Skip certs[0].
     for (it = certs.begin() + 1; it != certs.end(); ++it) {
@@ -44,6 +47,12 @@ class FakeSSLCertificate : public rtc::SSLCertificate {
     VERIFY(SSLIdentity::PemToDer(kPemTypeCertificate, data_, &der_string));
     der_buffer->SetData(der_string.c_str(), der_string.size());
   }
+  int64_t CertificateExpirationTime() const override {
+    return expiration_time_;
+  }
+  void SetCertificateExpirationTime(int64_t expiration_time) {
+    expiration_time_ = expiration_time;
+  }
   void set_digest_algorithm(const std::string& algorithm) {
     digest_algorithm_ = algorithm;
   }
@@ -77,6 +86,8 @@ class FakeSSLCertificate : public rtc::SSLCertificate {
   std::string data_;
   std::vector<FakeSSLCertificate> certs_;
   std::string digest_algorithm_;
+  // Expiration time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC).
+  int64_t expiration_time_;
 };
 
 class FakeSSLIdentity : public rtc::SSLIdentity {
diff --git a/media/webrtc/trunk/webrtc/base/faketaskrunner.h b/media/webrtc/trunk/webrtc/base/faketaskrunner.h
index 5408ab8b2c..88e48261b3 100644
--- a/media/webrtc/trunk/webrtc/base/faketaskrunner.h
+++ b/media/webrtc/trunk/webrtc/base/faketaskrunner.h
@@ -25,12 +25,12 @@ class FakeTaskRunner : public TaskRunner {
 
   virtual void WakeTasks() { RunTasks(); }
 
-  virtual int64 CurrentTime() {
+  virtual int64_t CurrentTime() {
     // Implement if needed.
     return current_time_++;
   }
 
-  int64 current_time_;
+  int64_t current_time_;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/filelock.cc b/media/webrtc/trunk/webrtc/base/filelock.cc
deleted file mode 100644
index fc921febcd..0000000000
--- a/media/webrtc/trunk/webrtc/base/filelock.cc
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/filelock.h"
-
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/stream.h"
-
-namespace rtc {
-
-FileLock::FileLock(const std::string& path, FileStream* file)
-    : path_(path), file_(file) {
-}
-
-FileLock::~FileLock() {
-  MaybeUnlock();
-}
-
-void FileLock::Unlock() {
-  LOG_F(LS_INFO);
-  MaybeUnlock();
-}
-
-void FileLock::MaybeUnlock() {
-  if (file_) {
-    LOG(LS_INFO) << "Unlocking:" << path_;
-    file_->Close();
-    Filesystem::DeleteFile(path_);
-    file_.reset();
-  }
-}
-
-FileLock* FileLock::TryLock(const std::string& path) {
-  FileStream* stream = new FileStream();
-  bool ok = false;
-#if defined(WEBRTC_WIN)
-  // Open and lock in a single operation.
-  ok = stream->OpenShare(path, "a", _SH_DENYRW, NULL);
-#else // WEBRTC_LINUX && !WEBRTC_ANDROID and WEBRTC_MAC && !defined(WEBRTC_IOS)
-  ok = stream->Open(path, "a", NULL) && stream->TryLock();
-#endif
-  if (ok) {
-    return new FileLock(path, stream);
-  } else {
-    // Something failed, either we didn't succeed to open the
-    // file or we failed to lock it. Anyway remove the heap
-    // allocated object and then return NULL to indicate failure.
-    delete stream;
-    return NULL;
-  }
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/filelock.h b/media/webrtc/trunk/webrtc/base/filelock.h
deleted file mode 100644
index 46c58ea4ab..0000000000
--- a/media/webrtc/trunk/webrtc/base/filelock.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_FILELOCK_H_
-#define WEBRTC_BASE_FILELOCK_H_
-
-#include <string>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace rtc {
-
-class FileStream;
-
-// Implements a very simple cross process lock based on a file.
-// When Lock(...) is called we try to open/create the file in read/write
-// mode without any sharing. (Or locking it with flock(...) on Unix)
-// If the process crash the OS will make sure that the file descriptor
-// is released and another process can accuire the lock.
-// This doesn't work on ancient OSX/Linux versions if used on NFS.
-// (Nfs-client before: ~2.6 and Linux Kernel < 2.6.)
-class FileLock {
- public:
-  virtual ~FileLock();
-
-  // Attempts to lock the file. The caller owns the returned
-  // lock object. Returns NULL if the file already was locked.
-  static FileLock* TryLock(const std::string& path);
-  void Unlock();
-
- protected:
-  FileLock(const std::string& path, FileStream* file);
-
- private:
-  void MaybeUnlock();
-
-  std::string path_;
-  scoped_ptr<FileStream> file_;
-
-  DISALLOW_EVIL_CONSTRUCTORS(FileLock);
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_FILELOCK_H_
diff --git a/media/webrtc/trunk/webrtc/base/filelock_unittest.cc b/media/webrtc/trunk/webrtc/base/filelock_unittest.cc
deleted file mode 100644
index cf2d3fef9b..0000000000
--- a/media/webrtc/trunk/webrtc/base/filelock_unittest.cc
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-
-#include "webrtc/base/event.h"
-#include "webrtc/base/filelock.h"
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace rtc {
-
-const static std::string kLockFile = "TestLockFile";
-const static int kTimeoutMS = 5000;
-
-class FileLockTest : public testing::Test, public Runnable {
- public:
-  FileLockTest() : done_(false, false), thread_lock_failed_(false) {
-  }
-
-  virtual void Run(Thread* t) {
-    scoped_ptr<FileLock> lock(FileLock::TryLock(temp_file_.pathname()));
-    // The lock is already owned by the main thread of
-    // this test, therefore the TryLock(...) call should fail.
-    thread_lock_failed_ = lock.get() == NULL;
-    done_.Set();
-  }
-
- protected:
-  virtual void SetUp() {
-    thread_lock_failed_ = false;
-    Pathname temp_dir;
-    Filesystem::GetAppTempFolder(&temp_dir);
-    temp_file_.SetPathname(rtc::Filesystem::TempFilename(temp_dir, kLockFile));
-  }
-
-  void LockOnThread() {
-    locker_.Start(this);
-    done_.Wait(kTimeoutMS);
-  }
-
-  Event done_;
-  Thread locker_;
-  bool thread_lock_failed_;
-  Pathname temp_file_;
-};
-
-TEST_F(FileLockTest, TestLockFileDeleted) {
-  scoped_ptr<FileLock> lock(FileLock::TryLock(temp_file_.pathname()));
-  EXPECT_TRUE(lock.get() != NULL);
-  EXPECT_FALSE(Filesystem::IsAbsent(temp_file_.pathname()));
-  lock->Unlock();
-  EXPECT_TRUE(Filesystem::IsAbsent(temp_file_.pathname()));
-}
-
-TEST_F(FileLockTest, TestLock) {
-  scoped_ptr<FileLock> lock(FileLock::TryLock(temp_file_.pathname()));
-  EXPECT_TRUE(lock.get() != NULL);
-}
-
-TEST_F(FileLockTest, TestLockX2) {
-  scoped_ptr<FileLock> lock1(FileLock::TryLock(temp_file_.pathname()));
-  EXPECT_TRUE(lock1.get() != NULL);
-
-  scoped_ptr<FileLock> lock2(FileLock::TryLock(temp_file_.pathname()));
-  EXPECT_TRUE(lock2.get() == NULL);
-}
-
-TEST_F(FileLockTest, TestThreadedLock) {
-  scoped_ptr<FileLock> lock(FileLock::TryLock(temp_file_.pathname()));
-  EXPECT_TRUE(lock.get() != NULL);
-
-  LockOnThread();
-  EXPECT_TRUE(thread_lock_failed_);
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/filerotatingstream.cc b/media/webrtc/trunk/webrtc/base/filerotatingstream.cc
new file mode 100644
index 0000000000..080999476b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/filerotatingstream.cc
@@ -0,0 +1,400 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/filerotatingstream.h"
+
+#include <algorithm>
+#include <iostream>
+#include <string>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/fileutils.h"
+#include "webrtc/base/pathutils.h"
+
+// Note: We use std::cerr for logging in the write paths of this stream to avoid
+// infinite loops when logging.
+
+namespace rtc {
+
+FileRotatingStream::FileRotatingStream(const std::string& dir_path,
+                                       const std::string& file_prefix)
+    : FileRotatingStream(dir_path, file_prefix, 0, 0, kRead) {
+}
+
+FileRotatingStream::FileRotatingStream(const std::string& dir_path,
+                                       const std::string& file_prefix,
+                                       size_t max_file_size,
+                                       size_t num_files)
+    : FileRotatingStream(dir_path,
+                         file_prefix,
+                         max_file_size,
+                         num_files,
+                         kWrite) {
+  RTC_DCHECK_GT(max_file_size, 0u);
+  RTC_DCHECK_GT(num_files, 1u);
+}
+
+FileRotatingStream::FileRotatingStream(const std::string& dir_path,
+                                       const std::string& file_prefix,
+                                       size_t max_file_size,
+                                       size_t num_files,
+                                       Mode mode)
+    : dir_path_(dir_path),
+      file_prefix_(file_prefix),
+      mode_(mode),
+      file_stream_(nullptr),
+      max_file_size_(max_file_size),
+      current_file_index_(0),
+      rotation_index_(0),
+      current_bytes_written_(0),
+      disable_buffering_(false) {
+  RTC_DCHECK(Filesystem::IsFolder(dir_path));
+  switch (mode) {
+    case kWrite: {
+      file_names_.clear();
+      for (size_t i = 0; i < num_files; ++i) {
+        file_names_.push_back(GetFilePath(i, num_files));
+      }
+      rotation_index_ = num_files - 1;
+      break;
+    }
+    case kRead: {
+      file_names_ = GetFilesWithPrefix();
+      std::sort(file_names_.begin(), file_names_.end());
+      if (file_names_.size() > 0) {
+        // |file_names_| is sorted newest first, so read from the end.
+        current_file_index_ = file_names_.size() - 1;
+      }
+      break;
+    }
+  }
+}
+
+FileRotatingStream::~FileRotatingStream() {
+}
+
+StreamState FileRotatingStream::GetState() const {
+  if (mode_ == kRead && current_file_index_ < file_names_.size()) {
+    return SS_OPEN;
+  }
+  if (!file_stream_) {
+    return SS_CLOSED;
+  }
+  return file_stream_->GetState();
+}
+
+StreamResult FileRotatingStream::Read(void* buffer,
+                                      size_t buffer_len,
+                                      size_t* read,
+                                      int* error) {
+  RTC_DCHECK(buffer);
+  if (mode_ != kRead) {
+    return SR_EOS;
+  }
+  if (current_file_index_ >= file_names_.size()) {
+    return SR_EOS;
+  }
+  // We will have no file stream initially, and when we are finished with the
+  // previous file.
+  if (!file_stream_) {
+    if (!OpenCurrentFile()) {
+      return SR_ERROR;
+    }
+  }
+  int local_error = 0;
+  if (!error) {
+    error = &local_error;
+  }
+  StreamResult result = file_stream_->Read(buffer, buffer_len, read, error);
+  if (result == SR_EOS || result == SR_ERROR) {
+    if (result == SR_ERROR) {
+      LOG(LS_ERROR) << "Failed to read from: "
+                    << file_names_[current_file_index_] << "Error: " << error;
+    }
+    // Reached the end of the file, read next file. If there is an error return
+    // the error status but allow for a next read by reading next file.
+    CloseCurrentFile();
+    if (current_file_index_ == 0) {
+      // Just finished reading the last file, signal EOS by setting index.
+      current_file_index_ = file_names_.size();
+    } else {
+      --current_file_index_;
+    }
+    if (read) {
+      *read = 0;
+    }
+    return result == SR_EOS ? SR_SUCCESS : result;
+  } else if (result == SR_SUCCESS) {
+    // Succeeded, continue reading from this file.
+    return SR_SUCCESS;
+  } else {
+    RTC_NOTREACHED();
+  }
+  return result;
+}
+
+StreamResult FileRotatingStream::Write(const void* data,
+                                       size_t data_len,
+                                       size_t* written,
+                                       int* error) {
+  if (mode_ != kWrite) {
+    return SR_EOS;
+  }
+  if (!file_stream_) {
+    std::cerr << "Open() must be called before Write." << std::endl;
+    return SR_ERROR;
+  }
+  // Write as much as will fit in to the current file.
+  RTC_DCHECK_LT(current_bytes_written_, max_file_size_);
+  size_t remaining_bytes = max_file_size_ - current_bytes_written_;
+  size_t write_length = std::min(data_len, remaining_bytes);
+  size_t local_written = 0;
+  if (!written) {
+    written = &local_written;
+  }
+  StreamResult result = file_stream_->Write(data, write_length, written, error);
+  current_bytes_written_ += *written;
+
+  // If we're done with this file, rotate it out.
+  if (current_bytes_written_ >= max_file_size_) {
+    RTC_DCHECK_EQ(current_bytes_written_, max_file_size_);
+    RotateFiles();
+  }
+  return result;
+}
+
+bool FileRotatingStream::Flush() {
+  if (!file_stream_) {
+    return false;
+  }
+  return file_stream_->Flush();
+}
+
+bool FileRotatingStream::GetSize(size_t* size) const {
+  if (mode_ != kRead) {
+    // Not possible to get accurate size on disk when writing because of
+    // potential buffering.
+    return false;
+  }
+  RTC_DCHECK(size);
+  *size = 0;
+  size_t total_size = 0;
+  for (auto file_name : file_names_) {
+    Pathname pathname(file_name);
+    size_t file_size = 0;
+    if (Filesystem::GetFileSize(file_name, &file_size)) {
+      total_size += file_size;
+    }
+  }
+  *size = total_size;
+  return true;
+}
+
+void FileRotatingStream::Close() {
+  CloseCurrentFile();
+}
+
+bool FileRotatingStream::Open() {
+  switch (mode_) {
+    case kRead:
+      // Defer opening to when we first read since we want to return read error
+      // if we fail to open next file.
+      return true;
+    case kWrite: {
+      // Delete existing files when opening for write.
+      std::vector<std::string> matching_files = GetFilesWithPrefix();
+      for (auto matching_file : matching_files) {
+        if (!Filesystem::DeleteFile(matching_file)) {
+          std::cerr << "Failed to delete: " << matching_file << std::endl;
+        }
+      }
+      return OpenCurrentFile();
+    }
+  }
+  return false;
+}
+
+bool FileRotatingStream::DisableBuffering() {
+  disable_buffering_ = true;
+  if (!file_stream_) {
+    std::cerr << "Open() must be called before DisableBuffering()."
+              << std::endl;
+    return false;
+  }
+  return file_stream_->DisableBuffering();
+}
+
+std::string FileRotatingStream::GetFilePath(size_t index) const {
+  RTC_DCHECK_LT(index, file_names_.size());
+  return file_names_[index];
+}
+
+bool FileRotatingStream::OpenCurrentFile() {
+  CloseCurrentFile();
+
+  // Opens the appropriate file in the appropriate mode.
+  RTC_DCHECK_LT(current_file_index_, file_names_.size());
+  std::string file_path = file_names_[current_file_index_];
+  file_stream_.reset(new FileStream());
+  const char* mode = nullptr;
+  switch (mode_) {
+    case kWrite:
+      mode = "w+";
+      // We should always we writing to the zero-th file.
+      RTC_DCHECK_EQ(current_file_index_, 0u);
+      break;
+    case kRead:
+      mode = "r";
+      break;
+  }
+  int error = 0;
+  if (!file_stream_->Open(file_path, mode, &error)) {
+    std::cerr << "Failed to open: " << file_path << "Error: " << error
+              << std::endl;
+    file_stream_.reset();
+    return false;
+  }
+  if (disable_buffering_) {
+    file_stream_->DisableBuffering();
+  }
+  return true;
+}
+
+void FileRotatingStream::CloseCurrentFile() {
+  if (!file_stream_) {
+    return;
+  }
+  current_bytes_written_ = 0;
+  file_stream_.reset();
+}
+
+void FileRotatingStream::RotateFiles() {
+  RTC_DCHECK_EQ(mode_, kWrite);
+  CloseCurrentFile();
+  // Rotates the files by deleting the file at |rotation_index_|, which is the
+  // oldest file and then renaming the newer files to have an incremented index.
+  // See header file comments for example.
+  RTC_DCHECK_LT(rotation_index_, file_names_.size());
+  std::string file_to_delete = file_names_[rotation_index_];
+  if (Filesystem::IsFile(file_to_delete)) {
+    if (!Filesystem::DeleteFile(file_to_delete)) {
+      std::cerr << "Failed to delete: " << file_to_delete << std::endl;
+    }
+  }
+  for (auto i = rotation_index_; i > 0; --i) {
+    std::string rotated_name = file_names_[i];
+    std::string unrotated_name = file_names_[i - 1];
+    if (Filesystem::IsFile(unrotated_name)) {
+      if (!Filesystem::MoveFile(unrotated_name, rotated_name)) {
+        std::cerr << "Failed to move: " << unrotated_name << " to "
+                  << rotated_name << std::endl;
+      }
+    }
+  }
+  // Create a new file for 0th index.
+  OpenCurrentFile();
+  OnRotation();
+}
+
+std::vector<std::string> FileRotatingStream::GetFilesWithPrefix() const {
+  std::vector<std::string> files;
+  // Iterate over the files in the directory.
+  DirectoryIterator it;
+  Pathname dir_path;
+  dir_path.SetFolder(dir_path_);
+  if (!it.Iterate(dir_path)) {
+    return files;
+  }
+  do {
+    std::string current_name = it.Name();
+    if (current_name.size() && !it.IsDirectory() &&
+        current_name.compare(0, file_prefix_.size(), file_prefix_) == 0) {
+      Pathname path(dir_path_, current_name);
+      files.push_back(path.pathname());
+    }
+  } while (it.Next());
+  return files;
+}
+
+std::string FileRotatingStream::GetFilePath(size_t index,
+                                            size_t num_files) const {
+  RTC_DCHECK_LT(index, num_files);
+  std::ostringstream file_name;
+  // The format will be "_%<num_digits>zu". We want to zero pad the index so
+  // that it will sort nicely.
+  size_t max_digits = ((num_files - 1) / 10) + 1;
+  size_t num_digits = (index / 10) + 1;
+  RTC_DCHECK_LE(num_digits, max_digits);
+  size_t padding = max_digits - num_digits;
+
+  file_name << file_prefix_ << "_";
+  for (size_t i = 0; i < padding; ++i) {
+    file_name << "0";
+  }
+  file_name << index;
+
+  Pathname file_path(dir_path_, file_name.str());
+  return file_path.pathname();
+}
+
+CallSessionFileRotatingStream::CallSessionFileRotatingStream(
+    const std::string& dir_path)
+    : FileRotatingStream(dir_path, kLogPrefix),
+      max_total_log_size_(0),
+      num_rotations_(0) {
+}
+
+CallSessionFileRotatingStream::CallSessionFileRotatingStream(
+    const std::string& dir_path,
+    size_t max_total_log_size)
+    : FileRotatingStream(dir_path,
+                         kLogPrefix,
+                         max_total_log_size / 2,
+                         GetNumRotatingLogFiles(max_total_log_size) + 1),
+      max_total_log_size_(max_total_log_size),
+      num_rotations_(0) {
+  RTC_DCHECK_GE(max_total_log_size, 4u);
+}
+
+const char* CallSessionFileRotatingStream::kLogPrefix = "webrtc_log";
+const size_t CallSessionFileRotatingStream::kRotatingLogFileDefaultSize =
+    1024 * 1024;
+
+void CallSessionFileRotatingStream::OnRotation() {
+  ++num_rotations_;
+  if (num_rotations_ == 1) {
+    // On the first rotation adjust the max file size so subsequent files after
+    // the first are smaller.
+    SetMaxFileSize(GetRotatingLogSize(max_total_log_size_));
+  } else if (num_rotations_ == (GetNumFiles() - 1)) {
+    // On the next rotation the very first file is going to be deleted. Change
+    // the rotation index so this doesn't happen.
+    SetRotationIndex(GetRotationIndex() - 1);
+  }
+}
+
+size_t CallSessionFileRotatingStream::GetRotatingLogSize(
+    size_t max_total_log_size) {
+  size_t num_rotating_log_files = GetNumRotatingLogFiles(max_total_log_size);
+  size_t rotating_log_size = num_rotating_log_files > 2
+                                 ? kRotatingLogFileDefaultSize
+                                 : max_total_log_size / 4;
+  return rotating_log_size;
+}
+
+size_t CallSessionFileRotatingStream::GetNumRotatingLogFiles(
+    size_t max_total_log_size) {
+  // At minimum have two rotating files. Otherwise split the available log size
+  // evenly across 1MB files.
+  return std::max((size_t)2,
+                  (max_total_log_size / 2) / kRotatingLogFileDefaultSize);
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/filerotatingstream.h b/media/webrtc/trunk/webrtc/base/filerotatingstream.h
new file mode 100644
index 0000000000..9e8e35ddd7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/filerotatingstream.h
@@ -0,0 +1,172 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FILEROTATINGSTREAM_H_
+#define WEBRTC_BASE_FILEROTATINGSTREAM_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/stream.h"
+
+namespace rtc {
+
+// FileRotatingStream writes to a file in the directory specified in the
+// constructor. It rotates the files once the current file is full. The
+// individual file size and the number of files used is configurable in the
+// constructor. Open() must be called before using this stream.
+class FileRotatingStream : public StreamInterface {
+ public:
+  // Use this constructor for reading a directory previously written to with
+  // this stream.
+  FileRotatingStream(const std::string& dir_path,
+                     const std::string& file_prefix);
+
+  // Use this constructor for writing to a directory. Files in the directory
+  // matching the prefix will be deleted on open.
+  FileRotatingStream(const std::string& dir_path,
+                     const std::string& file_prefix,
+                     size_t max_file_size,
+                     size_t num_files);
+
+  ~FileRotatingStream() override;
+
+  // StreamInterface methods.
+  StreamState GetState() const override;
+  StreamResult Read(void* buffer,
+                    size_t buffer_len,
+                    size_t* read,
+                    int* error) override;
+  StreamResult Write(const void* data,
+                     size_t data_len,
+                     size_t* written,
+                     int* error) override;
+  bool Flush() override;
+  // Returns the total file size currently used on disk.
+  bool GetSize(size_t* size) const override;
+  void Close() override;
+
+  // Opens the appropriate file(s). Call this before using the stream.
+  bool Open();
+
+  // Disabling buffering causes writes to block until disk is updated. This is
+  // enabled by default for performance.
+  bool DisableBuffering();
+
+  // Returns the path used for the i-th newest file, where the 0th file is the
+  // newest file. The file may or may not exist, this is just used for
+  // formatting. Index must be less than GetNumFiles().
+  std::string GetFilePath(size_t index) const;
+
+  // Returns the number of files that will used by this stream.
+  size_t GetNumFiles() { return file_names_.size(); }
+
+ protected:
+  size_t GetMaxFileSize() const { return max_file_size_; }
+
+  void SetMaxFileSize(size_t size) { max_file_size_ = size; }
+
+  size_t GetRotationIndex() const { return rotation_index_; }
+
+  void SetRotationIndex(size_t index) { rotation_index_ = index; }
+
+  virtual void OnRotation() {}
+
+ private:
+  enum Mode { kRead, kWrite };
+
+  FileRotatingStream(const std::string& dir_path,
+                     const std::string& file_prefix,
+                     size_t max_file_size,
+                     size_t num_files,
+                     Mode mode);
+
+  bool OpenCurrentFile();
+  void CloseCurrentFile();
+
+  // Rotates the files by creating a new current file, renaming the
+  // existing files, and deleting the oldest one. e.g.
+  // file_0 -> file_1
+  // file_1 -> file_2
+  // file_2 -> delete
+  // create new file_0
+  void RotateFiles();
+
+  // Returns a list of file names in the directory beginning with the prefix.
+  std::vector<std::string> GetFilesWithPrefix() const;
+  // Private version of GetFilePath.
+  std::string GetFilePath(size_t index, size_t num_files) const;
+
+  const std::string dir_path_;
+  const std::string file_prefix_;
+  const Mode mode_;
+
+  // FileStream is used to write to the current file.
+  scoped_ptr<FileStream> file_stream_;
+  // Convenience storage for file names so we don't generate them over and over.
+  std::vector<std::string> file_names_;
+  size_t max_file_size_;
+  size_t current_file_index_;
+  // The rotation index indicates the index of the file that will be
+  // deleted first on rotation. Indices lower than this index will be rotated.
+  size_t rotation_index_;
+  // Number of bytes written to current file. We need this because with
+  // buffering the file size read from disk might not be accurate.
+  size_t current_bytes_written_;
+  bool disable_buffering_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(FileRotatingStream);
+};
+
+// CallSessionFileRotatingStream is meant to be used in situations where we will
+// have limited disk space. Its purpose is to read and write logs up to a
+// maximum size. Once the maximum size is exceeded, logs from the middle are
+// deleted whereas logs from the beginning and end are preserved. The reason for
+// this is because we anticipate that in WebRTC the beginning and end of the
+// logs are most useful for call diagnostics.
+//
+// This implementation simply writes to a single file until
+// |max_total_log_size| / 2 bytes are written to it, and subsequently writes to
+// a set of rotating files. We do this by inheriting FileRotatingStream and
+// setting the appropriate internal variables so that we don't delete the last
+// (earliest) file on rotate, and that that file's size is bigger.
+//
+// Open() must be called before using this stream.
+class CallSessionFileRotatingStream : public FileRotatingStream {
+ public:
+  // Use this constructor for reading a directory previously written to with
+  // this stream.
+  explicit CallSessionFileRotatingStream(const std::string& dir_path);
+  // Use this constructor for writing to a directory. Files in the directory
+  // matching what's used by the stream will be deleted. |max_total_log_size|
+  // must be at least 4.
+  CallSessionFileRotatingStream(const std::string& dir_path,
+                                size_t max_total_log_size);
+  ~CallSessionFileRotatingStream() override {}
+
+ protected:
+  void OnRotation() override;
+
+ private:
+  static size_t GetRotatingLogSize(size_t max_total_log_size);
+  static size_t GetNumRotatingLogFiles(size_t max_total_log_size);
+  static const char* kLogPrefix;
+  static const size_t kRotatingLogFileDefaultSize;
+
+  const size_t max_total_log_size_;
+  size_t num_rotations_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(CallSessionFileRotatingStream);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_FILEROTATINGSTREAM_H_
diff --git a/media/webrtc/trunk/webrtc/base/filerotatingstream_unittest.cc b/media/webrtc/trunk/webrtc/base/filerotatingstream_unittest.cc
new file mode 100644
index 0000000000..09438f870e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/filerotatingstream_unittest.cc
@@ -0,0 +1,316 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/filerotatingstream.h"
+#include "webrtc/base/fileutils.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/pathutils.h"
+
+namespace rtc {
+
+class FileRotatingStreamTest : public ::testing::Test {
+ protected:
+  static const char* kFilePrefix;
+  static const size_t kMaxFileSize;
+
+  void Init(const std::string& dir_name,
+            const std::string& file_prefix,
+            size_t max_file_size,
+            size_t num_log_files) {
+    Pathname test_path;
+    ASSERT_TRUE(Filesystem::GetAppTempFolder(&test_path));
+    // Append per-test output path in order to run within gtest parallel.
+    test_path.AppendFolder(dir_name);
+    ASSERT_TRUE(Filesystem::CreateFolder(test_path));
+    dir_path_ = test_path.pathname();
+    ASSERT_TRUE(dir_path_.size());
+    stream_.reset(new FileRotatingStream(dir_path_, file_prefix, max_file_size,
+                                         num_log_files));
+  }
+
+  void TearDown() override {
+    stream_.reset();
+    if (dir_path_.size() && Filesystem::IsFolder(dir_path_) &&
+        Filesystem::IsTemporaryPath(dir_path_)) {
+      Filesystem::DeleteFolderAndContents(dir_path_);
+    }
+  }
+
+  // Writes the data to the stream and flushes it.
+  void WriteAndFlush(const void* data, const size_t data_len) {
+    EXPECT_EQ(SR_SUCCESS, stream_->WriteAll(data, data_len, nullptr, nullptr));
+    EXPECT_TRUE(stream_->Flush());
+  }
+
+  // Checks that the stream reads in the expected contents and then returns an
+  // end of stream result.
+  void VerifyStreamRead(const char* expected_contents,
+                        const size_t expected_length,
+                        const std::string& dir_path,
+                        const char* file_prefix) {
+    scoped_ptr<FileRotatingStream> stream;
+    stream.reset(new FileRotatingStream(dir_path, file_prefix));
+    ASSERT_TRUE(stream->Open());
+    size_t read = 0;
+    size_t stream_size = 0;
+    EXPECT_TRUE(stream->GetSize(&stream_size));
+    scoped_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
+    EXPECT_EQ(SR_SUCCESS,
+              stream->ReadAll(buffer.get(), expected_length, &read, nullptr));
+    EXPECT_EQ(0, memcmp(expected_contents, buffer.get(), expected_length));
+    EXPECT_EQ(SR_EOS, stream->ReadAll(buffer.get(), 1, nullptr, nullptr));
+    EXPECT_EQ(stream_size, read);
+  }
+
+  void VerifyFileContents(const char* expected_contents,
+                          const size_t expected_length,
+                          const std::string& file_path) {
+    scoped_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
+    scoped_ptr<FileStream> stream(Filesystem::OpenFile(file_path, "r"));
+    EXPECT_TRUE(stream);
+    if (!stream) {
+      return;
+    }
+    EXPECT_EQ(rtc::SR_SUCCESS,
+              stream->ReadAll(buffer.get(), expected_length, nullptr, nullptr));
+    EXPECT_EQ(0, memcmp(expected_contents, buffer.get(), expected_length));
+    size_t file_size = 0;
+    EXPECT_TRUE(stream->GetSize(&file_size));
+    EXPECT_EQ(file_size, expected_length);
+  }
+
+  scoped_ptr<FileRotatingStream> stream_;
+  std::string dir_path_;
+};
+
+const char* FileRotatingStreamTest::kFilePrefix = "FileRotatingStreamTest";
+const size_t FileRotatingStreamTest::kMaxFileSize = 2;
+
+// Tests that stream state is correct before and after Open / Close.
+TEST_F(FileRotatingStreamTest, State) {
+  Init("FileRotatingStreamTestState", kFilePrefix, kMaxFileSize, 3);
+
+  EXPECT_EQ(SS_CLOSED, stream_->GetState());
+  ASSERT_TRUE(stream_->Open());
+  EXPECT_EQ(SS_OPEN, stream_->GetState());
+  stream_->Close();
+  EXPECT_EQ(SS_CLOSED, stream_->GetState());
+}
+
+// Tests that nothing is written to file when data of length zero is written.
+TEST_F(FileRotatingStreamTest, EmptyWrite) {
+  Init("FileRotatingStreamTestEmptyWrite", kFilePrefix, kMaxFileSize, 3);
+
+  ASSERT_TRUE(stream_->Open());
+  WriteAndFlush("a", 0);
+
+  std::string logfile_path = stream_->GetFilePath(0);
+  scoped_ptr<FileStream> stream(Filesystem::OpenFile(logfile_path, "r"));
+  size_t file_size = 0;
+  EXPECT_TRUE(stream->GetSize(&file_size));
+  EXPECT_EQ(0u, file_size);
+}
+
+// Tests that a write operation followed by a read returns the expected data
+// and writes to the expected files.
+TEST_F(FileRotatingStreamTest, WriteAndRead) {
+  Init("FileRotatingStreamTestWriteAndRead", kFilePrefix, kMaxFileSize, 3);
+
+  ASSERT_TRUE(stream_->Open());
+  // The test is set up to create three log files of length 2. Write and check
+  // contents.
+  std::string messages[3] = {"aa", "bb", "cc"};
+  for (size_t i = 0; i < arraysize(messages); ++i) {
+    const std::string& message = messages[i];
+    WriteAndFlush(message.c_str(), message.size());
+    // Since the max log size is 2, we will be causing rotation. Read from the
+    // next file.
+    VerifyFileContents(message.c_str(), message.size(),
+                       stream_->GetFilePath(1));
+  }
+  // Check that exactly three files exist.
+  for (size_t i = 0; i < arraysize(messages); ++i) {
+    EXPECT_TRUE(Filesystem::IsFile(stream_->GetFilePath(i)));
+  }
+  std::string message("d");
+  WriteAndFlush(message.c_str(), message.size());
+  for (size_t i = 0; i < arraysize(messages); ++i) {
+    EXPECT_TRUE(Filesystem::IsFile(stream_->GetFilePath(i)));
+  }
+  // TODO(tkchin): Maybe check all the files in the dir.
+
+  // Reopen for read.
+  std::string expected_contents("bbccd");
+  VerifyStreamRead(expected_contents.c_str(), expected_contents.size(),
+                   dir_path_, kFilePrefix);
+}
+
+// Tests that writing data greater than the total capacity of the files
+// overwrites the files correctly and is read correctly after.
+TEST_F(FileRotatingStreamTest, WriteOverflowAndRead) {
+  Init("FileRotatingStreamTestWriteOverflowAndRead", kFilePrefix, kMaxFileSize,
+       3);
+  ASSERT_TRUE(stream_->Open());
+  // This should cause overflow across all three files, such that the first file
+  // we wrote to also gets overwritten.
+  std::string message("foobarbaz");
+  WriteAndFlush(message.c_str(), message.size());
+  std::string expected_file_contents("z");
+  VerifyFileContents(expected_file_contents.c_str(),
+                     expected_file_contents.size(), stream_->GetFilePath(0));
+  std::string expected_stream_contents("arbaz");
+  VerifyStreamRead(expected_stream_contents.c_str(),
+                   expected_stream_contents.size(), dir_path_, kFilePrefix);
+}
+
+// Tests that the returned file paths have the right folder and prefix.
+TEST_F(FileRotatingStreamTest, GetFilePath) {
+  Init("FileRotatingStreamTestGetFilePath", kFilePrefix, kMaxFileSize, 20);
+  for (auto i = 0; i < 20; ++i) {
+    Pathname path(stream_->GetFilePath(i));
+    EXPECT_EQ(0, path.folder().compare(dir_path_));
+    EXPECT_EQ(0, path.filename().compare(0, strlen(kFilePrefix), kFilePrefix));
+  }
+}
+
+class CallSessionFileRotatingStreamTest : public ::testing::Test {
+ protected:
+  void Init(const std::string& dir_name, size_t max_total_log_size) {
+    Pathname test_path;
+    ASSERT_TRUE(Filesystem::GetAppTempFolder(&test_path));
+    // Append per-test output path in order to run within gtest parallel.
+    test_path.AppendFolder(dir_name);
+    ASSERT_TRUE(Filesystem::CreateFolder(test_path));
+    dir_path_ = test_path.pathname();
+    ASSERT_TRUE(dir_path_.size());
+    stream_.reset(
+        new CallSessionFileRotatingStream(dir_path_, max_total_log_size));
+  }
+
+  virtual void TearDown() {
+    stream_.reset();
+    if (dir_path_.size() && Filesystem::IsFolder(dir_path_) &&
+        Filesystem::IsTemporaryPath(dir_path_)) {
+      Filesystem::DeleteFolderAndContents(dir_path_);
+    }
+  }
+
+  // Writes the data to the stream and flushes it.
+  void WriteAndFlush(const void* data, const size_t data_len) {
+    EXPECT_EQ(SR_SUCCESS, stream_->WriteAll(data, data_len, nullptr, nullptr));
+    EXPECT_TRUE(stream_->Flush());
+  }
+
+  // Checks that the stream reads in the expected contents and then returns an
+  // end of stream result.
+  void VerifyStreamRead(const char* expected_contents,
+                        const size_t expected_length,
+                        const std::string& dir_path) {
+    scoped_ptr<CallSessionFileRotatingStream> stream(
+        new CallSessionFileRotatingStream(dir_path));
+    ASSERT_TRUE(stream->Open());
+    size_t read = 0;
+    size_t stream_size = 0;
+    EXPECT_TRUE(stream->GetSize(&stream_size));
+    scoped_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
+    EXPECT_EQ(SR_SUCCESS,
+              stream->ReadAll(buffer.get(), expected_length, &read, nullptr));
+    EXPECT_EQ(0, memcmp(expected_contents, buffer.get(), expected_length));
+    EXPECT_EQ(SR_EOS, stream->ReadAll(buffer.get(), 1, nullptr, nullptr));
+    EXPECT_EQ(stream_size, read);
+  }
+
+  scoped_ptr<CallSessionFileRotatingStream> stream_;
+  std::string dir_path_;
+};
+
+// Tests that writing and reading to a stream with the smallest possible
+// capacity works.
+TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadSmallest) {
+  Init("CallSessionFileRotatingStreamTestWriteAndReadSmallest", 4);
+
+  ASSERT_TRUE(stream_->Open());
+  std::string message("abcde");
+  WriteAndFlush(message.c_str(), message.size());
+  std::string expected_contents("abe");
+  VerifyStreamRead(expected_contents.c_str(), expected_contents.size(),
+                   dir_path_);
+}
+
+// Tests that writing and reading to a stream with capacity lesser than 4MB
+// behaves correctly.
+TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadSmall) {
+  Init("CallSessionFileRotatingStreamTestWriteAndReadSmall", 8);
+
+  ASSERT_TRUE(stream_->Open());
+  std::string message("123456789");
+  WriteAndFlush(message.c_str(), message.size());
+  std::string expected_contents("1234789");
+  VerifyStreamRead(expected_contents.c_str(), expected_contents.size(),
+                   dir_path_);
+}
+
+// Tests that writing and reading to a stream with capacity greater than 4MB
+// behaves correctly.
+TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadLarge) {
+  Init("CallSessionFileRotatingStreamTestWriteAndReadLarge", 6 * 1024 * 1024);
+
+  ASSERT_TRUE(stream_->Open());
+  const size_t buffer_size = 1024 * 1024;
+  scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  for (int i = 0; i < 8; i++) {
+    memset(buffer.get(), i, buffer_size);
+    EXPECT_EQ(SR_SUCCESS,
+              stream_->WriteAll(buffer.get(), buffer_size, nullptr, nullptr));
+  }
+
+  stream_.reset(new CallSessionFileRotatingStream(dir_path_));
+  ASSERT_TRUE(stream_->Open());
+  scoped_ptr<uint8_t[]> expected_buffer(new uint8_t[buffer_size]);
+  int expected_vals[] = {0, 1, 2, 6, 7};
+  for (size_t i = 0; i < arraysize(expected_vals); ++i) {
+    memset(expected_buffer.get(), expected_vals[i], buffer_size);
+    EXPECT_EQ(SR_SUCCESS,
+              stream_->ReadAll(buffer.get(), buffer_size, nullptr, nullptr));
+    EXPECT_EQ(0, memcmp(buffer.get(), expected_buffer.get(), buffer_size));
+  }
+  EXPECT_EQ(SR_EOS, stream_->ReadAll(buffer.get(), 1, nullptr, nullptr));
+}
+
+// Tests that writing and reading to a stream where only the first file is
+// written to behaves correctly.
+TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadFirstHalf) {
+  Init("CallSessionFileRotatingStreamTestWriteAndReadFirstHalf",
+       6 * 1024 * 1024);
+  ASSERT_TRUE(stream_->Open());
+  const size_t buffer_size = 1024 * 1024;
+  scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  for (int i = 0; i < 2; i++) {
+    memset(buffer.get(), i, buffer_size);
+    EXPECT_EQ(SR_SUCCESS,
+              stream_->WriteAll(buffer.get(), buffer_size, nullptr, nullptr));
+  }
+
+  stream_.reset(new CallSessionFileRotatingStream(dir_path_));
+  ASSERT_TRUE(stream_->Open());
+  scoped_ptr<uint8_t[]> expected_buffer(new uint8_t[buffer_size]);
+  int expected_vals[] = {0, 1};
+  for (size_t i = 0; i < arraysize(expected_vals); ++i) {
+    memset(expected_buffer.get(), expected_vals[i], buffer_size);
+    EXPECT_EQ(SR_SUCCESS,
+              stream_->ReadAll(buffer.get(), buffer_size, nullptr, nullptr));
+    EXPECT_EQ(0, memcmp(buffer.get(), expected_buffer.get(), buffer_size));
+  }
+  EXPECT_EQ(SR_EOS, stream_->ReadAll(buffer.get(), 1, nullptr, nullptr));
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/fileutils.cc b/media/webrtc/trunk/webrtc/base/fileutils.cc
index 6f385d72b7..cb23153de7 100644
--- a/media/webrtc/trunk/webrtc/base/fileutils.cc
+++ b/media/webrtc/trunk/webrtc/base/fileutils.cc
@@ -10,6 +10,7 @@
 
 #include <assert.h>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/stringutils.h"
@@ -273,8 +274,8 @@ bool CreateUniqueFile(Pathname& path, bool create_empty) {
     }
     version += 1;
     char version_base[MAX_PATH];
-    sprintfn(version_base, ARRAY_SIZE(version_base), "%s-%u",
-             basename.c_str(), version);
+    sprintfn(version_base, arraysize(version_base), "%s-%u", basename.c_str(),
+             version);
     path.SetBasename(version_base);
   }
   return true;
diff --git a/media/webrtc/trunk/webrtc/base/fileutils.h b/media/webrtc/trunk/webrtc/base/fileutils.h
index 1533fb5c11..bf02571d93 100644
--- a/media/webrtc/trunk/webrtc/base/fileutils.h
+++ b/media/webrtc/trunk/webrtc/base/fileutils.h
@@ -234,7 +234,7 @@ class FilesystemInterface {
   // Delete the contents of the folder returned by GetAppTempFolder
   bool CleanAppTempFolder();
 
-  virtual bool GetDiskFreeSpace(const Pathname& path, int64 *freebytes) = 0;
+  virtual bool GetDiskFreeSpace(const Pathname& path, int64_t* freebytes) = 0;
 
   // Returns the absolute path of the current directory.
   virtual Pathname GetCurrentDirectory() = 0;
@@ -379,7 +379,7 @@ class Filesystem {
     return EnsureDefaultFilesystem()->CleanAppTempFolder();
   }
 
-  static bool GetDiskFreeSpace(const Pathname& path, int64 *freebytes) {
+  static bool GetDiskFreeSpace(const Pathname& path, int64_t* freebytes) {
     return EnsureDefaultFilesystem()->GetDiskFreeSpace(path, freebytes);
   }
 
@@ -407,7 +407,7 @@ class Filesystem {
   static FilesystemInterface* default_filesystem_;
 
   static FilesystemInterface *EnsureDefaultFilesystem();
-  DISALLOW_IMPLICIT_CONSTRUCTORS(Filesystem);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Filesystem);
 };
 
 class FilesystemScope{
@@ -420,7 +420,7 @@ class FilesystemScope{
   }
  private:
   FilesystemInterface* old_fs_;
-  DISALLOW_IMPLICIT_CONSTRUCTORS(FilesystemScope);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FilesystemScope);
 };
 
 // Generates a unique filename based on the input path.  If no path component
diff --git a/media/webrtc/trunk/webrtc/base/fileutils_mock.h b/media/webrtc/trunk/webrtc/base/fileutils_mock.h
index e9d20a75f4..428d444e12 100644
--- a/media/webrtc/trunk/webrtc/base/fileutils_mock.h
+++ b/media/webrtc/trunk/webrtc/base/fileutils_mock.h
@@ -237,7 +237,7 @@ class FakeFileSystem : public FilesystemInterface {
       EXPECT_TRUE(false) << "Unsupported operation";
       return false;
     }
-    bool GetDiskFreeSpace(const Pathname &path, int64 *freebytes) {
+    bool GetDiskFreeSpace(const Pathname& path, int64_t* freebytes) {
       EXPECT_TRUE(false) << "Unsupported operation";
       return false;
     }
diff --git a/media/webrtc/trunk/webrtc/base/fileutils_unittest.cc b/media/webrtc/trunk/webrtc/base/fileutils_unittest.cc
index 9076bc7870..6e98e14509 100644
--- a/media/webrtc/trunk/webrtc/base/fileutils_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/fileutils_unittest.cc
@@ -90,29 +90,29 @@ TEST(FilesystemTest, TestGetDiskFreeSpace) {
   Pathname path;
   ASSERT_TRUE(Filesystem::GetAppDataFolder(&path, true));
 
-  int64 free1 = 0;
+  int64_t free1 = 0;
   EXPECT_TRUE(Filesystem::IsFolder(path));
   EXPECT_FALSE(Filesystem::IsFile(path));
   EXPECT_TRUE(Filesystem::GetDiskFreeSpace(path, &free1));
   EXPECT_GT(free1, 0);
 
-  int64 free2 = 0;
+  int64_t free2 = 0;
   path.AppendFolder("this_folder_doesnt_exist");
   EXPECT_FALSE(Filesystem::IsFolder(path));
   EXPECT_TRUE(Filesystem::IsAbsent(path));
   EXPECT_TRUE(Filesystem::GetDiskFreeSpace(path, &free2));
   // These should be the same disk, and disk free space should not have changed
   // by more than 1% between the two calls.
-  EXPECT_LT(static_cast<int64>(free1 * .9), free2);
-  EXPECT_LT(free2, static_cast<int64>(free1 * 1.1));
+  EXPECT_LT(static_cast<int64_t>(free1 * .9), free2);
+  EXPECT_LT(free2, static_cast<int64_t>(free1 * 1.1));
 
-  int64 free3 = 0;
+  int64_t free3 = 0;
   path.clear();
   EXPECT_TRUE(path.empty());
   EXPECT_TRUE(Filesystem::GetDiskFreeSpace(path, &free3));
   // Current working directory may not be where exe is.
-  // EXPECT_LT(static_cast<int64>(free1 * .9), free3);
-  // EXPECT_LT(free3, static_cast<int64>(free1 * 1.1));
+  // EXPECT_LT(static_cast<int64_t>(free1 * .9), free3);
+  // EXPECT_LT(free3, static_cast<int64_t>(free1 * 1.1));
   EXPECT_GT(free3, 0);
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/firewallsocketserver.cc b/media/webrtc/trunk/webrtc/base/firewallsocketserver.cc
index c35a687fff..6339017e08 100644
--- a/media/webrtc/trunk/webrtc/base/firewallsocketserver.cc
+++ b/media/webrtc/trunk/webrtc/base/firewallsocketserver.cc
@@ -126,13 +126,13 @@ FirewallSocketServer::~FirewallSocketServer() {
 void FirewallSocketServer::AddRule(bool allow, FirewallProtocol p,
                                    FirewallDirection d,
                                    const SocketAddress& addr) {
-  SocketAddress src, dst;
-  if (d == FD_IN) {
-    dst = addr;
-  } else {
-    src = addr;
+  SocketAddress any;
+  if (d == FD_IN || d == FD_ANY) {
+    AddRule(allow, p, any, addr);
+  }
+  if (d == FD_OUT || d == FD_ANY) {
+    AddRule(allow, p, addr, any);
   }
-  AddRule(allow, p, src, dst);
 }
 
 
diff --git a/media/webrtc/trunk/webrtc/base/flags.cc b/media/webrtc/trunk/webrtc/base/flags.cc
index a5e1c45e5d..0c0f4491c8 100644
--- a/media/webrtc/trunk/webrtc/base/flags.cc
+++ b/media/webrtc/trunk/webrtc/base/flags.cc
@@ -163,7 +163,7 @@ void FlagList::SplitArgument(const char* arg,
     if (*arg == '=') {
       // make a copy so we can NUL-terminate flag name
       int n = static_cast<int>(arg - *name);
-      CHECK_LT(n, buffer_size);
+      RTC_CHECK_LT(n, buffer_size);
       memcpy(buffer, *name, n * sizeof(char));
       buffer[n] = '\0';
       *name = buffer;
@@ -257,7 +257,8 @@ int FlagList::SetFlagsFromCommandLine(int* argc, const char** argv,
 
 void FlagList::Register(Flag* flag) {
   assert(flag != NULL && strlen(flag->name()) > 0);
-  CHECK(!Lookup(flag->name())) << "flag " << flag->name() << " declared twice";
+  RTC_CHECK(!Lookup(flag->name())) << "flag " << flag->name()
+                                   << " declared twice";
   flag->next_ = list_;
   list_ = flag;
 }
diff --git a/media/webrtc/trunk/webrtc/base/flags.h b/media/webrtc/trunk/webrtc/base/flags.h
index 5cff1cc365..4ce857b74a 100644
--- a/media/webrtc/trunk/webrtc/base/flags.h
+++ b/media/webrtc/trunk/webrtc/base/flags.h
@@ -90,17 +90,17 @@ class Flag {
     assert(type_ == BOOL);
     return &variable_->b;
   }
-  
+
   int* int_variable() const {
     assert(type_ == INT);
     return &variable_->i;
   }
-  
+
   double* float_variable() const {
     assert(type_ == FLOAT);
     return &variable_->f;
   }
-  
+
   const char** string_variable() const {
     assert(type_ == STRING);
     return &variable_->s;
@@ -111,17 +111,17 @@ class Flag {
     assert(type_ == BOOL);
     return default_.b;
   }
-  
+
   int int_default() const {
     assert(type_ == INT);
     return default_.i;
   }
-  
+
   double float_default() const {
     assert(type_ == FLOAT);
     return default_.f;
   }
-  
+
   const char* string_default() const {
     assert(type_ == STRING);
     return default_.s;
@@ -261,9 +261,9 @@ class WindowsCommandLineArguments {
   char **argv_;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(WindowsCommandLineArguments);
+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowsCommandLineArguments);
 };
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 }  // namespace rtc
 
diff --git a/media/webrtc/trunk/webrtc/base/format_macros.h b/media/webrtc/trunk/webrtc/base/format_macros.h
index 5d7dcc36b9..90f86a686c 100644
--- a/media/webrtc/trunk/webrtc/base/format_macros.h
+++ b/media/webrtc/trunk/webrtc/base/format_macros.h
@@ -73,6 +73,8 @@
 
 #else  // WEBRTC_WIN
 
+#include <inttypes.h>
+
 #if !defined(PRId64)
 #define PRId64 "I64d"
 #endif
diff --git a/media/webrtc/trunk/webrtc/base/gunit.h b/media/webrtc/trunk/webrtc/base/gunit.h
index 7431fcf308..1a6c36374e 100644
--- a/media/webrtc/trunk/webrtc/base/gunit.h
+++ b/media/webrtc/trunk/webrtc/base/gunit.h
@@ -13,30 +13,29 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
-#if defined(WEBRTC_ANDROID) || defined(GTEST_RELATIVE_PATH)
+#if defined(GTEST_RELATIVE_PATH)
 #include "testing/gtest/include/gtest/gtest.h"
 #else
 #include "testing/base/public/gunit.h"
 #endif
 
 // Wait until "ex" is true, or "timeout" expires.
-#define WAIT(ex, timeout) \
-  for (uint32 start = rtc::Time(); \
-      !(ex) && rtc::Time() < start + timeout;) \
+#define WAIT(ex, timeout)                                                     \
+  for (uint32_t start = rtc::Time(); !(ex) && rtc::Time() < start + timeout;) \
     rtc::Thread::Current()->ProcessMessages(1);
 
 // This returns the result of the test in res, so that we don't re-evaluate
 // the expression in the XXXX_WAIT macros below, since that causes problems
 // when the expression is only true the first time you check it.
-#define WAIT_(ex, timeout, res) \
-  do { \
-    uint32 start = rtc::Time(); \
-    res = (ex); \
+#define WAIT_(ex, timeout, res)                     \
+  do {                                              \
+    uint32_t start = rtc::Time();                   \
+    res = (ex);                                     \
     while (!res && rtc::Time() < start + timeout) { \
-      rtc::Thread::Current()->ProcessMessages(1); \
-      res = (ex); \
-    } \
-  } while (0);
+      rtc::Thread::Current()->ProcessMessages(1);   \
+      res = (ex);                                   \
+    }                                               \
+  } while (0)
 
 // The typical EXPECT_XXXX and ASSERT_XXXXs, but done until true or a timeout.
 #define EXPECT_TRUE_WAIT(ex, timeout) \
@@ -44,28 +43,28 @@
     bool res; \
     WAIT_(ex, timeout, res); \
     if (!res) EXPECT_TRUE(ex); \
-  } while (0);
+  } while (0)
 
 #define EXPECT_EQ_WAIT(v1, v2, timeout) \
   do { \
     bool res; \
     WAIT_(v1 == v2, timeout, res); \
     if (!res) EXPECT_EQ(v1, v2); \
-  } while (0);
+  } while (0)
 
 #define ASSERT_TRUE_WAIT(ex, timeout) \
   do { \
     bool res; \
     WAIT_(ex, timeout, res); \
     if (!res) ASSERT_TRUE(ex); \
-  } while (0);
+  } while (0)
 
 #define ASSERT_EQ_WAIT(v1, v2, timeout) \
   do { \
     bool res; \
     WAIT_(v1 == v2, timeout, res); \
     if (!res) ASSERT_EQ(v1, v2); \
-  } while (0);
+  } while (0)
 
 // Version with a "soft" timeout and a margin. This logs if the timeout is
 // exceeded, but it only fails if the expression still isn't true after the
@@ -83,6 +82,6 @@
     if (!res) { \
       EXPECT_TRUE(ex); \
     } \
-  } while (0);
+  } while (0)
 
 #endif  // WEBRTC_BASE_GUNIT_H_
diff --git a/media/webrtc/trunk/webrtc/base/helpers.cc b/media/webrtc/trunk/webrtc/base/helpers.cc
index bd7ff96ac4..1ad5d0e12b 100644
--- a/media/webrtc/trunk/webrtc/base/helpers.cc
+++ b/media/webrtc/trunk/webrtc/base/helpers.cc
@@ -16,14 +16,12 @@
 #include "webrtc/base/sslconfig.h"
 #if defined(SSL_USE_OPENSSL)
 #include <openssl/rand.h>
-#elif defined(SSL_USE_NSS_RNG)
-#include "pk11func.h"
 #else
 #if defined(WEBRTC_WIN)
 #define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #include <ntsecapi.h>
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 #endif  // else
 #endif  // FEATURE_ENABLED_SSL
 
@@ -141,7 +139,7 @@ class SecureRandomGenerator : public RandomGenerator {
 
 #error No SSL implementation has been selected!
 
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 #endif
 
 // A test random generator, for predictable output.
@@ -154,7 +152,7 @@ class TestRandomGenerator : public RandomGenerator {
   bool Init(const void* seed, size_t len) override { return true; }
   bool Generate(void* buf, size_t len) override {
     for (size_t i = 0; i < len; ++i) {
-      static_cast<uint8*>(buf)[i] = static_cast<uint8>(GetRandom());
+      static_cast<uint8_t*>(buf)[i] = static_cast<uint8_t>(GetRandom());
     }
     return true;
   }
@@ -166,22 +164,26 @@ class TestRandomGenerator : public RandomGenerator {
   int seed_;
 };
 
-// TODO: Use Base64::Base64Table instead.
-static const char BASE64[64] = {
-  'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
-  'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
-  'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
-  'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
-  '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'
-};
-
 namespace {
 
+// TODO: Use Base64::Base64Table instead.
+static const char kBase64[64] = {
+    'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
+    'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
+    'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
+    'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
+    '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'};
+
+static const char kHex[16] = {'0', '1', '2', '3', '4', '5', '6', '7',
+                              '8', '9', 'a', 'b', 'c', 'd', 'e', 'f'};
+
+static const char kUuidDigit17[4] = {'8', '9', 'a', 'b'};
+
 // This round about way of creating a global RNG is to safe-guard against
 // indeterminant static initialization order.
 scoped_ptr<RandomGenerator>& GetGlobalRng() {
-  LIBJINGLE_DEFINE_STATIC_LOCAL(scoped_ptr<RandomGenerator>, global_rng,
-                                (new SecureRandomGenerator()));
+  RTC_DEFINE_STATIC_LOCAL(scoped_ptr<RandomGenerator>, global_rng,
+                          (new SecureRandomGenerator()));
   return global_rng;
 }
 
@@ -221,7 +223,7 @@ bool CreateRandomString(size_t len,
                         const char* table, int table_size,
                         std::string* str) {
   str->clear();
-  scoped_ptr<uint8[]> bytes(new uint8[len]);
+  scoped_ptr<uint8_t[]> bytes(new uint8_t[len]);
   if (!Rng().Generate(bytes.get(), len)) {
     LOG(LS_ERROR) << "Failed to generate random string!";
     return false;
@@ -234,7 +236,7 @@ bool CreateRandomString(size_t len,
 }
 
 bool CreateRandomString(size_t len, std::string* str) {
-  return CreateRandomString(len, BASE64, 64, str);
+  return CreateRandomString(len, kBase64, 64, str);
 }
 
 bool CreateRandomString(size_t len, const std::string& table,
@@ -243,20 +245,55 @@ bool CreateRandomString(size_t len, const std::string& table,
                             static_cast<int>(table.size()), str);
 }
 
-uint32 CreateRandomId() {
-  uint32 id;
+// Version 4 UUID is of the form:
+// xxxxxxxx-xxxx-4xxx-yxxx-xxxxxxxxxxxx
+// Where 'x' is a hex digit, and 'y' is 8, 9, a or b.
+std::string CreateRandomUuid() {
+  std::string str;
+  scoped_ptr<uint8_t[]> bytes(new uint8_t[31]);
+  if (!Rng().Generate(bytes.get(), 31)) {
+    LOG(LS_ERROR) << "Failed to generate random string!";
+    return str;
+  }
+  str.reserve(36);
+  for (size_t i = 0; i < 8; ++i) {
+    str.push_back(kHex[bytes[i] % 16]);
+  }
+  str.push_back('-');
+  for (size_t i = 8; i < 12; ++i) {
+    str.push_back(kHex[bytes[i] % 16]);
+  }
+  str.push_back('-');
+  str.push_back('4');
+  for (size_t i = 12; i < 15; ++i) {
+    str.push_back(kHex[bytes[i] % 16]);
+  }
+  str.push_back('-');
+  str.push_back(kUuidDigit17[bytes[15] % 4]);
+  for (size_t i = 16; i < 19; ++i) {
+    str.push_back(kHex[bytes[i] % 16]);
+  }
+  str.push_back('-');
+  for (size_t i = 19; i < 31; ++i) {
+    str.push_back(kHex[bytes[i] % 16]);
+  }
+  return str;
+}
+
+uint32_t CreateRandomId() {
+  uint32_t id;
   if (!Rng().Generate(&id, sizeof(id))) {
     LOG(LS_ERROR) << "Failed to generate random id!";
   }
   return id;
 }
 
-uint64 CreateRandomId64() {
-  return static_cast<uint64>(CreateRandomId()) << 32 | CreateRandomId();
+uint64_t CreateRandomId64() {
+  return static_cast<uint64_t>(CreateRandomId()) << 32 | CreateRandomId();
 }
 
-uint32 CreateRandomNonZeroId() {
-  uint32 id;
+uint32_t CreateRandomNonZeroId() {
+  uint32_t id;
   do {
     id = CreateRandomId();
   } while (id == 0);
@@ -264,8 +301,8 @@ uint32 CreateRandomNonZeroId() {
 }
 
 double CreateRandomDouble() {
-  return CreateRandomId() / (std::numeric_limits<uint32>::max() +
-      std::numeric_limits<double>::epsilon());
+  return CreateRandomId() / (std::numeric_limits<uint32_t>::max() +
+                             std::numeric_limits<double>::epsilon());
 }
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/helpers.h b/media/webrtc/trunk/webrtc/base/helpers.h
index e46d12a330..0e7937362a 100644
--- a/media/webrtc/trunk/webrtc/base/helpers.h
+++ b/media/webrtc/trunk/webrtc/base/helpers.h
@@ -39,14 +39,17 @@ bool CreateRandomString(size_t length, std::string* str);
 bool CreateRandomString(size_t length, const std::string& table,
                         std::string* str);
 
+// Generates a (cryptographically) random UUID version 4 string.
+std::string CreateRandomUuid();
+
 // Generates a random id.
-uint32 CreateRandomId();
+uint32_t CreateRandomId();
 
 // Generates a 64 bit random id.
-uint64 CreateRandomId64();
+uint64_t CreateRandomId64();
 
 // Generates a random id > 0.
-uint32 CreateRandomNonZeroId();
+uint32_t CreateRandomNonZeroId();
 
 // Generates a random double between 0.0 (inclusive) and 1.0 (exclusive).
 double CreateRandomDouble();
diff --git a/media/webrtc/trunk/webrtc/base/helpers_unittest.cc b/media/webrtc/trunk/webrtc/base/helpers_unittest.cc
index 6ea0167e98..83cc685919 100644
--- a/media/webrtc/trunk/webrtc/base/helpers_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/helpers_unittest.cc
@@ -43,16 +43,23 @@ TEST_F(RandomTest, TestCreateRandomString) {
   EXPECT_EQ(256U, random2.size());
 }
 
+TEST_F(RandomTest, TestCreateRandomUuid) {
+  std::string random = CreateRandomUuid();
+  EXPECT_EQ(36U, random.size());
+}
+
 TEST_F(RandomTest, TestCreateRandomForTest) {
   // Make sure we get the output we expect.
   SetRandomTestMode(true);
   EXPECT_EQ(2154761789U, CreateRandomId());
   EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
+  EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
 
   // Reset and make sure we get the same output.
   SetRandomTestMode(true);
   EXPECT_EQ(2154761789U, CreateRandomId());
   EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
+  EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
 
   // Test different character sets.
   SetRandomTestMode(true);
diff --git a/media/webrtc/trunk/webrtc/base/httpbase_unittest.cc b/media/webrtc/trunk/webrtc/base/httpbase_unittest.cc
index fd5f867331..8d8e09715f 100644
--- a/media/webrtc/trunk/webrtc/base/httpbase_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/httpbase_unittest.cc
@@ -145,7 +145,7 @@ void HttpBaseTest::VerifyHeaderComplete(size_t event_count, bool empty_doc) {
 
   std::string header;
   EXPECT_EQ(HVER_1_1, data.version);
-  EXPECT_EQ(static_cast<uint32>(HC_OK), data.scode);
+  EXPECT_EQ(static_cast<uint32_t>(HC_OK), data.scode);
   EXPECT_TRUE(data.hasHeader(HH_PROXY_AUTHORIZATION, &header));
   EXPECT_EQ("42", header);
   EXPECT_TRUE(data.hasHeader(HH_CONNECTION, &header));
@@ -215,7 +215,7 @@ void HttpBaseTest::VerifyDocumentStreamOpenEvent() {
 
   // HTTP headers haven't arrived yet
   EXPECT_EQ(0U, events.size());
-  EXPECT_EQ(static_cast<uint32>(HC_INTERNAL_SERVER_ERROR), data.scode);
+  EXPECT_EQ(static_cast<uint32_t>(HC_INTERNAL_SERVER_ERROR), data.scode);
   LOG_F(LS_VERBOSE) << "Exit";
 }
 
@@ -404,7 +404,7 @@ TEST_F(HttpBaseTest, SupportsReceiveViaStreamPull) {
 TEST_F(HttpBaseTest, DISABLED_AllowsCloseStreamBeforeDocumentIsComplete) {
 
   // TODO: Remove extra logging once test failure is understood
-  int old_sev = rtc::LogMessage::GetLogToDebug();
+  LoggingSeverity old_sev = rtc::LogMessage::GetLogToDebug();
   rtc::LogMessage::LogToDebug(LS_VERBOSE);
 
 
diff --git a/media/webrtc/trunk/webrtc/base/httpclient.h b/media/webrtc/trunk/webrtc/base/httpclient.h
index 2983d8c005..e7d2c5ce7d 100644
--- a/media/webrtc/trunk/webrtc/base/httpclient.h
+++ b/media/webrtc/trunk/webrtc/base/httpclient.h
@@ -68,7 +68,7 @@ public:
 
   void set_agent(const std::string& agent) { agent_ = agent; }
   const std::string& agent() const { return agent_; }
-  
+
   void set_proxy(const ProxyInfo& proxy) { proxy_ = proxy; }
   const ProxyInfo& proxy() const { return proxy_; }
 
@@ -83,11 +83,6 @@ public:
   enum RedirectAction { REDIRECT_DEFAULT, REDIRECT_ALWAYS, REDIRECT_NEVER };
   void set_redirect_action(RedirectAction action) { redirect_action_ = action; }
   RedirectAction redirect_action() const { return redirect_action_; }
-  // Deprecated
-  void set_fail_redirect(bool fail_redirect) {
-    redirect_action_ = REDIRECT_NEVER;
-  }
-  bool fail_redirect() const { return (REDIRECT_NEVER == redirect_action_); }
 
   enum UriForm { URI_DEFAULT, URI_ABSOLUTE, URI_RELATIVE };
   void set_uri_form(UriForm form) { uri_form_ = form; }
@@ -99,22 +94,20 @@ public:
   // reset clears the server, request, and response structures.  It will also
   // abort an active request.
   void reset();
-  
+
   void set_server(const SocketAddress& address);
   const SocketAddress& server() const { return server_; }
 
   // Note: in order for HttpClient to retry a POST in response to
   // an authentication challenge, a redirect response, or socket disconnection,
   // the request document must support 'replaying' by calling Rewind() on it.
-  // In the case where just a subset of a stream should be used as the request
-  // document, the stream may be wrapped with the StreamSegment adapter.
   HttpTransaction* transaction() { return transaction_; }
   const HttpTransaction* transaction() const { return transaction_; }
   HttpRequestData& request() { return transaction_->request; }
   const HttpRequestData& request() const { return transaction_->request; }
   HttpResponseData& response() { return transaction_->response; }
   const HttpResponseData& response() const { return transaction_->response; }
-  
+
   // convenience methods
   void prepare_get(const std::string& url);
   void prepare_post(const std::string& url, const std::string& content_type,
@@ -125,7 +118,7 @@ public:
 
   // After you finish setting up your request, call start.
   void start();
-  
+
   // Signalled when the header has finished downloading, before the document
   // content is processed.  You may change the response document in response
   // to this signal.  The second parameter indicates whether this is an
diff --git a/media/webrtc/trunk/webrtc/base/httpcommon-inl.h b/media/webrtc/trunk/webrtc/base/httpcommon-inl.h
index 2f525ce792..188d9e6509 100644
--- a/media/webrtc/trunk/webrtc/base/httpcommon-inl.h
+++ b/media/webrtc/trunk/webrtc/base/httpcommon-inl.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_BASE_HTTPCOMMON_INL_H__
 #define WEBRTC_BASE_HTTPCOMMON_INL_H__
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/httpcommon.h"
 
@@ -52,7 +53,7 @@ void Url<CTYPE>::do_set_address(const CTYPE* val, size_t len) {
     host_.assign(val, colon - val);
     // Note: In every case, we're guaranteed that colon is followed by a null,
     // or non-numeric character.
-    port_ = static_cast<uint16>(::strtoul(colon + 1, NULL, 10));
+    port_ = static_cast<uint16_t>(::strtoul(colon + 1, NULL, 10));
     // TODO: Consider checking for invalid data following port number.
   } else {
     host_.assign(val, len);
@@ -80,7 +81,7 @@ void Url<CTYPE>::do_set_full_path(const CTYPE* val, size_t len) {
 template<class CTYPE>
 void Url<CTYPE>::do_get_url(string* val) const {
   CTYPE protocol[9];
-  asccpyn(protocol, ARRAY_SIZE(protocol), secure_ ? "https://" : "http://");
+  asccpyn(protocol, arraysize(protocol), secure_ ? "https://" : "http://");
   val->append(protocol);
   do_get_address(val);
   do_get_full_path(val);
@@ -91,8 +92,8 @@ void Url<CTYPE>::do_get_address(string* val) const {
   val->append(host_);
   if (port_ != HttpDefaultPort(secure_)) {
     CTYPE format[5], port[32];
-    asccpyn(format, ARRAY_SIZE(format), ":%hu");
-    sprintfn(port, ARRAY_SIZE(port), format, port_);
+    asccpyn(format, arraysize(format), ":%hu");
+    sprintfn(port, arraysize(port), format, port_);
     val->append(port);
   }
 }
diff --git a/media/webrtc/trunk/webrtc/base/httpcommon.cc b/media/webrtc/trunk/webrtc/base/httpcommon.cc
index cff954cca5..c90bea51cc 100644
--- a/media/webrtc/trunk/webrtc/base/httpcommon.cc
+++ b/media/webrtc/trunk/webrtc/base/httpcommon.cc
@@ -21,6 +21,7 @@
 
 #include <algorithm>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/cryptstring.h"
@@ -149,12 +150,12 @@ bool FromString(HttpHeader& header, const std::string& str) {
   return Enum<HttpHeader>::Parse(header, str);
 }
 
-bool HttpCodeHasBody(uint32 code) {
+bool HttpCodeHasBody(uint32_t code) {
   return !HttpCodeIsInformational(code)
          && (code != HC_NO_CONTENT) && (code != HC_NOT_MODIFIED);
 }
 
-bool HttpCodeIsCacheable(uint32 code) {
+bool HttpCodeIsCacheable(uint32_t code) {
   switch (code) {
   case HC_OK:
   case HC_NON_AUTHORITATIVE:
@@ -377,7 +378,7 @@ bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
     gmt = non_gmt + ((zone[0] == '+') ? offset : -offset);
   } else {
     size_t zindex;
-    if (!find_string(zindex, zone, kTimeZones, ARRAY_SIZE(kTimeZones))) {
+    if (!find_string(zindex, zone, kTimeZones, arraysize(kTimeZones))) {
       return false;
     }
     gmt = non_gmt + kTimeZoneOffsets[zindex] * 60 * 60;
@@ -387,6 +388,10 @@ bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
   tm *tm_for_timezone = localtime(&gmt);
   *seconds = gmt + tm_for_timezone->tm_gmtoff;
 #else
+#if _MSC_VER >= 1900
+  long timezone = 0;
+  _get_timezone(&timezone);
+#endif
   *seconds = gmt - timezone;
 #endif
   return true;
@@ -595,32 +600,29 @@ HttpResponseData::copy(const HttpResponseData& src) {
   HttpData::copy(src);
 }
 
-void
-HttpResponseData::set_success(uint32 scode) {
+void HttpResponseData::set_success(uint32_t scode) {
   this->scode = scode;
   message.clear();
   setHeader(HH_CONTENT_LENGTH, "0", false);
 }
 
-void
-HttpResponseData::set_success(const std::string& content_type,
-                              StreamInterface* document,
-                              uint32 scode) {
+void HttpResponseData::set_success(const std::string& content_type,
+                                   StreamInterface* document,
+                                   uint32_t scode) {
   this->scode = scode;
   message.erase(message.begin(), message.end());
   setContent(content_type, document);
 }
 
-void
-HttpResponseData::set_redirect(const std::string& location, uint32 scode) {
+void HttpResponseData::set_redirect(const std::string& location,
+                                    uint32_t scode) {
   this->scode = scode;
   message.clear();
   setHeader(HH_LOCATION, location);
   setHeader(HH_CONTENT_LENGTH, "0", false);
 }
 
-void
-HttpResponseData::set_error(uint32 scode) {
+void HttpResponseData::set_error(uint32_t scode) {
   this->scode = scode;
   message.clear();
   setHeader(HH_CONTENT_LENGTH, "0", false);
@@ -907,7 +909,7 @@ HttpAuthResult HttpAuthenticate(
     bool specify_credentials = !username.empty();
     size_t steps = 0;
 
-    //uint32 now = Time();
+    // uint32_t now = Time();
 
     NegotiateAuthContext * neg = static_cast<NegotiateAuthContext *>(context);
     if (neg) {
diff --git a/media/webrtc/trunk/webrtc/base/httpcommon.h b/media/webrtc/trunk/webrtc/base/httpcommon.h
index 7b20facaa7..addc1bc30d 100644
--- a/media/webrtc/trunk/webrtc/base/httpcommon.h
+++ b/media/webrtc/trunk/webrtc/base/httpcommon.h
@@ -112,8 +112,8 @@ enum HttpHeader {
   HH_LAST = HH_WWW_AUTHENTICATE
 };
 
-const uint16 HTTP_DEFAULT_PORT = 80;
-const uint16 HTTP_SECURE_PORT = 443;
+const uint16_t HTTP_DEFAULT_PORT = 80;
+const uint16_t HTTP_SECURE_PORT = 443;
 
 //////////////////////////////////////////////////////////////////////
 // Utility Functions
@@ -132,14 +132,24 @@ bool FromString(HttpVerb& verb, const std::string& str);
 const char* ToString(HttpHeader header);
 bool FromString(HttpHeader& header, const std::string& str);
 
-inline bool HttpCodeIsInformational(uint32 code) { return ((code / 100) == 1); }
-inline bool HttpCodeIsSuccessful(uint32 code)    { return ((code / 100) == 2); }
-inline bool HttpCodeIsRedirection(uint32 code)   { return ((code / 100) == 3); }
-inline bool HttpCodeIsClientError(uint32 code)   { return ((code / 100) == 4); }
-inline bool HttpCodeIsServerError(uint32 code)   { return ((code / 100) == 5); }
+inline bool HttpCodeIsInformational(uint32_t code) {
+  return ((code / 100) == 1);
+}
+inline bool HttpCodeIsSuccessful(uint32_t code) {
+  return ((code / 100) == 2);
+}
+inline bool HttpCodeIsRedirection(uint32_t code) {
+  return ((code / 100) == 3);
+}
+inline bool HttpCodeIsClientError(uint32_t code) {
+  return ((code / 100) == 4);
+}
+inline bool HttpCodeIsServerError(uint32_t code) {
+  return ((code / 100) == 5);
+}
 
-bool HttpCodeHasBody(uint32 code);
-bool HttpCodeIsCacheable(uint32 code);
+bool HttpCodeHasBody(uint32_t code);
+bool HttpCodeIsCacheable(uint32_t code);
 bool HttpHeaderIsEndToEnd(HttpHeader header);
 bool HttpHeaderIsCollapsible(HttpHeader header);
 
@@ -163,7 +173,7 @@ bool HttpHasNthAttribute(HttpAttributeList& attributes,
 // Convert RFC1123 date (DoW, DD Mon YYYY HH:MM:SS TZ) to unix timestamp
 bool HttpDateToSeconds(const std::string& date, time_t* seconds);
 
-inline uint16 HttpDefaultPort(bool secure) {
+inline uint16_t HttpDefaultPort(bool secure) {
   return secure ? HTTP_SECURE_PORT : HTTP_DEFAULT_PORT;
 }
 
@@ -196,9 +206,10 @@ public:
   static int Decode(const string& source, string& destination);
 
   Url(const string& url) { do_set_url(url.c_str(), url.size()); }
-  Url(const string& path, const string& host, uint16 port = HTTP_DEFAULT_PORT)
-  : host_(host), port_(port), secure_(HTTP_SECURE_PORT == port)
-  { set_full_path(path); }
+  Url(const string& path, const string& host, uint16_t port = HTTP_DEFAULT_PORT)
+      : host_(host), port_(port), secure_(HTTP_SECURE_PORT == port) {
+    set_full_path(path);
+  }
 
   bool valid() const { return !host_.empty(); }
   void clear() {
@@ -233,8 +244,8 @@ public:
   void set_host(const string& val) { host_ = val; }
   const string& host() const { return host_; }
 
-  void set_port(uint16 val) { port_ = val; }
-  uint16 port() const { return port_; }
+  void set_port(uint16_t val) { port_ = val; }
+  uint16_t port() const { return port_; }
 
   void set_secure(bool val) { secure_ = val; }
   bool secure() const { return secure_; }
@@ -267,7 +278,7 @@ private:
   void do_get_full_path(string* val) const;
 
   string host_, path_, query_;
-  uint16 port_;
+  uint16_t port_;
   bool secure_;
 };
 
@@ -393,7 +404,7 @@ struct HttpRequestData : public HttpData {
 };
 
 struct HttpResponseData : public HttpData {
-  uint32 scode;
+  uint32_t scode;
   std::string message;
 
   HttpResponseData() : scode(HC_INTERNAL_SERVER_ERROR) { }
@@ -401,12 +412,13 @@ struct HttpResponseData : public HttpData {
   void copy(const HttpResponseData& src);
 
   // Convenience methods
-  void set_success(uint32 scode = HC_OK);
-  void set_success(const std::string& content_type, StreamInterface* document,
-                   uint32 scode = HC_OK);
+  void set_success(uint32_t scode = HC_OK);
+  void set_success(const std::string& content_type,
+                   StreamInterface* document,
+                   uint32_t scode = HC_OK);
   void set_redirect(const std::string& location,
-                    uint32 scode = HC_MOVED_TEMPORARILY);
-  void set_error(uint32 scode);
+                    uint32_t scode = HC_MOVED_TEMPORARILY);
+  void set_error(uint32_t scode);
 
   size_t formatLeader(char* buffer, size_t size) const override;
   HttpError parseLeader(const char* line, size_t len) override;
diff --git a/media/webrtc/trunk/webrtc/base/httprequest.cc b/media/webrtc/trunk/webrtc/base/httprequest.cc
index 3199c8f94f..0139f01fcf 100644
--- a/media/webrtc/trunk/webrtc/base/httprequest.cc
+++ b/media/webrtc/trunk/webrtc/base/httprequest.cc
@@ -48,11 +48,13 @@ void HttpMonitor::OnHttpClientComplete(HttpClient * http, HttpErrorType error) {
 
 const int kDefaultHTTPTimeout = 30 * 1000; // 30 sec
 
-HttpRequest::HttpRequest(const std::string &user_agent)
-    : firewall_(0), port_(80), secure_(false),
-      timeout_(kDefaultHTTPTimeout), fail_redirect_(false),
-      client_(user_agent.c_str(), NULL), error_(HE_NONE) {
-}
+HttpRequest::HttpRequest(const std::string& user_agent)
+    : firewall_(0),
+      port_(80),
+      secure_(false),
+      timeout_(kDefaultHTTPTimeout),
+      client_(user_agent.c_str(), NULL),
+      error_(HE_NONE) {}
 
 HttpRequest::~HttpRequest() = default;
 
@@ -82,7 +84,7 @@ void HttpRequest::Send() {
   if (transparent_proxy) {
     client_.set_proxy(proxy_);
   }
-  client_.set_fail_redirect(fail_redirect_);
+  client_.set_redirect_action(HttpClient::REDIRECT_ALWAYS);
 
   SocketAddress server(host_, port_);
   client_.set_server(server);
diff --git a/media/webrtc/trunk/webrtc/base/ifaddrs_converter.cc b/media/webrtc/trunk/webrtc/base/ifaddrs_converter.cc
new file mode 100644
index 0000000000..7dd35552f6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/ifaddrs_converter.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/ifaddrs_converter.h"
+
+namespace rtc {
+
+IfAddrsConverter::IfAddrsConverter() {}
+
+IfAddrsConverter::~IfAddrsConverter() {}
+
+bool IfAddrsConverter::ConvertIfAddrsToIPAddress(
+    const struct ifaddrs* interface,
+    InterfaceAddress* ip,
+    IPAddress* mask) {
+  switch (interface->ifa_addr->sa_family) {
+    case AF_INET: {
+      *ip = IPAddress(
+          reinterpret_cast<sockaddr_in*>(interface->ifa_addr)->sin_addr);
+      *mask = IPAddress(
+          reinterpret_cast<sockaddr_in*>(interface->ifa_netmask)->sin_addr);
+      return true;
+    }
+    case AF_INET6: {
+      int ip_attributes = IPV6_ADDRESS_FLAG_NONE;
+      if (!ConvertNativeAttributesToIPAttributes(interface, &ip_attributes)) {
+        return false;
+      }
+      *ip = InterfaceAddress(
+          reinterpret_cast<sockaddr_in6*>(interface->ifa_addr)->sin6_addr,
+          ip_attributes);
+      *mask = IPAddress(
+          reinterpret_cast<sockaddr_in6*>(interface->ifa_netmask)->sin6_addr);
+      return true;
+    }
+    default: { return false; }
+  }
+}
+
+bool IfAddrsConverter::ConvertNativeAttributesToIPAttributes(
+    const struct ifaddrs* interface,
+    int* ip_attributes) {
+  *ip_attributes = IPV6_ADDRESS_FLAG_NONE;
+  return true;
+}
+
+#if !defined(WEBRTC_MAC)
+// For MAC and IOS, it's defined in macifaddrs_converter.cc
+IfAddrsConverter* CreateIfAddrsConverter() {
+  return new IfAddrsConverter();
+}
+#endif
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/ifaddrs_converter.h b/media/webrtc/trunk/webrtc/base/ifaddrs_converter.h
new file mode 100644
index 0000000000..0a1cdb9e41
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/ifaddrs_converter.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_IFADDRS_CONVERTER_H_
+#define WEBRTC_BASE_IFADDRS_CONVERTER_H_
+
+#if defined(WEBRTC_ANDROID)
+#include "webrtc/base/ifaddrs-android.h"
+#else
+#include <ifaddrs.h>
+#endif  // WEBRTC_ANDROID
+
+#include "webrtc/base/ipaddress.h"
+
+namespace rtc {
+
+// This class converts native interface addresses to our internal IPAddress
+// class. Subclasses should override ConvertNativeToIPAttributes to implement
+// the different ways of retrieving IPv6 attributes for various POSIX platforms.
+class IfAddrsConverter {
+ public:
+  IfAddrsConverter();
+  virtual ~IfAddrsConverter();
+  virtual bool ConvertIfAddrsToIPAddress(const struct ifaddrs* interface,
+                                         InterfaceAddress* ipaddress,
+                                         IPAddress* mask);
+
+ protected:
+  virtual bool ConvertNativeAttributesToIPAttributes(
+      const struct ifaddrs* interface,
+      int* ip_attributes);
+};
+
+IfAddrsConverter* CreateIfAddrsConverter();
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_IFADDRS_CONVERTER_H_
diff --git a/media/webrtc/trunk/webrtc/base/ipaddress.cc b/media/webrtc/trunk/webrtc/base/ipaddress.cc
index ae3b38da8b..c92f33c74d 100644
--- a/media/webrtc/trunk/webrtc/base/ipaddress.cc
+++ b/media/webrtc/trunk/webrtc/base/ipaddress.cc
@@ -27,8 +27,10 @@
 
 #include "webrtc/base/ipaddress.h"
 #include "webrtc/base/byteorder.h"
-#include "webrtc/base/nethelpers.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/nethelpers.h"
+#include "webrtc/base/stringutils.h"
 #include "webrtc/base/win32.h"
 
 namespace rtc {
@@ -41,12 +43,10 @@ static const in6_addr kTeredoPrefix = {{{0x20, 0x01, 0x00, 0x00}}};
 static const in6_addr kV4CompatibilityPrefix = {{{0}}};
 static const in6_addr k6BonePrefix = {{{0x3f, 0xfe, 0}}};
 
-bool IPAddress::strip_sensitive_ = false;
-
-static bool IsPrivateV4(uint32 ip);
+static bool IsPrivateV4(uint32_t ip);
 static in_addr ExtractMappedAddress(const in6_addr& addr);
 
-uint32 IPAddress::v4AddressAsHostOrderInteger() const {
+uint32_t IPAddress::v4AddressAsHostOrderInteger() const {
   if (family_ == AF_INET) {
     return NetworkToHost32(u_.ip4.s_addr);
   } else {
@@ -54,6 +54,10 @@ uint32 IPAddress::v4AddressAsHostOrderInteger() const {
   }
 }
 
+bool IPAddress::IsNil() const {
+  return IPIsUnspec(*this);
+}
+
 size_t IPAddress::Size() const {
   switch (family_) {
     case AF_INET:
@@ -140,9 +144,10 @@ std::string IPAddress::ToString() const {
 }
 
 std::string IPAddress::ToSensitiveString() const {
-  if (!strip_sensitive_)
-    return ToString();
-
+#if !defined(NDEBUG)
+  // Return non-stripped in debug.
+  return ToString();
+#else
   switch (family_) {
     case AF_INET: {
       std::string address = ToString();
@@ -154,12 +159,20 @@ std::string IPAddress::ToSensitiveString() const {
       return address;
     }
     case AF_INET6: {
-      // TODO(grunell): Return a string of format 1:2:3:x:x:x:x:x or such
-      // instead of zeroing out.
-      return TruncateIP(*this, 128 - 80).ToString();
+      std::string result;
+      result.resize(INET6_ADDRSTRLEN);
+      in6_addr addr = ipv6_address();
+      size_t len =
+          rtc::sprintfn(&(result[0]), result.size(), "%x:%x:%x:x:x:x:x:x",
+                        (addr.s6_addr[0] << 8) + addr.s6_addr[1],
+                        (addr.s6_addr[2] << 8) + addr.s6_addr[3],
+                        (addr.s6_addr[4] << 8) + addr.s6_addr[5]);
+      result.resize(len);
+      return result;
     }
   }
   return std::string();
+#endif
 }
 
 IPAddress IPAddress::Normalized() const {
@@ -182,10 +195,6 @@ IPAddress IPAddress::AsIPv6Address() const {
   return IPAddress(v6addr);
 }
 
-void IPAddress::set_strip_sensitive(bool enable) {
-  strip_sensitive_ = enable;
-}
-
 bool InterfaceAddress::operator==(const InterfaceAddress &other) const {
   return ipv6_flags_ == other.ipv6_flags() &&
     static_cast<const IPAddress&>(*this) == other;
@@ -211,7 +220,7 @@ std::ostream& operator<<(std::ostream& os, const InterfaceAddress& ip) {
   return os;
 }
 
-bool IsPrivateV4(uint32 ip_in_host_order) {
+bool IsPrivateV4(uint32_t ip_in_host_order) {
   return ((ip_in_host_order >> 24) == 127) ||
       ((ip_in_host_order >> 24) == 10) ||
       ((ip_in_host_order >> 20) == ((172 << 4) | 1)) ||
@@ -317,8 +326,8 @@ size_t HashIP(const IPAddress& ip) {
     }
     case AF_INET6: {
       in6_addr v6addr = ip.ipv6_address();
-      const uint32* v6_as_ints =
-          reinterpret_cast<const uint32*>(&v6addr.s6_addr);
+      const uint32_t* v6_as_ints =
+          reinterpret_cast<const uint32_t*>(&v6addr.s6_addr);
       return v6_as_ints[0] ^ v6_as_ints[1] ^ v6_as_ints[2] ^ v6_as_ints[3];
     }
   }
@@ -337,7 +346,7 @@ IPAddress TruncateIP(const IPAddress& ip, int length) {
       return IPAddress(INADDR_ANY);
     }
     int mask = (0xFFFFFFFF << (32 - length));
-    uint32 host_order_ip = NetworkToHost32(ip.ipv4_address().s_addr);
+    uint32_t host_order_ip = NetworkToHost32(ip.ipv4_address().s_addr);
     in_addr masked;
     masked.s_addr = HostToNetwork32(host_order_ip & mask);
     return IPAddress(masked);
@@ -352,12 +361,11 @@ IPAddress TruncateIP(const IPAddress& ip, int length) {
     int position = length / 32;
     int inner_length = 32 - (length - (position * 32));
     // Note: 64bit mask constant needed to allow possible 32-bit left shift.
-    uint32 inner_mask = 0xFFFFFFFFLL  << inner_length;
-    uint32* v6_as_ints =
-        reinterpret_cast<uint32*>(&v6addr.s6_addr);
+    uint32_t inner_mask = 0xFFFFFFFFLL << inner_length;
+    uint32_t* v6_as_ints = reinterpret_cast<uint32_t*>(&v6addr.s6_addr);
     for (int i = 0; i < 4; ++i) {
       if (i == position) {
-        uint32 host_order_inner = NetworkToHost32(v6_as_ints[i]);
+        uint32_t host_order_inner = NetworkToHost32(v6_as_ints[i]);
         v6_as_ints[i] = HostToNetwork32(host_order_inner & inner_mask);
       } else if (i > position) {
         v6_as_ints[i] = 0;
@@ -369,7 +377,7 @@ IPAddress TruncateIP(const IPAddress& ip, int length) {
 }
 
 int CountIPMaskBits(IPAddress mask) {
-  uint32 word_to_count = 0;
+  uint32_t word_to_count = 0;
   int bits = 0;
   switch (mask.family()) {
     case AF_INET: {
@@ -378,8 +386,8 @@ int CountIPMaskBits(IPAddress mask) {
     }
     case AF_INET6: {
       in6_addr v6addr = mask.ipv6_address();
-      const uint32* v6_as_ints =
-          reinterpret_cast<const uint32*>(&v6addr.s6_addr);
+      const uint32_t* v6_as_ints =
+          reinterpret_cast<const uint32_t*>(&v6addr.s6_addr);
       int i = 0;
       for (; i < 4; ++i) {
         if (v6_as_ints[i] != 0xFFFFFFFF) {
@@ -404,7 +412,7 @@ int CountIPMaskBits(IPAddress mask) {
   // http://graphics.stanford.edu/~seander/bithacks.html
   // Counts the trailing 0s in the word.
   unsigned int zeroes = 32;
-  word_to_count &= -static_cast<int32>(word_to_count);
+  word_to_count &= -static_cast<int32_t>(word_to_count);
   if (word_to_count) zeroes--;
   if (word_to_count & 0x0000FFFF) zeroes -= 16;
   if (word_to_count & 0x00FF00FF) zeroes -= 8;
@@ -494,4 +502,24 @@ int IPAddressPrecedence(const IPAddress& ip) {
   return 0;
 }
 
-}  // Namespace talk base
+IPAddress GetLoopbackIP(int family) {
+  if (family == AF_INET) {
+    return rtc::IPAddress(INADDR_LOOPBACK);
+  }
+  if (family == AF_INET6) {
+    return rtc::IPAddress(in6addr_loopback);
+  }
+  return rtc::IPAddress();
+}
+
+IPAddress GetAnyIP(int family) {
+  if (family == AF_INET) {
+    return rtc::IPAddress(INADDR_ANY);
+  }
+  if (family == AF_INET6) {
+    return rtc::IPAddress(in6addr_any);
+  }
+  return rtc::IPAddress();
+}
+
+}  // Namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/ipaddress.h b/media/webrtc/trunk/webrtc/base/ipaddress.h
index fb222eeba3..ef1e3d8170 100644
--- a/media/webrtc/trunk/webrtc/base/ipaddress.h
+++ b/media/webrtc/trunk/webrtc/base/ipaddress.h
@@ -62,7 +62,7 @@ class IPAddress {
     u_.ip6 = ip6;
   }
 
-  explicit IPAddress(uint32 ip_in_host_byte_order) : family_(AF_INET) {
+  explicit IPAddress(uint32_t ip_in_host_byte_order) : family_(AF_INET) {
     memset(&u_, 0, sizeof(u_));
     u_.ip4.s_addr = HostToNetwork32(ip_in_host_byte_order);
   }
@@ -107,9 +107,10 @@ class IPAddress {
   IPAddress AsIPv6Address() const;
 
   // For socketaddress' benefit. Returns the IP in host byte order.
-  uint32 v4AddressAsHostOrderInteger() const;
+  uint32_t v4AddressAsHostOrderInteger() const;
 
-  static void set_strip_sensitive(bool enable);
+  // Whether this is an unspecified IP address.
+  bool IsNil() const;
 
  private:
   int family_;
@@ -117,8 +118,6 @@ class IPAddress {
     in_addr ip4;
     in6_addr ip6;
   } u_;
-
-  static bool strip_sensitive_;
 };
 
 // IP class which could represent IPv6 address flags which is only
@@ -176,6 +175,9 @@ int IPAddressPrecedence(const IPAddress& ip);
 // Returns 'ip' truncated to be 'length' bits long.
 IPAddress TruncateIP(const IPAddress& ip, int length);
 
+IPAddress GetLoopbackIP(int family);
+IPAddress GetAnyIP(int family);
+
 // Returns the number of contiguously set bits, counting from the MSB in network
 // byte order, in this IPAddress. Bits after the first 0 encountered are not
 // counted.
diff --git a/media/webrtc/trunk/webrtc/base/ipaddress_unittest.cc b/media/webrtc/trunk/webrtc/base/ipaddress_unittest.cc
index 3a8087a5d4..62773c143a 100644
--- a/media/webrtc/trunk/webrtc/base/ipaddress_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/ipaddress_unittest.cc
@@ -25,6 +25,10 @@ static const in6_addr kIPv6PublicAddr = {{{0x24, 0x01, 0xfa, 0x00,
                                            0x00, 0x04, 0x10, 0x00,
                                            0xbe, 0x30, 0x5b, 0xff,
                                            0xfe, 0xe5, 0x00, 0xc3}}};
+static const in6_addr kIPv6PublicAddr2 = {{{0x24, 0x01, 0x00, 0x00,
+                                            0x00, 0x00, 0x10, 0x00,
+                                            0xbe, 0x30, 0x5b, 0xff,
+                                            0xfe, 0xe5, 0x00, 0xc3}}};
 static const in6_addr kIPv4MappedAnyAddr = {{{0x00, 0x00, 0x00, 0x00,
                                               0x00, 0x00, 0x00, 0x00,
                                               0x00, 0x00, 0xff, 0xff,
@@ -52,7 +56,12 @@ static const std::string kIPv6TemporaryAddrString =
     "2620:0:1008:1201:2089:6dda:385e:80c0";
 static const std::string kIPv6PublicAddrString =
     "2401:fa00:4:1000:be30:5bff:fee5:c3";
-static const std::string kIPv6PublicAddrAnonymizedString = "2401:fa00:4::";
+static const std::string kIPv6PublicAddr2String =
+    "2401::1000:be30:5bff:fee5:c3";
+static const std::string kIPv6PublicAddrAnonymizedString =
+    "2401:fa00:4:x:x:x:x:x";
+static const std::string kIPv6PublicAddr2AnonymizedString =
+    "2401:0:0:x:x:x:x:x";
 static const std::string kIPv4MappedAnyAddrString = "::ffff:0:0";
 static const std::string kIPv4MappedRFC1918AddrString = "::ffff:c0a8:701";
 static const std::string kIPv4MappedLoopbackAddrString = "::ffff:7f00:1";
@@ -544,6 +553,19 @@ TEST(IPAddressTest, TestIsPrivate) {
   EXPECT_TRUE(IPIsPrivate(IPAddress(kIPv6LinkLocalAddr)));
 }
 
+TEST(IPAddressTest, TestIsNil) {
+  IPAddress addr;
+  EXPECT_TRUE(IPAddress().IsNil());
+
+  EXPECT_TRUE(IPFromString(kIPv6AnyAddrString, &addr));
+  EXPECT_FALSE(addr.IsNil());
+
+  EXPECT_TRUE(IPFromString(kIPv4AnyAddrString, &addr));
+  EXPECT_FALSE(addr.IsNil());
+
+  EXPECT_FALSE(IPAddress(kIPv4PublicAddr).IsNil());
+}
+
 TEST(IPAddressTest, TestIsLoopback) {
   EXPECT_FALSE(IPIsLoopback(IPAddress(INADDR_ANY)));
   EXPECT_FALSE(IPIsLoopback(IPAddress(kIPv4PublicAddr)));
@@ -875,20 +897,20 @@ TEST(IPAddressTest, TestCategorizeIPv6) {
 
 TEST(IPAddressTest, TestToSensitiveString) {
   IPAddress addr_v4 = IPAddress(kIPv4PublicAddr);
-  EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToString());
-  EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToSensitiveString());
-  IPAddress::set_strip_sensitive(true);
-  EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToString());
-  EXPECT_EQ(kIPv4PublicAddrAnonymizedString, addr_v4.ToSensitiveString());
-  IPAddress::set_strip_sensitive(false);
-
   IPAddress addr_v6 = IPAddress(kIPv6PublicAddr);
+  IPAddress addr_v6_2 = IPAddress(kIPv6PublicAddr2);
+  EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToString());
   EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToString());
-  EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToSensitiveString());
-  IPAddress::set_strip_sensitive(true);
-  EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToString());
+  EXPECT_EQ(kIPv6PublicAddr2String, addr_v6_2.ToString());
+#if defined(NDEBUG)
+  EXPECT_EQ(kIPv4PublicAddrAnonymizedString, addr_v4.ToSensitiveString());
   EXPECT_EQ(kIPv6PublicAddrAnonymizedString, addr_v6.ToSensitiveString());
-  IPAddress::set_strip_sensitive(false);
+  EXPECT_EQ(kIPv6PublicAddr2AnonymizedString, addr_v6_2.ToSensitiveString());
+#else
+  EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToSensitiveString());
+  EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToSensitiveString());
+  EXPECT_EQ(kIPv6PublicAddr2String, addr_v6_2.ToSensitiveString());
+#endif  // defined(NDEBUG)
 }
 
 TEST(IPAddressTest, TestInterfaceAddress) {
diff --git a/media/webrtc/trunk/webrtc/base/java/src/org/webrtc/Logging.java b/media/webrtc/trunk/webrtc/base/java/src/org/webrtc/Logging.java
new file mode 100644
index 0000000000..ea1bca3bdb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/java/src/org/webrtc/Logging.java
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.io.PrintWriter;
+import java.io.StringWriter;
+import java.util.EnumSet;
+import java.util.logging.Logger;
+import java.util.logging.Level;
+
+/** Java wrapper for WebRTC logging. */
+public class Logging {
+  private static final Logger fallbackLogger = Logger.getLogger("org.webrtc.Logging");
+  private static volatile boolean tracingEnabled;
+  private static volatile boolean nativeLibLoaded;
+
+  static {
+    try {
+      System.loadLibrary("jingle_peerconnection_so");
+      nativeLibLoaded = true;
+    } catch (UnsatisfiedLinkError t) {
+      // If native logging is unavailable, log to system log.
+      fallbackLogger.setLevel(Level.ALL);
+
+      fallbackLogger.log(Level.WARNING, "Failed to load jingle_peerconnection_so: ", t);
+    }
+  }
+
+  // Keep in sync with webrtc/common_types.h:TraceLevel.
+  public enum TraceLevel {
+    TRACE_NONE(0x0000),
+    TRACE_STATEINFO(0x0001),
+    TRACE_WARNING(0x0002),
+    TRACE_ERROR(0x0004),
+    TRACE_CRITICAL(0x0008),
+    TRACE_APICALL(0x0010),
+    TRACE_DEFAULT(0x00ff),
+    TRACE_MODULECALL(0x0020),
+    TRACE_MEMORY(0x0100),
+    TRACE_TIMER(0x0200),
+    TRACE_STREAM(0x0400),
+    TRACE_DEBUG(0x0800),
+    TRACE_INFO(0x1000),
+    TRACE_TERSEINFO(0x2000),
+    TRACE_ALL(0xffff);
+
+    public final int level;
+    TraceLevel(int level) {
+      this.level = level;
+    }
+  };
+
+  // Keep in sync with webrtc/base/logging.h:LoggingSeverity.
+  public enum Severity {
+    LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR,
+  };
+
+  public static void enableLogThreads() {
+    if (!nativeLibLoaded) {
+      fallbackLogger.log(Level.WARNING, "Cannot enable log thread because native lib not loaded.");
+      return;
+    }
+    nativeEnableLogThreads();
+  }
+
+  public static void enableLogTimeStamps() {
+    if (!nativeLibLoaded) {
+      fallbackLogger.log(Level.WARNING,
+                         "Cannot enable log timestamps because native lib not loaded.");
+      return;
+    }
+    nativeEnableLogTimeStamps();
+  }
+
+  // Enable tracing to |path| of messages of |levels| and |severity|.
+  // On Android, use "logcat:" for |path| to send output there.
+  public static synchronized void enableTracing(
+      String path, EnumSet<TraceLevel> levels, Severity severity) {
+    if (!nativeLibLoaded) {
+      fallbackLogger.log(Level.WARNING, "Cannot enable tracing because native lib not loaded.");
+      return;
+    }
+
+    if (tracingEnabled) {
+      return;
+    }
+    int nativeLevel = 0;
+    for (TraceLevel level : levels) {
+      nativeLevel |= level.level;
+    }
+    nativeEnableTracing(path, nativeLevel, severity.ordinal());
+    tracingEnabled = true;
+  }
+
+  public static void log(Severity severity, String tag, String message) {
+    if (tracingEnabled) {
+      nativeLog(severity.ordinal(), tag, message);
+      return;
+    }
+
+    // Fallback to system log.
+    Level level;
+    switch (severity) {
+      case LS_ERROR:
+        level = Level.SEVERE;
+        break;
+      case LS_WARNING:
+        level = Level.WARNING;
+        break;
+      case LS_INFO:
+        level = Level.INFO;
+        break;
+      default:
+        level = Level.FINE;
+        break;
+    }
+    fallbackLogger.log(level, tag + ": " + message);
+  }
+
+  public static void d(String tag, String message) {
+    log(Severity.LS_INFO, tag, message);
+  }
+
+  public static void e(String tag, String message) {
+    log(Severity.LS_ERROR, tag, message);
+  }
+
+  public static void w(String tag, String message) {
+    log(Severity.LS_WARNING, tag, message);
+  }
+
+  public static void e(String tag, String message, Throwable e) {
+    log(Severity.LS_ERROR, tag, message);
+    log(Severity.LS_ERROR, tag, e.toString());
+    log(Severity.LS_ERROR, tag, getStackTraceString(e));
+  }
+
+  public static void w(String tag, String message, Throwable e) {
+    log(Severity.LS_WARNING, tag, message);
+    log(Severity.LS_WARNING, tag, e.toString());
+    log(Severity.LS_WARNING, tag, getStackTraceString(e));
+  }
+
+  public static void v(String tag, String message) {
+    log(Severity.LS_VERBOSE, tag, message);
+  }
+
+  private static String getStackTraceString(Throwable e) {
+    if (e == null) {
+      return "";
+    }
+
+    StringWriter sw = new StringWriter();
+    PrintWriter pw = new PrintWriter(sw);
+    e.printStackTrace(pw);
+    return sw.toString();
+  }
+
+  private static native void nativeEnableTracing(
+      String path, int nativeLevels, int nativeSeverity);
+  private static native void nativeEnableLogThreads();
+  private static native void nativeEnableLogTimeStamps();
+  private static native void nativeLog(int severity, String tag, String message);
+}
diff --git a/media/webrtc/trunk/webrtc/base/keep_ref_until_done.h b/media/webrtc/trunk/webrtc/base/keep_ref_until_done.h
new file mode 100644
index 0000000000..269e1c8657
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/keep_ref_until_done.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
+#define WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/callback.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace rtc {
+
+namespace impl {
+template <class T>
+static inline void DoNothing(const scoped_refptr<T>& object) {}
+}  // namespace impl
+
+// KeepRefUntilDone keeps a reference to |object| until the returned
+// callback goes out of scope. If the returned callback is copied, the
+// reference will be released when the last callback goes out of scope.
+template <class ObjectT>
+static inline Callback0<void> KeepRefUntilDone(ObjectT* object) {
+  return rtc::Bind(&impl::DoNothing<ObjectT>, scoped_refptr<ObjectT>(object));
+}
+
+template <class ObjectT>
+static inline Callback0<void> KeepRefUntilDone(
+    const scoped_refptr<ObjectT>& object) {
+  return rtc::Bind(&impl::DoNothing<ObjectT>, object);
+}
+
+}  // namespace rtc
+
+
+#endif  // WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
diff --git a/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h b/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h
index c1f535cd2b..636e7d0707 100644
--- a/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h
+++ b/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h
@@ -61,7 +61,7 @@ class LateBindingSymbolTable {
   DllHandle handle_;
   bool undefined_symbols_;
 
-  DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h.def b/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h.def
index 39b515fbd1..bc6396b1f0 100644
--- a/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h.def
+++ b/media/webrtc/trunk/webrtc/base/latebindingsymboltable.h.def
@@ -76,7 +76,7 @@ LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST
 
   void *table_[SYMBOL_TABLE_SIZE];
 
-  DISALLOW_COPY_AND_ASSIGN(LATE_BINDING_SYMBOL_TABLE_CLASS_NAME);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LATE_BINDING_SYMBOL_TABLE_CLASS_NAME);
 };
 
 #undef LATE_BINDING_SYMBOL_TABLE_CLASS_NAME
diff --git a/media/webrtc/trunk/webrtc/base/latebindingsymboltable_unittest.cc b/media/webrtc/trunk/webrtc/base/latebindingsymboltable_unittest.cc
index 30ebd17cba..0079f20342 100644
--- a/media/webrtc/trunk/webrtc/base/latebindingsymboltable_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/latebindingsymboltable_unittest.cc
@@ -21,9 +21,10 @@ namespace rtc {
 
 #define LIBM_SYMBOLS_CLASS_NAME LibmTestSymbolTable
 #define LIBM_SYMBOLS_LIST \
-  X(acos) \
-  X(sin) \
-  X(tan)
+  X(acosf) \
+  X(sinf) \
+  X(tanf)
+
 
 #define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME LIBM_SYMBOLS_CLASS_NAME
 #define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST LIBM_SYMBOLS_LIST
@@ -39,9 +40,9 @@ TEST(LateBindingSymbolTable, libm) {
   EXPECT_FALSE(table.IsLoaded());
   ASSERT_TRUE(table.Load());
   EXPECT_TRUE(table.IsLoaded());
-  EXPECT_EQ(table.acos()(0.5), acos(0.5));
-  EXPECT_EQ(table.sin()(0.5), sin(0.5));
-  EXPECT_EQ(table.tan()(0.5), tan(0.5));
+  EXPECT_EQ(table.acosf()(0.5f), acosf(0.5f));
+  EXPECT_EQ(table.sinf()(0.5f), sinf(0.5f));
+  EXPECT_EQ(table.tanf()(0.5f), tanf(0.5f));
   // It would be nice to check that the addresses are the same, but the nature
   // of dynamic linking and relocation makes them actually be different.
   table.Unload();
diff --git a/media/webrtc/trunk/webrtc/base/linux.cc b/media/webrtc/trunk/webrtc/base/linux.cc
index 9800f471d5..0894d39c77 100644
--- a/media/webrtc/trunk/webrtc/base/linux.cc
+++ b/media/webrtc/trunk/webrtc/base/linux.cc
@@ -233,90 +233,6 @@ bool ConfigParser::ParseLine(std::string* key, std::string* value) {
   return true;
 }
 
-#if !defined(WEBRTC_CHROMIUM_BUILD)
-static bool ExpectLineFromStream(FileStream* stream,
-                                 std::string* out) {
-  StreamResult res = stream->ReadLine(out);
-  if (res != SR_SUCCESS) {
-    if (res != SR_EOS) {
-      LOG(LS_ERROR) << "Error when reading from stream";
-    } else {
-      LOG(LS_ERROR) << "Incorrect number of lines in stream";
-    }
-    return false;
-  }
-  return true;
-}
-
-static void ExpectEofFromStream(FileStream* stream) {
-  std::string unused;
-  StreamResult res = stream->ReadLine(&unused);
-  if (res == SR_SUCCESS) {
-    LOG(LS_WARNING) << "Ignoring unexpected extra lines from stream";
-  } else if (res != SR_EOS) {
-    LOG(LS_WARNING) << "Error when checking for extra lines from stream";
-  }
-}
-
-// For caching the lsb_release output (reading it invokes a sub-process and
-// hence is somewhat expensive).
-static std::string lsb_release_string;
-static CriticalSection lsb_release_string_critsec;
-
-std::string ReadLinuxLsbRelease() {
-  CritScope cs(&lsb_release_string_critsec);
-  if (!lsb_release_string.empty()) {
-    // Have cached result from previous call.
-    return lsb_release_string;
-  }
-  // No cached result. Run lsb_release and parse output.
-  POpenStream lsb_release_output;
-  if (!lsb_release_output.Open("lsb_release -idrcs", "r", NULL)) {
-    LOG_ERR(LS_ERROR) << "Can't run lsb_release";
-    return lsb_release_string;  // empty
-  }
-  // Read in the command's output and build the string.
-  std::ostringstream sstr;
-  std::string line;
-  int wait_status;
-
-  if (!ExpectLineFromStream(&lsb_release_output, &line)) {
-    return lsb_release_string;  // empty
-  }
-  sstr << "DISTRIB_ID=" << line;
-
-  if (!ExpectLineFromStream(&lsb_release_output, &line)) {
-    return lsb_release_string;  // empty
-  }
-  sstr << " DISTRIB_DESCRIPTION=\"" << line << '"';
-
-  if (!ExpectLineFromStream(&lsb_release_output, &line)) {
-    return lsb_release_string;  // empty
-  }
-  sstr << " DISTRIB_RELEASE=" << line;
-
-  if (!ExpectLineFromStream(&lsb_release_output, &line)) {
-    return lsb_release_string;  // empty
-  }
-  sstr << " DISTRIB_CODENAME=" << line;
-
-  // Should not be anything left.
-  ExpectEofFromStream(&lsb_release_output);
-
-  lsb_release_output.Close();
-  wait_status = lsb_release_output.GetWaitStatus();
-  if (wait_status == -1 ||
-      !WIFEXITED(wait_status) ||
-      WEXITSTATUS(wait_status) != 0) {
-    LOG(LS_WARNING) << "Unexpected exit status from lsb_release";
-  }
-
-  lsb_release_string = sstr.str();
-
-  return lsb_release_string;
-}
-#endif
-
 std::string ReadLinuxUname() {
   struct utsname buf;
   if (uname(&buf) < 0) {
diff --git a/media/webrtc/trunk/webrtc/base/linux.h b/media/webrtc/trunk/webrtc/base/linux.h
index dd863ba495..ba73b854ba 100644
--- a/media/webrtc/trunk/webrtc/base/linux.h
+++ b/media/webrtc/trunk/webrtc/base/linux.h
@@ -104,11 +104,6 @@ class ProcCpuInfo {
   ConfigParser::MapVector sections_;
 };
 
-#if !defined(WEBRTC_CHROMIUM_BUILD)
-// Builds a string containing the info from lsb_release on a single line.
-std::string ReadLinuxLsbRelease();
-#endif
-
 // Returns the output of "uname".
 std::string ReadLinuxUname();
 
diff --git a/media/webrtc/trunk/webrtc/base/linux_unittest.cc b/media/webrtc/trunk/webrtc/base/linux_unittest.cc
index 19401b439a..80d469f29d 100644
--- a/media/webrtc/trunk/webrtc/base/linux_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/linux_unittest.cc
@@ -88,14 +88,6 @@ TEST(ConfigParser, ParseConfig) {
   EXPECT_EQ(true, parser.Parse(&key_val_pairs));
 }
 
-#if !defined(WEBRTC_CHROMIUM_BUILD)
-TEST(ReadLinuxLsbRelease, ReturnsSomething) {
-  std::string str = ReadLinuxLsbRelease();
-  // ChromeOS don't have lsb_release
-  // EXPECT_FALSE(str.empty());
-}
-#endif
-
 TEST(ReadLinuxUname, ReturnsSomething) {
   std::string str = ReadLinuxUname();
   EXPECT_FALSE(str.empty());
diff --git a/media/webrtc/trunk/webrtc/base/logging.cc b/media/webrtc/trunk/webrtc/base/logging.cc
index 2cf567c552..686b9b2b02 100644
--- a/media/webrtc/trunk/webrtc/base/logging.cc
+++ b/media/webrtc/trunk/webrtc/base/logging.cc
@@ -9,7 +9,9 @@
  */
 
 #if defined(WEBRTC_WIN)
+#if !defined(WIN32_LEAN_AND_MEAN)
 #define WIN32_LEAN_AND_MEAN
+#endif
 #include <windows.h>
 #define snprintf _snprintf
 #undef ERROR  // wingdi.h
@@ -19,13 +21,14 @@
 #include <CoreServices/CoreServices.h>
 #elif defined(WEBRTC_ANDROID)
 #include <android/log.h>
-static const char kLibjingle[] = "libjingle";
 // Android has a 1024 limit on log inputs. We use 60 chars as an
 // approx for the header/tag portion.
 // See android/system/core/liblog/logd_write.c
 static const int kMaxLogLineSize = 1024 - 60;
 #endif  // WEBRTC_MAC && !defined(WEBRTC_IOS) || WEBRTC_ANDROID
 
+static const char kLibjingle[] = "libjingle";
+
 #include <time.h>
 #include <limits.h>
 
@@ -34,19 +37,34 @@ static const int kMaxLogLineSize = 1024 - 60;
 #include <ostream>
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/stream.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
 
 namespace rtc {
+namespace {
+
+// Return the filename portion of the string (that following the last slash).
+const char* FilenameFromPath(const char* file) {
+  const char* end1 = ::strrchr(file, '/');
+  const char* end2 = ::strrchr(file, '\\');
+  if (!end1 && !end2)
+    return file;
+  else
+    return (end1 > end2) ? end1 + 1 : end2 + 1;
+}
+
+}  // namespace
 
 /////////////////////////////////////////////////////////////////////////////
 // Constant Labels
 /////////////////////////////////////////////////////////////////////////////
 
-const char * FindLabel(int value, const ConstantLabel entries[]) {
+const char* FindLabel(int value, const ConstantLabel entries[]) {
   for (int i = 0; entries[i].label; ++i) {
     if (value == entries[i].value) {
       return entries[i].label;
@@ -55,12 +73,12 @@ const char * FindLabel(int value, const ConstantLabel entries[]) {
   return 0;
 }
 
-std::string ErrorName(int err, const ConstantLabel * err_table) {
+std::string ErrorName(int err, const ConstantLabel* err_table) {
   if (err == 0)
     return "No error";
 
   if (err_table != 0) {
-    if (const char * value = FindLabel(err, err_table))
+    if (const char* value = FindLabel(err, err_table))
       return value;
   }
 
@@ -73,42 +91,39 @@ std::string ErrorName(int err, const ConstantLabel * err_table) {
 // LogMessage
 /////////////////////////////////////////////////////////////////////////////
 
-const int LogMessage::NO_LOGGING = LS_ERROR + 1;
-
-#if _DEBUG
-static const int LOG_DEFAULT = LS_INFO;
-#else  // !_DEBUG
-static const int LOG_DEFAULT = LogMessage::NO_LOGGING;
-#endif  // !_DEBUG
-
-// Global lock for log subsystem, only needed to serialize access to streams_.
-CriticalSection LogMessage::crit_;
-
 // By default, release builds don't log, debug builds at info level
-int LogMessage::min_sev_ = LOG_DEFAULT;
-int LogMessage::dbg_sev_ = LOG_DEFAULT;
+#if !defined(NDEBUG)
+LoggingSeverity LogMessage::min_sev_ = LS_INFO;
+LoggingSeverity LogMessage::dbg_sev_ = LS_INFO;
+#else
+LoggingSeverity LogMessage::min_sev_ = LS_NONE;
+LoggingSeverity LogMessage::dbg_sev_ = LS_NONE;
+#endif
+bool LogMessage::log_to_stderr_ = true;
 
-// Don't bother printing context for the ubiquitous INFO log messages
-int LogMessage::ctx_sev_ = LS_WARNING;
+namespace {
+// Global lock for log subsystem, only needed to serialize access to streams_.
+CriticalSection g_log_crit;
+}  // namespace
 
 // The list of logging streams currently configured.
 // Note: we explicitly do not clean this up, because of the uncertain ordering
 // of destructors at program exit.  Let the person who sets the stream trigger
 // cleanup by setting to NULL, or let it leak (safe at program exit).
-LogMessage::StreamList LogMessage::streams_;
+LogMessage::StreamList LogMessage::streams_ GUARDED_BY(g_log_crit);
 
 // Boolean options default to false (0)
 bool LogMessage::thread_, LogMessage::timestamp_;
 
-// If we're in diagnostic mode, we'll be explicitly set that way; default=false.
-bool LogMessage::is_diagnostic_mode_ = false;
-
-LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev,
-                       LogErrorContext err_ctx, int err, const char* module)
-    : severity_(sev),
-      warn_slow_logs_delay_(WARN_SLOW_LOGS_DELAY) {
+LogMessage::LogMessage(const char* file,
+                       int line,
+                       LoggingSeverity sev,
+                       LogErrorContext err_ctx,
+                       int err,
+                       const char* module)
+    : severity_(sev), tag_(kLibjingle) {
   if (timestamp_) {
-    uint32 time = TimeSince(LogStartTime());
+    uint32_t time = TimeSince(LogStartTime());
     // Also ensure WallClockStartTime is initialized, so that it matches
     // LogStartTime.
     WallClockStartTime();
@@ -118,16 +133,12 @@ LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev,
   }
 
   if (thread_) {
-#if defined(WEBRTC_WIN)
-    DWORD id = GetCurrentThreadId();
-    print_stream_ << "[" << std::hex << id << std::dec << "] ";
-#endif  // WEBRTC_WIN 
+    PlatformThreadId id = CurrentThreadId();
+    print_stream_ << "[" << std::dec << id << "] ";
   }
 
-  if (severity_ >= ctx_sev_) {
-    print_stream_ << Describe(sev) << "(" << DescribeFile(file)
-                  << ":" << line << "): ";
-  }
+  if (file != NULL)
+    print_stream_ << "(" << FilenameFromPath(file)  << ":" << line << "): ";
 
   if (err_ctx != ERRCTX_NONE) {
     std::ostringstream tmp;
@@ -155,7 +166,7 @@ LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev,
         }
         break;
       }
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
       case ERRCTX_OSSTATUS: {
         tmp << " " << nonnull(GetMacOSStatusErrorString(err), "Unknown error");
@@ -172,6 +183,15 @@ LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev,
   }
 }
 
+LogMessage::LogMessage(const char* file,
+                       int line,
+                       LoggingSeverity sev,
+                       const std::string& tag)
+    : LogMessage(file, line, sev, ERRCTX_NONE, 0 /* err */, NULL /* module */) {
+  tag_ = tag;
+  print_stream_ << tag << ": ";
+}
+
 LogMessage::~LogMessage() {
   if (!extra_.empty())
     print_stream_ << " : " << extra_;
@@ -179,44 +199,27 @@ LogMessage::~LogMessage() {
 
   const std::string& str = print_stream_.str();
   if (severity_ >= dbg_sev_) {
-    OutputToDebug(str, severity_);
+    OutputToDebug(str, severity_, tag_);
   }
 
-  uint32 before = Time();
-  // Must lock streams_ before accessing
-  CritScope cs(&crit_);
-  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
-    if (severity_ >= it->second) {
-      OutputToStream(it->first, str);
+  CritScope cs(&g_log_crit);
+  for (auto& kv : streams_) {
+    if (severity_ >= kv.second) {
+      kv.first->OnLogMessage(str);
     }
   }
-  uint32 delay = TimeSince(before);
-  if (delay >= warn_slow_logs_delay_) {
-    LogMessage slow_log_warning =
-        rtc::LogMessage(__FILE__, __LINE__, LS_WARNING);
-    // If our warning is slow, we don't want to warn about it, because
-    // that would lead to inifinite recursion.  So, give a really big
-    // number for the delay threshold.
-    slow_log_warning.warn_slow_logs_delay_ = UINT_MAX;
-    slow_log_warning.stream() << "Slow log: took " << delay << "ms to write "
-                              << str.size() << " bytes.";
-  }
 }
 
-uint32 LogMessage::LogStartTime() {
-  static const uint32 g_start = Time();
+uint32_t LogMessage::LogStartTime() {
+  static const uint32_t g_start = Time();
   return g_start;
 }
 
-uint32 LogMessage::WallClockStartTime() {
-  static const uint32 g_start_wallclock = time(NULL);
+uint32_t LogMessage::WallClockStartTime() {
+  static const uint32_t g_start_wallclock = time(NULL);
   return g_start_wallclock;
 }
 
-void LogMessage::LogContext(int min_sev) {
-  ctx_sev_ = min_sev;
-}
-
 void LogMessage::LogThreads(bool on) {
   thread_ = on;
 }
@@ -225,43 +228,35 @@ void LogMessage::LogTimestamps(bool on) {
   timestamp_ = on;
 }
 
-void LogMessage::LogToDebug(int min_sev) {
+void LogMessage::LogToDebug(LoggingSeverity min_sev) {
   dbg_sev_ = min_sev;
+  CritScope cs(&g_log_crit);
   UpdateMinLogSeverity();
 }
 
-void LogMessage::LogToStream(StreamInterface* stream, int min_sev) {
-  CritScope cs(&crit_);
-  // Discard and delete all previously installed streams
-  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
-    delete it->first;
-  }
-  streams_.clear();
-  // Install the new stream, if specified
-  if (stream) {
-    AddLogToStream(stream, min_sev);
-  }
+void LogMessage::SetLogToStderr(bool log_to_stderr) {
+  log_to_stderr_ = log_to_stderr;
 }
 
-int LogMessage::GetLogToStream(StreamInterface* stream) {
-  CritScope cs(&crit_);
-  int sev = NO_LOGGING;
-  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
-    if (!stream || stream == it->first) {
-      sev = std::min(sev, it->second);
+int LogMessage::GetLogToStream(LogSink* stream) {
+  CritScope cs(&g_log_crit);
+  LoggingSeverity sev = LS_NONE;
+  for (auto& kv : streams_) {
+    if (!stream || stream == kv.first) {
+      sev = std::min(sev, kv.second);
     }
   }
   return sev;
 }
 
-void LogMessage::AddLogToStream(StreamInterface* stream, int min_sev) {
-  CritScope cs(&crit_);
+void LogMessage::AddLogToStream(LogSink* stream, LoggingSeverity min_sev) {
+  CritScope cs(&g_log_crit);
   streams_.push_back(std::make_pair(stream, min_sev));
   UpdateMinLogSeverity();
 }
 
-void LogMessage::RemoveLogToStream(StreamInterface* stream) {
-  CritScope cs(&crit_);
+void LogMessage::RemoveLogToStream(LogSink* stream) {
+  CritScope cs(&g_log_crit);
   for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
     if (stream == it->first) {
       streams_.erase(it);
@@ -271,48 +266,45 @@ void LogMessage::RemoveLogToStream(StreamInterface* stream) {
   UpdateMinLogSeverity();
 }
 
-void LogMessage::ConfigureLogging(const char* params, const char* filename) {
-  int current_level = LS_VERBOSE;
-  int debug_level = GetLogToDebug();
-  int file_level = GetLogToStream();
+void LogMessage::ConfigureLogging(const char* params) {
+  LoggingSeverity current_level = LS_VERBOSE;
+  LoggingSeverity debug_level = GetLogToDebug();
 
   std::vector<std::string> tokens;
   tokenize(params, ' ', &tokens);
 
-  for (size_t i = 0; i < tokens.size(); ++i) {
-    if (tokens[i].empty())
+  for (const std::string& token : tokens) {
+    if (token.empty())
       continue;
 
     // Logging features
-    if (tokens[i] == "tstamp") {
+    if (token == "tstamp") {
       LogTimestamps();
-    } else if (tokens[i] == "thread") {
+    } else if (token == "thread") {
       LogThreads();
 
     // Logging levels
-    } else if (tokens[i] == "sensitive") {
+    } else if (token == "sensitive") {
       current_level = LS_SENSITIVE;
-    } else if (tokens[i] == "verbose") {
+    } else if (token == "verbose") {
       current_level = LS_VERBOSE;
-    } else if (tokens[i] == "info") {
+    } else if (token == "info") {
       current_level = LS_INFO;
-    } else if (tokens[i] == "warning") {
+    } else if (token == "warning") {
       current_level = LS_WARNING;
-    } else if (tokens[i] == "error") {
+    } else if (token == "error") {
       current_level = LS_ERROR;
-    } else if (tokens[i] == "none") {
-      current_level = NO_LOGGING;
+    } else if (token == "none") {
+      current_level = LS_NONE;
 
     // Logging targets
-    } else if (tokens[i] == "file") {
-      file_level = current_level;
-    } else if (tokens[i] == "debug") {
+    } else if (token == "debug") {
       debug_level = current_level;
     }
   }
 
 #if defined(WEBRTC_WIN)
-  if ((NO_LOGGING != debug_level) && !::IsDebuggerPresent()) {
+  if ((LS_NONE != debug_level) && !::IsDebuggerPresent()) {
     // First, attempt to attach to our parent's console... so if you invoke
     // from the command line, we'll see the output there.  Otherwise, create
     // our own console window.
@@ -331,73 +323,24 @@ void LogMessage::ConfigureLogging(const char* params, const char* filename) {
       ::AllocConsole();
     }
   }
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
   LogToDebug(debug_level);
-
-#if !defined(__native_client__)  // No logging to file in NaCl.
-  scoped_ptr<FileStream> stream;
-  if (NO_LOGGING != file_level) {
-    stream.reset(new FileStream);
-    if (!stream->Open(filename, "wb", NULL) || !stream->DisableBuffering()) {
-      stream.reset();
-    }
-  }
-
-  LogToStream(stream.release(), file_level);
-#endif
 }
 
-int LogMessage::ParseLogSeverity(const std::string& value) {
-  int level = NO_LOGGING;
-  if (value == "LS_SENSITIVE") {
-    level = LS_SENSITIVE;
-  } else if (value == "LS_VERBOSE") {
-    level = LS_VERBOSE;
-  } else if (value == "LS_INFO") {
-    level = LS_INFO;
-  } else if (value == "LS_WARNING") {
-    level = LS_WARNING;
-  } else if (value == "LS_ERROR") {
-    level = LS_ERROR;
-  } else if (isdigit(value[0])) {
-    level = atoi(value.c_str());  // NOLINT
-  }
-  return level;
-}
-
-void LogMessage::UpdateMinLogSeverity() {
-  int min_sev = dbg_sev_;
-  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
-    min_sev = std::min(dbg_sev_, it->second);
+void LogMessage::UpdateMinLogSeverity() EXCLUSIVE_LOCKS_REQUIRED(g_log_crit) {
+  LoggingSeverity min_sev = dbg_sev_;
+  for (auto& kv : streams_) {
+    min_sev = std::min(dbg_sev_, kv.second);
   }
   min_sev_ = min_sev;
 }
 
-const char* LogMessage::Describe(LoggingSeverity sev) {
-  switch (sev) {
-  case LS_SENSITIVE: return "Sensitive";
-  case LS_VERBOSE:   return "Verbose";
-  case LS_INFO:      return "Info";
-  case LS_WARNING:   return "Warning";
-  case LS_ERROR:     return "Error";
-  default:           return "<unknown>";
-  }
-}
-
-const char* LogMessage::DescribeFile(const char* file) {
-  const char* end1 = ::strrchr(file, '/');
-  const char* end2 = ::strrchr(file, '\\');
-  if (!end1 && !end2)
-    return file;
-  else
-    return (end1 > end2) ? end1 + 1 : end2 + 1;
-}
-
 void LogMessage::OutputToDebug(const std::string& str,
-                               LoggingSeverity severity) {
-  bool log_to_stderr = true;
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && (!defined(DEBUG) || defined(NDEBUG))
+                               LoggingSeverity severity,
+                               const std::string& tag) {
+  bool log_to_stderr = log_to_stderr_;
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && defined(NDEBUG)
   // On the Mac, all stderr output goes to the Console log and causes clutter.
   // So in opt builds, don't log to stderr unless the user specifically sets
   // a preference to do so.
@@ -430,7 +373,7 @@ void LogMessage::OutputToDebug(const std::string& str,
                   &written, 0);
     }
   }
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 #if defined(WEBRTC_ANDROID)
   // Android's logging facility uses severity to log messages but we
   // need to map libjingle's severity levels to Android ones first.
@@ -439,7 +382,7 @@ void LogMessage::OutputToDebug(const std::string& str,
   int prio;
   switch (severity) {
     case LS_SENSITIVE:
-      __android_log_write(ANDROID_LOG_INFO, kLibjingle, "SENSITIVE");
+      __android_log_write(ANDROID_LOG_INFO, tag.c_str(), "SENSITIVE");
       if (log_to_stderr) {
         fprintf(stderr, "SENSITIVE");
         fflush(stderr);
@@ -466,13 +409,13 @@ void LogMessage::OutputToDebug(const std::string& str,
   int idx = 0;
   const int max_lines = size / kMaxLogLineSize + 1;
   if (max_lines == 1) {
-    __android_log_print(prio, kLibjingle, "%.*s", size, str.c_str());
+    __android_log_print(prio, tag.c_str(), "%.*s", size, str.c_str());
   } else {
     while (size > 0) {
       const int len = std::min(size, kMaxLogLineSize);
       // Use the size of the string in the format (str may have \0 in the
       // middle).
-      __android_log_print(prio, kLibjingle, "[%d/%d] %.*s",
+      __android_log_print(prio, tag.c_str(), "[%d/%d] %.*s",
                           line + 1, max_lines,
                           len, str.c_str() + idx);
       idx += len;
@@ -487,12 +430,6 @@ void LogMessage::OutputToDebug(const std::string& str,
   }
 }
 
-void LogMessage::OutputToStream(StreamInterface* stream,
-                                const std::string& str) {
-  // If write isn't fully successful, what are we going to do, log it? :)
-  stream->WriteAll(str.data(), str.size(), NULL, NULL);
-}
-
 //////////////////////////////////////////////////////////////////////
 // Logging Helpers
 //////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/base/logging.h b/media/webrtc/trunk/webrtc/base/logging.h
index e07045ff1d..e40ca4465f 100644
--- a/media/webrtc/trunk/webrtc/base/logging.h
+++ b/media/webrtc/trunk/webrtc/base/logging.h
@@ -10,7 +10,7 @@
 
 //   LOG(...) an ostream target that can be used to send formatted
 // output to a variety of logging targets, such as debugger console, stderr,
-// file, or any StreamInterface.
+// or any LogSink.
 //   The severity level passed as the first argument to the LOGging
 // functions is used as a filter, to limit the verbosity of the logging.
 //   Static members of LogMessage documented below are used to control the
@@ -54,13 +54,13 @@
 #include <sstream>
 #include <string>
 #include <utility>
+
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/thread_annotations.h"
 
 namespace rtc {
 
-class StreamInterface;
-
 ///////////////////////////////////////////////////////////////////////////////
 // ConstantLabel can be used to easily generate string names from constant
 // values.  This can be useful for logging descriptive names of error messages.
@@ -81,7 +81,7 @@ struct ConstantLabel { int value; const char * label; };
 #define TLABEL(x, y) { x, y }
 #define LASTLABEL { 0, 0 }
 
-const char * FindLabel(int value, const ConstantLabel entries[]);
+const char* FindLabel(int value, const ConstantLabel entries[]);
 std::string ErrorName(int err, const ConstantLabel* err_table);
 
 //////////////////////////////////////////////////////////////////////
@@ -96,10 +96,18 @@ std::string ErrorName(int err, const ConstantLabel* err_table);
 //   in debug builds.
 //  LS_WARNING: Something that may warrant investigation.
 //  LS_ERROR: Something that should not have occurred.
-enum LoggingSeverity { LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR,
-                       INFO = LS_INFO,
-                       WARNING = LS_WARNING,
-                       LERROR = LS_ERROR };
+//  LS_NONE: Don't log.
+enum LoggingSeverity {
+  LS_SENSITIVE,
+  LS_VERBOSE,
+  LS_INFO,
+  LS_WARNING,
+  LS_ERROR,
+  LS_NONE,
+  INFO = LS_INFO,
+  WARNING = LS_WARNING,
+  LERROR = LS_ERROR
+};
 
 // LogErrorContext assists in interpreting the meaning of an error value.
 enum LogErrorContext {
@@ -114,14 +122,25 @@ enum LogErrorContext {
   ERRCTX_OS = ERRCTX_OSSTATUS,  // LOG_E(sev, OS, x)
 };
 
+// Virtual sink interface that can receive log messages.
+class LogSink {
+ public:
+  LogSink() {}
+  virtual ~LogSink() {}
+  virtual void OnLogMessage(const std::string& message) = 0;
+};
+
 class LogMessage {
  public:
-  static const int NO_LOGGING;
-  static const uint32 WARN_SLOW_LOGS_DELAY = 50;  // ms
-
   LogMessage(const char* file, int line, LoggingSeverity sev,
              LogErrorContext err_ctx = ERRCTX_NONE, int err = 0,
              const char* module = NULL);
+
+  LogMessage(const char* file,
+             int line,
+             LoggingSeverity sev,
+             const std::string& tag);
+
   ~LogMessage();
 
   static inline bool Loggable(LoggingSeverity sev) { return (sev >= min_sev_); }
@@ -132,24 +151,25 @@ class LogMessage {
   // If this is not called externally, the LogMessage ctor also calls it, in
   // which case the logging start time will be the time of the first LogMessage
   // instance is created.
-  static uint32 LogStartTime();
+  static uint32_t LogStartTime();
 
   // Returns the wall clock equivalent of |LogStartTime|, in seconds from the
   // epoch.
-  static uint32 WallClockStartTime();
+  static uint32_t WallClockStartTime();
 
-  // These are attributes which apply to all logging channels
-  //  LogContext: Display the file and line number of the message
-  static void LogContext(int min_sev);
   //  LogThreads: Display the thread identifier of the current thread
   static void LogThreads(bool on = true);
+
   //  LogTimestamps: Display the elapsed time of the program
   static void LogTimestamps(bool on = true);
 
   // These are the available logging channels
   //  Debug: Debug console on Windows, otherwise stderr
-  static void LogToDebug(int min_sev);
-  static int GetLogToDebug() { return dbg_sev_; }
+  static void LogToDebug(LoggingSeverity min_sev);
+  static LoggingSeverity GetLogToDebug() { return dbg_sev_; }
+
+  // Sets whether logs will be directed to stderr in debug mode.
+  static void SetLogToStderr(bool log_to_stderr);
 
   //  Stream: Any non-blocking stream interface.  LogMessage takes ownership of
   //   the stream. Multiple streams may be specified by using AddLogToStream.
@@ -158,39 +178,29 @@ class LogMessage {
   //   GetLogToStream gets the severity for the specified stream, of if none
   //   is specified, the minimum stream severity.
   //   RemoveLogToStream removes the specified stream, without destroying it.
-  static void LogToStream(StreamInterface* stream, int min_sev);
-  static int GetLogToStream(StreamInterface* stream = NULL);
-  static void AddLogToStream(StreamInterface* stream, int min_sev);
-  static void RemoveLogToStream(StreamInterface* stream);
+  static int GetLogToStream(LogSink* stream = NULL);
+  static void AddLogToStream(LogSink* stream, LoggingSeverity min_sev);
+  static void RemoveLogToStream(LogSink* stream);
 
   // Testing against MinLogSeverity allows code to avoid potentially expensive
   // logging operations by pre-checking the logging level.
   static int GetMinLogSeverity() { return min_sev_; }
 
-  static void SetDiagnosticMode(bool f) { is_diagnostic_mode_ = f; }
-  static bool IsDiagnosticMode() { return is_diagnostic_mode_; }
-
   // Parses the provided parameter stream to configure the options above.
-  // Useful for configuring logging from the command line.  If file logging
-  // is enabled, it is output to the specified filename.
-  static void ConfigureLogging(const char* params, const char* filename);
-
-  // Convert the string to a LS_ value; also accept numeric values.
-  static int ParseLogSeverity(const std::string& value);
+  // Useful for configuring logging from the command line.
+  static void ConfigureLogging(const char* params);
 
  private:
-  typedef std::list<std::pair<StreamInterface*, int> > StreamList;
+  typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
+  typedef std::list<StreamAndSeverity> StreamList;
 
   // Updates min_sev_ appropriately when debug sinks change.
   static void UpdateMinLogSeverity();
 
-  // These assist in formatting some parts of the debug output.
-  static const char* Describe(LoggingSeverity sev);
-  static const char* DescribeFile(const char* file);
-
   // These write out the actual log messages.
-  static void OutputToDebug(const std::string& msg, LoggingSeverity severity_);
-  static void OutputToStream(StreamInterface* stream, const std::string& msg);
+  static void OutputToDebug(const std::string& msg,
+                            LoggingSeverity severity,
+                            const std::string& tag);
 
   // The ostream that buffers the formatted message before output
   std::ostringstream print_stream_;
@@ -198,23 +208,19 @@ class LogMessage {
   // The severity level of this message
   LoggingSeverity severity_;
 
+  // The Android debug output tag.
+  std::string tag_;
+
   // String data generated in the constructor, that should be appended to
   // the message before output.
   std::string extra_;
 
-  // If time it takes to write to stream is more than this, log one
-  // additional warning about it.
-  uint32 warn_slow_logs_delay_;
-
-  // Global lock for the logging subsystem
-  static CriticalSection crit_;
-
   // dbg_sev_ is the thresholds for those output targets
   // min_sev_ is the minimum (most verbose) of those levels, and is used
   //  as a short-circuit in the logging macros to identify messages that won't
   //  be logged.
   // ctx_sev_ is the minimum level at which file context is displayed
-  static int min_sev_, dbg_sev_, ctx_sev_;
+  static LoggingSeverity min_sev_, dbg_sev_, ctx_sev_;
 
   // The output streams and their associated severities
   static StreamList streams_;
@@ -222,10 +228,10 @@ class LogMessage {
   // Flags for formatting options
   static bool thread_, timestamp_;
 
-  // are we in diagnostic mode (as defined by the app)?
-  static bool is_diagnostic_mode_;
+  // Determines if logs will be directed to stderr in debug mode.
+  static bool log_to_stderr_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(LogMessage);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LogMessage);
 };
 
 //////////////////////////////////////////////////////////////////////
@@ -279,7 +285,7 @@ class LogMessageVoidify {
     rtc::LogMessage(__FILE__, __LINE__, sev).stream()
 
 // The _F version prefixes the message with the current function name.
-#if (defined(__GNUC__) && defined(_DEBUG)) || defined(WANT_PRETTY_LOG_F)
+#if (defined(__GNUC__) && !defined(NDEBUG)) || defined(WANT_PRETTY_LOG_F)
 #define LOG_F(sev) LOG(sev) << __PRETTY_FUNCTION__ << ": "
 #define LOG_T_F(sev) LOG(sev) << this << ": " << __PRETTY_FUNCTION__ << ": "
 #else
@@ -291,6 +297,7 @@ class LogMessageVoidify {
   rtc::LogCheckLevel(rtc::sev)
 #define LOG_CHECK_LEVEL_V(sev) \
   rtc::LogCheckLevel(sev)
+
 inline bool LogCheckLevel(LoggingSeverity sev) {
   return (LogMessage::GetMinLogSeverity() <= sev);
 }
@@ -335,7 +342,11 @@ inline bool LogCheckLevel(LoggingSeverity sev) {
   LOG_ERRNO(sev)
 #define LAST_SYSTEM_ERROR \
   (errno)
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
+
+#define LOG_TAG(sev, tag) \
+  LOG_SEVERITY_PRECONDITION(sev) \
+    rtc::LogMessage(NULL, 0, sev, tag).stream()
 
 #define PLOG(sev, err) \
   LOG_ERR_EX(sev, err)
diff --git a/media/webrtc/trunk/webrtc/base/logging_unittest.cc b/media/webrtc/trunk/webrtc/base/logging_unittest.cc
index 7d7c97ec5e..6047361bf5 100644
--- a/media/webrtc/trunk/webrtc/base/logging_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/logging_unittest.cc
@@ -14,17 +14,33 @@
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
+template <typename Base>
+class LogSinkImpl
+    : public LogSink,
+      public Base {
+ public:
+  LogSinkImpl() {}
+
+  template<typename P>
+  explicit LogSinkImpl(P* p) : Base(p) {}
+
+ private:
+  void OnLogMessage(const std::string& message) override {
+    static_cast<Base*>(this)->WriteAll(
+        message.data(), message.size(), nullptr, nullptr);
+  }
+};
+
 // Test basic logging operation. We should get the INFO log but not the VERBOSE.
 // We should restore the correct global state at the end.
 TEST(LogTest, SingleStream) {
   int sev = LogMessage::GetLogToStream(NULL);
 
   std::string str;
-  StringStream stream(str);
+  LogSinkImpl<StringStream> stream(&str);
   LogMessage::AddLogToStream(&stream, LS_INFO);
   EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream));
 
@@ -34,7 +50,7 @@ TEST(LogTest, SingleStream) {
   EXPECT_EQ(std::string::npos, str.find("VERBOSE"));
 
   LogMessage::RemoveLogToStream(&stream);
-  EXPECT_EQ(LogMessage::NO_LOGGING, LogMessage::GetLogToStream(&stream));
+  EXPECT_EQ(LS_NONE, LogMessage::GetLogToStream(&stream));
 
   EXPECT_EQ(sev, LogMessage::GetLogToStream(NULL));
 }
@@ -46,7 +62,7 @@ TEST(LogTest, MultipleStreams) {
   int sev = LogMessage::GetLogToStream(NULL);
 
   std::string str1, str2;
-  StringStream stream1(str1), stream2(str2);
+  LogSinkImpl<StringStream> stream1(&str1), stream2(&str2);
   LogMessage::AddLogToStream(&stream1, LS_INFO);
   LogMessage::AddLogToStream(&stream2, LS_VERBOSE);
   EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream1));
@@ -62,8 +78,8 @@ TEST(LogTest, MultipleStreams) {
 
   LogMessage::RemoveLogToStream(&stream2);
   LogMessage::RemoveLogToStream(&stream1);
-  EXPECT_EQ(LogMessage::NO_LOGGING, LogMessage::GetLogToStream(&stream2));
-  EXPECT_EQ(LogMessage::NO_LOGGING, LogMessage::GetLogToStream(&stream1));
+  EXPECT_EQ(LS_NONE, LogMessage::GetLogToStream(&stream2));
+  EXPECT_EQ(LS_NONE, LogMessage::GetLogToStream(&stream1));
 
   EXPECT_EQ(sev, LogMessage::GetLogToStream(NULL));
 }
@@ -91,7 +107,7 @@ TEST(LogTest, MultipleThreads) {
   thread2.Start();
   thread3.Start();
 
-  NullStream stream1, stream2, stream3;
+  LogSinkImpl<NullStream> stream1, stream2, stream3;
   for (int i = 0; i < 1000; ++i) {
     LogMessage::AddLogToStream(&stream1, LS_INFO);
     LogMessage::AddLogToStream(&stream2, LS_VERBOSE);
@@ -106,7 +122,7 @@ TEST(LogTest, MultipleThreads) {
 
 
 TEST(LogTest, WallClockStartTime) {
-  uint32 time = LogMessage::WallClockStartTime();
+  uint32_t time = LogMessage::WallClockStartTime();
   // Expect the time to be in a sensible range, e.g. > 2012-01-01.
   EXPECT_GT(time, 1325376000u);
 }
@@ -117,12 +133,12 @@ TEST(LogTest, Perf) {
   EXPECT_TRUE(Filesystem::GetTemporaryFolder(path, true, NULL));
   path.SetPathname(Filesystem::TempFilename(path, "ut"));
 
-  FileStream stream;
+  LogSinkImpl<FileStream> stream;
   EXPECT_TRUE(stream.Open(path.pathname(), "wb", NULL));
   stream.DisableBuffering();
   LogMessage::AddLogToStream(&stream, LS_SENSITIVE);
 
-  uint32 start = Time(), finish;
+  uint32_t start = Time(), finish;
   std::string message('X', 80);
   for (int i = 0; i < 1000; ++i) {
     LOG(LS_SENSITIVE) << message;
diff --git a/media/webrtc/trunk/webrtc/base/logsinks.cc b/media/webrtc/trunk/webrtc/base/logsinks.cc
new file mode 100644
index 0000000000..5a6db45caa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/logsinks.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/logsinks.h"
+
+#include <iostream>
+#include <string>
+
+#include "webrtc/base/checks.h"
+
+namespace rtc {
+
+FileRotatingLogSink::FileRotatingLogSink(const std::string& log_dir_path,
+                                         const std::string& log_prefix,
+                                         size_t max_log_size,
+                                         size_t num_log_files)
+    : FileRotatingLogSink(new FileRotatingStream(log_dir_path,
+                                                 log_prefix,
+                                                 max_log_size,
+                                                 num_log_files)) {
+}
+
+FileRotatingLogSink::FileRotatingLogSink(FileRotatingStream* stream)
+    : stream_(stream) {
+  RTC_DCHECK(stream);
+}
+
+FileRotatingLogSink::~FileRotatingLogSink() {
+}
+
+void FileRotatingLogSink::OnLogMessage(const std::string& message) {
+  if (stream_->GetState() != SS_OPEN) {
+    std::cerr << "Init() must be called before adding this sink." << std::endl;
+    return;
+  }
+  stream_->WriteAll(message.c_str(), message.size(), nullptr, nullptr);
+}
+
+bool FileRotatingLogSink::Init() {
+  return stream_->Open();
+}
+
+bool FileRotatingLogSink::DisableBuffering() {
+  return stream_->DisableBuffering();
+}
+
+CallSessionFileRotatingLogSink::CallSessionFileRotatingLogSink(
+    const std::string& log_dir_path,
+    size_t max_total_log_size)
+    : FileRotatingLogSink(
+          new CallSessionFileRotatingStream(log_dir_path, max_total_log_size)) {
+}
+
+CallSessionFileRotatingLogSink::~CallSessionFileRotatingLogSink() {
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/logsinks.h b/media/webrtc/trunk/webrtc/base/logsinks.h
new file mode 100644
index 0000000000..eabf056398
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/logsinks.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FILE_ROTATING_LOG_SINK_H_
+#define WEBRTC_BASE_FILE_ROTATING_LOG_SINK_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/filerotatingstream.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace rtc {
+
+// Log sink that uses a FileRotatingStream to write to disk.
+// Init() must be called before adding this sink.
+class FileRotatingLogSink : public LogSink {
+ public:
+  // |num_log_files| must be greater than 1 and |max_log_size| must be greater
+  // than 0.
+  FileRotatingLogSink(const std::string& log_dir_path,
+                      const std::string& log_prefix,
+                      size_t max_log_size,
+                      size_t num_log_files);
+  ~FileRotatingLogSink() override;
+
+  // Writes the message to the current file. It will spill over to the next
+  // file if needed.
+  void OnLogMessage(const std::string& message) override;
+
+  // Deletes any existing files in the directory and creates a new log file.
+  virtual bool Init();
+
+  // Disables buffering on the underlying stream.
+  bool DisableBuffering();
+
+ protected:
+  explicit FileRotatingLogSink(FileRotatingStream* stream);
+
+ private:
+  scoped_ptr<FileRotatingStream> stream_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(FileRotatingLogSink);
+};
+
+// Log sink that uses a CallSessionFileRotatingStream to write to disk.
+// Init() must be called before adding this sink.
+class CallSessionFileRotatingLogSink : public FileRotatingLogSink {
+ public:
+  CallSessionFileRotatingLogSink(const std::string& log_dir_path,
+                                 size_t max_total_log_size);
+  ~CallSessionFileRotatingLogSink() override;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(CallSessionFileRotatingLogSink);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_FILE_ROTATING_LOG_SINK_H_
diff --git a/media/webrtc/trunk/webrtc/base/macasyncsocket.cc b/media/webrtc/trunk/webrtc/base/macasyncsocket.cc
index ee982ffff1..8f811ea8b6 100644
--- a/media/webrtc/trunk/webrtc/base/macasyncsocket.cc
+++ b/media/webrtc/trunk/webrtc/base/macasyncsocket.cc
@@ -112,7 +112,7 @@ int MacAsyncSocket::Connect(const SocketAddress& addr) {
     SetError(EALREADY);
     return SOCKET_ERROR;
   }
-  if (addr.IsUnresolved()) {
+  if (addr.IsUnresolvedIP()) {
     LOG(LS_VERBOSE) << "Resolving addr in MacAsyncSocket::Connect";
     resolver_ = new AsyncResolver();
     resolver_->SignalWorkDone.connect(this,
@@ -276,7 +276,7 @@ int MacAsyncSocket::Close() {
   return 0;
 }
 
-int MacAsyncSocket::EstimateMTU(uint16* mtu) {
+int MacAsyncSocket::EstimateMTU(uint16_t* mtu) {
   ASSERT(false && "NYI");
   return -1;
 }
diff --git a/media/webrtc/trunk/webrtc/base/macasyncsocket.h b/media/webrtc/trunk/webrtc/base/macasyncsocket.h
index 1aa4fe122a..5861ee3276 100644
--- a/media/webrtc/trunk/webrtc/base/macasyncsocket.h
+++ b/media/webrtc/trunk/webrtc/base/macasyncsocket.h
@@ -49,7 +49,7 @@ class MacAsyncSocket : public AsyncSocket, public sigslot::has_slots<> {
   int GetError() const override;
   void SetError(int error) override;
   ConnState GetState() const override;
-  int EstimateMTU(uint16* mtu) override;
+  int EstimateMTU(uint16_t* mtu) override;
   int GetOption(Option opt, int* value) override;
   int SetOption(Option opt, int value) override;
 
@@ -90,7 +90,7 @@ class MacAsyncSocket : public AsyncSocket, public sigslot::has_slots<> {
   ConnState state_;
   AsyncResolver* resolver_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(MacAsyncSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MacAsyncSocket);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/maccocoasocketserver.h b/media/webrtc/trunk/webrtc/base/maccocoasocketserver.h
index 0e171b4628..0acf8d757a 100644
--- a/media/webrtc/trunk/webrtc/base/maccocoasocketserver.h
+++ b/media/webrtc/trunk/webrtc/base/maccocoasocketserver.h
@@ -40,7 +40,7 @@ class MacCocoaSocketServer : public MacBaseSocketServer {
   // The count of how many times we're inside the NSApplication main loop.
   int run_count_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(MacCocoaSocketServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MacCocoaSocketServer);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/maccocoasocketserver_unittest.mm b/media/webrtc/trunk/webrtc/base/maccocoasocketserver_unittest.mm
index 932b4a14f5..5401ffb329 100644
--- a/media/webrtc/trunk/webrtc/base/maccocoasocketserver_unittest.mm
+++ b/media/webrtc/trunk/webrtc/base/maccocoasocketserver_unittest.mm
@@ -32,7 +32,7 @@ class WakeThread : public Thread {
 // Test that MacCocoaSocketServer::Wait works as expected.
 TEST(MacCocoaSocketServer, TestWait) {
   MacCocoaSocketServer server;
-  uint32 start = Time();
+  uint32_t start = Time();
   server.Wait(1000, true);
   EXPECT_GE(TimeSince(start), 1000);
 }
@@ -41,7 +41,7 @@ TEST(MacCocoaSocketServer, TestWait) {
 TEST(MacCocoaSocketServer, TestWakeup) {
   MacCFSocketServer server;
   WakeThread thread(&server);
-  uint32 start = Time();
+  uint32_t start = Time();
   thread.Start();
   server.Wait(10000, true);
   EXPECT_LT(TimeSince(start), 10000);
diff --git a/media/webrtc/trunk/webrtc/base/macconversion.cc b/media/webrtc/trunk/webrtc/base/macconversion.cc
index 75d11a803c..c1eec03de7 100644
--- a/media/webrtc/trunk/webrtc/base/macconversion.cc
+++ b/media/webrtc/trunk/webrtc/base/macconversion.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
 
 #include <CoreFoundation/CoreFoundation.h>
 
@@ -156,4 +156,4 @@ bool p_isCFNumberTrue(CFNumberRef cfn) {
   return result;
 }
 
-#endif  // WEBRTC_MAC && !defined(WEBRTC_IOS)
+#endif  // WEBRTC_MAC || WEBRTC_IOS
diff --git a/media/webrtc/trunk/webrtc/base/macconversion.h b/media/webrtc/trunk/webrtc/base/macconversion.h
index a96ed22985..63b27cf930 100644
--- a/media/webrtc/trunk/webrtc/base/macconversion.h
+++ b/media/webrtc/trunk/webrtc/base/macconversion.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_BASE_MACCONVERSION_H_
 #define WEBRTC_BASE_MACCONVERSION_H_
 
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
 
 #include <CoreFoundation/CoreFoundation.h>
 
@@ -34,6 +34,6 @@ bool p_convertCFNumberToInt(CFNumberRef cfn, int* i);
 // given a CFNumberRef, determine if it represents a true value.
 bool p_isCFNumberTrue(CFNumberRef cfn);
 
-#endif  // WEBRTC_MAC && !defined(WEBRTC_IOS)
+#endif  // WEBRTC_MAC || WEBRTC_IOS
 
 #endif  // WEBRTC_BASE_MACCONVERSION_H_
diff --git a/media/webrtc/trunk/webrtc/base/macifaddrs_converter.cc b/media/webrtc/trunk/webrtc/base/macifaddrs_converter.cc
new file mode 100644
index 0000000000..0916cb5ba2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/macifaddrs_converter.cc
@@ -0,0 +1,281 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <net/if.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/ifaddrs_converter.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+
+#if !defined(WEBRTC_IOS)
+#include <net/if_media.h>
+#include <netinet/in_var.h>
+#else  // WEBRTC_IOS
+#define SCOPE6_ID_MAX 16
+
+struct in6_addrlifetime {
+  time_t ia6t_expire;    /* valid lifetime expiration time */
+  time_t ia6t_preferred; /* preferred lifetime expiration time */
+  u_int32_t ia6t_vltime; /* valid lifetime */
+  u_int32_t ia6t_pltime; /* prefix lifetime */
+};
+
+struct in6_ifstat {
+  u_quad_t ifs6_in_receive;      /* # of total input datagram */
+  u_quad_t ifs6_in_hdrerr;       /* # of datagrams with invalid hdr */
+  u_quad_t ifs6_in_toobig;       /* # of datagrams exceeded MTU */
+  u_quad_t ifs6_in_noroute;      /* # of datagrams with no route */
+  u_quad_t ifs6_in_addrerr;      /* # of datagrams with invalid dst */
+  u_quad_t ifs6_in_protounknown; /* # of datagrams with unknown proto */
+                                 /* NOTE: increment on final dst if */
+  u_quad_t ifs6_in_truncated;    /* # of truncated datagrams */
+  u_quad_t ifs6_in_discard;      /* # of discarded datagrams */
+                                 /* NOTE: fragment timeout is not here */
+  u_quad_t ifs6_in_deliver;      /* # of datagrams delivered to ULP */
+                                 /* NOTE: increment on final dst if */
+  u_quad_t ifs6_out_forward;     /* # of datagrams forwarded */
+                                 /* NOTE: increment on outgoing if */
+  u_quad_t ifs6_out_request;     /* # of outgoing datagrams from ULP */
+                                 /* NOTE: does not include forwrads */
+  u_quad_t ifs6_out_discard;     /* # of discarded datagrams */
+  u_quad_t ifs6_out_fragok;      /* # of datagrams fragmented */
+  u_quad_t ifs6_out_fragfail;    /* # of datagrams failed on fragment */
+  u_quad_t ifs6_out_fragcreat;   /* # of fragment datagrams */
+                                 /* NOTE: this is # after fragment */
+  u_quad_t ifs6_reass_reqd;      /* # of incoming fragmented packets */
+                                 /* NOTE: increment on final dst if */
+  u_quad_t ifs6_reass_ok;        /* # of reassembled packets */
+                                 /* NOTE: this is # after reass */
+                                 /* NOTE: increment on final dst if */
+  u_quad_t ifs6_reass_fail;      /* # of reass failures */
+                                 /* NOTE: may not be packet count */
+                                 /* NOTE: increment on final dst if */
+  u_quad_t ifs6_in_mcast;        /* # of inbound multicast datagrams */
+  u_quad_t ifs6_out_mcast;       /* # of outbound multicast datagrams */
+};
+struct icmp6_ifstat {
+  /*
+   * Input statistics
+   */
+  /* ipv6IfIcmpInMsgs, total # of input messages */
+  u_quad_t ifs6_in_msg;
+  /* ipv6IfIcmpInErrors, # of input error messages */
+  u_quad_t ifs6_in_error;
+  /* ipv6IfIcmpInDestUnreachs, # of input dest unreach errors */
+  u_quad_t ifs6_in_dstunreach;
+  /* ipv6IfIcmpInAdminProhibs, # of input admin. prohibited errs */
+  u_quad_t ifs6_in_adminprohib;
+  /* ipv6IfIcmpInTimeExcds, # of input time exceeded errors */
+  u_quad_t ifs6_in_timeexceed;
+  /* ipv6IfIcmpInParmProblems, # of input parameter problem errors */
+  u_quad_t ifs6_in_paramprob;
+  /* ipv6IfIcmpInPktTooBigs, # of input packet too big errors */
+  u_quad_t ifs6_in_pkttoobig;
+  /* ipv6IfIcmpInEchos, # of input echo requests */
+  u_quad_t ifs6_in_echo;
+  /* ipv6IfIcmpInEchoReplies, # of input echo replies */
+  u_quad_t ifs6_in_echoreply;
+  /* ipv6IfIcmpInRouterSolicits, # of input router solicitations */
+  u_quad_t ifs6_in_routersolicit;
+  /* ipv6IfIcmpInRouterAdvertisements, # of input router advertisements */
+  u_quad_t ifs6_in_routeradvert;
+  /* ipv6IfIcmpInNeighborSolicits, # of input neighbor solicitations */
+  u_quad_t ifs6_in_neighborsolicit;
+  /* ipv6IfIcmpInNeighborAdvertisements, # of input neighbor advs. */
+  u_quad_t ifs6_in_neighboradvert;
+  /* ipv6IfIcmpInRedirects, # of input redirects */
+  u_quad_t ifs6_in_redirect;
+  /* ipv6IfIcmpInGroupMembQueries, # of input MLD queries */
+  u_quad_t ifs6_in_mldquery;
+  /* ipv6IfIcmpInGroupMembResponses, # of input MLD reports */
+  u_quad_t ifs6_in_mldreport;
+  /* ipv6IfIcmpInGroupMembReductions, # of input MLD done */
+  u_quad_t ifs6_in_mlddone;
+
+  /*
+   * Output statistics. We should solve unresolved routing problem...
+   */
+  /* ipv6IfIcmpOutMsgs, total # of output messages */
+  u_quad_t ifs6_out_msg;
+  /* ipv6IfIcmpOutErrors, # of output error messages */
+  u_quad_t ifs6_out_error;
+  /* ipv6IfIcmpOutDestUnreachs, # of output dest unreach errors */
+  u_quad_t ifs6_out_dstunreach;
+  /* ipv6IfIcmpOutAdminProhibs, # of output admin. prohibited errs */
+  u_quad_t ifs6_out_adminprohib;
+  /* ipv6IfIcmpOutTimeExcds, # of output time exceeded errors */
+  u_quad_t ifs6_out_timeexceed;
+  /* ipv6IfIcmpOutParmProblems, # of output parameter problem errors */
+  u_quad_t ifs6_out_paramprob;
+  /* ipv6IfIcmpOutPktTooBigs, # of output packet too big errors */
+  u_quad_t ifs6_out_pkttoobig;
+  /* ipv6IfIcmpOutEchos, # of output echo requests */
+  u_quad_t ifs6_out_echo;
+  /* ipv6IfIcmpOutEchoReplies, # of output echo replies */
+  u_quad_t ifs6_out_echoreply;
+  /* ipv6IfIcmpOutRouterSolicits, # of output router solicitations */
+  u_quad_t ifs6_out_routersolicit;
+  /* ipv6IfIcmpOutRouterAdvertisements, # of output router advs. */
+  u_quad_t ifs6_out_routeradvert;
+  /* ipv6IfIcmpOutNeighborSolicits, # of output neighbor solicitations */
+  u_quad_t ifs6_out_neighborsolicit;
+  /* ipv6IfIcmpOutNeighborAdvertisements, # of output neighbor advs. */
+  u_quad_t ifs6_out_neighboradvert;
+  /* ipv6IfIcmpOutRedirects, # of output redirects */
+  u_quad_t ifs6_out_redirect;
+  /* ipv6IfIcmpOutGroupMembQueries, # of output MLD queries */
+  u_quad_t ifs6_out_mldquery;
+  /* ipv6IfIcmpOutGroupMembResponses, # of output MLD reports */
+  u_quad_t ifs6_out_mldreport;
+  /* ipv6IfIcmpOutGroupMembReductions, # of output MLD done */
+  u_quad_t ifs6_out_mlddone;
+};
+
+struct in6_ifreq {
+  char ifr_name[IFNAMSIZ];
+  union {
+    struct sockaddr_in6 ifru_addr;
+    struct sockaddr_in6 ifru_dstaddr;
+    int ifru_flags;
+    int ifru_flags6;
+    int ifru_metric;
+    int ifru_intval;
+    caddr_t ifru_data;
+    struct in6_addrlifetime ifru_lifetime;
+    struct in6_ifstat ifru_stat;
+    struct icmp6_ifstat ifru_icmp6stat;
+    u_int32_t ifru_scope_id[SCOPE6_ID_MAX];
+  } ifr_ifru;
+};
+
+#define SIOCGIFAFLAG_IN6 _IOWR('i', 73, struct in6_ifreq)
+
+#define IN6_IFF_ANYCAST 0x0001    /* anycast address */
+#define IN6_IFF_TENTATIVE 0x0002  /* tentative address */
+#define IN6_IFF_DUPLICATED 0x0004 /* DAD detected duplicate */
+#define IN6_IFF_DETACHED 0x0008   /* may be detached from the link */
+#define IN6_IFF_DEPRECATED 0x0010 /* deprecated address */
+#define IN6_IFF_TEMPORARY 0x0080  /* temporary (anonymous) address. */
+
+#endif  // WEBRTC_IOS
+
+namespace rtc {
+
+namespace {
+
+class IPv6AttributesGetter {
+ public:
+  IPv6AttributesGetter();
+  virtual ~IPv6AttributesGetter();
+  bool IsInitialized() const;
+  bool GetIPAttributes(const char* ifname,
+                       const sockaddr* sock_addr,
+                       int* native_attributes);
+
+ private:
+  // on MAC or IOS, we have to use ioctl with a socket to query an IPv6
+  // interface's attribute.
+  int ioctl_socket_;
+};
+
+IPv6AttributesGetter::IPv6AttributesGetter()
+    : ioctl_socket_(
+          socket(AF_INET6, SOCK_DGRAM, 0 /* unspecified protocol */)) {
+  RTC_DCHECK_GE(ioctl_socket_, 0);
+}
+
+bool IPv6AttributesGetter::IsInitialized() const {
+  return ioctl_socket_ >= 0;
+}
+
+IPv6AttributesGetter::~IPv6AttributesGetter() {
+  if (!IsInitialized()) {
+    return;
+  }
+  close(ioctl_socket_);
+}
+
+bool IPv6AttributesGetter::GetIPAttributes(const char* ifname,
+                                           const sockaddr* sock_addr,
+                                           int* native_attributes) {
+  if (!IsInitialized()) {
+    return false;
+  }
+
+  struct in6_ifreq ifr = {};
+  strncpy(ifr.ifr_name, ifname, sizeof(ifr.ifr_name) - 1);
+  memcpy(&ifr.ifr_ifru.ifru_addr, sock_addr, sock_addr->sa_len);
+  int rv = ioctl(ioctl_socket_, SIOCGIFAFLAG_IN6, &ifr);
+  if (rv >= 0) {
+    *native_attributes = ifr.ifr_ifru.ifru_flags;
+  } else {
+    LOG(LS_ERROR) << "ioctl returns " << errno;
+  }
+  return (rv >= 0);
+}
+
+// Converts native IPv6 address attributes to net IPv6 address attributes.  If
+// it returns false, the IP address isn't suitable for one-to-one communications
+// applications and should be ignored.
+bool ConvertNativeToIPAttributes(int native_attributes, int* net_attributes) {
+  // For MacOSX, we disallow addresses with attributes IN6_IFF_ANYCASE,
+  // IN6_IFF_DUPLICATED, IN6_IFF_TENTATIVE, and IN6_IFF_DETACHED as these are
+  // still progressing through duplicated address detection (DAD) or are not
+  // suitable for one-to-one communication applications.
+  if (native_attributes & (IN6_IFF_ANYCAST | IN6_IFF_DUPLICATED |
+                           IN6_IFF_TENTATIVE | IN6_IFF_DETACHED)) {
+    return false;
+  }
+
+  if (native_attributes & IN6_IFF_TEMPORARY) {
+    *net_attributes |= IPV6_ADDRESS_FLAG_TEMPORARY;
+  }
+
+  if (native_attributes & IN6_IFF_DEPRECATED) {
+    *net_attributes |= IPV6_ADDRESS_FLAG_DEPRECATED;
+  }
+
+  return true;
+}
+
+class MacIfAddrsConverter : public IfAddrsConverter {
+ public:
+  MacIfAddrsConverter() : ip_attribute_getter_(new IPv6AttributesGetter()) {}
+  ~MacIfAddrsConverter() override {}
+
+  bool ConvertNativeAttributesToIPAttributes(const struct ifaddrs* interface,
+                                             int* ip_attributes) override {
+    int native_attributes;
+    if (!ip_attribute_getter_->GetIPAttributes(
+            interface->ifa_name, interface->ifa_addr, &native_attributes)) {
+      return false;
+    }
+
+    if (!ConvertNativeToIPAttributes(native_attributes, ip_attributes)) {
+      return false;
+    }
+
+    return true;
+  }
+
+ private:
+  rtc::scoped_ptr<IPv6AttributesGetter> ip_attribute_getter_;
+};
+
+}  // namespace
+
+IfAddrsConverter* CreateIfAddrsConverter() {
+  return new MacIfAddrsConverter();
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/macsocketserver_unittest.cc b/media/webrtc/trunk/webrtc/base/macsocketserver_unittest.cc
index e98be918ca..ecb9a706b7 100644
--- a/media/webrtc/trunk/webrtc/base/macsocketserver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/macsocketserver_unittest.cc
@@ -35,7 +35,7 @@ class WakeThread : public Thread {
 // Test that MacCFSocketServer::Wait works as expected.
 TEST(MacCFSocketServerTest, TestWait) {
   MacCFSocketServer server;
-  uint32 start = Time();
+  uint32_t start = Time();
   server.Wait(1000, true);
   EXPECT_GE(TimeSince(start), 1000);
 }
@@ -44,7 +44,7 @@ TEST(MacCFSocketServerTest, TestWait) {
 TEST(MacCFSocketServerTest, TestWakeup) {
   MacCFSocketServer server;
   WakeThread thread(&server);
-  uint32 start = Time();
+  uint32_t start = Time();
   thread.Start();
   server.Wait(10000, true);
   EXPECT_LT(TimeSince(start), 10000);
@@ -53,7 +53,7 @@ TEST(MacCFSocketServerTest, TestWakeup) {
 // Test that MacCarbonSocketServer::Wait works as expected.
 TEST(MacCarbonSocketServerTest, TestWait) {
   MacCarbonSocketServer server;
-  uint32 start = Time();
+  uint32_t start = Time();
   server.Wait(1000, true);
   EXPECT_GE(TimeSince(start), 1000);
 }
@@ -62,7 +62,7 @@ TEST(MacCarbonSocketServerTest, TestWait) {
 TEST(MacCarbonSocketServerTest, TestWakeup) {
   MacCarbonSocketServer server;
   WakeThread thread(&server);
-  uint32 start = Time();
+  uint32_t start = Time();
   thread.Start();
   server.Wait(10000, true);
   EXPECT_LT(TimeSince(start), 10000);
@@ -71,7 +71,7 @@ TEST(MacCarbonSocketServerTest, TestWakeup) {
 // Test that MacCarbonAppSocketServer::Wait works as expected.
 TEST(MacCarbonAppSocketServerTest, TestWait) {
   MacCarbonAppSocketServer server;
-  uint32 start = Time();
+  uint32_t start = Time();
   server.Wait(1000, true);
   EXPECT_GE(TimeSince(start), 1000);
 }
@@ -80,7 +80,7 @@ TEST(MacCarbonAppSocketServerTest, TestWait) {
 TEST(MacCarbonAppSocketServerTest, TestWakeup) {
   MacCarbonAppSocketServer server;
   WakeThread thread(&server);
-  uint32 start = Time();
+  uint32_t start = Time();
   thread.Start();
   server.Wait(10000, true);
   EXPECT_LT(TimeSince(start), 10000);
@@ -123,7 +123,8 @@ TEST_F(MacAsyncSocketTest, DISABLED_TestConnectFailIPv4) {
   SocketTest::TestConnectFailIPv4();
 }
 
-TEST_F(MacAsyncSocketTest, TestConnectFailIPv6) {
+// Flaky. See webrtc:4738.
+TEST_F(MacAsyncSocketTest, DISABLED_TestConnectFailIPv6) {
   SocketTest::TestConnectFailIPv6();
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/macutils.cc b/media/webrtc/trunk/webrtc/base/macutils.cc
index 3d9a4cfd7d..becc330676 100644
--- a/media/webrtc/trunk/webrtc/base/macutils.cc
+++ b/media/webrtc/trunk/webrtc/base/macutils.cc
@@ -200,10 +200,10 @@ bool RunAppleScript(const std::string& script) {
     AECreateDesc(typeNull, NULL, 0, &result_data);
     OSAScriptError(component, kOSAErrorMessage, typeChar, &result_data);
     int len = AEGetDescDataSize(&result_data);
-    char* data = (char*) malloc(len);
+    char* data = (char*)malloc(len);
     if (data != NULL) {
       err = AEGetDescData(&result_data, data, len);
-      LOG(LS_ERROR) << "Script error: " << data;
+      LOG(LS_ERROR) << "Script error: " << std::string(data, len);
     }
     AEDisposeDesc(&script_desc);
     AEDisposeDesc(&result_data);
diff --git a/media/webrtc/trunk/webrtc/base/md5.cc b/media/webrtc/trunk/webrtc/base/md5.cc
index 54128907ad..fda6ddd238 100644
--- a/media/webrtc/trunk/webrtc/base/md5.cc
+++ b/media/webrtc/trunk/webrtc/base/md5.cc
@@ -23,14 +23,14 @@
 // TODO: Avoid memcmpy - hash directly from memory.
 #include <string.h>  // for memcpy().
 
-#include "webrtc/base/byteorder.h"  // for ARCH_CPU_LITTLE_ENDIAN.
+#include "webrtc/base/byteorder.h"  // for RTC_ARCH_CPU_LITTLE_ENDIAN.
 
 namespace rtc {
 
-#ifdef ARCH_CPU_LITTLE_ENDIAN
+#ifdef RTC_ARCH_CPU_LITTLE_ENDIAN
 #define ByteReverse(buf, len)  // Nothing.
-#else  // ARCH_CPU_BIG_ENDIAN
-static void ByteReverse(uint32* buf, int len) {
+#else  // RTC_ARCH_CPU_BIG_ENDIAN
+static void ByteReverse(uint32_t* buf, int len) {
   for (int i = 0; i < len; ++i) {
     buf[i] = rtc::GetLE32(&buf[i]);
   }
@@ -49,18 +49,18 @@ void MD5Init(MD5Context* ctx) {
 }
 
 // Update context to reflect the concatenation of another buffer full of bytes.
-void MD5Update(MD5Context* ctx, const uint8* buf, size_t len) {
+void MD5Update(MD5Context* ctx, const uint8_t* buf, size_t len) {
   // Update bitcount.
-  uint32 t = ctx->bits[0];
-  if ((ctx->bits[0] = t + (static_cast<uint32>(len) << 3)) < t) {
+  uint32_t t = ctx->bits[0];
+  if ((ctx->bits[0] = t + (static_cast<uint32_t>(len) << 3)) < t) {
     ctx->bits[1]++;  // Carry from low to high.
   }
-  ctx->bits[1] += static_cast<uint32>(len >> 29);
+  ctx->bits[1] += static_cast<uint32_t>(len >> 29);
   t = (t >> 3) & 0x3f;  // Bytes already in shsInfo->data.
 
   // Handle any leading odd-sized chunks.
   if (t) {
-    uint8* p = reinterpret_cast<uint8*>(ctx->in) + t;
+    uint8_t* p = reinterpret_cast<uint8_t*>(ctx->in) + t;
 
     t = 64-t;
     if (len < t) {
@@ -89,13 +89,13 @@ void MD5Update(MD5Context* ctx, const uint8* buf, size_t len) {
 
 // Final wrapup - pad to 64-byte boundary with the bit pattern.
 // 1 0* (64-bit count of bits processed, MSB-first)
-void MD5Final(MD5Context* ctx, uint8 digest[16]) {
+void MD5Final(MD5Context* ctx, uint8_t digest[16]) {
   // Compute number of bytes mod 64.
-  uint32 count = (ctx->bits[0] >> 3) & 0x3F;
+  uint32_t count = (ctx->bits[0] >> 3) & 0x3F;
 
   // Set the first char of padding to 0x80.  This is safe since there is
   // always at least one byte free.
-  uint8* p = reinterpret_cast<uint8*>(ctx->in) + count;
+  uint8_t* p = reinterpret_cast<uint8_t*>(ctx->in) + count;
   *p++ = 0x80;
 
   // Bytes of padding needed to make 64 bytes.
@@ -140,11 +140,11 @@ void MD5Final(MD5Context* ctx, uint8 digest[16]) {
 // The core of the MD5 algorithm, this alters an existing MD5 hash to
 // reflect the addition of 16 longwords of new data.  MD5Update blocks
 // the data and converts bytes into longwords for this routine.
-void MD5Transform(uint32 buf[4], const uint32 in[16]) {
-  uint32 a = buf[0];
-  uint32 b = buf[1];
-  uint32 c = buf[2];
-  uint32 d = buf[3];
+void MD5Transform(uint32_t buf[4], const uint32_t in[16]) {
+  uint32_t a = buf[0];
+  uint32_t b = buf[1];
+  uint32_t c = buf[2];
+  uint32_t d = buf[3];
 
   MD5STEP(F1, a, b, c, d, in[ 0] + 0xd76aa478, 7);
   MD5STEP(F1, d, a, b, c, in[ 1] + 0xe8c7b756, 12);
diff --git a/media/webrtc/trunk/webrtc/base/md5.h b/media/webrtc/trunk/webrtc/base/md5.h
index 80294bb483..45e00b73d1 100644
--- a/media/webrtc/trunk/webrtc/base/md5.h
+++ b/media/webrtc/trunk/webrtc/base/md5.h
@@ -18,24 +18,26 @@
 // Changes(fbarchard): Ported to C++ and Google style guide.
 // Made context first parameter in MD5Final for consistency with Sha1.
 // Changes(hellner): added rtc namespace
+// Changes(pbos): Reverted types back to uint32(8)_t with _t suffix.
 
 #ifndef WEBRTC_BASE_MD5_H_
 #define WEBRTC_BASE_MD5_H_
 
-#include "webrtc/base/basictypes.h"
+#include <stdint.h>
+#include <stdlib.h>
 
 namespace rtc {
 
 struct MD5Context {
-  uint32 buf[4];
-  uint32 bits[2];
-  uint32 in[16];
+  uint32_t buf[4];
+  uint32_t bits[2];
+  uint32_t in[16];
 };
 
 void MD5Init(MD5Context* context);
-void MD5Update(MD5Context* context, const uint8* data, size_t len);
-void MD5Final(MD5Context* context, uint8 digest[16]);
-void MD5Transform(uint32 buf[4], const uint32 in[16]);
+void MD5Update(MD5Context* context, const uint8_t* data, size_t len);
+void MD5Final(MD5Context* context, uint8_t digest[16]);
+void MD5Transform(uint32_t buf[4], const uint32_t in[16]);
 
 }  // namespace rtc
 
diff --git a/media/webrtc/trunk/webrtc/base/md5digest.cc b/media/webrtc/trunk/webrtc/base/md5digest.cc
index 1d014c3588..74f6bede29 100644
--- a/media/webrtc/trunk/webrtc/base/md5digest.cc
+++ b/media/webrtc/trunk/webrtc/base/md5digest.cc
@@ -17,14 +17,14 @@ size_t Md5Digest::Size() const {
 }
 
 void Md5Digest::Update(const void* buf, size_t len) {
-  MD5Update(&ctx_, static_cast<const uint8*>(buf), len);
+  MD5Update(&ctx_, static_cast<const uint8_t*>(buf), len);
 }
 
 size_t Md5Digest::Finish(void* buf, size_t len) {
   if (len < kSize) {
     return 0;
   }
-  MD5Final(&ctx_, static_cast<uint8*>(buf));
+  MD5Final(&ctx_, static_cast<uint8_t*>(buf));
   MD5Init(&ctx_);  // Reset for next use.
   return kSize;
 }
diff --git a/media/webrtc/trunk/webrtc/base/messagedigest.cc b/media/webrtc/trunk/webrtc/base/messagedigest.cc
index 8af60d9a96..0c2b4a16ac 100644
--- a/media/webrtc/trunk/webrtc/base/messagedigest.cc
+++ b/media/webrtc/trunk/webrtc/base/messagedigest.cc
@@ -117,7 +117,7 @@ size_t ComputeHmac(MessageDigest* digest,
   }
   // Copy the key to a block-sized buffer to simplify padding.
   // If the key is longer than a block, hash it and use the result instead.
-  scoped_ptr<uint8[]> new_key(new uint8[block_len]);
+  scoped_ptr<uint8_t[]> new_key(new uint8_t[block_len]);
   if (key_len > block_len) {
     ComputeDigest(digest, key, key_len, new_key.get(), block_len);
     memset(new_key.get() + digest->Size(), 0, block_len - digest->Size());
@@ -126,13 +126,14 @@ size_t ComputeHmac(MessageDigest* digest,
     memset(new_key.get() + key_len, 0, block_len - key_len);
   }
   // Set up the padding from the key, salting appropriately for each padding.
-  scoped_ptr<uint8[]> o_pad(new uint8[block_len]), i_pad(new uint8[block_len]);
+  scoped_ptr<uint8_t[]> o_pad(new uint8_t[block_len]);
+  scoped_ptr<uint8_t[]> i_pad(new uint8_t[block_len]);
   for (size_t i = 0; i < block_len; ++i) {
     o_pad[i] = 0x5c ^ new_key[i];
     i_pad[i] = 0x36 ^ new_key[i];
   }
   // Inner hash; hash the inner padding, and then the input buffer.
-  scoped_ptr<uint8[]> inner(new uint8[digest->Size()]);
+  scoped_ptr<uint8_t[]> inner(new uint8_t[digest->Size()]);
   digest->Update(i_pad.get(), block_len);
   digest->Update(input, in_len);
   digest->Finish(inner.get(), digest->Size());
diff --git a/media/webrtc/trunk/webrtc/base/messagehandler.h b/media/webrtc/trunk/webrtc/base/messagehandler.h
index 123c850973..b55b229a6d 100644
--- a/media/webrtc/trunk/webrtc/base/messagehandler.h
+++ b/media/webrtc/trunk/webrtc/base/messagehandler.h
@@ -11,7 +11,10 @@
 #ifndef WEBRTC_BASE_MESSAGEHANDLER_H_
 #define WEBRTC_BASE_MESSAGEHANDLER_H_
 
+#include <utility>
+
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
@@ -28,7 +31,7 @@ class MessageHandler {
   MessageHandler() {}
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(MessageHandler);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MessageHandler);
 };
 
 // Helper class to facilitate executing a functor on a thread.
@@ -47,6 +50,20 @@ class FunctorMessageHandler : public MessageHandler {
   ReturnT result_;
 };
 
+// Specialization for rtc::scoped_ptr<ReturnT>.
+template <class ReturnT, class FunctorT>
+class FunctorMessageHandler<class rtc::scoped_ptr<ReturnT>, FunctorT>
+    : public MessageHandler {
+ public:
+  explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
+  virtual void OnMessage(Message* msg) { result_ = std::move(functor_()); }
+  rtc::scoped_ptr<ReturnT> result() { return std::move(result_); }
+
+ private:
+  FunctorT functor_;
+  rtc::scoped_ptr<ReturnT> result_;
+};
+
 // Specialization for ReturnT of void.
 template <class FunctorT>
 class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
@@ -62,7 +79,6 @@ class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
   FunctorT functor_;
 };
 
-
 } // namespace rtc
 
 #endif // WEBRTC_BASE_MESSAGEHANDLER_H_
diff --git a/media/webrtc/trunk/webrtc/base/messagequeue.cc b/media/webrtc/trunk/webrtc/base/messagequeue.cc
index 53e451f632..857cf12927 100644
--- a/media/webrtc/trunk/webrtc/base/messagequeue.cc
+++ b/media/webrtc/trunk/webrtc/base/messagequeue.cc
@@ -27,7 +27,7 @@ typedef rtc::PhysicalSocketServer DefaultSocketServer;
 
 namespace rtc {
 
-const uint32 kMaxMsgLatency = 150;  // 150 ms
+const uint32_t kMaxMsgLatency = 150;  // 150 ms
 
 //------------------------------------------------------------------
 // MessageQueueManager
@@ -59,7 +59,7 @@ void MessageQueueManager::AddInternal(MessageQueue *message_queue) {
   // MessageQueueManager methods should be non-reentrant, so we
   // ASSERT that is the case.  If any of these ASSERT, please
   // contact bpm or jbeda.
-#if CS_TRACK_OWNER  // CurrentThreadIsOwner returns true by default.
+#if CS_DEBUG_CHECKS  // CurrentThreadIsOwner returns true by default.
   ASSERT(!crit_.CurrentThreadIsOwner());
 #endif
   CritScope cs(&crit_);
@@ -73,7 +73,7 @@ void MessageQueueManager::Remove(MessageQueue *message_queue) {
   return Instance()->RemoveInternal(message_queue);
 }
 void MessageQueueManager::RemoveInternal(MessageQueue *message_queue) {
-#if CS_TRACK_OWNER  // CurrentThreadIsOwner returns true by default.
+#if CS_DEBUG_CHECKS  // CurrentThreadIsOwner returns true by default.
   ASSERT(!crit_.CurrentThreadIsOwner());  // See note above.
 #endif
   // If this is the last MessageQueue, destroy the manager as well so that
@@ -104,7 +104,7 @@ void MessageQueueManager::Clear(MessageHandler *handler) {
   return Instance()->ClearInternal(handler);
 }
 void MessageQueueManager::ClearInternal(MessageHandler *handler) {
-#if CS_TRACK_OWNER  // CurrentThreadIsOwner returns true by default.
+#if CS_DEBUG_CHECKS  // CurrentThreadIsOwner returns true by default.
   ASSERT(!crit_.CurrentThreadIsOwner());  // See note above.
 #endif
   CritScope cs(&crit_);
@@ -188,8 +188,8 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
 
   int cmsTotal = cmsWait;
   int cmsElapsed = 0;
-  uint32 msStart = Time();
-  uint32 msCurrent = msStart;
+  uint32_t msStart = Time();
+  uint32_t msCurrent = msStart;
   while (true) {
     // Check for sent messages
     ReceiveSends();
@@ -227,7 +227,7 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
 
       // Log a warning for time-sensitive messages that we're late to deliver.
       if (pmsg->ts_sensitive) {
-        int32 delay = TimeDiff(msCurrent, pmsg->ts_sensitive);
+        int32_t delay = TimeDiff(msCurrent, pmsg->ts_sensitive);
         if (delay > 0) {
           LOG_F(LS_WARNING) << "id: " << pmsg->message_id << "  delay: "
                             << (delay + kMaxMsgLatency) << "ms";
@@ -276,8 +276,10 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
 void MessageQueue::ReceiveSends() {
 }
 
-void MessageQueue::Post(MessageHandler *phandler, uint32 id,
-    MessageData *pdata, bool time_sensitive) {
+void MessageQueue::Post(MessageHandler* phandler,
+                        uint32_t id,
+                        MessageData* pdata,
+                        bool time_sensitive) {
   if (fStop_)
     return;
 
@@ -299,20 +301,23 @@ void MessageQueue::Post(MessageHandler *phandler, uint32 id,
 
 void MessageQueue::PostDelayed(int cmsDelay,
                                MessageHandler* phandler,
-                               uint32 id,
+                               uint32_t id,
                                MessageData* pdata) {
   return DoDelayPost(cmsDelay, TimeAfter(cmsDelay), phandler, id, pdata);
 }
 
-void MessageQueue::PostAt(uint32 tstamp,
+void MessageQueue::PostAt(uint32_t tstamp,
                           MessageHandler* phandler,
-                          uint32 id,
+                          uint32_t id,
                           MessageData* pdata) {
   return DoDelayPost(TimeUntil(tstamp), tstamp, phandler, id, pdata);
 }
 
-void MessageQueue::DoDelayPost(int cmsDelay, uint32 tstamp,
-    MessageHandler *phandler, uint32 id, MessageData* pdata) {
+void MessageQueue::DoDelayPost(int cmsDelay,
+                               uint32_t tstamp,
+                               MessageHandler* phandler,
+                               uint32_t id,
+                               MessageData* pdata) {
   if (fStop_)
     return;
 
@@ -350,7 +355,8 @@ int MessageQueue::GetDelay() {
   return kForever;
 }
 
-void MessageQueue::Clear(MessageHandler *phandler, uint32 id,
+void MessageQueue::Clear(MessageHandler* phandler,
+                         uint32_t id,
                          MessageList* removed) {
   CritScope cs(&crit_);
 
diff --git a/media/webrtc/trunk/webrtc/base/messagequeue.h b/media/webrtc/trunk/webrtc/base/messagequeue.h
index e0cab8f5be..c3ab3b6669 100644
--- a/media/webrtc/trunk/webrtc/base/messagequeue.h
+++ b/media/webrtc/trunk/webrtc/base/messagequeue.h
@@ -123,8 +123,8 @@ class DisposeData : public MessageData {
   T* data_;
 };
 
-const uint32 MQID_ANY = static_cast<uint32>(-1);
-const uint32 MQID_DISPOSE = static_cast<uint32>(-2);
+const uint32_t MQID_ANY = static_cast<uint32_t>(-1);
+const uint32_t MQID_DISPOSE = static_cast<uint32_t>(-2);
 
 // No destructor
 
@@ -132,14 +132,14 @@ struct Message {
   Message() {
     memset(this, 0, sizeof(*this));
   }
-  inline bool Match(MessageHandler* handler, uint32 id) const {
+  inline bool Match(MessageHandler* handler, uint32_t id) const {
     return (handler == NULL || handler == phandler)
            && (id == MQID_ANY || id == message_id);
   }
   MessageHandler *phandler;
-  uint32 message_id;
+  uint32_t message_id;
   MessageData *pdata;
-  uint32 ts_sensitive;
+  uint32_t ts_sensitive;
 };
 
 typedef std::list<Message> MessageList;
@@ -149,8 +149,8 @@ typedef std::list<Message> MessageList;
 
 class DelayedMessage {
  public:
-  DelayedMessage(int delay, uint32 trigger, uint32 num, const Message& msg)
-  : cmsDelay_(delay), msTrigger_(trigger), num_(num), msg_(msg) { }
+  DelayedMessage(int delay, uint32_t trigger, uint32_t num, const Message& msg)
+      : cmsDelay_(delay), msTrigger_(trigger), num_(num), msg_(msg) {}
 
   bool operator< (const DelayedMessage& dmsg) const {
     return (dmsg.msTrigger_ < msTrigger_)
@@ -158,8 +158,8 @@ class DelayedMessage {
   }
 
   int cmsDelay_;  // for debugging
-  uint32 msTrigger_;
-  uint32 num_;
+  uint32_t msTrigger_;
+  uint32_t num_;
   Message msg_;
 };
 
@@ -190,17 +190,20 @@ class MessageQueue {
   virtual bool Get(Message *pmsg, int cmsWait = kForever,
                    bool process_io = true);
   virtual bool Peek(Message *pmsg, int cmsWait = 0);
-  virtual void Post(MessageHandler *phandler, uint32 id = 0,
-                    MessageData *pdata = NULL, bool time_sensitive = false);
+  virtual void Post(MessageHandler* phandler,
+                    uint32_t id = 0,
+                    MessageData* pdata = NULL,
+                    bool time_sensitive = false);
   virtual void PostDelayed(int cmsDelay,
                            MessageHandler* phandler,
-                           uint32 id = 0,
+                           uint32_t id = 0,
                            MessageData* pdata = NULL);
-  virtual void PostAt(uint32 tstamp,
+  virtual void PostAt(uint32_t tstamp,
                       MessageHandler* phandler,
-                      uint32 id = 0,
+                      uint32_t id = 0,
                       MessageData* pdata = NULL);
-  virtual void Clear(MessageHandler *phandler, uint32 id = MQID_ANY,
+  virtual void Clear(MessageHandler* phandler,
+                     uint32_t id = MQID_ANY,
                      MessageList* removed = NULL);
   virtual void Dispatch(Message *pmsg);
   virtual void ReceiveSends();
@@ -232,8 +235,11 @@ class MessageQueue {
     void reheap() { make_heap(c.begin(), c.end(), comp); }
   };
 
-  void DoDelayPost(int cmsDelay, uint32 tstamp, MessageHandler *phandler,
-                   uint32 id, MessageData* pdata);
+  void DoDelayPost(int cmsDelay,
+                   uint32_t tstamp,
+                   MessageHandler* phandler,
+                   uint32_t id,
+                   MessageData* pdata);
 
   // The SocketServer is not owned by MessageQueue.
   SocketServer* ss_;
@@ -244,11 +250,11 @@ class MessageQueue {
   Message msgPeek_;
   MessageList msgq_;
   PriorityQueue dmsgq_;
-  uint32 dmsgq_next_num_;
+  uint32_t dmsgq_next_num_;
   mutable CriticalSection crit_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(MessageQueue);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MessageQueue);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/messagequeue_unittest.cc b/media/webrtc/trunk/webrtc/base/messagequeue_unittest.cc
index 871542df28..78024e0b2d 100644
--- a/media/webrtc/trunk/webrtc/base/messagequeue_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/messagequeue_unittest.cc
@@ -16,7 +16,6 @@
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/nullsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using namespace rtc;
 
diff --git a/media/webrtc/trunk/webrtc/base/move.h b/media/webrtc/trunk/webrtc/base/move.h
deleted file mode 100644
index 198badf891..0000000000
--- a/media/webrtc/trunk/webrtc/base/move.h
+++ /dev/null
@@ -1,237 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Borrowed from Chromium's src/base/move.h.
-
-#ifndef WEBRTC_BASE_MOVE_H_
-#define WEBRTC_BASE_MOVE_H_
-
-#include "webrtc/typedefs.h"
-
-// Macro with the boilerplate that makes a type move-only in C++03.
-//
-// USAGE
-//
-// This macro should be used instead of DISALLOW_COPY_AND_ASSIGN to create
-// a "move-only" type.  Unlike DISALLOW_COPY_AND_ASSIGN, this macro should be
-// the first line in a class declaration.
-//
-// A class using this macro must call .Pass() (or somehow be an r-value already)
-// before it can be:
-//
-//   * Passed as a function argument
-//   * Used as the right-hand side of an assignment
-//   * Returned from a function
-//
-// Each class will still need to define their own "move constructor" and "move
-// operator=" to make this useful.  Here's an example of the macro, the move
-// constructor, and the move operator= from the scoped_ptr class:
-//
-//  template <typename T>
-//  class scoped_ptr {
-//     RTC_MOVE_ONLY_TYPE_FOR_CPP_03(scoped_ptr, RValue)
-//   public:
-//    scoped_ptr(RValue& other) : ptr_(other.release()) { }
-//    scoped_ptr& operator=(RValue& other) {
-//      swap(other);
-//      return *this;
-//    }
-//  };
-//
-// Note that the constructor must NOT be marked explicit.
-//
-// For consistency, the second parameter to the macro should always be RValue
-// unless you have a strong reason to do otherwise.  It is only exposed as a
-// macro parameter so that the move constructor and move operator= don't look
-// like they're using a phantom type.
-//
-//
-// HOW THIS WORKS
-//
-// For a thorough explanation of this technique, see:
-//
-//   http://en.wikibooks.org/wiki/More_C%2B%2B_Idioms/Move_Constructor
-//
-// The summary is that we take advantage of 2 properties:
-//
-//   1) non-const references will not bind to r-values.
-//   2) C++ can apply one user-defined conversion when initializing a
-//      variable.
-//
-// The first lets us disable the copy constructor and assignment operator
-// by declaring private version of them with a non-const reference parameter.
-//
-// For l-values, direct initialization still fails like in
-// DISALLOW_COPY_AND_ASSIGN because the copy constructor and assignment
-// operators are private.
-//
-// For r-values, the situation is different. The copy constructor and
-// assignment operator are not viable due to (1), so we are trying to call
-// a non-existent constructor and non-existing operator= rather than a private
-// one.  Since we have not committed an error quite yet, we can provide an
-// alternate conversion sequence and a constructor.  We add
-//
-//   * a private struct named "RValue"
-//   * a user-defined conversion "operator RValue()"
-//   * a "move constructor" and "move operator=" that take the RValue& as
-//     their sole parameter.
-//
-// Only r-values will trigger this sequence and execute our "move constructor"
-// or "move operator=."  L-values will match the private copy constructor and
-// operator= first giving a "private in this context" error.  This combination
-// gives us a move-only type.
-//
-// For signaling a destructive transfer of data from an l-value, we provide a
-// method named Pass() which creates an r-value for the current instance
-// triggering the move constructor or move operator=.
-//
-// Other ways to get r-values is to use the result of an expression like a
-// function call.
-//
-// Here's an example with comments explaining what gets triggered where:
-//
-//    class Foo {
-//      RTC_MOVE_ONLY_TYPE_FOR_CPP_03(Foo, RValue);
-//
-//     public:
-//       ... API ...
-//       Foo(RValue other);           // Move constructor.
-//       Foo& operator=(RValue rhs);  // Move operator=
-//    };
-//
-//    Foo MakeFoo();  // Function that returns a Foo.
-//
-//    Foo f;
-//    Foo f_copy(f);  // ERROR: Foo(Foo&) is private in this context.
-//    Foo f_assign;
-//    f_assign = f;   // ERROR: operator=(Foo&) is private in this context.
-//
-//
-//    Foo f(MakeFoo());      // R-value so alternate conversion executed.
-//    Foo f_copy(f.Pass());  // R-value so alternate conversion executed.
-//    f = f_copy.Pass();     // R-value so alternate conversion executed.
-//
-//
-// IMPLEMENTATION SUBTLETIES WITH RValue
-//
-// The RValue struct is just a container for a pointer back to the original
-// object. It should only ever be created as a temporary, and no external
-// class should ever declare it or use it in a parameter.
-//
-// It is tempting to want to use the RValue type in function parameters, but
-// excluding the limited usage here for the move constructor and move
-// operator=, doing so would mean that the function could take both r-values
-// and l-values equially which is unexpected.  See COMPARED To Boost.Move for
-// more details.
-//
-// An alternate, and incorrect, implementation of the RValue class used by
-// Boost.Move makes RValue a fieldless child of the move-only type. RValue&
-// is then used in place of RValue in the various operators.  The RValue& is
-// "created" by doing *reinterpret_cast<RValue*>(this).  This has the appeal
-// of never creating a temporary RValue struct even with optimizations
-// disabled.  Also, by virtue of inheritance you can treat the RValue
-// reference as if it were the move-only type itself.  Unfortunately,
-// using the result of this reinterpret_cast<> is actually undefined behavior
-// due to C++98 5.2.10.7. In certain compilers (e.g., NaCl) the optimizer
-// will generate non-working code.
-//
-// In optimized builds, both implementations generate the same assembly so we
-// choose the one that adheres to the standard.
-//
-//
-// WHY HAVE typedef void MoveOnlyTypeForCPP03
-//
-// Callback<>/Bind() needs to understand movable-but-not-copyable semantics
-// to call .Pass() appropriately when it is expected to transfer the value.
-// The cryptic typedef MoveOnlyTypeForCPP03 is added to make this check
-// easy and automatic in helper templates for Callback<>/Bind().
-// See IsMoveOnlyType template and its usage in base/callback_internal.h
-// for more details.
-//
-//
-// COMPARED TO C++11
-//
-// In C++11, you would implement this functionality using an r-value reference
-// and our .Pass() method would be replaced with a call to std::move().
-//
-// This emulation also has a deficiency where it uses up the single
-// user-defined conversion allowed by C++ during initialization.  This can
-// cause problems in some API edge cases.  For instance, in scoped_ptr, it is
-// impossible to make a function "void Foo(scoped_ptr<Parent> p)" accept a
-// value of type scoped_ptr<Child> even if you add a constructor to
-// scoped_ptr<> that would make it look like it should work.  C++11 does not
-// have this deficiency.
-//
-//
-// COMPARED TO Boost.Move
-//
-// Our implementation similar to Boost.Move, but we keep the RValue struct
-// private to the move-only type, and we don't use the reinterpret_cast<> hack.
-//
-// In Boost.Move, RValue is the boost::rv<> template.  This type can be used
-// when writing APIs like:
-//
-//   void MyFunc(boost::rv<Foo>& f)
-//
-// that can take advantage of rv<> to avoid extra copies of a type.  However you
-// would still be able to call this version of MyFunc with an l-value:
-//
-//   Foo f;
-//   MyFunc(f);  // Uh oh, we probably just destroyed |f| w/o calling Pass().
-//
-// unless someone is very careful to also declare a parallel override like:
-//
-//   void MyFunc(const Foo& f)
-//
-// that would catch the l-values first.  This was declared unsafe in C++11 and
-// a C++11 compiler will explicitly fail MyFunc(f).  Unfortunately, we cannot
-// ensure this in C++03.
-//
-// Since we have no need for writing such APIs yet, our implementation keeps
-// RValue private and uses a .Pass() method to do the conversion instead of
-// trying to write a version of "std::move()." Writing an API like std::move()
-// would require the RValue struct to be public.
-//
-//
-// CAVEATS
-//
-// If you include a move-only type as a field inside a class that does not
-// explicitly declare a copy constructor, the containing class's implicit
-// copy constructor will change from Containing(const Containing&) to
-// Containing(Containing&).  This can cause some unexpected errors.
-//
-//   http://llvm.org/bugs/show_bug.cgi?id=11528
-//
-// The workaround is to explicitly declare your copy constructor.
-//
-#define RTC_MOVE_ONLY_TYPE_FOR_CPP_03(type, rvalue_type) \
- private: \
-  struct rvalue_type { \
-    explicit rvalue_type(type* object) : object(object) {} \
-    type* object; \
-  }; \
-  type(type&); \
-  void operator=(type&); \
- public: \
-  operator rvalue_type() { return rvalue_type(this); } \
-  type Pass() WARN_UNUSED_RESULT { return type(rvalue_type(this)); } \
-  typedef void MoveOnlyTypeForCPP03; \
- private:
-
-#define RTC_MOVE_ONLY_TYPE_WITH_MOVE_CONSTRUCTOR_FOR_CPP_03(type) \
- private: \
-  type(type&); \
-  void operator=(type&); \
- public: \
-  type&& Pass() WARN_UNUSED_RESULT { return static_cast<type&&>(*this); } \
-  typedef void MoveOnlyTypeForCPP03; \
- private:
-
-#endif  // WEBRTC_BASE_MOVE_H_
diff --git a/media/webrtc/trunk/webrtc/base/multipart.h b/media/webrtc/trunk/webrtc/base/multipart.h
index 1eeef5ce2d..a099230cc5 100644
--- a/media/webrtc/trunk/webrtc/base/multipart.h
+++ b/media/webrtc/trunk/webrtc/base/multipart.h
@@ -75,7 +75,7 @@ class MultipartStream : public StreamInterface, public sigslot::has_slots<> {
   size_t current_;  // The index into parts_ of the current read position.
   size_t position_;  // The current read position in bytes.
 
-  DISALLOW_COPY_AND_ASSIGN(MultipartStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MultipartStream);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/multipart_unittest.cc b/media/webrtc/trunk/webrtc/base/multipart_unittest.cc
index 38e1114935..9db316b15d 100644
--- a/media/webrtc/trunk/webrtc/base/multipart_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/multipart_unittest.cc
@@ -91,7 +91,7 @@ TEST(MultipartTest, TestAddAndRead) {
 
   // Read the multipart stream into StringStream
   std::string str;
-  rtc::StringStream str_stream(str);
+  rtc::StringStream str_stream(&str);
   EXPECT_EQ(rtc::SR_SUCCESS,
             Flow(&multipart, buffer, sizeof(buffer), &str_stream));
   EXPECT_EQ(size, str.length());
diff --git a/media/webrtc/trunk/webrtc/base/nat_unittest.cc b/media/webrtc/trunk/webrtc/base/nat_unittest.cc
index 1d4ee413b4..8be1be9f05 100644
--- a/media/webrtc/trunk/webrtc/base/nat_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/nat_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
 #include <string>
 
 #include "webrtc/base/gunit.h"
@@ -18,8 +19,8 @@
 #include "webrtc/base/network.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/testclient.h"
+#include "webrtc/base/asynctcpsocket.h"
 #include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using namespace rtc;
 
@@ -36,6 +37,11 @@ TestClient* CreateTestClient(
   return new TestClient(socket);
 }
 
+TestClient* CreateTCPTestClient(AsyncSocket* socket) {
+  AsyncTCPSocket* packet_socket = new AsyncTCPSocket(socket, false);
+  return new TestClient(packet_socket);
+}
+
 // Tests that when sending from internal_addr to external_addrs through the
 // NAT type specified by nat_type, all external addrs receive the sent packet
 // and, if exp_same is true, all use the same mapped-address on the NAT.
@@ -48,10 +54,11 @@ void TestSend(
 
   SocketAddress server_addr = internal_addr;
   server_addr.SetPort(0);  // Auto-select a port
-  NATServer* nat = new NATServer(
-      nat_type, internal, server_addr, external, external_addrs[0]);
+  NATServer* nat = new NATServer(nat_type, internal, server_addr, server_addr,
+                                 external, external_addrs[0]);
   NATSocketFactory* natsf = new NATSocketFactory(internal,
-                                                 nat->internal_address());
+                                                 nat->internal_udp_address(),
+                                                 nat->internal_tcp_address());
 
   TestClient* in = CreateTestClient(natsf, internal_addr);
   TestClient* out[4];
@@ -99,10 +106,11 @@ void TestRecv(
 
   SocketAddress server_addr = internal_addr;
   server_addr.SetPort(0);  // Auto-select a port
-  NATServer* nat = new NATServer(
-      nat_type, internal, server_addr, external, external_addrs[0]);
+  NATServer* nat = new NATServer(nat_type, internal, server_addr, server_addr,
+                                 external, external_addrs[0]);
   NATSocketFactory* natsf = new NATSocketFactory(internal,
-                                                 nat->internal_address());
+                                                 nat->internal_udp_address(),
+                                                 nat->internal_tcp_address());
 
   TestClient* in = CreateTestClient(natsf, internal_addr);
   TestClient* out[4];
@@ -199,6 +207,12 @@ void TestPhysicalInternal(const SocketAddress& int_addr) {
 
   std::vector<Network*> networks;
   network_manager.GetNetworks(&networks);
+  networks.erase(std::remove_if(networks.begin(), networks.end(),
+                                [](rtc::Network* network) {
+                                  return rtc::kDefaultNetworkIgnoreMask &
+                                         network->type();
+                                }),
+                 networks.end());
   if (networks.empty()) {
     LOG(LS_WARNING) << "Not enough network adapters for test.";
     return;
@@ -249,6 +263,8 @@ TEST(NatTest, TestPhysicalIPv6) {
   }
 }
 
+namespace {
+
 class TestVirtualSocketServer : public VirtualSocketServer {
  public:
   explicit TestVirtualSocketServer(SocketServer* ss)
@@ -261,6 +277,8 @@ class TestVirtualSocketServer : public VirtualSocketServer {
   scoped_ptr<SocketServer> ss_;
 };
 
+}  // namespace
+
 void TestVirtualInternal(int family) {
   scoped_ptr<TestVirtualSocketServer> int_vss(new TestVirtualSocketServer(
       new PhysicalSocketServer()));
@@ -291,56 +309,86 @@ TEST(NatTest, TestVirtualIPv6) {
   }
 }
 
-// TODO: Finish this test
 class NatTcpTest : public testing::Test, public sigslot::has_slots<> {
  public:
-  NatTcpTest() : connected_(false) {}
-  virtual void SetUp() {
-    int_vss_ = new TestVirtualSocketServer(new PhysicalSocketServer());
-    ext_vss_ = new TestVirtualSocketServer(new PhysicalSocketServer());
-    nat_ = new NATServer(NAT_OPEN_CONE, int_vss_, SocketAddress(),
-                         ext_vss_, SocketAddress());
-    natsf_ = new NATSocketFactory(int_vss_, nat_->internal_address());
+  NatTcpTest()
+      : int_addr_("192.168.0.1", 0),
+        ext_addr_("10.0.0.1", 0),
+        connected_(false),
+        int_pss_(new PhysicalSocketServer()),
+        ext_pss_(new PhysicalSocketServer()),
+        int_vss_(new TestVirtualSocketServer(int_pss_)),
+        ext_vss_(new TestVirtualSocketServer(ext_pss_)),
+        int_thread_(new Thread(int_vss_.get())),
+        ext_thread_(new Thread(ext_vss_.get())),
+        nat_(new NATServer(NAT_OPEN_CONE, int_vss_.get(), int_addr_, int_addr_,
+                           ext_vss_.get(), ext_addr_)),
+        natsf_(new NATSocketFactory(int_vss_.get(),
+                                    nat_->internal_udp_address(),
+                                    nat_->internal_tcp_address())) {
+    int_thread_->Start();
+    ext_thread_->Start();
   }
+
   void OnConnectEvent(AsyncSocket* socket) {
     connected_ = true;
   }
+
   void OnAcceptEvent(AsyncSocket* socket) {
-    accepted_ = server_->Accept(NULL);
+    accepted_.reset(server_->Accept(NULL));
   }
+
   void OnCloseEvent(AsyncSocket* socket, int error) {
   }
+
   void ConnectEvents() {
     server_->SignalReadEvent.connect(this, &NatTcpTest::OnAcceptEvent);
     client_->SignalConnectEvent.connect(this, &NatTcpTest::OnConnectEvent);
   }
-  TestVirtualSocketServer* int_vss_;
-  TestVirtualSocketServer* ext_vss_;
-  NATServer* nat_;
-  NATSocketFactory* natsf_;
-  AsyncSocket* client_;
-  AsyncSocket* server_;
-  AsyncSocket* accepted_;
+
+  SocketAddress int_addr_;
+  SocketAddress ext_addr_;
   bool connected_;
+  PhysicalSocketServer* int_pss_;
+  PhysicalSocketServer* ext_pss_;
+  rtc::scoped_ptr<TestVirtualSocketServer> int_vss_;
+  rtc::scoped_ptr<TestVirtualSocketServer> ext_vss_;
+  rtc::scoped_ptr<Thread> int_thread_;
+  rtc::scoped_ptr<Thread> ext_thread_;
+  rtc::scoped_ptr<NATServer> nat_;
+  rtc::scoped_ptr<NATSocketFactory> natsf_;
+  rtc::scoped_ptr<AsyncSocket> client_;
+  rtc::scoped_ptr<AsyncSocket> server_;
+  rtc::scoped_ptr<AsyncSocket> accepted_;
 };
 
 TEST_F(NatTcpTest, DISABLED_TestConnectOut) {
-  server_ = ext_vss_->CreateAsyncSocket(SOCK_STREAM);
-  server_->Bind(SocketAddress());
+  server_.reset(ext_vss_->CreateAsyncSocket(SOCK_STREAM));
+  server_->Bind(ext_addr_);
   server_->Listen(5);
 
-  client_ = int_vss_->CreateAsyncSocket(SOCK_STREAM);
-  EXPECT_GE(0, client_->Bind(SocketAddress()));
+  client_.reset(natsf_->CreateAsyncSocket(SOCK_STREAM));
+  EXPECT_GE(0, client_->Bind(int_addr_));
   EXPECT_GE(0, client_->Connect(server_->GetLocalAddress()));
 
-
   ConnectEvents();
 
   EXPECT_TRUE_WAIT(connected_, 1000);
   EXPECT_EQ(client_->GetRemoteAddress(), server_->GetLocalAddress());
-  EXPECT_EQ(client_->GetRemoteAddress(), accepted_->GetLocalAddress());
-  EXPECT_EQ(client_->GetLocalAddress(), accepted_->GetRemoteAddress());
+  EXPECT_EQ(accepted_->GetRemoteAddress().ipaddr(), ext_addr_.ipaddr());
 
-  client_->Close();
+  rtc::scoped_ptr<rtc::TestClient> in(CreateTCPTestClient(client_.release()));
+  rtc::scoped_ptr<rtc::TestClient> out(
+      CreateTCPTestClient(accepted_.release()));
+
+  const char* buf = "test_packet";
+  size_t len = strlen(buf);
+
+  in->Send(buf, len);
+  SocketAddress trans_addr;
+  EXPECT_TRUE(out->CheckNextPacket(buf, len, &trans_addr));
+
+  out->Send(buf, len);
+  EXPECT_TRUE(in->CheckNextPacket(buf, len, &trans_addr));
 }
-//#endif
+// #endif
diff --git a/media/webrtc/trunk/webrtc/base/natserver.cc b/media/webrtc/trunk/webrtc/base/natserver.cc
index 0ce04d70b3..b071e014db 100644
--- a/media/webrtc/trunk/webrtc/base/natserver.cc
+++ b/media/webrtc/trunk/webrtc/base/natserver.cc
@@ -11,6 +11,7 @@
 #include "webrtc/base/natsocketfactory.h"
 #include "webrtc/base/natserver.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/socketadapters.h"
 
 namespace rtc {
 
@@ -63,14 +64,77 @@ bool AddrCmp::operator()(
   return false;
 }
 
+// Proxy socket that will capture the external destination address intended for
+// a TCP connection to the NAT server.
+class NATProxyServerSocket : public AsyncProxyServerSocket {
+ public:
+  NATProxyServerSocket(AsyncSocket* socket)
+      : AsyncProxyServerSocket(socket, kNATEncodedIPv6AddressSize) {
+    BufferInput(true);
+  }
+
+  void SendConnectResult(int err, const SocketAddress& addr) override {
+    char code = err ? 1 : 0;
+    BufferedReadAdapter::DirectSend(&code, sizeof(char));
+  }
+
+ protected:
+  void ProcessInput(char* data, size_t* len) override {
+    if (*len < 2) {
+      return;
+    }
+
+    int family = data[1];
+    ASSERT(family == AF_INET || family == AF_INET6);
+    if ((family == AF_INET && *len < kNATEncodedIPv4AddressSize) ||
+        (family == AF_INET6 && *len < kNATEncodedIPv6AddressSize)) {
+      return;
+    }
+
+    SocketAddress dest_addr;
+    size_t address_length = UnpackAddressFromNAT(data, *len, &dest_addr);
+
+    *len -= address_length;
+    if (*len > 0) {
+      memmove(data, data + address_length, *len);
+    }
+
+    bool remainder = (*len > 0);
+    BufferInput(false);
+    SignalConnectRequest(this, dest_addr);
+    if (remainder) {
+      SignalReadEvent(this);
+    }
+  }
+
+};
+
+class NATProxyServer : public ProxyServer {
+ public:
+  NATProxyServer(SocketFactory* int_factory, const SocketAddress& int_addr,
+                 SocketFactory* ext_factory, const SocketAddress& ext_ip)
+      : ProxyServer(int_factory, int_addr, ext_factory, ext_ip) {
+  }
+
+ protected:
+  AsyncProxyServerSocket* WrapSocket(AsyncSocket* socket) override {
+    return new NATProxyServerSocket(socket);
+  }
+};
+
 NATServer::NATServer(
-    NATType type, SocketFactory* internal, const SocketAddress& internal_addr,
+    NATType type, SocketFactory* internal,
+    const SocketAddress& internal_udp_addr,
+    const SocketAddress& internal_tcp_addr,
     SocketFactory* external, const SocketAddress& external_ip)
     : external_(external), external_ip_(external_ip.ipaddr(), 0) {
   nat_ = NAT::Create(type);
 
-  server_socket_ = AsyncUDPSocket::Create(internal, internal_addr);
-  server_socket_->SignalReadPacket.connect(this, &NATServer::OnInternalPacket);
+  udp_server_socket_ = AsyncUDPSocket::Create(internal, internal_udp_addr);
+  udp_server_socket_->SignalReadPacket.connect(this,
+                                               &NATServer::OnInternalUDPPacket);
+  tcp_proxy_server_ = new NATProxyServer(internal, internal_tcp_addr, external,
+                                         external_ip);
 
   int_map_ = new InternalMap(RouteCmp(nat_));
   ext_map_ = new ExternalMap();
@@ -83,15 +147,15 @@ NATServer::~NATServer() {
     delete iter->second;
 
   delete nat_;
-  delete server_socket_;
+  delete udp_server_socket_;
+  delete tcp_proxy_server_;
   delete int_map_;
   delete ext_map_;
 }
 
-void NATServer::OnInternalPacket(
+void NATServer::OnInternalUDPPacket(
     AsyncPacketSocket* socket, const char* buf, size_t size,
     const SocketAddress& addr, const PacketTime& packet_time) {
-
   // Read the intended destination from the wire.
   SocketAddress dest_addr;
   size_t length = UnpackAddressFromNAT(buf, size, &dest_addr);
@@ -113,10 +177,9 @@ void NATServer::OnInternalPacket(
   iter->second->socket->SendTo(buf + length, size - length, dest_addr, options);
 }
 
-void NATServer::OnExternalPacket(
+void NATServer::OnExternalUDPPacket(
     AsyncPacketSocket* socket, const char* buf, size_t size,
     const SocketAddress& remote_addr, const PacketTime& packet_time) {
-
   SocketAddress local_addr = socket->GetLocalAddress();
 
   // Find the translation for this addresses.
@@ -139,8 +202,8 @@ void NATServer::OnExternalPacket(
   // Copy the data part after the address.
   rtc::PacketOptions options;
   memcpy(real_buf.get() + addrlength, buf, size);
-  server_socket_->SendTo(real_buf.get(), size + addrlength,
-                         iter->second->route.source(), options);
+  udp_server_socket_->SendTo(real_buf.get(), size + addrlength,
+                             iter->second->route.source(), options);
 }
 
 void NATServer::Translate(const SocketAddressPair& route) {
@@ -154,7 +217,7 @@ void NATServer::Translate(const SocketAddressPair& route) {
   TransEntry* entry = new TransEntry(route, socket, nat_);
   (*int_map_)[route] = entry;
   (*ext_map_)[socket->GetLocalAddress()] = entry;
-  socket->SignalReadPacket.connect(this, &NATServer::OnExternalPacket);
+  socket->SignalReadPacket.connect(this, &NATServer::OnExternalUDPPacket);
 }
 
 bool NATServer::ShouldFilterOut(TransEntry* entry,
diff --git a/media/webrtc/trunk/webrtc/base/natserver.h b/media/webrtc/trunk/webrtc/base/natserver.h
index 16c6d93fdb..b6a02feca3 100644
--- a/media/webrtc/trunk/webrtc/base/natserver.h
+++ b/media/webrtc/trunk/webrtc/base/natserver.h
@@ -19,6 +19,7 @@
 #include "webrtc/base/thread.h"
 #include "webrtc/base/socketfactory.h"
 #include "webrtc/base/nattypes.h"
+#include "webrtc/base/proxyserver.h"
 
 namespace rtc {
 
@@ -46,27 +47,40 @@ struct AddrCmp {
 
 // Implements the NAT device.  It listens for packets on the internal network,
 // translates them, and sends them out over the external network.
+//
+// TCP connections initiated from the internal side of the NAT server are
+// also supported, by making a connection to the NAT server's TCP address and
+// then sending the remote address in quasi-STUN format. The connection status
+// will be indicated back to the client as a 1 byte status code, where '0'
+// indicates success.
 
-const int NAT_SERVER_PORT = 4237;
+const int NAT_SERVER_UDP_PORT = 4237;
+const int NAT_SERVER_TCP_PORT = 4238;
 
 class NATServer : public sigslot::has_slots<> {
  public:
   NATServer(
-      NATType type, SocketFactory* internal, const SocketAddress& internal_addr,
+      NATType type, SocketFactory* internal,
+      const SocketAddress& internal_udp_addr,
+      const SocketAddress& internal_tcp_addr,
       SocketFactory* external, const SocketAddress& external_ip);
   ~NATServer() override;
 
-  SocketAddress internal_address() const {
-    return server_socket_->GetLocalAddress();
+  SocketAddress internal_udp_address() const {
+    return udp_server_socket_->GetLocalAddress();
+  }
+
+  SocketAddress internal_tcp_address() const {
+    return tcp_proxy_server_->GetServerAddress();
   }
 
   // Packets received on one of the networks.
-  void OnInternalPacket(AsyncPacketSocket* socket, const char* buf,
-                        size_t size, const SocketAddress& addr,
-                        const PacketTime& packet_time);
-  void OnExternalPacket(AsyncPacketSocket* socket, const char* buf,
-                        size_t size, const SocketAddress& remote_addr,
-                        const PacketTime& packet_time);
+  void OnInternalUDPPacket(AsyncPacketSocket* socket, const char* buf,
+                           size_t size, const SocketAddress& addr,
+                           const PacketTime& packet_time);
+  void OnExternalUDPPacket(AsyncPacketSocket* socket, const char* buf,
+                           size_t size, const SocketAddress& remote_addr,
+                           const PacketTime& packet_time);
 
  private:
   typedef std::set<SocketAddress, AddrCmp> AddressSet;
@@ -95,14 +109,13 @@ class NATServer : public sigslot::has_slots<> {
   bool ShouldFilterOut(TransEntry* entry, const SocketAddress& ext_addr);
 
   NAT* nat_;
-  SocketFactory* internal_;
   SocketFactory* external_;
   SocketAddress external_ip_;
-  AsyncUDPSocket* server_socket_;
-  AsyncSocket* tcp_server_socket_;
+  AsyncUDPSocket* udp_server_socket_;
+  ProxyServer* tcp_proxy_server_;
   InternalMap* int_map_;
   ExternalMap* ext_map_;
-  DISALLOW_EVIL_CONSTRUCTORS(NATServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(NATServer);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/natsocketfactory.cc b/media/webrtc/trunk/webrtc/base/natsocketfactory.cc
index 9c6756ba98..0abd2a1b05 100644
--- a/media/webrtc/trunk/webrtc/base/natsocketfactory.cc
+++ b/media/webrtc/trunk/webrtc/base/natsocketfactory.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/base/natsocketfactory.h"
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/natserver.h"
 #include "webrtc/base/virtualsocketserver.h"
@@ -26,7 +27,7 @@ size_t PackAddressForNAT(char* buf, size_t buf_size,
   buf[0] = 0;
   buf[1] = family;
   // Writes the port.
-  *(reinterpret_cast<uint16*>(&buf[2])) = HostToNetwork16(remote_addr.port());
+  *(reinterpret_cast<uint16_t*>(&buf[2])) = HostToNetwork16(remote_addr.port());
   if (family == AF_INET) {
     ASSERT(buf_size >= kNATEncodedIPv4AddressSize);
     in_addr v4addr = ip.ipv4_address();
@@ -49,7 +50,8 @@ size_t UnpackAddressFromNAT(const char* buf, size_t buf_size,
   ASSERT(buf_size >= 8);
   ASSERT(buf[0] == 0);
   int family = buf[1];
-  uint16 port = NetworkToHost16(*(reinterpret_cast<const uint16*>(&buf[2])));
+  uint16_t port =
+      NetworkToHost16(*(reinterpret_cast<const uint16_t*>(&buf[2])));
   if (family == AF_INET) {
     const in_addr* v4addr = reinterpret_cast<const in_addr*>(&buf[4]);
     *remote_addr = SocketAddress(IPAddress(*v4addr), port);
@@ -179,8 +181,7 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
 
       // Decode the wire packet into the actual results.
       SocketAddress real_remote_addr;
-      size_t addrlength =
-          UnpackAddressFromNAT(buf_, result, &real_remote_addr);
+      size_t addrlength = UnpackAddressFromNAT(buf_, result, &real_remote_addr);
       memcpy(data, buf_ + addrlength, result - addrlength);
 
       // Make sure this packet should be delivered before returning it.
@@ -221,7 +222,7 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
   ConnState GetState() const override {
     return connected_ ? CS_CONNECTED : CS_CLOSED;
   }
-  int EstimateMTU(uint16* mtu) override { return socket_->EstimateMTU(mtu); }
+  int EstimateMTU(uint16_t* mtu) override { return socket_->EstimateMTU(mtu); }
   int GetOption(Option opt, int* value) override {
     return socket_->GetOption(opt, value);
   }
@@ -230,7 +231,7 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
   }
 
   void OnConnectEvent(AsyncSocket* socket) {
-    // If we're NATed, we need to send a request with the real addr to use.
+    // If we're NATed, we need to send a message with the real addr to use.
     ASSERT(socket == socket_);
     if (server_addr_.IsNil()) {
       connected_ = true;
@@ -269,8 +270,8 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
 
   // Sends the destination address to the server to tell it to connect.
   void SendConnectRequest() {
-    char buf[256];
-    size_t length = PackAddressForNAT(buf, ARRAY_SIZE(buf), remote_addr_);
+    char buf[kNATEncodedIPv6AddressSize];
+    size_t length = PackAddressForNAT(buf, arraysize(buf), remote_addr_);
     socket_->Send(buf, length);
   }
 
@@ -279,6 +280,7 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
     char code;
     socket_->Recv(&code, sizeof(code));
     if (code == 0) {
+      connected_ = true;
       SignalConnectEvent(this);
     } else {
       Close();
@@ -299,8 +301,10 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
 
 // NATSocketFactory
 NATSocketFactory::NATSocketFactory(SocketFactory* factory,
-                                   const SocketAddress& nat_addr)
-    : factory_(factory), nat_addr_(nat_addr) {
+                                   const SocketAddress& nat_udp_addr,
+                                   const SocketAddress& nat_tcp_addr)
+    : factory_(factory), nat_udp_addr_(nat_udp_addr),
+      nat_tcp_addr_(nat_tcp_addr) {
 }
 
 Socket* NATSocketFactory::CreateSocket(int type) {
@@ -321,7 +325,11 @@ AsyncSocket* NATSocketFactory::CreateAsyncSocket(int family, int type) {
 
 AsyncSocket* NATSocketFactory::CreateInternalSocket(int family, int type,
     const SocketAddress& local_addr, SocketAddress* nat_addr) {
-  *nat_addr = nat_addr_;
+  if (type == SOCK_STREAM) {
+    *nat_addr = nat_tcp_addr_;
+  } else {
+    *nat_addr = nat_udp_addr_;
+  }
   return factory_->CreateAsyncSocket(family, type);
 }
 
@@ -385,7 +393,7 @@ AsyncSocket* NATSocketServer::CreateInternalSocket(int family, int type,
   if (nat) {
     socket = nat->internal_factory()->CreateAsyncSocket(family, type);
     *nat_addr = (type == SOCK_STREAM) ?
-        nat->internal_tcp_address() : nat->internal_address();
+        nat->internal_tcp_address() : nat->internal_udp_address();
   } else {
     socket = server_->CreateAsyncSocket(family, type);
   }
@@ -403,7 +411,7 @@ NATSocketServer::Translator::Translator(
   VirtualSocketServer* internal_server = new VirtualSocketServer(server_);
   internal_server->SetMessageQueue(server_->queue());
   internal_factory_.reset(internal_server);
-  nat_server_.reset(new NATServer(type, internal_server, int_ip,
+  nat_server_.reset(new NATServer(type, internal_server, int_ip, int_ip,
                                   ext_factory, ext_ip));
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/natsocketfactory.h b/media/webrtc/trunk/webrtc/base/natsocketfactory.h
index cafd78cbd2..9ca0739440 100644
--- a/media/webrtc/trunk/webrtc/base/natsocketfactory.h
+++ b/media/webrtc/trunk/webrtc/base/natsocketfactory.h
@@ -37,7 +37,8 @@ class NATInternalSocketFactory {
 // from a socket factory, given to the constructor.
 class NATSocketFactory : public SocketFactory, public NATInternalSocketFactory {
  public:
-  NATSocketFactory(SocketFactory* factory, const SocketAddress& nat_addr);
+  NATSocketFactory(SocketFactory* factory, const SocketAddress& nat_udp_addr,
+                   const SocketAddress& nat_tcp_addr);
 
   // SocketFactory implementation
   Socket* CreateSocket(int type) override;
@@ -53,8 +54,9 @@ class NATSocketFactory : public SocketFactory, public NATInternalSocketFactory {
 
  private:
   SocketFactory* factory_;
-  SocketAddress nat_addr_;
-  DISALLOW_EVIL_CONSTRUCTORS(NATSocketFactory);
+  SocketAddress nat_udp_addr_;
+  SocketAddress nat_tcp_addr_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(NATSocketFactory);
 };
 
 // Creates sockets that will send traffic through a NAT depending on what
@@ -94,8 +96,8 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
     ~Translator();
 
     SocketFactory* internal_factory() { return internal_factory_.get(); }
-    SocketAddress internal_address() const {
-      return nat_server_->internal_address();
+    SocketAddress internal_udp_address() const {
+      return nat_server_->internal_udp_address();
     }
     SocketAddress internal_tcp_address() const {
       return SocketAddress();  // nat_server_->internal_tcp_address();
@@ -151,7 +153,7 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
   SocketServer* server_;
   MessageQueue* msg_queue_;
   TranslatorMap nats_;
-  DISALLOW_EVIL_CONSTRUCTORS(NATSocketServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(NATSocketServer);
 };
 
 // Free-standing NAT helper functions.
diff --git a/media/webrtc/trunk/webrtc/base/network.cc b/media/webrtc/trunk/webrtc/base/network.cc
index ff092366fa..488c475137 100644
--- a/media/webrtc/trunk/webrtc/base/network.cc
+++ b/media/webrtc/trunk/webrtc/base/network.cc
@@ -24,23 +24,13 @@
 #elif !defined(__native_client__)
 #include <net/if.h>
 #endif
-#include <sys/socket.h>
-#include <sys/utsname.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <errno.h>
-
-#if defined(WEBRTC_ANDROID)
-#include "webrtc/base/ifaddrs-android.h"
-#elif !defined(__native_client__)
-#include <ifaddrs.h>
-#endif
-
 #endif  // WEBRTC_POSIX
 
 #if defined(WEBRTC_WIN)
 #include "webrtc/base/win32.h"
 #include <Iphlpapi.h>
+#elif !defined(__native_client__)
+#include "webrtc/base/ifaddrs_converter.h"
 #endif
 
 #include <stdio.h>
@@ -48,6 +38,7 @@
 #include <algorithm>
 
 #include "webrtc/base/logging.h"
+#include "webrtc/base/networkmonitor.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socket.h"  // includes something that makes windows happy
 #include "webrtc/base/stream.h"
@@ -62,8 +53,8 @@ namespace {
 // limit of IPv6 networks but could be changed by set_max_ipv6_networks().
 const int kMaxIPv6Networks = 5;
 
-const uint32 kUpdateNetworksMessage = 1;
-const uint32 kSignalNetworksMessage = 2;
+const uint32_t kUpdateNetworksMessage = 1;
+const uint32_t kSignalNetworksMessage = 2;
 
 // Fetch list of networks every two seconds.
 const int kNetworksUpdateIntervalMs = 2000;
@@ -123,12 +114,13 @@ std::string AdapterTypeToString(AdapterType type) {
     case ADAPTER_TYPE_LOOPBACK:
       return "Loopback";
     default:
-      DCHECK(false) << "Invalid type " << type;
+      RTC_DCHECK(false) << "Invalid type " << type;
       return std::string();
   }
 }
 
-bool IsIgnoredIPv6(const IPAddress& ip) {
+#if !defined(__native_client__)
+bool IsIgnoredIPv6(const InterfaceAddress& ip) {
   if (ip.family() != AF_INET6) {
     return false;
   }
@@ -145,11 +137,23 @@ bool IsIgnoredIPv6(const IPAddress& ip) {
     return true;
   }
 
+  // Ignore deprecated IPv6.
+  if (ip.ipv6_flags() & IPV6_ADDRESS_FLAG_DEPRECATED) {
+    return true;
+  }
+
   return false;
 }
+#endif  // !defined(__native_client__)
 
 }  // namespace
 
+// These addresses are used as the targets to find out the default local address
+// on a multi-homed endpoint. They are actually DNS servers.
+const char kPublicIPv4Host[] = "8.8.8.8";
+const char kPublicIPv6Host[] = "2001:4860:4860::8888";
+const int kPublicPort = 53;  // DNS port.
+
 std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
                            int prefix_length) {
   std::ostringstream ost;
@@ -163,8 +167,19 @@ NetworkManager::NetworkManager() {
 NetworkManager::~NetworkManager() {
 }
 
+NetworkManager::EnumerationPermission NetworkManager::enumeration_permission()
+    const {
+  return ENUMERATION_ALLOWED;
+}
+
+bool NetworkManager::GetDefaultLocalAddress(int family, IPAddress* addr) const {
+  return false;
+}
+
 NetworkManagerBase::NetworkManagerBase()
-    : max_ipv6_networks_(kMaxIPv6Networks), ipv6_enabled_(true) {
+    : enumeration_permission_(NetworkManager::ENUMERATION_ALLOWED),
+      max_ipv6_networks_(kMaxIPv6Networks),
+      ipv6_enabled_(true) {
 }
 
 NetworkManagerBase::~NetworkManagerBase() {
@@ -173,11 +188,17 @@ NetworkManagerBase::~NetworkManagerBase() {
   }
 }
 
+NetworkManager::EnumerationPermission
+NetworkManagerBase::enumeration_permission() const {
+  return enumeration_permission_;
+}
+
 void NetworkManagerBase::GetAnyAddressNetworks(NetworkList* networks) {
   if (!ipv4_any_address_network_) {
     const rtc::IPAddress ipv4_any_address(INADDR_ANY);
     ipv4_any_address_network_.reset(
         new rtc::Network("any", "any", ipv4_any_address, 0));
+    ipv4_any_address_network_->set_default_local_address_provider(this);
     ipv4_any_address_network_->AddIP(ipv4_any_address);
   }
   networks->push_back(ipv4_any_address_network_.get());
@@ -187,6 +208,7 @@ void NetworkManagerBase::GetAnyAddressNetworks(NetworkList* networks) {
       const rtc::IPAddress ipv6_any_address(in6addr_any);
       ipv6_any_address_network_.reset(
           new rtc::Network("any", "any", ipv6_any_address, 0));
+      ipv6_any_address_network_->set_default_local_address_provider(this);
       ipv6_any_address_network_->AddIP(ipv6_any_address);
     }
     networks->push_back(ipv6_any_address_network_.get());
@@ -217,20 +239,12 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
 void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
                                           bool* changed,
                                           NetworkManager::Stats* stats) {
+  *changed = false;
   // AddressList in this map will track IP addresses for all Networks
   // with the same key.
   std::map<std::string, AddressList> consolidated_address_list;
   NetworkList list(new_networks);
-
-  // Result of Network merge. Element in this list should have unique key.
-  NetworkList merged_list;
   std::sort(list.begin(), list.end(), CompareNetworks);
-
-  *changed = false;
-
-  if (networks_.size() != list.size())
-    *changed = true;
-
   // First, build a set of network-keys to the ipaddresses.
   for (Network* network : list) {
     bool might_add_to_merged_list = false;
@@ -262,6 +276,8 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
   }
 
   // Next, look for existing network objects to re-use.
+  // Result of Network merge. Element in this list should have unique key.
+  NetworkList merged_list;
   for (const auto& kv : consolidated_address_list) {
     const std::string& key = kv.first;
     Network* net = kv.second.net;
@@ -276,17 +292,36 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
       *changed = true;
     } else {
       // This network exists in the map already. Reset its IP addresses.
-      *changed = existing->second->SetIPs(kv.second.ips, *changed);
-      merged_list.push_back(existing->second);
-      if (existing->second != net) {
+      Network* existing_net = existing->second;
+      *changed = existing_net->SetIPs(kv.second.ips, *changed);
+      merged_list.push_back(existing_net);
+      // If the existing network was not active, networks have changed.
+      if (!existing_net->active()) {
+        *changed = true;
+      }
+      ASSERT(net->active());
+      if (existing_net != net) {
         delete net;
       }
     }
   }
-  networks_ = merged_list;
+  // It may still happen that the merged list is a subset of |networks_|.
+  // To detect this change, we compare their sizes.
+  if (merged_list.size() != networks_.size()) {
+    *changed = true;
+  }
 
-  // If the network lists changes, we resort it.
+  // If the network list changes, we re-assign |networks_| to the merged list
+  // and re-sort it.
   if (*changed) {
+    networks_ = merged_list;
+    // Reset the active states of all networks.
+    for (const auto& kv : networks_map_) {
+      kv.second->set_active(false);
+    }
+    for (Network* network : networks_) {
+      network->set_active(true);
+    }
     std::sort(networks_.begin(), networks_.end(), SortNetworks);
     // Now network interfaces are sorted, we should set the preference value
     // for each of the interfaces we are planning to use.
@@ -308,15 +343,41 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
   }
 }
 
+void NetworkManagerBase::set_default_local_addresses(const IPAddress& ipv4,
+                                                     const IPAddress& ipv6) {
+  if (ipv4.family() == AF_INET) {
+    default_local_ipv4_address_ = ipv4;
+  }
+  if (ipv6.family() == AF_INET6) {
+    default_local_ipv6_address_ = ipv6;
+  }
+}
+
+bool NetworkManagerBase::GetDefaultLocalAddress(int family,
+                                                IPAddress* ipaddr) const {
+  if (family == AF_INET && !default_local_ipv4_address_.IsNil()) {
+    *ipaddr = default_local_ipv4_address_;
+    return true;
+  } else if (family == AF_INET6 && !default_local_ipv6_address_.IsNil()) {
+    *ipaddr = default_local_ipv6_address_;
+    return true;
+  }
+  return false;
+}
+
 BasicNetworkManager::BasicNetworkManager()
     : thread_(NULL), sent_first_update_(false), start_count_(0),
-      network_ignore_mask_(kDefaultNetworkIgnoreMask),
       ignore_non_default_routes_(false) {
 }
 
 BasicNetworkManager::~BasicNetworkManager() {
 }
 
+void BasicNetworkManager::OnNetworksChanged() {
+  LOG(LS_VERBOSE) << "Network change was observed at the network manager";
+  UpdateNetworksOnce();
+}
+
 #if defined(__native_client__)
 
 bool BasicNetworkManager::CreateNetworks(bool include_ignored,
@@ -328,49 +389,47 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
 
 #elif defined(WEBRTC_POSIX)
 void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
+                                         IfAddrsConverter* ifaddrs_converter,
                                          bool include_ignored,
                                          NetworkList* networks) const {
   NetworkMap current_networks;
+
   for (struct ifaddrs* cursor = interfaces;
        cursor != NULL; cursor = cursor->ifa_next) {
     IPAddress prefix;
     IPAddress mask;
-    IPAddress ip;
+    InterfaceAddress ip;
     int scope_id = 0;
 
     // Some interfaces may not have address assigned.
-    if (!cursor->ifa_addr || !cursor->ifa_netmask)
+    if (!cursor->ifa_addr || !cursor->ifa_netmask) {
       continue;
+    }
+    // Skip ones which are down.
+    if (!(cursor->ifa_flags & IFF_RUNNING)) {
+      continue;
+    }
+    // Skip unknown family.
+    if (cursor->ifa_addr->sa_family != AF_INET &&
+        cursor->ifa_addr->sa_family != AF_INET6) {
+      continue;
+    }
+    // Skip IPv6 if not enabled.
+    if (cursor->ifa_addr->sa_family == AF_INET6 && !ipv6_enabled()) {
+      continue;
+    }
+    // Convert to InterfaceAddress.
+    if (!ifaddrs_converter->ConvertIfAddrsToIPAddress(cursor, &ip, &mask)) {
+      continue;
+    }
 
-    switch (cursor->ifa_addr->sa_family) {
-      case AF_INET: {
-        ip = IPAddress(
-            reinterpret_cast<sockaddr_in*>(cursor->ifa_addr)->sin_addr);
-        mask = IPAddress(
-            reinterpret_cast<sockaddr_in*>(cursor->ifa_netmask)->sin_addr);
-        break;
-      }
-      case AF_INET6: {
-        if (ipv6_enabled()) {
-          ip = IPAddress(
-              reinterpret_cast<sockaddr_in6*>(cursor->ifa_addr)->sin6_addr);
-
-          if (IsIgnoredIPv6(ip)) {
-            continue;
-          }
-
-          mask = IPAddress(
-              reinterpret_cast<sockaddr_in6*>(cursor->ifa_netmask)->sin6_addr);
-          scope_id =
-              reinterpret_cast<sockaddr_in6*>(cursor->ifa_addr)->sin6_scope_id;
-          break;
-        } else {
-          continue;
-        }
-      }
-      default: {
+    // Special case for IPv6 address.
+    if (cursor->ifa_addr->sa_family == AF_INET6) {
+      if (IsIgnoredIPv6(ip)) {
         continue;
       }
+      scope_id =
+          reinterpret_cast<sockaddr_in6*>(cursor->ifa_addr)->sin6_scope_id;
     }
 
     int prefix_length = CountIPMaskBits(mask);
@@ -381,18 +440,24 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
     if (existing_network == current_networks.end()) {
       AdapterType adapter_type = ADAPTER_TYPE_UNKNOWN;
       if (cursor->ifa_flags & IFF_LOOPBACK) {
-        // TODO(phoglund): Need to recognize other types as well.
         adapter_type = ADAPTER_TYPE_LOOPBACK;
       }
+#if defined(WEBRTC_IOS)
+      // Cell networks are pdp_ipN on iOS.
+      if (strncmp(cursor->ifa_name, "pdp_ip", 6) == 0) {
+        adapter_type = ADAPTER_TYPE_CELLULAR;
+      }
+#endif
+      // TODO(phoglund): Need to recognize other types as well.
       scoped_ptr<Network> network(new Network(cursor->ifa_name,
-                                              cursor->ifa_name,
-                                              prefix,
-                                              prefix_length,
-                                              adapter_type));
+                                              cursor->ifa_name, prefix,
+                                              prefix_length, adapter_type));
+      network->set_default_local_address_provider(this);
       network->set_scope_id(scope_id);
       network->AddIP(ip);
       network->set_ignored(IsIgnoredNetwork(*network));
       if (include_ignored || !network->ignored()) {
+        current_networks[key] = network.get();
         networks->push_back(network.release());
       }
     } else {
@@ -410,7 +475,9 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
     return false;
   }
 
-  ConvertIfAddrs(interfaces, include_ignored, networks);
+  rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(CreateIfAddrsConverter());
+  ConvertIfAddrs(interfaces, ifaddrs_converter.get(), include_ignored,
+                 networks);
 
   freeifaddrs(interfaces);
   return true;
@@ -487,14 +554,14 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
       PIP_ADAPTER_PREFIX prefixlist = adapter_addrs->FirstPrefix;
       std::string name;
       std::string description;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
       name = ToUtf8(adapter_addrs->FriendlyName,
                     wcslen(adapter_addrs->FriendlyName));
 #endif
       description = ToUtf8(adapter_addrs->Description,
                            wcslen(adapter_addrs->Description));
       for (; address; address = address->Next) {
-#ifndef _DEBUG
+#if defined(NDEBUG)
         name = rtc::ToString(count);
 #endif
 
@@ -539,16 +606,15 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
             // TODO(phoglund): Need to recognize other types as well.
             adapter_type = ADAPTER_TYPE_LOOPBACK;
           }
-          scoped_ptr<Network> network(new Network(name,
-                                                  description,
-                                                  prefix,
-                                                  prefix_length,
-                                                  adapter_type));
+          scoped_ptr<Network> network(new Network(name, description, prefix,
+                                                  prefix_length, adapter_type));
+          network->set_default_local_address_provider(this);
           network->set_scope_id(scope_id);
           network->AddIP(ip);
           bool ignored = IsIgnoredNetwork(*network);
           network->set_ignored(ignored);
           if (include_ignored || !network->ignored()) {
+            current_networks[key] = network.get();
             networks->push_back(network.release());
           }
         } else {
@@ -600,9 +666,6 @@ bool BasicNetworkManager::IsIgnoredNetwork(const Network& network) const {
     }
   }
 
-  if (network_ignore_mask_ & network.type()) {
-    return true;
-  }
 #if defined(WEBRTC_POSIX)
   // Filter out VMware/VirtualBox interfaces, typically named vmnet1,
   // vmnet8, or vboxnet0.
@@ -645,6 +708,7 @@ void BasicNetworkManager::StartUpdating() {
       thread_->Post(this, kSignalNetworksMessage);
   } else {
     thread_->Post(this, kUpdateNetworksMessage);
+    StartNetworkMonitor();
   }
   ++start_count_;
 }
@@ -658,13 +722,36 @@ void BasicNetworkManager::StopUpdating() {
   if (!start_count_) {
     thread_->Clear(this);
     sent_first_update_ = false;
+    StopNetworkMonitor();
   }
 }
 
+void BasicNetworkManager::StartNetworkMonitor() {
+  NetworkMonitorFactory* factory = NetworkMonitorFactory::GetFactory();
+  if (factory == nullptr) {
+    return;
+  }
+  network_monitor_.reset(factory->CreateNetworkMonitor());
+  if (!network_monitor_) {
+    return;
+  }
+  network_monitor_->SignalNetworksChanged.connect(
+      this, &BasicNetworkManager::OnNetworksChanged);
+  network_monitor_->Start();
+}
+
+void BasicNetworkManager::StopNetworkMonitor() {
+  if (!network_monitor_) {
+    return;
+  }
+  network_monitor_->Stop();
+  network_monitor_.reset();
+}
+
 void BasicNetworkManager::OnMessage(Message* msg) {
   switch (msg->message_id) {
-    case kUpdateNetworksMessage:  {
-      DoUpdateNetworks();
+    case kUpdateNetworksMessage: {
+      UpdateNetworksContinually();
       break;
     }
     case kSignalNetworksMessage:  {
@@ -676,7 +763,26 @@ void BasicNetworkManager::OnMessage(Message* msg) {
   }
 }
 
-void BasicNetworkManager::DoUpdateNetworks() {
+IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
+  ASSERT(thread_ == Thread::Current());
+  ASSERT(thread_->socketserver() != nullptr);
+  ASSERT(family == AF_INET || family == AF_INET6);
+
+  scoped_ptr<AsyncSocket> socket(
+      thread_->socketserver()->CreateAsyncSocket(family, SOCK_DGRAM));
+  if (!socket) {
+    return IPAddress();
+  }
+
+  if (!socket->Connect(
+          SocketAddress(family == AF_INET ? kPublicIPv4Host : kPublicIPv6Host,
+                        kPublicPort))) {
+    return IPAddress();
+  }
+  return socket->GetLocalAddress().ipaddr();
+}
+
+void BasicNetworkManager::UpdateNetworksOnce() {
   if (!start_count_)
     return;
 
@@ -687,49 +793,61 @@ void BasicNetworkManager::DoUpdateNetworks() {
     SignalError();
   } else {
     bool changed;
-    MergeNetworkList(list, &changed);
+    NetworkManager::Stats stats;
+    MergeNetworkList(list, &changed, &stats);
+    set_default_local_addresses(QueryDefaultLocalAddress(AF_INET),
+                                QueryDefaultLocalAddress(AF_INET6));
     if (changed || !sent_first_update_) {
       SignalNetworksChanged();
       sent_first_update_ = true;
     }
   }
+}
 
+void BasicNetworkManager::UpdateNetworksContinually() {
+  UpdateNetworksOnce();
   thread_->PostDelayed(kNetworksUpdateIntervalMs, this, kUpdateNetworksMessage);
 }
 
-void BasicNetworkManager::DumpNetworks(bool include_ignored) {
+void BasicNetworkManager::DumpNetworks() {
   NetworkList list;
-  CreateNetworks(include_ignored, &list);
+  GetNetworks(&list);
   LOG(LS_INFO) << "NetworkManager detected " << list.size() << " networks:";
   for (const Network* network : list) {
-    if (!network->ignored() || include_ignored) {
-      LOG(LS_INFO) << network->ToString() << ": "
-                   << network->description()
-                   << ((network->ignored()) ? ", Ignored" : "");
-    }
-  }
-  // Release the network list created previously.
-  // Do this in a seperated for loop for better readability.
-  for (Network* network : list) {
-    delete network;
+    LOG(LS_INFO) << network->ToString() << ": " << network->description()
+                 << ", active ? " << network->active()
+                 << ((network->ignored()) ? ", Ignored" : "");
   }
 }
 
-Network::Network(const std::string& name, const std::string& desc,
-                 const IPAddress& prefix, int prefix_length)
-    : name_(name), description_(desc), prefix_(prefix),
+Network::Network(const std::string& name,
+                 const std::string& desc,
+                 const IPAddress& prefix,
+                 int prefix_length)
+    : name_(name),
+      description_(desc),
+      prefix_(prefix),
       prefix_length_(prefix_length),
-      key_(MakeNetworkKey(name, prefix, prefix_length)), scope_id_(0),
-      ignored_(false), type_(ADAPTER_TYPE_UNKNOWN), preference_(0) {
-}
+      key_(MakeNetworkKey(name, prefix, prefix_length)),
+      scope_id_(0),
+      ignored_(false),
+      type_(ADAPTER_TYPE_UNKNOWN),
+      preference_(0) {}
 
-Network::Network(const std::string& name, const std::string& desc,
-                 const IPAddress& prefix, int prefix_length, AdapterType type)
-    : name_(name), description_(desc), prefix_(prefix),
+Network::Network(const std::string& name,
+                 const std::string& desc,
+                 const IPAddress& prefix,
+                 int prefix_length,
+                 AdapterType type)
+    : name_(name),
+      description_(desc),
+      prefix_(prefix),
       prefix_length_(prefix_length),
-      key_(MakeNetworkKey(name, prefix, prefix_length)), scope_id_(0),
-      ignored_(false), type_(type), preference_(0) {
-}
+      key_(MakeNetworkKey(name, prefix, prefix_length)),
+      scope_id_(0),
+      ignored_(false),
+      type_(type),
+      preference_(0) {}
 
 Network::~Network() = default;
 
diff --git a/media/webrtc/trunk/webrtc/base/network.h b/media/webrtc/trunk/webrtc/base/network.h
index 8e5c8f0fb7..2f2e1b3a45 100644
--- a/media/webrtc/trunk/webrtc/base/network.h
+++ b/media/webrtc/trunk/webrtc/base/network.h
@@ -28,7 +28,12 @@ struct ifaddrs;
 
 namespace rtc {
 
+extern const char kPublicIPv4Host[];
+extern const char kPublicIPv6Host[];
+
+class IfAddrsConverter;
 class Network;
+class NetworkMonitorInterface;
 class Thread;
 
 enum AdapterType {
@@ -50,14 +55,32 @@ const int kDefaultNetworkIgnoreMask = ADAPTER_TYPE_LOOPBACK;
 std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
                            int prefix_length);
 
+class DefaultLocalAddressProvider {
+ public:
+  virtual ~DefaultLocalAddressProvider() = default;
+  // The default local address is the local address used in multi-homed endpoint
+  // when the any address (0.0.0.0 or ::) is used as the local address. It's
+  // important to check the return value as a IP family may not be enabled.
+  virtual bool GetDefaultLocalAddress(int family, IPAddress* ipaddr) const = 0;
+};
+
 // Generic network manager interface. It provides list of local
 // networks.
-class NetworkManager {
+class NetworkManager : public DefaultLocalAddressProvider {
  public:
   typedef std::vector<Network*> NetworkList;
 
+  // This enum indicates whether adapter enumeration is allowed.
+  enum EnumerationPermission {
+    ENUMERATION_ALLOWED,  // Adapter enumeration is allowed. Getting 0 network
+                          // from GetNetworks means that there is no network
+                          // available.
+    ENUMERATION_BLOCKED,  // Adapter enumeration is disabled.
+                          // GetAnyAddressNetworks() should be used instead.
+  };
+
   NetworkManager();
-  virtual ~NetworkManager();
+  ~NetworkManager() override;
 
   // Called when network list is updated.
   sigslot::signal0<> SignalNetworksChanged;
@@ -73,12 +96,15 @@ class NetworkManager {
   virtual void StopUpdating() = 0;
 
   // Returns the current list of networks available on this machine.
-  // UpdateNetworks() must be called before this method is called.
+  // StartUpdating() must be called before this method is called.
   // It makes sure that repeated calls return the same object for a
   // given network, so that quality is tracked appropriately. Does not
   // include ignored networks.
   virtual void GetNetworks(NetworkList* networks) const = 0;
 
+  // return the current permission state of GetNetworks()
+  virtual EnumerationPermission enumeration_permission() const;
+
   // "AnyAddressNetwork" is a network which only contains single "any address"
   // IP address.  (i.e. INADDR_ANY for IPv4 or in6addr_any for IPv6). This is
   // useful as binding to such interfaces allow default routing behavior like
@@ -86,8 +112,9 @@ class NetworkManager {
   // TODO(guoweis): remove this body when chromium implements this.
   virtual void GetAnyAddressNetworks(NetworkList* networks) {}
 
-  // Dumps a list of networks available to LS_INFO.
-  virtual void DumpNetworks(bool include_ignored) {}
+  // Dumps the current list of networks in the network manager.
+  virtual void DumpNetworks() {}
+  bool GetDefaultLocalAddress(int family, IPAddress* ipaddr) const override;
 
   struct Stats {
     int ipv4_network_count;
@@ -113,6 +140,10 @@ class NetworkManagerBase : public NetworkManager {
   void set_max_ipv6_networks(int networks) { max_ipv6_networks_ = networks; }
   int max_ipv6_networks() { return max_ipv6_networks_; }
 
+  EnumerationPermission enumeration_permission() const override;
+
+  bool GetDefaultLocalAddress(int family, IPAddress* ipaddr) const override;
+
  protected:
   typedef std::map<std::string, Network*> NetworkMap;
   // Updates |networks_| with the networks listed in |list|. If
@@ -127,9 +158,17 @@ class NetworkManagerBase : public NetworkManager {
                         bool* changed,
                         NetworkManager::Stats* stats);
 
+  void set_enumeration_permission(EnumerationPermission state) {
+    enumeration_permission_ = state;
+  }
+
+  void set_default_local_addresses(const IPAddress& ipv4,
+                                   const IPAddress& ipv6);
+
  private:
   friend class NetworkTest;
-  void DoUpdateNetworks();
+
+  EnumerationPermission enumeration_permission_;
 
   NetworkList networks_;
   int max_ipv6_networks_;
@@ -139,12 +178,16 @@ class NetworkManagerBase : public NetworkManager {
 
   rtc::scoped_ptr<rtc::Network> ipv4_any_address_network_;
   rtc::scoped_ptr<rtc::Network> ipv6_any_address_network_;
+
+  IPAddress default_local_ipv4_address_;
+  IPAddress default_local_ipv6_address_;
 };
 
 // Basic implementation of the NetworkManager interface that gets list
 // of networks using OS APIs.
 class BasicNetworkManager : public NetworkManagerBase,
-                            public MessageHandler {
+                            public MessageHandler,
+                            public sigslot::has_slots<> {
  public:
   BasicNetworkManager();
   ~BasicNetworkManager() override;
@@ -152,8 +195,7 @@ class BasicNetworkManager : public NetworkManagerBase,
   void StartUpdating() override;
   void StopUpdating() override;
 
-  // Logs the available networks.
-  void DumpNetworks(bool include_ignored) override;
+  void DumpNetworks() override;
 
   // MessageHandler interface.
   void OnMessage(Message* msg) override;
@@ -165,18 +207,6 @@ class BasicNetworkManager : public NetworkManagerBase,
     network_ignore_list_ = list;
   }
 
-  // Sets the network types to ignore. For instance, calling this with
-  // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
-  // loopback interfaces. Set to kDefaultNetworkIgnoreMask by default.
-  void set_network_ignore_mask(int network_ignore_mask) {
-    // TODO(phoglund): implement support for other types than loopback.
-    // See https://code.google.com/p/webrtc/issues/detail?id=4288.
-    // Then remove set_network_ignore_list.
-    network_ignore_mask_ = network_ignore_mask;
-  }
-
-  int network_ignore_mask() const { return network_ignore_mask_; }
-
 #if defined(WEBRTC_LINUX)
   // Sets the flag for ignoring non-default routes.
   void set_ignore_non_default_routes(bool value) {
@@ -188,6 +218,7 @@ class BasicNetworkManager : public NetworkManagerBase,
 #if defined(WEBRTC_POSIX)
   // Separated from CreateNetworks for tests.
   void ConvertIfAddrs(ifaddrs* interfaces,
+                      IfAddrsConverter* converter,
                       bool include_ignored,
                       NetworkList* networks) const;
 #endif  // defined(WEBRTC_POSIX)
@@ -199,29 +230,57 @@ class BasicNetworkManager : public NetworkManagerBase,
   // based on the network's property instead of any individual IP.
   bool IsIgnoredNetwork(const Network& network) const;
 
+  // This function connects a UDP socket to a public address and returns the
+  // local address associated it. Since it binds to the "any" address
+  // internally, it returns the default local address on a multi-homed endpoint.
+  IPAddress QueryDefaultLocalAddress(int family) const;
+
  private:
   friend class NetworkTest;
 
-  void DoUpdateNetworks();
+  // Creates a network monitor and listens for network updates.
+  void StartNetworkMonitor();
+  // Stops and removes the network monitor.
+  void StopNetworkMonitor();
+  // Called when it receives updates from the network monitor.
+  void OnNetworksChanged();
+
+  // Updates the networks and reschedules the next update.
+  void UpdateNetworksContinually();
+  // Only updates the networks; does not reschedule the next update.
+  void UpdateNetworksOnce();
 
   Thread* thread_;
   bool sent_first_update_;
   int start_count_;
   std::vector<std::string> network_ignore_list_;
-  int network_ignore_mask_;
   bool ignore_non_default_routes_;
+  scoped_ptr<NetworkMonitorInterface> network_monitor_;
 };
 
 // Represents a Unix-type network interface, with a name and single address.
 class Network {
  public:
-  Network(const std::string& name, const std::string& description,
-          const IPAddress& prefix, int prefix_length);
+  Network(const std::string& name,
+          const std::string& description,
+          const IPAddress& prefix,
+          int prefix_length);
 
-  Network(const std::string& name, const std::string& description,
-          const IPAddress& prefix, int prefix_length, AdapterType type);
+  Network(const std::string& name,
+          const std::string& description,
+          const IPAddress& prefix,
+          int prefix_length,
+          AdapterType type);
   ~Network();
 
+  const DefaultLocalAddressProvider* default_local_address_provider() {
+    return default_local_address_provider_;
+  }
+  void set_default_local_address_provider(
+      const DefaultLocalAddressProvider* provider) {
+    default_local_address_provider_ = provider;
+  }
+
   // Returns the name of the interface this network is associated wtih.
   const std::string& name() const { return name_; }
 
@@ -287,10 +346,17 @@ class Network {
   int preference() const { return preference_; }
   void set_preference(int preference) { preference_ = preference; }
 
+  // When we enumerate networks and find a previously-seen network is missing,
+  // we do not remove it (because it may be used elsewhere). Instead, we mark
+  // it inactive, so that we can detect network changes properly.
+  bool active() const { return active_; }
+  void set_active(bool active) { active_ = active; }
+
   // Debugging description of this network
   std::string ToString() const;
 
  private:
+  const DefaultLocalAddressProvider* default_local_address_provider_ = nullptr;
   std::string name_;
   std::string description_;
   IPAddress prefix_;
@@ -301,6 +367,7 @@ class Network {
   bool ignored_;
   AdapterType type_;
   int preference_;
+  bool active_ = true;
 
   friend class NetworkManager;
 };
diff --git a/media/webrtc/trunk/webrtc/base/network_unittest.cc b/media/webrtc/trunk/webrtc/base/network_unittest.cc
index fdf75caf37..7dd400b996 100644
--- a/media/webrtc/trunk/webrtc/base/network_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/network_unittest.cc
@@ -10,15 +10,14 @@
 
 #include "webrtc/base/network.h"
 
+#include "webrtc/base/nethelpers.h"
+#include "webrtc/base/networkmonitor.h"
 #include <vector>
 #if defined(WEBRTC_POSIX)
 #include <sys/types.h>
-#if !defined(WEBRTC_ANDROID)
-#include <ifaddrs.h>
-#else
-#include "webrtc/base/ifaddrs-android.h"
-#endif
-#endif
+#include <net/if.h>
+#include "webrtc/base/ifaddrs_converter.h"
+#endif  // defined(WEBRTC_POSIX)
 #include "webrtc/base/gunit.h"
 #if defined(WEBRTC_WIN)
 #include "webrtc/base/logging.h"  // For LOG_GLE
@@ -26,6 +25,24 @@
 
 namespace rtc {
 
+namespace {
+
+class FakeNetworkMonitor : public NetworkMonitorBase {
+ public:
+  void Start() override {}
+  void Stop() override {}
+};
+
+class FakeNetworkMonitorFactory : public NetworkMonitorFactory {
+ public:
+  FakeNetworkMonitorFactory() {}
+  NetworkMonitorInterface* CreateNetworkMonitor() {
+    return new FakeNetworkMonitor();
+  }
+};
+
+}  // namespace
+
 class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
  public:
   NetworkTest() : callback_called_(false) {}
@@ -55,13 +72,69 @@ class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
     return list;
   }
 
+  NetworkMonitorInterface* GetNetworkMonitor(
+      BasicNetworkManager& network_manager) {
+    return network_manager.network_monitor_.get();
+  }
+  void ClearNetworks(BasicNetworkManager& network_manager) {
+    for (const auto& kv : network_manager.networks_map_) {
+      delete kv.second;
+    }
+    network_manager.networks_.clear();
+    network_manager.networks_map_.clear();
+  }
+
 #if defined(WEBRTC_POSIX)
   // Separated from CreateNetworks for tests.
   static void CallConvertIfAddrs(const BasicNetworkManager& network_manager,
                                  struct ifaddrs* interfaces,
                                  bool include_ignored,
                                  NetworkManager::NetworkList* networks) {
-    network_manager.ConvertIfAddrs(interfaces, include_ignored, networks);
+    // Use the base IfAddrsConverter for test cases.
+    rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(new IfAddrsConverter());
+    network_manager.ConvertIfAddrs(interfaces, ifaddrs_converter.get(),
+                                   include_ignored, networks);
+  }
+
+  struct sockaddr_in6* CreateIpv6Addr(const std::string& ip_string,
+                                      uint32_t scope_id) {
+    struct sockaddr_in6* ipv6_addr = new struct sockaddr_in6;
+    memset(ipv6_addr, 0, sizeof(struct sockaddr_in6));
+    ipv6_addr->sin6_family = AF_INET6;
+    ipv6_addr->sin6_scope_id = scope_id;
+    IPAddress ip;
+    IPFromString(ip_string, &ip);
+    ipv6_addr->sin6_addr = ip.ipv6_address();
+    return ipv6_addr;
+  }
+
+  // Pointers created here need to be released via ReleaseIfAddrs.
+  struct ifaddrs* AddIpv6Address(struct ifaddrs* list,
+                                 char* if_name,
+                                 const std::string& ipv6_address,
+                                 const std::string& ipv6_netmask,
+                                 uint32_t scope_id) {
+    struct ifaddrs* if_addr = new struct ifaddrs;
+    memset(if_addr, 0, sizeof(struct ifaddrs));
+    if_addr->ifa_name = if_name;
+    if_addr->ifa_addr = reinterpret_cast<struct sockaddr*>(
+        CreateIpv6Addr(ipv6_address, scope_id));
+    if_addr->ifa_netmask =
+        reinterpret_cast<struct sockaddr*>(CreateIpv6Addr(ipv6_netmask, 0));
+    if_addr->ifa_next = list;
+    if_addr->ifa_flags = IFF_RUNNING;
+    return if_addr;
+  }
+
+  void ReleaseIfAddrs(struct ifaddrs* list) {
+    struct ifaddrs* if_addr = list;
+    while (if_addr != nullptr) {
+      struct ifaddrs* next_addr = if_addr->ifa_next;
+      delete if_addr->ifa_addr;
+      delete if_addr->ifa_netmask;
+      delete if_addr;
+      if_addr = next_addr;
+    }
   }
 #endif  // defined(WEBRTC_POSIX)
 
@@ -69,6 +142,12 @@ class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
   bool callback_called_;
 };
 
+class TestBasicNetworkManager : public BasicNetworkManager {
+ public:
+  using BasicNetworkManager::QueryDefaultLocalAddress;
+  using BasicNetworkManager::set_default_local_addresses;
+};
+
 // Test that the Network ctor works properly.
 TEST_F(NetworkTest, TestNetworkConstruct) {
   Network ipv4_network1("test_eth0", "Test Network Adapter 1",
@@ -80,26 +159,6 @@ TEST_F(NetworkTest, TestNetworkConstruct) {
   EXPECT_FALSE(ipv4_network1.ignored());
 }
 
-// Tests that our ignore function works properly.
-TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresOnlyLoopbackByDefault) {
-  Network ipv4_network1("test_eth0", "Test Network Adapter 1",
-                        IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET);
-  Network ipv4_network2("test_wlan0", "Test Network Adapter 2",
-                        IPAddress(0x12345601U), 16, ADAPTER_TYPE_WIFI);
-  Network ipv4_network3("test_cell0", "Test Network Adapter 3",
-                        IPAddress(0x12345602U), 16, ADAPTER_TYPE_CELLULAR);
-  Network ipv4_network4("test_vpn0", "Test Network Adapter 4",
-                        IPAddress(0x12345603U), 16, ADAPTER_TYPE_VPN);
-  Network ipv4_network5("test_lo", "Test Network Adapter 5",
-                        IPAddress(0x12345604U), 16, ADAPTER_TYPE_LOOPBACK);
-  BasicNetworkManager network_manager;
-  EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network1));
-  EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network2));
-  EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network3));
-  EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network4));
-  EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network5));
-}
-
 TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresIPsStartingWith0) {
   Network ipv4_network1("test_eth0", "Test Network Adapter 1",
                         IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET);
@@ -110,21 +169,6 @@ TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresIPsStartingWith0) {
   EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network2));
 }
 
-TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresNetworksAccordingToIgnoreMask) {
-  Network ipv4_network1("test_eth0", "Test Network Adapter 1",
-                        IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET);
-  Network ipv4_network2("test_wlan0", "Test Network Adapter 2",
-                        IPAddress(0x12345601U), 16, ADAPTER_TYPE_WIFI);
-  Network ipv4_network3("test_cell0", "Test Network Adapter 3",
-                        IPAddress(0x12345602U), 16, ADAPTER_TYPE_CELLULAR);
-  BasicNetworkManager network_manager;
-  network_manager.set_network_ignore_mask(
-      ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK | ADAPTER_TYPE_WIFI);
-  EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network1));
-  EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network2));
-  EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network3));
-}
-
 // TODO(phoglund): Remove when ignore list goes away.
 TEST_F(NetworkTest, TestIgnoreList) {
   Network ignore_me("ignore_me", "Ignore me please!",
@@ -177,11 +221,14 @@ TEST_F(NetworkTest, DISABLED_TestCreateNetworks) {
   }
 }
 
-// Test that UpdateNetworks succeeds.
+// Test StartUpdating() and StopUpdating(). network_permission_state starts with
+// ALLOWED.
 TEST_F(NetworkTest, TestUpdateNetworks) {
   BasicNetworkManager manager;
   manager.SignalNetworksChanged.connect(
       static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
+  EXPECT_EQ(NetworkManager::ENUMERATION_ALLOWED,
+            manager.enumeration_permission());
   manager.StartUpdating();
   Thread::Current()->ProcessMessages(0);
   EXPECT_TRUE(callback_called_);
@@ -195,6 +242,8 @@ TEST_F(NetworkTest, TestUpdateNetworks) {
   manager.StopUpdating();
   EXPECT_TRUE(manager.started());
   manager.StopUpdating();
+  EXPECT_EQ(NetworkManager::ENUMERATION_ALLOWED,
+            manager.enumeration_permission());
   EXPECT_FALSE(manager.started());
   manager.StopUpdating();
   EXPECT_FALSE(manager.started());
@@ -551,14 +600,23 @@ TEST_F(NetworkTest, TestMultiplePublicNetworksOnOneInterfaceMerge) {
   }
 }
 
-// Test that DumpNetworks works.
-TEST_F(NetworkTest, TestDumpNetworks) {
+// Test that DumpNetworks does not crash.
+TEST_F(NetworkTest, TestCreateAndDumpNetworks) {
   BasicNetworkManager manager;
-  manager.DumpNetworks(true);
+  NetworkManager::NetworkList list = GetNetworks(manager, true);
+  bool changed;
+  MergeNetworkList(manager, list, &changed);
+  manager.DumpNetworks();
 }
 
 // Test that we can toggle IPv6 on and off.
-TEST_F(NetworkTest, TestIPv6Toggle) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestIPv6Toggle DISABLED_TestIPv6Toggle
+#else
+#define MAYBE_TestIPv6Toggle TestIPv6Toggle
+#endif
+TEST_F(NetworkTest, MAYBE_TestIPv6Toggle) {
   BasicNetworkManager manager;
   bool ipv6_found = false;
   NetworkManager::NetworkList list;
@@ -655,6 +713,40 @@ TEST_F(NetworkTest, TestConvertIfAddrsNoAddress) {
   CallConvertIfAddrs(manager, &list, true, &result);
   EXPECT_TRUE(result.empty());
 }
+
+// Verify that if there are two addresses on one interface, only one network
+// is generated.
+TEST_F(NetworkTest, TestConvertIfAddrsMultiAddressesOnOneInterface) {
+  char if_name[20] = "rmnet0";
+  ifaddrs* list = nullptr;
+  list = AddIpv6Address(list, if_name, "1000:2000:3000:4000:0:0:0:1",
+                        "FFFF:FFFF:FFFF:FFFF::", 0);
+  list = AddIpv6Address(list, if_name, "1000:2000:3000:4000:0:0:0:2",
+                        "FFFF:FFFF:FFFF:FFFF::", 0);
+  NetworkManager::NetworkList result;
+  BasicNetworkManager manager;
+  CallConvertIfAddrs(manager, list, true, &result);
+  EXPECT_EQ(1U, result.size());
+  bool changed;
+  // This ensures we release the objects created in CallConvertIfAddrs.
+  MergeNetworkList(manager, result, &changed);
+  ReleaseIfAddrs(list);
+}
+
+TEST_F(NetworkTest, TestConvertIfAddrsNotRunning) {
+  ifaddrs list;
+  memset(&list, 0, sizeof(list));
+  list.ifa_name = const_cast<char*>("test_iface");
+  sockaddr ifa_addr;
+  sockaddr ifa_netmask;
+  list.ifa_addr = &ifa_addr;
+  list.ifa_netmask = &ifa_netmask;
+
+  NetworkManager::NetworkList result;
+  BasicNetworkManager manager;
+  CallConvertIfAddrs(manager, &list, true, &result);
+  EXPECT_TRUE(result.empty());
+}
 #endif  // defined(WEBRTC_POSIX)
 
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
@@ -738,6 +830,49 @@ TEST_F(NetworkTest, TestMergeNetworkList) {
   EXPECT_EQ(list2[0]->GetIPs()[1], ip2);
 }
 
+// Test that MergeNetworkList successfully detects the change if
+// a network becomes inactive and then active again.
+TEST_F(NetworkTest, TestMergeNetworkListWithInactiveNetworks) {
+  BasicNetworkManager manager;
+  Network network1("test_wifi", "Test Network Adapter 1",
+                   IPAddress(0x12345600U), 24);
+  Network network2("test_eth0", "Test Network Adapter 2",
+                   IPAddress(0x00010000U), 16);
+  network1.AddIP(IPAddress(0x12345678));
+  network2.AddIP(IPAddress(0x00010004));
+  NetworkManager::NetworkList list;
+  Network* net1 = new Network(network1);
+  list.push_back(net1);
+  bool changed;
+  MergeNetworkList(manager, list, &changed);
+  EXPECT_TRUE(changed);
+  list.clear();
+  manager.GetNetworks(&list);
+  ASSERT_EQ(1U, list.size());
+  EXPECT_EQ(net1, list[0]);
+
+  list.clear();
+  Network* net2 = new Network(network2);
+  list.push_back(net2);
+  MergeNetworkList(manager, list, &changed);
+  EXPECT_TRUE(changed);
+  list.clear();
+  manager.GetNetworks(&list);
+  ASSERT_EQ(1U, list.size());
+  EXPECT_EQ(net2, list[0]);
+
+  // Now network1 is inactive. Try to merge it again.
+  list.clear();
+  list.push_back(new Network(network1));
+  MergeNetworkList(manager, list, &changed);
+  EXPECT_TRUE(changed);
+  list.clear();
+  manager.GetNetworks(&list);
+  ASSERT_EQ(1U, list.size());
+  EXPECT_TRUE(list[0]->active());
+  EXPECT_EQ(net1, list[0]);
+}
+
 // Test that the filtering logic follows the defined ruleset in network.h.
 TEST_F(NetworkTest, TestIPv6Selection) {
   InterfaceAddress ip;
@@ -779,4 +914,60 @@ TEST_F(NetworkTest, TestIPv6Selection) {
   EXPECT_EQ(ipv6_network.GetBestIP(), static_cast<IPAddress>(ip));
 }
 
+TEST_F(NetworkTest, TestNetworkMonitoring) {
+  BasicNetworkManager manager;
+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
+                                        &NetworkTest::OnNetworksChanged);
+  FakeNetworkMonitorFactory* factory = new FakeNetworkMonitorFactory();
+  NetworkMonitorFactory::SetFactory(factory);
+  manager.StartUpdating();
+  NetworkMonitorInterface* network_monitor = GetNetworkMonitor(manager);
+  EXPECT_TRUE_WAIT(callback_called_, 1000);
+  callback_called_ = false;
+
+  // Clear the networks so that there will be network changes below.
+  ClearNetworks(manager);
+  // Network manager is started, so the callback is called when the network
+  // monitor fires the network-change event.
+  network_monitor->OnNetworksChanged();
+  EXPECT_TRUE_WAIT(callback_called_, 1000);
+
+  // Network manager is stopped; the network monitor is removed.
+  manager.StopUpdating();
+  EXPECT_TRUE(GetNetworkMonitor(manager) == nullptr);
+
+  NetworkMonitorFactory::ReleaseFactory(factory);
+}
+
+TEST_F(NetworkTest, DefaultLocalAddress) {
+  TestBasicNetworkManager manager;
+  manager.StartUpdating();
+  IPAddress ip;
+
+  // GetDefaultLocalAddress should return false when not set.
+  EXPECT_FALSE(manager.GetDefaultLocalAddress(AF_INET, &ip));
+  EXPECT_FALSE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
+
+  // Make sure we can query default local address when an address for such
+  // address family exists.
+  std::vector<Network*> networks;
+  manager.GetNetworks(&networks);
+  for (auto& network : networks) {
+    if (network->GetBestIP().family() == AF_INET) {
+      EXPECT_TRUE(manager.QueryDefaultLocalAddress(AF_INET) != IPAddress());
+    } else if (network->GetBestIP().family() == AF_INET6) {
+      EXPECT_TRUE(manager.QueryDefaultLocalAddress(AF_INET6) != IPAddress());
+    }
+  }
+
+  // GetDefaultLocalAddress should return the valid default address after set.
+  manager.set_default_local_addresses(GetLoopbackIP(AF_INET),
+                                      GetLoopbackIP(AF_INET6));
+  EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET, &ip));
+  EXPECT_EQ(ip, GetLoopbackIP(AF_INET));
+  EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
+  EXPECT_EQ(ip, GetLoopbackIP(AF_INET6));
+  manager.StopUpdating();
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/networkmonitor.cc b/media/webrtc/trunk/webrtc/base/networkmonitor.cc
new file mode 100644
index 0000000000..92bf0592b5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/networkmonitor.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/networkmonitor.h"
+
+#include "webrtc/base/common.h"
+
+namespace {
+const uint32_t UPDATE_NETWORKS_MESSAGE = 1;
+
+// This is set by NetworkMonitorFactory::SetFactory and the caller of
+// NetworkMonitorFactory::SetFactory must be responsible for calling
+// ReleaseFactory to destroy the factory.
+rtc::NetworkMonitorFactory* network_monitor_factory = nullptr;
+}  // namespace
+
+namespace rtc {
+NetworkMonitorInterface::NetworkMonitorInterface() {}
+
+NetworkMonitorInterface::~NetworkMonitorInterface() {}
+
+NetworkMonitorBase::NetworkMonitorBase() : thread_(Thread::Current()) {}
+NetworkMonitorBase::~NetworkMonitorBase() {}
+
+void NetworkMonitorBase::OnNetworksChanged() {
+  LOG(LS_VERBOSE) << "Network change is received at the network monitor";
+  thread_->Post(this, UPDATE_NETWORKS_MESSAGE);
+}
+
+void NetworkMonitorBase::OnMessage(Message* msg) {
+  ASSERT(msg->message_id == UPDATE_NETWORKS_MESSAGE);
+  SignalNetworksChanged();
+}
+
+NetworkMonitorFactory::NetworkMonitorFactory() {}
+NetworkMonitorFactory::~NetworkMonitorFactory() {}
+
+void NetworkMonitorFactory::SetFactory(NetworkMonitorFactory* factory) {
+  if (network_monitor_factory != nullptr) {
+    delete network_monitor_factory;
+  }
+  network_monitor_factory = factory;
+}
+
+void NetworkMonitorFactory::ReleaseFactory(NetworkMonitorFactory* factory) {
+  if (factory == network_monitor_factory) {
+    SetFactory(nullptr);
+  }
+}
+
+NetworkMonitorFactory* NetworkMonitorFactory::GetFactory() {
+  return network_monitor_factory;
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/networkmonitor.h b/media/webrtc/trunk/webrtc/base/networkmonitor.h
new file mode 100644
index 0000000000..c45c817040
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/networkmonitor.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NETWORKMONITOR_H_
+#define WEBRTC_BASE_NETWORKMONITOR_H_
+
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread.h"
+
+namespace rtc {
+/*
+ * Receives network-change events via |OnNetworksChanged| and signals the
+ * networks changed event.
+ *
+ * Threading consideration:
+ * It is expected that all upstream operations (from native to Java) are
+ * performed from the worker thread. This includes creating, starting and
+ * stopping the monitor. This avoids the potential race condition when creating
+ * the singleton Java NetworkMonitor class. Downstream operations can be from
+ * any thread, but this class will forward all the downstream operations onto
+ * the worker thread.
+ *
+ * Memory consideration:
+ * NetworkMonitor is owned by the caller (NetworkManager). The global network
+ * monitor factory is owned by the factory itself but needs to be released from
+ * the factory creator.
+ */
+// Generic network monitor interface. It starts and stops monitoring network
+// changes, and fires the SignalNetworksChanged event when networks change.
+class NetworkMonitorInterface {
+ public:
+  NetworkMonitorInterface();
+  virtual ~NetworkMonitorInterface();
+
+  sigslot::signal0<> SignalNetworksChanged;
+
+  virtual void Start() = 0;
+  virtual void Stop() = 0;
+
+  // Implementations should call this method on the base when networks change,
+  // and the base will fire SignalNetworksChanged on the right thread.
+  virtual void OnNetworksChanged() = 0;
+};
+
+class NetworkMonitorBase : public NetworkMonitorInterface,
+                           public MessageHandler,
+                           public sigslot::has_slots<> {
+ public:
+  NetworkMonitorBase();
+  ~NetworkMonitorBase() override;
+
+  void OnNetworksChanged() override;
+
+  void OnMessage(Message* msg) override;
+
+ private:
+  Thread* thread_;
+};
+
+/*
+ * NetworkMonitorFactory creates NetworkMonitors.
+ */
+class NetworkMonitorFactory {
+ public:
+  // This is not thread-safe; it should be called once (or once per audio/video
+  // call) during the call initialization.
+  static void SetFactory(NetworkMonitorFactory* factory);
+
+  static void ReleaseFactory(NetworkMonitorFactory* factory);
+  static NetworkMonitorFactory* GetFactory();
+
+  virtual NetworkMonitorInterface* CreateNetworkMonitor() = 0;
+
+  virtual ~NetworkMonitorFactory();
+
+ protected:
+  NetworkMonitorFactory();
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_NETWORKMONITOR_H_
diff --git a/media/webrtc/trunk/webrtc/base/nssidentity.cc b/media/webrtc/trunk/webrtc/base/nssidentity.cc
deleted file mode 100644
index b34ce1dbbd..0000000000
--- a/media/webrtc/trunk/webrtc/base/nssidentity.cc
+++ /dev/null
@@ -1,532 +0,0 @@
-/*
- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <string>
-#include <vector>
-
-#if HAVE_CONFIG_H
-#include "config.h"
-#endif  // HAVE_CONFIG_H
-
-#if HAVE_NSS_SSL_H
-
-#include "webrtc/base/nssidentity.h"
-
-#include "cert.h"
-#include "cryptohi.h"
-#include "keyhi.h"
-#include "nss.h"
-#include "pk11pub.h"
-#include "sechash.h"
-
-#include "webrtc/base/logging.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/nssstreamadapter.h"
-#include "webrtc/base/safe_conversions.h"
-
-namespace rtc {
-
-// Certificate validity lifetime in seconds.
-static const int CERTIFICATE_LIFETIME = 60*60*24*30;  // 30 days, arbitrarily
-// Certificate validity window in seconds.
-// This is to compensate for slightly incorrect system clocks.
-static const int CERTIFICATE_WINDOW = -60*60*24;
-
-NSSKeyPair::~NSSKeyPair() {
-  if (privkey_)
-    SECKEY_DestroyPrivateKey(privkey_);
-  if (pubkey_)
-    SECKEY_DestroyPublicKey(pubkey_);
-}
-
-NSSKeyPair *NSSKeyPair::Generate() {
-  SECKEYPrivateKey *privkey = NULL;
-  SECKEYPublicKey *pubkey = NULL;
-  PK11RSAGenParams rsaparams;
-  rsaparams.keySizeInBits = 1024;
-  rsaparams.pe = 0x010001;  // 65537 -- a common RSA public exponent.
-
-  privkey = PK11_GenerateKeyPair(NSSContext::GetSlot(),
-                                 CKM_RSA_PKCS_KEY_PAIR_GEN,
-                                 &rsaparams, &pubkey, PR_FALSE /*permanent*/,
-                                 PR_FALSE /*sensitive*/, NULL);
-  if (!privkey) {
-    LOG(LS_ERROR) << "Couldn't generate key pair";
-    return NULL;
-  }
-
-  return new NSSKeyPair(privkey, pubkey);
-}
-
-// Just make a copy.
-NSSKeyPair *NSSKeyPair::GetReference() {
-  SECKEYPrivateKey *privkey = SECKEY_CopyPrivateKey(privkey_);
-  if (!privkey)
-    return NULL;
-
-  SECKEYPublicKey *pubkey = SECKEY_CopyPublicKey(pubkey_);
-  if (!pubkey) {
-    SECKEY_DestroyPrivateKey(privkey);
-    return NULL;
-  }
-
-  return new NSSKeyPair(privkey, pubkey);
-}
-
-NSSCertificate::NSSCertificate(CERTCertificate* cert)
-    : certificate_(CERT_DupCertificate(cert)) {
-  ASSERT(certificate_ != NULL);
-}
-
-static void DeleteCert(SSLCertificate* cert) {
-  delete cert;
-}
-
-NSSCertificate::NSSCertificate(CERTCertList* cert_list) {
-  // Copy the first cert into certificate_.
-  CERTCertListNode* node = CERT_LIST_HEAD(cert_list);
-  certificate_ = CERT_DupCertificate(node->cert);
-
-  // Put any remaining certificates into the chain.
-  node = CERT_LIST_NEXT(node);
-  std::vector<SSLCertificate*> certs;
-  for (; !CERT_LIST_END(node, cert_list); node = CERT_LIST_NEXT(node)) {
-    certs.push_back(new NSSCertificate(node->cert));
-  }
-
-  if (!certs.empty())
-    chain_.reset(new SSLCertChain(certs));
-
-  // The SSLCertChain constructor copies its input, so now we have to delete
-  // the originals.
-  std::for_each(certs.begin(), certs.end(), DeleteCert);
-}
-
-NSSCertificate::NSSCertificate(CERTCertificate* cert, SSLCertChain* chain)
-    : certificate_(CERT_DupCertificate(cert)) {
-  ASSERT(certificate_ != NULL);
-  if (chain)
-    chain_.reset(chain->Copy());
-}
-
-NSSCertificate::~NSSCertificate() {
-  if (certificate_)
-    CERT_DestroyCertificate(certificate_);
-}
-
-NSSCertificate *NSSCertificate::FromPEMString(const std::string &pem_string) {
-  std::string der;
-  if (!SSLIdentity::PemToDer(kPemTypeCertificate, pem_string, &der))
-    return NULL;
-
-  SECItem der_cert;
-  der_cert.data = reinterpret_cast<unsigned char *>(const_cast<char *>(
-      der.data()));
-  der_cert.len = checked_cast<unsigned int>(der.size());
-  CERTCertificate *cert = CERT_NewTempCertificate(CERT_GetDefaultCertDB(),
-      &der_cert, NULL, PR_FALSE, PR_TRUE);
-
-  if (!cert)
-    return NULL;
-
-  NSSCertificate* ret = new NSSCertificate(cert);
-  CERT_DestroyCertificate(cert);
-  return ret;
-}
-
-NSSCertificate *NSSCertificate::GetReference() const {
-  return new NSSCertificate(certificate_, chain_.get());
-}
-
-std::string NSSCertificate::ToPEMString() const {
-  return SSLIdentity::DerToPem(kPemTypeCertificate,
-                               certificate_->derCert.data,
-                               certificate_->derCert.len);
-}
-
-void NSSCertificate::ToDER(Buffer* der_buffer) const {
-  der_buffer->SetData(certificate_->derCert.data, certificate_->derCert.len);
-}
-
-static bool Certifies(CERTCertificate* parent, CERTCertificate* child) {
-  // TODO(bemasc): Identify stricter validation checks to use here.  In the
-  // context of some future identity standard, it might make sense to check
-  // the certificates' roles, expiration dates, self-signatures (if
-  // self-signed), certificate transparency logging, or many other attributes.
-  // NOTE: Future changes to this validation may reject some previously allowed
-  // certificate chains.  Users should be advised not to deploy chained
-  // certificates except in controlled environments until the validity
-  // requirements are finalized.
-
-  // Check that the parent's name is the same as the child's claimed issuer.
-  SECComparison name_status =
-      CERT_CompareName(&child->issuer, &parent->subject);
-  if (name_status != SECEqual)
-    return false;
-
-  // Extract the parent's public key, or fail if the key could not be read
-  // (e.g. certificate is corrupted).
-  SECKEYPublicKey* parent_key = CERT_ExtractPublicKey(parent);
-  if (!parent_key)
-    return false;
-
-  // Check that the parent's privkey was actually used to generate the child's
-  // signature.
-  SECStatus verified = CERT_VerifySignedDataWithPublicKey(
-      &child->signatureWrap, parent_key, NULL);
-  SECKEY_DestroyPublicKey(parent_key);
-  return verified == SECSuccess;
-}
-
-bool NSSCertificate::IsValidChain(const CERTCertList* cert_list) {
-  CERTCertListNode* child = CERT_LIST_HEAD(cert_list);
-  for (CERTCertListNode* parent = CERT_LIST_NEXT(child);
-       !CERT_LIST_END(parent, cert_list);
-       child = parent, parent = CERT_LIST_NEXT(parent)) {
-    if (!Certifies(parent->cert, child->cert))
-      return false;
-  }
-  return true;
-}
-
-bool NSSCertificate::GetDigestLength(const std::string& algorithm,
-                                     size_t* length) {
-  const SECHashObject *ho;
-
-  if (!GetDigestObject(algorithm, &ho))
-    return false;
-
-  *length = ho->length;
-
-  return true;
-}
-
-bool NSSCertificate::GetSignatureDigestAlgorithm(std::string* algorithm) const {
-  // The function sec_DecodeSigAlg in NSS provides this mapping functionality.
-  // Unfortunately it is private, so the functionality must be duplicated here.
-  // See https://bugzilla.mozilla.org/show_bug.cgi?id=925165 .
-  SECOidTag sig_alg = SECOID_GetAlgorithmTag(&certificate_->signature);
-  switch (sig_alg) {
-    case SEC_OID_PKCS1_MD5_WITH_RSA_ENCRYPTION:
-      *algorithm = DIGEST_MD5;
-      break;
-    case SEC_OID_PKCS1_SHA1_WITH_RSA_ENCRYPTION:
-    case SEC_OID_ISO_SHA_WITH_RSA_SIGNATURE:
-    case SEC_OID_ISO_SHA1_WITH_RSA_SIGNATURE:
-    case SEC_OID_ANSIX9_DSA_SIGNATURE_WITH_SHA1_DIGEST:
-    case SEC_OID_BOGUS_DSA_SIGNATURE_WITH_SHA1_DIGEST:
-    case SEC_OID_ANSIX962_ECDSA_SHA1_SIGNATURE:
-    case SEC_OID_MISSI_DSS:
-    case SEC_OID_MISSI_KEA_DSS:
-    case SEC_OID_MISSI_KEA_DSS_OLD:
-    case SEC_OID_MISSI_DSS_OLD:
-      *algorithm = DIGEST_SHA_1;
-      break;
-    case SEC_OID_ANSIX962_ECDSA_SHA224_SIGNATURE:
-    case SEC_OID_PKCS1_SHA224_WITH_RSA_ENCRYPTION:
-    case SEC_OID_NIST_DSA_SIGNATURE_WITH_SHA224_DIGEST:
-      *algorithm = DIGEST_SHA_224;
-      break;
-    case SEC_OID_ANSIX962_ECDSA_SHA256_SIGNATURE:
-    case SEC_OID_PKCS1_SHA256_WITH_RSA_ENCRYPTION:
-    case SEC_OID_NIST_DSA_SIGNATURE_WITH_SHA256_DIGEST:
-      *algorithm = DIGEST_SHA_256;
-      break;
-    case SEC_OID_ANSIX962_ECDSA_SHA384_SIGNATURE:
-    case SEC_OID_PKCS1_SHA384_WITH_RSA_ENCRYPTION:
-      *algorithm = DIGEST_SHA_384;
-      break;
-    case SEC_OID_ANSIX962_ECDSA_SHA512_SIGNATURE:
-    case SEC_OID_PKCS1_SHA512_WITH_RSA_ENCRYPTION:
-      *algorithm = DIGEST_SHA_512;
-      break;
-    default:
-      // Unknown algorithm.  There are several unhandled options that are less
-      // common and more complex.
-      algorithm->clear();
-      return false;
-  }
-  return true;
-}
-
-bool NSSCertificate::ComputeDigest(const std::string& algorithm,
-                                   unsigned char* digest,
-                                   size_t size,
-                                   size_t* length) const {
-  const SECHashObject *ho;
-
-  if (!GetDigestObject(algorithm, &ho))
-    return false;
-
-  if (size < ho->length)  // Sanity check for fit
-    return false;
-
-  SECStatus rv = HASH_HashBuf(ho->type, digest,
-                              certificate_->derCert.data,
-                              certificate_->derCert.len);
-  if (rv != SECSuccess)
-    return false;
-
-  *length = ho->length;
-
-  return true;
-}
-
-bool NSSCertificate::GetChain(SSLCertChain** chain) const {
-  if (!chain_)
-    return false;
-
-  *chain = chain_->Copy();
-  return true;
-}
-
-bool NSSCertificate::Equals(const NSSCertificate *tocompare) const {
-  if (!certificate_->derCert.len)
-    return false;
-  if (!tocompare->certificate_->derCert.len)
-    return false;
-
-  if (certificate_->derCert.len != tocompare->certificate_->derCert.len)
-    return false;
-
-  return memcmp(certificate_->derCert.data,
-                tocompare->certificate_->derCert.data,
-                certificate_->derCert.len) == 0;
-}
-
-
-bool NSSCertificate::GetDigestObject(const std::string &algorithm,
-                                     const SECHashObject **hop) {
-  const SECHashObject *ho;
-  HASH_HashType hash_type;
-
-  if (algorithm == DIGEST_SHA_1) {
-    hash_type = HASH_AlgSHA1;
-  // HASH_AlgSHA224 is not supported in the chromium linux build system.
-#if 0
-  } else if (algorithm == DIGEST_SHA_224) {
-    hash_type = HASH_AlgSHA224;
-#endif
-  } else if (algorithm == DIGEST_SHA_256) {
-    hash_type = HASH_AlgSHA256;
-  } else if (algorithm == DIGEST_SHA_384) {
-    hash_type = HASH_AlgSHA384;
-  } else if (algorithm == DIGEST_SHA_512) {
-    hash_type = HASH_AlgSHA512;
-  } else {
-    return false;
-  }
-
-  ho = HASH_GetHashObject(hash_type);
-
-  ASSERT(ho->length >= 20);  // Can't happen
-  *hop = ho;
-
-  return true;
-}
-
-NSSIdentity::NSSIdentity(NSSKeyPair* keypair, NSSCertificate* cert)
-    : keypair_(keypair), certificate_(cert) {
-}
-
-NSSIdentity* NSSIdentity::GenerateInternal(const SSLIdentityParams& params) {
-  std::string subject_name_string = "CN=" + params.common_name;
-  CERTName *subject_name = CERT_AsciiToName(
-      const_cast<char *>(subject_name_string.c_str()));
-  NSSIdentity *identity = NULL;
-  CERTSubjectPublicKeyInfo *spki = NULL;
-  CERTCertificateRequest *certreq = NULL;
-  CERTValidity *validity = NULL;
-  CERTCertificate *certificate = NULL;
-  NSSKeyPair *keypair = NSSKeyPair::Generate();
-  SECItem inner_der;
-  SECStatus rv;
-  PLArenaPool* arena;
-  SECItem signed_cert;
-  PRTime now = PR_Now();
-  PRTime not_before =
-      now + static_cast<PRTime>(params.not_before) * PR_USEC_PER_SEC;
-  PRTime not_after =
-      now + static_cast<PRTime>(params.not_after) * PR_USEC_PER_SEC;
-
-  inner_der.len = 0;
-  inner_der.data = NULL;
-
-  if (!keypair) {
-    LOG(LS_ERROR) << "Couldn't generate key pair";
-    goto fail;
-  }
-
-  if (!subject_name) {
-    LOG(LS_ERROR) << "Couldn't convert subject name " << subject_name;
-    goto fail;
-  }
-
-  spki = SECKEY_CreateSubjectPublicKeyInfo(keypair->pubkey());
-  if (!spki) {
-    LOG(LS_ERROR) << "Couldn't create SPKI";
-    goto fail;
-  }
-
-  certreq = CERT_CreateCertificateRequest(subject_name, spki, NULL);
-  if (!certreq) {
-    LOG(LS_ERROR) << "Couldn't create certificate signing request";
-    goto fail;
-  }
-
-  validity = CERT_CreateValidity(not_before, not_after);
-  if (!validity) {
-    LOG(LS_ERROR) << "Couldn't create validity";
-    goto fail;
-  }
-
-  unsigned long serial;
-  // Note: This serial in principle could collide, but it's unlikely
-  rv = PK11_GenerateRandom(reinterpret_cast<unsigned char *>(&serial),
-                           sizeof(serial));
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Couldn't generate random serial";
-    goto fail;
-  }
-
-  certificate = CERT_CreateCertificate(serial, subject_name, validity, certreq);
-  if (!certificate) {
-    LOG(LS_ERROR) << "Couldn't create certificate";
-    goto fail;
-  }
-
-  arena = certificate->arena;
-
-  rv = SECOID_SetAlgorithmID(arena, &certificate->signature,
-                             SEC_OID_PKCS1_SHA1_WITH_RSA_ENCRYPTION, NULL);
-  if (rv != SECSuccess)
-    goto fail;
-
-  // Set version to X509v3.
-  *(certificate->version.data) = 2;
-  certificate->version.len = 1;
-
-  if (!SEC_ASN1EncodeItem(arena, &inner_der, certificate,
-                          SEC_ASN1_GET(CERT_CertificateTemplate)))
-    goto fail;
-
-  rv = SEC_DerSignData(arena, &signed_cert, inner_der.data, inner_der.len,
-                       keypair->privkey(),
-                       SEC_OID_PKCS1_SHA1_WITH_RSA_ENCRYPTION);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Couldn't sign certificate";
-    goto fail;
-  }
-  certificate->derCert = signed_cert;
-
-  identity = new NSSIdentity(keypair, new NSSCertificate(certificate));
-
-  goto done;
-
- fail:
-  delete keypair;
-
- done:
-  if (certificate) CERT_DestroyCertificate(certificate);
-  if (subject_name) CERT_DestroyName(subject_name);
-  if (spki) SECKEY_DestroySubjectPublicKeyInfo(spki);
-  if (certreq) CERT_DestroyCertificateRequest(certreq);
-  if (validity) CERT_DestroyValidity(validity);
-  return identity;
-}
-
-NSSIdentity* NSSIdentity::Generate(const std::string &common_name) {
-  SSLIdentityParams params;
-  params.common_name = common_name;
-  params.not_before = CERTIFICATE_WINDOW;
-  params.not_after = CERTIFICATE_LIFETIME;
-  return GenerateInternal(params);
-}
-
-NSSIdentity* NSSIdentity::GenerateForTest(const SSLIdentityParams& params) {
-  return GenerateInternal(params);
-}
-
-SSLIdentity* NSSIdentity::FromPEMStrings(const std::string& private_key,
-                                         const std::string& certificate) {
-  std::string private_key_der;
-  if (!SSLIdentity::PemToDer(
-      kPemTypeRsaPrivateKey, private_key, &private_key_der))
-    return NULL;
-
-  SECItem private_key_item;
-  private_key_item.data = reinterpret_cast<unsigned char *>(
-      const_cast<char *>(private_key_der.c_str()));
-  private_key_item.len = checked_cast<unsigned int>(private_key_der.size());
-
-  const unsigned int key_usage = KU_KEY_ENCIPHERMENT | KU_DATA_ENCIPHERMENT |
-      KU_DIGITAL_SIGNATURE;
-
-  SECKEYPrivateKey* privkey = NULL;
-  SECStatus rv =
-      PK11_ImportDERPrivateKeyInfoAndReturnKey(NSSContext::GetSlot(),
-                                               &private_key_item,
-                                               NULL, NULL, PR_FALSE, PR_FALSE,
-                                               key_usage, &privkey, NULL);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Couldn't import private key";
-    return NULL;
-  }
-
-  SECKEYPublicKey *pubkey = SECKEY_ConvertToPublicKey(privkey);
-  if (rv != SECSuccess) {
-    SECKEY_DestroyPrivateKey(privkey);
-    LOG(LS_ERROR) << "Couldn't convert private key to public key";
-    return NULL;
-  }
-
-  // Assign to a scoped_ptr so we don't leak on error.
-  scoped_ptr<NSSKeyPair> keypair(new NSSKeyPair(privkey, pubkey));
-
-  scoped_ptr<NSSCertificate> cert(NSSCertificate::FromPEMString(certificate));
-  if (!cert) {
-    LOG(LS_ERROR) << "Couldn't parse certificate";
-    return NULL;
-  }
-
-  // TODO(ekr@rtfm.com): Check the public key against the certificate.
-
-  return new NSSIdentity(keypair.release(), cert.release());
-}
-
-NSSIdentity::~NSSIdentity() {
-  LOG(LS_INFO) << "Destroying NSS identity";
-}
-
-NSSIdentity *NSSIdentity::GetReference() const {
-  NSSKeyPair *keypair = keypair_->GetReference();
-  if (!keypair)
-    return NULL;
-
-  NSSCertificate *certificate = certificate_->GetReference();
-  if (!certificate) {
-    delete keypair;
-    return NULL;
-  }
-
-  return new NSSIdentity(keypair, certificate);
-}
-
-
-NSSCertificate &NSSIdentity::certificate() const {
-  return *certificate_;
-}
-
-
-}  // rtc namespace
-
-#endif  // HAVE_NSS_SSL_H
-
diff --git a/media/webrtc/trunk/webrtc/base/nssidentity.h b/media/webrtc/trunk/webrtc/base/nssidentity.h
deleted file mode 100644
index bf0a15a711..0000000000
--- a/media/webrtc/trunk/webrtc/base/nssidentity.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_NSSIDENTITY_H_
-#define WEBRTC_BASE_NSSIDENTITY_H_
-
-#include <string>
-
-#include "cert.h"
-#include "nspr.h"
-#include "hasht.h"
-#include "keythi.h"
-
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/sslidentity.h"
-
-namespace rtc {
-
-class NSSKeyPair {
- public:
-  NSSKeyPair(SECKEYPrivateKey* privkey, SECKEYPublicKey* pubkey) :
-      privkey_(privkey), pubkey_(pubkey) {}
-  ~NSSKeyPair();
-
-  // Generate a 1024-bit RSA key pair.
-  static NSSKeyPair* Generate();
-  NSSKeyPair* GetReference();
-
-  SECKEYPrivateKey* privkey() const { return privkey_; }
-  SECKEYPublicKey * pubkey() const { return pubkey_; }
-
- private:
-  SECKEYPrivateKey* privkey_;
-  SECKEYPublicKey* pubkey_;
-
-  DISALLOW_EVIL_CONSTRUCTORS(NSSKeyPair);
-};
-
-
-class NSSCertificate : public SSLCertificate {
- public:
-  static NSSCertificate* FromPEMString(const std::string& pem_string);
-  // The caller retains ownership of the argument to all the constructors,
-  // and the constructor makes a copy.
-  explicit NSSCertificate(CERTCertificate* cert);
-  explicit NSSCertificate(CERTCertList* cert_list);
-  ~NSSCertificate() override;
-
-  NSSCertificate* GetReference() const override;
-
-  std::string ToPEMString() const override;
-
-  void ToDER(Buffer* der_buffer) const override;
-
-  bool GetSignatureDigestAlgorithm(std::string* algorithm) const override;
-
-  bool ComputeDigest(const std::string& algorithm,
-                     unsigned char* digest,
-                     size_t size,
-                     size_t* length) const override;
-
-  bool GetChain(SSLCertChain** chain) const override;
-
-  CERTCertificate* certificate() { return certificate_; }
-
-  // Performs minimal checks to determine if the list is a valid chain.  This
-  // only checks that each certificate certifies the preceding certificate,
-  // and ignores many other certificate features such as expiration dates.
-  static bool IsValidChain(const CERTCertList* cert_list);
-
-  // Helper function to get the length of a digest
-  static bool GetDigestLength(const std::string& algorithm, size_t* length);
-
-  // Comparison.  Only the certificate itself is considered, not the chain.
-  bool Equals(const NSSCertificate* tocompare) const;
-
- private:
-  NSSCertificate(CERTCertificate* cert, SSLCertChain* chain);
-  static bool GetDigestObject(const std::string& algorithm,
-                              const SECHashObject** hash_object);
-
-  CERTCertificate* certificate_;
-  scoped_ptr<SSLCertChain> chain_;
-
-  DISALLOW_EVIL_CONSTRUCTORS(NSSCertificate);
-};
-
-// Represents a SSL key pair and certificate for NSS.
-class NSSIdentity : public SSLIdentity {
- public:
-  static NSSIdentity* Generate(const std::string& common_name);
-  static NSSIdentity* GenerateForTest(const SSLIdentityParams& params);
-  static SSLIdentity* FromPEMStrings(const std::string& private_key,
-                                     const std::string& certificate);
-  ~NSSIdentity() override;
-
-  NSSIdentity* GetReference() const override;
-  NSSCertificate& certificate() const override;
-
-  NSSKeyPair* keypair() const { return keypair_.get(); }
-
- private:
-  NSSIdentity(NSSKeyPair* keypair, NSSCertificate* cert);
-
-  static NSSIdentity* GenerateInternal(const SSLIdentityParams& params);
-
-  rtc::scoped_ptr<NSSKeyPair> keypair_;
-  rtc::scoped_ptr<NSSCertificate> certificate_;
-
-  DISALLOW_EVIL_CONSTRUCTORS(NSSIdentity);
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_NSSIDENTITY_H_
diff --git a/media/webrtc/trunk/webrtc/base/nssstreamadapter.cc b/media/webrtc/trunk/webrtc/base/nssstreamadapter.cc
deleted file mode 100644
index fe1692cc31..0000000000
--- a/media/webrtc/trunk/webrtc/base/nssstreamadapter.cc
+++ /dev/null
@@ -1,1052 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <vector>
-
-#if HAVE_CONFIG_H
-#include "config.h"
-#endif  // HAVE_CONFIG_H
-
-#if HAVE_NSS_SSL_H
-
-#include "webrtc/base/nssstreamadapter.h"
-
-#include "keyhi.h"
-#include "nspr.h"
-#include "nss.h"
-#include "pk11pub.h"
-#include "secerr.h"
-
-#ifdef NSS_SSL_RELATIVE_PATH
-#include "ssl.h"
-#include "sslerr.h"
-#include "sslproto.h"
-#else
-#include "net/third_party/nss/ssl/ssl.h"
-#include "net/third_party/nss/ssl/sslerr.h"
-#include "net/third_party/nss/ssl/sslproto.h"
-#endif
-
-#include "webrtc/base/nssidentity.h"
-#include "webrtc/base/safe_conversions.h"
-#include "webrtc/base/thread.h"
-
-namespace rtc {
-
-PRDescIdentity NSSStreamAdapter::nspr_layer_identity = PR_INVALID_IO_LAYER;
-
-#define UNIMPLEMENTED \
-  PR_SetError(PR_NOT_IMPLEMENTED_ERROR, 0); \
-  LOG(LS_ERROR) \
-  << "Call to unimplemented function "<< __FUNCTION__; ASSERT(false)
-
-#ifdef SRTP_AES128_CM_HMAC_SHA1_80
-#define HAVE_DTLS_SRTP
-#endif
-
-#ifdef HAVE_DTLS_SRTP
-// SRTP cipher suite table
-struct SrtpCipherMapEntry {
-  const char* external_name;
-  PRUint16 cipher_id;
-};
-
-// This isn't elegant, but it's better than an external reference
-static const SrtpCipherMapEntry kSrtpCipherMap[] = {
-  {"AES_CM_128_HMAC_SHA1_80", SRTP_AES128_CM_HMAC_SHA1_80 },
-  {"AES_CM_128_HMAC_SHA1_32", SRTP_AES128_CM_HMAC_SHA1_32 },
-  {NULL, 0}
-};
-#endif
-
-// Default cipher used between NSS stream adapters.
-// This needs to be updated when the default of the SSL library changes.
-static const char kDefaultSslCipher[] = "TLS_RSA_WITH_AES_128_CBC_SHA";
-
-
-// Implementation of NSPR methods
-static PRStatus StreamClose(PRFileDesc *socket) {
-  ASSERT(!socket->lower);
-  socket->dtor(socket);
-  return PR_SUCCESS;
-}
-
-static PRInt32 StreamRead(PRFileDesc *socket, void *buf, PRInt32 length) {
-  StreamInterface *stream = reinterpret_cast<StreamInterface *>(socket->secret);
-  size_t read;
-  int error;
-  StreamResult result = stream->Read(buf, length, &read, &error);
-  if (result == SR_SUCCESS) {
-    return checked_cast<PRInt32>(read);
-  }
-
-  if (result == SR_EOS) {
-    return 0;
-  }
-
-  if (result == SR_BLOCK) {
-    PR_SetError(PR_WOULD_BLOCK_ERROR, 0);
-    return -1;
-  }
-
-  PR_SetError(PR_UNKNOWN_ERROR, error);
-  return -1;
-}
-
-static PRInt32 StreamWrite(PRFileDesc *socket, const void *buf,
-                           PRInt32 length) {
-  StreamInterface *stream = reinterpret_cast<StreamInterface *>(socket->secret);
-  size_t written;
-  int error;
-  StreamResult result = stream->Write(buf, length, &written, &error);
-  if (result == SR_SUCCESS) {
-    return checked_cast<PRInt32>(written);
-  }
-
-  if (result == SR_BLOCK) {
-    LOG(LS_INFO) <<
-        "NSSStreamAdapter: write to underlying transport would block";
-    PR_SetError(PR_WOULD_BLOCK_ERROR, 0);
-    return -1;
-  }
-
-  LOG(LS_ERROR) << "Write error";
-  PR_SetError(PR_UNKNOWN_ERROR, error);
-  return -1;
-}
-
-static PRInt32 StreamAvailable(PRFileDesc *socket) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-PRInt64 StreamAvailable64(PRFileDesc *socket) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRStatus StreamSync(PRFileDesc *socket) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PROffset32 StreamSeek(PRFileDesc *socket, PROffset32 offset,
-                             PRSeekWhence how) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PROffset64 StreamSeek64(PRFileDesc *socket, PROffset64 offset,
-                               PRSeekWhence how) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRStatus StreamFileInfo(PRFileDesc *socket, PRFileInfo *info) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRStatus StreamFileInfo64(PRFileDesc *socket, PRFileInfo64 *info) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRInt32 StreamWritev(PRFileDesc *socket, const PRIOVec *iov,
-                     PRInt32 iov_size, PRIntervalTime timeout) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRStatus StreamConnect(PRFileDesc *socket, const PRNetAddr *addr,
-                              PRIntervalTime timeout) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRFileDesc *StreamAccept(PRFileDesc *sd, PRNetAddr *addr,
-                                PRIntervalTime timeout) {
-  UNIMPLEMENTED;
-  return NULL;
-}
-
-static PRStatus StreamBind(PRFileDesc *socket, const PRNetAddr *addr) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRStatus StreamListen(PRFileDesc *socket, PRIntn depth) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRStatus StreamShutdown(PRFileDesc *socket, PRIntn how) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-// Note: this is always nonblocking and ignores the timeout.
-// TODO(ekr@rtfm.com): In future verify that the socket is
-// actually in non-blocking mode.
-// This function does not support peek.
-static PRInt32 StreamRecv(PRFileDesc *socket, void *buf, PRInt32 amount,
-                   PRIntn flags, PRIntervalTime to) {
-  ASSERT(flags == 0);
-
-  if (flags != 0) {
-    PR_SetError(PR_NOT_IMPLEMENTED_ERROR, 0);
-    return -1;
-  }
-
-  return StreamRead(socket, buf, amount);
-}
-
-// Note: this is always nonblocking and assumes a zero timeout.
-// This function does not support peek.
-static PRInt32 StreamSend(PRFileDesc *socket, const void *buf,
-                          PRInt32 amount, PRIntn flags,
-                          PRIntervalTime to) {
-  ASSERT(flags == 0);
-
-  return StreamWrite(socket, buf, amount);
-}
-
-static PRInt32 StreamRecvfrom(PRFileDesc *socket, void *buf,
-                              PRInt32 amount, PRIntn flags,
-                              PRNetAddr *addr, PRIntervalTime to) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRInt32 StreamSendto(PRFileDesc *socket, const void *buf,
-                            PRInt32 amount, PRIntn flags,
-                            const PRNetAddr *addr, PRIntervalTime to) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRInt16 StreamPoll(PRFileDesc *socket, PRInt16 in_flags,
-                          PRInt16 *out_flags) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRInt32 StreamAcceptRead(PRFileDesc *sd, PRFileDesc **nd,
-                                PRNetAddr **raddr,
-                                void *buf, PRInt32 amount, PRIntervalTime t) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRInt32 StreamTransmitFile(PRFileDesc *sd, PRFileDesc *socket,
-                                  const void *headers, PRInt32 hlen,
-                                  PRTransmitFileFlags flags, PRIntervalTime t) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRStatus StreamGetPeerName(PRFileDesc *socket, PRNetAddr *addr) {
-  // TODO(ekr@rtfm.com): Modify to return unique names for each channel
-  // somehow, as opposed to always the same static address. The current
-  // implementation messes up the session cache, which is why it's off
-  // elsewhere
-  addr->inet.family = PR_AF_INET;
-  addr->inet.port = 0;
-  addr->inet.ip = 0;
-
-  return PR_SUCCESS;
-}
-
-static PRStatus StreamGetSockName(PRFileDesc *socket, PRNetAddr *addr) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRStatus StreamGetSockOption(PRFileDesc *socket, PRSocketOptionData *opt) {
-  switch (opt->option) {
-    case PR_SockOpt_Nonblocking:
-      opt->value.non_blocking = PR_TRUE;
-      return PR_SUCCESS;
-    default:
-      UNIMPLEMENTED;
-      break;
-  }
-
-  return PR_FAILURE;
-}
-
-// Imitate setting socket options. These are mostly noops.
-static PRStatus StreamSetSockOption(PRFileDesc *socket,
-                                    const PRSocketOptionData *opt) {
-  switch (opt->option) {
-    case PR_SockOpt_Nonblocking:
-      return PR_SUCCESS;
-    case PR_SockOpt_NoDelay:
-      return PR_SUCCESS;
-    default:
-      UNIMPLEMENTED;
-      break;
-  }
-
-  return PR_FAILURE;
-}
-
-static PRInt32 StreamSendfile(PRFileDesc *out, PRSendFileData *in,
-                              PRTransmitFileFlags flags, PRIntervalTime to) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static PRStatus StreamConnectContinue(PRFileDesc *socket, PRInt16 flags) {
-  UNIMPLEMENTED;
-  return PR_FAILURE;
-}
-
-static PRIntn StreamReserved(PRFileDesc *socket) {
-  UNIMPLEMENTED;
-  return -1;
-}
-
-static const struct PRIOMethods nss_methods = {
-  PR_DESC_LAYERED,
-  StreamClose,
-  StreamRead,
-  StreamWrite,
-  StreamAvailable,
-  StreamAvailable64,
-  StreamSync,
-  StreamSeek,
-  StreamSeek64,
-  StreamFileInfo,
-  StreamFileInfo64,
-  StreamWritev,
-  StreamConnect,
-  StreamAccept,
-  StreamBind,
-  StreamListen,
-  StreamShutdown,
-  StreamRecv,
-  StreamSend,
-  StreamRecvfrom,
-  StreamSendto,
-  StreamPoll,
-  StreamAcceptRead,
-  StreamTransmitFile,
-  StreamGetSockName,
-  StreamGetPeerName,
-  StreamReserved,
-  StreamReserved,
-  StreamGetSockOption,
-  StreamSetSockOption,
-  StreamSendfile,
-  StreamConnectContinue,
-  StreamReserved,
-  StreamReserved,
-  StreamReserved,
-  StreamReserved
-};
-
-NSSStreamAdapter::NSSStreamAdapter(StreamInterface *stream)
-    : SSLStreamAdapterHelper(stream),
-      ssl_fd_(NULL),
-      cert_ok_(false) {
-}
-
-bool NSSStreamAdapter::Init() {
-  if (nspr_layer_identity == PR_INVALID_IO_LAYER) {
-    nspr_layer_identity = PR_GetUniqueIdentity("nssstreamadapter");
-  }
-  PRFileDesc *pr_fd = PR_CreateIOLayerStub(nspr_layer_identity, &nss_methods);
-  if (!pr_fd)
-    return false;
-  pr_fd->secret = reinterpret_cast<PRFilePrivate *>(stream());
-
-  PRFileDesc *ssl_fd;
-  if (ssl_mode_ == SSL_MODE_DTLS) {
-    ssl_fd = DTLS_ImportFD(NULL, pr_fd);
-  } else {
-    ssl_fd = SSL_ImportFD(NULL, pr_fd);
-  }
-  ASSERT(ssl_fd != NULL);  // This should never happen
-  if (!ssl_fd) {
-    PR_Close(pr_fd);
-    return false;
-  }
-
-  SECStatus rv;
-  // Turn on security.
-  rv = SSL_OptionSet(ssl_fd, SSL_SECURITY, PR_TRUE);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Error enabling security on SSL Socket";
-    return false;
-  }
-
-  // Disable SSLv2.
-  rv = SSL_OptionSet(ssl_fd, SSL_ENABLE_SSL2, PR_FALSE);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Error disabling SSL2";
-    return false;
-  }
-
-  // Disable caching.
-  // TODO(ekr@rtfm.com): restore this when I have the caching
-  // identity set.
-  rv = SSL_OptionSet(ssl_fd, SSL_NO_CACHE, PR_TRUE);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Error disabling cache";
-    return false;
-  }
-
-  // Disable session tickets.
-  rv = SSL_OptionSet(ssl_fd, SSL_ENABLE_SESSION_TICKETS, PR_FALSE);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Error enabling tickets";
-    return false;
-  }
-
-  // Disable renegotiation.
-  rv = SSL_OptionSet(ssl_fd, SSL_ENABLE_RENEGOTIATION,
-                     SSL_RENEGOTIATE_NEVER);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Error disabling renegotiation";
-    return false;
-  }
-
-  // Disable false start.
-  rv = SSL_OptionSet(ssl_fd, SSL_ENABLE_FALSE_START, PR_FALSE);
-  if (rv != SECSuccess) {
-    LOG(LS_ERROR) << "Error disabling false start";
-    return false;
-  }
-
-  ssl_fd_ = ssl_fd;
-
-  return true;
-}
-
-NSSStreamAdapter::~NSSStreamAdapter() {
-  if (ssl_fd_)
-    PR_Close(ssl_fd_);
-};
-
-
-int NSSStreamAdapter::BeginSSL() {
-  SECStatus rv;
-
-  if (!Init()) {
-    Error("Init", -1, false);
-    return -1;
-  }
-
-  ASSERT(state_ == SSL_CONNECTING);
-  // The underlying stream has been opened. If we are in peer-to-peer mode
-  // then a peer certificate must have been specified by now.
-  ASSERT(!ssl_server_name_.empty() ||
-         peer_certificate_.get() != NULL ||
-         !peer_certificate_digest_algorithm_.empty());
-  LOG(LS_INFO) << "BeginSSL: "
-               << (!ssl_server_name_.empty() ? ssl_server_name_ :
-                                               "with peer");
-
-  if (role_ == SSL_CLIENT) {
-    LOG(LS_INFO) << "BeginSSL: as client";
-
-    rv = SSL_GetClientAuthDataHook(ssl_fd_, GetClientAuthDataHook,
-                                   this);
-    if (rv != SECSuccess) {
-      Error("BeginSSL", -1, false);
-      return -1;
-    }
-  } else {
-    LOG(LS_INFO) << "BeginSSL: as server";
-    NSSIdentity *identity;
-
-    if (identity_.get()) {
-      identity = static_cast<NSSIdentity *>(identity_.get());
-    } else {
-      LOG(LS_ERROR) << "Can't be an SSL server without an identity";
-      Error("BeginSSL", -1, false);
-      return -1;
-    }
-    rv = SSL_ConfigSecureServer(ssl_fd_, identity->certificate().certificate(),
-                                identity->keypair()->privkey(),
-                                kt_rsa);
-    if (rv != SECSuccess) {
-      Error("BeginSSL", -1, false);
-      return -1;
-    }
-
-    // Insist on a certificate from the client
-    rv = SSL_OptionSet(ssl_fd_, SSL_REQUEST_CERTIFICATE, PR_TRUE);
-    if (rv != SECSuccess) {
-      Error("BeginSSL", -1, false);
-      return -1;
-    }
-
-    // TODO(juberti): Check for client_auth_enabled()
-
-    rv = SSL_OptionSet(ssl_fd_, SSL_REQUIRE_CERTIFICATE, PR_TRUE);
-    if (rv != SECSuccess) {
-      Error("BeginSSL", -1, false);
-      return -1;
-    }
-  }
-
-  // Set the version range.
-  SSLVersionRange vrange;
-  vrange.min =  (ssl_mode_ == SSL_MODE_DTLS) ?
-      SSL_LIBRARY_VERSION_TLS_1_1 :
-      SSL_LIBRARY_VERSION_TLS_1_0;
-  vrange.max = SSL_LIBRARY_VERSION_TLS_1_1;
-
-  rv = SSL_VersionRangeSet(ssl_fd_, &vrange);
-  if (rv != SECSuccess) {
-    Error("BeginSSL", -1, false);
-    return -1;
-  }
-
-  // SRTP
-#ifdef HAVE_DTLS_SRTP
-  if (!srtp_ciphers_.empty()) {
-    rv = SSL_SetSRTPCiphers(
-        ssl_fd_, &srtp_ciphers_[0],
-        checked_cast<unsigned int>(srtp_ciphers_.size()));
-    if (rv != SECSuccess) {
-      Error("BeginSSL", -1, false);
-      return -1;
-    }
-  }
-#endif
-
-  // Certificate validation
-  rv = SSL_AuthCertificateHook(ssl_fd_, AuthCertificateHook, this);
-  if (rv != SECSuccess) {
-    Error("BeginSSL", -1, false);
-    return -1;
-  }
-
-  // Now start the handshake
-  rv = SSL_ResetHandshake(ssl_fd_, role_ == SSL_SERVER ? PR_TRUE : PR_FALSE);
-  if (rv != SECSuccess) {
-    Error("BeginSSL", -1, false);
-    return -1;
-  }
-
-  return ContinueSSL();
-}
-
-int NSSStreamAdapter::ContinueSSL() {
-  LOG(LS_INFO) << "ContinueSSL";
-  ASSERT(state_ == SSL_CONNECTING);
-
-  // Clear the DTLS timer
-  Thread::Current()->Clear(this, MSG_DTLS_TIMEOUT);
-
-  SECStatus rv = SSL_ForceHandshake(ssl_fd_);
-
-  if (rv == SECSuccess) {
-    LOG(LS_INFO) << "Handshake complete";
-
-    ASSERT(cert_ok_);
-    if (!cert_ok_) {
-      Error("ContinueSSL", -1, true);
-      return -1;
-    }
-
-    state_ = SSL_CONNECTED;
-    StreamAdapterInterface::OnEvent(stream(), SE_OPEN|SE_READ|SE_WRITE, 0);
-    return 0;
-  }
-
-  PRInt32 err = PR_GetError();
-  switch (err) {
-    case SSL_ERROR_RX_MALFORMED_HANDSHAKE:
-      if (ssl_mode_ != SSL_MODE_DTLS) {
-        Error("ContinueSSL", -1, true);
-        return -1;
-      } else {
-        LOG(LS_INFO) << "Malformed DTLS message. Ignoring.";
-        FALLTHROUGH();  // Fall through
-      }
-    case PR_WOULD_BLOCK_ERROR:
-      LOG(LS_INFO) << "Would have blocked";
-      if (ssl_mode_ == SSL_MODE_DTLS) {
-        PRIntervalTime timeout;
-
-        SECStatus rv = DTLS_GetHandshakeTimeout(ssl_fd_, &timeout);
-        if (rv == SECSuccess) {
-          LOG(LS_INFO) << "Timeout is " << timeout << " ms";
-          Thread::Current()->PostDelayed(PR_IntervalToMilliseconds(timeout),
-                                         this, MSG_DTLS_TIMEOUT, 0);
-        }
-      }
-
-      return 0;
-    default:
-      LOG(LS_INFO) << "Error " << err;
-      break;
-  }
-
-  Error("ContinueSSL", -1, true);
-  return -1;
-}
-
-void NSSStreamAdapter::Cleanup() {
-  if (state_ != SSL_ERROR) {
-    state_ = SSL_CLOSED;
-  }
-
-  if (ssl_fd_) {
-    PR_Close(ssl_fd_);
-    ssl_fd_ = NULL;
-  }
-
-  identity_.reset();
-  peer_certificate_.reset();
-
-  Thread::Current()->Clear(this, MSG_DTLS_TIMEOUT);
-}
-
-bool NSSStreamAdapter::GetDigestLength(const std::string& algorithm,
-                                       size_t* length) {
-  return NSSCertificate::GetDigestLength(algorithm, length);
-}
-
-StreamResult NSSStreamAdapter::Read(void* data, size_t data_len,
-                                    size_t* read, int* error) {
-  // SSL_CONNECTED sanity check.
-  switch (state_) {
-    case SSL_NONE:
-    case SSL_WAIT:
-    case SSL_CONNECTING:
-      return SR_BLOCK;
-
-    case SSL_CONNECTED:
-      break;
-
-    case SSL_CLOSED:
-      return SR_EOS;
-
-    case SSL_ERROR:
-    default:
-      if (error)
-        *error = ssl_error_code_;
-      return SR_ERROR;
-  }
-
-  PRInt32 rv = PR_Read(ssl_fd_, data, checked_cast<PRInt32>(data_len));
-
-  if (rv == 0) {
-    return SR_EOS;
-  }
-
-  // Error
-  if (rv < 0) {
-    PRInt32 err = PR_GetError();
-
-    switch (err) {
-      case PR_WOULD_BLOCK_ERROR:
-        return SR_BLOCK;
-      default:
-        Error("Read", -1, false);
-        *error = err;  // libjingle semantics are that this is impl-specific
-        return SR_ERROR;
-    }
-  }
-
-  // Success
-  *read = rv;
-
-  return SR_SUCCESS;
-}
-
-StreamResult NSSStreamAdapter::Write(const void* data, size_t data_len,
-                                     size_t* written, int* error) {
-  // SSL_CONNECTED sanity check.
-  switch (state_) {
-    case SSL_NONE:
-    case SSL_WAIT:
-    case SSL_CONNECTING:
-      return SR_BLOCK;
-
-    case SSL_CONNECTED:
-      break;
-
-    case SSL_ERROR:
-    case SSL_CLOSED:
-    default:
-      if (error)
-        *error = ssl_error_code_;
-      return SR_ERROR;
-  }
-
-  PRInt32 rv = PR_Write(ssl_fd_, data, checked_cast<PRInt32>(data_len));
-
-  // Error
-  if (rv < 0) {
-    PRInt32 err = PR_GetError();
-
-    switch (err) {
-      case PR_WOULD_BLOCK_ERROR:
-        return SR_BLOCK;
-      default:
-        Error("Write", -1, false);
-        *error = err;  // libjingle semantics are that this is impl-specific
-        return SR_ERROR;
-    }
-  }
-
-  // Success
-  *written = rv;
-
-  return SR_SUCCESS;
-}
-
-void NSSStreamAdapter::OnEvent(StreamInterface* stream, int events,
-                               int err) {
-  int events_to_signal = 0;
-  int signal_error = 0;
-  ASSERT(stream == this->stream());
-  if ((events & SE_OPEN)) {
-    LOG(LS_INFO) << "NSSStreamAdapter::OnEvent SE_OPEN";
-    if (state_ != SSL_WAIT) {
-      ASSERT(state_ == SSL_NONE);
-      events_to_signal |= SE_OPEN;
-    } else {
-      state_ = SSL_CONNECTING;
-      if (int err = BeginSSL()) {
-        Error("BeginSSL", err, true);
-        return;
-      }
-    }
-  }
-  if ((events & (SE_READ|SE_WRITE))) {
-    LOG(LS_INFO) << "NSSStreamAdapter::OnEvent"
-                 << ((events & SE_READ) ? " SE_READ" : "")
-                 << ((events & SE_WRITE) ? " SE_WRITE" : "");
-    if (state_ == SSL_NONE) {
-      events_to_signal |= events & (SE_READ|SE_WRITE);
-    } else if (state_ == SSL_CONNECTING) {
-      if (int err = ContinueSSL()) {
-        Error("ContinueSSL", err, true);
-        return;
-      }
-    } else if (state_ == SSL_CONNECTED) {
-      if (events & SE_WRITE) {
-        LOG(LS_INFO) << " -- onStreamWriteable";
-        events_to_signal |= SE_WRITE;
-      }
-      if (events & SE_READ) {
-        LOG(LS_INFO) << " -- onStreamReadable";
-        events_to_signal |= SE_READ;
-      }
-    }
-  }
-  if ((events & SE_CLOSE)) {
-    LOG(LS_INFO) << "NSSStreamAdapter::OnEvent(SE_CLOSE, " << err << ")";
-    Cleanup();
-    events_to_signal |= SE_CLOSE;
-    // SE_CLOSE is the only event that uses the final parameter to OnEvent().
-    ASSERT(signal_error == 0);
-    signal_error = err;
-  }
-  if (events_to_signal)
-    StreamAdapterInterface::OnEvent(stream, events_to_signal, signal_error);
-}
-
-void NSSStreamAdapter::OnMessage(Message* msg) {
-  // Process our own messages and then pass others to the superclass
-  if (MSG_DTLS_TIMEOUT == msg->message_id) {
-    LOG(LS_INFO) << "DTLS timeout expired";
-    ContinueSSL();
-  } else {
-    StreamInterface::OnMessage(msg);
-  }
-}
-
-// Certificate verification callback. Called to check any certificate
-SECStatus NSSStreamAdapter::AuthCertificateHook(void *arg,
-                                                PRFileDesc *fd,
-                                                PRBool checksig,
-                                                PRBool isServer) {
-  LOG(LS_INFO) << "NSSStreamAdapter::AuthCertificateHook";
-  // SSL_PeerCertificate returns a pointer that is owned by the caller, and
-  // the NSSCertificate constructor copies its argument, so |raw_peer_cert|
-  // must be destroyed in this function.
-  CERTCertificate* raw_peer_cert = SSL_PeerCertificate(fd);
-  NSSCertificate peer_cert(raw_peer_cert);
-  CERT_DestroyCertificate(raw_peer_cert);
-
-  NSSStreamAdapter *stream = reinterpret_cast<NSSStreamAdapter *>(arg);
-  stream->cert_ok_ = false;
-
-  // Read the peer's certificate chain.
-  CERTCertList* cert_list = SSL_PeerCertificateChain(fd);
-  ASSERT(cert_list != NULL);
-
-  // If the peer provided multiple certificates, check that they form a valid
-  // chain as defined by RFC 5246 Section 7.4.2: "Each following certificate
-  // MUST directly certify the one preceding it.".  This check does NOT
-  // verify other requirements, such as whether the chain reaches a trusted
-  // root, self-signed certificates have valid signatures, certificates are not
-  // expired, etc.
-  // Even if the chain is valid, the leaf certificate must still match a
-  // provided certificate or digest.
-  if (!NSSCertificate::IsValidChain(cert_list)) {
-    CERT_DestroyCertList(cert_list);
-    PORT_SetError(SEC_ERROR_BAD_SIGNATURE);
-    return SECFailure;
-  }
-
-  if (stream->peer_certificate_.get()) {
-    LOG(LS_INFO) << "Checking against specified certificate";
-
-    // The peer certificate was specified
-    if (reinterpret_cast<NSSCertificate *>(stream->peer_certificate_.get())->
-        Equals(&peer_cert)) {
-      LOG(LS_INFO) << "Accepted peer certificate";
-      stream->cert_ok_ = true;
-    }
-  } else if (!stream->peer_certificate_digest_algorithm_.empty()) {
-    LOG(LS_INFO) << "Checking against specified digest";
-    // The peer certificate digest was specified
-    unsigned char digest[64];  // Maximum size
-    size_t digest_length;
-
-    if (!peer_cert.ComputeDigest(
-            stream->peer_certificate_digest_algorithm_,
-            digest, sizeof(digest), &digest_length)) {
-      LOG(LS_ERROR) << "Digest computation failed";
-    } else {
-      Buffer computed_digest(digest, digest_length);
-      if (computed_digest == stream->peer_certificate_digest_value_) {
-        LOG(LS_INFO) << "Accepted peer certificate";
-        stream->cert_ok_ = true;
-      }
-    }
-  } else {
-    // Other modes, but we haven't implemented yet
-    // TODO(ekr@rtfm.com): Implement real certificate validation
-    UNIMPLEMENTED;
-  }
-
-  if (!stream->cert_ok_ && stream->ignore_bad_cert()) {
-    LOG(LS_WARNING) << "Ignoring cert error while verifying cert chain";
-    stream->cert_ok_ = true;
-  }
-
-  if (stream->cert_ok_)
-    stream->peer_certificate_.reset(new NSSCertificate(cert_list));
-
-  CERT_DestroyCertList(cert_list);
-
-  if (stream->cert_ok_)
-    return SECSuccess;
-
-  PORT_SetError(SEC_ERROR_UNTRUSTED_CERT);
-  return SECFailure;
-}
-
-
-SECStatus NSSStreamAdapter::GetClientAuthDataHook(void *arg, PRFileDesc *fd,
-                                                  CERTDistNames *caNames,
-                                                  CERTCertificate **pRetCert,
-                                                  SECKEYPrivateKey **pRetKey) {
-  LOG(LS_INFO) << "Client cert requested";
-  NSSStreamAdapter *stream = reinterpret_cast<NSSStreamAdapter *>(arg);
-
-  if (!stream->identity_.get()) {
-    LOG(LS_ERROR) << "No identity available";
-    return SECFailure;
-  }
-
-  NSSIdentity *identity = static_cast<NSSIdentity *>(stream->identity_.get());
-  // Destroyed internally by NSS
-  *pRetCert = CERT_DupCertificate(identity->certificate().certificate());
-  *pRetKey = SECKEY_CopyPrivateKey(identity->keypair()->privkey());
-
-  return SECSuccess;
-}
-
-bool NSSStreamAdapter::GetSslCipher(std::string* cipher) {
-  ASSERT(state_ == SSL_CONNECTED);
-  if (state_ != SSL_CONNECTED)
-    return false;
-
-  SSLChannelInfo channel_info;
-  SECStatus rv = SSL_GetChannelInfo(ssl_fd_, &channel_info,
-                                    sizeof(channel_info));
-  if (rv == SECFailure)
-    return false;
-
-  SSLCipherSuiteInfo ciphersuite_info;
-  rv = SSL_GetCipherSuiteInfo(channel_info.cipherSuite, &ciphersuite_info,
-                              sizeof(ciphersuite_info));
-  if (rv == SECFailure)
-    return false;
-
-  *cipher = ciphersuite_info.cipherSuiteName;
-  return true;
-}
-
-// RFC 5705 Key Exporter
-bool NSSStreamAdapter::ExportKeyingMaterial(const std::string& label,
-                                            const uint8* context,
-                                            size_t context_len,
-                                            bool use_context,
-                                            uint8* result,
-                                            size_t result_len) {
-  SECStatus rv = SSL_ExportKeyingMaterial(
-      ssl_fd_,
-      label.c_str(),
-      checked_cast<unsigned int>(label.size()),
-      use_context,
-      context,
-      checked_cast<unsigned int>(context_len),
-      result,
-      checked_cast<unsigned int>(result_len));
-
-  return rv == SECSuccess;
-}
-
-bool NSSStreamAdapter::SetDtlsSrtpCiphers(
-    const std::vector<std::string>& ciphers) {
-#ifdef HAVE_DTLS_SRTP
-  std::vector<PRUint16> internal_ciphers;
-  if (state_ != SSL_NONE)
-    return false;
-
-  for (std::vector<std::string>::const_iterator cipher = ciphers.begin();
-       cipher != ciphers.end(); ++cipher) {
-    bool found = false;
-    for (const SrtpCipherMapEntry *entry = kSrtpCipherMap; entry->cipher_id;
-         ++entry) {
-      if (*cipher == entry->external_name) {
-        found = true;
-        internal_ciphers.push_back(entry->cipher_id);
-        break;
-      }
-    }
-
-    if (!found) {
-      LOG(LS_ERROR) << "Could not find cipher: " << *cipher;
-      return false;
-    }
-  }
-
-  if (internal_ciphers.empty())
-    return false;
-
-  srtp_ciphers_ = internal_ciphers;
-
-  return true;
-#else
-  return false;
-#endif
-}
-
-bool NSSStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
-#ifdef HAVE_DTLS_SRTP
-  ASSERT(state_ == SSL_CONNECTED);
-  if (state_ != SSL_CONNECTED)
-    return false;
-
-  PRUint16 selected_cipher;
-
-  SECStatus rv = SSL_GetSRTPCipher(ssl_fd_, &selected_cipher);
-  if (rv == SECFailure)
-    return false;
-
-  for (const SrtpCipherMapEntry *entry = kSrtpCipherMap;
-       entry->cipher_id; ++entry) {
-    if (selected_cipher == entry->cipher_id) {
-      *cipher = entry->external_name;
-      return true;
-    }
-  }
-
-  ASSERT(false);  // This should never happen
-#endif
-  return false;
-}
-
-
-GlobalLockPod NSSContext::lock;
-NSSContext *NSSContext::global_nss_context;
-
-// Static initialization and shutdown
-NSSContext *NSSContext::Instance() {
-  lock.Lock();
-  if (!global_nss_context) {
-    scoped_ptr<NSSContext> new_ctx(new NSSContext(PK11_GetInternalSlot()));
-    if (new_ctx->slot_)
-      global_nss_context = new_ctx.release();
-  }
-  lock.Unlock();
-
-  return global_nss_context;
-}
-
-bool NSSContext::InitializeSSL(VerificationCallback callback) {
-  ASSERT(!callback);
-
-  static bool initialized = false;
-
-  if (!initialized) {
-    SECStatus rv;
-
-    rv = NSS_NoDB_Init(NULL);
-    if (rv != SECSuccess) {
-      LOG(LS_ERROR) << "Couldn't initialize NSS error=" <<
-          PORT_GetError();
-      return false;
-    }
-
-    NSS_SetDomesticPolicy();
-
-    initialized = true;
-  }
-
-  return true;
-}
-
-bool NSSContext::InitializeSSLThread() {
-  // Not needed
-  return true;
-}
-
-bool NSSContext::CleanupSSL() {
-  // Not needed
-  return true;
-}
-
-bool NSSStreamAdapter::HaveDtls() {
-  return true;
-}
-
-bool NSSStreamAdapter::HaveDtlsSrtp() {
-#ifdef HAVE_DTLS_SRTP
-  return true;
-#else
-  return false;
-#endif
-}
-
-bool NSSStreamAdapter::HaveExporter() {
-  return true;
-}
-
-std::string NSSStreamAdapter::GetDefaultSslCipher() {
-  return kDefaultSslCipher;
-}
-
-}  // namespace rtc
-
-#endif  // HAVE_NSS_SSL_H
diff --git a/media/webrtc/trunk/webrtc/base/nssstreamadapter.h b/media/webrtc/trunk/webrtc/base/nssstreamadapter.h
deleted file mode 100644
index fcacb95398..0000000000
--- a/media/webrtc/trunk/webrtc/base/nssstreamadapter.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_NSSSTREAMADAPTER_H_
-#define WEBRTC_BASE_NSSSTREAMADAPTER_H_
-
-#include <string>
-#include <vector>
-
-#include "nspr.h"
-#include "nss.h"
-#include "secmodt.h"
-
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/nssidentity.h"
-#include "webrtc/base/ssladapter.h"
-#include "webrtc/base/sslstreamadapter.h"
-#include "webrtc/base/sslstreamadapterhelper.h"
-
-namespace rtc {
-
-// Singleton
-class NSSContext {
- public:
-  explicit NSSContext(PK11SlotInfo* slot) : slot_(slot) {}
-  ~NSSContext() {
-  }
-
-  static PK11SlotInfo *GetSlot() {
-    return Instance() ? Instance()->slot_: NULL;
-  }
-
-  static NSSContext *Instance();
-  static bool InitializeSSL(VerificationCallback callback);
-  static bool InitializeSSLThread();
-  static bool CleanupSSL();
-
- private:
-  PK11SlotInfo *slot_;                    // The PKCS-11 slot
-  static GlobalLockPod lock;              // To protect the global context
-  static NSSContext *global_nss_context;  // The global context
-};
-
-
-class NSSStreamAdapter : public SSLStreamAdapterHelper {
- public:
-  explicit NSSStreamAdapter(StreamInterface* stream);
-  ~NSSStreamAdapter() override;
-  bool Init();
-
-  StreamResult Read(void* data,
-                    size_t data_len,
-                    size_t* read,
-                    int* error) override;
-  StreamResult Write(const void* data,
-                     size_t data_len,
-                     size_t* written,
-                     int* error) override;
-  void OnMessage(Message* msg) override;
-
-  bool GetSslCipher(std::string* cipher) override;
-
-  // Key Extractor interface
-  bool ExportKeyingMaterial(const std::string& label,
-                            const uint8* context,
-                            size_t context_len,
-                            bool use_context,
-                            uint8* result,
-                            size_t result_len) override;
-
-  // DTLS-SRTP interface
-  bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers) override;
-  bool GetDtlsSrtpCipher(std::string* cipher) override;
-
-  // Capabilities interfaces
-  static bool HaveDtls();
-  static bool HaveDtlsSrtp();
-  static bool HaveExporter();
-  static std::string GetDefaultSslCipher();
-
- protected:
-  // Override SSLStreamAdapter
-  void OnEvent(StreamInterface* stream, int events, int err) override;
-
-  // Override SSLStreamAdapterHelper
-  int BeginSSL() override;
-  void Cleanup() override;
-  bool GetDigestLength(const std::string& algorithm, size_t* length) override;
-
- private:
-  int ContinueSSL();
-  static SECStatus AuthCertificateHook(void *arg, PRFileDesc *fd,
-                                       PRBool checksig, PRBool isServer);
-  static SECStatus GetClientAuthDataHook(void *arg, PRFileDesc *fd,
-                                         CERTDistNames *caNames,
-                                         CERTCertificate **pRetCert,
-                                         SECKEYPrivateKey **pRetKey);
-
-  PRFileDesc *ssl_fd_;              // NSS's SSL file descriptor
-  static bool initialized;          // Was InitializeSSL() called?
-  bool cert_ok_;                    // Did we get and check a cert
-  std::vector<PRUint16> srtp_ciphers_;  // SRTP cipher list
-
-  static PRDescIdentity nspr_layer_identity;  // The NSPR layer identity
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_NSSSTREAMADAPTER_H_
diff --git a/media/webrtc/trunk/webrtc/base/nullsocketserver_unittest.cc b/media/webrtc/trunk/webrtc/base/nullsocketserver_unittest.cc
index 4bb1d7f8eb..4f22c382d8 100644
--- a/media/webrtc/trunk/webrtc/base/nullsocketserver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/nullsocketserver_unittest.cc
@@ -10,11 +10,10 @@
 
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/nullsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
-static const uint32 kTimeout = 5000U;
+static const uint32_t kTimeout = 5000U;
 
 class NullSocketServerTest
     : public testing::Test,
@@ -38,7 +37,7 @@ TEST_F(NullSocketServerTest, WaitAndSet) {
 }
 
 TEST_F(NullSocketServerTest, TestWait) {
-  uint32 start = Time();
+  uint32_t start = Time();
   ss_.Wait(200, true);
   // The actual wait time is dependent on the resolution of the timer used by
   // the Event class. Allow for the event to signal ~20ms early.
diff --git a/media/webrtc/trunk/webrtc/base/objc/NSString+StdString.h b/media/webrtc/trunk/webrtc/base/objc/NSString+StdString.h
new file mode 100644
index 0000000000..8bf6cc94be
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/NSString+StdString.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include <string>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface NSString (StdString)
+
+@property(nonatomic, readonly) std::string stdString;
+
++ (std::string)stdStringForString:(NSString *)nsString;
++ (NSString *)stringForStdString:(const std::string&)stdString;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/media/webrtc/trunk/webrtc/base/objc/NSString+StdString.mm b/media/webrtc/trunk/webrtc/base/objc/NSString+StdString.mm
new file mode 100644
index 0000000000..3210ff0b65
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/NSString+StdString.mm
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "NSString+StdString.h"
+
+@implementation NSString (StdString)
+
+- (std::string)stdString {
+  return [NSString stdStringForString:self];
+}
+
++ (std::string)stdStringForString:(NSString *)nsString {
+  NSData *charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char *>(charData.bytes),
+                     charData.length);
+}
+
++ (NSString *)stringForStdString:(const std::string&)stdString {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/base/objc/OWNERS b/media/webrtc/trunk/webrtc/base/objc/OWNERS
new file mode 100644
index 0000000000..cd06158b7f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/OWNERS
@@ -0,0 +1 @@
+tkchin@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.h b/media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.h
new file mode 100644
index 0000000000..03e94c29ae
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+@class AVCaptureSession;
+@class RTCAVFoundationVideoSource;
+
+/** RTCCameraPreviewView is a view that renders local video from an
+ *  AVCaptureSession.
+ */
+@interface RTCCameraPreviewView : UIView
+
+/** The capture session being rendered in the view. Capture session
+ *  is assigned to AVCaptureVideoPreviewLayer async in the same
+ *  queue that the AVCaptureSession is started/stopped.
+ */
+@property(nonatomic, strong) AVCaptureSession *captureSession;
+
+@end
diff --git a/media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.m b/media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.m
new file mode 100644
index 0000000000..5a57483676
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.m
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
+
+#import "webrtc/base/objc/RTCCameraPreviewView.h"
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "webrtc/base/objc/RTCDispatcher.h"
+
+@implementation RTCCameraPreviewView
+
+@synthesize captureSession = _captureSession;
+
++ (Class)layerClass {
+  return [AVCaptureVideoPreviewLayer class];
+}
+
+- (void)setCaptureSession:(AVCaptureSession *)captureSession {
+  if (_captureSession == captureSession) {
+    return;
+  }
+  _captureSession = captureSession;
+  AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
+  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                               block:^{
+    previewLayer.session = captureSession;
+  }];
+}
+
+#pragma mark - Private
+
+- (AVCaptureVideoPreviewLayer *)previewLayer {
+  return (AVCaptureVideoPreviewLayer *)self.layer;
+}
+
+@end
diff --git a/media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.h b/media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.h
new file mode 100644
index 0000000000..c32b93d472
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
+  // Main dispatcher queue.
+  RTCDispatcherTypeMain,
+  // Used for starting/stopping AVCaptureSession, and assigning
+  // capture session to AVCaptureVideoPreviewLayer.
+  RTCDispatcherTypeCaptureSession,
+};
+
+/** Dispatcher that asynchronously dispatches blocks to a specific
+ *  shared dispatch queue.
+ */
+@interface RTCDispatcher : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Dispatch the block asynchronously on the queue for dispatchType.
+ *  @param dispatchType The queue type to dispatch on.
+ *  @param block The block to dispatch asynchronously.
+ */
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
+                      block:(dispatch_block_t)block;
+
+@end
diff --git a/media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.m b/media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.m
new file mode 100644
index 0000000000..065705a4ae
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.m
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDispatcher.h"
+
+static dispatch_queue_t kCaptureSessionQueue = nil;
+
+@implementation RTCDispatcher {
+  dispatch_queue_t _captureSessionQueue;
+}
+
++ (void)initialize {
+  static dispatch_once_t onceToken;
+  dispatch_once(&onceToken, ^{
+    kCaptureSessionQueue = dispatch_queue_create(
+        "org.webrtc.RTCDispatcherCaptureSession",
+        DISPATCH_QUEUE_SERIAL);
+  });
+}
+
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
+                      block:(dispatch_block_t)block {
+  dispatch_queue_t queue = [self dispatchQueueForType:dispatchType];
+  dispatch_async(queue, block);
+}
+
+#pragma mark - Private
+
++ (dispatch_queue_t)dispatchQueueForType:(RTCDispatcherQueueType)dispatchType {
+  switch (dispatchType) {
+    case RTCDispatcherTypeMain:
+      return dispatch_get_main_queue();
+    case RTCDispatcherTypeCaptureSession:
+      return kCaptureSessionQueue;
+  }
+}
+
+@end
+
diff --git a/media/webrtc/trunk/webrtc/base/objc/RTCLogging.h b/media/webrtc/trunk/webrtc/base/objc/RTCLogging.h
new file mode 100644
index 0000000000..19fade5cfc
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/RTCLogging.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+// Subset of rtc::LoggingSeverity.
+typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
+  kRTCLoggingSeverityVerbose,
+  kRTCLoggingSeverityInfo,
+  kRTCLoggingSeverityWarning,
+  kRTCLoggingSeverityError,
+};
+
+#if defined(__cplusplus)
+extern "C" void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+extern "C" void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+extern "C" NSString* RTCFileName(const char* filePath);
+#else
+
+// Wrapper for C++ LOG(sev) macros.
+// Logs the log string to the webrtc logstream for the given severity.
+extern void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+
+// Wrapper for rtc::LogMessage::LogToDebug.
+// Sets the minimum severity to be logged to console.
+extern void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+
+// Returns the filename with the path prefix removed.
+extern NSString* RTCFileName(const char* filePath);
+
+#endif
+
+// Some convenience macros.
+
+#define RTCLogString(format, ...)                    \
+  [NSString stringWithFormat:@"(%@:%d %s): " format, \
+      RTCFileName(__FILE__),                         \
+      __LINE__,                                      \
+      __FUNCTION__,                                  \
+      ##__VA_ARGS__]
+
+#define RTCLogFormat(severity, format, ...)                     \
+  do {                                                          \
+    NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
+    RTCLogEx(severity, log_string);                             \
+  } while (false)
+
+#define RTCLogVerbose(format, ...)                                \
+  RTCLogFormat(kRTCLoggingSeverityVerbose, format, ##__VA_ARGS__) \
+
+#define RTCLogInfo(format, ...)                                   \
+  RTCLogFormat(kRTCLoggingSeverityInfo, format, ##__VA_ARGS__)    \
+
+#define RTCLogWarning(format, ...)                                \
+  RTCLogFormat(kRTCLoggingSeverityWarning, format, ##__VA_ARGS__) \
+
+#define RTCLogError(format, ...)                                  \
+  RTCLogFormat(kRTCLoggingSeverityError, format, ##__VA_ARGS__)   \
+
+#if !defined(NDEBUG)
+#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
+#else
+#define RTCLogDebug(format, ...) \
+  do {                           \
+  } while (false)
+#endif
+
+#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/media/webrtc/trunk/webrtc/base/objc/RTCLogging.mm b/media/webrtc/trunk/webrtc/base/objc/RTCLogging.mm
new file mode 100644
index 0000000000..e9afe725d1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/objc/RTCLogging.mm
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCLogging.h"
+
+#include "webrtc/base/logging.h"
+
+rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
+  switch (severity) {
+    case kRTCLoggingSeverityVerbose:
+      return rtc::LS_VERBOSE;
+    case kRTCLoggingSeverityInfo:
+      return rtc::LS_INFO;
+    case kRTCLoggingSeverityWarning:
+      return rtc::LS_WARNING;
+    case kRTCLoggingSeverityError:
+      return rtc::LS_ERROR;
+  }
+}
+
+void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string) {
+  if (log_string.length) {
+    const char* utf8_string = log_string.UTF8String;
+    LOG_V(RTCGetNativeLoggingSeverity(severity)) << utf8_string;
+  }
+}
+
+void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity) {
+  rtc::LogMessage::LogToDebug(RTCGetNativeLoggingSeverity(severity));
+}
+
+NSString* RTCFileName(const char* file_path) {
+  NSString* ns_file_path =
+      [[NSString alloc] initWithBytesNoCopy:const_cast<char*>(file_path)
+                                     length:strlen(file_path)
+                                   encoding:NSUTF8StringEncoding
+                               freeWhenDone:NO];
+  return ns_file_path.lastPathComponent;
+}
+
diff --git a/media/webrtc/trunk/webrtc/base/openssladapter.cc b/media/webrtc/trunk/webrtc/base/openssladapter.cc
index 56610fa240..1f5fbbc4d1 100644
--- a/media/webrtc/trunk/webrtc/base/openssladapter.cc
+++ b/media/webrtc/trunk/webrtc/base/openssladapter.cc
@@ -31,6 +31,7 @@
 #include "config.h"
 #endif  // HAVE_CONFIG_H
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/openssl.h"
@@ -39,6 +40,8 @@
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 
+#ifndef OPENSSL_IS_BORINGSSL
+
 // TODO: Use a nicer abstraction for mutex.
 
 #if defined(WEBRTC_WIN)
@@ -63,6 +66,8 @@ struct CRYPTO_dynlock_value {
   MUTEX_TYPE mutex;
 };
 
+#endif  // #ifndef OPENSSL_IS_BORINGSSL
+
 //////////////////////////////////////////////////////////////////////
 // SocketBIO
 //////////////////////////////////////////////////////////////////////
@@ -172,6 +177,8 @@ static long socket_ctrl(BIO* b, int cmd, long num, void* ptr) {
 
 namespace rtc {
 
+#ifndef OPENSSL_IS_BORINGSSL
+
 // This array will store all of the mutexes available to OpenSSL.
 static MUTEX_TYPE* mutex_buf = NULL;
 
@@ -213,6 +220,8 @@ static void dyn_destroy_function(CRYPTO_dynlock_value* l,
   delete l;
 }
 
+#endif  // #ifndef OPENSSL_IS_BORINGSSL
+
 VerificationCallback OpenSSLAdapter::custom_verify_callback_ = NULL;
 
 bool OpenSSLAdapter::InitializeSSL(VerificationCallback callback) {
@@ -230,6 +239,9 @@ bool OpenSSLAdapter::InitializeSSL(VerificationCallback callback) {
 }
 
 bool OpenSSLAdapter::InitializeSSLThread() {
+  // BoringSSL is doing the locking internally, so the callbacks are not used
+  // in this case (and are no-ops anyways).
+#ifndef OPENSSL_IS_BORINGSSL
   mutex_buf = new MUTEX_TYPE[CRYPTO_num_locks()];
   if (!mutex_buf)
     return false;
@@ -243,10 +255,12 @@ bool OpenSSLAdapter::InitializeSSLThread() {
   CRYPTO_set_dynlock_create_callback(dyn_create_function);
   CRYPTO_set_dynlock_lock_callback(dyn_lock_function);
   CRYPTO_set_dynlock_destroy_callback(dyn_destroy_function);
+#endif  // #ifndef OPENSSL_IS_BORINGSSL
   return true;
 }
 
 bool OpenSSLAdapter::CleanupSSL() {
+#ifndef OPENSSL_IS_BORINGSSL
   if (!mutex_buf)
     return false;
   CRYPTO_set_id_callback(NULL);
@@ -258,6 +272,7 @@ bool OpenSSLAdapter::CleanupSSL() {
     MUTEX_CLEANUP(mutex_buf[i]);
   delete [] mutex_buf;
   mutex_buf = NULL;
+#endif  // #ifndef OPENSSL_IS_BORINGSSL
   return true;
 }
 
@@ -821,7 +836,7 @@ bool OpenSSLAdapter::SSLPostConnectionCheck(SSL* ssl, const char* host) {
   return ok;
 }
 
-#if _DEBUG
+#if !defined(NDEBUG)
 
 // We only use this for tracing and so it is only needed in debug mode
 
@@ -850,11 +865,11 @@ OpenSSLAdapter::SSLInfoCallback(const SSL* s, int where, int ret) {
   }
 }
 
-#endif  // _DEBUG
+#endif
 
 int
 OpenSSLAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
-#if _DEBUG
+#if !defined(NDEBUG)
   if (!ok) {
     char data[256];
     X509* cert = X509_STORE_CTX_get_current_cert(store);
@@ -901,7 +916,7 @@ OpenSSLAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
 bool OpenSSLAdapter::ConfigureTrustedRootCertificates(SSL_CTX* ctx) {
   // Add the root cert that we care about to the SSL context
   int count_of_added_certs = 0;
-  for (int i = 0; i < ARRAY_SIZE(kSSLCertCertificateList); i++) {
+  for (size_t i = 0; i < arraysize(kSSLCertCertificateList); i++) {
     const unsigned char* cert_buffer = kSSLCertCertificateList[i];
     size_t cert_buffer_len = kSSLCertCertificateSizeList[i];
     X509* cert = d2i_X509(NULL, &cert_buffer,
@@ -935,7 +950,7 @@ OpenSSLAdapter::SetupSSLContext() {
     return NULL;
   }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   SSL_CTX_set_info_callback(ctx, SSLInfoCallback);
 #endif
 
diff --git a/media/webrtc/trunk/webrtc/base/openssladapter.h b/media/webrtc/trunk/webrtc/base/openssladapter.h
index 3dcb1c5645..cdf45e603f 100644
--- a/media/webrtc/trunk/webrtc/base/openssladapter.h
+++ b/media/webrtc/trunk/webrtc/base/openssladapter.h
@@ -67,9 +67,9 @@ private:
   static bool VerifyServerName(SSL* ssl, const char* host,
                                bool ignore_bad_cert);
   bool SSLPostConnectionCheck(SSL* ssl, const char* host);
-#if _DEBUG
+#if !defined(NDEBUG)
   static void SSLInfoCallback(const SSL* s, int where, int ret);
-#endif  // !_DEBUG
+#endif
   static int SSLVerifyCallback(int ok, X509_STORE_CTX* store);
   static VerificationCallback custom_verify_callback_;
   friend class OpenSSLStreamAdapter;  // for custom_verify_callback_;
diff --git a/media/webrtc/trunk/webrtc/base/opensslidentity.cc b/media/webrtc/trunk/webrtc/base/opensslidentity.cc
index 3932680d4d..7185571102 100644
--- a/media/webrtc/trunk/webrtc/base/opensslidentity.cc
+++ b/media/webrtc/trunk/webrtc/base/opensslidentity.cc
@@ -33,9 +33,6 @@ namespace rtc {
 // We could have exposed a myriad of parameters for the crypto stuff,
 // but keeping it simple seems best.
 
-// Strength of generated keys. Those are RSA.
-static const int KEY_LENGTH = 1024;
-
 // Random bits for certificate serial number
 static const int SERIAL_RAND_BITS = 64;
 
@@ -46,23 +43,48 @@ static const int CERTIFICATE_LIFETIME = 60*60*24*30;  // 30 days, arbitrarily
 static const int CERTIFICATE_WINDOW = -60*60*24;
 
 // Generate a key pair. Caller is responsible for freeing the returned object.
-static EVP_PKEY* MakeKey() {
+static EVP_PKEY* MakeKey(const KeyParams& key_params) {
   LOG(LS_INFO) << "Making key pair";
   EVP_PKEY* pkey = EVP_PKEY_new();
-  // RSA_generate_key is deprecated. Use _ex version.
-  BIGNUM* exponent = BN_new();
-  RSA* rsa = RSA_new();
-  if (!pkey || !exponent || !rsa ||
-      !BN_set_word(exponent, 0x10001) ||  // 65537 RSA exponent
-      !RSA_generate_key_ex(rsa, KEY_LENGTH, exponent, NULL) ||
-      !EVP_PKEY_assign_RSA(pkey, rsa)) {
-    EVP_PKEY_free(pkey);
+  if (key_params.type() == KT_RSA) {
+    int key_length = key_params.rsa_params().mod_size;
+    BIGNUM* exponent = BN_new();
+    RSA* rsa = RSA_new();
+    if (!pkey || !exponent || !rsa ||
+        !BN_set_word(exponent, key_params.rsa_params().pub_exp) ||
+        !RSA_generate_key_ex(rsa, key_length, exponent, NULL) ||
+        !EVP_PKEY_assign_RSA(pkey, rsa)) {
+      EVP_PKEY_free(pkey);
+      BN_free(exponent);
+      RSA_free(rsa);
+      LOG(LS_ERROR) << "Failed to make RSA key pair";
+      return NULL;
+    }
+    // ownership of rsa struct was assigned, don't free it.
     BN_free(exponent);
-    RSA_free(rsa);
+  } else if (key_params.type() == KT_ECDSA) {
+    if (key_params.ec_curve() == EC_NIST_P256) {
+      EC_KEY* ec_key = EC_KEY_new_by_curve_name(NID_X9_62_prime256v1);
+      if (!pkey || !ec_key || !EC_KEY_generate_key(ec_key) ||
+          !EVP_PKEY_assign_EC_KEY(pkey, ec_key)) {
+        EVP_PKEY_free(pkey);
+        EC_KEY_free(ec_key);
+        LOG(LS_ERROR) << "Failed to make EC key pair";
+        return NULL;
+      }
+      // ownership of ec_key struct was assigned, don't free it.
+    } else {
+      // Add generation of any other curves here.
+      EVP_PKEY_free(pkey);
+      LOG(LS_ERROR) << "ECDSA key requested for unknown curve";
+      return NULL;
+    }
+  } else {
+    EVP_PKEY_free(pkey);
+    LOG(LS_ERROR) << "Key type requested not understood";
     return NULL;
   }
-  // ownership of rsa struct was assigned, don't free it.
-  BN_free(exponent);
+
   LOG(LS_INFO) << "Returning key pair";
   return pkey;
 }
@@ -74,6 +96,7 @@ static X509* MakeCertificate(EVP_PKEY* pkey, const SSLIdentityParams& params) {
   X509* x509 = NULL;
   BIGNUM* serial_number = NULL;
   X509_NAME* name = NULL;
+  time_t epoch_off = 0;  // Time offset since epoch.
 
   if ((x509=X509_new()) == NULL)
     goto error;
@@ -108,11 +131,11 @@ static X509* MakeCertificate(EVP_PKEY* pkey, const SSLIdentityParams& params) {
       !X509_set_issuer_name(x509, name))
     goto error;
 
-  if (!X509_gmtime_adj(X509_get_notBefore(x509), params.not_before) ||
-      !X509_gmtime_adj(X509_get_notAfter(x509), params.not_after))
+  if (!X509_time_adj(X509_get_notBefore(x509), params.not_before, &epoch_off) ||
+      !X509_time_adj(X509_get_notAfter(x509), params.not_after, &epoch_off))
     goto error;
 
-  if (!X509_sign(x509, pkey, EVP_sha1()))
+  if (!X509_sign(x509, pkey, EVP_sha256()))
     goto error;
 
   BN_free(serial_number);
@@ -138,8 +161,8 @@ static void LogSSLErrors(const std::string& prefix) {
   }
 }
 
-OpenSSLKeyPair* OpenSSLKeyPair::Generate() {
-  EVP_PKEY* pkey = MakeKey();
+OpenSSLKeyPair* OpenSSLKeyPair::Generate(const KeyParams& key_params) {
+  EVP_PKEY* pkey = MakeKey(key_params);
   if (!pkey) {
     LogSSLErrors("Generating key pair");
     return NULL;
@@ -157,10 +180,14 @@ OpenSSLKeyPair* OpenSSLKeyPair::GetReference() {
 }
 
 void OpenSSLKeyPair::AddReference() {
+#if defined(OPENSSL_IS_BORINGSSL)
+  EVP_PKEY_up_ref(pkey_);
+#else
   CRYPTO_add(&pkey_->references, 1, CRYPTO_LOCK_EVP_PKEY);
+#endif
 }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 // Print a certificate to the log, for debugging.
 static void PrintCert(X509* x509) {
   BIO* temp_memory_bio = BIO_new(BIO_s_mem());
@@ -189,7 +216,7 @@ OpenSSLCertificate* OpenSSLCertificate::Generate(
     LogSSLErrors("Generating certificate");
     return NULL;
   }
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   PrintCert(x509);
 #endif
   OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
@@ -203,8 +230,7 @@ OpenSSLCertificate* OpenSSLCertificate::FromPEMString(
   if (!bio)
     return NULL;
   BIO_set_mem_eof_return(bio, 0);
-  X509 *x509 = PEM_read_bio_X509(bio, NULL, NULL,
-                                 const_cast<char*>("\0"));
+  X509* x509 = PEM_read_bio_X509(bio, NULL, NULL, const_cast<char*>("\0"));
   BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
 
   if (!x509)
@@ -279,7 +305,7 @@ bool OpenSSLCertificate::ComputeDigest(const X509* x509,
                                        unsigned char* digest,
                                        size_t size,
                                        size_t* length) {
-  const EVP_MD *md;
+  const EVP_MD* md;
   unsigned int n;
 
   if (!OpenSSLDigest::GetDigestEVP(algorithm, &md))
@@ -322,7 +348,7 @@ std::string OpenSSLCertificate::ToPEMString() const {
 
 void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
   // In case of failure, make sure to leave the buffer empty.
-  der_buffer->SetData(NULL, 0);
+  der_buffer->SetSize(0);
 
   // Calculates the DER representation of the certificate, from scratch.
   BIO* bio = BIO_new(BIO_s_mem());
@@ -341,7 +367,27 @@ void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
 
 void OpenSSLCertificate::AddReference() const {
   ASSERT(x509_ != NULL);
+#if defined(OPENSSL_IS_BORINGSSL)
+  X509_up_ref(x509_);
+#else
   CRYPTO_add(&x509_->references, 1, CRYPTO_LOCK_X509);
+#endif
+}
+
+// Documented in sslidentity.h.
+int64_t OpenSSLCertificate::CertificateExpirationTime() const {
+  ASN1_TIME* expire_time = X509_get_notAfter(x509_);
+  bool long_format;
+
+  if (expire_time->type == V_ASN1_UTCTIME) {
+    long_format = false;
+  } else if (expire_time->type == V_ASN1_GENERALIZEDTIME) {
+    long_format = true;
+  } else {
+    return -1;
+  }
+
+  return ASN1TimeToSec(expire_time->data, expire_time->length, long_format);
 }
 
 OpenSSLIdentity::OpenSSLIdentity(OpenSSLKeyPair* key_pair,
@@ -355,10 +401,10 @@ OpenSSLIdentity::~OpenSSLIdentity() = default;
 
 OpenSSLIdentity* OpenSSLIdentity::GenerateInternal(
     const SSLIdentityParams& params) {
-  OpenSSLKeyPair *key_pair = OpenSSLKeyPair::Generate();
+  OpenSSLKeyPair* key_pair = OpenSSLKeyPair::Generate(params.key_params);
   if (key_pair) {
-    OpenSSLCertificate *certificate = OpenSSLCertificate::Generate(
-        key_pair, params);
+    OpenSSLCertificate* certificate =
+        OpenSSLCertificate::Generate(key_pair, params);
     if (certificate)
       return new OpenSSLIdentity(key_pair, certificate);
     delete key_pair;
@@ -367,11 +413,14 @@ OpenSSLIdentity* OpenSSLIdentity::GenerateInternal(
   return NULL;
 }
 
-OpenSSLIdentity* OpenSSLIdentity::Generate(const std::string& common_name) {
+OpenSSLIdentity* OpenSSLIdentity::Generate(const std::string& common_name,
+                                           const KeyParams& key_params) {
   SSLIdentityParams params;
+  params.key_params = key_params;
   params.common_name = common_name;
-  params.not_before = CERTIFICATE_WINDOW;
-  params.not_after = CERTIFICATE_LIFETIME;
+  time_t now = time(NULL);
+  params.not_before = now + CERTIFICATE_WINDOW;
+  params.not_after = now + CERTIFICATE_LIFETIME;
   return GenerateInternal(params);
 }
 
@@ -396,8 +445,8 @@ SSLIdentity* OpenSSLIdentity::FromPEMStrings(
     return NULL;
   }
   BIO_set_mem_eof_return(bio, 0);
-  EVP_PKEY *pkey = PEM_read_bio_PrivateKey(bio, NULL, NULL,
-                                           const_cast<char*>("\0"));
+  EVP_PKEY* pkey =
+      PEM_read_bio_PrivateKey(bio, NULL, NULL, const_cast<char*>("\0"));
   BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
 
   if (!pkey) {
diff --git a/media/webrtc/trunk/webrtc/base/opensslidentity.h b/media/webrtc/trunk/webrtc/base/opensslidentity.h
index 71a2c315d4..c8aa69a76e 100644
--- a/media/webrtc/trunk/webrtc/base/opensslidentity.h
+++ b/media/webrtc/trunk/webrtc/base/opensslidentity.h
@@ -32,7 +32,7 @@ class OpenSSLKeyPair {
     ASSERT(pkey_ != NULL);
   }
 
-  static OpenSSLKeyPair* Generate();
+  static OpenSSLKeyPair* Generate(const KeyParams& key_params);
 
   virtual ~OpenSSLKeyPair();
 
@@ -45,7 +45,7 @@ class OpenSSLKeyPair {
 
   EVP_PKEY* pkey_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(OpenSSLKeyPair);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLKeyPair);
 };
 
 // OpenSSLCertificate encapsulates an OpenSSL X509* certificate object,
@@ -87,19 +87,22 @@ class OpenSSLCertificate : public SSLCertificate {
   bool GetSignatureDigestAlgorithm(std::string* algorithm) const override;
   bool GetChain(SSLCertChain** chain) const override;
 
+  int64_t CertificateExpirationTime() const override;
+
  private:
   void AddReference() const;
 
   X509* x509_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(OpenSSLCertificate);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLCertificate);
 };
 
 // Holds a keypair and certificate together, and a method to generate
 // them consistently.
 class OpenSSLIdentity : public SSLIdentity {
  public:
-  static OpenSSLIdentity* Generate(const std::string& common_name);
+  static OpenSSLIdentity* Generate(const std::string& common_name,
+                                   const KeyParams& key_params);
   static OpenSSLIdentity* GenerateForTest(const SSLIdentityParams& params);
   static SSLIdentity* FromPEMStrings(const std::string& private_key,
                                      const std::string& certificate);
@@ -119,7 +122,7 @@ class OpenSSLIdentity : public SSLIdentity {
   scoped_ptr<OpenSSLKeyPair> key_pair_;
   scoped_ptr<OpenSSLCertificate> certificate_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(OpenSSLIdentity);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLIdentity);
 };
 
 
diff --git a/media/webrtc/trunk/webrtc/base/opensslstreamadapter.cc b/media/webrtc/trunk/webrtc/base/opensslstreamadapter.cc
index 619f3e1ea4..4b0fe02702 100644
--- a/media/webrtc/trunk/webrtc/base/opensslstreamadapter.cc
+++ b/media/webrtc/trunk/webrtc/base/opensslstreamadapter.cc
@@ -43,21 +43,23 @@ namespace rtc {
 #endif
 
 #ifdef HAVE_DTLS_SRTP
-// SRTP cipher suite table
+// SRTP cipher suite table. |internal_name| is used to construct a
+// colon-separated profile strings which is needed by
+// SSL_CTX_set_tlsext_use_srtp().
 struct SrtpCipherMapEntry {
-  const char* external_name;
   const char* internal_name;
+  const int id;
 };
 
 // This isn't elegant, but it's better than an external reference
 static SrtpCipherMapEntry SrtpCipherMap[] = {
-  {"AES_CM_128_HMAC_SHA1_80", "SRTP_AES128_CM_SHA1_80"},
-  {"AES_CM_128_HMAC_SHA1_32", "SRTP_AES128_CM_SHA1_32"},
-  {NULL, NULL}
-};
+    {"SRTP_AES128_CM_SHA1_80", SRTP_AES128_CM_SHA1_80},
+    {"SRTP_AES128_CM_SHA1_32", SRTP_AES128_CM_SHA1_32},
+    {nullptr, 0}};
 #endif
 
 #ifndef OPENSSL_IS_BORINGSSL
+
 // Cipher name table. Maps internal OpenSSL cipher ids to the RFC name.
 struct SslCipherMapEntry {
   uint32_t openssl_id;
@@ -139,9 +141,41 @@ static const SslCipherMapEntry kSslCipherMap[] = {
 };
 #endif  // #ifndef OPENSSL_IS_BORINGSSL
 
+#if defined(_MSC_VER)
+#pragma warning(push)
+#pragma warning(disable : 4309)
+#pragma warning(disable : 4310)
+#endif  // defined(_MSC_VER)
+
 // Default cipher used between OpenSSL/BoringSSL stream adapters.
 // This needs to be updated when the default of the SSL library changes.
-static const char kDefaultSslCipher[] = "TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA";
+// static_cast<uint16_t> causes build warnings on windows platform.
+static int kDefaultSslCipher10 =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_RSA_WITH_AES_256_CBC_SHA);
+static int kDefaultSslEcCipher10 =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_WITH_AES_256_CBC_SHA);
+#ifdef OPENSSL_IS_BORINGSSL
+static int kDefaultSslCipher12 =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_RSA_WITH_AES_128_GCM_SHA256);
+static int kDefaultSslEcCipher12 =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256);
+// Fallback cipher for DTLS 1.2 if hardware-accelerated AES-GCM is unavailable.
+static int kDefaultSslCipher12NoAesGcm =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256);
+static int kDefaultSslEcCipher12NoAesGcm =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256);
+#else  // !OPENSSL_IS_BORINGSSL
+// OpenSSL sorts differently than BoringSSL, so the default cipher doesn't
+// change between TLS 1.0 and TLS 1.2 with the current setup.
+static int kDefaultSslCipher12 =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_RSA_WITH_AES_256_CBC_SHA);
+static int kDefaultSslEcCipher12 =
+    static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_WITH_AES_256_CBC_SHA);
+#endif  // OPENSSL_IS_BORINGSSL
+
+#if defined(_MSC_VER)
+#pragma warning(pop)
+#endif  // defined(_MSC_VER)
 
 //////////////////////////////////////////////////////////////////////
 // StreamBIO
@@ -246,6 +280,12 @@ static long stream_ctrl(BIO* b, int cmd, long num, void* ptr) {
       return 0;
     case BIO_CTRL_FLUSH:
       return 1;
+    case BIO_CTRL_DGRAM_QUERY_MTU:
+      // openssl defaults to mtu=256 unless we return something here.
+      // The handshake doesn't actually need to send packets above 1k,
+      // so this seems like a sensible value that should work in most cases.
+      // Webrtc uses the same value for video packets.
+      return 1200;
     default:
       return 0;
   }
@@ -259,11 +299,13 @@ OpenSSLStreamAdapter::OpenSSLStreamAdapter(StreamInterface* stream)
     : SSLStreamAdapter(stream),
       state_(SSL_NONE),
       role_(SSL_CLIENT),
-      ssl_read_needs_write_(false), ssl_write_needs_read_(false),
-      ssl_(NULL), ssl_ctx_(NULL),
+      ssl_read_needs_write_(false),
+      ssl_write_needs_read_(false),
+      ssl_(NULL),
+      ssl_ctx_(NULL),
       custom_verification_succeeded_(false),
-      ssl_mode_(SSL_MODE_TLS) {
-}
+      ssl_mode_(SSL_MODE_TLS),
+      ssl_max_version_(SSL_PROTOCOL_TLS_12) {}
 
 OpenSSLStreamAdapter::~OpenSSLStreamAdapter() {
   Cleanup();
@@ -309,21 +351,28 @@ bool OpenSSLStreamAdapter::SetPeerCertificateDigest(const std::string
   return true;
 }
 
-#ifndef OPENSSL_IS_BORINGSSL
-const char* OpenSSLStreamAdapter::GetRfcSslCipherName(
-    const SSL_CIPHER* cipher) {
-  ASSERT(cipher != NULL);
+std::string OpenSSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
+#ifdef OPENSSL_IS_BORINGSSL
+  const SSL_CIPHER* ssl_cipher = SSL_get_cipher_by_value(cipher_suite);
+  if (!ssl_cipher) {
+    return std::string();
+  }
+  char* cipher_name = SSL_CIPHER_get_rfc_name(ssl_cipher);
+  std::string rfc_name = std::string(cipher_name);
+  OPENSSL_free(cipher_name);
+  return rfc_name;
+#else
   for (const SslCipherMapEntry* entry = kSslCipherMap; entry->rfc_name;
        ++entry) {
-    if (cipher->id == entry->openssl_id) {
+    if (cipher_suite == static_cast<int>(entry->openssl_id)) {
       return entry->rfc_name;
     }
   }
-  return NULL;
-}
+  return std::string();
 #endif
+}
 
-bool OpenSSLStreamAdapter::GetSslCipher(std::string* cipher) {
+bool OpenSSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) {
   if (state_ != SSL_CONNECTED)
     return false;
 
@@ -332,35 +381,22 @@ bool OpenSSLStreamAdapter::GetSslCipher(std::string* cipher) {
     return false;
   }
 
-#ifdef OPENSSL_IS_BORINGSSL
-  char* cipher_name = SSL_CIPHER_get_rfc_name(current_cipher);
-#else
-  const char* cipher_name = GetRfcSslCipherName(current_cipher);
-#endif
-  if (cipher_name == NULL) {
-    return false;
-  }
-
-  *cipher = cipher_name;
-#ifdef OPENSSL_IS_BORINGSSL
-  OPENSSL_free(cipher_name);
-#endif
+  *cipher_suite = static_cast<uint16_t>(SSL_CIPHER_get_id(current_cipher));
   return true;
 }
 
 // Key Extractor interface
 bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
-                                                const uint8* context,
+                                                const uint8_t* context,
                                                 size_t context_len,
                                                 bool use_context,
-                                                uint8* result,
+                                                uint8_t* result,
                                                 size_t result_len) {
 #ifdef HAVE_DTLS_SRTP
   int i;
 
-  i = SSL_export_keying_material(ssl_, result, result_len,
-                                 label.c_str(), label.length(),
-                                 const_cast<uint8 *>(context),
+  i = SSL_export_keying_material(ssl_, result, result_len, label.c_str(),
+                                 label.length(), const_cast<uint8_t*>(context),
                                  context_len, use_context);
 
   if (i != 1)
@@ -372,20 +408,20 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
 #endif
 }
 
-bool OpenSSLStreamAdapter::SetDtlsSrtpCiphers(
-    const std::vector<std::string>& ciphers) {
+bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
+    const std::vector<int>& ciphers) {
 #ifdef HAVE_DTLS_SRTP
   std::string internal_ciphers;
 
   if (state_ != SSL_NONE)
     return false;
 
-  for (std::vector<std::string>::const_iterator cipher = ciphers.begin();
+  for (std::vector<int>::const_iterator cipher = ciphers.begin();
        cipher != ciphers.end(); ++cipher) {
     bool found = false;
-    for (SrtpCipherMapEntry *entry = SrtpCipherMap; entry->internal_name;
+    for (SrtpCipherMapEntry* entry = SrtpCipherMap; entry->internal_name;
          ++entry) {
-      if (*cipher == entry->external_name) {
+      if (*cipher == entry->id) {
         found = true;
         if (!internal_ciphers.empty())
           internal_ciphers += ":";
@@ -410,7 +446,7 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCiphers(
 #endif
 }
 
-bool OpenSSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
+bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
 #ifdef HAVE_DTLS_SRTP
   ASSERT(state_ == SSL_CONNECTED);
   if (state_ != SSL_CONNECTED)
@@ -422,17 +458,9 @@ bool OpenSSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
   if (!srtp_profile)
     return false;
 
-  for (SrtpCipherMapEntry *entry = SrtpCipherMap;
-       entry->internal_name; ++entry) {
-    if (!strcmp(entry->internal_name, srtp_profile->name)) {
-      *cipher = entry->external_name;
-      return true;
-    }
-  }
-
-  ASSERT(false);  // This should never happen
-
-  return false;
+  *crypto_suite = srtp_profile->id;
+  ASSERT(!SrtpCryptoSuiteToName(*crypto_suite).empty());
+  return true;
 #else
   return false;
 #endif
@@ -455,6 +483,11 @@ void OpenSSLStreamAdapter::SetMode(SSLMode mode) {
   ssl_mode_ = mode;
 }
 
+void OpenSSLStreamAdapter::SetMaxProtocolVersion(SSLProtocolVersion version) {
+  ASSERT(ssl_ctx_ == NULL);
+  ssl_max_version_ = version;
+}
+
 //
 // StreamInterface Implementation
 //
@@ -739,6 +772,13 @@ int OpenSSLStreamAdapter::BeginSSL() {
   SSL_set_app_data(ssl_, this);
 
   SSL_set_bio(ssl_, bio, bio);  // the SSL object owns the bio now.
+#ifndef OPENSSL_IS_BORINGSSL
+  if (ssl_mode_ == SSL_MODE_DTLS) {
+    // Enable read-ahead for DTLS so whole packets are read from internal BIO
+    // before parsing. This is done internally by BoringSSL for DTLS.
+    SSL_set_read_ahead(ssl_, 1);
+  }
+#endif
 
   SSL_set_mode(ssl_, SSL_MODE_ENABLE_PARTIAL_WRITE |
                SSL_MODE_ACCEPT_MOVING_WRITE_BUFFER);
@@ -858,22 +898,98 @@ void OpenSSLStreamAdapter::OnMessage(Message* msg) {
 SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
   SSL_CTX *ctx = NULL;
 
-  if (role_ == SSL_CLIENT) {
+#ifdef OPENSSL_IS_BORINGSSL
     ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ?
-        DTLSv1_client_method() : TLSv1_client_method());
-  } else {
-    ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ?
-        DTLSv1_server_method() : TLSv1_server_method());
+        DTLS_method() : TLS_method());
+    // Version limiting for BoringSSL will be done below.
+#else
+  const SSL_METHOD* method;
+  switch (ssl_max_version_) {
+    case SSL_PROTOCOL_TLS_10:
+    case SSL_PROTOCOL_TLS_11:
+      // OpenSSL doesn't support setting min/max versions, so we always use
+      // (D)TLS 1.0 if a max. version below the max. available is requested.
+      if (ssl_mode_ == SSL_MODE_DTLS) {
+        if (role_ == SSL_CLIENT) {
+          method = DTLSv1_client_method();
+        } else {
+          method = DTLSv1_server_method();
+        }
+      } else {
+        if (role_ == SSL_CLIENT) {
+          method = TLSv1_client_method();
+        } else {
+          method = TLSv1_server_method();
+        }
+      }
+      break;
+    case SSL_PROTOCOL_TLS_12:
+    default:
+      if (ssl_mode_ == SSL_MODE_DTLS) {
+#if (OPENSSL_VERSION_NUMBER >= 0x10002000L)
+        // DTLS 1.2 only available starting from OpenSSL 1.0.2
+        if (role_ == SSL_CLIENT) {
+          method = DTLS_client_method();
+        } else {
+          method = DTLS_server_method();
+        }
+#else
+        if (role_ == SSL_CLIENT) {
+          method = DTLSv1_client_method();
+        } else {
+          method = DTLSv1_server_method();
+        }
+#endif
+      } else {
+#if (OPENSSL_VERSION_NUMBER >= 0x10100000L)
+        // New API only available starting from OpenSSL 1.1.0
+        if (role_ == SSL_CLIENT) {
+          method = TLS_client_method();
+        } else {
+          method = TLS_server_method();
+        }
+#else
+        if (role_ == SSL_CLIENT) {
+          method = SSLv23_client_method();
+        } else {
+          method = SSLv23_server_method();
+        }
+#endif
+      }
+      break;
   }
+  ctx = SSL_CTX_new(method);
+#endif  // OPENSSL_IS_BORINGSSL
+
   if (ctx == NULL)
     return NULL;
 
+#ifdef OPENSSL_IS_BORINGSSL
+  SSL_CTX_set_min_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
+      DTLS1_VERSION : TLS1_VERSION);
+  switch (ssl_max_version_) {
+    case SSL_PROTOCOL_TLS_10:
+      SSL_CTX_set_max_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
+          DTLS1_VERSION : TLS1_VERSION);
+      break;
+    case SSL_PROTOCOL_TLS_11:
+      SSL_CTX_set_max_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
+          DTLS1_VERSION : TLS1_1_VERSION);
+      break;
+    case SSL_PROTOCOL_TLS_12:
+    default:
+      SSL_CTX_set_max_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
+          DTLS1_2_VERSION : TLS1_2_VERSION);
+      break;
+  }
+#endif
+
   if (identity_ && !identity_->ConfigureIdentity(ctx)) {
     SSL_CTX_free(ctx);
     return NULL;
   }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   SSL_CTX_set_info_callback(ctx, OpenSSLAdapter::SSLInfoCallback);
 #endif
 
@@ -887,7 +1003,12 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
 
   SSL_CTX_set_verify(ctx, mode, SSLVerifyCallback);
   SSL_CTX_set_verify_depth(ctx, 4);
-  SSL_CTX_set_cipher_list(ctx, "ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH");
+  // Select list of available ciphers. Note that !SHA256 and !SHA384 only
+  // remove HMAC-SHA256 and HMAC-SHA384 cipher suites, not GCM cipher suites
+  // with SHA256 or SHA384 as the handshake hash.
+  // This matches the list of SSLClientSocketOpenSSL in Chromium.
+  SSL_CTX_set_cipher_list(ctx,
+      "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK");
 
 #ifdef HAVE_DTLS_SRTP
   if (!srtp_ciphers_.empty()) {
@@ -919,7 +1040,7 @@ int OpenSSLStreamAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
   // the digest.
   //
   // TODO(jiayl): Verify the chain is a proper chain and report the chain to
-  // |stream->peer_certificate_|, like what NSS does.
+  // |stream->peer_certificate_|.
   if (depth > 0) {
     LOG(LS_INFO) << "Ignored chained certificate at depth " << depth;
     return 1;
@@ -1003,8 +1124,46 @@ bool OpenSSLStreamAdapter::HaveExporter() {
 #endif
 }
 
-std::string OpenSSLStreamAdapter::GetDefaultSslCipher() {
-  return kDefaultSslCipher;
+int OpenSSLStreamAdapter::GetDefaultSslCipherForTest(SSLProtocolVersion version,
+                                                     KeyType key_type) {
+  if (key_type == KT_RSA) {
+    switch (version) {
+      case SSL_PROTOCOL_TLS_10:
+      case SSL_PROTOCOL_TLS_11:
+        return kDefaultSslCipher10;
+      case SSL_PROTOCOL_TLS_12:
+      default:
+#ifdef OPENSSL_IS_BORINGSSL
+        if (EVP_has_aes_hardware()) {
+          return kDefaultSslCipher12;
+        } else {
+          return kDefaultSslCipher12NoAesGcm;
+        }
+#else  // !OPENSSL_IS_BORINGSSL
+        return kDefaultSslCipher12;
+#endif
+    }
+  } else if (key_type == KT_ECDSA) {
+    switch (version) {
+      case SSL_PROTOCOL_TLS_10:
+      case SSL_PROTOCOL_TLS_11:
+        return kDefaultSslEcCipher10;
+      case SSL_PROTOCOL_TLS_12:
+      default:
+#ifdef OPENSSL_IS_BORINGSSL
+        if (EVP_has_aes_hardware()) {
+          return kDefaultSslEcCipher12;
+        } else {
+          return kDefaultSslEcCipher12NoAesGcm;
+        }
+#else  // !OPENSSL_IS_BORINGSSL
+        return kDefaultSslEcCipher12;
+#endif
+    }
+  } else {
+    RTC_NOTREACHED();
+    return kDefaultSslEcCipher12;
+  }
 }
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/opensslstreamadapter.h b/media/webrtc/trunk/webrtc/base/opensslstreamadapter.h
index 6b24c9b891..e57b2a3293 100644
--- a/media/webrtc/trunk/webrtc/base/opensslstreamadapter.h
+++ b/media/webrtc/trunk/webrtc/base/opensslstreamadapter.h
@@ -74,6 +74,7 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
   int StartSSLWithServer(const char* server_name) override;
   int StartSSLWithPeer() override;
   void SetMode(SSLMode mode) override;
+  void SetMaxProtocolVersion(SSLProtocolVersion version) override;
 
   StreamResult Read(void* data,
                     size_t data_len,
@@ -86,30 +87,31 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
   void Close() override;
   StreamState GetState() const override;
 
-#ifndef OPENSSL_IS_BORINGSSL
-  // Return the RFC (5246, 3268, etc.) cipher name for an OpenSSL cipher.
-  static const char* GetRfcSslCipherName(const SSL_CIPHER* cipher);
-#endif
+  // TODO(guoweis): Move this away from a static class method.
+  static std::string SslCipherSuiteToName(int crypto_suite);
 
-  bool GetSslCipher(std::string* cipher) override;
+  bool GetSslCipherSuite(int* cipher) override;
 
   // Key Extractor interface
   bool ExportKeyingMaterial(const std::string& label,
-                            const uint8* context,
+                            const uint8_t* context,
                             size_t context_len,
                             bool use_context,
-                            uint8* result,
+                            uint8_t* result,
                             size_t result_len) override;
 
   // DTLS-SRTP interface
-  bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers) override;
-  bool GetDtlsSrtpCipher(std::string* cipher) override;
+  bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites) override;
+  bool GetDtlsSrtpCryptoSuite(int* crypto_suite) override;
 
   // Capabilities interfaces
   static bool HaveDtls();
   static bool HaveDtlsSrtp();
   static bool HaveExporter();
-  static std::string GetDefaultSslCipher();
+
+  // TODO(guoweis): Move this away from a static class method.
+  static int GetDefaultSslCipherForTest(SSLProtocolVersion version,
+                                        KeyType key_type);
 
  protected:
   void OnEvent(StreamInterface* stream, int events, int err) override;
@@ -201,6 +203,9 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
 
   // Do DTLS or not
   SSLMode ssl_mode_;
+
+  // Max. allowed protocol version
+  SSLProtocolVersion ssl_max_version_;
 };
 
 /////////////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/base/optional.h b/media/webrtc/trunk/webrtc/base/optional.h
new file mode 100644
index 0000000000..b8071e6358
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/optional.h
@@ -0,0 +1,139 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPTIONAL_H_
+#define WEBRTC_BASE_OPTIONAL_H_
+
+#include <algorithm>
+#include <utility>
+
+#include "webrtc/base/checks.h"
+
+namespace rtc {
+
+// Simple std::experimental::optional-wannabe. It either contains a T or not.
+// In order to keep the implementation simple and portable, this implementation
+// actually contains a (default-constructed) T even when it supposedly doesn't
+// contain a value; use e.g. rtc::scoped_ptr<T> instead if that's too
+// expensive.
+//
+// A moved-from Optional<T> may only be destroyed, and assigned to if T allows
+// being assigned to after having been moved from. Specifically, you may not
+// assume that it just doesn't contain a value anymore.
+//
+// Examples of good places to use Optional:
+//
+// - As a class or struct member, when the member doesn't always have a value:
+//     struct Prisoner {
+//       std::string name;
+//       Optional<int> cell_number;  // Empty if not currently incarcerated.
+//     };
+//
+// - As a return value for functions that may fail to return a value on all
+//   allowed inputs. For example, a function that searches an array might
+//   return an Optional<size_t> (the index where it found the element, or
+//   nothing if it didn't find it); and a function that parses numbers might
+//   return Optional<double> (the parsed number, or nothing if parsing failed).
+//
+// Examples of bad places to use Optional:
+//
+// - As a return value for functions that may fail because of disallowed
+//   inputs. For example, a string length function should not return
+//   Optional<size_t> so that it can return nothing in case the caller passed
+//   it a null pointer; the function should probably use RTC_[D]CHECK instead,
+//   and return plain size_t.
+//
+// - As a return value for functions that may fail to return a value on all
+//   allowed inputs, but need to tell the caller what went wrong. Returning
+//   Optional<double> when parsing a single number as in the example above
+//   might make sense, but any larger parse job is probably going to need to
+//   tell the caller what the problem was, not just that there was one.
+//
+// TODO(kwiberg): Get rid of this class when the standard library has
+// std::optional (and we're allowed to use it).
+template <typename T>
+class Optional final {
+ public:
+  // Construct an empty Optional.
+  Optional() : has_value_(false) {}
+
+  // Construct an Optional that contains a value.
+  explicit Optional(const T& val) : value_(val), has_value_(true) {}
+  explicit Optional(T&& val) : value_(std::move(val)), has_value_(true) {}
+
+  // Copy and move constructors.
+  // TODO(kwiberg): =default the move constructor when MSVC supports it.
+  Optional(const Optional&) = default;
+  Optional(Optional&& m)
+      : value_(std::move(m.value_)), has_value_(m.has_value_) {}
+
+  // Assignment.
+  // TODO(kwiberg): =default the move assignment op when MSVC supports it.
+  Optional& operator=(const Optional&) = default;
+  Optional& operator=(Optional&& m) {
+    value_ = std::move(m.value_);
+    has_value_ = m.has_value_;
+    return *this;
+  }
+
+  friend void swap(Optional& m1, Optional& m2) {
+    using std::swap;
+    swap(m1.value_, m2.value_);
+    swap(m1.has_value_, m2.has_value_);
+  }
+
+  // Conversion to bool to test if we have a value.
+  explicit operator bool() const { return has_value_; }
+
+  // Dereferencing. Only allowed if we have a value.
+  const T* operator->() const {
+    RTC_DCHECK(has_value_);
+    return &value_;
+  }
+  T* operator->() {
+    RTC_DCHECK(has_value_);
+    return &value_;
+  }
+  const T& operator*() const {
+    RTC_DCHECK(has_value_);
+    return value_;
+  }
+  T& operator*() {
+    RTC_DCHECK(has_value_);
+    return value_;
+  }
+
+  // Dereference with a default value in case we don't have a value.
+  const T& value_or(const T& default_val) const {
+    return has_value_ ? value_ : default_val;
+  }
+
+  // Equality tests. Two Optionals are equal if they contain equivalent values,
+  // or
+  // if they're both empty.
+  friend bool operator==(const Optional& m1, const Optional& m2) {
+    return m1.has_value_ && m2.has_value_ ? m1.value_ == m2.value_
+                                          : m1.has_value_ == m2.has_value_;
+  }
+  friend bool operator!=(const Optional& m1, const Optional& m2) {
+    return m1.has_value_ && m2.has_value_ ? m1.value_ != m2.value_
+                                          : m1.has_value_ != m2.has_value_;
+  }
+
+ private:
+  // Invariant: Unless *this has been moved from, value_ is default-initialized
+  // (or copied or moved from a default-initialized T) if !has_value_.
+  T value_;
+  bool has_value_;
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_OPTIONAL_H_
diff --git a/media/webrtc/trunk/webrtc/base/optional_unittest.cc b/media/webrtc/trunk/webrtc/base/optional_unittest.cc
new file mode 100644
index 0000000000..eabf091e17
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/optional_unittest.cc
@@ -0,0 +1,489 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <sstream>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/optional.h"
+
+namespace rtc {
+
+namespace {
+
+// Class whose instances logs various method calls (constructor, destructor,
+// etc.). Each instance has a unique ID (a simple global sequence number) and
+// an origin ID. When a copy is made, the new object gets a fresh ID but copies
+// the origin ID from the original. When a new Logger is created from scratch,
+// it gets a fresh ID, and the origin ID is the same as the ID (default
+// constructor) or given as an argument (explicit constructor).
+class Logger {
+ public:
+  Logger() : id_(next_id_++), origin_(id_) { Log("default constructor"); }
+  explicit Logger(int origin) : id_(next_id_++), origin_(origin) {
+    Log("explicit constructor");
+  }
+  Logger(const Logger& other) : id_(next_id_++), origin_(other.origin_) {
+    LogFrom("copy constructor", other);
+  }
+  Logger(Logger&& other) : id_(next_id_++), origin_(other.origin_) {
+    LogFrom("move constructor", other);
+  }
+  ~Logger() { Log("destructor"); }
+  Logger& operator=(const Logger& other) {
+    origin_ = other.origin_;
+    LogFrom("operator= copy", other);
+    return *this;
+  }
+  Logger& operator=(Logger&& other) {
+    origin_ = other.origin_;
+    LogFrom("operator= move", other);
+    return *this;
+  }
+  friend void swap(Logger& a, Logger& b) {
+    using std::swap;
+    swap(a.origin_, b.origin_);
+    Log2("swap", a, b);
+  }
+  friend bool operator==(const Logger& a, const Logger& b) {
+    Log2("operator==", a, b);
+    return a.origin_ == b.origin_;
+  }
+  friend bool operator!=(const Logger& a, const Logger& b) {
+    Log2("operator!=", a, b);
+    return a.origin_ != b.origin_;
+  }
+  void Foo() { Log("Foo()"); }
+  void Foo() const { Log("Foo() const"); }
+  static rtc::scoped_ptr<std::vector<std::string>> Setup() {
+    auto s = rtc_make_scoped_ptr(new std::vector<std::string>);
+    Logger::log_ = s.get();
+    Logger::next_id_ = 0;
+    return s;
+  }
+
+ private:
+  int id_;
+  int origin_;
+  static std::vector<std::string>* log_;
+  static int next_id_;
+  void Log(const char* msg) const {
+    std::ostringstream oss;
+    oss << id_ << ':' << origin_ << ". " << msg;
+    log_->push_back(oss.str());
+  }
+  void LogFrom(const char* msg, const Logger& other) const {
+    std::ostringstream oss;
+    oss << id_ << ':' << origin_ << ". " << msg << " (from " << other.id_ << ':'
+        << other.origin_ << ")";
+    log_->push_back(oss.str());
+  }
+  static void Log2(const char* msg, const Logger& a, const Logger& b) {
+    std::ostringstream oss;
+    oss << msg << ' ' << a.id_ << ':' << a.origin_ << ", " << b.id_ << ':'
+        << b.origin_;
+    log_->push_back(oss.str());
+  }
+};
+
+std::vector<std::string>* Logger::log_ = nullptr;
+int Logger::next_id_ = 0;
+
+// Append all the other args to the vector pointed to by the first arg.
+template <typename T>
+void VectorAppend(std::vector<T>* v) {}
+template <typename T, typename... Ts>
+void VectorAppend(std::vector<T>* v, const T& e, Ts... es) {
+  v->push_back(e);
+  VectorAppend(v, es...);
+}
+
+// Create a vector of strings. Because we're not allowed to use
+// std::initializer_list.
+template <typename... Ts>
+std::vector<std::string> V(Ts... es) {
+  std::vector<std::string> strings;
+  VectorAppend(&strings, static_cast<std::string>(es)...);
+  return strings;
+}
+
+}  // namespace
+
+TEST(OptionalTest, TestConstructDefault) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    EXPECT_FALSE(x);
+  }
+  EXPECT_EQ(V("0:0. default constructor", "0:0. destructor"), *log);
+}
+
+TEST(OptionalTest, TestConstructCopyEmpty) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    EXPECT_FALSE(x);
+    auto y = x;
+    EXPECT_FALSE(y);
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
+              "1:0. destructor", "0:0. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestConstructCopyFull) {
+  auto log = Logger::Setup();
+  {
+    Logger a;
+    Optional<Logger> x(a);
+    EXPECT_TRUE(x);
+    log->push_back("---");
+    auto y = x;
+    EXPECT_TRUE(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
+              "---", "2:0. copy constructor (from 1:0)", "---",
+              "2:0. destructor", "1:0. destructor", "0:0. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestConstructMoveEmpty) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    EXPECT_FALSE(x);
+    auto y = std::move(x);
+    EXPECT_FALSE(y);
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:0. move constructor (from 0:0)",
+              "1:0. destructor", "0:0. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestConstructMoveFull) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    EXPECT_TRUE(x);
+    log->push_back("---");
+    auto y = std::move(x);
+    EXPECT_TRUE(x);
+    EXPECT_TRUE(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "---", "2:17. move constructor (from 1:17)", "---",
+        "2:17. destructor", "1:17. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToEmptyFromEmpty) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x, y;
+    x = y;
+  }
+  EXPECT_EQ(
+      V("0:0. default constructor", "1:1. default constructor",
+        "0:1. operator= copy (from 1:1)", "1:1. destructor", "0:1. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToFullFromEmpty) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    Optional<Logger> y;
+    log->push_back("---");
+    x = y;
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "2:2. default constructor", "---",
+        "1:2. operator= copy (from 2:2)", "---", "2:2. destructor",
+        "1:2. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToEmptyFromFull) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    Optional<Logger> y(Logger(17));
+    log->push_back("---");
+    x = y;
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
+              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
+              "0:17. operator= copy (from 2:17)", "---", "2:17. destructor",
+              "0:17. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToFullFromFull) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    Optional<Logger> y(Logger(42));
+    log->push_back("---");
+    x = y;
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "2:42. explicit constructor",
+        "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
+        "1:42. operator= copy (from 3:42)", "---", "3:42. destructor",
+        "1:42. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToEmptyFromT) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    Logger y(17);
+    log->push_back("---");
+    x = Optional<Logger>(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
+              "2:17. copy constructor (from 1:17)",
+              "0:17. operator= move (from 2:17)", "2:17. destructor", "---",
+              "1:17. destructor", "0:17. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToFullFromT) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    Logger y(42);
+    log->push_back("---");
+    x = Optional<Logger>(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "2:42. explicit constructor", "---",
+        "3:42. copy constructor (from 2:42)",
+        "1:42. operator= move (from 3:42)", "3:42. destructor", "---",
+        "2:42. destructor", "1:42. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToEmptyFromEmpty) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x, y;
+    x = std::move(y);
+  }
+  EXPECT_EQ(
+      V("0:0. default constructor", "1:1. default constructor",
+        "0:1. operator= move (from 1:1)", "1:1. destructor", "0:1. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToFullFromEmpty) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    Optional<Logger> y;
+    log->push_back("---");
+    x = std::move(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "2:2. default constructor", "---",
+        "1:2. operator= move (from 2:2)", "---", "2:2. destructor",
+        "1:2. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToEmptyFromFull) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    Optional<Logger> y(Logger(17));
+    log->push_back("---");
+    x = std::move(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
+              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
+              "0:17. operator= move (from 2:17)", "---", "2:17. destructor",
+              "0:17. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToFullFromFull) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    Optional<Logger> y(Logger(42));
+    log->push_back("---");
+    x = std::move(y);
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "2:42. explicit constructor",
+        "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
+        "1:42. operator= move (from 3:42)", "---", "3:42. destructor",
+        "1:42. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToEmptyFromT) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x;
+    Logger y(17);
+    log->push_back("---");
+    x = Optional<Logger>(std::move(y));
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
+              "2:17. move constructor (from 1:17)",
+              "0:17. operator= move (from 2:17)", "2:17. destructor", "---",
+              "1:17. destructor", "0:17. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToFullFromT) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(17));
+    Logger y(42);
+    log->push_back("---");
+    x = Optional<Logger>(std::move(y));
+    log->push_back("---");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "2:42. explicit constructor", "---",
+        "3:42. move constructor (from 2:42)",
+        "1:42. operator= move (from 3:42)", "3:42. destructor", "---",
+        "2:42. destructor", "1:42. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestDereference) {
+  auto log = Logger::Setup();
+  {
+    Optional<Logger> x(Logger(42));
+    const auto& y = x;
+    log->push_back("---");
+    x->Foo();
+    y->Foo();
+    std::move(x)->Foo();
+    std::move(y)->Foo();
+    log->push_back("---");
+    (*x).Foo();
+    (*y).Foo();
+    (*std::move(x)).Foo();
+    (*std::move(y)).Foo();
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:42. explicit constructor",
+              "1:42. move constructor (from 0:42)", "0:42. destructor", "---",
+              "1:42. Foo()", "1:42. Foo() const", "1:42. Foo()",
+              "1:42. Foo() const", "---", "1:42. Foo()", "1:42. Foo() const",
+              "1:42. Foo()", "1:42. Foo() const", "---", "1:42. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestDereferenceWithDefault) {
+  auto log = Logger::Setup();
+  {
+    const Logger a(17), b(42);
+    Optional<Logger> x(a);
+    Optional<Logger> y;
+    log->push_back("-1-");
+    EXPECT_EQ(a, x.value_or(Logger(42)));
+    log->push_back("-2-");
+    EXPECT_EQ(b, y.value_or(Logger(42)));
+    log->push_back("-3-");
+    EXPECT_EQ(a, Optional<Logger>(Logger(17)).value_or(b));
+    log->push_back("-4-");
+    EXPECT_EQ(b, Optional<Logger>().value_or(b));
+    log->push_back("-5-");
+  }
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:42. explicit constructor",
+        "2:17. copy constructor (from 0:17)", "3:3. default constructor", "-1-",
+        "4:42. explicit constructor", "operator== 0:17, 2:17",
+        "4:42. destructor", "-2-", "5:42. explicit constructor",
+        "operator== 1:42, 5:42", "5:42. destructor", "-3-",
+        "6:17. explicit constructor", "7:17. move constructor (from 6:17)",
+        "operator== 0:17, 7:17", "7:17. destructor", "6:17. destructor", "-4-",
+        "8:8. default constructor", "operator== 1:42, 1:42", "8:8. destructor",
+        "-5-", "3:3. destructor", "2:17. destructor", "1:42. destructor",
+        "0:17. destructor"),
+      *log);
+}
+
+TEST(OptionalTest, TestEquality) {
+  auto log = Logger::Setup();
+  {
+    Logger a(17), b(42);
+    Optional<Logger> ma1(a), ma2(a), mb(b), me1, me2;
+    log->push_back("---");
+    EXPECT_EQ(ma1, ma1);
+    EXPECT_EQ(ma1, ma2);
+    EXPECT_NE(ma1, mb);
+    EXPECT_NE(ma1, me1);
+    EXPECT_EQ(me1, me1);
+    EXPECT_EQ(me1, me2);
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
+              "2:17. copy constructor (from 0:17)",
+              "3:17. copy constructor (from 0:17)",
+              "4:42. copy constructor (from 1:42)", "5:5. default constructor",
+              "6:6. default constructor", "---", "operator== 2:17, 2:17",
+              "operator== 2:17, 3:17", "operator!= 2:17, 4:42", "---",
+              "6:6. destructor", "5:5. destructor", "4:42. destructor",
+              "3:17. destructor", "2:17. destructor", "1:42. destructor",
+              "0:17. destructor"),
+            *log);
+}
+
+TEST(OptionalTest, TestSwap) {
+  auto log = Logger::Setup();
+  {
+    Logger a(17), b(42);
+    Optional<Logger> x1(a), x2(b), y1(a), y2, z1, z2;
+    log->push_back("---");
+    swap(x1, x2);  // Swap full <-> full.
+    swap(y1, y2);  // Swap full <-> empty.
+    swap(z1, z2);  // Swap empty <-> empty.
+    log->push_back("---");
+  }
+  EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
+              "2:17. copy constructor (from 0:17)",
+              "3:42. copy constructor (from 1:42)",
+              "4:17. copy constructor (from 0:17)", "5:5. default constructor",
+              "6:6. default constructor", "7:7. default constructor", "---",
+              "swap 2:42, 3:17", "swap 4:5, 5:17", "swap 6:7, 7:6", "---",
+              "7:6. destructor", "6:7. destructor", "5:17. destructor",
+              "4:5. destructor", "3:17. destructor", "2:42. destructor",
+              "1:42. destructor", "0:17. destructor"),
+            *log);
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/pathutils.cc b/media/webrtc/trunk/webrtc/base/pathutils.cc
index 7671bfc29f..b5227ecb10 100644
--- a/media/webrtc/trunk/webrtc/base/pathutils.cc
+++ b/media/webrtc/trunk/webrtc/base/pathutils.cc
@@ -225,12 +225,13 @@ bool Pathname::SetFilename(const std::string& filename) {
 }
 
 #if defined(WEBRTC_WIN)
-bool Pathname::GetDrive(char *drive, uint32 bytes) const {
+bool Pathname::GetDrive(char* drive, uint32_t bytes) const {
   return GetDrive(drive, bytes, folder_);
 }
 
 // static
-bool Pathname::GetDrive(char *drive, uint32 bytes,
+bool Pathname::GetDrive(char* drive,
+                        uint32_t bytes,
                         const std::string& pathname) {
   // need at lease 4 bytes to save c:
   if (bytes < 4 || pathname.size() < 3) {
diff --git a/media/webrtc/trunk/webrtc/base/pathutils.h b/media/webrtc/trunk/webrtc/base/pathutils.h
index 8f07e1dbc0..2d5819f1b6 100644
--- a/media/webrtc/trunk/webrtc/base/pathutils.h
+++ b/media/webrtc/trunk/webrtc/base/pathutils.h
@@ -92,8 +92,10 @@ public:
   bool SetFilename(const std::string& filename);
 
 #if defined(WEBRTC_WIN)
-  bool GetDrive(char *drive, uint32 bytes) const;
-  static bool GetDrive(char *drive, uint32 bytes,const std::string& pathname);
+  bool GetDrive(char* drive, uint32_t bytes) const;
+  static bool GetDrive(char* drive,
+                       uint32_t bytes,
+                       const std::string& pathname);
 #endif
 
 private:
diff --git a/media/webrtc/trunk/webrtc/base/physicalsocketserver.cc b/media/webrtc/trunk/webrtc/base/physicalsocketserver.cc
index bc3fb32dd9..3e454527ca 100644
--- a/media/webrtc/trunk/webrtc/base/physicalsocketserver.cc
+++ b/media/webrtc/trunk/webrtc/base/physicalsocketserver.cc
@@ -39,11 +39,11 @@
 #include <algorithm>
 #include <map>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/nethelpers.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/winping.h"
@@ -68,26 +68,26 @@ namespace rtc {
 
 #if defined(WEBRTC_WIN)
 // Standard MTUs, from RFC 1191
-const uint16 PACKET_MAXIMUMS[] = {
-  65535,    // Theoretical maximum, Hyperchannel
-  32000,    // Nothing
-  17914,    // 16Mb IBM Token Ring
-  8166,     // IEEE 802.4
-  //4464,   // IEEE 802.5 (4Mb max)
-  4352,     // FDDI
-  //2048,   // Wideband Network
-  2002,     // IEEE 802.5 (4Mb recommended)
-  //1536,   // Expermental Ethernet Networks
-  //1500,   // Ethernet, Point-to-Point (default)
-  1492,     // IEEE 802.3
-  1006,     // SLIP, ARPANET
-  //576,    // X.25 Networks
-  //544,    // DEC IP Portal
-  //512,    // NETBIOS
-  508,      // IEEE 802/Source-Rt Bridge, ARCNET
-  296,      // Point-to-Point (low delay)
-  68,       // Official minimum
-  0,        // End of list marker
+const uint16_t PACKET_MAXIMUMS[] = {
+    65535,  // Theoretical maximum, Hyperchannel
+    32000,  // Nothing
+    17914,  // 16Mb IBM Token Ring
+    8166,   // IEEE 802.4
+    // 4464,   // IEEE 802.5 (4Mb max)
+    4352,   // FDDI
+    // 2048,   // Wideband Network
+    2002,   // IEEE 802.5 (4Mb recommended)
+    // 1536,   // Expermental Ethernet Networks
+    // 1500,   // Ethernet, Point-to-Point (default)
+    1492,   // IEEE 802.3
+    1006,   // SLIP, ARPANET
+    // 576,    // X.25 Networks
+    // 544,    // DEC IP Portal
+    // 512,    // NETBIOS
+    508,    // IEEE 802/Source-Rt Bridge, ARCNET
+    296,    // Point-to-Point (low delay)
+    68,     // Official minimum
+    0,      // End of list marker
 };
 
 static const int IP_HEADER_SIZE = 20u;
@@ -96,463 +96,669 @@ static const int ICMP_HEADER_SIZE = 8u;
 static const int ICMP_PING_TIMEOUT_MILLIS = 10000u;
 #endif
 
-class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
- public:
-  PhysicalSocket(PhysicalSocketServer* ss, SOCKET s = INVALID_SOCKET)
-    : ss_(ss), s_(s), enabled_events_(0), error_(0),
-      state_((s == INVALID_SOCKET) ? CS_CLOSED : CS_CONNECTED),
-      resolver_(NULL) {
+PhysicalSocket::PhysicalSocket(PhysicalSocketServer* ss, SOCKET s)
+  : ss_(ss), s_(s), enabled_events_(0), error_(0),
+    state_((s == INVALID_SOCKET) ? CS_CLOSED : CS_CONNECTED),
+    resolver_(nullptr) {
 #if defined(WEBRTC_WIN)
-    // EnsureWinsockInit() ensures that winsock is initialized. The default
-    // version of this function doesn't do anything because winsock is
-    // initialized by constructor of a static object. If neccessary libjingle
-    // users can link it with a different version of this function by replacing
-    // win32socketinit.cc. See win32socketinit.cc for more details.
-    EnsureWinsockInit();
+  // EnsureWinsockInit() ensures that winsock is initialized. The default
+  // version of this function doesn't do anything because winsock is
+  // initialized by constructor of a static object. If neccessary libjingle
+  // users can link it with a different version of this function by replacing
+  // win32socketinit.cc. See win32socketinit.cc for more details.
+  EnsureWinsockInit();
 #endif
-    if (s_ != INVALID_SOCKET) {
-      enabled_events_ = DE_READ | DE_WRITE;
+  if (s_ != INVALID_SOCKET) {
+    enabled_events_ = DE_READ | DE_WRITE;
 
-      int type = SOCK_STREAM;
-      socklen_t len = sizeof(type);
-      VERIFY(0 == getsockopt(s_, SOL_SOCKET, SO_TYPE, (SockOptArg)&type, &len));
-      udp_ = (SOCK_DGRAM == type);
-    }
-  }
-
-  ~PhysicalSocket() override {
-    Close();
-  }
-
-  // Creates the underlying OS socket (same as the "socket" function).
-  virtual bool Create(int family, int type) {
-    Close();
-    s_ = ::socket(family, type, 0);
+    int type = SOCK_STREAM;
+    socklen_t len = sizeof(type);
+    VERIFY(0 == getsockopt(s_, SOL_SOCKET, SO_TYPE, (SockOptArg)&type, &len));
     udp_ = (SOCK_DGRAM == type);
-    UpdateLastError();
-    if (udp_)
-      enabled_events_ = DE_READ | DE_WRITE;
-    return s_ != INVALID_SOCKET;
   }
+}
 
-  SocketAddress GetLocalAddress() const override {
-    sockaddr_storage addr_storage = {0};
-    socklen_t addrlen = sizeof(addr_storage);
-    sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
-    int result = ::getsockname(s_, addr, &addrlen);
-    SocketAddress address;
-    if (result >= 0) {
-      SocketAddressFromSockAddrStorage(addr_storage, &address);
-    } else {
-      LOG(LS_WARNING) << "GetLocalAddress: unable to get local addr, socket="
-                      << s_;
-    }
-    return address;
+PhysicalSocket::~PhysicalSocket() {
+  Close();
+}
+
+bool PhysicalSocket::Create(int family, int type) {
+  Close();
+  s_ = ::socket(family, type, 0);
+  udp_ = (SOCK_DGRAM == type);
+  UpdateLastError();
+  if (udp_)
+    enabled_events_ = DE_READ | DE_WRITE;
+  return s_ != INVALID_SOCKET;
+}
+
+SocketAddress PhysicalSocket::GetLocalAddress() const {
+  sockaddr_storage addr_storage = {0};
+  socklen_t addrlen = sizeof(addr_storage);
+  sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+  int result = ::getsockname(s_, addr, &addrlen);
+  SocketAddress address;
+  if (result >= 0) {
+    SocketAddressFromSockAddrStorage(addr_storage, &address);
+  } else {
+    LOG(LS_WARNING) << "GetLocalAddress: unable to get local addr, socket="
+                    << s_;
   }
+  return address;
+}
 
-  SocketAddress GetRemoteAddress() const override {
-    sockaddr_storage addr_storage = {0};
-    socklen_t addrlen = sizeof(addr_storage);
-    sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
-    int result = ::getpeername(s_, addr, &addrlen);
-    SocketAddress address;
-    if (result >= 0) {
-      SocketAddressFromSockAddrStorage(addr_storage, &address);
-    } else {
-      LOG(LS_WARNING) << "GetRemoteAddress: unable to get remote addr, socket="
-                      << s_;
-    }
-    return address;
+SocketAddress PhysicalSocket::GetRemoteAddress() const {
+  sockaddr_storage addr_storage = {0};
+  socklen_t addrlen = sizeof(addr_storage);
+  sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+  int result = ::getpeername(s_, addr, &addrlen);
+  SocketAddress address;
+  if (result >= 0) {
+    SocketAddressFromSockAddrStorage(addr_storage, &address);
+  } else {
+    LOG(LS_WARNING) << "GetRemoteAddress: unable to get remote addr, socket="
+                    << s_;
   }
+  return address;
+}
 
-  int Bind(const SocketAddress& bind_addr) override {
-    sockaddr_storage addr_storage;
-    size_t len = bind_addr.ToSockAddrStorage(&addr_storage);
-    sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
-    int err = ::bind(s_, addr, static_cast<int>(len));
-    UpdateLastError();
-#ifdef _DEBUG
-    if (0 == err) {
-      dbg_addr_ = "Bound @ ";
-      dbg_addr_.append(GetLocalAddress().ToString());
-    }
-#endif  // _DEBUG
-    return err;
+int PhysicalSocket::Bind(const SocketAddress& bind_addr) {
+  sockaddr_storage addr_storage;
+  size_t len = bind_addr.ToSockAddrStorage(&addr_storage);
+  sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+  int err = ::bind(s_, addr, static_cast<int>(len));
+  UpdateLastError();
+#if !defined(NDEBUG)
+  if (0 == err) {
+    dbg_addr_ = "Bound @ ";
+    dbg_addr_.append(GetLocalAddress().ToString());
   }
+#endif
+  return err;
+}
 
-  int Connect(const SocketAddress& addr) override {
-    // TODO: Implicit creation is required to reconnect...
-    // ...but should we make it more explicit?
-    if (state_ != CS_CLOSED) {
-      SetError(EALREADY);
-      return SOCKET_ERROR;
-    }
-    if (addr.IsUnresolved()) {
-      LOG(LS_VERBOSE) << "Resolving addr in PhysicalSocket::Connect";
-      resolver_ = new AsyncResolver();
-      resolver_->SignalDone.connect(this, &PhysicalSocket::OnResolveResult);
-      resolver_->Start(addr);
-      state_ = CS_CONNECTING;
-      return 0;
-    }
-
-    return DoConnect(addr);
+int PhysicalSocket::Connect(const SocketAddress& addr) {
+  // TODO(pthatcher): Implicit creation is required to reconnect...
+  // ...but should we make it more explicit?
+  if (state_ != CS_CLOSED) {
+    SetError(EALREADY);
+    return SOCKET_ERROR;
   }
-
-  int DoConnect(const SocketAddress& connect_addr) {
-    if ((s_ == INVALID_SOCKET) &&
-        !Create(connect_addr.family(), SOCK_STREAM)) {
-      return SOCKET_ERROR;
-    }
-    sockaddr_storage addr_storage;
-    size_t len = connect_addr.ToSockAddrStorage(&addr_storage);
-    sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
-    int err = ::connect(s_, addr, static_cast<int>(len));
-    UpdateLastError();
-    if (err == 0) {
-      state_ = CS_CONNECTED;
-    } else if (IsBlockingError(GetError())) {
-      state_ = CS_CONNECTING;
-      enabled_events_ |= DE_CONNECT;
-    } else {
-      return SOCKET_ERROR;
-    }
-
-    enabled_events_ |= DE_READ | DE_WRITE;
+  if (addr.IsUnresolvedIP()) {
+    LOG(LS_VERBOSE) << "Resolving addr in PhysicalSocket::Connect";
+    resolver_ = new AsyncResolver();
+    resolver_->SignalDone.connect(this, &PhysicalSocket::OnResolveResult);
+    resolver_->Start(addr);
+    state_ = CS_CONNECTING;
     return 0;
   }
 
-  int GetError() const override {
-    CritScope cs(&crit_);
-    return error_;
+  return DoConnect(addr);
+}
+
+int PhysicalSocket::DoConnect(const SocketAddress& connect_addr) {
+  if ((s_ == INVALID_SOCKET) &&
+      !Create(connect_addr.family(), SOCK_STREAM)) {
+    return SOCKET_ERROR;
+  }
+  sockaddr_storage addr_storage;
+  size_t len = connect_addr.ToSockAddrStorage(&addr_storage);
+  sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+  int err = ::connect(s_, addr, static_cast<int>(len));
+  UpdateLastError();
+  if (err == 0) {
+    state_ = CS_CONNECTED;
+  } else if (IsBlockingError(GetError())) {
+    state_ = CS_CONNECTING;
+    enabled_events_ |= DE_CONNECT;
+  } else {
+    return SOCKET_ERROR;
   }
 
-  void SetError(int error) override {
-    CritScope cs(&crit_);
-    error_ = error;
-  }
+  enabled_events_ |= DE_READ | DE_WRITE;
+  return 0;
+}
 
-  ConnState GetState() const override { return state_; }
+int PhysicalSocket::GetError() const {
+  CritScope cs(&crit_);
+  return error_;
+}
 
-  int GetOption(Option opt, int* value) override {
-    int slevel;
-    int sopt;
-    if (TranslateOption(opt, &slevel, &sopt) == -1)
-      return -1;
-    socklen_t optlen = sizeof(*value);
-    int ret = ::getsockopt(s_, slevel, sopt, (SockOptArg)value, &optlen);
-    if (ret != -1 && opt == OPT_DONTFRAGMENT) {
+void PhysicalSocket::SetError(int error) {
+  CritScope cs(&crit_);
+  error_ = error;
+}
+
+AsyncSocket::ConnState PhysicalSocket::GetState() const {
+  return state_;
+}
+
+int PhysicalSocket::GetOption(Option opt, int* value) {
+  int slevel;
+  int sopt;
+  if (TranslateOption(opt, &slevel, &sopt) == -1)
+    return -1;
+  socklen_t optlen = sizeof(*value);
+  int ret = ::getsockopt(s_, slevel, sopt, (SockOptArg)value, &optlen);
+  if (ret != -1 && opt == OPT_DONTFRAGMENT) {
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-      *value = (*value != IP_PMTUDISC_DONT) ? 1 : 0;
+    *value = (*value != IP_PMTUDISC_DONT) ? 1 : 0;
 #endif
-    }
-    return ret;
   }
+  return ret;
+}
 
-  int SetOption(Option opt, int value) override {
-    int slevel;
-    int sopt;
-    if (TranslateOption(opt, &slevel, &sopt) == -1)
-      return -1;
-    if (opt == OPT_DONTFRAGMENT) {
+int PhysicalSocket::SetOption(Option opt, int value) {
+  int slevel;
+  int sopt;
+  if (TranslateOption(opt, &slevel, &sopt) == -1)
+    return -1;
+  if (opt == OPT_DONTFRAGMENT) {
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-      value = (value) ? IP_PMTUDISC_DO : IP_PMTUDISC_DONT;
+    value = (value) ? IP_PMTUDISC_DO : IP_PMTUDISC_DONT;
 #endif
-    }
-    return ::setsockopt(s_, slevel, sopt, (SockOptArg)&value, sizeof(value));
   }
+  return ::setsockopt(s_, slevel, sopt, (SockOptArg)&value, sizeof(value));
+}
 
-  int Send(const void* pv, size_t cb) override {
-    int sent = ::send(s_, reinterpret_cast<const char *>(pv), (int)cb,
+int PhysicalSocket::Send(const void* pv, size_t cb) {
+  int sent = ::send(s_, reinterpret_cast<const char *>(pv), (int)cb,
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-        // Suppress SIGPIPE. Without this, attempting to send on a socket whose
-        // other end is closed will result in a SIGPIPE signal being raised to
-        // our process, which by default will terminate the process, which we
-        // don't want. By specifying this flag, we'll just get the error EPIPE
-        // instead and can handle the error gracefully.
-        MSG_NOSIGNAL
+      // Suppress SIGPIPE. Without this, attempting to send on a socket whose
+      // other end is closed will result in a SIGPIPE signal being raised to
+      // our process, which by default will terminate the process, which we
+      // don't want. By specifying this flag, we'll just get the error EPIPE
+      // instead and can handle the error gracefully.
+      MSG_NOSIGNAL
 #else
-        0
+      0
 #endif
-        );
-    UpdateLastError();
-    MaybeRemapSendError();
-    // We have seen minidumps where this may be false.
-    ASSERT(sent <= static_cast<int>(cb));
-    if ((sent < 0) && IsBlockingError(GetError())) {
-      enabled_events_ |= DE_WRITE;
-    }
-    return sent;
+      );
+  UpdateLastError();
+  MaybeRemapSendError();
+  // We have seen minidumps where this may be false.
+  ASSERT(sent <= static_cast<int>(cb));
+  if ((sent < 0) && IsBlockingError(GetError())) {
+    enabled_events_ |= DE_WRITE;
   }
+  return sent;
+}
 
-  int SendTo(const void* buffer,
-             size_t length,
-             const SocketAddress& addr) override {
-    sockaddr_storage saddr;
-    size_t len = addr.ToSockAddrStorage(&saddr);
-    int sent = ::sendto(
-        s_, static_cast<const char *>(buffer), static_cast<int>(length),
+int PhysicalSocket::SendTo(const void* buffer,
+                           size_t length,
+                           const SocketAddress& addr) {
+  sockaddr_storage saddr;
+  size_t len = addr.ToSockAddrStorage(&saddr);
+  int sent = ::sendto(
+      s_, static_cast<const char *>(buffer), static_cast<int>(length),
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-        // Suppress SIGPIPE. See above for explanation.
-        MSG_NOSIGNAL,
+      // Suppress SIGPIPE. See above for explanation.
+      MSG_NOSIGNAL,
 #else
-        0,
+      0,
 #endif
-        reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(len));
-    UpdateLastError();
-    MaybeRemapSendError();
-    // We have seen minidumps where this may be false.
-    ASSERT(sent <= static_cast<int>(length));
-    if ((sent < 0) && IsBlockingError(GetError())) {
-      enabled_events_ |= DE_WRITE;
-    }
-    return sent;
+      reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(len));
+  UpdateLastError();
+  MaybeRemapSendError();
+  // We have seen minidumps where this may be false.
+  ASSERT(sent <= static_cast<int>(length));
+  if ((sent < 0) && IsBlockingError(GetError())) {
+    enabled_events_ |= DE_WRITE;
   }
+  return sent;
+}
 
-  int Recv(void* buffer, size_t length) override {
-    int received = ::recv(s_, static_cast<char*>(buffer),
-                          static_cast<int>(length), 0);
-    if ((received == 0) && (length != 0)) {
-      // Note: on graceful shutdown, recv can return 0.  In this case, we
-      // pretend it is blocking, and then signal close, so that simplifying
-      // assumptions can be made about Recv.
-      LOG(LS_WARNING) << "EOF from socket; deferring close event";
-      // Must turn this back on so that the select() loop will notice the close
-      // event.
-      enabled_events_ |= DE_READ;
-      SetError(EWOULDBLOCK);
-      return SOCKET_ERROR;
-    }
-    UpdateLastError();
-    int error = GetError();
-    bool success = (received >= 0) || IsBlockingError(error);
-    if (udp_ || success) {
-      enabled_events_ |= DE_READ;
-    }
-    if (!success) {
-      LOG_F(LS_VERBOSE) << "Error = " << error;
-    }
-    return received;
+int PhysicalSocket::Recv(void* buffer, size_t length) {
+  int received = ::recv(s_, static_cast<char*>(buffer),
+                        static_cast<int>(length), 0);
+  if ((received == 0) && (length != 0)) {
+    // Note: on graceful shutdown, recv can return 0.  In this case, we
+    // pretend it is blocking, and then signal close, so that simplifying
+    // assumptions can be made about Recv.
+    LOG(LS_WARNING) << "EOF from socket; deferring close event";
+    // Must turn this back on so that the select() loop will notice the close
+    // event.
+    enabled_events_ |= DE_READ;
+    SetError(EWOULDBLOCK);
+    return SOCKET_ERROR;
   }
-
-  int RecvFrom(void* buffer, size_t length, SocketAddress* out_addr) override {
-    sockaddr_storage addr_storage;
-    socklen_t addr_len = sizeof(addr_storage);
-    sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
-    int received = ::recvfrom(s_, static_cast<char*>(buffer),
-                              static_cast<int>(length), 0, addr, &addr_len);
-    UpdateLastError();
-    if ((received >= 0) && (out_addr != NULL))
-      SocketAddressFromSockAddrStorage(addr_storage, out_addr);
-    int error = GetError();
-    bool success = (received >= 0) || IsBlockingError(error);
-    if (udp_ || success) {
-      enabled_events_ |= DE_READ;
-    }
-    if (!success) {
-      LOG_F(LS_VERBOSE) << "Error = " << error;
-    }
-    return received;
+  UpdateLastError();
+  int error = GetError();
+  bool success = (received >= 0) || IsBlockingError(error);
+  if (udp_ || success) {
+    enabled_events_ |= DE_READ;
   }
-
-  int Listen(int backlog) override {
-    int err = ::listen(s_, backlog);
-    UpdateLastError();
-    if (err == 0) {
-      state_ = CS_CONNECTING;
-      enabled_events_ |= DE_ACCEPT;
-#ifdef _DEBUG
-      dbg_addr_ = "Listening @ ";
-      dbg_addr_.append(GetLocalAddress().ToString());
-#endif  // _DEBUG
-    }
-    return err;
+  if (!success) {
+    LOG_F(LS_VERBOSE) << "Error = " << error;
   }
+  return received;
+}
 
-  AsyncSocket* Accept(SocketAddress* out_addr) override {
-    sockaddr_storage addr_storage;
-    socklen_t addr_len = sizeof(addr_storage);
-    sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
-    SOCKET s = ::accept(s_, addr, &addr_len);
-    UpdateLastError();
-    if (s == INVALID_SOCKET)
-      return NULL;
+int PhysicalSocket::RecvFrom(void* buffer,
+                             size_t length,
+                             SocketAddress* out_addr) {
+  sockaddr_storage addr_storage;
+  socklen_t addr_len = sizeof(addr_storage);
+  sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+  int received = ::recvfrom(s_, static_cast<char*>(buffer),
+                            static_cast<int>(length), 0, addr, &addr_len);
+  UpdateLastError();
+  if ((received >= 0) && (out_addr != nullptr))
+    SocketAddressFromSockAddrStorage(addr_storage, out_addr);
+  int error = GetError();
+  bool success = (received >= 0) || IsBlockingError(error);
+  if (udp_ || success) {
+    enabled_events_ |= DE_READ;
+  }
+  if (!success) {
+    LOG_F(LS_VERBOSE) << "Error = " << error;
+  }
+  return received;
+}
+
+int PhysicalSocket::Listen(int backlog) {
+  int err = ::listen(s_, backlog);
+  UpdateLastError();
+  if (err == 0) {
+    state_ = CS_CONNECTING;
     enabled_events_ |= DE_ACCEPT;
-    if (out_addr != NULL)
-      SocketAddressFromSockAddrStorage(addr_storage, out_addr);
-    return ss_->WrapSocket(s);
+#if !defined(NDEBUG)
+    dbg_addr_ = "Listening @ ";
+    dbg_addr_.append(GetLocalAddress().ToString());
+#endif
   }
+  return err;
+}
 
-  int Close() override {
-    if (s_ == INVALID_SOCKET)
-      return 0;
-    int err = ::closesocket(s_);
-    UpdateLastError();
-    s_ = INVALID_SOCKET;
-    state_ = CS_CLOSED;
-    enabled_events_ = 0;
-    if (resolver_) {
-      resolver_->Destroy(false);
-      resolver_ = NULL;
-    }
-    return err;
+AsyncSocket* PhysicalSocket::Accept(SocketAddress* out_addr) {
+  // Always re-subscribe DE_ACCEPT to make sure new incoming connections will
+  // trigger an event even if DoAccept returns an error here.
+  enabled_events_ |= DE_ACCEPT;
+  sockaddr_storage addr_storage;
+  socklen_t addr_len = sizeof(addr_storage);
+  sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+  SOCKET s = DoAccept(s_, addr, &addr_len);
+  UpdateLastError();
+  if (s == INVALID_SOCKET)
+    return nullptr;
+  if (out_addr != nullptr)
+    SocketAddressFromSockAddrStorage(addr_storage, out_addr);
+  return ss_->WrapSocket(s);
+}
+
+int PhysicalSocket::Close() {
+  if (s_ == INVALID_SOCKET)
+    return 0;
+  int err = ::closesocket(s_);
+  UpdateLastError();
+  s_ = INVALID_SOCKET;
+  state_ = CS_CLOSED;
+  enabled_events_ = 0;
+  if (resolver_) {
+    resolver_->Destroy(false);
+    resolver_ = nullptr;
   }
+  return err;
+}
 
-  int EstimateMTU(uint16* mtu) override {
-    SocketAddress addr = GetRemoteAddress();
-    if (addr.IsAny()) {
-      SetError(ENOTCONN);
-      return -1;
-    }
+int PhysicalSocket::EstimateMTU(uint16_t* mtu) {
+  SocketAddress addr = GetRemoteAddress();
+  if (addr.IsAnyIP()) {
+    SetError(ENOTCONN);
+    return -1;
+  }
 
 #if defined(WEBRTC_WIN)
-    // Gets the interface MTU (TTL=1) for the interface used to reach |addr|.
-    WinPing ping;
-    if (!ping.IsValid()) {
+  // Gets the interface MTU (TTL=1) for the interface used to reach |addr|.
+  WinPing ping;
+  if (!ping.IsValid()) {
+    SetError(EINVAL);  // can't think of a better error ID
+    return -1;
+  }
+  int header_size = ICMP_HEADER_SIZE;
+  if (addr.family() == AF_INET6) {
+    header_size += IPV6_HEADER_SIZE;
+  } else if (addr.family() == AF_INET) {
+    header_size += IP_HEADER_SIZE;
+  }
+
+  for (int level = 0; PACKET_MAXIMUMS[level + 1] > 0; ++level) {
+    int32_t size = PACKET_MAXIMUMS[level] - header_size;
+    WinPing::PingResult result = ping.Ping(addr.ipaddr(), size,
+                                           ICMP_PING_TIMEOUT_MILLIS,
+                                           1, false);
+    if (result == WinPing::PING_FAIL) {
       SetError(EINVAL);  // can't think of a better error ID
       return -1;
+    } else if (result != WinPing::PING_TOO_LARGE) {
+      *mtu = PACKET_MAXIMUMS[level];
+      return 0;
     }
-    int header_size = ICMP_HEADER_SIZE;
-    if (addr.family() == AF_INET6) {
-      header_size += IPV6_HEADER_SIZE;
-    } else if (addr.family() == AF_INET) {
-      header_size += IP_HEADER_SIZE;
-    }
+  }
 
-    for (int level = 0; PACKET_MAXIMUMS[level + 1] > 0; ++level) {
-      int32 size = PACKET_MAXIMUMS[level] - header_size;
-      WinPing::PingResult result = ping.Ping(addr.ipaddr(), size,
-                                             ICMP_PING_TIMEOUT_MILLIS,
-                                             1, false);
-      if (result == WinPing::PING_FAIL) {
-        SetError(EINVAL);  // can't think of a better error ID
-        return -1;
-      } else if (result != WinPing::PING_TOO_LARGE) {
-        *mtu = PACKET_MAXIMUMS[level];
-        return 0;
-      }
-    }
-
-    ASSERT(false);
-    return -1;
+  ASSERT(false);
+  return -1;
 #elif defined(WEBRTC_MAC)
-    // No simple way to do this on Mac OS X.
-    // SIOCGIFMTU would work if we knew which interface would be used, but
-    // figuring that out is pretty complicated. For now we'll return an error
-    // and let the caller pick a default MTU.
-    SetError(EINVAL);
-    return -1;
+  // No simple way to do this on Mac OS X.
+  // SIOCGIFMTU would work if we knew which interface would be used, but
+  // figuring that out is pretty complicated. For now we'll return an error
+  // and let the caller pick a default MTU.
+  SetError(EINVAL);
+  return -1;
 #elif defined(WEBRTC_LINUX)
-    // Gets the path MTU.
-    int value;
-    socklen_t vlen = sizeof(value);
-    int err = getsockopt(s_, IPPROTO_IP, IP_MTU, &value, &vlen);
-    if (err < 0) {
-      UpdateLastError();
-      return err;
-    }
+  // Gets the path MTU.
+  int value;
+  socklen_t vlen = sizeof(value);
+  int err = getsockopt(s_, IPPROTO_IP, IP_MTU, &value, &vlen);
+  if (err < 0) {
+    UpdateLastError();
+    return err;
+  }
 
-    ASSERT((0 <= value) && (value <= 65536));
-    *mtu = value;
-    return 0;
+  ASSERT((0 <= value) && (value <= 65536));
+  *mtu = value;
+  return 0;
 #elif defined(__native_client__)
-    // Most socket operations, including this, will fail in NaCl's sandbox.
-    error_ = EACCES;
-    return -1;
+  // Most socket operations, including this, will fail in NaCl's sandbox.
+  error_ = EACCES;
+  return -1;
 #endif
+}
+
+
+SOCKET PhysicalSocket::DoAccept(SOCKET socket,
+                                sockaddr* addr,
+                                socklen_t* addrlen) {
+  return ::accept(socket, addr, addrlen);
+}
+
+void PhysicalSocket::OnResolveResult(AsyncResolverInterface* resolver) {
+  if (resolver != resolver_) {
+    return;
   }
 
-  SocketServer* socketserver() { return ss_; }
-
- protected:
-  void OnResolveResult(AsyncResolverInterface* resolver) {
-    if (resolver != resolver_) {
-      return;
-    }
-
-    int error = resolver_->GetError();
-    if (error == 0) {
-      error = DoConnect(resolver_->address());
-    } else {
-      Close();
-    }
-
-    if (error) {
-      SetError(error);
-      SignalCloseEvent(this, error);
-    }
+  int error = resolver_->GetError();
+  if (error == 0) {
+    error = DoConnect(resolver_->address());
+  } else {
+    Close();
   }
 
-  void UpdateLastError() {
-    SetError(LAST_SYSTEM_ERROR);
+  if (error) {
+    SetError(error);
+    SignalCloseEvent(this, error);
   }
+}
 
-  void MaybeRemapSendError() {
+void PhysicalSocket::UpdateLastError() {
+  SetError(LAST_SYSTEM_ERROR);
+}
+
+void PhysicalSocket::MaybeRemapSendError() {
 #if defined(WEBRTC_MAC)
-    // https://developer.apple.com/library/mac/documentation/Darwin/
-    // Reference/ManPages/man2/sendto.2.html
-    // ENOBUFS - The output queue for a network interface is full.
-    // This generally indicates that the interface has stopped sending,
-    // but may be caused by transient congestion.
-    if (GetError() == ENOBUFS) {
-      SetError(EWOULDBLOCK);
-    }
-#endif
+  // https://developer.apple.com/library/mac/documentation/Darwin/
+  // Reference/ManPages/man2/sendto.2.html
+  // ENOBUFS - The output queue for a network interface is full.
+  // This generally indicates that the interface has stopped sending,
+  // but may be caused by transient congestion.
+  if (GetError() == ENOBUFS) {
+    SetError(EWOULDBLOCK);
   }
+#endif
+}
 
-  static int TranslateOption(Option opt, int* slevel, int* sopt) {
-    switch (opt) {
-      case OPT_DONTFRAGMENT:
+int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
+  switch (opt) {
+    case OPT_DONTFRAGMENT:
 #if defined(WEBRTC_WIN)
-        *slevel = IPPROTO_IP;
-        *sopt = IP_DONTFRAGMENT;
-        break;
+      *slevel = IPPROTO_IP;
+      *sopt = IP_DONTFRAGMENT;
+      break;
 #elif defined(WEBRTC_MAC) || defined(BSD) || defined(__native_client__)
-        LOG(LS_WARNING) << "Socket::OPT_DONTFRAGMENT not supported.";
-        return -1;
+      LOG(LS_WARNING) << "Socket::OPT_DONTFRAGMENT not supported.";
+      return -1;
 #elif defined(WEBRTC_POSIX)
-        *slevel = IPPROTO_IP;
-        *sopt = IP_MTU_DISCOVER;
-        break;
+      *slevel = IPPROTO_IP;
+      *sopt = IP_MTU_DISCOVER;
+      break;
 #endif
-      case OPT_RCVBUF:
-        *slevel = SOL_SOCKET;
-        *sopt = SO_RCVBUF;
-        break;
-      case OPT_SNDBUF:
-        *slevel = SOL_SOCKET;
-        *sopt = SO_SNDBUF;
-        break;
-      case OPT_NODELAY:
-        *slevel = IPPROTO_TCP;
-        *sopt = TCP_NODELAY;
-        break;
-      case OPT_DSCP:
-        LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
-        return -1;
-      case OPT_RTP_SENDTIME_EXTN_ID:
-        return -1;  // No logging is necessary as this not a OS socket option.
-      default:
-        ASSERT(false);
-        return -1;
-    }
-    return 0;
+    case OPT_RCVBUF:
+      *slevel = SOL_SOCKET;
+      *sopt = SO_RCVBUF;
+      break;
+    case OPT_SNDBUF:
+      *slevel = SOL_SOCKET;
+      *sopt = SO_SNDBUF;
+      break;
+    case OPT_NODELAY:
+      *slevel = IPPROTO_TCP;
+      *sopt = TCP_NODELAY;
+      break;
+    case OPT_DSCP:
+      LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
+      return -1;
+    case OPT_RTP_SENDTIME_EXTN_ID:
+      return -1;  // No logging is necessary as this not a OS socket option.
+    default:
+      ASSERT(false);
+      return -1;
   }
+  return 0;
+}
 
-  PhysicalSocketServer* ss_;
-  SOCKET s_;
-  uint8 enabled_events_;
-  bool udp_;
-  int error_;
-  // Protects |error_| that is accessed from different threads.
-  mutable CriticalSection crit_;
-  ConnState state_;
-  AsyncResolver* resolver_;
+SocketDispatcher::SocketDispatcher(PhysicalSocketServer *ss)
+#if defined(WEBRTC_WIN)
+  : PhysicalSocket(ss), id_(0), signal_close_(false)
+#else
+  : PhysicalSocket(ss)
+#endif
+{
+}
 
-#ifdef _DEBUG
-  std::string dbg_addr_;
-#endif  // _DEBUG;
-};
+SocketDispatcher::SocketDispatcher(SOCKET s, PhysicalSocketServer *ss)
+#if defined(WEBRTC_WIN)
+  : PhysicalSocket(ss, s), id_(0), signal_close_(false)
+#else
+  : PhysicalSocket(ss, s)
+#endif
+{
+}
+
+SocketDispatcher::~SocketDispatcher() {
+  Close();
+}
+
+bool SocketDispatcher::Initialize() {
+  ASSERT(s_ != INVALID_SOCKET);
+  // Must be a non-blocking
+#if defined(WEBRTC_WIN)
+  u_long argp = 1;
+  ioctlsocket(s_, FIONBIO, &argp);
+#elif defined(WEBRTC_POSIX)
+  fcntl(s_, F_SETFL, fcntl(s_, F_GETFL, 0) | O_NONBLOCK);
+#endif
+  ss_->Add(this);
+  return true;
+}
+
+bool SocketDispatcher::Create(int type) {
+  return Create(AF_INET, type);
+}
+
+bool SocketDispatcher::Create(int family, int type) {
+  // Change the socket to be non-blocking.
+  if (!PhysicalSocket::Create(family, type))
+    return false;
+
+  if (!Initialize())
+    return false;
+
+#if defined(WEBRTC_WIN)
+  do { id_ = ++next_id_; } while (id_ == 0);
+#endif
+  return true;
+}
+
+#if defined(WEBRTC_WIN)
+
+WSAEVENT SocketDispatcher::GetWSAEvent() {
+  return WSA_INVALID_EVENT;
+}
+
+SOCKET SocketDispatcher::GetSocket() {
+  return s_;
+}
+
+bool SocketDispatcher::CheckSignalClose() {
+  if (!signal_close_)
+    return false;
+
+  char ch;
+  if (recv(s_, &ch, 1, MSG_PEEK) > 0)
+    return false;
+
+  state_ = CS_CLOSED;
+  signal_close_ = false;
+  SignalCloseEvent(this, signal_err_);
+  return true;
+}
+
+int SocketDispatcher::next_id_ = 0;
+
+#elif defined(WEBRTC_POSIX)
+
+int SocketDispatcher::GetDescriptor() {
+  return s_;
+}
+
+bool SocketDispatcher::IsDescriptorClosed() {
+  // We don't have a reliable way of distinguishing end-of-stream
+  // from readability.  So test on each readable call.  Is this
+  // inefficient?  Probably.
+  char ch;
+  ssize_t res = ::recv(s_, &ch, 1, MSG_PEEK);
+  if (res > 0) {
+    // Data available, so not closed.
+    return false;
+  } else if (res == 0) {
+    // EOF, so closed.
+    return true;
+  } else {  // error
+    switch (errno) {
+      // Returned if we've already closed s_.
+      case EBADF:
+      // Returned during ungraceful peer shutdown.
+      case ECONNRESET:
+        return true;
+      default:
+        // Assume that all other errors are just blocking errors, meaning the
+        // connection is still good but we just can't read from it right now.
+        // This should only happen when connecting (and at most once), because
+        // in all other cases this function is only called if the file
+        // descriptor is already known to be in the readable state. However,
+        // it's not necessary a problem if we spuriously interpret a
+        // "connection lost"-type error as a blocking error, because typically
+        // the next recv() will get EOF, so we'll still eventually notice that
+        // the socket is closed.
+        LOG_ERR(LS_WARNING) << "Assuming benign blocking error";
+        return false;
+    }
+  }
+}
+
+#endif // WEBRTC_POSIX
+
+uint32_t SocketDispatcher::GetRequestedEvents() {
+  return enabled_events_;
+}
+
+void SocketDispatcher::OnPreEvent(uint32_t ff) {
+  if ((ff & DE_CONNECT) != 0)
+    state_ = CS_CONNECTED;
+
+#if defined(WEBRTC_WIN)
+  // We set CS_CLOSED from CheckSignalClose.
+#elif defined(WEBRTC_POSIX)
+  if ((ff & DE_CLOSE) != 0)
+    state_ = CS_CLOSED;
+#endif
+}
+
+#if defined(WEBRTC_WIN)
+
+void SocketDispatcher::OnEvent(uint32_t ff, int err) {
+  int cache_id = id_;
+  // Make sure we deliver connect/accept first. Otherwise, consumers may see
+  // something like a READ followed by a CONNECT, which would be odd.
+  if (((ff & DE_CONNECT) != 0) && (id_ == cache_id)) {
+    if (ff != DE_CONNECT)
+      LOG(LS_VERBOSE) << "Signalled with DE_CONNECT: " << ff;
+    enabled_events_ &= ~DE_CONNECT;
+#if !defined(NDEBUG)
+    dbg_addr_ = "Connected @ ";
+    dbg_addr_.append(GetRemoteAddress().ToString());
+#endif
+    SignalConnectEvent(this);
+  }
+  if (((ff & DE_ACCEPT) != 0) && (id_ == cache_id)) {
+    enabled_events_ &= ~DE_ACCEPT;
+    SignalReadEvent(this);
+  }
+  if ((ff & DE_READ) != 0) {
+    enabled_events_ &= ~DE_READ;
+    SignalReadEvent(this);
+  }
+  if (((ff & DE_WRITE) != 0) && (id_ == cache_id)) {
+    enabled_events_ &= ~DE_WRITE;
+    SignalWriteEvent(this);
+  }
+  if (((ff & DE_CLOSE) != 0) && (id_ == cache_id)) {
+    signal_close_ = true;
+    signal_err_ = err;
+  }
+}
+
+#elif defined(WEBRTC_POSIX)
+
+void SocketDispatcher::OnEvent(uint32_t ff, int err) {
+  // Make sure we deliver connect/accept first. Otherwise, consumers may see
+  // something like a READ followed by a CONNECT, which would be odd.
+  if ((ff & DE_CONNECT) != 0) {
+    enabled_events_ &= ~DE_CONNECT;
+    SignalConnectEvent(this);
+  }
+  if ((ff & DE_ACCEPT) != 0) {
+    enabled_events_ &= ~DE_ACCEPT;
+    SignalReadEvent(this);
+  }
+  if ((ff & DE_READ) != 0) {
+    enabled_events_ &= ~DE_READ;
+    SignalReadEvent(this);
+  }
+  if ((ff & DE_WRITE) != 0) {
+    enabled_events_ &= ~DE_WRITE;
+    SignalWriteEvent(this);
+  }
+  if ((ff & DE_CLOSE) != 0) {
+    // The socket is now dead to us, so stop checking it.
+    enabled_events_ = 0;
+    SignalCloseEvent(this, err);
+  }
+}
+
+#endif // WEBRTC_POSIX
+
+int SocketDispatcher::Close() {
+  if (s_ == INVALID_SOCKET)
+    return 0;
+
+#if defined(WEBRTC_WIN)
+  id_ = 0;
+  signal_close_ = false;
+#endif
+  ss_->Remove(this);
+  return PhysicalSocket::Close();
+}
 
 #if defined(WEBRTC_POSIX)
 class EventDispatcher : public Dispatcher {
@@ -572,28 +778,28 @@ class EventDispatcher : public Dispatcher {
   virtual void Signal() {
     CritScope cs(&crit_);
     if (!fSignaled_) {
-      const uint8 b[1] = { 0 };
+      const uint8_t b[1] = {0};
       if (VERIFY(1 == write(afd_[1], b, sizeof(b)))) {
         fSignaled_ = true;
       }
     }
   }
 
-  uint32 GetRequestedEvents() override { return DE_READ; }
+  uint32_t GetRequestedEvents() override { return DE_READ; }
 
-  void OnPreEvent(uint32 ff) override {
+  void OnPreEvent(uint32_t ff) override {
     // It is not possible to perfectly emulate an auto-resetting event with
     // pipes.  This simulates it by resetting before the event is handled.
 
     CritScope cs(&crit_);
     if (fSignaled_) {
-      uint8 b[4];  // Allow for reading more than 1 byte, but expect 1.
+      uint8_t b[4];  // Allow for reading more than 1 byte, but expect 1.
       VERIFY(1 == read(afd_[0], b, sizeof(b)));
       fSignaled_ = false;
     }
   }
 
-  void OnEvent(uint32 ff, int err) override { ASSERT(false); }
+  void OnEvent(uint32_t ff, int err) override { ASSERT(false); }
 
   int GetDescriptor() override { return afd_[0]; }
 
@@ -622,14 +828,14 @@ class PosixSignalHandler {
   // sort of user-defined void * parameter, so they can't access anything that
   // isn't global.)
   static PosixSignalHandler* Instance() {
-    LIBJINGLE_DEFINE_STATIC_LOCAL(PosixSignalHandler, instance, ());
+    RTC_DEFINE_STATIC_LOCAL(PosixSignalHandler, instance, ());
     return &instance;
   }
 
   // Returns true if the given signal number is set.
   bool IsSignalSet(int signum) const {
-    ASSERT(signum < ARRAY_SIZE(received_signal_));
-    if (signum < ARRAY_SIZE(received_signal_)) {
+    ASSERT(signum < static_cast<int>(arraysize(received_signal_)));
+    if (signum < static_cast<int>(arraysize(received_signal_))) {
       return received_signal_[signum];
     } else {
       return false;
@@ -638,8 +844,8 @@ class PosixSignalHandler {
 
   // Clears the given signal number.
   void ClearSignal(int signum) {
-    ASSERT(signum < ARRAY_SIZE(received_signal_));
-    if (signum < ARRAY_SIZE(received_signal_)) {
+    ASSERT(signum < static_cast<int>(arraysize(received_signal_)));
+    if (signum < static_cast<int>(arraysize(received_signal_))) {
       received_signal_[signum] = false;
     }
   }
@@ -654,14 +860,14 @@ class PosixSignalHandler {
   // user-level state of the process, since the handler could be executed at any
   // time on any thread.
   void OnPosixSignalReceived(int signum) {
-    if (signum >= ARRAY_SIZE(received_signal_)) {
+    if (signum >= static_cast<int>(arraysize(received_signal_))) {
       // We don't have space in our array for this.
       return;
     }
     // Set a flag saying we've seen this signal.
     received_signal_[signum] = true;
     // Notify application code that we got a signal.
-    const uint8 b[1] = { 0 };
+    const uint8_t b[1] = {0};
     if (-1 == write(afd_[1], b, sizeof(b))) {
       // Nothing we can do here. If there's an error somehow then there's
       // nothing we can safely do from a signal handler.
@@ -718,7 +924,7 @@ class PosixSignalHandler {
   // will still be handled, so this isn't a problem.
   // Volatile is not necessary here for correctness, but this data _is_ volatile
   // so I've marked it as such.
-  volatile uint8 received_signal_[kNumPosixSignals];
+  volatile uint8_t received_signal_[kNumPosixSignals];
 };
 
 class PosixSignalDispatcher : public Dispatcher {
@@ -731,12 +937,12 @@ class PosixSignalDispatcher : public Dispatcher {
     owner_->Remove(this);
   }
 
-  uint32 GetRequestedEvents() override { return DE_READ; }
+  uint32_t GetRequestedEvents() override { return DE_READ; }
 
-  void OnPreEvent(uint32 ff) override {
+  void OnPreEvent(uint32_t ff) override {
     // Events might get grouped if signals come very fast, so we read out up to
     // 16 bytes to make sure we keep the pipe empty.
-    uint8 b[16];
+    uint8_t b[16];
     ssize_t ret = read(GetDescriptor(), b, sizeof(b));
     if (ret < 0) {
       LOG_ERR(LS_WARNING) << "Error in read()";
@@ -745,7 +951,7 @@ class PosixSignalDispatcher : public Dispatcher {
     }
   }
 
-  void OnEvent(uint32 ff, int err) override {
+  void OnEvent(uint32_t ff, int err) override {
     for (int signum = 0; signum < PosixSignalHandler::kNumPosixSignals;
          ++signum) {
       if (PosixSignalHandler::Instance()->IsSignalSet(signum)) {
@@ -790,116 +996,6 @@ class PosixSignalDispatcher : public Dispatcher {
   PhysicalSocketServer *owner_;
 };
 
-class SocketDispatcher : public Dispatcher, public PhysicalSocket {
- public:
-  explicit SocketDispatcher(PhysicalSocketServer *ss) : PhysicalSocket(ss) {
-  }
-  SocketDispatcher(SOCKET s, PhysicalSocketServer *ss) : PhysicalSocket(ss, s) {
-  }
-
-  ~SocketDispatcher() override {
-    Close();
-  }
-
-  bool Initialize() {
-    ss_->Add(this);
-    fcntl(s_, F_SETFL, fcntl(s_, F_GETFL, 0) | O_NONBLOCK);
-    return true;
-  }
-
-  virtual bool Create(int type) {
-    return Create(AF_INET, type);
-  }
-
-  bool Create(int family, int type) override {
-    // Change the socket to be non-blocking.
-    if (!PhysicalSocket::Create(family, type))
-      return false;
-
-    return Initialize();
-  }
-
-  int GetDescriptor() override { return s_; }
-
-  bool IsDescriptorClosed() override {
-    // We don't have a reliable way of distinguishing end-of-stream
-    // from readability.  So test on each readable call.  Is this
-    // inefficient?  Probably.
-    char ch;
-    ssize_t res = ::recv(s_, &ch, 1, MSG_PEEK);
-    if (res > 0) {
-      // Data available, so not closed.
-      return false;
-    } else if (res == 0) {
-      // EOF, so closed.
-      return true;
-    } else {  // error
-      switch (errno) {
-        // Returned if we've already closed s_.
-        case EBADF:
-        // Returned during ungraceful peer shutdown.
-        case ECONNRESET:
-          return true;
-        default:
-          // Assume that all other errors are just blocking errors, meaning the
-          // connection is still good but we just can't read from it right now.
-          // This should only happen when connecting (and at most once), because
-          // in all other cases this function is only called if the file
-          // descriptor is already known to be in the readable state. However,
-          // it's not necessary a problem if we spuriously interpret a
-          // "connection lost"-type error as a blocking error, because typically
-          // the next recv() will get EOF, so we'll still eventually notice that
-          // the socket is closed.
-          LOG_ERR(LS_WARNING) << "Assuming benign blocking error";
-          return false;
-      }
-    }
-  }
-
-  uint32 GetRequestedEvents() override { return enabled_events_; }
-
-  void OnPreEvent(uint32 ff) override {
-    if ((ff & DE_CONNECT) != 0)
-      state_ = CS_CONNECTED;
-    if ((ff & DE_CLOSE) != 0)
-      state_ = CS_CLOSED;
-  }
-
-  void OnEvent(uint32 ff, int err) override {
-    // Make sure we deliver connect/accept first. Otherwise, consumers may see
-    // something like a READ followed by a CONNECT, which would be odd.
-    if ((ff & DE_CONNECT) != 0) {
-      enabled_events_ &= ~DE_CONNECT;
-      SignalConnectEvent(this);
-    }
-    if ((ff & DE_ACCEPT) != 0) {
-      enabled_events_ &= ~DE_ACCEPT;
-      SignalReadEvent(this);
-    }
-    if ((ff & DE_READ) != 0) {
-      enabled_events_ &= ~DE_READ;
-      SignalReadEvent(this);
-    }
-    if ((ff & DE_WRITE) != 0) {
-      enabled_events_ &= ~DE_WRITE;
-      SignalWriteEvent(this);
-    }
-    if ((ff & DE_CLOSE) != 0) {
-      // The socket is now dead to us, so stop checking it.
-      enabled_events_ = 0;
-      SignalCloseEvent(this, err);
-    }
-  }
-
-  int Close() override {
-    if (s_ == INVALID_SOCKET)
-      return 0;
-
-    ss_->Remove(this);
-    return PhysicalSocket::Close();
-  }
-};
-
 class FileDispatcher: public Dispatcher, public AsyncFile {
  public:
   FileDispatcher(int fd, PhysicalSocketServer *ss) : ss_(ss), fd_(fd) {
@@ -920,11 +1016,11 @@ class FileDispatcher: public Dispatcher, public AsyncFile {
 
   bool IsDescriptorClosed() override { return false; }
 
-  uint32 GetRequestedEvents() override { return flags_; }
+  uint32_t GetRequestedEvents() override { return flags_; }
 
-  void OnPreEvent(uint32 ff) override {}
+  void OnPreEvent(uint32_t ff) override {}
 
-  void OnEvent(uint32 ff, int err) override {
+  void OnEvent(uint32_t ff, int err) override {
     if ((ff & DE_READ) != 0)
       SignalReadEvent(this);
     if ((ff & DE_WRITE) != 0)
@@ -958,8 +1054,8 @@ AsyncFile* PhysicalSocketServer::CreateFile(int fd) {
 #endif // WEBRTC_POSIX
 
 #if defined(WEBRTC_WIN)
-static uint32 FlagsToEvents(uint32 events) {
-  uint32 ffFD = FD_CLOSE;
+static uint32_t FlagsToEvents(uint32_t events) {
+  uint32_t ffFD = FD_CLOSE;
   if (events & DE_READ)
     ffFD |= FD_READ;
   if (events & DE_WRITE)
@@ -993,16 +1089,11 @@ class EventDispatcher : public Dispatcher {
       WSASetEvent(hev_);
   }
 
-  virtual uint32 GetRequestedEvents() {
-    return 0;
-  }
+  virtual uint32_t GetRequestedEvents() { return 0; }
 
-  virtual void OnPreEvent(uint32 ff) {
-    WSAResetEvent(hev_);
-  }
+  virtual void OnPreEvent(uint32_t ff) { WSAResetEvent(hev_); }
 
-  virtual void OnEvent(uint32 ff, int err) {
-  }
+  virtual void OnEvent(uint32_t ff, int err) {}
 
   virtual WSAEVENT GetWSAEvent() {
     return hev_;
@@ -1018,132 +1109,6 @@ private:
   PhysicalSocketServer* ss_;
   WSAEVENT hev_;
 };
-
-class SocketDispatcher : public Dispatcher, public PhysicalSocket {
- public:
-  static int next_id_;
-  int id_;
-  bool signal_close_;
-  int signal_err_;
-
-  SocketDispatcher(PhysicalSocketServer* ss)
-      : PhysicalSocket(ss),
-        id_(0),
-        signal_close_(false) {
-  }
-
-  SocketDispatcher(SOCKET s, PhysicalSocketServer* ss)
-      : PhysicalSocket(ss, s),
-        id_(0),
-        signal_close_(false) {
-  }
-
-  virtual ~SocketDispatcher() {
-    Close();
-  }
-
-  bool Initialize() {
-    ASSERT(s_ != INVALID_SOCKET);
-    // Must be a non-blocking
-    u_long argp = 1;
-    ioctlsocket(s_, FIONBIO, &argp);
-    ss_->Add(this);
-    return true;
-  }
-
-  virtual bool Create(int type) {
-    return Create(AF_INET, type);
-  }
-
-  virtual bool Create(int family, int type) {
-    // Create socket
-    if (!PhysicalSocket::Create(family, type))
-      return false;
-
-    if (!Initialize())
-      return false;
-
-    do { id_ = ++next_id_; } while (id_ == 0);
-    return true;
-  }
-
-  virtual int Close() {
-    if (s_ == INVALID_SOCKET)
-      return 0;
-
-    id_ = 0;
-    signal_close_ = false;
-    ss_->Remove(this);
-    return PhysicalSocket::Close();
-  }
-
-  virtual uint32 GetRequestedEvents() {
-    return enabled_events_;
-  }
-
-  virtual void OnPreEvent(uint32 ff) {
-    if ((ff & DE_CONNECT) != 0)
-      state_ = CS_CONNECTED;
-    // We set CS_CLOSED from CheckSignalClose.
-  }
-
-  virtual void OnEvent(uint32 ff, int err) {
-    int cache_id = id_;
-    // Make sure we deliver connect/accept first. Otherwise, consumers may see
-    // something like a READ followed by a CONNECT, which would be odd.
-    if (((ff & DE_CONNECT) != 0) && (id_ == cache_id)) {
-      if (ff != DE_CONNECT)
-        LOG(LS_VERBOSE) << "Signalled with DE_CONNECT: " << ff;
-      enabled_events_ &= ~DE_CONNECT;
-#ifdef _DEBUG
-      dbg_addr_ = "Connected @ ";
-      dbg_addr_.append(GetRemoteAddress().ToString());
-#endif  // _DEBUG
-      SignalConnectEvent(this);
-    }
-    if (((ff & DE_ACCEPT) != 0) && (id_ == cache_id)) {
-      enabled_events_ &= ~DE_ACCEPT;
-      SignalReadEvent(this);
-    }
-    if ((ff & DE_READ) != 0) {
-      enabled_events_ &= ~DE_READ;
-      SignalReadEvent(this);
-    }
-    if (((ff & DE_WRITE) != 0) && (id_ == cache_id)) {
-      enabled_events_ &= ~DE_WRITE;
-      SignalWriteEvent(this);
-    }
-    if (((ff & DE_CLOSE) != 0) && (id_ == cache_id)) {
-      signal_close_ = true;
-      signal_err_ = err;
-    }
-  }
-
-  virtual WSAEVENT GetWSAEvent() {
-    return WSA_INVALID_EVENT;
-  }
-
-  virtual SOCKET GetSocket() {
-    return s_;
-  }
-
-  virtual bool CheckSignalClose() {
-    if (!signal_close_)
-      return false;
-
-    char ch;
-    if (recv(s_, &ch, 1, MSG_PEEK) > 0)
-      return false;
-
-    state_ = CS_CLOSED;
-    signal_close_ = false;
-    SignalCloseEvent(this, signal_err_);
-    return true;
-  }
-};
-
-int SocketDispatcher::next_id_ = 0;
-
 #endif  // WEBRTC_WIN 
 
 // Sets the value of a boolean value to false when signaled.
@@ -1154,7 +1119,7 @@ class Signaler : public EventDispatcher {
   }
   ~Signaler() override { }
 
-  void OnEvent(uint32 ff, int err) override {
+  void OnEvent(uint32_t ff, int err) override {
     if (pf_)
       *pf_ = false;
   }
@@ -1196,7 +1161,7 @@ Socket* PhysicalSocketServer::CreateSocket(int family, int type) {
     return socket;
   } else {
     delete socket;
-    return 0;
+    return nullptr;
   }
 }
 
@@ -1210,7 +1175,7 @@ AsyncSocket* PhysicalSocketServer::CreateAsyncSocket(int family, int type) {
     return dispatcher;
   } else {
     delete dispatcher;
-    return 0;
+    return nullptr;
   }
 }
 
@@ -1220,7 +1185,7 @@ AsyncSocket* PhysicalSocketServer::WrapSocket(SOCKET s) {
     return dispatcher;
   } else {
     delete dispatcher;
-    return 0;
+    return nullptr;
   }
 }
 
@@ -1312,7 +1277,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
         if (fd > fdmax)
           fdmax = fd;
 
-        uint32 ff = pdispatcher->GetRequestedEvents();
+        uint32_t ff = pdispatcher->GetRequestedEvents();
         if (ff & (DE_READ | DE_ACCEPT))
           FD_SET(fd, &fdsRead);
         if (ff & (DE_WRITE | DE_CONNECT))
@@ -1345,11 +1310,11 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
       for (size_t i = 0; i < dispatchers_.size(); ++i) {
         Dispatcher *pdispatcher = dispatchers_[i];
         int fd = pdispatcher->GetDescriptor();
-        uint32 ff = 0;
+        uint32_t ff = 0;
         int errcode = 0;
 
         // Reap any error code, which can be signaled through reads or writes.
-        // TODO: Should we set errcode if getsockopt fails?
+        // TODO(pthatcher): Should we set errcode if getsockopt fails?
         if (FD_ISSET(fd, &fdsRead) || FD_ISSET(fd, &fdsWrite)) {
           socklen_t len = sizeof(errcode);
           ::getsockopt(fd, SOL_SOCKET, SO_ERROR, &errcode, &len);
@@ -1358,7 +1323,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
         // Check readable descriptors. If we're waiting on an accept, signal
         // that. Otherwise we're waiting for data, check to see if we're
         // readable or really closed.
-        // TODO: Only peek at TCP descriptors.
+        // TODO(pthatcher): Only peek at TCP descriptors.
         if (FD_ISSET(fd, &fdsRead)) {
           FD_CLR(fd, &fdsRead);
           if (pdispatcher->GetRequestedEvents() & DE_ACCEPT) {
@@ -1479,7 +1444,7 @@ bool PhysicalSocketServer::InstallSignal(int signum, void (*handler)(int)) {
 bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
   int cmsTotal = cmsWait;
   int cmsElapsed = 0;
-  uint32 msStart = Time();
+  uint32_t msStart = Time();
 
   fWait_ = true;
   while (fWait_) {
@@ -1532,7 +1497,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
 
     if (dw == WSA_WAIT_FAILED) {
       // Failed?
-      // TODO: need a better strategy than this!
+      // TODO(pthatcher): need a better strategy than this!
       WSAGetLastError();
       ASSERT(false);
       return false;
@@ -1590,7 +1555,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
               }
             }
 #endif
-            uint32 ff = 0;
+            uint32_t ff = 0;
             int errcode = 0;
             if (wsaEvents.lNetworkEvents & FD_READ)
               ff |= DE_READ;
diff --git a/media/webrtc/trunk/webrtc/base/physicalsocketserver.h b/media/webrtc/trunk/webrtc/base/physicalsocketserver.h
index 15be789e26..ae1f10f596 100644
--- a/media/webrtc/trunk/webrtc/base/physicalsocketserver.h
+++ b/media/webrtc/trunk/webrtc/base/physicalsocketserver.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/asyncfile.h"
+#include "webrtc/base/nethelpers.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketserver.h"
 #include "webrtc/base/criticalsection.h"
@@ -41,9 +42,9 @@ class PosixSignalDispatcher;
 class Dispatcher {
  public:
   virtual ~Dispatcher() {}
-  virtual uint32 GetRequestedEvents() = 0;
-  virtual void OnPreEvent(uint32 ff) = 0;
-  virtual void OnEvent(uint32 ff, int err) = 0;
+  virtual uint32_t GetRequestedEvents() = 0;
+  virtual void OnPreEvent(uint32_t ff) = 0;
+  virtual void OnEvent(uint32_t ff, int err) = 0;
 #if defined(WEBRTC_WIN)
   virtual WSAEVENT GetWSAEvent() = 0;
   virtual SOCKET GetSocket() = 0;
@@ -115,6 +116,107 @@ class PhysicalSocketServer : public SocketServer {
 #endif
 };
 
+class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
+ public:
+  PhysicalSocket(PhysicalSocketServer* ss, SOCKET s = INVALID_SOCKET);
+  ~PhysicalSocket() override;
+
+  // Creates the underlying OS socket (same as the "socket" function).
+  virtual bool Create(int family, int type);
+
+  SocketAddress GetLocalAddress() const override;
+  SocketAddress GetRemoteAddress() const override;
+
+  int Bind(const SocketAddress& bind_addr) override;
+  int Connect(const SocketAddress& addr) override;
+
+  int GetError() const override;
+  void SetError(int error) override;
+
+  ConnState GetState() const override;
+
+  int GetOption(Option opt, int* value) override;
+  int SetOption(Option opt, int value) override;
+
+  int Send(const void* pv, size_t cb) override;
+  int SendTo(const void* buffer,
+             size_t length,
+             const SocketAddress& addr) override;
+
+  int Recv(void* buffer, size_t length) override;
+  int RecvFrom(void* buffer, size_t length, SocketAddress* out_addr) override;
+
+  int Listen(int backlog) override;
+  AsyncSocket* Accept(SocketAddress* out_addr) override;
+
+  int Close() override;
+
+  int EstimateMTU(uint16_t* mtu) override;
+
+  SocketServer* socketserver() { return ss_; }
+
+ protected:
+  int DoConnect(const SocketAddress& connect_addr);
+
+  // Make virtual so ::accept can be overwritten in tests.
+  virtual SOCKET DoAccept(SOCKET socket, sockaddr* addr, socklen_t* addrlen);
+
+  void OnResolveResult(AsyncResolverInterface* resolver);
+
+  void UpdateLastError();
+  void MaybeRemapSendError();
+
+  static int TranslateOption(Option opt, int* slevel, int* sopt);
+
+  PhysicalSocketServer* ss_;
+  SOCKET s_;
+  uint8_t enabled_events_;
+  bool udp_;
+  mutable CriticalSection crit_;
+  int error_ GUARDED_BY(crit_);
+  ConnState state_;
+  AsyncResolver* resolver_;
+
+#if !defined(NDEBUG)
+  std::string dbg_addr_;
+#endif
+};
+
+class SocketDispatcher : public Dispatcher, public PhysicalSocket {
+ public:
+  explicit SocketDispatcher(PhysicalSocketServer *ss);
+  SocketDispatcher(SOCKET s, PhysicalSocketServer *ss);
+  ~SocketDispatcher() override;
+
+  bool Initialize();
+
+  virtual bool Create(int type);
+  bool Create(int family, int type) override;
+
+#if defined(WEBRTC_WIN)
+  WSAEVENT GetWSAEvent() override;
+  SOCKET GetSocket() override;
+  bool CheckSignalClose() override;
+#elif defined(WEBRTC_POSIX)
+  int GetDescriptor() override;
+  bool IsDescriptorClosed() override;
+#endif
+
+  uint32_t GetRequestedEvents() override;
+  void OnPreEvent(uint32_t ff) override;
+  void OnEvent(uint32_t ff, int err) override;
+
+  int Close() override;
+
+#if defined(WEBRTC_WIN)
+ private:
+  static int next_id_;
+  int id_;
+  bool signal_close_;
+  int signal_err_;
+#endif // WEBRTC_WIN
+};
+
 } // namespace rtc
 
 #endif // WEBRTC_BASE_PHYSICALSOCKETSERVER_H__
diff --git a/media/webrtc/trunk/webrtc/base/physicalsocketserver_unittest.cc b/media/webrtc/trunk/webrtc/base/physicalsocketserver_unittest.cc
index 3b7ed7bff2..a2fde80b42 100644
--- a/media/webrtc/trunk/webrtc/base/physicalsocketserver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/physicalsocketserver_unittest.cc
@@ -18,18 +18,96 @@
 #include "webrtc/base/socket_unittest.h"
 #include "webrtc/base/testutils.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
-class PhysicalSocketTest : public SocketTest {
+class PhysicalSocketTest;
+
+class FakeSocketDispatcher : public SocketDispatcher {
+ public:
+  explicit FakeSocketDispatcher(PhysicalSocketServer* ss)
+    : SocketDispatcher(ss) {
+  }
+
+ protected:
+  SOCKET DoAccept(SOCKET socket, sockaddr* addr, socklen_t* addrlen) override;
 };
 
+class FakePhysicalSocketServer : public PhysicalSocketServer {
+ public:
+  explicit FakePhysicalSocketServer(PhysicalSocketTest* test)
+    : test_(test) {
+  }
+
+  AsyncSocket* CreateAsyncSocket(int type) override {
+    SocketDispatcher* dispatcher = new FakeSocketDispatcher(this);
+    if (dispatcher->Create(type)) {
+      return dispatcher;
+    } else {
+      delete dispatcher;
+      return nullptr;
+    }
+  }
+
+  AsyncSocket* CreateAsyncSocket(int family, int type) override {
+    SocketDispatcher* dispatcher = new FakeSocketDispatcher(this);
+    if (dispatcher->Create(family, type)) {
+      return dispatcher;
+    } else {
+      delete dispatcher;
+      return nullptr;
+    }
+  }
+
+  PhysicalSocketTest* GetTest() const { return test_; }
+
+ private:
+  PhysicalSocketTest* test_;
+};
+
+class PhysicalSocketTest : public SocketTest {
+ public:
+  // Set flag to simluate failures when calling "::accept" on a AsyncSocket.
+  void SetFailAccept(bool fail) { fail_accept_ = fail; }
+  bool FailAccept() const { return fail_accept_; }
+
+ protected:
+  PhysicalSocketTest()
+    : server_(new FakePhysicalSocketServer(this)),
+      scope_(server_.get()),
+      fail_accept_(false) {
+  }
+
+  void ConnectInternalAcceptError(const IPAddress& loopback);
+
+  rtc::scoped_ptr<FakePhysicalSocketServer> server_;
+  SocketServerScope scope_;
+  bool fail_accept_;
+};
+
+SOCKET FakeSocketDispatcher::DoAccept(SOCKET socket,
+                                      sockaddr* addr,
+                                      socklen_t* addrlen) {
+  FakePhysicalSocketServer* ss =
+      static_cast<FakePhysicalSocketServer*>(socketserver());
+  if (ss->GetTest()->FailAccept()) {
+    return INVALID_SOCKET;
+  }
+
+  return SocketDispatcher::DoAccept(socket, addr, addrlen);
+}
+
 TEST_F(PhysicalSocketTest, TestConnectIPv4) {
   SocketTest::TestConnectIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestConnectIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectIPv6 DISABLED_TestConnectIPv6
+#else
+#define MAYBE_TestConnectIPv6 TestConnectIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectIPv6) {
   SocketTest::TestConnectIPv6();
 }
 
@@ -45,7 +123,99 @@ TEST_F(PhysicalSocketTest, TestConnectFailIPv4) {
   SocketTest::TestConnectFailIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestConnectFailIPv6) {
+void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) {
+  testing::StreamSink sink;
+  SocketAddress accept_addr;
+
+  // Create two clients.
+  scoped_ptr<AsyncSocket> client1(server_->CreateAsyncSocket(loopback.family(),
+                                                             SOCK_STREAM));
+  sink.Monitor(client1.get());
+  EXPECT_EQ(AsyncSocket::CS_CLOSED, client1->GetState());
+  EXPECT_PRED1(IsUnspecOrEmptyIP, client1->GetLocalAddress().ipaddr());
+
+  scoped_ptr<AsyncSocket> client2(server_->CreateAsyncSocket(loopback.family(),
+                                                             SOCK_STREAM));
+  sink.Monitor(client2.get());
+  EXPECT_EQ(AsyncSocket::CS_CLOSED, client2->GetState());
+  EXPECT_PRED1(IsUnspecOrEmptyIP, client2->GetLocalAddress().ipaddr());
+
+  // Create server and listen.
+  scoped_ptr<AsyncSocket> server(
+      server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
+  sink.Monitor(server.get());
+  EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
+  EXPECT_EQ(0, server->Listen(5));
+  EXPECT_EQ(AsyncSocket::CS_CONNECTING, server->GetState());
+
+  // Ensure no pending server connections, since we haven't done anything yet.
+  EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ));
+  EXPECT_TRUE(nullptr == server->Accept(&accept_addr));
+  EXPECT_TRUE(accept_addr.IsNil());
+
+  // Attempt first connect to listening socket.
+  EXPECT_EQ(0, client1->Connect(server->GetLocalAddress()));
+  EXPECT_FALSE(client1->GetLocalAddress().IsNil());
+  EXPECT_NE(server->GetLocalAddress(), client1->GetLocalAddress());
+
+  // Client is connecting, outcome not yet determined.
+  EXPECT_EQ(AsyncSocket::CS_CONNECTING, client1->GetState());
+  EXPECT_FALSE(sink.Check(client1.get(), testing::SSE_OPEN));
+  EXPECT_FALSE(sink.Check(client1.get(), testing::SSE_CLOSE));
+
+  // Server has pending connection, try to accept it (will fail).
+  EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
+  // Simulate "::accept" returning an error.
+  SetFailAccept(true);
+  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  EXPECT_FALSE(accepted);
+  ASSERT_TRUE(accept_addr.IsNil());
+
+  // Ensure no more pending server connections.
+  EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ));
+  EXPECT_TRUE(nullptr == server->Accept(&accept_addr));
+  EXPECT_TRUE(accept_addr.IsNil());
+
+  // Attempt second connect to listening socket.
+  EXPECT_EQ(0, client2->Connect(server->GetLocalAddress()));
+  EXPECT_FALSE(client2->GetLocalAddress().IsNil());
+  EXPECT_NE(server->GetLocalAddress(), client2->GetLocalAddress());
+
+  // Client is connecting, outcome not yet determined.
+  EXPECT_EQ(AsyncSocket::CS_CONNECTING, client2->GetState());
+  EXPECT_FALSE(sink.Check(client2.get(), testing::SSE_OPEN));
+  EXPECT_FALSE(sink.Check(client2.get(), testing::SSE_CLOSE));
+
+  // Server has pending connection, try to accept it (will succeed).
+  EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
+  SetFailAccept(false);
+  scoped_ptr<AsyncSocket> accepted2(server->Accept(&accept_addr));
+  ASSERT_TRUE(accepted2);
+  EXPECT_FALSE(accept_addr.IsNil());
+  EXPECT_EQ(accepted2->GetRemoteAddress(), accept_addr);
+}
+
+TEST_F(PhysicalSocketTest, TestConnectAcceptErrorIPv4) {
+  ConnectInternalAcceptError(kIPv4Loopback);
+}
+
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectAcceptErrorIPv6 DISABLED_TestConnectAcceptErrorIPv6
+#else
+#define MAYBE_TestConnectAcceptErrorIPv6 TestConnectAcceptErrorIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectAcceptErrorIPv6) {
+  ConnectInternalAcceptError(kIPv6Loopback);
+}
+
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectFailIPv6 DISABLED_TestConnectFailIPv6
+#else
+#define MAYBE_TestConnectFailIPv6 TestConnectFailIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectFailIPv6) {
   SocketTest::TestConnectFailIPv6();
 }
 
@@ -53,8 +223,15 @@ TEST_F(PhysicalSocketTest, TestConnectWithDnsLookupFailIPv4) {
   SocketTest::TestConnectWithDnsLookupFailIPv4();
 }
 
-
-TEST_F(PhysicalSocketTest, TestConnectWithDnsLookupFailIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectWithDnsLookupFailIPv6 \
+  DISABLED_TestConnectWithDnsLookupFailIPv6
+#else
+#define MAYBE_TestConnectWithDnsLookupFailIPv6 \
+  TestConnectWithDnsLookupFailIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectWithDnsLookupFailIPv6) {
   SocketTest::TestConnectWithDnsLookupFailIPv6();
 }
 
@@ -63,7 +240,14 @@ TEST_F(PhysicalSocketTest, TestConnectWithClosedSocketIPv4) {
   SocketTest::TestConnectWithClosedSocketIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestConnectWithClosedSocketIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectWithClosedSocketIPv6 \
+  DISABLED_TestConnectWithClosedSocketIPv6
+#else
+#define MAYBE_TestConnectWithClosedSocketIPv6 TestConnectWithClosedSocketIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectWithClosedSocketIPv6) {
   SocketTest::TestConnectWithClosedSocketIPv6();
 }
 
@@ -71,7 +255,14 @@ TEST_F(PhysicalSocketTest, TestConnectWhileNotClosedIPv4) {
   SocketTest::TestConnectWhileNotClosedIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestConnectWhileNotClosedIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectWhileNotClosedIPv6 \
+  DISABLED_TestConnectWhileNotClosedIPv6
+#else
+#define MAYBE_TestConnectWhileNotClosedIPv6 TestConnectWhileNotClosedIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectWhileNotClosedIPv6) {
   SocketTest::TestConnectWhileNotClosedIPv6();
 }
 
@@ -79,7 +270,14 @@ TEST_F(PhysicalSocketTest, TestServerCloseDuringConnectIPv4) {
   SocketTest::TestServerCloseDuringConnectIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestServerCloseDuringConnectIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestServerCloseDuringConnectIPv6 \
+  DISABLED_TestServerCloseDuringConnectIPv6
+#else
+#define MAYBE_TestServerCloseDuringConnectIPv6 TestServerCloseDuringConnectIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestServerCloseDuringConnectIPv6) {
   SocketTest::TestServerCloseDuringConnectIPv6();
 }
 
@@ -87,7 +285,14 @@ TEST_F(PhysicalSocketTest, TestClientCloseDuringConnectIPv4) {
   SocketTest::TestClientCloseDuringConnectIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestClientCloseDuringConnectIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestClientCloseDuringConnectIPv6 \
+  DISABLED_TestClientCloseDuringConnectIPv6
+#else
+#define MAYBE_TestClientCloseDuringConnectIPv6 TestClientCloseDuringConnectIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestClientCloseDuringConnectIPv6) {
   SocketTest::TestClientCloseDuringConnectIPv6();
 }
 
@@ -95,7 +300,13 @@ TEST_F(PhysicalSocketTest, TestServerCloseIPv4) {
   SocketTest::TestServerCloseIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestServerCloseIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestServerCloseIPv6 DISABLED_TestServerCloseIPv6
+#else
+#define MAYBE_TestServerCloseIPv6 TestServerCloseIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestServerCloseIPv6) {
   SocketTest::TestServerCloseIPv6();
 }
 
@@ -103,7 +314,14 @@ TEST_F(PhysicalSocketTest, TestCloseInClosedCallbackIPv4) {
   SocketTest::TestCloseInClosedCallbackIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestCloseInClosedCallbackIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestCloseInClosedCallbackIPv6 \
+  DISABLED_TestCloseInClosedCallbackIPv6
+#else
+#define MAYBE_TestCloseInClosedCallbackIPv6 TestCloseInClosedCallbackIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestCloseInClosedCallbackIPv6) {
   SocketTest::TestCloseInClosedCallbackIPv6();
 }
 
@@ -111,7 +329,13 @@ TEST_F(PhysicalSocketTest, TestSocketServerWaitIPv4) {
   SocketTest::TestSocketServerWaitIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestSocketServerWaitIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestSocketServerWaitIPv6 DISABLED_TestSocketServerWaitIPv6
+#else
+#define MAYBE_TestSocketServerWaitIPv6 TestSocketServerWaitIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestSocketServerWaitIPv6) {
   SocketTest::TestSocketServerWaitIPv6();
 }
 
@@ -119,7 +343,13 @@ TEST_F(PhysicalSocketTest, TestTcpIPv4) {
   SocketTest::TestTcpIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestTcpIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestTcpIPv6 DISABLED_TestTcpIPv6
+#else
+#define MAYBE_TestTcpIPv6 TestTcpIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestTcpIPv6) {
   SocketTest::TestTcpIPv6();
 }
 
@@ -127,20 +357,35 @@ TEST_F(PhysicalSocketTest, TestUdpIPv4) {
   SocketTest::TestUdpIPv4();
 }
 
-TEST_F(PhysicalSocketTest, TestUdpIPv6) {
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestUdpIPv6 DISABLED_TestUdpIPv6
+#else
+#define MAYBE_TestUdpIPv6 TestUdpIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestUdpIPv6) {
   SocketTest::TestUdpIPv6();
 }
 
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=3498 for details.
-#if !defined(THREAD_SANITIZER)
-
-TEST_F(PhysicalSocketTest, TestUdpReadyToSendIPv4) {
+// Also disable for MSan, see:
+// https://code.google.com/p/webrtc/issues/detail?id=4958
+// TODO(deadbeef): Enable again once test is reimplemented to be unflaky.
+// Also disable for ASan.
+// Disabled on Android: https://code.google.com/p/webrtc/issues/detail?id=4364
+// Disabled on Linux: https://bugs.chromium.org/p/webrtc/issues/detail?id=5233
+#if defined(THREAD_SANITIZER) || defined(MEMORY_SANITIZER) || \
+    defined(ADDRESS_SANITIZER) || defined(WEBRTC_ANDROID) ||  \
+    defined(WEBRTC_LINUX)
+#define MAYBE_TestUdpReadyToSendIPv4 DISABLED_TestUdpReadyToSendIPv4
+#else
+#define MAYBE_TestUdpReadyToSendIPv4 TestUdpReadyToSendIPv4
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestUdpReadyToSendIPv4) {
   SocketTest::TestUdpReadyToSendIPv4();
 }
 
-#endif // if !defined(THREAD_SANITIZER)
-
 TEST_F(PhysicalSocketTest, TestUdpReadyToSendIPv6) {
   SocketTest::TestUdpReadyToSendIPv6();
 }
diff --git a/media/webrtc/trunk/webrtc/base/platform_thread.cc b/media/webrtc/trunk/webrtc/base/platform_thread.cc
new file mode 100644
index 0000000000..0a394331ff
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/platform_thread.cc
@@ -0,0 +1,362 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/platform_thread.h"
+
+#include "webrtc/base/checks.h"
+
+#if defined(WEBRTC_LINUX)
+#include <sys/prctl.h>
+#include <sys/syscall.h>
+#endif
+
+namespace rtc {
+
+#if defined(WEBRTC_WIN)
+// For use in ThreadWindowsUI callbacks
+static UINT static_reg_windows_msg = RegisterWindowMessageW(L"WebrtcWindowsUIThreadEvent");
+// timer id used in delayed callbacks
+static const UINT_PTR kTimerId = 1;
+static const wchar_t kThisProperty[] = L"ThreadWindowsUIPtr";
+static const wchar_t kThreadWindow[] = L"WebrtcWindowsUIThread";
+#endif
+
+PlatformThreadId CurrentThreadId() {
+  PlatformThreadId ret;
+#if defined(WEBRTC_WIN)
+  ret = GetCurrentThreadId();
+#elif defined(WEBRTC_POSIX)
+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+  ret = pthread_mach_thread_np(pthread_self());
+#elif defined(WEBRTC_LINUX)
+  ret =  syscall(__NR_gettid);
+#elif defined(WEBRTC_ANDROID)
+  ret = gettid();
+#else
+  // Default implementation for nacl and solaris.
+  ret = reinterpret_cast<pid_t>(pthread_self());
+#endif
+#endif  // defined(WEBRTC_POSIX)
+  RTC_DCHECK(ret);
+  return ret;
+}
+
+PlatformThreadRef CurrentThreadRef() {
+#if defined(WEBRTC_WIN)
+  return GetCurrentThreadId();
+#elif defined(WEBRTC_POSIX)
+  return pthread_self();
+#endif
+}
+
+bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b) {
+#if defined(WEBRTC_WIN)
+  return a == b;
+#elif defined(WEBRTC_POSIX)
+  return pthread_equal(a, b);
+#endif
+}
+
+void SetCurrentThreadName(const char* name) {
+#if defined(WEBRTC_WIN)
+  struct {
+    DWORD dwType;
+    LPCSTR szName;
+    DWORD dwThreadID;
+    DWORD dwFlags;
+  } threadname_info = {0x1000, name, static_cast<DWORD>(-1), 0};
+
+  __try {
+    ::RaiseException(0x406D1388, 0, sizeof(threadname_info) / sizeof(DWORD),
+                     reinterpret_cast<ULONG_PTR*>(&threadname_info));
+  } __except (EXCEPTION_EXECUTE_HANDLER) {
+  }
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
+  prctl(PR_SET_NAME, reinterpret_cast<unsigned long>(name));
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+  pthread_setname_np(name);
+#endif
+}
+
+namespace {
+#if defined(WEBRTC_WIN)
+void CALLBACK RaiseFlag(ULONG_PTR param) {
+  *reinterpret_cast<bool*>(param) = true;
+}
+#else
+struct ThreadAttributes {
+  ThreadAttributes() { pthread_attr_init(&attr); }
+  ~ThreadAttributes() { pthread_attr_destroy(&attr); }
+  pthread_attr_t* operator&() { return &attr; }
+  pthread_attr_t attr;
+};
+#endif  // defined(WEBRTC_WIN)
+}
+
+PlatformThread::PlatformThread(ThreadRunFunction func,
+                               void* obj,
+                               const char* thread_name)
+    : run_function_(func),
+      obj_(obj),
+      name_(thread_name ? thread_name : "webrtc"),
+#if defined(WEBRTC_WIN)
+      stop_(false),
+      thread_(NULL) {
+#else
+      stop_event_(false, false),
+      thread_(0) {
+#endif  // defined(WEBRTC_WIN)
+  RTC_DCHECK(func);
+  RTC_DCHECK(name_.length() < 64);
+}
+
+PlatformThread::~PlatformThread() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+#if defined(WEBRTC_WIN)
+  RTC_DCHECK(!thread_);
+#endif  // defined(WEBRTC_WIN)
+}
+
+#if defined(WEBRTC_WIN)
+bool PlatformUIThread::InternalInit() {
+  // Create an event window for use in generating callbacks to capture
+  // objects.
+  if (hwnd_ == NULL) {
+    WNDCLASSW wc;
+    HMODULE hModule = GetModuleHandle(NULL);
+    if (!GetClassInfoW(hModule, kThreadWindow, &wc)) {
+      ZeroMemory(&wc, sizeof(WNDCLASSW));
+      wc.hInstance = hModule;
+      wc.lpfnWndProc = EventWindowProc;
+      wc.lpszClassName = kThreadWindow;
+      RegisterClassW(&wc);
+    }
+    hwnd_ = CreateWindowW(kThreadWindow, L"",
+                          0, 0, 0, 0, 0,
+                          NULL, NULL, hModule, NULL);
+    assert(hwnd_);
+    SetPropW(hwnd_, kThisProperty, this);
+
+    if (timeout_) {
+      // if someone set the timer before we started
+      RequestCallbackTimer(timeout_);
+    }
+  }
+  return !!hwnd_;
+}
+
+void PlatformUIThread::RequestCallback() {
+  assert(hwnd_);
+  assert(static_reg_windows_msg);
+  PostMessage(hwnd_, static_reg_windows_msg, 0, 0);
+}
+
+bool PlatformUIThread::RequestCallbackTimer(unsigned int milliseconds) {
+  if (!hwnd_) {
+    assert(!thread_);
+    // set timer once thread starts
+  } else {
+    if (timerid_) {
+      KillTimer(hwnd_, timerid_);
+    }
+    timerid_ = SetTimer(hwnd_, kTimerId, milliseconds, NULL);
+  }
+  timeout_ = milliseconds;
+  return !!timerid_;
+}
+
+DWORD WINAPI PlatformThread::StartThread(void* param) {
+  static_cast<PlatformThread*>(param)->Run();
+  return 0;
+}
+#else
+void* PlatformThread::StartThread(void* param) {
+  static_cast<PlatformThread*>(param)->Run();
+  return 0;
+}
+#endif  // defined(WEBRTC_WIN)
+
+void PlatformThread::Start() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!thread_) << "Thread already started?";
+#if defined(WEBRTC_WIN)
+  stop_ = false;
+
+  // See bug 2902 for background on STACK_SIZE_PARAM_IS_A_RESERVATION.
+  // Set the reserved stack stack size to 1M, which is the default on Windows
+  // and Linux.
+  DWORD thread_id;
+  thread_ = ::CreateThread(NULL, 1024 * 1024, &StartThread, this,
+                           STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
+  RTC_CHECK(thread_) << "CreateThread failed";
+#else
+  ThreadAttributes attr;
+  // Set the stack stack size to 1M.
+  pthread_attr_setstacksize(&attr, 1024 * 1024);
+  RTC_CHECK_EQ(0, pthread_create(&thread_, &attr, &StartThread, this));
+#endif  // defined(WEBRTC_WIN)
+}
+
+bool PlatformThread::IsRunning() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+#if defined(WEBRTC_WIN)
+  return thread_ != nullptr;
+#else
+  return thread_ != 0;
+#endif  // defined(WEBRTC_WIN)
+}
+
+void PlatformThread::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!IsRunning())
+    return;
+
+#if defined(WEBRTC_WIN)
+  // Set stop_ to |true| on the worker thread.
+  QueueUserAPC(&RaiseFlag, thread_, reinterpret_cast<ULONG_PTR>(&stop_));
+  WaitForSingleObject(thread_, INFINITE);
+  CloseHandle(thread_);
+  thread_ = nullptr;
+#else
+  stop_event_.Set();
+  RTC_CHECK_EQ(0, pthread_join(thread_, nullptr));
+  thread_ = 0;
+#endif  // defined(WEBRTC_WIN)
+}
+
+#ifdef WEBRTC_WIN
+void PlatformUIThread::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  // Shut down the dispatch loop and let the background thread exit.
+  if (timerid_) {
+    KillTimer(hwnd_, timerid_);
+    timerid_ = 0;
+  }
+
+  PostMessage(hwnd_, WM_CLOSE, 0, 0);
+
+  PlatformThread::Stop();
+}
+#endif
+
+void PlatformThread::Run() {
+  if (!name_.empty())
+    rtc::SetCurrentThreadName(name_.c_str());
+  do {
+    // The interface contract of Start/Stop is that for a successfull call to
+    // Start, there should be at least one call to the run function.  So we
+    // call the function before checking |stop_|.
+    if (!run_function_(obj_))
+      break;
+#if defined(WEBRTC_WIN)
+    // Alertable sleep to permit RaiseFlag to run and update |stop_|.
+    SleepEx(0, true);
+  } while (!stop_);
+#else
+  } while (!stop_event_.Wait(0));
+#endif  // defined(WEBRTC_WIN)
+}
+
+#if defined(WEBRTC_WIN)
+void PlatformUIThread::Run() {
+  if (!InternalInit()) {
+    assert(false);
+  }
+  PlatformThread::Run();
+  // Don't need to DestroyWindow(hwnd_) due to WM_CLOSE->WM_DESTROY handling
+}
+
+void PlatformUIThread::NativeEventCallback() {
+  if (!run_function_) {
+    stop_ = true;
+    return;
+  }
+  stop_ = !run_function_(obj_);
+}
+
+/* static */
+LRESULT CALLBACK
+PlatformUIThread::EventWindowProc(HWND hwnd, UINT uMsg, WPARAM wParam, LPARAM lParam) {
+  if (uMsg == WM_DESTROY) {
+    RemovePropW(hwnd, kThisProperty);
+    PostQuitMessage(0);
+    return 0;
+  }
+
+  PlatformUIThread *twui = static_cast<PlatformUIThread*>(GetPropW(hwnd, kThisProperty));
+  if (!twui) {
+    return DefWindowProc(hwnd, uMsg, wParam, lParam);
+  }
+
+  if ((uMsg == static_reg_windows_msg && uMsg != WM_NULL) ||
+      (uMsg == WM_TIMER && wParam == kTimerId)) {
+    twui->NativeEventCallback();
+    return 0;
+  }
+
+  return DefWindowProc(hwnd, uMsg, wParam, lParam);
+}
+#endif
+
+bool PlatformThread::SetPriority(ThreadPriority priority) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(IsRunning());
+#if defined(WEBRTC_WIN)
+  return SetThreadPriority(thread_, priority) != FALSE;
+#elif defined(__native_client__)
+  // Setting thread priorities is not supported in NaCl.
+  return true;
+#elif defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_LINUX)
+  // TODO(tommi): Switch to the same mechanism as Chromium uses for changing
+  // thread priorities.
+  return true;
+#else
+#ifdef WEBRTC_THREAD_RR
+  const int policy = SCHED_RR;
+#else
+  const int policy = SCHED_FIFO;
+#endif
+  const int min_prio = sched_get_priority_min(policy);
+  const int max_prio = sched_get_priority_max(policy);
+  if (min_prio == -1 || max_prio == -1) {
+    return false;
+  }
+
+  if (max_prio - min_prio <= 2)
+    return false;
+
+  // Convert webrtc priority to system priorities:
+  sched_param param;
+  const int top_prio = max_prio - 1;
+  const int low_prio = min_prio + 1;
+  switch (priority) {
+    case kLowPriority:
+      param.sched_priority = low_prio;
+      break;
+    case kNormalPriority:
+      // The -1 ensures that the kHighPriority is always greater or equal to
+      // kNormalPriority.
+      param.sched_priority = (low_prio + top_prio - 1) / 2;
+      break;
+    case kHighPriority:
+      param.sched_priority = std::max(top_prio - 2, low_prio);
+      break;
+    case kHighestPriority:
+      param.sched_priority = std::max(top_prio - 1, low_prio);
+      break;
+    case kRealtimePriority:
+      param.sched_priority = top_prio;
+      break;
+  }
+  return pthread_setschedparam(thread_, policy, &param) == 0;
+#endif  // defined(WEBRTC_WIN)
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/platform_thread.h b/media/webrtc/trunk/webrtc/base/platform_thread.h
new file mode 100644
index 0000000000..b7d1023107
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/platform_thread.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PLATFORM_THREAD_H_
+#define WEBRTC_BASE_PLATFORM_THREAD_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread_types.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+
+namespace rtc {
+
+PlatformThreadId CurrentThreadId();
+PlatformThreadRef CurrentThreadRef();
+
+// Compares two thread identifiers for equality.
+bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b);
+
+// Sets the current thread name.
+void SetCurrentThreadName(const char* name);
+
+// Callback function that the spawned thread will enter once spawned.
+// A return value of false is interpreted as that the function has no
+// more work to do and that the thread can be released.
+typedef bool (*ThreadRunFunction)(void*);
+
+enum ThreadPriority {
+#ifdef WEBRTC_WIN
+  kLowPriority = THREAD_PRIORITY_BELOW_NORMAL,
+  kNormalPriority = THREAD_PRIORITY_NORMAL,
+  kHighPriority = THREAD_PRIORITY_ABOVE_NORMAL,
+  kHighestPriority = THREAD_PRIORITY_HIGHEST,
+  kRealtimePriority = THREAD_PRIORITY_TIME_CRITICAL
+#else
+  kLowPriority = 1,
+  kNormalPriority = 2,
+  kHighPriority = 3,
+  kHighestPriority = 4,
+  kRealtimePriority = 5
+#endif
+};
+
+// Represents a simple worker thread.  The implementation must be assumed
+// to be single threaded, meaning that all methods of the class, must be
+// called from the same thread, including instantiation.
+class PlatformThread {
+ public:
+  PlatformThread(ThreadRunFunction func, void* obj, const char* thread_name);
+  virtual ~PlatformThread();
+
+  // Spawns a thread and tries to set thread priority according to the priority
+  // from when CreateThread was called.
+  void Start();
+
+  bool IsRunning() const;
+
+  // Stops (joins) the spawned thread.
+  virtual void Stop();
+
+  // Set the priority of the thread. Must be called when thread is running.
+  bool SetPriority(ThreadPriority priority);
+
+ protected:
+  virtual void Run();
+
+  ThreadRunFunction const run_function_;
+  void* const obj_;
+  // TODO(pbos): Make sure call sites use string literals and update to a const
+  // char* instead of a std::string.
+  const std::string name_;
+  rtc::ThreadChecker thread_checker_;
+#if defined(WEBRTC_WIN)
+  static DWORD WINAPI StartThread(void* param);
+
+  bool stop_;
+  HANDLE thread_;
+#else
+  static void* StartThread(void* param);
+
+  rtc::Event stop_event_;
+
+  pthread_t thread_;
+#endif  // defined(WEBRTC_WIN)
+  RTC_DISALLOW_COPY_AND_ASSIGN(PlatformThread);
+};
+
+#if defined(WEBRTC_WIN)
+class PlatformUIThread : public PlatformThread {
+ public:
+  PlatformUIThread(ThreadRunFunction func, void* obj,
+		  const char* thread_name) :
+  PlatformThread(func, obj, thread_name),
+  hwnd_(nullptr),
+  timerid_(0),
+  timeout_(0) {
+ }
+ virtual ~PlatformUIThread() {}
+
+ virtual void Stop() override;
+
+ /**
+  * Request an async callback soon.
+  */
+ void RequestCallback();
+
+ /**
+  * Request a recurring callback.
+  */
+ bool RequestCallbackTimer(unsigned int milliseconds);
+
+ protected:
+  virtual void Run() override;
+
+ private:
+  static LRESULT CALLBACK EventWindowProc(HWND, UINT, WPARAM, LPARAM);
+  void NativeEventCallback();
+  bool InternalInit();
+
+  HWND hwnd_;
+  UINT_PTR timerid_;
+  unsigned int timeout_;
+};
+#endif
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_PLATFORM_THREAD_H_
diff --git a/media/webrtc/trunk/webrtc/base/platform_thread_types.h b/media/webrtc/trunk/webrtc/base/platform_thread_types.h
new file mode 100644
index 0000000000..546fffd96d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/platform_thread_types.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
+#define WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
+
+#if defined(WEBRTC_WIN)
+#include <winsock2.h>
+#include <windows.h>
+#elif defined(WEBRTC_POSIX)
+#include <pthread.h>
+#include <unistd.h>
+#endif
+
+namespace rtc {
+#if defined(WEBRTC_WIN)
+typedef DWORD PlatformThreadId;
+typedef DWORD PlatformThreadRef;
+#elif defined(WEBRTC_POSIX)
+typedef pid_t PlatformThreadId;
+typedef pthread_t PlatformThreadRef;
+#endif
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_unittest.cc b/media/webrtc/trunk/webrtc/base/platform_thread_unittest.cc
similarity index 66%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/thread_unittest.cc
rename to media/webrtc/trunk/webrtc/base/platform_thread_unittest.cc
index 854f98bbd9..f9db8e34a3 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/platform_thread_unittest.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 
 namespace webrtc {
 
@@ -22,11 +22,10 @@ bool NullRunFunction(void* obj) {
   return true;
 }
 
-TEST(ThreadTest, StartStop) {
-  rtc::scoped_ptr<ThreadWrapper> thread = ThreadWrapper::CreateThread(
-      &NullRunFunction, nullptr, "ThreadTest");
-  ASSERT_TRUE(thread->Start());
-  EXPECT_TRUE(thread->Stop());
+TEST(PlatformThreadTest, StartStop) {
+  rtc::PlatformThread thread(&NullRunFunction, nullptr, "PlatformThreadTest");
+  thread.Start();
+  thread.Stop();
 }
 
 // Function that sets a boolean.
@@ -37,14 +36,13 @@ bool SetFlagRunFunction(void* obj) {
   return true;
 }
 
-TEST(ThreadTest, RunFunctionIsCalled) {
+TEST(PlatformThreadTest, RunFunctionIsCalled) {
   bool flag = false;
-  rtc::scoped_ptr<ThreadWrapper> thread = ThreadWrapper::CreateThread(
-      &SetFlagRunFunction, &flag, "RunFunctionIsCalled");
-  ASSERT_TRUE(thread->Start());
+  rtc::PlatformThread thread(&SetFlagRunFunction, &flag, "RunFunctionIsCalled");
+  thread.Start();
 
   // At this point, the flag may be either true or false.
-  EXPECT_TRUE(thread->Stop());
+  thread.Stop();
 
   // We expect the thread to have run at least once.
   EXPECT_TRUE(flag);
diff --git a/media/webrtc/trunk/webrtc/base/profiler.cc b/media/webrtc/trunk/webrtc/base/profiler.cc
index e0bd431cf6..873b1989f7 100644
--- a/media/webrtc/trunk/webrtc/base/profiler.cc
+++ b/media/webrtc/trunk/webrtc/base/profiler.cc
@@ -55,7 +55,7 @@ void ProfilerEvent::Start() {
   ++start_count_;
 }
 
-void ProfilerEvent::Stop(uint64 stop_time) {
+void ProfilerEvent::Stop(uint64_t stop_time) {
   --start_count_;
   ASSERT(start_count_ >= 0);
   if (start_count_ == 0) {
@@ -89,7 +89,7 @@ double ProfilerEvent::standard_deviation() const {
 Profiler::~Profiler() = default;
 
 Profiler* Profiler::Instance() {
-  LIBJINGLE_DEFINE_STATIC_LOCAL(Profiler, instance, ());
+  RTC_DEFINE_STATIC_LOCAL(Profiler, instance, ());
   return &instance;
 }
 
@@ -114,7 +114,7 @@ void Profiler::StartEvent(const std::string& event_name) {
 
 void Profiler::StopEvent(const std::string& event_name) {
   // Get the time ASAP, then wait for the lock.
-  uint64 stop_time = TimeNanos();
+  uint64_t stop_time = TimeNanos();
   SharedScope scope(&lock_);
   EventMap::iterator it = events_.find(event_name);
   if (it != events_.end()) {
diff --git a/media/webrtc/trunk/webrtc/base/profiler.h b/media/webrtc/trunk/webrtc/base/profiler.h
index 6289035743..419763fc8a 100644
--- a/media/webrtc/trunk/webrtc/base/profiler.h
+++ b/media/webrtc/trunk/webrtc/base/profiler.h
@@ -91,7 +91,7 @@ class ProfilerEvent {
   ProfilerEvent();
   void Start();
   void Stop();
-  void Stop(uint64 stop_time);
+  void Stop(uint64_t stop_time);
   double standard_deviation() const;
   double total_time() const { return total_time_; }
   double mean() const { return mean_; }
@@ -101,7 +101,7 @@ class ProfilerEvent {
   bool is_started() const { return start_count_ > 0; }
 
  private:
-  uint64 current_start_time_;
+  uint64_t current_start_time_;
   double total_time_;
   double mean_;
   double sum_of_squared_differences_;
@@ -134,7 +134,7 @@ class Profiler {
   EventMap events_;
   mutable SharedExclusiveLock lock_;
 
-  DISALLOW_COPY_AND_ASSIGN(Profiler);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Profiler);
 };
 
 // Starts an event on construction and stops it on destruction.
@@ -151,7 +151,7 @@ class ProfilerScope {
  private:
   std::string event_name_;
 
-  DISALLOW_COPY_AND_ASSIGN(ProfilerScope);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ProfilerScope);
 };
 
 std::ostream& operator<<(std::ostream& stream,
diff --git a/media/webrtc/trunk/webrtc/base/proxy_unittest.cc b/media/webrtc/trunk/webrtc/base/proxy_unittest.cc
index 03dc154a6f..d8a523fe17 100644
--- a/media/webrtc/trunk/webrtc/base/proxy_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/proxy_unittest.cc
@@ -17,7 +17,6 @@
 #include "webrtc/base/testclient.h"
 #include "webrtc/base/testechoserver.h"
 #include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using rtc::Socket;
 using rtc::Thread;
diff --git a/media/webrtc/trunk/webrtc/base/proxydetect.cc b/media/webrtc/trunk/webrtc/base/proxydetect.cc
index 7265f4fd96..30959ca1d3 100644
--- a/media/webrtc/trunk/webrtc/base/proxydetect.cc
+++ b/media/webrtc/trunk/webrtc/base/proxydetect.cc
@@ -13,7 +13,7 @@
 #if defined(WEBRTC_WIN)
 #include "webrtc/base/win32.h"
 #include <shlobj.h>
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"
@@ -27,8 +27,14 @@
 #include "macconversion.h"
 #endif
 
+#ifdef WEBRTC_IOS
+#include <CFNetwork/CFNetwork.h>
+#include "macconversion.h"
+#endif
+
 #include <map>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/httpcommon.h"
 #include "webrtc/base/httpcommon-inl.h"
@@ -40,7 +46,7 @@
 #define _TRY_JSPROXY 0
 #define _TRY_WM_FINDPROXY 0
 #define _TRY_IE_LAN_SETTINGS 1
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 // For all platforms try Firefox.
 #define _TRY_FIREFOX 1
@@ -193,7 +199,7 @@ typedef std::string tstring;
 std::string Utf8String(const tstring& str) { return str; }
 
 #endif  // !_UNICODE
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 bool ProxyItemMatch(const Url<char>& url, char * item, size_t len) {
   // hostname:443
@@ -208,16 +214,16 @@ bool ProxyItemMatch(const Url<char>& url, char * item, size_t len) {
   int a, b, c, d, m;
   int match = sscanf(item, "%d.%d.%d.%d/%d", &a, &b, &c, &d, &m);
   if (match >= 4) {
-    uint32 ip = ((a & 0xFF) << 24) | ((b & 0xFF) << 16) | ((c & 0xFF) << 8) |
-        (d & 0xFF);
+    uint32_t ip = ((a & 0xFF) << 24) | ((b & 0xFF) << 16) | ((c & 0xFF) << 8) |
+                  (d & 0xFF);
     if ((match < 5) || (m > 32))
       m = 32;
     else if (m < 0)
       m = 0;
-    uint32 mask = (m == 0) ? 0 : (~0UL) << (32 - m);
+    uint32_t mask = (m == 0) ? 0 : (~0UL) << (32 - m);
     SocketAddress addr(url.host(), 0);
     // TODO: Support IPv6 proxyitems. This code block is IPv4 only anyway.
-    return !addr.IsUnresolved() &&
+    return !addr.IsUnresolvedIP() &&
         ((addr.ipaddr().v4AddressAsHostOrderInteger() & mask) == (ip & mask));
   }
 
@@ -284,7 +290,7 @@ bool ParseProxy(const std::string& saddress, ProxyInfo* proxy) {
 
   ProxyType ptype;
   std::string host;
-  uint16 port;
+  uint16_t port;
 
   const char* address = saddress.c_str();
   while (*address) {
@@ -318,7 +324,7 @@ bool ParseProxy(const std::string& saddress, ProxyInfo* proxy) {
 
     *colon = 0;
     char * endptr;
-    port = static_cast<uint16>(strtol(colon + 1, &endptr, 0));
+    port = static_cast<uint16_t>(strtol(colon + 1, &endptr, 0));
     if (*endptr != 0) {
       LOG(LS_WARNING) << "Proxy address with invalid port [" << buffer << "]";
       continue;
@@ -392,8 +398,8 @@ bool GetFirefoxProfilePath(Pathname* path) {
     return false;
   }
   char buffer[NAME_MAX + 1];
-  if (0 != FSRefMakePath(&fr, reinterpret_cast<uint8*>(buffer),
-                         ARRAY_SIZE(buffer))) {
+  if (0 != FSRefMakePath(&fr, reinterpret_cast<uint8_t*>(buffer),
+                         arraysize(buffer))) {
     LOG(LS_ERROR) << "FSRefMakePath failed";
     return false;
   }
@@ -407,7 +413,7 @@ bool GetFirefoxProfilePath(Pathname* path) {
   path->SetFolder(std::string(user_home));
   path->AppendFolder(".mozilla");
   path->AppendFolder("firefox");
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
   return true;
 }
 
@@ -939,7 +945,7 @@ bool GetIeProxySettings(const char* agent, const char* url, ProxyInfo* proxy) {
   return true;
 }
 
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)  // WEBRTC_MAC && !defined(WEBRTC_IOS) specific implementation for reading system wide
             // proxy settings.
@@ -1179,6 +1185,56 @@ bool GetMacProxySettings(ProxyInfo* proxy) {
 }
 #endif  // WEBRTC_MAC && !defined(WEBRTC_IOS)
 
+#ifdef WEBRTC_IOS
+// iOS has only http proxy
+bool GetiOSProxySettings(ProxyInfo* proxy) {
+
+  bool result = false;
+
+  CFDictionaryRef proxy_dict = CFNetworkCopySystemProxySettings();
+  if (!proxy_dict) {
+    LOG(LS_ERROR) << "CFNetworkCopySystemProxySettings failed";
+    return false;
+  }
+
+  CFNumberRef proxiesHTTPEnable = (CFNumberRef)CFDictionaryGetValue(
+    proxy_dict, kCFNetworkProxiesHTTPEnable);
+  if (!p_isCFNumberTrue(proxiesHTTPEnable)) {
+    CFRelease(proxy_dict);
+    return false;
+  }
+
+  CFStringRef proxy_address = (CFStringRef)CFDictionaryGetValue(
+    proxy_dict, kCFNetworkProxiesHTTPProxy);
+  CFNumberRef proxy_port = (CFNumberRef)CFDictionaryGetValue(
+    proxy_dict, kCFNetworkProxiesHTTPPort);
+
+  // the data we need to construct the SocketAddress for the proxy.
+  std::string hostname;
+  int port;
+  if (p_convertHostCFStringRefToCPPString(proxy_address, hostname) &&
+      p_convertCFNumberToInt(proxy_port, &port)) {
+      // We have something enabled, with a hostname and a port.
+      // That's sufficient to set up the proxy info.
+      // Finally, try HTTP proxy. Note that flute doesn't
+      // differentiate between HTTPS and HTTP, hence we are using the
+      // same flute type here, ie. PROXY_HTTPS.
+      proxy->type = PROXY_HTTPS;
+
+      proxy->address.SetIP(hostname);
+      proxy->address.SetPort(port);
+      result = true;
+  }
+
+  // We created the dictionary with something that had the
+  // word 'copy' in it, so we have to release it, according
+  // to the Carbon memory management standards.
+  CFRelease(proxy_dict);
+
+  return result;
+}
+#endif // WEBRTC_IOS
+
 bool AutoDetectProxySettings(const char* agent, const char* url,
                              ProxyInfo* proxy) {
 #if defined(WEBRTC_WIN)
@@ -1195,6 +1251,8 @@ bool GetSystemDefaultProxySettings(const char* agent, const char* url,
   return GetIeProxySettings(agent, url, proxy);
 #elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
   return GetMacProxySettings(proxy);
+#elif defined(WEBRTC_IOS)
+  return GetiOSProxySettings(proxy);
 #else
   // TODO: Get System settings if browser is not firefox.
   return GetFirefoxProxySettings(url, proxy);
@@ -1222,7 +1280,7 @@ bool GetProxySettingsForUrl(const char* agent, const char* url,
         result = GetIeProxySettings(agent, url, proxy);
       }
       break;
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
     default:
       result = GetSystemDefaultProxySettings(agent, url, proxy);
       break;
diff --git a/media/webrtc/trunk/webrtc/base/proxyserver.cc b/media/webrtc/trunk/webrtc/base/proxyserver.cc
index 8f12a99f98..d91a92fbde 100644
--- a/media/webrtc/trunk/webrtc/base/proxyserver.cc
+++ b/media/webrtc/trunk/webrtc/base/proxyserver.cc
@@ -36,6 +36,10 @@ ProxyServer::~ProxyServer() {
   }
 }
 
+SocketAddress ProxyServer::GetServerAddress() {
+  return server_socket_->GetLocalAddress();
+}
+
 void ProxyServer::OnAcceptEvent(AsyncSocket* socket) {
   ASSERT(socket != NULL && socket == server_socket_.get());
   AsyncSocket* int_socket = socket->Accept(NULL);
diff --git a/media/webrtc/trunk/webrtc/base/proxyserver.h b/media/webrtc/trunk/webrtc/base/proxyserver.h
index 5418a97c65..adb26ae9d0 100644
--- a/media/webrtc/trunk/webrtc/base/proxyserver.h
+++ b/media/webrtc/trunk/webrtc/base/proxyserver.h
@@ -55,7 +55,7 @@ class ProxyBinding : public sigslot::has_slots<> {
   bool connected_;
   FifoBuffer out_buffer_;
   FifoBuffer in_buffer_;
-  DISALLOW_EVIL_CONSTRUCTORS(ProxyBinding);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ProxyBinding);
 };
 
 class ProxyServer : public sigslot::has_slots<> {
@@ -64,6 +64,9 @@ class ProxyServer : public sigslot::has_slots<> {
               SocketFactory* ext_factory, const SocketAddress& ext_ip);
   ~ProxyServer() override;
 
+  // Returns the address to which the proxy server is bound
+  SocketAddress GetServerAddress();
+
  protected:
   void OnAcceptEvent(AsyncSocket* socket);
   virtual AsyncProxyServerSocket* WrapSocket(AsyncSocket* socket) = 0;
@@ -75,7 +78,7 @@ class ProxyServer : public sigslot::has_slots<> {
   SocketAddress ext_ip_;
   scoped_ptr<AsyncSocket> server_socket_;
   BindingList bindings_;
-  DISALLOW_EVIL_CONSTRUCTORS(ProxyServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ProxyServer);
 };
 
 // SocksProxyServer is a simple extension of ProxyServer to implement SOCKS.
@@ -87,7 +90,7 @@ class SocksProxyServer : public ProxyServer {
   }
  protected:
   AsyncProxyServerSocket* WrapSocket(AsyncSocket* socket) override;
-  DISALLOW_EVIL_CONSTRUCTORS(SocksProxyServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SocksProxyServer);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/random.cc b/media/webrtc/trunk/webrtc/base/random.cc
new file mode 100644
index 0000000000..14a9faf5b3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/random.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/base/random.h"
+
+#include <math.h>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+Random::Random(uint64_t seed) {
+  RTC_DCHECK(seed != 0x0ull);
+  state_ = seed;
+}
+
+uint32_t Random::Rand(uint32_t t) {
+  // Casting the output to 32 bits will give an almost uniform number.
+  // Pr[x=0] = (2^32-1) / (2^64-1)
+  // Pr[x=k] = 2^32 / (2^64-1) for k!=0
+  // Uniform would be Pr[x=k] = 2^32 / 2^64 for all 32-bit integers k.
+  uint32_t x = NextOutput();
+  // If x / 2^32 is uniform on [0,1), then x / 2^32 * (t+1) is uniform on
+  // the interval [0,t+1), so the integer part is uniform on [0,t].
+  uint64_t result = x * (static_cast<uint64_t>(t) + 1);
+  result >>= 32;
+  return result;
+}
+
+uint32_t Random::Rand(uint32_t low, uint32_t high) {
+  RTC_DCHECK(low <= high);
+  return Rand(high - low) + low;
+}
+
+int32_t Random::Rand(int32_t low, int32_t high) {
+  RTC_DCHECK(low <= high);
+  // We rely on subtraction (and addition) to be the same for signed and
+  // unsigned numbers in two-complement representation. Thus, although
+  // high - low might be negative as an int, it is the correct difference
+  // when interpreted as an unsigned.
+  return Rand(high - low) + low;
+}
+
+template <>
+float Random::Rand<float>() {
+  double result = NextOutput() - 1;
+  result = result / 0xFFFFFFFFFFFFFFFEull;
+  return static_cast<float>(result);
+}
+
+template <>
+double Random::Rand<double>() {
+  double result = NextOutput() - 1;
+  result = result / 0xFFFFFFFFFFFFFFFEull;
+  return result;
+}
+
+template <>
+bool Random::Rand<bool>() {
+  return Rand(0, 1) == 1;
+}
+
+double Random::Gaussian(double mean, double standard_deviation) {
+  // Creating a Normal distribution variable from two independent uniform
+  // variables based on the Box-Muller transform, which is defined on the
+  // interval (0, 1]. Note that we rely on NextOutput to generate integers
+  // in the range [1, 2^64-1]. Normally this behavior is a bit frustrating,
+  // but here it is exactly what we need.
+  const double kPi = 3.14159265358979323846;
+  double u1 = static_cast<double>(NextOutput()) / 0xFFFFFFFFFFFFFFFFull;
+  double u2 = static_cast<double>(NextOutput()) / 0xFFFFFFFFFFFFFFFFull;
+  return mean + standard_deviation * sqrt(-2 * log(u1)) * cos(2 * kPi * u2);
+}
+
+double Random::Exponential(double lambda) {
+  double uniform = Rand<double>();
+  return -log(uniform) / lambda;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/base/random.h b/media/webrtc/trunk/webrtc/base/random.h
new file mode 100644
index 0000000000..647b84c9c9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/random.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RANDOM_H_
+#define WEBRTC_BASE_RANDOM_H_
+
+#include <limits>
+
+#include "webrtc/typedefs.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+class Random {
+ public:
+  explicit Random(uint64_t seed);
+
+  // Return pseudo-random integer of the specified type.
+  // We need to limit the size to 32 bits to keep the output close to uniform.
+  template <typename T>
+  T Rand() {
+    static_assert(std::numeric_limits<T>::is_integer &&
+                      std::numeric_limits<T>::radix == 2 &&
+                      std::numeric_limits<T>::digits <= 32,
+                  "Rand is only supported for built-in integer types that are "
+                  "32 bits or smaller.");
+    return static_cast<T>(NextOutput());
+  }
+
+  // Uniformly distributed pseudo-random number in the interval [0, t].
+  uint32_t Rand(uint32_t t);
+
+  // Uniformly distributed pseudo-random number in the interval [low, high].
+  uint32_t Rand(uint32_t low, uint32_t high);
+
+  // Uniformly distributed pseudo-random number in the interval [low, high].
+  int32_t Rand(int32_t low, int32_t high);
+
+  // Normal Distribution.
+  double Gaussian(double mean, double standard_deviation);
+
+  // Exponential Distribution.
+  double Exponential(double lambda);
+
+ private:
+  // Outputs a nonzero 64-bit random number.
+  uint64_t NextOutput() {
+    state_ ^= state_ >> 12;
+    state_ ^= state_ << 25;
+    state_ ^= state_ >> 27;
+    RTC_DCHECK(state_ != 0x0ULL);
+    return state_ * 2685821657736338717ull;
+  }
+
+  uint64_t state_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Random);
+};
+
+// Return pseudo-random number in the interval [0.0, 1.0).
+template <>
+float Random::Rand<float>();
+
+// Return pseudo-random number in the interval [0.0, 1.0).
+template <>
+double Random::Rand<double>();
+
+// Return pseudo-random boolean value.
+template <>
+bool Random::Rand<bool>();
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_BASE_RANDOM_H_
diff --git a/media/webrtc/trunk/webrtc/base/random_unittest.cc b/media/webrtc/trunk/webrtc/base/random_unittest.cc
new file mode 100644
index 0000000000..febae1c28f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/random_unittest.cc
@@ -0,0 +1,302 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+
+#include <limits>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+
+namespace webrtc {
+
+namespace {
+// Computes the positive remainder of x/n.
+template <typename T>
+T fdiv_remainder(T x, T n) {
+  RTC_CHECK_GE(n, static_cast<T>(0));
+  T remainder = x % n;
+  if (remainder < 0)
+    remainder += n;
+  return remainder;
+}
+}  // namespace
+
+// Sample a number of random integers of type T. Divide them into buckets
+// based on the remainder when dividing by bucket_count and check that each
+// bucket gets roughly the expected number of elements.
+template <typename T>
+void UniformBucketTest(T bucket_count, int samples, Random* prng) {
+  std::vector<int> buckets(bucket_count, 0);
+
+  uint64_t total_values = 1ull << (std::numeric_limits<T>::digits +
+                                   std::numeric_limits<T>::is_signed);
+  T upper_limit =
+      std::numeric_limits<T>::max() -
+      static_cast<T>(total_values % static_cast<uint64_t>(bucket_count));
+  ASSERT_GT(upper_limit, std::numeric_limits<T>::max() / 2);
+
+  for (int i = 0; i < samples; i++) {
+    T sample;
+    do {
+      // We exclude a few numbers from the range so that it is divisible by
+      // the number of buckets. If we are unlucky and hit one of the excluded
+      // numbers we just resample. Note that if the number of buckets is a
+      // power of 2, then we don't have to exclude anything.
+      sample = prng->Rand<T>();
+    } while (sample > upper_limit);
+    buckets[fdiv_remainder(sample, bucket_count)]++;
+  }
+
+  for (T i = 0; i < bucket_count; i++) {
+    // Expect the result to be within 3 standard deviations of the mean.
+    EXPECT_NEAR(buckets[i], samples / bucket_count,
+                3 * sqrt(samples / bucket_count));
+  }
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestSignedChar) {
+  Random prng(7297352569824ull);
+  UniformBucketTest<signed char>(64, 640000, &prng);
+  UniformBucketTest<signed char>(11, 440000, &prng);
+  UniformBucketTest<signed char>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestUnsignedChar) {
+  Random prng(7297352569824ull);
+  UniformBucketTest<unsigned char>(64, 640000, &prng);
+  UniformBucketTest<unsigned char>(11, 440000, &prng);
+  UniformBucketTest<unsigned char>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestSignedShort) {
+  Random prng(7297352569824ull);
+  UniformBucketTest<int16_t>(64, 640000, &prng);
+  UniformBucketTest<int16_t>(11, 440000, &prng);
+  UniformBucketTest<int16_t>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestUnsignedShort) {
+  Random prng(7297352569824ull);
+  UniformBucketTest<uint16_t>(64, 640000, &prng);
+  UniformBucketTest<uint16_t>(11, 440000, &prng);
+  UniformBucketTest<uint16_t>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestSignedInt) {
+  Random prng(7297352569824ull);
+  UniformBucketTest<signed int>(64, 640000, &prng);
+  UniformBucketTest<signed int>(11, 440000, &prng);
+  UniformBucketTest<signed int>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestUnsignedInt) {
+  Random prng(7297352569824ull);
+  UniformBucketTest<unsigned int>(64, 640000, &prng);
+  UniformBucketTest<unsigned int>(11, 440000, &prng);
+  UniformBucketTest<unsigned int>(3, 270000, &prng);
+}
+
+// The range of the random numbers is divided into bucket_count intervals
+// of consecutive numbers. Check that approximately equally many numbers
+// from each inteval are generated.
+void BucketTestSignedInterval(unsigned int bucket_count,
+                              unsigned int samples,
+                              int32_t low,
+                              int32_t high,
+                              int sigma_level,
+                              Random* prng) {
+  std::vector<unsigned int> buckets(bucket_count, 0);
+
+  ASSERT_GE(high, low);
+  ASSERT_GE(bucket_count, 2u);
+  uint32_t interval = static_cast<uint32_t>(high - low + 1);
+  uint32_t numbers_per_bucket;
+  if (interval == 0) {
+    // The computation high - low + 1 should be 2^32 but overflowed
+    // Hence, bucket_count must be a power of 2
+    ASSERT_EQ(bucket_count & (bucket_count - 1), 0u);
+    numbers_per_bucket = (0x80000000u / bucket_count) * 2;
+  } else {
+    ASSERT_EQ(interval % bucket_count, 0u);
+    numbers_per_bucket = interval / bucket_count;
+  }
+
+  for (unsigned int i = 0; i < samples; i++) {
+    int32_t sample = prng->Rand(low, high);
+    EXPECT_LE(low, sample);
+    EXPECT_GE(high, sample);
+    buckets[static_cast<uint32_t>(sample - low) / numbers_per_bucket]++;
+  }
+
+  for (unsigned int i = 0; i < bucket_count; i++) {
+    // Expect the result to be within 3 standard deviations of the mean,
+    // or more generally, within sigma_level standard deviations of the mean.
+    double mean = static_cast<double>(samples) / bucket_count;
+    EXPECT_NEAR(buckets[i], mean, sigma_level * sqrt(mean));
+  }
+}
+
+// The range of the random numbers is divided into bucket_count intervals
+// of consecutive numbers. Check that approximately equally many numbers
+// from each inteval are generated.
+void BucketTestUnsignedInterval(unsigned int bucket_count,
+                                unsigned int samples,
+                                uint32_t low,
+                                uint32_t high,
+                                int sigma_level,
+                                Random* prng) {
+  std::vector<unsigned int> buckets(bucket_count, 0);
+
+  ASSERT_GE(high, low);
+  ASSERT_GE(bucket_count, 2u);
+  uint32_t interval = static_cast<uint32_t>(high - low + 1);
+  uint32_t numbers_per_bucket;
+  if (interval == 0) {
+    // The computation high - low + 1 should be 2^32 but overflowed
+    // Hence, bucket_count must be a power of 2
+    ASSERT_EQ(bucket_count & (bucket_count - 1), 0u);
+    numbers_per_bucket = (0x80000000u / bucket_count) * 2;
+  } else {
+    ASSERT_EQ(interval % bucket_count, 0u);
+    numbers_per_bucket = interval / bucket_count;
+  }
+
+  for (unsigned int i = 0; i < samples; i++) {
+    uint32_t sample = prng->Rand(low, high);
+    EXPECT_LE(low, sample);
+    EXPECT_GE(high, sample);
+    buckets[static_cast<uint32_t>(sample - low) / numbers_per_bucket]++;
+  }
+
+  for (unsigned int i = 0; i < bucket_count; i++) {
+    // Expect the result to be within 3 standard deviations of the mean,
+    // or more generally, within sigma_level standard deviations of the mean.
+    double mean = static_cast<double>(samples) / bucket_count;
+    EXPECT_NEAR(buckets[i], mean, sigma_level * sqrt(mean));
+  }
+}
+
+TEST(RandomNumberGeneratorTest, UniformUnsignedInterval) {
+  Random prng(299792458ull);
+  BucketTestUnsignedInterval(2, 100000, 0, 1, 3, &prng);
+  BucketTestUnsignedInterval(7, 100000, 1, 14, 3, &prng);
+  BucketTestUnsignedInterval(11, 100000, 1000, 1010, 3, &prng);
+  BucketTestUnsignedInterval(100, 100000, 0, 99, 3, &prng);
+  BucketTestUnsignedInterval(2, 100000, 0, 4294967295, 3, &prng);
+  BucketTestUnsignedInterval(17, 100000, 455, 2147484110, 3, &prng);
+  // 99.7% of all samples will be within 3 standard deviations of the mean,
+  // but since we test 1000 buckets we allow an interval of 4 sigma.
+  BucketTestUnsignedInterval(1000, 1000000, 0, 2147483999, 4, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, UniformSignedInterval) {
+  Random prng(66260695729ull);
+  BucketTestSignedInterval(2, 100000, 0, 1, 3, &prng);
+  BucketTestSignedInterval(7, 100000, -2, 4, 3, &prng);
+  BucketTestSignedInterval(11, 100000, 1000, 1010, 3, &prng);
+  BucketTestSignedInterval(100, 100000, 0, 99, 3, &prng);
+  BucketTestSignedInterval(2, 100000, std::numeric_limits<int32_t>::min(),
+                           std::numeric_limits<int32_t>::max(), 3, &prng);
+  BucketTestSignedInterval(17, 100000, -1073741826, 1073741829, 3, &prng);
+  // 99.7% of all samples will be within 3 standard deviations of the mean,
+  // but since we test 1000 buckets we allow an interval of 4 sigma.
+  BucketTestSignedInterval(1000, 1000000, -352, 2147483647, 4, &prng);
+}
+
+// The range of the random numbers is divided into bucket_count intervals
+// of consecutive numbers. Check that approximately equally many numbers
+// from each inteval are generated.
+void BucketTestFloat(unsigned int bucket_count,
+                     unsigned int samples,
+                     int sigma_level,
+                     Random* prng) {
+  ASSERT_GE(bucket_count, 2u);
+  std::vector<unsigned int> buckets(bucket_count, 0);
+
+  for (unsigned int i = 0; i < samples; i++) {
+    uint32_t sample = bucket_count * prng->Rand<float>();
+    EXPECT_LE(0u, sample);
+    EXPECT_GE(bucket_count - 1, sample);
+    buckets[sample]++;
+  }
+
+  for (unsigned int i = 0; i < bucket_count; i++) {
+    // Expect the result to be within 3 standard deviations of the mean,
+    // or more generally, within sigma_level standard deviations of the mean.
+    double mean = static_cast<double>(samples) / bucket_count;
+    EXPECT_NEAR(buckets[i], mean, sigma_level * sqrt(mean));
+  }
+}
+
+TEST(RandomNumberGeneratorTest, UniformFloatInterval) {
+  Random prng(1380648813ull);
+  BucketTestFloat(100, 100000, 3, &prng);
+  // 99.7% of all samples will be within 3 standard deviations of the mean,
+  // but since we test 1000 buckets we allow an interval of 4 sigma.
+  // BucketTestSignedInterval(1000, 1000000, -352, 2147483647, 4, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, SignedHasSameBitPattern) {
+  Random prng_signed(66738480ull), prng_unsigned(66738480ull);
+
+  for (int i = 0; i < 1000; i++) {
+    signed int s = prng_signed.Rand<signed int>();
+    unsigned int u = prng_unsigned.Rand<unsigned int>();
+    EXPECT_EQ(u, static_cast<unsigned int>(s));
+  }
+
+  for (int i = 0; i < 1000; i++) {
+    int16_t s = prng_signed.Rand<int16_t>();
+    uint16_t u = prng_unsigned.Rand<uint16_t>();
+    EXPECT_EQ(u, static_cast<uint16_t>(s));
+  }
+
+  for (int i = 0; i < 1000; i++) {
+    signed char s = prng_signed.Rand<signed char>();
+    unsigned char u = prng_unsigned.Rand<unsigned char>();
+    EXPECT_EQ(u, static_cast<unsigned char>(s));
+  }
+}
+
+TEST(RandomNumberGeneratorTest, Gaussian) {
+  const int kN = 100000;
+  const int kBuckets = 100;
+  const double kMean = 49;
+  const double kStddev = 10;
+
+  Random prng(1256637061);
+
+  std::vector<unsigned int> buckets(kBuckets, 0);
+  for (int i = 0; i < kN; i++) {
+    int index = prng.Gaussian(kMean, kStddev) + 0.5;
+    if (index >= 0 && index < kBuckets) {
+      buckets[index]++;
+    }
+  }
+
+  const double kPi = 3.14159265358979323846;
+  const double kScale = 1 / (kStddev * sqrt(2.0 * kPi));
+  const double kDiv = -2.0 * kStddev * kStddev;
+  for (int n = 0; n < kBuckets; ++n) {
+    // Use Simpsons rule to estimate the probability that a random gaussian
+    // sample is in the interval [n-0.5, n+0.5].
+    double f_left = kScale * exp((n - kMean - 0.5) * (n - kMean - 0.5) / kDiv);
+    double f_mid = kScale * exp((n - kMean) * (n - kMean) / kDiv);
+    double f_right = kScale * exp((n - kMean + 0.5) * (n - kMean + 0.5) / kDiv);
+    double normal_dist = (f_left + 4 * f_mid + f_right) / 6;
+    // Expect the number of samples to be within 3 standard deviations
+    // (rounded up) of the expected number of samples in the bucket.
+    EXPECT_NEAR(buckets[n], kN * normal_dist, 3 * sqrt(kN * normal_dist) + 1);
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/base/ratetracker.cc b/media/webrtc/trunk/webrtc/base/ratetracker.cc
index e03bfe0f29..35521a8d3d 100644
--- a/media/webrtc/trunk/webrtc/base/ratetracker.cc
+++ b/media/webrtc/trunk/webrtc/base/ratetracker.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -9,56 +9,140 @@
  */
 
 #include "webrtc/base/ratetracker.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
 #include "webrtc/base/timeutils.h"
 
 namespace rtc {
 
-RateTracker::RateTracker()
-    : total_units_(0), units_second_(0),
-      last_units_second_time_(~0u),
-      last_units_second_calc_(0) {
+RateTracker::RateTracker(uint32_t bucket_milliseconds, size_t bucket_count)
+    : bucket_milliseconds_(bucket_milliseconds),
+      bucket_count_(bucket_count),
+      sample_buckets_(new size_t[bucket_count + 1]),
+      total_sample_count_(0u),
+      bucket_start_time_milliseconds_(~0u) {
+  RTC_CHECK(bucket_milliseconds > 0u);
+  RTC_CHECK(bucket_count > 0u);
 }
 
-size_t RateTracker::total_units() const {
-  return total_units_;
+RateTracker::~RateTracker() {
+  delete[] sample_buckets_;
 }
 
-size_t RateTracker::units_second() {
-  // Snapshot units / second calculator. Determine how many seconds have
-  // elapsed since our last reference point. If over 1 second, establish
-  // a new reference point that is an integer number of seconds since the
-  // last one, and compute the units over that interval.
-  uint32 current_time = Time();
-  if (last_units_second_time_ == ~0u) {
-      last_units_second_time_ = current_time;
-      last_units_second_calc_ = total_units_;
-  } else {
-    int delta = rtc::TimeDiff(current_time, last_units_second_time_);
-    if (delta >= 1000) {
-      int fraction_time = delta % 1000;
-      int seconds = delta / 1000;
-      int fraction_units =
-          static_cast<int>(total_units_ - last_units_second_calc_) *
-              fraction_time / delta;
-      // Compute "units received during the interval" / "seconds in interval"
-      units_second_ =
-          (total_units_ - last_units_second_calc_ - fraction_units) / seconds;
-      last_units_second_time_ = current_time - fraction_time;
-      last_units_second_calc_ = total_units_ - fraction_units;
-    }
+double RateTracker::ComputeRateForInterval(
+    uint32_t interval_milliseconds) const {
+  if (bucket_start_time_milliseconds_ == ~0u) {
+    return 0.0;
   }
-
-  return units_second_;
+  uint32_t current_time = Time();
+  // Calculate which buckets to sum up given the current time.  If the time
+  // has passed to a new bucket then we have to skip some of the oldest buckets.
+  uint32_t available_interval_milliseconds = std::min<uint32_t>(
+      interval_milliseconds,
+      bucket_milliseconds_ * static_cast<uint32_t>(bucket_count_));
+  // number of old buckets (i.e. after the current bucket in the ring buffer)
+  // that are expired given our current time interval.
+  size_t buckets_to_skip;
+  // Number of milliseconds of the first bucket that are not a portion of the
+  // current interval.
+  uint32_t milliseconds_to_skip;
+  if (current_time >
+      initialization_time_milliseconds_ + available_interval_milliseconds) {
+    uint32_t time_to_skip =
+        current_time - bucket_start_time_milliseconds_ +
+        static_cast<uint32_t>(bucket_count_) * bucket_milliseconds_ -
+        available_interval_milliseconds;
+    buckets_to_skip = time_to_skip / bucket_milliseconds_;
+    milliseconds_to_skip = time_to_skip % bucket_milliseconds_;
+  } else {
+    buckets_to_skip = bucket_count_ - current_bucket_;
+    milliseconds_to_skip = 0u;
+    available_interval_milliseconds =
+        TimeDiff(current_time, initialization_time_milliseconds_);
+  }
+  // If we're skipping all buckets that means that there have been no samples
+  // within the sampling interval so report 0.
+  if (buckets_to_skip > bucket_count_ ||
+      available_interval_milliseconds == 0u) {
+    return 0.0;
+  }
+  size_t start_bucket = NextBucketIndex(current_bucket_ + buckets_to_skip);
+  // Only count a portion of the first bucket according to how much of the
+  // first bucket is within the current interval.
+  size_t total_samples = ((sample_buckets_[start_bucket] *
+      (bucket_milliseconds_ - milliseconds_to_skip)) +
+      (bucket_milliseconds_ >> 1)) /
+      bucket_milliseconds_;
+  // All other buckets in the interval are counted in their entirety.
+  for (size_t i = NextBucketIndex(start_bucket);
+      i != NextBucketIndex(current_bucket_);
+      i = NextBucketIndex(i)) {
+    total_samples += sample_buckets_[i];
+  }
+  // Convert to samples per second.
+  return static_cast<double>(total_samples * 1000u) /
+      static_cast<double>(available_interval_milliseconds);
 }
 
-void RateTracker::Update(size_t units) {
-  if (last_units_second_time_ == ~0u)
-    last_units_second_time_ = Time();
-  total_units_ += units;
+double RateTracker::ComputeTotalRate() const {
+  if (bucket_start_time_milliseconds_ == ~0u) {
+    return 0.0;
+  }
+  uint32_t current_time = Time();
+  if (TimeIsLaterOrEqual(current_time, initialization_time_milliseconds_)) {
+    return 0.0;
+  }
+  return static_cast<double>(total_sample_count_ * 1000u) /
+      static_cast<double>(
+          TimeDiff(current_time, initialization_time_milliseconds_));
 }
 
-uint32 RateTracker::Time() const {
+size_t RateTracker::TotalSampleCount() const {
+  return total_sample_count_;
+}
+
+void RateTracker::AddSamples(size_t sample_count) {
+  EnsureInitialized();
+  uint32_t current_time = Time();
+  // Advance the current bucket as needed for the current time, and reset
+  // bucket counts as we advance.
+  for (size_t i = 0u; i <= bucket_count_ &&
+      current_time >= bucket_start_time_milliseconds_ + bucket_milliseconds_;
+      ++i) {
+    bucket_start_time_milliseconds_ += bucket_milliseconds_;
+    current_bucket_ = NextBucketIndex(current_bucket_);
+    sample_buckets_[current_bucket_] = 0u;
+  }
+  // Ensure that bucket_start_time_milliseconds_ is updated appropriately if
+  // the entire buffer of samples has been expired.
+  bucket_start_time_milliseconds_ += bucket_milliseconds_ *
+      ((current_time - bucket_start_time_milliseconds_) / bucket_milliseconds_);
+  // Add all samples in the bucket that includes the current time.
+  sample_buckets_[current_bucket_] += sample_count;
+  total_sample_count_ += sample_count;
+}
+
+uint32_t RateTracker::Time() const {
   return rtc::Time();
 }
 
+void RateTracker::EnsureInitialized() {
+  if (bucket_start_time_milliseconds_ == ~0u) {
+    initialization_time_milliseconds_ = Time();
+    bucket_start_time_milliseconds_ = initialization_time_milliseconds_;
+    current_bucket_ = 0u;
+    // We only need to initialize the first bucket because we reset buckets when
+    // current_bucket_ increments.
+    sample_buckets_[current_bucket_] = 0u;
+  }
+}
+
+size_t RateTracker::NextBucketIndex(size_t bucket_index) const {
+  return (bucket_index + 1u) % (bucket_count_ + 1u);
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/ratetracker.h b/media/webrtc/trunk/webrtc/base/ratetracker.h
index 575bff75a4..d49d7cacdd 100644
--- a/media/webrtc/trunk/webrtc/base/ratetracker.h
+++ b/media/webrtc/trunk/webrtc/base/ratetracker.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -16,25 +16,52 @@
 
 namespace rtc {
 
-// Computes instantaneous units per second.
+// Computes units per second over a given interval by tracking the units over
+// each bucket of a given size and calculating the instantaneous rate assuming
+// that over each bucket the rate was constant.
 class RateTracker {
  public:
-  RateTracker();
-  virtual ~RateTracker() {}
+  RateTracker(uint32_t bucket_milliseconds, size_t bucket_count);
+  virtual ~RateTracker();
 
-  size_t total_units() const;
-  size_t units_second();
-  void Update(size_t units);
+  // Computes the average rate over the most recent interval_milliseconds,
+  // or if the first sample was added within this period, computes the rate
+  // since the first sample was added.
+  double ComputeRateForInterval(uint32_t interval_milliseconds) const;
+
+  // Computes the average rate over the rate tracker's recording interval
+  // of bucket_milliseconds * bucket_count.
+  double ComputeRate() const {
+    return ComputeRateForInterval(bucket_milliseconds_ *
+                                  static_cast<uint32_t>(bucket_count_));
+  }
+
+  // Computes the average rate since the first sample was added to the
+  // rate tracker.
+  double ComputeTotalRate() const;
+
+  // The total number of samples added.
+  size_t TotalSampleCount() const;
+
+  // Reads the current time in order to determine the appropriate bucket for
+  // these samples, and increments the count for that bucket by sample_count.
+  void AddSamples(size_t sample_count);
 
  protected:
   // overrideable for tests
-  virtual uint32 Time() const;
+  virtual uint32_t Time() const;
 
  private:
-  size_t total_units_;
-  size_t units_second_;
-  uint32 last_units_second_time_;
-  size_t last_units_second_calc_;
+  void EnsureInitialized();
+  size_t NextBucketIndex(size_t bucket_index) const;
+
+  const uint32_t bucket_milliseconds_;
+  const size_t bucket_count_;
+  size_t* sample_buckets_;
+  size_t total_sample_count_;
+  size_t current_bucket_;
+  uint32_t bucket_start_time_milliseconds_;
+  uint32_t initialization_time_milliseconds_;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/ratetracker_unittest.cc b/media/webrtc/trunk/webrtc/base/ratetracker_unittest.cc
index 1c20fd05a2..2187282cd3 100644
--- a/media/webrtc/trunk/webrtc/base/ratetracker_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/ratetracker_unittest.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2010 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -15,67 +15,148 @@ namespace rtc {
 
 class RateTrackerForTest : public RateTracker {
  public:
-  RateTrackerForTest() : time_(0) {}
-  virtual uint32 Time() const { return time_; }
-  void AdvanceTime(uint32 delta) { time_ += delta; }
+  RateTrackerForTest() : RateTracker(100u, 10u), time_(0) {}
+  virtual uint32_t Time() const { return time_; }
+  void AdvanceTime(uint32_t delta) { time_ += delta; }
 
  private:
-  uint32 time_;
+  uint32_t time_;
 };
 
-TEST(RateTrackerTest, TestBasics) {
+TEST(RateTrackerTest, Test30FPS) {
   RateTrackerForTest tracker;
-  EXPECT_EQ(0U, tracker.total_units());
-  EXPECT_EQ(0U, tracker.units_second());
+
+  for (int i = 0; i < 300; ++i) {
+    tracker.AddSamples(1);
+    tracker.AdvanceTime(33);
+    if (i % 3 == 0) {
+      tracker.AdvanceTime(1);
+    }
+  }
+  EXPECT_DOUBLE_EQ(30.0, tracker.ComputeRateForInterval(50000u));
+}
+
+TEST(RateTrackerTest, Test60FPS) {
+  RateTrackerForTest tracker;
+
+  for (int i = 0; i < 300; ++i) {
+    tracker.AddSamples(1);
+    tracker.AdvanceTime(16);
+    if (i % 3 != 0) {
+      tracker.AdvanceTime(1);
+    }
+  }
+  EXPECT_DOUBLE_EQ(60.0, tracker.ComputeRateForInterval(1000u));
+}
+
+TEST(RateTrackerTest, TestRateTrackerBasics) {
+  RateTrackerForTest tracker;
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(1000u));
 
   // Add a sample.
-  tracker.Update(1234);
+  tracker.AddSamples(1234);
   // Advance the clock by 100 ms.
   tracker.AdvanceTime(100);
-  // total_units should advance, but units_second should stay 0.
-  EXPECT_EQ(1234U, tracker.total_units());
-  EXPECT_EQ(0U, tracker.units_second());
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeTotalRate());
 
   // Repeat.
-  tracker.Update(1234);
+  tracker.AddSamples(1234);
   tracker.AdvanceTime(100);
-  EXPECT_EQ(1234U * 2, tracker.total_units());
-  EXPECT_EQ(0U, tracker.units_second());
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeTotalRate());
 
   // Advance the clock by 800 ms, so we've elapsed a full second.
   // units_second should now be filled in properly.
   tracker.AdvanceTime(800);
-  EXPECT_EQ(1234U * 2, tracker.total_units());
-  EXPECT_EQ(1234U * 2, tracker.units_second());
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeTotalRate());
 
   // Poll the tracker again immediately. The reported rate should stay the same.
-  EXPECT_EQ(1234U * 2, tracker.total_units());
-  EXPECT_EQ(1234U * 2, tracker.units_second());
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeTotalRate());
 
   // Do nothing and advance by a second. We should drop down to zero.
   tracker.AdvanceTime(1000);
-  EXPECT_EQ(1234U * 2, tracker.total_units());
-  EXPECT_EQ(0U, tracker.units_second());
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ(1234.0, tracker.ComputeTotalRate());
 
   // Send a bunch of data at a constant rate for 5.5 "seconds".
   // We should report the rate properly.
   for (int i = 0; i < 5500; i += 100) {
-    tracker.Update(9876U);
+    tracker.AddSamples(9876U);
     tracker.AdvanceTime(100);
   }
-  EXPECT_EQ(9876U * 10, tracker.units_second());
+  EXPECT_DOUBLE_EQ(9876.0 * 10.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(9876.0 * 10.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U * 2 + 9876U * 55, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ((1234.0 * 2.0 + 9876.0 * 55.0) / 7.5,
+      tracker.ComputeTotalRate());
 
   // Advance the clock by 500 ms. Since we sent nothing over this half-second,
   // the reported rate should be reduced by half.
   tracker.AdvanceTime(500);
-  EXPECT_EQ(9876U * 5, tracker.units_second());
+  EXPECT_DOUBLE_EQ(9876.0 * 5.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(9876.0 * 5.0, tracker.ComputeRate());
+  EXPECT_EQ(1234U * 2 + 9876U * 55, tracker.TotalSampleCount());
+  EXPECT_DOUBLE_EQ((1234.0 * 2.0 + 9876.0 * 55.0) / 8.0,
+      tracker.ComputeTotalRate());
+
+  // Rate over the last half second should be zero.
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(500u));
+}
+
+TEST(RateTrackerTest, TestLongPeriodBetweenSamples) {
+  RateTrackerForTest tracker;
+  tracker.AddSamples(1);
+  tracker.AdvanceTime(1000);
+  EXPECT_DOUBLE_EQ(1.0, tracker.ComputeRate());
+
+  tracker.AdvanceTime(2000);
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRate());
+
+  tracker.AdvanceTime(2000);
+  tracker.AddSamples(1);
+  EXPECT_DOUBLE_EQ(1.0, tracker.ComputeRate());
+}
+
+TEST(RateTrackerTest, TestRolloff) {
+  RateTrackerForTest tracker;
+  for (int i = 0; i < 10; ++i) {
+    tracker.AddSamples(1U);
+    tracker.AdvanceTime(100);
+  }
+  EXPECT_DOUBLE_EQ(10.0, tracker.ComputeRate());
+
+  for (int i = 0; i < 10; ++i) {
+    tracker.AddSamples(1U);
+    tracker.AdvanceTime(50);
+  }
+  EXPECT_DOUBLE_EQ(15.0, tracker.ComputeRate());
+  EXPECT_DOUBLE_EQ(20.0, tracker.ComputeRateForInterval(500u));
+
+  for (int i = 0; i < 10; ++i) {
+    tracker.AddSamples(1U);
+    tracker.AdvanceTime(50);
+  }
+  EXPECT_DOUBLE_EQ(20.0, tracker.ComputeRate());
 }
 
 TEST(RateTrackerTest, TestGetUnitSecondsAfterInitialValue) {
   RateTrackerForTest tracker;
-  tracker.Update(1234);
+  tracker.AddSamples(1234);
   tracker.AdvanceTime(1000);
-  EXPECT_EQ(1234u, tracker.units_second());
+  EXPECT_DOUBLE_EQ(1234.0, tracker.ComputeRateForInterval(1000u));
 }
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/refcount.h b/media/webrtc/trunk/webrtc/base/refcount.h
index e7306d3df9..55ce23a348 100644
--- a/media/webrtc/trunk/webrtc/base/refcount.h
+++ b/media/webrtc/trunk/webrtc/base/refcount.h
@@ -8,20 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_APP_BASE_REFCOUNT_H_
-#define TALK_APP_BASE_REFCOUNT_H_
+#ifndef WEBRTC_BASE_REFCOUNT_H_
+#define WEBRTC_BASE_REFCOUNT_H_
 
 #include <string.h>
 
-#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/atomicops.h"
 
 namespace rtc {
 
 // Reference count interface.
 class RefCountInterface {
  public:
-  virtual int AddRef() = 0;
-  virtual int Release() = 0;
+  virtual int AddRef() const = 0;
+  virtual int Release() const = 0;
  protected:
   virtual ~RefCountInterface() {}
 };
@@ -95,12 +95,12 @@ class RefCountedObject : public T {
   : T(p1, p2, p3, p4, p5, p6, p7, p8, p9, p10, p11), ref_count_(0) {
   }
 
-  virtual int AddRef() {
-    return rtc::AtomicOps::Increment(&ref_count_);
+  virtual int AddRef() const {
+    return AtomicOps::Increment(&ref_count_);
   }
 
-  virtual int Release() {
-    int count = rtc::AtomicOps::Decrement(&ref_count_);
+  virtual int Release() const {
+    int count = AtomicOps::Decrement(&ref_count_);
     if (!count) {
       delete this;
     }
@@ -114,16 +114,16 @@ class RefCountedObject : public T {
   // barrier needed for the owning thread to act on the object, knowing that it
   // has exclusive access to the object.
   virtual bool HasOneRef() const {
-    return rtc::AtomicOps::Load(&ref_count_) == 1;
+    return AtomicOps::AcquireLoad(&ref_count_) == 1;
   }
 
  protected:
   virtual ~RefCountedObject() {
   }
 
-  volatile int ref_count_;
+  mutable volatile int ref_count_;
 };
 
 }  // namespace rtc
 
-#endif  // TALK_APP_BASE_REFCOUNT_H_
+#endif  // WEBRTC_BASE_REFCOUNT_H_
diff --git a/media/webrtc/trunk/webrtc/base/referencecountedsingletonfactory.h b/media/webrtc/trunk/webrtc/base/referencecountedsingletonfactory.h
index 7138c8c5e1..f955986827 100644
--- a/media/webrtc/trunk/webrtc/base/referencecountedsingletonfactory.h
+++ b/media/webrtc/trunk/webrtc/base/referencecountedsingletonfactory.h
@@ -77,7 +77,7 @@ class ReferenceCountedSingletonFactory {
   CriticalSection crit_;
   int ref_count_;
 
-  DISALLOW_COPY_AND_ASSIGN(ReferenceCountedSingletonFactory);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ReferenceCountedSingletonFactory);
 };
 
 template <typename Interface>
@@ -149,7 +149,7 @@ class rcsf_ptr {
   Interface* instance_;
   ReferenceCountedSingletonFactory<Interface>* factory_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(rcsf_ptr);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(rcsf_ptr);
 };
 
 };  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/rollingaccumulator.h b/media/webrtc/trunk/webrtc/base/rollingaccumulator.h
index 25434fb1b9..e105380191 100644
--- a/media/webrtc/trunk/webrtc/base/rollingaccumulator.h
+++ b/media/webrtc/trunk/webrtc/base/rollingaccumulator.h
@@ -165,7 +165,7 @@ class RollingAccumulator {
   mutable bool min_stale_;
   std::vector<T> samples_;
 
-  DISALLOW_COPY_AND_ASSIGN(RollingAccumulator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RollingAccumulator);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/rtccertificate.cc b/media/webrtc/trunk/webrtc/base/rtccertificate.cc
new file mode 100644
index 0000000000..7b764bd72e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/rtccertificate.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/rtccertificate.h"
+
+#include "webrtc/base/checks.h"
+
+namespace rtc {
+
+scoped_refptr<RTCCertificate> RTCCertificate::Create(
+    scoped_ptr<SSLIdentity> identity) {
+  return new RefCountedObject<RTCCertificate>(identity.release());
+}
+
+RTCCertificate::RTCCertificate(SSLIdentity* identity)
+    : identity_(identity) {
+  RTC_DCHECK(identity_);
+}
+
+RTCCertificate::~RTCCertificate() {
+}
+
+uint64_t RTCCertificate::Expires() const {
+  int64_t expires = ssl_certificate().CertificateExpirationTime();
+  if (expires != -1)
+    return static_cast<uint64_t>(expires) * kNumMillisecsPerSec;
+  // If the expiration time could not be retrieved return an expired timestamp.
+  return 0;  // = 1970-01-01
+}
+
+bool RTCCertificate::HasExpired(uint64_t now) const {
+  return Expires() <= now;
+}
+
+const SSLCertificate& RTCCertificate::ssl_certificate() const {
+  return identity_->certificate();
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/rtccertificate.h b/media/webrtc/trunk/webrtc/base/rtccertificate.h
new file mode 100644
index 0000000000..600739bc86
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/rtccertificate.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RTCCERTIFICATE_H_
+#define WEBRTC_BASE_RTCCERTIFICATE_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sslidentity.h"
+
+namespace rtc {
+
+// A thin abstraction layer between "lower level crypto stuff" like
+// SSLCertificate and WebRTC usage. Takes ownership of some lower level objects,
+// reference counting protects these from premature destruction.
+class RTCCertificate : public RefCountInterface {
+ public:
+  // Takes ownership of |identity|.
+  static scoped_refptr<RTCCertificate> Create(scoped_ptr<SSLIdentity> identity);
+
+  // Returns the expiration time in ms relative to epoch, 1970-01-01T00:00:00Z.
+  uint64_t Expires() const;
+  // Checks if the certificate has expired, where |now| is expressed in ms
+  // relative to epoch, 1970-01-01T00:00:00Z.
+  bool HasExpired(uint64_t now) const;
+  const SSLCertificate& ssl_certificate() const;
+
+  // TODO(hbos): If possible, remove once RTCCertificate and its
+  // ssl_certificate() is used in all relevant places. Should not pass around
+  // raw SSLIdentity* for the sake of accessing SSLIdentity::certificate().
+  // However, some places might need SSLIdentity* for its public/private key...
+  SSLIdentity* identity() const { return identity_.get(); }
+
+ protected:
+  explicit RTCCertificate(SSLIdentity* identity);
+  ~RTCCertificate() override;
+
+ private:
+  // The SSLIdentity is the owner of the SSLCertificate. To protect our
+  // ssl_certificate() we take ownership of |identity_|.
+  scoped_ptr<SSLIdentity> identity_;
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_RTCCERTIFICATE_H_
diff --git a/media/webrtc/trunk/webrtc/base/rtccertificate_unittests.cc b/media/webrtc/trunk/webrtc/base/rtccertificate_unittests.cc
new file mode 100644
index 0000000000..84c854478b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/base/rtccertificate_unittests.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/fakesslidentity.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/rtccertificate.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/sslidentity.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/base/timeutils.h"
+
+namespace rtc {
+
+namespace {
+
+static const char* kTestCertCommonName = "RTCCertificateTest's certificate";
+
+}  // namespace
+
+class RTCCertificateTest : public testing::Test {
+ public:
+  RTCCertificateTest() {}
+  ~RTCCertificateTest() {}
+
+ protected:
+  // Timestamp note:
+  //   All timestamps in this unittest are expressed in number of seconds since
+  // epoch, 1970-01-01T00:00:00Z (UTC). The RTCCertificate interface uses ms,
+  // but only seconds-precision is supported by SSLCertificate. To make the
+  // tests clearer we convert everything to seconds since the precision matters
+  // when generating certificates or comparing timestamps.
+  //   As a result, ExpiresSeconds and HasExpiredSeconds are used instead of
+  // RTCCertificate::Expires and ::HasExpired for ms -> s conversion.
+
+  uint64_t NowSeconds() const {
+    return TimeNanos() / kNumNanosecsPerSec;
+  }
+
+  uint64_t ExpiresSeconds(const scoped_refptr<RTCCertificate>& cert) const {
+    uint64_t exp_ms = cert->Expires();
+    uint64_t exp_s = exp_ms / kNumMillisecsPerSec;
+    // Make sure this did not result in loss of precision.
+    RTC_CHECK_EQ(exp_s * kNumMillisecsPerSec, exp_ms);
+    return exp_s;
+  }
+
+  bool HasExpiredSeconds(const scoped_refptr<RTCCertificate>& cert,
+                         uint64_t now_s) const {
+    return cert->HasExpired(now_s * kNumMillisecsPerSec);
+  }
+
+  // An RTC_CHECK ensures that |expires_s| this is in valid range of time_t as
+  // is required by SSLIdentityParams. On some 32-bit systems time_t is limited
+  // to < 2^31. On such systems this will fail for expiration times of year 2038
+  // or later.
+  scoped_refptr<RTCCertificate> GenerateCertificateWithExpires(
+      uint64_t expires_s) const {
+    RTC_CHECK(IsValueInRangeForNumericType<time_t>(expires_s));
+
+    SSLIdentityParams params;
+    params.common_name = kTestCertCommonName;
+    params.not_before = 0;
+    params.not_after = static_cast<time_t>(expires_s);
+    // Certificate type does not matter for our purposes, using ECDSA because it
+    // is fast to generate.
+    params.key_params = KeyParams::ECDSA();
+
+    scoped_ptr<SSLIdentity> identity(SSLIdentity::GenerateForTest(params));
+    return RTCCertificate::Create(std::move(identity));
+  }
+};
+
+TEST_F(RTCCertificateTest, NewCertificateNotExpired) {
+  // Generate a real certificate without specifying the expiration time.
+  // Certificate type doesn't matter, using ECDSA because it's fast to generate.
+  scoped_ptr<SSLIdentity> identity(
+      SSLIdentity::Generate(kTestCertCommonName, KeyParams::ECDSA()));
+  scoped_refptr<RTCCertificate> certificate =
+      RTCCertificate::Create(std::move(identity));
+
+  uint64_t now = NowSeconds();
+  EXPECT_FALSE(HasExpiredSeconds(certificate, now));
+  // Even without specifying the expiration time we would expect it to be valid
+  // for at least half an hour.
+  EXPECT_FALSE(HasExpiredSeconds(certificate, now + 30*60));
+}
+
+TEST_F(RTCCertificateTest, UsesExpiresAskedFor) {
+  uint64_t now = NowSeconds();
+  scoped_refptr<RTCCertificate> certificate =
+      GenerateCertificateWithExpires(now);
+  EXPECT_EQ(now, ExpiresSeconds(certificate));
+}
+
+TEST_F(RTCCertificateTest, ExpiresInOneSecond) {
+  // Generate a certificate that expires in 1s.
+  uint64_t now = NowSeconds();
+  scoped_refptr<RTCCertificate> certificate =
+      GenerateCertificateWithExpires(now + 1);
+  // Now it should not have expired.
+  EXPECT_FALSE(HasExpiredSeconds(certificate, now));
+  // In 2s it should have expired.
+  EXPECT_TRUE(HasExpiredSeconds(certificate, now + 2));
+}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/safe_conversions.h b/media/webrtc/trunk/webrtc/base/safe_conversions.h
index 7fc67cb67a..51239bc65d 100644
--- a/media/webrtc/trunk/webrtc/base/safe_conversions.h
+++ b/media/webrtc/trunk/webrtc/base/safe_conversions.h
@@ -32,13 +32,13 @@ inline bool IsValueInRangeForNumericType(Src value) {
 // overflow or underflow. NaN source will always trigger a CHECK.
 template <typename Dst, typename Src>
 inline Dst checked_cast(Src value) {
-  CHECK(IsValueInRangeForNumericType<Dst>(value));
+  RTC_CHECK(IsValueInRangeForNumericType<Dst>(value));
   return static_cast<Dst>(value);
 }
 
 // saturated_cast<> is analogous to static_cast<> for numeric types, except
 // that the specified numeric conversion will saturate rather than overflow or
-// underflow. NaN assignment to an integral will trigger a CHECK condition.
+// underflow. NaN assignment to an integral will trigger a RTC_CHECK condition.
 template <typename Dst, typename Src>
 inline Dst saturated_cast(Src value) {
   // Optimization for floating point values, which already saturate.
diff --git a/media/webrtc/trunk/webrtc/base/schanneladapter.cc b/media/webrtc/trunk/webrtc/base/schanneladapter.cc
deleted file mode 100644
index f231bcfe30..0000000000
--- a/media/webrtc/trunk/webrtc/base/schanneladapter.cc
+++ /dev/null
@@ -1,714 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/win32.h"
-#define SECURITY_WIN32
-#include <security.h>
-#include <schannel.h>
-
-#include <algorithm>
-#include <iomanip>
-#include <vector>
-
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/schanneladapter.h"
-#include "webrtc/base/sec_buffer.h"
-#include "webrtc/base/thread.h"
-
-namespace rtc {
-
-/////////////////////////////////////////////////////////////////////////////
-// SChannelAdapter
-/////////////////////////////////////////////////////////////////////////////
-
-extern const ConstantLabel SECURITY_ERRORS[];
-
-const ConstantLabel SCHANNEL_BUFFER_TYPES[] = {
-  KLABEL(SECBUFFER_EMPTY),              //  0
-  KLABEL(SECBUFFER_DATA),               //  1
-  KLABEL(SECBUFFER_TOKEN),              //  2
-  KLABEL(SECBUFFER_PKG_PARAMS),         //  3
-  KLABEL(SECBUFFER_MISSING),            //  4
-  KLABEL(SECBUFFER_EXTRA),              //  5
-  KLABEL(SECBUFFER_STREAM_TRAILER),     //  6
-  KLABEL(SECBUFFER_STREAM_HEADER),      //  7
-  KLABEL(SECBUFFER_MECHLIST),           // 11
-  KLABEL(SECBUFFER_MECHLIST_SIGNATURE), // 12
-  KLABEL(SECBUFFER_TARGET),             // 13
-  KLABEL(SECBUFFER_CHANNEL_BINDINGS),   // 14
-  LASTLABEL
-};
-
-void DescribeBuffer(LoggingSeverity severity, const char* prefix,
-                    const SecBuffer& sb) {
-  LOG_V(severity)
-    << prefix
-    << "(" << sb.cbBuffer
-    << ", " << FindLabel(sb.BufferType & ~SECBUFFER_ATTRMASK,
-                          SCHANNEL_BUFFER_TYPES)
-    << ", " << sb.pvBuffer << ")";
-}
-
-void DescribeBuffers(LoggingSeverity severity, const char* prefix,
-                     const SecBufferDesc* sbd) {
-  if (!LOG_CHECK_LEVEL_V(severity))
-    return;
-  LOG_V(severity) << prefix << "(";
-  for (size_t i=0; i<sbd->cBuffers; ++i) {
-    DescribeBuffer(severity, "  ", sbd->pBuffers[i]);
-  }
-  LOG_V(severity) << ")";
-}
-
-const ULONG SSL_FLAGS_DEFAULT = ISC_REQ_ALLOCATE_MEMORY
-                              | ISC_REQ_CONFIDENTIALITY
-                              | ISC_REQ_EXTENDED_ERROR
-                              | ISC_REQ_INTEGRITY
-                              | ISC_REQ_REPLAY_DETECT
-                              | ISC_REQ_SEQUENCE_DETECT
-                              | ISC_REQ_STREAM;
-                              //| ISC_REQ_USE_SUPPLIED_CREDS;
-
-typedef std::vector<char> SChannelBuffer;
-
-struct SChannelAdapter::SSLImpl {
-  CredHandle cred;
-  CtxtHandle ctx;
-  bool cred_init, ctx_init;
-  SChannelBuffer inbuf, outbuf, readable;
-  SecPkgContext_StreamSizes sizes;
-
-  SSLImpl() : cred_init(false), ctx_init(false) { }
-};
-
-SChannelAdapter::SChannelAdapter(AsyncSocket* socket)
-  : SSLAdapter(socket), state_(SSL_NONE), mode_(SSL_MODE_TLS),
-    restartable_(false), signal_close_(false), message_pending_(false),
-    impl_(new SSLImpl) {
-}
-
-SChannelAdapter::~SChannelAdapter() {
-  Cleanup();
-}
-
-void
-SChannelAdapter::SetMode(SSLMode mode) {
-  // SSL_MODE_DTLS isn't supported.
-  ASSERT(mode == SSL_MODE_TLS);
-  mode_ = mode;
-}
-
-int
-SChannelAdapter::StartSSL(const char* hostname, bool restartable) {
-  if (state_ != SSL_NONE)
-    return -1;
-
-  if (mode_ != SSL_MODE_TLS)
-    return -1;
-
-  ssl_host_name_ = hostname;
-  restartable_ = restartable;
-
-  if (socket_->GetState() != Socket::CS_CONNECTED) {
-    state_ = SSL_WAIT;
-    return 0;
-  }
-
-  state_ = SSL_CONNECTING;
-  if (int err = BeginSSL()) {
-    Error("BeginSSL", err, false);
-    return err;
-  }
-
-  return 0;
-}
-
-int
-SChannelAdapter::BeginSSL() {
-  LOG(LS_VERBOSE) << "BeginSSL: " << ssl_host_name_;
-  ASSERT(state_ == SSL_CONNECTING);
-
-  SECURITY_STATUS ret;
-
-  SCHANNEL_CRED sc_cred = { 0 };
-  sc_cred.dwVersion = SCHANNEL_CRED_VERSION;
-  //sc_cred.dwMinimumCipherStrength = 128; // Note: use system default
-  sc_cred.dwFlags = SCH_CRED_NO_DEFAULT_CREDS | SCH_CRED_AUTO_CRED_VALIDATION;
-
-  ret = AcquireCredentialsHandle(NULL, const_cast<LPTSTR>(UNISP_NAME),
-                                 SECPKG_CRED_OUTBOUND, NULL, &sc_cred, NULL,
-                                 NULL, &impl_->cred, NULL);
-  if (ret != SEC_E_OK) {
-    LOG(LS_ERROR) << "AcquireCredentialsHandle error: "
-                  << ErrorName(ret, SECURITY_ERRORS);
-    return ret;
-  }
-  impl_->cred_init = true;
-
-  if (LOG_CHECK_LEVEL(LS_VERBOSE)) {
-    SecPkgCred_CipherStrengths cipher_strengths = { 0 };
-    ret = QueryCredentialsAttributes(&impl_->cred,
-                                     SECPKG_ATTR_CIPHER_STRENGTHS,
-                                     &cipher_strengths);
-    if (SUCCEEDED(ret)) {
-      LOG(LS_VERBOSE) << "SChannel cipher strength: "
-                  << cipher_strengths.dwMinimumCipherStrength << " - "
-                  << cipher_strengths.dwMaximumCipherStrength;
-    }
-
-    SecPkgCred_SupportedAlgs supported_algs = { 0 };
-    ret = QueryCredentialsAttributes(&impl_->cred,
-                                     SECPKG_ATTR_SUPPORTED_ALGS,
-                                     &supported_algs);
-    if (SUCCEEDED(ret)) {
-      LOG(LS_VERBOSE) << "SChannel supported algorithms:";
-      for (DWORD i=0; i<supported_algs.cSupportedAlgs; ++i) {
-        ALG_ID alg_id = supported_algs.palgSupportedAlgs[i];
-        PCCRYPT_OID_INFO oinfo = CryptFindOIDInfo(CRYPT_OID_INFO_ALGID_KEY,
-                                                  &alg_id, 0);
-        LPCWSTR alg_name = (NULL != oinfo) ? oinfo->pwszName : L"Unknown";
-        LOG(LS_VERBOSE) << "  " << ToUtf8(alg_name) << " (" << alg_id << ")";
-      }
-      CSecBufferBase::FreeSSPI(supported_algs.palgSupportedAlgs);
-    }
-  }
-
-  ULONG flags = SSL_FLAGS_DEFAULT, ret_flags = 0;
-  if (ignore_bad_cert())
-    flags |= ISC_REQ_MANUAL_CRED_VALIDATION;
-
-  CSecBufferBundle<2, CSecBufferBase::FreeSSPI> sb_out;
-  ret = InitializeSecurityContextA(&impl_->cred, NULL,
-                                   const_cast<char*>(ssl_host_name_.c_str()),
-                                   flags, 0, 0, NULL, 0,
-                                   &impl_->ctx, sb_out.desc(),
-                                   &ret_flags, NULL);
-  if (SUCCEEDED(ret))
-    impl_->ctx_init = true;
-  return ProcessContext(ret, NULL, sb_out.desc());
-}
-
-int
-SChannelAdapter::ContinueSSL() {
-  LOG(LS_VERBOSE) << "ContinueSSL";
-  ASSERT(state_ == SSL_CONNECTING);
-
-  SECURITY_STATUS ret;
-
-  CSecBufferBundle<2> sb_in;
-  sb_in[0].BufferType = SECBUFFER_TOKEN;
-  sb_in[0].cbBuffer = static_cast<unsigned long>(impl_->inbuf.size());
-  sb_in[0].pvBuffer = &impl_->inbuf[0];
-  //DescribeBuffers(LS_VERBOSE, "Input Buffer ", sb_in.desc());
-
-  ULONG flags = SSL_FLAGS_DEFAULT, ret_flags = 0;
-  if (ignore_bad_cert())
-    flags |= ISC_REQ_MANUAL_CRED_VALIDATION;
-
-  CSecBufferBundle<2, CSecBufferBase::FreeSSPI> sb_out;
-  ret = InitializeSecurityContextA(&impl_->cred, &impl_->ctx,
-                                   const_cast<char*>(ssl_host_name_.c_str()),
-                                   flags, 0, 0, sb_in.desc(), 0,
-                                   NULL, sb_out.desc(),
-                                   &ret_flags, NULL);
-  return ProcessContext(ret, sb_in.desc(), sb_out.desc());
-}
-
-int
-SChannelAdapter::ProcessContext(long int status, _SecBufferDesc* sbd_in,
-                                _SecBufferDesc* sbd_out) {
-  if (status != SEC_E_OK && status != SEC_I_CONTINUE_NEEDED &&
-      status != SEC_E_INCOMPLETE_MESSAGE) {
-    LOG(LS_ERROR)
-      << "InitializeSecurityContext error: "
-      << ErrorName(status, SECURITY_ERRORS);
-  }
-  //if (sbd_in)
-  //  DescribeBuffers(LS_VERBOSE, "Input Buffer ", sbd_in);
-  //if (sbd_out)
-  //  DescribeBuffers(LS_VERBOSE, "Output Buffer ", sbd_out);
-
-  if (status == SEC_E_INCOMPLETE_MESSAGE) {
-    // Wait for more input from server.
-    return Flush();
-  }
-
-  if (FAILED(status)) {
-    // We can't continue.  Common errors:
-    // SEC_E_CERT_EXPIRED - Typically, this means the computer clock is wrong.
-    return status;
-  }
-
-  // Note: we check both input and output buffers for SECBUFFER_EXTRA.
-  // Experience shows it appearing in the input, but the documentation claims
-  // it should appear in the output.
-  size_t extra = 0;
-  if (sbd_in) {
-    for (size_t i=0; i<sbd_in->cBuffers; ++i) {
-      SecBuffer& buffer = sbd_in->pBuffers[i];
-      if (buffer.BufferType == SECBUFFER_EXTRA) {
-        extra += buffer.cbBuffer;
-      }
-    }
-  }
-  if (sbd_out) {
-    for (size_t i=0; i<sbd_out->cBuffers; ++i) {
-      SecBuffer& buffer = sbd_out->pBuffers[i];
-      if (buffer.BufferType == SECBUFFER_EXTRA) {
-        extra += buffer.cbBuffer;
-      } else if (buffer.BufferType == SECBUFFER_TOKEN) {
-        impl_->outbuf.insert(impl_->outbuf.end(),
-          reinterpret_cast<char*>(buffer.pvBuffer),
-          reinterpret_cast<char*>(buffer.pvBuffer) + buffer.cbBuffer);
-      }
-    }
-  }
-
-  if (extra) {
-    ASSERT(extra <= impl_->inbuf.size());
-    size_t consumed = impl_->inbuf.size() - extra;
-    memmove(&impl_->inbuf[0], &impl_->inbuf[consumed], extra);
-    impl_->inbuf.resize(extra);
-  } else {
-    impl_->inbuf.clear();
-  }
-
-  if (SEC_I_CONTINUE_NEEDED == status) {
-    // Send data to server and wait for response.
-    // Note: ContinueSSL will result in a Flush, anyway.
-    return impl_->inbuf.empty() ? Flush() : ContinueSSL();
-  }
-
-  if (SEC_E_OK == status) {
-    LOG(LS_VERBOSE) << "QueryContextAttributes";
-    status = QueryContextAttributes(&impl_->ctx, SECPKG_ATTR_STREAM_SIZES,
-                                    &impl_->sizes);
-    if (FAILED(status)) {
-      LOG(LS_ERROR) << "QueryContextAttributes error: "
-                    << ErrorName(status, SECURITY_ERRORS);
-      return status;
-    }
-
-    state_ = SSL_CONNECTED;
-
-    if (int err = DecryptData()) {
-      return err;
-    } else if (int err = Flush()) {
-      return err;
-    } else {
-      // If we decrypted any data, queue up a notification here
-      PostEvent();
-      // Signal our connectedness
-      AsyncSocketAdapter::OnConnectEvent(this);
-    }
-    return 0;
-  }
-
-  if (SEC_I_INCOMPLETE_CREDENTIALS == status) {
-    // We don't support client authentication in schannel.
-    return status;
-  }
-
-  // We don't expect any other codes
-  ASSERT(false);
-  return status;
-}
-
-int
-SChannelAdapter::DecryptData() {
-  SChannelBuffer& inbuf = impl_->inbuf;
-  SChannelBuffer& readable = impl_->readable;
-
-  while (!inbuf.empty()) {
-    CSecBufferBundle<4> in_buf;
-    in_buf[0].BufferType = SECBUFFER_DATA;
-    in_buf[0].cbBuffer = static_cast<unsigned long>(inbuf.size());
-    in_buf[0].pvBuffer = &inbuf[0];
-
-    //DescribeBuffers(LS_VERBOSE, "Decrypt In ", in_buf.desc());
-    SECURITY_STATUS status = DecryptMessage(&impl_->ctx, in_buf.desc(), 0, 0);
-    //DescribeBuffers(LS_VERBOSE, "Decrypt Out ", in_buf.desc());
-
-    // Note: We are explicitly treating SEC_E_OK, SEC_I_CONTEXT_EXPIRED, and
-    // any other successful results as continue.
-    if (SUCCEEDED(status)) {
-      size_t data_len = 0, extra_len = 0;
-      for (size_t i=0; i<in_buf.desc()->cBuffers; ++i) {
-        if (in_buf[i].BufferType == SECBUFFER_DATA) {
-          data_len += in_buf[i].cbBuffer;
-          readable.insert(readable.end(),
-            reinterpret_cast<char*>(in_buf[i].pvBuffer),
-            reinterpret_cast<char*>(in_buf[i].pvBuffer) + in_buf[i].cbBuffer);
-        } else if (in_buf[i].BufferType == SECBUFFER_EXTRA) {
-          extra_len += in_buf[i].cbBuffer;
-        }
-      }
-      // There is a bug on Win2K where SEC_I_CONTEXT_EXPIRED is misclassified.
-      if ((data_len == 0) && (inbuf[0] == 0x15)) {
-        status = SEC_I_CONTEXT_EXPIRED;
-      }
-      if (extra_len) {
-        size_t consumed = inbuf.size() - extra_len;
-        memmove(&inbuf[0], &inbuf[consumed], extra_len);
-        inbuf.resize(extra_len);
-      } else {
-        inbuf.clear();
-      }
-      // TODO: Handle SEC_I_CONTEXT_EXPIRED to do clean shutdown
-      if (status != SEC_E_OK) {
-        LOG(LS_INFO) << "DecryptMessage returned continuation code: "
-                      << ErrorName(status, SECURITY_ERRORS);
-      }
-      continue;
-    }
-
-    if (status == SEC_E_INCOMPLETE_MESSAGE) {
-      break;
-    } else {
-      return status;
-    }
-  }
-
-  return 0;
-}
-
-void
-SChannelAdapter::Cleanup() {
-  if (impl_->ctx_init)
-    DeleteSecurityContext(&impl_->ctx);
-  if (impl_->cred_init)
-    FreeCredentialsHandle(&impl_->cred);
-  delete impl_;
-}
-
-void
-SChannelAdapter::PostEvent() {
-  // Check if there's anything notable to signal
-  if (impl_->readable.empty() && !signal_close_)
-    return;
-
-  // Only one post in the queue at a time
-  if (message_pending_)
-    return;
-
-  if (Thread* thread = Thread::Current()) {
-    message_pending_ = true;
-    thread->Post(this);
-  } else {
-    LOG(LS_ERROR) << "No thread context available for SChannelAdapter";
-    ASSERT(false);
-  }
-}
-
-void
-SChannelAdapter::Error(const char* context, int err, bool signal) {
-  LOG(LS_WARNING) << "SChannelAdapter::Error("
-                  << context << ", "
-                  << ErrorName(err, SECURITY_ERRORS) << ")";
-  state_ = SSL_ERROR;
-  SetError(err);
-  if (signal)
-    AsyncSocketAdapter::OnCloseEvent(this, err);
-}
-
-int
-SChannelAdapter::Read() {
-  char buffer[4096];
-  SChannelBuffer& inbuf = impl_->inbuf;
-  while (true) {
-    int ret = AsyncSocketAdapter::Recv(buffer, sizeof(buffer));
-    if (ret > 0) {
-      inbuf.insert(inbuf.end(), buffer, buffer + ret);
-    } else if (GetError() == EWOULDBLOCK) {
-      return 0;  // Blocking
-    } else {
-      return GetError();
-    }
-  }
-}
-
-int
-SChannelAdapter::Flush() {
-  int result = 0;
-  size_t pos = 0;
-  SChannelBuffer& outbuf = impl_->outbuf;
-  while (pos < outbuf.size()) {
-    int sent = AsyncSocketAdapter::Send(&outbuf[pos], outbuf.size() - pos);
-    if (sent > 0) {
-      pos += sent;
-    } else if (GetError() == EWOULDBLOCK) {
-      break;  // Blocking
-    } else {
-      result = GetError();
-      break;
-    }
-  }
-  if (int remainder = static_cast<int>(outbuf.size() - pos)) {
-    memmove(&outbuf[0], &outbuf[pos], remainder);
-    outbuf.resize(remainder);
-  } else {
-    outbuf.clear();
-  }
-  return result;
-}
-
-//
-// AsyncSocket Implementation
-//
-
-int
-SChannelAdapter::Send(const void* pv, size_t cb) {
-  switch (state_) {
-  case SSL_NONE:
-    return AsyncSocketAdapter::Send(pv, cb);
-
-  case SSL_WAIT:
-  case SSL_CONNECTING:
-    SetError(EWOULDBLOCK);
-    return SOCKET_ERROR;
-
-  case SSL_CONNECTED:
-    break;
-
-  case SSL_ERROR:
-  default:
-    return SOCKET_ERROR;
-  }
-
-  size_t written = 0;
-  SChannelBuffer& outbuf = impl_->outbuf;
-  while (written < cb) {
-    const size_t encrypt_len = std::min<size_t>(cb - written,
-                                                impl_->sizes.cbMaximumMessage);
-
-    CSecBufferBundle<4> out_buf;
-    out_buf[0].BufferType = SECBUFFER_STREAM_HEADER;
-    out_buf[0].cbBuffer = impl_->sizes.cbHeader;
-    out_buf[1].BufferType = SECBUFFER_DATA;
-    out_buf[1].cbBuffer = static_cast<unsigned long>(encrypt_len);
-    out_buf[2].BufferType = SECBUFFER_STREAM_TRAILER;
-    out_buf[2].cbBuffer = impl_->sizes.cbTrailer;
-
-    size_t packet_len = out_buf[0].cbBuffer
-                      + out_buf[1].cbBuffer
-                      + out_buf[2].cbBuffer;
-
-    SChannelBuffer message;
-    message.resize(packet_len);
-    out_buf[0].pvBuffer = &message[0];
-    out_buf[1].pvBuffer = &message[out_buf[0].cbBuffer];
-    out_buf[2].pvBuffer = &message[out_buf[0].cbBuffer + out_buf[1].cbBuffer];
-
-    memcpy(out_buf[1].pvBuffer,
-           static_cast<const char*>(pv) + written,
-           encrypt_len);
-
-    //DescribeBuffers(LS_VERBOSE, "Encrypt In ", out_buf.desc());
-    SECURITY_STATUS res = EncryptMessage(&impl_->ctx, 0, out_buf.desc(), 0);
-    //DescribeBuffers(LS_VERBOSE, "Encrypt Out ", out_buf.desc());
-
-    if (FAILED(res)) {
-      Error("EncryptMessage", res, false);
-      return SOCKET_ERROR;
-    }
-
-    // We assume that the header and data segments do not change length,
-    // or else encrypting the concatenated packet in-place is wrong.
-    ASSERT(out_buf[0].cbBuffer == impl_->sizes.cbHeader);
-    ASSERT(out_buf[1].cbBuffer == static_cast<unsigned long>(encrypt_len));
-
-    // However, the length of the trailer may change due to padding.
-    ASSERT(out_buf[2].cbBuffer <= impl_->sizes.cbTrailer);
-
-    packet_len = out_buf[0].cbBuffer
-               + out_buf[1].cbBuffer
-               + out_buf[2].cbBuffer;
-
-    written += encrypt_len;
-    outbuf.insert(outbuf.end(), &message[0], &message[packet_len-1]+1);
-  }
-
-  if (int err = Flush()) {
-    state_ = SSL_ERROR;
-    SetError(err);
-    return SOCKET_ERROR;
-  }
-
-  return static_cast<int>(written);
-}
-
-int
-SChannelAdapter::Recv(void* pv, size_t cb) {
-  switch (state_) {
-  case SSL_NONE:
-    return AsyncSocketAdapter::Recv(pv, cb);
-
-  case SSL_WAIT:
-  case SSL_CONNECTING:
-    SetError(EWOULDBLOCK);
-    return SOCKET_ERROR;
-
-  case SSL_CONNECTED:
-    break;
-
-  case SSL_ERROR:
-  default:
-    return SOCKET_ERROR;
-  }
-
-  SChannelBuffer& readable = impl_->readable;
-  if (readable.empty()) {
-    SetError(EWOULDBLOCK);
-    return SOCKET_ERROR;
-  }
-  size_t read = std::min(cb, readable.size());
-  memcpy(pv, &readable[0], read);
-  if (size_t remaining = readable.size() - read) {
-    memmove(&readable[0], &readable[read], remaining);
-    readable.resize(remaining);
-  } else {
-    readable.clear();
-  }
-
-  PostEvent();
-  return static_cast<int>(read);
-}
-
-int
-SChannelAdapter::Close() {
-  if (!impl_->readable.empty()) {
-    LOG(WARNING) << "SChannelAdapter::Close with readable data";
-    // Note: this isn't strictly an error, but we're using it temporarily to
-    // track bugs.
-    //ASSERT(false);
-  }
-  if (state_ == SSL_CONNECTED) {
-    DWORD token = SCHANNEL_SHUTDOWN;
-    CSecBufferBundle<1> sb_in;
-    sb_in[0].BufferType = SECBUFFER_TOKEN;
-    sb_in[0].cbBuffer = sizeof(token);
-    sb_in[0].pvBuffer = &token;
-    ApplyControlToken(&impl_->ctx, sb_in.desc());
-    // TODO: In theory, to do a nice shutdown, we need to begin shutdown
-    // negotiation with more calls to InitializeSecurityContext.  Since the
-    // socket api doesn't support nice shutdown at this point, we don't bother.
-  }
-  Cleanup();
-  impl_ = new SSLImpl;
-  state_ = restartable_ ? SSL_WAIT : SSL_NONE;
-  signal_close_ = false;
-  message_pending_ = false;
-  return AsyncSocketAdapter::Close();
-}
-
-Socket::ConnState
-SChannelAdapter::GetState() const {
-  if (signal_close_)
-    return CS_CONNECTED;
-  ConnState state = socket_->GetState();
-  if ((state == CS_CONNECTED)
-      && ((state_ == SSL_WAIT) || (state_ == SSL_CONNECTING)))
-    state = CS_CONNECTING;
-  return state;
-}
-
-void
-SChannelAdapter::OnConnectEvent(AsyncSocket* socket) {
-  LOG(LS_VERBOSE) << "SChannelAdapter::OnConnectEvent";
-  if (state_ != SSL_WAIT) {
-    ASSERT(state_ == SSL_NONE);
-    AsyncSocketAdapter::OnConnectEvent(socket);
-    return;
-  }
-
-  state_ = SSL_CONNECTING;
-  if (int err = BeginSSL()) {
-    Error("BeginSSL", err);
-  }
-}
-
-void
-SChannelAdapter::OnReadEvent(AsyncSocket* socket) {
-  if (state_ == SSL_NONE) {
-    AsyncSocketAdapter::OnReadEvent(socket);
-    return;
-  }
-
-  if (int err = Read()) {
-    Error("Read", err);
-    return;
-  }
-
-  if (impl_->inbuf.empty())
-    return;
-
-  if (state_ == SSL_CONNECTED) {
-    if (int err = DecryptData()) {
-      Error("DecryptData", err);
-    } else if (!impl_->readable.empty()) {
-      AsyncSocketAdapter::OnReadEvent(this);
-    }
-  } else if (state_ == SSL_CONNECTING) {
-    if (int err = ContinueSSL()) {
-      Error("ContinueSSL", err);
-    }
-  }
-}
-
-void
-SChannelAdapter::OnWriteEvent(AsyncSocket* socket) {
-  if (state_ == SSL_NONE) {
-    AsyncSocketAdapter::OnWriteEvent(socket);
-    return;
-  }
-
-  if (int err = Flush()) {
-    Error("Flush", err);
-    return;
-  }
-
-  // See if we have more data to write
-  if (!impl_->outbuf.empty())
-    return;
-
-  // Buffer is empty, submit notification
-  if (state_ == SSL_CONNECTED) {
-    AsyncSocketAdapter::OnWriteEvent(socket);
-  }
-}
-
-void
-SChannelAdapter::OnCloseEvent(AsyncSocket* socket, int err) {
-  if ((state_ == SSL_NONE) || impl_->readable.empty()) {
-    AsyncSocketAdapter::OnCloseEvent(socket, err);
-    return;
-  }
-
-  // If readable is non-empty, then we have a pending Message
-  // that will allow us to signal close (eventually).
-  signal_close_ = true;
-}
-
-void
-SChannelAdapter::OnMessage(Message* pmsg) {
-  if (!message_pending_)
-    return;  // This occurs when socket is closed
-
-  message_pending_ = false;
-  if (!impl_->readable.empty()) {
-    AsyncSocketAdapter::OnReadEvent(this);
-  } else if (signal_close_) {
-    signal_close_ = false;
-    AsyncSocketAdapter::OnCloseEvent(this, 0); // TODO: cache this error?
-  }
-}
-
-} // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/schanneladapter.h b/media/webrtc/trunk/webrtc/base/schanneladapter.h
deleted file mode 100644
index f6f73ad278..0000000000
--- a/media/webrtc/trunk/webrtc/base/schanneladapter.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_SCHANNELADAPTER_H__
-#define WEBRTC_BASE_SCHANNELADAPTER_H__
-
-#include <string>
-#include "webrtc/base/ssladapter.h"
-#include "webrtc/base/messagequeue.h"
-struct _SecBufferDesc;
-
-namespace rtc {
-
-///////////////////////////////////////////////////////////////////////////////
-
-class SChannelAdapter : public SSLAdapter, public MessageHandler {
-public:
-  SChannelAdapter(AsyncSocket* socket);
-  virtual ~SChannelAdapter();
-
-  virtual void SetMode(SSLMode mode);
-  virtual int StartSSL(const char* hostname, bool restartable);
-  virtual int Send(const void* pv, size_t cb);
-  virtual int Recv(void* pv, size_t cb);
-  virtual int Close();
-
-  // Note that the socket returns ST_CONNECTING while SSL is being negotiated.
-  virtual ConnState GetState() const;
-
-protected:
-  enum SSLState {
-    SSL_NONE, SSL_WAIT, SSL_CONNECTING, SSL_CONNECTED, SSL_ERROR
-  };
-  struct SSLImpl;
-
-  virtual void OnConnectEvent(AsyncSocket* socket);
-  virtual void OnReadEvent(AsyncSocket* socket);
-  virtual void OnWriteEvent(AsyncSocket* socket);
-  virtual void OnCloseEvent(AsyncSocket* socket, int err);
-  virtual void OnMessage(Message* pmsg);
-
-  int BeginSSL();
-  int ContinueSSL();
-  int ProcessContext(long int status, _SecBufferDesc* sbd_in,
-                     _SecBufferDesc* sbd_out);
-  int DecryptData();
-
-  int Read();
-  int Flush();
-  void Error(const char* context, int err, bool signal = true);
-  void Cleanup();
-
-  void PostEvent();
-
-private:
-  SSLState state_;
-  SSLMode mode_;
-  std::string ssl_host_name_;
-  // If true, socket will retain SSL configuration after Close.
-  bool restartable_;
-  // If true, we are delaying signalling close until all data is read.
-  bool signal_close_;
-  // If true, we are waiting to be woken up to signal readability or closure.
-  bool message_pending_;
-  SSLImpl* impl_;
-};
-
-/////////////////////////////////////////////////////////////////////////////
-
-} // namespace rtc
-
-#endif // WEBRTC_BASE_SCHANNELADAPTER_H__
diff --git a/media/webrtc/trunk/webrtc/base/scoped_autorelease_pool.h b/media/webrtc/trunk/webrtc/base/scoped_autorelease_pool.h
index d9cc3cb362..9aac112793 100644
--- a/media/webrtc/trunk/webrtc/base/scoped_autorelease_pool.h
+++ b/media/webrtc/trunk/webrtc/base/scoped_autorelease_pool.h
@@ -50,7 +50,7 @@ class ScopedAutoreleasePool {
 
   NSAutoreleasePool* pool_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(ScopedAutoreleasePool);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedAutoreleasePool);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/scoped_ptr.h b/media/webrtc/trunk/webrtc/base/scoped_ptr.h
index a3a6faf3cb..deeaf2eeb7 100644
--- a/media/webrtc/trunk/webrtc/base/scoped_ptr.h
+++ b/media/webrtc/trunk/webrtc/base/scoped_ptr.h
@@ -42,55 +42,39 @@
 //   }
 //
 // These scopers also implement part of the functionality of C++11 unique_ptr
-// in that they are "movable but not copyable."  You can use the scopers in
-// the parameter and return types of functions to signify ownership transfer
-// in to and out of a function.  When calling a function that has a scoper
-// as the argument type, it must be called with the result of an analogous
-// scoper's Pass() function or another function that generates a temporary;
-// passing by copy will NOT work.  Here is an example using scoped_ptr:
+// in that they are "movable but not copyable." You can use the scopers in the
+// parameter and return types of functions to signify ownership transfer in to
+// and out of a function. When calling a function that has a scoper as the
+// argument type, it must be called with the result of calling std::move on an
+// analogous scoper, or another function that generates a temporary; passing by
+// copy will NOT work. Here is an example using scoped_ptr:
 //
 //   void TakesOwnership(scoped_ptr<Foo> arg) {
 //     // Do something with arg
 //   }
 //   scoped_ptr<Foo> CreateFoo() {
-//     // No need for calling Pass() because we are constructing a temporary
+//     // No need for calling std::move because we are constructing a temporary
 //     // for the return value.
 //     return scoped_ptr<Foo>(new Foo("new"));
 //   }
 //   scoped_ptr<Foo> PassThru(scoped_ptr<Foo> arg) {
-//     return arg.Pass();
+//     return std::move(arg);
 //   }
 //
 //   {
 //     scoped_ptr<Foo> ptr(new Foo("yay"));  // ptr manages Foo("yay").
-//     TakesOwnership(ptr.Pass());           // ptr no longer owns Foo("yay").
+//     TakesOwnership(std::move(ptr));       // ptr no longer owns Foo("yay").
 //     scoped_ptr<Foo> ptr2 = CreateFoo();   // ptr2 owns the return Foo.
 //     scoped_ptr<Foo> ptr3 =                // ptr3 now owns what was in ptr2.
-//         PassThru(ptr2.Pass());            // ptr2 is correspondingly nullptr.
+//         PassThru(std::move(ptr2));        // ptr2 is correspondingly nullptr.
 //   }
 //
-// Notice that if you do not call Pass() when returning from PassThru(), or
+// Notice that if you do not call std::move when returning from PassThru(), or
 // when invoking TakesOwnership(), the code will not compile because scopers
 // are not copyable; they only implement move semantics which require calling
-// the Pass() function to signify a destructive transfer of state. CreateFoo()
-// is different though because we are constructing a temporary on the return
-// line and thus can avoid needing to call Pass().
-//
-// Pass() properly handles upcast in initialization, i.e. you can use a
-// scoped_ptr<Child> to initialize a scoped_ptr<Parent>:
-//
-//   scoped_ptr<Foo> foo(new Foo());
-//   scoped_ptr<FooParent> parent(foo.Pass());
-//
-// PassAs<>() should be used to upcast return value in return statement:
-//
-//   scoped_ptr<Foo> CreateFoo() {
-//     scoped_ptr<FooChild> result(new FooChild());
-//     return result.PassAs<Foo>();
-//   }
-//
-// Note that PassAs<>() is implemented only for scoped_ptr<T>, but not for
-// scoped_ptr<T[]>. This is because casting array pointers may not be safe.
+// std::move to signify a destructive transfer of state. CreateFoo() is
+// different though because we are constructing a temporary on the return line
+// and thus can avoid needing to call std::move.
 
 #ifndef WEBRTC_BASE_SCOPED_PTR_H__
 #define WEBRTC_BASE_SCOPED_PTR_H__
@@ -103,12 +87,13 @@
 #include <stdlib.h>
 
 #include <algorithm>  // For std::swap().
+#include <cstddef>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/move.h"
+#include "webrtc/base/deprecation.h"
 #include "webrtc/base/template_util.h"
 #include "webrtc/typedefs.h"
-
+ 
 // XXX This file creates unused typedefs as a way of doing static assertions,
 // both via COMPILE_ASSERT and via direct typedefs like
 // 'type_must_be_complete'. These trigger a GCC warning (enabled by -Wall in
@@ -312,7 +297,7 @@ class scoped_ptr_impl {
 
   Data data_;
 
-  DISALLOW_COPY_AND_ASSIGN(scoped_ptr_impl);
+  RTC_DISALLOW_COPY_AND_ASSIGN(scoped_ptr_impl);
 };
 
 }  // namespace internal
@@ -335,7 +320,6 @@ class scoped_ptr_impl {
 // types.
 template <class T, class D = rtc::DefaultDeleter<T> >
 class scoped_ptr {
-  RTC_MOVE_ONLY_TYPE_WITH_MOVE_CONSTRUCTOR_FOR_CPP_03(scoped_ptr)
 
   // TODO(ajm): If we ever import RefCountedBase, this check needs to be
   // enabled.
@@ -357,7 +341,7 @@ class scoped_ptr {
   scoped_ptr(element_type* p, const D& d) : impl_(p, d) {}
 
   // Constructor.  Allows construction from a nullptr.
-  scoped_ptr(decltype(nullptr)) : impl_(nullptr) {}
+  scoped_ptr(std::nullptr_t) : impl_(nullptr) {}
 
   // Constructor.  Allows construction from a scoped_ptr rvalue for a
   // convertible type and deleter.
@@ -394,11 +378,21 @@ class scoped_ptr {
 
   // operator=.  Allows assignment from a nullptr. Deletes the currently owned
   // object, if any.
-  scoped_ptr& operator=(decltype(nullptr)) {
+  scoped_ptr& operator=(std::nullptr_t) {
     reset();
     return *this;
   }
 
+  // Deleted copy constructor and copy assignment, to make the type move-only.
+  scoped_ptr(const scoped_ptr& other) = delete;
+  scoped_ptr& operator=(const scoped_ptr& other) = delete;
+
+  // Get an rvalue reference. (sp.Pass() does the same thing as std::move(sp).)
+  // Deprecated; remove in March 2016 (bug 5373).
+  RTC_DEPRECATED scoped_ptr&& Pass() {
+    return std::move(*this);
+  }
+
   // Reset.  Deletes the currently owned object, if any.
   // Then takes ownership of a new object, if given.
   void reset(element_type* p = nullptr) { impl_.reset(p); }
@@ -483,8 +477,6 @@ class scoped_ptr {
 
 template <class T, class D>
 class scoped_ptr<T[], D> {
-  RTC_MOVE_ONLY_TYPE_WITH_MOVE_CONSTRUCTOR_FOR_CPP_03(scoped_ptr)
-
  public:
   // The element and deleter types.
   typedef T element_type;
@@ -509,7 +501,7 @@ class scoped_ptr<T[], D> {
   explicit scoped_ptr(element_type* array) : impl_(array) {}
 
   // Constructor.  Allows construction from a nullptr.
-  scoped_ptr(decltype(nullptr)) : impl_(nullptr) {}
+  scoped_ptr(std::nullptr_t) : impl_(nullptr) {}
 
   // Constructor.  Allows construction from a scoped_ptr rvalue.
   scoped_ptr(scoped_ptr&& other) : impl_(&other.impl_) {}
@@ -522,11 +514,21 @@ class scoped_ptr<T[], D> {
 
   // operator=.  Allows assignment from a nullptr. Deletes the currently owned
   // array, if any.
-  scoped_ptr& operator=(decltype(nullptr)) {
+  scoped_ptr& operator=(std::nullptr_t) {
     reset();
     return *this;
   }
 
+  // Deleted copy constructor and copy assignment, to make the type move-only.
+  scoped_ptr(const scoped_ptr& other) = delete;
+  scoped_ptr& operator=(const scoped_ptr& other) = delete;
+
+  // Get an rvalue reference. (sp.Pass() does the same thing as std::move(sp).)
+  // Deprecated; remove in March 2016 (bug 5373).
+  RTC_DEPRECATED scoped_ptr&& Pass() {
+    return std::move(*this);
+  }
+
   // Reset.  Deletes the currently owned array, if any.
   // Then takes ownership of a new object, if given.
   void reset(element_type* array = nullptr) { impl_.reset(array); }
@@ -611,13 +613,13 @@ class scoped_ptr<T[], D> {
   template <class U> bool operator!=(scoped_ptr<U> const& p2) const;
 };
 
-}  // namespace rtc
-
 template <class T, class D>
 void swap(rtc::scoped_ptr<T, D>& p1, rtc::scoped_ptr<T, D>& p2) {
   p1.swap(p2);
 }
 
+}  // namespace rtc
+
 template <class T, class D>
 bool operator==(T* p1, const rtc::scoped_ptr<T, D>& p2) {
   return p1 == p2.get();
@@ -643,4 +645,11 @@ rtc::scoped_ptr<T> rtc_make_scoped_ptr(T* ptr) {
 #endif // not clang, and version >= 4.8
 #endif // GCC or clang
 
+// Pop off 'ignored "-Wunused-local-typedefs"':
+#if defined(__GNUC__)
+#if !defined(__clang__) && ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
+#pragma GCC diagnostic pop
+#endif // not clang, and version >= 4.8
+#endif // GCC or clang
+
 #endif  // #ifndef WEBRTC_BASE_SCOPED_PTR_H__
diff --git a/media/webrtc/trunk/webrtc/base/scopedptrcollection.h b/media/webrtc/trunk/webrtc/base/scopedptrcollection.h
index 47dff6503b..cfdb6f9673 100644
--- a/media/webrtc/trunk/webrtc/base/scopedptrcollection.h
+++ b/media/webrtc/trunk/webrtc/base/scopedptrcollection.h
@@ -52,7 +52,7 @@ class ScopedPtrCollection {
  private:
   VectorT collection_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScopedPtrCollection);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedPtrCollection);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/scopedptrcollection_unittest.cc b/media/webrtc/trunk/webrtc/base/scopedptrcollection_unittest.cc
index 30b8ed9ed0..933173e3fa 100644
--- a/media/webrtc/trunk/webrtc/base/scopedptrcollection_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/scopedptrcollection_unittest.cc
@@ -28,7 +28,7 @@ class InstanceCounter {
  private:
   int* num_instances_;
 
-  DISALLOW_COPY_AND_ASSIGN(InstanceCounter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(InstanceCounter);
 };
 
 }  // namespace
diff --git a/media/webrtc/trunk/webrtc/base/sec_buffer.h b/media/webrtc/trunk/webrtc/base/sec_buffer.h
index d4cda00d46..e6ffea4eb7 100644
--- a/media/webrtc/trunk/webrtc/base/sec_buffer.h
+++ b/media/webrtc/trunk/webrtc/base/sec_buffer.h
@@ -119,7 +119,7 @@ class CSecBufferBundle : public SecBufferBundleBase {
   }
 
   // Accessor for the descriptor
-  const PSecBufferDesc desc() const {
+  PSecBufferDesc desc() const {
     return &desc_;
   }
 
diff --git a/media/webrtc/trunk/webrtc/base/sha1.cc b/media/webrtc/trunk/webrtc/base/sha1.cc
index afc5569fd7..5816152b16 100644
--- a/media/webrtc/trunk/webrtc/base/sha1.cc
+++ b/media/webrtc/trunk/webrtc/base/sha1.cc
@@ -91,6 +91,16 @@
  *   84983E44 1C3BD26E BAAE4AA1 F95129E5 E54670F1
  * A million repetitions of "a"
  *   34AA973C D4C4DAA4 F61EEB2B DBAD2731 6534016F
+ *
+ * -----------------
+ * Modified 05/2015
+ * By Sergey Ulanov <sergeyu@chromium.org>
+ * Removed static buffer to make computation thread-safe.
+ *
+ * -----------------
+ * Modified 10/2015
+ * By Peter Boström <pbos@webrtc.org>
+ * Change uint32(8) back to uint32(8)_t (undoes (03/2012) change).
  */
 
 // Enabling SHA1HANDSOFF preserves the caller's data buffer.
@@ -104,14 +114,14 @@
 
 namespace rtc {
 
-void SHA1Transform(uint32 state[5], const uint8 buffer[64]);
+namespace {
 
 #define rol(value, bits) (((value) << (bits)) | ((value) >> (32 - (bits))))
 
 // blk0() and blk() perform the initial expand.
 // I got the idea of expanding during the round function from SSLeay
 // FIXME: can we do this in an endian-proof way?
-#ifdef ARCH_CPU_BIG_ENDIAN
+#ifdef RTC_ARCH_CPU_BIG_ENDIAN
 #define blk0(i) block->l[i]
 #else
 #define blk0(i) (block->l[i] = (rol(block->l[i], 24) & 0xFF00FF00) | \
@@ -151,13 +161,13 @@ void SHAPrintContext(SHA1_CTX *context, char *msg) {
 #endif /* VERBOSE */
 
 // Hash a single 512-bit block. This is the core of the algorithm.
-void SHA1Transform(uint32 state[5], const uint8 buffer[64]) {
+void SHA1Transform(uint32_t state[5], const uint8_t buffer[64]) {
   union CHAR64LONG16 {
-    uint8 c[64];
-    uint32 l[16];
+    uint8_t c[64];
+    uint32_t l[16];
   };
 #ifdef SHA1HANDSOFF
-  static uint8 workspace[64];
+  uint8_t workspace[64];
   memcpy(workspace, buffer, 64);
   CHAR64LONG16* block = reinterpret_cast<CHAR64LONG16*>(workspace);
 #else
@@ -167,11 +177,11 @@ void SHA1Transform(uint32 state[5], const uint8 buffer[64]) {
 #endif
 
   // Copy context->state[] to working vars.
-  uint32 a = state[0];
-  uint32 b = state[1];
-  uint32 c = state[2];
-  uint32 d = state[3];
-  uint32 e = state[4];
+  uint32_t a = state[0];
+  uint32_t b = state[1];
+  uint32_t c = state[2];
+  uint32_t d = state[3];
+  uint32_t e = state[4];
 
   // 4 rounds of 20 operations each. Loop unrolled.
   // Note(fbarchard): The following has lint warnings for multiple ; on
@@ -206,6 +216,8 @@ void SHA1Transform(uint32 state[5], const uint8 buffer[64]) {
   state[4] += e;
 }
 
+}  // namespace
+
 // SHA1Init - Initialize new context.
 void SHA1Init(SHA1_CTX* context) {
   // SHA1 initialization constants.
@@ -218,7 +230,7 @@ void SHA1Init(SHA1_CTX* context) {
 }
 
 // Run your data through this.
-void SHA1Update(SHA1_CTX* context, const uint8* data, size_t input_len) {
+void SHA1Update(SHA1_CTX* context, const uint8_t* data, size_t input_len) {
   size_t i = 0;
 
 #ifdef VERBOSE
@@ -229,15 +241,15 @@ void SHA1Update(SHA1_CTX* context, const uint8* data, size_t input_len) {
   size_t index = (context->count[0] >> 3) & 63;
 
   // Update number of bits.
-  // TODO: Use uint64 instead of 2 uint32 for count.
+  // TODO: Use uint64_t instead of 2 uint32_t for count.
   // count[0] has low 29 bits for byte count + 3 pad 0's making 32 bits for
   // bit count.
-  // Add bit count to low uint32
-  context->count[0] += static_cast<uint32>(input_len << 3);
-  if (context->count[0] < static_cast<uint32>(input_len << 3)) {
+  // Add bit count to low uint32_t
+  context->count[0] += static_cast<uint32_t>(input_len << 3);
+  if (context->count[0] < static_cast<uint32_t>(input_len << 3)) {
     ++context->count[1];  // if overlow (carry), add one to high word
   }
-  context->count[1] += static_cast<uint32>(input_len >> 29);
+  context->count[1] += static_cast<uint32_t>(input_len >> 29);
   if ((index + input_len) > 63) {
     i = 64 - index;
     memcpy(&context->buffer[index], data, i);
@@ -255,21 +267,21 @@ void SHA1Update(SHA1_CTX* context, const uint8* data, size_t input_len) {
 }
 
 // Add padding and return the message digest.
-void SHA1Final(SHA1_CTX* context, uint8 digest[SHA1_DIGEST_SIZE]) {
-  uint8 finalcount[8];
+void SHA1Final(SHA1_CTX* context, uint8_t digest[SHA1_DIGEST_SIZE]) {
+  uint8_t finalcount[8];
   for (int i = 0; i < 8; ++i) {
     // Endian independent
-    finalcount[i] = static_cast<uint8>(
-        (context->count[(i >= 4 ? 0 : 1)] >> ((3 - (i & 3)) * 8) ) & 255);
+    finalcount[i] = static_cast<uint8_t>(
+        (context->count[(i >= 4 ? 0 : 1)] >> ((3 - (i & 3)) * 8)) & 255);
   }
-  SHA1Update(context, reinterpret_cast<const uint8*>("\200"), 1);
+  SHA1Update(context, reinterpret_cast<const uint8_t*>("\200"), 1);
   while ((context->count[0] & 504) != 448) {
-    SHA1Update(context, reinterpret_cast<const uint8*>("\0"), 1);
+    SHA1Update(context, reinterpret_cast<const uint8_t*>("\0"), 1);
   }
   SHA1Update(context, finalcount, 8);  // Should cause a SHA1Transform().
   for (int i = 0; i < SHA1_DIGEST_SIZE; ++i) {
-    digest[i] = static_cast<uint8>(
-        (context->state[i >> 2] >> ((3 - (i & 3)) * 8) ) & 255);
+    digest[i] = static_cast<uint8_t>(
+        (context->state[i >> 2] >> ((3 - (i & 3)) * 8)) & 255);
   }
 
   // Wipe variables.
diff --git a/media/webrtc/trunk/webrtc/base/sha1.h b/media/webrtc/trunk/webrtc/base/sha1.h
index 4862a00498..aa5a6a5506 100644
--- a/media/webrtc/trunk/webrtc/base/sha1.h
+++ b/media/webrtc/trunk/webrtc/base/sha1.h
@@ -5,27 +5,28 @@
  *
 */
 
-// Ported to C++, Google style, under namespace rtc and uses basictypes.h
+// Ported to C++, Google style, under namespace rtc.
 
 #ifndef WEBRTC_BASE_SHA1_H_
 #define WEBRTC_BASE_SHA1_H_
 
-#include "webrtc/base/basictypes.h"
+#include <stdint.h>
+#include <stdlib.h>
 
 namespace rtc {
 
 struct SHA1_CTX {
-  uint32 state[5];
-  // TODO: Change bit count to uint64.
-  uint32 count[2];  // Bit count of input.
-  uint8 buffer[64];
+  uint32_t state[5];
+  // TODO: Change bit count to uint64_t.
+  uint32_t count[2];  // Bit count of input.
+  uint8_t buffer[64];
 };
 
 #define SHA1_DIGEST_SIZE 20
 
 void SHA1Init(SHA1_CTX* context);
-void SHA1Update(SHA1_CTX* context, const uint8* data, size_t len);
-void SHA1Final(SHA1_CTX* context, uint8 digest[SHA1_DIGEST_SIZE]);
+void SHA1Update(SHA1_CTX* context, const uint8_t* data, size_t len);
+void SHA1Final(SHA1_CTX* context, uint8_t digest[SHA1_DIGEST_SIZE]);
 
 #endif  // WEBRTC_BASE_SHA1_H_
 
diff --git a/media/webrtc/trunk/webrtc/base/sha1digest.cc b/media/webrtc/trunk/webrtc/base/sha1digest.cc
index 5ba0c54425..c090a06bb0 100644
--- a/media/webrtc/trunk/webrtc/base/sha1digest.cc
+++ b/media/webrtc/trunk/webrtc/base/sha1digest.cc
@@ -17,14 +17,14 @@ size_t Sha1Digest::Size() const {
 }
 
 void Sha1Digest::Update(const void* buf, size_t len) {
-  SHA1Update(&ctx_, static_cast<const uint8*>(buf), len);
+  SHA1Update(&ctx_, static_cast<const uint8_t*>(buf), len);
 }
 
 size_t Sha1Digest::Finish(void* buf, size_t len) {
   if (len < kSize) {
     return 0;
   }
-  SHA1Final(&ctx_, static_cast<uint8*>(buf));
+  SHA1Final(&ctx_, static_cast<uint8_t*>(buf));
   SHA1Init(&ctx_);  // Reset for next use.
   return kSize;
 }
diff --git a/media/webrtc/trunk/webrtc/base/sharedexclusivelock.h b/media/webrtc/trunk/webrtc/base/sharedexclusivelock.h
index aaaba3b83c..a6ca5735a2 100644
--- a/media/webrtc/trunk/webrtc/base/sharedexclusivelock.h
+++ b/media/webrtc/trunk/webrtc/base/sharedexclusivelock.h
@@ -36,7 +36,7 @@ class LOCKABLE SharedExclusiveLock {
   rtc::Event shared_count_is_zero_;
   int shared_count_;
 
-  DISALLOW_COPY_AND_ASSIGN(SharedExclusiveLock);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedExclusiveLock);
 };
 
 class SCOPED_LOCKABLE SharedScope {
@@ -51,7 +51,7 @@ class SCOPED_LOCKABLE SharedScope {
  private:
   SharedExclusiveLock* lock_;
 
-  DISALLOW_COPY_AND_ASSIGN(SharedScope);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedScope);
 };
 
 class SCOPED_LOCKABLE ExclusiveScope {
@@ -67,7 +67,7 @@ class SCOPED_LOCKABLE ExclusiveScope {
  private:
   SharedExclusiveLock* lock_;
 
-  DISALLOW_COPY_AND_ASSIGN(ExclusiveScope);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ExclusiveScope);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/sharedexclusivelock_unittest.cc b/media/webrtc/trunk/webrtc/base/sharedexclusivelock_unittest.cc
index c124db5c82..9b64ed760a 100644
--- a/media/webrtc/trunk/webrtc/base/sharedexclusivelock_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/sharedexclusivelock_unittest.cc
@@ -16,12 +16,11 @@
 #include "webrtc/base/sharedexclusivelock.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
-static const uint32 kMsgRead = 0;
-static const uint32 kMsgWrite = 0;
+static const uint32_t kMsgRead = 0;
+static const uint32_t kMsgWrite = 0;
 static const int kNoWaitThresholdInMs = 10;
 static const int kWaitThresholdInMs = 80;
 static const int kProcessTimeInMs = 100;
@@ -69,7 +68,7 @@ class ReadTask : public SharedExclusiveTask {
     TypedMessageData<int*>* message_data =
         static_cast<TypedMessageData<int*>*>(message->pdata);
 
-    uint32 start_time = Time();
+    uint32_t start_time = Time();
     {
       SharedScope ss(shared_exclusive_lock_);
       waiting_time_in_ms_ = TimeDiff(Time(), start_time);
@@ -102,7 +101,7 @@ class WriteTask : public SharedExclusiveTask {
     TypedMessageData<int>* message_data =
         static_cast<TypedMessageData<int>*>(message->pdata);
 
-    uint32 start_time = Time();
+    uint32_t start_time = Time();
     {
       ExclusiveScope es(shared_exclusive_lock_);
       waiting_time_in_ms_ = TimeDiff(Time(), start_time);
diff --git a/media/webrtc/trunk/webrtc/base/signalthread.cc b/media/webrtc/trunk/webrtc/base/signalthread.cc
index d03f386416..75f7b77315 100644
--- a/media/webrtc/trunk/webrtc/base/signalthread.cc
+++ b/media/webrtc/trunk/webrtc/base/signalthread.cc
@@ -39,13 +39,6 @@ bool SignalThread::SetName(const std::string& name, const void* obj) {
   return worker_.SetName(name, obj);
 }
 
-bool SignalThread::SetPriority(ThreadPriority priority) {
-  EnterExit ee(this);
-  ASSERT(main_->IsCurrent());
-  ASSERT(kInit == state_);
-  return worker_.SetPriority(priority);
-}
-
 void SignalThread::Start() {
   EnterExit ee(this);
   ASSERT(main_->IsCurrent());
diff --git a/media/webrtc/trunk/webrtc/base/signalthread.h b/media/webrtc/trunk/webrtc/base/signalthread.h
index 3a9205c4ca..ec250c6aad 100644
--- a/media/webrtc/trunk/webrtc/base/signalthread.h
+++ b/media/webrtc/trunk/webrtc/base/signalthread.h
@@ -45,9 +45,6 @@ class SignalThread
   // Context: Main Thread.  Call before Start to change the worker's name.
   bool SetName(const std::string& name, const void* obj);
 
-  // Context: Main Thread.  Call before Start to change the worker's priority.
-  bool SetPriority(ThreadPriority priority);
-
   // Context: Main Thread.  Call to begin the worker thread.
   void Start();
 
@@ -112,7 +109,7 @@ class SignalThread
    private:
     SignalThread* parent_;
 
-    DISALLOW_IMPLICIT_CONSTRUCTORS(Worker);
+    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Worker);
   };
 
   class SCOPED_LOCKABLE EnterExit {
@@ -135,7 +132,7 @@ class SignalThread
    private:
     SignalThread* t_;
 
-    DISALLOW_IMPLICIT_CONSTRUCTORS(EnterExit);
+    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EnterExit);
   };
 
   void Run();
@@ -147,7 +144,7 @@ class SignalThread
   State state_;
   int refcount_;
 
-  DISALLOW_COPY_AND_ASSIGN(SignalThread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SignalThread);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/base/signalthread_unittest.cc b/media/webrtc/trunk/webrtc/base/signalthread_unittest.cc
index 525ea62159..a583aefcb5 100644
--- a/media/webrtc/trunk/webrtc/base/signalthread_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/signalthread_unittest.cc
@@ -11,7 +11,6 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/signalthread.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using namespace rtc;
 
@@ -57,7 +56,7 @@ class SignalThreadTest : public testing::Test, public sigslot::has_slots<> {
 
    private:
     SignalThreadTest* harness_;
-    DISALLOW_EVIL_CONSTRUCTORS(SlowSignalThread);
+    RTC_DISALLOW_COPY_AND_ASSIGN(SlowSignalThread);
   };
 
   void OnWorkComplete(rtc::SignalThread* thread) {
@@ -128,7 +127,7 @@ class OwnerThread : public Thread, public sigslot::has_slots<> {
  private:
   SignalThreadTest* harness_;
   bool has_run_;
-  DISALLOW_EVIL_CONSTRUCTORS(OwnerThread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OwnerThread);
 };
 
 // Test for when the main thread goes away while the
diff --git a/media/webrtc/trunk/webrtc/base/sigslot.h b/media/webrtc/trunk/webrtc/base/sigslot.h
index 1c777f6371..0678f28692 100644
--- a/media/webrtc/trunk/webrtc/base/sigslot.h
+++ b/media/webrtc/trunk/webrtc/base/sigslot.h
@@ -534,7 +534,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -688,7 +688,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -827,7 +827,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -965,7 +965,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -1103,7 +1103,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -1243,7 +1243,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -1383,7 +1383,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -1523,7 +1523,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -1664,7 +1664,7 @@ namespace sigslot {
 			m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
 		}
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 			bool connected(has_slots_interface* pclass)
 		{
 			lock_block<mt_policy> lock(this);
@@ -2803,5 +2803,6 @@ namespace sigslot {
 
 }; // namespace sigslot
 
+#endif // TALK_BASE_SIGSLOT_H__
 #endif // TALK_BASE_SIGSLOT_H__
 #endif // WEBRTC_BASE_SIGSLOT_H__
diff --git a/media/webrtc/trunk/webrtc/base/sigslottester.h b/media/webrtc/trunk/webrtc/base/sigslottester.h
index ae781a97e3..cdbd44aa90 100644
--- a/media/webrtc/trunk/webrtc/base/sigslottester.h
+++ b/media/webrtc/trunk/webrtc/base/sigslottester.h
@@ -71,7 +71,7 @@ class SigslotTester1 : public sigslot::has_slots<> {
   int callback_count_;
   C1* capture1_;
 
-  DISALLOW_COPY_AND_ASSIGN(SigslotTester1);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester1);
 };
 
 template <class A1, class A2, class C1, class C2>
@@ -97,7 +97,7 @@ class SigslotTester2 : public sigslot::has_slots<> {
   C1* capture1_;
   C2* capture2_;
 
-  DISALLOW_COPY_AND_ASSIGN(SigslotTester2);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester2);
 };
 
 template <class A1, class A2, class A3, class C1, class C2, class C3>
@@ -125,7 +125,7 @@ class SigslotTester3 : public sigslot::has_slots<> {
   C2* capture2_;
   C3* capture3_;
 
-  DISALLOW_COPY_AND_ASSIGN(SigslotTester3);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester3);
 };
 
 template <class A1, class A2, class A3, class A4, class C1, class C2, class C3,
@@ -157,7 +157,7 @@ class SigslotTester4 : public sigslot::has_slots<> {
   C3* capture3_;
   C4* capture4_;
 
-  DISALLOW_COPY_AND_ASSIGN(SigslotTester4);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester4);
 };
 
 template <class A1, class A2, class A3, class A4, class A5, class C1, class C2,
@@ -192,7 +192,7 @@ class SigslotTester5 : public sigslot::has_slots<> {
   C4* capture4_;
   C5* capture5_;
 
-  DISALLOW_COPY_AND_ASSIGN(SigslotTester5);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester5);
 };
 }  // namespace rtc
 
diff --git a/media/webrtc/trunk/webrtc/base/sigslottester.h.pump b/media/webrtc/trunk/webrtc/base/sigslottester.h.pump
index 2fd9386a16..4410991d9e 100644
--- a/media/webrtc/trunk/webrtc/base/sigslottester.h.pump
+++ b/media/webrtc/trunk/webrtc/base/sigslottester.h.pump
@@ -76,7 +76,7 @@ class SigslotTester$i : public sigslot::has_slots<> {
   C$j* capture$j[[]]_;]]
 
 
-  DISALLOW_COPY_AND_ASSIGN(SigslotTester$i);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester$i);
 };
 
 ]]
diff --git a/media/webrtc/trunk/webrtc/base/socket.h b/media/webrtc/trunk/webrtc/base/socket.h
index 725bd45d10..22326cb997 100644
--- a/media/webrtc/trunk/webrtc/base/socket.h
+++ b/media/webrtc/trunk/webrtc/base/socket.h
@@ -26,6 +26,7 @@
 #endif
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socketaddress.h"
 
 // Rather than converting errors into a private namespace,
@@ -109,7 +110,7 @@
 #define EREMOTE WSAEREMOTE
 #undef EACCES
 #define SOCKET_EACCES WSAEACCES
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 #if defined(WEBRTC_POSIX)
 #define INVALID_SOCKET (-1)
@@ -123,6 +124,15 @@ inline bool IsBlockingError(int e) {
   return (e == EWOULDBLOCK) || (e == EAGAIN) || (e == EINPROGRESS);
 }
 
+struct SentPacket {
+  SentPacket() : packet_id(-1), send_time_ms(-1) {}
+  SentPacket(int packet_id, int64_t send_time_ms)
+      : packet_id(packet_id), send_time_ms(send_time_ms) {}
+
+  int packet_id;
+  int64_t send_time_ms;
+};
+
 // General interface for the socket implementations of various networks.  The
 // methods match those of normal UNIX sockets very closely.
 class Socket {
@@ -157,10 +167,10 @@ class Socket {
   };
   virtual ConnState GetState() const = 0;
 
-  // Fills in the given uint16 with the current estimate of the MTU along the
+  // Fills in the given uint16_t with the current estimate of the MTU along the
   // path to the address to which this socket is connected. NOTE: This method
   // can block for up to 10 seconds on Windows.
-  virtual int EstimateMTU(uint16* mtu) = 0;
+  virtual int EstimateMTU(uint16_t* mtu) = 0;
 
   enum Option {
     OPT_DONTFRAGMENT,
@@ -180,7 +190,7 @@ class Socket {
   Socket() {}
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(Socket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Socket);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/socket_unittest.cc b/media/webrtc/trunk/webrtc/base/socket_unittest.cc
index 6104eda4e4..8143823b86 100644
--- a/media/webrtc/trunk/webrtc/base/socket_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/socket_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/base/socket_unittest.h"
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/nethelpers.h"
@@ -827,7 +828,7 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
   // Fill the socket buffer.
   char buf[1024 * 16] = {0};
   int sends = 0;
-  while (++sends && accepted->Send(&buf, ARRAY_SIZE(buf)) != -1) {}
+  while (++sends && accepted->Send(&buf, arraysize(buf)) != -1) {}
   EXPECT_TRUE(accepted->IsBlocking());
 
   // Wait until data is available.
@@ -835,7 +836,7 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
 
   // Pull data.
   for (int i = 0; i < sends; ++i) {
-    client->Recv(buf, ARRAY_SIZE(buf));
+    client->Recv(buf, arraysize(buf));
   }
 
   // Expect at least one additional writable callback.
@@ -845,7 +846,7 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
   // callbacks.
   int extras = 0;
   for (int i = 0; i < 100; ++i) {
-    accepted->Send(&buf, ARRAY_SIZE(buf));
+    accepted->Send(&buf, arraysize(buf));
     rtc::Thread::Current()->ProcessMessages(1);
     if (sink.Check(accepted.get(), testing::SSE_WRITE)) {
       extras++;
@@ -929,7 +930,7 @@ void SocketTest::UdpReadyToSend(const IPAddress& loopback) {
   client->SetOption(rtc::Socket::OPT_SNDBUF, send_buffer_size);
 
   int error = 0;
-  uint32 start_ms = Time();
+  uint32_t start_ms = Time();
   int sent_packet_num = 0;
   int expected_error = EWOULDBLOCK;
   while (start_ms + kTimeout > Time()) {
@@ -990,7 +991,7 @@ void SocketTest::GetSetOptionsInternal(const IPAddress& loopback) {
         mtu_socket(
             ss_->CreateAsyncSocket(loopback.family(), SOCK_DGRAM));
     mtu_socket->Bind(SocketAddress(loopback, 0));
-    uint16 mtu;
+    uint16_t mtu;
     // should fail until we connect
     ASSERT_EQ(-1, mtu_socket->EstimateMTU(&mtu));
     mtu_socket->Connect(SocketAddress(loopback, 0));
diff --git a/media/webrtc/trunk/webrtc/base/socket_unittest.h b/media/webrtc/trunk/webrtc/base/socket_unittest.h
index d368afb3f5..e4a6b32705 100644
--- a/media/webrtc/trunk/webrtc/base/socket_unittest.h
+++ b/media/webrtc/trunk/webrtc/base/socket_unittest.h
@@ -21,8 +21,9 @@ namespace rtc {
 // socketserver, and call the SocketTest test methods.
 class SocketTest : public testing::Test {
  protected:
-  SocketTest() : ss_(NULL), kIPv4Loopback(INADDR_LOOPBACK),
-                 kIPv6Loopback(in6addr_loopback) {}
+  SocketTest() : kIPv4Loopback(INADDR_LOOPBACK),
+                 kIPv6Loopback(in6addr_loopback),
+                 ss_(nullptr) {}
   virtual void SetUp() { ss_ = Thread::Current()->socketserver(); }
   void TestConnectIPv4();
   void TestConnectIPv6();
@@ -57,6 +58,10 @@ class SocketTest : public testing::Test {
   void TestGetSetOptionsIPv4();
   void TestGetSetOptionsIPv6();
 
+  static const int kTimeout = 5000;  // ms
+  const IPAddress kIPv4Loopback;
+  const IPAddress kIPv6Loopback;
+
  private:
   void ConnectInternal(const IPAddress& loopback);
   void ConnectWithDnsLookupInternal(const IPAddress& loopback,
@@ -77,12 +82,13 @@ class SocketTest : public testing::Test {
   void UdpReadyToSend(const IPAddress& loopback);
   void GetSetOptionsInternal(const IPAddress& loopback);
 
-  static const int kTimeout = 5000;  // ms
   SocketServer* ss_;
-  const IPAddress kIPv4Loopback;
-  const IPAddress kIPv6Loopback;
 };
 
+// For unbound sockets, GetLocalAddress / GetRemoteAddress return AF_UNSPEC
+// values on Windows, but an empty address of the same family on Linux/MacOS X.
+bool IsUnspecOrEmptyIP(const IPAddress& address);
+
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_SOCKET_UNITTEST_H_
diff --git a/media/webrtc/trunk/webrtc/base/socketadapters.cc b/media/webrtc/trunk/webrtc/base/socketadapters.cc
index 4a2da0adfc..2b513dca63 100644
--- a/media/webrtc/trunk/webrtc/base/socketadapters.cc
+++ b/media/webrtc/trunk/webrtc/base/socketadapters.cc
@@ -81,10 +81,18 @@ int BufferedReadAdapter::Recv(void *pv, size_t cb) {
   // FIX: If cb == 0, we won't generate another read event
 
   int res = AsyncSocketAdapter::Recv(pv, cb);
-  if (res < 0)
-    return res;
+  if (res >= 0) {
+    // Read from socket and possibly buffer; return combined length
+    return res + static_cast<int>(read);
+  }
 
-  return res + static_cast<int>(read);
+  if (read > 0) {
+    // Failed to read from socket, but still read something from buffer
+    return static_cast<int>(read);
+  }
+
+  // Didn't read anything; return error from socket
+  return res;
 }
 
 void BufferedReadAdapter::BufferInput(bool on) {
@@ -129,41 +137,41 @@ AsyncProxyServerSocket::~AsyncProxyServerSocket() = default;
 // This is a SSL v2 CLIENT_HELLO message.
 // TODO: Should this have a session id? The response doesn't have a
 // certificate, so the hello should have a session id.
-static const uint8 kSslClientHello[] = {
-  0x80, 0x46,                                            // msg len
-  0x01,                                                  // CLIENT_HELLO
-  0x03, 0x01,                                            // SSL 3.1
-  0x00, 0x2d,                                            // ciphersuite len
-  0x00, 0x00,                                            // session id len
-  0x00, 0x10,                                            // challenge len
-  0x01, 0x00, 0x80, 0x03, 0x00, 0x80, 0x07, 0x00, 0xc0,  // ciphersuites
-  0x06, 0x00, 0x40, 0x02, 0x00, 0x80, 0x04, 0x00, 0x80,  //
-  0x00, 0x00, 0x04, 0x00, 0xfe, 0xff, 0x00, 0x00, 0x0a,  //
-  0x00, 0xfe, 0xfe, 0x00, 0x00, 0x09, 0x00, 0x00, 0x64,  //
-  0x00, 0x00, 0x62, 0x00, 0x00, 0x03, 0x00, 0x00, 0x06,  //
-  0x1f, 0x17, 0x0c, 0xa6, 0x2f, 0x00, 0x78, 0xfc,        // challenge
-  0x46, 0x55, 0x2e, 0xb1, 0x83, 0x39, 0xf1, 0xea         //
+static const uint8_t kSslClientHello[] = {
+    0x80, 0x46,                                            // msg len
+    0x01,                                                  // CLIENT_HELLO
+    0x03, 0x01,                                            // SSL 3.1
+    0x00, 0x2d,                                            // ciphersuite len
+    0x00, 0x00,                                            // session id len
+    0x00, 0x10,                                            // challenge len
+    0x01, 0x00, 0x80, 0x03, 0x00, 0x80, 0x07, 0x00, 0xc0,  // ciphersuites
+    0x06, 0x00, 0x40, 0x02, 0x00, 0x80, 0x04, 0x00, 0x80,  //
+    0x00, 0x00, 0x04, 0x00, 0xfe, 0xff, 0x00, 0x00, 0x0a,  //
+    0x00, 0xfe, 0xfe, 0x00, 0x00, 0x09, 0x00, 0x00, 0x64,  //
+    0x00, 0x00, 0x62, 0x00, 0x00, 0x03, 0x00, 0x00, 0x06,  //
+    0x1f, 0x17, 0x0c, 0xa6, 0x2f, 0x00, 0x78, 0xfc,        // challenge
+    0x46, 0x55, 0x2e, 0xb1, 0x83, 0x39, 0xf1, 0xea         //
 };
 
 // This is a TLSv1 SERVER_HELLO message.
-static const uint8 kSslServerHello[] = {
-  0x16,                                            // handshake message
-  0x03, 0x01,                                      // SSL 3.1
-  0x00, 0x4a,                                      // message len
-  0x02,                                            // SERVER_HELLO
-  0x00, 0x00, 0x46,                                // handshake len
-  0x03, 0x01,                                      // SSL 3.1
-  0x42, 0x85, 0x45, 0xa7, 0x27, 0xa9, 0x5d, 0xa0,  // server random
-  0xb3, 0xc5, 0xe7, 0x53, 0xda, 0x48, 0x2b, 0x3f,  //
-  0xc6, 0x5a, 0xca, 0x89, 0xc1, 0x58, 0x52, 0xa1,  //
-  0x78, 0x3c, 0x5b, 0x17, 0x46, 0x00, 0x85, 0x3f,  //
-  0x20,                                            // session id len
-  0x0e, 0xd3, 0x06, 0x72, 0x5b, 0x5b, 0x1b, 0x5f,  // session id
-  0x15, 0xac, 0x13, 0xf9, 0x88, 0x53, 0x9d, 0x9b,  //
-  0xe8, 0x3d, 0x7b, 0x0c, 0x30, 0x32, 0x6e, 0x38,  //
-  0x4d, 0xa2, 0x75, 0x57, 0x41, 0x6c, 0x34, 0x5c,  //
-  0x00, 0x04,                                      // RSA/RC4-128/MD5
-  0x00                                             // null compression
+static const uint8_t kSslServerHello[] = {
+    0x16,                                            // handshake message
+    0x03, 0x01,                                      // SSL 3.1
+    0x00, 0x4a,                                      // message len
+    0x02,                                            // SERVER_HELLO
+    0x00, 0x00, 0x46,                                // handshake len
+    0x03, 0x01,                                      // SSL 3.1
+    0x42, 0x85, 0x45, 0xa7, 0x27, 0xa9, 0x5d, 0xa0,  // server random
+    0xb3, 0xc5, 0xe7, 0x53, 0xda, 0x48, 0x2b, 0x3f,  //
+    0xc6, 0x5a, 0xca, 0x89, 0xc1, 0x58, 0x52, 0xa1,  //
+    0x78, 0x3c, 0x5b, 0x17, 0x46, 0x00, 0x85, 0x3f,  //
+    0x20,                                            // session id len
+    0x0e, 0xd3, 0x06, 0x72, 0x5b, 0x5b, 0x1b, 0x5f,  // session id
+    0x15, 0xac, 0x13, 0xf9, 0x88, 0x53, 0x9d, 0x9b,  //
+    0xe8, 0x3d, 0x7b, 0x0c, 0x30, 0x32, 0x6e, 0x38,  //
+    0x4d, 0xa2, 0x75, 0x57, 0x41, 0x6c, 0x34, 0x5c,  //
+    0x00, 0x04,                                      // RSA/RC4-128/MD5
+    0x00                                             // null compression
 };
 
 AsyncSSLSocket::AsyncSSLSocket(AsyncSocket* socket)
@@ -556,7 +564,7 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
   ByteBuffer response(data, *len);
 
   if (state_ == SS_HELLO) {
-    uint8 ver, method;
+    uint8_t ver, method;
     if (!response.ReadUInt8(&ver) ||
         !response.ReadUInt8(&method))
       return;
@@ -575,7 +583,7 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
       return;
     }
   } else if (state_ == SS_AUTH) {
-    uint8 ver, status;
+    uint8_t ver, status;
     if (!response.ReadUInt8(&ver) ||
         !response.ReadUInt8(&status))
       return;
@@ -587,7 +595,7 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
 
     SendConnect();
   } else if (state_ == SS_CONNECT) {
-    uint8 ver, rep, rsv, atyp;
+    uint8_t ver, rep, rsv, atyp;
     if (!response.ReadUInt8(&ver) ||
         !response.ReadUInt8(&rep) ||
         !response.ReadUInt8(&rsv) ||
@@ -599,15 +607,15 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
       return;
     }
 
-    uint16 port;
+    uint16_t port;
     if (atyp == 1) {
-      uint32 addr;
+      uint32_t addr;
       if (!response.ReadUInt32(&addr) ||
           !response.ReadUInt16(&port))
         return;
       LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port;
     } else if (atyp == 3) {
-      uint8 len;
+      uint8_t len;
       std::string addr;
       if (!response.ReadUInt8(&len) ||
           !response.ReadString(&addr, len) ||
@@ -662,9 +670,9 @@ void AsyncSocksProxySocket::SendHello() {
 void AsyncSocksProxySocket::SendAuth() {
   ByteBuffer request;
   request.WriteUInt8(1);           // Negotiation Version
-  request.WriteUInt8(static_cast<uint8>(user_.size()));
+  request.WriteUInt8(static_cast<uint8_t>(user_.size()));
   request.WriteString(user_);      // Username
-  request.WriteUInt8(static_cast<uint8>(pass_.GetLength()));
+  request.WriteUInt8(static_cast<uint8_t>(pass_.GetLength()));
   size_t len = pass_.GetLength() + 1;
   char * sensitive = new char[len];
   pass_.CopyTo(sensitive, true);
@@ -680,10 +688,10 @@ void AsyncSocksProxySocket::SendConnect() {
   request.WriteUInt8(5);              // Socks Version
   request.WriteUInt8(1);              // CONNECT
   request.WriteUInt8(0);              // Reserved
-  if (dest_.IsUnresolved()) {
+  if (dest_.IsUnresolvedIP()) {
     std::string hostname = dest_.hostname();
     request.WriteUInt8(3);            // DOMAINNAME
-    request.WriteUInt8(static_cast<uint8>(hostname.size()));
+    request.WriteUInt8(static_cast<uint8_t>(hostname.size()));
     request.WriteString(hostname);    // Destination Hostname
   } else {
     request.WriteUInt8(1);            // IPV4
@@ -730,7 +738,7 @@ void AsyncSocksProxyServerSocket::DirectSend(const ByteBuffer& buf) {
 }
 
 void AsyncSocksProxyServerSocket::HandleHello(ByteBuffer* request) {
-  uint8 ver, num_methods;
+  uint8_t ver, num_methods;
   if (!request->ReadUInt8(&ver) ||
       !request->ReadUInt8(&num_methods)) {
     Error(0);
@@ -743,7 +751,7 @@ void AsyncSocksProxyServerSocket::HandleHello(ByteBuffer* request) {
   }
 
   // Handle either no-auth (0) or user/pass auth (2)
-  uint8 method = 0xFF;
+  uint8_t method = 0xFF;
   if (num_methods > 0 && !request->ReadUInt8(&method)) {
     Error(0);
     return;
@@ -760,7 +768,7 @@ void AsyncSocksProxyServerSocket::HandleHello(ByteBuffer* request) {
   }
 }
 
-void AsyncSocksProxyServerSocket::SendHelloReply(uint8 method) {
+void AsyncSocksProxyServerSocket::SendHelloReply(uint8_t method) {
   ByteBuffer response;
   response.WriteUInt8(5);  // Socks Version
   response.WriteUInt8(method);  // Auth method
@@ -768,7 +776,7 @@ void AsyncSocksProxyServerSocket::SendHelloReply(uint8 method) {
 }
 
 void AsyncSocksProxyServerSocket::HandleAuth(ByteBuffer* request) {
-  uint8 ver, user_len, pass_len;
+  uint8_t ver, user_len, pass_len;
   std::string user, pass;
   if (!request->ReadUInt8(&ver) ||
       !request->ReadUInt8(&user_len) ||
@@ -784,7 +792,7 @@ void AsyncSocksProxyServerSocket::HandleAuth(ByteBuffer* request) {
   state_ = SS_CONNECT;
 }
 
-void AsyncSocksProxyServerSocket::SendAuthReply(uint8 result) {
+void AsyncSocksProxyServerSocket::SendAuthReply(uint8_t result) {
   ByteBuffer response;
   response.WriteUInt8(1);  // Negotiation Version
   response.WriteUInt8(result);
@@ -792,9 +800,9 @@ void AsyncSocksProxyServerSocket::SendAuthReply(uint8 result) {
 }
 
 void AsyncSocksProxyServerSocket::HandleConnect(ByteBuffer* request) {
-  uint8 ver, command, reserved, addr_type;
-  uint32 ip;
-  uint16 port;
+  uint8_t ver, command, reserved, addr_type;
+  uint32_t ip;
+  uint16_t port;
   if (!request->ReadUInt8(&ver) ||
       !request->ReadUInt8(&command) ||
       !request->ReadUInt8(&reserved) ||
diff --git a/media/webrtc/trunk/webrtc/base/socketadapters.h b/media/webrtc/trunk/webrtc/base/socketadapters.h
index a93d26ffa0..ece591d29f 100644
--- a/media/webrtc/trunk/webrtc/base/socketadapters.h
+++ b/media/webrtc/trunk/webrtc/base/socketadapters.h
@@ -50,7 +50,7 @@ class BufferedReadAdapter : public AsyncSocketAdapter {
   char * buffer_;
   size_t buffer_size_, data_len_;
   bool buffering_;
-  DISALLOW_EVIL_CONSTRUCTORS(BufferedReadAdapter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BufferedReadAdapter);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -78,7 +78,7 @@ class AsyncSSLSocket : public BufferedReadAdapter {
  protected:
   void OnConnectEvent(AsyncSocket* socket) override;
   void ProcessInput(char* data, size_t* len) override;
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncSSLSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncSSLSocket);
 };
 
 // Implements a socket adapter that performs the server side of a
@@ -89,7 +89,7 @@ class AsyncSSLServerSocket : public BufferedReadAdapter {
 
  protected:
   void ProcessInput(char* data, size_t* len) override;
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncSSLServerSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncSSLServerSocket);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -137,7 +137,7 @@ class AsyncHttpsProxySocket : public BufferedReadAdapter {
   } state_;
   HttpAuthContext * context_;
   std::string unknown_mechanisms_;
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncHttpsProxySocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncHttpsProxySocket);
 };
 
 /* TODO: Implement this.
@@ -148,7 +148,7 @@ class AsyncHttpsProxyServerSocket : public AsyncProxyServerSocket {
  private:
   virtual void ProcessInput(char * data, size_t& len);
   void Error(int error);
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncHttpsProxyServerSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncHttpsProxyServerSocket);
 };
 */
 
@@ -183,7 +183,7 @@ class AsyncSocksProxySocket : public BufferedReadAdapter {
   SocketAddress proxy_, dest_;
   std::string user_;
   CryptString pass_;
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncSocksProxySocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncSocksProxySocket);
 };
 
 // Implements a proxy server socket for the SOCKS protocol.
@@ -196,9 +196,9 @@ class AsyncSocksProxyServerSocket : public AsyncProxyServerSocket {
   void DirectSend(const ByteBuffer& buf);
 
   void HandleHello(ByteBuffer* request);
-  void SendHelloReply(uint8 method);
+  void SendHelloReply(uint8_t method);
   void HandleAuth(ByteBuffer* request);
-  void SendAuthReply(uint8 result);
+  void SendAuthReply(uint8_t result);
   void HandleConnect(ByteBuffer* request);
   void SendConnectResult(int result, const SocketAddress& addr) override;
 
@@ -209,7 +209,7 @@ class AsyncSocksProxyServerSocket : public AsyncProxyServerSocket {
     SS_HELLO, SS_AUTH, SS_CONNECT, SS_CONNECT_PENDING, SS_TUNNEL, SS_ERROR
   };
   State state_;
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncSocksProxyServerSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncSocksProxyServerSocket);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -235,7 +235,7 @@ class LoggingSocketAdapter : public AsyncSocketAdapter {
   std::string label_;
   bool hex_mode_;
   LogMultilineState lms_;
-  DISALLOW_EVIL_CONSTRUCTORS(LoggingSocketAdapter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LoggingSocketAdapter);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/base/socketaddress.cc b/media/webrtc/trunk/webrtc/base/socketaddress.cc
index b15c0c48b6..c5fd798cb1 100644
--- a/media/webrtc/trunk/webrtc/base/socketaddress.cc
+++ b/media/webrtc/trunk/webrtc/base/socketaddress.cc
@@ -47,7 +47,7 @@ SocketAddress::SocketAddress(const std::string& hostname, int port) {
   SetPort(port);
 }
 
-SocketAddress::SocketAddress(uint32 ip_as_host_order_integer, int port) {
+SocketAddress::SocketAddress(uint32_t ip_as_host_order_integer, int port) {
   SetIP(IPAddress(ip_as_host_order_integer));
   SetPort(port);
 }
@@ -86,7 +86,7 @@ SocketAddress& SocketAddress::operator=(const SocketAddress& addr) {
   return *this;
 }
 
-void SocketAddress::SetIP(uint32 ip_as_host_order_integer) {
+void SocketAddress::SetIP(uint32_t ip_as_host_order_integer) {
   hostname_.clear();
   literal_ = false;
   ip_ = IPAddress(ip_as_host_order_integer);
@@ -109,7 +109,7 @@ void SocketAddress::SetIP(const std::string& hostname) {
   scope_id_ = 0;
 }
 
-void SocketAddress::SetResolvedIP(uint32 ip_as_host_order_integer) {
+void SocketAddress::SetResolvedIP(uint32_t ip_as_host_order_integer) {
   ip_ = IPAddress(ip_as_host_order_integer);
   scope_id_ = 0;
 }
@@ -121,10 +121,10 @@ void SocketAddress::SetResolvedIP(const IPAddress& ip) {
 
 void SocketAddress::SetPort(int port) {
   ASSERT((0 <= port) && (port < 65536));
-  port_ = static_cast<uint16>(port);
+  port_ = static_cast<uint16_t>(port);
 }
 
-uint32 SocketAddress::ip() const {
+uint32_t SocketAddress::ip() const {
   return ip_.v4AddressAsHostOrderInteger();
 }
 
@@ -132,7 +132,7 @@ const IPAddress& SocketAddress::ipaddr() const {
   return ip_;
 }
 
-uint16 SocketAddress::port() const {
+uint16_t SocketAddress::port() const {
   return port_;
 }
 
@@ -279,7 +279,9 @@ bool SocketAddress::FromSockAddr(const sockaddr_in& saddr) {
 }
 
 static size_t ToSockAddrStorageHelper(sockaddr_storage* addr,
-                                      IPAddress ip, uint16 port, int scope_id) {
+                                      IPAddress ip,
+                                      uint16_t port,
+                                      int scope_id) {
   memset(addr, 0, sizeof(sockaddr_storage));
   addr->ss_family = static_cast<unsigned short>(ip.family());
   if (addr->ss_family == AF_INET6) {
@@ -305,47 +307,6 @@ size_t SocketAddress::ToSockAddrStorage(sockaddr_storage* addr) const {
   return ToSockAddrStorageHelper(addr, ip_, port_, scope_id_);
 }
 
-std::string SocketAddress::IPToString(uint32 ip_as_host_order_integer) {
-  return IPAddress(ip_as_host_order_integer).ToString();
-}
-
-std::string IPToSensitiveString(uint32 ip_as_host_order_integer) {
-  return IPAddress(ip_as_host_order_integer).ToSensitiveString();
-}
-
-bool SocketAddress::StringToIP(const std::string& hostname, uint32* ip) {
-  in_addr addr;
-  if (rtc::inet_pton(AF_INET, hostname.c_str(), &addr) == 0)
-    return false;
-  *ip = NetworkToHost32(addr.s_addr);
-  return true;
-}
-
-bool SocketAddress::StringToIP(const std::string& hostname, IPAddress* ip) {
-  in_addr addr4;
-  if (rtc::inet_pton(AF_INET, hostname.c_str(), &addr4) > 0) {
-    if (ip) {
-      *ip = IPAddress(addr4);
-    }
-    return true;
-  }
-
-  in6_addr addr6;
-  if (rtc::inet_pton(AF_INET6, hostname.c_str(), &addr6) > 0) {
-    if (ip) {
-      *ip = IPAddress(addr6);
-    }
-    return true;
-  }
-  return false;
-}
-
-uint32 SocketAddress::StringToIP(const std::string& hostname) {
-  uint32 ip = 0;
-  StringToIP(hostname, &ip);
-  return ip;
-}
-
 bool SocketAddressFromSockAddrStorage(const sockaddr_storage& addr,
                                       SocketAddress* out) {
   if (!out) {
diff --git a/media/webrtc/trunk/webrtc/base/socketaddress.h b/media/webrtc/trunk/webrtc/base/socketaddress.h
index f8256fc629..175d7a9d12 100644
--- a/media/webrtc/trunk/webrtc/base/socketaddress.h
+++ b/media/webrtc/trunk/webrtc/base/socketaddress.h
@@ -36,7 +36,7 @@ class SocketAddress {
 
   // Creates the address with the given IP and port.
   // IP is given as an integer in host byte order. V4 only, to be deprecated.
-  SocketAddress(uint32 ip_as_host_order_integer, int port);
+  SocketAddress(uint32_t ip_as_host_order_integer, int port);
 
   // Creates the address with the given IP and port.
   SocketAddress(const IPAddress& ip, int port);
@@ -58,7 +58,7 @@ class SocketAddress {
 
   // Changes the IP of this address to the given one, and clears the hostname
   // IP is given as an integer in host byte order. V4 only, to be deprecated..
-  void SetIP(uint32 ip_as_host_order_integer);
+  void SetIP(uint32_t ip_as_host_order_integer);
 
   // Changes the IP of this address to the given one, and clears the hostname.
   void SetIP(const IPAddress& ip);
@@ -70,7 +70,7 @@ class SocketAddress {
   // Sets the IP address while retaining the hostname.  Useful for bypassing
   // DNS for a pre-resolved IP.
   // IP is given as an integer in host byte order. V4 only, to be deprecated.
-  void SetResolvedIP(uint32 ip_as_host_order_integer);
+  void SetResolvedIP(uint32_t ip_as_host_order_integer);
 
   // Sets the IP address while retaining the hostname.  Useful for bypassing
   // DNS for a pre-resolved IP.
@@ -84,14 +84,14 @@ class SocketAddress {
 
   // Returns the IP address as a host byte order integer.
   // Returns 0 for non-v4 addresses.
-  uint32 ip() const;
+  uint32_t ip() const;
 
   const IPAddress& ipaddr() const;
 
   int family() const {return ip_.family(); }
 
   // Returns the port part of this address.
-  uint16 port() const;
+  uint16_t port() const;
 
   // Returns the scope ID associated with this address. Scope IDs are a
   // necessary addition to IPv6 link-local addresses, with different network
@@ -128,7 +128,6 @@ class SocketAddress {
   // That is, 0.0.0.0 or ::.
   // Hostname and/or port may be set.
   bool IsAnyIP() const;
-  inline bool IsAny() const { return IsAnyIP(); }  // deprecated
 
   // Determines whether the IP address refers to a loopback address.
   // For v4 addresses this means the address is in the range 127.0.0.0/8.
@@ -142,7 +141,6 @@ class SocketAddress {
 
   // Determines whether the hostname has been resolved to an IP.
   bool IsUnresolvedIP() const;
-  inline bool IsUnresolved() const { return IsUnresolvedIP(); }  // deprecated
 
   // Determines whether this address is identical to the given one.
   bool operator ==(const SocketAddress& addr) const;
@@ -178,29 +176,10 @@ class SocketAddress {
   size_t ToDualStackSockAddrStorage(sockaddr_storage* saddr) const;
   size_t ToSockAddrStorage(sockaddr_storage* saddr) const;
 
-  // Converts the IP address given in 'compact form' into dotted form.
-  // IP is given as an integer in host byte order. V4 only, to be deprecated.
-  // TODO: Deprecate this.
-  static std::string IPToString(uint32 ip_as_host_order_integer);
-
-  // Same as IPToString but anonymizes it by hiding the last part.
-  // TODO: Deprecate this.
-  static std::string IPToSensitiveString(uint32 ip_as_host_order_integer);
-
-  // Converts the IP address given in dotted form into compact form.
-  // Only dotted names (A.B.C.D) are  converted.
-  // Output integer is returned in host byte order.
-  // TODO: Deprecate, replace wth agnostic versions.
-  static bool StringToIP(const std::string& str, uint32* ip);
-  static uint32 StringToIP(const std::string& str);
-
-  // Converts the IP address given in printable form into an IPAddress.
-  static bool StringToIP(const std::string& str, IPAddress* ip);
-
  private:
   std::string hostname_;
   IPAddress ip_;
-  uint16 port_;
+  uint16_t port_;
   int scope_id_;
   bool literal_;  // Indicates that 'hostname_' contains a literal IP string.
 };
diff --git a/media/webrtc/trunk/webrtc/base/socketaddress_unittest.cc b/media/webrtc/trunk/webrtc/base/socketaddress_unittest.cc
index 6e9f089561..e235447dc7 100644
--- a/media/webrtc/trunk/webrtc/base/socketaddress_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/socketaddress_unittest.cc
@@ -27,10 +27,11 @@ const in6_addr kMappedV4Addr = { { {0x00, 0x00, 0x00, 0x00,
                                     0x00, 0x00, 0xFF, 0xFF,
                                     0x01, 0x02, 0x03, 0x04} } };
 const std::string kTestV6AddrString = "2001:db8:1020:3040:5060:7080:90a0:b0c0";
-const std::string kTestV6AddrAnonymizedString = "2001:db8:1020::";
+const std::string kTestV6AddrAnonymizedString = "2001:db8:1020:x:x:x:x:x";
 const std::string kTestV6AddrFullString =
     "[2001:db8:1020:3040:5060:7080:90a0:b0c0]:5678";
-const std::string kTestV6AddrFullAnonymizedString = "[2001:db8:1020::]:5678";
+const std::string kTestV6AddrFullAnonymizedString =
+    "[2001:db8:1020:x:x:x:x:x]:5678";
 
 TEST(SocketAddressTest, TestDefaultCtor) {
   SocketAddress addr;
@@ -325,23 +326,26 @@ TEST(SocketAddressTest, TestToSensitiveString) {
   SocketAddress addr_v4("1.2.3.4", 5678);
   EXPECT_EQ("1.2.3.4", addr_v4.HostAsURIString());
   EXPECT_EQ("1.2.3.4:5678", addr_v4.ToString());
-  EXPECT_EQ("1.2.3.4", addr_v4.HostAsSensitiveURIString());
-  EXPECT_EQ("1.2.3.4:5678", addr_v4.ToSensitiveString());
-  IPAddress::set_strip_sensitive(true);
+
+#if defined(NDEBUG)
   EXPECT_EQ("1.2.3.x", addr_v4.HostAsSensitiveURIString());
   EXPECT_EQ("1.2.3.x:5678", addr_v4.ToSensitiveString());
-  IPAddress::set_strip_sensitive(false);
+#else
+  EXPECT_EQ("1.2.3.4", addr_v4.HostAsSensitiveURIString());
+  EXPECT_EQ("1.2.3.4:5678", addr_v4.ToSensitiveString());
+#endif  // defined(NDEBUG)
 
   SocketAddress addr_v6(kTestV6AddrString, 5678);
   EXPECT_EQ("[" + kTestV6AddrString + "]", addr_v6.HostAsURIString());
   EXPECT_EQ(kTestV6AddrFullString, addr_v6.ToString());
-  EXPECT_EQ("[" + kTestV6AddrString + "]", addr_v6.HostAsSensitiveURIString());
-  EXPECT_EQ(kTestV6AddrFullString, addr_v6.ToSensitiveString());
-  IPAddress::set_strip_sensitive(true);
+#if defined(NDEBUG)
   EXPECT_EQ("[" + kTestV6AddrAnonymizedString + "]",
             addr_v6.HostAsSensitiveURIString());
   EXPECT_EQ(kTestV6AddrFullAnonymizedString, addr_v6.ToSensitiveString());
-  IPAddress::set_strip_sensitive(false);
+#else
+  EXPECT_EQ("[" + kTestV6AddrString + "]", addr_v6.HostAsSensitiveURIString());
+  EXPECT_EQ(kTestV6AddrFullString, addr_v6.ToSensitiveString());
+#endif  // defined(NDEBUG)
 }
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/socketstream.h b/media/webrtc/trunk/webrtc/base/socketstream.h
index 4e65cb2b87..fd8b559007 100644
--- a/media/webrtc/trunk/webrtc/base/socketstream.h
+++ b/media/webrtc/trunk/webrtc/base/socketstream.h
@@ -51,7 +51,7 @@ class SocketStream : public StreamInterface, public sigslot::has_slots<> {
 
   AsyncSocket* socket_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(SocketStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SocketStream);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/base/ssladapter.cc b/media/webrtc/trunk/webrtc/base/ssladapter.cc
index d83a2779e8..454a56637d 100644
--- a/media/webrtc/trunk/webrtc/base/ssladapter.cc
+++ b/media/webrtc/trunk/webrtc/base/ssladapter.cc
@@ -16,19 +16,11 @@
 
 #include "webrtc/base/sslconfig.h"
 
-#if SSL_USE_SCHANNEL
-
-#include "schanneladapter.h"
-
-#elif SSL_USE_OPENSSL  // && !SSL_USE_SCHANNEL
+#if SSL_USE_OPENSSL
 
 #include "openssladapter.h"
 
-#elif SSL_USE_NSS     // && !SSL_USE_CHANNEL && !SSL_USE_OPENSSL
-
-#include "nssstreamadapter.h"
-
-#endif  // SSL_USE_OPENSSL && !SSL_USE_SCHANNEL && !SSL_USE_NSS
+#endif
 
 ///////////////////////////////////////////////////////////////////////////////
 
@@ -36,14 +28,12 @@ namespace rtc {
 
 SSLAdapter*
 SSLAdapter::Create(AsyncSocket* socket) {
-#if SSL_USE_SCHANNEL
-  return new SChannelAdapter(socket);
-#elif SSL_USE_OPENSSL  // && !SSL_USE_SCHANNEL
+#if SSL_USE_OPENSSL
   return new OpenSSLAdapter(socket);
-#else  // !SSL_USE_OPENSSL && !SSL_USE_SCHANNEL
+#else  // !SSL_USE_OPENSSL
   delete socket;
   return NULL;
-#endif  // !SSL_USE_OPENSSL && !SSL_USE_SCHANNEL
+#endif  // SSL_USE_OPENSSL
 }
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -62,21 +52,7 @@ bool CleanupSSL() {
   return OpenSSLAdapter::CleanupSSL();
 }
 
-#elif SSL_USE_NSS  // !SSL_USE_OPENSSL
-
-bool InitializeSSL(VerificationCallback callback) {
-  return NSSContext::InitializeSSL(callback);
-}
-
-bool InitializeSSLThread() {
-  return NSSContext::InitializeSSLThread();
-}
-
-bool CleanupSSL() {
-  return NSSContext::CleanupSSL();
-}
-
-#else  // !SSL_USE_OPENSSL && !SSL_USE_NSS
+#else  // !SSL_USE_OPENSSL
 
 bool InitializeSSL(VerificationCallback callback) {
   return true;
@@ -90,7 +66,7 @@ bool CleanupSSL() {
   return true;
 }
 
-#endif  // !SSL_USE_OPENSSL && !SSL_USE_NSS
+#endif  // SSL_USE_OPENSSL
 
 ///////////////////////////////////////////////////////////////////////////////
 
diff --git a/media/webrtc/trunk/webrtc/base/ssladapter_unittest.cc b/media/webrtc/trunk/webrtc/base/ssladapter_unittest.cc
index 9a0548602e..7869b6eb63 100644
--- a/media/webrtc/trunk/webrtc/base/ssladapter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/ssladapter_unittest.cc
@@ -15,6 +15,7 @@
 #include "webrtc/base/socketstream.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
+#include "webrtc/base/sslidentity.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/virtualsocketserver.h"
 
@@ -129,10 +130,11 @@ class SSLAdapterTestDummyClient : public sigslot::has_slots<> {
 
 class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
  public:
-  explicit SSLAdapterTestDummyServer(const rtc::SSLMode& ssl_mode)
+  explicit SSLAdapterTestDummyServer(const rtc::SSLMode& ssl_mode,
+                                     const rtc::KeyParams& key_params)
       : ssl_mode_(ssl_mode) {
     // Generate a key pair and a certificate for this host.
-    ssl_identity_.reset(rtc::SSLIdentity::Generate(GetHostname()));
+    ssl_identity_.reset(rtc::SSLIdentity::Generate(GetHostname(), key_params));
 
     server_socket_.reset(CreateSocket(ssl_mode_));
 
@@ -268,13 +270,13 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
 class SSLAdapterTestBase : public testing::Test,
                            public sigslot::has_slots<> {
  public:
-  explicit SSLAdapterTestBase(const rtc::SSLMode& ssl_mode)
+  explicit SSLAdapterTestBase(const rtc::SSLMode& ssl_mode,
+                              const rtc::KeyParams& key_params)
       : ssl_mode_(ssl_mode),
         ss_scope_(new rtc::VirtualSocketServer(NULL)),
-        server_(new SSLAdapterTestDummyServer(ssl_mode_)),
+        server_(new SSLAdapterTestDummyServer(ssl_mode_, key_params)),
         client_(new SSLAdapterTestDummyClient(ssl_mode_)),
-        handshake_wait_(kTimeout) {
-  }
+        handshake_wait_(kTimeout) {}
 
   void SetHandshakeWait(int wait) {
     handshake_wait_ = wait;
@@ -343,43 +345,78 @@ class SSLAdapterTestBase : public testing::Test,
   int handshake_wait_;
 };
 
-class SSLAdapterTestTLS : public SSLAdapterTestBase {
+class SSLAdapterTestTLS_RSA : public SSLAdapterTestBase {
  public:
-  SSLAdapterTestTLS() : SSLAdapterTestBase(rtc::SSL_MODE_TLS) {}
+  SSLAdapterTestTLS_RSA()
+      : SSLAdapterTestBase(rtc::SSL_MODE_TLS, rtc::KeyParams::RSA()) {}
 };
 
-class SSLAdapterTestDTLS : public SSLAdapterTestBase {
+class SSLAdapterTestTLS_ECDSA : public SSLAdapterTestBase {
  public:
-  SSLAdapterTestDTLS() : SSLAdapterTestBase(rtc::SSL_MODE_DTLS) {}
+  SSLAdapterTestTLS_ECDSA()
+      : SSLAdapterTestBase(rtc::SSL_MODE_TLS, rtc::KeyParams::ECDSA()) {}
+};
+
+class SSLAdapterTestDTLS_RSA : public SSLAdapterTestBase {
+ public:
+  SSLAdapterTestDTLS_RSA()
+      : SSLAdapterTestBase(rtc::SSL_MODE_DTLS, rtc::KeyParams::RSA()) {}
+};
+
+class SSLAdapterTestDTLS_ECDSA : public SSLAdapterTestBase {
+ public:
+  SSLAdapterTestDTLS_ECDSA()
+      : SSLAdapterTestBase(rtc::SSL_MODE_DTLS, rtc::KeyParams::ECDSA()) {}
 };
 
 #if SSL_USE_OPENSSL
 
 // Basic tests: TLS
 
-// Test that handshake works
-TEST_F(SSLAdapterTestTLS, TestTLSConnect) {
+// Test that handshake works, using RSA
+TEST_F(SSLAdapterTestTLS_RSA, TestTLSConnect) {
   TestHandshake(true);
 }
 
-// Test transfer between client and server
-TEST_F(SSLAdapterTestTLS, TestTLSTransfer) {
+// Test that handshake works, using ECDSA
+TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSConnect) {
+  TestHandshake(true);
+}
+
+// Test transfer between client and server, using RSA
+TEST_F(SSLAdapterTestTLS_RSA, TestTLSTransfer) {
+  TestHandshake(true);
+  TestTransfer("Hello, world!");
+}
+
+// Test transfer between client and server, using ECDSA
+TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSTransfer) {
   TestHandshake(true);
   TestTransfer("Hello, world!");
 }
 
 // Basic tests: DTLS
 
-// Test that handshake works
-TEST_F(SSLAdapterTestDTLS, TestDTLSConnect) {
+// Test that handshake works, using RSA
+TEST_F(SSLAdapterTestDTLS_RSA, TestDTLSConnect) {
   TestHandshake(true);
 }
 
-// Test transfer between client and server
-TEST_F(SSLAdapterTestDTLS, TestDTLSTransfer) {
+// Test that handshake works, using ECDSA
+TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSConnect) {
+  TestHandshake(true);
+}
+
+// Test transfer between client and server, using RSA
+TEST_F(SSLAdapterTestDTLS_RSA, TestDTLSTransfer) {
+  TestHandshake(true);
+  TestTransfer("Hello, world!");
+}
+
+// Test transfer between client and server, using ECDSA
+TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSTransfer) {
   TestHandshake(true);
   TestTransfer("Hello, world!");
 }
 
 #endif  // SSL_USE_OPENSSL
-
diff --git a/media/webrtc/trunk/webrtc/base/sslconfig.h b/media/webrtc/trunk/webrtc/base/sslconfig.h
index d824ab0627..6aabad07a8 100644
--- a/media/webrtc/trunk/webrtc/base/sslconfig.h
+++ b/media/webrtc/trunk/webrtc/base/sslconfig.h
@@ -13,8 +13,7 @@
 
 // If no preference has been indicated, default to SChannel on Windows and
 // OpenSSL everywhere else, if it is available.
-#if !defined(SSL_USE_SCHANNEL) && !defined(SSL_USE_OPENSSL) && \
-    !defined(SSL_USE_NSS)
+#if !defined(SSL_USE_SCHANNEL) && !defined(SSL_USE_OPENSSL)
 #if defined(WEBRTC_WIN)
 
 #define SSL_USE_SCHANNEL 1
@@ -23,8 +22,6 @@
 
 #if defined(HAVE_OPENSSL_SSL_H)
 #define SSL_USE_OPENSSL 1
-#elif defined(HAVE_NSS_SSL_H)
-#define SSL_USE_NSS 1
 #endif
 
 #endif  // !defined(WEBRTC_WIN)
diff --git a/media/webrtc/trunk/webrtc/base/sslfingerprint.cc b/media/webrtc/trunk/webrtc/base/sslfingerprint.cc
index d45e7a068b..1939b4fd0b 100644
--- a/media/webrtc/trunk/webrtc/base/sslfingerprint.cc
+++ b/media/webrtc/trunk/webrtc/base/sslfingerprint.cc
@@ -30,7 +30,7 @@ SSLFingerprint* SSLFingerprint::Create(
 
 SSLFingerprint* SSLFingerprint::Create(
     const std::string& algorithm, const rtc::SSLCertificate* cert) {
-  uint8 digest_val[64];
+  uint8_t digest_val[64];
   size_t digest_len;
   bool ret = cert->ComputeDigest(
       algorithm, digest_val, sizeof(digest_val), &digest_len);
@@ -58,13 +58,13 @@ SSLFingerprint* SSLFingerprint::CreateFromRfc4572(
   if (!value_len)
     return NULL;
 
-  return new SSLFingerprint(algorithm,
-                            reinterpret_cast<uint8*>(value),
+  return new SSLFingerprint(algorithm, reinterpret_cast<uint8_t*>(value),
                             value_len);
 }
 
-SSLFingerprint::SSLFingerprint(
-    const std::string& algorithm, const uint8* digest_in, size_t digest_len)
+SSLFingerprint::SSLFingerprint(const std::string& algorithm,
+                               const uint8_t* digest_in,
+                               size_t digest_len)
     : algorithm(algorithm) {
   digest.SetData(digest_in, digest_len);
 }
@@ -79,7 +79,7 @@ bool SSLFingerprint::operator==(const SSLFingerprint& other) const {
 
 std::string SSLFingerprint::GetRfc4572Fingerprint() const {
   std::string fingerprint =
-      rtc::hex_encode_with_delimiter(digest.data(), digest.size(), ':');
+      rtc::hex_encode_with_delimiter(digest.data<char>(), digest.size(), ':');
   std::transform(fingerprint.begin(), fingerprint.end(),
                  fingerprint.begin(), ::toupper);
   return fingerprint;
diff --git a/media/webrtc/trunk/webrtc/base/sslfingerprint.h b/media/webrtc/trunk/webrtc/base/sslfingerprint.h
index a63b3dd875..735238dde6 100644
--- a/media/webrtc/trunk/webrtc/base/sslfingerprint.h
+++ b/media/webrtc/trunk/webrtc/base/sslfingerprint.h
@@ -13,6 +13,7 @@
 
 #include <string>
 
+#include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/sslidentity.h"
 
@@ -30,7 +31,8 @@ struct SSLFingerprint {
   static SSLFingerprint* CreateFromRfc4572(const std::string& algorithm,
                                            const std::string& fingerprint);
 
-  SSLFingerprint(const std::string& algorithm, const uint8* digest_in,
+  SSLFingerprint(const std::string& algorithm,
+                 const uint8_t* digest_in,
                  size_t digest_len);
 
   SSLFingerprint(const SSLFingerprint& from);
diff --git a/media/webrtc/trunk/webrtc/base/sslidentity.cc b/media/webrtc/trunk/webrtc/base/sslidentity.cc
index ea9f547a87..5f6b6869dd 100644
--- a/media/webrtc/trunk/webrtc/base/sslidentity.cc
+++ b/media/webrtc/trunk/webrtc/base/sslidentity.cc
@@ -15,28 +15,78 @@
 
 #include "webrtc/base/sslidentity.h"
 
+#include <ctime>
 #include <string>
 
 #include "webrtc/base/base64.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/sslconfig.h"
 
-#if SSL_USE_SCHANNEL
-
-#elif SSL_USE_OPENSSL  // !SSL_USE_SCHANNEL
+#if SSL_USE_OPENSSL
 
 #include "webrtc/base/opensslidentity.h"
 
-#elif SSL_USE_NSS  // !SSL_USE_SCHANNEL && !SSL_USE_OPENSSL
-
-#include "webrtc/base/nssidentity.h"
-
-#endif  // SSL_USE_SCHANNEL
+#endif  // SSL_USE_OPENSSL
 
 namespace rtc {
 
 const char kPemTypeCertificate[] = "CERTIFICATE";
 const char kPemTypeRsaPrivateKey[] = "RSA PRIVATE KEY";
+const char kPemTypeEcPrivateKey[] = "EC PRIVATE KEY";
+
+KeyParams::KeyParams(KeyType key_type) {
+  if (key_type == KT_ECDSA) {
+    type_ = KT_ECDSA;
+    params_.curve = EC_NIST_P256;
+  } else if (key_type == KT_RSA) {
+    type_ = KT_RSA;
+    params_.rsa.mod_size = kRsaDefaultModSize;
+    params_.rsa.pub_exp = kRsaDefaultExponent;
+  } else {
+    RTC_NOTREACHED();
+  }
+}
+
+// static
+KeyParams KeyParams::RSA(int mod_size, int pub_exp) {
+  KeyParams kt(KT_RSA);
+  kt.params_.rsa.mod_size = mod_size;
+  kt.params_.rsa.pub_exp = pub_exp;
+  return kt;
+}
+
+// static
+KeyParams KeyParams::ECDSA(ECCurve curve) {
+  KeyParams kt(KT_ECDSA);
+  kt.params_.curve = curve;
+  return kt;
+}
+
+bool KeyParams::IsValid() const {
+  if (type_ == KT_RSA) {
+    return (params_.rsa.mod_size >= kRsaMinModSize &&
+            params_.rsa.mod_size <= kRsaMaxModSize &&
+            params_.rsa.pub_exp > params_.rsa.mod_size);
+  } else if (type_ == KT_ECDSA) {
+    return (params_.curve == EC_NIST_P256);
+  }
+  return false;
+}
+
+RSAParams KeyParams::rsa_params() const {
+  RTC_DCHECK(type_ == KT_RSA);
+  return params_.rsa;
+}
+
+ECCurve KeyParams::ec_curve() const {
+  RTC_DCHECK(type_ == KT_ECDSA);
+  return params_.curve;
+}
+
+KeyType IntKeyTypeFamilyToKeyType(int key_type_family) {
+  return static_cast<KeyType>(key_type_family);
+}
 
 bool SSLIdentity::PemToDer(const std::string& pem_type,
                            const std::string& pem_string,
@@ -102,33 +152,15 @@ SSLCertChain::~SSLCertChain() {
   std::for_each(certs_.begin(), certs_.end(), DeleteCert);
 }
 
-#if SSL_USE_SCHANNEL
-
-SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
-  return NULL;
-}
-
-SSLIdentity* SSLIdentity::Generate(const std::string& common_name) {
-  return NULL;
-}
-
-SSLIdentity* GenerateForTest(const SSLIdentityParams& params) {
-  return NULL;
-}
-
-SSLIdentity* SSLIdentity::FromPEMStrings(const std::string& private_key,
-                                         const std::string& certificate) {
-  return NULL;
-}
-
-#elif SSL_USE_OPENSSL  // !SSL_USE_SCHANNEL
+#if SSL_USE_OPENSSL
 
 SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
   return OpenSSLCertificate::FromPEMString(pem_string);
 }
 
-SSLIdentity* SSLIdentity::Generate(const std::string& common_name) {
-  return OpenSSLIdentity::Generate(common_name);
+SSLIdentity* SSLIdentity::Generate(const std::string& common_name,
+                                   const KeyParams& key_params) {
+  return OpenSSLIdentity::Generate(common_name, key_params);
 }
 
 SSLIdentity* SSLIdentity::GenerateForTest(const SSLIdentityParams& params) {
@@ -140,29 +172,80 @@ SSLIdentity* SSLIdentity::FromPEMStrings(const std::string& private_key,
   return OpenSSLIdentity::FromPEMStrings(private_key, certificate);
 }
 
-#elif SSL_USE_NSS  // !SSL_USE_OPENSSL && !SSL_USE_SCHANNEL
-
-SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
-  return NSSCertificate::FromPEMString(pem_string);
-}
-
-SSLIdentity* SSLIdentity::Generate(const std::string& common_name) {
-  return NSSIdentity::Generate(common_name);
-}
-
-SSLIdentity* SSLIdentity::GenerateForTest(const SSLIdentityParams& params) {
-  return NSSIdentity::GenerateForTest(params);
-}
-
-SSLIdentity* SSLIdentity::FromPEMStrings(const std::string& private_key,
-                                         const std::string& certificate) {
-  return NSSIdentity::FromPEMStrings(private_key, certificate);
-}
-
-#else  // !SSL_USE_OPENSSL && !SSL_USE_SCHANNEL && !SSL_USE_NSS
+#else  // !SSL_USE_OPENSSL
 
 #error "No SSL implementation"
 
-#endif  // SSL_USE_SCHANNEL
+#endif  // SSL_USE_OPENSSL
+
+// Read |n| bytes from ASN1 number string at *|pp| and return the numeric value.
+// Update *|pp| and *|np| to reflect number of read bytes.
+static inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) {
+  const unsigned char* p = *pp;
+  int x = 0;
+  for (size_t i = 0; i < n; i++)
+    x = 10 * x + p[i] - '0';
+  *pp = p + n;
+  *np = *np - n;
+  return x;
+}
+
+int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format) {
+  size_t bytes_left = length;
+
+  // Make sure the string ends with Z.  Doing it here protects the strspn call
+  // from running off the end of the string in Z's absense.
+  if (length == 0 || s[length - 1] != 'Z')
+    return -1;
+
+  // Make sure we only have ASCII digits so that we don't need to clutter the
+  // code below and ASN1ReadInt with error checking.
+  size_t n = strspn(reinterpret_cast<const char*>(s), "0123456789");
+  if (n + 1 != length)
+    return -1;
+
+  int year;
+
+  // Read out ASN1 year, in either 2-char "UTCTIME" or 4-char "GENERALIZEDTIME"
+  // format.  Both format use UTC in this context.
+  if (long_format) {
+    // ASN1 format: yyyymmddhh[mm[ss[.fff]]]Z where the Z is literal, but
+    // RFC 5280 requires us to only support exactly yyyymmddhhmmssZ.
+
+    if (bytes_left < 11)
+      return -1;
+
+    year = ASN1ReadInt(&s, &bytes_left, 4);
+    year -= 1900;
+  } else {
+    // ASN1 format: yymmddhhmm[ss]Z where the Z is literal, but RFC 5280
+    // requires us to only support exactly yymmddhhmmssZ.
+
+    if (bytes_left < 9)
+      return -1;
+
+    year = ASN1ReadInt(&s, &bytes_left, 2);
+    if (year < 50)  // Per RFC 5280 4.1.2.5.1
+      year += 100;
+  }
+
+  std::tm tm;
+  tm.tm_year = year;
+
+  // Read out remaining ASN1 time data and store it in |tm| in documented
+  // std::tm format.
+  tm.tm_mon = ASN1ReadInt(&s, &bytes_left, 2) - 1;
+  tm.tm_mday = ASN1ReadInt(&s, &bytes_left, 2);
+  tm.tm_hour = ASN1ReadInt(&s, &bytes_left, 2);
+  tm.tm_min = ASN1ReadInt(&s, &bytes_left, 2);
+  tm.tm_sec = ASN1ReadInt(&s, &bytes_left, 2);
+
+  if (bytes_left != 1) {
+    // Now just Z should remain.  Its existence was asserted above.
+    return -1;
+  }
+
+  return TmToSeconds(tm);
+}
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/sslidentity.h b/media/webrtc/trunk/webrtc/base/sslidentity.h
index 320fe53d0f..a143ee4108 100644
--- a/media/webrtc/trunk/webrtc/base/sslidentity.h
+++ b/media/webrtc/trunk/webrtc/base/sslidentity.h
@@ -19,6 +19,7 @@
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/messagedigest.h"
+#include "webrtc/base/timeutils.h"
 
 namespace rtc {
 
@@ -68,6 +69,10 @@ class SSLCertificate {
                              unsigned char* digest,
                              size_t size,
                              size_t* length) const = 0;
+
+  // Returns the time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC),
+  // or -1 if an expiration time could not be retrieved.
+  virtual int64_t CertificateExpirationTime() const = 0;
 };
 
 // SSLCertChain is a simple wrapper for a vector of SSLCertificates. It serves
@@ -104,17 +109,73 @@ class SSLCertChain {
 
   std::vector<SSLCertificate*> certs_;
 
-  DISALLOW_COPY_AND_ASSIGN(SSLCertChain);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SSLCertChain);
 };
 
-// Parameters for generating an identity for testing. If common_name is
-// non-empty, it will be used for the certificate's subject and issuer name,
-// otherwise a random string will be used. |not_before| and |not_after| are
-// offsets to the current time in number of seconds.
+// KT_DEFAULT is currently an alias for KT_RSA.  This is likely to change.
+// KT_LAST is intended for vector declarations and loops over all key types;
+// it does not represent any key type in itself.
+// TODO(hbos,torbjorng): Don't change KT_DEFAULT without first updating
+// PeerConnectionFactory_nativeCreatePeerConnection's certificate generation
+// code.
+enum KeyType { KT_RSA, KT_ECDSA, KT_LAST, KT_DEFAULT = KT_RSA };
+
+static const int kRsaDefaultModSize = 1024;
+static const int kRsaDefaultExponent = 0x10001;  // = 2^16+1 = 65537
+static const int kRsaMinModSize = 1024;
+static const int kRsaMaxModSize = 8192;
+
+struct RSAParams {
+  unsigned int mod_size;
+  unsigned int pub_exp;
+};
+
+enum ECCurve { EC_NIST_P256, /* EC_FANCY, */ EC_LAST };
+
+class KeyParams {
+ public:
+  // Generate a KeyParams object from a simple KeyType, using default params.
+  explicit KeyParams(KeyType key_type = KT_DEFAULT);
+
+  // Generate a a KeyParams for RSA with explicit parameters.
+  static KeyParams RSA(int mod_size = kRsaDefaultModSize,
+                       int pub_exp = kRsaDefaultExponent);
+
+  // Generate a a KeyParams for ECDSA specifying the curve.
+  static KeyParams ECDSA(ECCurve curve = EC_NIST_P256);
+
+  // Check validity of a KeyParams object. Since the factory functions have
+  // no way of returning errors, this function can be called after creation
+  // to make sure the parameters are OK.
+  bool IsValid() const;
+
+  RSAParams rsa_params() const;
+
+  ECCurve ec_curve() const;
+
+  KeyType type() const { return type_; }
+
+ private:
+  KeyType type_;
+  union {
+    RSAParams rsa;
+    ECCurve curve;
+  } params_;
+};
+
+// TODO(hbos): Remove once rtc::KeyType (to be modified) and
+// blink::WebRTCKeyType (to be landed) match. By using this function in Chromium
+// appropriately we can change KeyType enum -> class without breaking Chromium.
+KeyType IntKeyTypeFamilyToKeyType(int key_type_family);
+
+// Parameters for generating a certificate. If |common_name| is non-empty, it
+// will be used for the certificate's subject and issuer name, otherwise a
+// random string will be used.
 struct SSLIdentityParams {
   std::string common_name;
-  int not_before;  // in seconds.
-  int not_after;  // in seconds.
+  time_t not_before;  // Absolute time since epoch in seconds.
+  time_t not_after;   // Absolute time since epoch in seconds.
+  KeyParams key_params;
 };
 
 // Our identity in an SSL negotiation: a keypair and certificate (both
@@ -127,7 +188,12 @@ class SSLIdentity {
   // subject and issuer name, otherwise a random string will be used.
   // Returns NULL on failure.
   // Caller is responsible for freeing the returned object.
-  static SSLIdentity* Generate(const std::string& common_name);
+  static SSLIdentity* Generate(const std::string& common_name,
+                               const KeyParams& key_param);
+  static SSLIdentity* Generate(const std::string& common_name,
+                               KeyType key_type) {
+    return Generate(common_name, KeyParams(key_type));
+  }
 
   // Generates an identity with the specified validity period.
   static SSLIdentity* GenerateForTest(const SSLIdentityParams& params);
@@ -141,6 +207,7 @@ class SSLIdentity {
   // Returns a new SSLIdentity object instance wrapping the same
   // identity information.
   // Caller is responsible for freeing the returned object.
+  // TODO(hbos,torbjorng): Rename to a less confusing name.
   virtual SSLIdentity* GetReference() const = 0;
 
   // Returns a temporary reference to the certificate.
@@ -155,8 +222,14 @@ class SSLIdentity {
                               size_t length);
 };
 
+// Convert from ASN1 time as restricted by RFC 5280 to seconds from 1970-01-01
+// 00.00 ("epoch").  If the ASN1 time cannot be read, return -1.  The data at
+// |s| is not 0-terminated; its char count is defined by |length|.
+int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format);
+
 extern const char kPemTypeCertificate[];
 extern const char kPemTypeRsaPrivateKey[];
+extern const char kPemTypeEcPrivateKey[];
 
 }  // namespace rtc
 
diff --git a/media/webrtc/trunk/webrtc/base/sslidentity_unittest.cc b/media/webrtc/trunk/webrtc/base/sslidentity_unittest.cc
index 3f756ef895..3582edb4a4 100644
--- a/media/webrtc/trunk/webrtc/base/sslidentity_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/sslidentity_unittest.cc
@@ -11,6 +11,7 @@
 #include <string>
 
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/helpers.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslidentity.h"
 
@@ -30,125 +31,195 @@ const char kTestCertificate[] = "-----BEGIN CERTIFICATE-----\n"
     "itAE+OjGF+PFKbwX8Q==\n"
     "-----END CERTIFICATE-----\n";
 
-const unsigned char kTestCertSha1[] = {0xA6, 0xC8, 0x59, 0xEA,
-                                       0xC3, 0x7E, 0x6D, 0x33,
-                                       0xCF, 0xE2, 0x69, 0x9D,
-                                       0x74, 0xE6, 0xF6, 0x8A,
-                                       0x9E, 0x47, 0xA7, 0xCA};
+const unsigned char kTestCertSha1[] = {
+    0xA6, 0xC8, 0x59, 0xEA, 0xC3, 0x7E, 0x6D, 0x33,
+    0xCF, 0xE2, 0x69, 0x9D, 0x74, 0xE6, 0xF6, 0x8A,
+    0x9E, 0x47, 0xA7, 0xCA};
+const unsigned char kTestCertSha224[] = {
+    0xd4, 0xce, 0xc6, 0xcf, 0x28, 0xcb, 0xe9, 0x77,
+    0x38, 0x36, 0xcf, 0xb1, 0x3b, 0x4a, 0xd7, 0xbd,
+    0xae, 0x24, 0x21, 0x08, 0xcf, 0x6a, 0x44, 0x0d,
+    0x3f, 0x94, 0x2a, 0x5b};
+const unsigned char kTestCertSha256[] = {
+    0x41, 0x6b, 0xb4, 0x93, 0x47, 0x79, 0x77, 0x24,
+    0x77, 0x0b, 0x8b, 0x2e, 0xa6, 0x2b, 0xe0, 0xf9,
+    0x0a, 0xed, 0x1f, 0x31, 0xa6, 0xf7, 0x5c, 0xa1,
+    0x5a, 0xc4, 0xb0, 0xa2, 0xa4, 0x78, 0xb9, 0x76};
+const unsigned char kTestCertSha384[] = {
+    0x42, 0x31, 0x9a, 0x79, 0x1d, 0xd6, 0x08, 0xbf,
+    0x3b, 0xba, 0x36, 0xd8, 0x37, 0x4a, 0x9a, 0x75,
+    0xd3, 0x25, 0x6e, 0x28, 0x92, 0xbe, 0x06, 0xb7,
+    0xc5, 0xa0, 0x83, 0xe3, 0x86, 0xb1, 0x03, 0xfc,
+    0x64, 0x47, 0xd6, 0xd8, 0xaa, 0xd9, 0x36, 0x60,
+    0x04, 0xcc, 0xbe, 0x7d, 0x6a, 0xe8, 0x34, 0x49};
+const unsigned char kTestCertSha512[] = {
+    0x51, 0x1d, 0xec, 0x02, 0x3d, 0x51, 0x45, 0xd3,
+    0xd8, 0x1d, 0xa4, 0x9d, 0x43, 0xc9, 0xee, 0x32,
+    0x6f, 0x4f, 0x37, 0xee, 0xab, 0x3f, 0x25, 0xdf,
+    0x72, 0xfc, 0x61, 0x1a, 0xd5, 0x92, 0xff, 0x6b,
+    0x28, 0x71, 0x58, 0xb3, 0xe1, 0x8a, 0x18, 0xcf,
+    0x61, 0x33, 0x0e, 0x14, 0xc3, 0x04, 0xaa, 0x07,
+    0xf6, 0xa5, 0xda, 0xdc, 0x42, 0x42, 0x22, 0x35,
+    0xce, 0x26, 0x58, 0x4a, 0x33, 0x6d, 0xbc, 0xb6};
 
 class SSLIdentityTest : public testing::Test {
  public:
-  SSLIdentityTest() :
-      identity1_(), identity2_() {
-  }
+  SSLIdentityTest() {}
 
   ~SSLIdentityTest() {
   }
 
   virtual void SetUp() {
-    identity1_.reset(SSLIdentity::Generate("test1"));
-    identity2_.reset(SSLIdentity::Generate("test2"));
+    identity_rsa1_.reset(SSLIdentity::Generate("test1", rtc::KT_RSA));
+    identity_rsa2_.reset(SSLIdentity::Generate("test2", rtc::KT_RSA));
+    identity_ecdsa1_.reset(SSLIdentity::Generate("test3", rtc::KT_ECDSA));
+    identity_ecdsa2_.reset(SSLIdentity::Generate("test4", rtc::KT_ECDSA));
 
-    ASSERT_TRUE(identity1_);
-    ASSERT_TRUE(identity2_);
+    ASSERT_TRUE(identity_rsa1_);
+    ASSERT_TRUE(identity_rsa2_);
+    ASSERT_TRUE(identity_ecdsa1_);
+    ASSERT_TRUE(identity_ecdsa2_);
 
-    test_cert_.reset(
-        rtc::SSLCertificate::FromPEMString(kTestCertificate));
+    test_cert_.reset(rtc::SSLCertificate::FromPEMString(kTestCertificate));
     ASSERT_TRUE(test_cert_);
   }
 
   void TestGetSignatureDigestAlgorithm() {
     std::string digest_algorithm;
-    // Both NSSIdentity::Generate and OpenSSLIdentity::Generate are
-    // hard-coded to generate RSA-SHA1 certificates.
-    ASSERT_TRUE(identity1_->certificate().GetSignatureDigestAlgorithm(
+
+    ASSERT_TRUE(identity_rsa1_->certificate().GetSignatureDigestAlgorithm(
         &digest_algorithm));
-    ASSERT_EQ(rtc::DIGEST_SHA_1, digest_algorithm);
-    ASSERT_TRUE(identity2_->certificate().GetSignatureDigestAlgorithm(
+    ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm);
+
+    ASSERT_TRUE(identity_rsa2_->certificate().GetSignatureDigestAlgorithm(
         &digest_algorithm));
-    ASSERT_EQ(rtc::DIGEST_SHA_1, digest_algorithm);
+    ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm);
+
+    ASSERT_TRUE(identity_ecdsa1_->certificate().GetSignatureDigestAlgorithm(
+        &digest_algorithm));
+    ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm);
+
+    ASSERT_TRUE(identity_ecdsa2_->certificate().GetSignatureDigestAlgorithm(
+        &digest_algorithm));
+    ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm);
 
     // The test certificate has an MD5-based signature.
     ASSERT_TRUE(test_cert_->GetSignatureDigestAlgorithm(&digest_algorithm));
     ASSERT_EQ(rtc::DIGEST_MD5, digest_algorithm);
   }
 
-  void TestDigest(const std::string &algorithm, size_t expected_len,
-                  const unsigned char *expected_digest = NULL) {
-    unsigned char digest1[64];
-    unsigned char digest1b[64];
-    unsigned char digest2[64];
-    size_t digest1_len;
-    size_t digest1b_len;
-    size_t digest2_len;
+  typedef unsigned char DigestType[rtc::MessageDigest::kMaxSize];
+
+  void TestDigestHelper(DigestType digest,
+                        const SSLIdentity* identity,
+                        const std::string& algorithm,
+                        size_t expected_len) {
+    DigestType digest1;
+    size_t digest_len;
     bool rv;
 
-    rv = identity1_->certificate().ComputeDigest(algorithm,
-                                                 digest1, sizeof(digest1),
-                                                 &digest1_len);
+    memset(digest, 0, expected_len);
+    rv = identity->certificate().ComputeDigest(algorithm, digest,
+                                               sizeof(DigestType), &digest_len);
     EXPECT_TRUE(rv);
-    EXPECT_EQ(expected_len, digest1_len);
+    EXPECT_EQ(expected_len, digest_len);
 
-    rv = identity1_->certificate().ComputeDigest(algorithm,
-                                                 digest1b, sizeof(digest1b),
-                                                 &digest1b_len);
+    // Repeat digest computation for the identity as a sanity check.
+    memset(digest1, 0xff, expected_len);
+    rv = identity->certificate().ComputeDigest(algorithm, digest1,
+                                               sizeof(DigestType), &digest_len);
     EXPECT_TRUE(rv);
-    EXPECT_EQ(expected_len, digest1b_len);
-    EXPECT_EQ(0, memcmp(digest1, digest1b, expected_len));
+    EXPECT_EQ(expected_len, digest_len);
 
+    EXPECT_EQ(0, memcmp(digest, digest1, expected_len));
+  }
 
-    rv = identity2_->certificate().ComputeDigest(algorithm,
-                                                 digest2, sizeof(digest2),
-                                                 &digest2_len);
-    EXPECT_TRUE(rv);
-    EXPECT_EQ(expected_len, digest2_len);
-    EXPECT_NE(0, memcmp(digest1, digest2, expected_len));
+  void TestDigestForGeneratedCert(const std::string& algorithm,
+                                  size_t expected_len) {
+    DigestType digest[4];
 
-    // If we have an expected hash for the test cert, check it.
-    if (expected_digest) {
-      unsigned char digest3[64];
-      size_t digest3_len;
+    ASSERT_TRUE(expected_len <= sizeof(DigestType));
 
-      rv = test_cert_->ComputeDigest(algorithm, digest3, sizeof(digest3),
-                                    &digest3_len);
-      EXPECT_TRUE(rv);
-      EXPECT_EQ(expected_len, digest3_len);
-      EXPECT_EQ(0, memcmp(digest3, expected_digest, expected_len));
+    TestDigestHelper(digest[0], identity_rsa1_.get(), algorithm, expected_len);
+    TestDigestHelper(digest[1], identity_rsa2_.get(), algorithm, expected_len);
+    TestDigestHelper(digest[2], identity_ecdsa1_.get(), algorithm,
+                     expected_len);
+    TestDigestHelper(digest[3], identity_ecdsa2_.get(), algorithm,
+                     expected_len);
+
+    // Sanity check that all four digests are unique.  This could theoretically
+    // fail, since cryptographic hash collisions have a non-zero probability.
+    for (int i = 0; i < 4; i++) {
+      for (int j = 0; j < 4; j++) {
+        if (i != j)
+          EXPECT_NE(0, memcmp(digest[i], digest[j], expected_len));
+      }
     }
   }
 
+  void TestDigestForFixedCert(const std::string& algorithm,
+                              size_t expected_len,
+                              const unsigned char* expected_digest) {
+    bool rv;
+    DigestType digest;
+    size_t digest_len;
+
+    ASSERT_TRUE(expected_len <= sizeof(DigestType));
+
+    rv = test_cert_->ComputeDigest(algorithm, digest, sizeof(digest),
+                                   &digest_len);
+    EXPECT_TRUE(rv);
+    EXPECT_EQ(expected_len, digest_len);
+    EXPECT_EQ(0, memcmp(digest, expected_digest, expected_len));
+  }
+
  private:
-  rtc::scoped_ptr<SSLIdentity> identity1_;
-  rtc::scoped_ptr<SSLIdentity> identity2_;
+  rtc::scoped_ptr<SSLIdentity> identity_rsa1_;
+  rtc::scoped_ptr<SSLIdentity> identity_rsa2_;
+  rtc::scoped_ptr<SSLIdentity> identity_ecdsa1_;
+  rtc::scoped_ptr<SSLIdentity> identity_ecdsa2_;
   rtc::scoped_ptr<rtc::SSLCertificate> test_cert_;
 };
 
-TEST_F(SSLIdentityTest, DigestSHA1) {
-  TestDigest(rtc::DIGEST_SHA_1, 20, kTestCertSha1);
+TEST_F(SSLIdentityTest, FixedDigestSHA1) {
+  TestDigestForFixedCert(rtc::DIGEST_SHA_1, 20, kTestCertSha1);
+}
+
+// HASH_AlgSHA224 is not supported in the chromium linux build.
+TEST_F(SSLIdentityTest, FixedDigestSHA224) {
+  TestDigestForFixedCert(rtc::DIGEST_SHA_224, 28, kTestCertSha224);
+}
+
+TEST_F(SSLIdentityTest, FixedDigestSHA256) {
+  TestDigestForFixedCert(rtc::DIGEST_SHA_256, 32, kTestCertSha256);
+}
+
+TEST_F(SSLIdentityTest, FixedDigestSHA384) {
+  TestDigestForFixedCert(rtc::DIGEST_SHA_384, 48, kTestCertSha384);
+}
+
+TEST_F(SSLIdentityTest, FixedDigestSHA512) {
+  TestDigestForFixedCert(rtc::DIGEST_SHA_512, 64, kTestCertSha512);
 }
 
 // HASH_AlgSHA224 is not supported in the chromium linux build.
-#if SSL_USE_NSS
-TEST_F(SSLIdentityTest, DISABLED_DigestSHA224) {
-#else
 TEST_F(SSLIdentityTest, DigestSHA224) {
-#endif
-  TestDigest(rtc::DIGEST_SHA_224, 28);
+  TestDigestForGeneratedCert(rtc::DIGEST_SHA_224, 28);
 }
 
 TEST_F(SSLIdentityTest, DigestSHA256) {
-  TestDigest(rtc::DIGEST_SHA_256, 32);
+  TestDigestForGeneratedCert(rtc::DIGEST_SHA_256, 32);
 }
 
 TEST_F(SSLIdentityTest, DigestSHA384) {
-  TestDigest(rtc::DIGEST_SHA_384, 48);
+  TestDigestForGeneratedCert(rtc::DIGEST_SHA_384, 48);
 }
 
 TEST_F(SSLIdentityTest, DigestSHA512) {
-  TestDigest(rtc::DIGEST_SHA_512, 64);
+  TestDigestForGeneratedCert(rtc::DIGEST_SHA_512, 64);
 }
 
-TEST_F(SSLIdentityTest, FromPEMStrings) {
+TEST_F(SSLIdentityTest, FromPEMStringsRSA) {
   static const char kRSA_PRIVATE_KEY_PEM[] =
       "-----BEGIN RSA PRIVATE KEY-----\n"
       "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
@@ -186,6 +257,33 @@ TEST_F(SSLIdentityTest, FromPEMStrings) {
   EXPECT_EQ(kCERT_PEM, identity->certificate().ToPEMString());
 }
 
+TEST_F(SSLIdentityTest, FromPEMStringsEC) {
+  static const char kRSA_PRIVATE_KEY_PEM[] =
+      "-----BEGIN EC PRIVATE KEY-----\n"
+      "MHcCAQEEIKkIztWLPbs4Y2zWv7VW2Ov4is2ifleCuPgRB8fRv3IkoAoGCCqGSM49\n"
+      "AwEHoUQDQgAEDPV33NrhSdhg9cBRkUWUXnVMXc3h17i9ARbSmNgminKcBXb8/y8L\n"
+      "A76cMWQPPM0ybHO8OS7ZVg2U/m+TwE1M2g==\n"
+      "-----END EC PRIVATE KEY-----\n";
+  static const char kCERT_PEM[] =
+      "-----BEGIN CERTIFICATE-----\n"
+      "MIIB0jCCAXmgAwIBAgIJAMCjpFt9t6LMMAoGCCqGSM49BAMCMEUxCzAJBgNVBAYT\n"
+      "AkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBXaWRn\n"
+      "aXRzIFB0eSBMdGQwIBcNMTUwNjMwMTMwMTIyWhgPMjI4OTA0MTMxMzAxMjJaMEUx\n"
+      "CzAJBgNVBAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRl\n"
+      "cm5ldCBXaWRnaXRzIFB0eSBMdGQwWTATBgcqhkjOPQIBBggqhkjOPQMBBwNCAAQM\n"
+      "9Xfc2uFJ2GD1wFGRRZRedUxdzeHXuL0BFtKY2CaKcpwFdvz/LwsDvpwxZA88zTJs\n"
+      "c7w5LtlWDZT+b5PATUzao1AwTjAdBgNVHQ4EFgQUYHq6nxNNIE832ZmaHc/noODO\n"
+      "rtAwHwYDVR0jBBgwFoAUYHq6nxNNIE832ZmaHc/noODOrtAwDAYDVR0TBAUwAwEB\n"
+      "/zAKBggqhkjOPQQDAgNHADBEAiAQRojsTyZG0BlKoU7gOt5h+yAMLl2cxmDtOIQr\n"
+      "GWP/PwIgJynB4AUDsPT0DWmethOXYijB5sY5UPd9DvgmiS/Mr6s=\n"
+      "-----END CERTIFICATE-----\n";
+
+  rtc::scoped_ptr<SSLIdentity> identity(
+      SSLIdentity::FromPEMStrings(kRSA_PRIVATE_KEY_PEM, kCERT_PEM));
+  EXPECT_TRUE(identity);
+  EXPECT_EQ(kCERT_PEM, identity->certificate().ToPEMString());
+}
+
 TEST_F(SSLIdentityTest, PemDerConversion) {
   std::string der;
   EXPECT_TRUE(SSLIdentity::PemToDer("CERTIFICATE", kTestCertificate, &der));
@@ -198,3 +296,119 @@ TEST_F(SSLIdentityTest, PemDerConversion) {
 TEST_F(SSLIdentityTest, GetSignatureDigestAlgorithm) {
   TestGetSignatureDigestAlgorithm();
 }
+
+class SSLIdentityExpirationTest : public testing::Test {
+ public:
+  SSLIdentityExpirationTest() {
+    // Set use of the test RNG to get deterministic expiration timestamp.
+    rtc::SetRandomTestMode(true);
+  }
+  ~SSLIdentityExpirationTest() {
+    // Put it back for the next test.
+    rtc::SetRandomTestMode(false);
+  }
+
+  void TestASN1TimeToSec() {
+    struct asn_example {
+      const char* string;
+      bool long_format;
+      int64_t want;
+    } static const data[] = {
+      // Valid examples.
+      {"19700101000000Z",  true,  0},
+      {"700101000000Z",    false, 0},
+      {"19700101000001Z",  true,  1},
+      {"700101000001Z",    false, 1},
+      {"19700101000100Z",  true,  60},
+      {"19700101000101Z",  true,  61},
+      {"19700101010000Z",  true,  3600},
+      {"19700101010001Z",  true,  3601},
+      {"19700101010100Z",  true,  3660},
+      {"19700101010101Z",  true,  3661},
+      {"710911012345Z",    false, 53400225},
+      {"20000101000000Z",  true,  946684800},
+      {"20000101000000Z",  true,  946684800},
+      {"20151130140156Z",  true,  1448892116},
+      {"151130140156Z",    false, 1448892116},
+      {"20491231235959Z",  true,  2524607999},
+      {"491231235959Z",    false, 2524607999},
+      {"20500101000000Z",  true,  2524607999+1},
+      {"20700101000000Z",  true,  3155760000},
+      {"21000101000000Z",  true,  4102444800},
+      {"24000101000000Z",  true,  13569465600},
+
+      // Invalid examples.
+      {"19700101000000",    true,  -1},  // missing Z long format
+      {"19700101000000X",   true,  -1},  // X instead of Z long format
+      {"197001010000000",   true,  -1},  // 0 instead of Z long format
+      {"1970010100000000Z", true,  -1},  // excess digits long format
+      {"700101000000",      false, -1},  // missing Z short format
+      {"700101000000X",     false, -1},  // X instead of Z short format
+      {"7001010000000",     false, -1},  // 0 instead of Z short format
+      {"70010100000000Z",   false, -1},  // excess digits short format
+      {":9700101000000Z",   true,  -1},  // invalid character
+      {"1:700101000001Z",   true,  -1},  // invalid character
+      {"19:00101000100Z",   true,  -1},  // invalid character
+      {"197:0101000101Z",   true,  -1},  // invalid character
+      {"1970:101010000Z",   true,  -1},  // invalid character
+      {"19700:01010001Z",   true,  -1},  // invalid character
+      {"197001:1010100Z",   true,  -1},  // invalid character
+      {"1970010:010101Z",   true,  -1},  // invalid character
+      {"70010100:000Z",     false, -1},  // invalid character
+      {"700101000:01Z",     false, -1},  // invalid character
+      {"2000010100:000Z",   true,  -1},  // invalid character
+      {"21000101000:00Z",   true,  -1},  // invalid character
+      {"240001010000:0Z",   true,  -1},  // invalid character
+      {"500101000000Z",     false, -1},  // but too old for epoch
+      {"691231235959Z",     false, -1},  // too old for epoch
+      {"19611118043000Z",   false, -1},  // way too old for epoch
+    };
+
+    unsigned char buf[20];
+
+    // Run all examples and check for the expected result.
+    for (const auto& entry : data) {
+      size_t length = strlen(entry.string);
+      memcpy(buf, entry.string, length);    // Copy the ASN1 string...
+      buf[length] = rtc::CreateRandomId();  // ...and terminate it with junk.
+      int64_t res = rtc::ASN1TimeToSec(buf, length, entry.long_format);
+      LOG(LS_VERBOSE) << entry.string;
+      ASSERT_EQ(entry.want, res);
+    }
+    // Run all examples again, but with an invalid length.
+    for (const auto& entry : data) {
+      size_t length = strlen(entry.string);
+      memcpy(buf, entry.string, length);    // Copy the ASN1 string...
+      buf[length] = rtc::CreateRandomId();  // ...and terminate it with junk.
+      int64_t res = rtc::ASN1TimeToSec(buf, length - 1, entry.long_format);
+      LOG(LS_VERBOSE) << entry.string;
+      ASSERT_EQ(-1, res);
+    }
+  }
+
+  void TestExpireTime(int times) {
+    for (int i = 0; i < times; i++) {
+      rtc::SSLIdentityParams params;
+      params.common_name = "";
+      params.not_before = 0;
+      // We limit the time to < 2^31 here, i.e., we stay before 2038, since else
+      // we hit time offset limitations in OpenSSL on some 32-bit systems.
+      params.not_after = rtc::CreateRandomId() % 0x80000000;
+      // We test just ECDSA here since what we're out to exercise here is the
+      // code for expiration setting and reading.
+      params.key_params = rtc::KeyParams::ECDSA(rtc::EC_NIST_P256);
+      SSLIdentity* identity = rtc::SSLIdentity::GenerateForTest(params);
+      EXPECT_EQ(params.not_after,
+                identity->certificate().CertificateExpirationTime());
+      delete identity;
+    }
+  }
+};
+
+TEST_F(SSLIdentityExpirationTest, TestASN1TimeToSec) {
+  TestASN1TimeToSec();
+}
+
+TEST_F(SSLIdentityExpirationTest, TestExpireTime) {
+  TestExpireTime(500);
+}
diff --git a/media/webrtc/trunk/webrtc/base/sslroots.h b/media/webrtc/trunk/webrtc/base/sslroots.h
index 31d601c169..0464ac8339 100644
--- a/media/webrtc/trunk/webrtc/base/sslroots.h
+++ b/media/webrtc/trunk/webrtc/base/sslroots.h
@@ -2,83 +2,813 @@
 // Google.
 
 // It was generated with the following command line:
-// > python //depot/googleclient/talk/tools/generate_sslroots.py
-//    //depot/google3/security/cacerts/for_connecting_to_google/roots.pem
+// > python tools/sslroots/generate_sslroots.py
+//    https://pki.google.com/roots.pem
 
-/* subject:/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
-/* issuer :/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
+/* subject:/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
+/* issuer :/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
 
-namespace rtc {
 
-const unsigned char AddTrust_External_Root_certificate[1082]={
-0x30,0x82,0x04,0x36,0x30,0x82,0x03,0x1E,0xA0,0x03,0x02,0x01,0x02,0x02,0x01,0x01,
+const unsigned char GlobalSign_Root_CA_certificate[889]={
+0x30,0x82,0x03,0x75,0x30,0x82,0x02,0x5D,0xA0,0x03,0x02,0x01,0x02,0x02,0x0B,0x04,
+0x00,0x00,0x00,0x00,0x01,0x15,0x4B,0x5A,0xC3,0x94,0x30,0x0D,0x06,0x09,0x2A,0x86,
+0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,0x57,0x31,0x0B,0x30,0x09,0x06,
+0x03,0x55,0x04,0x06,0x13,0x02,0x42,0x45,0x31,0x19,0x30,0x17,0x06,0x03,0x55,0x04,
+0x0A,0x13,0x10,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x20,0x6E,0x76,
+0x2D,0x73,0x61,0x31,0x10,0x30,0x0E,0x06,0x03,0x55,0x04,0x0B,0x13,0x07,0x52,0x6F,
+0x6F,0x74,0x20,0x43,0x41,0x31,0x1B,0x30,0x19,0x06,0x03,0x55,0x04,0x03,0x13,0x12,
+0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x20,0x52,0x6F,0x6F,0x74,0x20,
+0x43,0x41,0x30,0x1E,0x17,0x0D,0x39,0x38,0x30,0x39,0x30,0x31,0x31,0x32,0x30,0x30,
+0x30,0x30,0x5A,0x17,0x0D,0x32,0x38,0x30,0x31,0x32,0x38,0x31,0x32,0x30,0x30,0x30,
+0x30,0x5A,0x30,0x57,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x42,
+0x45,0x31,0x19,0x30,0x17,0x06,0x03,0x55,0x04,0x0A,0x13,0x10,0x47,0x6C,0x6F,0x62,
+0x61,0x6C,0x53,0x69,0x67,0x6E,0x20,0x6E,0x76,0x2D,0x73,0x61,0x31,0x10,0x30,0x0E,
+0x06,0x03,0x55,0x04,0x0B,0x13,0x07,0x52,0x6F,0x6F,0x74,0x20,0x43,0x41,0x31,0x1B,
+0x30,0x19,0x06,0x03,0x55,0x04,0x03,0x13,0x12,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,
+0x69,0x67,0x6E,0x20,0x52,0x6F,0x6F,0x74,0x20,0x43,0x41,0x30,0x82,0x01,0x22,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x01,0x05,0x00,0x03,0x82,
+0x01,0x0F,0x00,0x30,0x82,0x01,0x0A,0x02,0x82,0x01,0x01,0x00,0xDA,0x0E,0xE6,0x99,
+0x8D,0xCE,0xA3,0xE3,0x4F,0x8A,0x7E,0xFB,0xF1,0x8B,0x83,0x25,0x6B,0xEA,0x48,0x1F,
+0xF1,0x2A,0xB0,0xB9,0x95,0x11,0x04,0xBD,0xF0,0x63,0xD1,0xE2,0x67,0x66,0xCF,0x1C,
+0xDD,0xCF,0x1B,0x48,0x2B,0xEE,0x8D,0x89,0x8E,0x9A,0xAF,0x29,0x80,0x65,0xAB,0xE9,
+0xC7,0x2D,0x12,0xCB,0xAB,0x1C,0x4C,0x70,0x07,0xA1,0x3D,0x0A,0x30,0xCD,0x15,0x8D,
+0x4F,0xF8,0xDD,0xD4,0x8C,0x50,0x15,0x1C,0xEF,0x50,0xEE,0xC4,0x2E,0xF7,0xFC,0xE9,
+0x52,0xF2,0x91,0x7D,0xE0,0x6D,0xD5,0x35,0x30,0x8E,0x5E,0x43,0x73,0xF2,0x41,0xE9,
+0xD5,0x6A,0xE3,0xB2,0x89,0x3A,0x56,0x39,0x38,0x6F,0x06,0x3C,0x88,0x69,0x5B,0x2A,
+0x4D,0xC5,0xA7,0x54,0xB8,0x6C,0x89,0xCC,0x9B,0xF9,0x3C,0xCA,0xE5,0xFD,0x89,0xF5,
+0x12,0x3C,0x92,0x78,0x96,0xD6,0xDC,0x74,0x6E,0x93,0x44,0x61,0xD1,0x8D,0xC7,0x46,
+0xB2,0x75,0x0E,0x86,0xE8,0x19,0x8A,0xD5,0x6D,0x6C,0xD5,0x78,0x16,0x95,0xA2,0xE9,
+0xC8,0x0A,0x38,0xEB,0xF2,0x24,0x13,0x4F,0x73,0x54,0x93,0x13,0x85,0x3A,0x1B,0xBC,
+0x1E,0x34,0xB5,0x8B,0x05,0x8C,0xB9,0x77,0x8B,0xB1,0xDB,0x1F,0x20,0x91,0xAB,0x09,
+0x53,0x6E,0x90,0xCE,0x7B,0x37,0x74,0xB9,0x70,0x47,0x91,0x22,0x51,0x63,0x16,0x79,
+0xAE,0xB1,0xAE,0x41,0x26,0x08,0xC8,0x19,0x2B,0xD1,0x46,0xAA,0x48,0xD6,0x64,0x2A,
+0xD7,0x83,0x34,0xFF,0x2C,0x2A,0xC1,0x6C,0x19,0x43,0x4A,0x07,0x85,0xE7,0xD3,0x7C,
+0xF6,0x21,0x68,0xEF,0xEA,0xF2,0x52,0x9F,0x7F,0x93,0x90,0xCF,0x02,0x03,0x01,0x00,
+0x01,0xA3,0x42,0x30,0x40,0x30,0x0E,0x06,0x03,0x55,0x1D,0x0F,0x01,0x01,0xFF,0x04,
+0x04,0x03,0x02,0x01,0x06,0x30,0x0F,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,
+0x05,0x30,0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,
+0x14,0x60,0x7B,0x66,0x1A,0x45,0x0D,0x97,0xCA,0x89,0x50,0x2F,0x7D,0x04,0xCD,0x34,
+0xA8,0xFF,0xFC,0xFD,0x4B,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,
+0x01,0x05,0x05,0x00,0x03,0x82,0x01,0x01,0x00,0xD6,0x73,0xE7,0x7C,0x4F,0x76,0xD0,
+0x8D,0xBF,0xEC,0xBA,0xA2,0xBE,0x34,0xC5,0x28,0x32,0xB5,0x7C,0xFC,0x6C,0x9C,0x2C,
+0x2B,0xBD,0x09,0x9E,0x53,0xBF,0x6B,0x5E,0xAA,0x11,0x48,0xB6,0xE5,0x08,0xA3,0xB3,
+0xCA,0x3D,0x61,0x4D,0xD3,0x46,0x09,0xB3,0x3E,0xC3,0xA0,0xE3,0x63,0x55,0x1B,0xF2,
+0xBA,0xEF,0xAD,0x39,0xE1,0x43,0xB9,0x38,0xA3,0xE6,0x2F,0x8A,0x26,0x3B,0xEF,0xA0,
+0x50,0x56,0xF9,0xC6,0x0A,0xFD,0x38,0xCD,0xC4,0x0B,0x70,0x51,0x94,0x97,0x98,0x04,
+0xDF,0xC3,0x5F,0x94,0xD5,0x15,0xC9,0x14,0x41,0x9C,0xC4,0x5D,0x75,0x64,0x15,0x0D,
+0xFF,0x55,0x30,0xEC,0x86,0x8F,0xFF,0x0D,0xEF,0x2C,0xB9,0x63,0x46,0xF6,0xAA,0xFC,
+0xDF,0xBC,0x69,0xFD,0x2E,0x12,0x48,0x64,0x9A,0xE0,0x95,0xF0,0xA6,0xEF,0x29,0x8F,
+0x01,0xB1,0x15,0xB5,0x0C,0x1D,0xA5,0xFE,0x69,0x2C,0x69,0x24,0x78,0x1E,0xB3,0xA7,
+0x1C,0x71,0x62,0xEE,0xCA,0xC8,0x97,0xAC,0x17,0x5D,0x8A,0xC2,0xF8,0x47,0x86,0x6E,
+0x2A,0xC4,0x56,0x31,0x95,0xD0,0x67,0x89,0x85,0x2B,0xF9,0x6C,0xA6,0x5D,0x46,0x9D,
+0x0C,0xAA,0x82,0xE4,0x99,0x51,0xDD,0x70,0xB7,0xDB,0x56,0x3D,0x61,0xE4,0x6A,0xE1,
+0x5C,0xD6,0xF6,0xFE,0x3D,0xDE,0x41,0xCC,0x07,0xAE,0x63,0x52,0xBF,0x53,0x53,0xF4,
+0x2B,0xE9,0xC7,0xFD,0xB6,0xF7,0x82,0x5F,0x85,0xD2,0x41,0x18,0xDB,0x81,0xB3,0x04,
+0x1C,0xC5,0x1F,0xA4,0x80,0x6F,0x15,0x20,0xC9,0xDE,0x0C,0x88,0x0A,0x1D,0xD6,0x66,
+0x55,0xE2,0xFC,0x48,0xC9,0x29,0x26,0x69,0xE0,
+};
+
+
+/* subject:/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust RSA Certification Authority */
+/* issuer :/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust RSA Certification Authority */
+
+
+const unsigned char USERTrust_RSA_Certification_Authority_certificate[1506]={
+0x30,0x82,0x05,0xDE,0x30,0x82,0x03,0xC6,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x01,
+0xFD,0x6D,0x30,0xFC,0xA3,0xCA,0x51,0xA8,0x1B,0xBC,0x64,0x0E,0x35,0x03,0x2D,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x0C,0x05,0x00,0x30,0x81,
+0x88,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x13,
+0x30,0x11,0x06,0x03,0x55,0x04,0x08,0x13,0x0A,0x4E,0x65,0x77,0x20,0x4A,0x65,0x72,
+0x73,0x65,0x79,0x31,0x14,0x30,0x12,0x06,0x03,0x55,0x04,0x07,0x13,0x0B,0x4A,0x65,
+0x72,0x73,0x65,0x79,0x20,0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,0x55,
+0x04,0x0A,0x13,0x15,0x54,0x68,0x65,0x20,0x55,0x53,0x45,0x52,0x54,0x52,0x55,0x53,
+0x54,0x20,0x4E,0x65,0x74,0x77,0x6F,0x72,0x6B,0x31,0x2E,0x30,0x2C,0x06,0x03,0x55,
+0x04,0x03,0x13,0x25,0x55,0x53,0x45,0x52,0x54,0x72,0x75,0x73,0x74,0x20,0x52,0x53,
+0x41,0x20,0x43,0x65,0x72,0x74,0x69,0x66,0x69,0x63,0x61,0x74,0x69,0x6F,0x6E,0x20,
+0x41,0x75,0x74,0x68,0x6F,0x72,0x69,0x74,0x79,0x30,0x1E,0x17,0x0D,0x31,0x30,0x30,
+0x32,0x30,0x31,0x30,0x30,0x30,0x30,0x30,0x30,0x5A,0x17,0x0D,0x33,0x38,0x30,0x31,
+0x31,0x38,0x32,0x33,0x35,0x39,0x35,0x39,0x5A,0x30,0x81,0x88,0x31,0x0B,0x30,0x09,
+0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x13,0x30,0x11,0x06,0x03,0x55,
+0x04,0x08,0x13,0x0A,0x4E,0x65,0x77,0x20,0x4A,0x65,0x72,0x73,0x65,0x79,0x31,0x14,
+0x30,0x12,0x06,0x03,0x55,0x04,0x07,0x13,0x0B,0x4A,0x65,0x72,0x73,0x65,0x79,0x20,
+0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,0x55,0x04,0x0A,0x13,0x15,0x54,
+0x68,0x65,0x20,0x55,0x53,0x45,0x52,0x54,0x52,0x55,0x53,0x54,0x20,0x4E,0x65,0x74,
+0x77,0x6F,0x72,0x6B,0x31,0x2E,0x30,0x2C,0x06,0x03,0x55,0x04,0x03,0x13,0x25,0x55,
+0x53,0x45,0x52,0x54,0x72,0x75,0x73,0x74,0x20,0x52,0x53,0x41,0x20,0x43,0x65,0x72,
+0x74,0x69,0x66,0x69,0x63,0x61,0x74,0x69,0x6F,0x6E,0x20,0x41,0x75,0x74,0x68,0x6F,
+0x72,0x69,0x74,0x79,0x30,0x82,0x02,0x22,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,
+0xF7,0x0D,0x01,0x01,0x01,0x05,0x00,0x03,0x82,0x02,0x0F,0x00,0x30,0x82,0x02,0x0A,
+0x02,0x82,0x02,0x01,0x00,0x80,0x12,0x65,0x17,0x36,0x0E,0xC3,0xDB,0x08,0xB3,0xD0,
+0xAC,0x57,0x0D,0x76,0xED,0xCD,0x27,0xD3,0x4C,0xAD,0x50,0x83,0x61,0xE2,0xAA,0x20,
+0x4D,0x09,0x2D,0x64,0x09,0xDC,0xCE,0x89,0x9F,0xCC,0x3D,0xA9,0xEC,0xF6,0xCF,0xC1,
+0xDC,0xF1,0xD3,0xB1,0xD6,0x7B,0x37,0x28,0x11,0x2B,0x47,0xDA,0x39,0xC6,0xBC,0x3A,
+0x19,0xB4,0x5F,0xA6,0xBD,0x7D,0x9D,0xA3,0x63,0x42,0xB6,0x76,0xF2,0xA9,0x3B,0x2B,
+0x91,0xF8,0xE2,0x6F,0xD0,0xEC,0x16,0x20,0x90,0x09,0x3E,0xE2,0xE8,0x74,0xC9,0x18,
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+0xA8,0xFD,
+};
+
+
+/* subject:/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
+/* issuer :/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
+
+
+const unsigned char Starfield_Class_2_CA_certificate[1043]={
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+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
+
+
+const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+0xBB,0x1D,0xC1,0xD7,0xA5,0x6B,0xCC,0x2F,0xD2,0x42,0xFD,0x49,0xDD,0xA7,0x89,0xCF,
+0x53,0xBA,0xDA,0x00,0x5A,0x28,0xBF,0x82,0xDF,0xF8,0xBA,0x13,0x1D,0x50,0x86,0x82,
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+0x12,0x1A,0x12,0x68,0xB8,0xFB,0x66,0x99,0x14,0x14,0x45,0x5C,0xAE,0xE7,0xAE,0x69,
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+0x54,0x00,0xCF,0xF0,0xF1,0xC1,0xC7,0x98,0x30,0x1A,0x3B,0x36,0x16,0xDB,0xA3,0x6E,
+0xEA,0xFD,0xAD,0xB2,0xC2,0xDA,0xEF,0x02,0x47,0x13,0x8A,0xC0,0xF1,0xB3,0x31,0xAD,
+0x4F,0x1C,0xE1,0x4F,0x9C,0xAF,0x0F,0x0C,0x9D,0xF7,0x78,0x0D,0xD8,0xF4,0x35,0x56,
+0x80,0xDA,0xB7,0x6D,0x17,0x8F,0x9D,0x1E,0x81,0x64,0xE1,0xFE,0xC5,0x45,0xBA,0xAD,
+0x6B,0xB9,0x0A,0x7A,0x4E,0x4F,0x4B,0x84,0xEE,0x4B,0xF1,0x7D,0xDD,0x11,
+};
+
+
+/* subject:/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust ECC Certification Authority */
+/* issuer :/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust ECC Certification Authority */
+
+
+const unsigned char USERTrust_ECC_Certification_Authority_certificate[659]={
+0x30,0x82,0x02,0x8F,0x30,0x82,0x02,0x15,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x5C,
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+0xB3,0x7D,0x32,0x7F,0x8A,0x34,0xE3,0x79,0x79,0xA3,0x42,0x30,0x40,0x30,0x1D,0x06,
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+0xFA,0x64,0xB9,0xEE,0x16,0x42,0xB3,0xBF,0x8A,0xC2,0x09,0xC4,0xEC,0xE4,0xB1,0x4D,
+0x02,0x31,0x00,0xE9,0x2A,0x61,0x47,0x8C,0x52,0x4A,0x4B,0x4E,0x18,0x70,0xF6,0xD6,
+0x44,0xD6,0x6E,0xF5,0x83,0xBA,0x6D,0x58,0xBD,0x24,0xD9,0x56,0x48,0xEA,0xEF,0xC4,
+0xA2,0x46,0x81,0x88,0x6A,0x3A,0x46,0xD1,0xA9,0x9B,0x4D,0xC9,0x61,0xDA,0xD1,0x5D,
+0x57,0x6A,0x18,
+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
+
+
+const unsigned char GeoTrust_Global_CA_certificate[856]={
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+0x4E,0x9B,0x4B,0xEE,0x4A,0x6F,0x55,0x52,0xB3,0x24,0xA1,0x37,0x0A,0xEB,0x64,0x76,
+0x2A,0x2E,0x2C,0xF3,0xFD,0x3B,0x75,0x90,0xBF,0xFA,0x71,0xD8,0xC7,0x3D,0x37,0xD2,
+0xB5,0x05,0x95,0x62,0xB9,0xA6,0xDE,0x89,0x3D,0x36,0x7B,0x38,0x77,0x48,0x97,0xAC,
+0xA6,0x20,0x8F,0x2E,0xA6,0xC9,0x0C,0xC2,0xB2,0x99,0x45,0x00,0xC7,0xCE,0x11,0x51,
+0x22,0x22,0xE0,0xA5,0xEA,0xB6,0x15,0x48,0x09,0x64,0xEA,0x5E,0x4F,0x74,0xF7,0x05,
+0x3E,0xC7,0x8A,0x52,0x0C,0xDB,0x15,0xB4,0xBD,0x6D,0x9B,0xE5,0xC6,0xB1,0x54,0x68,
+0xA9,0xE3,0x69,0x90,0xB6,0x9A,0xA5,0x0F,0xB8,0xB9,0x3F,0x20,0x7D,0xAE,0x4A,0xB5,
+0xB8,0x9C,0xE4,0x1D,0xB6,0xAB,0xE6,0x94,0xA5,0xC1,0xC7,0x83,0xAD,0xDB,0xF5,0x27,
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+0xAE,0x39,0xA6,0x6A,0x74,0xE9,0xDA,0xC4,0xE7,0xBC,0x4D,0x34,0x1E,0xA9,0x5C,0x4D,
+0x33,0x5F,0x92,0x09,0x2F,0x88,0x66,0x5D,0x77,0x97,0xC7,0x1D,0x76,0x13,0xA9,0xD5,
+0xE5,0xF1,0x16,0x09,0x11,0x35,0xD5,0xAC,0xDB,0x24,0x71,0x70,0x2C,0x98,0x56,0x0B,
+0xD9,0x17,0xB4,0xD1,0xE3,0x51,0x2B,0x5E,0x75,0xE8,0xD5,0xD0,0xDC,0x4F,0x34,0xED,
+0xC2,0x05,0x66,0x80,0xA1,0xCB,0xE6,0x33,
+};
+
+
+/* subject:/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
+/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
+
+
+const unsigned char Starfield_Root_Certificate_Authority___G2_certificate[993]={
+0x30,0x82,0x03,0xDD,0x30,0x82,0x02,0xC5,0xA0,0x03,0x02,0x01,0x02,0x02,0x01,0x00,
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+0x79,0x51,0x4E,0xCD,0x90,0xF4,0x67,0x07,0x19,0xC8,0x3C,0x46,0x7A,0x0D,0x01,0x7D,
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+0x1F,0xC5,0xEC,0xFA,0x9C,0x7E,0xCF,0x7E,0xB1,0xF1,0x07,0x2D,0xB6,0xFC,0xBF,0xCA,
+0xA4,0xBF,0xD0,0x97,0x05,0x4A,0xBC,0xEA,0x18,0x28,0x02,0x90,0xBD,0x54,0x78,0x09,
+0x21,0x71,0xD3,0xD1,0x7D,0x1D,0xD9,0x16,0xB0,0xA9,0x61,0x3D,0xD0,0x0A,0x00,0x22,
+0xFC,0xC7,0x7B,0xCB,0x09,0x64,0x45,0x0B,0x3B,0x40,0x81,0xF7,0x7D,0x7C,0x32,0xF5,
+0x98,0xCA,0x58,0x8E,0x7D,0x2A,0xEE,0x90,0x59,0x73,0x64,0xF9,0x36,0x74,0x5E,0x25,
+0xA1,0xF5,0x66,0x05,0x2E,0x7F,0x39,0x15,0xA9,0x2A,0xFB,0x50,0x8B,0x8E,0x85,0x69,
+0xF4,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G3 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G3 */
+
+
+const unsigned char DigiCert_Global_Root_G3_certificate[579]={
+0x30,0x82,0x02,0x3F,0x30,0x82,0x01,0xC5,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x05,
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+0xB3,0x27,0x17,
+};
+
+
+/* subject:/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
+/* issuer :/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
+
+
+const unsigned char thawte_Primary_Root_CA___G2_certificate[652]={
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+0xF7,0x58,0x3F,0x2E,0x72,0x02,0x57,0xA3,0x8F,0xA1,0x14,0x2E,
+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
+
+
+const unsigned char VeriSign_Universal_Root_Certification_Authority_certificate[1213]={
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+0xAC,0xF1,0x63,0xCB,0xB8,0x4C,0x1D,0x59,0x90,0x4A,0xEF,0x90,0x16,0x28,0x1F,0x5A,
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+0xE3,0x21,0xC9,0x24,0x39,0xE9,0xFD,0x15,0x66,0x46,0xF4,0x1B,0x11,0xD0,0x4D,0x73,
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+0x2B,0x18,0x9D,0x81,0xB4,0xC4,0x28,0xDA,0x94,0x97,0xA5,0x70,0xEB,0xAC,0x1D,0xBE,
+0x07,0x11,0xF0,0xD5,0xDB,0xDD,0xE5,0x8C,0xF0,0xD5,0x32,0xB0,0x83,0xE6,0x57,0xE2,
+0x8F,0xBF,0xBE,0xA1,0xAA,0xBF,0x3D,0x1D,0xB5,0xD4,0x38,0xEA,0xD7,0xB0,0x5C,0x3A,
+0x4F,0x6A,0x3F,0x8F,0xC0,0x66,0x6C,0x63,0xAA,0xE9,0xD9,0xA4,0x16,0xF4,0x81,0xD1,
+0x95,0x14,0x0E,0x7D,0xCD,0x95,0x34,0xD9,0xD2,0x8F,0x70,0x73,0x81,0x7B,0x9C,0x7E,
+0xBD,0x98,0x61,0xD8,0x45,0x87,0x98,0x90,0xC5,0xEB,0x86,0x30,0xC6,0x35,0xBF,0xF0,
+0xFF,0xC3,0x55,0x88,0x83,0x4B,0xEF,0x05,0x92,0x06,0x71,0xF2,0xB8,0x98,0x93,0xB7,
+0xEC,0xCD,0x82,0x61,0xF1,0x38,0xE6,0x4F,0x97,0x98,0x2A,0x5A,0x8D,
+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
+
+
+const unsigned char VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate[904]={
+0x30,0x82,0x03,0x84,0x30,0x82,0x03,0x0A,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x2F,
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+0x2D,0xA7,0xD8,0x86,0x2A,0xDD,0x2E,0x10,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G2 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G2 */
+
+
+const unsigned char DigiCert_Global_Root_G2_certificate[914]={
+0x30,0x82,0x03,0x8E,0x30,0x82,0x02,0x76,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x03,
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+0xF2,0xB1,0x8E,0x99,0xA1,0x6F,0x13,0xB1,0x41,0x71,0xFE,0x88,0x2A,0xC8,0x4F,0x10,
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+0x76,0xEE,0x3C,0x8D,0xC4,0x5D,0x56,0x5B,0xA2,0xD9,0x66,0x6E,0xB3,0x35,0x37,0xE5,
+0x32,0xB6,
 };
 
 
@@ -156,6 +886,196 @@ const unsigned char AddTrust_Low_Value_Services_Root_certificate[1052]={
 };
 
 
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
+
+
+const unsigned char AffirmTrust_Premium_ECC_certificate[514]={
+0x30,0x82,0x01,0xFE,0x30,0x82,0x01,0x85,0xA0,0x03,0x02,0x01,0x02,0x02,0x08,0x74,
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+0xDB,0xF4,0x0F,0xA4,0x68,0x4C,0x55,0x72,0x6B,0x95,0x1D,0x4E,0x18,0x42,0x95,0x78,
+0xCC,0x37,0x3C,0x91,0xE2,0x9B,0x65,0x2B,0x29,0xA3,0x42,0x30,0x40,0x30,0x1D,0x06,
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+0x2F,0xB3,0xCA,0x09,0x3C,0x37,0x9D,0xD7,0xE7,0xB8,0x46,0xF1,0xFD,0xA1,0xE2,0x71,
+0x02,0x30,0x42,0x59,0x87,0x43,0xD4,0x51,0xDF,0xBA,0xD3,0x09,0x32,0x5A,0xCE,0x88,
+0x7E,0x57,0x3D,0x9C,0x5F,0x42,0x6B,0xF5,0x07,0x2D,0xB5,0xF0,0x82,0x93,0xF9,0x59,
+0x6F,0xAE,0x64,0xFA,0x58,0xE5,0x8B,0x1E,0xE3,0x63,0xBE,0xB5,0x81,0xCD,0x6F,0x02,
+0x8C,0x79,
+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
+
+
+const unsigned char Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+};
+
+
+/* subject:/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
+/* issuer :/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
+
+
+const unsigned char thawte_Primary_Root_CA_certificate[1060]={
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+};
+
+
 /* subject:/C=SE/O=AddTrust AB/OU=AddTrust TTP Network/CN=AddTrust Public CA Root */
 /* issuer :/C=SE/O=AddTrust AB/OU=AddTrust TTP Network/CN=AddTrust Public CA Root */
 
@@ -305,1822 +1225,6 @@ const unsigned char AddTrust_Qualified_Certificates_Root_certificate[1058]={
 };
 
 
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
-
-
-const unsigned char AffirmTrust_Commercial_certificate[848]={
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-};
-
-
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
-
-
-const unsigned char AffirmTrust_Networking_certificate[848]={
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-};
-
-
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
-
-
-const unsigned char AffirmTrust_Premium_certificate[1354]={
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-};
-
-
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
-
-
-const unsigned char AffirmTrust_Premium_ECC_certificate[514]={
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-};
-
-
-/* subject:/C=US/O=America Online Inc./CN=America Online Root Certification Authority 1 */
-/* issuer :/C=US/O=America Online Inc./CN=America Online Root Certification Authority 1 */
-
-
-const unsigned char America_Online_Root_Certification_Authority_1_certificate[936]={
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-};
-
-
-/* subject:/C=US/O=America Online Inc./CN=America Online Root Certification Authority 2 */
-/* issuer :/C=US/O=America Online Inc./CN=America Online Root Certification Authority 2 */
-
-
-const unsigned char America_Online_Root_Certification_Authority_2_certificate[1448]={
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-};
-
-
-/* subject:/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
-/* issuer :/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
-
-
-const unsigned char Baltimore_CyberTrust_Root_certificate[891]={
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-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
-
-
-const unsigned char Comodo_AAA_Services_root_certificate[1078]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
-
-
-const unsigned char COMODO_Certification_Authority_certificate[1057]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
-
-
-const unsigned char COMODO_ECC_Certification_Authority_certificate[653]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
-
-
-const unsigned char Comodo_Secure_Services_root_certificate[1091]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
-
-
-const unsigned char Comodo_Trusted_Services_root_certificate[1095]={
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-};
-
-
-/* subject:/O=Cybertrust, Inc/CN=Cybertrust Global Root */
-/* issuer :/O=Cybertrust, Inc/CN=Cybertrust Global Root */
-
-
-const unsigned char Cybertrust_Global_Root_certificate[933]={
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-};
-
-
-/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
-/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
-
-
-const unsigned char DigiCert_Assured_ID_Root_CA_certificate[955]={
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-0x10,0x71,0x9D,0xAD,0xE2,0xC3,0xF9,0xC3,0x99,0x51,0xB7,0x2B,0x07,0x08,0xCE,0x2E,
-0xE6,0x50,0xB2,0xA7,0xFA,0x0A,0x45,0x2F,0xA2,0xF0,0xF2,
-};
-
-
-/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
-/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
-
-
-const unsigned char DigiCert_Global_Root_CA_certificate[947]={
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-0x14,0xE9,0xD8,0x89,0xC1,0xB9,0x38,0x6C,0xE2,0x91,0x6C,0x8A,0xFF,0x64,0xB9,0x77,
-0x25,0x57,0x30,0xC0,0x1B,0x24,0xA3,0xE1,0xDC,0xE9,0xDF,0x47,0x7C,0xB5,0xB4,0x24,
-0x08,0x05,0x30,0xEC,0x2D,0xBD,0x0B,0xBF,0x45,0xBF,0x50,0xB9,0xA9,0xF3,0xEB,0x98,
-0x01,0x12,0xAD,0xC8,0x88,0xC6,0x98,0x34,0x5F,0x8D,0x0A,0x3C,0xC6,0xE9,0xD5,0x95,
-0x95,0x6D,0xDE,
-};
-
-
-/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
-/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
-
-
-const unsigned char DigiCert_High_Assurance_EV_Root_CA_certificate[969]={
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-};
-
-
-/* subject:/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
-/* issuer :/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
-
-
-const unsigned char Entrust_net_Premium_2048_Secure_Server_CA_certificate[1120]={
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-};
-
-
-/* subject:/C=US/O=Entrust.net/OU=www.entrust.net/CPS incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Secure Server Certification Authority */
-/* issuer :/C=US/O=Entrust.net/OU=www.entrust.net/CPS incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Secure Server Certification Authority */
-
-
-const unsigned char Entrust_net_Secure_Server_CA_certificate[1244]={
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-};
-
-
-/* subject:/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
-/* issuer :/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
-
-
-const unsigned char Entrust_Root_Certification_Authority_certificate[1173]={
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-};
-
-
-/* subject:/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
-/* issuer :/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
-
-
-const unsigned char Equifax_Secure_CA_certificate[804]={
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-0xAC,0x07,0x77,0x38,
-};
-
-
-/* subject:/C=US/O=Equifax Secure Inc./CN=Equifax Secure eBusiness CA-1 */
-/* issuer :/C=US/O=Equifax Secure Inc./CN=Equifax Secure eBusiness CA-1 */
-
-
-const unsigned char Equifax_Secure_eBusiness_CA_1_certificate[646]={
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-0x5A,0x2A,0x82,0xB2,0x37,0x79,
-};
-
-
-/* subject:/C=US/O=Equifax Secure/OU=Equifax Secure eBusiness CA-2 */
-/* issuer :/C=US/O=Equifax Secure/OU=Equifax Secure eBusiness CA-2 */
-
-
-const unsigned char Equifax_Secure_eBusiness_CA_2_certificate[804]={
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-0xD1,0xC1,0x3E,0x53,0x9F,0x03,0x44,0xB0,0x7E,0x4B,0xF4,0x6F,0xE4,0x7C,0x1F,0xE7,
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-0xED,0x33,0x6B,0xC4,0xD4,0xD7,0x3D,0x12,0x58,0xAB,0x7D,0x09,0x2D,0xCB,0x70,0xF5,
-0x13,0x8A,0x94,0xA1,0x27,0xA4,0xD6,0x70,0xC5,0x6D,0x94,0xB5,0xC9,0x7D,0x9D,0xA0,
-0xD2,0xC6,0x08,0x49,0xD9,0x66,0x9B,0xA6,0xD3,0xF4,0x0B,0xDC,0xC5,0x26,0x57,0xE1,
-0x91,0x30,0xEA,0xCD,
-};
-
-
-/* subject:/C=US/O=Equifax Secure Inc./CN=Equifax Secure Global eBusiness CA-1 */
-/* issuer :/C=US/O=Equifax Secure Inc./CN=Equifax Secure Global eBusiness CA-1 */
-
-
-const unsigned char Equifax_Secure_Global_eBusiness_CA_certificate[660]={
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-0x56,0x94,0xA9,0x55,
-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
-
-
-const unsigned char GeoTrust_Global_CA_certificate[856]={
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-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
-
-
-const unsigned char GeoTrust_Global_CA_2_certificate[874]={
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-0xE2,0x22,0x29,0xAE,0x7D,0x83,0x40,0xA8,0xBA,0x6C,
-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
-
-
-const unsigned char GeoTrust_Primary_Certification_Authority_certificate[896]={
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-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
-/* issuer :/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
-
-
-const unsigned char GeoTrust_Primary_Certification_Authority___G2_certificate[690]={
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-};
-
-
 /* subject:/C=US/O=GeoTrust Inc./OU=(c) 2008 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G3 */
 /* issuer :/C=US/O=GeoTrust Inc./OU=(c) 2008 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G3 */
 
@@ -2194,101 +1298,6 @@ const unsigned char GeoTrust_Primary_Certification_Authority___G3_certificate[10
 };
 
 
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
-
-
-const unsigned char GeoTrust_Universal_CA_certificate[1388]={
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-};
-
-
 /* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA 2 */
 /* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA 2 */
 
@@ -2384,67 +1393,67 @@ const unsigned char GeoTrust_Universal_CA_2_certificate[1392]={
 };
 
 
-/* subject:/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
-/* issuer :/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
+/* subject:/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
+/* issuer :/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
 
 
-const unsigned char GlobalSign_Root_CA_certificate[889]={
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-0xA8,0xFF,0xFC,0xFD,0x4B,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,
-0x01,0x05,0x05,0x00,0x03,0x82,0x01,0x01,0x00,0xD6,0x73,0xE7,0x7C,0x4F,0x76,0xD0,
-0x8D,0xBF,0xEC,0xBA,0xA2,0xBE,0x34,0xC5,0x28,0x32,0xB5,0x7C,0xFC,0x6C,0x9C,0x2C,
-0x2B,0xBD,0x09,0x9E,0x53,0xBF,0x6B,0x5E,0xAA,0x11,0x48,0xB6,0xE5,0x08,0xA3,0xB3,
-0xCA,0x3D,0x61,0x4D,0xD3,0x46,0x09,0xB3,0x3E,0xC3,0xA0,0xE3,0x63,0x55,0x1B,0xF2,
-0xBA,0xEF,0xAD,0x39,0xE1,0x43,0xB9,0x38,0xA3,0xE6,0x2F,0x8A,0x26,0x3B,0xEF,0xA0,
-0x50,0x56,0xF9,0xC6,0x0A,0xFD,0x38,0xCD,0xC4,0x0B,0x70,0x51,0x94,0x97,0x98,0x04,
-0xDF,0xC3,0x5F,0x94,0xD5,0x15,0xC9,0x14,0x41,0x9C,0xC4,0x5D,0x75,0x64,0x15,0x0D,
-0xFF,0x55,0x30,0xEC,0x86,0x8F,0xFF,0x0D,0xEF,0x2C,0xB9,0x63,0x46,0xF6,0xAA,0xFC,
-0xDF,0xBC,0x69,0xFD,0x2E,0x12,0x48,0x64,0x9A,0xE0,0x95,0xF0,0xA6,0xEF,0x29,0x8F,
-0x01,0xB1,0x15,0xB5,0x0C,0x1D,0xA5,0xFE,0x69,0x2C,0x69,0x24,0x78,0x1E,0xB3,0xA7,
-0x1C,0x71,0x62,0xEE,0xCA,0xC8,0x97,0xAC,0x17,0x5D,0x8A,0xC2,0xF8,0x47,0x86,0x6E,
-0x2A,0xC4,0x56,0x31,0x95,0xD0,0x67,0x89,0x85,0x2B,0xF9,0x6C,0xA6,0x5D,0x46,0x9D,
-0x0C,0xAA,0x82,0xE4,0x99,0x51,0xDD,0x70,0xB7,0xDB,0x56,0x3D,0x61,0xE4,0x6A,0xE1,
-0x5C,0xD6,0xF6,0xFE,0x3D,0xDE,0x41,0xCC,0x07,0xAE,0x63,0x52,0xBF,0x53,0x53,0xF4,
-0x2B,0xE9,0xC7,0xFD,0xB6,0xF7,0x82,0x5F,0x85,0xD2,0x41,0x18,0xDB,0x81,0xB3,0x04,
-0x1C,0xC5,0x1F,0xA4,0x80,0x6F,0x15,0x20,0xC9,0xDE,0x0C,0x88,0x0A,0x1D,0xD6,0x66,
-0x55,0xE2,0xFC,0x48,0xC9,0x29,0x26,0x69,0xE0,
+const unsigned char Baltimore_CyberTrust_Root_certificate[891]={
+0x30,0x82,0x03,0x77,0x30,0x82,0x02,0x5F,0xA0,0x03,0x02,0x01,0x02,0x02,0x04,0x02,
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+0x63,0xF5,0xA5,0x63,0xE0,0x90,0x0C,0x7D,0x5D,0xB2,0x06,0x7A,0xF3,0x85,0xEA,0xEB,
+0xD4,0x03,0xAE,0x5E,0x84,0x3E,0x5F,0xFF,0x15,0xED,0x69,0xBC,0xF9,0x39,0x36,0x72,
+0x75,0xCF,0x77,0x52,0x4D,0xF3,0xC9,0x90,0x2C,0xB9,0x3D,0xE5,0xC9,0x23,0x53,0x3F,
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+0xFF,0x04,0x04,0x03,0x02,0x01,0x06,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,
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+0x6F,0x51,0x68,0x42,0x05,0xA0,0xDD,0xBB,0x4F,0x27,0x25,0x84,0x03,0xBD,0xF7,0x64,
+0xFD,0x2D,0xD7,0x30,0xE3,0xA4,0x10,0x17,0xEB,0xDA,0x29,0x29,0xB6,0x79,0x3F,0x76,
+0xF6,0x19,0x13,0x23,0xB8,0x10,0x0A,0xF9,0x58,0xA4,0xD4,0x61,0x70,0xBD,0x04,0x61,
+0x6A,0x12,0x8A,0x17,0xD5,0x0A,0xBD,0xC5,0xBC,0x30,0x7C,0xD6,0xE9,0x0C,0x25,0x8D,
+0x86,0x40,0x4F,0xEC,0xCC,0xA3,0x7E,0x38,0xC6,0x37,0x11,0x4F,0xED,0xDD,0x68,0x31,
+0x8E,0x4C,0xD2,0xB3,0x01,0x74,0xEE,0xBE,0x75,0x5E,0x07,0x48,0x1A,0x7F,0x70,0xFF,
+0x16,0x5C,0x84,0xC0,0x79,0x85,0xB8,0x05,0xFD,0x7F,0xBE,0x65,0x11,0xA3,0x0F,0xC0,
+0x02,0xB4,0xF8,0x52,0x37,0x39,0x04,0xD5,0xA9,0x31,0x7A,0x18,0xBF,0xA0,0x2A,0xF4,
+0x12,0x99,0xF7,0xA3,0x45,0x82,0xE3,0x3C,0x5E,0xF5,0x9D,0x9E,0xB5,0xC8,0x9E,0x7C,
+0x2E,0xC8,0xA4,0x9E,0x4E,0x08,0x14,0x4B,0x6D,0xFD,0x70,0x6D,0x6B,0x1A,0x63,0xBD,
+0x64,0xE6,0x1F,0xB7,0xCE,0xF0,0xF2,0x9F,0x2E,0xBB,0x1B,0xB7,0xF2,0x50,0x88,0x73,
+0x92,0xC2,0xE2,0xE3,0x16,0x8D,0x9A,0x32,0x02,0xAB,0x8E,0x18,0xDD,0xE9,0x10,0x11,
+0xEE,0x7E,0x35,0xAB,0x90,0xAF,0x3E,0x30,0x94,0x7A,0xD0,0x33,0x3D,0xA7,0x65,0x0F,
+0xF5,0xFC,0x8E,0x9E,0x62,0xCF,0x47,0x44,0x2C,0x01,0x5D,0xBB,0x1D,0xB5,0x32,0xD2,
+0x47,0xD2,0x38,0x2E,0xD0,0xFE,0x81,0xDC,0x32,0x6A,0x1E,0xB5,0xEE,0x3C,0xD5,0xFC,
+0xE7,0x81,0x1D,0x19,0xC3,0x24,0x42,0xEA,0x63,0x39,0xA9,
 };
 
 
@@ -2579,1249 +1588,140 @@ const unsigned char GlobalSign_Root_CA___R3_certificate[867]={
 };
 
 
-/* subject:/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
-/* issuer :/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
 
 
-const unsigned char Go_Daddy_Class_2_CA_certificate[1028]={
-0x30,0x82,0x04,0x00,0x30,0x82,0x02,0xE8,0xA0,0x03,0x02,0x01,0x02,0x02,0x01,0x00,
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-0x63,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x21,
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-};
-
-
-/* subject:/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
-/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
-
-
-const unsigned char Go_Daddy_Root_Certificate_Authority___G2_certificate[969]={
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-};
-
-
-/* subject:/C=US/O=GTE Corporation/OU=GTE CyberTrust Solutions, Inc./CN=GTE CyberTrust Global Root */
-/* issuer :/C=US/O=GTE Corporation/OU=GTE CyberTrust Solutions, Inc./CN=GTE CyberTrust Global Root */
-
-
-const unsigned char GTE_CyberTrust_Global_Root_certificate[606]={
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-};
-
-
-/* subject:/C=US/O=Network Solutions L.L.C./CN=Network Solutions Certificate Authority */
-/* issuer :/C=US/O=Network Solutions L.L.C./CN=Network Solutions Certificate Authority */
-
-
-const unsigned char Network_Solutions_Certificate_Authority_certificate[1002]={
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-};
-
-
-/* subject:/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 3 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-/* issuer :/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 3 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-
-
-const unsigned char RSA_Root_Certificate_1_certificate[747]={
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-};
-
-
-/* subject:/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
-/* issuer :/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
-
-
-const unsigned char Starfield_Class_2_CA_certificate[1043]={
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-};
-
-
-/* subject:/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
-/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
-
-
-const unsigned char Starfield_Root_Certificate_Authority___G2_certificate[993]={
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-};
-
-
-/* subject:/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Services Root Certificate Authority - G2 */
-/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Services Root Certificate Authority - G2 */
-
-
-const unsigned char Starfield_Services_Root_Certificate_Authority___G2_certificate[1011]={
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-};
-
-
-/* subject:/C=IL/O=StartCom Ltd./OU=Secure Digital Certificate Signing/CN=StartCom Certification Authority */
-/* issuer :/C=IL/O=StartCom Ltd./OU=Secure Digital Certificate Signing/CN=StartCom Certification Authority */
-
-
-const unsigned char StartCom_Certification_Authority_certificate[1931]={
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-};
-
-
-/* subject:/C=IL/O=StartCom Ltd./CN=StartCom Certification Authority G2 */
-/* issuer :/C=IL/O=StartCom Ltd./CN=StartCom Certification Authority G2 */
-
-
-const unsigned char StartCom_Certification_Authority_G2_certificate[1383]={
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-};
-
-
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
-
-
-const unsigned char TC_TrustCenter_Class_2_CA_II_certificate[1198]={
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-
-
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 3 CA/CN=TC TrustCenter Class 3 CA II */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 3 CA/CN=TC TrustCenter Class 3 CA II */
-
-
-const unsigned char TC_TrustCenter_Class_3_CA_II_certificate[1198]={
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-};
-
-
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
-
-
-const unsigned char TC_TrustCenter_Universal_CA_I_certificate[993]={
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+const unsigned char AffirmTrust_Networking_certificate[848]={
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 };
 
 
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA III */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA III */
+/* subject:/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
+/* issuer :/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
 
 
-const unsigned char TC_TrustCenter_Universal_CA_III_certificate[997]={
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-};
-
-
-/* subject:/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Premium Server CA/emailAddress=premium-server@thawte.com */
-/* issuer :/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Premium Server CA/emailAddress=premium-server@thawte.com */
-
-
-const unsigned char Thawte_Premium_Server_CA_certificate[811]={
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-};
-
-
-/* subject:/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
-/* issuer :/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
-
-
-const unsigned char thawte_Primary_Root_CA_certificate[1060]={
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-};
-
-
-/* subject:/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
-/* issuer :/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
-
-
-const unsigned char thawte_Primary_Root_CA___G2_certificate[652]={
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+const unsigned char AddTrust_External_Root_certificate[1082]={
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 };
 
 
@@ -3900,570 +1800,208 @@ const unsigned char thawte_Primary_Root_CA___G3_certificate[1070]={
 };
 
 
-/* subject:/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Server CA/emailAddress=server-certs@thawte.com */
-/* issuer :/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Server CA/emailAddress=server-certs@thawte.com */
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
 
 
-const unsigned char Thawte_Server_CA_certificate[791]={
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+const unsigned char DigiCert_Assured_ID_Root_CA_certificate[955]={
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 };
 
 
-/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
-/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
+/* subject:/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
+/* issuer :/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
 
 
-const unsigned char UTN_DATACorp_SGC_Root_CA_certificate[1122]={
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-0xF4,0x08,
+const unsigned char Go_Daddy_Class_2_CA_certificate[1028]={
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+0x9E,0x0A,0xD5,0x95,0x62,0x9A,0x0D,0xCF,0x88,0x82,0xC5,0x32,0x0C,0xE4,0x2B,0x9F,
+0x45,0xE6,0x0D,0x9F,0x28,0x9C,0xB1,0xB9,0x2A,0x5A,0x57,0xAD,0x37,0x0F,0xAF,0x1D,
+0x7F,0xDB,0xBD,0x9F,
 };
 
 
-/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
-/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
 
 
-const unsigned char UTN_USERFirst_Hardware_Root_CA_certificate[1144]={
-0x30,0x82,0x04,0x74,0x30,0x82,0x03,0x5C,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x44,
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-};
-
-
-/* subject:/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 1 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-/* issuer :/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 1 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-
-
-const unsigned char ValiCert_Class_1_VA_certificate[747]={
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-};
-
-
-/* subject:/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 2 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-/* issuer :/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 2 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-
-
-const unsigned char ValiCert_Class_2_VA_certificate[747]={
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-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority */
-/* issuer :/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority */
-
-
-const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority_certificate[576]={
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-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority - G2/OU=(c) 1998 VeriSign, Inc. - For authorized use only/OU=VeriSign Trust Network */
-/* issuer :/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority - G2/OU=(c) 1998 VeriSign, Inc. - For authorized use only/OU=VeriSign Trust Network */
-
-
-const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority___G2_certificate[774]={
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-};
-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
-
-
-const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+const unsigned char GeoTrust_Primary_Certification_Authority_certificate[896]={
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-};
-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
-
-
-const unsigned char VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate[904]={
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 };
 
 
@@ -4553,380 +2091,2176 @@ const unsigned char VeriSign_Class_3_Public_Primary_Certification_Authority___G5
 };
 
 
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
+/* subject:/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
+/* issuer :/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
 
 
-const unsigned char Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+const unsigned char Equifax_Secure_CA_certificate[804]={
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+};
+
+
+/* subject:/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
+/* issuer :/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
+
+
+const unsigned char Entrust_net_Premium_2048_Secure_Server_CA_certificate[1120]={
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 };
 
 
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G3 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G3 */
 
 
-const unsigned char VeriSign_Universal_Root_Certification_Authority_certificate[1213]={
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-0x07,0x11,0xF0,0xD5,0xDB,0xDD,0xE5,0x8C,0xF0,0xD5,0x32,0xB0,0x83,0xE6,0x57,0xE2,
-0x8F,0xBF,0xBE,0xA1,0xAA,0xBF,0x3D,0x1D,0xB5,0xD4,0x38,0xEA,0xD7,0xB0,0x5C,0x3A,
-0x4F,0x6A,0x3F,0x8F,0xC0,0x66,0x6C,0x63,0xAA,0xE9,0xD9,0xA4,0x16,0xF4,0x81,0xD1,
-0x95,0x14,0x0E,0x7D,0xCD,0x95,0x34,0xD9,0xD2,0x8F,0x70,0x73,0x81,0x7B,0x9C,0x7E,
-0xBD,0x98,0x61,0xD8,0x45,0x87,0x98,0x90,0xC5,0xEB,0x86,0x30,0xC6,0x35,0xBF,0xF0,
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-0xEC,0xCD,0x82,0x61,0xF1,0x38,0xE6,0x4F,0x97,0x98,0x2A,0x5A,0x8D,
+const unsigned char DigiCert_Assured_ID_Root_G3_certificate[586]={
+0x30,0x82,0x02,0x46,0x30,0x82,0x01,0xCD,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x0B,
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+0xEA,0x96,0x63,0x6A,0x65,0x45,0x92,0x95,0x01,0xB4,
 };
 
 
-/* subject:/C=US/OU=www.xrampsecurity.com/O=XRamp Security Services Inc/CN=XRamp Global Certification Authority */
-/* issuer :/C=US/OU=www.xrampsecurity.com/O=XRamp Security Services Inc/CN=XRamp Global Certification Authority */
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
 
 
-const unsigned char XRamp_Global_CA_Root_certificate[1076]={
-0x30,0x82,0x04,0x30,0x30,0x82,0x03,0x18,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x50,
-0x94,0x6C,0xEC,0x18,0xEA,0xD5,0x9C,0x4D,0xD5,0x97,0xEF,0x75,0x8F,0xA0,0xAD,0x30,
+const unsigned char COMODO_Certification_Authority_certificate[1057]={
+0x30,0x82,0x04,0x1D,0x30,0x82,0x03,0x05,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x4E,
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+0xEC,0x2E,0x30,0x4E,0x94,0x35,0xB4,0x24,0x1F,0x4B,0x78,0x69,0xDA,0xF2,0x02,0x38,
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+0x04,0x0E,0xBE,0x0B,0x83,0xA6,0x17,0x83,0xB9,0x30,0x12,0xA0,0xC5,0x33,0x15,0x05,
+0xB9,0x0D,0xFB,0xC7,0x05,0x76,0xE3,0xD8,0x4A,0x8D,0xFC,0x34,0x17,0xA3,0xC6,0x21,
+0x28,0xBE,0x30,0x45,0x31,0x1E,0xC7,0x78,0xBE,0x58,0x61,0x38,0xAC,0x3B,0xE2,0x01,
+0x65,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
+
+
+const unsigned char DigiCert_Global_Root_CA_certificate[947]={
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+0x95,0x6D,0xDE,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
+
+
+const unsigned char Comodo_AAA_Services_root_certificate[1078]={
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+0xE3,0x2B,0x38,0x3A,0x09,0x81,0xB5,0x6B,0x85,0xD7,0xBE,0x2D,0xED,0x3F,0x1A,0xB7,
+0xB2,0x63,0xE2,0xF5,0x62,0x2C,0x82,0xD4,0x6A,0x00,0x41,0x50,0xF1,0x39,0x83,0x9F,
+0x95,0xE9,0x36,0x96,0x98,0x6E,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
+
+
+const unsigned char DigiCert_High_Assurance_EV_Root_CA_certificate[969]={
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+0xF7,0xEA,0x21,0xB9,0x68,0x12,0x9C,0x3C,0x9C,0xBF,0x9E,0xFC,0x80,0x5C,0x9B,0x63,
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+0xF8,0xE9,0x2E,0x13,0xA3,0x77,0xE8,0x1F,0x4A,
+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
+
+
+const unsigned char GeoTrust_Universal_CA_certificate[1388]={
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+0x6C,0x9C,0x4F,0x81,0x6A,0x7A,0x64,0xAC,0xBB,0x3E,0xD5,0xE7,0xCB,0x76,0x2E,0xC5,
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+0xA4,0xE6,0x8E,0xD8,0xF9,0x29,0x48,0x8A,0xCE,0x73,0xFE,0x2C,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
+
+
+const unsigned char COMODO_ECC_Certification_Authority_certificate[653]={
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+0x59,0x5B,0xA5,0x5B,0x0B,0x33,0x99,0xE2,0xC3,0x3D,0x89,0xF9,0x6A,0x2C,0xEF,0xB2,
+0xD3,0x06,0xE9,0xA3,0x42,0x30,0x40,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,
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+0xC4,0x48,0x77,0x99,0xD3,0x79,0x30,0x0E,0x06,0x03,0x55,0x1D,0x0F,0x01,0x01,0xFF,
+0x04,0x04,0x03,0x02,0x01,0x06,0x30,0x0F,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,
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+0xB7,0x78,0x0A,0x72,0xB7,0x88,0xDF,0xFF,0xB5,0x46,0x14,0x09,0x0A,0xFA,0xA0,0xE6,
+0x7D,0x08,0xC6,0x1A,0x87,0xBD,0x18,0xA8,0x73,0xBD,0x26,0xCA,0x60,0x0C,0x9D,0xCE,
+0x99,0x9F,0xCF,0x5C,0x0F,0x30,0xE1,0xBE,0x14,0x31,0xEA,0x02,0x30,0x14,0xF4,0x93,
+0x3C,0x49,0xA7,0x33,0x7A,0x90,0x46,0x47,0xB3,0x63,0x7D,0x13,0x9B,0x4E,0xB7,0x6F,
+0x18,0x37,0x80,0x53,0xFE,0xDD,0x20,0xE0,0x35,0x9A,0x36,0xD1,0xC7,0x01,0xB9,0xE6,
+0xDC,0xDD,0xF3,0xFF,0x1D,0x2C,0x3A,0x16,0x57,0xD9,0x92,0x39,0xD6,
+};
+
+
+/* subject:/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2009 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - G2 */
+/* issuer :/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2009 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - G2 */
+
+
+const unsigned char Entrust_Root_Certification_Authority___G2_certificate[1090]={
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+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G2 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G2 */
+
+
+const unsigned char DigiCert_Assured_ID_Root_G2_certificate[922]={
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+0x50,0xE3,0x03,0xDB,0x73,0x77,0x6F,0x4A,0x39,0x53,0xED,0xDE,0x26,0xC7,0xB5,0x7D,
+0xAF,0x2B,0x42,0xD1,0x75,0x62,0xE3,0x4A,0x2B,0x02,0xC7,0x50,0x4B,0xE0,0x69,0xE2,
+0x96,0x6C,0x0E,0x44,0x66,0x10,0x44,0x8F,0xAD,0x05,0xEB,0xF8,0x79,0xAC,0xA6,0x1B,
+0xE8,0x37,0x34,0x9D,0x53,0xC9,0x61,0xAA,0xA2,0x52,0xAF,0x4A,0x70,0x16,0x86,0xC2,
+0x3A,0xC8,0xB1,0x13,0x70,0x36,0xD8,0xCF,0xEE,0xF4,0x0A,0x34,0xD5,0x5B,0x4C,0xFD,
+0x07,0x9C,0xA2,0xBA,0xD9,0x01,0x72,0x5C,0xF3,0x4D,0xC1,0xDD,0x0E,0xB1,0x1C,0x0D,
+0xC4,0x63,0xBE,0xAD,0xF4,0x14,0xFB,0x89,0xEC,0xA2,0x41,0x0E,0x4C,0xCC,0xC8,0x57,
+0x40,0xD0,0x6E,0x03,0xAA,0xCD,0x0C,0x8E,0x89,0x99,0x99,0x6C,0xF0,0x3C,0x30,0xAF,
+0x38,0xDF,0x6F,0xBC,0xA3,0xBE,0x29,0x20,0x27,0xAB,0x74,0xFF,0x13,0x22,0x78,0xDE,
+0x97,0x52,0x55,0x1E,0x83,0xB5,0x54,0x20,0x03,0xEE,0xAE,0xC0,0x4F,0x56,0xDE,0x37,
+0xCC,0xC3,0x7F,0xAA,0x04,0x27,0xBB,0xD3,0x77,0xB8,0x62,0xDB,0x17,0x7C,0x9C,0x28,
+0x22,0x13,0x73,0x6C,0xCF,0x26,0xF5,0x8A,0x29,0xE7,
+};
+
+
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
+
+
+const unsigned char AffirmTrust_Commercial_certificate[848]={
+0x30,0x82,0x03,0x4C,0x30,0x82,0x02,0x34,0xA0,0x03,0x02,0x01,0x02,0x02,0x08,0x77,
+0x77,0x06,0x27,0x26,0xA9,0xB1,0x7C,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,
+0x0D,0x01,0x01,0x0B,0x05,0x00,0x30,0x44,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,
+0x06,0x13,0x02,0x55,0x53,0x31,0x14,0x30,0x12,0x06,0x03,0x55,0x04,0x0A,0x0C,0x0B,
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+0x30,0x31,0x32,0x33,0x31,0x31,0x34,0x30,0x36,0x30,0x36,0x5A,0x30,0x44,0x31,0x0B,
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+0xB7,0xAB,0x2C,0x4F,0xC6,0x41,0x32,0x94,0x57,0xFA,0x12,0xA7,0x5B,0x0E,0xE2,0x8F,
+0x1F,0x1E,0x86,0x19,0xA7,0xAA,0xB5,0x2D,0xB9,0x5F,0x0D,0x8A,0xC2,0xAF,0x85,0x35,
+0x79,0x32,0x2D,0xBB,0x1C,0x62,0x37,0xF2,0xB1,0x5B,0x4A,0x3D,0xCA,0xCD,0x71,0x5F,
+0xE9,0x42,0xBE,0x94,0xE8,0xC8,0xDE,0xF9,0x22,0x48,0x64,0xC6,0xE5,0xAB,0xC6,0x2B,
+0x6D,0xAD,0x05,0xF0,0xFA,0xD5,0x0B,0xCF,0x9A,0xE5,0xF0,0x50,0xA4,0x8B,0x3B,0x47,
+0xA5,0x23,0x5B,0x7A,0x7A,0xF8,0x33,0x3F,0xB8,0xEF,0x99,0x97,0xE3,0x20,0xC1,0xD6,
+0x28,0x89,0xCF,0x94,0xFB,0xB9,0x45,0xED,0xE3,0x40,0x17,0x11,0xD4,0x74,0xF0,0x0B,
+0x31,0xE2,0x2B,0x26,0x6A,0x9B,0x4C,0x57,0xAE,0xAC,0x20,0x3E,0xBA,0x45,0x7A,0x05,
+0xF3,0xBD,0x9B,0x69,0x15,0xAE,0x7D,0x4E,0x20,0x63,0xC4,0x35,0x76,0x3A,0x07,0x02,
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+0x35,0xAF,0x44,0xBA,0x54,0x1E,0x1C,0x30,0x32,0x65,0x9D,0xE6,0x3C,0x93,0x5E,0x50,
+0x4E,0x7A,0xE3,0x3A,0xD4,0x6E,0xCC,0x1A,0xFB,0xF9,0xD2,0x37,0xAE,0x24,0x2A,0xAB,
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+0x5D,0x43,0xC6,0x86,0x2E,0x3E,0x73,0xB2,0x26,0x7D,0x4F,0x93,0xA9,0xB6,0xC4,0x2A,
+0x9A,0xAB,0x21,0x97,0x14,0xB1,0xDE,0x8C,0xD3,0xAB,0x89,0x15,0xD8,0x6B,0x24,0xD4,
+0xF1,0x16,0xAE,0xD8,0xA4,0x5C,0xD4,0x7F,0x51,0x8E,0xED,0x18,0x01,0xB1,0x93,0x63,
+0xBD,0xBC,0xF8,0x61,0x80,0x9A,0x9E,0xB1,0xCE,0x42,0x70,0xE2,0xA9,0x7D,0x06,0x25,
+0x7D,0x27,0xA1,0xFE,0x6F,0xEC,0xB3,0x1E,0x24,0xDA,0xE3,0x4B,0x55,0x1A,0x00,0x3B,
+0x35,0xB4,0x3B,0xD9,0xD7,0x5D,0x30,0xFD,0x81,0x13,0x89,0xF2,0xC2,0x06,0x2B,0xED,
+0x67,0xC4,0x8E,0xC9,0x43,0xB2,0x5C,0x6B,0x15,0x89,0x02,0xBC,0x62,0xFC,0x4E,0xF2,
+0xB5,0x33,0xAA,0xB2,0x6F,0xD3,0x0A,0xA2,0x50,0xE3,0xF6,0x3B,0xE8,0x2E,0x44,0xC2,
+0xDB,0x66,0x38,0xA9,0x33,0x56,0x48,0xF1,0x6D,0x1B,0x33,0x8D,0x0D,0x8C,0x3F,0x60,
+0x37,0x9D,0xD3,0xCA,0x6D,0x7E,0x34,0x7E,0x0D,0x9F,0x72,0x76,0x8B,0x1B,0x9F,0x72,
+0xFD,0x52,0x35,0x41,0x45,0x02,0x96,0x2F,0x1C,0xB2,0x9A,0x73,0x49,0x21,0xB1,0x49,
+0x47,0x45,0x47,0xB4,0xEF,0x6A,0x34,0x11,0xC9,0x4D,0x9A,0xCC,0x59,0xB7,0xD6,0x02,
+0x9E,0x5A,0x4E,0x65,0xB5,0x94,0xAE,0x1B,0xDF,0x29,0xB0,0x16,0xF1,0xBF,0x00,0x9E,
+0x07,0x3A,0x17,0x64,0xB5,0x04,0xB5,0x23,0x21,0x99,0x0A,0x95,0x3B,0x97,0x7C,0xEF,
+};
+
+
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
+
+
+const unsigned char AffirmTrust_Premium_certificate[1354]={
+0x30,0x82,0x05,0x46,0x30,0x82,0x03,0x2E,0xA0,0x03,0x02,0x01,0x02,0x02,0x08,0x6D,
+0x8C,0x14,0x46,0xB1,0xA6,0x0A,0xEE,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,
+0x0D,0x01,0x01,0x0C,0x05,0x00,0x30,0x41,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,
+0x06,0x13,0x02,0x55,0x53,0x31,0x14,0x30,0x12,0x06,0x03,0x55,0x04,0x0A,0x0C,0x0B,
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+0x31,0x32,0x39,0x31,0x34,0x31,0x30,0x33,0x36,0x5A,0x17,0x0D,0x34,0x30,0x31,0x32,
+0x33,0x31,0x31,0x34,0x31,0x30,0x33,0x36,0x5A,0x30,0x41,0x31,0x0B,0x30,0x09,0x06,
+0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x14,0x30,0x12,0x06,0x03,0x55,0x04,
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+0xD2,0x76,0xE6,0xE1,0x8C,0xBD,0x3C,0xB3,0xD3,0x48,0x4B,0xE2,0xC8,0xF8,0x7F,0x92,
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+0x29,0xE0,0xB6,0xB8,0x09,0x68,0x19,0x1C,0x18,0x43,
+};
+
+
+/* subject:/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
+/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
+
+
+const unsigned char Go_Daddy_Root_Certificate_Authority___G2_certificate[969]={
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+0xE2,0xE2,0x44,0xBE,0x5C,0xF7,0xEA,0x1C,0xF5,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
+
+
+const unsigned char Comodo_Secure_Services_root_certificate[1091]={
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+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Trusted Root G4 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Trusted Root G4 */
+
+
+const unsigned char DigiCert_Trusted_Root_G4_certificate[1428]={
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+0x82,0x02,0x01,0x00,0xBB,0x61,0xD9,0x7D,0xA9,0x6C,0xBE,0x17,0xC4,0x91,0x1B,0xC3,
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+0x49,0x5D,0x44,0x63,0xE0,0xAD,0x9C,0x96,0x18,0x66,0x1B,0x23,0x0D,0x3D,0x79,0xE9,
+0x6D,0x6B,0xD6,0x54,0xF8,0xD2,0x3C,0xC1,0x43,0x40,0xAE,0x1D,0x50,0xF5,0x52,0xFC,
+0x90,0x3B,0xBB,0x98,0x99,0x69,0x6B,0xC7,0xC1,0xA7,0xA8,0x68,0xA4,0x27,0xDC,0x9D,
+0xF9,0x27,0xAE,0x30,0x85,0xB9,0xF6,0x67,0x4D,0x3A,0x3E,0x8F,0x59,0x39,0x22,0x53,
+0x44,0xEB,0xC8,0x5D,0x03,0xCA,0xED,0x50,0x7A,0x7D,0x62,0x21,0x0A,0x80,0xC8,0x73,
+0x66,0xD1,0xA0,0x05,0x60,0x5F,0xE8,0xA5,0xB4,0xA7,0xAF,0xA8,0xF7,0x6D,0x35,0x9C,
+0x7C,0x5A,0x8A,0xD6,0xA2,0x38,0x99,0xF3,0x78,0x8B,0xF4,0x4D,0xD2,0x20,0x0B,0xDE,
+0x04,0xEE,0x8C,0x9B,0x47,0x81,0x72,0x0D,0xC0,0x14,0x32,0xEF,0x30,0x59,0x2E,0xAE,
+0xE0,0x71,0xF2,0x56,0xE4,0x6A,0x97,0x6F,0x92,0x50,0x6D,0x96,0x8D,0x68,0x7A,0x9A,
+0xB2,0x36,0x14,0x7A,0x06,0xF2,0x24,0xB9,0x09,0x11,0x50,0xD7,0x08,0xB1,0xB8,0x89,
+0x7A,0x84,0x23,0x61,0x42,0x29,0xE5,0xA3,0xCD,0xA2,0x20,0x41,0xD7,0xD1,0x9C,0x64,
+0xD9,0xEA,0x26,0xA1,0x8B,0x14,0xD7,0x4C,0x19,0xB2,0x50,0x41,0x71,0x3D,0x3F,0x4D,
+0x70,0x23,0x86,0x0C,0x4A,0xDC,0x81,0xD2,0xCC,0x32,0x94,0x84,0x0D,0x08,0x09,0x97,
+0x1C,0x4F,0xC0,0xEE,0x6B,0x20,0x74,0x30,0xD2,0xE0,0x39,0x34,0x10,0x85,0x21,0x15,
+0x01,0x08,0xE8,0x55,0x32,0xDE,0x71,0x49,0xD9,0x28,0x17,0x50,0x4D,0xE6,0xBE,0x4D,
+0xD1,0x75,0xAC,0xD0,0xCA,0xFB,0x41,0xB8,0x43,0xA5,0xAA,0xD3,0xC3,0x05,0x44,0x4F,
+0x2C,0x36,0x9B,0xE2,0xFA,0xE2,0x45,0xB8,0x23,0x53,0x6C,0x06,0x6F,0x67,0x55,0x7F,
+0x46,0xB5,0x4C,0x3F,0x6E,0x28,0x5A,0x79,0x26,0xD2,0xA4,0xA8,0x62,0x97,0xD2,0x1E,
+0xE2,0xED,0x4A,0x8B,0xBC,0x1B,0xFD,0x47,0x4A,0x0D,0xDF,0x67,0x66,0x7E,0xB2,0x5B,
+0x41,0xD0,0x3B,0xE4,0xF4,0x3B,0xF4,0x04,0x63,0xE9,0xEF,0xC2,0x54,0x00,0x51,0xA0,
+0x8A,0x2A,0xC9,0xCE,0x78,0xCC,0xD5,0xEA,0x87,0x04,0x18,0xB3,0xCE,0xAF,0x49,0x88,
+0xAF,0xF3,0x92,0x99,0xB6,0xB3,0xE6,0x61,0x0F,0xD2,0x85,0x00,0xE7,0x50,0x1A,0xE4,
+0x1B,0x95,0x9D,0x19,0xA1,0xB9,0x9C,0xB1,0x9B,0xB1,0x00,0x1E,0xEF,0xD0,0x0F,0x4F,
+0x42,0x6C,0xC9,0x0A,0xBC,0xEE,0x43,0xFA,0x3A,0x71,0xA5,0xC8,0x4D,0x26,0xA5,0x35,
+0xFD,0x89,0x5D,0xBC,0x85,0x62,0x1D,0x32,0xD2,0xA0,0x2B,0x54,0xED,0x9A,0x57,0xC1,
+0xDB,0xFA,0x10,0xCF,0x19,0xB7,0x8B,0x4A,0x1B,0x8F,0x01,0xB6,0x27,0x95,0x53,0xE8,
+0xB6,0x89,0x6D,0x5B,0xBC,0x68,0xD4,0x23,0xE8,0x8B,0x51,0xA2,0x56,0xF9,0xF0,0xA6,
+0x80,0xA0,0xD6,0x1E,0xB3,0xBC,0x0F,0x0F,0x53,0x75,0x29,0xAA,0xEA,0x13,0x77,0xE4,
+0xDE,0x8C,0x81,0x21,0xAD,0x07,0x10,0x47,0x11,0xAD,0x87,0x3D,0x07,0xD1,0x75,0xBC,
+0xCF,0xF3,0x66,0x7E,
+};
+
+
+/* subject:/OU=GlobalSign ECC Root CA - R5/O=GlobalSign/CN=GlobalSign */
+/* issuer :/OU=GlobalSign ECC Root CA - R5/O=GlobalSign/CN=GlobalSign */
+
+
+const unsigned char GlobalSign_ECC_Root_CA___R5_certificate[546]={
+0x30,0x82,0x02,0x1E,0x30,0x82,0x01,0xA4,0xA0,0x03,0x02,0x01,0x02,0x02,0x11,0x60,
+0x59,0x49,0xE0,0x26,0x2E,0xBB,0x55,0xF9,0x0A,0x77,0x8A,0x71,0xF9,0x4A,0xD8,0x6C,
+0x30,0x0A,0x06,0x08,0x2A,0x86,0x48,0xCE,0x3D,0x04,0x03,0x03,0x30,0x50,0x31,0x24,
+0x30,0x22,0x06,0x03,0x55,0x04,0x0B,0x13,0x1B,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,
+0x69,0x67,0x6E,0x20,0x45,0x43,0x43,0x20,0x52,0x6F,0x6F,0x74,0x20,0x43,0x41,0x20,
+0x2D,0x20,0x52,0x35,0x31,0x13,0x30,0x11,0x06,0x03,0x55,0x04,0x0A,0x13,0x0A,0x47,
+0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x31,0x13,0x30,0x11,0x06,0x03,0x55,
+0x04,0x03,0x13,0x0A,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x30,0x1E,
+0x17,0x0D,0x31,0x32,0x31,0x31,0x31,0x33,0x30,0x30,0x30,0x30,0x30,0x30,0x5A,0x17,
+0x0D,0x33,0x38,0x30,0x31,0x31,0x39,0x30,0x33,0x31,0x34,0x30,0x37,0x5A,0x30,0x50,
+0x31,0x24,0x30,0x22,0x06,0x03,0x55,0x04,0x0B,0x13,0x1B,0x47,0x6C,0x6F,0x62,0x61,
+0x6C,0x53,0x69,0x67,0x6E,0x20,0x45,0x43,0x43,0x20,0x52,0x6F,0x6F,0x74,0x20,0x43,
+0x41,0x20,0x2D,0x20,0x52,0x35,0x31,0x13,0x30,0x11,0x06,0x03,0x55,0x04,0x0A,0x13,
+0x0A,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x31,0x13,0x30,0x11,0x06,
+0x03,0x55,0x04,0x03,0x13,0x0A,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,
+0x30,0x76,0x30,0x10,0x06,0x07,0x2A,0x86,0x48,0xCE,0x3D,0x02,0x01,0x06,0x05,0x2B,
+0x81,0x04,0x00,0x22,0x03,0x62,0x00,0x04,0x47,0x45,0x0E,0x96,0xFB,0x7D,0x5D,0xBF,
+0xE9,0x39,0xD1,0x21,0xF8,0x9F,0x0B,0xB6,0xD5,0x7B,0x1E,0x92,0x3A,0x48,0x59,0x1C,
+0xF0,0x62,0x31,0x2D,0xC0,0x7A,0x28,0xFE,0x1A,0xA7,0x5C,0xB3,0xB6,0xCC,0x97,0xE7,
+0x45,0xD4,0x58,0xFA,0xD1,0x77,0x6D,0x43,0xA2,0xC0,0x87,0x65,0x34,0x0A,0x1F,0x7A,
+0xDD,0xEB,0x3C,0x33,0xA1,0xC5,0x9D,0x4D,0xA4,0x6F,0x41,0x95,0x38,0x7F,0xC9,0x1E,
+0x84,0xEB,0xD1,0x9E,0x49,0x92,0x87,0x94,0x87,0x0C,0x3A,0x85,0x4A,0x66,0x9F,0x9D,
+0x59,0x93,0x4D,0x97,0x61,0x06,0x86,0x4A,0xA3,0x42,0x30,0x40,0x30,0x0E,0x06,0x03,
+0x55,0x1D,0x0F,0x01,0x01,0xFF,0x04,0x04,0x03,0x02,0x01,0x06,0x30,0x0F,0x06,0x03,
+0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,0x05,0x30,0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,
+0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,0x14,0x3D,0xE6,0x29,0x48,0x9B,0xEA,0x07,0xCA,
+0x21,0x44,0x4A,0x26,0xDE,0x6E,0xDE,0xD2,0x83,0xD0,0x9F,0x59,0x30,0x0A,0x06,0x08,
+0x2A,0x86,0x48,0xCE,0x3D,0x04,0x03,0x03,0x03,0x68,0x00,0x30,0x65,0x02,0x31,0x00,
+0xE5,0x69,0x12,0xC9,0x6E,0xDB,0xC6,0x31,0xBA,0x09,0x41,0xE1,0x97,0xF8,0xFB,0xFD,
+0x9A,0xE2,0x7D,0x12,0xC9,0xED,0x7C,0x64,0xD3,0xCB,0x05,0x25,0x8B,0x56,0xD9,0xA0,
+0xE7,0x5E,0x5D,0x4E,0x0B,0x83,0x9C,0x5B,0x76,0x29,0xA0,0x09,0x26,0x21,0x6A,0x62,
+0x02,0x30,0x71,0xD2,0xB5,0x8F,0x5C,0xEA,0x3B,0xE1,0x78,0x09,0x85,0xA8,0x75,0x92,
+0x3B,0xC8,0x5C,0xFD,0x48,0xEF,0x0D,0x74,0x22,0xA8,0x08,0xE2,0x6E,0xC5,0x49,0xCE,
+0xC7,0x0C,0xBC,0xA7,0x61,0x69,0xF1,0xF7,0x3B,0xE1,0x2A,0xCB,0xF9,0x2B,0xF3,0x66,
+0x90,0x37,
+};
+
+
+/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
+/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
+
+
+const unsigned char UTN_USERFirst_Hardware_Root_CA_certificate[1144]={
+0x30,0x82,0x04,0x74,0x30,0x82,0x03,0x5C,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x44,
+0xBE,0x0C,0x8B,0x50,0x00,0x24,0xB4,0x11,0xD3,0x36,0x2A,0xFE,0x65,0x0A,0xFD,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,0x81,
+0x97,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x0B,
+0x30,0x09,0x06,0x03,0x55,0x04,0x08,0x13,0x02,0x55,0x54,0x31,0x17,0x30,0x15,0x06,
+0x03,0x55,0x04,0x07,0x13,0x0E,0x53,0x61,0x6C,0x74,0x20,0x4C,0x61,0x6B,0x65,0x20,
+0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,0x55,0x04,0x0A,0x13,0x15,0x54,
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+0x32,0x9C,0x1E,0xBB,0x9D,0xF8,0x66,0xA8,
+};
+
+
+/* subject:/OU=GlobalSign ECC Root CA - R4/O=GlobalSign/CN=GlobalSign */
+/* issuer :/OU=GlobalSign ECC Root CA - R4/O=GlobalSign/CN=GlobalSign */
+
+
+const unsigned char GlobalSign_ECC_Root_CA___R4_certificate[485]={
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+0x7B,0x0B,0xF8,0x9F,0x84,
+};
+
+
+/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
+/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
+
+
+const unsigned char TC_TrustCenter_Universal_CA_I_certificate[993]={
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+0x98,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
+
+
+const unsigned char Comodo_Trusted_Services_root_certificate[1095]={
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+};
+
+
+/* subject:/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
+/* issuer :/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
+
+
+const unsigned char Entrust_Root_Certification_Authority_certificate[1173]={
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+};
+
+
+/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
+/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
+
+
+const unsigned char TC_TrustCenter_Class_2_CA_II_certificate[1198]={
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+};
+
+
+/* subject:/O=Cybertrust, Inc/CN=Cybertrust Global Root */
+/* issuer :/O=Cybertrust, Inc/CN=Cybertrust Global Root */
+
+
+const unsigned char Cybertrust_Global_Root_certificate[933]={
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+0xA6,0x88,0x38,0xCE,0x55,
+};
+
+
+/* subject:/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2012 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - EC1 */
+/* issuer :/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2012 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - EC1 */
+
+
+const unsigned char Entrust_Root_Certification_Authority___EC1_certificate[765]={
+0x30,0x82,0x02,0xF9,0x30,0x82,0x02,0x80,0xA0,0x03,0x02,0x01,0x02,0x02,0x0D,0x00,
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+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
+/* issuer :/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
+
+
+const unsigned char GeoTrust_Primary_Certification_Authority___G2_certificate[690]={
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+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
+
+
+const unsigned char GeoTrust_Global_CA_2_certificate[874]={
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+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO RSA Certification Authority */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO RSA Certification Authority */
+
+
+const unsigned char COMODO_RSA_Certification_Authority_certificate[1500]={
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+0x7B,0x7F,0x86,0xBE,0x1E,0x0E,0x32,0x46,0xFB,0xFB,0x13,0x8F,0x75,0x7F,0x4C,0x8B,
+0x4B,0x46,0x63,0xFE,0x00,0x34,0x40,0x70,0xC1,0xC3,0xB9,0xA1,0xDD,0xA6,0x70,0xE2,
+0x04,0xB3,0x41,0xBC,0xE9,0x80,0x91,0xEA,0x64,0x9C,0x7A,0xE1,0x22,0x03,0xA9,0x9C,
+0x6E,0x6F,0x0E,0x65,0x4F,0x6C,0x87,0x87,0x5E,0xF3,0x6E,0xA0,0xF9,0x75,0xA5,0x9B,
+0x40,0xE8,0x53,0xB2,0x27,0x9D,0x4A,0xB9,0xC0,0x77,0x21,0x8D,0xFF,0x87,0xF2,0xDE,
+0xBC,0x8C,0xEF,0x17,0xDF,0xB7,0x49,0x0B,0xD1,0xF2,0x6E,0x30,0x0B,0x1A,0x0E,0x4E,
+0x76,0xED,0x11,0xFC,0xF5,0xE9,0x56,0xB2,0x7D,0xBF,0xC7,0x6D,0x0A,0x93,0x8C,0xA5,
+0xD0,0xC0,0xB6,0x1D,0xBE,0x3A,0x4E,0x94,0xA2,0xD7,0x6E,0x6C,0x0B,0xC2,0x8A,0x7C,
+0xFA,0x20,0xF3,0xC4,0xE4,0xE5,0xCD,0x0D,0xA8,0xCB,0x91,0x92,0xB1,0x7C,0x85,0xEC,
+0xB5,0x14,0x69,0x66,0x0E,0x82,0xE7,0xCD,0xCE,0xC8,0x2D,0xA6,0x51,0x7F,0x21,0xC1,
+0x35,0x53,0x85,0x06,0x4A,0x5D,0x9F,0xAD,0xBB,0x1B,0x5F,0x74,
+};
+
+
+/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
+/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
+
+
+const unsigned char UTN_DATACorp_SGC_Root_CA_certificate[1122]={
+0x30,0x82,0x04,0x5E,0x30,0x82,0x03,0x46,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x44,
+0xBE,0x0C,0x8B,0x50,0x00,0x21,0xB4,0x11,0xD3,0x2A,0x68,0x06,0xA9,0xAD,0x69,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,0x81,
+0x93,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x0B,
+0x30,0x09,0x06,0x03,0x55,0x04,0x08,0x13,0x02,0x55,0x54,0x31,0x17,0x30,0x15,0x06,
+0x03,0x55,0x04,0x07,0x13,0x0E,0x53,0x61,0x6C,0x74,0x20,0x4C,0x61,0x6B,0x65,0x20,
+0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,0x55,0x04,0x0A,0x13,0x15,0x54,
+0x68,0x65,0x20,0x55,0x53,0x45,0x52,0x54,0x52,0x55,0x53,0x54,0x20,0x4E,0x65,0x74,
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+0x35,0x37,0x32,0x31,0x5A,0x17,0x0D,0x31,0x39,0x30,0x36,0x32,0x34,0x31,0x39,0x30,
+0x36,0x33,0x30,0x5A,0x30,0x81,0x93,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,
+0x13,0x02,0x55,0x53,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x08,0x13,0x02,0x55,
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+0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x01,0x05,0x00,0x03,0x82,0x01,
+0x0F,0x00,0x30,0x82,0x01,0x0A,0x02,0x82,0x01,0x01,0x00,0xDF,0xEE,0x58,0x10,0xA2,
+0x2B,0x6E,0x55,0xC4,0x8E,0xBF,0x2E,0x46,0x09,0xE7,0xE0,0x08,0x0F,0x2E,0x2B,0x7A,
+0x13,0x94,0x1B,0xBD,0xF6,0xB6,0x80,0x8E,0x65,0x05,0x93,0x00,0x1E,0xBC,0xAF,0xE2,
+0x0F,0x8E,0x19,0x0D,0x12,0x47,0xEC,0xAC,0xAD,0xA3,0xFA,0x2E,0x70,0xF8,0xDE,0x6E,
+0xFB,0x56,0x42,0x15,0x9E,0x2E,0x5C,0xEF,0x23,0xDE,0x21,0xB9,0x05,0x76,0x27,0x19,
+0x0F,0x4F,0xD6,0xC3,0x9C,0xB4,0xBE,0x94,0x19,0x63,0xF2,0xA6,0x11,0x0A,0xEB,0x53,
+0x48,0x9C,0xBE,0xF2,0x29,0x3B,0x16,0xE8,0x1A,0xA0,0x4C,0xA6,0xC9,0xF4,0x18,0x59,
+0x68,0xC0,0x70,0xF2,0x53,0x00,0xC0,0x5E,0x50,0x82,0xA5,0x56,0x6F,0x36,0xF9,0x4A,
+0xE0,0x44,0x86,0xA0,0x4D,0x4E,0xD6,0x47,0x6E,0x49,0x4A,0xCB,0x67,0xD7,0xA6,0xC4,
+0x05,0xB9,0x8E,0x1E,0xF4,0xFC,0xFF,0xCD,0xE7,0x36,0xE0,0x9C,0x05,0x6C,0xB2,0x33,
+0x22,0x15,0xD0,0xB4,0xE0,0xCC,0x17,0xC0,0xB2,0xC0,0xF4,0xFE,0x32,0x3F,0x29,0x2A,
+0x95,0x7B,0xD8,0xF2,0xA7,0x4E,0x0F,0x54,0x7C,0xA1,0x0D,0x80,0xB3,0x09,0x03,0xC1,
+0xFF,0x5C,0xDD,0x5E,0x9A,0x3E,0xBC,0xAE,0xBC,0x47,0x8A,0x6A,0xAE,0x71,0xCA,0x1F,
+0xB1,0x2A,0xB8,0x5F,0x42,0x05,0x0B,0xEC,0x46,0x30,0xD1,0x72,0x0B,0xCA,0xE9,0x56,
+0x6D,0xF5,0xEF,0xDF,0x78,0xBE,0x61,0xBA,0xB2,0xA5,0xAE,0x04,0x4C,0xBC,0xA8,0xAC,
+0x69,0x15,0x97,0xBD,0xEF,0xEB,0xB4,0x8C,0xBF,0x35,0xF8,0xD4,0xC3,0xD1,0x28,0x0E,
+0x5C,0x3A,0x9F,0x70,0x18,0x33,0x20,0x77,0xC4,0xA2,0xAF,0x02,0x03,0x01,0x00,0x01,
+0xA3,0x81,0xAB,0x30,0x81,0xA8,0x30,0x0B,0x06,0x03,0x55,0x1D,0x0F,0x04,0x04,0x03,
+0x02,0x01,0xC6,0x30,0x0F,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,0x05,0x30,
+0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,0x14,0x53,
+0x32,0xD1,0xB3,0xCF,0x7F,0xFA,0xE0,0xF1,0xA0,0x5D,0x85,0x4E,0x92,0xD2,0x9E,0x45,
+0x1D,0xB4,0x4F,0x30,0x3D,0x06,0x03,0x55,0x1D,0x1F,0x04,0x36,0x30,0x34,0x30,0x32,
+0xA0,0x30,0xA0,0x2E,0x86,0x2C,0x68,0x74,0x74,0x70,0x3A,0x2F,0x2F,0x63,0x72,0x6C,
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+0x54,0x4E,0x2D,0x44,0x41,0x54,0x41,0x43,0x6F,0x72,0x70,0x53,0x47,0x43,0x2E,0x63,
+0x72,0x6C,0x30,0x2A,0x06,0x03,0x55,0x1D,0x25,0x04,0x23,0x30,0x21,0x06,0x08,0x2B,
+0x06,0x01,0x05,0x05,0x07,0x03,0x01,0x06,0x0A,0x2B,0x06,0x01,0x04,0x01,0x82,0x37,
+0x0A,0x03,0x03,0x06,0x09,0x60,0x86,0x48,0x01,0x86,0xF8,0x42,0x04,0x01,0x30,0x0D,
+0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x03,0x82,0x01,
+0x01,0x00,0x27,0x35,0x97,0x00,0x8A,0x8B,0x28,0xBD,0xC6,0x33,0x30,0x1E,0x29,0xFC,
+0xE2,0xF7,0xD5,0x98,0xD4,0x40,0xBB,0x60,0xCA,0xBF,0xAB,0x17,0x2C,0x09,0x36,0x7F,
+0x50,0xFA,0x41,0xDC,0xAE,0x96,0x3A,0x0A,0x23,0x3E,0x89,0x59,0xC9,0xA3,0x07,0xED,
+0x1B,0x37,0xAD,0xFC,0x7C,0xBE,0x51,0x49,0x5A,0xDE,0x3A,0x0A,0x54,0x08,0x16,0x45,
+0xC2,0x99,0xB1,0x87,0xCD,0x8C,0x68,0xE0,0x69,0x03,0xE9,0xC4,0x4E,0x98,0xB2,0x3B,
+0x8C,0x16,0xB3,0x0E,0xA0,0x0C,0x98,0x50,0x9B,0x93,0xA9,0x70,0x09,0xC8,0x2C,0xA3,
+0x8F,0xDF,0x02,0xE4,0xE0,0x71,0x3A,0xF1,0xB4,0x23,0x72,0xA0,0xAA,0x01,0xDF,0xDF,
+0x98,0x3E,0x14,0x50,0xA0,0x31,0x26,0xBD,0x28,0xE9,0x5A,0x30,0x26,0x75,0xF9,0x7B,
+0x60,0x1C,0x8D,0xF3,0xCD,0x50,0x26,0x6D,0x04,0x27,0x9A,0xDF,0xD5,0x0D,0x45,0x47,
+0x29,0x6B,0x2C,0xE6,0x76,0xD9,0xA9,0x29,0x7D,0x32,0xDD,0xC9,0x36,0x3C,0xBD,0xAE,
+0x35,0xF1,0x11,0x9E,0x1D,0xBB,0x90,0x3F,0x12,0x47,0x4E,0x8E,0xD7,0x7E,0x0F,0x62,
+0x73,0x1D,0x52,0x26,0x38,0x1C,0x18,0x49,0xFD,0x30,0x74,0x9A,0xC4,0xE5,0x22,0x2F,
+0xD8,0xC0,0x8D,0xED,0x91,0x7A,0x4C,0x00,0x8F,0x72,0x7F,0x5D,0xDA,0xDD,0x1B,0x8B,
+0x45,0x6B,0xE7,0xDD,0x69,0x97,0xA8,0xC5,0x56,0x4C,0x0F,0x0C,0xF6,0x9F,0x7A,0x91,
+0x37,0xF6,0x97,0x82,0xE0,0xDD,0x71,0x69,0xFF,0x76,0x3F,0x60,0x4D,0x3C,0xCF,0xF7,
+0x99,0xF9,0xC6,0x57,0xF4,0xC9,0x55,0x39,0x78,0xBA,0x2C,0x79,0xC9,0xA6,0x88,0x2B,
+0xF4,0x08,
 };
 
 
 const unsigned char* kSSLCertCertificateList[] = {
- AddTrust_External_Root_certificate,
+ GlobalSign_Root_CA_certificate,
+ USERTrust_RSA_Certification_Authority_certificate,
+ Starfield_Class_2_CA_certificate,
+ Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate,
+ USERTrust_ECC_Certification_Authority_certificate,
+ GeoTrust_Global_CA_certificate,
+ Starfield_Root_Certificate_Authority___G2_certificate,
+ DigiCert_Global_Root_G3_certificate,
+ thawte_Primary_Root_CA___G2_certificate,
+ VeriSign_Universal_Root_Certification_Authority_certificate,
+ VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate,
+ DigiCert_Global_Root_G2_certificate,
  AddTrust_Low_Value_Services_Root_certificate,
+ AffirmTrust_Premium_ECC_certificate,
+ Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate,
+ thawte_Primary_Root_CA_certificate,
  AddTrust_Public_Services_Root_certificate,
  AddTrust_Qualified_Certificates_Root_certificate,
- AffirmTrust_Commercial_certificate,
- AffirmTrust_Networking_certificate,
- AffirmTrust_Premium_certificate,
- AffirmTrust_Premium_ECC_certificate,
- America_Online_Root_Certification_Authority_1_certificate,
- America_Online_Root_Certification_Authority_2_certificate,
- Baltimore_CyberTrust_Root_certificate,
- Comodo_AAA_Services_root_certificate,
- COMODO_Certification_Authority_certificate,
- COMODO_ECC_Certification_Authority_certificate,
- Comodo_Secure_Services_root_certificate,
- Comodo_Trusted_Services_root_certificate,
- Cybertrust_Global_Root_certificate,
- DigiCert_Assured_ID_Root_CA_certificate,
- DigiCert_Global_Root_CA_certificate,
- DigiCert_High_Assurance_EV_Root_CA_certificate,
- Entrust_net_Premium_2048_Secure_Server_CA_certificate,
- Entrust_net_Secure_Server_CA_certificate,
- Entrust_Root_Certification_Authority_certificate,
- Equifax_Secure_CA_certificate,
- Equifax_Secure_eBusiness_CA_1_certificate,
- Equifax_Secure_eBusiness_CA_2_certificate,
- Equifax_Secure_Global_eBusiness_CA_certificate,
- GeoTrust_Global_CA_certificate,
- GeoTrust_Global_CA_2_certificate,
- GeoTrust_Primary_Certification_Authority_certificate,
- GeoTrust_Primary_Certification_Authority___G2_certificate,
  GeoTrust_Primary_Certification_Authority___G3_certificate,
- GeoTrust_Universal_CA_certificate,
  GeoTrust_Universal_CA_2_certificate,
- GlobalSign_Root_CA_certificate,
+ Baltimore_CyberTrust_Root_certificate,
  GlobalSign_Root_CA___R2_certificate,
  GlobalSign_Root_CA___R3_certificate,
- Go_Daddy_Class_2_CA_certificate,
- Go_Daddy_Root_Certificate_Authority___G2_certificate,
- GTE_CyberTrust_Global_Root_certificate,
- Network_Solutions_Certificate_Authority_certificate,
- RSA_Root_Certificate_1_certificate,
- Starfield_Class_2_CA_certificate,
- Starfield_Root_Certificate_Authority___G2_certificate,
- Starfield_Services_Root_Certificate_Authority___G2_certificate,
- StartCom_Certification_Authority_certificate,
- StartCom_Certification_Authority_G2_certificate,
- TC_TrustCenter_Class_2_CA_II_certificate,
- TC_TrustCenter_Class_3_CA_II_certificate,
- TC_TrustCenter_Universal_CA_I_certificate,
- TC_TrustCenter_Universal_CA_III_certificate,
- Thawte_Premium_Server_CA_certificate,
- thawte_Primary_Root_CA_certificate,
- thawte_Primary_Root_CA___G2_certificate,
+ AffirmTrust_Networking_certificate,
+ AddTrust_External_Root_certificate,
  thawte_Primary_Root_CA___G3_certificate,
- Thawte_Server_CA_certificate,
- UTN_DATACorp_SGC_Root_CA_certificate,
- UTN_USERFirst_Hardware_Root_CA_certificate,
- ValiCert_Class_1_VA_certificate,
- ValiCert_Class_2_VA_certificate,
- Verisign_Class_3_Public_Primary_Certification_Authority_certificate,
- Verisign_Class_3_Public_Primary_Certification_Authority___G2_certificate,
- Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate,
- VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate,
+ DigiCert_Assured_ID_Root_CA_certificate,
+ Go_Daddy_Class_2_CA_certificate,
+ GeoTrust_Primary_Certification_Authority_certificate,
  VeriSign_Class_3_Public_Primary_Certification_Authority___G5_certificate,
- Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate,
- VeriSign_Universal_Root_Certification_Authority_certificate,
- XRamp_Global_CA_Root_certificate,
+ Equifax_Secure_CA_certificate,
+ Entrust_net_Premium_2048_Secure_Server_CA_certificate,
+ DigiCert_Assured_ID_Root_G3_certificate,
+ COMODO_Certification_Authority_certificate,
+ DigiCert_Global_Root_CA_certificate,
+ Comodo_AAA_Services_root_certificate,
+ DigiCert_High_Assurance_EV_Root_CA_certificate,
+ GeoTrust_Universal_CA_certificate,
+ COMODO_ECC_Certification_Authority_certificate,
+ Entrust_Root_Certification_Authority___G2_certificate,
+ DigiCert_Assured_ID_Root_G2_certificate,
+ AffirmTrust_Commercial_certificate,
+ AffirmTrust_Premium_certificate,
+ Go_Daddy_Root_Certificate_Authority___G2_certificate,
+ Comodo_Secure_Services_root_certificate,
+ DigiCert_Trusted_Root_G4_certificate,
+ GlobalSign_ECC_Root_CA___R5_certificate,
+ UTN_USERFirst_Hardware_Root_CA_certificate,
+ GlobalSign_ECC_Root_CA___R4_certificate,
+ TC_TrustCenter_Universal_CA_I_certificate,
+ Comodo_Trusted_Services_root_certificate,
+ Entrust_Root_Certification_Authority_certificate,
+ TC_TrustCenter_Class_2_CA_II_certificate,
+ Cybertrust_Global_Root_certificate,
+ Entrust_Root_Certification_Authority___EC1_certificate,
+ GeoTrust_Primary_Certification_Authority___G2_certificate,
+ GeoTrust_Global_CA_2_certificate,
+ COMODO_RSA_Certification_Authority_certificate,
+ UTN_DATACorp_SGC_Root_CA_certificate,
 };
 
 const size_t kSSLCertCertificateSizeList[] = {
-  1082,
+  889,
+  1506,
+  1043,
+  1054,
+  659,
+  856,
+  993,
+  579,
+  652,
+  1213,
+  904,
+  914,
   1052,
+  514,
+  1054,
+  1060,
   1049,
   1058,
-  848,
-  848,
-  1354,
-  514,
-  936,
-  1448,
-  891,
-  1078,
-  1057,
-  653,
-  1091,
-  1095,
-  933,
-  955,
-  947,
-  969,
-  1120,
-  1244,
-  1173,
-  804,
-  646,
-  804,
-  660,
-  856,
-  874,
-  896,
-  690,
   1026,
-  1388,
   1392,
-  889,
+  891,
   958,
   867,
-  1028,
-  969,
-  606,
-  1002,
-  747,
-  1043,
-  993,
-  1011,
-  1931,
-  1383,
-  1198,
-  1198,
-  993,
-  997,
-  811,
-  1060,
-  652,
+  848,
+  1082,
   1070,
-  791,
-  1122,
-  1144,
-  747,
-  747,
-  576,
-  774,
-  1054,
-  904,
+  955,
+  1028,
+  896,
   1239,
-  1054,
-  1213,
-  1076,
+  804,
+  1120,
+  586,
+  1057,
+  947,
+  1078,
+  969,
+  1388,
+  653,
+  1090,
+  922,
+  848,
+  1354,
+  969,
+  1091,
+  1428,
+  546,
+  1144,
+  485,
+  993,
+  1095,
+  1173,
+  1198,
+  933,
+  765,
+  690,
+  874,
+  1500,
+  1122,
 };
 
-}  // namspace rtc
diff --git a/media/webrtc/trunk/webrtc/base/sslstreamadapter.cc b/media/webrtc/trunk/webrtc/base/sslstreamadapter.cc
index a5922831a5..a2cff3e448 100644
--- a/media/webrtc/trunk/webrtc/base/sslstreamadapter.cc
+++ b/media/webrtc/trunk/webrtc/base/sslstreamadapter.cc
@@ -15,67 +15,68 @@
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/sslconfig.h"
 
-#if SSL_USE_SCHANNEL
-
-// SChannel support for DTLS and peer-to-peer mode are not
-// done.
-#elif SSL_USE_OPENSSL  // && !SSL_USE_SCHANNEL
+#if SSL_USE_OPENSSL
 
 #include "webrtc/base/opensslstreamadapter.h"
 
-#elif SSL_USE_NSS      // && !SSL_USE_SCHANNEL && !SSL_USE_OPENSSL
-
-#include "webrtc/base/nssstreamadapter.h"
-
-#endif  // !SSL_USE_OPENSSL && !SSL_USE_SCHANNEL && !SSL_USE_NSS
+#endif  // SSL_USE_OPENSSL
 
 ///////////////////////////////////////////////////////////////////////////////
 
 namespace rtc {
 
-SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
-#if SSL_USE_SCHANNEL
-  return NULL;
-#elif SSL_USE_OPENSSL  // !SSL_USE_SCHANNEL
-  return new OpenSSLStreamAdapter(stream);
-#elif SSL_USE_NSS     //  !SSL_USE_SCHANNEL && !SSL_USE_OPENSSL
-  return new NSSStreamAdapter(stream);
-#else  // !SSL_USE_SCHANNEL && !SSL_USE_OPENSSL && !SSL_USE_NSS
-  return NULL;
-#endif
+// TODO(guoweis): Move this to SDP layer and use int form internally.
+// webrtc:5043.
+const char CS_AES_CM_128_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80";
+const char CS_AES_CM_128_HMAC_SHA1_32[] = "AES_CM_128_HMAC_SHA1_32";
+
+std::string SrtpCryptoSuiteToName(int crypto_suite) {
+  if (crypto_suite == SRTP_AES128_CM_SHA1_32)
+    return CS_AES_CM_128_HMAC_SHA1_32;
+  if (crypto_suite == SRTP_AES128_CM_SHA1_80)
+    return CS_AES_CM_128_HMAC_SHA1_80;
+  return std::string();
 }
 
-bool SSLStreamAdapter::GetSslCipher(std::string* cipher) {
+int SrtpCryptoSuiteFromName(const std::string& crypto_suite) {
+  if (crypto_suite == CS_AES_CM_128_HMAC_SHA1_32)
+    return SRTP_AES128_CM_SHA1_32;
+  if (crypto_suite == CS_AES_CM_128_HMAC_SHA1_80)
+    return SRTP_AES128_CM_SHA1_80;
+  return SRTP_INVALID_CRYPTO_SUITE;
+}
+
+SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
+#if SSL_USE_OPENSSL
+  return new OpenSSLStreamAdapter(stream);
+#else  // !SSL_USE_OPENSSL
+  return NULL;
+#endif  // SSL_USE_OPENSSL
+}
+
+bool SSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) {
   return false;
 }
 
 bool SSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
-                                            const uint8* context,
+                                            const uint8_t* context,
                                             size_t context_len,
                                             bool use_context,
-                                            uint8* result,
+                                            uint8_t* result,
                                             size_t result_len) {
   return false;  // Default is unsupported
 }
 
-bool SSLStreamAdapter::SetDtlsSrtpCiphers(
-    const std::vector<std::string>& ciphers) {
+bool SSLStreamAdapter::SetDtlsSrtpCryptoSuites(
+    const std::vector<int>& crypto_suites) {
   return false;
 }
 
-bool SSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
+bool SSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
   return false;
 }
 
-// Note: this matches the logic above with SCHANNEL dominating
-#if SSL_USE_SCHANNEL
-bool SSLStreamAdapter::HaveDtls() { return false; }
-bool SSLStreamAdapter::HaveDtlsSrtp() { return false; }
-bool SSLStreamAdapter::HaveExporter() { return false; }
-std::string SSLStreamAdapter::GetDefaultSslCipher() {
-  return std::string();
-}
-#elif SSL_USE_OPENSSL
+#if SSL_USE_OPENSSL
 bool SSLStreamAdapter::HaveDtls() {
   return OpenSSLStreamAdapter::HaveDtls();
 }
@@ -85,23 +86,15 @@ bool SSLStreamAdapter::HaveDtlsSrtp() {
 bool SSLStreamAdapter::HaveExporter() {
   return OpenSSLStreamAdapter::HaveExporter();
 }
-std::string SSLStreamAdapter::GetDefaultSslCipher() {
-  return OpenSSLStreamAdapter::GetDefaultSslCipher();
+int SSLStreamAdapter::GetDefaultSslCipherForTest(SSLProtocolVersion version,
+                                                 KeyType key_type) {
+  return OpenSSLStreamAdapter::GetDefaultSslCipherForTest(version, key_type);
 }
-#elif SSL_USE_NSS
-bool SSLStreamAdapter::HaveDtls() {
-  return NSSStreamAdapter::HaveDtls();
+
+std::string SSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
+  return OpenSSLStreamAdapter::SslCipherSuiteToName(cipher_suite);
 }
-bool SSLStreamAdapter::HaveDtlsSrtp() {
-  return NSSStreamAdapter::HaveDtlsSrtp();
-}
-bool SSLStreamAdapter::HaveExporter() {
-  return NSSStreamAdapter::HaveExporter();
-}
-std::string SSLStreamAdapter::GetDefaultSslCipher() {
-  return NSSStreamAdapter::GetDefaultSslCipher();
-}
-#endif  // !SSL_USE_SCHANNEL && !SSL_USE_OPENSSL && !SSL_USE_NSS
+#endif  // SSL_USE_OPENSSL
 
 ///////////////////////////////////////////////////////////////////////////////
 
diff --git a/media/webrtc/trunk/webrtc/base/sslstreamadapter.h b/media/webrtc/trunk/webrtc/base/sslstreamadapter.h
index 2f819c859a..c57056b14a 100644
--- a/media/webrtc/trunk/webrtc/base/sslstreamadapter.h
+++ b/media/webrtc/trunk/webrtc/base/sslstreamadapter.h
@@ -19,6 +19,30 @@
 
 namespace rtc {
 
+// Constants for SSL profile.
+const int TLS_NULL_WITH_NULL_NULL = 0;
+
+// Constants for SRTP profiles.
+const int SRTP_INVALID_CRYPTO_SUITE = 0;
+const int SRTP_AES128_CM_SHA1_80 = 0x0001;
+const int SRTP_AES128_CM_SHA1_32 = 0x0002;
+
+// Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
+// in applications (voice) where the additional bandwidth may be significant.
+// A 80-bit HMAC is always used for SRTCP.
+// 128-bit AES with 80-bit SHA-1 HMAC.
+extern const char CS_AES_CM_128_HMAC_SHA1_80[];
+// 128-bit AES with 32-bit SHA-1 HMAC.
+extern const char CS_AES_CM_128_HMAC_SHA1_32[];
+
+// Given the DTLS-SRTP protection profile ID, as defined in
+// https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
+// name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2.
+std::string SrtpCryptoSuiteToName(int crypto_suite);
+
+// The reverse of above conversion.
+int SrtpCryptoSuiteFromName(const std::string& crypto_suite);
+
 // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
 // After SSL has been started, the stream will only open on successful
 // SSL verification of certificates, and the communication is
@@ -36,6 +60,13 @@ namespace rtc {
 
 enum SSLRole { SSL_CLIENT, SSL_SERVER };
 enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS };
+enum SSLProtocolVersion {
+  SSL_PROTOCOL_TLS_10,
+  SSL_PROTOCOL_TLS_11,
+  SSL_PROTOCOL_TLS_12,
+  SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11,
+  SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12,
+};
 
 // Errors for Read -- in the high range so no conflict with OpenSSL.
 enum { SSE_MSG_TRUNC = 0xff0001 };
@@ -74,6 +105,13 @@ class SSLStreamAdapter : public StreamAdapterInterface {
   // Do DTLS or TLS
   virtual void SetMode(SSLMode mode) = 0;
 
+  // Set maximum supported protocol version. The highest version supported by
+  // both ends will be used for the connection, i.e. if one party supports
+  // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
+  // If requested version is not supported by underlying crypto library, the
+  // next lower will be used.
+  virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0;
+
   // The mode of operation is selected by calling either
   // StartSSLWithServer or StartSSLWithPeer.
   // Use of the stream prior to calling either of these functions will
@@ -119,9 +157,9 @@ class SSLStreamAdapter : public StreamAdapterInterface {
   // chain. The returned certificate is owned by the caller.
   virtual bool GetPeerCertificate(SSLCertificate** cert) const = 0;
 
-  // Retrieves the name of the cipher suite used for the connection
-  // (e.g. "TLS_RSA_WITH_AES_128_CBC_SHA").
-  virtual bool GetSslCipher(std::string* cipher);
+  // Retrieves the IANA registration id of the cipher suite used for the
+  // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
+  virtual bool GetSslCipherSuite(int* cipher_suite);
 
   // Key Exporter interface from RFC 5705
   // Arguments are:
@@ -136,24 +174,31 @@ class SSLStreamAdapter : public StreamAdapterInterface {
   // result              -- where to put the computed value
   // result_len          -- the length of the computed value
   virtual bool ExportKeyingMaterial(const std::string& label,
-                                    const uint8* context,
+                                    const uint8_t* context,
                                     size_t context_len,
                                     bool use_context,
-                                    uint8* result,
+                                    uint8_t* result,
                                     size_t result_len);
 
   // DTLS-SRTP interface
-  virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers);
-  virtual bool GetDtlsSrtpCipher(std::string* cipher);
+  virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites);
+  virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite);
 
   // Capabilities testing
   static bool HaveDtls();
   static bool HaveDtlsSrtp();
   static bool HaveExporter();
 
-  // Returns the default Ssl cipher used between streams of this class.
-  // This is used by the unit tests.
-  static std::string GetDefaultSslCipher();
+  // Returns the default Ssl cipher used between streams of this class
+  // for the given protocol version. This is used by the unit tests.
+  // TODO(guoweis): Move this away from a static class method.
+  static int GetDefaultSslCipherForTest(SSLProtocolVersion version,
+                                        KeyType key_type);
+
+  // TODO(guoweis): Move this away from a static class method. Currently this is
+  // introduced such that any caller could depend on sslstreamadapter.h without
+  // depending on specific SSL implementation.
+  static std::string SslCipherSuiteToName(int cipher_suite);
 
  private:
   // If true, the server certificate need not match the configured
diff --git a/media/webrtc/trunk/webrtc/base/sslstreamadapter_unittest.cc b/media/webrtc/trunk/webrtc/base/sslstreamadapter_unittest.cc
index 677be35f67..a041c25211 100644
--- a/media/webrtc/trunk/webrtc/base/sslstreamadapter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/sslstreamadapter_unittest.cc
@@ -13,6 +13,7 @@
 #include <set>
 #include <string>
 
+#include "webrtc/base/bufferqueue.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -21,11 +22,13 @@
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stream.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+
+using ::testing::WithParamInterface;
+using ::testing::Values;
+using ::testing::Combine;
+using ::testing::tuple;
 
 static const int kBlockSize = 4096;
-static const char kAES_CM_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80";
-static const char kAES_CM_HMAC_SHA1_32[] = "AES_CM_128_HMAC_SHA1_32";
 static const char kExporterLabel[] = "label";
 static const unsigned char kExporterContext[] = "context";
 static int kExporterContextLen = sizeof(kExporterContext);
@@ -69,26 +72,26 @@ static const char kCERT_PEM[] =
 
 class SSLStreamAdapterTestBase;
 
-class SSLDummyStream : public rtc::StreamInterface,
-                       public sigslot::has_slots<> {
+class SSLDummyStreamBase : public rtc::StreamInterface,
+                           public sigslot::has_slots<> {
  public:
-  explicit SSLDummyStream(SSLStreamAdapterTestBase *test,
-                          const std::string &side,
-                          rtc::FifoBuffer *in,
-                          rtc::FifoBuffer *out) :
-      test_(test),
+  SSLDummyStreamBase(SSLStreamAdapterTestBase* test,
+                     const std::string &side,
+                     rtc::StreamInterface* in,
+                     rtc::StreamInterface* out) :
+      test_base_(test),
       side_(side),
       in_(in),
       out_(out),
       first_packet_(true) {
-    in_->SignalEvent.connect(this, &SSLDummyStream::OnEventIn);
-    out_->SignalEvent.connect(this, &SSLDummyStream::OnEventOut);
+    in_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventIn);
+    out_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventOut);
   }
 
-  virtual rtc::StreamState GetState() const { return rtc::SS_OPEN; }
+  rtc::StreamState GetState() const override { return rtc::SS_OPEN; }
 
-  virtual rtc::StreamResult Read(void* buffer, size_t buffer_len,
-                                       size_t* read, int* error) {
+  rtc::StreamResult Read(void* buffer, size_t buffer_len,
+                         size_t* read, int* error) override {
     rtc::StreamResult r;
 
     r = in_->Read(buffer, buffer_len, read, error);
@@ -106,22 +109,20 @@ class SSLDummyStream : public rtc::StreamInterface,
   }
 
   // Catch readability events on in and pass them up.
-  virtual void OnEventIn(rtc::StreamInterface *stream, int sig,
-                         int err) {
+  void OnEventIn(rtc::StreamInterface* stream, int sig, int err) {
     int mask = (rtc::SE_READ | rtc::SE_CLOSE);
 
     if (sig & mask) {
-      LOG(LS_INFO) << "SSLDummyStream::OnEvent side=" << side_ <<  " sig="
+      LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ <<  " sig="
         << sig << " forwarding upward";
       PostEvent(sig & mask, 0);
     }
   }
 
   // Catch writeability events on out and pass them up.
-  virtual void OnEventOut(rtc::StreamInterface *stream, int sig,
-                          int err) {
+  void OnEventOut(rtc::StreamInterface* stream, int sig, int err) {
     if (sig & rtc::SE_WRITE) {
-      LOG(LS_INFO) << "SSLDummyStream::OnEvent side=" << side_ <<  " sig="
+      LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ <<  " sig="
         << sig << " forwarding upward";
 
       PostEvent(sig & rtc::SE_WRITE, 0);
@@ -130,63 +131,120 @@ class SSLDummyStream : public rtc::StreamInterface,
 
   // Write to the outgoing FifoBuffer
   rtc::StreamResult WriteData(const void* data, size_t data_len,
-                                    size_t* written, int* error) {
+                              size_t* written, int* error) {
     return out_->Write(data, data_len, written, error);
   }
 
-  // Defined later
-  virtual rtc::StreamResult Write(const void* data, size_t data_len,
-                                        size_t* written, int* error);
+  rtc::StreamResult Write(const void* data, size_t data_len,
+                          size_t* written, int* error) override;
 
-  virtual void Close() {
+  void Close() override {
     LOG(LS_INFO) << "Closing outbound stream";
     out_->Close();
   }
 
- private:
-  SSLStreamAdapterTestBase *test_;
+ protected:
+  SSLStreamAdapterTestBase* test_base_;
   const std::string side_;
-  rtc::FifoBuffer *in_;
-  rtc::FifoBuffer *out_;
+  rtc::StreamInterface* in_;
+  rtc::StreamInterface* out_;
   bool first_packet_;
 };
 
+class SSLDummyStreamTLS : public SSLDummyStreamBase {
+ public:
+  SSLDummyStreamTLS(SSLStreamAdapterTestBase* test,
+                    const std::string& side,
+                    rtc::FifoBuffer* in,
+                    rtc::FifoBuffer* out) :
+      SSLDummyStreamBase(test, side, in, out) {
+  }
+};
+
+class BufferQueueStream : public rtc::BufferQueue,
+                          public rtc::StreamInterface {
+ public:
+  BufferQueueStream(size_t capacity, size_t default_size)
+      : rtc::BufferQueue(capacity, default_size) {
+  }
+
+  // Implementation of abstract StreamInterface methods.
+
+  // A buffer queue stream is always "open".
+  rtc::StreamState GetState() const override { return rtc::SS_OPEN; }
+
+  // Reading a buffer queue stream will either succeed or block.
+  rtc::StreamResult Read(void* buffer, size_t buffer_len,
+                         size_t* read, int* error) override {
+    if (!ReadFront(buffer, buffer_len, read)) {
+      return rtc::SR_BLOCK;
+    }
+    return rtc::SR_SUCCESS;
+  }
+
+  // Writing to a buffer queue stream will either succeed or block.
+  rtc::StreamResult Write(const void* data, size_t data_len,
+                          size_t* written, int* error) override {
+    if (!WriteBack(data, data_len, written)) {
+      return rtc::SR_BLOCK;
+    }
+    return rtc::SR_SUCCESS;
+  }
+
+  // A buffer queue stream can not be closed.
+  void Close() override {}
+
+ protected:
+  void NotifyReadableForTest() override {
+    PostEvent(rtc::SE_READ, 0);
+  }
+
+  void NotifyWritableForTest() override {
+    PostEvent(rtc::SE_WRITE, 0);
+  }
+};
+
+class SSLDummyStreamDTLS : public SSLDummyStreamBase {
+ public:
+  SSLDummyStreamDTLS(SSLStreamAdapterTestBase* test,
+                     const std::string& side,
+                     BufferQueueStream* in,
+                     BufferQueueStream* out) :
+      SSLDummyStreamBase(test, side, in, out) {
+  }
+};
+
 static const int kFifoBufferSize = 4096;
+static const int kBufferCapacity = 1;
+static const size_t kDefaultBufferSize = 2048;
 
 class SSLStreamAdapterTestBase : public testing::Test,
                                  public sigslot::has_slots<> {
  public:
-  SSLStreamAdapterTestBase(const std::string& client_cert_pem,
-                           const std::string& client_private_key_pem,
-                           bool dtls) :
-      client_buffer_(kFifoBufferSize), server_buffer_(kFifoBufferSize),
-      client_stream_(
-          new SSLDummyStream(this, "c2s", &client_buffer_, &server_buffer_)),
-      server_stream_(
-          new SSLDummyStream(this, "s2c", &server_buffer_, &client_buffer_)),
-      client_ssl_(rtc::SSLStreamAdapter::Create(client_stream_)),
-      server_ssl_(rtc::SSLStreamAdapter::Create(server_stream_)),
-      client_identity_(NULL), server_identity_(NULL),
-      delay_(0), mtu_(1460), loss_(0), lose_first_packet_(false),
-      damage_(false), dtls_(dtls),
-      handshake_wait_(5000), identities_set_(false) {
+  SSLStreamAdapterTestBase(
+      const std::string& client_cert_pem,
+      const std::string& client_private_key_pem,
+      bool dtls,
+      rtc::KeyParams client_key_type = rtc::KeyParams(rtc::KT_DEFAULT),
+      rtc::KeyParams server_key_type = rtc::KeyParams(rtc::KT_DEFAULT))
+      : client_cert_pem_(client_cert_pem),
+        client_private_key_pem_(client_private_key_pem),
+        client_key_type_(client_key_type),
+        server_key_type_(server_key_type),
+        client_stream_(NULL),
+        server_stream_(NULL),
+        client_identity_(NULL),
+        server_identity_(NULL),
+        delay_(0),
+        mtu_(1460),
+        loss_(0),
+        lose_first_packet_(false),
+        damage_(false),
+        dtls_(dtls),
+        handshake_wait_(5000),
+        identities_set_(false) {
     // Set use of the test RNG to get predictable loss patterns.
     rtc::SetRandomTestMode(true);
-
-    // Set up the slots
-    client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
-    server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
-
-    if (!client_cert_pem.empty() && !client_private_key_pem.empty()) {
-      client_identity_ = rtc::SSLIdentity::FromPEMStrings(
-          client_private_key_pem, client_cert_pem);
-    } else {
-      client_identity_ = rtc::SSLIdentity::Generate("client");
-    }
-    server_identity_ = rtc::SSLIdentity::Generate("server");
-
-    client_ssl_->SetIdentity(client_identity_);
-    server_ssl_->SetIdentity(server_identity_);
   }
 
   ~SSLStreamAdapterTestBase() {
@@ -194,14 +252,40 @@ class SSLStreamAdapterTestBase : public testing::Test,
     rtc::SetRandomTestMode(false);
   }
 
+  void SetUp() override {
+    CreateStreams();
+
+    client_ssl_.reset(rtc::SSLStreamAdapter::Create(client_stream_));
+    server_ssl_.reset(rtc::SSLStreamAdapter::Create(server_stream_));
+
+    // Set up the slots
+    client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
+    server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
+
+    if (!client_cert_pem_.empty() && !client_private_key_pem_.empty()) {
+      client_identity_ = rtc::SSLIdentity::FromPEMStrings(
+          client_private_key_pem_, client_cert_pem_);
+    } else {
+      client_identity_ = rtc::SSLIdentity::Generate("client", client_key_type_);
+    }
+    server_identity_ = rtc::SSLIdentity::Generate("server", server_key_type_);
+
+    client_ssl_->SetIdentity(client_identity_);
+    server_ssl_->SetIdentity(server_identity_);
+  }
+
+  void TearDown() override {
+    client_ssl_.reset(nullptr);
+    server_ssl_.reset(nullptr);
+  }
+
+  virtual void CreateStreams() = 0;
+
   // Recreate the client/server identities with the specified validity period.
   // |not_before| and |not_after| are offsets from the current time in number
   // of seconds.
   void ResetIdentitiesWithValidity(int not_before, int not_after) {
-    client_stream_ =
-        new SSLDummyStream(this, "c2s", &client_buffer_, &server_buffer_);
-    server_stream_ =
-        new SSLDummyStream(this, "s2c", &server_buffer_, &client_buffer_);
+    CreateStreams();
 
     client_ssl_.reset(rtc::SSLStreamAdapter::Create(client_stream_));
     server_ssl_.reset(rtc::SSLStreamAdapter::Create(server_stream_));
@@ -209,16 +293,20 @@ class SSLStreamAdapterTestBase : public testing::Test,
     client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
     server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
 
+    time_t now = time(nullptr);
+
     rtc::SSLIdentityParams client_params;
+    client_params.key_params = rtc::KeyParams(rtc::KT_DEFAULT);
     client_params.common_name = "client";
-    client_params.not_before = not_before;
-    client_params.not_after = not_after;
+    client_params.not_before = now + not_before;
+    client_params.not_after = now + not_after;
     client_identity_ = rtc::SSLIdentity::GenerateForTest(client_params);
 
     rtc::SSLIdentityParams server_params;
+    server_params.key_params = rtc::KeyParams(rtc::KT_DEFAULT);
     server_params.common_name = "server";
-    server_params.not_before = not_before;
-    server_params.not_after = not_after;
+    server_params.not_before = now + not_before;
+    server_params.not_after = now + not_after;
     server_identity_ = rtc::SSLIdentity::GenerateForTest(server_params);
 
     client_ssl_->SetIdentity(client_identity_);
@@ -271,6 +359,12 @@ class SSLStreamAdapterTestBase : public testing::Test,
     identities_set_ = true;
   }
 
+  void SetupProtocolVersions(rtc::SSLProtocolVersion server_version,
+                             rtc::SSLProtocolVersion client_version) {
+    server_ssl_->SetMaxProtocolVersion(server_version);
+    client_ssl_->SetMaxProtocolVersion(client_version);
+  }
+
   void TestHandshake(bool expect_success = true) {
     server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS :
                          rtc::SSL_MODE_TLS);
@@ -308,11 +402,11 @@ class SSLStreamAdapterTestBase : public testing::Test,
     }
   }
 
-  rtc::StreamResult DataWritten(SSLDummyStream *from, const void *data,
-                                      size_t data_len, size_t *written,
-                                      int *error) {
+  rtc::StreamResult DataWritten(SSLDummyStreamBase *from, const void *data,
+                                size_t data_len, size_t *written,
+                                int *error) {
     // Randomly drop loss_ percent of packets
-    if (rtc::CreateRandomId() % 100 < static_cast<uint32>(loss_)) {
+    if (rtc::CreateRandomId() % 100 < static_cast<uint32_t>(loss_)) {
       LOG(LS_INFO) << "Randomly dropping packet, size=" << data_len;
       *written = data_len;
       return rtc::SR_SUCCESS;
@@ -366,19 +460,18 @@ class SSLStreamAdapterTestBase : public testing::Test,
     handshake_wait_ = wait;
   }
 
-  void SetDtlsSrtpCiphers(const std::vector<std::string> &ciphers,
-    bool client) {
+  void SetDtlsSrtpCryptoSuites(const std::vector<int>& ciphers, bool client) {
     if (client)
-      client_ssl_->SetDtlsSrtpCiphers(ciphers);
+      client_ssl_->SetDtlsSrtpCryptoSuites(ciphers);
     else
-      server_ssl_->SetDtlsSrtpCiphers(ciphers);
+      server_ssl_->SetDtlsSrtpCryptoSuites(ciphers);
   }
 
-  bool GetDtlsSrtpCipher(bool client, std::string *retval) {
+  bool GetDtlsSrtpCryptoSuite(bool client, int* retval) {
     if (client)
-      return client_ssl_->GetDtlsSrtpCipher(retval);
+      return client_ssl_->GetDtlsSrtpCryptoSuite(retval);
     else
-      return server_ssl_->GetDtlsSrtpCipher(retval);
+      return server_ssl_->GetDtlsSrtpCryptoSuite(retval);
   }
 
   bool GetPeerCertificate(bool client, rtc::SSLCertificate** cert) {
@@ -388,11 +481,11 @@ class SSLStreamAdapterTestBase : public testing::Test,
       return server_ssl_->GetPeerCertificate(cert);
   }
 
-  bool GetSslCipher(bool client, std::string *retval) {
+  bool GetSslCipherSuite(bool client, int* retval) {
     if (client)
-      return client_ssl_->GetSslCipher(retval);
+      return client_ssl_->GetSslCipherSuite(retval);
     else
-      return server_ssl_->GetSslCipher(retval);
+      return server_ssl_->GetSslCipherSuite(retval);
   }
 
   bool ExportKeyingMaterial(const char *label,
@@ -420,10 +513,12 @@ class SSLStreamAdapterTestBase : public testing::Test,
   virtual void TestTransfer(int size) = 0;
 
  protected:
-  rtc::FifoBuffer client_buffer_;
-  rtc::FifoBuffer server_buffer_;
-  SSLDummyStream *client_stream_;  // freed by client_ssl_ destructor
-  SSLDummyStream *server_stream_;  // freed by server_ssl_ destructor
+  std::string client_cert_pem_;
+  std::string client_private_key_pem_;
+  rtc::KeyParams client_key_type_;
+  rtc::KeyParams server_key_type_;
+  SSLDummyStreamBase *client_stream_;  // freed by client_ssl_ destructor
+  SSLDummyStreamBase *server_stream_;  // freed by server_ssl_ destructor
   rtc::scoped_ptr<rtc::SSLStreamAdapter> client_ssl_;
   rtc::scoped_ptr<rtc::SSLStreamAdapter> server_ssl_;
   rtc::SSLIdentity *client_identity_;  // freed by client_ssl_ destructor
@@ -438,11 +533,26 @@ class SSLStreamAdapterTestBase : public testing::Test,
   bool identities_set_;
 };
 
-class SSLStreamAdapterTestTLS : public SSLStreamAdapterTestBase {
+class SSLStreamAdapterTestTLS
+    : public SSLStreamAdapterTestBase,
+      public WithParamInterface<tuple<rtc::KeyParams, rtc::KeyParams>> {
  public:
-  SSLStreamAdapterTestTLS() :
-      SSLStreamAdapterTestBase("", "", false) {
-  };
+  SSLStreamAdapterTestTLS()
+      : SSLStreamAdapterTestBase("",
+                                 "",
+                                 false,
+                                 ::testing::get<0>(GetParam()),
+                                 ::testing::get<1>(GetParam())),
+        client_buffer_(kFifoBufferSize),
+        server_buffer_(kFifoBufferSize) {
+  }
+
+  void CreateStreams() override {
+    client_stream_ =
+        new SSLDummyStreamTLS(this, "c2s", &client_buffer_, &server_buffer_);
+    server_stream_ =
+        new SSLDummyStreamTLS(this, "s2c", &server_buffer_, &client_buffer_);
+  }
 
   // Test data transfer for TLS
   virtual void TestTransfer(int size) {
@@ -521,7 +631,7 @@ class SSLStreamAdapterTestTLS : public SSLStreamAdapterTestBase {
 
       if (r == rtc::SR_ERROR || r == rtc::SR_EOS) {
         // Unfortunately, errors are the way that the stream adapter
-        // signals close in OpenSSL
+        // signals close in OpenSSL.
         stream->Close();
         return;
       }
@@ -537,32 +647,57 @@ class SSLStreamAdapterTestTLS : public SSLStreamAdapterTestBase {
   }
 
  private:
+  rtc::FifoBuffer client_buffer_;
+  rtc::FifoBuffer server_buffer_;
   rtc::MemoryStream send_stream_;
   rtc::MemoryStream recv_stream_;
 };
 
-class SSLStreamAdapterTestDTLS : public SSLStreamAdapterTestBase {
+class SSLStreamAdapterTestDTLS
+    : public SSLStreamAdapterTestBase,
+      public WithParamInterface<tuple<rtc::KeyParams, rtc::KeyParams>> {
  public:
-  SSLStreamAdapterTestDTLS() :
-      SSLStreamAdapterTestBase("", "", true),
-      packet_size_(1000), count_(0), sent_(0) {
-  }
+  SSLStreamAdapterTestDTLS()
+      : SSLStreamAdapterTestBase("",
+                                 "",
+                                 true,
+                                 ::testing::get<0>(GetParam()),
+                                 ::testing::get<1>(GetParam())),
+        client_buffer_(kBufferCapacity, kDefaultBufferSize),
+        server_buffer_(kBufferCapacity, kDefaultBufferSize),
+        packet_size_(1000),
+        count_(0),
+        sent_(0) {}
 
   SSLStreamAdapterTestDTLS(const std::string& cert_pem,
                            const std::string& private_key_pem) :
       SSLStreamAdapterTestBase(cert_pem, private_key_pem, true),
+      client_buffer_(kBufferCapacity, kDefaultBufferSize),
+      server_buffer_(kBufferCapacity, kDefaultBufferSize),
       packet_size_(1000), count_(0), sent_(0) {
   }
 
+  void CreateStreams() override {
+    client_stream_ =
+        new SSLDummyStreamDTLS(this, "c2s", &client_buffer_, &server_buffer_);
+    server_stream_ =
+        new SSLDummyStreamDTLS(this, "s2c", &server_buffer_, &client_buffer_);
+  }
+
   virtual void WriteData() {
     unsigned char *packet = new unsigned char[1600];
 
-    do {
-      memset(packet, sent_ & 0xff, packet_size_);
-      *(reinterpret_cast<uint32_t *>(packet)) = sent_;
+    while (sent_ < count_) {
+      unsigned int rand_state = sent_;
+      packet[0] = sent_;
+      for (size_t i = 1; i < packet_size_; i++) {
+        // This is a simple LC PRNG.  Keep in synch with identical code below.
+        rand_state = (rand_state * 251 + 19937) >> 7;
+        packet[i] = rand_state & 0xff;
+      }
 
       size_t sent;
-      int rv = client_ssl_->Write(packet, packet_size_, &sent, 0);
+      rtc::StreamResult rv = client_ssl_->Write(packet, packet_size_, &sent, 0);
       if (rv == rtc::SR_SUCCESS) {
         LOG(LS_VERBOSE) << "Sent: " << sent_;
         sent_++;
@@ -573,7 +708,7 @@ class SSLStreamAdapterTestDTLS : public SSLStreamAdapterTestBase {
         ADD_FAILURE();
         break;
       }
-    } while (sent_ < count_);
+    }
 
     delete [] packet;
   }
@@ -602,11 +737,13 @@ class SSLStreamAdapterTestDTLS : public SSLStreamAdapterTestBase {
 
       // Now parse the datagram
       ASSERT_EQ(packet_size_, bread);
-      unsigned char* ptr_to_buffer = buffer;
-      uint32_t packet_num = *(reinterpret_cast<uint32_t *>(ptr_to_buffer));
+      unsigned char packet_num = buffer[0];
 
-      for (size_t i = 4; i < packet_size_; i++) {
-        ASSERT_EQ((packet_num & 0xff), buffer[i]);
+      unsigned int rand_state = packet_num;
+      for (size_t i = 1; i < packet_size_; i++) {
+        // This is a simple LC PRNG.  Keep in synch with identical code above.
+        rand_state = (rand_state * 251 + 19937) >> 7;
+        ASSERT_EQ(rand_state & 0xff, buffer[i]);
       }
       received_.insert(packet_num);
     }
@@ -632,6 +769,8 @@ class SSLStreamAdapterTestDTLS : public SSLStreamAdapterTestBase {
   };
 
  private:
+  BufferQueueStream client_buffer_;
+  BufferQueueStream server_buffer_;
   size_t packet_size_;
   int count_;
   int sent_;
@@ -639,23 +778,20 @@ class SSLStreamAdapterTestDTLS : public SSLStreamAdapterTestBase {
 };
 
 
-rtc::StreamResult SSLDummyStream::Write(const void* data, size_t data_len,
+rtc::StreamResult SSLDummyStreamBase::Write(const void* data, size_t data_len,
                                               size_t* written, int* error) {
-  *written = data_len;
-
   LOG(LS_INFO) << "Writing to loopback " << data_len;
 
   if (first_packet_) {
     first_packet_ = false;
-    if (test_->GetLoseFirstPacket()) {
+    if (test_base_->GetLoseFirstPacket()) {
       LOG(LS_INFO) << "Losing initial packet of length " << data_len;
+      *written = data_len;  // Fake successful writing also to writer.
       return rtc::SR_SUCCESS;
     }
   }
 
-  return test_->DataWritten(this, data, data_len, written, error);
-
-  return rtc::SR_SUCCESS;
+  return test_base_->DataWritten(this, data, data_len, written, error);
 };
 
 class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS {
@@ -667,44 +803,26 @@ class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS {
 
 // Basic tests: TLS
 
-// Test that we cannot read/write if we have not yet handshaked.
-// This test only applies to NSS because OpenSSL has passthrough
-// semantics for I/O before the handshake is started.
-#if SSL_USE_NSS
-TEST_F(SSLStreamAdapterTestTLS, TestNoReadWriteBeforeConnect) {
-  rtc::StreamResult rv;
-  char block[kBlockSize];
-  size_t dummy;
-
-  rv = client_ssl_->Write(block, sizeof(block), &dummy, NULL);
-  ASSERT_EQ(rtc::SR_BLOCK, rv);
-
-  rv = client_ssl_->Read(block, sizeof(block), &dummy, NULL);
-  ASSERT_EQ(rtc::SR_BLOCK, rv);
-}
-#endif
-
-
 // Test that we can make a handshake work
-TEST_F(SSLStreamAdapterTestTLS, TestTLSConnect) {
+TEST_P(SSLStreamAdapterTestTLS, TestTLSConnect) {
   TestHandshake();
 };
 
 // Test that closing the connection on one side updates the other side.
-TEST_F(SSLStreamAdapterTestTLS, TestTLSClose) {
+TEST_P(SSLStreamAdapterTestTLS, TestTLSClose) {
   TestHandshake();
   client_ssl_->Close();
   EXPECT_EQ_WAIT(rtc::SS_CLOSED, server_ssl_->GetState(), handshake_wait_);
 };
 
 // Test transfer -- trivial
-TEST_F(SSLStreamAdapterTestTLS, TestTLSTransfer) {
+TEST_P(SSLStreamAdapterTestTLS, TestTLSTransfer) {
   TestHandshake();
   TestTransfer(100000);
 };
 
 // Test read-write after close.
-TEST_F(SSLStreamAdapterTestTLS, ReadWriteAfterClose) {
+TEST_P(SSLStreamAdapterTestTLS, ReadWriteAfterClose) {
   TestHandshake();
   TestTransfer(100000);
   client_ssl_->Close();
@@ -723,7 +841,7 @@ TEST_F(SSLStreamAdapterTestTLS, ReadWriteAfterClose) {
 };
 
 // Test a handshake with a bogus peer digest
-TEST_F(SSLStreamAdapterTestTLS, TestTLSBogusDigest) {
+TEST_P(SSLStreamAdapterTestTLS, TestTLSBogusDigest) {
   SetPeerIdentitiesByDigest(false);
   TestHandshake(false);
 };
@@ -732,7 +850,7 @@ TEST_F(SSLStreamAdapterTestTLS, TestTLSBogusDigest) {
 
 // Basic tests: DTLS
 // Test that we can make a handshake work
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSConnect) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) {
   MAYBE_SKIP_TEST(HaveDtls);
   TestHandshake();
 };
@@ -740,15 +858,14 @@ TEST_F(SSLStreamAdapterTestDTLS, TestDTLSConnect) {
 // Test that we can make a handshake work if the first packet in
 // each direction is lost. This gives us predictable loss
 // rather than having to tune random
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacket) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacket) {
   MAYBE_SKIP_TEST(HaveDtls);
   SetLoseFirstPacket(true);
   TestHandshake();
 };
 
 // Test a handshake with loss and delay
-TEST_F(SSLStreamAdapterTestDTLS,
-       TestDTLSConnectWithLostFirstPacketDelay2s) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) {
   MAYBE_SKIP_TEST(HaveDtls);
   SetLoseFirstPacket(true);
   SetDelay(2000);
@@ -758,7 +875,7 @@ TEST_F(SSLStreamAdapterTestDTLS,
 
 // Test a handshake with small MTU
 // Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3910
-TEST_F(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) {
+TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) {
   MAYBE_SKIP_TEST(HaveDtls);
   SetMtu(700);
   SetHandshakeWait(20000);
@@ -766,20 +883,20 @@ TEST_F(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) {
 };
 
 // Test transfer -- trivial
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSTransfer) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransfer) {
   MAYBE_SKIP_TEST(HaveDtls);
   TestHandshake();
   TestTransfer(100);
 };
 
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) {
   MAYBE_SKIP_TEST(HaveDtls);
   TestHandshake();
   SetLoss(10);
   TestTransfer(100);
 };
 
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) {
   MAYBE_SKIP_TEST(HaveDtls);
   SetDamage();  // Must be called first because first packet
                 // write happens at end of handshake.
@@ -788,80 +905,80 @@ TEST_F(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) {
 };
 
 // Test DTLS-SRTP with all high ciphers
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  std::vector<std::string> high;
-  high.push_back(kAES_CM_HMAC_SHA1_80);
-  SetDtlsSrtpCiphers(high, true);
-  SetDtlsSrtpCiphers(high, false);
+  std::vector<int> high;
+  high.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+  SetDtlsSrtpCryptoSuites(high, true);
+  SetDtlsSrtpCryptoSuites(high, false);
   TestHandshake();
 
-  std::string client_cipher;
-  ASSERT_TRUE(GetDtlsSrtpCipher(true, &client_cipher));
-  std::string server_cipher;
-  ASSERT_TRUE(GetDtlsSrtpCipher(false, &server_cipher));
+  int client_cipher;
+  ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
 
   ASSERT_EQ(client_cipher, server_cipher);
-  ASSERT_EQ(client_cipher, kAES_CM_HMAC_SHA1_80);
+  ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_80);
 };
 
 // Test DTLS-SRTP with all low ciphers
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  std::vector<std::string> low;
-  low.push_back(kAES_CM_HMAC_SHA1_32);
-  SetDtlsSrtpCiphers(low, true);
-  SetDtlsSrtpCiphers(low, false);
+  std::vector<int> low;
+  low.push_back(rtc::SRTP_AES128_CM_SHA1_32);
+  SetDtlsSrtpCryptoSuites(low, true);
+  SetDtlsSrtpCryptoSuites(low, false);
   TestHandshake();
 
-  std::string client_cipher;
-  ASSERT_TRUE(GetDtlsSrtpCipher(true, &client_cipher));
-  std::string server_cipher;
-  ASSERT_TRUE(GetDtlsSrtpCipher(false, &server_cipher));
+  int client_cipher;
+  ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
 
   ASSERT_EQ(client_cipher, server_cipher);
-  ASSERT_EQ(client_cipher, kAES_CM_HMAC_SHA1_32);
+  ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_32);
 };
 
 
 // Test DTLS-SRTP with a mismatch -- should not converge
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  std::vector<std::string> high;
-  high.push_back(kAES_CM_HMAC_SHA1_80);
-  std::vector<std::string> low;
-  low.push_back(kAES_CM_HMAC_SHA1_32);
-  SetDtlsSrtpCiphers(high, true);
-  SetDtlsSrtpCiphers(low, false);
+  std::vector<int> high;
+  high.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+  std::vector<int> low;
+  low.push_back(rtc::SRTP_AES128_CM_SHA1_32);
+  SetDtlsSrtpCryptoSuites(high, true);
+  SetDtlsSrtpCryptoSuites(low, false);
   TestHandshake();
 
-  std::string client_cipher;
-  ASSERT_FALSE(GetDtlsSrtpCipher(true, &client_cipher));
-  std::string server_cipher;
-  ASSERT_FALSE(GetDtlsSrtpCipher(false, &server_cipher));
+  int client_cipher;
+  ASSERT_FALSE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_FALSE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
 };
 
 // Test DTLS-SRTP with each side being mixed -- should select high
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  std::vector<std::string> mixed;
-  mixed.push_back(kAES_CM_HMAC_SHA1_80);
-  mixed.push_back(kAES_CM_HMAC_SHA1_32);
-  SetDtlsSrtpCiphers(mixed, true);
-  SetDtlsSrtpCiphers(mixed, false);
+  std::vector<int> mixed;
+  mixed.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+  mixed.push_back(rtc::SRTP_AES128_CM_SHA1_32);
+  SetDtlsSrtpCryptoSuites(mixed, true);
+  SetDtlsSrtpCryptoSuites(mixed, false);
   TestHandshake();
 
-  std::string client_cipher;
-  ASSERT_TRUE(GetDtlsSrtpCipher(true, &client_cipher));
-  std::string server_cipher;
-  ASSERT_TRUE(GetDtlsSrtpCipher(false, &server_cipher));
+  int client_cipher;
+  ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
 
   ASSERT_EQ(client_cipher, server_cipher);
-  ASSERT_EQ(client_cipher, kAES_CM_HMAC_SHA1_80);
+  ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_80);
 };
 
 // Test an exporter
-TEST_F(SSLStreamAdapterTestDTLS, TestDTLSExporter) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) {
   MAYBE_SKIP_TEST(HaveExporter);
   TestHandshake();
   unsigned char client_out[20];
@@ -884,7 +1001,7 @@ TEST_F(SSLStreamAdapterTestDTLS, TestDTLSExporter) {
 }
 
 // Test not yet valid certificates are not rejected.
-TEST_F(SSLStreamAdapterTestDTLS, TestCertNotYetValid) {
+TEST_P(SSLStreamAdapterTestDTLS, TestCertNotYetValid) {
   MAYBE_SKIP_TEST(HaveDtls);
   long one_day = 60 * 60 * 24;
   // Make the certificates not valid until one day later.
@@ -893,7 +1010,7 @@ TEST_F(SSLStreamAdapterTestDTLS, TestCertNotYetValid) {
 }
 
 // Test expired certificates are not rejected.
-TEST_F(SSLStreamAdapterTestDTLS, TestCertExpired) {
+TEST_P(SSLStreamAdapterTestDTLS, TestCertExpired) {
   MAYBE_SKIP_TEST(HaveDtls);
   long one_day = 60 * 60 * 24;
   // Make the certificates already expired.
@@ -948,15 +1065,97 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
 }
 
 // Test getting the used DTLS ciphers.
-TEST_F(SSLStreamAdapterTestDTLS, TestGetSslCipher) {
+// DTLS 1.2 enabled for neither client nor server -> DTLS 1.0 will be used.
+TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuite) {
   MAYBE_SKIP_TEST(HaveDtls);
+  SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10);
   TestHandshake();
 
-  std::string client_cipher;
-  ASSERT_TRUE(GetSslCipher(true, &client_cipher));
-  std::string server_cipher;
-  ASSERT_TRUE(GetSslCipher(false, &server_cipher));
+  int client_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher));
 
   ASSERT_EQ(client_cipher, server_cipher);
-  ASSERT_EQ(rtc::SSLStreamAdapter::GetDefaultSslCipher(), client_cipher);
+  ASSERT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
+          rtc::SSL_PROTOCOL_DTLS_10, ::testing::get<1>(GetParam()).type()),
+      server_cipher);
 }
+
+// Test getting the used DTLS 1.2 ciphers.
+// DTLS 1.2 enabled for client and server -> DTLS 1.2 will be used.
+TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Both) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12);
+  TestHandshake();
+
+  int client_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher));
+
+  ASSERT_EQ(client_cipher, server_cipher);
+  ASSERT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
+          rtc::SSL_PROTOCOL_DTLS_12, ::testing::get<1>(GetParam()).type()),
+      server_cipher);
+}
+
+// DTLS 1.2 enabled for client only -> DTLS 1.0 will be used.
+TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Client) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12);
+  TestHandshake();
+
+  int client_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher));
+
+  ASSERT_EQ(client_cipher, server_cipher);
+  ASSERT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
+          rtc::SSL_PROTOCOL_DTLS_10, ::testing::get<1>(GetParam()).type()),
+      server_cipher);
+}
+
+// DTLS 1.2 enabled for server only -> DTLS 1.0 will be used.
+TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Server) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10);
+  TestHandshake();
+
+  int client_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(true, &client_cipher));
+  int server_cipher;
+  ASSERT_TRUE(GetSslCipherSuite(false, &server_cipher));
+
+  ASSERT_EQ(client_cipher, server_cipher);
+  ASSERT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
+          rtc::SSL_PROTOCOL_DTLS_10, ::testing::get<1>(GetParam()).type()),
+      server_cipher);
+}
+
+// The RSA keysizes here might look strange, why not include the RFC's size
+// 2048?. The reason is test case slowness; testing two sizes to exercise
+// parametrization is sufficient.
+INSTANTIATE_TEST_CASE_P(
+    SSLStreamAdapterTestsTLS,
+    SSLStreamAdapterTestTLS,
+    Combine(Values(rtc::KeyParams::RSA(1024, 65537),
+                   rtc::KeyParams::RSA(1152, 65537),
+                   rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)),
+            Values(rtc::KeyParams::RSA(1024, 65537),
+                   rtc::KeyParams::RSA(1152, 65537),
+                   rtc::KeyParams::ECDSA(rtc::EC_NIST_P256))));
+INSTANTIATE_TEST_CASE_P(
+    SSLStreamAdapterTestsDTLS,
+    SSLStreamAdapterTestDTLS,
+    Combine(Values(rtc::KeyParams::RSA(1024, 65537),
+                   rtc::KeyParams::RSA(1152, 65537),
+                   rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)),
+            Values(rtc::KeyParams::RSA(1024, 65537),
+                   rtc::KeyParams::RSA(1152, 65537),
+                   rtc::KeyParams::ECDSA(rtc::EC_NIST_P256))));
diff --git a/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.cc b/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.cc
index 1ab7369613..61c0e43ff7 100644
--- a/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.cc
+++ b/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.cc
@@ -28,8 +28,8 @@ SSLStreamAdapterHelper::SSLStreamAdapterHelper(StreamInterface* stream)
       state_(SSL_NONE),
       role_(SSL_CLIENT),
       ssl_error_code_(0),  // Not meaningful yet
-      ssl_mode_(SSL_MODE_TLS) {
-}
+      ssl_mode_(SSL_MODE_TLS),
+      ssl_max_version_(SSL_PROTOCOL_TLS_12) {}
 
 SSLStreamAdapterHelper::~SSLStreamAdapterHelper() = default;
 
@@ -59,6 +59,10 @@ void SSLStreamAdapterHelper::SetMode(SSLMode mode) {
   ssl_mode_ = mode;
 }
 
+void SSLStreamAdapterHelper::SetMaxProtocolVersion(SSLProtocolVersion version) {
+  ssl_max_version_ = version;
+}
+
 StreamState SSLStreamAdapterHelper::GetState() const {
   switch (state_) {
     case SSL_WAIT:
diff --git a/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.h b/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.h
index 1c856e8993..c6979ba036 100644
--- a/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.h
+++ b/media/webrtc/trunk/webrtc/base/sslstreamadapterhelper.h
@@ -23,7 +23,7 @@ namespace rtc {
 
 // SSLStreamAdapterHelper : A stream adapter which implements much
 // of the logic that is common between the known implementations
-// (NSS and OpenSSL)
+// (OpenSSL and previously NSS)
 class SSLStreamAdapterHelper : public SSLStreamAdapter {
  public:
   explicit SSLStreamAdapterHelper(StreamInterface* stream);
@@ -33,6 +33,7 @@ class SSLStreamAdapterHelper : public SSLStreamAdapter {
   void SetIdentity(SSLIdentity* identity) override;
   void SetServerRole(SSLRole role = SSL_SERVER) override;
   void SetMode(SSLMode mode) override;
+  void SetMaxProtocolVersion(SSLProtocolVersion version) override;
 
   int StartSSLWithServer(const char* server_name) override;
   int StartSSLWithPeer() override;
@@ -101,6 +102,9 @@ class SSLStreamAdapterHelper : public SSLStreamAdapter {
   // Do DTLS or not
   SSLMode ssl_mode_;
 
+  // Maximum allowed protocol version.
+  SSLProtocolVersion ssl_max_version_;
+
  private:
   // Go from state SSL_NONE to either SSL_CONNECTING or SSL_WAIT,
   // depending on whether the underlying stream is already open or
diff --git a/media/webrtc/trunk/webrtc/base/stream.cc b/media/webrtc/trunk/webrtc/base/stream.cc
index 0fdb1fcd83..e22c3d8aa4 100644
--- a/media/webrtc/trunk/webrtc/base/stream.cc
+++ b/media/webrtc/trunk/webrtc/base/stream.cc
@@ -291,89 +291,6 @@ StreamResult StreamTap::Write(const void* data, size_t data_len,
   return res;
 }
 
-///////////////////////////////////////////////////////////////////////////////
-// StreamSegment
-///////////////////////////////////////////////////////////////////////////////
-
-StreamSegment::StreamSegment(StreamInterface* stream)
-    : StreamAdapterInterface(stream), start_(SIZE_UNKNOWN), pos_(0),
-    length_(SIZE_UNKNOWN) {
-  // It's ok for this to fail, in which case start_ is left as SIZE_UNKNOWN.
-  stream->GetPosition(&start_);
-}
-
-StreamSegment::StreamSegment(StreamInterface* stream, size_t length)
-    : StreamAdapterInterface(stream), start_(SIZE_UNKNOWN), pos_(0),
-    length_(length) {
-  // It's ok for this to fail, in which case start_ is left as SIZE_UNKNOWN.
-  stream->GetPosition(&start_);
-}
-
-StreamResult StreamSegment::Read(void* buffer, size_t buffer_len,
-                                 size_t* read, int* error) {
-  if (SIZE_UNKNOWN != length_) {
-    if (pos_ >= length_)
-      return SR_EOS;
-    buffer_len = std::min(buffer_len, length_ - pos_);
-  }
-  size_t backup_read;
-  if (!read) {
-    read = &backup_read;
-  }
-  StreamResult result = StreamAdapterInterface::Read(buffer, buffer_len,
-                                                     read, error);
-  if (SR_SUCCESS == result) {
-    pos_ += *read;
-  }
-  return result;
-}
-
-bool StreamSegment::SetPosition(size_t position) {
-  if (SIZE_UNKNOWN == start_)
-    return false;  // Not seekable
-  if ((SIZE_UNKNOWN != length_) && (position > length_))
-    return false;  // Seek past end of segment
-  if (!StreamAdapterInterface::SetPosition(start_ + position))
-    return false;
-  pos_ = position;
-  return true;
-}
-
-bool StreamSegment::GetPosition(size_t* position) const {
-  if (SIZE_UNKNOWN == start_)
-    return false;  // Not seekable
-  if (!StreamAdapterInterface::GetPosition(position))
-    return false;
-  if (position) {
-    ASSERT(*position >= start_);
-    *position -= start_;
-  }
-  return true;
-}
-
-bool StreamSegment::GetSize(size_t* size) const {
-  if (!StreamAdapterInterface::GetSize(size))
-    return false;
-  if (size) {
-    if (SIZE_UNKNOWN != start_) {
-      ASSERT(*size >= start_);
-      *size -= start_;
-    }
-    if (SIZE_UNKNOWN != length_) {
-      *size = std::min(*size, length_);
-    }
-  }
-  return true;
-}
-
-bool StreamSegment::GetAvailable(size_t* size) const {
-  if (!StreamAdapterInterface::GetAvailable(size))
-    return false;
-  if (size && (SIZE_UNKNOWN != length_))
-    *size = std::min(*size, length_ - pos_);
-  return true;
-}
-
 ///////////////////////////////////////////////////////////////////////////////
 // NullStream
 ///////////////////////////////////////////////////////////////////////////////
@@ -600,237 +517,6 @@ void FileStream::DoClose() {
   fclose(file_);
 }
 
-CircularFileStream::CircularFileStream(size_t max_size)
-  : max_write_size_(max_size),
-    position_(0),
-    marked_position_(max_size / 2),
-    last_write_position_(0),
-    read_segment_(READ_LATEST),
-    read_segment_available_(0) {
-}
-
-bool CircularFileStream::Open(
-    const std::string& filename, const char* mode, int* error) {
-  if (!FileStream::Open(filename.c_str(), mode, error))
-    return false;
-
-  if (strchr(mode, "r") != NULL) {  // Opened in read mode.
-    // Check if the buffer has been overwritten and determine how to read the
-    // log in time sequence.
-    size_t file_size;
-    GetSize(&file_size);
-    if (file_size == position_) {
-      // The buffer has not been overwritten yet. Read 0 .. file_size
-      read_segment_ = READ_LATEST;
-      read_segment_available_ = file_size;
-    } else {
-      // The buffer has been over written. There are three segments: The first
-      // one is 0 .. marked_position_, which is the marked earliest log. The
-      // second one is position_ .. file_size, which is the middle log. The
-      // last one is marked_position_ .. position_, which is the latest log.
-      read_segment_ = READ_MARKED;
-      read_segment_available_ = marked_position_;
-      last_write_position_ = position_;
-    }
-
-    // Read from the beginning.
-    position_ = 0;
-    SetPosition(position_);
-  }
-
-  return true;
-}
-
-StreamResult CircularFileStream::Read(void* buffer, size_t buffer_len,
-                                      size_t* read, int* error) {
-  if (read_segment_available_ == 0) {
-    size_t file_size;
-    switch (read_segment_) {
-      case READ_MARKED:  // Finished READ_MARKED and start READ_MIDDLE.
-        read_segment_ = READ_MIDDLE;
-        position_ = last_write_position_;
-        SetPosition(position_);
-        GetSize(&file_size);
-        read_segment_available_ = file_size - position_;
-        break;
-
-      case READ_MIDDLE:  // Finished READ_MIDDLE and start READ_LATEST.
-        read_segment_ = READ_LATEST;
-        position_ = marked_position_;
-        SetPosition(position_);
-        read_segment_available_ = last_write_position_ - position_;
-        break;
-
-      default:  // Finished READ_LATEST and return EOS.
-        return rtc::SR_EOS;
-    }
-  }
-
-  size_t local_read;
-  if (!read) read = &local_read;
-
-  size_t to_read = std::min(buffer_len, read_segment_available_);
-  rtc::StreamResult result
-    = rtc::FileStream::Read(buffer, to_read, read, error);
-  if (result == rtc::SR_SUCCESS) {
-    read_segment_available_ -= *read;
-    position_ += *read;
-  }
-  return result;
-}
-
-StreamResult CircularFileStream::Write(const void* data, size_t data_len,
-                                       size_t* written, int* error) {
-  if (position_ >= max_write_size_) {
-    ASSERT(position_ == max_write_size_);
-    position_ = marked_position_;
-    SetPosition(position_);
-  }
-
-  size_t local_written;
-  if (!written) written = &local_written;
-
-  size_t to_eof = max_write_size_ - position_;
-  size_t to_write = std::min(data_len, to_eof);
-  rtc::StreamResult result
-    = rtc::FileStream::Write(data, to_write, written, error);
-  if (result == rtc::SR_SUCCESS) {
-    position_ += *written;
-  }
-  return result;
-}
-
-AsyncWriteStream::AsyncWriteStream(StreamInterface* stream,
-                                   rtc::Thread* write_thread)
-    : stream_(stream),
-      write_thread_(write_thread),
-      state_(stream ? stream->GetState() : SS_CLOSED) {
-}
-
-AsyncWriteStream::~AsyncWriteStream() {
-  write_thread_->Clear(this, 0, NULL);
-  ClearBufferAndWrite();
-
-  CritScope cs(&crit_stream_);
-  stream_.reset();
-}
-
-StreamState AsyncWriteStream::GetState() const {
-  return state_;
-}
-
-// This is needed by some stream writers, such as RtpDumpWriter.
-bool AsyncWriteStream::GetPosition(size_t* position) const {
-  CritScope cs(&crit_stream_);
-  return stream_->GetPosition(position);
-}
-
-// This is needed by some stream writers, such as the plugin log writers.
-StreamResult AsyncWriteStream::Read(void* buffer, size_t buffer_len,
-                                    size_t* read, int* error) {
-  CritScope cs(&crit_stream_);
-  return stream_->Read(buffer, buffer_len, read, error);
-}
-
-void AsyncWriteStream::Close() {
-  if (state_ == SS_CLOSED) {
-    return;
-  }
-
-  write_thread_->Clear(this, 0, NULL);
-  ClearBufferAndWrite();
-
-  CritScope cs(&crit_stream_);
-  stream_->Close();
-  state_ = SS_CLOSED;
-}
-
-StreamResult AsyncWriteStream::Write(const void* data, size_t data_len,
-                                     size_t* written, int* error) {
-  if (state_ == SS_CLOSED) {
-    return SR_ERROR;
-  }
-
-  size_t previous_buffer_length = 0;
-  {
-    CritScope cs(&crit_buffer_);
-    previous_buffer_length = buffer_.size();
-    buffer_.AppendData(data, data_len);
-  }
-
-  if (previous_buffer_length == 0) {
-    // If there's stuff already in the buffer, then we already called
-    // Post and the write_thread_ hasn't pulled it out yet, so we
-    // don't need to re-Post.
-    write_thread_->Post(this, 0, NULL);
-  }
-  // Return immediately, assuming that it works.
-  if (written) {
-    *written = data_len;
-  }
-  return SR_SUCCESS;
-}
-
-void AsyncWriteStream::OnMessage(rtc::Message* pmsg) {
-  ClearBufferAndWrite();
-}
-
-bool AsyncWriteStream::Flush() {
-  if (state_ == SS_CLOSED) {
-    return false;
-  }
-
-  ClearBufferAndWrite();
-
-  CritScope cs(&crit_stream_);
-  return stream_->Flush();
-}
-
-void AsyncWriteStream::ClearBufferAndWrite() {
-  Buffer to_write;
-  {
-    CritScope cs_buffer(&crit_buffer_);
-    buffer_.TransferTo(&to_write);
-  }
-
-  if (to_write.size() > 0) {
-    CritScope cs(&crit_stream_);
-    stream_->WriteAll(to_write.data(), to_write.size(), NULL, NULL);
-  }
-}
-
-#if defined(WEBRTC_POSIX) && !defined(__native_client__)
-
-// Have to identically rewrite the FileStream destructor or else it would call
-// the base class's Close() instead of the sub-class's.
-POpenStream::~POpenStream() {
-  POpenStream::Close();
-}
-
-bool POpenStream::Open(const std::string& subcommand,
-                       const char* mode,
-                       int* error) {
-  Close();
-  file_ = popen(subcommand.c_str(), mode);
-  if (file_ == NULL) {
-    if (error)
-      *error = errno;
-    return false;
-  }
-  return true;
-}
-
-bool POpenStream::OpenShare(const std::string& subcommand, const char* mode,
-                            int shflag, int* error) {
-  return Open(subcommand, mode, error);
-}
-
-void POpenStream::DoClose() {
-  wait_status_ = pclose(file_);
-}
-
-#endif
-
 ///////////////////////////////////////////////////////////////////////////////
 // MemoryStream
 ///////////////////////////////////////////////////////////////////////////////
@@ -1276,8 +962,8 @@ void LoggingAdapter::OnEvent(StreamInterface* stream, int events, int err) {
 // StringStream - Reads/Writes to an external std::string
 ///////////////////////////////////////////////////////////////////////////////
 
-StringStream::StringStream(std::string& str)
-    : str_(str), read_pos_(0), read_only_(false) {
+StringStream::StringStream(std::string* str)
+    : str_(*str), read_pos_(0), read_only_(false) {
 }
 
 StringStream::StringStream(const std::string& str)
diff --git a/media/webrtc/trunk/webrtc/base/stream.h b/media/webrtc/trunk/webrtc/base/stream.h
index 2a29b2361e..c57daae76c 100644
--- a/media/webrtc/trunk/webrtc/base/stream.h
+++ b/media/webrtc/trunk/webrtc/base/stream.h
@@ -228,7 +228,7 @@ class StreamInterface : public MessageHandler {
   void OnMessage(Message* msg) override;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(StreamInterface);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamInterface);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -305,7 +305,7 @@ class StreamAdapterInterface : public StreamInterface,
  private:
   StreamInterface* stream_;
   bool owned_;
-  DISALLOW_EVIL_CONSTRUCTORS(StreamAdapterInterface);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamAdapterInterface);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -337,37 +337,7 @@ class StreamTap : public StreamAdapterInterface {
   scoped_ptr<StreamInterface> tap_;
   StreamResult tap_result_;
   int tap_error_;
-  DISALLOW_EVIL_CONSTRUCTORS(StreamTap);
-};
-
-///////////////////////////////////////////////////////////////////////////////
-// StreamSegment adapts a read stream, to expose a subset of the adapted
-// stream's data.  This is useful for cases where a stream contains multiple
-// documents concatenated together.  StreamSegment can expose a subset of
-// the data as an independent stream, including support for rewinding and
-// seeking.
-///////////////////////////////////////////////////////////////////////////////
-
-class StreamSegment : public StreamAdapterInterface {
- public:
-  // The current position of the adapted stream becomes the beginning of the
-  // segment.  If a length is specified, it bounds the length of the segment.
-  explicit StreamSegment(StreamInterface* stream);
-  explicit StreamSegment(StreamInterface* stream, size_t length);
-
-  // StreamAdapterInterface Interface
-  StreamResult Read(void* buffer,
-                    size_t buffer_len,
-                    size_t* read,
-                    int* error) override;
-  bool SetPosition(size_t position) override;
-  bool GetPosition(size_t* position) const override;
-  bool GetSize(size_t* size) const override;
-  bool GetAvailable(size_t* size) const override;
-
- private:
-  size_t start_, pos_, length_;
-  DISALLOW_EVIL_CONSTRUCTORS(StreamSegment);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamTap);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -445,113 +415,9 @@ class FileStream : public StreamInterface {
   FILE* file_;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(FileStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(FileStream);
 };
 
-// A stream that caps the output at a certain size, dropping content from the
-// middle of the logical stream and maintaining equal parts of the start/end of
-// the logical stream.
-class CircularFileStream : public FileStream {
- public:
-  explicit CircularFileStream(size_t max_size);
-
-  bool Open(const std::string& filename, const char* mode, int* error) override;
-  StreamResult Read(void* buffer,
-                    size_t buffer_len,
-                    size_t* read,
-                    int* error) override;
-  StreamResult Write(const void* data,
-                     size_t data_len,
-                     size_t* written,
-                     int* error) override;
-
- private:
-  enum ReadSegment {
-    READ_MARKED,  // Read 0 .. marked_position_
-    READ_MIDDLE,  // Read position_ .. file_size
-    READ_LATEST,  // Read marked_position_ .. position_ if the buffer was
-                  // overwritten or 0 .. position_ otherwise.
-  };
-
-  size_t max_write_size_;
-  size_t position_;
-  size_t marked_position_;
-  size_t last_write_position_;
-  ReadSegment read_segment_;
-  size_t read_segment_available_;
-};
-
-// A stream which pushes writes onto a separate thread and
-// returns from the write call immediately.
-class AsyncWriteStream : public StreamInterface {
- public:
-  // Takes ownership of the stream, but not the thread.
-  AsyncWriteStream(StreamInterface* stream, rtc::Thread* write_thread);
-  ~AsyncWriteStream() override;
-
-  // StreamInterface Interface
-  StreamState GetState() const override;
-  // This is needed by some stream writers, such as RtpDumpWriter.
-  bool GetPosition(size_t* position) const override;
-  StreamResult Read(void* buffer,
-                    size_t buffer_len,
-                    size_t* read,
-                    int* error) override;
-  StreamResult Write(const void* data,
-                     size_t data_len,
-                     size_t* written,
-                     int* error) override;
-  void Close() override;
-  bool Flush() override;
-
- protected:
-  // From MessageHandler
-  void OnMessage(rtc::Message* pmsg) override;
-  virtual void ClearBufferAndWrite();
-
- private:
-  rtc::scoped_ptr<StreamInterface> stream_;
-  Thread* write_thread_;
-  StreamState state_;
-  Buffer buffer_;
-  mutable CriticalSection crit_stream_;
-  CriticalSection crit_buffer_;
-
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncWriteStream);
-};
-
-
-#if defined(WEBRTC_POSIX) && !defined(__native_client__)
-// A FileStream that is actually not a file, but the output or input of a
-// sub-command. See "man 3 popen" for documentation of the underlying OS popen()
-// function.
-class POpenStream : public FileStream {
- public:
-  POpenStream() : wait_status_(-1) {}
-  ~POpenStream() override;
-
-  bool Open(const std::string& subcommand,
-            const char* mode,
-            int* error) override;
-  // Same as Open(). shflag is ignored.
-  bool OpenShare(const std::string& subcommand,
-                 const char* mode,
-                 int shflag,
-                 int* error) override;
-
-  // Returns the wait status from the last Close() of an Open()'ed stream, or
-  // -1 if no Open()+Close() has been done on this object. Meaning of the number
-  // is documented in "man 2 wait".
-  int GetWaitStatus() const { return wait_status_; }
-
- protected:
-  void DoClose() override;
-
- private:
-  int wait_status_;
-};
-#endif  // WEBRTC_POSIX
-
 ///////////////////////////////////////////////////////////////////////////////
 // MemoryStream is a simple implementation of a StreamInterface over in-memory
 // data.  Data is read and written at the current seek position.  Reads return
@@ -592,7 +458,7 @@ class MemoryStreamBase : public StreamInterface {
   size_t seek_position_;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(MemoryStreamBase);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MemoryStreamBase);
 };
 
 // MemoryStream dynamically resizes to accomodate written data.
@@ -690,7 +556,7 @@ class FifoBuffer : public StreamInterface {
   size_t read_position_;  // offset to the readable data
   Thread* owner_;  // stream callbacks are dispatched on this thread
   mutable CriticalSection crit_;  // object lock
-  DISALLOW_EVIL_CONSTRUCTORS(FifoBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(FifoBuffer);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -721,7 +587,7 @@ class LoggingAdapter : public StreamAdapterInterface {
   bool hex_mode_;
   LogMultilineState lms_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(LoggingAdapter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LoggingAdapter);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
@@ -730,7 +596,7 @@ class LoggingAdapter : public StreamAdapterInterface {
 
 class StringStream : public StreamInterface {
  public:
-  explicit StringStream(std::string& str);
+  explicit StringStream(std::string* str);
   explicit StringStream(const std::string& str);
 
   StreamState GetState() const override;
@@ -804,7 +670,7 @@ class StreamReference : public StreamAdapterInterface {
     StreamInterface* stream_;
     int ref_count_;
     CriticalSection cs_;
-    DISALLOW_EVIL_CONSTRUCTORS(StreamRefCount);
+    RTC_DISALLOW_COPY_AND_ASSIGN(StreamRefCount);
   };
 
   // Constructor for adding references
@@ -812,7 +678,7 @@ class StreamReference : public StreamAdapterInterface {
                            StreamInterface* stream);
 
   StreamRefCount* stream_ref_count_;
-  DISALLOW_EVIL_CONSTRUCTORS(StreamReference);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamReference);
 };
 
 ///////////////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/base/stream_unittest.cc b/media/webrtc/trunk/webrtc/base/stream_unittest.cc
index e31b092c45..8cfd052fe5 100644
--- a/media/webrtc/trunk/webrtc/base/stream_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/stream_unittest.cc
@@ -8,15 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/fileutils.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/pathutils.h"
 #include "webrtc/base/stream.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
-namespace {
-static const int kTimeoutMs = 10000;
-}  // namespace
 ///////////////////////////////////////////////////////////////////////////////
 // TestStream
 ///////////////////////////////////////////////////////////////////////////////
@@ -96,71 +94,6 @@ void SeekTest(StreamInterface* stream, const unsigned char value) {
   EXPECT_EQ(20U, bytes);
 }
 
-TEST(StreamSegment, TranslatesPosition) {
-  TestStream* test = new TestStream;
-  // Verify behavior of original stream
-  SeekTest(test, 0);
-  StreamSegment* segment = new StreamSegment(test);
-  // Verify behavior of adapted stream (all values offset by 20)
-  SeekTest(segment, 20);
-  delete segment;
-}
-
-TEST(StreamSegment, SupportsArtificialTermination) {
-  TestStream* test = new TestStream;
-
-  size_t bytes;
-  unsigned char buffer[5000] = { 0 };
-  const size_t kBufSize = sizeof(buffer);
-
-  {
-    StreamInterface* stream = test;
-
-    // Read a lot of bytes
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_SUCCESS);
-    EXPECT_EQ(bytes, kBufSize);
-    EXPECT_TRUE(VerifyTestBuffer(buffer, kBufSize, 0));
-
-    // Test seeking far ahead
-    EXPECT_TRUE(stream->SetPosition(12345));
-
-    // Read a bunch more bytes
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_SUCCESS);
-    EXPECT_EQ(bytes, kBufSize);
-    EXPECT_TRUE(VerifyTestBuffer(buffer, kBufSize, 12345 % 256));
-  }
-
-  // Create a segment of test stream in range [100,600)
-  EXPECT_TRUE(test->SetPosition(100));
-  StreamSegment* segment = new StreamSegment(test, 500);
-
-  {
-    StreamInterface* stream = segment;
-
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_SUCCESS);
-    EXPECT_EQ(500U, bytes);
-    EXPECT_TRUE(VerifyTestBuffer(buffer, 500, 100));
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_EOS);
-
-    // Test seeking past "end" of stream
-    EXPECT_FALSE(stream->SetPosition(12345));
-    EXPECT_FALSE(stream->SetPosition(501));
-
-    // Test seeking to end (edge case)
-    EXPECT_TRUE(stream->SetPosition(500));
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_EOS);
-
-    // Test seeking to start
-    EXPECT_TRUE(stream->SetPosition(0));
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_SUCCESS);
-    EXPECT_EQ(500U, bytes);
-    EXPECT_TRUE(VerifyTestBuffer(buffer, 500, 100));
-    EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, NULL), SR_EOS);
-  }
-
-  delete segment;
-}
-
 TEST(FifoBufferTest, TestAll) {
   const size_t kSize = 16;
   const char in[kSize * 2 + 1] = "0123456789ABCDEFGHIJKLMNOPQRSTUV";
@@ -438,60 +371,4 @@ TEST(FifoBufferTest, WriteOffsetAndReadOffset) {
   EXPECT_EQ(SR_BLOCK, buf.ReadOffset(out, 10, 16, NULL));
 }
 
-TEST(AsyncWriteTest, TestWrite) {
-  FifoBuffer* buf = new FifoBuffer(100);
-  AsyncWriteStream stream(buf, Thread::Current());
-  EXPECT_EQ(SS_OPEN, stream.GetState());
-
-  // Write "abc".  Will go to the logging thread, which is the current
-  // thread.
-  stream.Write("abc", 3, NULL, NULL);
-  char bytes[100];
-  size_t count;
-  // Messages on the thread's queue haven't been processed, so "abc"
-  // hasn't been written yet.
-  EXPECT_NE(SR_SUCCESS, buf->ReadOffset(&bytes, 3, 0, &count));
-  // Now we process the messages on the thread's queue, so "abc" has
-  // been written.
-  EXPECT_TRUE_WAIT(SR_SUCCESS == buf->ReadOffset(&bytes, 3, 0, &count),
-                   kTimeoutMs);
-  EXPECT_EQ(3u, count);
-  EXPECT_EQ(0, memcmp(bytes, "abc", 3));
-
-  // Write "def".  Will go to the logging thread, which is the current
-  // thread.
-  stream.Write("d", 1, &count, NULL);
-  stream.Write("e", 1, &count, NULL);
-  stream.Write("f", 1, &count, NULL);
-  EXPECT_EQ(1u, count);
-  // Messages on the thread's queue haven't been processed, so "def"
-  // hasn't been written yet.
-  EXPECT_NE(SR_SUCCESS, buf->ReadOffset(&bytes, 3, 3, &count));
-  // Flush() causes the message to be processed, so "def" has now been
-  // written.
-  stream.Flush();
-  EXPECT_EQ(SR_SUCCESS, buf->ReadOffset(&bytes, 3, 3, &count));
-  EXPECT_EQ(3u, count);
-  EXPECT_EQ(0, memcmp(bytes, "def", 3));
-
-  // Write "xyz".  Will go to the logging thread, which is the current
-  // thread.
-  stream.Write("xyz", 3, &count, NULL);
-  EXPECT_EQ(3u, count);
-  // Messages on the thread's queue haven't been processed, so "xyz"
-  // hasn't been written yet.
-  EXPECT_NE(SR_SUCCESS, buf->ReadOffset(&bytes, 3, 6, &count));
-  // Close() causes the message to be processed, so "xyz" has now been
-  // written.
-  stream.Close();
-  EXPECT_EQ(SR_SUCCESS, buf->ReadOffset(&bytes, 3, 6, &count));
-  EXPECT_EQ(3u, count);
-  EXPECT_EQ(0, memcmp(bytes, "xyz", 3));
-  EXPECT_EQ(SS_CLOSED, stream.GetState());
-
-  // Is't closed, so the writes should fail.
-  EXPECT_EQ(SR_ERROR, stream.Write("000", 3, NULL, NULL));
-
-}
-
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/stringencode.cc b/media/webrtc/trunk/webrtc/base/stringencode.cc
index 52b75da705..01b41a633a 100644
--- a/media/webrtc/trunk/webrtc/base/stringencode.cc
+++ b/media/webrtc/trunk/webrtc/base/stringencode.cc
@@ -26,7 +26,7 @@ namespace rtc {
 size_t escape(char * buffer, size_t buflen,
               const char * source, size_t srclen,
               const char * illegal, char escape) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen <= 0)
     return 0;
 
@@ -48,7 +48,7 @@ size_t escape(char * buffer, size_t buflen,
 size_t unescape(char * buffer, size_t buflen,
                 const char * source, size_t srclen,
                 char escape) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen <= 0)
     return 0;
 
@@ -67,7 +67,7 @@ size_t unescape(char * buffer, size_t buflen,
 size_t encode(char * buffer, size_t buflen,
               const char * source, size_t srclen,
               const char * illegal, char escape) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen <= 0)
     return 0;
 
@@ -119,8 +119,8 @@ const char* unsafe_filename_characters() {
 #if defined(WEBRTC_WIN)
   return "\\/:*?\"<>|";
 #else  // !WEBRTC_WIN
-  // TODO
-  DCHECK(false);
+  // TODO(grunell): Should this never be reached?
+  RTC_DCHECK(false);
   return "";
 #endif  // !WEBRTC_WIN
 }
@@ -257,7 +257,7 @@ size_t utf8_encode(char* buffer, size_t buflen, unsigned long value) {
 
 size_t html_encode(char * buffer, size_t buflen,
                    const char * source, size_t srclen) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen <= 0)
     return 0;
 
@@ -275,7 +275,7 @@ size_t html_encode(char * buffer, size_t buflen,
           case '\'': escseq = "&#39;";  esclen = 5; break;
           case '\"': escseq = "&quot;"; esclen = 6; break;
           case '&':  escseq = "&amp;";  esclen = 5; break;
-          default: DCHECK(false);
+          default: RTC_DCHECK(false);
         }
         if (bufpos + esclen >= buflen) {
           break;
@@ -310,13 +310,13 @@ size_t html_encode(char * buffer, size_t buflen,
 
 size_t html_decode(char * buffer, size_t buflen,
                    const char * source, size_t srclen) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   return xml_decode(buffer, buflen, source, srclen);
 }
 
 size_t xml_encode(char * buffer, size_t buflen,
                   const char * source, size_t srclen) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen <= 0)
     return 0;
 
@@ -332,7 +332,7 @@ size_t xml_encode(char * buffer, size_t buflen,
         case '\'': escseq = "&apos;"; esclen = 6; break;
         case '\"': escseq = "&quot;"; esclen = 6; break;
         case '&':  escseq = "&amp;";  esclen = 5; break;
-        default: DCHECK(false);
+        default: RTC_DCHECK(false);
       }
       if (bufpos + esclen >= buflen) {
         break;
@@ -349,7 +349,7 @@ size_t xml_encode(char * buffer, size_t buflen,
 
 size_t xml_decode(char * buffer, size_t buflen,
                   const char * source, size_t srclen) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen <= 0)
     return 0;
 
@@ -385,7 +385,7 @@ size_t xml_decode(char * buffer, size_t buflen,
         srcpos += 1;
       }
       char * ptr;
-      // TODO: Fix hack (ptr may go past end of data)
+      // TODO(grunell): Fix hack (ptr may go past end of data)
       unsigned long val = strtoul(source + srcpos + 1, &ptr, int_base);
       if ((static_cast<size_t>(ptr - source) < srclen) && (*ptr == ';')) {
         srcpos = ptr - source + 1;
@@ -411,7 +411,7 @@ size_t xml_decode(char * buffer, size_t buflen,
 static const char HEX[] = "0123456789abcdef";
 
 char hex_encode(unsigned char val) {
-  DCHECK_LT(val, 16);
+  RTC_DCHECK_LT(val, 16);
   return (val < 16) ? HEX[val] : '!';
 }
 
@@ -436,7 +436,7 @@ size_t hex_encode(char* buffer, size_t buflen,
 size_t hex_encode_with_delimiter(char* buffer, size_t buflen,
                                  const char* csource, size_t srclen,
                                  char delimiter) {
-  DCHECK(buffer);  // TODO: estimate output size
+  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
   if (buflen == 0)
     return 0;
 
@@ -466,6 +466,10 @@ size_t hex_encode_with_delimiter(char* buffer, size_t buflen,
   return bufpos;
 }
 
+std::string hex_encode(const std::string& str) {
+  return hex_encode(str.c_str(), str.size());
+}
+
 std::string hex_encode(const char* source, size_t srclen) {
   return hex_encode_with_delimiter(source, srclen, 0);
 }
@@ -476,7 +480,7 @@ std::string hex_encode_with_delimiter(const char* source, size_t srclen,
   char* buffer = STACK_ARRAY(char, kBufferSize);
   size_t length = hex_encode_with_delimiter(buffer, kBufferSize,
                                             source, srclen, delimiter);
-  DCHECK(srclen == 0 || length > 0);
+  RTC_DCHECK(srclen == 0 || length > 0);
   return std::string(buffer, length);
 }
 
@@ -488,7 +492,7 @@ size_t hex_decode(char * cbuffer, size_t buflen,
 size_t hex_decode_with_delimiter(char* cbuffer, size_t buflen,
                                  const char* source, size_t srclen,
                                  char delimiter) {
-  DCHECK(cbuffer);  // TODO: estimate output size
+  RTC_DCHECK(cbuffer);  // TODO(grunell): estimate output size
   if (buflen == 0)
     return 0;
 
@@ -552,7 +556,6 @@ std::string s_transform(const std::string& source, Transform t) {
 
 size_t tokenize(const std::string& source, char delimiter,
                 std::vector<std::string>* fields) {
-  DCHECK(fields);
   fields->clear();
   size_t last = 0;
   for (size_t i = 0; i < source.length(); ++i) {
@@ -569,6 +572,21 @@ size_t tokenize(const std::string& source, char delimiter,
   return fields->size();
 }
 
+size_t tokenize_with_empty_tokens(const std::string& source,
+                                  char delimiter,
+                                  std::vector<std::string>* fields) {
+  fields->clear();
+  size_t last = 0;
+  for (size_t i = 0; i < source.length(); ++i) {
+    if (source[i] == delimiter) {
+      fields->push_back(source.substr(last, i - last));
+      last = i + 1;
+    }
+  }
+  fields->push_back(source.substr(last, source.length() - last));
+  return fields->size();
+}
+
 size_t tokenize_append(const std::string& source, char delimiter,
                        std::vector<std::string>* fields) {
   if (!fields) return 0;
@@ -607,9 +625,30 @@ size_t tokenize(const std::string& source, char delimiter, char start_mark,
   return tokenize_append(remain_source, delimiter, fields);
 }
 
+bool tokenize_first(const std::string& source,
+                    const char delimiter,
+                    std::string* token,
+                    std::string* rest) {
+  // Find the first delimiter
+  size_t left_pos = source.find(delimiter);
+  if (left_pos == std::string::npos) {
+    return false;
+  }
+
+  // Look for additional occurrances of delimiter.
+  size_t right_pos = left_pos + 1;
+  while (source[right_pos] == delimiter) {
+    right_pos++;
+  }
+
+  *token = source.substr(0, left_pos);
+  *rest = source.substr(right_pos);
+  return true;
+}
+
 size_t split(const std::string& source, char delimiter,
              std::vector<std::string>* fields) {
-  DCHECK(fields);
+  RTC_DCHECK(fields);
   fields->clear();
   size_t last = 0;
   for (size_t i = 0; i < source.length(); ++i) {
diff --git a/media/webrtc/trunk/webrtc/base/stringencode.h b/media/webrtc/trunk/webrtc/base/stringencode.h
index 2e69a9c021..8f78ad1a64 100644
--- a/media/webrtc/trunk/webrtc/base/stringencode.h
+++ b/media/webrtc/trunk/webrtc/base/stringencode.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_BASE_STRINGENCODE_H_
 #define WEBRTC_BASE_STRINGENCODE_H_
 
-#include <string>
 #include <sstream>
+#include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
@@ -95,6 +95,7 @@ size_t hex_encode_with_delimiter(char* buffer, size_t buflen,
                                  char delimiter);
 
 // Helper functions for hex_encode.
+std::string hex_encode(const std::string& str);
 std::string hex_encode(const char* source, size_t srclen);
 std::string hex_encode_with_delimiter(const char* source, size_t srclen,
                                       char delimiter);
@@ -145,6 +146,11 @@ size_t split(const std::string& source, char delimiter,
 size_t tokenize(const std::string& source, char delimiter,
                 std::vector<std::string>* fields);
 
+// Tokenize, including the empty tokens.
+size_t tokenize_with_empty_tokens(const std::string& source,
+                                  char delimiter,
+                                  std::vector<std::string>* fields);
+
 // Tokenize and append the tokens to fields. Return the new size of fields.
 size_t tokenize_append(const std::string& source, char delimiter,
                        std::vector<std::string>* fields);
@@ -159,6 +165,14 @@ size_t tokenize_append(const std::string& source, char delimiter,
 size_t tokenize(const std::string& source, char delimiter, char start_mark,
                 char end_mark, std::vector<std::string>* fields);
 
+// Extract the first token from source as separated by delimiter, with
+// duplicates of delimiter ignored. Return false if the delimiter could not be
+// found, otherwise return true.
+bool tokenize_first(const std::string& source,
+                    const char delimiter,
+                    std::string* token,
+                    std::string* rest);
+
 // Safe sprintf to std::string
 //void sprintf(std::string& value, size_t maxlen, const char * format, ...)
 //     PRINTF_FORMAT(3);
@@ -167,7 +181,7 @@ size_t tokenize(const std::string& source, char delimiter, char start_mark,
 
 template <class T>
 static bool ToString(const T &t, std::string* s) {
-  DCHECK(s);
+  RTC_DCHECK(s);
   std::ostringstream oss;
   oss << std::boolalpha << t;
   *s = oss.str();
@@ -176,7 +190,7 @@ static bool ToString(const T &t, std::string* s) {
 
 template <class T>
 static bool FromString(const std::string& s, T* t) {
-  DCHECK(t);
+  RTC_DCHECK(t);
   std::istringstream iss(s);
   iss >> std::boolalpha >> *t;
   return !iss.fail();
diff --git a/media/webrtc/trunk/webrtc/base/stringencode_unittest.cc b/media/webrtc/trunk/webrtc/base/stringencode_unittest.cc
index c9e726ecb5..588e9d8ff5 100644
--- a/media/webrtc/trunk/webrtc/base/stringencode_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/stringencode_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/stringencode.h"
@@ -48,7 +49,7 @@ TEST(Utf8EncodeTest, EncodeDecode) {
     }
 
     char buffer[5];
-    memset(buffer, 0x01, ARRAY_SIZE(buffer));
+    memset(buffer, 0x01, arraysize(buffer));
     ASSERT_EQ(kTests[i].enclen, utf8_encode(buffer,
                                             kTests[i].encsize,
                                             kTests[i].decoded));
@@ -56,7 +57,7 @@ TEST(Utf8EncodeTest, EncodeDecode) {
     // Make sure remainder of buffer is unchanged
     ASSERT_TRUE(memory_check(buffer + kTests[i].enclen,
                              0x1,
-                             ARRAY_SIZE(buffer) - kTests[i].enclen));
+                             arraysize(buffer) - kTests[i].enclen));
   }
 }
 
@@ -298,6 +299,68 @@ TEST(TokenizeTest, TokenizeWithMarks) {
   ASSERT_STREQ("E F", fields.at(3).c_str());
 }
 
+TEST(TokenizeTest, TokenizeWithEmptyTokens) {
+  std::vector<std::string> fields;
+  EXPECT_EQ(3ul, tokenize_with_empty_tokens("a.b.c", '.', &fields));
+  EXPECT_EQ("a", fields[0]);
+  EXPECT_EQ("b", fields[1]);
+  EXPECT_EQ("c", fields[2]);
+
+  EXPECT_EQ(3ul, tokenize_with_empty_tokens("..c", '.', &fields));
+  EXPECT_TRUE(fields[0].empty());
+  EXPECT_TRUE(fields[1].empty());
+  EXPECT_EQ("c", fields[2]);
+
+  EXPECT_EQ(1ul, tokenize_with_empty_tokens("", '.', &fields));
+  EXPECT_TRUE(fields[0].empty());
+}
+
+TEST(TokenizeFirstTest, NoLeadingSpaces) {
+  std::string token;
+  std::string rest;
+
+  ASSERT_TRUE(tokenize_first("A &*${}", ' ', &token, &rest));
+  ASSERT_STREQ("A", token.c_str());
+  ASSERT_STREQ("&*${}", rest.c_str());
+
+  ASSERT_TRUE(tokenize_first("A B& *${}", ' ', &token, &rest));
+  ASSERT_STREQ("A", token.c_str());
+  ASSERT_STREQ("B& *${}", rest.c_str());
+
+  ASSERT_TRUE(tokenize_first("A    B& *${}    ", ' ', &token, &rest));
+  ASSERT_STREQ("A", token.c_str());
+  ASSERT_STREQ("B& *${}    ", rest.c_str());
+}
+
+TEST(TokenizeFirstTest, LeadingSpaces) {
+  std::string token;
+  std::string rest;
+
+  ASSERT_TRUE(tokenize_first("     A B C", ' ', &token, &rest));
+  ASSERT_STREQ("", token.c_str());
+  ASSERT_STREQ("A B C", rest.c_str());
+
+  ASSERT_TRUE(tokenize_first("     A    B   C    ", ' ', &token, &rest));
+  ASSERT_STREQ("", token.c_str());
+  ASSERT_STREQ("A    B   C    ", rest.c_str());
+}
+
+TEST(TokenizeFirstTest, SingleToken) {
+  std::string token;
+  std::string rest;
+
+  // In the case where we cannot find delimiter the whole string is a token.
+  ASSERT_FALSE(tokenize_first("ABC", ' ', &token, &rest));
+
+  ASSERT_TRUE(tokenize_first("ABC    ", ' ', &token, &rest));
+  ASSERT_STREQ("ABC", token.c_str());
+  ASSERT_STREQ("", rest.c_str());
+
+  ASSERT_TRUE(tokenize_first("    ABC    ", ' ', &token, &rest));
+  ASSERT_STREQ("", token.c_str());
+  ASSERT_STREQ("ABC    ", rest.c_str());
+}
+
 // Tests counting substrings.
 TEST(SplitTest, CountSubstrings) {
   std::vector<std::string> fields;
@@ -382,4 +445,5 @@ TEST(BoolTest, RoundTrip) {
   EXPECT_TRUE(FromString(ToString(false), &value));
   EXPECT_FALSE(value);
 }
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/stringutils.cc b/media/webrtc/trunk/webrtc/base/stringutils.cc
index cb99c25489..9580253d1b 100644
--- a/media/webrtc/trunk/webrtc/base/stringutils.cc
+++ b/media/webrtc/trunk/webrtc/base/stringutils.cc
@@ -57,7 +57,7 @@ int ascii_string_compare(const wchar_t* s1, const char* s2, size_t n,
     if (n-- == 0) return 0;
     c1 = transformation(*s1);
     // Double check that characters are not UTF-8
-    DCHECK_LT(static_cast<unsigned char>(*s2), 128);
+    RTC_DCHECK_LT(static_cast<unsigned char>(*s2), 128);
     // Note: *s2 gets implicitly promoted to wchar_t
     c2 = transformation(*s2);
     if (c1 != c2) return (c1 < c2) ? -1 : 1;
@@ -77,11 +77,11 @@ size_t asccpyn(wchar_t* buffer, size_t buflen,
   } else if (srclen >= buflen) {
     srclen = buflen - 1;
   }
-#if _DEBUG
+#if !defined(NDEBUG)
   // Double check that characters are not UTF-8
   for (size_t pos = 0; pos < srclen; ++pos)
-    DCHECK_LT(static_cast<unsigned char>(source[pos]), 128);
-#endif  // _DEBUG
+    RTC_DCHECK_LT(static_cast<unsigned char>(source[pos]), 128);
+#endif
   std::copy(source, source + srclen, buffer);
   buffer[srclen] = 0;
   return srclen;
diff --git a/media/webrtc/trunk/webrtc/base/stringutils.h b/media/webrtc/trunk/webrtc/base/stringutils.h
index 44b29d9acb..67ec335e01 100644
--- a/media/webrtc/trunk/webrtc/base/stringutils.h
+++ b/media/webrtc/trunk/webrtc/base/stringutils.h
@@ -292,7 +292,7 @@ struct Traits<char> {
 template<>
 struct Traits<wchar_t> {
   typedef std::wstring string;
-  inline static const wchar_t* Traits<wchar_t>::empty_str() { return L""; }
+  inline static const wchar_t* empty_str() { return L""; }
 };
 
 #endif  // WEBRTC_WIN 
diff --git a/media/webrtc/trunk/webrtc/base/systeminfo.cc b/media/webrtc/trunk/webrtc/base/systeminfo.cc
index 6d8d5ba751..b400aa08a8 100644
--- a/media/webrtc/trunk/webrtc/base/systeminfo.cc
+++ b/media/webrtc/trunk/webrtc/base/systeminfo.cc
@@ -12,6 +12,7 @@
 
 #if defined(WEBRTC_WIN)
 #include <winsock2.h>
+#include <windows.h>
 #ifndef EXCLUDE_D3D9
 #include <d3d9.h>
 #endif
@@ -26,14 +27,6 @@
 #include <sys/sysctl.h>
 #endif
 
-#if defined(WEBRTC_WIN)
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/win32.h"
-#elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-#include "webrtc/base/macconversion.h"
-#elif defined(WEBRTC_LINUX)
-#include "webrtc/base/linux.h"
-#endif
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
@@ -41,47 +34,7 @@
 namespace rtc {
 
 // See Also: http://msdn.microsoft.com/en-us/library/ms683194(v=vs.85).aspx
-#if defined(WEBRTC_WIN)
-typedef BOOL (WINAPI *LPFN_GLPI)(
-    PSYSTEM_LOGICAL_PROCESSOR_INFORMATION,
-    PDWORD);
-
-static void GetProcessorInformation(int* physical_cpus, int* cache_size) {
-  // GetLogicalProcessorInformation() is available on Windows XP SP3 and beyond.
-  LPFN_GLPI glpi = reinterpret_cast<LPFN_GLPI>(GetProcAddress(
-      GetModuleHandle(L"kernel32"),
-      "GetLogicalProcessorInformation"));
-  if (NULL == glpi) {
-    return;
-  }
-  // Determine buffer size, allocate and get processor information.
-  // Size can change between calls (unlikely), so a loop is done.
-  DWORD return_length = 0;
-  scoped_ptr<SYSTEM_LOGICAL_PROCESSOR_INFORMATION[]> infos;
-  while (!glpi(infos.get(), &return_length)) {
-    if (GetLastError() == ERROR_INSUFFICIENT_BUFFER) {
-      infos.reset(new SYSTEM_LOGICAL_PROCESSOR_INFORMATION[
-          return_length / sizeof(SYSTEM_LOGICAL_PROCESSOR_INFORMATION)]);
-    } else {
-      return;
-    }
-  }
-  *physical_cpus = 0;
-  *cache_size = 0;
-  for (size_t i = 0;
-      i < return_length / sizeof(SYSTEM_LOGICAL_PROCESSOR_INFORMATION); ++i) {
-    if (infos[i].Relationship == RelationProcessorCore) {
-      ++*physical_cpus;
-    } else if (infos[i].Relationship == RelationCache) {
-      int next_cache_size = static_cast<int>(infos[i].Cache.Size);
-      if (next_cache_size >= *cache_size) {
-        *cache_size = next_cache_size;
-      }
-    }
-  }
-  return;
-}
-#else
+#if !defined(WEBRTC_WIN)
 // TODO(fbarchard): Use gcc 4.4 provided cpuid intrinsic
 // 32 bit fpic requires ebx be preserved
 #if (defined(__pic__) || defined(__APPLE__)) && defined(__i386__)
@@ -103,134 +56,64 @@ static inline void __cpuid(int cpu_info[4], int info_type) {
   );  // NOLINT
 }
 #endif
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
-// Note(fbarchard):
-// Family and model are extended family and extended model.  8 bits each.
-SystemInfo::SystemInfo()
-    : physical_cpus_(1), logical_cpus_(1), cache_size_(0),
-      cpu_family_(0), cpu_model_(0), cpu_stepping_(0),
-      cpu_speed_(0), memory_(0) {
-  // Initialize the basic information.
-#if defined(__arm__) || defined(_M_ARM)
-  cpu_arch_ = SI_ARCH_ARM;
-#elif defined(__x86_64__) || defined(_M_X64)
-  cpu_arch_ = SI_ARCH_X64;
-#elif defined(__i386__) || defined(_M_IX86)
-  cpu_arch_ = SI_ARCH_X86;
-#else
-  cpu_arch_ = SI_ARCH_UNKNOWN;
-#endif
+static int DetectNumberOfCores() {
+  // We fall back on assuming a single core in case of errors.
+  int number_of_cores = 1;
 
 #if defined(WEBRTC_WIN)
   SYSTEM_INFO si;
   GetSystemInfo(&si);
-  logical_cpus_ = si.dwNumberOfProcessors;
-  GetProcessorInformation(&physical_cpus_, &cache_size_);
-  if (physical_cpus_ <= 0) {
-    physical_cpus_ = logical_cpus_;
-  }
-  cpu_family_ = si.wProcessorLevel;
-  cpu_model_ = si.wProcessorRevision >> 8;
-  cpu_stepping_ = si.wProcessorRevision & 0xFF;
+  number_of_cores = static_cast<int>(si.dwNumberOfProcessors);
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
+  number_of_cores = static_cast<int>(sysconf(_SC_NPROCESSORS_ONLN));
 #elif defined(WEBRTC_MAC)
-  uint32_t sysctl_value;
-  size_t length = sizeof(sysctl_value);
-  if (!sysctlbyname("hw.physicalcpu_max", &sysctl_value, &length, NULL, 0)) {
-    physical_cpus_ = static_cast<int>(sysctl_value);
-  }
-  length = sizeof(sysctl_value);
-  if (!sysctlbyname("hw.logicalcpu_max", &sysctl_value, &length, NULL, 0)) {
-    logical_cpus_ = static_cast<int>(sysctl_value);
-  }
-  uint64_t sysctl_value64;
-  length = sizeof(sysctl_value64);
-  if (!sysctlbyname("hw.l3cachesize", &sysctl_value64, &length, NULL, 0)) {
-    cache_size_ = static_cast<int>(sysctl_value64);
-  }
-  if (!cache_size_) {
-    length = sizeof(sysctl_value64);
-    if (!sysctlbyname("hw.l2cachesize", &sysctl_value64, &length, NULL, 0)) {
-      cache_size_ = static_cast<int>(sysctl_value64);
-    }
-  }
-  length = sizeof(sysctl_value);
-  if (!sysctlbyname("machdep.cpu.family", &sysctl_value, &length, NULL, 0)) {
-    cpu_family_ = static_cast<int>(sysctl_value);
-  }
-  length = sizeof(sysctl_value);
-  if (!sysctlbyname("machdep.cpu.model", &sysctl_value, &length, NULL, 0)) {
-    cpu_model_ = static_cast<int>(sysctl_value);
-  }
-  length = sizeof(sysctl_value);
-  if (!sysctlbyname("machdep.cpu.stepping", &sysctl_value, &length, NULL, 0)) {
-    cpu_stepping_ = static_cast<int>(sysctl_value);
-  }
-#elif defined(__native_client__)
-  // TODO(ryanpetrie): Implement this via PPAPI when it's available.
-#else  // WEBRTC_LINUX
-  ProcCpuInfo proc_info;
-  if (proc_info.LoadFromSystem()) {
-    proc_info.GetNumCpus(&logical_cpus_);
-    proc_info.GetNumPhysicalCpus(&physical_cpus_);
-    proc_info.GetCpuFamily(&cpu_family_);
-#if defined(CPU_X86)
-    // These values only apply to x86 systems.
-    proc_info.GetSectionIntValue(0, "model", &cpu_model_);
-    proc_info.GetSectionIntValue(0, "stepping", &cpu_stepping_);
-    proc_info.GetSectionIntValue(0, "cpu MHz", &cpu_speed_);
-    proc_info.GetSectionIntValue(0, "cache size", &cache_size_);
-    cache_size_ *= 1024;
-#endif
-  }
-  // ProcCpuInfo reads cpu speed from "cpu MHz" under /proc/cpuinfo.
-  // But that number is a moving target which can change on-the-fly according to
-  // many factors including system workload.
-  // See /sys/devices/system/cpu/cpu0/cpufreq/scaling_available_governors.
-  // The one in /sys/devices/system/cpu/cpu0/cpufreq/cpuinfo_max_freq is more
-  // accurate. We use it as our cpu speed when it is available.
-  // cpuinfo_max_freq is measured in KHz and requires conversion to MHz.
-  int max_freq = rtc::ReadCpuMaxFreq();
-  if (max_freq > 0) {
-    cpu_speed_ = max_freq / 1000;
-  }
-#endif
-// For L2 CacheSize see also
-// http://www.flounder.com/cpuid_explorer2.htm#CPUID(0x800000006)
-#ifdef CPU_X86
-  if (cache_size_ == 0) {
-    int cpu_info[4];
-    __cpuid(cpu_info, 0x80000000);  // query maximum extended cpuid function.
-    if (static_cast<uint32>(cpu_info[0]) >= 0x80000006) {
-      __cpuid(cpu_info, 0x80000006);
-      cache_size_ = (cpu_info[2] >> 16) * 1024;
-    }
+  int name[] = {CTL_HW, HW_AVAILCPU};
+  size_t size = sizeof(number_of_cores);
+  if (0 != sysctl(name, 2, &number_of_cores, &size, NULL, 0)) {
+    LOG(LS_ERROR) << "Failed to get number of cores";
+    number_of_cores = 1;
   }
+#else
+  LOG(LS_ERROR) << "No function to get number of cores";
 #endif
+
+  LOG(LS_INFO) << "Available number of cores: " << number_of_cores;
+
+  return number_of_cores;
+}
+
+// Statically cache the number of system cores available since if the process
+// is running in a sandbox, we may only be able to read the value once (before
+// the sandbox is initialized) and not thereafter.
+// For more information see crbug.com/176522.
+int SystemInfo::logical_cpus_ = 0;
+
+SystemInfo::SystemInfo() {
 }
 
 // Return the number of cpu threads available to the system.
+// static
 int SystemInfo::GetMaxCpus() {
+  if (!logical_cpus_)
+    logical_cpus_ = DetectNumberOfCores();
   return logical_cpus_;
 }
 
-// Return the number of cpu cores available to the system.
-int SystemInfo::GetMaxPhysicalCpus() {
-  return physical_cpus_;
-}
-
 // Return the number of cpus available to the process.  Since affinity can be
 // changed on the fly, do not cache this value.
 // Can be affected by heat.
 int SystemInfo::GetCurCpus() {
-  int cur_cpus;
+  int cur_cpus = 0;
 #if defined(WEBRTC_WIN)
-  DWORD_PTR process_mask, system_mask;
+  DWORD_PTR process_mask = 0;
+  DWORD_PTR system_mask = 0;
   ::GetProcessAffinityMask(::GetCurrentProcess(), &process_mask, &system_mask);
-  for (cur_cpus = 0; process_mask; ++cur_cpus) {
-    // Sparse-ones algorithm. There are slightly faster methods out there but
-    // they are unintuitive and won't make a difference on a single dword.
-    process_mask &= (process_mask - 1);
+  for (size_t i = 0; i < sizeof(DWORD_PTR) * 8; ++i) {
+    if (process_mask & 1)
+      ++cur_cpus;
+    process_mask >>= 1;
   }
 #elif defined(WEBRTC_MAC)
   uint32_t sysctl_value;
@@ -239,285 +122,92 @@ int SystemInfo::GetCurCpus() {
   cur_cpus = !error ? static_cast<int>(sysctl_value) : 1;
 #else
   // Linux, Solaris, WEBRTC_ANDROID
-  cur_cpus = static_cast<int>(sysconf(_SC_NPROCESSORS_ONLN));
+  cur_cpus = GetMaxCpus();
 #endif
   return cur_cpus;
 }
 
 // Return the type of this CPU.
 SystemInfo::Architecture SystemInfo::GetCpuArchitecture() {
-  return cpu_arch_;
+#if defined(__arm__) || defined(_M_ARM)
+  return SI_ARCH_ARM;
+#elif defined(__x86_64__) || defined(_M_X64)
+  return SI_ARCH_X64;
+#elif defined(__i386__) || defined(_M_IX86)
+  return SI_ARCH_X86;
+#else
+  return SI_ARCH_UNKNOWN;
+#endif
 }
 
 // Returns the vendor string from the cpu, e.g. "GenuineIntel", "AuthenticAMD".
 // See "Intel Processor Identification and the CPUID Instruction"
 // (Intel document number: 241618)
 std::string SystemInfo::GetCpuVendor() {
-  if (cpu_vendor_.empty()) {
 #if defined(CPU_X86)
-    int cpu_info[4];
-    __cpuid(cpu_info, 0);
-    cpu_info[0] = cpu_info[1];  // Reorder output
-    cpu_info[1] = cpu_info[3];
-    // cpu_info[2] = cpu_info[2];  // Avoid -Werror=self-assign
-    cpu_info[3] = 0;
-    cpu_vendor_ = std::string(reinterpret_cast<char*>(&cpu_info[0]));
+  int cpu_info[4];
+  __cpuid(cpu_info, 0);
+  cpu_info[0] = cpu_info[1];  // Reorder output
+  cpu_info[1] = cpu_info[3];
+  // cpu_info[2] = cpu_info[2];  // Avoid -Werror=self-assign
+  cpu_info[3] = 0;
+  return std::string(reinterpret_cast<char*>(&cpu_info[0]));
 #elif defined(CPU_ARM)
-    cpu_vendor_ = std::string("ARM");
+  return "ARM";
 #else
-    cpu_vendor_ = std::string("Undefined");
-#endif
-  }
-  return cpu_vendor_;
-}
-
-int SystemInfo::GetCpuCacheSize() {
-  return cache_size_;
-}
-
-// Return the "family" of this CPU.
-int SystemInfo::GetCpuFamily() {
-  return cpu_family_;
-}
-
-// Return the "model" of this CPU.
-int SystemInfo::GetCpuModel() {
-  return cpu_model_;
-}
-
-// Return the "stepping" of this CPU.
-int SystemInfo::GetCpuStepping() {
-  return cpu_stepping_;
-}
-
-// Return the clockrate of the primary processor in Mhz.  This value can be
-// cached.  Returns -1 on error.
-int SystemInfo::GetMaxCpuSpeed() {
-  if (cpu_speed_) {
-    return cpu_speed_;
-  }
-#if defined(WEBRTC_WIN)
-  HKEY key;
-  static const WCHAR keyName[] =
-      L"HARDWARE\\DESCRIPTION\\System\\CentralProcessor\\0";
-
-  if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, keyName , 0, KEY_QUERY_VALUE, &key)
-      == ERROR_SUCCESS) {
-    DWORD data, len;
-    len = sizeof(data);
-
-    if (RegQueryValueEx(key, L"~Mhz", 0, 0, reinterpret_cast<LPBYTE>(&data),
-                        &len) == ERROR_SUCCESS) {
-      cpu_speed_ = data;
-    } else {
-      LOG(LS_WARNING) << "Failed to query registry value HKLM\\" << keyName
-                      << "\\~Mhz";
-      cpu_speed_ = -1;
-    }
-
-    RegCloseKey(key);
-  } else {
-    LOG(LS_WARNING) << "Failed to open registry key HKLM\\" << keyName;
-    cpu_speed_ = -1;
-  }
-#elif defined(WEBRTC_MAC)
-  uint64_t sysctl_value;
-  size_t length = sizeof(sysctl_value);
-  int error = sysctlbyname("hw.cpufrequency_max", &sysctl_value, &length,
-                           NULL, 0);
-  cpu_speed_ = !error ? static_cast<int>(sysctl_value/1000000) : -1;
-#else
-  // TODO(fbarchard): Implement using proc/cpuinfo
-  cpu_speed_ = 0;
-#endif
-  return cpu_speed_;
-}
-
-// Dynamically check the current clockrate, which could be reduced because of
-// powersaving profiles.  Eventually for windows we want to query WMI for
-// root\WMI::ProcessorPerformance.InstanceName="Processor_Number_0".frequency
-int SystemInfo::GetCurCpuSpeed() {
-#if defined(WEBRTC_WIN)
-  // TODO(fbarchard): Add WMI check, requires COM initialization
-  // NOTE(fbarchard): Testable on Sandy Bridge.
-  return GetMaxCpuSpeed();
-#elif defined(WEBRTC_MAC)
-  uint64_t sysctl_value;
-  size_t length = sizeof(sysctl_value);
-  int error = sysctlbyname("hw.cpufrequency", &sysctl_value, &length, NULL, 0);
-  return !error ? static_cast<int>(sysctl_value/1000000) : GetMaxCpuSpeed();
-#else  // WEBRTC_LINUX
-  // TODO(fbarchard): Use proc/cpuinfo for Cur speed on Linux.
-  return GetMaxCpuSpeed();
+  return "Undefined";
 #endif
 }
 
 // Returns the amount of installed physical memory in Bytes.  Cacheable.
 // Returns -1 on error.
-int64 SystemInfo::GetMemorySize() {
-  if (memory_) {
-    return memory_;
-  }
+int64_t SystemInfo::GetMemorySize() {
+  int64_t memory = -1;
 
 #if defined(WEBRTC_WIN)
   MEMORYSTATUSEX status = {0};
   status.dwLength = sizeof(status);
 
   if (GlobalMemoryStatusEx(&status)) {
-    memory_ = status.ullTotalPhys;
+    memory = status.ullTotalPhys;
   } else {
     LOG_GLE(LS_WARNING) << "GlobalMemoryStatusEx failed.";
-    memory_ = -1;
   }
 
 #elif defined(WEBRTC_MAC)
-  size_t len = sizeof(memory_);
-  int error = sysctlbyname("hw.memsize", &memory_, &len, NULL, 0);
-  if (error || memory_ == 0) {
-    memory_ = -1;
-  }
-#else  // WEBRTC_LINUX
-  memory_ = static_cast<int64>(sysconf(_SC_PHYS_PAGES)) *
-      static_cast<int64>(sysconf(_SC_PAGESIZE));
-  if (memory_ < 0) {
+  size_t len = sizeof(memory);
+  int error = sysctlbyname("hw.memsize", &memory, &len, NULL, 0);
+  if (error || memory == 0)
+    memory = -1;
+#elif defined(WEBRTC_LINUX)
+  memory = static_cast<int64_t>(sysconf(_SC_PHYS_PAGES)) *
+           static_cast<int64_t>(sysconf(_SC_PAGESIZE));
+  if (memory < 0) {
     LOG(LS_WARNING) << "sysconf(_SC_PHYS_PAGES) failed."
                     << "sysconf(_SC_PHYS_PAGES) " << sysconf(_SC_PHYS_PAGES)
                     << "sysconf(_SC_PAGESIZE) " << sysconf(_SC_PAGESIZE);
-    memory_ = -1;
+    memory = -1;
   }
 #endif
 
-  return memory_;
+  return memory;
 }
 
-
 // Return the name of the machine model we are currently running on.
 // This is a human readable string that consists of the name and version
 // number of the hardware, i.e 'MacBookAir1,1'. Returns an empty string if
-// model can not be determined. The string is cached for subsequent calls.
+// model can not be determined.
 std::string SystemInfo::GetMachineModel() {
-  if (!machine_model_.empty()) {
-    return machine_model_;
-  }
-
 #if defined(WEBRTC_MAC)
   char buffer[128];
   size_t length = sizeof(buffer);
   int error = sysctlbyname("hw.model", buffer, &length, NULL, 0);
-  if (!error) {
-    machine_model_.assign(buffer, length - 1);
-  } else {
-    machine_model_.clear();
-  }
+  if (!error)
+    return std::string(buffer, length - 1);
+  return std::string();
 #else
-  machine_model_ = "Not available";
-#endif
-
-  return machine_model_;
-}
-
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-// Helper functions to query IOKit for video hardware properties.
-static CFTypeRef SearchForProperty(io_service_t port, CFStringRef name) {
-  return IORegistryEntrySearchCFProperty(port, kIOServicePlane,
-      name, kCFAllocatorDefault,
-      kIORegistryIterateRecursively | kIORegistryIterateParents);
-}
-
-static void GetProperty(io_service_t port, CFStringRef name, int* value) {
-  if (!value) return;
-  CFTypeRef ref = SearchForProperty(port, name);
-  if (ref) {
-    CFTypeID refType = CFGetTypeID(ref);
-    if (CFNumberGetTypeID() == refType) {
-      CFNumberRef number = reinterpret_cast<CFNumberRef>(ref);
-      p_convertCFNumberToInt(number, value);
-    } else if (CFDataGetTypeID() == refType) {
-      CFDataRef data = reinterpret_cast<CFDataRef>(ref);
-      if (CFDataGetLength(data) == sizeof(UInt32)) {
-        *value = *reinterpret_cast<const UInt32*>(CFDataGetBytePtr(data));
-      }
-    }
-    CFRelease(ref);
-  }
-}
-
-static void GetProperty(io_service_t port, CFStringRef name,
-                        std::string* value) {
-  if (!value) return;
-  CFTypeRef ref = SearchForProperty(port, name);
-  if (ref) {
-    CFTypeID refType = CFGetTypeID(ref);
-    if (CFStringGetTypeID() == refType) {
-      CFStringRef stringRef = reinterpret_cast<CFStringRef>(ref);
-      p_convertHostCFStringRefToCPPString(stringRef, *value);
-    } else if (CFDataGetTypeID() == refType) {
-      CFDataRef dataRef = reinterpret_cast<CFDataRef>(ref);
-      *value = std::string(reinterpret_cast<const char*>(
-          CFDataGetBytePtr(dataRef)), CFDataGetLength(dataRef));
-    }
-    CFRelease(ref);
-  }
-}
-#endif
-
-SystemInfo::GpuInfo::GpuInfo() : vendor_id(0), device_id(0) {
-}
-
-SystemInfo::GpuInfo::~GpuInfo() = default;
-
-// Fills a struct with information on the graphics adapater and returns true
-// iff successful.
-bool SystemInfo::GetGpuInfo(GpuInfo *info) {
-  if (!info) return false;
-#if defined(WEBRTC_WIN) && !defined(EXCLUDE_D3D9)
-  D3DADAPTER_IDENTIFIER9 identifier;
-  HRESULT hr = E_FAIL;
-  HINSTANCE d3d_lib = LoadLibrary(L"d3d9.dll");
-
-  if (d3d_lib) {
-    typedef IDirect3D9* (WINAPI *D3DCreate9Proc)(UINT);
-    D3DCreate9Proc d3d_create_proc = reinterpret_cast<D3DCreate9Proc>(
-        GetProcAddress(d3d_lib, "Direct3DCreate9"));
-    if (d3d_create_proc) {
-      IDirect3D9* d3d = d3d_create_proc(D3D_SDK_VERSION);
-      if (d3d) {
-        hr = d3d->GetAdapterIdentifier(D3DADAPTER_DEFAULT, 0, &identifier);
-        d3d->Release();
-      }
-    }
-    FreeLibrary(d3d_lib);
-  }
-
-  if (hr != D3D_OK) {
-    LOG(LS_ERROR) << "Failed to access Direct3D9 information.";
-    return false;
-  }
-
-  info->device_name = identifier.DeviceName;
-  info->description = identifier.Description;
-  info->vendor_id = identifier.VendorId;
-  info->device_id = identifier.DeviceId;
-  info->driver = identifier.Driver;
-  // driver_version format: product.version.subversion.build
-  std::stringstream ss;
-  ss << HIWORD(identifier.DriverVersion.HighPart) << "."
-     << LOWORD(identifier.DriverVersion.HighPart) << "."
-     << HIWORD(identifier.DriverVersion.LowPart) << "."
-     << LOWORD(identifier.DriverVersion.LowPart);
-  info->driver_version = ss.str();
-  return true;
-#elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-  // We'll query the IOKit for the gpu of the main display.
-  io_service_t display_service_port = CGDisplayIOServicePort(
-      kCGDirectMainDisplay);
-  GetProperty(display_service_port, CFSTR("vendor-id"), &info->vendor_id);
-  GetProperty(display_service_port, CFSTR("device-id"), &info->device_id);
-  GetProperty(display_service_port, CFSTR("model"), &info->description);
-  return true;
-#else  // WEBRTC_LINUX
-  // TODO(fbarchard): Implement this on Linux
-  return false;
+  return "Not available";
 #endif
 }
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/systeminfo.h b/media/webrtc/trunk/webrtc/base/systeminfo.h
index 47b93f6799..99d18b2960 100644
--- a/media/webrtc/trunk/webrtc/base/systeminfo.h
+++ b/media/webrtc/trunk/webrtc/base/systeminfo.h
@@ -28,53 +28,20 @@ class SystemInfo {
 
   SystemInfo();
 
-  // The number of CPU Cores in the system.
-  int GetMaxPhysicalCpus();
   // The number of CPU Threads in the system.
-  int GetMaxCpus();
+  static int GetMaxCpus();
   // The number of CPU Threads currently available to this process.
-  int GetCurCpus();
+  static int GetCurCpus();
   // Identity of the CPUs.
   Architecture GetCpuArchitecture();
   std::string GetCpuVendor();
-  int GetCpuFamily();
-  int GetCpuModel();
-  int GetCpuStepping();
-  // Return size of CPU cache in bytes.  Uses largest available cache (L3).
-  int GetCpuCacheSize();
-  // Estimated speed of the CPUs, in MHz.  e.g. 2400 for 2.4 GHz
-  int GetMaxCpuSpeed();
-  int GetCurCpuSpeed();
   // Total amount of physical memory, in bytes.
-  int64 GetMemorySize();
+  int64_t GetMemorySize();
   // The model name of the machine, e.g. "MacBookAir1,1"
   std::string GetMachineModel();
 
-  // The gpu identifier
-  struct GpuInfo {
-    GpuInfo();
-    ~GpuInfo();
-    std::string device_name;
-    std::string description;
-    int vendor_id;
-    int device_id;
-    std::string driver;
-    std::string driver_version;
-  };
-  bool GetGpuInfo(GpuInfo *info);
-
  private:
-  int physical_cpus_;
-  int logical_cpus_;
-  int cache_size_;
-  Architecture cpu_arch_;
-  std::string cpu_vendor_;
-  int cpu_family_;
-  int cpu_model_;
-  int cpu_stepping_;
-  int cpu_speed_;
-  int64 memory_;
-  std::string machine_model_;
+  static int logical_cpus_;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/systeminfo_unittest.cc b/media/webrtc/trunk/webrtc/base/systeminfo_unittest.cc
index fec553582a..b1fc65e091 100644
--- a/media/webrtc/trunk/webrtc/base/systeminfo_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/systeminfo_unittest.cc
@@ -52,14 +52,6 @@ TEST(SystemInfoTest, GetCpuArchitecture) {
 #endif
 }
 
-// Tests Cpu Cache Size
-TEST(SystemInfoTest, CpuCacheSize) {
-  rtc::SystemInfo info;
-  LOG(LS_INFO) << "CpuCacheSize: " << info.GetCpuCacheSize();
-  EXPECT_GE(info.GetCpuCacheSize(), 8192);  // 8 KB min cache
-  EXPECT_LE(info.GetCpuCacheSize(), 1024 * 1024 * 1024);  // 1 GB max cache
-}
-
 // Tests MachineModel is set.  On Mac test machine model is known.
 TEST(SystemInfoTest, MachineModelKnown) {
   rtc::SystemInfo info;
@@ -86,22 +78,6 @@ TEST(SystemInfoTest, MachineModelKnown) {
   }
 }
 
-// Tests maximum cpu clockrate.
-TEST(SystemInfoTest, CpuMaxCpuSpeed) {
-  rtc::SystemInfo info;
-  LOG(LS_INFO) << "MaxCpuSpeed: " << info.GetMaxCpuSpeed();
-  EXPECT_GT(info.GetMaxCpuSpeed(), 0);
-  EXPECT_LT(info.GetMaxCpuSpeed(), 100000);  // 100 Ghz
-}
-
-// Tests current cpu clockrate.
-TEST(SystemInfoTest, CpuCurCpuSpeed) {
-  rtc::SystemInfo info;
-  LOG(LS_INFO) << "MaxCurSpeed: " << info.GetCurCpuSpeed();
-  EXPECT_GT(info.GetCurCpuSpeed(), 0);
-  EXPECT_LT(info.GetMaxCpuSpeed(), 100000);
-}
-
 // Tests physical memory size.
 TEST(SystemInfoTest, MemorySize) {
   rtc::SystemInfo info;
@@ -116,14 +92,6 @@ TEST(SystemInfoTest, MaxCpus) {
   EXPECT_GT(info.GetMaxCpus(), 0);
 }
 
-// Tests number of physical cpus available to the system.
-TEST(SystemInfoTest, MaxPhysicalCpus) {
-  rtc::SystemInfo info;
-  LOG(LS_INFO) << "MaxPhysicalCpus: " << info.GetMaxPhysicalCpus();
-  EXPECT_GT(info.GetMaxPhysicalCpus(), 0);
-  EXPECT_LE(info.GetMaxPhysicalCpus(), info.GetMaxCpus());
-}
-
 // Tests number of logical cpus available to the process.
 TEST(SystemInfoTest, CurCpus) {
   rtc::SystemInfo info;
@@ -180,15 +148,3 @@ TEST(SystemInfoTest, CpuStepping) {
   EXPECT_EQ(0, info.GetCpuStepping());
 }
 #endif  // CPU_X86
-
-#if WEBRTC_WIN && !defined(EXCLUDE_D3D9)
-TEST(SystemInfoTest, GpuInfo) {
-  rtc::SystemInfo info;
-  rtc::SystemInfo::GpuInfo gi;
-  EXPECT_TRUE(info.GetGpuInfo(&gi));
-  LOG(LS_INFO) << "GpuDriver: " << gi.driver;
-  EXPECT_FALSE(gi.driver.empty());
-  LOG(LS_INFO) << "GpuDriverVersion: " << gi.driver_version;
-  EXPECT_FALSE(gi.driver_version.empty());
-}
-#endif
diff --git a/media/webrtc/trunk/webrtc/base/task.cc b/media/webrtc/trunk/webrtc/base/task.cc
index d81a6d2429..b09ced12b4 100644
--- a/media/webrtc/trunk/webrtc/base/task.cc
+++ b/media/webrtc/trunk/webrtc/base/task.cc
@@ -14,7 +14,7 @@
 
 namespace rtc {
 
-int32 Task::unique_id_seed_ = 0;
+int32_t Task::unique_id_seed_ = 0;
 
 Task::Task(TaskParent *parent)
     : TaskParent(this, parent),
@@ -48,11 +48,11 @@ Task::~Task() {
   }
 }
 
-int64 Task::CurrentTime() {
+int64_t Task::CurrentTime() {
   return GetRunner()->CurrentTime();
 }
 
-int64 Task::ElapsedTime() {
+int64_t Task::ElapsedTime() {
   return CurrentTime() - start_time_;
 }
 
@@ -68,7 +68,7 @@ void Task::Start() {
 
 void Task::Step() {
   if (done_) {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     // we do not know how !blocked_ happens when done_ - should be impossible.
     // But it causes problems, so in retail build, we force blocked_, and
     // under debug we assert.
@@ -88,7 +88,7 @@ void Task::Step() {
 //   SignalDone();
 
     Stop();
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     // verify that stop removed this from its parent
     ASSERT(!parent()->IsChildTask(this));
 #endif
@@ -125,7 +125,7 @@ void Task::Step() {
 //    SignalDone();
 
     Stop();
-#if _DEBUG
+#if !defined(NDEBUG)
     // verify that stop removed this from its parent
     ASSERT(!parent()->IsChildTask(this));
 #endif
@@ -150,7 +150,7 @@ void Task::Abort(bool nowake) {
     // "done_" is set before calling "Stop()" to ensure that this code 
     // doesn't execute more than once (recursively) for the same task.
     Stop();
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     // verify that stop removed this from its parent
     ASSERT(!parent()->IsChildTask(this));
 #endif
@@ -240,7 +240,7 @@ bool Task::TimedOut() {
 }
 
 void Task::ResetTimeout() {
-  int64 previous_timeout_time = timeout_time_;
+  int64_t previous_timeout_time = timeout_time_;
   bool timeout_allowed = (state_ != STATE_INIT)
                       && (state_ != STATE_DONE)
                       && (state_ != STATE_ERROR);
@@ -254,7 +254,7 @@ void Task::ResetTimeout() {
 }
 
 void Task::ClearTimeout() {
-  int64 previous_timeout_time = timeout_time_;
+  int64_t previous_timeout_time = timeout_time_;
   timeout_time_ = 0;
   GetRunner()->UpdateTaskTimeout(this, previous_timeout_time);
 }
diff --git a/media/webrtc/trunk/webrtc/base/task.h b/media/webrtc/trunk/webrtc/base/task.h
index 3e43e11dae..28702e49a7 100644
--- a/media/webrtc/trunk/webrtc/base/task.h
+++ b/media/webrtc/trunk/webrtc/base/task.h
@@ -95,7 +95,7 @@ class Task : public TaskParent {
   Task(TaskParent *parent);
   ~Task() override;
 
-  int32 unique_id() { return unique_id_; }
+  int32_t unique_id() { return unique_id_; }
 
   void Start();
   void Step();
@@ -103,14 +103,14 @@ class Task : public TaskParent {
   bool HasError() const { return (GetState() == STATE_ERROR); }
   bool Blocked() const { return blocked_; }
   bool IsDone() const { return done_; }
-  int64 ElapsedTime();
+  int64_t ElapsedTime();
 
   // Called from outside to stop task without any more callbacks
   void Abort(bool nowake = false);
 
   bool TimedOut();
 
-  int64 timeout_time() const { return timeout_time_; }
+  int64_t timeout_time() const { return timeout_time_; }
   int timeout_seconds() const { return timeout_seconds_; }
   void set_timeout_seconds(int timeout_seconds);
 
@@ -134,7 +134,7 @@ class Task : public TaskParent {
   // Called inside to advise that the task should wake and signal an error
   void Error();
 
-  int64 CurrentTime();
+  int64_t CurrentTime();
 
   virtual std::string GetStateName(int state) const;
   virtual int Process(int state);
@@ -160,13 +160,13 @@ class Task : public TaskParent {
   bool aborted_;
   bool busy_;
   bool error_;
-  int64 start_time_;
-  int64 timeout_time_;
+  int64_t start_time_;
+  int64_t timeout_time_;
   int timeout_seconds_;
   bool timeout_suspended_;
-  int32 unique_id_;
-  
-  static int32 unique_id_seed_;
+  int32_t unique_id_;
+
+  static int32_t unique_id_seed_;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/task_unittest.cc b/media/webrtc/trunk/webrtc/base/task_unittest.cc
index 5a3a60544a..7492436a5d 100644
--- a/media/webrtc/trunk/webrtc/base/task_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/task_unittest.cc
@@ -18,8 +18,9 @@
 
 #if defined(WEBRTC_WIN)
 #include "webrtc/base/win32.h"
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
@@ -27,12 +28,11 @@
 #include "webrtc/base/taskrunner.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace rtc {
 
-static int64 GetCurrentTime() {
-  return static_cast<int64>(Time()) * 10000;
+static int64_t GetCurrentTime() {
+  return static_cast<int64_t>(Time()) * 10000;
 }
 
 // feel free to change these numbers.  Note that '0' won't work, though
@@ -98,9 +98,7 @@ class HappyTask : public IdTimeoutTask {
 class MyTaskRunner : public TaskRunner {
  public:
   virtual void WakeTasks() { RunTasks(); }
-  virtual int64 CurrentTime() {
-    return GetCurrentTime();
-  }
+  virtual int64_t CurrentTime() { return GetCurrentTime(); }
 
   bool timeout_change() const {
     return timeout_change_;
@@ -270,7 +268,7 @@ class TaskTest : public sigslot::has_slots<> {
       EXPECT_TRUE(stuck_[i].timed_out_);
       if (!stuck_[i].timed_out_) {
         std::cout << "Stuck task #" << i << " timeout is at "
-            << stuck_[i].task_->timeout_time() << std::endl;        
+                  << stuck_[i].task_->timeout_time() << std::endl;
       }
     }
 
@@ -308,7 +306,7 @@ class AbortTask : public Task {
     return STATE_NEXT;
   }
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(AbortTask);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AbortTask);
 };
 
 class TaskAbortTest : public sigslot::has_slots<> {
@@ -333,7 +331,7 @@ class TaskAbortTest : public sigslot::has_slots<> {
   }
 
   MyTaskRunner task_runner_;
-  DISALLOW_EVIL_CONSTRUCTORS(TaskAbortTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TaskAbortTest);
 };
 
 TEST(start_task_test, Abort) {
@@ -363,7 +361,7 @@ class SetBoolOnDeleteTask : public Task {
 
  private:
   bool* set_when_deleted_;
-  DISALLOW_EVIL_CONSTRUCTORS(SetBoolOnDeleteTask);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SetBoolOnDeleteTask);
 };
 
 class AbortShouldWakeTest : public sigslot::has_slots<> {
@@ -396,7 +394,7 @@ class AbortShouldWakeTest : public sigslot::has_slots<> {
   }
 
   MyTaskRunner task_runner_;
-  DISALLOW_EVIL_CONSTRUCTORS(AbortShouldWakeTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AbortShouldWakeTest);
 };
 
 TEST(start_task_test, AbortShouldWake) {
@@ -410,7 +408,7 @@ TEST(start_task_test, AbortShouldWake) {
 class TimeoutChangeTest : public sigslot::has_slots<> {
  public:
   TimeoutChangeTest()
-    : task_count_(ARRAY_SIZE(stuck_tasks_)) {}
+    : task_count_(arraysize(stuck_tasks_)) {}
 
   // no need to delete any tasks; the task runner owns them
   ~TimeoutChangeTest() {}
@@ -465,7 +463,7 @@ class TimeoutChangeTest : public sigslot::has_slots<> {
 
  private:
   void OnTimeoutId(const int id) {
-    for (int i = 0; i < ARRAY_SIZE(stuck_tasks_); ++i) {
+    for (size_t i = 0; i < arraysize(stuck_tasks_); ++i) {
       if (stuck_tasks_[i] && stuck_tasks_[i]->unique_id() == id) {
         task_count_--;
         stuck_tasks_[i] = NULL;
@@ -477,7 +475,7 @@ class TimeoutChangeTest : public sigslot::has_slots<> {
   MyTaskRunner task_runner_;
   StuckTask* (stuck_tasks_[3]);
   int task_count_;
-  DISALLOW_EVIL_CONSTRUCTORS(TimeoutChangeTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TimeoutChangeTest);
 };
 
 TEST(start_task_test, TimeoutChange) {
@@ -490,11 +488,9 @@ class DeleteTestTaskRunner : public TaskRunner {
   DeleteTestTaskRunner() {
   }
   virtual void WakeTasks() { }
-  virtual int64 CurrentTime() {
-    return GetCurrentTime();
-  }
+  virtual int64_t CurrentTime() { return GetCurrentTime(); }
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(DeleteTestTaskRunner);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DeleteTestTaskRunner);
 };
 
 TEST(unstarted_task_test, DeleteTask) {
diff --git a/media/webrtc/trunk/webrtc/base/taskparent.cc b/media/webrtc/trunk/webrtc/base/taskparent.cc
index db6db37029..14d236dc42 100644
--- a/media/webrtc/trunk/webrtc/base/taskparent.cc
+++ b/media/webrtc/trunk/webrtc/base/taskparent.cc
@@ -46,7 +46,7 @@ void TaskParent::AddChild(Task *child) {
   children_->insert(child);
 }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 bool TaskParent::IsChildTask(Task *task) {
   ASSERT(task != NULL);
   return task->parent_ == this && children_->find(task) != children_->end();
@@ -69,7 +69,7 @@ bool TaskParent::AnyChildError() {
 
 void TaskParent::AbortAllChildren() {
   if (children_->size() > 0) {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     runner_->IncrementAbortCount();
 #endif
 
@@ -78,7 +78,7 @@ void TaskParent::AbortAllChildren() {
       (*it)->Abort(true);  // Note we do not wake
     }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     runner_->DecrementAbortCount();
 #endif
   }
diff --git a/media/webrtc/trunk/webrtc/base/taskparent.h b/media/webrtc/trunk/webrtc/base/taskparent.h
index f26d7970ff..41008fa98e 100644
--- a/media/webrtc/trunk/webrtc/base/taskparent.h
+++ b/media/webrtc/trunk/webrtc/base/taskparent.h
@@ -32,7 +32,7 @@ class TaskParent {
 
   bool AllChildrenDone();
   bool AnyChildError();
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   bool IsChildTask(Task *task);
 #endif
 
@@ -53,7 +53,7 @@ class TaskParent {
   bool child_error_;
   typedef std::set<Task *> ChildSet;
   scoped_ptr<ChildSet> children_;
-  DISALLOW_EVIL_CONSTRUCTORS(TaskParent);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TaskParent);
 };
 
 
diff --git a/media/webrtc/trunk/webrtc/base/taskrunner.cc b/media/webrtc/trunk/webrtc/base/taskrunner.cc
index bc4ab5e44f..c50c9f833e 100644
--- a/media/webrtc/trunk/webrtc/base/taskrunner.cc
+++ b/media/webrtc/trunk/webrtc/base/taskrunner.cc
@@ -23,7 +23,7 @@ TaskRunner::TaskRunner()
   : TaskParent(this),
     next_timeout_task_(NULL),
     tasks_running_(false)
-#ifdef _DEBUG
+#if !defined(NDEBUG)
     , abort_count_(0),
     deleting_task_(NULL)
 #endif
@@ -64,7 +64,7 @@ void TaskRunner::InternalRunTasks(bool in_destructor) {
 
   tasks_running_ = true;
 
-  int64 previous_timeout_time = next_task_timeout();
+  int64_t previous_timeout_time = next_task_timeout();
 
   int did_run = true;
   while (did_run) {
@@ -88,11 +88,11 @@ void TaskRunner::InternalRunTasks(bool in_destructor) {
         need_timeout_recalc = true;
       }
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
       deleting_task_ = task;
 #endif
       delete task;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
       deleting_task_ = NULL;
 #endif
       tasks_[i] = NULL;
@@ -135,7 +135,7 @@ void TaskRunner::PollTasks() {
   }
 }
 
-int64 TaskRunner::next_task_timeout() const {
+int64_t TaskRunner::next_task_timeout() const {
   if (next_timeout_task_) {
     return next_timeout_task_->timeout_time();
   }
@@ -150,9 +150,9 @@ int64 TaskRunner::next_task_timeout() const {
 // effectively making the task scheduler O-1 instead of O-N
 
 void TaskRunner::UpdateTaskTimeout(Task* task,
-                                   int64 previous_task_timeout_time) {
+                                   int64_t previous_task_timeout_time) {
   ASSERT(task != NULL);
-  int64 previous_timeout_time = next_task_timeout();
+  int64_t previous_timeout_time = next_task_timeout();
   bool task_is_timeout_task = next_timeout_task_ != NULL &&
       task->unique_id() == next_timeout_task_->unique_id();
   if (task_is_timeout_task) {
@@ -190,7 +190,7 @@ void TaskRunner::RecalcNextTimeout(Task *exclude_task) {
   //   we're not excluding it
   //   it has the closest timeout time
 
-  int64 next_timeout_time = 0;
+  int64_t next_timeout_time = 0;
   next_timeout_task_ = NULL;
 
   for (size_t i = 0; i < tasks_.size(); ++i) {
@@ -210,8 +210,8 @@ void TaskRunner::RecalcNextTimeout(Task *exclude_task) {
   }
 }
 
-void TaskRunner::CheckForTimeoutChange(int64 previous_timeout_time) {
-  int64 next_timeout = next_task_timeout();
+void TaskRunner::CheckForTimeoutChange(int64_t previous_timeout_time) {
+  int64_t next_timeout = next_task_timeout();
   bool timeout_change = (previous_timeout_time == 0 && next_timeout != 0) ||
       next_timeout < previous_timeout_time ||
       (previous_timeout_time <= CurrentTime() &&
diff --git a/media/webrtc/trunk/webrtc/base/taskrunner.h b/media/webrtc/trunk/webrtc/base/taskrunner.h
index bdcebc4bdc..e0cf17513a 100644
--- a/media/webrtc/trunk/webrtc/base/taskrunner.h
+++ b/media/webrtc/trunk/webrtc/base/taskrunner.h
@@ -20,9 +20,9 @@
 namespace rtc {
 class Task;
 
-const int64 kSecToMsec = 1000;
-const int64 kMsecTo100ns = 10000;
-const int64 kSecTo100ns = kSecToMsec * kMsecTo100ns;
+const int64_t kSecToMsec = 1000;
+const int64_t kMsecTo100ns = 10000;
+const int64_t kSecTo100ns = kSecToMsec * kMsecTo100ns;
 
 class TaskRunner : public TaskParent, public sigslot::has_slots<> {
  public:
@@ -36,15 +36,15 @@ class TaskRunner : public TaskParent, public sigslot::has_slots<> {
   // the units and that rollover while the computer is running.
   //
   // On Windows, GetSystemTimeAsFileTime is the typical implementation.
-  virtual int64 CurrentTime() = 0 ;
+  virtual int64_t CurrentTime() = 0;
 
   void StartTask(Task *task);
   void RunTasks();
   void PollTasks();
 
-  void UpdateTaskTimeout(Task *task, int64 previous_task_timeout_time);
+  void UpdateTaskTimeout(Task* task, int64_t previous_task_timeout_time);
 
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   bool is_ok_to_delete(Task* task) {
     return task == deleting_task_;
   }
@@ -60,7 +60,7 @@ class TaskRunner : public TaskParent, public sigslot::has_slots<> {
 
   // Returns the next absolute time when a task times out
   // OR "0" if there is no next timeout.
-  int64 next_task_timeout() const;
+  int64_t next_task_timeout() const;
 
  protected:
   // The primary usage of this method is to know if
@@ -82,12 +82,12 @@ class TaskRunner : public TaskParent, public sigslot::has_slots<> {
 
  private:
   void InternalRunTasks(bool in_destructor);
-  void CheckForTimeoutChange(int64 previous_timeout_time);
+  void CheckForTimeoutChange(int64_t previous_timeout_time);
 
   std::vector<Task *> tasks_;
   Task *next_timeout_task_;
   bool tasks_running_;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   int abort_count_;
   Task* deleting_task_;
 #endif
diff --git a/media/webrtc/trunk/webrtc/base/template_util.h b/media/webrtc/trunk/webrtc/base/template_util.h
index 86e541d8cf..31464cf35d 100644
--- a/media/webrtc/trunk/webrtc/base/template_util.h
+++ b/media/webrtc/trunk/webrtc/base/template_util.h
@@ -48,6 +48,19 @@ template <class T> struct is_non_const_reference<const T&> : false_type {};
 template <class T> struct is_void : false_type {};
 template <> struct is_void<void> : true_type {};
 
+template <class T>
+struct remove_reference {
+  typedef T type;
+};
+template <class T>
+struct remove_reference<T&> {
+  typedef T type;
+};
+template <class T>
+struct remove_reference<T&&> {
+  typedef T type;
+};
+
 namespace internal {
 
 // Types YesType and NoType are guaranteed such that sizeof(YesType) <
diff --git a/media/webrtc/trunk/webrtc/base/testclient.cc b/media/webrtc/trunk/webrtc/base/testclient.cc
index 8483c4e8f4..c7484fa541 100644
--- a/media/webrtc/trunk/webrtc/base/testclient.cc
+++ b/media/webrtc/trunk/webrtc/base/testclient.cc
@@ -34,7 +34,7 @@ TestClient::~TestClient() {
 
 bool TestClient::CheckConnState(AsyncPacketSocket::State state) {
   // Wait for our timeout value until the socket reaches the desired state.
-  uint32 end = TimeAfter(kTimeoutMs);
+  uint32_t end = TimeAfter(kTimeoutMs);
   while (socket_->GetState() != state && TimeUntil(end) > 0)
     Thread::Current()->ProcessMessages(1);
   return (socket_->GetState() == state);
@@ -63,7 +63,7 @@ TestClient::Packet* TestClient::NextPacket(int timeout_ms) {
   // Pumping another thread's queue could lead to messages being dispatched from
   // the wrong thread to non-thread-safe objects.
 
-  uint32 end = TimeAfter(timeout_ms);
+  uint32_t end = TimeAfter(timeout_ms);
   while (TimeUntil(end) > 0) {
     {
       CritScope cs(&crit_);
diff --git a/media/webrtc/trunk/webrtc/base/testclient.h b/media/webrtc/trunk/webrtc/base/testclient.h
index 8e692b6334..5d8ee98a6f 100644
--- a/media/webrtc/trunk/webrtc/base/testclient.h
+++ b/media/webrtc/trunk/webrtc/base/testclient.h
@@ -89,7 +89,7 @@ class TestClient : public sigslot::has_slots<> {
   AsyncPacketSocket* socket_;
   std::vector<Packet*>* packets_;
   bool ready_to_send_;
-  DISALLOW_EVIL_CONSTRUCTORS(TestClient);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TestClient);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/testclient_unittest.cc b/media/webrtc/trunk/webrtc/base/testclient_unittest.cc
index 1cb9a1a099..bdd06b329a 100644
--- a/media/webrtc/trunk/webrtc/base/testclient_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/testclient_unittest.cc
@@ -14,7 +14,6 @@
 #include "webrtc/base/testclient.h"
 #include "webrtc/base/testechoserver.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using namespace rtc;
 
@@ -56,7 +55,12 @@ TEST(TestClientTest, TestUdpIPv4) {
   TestUdpInternal(SocketAddress("127.0.0.1", 0));
 }
 
-TEST(TestClientTest, TestUdpIPv6) {
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestUdpIPv6 DISABLED_TestUdpIPv6
+#else
+#define MAYBE_TestUdpIPv6 TestUdpIPv6
+#endif
+TEST(TestClientTest, MAYBE_TestUdpIPv6) {
   if (HasIPv6Enabled()) {
     TestUdpInternal(SocketAddress("::1", 0));
   } else {
@@ -69,7 +73,12 @@ TEST(TestClientTest, TestTcpIPv4) {
   TestTcpInternal(SocketAddress("127.0.0.1", 0));
 }
 
-TEST(TestClientTest, TestTcpIPv6) {
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestTcpIPv6 DISABLED_TestTcpIPv6
+#else
+#define MAYBE_TestTcpIPv6 TestTcpIPv6
+#endif
+TEST(TestClientTest, MAYBE_TestTcpIPv6) {
   if (HasIPv6Enabled()) {
     TestTcpInternal(SocketAddress("::1", 0));
   } else {
diff --git a/media/webrtc/trunk/webrtc/base/testechoserver.h b/media/webrtc/trunk/webrtc/base/testechoserver.h
index 733b320dde..51d7d539e4 100644
--- a/media/webrtc/trunk/webrtc/base/testechoserver.h
+++ b/media/webrtc/trunk/webrtc/base/testechoserver.h
@@ -65,7 +65,7 @@ class TestEchoServer : public sigslot::has_slots<> {
   typedef std::list<AsyncTCPSocket*> ClientList;
   scoped_ptr<AsyncSocket> server_socket_;
   ClientList client_sockets_;
-  DISALLOW_EVIL_CONSTRUCTORS(TestEchoServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TestEchoServer);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/testutils.h b/media/webrtc/trunk/webrtc/base/testutils.h
index 4c978e7955..6e7e22a928 100644
--- a/media/webrtc/trunk/webrtc/base/testutils.h
+++ b/media/webrtc/trunk/webrtc/base/testutils.h
@@ -25,6 +25,7 @@
 #include <algorithm>
 #include <map>
 #include <vector>
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/asyncsocket.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
@@ -357,7 +358,7 @@ private:
   }
   void OnReadEvent(AsyncSocket* socket) {
     char data[64 * 1024];
-    int result = socket_->Recv(data, ARRAY_SIZE(data));
+    int result = socket_->Recv(data, arraysize(data));
     if (result > 0) {
       recv_buffer_.insert(recv_buffer_.end(), data, data + result);
     }
@@ -542,29 +543,33 @@ inline AssertionResult CmpHelperFileEq(const char* expected_expression,
 // order
 ///////////////////////////////////////////////////////////////////////////////
 
-#define BYTE_CAST(x) static_cast<uint8>((x) & 0xFF)
+#define BYTE_CAST(x) static_cast<uint8_t>((x)&0xFF)
 
 // Declare a N-bit integer as a little-endian sequence of bytes
-#define LE16(x) BYTE_CAST(((uint16)x) >>  0), BYTE_CAST(((uint16)x) >>  8)
+#define LE16(x) BYTE_CAST(((uint16_t)x) >> 0), BYTE_CAST(((uint16_t)x) >> 8)
 
-#define LE32(x) BYTE_CAST(((uint32)x) >>  0), BYTE_CAST(((uint32)x) >>  8), \
-                BYTE_CAST(((uint32)x) >> 16), BYTE_CAST(((uint32)x) >> 24)
+#define LE32(x) \
+  BYTE_CAST(((uint32_t)x) >> 0), BYTE_CAST(((uint32_t)x) >> 8), \
+      BYTE_CAST(((uint32_t)x) >> 16), BYTE_CAST(((uint32_t)x) >> 24)
 
-#define LE64(x) BYTE_CAST(((uint64)x) >>  0), BYTE_CAST(((uint64)x) >>  8), \
-                BYTE_CAST(((uint64)x) >> 16), BYTE_CAST(((uint64)x) >> 24), \
-                BYTE_CAST(((uint64)x) >> 32), BYTE_CAST(((uint64)x) >> 40), \
-                BYTE_CAST(((uint64)x) >> 48), BYTE_CAST(((uint64)x) >> 56)
+#define LE64(x) \
+  BYTE_CAST(((uint64_t)x) >> 0), BYTE_CAST(((uint64_t)x) >> 8),       \
+      BYTE_CAST(((uint64_t)x) >> 16), BYTE_CAST(((uint64_t)x) >> 24), \
+      BYTE_CAST(((uint64_t)x) >> 32), BYTE_CAST(((uint64_t)x) >> 40), \
+      BYTE_CAST(((uint64_t)x) >> 48), BYTE_CAST(((uint64_t)x) >> 56)
 
 // Declare a N-bit integer as a big-endian (Internet) sequence of bytes
-#define BE16(x) BYTE_CAST(((uint16)x) >>  8), BYTE_CAST(((uint16)x) >>  0)
+#define BE16(x) BYTE_CAST(((uint16_t)x) >> 8), BYTE_CAST(((uint16_t)x) >> 0)
 
-#define BE32(x) BYTE_CAST(((uint32)x) >> 24), BYTE_CAST(((uint32)x) >> 16), \
-                BYTE_CAST(((uint32)x) >>  8), BYTE_CAST(((uint32)x) >>  0)
+#define BE32(x) \
+  BYTE_CAST(((uint32_t)x) >> 24), BYTE_CAST(((uint32_t)x) >> 16), \
+      BYTE_CAST(((uint32_t)x) >> 8), BYTE_CAST(((uint32_t)x) >> 0)
 
-#define BE64(x) BYTE_CAST(((uint64)x) >> 56), BYTE_CAST(((uint64)x) >> 48), \
-                BYTE_CAST(((uint64)x) >> 40), BYTE_CAST(((uint64)x) >> 32), \
-                BYTE_CAST(((uint64)x) >> 24), BYTE_CAST(((uint64)x) >> 16), \
-                BYTE_CAST(((uint64)x) >>  8), BYTE_CAST(((uint64)x) >>  0)
+#define BE64(x) \
+  BYTE_CAST(((uint64_t)x) >> 56), BYTE_CAST(((uint64_t)x) >> 48),     \
+      BYTE_CAST(((uint64_t)x) >> 40), BYTE_CAST(((uint64_t)x) >> 32), \
+      BYTE_CAST(((uint64_t)x) >> 24), BYTE_CAST(((uint64_t)x) >> 16), \
+      BYTE_CAST(((uint64_t)x) >> 8), BYTE_CAST(((uint64_t)x) >> 0)
 
 // Declare a N-bit integer as a this-endian (local machine) sequence of bytes
 #ifndef BIG_ENDIAN
diff --git a/media/webrtc/trunk/webrtc/base/thread.cc b/media/webrtc/trunk/webrtc/base/thread.cc
index 6aded99f1f..4197d28175 100644
--- a/media/webrtc/trunk/webrtc/base/thread.cc
+++ b/media/webrtc/trunk/webrtc/base/thread.cc
@@ -22,6 +22,7 @@
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
 
@@ -35,7 +36,7 @@
 namespace rtc {
 
 ThreadManager* ThreadManager::Instance() {
-  LIBJINGLE_DEFINE_STATIC_LOCAL(ThreadManager, thread_manager, ());
+  RTC_DEFINE_STATIC_LOCAL(ThreadManager, thread_manager, ());
   return &thread_manager;
 }
 
@@ -139,7 +140,6 @@ Thread::ScopedDisallowBlockingCalls::~ScopedDisallowBlockingCalls() {
 
 Thread::Thread(SocketServer* ss)
     : MessageQueue(ss),
-      priority_(PRIORITY_NORMAL),
       running_(true, false),
 #if defined(WEBRTC_WIN)
       thread_(NULL),
@@ -187,34 +187,6 @@ bool Thread::SetName(const std::string& name, const void* obj) {
   return true;
 }
 
-bool Thread::SetPriority(ThreadPriority priority) {
-#if defined(WEBRTC_WIN)
-  if (running()) {
-    ASSERT(thread_ != NULL);
-    BOOL ret = FALSE;
-    if (priority == PRIORITY_NORMAL) {
-      ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_NORMAL);
-    } else if (priority == PRIORITY_HIGH) {
-      ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_HIGHEST);
-    } else if (priority == PRIORITY_ABOVE_NORMAL) {
-      ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_ABOVE_NORMAL);
-    } else if (priority == PRIORITY_IDLE) {
-      ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_IDLE);
-    }
-    if (!ret) {
-      return false;
-    }
-  }
-  priority_ = priority;
-  return true;
-#else
-  // TODO: Implement for Linux/Mac if possible.
-  if (running()) return false;
-  priority_ = priority;
-  return true;
-#endif
-}
-
 bool Thread::Start(Runnable* runnable) {
   ASSERT(owned_);
   if (!owned_) return false;
@@ -231,18 +203,10 @@ bool Thread::Start(Runnable* runnable) {
   init->thread = this;
   init->runnable = runnable;
 #if defined(WEBRTC_WIN)
-  DWORD flags = 0;
-  if (priority_ != PRIORITY_NORMAL) {
-    flags = CREATE_SUSPENDED;
-  }
-  thread_ = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)PreRun, init, flags,
+  thread_ = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)PreRun, init, 0,
                          &thread_id_);
   if (thread_) {
     running_.Set();
-    if (priority_ != PRIORITY_NORMAL) {
-      SetPriority(priority_);
-      ::ResumeThread(thread_);
-    }
   } else {
     return false;
   }
@@ -250,37 +214,6 @@ bool Thread::Start(Runnable* runnable) {
   pthread_attr_t attr;
   pthread_attr_init(&attr);
 
-  // Thread priorities are not supported in NaCl.
-#if !defined(__native_client__)
-  if (priority_ != PRIORITY_NORMAL) {
-    if (priority_ == PRIORITY_IDLE) {
-      // There is no POSIX-standard way to set a below-normal priority for an
-      // individual thread (only whole process), so let's not support it.
-      LOG(LS_WARNING) << "PRIORITY_IDLE not supported";
-    } else {
-      // Set real-time round-robin policy.
-      if (pthread_attr_setschedpolicy(&attr, SCHED_RR) != 0) {
-        LOG(LS_ERROR) << "pthread_attr_setschedpolicy";
-      }
-      struct sched_param param;
-      if (pthread_attr_getschedparam(&attr, &param) != 0) {
-        LOG(LS_ERROR) << "pthread_attr_getschedparam";
-      } else {
-        // The numbers here are arbitrary.
-        if (priority_ == PRIORITY_HIGH) {
-          param.sched_priority = 6;           // 6 = HIGH
-        } else {
-          ASSERT(priority_ == PRIORITY_ABOVE_NORMAL);
-          param.sched_priority = 4;           // 4 = ABOVE_NORMAL
-        }
-        if (pthread_attr_setschedparam(&attr, &param) != 0) {
-          LOG(LS_ERROR) << "pthread_attr_setschedparam";
-        }
-      }
-    }
-  }
-#endif  // !defined(__native_client__)
-
   int error_code = pthread_create(&thread_, &attr, PreRun, init);
   if (0 != error_code) {
     LOG(LS_ERROR) << "Unable to create pthread, error " << error_code;
@@ -344,47 +277,16 @@ bool Thread::SetAllowBlockingCalls(bool allow) {
 
 // static
 void Thread::AssertBlockingIsAllowedOnCurrentThread() {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
   Thread* current = Thread::Current();
   ASSERT(!current || current->blocking_calls_allowed_);
 #endif
 }
 
-#if defined(WEBRTC_WIN)
-// As seen on MSDN.
-// http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx
-#define MSDEV_SET_THREAD_NAME  0x406D1388
-typedef struct tagTHREADNAME_INFO {
-  DWORD dwType;
-  LPCSTR szName;
-  DWORD dwThreadID;
-  DWORD dwFlags;
-} THREADNAME_INFO;
-
-void SetThreadName(DWORD dwThreadID, LPCSTR szThreadName) {
-  THREADNAME_INFO info;
-  info.dwType = 0x1000;
-  info.szName = szThreadName;
-  info.dwThreadID = dwThreadID;
-  info.dwFlags = 0;
-
-  __try {
-    RaiseException(MSDEV_SET_THREAD_NAME, 0, sizeof(info) / sizeof(DWORD),
-                   reinterpret_cast<ULONG_PTR*>(&info));
-  }
-  __except(EXCEPTION_CONTINUE_EXECUTION) {
-  }
-}
-#endif  // WEBRTC_WIN
-
 void* Thread::PreRun(void* pv) {
   ThreadInit* init = static_cast<ThreadInit*>(pv);
   ThreadManager::Instance()->SetCurrentThread(init->thread);
-#if defined(WEBRTC_WIN)
-  SetThreadName(GetCurrentThreadId(), init->thread->name_.c_str());
-#elif defined(WEBRTC_POSIX)
-  // TODO: See if naming exists for pthreads.
-#endif
+  rtc::SetCurrentThreadName(init->thread->name_.c_str());
 #if __has_feature(objc_arc)
   @autoreleasepool
 #elif defined(WEBRTC_MAC)
@@ -415,7 +317,7 @@ void Thread::Stop() {
   Join();
 }
 
-void Thread::Send(MessageHandler *phandler, uint32 id, MessageData *pdata) {
+void Thread::Send(MessageHandler* phandler, uint32_t id, MessageData* pdata) {
   if (fStop_)
     return;
 
@@ -525,7 +427,8 @@ void Thread::InvokeEnd() {
   TRACE_EVENT_END0("webrtc", "Thread::Invoke");
 }
 
-void Thread::Clear(MessageHandler *phandler, uint32 id,
+void Thread::Clear(MessageHandler* phandler,
+                   uint32_t id,
                    MessageList* removed) {
   CritScope cs(&crit_);
 
@@ -554,7 +457,7 @@ void Thread::Clear(MessageHandler *phandler, uint32 id,
 }
 
 bool Thread::ProcessMessages(int cmsLoop) {
-  uint32 msEnd = (kForever == cmsLoop) ? 0 : TimeAfter(cmsLoop);
+  uint32_t msEnd = (kForever == cmsLoop) ? 0 : TimeAfter(cmsLoop);
   int cmsNext = cmsLoop;
 
   while (true) {
diff --git a/media/webrtc/trunk/webrtc/base/thread.h b/media/webrtc/trunk/webrtc/base/thread.h
index 3a9efb0c99..f91aa56733 100644
--- a/media/webrtc/trunk/webrtc/base/thread.h
+++ b/media/webrtc/trunk/webrtc/base/thread.h
@@ -68,7 +68,7 @@ class ThreadManager {
   DWORD key_;
 #endif
 
-  DISALLOW_COPY_AND_ASSIGN(ThreadManager);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ThreadManager);
 };
 
 struct _SendMessage {
@@ -78,13 +78,6 @@ struct _SendMessage {
   bool *ready;
 };
 
-enum ThreadPriority {
-  PRIORITY_IDLE = -1,
-  PRIORITY_NORMAL = 0,
-  PRIORITY_ABOVE_NORMAL = 1,
-  PRIORITY_HIGH = 2,
-};
-
 class Runnable {
  public:
   virtual ~Runnable() {}
@@ -94,7 +87,7 @@ class Runnable {
   Runnable() {}
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(Runnable);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Runnable);
 };
 
 // WARNING! SUBCLASSES MUST CALL Stop() IN THEIR DESTRUCTORS!  See ~Thread().
@@ -137,10 +130,6 @@ class Thread : public MessageQueue {
   const std::string& name() const { return name_; }
   bool SetName(const std::string& name, const void* obj);
 
-  // Sets the thread's priority. Must be called before Start().
-  ThreadPriority priority() const { return priority_; }
-  bool SetPriority(ThreadPriority priority);
-
   // Starts the execution of the thread.
   bool Start(Runnable* runnable = NULL);
 
@@ -155,8 +144,9 @@ class Thread : public MessageQueue {
   // ProcessMessages occasionally.
   virtual void Run();
 
-  virtual void Send(MessageHandler *phandler, uint32 id = 0,
-      MessageData *pdata = NULL);
+  virtual void Send(MessageHandler* phandler,
+                    uint32_t id = 0,
+                    MessageData* pdata = NULL);
 
   // Convenience method to invoke a functor on another thread.  Caller must
   // provide the |ReturnT| template argument, which cannot (easily) be deduced.
@@ -176,7 +166,7 @@ class Thread : public MessageQueue {
 
   // From MessageQueue
   void Clear(MessageHandler* phandler,
-             uint32 id = MQID_ANY,
+             uint32_t id = MQID_ANY,
              MessageList* removed = NULL) override;
   void ReceiveSends() override;
 
@@ -270,7 +260,6 @@ class Thread : public MessageQueue {
 
   std::list<_SendMessage> sendlist_;
   std::string name_;
-  ThreadPriority priority_;
   Event running_;  // Signalled means running.
 
 #if defined(WEBRTC_POSIX)
@@ -287,7 +276,7 @@ class Thread : public MessageQueue {
 
   friend class ThreadManager;
 
-  DISALLOW_COPY_AND_ASSIGN(Thread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Thread);
 };
 
 // AutoThread automatically installs itself at construction
@@ -300,7 +289,7 @@ class AutoThread : public Thread {
   ~AutoThread() override;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(AutoThread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AutoThread);
 };
 
 // Win32 extension for threads that need to use COM
@@ -314,7 +303,7 @@ class ComThread : public Thread {
   virtual void Run();
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(ComThread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ComThread);
 };
 #endif
 
@@ -332,7 +321,7 @@ class SocketServerScope {
  private:
   SocketServer* old_ss_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(SocketServerScope);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SocketServerScope);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/thread_checker.h b/media/webrtc/trunk/webrtc/base/thread_checker.h
index eee9315533..6cd7d7b9e0 100644
--- a/media/webrtc/trunk/webrtc/base/thread_checker.h
+++ b/media/webrtc/trunk/webrtc/base/thread_checker.h
@@ -18,10 +18,10 @@
 // with this define will get the same level of thread checking as
 // debug bots.
 //
-// Note that this does not perfectly match situations where DCHECK is
+// Note that this does not perfectly match situations where RTC_DCHECK is
 // enabled.  For example a non-official release build may have
 // DCHECK_ALWAYS_ON undefined (and therefore ThreadChecker would be
-// disabled) but have DCHECKs enabled at runtime.
+// disabled) but have RTC_DCHECKs enabled at runtime.
 #if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
 #define ENABLE_THREAD_CHECKER 1
 #else
@@ -67,7 +67,7 @@ class ThreadCheckerDoNothing {
 // class MyClass {
 //  public:
 //   void Foo() {
-//     DCHECK(thread_checker_.CalledOnValidThread());
+//     RTC_DCHECK(thread_checker_.CalledOnValidThread());
 //     ... (do stuff) ...
 //   }
 //
diff --git a/media/webrtc/trunk/webrtc/base/thread_checker_impl.cc b/media/webrtc/trunk/webrtc/base/thread_checker_impl.cc
index 7098a51cf6..79be606445 100644
--- a/media/webrtc/trunk/webrtc/base/thread_checker_impl.cc
+++ b/media/webrtc/trunk/webrtc/base/thread_checker_impl.cc
@@ -12,64 +12,10 @@
 
 #include "webrtc/base/thread_checker_impl.h"
 
-#include "webrtc/base/checks.h"
-
-#if defined(WEBRTC_LINUX)
-#include <sys/syscall.h>
-#endif
-
-#if defined(__NetBSD__)
-#include <lwp.h>
-#elif defined(__FreeBSD__)
-#include <pthread_np.h>
-#endif
+#include "webrtc/base/platform_thread.h"
 
 namespace rtc {
 
-PlatformThreadId CurrentThreadId() {
-  PlatformThreadId ret;
-#if defined(WEBRTC_WIN)
-  ret = GetCurrentThreadId();
-#elif defined(WEBRTC_POSIX)
-#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
-  ret = pthread_mach_thread_np(pthread_self());
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_GONK)
-  ret =  syscall(__NR_gettid);
-#elif defined(WEBRTC_ANDROID)
-  ret = gettid();
-#elif defined(__NetBSD__)
-  ret = _lwp_self();
-#elif defined(__DragonFly__)
-  ret = lwp_gettid();
-#elif defined(__OpenBSD__)
-  ret = reinterpret_cast<uintptr_t> (pthread_self());
-#elif defined(__FreeBSD__)
-  ret = pthread_getthreadid_np();
-#else
-  // Default implementation for nacl and solaris.
-  ret = reinterpret_cast<pid_t>(pthread_self());
-#endif
-#endif  // defined(WEBRTC_POSIX)
-  DCHECK(ret);
-  return ret;
-}
-
-PlatformThreadRef CurrentThreadRef() {
-#if defined(WEBRTC_WIN)
-  return GetCurrentThreadId();
-#elif defined(WEBRTC_POSIX)
-  return pthread_self();
-#endif
-}
-
-bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b) {
-#if defined(WEBRTC_WIN)
-  return a == b;
-#elif defined(WEBRTC_POSIX)
-  return pthread_equal(a, b);
-#endif
-}
-
 ThreadCheckerImpl::ThreadCheckerImpl() : valid_thread_(CurrentThreadRef()) {
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/thread_checker_impl.h b/media/webrtc/trunk/webrtc/base/thread_checker_impl.h
index 5ad15bb515..045583591d 100644
--- a/media/webrtc/trunk/webrtc/base/thread_checker_impl.h
+++ b/media/webrtc/trunk/webrtc/base/thread_checker_impl.h
@@ -13,33 +13,13 @@
 #ifndef WEBRTC_BASE_THREAD_CHECKER_IMPL_H_
 #define WEBRTC_BASE_THREAD_CHECKER_IMPL_H_
 
-#if defined(WEBRTC_POSIX)
-#include <pthread.h>
-#include <unistd.h>
-#endif
-
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread_types.h"
 
 namespace rtc {
 
-// Used for identifying the current thread. Always an integer value.
-#if defined(WEBRTC_WIN)
-typedef DWORD PlatformThreadId;
-typedef DWORD PlatformThreadRef;
-#elif defined(WEBRTC_POSIX)
-typedef pid_t PlatformThreadId;
-typedef pthread_t PlatformThreadRef;
-#endif
-
-// TODO(tommi): This+PlatformThreadId belongs in a common thread related header.
-PlatformThreadId CurrentThreadId();
-PlatformThreadRef CurrentThreadRef();
-
-// Compares two thread identifiers for equality.
-bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b);
-
 // Real implementation of ThreadChecker, for use in debug mode, or
-// for temporary use in release mode (e.g. to CHECK on a threading issue
+// for temporary use in release mode (e.g. to RTC_CHECK on a threading issue
 // seen only in the wild).
 //
 // Note: You should almost always use the ThreadChecker class to get the
diff --git a/media/webrtc/trunk/webrtc/base/thread_checker_unittest.cc b/media/webrtc/trunk/webrtc/base/thread_checker_unittest.cc
index 3c496314fa..338190093d 100644
--- a/media/webrtc/trunk/webrtc/base/thread_checker_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/thread_checker_unittest.cc
@@ -15,7 +15,6 @@
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 // Duplicated from base/threading/thread_checker.h so that we can be
 // good citizens there and undef the macro.
@@ -37,9 +36,7 @@ class ThreadCheckerClass : public ThreadChecker {
   ThreadCheckerClass() {}
 
   // Verifies that it was called on the same thread as the constructor.
-  void DoStuff() {
-    DCHECK(CalledOnValidThread());
-  }
+  void DoStuff() { RTC_DCHECK(CalledOnValidThread()); }
 
   void DetachFromThread() {
     ThreadChecker::DetachFromThread();
@@ -49,7 +46,7 @@ class ThreadCheckerClass : public ThreadChecker {
   static void DetachThenCallFromDifferentThreadImpl();
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(ThreadCheckerClass);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ThreadCheckerClass);
 };
 
 // Calls ThreadCheckerClass::DoStuff on another thread.
@@ -72,7 +69,7 @@ class CallDoStuffOnThread : public Thread {
  private:
   ThreadCheckerClass* thread_checker_class_;
 
-  DISALLOW_COPY_AND_ASSIGN(CallDoStuffOnThread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(CallDoStuffOnThread);
 };
 
 // Deletes ThreadCheckerClass on a different thread.
@@ -96,7 +93,7 @@ class DeleteThreadCheckerClassOnThread : public Thread {
  private:
   scoped_ptr<ThreadCheckerClass> thread_checker_class_;
 
-  DISALLOW_COPY_AND_ASSIGN(DeleteThreadCheckerClassOnThread);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DeleteThreadCheckerClassOnThread);
 };
 
 }  // namespace
diff --git a/media/webrtc/trunk/webrtc/base/thread_unittest.cc b/media/webrtc/trunk/webrtc/base/thread_unittest.cc
index 951a7e14f3..7ed4326724 100644
--- a/media/webrtc/trunk/webrtc/base/thread_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/thread_unittest.cc
@@ -15,7 +15,6 @@
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 #if defined(WEBRTC_WIN)
 #include <comdef.h>  // NOLINT
@@ -66,9 +65,9 @@ class SocketClient : public TestGenerator, public sigslot::has_slots<> {
   void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t size,
                 const SocketAddress& remote_addr,
                 const PacketTime& packet_time) {
-    EXPECT_EQ(size, sizeof(uint32));
-    uint32 prev = reinterpret_cast<const uint32*>(buf)[0];
-    uint32 result = Next(prev);
+    EXPECT_EQ(size, sizeof(uint32_t));
+    uint32_t prev = reinterpret_cast<const uint32_t*>(buf)[0];
+    uint32_t result = Next(prev);
 
     post_thread_->PostDelayed(200, post_handler_, 0, new TestMessage(result));
   }
@@ -137,16 +136,48 @@ class SignalWhenDestroyedThread : public Thread {
   Event* event_;
 };
 
+// A bool wrapped in a mutex, to avoid data races. Using a volatile
+// bool should be sufficient for correct code ("eventual consistency"
+// between caches is sufficient), but we can't tell the compiler about
+// that, and then tsan complains about a data race.
+
+// See also discussion at
+// http://stackoverflow.com/questions/7223164/is-mutex-needed-to-synchronize-a-simple-flag-between-pthreads
+
+// Using std::atomic<bool> or std::atomic_flag in C++11 is probably
+// the right thing to do, but those features are not yet allowed. Or
+// rtc::AtomicInt, if/when that is added. Since the use isn't
+// performance critical, use a plain critical section for the time
+// being.
+
+class AtomicBool {
+ public:
+  explicit AtomicBool(bool value = false) : flag_(value) {}
+  AtomicBool& operator=(bool value) {
+    CritScope scoped_lock(&cs_);
+    flag_ = value;
+    return *this;
+  }
+  bool get() const {
+    CritScope scoped_lock(&cs_);
+    return flag_;
+  }
+
+ private:
+  mutable CriticalSection cs_;
+  bool flag_;
+};
+
 // Function objects to test Thread::Invoke.
 struct FunctorA {
   int operator()() { return 42; }
 };
 class FunctorB {
  public:
-  explicit FunctorB(bool* flag) : flag_(flag) {}
+  explicit FunctorB(AtomicBool* flag) : flag_(flag) {}
   void operator()() { if (flag_) *flag_ = true; }
  private:
-  bool* flag_;
+  AtomicBool* flag_;
 };
 struct FunctorC {
   int operator()() {
@@ -220,33 +251,6 @@ TEST(ThreadTest, Names) {
   delete thread;
 }
 
-// Test that setting thread priorities doesn't cause a malfunction.
-// There's no easy way to verify the priority was set properly at this time.
-TEST(ThreadTest, Priorities) {
-  Thread *thread;
-  thread = new Thread();
-  EXPECT_TRUE(thread->SetPriority(PRIORITY_HIGH));
-  EXPECT_TRUE(thread->Start());
-  thread->Stop();
-  delete thread;
-  thread = new Thread();
-  EXPECT_TRUE(thread->SetPriority(PRIORITY_ABOVE_NORMAL));
-  EXPECT_TRUE(thread->Start());
-  thread->Stop();
-  delete thread;
-
-  thread = new Thread();
-  EXPECT_TRUE(thread->Start());
-#if defined(WEBRTC_WIN)
-  EXPECT_TRUE(thread->SetPriority(PRIORITY_ABOVE_NORMAL));
-#else
-  EXPECT_FALSE(thread->SetPriority(PRIORITY_ABOVE_NORMAL));
-#endif
-  thread->Stop();
-  delete thread;
-
-}
-
 TEST(ThreadTest, Wrap) {
   Thread* current_thread = Thread::Current();
   current_thread->UnwrapCurrent();
@@ -266,10 +270,10 @@ TEST(ThreadTest, Invoke) {
   thread.Start();
   // Try calling functors.
   EXPECT_EQ(42, thread.Invoke<int>(FunctorA()));
-  bool called = false;
+  AtomicBool called;
   FunctorB f2(&called);
   thread.Invoke<void>(f2);
-  EXPECT_TRUE(called);
+  EXPECT_TRUE(called.get());
   // Try calling bare functions.
   struct LocalFuncs {
     static int Func1() { return 999; }
@@ -408,9 +412,9 @@ TEST_F(AsyncInvokeTest, FireAndForget) {
   Thread thread;
   thread.Start();
   // Try calling functor.
-  bool called = false;
+  AtomicBool called;
   invoker.AsyncInvoke<void>(&thread, FunctorB(&called));
-  EXPECT_TRUE_WAIT(called, kWaitTimeout);
+  EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
 }
 
 TEST_F(AsyncInvokeTest, WithCallback) {
@@ -478,26 +482,26 @@ TEST_F(AsyncInvokeTest, KillInvokerBeforeExecute) {
 
 TEST_F(AsyncInvokeTest, Flush) {
   AsyncInvoker invoker;
-  bool flag1 = false;
-  bool flag2 = false;
+  AtomicBool flag1;
+  AtomicBool flag2;
   // Queue two async calls to the current thread.
   invoker.AsyncInvoke<void>(Thread::Current(),
                             FunctorB(&flag1));
   invoker.AsyncInvoke<void>(Thread::Current(),
                             FunctorB(&flag2));
   // Because we haven't pumped messages, these should not have run yet.
-  EXPECT_FALSE(flag1);
-  EXPECT_FALSE(flag2);
+  EXPECT_FALSE(flag1.get());
+  EXPECT_FALSE(flag2.get());
   // Force them to run now.
   invoker.Flush(Thread::Current());
-  EXPECT_TRUE(flag1);
-  EXPECT_TRUE(flag2);
+  EXPECT_TRUE(flag1.get());
+  EXPECT_TRUE(flag2.get());
 }
 
 TEST_F(AsyncInvokeTest, FlushWithIds) {
   AsyncInvoker invoker;
-  bool flag1 = false;
-  bool flag2 = false;
+  AtomicBool flag1;
+  AtomicBool flag2;
   // Queue two async calls to the current thread, one with a message id.
   invoker.AsyncInvoke<void>(Thread::Current(),
                             FunctorB(&flag1),
@@ -505,19 +509,195 @@ TEST_F(AsyncInvokeTest, FlushWithIds) {
   invoker.AsyncInvoke<void>(Thread::Current(),
                             FunctorB(&flag2));
   // Because we haven't pumped messages, these should not have run yet.
-  EXPECT_FALSE(flag1);
-  EXPECT_FALSE(flag2);
+  EXPECT_FALSE(flag1.get());
+  EXPECT_FALSE(flag2.get());
   // Execute pending calls with id == 5.
   invoker.Flush(Thread::Current(), 5);
-  EXPECT_TRUE(flag1);
-  EXPECT_FALSE(flag2);
+  EXPECT_TRUE(flag1.get());
+  EXPECT_FALSE(flag2.get());
   flag1 = false;
   // Execute all pending calls. The id == 5 call should not execute again.
   invoker.Flush(Thread::Current());
-  EXPECT_FALSE(flag1);
-  EXPECT_TRUE(flag2);
+  EXPECT_FALSE(flag1.get());
+  EXPECT_TRUE(flag2.get());
 }
 
+class GuardedAsyncInvokeTest : public testing::Test {
+ public:
+  void IntCallback(int value) {
+    EXPECT_EQ(expected_thread_, Thread::Current());
+    int_value_ = value;
+  }
+  void AsyncInvokeIntCallback(GuardedAsyncInvoker* invoker, Thread* thread) {
+    expected_thread_ = thread;
+    invoker->AsyncInvoke(FunctorC(), &GuardedAsyncInvokeTest::IntCallback,
+                         static_cast<GuardedAsyncInvokeTest*>(this));
+    invoke_started_.Set();
+  }
+  void SetExpectedThreadForIntCallback(Thread* thread) {
+    expected_thread_ = thread;
+  }
+
+ protected:
+  const static int kWaitTimeout = 1000;
+  GuardedAsyncInvokeTest()
+      : int_value_(0),
+        invoke_started_(true, false),
+        expected_thread_(nullptr) {}
+
+  int int_value_;
+  Event invoke_started_;
+  Thread* expected_thread_;
+};
+
+// Functor for creating an invoker.
+struct CreateInvoker {
+  CreateInvoker(scoped_ptr<GuardedAsyncInvoker>* invoker) : invoker_(invoker) {}
+  void operator()() { invoker_->reset(new GuardedAsyncInvoker()); }
+  scoped_ptr<GuardedAsyncInvoker>* invoker_;
+};
+
+// Test that we can call AsyncInvoke<void>() after the thread died.
+TEST_F(GuardedAsyncInvokeTest, KillThreadFireAndForget) {
+  // Create and start the thread.
+  scoped_ptr<Thread> thread(new Thread());
+  thread->Start();
+  scoped_ptr<GuardedAsyncInvoker> invoker;
+  // Create the invoker on |thread|.
+  thread->Invoke<void>(CreateInvoker(&invoker));
+  // Kill |thread|.
+  thread = nullptr;
+  // Try calling functor.
+  AtomicBool called;
+  EXPECT_FALSE(invoker->AsyncInvoke<void>(FunctorB(&called)));
+  // With thread gone, nothing should happen.
+  WAIT(called.get(), kWaitTimeout);
+  EXPECT_FALSE(called.get());
+}
+
+// Test that we can call AsyncInvoke with callback after the thread died.
+TEST_F(GuardedAsyncInvokeTest, KillThreadWithCallback) {
+  // Create and start the thread.
+  scoped_ptr<Thread> thread(new Thread());
+  thread->Start();
+  scoped_ptr<GuardedAsyncInvoker> invoker;
+  // Create the invoker on |thread|.
+  thread->Invoke<void>(CreateInvoker(&invoker));
+  // Kill |thread|.
+  thread = nullptr;
+  // Try calling functor.
+  EXPECT_FALSE(
+      invoker->AsyncInvoke(FunctorC(), &GuardedAsyncInvokeTest::IntCallback,
+                           static_cast<GuardedAsyncInvokeTest*>(this)));
+  // With thread gone, callback should be cancelled.
+  Thread::Current()->ProcessMessages(kWaitTimeout);
+  EXPECT_EQ(0, int_value_);
+}
+
+// The remaining tests check that GuardedAsyncInvoker behaves as AsyncInvoker
+// when Thread is still alive.
+TEST_F(GuardedAsyncInvokeTest, FireAndForget) {
+  GuardedAsyncInvoker invoker;
+  // Try calling functor.
+  AtomicBool called;
+  EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&called)));
+  EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
+}
+
+TEST_F(GuardedAsyncInvokeTest, WithCallback) {
+  GuardedAsyncInvoker invoker;
+  // Try calling functor.
+  SetExpectedThreadForIntCallback(Thread::Current());
+  EXPECT_TRUE(invoker.AsyncInvoke(FunctorA(),
+                                  &GuardedAsyncInvokeTest::IntCallback,
+                                  static_cast<GuardedAsyncInvokeTest*>(this)));
+  EXPECT_EQ_WAIT(42, int_value_, kWaitTimeout);
+}
+
+TEST_F(GuardedAsyncInvokeTest, CancelInvoker) {
+  // Try destroying invoker during call.
+  {
+    GuardedAsyncInvoker invoker;
+    EXPECT_TRUE(
+        invoker.AsyncInvoke(FunctorC(), &GuardedAsyncInvokeTest::IntCallback,
+                            static_cast<GuardedAsyncInvokeTest*>(this)));
+  }
+  // With invoker gone, callback should be cancelled.
+  Thread::Current()->ProcessMessages(kWaitTimeout);
+  EXPECT_EQ(0, int_value_);
+}
+
+TEST_F(GuardedAsyncInvokeTest, CancelCallingThread) {
+  GuardedAsyncInvoker invoker;
+  // Try destroying calling thread during call.
+  {
+    Thread thread;
+    thread.Start();
+    // Try calling functor.
+    thread.Invoke<void>(Bind(&GuardedAsyncInvokeTest::AsyncInvokeIntCallback,
+                             static_cast<GuardedAsyncInvokeTest*>(this),
+                             &invoker, Thread::Current()));
+    // Wait for the call to begin.
+    ASSERT_TRUE(invoke_started_.Wait(kWaitTimeout));
+  }
+  // Calling thread is gone. Return message shouldn't happen.
+  Thread::Current()->ProcessMessages(kWaitTimeout);
+  EXPECT_EQ(0, int_value_);
+}
+
+TEST_F(GuardedAsyncInvokeTest, KillInvokerBeforeExecute) {
+  Thread thread;
+  thread.Start();
+  {
+    GuardedAsyncInvoker invoker;
+    // Try calling functor.
+    thread.Invoke<void>(Bind(&GuardedAsyncInvokeTest::AsyncInvokeIntCallback,
+                             static_cast<GuardedAsyncInvokeTest*>(this),
+                             &invoker, Thread::Current()));
+    // Wait for the call to begin.
+    ASSERT_TRUE(invoke_started_.Wait(kWaitTimeout));
+  }
+  // Invoker is destroyed. Function should not execute.
+  Thread::Current()->ProcessMessages(kWaitTimeout);
+  EXPECT_EQ(0, int_value_);
+}
+
+TEST_F(GuardedAsyncInvokeTest, Flush) {
+  GuardedAsyncInvoker invoker;
+  AtomicBool flag1;
+  AtomicBool flag2;
+  // Queue two async calls to the current thread.
+  EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag1)));
+  EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag2)));
+  // Because we haven't pumped messages, these should not have run yet.
+  EXPECT_FALSE(flag1.get());
+  EXPECT_FALSE(flag2.get());
+  // Force them to run now.
+  EXPECT_TRUE(invoker.Flush());
+  EXPECT_TRUE(flag1.get());
+  EXPECT_TRUE(flag2.get());
+}
+
+TEST_F(GuardedAsyncInvokeTest, FlushWithIds) {
+  GuardedAsyncInvoker invoker;
+  AtomicBool flag1;
+  AtomicBool flag2;
+  // Queue two async calls to the current thread, one with a message id.
+  EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag1), 5));
+  EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag2)));
+  // Because we haven't pumped messages, these should not have run yet.
+  EXPECT_FALSE(flag1.get());
+  EXPECT_FALSE(flag2.get());
+  // Execute pending calls with id == 5.
+  EXPECT_TRUE(invoker.Flush(5));
+  EXPECT_TRUE(flag1.get());
+  EXPECT_FALSE(flag2.get());
+  flag1 = false;
+  // Execute all pending calls. The id == 5 call should not execute again.
+  EXPECT_TRUE(invoker.Flush());
+  EXPECT_FALSE(flag1.get());
+  EXPECT_TRUE(flag2.get());
+}
 
 #if defined(WEBRTC_WIN)
 class ComThreadTest : public testing::Test, public MessageHandler {
diff --git a/media/webrtc/trunk/webrtc/base/timeutils.cc b/media/webrtc/trunk/webrtc/base/timeutils.cc
index 64dae2f975..24b04ee2ee 100644
--- a/media/webrtc/trunk/webrtc/base/timeutils.cc
+++ b/media/webrtc/trunk/webrtc/base/timeutils.cc
@@ -32,17 +32,17 @@
 
 namespace rtc {
 
-const uint32 HALF = 0x80000000;
+const uint32_t HALF = 0x80000000;
 
-uint64 TimeNanos() {
-  int64 ticks = 0;
+uint64_t TimeNanos() {
+  int64_t ticks = 0;
 #if defined(WEBRTC_MAC)
   static mach_timebase_info_data_t timebase;
   if (timebase.denom == 0) {
     // Get the timebase if this is the first time we run.
     // Recommended by Apple's QA1398.
     if (mach_timebase_info(&timebase) != KERN_SUCCESS) {
-      DCHECK(false);
+      RTC_DCHECK(false);
     }
   }
   // Use timebase to convert absolute time tick units into nanoseconds.
@@ -52,11 +52,11 @@ uint64 TimeNanos() {
   // TODO: Do we need to handle the case when CLOCK_MONOTONIC
   // is not supported?
   clock_gettime(CLOCK_MONOTONIC, &ts);
-  ticks = kNumNanosecsPerSec * static_cast<int64>(ts.tv_sec) +
-      static_cast<int64>(ts.tv_nsec);
+  ticks = kNumNanosecsPerSec * static_cast<int64_t>(ts.tv_sec) +
+          static_cast<int64_t>(ts.tv_nsec);
 #elif defined(WEBRTC_WIN)
   static volatile LONG last_timegettime = 0;
-  static volatile int64 num_wrap_timegettime = 0;
+  static volatile int64_t num_wrap_timegettime = 0;
   volatile LONG* last_timegettime_ptr = &last_timegettime;
   DWORD now = timeGetTime();
   // Atomically update the last gotten time
@@ -74,20 +74,22 @@ uint64 TimeNanos() {
   // TODO: Calculate with nanosecond precision.  Otherwise, we're just
   // wasting a multiply and divide when doing Time() on Windows.
   ticks = ticks * kNumNanosecsPerMillisec;
+#else
+#error Unsupported platform.
 #endif
   return ticks;
 }
 
-uint32 Time() {
-  return static_cast<uint32>(TimeNanos() / kNumNanosecsPerMillisec);
+uint32_t Time() {
+  return static_cast<uint32_t>(TimeNanos() / kNumNanosecsPerMillisec);
 }
 
-uint64 TimeMicros() {
-  return static_cast<uint64>(TimeNanos() / kNumNanosecsPerMicrosec);
+uint64_t TimeMicros() {
+  return static_cast<uint64_t>(TimeNanos() / kNumNanosecsPerMicrosec);
 }
 
 #if defined(WEBRTC_WIN)
-static const uint64 kFileTimeToUnixTimeEpochOffset = 116444736000000000ULL;
+static const uint64_t kFileTimeToUnixTimeEpochOffset = 116444736000000000ULL;
 
 struct timeval {
   long tv_sec, tv_usec;  // NOLINT
@@ -105,7 +107,7 @@ static int gettimeofday(struct timeval *tv, void *tz) {
   li.HighPart = ft.dwHighDateTime;
 
   // Convert to seconds and microseconds since Unix time Epoch.
-  int64 micros = (li.QuadPart - kFileTimeToUnixTimeEpochOffset) / 10;
+  int64_t micros = (li.QuadPart - kFileTimeToUnixTimeEpochOffset) / 10;
   tv->tv_sec = static_cast<long>(micros / kNumMicrosecsPerSec);  // NOLINT
   tv->tv_usec = static_cast<long>(micros % kNumMicrosecsPerSec); // NOLINT
 
@@ -135,13 +137,13 @@ void CurrentTmTime(struct tm *tm, int *microseconds) {
   *microseconds = timeval.tv_usec;
 }
 
-uint32 TimeAfter(int32 elapsed) {
-  DCHECK_GE(elapsed, 0);
-  DCHECK_LT(static_cast<uint32>(elapsed), HALF);
+uint32_t TimeAfter(int32_t elapsed) {
+  RTC_DCHECK_GE(elapsed, 0);
+  RTC_DCHECK_LT(static_cast<uint32_t>(elapsed), HALF);
   return Time() + elapsed;
 }
 
-bool TimeIsBetween(uint32 earlier, uint32 middle, uint32 later) {
+bool TimeIsBetween(uint32_t earlier, uint32_t middle, uint32_t later) {
   if (earlier <= later) {
     return ((earlier <= middle) && (middle <= later));
   } else {
@@ -149,27 +151,27 @@ bool TimeIsBetween(uint32 earlier, uint32 middle, uint32 later) {
   }
 }
 
-bool TimeIsLaterOrEqual(uint32 earlier, uint32 later) {
+bool TimeIsLaterOrEqual(uint32_t earlier, uint32_t later) {
 #if EFFICIENT_IMPLEMENTATION
-  int32 diff = later - earlier;
-  return (diff >= 0 && static_cast<uint32>(diff) < HALF);
+  int32_t diff = later - earlier;
+  return (diff >= 0 && static_cast<uint32_t>(diff) < HALF);
 #else
   const bool later_or_equal = TimeIsBetween(earlier, later, earlier + HALF);
   return later_or_equal;
 #endif
 }
 
-bool TimeIsLater(uint32 earlier, uint32 later) {
+bool TimeIsLater(uint32_t earlier, uint32_t later) {
 #if EFFICIENT_IMPLEMENTATION
-  int32 diff = later - earlier;
-  return (diff > 0 && static_cast<uint32>(diff) < HALF);
+  int32_t diff = later - earlier;
+  return (diff > 0 && static_cast<uint32_t>(diff) < HALF);
 #else
   const bool earlier_or_equal = TimeIsBetween(later, earlier, later + HALF);
   return !earlier_or_equal;
 #endif
 }
 
-int32 TimeDiff(uint32 later, uint32 earlier) {
+int32_t TimeDiff(uint32_t later, uint32_t earlier) {
 #if EFFICIENT_IMPLEMENTATION
   return later - earlier;
 #else
@@ -193,7 +195,7 @@ int32 TimeDiff(uint32 later, uint32 earlier) {
 TimestampWrapAroundHandler::TimestampWrapAroundHandler()
     : last_ts_(0), num_wrap_(0) {}
 
-int64 TimestampWrapAroundHandler::Unwrap(uint32 ts) {
+int64_t TimestampWrapAroundHandler::Unwrap(uint32_t ts) {
   if (ts < last_ts_) {
     if (last_ts_ > 0xf0000000 && ts < 0x0fffffff) {
       ++num_wrap_;
@@ -204,4 +206,48 @@ int64 TimestampWrapAroundHandler::Unwrap(uint32 ts) {
   return unwrapped_ts;
 }
 
+int64_t TmToSeconds(const std::tm& tm) {
+  static short int mdays[12] = {31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31};
+  static short int cumul_mdays[12] = {0,   31,  59,  90,  120, 151,
+                                      181, 212, 243, 273, 304, 334};
+  int year = tm.tm_year + 1900;
+  int month = tm.tm_mon;
+  int day = tm.tm_mday - 1;  // Make 0-based like the rest.
+  int hour = tm.tm_hour;
+  int min = tm.tm_min;
+  int sec = tm.tm_sec;
+
+  bool expiry_in_leap_year = (year % 4 == 0 &&
+                              (year % 100 != 0 || year % 400 == 0));
+
+  if (year < 1970)
+    return -1;
+  if (month < 0 || month > 11)
+    return -1;
+  if (day < 0 || day >= mdays[month] + (expiry_in_leap_year && month == 2 - 1))
+    return -1;
+  if (hour < 0 || hour > 23)
+    return -1;
+  if (min < 0 || min > 59)
+    return -1;
+  if (sec < 0 || sec > 59)
+    return -1;
+
+  day += cumul_mdays[month];
+
+  // Add number of leap days between 1970 and the expiration year, inclusive.
+  day += ((year / 4 - 1970 / 4) - (year / 100 - 1970 / 100) +
+          (year / 400 - 1970 / 400));
+
+  // We will have added one day too much above if expiration is during a leap
+  // year, and expiration is in January or February.
+  if (expiry_in_leap_year && month <= 2 - 1) // |month| is zero based.
+    day -= 1;
+
+  // Combine all variables into seconds from 1970-01-01 00:00 (except |month|
+  // which was accumulated into |day| above).
+  return (((static_cast<int64_t>
+            (year - 1970) * 365 + day) * 24 + hour) * 60 + min) * 60 + sec;
+}
+
 } // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/timeutils.h b/media/webrtc/trunk/webrtc/base/timeutils.h
index ca041a7d11..3ade430947 100644
--- a/media/webrtc/trunk/webrtc/base/timeutils.h
+++ b/media/webrtc/trunk/webrtc/base/timeutils.h
@@ -11,72 +11,75 @@
 #ifndef WEBRTC_BASE_TIMEUTILS_H_
 #define WEBRTC_BASE_TIMEUTILS_H_
 
+#include <ctime>
 #include <time.h>
 
 #include "webrtc/base/basictypes.h"
 
 namespace rtc {
 
-static const int64 kNumMillisecsPerSec = INT64_C(1000);
-static const int64 kNumMicrosecsPerSec = INT64_C(1000000);
-static const int64 kNumNanosecsPerSec = INT64_C(1000000000);
+static const int64_t kNumMillisecsPerSec = INT64_C(1000);
+static const int64_t kNumMicrosecsPerSec = INT64_C(1000000);
+static const int64_t kNumNanosecsPerSec = INT64_C(1000000000);
 
-static const int64 kNumMicrosecsPerMillisec = kNumMicrosecsPerSec /
-    kNumMillisecsPerSec;
-static const int64 kNumNanosecsPerMillisec =  kNumNanosecsPerSec /
-    kNumMillisecsPerSec;
-static const int64 kNumNanosecsPerMicrosec =  kNumNanosecsPerSec /
-    kNumMicrosecsPerSec;
+static const int64_t kNumMicrosecsPerMillisec =
+    kNumMicrosecsPerSec / kNumMillisecsPerSec;
+static const int64_t kNumNanosecsPerMillisec =
+    kNumNanosecsPerSec / kNumMillisecsPerSec;
+static const int64_t kNumNanosecsPerMicrosec =
+    kNumNanosecsPerSec / kNumMicrosecsPerSec;
 
 // January 1970, in NTP milliseconds.
-static const int64 kJan1970AsNtpMillisecs = INT64_C(2208988800000);
+static const int64_t kJan1970AsNtpMillisecs = INT64_C(2208988800000);
 
-typedef uint32 TimeStamp;
+typedef uint32_t TimeStamp;
 
 // Returns the current time in milliseconds.
-uint32 Time();
+uint32_t Time();
 // Returns the current time in microseconds.
-uint64 TimeMicros();
+uint64_t TimeMicros();
 // Returns the current time in nanoseconds.
-uint64 TimeNanos();
+uint64_t TimeNanos();
 
 // Stores current time in *tm and microseconds in *microseconds.
 void CurrentTmTime(struct tm *tm, int *microseconds);
 
 // Returns a future timestamp, 'elapsed' milliseconds from now.
-uint32 TimeAfter(int32 elapsed);
+uint32_t TimeAfter(int32_t elapsed);
 
 // Comparisons between time values, which can wrap around.
-bool TimeIsBetween(uint32 earlier, uint32 middle, uint32 later);  // Inclusive
-bool TimeIsLaterOrEqual(uint32 earlier, uint32 later);  // Inclusive
-bool TimeIsLater(uint32 earlier, uint32 later);  // Exclusive
+bool TimeIsBetween(uint32_t earlier,
+                   uint32_t middle,
+                   uint32_t later);                         // Inclusive
+bool TimeIsLaterOrEqual(uint32_t earlier, uint32_t later);  // Inclusive
+bool TimeIsLater(uint32_t earlier, uint32_t later);         // Exclusive
 
 // Returns the later of two timestamps.
-inline uint32 TimeMax(uint32 ts1, uint32 ts2) {
+inline uint32_t TimeMax(uint32_t ts1, uint32_t ts2) {
   return TimeIsLaterOrEqual(ts1, ts2) ? ts2 : ts1;
 }
 
 // Returns the earlier of two timestamps.
-inline uint32 TimeMin(uint32 ts1, uint32 ts2) {
+inline uint32_t TimeMin(uint32_t ts1, uint32_t ts2) {
   return TimeIsLaterOrEqual(ts1, ts2) ? ts1 : ts2;
 }
 
 // Number of milliseconds that would elapse between 'earlier' and 'later'
 // timestamps.  The value is negative if 'later' occurs before 'earlier'.
-int32 TimeDiff(uint32 later, uint32 earlier);
+int32_t TimeDiff(uint32_t later, uint32_t earlier);
 
 // The number of milliseconds that have elapsed since 'earlier'.
-inline int32 TimeSince(uint32 earlier) {
+inline int32_t TimeSince(uint32_t earlier) {
   return TimeDiff(Time(), earlier);
 }
 
 // The number of milliseconds that will elapse between now and 'later'.
-inline int32 TimeUntil(uint32 later) {
+inline int32_t TimeUntil(uint32_t later) {
   return TimeDiff(later, Time());
 }
 
 // Converts a unix timestamp in nanoseconds to an NTP timestamp in ms.
-inline int64 UnixTimestampNanosecsToNtpMillisecs(int64 unix_ts_ns) {
+inline int64_t UnixTimestampNanosecsToNtpMillisecs(int64_t unix_ts_ns) {
   return unix_ts_ns / kNumNanosecsPerMillisec + kJan1970AsNtpMillisecs;
 }
 
@@ -84,13 +87,18 @@ class TimestampWrapAroundHandler {
  public:
   TimestampWrapAroundHandler();
 
-  int64 Unwrap(uint32 ts);
+  int64_t Unwrap(uint32_t ts);
 
  private:
-  uint32 last_ts_;
-  int64 num_wrap_;
+  uint32_t last_ts_;
+  int64_t num_wrap_;
 };
 
+// Convert from std::tm, which is relative to 1900-01-01 00:00 to number of
+// seconds from 1970-01-01 00:00 ("epoch").  Don't return time_t since that
+// is still 32 bits on many systems.
+int64_t TmToSeconds(const std::tm& tm);
+
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_TIMEUTILS_H_
diff --git a/media/webrtc/trunk/webrtc/base/timeutils_unittest.cc b/media/webrtc/trunk/webrtc/base/timeutils_unittest.cc
index 087fb0c28b..688658b32f 100644
--- a/media/webrtc/trunk/webrtc/base/timeutils_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/timeutils_unittest.cc
@@ -10,15 +10,16 @@
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/helpers.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
 
 namespace rtc {
 
 TEST(TimeTest, TimeInMs) {
-  uint32 ts_earlier = Time();
+  uint32_t ts_earlier = Time();
   Thread::SleepMs(100);
-  uint32 ts_now = Time();
+  uint32_t ts_now = Time();
   // Allow for the thread to wakeup ~20ms early.
   EXPECT_GE(ts_now, ts_earlier + 80);
   // Make sure the Time is not returning in smaller unit like microseconds.
@@ -152,8 +153,8 @@ class TimestampWrapAroundHandlerTest : public testing::Test {
 };
 
 TEST_F(TimestampWrapAroundHandlerTest, Unwrap) {
-  uint32 ts = 0xfffffff2;
-  int64 unwrapped_ts = ts;
+  uint32_t ts = 0xfffffff2;
+  int64_t unwrapped_ts = ts;
   EXPECT_EQ(ts, wraparound_handler_.Unwrap(ts));
   ts = 2;
   unwrapped_ts += 0x10;
@@ -166,4 +167,99 @@ TEST_F(TimestampWrapAroundHandlerTest, Unwrap) {
   EXPECT_EQ(unwrapped_ts, wraparound_handler_.Unwrap(ts));
 }
 
+class TmToSeconds : public testing::Test {
+ public:
+  TmToSeconds() {
+    // Set use of the test RNG to get deterministic expiration timestamp.
+    rtc::SetRandomTestMode(true);
+  }
+  ~TmToSeconds() {
+    // Put it back for the next test.
+    rtc::SetRandomTestMode(false);
+  }
+
+  void TestTmToSeconds(int times) {
+    static char mdays[12] = {31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31};
+    for (int i = 0; i < times; i++) {
+
+      // First generate something correct and check that TmToSeconds is happy.
+      int year = rtc::CreateRandomId() % 400 + 1970;
+
+      bool leap_year = false;
+      if (year % 4 == 0)
+        leap_year = true;
+      if (year % 100 == 0)
+        leap_year = false;
+      if (year % 400 == 0)
+        leap_year = true;
+
+      std::tm tm;
+      tm.tm_year = year - 1900;  // std::tm is year 1900 based.
+      tm.tm_mon = rtc::CreateRandomId() % 12;
+      tm.tm_mday = rtc::CreateRandomId() % mdays[tm.tm_mon] + 1;
+      tm.tm_hour = rtc::CreateRandomId() % 24;
+      tm.tm_min = rtc::CreateRandomId() % 60;
+      tm.tm_sec = rtc::CreateRandomId() % 60;
+      int64_t t = rtc::TmToSeconds(tm);
+      EXPECT_TRUE(t >= 0);
+
+      // Now damage a random field and check that TmToSeconds is unhappy.
+      switch (rtc::CreateRandomId() % 11) {
+        case 0:
+          tm.tm_year = 1969 - 1900;
+          break;
+        case 1:
+          tm.tm_mon = -1;
+          break;
+        case 2:
+          tm.tm_mon = 12;
+          break;
+        case 3:
+          tm.tm_mday = 0;
+          break;
+        case 4:
+          tm.tm_mday = mdays[tm.tm_mon] + (leap_year && tm.tm_mon == 1) + 1;
+          break;
+        case 5:
+          tm.tm_hour = -1;
+          break;
+        case 6:
+          tm.tm_hour = 24;
+          break;
+        case 7:
+          tm.tm_min = -1;
+          break;
+        case 8:
+          tm.tm_min = 60;
+          break;
+        case 9:
+          tm.tm_sec = -1;
+          break;
+        case 10:
+          tm.tm_sec = 60;
+          break;
+      }
+      EXPECT_EQ(rtc::TmToSeconds(tm), -1);
+    }
+    // Check consistency with the system gmtime_r.  With time_t, we can only
+    // portably test dates until 2038, which is achieved by the % 0x80000000.
+    for (int i = 0; i < times; i++) {
+      time_t t = rtc::CreateRandomId() % 0x80000000;
+#if defined(WEBRTC_WIN)
+      std::tm* tm = std::gmtime(&t);
+      EXPECT_TRUE(tm);
+      EXPECT_TRUE(rtc::TmToSeconds(*tm) == t);
+#else
+      std::tm tm;
+      EXPECT_TRUE(gmtime_r(&t, &tm));
+      EXPECT_TRUE(rtc::TmToSeconds(tm) == t);
+#endif
+    }
+  }
+};
+
+TEST_F(TmToSeconds, TestTmToSeconds) {
+  TestTmToSeconds(100000);
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/trace_event.h b/media/webrtc/trunk/webrtc/base/trace_event.h
index c14cbff030..3916af4fb6 100644
--- a/media/webrtc/trunk/webrtc/base/trace_event.h
+++ b/media/webrtc/trunk/webrtc/base/trace_event.h
@@ -701,7 +701,7 @@ class TraceID {
 
   explicit TraceID(const void* id, unsigned char* flags)
       : data_(static_cast<unsigned long long>(
-              reinterpret_cast<unsigned long>(id))) {
+              reinterpret_cast<uintptr_t>(id))) {
     *flags |= TRACE_EVENT_FLAG_MANGLE_ID;
   }
   explicit TraceID(ForceMangle id, unsigned char* flags) : data_(id.data()) {
diff --git a/media/webrtc/trunk/webrtc/base/unittest_main.cc b/media/webrtc/trunk/webrtc/base/unittest_main.cc
index c9864fe08b..167570d449 100644
--- a/media/webrtc/trunk/webrtc/base/unittest_main.cc
+++ b/media/webrtc/trunk/webrtc/base/unittest_main.cc
@@ -19,9 +19,16 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/ssladapter.h"
+#include "webrtc/test/field_trial.h"
 
 DEFINE_bool(help, false, "prints this message");
 DEFINE_string(log, "", "logging options to use");
+DEFINE_string(
+    force_fieldtrials,
+    "",
+    "Field trials control experimental feature code which can be forced. "
+    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+    " will assign the group Enable to field trial WebRTC-FooFeature.");
 #if defined(WEBRTC_WIN)
 DEFINE_int(crt_break_alloc, -1, "memory allocation to break on");
 DEFINE_bool(default_error_handlers, false,
@@ -51,7 +58,7 @@ int TestCrtReportHandler(int report_type, char* msg, int* retval) {
     return TRUE;
   }
 }
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 int main(int argc, char** argv) {
   testing::InitGoogleTest(&argc, argv);
@@ -61,6 +68,8 @@ int main(int argc, char** argv) {
     return 0;
   }
 
+  webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
+
 #if defined(WEBRTC_WIN)
   if (!FLAG_default_error_handlers) {
     // Make sure any errors don't throw dialogs hanging the test run.
@@ -69,13 +78,13 @@ int main(int argc, char** argv) {
     _CrtSetReportHook2(_CRT_RPTHOOK_INSTALL, TestCrtReportHandler);
   }
 
-#ifdef _DEBUG  // Turn on memory leak checking on Windows.
+#if !defined(NDEBUG)  // Turn on memory leak checking on Windows.
   _CrtSetDbgFlag(_CRTDBG_ALLOC_MEM_DF |_CRTDBG_LEAK_CHECK_DF);
   if (FLAG_crt_break_alloc >= 0) {
     _crtBreakAlloc = FLAG_crt_break_alloc;
   }
-#endif  // _DEBUG
-#endif  // WEBRTC_WIN 
+#endif
+#endif  // WEBRTC_WIN
 
   rtc::Filesystem::SetOrganizationName("google");
   rtc::Filesystem::SetApplicationName("unittest");
@@ -83,7 +92,11 @@ int main(int argc, char** argv) {
   // By default, log timestamps. Allow overrides by used of a --log flag.
   rtc::LogMessage::LogTimestamps();
   if (*FLAG_log != '\0') {
-    rtc::LogMessage::ConfigureLogging(FLAG_log, "unittest.log");
+    rtc::LogMessage::ConfigureLogging(FLAG_log);
+  } else if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO) {
+    // Default to LS_INFO, even for release builds to provide better test
+    // logging.
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
   }
 
   // Initialize SSL which are used by several tests.
@@ -94,7 +107,7 @@ int main(int argc, char** argv) {
   rtc::CleanupSSL();
 
   // clean up logging so we don't appear to leak memory.
-  rtc::LogMessage::ConfigureLogging("", "");
+  rtc::LogMessage::ConfigureLogging("");
 
 #if defined(WEBRTC_WIN)
   // Unhook crt function so that we don't ever log after statics have been
diff --git a/media/webrtc/trunk/webrtc/base/unixfilesystem.cc b/media/webrtc/trunk/webrtc/base/unixfilesystem.cc
index 081d561dba..734e880d9e 100644
--- a/media/webrtc/trunk/webrtc/base/unixfilesystem.cc
+++ b/media/webrtc/trunk/webrtc/base/unixfilesystem.cc
@@ -44,6 +44,7 @@
 #include <sys/syslimits.h>
 #endif
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/stream.h"
@@ -176,7 +177,7 @@ bool UnixFilesystem::GetTemporaryFolder(Pathname &pathname, bool create,
                         kCreateFolder, &fr))
     return false;
   unsigned char buffer[NAME_MAX+1];
-  if (0 != FSRefMakePath(&fr, buffer, ARRAY_SIZE(buffer)))
+  if (0 != FSRefMakePath(&fr, buffer, arraysize(buffer)))
     return false;
   pathname.SetPathname(reinterpret_cast<char*>(buffer), "");
 #elif defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
@@ -303,7 +304,7 @@ bool UnixFilesystem::IsTemporaryPath(const Pathname& pathname) {
 #endif  // WEBRTC_MAC && !defined(WEBRTC_IOS)
 #endif  // WEBRTC_ANDROID || WEBRTC_IOS
   };
-  for (size_t i = 0; i < ARRAY_SIZE(kTempPrefixes); ++i) {
+  for (size_t i = 0; i < arraysize(kTempPrefixes); ++i) {
     if (0 == strncmp(pathname.pathname().c_str(), kTempPrefixes[i],
                      strlen(kTempPrefixes[i])))
       return true;
@@ -377,7 +378,7 @@ bool UnixFilesystem::GetAppPathname(Pathname* path) {
   return true;
 #else  // WEBRTC_MAC && !defined(WEBRTC_IOS)
   char buffer[PATH_MAX + 2];
-  ssize_t len = readlink("/proc/self/exe", buffer, ARRAY_SIZE(buffer) - 1);
+  ssize_t len = readlink("/proc/self/exe", buffer, arraysize(buffer) - 1);
   if ((len <= 0) || (len == PATH_MAX + 1))
     return false;
   buffer[len] = '\0';
@@ -399,7 +400,7 @@ bool UnixFilesystem::GetAppDataFolder(Pathname* path, bool per_user) {
                           kCreateFolder, &fr))
       return false;
     unsigned char buffer[NAME_MAX+1];
-    if (0 != FSRefMakePath(&fr, buffer, ARRAY_SIZE(buffer)))
+    if (0 != FSRefMakePath(&fr, buffer, arraysize(buffer)))
       return false;
     path->SetPathname(reinterpret_cast<char*>(buffer), "");
   } else {
@@ -487,7 +488,7 @@ bool UnixFilesystem::GetAppTempFolder(Pathname* path) {
 
   // Create a random directory as /tmp/<appname>-<pid>-<timestamp>
   char buffer[128];
-  sprintfn(buffer, ARRAY_SIZE(buffer), "-%d-%d",
+  sprintfn(buffer, arraysize(buffer), "-%d-%d",
            static_cast<int>(getpid()),
            static_cast<int>(time(0)));
   std::string folder(application_name_);
@@ -502,7 +503,8 @@ bool UnixFilesystem::GetAppTempFolder(Pathname* path) {
 #endif
 }
 
-bool UnixFilesystem::GetDiskFreeSpace(const Pathname& path, int64 *freebytes) {
+bool UnixFilesystem::GetDiskFreeSpace(const Pathname& path,
+                                      int64_t* freebytes) {
 #ifdef __native_client__
   return false;
 #else  // __native_client__
@@ -526,9 +528,9 @@ bool UnixFilesystem::GetDiskFreeSpace(const Pathname& path, int64 *freebytes) {
     return false;
 #endif  // WEBRTC_ANDROID
 #if defined(WEBRTC_LINUX)
-  *freebytes = static_cast<int64>(vfs.f_bsize) * vfs.f_bavail;
+  *freebytes = static_cast<int64_t>(vfs.f_bsize) * vfs.f_bavail;
 #elif defined(WEBRTC_MAC)
-  *freebytes = static_cast<int64>(vfs.f_frsize) * vfs.f_bavail;
+  *freebytes = static_cast<int64_t>(vfs.f_frsize) * vfs.f_bavail;
 #endif
 
   return true;
diff --git a/media/webrtc/trunk/webrtc/base/unixfilesystem.h b/media/webrtc/trunk/webrtc/base/unixfilesystem.h
index e220911187..dbfbaf0a7d 100644
--- a/media/webrtc/trunk/webrtc/base/unixfilesystem.h
+++ b/media/webrtc/trunk/webrtc/base/unixfilesystem.h
@@ -107,7 +107,7 @@ class UnixFilesystem : public FilesystemInterface {
   // Get a temporary folder that is unique to the current user and application.
   bool GetAppTempFolder(Pathname* path) override;
 
-  bool GetDiskFreeSpace(const Pathname& path, int64* freebytes) override;
+  bool GetDiskFreeSpace(const Pathname& path, int64_t* freebytes) override;
 
   // Returns the absolute path of the current directory.
   Pathname GetCurrentDirectory() override;
diff --git a/media/webrtc/trunk/webrtc/base/urlencode_unittest.cc b/media/webrtc/trunk/webrtc/base/urlencode_unittest.cc
index 52169132e2..6a61db3ae3 100644
--- a/media/webrtc/trunk/webrtc/base/urlencode_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/urlencode_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/thread.h"
@@ -19,7 +20,7 @@ TEST(Urlencode, SourceTooLong) {
   char source[] = "^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^"
       "^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^";
   char dest[1];
-  ASSERT_EQ(0, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+  ASSERT_EQ(0, UrlEncode(source, dest, arraysize(dest)));
   ASSERT_EQ('\0', dest[0]);
 
   dest[0] = 'a';
@@ -30,7 +31,7 @@ TEST(Urlencode, SourceTooLong) {
 TEST(Urlencode, OneCharacterConversion) {
   char source[] = "^";
   char dest[4];
-  ASSERT_EQ(3, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+  ASSERT_EQ(3, UrlEncode(source, dest, arraysize(dest)));
   ASSERT_STREQ("%5E", dest);
 }
 
@@ -40,7 +41,7 @@ TEST(Urlencode, ShortDestinationNoEncoding) {
   // hold the text given.
   char source[] = "aa";
   char dest[3];
-  ASSERT_EQ(2, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+  ASSERT_EQ(2, UrlEncode(source, dest, arraysize(dest)));
   ASSERT_STREQ("aa", dest);
 }
 
@@ -49,14 +50,14 @@ TEST(Urlencode, ShortDestinationEncoding) {
   // big enough to hold the encoding.
   char source[] = "&";
   char dest[3];
-  ASSERT_EQ(0, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+  ASSERT_EQ(0, UrlEncode(source, dest, arraysize(dest)));
   ASSERT_EQ('\0', dest[0]);
 }
 
 TEST(Urlencode, Encoding1) {
   char source[] = "A^ ";
   char dest[8];
-  ASSERT_EQ(5, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+  ASSERT_EQ(5, UrlEncode(source, dest, arraysize(dest)));
   ASSERT_STREQ("A%5E+", dest);
 }
 
@@ -64,7 +65,7 @@ TEST(Urlencode, Encoding2) {
   char source[] = "A^ ";
   char dest[8];
   ASSERT_EQ(7, rtc::UrlEncodeWithoutEncodingSpaceAsPlus(source, dest,
-                                                        ARRAY_SIZE(dest)));
+                                                        arraysize(dest)));
   ASSERT_STREQ("A%5E%20", dest);
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/virtualsocket_unittest.cc b/media/webrtc/trunk/webrtc/base/virtualsocket_unittest.cc
index e9d57f8f30..2cd2b5e4de 100644
--- a/media/webrtc/trunk/webrtc/base/virtualsocket_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/virtualsocket_unittest.cc
@@ -14,6 +14,7 @@
 #include <netinet/in.h>
 #endif
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/testclient.h"
@@ -21,21 +22,23 @@
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using namespace rtc;
 
 // Sends at a constant rate but with random packet sizes.
 struct Sender : public MessageHandler {
-  Sender(Thread* th, AsyncSocket* s, uint32 rt)
-      : thread(th), socket(new AsyncUDPSocket(s)),
-        done(false), rate(rt), count(0) {
+  Sender(Thread* th, AsyncSocket* s, uint32_t rt)
+      : thread(th),
+        socket(new AsyncUDPSocket(s)),
+        done(false),
+        rate(rt),
+        count(0) {
     last_send = rtc::Time();
     thread->PostDelayed(NextDelay(), this, 1);
   }
 
-  uint32 NextDelay() {
-    uint32 size = (rand() % 4096) + 1;
+  uint32_t NextDelay() {
+    uint32_t size = (rand() % 4096) + 1;
     return 1000 * size / rate;
   }
 
@@ -45,11 +48,11 @@ struct Sender : public MessageHandler {
     if (done)
       return;
 
-    uint32 cur_time = rtc::Time();
-    uint32 delay = cur_time - last_send;
-    uint32 size = rate * delay / 1000;
-    size = std::min<uint32>(size, 4096);
-    size = std::max<uint32>(size, sizeof(uint32));
+    uint32_t cur_time = rtc::Time();
+    uint32_t delay = cur_time - last_send;
+    uint32_t size = rate * delay / 1000;
+    size = std::min<uint32_t>(size, 4096);
+    size = std::max<uint32_t>(size, sizeof(uint32_t));
 
     count += size;
     memcpy(dummy, &cur_time, sizeof(cur_time));
@@ -63,16 +66,23 @@ struct Sender : public MessageHandler {
   scoped_ptr<AsyncUDPSocket> socket;
   rtc::PacketOptions options;
   bool done;
-  uint32 rate;  // bytes per second
-  uint32 count;
-  uint32 last_send;
+  uint32_t rate;  // bytes per second
+  uint32_t count;
+  uint32_t last_send;
   char dummy[4096];
 };
 
 struct Receiver : public MessageHandler, public sigslot::has_slots<> {
-  Receiver(Thread* th, AsyncSocket* s, uint32 bw)
-      : thread(th), socket(new AsyncUDPSocket(s)), bandwidth(bw), done(false),
-        count(0), sec_count(0), sum(0), sum_sq(0), samples(0) {
+  Receiver(Thread* th, AsyncSocket* s, uint32_t bw)
+      : thread(th),
+        socket(new AsyncUDPSocket(s)),
+        bandwidth(bw),
+        done(false),
+        count(0),
+        sec_count(0),
+        sum(0),
+        sum_sq(0),
+        samples(0) {
     socket->SignalReadPacket.connect(this, &Receiver::OnReadPacket);
     thread->PostDelayed(1000, this, 1);
   }
@@ -90,9 +100,9 @@ struct Receiver : public MessageHandler, public sigslot::has_slots<> {
     count += size;
     sec_count += size;
 
-    uint32 send_time = *reinterpret_cast<const uint32*>(data);
-    uint32 recv_time = rtc::Time();
-    uint32 delay = recv_time - send_time;
+    uint32_t send_time = *reinterpret_cast<const uint32_t*>(data);
+    uint32_t recv_time = rtc::Time();
+    uint32_t delay = recv_time - send_time;
     sum += delay;
     sum_sq += delay * delay;
     samples += 1;
@@ -114,13 +124,13 @@ struct Receiver : public MessageHandler, public sigslot::has_slots<> {
 
   Thread* thread;
   scoped_ptr<AsyncUDPSocket> socket;
-  uint32 bandwidth;
+  uint32_t bandwidth;
   bool done;
   size_t count;
   size_t sec_count;
   double sum;
   double sum_sq;
-  uint32 samples;
+  uint32_t samples;
 };
 
 class VirtualSocketServerTest : public testing::Test {
@@ -143,12 +153,47 @@ class VirtualSocketServerTest : public testing::Test {
     } else if (post_ip.family() == AF_INET6) {
       in6_addr post_ip6 = post_ip.ipv6_address();
       in6_addr pre_ip6 = pre_ip.ipv6_address();
-      uint32* post_as_ints = reinterpret_cast<uint32*>(&post_ip6.s6_addr);
-      uint32* pre_as_ints = reinterpret_cast<uint32*>(&pre_ip6.s6_addr);
+      uint32_t* post_as_ints = reinterpret_cast<uint32_t*>(&post_ip6.s6_addr);
+      uint32_t* pre_as_ints = reinterpret_cast<uint32_t*>(&pre_ip6.s6_addr);
       EXPECT_EQ(post_as_ints[3], pre_as_ints[3]);
     }
   }
 
+  // Test a client can bind to the any address, and all sent packets will have
+  // the default route as the source address. Also, it can receive packets sent
+  // to the default route.
+  void TestDefaultRoute(const IPAddress& default_route) {
+    ss_->SetDefaultRoute(default_route);
+
+    // Create client1 bound to the any address.
+    AsyncSocket* socket =
+        ss_->CreateAsyncSocket(default_route.family(), SOCK_DGRAM);
+    socket->Bind(EmptySocketAddressWithFamily(default_route.family()));
+    SocketAddress client1_any_addr = socket->GetLocalAddress();
+    EXPECT_TRUE(client1_any_addr.IsAnyIP());
+    TestClient* client1 = new TestClient(new AsyncUDPSocket(socket));
+
+    // Create client2 bound to the default route.
+    AsyncSocket* socket2 =
+        ss_->CreateAsyncSocket(default_route.family(), SOCK_DGRAM);
+    socket2->Bind(SocketAddress(default_route, 0));
+    SocketAddress client2_addr = socket2->GetLocalAddress();
+    EXPECT_FALSE(client2_addr.IsAnyIP());
+    TestClient* client2 = new TestClient(new AsyncUDPSocket(socket2));
+
+    // Client1 sends to client2, client2 should see the default route as
+    // client1's address.
+    SocketAddress client1_addr;
+    EXPECT_EQ(6, client1->SendTo("bizbaz", 6, client2_addr));
+    EXPECT_TRUE(client2->CheckNextPacket("bizbaz", 6, &client1_addr));
+    EXPECT_EQ(client1_addr,
+              SocketAddress(default_route, client1_any_addr.port()));
+
+    // Client2 can send back to client1's default route address.
+    EXPECT_EQ(3, client2->SendTo("foo", 3, client1_addr));
+    EXPECT_TRUE(client1->CheckNextPacket("foo", 3, &client2_addr));
+  }
+
   void BasicTest(const SocketAddress& initial_addr) {
     AsyncSocket* socket = ss_->CreateAsyncSocket(initial_addr.family(),
                                                  SOCK_DGRAM);
@@ -585,8 +630,8 @@ class VirtualSocketServerTest : public testing::Test {
     }
 
     // Next, deliver packets at random intervals
-    const uint32 mean = 50;
-    const uint32 stddev = 50;
+    const uint32_t mean = 50;
+    const uint32_t stddev = 50;
 
     ss_->set_delay_mean(mean);
     ss_->set_delay_stddev(stddev);
@@ -619,7 +664,7 @@ class VirtualSocketServerTest : public testing::Test {
     EXPECT_EQ(recv_socket->GetLocalAddress().family(), initial_addr.family());
     ASSERT_EQ(0, send_socket->Connect(recv_socket->GetLocalAddress()));
 
-    uint32 bandwidth = 64 * 1024;
+    uint32_t bandwidth = 64 * 1024;
     ss_->set_bandwidth(bandwidth);
 
     Thread* pthMain = Thread::Current();
@@ -644,8 +689,8 @@ class VirtualSocketServerTest : public testing::Test {
     LOG(LS_VERBOSE) << "seed = " << seed;
     srand(static_cast<unsigned int>(seed));
 
-    const uint32 mean = 2000;
-    const uint32 stddev = 500;
+    const uint32_t mean = 2000;
+    const uint32_t stddev = 500;
 
     ss_->set_delay_mean(mean);
     ss_->set_delay_stddev(stddev);
@@ -791,6 +836,18 @@ TEST_F(VirtualSocketServerTest, basic_v6) {
   BasicTest(ipv6_test_addr);
 }
 
+TEST_F(VirtualSocketServerTest, TestDefaultRoute_v4) {
+  IPAddress ipv4_default_addr(0x01020304);
+  TestDefaultRoute(ipv4_default_addr);
+}
+
+TEST_F(VirtualSocketServerTest, TestDefaultRoute_v6) {
+  IPAddress ipv6_default_addr;
+  EXPECT_TRUE(
+      IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", &ipv6_default_addr));
+  TestDefaultRoute(ipv6_default_addr);
+}
+
 TEST_F(VirtualSocketServerTest, connect_v4) {
   ConnectTest(kIPv4AnyAddress);
 }
@@ -961,16 +1018,16 @@ TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv6ToIPv4Any) {
 }
 
 TEST_F(VirtualSocketServerTest, CreatesStandardDistribution) {
-  const uint32 kTestMean[] = { 10, 100, 333, 1000 };
+  const uint32_t kTestMean[] = {10, 100, 333, 1000};
   const double kTestDev[] = { 0.25, 0.1, 0.01 };
   // TODO: The current code only works for 1000 data points or more.
-  const uint32 kTestSamples[] = { /*10, 100,*/ 1000 };
-  for (size_t midx = 0; midx < ARRAY_SIZE(kTestMean); ++midx) {
-    for (size_t didx = 0; didx < ARRAY_SIZE(kTestDev); ++didx) {
-      for (size_t sidx = 0; sidx < ARRAY_SIZE(kTestSamples); ++sidx) {
+  const uint32_t kTestSamples[] = {/*10, 100,*/ 1000};
+  for (size_t midx = 0; midx < arraysize(kTestMean); ++midx) {
+    for (size_t didx = 0; didx < arraysize(kTestDev); ++didx) {
+      for (size_t sidx = 0; sidx < arraysize(kTestSamples); ++sidx) {
         ASSERT_LT(0u, kTestSamples[sidx]);
-        const uint32 kStdDev =
-            static_cast<uint32>(kTestDev[didx] * kTestMean[midx]);
+        const uint32_t kStdDev =
+            static_cast<uint32_t>(kTestDev[didx] * kTestMean[midx]);
         VirtualSocketServer::Function* f =
             VirtualSocketServer::CreateDistribution(kTestMean[midx],
                                                     kStdDev,
@@ -978,12 +1035,12 @@ TEST_F(VirtualSocketServerTest, CreatesStandardDistribution) {
         ASSERT_TRUE(NULL != f);
         ASSERT_EQ(kTestSamples[sidx], f->size());
         double sum = 0;
-        for (uint32 i = 0; i < f->size(); ++i) {
+        for (uint32_t i = 0; i < f->size(); ++i) {
           sum += (*f)[i].second;
         }
         const double mean = sum / f->size();
         double sum_sq_dev = 0;
-        for (uint32 i = 0; i < f->size(); ++i) {
+        for (uint32_t i = 0; i < f->size(); ++i) {
           double dev = (*f)[i].second - mean;
           sum_sq_dev += dev * dev;
         }
diff --git a/media/webrtc/trunk/webrtc/base/virtualsocketserver.cc b/media/webrtc/trunk/webrtc/base/virtualsocketserver.cc
index 8cb431dcf2..c6d402f1f3 100644
--- a/media/webrtc/trunk/webrtc/base/virtualsocketserver.cc
+++ b/media/webrtc/trunk/webrtc/base/virtualsocketserver.cc
@@ -17,6 +17,7 @@
 #include <map>
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/physicalsocketserver.h"
@@ -36,15 +37,16 @@ const in6_addr kInitialNextIPv6 = { { {
       0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 2
     } } };
 
-const uint16 kFirstEphemeralPort = 49152;
-const uint16 kLastEphemeralPort = 65535;
-const uint16 kEphemeralPortCount = kLastEphemeralPort - kFirstEphemeralPort + 1;
-const uint32 kDefaultNetworkCapacity = 64 * 1024;
-const uint32 kDefaultTcpBufferSize = 32 * 1024;
+const uint16_t kFirstEphemeralPort = 49152;
+const uint16_t kLastEphemeralPort = 65535;
+const uint16_t kEphemeralPortCount =
+    kLastEphemeralPort - kFirstEphemeralPort + 1;
+const uint32_t kDefaultNetworkCapacity = 64 * 1024;
+const uint32_t kDefaultTcpBufferSize = 32 * 1024;
 
-const uint32 UDP_HEADER_SIZE = 28;  // IP + UDP headers
-const uint32 TCP_HEADER_SIZE = 40;  // IP + TCP headers
-const uint32 TCP_MSS = 1400;  // Maximum segment size
+const uint32_t UDP_HEADER_SIZE = 28;  // IP + UDP headers
+const uint32_t TCP_HEADER_SIZE = 40;  // IP + TCP headers
+const uint32_t TCP_MSS = 1400;        // Maximum segment size
 
 // Note: The current algorithm doesn't work for sample sizes smaller than this.
 const int NUM_SAMPLES = 1000;
@@ -97,7 +99,6 @@ VirtualSocket::VirtualSocket(VirtualSocketServer* server,
                              int type,
                              bool async)
     : server_(server),
-      family_(family),
       type_(type),
       async_(async),
       state_(CS_CLOSED),
@@ -375,7 +376,7 @@ int VirtualSocket::SetOption(Option opt, int value) {
   return 0;  // 0 is success to emulate setsockopt()
 }
 
-int VirtualSocket::EstimateMTU(uint16* mtu) {
+int VirtualSocket::EstimateMTU(uint16_t* mtu) {
   if (CS_CONNECTED != state_)
     return ENOTCONN;
   else
@@ -384,7 +385,7 @@ int VirtualSocket::EstimateMTU(uint16* mtu) {
 
 void VirtualSocket::OnMessage(Message* pmsg) {
   if (pmsg->message_id == MSG_ID_PACKET) {
-    // ASSERT(!local_addr_.IsAny());
+    // ASSERT(!local_addr_.IsAnyIP());
     ASSERT(NULL != pmsg->pdata);
     Packet* packet = static_cast<Packet*>(pmsg->pdata);
 
@@ -532,15 +533,15 @@ IPAddress VirtualSocketServer::GetNextIP(int family) {
     return next_ip;
   } else if (family == AF_INET6) {
     IPAddress next_ip(next_ipv6_);
-    uint32* as_ints = reinterpret_cast<uint32*>(&next_ipv6_.s6_addr);
+    uint32_t* as_ints = reinterpret_cast<uint32_t*>(&next_ipv6_.s6_addr);
     as_ints[3] += 1;
     return next_ip;
   }
   return IPAddress();
 }
 
-uint16 VirtualSocketServer::GetNextPort() {
-  uint16 port = next_port_;
+uint16_t VirtualSocketServer::GetNextPort() {
+  uint16_t port = next_port_;
   if (next_port_ < kLastEphemeralPort) {
     ++next_port_;
   } else {
@@ -602,10 +603,26 @@ bool VirtualSocketServer::ProcessMessagesUntilIdle() {
   return !msg_queue_->IsQuitting();
 }
 
-void VirtualSocketServer::SetNextPortForTesting(uint16 port) {
+void VirtualSocketServer::SetNextPortForTesting(uint16_t port) {
   next_port_ = port;
 }
 
+bool VirtualSocketServer::CloseTcpConnections(
+    const SocketAddress& addr_local,
+    const SocketAddress& addr_remote) {
+  VirtualSocket* socket = LookupConnection(addr_local, addr_remote);
+  if (!socket) {
+    return false;
+  }
+  // Signal the close event on the local connection first.
+  socket->SignalCloseEvent(socket, 0);
+
+  // Trigger the remote connection's close event.
+  socket->Close();
+
+  return true;
+}
+
 int VirtualSocketServer::Bind(VirtualSocket* socket,
                               const SocketAddress& addr) {
   ASSERT(NULL != socket);
@@ -645,7 +662,22 @@ VirtualSocket* VirtualSocketServer::LookupBinding(const SocketAddress& addr) {
   SocketAddress normalized(addr.ipaddr().Normalized(),
                            addr.port());
   AddressMap::iterator it = bindings_->find(normalized);
-  return (bindings_->end() != it) ? it->second : NULL;
+  if (it != bindings_->end()) {
+    return it->second;
+  }
+
+  IPAddress default_ip = GetDefaultRoute(addr.ipaddr().family());
+  if (!IPIsUnspec(default_ip) && addr.ipaddr() == default_ip) {
+    // If we can't find a binding for the packet which is sent to the interface
+    // corresponding to the default route, it should match a binding with the
+    // correct port to the any address.
+    SocketAddress sock_addr =
+        EmptySocketAddressWithFamily(addr.ipaddr().family());
+    sock_addr.SetPort(addr.port());
+    return LookupBinding(sock_addr);
+  }
+
+  return nullptr;
 }
 
 int VirtualSocketServer::Unbind(const SocketAddress& addr,
@@ -700,7 +732,7 @@ static double Random() {
 int VirtualSocketServer::Connect(VirtualSocket* socket,
                                  const SocketAddress& remote_addr,
                                  bool use_delay) {
-  uint32 delay = use_delay ? GetRandomTransitDelay() : 0;
+  uint32_t delay = use_delay ? GetRandomTransitDelay() : 0;
   VirtualSocket* remote = LookupBinding(remote_addr);
   if (!CanInteractWith(socket, remote)) {
     LOG(LS_INFO) << "Address family mismatch between "
@@ -759,7 +791,7 @@ int VirtualSocketServer::SendUdp(VirtualSocket* socket,
 
   CritScope cs(&socket->crit_);
 
-  uint32 cur_time = Time();
+  uint32_t cur_time = Time();
   PurgeNetworkPackets(socket, cur_time);
 
   // Determine whether we have enough bandwidth to accept this packet.  To do
@@ -799,7 +831,7 @@ void VirtualSocketServer::SendTcp(VirtualSocket* socket) {
 
   CritScope cs(&socket->crit_);
 
-  uint32 cur_time = Time();
+  uint32_t cur_time = Time();
   PurgeNetworkPackets(socket, cur_time);
 
   while (true) {
@@ -834,7 +866,7 @@ void VirtualSocketServer::SendTcp(VirtualSocket* socket) {
 
 void VirtualSocketServer::AddPacketToNetwork(VirtualSocket* sender,
                                              VirtualSocket* recipient,
-                                             uint32 cur_time,
+                                             uint32_t cur_time,
                                              const char* data,
                                              size_t data_size,
                                              size_t header_size,
@@ -843,16 +875,26 @@ void VirtualSocketServer::AddPacketToNetwork(VirtualSocket* sender,
   entry.size = data_size + header_size;
 
   sender->network_size_ += entry.size;
-  uint32 send_delay = SendDelay(static_cast<uint32>(sender->network_size_));
+  uint32_t send_delay = SendDelay(static_cast<uint32_t>(sender->network_size_));
   entry.done_time = cur_time + send_delay;
   sender->network_.push_back(entry);
 
   // Find the delay for crossing the many virtual hops of the network.
-  uint32 transit_delay = GetRandomTransitDelay();
+  uint32_t transit_delay = GetRandomTransitDelay();
+
+  // When the incoming packet is from a binding of the any address, translate it
+  // to the default route here such that the recipient will see the default
+  // route.
+  SocketAddress sender_addr = sender->local_addr_;
+  IPAddress default_ip = GetDefaultRoute(sender_addr.ipaddr().family());
+  if (sender_addr.IsAnyIP() && !IPIsUnspec(default_ip)) {
+    sender_addr.SetIP(default_ip);
+  }
 
   // Post the packet as a message to be delivered (on our own thread)
-  Packet* p = new Packet(data, data_size, sender->local_addr_);
-  uint32 ts = TimeAfter(send_delay + transit_delay);
+  Packet* p = new Packet(data, data_size, sender_addr);
+
+  uint32_t ts = TimeAfter(send_delay + transit_delay);
   if (ordered) {
     // Ensure that new packets arrive after previous ones
     // TODO: consider ordering on a per-socket basis, since this
@@ -864,7 +906,7 @@ void VirtualSocketServer::AddPacketToNetwork(VirtualSocket* sender,
 }
 
 void VirtualSocketServer::PurgeNetworkPackets(VirtualSocket* socket,
-                                              uint32 cur_time) {
+                                              uint32_t cur_time) {
   while (!socket->network_.empty() &&
          (socket->network_.front().done_time <= cur_time)) {
     ASSERT(socket->network_size_ >= socket->network_.front().size);
@@ -873,7 +915,7 @@ void VirtualSocketServer::PurgeNetworkPackets(VirtualSocket* socket,
   }
 }
 
-uint32 VirtualSocketServer::SendDelay(uint32 size) {
+uint32_t VirtualSocketServer::SendDelay(uint32_t size) {
   if (bandwidth_ == 0)
     return 0;
   else
@@ -884,14 +926,14 @@ uint32 VirtualSocketServer::SendDelay(uint32 size) {
 void PrintFunction(std::vector<std::pair<double, double> >* f) {
   return;
   double sum = 0;
-  for (uint32 i = 0; i < f->size(); ++i) {
+  for (uint32_t i = 0; i < f->size(); ++i) {
     std::cout << (*f)[i].first << '\t' << (*f)[i].second << std::endl;
     sum += (*f)[i].second;
   }
   if (!f->empty()) {
     const double mean = sum / f->size();
     double sum_sq_dev = 0;
-    for (uint32 i = 0; i < f->size(); ++i) {
+    for (uint32_t i = 0; i < f->size(); ++i) {
       double dev = (*f)[i].second - mean;
       sum_sq_dev += dev * dev;
     }
@@ -929,7 +971,9 @@ static double Pareto(double x, double min, double k) {
 #endif
 
 VirtualSocketServer::Function* VirtualSocketServer::CreateDistribution(
-    uint32 mean, uint32 stddev, uint32 samples) {
+    uint32_t mean,
+    uint32_t stddev,
+    uint32_t samples) {
   Function* f = new Function();
 
   if (0 == stddev) {
@@ -940,7 +984,7 @@ VirtualSocketServer::Function* VirtualSocketServer::CreateDistribution(
       start = mean - 4 * static_cast<double>(stddev);
     double end = mean + 4 * static_cast<double>(stddev);
 
-    for (uint32 i = 0; i < samples; i++) {
+    for (uint32_t i = 0; i < samples; i++) {
       double x = start + (end - start) * i / (samples - 1);
       double y = Normal(x, mean, stddev);
       f->push_back(Point(x, y));
@@ -949,11 +993,11 @@ VirtualSocketServer::Function* VirtualSocketServer::CreateDistribution(
   return Resample(Invert(Accumulate(f)), 0, 1, samples);
 }
 
-uint32 VirtualSocketServer::GetRandomTransitDelay() {
+uint32_t VirtualSocketServer::GetRandomTransitDelay() {
   size_t index = rand() % delay_dist_->size();
   double delay = (*delay_dist_)[index].second;
   //LOG_F(LS_INFO) << "random[" << index << "] = " << delay;
-  return static_cast<uint32>(delay);
+  return static_cast<uint32_t>(delay);
 }
 
 struct FunctionDomainCmp {
@@ -990,8 +1034,10 @@ VirtualSocketServer::Function* VirtualSocketServer::Invert(Function* f) {
   return f;
 }
 
-VirtualSocketServer::Function* VirtualSocketServer::Resample(
-    Function* f, double x1, double x2, uint32 samples) {
+VirtualSocketServer::Function* VirtualSocketServer::Resample(Function* f,
+                                                             double x1,
+                                                             double x2,
+                                                             uint32_t samples) {
   Function* g = new Function();
 
   for (size_t i = 0; i < samples; i++) {
@@ -1064,4 +1110,22 @@ bool VirtualSocketServer::CanInteractWith(VirtualSocket* local,
   return false;
 }
 
+IPAddress VirtualSocketServer::GetDefaultRoute(int family) {
+  if (family == AF_INET) {
+    return default_route_v4_;
+  }
+  if (family == AF_INET6) {
+    return default_route_v6_;
+  }
+  return IPAddress();
+}
+void VirtualSocketServer::SetDefaultRoute(const IPAddress& from_addr) {
+  RTC_DCHECK(!IPIsAny(from_addr));
+  if (from_addr.family() == AF_INET) {
+    default_route_v4_ = from_addr;
+  } else if (from_addr.family() == AF_INET6) {
+    default_route_v6_ = from_addr;
+  }
+}
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/virtualsocketserver.h b/media/webrtc/trunk/webrtc/base/virtualsocketserver.h
index f2c8bd765d..daf0145a26 100644
--- a/media/webrtc/trunk/webrtc/base/virtualsocketserver.h
+++ b/media/webrtc/trunk/webrtc/base/virtualsocketserver.h
@@ -38,41 +38,42 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
 
   SocketServer* socketserver() { return server_; }
 
+  // The default route indicates which local address to use when a socket is
+  // bound to the 'any' address, e.g. 0.0.0.0.
+  IPAddress GetDefaultRoute(int family);
+  void SetDefaultRoute(const IPAddress& from_addr);
+
   // Limits the network bandwidth (maximum bytes per second).  Zero means that
   // all sends occur instantly.  Defaults to 0.
-  uint32 bandwidth() const { return bandwidth_; }
-  void set_bandwidth(uint32 bandwidth) { bandwidth_ = bandwidth; }
+  uint32_t bandwidth() const { return bandwidth_; }
+  void set_bandwidth(uint32_t bandwidth) { bandwidth_ = bandwidth; }
 
   // Limits the amount of data which can be in flight on the network without
   // packet loss (on a per sender basis).  Defaults to 64 KB.
-  uint32 network_capacity() const { return network_capacity_; }
-  void set_network_capacity(uint32 capacity) {
-    network_capacity_ = capacity;
-  }
+  uint32_t network_capacity() const { return network_capacity_; }
+  void set_network_capacity(uint32_t capacity) { network_capacity_ = capacity; }
 
   // The amount of data which can be buffered by tcp on the sender's side
-  uint32 send_buffer_capacity() const { return send_buffer_capacity_; }
-  void set_send_buffer_capacity(uint32 capacity) {
+  uint32_t send_buffer_capacity() const { return send_buffer_capacity_; }
+  void set_send_buffer_capacity(uint32_t capacity) {
     send_buffer_capacity_ = capacity;
   }
 
   // The amount of data which can be buffered by tcp on the receiver's side
-  uint32 recv_buffer_capacity() const { return recv_buffer_capacity_; }
-  void set_recv_buffer_capacity(uint32 capacity) {
+  uint32_t recv_buffer_capacity() const { return recv_buffer_capacity_; }
+  void set_recv_buffer_capacity(uint32_t capacity) {
     recv_buffer_capacity_ = capacity;
   }
 
   // Controls the (transit) delay for packets sent in the network.  This does
   // not inclue the time required to sit in the send queue.  Both of these
   // values are measured in milliseconds.  Defaults to no delay.
-  uint32 delay_mean() const { return delay_mean_; }
-  uint32 delay_stddev() const { return delay_stddev_; }
-  uint32 delay_samples() const { return delay_samples_; }
-  void set_delay_mean(uint32 delay_mean) { delay_mean_ = delay_mean; }
-  void set_delay_stddev(uint32 delay_stddev) {
-    delay_stddev_ = delay_stddev;
-  }
-  void set_delay_samples(uint32 delay_samples) {
+  uint32_t delay_mean() const { return delay_mean_; }
+  uint32_t delay_stddev() const { return delay_stddev_; }
+  uint32_t delay_samples() const { return delay_samples_; }
+  void set_delay_mean(uint32_t delay_mean) { delay_mean_ = delay_mean; }
+  void set_delay_stddev(uint32_t delay_stddev) { delay_stddev_ = delay_stddev; }
+  void set_delay_samples(uint32_t delay_samples) {
     delay_samples_ = delay_samples;
   }
 
@@ -103,8 +104,9 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   typedef std::pair<double, double> Point;
   typedef std::vector<Point> Function;
 
-  static Function* CreateDistribution(uint32 mean, uint32 stddev,
-                                      uint32 samples);
+  static Function* CreateDistribution(uint32_t mean,
+                                      uint32_t stddev,
+                                      uint32_t samples);
 
   // Similar to Thread::ProcessMessages, but it only processes messages until
   // there are no immediate messages or pending network traffic.  Returns false
@@ -112,12 +114,17 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   bool ProcessMessagesUntilIdle();
 
   // Sets the next port number to use for testing.
-  void SetNextPortForTesting(uint16 port);
+  void SetNextPortForTesting(uint16_t port);
+
+  // Close a pair of Tcp connections by addresses. Both connections will have
+  // its own OnClose invoked.
+  bool CloseTcpConnections(const SocketAddress& addr_local,
+                           const SocketAddress& addr_remote);
 
  protected:
   // Returns a new IP not used before in this network.
   IPAddress GetNextIP(int family);
-  uint16 GetNextPort();
+  uint16_t GetNextPort();
 
   VirtualSocket* CreateSocketInternal(int family, int type);
 
@@ -159,24 +166,31 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   void SendTcp(VirtualSocket* socket);
 
   // Places a packet on the network.
-  void AddPacketToNetwork(VirtualSocket* socket, VirtualSocket* recipient,
-                          uint32 cur_time, const char* data, size_t data_size,
-                          size_t header_size, bool ordered);
+  void AddPacketToNetwork(VirtualSocket* socket,
+                          VirtualSocket* recipient,
+                          uint32_t cur_time,
+                          const char* data,
+                          size_t data_size,
+                          size_t header_size,
+                          bool ordered);
 
   // Removes stale packets from the network
-  void PurgeNetworkPackets(VirtualSocket* socket, uint32 cur_time);
+  void PurgeNetworkPackets(VirtualSocket* socket, uint32_t cur_time);
 
   // Computes the number of milliseconds required to send a packet of this size.
-  uint32 SendDelay(uint32 size);
+  uint32_t SendDelay(uint32_t size);
 
   // Returns a random transit delay chosen from the appropriate distribution.
-  uint32 GetRandomTransitDelay();
+  uint32_t GetRandomTransitDelay();
 
   // Basic operations on functions.  Those that return a function also take
   // ownership of the function given (and hence, may modify or delete it).
   static Function* Accumulate(Function* f);
   static Function* Invert(Function* f);
-  static Function* Resample(Function* f, double x1, double x2, uint32 samples);
+  static Function* Resample(Function* f,
+                            double x1,
+                            double x2,
+                            uint32_t samples);
   static double Evaluate(Function* f, double x);
 
   // NULL out our message queue if it goes away. Necessary in the case where
@@ -212,25 +226,28 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   bool server_owned_;
   MessageQueue* msg_queue_;
   bool stop_on_idle_;
-  uint32 network_delay_;
+  uint32_t network_delay_;
   in_addr next_ipv4_;
   in6_addr next_ipv6_;
-  uint16 next_port_;
+  uint16_t next_port_;
   AddressMap* bindings_;
   ConnectionMap* connections_;
 
-  uint32 bandwidth_;
-  uint32 network_capacity_;
-  uint32 send_buffer_capacity_;
-  uint32 recv_buffer_capacity_;
-  uint32 delay_mean_;
-  uint32 delay_stddev_;
-  uint32 delay_samples_;
+  IPAddress default_route_v4_;
+  IPAddress default_route_v6_;
+
+  uint32_t bandwidth_;
+  uint32_t network_capacity_;
+  uint32_t send_buffer_capacity_;
+  uint32_t recv_buffer_capacity_;
+  uint32_t delay_mean_;
+  uint32_t delay_stddev_;
+  uint32_t delay_samples_;
   Function* delay_dist_;
   CriticalSection delay_crit_;
 
   double drop_prob_;
-  DISALLOW_EVIL_CONSTRUCTORS(VirtualSocketServer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(VirtualSocketServer);
 };
 
 // Implements the socket interface using the virtual network.  Packets are
@@ -243,9 +260,6 @@ class VirtualSocket : public AsyncSocket, public MessageHandler {
   SocketAddress GetLocalAddress() const override;
   SocketAddress GetRemoteAddress() const override;
 
-  // Used by server sockets to set the local address without binding.
-  void SetLocalAddress(const SocketAddress& addr);
-
   // Used by TurnPortTest to mimic a case where proxy returns local host address
   // instead of the original one TurnPort was bound against. Please see WebRTC
   // issue 3927 for more detail.
@@ -266,7 +280,7 @@ class VirtualSocket : public AsyncSocket, public MessageHandler {
   ConnState GetState() const override;
   int GetOption(Option opt, int* value) override;
   int SetOption(Option opt, int value) override;
-  int EstimateMTU(uint16* mtu) override;
+  int EstimateMTU(uint16_t* mtu) override;
   void OnMessage(Message* pmsg) override;
 
   bool was_any() { return was_any_; }
@@ -278,7 +292,7 @@ class VirtualSocket : public AsyncSocket, public MessageHandler {
  private:
   struct NetworkEntry {
     size_t size;
-    uint32 done_time;
+    uint32_t done_time;
   };
 
   typedef std::deque<SocketAddress> ListenQueue;
@@ -292,8 +306,10 @@ class VirtualSocket : public AsyncSocket, public MessageHandler {
   int SendUdp(const void* pv, size_t cb, const SocketAddress& addr);
   int SendTcp(const void* pv, size_t cb);
 
+  // Used by server sockets to set the local address without binding.
+  void SetLocalAddress(const SocketAddress& addr);
+
   VirtualSocketServer* server_;
-  int family_;
   int type_;
   bool async_;
   ConnState state_;
diff --git a/media/webrtc/trunk/webrtc/base/win32.cc b/media/webrtc/trunk/webrtc/base/win32.cc
index c1b55bf63e..182b84f482 100644
--- a/media/webrtc/trunk/webrtc/base/win32.cc
+++ b/media/webrtc/trunk/webrtc/base/win32.cc
@@ -14,6 +14,7 @@
 #include <ws2tcpip.h>
 #include <algorithm>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
@@ -82,13 +83,12 @@ const char* inet_ntop_v6(const void* src, char* dst, socklen_t size) {
   if (size < INET6_ADDRSTRLEN) {
     return NULL;
   }
-  const uint16* as_shorts =
-      reinterpret_cast<const uint16*>(src);
+  const uint16_t* as_shorts = reinterpret_cast<const uint16_t*>(src);
   int runpos[8];
   int current = 1;
   int max = 0;
   int maxpos = -1;
-  int run_array_size = ARRAY_SIZE(runpos);
+  int run_array_size = arraysize(runpos);
   // Run over the address marking runs of 0s.
   for (int i = 0; i < run_array_size; ++i) {
     if (as_shorts[i] == 0) {
@@ -214,8 +214,8 @@ int inet_pton_v6(const char* src, void* dst) {
   struct in6_addr an_addr;
   memset(&an_addr, 0, sizeof(an_addr));
 
-  uint16* addr_cursor = reinterpret_cast<uint16*>(&an_addr.s6_addr[0]);
-  uint16* addr_end = reinterpret_cast<uint16*>(&an_addr.s6_addr[16]);
+  uint16_t* addr_cursor = reinterpret_cast<uint16_t*>(&an_addr.s6_addr[0]);
+  uint16_t* addr_end = reinterpret_cast<uint16_t*>(&an_addr.s6_addr[16]);
   bool seencompressed = false;
 
   // Addresses that start with "::" (i.e., a run of initial zeros) or
@@ -228,7 +228,7 @@ int inet_pton_v6(const char* src, void* dst) {
     if (rtc::strchr(addrstart, ".")) {
       const char* colon = rtc::strchr(addrstart, "::");
       if (colon) {
-        uint16 a_short;
+        uint16_t a_short;
         int bytesread = 0;
         if (sscanf(addrstart, "%hx%n", &a_short, &bytesread) != 1 ||
             a_short != 0xFFFF || bytesread != 4) {
@@ -283,7 +283,7 @@ int inet_pton_v6(const char* src, void* dst) {
         ++readcursor;
       }
     } else {
-      uint16 word;
+      uint16_t word;
       int bytesread = 0;
       if (sscanf(readcursor, "%hx%n", &word, &bytesread) != 1) {
         return 0;
@@ -362,7 +362,7 @@ void UnixTimeToFileTime(const time_t& ut, FILETIME* ft) {
   // base date value.
   const ULONGLONG RATIO = 10000000;
   ULARGE_INTEGER current_ul;
-  current_ul.QuadPart = base_ul.QuadPart + static_cast<int64>(ut) * RATIO;
+  current_ul.QuadPart = base_ul.QuadPart + static_cast<int64_t>(ut) * RATIO;
   memcpy(ft, &current_ul, sizeof(FILETIME));
 }
 
@@ -453,4 +453,5 @@ bool GetCurrentProcessIntegrityLevel(int* level) {
   }
   return ret;
 }
+
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/win32.h b/media/webrtc/trunk/webrtc/base/win32.h
index 07e1e1ea51..dba9b773b5 100644
--- a/media/webrtc/trunk/webrtc/base/win32.h
+++ b/media/webrtc/trunk/webrtc/base/win32.h
@@ -46,8 +46,6 @@ namespace rtc {
 const char* win32_inet_ntop(int af, const void *src, char* dst, socklen_t size);
 int win32_inet_pton(int af, const char* src, void *dst);
 
-///////////////////////////////////////////////////////////////////////////////
-
 inline std::wstring ToUtf16(const char* utf8, size_t len) {
   int len16 = ::MultiByteToWideChar(CP_UTF8, 0, utf8, static_cast<int>(len),
                                     NULL, 0);
@@ -87,8 +85,8 @@ void UnixTimeToFileTime(const time_t& ut, FILETIME * ft);
 bool Utf8ToWindowsFilename(const std::string& utf8, std::wstring* filename);
 
 // Convert a FILETIME to a UInt64
-inline uint64 ToUInt64(const FILETIME& ft) {
-  ULARGE_INTEGER r = {ft.dwLowDateTime, ft.dwHighDateTime};
+inline uint64_t ToUInt64(const FILETIME& ft) {
+  ULARGE_INTEGER r = {{ft.dwLowDateTime, ft.dwHighDateTime}};
   return r.QuadPart;
 }
 
@@ -128,8 +126,6 @@ inline bool IsCurrentProcessLowIntegrity() {
 
 bool AdjustCurrentProcessPrivilege(const TCHAR* privilege, bool to_enable);
 
-///////////////////////////////////////////////////////////////////////////////
-
 }  // namespace rtc
 
 #endif  // WEBRTC_WIN
diff --git a/media/webrtc/trunk/webrtc/base/win32_unittest.cc b/media/webrtc/trunk/webrtc/base/win32_unittest.cc
index 2bd93acc1c..15b2614111 100644
--- a/media/webrtc/trunk/webrtc/base/win32_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/win32_unittest.cc
@@ -32,7 +32,7 @@ TEST_F(Win32Test, FileTimeToUInt64Test) {
   ft.dwHighDateTime = 0xBAADF00D;
   ft.dwLowDateTime = 0xFEED3456;
 
-  uint64 expected = 0xBAADF00DFEED3456;
+  uint64_t expected = 0xBAADF00DFEED3456;
   EXPECT_EQ(expected, ToUInt64(ft));
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/win32filesystem.cc b/media/webrtc/trunk/webrtc/base/win32filesystem.cc
index 9ca4c996d6..b731974bac 100644
--- a/media/webrtc/trunk/webrtc/base/win32filesystem.cc
+++ b/media/webrtc/trunk/webrtc/base/win32filesystem.cc
@@ -15,6 +15,7 @@
 #include <shlobj.h>
 #include <tchar.h>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/pathutils.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -197,16 +198,16 @@ bool Win32Filesystem::DeleteEmptyFolder(const Pathname &folder) {
 bool Win32Filesystem::GetTemporaryFolder(Pathname &pathname, bool create,
                                          const std::string *append) {
   wchar_t buffer[MAX_PATH + 1];
-  if (!::GetTempPath(ARRAY_SIZE(buffer), buffer))
+  if (!::GetTempPath(arraysize(buffer), buffer))
     return false;
   if (!IsCurrentProcessLowIntegrity() &&
-      !::GetLongPathName(buffer, buffer, ARRAY_SIZE(buffer)))
+      !::GetLongPathName(buffer, buffer, arraysize(buffer)))
     return false;
   size_t len = strlen(buffer);
   if ((len > 0) && (buffer[len-1] != '\\')) {
-    len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len, L"\\");
+    len += strcpyn(buffer + len, arraysize(buffer) - len, L"\\");
   }
-  if (len >= ARRAY_SIZE(buffer) - 1)
+  if (len >= arraysize(buffer) - 1)
     return false;
   pathname.clear();
   pathname.SetFolder(ToUtf8(buffer));
@@ -295,10 +296,10 @@ bool Win32Filesystem::CopyFile(const Pathname &old_path,
 
 bool Win32Filesystem::IsTemporaryPath(const Pathname& pathname) {
   TCHAR buffer[MAX_PATH + 1];
-  if (!::GetTempPath(ARRAY_SIZE(buffer), buffer))
+  if (!::GetTempPath(arraysize(buffer), buffer))
     return false;
   if (!IsCurrentProcessLowIntegrity() &&
-      !::GetLongPathName(buffer, buffer, ARRAY_SIZE(buffer)))
+      !::GetLongPathName(buffer, buffer, arraysize(buffer)))
     return false;
   return (::strnicmp(ToUtf16(pathname.pathname()).c_str(),
                      buffer, strlen(buffer)) == 0);
@@ -337,7 +338,7 @@ bool Win32Filesystem::GetFileTime(const Pathname& path, FileTimeType which,
 
 bool Win32Filesystem::GetAppPathname(Pathname* path) {
   TCHAR buffer[MAX_PATH + 1];
-  if (0 == ::GetModuleFileName(NULL, buffer, ARRAY_SIZE(buffer)))
+  if (0 == ::GetModuleFileName(NULL, buffer, arraysize(buffer)))
     return false;
   path->SetPathname(ToUtf8(buffer));
   return true;
@@ -351,20 +352,20 @@ bool Win32Filesystem::GetAppDataFolder(Pathname* path, bool per_user) {
   if (!::SHGetSpecialFolderPath(NULL, buffer, csidl, TRUE))
     return false;
   if (!IsCurrentProcessLowIntegrity() &&
-      !::GetLongPathName(buffer, buffer, ARRAY_SIZE(buffer)))
+      !::GetLongPathName(buffer, buffer, arraysize(buffer)))
     return false;
-  size_t len = strcatn(buffer, ARRAY_SIZE(buffer), __T("\\"));
-  len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len,
+  size_t len = strcatn(buffer, arraysize(buffer), __T("\\"));
+  len += strcpyn(buffer + len, arraysize(buffer) - len,
                  ToUtf16(organization_name_).c_str());
   if ((len > 0) && (buffer[len-1] != __T('\\'))) {
-    len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len, __T("\\"));
+    len += strcpyn(buffer + len, arraysize(buffer) - len, __T("\\"));
   }
-  len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len,
+  len += strcpyn(buffer + len, arraysize(buffer) - len,
                  ToUtf16(application_name_).c_str());
   if ((len > 0) && (buffer[len-1] != __T('\\'))) {
-    len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len, __T("\\"));
+    len += strcpyn(buffer + len, arraysize(buffer) - len, __T("\\"));
   }
-  if (len >= ARRAY_SIZE(buffer) - 1)
+  if (len >= arraysize(buffer) - 1)
     return false;
   path->clear();
   path->SetFolder(ToUtf8(buffer));
@@ -379,7 +380,7 @@ bool Win32Filesystem::GetAppTempFolder(Pathname* path) {
 }
 
 bool Win32Filesystem::GetDiskFreeSpace(const Pathname& path,
-                                       int64 *free_bytes) {
+                                       int64_t* free_bytes) {
   if (!free_bytes) {
     return false;
   }
@@ -405,11 +406,11 @@ bool Win32Filesystem::GetDiskFreeSpace(const Pathname& path,
     return false;
   }
 
-  int64 total_number_of_bytes;  // receives the number of bytes on disk
-  int64 total_number_of_free_bytes;  // receives the free bytes on disk
+  int64_t total_number_of_bytes;       // receives the number of bytes on disk
+  int64_t total_number_of_free_bytes;  // receives the free bytes on disk
   // make sure things won't change in 64 bit machine
   // TODO replace with compile time assert
-  ASSERT(sizeof(ULARGE_INTEGER) == sizeof(uint64));  //NOLINT
+  ASSERT(sizeof(ULARGE_INTEGER) == sizeof(uint64_t));  // NOLINT
   if (::GetDiskFreeSpaceEx(target_drive,
                            (PULARGE_INTEGER)free_bytes,
                            (PULARGE_INTEGER)&total_number_of_bytes,
diff --git a/media/webrtc/trunk/webrtc/base/win32filesystem.h b/media/webrtc/trunk/webrtc/base/win32filesystem.h
index 0ae921843e..439b2c6268 100644
--- a/media/webrtc/trunk/webrtc/base/win32filesystem.h
+++ b/media/webrtc/trunk/webrtc/base/win32filesystem.h
@@ -91,7 +91,7 @@ class Win32Filesystem : public FilesystemInterface {
   // Get a temporary folder that is unique to the current user and application.
   virtual bool GetAppTempFolder(Pathname* path);
 
-  virtual bool GetDiskFreeSpace(const Pathname& path, int64 *free_bytes);
+  virtual bool GetDiskFreeSpace(const Pathname& path, int64_t* free_bytes);
 
   virtual Pathname GetCurrentDirectory();
 };
diff --git a/media/webrtc/trunk/webrtc/base/win32regkey.cc b/media/webrtc/trunk/webrtc/base/win32regkey.cc
index 1ed0d4ea29..ccf931c14a 100644
--- a/media/webrtc/trunk/webrtc/base/win32regkey.cc
+++ b/media/webrtc/trunk/webrtc/base/win32regkey.cc
@@ -100,22 +100,22 @@ HRESULT RegKey::SetValue(const wchar_t* full_key_name,
 
 HRESULT RegKey::SetValue(const wchar_t* full_key_name,
                          const wchar_t* value_name,
-                         const uint8* value,
+                         const uint8_t* value,
                          DWORD byte_count) {
   ASSERT(full_key_name != NULL);
 
   return SetValueStaticHelper(full_key_name, value_name, REG_BINARY,
-                              const_cast<uint8*>(value), byte_count);
+                              const_cast<uint8_t*>(value), byte_count);
 }
 
 HRESULT RegKey::SetValueMultiSZ(const wchar_t* full_key_name,
                                 const wchar_t* value_name,
-                                const uint8* value,
+                                const uint8_t* value,
                                 DWORD byte_count) {
   ASSERT(full_key_name != NULL);
 
   return SetValueStaticHelper(full_key_name, value_name, REG_MULTI_SZ,
-                              const_cast<uint8*>(value), byte_count);
+                              const_cast<uint8_t*>(value), byte_count);
 }
 
 HRESULT RegKey::GetValue(const wchar_t* full_key_name,
@@ -208,7 +208,7 @@ HRESULT RegKey::GetValue(const wchar_t* full_key_name,
 
 HRESULT RegKey::GetValue(const wchar_t* full_key_name,
                          const wchar_t* value_name,
-                         uint8** value,
+                         uint8_t** value,
                          DWORD* byte_count) {
   ASSERT(full_key_name != NULL);
   ASSERT(value != NULL);
@@ -407,11 +407,11 @@ HRESULT RegKey::SetValueStaticHelper(const wchar_t* full_key_name,
           hr = key.SetValue(value_name, static_cast<const wchar_t*>(value));
           break;
         case REG_BINARY:
-          hr = key.SetValue(value_name, static_cast<const uint8*>(value),
+          hr = key.SetValue(value_name, static_cast<const uint8_t*>(value),
                             byte_count);
           break;
         case REG_MULTI_SZ:
-          hr = key.SetValue(value_name, static_cast<const uint8*>(value),
+          hr = key.SetValue(value_name, static_cast<const uint8_t*>(value),
                             byte_count, type);
           break;
         default:
@@ -461,7 +461,7 @@ HRESULT RegKey::GetValueStaticHelper(const wchar_t* full_key_name,
                                             std::vector<std::wstring>*>(value));
           break;
         case REG_BINARY:
-          hr = key.GetValue(value_name, reinterpret_cast<uint8**>(value),
+          hr = key.GetValue(value_name, reinterpret_cast<uint8_t**>(value),
                             byte_count);
           break;
         default:
@@ -482,7 +482,7 @@ HRESULT RegKey::GetValueStaticHelper(const wchar_t* full_key_name,
 // GET helper
 HRESULT RegKey::GetValueHelper(const wchar_t* value_name,
                                DWORD* type,
-                               uint8** value,
+                               uint8_t** value,
                                DWORD* byte_count) const {
   ASSERT(byte_count != NULL);
   ASSERT(value != NULL);
@@ -608,7 +608,7 @@ HRESULT RegKey::GetValue(const wchar_t* value_name, std::wstring* value) const {
 }
 
 // convert REG_MULTI_SZ bytes to string array
-HRESULT RegKey::MultiSZBytesToStringArray(const uint8* buffer,
+HRESULT RegKey::MultiSZBytesToStringArray(const uint8_t* buffer,
                                           DWORD byte_count,
                                           std::vector<std::wstring>* value) {
   ASSERT(buffer != NULL);
@@ -640,7 +640,7 @@ HRESULT RegKey::GetValue(const wchar_t* value_name,
 
   DWORD byte_count = 0;
   DWORD type = 0;
-  uint8* buffer = 0;
+  uint8_t* buffer = 0;
 
   // first get the size of the buffer
   HRESULT hr = GetValueHelper(value_name, &type, &buffer, &byte_count);
@@ -655,7 +655,7 @@ HRESULT RegKey::GetValue(const wchar_t* value_name,
 
 // Binary data Get
 HRESULT RegKey::GetValue(const wchar_t* value_name,
-                         uint8** value,
+                         uint8_t** value,
                          DWORD* byte_count) const {
   ASSERT(byte_count != NULL);
   ASSERT(value != NULL);
@@ -668,9 +668,9 @@ HRESULT RegKey::GetValue(const wchar_t* value_name,
 
 // Raw data get
 HRESULT RegKey::GetValue(const wchar_t* value_name,
-                         uint8** value,
+                         uint8_t** value,
                          DWORD* byte_count,
-                         DWORD*type) const {
+                         DWORD* type) const {
   ASSERT(type != NULL);
   ASSERT(byte_count != NULL);
   ASSERT(value != NULL);
@@ -682,9 +682,9 @@ HRESULT RegKey::GetValue(const wchar_t* value_name,
 HRESULT RegKey::SetValue(const wchar_t* value_name, DWORD value) const {
   ASSERT(h_key_ != NULL);
 
-  LONG res = ::RegSetValueEx(h_key_, value_name, NULL, REG_DWORD,
-                             reinterpret_cast<const uint8*>(&value),
-                             sizeof(DWORD));
+  LONG res =
+      ::RegSetValueEx(h_key_, value_name, NULL, REG_DWORD,
+                      reinterpret_cast<const uint8_t*>(&value), sizeof(DWORD));
   return HRESULT_FROM_WIN32(res);
 }
 
@@ -693,7 +693,7 @@ HRESULT RegKey::SetValue(const wchar_t* value_name, DWORD64 value) const {
   ASSERT(h_key_ != NULL);
 
   LONG res = ::RegSetValueEx(h_key_, value_name, NULL, REG_QWORD,
-                             reinterpret_cast<const uint8*>(&value),
+                             reinterpret_cast<const uint8_t*>(&value),
                              sizeof(DWORD64));
   return HRESULT_FROM_WIN32(res);
 }
@@ -705,14 +705,14 @@ HRESULT RegKey::SetValue(const wchar_t* value_name,
   ASSERT(h_key_ != NULL);
 
   LONG res = ::RegSetValueEx(h_key_, value_name, NULL, REG_SZ,
-                             reinterpret_cast<const uint8*>(value),
+                             reinterpret_cast<const uint8_t*>(value),
                              (lstrlen(value) + 1) * sizeof(wchar_t));
   return HRESULT_FROM_WIN32(res);
 }
 
 // Binary data set
 HRESULT RegKey::SetValue(const wchar_t* value_name,
-                         const uint8* value,
+                         const uint8_t* value,
                          DWORD byte_count) const {
   ASSERT(h_key_ != NULL);
 
@@ -728,7 +728,7 @@ HRESULT RegKey::SetValue(const wchar_t* value_name,
 
 // Raw data set
 HRESULT RegKey::SetValue(const wchar_t* value_name,
-                         const uint8* value,
+                         const uint8_t* value,
                          DWORD byte_count,
                          DWORD type) const {
   ASSERT(value != NULL);
@@ -964,7 +964,7 @@ std::wstring RegKey::GetParentKeyInfo(std::wstring* key_name) {
 }
 
 // get the number of values for this key
-uint32 RegKey::GetValueCount() {
+uint32_t RegKey::GetValueCount() {
   DWORD num_values = 0;
 
   if (ERROR_SUCCESS != ::RegQueryInfoKey(
@@ -1007,7 +1007,7 @@ HRESULT RegKey::GetValueNameAt(int index, std::wstring* value_name,
   return HRESULT_FROM_WIN32(res);
 }
 
-uint32 RegKey::GetSubkeyCount() {
+uint32_t RegKey::GetSubkeyCount() {
   // number of values for key
   DWORD num_subkeys = 0;
 
diff --git a/media/webrtc/trunk/webrtc/base/win32regkey.h b/media/webrtc/trunk/webrtc/base/win32regkey.h
index b33d4dc2b3..d5c51b9b06 100644
--- a/media/webrtc/trunk/webrtc/base/win32regkey.h
+++ b/media/webrtc/trunk/webrtc/base/win32regkey.h
@@ -24,6 +24,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/win32.h"
 
 namespace rtc {
@@ -63,7 +64,7 @@ class RegKey {
   bool HasValue(const wchar_t* value_name) const;
 
   // get the number of values for this key
-  uint32 GetValueCount();
+  uint32_t GetValueCount();
 
   // Called to get the value name for the given value name index
   // Use GetValueCount() to get the total value_name count for this key
@@ -79,7 +80,7 @@ class RegKey {
   bool HasSubkey(const wchar_t* key_name) const;
 
   // get the number of subkeys for this key
-  uint32 GetSubkeyCount();
+  uint32_t GetSubkeyCount();
 
   // Called to get the key name for the given key index
   // Use GetSubkeyCount() to get the total count for this key
@@ -91,10 +92,10 @@ class RegKey {
 
   // SETTERS
 
-  // set an int32 value - use when reading multiple values from a key
+  // set an int32_t value - use when reading multiple values from a key
   HRESULT SetValue(const wchar_t* value_name, DWORD value) const;
 
-  // set an int64 value
+  // set an int64_t value
   HRESULT SetValue(const wchar_t* value_name, DWORD64 value) const;
 
   // set a string value
@@ -102,21 +103,21 @@ class RegKey {
 
   // set binary data
   HRESULT SetValue(const wchar_t* value_name,
-                   const uint8* value,
+                   const uint8_t* value,
                    DWORD byte_count) const;
 
   // set raw data, including type
   HRESULT SetValue(const wchar_t* value_name,
-                   const uint8* value,
+                   const uint8_t* value,
                    DWORD byte_count,
                    DWORD type) const;
 
   // GETTERS
 
-  // get an int32 value
+  // get an int32_t value
   HRESULT GetValue(const wchar_t* value_name, DWORD* value) const;
 
-  // get an int64 value
+  // get an int64_t value
   HRESULT GetValue(const wchar_t* value_name, DWORD64* value) const;
 
   // get a string value - the caller must free the return buffer
@@ -131,12 +132,12 @@ class RegKey {
 
   // get binary data - the caller must free the return buffer
   HRESULT GetValue(const wchar_t* value_name,
-                   uint8** value,
+                   uint8_t** value,
                    DWORD* byte_count) const;
 
   // get raw data, including type - the caller must free the return buffer
   HRESULT GetValue(const wchar_t* value_name,
-                   uint8** value,
+                   uint8_t** value,
                    DWORD* byte_count,
                    DWORD* type) const;
 
@@ -153,12 +154,12 @@ class RegKey {
 
   // SETTERS
 
-  // STATIC int32 set
+  // STATIC int32_t set
   static HRESULT SetValue(const wchar_t* full_key_name,
                           const wchar_t* value_name,
                           DWORD value);
 
-  // STATIC int64 set
+  // STATIC int64_t set
   static HRESULT SetValue(const wchar_t* full_key_name,
                           const wchar_t* value_name,
                           DWORD64 value);
@@ -181,23 +182,23 @@ class RegKey {
   // STATIC binary data set
   static HRESULT SetValue(const wchar_t* full_key_name,
                           const wchar_t* value_name,
-                          const uint8* value,
+                          const uint8_t* value,
                           DWORD byte_count);
 
   // STATIC multi-string set
   static HRESULT SetValueMultiSZ(const wchar_t* full_key_name,
                                  const TCHAR* value_name,
-                                 const uint8* value,
+                                 const uint8_t* value,
                                  DWORD byte_count);
 
   // GETTERS
 
-  // STATIC int32 get
+  // STATIC int32_t get
   static HRESULT GetValue(const wchar_t* full_key_name,
                           const wchar_t* value_name,
                           DWORD* value);
 
-  // STATIC int64 get
+  // STATIC int64_t get
   //
   // Note: if you are using time64 you should
   // likely use GetLimitedTimeValue (util.h) instead of this method.
@@ -232,7 +233,7 @@ class RegKey {
   // STATIC get binary data - the caller must free the return buffer
   static HRESULT GetValue(const wchar_t* full_key_name,
                           const wchar_t* value_name,
-                          uint8** value,
+                          uint8_t** value,
                           DWORD* byte_count);
 
   // Get type of a registry value
@@ -296,7 +297,8 @@ class RegKey {
   // helper function to get any value from the registry
   // used when the size of the data is unknown
   HRESULT GetValueHelper(const wchar_t* value_name,
-                         DWORD* type, uint8** value,
+                         DWORD* type,
+                         uint8_t** value,
                          DWORD* byte_count) const;
 
   // helper function to get the parent key name and the subkey from a string
@@ -319,7 +321,7 @@ class RegKey {
                                       DWORD* byte_count = NULL);
 
   // convert REG_MULTI_SZ bytes to string array
-  static HRESULT MultiSZBytesToStringArray(const uint8* buffer,
+  static HRESULT MultiSZBytesToStringArray(const uint8_t* buffer,
                                            DWORD byte_count,
                                            std::vector<std::wstring>* value);
 
@@ -329,7 +331,7 @@ class RegKey {
   // for unittest
   friend void RegKeyHelperFunctionsTest();
 
-  DISALLOW_EVIL_CONSTRUCTORS(RegKey);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RegKey);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/win32regkey_unittest.cc b/media/webrtc/trunk/webrtc/base/win32regkey_unittest.cc
index d26305147f..1702ef741d 100644
--- a/media/webrtc/trunk/webrtc/base/win32regkey_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/win32regkey_unittest.cc
@@ -10,6 +10,7 @@
 
 // Unittest for registry access API
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/win32regkey.h"
@@ -150,18 +151,18 @@ void RegKeyHelperFunctionsTest() {
 
   std::vector<std::wstring> result;
   EXPECT_SUCCEEDED(RegKey::MultiSZBytesToStringArray(
-      reinterpret_cast<const uint8*>(kMultiSZ), sizeof(kMultiSZ), &result));
+      reinterpret_cast<const uint8_t*>(kMultiSZ), sizeof(kMultiSZ), &result));
   EXPECT_EQ(result.size(), 3);
   EXPECT_STREQ(result[0].c_str(), L"abc");
   EXPECT_STREQ(result[1].c_str(), L"def");
   EXPECT_STREQ(result[2].c_str(), L"P12345");
 
   EXPECT_SUCCEEDED(RegKey::MultiSZBytesToStringArray(
-      reinterpret_cast<const uint8*>(kEmptyMultiSZ),
-      sizeof(kEmptyMultiSZ), &result));
+      reinterpret_cast<const uint8_t*>(kEmptyMultiSZ), sizeof(kEmptyMultiSZ),
+      &result));
   EXPECT_EQ(result.size(), 0);
   EXPECT_FALSE(SUCCEEDED(RegKey::MultiSZBytesToStringArray(
-      reinterpret_cast<const uint8*>(kInvalidMultiSZ),
+      reinterpret_cast<const uint8_t*>(kInvalidMultiSZ),
       sizeof(kInvalidMultiSZ), &result)));
 }
 
@@ -173,7 +174,7 @@ void RegKeyNonStaticFunctionsTest() {
   DWORD int_val = 0;
   DWORD64 int64_val = 0;
   wchar_t* str_val = NULL;
-  uint8* binary_val = NULL;
+  uint8_t* binary_val = NULL;
   DWORD uint8_count = 0;
 
   // Just in case...
@@ -265,7 +266,8 @@ void RegKeyNonStaticFunctionsTest() {
 
   // set a binary value
   EXPECT_SUCCEEDED(r_key.SetValue(kValNameBinary,
-      reinterpret_cast<const uint8*>(kBinaryVal), sizeof(kBinaryVal) - 1));
+                                  reinterpret_cast<const uint8_t*>(kBinaryVal),
+                                  sizeof(kBinaryVal) - 1));
 
   // check that the value exists
   EXPECT_TRUE(r_key.HasValue(kValNameBinary));
@@ -277,7 +279,8 @@ void RegKeyNonStaticFunctionsTest() {
 
   // set it again
   EXPECT_SUCCEEDED(r_key.SetValue(kValNameBinary,
-      reinterpret_cast<const uint8*>(kBinaryVal2), sizeof(kBinaryVal) - 1));
+                                  reinterpret_cast<const uint8_t*>(kBinaryVal2),
+                                  sizeof(kBinaryVal) - 1));
 
   // read it again
   EXPECT_SUCCEEDED(r_key.GetValue(kValNameBinary, &binary_val, &uint8_count));
@@ -303,10 +306,11 @@ void RegKeyNonStaticFunctionsTest() {
 
   // set a binary value
   EXPECT_SUCCEEDED(r_key.SetValue(kValNameBinary,
-      reinterpret_cast<const uint8*>(kBinaryVal), sizeof(kBinaryVal) - 1));
+                                  reinterpret_cast<const uint8_t*>(kBinaryVal),
+                                  sizeof(kBinaryVal) - 1));
 
   // get the value count
-  uint32 value_count = r_key.GetValueCount();
+  uint32_t value_count = r_key.GetValueCount();
   EXPECT_EQ(value_count, 4);
 
   // check the value names
@@ -332,7 +336,7 @@ void RegKeyNonStaticFunctionsTest() {
   // check that there are no more values
   EXPECT_FAILED(r_key.GetValueNameAt(4, &value_name, &type));
 
-  uint32 subkey_count = r_key.GetSubkeyCount();
+  uint32_t subkey_count = r_key.GetSubkeyCount();
   EXPECT_EQ(subkey_count, 0);
 
   // now create a subkey and make sure we can get the name
@@ -366,7 +370,7 @@ void RegKeyStaticFunctionsTest() {
   double double_val = 0;
   wchar_t* str_val = NULL;
   std::wstring wstr_val;
-  uint8* binary_val = NULL;
+  uint8_t* binary_val = NULL;
   DWORD uint8_count = 0;
 
   // Just in case...
@@ -377,7 +381,7 @@ void RegKeyStaticFunctionsTest() {
   EXPECT_EQ(RegKey::GetValue(kFullRkey1, kValNameInt, &int_val),
             HRESULT_FROM_WIN32(ERROR_FILE_NOT_FOUND));
 
-  // set int32
+  // set int32_t
   EXPECT_SUCCEEDED(RegKey::SetValue(kFullRkey1, kValNameInt, kIntVal));
 
   // check that the value exists
@@ -397,7 +401,7 @@ void RegKeyStaticFunctionsTest() {
   // check that the value is gone
   EXPECT_FALSE(RegKey::HasValue(kFullRkey1, kValNameInt));
 
-  // set int64
+  // set int64_t
   EXPECT_SUCCEEDED(RegKey::SetValue(kFullRkey1, kValNameInt64, kIntVal64));
 
   // check that the value exists
@@ -473,8 +477,9 @@ void RegKeyStaticFunctionsTest() {
   EXPECT_FALSE(RegKey::HasValue(kFullRkey1, kValNameStr));
 
   // set binary
-  EXPECT_SUCCEEDED(RegKey::SetValue(kFullRkey1, kValNameBinary,
-      reinterpret_cast<const uint8*>(kBinaryVal), sizeof(kBinaryVal)-1));
+  EXPECT_SUCCEEDED(RegKey::SetValue(
+      kFullRkey1, kValNameBinary, reinterpret_cast<const uint8_t*>(kBinaryVal),
+      sizeof(kBinaryVal) - 1));
 
   // check that the value exists
   EXPECT_TRUE(RegKey::HasValue(kFullRkey1, kValNameBinary));
@@ -492,8 +497,9 @@ void RegKeyStaticFunctionsTest() {
   EXPECT_FALSE(RegKey::HasValue(kFullRkey1, kValNameBinary));
 
   // special case - set a binary value with length 0
-  EXPECT_SUCCEEDED(RegKey::SetValue(kFullRkey1, kValNameBinary,
-      reinterpret_cast<const uint8*>(kBinaryVal), 0));
+  EXPECT_SUCCEEDED(
+      RegKey::SetValue(kFullRkey1, kValNameBinary,
+                       reinterpret_cast<const uint8_t*>(kBinaryVal), 0));
 
   // check that the value exists
   EXPECT_TRUE(RegKey::HasValue(kFullRkey1, kValNameBinary));
@@ -532,20 +538,24 @@ void RegKeyStaticFunctionsTest() {
 
   // test read/write REG_MULTI_SZ value
   std::vector<std::wstring> result;
-  EXPECT_SUCCEEDED(RegKey::SetValueMultiSZ(kFullRkey1, kValNameMultiStr,
-      reinterpret_cast<const uint8*>(kMultiSZ), sizeof(kMultiSZ)));
+  EXPECT_SUCCEEDED(RegKey::SetValueMultiSZ(
+      kFullRkey1, kValNameMultiStr, reinterpret_cast<const uint8_t*>(kMultiSZ),
+      sizeof(kMultiSZ)));
   EXPECT_SUCCEEDED(RegKey::GetValue(kFullRkey1, kValNameMultiStr, &result));
   EXPECT_EQ(result.size(), 3);
   EXPECT_STREQ(result[0].c_str(), L"abc");
   EXPECT_STREQ(result[1].c_str(), L"def");
   EXPECT_STREQ(result[2].c_str(), L"P12345");
-  EXPECT_SUCCEEDED(RegKey::SetValueMultiSZ(kFullRkey1, kValNameMultiStr,
-      reinterpret_cast<const uint8*>(kEmptyMultiSZ), sizeof(kEmptyMultiSZ)));
+  EXPECT_SUCCEEDED(RegKey::SetValueMultiSZ(
+      kFullRkey1, kValNameMultiStr,
+      reinterpret_cast<const uint8_t*>(kEmptyMultiSZ), sizeof(kEmptyMultiSZ)));
   EXPECT_SUCCEEDED(RegKey::GetValue(kFullRkey1, kValNameMultiStr, &result));
   EXPECT_EQ(result.size(), 0);
   // writing REG_MULTI_SZ value will automatically add ending null characters
-  EXPECT_SUCCEEDED(RegKey::SetValueMultiSZ(kFullRkey1, kValNameMultiStr,
-      reinterpret_cast<const uint8*>(kInvalidMultiSZ), sizeof(kInvalidMultiSZ)));
+  EXPECT_SUCCEEDED(
+      RegKey::SetValueMultiSZ(kFullRkey1, kValNameMultiStr,
+                              reinterpret_cast<const uint8_t*>(kInvalidMultiSZ),
+                              sizeof(kInvalidMultiSZ)));
   EXPECT_SUCCEEDED(RegKey::GetValue(kFullRkey1, kValNameMultiStr, &result));
   EXPECT_EQ(result.size(), 1);
   EXPECT_STREQ(result[0].c_str(), L"678");
@@ -555,8 +565,8 @@ void RegKeyStaticFunctionsTest() {
 #ifdef IS_PRIVATE_BUILD
   // get a temp file name
   wchar_t temp_path[MAX_PATH] = {0};
-  EXPECT_LT(::GetTempPath(ARRAY_SIZE(temp_path), temp_path),
-            static_cast<DWORD>(ARRAY_SIZE(temp_path)));
+  EXPECT_LT(::GetTempPath(arraysize(temp_path), temp_path),
+            static_cast<DWORD>(arraysize(temp_path)));
   wchar_t temp_file[MAX_PATH] = {0};
   EXPECT_NE(::GetTempFileName(temp_path, L"rkut_",
                               ::GetTickCount(), temp_file), 0);
diff --git a/media/webrtc/trunk/webrtc/base/win32socketserver.cc b/media/webrtc/trunk/webrtc/base/win32socketserver.cc
index 2adb0d3d27..72ce4ebb7c 100644
--- a/media/webrtc/trunk/webrtc/base/win32socketserver.cc
+++ b/media/webrtc/trunk/webrtc/base/win32socketserver.cc
@@ -28,26 +28,26 @@ namespace rtc {
 // TODO: Move this to a common place where PhysicalSocketServer can
 // share it.
 // Standard MTUs
-static const uint16 PACKET_MAXIMUMS[] = {
-  65535,    // Theoretical maximum, Hyperchannel
-  32000,    // Nothing
-  17914,    // 16Mb IBM Token Ring
-  8166,     // IEEE 802.4
-  // 4464   // IEEE 802.5 (4Mb max)
-  4352,     // FDDI
-  // 2048,  // Wideband Network
-  2002,     // IEEE 802.5 (4Mb recommended)
-  // 1536,  // Expermental Ethernet Networks
-  // 1500,  // Ethernet, Point-to-Point (default)
-  1492,     // IEEE 802.3
-  1006,     // SLIP, ARPANET
-  // 576,   // X.25 Networks
-  // 544,   // DEC IP Portal
-  // 512,   // NETBIOS
-  508,      // IEEE 802/Source-Rt Bridge, ARCNET
-  296,      // Point-to-Point (low delay)
-  68,       // Official minimum
-  0,        // End of list marker
+static const uint16_t PACKET_MAXIMUMS[] = {
+    65535,  // Theoretical maximum, Hyperchannel
+    32000,  // Nothing
+    17914,  // 16Mb IBM Token Ring
+    8166,   // IEEE 802.4
+    // 4464   // IEEE 802.5 (4Mb max)
+    4352,   // FDDI
+    // 2048,  // Wideband Network
+    2002,   // IEEE 802.5 (4Mb recommended)
+    // 1536,  // Expermental Ethernet Networks
+    // 1500,  // Ethernet, Point-to-Point (default)
+    1492,   // IEEE 802.3
+    1006,   // SLIP, ARPANET
+    // 576,   // X.25 Networks
+    // 544,   // DEC IP Portal
+    // 512,   // NETBIOS
+    508,    // IEEE 802/Source-Rt Bridge, ARCNET
+    296,    // Point-to-Point (low delay)
+    68,     // Official minimum
+    0,      // End of list marker
 };
 
 static const int IP_HEADER_SIZE = 20u;
@@ -55,7 +55,7 @@ static const int ICMP_HEADER_SIZE = 8u;
 static const int ICMP_PING_TIMEOUT_MILLIS = 10000u;
 
 // TODO: Enable for production builds also? Use FormatMessage?
-#ifdef _DEBUG
+#if !defined(NDEBUG)
 LPCSTR WSAErrorToString(int error, LPCSTR *description_result) {
   LPCSTR string = "Unspecified";
   LPCSTR description = "Unspecified description";
@@ -143,7 +143,7 @@ void ReportWSAError(LPCSTR context, int error, const SocketAddress& address) {}
 
 struct Win32Socket::DnsLookup {
   HANDLE handle;
-  uint16 port;
+  uint16_t port;
   char buffer[MAXGETHOSTSTRUCT];
 };
 
@@ -512,9 +512,9 @@ int Win32Socket::Close() {
   return err;
 }
 
-int Win32Socket::EstimateMTU(uint16* mtu) {
+int Win32Socket::EstimateMTU(uint16_t* mtu) {
   SocketAddress addr = GetRemoteAddress();
-  if (addr.IsAny()) {
+  if (addr.IsAnyIP()) {
     error_ = ENOTCONN;
     return -1;
   }
@@ -526,7 +526,7 @@ int Win32Socket::EstimateMTU(uint16* mtu) {
   }
 
   for (int level = 0; PACKET_MAXIMUMS[level + 1] > 0; ++level) {
-    int32 size = PACKET_MAXIMUMS[level] - IP_HEADER_SIZE - ICMP_HEADER_SIZE;
+    int32_t size = PACKET_MAXIMUMS[level] - IP_HEADER_SIZE - ICMP_HEADER_SIZE;
     WinPing::PingResult result = ping.Ping(addr.ipaddr(), size,
                                            ICMP_PING_TIMEOUT_MILLIS, 1, false);
     if (result == WinPing::PING_FAIL) {
@@ -626,8 +626,8 @@ void Win32Socket::OnSocketNotify(SOCKET socket, int event, int error) {
     case FD_CONNECT:
       if (error != ERROR_SUCCESS) {
         ReportWSAError("WSAAsync:connect notify", error, addr_);
-#ifdef _DEBUG
-        int32 duration = TimeSince(connect_time_);
+#if !defined(NDEBUG)
+        int32_t duration = TimeSince(connect_time_);
         LOG(LS_INFO) << "WSAAsync:connect error (" << duration
                      << " ms), faking close";
 #endif
@@ -639,8 +639,8 @@ void Win32Socket::OnSocketNotify(SOCKET socket, int event, int error) {
         // though the connect event never did occur.
         SignalCloseEvent(this, error);
       } else {
-#ifdef _DEBUG
-        int32 duration = TimeSince(connect_time_);
+#if !defined(NDEBUG)
+        int32_t duration = TimeSince(connect_time_);
         LOG(LS_INFO) << "WSAAsync:connect (" << duration << " ms)";
 #endif
         state_ = CS_CONNECTED;
@@ -679,10 +679,10 @@ void Win32Socket::OnDnsNotify(HANDLE task, int error) {
   if (!dns_ || dns_->handle != task)
     return;
 
-  uint32 ip = 0;
+  uint32_t ip = 0;
   if (error == 0) {
     hostent* pHost = reinterpret_cast<hostent*>(dns_->buffer);
-    uint32 net_ip = *reinterpret_cast<uint32*>(pHost->h_addr_list[0]);
+    uint32_t net_ip = *reinterpret_cast<uint32_t*>(pHost->h_addr_list[0]);
     ip = NetworkToHost32(net_ip);
   }
 
@@ -762,7 +762,7 @@ bool Win32SocketServer::Wait(int cms, bool process_io) {
   if (process_io) {
     // Spin the Win32 message pump at least once, and as long as requested.
     // This is the Thread::ProcessMessages case.
-    uint32 start = Time();
+    uint32_t start = Time();
     do {
       MSG msg;
       SetTimer(wnd_.handle(), 0, cms, NULL);
diff --git a/media/webrtc/trunk/webrtc/base/win32socketserver.h b/media/webrtc/trunk/webrtc/base/win32socketserver.h
index a03f6c028c..b468cfd9e3 100644
--- a/media/webrtc/trunk/webrtc/base/win32socketserver.h
+++ b/media/webrtc/trunk/webrtc/base/win32socketserver.h
@@ -52,7 +52,7 @@ class Win32Socket : public AsyncSocket {
   virtual int GetError() const;
   virtual void SetError(int error);
   virtual ConnState GetState() const;
-  virtual int EstimateMTU(uint16* mtu);
+  virtual int EstimateMTU(uint16_t* mtu);
   virtual int GetOption(Option opt, int* value);
   virtual int SetOption(Option opt, int value);
 
@@ -72,7 +72,7 @@ class Win32Socket : public AsyncSocket {
   int error_;
   ConnState state_;
   SocketAddress addr_;         // address that we connected to (see DoConnect)
-  uint32 connect_time_;
+  uint32_t connect_time_;
   bool closing_;
   int close_error_;
 
diff --git a/media/webrtc/trunk/webrtc/base/win32socketserver_unittest.cc b/media/webrtc/trunk/webrtc/base/win32socketserver_unittest.cc
index 1d3ef2ea37..daf9e70d1f 100644
--- a/media/webrtc/trunk/webrtc/base/win32socketserver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/win32socketserver_unittest.cc
@@ -17,7 +17,7 @@ namespace rtc {
 // Test that Win32SocketServer::Wait works as expected.
 TEST(Win32SocketServerTest, TestWait) {
   Win32SocketServer server(NULL);
-  uint32 start = Time();
+  uint32_t start = Time();
   server.Wait(1000, true);
   EXPECT_GE(TimeSince(start), 1000);
 }
diff --git a/media/webrtc/trunk/webrtc/base/win32toolhelp.h b/media/webrtc/trunk/webrtc/base/win32toolhelp.h
deleted file mode 100644
index dfafdb317f..0000000000
--- a/media/webrtc/trunk/webrtc/base/win32toolhelp.h
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- *  Copyright 2010 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_BASE_WIN32TOOLHELP_H_
-#define WEBRTC_BASE_WIN32TOOLHELP_H_
-
-#if !defined(WEBRTC_WIN)
-#error WEBRTC_WIN Only
-#endif
-
-#include "webrtc/base/win32.h"
-
-// Should be included first, but that causes redefinitions.
-#include <tlhelp32.h>
-
-#include "webrtc/base/constructormagic.h"
-
-namespace rtc {
-
-// The toolhelp api used to enumerate processes and their modules
-// on Windows is very repetetive and clunky to use. This little
-// template wraps it to make it a little more programmer friendly.
-//
-// Traits: Traits type that adapts the enumerator to the corresponding
-//         win32 toolhelp api. Each traits class need to:
-//         - define the type of the enumerated data as a public symbol Type
-//
-//         - implement bool First(HANDLE, T*) normally calls a
-//           Xxxx32First method in the toolhelp API. Ex Process32First(...)
-//
-//         - implement bool Next(HANDLE, T*) normally calls a
-//           Xxxx32Next method in the toolhelp API. Ex Process32Next(...)
-//
-//         - implement bool CloseHandle(HANDLE)
-//
-template<typename Traits>
-class ToolhelpEnumeratorBase {
- public:
-  ToolhelpEnumeratorBase(HANDLE snapshot)
-      : snapshot_(snapshot), broken_(false), first_(true) {
-
-    // Clear out the Traits::Type structure instance.
-    Zero(&current_);
-  }
-
-  virtual ~ToolhelpEnumeratorBase() {
-    Close();
-  }
-
-  // Moves forward to the next object using the First and Next
-  // pointers. If either First or Next ever indicates an failure
-  // all subsequent calls to this method will fail; the enumerator
-  // object is considered broken.
-  bool Next() {
-    if (!Valid()) {
-      return false;
-    }
-
-    // Move the iteration forward.
-    current_.dwSize = sizeof(typename Traits::Type);
-    bool incr_ok = false;
-    if (first_) {
-      incr_ok = Traits::First(snapshot_, &current_);
-      first_ = false;
-    } else {
-      incr_ok = Traits::Next(snapshot_, &current_);
-    }
-
-    if (!incr_ok) {
-      Zero(&current_);
-      broken_ = true;
-    }
-
-    return incr_ok;
-  }
-
-  const typename Traits::Type& current() const {
-    return current_;
-  }
-
-  void Close() {
-    if (snapshot_ != INVALID_HANDLE_VALUE) {
-      Traits::CloseHandle(snapshot_);
-      snapshot_ = INVALID_HANDLE_VALUE;
-    }
-  }
-
- private:
-  // Checks the state of the snapshot handle.
-  bool Valid() {
-    return snapshot_ != INVALID_HANDLE_VALUE && !broken_;
-  }
-
-  static void Zero(typename Traits::Type* buff) {
-    ZeroMemory(buff, sizeof(typename Traits::Type));
-  }
-
-  HANDLE snapshot_;
-  typename Traits::Type current_;
-  bool broken_;
-  bool first_;
-};
-
-class ToolhelpTraits {
- public:
-  static HANDLE CreateSnapshot(uint32 flags, uint32 process_id) {
-    return CreateToolhelp32Snapshot(flags, process_id);
-  }
-
-  static bool CloseHandle(HANDLE handle) {
-    return ::CloseHandle(handle) == TRUE;
-  }
-};
-
-class ToolhelpProcessTraits : public ToolhelpTraits {
- public:
-  typedef PROCESSENTRY32 Type;
-
-  static bool First(HANDLE handle, Type* t) {
-    return ::Process32First(handle, t) == TRUE;
-  }
-
-  static bool Next(HANDLE handle, Type* t) {
-    return ::Process32Next(handle, t) == TRUE;
-  }
-};
-
-class ProcessEnumerator : public ToolhelpEnumeratorBase<ToolhelpProcessTraits> {
- public:
-  ProcessEnumerator()
-      : ToolhelpEnumeratorBase(
-           ToolhelpProcessTraits::CreateSnapshot(TH32CS_SNAPPROCESS, 0)) {
-  }
-
- private:
-  DISALLOW_EVIL_CONSTRUCTORS(ProcessEnumerator);
-};
-
-class ToolhelpModuleTraits : public ToolhelpTraits {
- public:
-  typedef MODULEENTRY32 Type;
-
-  static bool First(HANDLE handle, Type* t) {
-    return ::Module32First(handle, t) == TRUE;
-  }
-
-  static bool Next(HANDLE handle, Type* t) {
-    return ::Module32Next(handle, t) == TRUE;
-  }
-};
-
-class ModuleEnumerator : public ToolhelpEnumeratorBase<ToolhelpModuleTraits> {
- public:
-  explicit ModuleEnumerator(uint32 process_id)
-      : ToolhelpEnumeratorBase(
-            ToolhelpModuleTraits::CreateSnapshot(TH32CS_SNAPMODULE,
-                                                 process_id)) {
-  }
-
- private:
-  DISALLOW_EVIL_CONSTRUCTORS(ModuleEnumerator);
-};
-
-}  // namespace rtc
-
-#endif  // WEBRTC_BASE_WIN32TOOLHELP_H_
diff --git a/media/webrtc/trunk/webrtc/base/win32toolhelp_unittest.cc b/media/webrtc/trunk/webrtc/base/win32toolhelp_unittest.cc
deleted file mode 100644
index 280f2ec98d..0000000000
--- a/media/webrtc/trunk/webrtc/base/win32toolhelp_unittest.cc
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- *  Copyright 2010 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/win32toolhelp.h"
-
-namespace rtc {
-
-typedef struct {
-  // Required to match the toolhelp api struct 'design'.
-  DWORD dwSize;
-  int a;
-  uint32 b;
-} TestData;
-
-class Win32ToolhelpTest : public testing::Test {
- public:
-  Win32ToolhelpTest() {
-  }
-
-  HANDLE AsHandle() {
-    return reinterpret_cast<HANDLE>(this);
-  }
-
-  static Win32ToolhelpTest* AsFixture(HANDLE handle) {
-    return reinterpret_cast<Win32ToolhelpTest*>(handle);
-  }
-
-  static bool First(HANDLE handle, TestData* d) {
-    Win32ToolhelpTest* tst = Win32ToolhelpTest::AsFixture(handle);
-    // This method should be called only once for every test.
-    // If it is called more than once it return false which
-    // should break the test.
-    EXPECT_EQ(0, tst->first_called_); // Just to be safe.
-    if (tst->first_called_ > 0) {
-      return false;
-    }
-
-    *d = kTestData[0];
-    tst->index_ = 1;
-    ++(tst->first_called_);
-    return true;
-  }
-
-  static bool Next(HANDLE handle, TestData* d) {
-    Win32ToolhelpTest* tst = Win32ToolhelpTest::AsFixture(handle);
-    ++(tst->next_called_);
-
-    if (tst->index_ >= kTestDataSize) {
-      return FALSE;
-    }
-
-    *d = kTestData[tst->index_];
-    ++(tst->index_);
-    return true;
-  }
-
-  static bool Fail(HANDLE handle, TestData* d) {
-    Win32ToolhelpTest* tst = Win32ToolhelpTest::AsFixture(handle);
-    ++(tst->fail_called_);
-    return false;
-  }
-
-  static bool CloseHandle(HANDLE handle) {
-    Win32ToolhelpTest* tst = Win32ToolhelpTest::AsFixture(handle);
-    ++(tst->close_handle_called_);
-    return true;
-  }
-
- protected:
-  virtual void SetUp() {
-    fail_called_ = 0;
-    first_called_ = 0;
-    next_called_ = 0;
-    close_handle_called_ = 0;
-    index_ = 0;
-  }
-
-  static bool AllZero(const TestData& data) {
-    return data.dwSize == 0 && data.a == 0 && data.b == 0;
-  }
-
-  static bool Equals(const TestData& expected, const TestData& actual) {
-    return expected.dwSize == actual.dwSize
-        && expected.a == actual.a
-        && expected.b == actual.b;
-  }
-
-  bool CheckCallCounters(int first, int next, int fail, int close) {
-    bool match = first_called_ == first && next_called_ == next
-      && fail_called_ == fail && close_handle_called_ == close;
-
-    if (!match) {
-      LOG(LS_ERROR) << "Expected: ("
-                    << first << ", "
-                    << next << ", "
-                    << fail << ", "
-                    << close << ")";
-
-      LOG(LS_ERROR) << "Actual: ("
-                    << first_called_ << ", "
-                    << next_called_ << ", "
-                    << fail_called_ << ", "
-                    << close_handle_called_ << ")";
-    }
-    return match;
-  }
-
-  static const int kTestDataSize = 3;
-  static const TestData kTestData[];
-  int index_;
-  int first_called_;
-  int fail_called_;
-  int next_called_;
-  int close_handle_called_;
-};
-
-const TestData Win32ToolhelpTest::kTestData[] = {
-  {1, 1, 1}, {2, 2, 2}, {3, 3, 3}
-};
-
-
-class TestTraits {
- public:
-  typedef TestData Type;
-
-  static bool First(HANDLE handle, Type* t) {
-    return Win32ToolhelpTest::First(handle, t);
-  }
-
-  static bool Next(HANDLE handle, Type* t) {
-    return Win32ToolhelpTest::Next(handle, t);
-  }
-
-  static bool CloseHandle(HANDLE handle) {
-    return Win32ToolhelpTest::CloseHandle(handle);
-  }
-};
-
-class BadFirstTraits {
- public:
-  typedef TestData Type;
-
-  static bool First(HANDLE handle, Type* t) {
-    return Win32ToolhelpTest::Fail(handle, t);
-  }
-
-  static bool Next(HANDLE handle, Type* t) {
-    // This should never be called.
-    ADD_FAILURE();
-    return false;
-  }
-
-  static bool CloseHandle(HANDLE handle) {
-    return Win32ToolhelpTest::CloseHandle(handle);
-  }
-};
-
-class BadNextTraits {
- public:
-  typedef TestData Type;
-
-  static bool First(HANDLE handle, Type* t) {
-    return Win32ToolhelpTest::First(handle, t);
-  }
-
-  static bool Next(HANDLE handle, Type* t) {
-    return Win32ToolhelpTest::Fail(handle, t);
-  }
-
-  static bool CloseHandle(HANDLE handle) {
-    return Win32ToolhelpTest::CloseHandle(handle);
-  }
-};
-
-// The toolhelp in normally inherited but most of
-// these tests only excercise the methods from the
-// traits therefore I use a typedef to make the
-// test code easier to read.
-typedef rtc::ToolhelpEnumeratorBase<TestTraits> EnumeratorForTest;
-
-TEST_F(Win32ToolhelpTest, TestNextWithInvalidCtorHandle) {
-  EnumeratorForTest t(INVALID_HANDLE_VALUE);
-
-  EXPECT_FALSE(t.Next());
-  EXPECT_TRUE(CheckCallCounters(0, 0, 0, 0));
-}
-
-// Tests that Next() returns false if the first-pointer
-// function fails.
-TEST_F(Win32ToolhelpTest, TestNextFirstFails) {
-  typedef rtc::ToolhelpEnumeratorBase<BadFirstTraits> BadEnumerator;
-  rtc::scoped_ptr<BadEnumerator> t(new BadEnumerator(AsHandle()));
-
-  // If next ever fails it shall always fail.
-  EXPECT_FALSE(t->Next());
-  EXPECT_FALSE(t->Next());
-  EXPECT_FALSE(t->Next());
-  t.reset();
-  EXPECT_TRUE(CheckCallCounters(0, 0, 1, 1));
-}
-
-// Tests that Next() returns false if the next-pointer
-// function fails.
-TEST_F(Win32ToolhelpTest, TestNextNextFails) {
-  typedef rtc::ToolhelpEnumeratorBase<BadNextTraits> BadEnumerator;
-  rtc::scoped_ptr<BadEnumerator> t(new BadEnumerator(AsHandle()));
-
-  // If next ever fails it shall always fail. No more calls
-  // shall be dispatched to Next(...).
-  EXPECT_TRUE(t->Next());
-  EXPECT_FALSE(t->Next());
-  EXPECT_FALSE(t->Next());
-  t.reset();
-  EXPECT_TRUE(CheckCallCounters(1, 0, 1, 1));
-}
-
-
-// Tests that current returns an object is all zero's
-// if Next() hasn't been called.
-TEST_F(Win32ToolhelpTest, TestCurrentNextNotCalled) {
-  rtc::scoped_ptr<EnumeratorForTest> t(new EnumeratorForTest(AsHandle()));
-  EXPECT_TRUE(AllZero(t->current()));
-  t.reset();
-  EXPECT_TRUE(CheckCallCounters(0, 0, 0, 1));
-}
-
-// Tests the simple everything works path through the code.
-TEST_F(Win32ToolhelpTest, TestCurrentNextCalled) {
-  rtc::scoped_ptr<EnumeratorForTest> t(new EnumeratorForTest(AsHandle()));
-
-  EXPECT_TRUE(t->Next());
-  EXPECT_TRUE(Equals(t->current(), kTestData[0]));
-  EXPECT_TRUE(t->Next());
-  EXPECT_TRUE(Equals(t->current(), kTestData[1]));
-  EXPECT_TRUE(t->Next());
-  EXPECT_TRUE(Equals(t->current(), kTestData[2]));
-  EXPECT_FALSE(t->Next());
-  t.reset();
-  EXPECT_TRUE(CheckCallCounters(1, 3, 0, 1));
-}
-
-TEST_F(Win32ToolhelpTest, TestCurrentProcess) {
-  WCHAR buf[MAX_PATH];
-  GetModuleFileName(NULL, buf, ARRAY_SIZE(buf));
-  std::wstring name = ToUtf16(Pathname(ToUtf8(buf)).filename());
-
-  rtc::ProcessEnumerator processes;
-  bool found = false;
-  while (processes.Next()) {
-    if (!name.compare(processes.current().szExeFile)) {
-      found = true;
-      break;
-    }
-  }
-  EXPECT_TRUE(found);
-
-  rtc::ModuleEnumerator modules(processes.current().th32ProcessID);
-  found = false;
-  while (modules.Next()) {
-    if (!name.compare(modules.current().szModule)) {
-      found = true;
-      break;
-    }
-  }
-  EXPECT_TRUE(found);
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/win32windowpicker.cc b/media/webrtc/trunk/webrtc/base/win32windowpicker.cc
index b4550ae4a4..da05a5c65c 100644
--- a/media/webrtc/trunk/webrtc/base/win32windowpicker.cc
+++ b/media/webrtc/trunk/webrtc/base/win32windowpicker.cc
@@ -12,6 +12,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 
@@ -58,7 +59,7 @@ BOOL CALLBACK Win32WindowPicker::EnumProc(HWND hwnd, LPARAM l_param) {
   }
 
   TCHAR window_title[500];
-  GetWindowText(hwnd, window_title, ARRAY_SIZE(window_title));
+  GetWindowText(hwnd, window_title, arraysize(window_title));
   std::string title = ToUtf8(window_title);
 
   WindowId id(hwnd);
diff --git a/media/webrtc/trunk/webrtc/base/win32windowpicker_unittest.cc b/media/webrtc/trunk/webrtc/base/win32windowpicker_unittest.cc
index 71e8af6bf2..701bb27d42 100644
--- a/media/webrtc/trunk/webrtc/base/win32windowpicker_unittest.cc
+++ b/media/webrtc/trunk/webrtc/base/win32windowpicker_unittest.cc
@@ -7,6 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
@@ -71,7 +72,7 @@ TEST(Win32WindowPickerTest, TestGetWindowList) {
   EXPECT_EQ(window_picker.visible_window()->handle(), desc.id().id());
   TCHAR window_title[500];
   GetWindowText(window_picker.visible_window()->handle(), window_title,
-                ARRAY_SIZE(window_title));
+                arraysize(window_title));
   EXPECT_EQ(0, wcscmp(window_title, kVisibleWindowTitle));
 }
 
diff --git a/media/webrtc/trunk/webrtc/base/window.h b/media/webrtc/trunk/webrtc/base/window.h
index 9f4381a610..b1f1724e63 100644
--- a/media/webrtc/trunk/webrtc/base/window.h
+++ b/media/webrtc/trunk/webrtc/base/window.h
@@ -40,7 +40,7 @@ class WindowId {
   typedef unsigned int WindowT;
 #endif
 
-  static WindowId Cast(uint64 id) {
+  static WindowId Cast(uint64_t id) {
 #if defined(WEBRTC_WIN)
     return WindowId(reinterpret_cast<WindowId::WindowT>(id));
 #else
@@ -48,11 +48,11 @@ class WindowId {
 #endif
   }
 
-  static uint64 Format(const WindowT& id) {
+  static uint64_t Format(const WindowT& id) {
 #if defined(WEBRTC_WIN)
-    return static_cast<uint64>(reinterpret_cast<uintptr_t>(id));
+    return static_cast<uint64_t>(reinterpret_cast<uintptr_t>(id));
 #else
-    return static_cast<uint64>(id);
+    return static_cast<uint64_t>(id);
 #endif
   }
 
diff --git a/media/webrtc/trunk/webrtc/base/winping.cc b/media/webrtc/trunk/webrtc/base/winping.cc
index 7d6ee22a95..be436c3cb0 100644
--- a/media/webrtc/trunk/webrtc/base/winping.cc
+++ b/media/webrtc/trunk/webrtc/base/winping.cc
@@ -124,14 +124,13 @@ const char * const ICMP_CREATE_FUNC = "IcmpCreateFile";
 const char * const ICMP_CLOSE_FUNC = "IcmpCloseHandle";
 const char * const ICMP_SEND_FUNC = "IcmpSendEcho";
 const char * const ICMP6_CREATE_FUNC = "Icmp6CreateFile";
-const char * const ICMP6_CLOSE_FUNC = "Icmp6CloseHandle";
 const char * const ICMP6_SEND_FUNC = "Icmp6SendEcho2";
 
-inline uint32 ReplySize(uint32 data_size, int family) {
+inline uint32_t ReplySize(uint32_t data_size, int family) {
   if (family == AF_INET) {
     // A ping error message is 8 bytes long, so make sure we allow for at least
     // 8 bytes of reply data.
-    return sizeof(ICMP_ECHO_REPLY) + std::max<uint32>(8, data_size);
+    return sizeof(ICMP_ECHO_REPLY) + std::max<uint32_t>(8, data_size);
   } else if (family == AF_INET6) {
     // Per MSDN, Send6IcmpEcho2 needs at least one ICMPV6_ECHO_REPLY,
     // 8 bytes for ICMP header, _and_ an IO_BLOCK_STATUS (2 pointers),
@@ -209,10 +208,11 @@ WinPing::~WinPing() {
   delete[] reply_;
 }
 
-WinPing::PingResult WinPing::Ping(
-    IPAddress ip, uint32 data_size, uint32 timeout, uint8 ttl,
-    bool allow_fragments) {
-
+WinPing::PingResult WinPing::Ping(IPAddress ip,
+                                  uint32_t data_size,
+                                  uint32_t timeout,
+                                  uint8_t ttl,
+                                  bool allow_fragments) {
   if (data_size == 0 || timeout == 0 || ttl == 0) {
     LOG(LERROR) << "IcmpSendEcho: data_size/timeout/ttl is 0.";
     return PING_INVALID_PARAMS;
@@ -226,7 +226,7 @@ WinPing::PingResult WinPing::Ping(
     ipopt.Flags |= IP_FLAG_DF;
   ipopt.Ttl = ttl;
 
-  uint32 reply_size = ReplySize(data_size, ip.family());
+  uint32_t reply_size = ReplySize(data_size, ip.family());
 
   if (data_size > dlen_) {
     delete [] data_;
@@ -242,19 +242,16 @@ WinPing::PingResult WinPing::Ping(
   }
   DWORD result = 0;
   if (ip.family() == AF_INET) {
-    result = send_(hping_, ip.ipv4_address().S_un.S_addr,
-                   data_, uint16(data_size), &ipopt,
-                   reply_, reply_size, timeout);
+    result = send_(hping_, ip.ipv4_address().S_un.S_addr, data_,
+                   uint16_t(data_size), &ipopt, reply_, reply_size, timeout);
   } else if (ip.family() == AF_INET6) {
     sockaddr_in6 src = {0};
     sockaddr_in6 dst = {0};
     src.sin6_family = AF_INET6;
     dst.sin6_family = AF_INET6;
     dst.sin6_addr = ip.ipv6_address();
-    result = send6_(hping6_, NULL, NULL, NULL,
-                    &src, &dst,
-                    data_, int16(data_size), &ipopt,
-                    reply_, reply_size, timeout);
+    result = send6_(hping6_, NULL, NULL, NULL, &src, &dst, data_,
+                    int16_t(data_size), &ipopt, reply_, reply_size, timeout);
   }
   if (result == 0) {
     DWORD error = GetLastError();
diff --git a/media/webrtc/trunk/webrtc/base/winping.h b/media/webrtc/trunk/webrtc/base/winping.h
index 75f82b7b4a..ddaefc5253 100644
--- a/media/webrtc/trunk/webrtc/base/winping.h
+++ b/media/webrtc/trunk/webrtc/base/winping.h
@@ -76,9 +76,11 @@ public:
     // Attempts to send a ping with the given parameters.
     enum PingResult { PING_FAIL, PING_INVALID_PARAMS,
                       PING_TOO_LARGE, PING_TIMEOUT, PING_SUCCESS };
-    PingResult Ping(
-        IPAddress ip, uint32 data_size, uint32 timeout_millis, uint8 ttl,
-        bool allow_fragments);
+    PingResult Ping(IPAddress ip,
+                    uint32_t data_size,
+                    uint32_t timeout_millis,
+                    uint8_t ttl,
+                    bool allow_fragments);
 
 private:
     HMODULE dll_;
@@ -90,9 +92,9 @@ private:
     PIcmp6CreateFile create6_;
     PIcmp6SendEcho2 send6_;
     char* data_;
-    uint32 dlen_;
+    uint32_t dlen_;
     char* reply_;
-    uint32 rlen_;
+    uint32_t rlen_;
     bool valid_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/base/worker.h b/media/webrtc/trunk/webrtc/base/worker.h
index d5594e3b87..17ae8cf67b 100644
--- a/media/webrtc/trunk/webrtc/base/worker.h
+++ b/media/webrtc/trunk/webrtc/base/worker.h
@@ -64,7 +64,7 @@ class Worker : private MessageHandler {
   // The thread that is currently doing the work.
   Thread *worker_thread_;
 
-  DISALLOW_COPY_AND_ASSIGN(Worker);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Worker);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/base/x11windowpicker.cc b/media/webrtc/trunk/webrtc/base/x11windowpicker.cc
index 918913a92b..21f71c61e3 100644
--- a/media/webrtc/trunk/webrtc/base/x11windowpicker.cc
+++ b/media/webrtc/trunk/webrtc/base/x11windowpicker.cc
@@ -73,7 +73,7 @@ class XWindowProperty {
   unsigned long size_;  // NOLINT: type required by XGetWindowProperty
   unsigned char* data_;
 
-  DISALLOW_COPY_AND_ASSIGN(XWindowProperty);
+  RTC_DISALLOW_COPY_AND_ASSIGN(XWindowProperty);
 };
 
 // Stupid X11.  It seems none of the synchronous returns codes from X11 calls
@@ -118,7 +118,7 @@ class XErrorSuppressor {
   Display* display_;
   XErrorHandler original_error_handler_;
 
-  DISALLOW_COPY_AND_ASSIGN(XErrorSuppressor);
+  RTC_DISALLOW_COPY_AND_ASSIGN(XErrorSuppressor);
 };
 
 // Hiding all X11 specifics inside its own class. This to avoid
@@ -277,7 +277,7 @@ class XWindowEnumerator {
     return true;
   }
 
-  uint8* GetWindowIcon(const WindowId& id, int* width, int* height) {
+  uint8_t* GetWindowIcon(const WindowId& id, int* width, int* height) {
     if (!Init()) {
       return NULL;
     }
@@ -297,14 +297,14 @@ class XWindowEnumerator {
       LOG(LS_ERROR) << "Failed to get size of the icon.";
       return NULL;
     }
-    // Get the icon data, the format is one uint32 each for width and height,
+    // Get the icon data, the format is one uint32_t each for width and height,
     // followed by the actual pixel data.
     if (size >= 2 &&
         XGetWindowProperty(
             display_, id.id(), net_wm_icon_, 0, size, False, XA_CARDINAL,
             &ret_type, &format, &length, &bytes_after, &data) == Success &&
         data) {
-      uint32* data_ptr = reinterpret_cast<uint32*>(data);
+      uint32_t* data_ptr = reinterpret_cast<uint32_t*>(data);
       int w, h;
       w = data_ptr[0];
       h = data_ptr[1];
@@ -313,8 +313,7 @@ class XWindowEnumerator {
         LOG(LS_ERROR) << "Not a vaild icon.";
         return NULL;
       }
-      uint8* rgba =
-          ArgbToRgba(&data_ptr[2], 0, 0, w, h, w, h, true);
+      uint8_t* rgba = ArgbToRgba(&data_ptr[2], 0, 0, w, h, w, h, true);
       XFree(data);
       *width = w;
       *height = h;
@@ -325,7 +324,7 @@ class XWindowEnumerator {
     }
   }
 
-  uint8* GetWindowThumbnail(const WindowId& id, int width, int height) {
+  uint8_t* GetWindowThumbnail(const WindowId& id, int width, int height) {
     if (!Init()) {
       return NULL;
     }
@@ -390,12 +389,8 @@ class XWindowEnumerator {
       return NULL;
     }
 
-    uint8* data = GetDrawableThumbnail(src_pixmap,
-                                       attr.visual,
-                                       src_width,
-                                       src_height,
-                                       width,
-                                       height);
+    uint8_t* data = GetDrawableThumbnail(src_pixmap, attr.visual, src_width,
+                                         src_height, width, height);
     XFreePixmap(display_, src_pixmap);
     return data;
   }
@@ -408,7 +403,7 @@ class XWindowEnumerator {
     return XScreenCount(display_);
   }
 
-  uint8* GetDesktopThumbnail(const DesktopId& id, int width, int height) {
+  uint8_t* GetDesktopThumbnail(const DesktopId& id, int width, int height) {
     if (!Init()) {
       return NULL;
     }
@@ -445,12 +440,12 @@ class XWindowEnumerator {
   }
 
  private:
-  uint8* GetDrawableThumbnail(Drawable src_drawable,
-                              Visual* visual,
-                              int src_width,
-                              int src_height,
-                              int dst_width,
-                              int dst_height) {
+  uint8_t* GetDrawableThumbnail(Drawable src_drawable,
+                                Visual* visual,
+                                int src_width,
+                                int src_height,
+                                int dst_width,
+                                int dst_height) {
     if (!has_render_extension_) {
       // Without the Xrender extension we would have to read the full window and
       // scale it down in our process. Xrender is over a decade old so we aren't
@@ -561,14 +556,9 @@ class XWindowEnumerator {
                               dst_width,
                               dst_height,
                               AllPlanes, ZPixmap);
-    uint8* data = ArgbToRgba(reinterpret_cast<uint32*>(image->data),
-                             centered_x,
-                             centered_y,
-                             scaled_width,
-                             scaled_height,
-                             dst_width,
-                             dst_height,
-                             false);
+    uint8_t* data = ArgbToRgba(reinterpret_cast<uint32_t*>(image->data),
+                               centered_x, centered_y, scaled_width,
+                               scaled_height, dst_width, dst_height, false);
     XDestroyImage(image);
     XRenderFreePicture(display_, dst);
     XFreePixmap(display_, dst_pixmap);
@@ -576,17 +566,23 @@ class XWindowEnumerator {
     return data;
   }
 
-  uint8* ArgbToRgba(uint32* argb_data, int x, int y, int w, int h,
-                    int stride_x, int stride_y, bool has_alpha) {
-    uint8* p;
+  uint8_t* ArgbToRgba(uint32_t* argb_data,
+                      int x,
+                      int y,
+                      int w,
+                      int h,
+                      int stride_x,
+                      int stride_y,
+                      bool has_alpha) {
+    uint8_t* p;
     int len = stride_x * stride_y * 4;
-    uint8* data = new uint8[len];
+    uint8_t* data = new uint8_t[len];
     memset(data, 0, len);
     p = data + 4 * (y * stride_x + x);
     for (int i = 0; i < h; ++i) {
       for (int j = 0; j < w; ++j) {
-        uint32 argb;
-        uint32 rgba;
+        uint32_t argb;
+        uint32_t rgba;
         argb = argb_data[stride_x * (y + i) + x + j];
         rgba = (argb << 8) | (argb >> 24);
         *p = rgba >> 24;
@@ -691,7 +687,7 @@ class XWindowEnumerator {
       return 0;
     }
     if (type != None) {
-      int64 state = static_cast<int64>(*data);
+      int64_t state = static_cast<int64_t>(*data);
       XFree(data);
       return state == NormalState ? window : 0;
     }
@@ -789,13 +785,14 @@ bool X11WindowPicker::MoveToFront(const WindowId& id) {
   return enumerator_->MoveToFront(id);
 }
 
-
-uint8* X11WindowPicker::GetWindowIcon(const WindowId& id, int* width,
+uint8_t* X11WindowPicker::GetWindowIcon(const WindowId& id,
+                                        int* width,
                                         int* height) {
   return enumerator_->GetWindowIcon(id, width, height);
 }
 
-uint8* X11WindowPicker::GetWindowThumbnail(const WindowId& id, int width,
+uint8_t* X11WindowPicker::GetWindowThumbnail(const WindowId& id,
+                                             int width,
                                              int height) {
   return enumerator_->GetWindowThumbnail(id, width, height);
 }
@@ -804,7 +801,7 @@ int X11WindowPicker::GetNumDesktops() {
   return enumerator_->GetNumDesktops();
 }
 
-uint8* X11WindowPicker::GetDesktopThumbnail(const DesktopId& id,
+uint8_t* X11WindowPicker::GetDesktopThumbnail(const DesktopId& id,
                                               int width,
                                               int height) {
   return enumerator_->GetDesktopThumbnail(id, width, height);
diff --git a/media/webrtc/trunk/webrtc/base/x11windowpicker.h b/media/webrtc/trunk/webrtc/base/x11windowpicker.h
index b340b88429..501adf5820 100644
--- a/media/webrtc/trunk/webrtc/base/x11windowpicker.h
+++ b/media/webrtc/trunk/webrtc/base/x11windowpicker.h
@@ -38,10 +38,10 @@ class X11WindowPicker : public WindowPicker {
   bool GetDesktopDimensions(const DesktopId& id,
                             int* width,
                             int* height) override;
-  uint8* GetWindowIcon(const WindowId& id, int* width, int* height);
-  uint8* GetWindowThumbnail(const WindowId& id, int width, int height);
+  uint8_t* GetWindowIcon(const WindowId& id, int* width, int* height);
+  uint8_t* GetWindowThumbnail(const WindowId& id, int width, int height);
   int GetNumDesktops();
-  uint8* GetDesktopThumbnail(const DesktopId& id, int width, int height);
+  uint8_t* GetDesktopThumbnail(const DesktopId& id, int width, int height);
 
  private:
   scoped_ptr<XWindowEnumerator> enumerator_;
diff --git a/media/webrtc/trunk/webrtc/build/android/AndroidManifest.xml b/media/webrtc/trunk/webrtc/build/android/AndroidManifest.xml
new file mode 100644
index 0000000000..0dcf2faffb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/build/android/AndroidManifest.xml
@@ -0,0 +1,14 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+  This is a dummy manifest which is required by:
+  1. aapt when generating R.java in java.gypi:
+     Nothing in the manifest is used, but it is still required by aapt.
+  2. lint: [min|target]SdkVersion are required by lint and should
+     be kept up-to-date.
+-->
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+    package="dummy.package">
+
+      <uses-sdk android:minSdkVersion="15" android:targetSdkVersion="23" />
+
+</manifest>
diff --git a/media/webrtc/trunk/webrtc/build/android/suppressions.xml b/media/webrtc/trunk/webrtc/build/android/suppressions.xml
new file mode 100644
index 0000000000..0fc22e0813
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/build/android/suppressions.xml
@@ -0,0 +1,23 @@
+<?xml version="1.0" encoding="utf-8"?>
+<lint>
+  <!-- These lint settings is for the Android linter that gets run by
+       lint_action.gypi on compile of WebRTC java code. All WebRTC java code
+       should lint cleanly for the issues below. -->
+  <!-- TODO(phoglund): make work with suppress.py or remove printout referring
+       to suppress.py. -->
+  <issue id="NewApi"></issue>
+
+  <issue id="Locale" severity="ignore"/>
+  <issue id="SdCardPath" severity="ignore"/>
+  <issue id="UseValueOf" severity="ignore"/>
+  <issue id="InlinedApi" severity="ignore"/>
+  <issue id="DefaultLocale" severity="ignore"/>
+  <issue id="Assert" severity="ignore"/>
+  <issue id="UseSparseArrays" severity="ignore"/>
+
+  <!-- These are just from the dummy AndroidManifest.xml we use for linting.
+       It's in the same directory as this file. -->
+  <issue id="MissingApplicationIcon" severity="ignore"/>
+  <issue id="AllowBackup" severity="ignore"/>
+  <issue id="MissingVersion" severity="ignore"/>
+</lint>
diff --git a/media/webrtc/trunk/webrtc/build/android/test_runner.py b/media/webrtc/trunk/webrtc/build/android/test_runner.py
index 6169bbc7e9..78a7a190b2 100644
--- a/media/webrtc/trunk/webrtc/build/android/test_runner.py
+++ b/media/webrtc/trunk/webrtc/build/android/test_runner.py
@@ -25,38 +25,21 @@ CHROMIUM_BUILD_ANDROID_DIR = os.path.join(SRC_DIR, 'build', 'android')
 sys.path.insert(0, CHROMIUM_BUILD_ANDROID_DIR)
 
 
-import test_runner
-from pylib.gtest import gtest_config
-from pylib.gtest import setup
+import test_runner  # pylint: disable=W0406
+from pylib.gtest import gtest_test_instance
 
 def main():
-  # Override the stable test suites with the WebRTC tests.
-  gtest_config.STABLE_TEST_SUITES = [
-    'audio_decoder_unittests',
-    'common_audio_unittests',
-    'common_video_unittests',
-    'modules_tests',
-    'modules_unittests',
-    'rtc_unittests',
-    'system_wrappers_unittests',
-    'test_support_unittests',
-    'tools_unittests',
-    'video_capture_tests',
-    'video_engine_tests',
-    'video_engine_core_unittests',
-    'voice_engine_unittests',
-    'webrtc_perf_tests',
-  ]
-  gtest_config.EXPERIMENTAL_TEST_SUITES = []
-
   # Set our own paths to the .isolate files.
-  setup.ISOLATE_FILE_PATHS = {
+  # pylint: disable=protected-access
+  gtest_test_instance._DEFAULT_ISOLATE_FILE_PATHS.update({
     'audio_decoder_unittests':
-        'webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate',
+        'webrtc/modules/audio_decoder_unittests.isolate',
     'common_audio_unittests':
         'webrtc/common_audio/common_audio_unittests.isolate',
     'common_video_unittests':
         'webrtc/common_video/common_video_unittests.isolate',
+    'libjingle_peerconnection_unittest':
+        'talk/libjingle_peerconnection_unittest.isolate',
     'modules_tests': 'webrtc/modules/modules_tests.isolate',
     'modules_unittests': 'webrtc/modules/modules_unittests.isolate',
     'rtc_unittests': 'webrtc/rtc_unittests.isolate',
@@ -67,12 +50,11 @@ def main():
     'video_capture_tests':
         'webrtc/modules/video_capture/video_capture_tests.isolate',
     'video_engine_tests': 'webrtc/video_engine_tests.isolate',
-    'video_engine_core_unittests':
-        'webrtc/video_engine/video_engine_core_unittests.isolate',
     'voice_engine_unittests':
         'webrtc/voice_engine/voice_engine_unittests.isolate',
+    'webrtc_nonparallel_tests': 'webrtc/webrtc_nonparallel_tests.isolate',
     'webrtc_perf_tests': 'webrtc/webrtc_perf_tests.isolate',
-  }
+  })
   # Override environment variable to make it possible for the scripts to find
   # the root directory (our symlinking of the Chromium build toolchain would
   # otherwise make them fail to do so).
diff --git a/media/webrtc/trunk/webrtc/build/apk_test.gypi b/media/webrtc/trunk/webrtc/build/apk_test.gypi
new file mode 100644
index 0000000000..a41e436a48
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/build/apk_test.gypi
@@ -0,0 +1,40 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# This is almost an identical copy of src/build/apk_test.gypi with minor
+# modifications to allow test executables starting with "lib".
+# See http://crbug.com/543820 for more details.
+
+{
+  'dependencies': [
+    '<(DEPTH)/base/base.gyp:base_java',
+    '<(DEPTH)/build/android/pylib/device/commands/commands.gyp:chromium_commands',
+    '<(DEPTH)/build/android/pylib/remote/device/dummy/dummy.gyp:remote_device_dummy_apk',
+    '<(DEPTH)/testing/android/appurify_support.gyp:appurify_support_java',
+    '<(DEPTH)/testing/android/on_device_instrumentation.gyp:reporter_java',
+    '<(DEPTH)/tools/android/android_tools.gyp:android_tools',
+  ],
+  'conditions': [
+     ['OS == "android"', {
+       'variables': {
+         # These are used to configure java_apk.gypi included below.
+         'test_type': 'gtest',
+         'apk_name': '<(test_suite_name)',
+         'intermediate_dir': '<(PRODUCT_DIR)/<(test_suite_name)_apk',
+         'final_apk_path': '<(intermediate_dir)/<(test_suite_name)-debug.apk',
+         'java_in_dir': '<(DEPTH)/testing/android/native_test/java',
+         'native_lib_target': '<(test_suite_name)',
+         'gyp_managed_install': 0,
+       },
+       'includes': [
+         '../../build/java_apk.gypi',
+         '../../build/android/test_runner.gypi',
+       ],
+     }],  # 'OS == "android"
+  ],  # conditions
+}
diff --git a/media/webrtc/trunk/webrtc/build/apk_tests.gyp b/media/webrtc/trunk/webrtc/build/apk_tests.gyp
index 98e005ec97..02a13421f9 100644
--- a/media/webrtc/trunk/webrtc/build/apk_tests.gyp
+++ b/media/webrtc/trunk/webrtc/build/apk_tests.gyp
@@ -60,6 +60,23 @@
         '../../build/apk_test.gypi',
       ],
     },
+    {
+      'target_name': 'libjingle_peerconnection_unittest_apk',
+      'type': 'none',
+      'variables': {
+        'test_suite_name': 'libjingle_peerconnection_unittest',
+        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)libjingle_peerconnection_unittest<(SHARED_LIB_SUFFIX)',
+      },
+      'dependencies': [
+        '<(DEPTH)/talk/libjingle_tests.gyp:libjingle_peerconnection_unittest',
+        '<(DEPTH)/talk/libjingle.gyp:libjingle_peerconnection_java',
+      ],
+      'includes': [
+        # Use webrtc copy of apk_test.gypi to allow test executables starting
+        # with "lib". See http://crbug.com/543820 for more details.
+        '../build/apk_test.gypi',
+      ],
+    },
     {
       'target_name': 'modules_tests_apk',
       'type': 'none',
@@ -145,20 +162,6 @@
         '../../build/apk_test.gypi',
       ],
     },
-    {
-      'target_name': 'video_engine_core_unittests_apk',
-      'type': 'none',
-      'variables': {
-        'test_suite_name': 'video_engine_core_unittests',
-        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)video_engine_core_unittests<(SHARED_LIB_SUFFIX)',
-      },
-      'dependencies': [
-        '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core_unittests',
-      ],
-      'includes': [
-        '../../build/apk_test.gypi',
-      ],
-    },
     {
       'target_name': 'video_engine_tests_apk',
       'type': 'none',
@@ -201,6 +204,20 @@
         '../../build/apk_test.gypi',
       ],
     },
+    {
+      'target_name': 'webrtc_nonparallel_tests_apk',
+      'type': 'none',
+      'variables': {
+        'test_suite_name': 'webrtc_nonparallel_tests',
+        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)webrtc_nonparallel_tests<(SHARED_LIB_SUFFIX)',
+      },
+      'dependencies': [
+        '<(webrtc_root)/webrtc.gyp:webrtc_nonparallel_tests',
+      ],
+      'includes': [
+        '../../build/apk_test.gypi',
+      ],
+    },
     {
       'target_name': 'audio_codec_speed_tests_apk',
       'type': 'none',
@@ -215,40 +232,12 @@
         '../../build/apk_test.gypi',
       ],
     },
-    {
-      'target_name': 'video_capture_tests_apk',
-      'type': 'none',
-       'variables': {
-         'test_suite_name': 'video_capture_tests',
-         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)video_capture_tests<(SHARED_LIB_SUFFIX)',
-       },
-       'dependencies': [
-         '<(webrtc_root)/modules/modules.gyp:video_capture_tests',
-         'video_capture_java',
-       ],
-       'includes': [
-         '../../build/apk_test.gypi',
-       ],
-    },
-    {
-      # Used only by video_capture_tests_apk above, and impossible to use in the
-      # standalone build, which is why it's declared here instead of under
-      # modules/video_capture/ (to avoid the need for a forked _noop.gyp file
-      # like this file has; see comment at the top of this file).
-      'target_name': 'video_capture_java',
-      'type': 'none',
-      'variables': {
-        'java_in_dir': '<(webrtc_root)/modules/video_capture/android/java',
-      },
-      'includes': [
-        '../../build/java.gypi',
-      ],
-    },
     {
       'target_name': 'audio_device_java',
       'type': 'none',
       'variables': {
         'java_in_dir': '<(webrtc_root)/modules/audio_device/android/java',
+        'additional_src_dirs': [ '<(webrtc_root)/base/java/src', ],
         'never_lint': 1,
       },
       'includes': [
diff --git a/media/webrtc/trunk/webrtc/build/apk_tests_noop.gyp b/media/webrtc/trunk/webrtc/build/apk_tests_noop.gyp
index 7c1a6aa924..ed9249aec5 100644
--- a/media/webrtc/trunk/webrtc/build/apk_tests_noop.gyp
+++ b/media/webrtc/trunk/webrtc/build/apk_tests_noop.gyp
@@ -21,6 +21,10 @@
       'target_name': 'common_video_unittests_apk',
       'type': 'none',
     },
+    {
+      'target_name': 'libjingle_peerconnection_unittest_apk',
+      'type': 'none',
+    },
     {
       'target_name': 'modules_tests_apk',
       'type': 'none',
@@ -45,10 +49,6 @@
       'target_name': 'tools_unittests_apk',
       'type': 'none',
     },
-    {
-      'target_name': 'video_engine_core_unittests_apk',
-      'type': 'none',
-    },
     {
       'target_name': 'video_engine_tests_apk',
       'type': 'none',
@@ -57,6 +57,10 @@
       'target_name': 'voice_engine_unittests_apk',
       'type': 'none',
     },
+    {
+      'target_name': 'webrtc_nonparallel_tests_apk',
+      'type': 'none',
+    },
     {
       'target_name': 'webrtc_perf_tests_apk',
       'type': 'none',
@@ -65,9 +69,5 @@
       'target_name': 'audio_codec_speed_tests_apk',
       'type': 'none',
     },
-    {
-      'target_name': 'video_capture_tests_apk',
-      'type': 'none',
-    },
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/build/arm_neon.gypi b/media/webrtc/trunk/webrtc/build/arm_neon.gypi
index a72c8e76e7..41dd50f19a 100644
--- a/media/webrtc/trunk/webrtc/build/arm_neon.gypi
+++ b/media/webrtc/trunk/webrtc/build/arm_neon.gypi
@@ -33,7 +33,7 @@
     '-mfpu=vfpv3-d16',
   ],
   'conditions': [
-    # "-mfpu=neon" is not requried for arm64 in GCC.
+    # "-mfpu=neon" is not required for arm64 in GCC.
     ['target_arch!="arm64"', {
       'cflags': [
         '-mfpu=neon',
@@ -48,5 +48,13 @@
         '-mfpu=neon',
       ],
     }],
+    # Disable GCC LTO on NEON targets due to compiler bug.
+    # TODO(fdegans): Enable this. See crbug.com/408997.
+    ['clang==0 and use_lto==1', {
+      'cflags!': [
+        '-flto',
+        '-ffat-lto-objects',
+      ],
+    }],
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/build/common.gypi b/media/webrtc/trunk/webrtc/build/common.gypi
index 97bf78628e..5793a10e0d 100644
--- a/media/webrtc/trunk/webrtc/build/common.gypi
+++ b/media/webrtc/trunk/webrtc/build/common.gypi
@@ -21,12 +21,10 @@
 
         'conditions': [
           ['build_with_chromium==1', {
-            'build_with_libjingle': 1,
             'webrtc_root%': '<(DEPTH)/third_party/webrtc',
             'apk_tests_path%': '<(DEPTH)/third_party/webrtc/build/apk_tests_noop.gyp',
             'modules_java_gyp_path%': '<(DEPTH)/third_party/webrtc/modules/modules_java_chromium.gyp',
           }, {
-            'build_with_libjingle%': 0,
             'webrtc_root%': '<(DEPTH)/webrtc',
             'apk_tests_path%': '<(DEPTH)/webrtc/build/apk_tests.gyp',
             'modules_java_gyp_path%': '<(DEPTH)/webrtc/modules/modules_java.gyp',
@@ -34,7 +32,6 @@
         ],
       },
       'build_with_chromium%': '<(build_with_chromium)',
-      'build_with_libjingle%': '<(build_with_libjingle)',
       'webrtc_root%': '<(webrtc_root)',
       'apk_tests_path%': '<(apk_tests_path)',
       'modules_java_gyp_path%': '<(modules_java_gyp_path)',
@@ -48,9 +45,12 @@
       'include_isac%': 1,
       'include_pcm16b%': 1,
       'opus_dir%': '<(DEPTH)/third_party/opus',
+
+      # Enable to use the Mozilla internal settings.
+      'build_with_mozilla%': 0,
     },
     'build_with_chromium%': '<(build_with_chromium)',
-    'build_with_libjingle%': '<(build_with_libjingle)',
+    'build_with_mozilla%': '<(build_with_mozilla)',
     'webrtc_root%': '<(webrtc_root)',
     'apk_tests_path%': '<(apk_tests_path)',
     'modules_java_gyp_path%': '<(modules_java_gyp_path)',
@@ -98,23 +98,21 @@
     'enable_protobuf%': 1,
 
     # Disable these to not build components which can be externally provided.
+    'build_expat%': 1,
     'build_json%': 1,
     'build_libjpeg%': 1,
-    'build_libyuv%': 1,
     'build_libvpx%': 1,
-    'build_vp9%': 1,
-    'build_ssl%': 1,
+    'build_libyuv%': 1,
     'build_openmax_dl%': 1,
     'build_opus%': 1,
+    'build_protobuf%': 1,
+    'build_ssl%': 1,
 
     # Disable by default
     'have_dbus_glib%': 0,
 
-    # Enable to use the Mozilla internal settings.
-    'build_with_mozilla%': 0,
-
     # Make it possible to provide custom locations for some libraries.
-    'libvpx_dir%': '<(DEPTH)/third_party/libvpx',
+    'libvpx_dir%': '<(DEPTH)/third_party/libvpx_new',
     'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
     'opus_dir%': '<(opus_dir)',
 
@@ -133,6 +131,24 @@
     # enable schannel on windows.
     'use_legacy_ssl_defaults%': 0,
 
+    # Determines whether NEON code will be built.
+    'build_with_neon%': 0,
+
+    # Enable this to use HW H.264 encoder/decoder on iOS/Mac PeerConnections.
+    # Enabling this may break interop with Android clients that support H264.
+    'use_objc_h264%': 0,
+
+    # Enable this to build H.264 encoder/decoder using third party libraries.
+    # Encoding uses OpenH264 and decoding uses FFmpeg. Because of this, OpenH264
+    # and FFmpeg have to be correctly enabled separately.
+    # - use_openh264=1 is required for OpenH264 targets to be defined.
+    # - ffmpeg_branding=Chrome is one way to support H.264 decoding in FFmpeg.
+    #   FFmpeg can be built with/without H.264 support, see 'ffmpeg_branding'.
+    #   Without it, it compiles but H264DecoderImpl fails to initialize.
+    # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
+    # http://www.openh264.org, https://www.ffmpeg.org/
+    'use_third_party_h264%': 0,  # TODO(hbos): To be used in follow-up CL(s).
+
     'conditions': [
       ['build_with_chromium==1', {
         # Exclude pulse audio on Chromium since its prerequisites don't require
@@ -143,6 +159,10 @@
         'include_internal_audio_device%': 0,
 
         'include_ndk_cpu_features%': 0,
+
+        # Remove tests for Chromium to avoid slowing down GYP generation.
+        'include_tests%': 0,
+        'restrict_webrtc_logging%': 1,
       }, {  # Settings for the standalone (not-in-Chromium) build.
         # TODO(andrew): For now, disable the Chrome plugins, which causes a
         # flood of chromium-style warnings. Investigate enabling them:
@@ -152,13 +172,23 @@
         'include_pulse_audio%': 1,
         'include_internal_audio_device%': 1,
         'include_ndk_cpu_features%': 0,
-      }],
-      ['build_with_libjingle==1', {
-        'include_tests%': 0,
-        'restrict_webrtc_logging%': 1,
-      }, {
-        'include_tests%': 1,
-        'restrict_webrtc_logging%': 0,
+        'conditions': [
+          ['build_with_mozilla==1', {
+            'include_tests%': 0,
+            'conditions': [
+	      # silly gyp won't let me do 'a': !'b'
+	      # suppress TRACE logging in non-debug builds
+              ['debug==1', {
+                'restrict_webrtc_logging%': 0,
+              }, {
+                'restrict_webrtc_logging%': 1,
+              }],
+	    ],
+          }, {
+            'include_tests%': 1,
+            'restrict_webrtc_logging%': 0,
+	  }],
+	],
       }],
       ['OS=="linux"', {
         'include_alsa_audio%': 1,
@@ -182,11 +212,13 @@
       }],
       ['OS=="ios"', {
         'build_libjpeg%': 0,
-        'enable_protobuf%': 0,
       }],
       ['target_arch=="arm" or target_arch=="arm64"', {
         'prefer_fixed_point%': 1,
       }],
+      ['(target_arch=="arm" and (arm_neon==1 or arm_neon_optional==1)) or target_arch=="arm64"', {
+        'build_with_neon%': 1,
+      }],
       ['OS!="ios" and (target_arch!="arm" or arm_version>=7) and target_arch!="mips64el" and build_with_mozilla==0', {
         'rtc_use_openmax_dl%': 1,
       }, {
@@ -248,7 +280,7 @@
           '<(DEPTH)',
           # The overrides must be included before the WebRTC root as that's the
           # mechanism for selecting the override headers in Chromium.
-          '../overrides',
+          '../../webrtc_overrides',
           # The WebRTC root is needed to allow includes in the WebRTC code base
           # to be prefixed with webrtc/.
           '../..',
@@ -294,13 +326,8 @@
       }],
       ['target_arch=="arm64"', {
         'defines': [
-          'WEBRTC_ARCH_ARM',
-          # TODO(zhongwei) Defining an unique WEBRTC_NEON and
-          # distinguishing ARMv7 NEON and ARM64 NEON by
-          # WEBRTC_ARCH_ARM_V7 and WEBRTC_ARCH_ARM64 should be better.
-
-          # This macro is used to distinguish ARMv7 NEON and ARM64 NEON
-          'WEBRTC_ARCH_ARM64_NEON',
+          'WEBRTC_ARCH_ARM64',
+          'WEBRTC_HAS_NEON',
         ],
       }],
       ['target_arch=="arm"', {
@@ -313,10 +340,10 @@
                         'WEBRTC_BUILD_NEON_LIBS'],
             'conditions': [
               ['arm_neon==1', {
-                'defines': ['WEBRTC_ARCH_ARM_NEON',],
+                'defines': ['WEBRTC_HAS_NEON',],
               }],
-              ['arm_neon==0 and arm_neon_optional==1', {
-                'defines': ['WEBRTC_DETECT_ARM_NEON',],
+              ['arm_neon==0 and (OS=="android")', {
+                'defines': ['WEBRTC_DETECT_NEON',],
               }],
             ],
           }],
@@ -328,13 +355,19 @@
           'WEBRTC_THREAD_RR',
         ],
       }],
+      ['OS=="dragonfly" or OS=="netbsd"', {
+        'defines': [
+          # doesn't support pthread_condattr_setclock
+          'WEBRTC_CLOCK_TYPE_REALTIME',
+        ],
+      }],
       ['OS=="openbsd"', {
         'defines' : [
           'WEBRTC_AUDIO_SNDIO',
         ],
       }],
       # Mozilla: if we support Mozilla on MIPS, we'll need to mod the cflags entries here
-      ['target_arch=="mipsel" and mips_arch_variant!="r6" and android_webview_build==0', {
+      ['target_arch=="mipsel" and mips_arch_variant!="r6"', {
         'defines': [
           'MIPS32_LE',
         ],
@@ -380,14 +413,12 @@
           'WEBRTC_IOS',
         ],
       }],
+      ['OS=="ios" and use_objc_h264==1', {
+        'defines': [
+          'WEBRTC_OBJC_H264',
+        ],
+      }],
       ['OS=="linux"', {
-#        'conditions': [
-#          ['have_clock_monotonic==1', {
-#            'defines': [
-#              'WEBRTC_CLOCK_TYPE_REALTIME',
-#            ],
-#          }],
-#        ],
         'defines': [
           'WEBRTC_LINUX',
         ],
@@ -412,7 +443,7 @@
         'msvs_disabled_warnings!': [4189,],
       }],
       # used on GONK as well
-      ['enable_android_opensl==1 and OS=="android"', {
+      ['enable_android_opensl==1 and (OS=="android")', {
         'defines': [
           'WEBRTC_ANDROID_OPENSLES',
         ],
@@ -428,7 +459,7 @@
           'WEBRTC_ANDROID',
          ],
          'conditions': [
-           ['clang!=1', {
+           ['clang==0', {
              # The Android NDK doesn't provide optimized versions of these
              # functions. Ensure they are disabled for all compilers.
              'cflags': [
@@ -440,6 +471,11 @@
            }],
          ],
       }],
+      ['include_internal_audio_device==1', {
+        'defines': [
+          'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE',
+        ],
+      }],
     ], # conditions
     'direct_dependent_settings': {
       'conditions': [
@@ -455,9 +491,9 @@
             'WEBRTC_CHROMIUM_BUILD',
           ],
           'include_dirs': [
-            # overrides must be included first as that is the mechanism for
+            # The overrides must be included first as that is the mechanism for
             # selecting the override headers in Chromium.
-            '../overrides',
+            '../../webrtc_overrides',
             '../..',
           ],
         }, {
@@ -491,13 +527,6 @@
             'WEBRTC_LINUX',
             'WEBRTC_ANDROID',
            ],
-           'conditions': [
-             ['enable_android_opensl==1', {
-               'defines': [
-                 'WEBRTC_ANDROID_OPENSLES',
-               ],
-             }]
-           ],
         }],
         ['os_posix==1', {
           # For access to standard POSIXish features, use WEBRTC_POSIX instead
diff --git a/media/webrtc/trunk/webrtc/build/get_landmines.py b/media/webrtc/trunk/webrtc/build/get_landmines.py
new file mode 100644
index 0000000000..748da5da03
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/build/get_landmines.py
@@ -0,0 +1,50 @@
+#!/usr/bin/env python
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""
+This file emits the list of reasons why a particular build needs to be clobbered
+(or a list of 'landmines').
+"""
+
+import os
+import sys
+
+script_dir = os.path.dirname(os.path.realpath(__file__))
+checkout_root = os.path.abspath(os.path.join(script_dir, os.pardir, os.pardir))
+sys.path.insert(0, os.path.join(checkout_root, 'build'))
+import landmine_utils
+
+
+builder = landmine_utils.builder
+distributor = landmine_utils.distributor
+gyp_defines = landmine_utils.gyp_defines
+gyp_msvs_version = landmine_utils.gyp_msvs_version
+platform = landmine_utils.platform
+
+
+def print_landmines():
+  """
+  ALL LANDMINES ARE EMITTED FROM HERE.
+  """
+  # DO NOT add landmines as part of a regular CL. Landmines are a last-effort
+  # bandaid fix if a CL that got landed has a build dependency bug and all bots
+  # need to be cleaned up. If you're writing a new CL that causes build
+  # dependency problems, fix the dependency problems instead of adding a
+  # landmine.
+  # See the Chromium version in src/build/get_landmines.py for usage examples.
+  print 'Clobber to remove out/{Debug,Release}/args.gn (webrtc:5070)'
+
+
+def main():
+  print_landmines()
+  return 0
+
+
+if __name__ == '__main__':
+  sys.exit(main())
diff --git a/media/webrtc/trunk/webrtc/build/gyp_webrtc b/media/webrtc/trunk/webrtc/build/gyp_webrtc
index edc6b36e49..cfe9ace1fc 100644
--- a/media/webrtc/trunk/webrtc/build/gyp_webrtc
+++ b/media/webrtc/trunk/webrtc/build/gyp_webrtc
@@ -38,6 +38,13 @@ def GetSupplementalFiles():
 if __name__ == '__main__':
   args = sys.argv[1:]
 
+  use_analyzer = len(args) and args[0] == '--analyzer'
+  if use_analyzer:
+    args.pop(0)
+    os.environ['GYP_GENERATORS'] = 'analyzer'
+    args.append('-Gconfig_path=' + args.pop(0))
+    args.append('-Ganalyzer_output_path=' + args.pop(0))
+
   if int(os.environ.get('GYP_CHROMIUM_NO_ACTION', 0)):
     print 'Skipping gyp_webrtc due to GYP_CHROMIUM_NO_ACTION env var.'
     sys.exit(0)
@@ -70,6 +77,13 @@ if __name__ == '__main__':
   if not os.environ.get('GYP_GENERATORS'):
     os.environ['GYP_GENERATORS'] = 'ninja'
 
+  # Enable check for missing sources in GYP files on Windows.
+  if sys.platform.startswith('win'):
+    gyp_generator_flags = os.getenv('GYP_GENERATOR_FLAGS', '')
+    if not 'msvs_error_on_missing_sources' in gyp_generator_flags:
+      os.environ['GYP_GENERATOR_FLAGS'] = (
+          gyp_generator_flags + ' msvs_error_on_missing_sources=1')
+
   vs2013_runtime_dll_dirs = None
   if int(os.environ.get('DEPOT_TOOLS_WIN_TOOLCHAIN', '1')):
     vs2013_runtime_dll_dirs = vs_toolchain.SetEnvironmentAndGetRuntimeDllDirs()
@@ -78,11 +92,11 @@ if __name__ == '__main__':
   args.append('--check')
 
   supplemental_includes = GetSupplementalFiles()
-  gn_vars_dict = gyp_chromium.GetGypVars(supplemental_includes)
+  gyp_vars = gyp_chromium.GetGypVars(supplemental_includes)
 
   # Automatically turn on crosscompile support for platforms that need it.
   if all(('ninja' in os.environ.get('GYP_GENERATORS', ''),
-          gn_vars_dict.get('OS') in ['android', 'ios'],
+          gyp_vars.get('OS') in ['android', 'ios'],
           'GYP_CROSSCOMPILE' not in os.environ)):
     os.environ['GYP_CROSSCOMPILE'] = '1'
 
@@ -93,13 +107,14 @@ if __name__ == '__main__':
   # Set the gyp depth variable to the root of the checkout.
   args.append('--depth=' + os.path.relpath(checkout_root))
 
-  print 'Updating projects from gyp files...'
-  sys.stdout.flush()
+  if not use_analyzer:
+    print 'Updating projects from gyp files...'
+    sys.stdout.flush()
 
   # Off we go...
   gyp_rc = gyp.main(args)
 
-  if vs2013_runtime_dll_dirs:
+  if vs2013_runtime_dll_dirs and not use_analyzer:
     x64_runtime, x86_runtime = vs2013_runtime_dll_dirs
     vs_toolchain.CopyVsRuntimeDlls(
         os.path.join(checkout_root, gyp_chromium.GetOutputDirectory()),
diff --git a/media/webrtc/trunk/webrtc/build/isolate.gypi b/media/webrtc/trunk/webrtc/build/isolate.gypi
index 86169fd094..ea44e2cc7f 100644
--- a/media/webrtc/trunk/webrtc/build/isolate.gypi
+++ b/media/webrtc/trunk/webrtc/build/isolate.gypi
@@ -8,14 +8,14 @@
 
 # Copied from Chromium's src/build/isolate.gypi
 #
-# It was necessary to copy this file to WebRTC, because the path to
-# build/common.gypi is different for the standalone and Chromium builds. Gyp
-# doesn't permit conditional inclusion or variable expansion in include paths.
+# It was necessary to copy this file because the path to build/common.gypi is
+# different for the standalone and Chromium builds. Gyp doesn't permit
+# conditional inclusion or variable expansion in include paths.
 # http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
 #
 # Local modifications:
 # * Removed include of '../chrome/version.gypi'.
-# * Removal passing of version_full variable created in version.gypi:
+# * Removed passing of version_full variable created in version.gypi:
 #   '--extra-variable', 'version_full=<(version_full)',
 
 # This file is meant to be included into a target to provide a rule
@@ -60,52 +60,82 @@
       'extension': 'isolate',
       'inputs': [
         # Files that are known to be involved in this step.
+        '<(DEPTH)/tools/isolate_driver.py',
         '<(DEPTH)/tools/swarming_client/isolate.py',
         '<(DEPTH)/tools/swarming_client/run_isolated.py',
       ],
-      'outputs': [
-        '<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated',
-      ],
+      'outputs': [],
       'action': [
         'python',
-        '<(DEPTH)/tools/swarming_client/isolate.py',
+        '<(DEPTH)/tools/isolate_driver.py',
         '<(test_isolation_mode)',
-        '--result', '<@(_outputs)',
+        '--isolated', '<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated',
         '--isolate', '<(RULE_INPUT_PATH)',
 
         # Variables should use the -V FOO=<(FOO) form so frequent values,
         # like '0' or '1', aren't stripped out by GYP. Run 'isolate.py help' for
         # more details.
-        #
-        # This list needs to be kept in sync with the cmd line options
-        # in src/build/android/pylib/gtest/setup.py.
 
         # Path variables are used to replace file paths when loading a .isolate
         # file
         '--path-variable', 'DEPTH', '<(DEPTH)',
         '--path-variable', 'PRODUCT_DIR', '<(PRODUCT_DIR) ',
 
+        # Note: This list must match DefaultConfigVariables()
+        # in build/android/pylib/utils/isolator.py
+        '--config-variable', 'CONFIGURATION_NAME=<(CONFIGURATION_NAME)',
         '--config-variable', 'OS=<(OS)',
+        '--config-variable', 'asan=<(asan)',
+        '--config-variable', 'branding=<(branding)',
         '--config-variable', 'chromeos=<(chromeos)',
         '--config-variable', 'component=<(component)',
+        '--config-variable', 'disable_nacl=<(disable_nacl)',
+        '--config-variable', 'enable_pepper_cdms=<(enable_pepper_cdms)',
+        '--config-variable', 'enable_plugins=<(enable_plugins)',
+        '--config-variable', 'fastbuild=<(fastbuild)',
+        '--config-variable', 'icu_use_data_file_flag=<(icu_use_data_file_flag)',
         # TODO(kbr): move this to chrome_tests.gypi:gles2_conform_tests_run
         # once support for user-defined config variables is added.
         '--config-variable',
           'internal_gles2_conform_tests=<(internal_gles2_conform_tests)',
-        '--config-variable', 'icu_use_data_file_flag=<(icu_use_data_file_flag)',
+        '--config-variable', 'kasko=<(kasko)',
+        '--config-variable', 'lsan=<(lsan)',
+        '--config-variable', 'msan=<(msan)',
+        '--config-variable', 'target_arch=<(target_arch)',
+        '--config-variable', 'tsan=<(tsan)',
+        '--config-variable', 'use_custom_libcxx=<(use_custom_libcxx)',
+        '--config-variable', 'use_instrumented_libraries=<(use_instrumented_libraries)',
+        '--config-variable',
+        'use_prebuilt_instrumented_libraries=<(use_prebuilt_instrumented_libraries)',
         '--config-variable', 'use_openssl=<(use_openssl)',
+        '--config-variable', 'use_ozone=<(use_ozone)',
+        '--config-variable', 'use_x11=<(use_x11)',
+        '--config-variable', 'v8_use_external_startup_data=<(v8_use_external_startup_data)',
       ],
       'conditions': [
         # Note: When gyp merges lists, it appends them to the old value.
         ['OS=="mac"', {
-          # <(mac_product_name) can contain a space, so don't use FOO=<(FOO)
-          # form.
           'action': [
-            '--extra-variable', 'mac_product_name', '<(mac_product_name)',
+            '--extra-variable', 'mac_product_name=<(mac_product_name)',
           ],
         }],
-        ["test_isolation_outdir!=''", {
-          'action': [ '--isolate-server', '<(test_isolation_outdir)' ],
+        ["test_isolation_mode == 'prepare'", {
+          'outputs': [
+            '<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated.gen.json',
+          ],
+        }, {
+          'outputs': [
+            '<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated',
+          ],
+        }],
+        ['OS=="win"', {
+          'action': [
+            '--config-variable', 'msvs_version=<(MSVS_VERSION)',
+          ],
+        }, {
+          'action': [
+            '--config-variable', 'msvs_version=0',
+          ],
         }],
       ],
     },
diff --git a/media/webrtc/trunk/webrtc/build/merge_libs.gyp b/media/webrtc/trunk/webrtc/build/merge_libs.gyp
index 3151c79478..de00a4efcf 100644
--- a/media/webrtc/trunk/webrtc/build/merge_libs.gyp
+++ b/media/webrtc/trunk/webrtc/build/merge_libs.gyp
@@ -49,6 +49,8 @@
       ],
     },
 #      }],
+#    ],
+#      }],
 #    ],
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/build/merge_libs.py b/media/webrtc/trunk/webrtc/build/merge_libs.py
index 84904d61ce..066e3ab39b 100644
--- a/media/webrtc/trunk/webrtc/build/merge_libs.py
+++ b/media/webrtc/trunk/webrtc/build/merge_libs.py
@@ -16,7 +16,7 @@ import os
 import subprocess
 import sys
 
-IGNORE_PATTERNS = ['do_not_use', 'protoc']
+IGNORE_PATTERNS = ['do_not_use', 'protoc', 'genperf']
 
 def FindFiles(path, pattern):
   """Finds files matching |pattern| under |path|.
@@ -36,7 +36,7 @@ def FindFiles(path, pattern):
   files = []
   for root, _, filenames in os.walk(path):
     for filename in fnmatch.filter(filenames, pattern):
-      if filename not in IGNORE_PATTERNS:
+      if all(pattern not in filename for pattern in IGNORE_PATTERNS):
         # We use the relative path here to avoid "argument list too
         # long" errors on Linux.  Note: This doesn't always work, so
         # we use the find command on Linux.
diff --git a/media/webrtc/trunk/webrtc/build/protoc.gypi b/media/webrtc/trunk/webrtc/build/protoc.gypi
index 5e486f16c2..682bc22cc5 100644
--- a/media/webrtc/trunk/webrtc/build/protoc.gypi
+++ b/media/webrtc/trunk/webrtc/build/protoc.gypi
@@ -109,10 +109,6 @@
       'process_outputs_as_sources': 1,
     },
   ],
-  'dependencies': [
-    '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
-    '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
-  ],
   'include_dirs': [
     '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
     '<(DEPTH)',
@@ -123,12 +119,20 @@
       '<(DEPTH)',
     ]
   },
-  'export_dependent_settings': [
-    # The generated headers reference headers within protobuf_lite,
-    # so dependencies must be able to find those headers too.
-    '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
-  ],
   # This target exports a hard dependency because it generates header
   # files.
   'hard_dependency': 1,
+  'conditions': [
+    ['build_protobuf==1', {
+      'dependencies': [
+        '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
+        '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+      ],
+      'export_dependent_settings': [
+        # The generated headers reference headers within protobuf_lite,
+        # so dependencies must be able to find those headers too.
+        '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+      ],
+    }],
+  ],
 }
diff --git a/media/webrtc/trunk/webrtc/build/sanitizers/lsan_suppressions_webrtc.cc b/media/webrtc/trunk/webrtc/build/sanitizers/lsan_suppressions_webrtc.cc
index 6d19a34881..61fdbbc20b 100644
--- a/media/webrtc/trunk/webrtc/build/sanitizers/lsan_suppressions_webrtc.cc
+++ b/media/webrtc/trunk/webrtc/build/sanitizers/lsan_suppressions_webrtc.cc
@@ -31,23 +31,6 @@ char kLSanDefaultSuppressions[] =
 // Leaks in Nvidia's libGL.
 "leak:libGL.so\n"
 
-// TODO(earthdok): revisit NSS suppressions after the switch to BoringSSL
-// NSS leaks in CertDatabaseNSSTest tests. http://crbug.com/51988
-"leak:net::NSSCertDatabase::ImportFromPKCS12\n"
-"leak:net::NSSCertDatabase::ListCerts\n"
-"leak:net::NSSCertDatabase::DeleteCertAndKey\n"
-"leak:crypto::ScopedTestNSSDB::ScopedTestNSSDB\n"
-// Another leak due to not shutting down NSS properly. http://crbug.com/124445
-"leak:error_get_my_stack\n"
-// The NSS suppressions above will not fire when the fast stack unwinder is
-// used, because it can't unwind through NSS libraries. Apply blanket
-// suppressions for now.
-"leak:libnssutil3\n"
-"leak:libnspr4\n"
-"leak:libnss3\n"
-"leak:libplds4\n"
-"leak:libnssckbi\n"
-
 // XRandR has several one time leaks.
 "leak:libxrandr\n"
 
diff --git a/media/webrtc/trunk/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc b/media/webrtc/trunk/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
index 38e62e927b..115099099f 100644
--- a/media/webrtc/trunk/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
+++ b/media/webrtc/trunk/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
@@ -27,7 +27,7 @@ char kTSanDefaultSuppressions[] =
 "race:rtc::MessageQueue::Quit\n"
 "race:FileVideoCapturerTest::VideoCapturerListener::OnFrameCaptured\n"
 "race:vp8cx_remove_encoder_threads\n"
-"race:third_party/libvpx/source/libvpx/vp9/common/vp9_scan.h\n"
+"race:third_party/libvpx_new/source/libvpx/vp9/common/vp9_scan.h\n"
 
 // Usage of trace callback and trace level is racy in libjingle_media_unittests.
 // https://code.google.com/p/webrtc/issues/detail?id=3372
@@ -42,12 +42,18 @@ char kTSanDefaultSuppressions[] =
 "race:webrtc/modules/audio_processing/aec/aec_core.c\n"
 "race:webrtc/modules/audio_processing/aec/aec_rdft.c\n"
 
+// Race in pulse initialization.
+// https://code.google.com/p/webrtc/issues/detail?id=5152
+"race:webrtc::AudioDeviceLinuxPulse::Init\n"
+
 // rtc_unittest
 // https://code.google.com/p/webrtc/issues/detail?id=3911 for details.
 "race:rtc::AsyncInvoker::OnMessage\n"
 "race:rtc::FireAndForgetAsyncClosure<FunctorB>::Execute\n"
 "race:rtc::MessageQueueManager::Clear\n"
 "race:rtc::Thread::Clear\n"
+// https://code.google.com/p/webrtc/issues/detail?id=3914
+"race:rtc::AsyncInvoker::~AsyncInvoker\n"
 // https://code.google.com/p/webrtc/issues/detail?id=2080
 "race:webrtc/base/logging.cc\n"
 "race:webrtc/base/sharedexclusivelock_unittest.cc\n"
@@ -66,21 +72,28 @@ char kTSanDefaultSuppressions[] =
 // TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492
 "race:user_sctp_timer_iterate\n"
 
+// https://code.google.com/p/webrtc/issues/detail?id=5151
+"race:sctp_close\n"
+
 // Potential deadlocks detected after roll in r6516.
 // https://code.google.com/p/webrtc/issues/detail?id=3509
 "deadlock:webrtc::RTCPReceiver::SetSsrcs\n"
 "deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n"
 "deadlock:webrtc::vcm::VideoReceiver::RegisterPacketRequestCallback\n"
-"deadlock:webrtc::ViECaptureImpl::ConnectCaptureDevice\n"
-"deadlock:webrtc::ViEChannel::StartSend\n"
-"deadlock:webrtc::ViECodecImpl::GetSendSideDelay\n"
 "deadlock:webrtc::ViEEncoder::OnLocalSsrcChanged\n"
-"deadlock:webrtc::ViESender::RegisterSendTransport\n"
 
 // TODO(pbos): Trace events are racy due to lack of proper POD atomics.
 // https://code.google.com/p/webrtc/issues/detail?id=2497
 "race:*trace_event_unique_catstatic*\n"
 
+// https://code.google.com/p/webrtc/issues/detail?id=4719
+"race:webrtc::voe::TransmitMixer::PrepareDemux\n"
+"race:webrtc::voe::TransmitMixer::EnableStereoChannelSwapping\n"
+
+// Race between InitCpuFlags and TestCpuFlag in libyuv.
+// https://code.google.com/p/libyuv/issues/detail?id=508
+"race:InitCpuFlags\n"
+
 // End of suppressions.
 ;  // Please keep this semicolon.
 
diff --git a/media/webrtc/trunk/webrtc/build/webrtc.gni b/media/webrtc/trunk/webrtc/build/webrtc.gni
index 2315f11eec..c55f4230bd 100644
--- a/media/webrtc/trunk/webrtc/build/webrtc.gni
+++ b/media/webrtc/trunk/webrtc/build/webrtc.gni
@@ -8,11 +8,9 @@
 
 import("//build/config/arm.gni")
 import("//build/config/mips.gni")
+import("//build_overrides/webrtc.gni")
 
 declare_args() {
-  # Assume Chromium build for now, since that's the priority case for getting GN
-  # up and running with WebRTC.
-  build_with_chromium = true
   build_with_libjingle = true
 
   # Disable this to avoid building the Opus audio codec.
@@ -37,13 +35,14 @@ declare_args() {
   rtc_enable_protobuf = true
 
   # Disable these to not build components which can be externally provided.
+  rtc_build_expat = true
   rtc_build_json = true
   rtc_build_libjpeg = true
-  rtc_build_libyuv = true
   rtc_build_libvpx = true
-  rtc_build_vp9 = true
-  rtc_build_ssl = true
+  rtc_build_libyuv = true
+  rtc_build_openmax_dl = true
   rtc_build_opus = true
+  rtc_build_ssl = true
 
   # Disable by default.
   rtc_have_dbus_glib = false
@@ -58,26 +57,6 @@ declare_args() {
   # https://gcc.gnu.org/wiki/LinkTimeOptimization
   rtc_use_lto = false
 
-  if (build_with_chromium) {
-    # Exclude pulse audio on Chromium since its prerequisites don't require
-    # pulse audio.
-    rtc_include_pulse_audio = false
-
-    # Exclude internal ADM since Chromium uses its own IO handling.
-    rtc_include_internal_audio_device = false
-
-  } else {
-    # Settings for the standalone (not-in-Chromium) build.
-
-    # TODO(andrew): For now, disable the Chrome plugins, which causes a
-    # flood of chromium-style warnings. Investigate enabling them:
-    # http://code.google.com/p/webrtc/issues/detail?id=163
-    clang_use_chrome_plugins = false
-
-    rtc_include_pulse_audio = true
-    rtc_include_internal_audio_device = true
-  }
-
   if (build_with_libjingle) {
     rtc_include_tests = false
     rtc_restrict_logging = true
@@ -103,14 +82,29 @@ declare_args() {
     rtc_use_openmax_dl = false
   }
 
-  # WebRTC builds ARM v7 Neon instruction set optimized code for both iOS and
-  # Android, which is why we currently cannot use the variables in
-  # //build/config/arm.gni (since it disables Neon for Android).
-  rtc_build_armv7_neon = (current_cpu == "arm" && arm_version >= 7)
+  # Determines whether NEON code will be built.
+  rtc_build_with_neon =
+      (current_cpu == "arm" && (arm_use_neon || arm_optionally_use_neon)) ||
+      current_cpu == "arm64"
+
+  # Enable this to use HW H.264 encoder/decoder on iOS PeerConnections.
+  # Enabling this may break interop with Android clients that support H264.
+  rtc_use_objc_h264 = false
+
+  # Enable this to build H.264 encoder/decoder using third party libraries.
+  # Encoding uses OpenH264 and decoding uses FFmpeg. Because of this, OpenH264
+  # and FFmpeg have to be correctly enabled separately.
+  # - use_openh264=true is required for OpenH264 targets to be defined.
+  # - ffmpeg_branding="Chrome" is one way to support H.264 decoding in FFmpeg.
+  #   FFmpeg can be built with/without H.264 support, see 'ffmpeg_branding'.
+  #   Without it, it compiles but H264DecoderImpl fails to initialize.
+  # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
+  # http://www.openh264.org, https://www.ffmpeg.org/
+  use_third_party_h264 = false  # TODO(hbos): To be used in follow-up CL(s).
 }
 
 # Make it possible to provide custom locations for some libraries (move these
 # up into declare_args should we need to actually use them for the GN build).
-rtc_libvpx_dir = "//third_party/libvpx"
+rtc_libvpx_dir = "//third_party/libvpx_new"
 rtc_libyuv_dir = "//third_party/libyuv"
 rtc_opus_dir = "//third_party/opus"
diff --git a/media/webrtc/trunk/webrtc/call.h b/media/webrtc/trunk/webrtc/call.h
index f5b7fed545..f4cc29d61e 100644
--- a/media/webrtc/trunk/webrtc/call.h
+++ b/media/webrtc/trunk/webrtc/call.h
@@ -14,15 +14,26 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
+#include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio_send_stream.h"
+#include "webrtc/audio_state.h"
+#include "webrtc/base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
-class VoiceEngine;
+class AudioProcessing;
 
 const char* Version();
 
+enum class MediaType {
+  ANY,
+  AUDIO,
+  VIDEO,
+  DATA
+};
+
 class PacketReceiver {
  public:
   enum DeliveryStatus {
@@ -31,8 +42,10 @@ class PacketReceiver {
     DELIVERY_PACKET_ERROR,
   };
 
-  virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                       size_t length) = 0;
+  virtual DeliveryStatus DeliverPacket(MediaType media_type,
+                                       const uint8_t* packet,
+                                       size_t length,
+                                       const PacketTime& packet_time) = 0;
 
  protected:
   virtual ~PacketReceiver() {}
@@ -56,65 +69,47 @@ class LoadObserver {
 // etc.
 class Call {
  public:
-  enum NetworkState {
-    kNetworkUp,
-    kNetworkDown,
-  };
   struct Config {
-    explicit Config(newapi::Transport* send_transport)
-        : webrtc_config(NULL),
-          send_transport(send_transport),
-          voice_engine(NULL),
-          overuse_callback(NULL) {}
-
     static const int kDefaultStartBitrateBps;
 
-    webrtc::Config* webrtc_config;
-
-    newapi::Transport* send_transport;
-
-    // VoiceEngine used for audio/video synchronization for this Call.
-    VoiceEngine* voice_engine;
-
-    // Callback for overuse and normal usage based on the jitter of incoming
-    // captured frames. 'NULL' disables the callback.
-    LoadObserver* overuse_callback;
-
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
-      BitrateConfig()
-          : min_bitrate_bps(0),
-            start_bitrate_bps(kDefaultStartBitrateBps),
-            max_bitrate_bps(-1) {}
-      int min_bitrate_bps;
-      int start_bitrate_bps;
-      int max_bitrate_bps;
+      int min_bitrate_bps = 0;
+      int start_bitrate_bps = kDefaultStartBitrateBps;
+      int max_bitrate_bps = -1;
     } bitrate_config;
+
+    // AudioState which is possibly shared between multiple calls.
+    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+    rtc::scoped_refptr<AudioState> audio_state;
+
+    // Audio Processing Module to be used in this call.
+    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+    AudioProcessing* audio_processing = nullptr;
   };
 
   struct Stats {
-    Stats()
-        : send_bandwidth_bps(0),
-          recv_bandwidth_bps(0),
-          pacer_delay_ms(0),
-          rtt_ms(-1) {}
-
-    int send_bandwidth_bps;
-    int recv_bandwidth_bps;
-    int64_t pacer_delay_ms;
-    int64_t rtt_ms;
+    int send_bandwidth_bps = 0;
+    int recv_bandwidth_bps = 0;
+    int64_t pacer_delay_ms = 0;
+    int64_t rtt_ms = -1;
   };
 
   static Call* Create(const Call::Config& config);
 
-  static Call* Create(const Call::Config& config,
-                      const webrtc::Config& webrtc_config);
+  virtual AudioSendStream* CreateAudioSendStream(
+      const AudioSendStream::Config& config) = 0;
+  virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
+
+  virtual AudioReceiveStream* CreateAudioReceiveStream(
+      const AudioReceiveStream::Config& config) = 0;
+  virtual void DestroyAudioReceiveStream(
+      AudioReceiveStream* receive_stream) = 0;
 
   virtual VideoSendStream* CreateVideoSendStream(
       const VideoSendStream::Config& config,
       const VideoEncoderConfig& encoder_config) = 0;
-
   virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
 
   virtual VideoReceiveStream* CreateVideoReceiveStream(
@@ -140,8 +135,13 @@ class Call {
       const Config::BitrateConfig& bitrate_config) = 0;
   virtual void SignalNetworkState(NetworkState state) = 0;
 
+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
+
+  virtual VoiceEngine* voice_engine() = 0;
+
   virtual ~Call() {}
 };
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_
diff --git a/media/webrtc/trunk/webrtc/call/BUILD.gn b/media/webrtc/trunk/webrtc/call/BUILD.gn
new file mode 100644
index 0000000000..498c724900
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/BUILD.gn
@@ -0,0 +1,35 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+
+source_set("call") {
+  sources = [
+    "bitrate_allocator.cc",
+    "call.cc",
+    "congestion_controller.cc",
+    "transport_adapter.cc",
+    "transport_adapter.h",
+  ]
+
+  configs += [ "..:common_config" ]
+  public_configs = [ "..:common_inherited_config" ]
+
+  if (is_clang) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+
+  deps = [
+    "..:rtc_event_log",
+    "..:webrtc_common",
+    "../modules/rtp_rtcp",
+    "../system_wrappers",
+  ]
+}
diff --git a/media/webrtc/trunk/webrtc/video_engine/OWNERS b/media/webrtc/trunk/webrtc/call/OWNERS
similarity index 85%
rename from media/webrtc/trunk/webrtc/video_engine/OWNERS
rename to media/webrtc/trunk/webrtc/call/OWNERS
index 5e47b1ae06..792de19042 100644
--- a/media/webrtc/trunk/webrtc/video_engine/OWNERS
+++ b/media/webrtc/trunk/webrtc/call/OWNERS
@@ -1,12 +1,11 @@
 mflodman@webrtc.org
+pbos@webrtc.org
+solenberg@webrtc.org
 stefan@webrtc.org
 
-per-file *.isolate=kjellander@webrtc.org
-
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
 per-file *.gyp=*
 per-file *.gypi=*
 
 per-file BUILD.gn=kjellander@webrtc.org
-
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator.cc b/media/webrtc/trunk/webrtc/call/bitrate_allocator.cc
similarity index 71%
rename from media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator.cc
rename to media/webrtc/trunk/webrtc/call/bitrate_allocator.cc
index fc83e060a8..b3789d3bb6 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator.cc
+++ b/media/webrtc/trunk/webrtc/call/bitrate_allocator.cc
@@ -9,7 +9,7 @@
  *
  */
 
-#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
+#include "webrtc/call/bitrate_allocator.h"
 
 #include <algorithm>
 #include <utility>
@@ -26,23 +26,26 @@ const int kDefaultBitrateBps = 300000;
 BitrateAllocator::BitrateAllocator()
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       bitrate_observers_(),
+      bitrate_observers_modified_(false),
       enforce_min_bitrate_(true),
       last_bitrate_bps_(kDefaultBitrateBps),
       last_fraction_loss_(0),
-      last_rtt_(0) {
-}
+      last_rtt_(0) {}
 
-
-void BitrateAllocator::OnNetworkChanged(uint32_t bitrate,
-                                        uint8_t fraction_loss,
-                                        int64_t rtt) {
+uint32_t BitrateAllocator::OnNetworkChanged(uint32_t bitrate,
+                                            uint8_t fraction_loss,
+                                            int64_t rtt) {
   CriticalSectionScoped lock(crit_sect_.get());
   last_bitrate_bps_ = bitrate;
   last_fraction_loss_ = fraction_loss;
   last_rtt_ = rtt;
+  uint32_t allocated_bitrate_bps = 0;
   ObserverBitrateMap allocation = AllocateBitrates();
-  for (const auto& kv : allocation)
+  for (const auto& kv : allocation) {
     kv.first->OnNetworkChanged(kv.second, last_fraction_loss_, last_rtt_);
+    allocated_bitrate_bps += kv.second;
+  }
+  return allocated_bitrate_bps;
 }
 
 BitrateAllocator::ObserverBitrateMap BitrateAllocator::AllocateBitrates() {
@@ -51,7 +54,7 @@ BitrateAllocator::ObserverBitrateMap BitrateAllocator::AllocateBitrates() {
 
   uint32_t sum_min_bitrates = 0;
   for (const auto& observer : bitrate_observers_)
-    sum_min_bitrates += observer.second.min_bitrate_;
+    sum_min_bitrates += observer.second.min_bitrate;
   if (last_bitrate_bps_ <= sum_min_bitrates)
     return LowRateAllocation(last_bitrate_bps_);
   else
@@ -59,10 +62,8 @@ BitrateAllocator::ObserverBitrateMap BitrateAllocator::AllocateBitrates() {
 }
 
 int BitrateAllocator::AddBitrateObserver(BitrateObserver* observer,
-                                         uint32_t start_bitrate_bps,
                                          uint32_t min_bitrate_bps,
-                                         uint32_t max_bitrate_bps,
-                                         int* new_observer_bitrate_bps) {
+                                         uint32_t max_bitrate_bps) {
   CriticalSectionScoped lock(crit_sect_.get());
 
   BitrateObserverConfList::iterator it =
@@ -73,43 +74,25 @@ int BitrateAllocator::AddBitrateObserver(BitrateObserver* observer,
   // properly allocate bitrate. The allocator should instead distribute any
   // extra bitrate after all streams have maxed out.
   max_bitrate_bps *= kTransmissionMaxBitrateMultiplier;
-  int new_bwe_candidate_bps = 0;
   if (it != bitrate_observers_.end()) {
     // Update current configuration.
-    it->second.start_bitrate_ = start_bitrate_bps;
-    it->second.min_bitrate_ = min_bitrate_bps;
-    it->second.max_bitrate_ = max_bitrate_bps;
-    // Set the send-side bandwidth to the max of the sum of start bitrates and
-    // the current estimate, so that if the user wants to immediately use more
-    // bandwidth, that can be enforced.
-    for (const auto& observer : bitrate_observers_)
-      new_bwe_candidate_bps += observer.second.start_bitrate_;
+    it->second.min_bitrate = min_bitrate_bps;
+    it->second.max_bitrate = max_bitrate_bps;
   } else {
     // Add new settings.
     bitrate_observers_.push_back(BitrateObserverConfiguration(
-        observer, BitrateConfiguration(start_bitrate_bps, min_bitrate_bps,
-                                       max_bitrate_bps)));
+        observer, BitrateConfiguration(min_bitrate_bps, max_bitrate_bps)));
     bitrate_observers_modified_ = true;
-
-    // TODO(andresp): This is a ugly way to set start bitrate.
-    //
-    // Only change start bitrate if we have exactly one observer. By definition
-    // you can only have one start bitrate, once we have our first estimate we
-    // will adapt from there.
-    if (bitrate_observers_.size() == 1)
-      new_bwe_candidate_bps = start_bitrate_bps;
   }
 
-  last_bitrate_bps_ = std::max<int>(new_bwe_candidate_bps, last_bitrate_bps_);
-
   ObserverBitrateMap allocation = AllocateBitrates();
-  *new_observer_bitrate_bps = 0;
+  int new_observer_bitrate_bps = 0;
   for (auto& kv : allocation) {
     kv.first->OnNetworkChanged(kv.second, last_fraction_loss_, last_rtt_);
     if (kv.first == observer)
-      *new_observer_bitrate_bps = kv.second;
+      new_observer_bitrate_bps = kv.second;
   }
-  return last_bitrate_bps_;
+  return new_observer_bitrate_bps;
 }
 
 void BitrateAllocator::RemoveBitrateObserver(BitrateObserver* observer) {
@@ -129,8 +112,8 @@ void BitrateAllocator::GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
 
   CriticalSectionScoped lock(crit_sect_.get());
   for (const auto& observer : bitrate_observers_) {
-    *min_bitrate_sum_bps += observer.second.min_bitrate_;
-    *max_bitrate_sum_bps += observer.second.max_bitrate_;
+    *min_bitrate_sum_bps += observer.second.min_bitrate;
+    *max_bitrate_sum_bps += observer.second.max_bitrate;
   }
 }
 
@@ -153,22 +136,23 @@ void BitrateAllocator::EnforceMinBitrate(bool enforce_min_bitrate) {
 BitrateAllocator::ObserverBitrateMap BitrateAllocator::NormalRateAllocation(
     uint32_t bitrate,
     uint32_t sum_min_bitrates) {
-  uint32_t number_of_observers = bitrate_observers_.size();
+  uint32_t number_of_observers =
+      static_cast<uint32_t>(bitrate_observers_.size());
   uint32_t bitrate_per_observer =
       (bitrate - sum_min_bitrates) / number_of_observers;
   // Use map to sort list based on max bitrate.
   ObserverSortingMap list_max_bitrates;
   for (const auto& observer : bitrate_observers_) {
     list_max_bitrates.insert(std::pair<uint32_t, ObserverConfiguration>(
-        observer.second.max_bitrate_,
-        ObserverConfiguration(observer.first, observer.second.min_bitrate_)));
+        observer.second.max_bitrate,
+        ObserverConfiguration(observer.first, observer.second.min_bitrate)));
   }
   ObserverBitrateMap allocation;
   ObserverSortingMap::iterator max_it = list_max_bitrates.begin();
   while (max_it != list_max_bitrates.end()) {
     number_of_observers--;
     uint32_t observer_allowance =
-        max_it->second.min_bitrate_ + bitrate_per_observer;
+        max_it->second.min_bitrate + bitrate_per_observer;
     if (max_it->first < observer_allowance) {
       // We have more than enough for this observer.
       // Carry the remainder forward.
@@ -176,9 +160,9 @@ BitrateAllocator::ObserverBitrateMap BitrateAllocator::NormalRateAllocation(
       if (number_of_observers != 0) {
         bitrate_per_observer += remainder / number_of_observers;
       }
-      allocation[max_it->second.observer_] = max_it->first;
+      allocation[max_it->second.observer] = max_it->first;
     } else {
-      allocation[max_it->second.observer_] = observer_allowance;
+      allocation[max_it->second.observer] = observer_allowance;
     }
     list_max_bitrates.erase(max_it);
     // Prepare next iteration.
@@ -193,14 +177,14 @@ BitrateAllocator::ObserverBitrateMap BitrateAllocator::LowRateAllocation(
   if (enforce_min_bitrate_) {
     // Min bitrate to all observers.
     for (const auto& observer : bitrate_observers_)
-      allocation[observer.first] = observer.second.min_bitrate_;
+      allocation[observer.first] = observer.second.min_bitrate;
   } else {
-    // Allocate up to |min_bitrate_| to one observer at a time, until
+    // Allocate up to |min_bitrate| to one observer at a time, until
     // |bitrate| is depleted.
     uint32_t remainder = bitrate;
     for (const auto& observer : bitrate_observers_) {
       uint32_t allocated_bitrate =
-          std::min(remainder, observer.second.min_bitrate_);
+          std::min(remainder, observer.second.min_bitrate);
       allocation[observer.first] = allocated_bitrate;
       remainder -= allocated_bitrate;
     }
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_allocator.h b/media/webrtc/trunk/webrtc/call/bitrate_allocator.h
similarity index 71%
rename from media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_allocator.h
rename to media/webrtc/trunk/webrtc/call/bitrate_allocator.h
index 9cc4b74711..4a3fd59d49 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_allocator.h
+++ b/media/webrtc/trunk/webrtc/call/bitrate_allocator.h
@@ -12,8 +12,8 @@
  *  and push the result to the encoders via BitrateObserver(s).
  */
 
-#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_ALLOCATOR_H_
-#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_ALLOCATOR_H_
+#ifndef WEBRTC_CALL_BITRATE_ALLOCATOR_H_
+#define WEBRTC_CALL_BITRATE_ALLOCATOR_H_
 
 #include <list>
 #include <map>
@@ -21,7 +21,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -31,21 +31,22 @@ class BitrateAllocator {
  public:
   BitrateAllocator();
 
-  void OnNetworkChanged(uint32_t target_bitrate,
-                        uint8_t fraction_loss,
-                        int64_t rtt);
+  // Allocate target_bitrate across the registered BitrateObservers.
+  // Returns actual bitrate allocated (might be higher than target_bitrate if
+  // for instance EnforceMinBitrate() is enabled.
+  uint32_t OnNetworkChanged(uint32_t target_bitrate,
+                            uint8_t fraction_loss,
+                            int64_t rtt);
 
   // Set the start and max send bitrate used by the bandwidth management.
   //
-  // observer, updates bitrates if already in use.
-  // min_bitrate_bps = 0 equals no min bitrate.
-  // max_bitrate_bps = 0 equals no max bitrate.
-  // TODO(holmer): Remove start_bitrate_bps when old API is gone.
+  // |observer| updates bitrates if already in use.
+  // |min_bitrate_bps| = 0 equals no min bitrate.
+  // |max_bitrate_bps| = 0 equals no max bitrate.
+  // Returns bitrate allocated for the bitrate observer.
   int AddBitrateObserver(BitrateObserver* observer,
-                         uint32_t start_bitrate_bps,
                          uint32_t min_bitrate_bps,
-                         uint32_t max_bitrate_bps,
-                         int* new_observer_bitrate_bps);
+                         uint32_t max_bitrate_bps);
 
   void RemoveBitrateObserver(BitrateObserver* observer);
 
@@ -61,21 +62,16 @@ class BitrateAllocator {
 
  private:
   struct BitrateConfiguration {
-    BitrateConfiguration(uint32_t start_bitrate,
-                         uint32_t min_bitrate,
-                         uint32_t max_bitrate)
-        : start_bitrate_(start_bitrate),
-          min_bitrate_(min_bitrate),
-          max_bitrate_(max_bitrate) {}
-    uint32_t start_bitrate_;
-    uint32_t min_bitrate_;
-    uint32_t max_bitrate_;
+    BitrateConfiguration(uint32_t min_bitrate, uint32_t max_bitrate)
+        : min_bitrate(min_bitrate), max_bitrate(max_bitrate) {}
+    uint32_t min_bitrate;
+    uint32_t max_bitrate;
   };
   struct ObserverConfiguration {
     ObserverConfiguration(BitrateObserver* observer, uint32_t bitrate)
-        : observer_(observer), min_bitrate_(bitrate) {}
-    BitrateObserver* observer_;
-    uint32_t min_bitrate_;
+        : observer(observer), min_bitrate(bitrate) {}
+    BitrateObserver* const observer;
+    uint32_t min_bitrate;
   };
   typedef std::pair<BitrateObserver*, BitrateConfiguration>
       BitrateObserverConfiguration;
@@ -103,4 +99,4 @@ class BitrateAllocator {
   int64_t last_rtt_ GUARDED_BY(crit_sect_);
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_ALLOCATOR_H_
+#endif  // WEBRTC_CALL_BITRATE_ALLOCATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc b/media/webrtc/trunk/webrtc/call/bitrate_allocator_unittest.cc
similarity index 78%
rename from media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc
rename to media/webrtc/trunk/webrtc/call/bitrate_allocator_unittest.cc
index b69247e861..86f75a4380 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_allocator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/call/bitrate_allocator_unittest.cc
@@ -12,7 +12,7 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
+#include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 
 namespace webrtc {
@@ -46,22 +46,24 @@ class BitrateAllocatorTest : public ::testing::Test {
 
 TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
   TestBitrateObserver bitrate_observer;
-  int start_bitrate;
-  allocator_->AddBitrateObserver(&bitrate_observer, 200000, 100000, 1500000,
-                                 &start_bitrate);
+  int start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer, 100000, 1500000);
   EXPECT_EQ(300000, start_bitrate);
   allocator_->OnNetworkChanged(200000, 0, 0);
   EXPECT_EQ(200000u, bitrate_observer.last_bitrate_);
 
-  allocator_->AddBitrateObserver(&bitrate_observer, 1500000, 100000, 1500000,
-                                 &start_bitrate);
-  EXPECT_EQ(1500000, start_bitrate);
-  allocator_->OnNetworkChanged(1500000, 0, 0);
-  EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_);
+  // TODO(pbos): Expect capping to 1.5M instead of 3M when not boosting the max
+  // bitrate for FEC/retransmissions (see todo in BitrateAllocator).
+  allocator_->OnNetworkChanged(4000000, 0, 0);
+  EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer, 100000, 4000000);
+  EXPECT_EQ(4000000, start_bitrate);
 
-  allocator_->AddBitrateObserver(&bitrate_observer, 500000, 100000, 1500000,
-                                 &start_bitrate);
-  EXPECT_EQ(1500000, start_bitrate);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer, 100000, 1500000);
+  EXPECT_EQ(3000000, start_bitrate);
+  EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
   allocator_->OnNetworkChanged(1500000, 0, 0);
   EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_);
 }
@@ -69,12 +71,11 @@ TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
 TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
   TestBitrateObserver bitrate_observer_1;
   TestBitrateObserver bitrate_observer_2;
-  int start_bitrate;
-  allocator_->AddBitrateObserver(&bitrate_observer_1, 200000, 100000, 300000,
-                                 &start_bitrate);
+  int start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 300000);
   EXPECT_EQ(300000, start_bitrate);
-  allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 200000, 300000,
-                                 &start_bitrate);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 300000);
   EXPECT_EQ(200000, start_bitrate);
 
   // Test too low start bitrate, hence lower than sum of min. Min bitrates will
@@ -114,9 +115,8 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
 // as intended.
 TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
   TestBitrateObserver bitrate_observer_1;
-  int start_bitrate;
-  allocator_->AddBitrateObserver(&bitrate_observer_1, 200000, 100000, 400000,
-                                 &start_bitrate);
+  int start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
   EXPECT_EQ(300000, start_bitrate);
 
   // High REMB.
@@ -135,18 +135,17 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
   TestBitrateObserver bitrate_observer_2;
   TestBitrateObserver bitrate_observer_3;
   // Set up the observers with min bitrates at 100000, 200000, and 300000.
-  int start_bitrate;
-  allocator_->AddBitrateObserver(&bitrate_observer_1, 200000, 100000, 400000,
-                                 &start_bitrate);
+  int start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
   EXPECT_EQ(300000, start_bitrate);
 
-  allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 200000, 400000,
-                                 &start_bitrate);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 400000);
   EXPECT_EQ(200000, start_bitrate);
   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
 
-  allocator_->AddBitrateObserver(&bitrate_observer_3, 200000, 300000, 400000,
-                                 &start_bitrate);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_3, 300000, 400000);
   EXPECT_EQ(0, start_bitrate);
   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
   EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);
@@ -185,18 +184,17 @@ TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowRembEnforceMin) {
   TestBitrateObserver bitrate_observer_1;
   TestBitrateObserver bitrate_observer_2;
   TestBitrateObserver bitrate_observer_3;
-  int start_bitrate;
-  allocator_->AddBitrateObserver(&bitrate_observer_1, 200000, 100000, 400000,
-                                 &start_bitrate);
+  int start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
   EXPECT_EQ(300000, start_bitrate);
 
-  allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 200000, 400000,
-                                 &start_bitrate);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 400000);
   EXPECT_EQ(200000, start_bitrate);
   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
 
-  allocator_->AddBitrateObserver(&bitrate_observer_3, 200000, 300000, 400000,
-                                 &start_bitrate);
+  start_bitrate =
+      allocator_->AddBitrateObserver(&bitrate_observer_3, 300000, 400000);
   EXPECT_EQ(300000, start_bitrate);
   EXPECT_EQ(100000, static_cast<int>(bitrate_observer_1.last_bitrate_));
   EXPECT_EQ(200000, static_cast<int>(bitrate_observer_2.last_bitrate_));
diff --git a/media/webrtc/trunk/webrtc/call/bitrate_estimator_tests.cc b/media/webrtc/trunk/webrtc/call/bitrate_estimator_tests.cc
new file mode 100644
index 0000000000..4b24bbd5ef
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/bitrate_estimator_tests.cc
@@ -0,0 +1,353 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <functional>
+#include <list>
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio_state.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/call.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/call_test.h"
+#include "webrtc/test/direct_transport.h"
+#include "webrtc/test/encoder_settings.h"
+#include "webrtc/test/fake_decoder.h"
+#include "webrtc/test/fake_encoder.h"
+#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/test/frame_generator_capturer.h"
+
+namespace webrtc {
+namespace {
+// Note: If you consider to re-use this class, think twice and instead consider
+// writing tests that don't depend on the logging system.
+class LogObserver {
+ public:
+  LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
+
+  ~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
+
+  void PushExpectedLogLine(const std::string& expected_log_line) {
+    callback_.PushExpectedLogLine(expected_log_line);
+  }
+
+  bool Wait() { return callback_.Wait(); }
+
+ private:
+  class Callback : public rtc::LogSink {
+   public:
+    Callback() : done_(false, false) {}
+
+    void OnLogMessage(const std::string& message) override {
+      rtc::CritScope lock(&crit_sect_);
+      // Ignore log lines that are due to missing AST extensions, these are
+      // logged when we switch back from AST to TOF until the wrapping bitrate
+      // estimator gives up on using AST.
+      if (message.find("BitrateEstimator") != std::string::npos &&
+          message.find("packet is missing") == std::string::npos) {
+        received_log_lines_.push_back(message);
+      }
+
+      int num_popped = 0;
+      while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
+        std::string a = received_log_lines_.front();
+        std::string b = expected_log_lines_.front();
+        received_log_lines_.pop_front();
+        expected_log_lines_.pop_front();
+        num_popped++;
+        EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
+      }
+      if (expected_log_lines_.size() <= 0) {
+        if (num_popped > 0) {
+          done_.Set();
+        }
+        return;
+      }
+    }
+
+    bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
+
+    void PushExpectedLogLine(const std::string& expected_log_line) {
+      rtc::CritScope lock(&crit_sect_);
+      expected_log_lines_.push_back(expected_log_line);
+    }
+
+   private:
+    typedef std::list<std::string> Strings;
+    rtc::CriticalSection crit_sect_;
+    Strings received_log_lines_ GUARDED_BY(crit_sect_);
+    Strings expected_log_lines_ GUARDED_BY(crit_sect_);
+    rtc::Event done_;
+  };
+
+  Callback callback_;
+};
+}  // namespace
+
+static const int kTOFExtensionId = 4;
+static const int kASTExtensionId = 5;
+
+class BitrateEstimatorTest : public test::CallTest {
+ public:
+  BitrateEstimatorTest() : receive_config_(nullptr) {}
+
+  virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
+
+  virtual void SetUp() {
+    AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = &mock_voice_engine_;
+    Call::Config config;
+    config.audio_state = AudioState::Create(audio_state_config);
+    receiver_call_.reset(Call::Create(config));
+    sender_call_.reset(Call::Create(config));
+
+    send_transport_.reset(new test::DirectTransport(sender_call_.get()));
+    send_transport_->SetReceiver(receiver_call_->Receiver());
+    receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
+    receive_transport_->SetReceiver(sender_call_->Receiver());
+
+    video_send_config_ = VideoSendStream::Config(send_transport_.get());
+    video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
+    // Encoders will be set separately per stream.
+    video_send_config_.encoder_settings.encoder = nullptr;
+    video_send_config_.encoder_settings.payload_name = "FAKE";
+    video_send_config_.encoder_settings.payload_type =
+        kFakeVideoSendPayloadType;
+    video_encoder_config_.streams = test::CreateVideoStreams(1);
+
+    receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
+    // receive_config_.decoders will be set by every stream separately.
+    receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
+    receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+    receive_config_.rtp.remb = true;
+    receive_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+    receive_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+  }
+
+  virtual void TearDown() {
+    std::for_each(streams_.begin(), streams_.end(),
+                  std::mem_fun(&Stream::StopSending));
+
+    send_transport_->StopSending();
+    receive_transport_->StopSending();
+
+    while (!streams_.empty()) {
+      delete streams_.back();
+      streams_.pop_back();
+    }
+
+    receiver_call_.reset();
+    sender_call_.reset();
+  }
+
+ protected:
+  friend class Stream;
+
+  class Stream {
+   public:
+    Stream(BitrateEstimatorTest* test, bool receive_audio)
+        : test_(test),
+          is_sending_receiving_(false),
+          send_stream_(nullptr),
+          audio_receive_stream_(nullptr),
+          video_receive_stream_(nullptr),
+          frame_generator_capturer_(),
+          fake_encoder_(Clock::GetRealTimeClock()),
+          fake_decoder_() {
+      test_->video_send_config_.rtp.ssrcs[0]++;
+      test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
+      send_stream_ = test_->sender_call_->CreateVideoSendStream(
+          test_->video_send_config_, test_->video_encoder_config_);
+      RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
+      frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
+          send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
+          test_->video_encoder_config_.streams[0].height, 30,
+          Clock::GetRealTimeClock()));
+      send_stream_->Start();
+      frame_generator_capturer_->Start();
+
+      if (receive_audio) {
+        AudioReceiveStream::Config receive_config;
+        receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
+        // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
+        // the AudioReceiveStream. Every receive stream has to correspond to
+        // an underlying channel id.
+        receive_config.voe_channel_id = 0;
+        receive_config.rtp.extensions.push_back(
+            RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+        receive_config.combined_audio_video_bwe = true;
+        audio_receive_stream_ =
+            test_->receiver_call_->CreateAudioReceiveStream(receive_config);
+      } else {
+        VideoReceiveStream::Decoder decoder;
+        decoder.decoder = &fake_decoder_;
+        decoder.payload_type =
+            test_->video_send_config_.encoder_settings.payload_type;
+        decoder.payload_name =
+            test_->video_send_config_.encoder_settings.payload_name;
+        test_->receive_config_.decoders.clear();
+        test_->receive_config_.decoders.push_back(decoder);
+        test_->receive_config_.rtp.remote_ssrc =
+            test_->video_send_config_.rtp.ssrcs[0];
+        test_->receive_config_.rtp.local_ssrc++;
+        video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
+            test_->receive_config_);
+        video_receive_stream_->Start();
+      }
+      is_sending_receiving_ = true;
+    }
+
+    ~Stream() {
+      EXPECT_FALSE(is_sending_receiving_);
+      frame_generator_capturer_.reset(nullptr);
+      test_->sender_call_->DestroyVideoSendStream(send_stream_);
+      send_stream_ = nullptr;
+      if (audio_receive_stream_) {
+        test_->receiver_call_->DestroyAudioReceiveStream(audio_receive_stream_);
+        audio_receive_stream_ = nullptr;
+      }
+      if (video_receive_stream_) {
+        test_->receiver_call_->DestroyVideoReceiveStream(video_receive_stream_);
+        video_receive_stream_ = nullptr;
+      }
+    }
+
+    void StopSending() {
+      if (is_sending_receiving_) {
+        frame_generator_capturer_->Stop();
+        send_stream_->Stop();
+        if (video_receive_stream_) {
+          video_receive_stream_->Stop();
+        }
+        is_sending_receiving_ = false;
+      }
+    }
+
+   private:
+    BitrateEstimatorTest* test_;
+    bool is_sending_receiving_;
+    VideoSendStream* send_stream_;
+    AudioReceiveStream* audio_receive_stream_;
+    VideoReceiveStream* video_receive_stream_;
+    rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
+    test::FakeEncoder fake_encoder_;
+    test::FakeDecoder fake_decoder_;
+  };
+
+  testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
+  LogObserver receiver_log_;
+  rtc::scoped_ptr<test::DirectTransport> send_transport_;
+  rtc::scoped_ptr<test::DirectTransport> receive_transport_;
+  rtc::scoped_ptr<Call> sender_call_;
+  rtc::scoped_ptr<Call> receiver_call_;
+  VideoReceiveStream::Config receive_config_;
+  std::vector<Stream*> streams_;
+};
+
+static const char* kAbsSendTimeLog =
+    "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
+static const char* kSingleStreamLog =
+    "RemoteBitrateEstimatorSingleStream: Instantiating.";
+
+TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  streams_.push_back(new Stream(this, false));
+  EXPECT_TRUE(receiver_log_.Wait());
+}
+
+TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+  streams_.push_back(new Stream(this, true));
+  EXPECT_TRUE(receiver_log_.Wait());
+}
+
+TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+  streams_.push_back(new Stream(this, false));
+  EXPECT_TRUE(receiver_log_.Wait());
+}
+
+TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  streams_.push_back(new Stream(this, true));
+  EXPECT_TRUE(receiver_log_.Wait());
+
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+  streams_.push_back(new Stream(this, true));
+  EXPECT_TRUE(receiver_log_.Wait());
+}
+
+TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  streams_.push_back(new Stream(this, false));
+  EXPECT_TRUE(receiver_log_.Wait());
+
+  video_send_config_.rtp.extensions[0] =
+      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+  streams_.push_back(new Stream(this, false));
+  EXPECT_TRUE(receiver_log_.Wait());
+}
+
+TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  streams_.push_back(new Stream(this, false));
+  EXPECT_TRUE(receiver_log_.Wait());
+
+  video_send_config_.rtp.extensions[0] =
+      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
+  receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+  receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
+  streams_.push_back(new Stream(this, false));
+  EXPECT_TRUE(receiver_log_.Wait());
+
+  video_send_config_.rtp.extensions[0] =
+      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
+  receiver_log_.PushExpectedLogLine(
+      "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
+  receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+  streams_.push_back(new Stream(this, false));
+  streams_[0]->StopSending();
+  streams_[1]->StopSending();
+  EXPECT_TRUE(receiver_log_.Wait());
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/call/call.cc b/media/webrtc/trunk/webrtc/call/call.cc
new file mode 100644
index 0000000000..44e4782f41
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/call.cc
@@ -0,0 +1,747 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include <map>
+#include <vector>
+
+#include "webrtc/audio/audio_receive_stream.h"
+#include "webrtc/audio/audio_send_stream.h"
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/call.h"
+#include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/common.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/cpu_info.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/video_receive_stream.h"
+#include "webrtc/video/video_send_stream.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+
+namespace webrtc {
+
+const int Call::Config::kDefaultStartBitrateBps = 300000;
+
+namespace internal {
+
+class Call : public webrtc::Call, public PacketReceiver,
+             public BitrateObserver {
+ public:
+  explicit Call(const Call::Config& config);
+  virtual ~Call();
+
+  PacketReceiver* Receiver() override;
+
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
+  webrtc::AudioReceiveStream* CreateAudioReceiveStream(
+      const webrtc::AudioReceiveStream::Config& config) override;
+  void DestroyAudioReceiveStream(
+      webrtc::AudioReceiveStream* receive_stream) override;
+
+  webrtc::VideoSendStream* CreateVideoSendStream(
+      const webrtc::VideoSendStream::Config& config,
+      const VideoEncoderConfig& encoder_config) override;
+  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
+
+  webrtc::VideoReceiveStream* CreateVideoReceiveStream(
+      const webrtc::VideoReceiveStream::Config& config) override;
+  void DestroyVideoReceiveStream(
+      webrtc::VideoReceiveStream* receive_stream) override;
+
+  Stats GetStats() const override;
+
+  DeliveryStatus DeliverPacket(MediaType media_type,
+                               const uint8_t* packet,
+                               size_t length,
+                               const PacketTime& packet_time) override;
+
+  void SetBitrateConfig(
+      const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+  void SignalNetworkState(NetworkState state) override;
+
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+
+  // Implements BitrateObserver.
+  void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
+                        int64_t rtt_ms) override;
+
+  VoiceEngine* voice_engine() {
+    internal::AudioState* audio_state =
+        static_cast<internal::AudioState*>(config_.audio_state.get());
+    if (audio_state)
+      return audio_state->voice_engine();
+    else
+      return nullptr;
+  };
+
+ private:
+  DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
+                             size_t length);
+  DeliveryStatus DeliverRtp(MediaType media_type,
+                            const uint8_t* packet,
+                            size_t length,
+                            const PacketTime& packet_time);
+
+  void ConfigureSync(const std::string& sync_group)
+      EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
+
+
+  void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
+  void UpdateReceiveHistograms();
+
+  Clock* const clock_;
+
+  const int num_cpu_cores_;
+  const rtc::scoped_ptr<ProcessThread> module_process_thread_;
+  const rtc::scoped_ptr<CallStats> call_stats_;
+  const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
+  Call::Config config_;
+  rtc::ThreadChecker configuration_thread_checker_;
+
+  bool network_enabled_;
+
+  rtc::scoped_ptr<RWLockWrapper> receive_crit_;
+  // Audio and Video receive streams are owned by the client that creates them.
+  std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
+      GUARDED_BY(receive_crit_);
+  std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
+      GUARDED_BY(receive_crit_);
+  std::set<VideoReceiveStream*> video_receive_streams_
+      GUARDED_BY(receive_crit_);
+  std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
+      GUARDED_BY(receive_crit_);
+
+  rtc::scoped_ptr<RWLockWrapper> send_crit_;
+  // Audio and Video send streams are owned by the client that creates them.
+  std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
+  std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
+  std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
+
+  VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
+
+  RtcEventLog* event_log_ = nullptr;
+
+  // The following members are only accessed (exclusively) from one thread and
+  // from the destructor, and therefore doesn't need any explicit
+  // synchronization.
+  int64_t received_video_bytes_;
+  int64_t received_audio_bytes_;
+  int64_t received_rtcp_bytes_;
+  int64_t first_rtp_packet_received_ms_;
+  int64_t last_rtp_packet_received_ms_;
+  int64_t first_packet_sent_ms_;
+
+  // TODO(holmer): Remove this lock once BitrateController no longer calls
+  // OnNetworkChanged from multiple threads.
+  rtc::CriticalSection bitrate_crit_;
+  int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+  int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+  int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
+
+  const rtc::scoped_ptr<CongestionController> congestion_controller_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Call);
+};
+}  // namespace internal
+
+Call* Call::Create(const Call::Config& config) {
+  return new internal::Call(config);
+}
+
+namespace internal {
+
+Call::Call(const Call::Config& config)
+    : clock_(Clock::GetRealTimeClock()),
+      num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
+      module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
+      call_stats_(new CallStats(clock_)),
+      bitrate_allocator_(new BitrateAllocator()),
+      config_(config),
+      network_enabled_(true),
+      receive_crit_(RWLockWrapper::CreateRWLock()),
+      send_crit_(RWLockWrapper::CreateRWLock()),
+      received_video_bytes_(0),
+      received_audio_bytes_(0),
+      received_rtcp_bytes_(0),
+      first_rtp_packet_received_ms_(-1),
+      last_rtp_packet_received_ms_(-1),
+      first_packet_sent_ms_(-1),
+      estimated_send_bitrate_sum_kbits_(0),
+      pacer_bitrate_sum_kbits_(0),
+      num_bitrate_updates_(0),
+      congestion_controller_(
+          new CongestionController(module_process_thread_.get(),
+                                   call_stats_.get(),
+                                   this)) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
+  RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
+                config.bitrate_config.min_bitrate_bps);
+  if (config.bitrate_config.max_bitrate_bps != -1) {
+    RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
+                  config.bitrate_config.start_bitrate_bps);
+  }
+  if (config.audio_state.get()) {
+    ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
+    event_log_ = voe_codec->GetEventLog();
+  }
+
+  Trace::CreateTrace();
+  module_process_thread_->Start();
+  module_process_thread_->RegisterModule(call_stats_.get());
+
+  congestion_controller_->SetBweBitrates(
+      config_.bitrate_config.min_bitrate_bps,
+      config_.bitrate_config.start_bitrate_bps,
+      config_.bitrate_config.max_bitrate_bps);
+
+  congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
+}
+
+Call::~Call() {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  UpdateSendHistograms();
+  UpdateReceiveHistograms();
+  RTC_CHECK(audio_send_ssrcs_.empty());
+  RTC_CHECK(video_send_ssrcs_.empty());
+  RTC_CHECK(video_send_streams_.empty());
+  RTC_CHECK(audio_receive_ssrcs_.empty());
+  RTC_CHECK(video_receive_ssrcs_.empty());
+  RTC_CHECK(video_receive_streams_.empty());
+
+  module_process_thread_->DeRegisterModule(call_stats_.get());
+  module_process_thread_->Stop();
+  Trace::ReturnTrace();
+}
+
+void Call::UpdateSendHistograms() {
+  if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
+    return;
+  int64_t elapsed_sec =
+      (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+    return;
+  int send_bitrate_kbps =
+      estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
+  int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
+  if (send_bitrate_kbps > 0) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+                                       send_bitrate_kbps);
+  }
+  if (pacer_bitrate_kbps > 0) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.PacerBitrateInKbps",
+                                       pacer_bitrate_kbps);
+  }
+}
+
+void Call::UpdateReceiveHistograms() {
+  if (first_rtp_packet_received_ms_ == -1)
+    return;
+  int64_t elapsed_sec =
+      (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+    return;
+  int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000;
+  int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000;
+  int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec;
+  if (video_bitrate_kbps > 0) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+                                       video_bitrate_kbps);
+  }
+  if (audio_bitrate_kbps > 0) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+                                       audio_bitrate_kbps);
+  }
+  if (rtcp_bitrate_bps > 0) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+                                       rtcp_bitrate_bps);
+  }
+  RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+      "WebRTC.Call.BitrateReceivedInKbps",
+      audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
+}
+
+PacketReceiver* Call::Receiver() {
+  // TODO(solenberg): Some test cases in EndToEndTest use this from a different
+  // thread. Re-enable once that is fixed.
+  // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  return this;
+}
+
+webrtc::AudioSendStream* Call::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  AudioSendStream* send_stream = new AudioSendStream(
+      config, config_.audio_state, congestion_controller_.get());
+  if (!network_enabled_)
+    send_stream->SignalNetworkState(kNetworkDown);
+  {
+    WriteLockScoped write_lock(*send_crit_);
+    RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
+               audio_send_ssrcs_.end());
+    audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
+  }
+  return send_stream;
+}
+
+void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(send_stream != nullptr);
+
+  send_stream->Stop();
+
+  webrtc::internal::AudioSendStream* audio_send_stream =
+      static_cast<webrtc::internal::AudioSendStream*>(send_stream);
+  {
+    WriteLockScoped write_lock(*send_crit_);
+    size_t num_deleted = audio_send_ssrcs_.erase(
+        audio_send_stream->config().rtp.ssrc);
+    RTC_DCHECK(num_deleted == 1);
+  }
+  delete audio_send_stream;
+}
+
+webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
+    const webrtc::AudioReceiveStream::Config& config) {
+  TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  AudioReceiveStream* receive_stream = new AudioReceiveStream(
+      congestion_controller_.get(), config, config_.audio_state);
+  {
+    WriteLockScoped write_lock(*receive_crit_);
+    RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
+               audio_receive_ssrcs_.end());
+    audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
+    ConfigureSync(config.sync_group);
+  }
+  return receive_stream;
+}
+
+void Call::DestroyAudioReceiveStream(
+    webrtc::AudioReceiveStream* receive_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(receive_stream != nullptr);
+  webrtc::internal::AudioReceiveStream* audio_receive_stream =
+      static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
+  {
+    WriteLockScoped write_lock(*receive_crit_);
+    size_t num_deleted = audio_receive_ssrcs_.erase(
+        audio_receive_stream->config().rtp.remote_ssrc);
+    RTC_DCHECK(num_deleted == 1);
+    const std::string& sync_group = audio_receive_stream->config().sync_group;
+    const auto it = sync_stream_mapping_.find(sync_group);
+    if (it != sync_stream_mapping_.end() &&
+        it->second == audio_receive_stream) {
+      sync_stream_mapping_.erase(it);
+      ConfigureSync(sync_group);
+    }
+  }
+  delete audio_receive_stream;
+}
+
+webrtc::VideoSendStream* Call::CreateVideoSendStream(
+    const webrtc::VideoSendStream::Config& config,
+    const VideoEncoderConfig& encoder_config) {
+  TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+
+  // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
+  // the call has already started.
+  VideoSendStream* send_stream = new VideoSendStream(
+      num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
+      congestion_controller_.get(), bitrate_allocator_.get(), config,
+      encoder_config, suspended_video_send_ssrcs_);
+
+  if (!network_enabled_)
+    send_stream->SignalNetworkState(kNetworkDown);
+
+  WriteLockScoped write_lock(*send_crit_);
+  for (uint32_t ssrc : config.rtp.ssrcs) {
+    RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
+    video_send_ssrcs_[ssrc] = send_stream;
+  }
+  video_send_streams_.insert(send_stream);
+
+  if (event_log_)
+    event_log_->LogVideoSendStreamConfig(config);
+
+  return send_stream;
+}
+
+void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
+  RTC_DCHECK(send_stream != nullptr);
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+
+  send_stream->Stop();
+
+  VideoSendStream* send_stream_impl = nullptr;
+  {
+    WriteLockScoped write_lock(*send_crit_);
+    auto it = video_send_ssrcs_.begin();
+    while (it != video_send_ssrcs_.end()) {
+      if (it->second == static_cast<VideoSendStream*>(send_stream)) {
+        send_stream_impl = it->second;
+        video_send_ssrcs_.erase(it++);
+      } else {
+        ++it;
+      }
+    }
+    video_send_streams_.erase(send_stream_impl);
+  }
+  RTC_CHECK(send_stream_impl != nullptr);
+
+  VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates();
+
+  for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
+       it != rtp_state.end();
+       ++it) {
+    suspended_video_send_ssrcs_[it->first] = it->second;
+  }
+
+  delete send_stream_impl;
+}
+
+webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
+    const webrtc::VideoReceiveStream::Config& config) {
+  TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  VideoReceiveStream* receive_stream = new VideoReceiveStream(
+      num_cpu_cores_, congestion_controller_.get(), config,
+      voice_engine(), module_process_thread_.get(), call_stats_.get());
+
+  WriteLockScoped write_lock(*receive_crit_);
+  RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
+             video_receive_ssrcs_.end());
+  video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
+  // TODO(pbos): Configure different RTX payloads per receive payload.
+  VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
+      config.rtp.rtx.begin();
+  if (it != config.rtp.rtx.end())
+    video_receive_ssrcs_[it->second.ssrc] = receive_stream;
+  video_receive_streams_.insert(receive_stream);
+
+  ConfigureSync(config.sync_group);
+
+  if (!network_enabled_)
+    receive_stream->SignalNetworkState(kNetworkDown);
+
+  if (event_log_)
+    event_log_->LogVideoReceiveStreamConfig(config);
+
+  return receive_stream;
+}
+
+void Call::DestroyVideoReceiveStream(
+    webrtc::VideoReceiveStream* receive_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(receive_stream != nullptr);
+  VideoReceiveStream* receive_stream_impl = nullptr;
+  {
+    WriteLockScoped write_lock(*receive_crit_);
+    // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
+    // separate SSRC there can be either one or two.
+    auto it = video_receive_ssrcs_.begin();
+    while (it != video_receive_ssrcs_.end()) {
+      if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
+        if (receive_stream_impl != nullptr)
+          RTC_DCHECK(receive_stream_impl == it->second);
+        receive_stream_impl = it->second;
+        video_receive_ssrcs_.erase(it++);
+      } else {
+        ++it;
+      }
+    }
+    video_receive_streams_.erase(receive_stream_impl);
+    RTC_CHECK(receive_stream_impl != nullptr);
+    ConfigureSync(receive_stream_impl->config().sync_group);
+  }
+  delete receive_stream_impl;
+}
+
+Call::Stats Call::GetStats() const {
+  // TODO(solenberg): Some test cases in EndToEndTest use this from a different
+  // thread. Re-enable once that is fixed.
+  // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  Stats stats;
+  // Fetch available send/receive bitrates.
+  uint32_t send_bandwidth = 0;
+  congestion_controller_->GetBitrateController()->AvailableBandwidth(
+      &send_bandwidth);
+  std::vector<unsigned int> ssrcs;
+  uint32_t recv_bandwidth = 0;
+  congestion_controller_->GetRemoteBitrateEstimator(false)->LatestEstimate(
+      &ssrcs, &recv_bandwidth);
+  stats.send_bandwidth_bps = send_bandwidth;
+  stats.recv_bandwidth_bps = recv_bandwidth;
+  stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();
+  {
+    ReadLockScoped read_lock(*send_crit_);
+    // TODO(solenberg): Add audio send streams.
+    for (const auto& kv : video_send_ssrcs_) {
+      int rtt_ms = kv.second->GetRtt();
+      if (rtt_ms > 0)
+        stats.rtt_ms = rtt_ms;
+    }
+  }
+  return stats;
+}
+
+void Call::SetBitrateConfig(
+    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
+  TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
+  if (bitrate_config.max_bitrate_bps != -1)
+    RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
+  if (config_.bitrate_config.min_bitrate_bps ==
+          bitrate_config.min_bitrate_bps &&
+      (bitrate_config.start_bitrate_bps <= 0 ||
+       config_.bitrate_config.start_bitrate_bps ==
+           bitrate_config.start_bitrate_bps) &&
+      config_.bitrate_config.max_bitrate_bps ==
+          bitrate_config.max_bitrate_bps) {
+    // Nothing new to set, early abort to avoid encoder reconfigurations.
+    return;
+  }
+  config_.bitrate_config = bitrate_config;
+  congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
+                                         bitrate_config.start_bitrate_bps,
+                                         bitrate_config.max_bitrate_bps);
+}
+
+void Call::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  network_enabled_ = state == kNetworkUp;
+  congestion_controller_->SignalNetworkState(state);
+  {
+    ReadLockScoped write_lock(*send_crit_);
+    for (auto& kv : audio_send_ssrcs_) {
+      kv.second->SignalNetworkState(state);
+    }
+    for (auto& kv : video_send_ssrcs_) {
+      kv.second->SignalNetworkState(state);
+    }
+  }
+  {
+    ReadLockScoped write_lock(*receive_crit_);
+    for (auto& kv : video_receive_ssrcs_) {
+      kv.second->SignalNetworkState(state);
+    }
+  }
+}
+
+void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  if (first_packet_sent_ms_ == -1)
+    first_packet_sent_ms_ = clock_->TimeInMilliseconds();
+  congestion_controller_->OnSentPacket(sent_packet);
+}
+
+void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
+                            int64_t rtt_ms) {
+  uint32_t allocated_bitrate_bps = bitrate_allocator_->OnNetworkChanged(
+      target_bitrate_bps, fraction_loss, rtt_ms);
+
+  int pad_up_to_bitrate_bps = 0;
+  {
+    ReadLockScoped read_lock(*send_crit_);
+    // No need to update as long as we're not sending.
+    if (video_send_streams_.empty())
+      return;
+
+    for (VideoSendStream* stream : video_send_streams_)
+      pad_up_to_bitrate_bps += stream->GetPaddingNeededBps();
+  }
+  // Allocated bitrate might be higher than bitrate estimate if enforcing min
+  // bitrate, or lower if estimate is higher than the sum of max bitrates, so
+  // set the pacer bitrate to the maximum of the two.
+  uint32_t pacer_bitrate_bps =
+      std::max(target_bitrate_bps, allocated_bitrate_bps);
+  {
+    rtc::CritScope lock(&bitrate_crit_);
+    // We only update these stats if we have send streams, and assume that
+    // OnNetworkChanged is called roughly with a fixed frequency.
+    estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
+    pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
+    ++num_bitrate_updates_;
+  }
+  congestion_controller_->UpdatePacerBitrate(
+      target_bitrate_bps / 1000,
+      PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
+      pad_up_to_bitrate_bps / 1000);
+}
+
+void Call::ConfigureSync(const std::string& sync_group) {
+  // Set sync only if there was no previous one.
+  if (voice_engine() == nullptr || sync_group.empty())
+    return;
+
+  AudioReceiveStream* sync_audio_stream = nullptr;
+  // Find existing audio stream.
+  const auto it = sync_stream_mapping_.find(sync_group);
+  if (it != sync_stream_mapping_.end()) {
+    sync_audio_stream = it->second;
+  } else {
+    // No configured audio stream, see if we can find one.
+    for (const auto& kv : audio_receive_ssrcs_) {
+      if (kv.second->config().sync_group == sync_group) {
+        if (sync_audio_stream != nullptr) {
+          LOG(LS_WARNING) << "Attempting to sync more than one audio stream "
+                             "within the same sync group. This is not "
+                             "supported in the current implementation.";
+          break;
+        }
+        sync_audio_stream = kv.second;
+      }
+    }
+  }
+  if (sync_audio_stream)
+    sync_stream_mapping_[sync_group] = sync_audio_stream;
+  size_t num_synced_streams = 0;
+  for (VideoReceiveStream* video_stream : video_receive_streams_) {
+    if (video_stream->config().sync_group != sync_group)
+      continue;
+    ++num_synced_streams;
+    if (num_synced_streams > 1) {
+      // TODO(pbos): Support synchronizing more than one A/V pair.
+      // https://code.google.com/p/webrtc/issues/detail?id=4762
+      LOG(LS_WARNING) << "Attempting to sync more than one audio/video pair "
+                         "within the same sync group. This is not supported in "
+                         "the current implementation.";
+    }
+    // Only sync the first A/V pair within this sync group.
+    if (sync_audio_stream != nullptr && num_synced_streams == 1) {
+      video_stream->SetSyncChannel(voice_engine(),
+                                   sync_audio_stream->config().voe_channel_id);
+    } else {
+      video_stream->SetSyncChannel(voice_engine(), -1);
+    }
+  }
+}
+
+PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
+                                                 const uint8_t* packet,
+                                                 size_t length) {
+  TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
+  // TODO(pbos): Figure out what channel needs it actually.
+  //             Do NOT broadcast! Also make sure it's a valid packet.
+  //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
+  //             there's no receiver of the packet.
+  received_rtcp_bytes_ += length;
+  bool rtcp_delivered = false;
+  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+    ReadLockScoped read_lock(*receive_crit_);
+    for (VideoReceiveStream* stream : video_receive_streams_) {
+      if (stream->DeliverRtcp(packet, length)) {
+        rtcp_delivered = true;
+        if (event_log_)
+          event_log_->LogRtcpPacket(true, media_type, packet, length);
+      }
+    }
+  }
+  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+    ReadLockScoped read_lock(*send_crit_);
+    for (VideoSendStream* stream : video_send_streams_) {
+      if (stream->DeliverRtcp(packet, length)) {
+        rtcp_delivered = true;
+        if (event_log_)
+          event_log_->LogRtcpPacket(false, media_type, packet, length);
+      }
+    }
+  }
+  return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
+}
+
+PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
+                                                const uint8_t* packet,
+                                                size_t length,
+                                                const PacketTime& packet_time) {
+  TRACE_EVENT0("webrtc", "Call::DeliverRtp");
+  // Minimum RTP header size.
+  if (length < 12)
+    return DELIVERY_PACKET_ERROR;
+
+  last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
+  if (first_rtp_packet_received_ms_ == -1)
+    first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
+
+  uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
+  ReadLockScoped read_lock(*receive_crit_);
+  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+    auto it = audio_receive_ssrcs_.find(ssrc);
+    if (it != audio_receive_ssrcs_.end()) {
+      received_audio_bytes_ += length;
+      auto status = it->second->DeliverRtp(packet, length, packet_time)
+                        ? DELIVERY_OK
+                        : DELIVERY_PACKET_ERROR;
+      if (status == DELIVERY_OK && event_log_)
+        event_log_->LogRtpHeader(true, media_type, packet, length);
+      return status;
+    }
+  }
+  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+    auto it = video_receive_ssrcs_.find(ssrc);
+    if (it != video_receive_ssrcs_.end()) {
+      received_video_bytes_ += length;
+      auto status = it->second->DeliverRtp(packet, length, packet_time)
+                        ? DELIVERY_OK
+                        : DELIVERY_PACKET_ERROR;
+      if (status == DELIVERY_OK && event_log_)
+        event_log_->LogRtpHeader(true, media_type, packet, length);
+      return status;
+    }
+  }
+  LOG(LS_WARNING) << __FUNCTION__ <<": found unknown SSRC: " << ssrc;
+  return DELIVERY_UNKNOWN_SSRC;
+}
+
+PacketReceiver::DeliveryStatus Call::DeliverPacket(
+    MediaType media_type,
+    const uint8_t* packet,
+    size_t length,
+    const PacketTime& packet_time) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
+  if (RtpHeaderParser::IsRtcp(packet, length))
+    return DeliverRtcp(media_type, packet, length);
+
+  return DeliverRtp(media_type, packet, length, packet_time);
+}
+
+}  // namespace internal
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/call_perf_tests.cc b/media/webrtc/trunk/webrtc/call/call_perf_tests.cc
similarity index 67%
rename from media/webrtc/trunk/webrtc/video/call_perf_tests.cc
rename to media/webrtc/trunk/webrtc/call/call_perf_tests.cc
index 182a83edce..3adcb10b09 100644
--- a/media/webrtc/trunk/webrtc/video/call_perf_tests.cc
+++ b/media/webrtc/trunk/webrtc/call/call_perf_tests.cc
@@ -17,11 +17,14 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/call/transport_adapter.h"
+#include "webrtc/common.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
@@ -33,7 +36,6 @@
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video/transport_adapter.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_network.h"
@@ -44,7 +46,7 @@ namespace webrtc {
 
 class CallPerfTest : public test::CallTest {
  protected:
-  void TestAudioVideoSync(bool fec);
+  void TestAudioVideoSync(bool fec, bool create_audio_first);
 
   void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
 
@@ -58,18 +60,16 @@ class CallPerfTest : public test::CallTest {
 
 class SyncRtcpObserver : public test::RtpRtcpObserver {
  public:
-  explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config)
-      : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs, config),
-        crit_(CriticalSectionWrapper::CreateCriticalSection()) {}
+  SyncRtcpObserver() : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs) {}
 
   Action OnSendRtcp(const uint8_t* packet, size_t length) override {
     RTCPUtility::RTCPParserV2 parser(packet, length, true);
     EXPECT_TRUE(parser.IsValid());
 
     for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-         packet_type != RTCPUtility::kRtcpNotValidCode;
+         packet_type != RTCPUtility::RTCPPacketTypes::kInvalid;
          packet_type = parser.Iterate()) {
-      if (packet_type == RTCPUtility::kRtcpSrCode) {
+      if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
         const RTCPUtility::RTCPPacket& packet = parser.Packet();
         RtcpMeasurement ntp_rtp_pair(
             packet.SR.NTPMostSignificant,
@@ -82,7 +82,7 @@ class SyncRtcpObserver : public test::RtpRtcpObserver {
   }
 
   int64_t RtpTimestampToNtp(uint32_t timestamp) const {
-    CriticalSectionScoped lock(crit_.get());
+    rtc::CritScope lock(&crit_);
     int64_t timestamp_in_ms = -1;
     if (ntp_rtp_pairs_.size() == 2) {
       // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
@@ -96,7 +96,7 @@ class SyncRtcpObserver : public test::RtpRtcpObserver {
 
  private:
   void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) {
-    CriticalSectionScoped lock(crit_.get());
+    rtc::CritScope lock(&crit_);
     for (RtcpList::iterator it = ntp_rtp_pairs_.begin();
          it != ntp_rtp_pairs_.end();
          ++it) {
@@ -114,7 +114,7 @@ class SyncRtcpObserver : public test::RtpRtcpObserver {
     ntp_rtp_pairs_.push_front(ntp_rtp_pair);
   }
 
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
   RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
 };
 
@@ -128,15 +128,14 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
                            int voe_channel,
                            VoEVideoSync* voe_sync,
                            SyncRtcpObserver* audio_observer)
-      : SyncRtcpObserver(FakeNetworkPipe::Config()),
-        clock_(clock),
+      : clock_(clock),
         voe_channel_(voe_channel),
         voe_sync_(voe_sync),
         audio_observer_(audio_observer),
         creation_time_ms_(clock_->TimeInMilliseconds()),
         first_time_in_sync_(-1) {}
 
-  void RenderFrame(const I420VideoFrame& video_frame,
+  void RenderFrame(const VideoFrame& video_frame,
                    int time_to_render_ms) override {
     int64_t now_ms = clock_->TimeInMilliseconds();
     uint32_t playout_timestamp = 0;
@@ -175,7 +174,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
                                   false);
       }
       if (time_since_creation > kMinRunTimeMs)
-        observation_complete_->Set();
+        observation_complete_.Set();
     }
   }
 
@@ -183,22 +182,29 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
 
  private:
   Clock* const clock_;
-  int voe_channel_;
-  VoEVideoSync* voe_sync_;
-  SyncRtcpObserver* audio_observer_;
-  int64_t creation_time_ms_;
+  const int voe_channel_;
+  VoEVideoSync* const voe_sync_;
+  SyncRtcpObserver* const audio_observer_;
+  const int64_t creation_time_ms_;
   int64_t first_time_in_sync_;
 };
 
-void CallPerfTest::TestAudioVideoSync(bool fec) {
+void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
+  const char* kSyncGroup = "av_sync";
+  const uint32_t kAudioSendSsrc = 1234;
+  const uint32_t kAudioRecvSsrc = 5678;
   class AudioPacketReceiver : public PacketReceiver {
    public:
     AudioPacketReceiver(int channel, VoENetwork* voe_network)
         : channel_(channel),
           voe_network_(voe_network),
           parser_(RtpHeaderParser::Create()) {}
-    DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                 size_t length) override {
+    DeliveryStatus DeliverPacket(MediaType media_type,
+                                 const uint8_t* packet,
+                                 size_t length,
+                                 const PacketTime& packet_time) override {
+      EXPECT_TRUE(media_type == MediaType::ANY ||
+                  media_type == MediaType::AUDIO);
       int ret;
       if (parser_->IsRtcp(packet, length)) {
         ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
@@ -226,90 +232,149 @@ void CallPerfTest::TestAudioVideoSync(bool fec) {
   test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
                                           audio_filename);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
-  int channel = voe_base->CreateChannel();
+  Config voe_config;
+  voe_config.Set<VoicePacing>(new VoicePacing(true));
+  int send_channel_id = voe_base->CreateChannel(voe_config);
+  int recv_channel_id = voe_base->CreateChannel();
+
+  SyncRtcpObserver audio_observer;
+
+  AudioState::Config send_audio_state_config;
+  send_audio_state_config.voice_engine = voice_engine;
+  Call::Config sender_config;
+  sender_config.audio_state = AudioState::Create(send_audio_state_config);
+  Call::Config receiver_config;
+  receiver_config.audio_state = sender_config.audio_state;
+  CreateCalls(sender_config, receiver_config);
+
+  AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
+  AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
 
   FakeNetworkPipe::Config net_config;
   net_config.queue_delay_ms = 500;
   net_config.loss_percent = 5;
-  SyncRtcpObserver audio_observer(net_config);
-  VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
-                                    channel,
-                                    voe_sync,
-                                    &audio_observer);
+  test::PacketTransport audio_send_transport(
+      nullptr, &audio_observer, test::PacketTransport::kSender, net_config);
+  audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
+  test::PacketTransport audio_receive_transport(
+      nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config);
+  audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
 
-  Call::Config receiver_config(observer.ReceiveTransport());
-  receiver_config.voice_engine = voice_engine;
-  CreateCalls(Call::Config(observer.SendTransport()), receiver_config);
+  internal::TransportAdapter send_transport_adapter(&audio_send_transport);
+  send_transport_adapter.Enable();
+  EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
+                                                      send_transport_adapter));
 
-  CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
-  EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
+  internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
+  recv_transport_adapter.Enable();
+  EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
+                                                      recv_transport_adapter));
 
-  AudioPacketReceiver voe_packet_receiver(channel, voe_network);
-  audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver);
+  VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), recv_channel_id,
+                                    voe_sync, &audio_observer);
 
-  internal::TransportAdapter transport_adapter(audio_observer.SendTransport());
-  transport_adapter.Enable();
-  EXPECT_EQ(0,
-            voe_network->RegisterExternalTransport(channel, transport_adapter));
-
-  observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
+  test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
+                                            test::PacketTransport::kSender,
+                                            FakeNetworkPipe::Config());
+  sync_send_transport.SetReceiver(receiver_call_->Receiver());
+  test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,
+                                               test::PacketTransport::kReceiver,
+                                               FakeNetworkPipe::Config());
+  sync_receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateSendConfig(1, 0, &sync_send_transport);
+  CreateMatchingReceiveConfigs(&sync_receive_transport);
 
-  send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+  AudioSendStream::Config audio_send_config(&audio_send_transport);
+  audio_send_config.voe_channel_id = send_channel_id;
+  audio_send_config.rtp.ssrc = kAudioSendSsrc;
+  AudioSendStream* audio_send_stream =
+      sender_call_->CreateAudioSendStream(audio_send_config);
+
+  CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
+  EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
+
+  video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   if (fec) {
-    send_config_.rtp.fec.red_payload_type = kRedPayloadType;
-    send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
-    receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
-    receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+    video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
+    video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+    video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
+    video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
   }
-  receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
-  receive_configs_[0].renderer = &observer;
-  receive_configs_[0].audio_channel_id = channel;
+  video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
+  video_receive_configs_[0].renderer = &observer;
+  video_receive_configs_[0].sync_group = kSyncGroup;
 
-  CreateStreams();
+  AudioReceiveStream::Config audio_recv_config;
+  audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
+  audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
+  audio_recv_config.voe_channel_id = recv_channel_id;
+  audio_recv_config.sync_group = kSyncGroup;
+
+  AudioReceiveStream* audio_receive_stream;
+
+  if (create_audio_first) {
+    audio_receive_stream =
+        receiver_call_->CreateAudioReceiveStream(audio_recv_config);
+    CreateVideoStreams();
+  } else {
+    CreateVideoStreams();
+    audio_receive_stream =
+        receiver_call_->CreateAudioReceiveStream(audio_recv_config);
+  }
 
   CreateFrameGeneratorCapturer();
 
   Start();
 
   fake_audio_device.Start();
-  EXPECT_EQ(0, voe_base->StartPlayout(channel));
-  EXPECT_EQ(0, voe_base->StartReceive(channel));
-  EXPECT_EQ(0, voe_base->StartSend(channel));
+  EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
+  EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
+  EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
 
-  EXPECT_EQ(kEventSignaled, observer.Wait())
+  EXPECT_TRUE(observer.Wait())
       << "Timed out while waiting for audio and video to be synchronized.";
 
-  EXPECT_EQ(0, voe_base->StopSend(channel));
-  EXPECT_EQ(0, voe_base->StopReceive(channel));
-  EXPECT_EQ(0, voe_base->StopPlayout(channel));
+  EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
+  EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
+  EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
   fake_audio_device.Stop();
 
   Stop();
-  observer.StopSending();
-  audio_observer.StopSending();
+  sync_send_transport.StopSending();
+  sync_receive_transport.StopSending();
+  audio_send_transport.StopSending();
+  audio_receive_transport.StopSending();
 
-  voe_base->DeleteChannel(channel);
+  DestroyStreams();
+
+  sender_call_->DestroyAudioSendStream(audio_send_stream);
+  receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
+
+  voe_base->DeleteChannel(send_channel_id);
+  voe_base->DeleteChannel(recv_channel_id);
   voe_base->Release();
   voe_codec->Release();
   voe_network->Release();
   voe_sync->Release();
 
-  DestroyStreams();
+  DestroyCalls();
 
   VoiceEngine::Delete(voice_engine);
 }
 
-TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
-  TestAudioVideoSync(false);
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
+  TestAudioVideoSync(false, true);
+}
+
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
+  TestAudioVideoSync(false, false);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
-  TestAudioVideoSync(true);
+  TestAudioVideoSync(true, false);
 }
 
 void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
@@ -319,11 +384,12 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
   class CaptureNtpTimeObserver : public test::EndToEndTest,
                                  public VideoRenderer {
    public:
-    CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config,
+    CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
                            int threshold_ms,
                            int start_time_ms,
                            int run_time_ms)
-        : EndToEndTest(kLongTimeoutMs, config),
+        : EndToEndTest(kLongTimeoutMs),
+          net_config_(net_config),
           clock_(Clock::GetRealTimeClock()),
           threshold_ms_(threshold_ms),
           start_time_ms_(start_time_ms),
@@ -334,8 +400,19 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
           rtp_start_timestamp_(0) {}
 
    private:
-    void RenderFrame(const I420VideoFrame& video_frame,
+    test::PacketTransport* CreateSendTransport(Call* sender_call) override {
+      return new test::PacketTransport(
+          sender_call, this, test::PacketTransport::kSender, net_config_);
+    }
+
+    test::PacketTransport* CreateReceiveTransport() override {
+      return new test::PacketTransport(
+          nullptr, this, test::PacketTransport::kReceiver, net_config_);
+    }
+
+    void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
+      rtc::CritScope lock(&crit_);
       if (video_frame.ntp_time_ms() <= 0) {
         // Haven't got enough RTCP SR in order to calculate the capture ntp
         // time.
@@ -350,7 +427,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
       }
 
       if (time_since_creation > run_time_ms_) {
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
 
       FrameCaptureTimeList::iterator iter =
@@ -376,6 +453,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
     bool IsTextureSupported() const override { return false; }
 
     virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -400,21 +478,24 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
       capturer_ = frame_generator_capturer;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       (*receive_configs)[0].renderer = this;
       // Enable the receiver side rtt calculation.
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
-                                           "estimated capture NTP time to be "
-                                           "within bounds.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for "
+                             "estimated capture NTP time to be "
+                             "within bounds.";
     }
 
-    Clock* clock_;
+    rtc::CriticalSection crit_;
+    const FakeNetworkPipe::Config net_config_;
+    Clock* const clock_;
     int threshold_ms_;
     int start_time_ms_;
     int run_time_ms_;
@@ -423,7 +504,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
     bool rtp_start_timestamp_set_;
     uint32_t rtp_start_timestamp_;
     typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
-    FrameCaptureTimeList capture_time_list_;
+    FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
   } test(net_config, threshold_ms, start_time_ms, run_time_ms);
 
   RunBaseTest(&test);
@@ -463,24 +544,19 @@ void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
 
     void OnLoadUpdate(Load load) override {
       if (load == tested_load_)
-        observation_complete_->Set();
+        observation_complete_.Set();
     }
 
-    Call::Config GetSenderCallConfig() override {
-      Call::Config config(SendTransport());
-      config.overuse_callback = this;
-      return config;
-    }
-
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->overuse_callback = this;
       send_config->encoder_settings.encoder = &encoder_;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out before receiving an overuse callback.";
+      EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
     }
 
     LoadObserver::Load tested_load_;
@@ -506,27 +582,21 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
   static const int kMinAcceptableTransmitBitrate = 130;
   static const int kMaxAcceptableTransmitBitrate = 170;
   static const int kNumBitrateObservationsInRange = 100;
-  class BitrateObserver : public test::EndToEndTest, public PacketReceiver {
+  static const int kAcceptableBitrateErrorMargin = 15;  // +- 7
+  class BitrateObserver : public test::EndToEndTest {
    public:
     explicit BitrateObserver(bool using_min_transmit_bitrate)
         : EndToEndTest(kLongTimeoutMs),
           send_stream_(nullptr),
-          send_transport_receiver_(nullptr),
           pad_to_min_bitrate_(using_min_transmit_bitrate),
           num_bitrate_observations_in_range_(0) {}
 
    private:
-    void SetReceivers(PacketReceiver* send_transport_receiver,
-                      PacketReceiver* receive_transport_receiver) override {
-      send_transport_receiver_ = send_transport_receiver;
-      test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
-    }
-
-    DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                 size_t length) override {
+    // TODO(holmer): Run this with a timer instead of once per packet.
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       VideoSendStream::Stats stats = send_stream_->GetStats();
       if (stats.substreams.size() > 0) {
-        DCHECK_EQ(1u, stats.substreams.size());
+        RTC_DCHECK_EQ(1u, stats.substreams.size());
         int bitrate_kbps =
             stats.substreams.begin()->second.total_bitrate_bps / 1000;
         if (bitrate_kbps > 0) {
@@ -545,42 +615,43 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
             }
           } else {
             // Expect bitrate stats to roughly match the max encode bitrate.
-            if (bitrate_kbps > kMaxEncodeBitrateKbps - 5 &&
-                bitrate_kbps < kMaxEncodeBitrateKbps + 5) {
+            if (bitrate_kbps > (kMaxEncodeBitrateKbps -
+                                kAcceptableBitrateErrorMargin / 2) &&
+                bitrate_kbps < (kMaxEncodeBitrateKbps +
+                                kAcceptableBitrateErrorMargin / 2)) {
               ++num_bitrate_observations_in_range_;
             }
           }
           if (num_bitrate_observations_in_range_ ==
               kNumBitrateObservationsInRange)
-            observation_complete_->Set();
+            observation_complete_.Set();
         }
       }
-      return send_transport_receiver_->DeliverPacket(packet, length);
+      return SEND_PACKET;
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       if (pad_to_min_bitrate_) {
         encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
       } else {
-        DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
+        RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
       }
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timeout while waiting for send-bitrate stats.";
+      EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
     }
 
     VideoSendStream* send_stream_;
-    PacketReceiver* send_transport_receiver_;
     const bool pad_to_min_bitrate_;
     int num_bitrate_observations_in_range_;
   } test(pad_to_min_bitrate);
@@ -605,8 +676,10 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
     BitrateObserver()
         : EndToEndTest(kDefaultTimeoutMs),
           FakeEncoder(Clock::GetRealTimeClock()),
-          time_to_reconfigure_(webrtc::EventWrapper::Create()),
-          encoder_inits_(0) {}
+          time_to_reconfigure_(false, false),
+          encoder_inits_(0),
+          last_set_bitrate_(0),
+          send_stream_(nullptr) {}
 
     int32_t InitEncode(const VideoCodec* config,
                        int32_t number_of_cores,
@@ -622,7 +695,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
                     last_set_bitrate_,
                     kPermittedReconfiguredBitrateDiffKbps)
             << "Encoder reconfigured with bitrate too far away from last set.";
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
       return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
     }
@@ -632,7 +705,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
       last_set_bitrate_ = new_target_bitrate_kbps;
       if (encoder_inits_ == 1 &&
           new_target_bitrate_kbps > kReconfigureThresholdKbps) {
-        time_to_reconfigure_->Set();
+        time_to_reconfigure_.Set();
       }
       return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
     }
@@ -643,9 +716,10 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
       return config;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
@@ -654,25 +728,25 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
       encoder_config_ = *encoder_config;
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
     void PerformTest() override {
-      ASSERT_EQ(kEventSignaled, time_to_reconfigure_->Wait(kDefaultTimeoutMs))
+      ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
           << "Timed out before receiving an initial high bitrate.";
       encoder_config_.streams[0].width *= 2;
       encoder_config_.streams[0].height *= 2;
       EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_));
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for a couple of high bitrate estimates "
              "after reconfiguring the send stream.";
     }
 
    private:
-    rtc::scoped_ptr<webrtc::EventWrapper> time_to_reconfigure_;
+    rtc::Event time_to_reconfigure_;
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
diff --git a/media/webrtc/trunk/webrtc/call/call_unittest.cc b/media/webrtc/trunk/webrtc/call/call_unittest.cc
new file mode 100644
index 0000000000..75c8238a5b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/call_unittest.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <list>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio_state.h"
+#include "webrtc/call.h"
+#include "webrtc/test/mock_voice_engine.h"
+
+namespace {
+
+struct CallHelper {
+  CallHelper() {
+    webrtc::AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = &voice_engine_;
+    webrtc::Call::Config config;
+    config.audio_state = webrtc::AudioState::Create(audio_state_config);
+    call_.reset(webrtc::Call::Create(config));
+  }
+
+  webrtc::Call* operator->() { return call_.get(); }
+
+ private:
+  testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
+  rtc::scoped_ptr<webrtc::Call> call_;
+};
+}  // namespace
+
+namespace webrtc {
+
+TEST(CallTest, ConstructDestruct) {
+  CallHelper call;
+}
+
+TEST(CallTest, CreateDestroy_AudioSendStream) {
+  CallHelper call;
+  AudioSendStream::Config config(nullptr);
+  config.rtp.ssrc = 42;
+  config.voe_channel_id = 123;
+  AudioSendStream* stream = call->CreateAudioSendStream(config);
+  EXPECT_NE(stream, nullptr);
+  call->DestroyAudioSendStream(stream);
+}
+
+TEST(CallTest, CreateDestroy_AudioReceiveStream) {
+  CallHelper call;
+  AudioReceiveStream::Config config;
+  config.rtp.remote_ssrc = 42;
+  config.voe_channel_id = 123;
+  AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
+  EXPECT_NE(stream, nullptr);
+  call->DestroyAudioReceiveStream(stream);
+}
+
+TEST(CallTest, CreateDestroy_AudioSendStreams) {
+  CallHelper call;
+  AudioSendStream::Config config(nullptr);
+  config.voe_channel_id = 123;
+  std::list<AudioSendStream*> streams;
+  for (int i = 0; i < 2; ++i) {
+    for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
+      config.rtp.ssrc = ssrc;
+      AudioSendStream* stream = call->CreateAudioSendStream(config);
+      EXPECT_NE(stream, nullptr);
+      if (ssrc & 1) {
+        streams.push_back(stream);
+      } else {
+        streams.push_front(stream);
+      }
+    }
+    for (auto s : streams) {
+      call->DestroyAudioSendStream(s);
+    }
+    streams.clear();
+  }
+}
+
+TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
+  CallHelper call;
+  AudioReceiveStream::Config config;
+  config.voe_channel_id = 123;
+  std::list<AudioReceiveStream*> streams;
+  for (int i = 0; i < 2; ++i) {
+    for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
+      config.rtp.remote_ssrc = ssrc;
+      AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
+      EXPECT_NE(stream, nullptr);
+      if (ssrc & 1) {
+        streams.push_back(stream);
+      } else {
+        streams.push_front(stream);
+      }
+    }
+    for (auto s : streams) {
+      call->DestroyAudioReceiveStream(s);
+    }
+    streams.clear();
+  }
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/call/congestion_controller.cc b/media/webrtc/trunk/webrtc/call/congestion_controller.cc
new file mode 100644
index 0000000000..c442667ae0
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/congestion_controller.cc
@@ -0,0 +1,294 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/congestion_controller.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video/vie_remb.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
+
+namespace webrtc {
+namespace {
+
+static const uint32_t kTimeOffsetSwitchThreshold = 30;
+
+class WrappingBitrateEstimator : public RemoteBitrateEstimator {
+ public:
+  WrappingBitrateEstimator(RemoteBitrateObserver* observer, Clock* clock)
+      : observer_(observer),
+        clock_(clock),
+        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+        rbe_(new RemoteBitrateEstimatorSingleStream(observer_, clock_)),
+        using_absolute_send_time_(false),
+        packets_since_absolute_send_time_(0),
+        min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {}
+
+  virtual ~WrappingBitrateEstimator() {}
+
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header,
+                      bool was_paced) override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    PickEstimatorFromHeader(header);
+    rbe_->IncomingPacket(arrival_time_ms, payload_size, header, was_paced);
+  }
+
+  int32_t Process() override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return rbe_->Process();
+  }
+
+  int64_t TimeUntilNextProcess() override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return rbe_->TimeUntilNextProcess();
+  }
+
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+  }
+
+  void RemoveStream(unsigned int ssrc) override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    rbe_->RemoveStream(ssrc);
+  }
+
+  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
+                      unsigned int* bitrate_bps) const override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return rbe_->LatestEstimate(ssrcs, bitrate_bps);
+  }
+
+  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return rbe_->GetStats(output);
+  }
+
+  void SetMinBitrate(int min_bitrate_bps) {
+    CriticalSectionScoped cs(crit_sect_.get());
+    rbe_->SetMinBitrate(min_bitrate_bps);
+    min_bitrate_bps_ = min_bitrate_bps;
+  }
+
+ private:
+  void PickEstimatorFromHeader(const RTPHeader& header)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
+    if (header.extension.hasAbsoluteSendTime) {
+      // If we see AST in header, switch RBE strategy immediately.
+      if (!using_absolute_send_time_) {
+        LOG(LS_INFO) <<
+            "WrappingBitrateEstimator: Switching to absolute send time RBE.";
+        using_absolute_send_time_ = true;
+        PickEstimator();
+      }
+      packets_since_absolute_send_time_ = 0;
+    } else {
+      // When we don't see AST, wait for a few packets before going back to TOF.
+      if (using_absolute_send_time_) {
+        ++packets_since_absolute_send_time_;
+        if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
+          LOG(LS_INFO) << "WrappingBitrateEstimator: Switching to transmission "
+                       << "time offset RBE.";
+          using_absolute_send_time_ = false;
+          PickEstimator();
+        }
+      }
+    }
+  }
+
+  // Instantiate RBE for Time Offset or Absolute Send Time extensions.
+  void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
+    if (using_absolute_send_time_) {
+      rbe_.reset(new RemoteBitrateEstimatorAbsSendTime(observer_, clock_));
+    } else {
+      rbe_.reset(new RemoteBitrateEstimatorSingleStream(observer_, clock_));
+    }
+    rbe_->SetMinBitrate(min_bitrate_bps_);
+  }
+
+  RemoteBitrateObserver* observer_;
+  Clock* clock_;
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  rtc::scoped_ptr<RemoteBitrateEstimator> rbe_;
+  bool using_absolute_send_time_;
+  uint32_t packets_since_absolute_send_time_;
+  int min_bitrate_bps_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
+};
+
+}  // namespace
+
+CongestionController::CongestionController(ProcessThread* process_thread,
+                                           CallStats* call_stats,
+                                           BitrateObserver* bitrate_observer)
+    : remb_(new VieRemb(Clock::GetRealTimeClock())),
+      packet_router_(new PacketRouter()),
+      pacer_(new PacedSender(Clock::GetRealTimeClock(),
+                             packet_router_.get(),
+                             BitrateController::kDefaultStartBitrateKbps,
+                             PacedSender::kDefaultPaceMultiplier *
+                                 BitrateController::kDefaultStartBitrateKbps,
+                             0)),
+      remote_bitrate_estimator_(
+          new WrappingBitrateEstimator(remb_.get(), Clock::GetRealTimeClock())),
+      remote_estimator_proxy_(
+          new RemoteEstimatorProxy(Clock::GetRealTimeClock(),
+                                   packet_router_.get())),
+      process_thread_(process_thread),
+      call_stats_(call_stats),
+      pacer_thread_(ProcessThread::Create("PacerThread")),
+      // Constructed last as this object calls the provided callback on
+      // construction.
+      bitrate_controller_(
+          BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
+                                                     bitrate_observer)),
+      min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {
+  call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
+
+  pacer_thread_->RegisterModule(pacer_.get());
+  pacer_thread_->Start();
+
+  process_thread->RegisterModule(remote_estimator_proxy_.get());
+  process_thread->RegisterModule(remote_bitrate_estimator_.get());
+  process_thread->RegisterModule(bitrate_controller_.get());
+}
+
+CongestionController::~CongestionController() {
+  pacer_thread_->Stop();
+  pacer_thread_->DeRegisterModule(pacer_.get());
+  process_thread_->DeRegisterModule(bitrate_controller_.get());
+  process_thread_->DeRegisterModule(remote_bitrate_estimator_.get());
+  process_thread_->DeRegisterModule(remote_estimator_proxy_.get());
+  call_stats_->DeregisterStatsObserver(remote_bitrate_estimator_.get());
+  if (transport_feedback_adapter_.get())
+    call_stats_->DeregisterStatsObserver(transport_feedback_adapter_.get());
+  RTC_DCHECK(!remb_->InUse());
+  RTC_DCHECK(encoders_.empty());
+}
+
+void CongestionController::AddEncoder(ViEEncoder* encoder) {
+  rtc::CritScope lock(&encoder_crit_);
+  encoders_.push_back(encoder);
+}
+
+void CongestionController::RemoveEncoder(ViEEncoder* encoder) {
+  rtc::CritScope lock(&encoder_crit_);
+  for (auto it = encoders_.begin(); it != encoders_.end(); ++it) {
+    if (*it == encoder) {
+      encoders_.erase(it);
+      return;
+    }
+  }
+}
+
+void CongestionController::SetBweBitrates(int min_bitrate_bps,
+                                          int start_bitrate_bps,
+                                          int max_bitrate_bps) {
+  if (start_bitrate_bps > 0)
+    bitrate_controller_->SetStartBitrate(start_bitrate_bps);
+  bitrate_controller_->SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
+  if (remote_bitrate_estimator_.get())
+    remote_bitrate_estimator_->SetMinBitrate(min_bitrate_bps);
+  if (transport_feedback_adapter_.get())
+    transport_feedback_adapter_->GetBitrateEstimator()->SetMinBitrate(
+        min_bitrate_bps);
+  min_bitrate_bps_ = min_bitrate_bps;
+}
+
+BitrateController* CongestionController::GetBitrateController() const {
+  return bitrate_controller_.get();
+}
+
+RemoteBitrateEstimator* CongestionController::GetRemoteBitrateEstimator(
+    bool send_side_bwe) const {
+
+  if (send_side_bwe)
+    return remote_estimator_proxy_.get();
+  else
+    return remote_bitrate_estimator_.get();
+}
+
+TransportFeedbackObserver*
+CongestionController::GetTransportFeedbackObserver() {
+  if (transport_feedback_adapter_.get() == nullptr) {
+    transport_feedback_adapter_.reset(new TransportFeedbackAdapter(
+        bitrate_controller_->CreateRtcpBandwidthObserver(),
+        Clock::GetRealTimeClock(), process_thread_));
+    transport_feedback_adapter_->SetBitrateEstimator(
+        new RemoteBitrateEstimatorAbsSendTime(
+            transport_feedback_adapter_.get(), Clock::GetRealTimeClock()));
+    transport_feedback_adapter_->GetBitrateEstimator()->SetMinBitrate(
+        min_bitrate_bps_);
+    call_stats_->RegisterStatsObserver(transport_feedback_adapter_.get());
+  }
+  return transport_feedback_adapter_.get();
+}
+
+void CongestionController::UpdatePacerBitrate(int bitrate_kbps,
+                                              int max_bitrate_kbps,
+                                              int min_bitrate_kbps) {
+  pacer_->UpdateBitrate(bitrate_kbps, max_bitrate_kbps, min_bitrate_kbps);
+}
+
+int64_t CongestionController::GetPacerQueuingDelayMs() const {
+  return pacer_->QueueInMs();
+}
+
+// TODO(mflodman): Move out of this class.
+void CongestionController::SetChannelRembStatus(bool sender,
+                                                bool receiver,
+                                                RtpRtcp* rtp_module) {
+  rtp_module->SetREMBStatus(sender || receiver);
+  if (sender) {
+    remb_->AddRembSender(rtp_module);
+  } else {
+    remb_->RemoveRembSender(rtp_module);
+  }
+  if (receiver) {
+    remb_->AddReceiveChannel(rtp_module);
+  } else {
+    remb_->RemoveReceiveChannel(rtp_module);
+  }
+}
+
+void CongestionController::SignalNetworkState(NetworkState state) {
+  if (state == kNetworkUp) {
+    pacer_->Resume();
+  } else {
+    pacer_->Pause();
+  }
+}
+
+void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  if (transport_feedback_adapter_) {
+    transport_feedback_adapter_->OnSentPacket(sent_packet.packet_id,
+                                              sent_packet.send_time_ms);
+  }
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/call/congestion_controller.h b/media/webrtc/trunk/webrtc/call/congestion_controller.h
new file mode 100644
index 0000000000..b77c46faa3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/congestion_controller.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_CONGESTION_CONTROLLER_H_
+#define WEBRTC_CALL_CONGESTION_CONTROLLER_H_
+
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/socket.h"
+#include "webrtc/stream.h"
+
+namespace webrtc {
+
+class BitrateController;
+class BitrateObserver;
+class CallStats;
+class Config;
+class PacedSender;
+class PacketRouter;
+class ProcessThread;
+class RemoteBitrateEstimator;
+class RemoteEstimatorProxy;
+class RtpRtcp;
+class SendStatisticsProxy;
+class TransportFeedbackAdapter;
+class TransportFeedbackObserver;
+class ViEEncoder;
+class VieRemb;
+
+class CongestionController {
+ public:
+  CongestionController(ProcessThread* process_thread, CallStats* call_stats,
+                       BitrateObserver* bitrate_observer);
+  virtual ~CongestionController();
+  virtual void AddEncoder(ViEEncoder* encoder);
+  virtual void RemoveEncoder(ViEEncoder* encoder);
+  virtual void SetBweBitrates(int min_bitrate_bps,
+                              int start_bitrate_bps,
+                              int max_bitrate_bps);
+
+  virtual void SetChannelRembStatus(bool sender,
+                                    bool receiver,
+                                    RtpRtcp* rtp_module);
+
+  virtual void SignalNetworkState(NetworkState state);
+
+  virtual BitrateController* GetBitrateController() const;
+  virtual RemoteBitrateEstimator* GetRemoteBitrateEstimator(
+      bool send_side_bwe) const;
+  virtual int64_t GetPacerQueuingDelayMs() const;
+  virtual PacedSender* pacer() const { return pacer_.get(); }
+  virtual PacketRouter* packet_router() const { return packet_router_.get(); }
+  virtual TransportFeedbackObserver* GetTransportFeedbackObserver();
+
+  virtual void UpdatePacerBitrate(int bitrate_kbps,
+                                  int max_bitrate_kbps,
+                                  int min_bitrate_kbps);
+
+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet);
+
+ private:
+  rtc::scoped_ptr<VieRemb> remb_;
+  rtc::scoped_ptr<PacketRouter> packet_router_;
+  rtc::scoped_ptr<PacedSender> pacer_;
+  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  rtc::scoped_ptr<RemoteEstimatorProxy> remote_estimator_proxy_;
+
+  mutable rtc::CriticalSection encoder_crit_;
+  std::vector<ViEEncoder*> encoders_ GUARDED_BY(encoder_crit_);
+
+  // Registered at construct time and assumed to outlive this class.
+  ProcessThread* const process_thread_;
+  CallStats* const call_stats_;
+
+  rtc::scoped_ptr<ProcessThread> pacer_thread_;
+
+  rtc::scoped_ptr<BitrateController> bitrate_controller_;
+  rtc::scoped_ptr<TransportFeedbackAdapter> transport_feedback_adapter_;
+  int min_bitrate_bps_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CongestionController);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_CONGESTION_CONTROLLER_H_
diff --git a/media/webrtc/trunk/webrtc/call/mock/mock_congestion_controller.h b/media/webrtc/trunk/webrtc/call/mock/mock_congestion_controller.h
new file mode 100644
index 0000000000..54014da339
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/mock/mock_congestion_controller.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_MOCK_MOCK_CONGESTION_CONTROLLER_H_
+#define WEBRTC_CALL_MOCK_MOCK_CONGESTION_CONTROLLER_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/call/congestion_controller.h"
+
+namespace webrtc {
+namespace test {
+
+class MockCongestionController : public CongestionController {
+ public:
+  MockCongestionController(ProcessThread* process_thread,
+                           CallStats* call_stats,
+                           BitrateObserver* bitrate_observer)
+      : CongestionController(process_thread, call_stats, bitrate_observer) {}
+  MOCK_METHOD1(AddEncoder, void(ViEEncoder* encoder));
+  MOCK_METHOD1(RemoveEncoder, void(ViEEncoder* encoder));
+  MOCK_METHOD3(SetBweBitrates,
+               void(int min_bitrate_bps,
+                    int start_bitrate_bps,
+                    int max_bitrate_bps));
+  MOCK_METHOD3(SetChannelRembStatus,
+               void(bool sender, bool receiver, RtpRtcp* rtp_module));
+  MOCK_METHOD1(SignalNetworkState, void(NetworkState state));
+  MOCK_CONST_METHOD0(GetBitrateController, BitrateController*());
+  MOCK_CONST_METHOD1(GetRemoteBitrateEstimator,
+                     RemoteBitrateEstimator*(bool send_side_bwe));
+  MOCK_CONST_METHOD0(GetPacerQueuingDelayMs, int64_t());
+  MOCK_CONST_METHOD0(pacer, PacedSender*());
+  MOCK_CONST_METHOD0(packet_router, PacketRouter*());
+  MOCK_METHOD0(GetTransportFeedbackObserver, TransportFeedbackObserver*());
+  MOCK_METHOD3(UpdatePacerBitrate,
+               void(int bitrate_kbps,
+                    int max_bitrate_kbps,
+                    int min_bitrate_kbps));
+  MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket& sent_packet));
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MockCongestionController);
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_CALL_MOCK_MOCK_CONGESTION_CONTROLLER_H_
diff --git a/media/webrtc/trunk/webrtc/call/packet_injection_tests.cc b/media/webrtc/trunk/webrtc/call/packet_injection_tests.cc
new file mode 100644
index 0000000000..277cd3e4df
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/packet_injection_tests.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/test/call_test.h"
+#include "webrtc/test/null_transport.h"
+
+namespace webrtc {
+
+class PacketInjectionTest : public test::CallTest {
+ protected:
+  enum class CodecType {
+    kVp8,
+    kH264,
+  };
+
+  PacketInjectionTest() : rtp_header_parser_(RtpHeaderParser::Create()) {}
+
+  void InjectIncorrectPacket(CodecType codec_type,
+                             uint8_t packet_type,
+                             const uint8_t* packet,
+                             size_t length);
+
+  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+};
+
+void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,
+                                                uint8_t payload_type,
+                                                const uint8_t* packet,
+                                                size_t length) {
+  CreateSenderCall(Call::Config());
+  CreateReceiverCall(Call::Config());
+
+  test::NullTransport null_transport;
+  CreateSendConfig(1, 0, &null_transport);
+  CreateMatchingReceiveConfigs(&null_transport);
+  video_receive_configs_[0].decoders[0].payload_type = payload_type;
+  switch (codec_type) {
+    case CodecType::kVp8:
+      video_receive_configs_[0].decoders[0].payload_name = "VP8";
+      break;
+    case CodecType::kH264:
+      video_receive_configs_[0].decoders[0].payload_name = "H264";
+      break;
+  }
+  CreateVideoStreams();
+
+  RTPHeader header;
+  EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
+  EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc)
+      << "Packet should have configured SSRC to not be dropped early.";
+  EXPECT_EQ(payload_type, header.payloadType);
+  Start();
+  EXPECT_EQ(PacketReceiver::DELIVERY_PACKET_ERROR,
+            receiver_call_->Receiver()->DeliverPacket(MediaType::VIDEO, packet,
+                                                      length, PacketTime()));
+  Stop();
+
+  DestroyStreams();
+}
+
+TEST_F(PacketInjectionTest, StapAPacketWithTruncatedNalUnits) {
+  const uint8_t kPacket[] = {0x80,
+                             0xE5,
+                             0xE6,
+                             0x0,
+                             0x0,
+                             0xED,
+                             0x23,
+                             0x4,
+                             0x00,
+                             0xC0,
+                             0xFF,
+                             0xED,
+                             0x58,
+                             0xCB,
+                             0xED,
+                             0xDF};
+
+  InjectIncorrectPacket(CodecType::kH264, 101, kPacket, sizeof(kPacket));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/call/rampup_tests.cc b/media/webrtc/trunk/webrtc/call/rampup_tests.cc
new file mode 100644
index 0000000000..81f1e81c68
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rampup_tests.cc
@@ -0,0 +1,587 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/rampup_tests.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace {
+
+static const int64_t kPollIntervalMs = 20;
+
+std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
+  std::vector<uint32_t> ssrcs;
+  for (size_t i = 0; i != num_streams; ++i)
+    ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
+  return ssrcs;
+}
+}  // namespace
+
+RampUpTester::RampUpTester(size_t num_video_streams,
+                           size_t num_audio_streams,
+                           unsigned int start_bitrate_bps,
+                           const std::string& extension_type,
+                           bool rtx,
+                           bool red)
+    : EndToEndTest(test::CallTest::kLongTimeoutMs),
+      event_(false, false),
+      clock_(Clock::GetRealTimeClock()),
+      num_video_streams_(num_video_streams),
+      num_audio_streams_(num_audio_streams),
+      rtx_(rtx),
+      red_(red),
+      send_stream_(nullptr),
+      start_bitrate_bps_(start_bitrate_bps),
+      start_bitrate_verified_(false),
+      expected_bitrate_bps_(0),
+      test_start_ms_(-1),
+      ramp_up_finished_ms_(-1),
+      extension_type_(extension_type),
+      video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)),
+      video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)),
+      audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)),
+      poller_thread_(&BitrateStatsPollingThread,
+                     this,
+                     "BitrateStatsPollingThread"),
+      sender_call_(nullptr) {
+  EXPECT_LE(num_audio_streams_, 1u);
+  if (rtx_) {
+    for (size_t i = 0; i < video_ssrcs_.size(); ++i)
+      rtx_ssrc_map_[video_rtx_ssrcs_[i]] = video_ssrcs_[i];
+  }
+}
+
+RampUpTester::~RampUpTester() {
+  event_.Set();
+}
+
+Call::Config RampUpTester::GetSenderCallConfig() {
+  Call::Config call_config;
+  if (start_bitrate_bps_ != 0) {
+    call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
+  }
+  call_config.bitrate_config.min_bitrate_bps = 10000;
+  return call_config;
+}
+
+void RampUpTester::OnVideoStreamsCreated(
+    VideoSendStream* send_stream,
+    const std::vector<VideoReceiveStream*>& receive_streams) {
+  send_stream_ = send_stream;
+}
+
+test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
+  send_transport_ = new test::PacketTransport(sender_call, this,
+                                              test::PacketTransport::kSender,
+                                              forward_transport_config_);
+  return send_transport_;
+}
+
+size_t RampUpTester::GetNumVideoStreams() const {
+  return num_video_streams_;
+}
+
+size_t RampUpTester::GetNumAudioStreams() const {
+  return num_audio_streams_;
+}
+
+void RampUpTester::ModifyVideoConfigs(
+    VideoSendStream::Config* send_config,
+    std::vector<VideoReceiveStream::Config>* receive_configs,
+    VideoEncoderConfig* encoder_config) {
+  send_config->suspend_below_min_bitrate = true;
+
+  if (num_video_streams_ == 1) {
+    encoder_config->streams[0].target_bitrate_bps =
+        encoder_config->streams[0].max_bitrate_bps = 2000000;
+    // For single stream rampup until 1mbps
+    expected_bitrate_bps_ = kSingleStreamTargetBps;
+  } else {
+    // For multi stream rampup until all streams are being sent. That means
+    // enough birate to send all the target streams plus the min bitrate of
+    // the last one.
+    expected_bitrate_bps_ = encoder_config->streams.back().min_bitrate_bps;
+    for (size_t i = 0; i < encoder_config->streams.size() - 1; ++i) {
+      expected_bitrate_bps_ += encoder_config->streams[i].target_bitrate_bps;
+    }
+  }
+
+  send_config->rtp.extensions.clear();
+
+  bool remb;
+  bool transport_cc;
+  if (extension_type_ == RtpExtension::kAbsSendTime) {
+    remb = true;
+    transport_cc = false;
+    send_config->rtp.extensions.push_back(
+        RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
+  } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
+    remb = false;
+    transport_cc = true;
+    send_config->rtp.extensions.push_back(RtpExtension(
+        extension_type_.c_str(), kTransportSequenceNumberExtensionId));
+  } else {
+    remb = true;
+    transport_cc = false;
+    send_config->rtp.extensions.push_back(RtpExtension(
+        extension_type_.c_str(), kTransmissionTimeOffsetExtensionId));
+  }
+
+  send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
+  send_config->rtp.ssrcs = video_ssrcs_;
+  if (rtx_) {
+    send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
+    send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
+  }
+  if (red_) {
+    send_config->rtp.fec.ulpfec_payload_type =
+        test::CallTest::kUlpfecPayloadType;
+    send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType;
+  }
+
+  size_t i = 0;
+  for (VideoReceiveStream::Config& recv_config : *receive_configs) {
+    recv_config.rtp.remb = remb;
+    recv_config.rtp.transport_cc = transport_cc;
+    recv_config.rtp.extensions = send_config->rtp.extensions;
+
+    recv_config.rtp.remote_ssrc = video_ssrcs_[i];
+    recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
+
+    if (red_) {
+      recv_config.rtp.fec.red_payload_type =
+          send_config->rtp.fec.red_payload_type;
+      recv_config.rtp.fec.ulpfec_payload_type =
+          send_config->rtp.fec.ulpfec_payload_type;
+    }
+
+    if (rtx_) {
+      recv_config.rtp.rtx[send_config->encoder_settings.payload_type].ssrc =
+          video_rtx_ssrcs_[i];
+      recv_config.rtp.rtx[send_config->encoder_settings.payload_type]
+          .payload_type = send_config->rtp.rtx.payload_type;
+    }
+    ++i;
+  }
+}
+
+void RampUpTester::ModifyAudioConfigs(
+    AudioSendStream::Config* send_config,
+    std::vector<AudioReceiveStream::Config>* receive_configs) {
+  if (num_audio_streams_ == 0)
+    return;
+
+  EXPECT_NE(RtpExtension::kTOffset, extension_type_)
+      << "Audio BWE not supported with toffset.";
+
+  send_config->rtp.ssrc = audio_ssrcs_[0];
+  send_config->rtp.extensions.clear();
+
+  bool transport_cc = false;
+  if (extension_type_ == RtpExtension::kAbsSendTime) {
+    transport_cc = false;
+    send_config->rtp.extensions.push_back(
+        RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
+  } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
+    transport_cc = true;
+    send_config->rtp.extensions.push_back(RtpExtension(
+        extension_type_.c_str(), kTransportSequenceNumberExtensionId));
+  }
+
+  for (AudioReceiveStream::Config& recv_config : *receive_configs) {
+    recv_config.combined_audio_video_bwe = true;
+    recv_config.rtp.transport_cc = transport_cc;
+    recv_config.rtp.extensions = send_config->rtp.extensions;
+    recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
+  }
+}
+
+void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
+  sender_call_ = sender_call;
+}
+
+bool RampUpTester::BitrateStatsPollingThread(void* obj) {
+  return static_cast<RampUpTester*>(obj)->PollStats();
+}
+
+bool RampUpTester::PollStats() {
+  if (sender_call_) {
+    Call::Stats stats = sender_call_->GetStats();
+
+    RTC_DCHECK_GT(expected_bitrate_bps_, 0);
+    if (!start_bitrate_verified_ && start_bitrate_bps_ != 0) {
+      // For tests with an explicitly set start bitrate, verify the first
+      // bitrate estimate is close to the start bitrate and lower than the
+      // test target bitrate. This is to verify a call respects the configured
+      // start bitrate, but due to the BWE implementation we can't guarantee the
+      // first estimate really is as high as the start bitrate.
+      EXPECT_GT(stats.send_bandwidth_bps, 0.9 * start_bitrate_bps_);
+      start_bitrate_verified_ = true;
+    }
+    if (stats.send_bandwidth_bps >= expected_bitrate_bps_) {
+      ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
+      observation_complete_.Set();
+    }
+  }
+
+  return !event_.Wait(kPollIntervalMs);
+}
+
+void RampUpTester::ReportResult(const std::string& measurement,
+                                size_t value,
+                                const std::string& units) const {
+  webrtc::test::PrintResult(
+      measurement, "",
+      ::testing::UnitTest::GetInstance()->current_test_info()->name(), value,
+      units, false);
+}
+
+void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream,
+                                   size_t* total_packets_sent,
+                                   size_t* total_sent,
+                                   size_t* padding_sent,
+                                   size_t* media_sent) const {
+  *total_packets_sent += stream.rtp_stats.transmitted.packets +
+                         stream.rtp_stats.retransmitted.packets +
+                         stream.rtp_stats.fec.packets;
+  *total_sent += stream.rtp_stats.transmitted.TotalBytes() +
+                 stream.rtp_stats.retransmitted.TotalBytes() +
+                 stream.rtp_stats.fec.TotalBytes();
+  *padding_sent += stream.rtp_stats.transmitted.padding_bytes +
+                   stream.rtp_stats.retransmitted.padding_bytes +
+                   stream.rtp_stats.fec.padding_bytes;
+  *media_sent += stream.rtp_stats.MediaPayloadBytes();
+}
+
+void RampUpTester::TriggerTestDone() {
+  RTC_DCHECK_GE(test_start_ms_, 0);
+
+  // TODO(holmer): Add audio send stats here too when those APIs are available.
+  VideoSendStream::Stats send_stats = send_stream_->GetStats();
+
+  size_t total_packets_sent = 0;
+  size_t total_sent = 0;
+  size_t padding_sent = 0;
+  size_t media_sent = 0;
+  for (uint32_t ssrc : video_ssrcs_) {
+    AccumulateStats(send_stats.substreams[ssrc], &total_packets_sent,
+                    &total_sent, &padding_sent, &media_sent);
+  }
+
+  size_t rtx_total_packets_sent = 0;
+  size_t rtx_total_sent = 0;
+  size_t rtx_padding_sent = 0;
+  size_t rtx_media_sent = 0;
+  for (uint32_t rtx_ssrc : video_rtx_ssrcs_) {
+    AccumulateStats(send_stats.substreams[rtx_ssrc], &rtx_total_packets_sent,
+                    &rtx_total_sent, &rtx_padding_sent, &rtx_media_sent);
+  }
+
+  ReportResult("ramp-up-total-packets-sent", total_packets_sent, "packets");
+  ReportResult("ramp-up-total-sent", total_sent, "bytes");
+  ReportResult("ramp-up-media-sent", media_sent, "bytes");
+  ReportResult("ramp-up-padding-sent", padding_sent, "bytes");
+  ReportResult("ramp-up-rtx-total-packets-sent", rtx_total_packets_sent,
+               "packets");
+  ReportResult("ramp-up-rtx-total-sent", rtx_total_sent, "bytes");
+  ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, "bytes");
+  ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes");
+  if (ramp_up_finished_ms_ >= 0) {
+    ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
+                 "milliseconds");
+  }
+  ReportResult("ramp-up-average-network-latency",
+               send_transport_->GetAverageDelayMs(), "milliseconds");
+}
+
+void RampUpTester::PerformTest() {
+  test_start_ms_ = clock_->TimeInMilliseconds();
+  poller_thread_.Start();
+  EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete.";
+  TriggerTestDone();
+  poller_thread_.Stop();
+}
+
+RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams,
+                                       size_t num_audio_streams,
+                                       unsigned int start_bitrate_bps,
+                                       const std::string& extension_type,
+                                       bool rtx,
+                                       bool red)
+    : RampUpTester(num_video_streams,
+                   num_audio_streams,
+                   start_bitrate_bps,
+                   extension_type,
+                   rtx,
+                   red),
+      test_state_(kFirstRampup),
+      state_start_ms_(clock_->TimeInMilliseconds()),
+      interval_start_ms_(clock_->TimeInMilliseconds()),
+      sent_bytes_(0) {
+  forward_transport_config_.link_capacity_kbps = kHighBandwidthLimitBps / 1000;
+}
+
+RampUpDownUpTester::~RampUpDownUpTester() {}
+
+bool RampUpDownUpTester::PollStats() {
+  if (send_stream_) {
+    webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
+    int transmit_bitrate_bps = 0;
+    for (auto it : stats.substreams) {
+      transmit_bitrate_bps += it.second.total_bitrate_bps;
+    }
+
+    EvolveTestState(transmit_bitrate_bps, stats.suspended);
+  }
+
+  return !event_.Wait(kPollIntervalMs);
+}
+
+Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
+  Call::Config config;
+  config.bitrate_config.min_bitrate_bps = 10000;
+  return config;
+}
+
+std::string RampUpDownUpTester::GetModifierString() const {
+  std::string str("_");
+  if (num_video_streams_ > 0) {
+    std::ostringstream s;
+    s << num_video_streams_;
+    str += s.str();
+    str += "stream";
+    str += (num_video_streams_ > 1 ? "s" : "");
+    str += "_";
+  }
+  if (num_audio_streams_ > 0) {
+    std::ostringstream s;
+    s << num_audio_streams_;
+    str += s.str();
+    str += "stream";
+    str += (num_audio_streams_ > 1 ? "s" : "");
+    str += "_";
+  }
+  str += (rtx_ ? "" : "no");
+  str += "rtx";
+  return str;
+}
+
+void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
+  int64_t now = clock_->TimeInMilliseconds();
+  switch (test_state_) {
+    case kFirstRampup: {
+      EXPECT_FALSE(suspended);
+      if (bitrate_bps > kExpectedHighBitrateBps) {
+        // The first ramp-up has reached the target bitrate. Change the
+        // channel limit, and move to the next test state.
+        forward_transport_config_.link_capacity_kbps =
+            kLowBandwidthLimitBps / 1000;
+        send_transport_->SetConfig(forward_transport_config_);
+        test_state_ = kLowRate;
+        webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
+                                  "first_rampup", now - state_start_ms_, "ms",
+                                  false);
+        state_start_ms_ = now;
+        interval_start_ms_ = now;
+        sent_bytes_ = 0;
+      }
+      break;
+    }
+    case kLowRate: {
+      if (bitrate_bps < kExpectedLowBitrateBps && suspended) {
+        // The ramp-down was successful. Change the channel limit back to a
+        // high value, and move to the next test state.
+        forward_transport_config_.link_capacity_kbps =
+            kHighBandwidthLimitBps / 1000;
+        send_transport_->SetConfig(forward_transport_config_);
+        test_state_ = kSecondRampup;
+        webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
+                                  "rampdown", now - state_start_ms_, "ms",
+                                  false);
+        state_start_ms_ = now;
+        interval_start_ms_ = now;
+        sent_bytes_ = 0;
+      }
+      break;
+    }
+    case kSecondRampup: {
+      if (bitrate_bps > kExpectedHighBitrateBps && !suspended) {
+        webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
+                                  "second_rampup", now - state_start_ms_, "ms",
+                                  false);
+        ReportResult("ramp-up-down-up-average-network-latency",
+                     send_transport_->GetAverageDelayMs(), "milliseconds");
+        observation_complete_.Set();
+      }
+      break;
+    }
+  }
+}
+
+class RampUpTest : public test::CallTest {
+ public:
+  RampUpTest() {}
+
+  virtual ~RampUpTest() {
+    EXPECT_EQ(nullptr, video_send_stream_);
+    EXPECT_TRUE(video_receive_streams_.empty());
+  }
+};
+
+TEST_F(RampUpTest, SingleStream) {
+  RampUpTester test(1, 0, 0, RtpExtension::kTOffset, false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, Simulcast) {
+  RampUpTester test(3, 0, 0, RtpExtension::kTOffset, false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SimulcastWithRtx) {
+  RampUpTester test(3, 0, 0, RtpExtension::kTOffset, true, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SimulcastByRedWithRtx) {
+  RampUpTester test(3, 0, 0, RtpExtension::kTOffset, true, true);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
+  RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset,
+                    false, false);
+  RunBaseTest(&test);
+}
+
+// Disabled on Mac due to flakiness, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5407
+#ifndef WEBRTC_MAC
+
+static const uint32_t kStartBitrateBps = 60000;
+
+TEST_F(RampUpTest, UpDownUpOneStream) {
+  RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+                          false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpThreeStreams) {
+  RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+                          false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpOneStreamRtx) {
+  RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+                          true, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) {
+  RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+                          true, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpOneStreamByRedRtx) {
+  RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+                          true, true);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpThreeStreamsByRedRtx) {
+  RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+                          true, true);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SendSideVideoUpDownUpRtx) {
+  RampUpDownUpTester test(3, 0, kStartBitrateBps,
+                          RtpExtension::kTransportSequenceNumber, true, false);
+  RunBaseTest(&test);
+}
+
+// TODO(holmer): Enable when audio bitrates are included in the bitrate
+//               allocation.
+TEST_F(RampUpTest, DISABLED_SendSideAudioVideoUpDownUpRtx) {
+  RampUpDownUpTester test(3, 1, kStartBitrateBps,
+                          RtpExtension::kTransportSequenceNumber, true, false);
+  RunBaseTest(&test);
+}
+
+#endif
+
+TEST_F(RampUpTest, AbsSendTimeSingleStream) {
+  RampUpTester test(1, 0, 0, RtpExtension::kAbsSendTime, false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcast) {
+  RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTime, false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcastWithRtx) {
+  RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTime, true, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
+  RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTime, true, true);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
+  RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
+                    RtpExtension::kAbsSendTime, false, false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSingleStream) {
+  RampUpTester test(1, 0, 0, RtpExtension::kTransportSequenceNumber, false,
+                    false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
+  RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, false,
+                    false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSimulcastWithRtx) {
+  RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true,
+                    false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AudioVideoTransportSequenceNumberSimulcastWithRtx) {
+  RampUpTester test(3, 1, 0, RtpExtension::kTransportSequenceNumber, true,
+                    false);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSimulcastByRedWithRtx) {
+  RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true,
+                    true);
+  RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
+  RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
+                    RtpExtension::kTransportSequenceNumber, false, false);
+  RunBaseTest(&test);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/call/rampup_tests.h b/media/webrtc/trunk/webrtc/call/rampup_tests.h
new file mode 100644
index 0000000000..31a0a0296e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rampup_tests.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RAMPUP_TESTS_H_
+#define WEBRTC_CALL_RAMPUP_TESTS_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/event.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call.h"
+#include "webrtc/test/call_test.h"
+
+namespace webrtc {
+
+static const int kTransmissionTimeOffsetExtensionId = 6;
+static const int kAbsSendTimeExtensionId = 7;
+static const int kTransportSequenceNumberExtensionId = 8;
+static const unsigned int kSingleStreamTargetBps = 1000000;
+
+class Clock;
+
+class RampUpTester : public test::EndToEndTest {
+ public:
+  RampUpTester(size_t num_video_streams,
+               size_t num_audio_streams,
+               unsigned int start_bitrate_bps,
+               const std::string& extension_type,
+               bool rtx,
+               bool red);
+  ~RampUpTester() override;
+
+  size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
+
+  void PerformTest() override;
+
+ protected:
+  virtual bool PollStats();
+
+  void AccumulateStats(const VideoSendStream::StreamStats& stream,
+                       size_t* total_packets_sent,
+                       size_t* total_sent,
+                       size_t* padding_sent,
+                       size_t* media_sent) const;
+
+  void ReportResult(const std::string& measurement,
+                    size_t value,
+                    const std::string& units) const;
+  void TriggerTestDone();
+
+  rtc::Event event_;
+  Clock* const clock_;
+  FakeNetworkPipe::Config forward_transport_config_;
+  const size_t num_video_streams_;
+  const size_t num_audio_streams_;
+  const bool rtx_;
+  const bool red_;
+  VideoSendStream* send_stream_;
+  test::PacketTransport* send_transport_;
+
+ private:
+  typedef std::map<uint32_t, uint32_t> SsrcMap;
+
+  Call::Config GetSenderCallConfig() override;
+  void OnVideoStreamsCreated(
+      VideoSendStream* send_stream,
+      const std::vector<VideoReceiveStream*>& receive_streams) override;
+  test::PacketTransport* CreateSendTransport(Call* sender_call) override;
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override;
+  void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStream::Config>* receive_configs) override;
+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
+
+  static bool BitrateStatsPollingThread(void* obj);
+
+  const int start_bitrate_bps_;
+  bool start_bitrate_verified_;
+  int expected_bitrate_bps_;
+  int64_t test_start_ms_;
+  int64_t ramp_up_finished_ms_;
+
+  const std::string extension_type_;
+  std::vector<uint32_t> video_ssrcs_;
+  std::vector<uint32_t> video_rtx_ssrcs_;
+  std::vector<uint32_t> audio_ssrcs_;
+  SsrcMap rtx_ssrc_map_;
+
+  rtc::PlatformThread poller_thread_;
+  Call* sender_call_;
+};
+
+class RampUpDownUpTester : public RampUpTester {
+ public:
+  RampUpDownUpTester(size_t num_video_streams,
+                     size_t num_audio_streams,
+                     unsigned int start_bitrate_bps,
+                     const std::string& extension_type,
+                     bool rtx,
+                     bool red);
+  ~RampUpDownUpTester() override;
+
+ protected:
+  bool PollStats() override;
+
+ private:
+  static const int kHighBandwidthLimitBps = 80000;
+  static const int kExpectedHighBitrateBps = 60000;
+  static const int kLowBandwidthLimitBps = 20000;
+  static const int kExpectedLowBitrateBps = 20000;
+  enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
+
+  Call::Config GetReceiverCallConfig() override;
+
+  std::string GetModifierString() const;
+  void EvolveTestState(int bitrate_bps, bool suspended);
+
+  TestStates test_state_;
+  int64_t state_start_ms_;
+  int64_t interval_start_ms_;
+  int sent_bytes_;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_CALL_RAMPUP_TESTS_H_
diff --git a/media/webrtc/trunk/webrtc/call/rtc_event_log.cc b/media/webrtc/trunk/webrtc/call/rtc_event_log.cc
new file mode 100644
index 0000000000..9f592ce479
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rtc_event_log.cc
@@ -0,0 +1,523 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/rtc_event_log.h"
+
+#include <deque>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/call.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+
+#ifdef ENABLE_RTC_EVENT_LOG
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+#endif
+
+namespace webrtc {
+
+#ifndef ENABLE_RTC_EVENT_LOG
+
+// No-op implementation if flag is not set.
+class RtcEventLogImpl final : public RtcEventLog {
+ public:
+  void SetBufferDuration(int64_t buffer_duration_us) override {}
+  void StartLogging(const std::string& file_name, int duration_ms) override {}
+  bool StartLogging(rtc::PlatformFile log_file) override { return false; }
+  void StopLogging(void) override {}
+  void LogVideoReceiveStreamConfig(
+      const VideoReceiveStream::Config& config) override {}
+  void LogVideoSendStreamConfig(
+      const VideoSendStream::Config& config) override {}
+  void LogRtpHeader(bool incoming,
+                    MediaType media_type,
+                    const uint8_t* header,
+                    size_t packet_length) override {}
+  void LogRtcpPacket(bool incoming,
+                     MediaType media_type,
+                     const uint8_t* packet,
+                     size_t length) override {}
+  void LogAudioPlayout(uint32_t ssrc) override {}
+  void LogBwePacketLossEvent(int32_t bitrate,
+                             uint8_t fraction_loss,
+                             int32_t total_packets) override {}
+};
+
+#else  // ENABLE_RTC_EVENT_LOG is defined
+
+class RtcEventLogImpl final : public RtcEventLog {
+ public:
+  RtcEventLogImpl();
+
+  void SetBufferDuration(int64_t buffer_duration_us) override;
+  void StartLogging(const std::string& file_name, int duration_ms) override;
+  bool StartLogging(rtc::PlatformFile log_file) override;
+  void StopLogging() override;
+  void LogVideoReceiveStreamConfig(
+      const VideoReceiveStream::Config& config) override;
+  void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override;
+  void LogRtpHeader(bool incoming,
+                    MediaType media_type,
+                    const uint8_t* header,
+                    size_t packet_length) override;
+  void LogRtcpPacket(bool incoming,
+                     MediaType media_type,
+                     const uint8_t* packet,
+                     size_t length) override;
+  void LogAudioPlayout(uint32_t ssrc) override;
+  void LogBwePacketLossEvent(int32_t bitrate,
+                             uint8_t fraction_loss,
+                             int32_t total_packets) override;
+
+ private:
+  // Starts logging. This function assumes the file_ has been opened succesfully
+  // and that the start_time_us_ and _duration_us_ have been set.
+  void StartLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Stops logging and clears the stored data and buffers.
+  void StopLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Adds a new event to the logfile if logging is active, or adds it to the
+  // list of recent log events otherwise.
+  void HandleEvent(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Writes the event to the file. Note that this will destroy the state of the
+  // input argument.
+  void StoreToFile(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Adds the event to the list of recent events, and removes any events that
+  // are too old and no longer fall in the time window.
+  void AddRecentEvent(const rtclog::Event& event)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  rtc::CriticalSection crit_;
+  rtc::scoped_ptr<FileWrapper> file_ GUARDED_BY(crit_) =
+      rtc::scoped_ptr<FileWrapper>(FileWrapper::Create());
+  rtc::PlatformFile platform_file_ GUARDED_BY(crit_) =
+      rtc::kInvalidPlatformFileValue;
+  rtclog::EventStream stream_ GUARDED_BY(crit_);
+  std::deque<rtclog::Event> recent_log_events_ GUARDED_BY(crit_);
+  std::vector<rtclog::Event> config_events_ GUARDED_BY(crit_);
+
+  // Microseconds to record log events, before starting the actual log.
+  int64_t buffer_duration_us_ GUARDED_BY(crit_);
+  bool currently_logging_ GUARDED_BY(crit_);
+  int64_t start_time_us_ GUARDED_BY(crit_);
+  int64_t duration_us_ GUARDED_BY(crit_);
+  const Clock* const clock_;
+};
+
+namespace {
+// The functions in this namespace convert enums from the runtime format
+// that the rest of the WebRtc project can use, to the corresponding
+// serialized enum which is defined by the protobuf.
+
+// Do not add default return values to the conversion functions in this
+// unnamed namespace. The intention is to make the compiler warn if anyone
+// adds unhandled new events/modes/etc.
+
+rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
+  switch (rtcp_mode) {
+    case RtcpMode::kCompound:
+      return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
+    case RtcpMode::kReducedSize:
+      return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
+    case RtcpMode::kOff:
+      RTC_NOTREACHED();
+      return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
+  }
+  RTC_NOTREACHED();
+  return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
+}
+
+rtclog::MediaType ConvertMediaType(MediaType media_type) {
+  switch (media_type) {
+    case MediaType::ANY:
+      return rtclog::MediaType::ANY;
+    case MediaType::AUDIO:
+      return rtclog::MediaType::AUDIO;
+    case MediaType::VIDEO:
+      return rtclog::MediaType::VIDEO;
+    case MediaType::DATA:
+      return rtclog::MediaType::DATA;
+  }
+  RTC_NOTREACHED();
+  return rtclog::ANY;
+}
+
+}  // namespace
+
+namespace {
+bool IsConfigEvent(const rtclog::Event& event) {
+  rtclog::Event_EventType event_type = event.type();
+  return event_type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT ||
+         event_type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT ||
+         event_type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT ||
+         event_type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT;
+}
+}  // namespace
+
+// RtcEventLogImpl member functions.
+RtcEventLogImpl::RtcEventLogImpl()
+    : file_(FileWrapper::Create()),
+      stream_(),
+      buffer_duration_us_(10000000),
+      currently_logging_(false),
+      start_time_us_(0),
+      duration_us_(0),
+      clock_(Clock::GetRealTimeClock()) {
+}
+
+void RtcEventLogImpl::SetBufferDuration(int64_t buffer_duration_us) {
+  rtc::CritScope lock(&crit_);
+  buffer_duration_us_ = buffer_duration_us;
+}
+
+void RtcEventLogImpl::StartLogging(const std::string& file_name,
+                                   int duration_ms) {
+  rtc::CritScope lock(&crit_);
+  if (currently_logging_) {
+    StopLoggingLocked();
+  }
+  if (file_->OpenFile(file_name.c_str(), false) != 0) {
+    return;
+  }
+  start_time_us_ = clock_->TimeInMicroseconds();
+  duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
+  StartLoggingLocked();
+}
+
+bool RtcEventLogImpl::StartLogging(rtc::PlatformFile log_file) {
+  rtc::CritScope lock(&crit_);
+
+  if (currently_logging_) {
+    StopLoggingLocked();
+  }
+  RTC_DCHECK(platform_file_ == rtc::kInvalidPlatformFileValue);
+
+  FILE* file_stream = rtc::FdopenPlatformFileForWriting(log_file);
+  if (!file_stream) {
+    rtc::ClosePlatformFile(log_file);
+    return false;
+  }
+
+  if (file_->OpenFromFileHandle(file_stream, true, false) != 0) {
+    rtc::ClosePlatformFile(log_file);
+    return false;
+  }
+  platform_file_ = log_file;
+  // Set the start time and duration to keep logging for 10 minutes.
+  start_time_us_ = clock_->TimeInMicroseconds();
+  duration_us_ = 10 * 60 * 1000000;
+  StartLoggingLocked();
+  return true;
+}
+
+void RtcEventLogImpl::StartLoggingLocked() {
+  currently_logging_ = true;
+
+  // Write all old configuration events to the log file.
+  for (auto& event : config_events_) {
+    StoreToFile(&event);
+  }
+  // Write all recent configuration events to the log file, and
+  // write all other recent events to the log file, ignoring any old events.
+  for (auto& event : recent_log_events_) {
+    if (IsConfigEvent(event)) {
+      StoreToFile(&event);
+      config_events_.push_back(event);
+    } else if (event.timestamp_us() >= start_time_us_ - buffer_duration_us_) {
+      StoreToFile(&event);
+    }
+  }
+  recent_log_events_.clear();
+  // Write a LOG_START event to the file.
+  rtclog::Event start_event;
+  start_event.set_timestamp_us(start_time_us_);
+  start_event.set_type(rtclog::Event::LOG_START);
+  StoreToFile(&start_event);
+}
+
+void RtcEventLogImpl::StopLogging() {
+  rtc::CritScope lock(&crit_);
+  StopLoggingLocked();
+}
+
+void RtcEventLogImpl::LogVideoReceiveStreamConfig(
+    const VideoReceiveStream::Config& config) {
+  rtc::CritScope lock(&crit_);
+
+  rtclog::Event event;
+  event.set_timestamp_us(clock_->TimeInMicroseconds());
+  event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
+
+  rtclog::VideoReceiveConfig* receiver_config =
+      event.mutable_video_receiver_config();
+  receiver_config->set_remote_ssrc(config.rtp.remote_ssrc);
+  receiver_config->set_local_ssrc(config.rtp.local_ssrc);
+
+  receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode));
+  receiver_config->set_remb(config.rtp.remb);
+
+  for (const auto& kv : config.rtp.rtx) {
+    rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
+    rtx->set_payload_type(kv.first);
+    rtx->mutable_config()->set_rtx_ssrc(kv.second.ssrc);
+    rtx->mutable_config()->set_rtx_payload_type(kv.second.payload_type);
+  }
+
+  for (const auto& e : config.rtp.extensions) {
+    rtclog::RtpHeaderExtension* extension =
+        receiver_config->add_header_extensions();
+    extension->set_name(e.name);
+    extension->set_id(e.id);
+  }
+
+  for (const auto& d : config.decoders) {
+    rtclog::DecoderConfig* decoder = receiver_config->add_decoders();
+    decoder->set_name(d.payload_name);
+    decoder->set_payload_type(d.payload_type);
+  }
+  HandleEvent(&event);
+}
+
+void RtcEventLogImpl::LogVideoSendStreamConfig(
+    const VideoSendStream::Config& config) {
+  rtc::CritScope lock(&crit_);
+
+  rtclog::Event event;
+  event.set_timestamp_us(clock_->TimeInMicroseconds());
+  event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
+
+  rtclog::VideoSendConfig* sender_config = event.mutable_video_sender_config();
+
+  for (const auto& ssrc : config.rtp.ssrcs) {
+    sender_config->add_ssrcs(ssrc);
+  }
+
+  for (const auto& e : config.rtp.extensions) {
+    rtclog::RtpHeaderExtension* extension =
+        sender_config->add_header_extensions();
+    extension->set_name(e.name);
+    extension->set_id(e.id);
+  }
+
+  for (const auto& rtx_ssrc : config.rtp.rtx.ssrcs) {
+    sender_config->add_rtx_ssrcs(rtx_ssrc);
+  }
+  sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
+
+  rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
+  encoder->set_name(config.encoder_settings.payload_name);
+  encoder->set_payload_type(config.encoder_settings.payload_type);
+  HandleEvent(&event);
+}
+
+void RtcEventLogImpl::LogRtpHeader(bool incoming,
+                                   MediaType media_type,
+                                   const uint8_t* header,
+                                   size_t packet_length) {
+  // Read header length (in bytes) from packet data.
+  if (packet_length < 12u) {
+    return;  // Don't read outside the packet.
+  }
+  const bool x = (header[0] & 0x10) != 0;
+  const uint8_t cc = header[0] & 0x0f;
+  size_t header_length = 12u + cc * 4u;
+
+  if (x) {
+    if (packet_length < 12u + cc * 4u + 4u) {
+      return;  // Don't read outside the packet.
+    }
+    size_t x_len = ByteReader<uint16_t>::ReadBigEndian(header + 14 + cc * 4);
+    header_length += (x_len + 1) * 4;
+  }
+
+  rtc::CritScope lock(&crit_);
+  rtclog::Event rtp_event;
+  rtp_event.set_timestamp_us(clock_->TimeInMicroseconds());
+  rtp_event.set_type(rtclog::Event::RTP_EVENT);
+  rtp_event.mutable_rtp_packet()->set_incoming(incoming);
+  rtp_event.mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
+  rtp_event.mutable_rtp_packet()->set_packet_length(packet_length);
+  rtp_event.mutable_rtp_packet()->set_header(header, header_length);
+  HandleEvent(&rtp_event);
+}
+
+void RtcEventLogImpl::LogRtcpPacket(bool incoming,
+                                    MediaType media_type,
+                                    const uint8_t* packet,
+                                    size_t length) {
+  rtc::CritScope lock(&crit_);
+  rtclog::Event rtcp_event;
+  rtcp_event.set_timestamp_us(clock_->TimeInMicroseconds());
+  rtcp_event.set_type(rtclog::Event::RTCP_EVENT);
+  rtcp_event.mutable_rtcp_packet()->set_incoming(incoming);
+  rtcp_event.mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
+
+  RTCPUtility::RtcpCommonHeader header;
+  const uint8_t* block_begin = packet;
+  const uint8_t* packet_end = packet + length;
+  RTC_DCHECK(length <= IP_PACKET_SIZE);
+  uint8_t buffer[IP_PACKET_SIZE];
+  uint32_t buffer_length = 0;
+  while (block_begin < packet_end) {
+    if (!RtcpParseCommonHeader(block_begin, packet_end - block_begin,
+                               &header)) {
+      break;  // Incorrect message header.
+    }
+    uint32_t block_size = header.BlockSize();
+    switch (header.packet_type) {
+      case RTCPUtility::PT_SR:
+        FALLTHROUGH();
+      case RTCPUtility::PT_RR:
+        FALLTHROUGH();
+      case RTCPUtility::PT_BYE:
+        FALLTHROUGH();
+      case RTCPUtility::PT_IJ:
+        FALLTHROUGH();
+      case RTCPUtility::PT_RTPFB:
+        FALLTHROUGH();
+      case RTCPUtility::PT_PSFB:
+        FALLTHROUGH();
+      case RTCPUtility::PT_XR:
+        // We log sender reports, receiver reports, bye messages
+        // inter-arrival jitter, third-party loss reports, payload-specific
+        // feedback and extended reports.
+        memcpy(buffer + buffer_length, block_begin, block_size);
+        buffer_length += block_size;
+        break;
+      case RTCPUtility::PT_SDES:
+        FALLTHROUGH();
+      case RTCPUtility::PT_APP:
+        FALLTHROUGH();
+      default:
+        // We don't log sender descriptions, application defined messages
+        // or message blocks of unknown type.
+        break;
+    }
+
+    block_begin += block_size;
+  }
+  rtcp_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
+  HandleEvent(&rtcp_event);
+}
+
+void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
+  rtc::CritScope lock(&crit_);
+  rtclog::Event event;
+  event.set_timestamp_us(clock_->TimeInMicroseconds());
+  event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
+  auto playout_event = event.mutable_audio_playout_event();
+  playout_event->set_local_ssrc(ssrc);
+  HandleEvent(&event);
+}
+
+void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
+                                            uint8_t fraction_loss,
+                                            int32_t total_packets) {
+  rtc::CritScope lock(&crit_);
+  rtclog::Event event;
+  event.set_timestamp_us(clock_->TimeInMicroseconds());
+  event.set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
+  auto bwe_event = event.mutable_bwe_packet_loss_event();
+  bwe_event->set_bitrate(bitrate);
+  bwe_event->set_fraction_loss(fraction_loss);
+  bwe_event->set_total_packets(total_packets);
+  HandleEvent(&event);
+}
+
+void RtcEventLogImpl::StopLoggingLocked() {
+  if (currently_logging_) {
+    currently_logging_ = false;
+    // Create a LogEnd event
+    rtclog::Event event;
+    event.set_timestamp_us(clock_->TimeInMicroseconds());
+    event.set_type(rtclog::Event::LOG_END);
+    // Store the event and close the file
+    RTC_DCHECK(file_->Open());
+    StoreToFile(&event);
+    file_->CloseFile();
+    if (platform_file_ != rtc::kInvalidPlatformFileValue) {
+      rtc::ClosePlatformFile(platform_file_);
+      platform_file_ = rtc::kInvalidPlatformFileValue;
+    }
+  }
+  RTC_DCHECK(!file_->Open());
+  stream_.Clear();
+}
+
+void RtcEventLogImpl::HandleEvent(rtclog::Event* event) {
+  if (currently_logging_) {
+    if (clock_->TimeInMicroseconds() < start_time_us_ + duration_us_) {
+      StoreToFile(event);
+      return;
+    }
+    StopLoggingLocked();
+  }
+  AddRecentEvent(*event);
+}
+
+void RtcEventLogImpl::StoreToFile(rtclog::Event* event) {
+  // Reuse the same object at every log event.
+  if (stream_.stream_size() < 1) {
+    stream_.add_stream();
+  }
+  RTC_DCHECK_EQ(stream_.stream_size(), 1);
+  stream_.mutable_stream(0)->Swap(event);
+  // TODO(terelius): Doesn't this create a new EventStream per event?
+  // Is this guaranteed to work e.g. in future versions of protobuf?
+  std::string dump_buffer;
+  stream_.SerializeToString(&dump_buffer);
+  file_->Write(dump_buffer.data(), dump_buffer.size());
+}
+
+void RtcEventLogImpl::AddRecentEvent(const rtclog::Event& event) {
+  recent_log_events_.push_back(event);
+  while (recent_log_events_.front().timestamp_us() <
+         event.timestamp_us() - buffer_duration_us_) {
+    if (IsConfigEvent(recent_log_events_.front())) {
+      config_events_.push_back(recent_log_events_.front());
+    }
+    recent_log_events_.pop_front();
+  }
+}
+
+bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
+                                   rtclog::EventStream* result) {
+  char tmp_buffer[1024];
+  int bytes_read = 0;
+  rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create());
+  if (dump_file->OpenFile(file_name.c_str(), true) != 0) {
+    return false;
+  }
+  std::string dump_buffer;
+  while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
+    dump_buffer.append(tmp_buffer, bytes_read);
+  }
+  dump_file->CloseFile();
+  return result->ParseFromString(dump_buffer);
+}
+
+#endif  // ENABLE_RTC_EVENT_LOG
+
+// RtcEventLog member functions.
+rtc::scoped_ptr<RtcEventLog> RtcEventLog::Create() {
+  return rtc::scoped_ptr<RtcEventLog>(new RtcEventLogImpl());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/call/rtc_event_log.h b/media/webrtc/trunk/webrtc/call/rtc_event_log.h
new file mode 100644
index 0000000000..489687a195
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rtc_event_log.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RTC_EVENT_LOG_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_H_
+
+#include <string>
+
+#include "webrtc/base/platform_file.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/video_receive_stream.h"
+#include "webrtc/video_send_stream.h"
+
+namespace webrtc {
+
+// Forward declaration of storage class that is automatically generated from
+// the protobuf file.
+namespace rtclog {
+class EventStream;
+}  // namespace rtclog
+
+class RtcEventLogImpl;
+
+enum class MediaType;
+
+class RtcEventLog {
+ public:
+  virtual ~RtcEventLog() {}
+
+  static rtc::scoped_ptr<RtcEventLog> Create();
+
+  // Sets the time that events are stored in the internal event buffer
+  // before the user calls StartLogging.  The default is 10 000 000 us = 10 s
+  virtual void SetBufferDuration(int64_t buffer_duration_us) = 0;
+
+  // Starts logging for the specified duration to the specified file.
+  // The logging will stop automatically after the specified duration.
+  // If the file already exists it will be overwritten.
+  // If the file cannot be opened, the RtcEventLog will not start logging.
+  virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
+
+  // Starts logging until either the 10 minute timer runs out or the StopLogging
+  // function is called. The RtcEventLog takes ownership of the supplied
+  // rtc::PlatformFile.
+  virtual bool StartLogging(rtc::PlatformFile log_file) = 0;
+
+  virtual void StopLogging() = 0;
+
+  // Logs configuration information for webrtc::VideoReceiveStream
+  virtual void LogVideoReceiveStreamConfig(
+      const webrtc::VideoReceiveStream::Config& config) = 0;
+
+  // Logs configuration information for webrtc::VideoSendStream
+  virtual void LogVideoSendStreamConfig(
+      const webrtc::VideoSendStream::Config& config) = 0;
+
+  // Logs the header of an incoming or outgoing RTP packet. packet_length
+  // is the total length of the packet, including both header and payload.
+  virtual void LogRtpHeader(bool incoming,
+                            MediaType media_type,
+                            const uint8_t* header,
+                            size_t packet_length) = 0;
+
+  // Logs an incoming or outgoing RTCP packet.
+  virtual void LogRtcpPacket(bool incoming,
+                             MediaType media_type,
+                             const uint8_t* packet,
+                             size_t length) = 0;
+
+  // Logs an audio playout event
+  virtual void LogAudioPlayout(uint32_t ssrc) = 0;
+
+  // Logs a bitrate update from the bandwidth estimator based on packet loss.
+  virtual void LogBwePacketLossEvent(int32_t bitrate,
+                                     uint8_t fraction_loss,
+                                     int32_t total_packets) = 0;
+
+  // Reads an RtcEventLog file and returns true when reading was successful.
+  // The result is stored in the given EventStream object.
+  static bool ParseRtcEventLog(const std::string& file_name,
+                               rtclog::EventStream* result);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTC_EVENT_LOG_H_
diff --git a/media/webrtc/trunk/webrtc/call/rtc_event_log.proto b/media/webrtc/trunk/webrtc/call/rtc_event_log.proto
new file mode 100644
index 0000000000..b14306e362
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rtc_event_log.proto
@@ -0,0 +1,242 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.rtclog;
+
+
+enum MediaType {
+  ANY = 0;
+  AUDIO = 1;
+  VIDEO = 2;
+  DATA = 3;
+}
+
+
+// This is the main message to dump to a file, it can contain multiple event
+// messages, but it is possible to append multiple EventStreams (each with a
+// single event) to a file.
+// This has the benefit that there's no need to keep all data in memory.
+message EventStream {
+  repeated Event stream = 1;
+}
+
+
+message Event {
+  // required - Elapsed wallclock time in us since the start of the log.
+  optional int64 timestamp_us = 1;
+
+  // The different types of events that can occur, the UNKNOWN_EVENT entry
+  // is added in case future EventTypes are added, in that case old code will
+  // receive the new events as UNKNOWN_EVENT.
+  enum EventType {
+    UNKNOWN_EVENT = 0;
+    LOG_START = 1;
+    LOG_END = 2;
+    RTP_EVENT = 3;
+    RTCP_EVENT = 4;
+    AUDIO_PLAYOUT_EVENT = 5;
+    BWE_PACKET_LOSS_EVENT = 6;
+    BWE_PACKET_DELAY_EVENT = 7;
+    VIDEO_RECEIVER_CONFIG_EVENT = 8;
+    VIDEO_SENDER_CONFIG_EVENT = 9;
+    AUDIO_RECEIVER_CONFIG_EVENT = 10;
+    AUDIO_SENDER_CONFIG_EVENT = 11;
+  }
+
+  // required - Indicates the type of this event
+  optional EventType type = 2;
+
+  // optional - but required if type == RTP_EVENT
+  optional RtpPacket rtp_packet = 3;
+
+  // optional - but required if type == RTCP_EVENT
+  optional RtcpPacket rtcp_packet = 4;
+
+  // optional - but required if type == AUDIO_PLAYOUT_EVENT
+  optional AudioPlayoutEvent audio_playout_event = 5;
+
+  // optional - but required if type == BWE_PACKET_LOSS_EVENT
+  optional BwePacketLossEvent bwe_packet_loss_event = 6;
+
+  // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
+  optional VideoReceiveConfig video_receiver_config = 8;
+
+  // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
+  optional VideoSendConfig video_sender_config = 9;
+
+  // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
+  optional AudioReceiveConfig audio_receiver_config = 10;
+
+  // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
+  optional AudioSendConfig audio_sender_config = 11;
+}
+
+
+message RtpPacket {
+  // required - True if the packet is incoming w.r.t. the user logging the data
+  optional bool incoming = 1;
+
+  // required
+  optional MediaType type = 2;
+
+  // required - The size of the packet including both payload and header.
+  optional uint32 packet_length = 3;
+
+  // required - The RTP header only.
+  optional bytes header = 4;
+
+  // Do not add code to log user payload data without a privacy review!
+}
+
+
+message RtcpPacket {
+  // required - True if the packet is incoming w.r.t. the user logging the data
+  optional bool incoming = 1;
+
+  // required
+  optional MediaType type = 2;
+
+  // required - The whole packet including both payload and header.
+  optional bytes packet_data = 3;
+}
+
+message AudioPlayoutEvent {
+  // required - The SSRC of the audio stream associated with the playout event.
+  optional uint32 local_ssrc = 2;
+}
+
+message BwePacketLossEvent {
+  // required - Bandwidth estimate (in bps) after the update.
+  optional int32 bitrate = 1;
+
+  // required - Fraction of lost packets since last receiver report
+  // computed as floor( 256 * (#lost_packets / #total_packets) ).
+  // The possible values range from 0 to 255.
+  optional uint32 fraction_loss = 2;
+
+  // TODO(terelius): Is this really needed? Remove or make optional?
+  // required - Total number of packets that the BWE update is based on.
+  optional int32 total_packets = 3;
+}
+
+// TODO(terelius): Video and audio streams could in principle share SSRC,
+// so identifying a stream based only on SSRC might not work.
+// It might be better to use a combination of SSRC and media type
+// or SSRC and port number, but for now we will rely on SSRC only.
+message VideoReceiveConfig {
+  // required - Synchronization source (stream identifier) to be received.
+  optional uint32 remote_ssrc = 1;
+  // required - Sender SSRC used for sending RTCP (such as receiver reports).
+  optional uint32 local_ssrc = 2;
+
+  // Compound mode is described by RFC 4585 and reduced-size
+  // RTCP mode is described by RFC 5506.
+  enum RtcpMode {
+    RTCP_COMPOUND = 1;
+    RTCP_REDUCEDSIZE = 2;
+  }
+  // required - RTCP mode to use.
+  optional RtcpMode rtcp_mode = 3;
+
+  // required - Receiver estimated maximum bandwidth.
+  optional bool remb = 4;
+
+  // Map from video RTP payload type -> RTX config.
+  repeated RtxMap rtx_map = 5;
+
+  // RTP header extensions used for the received stream.
+  repeated RtpHeaderExtension header_extensions = 6;
+
+  // List of decoders associated with the stream.
+  repeated DecoderConfig decoders = 7;
+}
+
+
+// Maps decoder names to payload types.
+message DecoderConfig {
+  // required
+  optional string name = 1;
+
+  // required
+  optional int32 payload_type = 2;
+}
+
+
+// Maps RTP header extension names to numerical IDs.
+message RtpHeaderExtension {
+  // required
+  optional string name = 1;
+
+  // required
+  optional int32 id = 2;
+}
+
+
+// RTX settings for incoming video payloads that may be received.
+// RTX is disabled if there's no config present.
+message RtxConfig {
+  // required - SSRC to use for the RTX stream.
+  optional uint32 rtx_ssrc = 1;
+
+  // required - Payload type to use for the RTX stream.
+  optional int32 rtx_payload_type = 2;
+}
+
+
+message RtxMap {
+  // required
+  optional int32 payload_type = 1;
+
+  // required
+  optional RtxConfig config = 2;
+}
+
+
+message VideoSendConfig {
+  // Synchronization source (stream identifier) for outgoing stream.
+  // One stream can have several ssrcs for e.g. simulcast.
+  // At least one ssrc is required.
+  repeated uint32 ssrcs = 1;
+
+  // RTP header extensions used for the outgoing stream.
+  repeated RtpHeaderExtension header_extensions = 2;
+
+  // List of SSRCs for retransmitted packets.
+  repeated uint32 rtx_ssrcs = 3;
+
+  // required if rtx_ssrcs is used - Payload type for retransmitted packets.
+  optional int32 rtx_payload_type = 4;
+
+  // required - Encoder associated with the stream.
+  optional EncoderConfig encoder = 5;
+}
+
+
+// Maps encoder names to payload types.
+message EncoderConfig {
+  // required
+  optional string name = 1;
+
+  // required
+  optional int32 payload_type = 2;
+}
+
+
+message AudioReceiveConfig {
+  // required - Synchronization source (stream identifier) to be received.
+  optional uint32 remote_ssrc = 1;
+
+  // required - Sender SSRC used for sending RTCP (such as receiver reports).
+  optional uint32 local_ssrc = 2;
+
+  // RTP header extensions used for the received audio stream.
+  repeated RtpHeaderExtension header_extensions = 3;
+}
+
+
+message AudioSendConfig {
+  // required - Synchronization source (stream identifier) for outgoing stream.
+  optional uint32 ssrc = 1;
+
+  // RTP header extensions used for the outgoing audio stream.
+  repeated RtpHeaderExtension header_extensions = 2;
+}
diff --git a/media/webrtc/trunk/webrtc/call/rtc_event_log2rtp_dump.cc b/media/webrtc/trunk/webrtc/call/rtc_event_log2rtp_dump.cc
new file mode 100644
index 0000000000..8357d4856a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rtc_event_log2rtp_dump.cc
@@ -0,0 +1,207 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <iostream>
+#include <sstream>
+#include <string>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/rtp_file_writer.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace {
+
+DEFINE_bool(noaudio,
+            false,
+            "Excludes audio packets from the converted RTPdump file.");
+DEFINE_bool(novideo,
+            false,
+            "Excludes video packets from the converted RTPdump file.");
+DEFINE_bool(nodata,
+            false,
+            "Excludes data packets from the converted RTPdump file.");
+DEFINE_bool(nortp,
+            false,
+            "Excludes RTP packets from the converted RTPdump file.");
+DEFINE_bool(nortcp,
+            false,
+            "Excludes RTCP packets from the converted RTPdump file.");
+DEFINE_string(ssrc,
+              "",
+              "Store only packets with this SSRC (decimal or hex, the latter "
+              "starting with 0x).");
+
+// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
+// written to the output variable |ssrc|, and true is returned. Otherwise,
+// false is returned.
+// The empty string must be validated as true, because it is the default value
+// of the command-line flag. In this case, no value is written to the output
+// variable.
+bool ParseSsrc(std::string str, uint32_t* ssrc) {
+  // If the input string starts with 0x or 0X it indicates a hexadecimal number.
+  auto read_mode = std::dec;
+  if (str.size() > 2 &&
+      (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
+    read_mode = std::hex;
+    str = str.substr(2);
+  }
+  std::stringstream ss(str);
+  ss >> read_mode >> *ssrc;
+  return str.empty() || (!ss.fail() && ss.eof());
+}
+
+}  // namespace
+
+// This utility will convert a stored event log to the rtpdump format.
+int main(int argc, char* argv[]) {
+  std::string program_name = argv[0];
+  std::string usage =
+      "Tool for converting an RtcEventLog file to an RTP dump file.\n"
+      "Run " +
+      program_name +
+      " --helpshort for usage.\n"
+      "Example usage:\n" +
+      program_name + " input.rel output.rtp\n";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 3) {
+    std::cout << google::ProgramUsage();
+    return 0;
+  }
+  std::string input_file = argv[1];
+  std::string output_file = argv[2];
+
+  uint32_t ssrc_filter = 0;
+  if (!FLAGS_ssrc.empty())
+    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
+        << "Flag verification has failed.";
+
+  webrtc::rtclog::EventStream event_stream;
+  if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
+    std::cerr << "Error while parsing input file: " << input_file << std::endl;
+    return -1;
+  }
+
+  rtc::scoped_ptr<webrtc::test::RtpFileWriter> rtp_writer(
+      webrtc::test::RtpFileWriter::Create(
+          webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
+
+  if (!rtp_writer.get()) {
+    std::cerr << "Error while opening output file: " << output_file
+              << std::endl;
+    return -1;
+  }
+
+  std::cout << "Found " << event_stream.stream_size()
+            << " events in the input file." << std::endl;
+  int rtp_counter = 0, rtcp_counter = 0;
+  bool header_only = false;
+  // TODO(ivoc): This can be refactored once the packet interpretation
+  //             functions are finished.
+  for (int i = 0; i < event_stream.stream_size(); i++) {
+    const webrtc::rtclog::Event& event = event_stream.stream(i);
+    if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) {
+      if (event.has_timestamp_us() && event.has_rtp_packet() &&
+          event.rtp_packet().has_header() &&
+          event.rtp_packet().header().size() >= 12 &&
+          event.rtp_packet().has_packet_length() &&
+          event.rtp_packet().has_type()) {
+        const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+        if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
+          continue;
+        if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
+          continue;
+        if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
+          continue;
+        if (!FLAGS_ssrc.empty()) {
+          const uint32_t packet_ssrc =
+              webrtc::ByteReader<uint32_t>::ReadBigEndian(
+                  reinterpret_cast<const uint8_t*>(rtp_packet.header().data() +
+                                                   8));
+          if (packet_ssrc != ssrc_filter)
+            continue;
+        }
+
+        webrtc::test::RtpPacket packet;
+        packet.length = rtp_packet.header().size();
+        if (packet.length > packet.kMaxPacketBufferSize) {
+          std::cout << "Skipping packet with size " << packet.length
+                    << ", the maximum supported size is "
+                    << packet.kMaxPacketBufferSize << std::endl;
+          continue;
+        }
+        packet.original_length = rtp_packet.packet_length();
+        if (packet.original_length > packet.length)
+          header_only = true;
+        packet.time_ms = event.timestamp_us() / 1000;
+        memcpy(packet.data, rtp_packet.header().data(), packet.length);
+        rtp_writer->WritePacket(&packet);
+        rtp_counter++;
+      } else {
+        std::cout << "Skipping malformed event." << std::endl;
+      }
+    }
+    if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) {
+      if (event.has_timestamp_us() && event.has_rtcp_packet() &&
+          event.rtcp_packet().has_type() &&
+          event.rtcp_packet().has_packet_data() &&
+          event.rtcp_packet().packet_data().size() > 0) {
+        const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
+        if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
+          continue;
+        if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
+          continue;
+        if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
+          continue;
+        if (!FLAGS_ssrc.empty()) {
+          const uint32_t packet_ssrc =
+              webrtc::ByteReader<uint32_t>::ReadBigEndian(
+                  reinterpret_cast<const uint8_t*>(
+                      rtcp_packet.packet_data().data() + 4));
+          if (packet_ssrc != ssrc_filter)
+            continue;
+        }
+
+        webrtc::test::RtpPacket packet;
+        packet.length = rtcp_packet.packet_data().size();
+        if (packet.length > packet.kMaxPacketBufferSize) {
+          std::cout << "Skipping packet with size " << packet.length
+                    << ", the maximum supported size is "
+                    << packet.kMaxPacketBufferSize << std::endl;
+          continue;
+        }
+        // For RTCP packets the original_length should be set to 0 in the
+        // RTPdump format.
+        packet.original_length = 0;
+        packet.time_ms = event.timestamp_us() / 1000;
+        memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length);
+        rtp_writer->WritePacket(&packet);
+        rtcp_counter++;
+      } else {
+        std::cout << "Skipping malformed event." << std::endl;
+      }
+    }
+  }
+  std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
+            << " RTP packets and " << rtcp_counter << " RTCP packets to the "
+            << "output file." << std::endl;
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/call/rtc_event_log_unittest.cc b/media/webrtc/trunk/webrtc/call/rtc_event_log_unittest.cc
new file mode 100644
index 0000000000..f590f669a2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/rtc_event_log_unittest.cc
@@ -0,0 +1,690 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifdef ENABLE_RTC_EVENT_LOG
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/random.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/test_suite.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+const RTPExtensionType kExtensionTypes[] = {
+    RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
+    RTPExtensionType::kRtpExtensionAudioLevel,
+    RTPExtensionType::kRtpExtensionAbsoluteSendTime,
+    RTPExtensionType::kRtpExtensionVideoRotation,
+    RTPExtensionType::kRtpExtensionTransportSequenceNumber};
+const char* kExtensionNames[] = {RtpExtension::kTOffset,
+                                 RtpExtension::kAudioLevel,
+                                 RtpExtension::kAbsSendTime,
+                                 RtpExtension::kVideoRotation,
+                                 RtpExtension::kTransportSequenceNumber};
+const size_t kNumExtensions = 5;
+
+}  // namespace
+
+// TODO(terelius): Place this definition with other parsing functions?
+MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
+  switch (media_type) {
+    case rtclog::MediaType::ANY:
+      return MediaType::ANY;
+    case rtclog::MediaType::AUDIO:
+      return MediaType::AUDIO;
+    case rtclog::MediaType::VIDEO:
+      return MediaType::VIDEO;
+    case rtclog::MediaType::DATA:
+      return MediaType::DATA;
+  }
+  RTC_NOTREACHED();
+  return MediaType::ANY;
+}
+
+// Checks that the event has a timestamp, a type and exactly the data field
+// corresponding to the type.
+::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
+  if (!event.has_timestamp_us())
+    return ::testing::AssertionFailure() << "Event has no timestamp";
+  if (!event.has_type())
+    return ::testing::AssertionFailure() << "Event has no event type";
+  rtclog::Event_EventType type = event.type();
+  if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet())
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
+  if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet())
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
+  if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
+      event.has_audio_playout_event())
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_audio_playout_event() ? "" : "no ")
+           << "audio_playout event";
+  if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
+      event.has_video_receiver_config())
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_video_receiver_config() ? "" : "no ")
+           << "receiver config";
+  if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
+      event.has_video_sender_config())
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_video_sender_config() ? "" : "no ") << "sender config";
+  if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
+      event.has_audio_receiver_config()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_audio_receiver_config() ? "" : "no ")
+           << "audio receiver config";
+  }
+  if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
+      event.has_audio_sender_config()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_audio_sender_config() ? "" : "no ")
+           << "audio sender config";
+  }
+  return ::testing::AssertionSuccess();
+}
+
+void VerifyReceiveStreamConfig(const rtclog::Event& event,
+                               const VideoReceiveStream::Config& config) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
+  const rtclog::VideoReceiveConfig& receiver_config =
+      event.video_receiver_config();
+  // Check SSRCs.
+  ASSERT_TRUE(receiver_config.has_remote_ssrc());
+  EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
+  ASSERT_TRUE(receiver_config.has_local_ssrc());
+  EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
+  // Check RTCP settings.
+  ASSERT_TRUE(receiver_config.has_rtcp_mode());
+  if (config.rtp.rtcp_mode == RtcpMode::kCompound)
+    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
+              receiver_config.rtcp_mode());
+  else
+    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
+              receiver_config.rtcp_mode());
+  ASSERT_TRUE(receiver_config.has_remb());
+  EXPECT_EQ(config.rtp.remb, receiver_config.remb());
+  // Check RTX map.
+  ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
+            receiver_config.rtx_map_size());
+  for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
+    ASSERT_TRUE(rtx_map.has_payload_type());
+    ASSERT_TRUE(rtx_map.has_config());
+    EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
+    const rtclog::RtxConfig& rtx_config = rtx_map.config();
+    const VideoReceiveStream::Config::Rtp::Rtx& rtx =
+        config.rtp.rtx.at(rtx_map.payload_type());
+    ASSERT_TRUE(rtx_config.has_rtx_ssrc());
+    ASSERT_TRUE(rtx_config.has_rtx_payload_type());
+    EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
+    EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
+  }
+  // Check header extensions.
+  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+            receiver_config.header_extensions_size());
+  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
+    ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
+    ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
+    const std::string& name = receiver_config.header_extensions(i).name();
+    int id = receiver_config.header_extensions(i).id();
+    EXPECT_EQ(config.rtp.extensions[i].id, id);
+    EXPECT_EQ(config.rtp.extensions[i].name, name);
+  }
+  // Check decoders.
+  ASSERT_EQ(static_cast<int>(config.decoders.size()),
+            receiver_config.decoders_size());
+  for (int i = 0; i < receiver_config.decoders_size(); i++) {
+    ASSERT_TRUE(receiver_config.decoders(i).has_name());
+    ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
+    const std::string& decoder_name = receiver_config.decoders(i).name();
+    int decoder_type = receiver_config.decoders(i).payload_type();
+    EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
+    EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
+  }
+}
+
+void VerifySendStreamConfig(const rtclog::Event& event,
+                            const VideoSendStream::Config& config) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
+  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
+  // Check SSRCs.
+  ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
+            sender_config.ssrcs_size());
+  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
+    EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
+  }
+  // Check header extensions.
+  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+            sender_config.header_extensions_size());
+  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
+    ASSERT_TRUE(sender_config.header_extensions(i).has_name());
+    ASSERT_TRUE(sender_config.header_extensions(i).has_id());
+    const std::string& name = sender_config.header_extensions(i).name();
+    int id = sender_config.header_extensions(i).id();
+    EXPECT_EQ(config.rtp.extensions[i].id, id);
+    EXPECT_EQ(config.rtp.extensions[i].name, name);
+  }
+  // Check RTX settings.
+  ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
+            sender_config.rtx_ssrcs_size());
+  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
+    EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
+  }
+  if (sender_config.rtx_ssrcs_size() > 0) {
+    ASSERT_TRUE(sender_config.has_rtx_payload_type());
+    EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
+  }
+  // Check encoder.
+  ASSERT_TRUE(sender_config.has_encoder());
+  ASSERT_TRUE(sender_config.encoder().has_name());
+  ASSERT_TRUE(sender_config.encoder().has_payload_type());
+  EXPECT_EQ(config.encoder_settings.payload_name,
+            sender_config.encoder().name());
+  EXPECT_EQ(config.encoder_settings.payload_type,
+            sender_config.encoder().payload_type());
+}
+
+void VerifyRtpEvent(const rtclog::Event& event,
+                    bool incoming,
+                    MediaType media_type,
+                    const uint8_t* header,
+                    size_t header_size,
+                    size_t total_size) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
+  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+  ASSERT_TRUE(rtp_packet.has_incoming());
+  EXPECT_EQ(incoming, rtp_packet.incoming());
+  ASSERT_TRUE(rtp_packet.has_type());
+  EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
+  ASSERT_TRUE(rtp_packet.has_packet_length());
+  EXPECT_EQ(total_size, rtp_packet.packet_length());
+  ASSERT_TRUE(rtp_packet.has_header());
+  ASSERT_EQ(header_size, rtp_packet.header().size());
+  for (size_t i = 0; i < header_size; i++) {
+    EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
+  }
+}
+
+void VerifyRtcpEvent(const rtclog::Event& event,
+                     bool incoming,
+                     MediaType media_type,
+                     const uint8_t* packet,
+                     size_t total_size) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
+  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
+  ASSERT_TRUE(rtcp_packet.has_incoming());
+  EXPECT_EQ(incoming, rtcp_packet.incoming());
+  ASSERT_TRUE(rtcp_packet.has_type());
+  EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
+  ASSERT_TRUE(rtcp_packet.has_packet_data());
+  ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
+  for (size_t i = 0; i < total_size; i++) {
+    EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
+  }
+}
+
+void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
+  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
+  ASSERT_TRUE(playout_event.has_local_ssrc());
+  EXPECT_EQ(ssrc, playout_event.local_ssrc());
+}
+
+void VerifyBweLossEvent(const rtclog::Event& event,
+                        int32_t bitrate,
+                        uint8_t fraction_loss,
+                        int32_t total_packets) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
+  const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
+  ASSERT_TRUE(bwe_event.has_bitrate());
+  EXPECT_EQ(bitrate, bwe_event.bitrate());
+  ASSERT_TRUE(bwe_event.has_fraction_loss());
+  EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
+  ASSERT_TRUE(bwe_event.has_total_packets());
+  EXPECT_EQ(total_packets, bwe_event.total_packets());
+}
+
+void VerifyLogStartEvent(const rtclog::Event& event) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  EXPECT_EQ(rtclog::Event::LOG_START, event.type());
+}
+
+/*
+ * Bit number i of extension_bitvector is set to indicate the
+ * presence of extension number i from kExtensionTypes / kExtensionNames.
+ * The least significant bit extension_bitvector has number 0.
+ */
+size_t GenerateRtpPacket(uint32_t extensions_bitvector,
+                         uint32_t csrcs_count,
+                         uint8_t* packet,
+                         size_t packet_size,
+                         Random* prng) {
+  RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
+  Clock* clock = Clock::GetRealTimeClock();
+
+  RTPSender rtp_sender(false,     // bool audio
+                       clock,     // Clock* clock
+                       nullptr,   // Transport*
+                       nullptr,   // RtpAudioFeedback*
+                       nullptr,   // PacedSender*
+                       nullptr,   // PacketRouter*
+                       nullptr,   // SendTimeObserver*
+                       nullptr,   // BitrateStatisticsObserver*
+                       nullptr,   // FrameCountObserver*
+                       nullptr);  // SendSideDelayObserver*
+
+  std::vector<uint32_t> csrcs;
+  for (unsigned i = 0; i < csrcs_count; i++) {
+    csrcs.push_back(prng->Rand<uint32_t>());
+  }
+  rtp_sender.SetCsrcs(csrcs);
+  rtp_sender.SetSSRC(prng->Rand<uint32_t>());
+  rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
+  rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
+
+  for (unsigned i = 0; i < kNumExtensions; i++) {
+    if (extensions_bitvector & (1u << i)) {
+      rtp_sender.RegisterRtpHeaderExtension(kExtensionTypes[i], i + 1);
+    }
+  }
+
+  int8_t payload_type = prng->Rand(0, 127);
+  bool marker_bit = prng->Rand<bool>();
+  uint32_t capture_timestamp = prng->Rand<uint32_t>();
+  int64_t capture_time_ms = prng->Rand<uint32_t>();
+  bool timestamp_provided = prng->Rand<bool>();
+  bool inc_sequence_number = prng->Rand<bool>();
+
+  size_t header_size = rtp_sender.BuildRTPheader(
+      packet, payload_type, marker_bit, capture_timestamp, capture_time_ms,
+      timestamp_provided, inc_sequence_number);
+
+  for (size_t i = header_size; i < packet_size; i++) {
+    packet[i] = prng->Rand<uint8_t>();
+  }
+
+  return header_size;
+}
+
+rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(Random* prng) {
+  rtcp::ReportBlock report_block;
+  report_block.To(prng->Rand<uint32_t>());  // Remote SSRC.
+  report_block.WithFractionLost(prng->Rand(50));
+
+  rtcp::SenderReport sender_report;
+  sender_report.From(prng->Rand<uint32_t>());  // Sender SSRC.
+  sender_report.WithNtpSec(prng->Rand<uint32_t>());
+  sender_report.WithNtpFrac(prng->Rand<uint32_t>());
+  sender_report.WithPacketCount(prng->Rand<uint32_t>());
+  sender_report.WithReportBlock(report_block);
+
+  return sender_report.Build();
+}
+
+void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
+                                VideoReceiveStream::Config* config,
+                                Random* prng) {
+  // Create a map from a payload type to an encoder name.
+  VideoReceiveStream::Decoder decoder;
+  decoder.payload_type = prng->Rand(0, 127);
+  decoder.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
+  config->decoders.push_back(decoder);
+  // Add SSRCs for the stream.
+  config->rtp.remote_ssrc = prng->Rand<uint32_t>();
+  config->rtp.local_ssrc = prng->Rand<uint32_t>();
+  // Add extensions and settings for RTCP.
+  config->rtp.rtcp_mode =
+      prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
+  config->rtp.remb = prng->Rand<bool>();
+  // Add a map from a payload type to a new ssrc and a new payload type for RTX.
+  VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
+  rtx_pair.ssrc = prng->Rand<uint32_t>();
+  rtx_pair.payload_type = prng->Rand(0, 127);
+  config->rtp.rtx.insert(std::make_pair(prng->Rand(0, 127), rtx_pair));
+  // Add header extensions.
+  for (unsigned i = 0; i < kNumExtensions; i++) {
+    if (extensions_bitvector & (1u << i)) {
+      config->rtp.extensions.push_back(
+          RtpExtension(kExtensionNames[i], prng->Rand<int>()));
+    }
+  }
+}
+
+void GenerateVideoSendConfig(uint32_t extensions_bitvector,
+                             VideoSendStream::Config* config,
+                             Random* prng) {
+  // Create a map from a payload type to an encoder name.
+  config->encoder_settings.payload_type = prng->Rand(0, 127);
+  config->encoder_settings.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
+  // Add SSRCs for the stream.
+  config->rtp.ssrcs.push_back(prng->Rand<uint32_t>());
+  // Add a map from a payload type to new ssrcs and a new payload type for RTX.
+  config->rtp.rtx.ssrcs.push_back(prng->Rand<uint32_t>());
+  config->rtp.rtx.payload_type = prng->Rand(0, 127);
+  // Add header extensions.
+  for (unsigned i = 0; i < kNumExtensions; i++) {
+    if (extensions_bitvector & (1u << i)) {
+      config->rtp.extensions.push_back(
+          RtpExtension(kExtensionNames[i], prng->Rand<int>()));
+    }
+  }
+}
+
+// Test for the RtcEventLog class. Dumps some RTP packets and other events
+// to disk, then reads them back to see if they match.
+void LogSessionAndReadBack(size_t rtp_count,
+                           size_t rtcp_count,
+                           size_t playout_count,
+                           size_t bwe_loss_count,
+                           uint32_t extensions_bitvector,
+                           uint32_t csrcs_count,
+                           unsigned int random_seed) {
+  ASSERT_LE(rtcp_count, rtp_count);
+  ASSERT_LE(playout_count, rtp_count);
+  ASSERT_LE(bwe_loss_count, rtp_count);
+  std::vector<rtc::Buffer> rtp_packets;
+  std::vector<rtc::scoped_ptr<rtcp::RawPacket> > rtcp_packets;
+  std::vector<size_t> rtp_header_sizes;
+  std::vector<uint32_t> playout_ssrcs;
+  std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
+
+  VideoReceiveStream::Config receiver_config(nullptr);
+  VideoSendStream::Config sender_config(nullptr);
+
+  Random prng(random_seed);
+
+  // Create rtp_count RTP packets containing random data.
+  for (size_t i = 0; i < rtp_count; i++) {
+    size_t packet_size = prng.Rand(1000, 1100);
+    rtp_packets.push_back(rtc::Buffer(packet_size));
+    size_t header_size =
+        GenerateRtpPacket(extensions_bitvector, csrcs_count,
+                          rtp_packets[i].data(), packet_size, &prng);
+    rtp_header_sizes.push_back(header_size);
+  }
+  // Create rtcp_count RTCP packets containing random data.
+  for (size_t i = 0; i < rtcp_count; i++) {
+    rtcp_packets.push_back(GenerateRtcpPacket(&prng));
+  }
+  // Create playout_count random SSRCs to use when logging AudioPlayout events.
+  for (size_t i = 0; i < playout_count; i++) {
+    playout_ssrcs.push_back(prng.Rand<uint32_t>());
+  }
+  // Create bwe_loss_count random bitrate updates for BwePacketLoss.
+  for (size_t i = 0; i < bwe_loss_count; i++) {
+    bwe_loss_updates.push_back(
+        std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>()));
+  }
+  // Create configurations for the video streams.
+  GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
+  GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
+  const int config_count = 2;
+
+  // Find the name of the current test, in order to use it as a temporary
+  // filename.
+  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
+  const std::string temp_filename =
+      test::OutputPath() + test_info->test_case_name() + test_info->name();
+
+  // When log_dumper goes out of scope, it causes the log file to be flushed
+  // to disk.
+  {
+    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    log_dumper->LogVideoReceiveStreamConfig(receiver_config);
+    log_dumper->LogVideoSendStreamConfig(sender_config);
+    size_t rtcp_index = 1;
+    size_t playout_index = 1;
+    size_t bwe_loss_index = 1;
+    for (size_t i = 1; i <= rtp_count; i++) {
+      log_dumper->LogRtpHeader(
+          (i % 2 == 0),  // Every second packet is incoming.
+          (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+          rtp_packets[i - 1].data(), rtp_packets[i - 1].size());
+      if (i * rtcp_count >= rtcp_index * rtp_count) {
+        log_dumper->LogRtcpPacket(
+            rtcp_index % 2 == 0,  // Every second packet is incoming
+            rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
+            rtcp_packets[rtcp_index - 1]->Buffer(),
+            rtcp_packets[rtcp_index - 1]->Length());
+        rtcp_index++;
+      }
+      if (i * playout_count >= playout_index * rtp_count) {
+        log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
+        playout_index++;
+      }
+      if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
+        log_dumper->LogBwePacketLossEvent(
+            bwe_loss_updates[bwe_loss_index - 1].first,
+            bwe_loss_updates[bwe_loss_index - 1].second, i);
+        bwe_loss_index++;
+      }
+      if (i == rtp_count / 2) {
+        log_dumper->StartLogging(temp_filename, 10000000);
+      }
+    }
+  }
+
+  // Read the generated file from disk.
+  rtclog::EventStream parsed_stream;
+
+  ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+
+  // Verify that what we read back from the event log is the same as
+  // what we wrote down. For RTCP we log the full packets, but for
+  // RTP we should only log the header.
+  const int event_count = config_count + playout_count + bwe_loss_count +
+                          rtcp_count + rtp_count + 1;
+  EXPECT_EQ(event_count, parsed_stream.stream_size());
+  VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
+  VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
+  size_t event_index = config_count;
+  size_t rtcp_index = 1;
+  size_t playout_index = 1;
+  size_t bwe_loss_index = 1;
+  for (size_t i = 1; i <= rtp_count; i++) {
+    VerifyRtpEvent(parsed_stream.stream(event_index),
+                   (i % 2 == 0),  // Every second packet is incoming.
+                   (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+                   rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
+                   rtp_packets[i - 1].size());
+    event_index++;
+    if (i * rtcp_count >= rtcp_index * rtp_count) {
+      VerifyRtcpEvent(parsed_stream.stream(event_index),
+                      rtcp_index % 2 == 0,  // Every second packet is incoming.
+                      rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
+                      rtcp_packets[rtcp_index - 1]->Buffer(),
+                      rtcp_packets[rtcp_index - 1]->Length());
+      event_index++;
+      rtcp_index++;
+    }
+    if (i * playout_count >= playout_index * rtp_count) {
+      VerifyPlayoutEvent(parsed_stream.stream(event_index),
+                         playout_ssrcs[playout_index - 1]);
+      event_index++;
+      playout_index++;
+    }
+    if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
+      VerifyBweLossEvent(parsed_stream.stream(event_index),
+                         bwe_loss_updates[bwe_loss_index - 1].first,
+                         bwe_loss_updates[bwe_loss_index - 1].second, i);
+      event_index++;
+      bwe_loss_index++;
+    }
+    if (i == rtp_count / 2) {
+      VerifyLogStartEvent(parsed_stream.stream(event_index));
+      event_index++;
+    }
+  }
+
+  // Clean up temporary file - can be pretty slow.
+  remove(temp_filename.c_str());
+}
+
+TEST(RtcEventLogTest, LogSessionAndReadBack) {
+  // Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events
+  // with no header extensions or CSRCS.
+  LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321);
+
+  // Enable AbsSendTime and TransportSequenceNumbers.
+  uint32_t extensions = 0;
+  for (uint32_t i = 0; i < kNumExtensions; i++) {
+    if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime ||
+        kExtensionTypes[i] ==
+            RTPExtensionType::kRtpExtensionTransportSequenceNumber) {
+      extensions |= 1u << i;
+    }
+  }
+  LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u);
+
+  extensions = (1u << kNumExtensions) - 1;  // Enable all header extensions.
+  LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u);
+
+  // Try all combinations of header extensions and up to 2 CSRCS.
+  for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) {
+    for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) {
+      LogSessionAndReadBack(5 + extensions,   // Number of RTP packets.
+                            2 + csrcs_count,  // Number of RTCP packets.
+                            3 + csrcs_count,  // Number of playout events.
+                            1 + csrcs_count,  // Number of BWE loss events.
+                            extensions,       // Bit vector choosing extensions.
+                            csrcs_count,      // Number of contributing sources.
+                            extensions * 3 + csrcs_count + 1);  // Random seed.
+    }
+  }
+}
+
+// Tests that the event queue works correctly, i.e. drops old RTP, RTCP and
+// debug events, but keeps config events even if they are older than the limit.
+void DropOldEvents(uint32_t extensions_bitvector,
+                   uint32_t csrcs_count,
+                   unsigned int random_seed) {
+  rtc::Buffer old_rtp_packet;
+  rtc::Buffer recent_rtp_packet;
+  rtc::scoped_ptr<rtcp::RawPacket> old_rtcp_packet;
+  rtc::scoped_ptr<rtcp::RawPacket> recent_rtcp_packet;
+
+  VideoReceiveStream::Config receiver_config(nullptr);
+  VideoSendStream::Config sender_config(nullptr);
+
+  Random prng(random_seed);
+
+  // Create two RTP packets containing random data.
+  size_t packet_size = prng.Rand(1000, 1100);
+  old_rtp_packet.SetSize(packet_size);
+  GenerateRtpPacket(extensions_bitvector, csrcs_count, old_rtp_packet.data(),
+                    packet_size, &prng);
+  packet_size = prng.Rand(1000, 1100);
+  recent_rtp_packet.SetSize(packet_size);
+  size_t recent_header_size =
+      GenerateRtpPacket(extensions_bitvector, csrcs_count,
+                        recent_rtp_packet.data(), packet_size, &prng);
+
+  // Create two RTCP packets containing random data.
+  old_rtcp_packet = GenerateRtcpPacket(&prng);
+  recent_rtcp_packet = GenerateRtcpPacket(&prng);
+
+  // Create configurations for the video streams.
+  GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
+  GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
+
+  // Find the name of the current test, in order to use it as a temporary
+  // filename.
+  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
+  const std::string temp_filename =
+      test::OutputPath() + test_info->test_case_name() + test_info->name();
+
+  // The log file will be flushed to disk when the log_dumper goes out of scope.
+  {
+    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    // Reduce the time old events are stored to 50 ms.
+    log_dumper->SetBufferDuration(50000);
+    log_dumper->LogVideoReceiveStreamConfig(receiver_config);
+    log_dumper->LogVideoSendStreamConfig(sender_config);
+    log_dumper->LogRtpHeader(false, MediaType::AUDIO, old_rtp_packet.data(),
+                             old_rtp_packet.size());
+    log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet->Buffer(),
+                              old_rtcp_packet->Length());
+    // Sleep 55 ms to let old events be removed from the queue.
+    rtc::Thread::SleepMs(55);
+    log_dumper->StartLogging(temp_filename, 10000000);
+    log_dumper->LogRtpHeader(true, MediaType::VIDEO, recent_rtp_packet.data(),
+                             recent_rtp_packet.size());
+    log_dumper->LogRtcpPacket(false, MediaType::VIDEO,
+                              recent_rtcp_packet->Buffer(),
+                              recent_rtcp_packet->Length());
+  }
+
+  // Read the generated file from disk.
+  rtclog::EventStream parsed_stream;
+  ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+
+  // Verify that what we read back from the event log is the same as
+  // what we wrote. Old RTP and RTCP events should have been discarded,
+  // but old configuration events should still be available.
+  EXPECT_EQ(5, parsed_stream.stream_size());
+  VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
+  VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
+  VerifyLogStartEvent(parsed_stream.stream(2));
+  VerifyRtpEvent(parsed_stream.stream(3), true, MediaType::VIDEO,
+                 recent_rtp_packet.data(), recent_header_size,
+                 recent_rtp_packet.size());
+  VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO,
+                  recent_rtcp_packet->Buffer(), recent_rtcp_packet->Length());
+
+  // Clean up temporary file - can be pretty slow.
+  remove(temp_filename.c_str());
+}
+
+TEST(RtcEventLogTest, DropOldEvents) {
+  // Enable all header extensions
+  uint32_t extensions = (1u << kNumExtensions) - 1;
+  uint32_t csrcs_count = 2;
+  DropOldEvents(extensions, csrcs_count, 141421356);
+  DropOldEvents(extensions, csrcs_count, 173205080);
+}
+
+}  // namespace webrtc
+
+#endif  // ENABLE_RTC_EVENT_LOG
diff --git a/media/webrtc/trunk/webrtc/video/transport_adapter.cc b/media/webrtc/trunk/webrtc/call/transport_adapter.cc
similarity index 51%
rename from media/webrtc/trunk/webrtc/video/transport_adapter.cc
rename to media/webrtc/trunk/webrtc/call/transport_adapter.cc
index 7b5a6962c4..5e59b7b700 100644
--- a/media/webrtc/trunk/webrtc/video/transport_adapter.cc
+++ b/media/webrtc/trunk/webrtc/call/transport_adapter.cc
@@ -8,34 +8,32 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video/transport_adapter.h"
+#include "webrtc/call/transport_adapter.h"
+
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 namespace internal {
 
-TransportAdapter::TransportAdapter(newapi::Transport* transport)
-    : transport_(transport), enabled_(0) {}
-
-int TransportAdapter::SendPacket(int /*channel*/,
-                                 const void* packet,
-                                 size_t length) {
-  if (enabled_.Value() == 0)
-    return false;
-
-  bool success = transport_->SendRtp(static_cast<const uint8_t*>(packet),
-                                     length);
-  return success ? static_cast<int>(length) : -1;
+TransportAdapter::TransportAdapter(Transport* transport)
+    : transport_(transport), enabled_(0) {
+  RTC_DCHECK(nullptr != transport);
 }
 
-int TransportAdapter::SendRTCPPacket(int /*channel*/,
-                                     const void* packet,
-                                     size_t length) {
+bool TransportAdapter::SendRtp(const uint8_t* packet,
+                               size_t length,
+                               const PacketOptions& options) {
   if (enabled_.Value() == 0)
     return false;
 
-  bool success = transport_->SendRtcp(static_cast<const uint8_t*>(packet),
-                                      length);
-  return success ? static_cast<int>(length) : -1;
+  return transport_->SendRtp(packet, length, options);
+}
+
+bool TransportAdapter::SendRtcp(const uint8_t* packet, size_t length) {
+  if (enabled_.Value() == 0)
+    return false;
+
+  return transport_->SendRtcp(packet, length);
 }
 
 void TransportAdapter::Enable() {
diff --git a/media/webrtc/trunk/webrtc/video/transport_adapter.h b/media/webrtc/trunk/webrtc/call/transport_adapter.h
similarity index 54%
rename from media/webrtc/trunk/webrtc/video/transport_adapter.h
rename to media/webrtc/trunk/webrtc/call/transport_adapter.h
index cd27d7cfe0..583cdf9585 100644
--- a/media/webrtc/trunk/webrtc/video/transport_adapter.h
+++ b/media/webrtc/trunk/webrtc/call/transport_adapter.h
@@ -7,33 +7,33 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
-#define WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
+#ifndef WEBRTC_CALL_TRANSPORT_ADAPTER_H_
+#define WEBRTC_CALL_TRANSPORT_ADAPTER_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 namespace internal {
 
-class TransportAdapter : public webrtc::Transport {
+class TransportAdapter : public Transport {
  public:
-  explicit TransportAdapter(newapi::Transport* transport);
+  explicit TransportAdapter(Transport* transport);
 
-  int SendPacket(int /*channel*/, const void* packet, size_t length) override;
-  int SendRTCPPacket(int /*channel*/,
-                     const void* packet,
-                     size_t length) override;
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
 
   void Enable();
   void Disable();
 
  private:
-  newapi::Transport *transport_;
+  Transport *transport_;
   Atomic32 enabled_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
+#endif  // WEBRTC_CALL_TRANSPORT_ADAPTER_H_
diff --git a/media/webrtc/trunk/webrtc/call/webrtc_call.gypi b/media/webrtc/trunk/webrtc/call/webrtc_call.gypi
new file mode 100644
index 0000000000..0c3efff43a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/call/webrtc_call.gypi
@@ -0,0 +1,24 @@
+# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'variables': {
+    'webrtc_call_dependencies': [
+      '<(webrtc_root)/common.gyp:webrtc_common',
+      '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+    ],
+    'webrtc_call_sources': [
+      'call/bitrate_allocator.cc',
+      'call/call.cc',
+      'call/congestion_controller.cc',
+      'call/transport_adapter.cc',
+      'call/transport_adapter.h',
+    ],
+  },
+}
diff --git a/media/webrtc/trunk/webrtc/codereview.settings b/media/webrtc/trunk/webrtc/codereview.settings
index 607eadfe09..c441cc61bc 100644
--- a/media/webrtc/trunk/webrtc/codereview.settings
+++ b/media/webrtc/trunk/webrtc/codereview.settings
@@ -1,9 +1,5 @@
-# This file is used by gcl to get repository specific information.
-CODE_REVIEW_SERVER: webrtc-codereview.appspot.com
-CC_LIST: webrtc-reviews@webrtc.org
-VIEW_VC: http://code.google.com/p/webrtc/source/detail?r=
-TRY_ON_UPLOAD: False
-TRYSERVER_SVN_URL: svn://svn.chromium.org/chrome-try/try-webrtc
-TRYSERVER_ROOT: src/webrtc
-PROJECT: webrtc
-FORCE_HTTPS_COMMIT_URL: True
+
+Creating CLs from this location is not supported! Please make sure the current
+working directory is the parent directory of this directory.
+If you're working with a Chromium checkout, you'll have to create a full WebRTC
+checkout and upload a CL from that. See http://www.webrtc.org for instructions.
diff --git a/media/webrtc/trunk/webrtc/common.h b/media/webrtc/trunk/webrtc/common.h
index 6ead4098d3..8789243000 100644
--- a/media/webrtc/trunk/webrtc/common.h
+++ b/media/webrtc/trunk/webrtc/common.h
@@ -13,8 +13,28 @@
 
 #include <map>
 
+#include "webrtc/base/basictypes.h"
+
 namespace webrtc {
 
+// Only add new values to the end of the enumeration and never remove (only
+// deprecate) to maintain binary compatibility.
+enum class ConfigOptionID {
+  kMyExperimentForTest,
+  kAlgo1CostFunctionForTest,
+  kTemporalLayersFactory,
+  kNetEqCapacityConfig,
+  kNetEqFastAccelerate,
+  kVoicePacing,
+  kExtendedFilter,
+  kDelayAgnostic,
+  kExperimentalAgc,
+  kExperimentalNs,
+  kBeamforming,
+  kIntelligibility,
+  kCaptureDeviceInfo
+};
+
 // Class Config is designed to ease passing a set of options across webrtc code.
 // Options are identified by typename in order to avoid incorrect casts.
 //
@@ -59,8 +79,6 @@ class Config {
   }
 
  private:
-  typedef void* OptionIdentifier;
-
   struct BaseOption {
     virtual ~BaseOption() {}
   };
@@ -74,11 +92,9 @@ class Config {
     T* value;
   };
 
-  // Own implementation of rtti-subset to avoid depending on rtti and its costs.
   template<typename T>
-  static OptionIdentifier identifier() {
-    static char id_placeholder;
-    return &id_placeholder;
+  static ConfigOptionID identifier() {
+    return T::identifier;
   }
 
   // Used to instantiate a default constructed object that doesn't needs to be
@@ -86,14 +102,14 @@ class Config {
   // locks.
   template<typename T>
   static const T& default_value() {
-    static const T def;
+    RTC_DEFINE_STATIC_LOCAL(const T, def, ());
     return def;
   }
 
-  typedef std::map<OptionIdentifier, BaseOption*> OptionMap;
+  typedef std::map<ConfigOptionID, BaseOption*> OptionMap;
   OptionMap options_;
 
-  // DISALLOW_COPY_AND_ASSIGN
+  // RTC_DISALLOW_COPY_AND_ASSIGN
   Config(const Config&);
   void operator=(const Config&);
 };
diff --git a/media/webrtc/trunk/webrtc/common_audio/BUILD.gn b/media/webrtc/trunk/webrtc/common_audio/BUILD.gn
index 34e1575b24..b4ec1d71ef 100644
--- a/media/webrtc/trunk/webrtc/common_audio/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/common_audio/BUILD.gn
@@ -51,9 +51,6 @@ source_set("common_audio") {
     "resampler/sinc_resampler.h",
     "ring_buffer.c",
     "ring_buffer.h",
-    "signal_processing/include/real_fft.h",
-    "signal_processing/include/signal_processing_library.h",
-    "signal_processing/include/spl_inl.h",
     "signal_processing/auto_corr_to_refl_coef.c",
     "signal_processing/auto_correlation.c",
     "signal_processing/complex_fft_tables.h",
@@ -68,12 +65,15 @@ source_set("common_audio") {
     "signal_processing/get_hanning_window.c",
     "signal_processing/get_scaling_square.c",
     "signal_processing/ilbc_specific_functions.c",
+    "signal_processing/include/real_fft.h",
+    "signal_processing/include/signal_processing_library.h",
+    "signal_processing/include/spl_inl.h",
     "signal_processing/levinson_durbin.c",
     "signal_processing/lpc_to_refl_coef.c",
     "signal_processing/min_max_operations.c",
     "signal_processing/randomization_functions.c",
-    "signal_processing/refl_coef_to_lpc.c",
     "signal_processing/real_fft.c",
+    "signal_processing/refl_coef_to_lpc.c",
     "signal_processing/resample.c",
     "signal_processing/resample_48khz.c",
     "signal_processing/resample_by_2.c",
@@ -85,10 +85,12 @@ source_set("common_audio") {
     "signal_processing/splitting_filter.c",
     "signal_processing/sqrt_of_one_minus_x_squared.c",
     "signal_processing/vector_scaling_operations.c",
+    "sparse_fir_filter.cc",
+    "sparse_fir_filter.h",
+    "swap_queue.h",
     "vad/include/vad.h",
     "vad/include/webrtc_vad.h",
     "vad/vad.cc",
-    "vad/webrtc_vad.c",
     "vad/vad_core.c",
     "vad/vad_core.h",
     "vad/vad_filterbank.c",
@@ -97,15 +99,18 @@ source_set("common_audio") {
     "vad/vad_gmm.h",
     "vad/vad_sp.c",
     "vad/vad_sp.h",
-    "wav_header.cc",
-    "wav_header.h",
+    "vad/webrtc_vad.c",
     "wav_file.cc",
     "wav_file.h",
+    "wav_header.cc",
+    "wav_header.h",
     "window_generator.cc",
     "window_generator.h",
   ]
 
-  deps = [ "../system_wrappers" ]
+  deps = [
+    "../system_wrappers",
+  ]
 
   defines = []
   if (rtc_use_openmax_dl) {
@@ -114,7 +119,9 @@ source_set("common_audio") {
       "real_fourier_openmax.h",
     ]
     defines += [ "RTC_USE_OPENMAX_DL" ]
-    deps += [ "//third_party/openmax_dl/dl" ]
+    if (rtc_build_openmax_dl) {
+      deps += [ "//third_party/openmax_dl/dl" ]
+    }
   }
 
   if (current_cpu == "arm") {
@@ -124,25 +131,24 @@ source_set("common_audio") {
     ]
 
     if (arm_version >= 7) {
-      deps += [ ":common_audio_neon" ]
       sources += [ "signal_processing/filter_ar_fast_q12_armv7.S" ]
     } else {
       sources += [ "signal_processing/filter_ar_fast_q12.c" ]
     }
   }
 
-  if (current_cpu == "arm64") {
+  if (rtc_build_with_neon) {
     deps += [ ":common_audio_neon" ]
   }
 
   if (current_cpu == "mipsel") {
     sources += [
-      "signal_processing/include/spl_inl_mips.h",
       "signal_processing/complex_bit_reverse_mips.c",
       "signal_processing/complex_fft_mips.c",
       "signal_processing/cross_correlation_mips.c",
       "signal_processing/downsample_fast_mips.c",
       "signal_processing/filter_ar_fast_q12_mips.c",
+      "signal_processing/include/spl_inl_mips.h",
       "signal_processing/min_max_operations_mips.c",
       "signal_processing/resample_by_2_mips.c",
       "signal_processing/spl_sqrt_floor_mips.c",
@@ -163,9 +169,7 @@ source_set("common_audio") {
   }
 
   if (is_win) {
-    cflags = [
-      "/wd4334",  # Ignore warning on shift operator promotion.
-    ]
+    cflags = [ "/wd4334" ]  # Ignore warning on shift operator promotion.
   }
 
   configs += [ "..:common_config" ]
@@ -193,7 +197,9 @@ if (current_cpu == "x86" || current_cpu == "x64") {
       "resampler/sinc_resampler_sse.cc",
     ]
 
-    cflags = [ "-msse2" ]
+    if (is_posix) {
+      cflags = [ "-msse2" ]
+    }
 
     configs += [ "..:common_inherited_config" ]
 
@@ -205,7 +211,7 @@ if (current_cpu == "x86" || current_cpu == "x64") {
   }
 }
 
-if (rtc_build_armv7_neon || current_cpu == "arm64") {
+if (rtc_build_with_neon) {
   source_set("common_audio_neon") {
     sources = [
       "fir_filter_neon.cc",
@@ -215,20 +221,24 @@ if (rtc_build_armv7_neon || current_cpu == "arm64") {
       "signal_processing/min_max_operations_neon.c",
     ]
 
-    configs += [ "..:common_config" ]
-    public_configs = [ "..:common_inherited_config" ]
-
-    if (!arm_use_neon) {
+    if (current_cpu != "arm64") {
+      # Enable compilation for the NEON instruction set. This is needed
+      # since //build/config/arm.gni only enables NEON for iOS, not Android.
+      # This provides the same functionality as webrtc/build/arm_neon.gypi.
       configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
       cflags = [ "-mfpu=neon" ]
     }
 
-    # Disable LTO in audio_processing_neon target due to compiler bug.
+    # Disable LTO on NEON targets due to compiler bug.
+    # TODO(fdegans): Enable this. See crbug.com/408997.
     if (rtc_use_lto) {
       cflags -= [
         "-flto",
         "-ffat-lto-objects",
       ]
     }
+
+    configs += [ "..:common_config" ]
+    public_configs = [ "..:common_inherited_config" ]
   }
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/OWNERS b/media/webrtc/trunk/webrtc/common_audio/OWNERS
index 98ec4728b5..208a7c5635 100644
--- a/media/webrtc/trunk/webrtc/common_audio/OWNERS
+++ b/media/webrtc/trunk/webrtc/common_audio/OWNERS
@@ -1,7 +1,7 @@
-bjornv@webrtc.org
-tina.legrand@webrtc.org
+henrik.lundin@webrtc.org
 jan.skoglund@webrtc.org
-andrew@webrtc.org
+kwiberg@webrtc.org
+tina.legrand@webrtc.org
 
 per-file *.isolate=kjellander@webrtc.org
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_converter.cc b/media/webrtc/trunk/webrtc/common_audio/audio_converter.cc
index 7e043b77e0..9ebfabc286 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_converter.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_converter.cc
@@ -11,12 +11,13 @@
 #include "webrtc/common_audio/audio_converter.h"
 
 #include <cstring>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
 
 using rtc::checked_cast;
 
@@ -24,8 +25,8 @@ namespace webrtc {
 
 class CopyConverter : public AudioConverter {
  public:
-  CopyConverter(int src_channels, int src_frames, int dst_channels,
-                int dst_frames)
+  CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
+                size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
   ~CopyConverter() override {};
 
@@ -33,7 +34,7 @@ class CopyConverter : public AudioConverter {
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
     if (src != dst) {
-      for (int i = 0; i < src_channels(); ++i)
+      for (size_t i = 0; i < src_channels(); ++i)
         std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
     }
   }
@@ -41,17 +42,17 @@ class CopyConverter : public AudioConverter {
 
 class UpmixConverter : public AudioConverter {
  public:
-  UpmixConverter(int src_channels, int src_frames, int dst_channels,
-                 int dst_frames)
+  UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
+                 size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
   ~UpmixConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
-    for (int i = 0; i < dst_frames(); ++i) {
+    for (size_t i = 0; i < dst_frames(); ++i) {
       const float value = src[0][i];
-      for (int j = 0; j < dst_channels(); ++j)
+      for (size_t j = 0; j < dst_channels(); ++j)
         dst[j][i] = value;
     }
   }
@@ -59,8 +60,8 @@ class UpmixConverter : public AudioConverter {
 
 class DownmixConverter : public AudioConverter {
  public:
-  DownmixConverter(int src_channels, int src_frames, int dst_channels,
-                   int dst_frames)
+  DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
+                   size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
   }
   ~DownmixConverter() override {};
@@ -69,9 +70,9 @@ class DownmixConverter : public AudioConverter {
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
     float* dst_mono = dst[0];
-    for (int i = 0; i < src_frames(); ++i) {
+    for (size_t i = 0; i < src_frames(); ++i) {
       float sum = 0;
-      for (int j = 0; j < src_channels(); ++j)
+      for (size_t j = 0; j < src_channels(); ++j)
         sum += src[j][i];
       dst_mono[i] = sum / src_channels();
     }
@@ -80,11 +81,11 @@ class DownmixConverter : public AudioConverter {
 
 class ResampleConverter : public AudioConverter {
  public:
-  ResampleConverter(int src_channels, int src_frames, int dst_channels,
-                    int dst_frames)
+  ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
+                    size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
     resamplers_.reserve(src_channels);
-    for (int i = 0; i < src_channels; ++i)
+    for (size_t i = 0; i < src_channels; ++i)
       resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
   }
   ~ResampleConverter() override {};
@@ -105,8 +106,8 @@ class ResampleConverter : public AudioConverter {
 class CompositionConverter : public AudioConverter {
  public:
   CompositionConverter(ScopedVector<AudioConverter> converters)
-      : converters_(converters.Pass()) {
-    CHECK_GE(converters_.size(), 2u);
+      : converters_(std::move(converters)) {
+    RTC_CHECK_GE(converters_.size(), 2u);
     // We need an intermediate buffer after every converter.
     for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
       buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
@@ -135,10 +136,10 @@ class CompositionConverter : public AudioConverter {
   ScopedVector<ChannelBuffer<float>> buffers_;
 };
 
-rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
-                                                       int src_frames,
-                                                       int dst_channels,
-                                                       int dst_frames) {
+rtc::scoped_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
+                                                       size_t src_frames,
+                                                       size_t dst_channels,
+                                                       size_t dst_frames) {
   rtc::scoped_ptr<AudioConverter> sp;
   if (src_channels > dst_channels) {
     if (src_frames != dst_frames) {
@@ -147,7 +148,7 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
                                                 dst_channels, src_frames));
       converters.push_back(new ResampleConverter(dst_channels, src_frames,
                                                  dst_channels, dst_frames));
-      sp.reset(new CompositionConverter(converters.Pass()));
+      sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
                                     dst_frames));
@@ -159,7 +160,7 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
                                                  src_channels, dst_frames));
       converters.push_back(new UpmixConverter(src_channels, dst_frames,
                                               dst_channels, dst_frames));
-      sp.reset(new CompositionConverter(converters.Pass()));
+      sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
                                   dst_frames));
@@ -172,7 +173,7 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
                                dst_frames));
   }
 
-  return sp.Pass();
+  return sp;
 }
 
 // For CompositionConverter.
@@ -182,18 +183,19 @@ AudioConverter::AudioConverter()
       dst_channels_(0),
       dst_frames_(0) {}
 
-AudioConverter::AudioConverter(int src_channels, int src_frames,
-                               int dst_channels, int dst_frames)
+AudioConverter::AudioConverter(size_t src_channels, size_t src_frames,
+                               size_t dst_channels, size_t dst_frames)
     : src_channels_(src_channels),
       src_frames_(src_frames),
       dst_channels_(dst_channels),
       dst_frames_(dst_frames) {
-  CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
+  RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
+            src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
-  CHECK_EQ(src_size, checked_cast<size_t>(src_channels() * src_frames()));
-  CHECK_GE(dst_capacity, checked_cast<size_t>(dst_channels() * dst_frames()));
+  RTC_CHECK_EQ(src_size, src_channels() * src_frames());
+  RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_converter.h b/media/webrtc/trunk/webrtc/common_audio/audio_converter.h
index 772872fcd6..c5f08c1d9b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_converter.h
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_converter.h
@@ -26,10 +26,10 @@ class AudioConverter {
  public:
   // Returns a new AudioConverter, which will use the supplied format for its
   // lifetime. Caller is responsible for the memory.
-  static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
-                                                int src_frames,
-                                                int dst_channels,
-                                                int dst_frames);
+  static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
+                                                size_t src_frames,
+                                                size_t dst_channels,
+                                                size_t dst_frames);
   virtual ~AudioConverter() {};
 
   // Convert |src|, containing |src_size| samples, to |dst|, having a sample
@@ -39,26 +39,26 @@ class AudioConverter {
   virtual void Convert(const float* const* src, size_t src_size,
                        float* const* dst, size_t dst_capacity) = 0;
 
-  int src_channels() const { return src_channels_; }
-  int src_frames() const { return src_frames_; }
-  int dst_channels() const { return dst_channels_; }
-  int dst_frames() const { return dst_frames_; }
+  size_t src_channels() const { return src_channels_; }
+  size_t src_frames() const { return src_frames_; }
+  size_t dst_channels() const { return dst_channels_; }
+  size_t dst_frames() const { return dst_frames_; }
 
  protected:
   AudioConverter();
-  AudioConverter(int src_channels, int src_frames, int dst_channels,
-                 int dst_frames);
+  AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
+                 size_t dst_frames);
 
-  // Helper to CHECK that inputs are correctly sized.
+  // Helper to RTC_CHECK that inputs are correctly sized.
   void CheckSizes(size_t src_size, size_t dst_capacity) const;
 
  private:
-  const int src_channels_;
-  const int src_frames_;
-  const int dst_channels_;
-  const int dst_frames_;
+  const size_t src_channels_;
+  const size_t src_frames_;
+  const size_t dst_channels_;
+  const size_t dst_frames_;
 
-  DISALLOW_COPY_AND_ASSIGN(AudioConverter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_converter_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/audio_converter_unittest.cc
index 6da339f08a..dace0bdccf 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_converter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_converter_unittest.cc
@@ -13,6 +13,8 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
@@ -23,11 +25,11 @@ namespace webrtc {
 typedef rtc::scoped_ptr<ChannelBuffer<float>> ScopedBuffer;
 
 // Sets the signal value to increase by |data| with every sample.
-ScopedBuffer CreateBuffer(const std::vector<float>& data, int frames) {
-  const int num_channels = static_cast<int>(data.size());
+ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
+  const size_t num_channels = data.size();
   ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
-  for (int i = 0; i < num_channels; ++i)
-    for (int j = 0; j < frames; ++j)
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < frames; ++j)
       sb->channels()[i][j] = data[i] * j;
   return sb;
 }
@@ -43,20 +45,20 @@ void VerifyParams(const ChannelBuffer<float>& ref,
 // signals to compensate for the resampling delay.
 float ComputeSNR(const ChannelBuffer<float>& ref,
                  const ChannelBuffer<float>& test,
-                 int expected_delay) {
+                 size_t expected_delay) {
   VerifyParams(ref, test);
   float best_snr = 0;
-  int best_delay = 0;
+  size_t best_delay = 0;
 
   // Search within one sample of the expected delay.
-  for (int delay = std::max(expected_delay - 1, 0);
+  for (size_t delay = std::max(expected_delay, static_cast<size_t>(1)) - 1;
        delay <= std::min(expected_delay + 1, ref.num_frames());
        ++delay) {
     float mse = 0;
     float variance = 0;
     float mean = 0;
-    for (int i = 0; i < ref.num_channels(); ++i) {
-      for (int j = 0; j < ref.num_frames() - delay; ++j) {
+    for (size_t i = 0; i < ref.num_channels(); ++i) {
+      for (size_t j = 0; j < ref.num_frames() - delay; ++j) {
         float error = ref.channels()[i][j] - test.channels()[i][j + delay];
         mse += error * error;
         variance += ref.channels()[i][j] * ref.channels()[i][j];
@@ -64,7 +66,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
       }
     }
 
-    const int length = ref.num_channels() * (ref.num_frames() - delay);
+    const size_t length = ref.num_channels() * (ref.num_frames() - delay);
     mse /= length;
     variance /= length;
     mean /= length;
@@ -77,16 +79,16 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
       best_delay = delay;
     }
   }
-  printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
+  printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
   return best_snr;
 }
 
 // Sets the source to a linearly increasing signal for which we can easily
 // generate a reference. Runs the AudioConverter and ensures the output has
 // sufficiently high SNR relative to the reference.
-void RunAudioConverterTest(int src_channels,
+void RunAudioConverterTest(size_t src_channels,
                            int src_sample_rate_hz,
-                           int dst_channels,
+                           size_t dst_channels,
                            int dst_sample_rate_hz) {
   const float kSrcLeft = 0.0002f;
   const float kSrcRight = 0.0001f;
@@ -95,8 +97,8 @@ void RunAudioConverterTest(int src_channels,
   const float dst_left = resampling_factor * kSrcLeft;
   const float dst_right = resampling_factor * kSrcRight;
   const float dst_mono = (dst_left + dst_right) / 2;
-  const int src_frames = src_sample_rate_hz / 100;
-  const int dst_frames = dst_sample_rate_hz / 100;
+  const size_t src_frames = static_cast<size_t>(src_sample_rate_hz / 100);
+  const size_t dst_frames = static_cast<size_t>(dst_sample_rate_hz / 100);
 
   std::vector<float> src_data(1, kSrcLeft);
   if (src_channels == 2)
@@ -122,11 +124,13 @@ void RunAudioConverterTest(int src_channels,
   ScopedBuffer ref_buffer = CreateBuffer(ref_data, dst_frames);
 
   // The sinc resampler has a known delay, which we compute here.
-  const int delay_frames = src_sample_rate_hz == dst_sample_rate_hz ? 0 :
-      PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
-          dst_sample_rate_hz;
-  printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
-      src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
+  const size_t delay_frames = src_sample_rate_hz == dst_sample_rate_hz ? 0 :
+      static_cast<size_t>(
+          PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
+          dst_sample_rate_hz);
+  // SNR reported on the same line later.
+  printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ",
+         src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
 
   rtc::scoped_ptr<AudioConverter> converter = AudioConverter::Create(
       src_channels, src_frames, dst_channels, dst_frames);
@@ -139,13 +143,13 @@ void RunAudioConverterTest(int src_channels,
 
 TEST(AudioConverterTest, ConversionsPassSNRThreshold) {
   const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000};
-  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
-  const int kChannels[] = {1, 2};
-  const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
-  for (int src_rate = 0; src_rate < kSampleRatesSize; ++src_rate) {
-    for (int dst_rate = 0; dst_rate < kSampleRatesSize; ++dst_rate) {
-      for (int src_channel = 0; src_channel < kChannelsSize; ++src_channel) {
-        for (int dst_channel = 0; dst_channel < kChannelsSize; ++dst_channel) {
+  const size_t kChannels[] = {1, 2};
+  for (size_t src_rate = 0; src_rate < arraysize(kSampleRates); ++src_rate) {
+    for (size_t dst_rate = 0; dst_rate < arraysize(kSampleRates); ++dst_rate) {
+      for (size_t src_channel = 0; src_channel < arraysize(kChannels);
+           ++src_channel) {
+        for (size_t dst_channel = 0; dst_channel < arraysize(kChannels);
+             ++dst_channel) {
           RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
                                 kChannels[dst_channel], kSampleRates[dst_rate]);
         }
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.cc b/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.cc
index 0ec53a34b8..a29e53a61c 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.cc
@@ -18,6 +18,7 @@
 namespace webrtc {
 
 AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
+  buffers_.reserve(channels);
   for (size_t i = 0; i < channels; ++i)
     buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
 }
@@ -29,18 +30,19 @@ AudioRingBuffer::~AudioRingBuffer() {
 
 void AudioRingBuffer::Write(const float* const* data, size_t channels,
                             size_t frames) {
-  DCHECK_EQ(buffers_.size(), channels);
+  RTC_DCHECK_EQ(buffers_.size(), channels);
   for (size_t i = 0; i < channels; ++i) {
-    size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
-    CHECK_EQ(written, frames);
+    const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
+    RTC_CHECK_EQ(written, frames);
   }
 }
 
 void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
-  DCHECK_EQ(buffers_.size(), channels);
+  RTC_DCHECK_EQ(buffers_.size(), channels);
   for (size_t i = 0; i < channels; ++i) {
-    size_t read = WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
-    CHECK_EQ(read, frames);
+    const size_t read =
+        WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
+    RTC_CHECK_EQ(read, frames);
   }
 }
 
@@ -54,10 +56,19 @@ size_t AudioRingBuffer::WriteFramesAvailable() const {
   return WebRtc_available_write(buffers_[0]);
 }
 
-void AudioRingBuffer::MoveReadPosition(int frames) {
+void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
   for (auto buf : buffers_) {
-    int moved = WebRtc_MoveReadPtr(buf, frames);
-    CHECK_EQ(moved, frames);
+    const size_t moved =
+        static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
+  for (auto buf : buffers_) {
+    const size_t moved = static_cast<size_t>(
+        -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.h b/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.h
index 60192a1690..58e543adea 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.h
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer.h
@@ -7,9 +7,9 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
 #ifndef WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
 #define WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
+
 #include <stddef.h>
 #include <vector>
 
@@ -27,20 +27,23 @@ class AudioRingBuffer final {
   AudioRingBuffer(size_t channels, size_t max_frames);
   ~AudioRingBuffer();
 
-  // Copy |data| to the buffer and advance the write pointer. |channels| must
+  // Copies |data| to the buffer and advances the write pointer. |channels| must
   // be the same as at creation time.
   void Write(const float* const* data, size_t channels, size_t frames);
 
-  // Copy from the buffer to |data| and advance the read pointer. |channels|
+  // Copies from the buffer to |data| and advances the read pointer. |channels|
   // must be the same as at creation time.
   void Read(float* const* data, size_t channels, size_t frames);
 
   size_t ReadFramesAvailable() const;
   size_t WriteFramesAvailable() const;
 
-  // Positive values advance the read pointer and negative values withdraw
-  // the read pointer (i.e. flush and stuff the buffer respectively.)
-  void MoveReadPosition(int frames);
+  // Moves the read position. The forward version advances the read pointer
+  // towards the write pointer and the backward verison withdraws the read
+  // pointer away from the write pointer (i.e. flushing and stuffing the buffer
+  // respectively.)
+  void MoveReadPositionForward(size_t frames);
+  void MoveReadPositionBackward(size_t frames);
 
  private:
   // We don't use a ScopedVector because it doesn't support a specialized
@@ -49,4 +52,5 @@ class AudioRingBuffer final {
 };
 
 }  // namespace webrtc
-#endif
+
+#endif  // WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer_unittest.cc
index cc1922c69a..a7a6a9442b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_ring_buffer_unittest.cc
@@ -34,26 +34,28 @@ void ReadAndWriteTest(const ChannelBuffer<float>& input,
   while (input_pos + buf.WriteFramesAvailable() < total_frames) {
     // Write until the buffer is as full as possible.
     while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
-      buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)),
-                num_channels, num_write_chunk_frames);
+      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+                num_write_chunk_frames);
       input_pos += num_write_chunk_frames;
     }
     // Read until the buffer is as empty as possible.
     while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
       EXPECT_LT(output_pos, total_frames);
-      buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)),
-               num_channels, num_read_chunk_frames);
+      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+               num_read_chunk_frames);
       output_pos += num_read_chunk_frames;
     }
   }
 
   // Write and read the last bit.
-  if (input_pos < total_frames)
-    buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)),
-              num_channels, total_frames - input_pos);
-  if (buf.ReadFramesAvailable())
-    buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)),
-             num_channels, buf.ReadFramesAvailable());
+  if (input_pos < total_frames) {
+    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+              total_frames - input_pos);
+  }
+  if (buf.ReadFramesAvailable()) {
+    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+             buf.ReadFramesAvailable());
+  }
   EXPECT_EQ(0u, buf.ReadFramesAvailable());
 }
 
@@ -96,11 +98,11 @@ TEST_F(AudioRingBufferTest, MoveReadPosition) {
   AudioRingBuffer buf(kNumChannels, kNumFrames);
   buf.Write(input.channels(), kNumChannels, kNumFrames);
 
-  buf.MoveReadPosition(3);
+  buf.MoveReadPositionForward(3);
   ChannelBuffer<float> output(1, kNumChannels);
   buf.Read(output.channels(), kNumChannels, 1);
   EXPECT_EQ(4, output.channels()[0][0]);
-  buf.MoveReadPosition(-3);
+  buf.MoveReadPositionBackward(3);
   buf.Read(output.channels(), kNumChannels, 1);
   EXPECT_EQ(2, output.channels()[0][0]);
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_util.cc b/media/webrtc/trunk/webrtc/common_audio/audio_util.cc
index 2047295cb9..2ce2eba994 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_util.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_util.cc
@@ -39,4 +39,13 @@ void FloatS16ToFloat(const float* src, size_t size, float* dest) {
     dest[i] = FloatS16ToFloat(src[i]);
 }
 
+template <>
+void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
+                                       size_t num_frames,
+                                       int num_channels,
+                                       int16_t* deinterleaved) {
+  DownmixInterleavedToMonoImpl<int16_t, int32_t>(interleaved, num_frames,
+                                                 num_channels, deinterleaved);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/audio_util_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/audio_util_unittest.cc
index 2cdf53813c..5583778b28 100644
--- a/media/webrtc/trunk/webrtc/common_audio/audio_util_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/audio_util_unittest.cc
@@ -8,38 +8,48 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace {
 
-void ExpectArraysEq(const int16_t* ref, const int16_t* test, int length) {
-  for (int i = 0; i < length; ++i) {
+using ::testing::ElementsAreArray;
+
+void ExpectArraysEq(const int16_t* ref, const int16_t* test, size_t length) {
+  for (size_t i = 0; i < length; ++i) {
     EXPECT_EQ(ref[i], test[i]);
   }
 }
 
-void ExpectArraysEq(const float* ref, const float* test, int length) {
-  for (int i = 0; i < length; ++i) {
+void ExpectArraysEq(const float* ref, const float* test, size_t length) {
+  for (size_t i = 0; i < length; ++i) {
     EXPECT_FLOAT_EQ(ref[i], test[i]);
   }
 }
 
 TEST(AudioUtilTest, FloatToS16) {
-  const int kSize = 9;
-  const float kInput[kSize] = {
-      0.f, 0.4f / 32767.f, 0.6f / 32767.f, -0.4f / 32768.f, -0.6f / 32768.f,
-      1.f, -1.f, 1.1f, -1.1f};
-  const int16_t kReference[kSize] = {
-    0, 0, 1, 0, -1, 32767, -32768, 32767, -32768};
+  const size_t kSize = 9;
+  const float kInput[kSize] = {0.f,
+                               0.4f / 32767.f,
+                               0.6f / 32767.f,
+                               -0.4f / 32768.f,
+                               -0.6f / 32768.f,
+                               1.f,
+                               -1.f,
+                               1.1f,
+                               -1.1f};
+  const int16_t kReference[kSize] = {0,     0,      1,     0,     -1,
+                                     32767, -32768, 32767, -32768};
   int16_t output[kSize];
   FloatToS16(kInput, kSize, output);
   ExpectArraysEq(kReference, output, kSize);
 }
 
 TEST(AudioUtilTest, S16ToFloat) {
-  const int kSize = 7;
+  const size_t kSize = 7;
   const int16_t kInput[kSize] = {0, 1, -1, 16384, -16384, 32767, -32768};
   const float kReference[kSize] = {
       0.f, 1.f / 32767.f, -1.f / 32768.f, 16384.f / 32767.f, -0.5f, 1.f, -1.f};
@@ -49,9 +59,9 @@ TEST(AudioUtilTest, S16ToFloat) {
 }
 
 TEST(AudioUtilTest, FloatS16ToS16) {
-  const int kSize = 7;
-  const float kInput[kSize] = {
-      0.f, 0.4f, 0.5f, -0.4f, -0.5f, 32768.f, -32769.f};
+  const size_t kSize = 7;
+  const float kInput[kSize] = {0.f,   0.4f,    0.5f,    -0.4f,
+                               -0.5f, 32768.f, -32769.f};
   const int16_t kReference[kSize] = {0, 0, 1, 0, -1, 32767, -32768};
   int16_t output[kSize];
   FloatS16ToS16(kInput, kSize, output);
@@ -59,24 +69,36 @@ TEST(AudioUtilTest, FloatS16ToS16) {
 }
 
 TEST(AudioUtilTest, FloatToFloatS16) {
-  const int kSize = 9;
-  const float kInput[kSize] = {
-      0.f, 0.4f / 32767.f, 0.6f / 32767.f, -0.4f / 32768.f, -0.6f / 32768.f,
-      1.f, -1.f, 1.1f, -1.1f};
-  const float kReference[kSize] = {
-    0.f, 0.4f, 0.6f, -0.4f, -0.6f, 32767.f, -32768.f, 36043.7f, -36044.8f};
+  const size_t kSize = 9;
+  const float kInput[kSize] = {0.f,
+                               0.4f / 32767.f,
+                               0.6f / 32767.f,
+                               -0.4f / 32768.f,
+                               -0.6f / 32768.f,
+                               1.f,
+                               -1.f,
+                               1.1f,
+                               -1.1f};
+  const float kReference[kSize] = {0.f,     0.4f,     0.6f,     -0.4f,    -0.6f,
+                                   32767.f, -32768.f, 36043.7f, -36044.8f};
   float output[kSize];
   FloatToFloatS16(kInput, kSize, output);
   ExpectArraysEq(kReference, output, kSize);
 }
 
 TEST(AudioUtilTest, FloatS16ToFloat) {
-  const int kSize = 9;
-  const float kInput[kSize] = {
-    0.f, 0.4f, 0.6f, -0.4f, -0.6f, 32767.f, -32768.f, 36043.7f, -36044.8f};
-  const float kReference[kSize] = {
-      0.f, 0.4f / 32767.f, 0.6f / 32767.f, -0.4f / 32768.f, -0.6f / 32768.f,
-      1.f, -1.f, 1.1f, -1.1f};
+  const size_t kSize = 9;
+  const float kInput[kSize] = {0.f,     0.4f,     0.6f,     -0.4f,    -0.6f,
+                               32767.f, -32768.f, 36043.7f, -36044.8f};
+  const float kReference[kSize] = {0.f,
+                                   0.4f / 32767.f,
+                                   0.6f / 32767.f,
+                                   -0.4f / 32768.f,
+                                   -0.6f / 32768.f,
+                                   1.f,
+                                   -1.f,
+                                   1.1f,
+                                   -1.1f};
   float output[kSize];
   FloatS16ToFloat(kInput, kSize, output);
   ExpectArraysEq(kReference, output, kSize);
@@ -84,9 +106,9 @@ TEST(AudioUtilTest, FloatS16ToFloat) {
 
 TEST(AudioUtilTest, InterleavingStereo) {
   const int16_t kInterleaved[] = {2, 3, 4, 9, 8, 27, 16, 81};
-  const int kSamplesPerChannel = 4;
+  const size_t kSamplesPerChannel = 4;
   const int kNumChannels = 2;
-  const int kLength = kSamplesPerChannel * kNumChannels;
+  const size_t kLength = kSamplesPerChannel * kNumChannels;
   int16_t left[kSamplesPerChannel], right[kSamplesPerChannel];
   int16_t* deinterleaved[] = {left, right};
   Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
@@ -102,7 +124,7 @@ TEST(AudioUtilTest, InterleavingStereo) {
 
 TEST(AudioUtilTest, InterleavingMonoIsIdentical) {
   const int16_t kInterleaved[] = {1, 2, 3, 4, 5};
-  const int kSamplesPerChannel = 5;
+  const size_t kSamplesPerChannel = 5;
   const int kNumChannels = 1;
   int16_t mono[kSamplesPerChannel];
   int16_t* deinterleaved[] = {mono};
@@ -114,4 +136,96 @@ TEST(AudioUtilTest, InterleavingMonoIsIdentical) {
   ExpectArraysEq(mono, interleaved, kSamplesPerChannel);
 }
 
+TEST(AudioUtilTest, DownmixInterleavedToMono) {
+  {
+    const size_t kNumFrames = 4;
+    const int kNumChannels = 1;
+    const int16_t interleaved[kNumChannels * kNumFrames] = {1, 2, -1, -3};
+    int16_t deinterleaved[kNumFrames];
+
+    DownmixInterleavedToMono(interleaved, kNumFrames, kNumChannels,
+                             deinterleaved);
+
+    EXPECT_THAT(deinterleaved, ElementsAreArray(interleaved));
+  }
+  {
+    const size_t kNumFrames = 2;
+    const int kNumChannels = 2;
+    const int16_t interleaved[kNumChannels * kNumFrames] = {10, 20, -10, -30};
+    int16_t deinterleaved[kNumFrames];
+
+    DownmixInterleavedToMono(interleaved, kNumFrames, kNumChannels,
+                             deinterleaved);
+    const int16_t expected[kNumFrames] = {15, -20};
+
+    EXPECT_THAT(deinterleaved, ElementsAreArray(expected));
+  }
+  {
+    const size_t kNumFrames = 3;
+    const int kNumChannels = 3;
+    const int16_t interleaved[kNumChannels * kNumFrames] = {
+        30000, 30000, 24001, -5, -10, -20, -30000, -30999, -30000};
+    int16_t deinterleaved[kNumFrames];
+
+    DownmixInterleavedToMono(interleaved, kNumFrames, kNumChannels,
+                             deinterleaved);
+    const int16_t expected[kNumFrames] = {28000, -11, -30333};
+
+    EXPECT_THAT(deinterleaved, ElementsAreArray(expected));
+  }
+}
+
+TEST(AudioUtilTest, DownmixToMonoTest) {
+  {
+    const size_t kNumFrames = 4;
+    const int kNumChannels = 1;
+    const float input_data[kNumChannels][kNumFrames] = {{1.f, 2.f, -1.f, -3.f}};
+    const float* input[kNumChannels];
+    for (int i = 0; i < kNumChannels; ++i) {
+      input[i] = input_data[i];
+    }
+
+    float downmixed[kNumFrames];
+
+    DownmixToMono<float, float>(input, kNumFrames, kNumChannels, downmixed);
+
+    EXPECT_THAT(downmixed, ElementsAreArray(input_data[0]));
+  }
+  {
+    const size_t kNumFrames = 3;
+    const int kNumChannels = 2;
+    const float input_data[kNumChannels][kNumFrames] = {{1.f, 2.f, -1.f},
+                                                        {3.f, 0.f, 1.f}};
+    const float* input[kNumChannels];
+    for (int i = 0; i < kNumChannels; ++i) {
+      input[i] = input_data[i];
+    }
+
+    float downmixed[kNumFrames];
+    const float expected[kNumFrames] = {2.f, 1.f, 0.f};
+
+    DownmixToMono<float, float>(input, kNumFrames, kNumChannels, downmixed);
+
+    EXPECT_THAT(downmixed, ElementsAreArray(expected));
+  }
+  {
+    const size_t kNumFrames = 3;
+    const int kNumChannels = 3;
+    const int16_t input_data[kNumChannels][kNumFrames] = {
+        {30000, -5, -30000}, {30000, -10, -30999}, {24001, -20, -30000}};
+    const int16_t* input[kNumChannels];
+    for (int i = 0; i < kNumChannels; ++i) {
+      input[i] = input_data[i];
+    }
+
+    int16_t downmixed[kNumFrames];
+    const int16_t expected[kNumFrames] = {28000, -11, -30333};
+
+    DownmixToMono<int16_t, int32_t>(input, kNumFrames, kNumChannels, downmixed);
+
+    EXPECT_THAT(downmixed, ElementsAreArray(expected));
+  }
+}
+
+}  // namespace
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/blocker.cc b/media/webrtc/trunk/webrtc/common_audio/blocker.cc
index 9569df4701..13432f2e7a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/blocker.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/blocker.cc
@@ -18,15 +18,15 @@ namespace {
 
 // Adds |a| and |b| frame by frame into |result| (basically matrix addition).
 void AddFrames(const float* const* a,
-               int a_start_index,
+               size_t a_start_index,
                const float* const* b,
                int b_start_index,
-               int num_frames,
-               int num_channels,
+               size_t num_frames,
+               size_t num_channels,
                float* const* result,
-               int result_start_index) {
-  for (int i = 0; i < num_channels; ++i) {
-    for (int j = 0; j < num_frames; ++j) {
+               size_t result_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
       result[i][j + result_start_index] =
           a[i][j + a_start_index] + b[i][j + b_start_index];
     }
@@ -35,12 +35,12 @@ void AddFrames(const float* const* a,
 
 // Copies |src| into |dst| channel by channel.
 void CopyFrames(const float* const* src,
-                int src_start_index,
-                int num_frames,
-                int num_channels,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
                 float* const* dst,
-                int dst_start_index) {
-  for (int i = 0; i < num_channels; ++i) {
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
     memcpy(&dst[i][dst_start_index],
            &src[i][src_start_index],
            num_frames * sizeof(dst[i][dst_start_index]));
@@ -49,12 +49,12 @@ void CopyFrames(const float* const* src,
 
 // Moves |src| into |dst| channel by channel.
 void MoveFrames(const float* const* src,
-                int src_start_index,
-                int num_frames,
-                int num_channels,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
                 float* const* dst,
-                int dst_start_index) {
-  for (int i = 0; i < num_channels; ++i) {
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
     memmove(&dst[i][dst_start_index],
             &src[i][src_start_index],
             num_frames * sizeof(dst[i][dst_start_index]));
@@ -62,10 +62,10 @@ void MoveFrames(const float* const* src,
 }
 
 void ZeroOut(float* const* buffer,
-             int starting_idx,
-             int num_frames,
-             int num_channels) {
-  for (int i = 0; i < num_channels; ++i) {
+             size_t starting_idx,
+             size_t num_frames,
+             size_t num_channels) {
+  for (size_t i = 0; i < num_channels; ++i) {
     memset(&buffer[i][starting_idx], 0,
            num_frames * sizeof(buffer[i][starting_idx]));
   }
@@ -74,18 +74,18 @@ void ZeroOut(float* const* buffer,
 // Pointwise multiplies each channel of |frames| with |window|. Results are
 // stored in |frames|.
 void ApplyWindow(const float* window,
-                 int num_frames,
-                 int num_channels,
+                 size_t num_frames,
+                 size_t num_channels,
                  float* const* frames) {
-  for (int i = 0; i < num_channels; ++i) {
-    for (int j = 0; j < num_frames; ++j) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
       frames[i][j] = frames[i][j] * window[j];
     }
   }
 }
 
-int gcd(int a, int b) {
-  int tmp;
+size_t gcd(size_t a, size_t b) {
+  size_t tmp;
   while (b) {
      tmp = a;
      a = b;
@@ -98,12 +98,12 @@ int gcd(int a, int b) {
 
 namespace webrtc {
 
-Blocker::Blocker(int chunk_size,
-                 int block_size,
-                 int num_input_channels,
-                 int num_output_channels,
+Blocker::Blocker(size_t chunk_size,
+                 size_t block_size,
+                 size_t num_input_channels,
+                 size_t num_output_channels,
                  const float* window,
-                 int shift_amount,
+                 size_t shift_amount,
                  BlockerCallback* callback)
     : chunk_size_(chunk_size),
       block_size_(block_size),
@@ -118,11 +118,11 @@ Blocker::Blocker(int chunk_size,
       window_(new float[block_size_]),
       shift_amount_(shift_amount),
       callback_(callback) {
-  CHECK_LE(num_output_channels_, num_input_channels_);
-  CHECK(window);
+  RTC_CHECK_LE(num_output_channels_, num_input_channels_);
+  RTC_CHECK_LE(shift_amount_, block_size_);
 
   memcpy(window_.get(), window, block_size_ * sizeof(*window_.get()));
-  input_buffer_.MoveReadPosition(-initial_delay_);
+  input_buffer_.MoveReadPositionBackward(initial_delay_);
 }
 
 // When block_size < chunk_size the input and output buffers look like this:
@@ -165,22 +165,22 @@ Blocker::Blocker(int chunk_size,
 //
 // TODO(claguna): Look at using ring buffers to eliminate some copies.
 void Blocker::ProcessChunk(const float* const* input,
-                           int chunk_size,
-                           int num_input_channels,
-                           int num_output_channels,
+                           size_t chunk_size,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
                            float* const* output) {
-  CHECK_EQ(chunk_size, chunk_size_);
-  CHECK_EQ(num_input_channels, num_input_channels_);
-  CHECK_EQ(num_output_channels, num_output_channels_);
+  RTC_CHECK_EQ(chunk_size, chunk_size_);
+  RTC_CHECK_EQ(num_input_channels, num_input_channels_);
+  RTC_CHECK_EQ(num_output_channels, num_output_channels_);
 
   input_buffer_.Write(input, num_input_channels, chunk_size_);
-  int first_frame_in_block = frame_offset_;
+  size_t first_frame_in_block = frame_offset_;
 
   // Loop through blocks.
   while (first_frame_in_block < chunk_size_) {
     input_buffer_.Read(input_block_.channels(), num_input_channels,
                        block_size_);
-    input_buffer_.MoveReadPosition(-block_size_ + shift_amount_);
+    input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_);
 
     ApplyWindow(window_.get(),
                 block_size_,
diff --git a/media/webrtc/trunk/webrtc/common_audio/blocker.h b/media/webrtc/trunk/webrtc/common_audio/blocker.h
index fbd7973837..3a67c134d0 100644
--- a/media/webrtc/trunk/webrtc/common_audio/blocker.h
+++ b/media/webrtc/trunk/webrtc/common_audio/blocker.h
@@ -25,9 +25,9 @@ class BlockerCallback {
   virtual ~BlockerCallback() {}
 
   virtual void ProcessBlock(const float* const* input,
-                            int num_frames,
-                            int num_input_channels,
-                            int num_output_channels,
+                            size_t num_frames,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
                             float* const* output) = 0;
 };
 
@@ -63,34 +63,34 @@ class BlockerCallback {
 // copy of window and does not attempt to delete it.
 class Blocker {
  public:
-  Blocker(int chunk_size,
-          int block_size,
-          int num_input_channels,
-          int num_output_channels,
+  Blocker(size_t chunk_size,
+          size_t block_size,
+          size_t num_input_channels,
+          size_t num_output_channels,
           const float* window,
-          int shift_amount,
+          size_t shift_amount,
           BlockerCallback* callback);
 
   void ProcessChunk(const float* const* input,
-                    int num_frames,
-                    int num_input_channels,
-                    int num_output_channels,
+                    size_t chunk_size,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
                     float* const* output);
 
  private:
-  const int chunk_size_;
-  const int block_size_;
-  const int num_input_channels_;
-  const int num_output_channels_;
+  const size_t chunk_size_;
+  const size_t block_size_;
+  const size_t num_input_channels_;
+  const size_t num_output_channels_;
 
   // The number of frames of delay to add at the beginning of the first chunk.
-  const int initial_delay_;
+  const size_t initial_delay_;
 
   // The frame index into the input buffer where the first block should be read
   // from. This is necessary because shift_amount_ is not necessarily a
   // multiple of chunk_size_, so blocks won't line up at the start of the
   // buffer.
-  int frame_offset_;
+  size_t frame_offset_;
 
   // Since blocks nearly always overlap, there are certain blocks that require
   // frames from the end of one chunk and the beginning of the next chunk. The
@@ -113,7 +113,7 @@ class Blocker {
 
   // The amount of frames between the start of contiguous blocks. For example,
   // |shift_amount_| = |block_size_| / 2 for a Hann window.
-  int shift_amount_;
+  size_t shift_amount_;
 
   BlockerCallback* callback_;
 };
diff --git a/media/webrtc/trunk/webrtc/common_audio/blocker_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/blocker_unittest.cc
index 9e9988612a..a5a7b56282 100644
--- a/media/webrtc/trunk/webrtc/common_audio/blocker_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/blocker_unittest.cc
@@ -11,6 +11,7 @@
 #include "webrtc/common_audio/blocker.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
 
 namespace {
 
@@ -18,12 +19,12 @@ namespace {
 class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
  public:
   void ProcessBlock(const float* const* input,
-                    int num_frames,
-                    int num_input_channels,
-                    int num_output_channels,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
                     float* const* output) override {
-    for (int i = 0; i < num_output_channels; ++i) {
-      for (int j = 0; j < num_frames; ++j) {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
         output[i][j] = input[i][j] + 3;
       }
     }
@@ -34,12 +35,12 @@ class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
 class CopyBlockerCallback : public webrtc::BlockerCallback {
  public:
   void ProcessBlock(const float* const* input,
-                    int num_frames,
-                    int num_input_channels,
-                    int num_output_channels,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
                     float* const* output) override {
-    for (int i = 0; i < num_output_channels; ++i) {
-      for (int j = 0; j < num_frames; ++j) {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
         output[i][j] = input[i][j];
       }
     }
@@ -56,16 +57,16 @@ namespace webrtc {
 class BlockerTest : public ::testing::Test {
  protected:
   void RunTest(Blocker* blocker,
-               int chunk_size,
-               int num_frames,
+               size_t chunk_size,
+               size_t num_frames,
                const float* const* input,
                float* const* input_chunk,
                float* const* output,
                float* const* output_chunk,
-               int num_input_channels,
-               int num_output_channels) {
-    int start = 0;
-    int end = chunk_size - 1;
+               size_t num_input_channels,
+               size_t num_output_channels) {
+    size_t start = 0;
+    size_t end = chunk_size - 1;
     while (end < num_frames) {
       CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input);
       blocker->ProcessChunk(input_chunk,
@@ -75,28 +76,28 @@ class BlockerTest : public ::testing::Test {
                             output_chunk);
       CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk);
 
-      start = start + chunk_size;
-      end = end + chunk_size;
+      start += chunk_size;
+      end += chunk_size;
     }
   }
 
   void ValidateSignalEquality(const float* const* expected,
                               const float* const* actual,
-                              int num_channels,
-                              int num_frames) {
-    for (int i = 0; i < num_channels; ++i) {
-      for (int j = 0; j < num_frames; ++j) {
+                              size_t num_channels,
+                              size_t num_frames) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
         EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]);
       }
     }
   }
 
   void ValidateInitialDelay(const float* const* output,
-                            int num_channels,
-                            int num_frames,
-                            int initial_delay) {
-    for (int i = 0; i < num_channels; ++i) {
-      for (int j = 0; j < num_frames; ++j) {
+                            size_t num_channels,
+                            size_t num_frames,
+                            size_t initial_delay) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
         if (j < initial_delay) {
           EXPECT_FLOAT_EQ(output[i][j], 0.f);
         } else {
@@ -107,12 +108,12 @@ class BlockerTest : public ::testing::Test {
   }
 
   static void CopyTo(float* const* dst,
-                     int start_index_dst,
-                     int start_index_src,
-                     int num_channels,
-                     int num_frames,
+                     size_t start_index_dst,
+                     size_t start_index_src,
+                     size_t num_channels,
+                     size_t num_frames,
                      const float* const* src) {
-    for (int i = 0; i < num_channels; ++i) {
+    for (size_t i = 0; i < num_channels; ++i) {
       memcpy(&dst[i][start_index_dst],
              &src[i][start_index_src],
              num_frames * sizeof(float));
@@ -121,12 +122,12 @@ class BlockerTest : public ::testing::Test {
 };
 
 TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
-  const int kNumInputChannels = 3;
-  const int kNumOutputChannels = 2;
-  const int kNumFrames = 10;
-  const int kBlockSize = 4;
-  const int kChunkSize = 5;
-  const int kShiftAmount = 2;
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 10;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 5;
+  const size_t kShiftAmount = 2;
 
   const float kInput[kNumInputChannels][kNumFrames] = {
       {1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
@@ -174,12 +175,12 @@ TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
 }
 
 TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
-  const int kNumInputChannels = 3;
-  const int kNumOutputChannels = 2;
-  const int kNumFrames = 12;
-  const int kBlockSize = 4;
-  const int kChunkSize = 6;
-  const int kShiftAmount = 3;
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 6;
+  const size_t kShiftAmount = 3;
 
   const float kInput[kNumInputChannels][kNumFrames] = {
       {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
@@ -227,12 +228,12 @@ TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
 }
 
 TEST_F(BlockerTest, TestBlockerNoOverlap) {
-  const int kNumInputChannels = 3;
-  const int kNumOutputChannels = 2;
-  const int kNumFrames = 12;
-  const int kBlockSize = 4;
-  const int kChunkSize = 4;
-  const int kShiftAmount = 4;
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 4;
+  const size_t kShiftAmount = 4;
 
   const float kInput[kNumInputChannels][kNumFrames] = {
       {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
@@ -280,21 +281,21 @@ TEST_F(BlockerTest, TestBlockerNoOverlap) {
 }
 
 TEST_F(BlockerTest, InitialDelaysAreMinimum) {
-  const int kNumInputChannels = 3;
-  const int kNumOutputChannels = 2;
-  const int kNumFrames = 1280;
-  const int kChunkSize[] =
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 1280;
+  const size_t kChunkSize[] =
       {80, 80, 80, 80, 80, 80, 160, 160, 160, 160, 160, 160};
-  const int kBlockSize[] =
+  const size_t kBlockSize[] =
       {64, 64, 64, 128, 128, 128, 128, 128, 128, 256, 256, 256};
-  const int kShiftAmount[] =
+  const size_t kShiftAmount[] =
       {16, 32, 64, 32, 64, 128, 32, 64, 128, 64, 128, 256};
-  const int kInitialDelay[] =
+  const size_t kInitialDelay[] =
       {48, 48, 48, 112, 112, 112, 96, 96, 96, 224, 224, 224};
 
   float input[kNumInputChannels][kNumFrames];
-  for (int i = 0; i < kNumInputChannels; ++i) {
-    for (int j = 0; j < kNumFrames; ++j) {
+  for (size_t i = 0; i < kNumInputChannels; ++i) {
+    for (size_t j = 0; j < kNumFrames; ++j) {
       input[i][j] = i + 1;
     }
   }
@@ -305,9 +306,9 @@ TEST_F(BlockerTest, InitialDelaysAreMinimum) {
 
   CopyBlockerCallback callback;
 
-  for (size_t i = 0; i < (sizeof(kChunkSize) / sizeof(*kChunkSize)); ++i) {
+  for (size_t i = 0; i < arraysize(kChunkSize); ++i) {
     rtc::scoped_ptr<float[]> window(new float[kBlockSize[i]]);
-    for (int j = 0; j < kBlockSize[i]; ++j) {
+    for (size_t j = 0; j < kBlockSize[i]; ++j) {
       window[j] = 1.f;
     }
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/channel_buffer.cc b/media/webrtc/trunk/webrtc/common_audio/channel_buffer.cc
index 14aaa7af09..44520c6100 100644
--- a/media/webrtc/trunk/webrtc/common_audio/channel_buffer.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/channel_buffer.cc
@@ -12,9 +12,9 @@
 
 namespace webrtc {
 
-IFChannelBuffer::IFChannelBuffer(int num_frames,
-                                 int num_channels,
-                                 int num_bands)
+IFChannelBuffer::IFChannelBuffer(size_t num_frames,
+                                 size_t num_channels,
+                                 size_t num_bands)
     : ivalid_(true),
       ibuf_(num_frames, num_channels, num_bands),
       fvalid_(true),
@@ -47,8 +47,8 @@ void IFChannelBuffer::RefreshF() const {
     assert(ivalid_);
     const int16_t* const* int_channels = ibuf_.channels();
     float* const* float_channels = fbuf_.channels();
-    for (int i = 0; i < ibuf_.num_channels(); ++i) {
-      for (int j = 0; j < ibuf_.num_frames(); ++j) {
+    for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
+      for (size_t j = 0; j < ibuf_.num_frames(); ++j) {
         float_channels[i][j] = int_channels[i][j];
       }
     }
@@ -61,7 +61,7 @@ void IFChannelBuffer::RefreshI() const {
     assert(fvalid_);
     int16_t* const* int_channels = ibuf_.channels();
     const float* const* float_channels = fbuf_.channels();
-    for (int i = 0; i < ibuf_.num_channels(); ++i) {
+    for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
       FloatS16ToS16(float_channels[i],
                     ibuf_.num_frames(),
                     int_channels[i]);
diff --git a/media/webrtc/trunk/webrtc/common_audio/channel_buffer.h b/media/webrtc/trunk/webrtc/common_audio/channel_buffer.h
index a5dcc6c264..d9069163fa 100644
--- a/media/webrtc/trunk/webrtc/common_audio/channel_buffer.h
+++ b/media/webrtc/trunk/webrtc/common_audio/channel_buffer.h
@@ -39,9 +39,9 @@ namespace webrtc {
 template <typename T>
 class ChannelBuffer {
  public:
-  ChannelBuffer(int num_frames,
-                int num_channels,
-                int num_bands = 1)
+  ChannelBuffer(size_t num_frames,
+                size_t num_channels,
+                size_t num_bands = 1)
       : data_(new T[num_frames * num_channels]()),
         channels_(new T*[num_channels * num_bands]),
         bands_(new T*[num_channels * num_bands]),
@@ -49,8 +49,8 @@ class ChannelBuffer {
         num_frames_per_band_(num_frames / num_bands),
         num_channels_(num_channels),
         num_bands_(num_bands) {
-    for (int i = 0; i < num_channels_; ++i) {
-      for (int j = 0; j < num_bands_; ++j) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      for (size_t j = 0; j < num_bands_; ++j) {
         channels_[j * num_channels_ + i] =
             &data_[i * num_frames_ + j * num_frames_per_band_];
         bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i];
@@ -74,12 +74,11 @@ class ChannelBuffer {
   // 0 <= band < |num_bands_|
   // 0 <= channel < |num_channels_|
   // 0 <= sample < |num_frames_per_band_|
-  const T* const* channels(int band) const {
-    DCHECK_LT(band, num_bands_);
-    DCHECK_GE(band, 0);
+  const T* const* channels(size_t band) const {
+    RTC_DCHECK_LT(band, num_bands_);
     return &channels_[band * num_channels_];
   }
-  T* const* channels(int band) {
+  T* const* channels(size_t band) {
     const ChannelBuffer<T>* t = this;
     return const_cast<T* const*>(t->channels(band));
   }
@@ -91,37 +90,37 @@ class ChannelBuffer {
   // 0 <= channel < |num_channels_|
   // 0 <= band < |num_bands_|
   // 0 <= sample < |num_frames_per_band_|
-  const T* const* bands(int channel) const {
-    DCHECK_LT(channel, num_channels_);
-    DCHECK_GE(channel, 0);
+  const T* const* bands(size_t channel) const {
+    RTC_DCHECK_LT(channel, num_channels_);
+    RTC_DCHECK_GE(channel, 0u);
     return &bands_[channel * num_bands_];
   }
-  T* const* bands(int channel) {
+  T* const* bands(size_t channel) {
     const ChannelBuffer<T>* t = this;
     return const_cast<T* const*>(t->bands(channel));
   }
 
   // Sets the |slice| pointers to the |start_frame| position for each channel.
   // Returns |slice| for convenience.
-  const T* const* Slice(T** slice, int start_frame) const {
-    DCHECK_LT(start_frame, num_frames_);
-    for (int i = 0; i < num_channels_; ++i)
+  const T* const* Slice(T** slice, size_t start_frame) const {
+    RTC_DCHECK_LT(start_frame, num_frames_);
+    for (size_t i = 0; i < num_channels_; ++i)
       slice[i] = &channels_[i][start_frame];
     return slice;
   }
-  T** Slice(T** slice, int start_frame) {
+  T** Slice(T** slice, size_t start_frame) {
     const ChannelBuffer<T>* t = this;
     return const_cast<T**>(t->Slice(slice, start_frame));
   }
 
-  int num_frames() const { return num_frames_; }
-  int num_frames_per_band() const { return num_frames_per_band_; }
-  int num_channels() const { return num_channels_; }
-  int num_bands() const { return num_bands_; }
+  size_t num_frames() const { return num_frames_; }
+  size_t num_frames_per_band() const { return num_frames_per_band_; }
+  size_t num_channels() const { return num_channels_; }
+  size_t num_bands() const { return num_bands_; }
   size_t size() const {return num_frames_ * num_channels_; }
 
   void SetDataForTesting(const T* data, size_t size) {
-    CHECK_EQ(size, this->size());
+    RTC_CHECK_EQ(size, this->size());
     memcpy(data_.get(), data, size * sizeof(*data));
   }
 
@@ -129,10 +128,10 @@ class ChannelBuffer {
   rtc::scoped_ptr<T[]> data_;
   rtc::scoped_ptr<T* []> channels_;
   rtc::scoped_ptr<T* []> bands_;
-  const int num_frames_;
-  const int num_frames_per_band_;
-  const int num_channels_;
-  const int num_bands_;
+  const size_t num_frames_;
+  const size_t num_frames_per_band_;
+  const size_t num_channels_;
+  const size_t num_bands_;
 };
 
 // One int16_t and one float ChannelBuffer that are kept in sync. The sync is
@@ -143,17 +142,17 @@ class ChannelBuffer {
 // fbuf() until the next call to any of the other functions.
 class IFChannelBuffer {
  public:
-  IFChannelBuffer(int num_frames, int num_channels, int num_bands = 1);
+  IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1);
 
   ChannelBuffer<int16_t>* ibuf();
   ChannelBuffer<float>* fbuf();
   const ChannelBuffer<int16_t>* ibuf_const() const;
   const ChannelBuffer<float>* fbuf_const() const;
 
-  int num_frames() const { return ibuf_.num_frames(); }
-  int num_frames_per_band() const { return ibuf_.num_frames_per_band(); }
-  int num_channels() const { return ibuf_.num_channels(); }
-  int num_bands() const { return ibuf_.num_bands(); }
+  size_t num_frames() const { return ibuf_.num_frames(); }
+  size_t num_frames_per_band() const { return ibuf_.num_frames_per_band(); }
+  size_t num_channels() const { return ibuf_.num_channels(); }
+  size_t num_bands() const { return ibuf_.num_bands(); }
 
  private:
   void RefreshF() const;
diff --git a/media/webrtc/trunk/webrtc/common_audio/common_audio.gyp b/media/webrtc/trunk/webrtc/common_audio/common_audio.gyp
index a9b1ac0ad0..a1c6717944 100644
--- a/media/webrtc/trunk/webrtc/common_audio/common_audio.gyp
+++ b/media/webrtc/trunk/webrtc/common_audio/common_audio.gyp
@@ -99,6 +99,9 @@
         'signal_processing/splitting_filter.c',
         'signal_processing/sqrt_of_one_minus_x_squared.c',
         'signal_processing/vector_scaling_operations.c',
+        'sparse_fir_filter.cc',
+        'sparse_fir_filter.h',
+        'swap_queue.h',
         'vad/include/vad.h',
         'vad/include/webrtc_vad.h',
         'vad/vad.cc',
@@ -134,6 +137,9 @@
         ['target_arch=="ia32" or target_arch=="x64"', {
           'dependencies': ['common_audio_sse2',],
         }],
+        ['build_with_neon==1', {
+          'dependencies': ['common_audio_neon',],
+        }],
         ['target_arch=="arm"', {
           'sources': [
             'signal_processing/complex_bit_reverse_arm.S',
@@ -145,7 +151,6 @@
           ],
           'conditions': [
             ['arm_version>=7', {
-              'dependencies': ['common_audio_neon',],
               'sources': [
                 'signal_processing/filter_ar_fast_q12_armv7.S',
               ],
@@ -155,10 +160,7 @@
             }],
           ],  # conditions
         }],
-        ['target_arch=="arm64"', {
-          'dependencies': ['common_audio_neon',],
-        }],
-        ['target_arch=="mipsel" and mips_arch_variant!="r6" and android_webview_build==0', {
+        ['target_arch=="mipsel" and mips_arch_variant!="r6"', {
           'sources': [
             'signal_processing/include/spl_inl_mips.h',
             'signal_processing/complex_bit_reverse_mips.c',
@@ -199,19 +201,19 @@
             'fir_filter_sse.cc',
             'resampler/sinc_resampler_sse.cc',
           ],
-          'cflags': ['-msse2',],
           'conditions': [
-            [ 'os_posix == 1', {
+            ['os_posix==1', {
+              'cflags': [ '-msse2', ],
               'cflags_mozilla': ['-msse2',],
+              'xcode_settings': {
+                'OTHER_CFLAGS': [ '-msse2', ],
+              },
             }],
           ],
-          'xcode_settings': {
-            'OTHER_CFLAGS': ['-msse2',],
-          },
         },
       ],  # targets
     }],
-    ['target_arch=="arm" and arm_version>=7 or target_arch=="arm64"', {
+    ['build_with_neon==1', {
       'targets': [
         {
           'target_name': 'common_audio_neon',
@@ -224,21 +226,13 @@
             'signal_processing/downsample_fast_neon.c',
             'signal_processing/min_max_operations_neon.c',
           ],
-          'conditions': [
-            # Disable LTO in common_audio_neon target due to compiler bug
-            ['use_lto==1', {
-              'cflags!': [
-                '-flto',
-                '-ffat-lto-objects',
-              ],
-            }],
-          ],
         },
       ],  # targets
     }],
-    ['include_tests==1', {
+    ['include_tests==1 and OS!="ios"', {
       'targets' : [
         {
+          # Does not compile on iOS: webrtc:4755.
           'target_name': 'common_audio_unittests',
           'type': '<(gtest_target_type)',
           'dependencies': [
@@ -264,6 +258,8 @@
             'ring_buffer_unittest.cc',
             'signal_processing/real_fft_unittest.cc',
             'signal_processing/signal_processing_unittest.cc',
+            'sparse_fir_filter_unittest.cc',
+            'swap_queue_unittest.cc',
             'vad/vad_core_unittest.cc',
             'vad/vad_filterbank_unittest.cc',
             'vad/vad_gmm_unittest.cc',
diff --git a/media/webrtc/trunk/webrtc/common_audio/fft4g.c b/media/webrtc/trunk/webrtc/common_audio/fft4g.c
index cbc4dc31eb..9cf7b9f6ca 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fft4g.c
+++ b/media/webrtc/trunk/webrtc/common_audio/fft4g.c
@@ -27,7 +27,7 @@ functions
     dfst: Sine Transform of RDFT (Real Anti-symmetric DFT)
 function prototypes
     void cdft(int, int, float *, int *, float *);
-    void rdft(int, int, float *, int *, float *);
+    void rdft(size_t, int, float *, size_t *, float *);
     void ddct(int, int, float *, int *, float *);
     void ddst(int, int, float *, int *, float *);
     void dfct(int, float *, float *, int *, float *);
@@ -94,7 +94,7 @@ function prototypes
             ip[0] = 0; // first time only
             rdft(n, -1, a, ip, w);
     [parameters]
-        n              :data length (int)
+        n              :data length (size_t)
                         n >= 2, n = power of 2
         a[0...n-1]     :input/output data (float *)
                         <case1>
@@ -107,7 +107,7 @@ function prototypes
                                 a[2*j] = R[j], 0<=j<n/2
                                 a[2*j+1] = I[j], 0<j<n/2
                                 a[1] = R[n/2]
-        ip[0...*]      :work area for bit reversal (int *)
+        ip[0...*]      :work area for bit reversal (size_t *)
                         length of ip >= 2+sqrt(n/2)
                         strictly,
                         length of ip >=
@@ -286,22 +286,27 @@ Appendix :
     w[] and ip[] are compatible with all routines.
 */
 
-static void makewt(int nw, int *ip, float *w);
-static void makect(int nc, int *ip, float *c);
-static void bitrv2(int n, int *ip, float *a);
+#include <stddef.h>
+
+static void makewt(size_t nw, size_t *ip, float *w);
+static void makect(size_t nc, size_t *ip, float *c);
+static void bitrv2(size_t n, size_t *ip, float *a);
+#if 0  // Not used.
 static void bitrv2conj(int n, int *ip, float *a);
-static void cftfsub(int n, float *a, float *w);
-static void cftbsub(int n, float *a, float *w);
-static void cft1st(int n, float *a, float *w);
-static void cftmdl(int n, int l, float *a, float *w);
-static void rftfsub(int n, float *a, int nc, float *c);
-static void rftbsub(int n, float *a, int nc, float *c);
+#endif
+static void cftfsub(size_t n, float *a, float *w);
+static void cftbsub(size_t n, float *a, float *w);
+static void cft1st(size_t n, float *a, float *w);
+static void cftmdl(size_t n, size_t l, float *a, float *w);
+static void rftfsub(size_t n, float *a, size_t nc, float *c);
+static void rftbsub(size_t n, float *a, size_t nc, float *c);
 #if 0  // Not used.
 static void dctsub(int n, float *a, int nc, float *c)
 static void dstsub(int n, float *a, int nc, float *c)
 #endif
 
 
+#if 0  // Not used.
 void WebRtc_cdft(int n, int isgn, float *a, int *ip, float *w)
 {
     if (n > (ip[0] << 2)) {
@@ -319,11 +324,12 @@ void WebRtc_cdft(int n, int isgn, float *a, int *ip, float *w)
         cftfsub(n, a, w);
     }
 }
+#endif
 
 
-void WebRtc_rdft(int n, int isgn, float *a, int *ip, float *w)
+void WebRtc_rdft(size_t n, int isgn, float *a, size_t *ip, float *w)
 {
-    int nw, nc;
+    size_t nw, nc;
     float xi;
 
     nw = ip[0];
@@ -639,16 +645,16 @@ static void dfst(int n, float *a, float *t, int *ip, float *w)
 
 #include <math.h>
 
-static void makewt(int nw, int *ip, float *w)
+static void makewt(size_t nw, size_t *ip, float *w)
 {
-    int j, nwh;
+    size_t j, nwh;
     float delta, x, y;
 
     ip[0] = nw;
     ip[1] = 1;
     if (nw > 2) {
         nwh = nw >> 1;
-        delta = (float)atan(1.0f) / nwh;
+        delta = atanf(1.0f) / nwh;
         w[0] = 1;
         w[1] = 0;
         w[nwh] = (float)cos(delta * nwh);
@@ -668,15 +674,15 @@ static void makewt(int nw, int *ip, float *w)
 }
 
 
-static void makect(int nc, int *ip, float *c)
+static void makect(size_t nc, size_t *ip, float *c)
 {
-    int j, nch;
+    size_t j, nch;
     float delta;
 
     ip[1] = nc;
     if (nc > 1) {
         nch = nc >> 1;
-        delta = (float)atan(1.0f) / nch;
+        delta = atanf(1.0f) / nch;
         c[0] = (float)cos(delta * nch);
         c[nch] = 0.5f * c[0];
         for (j = 1; j < nch; j++) {
@@ -690,9 +696,9 @@ static void makect(int nc, int *ip, float *c)
 /* -------- child routines -------- */
 
 
-static void bitrv2(int n, int *ip, float *a)
+static void bitrv2(size_t n, size_t *ip, float *a)
 {
-    int j, j1, k, k1, l, m, m2;
+    size_t j, j1, k, k1, l, m, m2;
     float xr, xi, yr, yi;
 
     ip[0] = 0;
@@ -789,7 +795,7 @@ static void bitrv2(int n, int *ip, float *a)
     }
 }
 
-
+#if 0  // Not used.
 static void bitrv2conj(int n, int *ip, float *a)
 {
     int j, j1, k, k1, l, m, m2;
@@ -897,11 +903,11 @@ static void bitrv2conj(int n, int *ip, float *a)
         }
     }
 }
+#endif
 
-
-static void cftfsub(int n, float *a, float *w)
+static void cftfsub(size_t n, float *a, float *w)
 {
-    int j, j1, j2, j3, l;
+    size_t j, j1, j2, j3, l;
     float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
 
     l = 2;
@@ -949,9 +955,9 @@ static void cftfsub(int n, float *a, float *w)
 }
 
 
-static void cftbsub(int n, float *a, float *w)
+static void cftbsub(size_t n, float *a, float *w)
 {
-    int j, j1, j2, j3, l;
+    size_t j, j1, j2, j3, l;
     float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
 
     l = 2;
@@ -999,9 +1005,9 @@ static void cftbsub(int n, float *a, float *w)
 }
 
 
-static void cft1st(int n, float *a, float *w)
+static void cft1st(size_t n, float *a, float *w)
 {
-    int j, k1, k2;
+    size_t j, k1, k2;
     float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
     float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
 
@@ -1104,9 +1110,9 @@ static void cft1st(int n, float *a, float *w)
 }
 
 
-static void cftmdl(int n, int l, float *a, float *w)
+static void cftmdl(size_t n, size_t l, float *a, float *w)
 {
-    int j, j1, j2, j3, k, k1, k2, m, m2;
+    size_t j, j1, j2, j3, k, k1, k2, m, m2;
     float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
     float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
 
@@ -1231,9 +1237,9 @@ static void cftmdl(int n, int l, float *a, float *w)
 }
 
 
-static void rftfsub(int n, float *a, int nc, float *c)
+static void rftfsub(size_t n, float *a, size_t nc, float *c)
 {
-    int j, k, kk, ks, m;
+    size_t j, k, kk, ks, m;
     float wkr, wki, xr, xi, yr, yi;
 
     m = n >> 1;
@@ -1256,9 +1262,9 @@ static void rftfsub(int n, float *a, int nc, float *c)
 }
 
 
-static void rftbsub(int n, float *a, int nc, float *c)
+static void rftbsub(size_t n, float *a, size_t nc, float *c)
 {
-    int j, k, kk, ks, m;
+    size_t j, k, kk, ks, m;
     float wkr, wki, xr, xi, yr, yi;
 
     a[1] = -a[1];
diff --git a/media/webrtc/trunk/webrtc/common_audio/fft4g.h b/media/webrtc/trunk/webrtc/common_audio/fft4g.h
index 90fefa0207..6dd792f630 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fft4g.h
+++ b/media/webrtc/trunk/webrtc/common_audio/fft4g.h
@@ -16,8 +16,7 @@ extern "C" {
 #endif
 
 // Refer to fft4g.c for documentation.
-void WebRtc_rdft(int n, int isgn, float *a, int *ip, float *w);
-void WebRtc_cdft(int n, int isgn, float *a, int *ip, float *w);
+void WebRtc_rdft(size_t n, int isgn, float *a, size_t *ip, float *w);
 
 #if defined(__cplusplus)
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/fir_filter.cc b/media/webrtc/trunk/webrtc/common_audio/fir_filter.cc
index c651235410..dc1b776f99 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fir_filter.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/fir_filter.cc
@@ -16,7 +16,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/fir_filter_neon.h"
 #include "webrtc/common_audio/fir_filter_sse.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
 namespace webrtc {
 
@@ -57,19 +57,16 @@ FIRFilter* FIRFilter::Create(const float* coefficients,
     filter = new FIRFilterC(coefficients, coefficients_length);
   }
 #endif
-#elif defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
-#if defined(WEBRTC_ARCH_ARM_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   filter =
       new FIRFilterNEON(coefficients, coefficients_length, max_input_length);
-#else
-  // ARM CPU detection required.
+#elif defined(WEBRTC_DETECT_NEON)
   if (WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) {
     filter =
         new FIRFilterNEON(coefficients, coefficients_length, max_input_length);
   } else {
     filter = new FIRFilterC(coefficients, coefficients_length);
   }
-#endif
 #else
   filter = new FIRFilterC(coefficients, coefficients_length);
 #endif
diff --git a/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.cc b/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.cc
index 97a75db0f2..a81562655b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.cc
@@ -14,7 +14,7 @@
 #include <assert.h>
 #include <string.h>
 
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.h b/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.h
index d7399ad977..3aa6168dd2 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.h
+++ b/media/webrtc/trunk/webrtc/common_audio/fir_filter_neon.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/fir_filter.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.cc b/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.cc
index 6e7ae70e3f..adbb2b75cc 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.cc
@@ -14,7 +14,7 @@
 #include <string.h>
 #include <xmmintrin.h>
 
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.h b/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.h
index d3968310b8..a3325cd01d 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.h
+++ b/media/webrtc/trunk/webrtc/common_audio/fir_filter_sse.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/fir_filter.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/fir_filter_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/fir_filter_unittest.cc
index 1bd01bea81..13f79d9482 100644
--- a/media/webrtc/trunk/webrtc/common_audio/fir_filter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/fir_filter_unittest.cc
@@ -16,6 +16,7 @@
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
+namespace {
 
 static const float kCoefficients[] = {0.2f, 0.3f, 0.5f, 0.7f, 0.11f};
 static const size_t kCoefficientsLength = sizeof(kCoefficients) /
@@ -34,6 +35,8 @@ void VerifyOutput(const float* expected_output,
                       length * sizeof(expected_output[0])));
 }
 
+}  // namespace
+
 TEST(FIRFilterTest, FilterAsIdentity) {
   const float kCoefficients[] = {1.f, 0.f, 0.f, 0.f, 0.f};
   float output[kInputLength];
diff --git a/media/webrtc/trunk/webrtc/common_audio/include/audio_util.h b/media/webrtc/trunk/webrtc/common_audio/include/audio_util.h
index 8262649145..55dfc06a31 100644
--- a/media/webrtc/trunk/webrtc/common_audio/include/audio_util.h
+++ b/media/webrtc/trunk/webrtc/common_audio/include/audio_util.h
@@ -12,7 +12,9 @@
 #define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
 
 #include <limits>
+#include <cstring>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
@@ -26,10 +28,10 @@ typedef std::numeric_limits<int16_t> limits_int16;
 // FloatS16: float   [-32768.0, 32767.0]
 static inline int16_t FloatToS16(float v) {
   if (v > 0)
-    return v >= 1 ? limits_int16::max() :
-                    static_cast<int16_t>(v * limits_int16::max() + 0.5f);
-  return v <= -1 ? limits_int16::min() :
-                   static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
+    return v >= 1 ? limits_int16::max()
+                  : static_cast<int16_t>(v * limits_int16::max() + 0.5f);
+  return v <= -1 ? limits_int16::min()
+                 : static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
 }
 
 static inline float S16ToFloat(int16_t v) {
@@ -42,10 +44,9 @@ static inline int16_t FloatS16ToS16(float v) {
   static const float kMaxRound = limits_int16::max() - 0.5f;
   static const float kMinRound = limits_int16::min() + 0.5f;
   if (v > 0)
-    return v >= kMaxRound ? limits_int16::max() :
-                            static_cast<int16_t>(v + 0.5f);
-  return v <= kMinRound ? limits_int16::min() :
-                          static_cast<int16_t>(v - 0.5f);
+    return v >= kMaxRound ? limits_int16::max()
+                          : static_cast<int16_t>(v + 0.5f);
+  return v <= kMinRound ? limits_int16::min() : static_cast<int16_t>(v - 0.5f);
 }
 
 static inline float FloatToFloatS16(float v) {
@@ -64,17 +65,34 @@ void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
 void FloatToFloatS16(const float* src, size_t size, float* dest);
 void FloatS16ToFloat(const float* src, size_t size, float* dest);
 
+// Copy audio from |src| channels to |dest| channels unless |src| and |dest|
+// point to the same address. |src| and |dest| must have the same number of
+// channels, and there must be sufficient space allocated in |dest|.
+template <typename T>
+void CopyAudioIfNeeded(const T* const* src,
+                       int num_frames,
+                       int num_channels,
+                       T* const* dest) {
+  for (int i = 0; i < num_channels; ++i) {
+    if (src[i] != dest[i]) {
+      std::copy(src[i], src[i] + num_frames, dest[i]);
+    }
+  }
+}
+
 // Deinterleave audio from |interleaved| to the channel buffers pointed to
 // by |deinterleaved|. There must be sufficient space allocated in the
 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
 // per buffer).
 template <typename T>
-void Deinterleave(const T* interleaved, int samples_per_channel,
-                  int num_channels, T* const* deinterleaved) {
-  for (int i = 0; i < num_channels; ++i) {
+void Deinterleave(const T* interleaved,
+                  size_t samples_per_channel,
+                  size_t num_channels,
+                  T* const* deinterleaved) {
+  for (size_t i = 0; i < num_channels; ++i) {
     T* channel = deinterleaved[i];
-    int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel; ++j) {
+    size_t interleaved_idx = i;
+    for (size_t j = 0; j < samples_per_channel; ++j) {
       channel[j] = interleaved[interleaved_idx];
       interleaved_idx += num_channels;
     }
@@ -85,18 +103,86 @@ void Deinterleave(const T* interleaved, int samples_per_channel,
 // |interleaved|. There must be sufficient space allocated in |interleaved|
 // (|samples_per_channel| * |num_channels|).
 template <typename T>
-void Interleave(const T* const* deinterleaved, int samples_per_channel,
-                int num_channels, T* interleaved) {
-  for (int i = 0; i < num_channels; ++i) {
+void Interleave(const T* const* deinterleaved,
+                size_t samples_per_channel,
+                size_t num_channels,
+                T* interleaved) {
+  for (size_t i = 0; i < num_channels; ++i) {
     const T* channel = deinterleaved[i];
-    int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel; ++j) {
+    size_t interleaved_idx = i;
+    for (size_t j = 0; j < samples_per_channel; ++j) {
       interleaved[interleaved_idx] = channel[j];
       interleaved_idx += num_channels;
     }
   }
 }
 
+// Copies audio from a single channel buffer pointed to by |mono| to each
+// channel of |interleaved|. There must be sufficient space allocated in
+// |interleaved| (|samples_per_channel| * |num_channels|).
+template <typename T>
+void UpmixMonoToInterleaved(const T* mono,
+                            int num_frames,
+                            int num_channels,
+                            T* interleaved) {
+  int interleaved_idx = 0;
+  for (int i = 0; i < num_frames; ++i) {
+    for (int j = 0; j < num_channels; ++j) {
+      interleaved[interleaved_idx++] = mono[i];
+    }
+  }
+}
+
+template <typename T, typename Intermediate>
+void DownmixToMono(const T* const* input_channels,
+                   size_t num_frames,
+                   int num_channels,
+                   T* out) {
+  for (size_t i = 0; i < num_frames; ++i) {
+    Intermediate value = input_channels[0][i];
+    for (int j = 1; j < num_channels; ++j) {
+      value += input_channels[j][i];
+    }
+    out[i] = value / num_channels;
+  }
+}
+
+// Downmixes an interleaved multichannel signal to a single channel by averaging
+// all channels.
+template <typename T, typename Intermediate>
+void DownmixInterleavedToMonoImpl(const T* interleaved,
+                                  size_t num_frames,
+                                  int num_channels,
+                                  T* deinterleaved) {
+  RTC_DCHECK_GT(num_channels, 0);
+  RTC_DCHECK_GT(num_frames, 0u);
+
+  const T* const end = interleaved + num_frames * num_channels;
+
+  while (interleaved < end) {
+    const T* const frame_end = interleaved + num_channels;
+
+    Intermediate value = *interleaved++;
+    while (interleaved < frame_end) {
+      value += *interleaved++;
+    }
+
+    *deinterleaved++ = value / num_channels;
+  }
+}
+
+template <typename T>
+void DownmixInterleavedToMono(const T* interleaved,
+                              size_t num_frames,
+                              int num_channels,
+                              T* deinterleaved);
+
+template <>
+void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
+                                       size_t num_frames,
+                                       int num_channels,
+                                       int16_t* deinterleaved);
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/lapped_transform.cc b/media/webrtc/trunk/webrtc/common_audio/lapped_transform.cc
index 3883582fc1..5ab1db1b25 100644
--- a/media/webrtc/trunk/webrtc/common_audio/lapped_transform.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/lapped_transform.cc
@@ -20,31 +20,31 @@
 namespace webrtc {
 
 void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
-                                               int num_frames,
-                                               int num_input_channels,
-                                               int num_output_channels,
+                                               size_t num_frames,
+                                               size_t num_input_channels,
+                                               size_t num_output_channels,
                                                float* const* output) {
-  CHECK_EQ(num_input_channels, parent_->in_channels_);
-  CHECK_EQ(num_output_channels, parent_->out_channels_);
-  CHECK_EQ(parent_->block_length_, num_frames);
+  RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_);
+  RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_);
+  RTC_CHECK_EQ(parent_->block_length_, num_frames);
 
-  for (int i = 0; i < num_input_channels; ++i) {
+  for (size_t i = 0; i < num_input_channels; ++i) {
     memcpy(parent_->real_buf_.Row(i), input[i],
            num_frames * sizeof(*input[0]));
     parent_->fft_->Forward(parent_->real_buf_.Row(i),
                            parent_->cplx_pre_.Row(i));
   }
 
-  int block_length = RealFourier::ComplexLength(
+  size_t block_length = RealFourier::ComplexLength(
       RealFourier::FftOrder(num_frames));
-  CHECK_EQ(parent_->cplx_length_, block_length);
+  RTC_CHECK_EQ(parent_->cplx_length_, block_length);
   parent_->block_processor_->ProcessAudioBlock(parent_->cplx_pre_.Array(),
                                                num_input_channels,
                                                parent_->cplx_length_,
                                                num_output_channels,
                                                parent_->cplx_post_.Array());
 
-  for (int i = 0; i < num_output_channels; ++i) {
+  for (size_t i = 0; i < num_output_channels; ++i) {
     parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
                            parent_->real_buf_.Row(i));
     memcpy(output[i], parent_->real_buf_.Row(i),
@@ -52,38 +52,50 @@ void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
   }
 }
 
-LappedTransform::LappedTransform(int in_channels, int out_channels,
-                                 int chunk_length, const float* window,
-                                 int block_length, int shift_amount,
+LappedTransform::LappedTransform(size_t num_in_channels,
+                                 size_t num_out_channels,
+                                 size_t chunk_length,
+                                 const float* window,
+                                 size_t block_length,
+                                 size_t shift_amount,
                                  Callback* callback)
     : blocker_callback_(this),
-      in_channels_(in_channels),
-      out_channels_(out_channels),
+      num_in_channels_(num_in_channels),
+      num_out_channels_(num_out_channels),
       block_length_(block_length),
       chunk_length_(chunk_length),
       block_processor_(callback),
-      blocker_(
-        chunk_length_, block_length_, in_channels_, out_channels_, window,
-        shift_amount, &blocker_callback_),
+      blocker_(chunk_length_,
+               block_length_,
+               num_in_channels_,
+               num_out_channels_,
+               window,
+               shift_amount,
+               &blocker_callback_),
       fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))),
       cplx_length_(RealFourier::ComplexLength(fft_->order())),
-      real_buf_(in_channels, block_length_, RealFourier::kFftBufferAlignment),
-      cplx_pre_(in_channels, cplx_length_, RealFourier::kFftBufferAlignment),
-      cplx_post_(out_channels, cplx_length_, RealFourier::kFftBufferAlignment) {
-  CHECK(in_channels_ > 0 && out_channels_ > 0);
-  CHECK_GT(block_length_, 0);
-  CHECK_GT(chunk_length_, 0);
-  CHECK(block_processor_);
+      real_buf_(num_in_channels,
+                block_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_pre_(num_in_channels,
+                cplx_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_post_(num_out_channels,
+                 cplx_length_,
+                 RealFourier::kFftBufferAlignment) {
+  RTC_CHECK(num_in_channels_ > 0 && num_out_channels_ > 0);
+  RTC_CHECK_GT(block_length_, 0u);
+  RTC_CHECK_GT(chunk_length_, 0u);
+  RTC_CHECK(block_processor_);
 
   // block_length_ power of 2?
-  CHECK_EQ(0, block_length_ & (block_length_ - 1));
+  RTC_CHECK_EQ(0u, block_length_ & (block_length_ - 1));
 }
 
 void LappedTransform::ProcessChunk(const float* const* in_chunk,
                                    float* const* out_chunk) {
-  blocker_.ProcessChunk(in_chunk, chunk_length_, in_channels_, out_channels_,
-                        out_chunk);
+  blocker_.ProcessChunk(in_chunk, chunk_length_, num_in_channels_,
+                        num_out_channels_, out_chunk);
 }
 
 }  // namespace webrtc
-
diff --git a/media/webrtc/trunk/webrtc/common_audio/lapped_transform.h b/media/webrtc/trunk/webrtc/common_audio/lapped_transform.h
index 9f6b302832..1373ca10e1 100644
--- a/media/webrtc/trunk/webrtc/common_audio/lapped_transform.h
+++ b/media/webrtc/trunk/webrtc/common_audio/lapped_transform.h
@@ -16,7 +16,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/blocker.h"
 #include "webrtc/common_audio/real_fourier.h"
-#include "webrtc/system_wrappers/interface/aligned_array.h"
+#include "webrtc/system_wrappers/include/aligned_array.h"
 
 namespace webrtc {
 
@@ -35,8 +35,8 @@ class LappedTransform {
     virtual ~Callback() {}
 
     virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
-                                   int in_channels, int frames,
-                                   int out_channels,
+                                   size_t num_in_channels, size_t frames,
+                                   size_t num_out_channels,
                                    std::complex<float>* const* out_block) = 0;
   };
 
@@ -46,8 +46,12 @@ class LappedTransform {
   // |block_length| defines the length of a block, in samples.
   // |shift_amount| is in samples. |callback| is the caller-owned audio
   // processing function called for each block of the input chunk.
-  LappedTransform(int in_channels, int out_channels, int chunk_length,
-                  const float* window, int block_length, int shift_amount,
+  LappedTransform(size_t num_in_channels,
+                  size_t num_out_channels,
+                  size_t chunk_length,
+                  const float* window,
+                  size_t block_length,
+                  size_t shift_amount,
                   Callback* callback);
   ~LappedTransform() {}
 
@@ -57,6 +61,31 @@ class LappedTransform {
   // |out_chunk|. Both buffers are caller-owned.
   void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
 
+  // Get the chunk length.
+  //
+  // The chunk length is the number of samples per channel that must be passed
+  // to ProcessChunk via the parameter in_chunk.
+  //
+  // Returns the same chunk_length passed to the LappedTransform constructor.
+  size_t chunk_length() const { return chunk_length_; }
+
+  // Get the number of input channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // in_chunk.
+  //
+  // Returns the same num_in_channels passed to the LappedTransform constructor.
+  size_t num_in_channels() const { return num_in_channels_; }
+
+  // Get the number of output channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // out_chunk.
+  //
+  // Returns the same num_out_channels passed to the LappedTransform
+  // constructor.
+  size_t num_out_channels() const { return num_out_channels_; }
+
  private:
   // Internal middleware callback, given to the blocker. Transforms each block
   // and hands it over to the processing method given at construction time.
@@ -64,25 +93,27 @@ class LappedTransform {
    public:
     explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
 
-    virtual void ProcessBlock(const float* const* input, int num_frames,
-                              int num_input_channels, int num_output_channels,
+    virtual void ProcessBlock(const float* const* input,
+                              size_t num_frames,
+                              size_t num_input_channels,
+                              size_t num_output_channels,
                               float* const* output);
 
    private:
     LappedTransform* const parent_;
   } blocker_callback_;
 
-  const int in_channels_;
-  const int out_channels_;
+  const size_t num_in_channels_;
+  const size_t num_out_channels_;
 
-  const int block_length_;
-  const int chunk_length_;
+  const size_t block_length_;
+  const size_t chunk_length_;
 
   Callback* const block_processor_;
   Blocker blocker_;
 
   rtc::scoped_ptr<RealFourier> fft_;
-  const int cplx_length_;
+  const size_t cplx_length_;
   AlignedArray<float> real_buf_;
   AlignedArray<std::complex<float> > cplx_pre_;
   AlignedArray<std::complex<float> > cplx_post_;
diff --git a/media/webrtc/trunk/webrtc/common_audio/lapped_transform_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/lapped_transform_unittest.cc
index 1bfb3b4b65..a78488e326 100644
--- a/media/webrtc/trunk/webrtc/common_audio/lapped_transform_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/lapped_transform_unittest.cc
@@ -25,21 +25,23 @@ class NoopCallback : public webrtc::LappedTransform::Callback {
   NoopCallback() : block_num_(0) {}
 
   virtual void ProcessAudioBlock(const complex<float>* const* in_block,
-                                 int in_channels, int frames, int out_channels,
+                                 size_t in_channels,
+                                 size_t frames,
+                                 size_t out_channels,
                                  complex<float>* const* out_block) {
-    CHECK_EQ(in_channels, out_channels);
-    for (int i = 0; i < out_channels; ++i) {
+    RTC_CHECK_EQ(in_channels, out_channels);
+    for (size_t i = 0; i < out_channels; ++i) {
       memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
     }
     ++block_num_;
   }
 
-  int block_num() {
+  size_t block_num() {
     return block_num_;
   }
 
  private:
-  int block_num_;
+  size_t block_num_;
 };
 
 class FftCheckerCallback : public webrtc::LappedTransform::Callback {
@@ -47,29 +49,32 @@ class FftCheckerCallback : public webrtc::LappedTransform::Callback {
   FftCheckerCallback() : block_num_(0) {}
 
   virtual void ProcessAudioBlock(const complex<float>* const* in_block,
-                                 int in_channels, int frames, int out_channels,
+                                 size_t in_channels,
+                                 size_t frames,
+                                 size_t out_channels,
                                  complex<float>* const* out_block) {
-    CHECK_EQ(in_channels, out_channels);
+    RTC_CHECK_EQ(in_channels, out_channels);
 
-    float full_length = (frames - 1) * 2;
+    size_t full_length = (frames - 1) * 2;
     ++block_num_;
 
     if (block_num_ > 0) {
-      ASSERT_NEAR(in_block[0][0].real(), full_length, 1e-5f);
+      ASSERT_NEAR(in_block[0][0].real(), static_cast<float>(full_length),
+                  1e-5f);
       ASSERT_NEAR(in_block[0][0].imag(), 0.0f, 1e-5f);
-      for (int i = 1; i < frames; ++i) {
+      for (size_t i = 1; i < frames; ++i) {
         ASSERT_NEAR(in_block[0][i].real(), 0.0f, 1e-5f);
         ASSERT_NEAR(in_block[0][i].imag(), 0.0f, 1e-5f);
       }
     }
   }
 
-  int block_num() {
+  size_t block_num() {
     return block_num_;
   }
 
  private:
-  int block_num_;
+  size_t block_num_;
 };
 
 void SetFloatArray(float value, int rows, int cols, float* const* array) {
@@ -85,10 +90,10 @@ void SetFloatArray(float value, int rows, int cols, float* const* array) {
 namespace webrtc {
 
 TEST(LappedTransformTest, Windowless) {
-  const int kChannels = 3;
-  const int kChunkLength = 512;
-  const int kBlockLength = 64;
-  const int kShiftAmount = 64;
+  const size_t kChannels = 3;
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 64;
   NoopCallback noop;
 
   // Rectangular window.
@@ -113,8 +118,8 @@ TEST(LappedTransformTest, Windowless) {
 
   trans.ProcessChunk(in_chunk, out_chunk);
 
-  for (int i = 0; i < kChannels; ++i) {
-    for (int j = 0; j < kChunkLength; ++j) {
+  for (size_t i = 0; i < kChannels; ++i) {
+    for (size_t j = 0; j < kChunkLength; ++j) {
       ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f);
     }
   }
@@ -123,9 +128,9 @@ TEST(LappedTransformTest, Windowless) {
 }
 
 TEST(LappedTransformTest, IdentityProcessor) {
-  const int kChunkLength = 512;
-  const int kBlockLength = 64;
-  const int kShiftAmount = 32;
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 32;
   NoopCallback noop;
 
   // Identity window for |overlap = block_size / 2|.
@@ -144,7 +149,7 @@ TEST(LappedTransformTest, IdentityProcessor) {
 
   trans.ProcessChunk(&in_chunk, &out_chunk);
 
-  for (int i = 0; i < kChunkLength; ++i) {
+  for (size_t i = 0; i < kChunkLength; ++i) {
     ASSERT_NEAR(out_chunk[i],
                 (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
                 1e-5f);
@@ -154,8 +159,8 @@ TEST(LappedTransformTest, IdentityProcessor) {
 }
 
 TEST(LappedTransformTest, Callbacks) {
-  const int kChunkLength = 512;
-  const int kBlockLength = 64;
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
   FftCheckerCallback call;
 
   // Rectangular window.
@@ -177,5 +182,27 @@ TEST(LappedTransformTest, Callbacks) {
   ASSERT_EQ(kChunkLength / kBlockLength, call.block_num());
 }
 
-}  // namespace webrtc
+TEST(LappedTransformTest, chunk_length) {
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+  const float window[kBlockLength] = {};
 
+  // Make sure that chunk_length returns the same value passed to the
+  // LappedTransform constructor.
+  {
+    const size_t kExpectedChunkLength = 512;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+  {
+    const size_t kExpectedChunkLength = 160;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/real_fourier.cc b/media/webrtc/trunk/webrtc/common_audio/real_fourier.cc
index dec2be6d60..55ec49cba2 100644
--- a/media/webrtc/trunk/webrtc/common_audio/real_fourier.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/real_fourier.cc
@@ -19,7 +19,7 @@ namespace webrtc {
 
 using std::complex;
 
-const int RealFourier::kFftBufferAlignment = 32;
+const size_t RealFourier::kFftBufferAlignment = 32;
 
 rtc::scoped_ptr<RealFourier> RealFourier::Create(int fft_order) {
 #if defined(RTC_USE_OPENMAX_DL)
@@ -29,19 +29,18 @@ rtc::scoped_ptr<RealFourier> RealFourier::Create(int fft_order) {
 #endif
 }
 
-int RealFourier::FftOrder(int length) {
-  CHECK_GT(length, 0);
-  return WebRtcSpl_GetSizeInBits(length - 1);
+int RealFourier::FftOrder(size_t length) {
+  RTC_CHECK_GT(length, 0U);
+  return WebRtcSpl_GetSizeInBits(static_cast<uint32_t>(length - 1));
 }
 
-int RealFourier::FftLength(int order) {
-  CHECK_GE(order, 0);
-  return 1 << order;
+size_t RealFourier::FftLength(int order) {
+  RTC_CHECK_GE(order, 0);
+  return static_cast<size_t>(1 << order);
 }
 
-int RealFourier::ComplexLength(int order) {
-  CHECK_GE(order, 0);
-  return (1 << order) / 2 + 1;
+size_t RealFourier::ComplexLength(int order) {
+  return FftLength(order) / 2 + 1;
 }
 
 RealFourier::fft_real_scoper RealFourier::AllocRealBuffer(int count) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/real_fourier.h b/media/webrtc/trunk/webrtc/common_audio/real_fourier.h
index cc49dbf379..0be56a58b0 100644
--- a/media/webrtc/trunk/webrtc/common_audio/real_fourier.h
+++ b/media/webrtc/trunk/webrtc/common_audio/real_fourier.h
@@ -14,7 +14,7 @@
 #include <complex>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 // Uniform interface class for the real DFT and its inverse, for power-of-2
 // input lengths. Also contains helper functions for buffer allocation, taking
@@ -30,7 +30,7 @@ class RealFourier {
       fft_cplx_scoper;
 
   // The alignment required for all input and output buffers, in bytes.
-  static const int kFftBufferAlignment;
+  static const size_t kFftBufferAlignment;
 
   // Construct a wrapper instance for the given input order, which must be
   // between 1 and kMaxFftOrder, inclusively.
@@ -39,14 +39,14 @@ class RealFourier {
 
   // Helper to compute the smallest FFT order (a power of 2) which will contain
   // the given input length.
-  static int FftOrder(int length);
+  static int FftOrder(size_t length);
 
   // Helper to compute the input length from the FFT order.
-  static int FftLength(int order);
+  static size_t FftLength(int order);
 
   // Helper to compute the exact length, in complex floats, of the transform
   // output (i.e. |2^order / 2 + 1|).
-  static int ComplexLength(int order);
+  static size_t ComplexLength(int order);
 
   // Buffer allocation helpers. The buffers are large enough to hold |count|
   // floats/complexes and suitably aligned for use by the implementation.
diff --git a/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.cc b/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.cc
index 6f76516432..8cd4c86b5b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.cc
@@ -22,12 +22,12 @@ using std::complex;
 
 namespace {
 
-void Conjugate(complex<float>* array, int complex_length) {
+void Conjugate(complex<float>* array, size_t complex_length) {
   std::for_each(array, array + complex_length,
                 [=](complex<float>& v) { v = std::conj(v); });
 }
 
-size_t ComputeWorkIpSize(int fft_length) {
+size_t ComputeWorkIpSize(size_t fft_length) {
   return static_cast<size_t>(2 + std::ceil(std::sqrt(
       static_cast<float>(fft_length))));
 }
@@ -40,9 +40,9 @@ RealFourierOoura::RealFourierOoura(int fft_order)
       complex_length_(ComplexLength(order_)),
       // Zero-initializing work_ip_ will cause rdft to initialize these work
       // arrays on the first call.
-      work_ip_(new int[ComputeWorkIpSize(length_)]()),
+      work_ip_(new size_t[ComputeWorkIpSize(length_)]()),
       work_w_(new float[complex_length_]()) {
-  CHECK_GE(fft_order, 1);
+  RTC_CHECK_GE(fft_order, 1);
 }
 
 void RealFourierOoura::Forward(const float* src, complex<float>* dest) const {
@@ -66,7 +66,7 @@ void RealFourierOoura::Inverse(const complex<float>* src, float* dest) const {
     auto dest_complex = reinterpret_cast<complex<float>*>(dest);
     // The real output array is shorter than the input complex array by one
     // complex element.
-    const int dest_complex_length = complex_length_ - 1;
+    const size_t dest_complex_length = complex_length_ - 1;
     std::copy(src, src + dest_complex_length, dest_complex);
     // Restore Ooura's conjugate definition.
     Conjugate(dest_complex, dest_complex_length);
diff --git a/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.h b/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.h
index 67b3ffd77b..8d094bf494 100644
--- a/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.h
+++ b/media/webrtc/trunk/webrtc/common_audio/real_fourier_ooura.h
@@ -31,11 +31,11 @@ class RealFourierOoura : public RealFourier {
 
  private:
   const int order_;
-  const int length_;
-  const int complex_length_;
+  const size_t length_;
+  const size_t complex_length_;
   // These are work arrays for Ooura. The names are based on the comments in
   // fft4g.c.
-  const rtc::scoped_ptr<int[]> work_ip_;
+  const rtc::scoped_ptr<size_t[]> work_ip_;
   const rtc::scoped_ptr<float[]> work_w_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/real_fourier_openmax.cc b/media/webrtc/trunk/webrtc/common_audio/real_fourier_openmax.cc
index f7a0f64e03..bc3e7347cb 100644
--- a/media/webrtc/trunk/webrtc/common_audio/real_fourier_openmax.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/real_fourier_openmax.cc
@@ -23,19 +23,19 @@ namespace {
 
 // Creates and initializes the Openmax state. Transfers ownership to caller.
 OMXFFTSpec_R_F32* CreateOpenmaxState(int order) {
-  CHECK_GE(order, 1);
+  RTC_CHECK_GE(order, 1);
   // The omx implementation uses this macro to check order validity.
-  CHECK_LE(order, TWIDDLE_TABLE_ORDER);
+  RTC_CHECK_LE(order, TWIDDLE_TABLE_ORDER);
 
   OMX_INT buffer_size;
   OMXResult r = omxSP_FFTGetBufSize_R_F32(order, &buffer_size);
-  CHECK_EQ(r, OMX_Sts_NoErr);
+  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
 
   OMXFFTSpec_R_F32* omx_spec = malloc(buffer_size);
-  DCHECK(omx_spec);
+  RTC_DCHECK(omx_spec);
 
   r = omxSP_FFTInit_R_F32(omx_spec, order);
-  CHECK_EQ(r, OMX_Sts_NoErr);
+  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
   return omx_spec;
 }
 
@@ -55,14 +55,14 @@ void RealFourierOpenmax::Forward(const float* src, complex<float>* dest) const {
   // http://en.cppreference.com/w/cpp/numeric/complex
   OMXResult r =
       omxSP_FFTFwd_RToCCS_F32(src, reinterpret_cast<OMX_F32*>(dest), omx_spec_);
-  CHECK_EQ(r, OMX_Sts_NoErr);
+  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
 }
 
 void RealFourierOpenmax::Inverse(const complex<float>* src, float* dest) const {
   OMXResult r =
       omxSP_FFTInv_CCSToR_F32(reinterpret_cast<const OMX_F32*>(src), dest,
                               omx_spec_);
-  CHECK_EQ(r, OMX_Sts_NoErr);
+  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/real_fourier_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/real_fourier_unittest.cc
index a66344187e..eb5880ee8a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/real_fourier_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/real_fourier_unittest.cc
@@ -26,32 +26,32 @@ TEST(RealFourierStaticsTest, AllocatorAlignment) {
     RealFourier::fft_real_scoper real;
     real = RealFourier::AllocRealBuffer(3);
     ASSERT_TRUE(real.get() != nullptr);
-    int64_t ptr_value = reinterpret_cast<int64_t>(real.get());
-    EXPECT_EQ(ptr_value % RealFourier::kFftBufferAlignment, 0);
+    uintptr_t ptr_value = reinterpret_cast<uintptr_t>(real.get());
+    EXPECT_EQ(0u, ptr_value % RealFourier::kFftBufferAlignment);
   }
   {
     RealFourier::fft_cplx_scoper cplx;
     cplx = RealFourier::AllocCplxBuffer(3);
     ASSERT_TRUE(cplx.get() != nullptr);
-    int64_t ptr_value = reinterpret_cast<int64_t>(cplx.get());
-    EXPECT_EQ(ptr_value % RealFourier::kFftBufferAlignment, 0);
+    uintptr_t ptr_value = reinterpret_cast<uintptr_t>(cplx.get());
+    EXPECT_EQ(0u, ptr_value % RealFourier::kFftBufferAlignment);
   }
 }
 
 TEST(RealFourierStaticsTest, OrderComputation) {
-  EXPECT_EQ(RealFourier::FftOrder(13), 4);
-  EXPECT_EQ(RealFourier::FftOrder(32), 5);
-  EXPECT_EQ(RealFourier::FftOrder(2), 1);
-  EXPECT_EQ(RealFourier::FftOrder(1), 0);
+  EXPECT_EQ(4, RealFourier::FftOrder(13));
+  EXPECT_EQ(5, RealFourier::FftOrder(32));
+  EXPECT_EQ(1, RealFourier::FftOrder(2));
+  EXPECT_EQ(0, RealFourier::FftOrder(1));
 }
 
 TEST(RealFourierStaticsTest, ComplexLengthComputation) {
-  EXPECT_EQ(RealFourier::ComplexLength(1), 2);
-  EXPECT_EQ(RealFourier::ComplexLength(2), 3);
-  EXPECT_EQ(RealFourier::ComplexLength(3), 5);
-  EXPECT_EQ(RealFourier::ComplexLength(4), 9);
-  EXPECT_EQ(RealFourier::ComplexLength(5), 17);
-  EXPECT_EQ(RealFourier::ComplexLength(7), 65);
+  EXPECT_EQ(2U, RealFourier::ComplexLength(1));
+  EXPECT_EQ(3U, RealFourier::ComplexLength(2));
+  EXPECT_EQ(5U, RealFourier::ComplexLength(3));
+  EXPECT_EQ(9U, RealFourier::ComplexLength(4));
+  EXPECT_EQ(17U, RealFourier::ComplexLength(5));
+  EXPECT_EQ(65U, RealFourier::ComplexLength(7));
 }
 
 template <typename T>
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/include/push_resampler.h b/media/webrtc/trunk/webrtc/common_audio/resampler/include/push_resampler.h
index a4e57e4b64..eeda790497 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/include/push_resampler.h
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/include/push_resampler.h
@@ -29,18 +29,18 @@ class PushResampler {
   // Must be called whenever the parameters change. Free to be called at any
   // time as it is a no-op if parameters have not changed since the last call.
   int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
-                         int num_channels);
+                         size_t num_channels);
 
   // Returns the total number of samples provided in destination (e.g. 32 kHz,
   // 2 channel audio gives 640 samples).
-  int Resample(const T* src, int src_length, T* dst, int dst_capacity);
+  int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
 
  private:
   rtc::scoped_ptr<PushSincResampler> sinc_resampler_;
   rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
   int src_sample_rate_hz_;
   int dst_sample_rate_hz_;
-  int num_channels_;
+  size_t num_channels_;
   rtc::scoped_ptr<T[]> src_left_;
   rtc::scoped_ptr<T[]> src_right_;
   rtc::scoped_ptr<T[]> dst_left_;
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/include/resampler.h b/media/webrtc/trunk/webrtc/common_audio/resampler/include/resampler.h
index 74874d1360..d0c967bb6b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/include/resampler.h
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/include/resampler.h
@@ -16,11 +16,13 @@
 #ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
 #define WEBRTC_RESAMPLER_RESAMPLER_H_
 
+#include <stddef.h>
+
 #include "webrtc/typedefs.h"
 #include <speex/speex_resampler.h>
 
 namespace webrtc {
-
+ 
 #define FIXED_RATE_RESAMPLER 0x10
 
 // All methods return 0 on success and -1 on failure.
@@ -29,18 +31,18 @@ class Resampler
 
 public:
     Resampler();
-    Resampler(int inFreq, int outFreq, int num_channels);
+    Resampler(int inFreq, int outFreq, size_t num_channels);
     ~Resampler();
 
     // Reset all states
-    int Reset(int inFreq, int outFreq, int num_channels);
+    int Reset(int inFreq, int outFreq, size_t num_channels);
 
     // Reset all states if any parameter has changed
-    int ResetIfNeeded(int inFreq, int outFreq, int num_channels);
+    int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
 
     // Resample samplesIn to samplesOut.
-    int Push(const int16_t* samplesIn, int lengthIn, int16_t* samplesOut,
-             int maxLen, int &outLen);
+    int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
+             size_t maxLen, size_t &outLen);
 
 private:
     SpeexResamplerState* state_;
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/push_resampler.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/push_resampler.cc
index 6f90569aef..afacd00f73 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/push_resampler.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/push_resampler.cc
@@ -31,7 +31,7 @@ PushResampler<T>::~PushResampler() {
 template <typename T>
 int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
                                          int dst_sample_rate_hz,
-                                         int num_channels) {
+                                         size_t num_channels) {
   if (src_sample_rate_hz == src_sample_rate_hz_ &&
       dst_sample_rate_hz == dst_sample_rate_hz_ &&
       num_channels == num_channels_)
@@ -46,8 +46,10 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
   dst_sample_rate_hz_ = dst_sample_rate_hz;
   num_channels_ = num_channels;
 
-  const int src_size_10ms_mono = src_sample_rate_hz / 100;
-  const int dst_size_10ms_mono = dst_sample_rate_hz / 100;
+  const size_t src_size_10ms_mono =
+      static_cast<size_t>(src_sample_rate_hz / 100);
+  const size_t dst_size_10ms_mono =
+      static_cast<size_t>(dst_sample_rate_hz / 100);
   sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
                                               dst_size_10ms_mono));
   if (num_channels_ == 2) {
@@ -63,10 +65,10 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
 }
 
 template <typename T>
-int PushResampler<T>::Resample(const T* src, int src_length, T* dst,
-                               int dst_capacity) {
-  const int src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
-  const int dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
+int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
+                               size_t dst_capacity) {
+  const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
+  const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
   if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
     return -1;
 
@@ -74,15 +76,15 @@ int PushResampler<T>::Resample(const T* src, int src_length, T* dst,
     // The old resampler provides this memcpy facility in the case of matching
     // sample rates, so reproduce it here for the sinc resampler.
     memcpy(dst, src, src_length * sizeof(T));
-    return src_length;
+    return static_cast<int>(src_length);
   }
   if (num_channels_ == 2) {
-    const int src_length_mono = src_length / num_channels_;
-    const int dst_capacity_mono = dst_capacity / num_channels_;
+    const size_t src_length_mono = src_length / num_channels_;
+    const size_t dst_capacity_mono = dst_capacity / num_channels_;
     T* deinterleaved[] = {src_left_.get(), src_right_.get()};
     Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
 
-    int dst_length_mono =
+    size_t dst_length_mono =
         sinc_resampler_->Resample(src_left_.get(), src_length_mono,
                                   dst_left_.get(), dst_capacity_mono);
     sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
@@ -91,9 +93,10 @@ int PushResampler<T>::Resample(const T* src, int src_length, T* dst,
     deinterleaved[0] = dst_left_.get();
     deinterleaved[1] = dst_right_.get();
     Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
-    return dst_length_mono * num_channels_;
+    return static_cast<int>(dst_length_mono * num_channels_);
   } else {
-    return sinc_resampler_->Resample(src, src_length, dst, dst_capacity);
+    return static_cast<int>(
+        sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.cc
index 7d372028b6..a740423eec 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.cc
@@ -17,7 +17,8 @@
 
 namespace webrtc {
 
-PushSincResampler::PushSincResampler(int source_frames, int destination_frames)
+PushSincResampler::PushSincResampler(size_t source_frames,
+                                     size_t destination_frames)
     : resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
                                    source_frames,
                                    this)),
@@ -30,10 +31,10 @@ PushSincResampler::PushSincResampler(int source_frames, int destination_frames)
 PushSincResampler::~PushSincResampler() {
 }
 
-int PushSincResampler::Resample(const int16_t* source,
-                                int source_length,
-                                int16_t* destination,
-                                int destination_capacity) {
+size_t PushSincResampler::Resample(const int16_t* source,
+                                   size_t source_length,
+                                   int16_t* destination,
+                                   size_t destination_capacity) {
   if (!float_buffer_.get())
     float_buffer_.reset(new float[destination_frames_]);
 
@@ -45,12 +46,12 @@ int PushSincResampler::Resample(const int16_t* source,
   return destination_frames_;
 }
 
-int PushSincResampler::Resample(const float* source,
-                                int source_length,
-                                float* destination,
-                                int destination_capacity) {
-  CHECK_EQ(source_length, resampler_->request_frames());
-  CHECK_GE(destination_capacity, destination_frames_);
+size_t PushSincResampler::Resample(const float* source,
+                                   size_t source_length,
+                                   float* destination,
+                                   size_t destination_capacity) {
+  RTC_CHECK_EQ(source_length, resampler_->request_frames());
+  RTC_CHECK_GE(destination_capacity, destination_frames_);
   // Cache the source pointer. Calling Resample() will immediately trigger
   // the Run() callback whereupon we provide the cached value.
   source_ptr_ = source;
@@ -77,10 +78,10 @@ int PushSincResampler::Resample(const float* source,
   return destination_frames_;
 }
 
-void PushSincResampler::Run(int frames, float* destination) {
+void PushSincResampler::Run(size_t frames, float* destination) {
   // Ensure we are only asked for the available samples. This would fail if
   // Run() was triggered more than once per Resample() call.
-  CHECK_EQ(source_available_, frames);
+  RTC_CHECK_EQ(source_available_, frames);
 
   if (first_pass_) {
     // Provide dummy input on the first pass, the output of which will be
@@ -93,7 +94,7 @@ void PushSincResampler::Run(int frames, float* destination) {
   if (source_ptr_) {
     std::memcpy(destination, source_ptr_, frames * sizeof(*destination));
   } else {
-    for (int i = 0; i < frames; ++i)
+    for (size_t i = 0; i < frames; ++i)
       destination[i] = static_cast<float>(source_ptr_int_[i]);
   }
   source_available_ -= frames;
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.h b/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.h
index c48ec71056..cefc62aa2a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.h
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler.h
@@ -27,7 +27,7 @@ class PushSincResampler : public SincResamplerCallback {
   // Provide the size of the source and destination blocks in samples. These
   // must correspond to the same time duration (typically 10 ms) as the sample
   // ratio is inferred from them.
-  PushSincResampler(int source_frames, int destination_frames);
+  PushSincResampler(size_t source_frames, size_t destination_frames);
   ~PushSincResampler() override;
 
   // Perform the resampling. |source_frames| must always equal the
@@ -35,12 +35,12 @@ class PushSincResampler : public SincResamplerCallback {
   // at least as large as |destination_frames|. Returns the number of samples
   // provided in destination (for convenience, since this will always be equal
   // to |destination_frames|).
-  int Resample(const int16_t* source, int source_frames,
-               int16_t* destination, int destination_capacity);
-  int Resample(const float* source,
-               int source_frames,
-               float* destination,
-               int destination_capacity);
+  size_t Resample(const int16_t* source, size_t source_frames,
+                  int16_t* destination, size_t destination_capacity);
+  size_t Resample(const float* source,
+                  size_t source_frames,
+                  float* destination,
+                  size_t destination_capacity);
 
   // Delay due to the filter kernel. Essentially, the time after which an input
   // sample will appear in the resampled output.
@@ -50,7 +50,7 @@ class PushSincResampler : public SincResamplerCallback {
 
  protected:
   // Implements SincResamplerCallback.
-  void Run(int frames, float* destination) override;
+  void Run(size_t frames, float* destination) override;
 
  private:
   friend class PushSincResamplerTest;
@@ -60,15 +60,15 @@ class PushSincResampler : public SincResamplerCallback {
   rtc::scoped_ptr<float[]> float_buffer_;
   const float* source_ptr_;
   const int16_t* source_ptr_int_;
-  const int destination_frames_;
+  const size_t destination_frames_;
 
   // True on the first call to Resample(), to prime the SincResampler buffer.
   bool first_pass_;
 
   // Used to assert we are only requested for as much data as is available.
-  int source_available_;
+  size_t source_available_;
 
-  DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
index f955a682b5..17e3dba1e2 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
@@ -17,7 +17,7 @@
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -58,14 +58,14 @@ class PushSincResamplerTest : public ::testing::TestWithParam<
 
 class ZeroSource : public SincResamplerCallback {
  public:
-  void Run(int frames, float* destination) {
+  void Run(size_t frames, float* destination) {
     std::memset(destination, 0, sizeof(float) * frames);
   }
 };
 
 void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
-  const int input_samples = input_rate_ / 100;
-  const int output_samples = output_rate_ / 100;
+  const size_t input_samples = static_cast<size_t>(input_rate_ / 100);
+  const size_t output_samples = static_cast<size_t>(output_rate_ / 100);
   const int kResampleIterations = 500000;
 
   // Source for data to be resampled.
@@ -77,7 +77,7 @@ void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
   rtc::scoped_ptr<int16_t[]> destination_int(new int16_t[output_samples]);
 
   resampler_source.Run(input_samples, source.get());
-  for (int i = 0; i < input_samples; ++i) {
+  for (size_t i = 0; i < input_samples; ++i) {
     source_int[i] = static_cast<int16_t>(floor(32767 * source[i] + 0.5));
   }
 
@@ -134,11 +134,13 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
   // Make comparisons using one second of data.
   static const double kTestDurationSecs = 1;
   // 10 ms blocks.
-  const int kNumBlocks = kTestDurationSecs * 100;
-  const int input_block_size = input_rate_ / 100;
-  const int output_block_size = output_rate_ / 100;
-  const int input_samples = kTestDurationSecs * input_rate_;
-  const int output_samples = kTestDurationSecs * output_rate_;
+  const size_t kNumBlocks = static_cast<size_t>(kTestDurationSecs * 100);
+  const size_t input_block_size = static_cast<size_t>(input_rate_ / 100);
+  const size_t output_block_size = static_cast<size_t>(output_rate_ / 100);
+  const size_t input_samples =
+      static_cast<size_t>(kTestDurationSecs * input_rate_);
+  const size_t output_samples =
+      static_cast<size_t>(kTestDurationSecs * output_rate_);
 
   // Nyquist frequency for the input sampling rate.
   const double input_nyquist_freq = 0.5 * input_rate_;
@@ -163,7 +165,7 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
   // deal with it in the test by delaying the "pure" source to match. It must be
   // checked before the first call to Resample(), because ChunkSize() will
   // change afterwards.
-  const int output_delay_samples = output_block_size -
+  const size_t output_delay_samples = output_block_size -
       resampler.get_resampler_for_testing()->ChunkSize();
 
   // Generate resampled signal.
@@ -171,7 +173,7 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
   // rather than in a single pass, to exercise how it will be used in WebRTC.
   resampler_source.Run(input_samples, source.get());
   if (int_format) {
-    for (int i = 0; i < kNumBlocks; ++i) {
+    for (size_t i = 0; i < kNumBlocks; ++i) {
       FloatToS16(&source[i * input_block_size], input_block_size,
                source_int.get());
       EXPECT_EQ(output_block_size,
@@ -183,7 +185,7 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
                &resampled_destination[i * output_block_size]);
     }
   } else {
-    for (int i = 0; i < kNumBlocks; ++i) {
+    for (size_t i = 0; i < kNumBlocks; ++i) {
       EXPECT_EQ(
           output_block_size,
           resampler.Resample(&source[i * input_block_size],
@@ -211,7 +213,7 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
   double low_frequency_range = kLowFrequencyNyquistRange * 0.5 * minimum_rate;
   double high_frequency_range = kHighFrequencyNyquistRange * 0.5 * minimum_rate;
 
-  for (int i = 0; i < output_samples; ++i) {
+  for (size_t i = 0; i < output_samples; ++i) {
     double error = fabs(resampled_destination[i] - pure_destination[i]);
 
     if (pure_source.Frequency(i) < low_frequency_range) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/resampler.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/resampler.cc
index 88e6055535..caaad7898b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/resampler.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/resampler.cc
@@ -36,7 +36,7 @@ Resampler::Resampler() : state_(NULL), channels_(0)
   // Note: Push will fail until Reset() is called
 }
 
-Resampler::Resampler(int inFreq, int outFreq, int num_channels)
+Resampler::Resampler(int inFreq, int outFreq, size_t num_channels)
     : Resampler() {
   Reset(inFreq, outFreq, num_channels);
 }
@@ -49,7 +49,7 @@ Resampler::~Resampler()
   }
 }
 
-int Resampler::ResetIfNeeded(int inFreq, int outFreq, int num_channels)
+int Resampler::ResetIfNeeded(int inFreq, int outFreq, size_t num_channels)
 {
   if (!state_ || channels_ != num_channels ||
       inFreq != in_freq_ || outFreq != out_freq_)
@@ -64,7 +64,7 @@ int Resampler::ResetIfNeeded(int inFreq, int outFreq, int num_channels)
   }
 }
 
-int Resampler::Reset(int inFreq, int outFreq, int num_channels)
+int Resampler::Reset(int inFreq, int outFreq, size_t num_channels)
 {
   if (num_channels != 1 && num_channels != 2) {
     return -1;
@@ -93,8 +93,8 @@ int Resampler::Reset(int inFreq, int outFreq, int num_channels)
 
 // Synchronous resampling, all output samples are written to samplesOut
 // TODO(jesup) Change to take samples-per-channel in and out
-int Resampler::Push(const int16_t* samplesIn, int lengthIn, int16_t* samplesOut,
-                    int maxLen, int &outLen)
+int Resampler::Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
+                    size_t maxLen, size_t &outLen)
 {
   if (maxLen < lengthIn)
   {
@@ -127,5 +127,4 @@ int Resampler::Push(const int16_t* samplesIn, int lengthIn, int16_t* samplesOut,
   outLen = (int) (channels_ * out);
   return 0;
 }
-
-}  // namespace webrtc
+}// namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/resampler_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/resampler_unittest.cc
index 5a272dac27..712c1fc9bf 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/resampler_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/resampler_unittest.cc
@@ -176,7 +176,6 @@ void ResamplerTest::RunResampleTest(int channels,
 }
 
 TEST_F(ResamplerTest, Mono) {
-  const int kChannels = 1;
   // We don't attempt to be exhaustive here, but just get good coverage. Some
   // combinations of rates will not be resampled, and some give an odd
   // resampling factor which makes it more difficult to evaluate.
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.cc
index 0e18ac55e2..6e392a043d 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.cc
@@ -83,11 +83,9 @@
 // |virtual_source_idx_|, etc.
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
+#ifndef _USE_MATH_DEFINES
 #define _USE_MATH_DEFINES
-
-#include "webrtc/common_audio/resampler/sinc_resampler.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
-#include "webrtc/typedefs.h"
+#endif
 
 #include <assert.h>
 #include <math.h>
@@ -95,9 +93,14 @@
 
 #include <limits>
 
-namespace webrtc {
+#include "webrtc/common_audio/resampler/sinc_resampler.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/typedefs.h"
 
-static double SincScaleFactor(double io_ratio) {
+
+namespace {
+
+double SincScaleFactor(double io_ratio) {
   // |sinc_scale_factor| is basically the normalized cutoff frequency of the
   // low-pass filter.
   double sinc_scale_factor = io_ratio > 1.0 ? 1.0 / io_ratio : 1.0;
@@ -113,6 +116,10 @@ static double SincScaleFactor(double io_ratio) {
   return sinc_scale_factor;
 }
 
+}  // namespace
+
+namespace webrtc {
+  
 // If we know the minimum architecture at compile time, avoid CPU detection.
 #if defined(WEBRTC_ARCH_X86_FAMILY)
 #if defined(__SSE2__)
@@ -127,29 +134,24 @@ void SincResampler::InitializeCPUSpecificFeatures() {}
 void SincResampler::InitializeCPUSpecificFeatures() {
   convolve_proc_ = WebRtc_GetCPUInfo(kSSE2) ? Convolve_SSE : Convolve_C;
 }
-#endif
-#elif defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
-#if defined(WEBRTC_ARCH_ARM_NEON)
+#endif // defined(__SSE2__)
+#elif defined(WEBRTC_HAS_NEON)
 #define CONVOLVE_FUNC Convolve_NEON
 void SincResampler::InitializeCPUSpecificFeatures() {}
-#else
-// ARM CPU detection required.  Function will be set by
-// InitializeCPUSpecificFeatures().
+#elif defined(WEBRTC_DETECT_NEON)
 #define CONVOLVE_FUNC convolve_proc_
-
 void SincResampler::InitializeCPUSpecificFeatures() {
   convolve_proc_ = WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON ?
       Convolve_NEON : Convolve_C;
 }
-#endif
 #else
 // Unknown architecture.
 #define CONVOLVE_FUNC Convolve_C
 void SincResampler::InitializeCPUSpecificFeatures() {}
-#endif
+#endif // defined(WEBRTC_ARCH_X86_FAMILY
 
 SincResampler::SincResampler(double io_sample_rate_ratio,
-                             int request_frames,
+                             size_t request_frames,
                              SincResamplerCallback* read_cb)
     : io_sample_rate_ratio_(io_sample_rate_ratio),
       read_cb_(read_cb),
@@ -215,14 +217,15 @@ void SincResampler::InitializeKernel() {
   // Generates a set of windowed sinc() kernels.
   // We generate a range of sub-sample offsets from 0.0 to 1.0.
   const double sinc_scale_factor = SincScaleFactor(io_sample_rate_ratio_);
-  for (int offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
+  for (size_t offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
     const float subsample_offset =
         static_cast<float>(offset_idx) / kKernelOffsetCount;
 
-    for (int i = 0; i < kKernelSize; ++i) {
-      const int idx = i + offset_idx * kKernelSize;
-      const float pre_sinc =
-          static_cast<float>(M_PI * (i - kKernelSize / 2 - subsample_offset));
+    for (size_t i = 0; i < kKernelSize; ++i) {
+      const size_t idx = i + offset_idx * kKernelSize;
+      const float pre_sinc = static_cast<float>(M_PI *
+          (static_cast<int>(i) - static_cast<int>(kKernelSize / 2) -
+           subsample_offset));
       kernel_pre_sinc_storage_[idx] = pre_sinc;
 
       // Compute Blackman window, matching the offset of the sinc().
@@ -252,9 +255,9 @@ void SincResampler::SetRatio(double io_sample_rate_ratio) {
   // Optimize reinitialization by reusing values which are independent of
   // |sinc_scale_factor|.  Provides a 3x speedup.
   const double sinc_scale_factor = SincScaleFactor(io_sample_rate_ratio_);
-  for (int offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
-    for (int i = 0; i < kKernelSize; ++i) {
-      const int idx = i + offset_idx * kKernelSize;
+  for (size_t offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
+    for (size_t i = 0; i < kKernelSize; ++i) {
+      const size_t idx = i + offset_idx * kKernelSize;
       const float window = kernel_window_storage_[idx];
       const float pre_sinc = kernel_pre_sinc_storage_[idx];
 
@@ -266,8 +269,8 @@ void SincResampler::SetRatio(double io_sample_rate_ratio) {
   }
 }
 
-void SincResampler::Resample(int frames, float* destination) {
-  int remaining_frames = frames;
+void SincResampler::Resample(size_t frames, float* destination) {
+  size_t remaining_frames = frames;
 
   // Step (1) -- Prime the input buffer at the start of the input stream.
   if (!buffer_primed_ && remaining_frames) {
@@ -343,8 +346,8 @@ void SincResampler::Resample(int frames, float* destination) {
 
 #undef CONVOLVE_FUNC
 
-int SincResampler::ChunkSize() const {
-  return static_cast<int>(block_size_ / io_sample_rate_ratio_);
+size_t SincResampler::ChunkSize() const {
+  return static_cast<size_t>(block_size_ / io_sample_rate_ratio_);
 }
 
 void SincResampler::Flush() {
@@ -363,7 +366,7 @@ float SincResampler::Convolve_C(const float* input_ptr, const float* k1,
 
   // Generate a single output sample.  Unrolling this loop hurt performance in
   // local testing.
-  int n = kKernelSize;
+  size_t n = kKernelSize;
   while (n--) {
     sum1 += *input_ptr * *k1++;
     sum2 += *input_ptr++ * *k2++;
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.h b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.h
index be84a99624..45ade0cc69 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.h
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler.h
@@ -16,7 +16,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 #include "webrtc/typedefs.h"
 
@@ -28,28 +28,27 @@ namespace webrtc {
 class SincResamplerCallback {
  public:
   virtual ~SincResamplerCallback() {}
-  virtual void Run(int frames, float* destination) = 0;
+  virtual void Run(size_t frames, float* destination) = 0;
 };
 
 // SincResampler is a high-quality single-channel sample-rate converter.
 class SincResampler {
  public:
-  enum {
-    // The kernel size can be adjusted for quality (higher is better) at the
-    // expense of performance.  Must be a multiple of 32.
-    // TODO(dalecurtis): Test performance to see if we can jack this up to 64+.
-    kKernelSize = 32,
+  // The kernel size can be adjusted for quality (higher is better) at the
+  // expense of performance.  Must be a multiple of 32.
+  // TODO(dalecurtis): Test performance to see if we can jack this up to 64+.
+  static const size_t kKernelSize = 32;
 
-    // Default request size.  Affects how often and for how much SincResampler
-    // calls back for input.  Must be greater than kKernelSize.
-    kDefaultRequestSize = 512,
+  // Default request size.  Affects how often and for how much SincResampler
+  // calls back for input.  Must be greater than kKernelSize.
+  static const size_t kDefaultRequestSize = 512;
 
-    // The kernel offset count is used for interpolation and is the number of
-    // sub-sample kernel shifts.  Can be adjusted for quality (higher is better)
-    // at the expense of allocating more memory.
-    kKernelOffsetCount = 32,
-    kKernelStorageSize = kKernelSize * (kKernelOffsetCount + 1),
-  };
+  // The kernel offset count is used for interpolation and is the number of
+  // sub-sample kernel shifts.  Can be adjusted for quality (higher is better)
+  // at the expense of allocating more memory.
+  static const size_t kKernelOffsetCount = 32;
+  static const size_t kKernelStorageSize =
+      kKernelSize * (kKernelOffsetCount + 1);
 
   // Constructs a SincResampler with the specified |read_cb|, which is used to
   // acquire audio data for resampling.  |io_sample_rate_ratio| is the ratio
@@ -58,18 +57,18 @@ class SincResampler {
   // greater than kKernelSize.  Specify kDefaultRequestSize if there are no
   // request size constraints.
   SincResampler(double io_sample_rate_ratio,
-                int request_frames,
+                size_t request_frames,
                 SincResamplerCallback* read_cb);
   virtual ~SincResampler();
 
   // Resample |frames| of data from |read_cb_| into |destination|.
-  void Resample(int frames, float* destination);
+  void Resample(size_t frames, float* destination);
 
   // The maximum size in frames that guarantees Resample() will only make a
   // single call to |read_cb_| for more data.
-  int ChunkSize() const;
+  size_t ChunkSize() const;
 
-  int request_frames() const { return request_frames_; }
+  size_t request_frames() const { return request_frames_; }
 
   // Flush all buffered data and reset internal indices.  Not thread safe, do
   // not call while Resample() is in progress.
@@ -107,7 +106,7 @@ class SincResampler {
   static float Convolve_SSE(const float* input_ptr, const float* k1,
                             const float* k2,
                             double kernel_interpolation_factor);
-#elif defined(WEBRTC_ARCH_ARM_V7) || defined(WEBRTC_ARCH_ARM64_NEON)
+#elif defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
   static float Convolve_NEON(const float* input_ptr, const float* k1,
                              const float* k2,
                              double kernel_interpolation_factor);
@@ -127,13 +126,13 @@ class SincResampler {
   SincResamplerCallback* read_cb_;
 
   // The size (in samples) to request from each |read_cb_| execution.
-  const int request_frames_;
+  const size_t request_frames_;
 
   // The number of source frames processed per pass.
-  int block_size_;
+  size_t block_size_;
 
   // The size (in samples) of the internal buffer used by the resampler.
-  const int input_buffer_size_;
+  const size_t input_buffer_size_;
 
   // Contains kKernelOffsetCount kernels back-to-back, each of size kKernelSize.
   // The kernel offsets are sub-sample shifts of a windowed sinc shifted from
@@ -163,7 +162,7 @@ class SincResampler {
   float* r3_;
   float* r4_;
 
-  DISALLOW_COPY_AND_ASSIGN(SincResampler);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SincResampler);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_sse.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_sse.cc
index e00e0e5dbe..9e3953fede 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_sse.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_sse.cc
@@ -27,13 +27,13 @@ float SincResampler::Convolve_SSE(const float* input_ptr, const float* k1,
   // Based on |input_ptr| alignment, we need to use loadu or load.  Unrolling
   // these loops hurt performance in local testing.
   if (reinterpret_cast<uintptr_t>(input_ptr) & 0x0F) {
-    for (int i = 0; i < kKernelSize; i += 4) {
+    for (size_t i = 0; i < kKernelSize; i += 4) {
       m_input = _mm_loadu_ps(input_ptr + i);
       m_sums1 = _mm_add_ps(m_sums1, _mm_mul_ps(m_input, _mm_load_ps(k1 + i)));
       m_sums2 = _mm_add_ps(m_sums2, _mm_mul_ps(m_input, _mm_load_ps(k2 + i)));
     }
   } else {
-    for (int i = 0; i < kKernelSize; i += 4) {
+    for (size_t i = 0; i < kKernelSize; i += 4) {
       m_input = _mm_load_ps(input_ptr + i);
       m_sums1 = _mm_add_ps(m_sums1, _mm_mul_ps(m_input, _mm_load_ps(k1 + i)));
       m_sums2 = _mm_add_ps(m_sums2, _mm_mul_ps(m_input, _mm_load_ps(k2 + i)));
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
index 1aea902dd0..b8d6c341a2 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
@@ -21,9 +21,9 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
-#include "webrtc/system_wrappers/interface/stringize_macros.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/stringize_macros.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/test_suite.h"
 
 using testing::_;
@@ -36,7 +36,7 @@ static const double kKernelInterpolationFactor = 0.5;
 // Helper class to ensure ChunkedResample() functions properly.
 class MockSource : public SincResamplerCallback {
  public:
-  MOCK_METHOD2(Run, void(int frames, float* destination));
+  MOCK_METHOD2(Run, void(size_t frames, float* destination));
 };
 
 ACTION(ClearBuffer) {
@@ -61,7 +61,7 @@ TEST(SincResamplerTest, ChunkedResample) {
                           &mock_source);
 
   static const int kChunks = 2;
-  int max_chunk_size = resampler.ChunkSize() * kChunks;
+  size_t max_chunk_size = resampler.ChunkSize() * kChunks;
   rtc::scoped_ptr<float[]> resampled_destination(new float[max_chunk_size]);
 
   // Verify requesting ChunkSize() frames causes a single callback.
@@ -96,7 +96,7 @@ TEST(SincResamplerTest, Flush) {
   EXPECT_CALL(mock_source, Run(_, _))
       .Times(1).WillOnce(ClearBuffer());
   resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get());
-  for (int i = 0; i < resampler.ChunkSize() / 2; ++i)
+  for (size_t i = 0; i < resampler.ChunkSize() / 2; ++i)
     ASSERT_FLOAT_EQ(resampled_destination[i], 0);
 }
 
@@ -163,8 +163,8 @@ TEST(SincResamplerTest, Convolve) {
 #endif
 
 // Benchmark for the various Convolve() methods.  Make sure to build with
-// branding=Chrome so that DCHECKs are compiled out when benchmarking.  Original
-// benchmarks were run with --convolve-iterations=50000000.
+// branding=Chrome so that RTC_DCHECKs are compiled out when benchmarking.
+// Original benchmarks were run with --convolve-iterations=50000000.
 TEST(SincResamplerTest, ConvolveBenchmark) {
   // Initialize a dummy resampler.
   MockSource mock_source;
@@ -251,8 +251,10 @@ class SincResamplerTest
 TEST_P(SincResamplerTest, Resample) {
   // Make comparisons using one second of data.
   static const double kTestDurationSecs = 1;
-  const int input_samples = kTestDurationSecs * input_rate_;
-  const int output_samples = kTestDurationSecs * output_rate_;
+  const size_t input_samples =
+      static_cast<size_t>(kTestDurationSecs * input_rate_);
+  const size_t output_samples =
+      static_cast<size_t>(kTestDurationSecs * output_rate_);
 
   // Nyquist frequency for the input sampling rate.
   const double input_nyquist_freq = 0.5 * input_rate_;
@@ -302,7 +304,7 @@ TEST_P(SincResamplerTest, Resample) {
   int minimum_rate = std::min(input_rate_, output_rate_);
   double low_frequency_range = kLowFrequencyNyquistRange * 0.5 * minimum_rate;
   double high_frequency_range = kHighFrequencyNyquistRange * 0.5 * minimum_rate;
-  for (int i = 0; i < output_samples; ++i) {
+  for (size_t i = 0; i < output_samples; ++i) {
     double error = fabs(resampled_destination[i] - pure_destination[i]);
 
     if (pure_source.Frequency(i) < low_frequency_range) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
index d38263c682..dc3f7c92f0 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
@@ -9,7 +9,9 @@
  */
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
+#ifndef _USE_MATH_DEFINES
 #define _USE_MATH_DEFINES
+#endif
 
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
 
@@ -18,7 +20,9 @@
 namespace webrtc {
 
 SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate,
-  int samples, double max_frequency, double delay_samples)
+                                                         size_t samples,
+                                                         double max_frequency,
+                                                         double delay_samples)
     : sample_rate_(sample_rate),
       total_samples_(samples),
       max_frequency_(max_frequency),
@@ -29,19 +33,18 @@ SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate,
   k_ = (max_frequency_ - kMinFrequency) / duration;
 }
 
-void SinusoidalLinearChirpSource::Run(int frames, float* destination) {
-  for (int i = 0; i < frames; ++i, ++current_index_) {
+void SinusoidalLinearChirpSource::Run(size_t frames, float* destination) {
+  for (size_t i = 0; i < frames; ++i, ++current_index_) {
     // Filter out frequencies higher than Nyquist.
     if (Frequency(current_index_) > 0.5 * sample_rate_) {
       destination[i] = 0;
     } else {
       // Calculate time in seconds.
-      double t = (static_cast<double>(current_index_) - delay_samples_) /
-          sample_rate_;
-      if (t < 0) {
+      if (current_index_ < delay_samples_) {
         destination[i] = 0;
       } else {
         // Sinusoidal linear chirp.
+        double t = (current_index_ - delay_samples_) / sample_rate_;
         destination[i] =
             sin(2 * M_PI * (kMinFrequency * t + (k_ / 2) * t * t));
       }
@@ -49,7 +52,7 @@ void SinusoidalLinearChirpSource::Run(int frames, float* destination) {
   }
 }
 
-double SinusoidalLinearChirpSource::Frequency(int position) {
+double SinusoidalLinearChirpSource::Frequency(size_t position) {
   return kMinFrequency + (position - delay_samples_) *
       (max_frequency_ - kMinFrequency) / total_samples_;
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h b/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h
index d6b8ce31e9..1807f86a19 100644
--- a/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h
+++ b/media/webrtc/trunk/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h
@@ -26,28 +26,28 @@ class SinusoidalLinearChirpSource : public SincResamplerCallback {
  public:
   // |delay_samples| can be used to insert a fractional sample delay into the
   // source.  It will produce zeros until non-negative time is reached.
-  SinusoidalLinearChirpSource(int sample_rate, int samples,
+  SinusoidalLinearChirpSource(int sample_rate, size_t samples,
                               double max_frequency, double delay_samples);
 
   virtual ~SinusoidalLinearChirpSource() {}
 
-  void Run(int frames, float* destination) override;
+  void Run(size_t frames, float* destination) override;
 
-  double Frequency(int position);
+  double Frequency(size_t position);
 
  private:
   enum {
     kMinFrequency = 5
   };
 
-  double sample_rate_;
-  int total_samples_;
+  int sample_rate_;
+  size_t total_samples_;
   double max_frequency_;
   double k_;
-  int current_index_;
+  size_t current_index_;
   double delay_samples_;
 
-  DISALLOW_COPY_AND_ASSIGN(SinusoidalLinearChirpSource);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SinusoidalLinearChirpSource);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/auto_correlation.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/auto_correlation.c
index 9fb9824eb8..fda4fffeed 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/auto_correlation.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/auto_correlation.c
@@ -10,22 +10,19 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
-                              int in_vector_length,
-                              int order,
-                              int32_t* result,
-                              int* scale) {
+#include <assert.h>
+
+size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
+                                 size_t in_vector_length,
+                                 size_t order,
+                                 int32_t* result,
+                                 int* scale) {
   int32_t sum = 0;
-  int i = 0, j = 0;
+  size_t i = 0, j = 0;
   int16_t smax = 0;
   int scaling = 0;
 
-  if (order > in_vector_length) {
-    /* Undefined */
-    return -1;
-  } else if (order < 0) {
-    order = in_vector_length;
-  }
+  assert(order <= in_vector_length);
 
   // Find the maximum absolute value of the samples.
   smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
@@ -36,7 +33,7 @@ int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
     scaling = 0;
   } else {
     // Number of bits in the sum loop.
-    int nbits = WebRtcSpl_GetSizeInBits(in_vector_length);
+    int nbits = WebRtcSpl_GetSizeInBits((uint32_t)in_vector_length);
     // Number of bits to normalize smax.
     int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
 
@@ -51,7 +48,7 @@ int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
   for (i = 0; i < order + 1; i++) {
     sum = 0;
     /* Unroll the loop to improve performance. */
-    for (j = 0; j < in_vector_length - i - 3; j += 4) {
+    for (j = 0; i + j + 3 < in_vector_length; j += 4) {
       sum += (in_vector[j + 0] * in_vector[i + j + 0]) >> scaling;
       sum += (in_vector[j + 1] * in_vector[i + j + 1]) >> scaling;
       sum += (in_vector[j + 2] * in_vector[i + j + 2]) >> scaling;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S b/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
index e7f8a819bd..93de99f51b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
@@ -12,7 +12,7 @@
 @ for ARMv5 platforms.
 @ Reference C code is in file complex_bit_reverse.c. Bit-exact.
 
-#include "webrtc/system_wrappers/interface/asm_defines.h"
+#include "webrtc/system_wrappers/include/asm_defines.h"
 
 GLOBAL_FUNCTION WebRtcSpl_ComplexBitReverse
 .align  2
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_fft.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_fft.c
index 74b4258a8e..97ebacc498 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_fft.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/complex_fft.c
@@ -126,11 +126,9 @@ int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
                          [wri]"r"(wri),
                          [cfftrnd]"r"(CFFTRND));
 #else
-                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
+                    tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CFFTRND;
 
-                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
+                    ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CFFTRND;
 #endif
 
                     tr32 >>= 15 - CFFTSFT;
@@ -159,7 +157,8 @@ int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
 
 int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
 {
-    int i, j, l, k, istep, n, m, scale, shift;
+    size_t i, j, l, istep, n, m;
+    int k, scale, shift;
     int16_t wr, wi;
     int32_t tr32, ti32, qr32, qi32;
     int32_t tmp32, round2;
@@ -183,7 +182,7 @@ int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
         shift = 0;
         round2 = 8192;
 
-        tmp32 = (int32_t)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
+        tmp32 = WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
         if (tmp32 > 13573)
         {
             shift++;
@@ -270,11 +269,9 @@ int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
                     );
 #else
 
-                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
+                    tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CIFFTRND;
 
-                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
+                    ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CIFFTRND;
 #endif
                     tr32 >>= 15 - CIFFTSFT;
                     ti32 >>= 15 - CIFFTSFT;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/copy_set_operations.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/copy_set_operations.c
index 84d3bc429c..9d7cf47e3b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/copy_set_operations.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/copy_set_operations.c
@@ -26,9 +26,9 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 
-void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, int length)
+void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, size_t length)
 {
-    int j;
+    size_t j;
     int16_t *arrptr = ptr;
 
     for (j = length; j > 0; j--)
@@ -37,9 +37,9 @@ void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, int length)
     }
 }
 
-void WebRtcSpl_MemSetW32(int32_t *ptr, int32_t set_value, int length)
+void WebRtcSpl_MemSetW32(int32_t *ptr, int32_t set_value, size_t length)
 {
-    int j;
+    size_t j;
     int32_t *arrptr = ptr;
 
     for (j = length; j > 0; j--)
@@ -48,9 +48,11 @@ void WebRtcSpl_MemSetW32(int32_t *ptr, int32_t set_value, int length)
     }
 }
 
-void WebRtcSpl_MemCpyReversedOrder(int16_t* dest, int16_t* source, int length)
+void WebRtcSpl_MemCpyReversedOrder(int16_t* dest,
+                                   int16_t* source,
+                                   size_t length)
 {
-    int j;
+    size_t j;
     int16_t* destPtr = dest;
     int16_t* sourcePtr = source;
 
@@ -61,20 +63,20 @@ void WebRtcSpl_MemCpyReversedOrder(int16_t* dest, int16_t* source, int length)
 }
 
 void WebRtcSpl_CopyFromEndW16(const int16_t *vector_in,
-                              int length,
-                              int samples,
+                              size_t length,
+                              size_t samples,
                               int16_t *vector_out)
 {
     // Copy the last <samples> of the input vector to vector_out
     WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
 }
 
-void WebRtcSpl_ZerosArrayW16(int16_t *vector, int length)
+void WebRtcSpl_ZerosArrayW16(int16_t *vector, size_t length)
 {
     WebRtcSpl_MemSetW16(vector, 0, length);
 }
 
-void WebRtcSpl_ZerosArrayW32(int32_t *vector, int length)
+void WebRtcSpl_ZerosArrayW32(int32_t *vector, size_t length)
 {
     WebRtcSpl_MemSetW32(vector, 0, length);
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation.c
index 42000d608d..d7c9f2b9af 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation.c
@@ -14,18 +14,17 @@
 void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
                                  const int16_t* seq1,
                                  const int16_t* seq2,
-                                 int16_t dim_seq,
-                                 int16_t dim_cross_correlation,
-                                 int16_t right_shifts,
-                                 int16_t step_seq2) {
-  int i = 0, j = 0;
+                                 size_t dim_seq,
+                                 size_t dim_cross_correlation,
+                                 int right_shifts,
+                                 int step_seq2) {
+  size_t i = 0, j = 0;
 
   for (i = 0; i < dim_cross_correlation; i++) {
-    *cross_correlation = 0;
-    /* Unrolling doesn't seem to improve performance. */
-    for (j = 0; j < dim_seq; j++) {
-      *cross_correlation += (seq1[j] * seq2[step_seq2 * i + j]) >> right_shifts;
-    }
-    cross_correlation++;
+    int32_t corr = 0;
+    for (j = 0; j < dim_seq; j++)
+      corr += (seq1[j] * seq2[j]) >> right_shifts;
+    seq2 += step_seq2;
+    *cross_correlation++ = corr;
   }
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_mips.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_mips.c
index 7d9a6c6442..b2364026c6 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_mips.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_mips.c
@@ -13,10 +13,10 @@
 void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
                                      const int16_t* seq1,
                                      const int16_t* seq2,
-                                     int16_t dim_seq,
-                                     int16_t dim_cross_correlation,
-                                     int16_t right_shifts,
-                                     int16_t step_seq2) {
+                                     size_t dim_seq,
+                                     size_t dim_cross_correlation,
+                                     int right_shifts,
+                                     int step_seq2) {
 
   int32_t t0 = 0, t1 = 0, t2 = 0, t3 = 0, sum = 0;
   int16_t *pseq2 = NULL;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_neon.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_neon.c
index c358c701af..918b6715cd 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_neon.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/cross_correlation_neon.c
@@ -15,19 +15,14 @@
 static inline void DotProductWithScaleNeon(int32_t* cross_correlation,
                                            const int16_t* vector1,
                                            const int16_t* vector2,
-                                           int length,
+                                           size_t length,
                                            int scaling) {
-  int i = 0;
-  int len1 = length >> 3;
-  int len2 = length & 7;
+  size_t i = 0;
+  size_t len1 = length >> 3;
+  size_t len2 = length & 7;
   int64x2_t sum0 = vdupq_n_s64(0);
   int64x2_t sum1 = vdupq_n_s64(0);
 
-  if (length < 0) {
-    *cross_correlation = 0;
-    return;
-  }
-
   for (i = len1; i > 0; i -= 1) {
     int16x8_t seq1_16x8 = vld1q_s16(vector1);
     int16x8_t seq2_16x8 = vld1q_s16(vector2);
@@ -72,11 +67,11 @@ static inline void DotProductWithScaleNeon(int32_t* cross_correlation,
 void WebRtcSpl_CrossCorrelationNeon(int32_t* cross_correlation,
                                     const int16_t* seq1,
                                     const int16_t* seq2,
-                                    int16_t dim_seq,
-                                    int16_t dim_cross_correlation,
-                                    int16_t right_shifts,
-                                    int16_t step_seq2) {
-  int i = 0;
+                                    size_t dim_seq,
+                                    size_t dim_cross_correlation,
+                                    int right_shifts,
+                                    int step_seq2) {
+  size_t i = 0;
 
   for (i = 0; i < dim_cross_correlation; i++) {
     const int16_t* seq1_ptr = seq1;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/division_operations.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/division_operations.c
index 6aeb0fb2bf..eaa06a1ff9 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/division_operations.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/division_operations.c
@@ -106,8 +106,7 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
     // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
 
     // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
-            + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
+    tmpW32 = (den_hi * approx << 1) + ((den_low * approx >> 15) << 1);
     // tmpW32 = den * approx
 
     tmpW32 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
@@ -117,8 +116,7 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
     tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
 
     // tmpW32 = 1/den in Q29
-    tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
-            >> 15)) << 1);
+    tmpW32 = (tmp_hi * approx + (tmp_low * approx >> 15)) << 1;
 
     // 1/den in hi and low format
     tmp_hi = (int16_t)(tmpW32 >> 16);
@@ -130,8 +128,8 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
 
     // num * (1/den) by 32 bit multiplication (result in Q28)
 
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
-            >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
+    tmpW32 = num_hi * tmp_hi + (num_hi * tmp_low >> 15) +
+        (num_low * tmp_hi >> 15);
 
     // Put result in Q31 (convert from Q28)
     tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/dot_product_with_scale.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/dot_product_with_scale.c
index 389bcf0578..1302d62541 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/dot_product_with_scale.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/dot_product_with_scale.c
@@ -12,13 +12,13 @@
 
 int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
                                       const int16_t* vector2,
-                                      int length,
+                                      size_t length,
                                       int scaling) {
   int32_t sum = 0;
-  int i = 0;
+  size_t i = 0;
 
   /* Unroll the loop to improve performance. */
-  for (i = 0; i < length - 3; i += 4) {
+  for (i = 0; i + 3 < length; i += 4) {
     sum += (vector1[i + 0] * vector2[i + 0]) >> scaling;
     sum += (vector1[i + 1] * vector2[i + 1]) >> scaling;
     sum += (vector1[i + 2] * vector2[i + 2]) >> scaling;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast.c
index 179c36a25c..726a88819a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast.c
@@ -13,20 +13,20 @@
 // TODO(Bjornv): Change the function parameter order to WebRTC code style.
 // C version of WebRtcSpl_DownsampleFast() for generic platforms.
 int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
-                              int data_in_length,
+                              size_t data_in_length,
                               int16_t* data_out,
-                              int data_out_length,
+                              size_t data_out_length,
                               const int16_t* __restrict coefficients,
-                              int coefficients_length,
+                              size_t coefficients_length,
                               int factor,
-                              int delay) {
-  int i = 0;
-  int j = 0;
+                              size_t delay) {
+  size_t i = 0;
+  size_t j = 0;
   int32_t out_s32 = 0;
-  int endpos = delay + factor * (data_out_length - 1) + 1;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
 
   // Return error if any of the running conditions doesn't meet.
-  if (data_out_length <= 0 || coefficients_length <= 0
+  if (data_out_length == 0 || coefficients_length == 0
                            || data_in_length < endpos) {
     return -1;
   }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_mips.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_mips.c
index dbde43d30b..ac39401abb 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_mips.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_mips.c
@@ -12,18 +12,18 @@
 
 // Version of WebRtcSpl_DownsampleFast() for MIPS platforms.
 int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in,
-                                  int data_in_length,
+                                  size_t data_in_length,
                                   int16_t* data_out,
-                                  int data_out_length,
+                                  size_t data_out_length,
                                   const int16_t* __restrict coefficients,
-                                  int coefficients_length,
+                                  size_t coefficients_length,
                                   int factor,
-                                  int delay) {
+                                  size_t delay) {
   int i;
   int j;
   int k;
   int32_t out_s32 = 0;
-  int endpos = delay + factor * (data_out_length - 1) + 1;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
 
   int32_t  tmp1, tmp2, tmp3, tmp4, factor_2;
   int16_t* p_coefficients;
@@ -36,7 +36,7 @@ int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in,
 #endif  // #if !defined(MIPS_DSP_R1_LE)
 
   // Return error if any of the running conditions doesn't meet.
-  if (data_out_length <= 0 || coefficients_length <= 0
+  if (data_out_length == 0 || coefficients_length == 0
                            || data_in_length < endpos) {
     return -1;
   }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_neon.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_neon.c
index f775e6936a..58732dab1c 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_neon.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/downsample_fast_neon.c
@@ -15,22 +15,22 @@
 // NEON intrinsics version of WebRtcSpl_DownsampleFast()
 // for ARM 32-bit/64-bit platforms.
 int WebRtcSpl_DownsampleFastNeon(const int16_t* data_in,
-                                 int data_in_length,
+                                 size_t data_in_length,
                                  int16_t* data_out,
-                                 int data_out_length,
+                                 size_t data_out_length,
                                  const int16_t* __restrict coefficients,
-                                 int coefficients_length,
+                                 size_t coefficients_length,
                                  int factor,
-                                 int delay) {
-  int i = 0;
-  int j = 0;
+                                 size_t delay) {
+  size_t i = 0;
+  size_t j = 0;
   int32_t out_s32 = 0;
-  int endpos = delay + factor * (data_out_length - 1) + 1;
-  int res = data_out_length & 0x7;
-  int endpos1 = endpos - factor * res;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
+  size_t res = data_out_length & 0x7;
+  size_t endpos1 = endpos - factor * res;
 
   // Return error if any of the running conditions doesn't meet.
-  if (data_out_length <= 0 || coefficients_length <= 0
+  if (data_out_length == 0 || coefficients_length == 0
                            || data_in_length < endpos) {
     return -1;
   }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/energy.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/energy.c
index 0611ad3e92..e83f1a698f 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/energy.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/energy.c
@@ -17,12 +17,15 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor)
+int32_t WebRtcSpl_Energy(int16_t* vector,
+                         size_t vector_length,
+                         int* scale_factor)
 {
     int32_t en = 0;
-    int i;
-    int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
-    int looptimes = vector_length;
+    size_t i;
+    int scaling =
+        WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
+    size_t looptimes = vector_length;
     int16_t *vectorptr = vector;
 
     for (i = 0; i < looptimes; i++)
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar.c
index 7386808c61..dfbc4c2f7a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar.c
@@ -17,21 +17,21 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-int WebRtcSpl_FilterAR(const int16_t* a,
-                       int a_length,
-                       const int16_t* x,
-                       int x_length,
-                       int16_t* state,
-                       int state_length,
-                       int16_t* state_low,
-                       int state_low_length,
-                       int16_t* filtered,
-                       int16_t* filtered_low,
-                       int filtered_low_length)
+size_t WebRtcSpl_FilterAR(const int16_t* a,
+                          size_t a_length,
+                          const int16_t* x,
+                          size_t x_length,
+                          int16_t* state,
+                          size_t state_length,
+                          int16_t* state_low,
+                          size_t state_low_length,
+                          int16_t* filtered,
+                          int16_t* filtered_low,
+                          size_t filtered_low_length)
 {
     int32_t o;
     int32_t oLOW;
-    int i, j, stop;
+    size_t i, j, stop;
     const int16_t* x_ptr = &x[0];
     int16_t* filteredFINAL_ptr = filtered;
     int16_t* filteredFINAL_LOW_ptr = filtered_low;
@@ -51,13 +51,13 @@ int WebRtcSpl_FilterAR(const int16_t* a,
         stop = (i < a_length) ? i + 1 : a_length;
         for (j = 1; j < stop; j++)
         {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
+          o -= *a_ptr * *filtered_ptr--;
+          oLOW -= *a_ptr++ * *filtered_low_ptr--;
         }
         for (j = i + 1; j < a_length; j++)
         {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
+          o -= *a_ptr * *state_ptr--;
+          oLOW -= *a_ptr++ * *state_low_ptr--;
         }
 
         o += (oLOW >> 12);
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
index cfd82ca8cf..70001a0882 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -16,10 +16,10 @@
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
                                int16_t* data_out,
                                const int16_t* __restrict coefficients,
-                               int coefficients_length,
-                               int data_length) {
-  int i = 0;
-  int j = 0;
+                               size_t coefficients_length,
+                               size_t data_length) {
+  size_t i = 0;
+  size_t j = 0;
 
   assert(data_length > 0);
   assert(coefficients_length > 1);
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
index ff60cc6198..f16362738a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
@@ -35,7 +35,7 @@
 @ r11: Scratch
 @ r12: &coefficients[j]
 
-#include "webrtc/system_wrappers/interface/asm_defines.h"
+#include "webrtc/system_wrappers/include/asm_defines.h"
 
 GLOBAL_FUNCTION WebRtcSpl_FilterARFastQ12
 .align  2
@@ -155,10 +155,13 @@ END:
 @void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
 @                               int16_t* data_out,
 @                               int16_t* __restrict coefficients,
-@                               int coefficients_length,
-@                               int data_length) {
-@  int i = 0;
-@  int j = 0;
+@                               size_t coefficients_length,
+@                               size_t data_length) {
+@  size_t i = 0;
+@  size_t j = 0;
+@
+@  assert(data_length > 0);
+@  assert(coefficients_length > 1);
 @
 @  for (i = 0; i < data_length - 1; i += 2) {
 @    int32_t output1 = 0;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
index e77e1f578c..03847018e3 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
@@ -14,8 +14,8 @@
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
                                int16_t* data_out,
                                const int16_t* __restrict coefficients,
-                               int coefficients_length,
-                               int data_length) {
+                               size_t coefficients_length,
+                               size_t data_length) {
   int r0, r1, r2, r3;
   int coef0, offset;
   int i, j, k;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
index 943b01c1d9..f4d9a3d303 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
@@ -20,21 +20,17 @@
 void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
                                int16_t* out_ptr,
                                const int16_t* B,
-                               int16_t B_length,
-                               int16_t length)
+                               size_t B_length,
+                               size_t length)
 {
-    int32_t o;
-    int i, j;
+    size_t i, j;
     for (i = 0; i < length; i++)
     {
-        const int16_t* b_ptr = &B[0];
-        const int16_t* x_ptr = &in_ptr[i];
-
-        o = (int32_t)0;
+        int32_t o = 0;
 
         for (j = 0; j < B_length; j++)
         {
-            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
+          o += B[j] * in_ptr[i - j];
         }
 
         // If output is higher than 32768, saturate it. Same with negative side
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_hanning_window.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_hanning_window.c
index 519b665843..d83ac21682 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_hanning_window.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_hanning_window.c
@@ -53,15 +53,15 @@ static const int16_t kHanningTable[] = {
 16354,  16362,  16369,  16374,  16378,  16382,  16383,  16384
 };
 
-void WebRtcSpl_GetHanningWindow(int16_t *v, int16_t size)
+void WebRtcSpl_GetHanningWindow(int16_t *v, size_t size)
 {
-    int jj;
+    size_t jj;
     int16_t *vptr1;
 
     int32_t index;
     int32_t factor = ((int32_t)0x40000000);
 
-    factor = WebRtcSpl_DivW32W16(factor, size);
+    factor = WebRtcSpl_DivW32W16(factor, (int16_t)size);
     if (size < 513)
         index = (int32_t)-0x200000;
     else
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_scaling_square.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_scaling_square.c
index 9b6049c24f..82e3c8b09c 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_scaling_square.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/get_scaling_square.c
@@ -18,16 +18,16 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
-                                   int in_vector_length,
-                                   int times)
+                                   size_t in_vector_length,
+                                   size_t times)
 {
-    int16_t nbits = WebRtcSpl_GetSizeInBits(times);
-    int i;
+    int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
+    size_t i;
     int16_t smax = -1;
     int16_t sabs;
     int16_t *sptr = in_vector;
     int16_t t;
-    int looptimes = in_vector_length;
+    size_t looptimes = in_vector_length;
 
     for (i = looptimes; i > 0; i--)
     {
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/ilbc_specific_functions.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
index de870b239b..301a922d79 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
@@ -23,10 +23,10 @@
 
 void WebRtcSpl_ReverseOrderMultArrayElements(int16_t *out, const int16_t *in,
                                              const int16_t *win,
-                                             int16_t vector_length,
+                                             size_t vector_length,
                                              int16_t right_shifts)
 {
-    int i;
+    size_t i;
     int16_t *outptr = out;
     const int16_t *inptr = in;
     const int16_t *winptr = win;
@@ -37,10 +37,10 @@ void WebRtcSpl_ReverseOrderMultArrayElements(int16_t *out, const int16_t *in,
 }
 
 void WebRtcSpl_ElementwiseVectorMult(int16_t *out, const int16_t *in,
-                                     const int16_t *win, int16_t vector_length,
+                                     const int16_t *win, size_t vector_length,
                                      int16_t right_shifts)
 {
-    int i;
+    size_t i;
     int16_t *outptr = out;
     const int16_t *inptr = in;
     const int16_t *winptr = win;
@@ -51,10 +51,10 @@ void WebRtcSpl_ElementwiseVectorMult(int16_t *out, const int16_t *in,
 }
 
 void WebRtcSpl_AddVectorsAndShift(int16_t *out, const int16_t *in1,
-                                  const int16_t *in2, int16_t vector_length,
+                                  const int16_t *in2, size_t vector_length,
                                   int16_t right_shifts)
 {
-    int i;
+    size_t i;
     int16_t *outptr = out;
     const int16_t *in1ptr = in1;
     const int16_t *in2ptr = in2;
@@ -66,34 +66,25 @@ void WebRtcSpl_AddVectorsAndShift(int16_t *out, const int16_t *in1,
 
 void WebRtcSpl_AddAffineVectorToVector(int16_t *out, int16_t *in,
                                        int16_t gain, int32_t add_constant,
-                                       int16_t right_shifts, int vector_length)
+                                       int16_t right_shifts,
+                                       size_t vector_length)
 {
-    int16_t *inPtr;
-    int16_t *outPtr;
-    int i;
+    size_t i;
 
-    inPtr = in;
-    outPtr = out;
     for (i = 0; i < vector_length; i++)
     {
-        (*outPtr++) += (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (int32_t)add_constant) >> right_shifts);
+      out[i] += (int16_t)((in[i] * gain + add_constant) >> right_shifts);
     }
 }
 
 void WebRtcSpl_AffineTransformVector(int16_t *out, int16_t *in,
                                      int16_t gain, int32_t add_constant,
-                                     int16_t right_shifts, int vector_length)
+                                     int16_t right_shifts, size_t vector_length)
 {
-    int16_t *inPtr;
-    int16_t *outPtr;
-    int i;
+    size_t i;
 
-    inPtr = in;
-    outPtr = out;
     for (i = 0; i < vector_length; i++)
     {
-        (*outPtr++) = (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (int32_t)add_constant) >> right_shifts);
+      out[i] = (int16_t)((in[i] * gain + add_constant) >> right_shifts);
     }
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/signal_processing_library.h
index 7014fa1df5..2e96883e6d 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -105,9 +105,9 @@ extern "C" {
 #include "webrtc/common_audio/signal_processing/include/spl_inl.h"
 
 // Initialize SPL. Currently it contains only function pointer initialization.
-// If the underlying platform is known to be ARM-Neon (WEBRTC_ARCH_ARM_NEON
-// defined), the pointers will be assigned to code optimized for Neon; otherwise
-// if run-time Neon detection (WEBRTC_DETECT_ARM_NEON) is enabled, the pointers
+// If the underlying platform is known to be ARM-Neon (WEBRTC_HAS_NEON defined),
+// the pointers will be assigned to code optimized for Neon; otherwise
+// if run-time Neon detection (WEBRTC_DETECT_NEON) is enabled, the pointers
 // will be assigned to either Neon code or generic C code; otherwise, generic C
 // code will be assigned.
 // Note that this function MUST be called in any application that uses SPL
@@ -115,28 +115,28 @@ extern "C" {
 void WebRtcSpl_Init();
 
 int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
-                                   int in_vector_length,
-                                   int times);
+                                   size_t in_vector_length,
+                                   size_t times);
 
 // Copy and set operations. Implementation in copy_set_operations.c.
 // Descriptions at bottom of file.
 void WebRtcSpl_MemSetW16(int16_t* vector,
                          int16_t set_value,
-                         int vector_length);
+                         size_t vector_length);
 void WebRtcSpl_MemSetW32(int32_t* vector,
                          int32_t set_value,
-                         int vector_length);
+                         size_t vector_length);
 void WebRtcSpl_MemCpyReversedOrder(int16_t* out_vector,
                                    int16_t* in_vector,
-                                   int vector_length);
+                                   size_t vector_length);
 void WebRtcSpl_CopyFromEndW16(const int16_t* in_vector,
-                              int in_vector_length,
-                              int samples,
+                              size_t in_vector_length,
+                              size_t samples,
                               int16_t* out_vector);
 void WebRtcSpl_ZerosArrayW16(int16_t* vector,
-                             int vector_length);
+                             size_t vector_length);
 void WebRtcSpl_ZerosArrayW32(int32_t* vector,
-                             int vector_length);
+                             size_t vector_length);
 // End: Copy and set operations.
 
 
@@ -149,17 +149,15 @@ void WebRtcSpl_ZerosArrayW32(int32_t* vector,
 //      - vector : 16-bit input vector.
 //      - length : Number of samples in vector.
 //
-// Return value  : Maximum absolute value in vector;
-//                 or -1, if (vector == NULL || length <= 0).
-typedef int16_t (*MaxAbsValueW16)(const int16_t* vector, int length);
+// Return value  : Maximum absolute value in vector.
+typedef int16_t (*MaxAbsValueW16)(const int16_t* vector, size_t length);
 extern MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
-int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, int length);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
-int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, int length);
+int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
-int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length);
+int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, size_t length);
 #endif
 
 // Returns the largest absolute value in a signed 32-bit vector.
@@ -168,17 +166,15 @@ int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length);
 //      - vector : 32-bit input vector.
 //      - length : Number of samples in vector.
 //
-// Return value  : Maximum absolute value in vector;
-//                 or -1, if (vector == NULL || length <= 0).
-typedef int32_t (*MaxAbsValueW32)(const int32_t* vector, int length);
+// Return value  : Maximum absolute value in vector.
+typedef int32_t (*MaxAbsValueW32)(const int32_t* vector, size_t length);
 extern MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32;
-int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, int length);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
-int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, int length);
+int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, size_t length);
 #endif
 #if defined(MIPS_DSP_R1_LE)
-int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, int length);
+int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, size_t length);
 #endif
 
 // Returns the maximum value of a 16-bit vector.
@@ -188,18 +184,14 @@ int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Maximum sample value in |vector|.
-//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD16_MIN
-//                 is returned. Note that WEBRTC_SPL_WORD16_MIN is a feasible
-//                 value and we can't catch errors purely based on it.
-typedef int16_t (*MaxValueW16)(const int16_t* vector, int length);
+typedef int16_t (*MaxValueW16)(const int16_t* vector, size_t length);
 extern MaxValueW16 WebRtcSpl_MaxValueW16;
-int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, int length);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
-int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, int length);
+int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
-int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, int length);
+int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length);
 #endif
 
 // Returns the maximum value of a 32-bit vector.
@@ -209,18 +201,14 @@ int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Maximum sample value in |vector|.
-//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD32_MIN
-//                 is returned. Note that WEBRTC_SPL_WORD32_MIN is a feasible
-//                 value and we can't catch errors purely based on it.
-typedef int32_t (*MaxValueW32)(const int32_t* vector, int length);
+typedef int32_t (*MaxValueW32)(const int32_t* vector, size_t length);
 extern MaxValueW32 WebRtcSpl_MaxValueW32;
-int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, int length);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
-int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, int length);
+int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
-int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, int length);
+int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length);
 #endif
 
 // Returns the minimum value of a 16-bit vector.
@@ -230,18 +218,14 @@ int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Minimum sample value in |vector|.
-//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD16_MAX
-//                 is returned. Note that WEBRTC_SPL_WORD16_MAX is a feasible
-//                 value and we can't catch errors purely based on it.
-typedef int16_t (*MinValueW16)(const int16_t* vector, int length);
+typedef int16_t (*MinValueW16)(const int16_t* vector, size_t length);
 extern MinValueW16 WebRtcSpl_MinValueW16;
-int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, int length);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
-int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, int length);
+int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
-int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, int length);
+int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length);
 #endif
 
 // Returns the minimum value of a 32-bit vector.
@@ -251,18 +235,14 @@ int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Minimum sample value in |vector|.
-//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD32_MAX
-//                 is returned. Note that WEBRTC_SPL_WORD32_MAX is a feasible
-//                 value and we can't catch errors purely based on it.
-typedef int32_t (*MinValueW32)(const int32_t* vector, int length);
+typedef int32_t (*MinValueW32)(const int32_t* vector, size_t length);
 extern MinValueW32 WebRtcSpl_MinValueW32;
-int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, int length);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
-int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, int length);
+int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
-int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, int length);
+int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, size_t length);
 #endif
 
 // Returns the vector index to the largest absolute value of a 16-bit vector.
@@ -271,12 +251,11 @@ int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, int length);
 //      - vector : 16-bit input vector.
 //      - length : Number of samples in vector.
 //
-// Return value  : Index to the maximum absolute value in vector, or -1,
-//                 if (vector == NULL || length <= 0).
+// Return value  : Index to the maximum absolute value in vector.
 //                 If there are multiple equal maxima, return the index of the
 //                 first. -32768 will always have precedence over 32767 (despite
-//                 -32768 presenting an int16 absolute value of 32767);
-int WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, int length);
+//                 -32768 presenting an int16 absolute value of 32767).
+size_t WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, size_t length);
 
 // Returns the vector index to the maximum sample value of a 16-bit vector.
 //
@@ -285,9 +264,8 @@ int WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Index to the maximum value in vector (if multiple
-//                 indexes have the maximum, return the first);
-//                 or -1, if (vector == NULL || length <= 0).
-int WebRtcSpl_MaxIndexW16(const int16_t* vector, int length);
+//                 indexes have the maximum, return the first).
+size_t WebRtcSpl_MaxIndexW16(const int16_t* vector, size_t length);
 
 // Returns the vector index to the maximum sample value of a 32-bit vector.
 //
@@ -296,9 +274,8 @@ int WebRtcSpl_MaxIndexW16(const int16_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Index to the maximum value in vector (if multiple
-//                 indexes have the maximum, return the first);
-//                 or -1, if (vector == NULL || length <= 0).
-int WebRtcSpl_MaxIndexW32(const int32_t* vector, int length);
+//                 indexes have the maximum, return the first).
+size_t WebRtcSpl_MaxIndexW32(const int32_t* vector, size_t length);
 
 // Returns the vector index to the minimum sample value of a 16-bit vector.
 //
@@ -307,9 +284,8 @@ int WebRtcSpl_MaxIndexW32(const int32_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Index to the mimimum value in vector  (if multiple
-//                 indexes have the minimum, return the first);
-//                 or -1, if (vector == NULL || length <= 0).
-int WebRtcSpl_MinIndexW16(const int16_t* vector, int length);
+//                 indexes have the minimum, return the first).
+size_t WebRtcSpl_MinIndexW16(const int16_t* vector, size_t length);
 
 // Returns the vector index to the minimum sample value of a 32-bit vector.
 //
@@ -318,9 +294,8 @@ int WebRtcSpl_MinIndexW16(const int16_t* vector, int length);
 //      - length : Number of samples in vector.
 //
 // Return value  : Index to the mimimum value in vector  (if multiple
-//                 indexes have the minimum, return the first);
-//                 or -1, if (vector == NULL || length <= 0).
-int WebRtcSpl_MinIndexW32(const int32_t* vector, int length);
+//                 indexes have the minimum, return the first).
+size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length);
 
 // End: Minimum and maximum operations.
 
@@ -328,33 +303,33 @@ int WebRtcSpl_MinIndexW32(const int32_t* vector, int length);
 // Vector scaling operations. Implementation in vector_scaling_operations.c.
 // Description at bottom of file.
 void WebRtcSpl_VectorBitShiftW16(int16_t* out_vector,
-                                 int16_t vector_length,
+                                 size_t vector_length,
                                  const int16_t* in_vector,
                                  int16_t right_shifts);
 void WebRtcSpl_VectorBitShiftW32(int32_t* out_vector,
-                                 int16_t vector_length,
+                                 size_t vector_length,
                                  const int32_t* in_vector,
                                  int16_t right_shifts);
 void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out_vector,
-                                      int vector_length,
+                                      size_t vector_length,
                                       const int32_t* in_vector,
                                       int right_shifts);
 void WebRtcSpl_ScaleVector(const int16_t* in_vector,
                            int16_t* out_vector,
                            int16_t gain,
-                           int16_t vector_length,
+                           size_t vector_length,
                            int16_t right_shifts);
 void WebRtcSpl_ScaleVectorWithSat(const int16_t* in_vector,
                                   int16_t* out_vector,
                                   int16_t gain,
-                                  int16_t vector_length,
+                                  size_t vector_length,
                                   int16_t right_shifts);
 void WebRtcSpl_ScaleAndAddVectors(const int16_t* in_vector1,
                                   int16_t gain1, int right_shifts1,
                                   const int16_t* in_vector2,
                                   int16_t gain2, int right_shifts2,
                                   int16_t* out_vector,
-                                  int vector_length);
+                                  size_t vector_length);
 
 // The functions (with related pointer) perform the vector operation:
 //   out_vector[k] = ((scale1 * in_vector1[k]) + (scale2 * in_vector2[k])
@@ -380,7 +355,7 @@ typedef int (*ScaleAndAddVectorsWithRound)(const int16_t* in_vector1,
                                            int16_t in_vector2_scale,
                                            int right_shifts,
                                            int16_t* out_vector,
-                                           int length);
+                                           size_t length);
 extern ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
 int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
                                            int16_t in_vector1_scale,
@@ -388,7 +363,7 @@ int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
                                            int16_t in_vector2_scale,
                                            int right_shifts,
                                            int16_t* out_vector,
-                                           int length);
+                                           size_t length);
 #if defined(MIPS_DSP_R1_LE)
 int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(const int16_t* in_vector1,
                                                int16_t in_vector1_scale,
@@ -396,7 +371,7 @@ int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(const int16_t* in_vector1,
                                                int16_t in_vector2_scale,
                                                int right_shifts,
                                                int16_t* out_vector,
-                                               int length);
+                                               size_t length);
 #endif
 // End: Vector scaling operations.
 
@@ -405,30 +380,30 @@ int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(const int16_t* in_vector1,
 void WebRtcSpl_ReverseOrderMultArrayElements(int16_t* out_vector,
                                              const int16_t* in_vector,
                                              const int16_t* window,
-                                             int16_t vector_length,
+                                             size_t vector_length,
                                              int16_t right_shifts);
 void WebRtcSpl_ElementwiseVectorMult(int16_t* out_vector,
                                      const int16_t* in_vector,
                                      const int16_t* window,
-                                     int16_t vector_length,
+                                     size_t vector_length,
                                      int16_t right_shifts);
 void WebRtcSpl_AddVectorsAndShift(int16_t* out_vector,
                                   const int16_t* in_vector1,
                                   const int16_t* in_vector2,
-                                  int16_t vector_length,
+                                  size_t vector_length,
                                   int16_t right_shifts);
 void WebRtcSpl_AddAffineVectorToVector(int16_t* out_vector,
                                        int16_t* in_vector,
                                        int16_t gain,
                                        int32_t add_constant,
                                        int16_t right_shifts,
-                                       int vector_length);
+                                       size_t vector_length);
 void WebRtcSpl_AffineTransformVector(int16_t* out_vector,
                                      int16_t* in_vector,
                                      int16_t gain,
                                      int32_t add_constant,
                                      int16_t right_shifts,
-                                     int vector_length);
+                                     size_t vector_length);
 // End: iLBC specific functions.
 
 // Signal processing operations.
@@ -449,14 +424,12 @@ void WebRtcSpl_AffineTransformVector(int16_t* out_vector,
 //      - scale            : The number of left shifts required to obtain the
 //                           auto-correlation in Q0
 //
-// Return value            :
-//      - -1, if |order| > |in_vector_length|;
-//      - Number of samples in |result|, i.e. (order+1), otherwise.
-int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
-                              int in_vector_length,
-                              int order,
-                              int32_t* result,
-                              int* scale);
+// Return value            : Number of samples in |result|, i.e. (order+1)
+size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
+                                 size_t in_vector_length,
+                                 size_t order,
+                                 int32_t* result,
+                                 int* scale);
 
 // A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
 // does NOT use the 64 bit class
@@ -473,7 +446,7 @@ int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
 int16_t WebRtcSpl_LevinsonDurbin(const int32_t* auto_corr,
                                  int16_t* lpc_coef,
                                  int16_t* refl_coef,
-                                 int16_t order);
+                                 size_t order);
 
 // Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
 // This version is a 16 bit operation.
@@ -546,36 +519,35 @@ void WebRtcSpl_AutoCorrToReflCoef(const int32_t* auto_corr,
 typedef void (*CrossCorrelation)(int32_t* cross_correlation,
                                  const int16_t* seq1,
                                  const int16_t* seq2,
-                                 int16_t dim_seq,
-                                 int16_t dim_cross_correlation,
-                                 int16_t right_shifts,
-                                 int16_t step_seq2);
+                                 size_t dim_seq,
+                                 size_t dim_cross_correlation,
+                                 int right_shifts,
+                                 int step_seq2);
 extern CrossCorrelation WebRtcSpl_CrossCorrelation;
 void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
                                  const int16_t* seq1,
                                  const int16_t* seq2,
-                                 int16_t dim_seq,
-                                 int16_t dim_cross_correlation,
-                                 int16_t right_shifts,
-                                 int16_t step_seq2);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
+                                 size_t dim_seq,
+                                 size_t dim_cross_correlation,
+                                 int right_shifts,
+                                 int step_seq2);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 void WebRtcSpl_CrossCorrelationNeon(int32_t* cross_correlation,
                                     const int16_t* seq1,
                                     const int16_t* seq2,
-                                    int16_t dim_seq,
-                                    int16_t dim_cross_correlation,
-                                    int16_t right_shifts,
-                                    int16_t step_seq2);
+                                    size_t dim_seq,
+                                    size_t dim_cross_correlation,
+                                    int right_shifts,
+                                    int step_seq2);
 #endif
 #if defined(MIPS32_LE)
 void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
                                      const int16_t* seq1,
                                      const int16_t* seq2,
-                                     int16_t dim_seq,
-                                     int16_t dim_cross_correlation,
-                                     int16_t right_shifts,
-                                     int16_t step_seq2);
+                                     size_t dim_seq,
+                                     size_t dim_cross_correlation,
+                                     int right_shifts,
+                                     int step_seq2);
 #endif
 
 // Creates (the first half of) a Hanning window. Size must be at least 1 and
@@ -586,7 +558,7 @@ void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
 //
 // Output:
 //      - window    : Hanning vector in Q14.
-void WebRtcSpl_GetHanningWindow(int16_t* window, int16_t size);
+void WebRtcSpl_GetHanningWindow(int16_t* window, size_t size);
 
 // Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
 // |in_vector|. Input and output values are in Q15.
@@ -598,7 +570,7 @@ void WebRtcSpl_GetHanningWindow(int16_t* window, int16_t size);
 // Output:
 //      - out_vector    : Output values in Q15
 void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t* in_vector,
-                                      int vector_length,
+                                      size_t vector_length,
                                       int16_t* out_vector);
 // End: Signal processing operations.
 
@@ -624,7 +596,9 @@ int32_t WebRtcSpl_DivResultInQ31(int32_t num, int32_t den);
 int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low);
 // End: Divisions.
 
-int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor);
+int32_t WebRtcSpl_Energy(int16_t* vector,
+                         size_t vector_length,
+                         int* scale_factor);
 
 // Calculates the dot product between two (int16_t) vectors.
 //
@@ -639,21 +613,21 @@ int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor);
 // Return value         : The dot product in Q(-scaling)
 int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
                                       const int16_t* vector2,
-                                      int length,
+                                      size_t length,
                                       int scaling);
 
 // Filter operations.
-int WebRtcSpl_FilterAR(const int16_t* ar_coef,
-                       int ar_coef_length,
-                       const int16_t* in_vector,
-                       int in_vector_length,
-                       int16_t* filter_state,
-                       int filter_state_length,
-                       int16_t* filter_state_low,
-                       int filter_state_low_length,
-                       int16_t* out_vector,
-                       int16_t* out_vector_low,
-                       int out_vector_low_length);
+size_t WebRtcSpl_FilterAR(const int16_t* ar_coef,
+                          size_t ar_coef_length,
+                          const int16_t* in_vector,
+                          size_t in_vector_length,
+                          int16_t* filter_state,
+                          size_t filter_state_length,
+                          int16_t* filter_state_low,
+                          size_t filter_state_low_length,
+                          int16_t* out_vector,
+                          int16_t* out_vector_low,
+                          size_t out_vector_low_length);
 
 // WebRtcSpl_FilterMAFastQ12(...)
 //
@@ -672,8 +646,8 @@ int WebRtcSpl_FilterAR(const int16_t* ar_coef,
 void WebRtcSpl_FilterMAFastQ12(const int16_t* in_vector,
                                int16_t* out_vector,
                                const int16_t* ma_coef,
-                               int16_t ma_coef_length,
-                               int16_t vector_length);
+                               size_t ma_coef_length,
+                               size_t vector_length);
 
 // Performs a AR filtering on a vector in Q12
 // Input:
@@ -688,8 +662,8 @@ void WebRtcSpl_FilterMAFastQ12(const int16_t* in_vector,
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
                                int16_t* data_out,
                                const int16_t* __restrict coefficients,
-                               int coefficients_length,
-                               int data_length);
+                               size_t coefficients_length,
+                               size_t data_length);
 
 // The functions (with related pointer) perform a MA down sampling filter
 // on a vector.
@@ -708,42 +682,41 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
 //      - data_out           : Filtered samples
 // Return value              : 0 if OK, -1 if |in_vector| is too short
 typedef int (*DownsampleFast)(const int16_t* data_in,
-                              int data_in_length,
+                              size_t data_in_length,
                               int16_t* data_out,
-                              int data_out_length,
+                              size_t data_out_length,
                               const int16_t* __restrict coefficients,
-                              int coefficients_length,
+                              size_t coefficients_length,
                               int factor,
-                              int delay);
+                              size_t delay);
 extern DownsampleFast WebRtcSpl_DownsampleFast;
 int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
-                              int data_in_length,
+                              size_t data_in_length,
                               int16_t* data_out,
-                              int data_out_length,
+                              size_t data_out_length,
                               const int16_t* __restrict coefficients,
-                              int coefficients_length,
+                              size_t coefficients_length,
                               int factor,
-                              int delay);
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
+                              size_t delay);
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 int WebRtcSpl_DownsampleFastNeon(const int16_t* data_in,
-                                 int data_in_length,
+                                 size_t data_in_length,
                                  int16_t* data_out,
-                                 int data_out_length,
+                                 size_t data_out_length,
                                  const int16_t* __restrict coefficients,
-                                 int coefficients_length,
+                                 size_t coefficients_length,
                                  int factor,
-                                 int delay);
+                                 size_t delay);
 #endif
 #if defined(MIPS32_LE)
 int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in,
-                                  int data_in_length,
+                                  size_t data_in_length,
                                   int16_t* data_out,
-                                  int data_out_length,
+                                  size_t data_out_length,
                                   const int16_t* __restrict coefficients,
-                                  int coefficients_length,
+                                  size_t coefficients_length,
                                   int factor,
-                                  int delay);
+                                  size_t delay);
 #endif
 
 // End: Filter operations.
@@ -852,14 +825,11 @@ void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
  *
  ******************************************************************/
 
-void WebRtcSpl_Resample48khzTo32khz(const int32_t* In, int32_t* Out,
-                                    int32_t K);
+void WebRtcSpl_Resample48khzTo32khz(const int32_t* In, int32_t* Out, size_t K);
 
-void WebRtcSpl_Resample32khzTo24khz(const int32_t* In, int32_t* Out,
-                                    int32_t K);
+void WebRtcSpl_Resample32khzTo24khz(const int32_t* In, int32_t* Out, size_t K);
 
-void WebRtcSpl_Resample44khzTo32khz(const int32_t* In, int32_t* Out,
-                                    int32_t K);
+void WebRtcSpl_Resample44khzTo32khz(const int32_t* In, int32_t* Out, size_t K);
 
 /*******************************************************************
  * resample_48khz.c
@@ -929,24 +899,24 @@ void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
  *
  ******************************************************************/
 
-void WebRtcSpl_DownsampleBy2(const int16_t* in, int len,
+void WebRtcSpl_DownsampleBy2(const int16_t* in, size_t len,
                              int16_t* out, int32_t* filtState);
 
-void WebRtcSpl_UpsampleBy2(const int16_t* in, int len,
+void WebRtcSpl_UpsampleBy2(const int16_t* in, size_t len,
                            int16_t* out, int32_t* filtState);
 
 /************************************************************
  * END OF RESAMPLING FUNCTIONS
  ************************************************************/
 void WebRtcSpl_AnalysisQMF(const int16_t* in_data,
-                           int in_data_length,
+                           size_t in_data_length,
                            int16_t* low_band,
                            int16_t* high_band,
                            int32_t* filter_state1,
                            int32_t* filter_state2);
 void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
                             const int16_t* high_band,
-                            int band_length,
+                            size_t band_length,
                             int16_t* out_data,
                             int32_t* filter_state1,
                             int32_t* filter_state2);
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/levinson_durbin.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/levinson_durbin.c
index e07af5d3b6..d46e551367 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/levinson_durbin.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/levinson_durbin.c
@@ -20,9 +20,9 @@
 #define SPL_LEVINSON_MAXORDER 20
 
 int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
-                                 int16_t order)
+                                 size_t order)
 {
-    int16_t i, j;
+    size_t i, j;
     // Auto-correlation coefficients in high precision
     int16_t R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
     // LPC coefficients in high precision
@@ -41,7 +41,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
     norm = WebRtcSpl_NormW32(R[0]);
 
-    for (i = order; i >= 0; i--)
+    for (i = 0; i <= order; ++i)
     {
         temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
         // Put R in hi and low format
@@ -76,8 +76,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
     // Alpha = R[0] * (1-K^2)
 
-    temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
-            << 1); // temp1W32 = k^2 in Q31
+    temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) << 1;  // = k^2 in Q31
 
     temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
     temp1W32 = (int32_t)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
@@ -87,9 +86,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
     tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
 
     // Calculate Alpha in Q31
-    temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
-            + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
-            + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
+    temp1W32 = (R_hi[0] * tmp_hi + (R_hi[0] * tmp_low >> 15) +
+        (R_low[0] * tmp_hi >> 15)) << 1;
 
     // Normalize Alpha and put it in hi and low format
 
@@ -113,10 +111,10 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
         for (j = 1; j < i; j++)
         {
-            // temp1W32 is in Q31
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
-                    + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
-                            + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
+          // temp1W32 is in Q31
+          temp1W32 += (R_hi[j] * A_hi[i - j] << 1) +
+              (((R_hi[j] * A_low[i - j] >> 15) +
+              (R_low[j] * A_hi[i - j] >> 15)) << 1);
         }
 
         temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
@@ -177,9 +175,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
                     + WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[j],1);
 
             // temp1W32 += K*A[i-j] in Q27
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
-                    + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
-                    + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
+            temp1W32 += (K_hi * A_hi[i - j] + (K_hi * A_low[i - j] >> 15) +
+                (K_low * A_hi[i - j] >> 15)) << 1;
 
             // Put Anew in hi and low format
             A_upd_hi[j] = (int16_t)(temp1W32 >> 16);
@@ -197,8 +194,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
 
         // Alpha = Alpha * (1-K^2)
 
-        temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
-                + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
+        temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) << 1;  // K*K in Q31
 
         temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
         temp1W32 = (int32_t)0x7fffffffL - temp1W32; // 1 - K*K  in Q31
@@ -208,9 +204,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
         tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
 
         // Calculate Alpha = Alpha * (1-K^2) in Q31
-        temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
+        temp1W32 = (Alpha_hi * tmp_hi + (Alpha_hi * tmp_low >> 15) +
+            (Alpha_low * tmp_hi >> 15)) << 1;
 
         // Normalize Alpha and store it on hi and low format
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
index 5fb4d8596b..edcebd4e63 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
@@ -30,7 +30,7 @@ void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
     for (m = use_order - 1; m > 0; m--)
     {
         // (1 - k^2) in Q30
-        tmp_inv_denom32 = ((int32_t)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
+        tmp_inv_denom32 = 1073741823 - k16[m] * k16[m];
         // (1 - k^2) in Q15
         tmp_inv_denom16 = (int16_t)(tmp_inv_denom32 >> 15);
 
@@ -39,8 +39,7 @@ void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
             // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
 
             // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
-            tmp32[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)a16[k], 16)
-                    - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
+            tmp32[k] = (a16[k] << 16) - (k16[m] * a16[m - k + 1] << 1);
 
             tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
         }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations.c
index f6de072a22..4a962f86a0 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations.c
@@ -24,21 +24,21 @@
  *
  */
 
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-
+#include <assert.h>
 #include <stdlib.h>
 
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
 // TODO(bjorn/kma): Consolidate function pairs (e.g. combine
 //   WebRtcSpl_MaxAbsValueW16C and WebRtcSpl_MaxAbsIndexW16 into a single one.)
 // TODO(kma): Move the next six functions into min_max_operations_c.c.
 
 // Maximum absolute value of word16 vector. C version for generic platforms.
-int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, int length) {
-  int i = 0, absolute = 0, maximum = 0;
+int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length) {
+  size_t i = 0;
+  int absolute = 0, maximum = 0;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     absolute = abs((int)vector[i]);
@@ -57,16 +57,14 @@ int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, int length) {
 }
 
 // Maximum absolute value of word32 vector. C version for generic platforms.
-int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length) {
   // Use uint32_t for the local variables, to accommodate the return value
   // of abs(0x80000000), which is 0x80000000.
 
   uint32_t absolute = 0, maximum = 0;
-  int i = 0;
+  size_t i = 0;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     absolute = abs((int)vector[i]);
@@ -81,13 +79,11 @@ int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, int length) {
 }
 
 // Maximum value of word16 vector. C version for generic platforms.
-int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length) {
   int16_t maximum = WEBRTC_SPL_WORD16_MIN;
-  int i = 0;
+  size_t i = 0;
 
-  if (vector == NULL || length <= 0) {
-    return maximum;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum)
@@ -97,13 +93,11 @@ int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, int length) {
 }
 
 // Maximum value of word32 vector. C version for generic platforms.
-int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length) {
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
-  int i = 0;
+  size_t i = 0;
 
-  if (vector == NULL || length <= 0) {
-    return maximum;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum)
@@ -113,13 +107,11 @@ int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, int length) {
 }
 
 // Minimum value of word16 vector. C version for generic platforms.
-int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length) {
   int16_t minimum = WEBRTC_SPL_WORD16_MAX;
-  int i = 0;
+  size_t i = 0;
 
-  if (vector == NULL || length <= 0) {
-    return minimum;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum)
@@ -129,13 +121,11 @@ int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, int length) {
 }
 
 // Minimum value of word32 vector. C version for generic platforms.
-int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length) {
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
-  int i = 0;
+  size_t i = 0;
 
-  if (vector == NULL || length <= 0) {
-    return minimum;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum)
@@ -145,14 +135,13 @@ int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, int length) {
 }
 
 // Index of maximum absolute value in a word16 vector.
-int WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, int length) {
+size_t WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, size_t length) {
   // Use type int for local variables, to accomodate the value of abs(-32768).
 
-  int i = 0, absolute = 0, maximum = 0, index = 0;
+  size_t i = 0, index = 0;
+  int absolute = 0, maximum = 0;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     absolute = abs((int)vector[i]);
@@ -167,13 +156,11 @@ int WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, int length) {
 }
 
 // Index of maximum value in a word16 vector.
-int WebRtcSpl_MaxIndexW16(const int16_t* vector, int length) {
-  int i = 0, index = 0;
+size_t WebRtcSpl_MaxIndexW16(const int16_t* vector, size_t length) {
+  size_t i = 0, index = 0;
   int16_t maximum = WEBRTC_SPL_WORD16_MIN;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum) {
@@ -186,13 +173,11 @@ int WebRtcSpl_MaxIndexW16(const int16_t* vector, int length) {
 }
 
 // Index of maximum value in a word32 vector.
-int WebRtcSpl_MaxIndexW32(const int32_t* vector, int length) {
-  int i = 0, index = 0;
+size_t WebRtcSpl_MaxIndexW32(const int32_t* vector, size_t length) {
+  size_t i = 0, index = 0;
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] > maximum) {
@@ -205,13 +190,11 @@ int WebRtcSpl_MaxIndexW32(const int32_t* vector, int length) {
 }
 
 // Index of minimum value in a word16 vector.
-int WebRtcSpl_MinIndexW16(const int16_t* vector, int length) {
-  int i = 0, index = 0;
+size_t WebRtcSpl_MinIndexW16(const int16_t* vector, size_t length) {
+  size_t i = 0, index = 0;
   int16_t minimum = WEBRTC_SPL_WORD16_MAX;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum) {
@@ -224,13 +207,11 @@ int WebRtcSpl_MinIndexW16(const int16_t* vector, int length) {
 }
 
 // Index of minimum value in a word32 vector.
-int WebRtcSpl_MinIndexW32(const int32_t* vector, int length) {
-  int i = 0, index = 0;
+size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length) {
+  size_t i = 0, index = 0;
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   for (i = 0; i < length; i++) {
     if (vector[i] < minimum) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_mips.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_mips.c
index 5fd8600833..28de45b3a5 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_mips.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_mips.c
@@ -16,17 +16,18 @@
  *
  */
 
+#include <assert.h>
+
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // Maximum absolute value of word16 vector.
-int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, size_t length) {
   int32_t totMax = 0;
   int32_t tmp32_0, tmp32_1, tmp32_2, tmp32_3;
-  int i, loop_size;
+  size_t i, loop_size;
+
+  assert(length > 0);
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
 #if defined(MIPS_DSP_R1)
   const int32_t* tmpvec32 = (int32_t*)vector;
   loop_size = length >> 4;
@@ -222,16 +223,14 @@ int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length) {
 
 #if defined(MIPS_DSP_R1_LE)
 // Maximum absolute value of word32 vector. Version for MIPS platform.
-int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, size_t length) {
   // Use uint32_t for the local variables, to accommodate the return value
   // of abs(0x80000000), which is 0x80000000.
 
   uint32_t absolute = 0, maximum = 0;
   int tmp1 = 0, max_value = 0x7fffffff;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -260,14 +259,12 @@ int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, int length) {
 #endif  // #if defined(MIPS_DSP_R1_LE)
 
 // Maximum value of word16 vector. Version for MIPS platform.
-int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length) {
   int16_t maximum = WEBRTC_SPL_WORD16_MIN;
   int tmp1;
   int16_t value;
 
-  if (vector == NULL || length <= 0) {
-    return maximum;
-  }
+  assert(length > 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -291,13 +288,11 @@ int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, int length) {
 }
 
 // Maximum value of word32 vector. Version for MIPS platform.
-int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length) {
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
   int tmp1, value;
 
-  if (vector == NULL || length <= 0) {
-    return maximum;
-  }
+  assert(length > 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -322,14 +317,12 @@ int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, int length) {
 }
 
 // Minimum value of word16 vector. Version for MIPS platform.
-int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length) {
   int16_t minimum = WEBRTC_SPL_WORD16_MAX;
   int tmp1;
   int16_t value;
 
-  if (vector == NULL || length <= 0) {
-    return minimum;
-  }
+  assert(length > 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
@@ -354,13 +347,11 @@ int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, int length) {
 }
 
 // Minimum value of word32 vector. Version for MIPS platform.
-int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, size_t length) {
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
   int tmp1, value;
 
-  if (vector == NULL || length <= 0) {
-    return minimum;
-  }
+  assert(length > 0);
 
   __asm__ volatile (
     ".set push                                                        \n\t"
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_neon.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_neon.c
index dec31ad315..6fbbf94ee0 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_neon.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/min_max_operations_neon.c
@@ -9,20 +9,19 @@
  */
 
 #include <arm_neon.h>
+#include <assert.h>
 #include <stdlib.h>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 // Maximum absolute value of word16 vector. C version for generic platforms.
-int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, size_t length) {
   int absolute = 0, maximum = 0;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   const int16_t* p_start = vector;
-  int rest = length & 7;
+  size_t rest = length & 7;
   const int16_t* p_end = vector + length - rest;
 
   int16x8_t v;
@@ -69,24 +68,22 @@ int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, int length) {
 
 // Maximum absolute value of word32 vector. NEON intrinsics version for
 // ARM 32-bit/64-bit platforms.
-int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, size_t length) {
   // Use uint32_t for the local variables, to accommodate the return value
   // of abs(0x80000000), which is 0x80000000.
 
   uint32_t absolute = 0, maximum = 0;
-  int i = 0;
-  int residual = length & 0x7;
+  size_t i = 0;
+  size_t residual = length & 0x7;
 
-  if (vector == NULL || length <= 0) {
-    return -1;
-  }
+  assert(length > 0);
 
   const int32_t* p_start = vector;
   uint32x4_t max32x4_0 = vdupq_n_u32(0);
   uint32x4_t max32x4_1 = vdupq_n_u32(0);
 
   // First part, unroll the loop 8 times.
-  for (i = length - residual; i >0; i -= 8) {
+  for (i = 0; i < length - residual; i += 8) {
     int32x4_t in32x4_0 = vld1q_s32(p_start);
     p_start += 4;
     int32x4_t in32x4_1 = vld1q_s32(p_start);
@@ -126,20 +123,18 @@ int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, int length) {
 
 // Maximum value of word16 vector. NEON intrinsics version for
 // ARM 32-bit/64-bit platforms.
-int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, size_t length) {
   int16_t maximum = WEBRTC_SPL_WORD16_MIN;
-  int i = 0;
-  int residual = length & 0x7;
+  size_t i = 0;
+  size_t residual = length & 0x7;
 
-  if (vector == NULL || length <= 0) {
-    return maximum;
-  }
+  assert(length > 0);
 
   const int16_t* p_start = vector;
   int16x8_t max16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MIN);
 
   // First part, unroll the loop 8 times.
-  for (i = length - residual; i >0; i -= 8) {
+  for (i = 0; i < length - residual; i += 8) {
     int16x8_t in16x8 = vld1q_s16(p_start);
     max16x8 = vmaxq_s16(max16x8, in16x8);
     p_start += 8;
@@ -166,21 +161,19 @@ int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, int length) {
 
 // Maximum value of word32 vector. NEON intrinsics version for
 // ARM 32-bit/64-bit platforms.
-int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, size_t length) {
   int32_t maximum = WEBRTC_SPL_WORD32_MIN;
-  int i = 0;
-  int residual = length & 0x7;
+  size_t i = 0;
+  size_t residual = length & 0x7;
 
-  if (vector == NULL || length <= 0) {
-    return maximum;
-  }
+  assert(length > 0);
 
   const int32_t* p_start = vector;
   int32x4_t max32x4_0 = vdupq_n_s32(WEBRTC_SPL_WORD32_MIN);
   int32x4_t max32x4_1 = vdupq_n_s32(WEBRTC_SPL_WORD32_MIN);
 
   // First part, unroll the loop 8 times.
-  for (i = length - residual; i >0; i -= 8) {
+  for (i = 0; i < length - residual; i += 8) {
     int32x4_t in32x4_0 = vld1q_s32(p_start);
     p_start += 4;
     int32x4_t in32x4_1 = vld1q_s32(p_start);
@@ -210,20 +203,18 @@ int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, int length) {
 
 // Minimum value of word16 vector. NEON intrinsics version for
 // ARM 32-bit/64-bit platforms.
-int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, int length) {
+int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, size_t length) {
   int16_t minimum = WEBRTC_SPL_WORD16_MAX;
-  int i = 0;
-  int residual = length & 0x7;
+  size_t i = 0;
+  size_t residual = length & 0x7;
 
-  if (vector == NULL || length <= 0) {
-    return minimum;
-  }
+  assert(length > 0);
 
   const int16_t* p_start = vector;
   int16x8_t min16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MAX);
 
   // First part, unroll the loop 8 times.
-  for (i = length - residual; i >0; i -= 8) {
+  for (i = 0; i < length - residual; i += 8) {
     int16x8_t in16x8 = vld1q_s16(p_start);
     min16x8 = vminq_s16(min16x8, in16x8);
     p_start += 8;
@@ -250,21 +241,19 @@ int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, int length) {
 
 // Minimum value of word32 vector. NEON intrinsics version for
 // ARM 32-bit/64-bit platforms.
-int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, int length) {
+int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, size_t length) {
   int32_t minimum = WEBRTC_SPL_WORD32_MAX;
-  int i = 0;
-  int residual = length & 0x7;
+  size_t i = 0;
+  size_t residual = length & 0x7;
 
-  if (vector == NULL || length <= 0) {
-    return minimum;
-  }
+  assert(length > 0);
 
   const int32_t* p_start = vector;
   int32x4_t min32x4_0 = vdupq_n_s32(WEBRTC_SPL_WORD32_MAX);
   int32x4_t min32x4_1 = vdupq_n_s32(WEBRTC_SPL_WORD32_MAX);
 
   // First part, unroll the loop 8 times.
-  for (i = length - residual; i >0; i -= 8) {
+  for (i = 0; i < length - residual; i += 8) {
     int32x4_t in32x4_0 = vld1q_s32(p_start);
     p_start += 4;
     int32x4_t in32x4_1 = vld1q_s32(p_start);
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/real_fft_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/signal_processing/real_fft_unittest.cc
index 9bd35cd68b..fa98836b9a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/real_fft_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/real_fft_unittest.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2.c
index 9c0784edc7..dcba82e35f 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2.c
@@ -67,10 +67,10 @@ static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
 
 // decimator
 #if !defined(MIPS32_LE)
-void WebRtcSpl_DownsampleBy2(const int16_t* in, int len,
+void WebRtcSpl_DownsampleBy2(const int16_t* in, size_t len,
                              int16_t* out, int32_t* filtState) {
   int32_t tmp1, tmp2, diff, in32, out32;
-  int i;
+  size_t i;
 
   register int32_t state0 = filtState[0];
   register int32_t state1 = filtState[1];
@@ -125,10 +125,10 @@ void WebRtcSpl_DownsampleBy2(const int16_t* in, int len,
 #endif  // #if defined(MIPS32_LE)
 
 
-void WebRtcSpl_UpsampleBy2(const int16_t* in, int len,
+void WebRtcSpl_UpsampleBy2(const int16_t* in, size_t len,
                            int16_t* out, int32_t* filtState) {
   int32_t tmp1, tmp2, diff, in32, out32;
-  int i;
+  size_t i;
 
   register int32_t state0 = filtState[0];
   register int32_t state1 = filtState[1];
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2_mips.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2_mips.c
index 6ffce551f0..ec5fc8b3b6 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2_mips.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_by_2_mips.c
@@ -29,11 +29,11 @@ static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
 
 // decimator
 void WebRtcSpl_DownsampleBy2(const int16_t* in,
-                             int len,
+                             size_t len,
                              int16_t* out,
                              int32_t* filtState) {
   int32_t out32;
-  int i, len1;
+  size_t i, len1;
 
   register int32_t state0 = filtState[0];
   register int32_t state1 = filtState[1];
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_fractional.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_fractional.c
index c7b5edbffb..6409fbac47 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_fractional.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/resample_fractional.c
@@ -41,8 +41,7 @@ static const int16_t kCoefficients44To32[4][9] = {
 // output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 2 * K
 //      K: number of blocks
 
-void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out,
-                                    int32_t K)
+void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out, size_t K)
 {
     /////////////////////////////////////////////////////////////
     // Filter operation:
@@ -50,7 +49,7 @@ void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out,
     // Perform resampling (3 input samples -> 2 output samples);
     // process in sub blocks of size 3 samples.
     int32_t tmp;
-    int32_t m;
+    size_t m;
 
     for (m = 0; m < K; m++)
     {
@@ -87,15 +86,14 @@ void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out,
 // output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 3 * K
 //      K: number of blocks
 
-void WebRtcSpl_Resample32khzTo24khz(const int32_t *In, int32_t *Out,
-                                    int32_t K)
+void WebRtcSpl_Resample32khzTo24khz(const int32_t *In, int32_t *Out, size_t K)
 {
     /////////////////////////////////////////////////////////////
     // Filter operation:
     //
     // Perform resampling (4 input samples -> 3 output samples);
     // process in sub blocks of size 4 samples.
-    int32_t m;
+    size_t m;
     int32_t tmp;
 
     for (m = 0; m < K; m++)
@@ -196,8 +194,7 @@ static void WebRtcSpl_ResampDotProduct(const int32_t *in1, const int32_t *in2,
 // output: int32_t (shifted 15 positions to the left, + offset 16384) :: size  8 * K
 //      K: number of blocks
 
-void WebRtcSpl_Resample44khzTo32khz(const int32_t *In, int32_t *Out,
-                                    int32_t K)
+void WebRtcSpl_Resample44khzTo32khz(const int32_t *In, int32_t *Out, size_t K)
 {
     /////////////////////////////////////////////////////////////
     // Filter operation:
@@ -205,7 +202,7 @@ void WebRtcSpl_Resample44khzTo32khz(const int32_t *In, int32_t *Out,
     // Perform resampling (11 input samples -> 8 output samples);
     // process in sub blocks of size 11 samples.
     int32_t tmp;
-    int32_t m;
+    size_t m;
 
     for (m = 0; m < K; m++)
     {
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index 12b2fffbee..108f459c89 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -11,7 +11,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-static const int kVector16Size = 9;
+static const size_t kVector16Size = 9;
 static const int16_t vector16[kVector16Size] = {1, -15511, 4323, 1963,
   WEBRTC_SPL_WORD16_MAX, 0, WEBRTC_SPL_WORD16_MIN + 5, -3333, 345};
 
@@ -157,7 +157,7 @@ TEST_F(SplTest, MathOperationsTest) {
 }
 
 TEST_F(SplTest, BasicArrayOperationsTest) {
-    const int kVectorSize = 4;
+    const size_t kVectorSize = 4;
     int B[] = {4, 12, 133, 1100};
     int16_t b16[kVectorSize];
     int32_t b32[kVectorSize];
@@ -166,27 +166,27 @@ TEST_F(SplTest, BasicArrayOperationsTest) {
     int32_t bTmp32[kVectorSize];
 
     WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(3, b16[kk]);
     }
     WebRtcSpl_ZerosArrayW16(b16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(0, b16[kk]);
     }
     WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(3, b32[kk]);
     }
     WebRtcSpl_ZerosArrayW32(b32, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(0, b32[kk]);
     }
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         bTmp16[kk] = (int16_t)kk;
         bTmp32[kk] = (int32_t)kk;
     }
     WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(b16[kk], bTmp16[kk]);
     }
 //    WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
@@ -194,65 +194,35 @@ TEST_F(SplTest, BasicArrayOperationsTest) {
 //        EXPECT_EQ(b32[kk], bTmp32[kk]);
 //    }
     WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16);
-    for (int kk = 0; kk < 2; ++kk) {
-        EXPECT_EQ(kk+2, bTmp16[kk]);
+    for (size_t kk = 0; kk < 2; ++kk) {
+        EXPECT_EQ(static_cast<int16_t>(kk+2), bTmp16[kk]);
     }
 
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         b32[kk] = B[kk];
         b16[kk] = (int16_t)B[kk];
     }
     WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
     }
     WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
     }
     WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
     }
 
     WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(b16[3-kk], bTmp16[kk]);
     }
 }
 
-TEST_F(SplTest, ExeptionsHandlingMinMaxOperationsTest) {
-  // Test how the functions handle exceptional cases.
-  const int kVectorSize = 2;
-  int16_t vector16[kVectorSize] = {0};
-  int32_t vector32[kVectorSize] = {0};
-
-  EXPECT_EQ(-1, WebRtcSpl_MaxAbsValueW16(vector16, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MaxAbsValueW16(NULL, kVectorSize));
-  EXPECT_EQ(WEBRTC_SPL_WORD16_MIN, WebRtcSpl_MaxValueW16(vector16, 0));
-  EXPECT_EQ(WEBRTC_SPL_WORD16_MIN, WebRtcSpl_MaxValueW16(NULL, kVectorSize));
-  EXPECT_EQ(WEBRTC_SPL_WORD16_MAX, WebRtcSpl_MinValueW16(vector16, 0));
-  EXPECT_EQ(WEBRTC_SPL_WORD16_MAX, WebRtcSpl_MinValueW16(NULL, kVectorSize));
-  EXPECT_EQ(-1, WebRtcSpl_MaxAbsValueW32(vector32, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MaxAbsValueW32(NULL, kVectorSize));
-  EXPECT_EQ(WEBRTC_SPL_WORD32_MIN, WebRtcSpl_MaxValueW32(vector32, 0));
-  EXPECT_EQ(WEBRTC_SPL_WORD32_MIN, WebRtcSpl_MaxValueW32(NULL, kVectorSize));
-  EXPECT_EQ(WEBRTC_SPL_WORD32_MAX, WebRtcSpl_MinValueW32(vector32, 0));
-  EXPECT_EQ(WEBRTC_SPL_WORD32_MAX, WebRtcSpl_MinValueW32(NULL, kVectorSize));
-  EXPECT_EQ(-1, WebRtcSpl_MaxAbsIndexW16(vector16, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MaxAbsIndexW16(NULL, kVectorSize));
-  EXPECT_EQ(-1, WebRtcSpl_MaxIndexW16(vector16, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MaxIndexW16(NULL, kVectorSize));
-  EXPECT_EQ(-1, WebRtcSpl_MaxIndexW32(vector32, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MaxIndexW32(NULL, kVectorSize));
-  EXPECT_EQ(-1, WebRtcSpl_MinIndexW16(vector16, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MinIndexW16(NULL, kVectorSize));
-  EXPECT_EQ(-1, WebRtcSpl_MinIndexW32(vector32, 0));
-  EXPECT_EQ(-1, WebRtcSpl_MinIndexW32(NULL, kVectorSize));
-}
-
 TEST_F(SplTest, MinMaxOperationsTest) {
-  const int kVectorSize = 17;
+  const size_t kVectorSize = 17;
 
   // Vectors to test the cases where minimum values have to be caught
   // outside of the unrolled loops in ARM-Neon.
@@ -307,67 +277,67 @@ TEST_F(SplTest, MinMaxOperationsTest) {
             WebRtcSpl_MaxValueW32(vector32, kVectorSize));
   EXPECT_EQ(WEBRTC_SPL_WORD32_MIN,
             WebRtcSpl_MinValueW32(vector32, kVectorSize));
-  EXPECT_EQ(6, WebRtcSpl_MaxAbsIndexW16(vector16, kVectorSize));
-  EXPECT_EQ(1, WebRtcSpl_MaxIndexW16(vector16, kVectorSize));
-  EXPECT_EQ(1, WebRtcSpl_MaxIndexW32(vector32, kVectorSize));
-  EXPECT_EQ(6, WebRtcSpl_MinIndexW16(vector16, kVectorSize));
-  EXPECT_EQ(6, WebRtcSpl_MinIndexW32(vector32, kVectorSize));
+  EXPECT_EQ(6u, WebRtcSpl_MaxAbsIndexW16(vector16, kVectorSize));
+  EXPECT_EQ(1u, WebRtcSpl_MaxIndexW16(vector16, kVectorSize));
+  EXPECT_EQ(1u, WebRtcSpl_MaxIndexW32(vector32, kVectorSize));
+  EXPECT_EQ(6u, WebRtcSpl_MinIndexW16(vector16, kVectorSize));
+  EXPECT_EQ(6u, WebRtcSpl_MinIndexW32(vector32, kVectorSize));
 }
 
 TEST_F(SplTest, VectorOperationsTest) {
-    const int kVectorSize = 4;
+    const size_t kVectorSize = 4;
     int B[] = {4, 12, 133, 1100};
     int16_t a16[kVectorSize];
     int16_t b16[kVectorSize];
     int16_t bTmp16[kVectorSize];
 
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         a16[kk] = B[kk];
         b16[kk] = B[kk];
     }
 
     WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
     }
     WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
     }
 
     WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
     }
 
     WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
     }
     WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
     }
     WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
     }
 
     WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
     }
     WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
     }
     WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
     }
 
     WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
-    for (int kk = 0; kk < kVectorSize - 1; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize - 1; ++kk) {
         EXPECT_EQ(32767, bTmp16[kk]);
     }
     EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
@@ -376,7 +346,7 @@ TEST_F(SplTest, VectorOperationsTest) {
 }
 
 TEST_F(SplTest, EstimatorsTest) {
-  const int16_t kOrder = 2;
+  const size_t kOrder = 2;
   const int32_t unstable_filter[] = { 4, 12, 133, 1100 };
   const int32_t stable_filter[] = { 1100, 133, 12, 4 };
   int16_t lpc[kOrder + 2] = { 0 };
@@ -386,15 +356,15 @@ TEST_F(SplTest, EstimatorsTest) {
 
   EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(unstable_filter, lpc, refl, kOrder));
   EXPECT_EQ(1, WebRtcSpl_LevinsonDurbin(stable_filter, lpc, refl, kOrder));
-  for (int i = 0; i < kOrder + 2; ++i) {
+  for (size_t i = 0; i < kOrder + 2; ++i) {
     EXPECT_EQ(lpc_result[i], lpc[i]);
     EXPECT_EQ(refl_result[i], refl[i]);
   }
 }
 
 TEST_F(SplTest, FilterTest) {
-    const int kVectorSize = 4;
-    const int kFilterOrder = 3;
+    const size_t kVectorSize = 4;
+    const size_t kFilterOrder = 3;
     int16_t A[] = {1, 2, 33, 100};
     int16_t A5[] = {1, 2, 33, 100, -5};
     int16_t B[] = {4, 12, 133, 110};
@@ -407,7 +377,7 @@ TEST_F(SplTest, FilterTest) {
     WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
     WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
 
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         data_in[kk] = A[kk];
         data_out[kk] = 0;
     }
@@ -460,10 +430,10 @@ TEST_F(SplTest, DotProductWithScaleTest) {
 
 TEST_F(SplTest, CrossCorrelationTest) {
   // Note the function arguments relation specificed by API.
-  const int kCrossCorrelationDimension = 3;
+  const size_t kCrossCorrelationDimension = 3;
   const int kShift = 2;
   const int kStep = 1;
-  const int kSeqDimension = 6;
+  const size_t kSeqDimension = 6;
 
   const int16_t kVector16[kVector16Size] = {1, 4323, 1963,
     WEBRTC_SPL_WORD16_MAX, WEBRTC_SPL_WORD16_MIN + 5, -3333, -876, 8483, 142};
@@ -484,7 +454,7 @@ TEST_F(SplTest, CrossCorrelationTest) {
     expected = kExpectedNeon;
   }
 #endif
-  for (int i = 0; i < kCrossCorrelationDimension; ++i) {
+  for (size_t i = 0; i < kCrossCorrelationDimension; ++i) {
     EXPECT_EQ(expected[i], vector32[i]);
   }
 }
@@ -495,18 +465,17 @@ TEST_F(SplTest, AutoCorrelationTest) {
   const int32_t expected[kVector16Size] = {302681398, 14223410, -121705063,
     -85221647, -17104971, 61806945, 6644603, -669329, 43};
 
-  EXPECT_EQ(-1, WebRtcSpl_AutoCorrelation(vector16,
-      kVector16Size, kVector16Size + 1, vector32, &scale));
-  EXPECT_EQ(kVector16Size, WebRtcSpl_AutoCorrelation(vector16,
-      kVector16Size, kVector16Size - 1, vector32, &scale));
+  EXPECT_EQ(kVector16Size,
+            WebRtcSpl_AutoCorrelation(vector16, kVector16Size,
+                                      kVector16Size - 1, vector32, &scale));
   EXPECT_EQ(3, scale);
-  for (int i = 0; i < kVector16Size; ++i) {
+  for (size_t i = 0; i < kVector16Size; ++i) {
     EXPECT_EQ(expected[i], vector32[i]);
   }
 }
 
 TEST_F(SplTest, SignalProcessingTest) {
-    const int kVectorSize = 4;
+    const size_t kVectorSize = 4;
     int A[] = {1, 2, 33, 100};
     const int16_t kHanning[4] = { 2399, 8192, 13985, 16384 };
     int16_t b16[kVectorSize];
@@ -515,7 +484,7 @@ TEST_F(SplTest, SignalProcessingTest) {
 
     int bScale = 0;
 
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         b16[kk] = A[kk];
     }
 
@@ -534,11 +503,11 @@ TEST_F(SplTest, SignalProcessingTest) {
 ////    }
 
     WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         EXPECT_EQ(kHanning[kk], bTmp16[kk]);
     }
 
-    for (int kk = 0; kk < kVectorSize; ++kk) {
+    for (size_t kk = 0; kk < kVectorSize; ++kk) {
         b16[kk] = A[kk];
     }
     EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
@@ -568,7 +537,7 @@ TEST_F(SplTest, FFTTest) {
 TEST_F(SplTest, Resample48WithSaturationTest) {
   // The test resamples 3*kBlockSize number of samples to 2*kBlockSize number
   // of samples.
-  const int kBlockSize = 16;
+  const size_t kBlockSize = 16;
 
   // Saturated input vector of 48 samples.
   const int32_t kVectorSaturated[3 * kBlockSize + 7] = {
@@ -599,11 +568,11 @@ TEST_F(SplTest, Resample48WithSaturationTest) {
 
   // Comparing output values against references. The values at position
   // 12-15 are skipped to account for the filter lag.
-  for (int i = 0; i < 12; ++i) {
+  for (size_t i = 0; i < 12; ++i) {
     EXPECT_EQ(kRefValue32kHz1, out_vector[i]);
     EXPECT_EQ(kRefValue16kHz1, out_vector_w16[i]);
   }
-  for (int i = 16; i < 2 * kBlockSize; ++i) {
+  for (size_t i = 16; i < 2 * kBlockSize; ++i) {
     EXPECT_EQ(kRefValue32kHz2, out_vector[i]);
     EXPECT_EQ(kRefValue16kHz2, out_vector_w16[i]);
   }
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_init.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_init.c
index 0a493796cb..fdab038399 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_init.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_init.c
@@ -15,7 +15,7 @@
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
 /* Declare function pointers. */
 MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
@@ -28,8 +28,8 @@ CrossCorrelation WebRtcSpl_CrossCorrelation;
 DownsampleFast WebRtcSpl_DownsampleFast;
 ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
 
-#if (defined(WEBRTC_DETECT_ARM_NEON) || !defined(WEBRTC_ARCH_ARM_NEON)) && \
-    !defined(MIPS32_LE) && !defined(WEBRTC_ARCH_ARM64_NEON)
+#if (defined(WEBRTC_DETECT_NEON) || !defined(WEBRTC_HAS_NEON)) && \
+    !defined(MIPS32_LE)
 /* Initialize function pointers to the generic C version. */
 static void InitPointersToC() {
   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16C;
@@ -45,8 +45,7 @@ static void InitPointersToC() {
 }
 #endif
 
-#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON) || \
-  (defined WEBRTC_ARCH_ARM64_NEON)
+#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 /* Initialize function pointers to the Neon version. */
 static void InitPointersToNeon() {
   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16Neon;
@@ -57,8 +56,6 @@ static void InitPointersToNeon() {
   WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32Neon;
   WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationNeon;
   WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastNeon;
-  /* TODO(henrik.lundin): re-enable NEON when the crash from bug 3243 is
-     understood. */
   WebRtcSpl_ScaleAndAddVectorsWithRound =
       WebRtcSpl_ScaleAndAddVectorsWithRoundC;
 }
@@ -87,19 +84,19 @@ static void InitPointersToMIPS() {
 #endif
 
 static void InitFunctionPointers(void) {
-#if defined(WEBRTC_DETECT_ARM_NEON)
+#if defined(WEBRTC_DETECT_NEON)
   if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
     InitPointersToNeon();
   } else {
     InitPointersToC();
   }
-#elif defined(WEBRTC_ARCH_ARM_NEON) || defined(WEBRTC_ARCH_ARM64_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   InitPointersToNeon();
 #elif defined(MIPS32_LE)
   InitPointersToMIPS();
 #else
   InitPointersToC();
-#endif  /* WEBRTC_DETECT_ARM_NEON */
+#endif  /* WEBRTC_DETECT_NEON */
 }
 
 #if defined(WEBRTC_POSIX)
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt.c
index 1de6ccd713..24db4f822c 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt.c
@@ -49,16 +49,16 @@ int32_t WebRtcSpl_SqrtLocal(int32_t in)
     A >>= 16;
     A = A * A * 2; // A = (x/2)^4
     t16 = (int16_t)(A >> 16);
-    B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
+    B += -20480 * t16 * 2;  // B = B - 0.625*A
     // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
 
-    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
+    A = x_half * t16 * 2;  // A = (x/2)^5
     t16 = (int16_t)(A >> 16);
-    B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
+    B += 28672 * t16 * 2;  // B = B + 0.875*A
     // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
 
     t16 = (int16_t)(x2 >> 16);
-    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
+    A = x_half * t16 * 2;  // A = x/2^3
 
     B = B + (A >> 1); // B = B + 0.5*A
     // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
@@ -166,7 +166,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
 
         t16 = (int16_t)(A >> 16);  // t16 = AH
 
-        A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
+        A = k_sqrt_2 * t16 * 2;  // A = 1/sqrt(2)*t16
         A = A + ((int32_t)32768); // Round off
         A = A & ((int32_t)0x7fff0000); // Round off
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S b/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
index f44ddd464c..72cd2d9a0a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
@@ -32,7 +32,7 @@
 @ Output:             r0 = INT (SQRT (r0)), precision is 16 bits
 @ Registers touched:  r1, r2
 
-#include "webrtc/system_wrappers/interface/asm_defines.h"
+#include "webrtc/system_wrappers/include/asm_defines.h"
 
 GLOBAL_FUNCTION WebRtcSpl_SqrtFloor
 .align  2
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/splitting_filter.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/splitting_filter.c
index 7ae281c2ee..36fcf355ec 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/splitting_filter.c
@@ -45,7 +45,7 @@ static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
 //                            |data_length|
 //
 
-void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
+void WebRtcSpl_AllPassQMF(int32_t* in_data, size_t data_length,
                           int32_t* out_data, const uint16_t* filter_coefficients,
                           int32_t* filter_state)
 {
@@ -65,7 +65,7 @@ void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
     // filter operation takes the |in_data| (which is the output from the previous cascade
     // filter) and store the output in |out_data|.
     // Note that the input vector values are changed during the process.
-    int k;
+    size_t k;
     int32_t diff;
     // First all-pass cascade; filter from in_data to out_data.
 
@@ -124,18 +124,18 @@ void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
     filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
 }
 
-void WebRtcSpl_AnalysisQMF(const int16_t* in_data, int in_data_length,
+void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
                            int16_t* low_band, int16_t* high_band,
                            int32_t* filter_state1, int32_t* filter_state2)
 {
-    int16_t i;
+    size_t i;
     int16_t k;
     int32_t tmp;
     int32_t half_in1[kMaxBandFrameLength];
     int32_t half_in2[kMaxBandFrameLength];
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
-    const int band_length = in_data_length / 2;
+    const size_t band_length = in_data_length / 2;
     assert(in_data_length % 2 == 0);
     assert(band_length <= kMaxBandFrameLength);
 
@@ -165,7 +165,7 @@ void WebRtcSpl_AnalysisQMF(const int16_t* in_data, int in_data_length,
 }
 
 void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
-                            int band_length, int16_t* out_data,
+                            size_t band_length, int16_t* out_data,
                             int32_t* filter_state1, int32_t* filter_state2)
 {
     int32_t tmp;
@@ -173,7 +173,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
     int32_t half_in2[kMaxBandFrameLength];
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
-    int16_t i;
+    size_t i;
     int16_t k;
     assert(band_length <= kMaxBandFrameLength);
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
index fc438c6373..ff78b5228f 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@@ -17,17 +17,17 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length,
+void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, size_t vector_length,
                                       int16_t *yQ15)
 {
     int32_t sq;
-    int m;
+    size_t m;
     int16_t tmp;
 
     for (m = 0; m < vector_length; m++)
     {
         tmp = xQ15[m];
-        sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
+        sq = tmp * tmp;  // x^2 in Q30
         sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
         sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
         yQ15[m] = (int16_t)sq;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations.c
index 736f62c71f..fdefd06760 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations.c
@@ -22,10 +22,10 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-void WebRtcSpl_VectorBitShiftW16(int16_t *res, int16_t length,
+void WebRtcSpl_VectorBitShiftW16(int16_t *res, size_t length,
                                  const int16_t *in, int16_t right_shifts)
 {
-    int i;
+    size_t i;
 
     if (right_shifts > 0)
     {
@@ -43,11 +43,11 @@ void WebRtcSpl_VectorBitShiftW16(int16_t *res, int16_t length,
 }
 
 void WebRtcSpl_VectorBitShiftW32(int32_t *out_vector,
-                                 int16_t vector_length,
+                                 size_t vector_length,
                                  const int32_t *in_vector,
                                  int16_t right_shifts)
 {
-    int i;
+    size_t i;
 
     if (right_shifts > 0)
     {
@@ -64,9 +64,9 @@ void WebRtcSpl_VectorBitShiftW32(int32_t *out_vector,
     }
 }
 
-void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, int length,
+void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, size_t length,
                                       const int32_t* in, int right_shifts) {
-  int i;
+  size_t i;
   int32_t tmp_w32;
 
   if (right_shifts >= 0) {
@@ -84,11 +84,11 @@ void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, int length,
 }
 
 void WebRtcSpl_ScaleVector(const int16_t *in_vector, int16_t *out_vector,
-                           int16_t gain, int16_t in_vector_length,
+                           int16_t gain, size_t in_vector_length,
                            int16_t right_shifts)
 {
     // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
-    int i;
+    size_t i;
     const int16_t *inptr;
     int16_t *outptr;
 
@@ -102,11 +102,11 @@ void WebRtcSpl_ScaleVector(const int16_t *in_vector, int16_t *out_vector,
 }
 
 void WebRtcSpl_ScaleVectorWithSat(const int16_t *in_vector, int16_t *out_vector,
-                                 int16_t gain, int16_t in_vector_length,
+                                 int16_t gain, size_t in_vector_length,
                                  int16_t right_shifts)
 {
     // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
-    int i;
+    size_t i;
     const int16_t *inptr;
     int16_t *outptr;
 
@@ -120,10 +120,10 @@ void WebRtcSpl_ScaleVectorWithSat(const int16_t *in_vector, int16_t *out_vector,
 
 void WebRtcSpl_ScaleAndAddVectors(const int16_t *in1, int16_t gain1, int shift1,
                                   const int16_t *in2, int16_t gain2, int shift2,
-                                  int16_t *out, int vector_length)
+                                  int16_t *out, size_t vector_length)
 {
     // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
-    int i;
+    size_t i;
     const int16_t *in1ptr;
     const int16_t *in2ptr;
     int16_t *outptr;
@@ -146,20 +146,19 @@ int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
                                            int16_t in_vector2_scale,
                                            int right_shifts,
                                            int16_t* out_vector,
-                                           int length) {
-  int i = 0;
+                                           size_t length) {
+  size_t i = 0;
   int round_value = (1 << right_shifts) >> 1;
 
   if (in_vector1 == NULL || in_vector2 == NULL || out_vector == NULL ||
-      length <= 0 || right_shifts < 0) {
+      length == 0 || right_shifts < 0) {
     return -1;
   }
 
   for (i = 0; i < length; i++) {
     out_vector[i] = (int16_t)((
-        WEBRTC_SPL_MUL_16_16(in_vector1[i], in_vector1_scale)
-        + WEBRTC_SPL_MUL_16_16(in_vector2[i], in_vector2_scale)
-        + round_value) >> right_shifts);
+        in_vector1[i] * in_vector1_scale + in_vector2[i] * in_vector2_scale +
+        round_value) >> right_shifts);
   }
 
   return 0;
diff --git a/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations_mips.c b/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations_mips.c
index 5ddcd2df7a..dd73eeaebb 100644
--- a/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations_mips.c
+++ b/media/webrtc/trunk/webrtc/common_audio/signal_processing/vector_scaling_operations_mips.c
@@ -22,15 +22,16 @@ int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(const int16_t* in_vector1,
                                                int16_t in_vector2_scale,
                                                int right_shifts,
                                                int16_t* out_vector,
-                                               int length) {
+                                               size_t length) {
   int16_t r0 = 0, r1 = 0;
   int16_t *in1 = (int16_t*)in_vector1;
   int16_t *in2 = (int16_t*)in_vector2;
   int16_t *out = out_vector;
-  int i = 0, value32 = 0;
+  size_t i = 0;
+  int value32 = 0;
 
   if (in_vector1 == NULL || in_vector2 == NULL || out_vector == NULL ||
-      length <= 0 || right_shifts < 0) {
+      length == 0 || right_shifts < 0) {
     return -1;
   }
   for (i = 0; i < length; i++) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter.cc b/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter.cc
new file mode 100644
index 0000000000..5862b7cc6b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_audio/sparse_fir_filter.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+SparseFIRFilter::SparseFIRFilter(const float* nonzero_coeffs,
+                                 size_t num_nonzero_coeffs,
+                                 size_t sparsity,
+                                 size_t offset)
+    : sparsity_(sparsity),
+      offset_(offset),
+      nonzero_coeffs_(nonzero_coeffs, nonzero_coeffs + num_nonzero_coeffs),
+      state_(sparsity_ * (num_nonzero_coeffs - 1) + offset_, 0.f) {
+  RTC_CHECK_GE(num_nonzero_coeffs, 1u);
+  RTC_CHECK_GE(sparsity, 1u);
+}
+
+void SparseFIRFilter::Filter(const float* in, size_t length, float* out) {
+  // Convolves the input signal |in| with the filter kernel |nonzero_coeffs_|
+  // taking into account the previous state.
+  for (size_t i = 0; i < length; ++i) {
+    out[i] = 0.f;
+    size_t j;
+    for (j = 0; i >= j * sparsity_ + offset_ &&
+                j < nonzero_coeffs_.size(); ++j) {
+      out[i] += in[i - j * sparsity_ - offset_] * nonzero_coeffs_[j];
+    }
+    for (; j < nonzero_coeffs_.size(); ++j) {
+      out[i] += state_[i + (nonzero_coeffs_.size() - j - 1) * sparsity_] *
+                nonzero_coeffs_[j];
+    }
+  }
+
+  // Update current state.
+  if (state_.size() > 0u) {
+    if (length >= state_.size()) {
+      std::memcpy(&state_[0],
+                  &in[length - state_.size()],
+                  state_.size() * sizeof(*in));
+    } else {
+      std::memmove(&state_[0],
+                   &state_[length],
+                   (state_.size() - length) * sizeof(state_[0]));
+      std::memcpy(&state_[state_.size() - length], in, length * sizeof(*in));
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter.h b/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter.h
new file mode 100644
index 0000000000..2ba5cf4600
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_SPARSE_FIR_FILTER_H_
+#define WEBRTC_COMMON_AUDIO_SPARSE_FIR_FILTER_H_
+
+#include <cstring>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+
+namespace webrtc {
+
+// A Finite Impulse Response filter implementation which takes advantage of a
+// sparse structure with uniformly distributed non-zero coefficients.
+class SparseFIRFilter final {
+ public:
+  // |num_nonzero_coeffs| is the number of non-zero coefficients,
+  // |nonzero_coeffs|. They are assumed to be uniformly distributed every
+  // |sparsity| samples and with an initial |offset|. The rest of the filter
+  // coefficients will be assumed zeros. For example, with sparsity = 3, and
+  // offset = 1 the filter coefficients will be:
+  // B = [0 coeffs[0] 0 0 coeffs[1] 0 0 coeffs[2] ... ]
+  // All initial state values will be zeros.
+  SparseFIRFilter(const float* nonzero_coeffs,
+                  size_t num_nonzero_coeffs,
+                  size_t sparsity,
+                  size_t offset);
+
+  // Filters the |in| data supplied.
+  // |out| must be previously allocated and it must be at least of |length|.
+  void Filter(const float* in, size_t length, float* out);
+
+ private:
+  const size_t sparsity_;
+  const size_t offset_;
+  const std::vector<float> nonzero_coeffs_;
+  std::vector<float> state_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(SparseFIRFilter);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_SPARSE_FIR_FILTER_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter_unittest.cc
new file mode 100644
index 0000000000..82a53a5287
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_audio/sparse_fir_filter_unittest.cc
@@ -0,0 +1,231 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_audio/sparse_fir_filter.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/fir_filter.h"
+
+namespace webrtc {
+namespace {
+
+static const float kCoeffs[] = {0.2f, 0.3f, 0.5f, 0.7f, 0.11f};
+static const float kInput[] =
+    {1.f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f, 8.f, 9.f, 10.f};
+
+template <size_t N>
+void VerifyOutput(const float (&expected_output)[N], const float (&output)[N]) {
+  EXPECT_EQ(0, memcmp(expected_output, output, sizeof(output)));
+}
+
+}  // namespace
+
+TEST(SparseFIRFilterTest, FilterAsIdentity) {
+  const float kCoeff = 1.f;
+  const size_t kNumCoeff = 1;
+  const size_t kSparsity = 3;
+  const size_t kOffset = 0;
+  float output[arraysize(kInput)];
+  SparseFIRFilter filter(&kCoeff, kNumCoeff, kSparsity, kOffset);
+  filter.Filter(kInput, arraysize(kInput), output);
+  VerifyOutput(kInput, output);
+}
+
+TEST(SparseFIRFilterTest, SameOutputForScalarCoefficientAndDifferentSparsity) {
+  const float kCoeff = 2.f;
+  const size_t kNumCoeff = 1;
+  const size_t kLowSparsity = 1;
+  const size_t kHighSparsity = 7;
+  const size_t kOffset = 0;
+  float low_sparsity_output[arraysize(kInput)];
+  float high_sparsity_output[arraysize(kInput)];
+  SparseFIRFilter low_sparsity_filter(&kCoeff,
+                                      kNumCoeff,
+                                      kLowSparsity,
+                                      kOffset);
+  SparseFIRFilter high_sparsity_filter(&kCoeff,
+                                       kNumCoeff,
+                                       kHighSparsity,
+                                       kOffset);
+  low_sparsity_filter.Filter(kInput, arraysize(kInput), low_sparsity_output);
+  high_sparsity_filter.Filter(kInput, arraysize(kInput), high_sparsity_output);
+  VerifyOutput(low_sparsity_output, high_sparsity_output);
+}
+
+TEST(SparseFIRFilterTest, FilterUsedAsScalarMultiplication) {
+  const float kCoeff = 5.f;
+  const size_t kNumCoeff = 1;
+  const size_t kSparsity = 5;
+  const size_t kOffset = 0;
+  float output[arraysize(kInput)];
+  SparseFIRFilter filter(&kCoeff, kNumCoeff, kSparsity, kOffset);
+  filter.Filter(kInput, arraysize(kInput), output);
+  EXPECT_FLOAT_EQ(5.f, output[0]);
+  EXPECT_FLOAT_EQ(20.f, output[3]);
+  EXPECT_FLOAT_EQ(25.f, output[4]);
+  EXPECT_FLOAT_EQ(50.f, output[arraysize(kInput) - 1]);
+}
+
+TEST(SparseFIRFilterTest, FilterUsedAsInputShifting) {
+  const float kCoeff = 1.f;
+  const size_t kNumCoeff = 1;
+  const size_t kSparsity = 1;
+  const size_t kOffset = 4;
+  float output[arraysize(kInput)];
+  SparseFIRFilter filter(&kCoeff, kNumCoeff, kSparsity, kOffset);
+  filter.Filter(kInput, arraysize(kInput), output);
+  EXPECT_FLOAT_EQ(0.f, output[0]);
+  EXPECT_FLOAT_EQ(0.f, output[3]);
+  EXPECT_FLOAT_EQ(1.f, output[4]);
+  EXPECT_FLOAT_EQ(2.f, output[5]);
+  EXPECT_FLOAT_EQ(6.f, output[arraysize(kInput) - 1]);
+}
+
+TEST(SparseFIRFilterTest, FilterUsedAsArbitraryWeighting) {
+  const size_t kSparsity = 2;
+  const size_t kOffset = 1;
+  float output[arraysize(kInput)];
+  SparseFIRFilter filter(kCoeffs, arraysize(kCoeffs), kSparsity, kOffset);
+  filter.Filter(kInput, arraysize(kInput), output);
+  EXPECT_FLOAT_EQ(0.f, output[0]);
+  EXPECT_FLOAT_EQ(0.9f, output[3]);
+  EXPECT_FLOAT_EQ(1.4f, output[4]);
+  EXPECT_FLOAT_EQ(2.4f, output[5]);
+  EXPECT_FLOAT_EQ(8.61f, output[arraysize(kInput) - 1]);
+}
+
+TEST(SparseFIRFilterTest, FilterInLengthLesserOrEqualToCoefficientsLength) {
+  const size_t kSparsity = 1;
+  const size_t kOffset = 0;
+  float output[arraysize(kInput)];
+  SparseFIRFilter filter(kCoeffs, arraysize(kCoeffs), kSparsity, kOffset);
+  filter.Filter(kInput, 2, output);
+  EXPECT_FLOAT_EQ(0.2f, output[0]);
+  EXPECT_FLOAT_EQ(0.7f, output[1]);
+}
+
+TEST(SparseFIRFilterTest, MultipleFilterCalls) {
+  const size_t kSparsity = 1;
+  const size_t kOffset = 0;
+  float output[arraysize(kInput)];
+  SparseFIRFilter filter(kCoeffs, arraysize(kCoeffs), kSparsity, kOffset);
+  filter.Filter(kInput, 2, output);
+  EXPECT_FLOAT_EQ(0.2f, output[0]);
+  EXPECT_FLOAT_EQ(0.7f, output[1]);
+  filter.Filter(kInput, 2, output);
+  EXPECT_FLOAT_EQ(1.3f, output[0]);
+  EXPECT_FLOAT_EQ(2.4f, output[1]);
+  filter.Filter(kInput, 2, output);
+  EXPECT_FLOAT_EQ(2.81f, output[0]);
+  EXPECT_FLOAT_EQ(2.62f, output[1]);
+  filter.Filter(kInput, 2, output);
+  EXPECT_FLOAT_EQ(2.81f, output[0]);
+  EXPECT_FLOAT_EQ(2.62f, output[1]);
+  filter.Filter(&kInput[3], 3, output);
+  EXPECT_FLOAT_EQ(3.41f, output[0]);
+  EXPECT_FLOAT_EQ(4.12f, output[1]);
+  EXPECT_FLOAT_EQ(6.21f, output[2]);
+  filter.Filter(&kInput[3], 3, output);
+  EXPECT_FLOAT_EQ(8.12f, output[0]);
+  EXPECT_FLOAT_EQ(9.14f, output[1]);
+  EXPECT_FLOAT_EQ(9.45f, output[2]);
+}
+
+TEST(SparseFIRFilterTest, VerifySampleBasedVsBlockBasedFiltering) {
+  const size_t kSparsity = 3;
+  const size_t kOffset = 1;
+  float output_block_based[arraysize(kInput)];
+  SparseFIRFilter filter_block(kCoeffs,
+                               arraysize(kCoeffs),
+                               kSparsity,
+                               kOffset);
+  filter_block.Filter(kInput, arraysize(kInput), output_block_based);
+  float output_sample_based[arraysize(kInput)];
+  SparseFIRFilter filter_sample(kCoeffs,
+                                arraysize(kCoeffs),
+                                kSparsity,
+                                kOffset);
+  for (size_t i = 0; i < arraysize(kInput); ++i)
+    filter_sample.Filter(&kInput[i], 1, &output_sample_based[i]);
+  VerifyOutput(output_block_based, output_sample_based);
+}
+
+TEST(SparseFIRFilterTest, SimpleHighPassFilter) {
+  const size_t kSparsity = 2;
+  const size_t kOffset = 2;
+  const float kHPCoeffs[] = {1.f, -1.f};
+  const float kConstantInput[] =
+      {1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f};
+  float output[arraysize(kConstantInput)];
+  SparseFIRFilter filter(kHPCoeffs, arraysize(kHPCoeffs), kSparsity, kOffset);
+  filter.Filter(kConstantInput, arraysize(kConstantInput), output);
+  EXPECT_FLOAT_EQ(0.f, output[0]);
+  EXPECT_FLOAT_EQ(0.f, output[1]);
+  EXPECT_FLOAT_EQ(1.f, output[2]);
+  EXPECT_FLOAT_EQ(1.f, output[3]);
+  for (size_t i = kSparsity + kOffset; i < arraysize(kConstantInput); ++i)
+    EXPECT_FLOAT_EQ(0.f, output[i]);
+}
+
+TEST(SparseFIRFilterTest, SimpleLowPassFilter) {
+  const size_t kSparsity = 2;
+  const size_t kOffset = 2;
+  const float kLPCoeffs[] = {1.f, 1.f};
+  const float kHighFrequencyInput[] =
+      {1.f, 1.f, -1.f, -1.f, 1.f, 1.f, -1.f, -1.f, 1.f, 1.f};
+  float output[arraysize(kHighFrequencyInput)];
+  SparseFIRFilter filter(kLPCoeffs, arraysize(kLPCoeffs), kSparsity, kOffset);
+  filter.Filter(kHighFrequencyInput, arraysize(kHighFrequencyInput), output);
+  EXPECT_FLOAT_EQ(0.f, output[0]);
+  EXPECT_FLOAT_EQ(0.f, output[1]);
+  EXPECT_FLOAT_EQ(1.f, output[2]);
+  EXPECT_FLOAT_EQ(1.f, output[3]);
+  for (size_t i = kSparsity + kOffset; i < arraysize(kHighFrequencyInput); ++i)
+    EXPECT_FLOAT_EQ(0.f, output[i]);
+}
+
+TEST(SparseFIRFilterTest, SameOutputWhenSwappedCoefficientsAndInput) {
+  const size_t kSparsity = 1;
+  const size_t kOffset = 0;
+  float output[arraysize(kCoeffs)];
+  float output_swapped[arraysize(kCoeffs)];
+  SparseFIRFilter filter(kCoeffs, arraysize(kCoeffs), kSparsity, kOffset);
+  // Use arraysize(kCoeffs) for in_length to get same-length outputs.
+  filter.Filter(kInput, arraysize(kCoeffs), output);
+  SparseFIRFilter filter_swapped(kInput,
+                                 arraysize(kCoeffs),
+                                 kSparsity,
+                                 kOffset);
+  filter_swapped.Filter(kCoeffs, arraysize(kCoeffs), output_swapped);
+  VerifyOutput(output, output_swapped);
+}
+
+TEST(SparseFIRFilterTest, SameOutputAsFIRFilterWhenSparsityOneAndOffsetZero) {
+  const size_t kSparsity = 1;
+  const size_t kOffset = 0;
+  float output[arraysize(kInput)];
+  float sparse_output[arraysize(kInput)];
+  rtc::scoped_ptr<FIRFilter> filter(FIRFilter::Create(kCoeffs,
+                                                      arraysize(kCoeffs),
+                                                      arraysize(kInput)));
+  SparseFIRFilter sparse_filter(kCoeffs,
+                                arraysize(kCoeffs),
+                                kSparsity,
+                                kOffset);
+  filter->Filter(kInput, arraysize(kInput), output);
+  sparse_filter.Filter(kInput, arraysize(kInput), sparse_output);
+  for (size_t i = 0; i < arraysize(kInput); ++i) {
+    EXPECT_FLOAT_EQ(output[i], sparse_output[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/swap_queue.h b/media/webrtc/trunk/webrtc/common_audio/swap_queue.h
new file mode 100644
index 0000000000..d8bb5c024e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_audio/swap_queue.h
@@ -0,0 +1,210 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_SWAP_QUEUE_H_
+#define WEBRTC_COMMON_AUDIO_SWAP_QUEUE_H_
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+
+namespace webrtc {
+
+namespace internal {
+
+// (Internal; please don't use outside this file.)
+template <typename T>
+bool NoopSwapQueueItemVerifierFunction(const T&) {
+  return true;
+}
+
+}  // namespace internal
+
+// Functor to use when supplying a verifier function for the queue.
+template <typename T,
+          bool (*QueueItemVerifierFunction)(const T&) =
+              internal::NoopSwapQueueItemVerifierFunction>
+class SwapQueueItemVerifier {
+ public:
+  bool operator()(const T& t) const { return QueueItemVerifierFunction(t); }
+};
+
+// This class is a fixed-size queue. A producer calls Insert() to insert
+// an element of type T at the back of the queue, and a consumer calls
+// Remove() to remove an element from the front of the queue. It's safe
+// for the producer(s) and the consumer(s) to access the queue
+// concurrently, from different threads.
+//
+// To avoid the construction, copying, and destruction of Ts that a naive
+// queue implementation would require, for each "full" T passed from
+// producer to consumer, SwapQueue<T> passes an "empty" T in the other
+// direction (an "empty" T is one that contains nothing of value for the
+// consumer). This bidirectional movement is implemented with swap().
+//
+// // Create queue:
+// Bottle proto(568);  // Prepare an empty Bottle. Heap allocates space for
+//                     // 568 ml.
+// SwapQueue<Bottle> q(N, proto);  // Init queue with N copies of proto.
+//                                 // Each copy allocates on the heap.
+// // Producer pseudo-code:
+// Bottle b(568); // Prepare an empty Bottle. Heap allocates space for 568 ml.
+// loop {
+//   b.Fill(amount);  // Where amount <= 568 ml.
+//   q.Insert(&b);    // Swap our full Bottle for an empty one from q.
+// }
+//
+// // Consumer pseudo-code:
+// Bottle b(568);  // Prepare an empty Bottle. Heap allocates space for 568 ml.
+// loop {
+//   q.Remove(&b); // Swap our empty Bottle for the next-in-line full Bottle.
+//   Drink(&b);
+// }
+//
+// For a well-behaved Bottle class, there are no allocations in the
+// producer, since it just fills an empty Bottle that's already large
+// enough; no deallocations in the consumer, since it returns each empty
+// Bottle to the queue after having drunk it; and no copies along the
+// way, since the queue uses swap() everywhere to move full Bottles in
+// one direction and empty ones in the other.
+template <typename T, typename QueueItemVerifier = SwapQueueItemVerifier<T>>
+class SwapQueue {
+ public:
+  // Creates a queue of size size and fills it with default constructed Ts.
+  explicit SwapQueue(size_t size) : queue_(size) {
+    RTC_DCHECK(VerifyQueueSlots());
+  }
+
+  // Same as above and accepts an item verification functor.
+  SwapQueue(size_t size, const QueueItemVerifier& queue_item_verifier)
+      : queue_item_verifier_(queue_item_verifier), queue_(size) {
+    RTC_DCHECK(VerifyQueueSlots());
+  }
+
+  // Creates a queue of size size and fills it with copies of prototype.
+  SwapQueue(size_t size, const T& prototype) : queue_(size, prototype) {
+    RTC_DCHECK(VerifyQueueSlots());
+  }
+
+  // Same as above and accepts an item verification functor.
+  SwapQueue(size_t size,
+            const T& prototype,
+            const QueueItemVerifier& queue_item_verifier)
+      : queue_item_verifier_(queue_item_verifier), queue_(size, prototype) {
+    RTC_DCHECK(VerifyQueueSlots());
+  }
+
+  // Resets the queue to have zero content wile maintaining the queue size.
+  void Clear() {
+    rtc::CritScope cs(&crit_queue_);
+    next_write_index_ = 0;
+    next_read_index_ = 0;
+    num_elements_ = 0;
+  }
+
+  // Inserts a "full" T at the back of the queue by swapping *input with an
+  // "empty" T from the queue.
+  // Returns true if the item was inserted or false if not (the queue was full).
+  // When specified, the T given in *input must pass the ItemVerifier() test.
+  // The contents of *input after the call are then also guaranteed to pass the
+  // ItemVerifier() test.
+  bool Insert(T* input) WARN_UNUSED_RESULT {
+    RTC_DCHECK(input);
+
+    rtc::CritScope cs(&crit_queue_);
+
+    RTC_DCHECK(queue_item_verifier_(*input));
+
+    if (num_elements_ == queue_.size()) {
+      return false;
+    }
+
+    using std::swap;
+    swap(*input, queue_[next_write_index_]);
+
+    ++next_write_index_;
+    if (next_write_index_ == queue_.size()) {
+      next_write_index_ = 0;
+    }
+
+    ++num_elements_;
+
+    RTC_DCHECK_LT(next_write_index_, queue_.size());
+    RTC_DCHECK_LE(num_elements_, queue_.size());
+
+    return true;
+  }
+
+  // Removes the frontmost "full" T from the queue by swapping it with
+  // the "empty" T in *output.
+  // Returns true if an item could be removed or false if not (the queue was
+  // empty). When specified, The T given in *output must pass the ItemVerifier()
+  // test and the contents of *output after the call are then also guaranteed to
+  // pass the ItemVerifier() test.
+  bool Remove(T* output) WARN_UNUSED_RESULT {
+    RTC_DCHECK(output);
+
+    rtc::CritScope cs(&crit_queue_);
+
+    RTC_DCHECK(queue_item_verifier_(*output));
+
+    if (num_elements_ == 0) {
+      return false;
+    }
+
+    using std::swap;
+    swap(*output, queue_[next_read_index_]);
+
+    ++next_read_index_;
+    if (next_read_index_ == queue_.size()) {
+      next_read_index_ = 0;
+    }
+
+    --num_elements_;
+
+    RTC_DCHECK_LT(next_read_index_, queue_.size());
+    RTC_DCHECK_LE(num_elements_, queue_.size());
+
+    return true;
+  }
+
+ private:
+  // Verify that the queue slots complies with the ItemVerifier test.
+  bool VerifyQueueSlots() {
+    rtc::CritScope cs(&crit_queue_);
+    for (const auto& v : queue_) {
+      RTC_DCHECK(queue_item_verifier_(v));
+    }
+    return true;
+  }
+
+  rtc::CriticalSection crit_queue_;
+
+  // TODO(peah): Change this to use std::function() once we can use C++11 std
+  // lib.
+  QueueItemVerifier queue_item_verifier_ GUARDED_BY(crit_queue_);
+
+  // (next_read_index_ + num_elements_) % queue_.size() =
+  //  next_write_index_
+  size_t next_write_index_ GUARDED_BY(crit_queue_) = 0;
+  size_t next_read_index_ GUARDED_BY(crit_queue_) = 0;
+  size_t num_elements_ GUARDED_BY(crit_queue_) = 0;
+
+  // queue_.size() is constant.
+  std::vector<T> queue_ GUARDED_BY(crit_queue_);
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(SwapQueue);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_SWAP_QUEUE_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/swap_queue_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/swap_queue_unittest.cc
new file mode 100644
index 0000000000..104e494bc6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_audio/swap_queue_unittest.cc
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_audio/swap_queue.h"
+
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Test parameter for the basic sample based SwapQueue Tests.
+const size_t kChunkSize = 3;
+
+// Queue item verification function for the vector test.
+bool LengthVerifierFunction(const std::vector<int>& v) {
+  return v.size() == kChunkSize;
+}
+
+// Queue item verifier for the vector test.
+class LengthVerifierFunctor {
+ public:
+  explicit LengthVerifierFunctor(size_t length) : length_(length) {}
+
+  bool operator()(const std::vector<int>& v) const {
+    return v.size() == length_;
+  }
+
+ private:
+  size_t length_;
+};
+
+}  // anonymous namespace
+
+TEST(SwapQueueTest, BasicOperation) {
+  std::vector<int> i(kChunkSize, 0);
+  SwapQueue<std::vector<int>> queue(2, i);
+
+  EXPECT_TRUE(queue.Insert(&i));
+  EXPECT_EQ(i.size(), kChunkSize);
+  EXPECT_TRUE(queue.Insert(&i));
+  EXPECT_EQ(i.size(), kChunkSize);
+  EXPECT_TRUE(queue.Remove(&i));
+  EXPECT_EQ(i.size(), kChunkSize);
+  EXPECT_TRUE(queue.Remove(&i));
+  EXPECT_EQ(i.size(), kChunkSize);
+}
+
+TEST(SwapQueueTest, FullQueue) {
+  SwapQueue<int> queue(2);
+
+  // Fill the queue.
+  int i = 0;
+  EXPECT_TRUE(queue.Insert(&i));
+  i = 1;
+  EXPECT_TRUE(queue.Insert(&i));
+
+  // Ensure that the value is not swapped when doing an Insert
+  // on a full queue.
+  i = 2;
+  EXPECT_FALSE(queue.Insert(&i));
+  EXPECT_EQ(i, 2);
+
+  // Ensure that the Insert didn't overwrite anything in the queue.
+  EXPECT_TRUE(queue.Remove(&i));
+  EXPECT_EQ(i, 0);
+  EXPECT_TRUE(queue.Remove(&i));
+  EXPECT_EQ(i, 1);
+}
+
+TEST(SwapQueueTest, EmptyQueue) {
+  SwapQueue<int> queue(2);
+  int i = 0;
+  EXPECT_FALSE(queue.Remove(&i));
+  EXPECT_TRUE(queue.Insert(&i));
+  EXPECT_TRUE(queue.Remove(&i));
+  EXPECT_FALSE(queue.Remove(&i));
+}
+
+TEST(SwapQueueTest, Clear) {
+  SwapQueue<int> queue(2);
+  int i = 0;
+
+  // Fill the queue.
+  EXPECT_TRUE(queue.Insert(&i));
+  EXPECT_TRUE(queue.Insert(&i));
+
+  // Ensure full queue.
+  EXPECT_FALSE(queue.Insert(&i));
+
+  // Empty the queue.
+  queue.Clear();
+
+  // Ensure that the queue is empty
+  EXPECT_FALSE(queue.Remove(&i));
+
+  // Ensure that the queue is no longer full.
+  EXPECT_TRUE(queue.Insert(&i));
+}
+
+TEST(SwapQueueTest, SuccessfulItemVerifyFunction) {
+  std::vector<int> template_element(kChunkSize);
+  SwapQueue<std::vector<int>,
+            SwapQueueItemVerifier<std::vector<int>, LengthVerifierFunction>>
+      queue(2, template_element);
+  std::vector<int> valid_chunk(kChunkSize, 0);
+
+  EXPECT_TRUE(queue.Insert(&valid_chunk));
+  EXPECT_EQ(valid_chunk.size(), kChunkSize);
+  EXPECT_TRUE(queue.Remove(&valid_chunk));
+  EXPECT_EQ(valid_chunk.size(), kChunkSize);
+}
+
+TEST(SwapQueueTest, SuccessfulItemVerifyFunctor) {
+  std::vector<int> template_element(kChunkSize);
+  LengthVerifierFunctor verifier(kChunkSize);
+  SwapQueue<std::vector<int>, LengthVerifierFunctor> queue(2, template_element,
+                                                           verifier);
+  std::vector<int> valid_chunk(kChunkSize, 0);
+
+  EXPECT_TRUE(queue.Insert(&valid_chunk));
+  EXPECT_EQ(valid_chunk.size(), kChunkSize);
+  EXPECT_TRUE(queue.Remove(&valid_chunk));
+  EXPECT_EQ(valid_chunk.size(), kChunkSize);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(SwapQueueTest, UnsuccessfulItemVerifyFunctor) {
+  // Queue item verifier for the test.
+  auto minus_2_verifier = [](const int& i) { return i > -2; };
+  SwapQueue<int, decltype(minus_2_verifier)> queue(2, minus_2_verifier);
+
+  int valid_value = 1;
+  int invalid_value = -4;
+  EXPECT_TRUE(queue.Insert(&valid_value));
+  EXPECT_TRUE(queue.Remove(&valid_value));
+  bool result;
+  EXPECT_DEATH(result = queue.Insert(&invalid_value), "");
+}
+
+TEST(SwapQueueTest, UnSuccessfulItemVerifyInsert) {
+  std::vector<int> template_element(kChunkSize);
+  SwapQueue<std::vector<int>,
+            SwapQueueItemVerifier<std::vector<int>, &LengthVerifierFunction>>
+      queue(2, template_element);
+  std::vector<int> invalid_chunk(kChunkSize - 1, 0);
+  bool result;
+  EXPECT_DEATH(result = queue.Insert(&invalid_chunk), "");
+}
+
+TEST(SwapQueueTest, UnSuccessfulItemVerifyRemove) {
+  std::vector<int> template_element(kChunkSize);
+  SwapQueue<std::vector<int>,
+            SwapQueueItemVerifier<std::vector<int>, &LengthVerifierFunction>>
+      queue(2, template_element);
+  std::vector<int> invalid_chunk(kChunkSize - 1, 0);
+  std::vector<int> valid_chunk(kChunkSize, 0);
+  EXPECT_TRUE(queue.Insert(&valid_chunk));
+  EXPECT_EQ(valid_chunk.size(), kChunkSize);
+  bool result;
+  EXPECT_DEATH(result = queue.Remove(&invalid_chunk), "");
+}
+#endif
+
+TEST(SwapQueueTest, VectorContentTest) {
+  const size_t kQueueSize = 10;
+  const size_t kFrameLength = 160;
+  const size_t kDataLength = kQueueSize * kFrameLength;
+  std::vector<int16_t> buffer_reader(kFrameLength, 0);
+  std::vector<int16_t> buffer_writer(kFrameLength, 0);
+  SwapQueue<std::vector<int16_t>> queue(kQueueSize,
+                                        std::vector<int16_t>(kFrameLength));
+  std::vector<int16_t> samples(kDataLength);
+
+  for (size_t k = 0; k < kDataLength; k++) {
+    samples[k] = k % 9;
+  }
+
+  for (size_t k = 0; k < kQueueSize; k++) {
+    buffer_writer.clear();
+    buffer_writer.insert(buffer_writer.end(), &samples[0] + k * kFrameLength,
+                         &samples[0] + (k + 1) * kFrameLength);
+
+    EXPECT_TRUE(queue.Insert(&buffer_writer));
+  }
+
+  for (size_t k = 0; k < kQueueSize; k++) {
+    EXPECT_TRUE(queue.Remove(&buffer_reader));
+
+    for (size_t j = 0; j < buffer_reader.size(); j++) {
+      EXPECT_EQ(buffer_reader[j], samples[k * kFrameLength + j]);
+    }
+  }
+}
+
+TEST(SwapQueueTest, ZeroSlotQueue) {
+  SwapQueue<int> queue(0);
+  int i = 42;
+  EXPECT_FALSE(queue.Insert(&i));
+  EXPECT_FALSE(queue.Remove(&i));
+  EXPECT_EQ(i, 42);
+}
+
+TEST(SwapQueueTest, OneSlotQueue) {
+  SwapQueue<int> queue(1);
+  int i = 42;
+  EXPECT_TRUE(queue.Insert(&i));
+  i = 43;
+  EXPECT_FALSE(queue.Insert(&i));
+  EXPECT_EQ(i, 43);
+  EXPECT_TRUE(queue.Remove(&i));
+  EXPECT_EQ(i, 42);
+  EXPECT_FALSE(queue.Remove(&i));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/include/vad.h b/media/webrtc/trunk/webrtc/common_audio/vad/include/vad.h
index 1944f9dc5a..087970f58e 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/include/vad.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/include/vad.h
@@ -12,12 +12,12 @@
 #define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_VAD_H_
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-// This is a C++ wrapper class for WebRtcVad.
 class Vad {
  public:
   enum Aggressiveness {
@@ -29,17 +29,22 @@ class Vad {
 
   enum Activity { kPassive = 0, kActive = 1, kError = -1 };
 
-  explicit Vad(enum Aggressiveness mode);
-
-  virtual ~Vad();
+  virtual ~Vad() = default;
 
+  // Calculates a VAD decision for the given audio frame. Valid sample rates
+  // are 8000, 16000, and 32000 Hz; the number of samples must be such that the
+  // frame is 10, 20, or 30 ms long.
   virtual Activity VoiceActivity(const int16_t* audio,
                                  size_t num_samples,
-                                 int sample_rate_hz);
+                                 int sample_rate_hz) = 0;
 
- private:
-  VadInst* handle_;
+  // Resets VAD state.
+  virtual void Reset() = 0;
 };
 
+// Returns a Vad instance that's implemented on top of WebRtcVad.
+rtc::scoped_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness);
+
 }  // namespace webrtc
+
 #endif  // WEBRTC_COMMON_AUDIO_VAD_INCLUDE_VAD_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/include/webrtc_vad.h b/media/webrtc/trunk/webrtc/common_audio/vad/include/webrtc_vad.h
index 053827303b..91308eef12 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/include/webrtc_vad.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/include/webrtc_vad.h
@@ -16,6 +16,8 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
 #define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
 
+#include <stddef.h>
+
 #include "webrtc/typedefs.h"
 
 typedef struct WebRtcVadInst VadInst;
@@ -25,11 +27,7 @@ extern "C" {
 #endif
 
 // Creates an instance to the VAD structure.
-//
-// - handle [o] : Pointer to the VAD instance that should be created.
-//
-// returns      : 0 - (OK), -1 - (Error)
-int WebRtcVad_Create(VadInst** handle);
+VadInst* WebRtcVad_Create();
 
 // Frees the dynamic memory of a specified VAD instance.
 //
@@ -70,7 +68,7 @@ int WebRtcVad_set_mode(VadInst* handle, int mode);
 //                        0 - (Non-active Voice),
 //                       -1 - (Error)
 int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
-                      int frame_length);
+                      size_t frame_length);
 
 // Checks for valid combinations of |rate| and |frame_length|. We support 10,
 // 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
@@ -79,7 +77,7 @@ int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
 // - frame_length [i] : Speech frame buffer length in number of samples.
 //
 // returns            : 0 - (valid combination), -1 - (invalid combination)
-int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length);
+int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length);
 
 #ifdef __cplusplus
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/mock/mock_vad.h b/media/webrtc/trunk/webrtc/common_audio/vad/mock/mock_vad.h
index 7a7de0fa7d..bc763bb9d9 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/mock/mock_vad.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/mock/mock_vad.h
@@ -19,7 +19,6 @@ namespace webrtc {
 
 class MockVad : public Vad {
  public:
-  explicit MockVad(enum Aggressiveness mode) : Vad(mode) {}
   virtual ~MockVad() { Die(); }
   MOCK_METHOD0(Die, void());
 
@@ -27,6 +26,7 @@ class MockVad : public Vad {
                enum Activity(const int16_t* audio,
                              size_t num_samples,
                              int sample_rate_hz));
+  MOCK_METHOD0(Reset, void());
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad.cc b/media/webrtc/trunk/webrtc/common_audio/vad/vad.cc
index 9cc0c19877..95a162fb92 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad.cc
@@ -14,30 +14,50 @@
 
 namespace webrtc {
 
-Vad::Vad(enum Aggressiveness mode) {
-  CHECK_EQ(WebRtcVad_Create(&handle_), 0);
-  CHECK_EQ(WebRtcVad_Init(handle_), 0);
-  CHECK_EQ(WebRtcVad_set_mode(handle_, mode), 0);
-}
+namespace {
 
-Vad::~Vad() {
-  WebRtcVad_Free(handle_);
-}
-
-enum Vad::Activity Vad::VoiceActivity(const int16_t* audio,
-                                      size_t num_samples,
-                                      int sample_rate_hz) {
-  int ret = WebRtcVad_Process(
-      handle_, sample_rate_hz, audio, static_cast<int>(num_samples));
-  switch (ret) {
-    case 0:
-      return kPassive;
-    case 1:
-      return kActive;
-    default:
-      DCHECK(false) << "WebRtcVad_Process returned an error.";
-      return kError;
+class VadImpl final : public Vad {
+ public:
+  explicit VadImpl(Aggressiveness aggressiveness)
+      : handle_(nullptr), aggressiveness_(aggressiveness) {
+    Reset();
   }
+
+  ~VadImpl() override { WebRtcVad_Free(handle_); }
+
+  Activity VoiceActivity(const int16_t* audio,
+                         size_t num_samples,
+                         int sample_rate_hz) override {
+    int ret = WebRtcVad_Process(handle_, sample_rate_hz, audio, num_samples);
+    switch (ret) {
+      case 0:
+        return kPassive;
+      case 1:
+        return kActive;
+      default:
+        RTC_DCHECK(false) << "WebRtcVad_Process returned an error.";
+        return kError;
+    }
+  }
+
+  void Reset() override {
+    if (handle_)
+      WebRtcVad_Free(handle_);
+    handle_ = WebRtcVad_Create();
+    RTC_CHECK(handle_);
+    RTC_CHECK_EQ(WebRtcVad_Init(handle_), 0);
+    RTC_CHECK_EQ(WebRtcVad_set_mode(handle_, aggressiveness_), 0);
+  }
+
+ private:
+  VadInst* handle_;
+  Aggressiveness aggressiveness_;
+};
+
+}  // namespace
+
+rtc::scoped_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness) {
+  return rtc::scoped_ptr<Vad>(new VadImpl(aggressiveness));
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.c b/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.c
index d246a4d954..51797eed54 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.c
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.c
@@ -122,7 +122,7 @@ static int32_t WeightedAverage(int16_t* data, int16_t offset,
 //
 // - returns              : the VAD decision (0 - noise, 1 - speech).
 static int16_t GmmProbability(VadInstT* self, int16_t* features,
-                              int16_t total_power, int frame_length) {
+                              int16_t total_power, size_t frame_length) {
   int channel, k;
   int16_t feature_minimum;
   int16_t h0, h1;
@@ -596,16 +596,16 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
 // probability for both speech and background noise.
 
 int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
-                           int frame_length) {
+                           size_t frame_length) {
   int vad;
-  int i;
+  size_t i;
   int16_t speech_nb[240];  // 30 ms in 8 kHz.
   // |tmp_mem| is a temporary memory used by resample function, length is
   // frame length in 10 ms (480 samples) + 256 extra.
   int32_t tmp_mem[480 + 256] = { 0 };
-  const int kFrameLen10ms48khz = 480;
-  const int kFrameLen10ms8khz = 80;
-  int num_10ms_frames = frame_length / kFrameLen10ms48khz;
+  const size_t kFrameLen10ms48khz = 480;
+  const size_t kFrameLen10ms8khz = 80;
+  size_t num_10ms_frames = frame_length / kFrameLen10ms48khz;
 
   for (i = 0; i < num_10ms_frames; i++) {
     WebRtcSpl_Resample48khzTo8khz(speech_frame,
@@ -621,9 +621,10 @@ int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
 }
 
 int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
-                           int frame_length)
+                           size_t frame_length)
 {
-    int len, vad;
+    size_t len;
+    int vad;
     int16_t speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
     int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
 
@@ -643,9 +644,10 @@ int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
 }
 
 int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
-                           int frame_length)
+                           size_t frame_length)
 {
-    int len, vad;
+    size_t len;
+    int vad;
     int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
 
     // Wideband: Downsample signal before doing VAD
@@ -659,7 +661,7 @@ int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
 }
 
 int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
-                          int frame_length)
+                          size_t frame_length)
 {
     int16_t feature_vector[kNumChannels], total_power;
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.h b/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.h
index 202963d8c6..b38c515ea1 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.h
@@ -104,12 +104,12 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode);
  *                        1-6 - Active speech
  */
 int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
-                           int frame_length);
+                           size_t frame_length);
 int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
-                           int frame_length);
+                           size_t frame_length);
 int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
-                           int frame_length);
+                           size_t frame_length);
 int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
-                          int frame_length);
+                          size_t frame_length);
 
 #endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_core_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/vad/vad_core_unittest.cc
index 77db3d8ed2..ee69484f0a 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_core_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_core_unittest.cc
@@ -82,8 +82,8 @@ TEST_F(VadTest, CalcVad) {
 
   // Construct a speech signal that will trigger the VAD in all modes. It is
   // known that (i * i) will wrap around, but that doesn't matter in this case.
-  for (int16_t i = 0; i < kMaxFrameLength; ++i) {
-    speech[i] = (i * i);
+  for (size_t i = 0; i < kMaxFrameLength; ++i) {
+    speech[i] = static_cast<int16_t>(i * i);
   }
   for (size_t j = 0; j < kFrameLengthsSize; ++j) {
     if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.c b/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.c
index 310678afc4..8b9df93b00 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.c
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.c
@@ -38,9 +38,9 @@ static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
 // - filter_state [i/o] : State of the filter.
 // - data_out     [o]   : Output audio data in the frequency interval
 //                        80 - 250 Hz.
-static void HighPassFilter(const int16_t* data_in, int data_length,
+static void HighPassFilter(const int16_t* data_in, size_t data_length,
                            int16_t* filter_state, int16_t* data_out) {
-  int i;
+  size_t i;
   const int16_t* in_ptr = data_in;
   int16_t* out_ptr = data_out;
   int32_t tmp32 = 0;
@@ -80,7 +80,7 @@ static void HighPassFilter(const int16_t* data_in, int data_length,
 // - filter_coefficient [i]   : Given in Q15.
 // - filter_state       [i/o] : State of the filter given in Q(-1).
 // - data_out           [o]   : Output audio signal given in Q(-1).
-static void AllPassFilter(const int16_t* data_in, int data_length,
+static void AllPassFilter(const int16_t* data_in, size_t data_length,
                           int16_t filter_coefficient, int16_t* filter_state,
                           int16_t* data_out) {
   // The filter can only cause overflow (in the w16 output variable)
@@ -89,17 +89,16 @@ static void AllPassFilter(const int16_t* data_in, int data_length,
   // First 6 taps of the impulse response:
   // 0.6399 0.5905 -0.3779 0.2418 -0.1547 0.0990
 
-  int i;
+  size_t i;
   int16_t tmp16 = 0;
   int32_t tmp32 = 0;
   int32_t state32 = ((int32_t) (*filter_state) << 16);  // Q15
 
   for (i = 0; i < data_length; i++) {
-    tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficient, *data_in);
+    tmp32 = state32 + filter_coefficient * *data_in;
     tmp16 = (int16_t) (tmp32 >> 16);  // Q(-1)
     *data_out++ = tmp16;
-    state32 = (((int32_t) (*data_in)) << 14); // Q14
-    state32 -= WEBRTC_SPL_MUL_16_16(filter_coefficient, tmp16);  // Q14
+    state32 = (*data_in << 14) - filter_coefficient * tmp16;  // Q14
     state32 <<= 1;  // Q15.
     data_in += 2;
   }
@@ -118,11 +117,11 @@ static void AllPassFilter(const int16_t* data_in, int data_length,
 //                        The length is |data_length| / 2.
 // - lp_data_out  [o]   : Output audio data of the lower half of the spectrum.
 //                        The length is |data_length| / 2.
-static void SplitFilter(const int16_t* data_in, int data_length,
+static void SplitFilter(const int16_t* data_in, size_t data_length,
                         int16_t* upper_state, int16_t* lower_state,
                         int16_t* hp_data_out, int16_t* lp_data_out) {
-  int i;
-  int half_length = data_length >> 1;  // Downsampling by 2.
+  size_t i;
+  size_t half_length = data_length >> 1;  // Downsampling by 2.
   int16_t tmp_out;
 
   // All-pass filtering upper branch.
@@ -152,7 +151,7 @@ static void SplitFilter(const int16_t* data_in, int data_length,
 //                        NOTE: |total_energy| is only updated if
 //                        |total_energy| <= |kMinEnergy|.
 // - log_energy   [o]   : 10 * log10("energy of |data_in|") given in Q4.
-static void LogOfEnergy(const int16_t* data_in, int data_length,
+static void LogOfEnergy(const int16_t* data_in, size_t data_length,
                         int16_t offset, int16_t* total_energy,
                         int16_t* log_energy) {
   // |tot_rshifts| accumulates the number of right shifts performed on |energy|.
@@ -244,7 +243,7 @@ static void LogOfEnergy(const int16_t* data_in, int data_length,
 }
 
 int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
-                                    int data_length, int16_t* features) {
+                                    size_t data_length, int16_t* features) {
   int16_t total_energy = 0;
   // We expect |data_length| to be 80, 160 or 240 samples, which corresponds to
   // 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
@@ -252,9 +251,9 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
   // the second split.
   int16_t hp_120[120], lp_120[120];
   int16_t hp_60[60], lp_60[60];
-  const int half_data_length = data_length >> 1;
-  int length = half_data_length;  // |data_length| / 2, corresponds to
-                                  // bandwidth = 2000 Hz after downsampling.
+  const size_t half_data_length = data_length >> 1;
+  size_t length = half_data_length;  // |data_length| / 2, corresponds to
+                                     // bandwidth = 2000 Hz after downsampling.
 
   // Initialize variables for the first SplitFilter().
   int frequency_band = 0;
@@ -262,7 +261,6 @@ int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
   int16_t* hp_out_ptr = hp_120;  // [2000 - 4000] Hz.
   int16_t* lp_out_ptr = lp_120;  // [0 - 2000] Hz.
 
-  assert(data_length >= 0);
   assert(data_length <= 240);
   assert(4 < kNumChannels - 1);  // Checking maximum |frequency_band|.
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.h b/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.h
index e9195e5e37..42bf3fc331 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.h
@@ -39,6 +39,6 @@
 // - returns            : Total energy of the signal (NOTE! This value is not
 //                        exact. It is only used in a comparison.)
 int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
-                                    int data_length, int16_t* features);
+                                    size_t data_length, int16_t* features);
 
 #endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank_unittest.cc
index d274c4b131..11b503a196 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank_unittest.cc
@@ -38,8 +38,8 @@ TEST_F(VadTest, vad_filterbank) {
   // Construct a speech signal that will trigger the VAD in all modes. It is
   // known that (i * i) will wrap around, but that doesn't matter in this case.
   int16_t speech[kMaxFrameLength];
-  for (int16_t i = 0; i < kMaxFrameLength; ++i) {
-    speech[i] = (i * i);
+  for (size_t i = 0; i < kMaxFrameLength; ++i) {
+    speech[i] = static_cast<int16_t>(i * i);
   }
 
   int frame_length_index = 0;
@@ -73,7 +73,7 @@ TEST_F(VadTest, vad_filterbank) {
 
   // Verify that all ones in gives kOffsetVector out. Any other constant input
   // will have a small impact in the sub bands.
-  for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+  for (size_t i = 0; i < kMaxFrameLength; ++i) {
     speech[i] = 1;
   }
   for (size_t j = 0; j < kFrameLengthsSize; ++j) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.c b/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.c
index 217ef26566..a54be17daa 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.c
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.c
@@ -27,12 +27,13 @@ static const int16_t kSmoothingUp = 32439;  // 0.99 in Q15.
 void WebRtcVad_Downsampling(const int16_t* signal_in,
                             int16_t* signal_out,
                             int32_t* filter_state,
-                            int in_length) {
+                            size_t in_length) {
   int16_t tmp16_1 = 0, tmp16_2 = 0;
   int32_t tmp32_1 = filter_state[0];
   int32_t tmp32_2 = filter_state[1];
-  int n = 0;
-  int half_length = (in_length >> 1);  // Downsampling by 2 gives half length.
+  size_t n = 0;
+  // Downsampling by 2 gives half length.
+  size_t half_length = (in_length >> 1);
 
   // Filter coefficients in Q13, filter state in Q0.
   for (n = 0; n < half_length; n++) {
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.h b/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.h
index b5e62593c0..4d2b02a1ef 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp.h
@@ -33,7 +33,7 @@
 void WebRtcVad_Downsampling(const int16_t* signal_in,
                             int16_t* signal_out,
                             int32_t* filter_state,
-                            int in_length);
+                            size_t in_length);
 
 // Updates and returns the smoothed feature minimum. As minimum we use the
 // median of the five smallest feature values in a 100 frames long window.
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp_unittest.cc
index d893138ad3..6d5e2a646b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_sp_unittest.cc
@@ -23,7 +23,7 @@ namespace {
 
 TEST_F(VadTest, vad_sp) {
   VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
-  const int kMaxFrameLenSp = 960;  // Maximum frame length in this unittest.
+  const size_t kMaxFrameLenSp = 960;  // Maximum frame length in this unittest.
   int16_t zeros[kMaxFrameLenSp] = { 0 };
   int32_t state[2] = { 0 };
   int16_t data_in[kMaxFrameLenSp];
@@ -40,14 +40,14 @@ TEST_F(VadTest, vad_sp) {
 
   // Construct a speech signal that will trigger the VAD in all modes. It is
   // known that (i * i) will wrap around, but that doesn't matter in this case.
-  for (int16_t i = 0; i < kMaxFrameLenSp; ++i) {
-    data_in[i] = (i * i);
+  for (size_t i = 0; i < kMaxFrameLenSp; ++i) {
+    data_in[i] = static_cast<int16_t>(i * i);
   }
   // Input values all zeros, expect all zeros out.
   WebRtcVad_Downsampling(zeros, data_out, state, kMaxFrameLenSp);
   EXPECT_EQ(0, state[0]);
   EXPECT_EQ(0, state[1]);
-  for (int16_t i = 0; i < kMaxFrameLenSp / 2; ++i) {
+  for (size_t i = 0; i < kMaxFrameLenSp / 2; ++i) {
     EXPECT_EQ(0, data_out[i]);
   }
   // Make a simple non-zero data test.
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.cc
index a1127ad244..a0e16b1ce5 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.cc
@@ -14,6 +14,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/typedefs.h"
@@ -25,7 +27,7 @@ void VadTest::SetUp() {}
 void VadTest::TearDown() {}
 
 // Returns true if the rate and frame length combination is valid.
-bool VadTest::ValidRatesAndFrameLengths(int rate, int frame_length) {
+bool VadTest::ValidRatesAndFrameLengths(int rate, size_t frame_length) {
   if (rate == 8000) {
     if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
       return true;
@@ -57,24 +59,24 @@ TEST_F(VadTest, ApiTest) {
   // This API test runs through the APIs for all possible valid and invalid
   // combinations.
 
-  VadInst* handle = NULL;
+  VadInst* handle = WebRtcVad_Create();
   int16_t zeros[kMaxFrameLength] = { 0 };
 
   // Construct a speech signal that will trigger the VAD in all modes. It is
   // known that (i * i) will wrap around, but that doesn't matter in this case.
   int16_t speech[kMaxFrameLength];
-  for (int16_t i = 0; i < kMaxFrameLength; i++) {
-    speech[i] = (i * i);
+  for (size_t i = 0; i < kMaxFrameLength; i++) {
+    speech[i] = static_cast<int16_t>(i * i);
   }
 
-  // NULL instance tests
-  EXPECT_EQ(-1, WebRtcVad_Create(NULL));
-  EXPECT_EQ(-1, WebRtcVad_Init(NULL));
-  EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, kModes[0]));
-  EXPECT_EQ(-1, WebRtcVad_Process(NULL, kRates[0], speech, kFrameLengths[0]));
+  // nullptr instance tests
+  EXPECT_EQ(-1, WebRtcVad_Init(nullptr));
+  EXPECT_EQ(-1, WebRtcVad_set_mode(nullptr, kModes[0]));
+  EXPECT_EQ(-1,
+            WebRtcVad_Process(nullptr, kRates[0], speech, kFrameLengths[0]));
 
   // WebRtcVad_Create()
-  ASSERT_EQ(0, WebRtcVad_Create(&handle));
+  RTC_CHECK(handle);
 
   // Not initialized tests
   EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], speech, kFrameLengths[0]));
@@ -93,8 +95,9 @@ TEST_F(VadTest, ApiTest) {
                                                          kModesSize) + 1));
 
   // WebRtcVad_Process() tests
-  // NULL speech pointer
-  EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], NULL, kFrameLengths[0]));
+  // nullptr as speech pointer
+  EXPECT_EQ(-1,
+            WebRtcVad_Process(handle, kRates[0], nullptr, kFrameLengths[0]));
   // Invalid sampling rate
   EXPECT_EQ(-1, WebRtcVad_Process(handle, 9999, speech, kFrameLengths[0]));
   // All zeros as input should work
@@ -127,18 +130,16 @@ TEST_F(VadTest, ValidRatesFrameLengths) {
   // This test verifies valid and invalid rate/frame_length combinations. We
   // loop through some sampling rates and frame lengths from negative values to
   // values larger than possible.
-  const int kNumRates = 12;
-  const int kRates[kNumRates] = {
+  const int kRates[] = {
     -8000, -4000, 0, 4000, 8000, 8001, 15999, 16000, 32000, 48000, 48001, 96000
   };
 
-  const int kNumFrameLengths = 13;
-  const int kFrameLengths[kNumFrameLengths] = {
-    -10, 0, 80, 81, 159, 160, 240, 320, 480, 640, 960, 1440, 2000
+  const size_t kFrameLengths[] = {
+    0, 80, 81, 159, 160, 240, 320, 480, 640, 960, 1440, 2000
   };
 
-  for (int i = 0; i < kNumRates; i++) {
-    for (int j = 0; j < kNumFrameLengths; j++) {
+  for (size_t i = 0; i < arraysize(kRates); i++) {
+    for (size_t j = 0; j < arraysize(kFrameLengths); j++) {
       if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
         EXPECT_EQ(0, WebRtcVad_ValidRateAndFrameLength(kRates[i],
                                                        kFrameLengths[j]));
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.h b/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.h
index 5fb726d47f..3efe61b632 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.h
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/vad_unittest.h
@@ -28,8 +28,8 @@ const int kRates[] = { 8000, 12000, 16000, 24000, 32000, 48000 };
 const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
 
 // Frame lengths we support.
-const int kMaxFrameLength = 1440;
-const int kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640, 960,
+const size_t kMaxFrameLength = 1440;
+const size_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640, 960,
     kMaxFrameLength };
 const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
 
@@ -42,7 +42,7 @@ class VadTest : public ::testing::Test {
   virtual void TearDown();
 
   // Returns true if the rate and frame length combination is valid.
-  bool ValidRatesAndFrameLengths(int rate, int frame_length);
+  bool ValidRatesAndFrameLengths(int rate, size_t frame_length);
 };
 
 #endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_UNITTEST_H
diff --git a/media/webrtc/trunk/webrtc/common_audio/vad/webrtc_vad.c b/media/webrtc/trunk/webrtc/common_audio/vad/webrtc_vad.c
index 8a9b9317d8..80c8f3c88d 100644
--- a/media/webrtc/trunk/webrtc/common_audio/vad/webrtc_vad.c
+++ b/media/webrtc/trunk/webrtc/common_audio/vad/webrtc_vad.c
@@ -22,26 +22,13 @@ static const int kValidRates[] = { 8000, 16000, 32000, 48000 };
 static const size_t kRatesSize = sizeof(kValidRates) / sizeof(*kValidRates);
 static const int kMaxFrameLengthMs = 30;
 
-int WebRtcVad_Create(VadInst** handle) {
-  VadInstT* self = NULL;
-
-  if (handle == NULL) {
-    return -1;
-  }
-
-  *handle = NULL;
-  self = (VadInstT*) malloc(sizeof(VadInstT));
-  *handle = (VadInst*) self;
-
-  if (self == NULL) {
-    return -1;
-  }
+VadInst* WebRtcVad_Create() {
+  VadInstT* self = (VadInstT*)malloc(sizeof(VadInstT));
 
   WebRtcSpl_Init();
-
   self->init_flag = 0;
 
-  return 0;
+  return (VadInst*)self;
 }
 
 void WebRtcVad_Free(VadInst* handle) {
@@ -69,7 +56,7 @@ int WebRtcVad_set_mode(VadInst* handle, int mode) {
 }
 
 int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
-                      int frame_length) {
+                      size_t frame_length) {
   int vad = -1;
   VadInstT* self = (VadInstT*) handle;
 
@@ -103,11 +90,11 @@ int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
   return vad;
 }
 
-int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length) {
+int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length) {
   int return_value = -1;
   size_t i;
   int valid_length_ms;
-  int valid_length;
+  size_t valid_length;
 
   // We only allow 10, 20 or 30 ms frames. Loop through valid frame rates and
   // see if we have a matching pair.
@@ -115,7 +102,7 @@ int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length) {
     if (kValidRates[i] == rate) {
       for (valid_length_ms = 10; valid_length_ms <= kMaxFrameLengthMs;
           valid_length_ms += 10) {
-        valid_length = (kValidRates[i] / 1000 * valid_length_ms);
+        valid_length = (size_t)(kValidRates[i] / 1000 * valid_length_ms);
         if (frame_length == valid_length) {
           return_value = 0;
           break;
diff --git a/media/webrtc/trunk/webrtc/common_audio/wav_file.cc b/media/webrtc/trunk/webrtc/common_audio/wav_file.cc
index 74329762ff..41a0dffa90 100644
--- a/media/webrtc/trunk/webrtc/common_audio/wav_file.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/wav_file.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 #include <cstdio>
 #include <limits>
+#include <sstream>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
@@ -23,7 +24,7 @@ namespace webrtc {
 
 // We write 16-bit PCM WAV files.
 static const WavFormat kWavFormat = kWavFormatPcm;
-static const int kBytesPerSample = 2;
+static const size_t kBytesPerSample = 2;
 
 // Doesn't take ownership of the file handle and won't close it.
 class ReadableWavFile : public ReadableWav {
@@ -37,18 +38,26 @@ class ReadableWavFile : public ReadableWav {
   FILE* file_;
 };
 
+std::string WavFile::FormatAsString() const {
+  std::ostringstream s;
+  s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
+    << ", Duration: "
+    << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
+  return s.str();
+}
+
 WavReader::WavReader(const std::string& filename)
     : file_handle_(fopen(filename.c_str(), "rb")) {
-  CHECK(file_handle_ && "Could not open wav file for reading.");
+  RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
 
   ReadableWavFile readable(file_handle_);
   WavFormat format;
-  int bytes_per_sample;
-  CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
-                      &bytes_per_sample, &num_samples_));
+  size_t bytes_per_sample;
+  RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
+                          &bytes_per_sample, &num_samples_));
   num_samples_remaining_ = num_samples_;
-  CHECK_EQ(kWavFormat, format);
-  CHECK_EQ(kBytesPerSample, bytes_per_sample);
+  RTC_CHECK_EQ(kWavFormat, format);
+  RTC_CHECK_EQ(kBytesPerSample, bytes_per_sample);
 }
 
 WavReader::~WavReader() {
@@ -57,14 +66,13 @@ WavReader::~WavReader() {
 
 size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
   // There could be metadata after the audio; ensure we don't read it.
-  num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
-                         num_samples_remaining_);
+  num_samples = std::min(num_samples, num_samples_remaining_);
   const size_t read =
       fread(samples, sizeof(*samples), num_samples, file_handle_);
   // If we didn't read what was requested, ensure we've reached the EOF.
-  CHECK(read == num_samples || feof(file_handle_));
-  CHECK_LE(read, num_samples_remaining_);
-  num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
+  RTC_CHECK(read == num_samples || feof(file_handle_));
+  RTC_CHECK_LE(read, num_samples_remaining_);
+  num_samples_remaining_ -= read;
 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
   //convert to big-endian
   for(size_t idx = 0; idx < num_samples; idx++) {
@@ -89,28 +97,27 @@ size_t WavReader::ReadSamples(size_t num_samples, float* samples) {
 }
 
 void WavReader::Close() {
-  CHECK_EQ(0, fclose(file_handle_));
+  RTC_CHECK_EQ(0, fclose(file_handle_));
   file_handle_ = NULL;
 }
 
 WavWriter::WavWriter(const std::string& filename, int sample_rate,
-                     int num_channels)
+                     size_t num_channels)
     : sample_rate_(sample_rate),
       num_channels_(num_channels),
       num_samples_(0),
       file_handle_(fopen(filename.c_str(), "wb")) {
   if (file_handle_) {
-    CHECK(file_handle_ && "Could not open wav file for writing.");
-    CHECK(CheckWavParameters(num_channels_,
+    RTC_CHECK(file_handle_ && "Could not open wav file for writing.");
+    RTC_CHECK(CheckWavParameters(num_channels_,
                              sample_rate_,
                              kWavFormat,
                              kBytesPerSample,
                              num_samples_));
-
     // Write a blank placeholder header, since we need to know the total number
     // of samples before we can fill in the real data.
     static const uint8_t blank_header[kWavHeaderSize] = {0};
-    CHECK_EQ(1u, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
+    RTC_CHECK_EQ(1u, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
   }
 }
 
@@ -134,15 +141,9 @@ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
   const size_t written =
       fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 #endif
-  CHECK_EQ(num_samples, written);
-  num_samples_ += static_cast<uint32_t>(written);
-  CHECK(written <= std::numeric_limits<uint32_t>::max() ||
-        num_samples_ >= written);  // detect uint32_t overflow
-  CHECK(CheckWavParameters(num_channels_,
-                           sample_rate_,
-                           kWavFormat,
-                           kBytesPerSample,
-                           num_samples_));
+  RTC_CHECK_EQ(num_samples, written);
+  num_samples_ += written;
+  RTC_CHECK(num_samples_ >= written);  // detect size_t overflow
 }
 
 void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
@@ -159,12 +160,12 @@ void WavWriter::Close() {
   if (!file_handle_) {
     return;
   }
-  CHECK_EQ(0, fseek(file_handle_, 0, SEEK_SET));
+  RTC_CHECK_EQ(0, fseek(file_handle_, 0, SEEK_SET));
   uint8_t header[kWavHeaderSize];
   WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat,
                  kBytesPerSample, num_samples_);
-  CHECK_EQ(1u, fwrite(header, kWavHeaderSize, 1, file_handle_));
-  CHECK_EQ(0, fclose(file_handle_));
+  RTC_CHECK_EQ(1u, fwrite(header, kWavHeaderSize, 1, file_handle_));
+  RTC_CHECK_EQ(0, fclose(file_handle_));
   file_handle_ = NULL;
 }
 
@@ -172,7 +173,7 @@ void WavWriter::Close() {
 
 rtc_WavWriter* rtc_WavOpen(const char* filename,
                            int sample_rate,
-                           int num_channels) {
+                           size_t num_channels) {
   return reinterpret_cast<rtc_WavWriter*>(
       new webrtc::WavWriter(filename, sample_rate, num_channels));
 }
@@ -191,10 +192,10 @@ int rtc_WavSampleRate(const rtc_WavWriter* wf) {
   return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
 }
 
-int rtc_WavNumChannels(const rtc_WavWriter* wf) {
+size_t rtc_WavNumChannels(const rtc_WavWriter* wf) {
   return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
 }
 
-uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
+size_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
   return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
 }
diff --git a/media/webrtc/trunk/webrtc/common_audio/wav_file.h b/media/webrtc/trunk/webrtc/common_audio/wav_file.h
index 1fbf9541fa..e656eb8643 100644
--- a/media/webrtc/trunk/webrtc/common_audio/wav_file.h
+++ b/media/webrtc/trunk/webrtc/common_audio/wav_file.h
@@ -17,14 +17,29 @@
 #include <cstddef>
 #include <string>
 
+#include "webrtc/base/constructormagic.h"
+
 namespace webrtc {
 
+// Interface to provide access to WAV file parameters.
+class WavFile {
+ public:
+  virtual ~WavFile() {}
+
+  virtual int sample_rate() const = 0;
+  virtual size_t num_channels() const = 0;
+  virtual size_t num_samples() const = 0;
+
+  // Returns a human-readable string containing the audio format.
+  std::string FormatAsString() const;
+};
+
 // Simple C++ class for writing 16-bit PCM WAV files. All error handling is
-// by calls to CHECK(), making it unsuitable for anything but debug code.
-class WavWriter {
+// by calls to RTC_CHECK(), making it unsuitable for anything but debug code.
+class WavWriter final : public WavFile {
  public:
   // Open a new WAV file for writing.
-  WavWriter(const std::string& filename, int sample_rate, int num_channels);
+  WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
 
   // Close the WAV file, after writing its header.
   ~WavWriter();
@@ -35,20 +50,22 @@ class WavWriter {
   void WriteSamples(const float* samples, size_t num_samples);
   void WriteSamples(const int16_t* samples, size_t num_samples);
 
-  int sample_rate() const { return sample_rate_; }
-  int num_channels() const { return num_channels_; }
-  uint32_t num_samples() const { return num_samples_; }
+  int sample_rate() const override { return sample_rate_; }
+  size_t num_channels() const override { return num_channels_; }
+  size_t num_samples() const override { return num_samples_; }
 
  private:
   void Close();
   const int sample_rate_;
-  const int num_channels_;
-  uint32_t num_samples_;  // Total number of samples written to file.
+  const size_t num_channels_;
+  size_t num_samples_;  // Total number of samples written to file.
   FILE* file_handle_;  // Output file, owned by this class
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(WavWriter);
 };
 
 // Follows the conventions of WavWriter.
-class WavReader {
+class WavReader final : public WavFile {
  public:
   // Opens an existing WAV file for reading.
   explicit WavReader(const std::string& filename);
@@ -61,17 +78,19 @@ class WavReader {
   size_t ReadSamples(size_t num_samples, float* samples);
   size_t ReadSamples(size_t num_samples, int16_t* samples);
 
-  int sample_rate() const { return sample_rate_; }
-  int num_channels() const { return num_channels_; }
-  uint32_t num_samples() const { return num_samples_; }
+  int sample_rate() const override { return sample_rate_; }
+  size_t num_channels() const override { return num_channels_; }
+  size_t num_samples() const override { return num_samples_; }
 
  private:
   void Close();
   int sample_rate_;
-  int num_channels_;
-  uint32_t num_samples_;  // Total number of samples in the file.
-  uint32_t num_samples_remaining_;
+  size_t num_channels_;
+  size_t num_samples_;  // Total number of samples in the file.
+  size_t num_samples_remaining_;
   FILE* file_handle_;  // Input file, owned by this class.
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(WavReader);
 };
 
 }  // namespace webrtc
@@ -83,14 +102,14 @@ extern "C" {
 typedef struct rtc_WavWriter rtc_WavWriter;
 rtc_WavWriter* rtc_WavOpen(const char* filename,
                            int sample_rate,
-                           int num_channels);
+                           size_t num_channels);
 void rtc_WavClose(rtc_WavWriter* wf);
 void rtc_WavWriteSamples(rtc_WavWriter* wf,
                          const float* samples,
                          size_t num_samples);
 int rtc_WavSampleRate(const rtc_WavWriter* wf);
-int rtc_WavNumChannels(const rtc_WavWriter* wf);
-uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf);
+size_t rtc_WavNumChannels(const rtc_WavWriter* wf);
+size_t rtc_WavNumSamples(const rtc_WavWriter* wf);
 
 #ifdef __cplusplus
 }  // extern "C"
diff --git a/media/webrtc/trunk/webrtc/common_audio/wav_file_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/wav_file_unittest.cc
index 78b0a34de9..ba1db1c296 100644
--- a/media/webrtc/trunk/webrtc/common_audio/wav_file_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/wav_file_unittest.cc
@@ -26,11 +26,11 @@ static const float kSamples[] = {0.0, 10.0, 4e4, -1e9};
 // Write a tiny WAV file with the C++ interface and verify the result.
 TEST(WavWriterTest, CPP) {
   const std::string outfile = test::OutputPath() + "wavtest1.wav";
-  static const uint32_t kNumSamples = 3;
+  static const size_t kNumSamples = 3;
   {
     WavWriter w(outfile, 14099, 1);
     EXPECT_EQ(14099, w.sample_rate());
-    EXPECT_EQ(1, w.num_channels());
+    EXPECT_EQ(1u, w.num_channels());
     EXPECT_EQ(0u, w.num_samples());
     w.WriteSamples(kSamples, kNumSamples);
     EXPECT_EQ(kNumSamples, w.num_samples());
@@ -64,10 +64,10 @@ TEST(WavWriterTest, CPP) {
     0xff, 0x7f,  // third sample: 4e4 (saturated)
     kMetadata[0], kMetadata[1],
   };
-  static const int kContentSize =
+  static const size_t kContentSize =
       kWavHeaderSize + kNumSamples * sizeof(int16_t) + sizeof(kMetadata);
   static_assert(sizeof(kExpectedContents) == kContentSize, "content size");
-  EXPECT_EQ(size_t(kContentSize), test::GetFileSize(outfile));
+  EXPECT_EQ(kContentSize, test::GetFileSize(outfile));
   FILE* f = fopen(outfile.c_str(), "rb");
   ASSERT_TRUE(f);
   uint8_t contents[kContentSize];
@@ -78,7 +78,7 @@ TEST(WavWriterTest, CPP) {
   {
     WavReader r(outfile);
     EXPECT_EQ(14099, r.sample_rate());
-    EXPECT_EQ(1, r.num_channels());
+    EXPECT_EQ(1u, r.num_channels());
     EXPECT_EQ(kNumSamples, r.num_samples());
     static const float kTruncatedSamples[] = {0.0, 10.0, 32767.0};
     float samples[kNumSamples];
@@ -93,9 +93,9 @@ TEST(WavWriterTest, C) {
   const std::string outfile = test::OutputPath() + "wavtest2.wav";
   rtc_WavWriter* w = rtc_WavOpen(outfile.c_str(), 11904, 2);
   EXPECT_EQ(11904, rtc_WavSampleRate(w));
-  EXPECT_EQ(2, rtc_WavNumChannels(w));
+  EXPECT_EQ(2u, rtc_WavNumChannels(w));
   EXPECT_EQ(0u, rtc_WavNumSamples(w));
-  static const uint32_t kNumSamples = 4;
+  static const size_t kNumSamples = 4;
   rtc_WavWriteSamples(w, &kSamples[0], 2);
   EXPECT_EQ(2u, rtc_WavNumSamples(w));
   rtc_WavWriteSamples(w, &kSamples[2], kNumSamples - 2);
@@ -120,10 +120,10 @@ TEST(WavWriterTest, C) {
     0xff, 0x7f,  // third sample: 4e4 (saturated)
     0, 0x80,  // fourth sample: -1e9 (saturated)
   };
-  static const int kContentSize =
+  static const size_t kContentSize =
       kWavHeaderSize + kNumSamples * sizeof(int16_t);
   static_assert(sizeof(kExpectedContents) == kContentSize, "content size");
-  EXPECT_EQ(size_t(kContentSize), test::GetFileSize(outfile));
+  EXPECT_EQ(kContentSize, test::GetFileSize(outfile));
   FILE* f = fopen(outfile.c_str(), "rb");
   ASSERT_TRUE(f);
   uint8_t contents[kContentSize];
@@ -136,10 +136,10 @@ TEST(WavWriterTest, C) {
 TEST(WavWriterTest, LargeFile) {
   std::string outfile = test::OutputPath() + "wavtest3.wav";
   static const int kSampleRate = 8000;
-  static const int kNumChannels = 2;
-  static const uint32_t kNumSamples = 3 * kSampleRate * kNumChannels;
+  static const size_t kNumChannels = 2;
+  static const size_t kNumSamples = 3 * kSampleRate * kNumChannels;
   float samples[kNumSamples];
-  for (uint32_t i = 0; i < kNumSamples; i += kNumChannels) {
+  for (size_t i = 0; i < kNumSamples; i += kNumChannels) {
     // A nice periodic beeping sound.
     static const double kToneHz = 440;
     const double t = static_cast<double>(i) / (kNumChannels * kSampleRate);
diff --git a/media/webrtc/trunk/webrtc/common_audio/wav_header.cc b/media/webrtc/trunk/webrtc/common_audio/wav_header.cc
index 1388dfacfd..b11ee58c6b 100644
--- a/media/webrtc/trunk/webrtc/common_audio/wav_header.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/wav_header.cc
@@ -59,20 +59,19 @@ static_assert(sizeof(WavHeader) == kWavHeaderSize, "no padding in header");
 
 }  // namespace
 
-bool CheckWavParameters(int num_channels,
+bool CheckWavParameters(size_t num_channels,
                         int sample_rate,
                         WavFormat format,
-                        int bytes_per_sample,
-                        uint32_t num_samples) {
+                        size_t bytes_per_sample,
+                        size_t num_samples) {
   // num_channels, sample_rate, and bytes_per_sample must be positive, must fit
   // in their respective fields, and their product must fit in the 32-bit
   // ByteRate field.
-  if (num_channels <= 0 || sample_rate <= 0 || bytes_per_sample <= 0)
+  if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0)
     return false;
   if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max())
     return false;
-  if (static_cast<uint64_t>(num_channels) >
-      std::numeric_limits<uint16_t>::max())
+  if (num_channels > std::numeric_limits<uint16_t>::max())
     return false;
   if (static_cast<uint64_t>(bytes_per_sample) * 8 >
       std::numeric_limits<uint16_t>::max())
@@ -99,10 +98,9 @@ bool CheckWavParameters(int num_channels,
 
   // The number of bytes in the file, not counting the first ChunkHeader, must
   // be less than 2^32; otherwise, the ChunkSize field overflows.
-  const uint32_t max_samples =
-      (std::numeric_limits<uint32_t>::max()
-       - (kWavHeaderSize - sizeof(ChunkHeader))) /
-      bytes_per_sample;
+  const size_t header_size = kWavHeaderSize - sizeof(ChunkHeader);
+  const size_t max_samples =
+      (std::numeric_limits<uint32_t>::max() - header_size) / bytes_per_sample;
   if (num_samples > max_samples)
     return false;
 
@@ -164,30 +162,32 @@ static inline std::string ReadFourCC(uint32_t x) {
 }
 #endif
 
-static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
-  return bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader);
+static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
+  return static_cast<uint32_t>(
+      bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader));
 }
 
-static inline uint32_t ByteRate(int num_channels, int sample_rate,
-                                int bytes_per_sample) {
-  return static_cast<uint32_t>(num_channels) * sample_rate * bytes_per_sample;
+static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
+                                size_t bytes_per_sample) {
+  return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
 }
 
-static inline uint16_t BlockAlign(int num_channels, int bytes_per_sample) {
-  return num_channels * bytes_per_sample;
+static inline uint16_t BlockAlign(size_t num_channels,
+                                  size_t bytes_per_sample) {
+  return static_cast<uint16_t>(num_channels * bytes_per_sample);
 }
 
 void WriteWavHeader(uint8_t* buf,
-                    int num_channels,
+                    size_t num_channels,
                     int sample_rate,
                     WavFormat format,
-                    int bytes_per_sample,
-                    uint32_t num_samples) {
-  CHECK(CheckWavParameters(num_channels, sample_rate, format,
-                           bytes_per_sample, num_samples));
+                    size_t bytes_per_sample,
+                    size_t num_samples) {
+  RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format,
+                               bytes_per_sample, num_samples));
 
   WavHeader header;
-  const uint32_t bytes_in_payload = bytes_per_sample * num_samples;
+  const size_t bytes_in_payload = bytes_per_sample * num_samples;
 
   WriteFourCC(&header.riff.header.ID, 'R', 'I', 'F', 'F');
   WriteLE32(&header.riff.header.Size, RiffChunkSize(bytes_in_payload));
@@ -196,15 +196,16 @@ void WriteWavHeader(uint8_t* buf,
   WriteFourCC(&header.fmt.header.ID, 'f', 'm', 't', ' ');
   WriteLE32(&header.fmt.header.Size, kFmtSubchunkSize);
   WriteLE16(&header.fmt.AudioFormat, format);
-  WriteLE16(&header.fmt.NumChannels, num_channels);
+  WriteLE16(&header.fmt.NumChannels, static_cast<uint16_t>(num_channels));
   WriteLE32(&header.fmt.SampleRate, sample_rate);
   WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate,
                                            bytes_per_sample));
   WriteLE16(&header.fmt.BlockAlign, BlockAlign(num_channels, bytes_per_sample));
-  WriteLE16(&header.fmt.BitsPerSample, 8 * bytes_per_sample);
+  WriteLE16(&header.fmt.BitsPerSample,
+            static_cast<uint16_t>(8 * bytes_per_sample));
 
   WriteFourCC(&header.data.header.ID, 'd', 'a', 't', 'a');
-  WriteLE32(&header.data.header.Size, bytes_in_payload);
+  WriteLE32(&header.data.header.Size, static_cast<uint32_t>(bytes_in_payload));
 
   // Do an extra copy rather than writing everything to buf directly, since buf
   // might not be correctly aligned.
@@ -212,11 +213,11 @@ void WriteWavHeader(uint8_t* buf,
 }
 
 bool ReadWavHeader(ReadableWav* readable,
-                   int* num_channels,
+                   size_t* num_channels,
                    int* sample_rate,
                    WavFormat* format,
-                   int* bytes_per_sample,
-                   uint32_t* num_samples) {
+                   size_t* bytes_per_sample,
+                   size_t* num_samples) {
   WavHeader header;
   if (readable->Read(&header, kWavHeaderSize - sizeof(header.data)) !=
       kWavHeaderSize - sizeof(header.data))
@@ -242,8 +243,8 @@ bool ReadWavHeader(ReadableWav* readable,
   *num_channels = ReadLE16(header.fmt.NumChannels);
   *sample_rate = ReadLE32(header.fmt.SampleRate);
   *bytes_per_sample = ReadLE16(header.fmt.BitsPerSample) / 8;
-  const uint32_t bytes_in_payload = ReadLE32(header.data.header.Size);
-  if (*bytes_per_sample <= 0)
+  const size_t bytes_in_payload = ReadLE32(header.data.header.Size);
+  if (*bytes_per_sample == 0)
     return false;
   *num_samples = bytes_in_payload / *bytes_per_sample;
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/wav_header.h b/media/webrtc/trunk/webrtc/common_audio/wav_header.h
index 1a0fd7c81d..6844306941 100644
--- a/media/webrtc/trunk/webrtc/common_audio/wav_header.h
+++ b/media/webrtc/trunk/webrtc/common_audio/wav_header.h
@@ -32,32 +32,32 @@ enum WavFormat {
 };
 
 // Return true if the given parameters will make a well-formed WAV header.
-bool CheckWavParameters(int num_channels,
+bool CheckWavParameters(size_t num_channels,
                         int sample_rate,
                         WavFormat format,
-                        int bytes_per_sample,
-                        uint32_t num_samples);
+                        size_t bytes_per_sample,
+                        size_t num_samples);
 
 // Write a kWavHeaderSize bytes long WAV header to buf. The payload that
 // follows the header is supposed to have the specified number of interleaved
 // channels and contain the specified total number of samples of the specified
 // type. CHECKs the input parameters for validity.
 void WriteWavHeader(uint8_t* buf,
-                    int num_channels,
+                    size_t num_channels,
                     int sample_rate,
                     WavFormat format,
-                    int bytes_per_sample,
-                    uint32_t num_samples);
+                    size_t bytes_per_sample,
+                    size_t num_samples);
 
 // Read a WAV header from an implemented ReadableWav and parse the values into
 // the provided output parameters. ReadableWav is used because the header can
 // be variably sized. Returns false if the header is invalid.
 bool ReadWavHeader(ReadableWav* readable,
-                   int* num_channels,
+                   size_t* num_channels,
                    int* sample_rate,
                    WavFormat* format,
-                   int* bytes_per_sample,
-                   uint32_t* num_samples);
+                   size_t* bytes_per_sample,
+                   size_t* num_samples);
 
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/wav_header_unittest.cc b/media/webrtc/trunk/webrtc/common_audio/wav_header_unittest.cc
index e03cb303aa..8527939eac 100644
--- a/media/webrtc/trunk/webrtc/common_audio/wav_header_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/wav_header_unittest.cc
@@ -70,7 +70,7 @@ TEST(WavHeaderTest, CheckWavParameters) {
   // Try some really stupid values for one parameter at a time.
   EXPECT_TRUE(CheckWavParameters(1, 8000, kWavFormatPcm, 1, 0));
   EXPECT_FALSE(CheckWavParameters(0, 8000, kWavFormatPcm, 1, 0));
-  EXPECT_FALSE(CheckWavParameters(-1, 8000, kWavFormatPcm, 1, 0));
+  EXPECT_FALSE(CheckWavParameters(0x10000, 8000, kWavFormatPcm, 1, 0));
   EXPECT_FALSE(CheckWavParameters(1, 0, kWavFormatPcm, 1, 0));
   EXPECT_FALSE(CheckWavParameters(1, 8000, WavFormat(0), 1, 0));
   EXPECT_FALSE(CheckWavParameters(1, 8000, kWavFormatPcm, 0, 0));
@@ -91,11 +91,11 @@ TEST(WavHeaderTest, CheckWavParameters) {
 }
 
 TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
-  int num_channels = 0;
+  size_t num_channels = 0;
   int sample_rate = 0;
   WavFormat format = kWavFormatPcm;
-  int bytes_per_sample = 0;
-  uint32_t num_samples = 0;
+  size_t bytes_per_sample = 0;
+  size_t num_samples = 0;
 
   // Test a few ways the header can be invalid. We start with the valid header
   // used in WriteAndReadWavHeader, and invalidate one field per test. The
@@ -268,19 +268,19 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
   static_assert(sizeof(kExpectedBuf) == kSize, "buffer size");
   EXPECT_EQ(0, memcmp(kExpectedBuf, buf, kSize));
 
-  int num_channels = 0;
+  size_t num_channels = 0;
   int sample_rate = 0;
   WavFormat format = kWavFormatPcm;
-  int bytes_per_sample = 0;
-  uint32_t num_samples = 0;
+  size_t bytes_per_sample = 0;
+  size_t num_samples = 0;
   ReadableWavBuffer r(buf + 4, sizeof(buf) - 8);
   EXPECT_TRUE(
       ReadWavHeader(&r, &num_channels, &sample_rate, &format,
                     &bytes_per_sample, &num_samples));
-  EXPECT_EQ(17, num_channels);
+  EXPECT_EQ(17u, num_channels);
   EXPECT_EQ(12345, sample_rate);
   EXPECT_EQ(kWavFormatALaw, format);
-  EXPECT_EQ(1, bytes_per_sample);
+  EXPECT_EQ(1u, bytes_per_sample);
   EXPECT_EQ(123457689u, num_samples);
 }
 
@@ -304,19 +304,19 @@ TEST(WavHeaderTest, ReadAtypicalWavHeader) {
     0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
   };
 
-  int num_channels = 0;
+  size_t num_channels = 0;
   int sample_rate = 0;
   WavFormat format = kWavFormatPcm;
-  int bytes_per_sample = 0;
-  uint32_t num_samples = 0;
+  size_t bytes_per_sample = 0;
+  size_t num_samples = 0;
   ReadableWavBuffer r(kBuf, sizeof(kBuf));
   EXPECT_TRUE(
       ReadWavHeader(&r, &num_channels, &sample_rate, &format,
                     &bytes_per_sample, &num_samples));
-  EXPECT_EQ(17, num_channels);
+  EXPECT_EQ(17u, num_channels);
   EXPECT_EQ(12345, sample_rate);
   EXPECT_EQ(kWavFormatALaw, format);
-  EXPECT_EQ(1, bytes_per_sample);
+  EXPECT_EQ(1u, bytes_per_sample);
   EXPECT_EQ(123457689u, num_samples);
 }
 
diff --git a/media/webrtc/trunk/webrtc/common_audio/window_generator.cc b/media/webrtc/trunk/webrtc/common_audio/window_generator.cc
index 1d61368c19..3da9a05786 100644
--- a/media/webrtc/trunk/webrtc/common_audio/window_generator.cc
+++ b/media/webrtc/trunk/webrtc/common_audio/window_generator.cc
@@ -8,7 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef _USE_MATH_DEFINES
 #define _USE_MATH_DEFINES
+#endif
 
 #include "webrtc/common_audio/window_generator.h"
 
@@ -38,28 +40,28 @@ complex<float> I0(complex<float> x) {
 namespace webrtc {
 
 void WindowGenerator::Hanning(int length, float* window) {
-  CHECK_GT(length, 1);
-  CHECK(window != nullptr);
+  RTC_CHECK_GT(length, 1);
+  RTC_CHECK(window != nullptr);
   for (int i = 0; i < length; ++i) {
     window[i] = 0.5f * (1 - cosf(2 * static_cast<float>(M_PI) * i /
                                  (length - 1)));
   }
 }
 
-void WindowGenerator::KaiserBesselDerived(float alpha, int length,
+void WindowGenerator::KaiserBesselDerived(float alpha, size_t length,
                                           float* window) {
-  CHECK_GT(length, 1);
-  CHECK(window != nullptr);
+  RTC_CHECK_GT(length, 1U);
+  RTC_CHECK(window != nullptr);
 
-  const int half = (length + 1) / 2;
+  const size_t half = (length + 1) / 2;
   float sum = 0.0f;
 
-  for (int i = 0; i <= half; ++i) {
+  for (size_t i = 0; i <= half; ++i) {
     complex<float> r = (4.0f * i) / length - 1.0f;
     sum += I0(static_cast<float>(M_PI) * alpha * sqrt(1.0f - r * r)).real();
     window[i] = sum;
   }
-  for (int i = length - 1; i >= half; --i) {
+  for (size_t i = length - 1; i >= half; --i) {
     window[length - i - 1] = sqrtf(window[length - i - 1] / sum);
     window[i] = window[length - i - 1];
   }
diff --git a/media/webrtc/trunk/webrtc/common_audio/window_generator.h b/media/webrtc/trunk/webrtc/common_audio/window_generator.h
index ee0acada52..25dd233b44 100644
--- a/media/webrtc/trunk/webrtc/common_audio/window_generator.h
+++ b/media/webrtc/trunk/webrtc/common_audio/window_generator.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_COMMON_AUDIO_WINDOW_GENERATOR_H_
 #define WEBRTC_COMMON_AUDIO_WINDOW_GENERATOR_H_
 
+#include <stddef.h>
+
 #include "webrtc/base/constructormagic.h"
 
 namespace webrtc {
@@ -19,10 +21,10 @@ namespace webrtc {
 class WindowGenerator {
  public:
   static void Hanning(int length, float* window);
-  static void KaiserBesselDerived(float alpha, int length, float* window);
+  static void KaiserBesselDerived(float alpha, size_t length, float* window);
 
  private:
-  DISALLOW_IMPLICIT_CONSTRUCTORS(WindowGenerator);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WindowGenerator);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_types.cc b/media/webrtc/trunk/webrtc/common_types.cc
index 3ed9a19702..59fb6046f5 100644
--- a/media/webrtc/trunk/webrtc/common_types.cc
+++ b/media/webrtc/trunk/webrtc/common_types.cc
@@ -28,6 +28,7 @@ RTPHeaderExtension::RTPHeaderExtension()
       hasTransportSequenceNumber(false),
       transportSequenceNumber(0),
       hasAudioLevel(false),
+      voiceActivity(false),
       audioLevel(0),
       hasVideoRotation(false),
       videoRotation(0),
diff --git a/media/webrtc/trunk/webrtc/common_types.h b/media/webrtc/trunk/webrtc/common_types.h
index a82b109719..01ae55e61e 100644
--- a/media/webrtc/trunk/webrtc/common_types.h
+++ b/media/webrtc/trunk/webrtc/common_types.h
@@ -156,25 +156,12 @@ enum ProcessingTypes
     kRecordingPreprocessing
 };
 
-enum FrameType
-{
-    kFrameEmpty            = 0,
-    kAudioFrameSpeech      = 1,
-    kAudioFrameCN          = 2,
-    kVideoFrameKey         = 3,    // independent frame
-    kVideoFrameDelta       = 4,    // depends on the previus frame
-};
-
-// External transport callback interface
-class Transport
-{
-public:
-    virtual int SendPacket(int channel, const void *data, size_t len) = 0;
-    virtual int SendRTCPPacket(int channel, const void *data, size_t len) = 0;
-
-protected:
-    virtual ~Transport() {}
-    Transport() {}
+enum FrameType {
+  kEmptyFrame = 0,
+  kAudioFrameSpeech = 1,
+  kAudioFrameCN = 2,
+  kVideoFrameKey = 3,
+  kVideoFrameDelta = 4,
 };
 
 // Statistics for an RTCP channel
@@ -304,7 +291,7 @@ struct CodecInst {
   char plname[RTP_PAYLOAD_NAME_SIZE];
   int plfreq;
   int pacsize;
-  int channels;
+  size_t channels;
   int rate;  // bits/sec unlike {start,min,max}Bitrate elsewhere in this file!
 
   bool operator==(const CodecInst& other) const {
@@ -324,12 +311,6 @@ struct CodecInst {
 // RTP
 enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
 
-enum RTPDirections
-{
-    kRtpIncoming = 0,
-    kRtpOutgoing
-};
-
 enum PayloadFrequencies
 {
     kFreq8000Hz = 8000,
@@ -381,7 +362,7 @@ struct NetworkStatistics           // NETEQ statistics
     // max packet waiting time in the jitter buffer (ms)
     int maxWaitingTimeMs;
     // added samples in off mode due to packet loss
-    int addedSamples;
+    size_t addedSamples;
 };
 
 // Statistics for calls to AudioCodingModule::PlayoutData10Ms().
@@ -571,6 +552,7 @@ enum VideoReceiveState
 enum { kConfigParameterSize = 128};
 enum { kPayloadNameSize = 32};
 enum { kMaxSimulcastStreams = 4};
+enum { kMaxSpatialLayers = 5 };
 enum { kMaxTemporalStreams = 4};
 enum { kRIDSize = 32};
 
@@ -630,7 +612,7 @@ struct VideoCodecVP8 {
   }
 };
 
-// VP9 specific
+// VP9 specific.
 struct VideoCodecVP9 {
   VideoCodecComplexity complexity;
   int                  resilience;
@@ -653,6 +635,7 @@ struct VideoCodecH264 {
   uint8_t        packetizationMode; // 0 or 1
   bool           frameDroppingOn;
   int            keyFrameInterval;
+  double         scaleDownBy;
   // These are NULL/0 if not externally negotiated.
   const uint8_t* spsData;
   size_t         spsLen;
@@ -711,6 +694,13 @@ struct SimulcastStream {
   }
 };
 
+struct SpatialLayer {
+  int scaling_factor_num;
+  int scaling_factor_den;
+  int target_bitrate_bps;
+  // TODO(ivica): Add max_quantizer and min_quantizer?
+};
+
 enum VideoCodecMode {
   kRealtimeVideo,
   kScreensharing
@@ -740,6 +730,7 @@ struct VideoCodec {
   unsigned char       numberOfSimulcastStreams;
   unsigned char       ridId;
   SimulcastStream     simulcastStream[kMaxSimulcastStreams];
+  SpatialLayer spatialLayers[kMaxSpatialLayers];
 
   VideoCodecMode      mode;
 
@@ -786,12 +777,11 @@ struct OverUseDetectorOptions {
         initial_e(),
         initial_process_noise(),
         initial_avg_noise(0.0),
-        initial_var_noise(50),
-        initial_threshold(25.0) {
+        initial_var_noise(50) {
     initial_e[0][0] = 100;
     initial_e[1][1] = 1e-1;
     initial_e[0][1] = initial_e[1][0] = 0;
-    initial_process_noise[0] = 1e-10;
+    initial_process_noise[0] = 1e-13;
     initial_process_noise[1] = 1e-2;
   }
   double initial_slope;
@@ -800,7 +790,6 @@ struct OverUseDetectorOptions {
   double initial_process_noise[2];
   double initial_avg_noise;
   double initial_var_noise;
-  double initial_threshold;
 };
 
 enum CPULoadState {
@@ -852,6 +841,7 @@ struct RTPHeaderExtension {
   // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
   // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
   bool hasAudioLevel;
+  bool voiceActivity;
   uint8_t audioLevel;
 
   // For Coordination of Video Orientation. See
@@ -955,6 +945,11 @@ class StreamDataCountersCallback {
   virtual void DataCountersUpdated(const StreamDataCounters& counters,
                                    uint32_t ssrc) = 0;
 };
+
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+// RTCP mode is described by RFC 5506.
+enum class RtcpMode { kOff, kCompound, kReducedSize };
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/BUILD.gn b/media/webrtc/trunk/webrtc/common_video/BUILD.gn
index 24423f5d33..4ef968d60f 100644
--- a/media/webrtc/trunk/webrtc/common_video/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/common_video/BUILD.gn
@@ -10,7 +10,7 @@ import("../build/webrtc.gni")
 
 config("common_video_config") {
   include_dirs = [
-    "interface",
+    "include",
     "libyuv/include",
   ]
 }
@@ -18,16 +18,18 @@ config("common_video_config") {
 source_set("common_video") {
   sources = [
     "i420_buffer_pool.cc",
-    "i420_video_frame.cc",
-    "interface/i420_video_frame.h",
-    "interface/i420_buffer_pool.h",
-    "interface/native_handle.h",
-    "interface/video_frame_buffer.h",
+    "include/i420_buffer_pool.h",
+    "include/incoming_video_stream.h",
+    "include/video_frame_buffer.h",
+    "incoming_video_stream.cc",
     "libyuv/include/scaler.h",
     "libyuv/include/webrtc_libyuv.h",
     "libyuv/scaler.cc",
     "libyuv/webrtc_libyuv.cc",
+    "video_frame.cc",
     "video_frame_buffer.cc",
+    "video_render_frames.cc",
+    "video_render_frames.h",
   ]
 
   include_dirs = [ "../modules/interface" ]
@@ -51,7 +53,9 @@ source_set("common_video") {
 
   if (rtc_build_libyuv) {
     deps += [ "$rtc_libyuv_dir" ]
-    public_deps = [ "$rtc_libyuv_dir" ]
+    public_deps = [
+      "$rtc_libyuv_dir",
+    ]
   } else {
     # Need to add a directory normally exported by libyuv.
     include_dirs += [ "$rtc_libyuv_dir/include" ]
diff --git a/media/webrtc/trunk/webrtc/common_video/common_video.gyp b/media/webrtc/trunk/webrtc/common_video/common_video.gyp
index 5a412014ea..fe14da1d2e 100644
--- a/media/webrtc/trunk/webrtc/common_video/common_video.gyp
+++ b/media/webrtc/trunk/webrtc/common_video/common_video.gyp
@@ -14,7 +14,7 @@
       'type': 'static_library',
       'include_dirs': [
         '<(webrtc_root)/modules/interface/',
-        'interface',
+        'include',
         'libyuv/include',
       ],
       'dependencies': [
@@ -23,7 +23,7 @@
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          'interface',
+          'include',
           'libyuv/include',
         ],
       },
@@ -39,17 +39,19 @@
         }],
       ],
       'sources': [
-        'interface/i420_buffer_pool.h',
-        'interface/i420_video_frame.h',
-        'interface/native_handle.h',
-        'interface/video_frame_buffer.h',
         'i420_buffer_pool.cc',
-        'i420_video_frame.cc',
-        'libyuv/include/webrtc_libyuv.h',
+        'video_frame.cc',
+        'incoming_video_stream.cc',
+        'include/i420_buffer_pool.h',
+        'include/incoming_video_stream.h',
+        'include/video_frame_buffer.h',
         'libyuv/include/scaler.h',
-        'libyuv/webrtc_libyuv.cc',
+        'libyuv/include/webrtc_libyuv.h',
         'libyuv/scaler.cc',
+        'libyuv/webrtc_libyuv.cc',
         'video_frame_buffer.cc',
+        'video_render_frames.cc',
+        'video_render_frames.h',
       ],
     },
   ],  # targets
diff --git a/media/webrtc/trunk/webrtc/common_video/common_video_unittests.gyp b/media/webrtc/trunk/webrtc/common_video/common_video_unittests.gyp
index beeab5ddca..b5e892caf0 100644
--- a/media/webrtc/trunk/webrtc/common_video/common_video_unittests.gyp
+++ b/media/webrtc/trunk/webrtc/common_video/common_video_unittests.gyp
@@ -17,6 +17,7 @@
          '<(DEPTH)/testing/gtest.gyp:gtest',
          '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
          '<(webrtc_root)/test/test.gyp:test_support_main',
+         '<(webrtc_root)/test/test.gyp:fake_video_frames',
       ],
       'sources': [
         'i420_buffer_pool_unittest.cc',
diff --git a/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool.cc b/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool.cc
index 04a0ab9b7f..98daec99f6 100644
--- a/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool.cc
+++ b/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
 
 #include "webrtc/base/checks.h"
 
@@ -27,18 +27,21 @@ class PooledI420Buffer : public webrtc::VideoFrameBuffer {
   int width() const override { return buffer_->width(); }
   int height() const override { return buffer_->height(); }
   const uint8_t* data(webrtc::PlaneType type) const override {
-    const webrtc::I420Buffer* cbuffer = buffer_.get();
-    return cbuffer->data(type);
+    return buffer_->data(type);
   }
-  uint8_t* data(webrtc::PlaneType type) {
-    DCHECK(HasOneRef());
-    const webrtc::I420Buffer* cbuffer = buffer_.get();
-    return const_cast<uint8_t*>(cbuffer->data(type));
+  uint8_t* MutableData(webrtc::PlaneType type) override {
+    // Make the HasOneRef() check here instead of in |buffer_|, because the pool
+    // also has a reference to |buffer_|.
+    RTC_DCHECK(HasOneRef());
+    return const_cast<uint8_t*>(buffer_->data(type));
   }
   int stride(webrtc::PlaneType type) const override {
     return buffer_->stride(type);
   }
-  rtc::scoped_refptr<webrtc::NativeHandle> native_handle() const override {
+  void* native_handle() const override { return nullptr; }
+
+  rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override {
+    RTC_NOTREACHED();
     return nullptr;
   }
 
@@ -61,7 +64,7 @@ void I420BufferPool::Release() {
 
 rtc::scoped_refptr<VideoFrameBuffer> I420BufferPool::CreateBuffer(int width,
                                                                   int height) {
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   // Release buffers with wrong resolution.
   for (auto it = buffers_.begin(); it != buffers_.end();) {
     if ((*it)->width() != width || (*it)->height() != height)
diff --git a/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool_unittest.cc b/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool_unittest.cc
index 625160be11..b030ee774a 100644
--- a/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_video/i420_buffer_pool_unittest.cc
@@ -11,7 +11,7 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
 
 namespace webrtc {
 
@@ -68,7 +68,7 @@ TEST(TestI420BufferPool, FrameValidAfterPoolDestruction) {
   EXPECT_EQ(16, buffer->width());
   EXPECT_EQ(16, buffer->height());
   // Try to trigger use-after-free errors by writing to y-plane.
-  memset(buffer->data(kYPlane), 0xA5, 16 * buffer->stride(kYPlane));
+  memset(buffer->MutableData(kYPlane), 0xA5, 16 * buffer->stride(kYPlane));
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/i420_video_frame.cc b/media/webrtc/trunk/webrtc/common_video/i420_video_frame.cc
deleted file mode 100644
index 25e1bf7651..0000000000
--- a/media/webrtc/trunk/webrtc/common_video/i420_video_frame.cc
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_video/interface/i420_video_frame.h"
-
-#include <string.h>
-
-#include <algorithm>  // swap
-
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-I420VideoFrame::I420VideoFrame() {
-  // Intentionally using Reset instead of initializer list so that any missed
-  // fields in Reset will be caught by memory checkers.
-  Reset();
-}
-
-I420VideoFrame::I420VideoFrame(
-    const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
-    uint32_t timestamp,
-    int64_t render_time_ms,
-    VideoRotation rotation)
-    : video_frame_buffer_(buffer),
-      timestamp_(timestamp),
-      ntp_time_ms_(0),
-      render_time_ms_(render_time_ms),
-      rotation_(rotation) {
-}
-
-I420VideoFrame::I420VideoFrame(NativeHandle* handle,
-                               int width,
-                               int height,
-                               uint32_t timestamp,
-                               int64_t render_time_ms)
-    : video_frame_buffer_(
-          new rtc::RefCountedObject<TextureBuffer>(handle, width, height)),
-      timestamp_(timestamp),
-      ntp_time_ms_(0),
-      render_time_ms_(render_time_ms),
-      rotation_(kVideoRotation_0) {
-  DCHECK(handle != nullptr);
-  DCHECK_GT(width, 0);
-  DCHECK_GT(height, 0);
-}
-
-int I420VideoFrame::CreateEmptyFrame(int width, int height,
-                                     int stride_y, int stride_u, int stride_v) {
-  const int half_width = (width + 1) / 2;
-  DCHECK_GT(width, 0);
-  DCHECK_GT(height, 0);
-  DCHECK_GE(stride_y, width);
-  DCHECK_GE(stride_u, half_width);
-  DCHECK_GE(stride_v, half_width);
-
-  // Creating empty frame - reset all values.
-  timestamp_ = 0;
-  ntp_time_ms_ = 0;
-  render_time_ms_ = 0;
-  rotation_ = kVideoRotation_0;
-
-  // Check if it's safe to reuse allocation.
-  if (video_frame_buffer_ &&
-      video_frame_buffer_->HasOneRef() &&
-      !video_frame_buffer_->native_handle() &&
-      width == video_frame_buffer_->width() &&
-      height == video_frame_buffer_->height() &&
-      stride_y == stride(kYPlane) &&
-      stride_u == stride(kUPlane) &&
-      stride_v == stride(kVPlane)) {
-    return 0;
-  }
-
-  // Need to allocate new buffer.
-  video_frame_buffer_ = new rtc::RefCountedObject<I420Buffer>(
-      width, height, stride_y, stride_u, stride_v);
-  return 0;
-}
-
-int I420VideoFrame::CreateFrame(const uint8_t* buffer_y,
-                                const uint8_t* buffer_u,
-                                const uint8_t* buffer_v,
-                                int width, int height,
-                                int stride_y,
-                                int stride_u,
-                                int stride_v) {
-  return CreateFrame(buffer_y, buffer_u, buffer_v,
-                     width, height, stride_y, stride_u, stride_v,
-                     kVideoRotation_0);
-}
-
-int I420VideoFrame::CreateFrame(const uint8_t* buffer_y,
-                                const uint8_t* buffer_u,
-                                const uint8_t* buffer_v,
-                                int width,
-                                int height,
-                                int stride_y,
-                                int stride_u,
-                                int stride_v,
-                                VideoRotation rotation) {
-  const int half_height = (height + 1) / 2;
-  const int expected_size_y = height * stride_y;
-  const int expected_size_u = half_height * stride_u;
-  const int expected_size_v = half_height * stride_v;
-  CreateEmptyFrame(width, height, stride_y, stride_u, stride_v);
-  memcpy(buffer(kYPlane), buffer_y, expected_size_y);
-  memcpy(buffer(kUPlane), buffer_u, expected_size_u);
-  memcpy(buffer(kVPlane), buffer_v, expected_size_v);
-  rotation_ = rotation;
-  return 0;
-}
-
-int I420VideoFrame::CreateFrame(const uint8_t* buffer,
-                                int width,
-                                int height,
-                                VideoRotation rotation) {
-  const int stride_y = width;
-  const int stride_uv = (width + 1) / 2;
-
-  const uint8_t* buffer_y = buffer;
-  const uint8_t* buffer_u = buffer_y + stride_y * height;
-  const uint8_t* buffer_v = buffer_u + stride_uv * ((height + 1) / 2);
-  return CreateFrame(buffer_y, buffer_u, buffer_v, width, height, stride_y,
-                     stride_uv, stride_uv, rotation);
-}
-
-int I420VideoFrame::CopyFrame(const I420VideoFrame& videoFrame) {
-  if (videoFrame.IsZeroSize()) {
-    video_frame_buffer_ = nullptr;
-  } else if (videoFrame.native_handle()) {
-    video_frame_buffer_ = videoFrame.video_frame_buffer();
-  } else {
-    CreateFrame(videoFrame.buffer(kYPlane), videoFrame.buffer(kUPlane),
-                videoFrame.buffer(kVPlane), videoFrame.width(),
-                videoFrame.height(), videoFrame.stride(kYPlane),
-                videoFrame.stride(kUPlane), videoFrame.stride(kVPlane));
-  }
-
-  timestamp_ = videoFrame.timestamp_;
-  ntp_time_ms_ = videoFrame.ntp_time_ms_;
-  render_time_ms_ = videoFrame.render_time_ms_;
-  rotation_ = videoFrame.rotation_;
-  return 0;
-}
-
-void I420VideoFrame::ShallowCopy(const I420VideoFrame& videoFrame) {
-  video_frame_buffer_ = videoFrame.video_frame_buffer();
-  timestamp_ = videoFrame.timestamp_;
-  ntp_time_ms_ = videoFrame.ntp_time_ms_;
-  render_time_ms_ = videoFrame.render_time_ms_;
-  rotation_ = videoFrame.rotation_;
-}
-
-void I420VideoFrame::Reset() {
-  video_frame_buffer_ = nullptr;
-  timestamp_ = 0;
-  ntp_time_ms_ = 0;
-  render_time_ms_ = 0;
-  rotation_ = kVideoRotation_0;
-}
-
-uint8_t* I420VideoFrame::buffer(PlaneType type) {
-  return video_frame_buffer_ ? video_frame_buffer_->data(type) : nullptr;
-}
-
-const uint8_t* I420VideoFrame::buffer(PlaneType type) const {
-  // Const cast to call the correct const-version of data.
-  const VideoFrameBuffer* const_buffer = video_frame_buffer_.get();
-  return const_buffer ? const_buffer->data(type) : nullptr;
-}
-
-int I420VideoFrame::allocated_size(PlaneType type) const {
-  const int plane_height = (type == kYPlane) ? height() : (height() + 1) / 2;
-  return plane_height * stride(type);
-}
-
-int I420VideoFrame::stride(PlaneType type) const {
-  return video_frame_buffer_ ? video_frame_buffer_->stride(type) : 0;
-}
-
-int I420VideoFrame::width() const {
-  return video_frame_buffer_ ? video_frame_buffer_->width() : 0;
-}
-
-int I420VideoFrame::height() const {
-  return video_frame_buffer_ ? video_frame_buffer_->height() : 0;
-}
-
-bool I420VideoFrame::IsZeroSize() const {
-  return !video_frame_buffer_;
-}
-
-void* I420VideoFrame::native_handle() const {
-  return video_frame_buffer_ ? video_frame_buffer_->native_handle() : nullptr;
-}
-
-rtc::scoped_refptr<VideoFrameBuffer> I420VideoFrame::video_frame_buffer()
-    const {
-  return video_frame_buffer_;
-}
-
-void I420VideoFrame::set_video_frame_buffer(
-    const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer) {
-  video_frame_buffer_ = buffer;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/i420_video_frame_unittest.cc b/media/webrtc/trunk/webrtc/common_video/i420_video_frame_unittest.cc
index 8273afc620..1ec451cb79 100644
--- a/media/webrtc/trunk/webrtc/common_video/i420_video_frame_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_video/i420_video_frame_unittest.cc
@@ -12,48 +12,34 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/bind.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
-class NativeHandleImpl : public NativeHandle {
- public:
-  NativeHandleImpl() : ref_count_(0) {}
-  virtual ~NativeHandleImpl() {}
-  virtual int32_t AddRef() { return ++ref_count_; }
-  virtual int32_t Release() { return --ref_count_; }
-  virtual void* GetHandle() { return NULL; }
-
-  int32_t ref_count() { return ref_count_; }
- private:
-  int32_t ref_count_;
-};
-
 bool EqualPlane(const uint8_t* data1,
                 const uint8_t* data2,
                 int stride,
                 int width,
                 int height);
-bool EqualFrames(const I420VideoFrame& frame1, const I420VideoFrame& frame2);
-bool EqualTextureFrames(const I420VideoFrame& frame1,
-                        const I420VideoFrame& frame2);
 int ExpectedSize(int plane_stride, int image_height, PlaneType type);
 
-TEST(TestI420VideoFrame, InitialValues) {
-  I420VideoFrame frame;
+TEST(TestVideoFrame, InitialValues) {
+  VideoFrame frame;
   EXPECT_TRUE(frame.IsZeroSize());
   EXPECT_EQ(kVideoRotation_0, frame.rotation());
 }
 
-TEST(TestI420VideoFrame, CopiesInitialFrameWithoutCrashing) {
-  I420VideoFrame frame;
-  I420VideoFrame frame2;
+TEST(TestVideoFrame, CopiesInitialFrameWithoutCrashing) {
+  VideoFrame frame;
+  VideoFrame frame2;
   frame2.CopyFrame(frame);
 }
 
-TEST(TestI420VideoFrame, WidthHeightValues) {
-  I420VideoFrame frame;
+TEST(TestVideoFrame, WidthHeightValues) {
+  VideoFrame frame;
   const int valid_value = 10;
   EXPECT_EQ(0, frame.CreateEmptyFrame(10, 10, 10, 14, 90));
   EXPECT_EQ(valid_value, frame.width());
@@ -66,8 +52,8 @@ TEST(TestI420VideoFrame, WidthHeightValues) {
   EXPECT_EQ(789, frame.render_time_ms());
 }
 
-TEST(TestI420VideoFrame, SizeAllocation) {
-  I420VideoFrame frame;
+TEST(TestVideoFrame, SizeAllocation) {
+  VideoFrame frame;
   EXPECT_EQ(0, frame. CreateEmptyFrame(10, 10, 12, 14, 220));
   int height = frame.height();
   int stride_y = frame.stride(kYPlane);
@@ -82,7 +68,7 @@ TEST(TestI420VideoFrame, SizeAllocation) {
             frame.allocated_size(kVPlane));
 }
 
-TEST(TestI420VideoFrame, CopyFrame) {
+TEST(TestVideoFrame, CopyFrame) {
   uint32_t timestamp = 1;
   int64_t ntp_time_ms = 2;
   int64_t render_time_ms = 3;
@@ -92,7 +78,7 @@ TEST(TestI420VideoFrame, CopyFrame) {
   int width = 15;
   int height = 15;
   // Copy frame.
-  I420VideoFrame small_frame;
+  VideoFrame small_frame;
   EXPECT_EQ(0, small_frame.CreateEmptyFrame(width, height,
                                             stride_y, stride_u, stride_v));
   small_frame.set_timestamp(timestamp);
@@ -108,14 +94,14 @@ TEST(TestI420VideoFrame, CopyFrame) {
   memset(buffer_y, 16, kSizeY);
   memset(buffer_u, 8, kSizeU);
   memset(buffer_v, 4, kSizeV);
-  I420VideoFrame big_frame;
+  VideoFrame big_frame;
   EXPECT_EQ(0,
             big_frame.CreateFrame(buffer_y, buffer_u, buffer_v,
                                   width + 5, height + 5, stride_y + 5,
                                   stride_u, stride_v, kRotation));
   // Frame of smaller dimensions.
   EXPECT_EQ(0, small_frame.CopyFrame(big_frame));
-  EXPECT_TRUE(EqualFrames(small_frame, big_frame));
+  EXPECT_TRUE(small_frame.EqualsFrame(big_frame));
   EXPECT_EQ(kRotation, small_frame.rotation());
 
   // Frame of larger dimensions.
@@ -125,10 +111,10 @@ TEST(TestI420VideoFrame, CopyFrame) {
   memset(small_frame.buffer(kUPlane), 2, small_frame.allocated_size(kUPlane));
   memset(small_frame.buffer(kVPlane), 3, small_frame.allocated_size(kVPlane));
   EXPECT_EQ(0, big_frame.CopyFrame(small_frame));
-  EXPECT_TRUE(EqualFrames(small_frame, big_frame));
+  EXPECT_TRUE(small_frame.EqualsFrame(big_frame));
 }
 
-TEST(TestI420VideoFrame, ShallowCopy) {
+TEST(TestVideoFrame, ShallowCopy) {
   uint32_t timestamp = 1;
   int64_t ntp_time_ms = 2;
   int64_t render_time_ms = 3;
@@ -148,18 +134,18 @@ TEST(TestI420VideoFrame, ShallowCopy) {
   memset(buffer_y, 16, kSizeY);
   memset(buffer_u, 8, kSizeU);
   memset(buffer_v, 4, kSizeV);
-  I420VideoFrame frame1;
+  VideoFrame frame1;
   EXPECT_EQ(0, frame1.CreateFrame(buffer_y, buffer_u, buffer_v, width, height,
                                   stride_y, stride_u, stride_v, kRotation));
   frame1.set_timestamp(timestamp);
   frame1.set_ntp_time_ms(ntp_time_ms);
   frame1.set_render_time_ms(render_time_ms);
-  I420VideoFrame frame2;
+  VideoFrame frame2;
   frame2.ShallowCopy(frame1);
 
   // To be able to access the buffers, we need const pointers to the frames.
-  const I420VideoFrame* const_frame1_ptr = &frame1;
-  const I420VideoFrame* const_frame2_ptr = &frame2;
+  const VideoFrame* const_frame1_ptr = &frame1;
+  const VideoFrame* const_frame2_ptr = &frame2;
 
   EXPECT_TRUE(const_frame1_ptr->buffer(kYPlane) ==
               const_frame2_ptr->buffer(kYPlane));
@@ -184,9 +170,9 @@ TEST(TestI420VideoFrame, ShallowCopy) {
   EXPECT_NE(frame2.rotation(), frame1.rotation());
 }
 
-TEST(TestI420VideoFrame, Reset) {
-  I420VideoFrame frame;
-  ASSERT_TRUE(frame.CreateEmptyFrame(5, 5, 5, 5, 5) == 0);
+TEST(TestVideoFrame, Reset) {
+  VideoFrame frame;
+  ASSERT_EQ(frame.CreateEmptyFrame(5, 5, 5, 5, 5), 0);
   frame.set_ntp_time_ms(1);
   frame.set_timestamp(2);
   frame.set_render_time_ms(3);
@@ -199,8 +185,8 @@ TEST(TestI420VideoFrame, Reset) {
   EXPECT_TRUE(frame.video_frame_buffer() == NULL);
 }
 
-TEST(TestI420VideoFrame, CopyBuffer) {
-  I420VideoFrame frame1, frame2;
+TEST(TestVideoFrame, CopyBuffer) {
+  VideoFrame frame1, frame2;
   int width = 15;
   int height = 15;
   int stride_y = 15;
@@ -228,8 +214,8 @@ TEST(TestI420VideoFrame, CopyBuffer) {
   EXPECT_LE(kSizeUv, frame2.allocated_size(kVPlane));
 }
 
-TEST(TestI420VideoFrame, ReuseAllocation) {
-  I420VideoFrame frame;
+TEST(TestVideoFrame, ReuseAllocation) {
+  VideoFrame frame;
   frame.CreateEmptyFrame(640, 320, 640, 320, 320);
   const uint8_t* y = frame.buffer(kYPlane);
   const uint8_t* u = frame.buffer(kUPlane);
@@ -240,28 +226,29 @@ TEST(TestI420VideoFrame, ReuseAllocation) {
   EXPECT_EQ(v, frame.buffer(kVPlane));
 }
 
-TEST(TestI420VideoFrame, FailToReuseAllocation) {
-  I420VideoFrame frame1;
+TEST(TestVideoFrame, FailToReuseAllocation) {
+  VideoFrame frame1;
   frame1.CreateEmptyFrame(640, 320, 640, 320, 320);
   const uint8_t* y = frame1.buffer(kYPlane);
   const uint8_t* u = frame1.buffer(kUPlane);
   const uint8_t* v = frame1.buffer(kVPlane);
   // Make a shallow copy of |frame1|.
-  I420VideoFrame frame2(frame1.video_frame_buffer(), 0, 0, kVideoRotation_0);
+  VideoFrame frame2(frame1.video_frame_buffer(), 0, 0, kVideoRotation_0);
   frame1.CreateEmptyFrame(640, 320, 640, 320, 320);
   EXPECT_NE(y, frame1.buffer(kYPlane));
   EXPECT_NE(u, frame1.buffer(kUPlane));
   EXPECT_NE(v, frame1.buffer(kVPlane));
 }
 
-TEST(TestI420VideoFrame, TextureInitialValues) {
-  NativeHandleImpl handle;
-  I420VideoFrame frame(&handle, 640, 480, 100, 10);
+TEST(TestVideoFrame, TextureInitialValues) {
+  test::FakeNativeHandle* handle = new test::FakeNativeHandle();
+  VideoFrame frame = test::FakeNativeHandle::CreateFrame(
+      handle, 640, 480, 100, 10, webrtc::kVideoRotation_0);
   EXPECT_EQ(640, frame.width());
   EXPECT_EQ(480, frame.height());
   EXPECT_EQ(100u, frame.timestamp());
   EXPECT_EQ(10, frame.render_time_ms());
-  EXPECT_EQ(&handle, frame.native_handle());
+  EXPECT_EQ(handle, frame.native_handle());
 
   frame.set_timestamp(200);
   EXPECT_EQ(200u, frame.timestamp());
@@ -269,66 +256,4 @@ TEST(TestI420VideoFrame, TextureInitialValues) {
   EXPECT_EQ(20, frame.render_time_ms());
 }
 
-TEST(TestI420VideoFrame, RefCount) {
-  NativeHandleImpl handle;
-  EXPECT_EQ(0, handle.ref_count());
-  I420VideoFrame *frame = new I420VideoFrame(&handle, 640, 480, 100, 200);
-  EXPECT_EQ(1, handle.ref_count());
-  delete frame;
-  EXPECT_EQ(0, handle.ref_count());
-}
-
-bool EqualPlane(const uint8_t* data1,
-                const uint8_t* data2,
-                int stride,
-                int width,
-                int height) {
-  for (int y = 0; y < height; ++y) {
-    if (memcmp(data1, data2, width) != 0)
-      return false;
-    data1 += stride;
-    data2 += stride;
-  }
-  return true;
-}
-
-bool EqualFrames(const I420VideoFrame& frame1, const I420VideoFrame& frame2) {
-  if ((frame1.width() != frame2.width()) ||
-      (frame1.height() != frame2.height()) ||
-      (frame1.stride(kYPlane) != frame2.stride(kYPlane)) ||
-      (frame1.stride(kUPlane) != frame2.stride(kUPlane)) ||
-      (frame1.stride(kVPlane) != frame2.stride(kVPlane)) ||
-      (frame1.timestamp() != frame2.timestamp()) ||
-      (frame1.ntp_time_ms() != frame2.ntp_time_ms()) ||
-      (frame1.render_time_ms() != frame2.render_time_ms())) {
-    return false;
-  }
-  const int half_width = (frame1.width() + 1) / 2;
-  const int half_height = (frame1.height() + 1) / 2;
-  return EqualPlane(frame1.buffer(kYPlane), frame2.buffer(kYPlane),
-                    frame1.stride(kYPlane), frame1.width(), frame1.height()) &&
-         EqualPlane(frame1.buffer(kUPlane), frame2.buffer(kUPlane),
-                    frame1.stride(kUPlane), half_width, half_height) &&
-         EqualPlane(frame1.buffer(kVPlane), frame2.buffer(kVPlane),
-                    frame1.stride(kVPlane), half_width, half_height);
-}
-
-bool EqualTextureFrames(const I420VideoFrame& frame1,
-                        const I420VideoFrame& frame2) {
-  return ((frame1.native_handle() == frame2.native_handle()) &&
-          (frame1.width() == frame2.width()) &&
-          (frame1.height() == frame2.height()) &&
-          (frame1.timestamp() == frame2.timestamp()) &&
-          (frame1.render_time_ms() == frame2.render_time_ms()));
-}
-
-int ExpectedSize(int plane_stride, int image_height, PlaneType type) {
-  if (type == kYPlane) {
-    return (plane_stride * image_height);
-  } else {
-    int half_height = (image_height + 1) / 2;
-    return (plane_stride * half_height);
-  }
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/interface/i420_buffer_pool.h b/media/webrtc/trunk/webrtc/common_video/include/i420_buffer_pool.h
similarity index 85%
rename from media/webrtc/trunk/webrtc/common_video/interface/i420_buffer_pool.h
rename to media/webrtc/trunk/webrtc/common_video/include/i420_buffer_pool.h
index df862cdba5..5ab1510689 100644
--- a/media/webrtc/trunk/webrtc/common_video/interface/i420_buffer_pool.h
+++ b/media/webrtc/trunk/webrtc/common_video/include/i420_buffer_pool.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_COMMON_VIDEO_INTERFACE_I420_BUFFER_POOL_H_
-#define WEBRTC_COMMON_VIDEO_INTERFACE_I420_BUFFER_POOL_H_
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
 
 #include <list>
 
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
 
 namespace webrtc {
 
@@ -40,4 +40,4 @@ class I420BufferPool {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_COMMON_VIDEO_INTERFACE_I420_BUFFER_POOL_H_
+#endif  // WEBRTC_COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/include/incoming_video_stream.h b/media/webrtc/trunk/webrtc/common_video/include/incoming_video_stream.h
new file mode 100644
index 0000000000..e3147eb871
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_video/include/incoming_video_stream.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
+
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_video/video_render_frames.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventTimerWrapper;
+
+class VideoRenderCallback {
+ public:
+  virtual int32_t RenderFrame(const uint32_t streamId,
+                              const VideoFrame& videoFrame) = 0;
+
+ protected:
+  virtual ~VideoRenderCallback() {}
+};
+
+class IncomingVideoStream : public VideoRenderCallback {
+ public:
+  IncomingVideoStream(uint32_t stream_id, bool disable_prerenderer_smoothing);
+  ~IncomingVideoStream();
+
+  // Get callback to deliver frames to the module.
+  VideoRenderCallback* ModuleCallback();
+  virtual int32_t RenderFrame(const uint32_t stream_id,
+                              const VideoFrame& video_frame);
+
+  // Set callback to the platform dependent code.
+  void SetRenderCallback(VideoRenderCallback* render_callback);
+
+  // Callback for file recording, snapshot, ...
+  void SetExternalCallback(VideoRenderCallback* render_object);
+
+  // Start/Stop.
+  int32_t Start();
+  int32_t Stop();
+
+  // Clear all buffers.
+  int32_t Reset();
+
+  // Properties.
+  uint32_t StreamId() const;
+  uint32_t IncomingRate() const;
+
+  int32_t SetStartImage(const VideoFrame& video_frame);
+
+  int32_t SetTimeoutImage(const VideoFrame& video_frame,
+                          const uint32_t timeout);
+
+  int32_t SetExpectedRenderDelay(int32_t delay_ms);
+
+ protected:
+  static bool IncomingVideoStreamThreadFun(void* obj);
+  bool IncomingVideoStreamProcess();
+
+ private:
+  enum { kEventStartupTimeMs = 10 };
+  enum { kEventMaxWaitTimeMs = 100 };
+  enum { kFrameRatePeriodMs = 1000 };
+
+  void DeliverFrame(const VideoFrame& video_frame);
+
+  uint32_t const stream_id_;
+  const bool disable_prerenderer_smoothing_;
+  // Critsects in allowed to enter order.
+  const rtc::scoped_ptr<CriticalSectionWrapper> stream_critsect_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> thread_critsect_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> buffer_critsect_;
+  // TODO(pbos): Make plain member and stop resetting this thread, just
+  // start/stoping it is enough.
+  rtc::scoped_ptr<rtc::PlatformThread> incoming_render_thread_
+      GUARDED_BY(thread_critsect_);
+  rtc::scoped_ptr<EventTimerWrapper> deliver_buffer_event_;
+
+  bool running_ GUARDED_BY(stream_critsect_);
+  VideoRenderCallback* external_callback_ GUARDED_BY(thread_critsect_);
+  VideoRenderCallback* render_callback_ GUARDED_BY(thread_critsect_);
+  const rtc::scoped_ptr<VideoRenderFrames> render_buffers_
+      GUARDED_BY(buffer_critsect_);
+
+  uint32_t incoming_rate_ GUARDED_BY(stream_critsect_);
+  int64_t last_rate_calculation_time_ms_ GUARDED_BY(stream_critsect_);
+  uint16_t num_frames_since_last_calculation_ GUARDED_BY(stream_critsect_);
+  int64_t last_render_time_ms_ GUARDED_BY(thread_critsect_);
+  VideoFrame temp_frame_ GUARDED_BY(thread_critsect_);
+  VideoFrame start_image_ GUARDED_BY(thread_critsect_);
+  VideoFrame timeout_image_ GUARDED_BY(thread_critsect_);
+  uint32_t timeout_time_ GUARDED_BY(thread_critsect_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/interface/video_frame_buffer.h b/media/webrtc/trunk/webrtc/common_video/include/video_frame_buffer.h
similarity index 62%
rename from media/webrtc/trunk/webrtc/common_video/interface/video_frame_buffer.h
rename to media/webrtc/trunk/webrtc/common_video/include/video_frame_buffer.h
index e7cae48a9f..710d2862f0 100644
--- a/media/webrtc/trunk/webrtc/common_video/interface/video_frame_buffer.h
+++ b/media/webrtc/trunk/webrtc/common_video/include/video_frame_buffer.h
@@ -8,15 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_FRAME_BUFFER_H_
-#define WEBRTC_VIDEO_FRAME_BUFFER_H_
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
 
 #include "webrtc/base/callback.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/common_video/interface/native_handle.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 namespace webrtc {
 
@@ -43,15 +42,20 @@ class VideoFrameBuffer : public rtc::RefCountInterface {
   // the VideoFrameBuffer object and must not be freed by the caller.
   virtual const uint8_t* data(PlaneType type) const = 0;
 
-  // Non-const data access is only allowed if |HasOneRef| is true.
-  virtual uint8_t* data(PlaneType type) = 0;
+  // Non-const data access is disallowed by default. You need to make sure you
+  // have exclusive access and a writable buffer before calling this function.
+  virtual uint8_t* MutableData(PlaneType type);
 
   // Returns the number of bytes between successive rows for a given plane.
   virtual int stride(PlaneType type) const = 0;
 
   // Return the handle of the underlying video frame. This is used when the
   // frame is backed by a texture.
-  virtual rtc::scoped_refptr<NativeHandle> native_handle() const = 0;
+  virtual void* native_handle() const = 0;
+
+  // Returns a new memory-backed frame buffer converted from this buffer's
+  // native handle.
+  virtual rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() = 0;
 
  protected:
   virtual ~VideoFrameBuffer();
@@ -66,9 +70,12 @@ class I420Buffer : public VideoFrameBuffer {
   int width() const override;
   int height() const override;
   const uint8_t* data(PlaneType type) const override;
-  uint8_t* data(PlaneType type) override;
+  // Non-const data access is only allowed if HasOneRef() is true to protect
+  // against unexpected overwrites.
+  uint8_t* MutableData(PlaneType type) override;
   int stride(PlaneType type) const override;
-  rtc::scoped_refptr<NativeHandle> native_handle() const override;
+  void* native_handle() const override;
+  rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
 
  protected:
   ~I420Buffer() override;
@@ -82,34 +89,29 @@ class I420Buffer : public VideoFrameBuffer {
   const rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> data_;
 };
 
-// Texture buffer around a NativeHandle.
-class TextureBuffer : public VideoFrameBuffer {
+// Base class for native-handle buffer is a wrapper around a |native_handle|.
+// This is used for convenience as most native-handle implementations can share
+// many VideoFrame implementations, but need to implement a few others (such
+// as their own destructors or conversion methods back to software I420).
+class NativeHandleBuffer : public VideoFrameBuffer {
  public:
-  TextureBuffer(const rtc::scoped_refptr<NativeHandle>& native_handle,
-                int width,
-                int height);
+  NativeHandleBuffer(void* native_handle, int width, int height);
 
   int width() const override;
   int height() const override;
   const uint8_t* data(PlaneType type) const override;
-  uint8_t* data(PlaneType type) override;
   int stride(PlaneType type) const override;
-  rtc::scoped_refptr<NativeHandle> native_handle() const override;
+  void* native_handle() const override;
 
- private:
-  friend class rtc::RefCountedObject<TextureBuffer>;
-  ~TextureBuffer() override;
-
-  const rtc::scoped_refptr<NativeHandle> native_handle_;
+ protected:
+  void* native_handle_;
   const int width_;
   const int height_;
 };
 
 class WrappedI420Buffer : public webrtc::VideoFrameBuffer {
  public:
-  WrappedI420Buffer(int desired_width,
-                    int desired_height,
-                    int width,
+  WrappedI420Buffer(int width,
                     int height,
                     const uint8_t* y_plane,
                     int y_stride,
@@ -122,26 +124,34 @@ class WrappedI420Buffer : public webrtc::VideoFrameBuffer {
   int height() const override;
 
   const uint8_t* data(PlaneType type) const override;
-  uint8_t* data(PlaneType type) override;
 
   int stride(PlaneType type) const override;
-  rtc::scoped_refptr<NativeHandle> native_handle() const override;
+  void* native_handle() const override;
+
+  rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
 
  private:
   friend class rtc::RefCountedObject<WrappedI420Buffer>;
   ~WrappedI420Buffer() override;
 
-  int width_;
-  int height_;
-  const uint8_t* y_plane_;
-  const uint8_t* u_plane_;
-  const uint8_t* v_plane_;
+  const int width_;
+  const int height_;
+  const uint8_t* const y_plane_;
+  const uint8_t* const u_plane_;
+  const uint8_t* const v_plane_;
   const int y_stride_;
   const int u_stride_;
   const int v_stride_;
   rtc::Callback0<void> no_longer_used_cb_;
 };
 
+// Helper function to crop |buffer| without making a deep copy. May only be used
+// for non-native frames.
+rtc::scoped_refptr<VideoFrameBuffer> ShallowCenterCrop(
+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
+    int cropped_width,
+    int cropped_height);
+
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_FRAME_BUFFER_H_
+#endif  // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/interface/video_image.h b/media/webrtc/trunk/webrtc/common_video/include/video_image.h
similarity index 76%
rename from media/webrtc/trunk/webrtc/common_video/interface/video_image.h
rename to media/webrtc/trunk/webrtc/common_video/include/video_image.h
index 4cbf23f1a1..4a6e451c0f 100644
--- a/media/webrtc/trunk/webrtc/common_video/interface/video_image.h
+++ b/media/webrtc/trunk/webrtc/common_video/include/video_image.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
-#define COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_IMAGE_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_IMAGE_H_
 
 // TODO(pbos): Remove this file and include webrtc/video_frame.h instead.
 #include "webrtc/video_frame.h"
 
-#endif // COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
+#endif  // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_IMAGE_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/incoming_video_stream.cc b/media/webrtc/trunk/webrtc/common_video/incoming_video_stream.cc
new file mode 100644
index 0000000000..1272ecc5bb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_video/incoming_video_stream.cc
@@ -0,0 +1,263 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_video/include/incoming_video_stream.h"
+
+#include <assert.h>
+
+#if defined(_WIN32)
+#include <windows.h>
+#elif defined(WEBRTC_LINUX)
+#include <sys/time.h>
+#include <time.h>
+#else
+#include <sys/time.h>
+#endif
+
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/common_video/video_render_frames.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/video_renderer.h"
+
+namespace webrtc {
+
+IncomingVideoStream::IncomingVideoStream(uint32_t stream_id,
+                                         bool disable_prerenderer_smoothing)
+    : stream_id_(stream_id),
+      disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
+      stream_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+      thread_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+      buffer_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+      incoming_render_thread_(),
+      deliver_buffer_event_(EventTimerWrapper::Create()),
+      running_(false),
+      external_callback_(nullptr),
+      render_callback_(nullptr),
+      render_buffers_(new VideoRenderFrames()),
+      incoming_rate_(0),
+      last_rate_calculation_time_ms_(0),
+      num_frames_since_last_calculation_(0),
+      last_render_time_ms_(0),
+      temp_frame_(),
+      start_image_(),
+      timeout_image_(),
+      timeout_time_() {}
+
+IncomingVideoStream::~IncomingVideoStream() {
+  Stop();
+}
+
+VideoRenderCallback* IncomingVideoStream::ModuleCallback() {
+  CriticalSectionScoped cs(stream_critsect_.get());
+  return this;
+}
+
+int32_t IncomingVideoStream::RenderFrame(const uint32_t stream_id,
+                                         const VideoFrame& video_frame) {
+  CriticalSectionScoped csS(stream_critsect_.get());
+
+  if (!running_) {
+    return -1;
+  }
+
+  // Rate statistics.
+  num_frames_since_last_calculation_++;
+  int64_t now_ms = TickTime::MillisecondTimestamp();
+  if (now_ms >= last_rate_calculation_time_ms_ + kFrameRatePeriodMs) {
+    incoming_rate_ =
+        static_cast<uint32_t>(1000 * num_frames_since_last_calculation_ /
+                              (now_ms - last_rate_calculation_time_ms_));
+    num_frames_since_last_calculation_ = 0;
+    last_rate_calculation_time_ms_ = now_ms;
+  }
+
+  // Hand over or insert frame.
+  if (disable_prerenderer_smoothing_) {
+    DeliverFrame(video_frame);
+  } else {
+    CriticalSectionScoped csB(buffer_critsect_.get());
+    if (render_buffers_->AddFrame(video_frame) == 1) {
+      deliver_buffer_event_->Set();
+    }
+  }
+  return 0;
+}
+
+int32_t IncomingVideoStream::SetStartImage(const VideoFrame& video_frame) {
+  CriticalSectionScoped csS(thread_critsect_.get());
+  return start_image_.CopyFrame(video_frame);
+}
+
+int32_t IncomingVideoStream::SetTimeoutImage(const VideoFrame& video_frame,
+                                             const uint32_t timeout) {
+  CriticalSectionScoped csS(thread_critsect_.get());
+  timeout_time_ = timeout;
+  return timeout_image_.CopyFrame(video_frame);
+}
+
+void IncomingVideoStream::SetRenderCallback(
+    VideoRenderCallback* render_callback) {
+  CriticalSectionScoped cs(thread_critsect_.get());
+  render_callback_ = render_callback;
+}
+
+int32_t IncomingVideoStream::SetExpectedRenderDelay(
+    int32_t delay_ms) {
+  CriticalSectionScoped csS(stream_critsect_.get());
+  if (running_) {
+    return -1;
+  }
+  CriticalSectionScoped cs(buffer_critsect_.get());
+  return render_buffers_->SetRenderDelay(delay_ms);
+}
+
+void IncomingVideoStream::SetExternalCallback(
+    VideoRenderCallback* external_callback) {
+  CriticalSectionScoped cs(thread_critsect_.get());
+  external_callback_ = external_callback;
+}
+
+int32_t IncomingVideoStream::Start() {
+  CriticalSectionScoped csS(stream_critsect_.get());
+  if (running_) {
+    return 0;
+  }
+
+  if (!disable_prerenderer_smoothing_) {
+    CriticalSectionScoped csT(thread_critsect_.get());
+    assert(incoming_render_thread_ == NULL);
+
+    incoming_render_thread_.reset(new rtc::PlatformThread(
+        IncomingVideoStreamThreadFun, this, "IncomingVideoStreamThread"));
+    if (!incoming_render_thread_) {
+      return -1;
+    }
+
+    incoming_render_thread_->Start();
+    incoming_render_thread_->SetPriority(rtc::kRealtimePriority);
+    deliver_buffer_event_->StartTimer(false, kEventStartupTimeMs);
+  }
+
+  running_ = true;
+  return 0;
+}
+
+int32_t IncomingVideoStream::Stop() {
+  CriticalSectionScoped cs_stream(stream_critsect_.get());
+
+  if (!running_) {
+    return 0;
+  }
+
+  rtc::PlatformThread* thread = NULL;
+  {
+    CriticalSectionScoped cs_thread(thread_critsect_.get());
+    if (incoming_render_thread_) {
+      // Setting the incoming render thread to NULL marks that we're performing
+      // a shutdown and will make IncomingVideoStreamProcess abort after wakeup.
+      thread = incoming_render_thread_.release();
+      deliver_buffer_event_->StopTimer();
+      // Set the event to allow the thread to wake up and shut down without
+      // waiting for a timeout.
+      deliver_buffer_event_->Set();
+    }
+  }
+  if (thread) {
+    thread->Stop();
+    delete thread;
+  }
+  running_ = false;
+  return 0;
+}
+
+int32_t IncomingVideoStream::Reset() {
+  CriticalSectionScoped cs_buffer(buffer_critsect_.get());
+  render_buffers_->ReleaseAllFrames();
+  return 0;
+}
+
+uint32_t IncomingVideoStream::StreamId() const {
+  return stream_id_;
+}
+
+uint32_t IncomingVideoStream::IncomingRate() const {
+  CriticalSectionScoped cs(stream_critsect_.get());
+  return incoming_rate_;
+}
+
+bool IncomingVideoStream::IncomingVideoStreamThreadFun(void* obj) {
+  return static_cast<IncomingVideoStream*>(obj)->IncomingVideoStreamProcess();
+}
+
+bool IncomingVideoStream::IncomingVideoStreamProcess() {
+  if (kEventError != deliver_buffer_event_->Wait(kEventMaxWaitTimeMs)) {
+    CriticalSectionScoped cs(thread_critsect_.get());
+    if (incoming_render_thread_ == NULL) {
+      // Terminating
+      return false;
+    }
+
+    // Get a new frame to render and the time for the frame after this one.
+    VideoFrame frame_to_render;
+    uint32_t wait_time;
+    {
+      CriticalSectionScoped cs(buffer_critsect_.get());
+      frame_to_render = render_buffers_->FrameToRender();
+      wait_time = render_buffers_->TimeToNextFrameRelease();
+    }
+
+    // Set timer for next frame to render.
+    if (wait_time > kEventMaxWaitTimeMs) {
+      wait_time = kEventMaxWaitTimeMs;
+    }
+    deliver_buffer_event_->StartTimer(false, wait_time);
+
+    DeliverFrame(frame_to_render);
+  }
+  return true;
+}
+
+void IncomingVideoStream::DeliverFrame(const VideoFrame& video_frame) {
+  CriticalSectionScoped cs(thread_critsect_.get());
+  if (video_frame.IsZeroSize()) {
+    if (render_callback_) {
+      if (last_render_time_ms_ == 0 && !start_image_.IsZeroSize()) {
+        // We have not rendered anything and have a start image.
+        temp_frame_.CopyFrame(start_image_);
+        render_callback_->RenderFrame(stream_id_, temp_frame_);
+      } else if (!timeout_image_.IsZeroSize() &&
+                 last_render_time_ms_ + timeout_time_ <
+                     TickTime::MillisecondTimestamp()) {
+        // Render a timeout image.
+        temp_frame_.CopyFrame(timeout_image_);
+        render_callback_->RenderFrame(stream_id_, temp_frame_);
+      }
+    }
+
+    // No frame.
+    return;
+  }
+
+  // Send frame for rendering.
+  if (external_callback_) {
+    external_callback_->RenderFrame(stream_id_, video_frame);
+  } else if (render_callback_) {
+    render_callback_->RenderFrame(stream_id_, video_frame);
+  }
+
+  // We're done with this frame.
+  last_render_time_ms_ = video_frame.render_time_ms();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/interface/i420_video_frame.h b/media/webrtc/trunk/webrtc/common_video/interface/i420_video_frame.h
deleted file mode 100644
index ba23c87a8f..0000000000
--- a/media/webrtc/trunk/webrtc/common_video/interface/i420_video_frame.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef COMMON_VIDEO_INTERFACE_I420_VIDEO_FRAME_H
-#define COMMON_VIDEO_INTERFACE_I420_VIDEO_FRAME_H
-
-// TODO(pbos): Remove this file and include webrtc/video_frame.h instead.
-#include "webrtc/video_frame.h"
-
-#endif  // COMMON_VIDEO_INTERFACE_I420_VIDEO_FRAME_H
diff --git a/media/webrtc/trunk/webrtc/common_video/interface/native_handle.h b/media/webrtc/trunk/webrtc/common_video/interface/native_handle.h
deleted file mode 100644
index da1feabbd8..0000000000
--- a/media/webrtc/trunk/webrtc/common_video/interface/native_handle.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef COMMON_VIDEO_INTERFACE_NATIVEHANDLE_H_
-#define COMMON_VIDEO_INTERFACE_NATIVEHANDLE_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// A class to store an opaque handle of the underlying video frame. This is used
-// when the frame is backed by a texture. WebRTC carries the handle in
-// TextureBuffer. This object keeps a reference to the handle. The reference
-// is cleared when the object is destroyed. It is important to destroy the
-// object as soon as possible so the texture can be recycled.
-class NativeHandle {
- public:
-  virtual ~NativeHandle() {}
-  // For scoped_refptr
-  virtual int32_t AddRef() = 0;
-  virtual int32_t Release() = 0;
-
-  // Gets the handle.
-  virtual void* GetHandle() = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // COMMON_VIDEO_INTERFACE_NATIVEHANDLE_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/libyuv/include/scaler.h b/media/webrtc/trunk/webrtc/common_video/libyuv/include/scaler.h
index 5dff5095a1..2b92f8148b 100644
--- a/media/webrtc/trunk/webrtc/common_video/libyuv/include/scaler.h
+++ b/media/webrtc/trunk/webrtc/common_video/libyuv/include/scaler.h
@@ -15,10 +15,10 @@
 #ifndef WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
 #define WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
 
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -48,8 +48,7 @@ class Scaler {
   // Return value: 0 - OK,
   //               -1 - parameter error
   //               -2 - scaler not set
-  int Scale(const I420VideoFrame& src_frame,
-            I420VideoFrame* dst_frame);
+  int Scale(const VideoFrame& src_frame, VideoFrame* dst_frame);
 
  private:
   // Determine if the VideoTypes are currently supported.
diff --git a/media/webrtc/trunk/webrtc/common_video/libyuv/include/webrtc_libyuv.h b/media/webrtc/trunk/webrtc/common_video/libyuv/include/webrtc_libyuv.h
index d8e931d1df..d66736fb24 100644
--- a/media/webrtc/trunk/webrtc/common_video/libyuv/include/webrtc_libyuv.h
+++ b/media/webrtc/trunk/webrtc/common_video/libyuv/include/webrtc_libyuv.h
@@ -19,8 +19,8 @@
 
 #include "webrtc/common_types.h"  // RawVideoTypes.
 #include "webrtc/common_video/rotation.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -77,23 +77,22 @@ void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv);
 size_t CalcBufferSize(VideoType type, int width, int height);
 
 // TODO(mikhal): Add unit test for these two functions and determine location.
-// Print I420VideoFrame to file
+// Print VideoFrame to file
 // Input:
 //    - frame       : Reference to video frame.
 //    - file        : pointer to file object. It is assumed that the file is
 //                    already open for writing.
 // Return value: 0 if OK, < 0 otherwise.
-int PrintI420VideoFrame(const I420VideoFrame& frame, FILE* file);
+int PrintVideoFrame(const VideoFrame& frame, FILE* file);
 
-// Extract buffer from I420VideoFrame (consecutive planes, no stride)
+// Extract buffer from VideoFrame (consecutive planes, no stride)
 // Input:
 //   - frame       : Reference to video frame.
 //   - size        : pointer to the size of the allocated buffer. If size is
 //                   insufficient, an error will be returned.
 //   - buffer      : Pointer to buffer
 // Return value: length of buffer if OK, < 0 otherwise.
-int ExtractBuffer(const I420VideoFrame& input_frame,
-                  size_t size, uint8_t* buffer);
+int ExtractBuffer(const VideoFrame& input_frame, size_t size, uint8_t* buffer);
 // Convert To I420
 // Input:
 //   - src_video_type   : Type of input video.
@@ -115,7 +114,7 @@ int ConvertToI420(VideoType src_video_type,
                   int src_height,
                   size_t sample_size,
                   VideoRotation rotation,
-                  I420VideoFrame* dst_frame);
+                  VideoFrame* dst_frame);
 
 // Convert From I420
 // Input:
@@ -125,13 +124,15 @@ int ConvertToI420(VideoType src_video_type,
 //   - dst_frame        : Pointer to a destination frame.
 // Return value: 0 if OK, < 0 otherwise.
 // It is assumed that source and destination have equal height.
-int ConvertFromI420(const I420VideoFrame& src_frame,
-                    VideoType dst_video_type, int dst_sample_size,
+int ConvertFromI420(const VideoFrame& src_frame,
+                    VideoType dst_video_type,
+                    int dst_sample_size,
                     uint8_t* dst_frame);
 // ConvertFrom YV12.
 // Interface - same as above.
-int ConvertFromYV12(const I420VideoFrame& src_frame,
-                    VideoType dst_video_type, int dst_sample_size,
+int ConvertFromYV12(const VideoFrame& src_frame,
+                    VideoType dst_video_type,
+                    int dst_sample_size,
                     uint8_t* dst_frame);
 
 // The following list describes designated conversion functions which
@@ -148,11 +149,10 @@ int ConvertNV12ToRGB565(const uint8_t* src_frame,
 
 // Compute PSNR for an I420 frame (all planes).
 // Returns the PSNR in decibel, to a maximum of kInfinitePSNR.
-double I420PSNR(const I420VideoFrame* ref_frame,
-                const I420VideoFrame* test_frame);
+double I420PSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame);
 // Compute SSIM for an I420 frame (all planes).
-double I420SSIM(const I420VideoFrame* ref_frame,
-                const I420VideoFrame* test_frame);
-}
+double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame);
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_
diff --git a/media/webrtc/trunk/webrtc/common_video/libyuv/libyuv_unittest.cc b/media/webrtc/trunk/webrtc/common_video/libyuv/libyuv_unittest.cc
index 583c81adb7..e7cf0759f6 100644
--- a/media/webrtc/trunk/webrtc/common_video/libyuv/libyuv_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_video/libyuv/libyuv_unittest.cc
@@ -13,10 +13,10 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -37,8 +37,7 @@ int PrintBuffer(const uint8_t* buffer, int width, int height, int stride) {
   return 0;
 }
 
-
-int PrintFrame(const I420VideoFrame* frame, const char* str) {
+int PrintFrame(const VideoFrame* frame, const char* str) {
   if (frame == NULL)
      return -1;
   printf("%s %dx%d \n", str, frame->width(), frame->height());
@@ -57,7 +56,7 @@ int PrintFrame(const I420VideoFrame* frame, const char* str) {
 
 // Create an image from on a YUV frame. Every plane value starts with a start
 // value, and will be set to increasing values.
-void CreateImage(I420VideoFrame* frame, int plane_offset[kNumOfPlanes]) {
+void CreateImage(VideoFrame* frame, int plane_offset[kNumOfPlanes]) {
   if (frame == NULL)
     return;
   for (int plane_num = 0; plane_num < kNumOfPlanes; ++plane_num) {
@@ -83,7 +82,7 @@ class TestLibYuv : public ::testing::Test {
   virtual void TearDown();
 
   FILE* source_file_;
-  I420VideoFrame orig_frame_;
+  VideoFrame orig_frame_;
   rtc::scoped_ptr<uint8_t[]> orig_buffer_;
   const int width_;
   const int height_;
@@ -98,7 +97,7 @@ TestLibYuv::TestLibYuv()
       width_(352),
       height_(288),
       size_y_(width_ * height_),
-      size_uv_(((width_ + 1 ) / 2) * ((height_ + 1) / 2)),
+      size_uv_(((width_ + 1) / 2) * ((height_ + 1) / 2)),
       frame_length_(CalcBufferSize(kI420, 352, 288)) {
   orig_buffer_.reset(new uint8_t[frame_length_]);
 }
@@ -142,10 +141,10 @@ TEST_F(TestLibYuv, ConvertTest) {
 
   double psnr = 0.0;
 
-  I420VideoFrame res_i420_frame;
-  EXPECT_EQ(0,res_i420_frame.CreateEmptyFrame(width_, height_, width_,
-                                              (width_ + 1) / 2,
-                                              (width_ + 1) / 2));
+  VideoFrame res_i420_frame;
+  EXPECT_EQ(0, res_i420_frame.CreateEmptyFrame(width_, height_, width_,
+                                               (width_ + 1) / 2,
+                                               (width_ + 1) / 2));
   printf("\nConvert #%d I420 <-> I420 \n", j);
   rtc::scoped_ptr<uint8_t[]> out_i420_buffer(new uint8_t[frame_length_]);
   EXPECT_EQ(0, ConvertFromI420(orig_frame_, kI420, 0,
@@ -153,7 +152,7 @@ TEST_F(TestLibYuv, ConvertTest) {
   EXPECT_EQ(0, ConvertToI420(kI420, out_i420_buffer.get(), 0, 0, width_,
                              height_, 0, kVideoRotation_0, &res_i420_frame));
 
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
   psnr = I420PSNR(&orig_frame_, &res_i420_frame);
@@ -173,7 +172,7 @@ TEST_F(TestLibYuv, ConvertTest) {
   EXPECT_EQ(0, ConvertToI420(kRGB24, res_rgb_buffer2.get(), 0, 0, width_,
                              height_, 0, kVideoRotation_0, &res_i420_frame));
 
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
   psnr = I420PSNR(&orig_frame_, &res_i420_frame);
@@ -189,7 +188,7 @@ TEST_F(TestLibYuv, ConvertTest) {
                              height_, 0, kVideoRotation_0, &res_i420_frame));
   psnr = I420PSNR(&orig_frame_, &res_i420_frame);
   EXPECT_EQ(48.0, psnr);
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
   j++;
@@ -197,7 +196,7 @@ TEST_F(TestLibYuv, ConvertTest) {
   printf("\nConvert #%d I420 <-> YV12\n", j);
   rtc::scoped_ptr<uint8_t[]> outYV120Buffer(new uint8_t[frame_length_]);
   rtc::scoped_ptr<uint8_t[]> res_i420_buffer(new uint8_t[frame_length_]);
-  I420VideoFrame yv12_frame;
+  VideoFrame yv12_frame;
   EXPECT_EQ(0, ConvertFromI420(orig_frame_, kYV12, 0, outYV120Buffer.get()));
   yv12_frame.CreateFrame(outYV120Buffer.get(),
                          outYV120Buffer.get() + size_y_,
@@ -223,7 +222,7 @@ TEST_F(TestLibYuv, ConvertTest) {
   EXPECT_EQ(0, ConvertToI420(kYUY2, out_yuy2_buffer.get(), 0, 0, width_,
                              height_, 0, kVideoRotation_0, &res_i420_frame));
 
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
 
@@ -238,7 +237,7 @@ TEST_F(TestLibYuv, ConvertTest) {
   EXPECT_EQ(0, ConvertToI420(kRGB565, out_rgb565_buffer.get(), 0, 0, width_,
                              height_, 0, kVideoRotation_0, &res_i420_frame));
 
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
   j++;
@@ -258,7 +257,7 @@ TEST_F(TestLibYuv, ConvertTest) {
   EXPECT_EQ(0, ConvertToI420(kARGB, out_argb8888_buffer.get(), 0, 0, width_,
                              height_, 0, kVideoRotation_0, &res_i420_frame));
 
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
 
@@ -278,19 +277,19 @@ TEST_F(TestLibYuv, ConvertAlignedFrame) {
 
   double psnr = 0.0;
 
-  I420VideoFrame res_i420_frame;
+  VideoFrame res_i420_frame;
   int stride_y = 0;
   int stride_uv = 0;
   Calc16ByteAlignedStride(width_, &stride_y, &stride_uv);
-  EXPECT_EQ(0,res_i420_frame.CreateEmptyFrame(width_, height_,
-                                              stride_y, stride_uv, stride_uv));
+  EXPECT_EQ(0, res_i420_frame.CreateEmptyFrame(width_, height_,
+                                               stride_y, stride_uv, stride_uv));
   rtc::scoped_ptr<uint8_t[]> out_i420_buffer(new uint8_t[frame_length_]);
   EXPECT_EQ(0, ConvertFromI420(orig_frame_, kI420, 0,
                                out_i420_buffer.get()));
   EXPECT_EQ(0, ConvertToI420(kI420, out_i420_buffer.get(), 0, 0, width_,
                              height_, 0, kVideoRotation_0, &res_i420_frame));
 
-  if (PrintI420VideoFrame(res_i420_frame, output_file) < 0) {
+  if (PrintVideoFrame(res_i420_frame, output_file) < 0) {
     return;
   }
   psnr = I420PSNR(&orig_frame_, &res_i420_frame);
@@ -301,30 +300,30 @@ TEST_F(TestLibYuv, ConvertAlignedFrame) {
 TEST_F(TestLibYuv, RotateTest) {
   // Use ConvertToI420 for multiple roatations - see that nothing breaks, all
   // memory is properly allocated and end result is equal to the starting point.
-  I420VideoFrame rotated_res_i420_frame;
+  VideoFrame rotated_res_i420_frame;
   int rotated_width = height_;
   int rotated_height = width_;
-  int stride_y ;
+  int stride_y;
   int stride_uv;
   Calc16ByteAlignedStride(rotated_width, &stride_y, &stride_uv);
-  EXPECT_EQ(0,rotated_res_i420_frame.CreateEmptyFrame(rotated_width,
-                                                      rotated_height,
-                                                      stride_y,
-                                                      stride_uv,
-                                                      stride_uv));
+  EXPECT_EQ(0, rotated_res_i420_frame.CreateEmptyFrame(rotated_width,
+                                                       rotated_height,
+                                                       stride_y,
+                                                       stride_uv,
+                                                       stride_uv));
   EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer_.get(), 0, 0, width_, height_,
                              0, kVideoRotation_90, &rotated_res_i420_frame));
   EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer_.get(), 0, 0, width_, height_,
                              0, kVideoRotation_270, &rotated_res_i420_frame));
-  EXPECT_EQ(0,rotated_res_i420_frame.CreateEmptyFrame(width_, height_,
-                                                      width_, (width_ + 1) / 2,
-                                                      (width_ + 1) / 2));
+  EXPECT_EQ(0, rotated_res_i420_frame.CreateEmptyFrame(width_, height_,
+                                                       width_, (width_ + 1) / 2,
+                                                       (width_ + 1) / 2));
   EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer_.get(), 0, 0, width_, height_,
                              0, kVideoRotation_180, &rotated_res_i420_frame));
 }
 
 TEST_F(TestLibYuv, alignment) {
-  int value = 0x3FF; // 1023
+  int value = 0x3FF;  // 1023
   EXPECT_EQ(0x400, AlignInt(value, 128));  // Low 7 bits are zero.
   EXPECT_EQ(0x400, AlignInt(value, 64));  // Low 6 bits are zero.
   EXPECT_EQ(0x400, AlignInt(value, 32));  // Low 5 bits are zero.
@@ -347,4 +346,4 @@ TEST_F(TestLibYuv, StrideAlignment) {
   EXPECT_EQ(64, stride_uv);
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/libyuv/scaler.cc b/media/webrtc/trunk/webrtc/common_video/libyuv/scaler.cc
index 598c1d018f..c6adbf9507 100644
--- a/media/webrtc/trunk/webrtc/common_video/libyuv/scaler.cc
+++ b/media/webrtc/trunk/webrtc/common_video/libyuv/scaler.cc
@@ -47,8 +47,7 @@ int Scaler::Set(int src_width, int src_height,
   return 0;
 }
 
-int Scaler::Scale(const I420VideoFrame& src_frame,
-                  I420VideoFrame* dst_frame) {
+int Scaler::Scale(const VideoFrame& src_frame, VideoFrame* dst_frame) {
   assert(dst_frame);
   if (src_frame.IsZeroSize())
     return -1;
diff --git a/media/webrtc/trunk/webrtc/common_video/libyuv/scaler_unittest.cc b/media/webrtc/trunk/webrtc/common_video/libyuv/scaler_unittest.cc
index 19bcff652a..6d026383a2 100644
--- a/media/webrtc/trunk/webrtc/common_video/libyuv/scaler_unittest.cc
+++ b/media/webrtc/trunk/webrtc/common_video/libyuv/scaler_unittest.cc
@@ -13,9 +13,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_video/libyuv/include/scaler.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -37,7 +36,7 @@ class TestScaler : public ::testing::Test {
 
   Scaler test_scaler_;
   FILE* source_file_;
-  I420VideoFrame test_frame_;
+  VideoFrame test_frame_;
   const int width_;
   const int half_width_;
   const int height_;
@@ -88,7 +87,7 @@ TEST_F(TestScaler, ScaleBadInitialValues) {
 }
 
 TEST_F(TestScaler, ScaleSendingNullSourcePointer) {
-  I420VideoFrame null_src_frame;
+  VideoFrame null_src_frame;
   EXPECT_EQ(-1, test_scaler_.Scale(null_src_frame, &test_frame_));
 }
 
@@ -98,7 +97,7 @@ TEST_F(TestScaler, ScaleSendingBufferTooSmall) {
                                 half_width_, half_height_,
                                 kI420, kI420,
                                 kScalePoint));
-  I420VideoFrame test_frame2;
+  VideoFrame test_frame2;
   rtc::scoped_ptr<uint8_t[]> orig_buffer(new uint8_t[frame_length_]);
   EXPECT_GT(fread(orig_buffer.get(), 1, frame_length_, source_file_), 0U);
   test_frame_.CreateFrame(orig_buffer.get(),
@@ -114,8 +113,13 @@ TEST_F(TestScaler, ScaleSendingBufferTooSmall) {
   EXPECT_EQ(half_height_, test_frame2.height());
 }
 
-//TODO (mikhal): Converge the test into one function that accepts the method.
-TEST_F(TestScaler, DISABLED_ON_ANDROID(PointScaleTest)) {
+// TODO(mikhal): Converge the test into one function that accepts the method.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PointScaleTest DISABLED_PointScaleTest
+#else
+#define MAYBE_PointScaleTest PointScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_PointScaleTest) {
   double avg_psnr;
   FILE* source_file2;
   ScaleMethod method = kScalePoint;
@@ -182,7 +186,12 @@ TEST_F(TestScaler, DISABLED_ON_ANDROID(PointScaleTest)) {
   ASSERT_EQ(0, fclose(source_file2));
 }
 
-TEST_F(TestScaler, DISABLED_ON_ANDROID(BiLinearScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_BilinearScaleTest DISABLED_BiLinearScaleTest
+#else
+#define MAYBE_BilinearScaleTest BiLinearScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_BiLinearScaleTest) {
   double avg_psnr;
   FILE* source_file2;
   ScaleMethod method = kScaleBilinear;
@@ -234,7 +243,12 @@ TEST_F(TestScaler, DISABLED_ON_ANDROID(BiLinearScaleTest)) {
                 400, 300);
 }
 
-TEST_F(TestScaler, DISABLED_ON_ANDROID(BoxScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_BoxScaleTest DISABLED_BoxScaleTest
+#else
+#define MAYBE_BoxScaleTest BoxScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_BoxScaleTest) {
   double avg_psnr;
   FILE* source_file2;
   ScaleMethod method = kScaleBox;
@@ -296,7 +310,7 @@ double TestScaler::ComputeAvgSequencePSNR(FILE* input_file,
 
   int frame_count = 0;
   double avg_psnr = 0;
-  I420VideoFrame in_frame, out_frame;
+  VideoFrame in_frame, out_frame;
   const int half_width = (width + 1) / 2;
   in_frame.CreateEmptyFrame(width, height, width, half_width, half_width);
   out_frame.CreateEmptyFrame(width, height, width, half_width, half_width);
@@ -322,7 +336,7 @@ double TestScaler::ComputeAvgSequencePSNR(FILE* input_file,
   return avg_psnr;
 }
 
-// TODO (mikhal): Move part to a separate scale test.
+// TODO(mikhal): Move part to a separate scale test.
 void TestScaler::ScaleSequence(ScaleMethod method,
                    FILE* source_file, std::string out_name,
                    int src_width, int src_height,
@@ -337,8 +351,8 @@ void TestScaler::ScaleSequence(ScaleMethod method,
 
   rewind(source_file);
 
-  I420VideoFrame input_frame;
-  I420VideoFrame output_frame;
+  VideoFrame input_frame;
+  VideoFrame output_frame;
   int64_t start_clock, total_clock;
   total_clock = 0;
   int frame_count = 0;
@@ -363,7 +377,7 @@ void TestScaler::ScaleSequence(ScaleMethod method,
     start_clock = TickTime::MillisecondTimestamp();
     EXPECT_EQ(0, test_scaler_.Scale(input_frame, &output_frame));
     total_clock += TickTime::MillisecondTimestamp() - start_clock;
-    if (PrintI420VideoFrame(output_frame, output_file) < 0) {
+    if (PrintVideoFrame(output_frame, output_file) < 0) {
         return;
     }
     frame_count++;
diff --git a/media/webrtc/trunk/webrtc/common_video/libyuv/webrtc_libyuv.cc b/media/webrtc/trunk/webrtc/common_video/libyuv/webrtc_libyuv.cc
index b05d1315e2..96f0d28c1f 100644
--- a/media/webrtc/trunk/webrtc/common_video/libyuv/webrtc_libyuv.cc
+++ b/media/webrtc/trunk/webrtc/common_video/libyuv/webrtc_libyuv.cc
@@ -59,7 +59,7 @@ VideoType RawVideoTypeToCommonVideoVideoType(RawVideoType type) {
 
 int AlignInt(int value, int alignment) {
   assert(!((alignment - 1) & alignment));
-  return ((value + alignment - 1) & ~ (alignment - 1));
+  return ((value + alignment - 1) & ~(alignment - 1));
 }
 
 void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv) {
@@ -108,13 +108,12 @@ size_t CalcBufferSize(VideoType type, int width, int height) {
       break;
     default:
       assert(false);
-      buffer_size = SIZE_MAX;
       break;
   }
   return buffer_size;
 }
 
-int PrintI420VideoFrame(const I420VideoFrame& frame, FILE* file) {
+int PrintVideoFrame(const VideoFrame& frame, FILE* file) {
   if (file == NULL)
     return -1;
   if (frame.IsZeroSize())
@@ -131,12 +130,11 @@ int PrintI420VideoFrame(const I420VideoFrame& frame, FILE* file) {
        }
        plane_buffer += frame.stride(plane_type);
     }
- }
- return 0;
+  }
+  return 0;
 }
 
-int ExtractBuffer(const I420VideoFrame& input_frame,
-                  size_t size, uint8_t* buffer) {
+int ExtractBuffer(const VideoFrame& input_frame, size_t size, uint8_t* buffer) {
   assert(buffer);
   if (input_frame.IsZeroSize())
     return -1;
@@ -189,7 +187,7 @@ int ConvertRGB24ToARGB(const uint8_t* src_frame, uint8_t* dst_frame,
 }
 
 libyuv::RotationMode ConvertRotationMode(VideoRotation rotation) {
-  switch(rotation) {
+  switch (rotation) {
     case kVideoRotation_0:
       return libyuv::kRotate0;
     case kVideoRotation_90:
@@ -204,7 +202,7 @@ libyuv::RotationMode ConvertRotationMode(VideoRotation rotation) {
 }
 
 int ConvertVideoType(VideoType video_type) {
-  switch(video_type) {
+  switch (video_type) {
     case kUnknown:
       return libyuv::FOURCC_ANY;
     case  kI420:
@@ -249,14 +247,14 @@ int ConvertToI420(VideoType src_video_type,
                   int src_height,
                   size_t sample_size,
                   VideoRotation rotation,
-                  I420VideoFrame* dst_frame) {
+                  VideoFrame* dst_frame) {
   int dst_width = dst_frame->width();
   int dst_height = dst_frame->height();
   // LibYuv expects pre-rotation values for dst.
   // Stride values should correspond to the destination values.
   if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) {
     dst_width = dst_frame->height();
-    dst_height =dst_frame->width();
+    dst_height = dst_frame->width();
   }
 #ifdef WEBRTC_GONK
   if (src_video_type == kYV12) {
@@ -266,7 +264,7 @@ int ConvertToI420(VideoType src_video_type,
     int stride_uv = (((stride_y + 1) / 2) + 15) & ~0x0F;
     return libyuv::I420Rotate(src_frame,
                               stride_y,
-                              src_frame + (stride_y * src_height) + (stride_uv * ((src_height + 1) / 2)),
+                              src_frame + (stride_y * src_height) + (stride_uv * ((src_height + 1 / 2)),
                               stride_uv,
                               src_frame + (stride_y * src_height),
                               stride_uv,
@@ -294,8 +292,9 @@ int ConvertToI420(VideoType src_video_type,
                                ConvertVideoType(src_video_type));
 }
 
-int ConvertFromI420(const I420VideoFrame& src_frame,
-                    VideoType dst_video_type, int dst_sample_size,
+int ConvertFromI420(const VideoFrame& src_frame,
+                    VideoType dst_video_type,
+                    int dst_sample_size,
                     uint8_t* dst_frame) {
   return libyuv::ConvertFromI420(src_frame.buffer(kYPlane),
                                  src_frame.stride(kYPlane),
@@ -309,8 +308,9 @@ int ConvertFromI420(const I420VideoFrame& src_frame,
 }
 
 // TODO(mikhal): Create a designated VideoFrame for non I420.
-int ConvertFromYV12(const I420VideoFrame& src_frame,
-                    VideoType dst_video_type, int dst_sample_size,
+int ConvertFromYV12(const VideoFrame& src_frame,
+                    VideoType dst_video_type,
+                    int dst_sample_size,
                     uint8_t* dst_frame) {
   // YV12 = Y, V, U
   return libyuv::ConvertFromI420(src_frame.buffer(kYPlane),
@@ -325,8 +325,7 @@ int ConvertFromYV12(const I420VideoFrame& src_frame,
 }
 
 // Compute PSNR for an I420 frame (all planes)
-double I420PSNR(const I420VideoFrame* ref_frame,
-                const I420VideoFrame* test_frame) {
+double I420PSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
   if (!ref_frame || !test_frame)
     return -1;
   else if ((ref_frame->width() !=  test_frame->width()) ||
@@ -354,8 +353,7 @@ double I420PSNR(const I420VideoFrame* ref_frame,
 }
 
 // Compute SSIM for an I420 frame (all planes)
-double I420SSIM(const I420VideoFrame* ref_frame,
-                const I420VideoFrame* test_frame) {
+double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
   if (!ref_frame || !test_frame)
     return -1;
   else if ((ref_frame->width() !=  test_frame->width()) ||
diff --git a/media/webrtc/trunk/webrtc/common_video/plane.cc b/media/webrtc/trunk/webrtc/common_video/plane.cc
deleted file mode 100644
index 15ebddf440..0000000000
--- a/media/webrtc/trunk/webrtc/common_video/plane.cc
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_video/plane.h"
-
-#include <string.h>  // memcpy
-
-#include <algorithm>  // swap
-
-namespace webrtc {
-
-// Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
-static const int kBufferAlignment =  64;
-
-Plane::Plane()
-    : allocated_size_(0),
-      plane_size_(0),
-      stride_(0) {}
-
-Plane::~Plane() {}
-
-int Plane::CreateEmptyPlane(int allocated_size, int stride, int plane_size) {
-  if (allocated_size < 1 || stride < 1 || plane_size < 1)
-    return -1;
-  stride_ = stride;
-  if (MaybeResize(allocated_size) < 0)
-    return -1;
-  plane_size_ = plane_size;
-  return 0;
-}
-
-int Plane::MaybeResize(int new_size) {
-  if (new_size <= 0)
-    return -1;
-  if (new_size <= allocated_size_)
-    return 0;
-  rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> new_buffer(
-      static_cast<uint8_t*>(AlignedMalloc(new_size, kBufferAlignment)));
-
-  if (!new_buffer.get()) {
-    return -1;
-  }
-
-  if (buffer_.get()) {
-    memcpy(new_buffer.get(), buffer_.get(), plane_size_);
-  }
-  buffer_.reset(new_buffer.release());
-  allocated_size_ = new_size;
-  return 0;
-}
-
-int Plane::Copy(const Plane& plane) {
-  if (MaybeResize(plane.allocated_size_) < 0)
-    return -1;
-  if (plane.buffer_.get())
-    memcpy(buffer_.get(), plane.buffer_.get(), plane.plane_size_);
-  stride_ = plane.stride_;
-  plane_size_ = plane.plane_size_;
-  return 0;
-}
-
-int Plane::Copy(int size, int stride, const uint8_t* buffer) {
-  if (MaybeResize(size) < 0)
-    return -1;
-  memcpy(buffer_.get(), buffer, size);
-  plane_size_ = size;
-  stride_ = stride;
-  return 0;
-}
-
-void Plane::Swap(Plane& plane) {
-  std::swap(stride_, plane.stride_);
-  std::swap(allocated_size_, plane.allocated_size_);
-  std::swap(plane_size_, plane.plane_size_);
-  buffer_.swap(plane.buffer_);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/plane.h b/media/webrtc/trunk/webrtc/common_video/plane.h
deleted file mode 100644
index 66b8a5a082..0000000000
--- a/media/webrtc/trunk/webrtc/common_video/plane.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef COMMON_VIDEO_PLANE_H
-#define COMMON_VIDEO_PLANE_H
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Helper class for I420VideoFrame: Store plane data and perform basic plane
-// operations.
-class Plane {
- public:
-  Plane();
-  ~Plane();
-  // CreateEmptyPlane - set allocated size, actual plane size and stride:
-  // If current size is smaller than current size, then a buffer of sufficient
-  // size will be allocated.
-  // Return value: 0 on success ,-1 on error.
-  int CreateEmptyPlane(int allocated_size, int stride, int plane_size);
-
-  // Copy the entire plane data.
-  // Return value: 0 on success ,-1 on error.
-  int Copy(const Plane& plane);
-
-  // Copy buffer: If current size is smaller
-  // than current size, then a buffer of sufficient size will be allocated.
-  // Return value: 0 on success ,-1 on error.
-  int Copy(int size, int stride, const uint8_t* buffer);
-
-  // Swap plane data.
-  void Swap(Plane& plane);
-
-  // Get allocated size.
-  int allocated_size() const {return allocated_size_;}
-
-  // Set actual size.
-  void ResetSize() {plane_size_ = 0;}
-
-  // Return true is plane size is zero, false if not.
-  bool IsZeroSize() const {return plane_size_ == 0;}
-
-  // Get stride value.
-  int stride() const {return stride_;}
-
-  // Return data pointer.
-  const uint8_t* buffer() const {return buffer_.get();}
-  // Overloading with non-const.
-  uint8_t* buffer() {return buffer_.get();}
-
- private:
-  // Resize when needed: If current allocated size is less than new_size, buffer
-  // will be updated. Old data will be copied to new buffer.
-  // Return value: 0 on success ,-1 on error.
-  int MaybeResize(int new_size);
-
-  rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> buffer_;
-  int allocated_size_;
-  int plane_size_;
-  int stride_;
-};  // Plane
-
-}  // namespace webrtc
-
-#endif  // COMMON_VIDEO_PLANE_H
diff --git a/media/webrtc/trunk/webrtc/common_video/video_frame.cc b/media/webrtc/trunk/webrtc/common_video/video_frame.cc
new file mode 100644
index 0000000000..8ccd821d09
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/common_video/video_frame.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_frame.h"
+
+#include <string.h>
+
+#include <algorithm>  // swap
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+bool EqualPlane(const uint8_t* data1,
+                const uint8_t* data2,
+                int stride,
+                int width,
+                int height) {
+  for (int y = 0; y < height; ++y) {
+    if (memcmp(data1, data2, width) != 0)
+      return false;
+    data1 += stride;
+    data2 += stride;
+  }
+  return true;
+}
+
+int ExpectedSize(int plane_stride, int image_height, PlaneType type) {
+  if (type == kYPlane)
+    return plane_stride * image_height;
+  return plane_stride * ((image_height + 1) / 2);
+}
+
+VideoFrame::VideoFrame() {
+  // Intentionally using Reset instead of initializer list so that any missed
+  // fields in Reset will be caught by memory checkers.
+  Reset();
+}
+
+VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
+                       uint32_t timestamp,
+                       int64_t render_time_ms,
+                       VideoRotation rotation)
+    : video_frame_buffer_(buffer),
+      timestamp_(timestamp),
+      ntp_time_ms_(0),
+      render_time_ms_(render_time_ms),
+      rotation_(rotation) {
+}
+
+int VideoFrame::CreateEmptyFrame(int width,
+                                 int height,
+                                 int stride_y,
+                                 int stride_u,
+                                 int stride_v) {
+  const int half_width = (width + 1) / 2;
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+  RTC_DCHECK_GE(stride_y, width);
+  RTC_DCHECK_GE(stride_u, half_width);
+  RTC_DCHECK_GE(stride_v, half_width);
+
+  // Creating empty frame - reset all values.
+  timestamp_ = 0;
+  ntp_time_ms_ = 0;
+  render_time_ms_ = 0;
+  rotation_ = kVideoRotation_0;
+
+  // Check if it's safe to reuse allocation.
+  if (video_frame_buffer_ && video_frame_buffer_->HasOneRef() &&
+      !video_frame_buffer_->native_handle() &&
+      width == video_frame_buffer_->width() &&
+      height == video_frame_buffer_->height() && stride_y == stride(kYPlane) &&
+      stride_u == stride(kUPlane) && stride_v == stride(kVPlane)) {
+    return 0;
+  }
+
+  // Need to allocate new buffer.
+  video_frame_buffer_ = new rtc::RefCountedObject<I420Buffer>(
+      width, height, stride_y, stride_u, stride_v);
+  return 0;
+}
+
+int VideoFrame::CreateFrame(const uint8_t* buffer_y,
+                            const uint8_t* buffer_u,
+                            const uint8_t* buffer_v,
+                            int width,
+                            int height,
+                            int stride_y,
+                            int stride_u,
+                            int stride_v) {
+  return CreateFrame(buffer_y, buffer_u, buffer_v, width, height, stride_y,
+                     stride_u, stride_v, kVideoRotation_0);
+}
+
+int VideoFrame::CreateFrame(const uint8_t* buffer_y,
+                            const uint8_t* buffer_u,
+                            const uint8_t* buffer_v,
+                            int width,
+                            int height,
+                            int stride_y,
+                            int stride_u,
+                            int stride_v,
+                            VideoRotation rotation) {
+  const int half_height = (height + 1) / 2;
+  const int expected_size_y = height * stride_y;
+  const int expected_size_u = half_height * stride_u;
+  const int expected_size_v = half_height * stride_v;
+  CreateEmptyFrame(width, height, stride_y, stride_u, stride_v);
+  memcpy(buffer(kYPlane), buffer_y, expected_size_y);
+  memcpy(buffer(kUPlane), buffer_u, expected_size_u);
+  memcpy(buffer(kVPlane), buffer_v, expected_size_v);
+  rotation_ = rotation;
+  return 0;
+}
+
+int VideoFrame::CreateFrame(const uint8_t* buffer,
+                            int width,
+                            int height,
+                            VideoRotation rotation) {
+  const int stride_y = width;
+  const int stride_uv = (width + 1) / 2;
+
+  const uint8_t* buffer_y = buffer;
+  const uint8_t* buffer_u = buffer_y + stride_y * height;
+  const uint8_t* buffer_v = buffer_u + stride_uv * ((height + 1) / 2);
+  return CreateFrame(buffer_y, buffer_u, buffer_v, width, height, stride_y,
+                     stride_uv, stride_uv, rotation);
+}
+
+int VideoFrame::CopyFrame(const VideoFrame& videoFrame) {
+  if (videoFrame.IsZeroSize()) {
+    video_frame_buffer_ = nullptr;
+  } else if (videoFrame.native_handle()) {
+    video_frame_buffer_ = videoFrame.video_frame_buffer();
+  } else {
+    CreateFrame(videoFrame.buffer(kYPlane), videoFrame.buffer(kUPlane),
+                videoFrame.buffer(kVPlane), videoFrame.width(),
+                videoFrame.height(), videoFrame.stride(kYPlane),
+                videoFrame.stride(kUPlane), videoFrame.stride(kVPlane));
+  }
+
+  timestamp_ = videoFrame.timestamp_;
+  ntp_time_ms_ = videoFrame.ntp_time_ms_;
+  render_time_ms_ = videoFrame.render_time_ms_;
+  rotation_ = videoFrame.rotation_;
+  return 0;
+}
+
+void VideoFrame::ShallowCopy(const VideoFrame& videoFrame) {
+  video_frame_buffer_ = videoFrame.video_frame_buffer();
+  timestamp_ = videoFrame.timestamp_;
+  ntp_time_ms_ = videoFrame.ntp_time_ms_;
+  render_time_ms_ = videoFrame.render_time_ms_;
+  rotation_ = videoFrame.rotation_;
+}
+
+void VideoFrame::Reset() {
+  video_frame_buffer_ = nullptr;
+  timestamp_ = 0;
+  ntp_time_ms_ = 0;
+  render_time_ms_ = 0;
+  rotation_ = kVideoRotation_0;
+}
+
+uint8_t* VideoFrame::buffer(PlaneType type) {
+  return video_frame_buffer_ ? video_frame_buffer_->MutableData(type)
+                             : nullptr;
+}
+
+const uint8_t* VideoFrame::buffer(PlaneType type) const {
+  return video_frame_buffer_ ? video_frame_buffer_->data(type) : nullptr;
+}
+
+int VideoFrame::allocated_size(PlaneType type) const {
+  const int plane_height = (type == kYPlane) ? height() : (height() + 1) / 2;
+  return plane_height * stride(type);
+}
+
+int VideoFrame::stride(PlaneType type) const {
+  return video_frame_buffer_ ? video_frame_buffer_->stride(type) : 0;
+}
+
+int VideoFrame::width() const {
+  return video_frame_buffer_ ? video_frame_buffer_->width() : 0;
+}
+
+int VideoFrame::height() const {
+  return video_frame_buffer_ ? video_frame_buffer_->height() : 0;
+}
+
+bool VideoFrame::IsZeroSize() const {
+  return !video_frame_buffer_;
+}
+
+void* VideoFrame::native_handle() const {
+  return video_frame_buffer_ ? video_frame_buffer_->native_handle() : nullptr;
+}
+
+rtc::scoped_refptr<VideoFrameBuffer> VideoFrame::video_frame_buffer() const {
+  return video_frame_buffer_;
+}
+
+void VideoFrame::set_video_frame_buffer(
+    const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer) {
+  video_frame_buffer_ = buffer;
+}
+
+VideoFrame VideoFrame::ConvertNativeToI420Frame() const {
+  RTC_DCHECK(native_handle());
+  VideoFrame frame;
+  frame.ShallowCopy(*this);
+  frame.set_video_frame_buffer(video_frame_buffer_->NativeToI420Buffer());
+  return frame;
+}
+
+bool VideoFrame::EqualsFrame(const VideoFrame& frame) const {
+  if (width() != frame.width() || height() != frame.height() ||
+      stride(kYPlane) != frame.stride(kYPlane) ||
+      stride(kUPlane) != frame.stride(kUPlane) ||
+      stride(kVPlane) != frame.stride(kVPlane) ||
+      timestamp() != frame.timestamp() ||
+      ntp_time_ms() != frame.ntp_time_ms() ||
+      render_time_ms() != frame.render_time_ms()) {
+    return false;
+  }
+  const int half_width = (width() + 1) / 2;
+  const int half_height = (height() + 1) / 2;
+  return EqualPlane(buffer(kYPlane), frame.buffer(kYPlane),
+                    stride(kYPlane), width(), height()) &&
+         EqualPlane(buffer(kUPlane), frame.buffer(kUPlane),
+                    stride(kUPlane), half_width, half_height) &&
+         EqualPlane(buffer(kVPlane), frame.buffer(kVPlane),
+                    stride(kVPlane), half_width, half_height);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/common_video/video_frame_buffer.cc b/media/webrtc/trunk/webrtc/common_video/video_frame_buffer.cc
index cca685b923..492bc49587 100644
--- a/media/webrtc/trunk/webrtc/common_video/video_frame_buffer.cc
+++ b/media/webrtc/trunk/webrtc/common_video/video_frame_buffer.cc
@@ -8,15 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/keep_ref_until_done.h"
 
 // Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
 static const int kBufferAlignment = 64;
 
 namespace webrtc {
 
+uint8_t* VideoFrameBuffer::MutableData(PlaneType type) {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
 VideoFrameBuffer::~VideoFrameBuffer() {}
 
 I420Buffer::I420Buffer(int width, int height)
@@ -36,11 +42,11 @@ I420Buffer::I420Buffer(int width,
       data_(static_cast<uint8_t*>(AlignedMalloc(
           stride_y * height + (stride_u + stride_v) * ((height + 1) / 2),
           kBufferAlignment))) {
-  DCHECK_GT(width, 0);
-  DCHECK_GT(height, 0);
-  DCHECK_GE(stride_y, width);
-  DCHECK_GE(stride_u, (width + 1) / 2);
-  DCHECK_GE(stride_v, (width + 1) / 2);
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+  RTC_DCHECK_GE(stride_y, width);
+  RTC_DCHECK_GE(stride_u, (width + 1) / 2);
+  RTC_DCHECK_GE(stride_v, (width + 1) / 2);
 }
 
 I420Buffer::~I420Buffer() {
@@ -69,8 +75,8 @@ const uint8_t* I420Buffer::data(PlaneType type) const {
   }
 }
 
-uint8_t* I420Buffer::data(PlaneType type) {
-  DCHECK(HasOneRef());
+uint8_t* I420Buffer::MutableData(PlaneType type) {
+  RTC_DCHECK(HasOneRef());
   return const_cast<uint8_t*>(
       static_cast<const VideoFrameBuffer*>(this)->data(type));
 }
@@ -89,54 +95,47 @@ int I420Buffer::stride(PlaneType type) const {
   }
 }
 
-rtc::scoped_refptr<NativeHandle> I420Buffer::native_handle() const {
+void* I420Buffer::native_handle() const {
   return nullptr;
 }
 
-TextureBuffer::TextureBuffer(
-    const rtc::scoped_refptr<NativeHandle>& native_handle,
-    int width,
-    int height)
+rtc::scoped_refptr<VideoFrameBuffer> I420Buffer::NativeToI420Buffer() {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+NativeHandleBuffer::NativeHandleBuffer(void* native_handle,
+                                       int width,
+                                       int height)
     : native_handle_(native_handle), width_(width), height_(height) {
-  DCHECK(native_handle.get());
-  DCHECK_GT(width, 0);
-  DCHECK_GT(height, 0);
+  RTC_DCHECK(native_handle != nullptr);
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
 }
 
-TextureBuffer::~TextureBuffer() {
-}
-
-int TextureBuffer::width() const {
+int NativeHandleBuffer::width() const {
   return width_;
 }
 
-int TextureBuffer::height() const {
+int NativeHandleBuffer::height() const {
   return height_;
 }
 
-const uint8_t* TextureBuffer::data(PlaneType type) const {
+const uint8_t* NativeHandleBuffer::data(PlaneType type) const {
   RTC_NOTREACHED();  // Should not be called.
   return nullptr;
 }
 
-uint8_t* TextureBuffer::data(PlaneType type) {
-  RTC_NOTREACHED();  // Should not be called.
-  return nullptr;
-}
-
-int TextureBuffer::stride(PlaneType type) const {
+int NativeHandleBuffer::stride(PlaneType type) const {
   RTC_NOTREACHED();  // Should not be called.
   return 0;
 }
 
-rtc::scoped_refptr<NativeHandle> TextureBuffer::native_handle() const {
+void* NativeHandleBuffer::native_handle() const {
   return native_handle_;
 }
 
-
-WrappedI420Buffer::WrappedI420Buffer(int desired_width,
-                                     int desired_height,
-                                     int width,
+WrappedI420Buffer::WrappedI420Buffer(int width,
                                      int height,
                                      const uint8_t* y_plane,
                                      int y_stride,
@@ -145,31 +144,21 @@ WrappedI420Buffer::WrappedI420Buffer(int desired_width,
                                      const uint8_t* v_plane,
                                      int v_stride,
                                      const rtc::Callback0<void>& no_longer_used)
-  :  width_(desired_width),
-     height_(desired_height),
-     y_plane_(y_plane),
-     u_plane_(u_plane),
-     v_plane_(v_plane),
-     y_stride_(y_stride),
-     u_stride_(u_stride),
-     v_stride_(v_stride),
-     no_longer_used_cb_(no_longer_used) {
-  CHECK(width >= desired_width && height >= desired_height);
-
-  // Center crop to |desired_width| x |desired_height|.
-  // Make sure offset is even so that u/v plane becomes aligned.
-  const int offset_x = ((width - desired_width) / 2) & ~1;
-  const int offset_y = ((height - desired_height) / 2) & ~1;
-  y_plane_ += y_stride_ * offset_y + offset_x;
-  u_plane_ += u_stride_ * (offset_y / 2) + (offset_x / 2);
-  v_plane_ += v_stride_ * (offset_y / 2) + (offset_x / 2);
+    : width_(width),
+      height_(height),
+      y_plane_(y_plane),
+      u_plane_(u_plane),
+      v_plane_(v_plane),
+      y_stride_(y_stride),
+      u_stride_(u_stride),
+      v_stride_(v_stride),
+      no_longer_used_cb_(no_longer_used) {
 }
 
 WrappedI420Buffer::~WrappedI420Buffer() {
   no_longer_used_cb_();
 }
 
-
 int WrappedI420Buffer::width() const {
   return width_;
 }
@@ -192,11 +181,6 @@ const uint8_t* WrappedI420Buffer::data(PlaneType type) const {
   }
 }
 
-uint8_t* WrappedI420Buffer::data(PlaneType type) {
-  RTC_NOTREACHED();
-  return nullptr;
-}
-
 int WrappedI420Buffer::stride(PlaneType type) const {
   switch (type) {
     case kYPlane:
@@ -211,8 +195,44 @@ int WrappedI420Buffer::stride(PlaneType type) const {
   }
 }
 
-rtc::scoped_refptr<NativeHandle> WrappedI420Buffer::native_handle() const {
+void* WrappedI420Buffer::native_handle() const {
   return nullptr;
 }
 
+rtc::scoped_refptr<VideoFrameBuffer> WrappedI420Buffer::NativeToI420Buffer() {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+rtc::scoped_refptr<VideoFrameBuffer> ShallowCenterCrop(
+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
+    int cropped_width,
+    int cropped_height) {
+  RTC_CHECK(buffer->native_handle() == nullptr);
+  RTC_CHECK_LE(cropped_width, buffer->width());
+  RTC_CHECK_LE(cropped_height, buffer->height());
+  if (buffer->width() == cropped_width && buffer->height() == cropped_height)
+    return buffer;
+
+  // Center crop to |cropped_width| x |cropped_height|.
+  // Make sure offset is even so that u/v plane becomes aligned.
+  const int uv_offset_x = (buffer->width() - cropped_width) / 4;
+  const int uv_offset_y = (buffer->height() - cropped_height) / 4;
+  const int offset_x = uv_offset_x * 2;
+  const int offset_y = uv_offset_y * 2;
+
+  const uint8_t* y_plane = buffer->data(kYPlane) +
+                           buffer->stride(kYPlane) * offset_y + offset_x;
+  const uint8_t* u_plane = buffer->data(kUPlane) +
+                           buffer->stride(kUPlane) * uv_offset_y + uv_offset_x;
+  const uint8_t* v_plane = buffer->data(kVPlane) +
+                           buffer->stride(kVPlane) * uv_offset_y + uv_offset_x;
+  return new rtc::RefCountedObject<WrappedI420Buffer>(
+      cropped_width, cropped_height,
+      y_plane, buffer->stride(kYPlane),
+      u_plane, buffer->stride(kUPlane),
+      v_plane, buffer->stride(kVPlane),
+      rtc::KeepRefUntilDone(buffer));
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.cc b/media/webrtc/trunk/webrtc/common_video/video_render_frames.cc
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.cc
rename to media/webrtc/trunk/webrtc/common_video/video_render_frames.cc
index 107625c88b..8b447cb10f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.cc
+++ b/media/webrtc/trunk/webrtc/common_video/video_render_frames.cc
@@ -8,25 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_render/video_render_frames.h"
+#include "webrtc/common_video/video_render_frames.h"
 
 #include <assert.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
 const uint32_t KEventMaxWaitTimeMs = 200;
 const uint32_t kMinRenderDelayMs = 10;
-const uint32_t kMaxRenderDelayMs= 500;
+const uint32_t kMaxRenderDelayMs = 500;
 
 VideoRenderFrames::VideoRenderFrames()
     : render_delay_ms_(10) {
 }
 
-int32_t VideoRenderFrames::AddFrame(const I420VideoFrame& new_frame) {
+int32_t VideoRenderFrames::AddFrame(const VideoFrame& new_frame) {
   const int64_t time_now = TickTime::MillisecondTimestamp();
 
   // Drop old frames only when there are other frames in the queue, otherwise, a
@@ -53,8 +53,8 @@ int32_t VideoRenderFrames::AddFrame(const I420VideoFrame& new_frame) {
   return static_cast<int32_t>(incoming_frames_.size());
 }
 
-I420VideoFrame VideoRenderFrames::FrameToRender() {
-  I420VideoFrame render_frame;
+VideoFrame VideoRenderFrames::FrameToRender() {
+  VideoFrame render_frame;
   // Get the newest frame that can be released for rendering.
   while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
     render_frame = incoming_frames_.front();
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.h b/media/webrtc/trunk/webrtc/common_video/video_render_frames.h
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.h
rename to media/webrtc/trunk/webrtc/common_video/video_render_frames.h
index 4094074c69..450c1f2215 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/video_render_frames.h
+++ b/media/webrtc/trunk/webrtc/common_video/video_render_frames.h
@@ -8,12 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_  // NOLINT
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_  // NOLINT
+#ifndef WEBRTC_COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
+#define WEBRTC_COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
+
+#include <stdint.h>
 
 #include <list>
 
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -23,10 +25,10 @@ class VideoRenderFrames {
   VideoRenderFrames();
 
   // Add a frame to the render queue
-  int32_t AddFrame(const I420VideoFrame& new_frame);
+  int32_t AddFrame(const VideoFrame& new_frame);
 
   // Get a frame for rendering, or a zero-size frame if it's not time to render.
-  I420VideoFrame FrameToRender();
+  VideoFrame FrameToRender();
 
   // Releases all frames
   int32_t ReleaseAllFrames();
@@ -46,7 +48,7 @@ class VideoRenderFrames {
   enum { KFutureRenderTimestampMS = 10000 };
 
   // Sorted list with framed to be rendered, oldest first.
-  std::list<I420VideoFrame> incoming_frames_;
+  std::list<VideoFrame> incoming_frames_;
 
   // Estimated delay from a frame is released until it's rendered.
   uint32_t render_delay_ms_;
@@ -54,4 +56,4 @@ class VideoRenderFrames {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_  // NOLINT
+#endif  // WEBRTC_COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
diff --git a/media/webrtc/trunk/webrtc/config.cc b/media/webrtc/trunk/webrtc/config.cc
index 7b75a68de8..bfd7a202d0 100644
--- a/media/webrtc/trunk/webrtc/config.cc
+++ b/media/webrtc/trunk/webrtc/config.cc
@@ -29,6 +29,33 @@ std::string RtpExtension::ToString() const {
   return ss.str();
 }
 
+const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
+const char* RtpExtension::kAbsSendTime =
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
+const char* RtpExtension::kAudioLevel =
+    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+const char* RtpExtension::kTransportSequenceNumber =
+    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
+const char* RtpExtension::kRtpStreamId =
+  "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+
+bool RtpExtension::IsSupportedForAudio(const std::string& name) {
+  return name == webrtc::RtpExtension::kAbsSendTime ||
+         name == webrtc::RtpExtension::kAudioLevel ||
+         name == webrtc::RtpExtension::kTransportSequenceNumber ||
+         name == webrtc::RtpExtension::kRtpStreamId;
+}
+
+bool RtpExtension::IsSupportedForVideo(const std::string& name) {
+  return name == webrtc::RtpExtension::kTOffset ||
+         name == webrtc::RtpExtension::kAbsSendTime ||
+         name == webrtc::RtpExtension::kVideoRotation ||
+         name == webrtc::RtpExtension::kTransportSequenceNumber ||
+         name == webrtc::RtpExtension::kRtpStreamId;
+
+}
+
 VideoStream::VideoStream()
     : width(0),
       height(0),
@@ -63,9 +90,10 @@ std::string VideoStream::ToString() const {
 }
 
 VideoEncoderConfig::VideoEncoderConfig()
-    : content_type(kRealtimeVideo),
+    : content_type(ContentType::kRealtimeVideo),
       encoder_specific_settings(NULL),
-      min_transmit_bitrate_bps(0) {}
+      min_transmit_bitrate_bps(0) {
+}
 
 VideoEncoderConfig::~VideoEncoderConfig() = default;
 
@@ -81,10 +109,10 @@ std::string VideoEncoderConfig::ToString() const {
   ss << ']';
   ss << ", content_type: ";
   switch (content_type) {
-    case kRealtimeVideo:
+    case ContentType::kRealtimeVideo:
       ss << "kRealtimeVideo";
       break;
-    case kScreenshare:
+    case ContentType::kScreen:
       ss << "kScreenshare";
       break;
   }
diff --git a/media/webrtc/trunk/webrtc/config.h b/media/webrtc/trunk/webrtc/config.h
index 9746494b37..290b552caf 100644
--- a/media/webrtc/trunk/webrtc/config.h
+++ b/media/webrtc/trunk/webrtc/config.h
@@ -15,8 +15,10 @@
 
 #include <string>
 #include <vector>
+#include <string.h>
 #include <algorithm>
 
+#include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
 
@@ -35,24 +37,36 @@ struct NackConfig {
 // Settings for forward error correction, see RFC 5109 for details. Set the
 // payload types to '-1' to disable.
 struct FecConfig {
-  FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
+  FecConfig()
+      : ulpfec_payload_type(-1),
+        red_payload_type(-1),
+        red_rtx_payload_type(-1) {}
   std::string ToString() const;
   // Payload type used for ULPFEC packets.
   int ulpfec_payload_type;
 
   // Payload type used for RED packets.
   int red_payload_type;
+
+  // RTX payload type for RED payload.
+  int red_rtx_payload_type;
 };
 
-// RTP header extension to use for the video stream, see RFC 5285.
+// RTP header extension, see RFC 5285.
 struct RtpExtension {
   RtpExtension(const std::string& name, int id) : name(name), id(id) {}
   std::string ToString() const;
-  static bool IsSupported(const std::string& name);
+  bool operator==(const RtpExtension& rhs) const {
+    return name == rhs.name && id == rhs.id;
+  }
+  static bool IsSupportedForAudio(const std::string& name);
+  static bool IsSupportedForVideo(const std::string& name);
 
   static const char* kTOffset;
   static const char* kAbsSendTime;
   static const char* kVideoRotation;
+  static const char* kAudioLevel;
+  static const char* kTransportSequenceNumber;
   static const char* kRtpStreamId;
   std::string name;
   int id;
@@ -79,7 +93,7 @@ struct VideoStream {
     return std::string(rid);
   }
 
-  void SetRid(const std::string& aRid) {
+  void SetRid(const std::string & aRid) {
     static_assert(sizeof(rid) > kRIDSize,
       "mRid must be large enought to hold a RID + null termination");
     strncpy(&rid[0], aRid.c_str(), std::min((size_t)kRIDSize, aRid.length()));
@@ -99,9 +113,9 @@ struct VideoStream {
 };
 
 struct VideoEncoderConfig {
-  enum ContentType {
+  enum class ContentType {
     kRealtimeVideo,
-    kScreenshare,
+    kScreen,
   };
 
   VideoEncoderConfig();
@@ -109,8 +123,10 @@ struct VideoEncoderConfig {
   std::string ToString() const;
 
   std::vector<VideoStream> streams;
+  std::vector<SpatialLayer> spatial_layers;
   ContentType content_type;
   void* encoder_specific_settings;
+  unsigned char resolution_divisor;
 
   // Padding will be used up to this bitrate regardless of the bitrate produced
   // by the encoder. Padding above what's actually produced by the encoder helps
@@ -119,6 +135,35 @@ struct VideoEncoderConfig {
   int min_transmit_bitrate_bps;
 };
 
+// Controls the capacity of the packet buffer in NetEq. The capacity is the
+// maximum number of packets that the buffer can contain. If the limit is
+// exceeded, the buffer will be flushed. The capacity does not affect the actual
+// audio delay in the general case, since this is governed by the target buffer
+// level (calculated from the jitter profile). It is only in the rare case of
+// severe network freezes that a higher capacity will lead to a (transient)
+// increase in audio delay.
+struct NetEqCapacityConfig {
+  NetEqCapacityConfig() : enabled(false), capacity(0) {}
+  explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kNetEqCapacityConfig;
+  bool enabled;
+  int capacity;
+};
+
+struct NetEqFastAccelerate {
+  NetEqFastAccelerate() : enabled(false) {}
+  explicit NetEqFastAccelerate(bool value) : enabled(value) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kNetEqFastAccelerate;
+  bool enabled;
+};
+
+struct VoicePacing {
+  VoicePacing() : enabled(false) {}
+  explicit VoicePacing(bool value) : enabled(value) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
+  bool enabled;
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_CONFIG_H_
diff --git a/media/webrtc/trunk/webrtc/engine_configurations.h b/media/webrtc/trunk/webrtc/engine_configurations.h
index edd819b394..cbd42cff79 100644
--- a/media/webrtc/trunk/webrtc/engine_configurations.h
+++ b/media/webrtc/trunk/webrtc/engine_configurations.h
@@ -13,49 +13,6 @@
 
 #include "webrtc/typedefs.h"
 
-// ============================================================================
-//                              Voice and Video
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-//  [Voice] Codec settings
-// ----------------------------------------------------------------------------
-
-// iSAC and G722 are not included in the Mozilla build, but in all other builds.
-#ifndef WEBRTC_MOZILLA_BUILD
-#ifdef WEBRTC_ARCH_ARM
-#define WEBRTC_CODEC_ISACFX  // Fix-point iSAC implementation.
-#else
-#define WEBRTC_CODEC_ISAC  // Floating-point iSAC implementation (default).
-#endif  // WEBRTC_ARCH_ARM
-#define WEBRTC_CODEC_G722
-#endif  // !WEBRTC_MOZILLA_BUILD
-
-// AVT is included in all builds, along with G.711, NetEQ and CNG
-// (which are mandatory and don't have any defines).
-#define WEBRTC_CODEC_AVT
-
-// PCM16 is useful for testing and incurs only a small binary size cost.
-#ifndef WEBRTC_CODEC_PCM16
-#define WEBRTC_CODEC_PCM16
-#endif
-
-// iLBC and Redundancy coding are excluded from Chromium and Mozilla
-// builds to reduce binary size.
-#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_MOZILLA_BUILD)
-#define WEBRTC_CODEC_ILBC
-#define WEBRTC_CODEC_RED
-#endif  // !WEBRTC_CHROMIUM_BUILD && !WEBRTC_MOZILLA_BUILD
-
-// ----------------------------------------------------------------------------
-//  [Video] Codec settings
-// ----------------------------------------------------------------------------
-
-#define VIDEOCODEC_I420
-#define VIDEOCODEC_VP8
-#define VIDEOCODEC_VP9
-#define VIDEOCODEC_H264
-
 // ============================================================================
 //                                 VoiceEngine
 // ============================================================================
@@ -88,27 +45,6 @@
 #define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
 #define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
 
-// ============================================================================
-//                                 VideoEngine
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-//  Settings for special VideoEngine configurations
-// ----------------------------------------------------------------------------
-// ----------------------------------------------------------------------------
-//  VideoEngine sub-API:s
-// ----------------------------------------------------------------------------
-
-#define WEBRTC_VIDEO_ENGINE_CAPTURE_API
-#define WEBRTC_VIDEO_ENGINE_CODEC_API
-#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-#define WEBRTC_VIDEO_ENGINE_RENDER_API
-#define WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-#define WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-
-// Now handled by gyp:
-// WEBRTC_VIDEO_ENGINE_FILE_API
-
 // ============================================================================
 //                       Platform specific configurations
 // ============================================================================
@@ -138,10 +74,4 @@
 #define EAGL_RENDERING
 #endif
 
-// ----------------------------------------------------------------------------
-//  Deprecated
-// ----------------------------------------------------------------------------
-
-// #define WEBRTC_DTMF_DETECTION
-
 #endif  // WEBRTC_ENGINE_CONFIGURATIONS_H_
diff --git a/media/webrtc/trunk/webrtc/experiments.h b/media/webrtc/trunk/webrtc/experiments.h
deleted file mode 100644
index ec871f2aa3..0000000000
--- a/media/webrtc/trunk/webrtc/experiments.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_EXPERIMENTS_H_
-#define WEBRTC_EXPERIMENTS_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-struct RemoteBitrateEstimatorMinRate {
-  RemoteBitrateEstimatorMinRate() : min_rate(30000) {}
-  RemoteBitrateEstimatorMinRate(uint32_t min_rate) : min_rate(min_rate) {}
-
-  uint32_t min_rate;
-};
-
-struct AimdRemoteRateControl {
-  AimdRemoteRateControl() : enabled(false) {}
-  explicit AimdRemoteRateControl(bool set_enabled)
-    : enabled(set_enabled) {}
-  virtual ~AimdRemoteRateControl() {}
-
-  const bool enabled;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_EXPERIMENTS_H_
diff --git a/media/webrtc/trunk/webrtc/frame_callback.h b/media/webrtc/trunk/webrtc/frame_callback.h
index 1cd077a7ca..b7f2210334 100644
--- a/media/webrtc/trunk/webrtc/frame_callback.h
+++ b/media/webrtc/trunk/webrtc/frame_callback.h
@@ -17,11 +17,11 @@
 
 namespace webrtc {
 
-class I420VideoFrame;
+class VideoFrame;
 
 struct EncodedFrame {
  public:
-  EncodedFrame() : data_(NULL), length_(0), frame_type_(kFrameEmpty) {}
+  EncodedFrame() : data_(NULL), length_(0), frame_type_(kEmptyFrame) {}
   EncodedFrame(const uint8_t* data, size_t length, FrameType frame_type)
     : data_(data), length_(length), frame_type_(frame_type) {}
 
@@ -34,7 +34,7 @@ class I420FrameCallback {
  public:
   // This function is called with a I420 frame allowing the user to modify the
   // frame content.
-  virtual void FrameCallback(I420VideoFrame* video_frame) = 0;
+  virtual void FrameCallback(VideoFrame* video_frame) = 0;
 
  protected:
   virtual ~I420FrameCallback() {}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate b/media/webrtc/trunk/webrtc/modules/audio_codec_speed_tests.isolate
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate
rename to media/webrtc/trunk/webrtc/modules/audio_codec_speed_tests.isolate
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn b/media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn
index d18ed66777..000dd394df 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn
@@ -9,37 +9,64 @@
 import("//build/config/arm.gni")
 import("../../build/webrtc.gni")
 
+source_set("rent_a_codec") {
+  sources = [
+    "acm2/acm_codec_database.cc",
+    "acm2/acm_codec_database.h",
+    "acm2/rent_a_codec.cc",
+    "acm2/rent_a_codec.h",
+  ]
+  configs += [ "../..:common_config" ]
+  public_configs = [ "../..:common_inherited_config" ]
+  deps = [
+    "../..:webrtc_common",
+  ]
+
+  defines = []
+  if (rtc_include_opus) {
+    defines += [ "WEBRTC_CODEC_OPUS" ]
+  }
+  if (!build_with_mozilla) {
+    if (current_cpu == "arm") {
+      defines += [ "WEBRTC_CODEC_ISACFX" ]
+    } else {
+      defines += [ "WEBRTC_CODEC_ISAC" ]
+    }
+    defines += [ "WEBRTC_CODEC_G722" ]
+  }
+  if (!build_with_mozilla && !build_with_chromium) {
+    defines += [
+      "WEBRTC_CODEC_ILBC",
+      "WEBRTC_CODEC_RED",
+    ]
+  }
+}
+
 config("audio_coding_config") {
   include_dirs = [
-    "main/interface",
-    "../interface",
+    "include",
+    "../include",
   ]
 }
 
 source_set("audio_coding") {
   sources = [
-    "main/acm2/acm_codec_database.cc",
-    "main/acm2/acm_codec_database.h",
-    "main/acm2/acm_common_defs.h",
-    "main/acm2/acm_generic_codec.cc",
-    "main/acm2/acm_generic_codec.h",
-    "main/acm2/acm_receiver.cc",
-    "main/acm2/acm_receiver.h",
-    "main/acm2/acm_resampler.cc",
-    "main/acm2/acm_resampler.h",
-    "main/acm2/audio_coding_module.cc",
-    "main/acm2/audio_coding_module_impl.cc",
-    "main/acm2/audio_coding_module_impl.h",
-    "main/acm2/call_statistics.cc",
-    "main/acm2/call_statistics.h",
-    "main/acm2/codec_manager.cc",
-    "main/acm2/codec_manager.h",
-    "main/acm2/initial_delay_manager.cc",
-    "main/acm2/initial_delay_manager.h",
-    "main/acm2/nack.cc",
-    "main/acm2/nack.h",
-    "main/interface/audio_coding_module.h",
-    "main/interface/audio_coding_module_typedefs.h",
+    "acm2/acm_common_defs.h",
+    "acm2/acm_receiver.cc",
+    "acm2/acm_receiver.h",
+    "acm2/acm_resampler.cc",
+    "acm2/acm_resampler.h",
+    "acm2/audio_coding_module.cc",
+    "acm2/audio_coding_module_impl.cc",
+    "acm2/audio_coding_module_impl.h",
+    "acm2/call_statistics.cc",
+    "acm2/call_statistics.h",
+    "acm2/codec_manager.cc",
+    "acm2/codec_manager.h",
+    "acm2/initial_delay_manager.cc",
+    "acm2/initial_delay_manager.h",
+    "include/audio_coding_module.h",
+    "include/audio_coding_module_typedefs.h",
   ]
 
   defines = []
@@ -51,12 +78,6 @@ source_set("audio_coding") {
     ":audio_coding_config",
   ]
 
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-
   if (is_win) {
     cflags = [
       # TODO(kjellander): Bug 261: fix this warning.
@@ -64,16 +85,19 @@ source_set("audio_coding") {
     ]
   }
 
+  if (is_clang) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+
   deps = [
     ":cng",
     ":g711",
-    ":g722",
-    ":ilbc",
-    ":isac",
-    ":isacfix",
     ":neteq",
     ":pcm16b",
-    ":red",
+    ":rent_a_codec",
+    "../..:rtc_event_log",
     "../..:webrtc_common",
     "../../common_audio",
     "../../system_wrappers",
@@ -83,6 +107,27 @@ source_set("audio_coding") {
     defines += [ "WEBRTC_CODEC_OPUS" ]
     deps += [ ":webrtc_opus" ]
   }
+  if (!build_with_mozilla) {
+    if (current_cpu == "arm") {
+      defines += [ "WEBRTC_CODEC_ISACFX" ]
+      deps += [ ":isac_fix" ]
+    } else {
+      defines += [ "WEBRTC_CODEC_ISAC" ]
+      deps += [ ":isac" ]
+    }
+    defines += [ "WEBRTC_CODEC_G722" ]
+    deps += [ ":g722" ]
+  }
+  if (!build_with_mozilla && !build_with_chromium) {
+    defines += [
+      "WEBRTC_CODEC_ILBC",
+      "WEBRTC_CODEC_RED",
+    ]
+    deps += [
+      ":ilbc",
+      ":red",
+    ]
+  }
 }
 
 source_set("audio_decoder_interface") {
@@ -92,7 +137,9 @@ source_set("audio_decoder_interface") {
   ]
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
-  deps = [ "../..:webrtc_common" ]
+  deps = [
+    "../..:webrtc_common",
+  ]
 }
 
 source_set("audio_encoder_interface") {
@@ -102,7 +149,9 @@ source_set("audio_encoder_interface") {
   ]
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
-  deps = [ "../..:webrtc_common" ]
+  deps = [
+    "../..:webrtc_common",
+  ]
 }
 
 config("cng_config") {
@@ -115,11 +164,11 @@ config("cng_config") {
 source_set("cng") {
   sources = [
     "codecs/cng/audio_encoder_cng.cc",
+    "codecs/cng/audio_encoder_cng.h",
     "codecs/cng/cng_helpfuns.c",
     "codecs/cng/cng_helpfuns.h",
-    "codecs/cng/include/audio_encoder_cng.h",
-    "codecs/cng/include/webrtc_cng.h",
     "codecs/cng/webrtc_cng.c",
+    "codecs/cng/webrtc_cng.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -130,15 +179,13 @@ source_set("cng") {
   ]
 
   deps = [
-    "../../common_audio",
     ":audio_encoder_interface",
+    "../../common_audio",
   ]
 }
 
 config("red_config") {
-  include_dirs = [
-    "codecs/red",
-  ]
+  include_dirs = [ "codecs/red" ]
 }
 
 source_set("red") {
@@ -155,8 +202,8 @@ source_set("red") {
   ]
 
   deps = [
-    "../../common_audio",
     ":audio_encoder_interface",
+    "../../common_audio",
   ]
 }
 
@@ -169,12 +216,14 @@ config("g711_config") {
 
 source_set("g711") {
   sources = [
-    "codecs/g711/include/audio_encoder_pcm.h",
-    "codecs/g711/include/g711_interface.h",
+    "codecs/g711/audio_decoder_pcm.cc",
+    "codecs/g711/audio_decoder_pcm.h",
     "codecs/g711/audio_encoder_pcm.cc",
-    "codecs/g711/g711_interface.c",
+    "codecs/g711/audio_encoder_pcm.h",
     "codecs/g711/g711.c",
     "codecs/g711/g711.h",
+    "codecs/g711/g711_interface.c",
+    "codecs/g711/g711_interface.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -184,7 +233,10 @@ source_set("g711") {
     ":g711_config",
   ]
 
-  deps = [ ":audio_encoder_interface" ]
+  deps = [
+    ":audio_decoder_interface",
+    ":audio_encoder_interface",
+  ]
 }
 
 config("g722_config") {
@@ -196,13 +248,15 @@ config("g722_config") {
 
 source_set("g722") {
   sources = [
+    "codecs/g722/audio_decoder_g722.cc",
+    "codecs/g722/audio_decoder_g722.h",
     "codecs/g722/audio_encoder_g722.cc",
-    "codecs/g722/include/audio_encoder_g722.h",
-    "codecs/g722/include/g722_interface.h",
-    "codecs/g722/g722_interface.c",
-    "codecs/g722/g722_encode.c",
+    "codecs/g722/audio_encoder_g722.h",
     "codecs/g722/g722_decode.c",
     "codecs/g722/g722_enc_dec.h",
+    "codecs/g722/g722_encode.c",
+    "codecs/g722/g722_interface.c",
+    "codecs/g722/g722_interface.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -212,40 +266,45 @@ source_set("g722") {
     ":g722_config",
   ]
 
-  deps = [ ":audio_encoder_interface" ]
+  deps = [
+    ":audio_decoder_interface",
+    ":audio_encoder_interface",
+  ]
 }
 
 config("ilbc_config") {
   include_dirs = [
     "../../..",
-    "codecs/ilbc/interface",
+    "codecs/ilbc/include",
   ]
 }
 
 source_set("ilbc") {
   sources = [
-    "codecs/ilbc/audio_encoder_ilbc.cc",
-    "codecs/ilbc/include/audio_encoder_ilbc.h",
     "codecs/ilbc/abs_quant.c",
     "codecs/ilbc/abs_quant.h",
     "codecs/ilbc/abs_quant_loop.c",
     "codecs/ilbc/abs_quant_loop.h",
+    "codecs/ilbc/audio_decoder_ilbc.cc",
+    "codecs/ilbc/audio_decoder_ilbc.h",
+    "codecs/ilbc/audio_encoder_ilbc.cc",
+    "codecs/ilbc/audio_encoder_ilbc.h",
     "codecs/ilbc/augmented_cb_corr.c",
     "codecs/ilbc/augmented_cb_corr.h",
     "codecs/ilbc/bw_expand.c",
     "codecs/ilbc/bw_expand.h",
     "codecs/ilbc/cb_construct.c",
     "codecs/ilbc/cb_construct.h",
+    "codecs/ilbc/cb_mem_energy.c",
+    "codecs/ilbc/cb_mem_energy.h",
     "codecs/ilbc/cb_mem_energy_augmentation.c",
     "codecs/ilbc/cb_mem_energy_augmentation.h",
-    "codecs/ilbc/cb_mem_energy.c",
     "codecs/ilbc/cb_mem_energy_calc.c",
     "codecs/ilbc/cb_mem_energy_calc.h",
-    "codecs/ilbc/cb_mem_energy.h",
     "codecs/ilbc/cb_search.c",
+    "codecs/ilbc/cb_search.h",
     "codecs/ilbc/cb_search_core.c",
     "codecs/ilbc/cb_search_core.h",
-    "codecs/ilbc/cb_search.h",
     "codecs/ilbc/cb_update_best_index.c",
     "codecs/ilbc/cb_update_best_index.h",
     "codecs/ilbc/chebyshev.c",
@@ -269,12 +328,12 @@ source_set("ilbc") {
     "codecs/ilbc/encode.h",
     "codecs/ilbc/energy_inverse.c",
     "codecs/ilbc/energy_inverse.h",
+    "codecs/ilbc/enh_upsample.c",
+    "codecs/ilbc/enh_upsample.h",
     "codecs/ilbc/enhancer.c",
     "codecs/ilbc/enhancer.h",
     "codecs/ilbc/enhancer_interface.c",
     "codecs/ilbc/enhancer_interface.h",
-    "codecs/ilbc/enh_upsample.c",
-    "codecs/ilbc/enh_upsample.h",
     "codecs/ilbc/filtered_cb_vecs.c",
     "codecs/ilbc/filtered_cb_vecs.h",
     "codecs/ilbc/frame_classify.c",
@@ -294,6 +353,7 @@ source_set("ilbc") {
     "codecs/ilbc/hp_output.c",
     "codecs/ilbc/hp_output.h",
     "codecs/ilbc/ilbc.c",
+    "codecs/ilbc/ilbc.h",
     "codecs/ilbc/index_conv_dec.c",
     "codecs/ilbc/index_conv_dec.h",
     "codecs/ilbc/index_conv_enc.c",
@@ -302,7 +362,6 @@ source_set("ilbc") {
     "codecs/ilbc/init_decode.h",
     "codecs/ilbc/init_encode.c",
     "codecs/ilbc/init_encode.h",
-    "codecs/ilbc/interface/ilbc.h",
     "codecs/ilbc/interpolate.c",
     "codecs/ilbc/interpolate.h",
     "codecs/ilbc/interpolate_samples.c",
@@ -375,36 +434,47 @@ source_set("ilbc") {
   ]
 
   deps = [
-    "../../common_audio",
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    "../../common_audio",
   ]
 }
 
+source_set("isac_common") {
+  sources = [
+    "codecs/isac/audio_encoder_isac_t.h",
+    "codecs/isac/audio_encoder_isac_t_impl.h",
+    "codecs/isac/locked_bandwidth_info.cc",
+    "codecs/isac/locked_bandwidth_info.h",
+  ]
+  public_configs = [ "../..:common_inherited_config" ]
+}
+
 config("isac_config") {
   include_dirs = [
     "../../..",
-    "codecs/isac/main/interface",
+    "codecs/isac/main/include",
   ]
 }
 
 source_set("isac") {
   sources = [
-    "codecs/isac/audio_encoder_isac_t.h",
-    "codecs/isac/audio_encoder_isac_t_impl.h",
-    "codecs/isac/main/interface/audio_encoder_isac.h",
-    "codecs/isac/main/interface/isac.h",
+    "codecs/isac/main/include/audio_decoder_isac.h",
+    "codecs/isac/main/include/audio_encoder_isac.h",
+    "codecs/isac/main/include/isac.h",
     "codecs/isac/main/source/arith_routines.c",
     "codecs/isac/main/source/arith_routines.h",
     "codecs/isac/main/source/arith_routines_hist.c",
     "codecs/isac/main/source/arith_routines_logist.c",
+    "codecs/isac/main/source/audio_decoder_isac.cc",
     "codecs/isac/main/source/audio_encoder_isac.cc",
     "codecs/isac/main/source/bandwidth_estimator.c",
     "codecs/isac/main/source/bandwidth_estimator.h",
     "codecs/isac/main/source/codec.h",
     "codecs/isac/main/source/crc.c",
     "codecs/isac/main/source/crc.h",
-    "codecs/isac/main/source/decode_bwe.c",
     "codecs/isac/main/source/decode.c",
+    "codecs/isac/main/source/decode_bwe.c",
     "codecs/isac/main/source/encode.c",
     "codecs/isac/main/source/encode_lpc_swb.c",
     "codecs/isac/main/source/encode_lpc_swb.h",
@@ -412,12 +482,13 @@ source_set("isac") {
     "codecs/isac/main/source/entropy_coding.h",
     "codecs/isac/main/source/fft.c",
     "codecs/isac/main/source/fft.h",
-    "codecs/isac/main/source/filterbanks.c",
+    "codecs/isac/main/source/filter_functions.c",
     "codecs/isac/main/source/filterbank_tables.c",
     "codecs/isac/main/source/filterbank_tables.h",
-    "codecs/isac/main/source/filter_functions.c",
+    "codecs/isac/main/source/filterbanks.c",
     "codecs/isac/main/source/intialize.c",
     "codecs/isac/main/source/isac.c",
+    "codecs/isac/main/source/isac_float_type.h",
     "codecs/isac/main/source/lattice.c",
     "codecs/isac/main/source/lpc_analysis.c",
     "codecs/isac/main/source/lpc_analysis.h",
@@ -458,6 +529,7 @@ source_set("isac") {
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
+    ":isac_common",
     "../../common_audio",
   ]
 }
@@ -465,46 +537,50 @@ source_set("isac") {
 config("isac_fix_config") {
   include_dirs = [
     "../../..",
-    "codecs/isac/fix/interface",
+    "codecs/isac/fix/include",
   ]
 }
 
-source_set("isacfix") {
+source_set("isac_fix") {
   sources = [
-    "codecs/isac/audio_encoder_isac_t.h",
-    "codecs/isac/audio_encoder_isac_t_impl.h",
-    "codecs/isac/fix/interface/audio_encoder_isacfix.h",
-    "codecs/isac/fix/interface/isacfix.h",
+    "codecs/isac/fix/include/audio_decoder_isacfix.h",
+    "codecs/isac/fix/include/audio_encoder_isacfix.h",
+    "codecs/isac/fix/include/isacfix.h",
     "codecs/isac/fix/source/arith_routines.c",
     "codecs/isac/fix/source/arith_routines_hist.c",
     "codecs/isac/fix/source/arith_routines_logist.c",
     "codecs/isac/fix/source/arith_routins.h",
+    "codecs/isac/fix/source/audio_decoder_isacfix.cc",
     "codecs/isac/fix/source/audio_encoder_isacfix.cc",
     "codecs/isac/fix/source/bandwidth_estimator.c",
     "codecs/isac/fix/source/bandwidth_estimator.h",
     "codecs/isac/fix/source/codec.h",
-    "codecs/isac/fix/source/decode_bwe.c",
     "codecs/isac/fix/source/decode.c",
+    "codecs/isac/fix/source/decode_bwe.c",
     "codecs/isac/fix/source/decode_plc.c",
     "codecs/isac/fix/source/encode.c",
     "codecs/isac/fix/source/entropy_coding.c",
     "codecs/isac/fix/source/entropy_coding.h",
     "codecs/isac/fix/source/fft.c",
     "codecs/isac/fix/source/fft.h",
-    "codecs/isac/fix/source/filterbanks.c",
     "codecs/isac/fix/source/filterbank_tables.c",
     "codecs/isac/fix/source/filterbank_tables.h",
+    "codecs/isac/fix/source/filterbanks.c",
     "codecs/isac/fix/source/filters.c",
     "codecs/isac/fix/source/initialize.c",
+    "codecs/isac/fix/source/isac_fix_type.h",
     "codecs/isac/fix/source/isacfix.c",
     "codecs/isac/fix/source/lattice.c",
+    "codecs/isac/fix/source/lattice_c.c",
     "codecs/isac/fix/source/lpc_masking_model.c",
     "codecs/isac/fix/source/lpc_masking_model.h",
     "codecs/isac/fix/source/lpc_tables.c",
     "codecs/isac/fix/source/lpc_tables.h",
     "codecs/isac/fix/source/pitch_estimator.c",
     "codecs/isac/fix/source/pitch_estimator.h",
+    "codecs/isac/fix/source/pitch_estimator_c.c",
     "codecs/isac/fix/source/pitch_filter.c",
+    "codecs/isac/fix/source/pitch_filter_c.c",
     "codecs/isac/fix/source/pitch_gain_tables.c",
     "codecs/isac/fix/source/pitch_gain_tables.h",
     "codecs/isac/fix/source/pitch_lag_tables.c",
@@ -529,31 +605,26 @@ source_set("isacfix") {
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
+    ":isac_common",
     "../../common_audio",
     "../../system_wrappers",
   ]
 
-  if (rtc_build_armv7_neon) {
+  if (rtc_build_with_neon) {
     deps += [ ":isac_neon" ]
+  }
 
-    # Enable compilation for the ARM v7 Neon instruction set. This is needed
-    # since //build/config/arm.gni only enables Neon for iOS, not Android.
-    # This provides the same functionality as webrtc/build/arm_neon.gypi.
-    # TODO(kjellander): Investigate if this can be moved into webrtc.gni or
-    # //build/config/arm.gni instead, to reduce code duplication.
-    # Remove the -mfpu=vfpv3-d16 cflag.
-    configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
-    cflags = [
-      "-mfpu=neon",
-    ]
-
+  if (current_cpu == "arm" && arm_version >= 7) {
     sources += [
       "codecs/isac/fix/source/lattice_armv7.S",
       "codecs/isac/fix/source/pitch_filter_armv6.S",
     ]
-  } else {
-    sources += [ "codecs/isac/fix/source/pitch_filter_c.c" ]
+    sources -= [
+      "codecs/isac/fix/source/lattice_c.c",
+      "codecs/isac/fix/source/pitch_filter_c.c",
+    ]
   }
 
   if (current_cpu == "mipsel") {
@@ -564,6 +635,10 @@ source_set("isacfix") {
       "codecs/isac/fix/source/pitch_estimator_mips.c",
       "codecs/isac/fix/source/transform_mips.c",
     ]
+    sources -= [
+      "codecs/isac/fix/source/lattice_c.c",
+      "codecs/isac/fix/source/pitch_estimator_c.c",
+    ]
     if (mips_dsp_rev > 0) {
       sources += [ "codecs/isac/fix/source/filterbanks_mips.c" ]
     }
@@ -572,34 +647,31 @@ source_set("isacfix") {
         "codecs/isac/fix/source/lpc_masking_model_mips.c",
         "codecs/isac/fix/source/pitch_filter_mips.c",
       ]
-    } else {
-      sources += [ "codecs/isac/fix/source/pitch_filter_c.c" ]
+      sources -= [ "codecs/isac/fix/source/pitch_filter_c.c" ]
     }
-  } else {
-    sources += [ "codecs/isac/fix/source/pitch_estimator_c.c" ]
-  }
-
-  if (!rtc_build_armv7_neon && current_cpu != "mipsel") {
-    sources += [ "codecs/isac/fix/source/lattice_c.c" ]
   }
 }
 
-if (rtc_build_armv7_neon) {
+if (rtc_build_with_neon) {
   source_set("isac_neon") {
     sources = [
       "codecs/isac/fix/source/entropy_coding_neon.c",
-      "codecs/isac/fix/source/filterbanks_neon.S",
-      "codecs/isac/fix/source/filters_neon.S",
-      "codecs/isac/fix/source/lattice_neon.S",
-      "codecs/isac/fix/source/lpc_masking_model_neon.S",
-      "codecs/isac/fix/source/transform_neon.S",
+      "codecs/isac/fix/source/filterbanks_neon.c",
+      "codecs/isac/fix/source/filters_neon.c",
+      "codecs/isac/fix/source/lattice_neon.c",
+      "codecs/isac/fix/source/transform_neon.c",
     ]
 
-    include_dirs = [
-      "../../..",
-    ]
+    if (current_cpu != "arm64") {
+      # Enable compilation for the NEON instruction set. This is needed
+      # since //build/config/arm.gni only enables NEON for iOS, not Android.
+      # This provides the same functionality as webrtc/build/arm_neon.gypi.
+      configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
+    }
 
-    # Disable LTO in audio_processing_neon target due to compiler bug.
+    # Disable LTO on NEON targets due to compiler bug.
+    # TODO(fdegans): Enable this. See crbug.com/408997.
     if (rtc_use_lto) {
       cflags -= [
         "-flto",
@@ -607,21 +679,12 @@ if (rtc_build_armv7_neon) {
       ]
     }
 
-    # Enable compilation for the ARM v7 Neon instruction set. This is needed
-    # since //build/config/arm.gni only enables Neon for iOS, not Android.
-    # This provides the same functionality as webrtc/build/arm_neon.gypi.
-    # TODO(kjellander): Investigate if this can be moved into webrtc.gni or
-    # //build/config/arm.gni instead, to reduce code duplication.
-    # Remove the -mfpu=vfpv3-d16 cflag.
-    configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
-    cflags = [
-      "-mfpu=neon",
-    ]
-
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
-    deps = [ "../../common_audio" ]
+    deps = [
+      "../../common_audio",
+    ]
   }
 }
 
@@ -634,13 +697,16 @@ config("pcm16b_config") {
 
 source_set("pcm16b") {
   sources = [
-    "codecs/pcm16b/include/audio_encoder_pcm16b.h",
-    "codecs/pcm16b/include/pcm16b.h",
+    "codecs/pcm16b/audio_decoder_pcm16b.cc",
+    "codecs/pcm16b/audio_decoder_pcm16b.h",
     "codecs/pcm16b/audio_encoder_pcm16b.cc",
+    "codecs/pcm16b/audio_encoder_pcm16b.h",
     "codecs/pcm16b/pcm16b.c",
+    "codecs/pcm16b/pcm16b.h",
   ]
 
   deps = [
+    ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":g711",
   ]
@@ -659,21 +725,28 @@ config("opus_config") {
 
 source_set("webrtc_opus") {
   sources = [
+    "codecs/opus/audio_decoder_opus.cc",
+    "codecs/opus/audio_decoder_opus.h",
     "codecs/opus/audio_encoder_opus.cc",
-    "codecs/opus/interface/audio_encoder_opus.h",
-    "codecs/opus/interface/opus_interface.h",
+    "codecs/opus/audio_encoder_opus.h",
     "codecs/opus/opus_inst.h",
     "codecs/opus/opus_interface.c",
+    "codecs/opus/opus_interface.h",
   ]
 
-  deps = [ ":audio_encoder_interface" ]
+  deps = [
+    ":audio_decoder_interface",
+    ":audio_encoder_interface",
+    "../../base:rtc_base_approved",
+  ]
 
   if (rtc_build_opus) {
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
-    deps += [ rtc_opus_dir ]
-    forward_dependent_configs_from = [ rtc_opus_dir ]
+    public_deps = [
+      rtc_opus_dir,
+    ]
   } else if (build_with_mozilla) {
     include_dirs = [ getenv("DIST") + "/include/opus" ]
   }
@@ -689,7 +762,6 @@ config("neteq_config") {
 
 source_set("neteq") {
   sources = [
-    "neteq/interface/neteq.h",
     "neteq/accelerate.cc",
     "neteq/accelerate.h",
     "neteq/audio_classifier.cc",
@@ -727,13 +799,14 @@ source_set("neteq") {
     "neteq/dtmf_tone_generator.h",
     "neteq/expand.cc",
     "neteq/expand.h",
+    "neteq/include/neteq.h",
     "neteq/merge.cc",
     "neteq/merge.h",
+    "neteq/nack.cc",
+    "neteq/nack.h",
+    "neteq/neteq.cc",
     "neteq/neteq_impl.cc",
     "neteq/neteq_impl.h",
-    "neteq/neteq.cc",
-    "neteq/statistics_calculator.cc",
-    "neteq/statistics_calculator.h",
     "neteq/normal.cc",
     "neteq/normal.h",
     "neteq/packet_buffer.cc",
@@ -748,12 +821,14 @@ source_set("neteq") {
     "neteq/random_vector.h",
     "neteq/rtcp.cc",
     "neteq/rtcp.h",
+    "neteq/statistics_calculator.cc",
+    "neteq/statistics_calculator.h",
     "neteq/sync_buffer.cc",
     "neteq/sync_buffer.h",
-    "neteq/timestamp_scaler.cc",
-    "neteq/timestamp_scaler.h",
     "neteq/time_stretch.cc",
     "neteq/time_stretch.h",
+    "neteq/timestamp_scaler.cc",
+    "neteq/timestamp_scaler.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -763,20 +838,10 @@ source_set("neteq") {
     ":neteq_config",
   ]
 
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-
   deps = [
     ":audio_decoder_interface",
     ":cng",
     ":g711",
-    ":g722",
-    ":ilbc",
-    ":isac",
-    ":isacfix",
     ":pcm16b",
     "../..:webrtc_common",
     "../../common_audio",
@@ -789,4 +854,19 @@ source_set("neteq") {
     defines += [ "WEBRTC_CODEC_OPUS" ]
     deps += [ ":webrtc_opus" ]
   }
+  if (!build_with_mozilla) {
+    if (current_cpu == "arm") {
+      defines += [ "WEBRTC_CODEC_ISACFX" ]
+      deps += [ ":isac_fix" ]
+    } else {
+      defines += [ "WEBRTC_CODEC_ISAC" ]
+      deps += [ ":isac" ]
+    }
+    defines += [ "WEBRTC_CODEC_G722" ]
+    deps += [ ":g722" ]
+  }
+  if (!build_with_mozilla && !build_with_chromium) {
+    defines += [ "WEBRTC_CODEC_ILBC" ]
+    deps += [ ":ilbc" ]
+  }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_coding/OWNERS
index ee76c69701..77db17d1c5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/OWNERS
@@ -2,8 +2,8 @@ tina.legrand@webrtc.org
 turaj@webrtc.org
 henrik.lundin@webrtc.org
 kwiberg@webrtc.org
-
-per-file *.isolate=kjellander@webrtc.org
+minyue@webrtc.org
+jan.skoglund@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_codec_database.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
new file mode 100644
index 0000000000..5f3c07802b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
@@ -0,0 +1,333 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file generates databases with information about all supported audio
+ * codecs.
+ */
+
+// TODO(tlegrand): Change constant input pointers in all functions to constant
+// references, where appropriate.
+#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
+
+#include <assert.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+namespace {
+
+// Checks if the bitrate is valid for iSAC.
+bool IsISACRateValid(int rate) {
+  return (rate == -1) || ((rate <= 56000) && (rate >= 10000));
+}
+
+// Checks if the bitrate is valid for iLBC.
+bool IsILBCRateValid(int rate, int frame_size_samples) {
+  if (((frame_size_samples == 240) || (frame_size_samples == 480)) &&
+      (rate == 13300)) {
+    return true;
+  } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
+      (rate == 15200)) {
+    return true;
+  } else {
+    return false;
+  }
+}
+
+// Checks if the bitrate is valid for Opus.
+bool IsOpusRateValid(int rate) {
+  return (rate >= 6000) && (rate <= 510000);
+}
+
+}  // namespace
+
+// Not yet used payload-types.
+// 83,  82,  81, 80, 79,  78,  77,  76,  75,  74,  73,  72,  71,  70,  69, 68,
+// 67, 66, 65
+
+const CodecInst ACMCodecDB::database_[] = {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+  {103, "ISAC", 16000, kIsacPacSize480, 1, kIsacWbDefaultRate},
+# if (defined(WEBRTC_CODEC_ISAC))
+  {104, "ISAC", 32000, kIsacPacSize960, 1, kIsacSwbDefaultRate},
+# endif
+#endif
+  // Mono
+  {107, "L16", 8000, 80, 1, 128000},
+  {108, "L16", 16000, 160, 1, 256000},
+  {109, "L16", 32000, 320, 1, 512000},
+  // Stereo
+  {111, "L16", 8000, 80, 2, 128000},
+  {112, "L16", 16000, 160, 2, 256000},
+  {113, "L16", 32000, 320, 2, 512000},
+  // G.711, PCM mu-law and A-law.
+  // Mono
+  {0, "PCMU", 8000, 160, 1, 64000},
+  {8, "PCMA", 8000, 160, 1, 64000},
+  // Stereo
+  {110, "PCMU", 8000, 160, 2, 64000},
+  {118, "PCMA", 8000, 160, 2, 64000},
+#ifdef WEBRTC_CODEC_ILBC
+  {102, "ILBC", 8000, 240, 1, 13300},
+#endif
+#ifdef WEBRTC_CODEC_G722
+  // Mono
+  {9, "G722", 16000, 320, 1, 64000},
+  // Stereo
+  {119, "G722", 16000, 320, 2, 64000},
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  // Opus internally supports 48, 24, 16, 12, 8 kHz.
+  // Mono and stereo.
+  {120, "opus", 48000, 960, 2, 64000},
+#endif
+  // Comfort noise for four different sampling frequencies.
+  {13, "CN", 8000, 240, 1, 0},
+  {98, "CN", 16000, 480, 1, 0},
+  {99, "CN", 32000, 960, 1, 0},
+#ifdef ENABLE_48000_HZ
+  {100, "CN", 48000, 1440, 1, 0},
+#endif
+  {106, "telephone-event", 8000, 240, 1, 0},
+#ifdef WEBRTC_CODEC_RED
+  {127, "red", 8000, 0, 1, 0},
+#endif
+  // To prevent compile errors due to trailing commas.
+  {-1, "Null", -1, -1, 0, -1}
+};
+
+// Create database with all codec settings at compile time.
+// Each entry needs the following parameters in the given order:
+// Number of allowed packet sizes, a vector with the allowed packet sizes,
+// Basic block samples, max number of channels that are supported.
+const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+    {2, {kIsacPacSize480, kIsacPacSize960}, 0, 1},
+# if (defined(WEBRTC_CODEC_ISAC))
+    {1, {kIsacPacSize960}, 0, 1},
+# endif
+#endif
+    // Mono
+    {4, {80, 160, 240, 320}, 0, 2},
+    {4, {160, 320, 480, 640}, 0, 2},
+    {2, {320, 640}, 0, 2},
+    // Stereo
+    {4, {80, 160, 240, 320}, 0, 2},
+    {4, {160, 320, 480, 640}, 0, 2},
+    {2, {320, 640}, 0, 2},
+    // G.711, PCM mu-law and A-law.
+    // Mono
+    {6, {80, 160, 240, 320, 400, 480}, 0, 2},
+    {6, {80, 160, 240, 320, 400, 480}, 0, 2},
+    // Stereo
+    {6, {80, 160, 240, 320, 400, 480}, 0, 2},
+    {6, {80, 160, 240, 320, 400, 480}, 0, 2},
+#ifdef WEBRTC_CODEC_ILBC
+    {4, {160, 240, 320, 480}, 0, 1},
+#endif
+#ifdef WEBRTC_CODEC_G722
+    // Mono
+    {6, {160, 320, 480, 640, 800, 960}, 0, 2},
+    // Stereo
+    {6, {160, 320, 480, 640, 800, 960}, 0, 2},
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    // Opus supports frames shorter than 10ms,
+    // but it doesn't help us to use them.
+    // Mono and stereo.
+    {4, {480, 960, 1920, 2880}, 0, 2},
+#endif
+    // Comfort noise for three different sampling frequencies.
+    {1, {240}, 240, 1},
+    {1, {480}, 480, 1},
+    {1, {960}, 960, 1},
+#ifdef ENABLE_48000_HZ
+    {1, {1440}, 1440, 1},
+#endif
+    {1, {240}, 240, 1},
+#ifdef WEBRTC_CODEC_RED
+    {1, {0}, 0, 1},
+#endif
+    // To prevent compile errors due to trailing commas.
+    {-1, {-1}, -1, 0}
+};
+
+// Create a database of all NetEQ decoders at compile time.
+const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+    NetEqDecoder::kDecoderISAC,
+# if (defined(WEBRTC_CODEC_ISAC))
+    NetEqDecoder::kDecoderISACswb,
+# endif
+#endif
+    // Mono
+    NetEqDecoder::kDecoderPCM16B, NetEqDecoder::kDecoderPCM16Bwb,
+    NetEqDecoder::kDecoderPCM16Bswb32kHz,
+    // Stereo
+    NetEqDecoder::kDecoderPCM16B_2ch, NetEqDecoder::kDecoderPCM16Bwb_2ch,
+    NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch,
+    // G.711, PCM mu-las and A-law.
+    // Mono
+    NetEqDecoder::kDecoderPCMu, NetEqDecoder::kDecoderPCMa,
+    // Stereo
+    NetEqDecoder::kDecoderPCMu_2ch, NetEqDecoder::kDecoderPCMa_2ch,
+#ifdef WEBRTC_CODEC_ILBC
+    NetEqDecoder::kDecoderILBC,
+#endif
+#ifdef WEBRTC_CODEC_G722
+    // Mono
+    NetEqDecoder::kDecoderG722,
+    // Stereo
+    NetEqDecoder::kDecoderG722_2ch,
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    // Mono and stereo.
+    NetEqDecoder::kDecoderOpus,
+#endif
+    // Comfort noise for three different sampling frequencies.
+    NetEqDecoder::kDecoderCNGnb, NetEqDecoder::kDecoderCNGwb,
+    NetEqDecoder::kDecoderCNGswb32kHz,
+#ifdef ENABLE_48000_HZ
+    NetEqDecoder::kDecoderCNGswb48kHz,
+#endif
+    NetEqDecoder::kDecoderAVT,
+#ifdef WEBRTC_CODEC_RED
+    NetEqDecoder::kDecoderRED,
+#endif
+};
+
+// Enumerator for error codes when asking for codec database id.
+enum {
+  kInvalidCodec = -10,
+  kInvalidPayloadtype = -30,
+  kInvalidPacketSize = -40,
+  kInvalidRate = -50
+};
+
+// Gets the codec id number from the database. If there is some mismatch in
+// the codec settings, the function will return an error code.
+// NOTE! The first mismatch found will generate the return value.
+int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
+  // Look for a matching codec in the database.
+  int codec_id = CodecId(codec_inst);
+
+  // Checks if we found a matching codec.
+  if (codec_id == -1) {
+    return kInvalidCodec;
+  }
+
+  // Checks the validity of payload type
+  if (!RentACodec::IsPayloadTypeValid(codec_inst.pltype)) {
+    return kInvalidPayloadtype;
+  }
+
+  // Comfort Noise is special case, packet-size & rate is not checked.
+  if (STR_CASE_CMP(database_[codec_id].plname, "CN") == 0) {
+    return codec_id;
+  }
+
+  // RED is special case, packet-size & rate is not checked.
+  if (STR_CASE_CMP(database_[codec_id].plname, "red") == 0) {
+    return codec_id;
+  }
+
+  // Checks the validity of packet size.
+  if (codec_settings_[codec_id].num_packet_sizes > 0) {
+    bool packet_size_ok = false;
+    int i;
+    int packet_size_samples;
+    for (i = 0; i < codec_settings_[codec_id].num_packet_sizes; i++) {
+      packet_size_samples =
+          codec_settings_[codec_id].packet_sizes_samples[i];
+      if (codec_inst.pacsize == packet_size_samples) {
+        packet_size_ok = true;
+        break;
+      }
+    }
+
+    if (!packet_size_ok) {
+      return kInvalidPacketSize;
+    }
+  }
+
+  if (codec_inst.pacsize < 1) {
+    return kInvalidPacketSize;
+  }
+
+  // Check the validity of rate. Codecs with multiple rates have their own
+  // function for this.
+  if (STR_CASE_CMP("isac", codec_inst.plname) == 0) {
+    return IsISACRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
+  } else if (STR_CASE_CMP("ilbc", codec_inst.plname) == 0) {
+    return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize)
+        ? codec_id : kInvalidRate;
+  } else if (STR_CASE_CMP("opus", codec_inst.plname) == 0) {
+    return IsOpusRateValid(codec_inst.rate)
+        ? codec_id : kInvalidRate;
+  }
+
+  return database_[codec_id].rate == codec_inst.rate ? codec_id : kInvalidRate;
+}
+
+// Looks for a matching payload name, frequency, and channels in the
+// codec list. Need to check all three since some codecs have several codec
+// entries with different frequencies and/or channels.
+// Does not check other codec settings, such as payload type and packet size.
+// Returns the id of the codec, or -1 if no match is found.
+int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
+  return (CodecId(codec_inst.plname, codec_inst.plfreq,
+                  codec_inst.channels));
+}
+
+int ACMCodecDB::CodecId(const char* payload_name,
+                        int frequency,
+                        size_t channels) {
+  for (const CodecInst& ci : RentACodec::Database()) {
+    bool name_match = false;
+    bool frequency_match = false;
+    bool channels_match = false;
+
+    // Payload name, sampling frequency and number of channels need to match.
+    // NOTE! If |frequency| is -1, the frequency is not applicable, and is
+    // always treated as true, like for RED.
+    name_match = (STR_CASE_CMP(ci.plname, payload_name) == 0);
+    frequency_match = (frequency == ci.plfreq) || (frequency == -1);
+    // The number of channels must match for all codecs but Opus.
+    if (STR_CASE_CMP(payload_name, "opus") != 0) {
+      channels_match = (channels == ci.channels);
+    } else {
+      // For opus we just check that number of channels is valid.
+      channels_match = (channels == 1 || channels == 2);
+    }
+
+    if (name_match && frequency_match && channels_match) {
+      // We have found a matching codec in the list.
+      return &ci - RentACodec::Database().data();
+    }
+  }
+
+  // We didn't find a matching codec.
+  return -1;
+}
+// Gets codec id number from database for the receiver.
+int ACMCodecDB::ReceiverCodecNumber(const CodecInst& codec_inst) {
+  // Look for a matching codec in the database.
+  return CodecId(codec_inst);
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_codec_database.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_codec_database.h
new file mode 100644
index 0000000000..6c2db9cfc8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_codec_database.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file generates databases with information about all supported audio
+ * codecs.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+// TODO(tlegrand): replace class ACMCodecDB with a namespace.
+class ACMCodecDB {
+ public:
+  // kMaxNumCodecs - Maximum number of codecs that can be activated in one
+  //                 build.
+  // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec.
+  // These might need to be increased if adding a new codec to the database
+  static const int kMaxNumCodecs =  50;
+  static const int kMaxNumPacketSize = 6;
+
+  // Codec specific settings
+  //
+  // num_packet_sizes     - number of allowed packet sizes.
+  // packet_sizes_samples - list of the allowed packet sizes.
+  // basic_block_samples  - assigned a value different from 0 if the codec
+  //                        requires to be fed with a specific number of samples
+  //                        that can be different from packet size.
+  // channel_support      - number of channels supported to encode;
+  //                        1 = mono, 2 = stereo, etc.
+  struct CodecSettings {
+    int num_packet_sizes;
+    int packet_sizes_samples[kMaxNumPacketSize];
+    int basic_block_samples;
+    size_t channel_support;
+  };
+
+  // Returns codec id from database, given the information received in the input
+  // [codec_inst].
+  // Input:
+  //   [codec_inst] - Information about the codec for which we require the
+  //                  database id.
+  // Return:
+  //   codec id if successful, otherwise < 0.
+  static int CodecNumber(const CodecInst& codec_inst);
+  static int CodecId(const CodecInst& codec_inst);
+  static int CodecId(const char* payload_name, int frequency, size_t channels);
+  static int ReceiverCodecNumber(const CodecInst& codec_inst);
+
+  // Databases with information about the supported codecs
+  // database_ - stored information about all codecs: payload type, name,
+  //             sampling frequency, packet size in samples, default channel
+  //             support, and default rate.
+  // codec_settings_ - stored codec settings: number of allowed packet sizes,
+  //                   a vector with the allowed packet sizes, basic block
+  //                   samples, and max number of channels that are supported.
+  // neteq_decoders_ - list of supported decoders in NetEQ.
+  static const CodecInst database_[kMaxNumCodecs];
+  static const CodecSettings codec_settings_[kMaxNumCodecs];
+  static const NetEqDecoder neteq_decoders_[kMaxNumCodecs];
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_common_defs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_common_defs.h
new file mode 100644
index 0000000000..483bdd93f1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_common_defs.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
+
+#include "webrtc/engine_configurations.h"
+
+// Checks for enabled codecs, we prevent enabling codecs which are not
+// compatible.
+#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
+#error iSAC and iSACFX codecs cannot be enabled at the same time
+#endif
+
+namespace webrtc {
+
+// General codec specific defines
+const int kIsacWbDefaultRate = 32000;
+const int kIsacSwbDefaultRate = 56000;
+const int kIsacPacSize480 = 480;
+const int kIsacPacSize960 = 960;
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
similarity index 92%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index 96a1fc5fdc..855a39e675 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
@@ -47,7 +47,6 @@ bool ModifyAndUseThisCodec(CodecInst* codec_param) {
 // iLBC = 102
 // iSAC wideband = 103
 // iSAC super-wideband = 104
-// iSAC fullband = 124
 // AVT/DTMF = 106
 // RED = 117
 // PCM16b 8 kHz = 93
@@ -56,7 +55,7 @@ bool ModifyAndUseThisCodec(CodecInst* codec_param) {
 // G.722 = 94
 bool RemapPltypeAndUseThisCodec(const char* plname,
                                 int plfreq,
-                                int channels,
+                                size_t channels,
                                 int* pltype) {
   if (channels != 1)
     return false;  // Don't use non-mono codecs.
@@ -78,8 +77,6 @@ bool RemapPltypeAndUseThisCodec(const char* plname,
     *pltype = 103;
   } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
     *pltype = 104;
-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 48000) {
-    *pltype = 124;
   } else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
     *pltype = 106;
   } else if (STR_CASE_CMP(plname, "red") == 0) {
@@ -143,6 +140,16 @@ void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
   }
 }
 
+int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
+    int rtp_payload_type,
+    AudioDecoder* external_decoder,
+    int sample_rate_hz,
+    int num_channels,
+    const std::string& name) {
+  return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
+                                            sample_rate_hz, num_channels, name);
+}
+
 void AcmReceiveTestOldApi::Run() {
   for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
        packet.reset(packet_source_->NextPacket())) {
@@ -151,7 +158,8 @@ void AcmReceiveTestOldApi::Run() {
       AudioFrame output_frame;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
       EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
-      const int samples_per_block = output_freq_hz_ * 10 / 1000;
+      const size_t samples_per_block =
+          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
       if (exptected_output_channels_ != kArbitraryChannels) {
         if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
similarity index 78%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
index 5e5ff9a0a0..3010ec72b1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
@@ -8,15 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 class AudioCodingModule;
+class AudioDecoder;
 struct CodecInst;
 
 namespace test {
@@ -44,6 +47,12 @@ class AcmReceiveTestOldApi {
   // files.
   void RegisterNetEqTestCodecs();
 
+  int RegisterExternalReceiveCodec(int rtp_payload_type,
+                                   AudioDecoder* external_decoder,
+                                   int sample_rate_hz,
+                                   int num_channels,
+                                   const std::string& name);
+
   // Runs the test and returns true if successful.
   void Run();
 
@@ -58,7 +67,7 @@ class AcmReceiveTestOldApi {
   int output_freq_hz_;
   NumOutputChannels exptected_output_channels_;
 
-  DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
 };
 
 // This test toggles the output frequency every |toggle_period_ms|. The test
@@ -85,4 +94,4 @@ class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.cc
new file mode 100644
index 0000000000..f45d5d3414
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -0,0 +1,541 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
+
+#include <stdlib.h>  // malloc
+
+#include <algorithm>  // sort
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+namespace {
+
+// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
+// before the call to this function.
+void SetAudioFrameActivityAndType(bool vad_enabled,
+                                  NetEqOutputType type,
+                                  AudioFrame* audio_frame) {
+  if (vad_enabled) {
+    switch (type) {
+      case kOutputNormal: {
+        audio_frame->vad_activity_ = AudioFrame::kVadActive;
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        break;
+      }
+      case kOutputVADPassive: {
+        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        break;
+      }
+      case kOutputCNG: {
+        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+        audio_frame->speech_type_ = AudioFrame::kCNG;
+        break;
+      }
+      case kOutputPLC: {
+        // Don't change |audio_frame->vad_activity_|, it should be the same as
+        // |previous_audio_activity_|.
+        audio_frame->speech_type_ = AudioFrame::kPLC;
+        break;
+      }
+      case kOutputPLCtoCNG: {
+        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+        audio_frame->speech_type_ = AudioFrame::kPLCCNG;
+        break;
+      }
+      default:
+        assert(false);
+    }
+  } else {
+    // Always return kVadUnknown when receive VAD is inactive
+    audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
+    switch (type) {
+      case kOutputNormal: {
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        break;
+      }
+      case kOutputCNG: {
+        audio_frame->speech_type_ = AudioFrame::kCNG;
+        break;
+      }
+      case kOutputPLC: {
+        audio_frame->speech_type_ = AudioFrame::kPLC;
+        break;
+      }
+      case kOutputPLCtoCNG: {
+        audio_frame->speech_type_ = AudioFrame::kPLCCNG;
+        break;
+      }
+      case kOutputVADPassive: {
+        // Normally, we should no get any VAD decision if post-decoding VAD is
+        // not active. However, if post-decoding VAD has been active then
+        // disabled, we might be here for couple of frames.
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        LOG(WARNING) << "Post-decoding VAD is disabled but output is "
+            << "labeled VAD-passive";
+        break;
+      }
+      default:
+        assert(false);
+    }
+  }
+}
+
+// Is the given codec a CNG codec?
+// TODO(kwiberg): Move to RentACodec.
+bool IsCng(int codec_id) {
+  auto i = RentACodec::CodecIdFromIndex(codec_id);
+  return (i && (*i == RentACodec::CodecId::kCNNB ||
+                *i == RentACodec::CodecId::kCNWB ||
+                *i == RentACodec::CodecId::kCNSWB ||
+                *i == RentACodec::CodecId::kCNFB));
+}
+
+}  // namespace
+
+AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
+    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+      id_(config.id),
+      last_audio_decoder_(nullptr),
+      previous_audio_activity_(AudioFrame::kVadPassive),
+      audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
+      last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
+      neteq_(NetEq::Create(config.neteq_config)),
+      vad_enabled_(config.neteq_config.enable_post_decode_vad),
+      clock_(config.clock),
+      resampled_last_output_frame_(true) {
+  assert(clock_);
+  memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
+  memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
+}
+
+AcmReceiver::~AcmReceiver() {
+  delete neteq_;
+}
+
+int AcmReceiver::SetMinimumDelay(int delay_ms) {
+  if (neteq_->SetMinimumDelay(delay_ms))
+    return 0;
+  LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+  return -1;
+}
+
+int AcmReceiver::SetMaximumDelay(int delay_ms) {
+  if (neteq_->SetMaximumDelay(delay_ms))
+    return 0;
+  LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+  return -1;
+}
+
+int AcmReceiver::LeastRequiredDelayMs() const {
+  return neteq_->LeastRequiredDelayMs();
+}
+
+rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  return last_packet_sample_rate_hz_;
+}
+
+int AcmReceiver::last_output_sample_rate_hz() const {
+  return neteq_->last_output_sample_rate_hz();
+}
+
+int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
+                              rtc::ArrayView<const uint8_t> incoming_payload) {
+  uint32_t receive_timestamp = 0;
+  const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
+
+  {
+    CriticalSectionScoped lock(crit_sect_.get());
+
+    const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload[0]);
+    if (!decoder) {
+      LOG_F(LS_ERROR) << "Payload-type "
+                      << static_cast<int>(header->payloadType)
+                      << " is not registered.";
+      return -1;
+    }
+    const int sample_rate_hz = [&decoder] {
+      const auto ci = RentACodec::CodecIdFromIndex(decoder->acm_codec_id);
+      return ci ? RentACodec::CodecInstById(*ci)->plfreq : -1;
+    }();
+    receive_timestamp = NowInTimestamp(sample_rate_hz);
+
+    // If this is a CNG while the audio codec is not mono, skip pushing in
+    // packets into NetEq.
+    if (IsCng(decoder->acm_codec_id) && last_audio_decoder_ &&
+        last_audio_decoder_->channels > 1)
+        return 0;
+    if (!IsCng(decoder->acm_codec_id) &&
+        decoder->acm_codec_id !=
+            *RentACodec::CodecIndexFromId(RentACodec::CodecId::kAVT)) {
+      last_audio_decoder_ = decoder;
+      last_packet_sample_rate_hz_ = rtc::Optional<int>(decoder->sample_rate_hz);
+    }
+
+  }  // |crit_sect_| is released.
+
+  if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
+      0) {
+    LOG(LERROR) << "AcmReceiver::InsertPacket "
+                << static_cast<int>(header->payloadType)
+                << " Failed to insert packet";
+    return -1;
+  }
+  return 0;
+}
+
+int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
+  enum NetEqOutputType type;
+  size_t samples_per_channel;
+  size_t num_channels;
+
+  // Accessing members, take the lock.
+  CriticalSectionScoped lock(crit_sect_.get());
+
+  // Always write the output to |audio_buffer_| first.
+  if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
+                       audio_buffer_.get(),
+                       &samples_per_channel,
+                       &num_channels,
+                       &type) != NetEq::kOK) {
+    LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
+    return -1;
+  }
+
+  const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
+
+  // Update if resampling is required.
+  const bool need_resampling =
+      (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
+
+  if (need_resampling && !resampled_last_output_frame_) {
+    // Prime the resampler with the last frame.
+    int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        num_channels, AudioFrame::kMaxDataSizeSamples, temp_output);
+    if (samples_per_channel_int < 0) {
+      LOG(LERROR) << "AcmReceiver::GetAudio - "
+                     "Resampling last_audio_buffer_ failed.";
+      return -1;
+    }
+    samples_per_channel = static_cast<size_t>(samples_per_channel_int);
+  }
+
+  // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
+  // through resampling, or through straight memcpy.
+  // TODO(henrik.lundin) Glitches in the output may appear if the output rate
+  // from NetEq changes. See WebRTC issue 3923.
+  if (need_resampling) {
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        num_channels, AudioFrame::kMaxDataSizeSamples, audio_frame->data_);
+    if (samples_per_channel_int < 0) {
+      LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
+      return -1;
+    }
+    samples_per_channel = static_cast<size_t>(samples_per_channel_int);
+    resampled_last_output_frame_ = true;
+  } else {
+    resampled_last_output_frame_ = false;
+    // We might end up here ONLY if codec is changed.
+    memcpy(audio_frame->data_,
+           audio_buffer_.get(),
+           samples_per_channel * num_channels * sizeof(int16_t));
+  }
+
+  // Swap buffers, so that the current audio is stored in |last_audio_buffer_|
+  // for next time.
+  audio_buffer_.swap(last_audio_buffer_);
+
+  audio_frame->num_channels_ = num_channels;
+  audio_frame->samples_per_channel_ = samples_per_channel;
+  audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
+
+  // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
+  audio_frame->vad_activity_ = previous_audio_activity_;
+  SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
+  previous_audio_activity_ = audio_frame->vad_activity_;
+  call_stats_.DecodedByNetEq(audio_frame->speech_type_);
+
+  // Computes the RTP timestamp of the first sample in |audio_frame| from
+  // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
+  // |audio_frame|.
+  uint32_t playout_timestamp = 0;
+  if (GetPlayoutTimestamp(&playout_timestamp)) {
+    audio_frame->timestamp_ = playout_timestamp -
+        static_cast<uint32_t>(audio_frame->samples_per_channel_);
+  } else {
+    // Remain 0 until we have a valid |playout_timestamp|.
+    audio_frame->timestamp_ = 0;
+  }
+
+  return 0;
+}
+
+int32_t AcmReceiver::AddCodec(int acm_codec_id,
+                              uint8_t payload_type,
+                              size_t channels,
+                              int sample_rate_hz,
+                              AudioDecoder* audio_decoder,
+                              const std::string& name) {
+  const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
+    if (acm_codec_id == -1)
+      return NetEqDecoder::kDecoderArbitrary;  // External decoder.
+    const rtc::Optional<RentACodec::CodecId> cid =
+        RentACodec::CodecIdFromIndex(acm_codec_id);
+    RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
+    const rtc::Optional<NetEqDecoder> ned =
+        RentACodec::NetEqDecoderFromCodecId(*cid, channels);
+    RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
+    return *ned;
+  }();
+
+  CriticalSectionScoped lock(crit_sect_.get());
+
+  // The corresponding NetEq decoder ID.
+  // If this codec has been registered before.
+  auto it = decoders_.find(payload_type);
+  if (it != decoders_.end()) {
+    const Decoder& decoder = it->second;
+    if (acm_codec_id != -1 && decoder.acm_codec_id == acm_codec_id &&
+        decoder.channels == channels &&
+        decoder.sample_rate_hz == sample_rate_hz) {
+      // Re-registering the same codec. Do nothing and return.
+      return 0;
+    }
+
+    // Changing codec. First unregister the old codec, then register the new
+    // one.
+    if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
+      LOG(LERROR) << "Cannot remove payload " << static_cast<int>(payload_type);
+      return -1;
+    }
+
+    decoders_.erase(it);
+  }
+
+  int ret_val;
+  if (!audio_decoder) {
+    ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
+  } else {
+    ret_val = neteq_->RegisterExternalDecoder(
+        audio_decoder, neteq_decoder, name, payload_type, sample_rate_hz);
+  }
+  if (ret_val != NetEq::kOK) {
+    LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
+                << static_cast<int>(payload_type)
+                << " channels: " << channels;
+    return -1;
+  }
+
+  Decoder decoder;
+  decoder.acm_codec_id = acm_codec_id;
+  decoder.payload_type = payload_type;
+  decoder.channels = channels;
+  decoder.sample_rate_hz = sample_rate_hz;
+  decoders_[payload_type] = decoder;
+  return 0;
+}
+
+void AcmReceiver::EnableVad() {
+  neteq_->EnableVad();
+  CriticalSectionScoped lock(crit_sect_.get());
+  vad_enabled_ = true;
+}
+
+void AcmReceiver::DisableVad() {
+  neteq_->DisableVad();
+  CriticalSectionScoped lock(crit_sect_.get());
+  vad_enabled_ = false;
+}
+
+void AcmReceiver::FlushBuffers() {
+  neteq_->FlushBuffers();
+}
+
+// If failed in removing one of the codecs, this method continues to remove as
+// many as it can.
+int AcmReceiver::RemoveAllCodecs() {
+  int ret_val = 0;
+  CriticalSectionScoped lock(crit_sect_.get());
+  for (auto it = decoders_.begin(); it != decoders_.end(); ) {
+    auto cur = it;
+    ++it;  // it will be valid even if we erase cur
+    if (neteq_->RemovePayloadType(cur->second.payload_type) == 0) {
+      decoders_.erase(cur);
+    } else {
+      LOG_F(LS_ERROR) << "Cannot remove payload "
+                      << static_cast<int>(cur->second.payload_type);
+      ret_val = -1;
+    }
+  }
+
+  // No codec is registered, invalidate last audio decoder.
+  last_audio_decoder_ = nullptr;
+  last_packet_sample_rate_hz_ = rtc::Optional<int>();
+  return ret_val;
+}
+
+int AcmReceiver::RemoveCodec(uint8_t payload_type) {
+  CriticalSectionScoped lock(crit_sect_.get());
+  auto it = decoders_.find(payload_type);
+  if (it == decoders_.end()) {  // Such a payload-type is not registered.
+    return 0;
+  }
+  if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
+    LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
+    return -1;
+  }
+  if (last_audio_decoder_ == &it->second) {
+    last_audio_decoder_ = nullptr;
+    last_packet_sample_rate_hz_ = rtc::Optional<int>();
+  }
+  decoders_.erase(it);
+  return 0;
+}
+
+void AcmReceiver::set_id(int id) {
+  CriticalSectionScoped lock(crit_sect_.get());
+  id_ = id;
+}
+
+bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
+  return neteq_->GetPlayoutTimestamp(timestamp);
+}
+
+int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  if (!last_audio_decoder_) {
+    return -1;
+  }
+  *codec = *RentACodec::CodecInstById(
+      *RentACodec::CodecIdFromIndex(last_audio_decoder_->acm_codec_id));
+  codec->pltype = last_audio_decoder_->payload_type;
+  codec->channels = last_audio_decoder_->channels;
+  codec->plfreq = last_audio_decoder_->sample_rate_hz;
+  return 0;
+}
+
+void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
+  NetEqNetworkStatistics neteq_stat;
+  // NetEq function always returns zero, so we don't check the return value.
+  neteq_->NetworkStatistics(&neteq_stat);
+
+  acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
+  acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
+  acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
+  acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
+  acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
+  acm_stat->currentExpandRate = neteq_stat.expand_rate;
+  acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
+  acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
+  acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
+  acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
+  acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
+  acm_stat->addedSamples = neteq_stat.added_zero_samples;
+  acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
+  acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
+  acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
+  acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
+}
+
+int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
+                                      CodecInst* codec) const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  auto it = decoders_.find(payload_type);
+  if (it == decoders_.end()) {
+    LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
+                << static_cast<int>(payload_type);
+    return -1;
+  }
+  const Decoder& decoder = it->second;
+  *codec = *RentACodec::CodecInstById(
+      *RentACodec::CodecIdFromIndex(decoder.acm_codec_id));
+  codec->pltype = decoder.payload_type;
+  codec->channels = decoder.channels;
+  codec->plfreq = decoder.sample_rate_hz;
+  return 0;
+}
+
+int AcmReceiver::EnableNack(size_t max_nack_list_size) {
+  neteq_->EnableNack(max_nack_list_size);
+  return 0;
+}
+
+void AcmReceiver::DisableNack() {
+  neteq_->DisableNack();
+}
+
+std::vector<uint16_t> AcmReceiver::GetNackList(
+    int64_t round_trip_time_ms) const {
+  return neteq_->GetNackList(round_trip_time_ms);
+}
+
+void AcmReceiver::ResetInitialDelay() {
+  neteq_->SetMinimumDelay(0);
+  // TODO(turajs): Should NetEq Buffer be flushed?
+}
+
+const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder(
+    const RTPHeader& rtp_header,
+    uint8_t payload_type) const {
+  auto it = decoders_.find(rtp_header.payloadType);
+  const auto red_index =
+      RentACodec::CodecIndexFromId(RentACodec::CodecId::kRED);
+  if (red_index &&  // This ensures that RED is defined in WebRTC.
+      it != decoders_.end() && it->second.acm_codec_id == *red_index) {
+    // This is a RED packet, get the payload of the audio codec.
+    it = decoders_.find(payload_type & 0x7F);
+  }
+
+  // Check if the payload is registered.
+  return it != decoders_.end() ? &it->second : nullptr;
+}
+
+uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
+  // Down-cast the time to (32-6)-bit since we only care about
+  // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
+  // We masked 6 most significant bits of 32-bit so there is no overflow in
+  // the conversion from milliseconds to timestamp.
+  const uint32_t now_in_ms = static_cast<uint32_t>(
+      clock_->TimeInMilliseconds() & 0x03ffffff);
+  return static_cast<uint32_t>(
+      (decoder_sampling_rate / 1000) * now_in_ms);
+}
+
+void AcmReceiver::GetDecodingCallStatistics(
+    AudioDecodingCallStats* stats) const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  *stats = call_stats_.GetDecodingStatistics();
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.h
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.h
index bdc4b844e2..b150612f69 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -8,23 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
 
 #include <map>
+#include <string>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -35,8 +37,6 @@ class NetEq;
 
 namespace acm2 {
 
-class Nack;
-
 class AcmReceiver {
  public:
   struct Decoder {
@@ -44,7 +44,8 @@ class AcmReceiver {
     uint8_t payload_type;
     // This field is meaningful for codecs where both mono and
     // stereo versions are registered under the same ID.
-    int channels;
+    size_t channels;
+    int sample_rate_hz;
   };
 
   // Constructor of the class
@@ -67,8 +68,7 @@ class AcmReceiver {
   //                           <0 if NetEq returned an error.
   //
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* incoming_payload,
-                   size_t length_payload);
+                   rtc::ArrayView<const uint8_t> incoming_payload);
 
   //
   // Asks NetEq for 10 milliseconds of decoded audio.
@@ -92,28 +92,34 @@ class AcmReceiver {
   // Adds a new codec to the NetEq codec database.
   //
   // Input:
-  //   - acm_codec_id        : ACM codec ID.
+  //   - acm_codec_id        : ACM codec ID; -1 means external decoder.
   //   - payload_type        : payload type.
-  //   - audio_decoder       : pointer to a decoder object. If it is NULL
-  //                           then NetEq will internally create the decoder
-  //                           object. Otherwise, NetEq will store this pointer
-  //                           as the decoder corresponding with the given
-  //                           payload type. NetEq won't acquire the ownership
-  //                           of this pointer. It is up to the client of this
-  //                           class (ACM) to delete it. By providing
-  //                           |audio_decoder| ACM will have control over the
-  //                           decoder instance of the codec. This is essential
-  //                           for a codec like iSAC which encoder/decoder
-  //                           encoder has to know about decoder (bandwidth
-  //                           estimator that is updated at decoding time).
+  //   - sample_rate_hz      : sample rate.
+  //   - audio_decoder       : pointer to a decoder object. If it's null, then
+  //                           NetEq will internally create a decoder object
+  //                           based on the value of |acm_codec_id| (which
+  //                           mustn't be -1). Otherwise, NetEq will use the
+  //                           given decoder for the given payload type. NetEq
+  //                           won't take ownership of the decoder; it's up to
+  //                           the caller to delete it when it's no longer
+  //                           needed.
+  //
+  //                           Providing an existing decoder object here is
+  //                           necessary for external decoders, but may also be
+  //                           used for built-in decoders if NetEq doesn't have
+  //                           all the info it needs to construct them properly
+  //                           (e.g. iSAC, where the decoder needs to be paired
+  //                           with an encoder).
   //
   // Return value             : 0 if OK.
   //                           <0 if NetEq returned an error.
   //
   int AddCodec(int acm_codec_id,
                uint8_t payload_type,
-               int channels,
-               AudioDecoder* audio_decoder);
+               size_t channels,
+               int sample_rate_hz,
+               AudioDecoder* audio_decoder,
+               const std::string& name);
 
   //
   // Sets a minimum delay for packet buffer. The given delay is maintained,
@@ -146,45 +152,19 @@ class AcmReceiver {
   //
   int LeastRequiredDelayMs() const;
 
-  //
-  // Sets an initial delay of |delay_ms| milliseconds. This introduces a playout
-  // delay. Silence (zero signal) is played out until equivalent of |delay_ms|
-  // millisecond of audio is buffered. Then, NetEq maintains the delay.
-  //
-  // Input:
-  //   - delay_ms             : initial delay in milliseconds.
-  //
-  // Return value             : 0 if OK.
-  //                           <0 if NetEq returned an error.
-  //
-  int SetInitialDelay(int delay_ms);
-
   //
   // Resets the initial delay to zero.
   //
   void ResetInitialDelay();
 
-  //
-  // Get the current sampling frequency in Hz.
-  //
-  // Return value             : Sampling frequency in Hz.
-  //
-  int current_sample_rate_hz() const;
+  // Returns the sample rate of the decoder associated with the last incoming
+  // packet. If no packet of a registered non-CNG codec has been received, the
+  // return value is empty. Also, if the decoder was unregistered since the last
+  // packet was inserted, the return value is empty.
+  rtc::Optional<int> last_packet_sample_rate_hz() const;
 
-  //
-  // Sets the playout mode.
-  //
-  // Input:
-  //   - mode                 : an enumerator specifying the playout mode.
-  //
-  void SetPlayoutMode(AudioPlayoutMode mode);
-
-  //
-  // Get the current playout mode.
-  //
-  // Return value             : The current playout mode.
-  //
-  AudioPlayoutMode PlayoutMode() const;
+  // Returns last_output_sample_rate_hz from the NetEq instance.
+  int last_output_sample_rate_hz() const;
 
   //
   // Get the current network statistics from NetEq.
@@ -241,13 +221,6 @@ class AcmReceiver {
   //
   bool GetPlayoutTimestamp(uint32_t* timestamp);
 
-  //
-  // Return the index of the codec associated with the last non-CNG/non-DTMF
-  // received payload. If no non-CNG/non-DTMF payload is received -1 is
-  // returned.
-  //
-  int last_audio_codec_id() const;  // TODO(turajs): can be inline.
-
   //
   // Get the audio codec associated with the last non-CNG/non-DTMF received
   // payload. If no non-CNG/non-DTMF packet is received -1 is returned,
@@ -255,11 +228,6 @@ class AcmReceiver {
   //
   int LastAudioCodec(CodecInst* codec) const;
 
-  //
-  // Return payload type of RED if it is registered, otherwise return -1;
-  //
-  int RedPayloadType() const;
-
   //
   // Get a decoder given its registered payload-type.
   //
@@ -307,31 +275,21 @@ class AcmReceiver {
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
  private:
-  bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  int GetNumSyncPacketToInsert(uint16_t received_squence_number);
-
   const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
-                                    const uint8_t* payload) const
+                                    uint8_t payload_type) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
-  void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);
-
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
-  int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   // Used in GetAudio, declared as member to avoid allocating every 10ms.
   // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
   rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
-  rtc::scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
-  bool nack_enabled_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   // Decoders map is keyed by payload type
@@ -339,23 +297,11 @@ class AcmReceiver {
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
-
-  // Indicates if a non-zero initial delay is set, and the receiver is in
-  // AV-sync mode.
-  bool av_sync_;
-  rtc::scoped_ptr<InitialDelayManager> initial_delay_manager_;
-
-  // The following are defined as members to avoid creating them in every
-  // iteration. |missing_packets_sync_stream_| is *ONLY* used in InsertPacket().
-  // |late_packets_sync_stream_| is only used in GetAudio(). Both of these
-  // member variables are allocated only when we AV-sync is enabled, i.e.
-  // initial delay is set.
-  rtc::scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_;
-  rtc::scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
+  rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
similarity index 50%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index 269d19c81d..24ecc694ff 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -8,20 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 
 #include <algorithm>  // std::min
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -37,6 +35,19 @@ bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) {
     return true;
 }
 
+struct CodecIdInst {
+  explicit CodecIdInst(RentACodec::CodecId codec_id) {
+    const auto codec_ix = RentACodec::CodecIndexFromId(codec_id);
+    EXPECT_TRUE(codec_ix);
+    id = *codec_ix;
+    const auto codec_inst = RentACodec::CodecInstById(codec_id);
+    EXPECT_TRUE(codec_inst);
+    inst = *codec_inst;
+  }
+  int id;
+  CodecInst inst;
+};
+
 }  // namespace
 
 class AcmReceiverTestOldApi : public AudioPacketizationCallback,
@@ -46,7 +57,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
       : timestamp_(0),
         packet_sent_(false),
         last_packet_send_timestamp_(timestamp_),
-        last_frame_type_(kFrameEmpty) {
+        last_frame_type_(kEmptyFrame) {
     AudioCodingModule::Config config;
     acm_.reset(new AudioCodingModuleImpl(config));
     receiver_.reset(new AcmReceiver(config));
@@ -57,9 +68,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   void SetUp() override {
     ASSERT_TRUE(receiver_.get() != NULL);
     ASSERT_TRUE(acm_.get() != NULL);
-    for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) {
-      ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n]));
-    }
+    codecs_ = RentACodec::Database();
 
     acm_->InitializeReceiver();
     acm_->RegisterTransportCallback(this);
@@ -77,14 +86,14 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   void TearDown() override {}
 
   void InsertOnePacketOfSilence(int codec_id) {
-    CodecInst codec;
-    ACMCodecDB::Codec(codec_id, &codec);
+    CodecInst codec =
+        *RentACodec::CodecInstById(*RentACodec::CodecIdFromIndex(codec_id));
     if (timestamp_ == 0) {  // This is the first time inserting audio.
       ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     } else {
-      CodecInst current_codec;
-      ASSERT_EQ(0, acm_->SendCodec(&current_codec));
-      if (!CodecsEqual(codec, current_codec))
+      auto current_codec = acm_->SendCodec();
+      ASSERT_TRUE(current_codec);
+      if (!CodecsEqual(codec, *current_codec))
         ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     }
     AudioFrame frame;
@@ -103,13 +112,14 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
     }
   }
 
-  // Last element of id should be negative.
-  void AddSetOfCodecs(const int* id) {
-    int n = 0;
-    while (id[n] >= 0) {
-      ASSERT_EQ(0, receiver_->AddCodec(id[n], codecs_[id[n]].pltype,
-                                       codecs_[id[n]].channels, NULL));
-      ++n;
+  template <size_t N>
+  void AddSetOfCodecs(const RentACodec::CodecId(&ids)[N]) {
+    for (auto id : ids) {
+      const auto i = RentACodec::CodecIndexFromId(id);
+      ASSERT_TRUE(i);
+      ASSERT_EQ(
+          0, receiver_->AddCodec(*i, codecs_[*i].pltype, codecs_[*i].channels,
+                                 codecs_[*i].plfreq, nullptr, ""));
     }
   }
 
@@ -119,7 +129,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
                const uint8_t* payload_data,
                size_t payload_len_bytes,
                const RTPFragmentationHeader* fragmentation) override {
-    if (frame_type == kFrameEmpty)
+    if (frame_type == kEmptyFrame)
       return 0;
 
     rtp_header_.header.payloadType = payload_type;
@@ -130,8 +140,9 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
       rtp_header_.type.Audio.isCNG = true;
     rtp_header_.header.timestamp = timestamp;
 
-    int ret_val = receiver_->InsertPacket(rtp_header_, payload_data,
-                                          payload_len_bytes);
+    int ret_val = receiver_->InsertPacket(
+        rtp_header_,
+        rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
     if (ret_val < 0) {
       assert(false);
       return -1;
@@ -143,7 +154,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   }
 
   rtc::scoped_ptr<AcmReceiver> receiver_;
-  CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
+  rtc::ArrayView<const CodecInst> codecs_;
   rtc::scoped_ptr<AudioCodingModule> acm_;
   WebRtcRTPHeader rtp_header_;
   uint32_t timestamp_;
@@ -152,15 +163,21 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
   FrameType last_frame_type_;
 };
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
+#else
+#define MAYBE_AddCodecGetCodec AddCodecGetCodec
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
   // Add codec.
-  for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) {
+  for (size_t n = 0; n < codecs_.size(); ++n) {
     if (n & 0x1)  // Just add codecs with odd index.
-      EXPECT_EQ(0, receiver_->AddCodec(n, codecs_[n].pltype,
-                                       codecs_[n].channels, NULL));
+      EXPECT_EQ(0,
+                receiver_->AddCodec(n, codecs_[n].pltype, codecs_[n].channels,
+                                    codecs_[n].plfreq, NULL, ""));
   }
   // Get codec and compare.
-  for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) {
+  for (size_t n = 0; n < codecs_.size(); ++n) {
     CodecInst my_codec;
     if (n & 0x1) {
       // Codecs with odd index should match the reference.
@@ -175,55 +192,68 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
   }
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
-  const int codec_id = ACMCodecDB::kPCMA;
-  CodecInst ref_codec1;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &ref_codec1));
-  CodecInst ref_codec2 = ref_codec1;
-  ++ref_codec2.pltype;
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
+#else
+#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
+  const CodecIdInst codec1(RentACodec::CodecId::kPCMA);
+  CodecInst codec2 = codec1.inst;
+  ++codec2.pltype;
   CodecInst test_codec;
 
   // Register the same codec with different payloads.
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec1.pltype,
-                                   ref_codec1.channels, NULL));
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec2.pltype,
-                                   ref_codec2.channels, NULL));
+  EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec1.inst.pltype,
+                                   codec1.inst.channels, codec1.inst.plfreq,
+                                   nullptr, ""));
+  EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec2.pltype, codec2.channels,
+                                   codec2.plfreq, NULL, ""));
 
   // Both payload types should exist.
-  EXPECT_EQ(0, receiver_->DecoderByPayloadType(ref_codec1.pltype, &test_codec));
-  EXPECT_EQ(true, CodecsEqual(ref_codec1, test_codec));
-  EXPECT_EQ(0, receiver_->DecoderByPayloadType(ref_codec2.pltype, &test_codec));
-  EXPECT_EQ(true, CodecsEqual(ref_codec2, test_codec));
+  EXPECT_EQ(0,
+            receiver_->DecoderByPayloadType(codec1.inst.pltype, &test_codec));
+  EXPECT_EQ(true, CodecsEqual(codec1.inst, test_codec));
+  EXPECT_EQ(0, receiver_->DecoderByPayloadType(codec2.pltype, &test_codec));
+  EXPECT_EQ(true, CodecsEqual(codec2, test_codec));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
-  const int codec_id1 = ACMCodecDB::kPCMU;
-  CodecInst ref_codec1;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id1, &ref_codec1));
-  const int codec_id2 = ACMCodecDB::kPCMA;
-  CodecInst ref_codec2;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id2, &ref_codec2));
-  ref_codec2.pltype = ref_codec1.pltype;
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
+#else
+#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
+#endif
+TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
+  const CodecIdInst codec1(RentACodec::CodecId::kPCMU);
+  CodecIdInst codec2(RentACodec::CodecId::kPCMA);
+  codec2.inst.pltype = codec1.inst.pltype;
   CodecInst test_codec;
 
   // Register the same payload type with different codec ID.
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id1, ref_codec1.pltype,
-                                   ref_codec1.channels, NULL));
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id2, ref_codec2.pltype,
-                                   ref_codec2.channels, NULL));
+  EXPECT_EQ(0, receiver_->AddCodec(codec1.id, codec1.inst.pltype,
+                                   codec1.inst.channels, codec1.inst.plfreq,
+                                   nullptr, ""));
+  EXPECT_EQ(0, receiver_->AddCodec(codec2.id, codec2.inst.pltype,
+                                   codec2.inst.channels, codec2.inst.plfreq,
+                                   nullptr, ""));
 
   // Make sure that the last codec is used.
-  EXPECT_EQ(0, receiver_->DecoderByPayloadType(ref_codec2.pltype, &test_codec));
-  EXPECT_EQ(true, CodecsEqual(ref_codec2, test_codec));
+  EXPECT_EQ(0,
+            receiver_->DecoderByPayloadType(codec2.inst.pltype, &test_codec));
+  EXPECT_EQ(true, CodecsEqual(codec2.inst, test_codec));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
-  CodecInst codec;
-  const int codec_id = ACMCodecDB::kPCMA;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &codec));
-  const int payload_type = codec.pltype;
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id, codec.pltype,
-                                   codec.channels, NULL));
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
+#else
+#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
+  const CodecIdInst codec(RentACodec::CodecId::kPCMA);
+  const int payload_type = codec.inst.pltype;
+  EXPECT_EQ(
+      0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
+                             codec.inst.plfreq, nullptr, ""));
 
   // Remove non-existing codec should not fail. ACM1 legacy.
   EXPECT_EQ(0, receiver_->RemoveCodec(payload_type + 1));
@@ -232,61 +262,52 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
   EXPECT_EQ(0, receiver_->RemoveCodec(payload_type));
 
   // Ask for the removed codec, must fail.
-  EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &codec));
+  CodecInst ci;
+  EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &ci));
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
-  const int kCodecId[] = {
-      ACMCodecDB::kISAC, ACMCodecDB::kISACSWB, ACMCodecDB::kISACFB,
-      -1  // Terminator.
-  };
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SampleRate DISABLED_SampleRate
+#else
+#define MAYBE_SampleRate SampleRate
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
+  const RentACodec::CodecId kCodecId[] = {RentACodec::CodecId::kISAC,
+                                          RentACodec::CodecId::kISACSWB};
   AddSetOfCodecs(kCodecId);
 
   AudioFrame frame;
   const int kOutSampleRateHz = 8000;  // Different than codec sample rate.
-  int n = 0;
-  while (kCodecId[n] >= 0) {
-    const int num_10ms_frames = codecs_[kCodecId[n]].pacsize /
-        (codecs_[kCodecId[n]].plfreq / 100);
-    InsertOnePacketOfSilence(kCodecId[n]);
+  for (const auto codec_id : kCodecId) {
+    const CodecIdInst codec(codec_id);
+    const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
+    InsertOnePacketOfSilence(codec.id);
     for (int k = 0; k < num_10ms_frames; ++k) {
       EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
     }
-    EXPECT_EQ(std::min(32000, codecs_[kCodecId[n]].plfreq),
-              receiver_->current_sample_rate_hz());
-    ++n;
+    EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
   }
 }
 
-// Verify that the playout mode is set correctly.
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PlayoutMode)) {
-  receiver_->SetPlayoutMode(voice);
-  EXPECT_EQ(voice, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(streaming);
-  EXPECT_EQ(streaming, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(fax);
-  EXPECT_EQ(fax, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(off);
-  EXPECT_EQ(off, receiver_->PlayoutMode());
-}
-
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
   receiver_->EnableVad();
   EXPECT_TRUE(receiver_->vad_enabled());
-
-  const int id = ACMCodecDB::kPCM16Bwb;
-  ASSERT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels,
-                                   NULL));
+  const CodecIdInst codec(RentACodec::CodecId::kPCM16Bwb);
+  ASSERT_EQ(
+      0, receiver_->AddCodec(codec.id, codec.inst.pltype, codec.inst.channels,
+                             codec.inst.plfreq, nullptr, ""));
   const int kNumPackets = 5;
-  const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100);
+  const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
   AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
-    InsertOnePacketOfSilence(id);
+    InsertOnePacketOfSilence(codec.id);
     for (int k = 0; k < num_10ms_frames; ++k)
-      ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
+      ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
   }
   EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
 
@@ -294,33 +315,34 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
   EXPECT_FALSE(receiver_->vad_enabled());
 
   for (int n = 0; n < kNumPackets; ++n) {
-    InsertOnePacketOfSilence(id);
+    InsertOnePacketOfSilence(codec.id);
     for (int k = 0; k < num_10ms_frames; ++k)
-      ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
+      ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
   }
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }
 
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(LastAudioCodec)) {
-  const int kCodecId[] = {
-      ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB,
-      ACMCodecDB::kPCM16Bswb32kHz,
-      -1  // Terminator.
-  };
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
+#else
+#define MAYBE_LastAudioCodec LastAudioCodec
+#endif
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
+  const RentACodec::CodecId kCodecId[] = {
+      RentACodec::CodecId::kISAC, RentACodec::CodecId::kPCMA,
+      RentACodec::CodecId::kISACSWB, RentACodec::CodecId::kPCM16Bswb32kHz};
   AddSetOfCodecs(kCodecId);
 
-  const int kCngId[] = {  // Not including full-band.
-      ACMCodecDB::kCNNB, ACMCodecDB::kCNWB, ACMCodecDB::kCNSWB,
-      -1  // Terminator.
-  };
+  const RentACodec::CodecId kCngId[] = {
+      // Not including full-band.
+      RentACodec::CodecId::kCNNB, RentACodec::CodecId::kCNWB,
+      RentACodec::CodecId::kCNSWB};
   AddSetOfCodecs(kCngId);
 
   // Register CNG at sender side.
-  int n = 0;
-  while (kCngId[n] > 0) {
-    ASSERT_EQ(0, acm_->RegisterSendCodec(codecs_[kCngId[n]]));
-    ++n;
-  }
+  for (auto id : kCngId)
+    ASSERT_EQ(0, acm_->RegisterSendCodec(CodecIdInst(id).inst));
 
   CodecInst codec;
   // No audio payload is received.
@@ -329,26 +351,29 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(LastAudioCodec)) {
   // Start with sending DTX.
   ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr));
   packet_sent_ = false;
-  InsertOnePacketOfSilence(kCodecId[0]);  // Enough to test with one codec.
+  InsertOnePacketOfSilence(CodecIdInst(kCodecId[0]).id);  // Enough to test
+                                                          // with one codec.
   ASSERT_TRUE(packet_sent_);
   EXPECT_EQ(kAudioFrameCN, last_frame_type_);
 
   // Has received, only, DTX. Last Audio codec is undefined.
   EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
-  EXPECT_EQ(-1, receiver_->last_audio_codec_id());
+  EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());
+
+  for (auto id : kCodecId) {
+    const CodecIdInst c(id);
 
-  n = 0;
-  while (kCodecId[n] >= 0) {  // Loop over codecs.
     // Set DTX off to send audio payload.
     acm_->SetVAD(false, false, VADAggr);
     packet_sent_ = false;
-    InsertOnePacketOfSilence(kCodecId[n]);
+    InsertOnePacketOfSilence(c.id);
 
     // Sanity check if Actually an audio payload received, and it should be
     // of type "speech."
     ASSERT_TRUE(packet_sent_);
     ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
-    EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
+    EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
+              receiver_->last_packet_sample_rate_hz());
 
     // Set VAD on to send DTX. Then check if the "Last Audio codec" returns
     // the expected codec.
@@ -357,15 +382,16 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(LastAudioCodec)) {
     // Do as many encoding until a DTX is sent.
     while (last_frame_type_ != kAudioFrameCN) {
       packet_sent_ = false;
-      InsertOnePacketOfSilence(kCodecId[n]);
+      InsertOnePacketOfSilence(c.id);
       ASSERT_TRUE(packet_sent_);
     }
-    EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
+    EXPECT_EQ(rtc::Optional<int>(c.inst.plfreq),
+              receiver_->last_packet_sample_rate_hz());
     EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
-    EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec));
-    ++n;
+    EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
+#endif
 
 }  // namespace acm2
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.cc
similarity index 64%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.cc
index 97d87b1b3a..dfc3ef7e27 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 
 #include <assert.h>
 #include <string.h>
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 namespace acm2 {
@@ -28,40 +28,35 @@ ACMResampler::~ACMResampler() {
 int ACMResampler::Resample10Msec(const int16_t* in_audio,
                                  int in_freq_hz,
                                  int out_freq_hz,
-                                 int num_audio_channels,
-                                 int out_capacity_samples,
+                                 size_t num_audio_channels,
+                                 size_t out_capacity_samples,
                                  int16_t* out_audio) {
-  int in_length = in_freq_hz * num_audio_channels / 100;
-  int out_length = out_freq_hz * num_audio_channels / 100;
+  size_t in_length = in_freq_hz * num_audio_channels / 100;
   if (in_freq_hz == out_freq_hz) {
     if (out_capacity_samples < in_length) {
       assert(false);
       return -1;
     }
     memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
-    return in_length / num_audio_channels;
+    return static_cast<int>(in_length / num_audio_channels);
   }
 
   if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
                                     num_audio_channels) != 0) {
-    LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
-              num_audio_channels);
+    LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", " << out_freq_hz
+                  << ", " << num_audio_channels << ") failed.";
     return -1;
   }
 
-  out_length =
+  int out_length =
       resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
   if (out_length == -1) {
-    LOG_FERR4(LS_ERROR,
-              Resample,
-              in_audio,
-              in_length,
-              out_audio,
-              out_capacity_samples);
+    LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
+                  << out_audio << ", " << out_capacity_samples << ") failed.";
     return -1;
   }
 
-  return out_length / num_audio_channels;
+  return static_cast<int>(out_length / num_audio_channels);
 }
 
 }  // namespace acm2
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.h
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.h
index a8fc6b6f26..268db8b752 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_resampler.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/typedefs.h"
@@ -25,8 +25,8 @@ class ACMResampler {
   int Resample10Msec(const int16_t* in_audio,
                      int in_freq_hz,
                      int out_freq_hz,
-                     int num_audio_channels,
-                     int out_capacity_samples,
+                     size_t num_audio_channels,
+                     size_t out_capacity_samples,
                      int16_t* out_audio);
 
  private:
@@ -36,4 +36,4 @@ class ACMResampler {
 }  // namespace acm2
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
similarity index 80%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
index 41e0feb5ff..3a89a77487 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
@@ -16,7 +16,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 
@@ -30,7 +31,8 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
       acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
       audio_source_(audio_source),
       source_rate_hz_(source_rate_hz),
-      input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
+      input_block_size_samples_(
+          static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
       codec_registered_(false),
       test_duration_ms_(test_duration_ms),
       frame_type_(kAudioFrameSpeech),
@@ -50,18 +52,27 @@ bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
                                       int channels,
                                       int payload_type,
                                       int frame_size_samples) {
-  CHECK_EQ(0,
-           AudioCodingModule::Codec(
-               payload_name, &codec_, sampling_freq_hz, channels));
-  codec_.pltype = payload_type;
-  codec_.pacsize = frame_size_samples;
-  codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0);
+  CodecInst codec;
+  RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
+                                           sampling_freq_hz, channels));
+  codec.pltype = payload_type;
+  codec.pacsize = frame_size_samples;
+  codec_registered_ = (acm_->RegisterSendCodec(codec) == 0);
   input_frame_.num_channels_ = channels;
   assert(input_block_size_samples_ * input_frame_.num_channels_ <=
          AudioFrame::kMaxDataSizeSamples);
   return codec_registered_;
 }
 
+bool AcmSendTestOldApi::RegisterExternalCodec(
+    AudioEncoder* external_speech_encoder) {
+  acm_->RegisterExternalSendCodec(external_speech_encoder);
+  input_frame_.num_channels_ = external_speech_encoder->NumChannels();
+  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
+         AudioFrame::kMaxDataSizeSamples);
+  return codec_registered_ = true;
+}
+
 Packet* AcmSendTestOldApi::NextPacket() {
   assert(codec_registered_);
   if (filter_.test(static_cast<size_t>(payload_type_))) {
@@ -73,7 +84,8 @@ Packet* AcmSendTestOldApi::NextPacket() {
   // Insert audio and process until one packet is produced.
   while (clock_.TimeInMilliseconds() < test_duration_ms_) {
     clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
-    CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_));
+    RTC_CHECK(
+        audio_source_->Read(input_block_size_samples_, input_frame_.data_));
     if (input_frame_.num_channels_ > 1) {
       InputAudioFile::DuplicateInterleaved(input_frame_.data_,
                                            input_block_size_samples_,
@@ -81,8 +93,8 @@ Packet* AcmSendTestOldApi::NextPacket() {
                                            input_frame_.data_);
     }
     data_to_send_ = false;
-    CHECK_GE(acm_->Add10MsData(input_frame_), 0);
-    input_frame_.timestamp_ += input_block_size_samples_;
+    RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
     if (data_to_send_) {
       // Encoded packet received.
       return CreatePacket();
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
similarity index 81%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
index 52cb415ebf..ce68196a3f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -8,18 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
 
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
+class AudioEncoder;
 
 namespace test {
 class InputAudioFile;
@@ -40,6 +41,9 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
                      int payload_type,
                      int frame_size_samples);
 
+  // Registers an external send codec. Returns true on success, false otherwise.
+  bool RegisterExternalCodec(AudioEncoder* external_speech_encoder);
+
   // Returns the next encoded packet. Returns NULL if the test duration was
   // exceeded. Ownership of the packet is handed over to the caller.
   // Inherited from PacketSource.
@@ -67,9 +71,8 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
   rtc::scoped_ptr<AudioCodingModule> acm_;
   InputAudioFile* audio_source_;
   int source_rate_hz_;
-  const int input_block_size_samples_;
+  const size_t input_block_size_samples_;
   AudioFrame input_frame_;
-  CodecInst codec_;
   bool codec_registered_;
   int test_duration_ms_;
   // The following member variables are set whenever SendData() is called.
@@ -80,9 +83,9 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
   std::vector<uint8_t> last_payload_vec_;
   bool data_to_send_;
 
-  DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
 };
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
new file mode 100644
index 0000000000..c4dd349cc4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+
+// Create module
+AudioCodingModule* AudioCodingModule::Create(int id) {
+  Config config;
+  config.id = id;
+  config.clock = Clock::GetRealTimeClock();
+  return Create(config);
+}
+
+AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
+  Config config;
+  config.id = id;
+  config.clock = clock;
+  return Create(config);
+}
+
+AudioCodingModule* AudioCodingModule::Create(const Config& config) {
+  return new acm2::AudioCodingModuleImpl(config);
+}
+
+int AudioCodingModule::NumberOfCodecs() {
+  return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
+}
+
+int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
+  auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
+  if (!codec_id)
+    return -1;
+  auto ci = acm2::RentACodec::CodecInstById(*codec_id);
+  if (!ci)
+    return -1;
+  *codec = *ci;
+  return 0;
+}
+
+int AudioCodingModule::Codec(const char* payload_name,
+                             CodecInst* codec,
+                             int sampling_freq_hz,
+                             size_t channels) {
+  rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
+      payload_name, sampling_freq_hz, channels);
+  if (ci) {
+    *codec = *ci;
+    return 0;
+  } else {
+    // We couldn't find a matching codec, so set the parameters to unacceptable
+    // values and return.
+    codec->plname[0] = '\0';
+    codec->pltype = -1;
+    codec->pacsize = 0;
+    codec->rate = 0;
+    codec->plfreq = 0;
+    return -1;
+  }
+}
+
+int AudioCodingModule::Codec(const char* payload_name,
+                             int sampling_freq_hz,
+                             size_t channels) {
+  rtc::Optional<acm2::RentACodec::CodecId> ci =
+      acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
+                                        channels);
+  if (!ci)
+    return -1;
+  rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
+  return i ? *i : -1;
+}
+
+// Checks the validity of the parameters of the given codec
+bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
+  bool valid = acm2::RentACodec::IsCodecValid(codec);
+  if (!valid)
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
+                 "Invalid codec setting");
+  return valid;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
new file mode 100644
index 0000000000..ac302f0fe3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -0,0 +1,828 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+namespace {
+
+// TODO(turajs): the same functionality is used in NetEq. If both classes
+// need them, make it a static function in ACMCodecDB.
+bool IsCodecRED(const CodecInst& codec) {
+  return (STR_CASE_CMP(codec.plname, "RED") == 0);
+}
+
+bool IsCodecCN(const CodecInst& codec) {
+  return (STR_CASE_CMP(codec.plname, "CN") == 0);
+}
+
+// Stereo-to-mono can be used as in-place.
+int DownMix(const AudioFrame& frame,
+            size_t length_out_buff,
+            int16_t* out_buff) {
+  if (length_out_buff < frame.samples_per_channel_) {
+    return -1;
+  }
+  for (size_t n = 0; n < frame.samples_per_channel_; ++n)
+    out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
+  return 0;
+}
+
+// Mono-to-stereo can be used as in-place.
+int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
+  if (length_out_buff < frame.samples_per_channel_) {
+    return -1;
+  }
+  for (size_t n = frame.samples_per_channel_; n != 0; --n) {
+    size_t i = n - 1;
+    int16_t sample = frame.data_[i];
+    out_buff[2 * i + 1] = sample;
+    out_buff[2 * i] = sample;
+  }
+  return 0;
+}
+
+void ConvertEncodedInfoToFragmentationHeader(
+    const AudioEncoder::EncodedInfo& info,
+    RTPFragmentationHeader* frag) {
+  if (info.redundant.empty()) {
+    frag->fragmentationVectorSize = 0;
+    return;
+  }
+
+  frag->VerifyAndAllocateFragmentationHeader(
+      static_cast<uint16_t>(info.redundant.size()));
+  frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size());
+  size_t offset = 0;
+  for (size_t i = 0; i < info.redundant.size(); ++i) {
+    frag->fragmentationOffset[i] = offset;
+    offset += info.redundant[i].encoded_bytes;
+    frag->fragmentationLength[i] = info.redundant[i].encoded_bytes;
+    frag->fragmentationTimeDiff[i] = rtc::checked_cast<uint16_t>(
+        info.encoded_timestamp - info.redundant[i].encoded_timestamp);
+    frag->fragmentationPlType[i] = info.redundant[i].payload_type;
+  }
+}
+}  // namespace
+
+void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
+  if (value != last_value_ || first_time_) {
+    first_time_ = false;
+    last_value_ = value;
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
+  }
+}
+
+AudioCodingModuleImpl::AudioCodingModuleImpl(
+    const AudioCodingModule::Config& config)
+    : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+      id_(config.id),
+      expected_codec_ts_(0xD87F3F9F),
+      expected_in_ts_(0xD87F3F9F),
+      receiver_(config),
+      bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
+      previous_pltype_(255),
+      receiver_initialized_(false),
+      first_10ms_data_(false),
+      first_frame_(true),
+      callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+      packetization_callback_(NULL),
+      vad_callback_(NULL) {
+  if (InitializeReceiverSafe() < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Cannot initialize receiver");
+  }
+  WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
+}
+
+AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
+
+int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
+  AudioEncoder::EncodedInfo encoded_info;
+  uint8_t previous_pltype;
+
+  // Check if there is an encoder before.
+  if (!HaveValidEncoder("Process"))
+    return -1;
+
+  AudioEncoder* audio_encoder = rent_a_codec_.GetEncoderStack();
+  // Scale the timestamp to the codec's RTP timestamp rate.
+  uint32_t rtp_timestamp =
+      first_frame_ ? input_data.input_timestamp
+                   : last_rtp_timestamp_ +
+                         rtc::CheckedDivExact(
+                             input_data.input_timestamp - last_timestamp_,
+                             static_cast<uint32_t>(rtc::CheckedDivExact(
+                                 audio_encoder->SampleRateHz(),
+                                 audio_encoder->RtpTimestampRateHz())));
+  last_timestamp_ = input_data.input_timestamp;
+  last_rtp_timestamp_ = rtp_timestamp;
+  first_frame_ = false;
+
+  encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
+  encoded_info = audio_encoder->Encode(
+      rtp_timestamp, rtc::ArrayView<const int16_t>(
+                         input_data.audio, input_data.audio_channel *
+                                               input_data.length_per_channel),
+      encode_buffer_.size(), encode_buffer_.data());
+  encode_buffer_.SetSize(encoded_info.encoded_bytes);
+  bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
+  if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
+    // Not enough data.
+    return 0;
+  }
+  previous_pltype = previous_pltype_;  // Read it while we have the critsect.
+
+  RTPFragmentationHeader my_fragmentation;
+  ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
+  FrameType frame_type;
+  if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
+    frame_type = kEmptyFrame;
+    encoded_info.payload_type = previous_pltype;
+  } else {
+    RTC_DCHECK_GT(encode_buffer_.size(), 0u);
+    frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
+  }
+
+  {
+    CriticalSectionScoped lock(callback_crit_sect_.get());
+    if (packetization_callback_) {
+      packetization_callback_->SendData(
+          frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
+          encode_buffer_.data(), encode_buffer_.size(),
+          my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
+                                                       : nullptr);
+    }
+
+    if (vad_callback_) {
+      // Callback with VAD decision.
+      vad_callback_->InFrameType(frame_type);
+    }
+  }
+  previous_pltype_ = encoded_info.payload_type;
+  return static_cast<int32_t>(encode_buffer_.size());
+}
+
+/////////////////////////////////////////
+//   Sender
+//
+
+// Can be called multiple times for Codec, CNG, RED.
+int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!codec_manager_.RegisterEncoder(send_codec)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (!sp->speech_encoder && codec_manager_.GetCodecInst()) {
+    // We have no speech encoder, but we have a specification for making one.
+    AudioEncoder* enc =
+        rent_a_codec_.RentEncoder(*codec_manager_.GetCodecInst());
+    if (!enc)
+      return -1;
+    sp->speech_encoder = enc;
+  }
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  return 0;
+}
+
+void AudioCodingModuleImpl::RegisterExternalSendCodec(
+    AudioEncoder* external_speech_encoder) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  auto* sp = codec_manager_.GetStackParams();
+  sp->speech_encoder = external_speech_encoder;
+  rent_a_codec_.RentEncoderStack(sp);
+}
+
+// Get current send codec.
+rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  auto* ci = codec_manager_.GetCodecInst();
+  if (ci) {
+    return rtc::Optional<CodecInst>(*ci);
+  }
+  auto* enc = codec_manager_.GetStackParams()->speech_encoder;
+  if (enc) {
+    return rtc::Optional<CodecInst>(CodecManager::ForgeCodecInst(enc));
+  }
+  return rtc::Optional<CodecInst>();
+}
+
+// Get current send frequency.
+int AudioCodingModuleImpl::SendFrequency() const {
+  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
+               "SendFrequency()");
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+
+  const auto* enc = rent_a_codec_.GetEncoderStack();
+  if (!enc) {
+    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
+                 "SendFrequency Failed, no codec is registered");
+    return -1;
+  }
+
+  return enc->SampleRateHz();
+}
+
+void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  auto* enc = rent_a_codec_.GetEncoderStack();
+  if (enc) {
+    enc->SetTargetBitrate(bitrate_bps);
+  }
+}
+
+// Register a transport callback which will be called to deliver
+// the encoded buffers.
+int AudioCodingModuleImpl::RegisterTransportCallback(
+    AudioPacketizationCallback* transport) {
+  CriticalSectionScoped lock(callback_crit_sect_.get());
+  packetization_callback_ = transport;
+  return 0;
+}
+
+// Add 10MS of raw (PCM) audio data to the encoder.
+int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
+  InputData input_data;
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  int r = Add10MsDataInternal(audio_frame, &input_data);
+  return r < 0 ? r : Encode(input_data);
+}
+
+int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
+                                               InputData* input_data) {
+  if (audio_frame.samples_per_channel_ == 0) {
+    assert(false);
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Cannot Add 10 ms audio, payload length is zero");
+    return -1;
+  }
+
+  if (audio_frame.sample_rate_hz_ > 48000) {
+    assert(false);
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Cannot Add 10 ms audio, input frequency not valid");
+    return -1;
+  }
+
+  // If the length and frequency matches. We currently just support raw PCM.
+  if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
+      audio_frame.samples_per_channel_) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Cannot Add 10 ms audio, input frequency and length doesn't"
+                 " match");
+    return -1;
+  }
+
+  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Cannot Add 10 ms audio, invalid number of channels.");
+    return -1;
+  }
+
+  // Do we have a codec registered?
+  if (!HaveValidEncoder("Add10MsData")) {
+    return -1;
+  }
+
+  const AudioFrame* ptr_frame;
+  // Perform a resampling, also down-mix if it is required and can be
+  // performed before resampling (a down mix prior to resampling will take
+  // place if both primary and secondary encoders are mono and input is in
+  // stereo).
+  if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
+    return -1;
+  }
+
+  // Check whether we need an up-mix or down-mix?
+  const size_t current_num_channels =
+      rent_a_codec_.GetEncoderStack()->NumChannels();
+  const bool same_num_channels =
+      ptr_frame->num_channels_ == current_num_channels;
+
+  if (!same_num_channels) {
+    if (ptr_frame->num_channels_ == 1) {
+      if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
+        return -1;
+    } else {
+      if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
+        return -1;
+    }
+  }
+
+  // When adding data to encoders this pointer is pointing to an audio buffer
+  // with correct number of channels.
+  const int16_t* ptr_audio = ptr_frame->data_;
+
+  // For pushing data to primary, point the |ptr_audio| to correct buffer.
+  if (!same_num_channels)
+    ptr_audio = input_data->buffer;
+
+  input_data->input_timestamp = ptr_frame->timestamp_;
+  input_data->audio = ptr_audio;
+  input_data->length_per_channel = ptr_frame->samples_per_channel_;
+  input_data->audio_channel = current_num_channels;
+
+  return 0;
+}
+
+// Perform a resampling and down-mix if required. We down-mix only if
+// encoder is mono and input is stereo. In case of dual-streaming, both
+// encoders has to be mono for down-mix to take place.
+// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
+// is required, |*ptr_out| points to |in_frame|.
+int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
+                                               const AudioFrame** ptr_out) {
+  const auto* enc = rent_a_codec_.GetEncoderStack();
+  const bool resample = in_frame.sample_rate_hz_ != enc->SampleRateHz();
+
+  // This variable is true if primary codec and secondary codec (if exists)
+  // are both mono and input is stereo.
+  // TODO(henrik.lundin): This condition should probably be
+  //   in_frame.num_channels_ > enc->NumChannels()
+  const bool down_mix = in_frame.num_channels_ == 2 && enc->NumChannels() == 1;
+
+  if (!first_10ms_data_) {
+    expected_in_ts_ = in_frame.timestamp_;
+    expected_codec_ts_ = in_frame.timestamp_;
+    first_10ms_data_ = true;
+  } else if (in_frame.timestamp_ != expected_in_ts_) {
+    // TODO(turajs): Do we need a warning here.
+    expected_codec_ts_ +=
+        (in_frame.timestamp_ - expected_in_ts_) *
+        static_cast<uint32_t>(static_cast<double>(enc->SampleRateHz()) /
+                              static_cast<double>(in_frame.sample_rate_hz_));
+    expected_in_ts_ = in_frame.timestamp_;
+  }
+
+
+  if (!down_mix && !resample) {
+    // No pre-processing is required.
+    expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    *ptr_out = &in_frame;
+    return 0;
+  }
+
+  *ptr_out = &preprocess_frame_;
+  preprocess_frame_.num_channels_ = in_frame.num_channels_;
+  int16_t audio[WEBRTC_10MS_PCM_AUDIO];
+  const int16_t* src_ptr_audio = in_frame.data_;
+  int16_t* dest_ptr_audio = preprocess_frame_.data_;
+  if (down_mix) {
+    // If a resampling is required the output of a down-mix is written into a
+    // local buffer, otherwise, it will be written to the output frame.
+    if (resample)
+      dest_ptr_audio = audio;
+    if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
+      return -1;
+    preprocess_frame_.num_channels_ = 1;
+    // Set the input of the resampler is the down-mixed signal.
+    src_ptr_audio = audio;
+  }
+
+  preprocess_frame_.timestamp_ = expected_codec_ts_;
+  preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
+  preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
+  // If it is required, we have to do a resampling.
+  if (resample) {
+    // The result of the resampler is written to output frame.
+    dest_ptr_audio = preprocess_frame_.data_;
+
+    int samples_per_channel = resampler_.Resample10Msec(
+        src_ptr_audio, in_frame.sample_rate_hz_, enc->SampleRateHz(),
+        preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
+        dest_ptr_audio);
+
+    if (samples_per_channel < 0) {
+      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                   "Cannot add 10 ms audio, resampling failed");
+      return -1;
+    }
+    preprocess_frame_.samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel);
+    preprocess_frame_.sample_rate_hz_ = enc->SampleRateHz();
+  }
+
+  expected_codec_ts_ +=
+      static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
+  expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+
+  return 0;
+}
+
+/////////////////////////////////////////
+//   (RED) Redundant Coding
+//
+
+bool AudioCodingModuleImpl::REDStatus() const {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  return codec_manager_.GetStackParams()->use_red;
+}
+
+// Configure RED status i.e on/off.
+int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
+#ifdef WEBRTC_CODEC_RED
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!codec_manager_.SetCopyRed(enable_red)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  return 0;
+#else
+  WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
+               "  WEBRTC_CODEC_RED is undefined");
+  return -1;
+#endif
+}
+
+/////////////////////////////////////////
+//   (FEC) Forward Error Correction (codec internal)
+//
+
+bool AudioCodingModuleImpl::CodecFEC() const {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  return codec_manager_.GetStackParams()->use_codec_fec;
+}
+
+int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!codec_manager_.SetCodecFEC(enable_codec_fec)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  if (enable_codec_fec) {
+    return sp->use_codec_fec ? 0 : -1;
+  } else {
+    RTC_DCHECK(!sp->use_codec_fec);
+    return 0;
+  }
+}
+
+int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (HaveValidEncoder("SetPacketLossRate")) {
+    rent_a_codec_.GetEncoderStack()->SetProjectedPacketLossRate(loss_rate /
+                                                                100.0);
+  }
+  return 0;
+}
+
+/////////////////////////////////////////
+//   (VAD) Voice Activity Detection
+//
+int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
+                                  bool enable_vad,
+                                  ACMVADMode mode) {
+  // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
+  RTC_DCHECK_EQ(enable_dtx, enable_vad);
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!codec_manager_.SetVAD(enable_dtx, mode)) {
+    return -1;
+  }
+  auto* sp = codec_manager_.GetStackParams();
+  if (sp->speech_encoder)
+    rent_a_codec_.RentEncoderStack(sp);
+  return 0;
+}
+
+// Get VAD/DTX settings.
+int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
+                               ACMVADMode* mode) const {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  const auto* sp = codec_manager_.GetStackParams();
+  *dtx_enabled = *vad_enabled = sp->use_cng;
+  *mode = sp->vad_mode;
+  return 0;
+}
+
+/////////////////////////////////////////
+//   Receiver
+//
+
+int AudioCodingModuleImpl::InitializeReceiver() {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  return InitializeReceiverSafe();
+}
+
+// Initialize receiver, resets codec database etc.
+int AudioCodingModuleImpl::InitializeReceiverSafe() {
+  // If the receiver is already initialized then we want to destroy any
+  // existing decoders. After a call to this function, we should have a clean
+  // start-up.
+  if (receiver_initialized_) {
+    if (receiver_.RemoveAllCodecs() < 0)
+      return -1;
+  }
+  receiver_.set_id(id_);
+  receiver_.ResetInitialDelay();
+  receiver_.SetMinimumDelay(0);
+  receiver_.SetMaximumDelay(0);
+  receiver_.FlushBuffers();
+
+  // Register RED and CN.
+  auto db = RentACodec::Database();
+  for (size_t i = 0; i < db.size(); i++) {
+    if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
+      if (receiver_.AddCodec(static_cast<int>(i),
+                             static_cast<uint8_t>(db[i].pltype), 1,
+                             db[i].plfreq, nullptr, db[i].plname) < 0) {
+        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                     "Cannot register master codec.");
+        return -1;
+      }
+    }
+  }
+  receiver_initialized_ = true;
+  return 0;
+}
+
+// Get current receive frequency.
+int AudioCodingModuleImpl::ReceiveFrequency() const {
+  const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
+  return last_packet_sample_rate ? *last_packet_sample_rate
+                                 : receiver_.last_output_sample_rate_hz();
+}
+
+// Get current playout frequency.
+int AudioCodingModuleImpl::PlayoutFrequency() const {
+  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
+               "PlayoutFrequency()");
+  return receiver_.last_output_sample_rate_hz();
+}
+
+// Register possible receive codecs, can be called multiple times,
+// for codecs, CNG (NB, WB and SWB), DTMF, RED.
+int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  RTC_DCHECK(receiver_initialized_);
+  if (codec.channels > 2) {
+    LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
+    return -1;
+  }
+
+  auto codec_id =
+      RentACodec::CodecIdByParams(codec.plname, codec.plfreq, codec.channels);
+  if (!codec_id) {
+    LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec";
+    return -1;
+  }
+  auto codec_index = RentACodec::CodecIndexFromId(*codec_id);
+  RTC_CHECK(codec_index) << "Invalid codec ID: " << static_cast<int>(*codec_id);
+
+  // Check if the payload-type is valid.
+  if (!RentACodec::IsPayloadTypeValid(codec.pltype)) {
+    LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
+                    << codec.plname;
+    return -1;
+  }
+
+  // Get |decoder| associated with |codec|. |decoder| is NULL if |codec| does
+  // not own its decoder.
+  return receiver_.AddCodec(
+      *codec_index, codec.pltype, codec.channels, codec.plfreq,
+      STR_CASE_CMP(codec.plname, "isac") == 0 ? rent_a_codec_.RentIsacDecoder()
+                                              : nullptr,
+      codec.plname);
+}
+
+int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
+    int rtp_payload_type,
+    AudioDecoder* external_decoder,
+    int sample_rate_hz,
+    int num_channels,
+    const std::string& name) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  RTC_DCHECK(receiver_initialized_);
+  if (num_channels > 2 || num_channels < 0) {
+    LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
+    return -1;
+  }
+
+  // Check if the payload-type is valid.
+  if (!RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
+    LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
+                    << " for external decoder.";
+    return -1;
+  }
+
+  return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
+                            sample_rate_hz, external_decoder, name);
+}
+
+// Get current received codec.
+int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  return receiver_.LastAudioCodec(current_codec);
+}
+
+// Incoming packet from network parsed and ready for decode.
+int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
+                                          const size_t payload_length,
+                                          const WebRtcRTPHeader& rtp_header) {
+  return receiver_.InsertPacket(
+      rtp_header,
+      rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
+}
+
+// Minimum playout delay (Used for lip-sync).
+int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
+  if ((time_ms < 0) || (time_ms > 10000)) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Delay must be in the range of 0-1000 milliseconds.");
+    return -1;
+  }
+  return receiver_.SetMinimumDelay(time_ms);
+}
+
+int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
+  if ((time_ms < 0) || (time_ms > 10000)) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "Delay must be in the range of 0-1000 milliseconds.");
+    return -1;
+  }
+  return receiver_.SetMaximumDelay(time_ms);
+}
+
+// Get 10 milliseconds of raw audio data to play out.
+// Automatic resample to the requested frequency.
+int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
+                                           AudioFrame* audio_frame) {
+  // GetAudio always returns 10 ms, at the requested sample rate.
+  if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "PlayoutData failed, RecOut Failed");
+    return -1;
+  }
+  audio_frame->id_ = id_;
+  return 0;
+}
+
+/////////////////////////////////////////
+//   Statistics
+//
+
+// TODO(turajs) change the return value to void. Also change the corresponding
+// NetEq function.
+int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
+  receiver_.GetNetworkStatistics(statistics);
+  return 0;
+}
+
+int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
+  WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
+               "RegisterVADCallback()");
+  CriticalSectionScoped lock(callback_crit_sect_.get());
+  vad_callback_ = vad_callback;
+  return 0;
+}
+
+// TODO(kwiberg): Remove this method, and have callers call IncomingPacket
+// instead. The translation logic and state belong with them, not with
+// AudioCodingModuleImpl.
+int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
+                                           size_t payload_length,
+                                           uint8_t payload_type,
+                                           uint32_t timestamp) {
+  // We are not acquiring any lock when interacting with |aux_rtp_header_| no
+  // other method uses this member variable.
+  if (!aux_rtp_header_) {
+    // This is the first time that we are using |dummy_rtp_header_|
+    // so we have to create it.
+    aux_rtp_header_.reset(new WebRtcRTPHeader);
+    aux_rtp_header_->header.payloadType = payload_type;
+    // Don't matter in this case.
+    aux_rtp_header_->header.ssrc = 0;
+    aux_rtp_header_->header.markerBit = false;
+    // Start with random numbers.
+    aux_rtp_header_->header.sequenceNumber = 0x1234;  // Arbitrary.
+    aux_rtp_header_->type.Audio.channel = 1;
+  }
+
+  aux_rtp_header_->header.timestamp = timestamp;
+  IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
+  // Get ready for the next payload.
+  aux_rtp_header_->header.sequenceNumber++;
+  return 0;
+}
+
+int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!HaveValidEncoder("SetOpusApplication")) {
+    return -1;
+  }
+  AudioEncoder::Application app;
+  switch (application) {
+    case kVoip:
+      app = AudioEncoder::Application::kSpeech;
+      break;
+    case kAudio:
+      app = AudioEncoder::Application::kAudio;
+      break;
+    default:
+      FATAL();
+      return 0;
+  }
+  return rent_a_codec_.GetEncoderStack()->SetApplication(app) ? 0 : -1;
+}
+
+// Informs Opus encoder of the maximum playback rate the receiver will render.
+int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
+    return -1;
+  }
+  rent_a_codec_.GetEncoderStack()->SetMaxPlaybackRate(frequency_hz);
+  return 0;
+}
+
+int AudioCodingModuleImpl::EnableOpusDtx() {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!HaveValidEncoder("EnableOpusDtx")) {
+    return -1;
+  }
+  return rent_a_codec_.GetEncoderStack()->SetDtx(true) ? 0 : -1;
+}
+
+int AudioCodingModuleImpl::DisableOpusDtx() {
+  CriticalSectionScoped lock(acm_crit_sect_.get());
+  if (!HaveValidEncoder("DisableOpusDtx")) {
+    return -1;
+  }
+  return rent_a_codec_.GetEncoderStack()->SetDtx(false) ? 0 : -1;
+}
+
+int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
+  return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1;
+}
+
+bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
+  if (!rent_a_codec_.GetEncoderStack()) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
+                 "%s failed: No send codec is registered.", caller_name);
+    return false;
+  }
+  return true;
+}
+
+int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
+  return receiver_.RemoveCodec(payload_type);
+}
+
+int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
+  return receiver_.EnableNack(max_nack_list_size);
+}
+
+void AudioCodingModuleImpl::DisableNack() {
+  receiver_.DisableNack();
+}
+
+std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
+    int64_t round_trip_time_ms) const {
+  return receiver_.GetNackList(round_trip_time_ms);
+}
+
+int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
+  return receiver_.LeastRequiredDelayMs();
+}
+
+void AudioCodingModuleImpl::GetDecodingCallStatistics(
+      AudioDecodingCallStats* call_stats) const {
+  receiver_.GetDecodingCallStatistics(call_stats);
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
similarity index 53%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 1e9cbb989b..926671f199 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -8,19 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 
+#include <string>
 #include <vector>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 
 namespace webrtc {
 
@@ -29,40 +30,33 @@ class AudioCodingImpl;
 
 namespace acm2 {
 
-class ACMDTMFDetection;
-class ACMGenericCodec;
-
-class AudioCodingModuleImpl : public AudioCodingModule {
+class AudioCodingModuleImpl final : public AudioCodingModule {
  public:
   friend webrtc::AudioCodingImpl;
 
   explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
-  ~AudioCodingModuleImpl();
+  ~AudioCodingModuleImpl() override;
 
   /////////////////////////////////////////
   //   Sender
   //
 
-  // Reset send codec.
-  int ResetEncoder() override;
-
   // Can be called multiple times for Codec, CNG, RED.
   int RegisterSendCodec(const CodecInst& send_codec) override;
 
+  void RegisterExternalSendCodec(
+      AudioEncoder* external_speech_encoder) override;
+
   // Get current send codec.
-  int SendCodec(CodecInst* current_codec) const override;
+  rtc::Optional<CodecInst> SendCodec() const override;
 
   // Get current send frequency.
   int SendFrequency() const override;
 
-  // Get encode bit-rate.
-  // Adaptive rate codecs return their current encode target rate, while other
-  // codecs return there long-term average or their fixed rate.
-  int SendBitrate() const override;
-
-  // Set available bandwidth, inform the encoder about the
-  // estimated bandwidth received from the remote party.
-  int SetReceivedEstimatedBandwidth(int bw) override;
+  // Sets the bitrate to the specified value in bits/sec. In case the codec does
+  // not support the requested value it will choose an appropriate value
+  // instead.
+  void SetBitRate(int bitrate_bps) override;
 
   // Register a transport callback which will be
   // called to deliver the encoded buffers.
@@ -117,9 +111,6 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   // Initialize receiver, resets codec database etc.
   int InitializeReceiver() override;
 
-  // Reset the decoder state.
-  int ResetDecoder() override;
-
   // Get current receive frequency.
   int ReceiveFrequency() const override;
 
@@ -130,14 +121,15 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   // for codecs, CNG, DTMF, RED.
   int RegisterReceiveCodec(const CodecInst& receive_codec) override;
 
+  int RegisterExternalReceiveCodec(int rtp_payload_type,
+                                   AudioDecoder* external_decoder,
+                                   int sample_rate_hz,
+                                   int num_channels,
+                                   const std::string& name) override;
+
   // Get current received codec.
   int ReceiveCodec(CodecInst* current_codec) const override;
 
-  int RegisterDecoder(int acm_codec_id,
-                      uint8_t payload_type,
-                      int channels,
-                      AudioDecoder* audio_decoder);
-
   // Incoming packet from network parsed and ready for decode.
   int IncomingPacket(const uint8_t* incoming_payload,
                      const size_t payload_length,
@@ -159,31 +151,6 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   // Smallest latency NetEq will maintain.
   int LeastRequiredDelayMs() const override;
 
-  // Impose an initial delay on playout. ACM plays silence until |delay_ms|
-  // audio is accumulated in NetEq buffer, then starts decoding payloads.
-  int SetInitialPlayoutDelay(int delay_ms) override;
-
-  // TODO(turajs): DTMF playout is always activated in NetEq these APIs should
-  // be removed, as well as all VoE related APIs and methods.
-  //
-  // Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
-  // tone.
-  int SetDtmfPlayoutStatus(bool enable) override { return 0; }
-
-  // Get Dtmf playout status.
-  bool DtmfPlayoutStatus() const override { return true; }
-
-  // Estimate the Bandwidth based on the incoming stream, needed
-  // for one way audio where the RTCP send the BW estimate.
-  // This is also done in the RTP module .
-  int DecoderEstimatedBandwidth() const override;
-
-  // Set playout mode voice, fax.
-  int SetPlayoutMode(AudioPlayoutMode mode) override;
-
-  // Get playout mode voice, fax.
-  AudioPlayoutMode PlayoutMode() const override;
-
   // Get playout timestamp.
   int PlayoutTimestamp(uint32_t* timestamp) override;
 
@@ -197,34 +164,13 @@ class AudioCodingModuleImpl : public AudioCodingModule {
 
   int GetNetworkStatistics(NetworkStatistics* statistics) override;
 
-  // GET RED payload for iSAC. The method id called when 'this' ACM is
-  // the default ACM.
-  // TODO(henrik.lundin) Not used. Remove?
-  int REDPayloadISAC(int isac_rate,
-                     int isac_bw_estimate,
-                     uint8_t* payload,
-                     int16_t* length_bytes);
-
-  int ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) override;
-
-  int IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx) override;
-
-  int SetISACMaxRate(int max_bit_per_sec) override;
-
-  int SetISACMaxPayloadSize(int max_size_bytes) override;
-
-  int ConfigISACBandwidthEstimator(int frame_size_ms,
-                                   int rate_bit_per_sec,
-                                   bool enforce_frame_size = false) override;
-
-  int SetOpusApplication(OpusApplicationMode application,
-                         bool disable_dtx_if_needed) override;
+  int SetOpusApplication(OpusApplicationMode application) override;
 
   // If current send codec is Opus, informs it about the maximum playback rate
   // the receiver will render.
   int SetOpusMaxPlaybackRate(int frequency_hz) override;
 
-  int EnableOpusDtx(bool force_voip) override;
+  int EnableOpusDtx() override;
 
   int DisableOpusDtx() override;
 
@@ -242,15 +188,33 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   struct InputData {
     uint32_t input_timestamp;
     const int16_t* audio;
-    uint16_t length_per_channel;
-    uint8_t audio_channel;
+    size_t length_per_channel;
+    size_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
   };
 
-  int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data);
-  int Encode(const InputData& input_data);
+  // This member class writes values to the named UMA histogram, but only if
+  // the value has changed since the last time (and always for the first call).
+  class ChangeLogger {
+   public:
+    explicit ChangeLogger(const std::string& histogram_name)
+        : histogram_name_(histogram_name) {}
+    // Logs the new value if it is different from the last logged value, or if
+    // this is the first call.
+    void MaybeLog(int value);
+
+   private:
+    int last_value_ = 0;
+    int first_time_ = true;
+    const std::string histogram_name_;
+  };
+
+  int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
+      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+  int Encode(const InputData& input_data)
+      EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
@@ -276,13 +240,16 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   // to |index|.
   int UpdateUponReceivingCodec(int index);
 
-  CriticalSectionWrapper* acm_crit_sect_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> acm_crit_sect_;
+  rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
   int id_;  // TODO(henrik.lundin) Make const.
   uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
   uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
   ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
   AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
+  ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
   CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_);
+  RentACodec rent_a_codec_ GUARDED_BY(acm_crit_sect_);
 
   // This is to keep track of CN instances where we can send DTMFs.
   uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
@@ -293,7 +260,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   // IMPORTANT: this variable is only used in IncomingPayload(), therefore,
   // no lock acquired when interacting with this variable. If it is going to
   // be used in other methods, locks need to be taken.
-  WebRtcRTPHeader* aux_rtp_header_;
+  rtc::scoped_ptr<WebRtcRTPHeader> aux_rtp_header_;
 
   bool receiver_initialized_ GUARDED_BY(acm_crit_sect_);
 
@@ -304,97 +271,13 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
   uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
 
-  CriticalSectionWrapper* callback_crit_sect_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
 };
 
 }  // namespace acm2
-
-class AudioCodingImpl : public AudioCoding {
- public:
-  AudioCodingImpl(const Config& config) {
-    AudioCodingModule::Config config_old = config.ToOldConfig();
-    acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
-    acm_old_->RegisterTransportCallback(config.transport);
-    acm_old_->RegisterVADCallback(config.vad_callback);
-    acm_old_->SetDtmfPlayoutStatus(config.play_dtmf);
-    if (config.initial_playout_delay_ms > 0) {
-      acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms);
-    }
-    playout_frequency_hz_ = config.playout_frequency_hz;
-  }
-
-  ~AudioCodingImpl() override{};
-
-  bool RegisterSendCodec(AudioEncoder* send_codec) override;
-
-  bool RegisterSendCodec(int encoder_type,
-                         uint8_t payload_type,
-                         int frame_size_samples = 0) override;
-
-  const AudioEncoder* GetSenderInfo() const override;
-
-  const CodecInst* GetSenderCodecInst() override;
-
-  int Add10MsAudio(const AudioFrame& audio_frame) override;
-
-  const ReceiverInfo* GetReceiverInfo() const override;
-
-  bool RegisterReceiveCodec(AudioDecoder* receive_codec) override;
-
-  bool RegisterReceiveCodec(int decoder_type, uint8_t payload_type) override;
-
-  bool InsertPacket(const uint8_t* incoming_payload,
-                    size_t payload_len_bytes,
-                    const WebRtcRTPHeader& rtp_info) override;
-
-  bool InsertPayload(const uint8_t* incoming_payload,
-                     size_t payload_len_byte,
-                     uint8_t payload_type,
-                     uint32_t timestamp) override;
-
-  bool SetMinimumPlayoutDelay(int time_ms) override;
-
-  bool SetMaximumPlayoutDelay(int time_ms) override;
-
-  int LeastRequiredDelayMs() const override;
-
-  bool PlayoutTimestamp(uint32_t* timestamp) override;
-
-  bool Get10MsAudio(AudioFrame* audio_frame) override;
-
-  bool GetNetworkStatistics(NetworkStatistics* network_statistics) override;
-
-  bool EnableNack(size_t max_nack_list_size) override;
-
-  void DisableNack() override;
-
-  bool SetVad(bool enable_dtx, bool enable_vad, ACMVADMode vad_mode) override;
-
-  std::vector<uint16_t> GetNackList(int round_trip_time_ms) const override;
-
-  void GetDecodingCallStatistics(
-      AudioDecodingCallStats* call_stats) const override;
-
- private:
-  // Temporary method to be used during redesign phase.
-  // Maps |codec_type| (a value from the anonymous enum in acm2::ACMCodecDB) to
-  // |codec_name|, |sample_rate_hz|, and |channels|.
-  // TODO(henrik.lundin) Remove this when no longer needed.
-  static bool MapCodecTypeToParameters(int codec_type,
-                                       std::string* codec_name,
-                                       int* sample_rate_hz,
-                                       int* channels);
-
-  int playout_frequency_hz_;
-  // TODO(henrik.lundin): All members below this line are temporary and should
-  // be removed after refactoring is completed.
-  rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
-  CodecInst current_send_codec_;
-};
-
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
similarity index 58%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 81ae8aad13..6f82a96ee5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -8,17 +8,26 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <stdio.h>
 #include <string.h>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/md5digest.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
@@ -26,14 +35,16 @@
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::_;
 
 namespace webrtc {
 
@@ -81,7 +92,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
  public:
   PacketizationCallbackStubOldApi()
       : num_calls_(0),
-        last_frame_type_(kFrameEmpty),
+        last_frame_type_(kEmptyFrame),
         last_payload_type_(-1),
         last_timestamp_(0),
         crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
@@ -226,7 +237,12 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
 
 // Check if the statistics are initialized correctly. Before any call to ACM
 // all fields have to be zero.
-TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_InitializedToZero DISABLED_InitializedToZero
+#else
+#define MAYBE_InitializedToZero InitializedToZero
+#endif
+TEST_F(AudioCodingModuleTestOldApi, MAYBE_InitializedToZero) {
   RegisterCodec();
   AudioDecodingCallStats stats;
   acm_->GetDecodingCallStatistics(&stats);
@@ -238,34 +254,15 @@ TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) {
   EXPECT_EQ(0, stats.decoded_plc_cng);
 }
 
-// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
-// should result in generating silence, check the associated field.
-TEST_F(AudioCodingModuleTestOldApi,
-       DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
-  RegisterCodec();
-  AudioDecodingCallStats stats;
-  const int kInitialDelay = 100;
-
-  acm_->SetInitialPlayoutDelay(kInitialDelay);
-
-  int num_calls = 0;
-  for (int time_ms = 0; time_ms < kInitialDelay;
-       time_ms += kFrameSizeMs, ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
 // Insert some packets and pull audio. Check statistics are valid. Then,
 // simulate packet loss and check if PLC and PLC-to-CNG statistics are
 // correctly updated.
-TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NetEqCalls DISABLED_NetEqCalls
+#else
+#define MAYBE_NetEqCalls NetEqCalls
+#endif
+TEST_F(AudioCodingModuleTestOldApi, MAYBE_NetEqCalls) {
   RegisterCodec();
   AudioDecodingCallStats stats;
   const int kNumNormalCalls = 10;
@@ -303,8 +300,9 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
   EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
-  EXPECT_GT(audio_frame.num_channels_, 0);
-  EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
+  EXPECT_GT(audio_frame.num_channels_, 0u);
+  EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
+            audio_frame.samples_per_channel_);
   EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
 }
 
@@ -330,6 +328,7 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
   EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
 }
 
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
 // Verifies that the RTP timestamp series is not reset when the codec is
 // changed.
 TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
@@ -364,6 +363,7 @@ TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
     expected_ts += codec_.pacsize;
   }
 }
+#endif
 
 // Introduce this class to set different expectations on the number of encoded
 // bytes. This class expects all encoded packets to be 9 bytes (matching one
@@ -397,18 +397,18 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
       int ix;
       FrameType type;
     } expectation[] = {{2, kAudioFrameCN},
-                       {5, kFrameEmpty},
-                       {8, kFrameEmpty},
+                       {5, kEmptyFrame},
+                       {8, kEmptyFrame},
                        {11, kAudioFrameCN},
-                       {14, kFrameEmpty},
-                       {17, kFrameEmpty},
+                       {14, kEmptyFrame},
+                       {17, kEmptyFrame},
                        {20, kAudioFrameCN},
-                       {23, kFrameEmpty},
-                       {26, kFrameEmpty},
-                       {29, kFrameEmpty},
+                       {23, kEmptyFrame},
+                       {26, kEmptyFrame},
+                       {29, kEmptyFrame},
                        {32, kAudioFrameCN},
-                       {35, kFrameEmpty},
-                       {38, kFrameEmpty}};
+                       {35, kEmptyFrame},
+                       {38, kEmptyFrame}};
     for (int i = 0; i < kLoops; ++i) {
       int num_calls_before = packet_cb_.num_calls();
       EXPECT_EQ(i / blocks_per_packet, num_calls_before);
@@ -428,7 +428,7 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
 
 // Checks that the transport callback is invoked once per frame period of the
 // underlying speech encoder, even when comfort noise is produced.
-// Also checks that the frame type is kAudioFrameCN or kFrameEmpty.
+// Also checks that the frame type is kAudioFrameCN or kEmptyFrame.
 // This test and the next check the same thing, but differ in the order of
 // speech codec and CNG registration.
 TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
@@ -462,11 +462,9 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
 
   AudioCodingModuleMtTestOldApi()
       : AudioCodingModuleTestOldApi(),
-        send_thread_(ThreadWrapper::CreateThread(CbSendThread, this, "send")),
-        insert_packet_thread_(ThreadWrapper::CreateThread(
-            CbInsertPacketThread, this, "insert_packet")),
-        pull_audio_thread_(ThreadWrapper::CreateThread(
-            CbPullAudioThread, this, "pull_audio")),
+        send_thread_(CbSendThread, this, "send"),
+        insert_packet_thread_(CbInsertPacketThread, this, "insert_packet"),
+        pull_audio_thread_(CbPullAudioThread, this, "pull_audio"),
         test_complete_(EventWrapper::Create()),
         send_count_(0),
         insert_packet_count_(0),
@@ -484,19 +482,19 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   }
 
   void StartThreads() {
-    ASSERT_TRUE(send_thread_->Start());
-    send_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(insert_packet_thread_->Start());
-    insert_packet_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(pull_audio_thread_->Start());
-    pull_audio_thread_->SetPriority(kRealtimePriority);
+    send_thread_.Start();
+    send_thread_.SetPriority(rtc::kRealtimePriority);
+    insert_packet_thread_.Start();
+    insert_packet_thread_.SetPriority(rtc::kRealtimePriority);
+    pull_audio_thread_.Start();
+    pull_audio_thread_.SetPriority(rtc::kRealtimePriority);
   }
 
   void TearDown() {
     AudioCodingModuleTestOldApi::TearDown();
-    pull_audio_thread_->Stop();
-    send_thread_->Stop();
-    insert_packet_thread_->Stop();
+    pull_audio_thread_.Stop();
+    send_thread_.Stop();
+    insert_packet_thread_.Stop();
   }
 
   EventTypeWrapper RunTest() {
@@ -576,9 +574,9 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
     return true;
   }
 
-  rtc::scoped_ptr<ThreadWrapper> send_thread_;
-  rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
-  rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
+  rtc::PlatformThread send_thread_;
+  rtc::PlatformThread insert_packet_thread_;
+  rtc::PlatformThread pull_audio_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
   int send_count_;
   int insert_packet_count_;
@@ -588,7 +586,12 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
 
-TEST_F(AudioCodingModuleMtTestOldApi, DoTest) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
 
@@ -661,7 +664,11 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   }
 
   void InsertAudio() {
-    memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
+    // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
+    // this call confuses the number of samples with the number of bytes, and
+    // ends up copying only half of what it should.
+    memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
+           kNumSamples10ms);
     AudioCodingModuleTestOldApi::InsertAudio();
   }
 
@@ -688,26 +695,209 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmIsacMtTestOldApi, DoTest) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
+#endif
+
+class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
+ protected:
+  static const int kRegisterAfterNumPackets = 5;
+  static const int kNumPackets = 10;
+  static const int kPacketSizeMs = 30;
+  static const int kPacketSizeSamples = kPacketSizeMs * 16;
+
+  AcmReRegisterIsacMtTestOldApi()
+      : AudioCodingModuleTestOldApi(),
+        receive_thread_(CbReceiveThread, this, "receive"),
+        codec_registration_thread_(CbCodecRegistrationThread,
+                                   this,
+                                   "codec_registration"),
+        test_complete_(EventWrapper::Create()),
+        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+        codec_registered_(false),
+        receive_packet_count_(0),
+        next_insert_packet_time_ms_(0),
+        fake_clock_(new SimulatedClock(0)) {
+    AudioEncoderIsac::Config config;
+    config.payload_type = kPayloadType;
+    isac_encoder_.reset(new AudioEncoderIsac(config));
+    clock_ = fake_clock_.get();
+  }
+
+  void SetUp() {
+    AudioCodingModuleTestOldApi::SetUp();
+    // Set up input audio source to read from specified file, loop after 5
+    // seconds, and deliver blocks of 10 ms.
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
+    audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
+    RegisterCodec();  // Must be called before the threads start below.
+    StartThreads();
+  }
+
+  void RegisterCodec() override {
+    static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
+    AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
+    codec_.pltype = kPayloadType;
+
+    // Register iSAC codec in ACM, effectively unregistering the PCM16B codec
+    // registered in AudioCodingModuleTestOldApi::SetUp();
+    // Only register the decoder for now. The encoder is registered later.
+    ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
+  }
+
+  void StartThreads() {
+    receive_thread_.Start();
+    receive_thread_.SetPriority(rtc::kRealtimePriority);
+    codec_registration_thread_.Start();
+    codec_registration_thread_.SetPriority(rtc::kRealtimePriority);
+  }
+
+  void TearDown() {
+    AudioCodingModuleTestOldApi::TearDown();
+    receive_thread_.Stop();
+    codec_registration_thread_.Stop();
+  }
+
+  EventTypeWrapper RunTest() {
+    return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
+  }
+
+  static bool CbReceiveThread(void* context) {
+    return reinterpret_cast<AcmReRegisterIsacMtTestOldApi*>(context)
+        ->CbReceiveImpl();
+  }
+
+  bool CbReceiveImpl() {
+    SleepMs(1);
+    const size_t max_encoded_bytes = isac_encoder_->MaxEncodedBytes();
+    rtc::scoped_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
+    AudioEncoder::EncodedInfo info;
+    {
+      CriticalSectionScoped lock(crit_sect_.get());
+      if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
+        return true;
+      }
+      next_insert_packet_time_ms_ += kPacketSizeMs;
+      ++receive_packet_count_;
+
+      // Encode new frame.
+      uint32_t input_timestamp = rtp_header_.header.timestamp;
+      while (info.encoded_bytes == 0) {
+        info =
+            isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
+                                  max_encoded_bytes, encoded.get());
+        input_timestamp += 160;  // 10 ms at 16 kHz.
+      }
+      EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
+                input_timestamp);
+      EXPECT_EQ(rtp_header_.header.timestamp, info.encoded_timestamp);
+      EXPECT_EQ(rtp_header_.header.payloadType, info.payload_type);
+    }
+    // Now we're not holding the crit sect when calling ACM.
+
+    // Insert into ACM.
+    EXPECT_EQ(0, acm_->IncomingPacket(encoded.get(), info.encoded_bytes,
+                                      rtp_header_));
+
+    // Pull audio.
+    for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
+      AudioFrame audio_frame;
+      EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
+                                         &audio_frame));
+      fake_clock_->AdvanceTimeMilliseconds(10);
+    }
+    rtp_utility_->Forward(&rtp_header_);
+    return true;
+  }
+
+  static bool CbCodecRegistrationThread(void* context) {
+    return reinterpret_cast<AcmReRegisterIsacMtTestOldApi*>(context)
+        ->CbCodecRegistrationImpl();
+  }
+
+  bool CbCodecRegistrationImpl() {
+    SleepMs(1);
+    if (HasFatalFailure()) {
+      // End the test early if a fatal failure (ASSERT_*) has occurred.
+      test_complete_->Set();
+    }
+    CriticalSectionScoped lock(crit_sect_.get());
+    if (!codec_registered_ &&
+        receive_packet_count_ > kRegisterAfterNumPackets) {
+      // Register the iSAC encoder.
+      EXPECT_EQ(0, acm_->RegisterSendCodec(codec_));
+      codec_registered_ = true;
+    }
+    if (codec_registered_ && receive_packet_count_ > kNumPackets) {
+      test_complete_->Set();
+    }
+    return true;
+  }
+
+  rtc::PlatformThread receive_thread_;
+  rtc::PlatformThread codec_registration_thread_;
+  const rtc::scoped_ptr<EventWrapper> test_complete_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  bool codec_registered_ GUARDED_BY(crit_sect_);
+  int receive_packet_count_ GUARDED_BY(crit_sect_);
+  int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
+  rtc::scoped_ptr<AudioEncoderIsac> isac_encoder_;
+  rtc::scoped_ptr<SimulatedClock> fake_clock_;
+  test::AudioLoop audio_loop_;
+};
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
+  EXPECT_EQ(kEventSignaled, RunTest());
+}
+#endif
+
+// Disabling all of these tests on iOS until file support has been added.
+// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
+#if !defined(WEBRTC_IOS)
 
 class AcmReceiverBitExactnessOldApi : public ::testing::Test {
  public:
-  static std::string PlatformChecksum(std::string win64,
-                                      std::string android,
-                                      std::string others) {
+  static std::string PlatformChecksum(std::string others,
+                                      std::string win64,
+                                      std::string android_arm32,
+                                      std::string android_arm64) {
 #if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
     return win64;
-#elif defined(WEBRTC_ANDROID)
-    return android;
+#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
+    return android_arm32;
+#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
+    return android_arm64;
 #else
     return others;
 #endif
   }
 
  protected:
-  void Run(int output_freq_hz, const std::string& checksum_ref) {
+  struct ExternalDecoder {
+    int rtp_payload_type;
+    AudioDecoder* external_decoder;
+    int sample_rate_hz;
+    int num_channels;
+    std::string name;
+  };
+
+  void Run(int output_freq_hz,
+           const std::string& checksum_ref,
+           const std::vector<ExternalDecoder>& external_decoders) {
     const std::string input_file_name =
         webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
     rtc::scoped_ptr<test::RtpFileSource> packet_source(
@@ -735,65 +925,116 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
         output_freq_hz,
         test::AcmReceiveTestOldApi::kArbitraryChannels);
     ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
+    for (const auto& ed : external_decoders) {
+      ASSERT_EQ(0, test.RegisterExternalReceiveCodec(
+                       ed.rtp_payload_type, ed.external_decoder,
+                       ed.sample_rate_hz, ed.num_channels, ed.name));
+    }
     test.Run();
 
     std::string checksum_string = checksum.Finish();
     EXPECT_EQ(checksum_ref, checksum_string);
+
+    // Delete the output file.
+    remove(output_file_name.c_str());
   }
 };
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_8kHzOutput DISABLED_8kHzOutput
-#else
-#define MAYBE_8kHzOutput 8kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_8kHzOutput) {
-  Run(8000,
-      PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
-                       "adc92e173f908f93b96ba5844209815a",
-                       "908002dc01fc4eb1d2be24eb1d3f354b"));
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
+  Run(8000, PlatformChecksum("908002dc01fc4eb1d2be24eb1d3f354b",
+                             "dcee98c623b147ebe1b40dd30efa896e",
+                             "adc92e173f908f93b96ba5844209815a",
+                             "ba16137d3a5a1e637252289c57522bfe"),
+      std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_16kHzOutput DISABLED_16kHzOutput
-#else
-#define MAYBE_16kHzOutput 16kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_16kHzOutput) {
-  Run(16000,
-      PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
-                       "8cffa6abcb3e18e33b9d857666dff66a",
-                       "a909560b5ca49fa472b17b7b277195e9"));
+TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
+  Run(16000, PlatformChecksum("a909560b5ca49fa472b17b7b277195e9",
+                              "f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
+                              "8cffa6abcb3e18e33b9d857666dff66a",
+                              "66ee001e23534d4dcf5d0f81f916c93b"),
+      std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_32kHzOutput DISABLED_32kHzOutput
-#else
-#define MAYBE_32kHzOutput 32kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_32kHzOutput) {
-  Run(32000,
-      PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
-                       "3e126fe894720c3f85edadcc91964ba5",
-                       "441aab4b347fb3db4e9244337aca8d8e"));
+TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
+  Run(32000, PlatformChecksum("441aab4b347fb3db4e9244337aca8d8e",
+                              "306e0d990ee6e92de3fbecc0123ece37",
+                              "3e126fe894720c3f85edadcc91964ba5",
+                              "9c6ff204b14152c48fe41d5ab757943b"),
+      std::vector<ExternalDecoder>());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_48kHzOutput DISABLED_48kHzOutput
-#else
-#define MAYBE_48kHzOutput 48kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutput) {
-  Run(48000,
-      PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
-                       "0155665e93067c4e89256b944dd11999",
-                       "4ee2730fa1daae755e8a8fd3abd779ec"));
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
+  Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
+                              "aa7c232f63a67b2a72703593bdd172e0",
+                              "0155665e93067c4e89256b944dd11999",
+                              "fc4f0da8844cd808d822bbddf3b9c285"),
+      std::vector<ExternalDecoder>());
 }
 
+TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
+  // Class intended to forward a call from a mock DecodeInternal to Decode on
+  // the real decoder's Decode. DecodeInternal for the real decoder isn't
+  // public.
+  class DecodeForwarder {
+   public:
+    DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {}
+    int Decode(const uint8_t* encoded,
+               size_t encoded_len,
+               int sample_rate_hz,
+               int16_t* decoded,
+               AudioDecoder::SpeechType* speech_type) {
+      return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
+                              decoder_->PacketDuration(encoded, encoded_len) *
+                                  decoder_->Channels() * sizeof(int16_t),
+                              decoded, speech_type);
+    }
+
+   private:
+    AudioDecoder* const decoder_;
+  };
+
+  AudioDecoderPcmU decoder(1);
+  DecodeForwarder decode_forwarder(&decoder);
+  MockAudioDecoder mock_decoder;
+  // Set expectations on the mock decoder and also delegate the calls to the
+  // real decoder.
+  EXPECT_CALL(mock_decoder, IncomingPacket(_, _, _, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::IncomingPacket));
+  EXPECT_CALL(mock_decoder, Channels())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Channels));
+  EXPECT_CALL(mock_decoder, DecodeInternal(_, _, _, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&decode_forwarder, &DecodeForwarder::Decode));
+  EXPECT_CALL(mock_decoder, HasDecodePlc())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration));
+  ExternalDecoder ed;
+  ed.rtp_payload_type = 0;
+  ed.external_decoder = &mock_decoder;
+  ed.sample_rate_hz = 8000;
+  ed.num_channels = 1;
+  ed.name = "MockPCMU";
+  std::vector<ExternalDecoder> external_decoders;
+  external_decoders.push_back(ed);
+
+  Run(48000, PlatformChecksum("4ee2730fa1daae755e8a8fd3abd779ec",
+                              "aa7c232f63a67b2a72703593bdd172e0",
+                              "0155665e93067c4e89256b944dd11999",
+                              "fc4f0da8844cd808d822bbddf3b9c285"),
+      external_decoders);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+#endif
+
 // This test verifies bit exactness for the send-side of ACM. The test setup is
 // a chain of three different test classes:
 //
@@ -849,6 +1090,15 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
                                      frame_size_samples);
   }
 
+  bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
+                                 int payload_type) {
+    payload_type_ = payload_type;
+    frame_size_rtp_timestamps_ =
+        external_speech_encoder->Num10MsFramesInNextPacket() *
+        external_speech_encoder->RtpTimestampRateHz() / 100;
+    return send_test_->RegisterExternalCodec(external_speech_encoder);
+  }
+
   // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
   // before calling this method.
   void Run(const std::string& audio_checksum_ref,
@@ -888,6 +1138,9 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
 
     // Verify number of packets produced.
     EXPECT_EQ(expected_packets, packet_count_);
+
+    // Delete the output file.
+    remove(output_file_name.c_str());
   }
 
   // Returns a pointer to the next packet. Returns NULL if the source is
@@ -942,6 +1195,13 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
                                   codec_frame_size_rtp_timestamps));
   }
 
+  void SetUpTestExternalEncoder(AudioEncoder* external_speech_encoder,
+                                int payload_type) {
+    ASSERT_TRUE(SetUpSender());
+    ASSERT_TRUE(
+        RegisterExternalSendCodec(external_speech_encoder, payload_type));
+  }
+
   rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
   rtc::scoped_ptr<test::InputAudioFile> audio_source_;
   uint32_t frame_size_rtp_timestamps_;
@@ -952,58 +1212,57 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   rtc::Md5Digest payload_checksum_;
 };
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_IsacWb30ms DISABLED_IsacWb30ms
-#else
-#define MAYBE_IsacWb30ms IsacWb30ms
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb30ms) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "0b58f9eeee43d5891f5f6c75e77984a3",
           "c7e5bdadfa2871df95639fcc297cf23d",
           "0499ca260390769b3172136faad925b9",
-          "0b58f9eeee43d5891f5f6c75e77984a3"),
+          "866abf524acd2807efbe65e133c23f95"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "3c79f16f34218271f3dca4e2b1dfe1bb",
           "d42cb5195463da26c8129bbfe73a22e6",
           "83de248aea9c3c2bd680b6952401b4ca",
           "3c79f16f34218271f3dca4e2b1dfe1bb"),
-      33,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+      33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_IsacWb60ms DISABLED_IsacWb60ms
-#else
-#define MAYBE_IsacWb60ms IsacWb60ms
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb60ms) {
+TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "1ad29139a04782a33daad8c2b9b35875",
           "14d63c5f08127d280e722e3191b73bdd",
           "8da003e16c5371af2dc2be79a50f9076",
-          "1ad29139a04782a33daad8c2b9b35875"),
+          "ef75e900e6f375e3061163c53fd09a63"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "9e0a0ab743ad987b55b8e14802769c56",
           "ebe04a819d3a9d83a83a17f271e1139a",
           "97aeef98553b5a4b5a68f8b716e8eaf0",
           "9e0a0ab743ad987b55b8e14802769c56"),
-      16,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+      16, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms
+#else
+#define MAYBE_IsacSwb30ms IsacSwb30ms
+#endif
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
-          "2b3c387d06f00b7b7aad4c9be56fb83d",
-          "",
-          "5683b58da0fbf2063c7adc2e6bfb3fb8"),
+          "5683b58da0fbf2063c7adc2e6bfb3fb8",
+          "2b3c387d06f00b7b7aad4c9be56fb83d", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
-          "bcc2041e7744c7ebd9f701866856849c",
-          "",
-          "ce86106a93419aefb063097108ec94ab"),
+          "ce86106a93419aefb063097108ec94ab",
+          "bcc2041e7744c7ebd9f701866856849c", "android_arm32_payload",
+          "android_arm64_payload"),
       33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
 TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
@@ -1085,88 +1344,327 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
       test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(Ilbc_30ms)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_Ilbc_30ms DISABLED_Ilbc_30ms
+#else
+#define MAYBE_Ilbc_30ms Ilbc_30ms
+#endif
+#if defined(WEBRTC_CODEC_ILBC)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7b6ec10910debd9af08011d3ed5249f7",
-          "android_audio",
-          "7b6ec10910debd9af08011d3ed5249f7"),
+          "7b6ec10910debd9af08011d3ed5249f7", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "cfae2e9f6aba96e145f2bcdd5050ce78",
-          "android_payload",
-          "cfae2e9f6aba96e145f2bcdd5050ce78"),
-      33,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+          "cfae2e9f6aba96e145f2bcdd5050ce78", "android_arm32_payload",
+          "android_arm64_payload"),
+      33, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_20ms)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_20ms DISABLED_G722_20ms
+#else
+#define MAYBE_G722_20ms G722_20ms
+#endif
+#if defined(WEBRTC_CODEC_G722)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7d759436f2533582950d148b5161a36c",
-          "android_audio",
-          "7d759436f2533582950d148b5161a36c"),
+          "7d759436f2533582950d148b5161a36c", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "fc68a87e1380614e658087cb35d5ca10",
-          "android_payload",
-          "fc68a87e1380614e658087cb35d5ca10"),
-      50,
-      test::AcmReceiveTestOldApi::kMonoOutput);
+          "fc68a87e1380614e658087cb35d5ca10", "android_arm32_payload",
+          "android_arm64_payload"),
+      50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
+#endif
 
-TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
+#else
+#define MAYBE_G722_stereo_20ms G722_stereo_20ms
+#endif
+#if defined(WEBRTC_CODEC_G722)
+TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "7190ee718ab3d80eca181e5f7140c210",
-          "android_audio",
-          "7190ee718ab3d80eca181e5f7140c210"),
+          "7190ee718ab3d80eca181e5f7140c210", "android_arm32_audio",
+          "android_arm64_audio"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "66516152eeaa1e650ad94ff85f668dac",
-          "android_payload",
-          "66516152eeaa1e650ad94ff85f668dac"),
-      50,
-      test::AcmReceiveTestOldApi::kStereoOutput);
+          "66516152eeaa1e650ad94ff85f668dac", "android_arm32_payload",
+          "android_arm64_payload"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
-#else
-#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
 #endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
+
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "855041f2490b887302bce9d544731849",
           "855041f2490b887302bce9d544731849",
           "1e1a0fce893fef2d66886a7f09e2ebce",
-          "855041f2490b887302bce9d544731849"),
+          "7417a66c28be42d5d9b2d64e0c191585"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "d781cce1ab986b618d0da87226cdde30",
           "d781cce1ab986b618d0da87226cdde30",
           "1a1fe04dd12e755949987c8d729fb3e0",
-          "d781cce1ab986b618d0da87226cdde30"),
-      50,
-      test::AcmReceiveTestOldApi::kStereoOutput);
+          "47b0b04f1d03076b857c86c72c2c298b"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
-#else
-#define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
-#endif
-TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
   // If not set, default will be kAudio in case of stereo.
-  EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip, false));
+  EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "9b9e12bc3cc793740966e11cbfa8b35b",
           "9b9e12bc3cc793740966e11cbfa8b35b",
           "57412a4b5771d19ff03ec35deffe7067",
-          "9b9e12bc3cc793740966e11cbfa8b35b"),
+          "7ad0bbefcaa87e23187bf4a56d2f3513"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "c7340b1189652ab6b5e80dade7390cb4",
           "c7340b1189652ab6b5e80dade7390cb4",
           "cdfe85939c411d12b61701c566e22d26",
-          "c7340b1189652ab6b5e80dade7390cb4"),
-      50,
-      test::AcmReceiveTestOldApi::kStereoOutput);
+          "7a678fbe46df5bf0c67e88264a2d9275"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+// This test is for verifying the SetBitRate function. The bitrate is changed at
+// the beginning, and the number of generated bytes are checked.
+class AcmSetBitRateOldApi : public ::testing::Test {
+ protected:
+  static const int kTestDurationMs = 1000;
+
+  // Sets up the test::AcmSendTest object. Returns true on success, otherwise
+  // false.
+  bool SetUpSender() {
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+    // Note that |audio_source_| will loop forever. The test duration is set
+    // explicitly by |kTestDurationMs|.
+    audio_source_.reset(new test::InputAudioFile(input_file_name));
+    static const int kSourceRateHz = 32000;
+    send_test_.reset(new test::AcmSendTestOldApi(
+        audio_source_.get(), kSourceRateHz, kTestDurationMs));
+    return send_test_.get();
+  }
+
+  // Registers a send codec in the test::AcmSendTest object. Returns true on
+  // success, false on failure.
+  virtual bool RegisterSendCodec(const char* payload_name,
+                                 int sampling_freq_hz,
+                                 int channels,
+                                 int payload_type,
+                                 int frame_size_samples,
+                                 int frame_size_rtp_timestamps) {
+    return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
+                                     payload_type, frame_size_samples);
+  }
+
+  // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
+  // before calling this method.
+  void Run(int target_bitrate_bps, int expected_total_bits) {
+    ASSERT_TRUE(send_test_->acm());
+    send_test_->acm()->SetBitRate(target_bitrate_bps);
+    int nr_bytes = 0;
+    while (test::Packet* next_packet = send_test_->NextPacket()) {
+      nr_bytes += next_packet->payload_length_bytes();
+      delete next_packet;
+    }
+    EXPECT_EQ(expected_total_bits, nr_bytes * 8);
+  }
+
+  void SetUpTest(const char* codec_name,
+                 int codec_sample_rate_hz,
+                 int channels,
+                 int payload_type,
+                 int codec_frame_size_samples,
+                 int codec_frame_size_rtp_timestamps) {
+    ASSERT_TRUE(SetUpSender());
+    ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
+                                  payload_type, codec_frame_size_samples,
+                                  codec_frame_size_rtp_timestamps));
+  }
+
+  rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
+  rtc::scoped_ptr<test::InputAudioFile> audio_source_;
+};
+
+TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
+#if defined(WEBRTC_ANDROID)
+  Run(10000, 9328);
+#else
+  Run(10000, 9072);
+#endif // WEBRTC_ANDROID
+
+}
+
+TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
+#if defined(WEBRTC_ANDROID)
+  Run(50000, 47952);
+#else
+  Run(50000, 49600);
+#endif // WEBRTC_ANDROID
+}
+
+// The result on the Android platforms is inconsistent for this test case.
+// On android_rel the result is different from android and android arm64 rel.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
+#else
+#define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
+#endif
+TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
+  Run(100000, 100888);
+}
+
+// These next 2 tests ensure that the SetBitRate function has no effect on PCM
+TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
+  Run(8000, 128000);
+}
+
+TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
+  Run(32000, 128000);
+}
+
+// This test is for verifying the SetBitRate function. The bitrate is changed
+// in the middle, and the number of generated bytes are before and after the
+// change are checked.
+class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
+ protected:
+  AcmChangeBitRateOldApi() : sampling_freq_hz_(0), frame_size_samples_(0) {}
+
+  // Registers a send codec in the test::AcmSendTest object. Returns true on
+  // success, false on failure.
+  bool RegisterSendCodec(const char* payload_name,
+                         int sampling_freq_hz,
+                         int channels,
+                         int payload_type,
+                         int frame_size_samples,
+                         int frame_size_rtp_timestamps) override {
+    frame_size_samples_ = frame_size_samples;
+    sampling_freq_hz_ = sampling_freq_hz;
+    return AcmSetBitRateOldApi::RegisterSendCodec(
+        payload_name, sampling_freq_hz, channels, payload_type,
+        frame_size_samples, frame_size_rtp_timestamps);
+  }
+
+  // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
+  // before calling this method.
+  void Run(int target_bitrate_bps,
+           int expected_before_switch_bits,
+           int expected_after_switch_bits) {
+    ASSERT_TRUE(send_test_->acm());
+    int nr_packets =
+        sampling_freq_hz_ * kTestDurationMs / (frame_size_samples_ * 1000);
+    int nr_bytes_before = 0, nr_bytes_after = 0;
+    int packet_counter = 0;
+    while (test::Packet* next_packet = send_test_->NextPacket()) {
+      if (packet_counter == nr_packets / 2)
+        send_test_->acm()->SetBitRate(target_bitrate_bps);
+      if (packet_counter < nr_packets / 2)
+        nr_bytes_before += next_packet->payload_length_bytes();
+      else
+        nr_bytes_after += next_packet->payload_length_bytes();
+      packet_counter++;
+      delete next_packet;
+    }
+    EXPECT_EQ(expected_before_switch_bits, nr_bytes_before * 8);
+    EXPECT_EQ(expected_after_switch_bits, nr_bytes_after * 8);
+  }
+
+  uint32_t sampling_freq_hz_;
+  uint32_t frame_size_samples_;
+};
+
+TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
+#if defined(WEBRTC_ANDROID)
+  Run(10000, 32200, 5496);
+#else
+  Run(10000, 32200, 5432);
+#endif // WEBRTC_ANDROID
+}
+
+TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
+#if defined(WEBRTC_ANDROID)
+  Run(50000, 32200, 24912);
+#else
+  Run(50000, 32200, 24792);
+#endif // WEBRTC_ANDROID
+}
+
+TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
+#if defined(WEBRTC_ANDROID)
+  Run(100000, 32200, 51480);
+#else
+  Run(100000, 32200, 50584);
+#endif // WEBRTC_ANDROID
+}
+
+// These next 2 tests ensure that the SetBitRate function has no effect on PCM
+TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
+  Run(8000, 64000, 64000);
+}
+
+TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
+  Run(32000, 64000, 64000);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
+  CodecInst codec_inst;
+  codec_inst.channels = 1;
+  codec_inst.pacsize = 160;
+  codec_inst.pltype = 0;
+  AudioEncoderPcmU encoder(codec_inst);
+  MockAudioEncoder mock_encoder;
+  // Set expectations on the mock encoder and also delegate the calls to the
+  // real encoder.
+  EXPECT_CALL(mock_encoder, MaxEncodedBytes())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::MaxEncodedBytes));
+  EXPECT_CALL(mock_encoder, SampleRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
+  EXPECT_CALL(mock_encoder, NumChannels())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels));
+  EXPECT_CALL(mock_encoder, RtpTimestampRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz));
+  EXPECT_CALL(mock_encoder, Num10MsFramesInNextPacket())
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
+  EXPECT_CALL(mock_encoder, GetTargetBitrate())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
+  EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::EncodeInternal));
+  EXPECT_CALL(mock_encoder, SetFec(_))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SetFec));
+  ASSERT_NO_FATAL_FAILURE(
+      SetUpTestExternalEncoder(&mock_encoder, codec_inst.pltype));
+  Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
+      50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
 // This test fixture is implemented to run ACM and change the desired output
@@ -1220,6 +1718,9 @@ class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
 
     // This is where the actual test is executed.
     receive_test.Run();
+
+    // Delete output file.
+    remove(output_file_name.c_str());
   }
 
   // Inherited from test::PacketSource.
@@ -1282,4 +1783,7 @@ TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) {
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
+
+#endif
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.cc
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.cc
index 4c3e9fc393..4441932c8c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
 #include <assert.h>
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.h
similarity index 88%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.h
index 2aece0ff40..888afea0a7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 //
 // This class is for book keeping of calls to ACM. It is not useful to log API
@@ -60,4 +60,4 @@ class CallStatistics {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
index 2bee96465d..9ba0774ce1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
@@ -9,7 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager.cc
new file mode 100644
index 0000000000..ad67377d42
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager.cc
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+namespace acm2 {
+
+namespace {
+
+// Check if the given codec is a valid to be registered as send codec.
+int IsValidSendCodec(const CodecInst& send_codec) {
+  int dummy_id = 0;
+  if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "Wrong number of channels (%" PRIuS ", only mono and stereo "
+                 "are supported)",
+                 send_codec.channels);
+    return -1;
+  }
+
+  auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec);
+  if (!maybe_codec_id) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "Invalid codec setting for the send codec.");
+    return -1;
+  }
+
+  // Telephone-event cannot be a send codec.
+  if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "telephone-event cannot be a send codec");
+    return -1;
+  }
+
+  if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
+           .value_or(false)) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                 "%" PRIuS " number of channels not supportedn for %s.",
+                 send_codec.channels, send_codec.plname);
+    return -1;
+  }
+  return RentACodec::CodecIndexFromId(*maybe_codec_id).value_or(-1);
+}
+
+bool IsOpus(const CodecInst& codec) {
+  return
+#ifdef WEBRTC_CODEC_OPUS
+      !STR_CASE_CMP(codec.plname, "opus") ||
+#endif
+      false;
+}
+
+}  // namespace
+
+CodecManager::CodecManager() {
+  thread_checker_.DetachFromThread();
+}
+
+CodecManager::~CodecManager() = default;
+
+bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int codec_id = IsValidSendCodec(send_codec);
+
+  // Check for reported errors from function IsValidSendCodec().
+  if (codec_id < 0) {
+    return false;
+  }
+
+  int dummy_id = 0;
+  switch (RentACodec::RegisterRedPayloadType(
+      &codec_stack_params_.red_payload_types, send_codec)) {
+    case RentACodec::RegistrationResult::kOk:
+      return true;
+    case RentACodec::RegistrationResult::kBadFreq:
+      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                   "RegisterSendCodec() failed, invalid frequency for RED"
+                   " registration");
+      return false;
+    case RentACodec::RegistrationResult::kSkip:
+      break;
+  }
+  switch (RentACodec::RegisterCngPayloadType(
+      &codec_stack_params_.cng_payload_types, send_codec)) {
+    case RentACodec::RegistrationResult::kOk:
+      return true;
+    case RentACodec::RegistrationResult::kBadFreq:
+      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
+                   "RegisterSendCodec() failed, invalid frequency for CNG"
+                   " registration");
+      return false;
+    case RentACodec::RegistrationResult::kSkip:
+      break;
+  }
+
+  if (IsOpus(send_codec)) {
+    // VAD/DTX not supported.
+    codec_stack_params_.use_cng = false;
+  }
+
+  send_codec_inst_ = rtc::Optional<CodecInst>(send_codec);
+  codec_stack_params_.speech_encoder = nullptr;  // Caller must recreate it.
+  return true;
+}
+
+CodecInst CodecManager::ForgeCodecInst(
+    const AudioEncoder* external_speech_encoder) {
+  CodecInst ci;
+  ci.channels = external_speech_encoder->NumChannels();
+  ci.plfreq = external_speech_encoder->SampleRateHz();
+  ci.pacsize = rtc::CheckedDivExact(
+      static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
+                       ci.plfreq),
+      100);
+  ci.pltype = -1;  // Not valid.
+  ci.rate = -1;    // Not valid.
+  static const char kName[] = "external";
+  memcpy(ci.plname, kName, sizeof(kName));
+  return ci;
+}
+
+bool CodecManager::SetCopyRed(bool enable) {
+  if (enable && codec_stack_params_.use_codec_fec) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
+                 "Codec internal FEC and RED cannot be co-enabled.");
+    return false;
+  }
+  if (enable && send_codec_inst_ &&
+      codec_stack_params_.red_payload_types.count(send_codec_inst_->plfreq) <
+          1) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
+                 "Cannot enable RED at %i Hz.", send_codec_inst_->plfreq);
+    return false;
+  }
+  codec_stack_params_.use_red = enable;
+  return true;
+}
+
+bool CodecManager::SetVAD(bool enable, ACMVADMode mode) {
+  // Sanity check of the mode.
+  RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
+             mode == VADVeryAggr);
+
+  // Check that the send codec is mono. We don't support VAD/DTX for stereo
+  // sending.
+  const bool stereo_send =
+      codec_stack_params_.speech_encoder
+          ? (codec_stack_params_.speech_encoder->NumChannels() != 1)
+          : false;
+  if (enable && stereo_send) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
+                 "VAD/DTX not supported for stereo sending");
+    return false;
+  }
+
+  // TODO(kwiberg): This doesn't protect Opus when injected as an external
+  // encoder.
+  if (send_codec_inst_ && IsOpus(*send_codec_inst_)) {
+    // VAD/DTX not supported, but don't fail.
+    enable = false;
+  }
+
+  codec_stack_params_.use_cng = enable;
+  codec_stack_params_.vad_mode = mode;
+  return true;
+}
+
+bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
+  if (enable_codec_fec && codec_stack_params_.use_red) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
+                 "Codec internal FEC and RED cannot be co-enabled.");
+    return false;
+  }
+
+  codec_stack_params_.use_codec_fec = enable_codec_fec;
+  return true;
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager.h
new file mode 100644
index 0000000000..9227e13f09
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+
+#include <map>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+class AudioDecoder;
+class AudioEncoder;
+
+namespace acm2 {
+
+class CodecManager final {
+ public:
+  CodecManager();
+  ~CodecManager();
+
+  // Parses the given specification. On success, returns true and updates the
+  // stored CodecInst and stack parameters; on error, returns false.
+  bool RegisterEncoder(const CodecInst& send_codec);
+
+  static CodecInst ForgeCodecInst(const AudioEncoder* external_speech_encoder);
+
+  const CodecInst* GetCodecInst() const {
+    return send_codec_inst_ ? &*send_codec_inst_ : nullptr;
+  }
+  const RentACodec::StackParameters* GetStackParams() const {
+    return &codec_stack_params_;
+  }
+  RentACodec::StackParameters* GetStackParams() { return &codec_stack_params_; }
+
+  bool SetCopyRed(bool enable);
+
+  bool SetVAD(bool enable, ACMVADMode mode);
+
+  bool SetCodecFEC(bool enable_codec_fec);
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  rtc::Optional<CodecInst> send_codec_inst_;
+  RentACodec::StackParameters codec_stack_params_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(CodecManager);
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
new file mode 100644
index 0000000000..dce8f38842
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+
+namespace webrtc {
+namespace acm2 {
+
+using ::testing::Return;
+
+namespace {
+
+// Create a MockAudioEncoder with some reasonable default behavior.
+rtc::scoped_ptr<MockAudioEncoder> CreateMockEncoder() {
+  auto enc = rtc_make_scoped_ptr(new MockAudioEncoder);
+  EXPECT_CALL(*enc, SampleRateHz()).WillRepeatedly(Return(8000));
+  EXPECT_CALL(*enc, NumChannels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(*enc, Max10MsFramesInAPacket()).WillRepeatedly(Return(1));
+  EXPECT_CALL(*enc, Die());
+  return enc;
+}
+
+}  // namespace
+
+TEST(CodecManagerTest, ExternalEncoderFec) {
+  auto enc0 = CreateMockEncoder();
+  auto enc1 = CreateMockEncoder();
+  {
+    ::testing::InSequence s;
+    EXPECT_CALL(*enc0, SetFec(false)).WillOnce(Return(true));
+    EXPECT_CALL(*enc0, Mark("A"));
+    EXPECT_CALL(*enc0, SetFec(true)).WillOnce(Return(true));
+    EXPECT_CALL(*enc1, SetFec(true)).WillOnce(Return(true));
+    EXPECT_CALL(*enc1, SetFec(false)).WillOnce(Return(true));
+    EXPECT_CALL(*enc0, Mark("B"));
+    EXPECT_CALL(*enc0, SetFec(false)).WillOnce(Return(true));
+  }
+
+  CodecManager cm;
+  RentACodec rac;
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+  cm.GetStackParams()->speech_encoder = enc0.get();
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+  enc0->Mark("A");
+  EXPECT_EQ(true, cm.SetCodecFEC(true));
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_TRUE(cm.GetStackParams()->use_codec_fec);
+  cm.GetStackParams()->speech_encoder = enc1.get();
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_TRUE(cm.GetStackParams()->use_codec_fec);
+
+  EXPECT_EQ(true, cm.SetCodecFEC(false));
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  enc0->Mark("B");
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+  cm.GetStackParams()->speech_encoder = enc0.get();
+  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
+  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
similarity index 99%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
index 786fb2e527..0c31b83eb3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
similarity index 93%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
index c6942ec285..32dd1260f1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -117,4 +117,4 @@ class InitialDelayManager {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
similarity index 99%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
index e973593eb4..d86d221851 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
@@ -11,7 +11,7 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
new file mode 100644
index 0000000000..5695fd6e08
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -0,0 +1,307 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+
+#include <utility>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#ifdef WEBRTC_CODEC_G722
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#endif
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#ifdef WEBRTC_CODEC_RED
+#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#endif
+#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+
+namespace webrtc {
+namespace acm2 {
+
+rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
+    const char* payload_name,
+    int sampling_freq_hz,
+    size_t channels) {
+  return CodecIdFromIndex(
+      ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
+}
+
+rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
+  rtc::Optional<int> mi = CodecIndexFromId(codec_id);
+  return mi ? rtc::Optional<CodecInst>(Database()[*mi])
+            : rtc::Optional<CodecInst>();
+}
+
+rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
+    const CodecInst& codec_inst) {
+  return CodecIdFromIndex(ACMCodecDB::CodecNumber(codec_inst));
+}
+
+rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
+                                                       int sampling_freq_hz,
+                                                       size_t channels) {
+  rtc::Optional<CodecId> codec_id =
+      CodecIdByParams(payload_name, sampling_freq_hz, channels);
+  if (!codec_id)
+    return rtc::Optional<CodecInst>();
+  rtc::Optional<CodecInst> ci = CodecInstById(*codec_id);
+  RTC_DCHECK(ci);
+
+  // Keep the number of channels from the function call. For most codecs it
+  // will be the same value as in default codec settings, but not for all.
+  ci->channels = channels;
+
+  return ci;
+}
+
+bool RentACodec::IsCodecValid(const CodecInst& codec_inst) {
+  return ACMCodecDB::CodecNumber(codec_inst) >= 0;
+}
+
+rtc::Optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
+                                                       size_t num_channels) {
+  auto i = CodecIndexFromId(codec_id);
+  return i ? rtc::Optional<bool>(
+                 ACMCodecDB::codec_settings_[*i].channel_support >=
+                 num_channels)
+           : rtc::Optional<bool>();
+}
+
+rtc::ArrayView<const CodecInst> RentACodec::Database() {
+  return rtc::ArrayView<const CodecInst>(ACMCodecDB::database_,
+                                         NumberOfCodecs());
+}
+
+rtc::Optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
+    CodecId codec_id,
+    size_t num_channels) {
+  rtc::Optional<int> i = CodecIndexFromId(codec_id);
+  if (!i)
+    return rtc::Optional<NetEqDecoder>();
+  const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
+  return rtc::Optional<NetEqDecoder>(
+      (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
+          ? NetEqDecoder::kDecoderOpus_2ch
+          : ned);
+}
+
+RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
+    std::map<int, int>* pt_map,
+    const CodecInst& codec_inst) {
+  if (STR_CASE_CMP(codec_inst.plname, "CN") != 0)
+    return RegistrationResult::kSkip;
+  switch (codec_inst.plfreq) {
+    case 8000:
+    case 16000:
+    case 32000:
+    case 48000:
+      (*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
+      return RegistrationResult::kOk;
+    default:
+      return RegistrationResult::kBadFreq;
+  }
+}
+
+RentACodec::RegistrationResult RentACodec::RegisterRedPayloadType(
+    std::map<int, int>* pt_map,
+    const CodecInst& codec_inst) {
+  if (STR_CASE_CMP(codec_inst.plname, "RED") != 0)
+    return RegistrationResult::kSkip;
+  switch (codec_inst.plfreq) {
+    case 8000:
+      (*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
+      return RegistrationResult::kOk;
+    default:
+      return RegistrationResult::kBadFreq;
+  }
+}
+
+namespace {
+
+// Returns a new speech encoder, or null on error.
+// TODO(kwiberg): Don't handle errors here (bug 5033)
+rtc::scoped_ptr<AudioEncoder> CreateEncoder(
+    const CodecInst& speech_inst,
+    LockedIsacBandwidthInfo* bwinfo) {
+#if defined(WEBRTC_CODEC_ISACFX)
+  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo));
+#endif
+#if defined(WEBRTC_CODEC_ISAC)
+  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderOpus(speech_inst));
+#endif
+  if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderPcmU(speech_inst));
+  if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderPcmA(speech_inst));
+  if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderPcm16B(speech_inst));
+#ifdef WEBRTC_CODEC_ILBC
+  if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderIlbc(speech_inst));
+#endif
+#ifdef WEBRTC_CODEC_G722
+  if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
+    return rtc_make_scoped_ptr(new AudioEncoderG722(speech_inst));
+#endif
+  LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
+  return rtc::scoped_ptr<AudioEncoder>();
+}
+
+rtc::scoped_ptr<AudioEncoder> CreateRedEncoder(AudioEncoder* encoder,
+                                               int red_payload_type) {
+#ifdef WEBRTC_CODEC_RED
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type;
+  config.speech_encoder = encoder;
+  return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCopyRed(config));
+#else
+  return rtc::scoped_ptr<AudioEncoder>();
+#endif
+}
+
+rtc::scoped_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder,
+                                               int payload_type,
+                                               ACMVADMode vad_mode) {
+  AudioEncoderCng::Config config;
+  config.num_channels = encoder->NumChannels();
+  config.payload_type = payload_type;
+  config.speech_encoder = encoder;
+  switch (vad_mode) {
+    case VADNormal:
+      config.vad_mode = Vad::kVadNormal;
+      break;
+    case VADLowBitrate:
+      config.vad_mode = Vad::kVadLowBitrate;
+      break;
+    case VADAggr:
+      config.vad_mode = Vad::kVadAggressive;
+      break;
+    case VADVeryAggr:
+      config.vad_mode = Vad::kVadVeryAggressive;
+      break;
+    default:
+      FATAL();
+  }
+  return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCng(config));
+}
+
+rtc::scoped_ptr<AudioDecoder> CreateIsacDecoder(
+    LockedIsacBandwidthInfo* bwinfo) {
+#if defined(WEBRTC_CODEC_ISACFX)
+  return rtc_make_scoped_ptr(new AudioDecoderIsacFix(bwinfo));
+#elif defined(WEBRTC_CODEC_ISAC)
+  return rtc_make_scoped_ptr(new AudioDecoderIsac(bwinfo));
+#else
+  FATAL() << "iSAC is not supported.";
+  return rtc::scoped_ptr<AudioDecoder>();
+#endif
+}
+
+}  // namespace
+
+RentACodec::RentACodec() = default;
+RentACodec::~RentACodec() = default;
+
+AudioEncoder* RentACodec::RentEncoder(const CodecInst& codec_inst) {
+  rtc::scoped_ptr<AudioEncoder> enc =
+      CreateEncoder(codec_inst, &isac_bandwidth_info_);
+  if (!enc)
+    return nullptr;
+  speech_encoder_ = std::move(enc);
+  return speech_encoder_.get();
+}
+
+RentACodec::StackParameters::StackParameters() {
+  // Register the default payload types for RED and CNG.
+  for (const CodecInst& ci : RentACodec::Database()) {
+    RentACodec::RegisterCngPayloadType(&cng_payload_types, ci);
+    RentACodec::RegisterRedPayloadType(&red_payload_types, ci);
+  }
+}
+
+RentACodec::StackParameters::~StackParameters() = default;
+
+AudioEncoder* RentACodec::RentEncoderStack(StackParameters* param) {
+  RTC_DCHECK(param->speech_encoder);
+
+  if (param->use_codec_fec) {
+    // Switch FEC on. On failure, remember that FEC is off.
+    if (!param->speech_encoder->SetFec(true))
+      param->use_codec_fec = false;
+  } else {
+    // Switch FEC off. This shouldn't fail.
+    const bool success = param->speech_encoder->SetFec(false);
+    RTC_DCHECK(success);
+  }
+
+  auto pt = [&param](const std::map<int, int>& m) {
+    auto it = m.find(param->speech_encoder->SampleRateHz());
+    return it == m.end() ? rtc::Optional<int>()
+                         : rtc::Optional<int>(it->second);
+  };
+  auto cng_pt = pt(param->cng_payload_types);
+  param->use_cng =
+      param->use_cng && cng_pt && param->speech_encoder->NumChannels() == 1;
+  auto red_pt = pt(param->red_payload_types);
+  param->use_red = param->use_red && red_pt;
+
+  if (param->use_cng || param->use_red) {
+    // The RED and CNG encoders need to be in sync with the speech encoder, so
+    // reset the latter to ensure its buffer is empty.
+    param->speech_encoder->Reset();
+  }
+  encoder_stack_ = param->speech_encoder;
+  if (param->use_red) {
+    red_encoder_ = CreateRedEncoder(encoder_stack_, *red_pt);
+    if (red_encoder_)
+      encoder_stack_ = red_encoder_.get();
+  } else {
+    red_encoder_.reset();
+  }
+  if (param->use_cng) {
+    cng_encoder_ = CreateCngEncoder(encoder_stack_, *cng_pt, param->vad_mode);
+    encoder_stack_ = cng_encoder_.get();
+  } else {
+    cng_encoder_.reset();
+  }
+  return encoder_stack_;
+}
+
+AudioDecoder* RentACodec::RentIsacDecoder() {
+  if (!isac_decoder_)
+    isac_decoder_ = CreateIsacDecoder(&isac_bandwidth_info_);
+  return isac_decoder_.get();
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec.h
new file mode 100644
index 0000000000..b1dcc9196c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec.h
@@ -0,0 +1,249 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+
+#include <stddef.h>
+#include <map>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/typedefs.h"
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#else
+// Dummy implementation, for when we don't have iSAC.
+namespace webrtc {
+class LockedIsacBandwidthInfo {};
+}
+#endif
+
+namespace webrtc {
+
+struct CodecInst;
+
+namespace acm2 {
+
+class RentACodec {
+ public:
+  enum class CodecId {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    kISAC,
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    kISACSWB,
+#endif
+    // Mono
+    kPCM16B,
+    kPCM16Bwb,
+    kPCM16Bswb32kHz,
+    // Stereo
+    kPCM16B_2ch,
+    kPCM16Bwb_2ch,
+    kPCM16Bswb32kHz_2ch,
+    // Mono
+    kPCMU,
+    kPCMA,
+    // Stereo
+    kPCMU_2ch,
+    kPCMA_2ch,
+#ifdef WEBRTC_CODEC_ILBC
+    kILBC,
+#endif
+#ifdef WEBRTC_CODEC_G722
+    kG722,      // Mono
+    kG722_2ch,  // Stereo
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    kOpus,  // Mono and stereo
+#endif
+    kCNNB,
+    kCNWB,
+    kCNSWB,
+#ifdef ENABLE_48000_HZ
+    kCNFB,
+#endif
+    kAVT,
+#ifdef WEBRTC_CODEC_RED
+    kRED,
+#endif
+    kNumCodecs,  // Implementation detail. Don't use.
+
+// Set unsupported codecs to -1.
+#if !defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)
+    kISAC = -1,
+#endif
+#ifndef WEBRTC_CODEC_ISAC
+    kISACSWB = -1,
+#endif
+    // 48 kHz not supported, always set to -1.
+    kPCM16Bswb48kHz = -1,
+#ifndef WEBRTC_CODEC_ILBC
+    kILBC = -1,
+#endif
+#ifndef WEBRTC_CODEC_G722
+    kG722 = -1,      // Mono
+    kG722_2ch = -1,  // Stereo
+#endif
+#ifndef WEBRTC_CODEC_OPUS
+    kOpus = -1,  // Mono and stereo
+#endif
+#ifndef WEBRTC_CODEC_RED
+    kRED = -1,
+#endif
+#ifndef ENABLE_48000_HZ
+    kCNFB = -1,
+#endif
+
+    kNone = -1
+  };
+
+  enum class NetEqDecoder {
+    kDecoderPCMu,
+    kDecoderPCMa,
+    kDecoderPCMu_2ch,
+    kDecoderPCMa_2ch,
+    kDecoderILBC,
+    kDecoderISAC,
+    kDecoderISACswb,
+    kDecoderPCM16B,
+    kDecoderPCM16Bwb,
+    kDecoderPCM16Bswb32kHz,
+    kDecoderPCM16Bswb48kHz,
+    kDecoderPCM16B_2ch,
+    kDecoderPCM16Bwb_2ch,
+    kDecoderPCM16Bswb32kHz_2ch,
+    kDecoderPCM16Bswb48kHz_2ch,
+    kDecoderPCM16B_5ch,
+    kDecoderG722,
+    kDecoderG722_2ch,
+    kDecoderRED,
+    kDecoderAVT,
+    kDecoderCNGnb,
+    kDecoderCNGwb,
+    kDecoderCNGswb32kHz,
+    kDecoderCNGswb48kHz,
+    kDecoderArbitrary,
+    kDecoderOpus,
+    kDecoderOpus_2ch,
+  };
+
+  static inline size_t NumberOfCodecs() {
+    return static_cast<size_t>(CodecId::kNumCodecs);
+  }
+
+  static inline rtc::Optional<int> CodecIndexFromId(CodecId codec_id) {
+    const int i = static_cast<int>(codec_id);
+    return i >= 0 && i < static_cast<int>(NumberOfCodecs())
+               ? rtc::Optional<int>(i)
+               : rtc::Optional<int>();
+  }
+
+  static inline rtc::Optional<CodecId> CodecIdFromIndex(int codec_index) {
+    return static_cast<size_t>(codec_index) < NumberOfCodecs()
+               ? rtc::Optional<RentACodec::CodecId>(
+                     static_cast<RentACodec::CodecId>(codec_index))
+               : rtc::Optional<RentACodec::CodecId>();
+  }
+
+  static rtc::Optional<CodecId> CodecIdByParams(const char* payload_name,
+                                                int sampling_freq_hz,
+                                                size_t channels);
+  static rtc::Optional<CodecInst> CodecInstById(CodecId codec_id);
+  static rtc::Optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
+  static rtc::Optional<CodecInst> CodecInstByParams(const char* payload_name,
+                                                    int sampling_freq_hz,
+                                                    size_t channels);
+  static bool IsCodecValid(const CodecInst& codec_inst);
+
+  static inline bool IsPayloadTypeValid(int payload_type) {
+    return payload_type >= 0 && payload_type <= 127;
+  }
+
+  static rtc::ArrayView<const CodecInst> Database();
+
+  static rtc::Optional<bool> IsSupportedNumChannels(CodecId codec_id,
+                                                    size_t num_channels);
+
+  static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(
+      CodecId codec_id,
+      size_t num_channels);
+
+  // Parse codec_inst and extract payload types. If the given CodecInst was for
+  // the wrong sort of codec, return kSkip; otherwise, if the rate was illegal,
+  // return kBadFreq; otherwise, update the given RTP timestamp rate (Hz) ->
+  // payload type map and return kOk.
+  enum class RegistrationResult { kOk, kSkip, kBadFreq };
+  static RegistrationResult RegisterCngPayloadType(std::map<int, int>* pt_map,
+                                                   const CodecInst& codec_inst);
+  static RegistrationResult RegisterRedPayloadType(std::map<int, int>* pt_map,
+                                                   const CodecInst& codec_inst);
+
+  RentACodec();
+  ~RentACodec();
+
+  // Creates and returns an audio encoder built to the given specification.
+  // Returns null in case of error. The returned encoder is live until the next
+  // successful call to this function, or until the Rent-A-Codec is destroyed.
+  AudioEncoder* RentEncoder(const CodecInst& codec_inst);
+
+  struct StackParameters {
+    StackParameters();
+    ~StackParameters();
+
+    AudioEncoder* speech_encoder = nullptr;
+    bool use_codec_fec = false;
+    bool use_red = false;
+    bool use_cng = false;
+    ACMVADMode vad_mode = VADNormal;
+
+    // Maps from RTP timestamp rate (in Hz) to payload type.
+    std::map<int, int> cng_payload_types;
+    std::map<int, int> red_payload_types;
+  };
+
+  // Creates and returns an audio encoder stack constructed to the given
+  // specification. If the specification isn't compatible with the encoder, it
+  // will be changed to match (things will be switched off). The returned
+  // encoder is live until the next successful call to this function, or until
+  // the Rent-A-Codec is destroyed.
+  AudioEncoder* RentEncoderStack(StackParameters* param);
+
+  // The last return value of RentEncoderStack, or null if it hasn't been
+  // called.
+  AudioEncoder* GetEncoderStack() const { return encoder_stack_; }
+
+  // Creates and returns an iSAC decoder, which will remain live until the
+  // Rent-A-Codec is destroyed. Subsequent calls will simply return the same
+  // object.
+  AudioDecoder* RentIsacDecoder();
+
+ private:
+  rtc::scoped_ptr<AudioEncoder> speech_encoder_;
+  rtc::scoped_ptr<AudioEncoder> cng_encoder_;
+  rtc::scoped_ptr<AudioEncoder> red_encoder_;
+  rtc::scoped_ptr<AudioDecoder> isac_decoder_;
+  AudioEncoder* encoder_stack_ = nullptr;
+  LockedIsacBandwidthInfo isac_bandwidth_info_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RentACodec);
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
new file mode 100644
index 0000000000..e838488e53
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
@@ -0,0 +1,222 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+
+namespace webrtc {
+namespace acm2 {
+
+using ::testing::Return;
+
+namespace {
+const int kDataLengthSamples = 80;
+const int kPacketSizeSamples = 2 * kDataLengthSamples;
+const int16_t kZeroData[kDataLengthSamples] = {0};
+const CodecInst kDefaultCodecInst = {0, "pcmu", 8000, kPacketSizeSamples,
+                                     1, 64000};
+const int kCngPt = 13;
+}  // namespace
+
+class RentACodecTestF : public ::testing::Test {
+ protected:
+  void CreateCodec() {
+    speech_encoder_ = rent_a_codec_.RentEncoder(kDefaultCodecInst);
+    ASSERT_TRUE(speech_encoder_);
+    RentACodec::StackParameters param;
+    param.use_cng = true;
+    param.speech_encoder = speech_encoder_;
+    encoder_ = rent_a_codec_.RentEncoderStack(&param);
+  }
+
+  void EncodeAndVerify(size_t expected_out_length,
+                       uint32_t expected_timestamp,
+                       int expected_payload_type,
+                       int expected_send_even_if_empty) {
+    uint8_t out[kPacketSizeSamples];
+    AudioEncoder::EncodedInfo encoded_info;
+    encoded_info =
+        encoder_->Encode(timestamp_, kZeroData, kPacketSizeSamples, out);
+    timestamp_ += kDataLengthSamples;
+    EXPECT_TRUE(encoded_info.redundant.empty());
+    EXPECT_EQ(expected_out_length, encoded_info.encoded_bytes);
+    EXPECT_EQ(expected_timestamp, encoded_info.encoded_timestamp);
+    if (expected_payload_type >= 0)
+      EXPECT_EQ(expected_payload_type, encoded_info.payload_type);
+    if (expected_send_even_if_empty >= 0)
+      EXPECT_EQ(static_cast<bool>(expected_send_even_if_empty),
+                encoded_info.send_even_if_empty);
+  }
+
+  RentACodec rent_a_codec_;
+  AudioEncoder* speech_encoder_ = nullptr;
+  AudioEncoder* encoder_ = nullptr;
+  uint32_t timestamp_ = 0;
+};
+
+// This test verifies that CNG frames are delivered as expected. Since the frame
+// size is set to 20 ms, we expect the first encode call to produce no output
+// (which is signaled as 0 bytes output of type kNoEncoding). The next encode
+// call should produce one SID frame of 9 bytes. The third call should not
+// result in any output (just like the first one). The fourth and final encode
+// call should produce an "empty frame", which is like no output, but with
+// AudioEncoder::EncodedInfo::send_even_if_empty set to true. (The reason to
+// produce an empty frame is to drive sending of DTMF packets in the RTP/RTCP
+// module.)
+TEST_F(RentACodecTestF, VerifyCngFrames) {
+  CreateCodec();
+  uint32_t expected_timestamp = timestamp_;
+  // Verify no frame.
+  {
+    SCOPED_TRACE("First encoding");
+    EncodeAndVerify(0, expected_timestamp, -1, -1);
+  }
+
+  // Verify SID frame delivered.
+  {
+    SCOPED_TRACE("Second encoding");
+    EncodeAndVerify(9, expected_timestamp, kCngPt, 1);
+  }
+
+  // Verify no frame.
+  {
+    SCOPED_TRACE("Third encoding");
+    EncodeAndVerify(0, expected_timestamp, -1, -1);
+  }
+
+  // Verify NoEncoding.
+  expected_timestamp += 2 * kDataLengthSamples;
+  {
+    SCOPED_TRACE("Fourth encoding");
+    EncodeAndVerify(0, expected_timestamp, kCngPt, 1);
+  }
+}
+
+TEST(RentACodecTest, ExternalEncoder) {
+  const int kSampleRateHz = 8000;
+  MockAudioEncoder external_encoder;
+  EXPECT_CALL(external_encoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(external_encoder, NumChannels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(external_encoder, SetFec(false)).WillRepeatedly(Return(true));
+
+  RentACodec rac;
+  RentACodec::StackParameters param;
+  param.speech_encoder = &external_encoder;
+  EXPECT_EQ(&external_encoder, rac.RentEncoderStack(&param));
+  const int kPacketSizeSamples = kSampleRateHz / 100;
+  int16_t audio[kPacketSizeSamples] = {0};
+  uint8_t encoded[kPacketSizeSamples];
+  AudioEncoder::EncodedInfo info;
+
+  {
+    ::testing::InSequence s;
+    info.encoded_timestamp = 0;
+    EXPECT_CALL(external_encoder,
+                EncodeInternal(0, rtc::ArrayView<const int16_t>(audio),
+                               arraysize(encoded), encoded))
+        .WillOnce(Return(info));
+    EXPECT_CALL(external_encoder, Mark("A"));
+    EXPECT_CALL(external_encoder, Mark("B"));
+    info.encoded_timestamp = 2;
+    EXPECT_CALL(external_encoder,
+                EncodeInternal(2, rtc::ArrayView<const int16_t>(audio),
+                               arraysize(encoded), encoded))
+        .WillOnce(Return(info));
+    EXPECT_CALL(external_encoder, Die());
+  }
+
+  info = rac.GetEncoderStack()->Encode(0, audio, arraysize(encoded), encoded);
+  EXPECT_EQ(0u, info.encoded_timestamp);
+  external_encoder.Mark("A");
+
+  // Change to internal encoder.
+  CodecInst codec_inst = kDefaultCodecInst;
+  codec_inst.pacsize = kPacketSizeSamples;
+  param.speech_encoder = rac.RentEncoder(codec_inst);
+  ASSERT_TRUE(param.speech_encoder);
+  EXPECT_EQ(param.speech_encoder, rac.RentEncoderStack(&param));
+
+  // Don't expect any more calls to the external encoder.
+  info = rac.GetEncoderStack()->Encode(1, audio, arraysize(encoded), encoded);
+  external_encoder.Mark("B");
+
+  // Change back to external encoder again.
+  param.speech_encoder = &external_encoder;
+  EXPECT_EQ(&external_encoder, rac.RentEncoderStack(&param));
+  info = rac.GetEncoderStack()->Encode(2, audio, arraysize(encoded), encoded);
+  EXPECT_EQ(2u, info.encoded_timestamp);
+}
+
+// Verify that the speech encoder's Reset method is called when CNG or RED
+// (or both) are switched on, but not when they're switched off.
+void TestCngAndRedResetSpeechEncoder(bool use_cng, bool use_red) {
+  MockAudioEncoder speech_encoder;
+  EXPECT_CALL(speech_encoder, NumChannels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(speech_encoder, Max10MsFramesInAPacket())
+      .WillRepeatedly(Return(2));
+  EXPECT_CALL(speech_encoder, SampleRateHz()).WillRepeatedly(Return(8000));
+  EXPECT_CALL(speech_encoder, SetFec(false)).WillRepeatedly(Return(true));
+  {
+    ::testing::InSequence s;
+    EXPECT_CALL(speech_encoder, Mark("disabled"));
+    EXPECT_CALL(speech_encoder, Mark("enabled"));
+    if (use_cng || use_red)
+      EXPECT_CALL(speech_encoder, Reset());
+    EXPECT_CALL(speech_encoder, Die());
+  }
+
+  RentACodec::StackParameters param1, param2;
+  param1.speech_encoder = &speech_encoder;
+  param2.speech_encoder = &speech_encoder;
+  param2.use_cng = use_cng;
+  param2.use_red = use_red;
+  speech_encoder.Mark("disabled");
+  RentACodec rac;
+  rac.RentEncoderStack(&param1);
+  speech_encoder.Mark("enabled");
+  rac.RentEncoderStack(&param2);
+}
+
+TEST(RentACodecTest, CngResetsSpeechEncoder) {
+  TestCngAndRedResetSpeechEncoder(true, false);
+}
+
+TEST(RentACodecTest, RedResetsSpeechEncoder) {
+  TestCngAndRedResetSpeechEncoder(false, true);
+}
+
+TEST(RentACodecTest, CngAndRedResetsSpeechEncoder) {
+  TestCngAndRedResetSpeechEncoder(true, true);
+}
+
+TEST(RentACodecTest, NoCngAndRedNoSpeechEncoderReset) {
+  TestCngAndRedResetSpeechEncoder(false, false);
+}
+
+TEST(RentACodecTest, RentEncoderError) {
+  const CodecInst codec_inst = {
+      0, "Robert'); DROP TABLE Students;", 8000, 160, 1, 64000};
+  RentACodec rent_a_codec;
+  EXPECT_FALSE(rent_a_codec.RentEncoder(codec_inst));
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(RentACodecTest, RentEncoderStackWithoutSpeechEncoder) {
+  RentACodec::StackParameters sp;
+  EXPECT_EQ(nullptr, sp.speech_encoder);
+  EXPECT_DEATH(RentACodec().RentEncoderStack(&sp), "");
+}
+#endif
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding.gypi
index b694e37962..89994fa547 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding.gypi
@@ -12,24 +12,214 @@
     'codecs/interfaces.gypi',
     'codecs/cng/cng.gypi',
     'codecs/g711/g711.gypi',
+    'codecs/g722/g722.gypi',
+    'codecs/ilbc/ilbc.gypi',
+    'codecs/isac/isac.gypi',
+    'codecs/isac/isac_common.gypi',
+    'codecs/isac/isacfix.gypi',
     'codecs/pcm16b/pcm16b.gypi',
     'codecs/red/red.gypi',
-    'main/acm2/audio_coding_module.gypi',
     'neteq/neteq.gypi',
   ],
+  'variables': {
+    'audio_coding_dependencies': [
+      'cng',
+      'g711',
+      'pcm16b',
+      '<(webrtc_root)/common.gyp:webrtc_common',
+      '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+    ],
+    'audio_coding_defines': [],
+    'conditions': [
+      ['include_g722==1', {
+        'audio_coding_dependencies': ['g722',],
+        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
+      }],
+      ['include_ilbc==1', {
+        'audio_coding_dependencies': ['ilbc', 'red',],
+        'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
+      }],
+      ['include_isac==1', {
+        'audio_coding_dependencies': ['isac',],
+        'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
+      }],
+      ['include_opus==1', {
+        'audio_coding_dependencies': ['webrtc_opus',],
+        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+      }],
+      ['build_with_mozilla==0', {
+        'conditions': [
+          ['target_arch=="arm"', {
+            'audio_coding_dependencies': ['isac_fix',],
+            'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
+          }, {
+            'audio_coding_dependencies': ['isac',],
+            'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
+          }],
+        ],
+        'audio_coding_dependencies': ['g722',],
+        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
+      }],
+      ['build_with_mozilla==0 and build_with_chromium==0', {
+        'audio_coding_dependencies': ['ilbc', 'red',],
+        'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
+      }],
+    ],
+  },
+  'targets': [
+    {
+      'target_name': 'rent_a_codec',
+      'type': 'static_library',
+      'defines': [
+        '<@(audio_coding_defines)',
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
+      ],
+      'include_dirs': [
+        '<(webrtc_root)',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '<(webrtc_root)',
+        ],
+      },
+      'sources': [
+        'acm2/acm_codec_database.cc',
+        'acm2/acm_codec_database.h',
+        'acm2/rent_a_codec.cc',
+        'acm2/rent_a_codec.h',
+      ],
+    },
+    {
+      'target_name': 'audio_coding_module',
+      'type': 'static_library',
+      'defines': [
+        '<@(audio_coding_defines)',
+      ],
+      'dependencies': [
+        '<@(audio_coding_dependencies)',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+        'neteq',
+        'rent_a_codec',
+      ],
+      'include_dirs': [
+        'include',
+        '../include',
+        '<(webrtc_root)',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          'include',
+          '../include',
+          '<(webrtc_root)',
+        ],
+      },
+      'conditions': [
+        ['include_opus==1', {
+          'export_dependent_settings': ['webrtc_opus'],
+        }],
+      ],
+      'sources': [
+        'acm2/acm_common_defs.h',
+        'acm2/acm_receiver.cc',
+        'acm2/acm_receiver.h',
+        'acm2/acm_resampler.cc',
+        'acm2/acm_resampler.h',
+        'acm2/audio_coding_module.cc',
+        'acm2/audio_coding_module_impl.cc',
+        'acm2/audio_coding_module_impl.h',
+        'acm2/call_statistics.cc',
+        'acm2/call_statistics.h',
+        'acm2/codec_manager.cc',
+        'acm2/codec_manager.h',
+        'acm2/initial_delay_manager.cc',
+        'acm2/initial_delay_manager.h',
+        'include/audio_coding_module.h',
+        'include/audio_coding_module_typedefs.h',
+      ],
+    },
+  ],
   'conditions': [
-    ['include_g722==1', {
-      'includes': ['codecs/g722/g722.gypi',],
-    }],
-    ['include_ilbc==1', {
-      'includes': ['codecs/ilbc/ilbc.gypi',],
-    }],
-    ['include_isac==1', {
-      'includes': ['codecs/isac/isac.gypi',
-                   'codecs/isac/isacfix.gypi',],
-    }],
     ['include_opus==1', {
       'includes': ['codecs/opus/opus.gypi',],
     }],
+    ['include_tests==1', {
+      'targets': [
+        {
+          'target_name': 'acm_receive_test',
+          'type': 'static_library',
+          'defines': [
+            '<@(audio_coding_defines)',
+          ],
+          'dependencies': [
+            '<@(audio_coding_dependencies)',
+            'audio_coding_module',
+            'neteq_unittest_tools',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'acm2/acm_receive_test_oldapi.cc',
+            'acm2/acm_receive_test_oldapi.h',
+          ],
+        }, # acm_receive_test
+        {
+          'target_name': 'acm_send_test',
+          'type': 'static_library',
+          'defines': [
+            '<@(audio_coding_defines)',
+          ],
+          'dependencies': [
+            '<@(audio_coding_dependencies)',
+            'audio_coding_module',
+            'neteq_unittest_tools',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'acm2/acm_send_test_oldapi.cc',
+            'acm2/acm_send_test_oldapi.h',
+          ],
+        }, # acm_send_test
+        {
+          'target_name': 'delay_test',
+          'type': 'executable',
+          'dependencies': [
+            'audio_coding_module',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/common.gyp:webrtc_common',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+             'test/delay_test.cc',
+             'test/Channel.cc',
+             'test/PCMFile.cc',
+             'test/utility.cc',
+           ],
+        }, # delay_test
+        {
+          'target_name': 'insert_packet_with_timing',
+          'type': 'executable',
+          'dependencies': [
+            'audio_coding_module',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/common.gyp:webrtc_common',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+             'test/insert_packet_with_timing.cc',
+             'test/Channel.cc',
+             'test/PCMFile.cc',
+           ],
+        }, # delay_test
+      ],
+    }],
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding_tests.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding_tests.gypi
index 86a92c595d..e60309a6df 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding_tests.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/audio_coding_tests.gypi
@@ -18,7 +18,7 @@
       'type': '<(gtest_target_type)',
       'dependencies': [
         'audio_processing',
-        'iSACFix',
+        'isac_fix',
         'webrtc_opus',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
@@ -51,22 +51,5 @@
         },
       ],
     }],
-    ['test_isolation_mode != "noop"', {
-      'targets': [
-        {
-          'target_name': 'audio_codec_speed_tests_run',
-          'type': 'none',
-          'dependencies': [
-            'audio_codec_speed_tests',
-          ],
-          'includes': [
-            '../../build/isolate.gypi',
-          ],
-          'sources': [
-            'audio_codec_speed_tests.isolate',
-          ],
-        },
-      ],
-    }],
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/OWNERS
index 906df28cf6..b98e52763d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/OWNERS
@@ -1,8 +1,3 @@
-tina.legrand@webrtc.org
-turaj@webrtc.org
-jan.skoglund@webrtc.org
-henrik.lundin@webrtc.org
-
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
 per-file *.gyp=*
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index 1ab2a7fec1..d2984b97b0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -13,14 +13,17 @@
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
 int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
                          int sample_rate_hz, size_t max_decoded_bytes,
                          int16_t* decoded, SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
   int duration = PacketDuration(encoded, encoded_len);
-  if (duration >= 0 && duration * sizeof(int16_t) > max_decoded_bytes) {
+  if (duration >= 0 &&
+      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
     return -1;
   }
   return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
@@ -30,20 +33,16 @@ int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
 int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
                                   int sample_rate_hz, size_t max_decoded_bytes,
                                   int16_t* decoded, SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
   int duration = PacketDurationRedundant(encoded, encoded_len);
-  if (duration >= 0 && duration * sizeof(int16_t) > max_decoded_bytes) {
+  if (duration >= 0 &&
+      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
     return -1;
   }
   return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
                                  speech_type);
 }
 
-int AudioDecoder::DecodeInternal(const uint8_t* encoded, size_t encoded_len,
-                                 int sample_rate_hz, int16_t* decoded,
-                                 SpeechType* speech_type) {
-  return kNotImplemented;
-}
-
 int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
                                           size_t encoded_len,
                                           int sample_rate_hz, int16_t* decoded,
@@ -54,7 +53,9 @@ int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
 
 bool AudioDecoder::HasDecodePlc() const { return false; }
 
-int AudioDecoder::DecodePlc(int num_frames, int16_t* decoded) { return -1; }
+size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return 0;
+}
 
 int AudioDecoder::IncomingPacket(const uint8_t* payload,
                                  size_t payload_len,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 30359a9777..9ae9150adf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
 
 #include <stdlib.h>  // NULL
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -36,36 +36,37 @@ class AudioDecoder {
 
   // Decodes |encode_len| bytes from |encoded| and writes the result in
   // |decoded|. The maximum bytes allowed to be written into |decoded| is
-  // |max_decoded_bytes|. The number of samples from all channels produced is
-  // in the return value. If the decoder produced comfort noise, |speech_type|
+  // |max_decoded_bytes|. Returns the total number of samples across all
+  // channels. If the decoder produced comfort noise, |speech_type|
   // is set to kComfortNoise, otherwise it is kSpeech. The desired output
   // sample rate is provided in |sample_rate_hz|, which must be valid for the
   // codec at hand.
-  virtual int Decode(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     size_t max_decoded_bytes,
-                     int16_t* decoded,
-                     SpeechType* speech_type);
+  int Decode(const uint8_t* encoded,
+             size_t encoded_len,
+             int sample_rate_hz,
+             size_t max_decoded_bytes,
+             int16_t* decoded,
+             SpeechType* speech_type);
 
   // Same as Decode(), but interfaces to the decoders redundant decode function.
   // The default implementation simply calls the regular Decode() method.
-  virtual int DecodeRedundant(const uint8_t* encoded,
-                              size_t encoded_len,
-                              int sample_rate_hz,
-                              size_t max_decoded_bytes,
-                              int16_t* decoded,
-                              SpeechType* speech_type);
+  int DecodeRedundant(const uint8_t* encoded,
+                      size_t encoded_len,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded,
+                      SpeechType* speech_type);
 
   // Indicates if the decoder implements the DecodePlc method.
   virtual bool HasDecodePlc() const;
 
   // Calls the packet-loss concealment of the decoder to update the state after
-  // one or several lost packets.
-  virtual int DecodePlc(int num_frames, int16_t* decoded);
+  // one or several lost packets. The caller has to make sure that the
+  // memory allocated in |decoded| should accommodate |num_frames| frames.
+  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
 
-  // Initializes the decoder.
-  virtual int Init() = 0;
+  // Resets the decoder state (empty buffers etc.).
+  virtual void Reset() = 0;
 
   // Notifies the decoder of an incoming packet to NetEQ.
   virtual int IncomingPacket(const uint8_t* payload,
@@ -77,14 +78,14 @@ class AudioDecoder {
   // Returns the last error code from the decoder.
   virtual int ErrorCode();
 
-  // Returns the duration in samples of the payload in |encoded| which is
-  // |encoded_len| bytes long. Returns kNotImplemented if no duration estimate
-  // is available, or -1 in case of an error.
+  // Returns the duration in samples-per-channel of the payload in |encoded|
+  // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
+  // estimate is available, or -1 in case of an error.
   virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
 
-  // Returns the duration in samples of the redandant payload in |encoded| which
-  // is |encoded_len| bytes long. Returns kNotImplemented if no duration
-  // estimate is available, or -1 in case of an error.
+  // Returns the duration in samples-per-channel of the redandant payload in
+  // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
+  // duration estimate is available, or -1 in case of an error.
   virtual int PacketDurationRedundant(const uint8_t* encoded,
                                       size_t encoded_len) const;
 
@@ -106,7 +107,7 @@ class AudioDecoder {
                              size_t encoded_len,
                              int sample_rate_hz,
                              int16_t* decoded,
-                             SpeechType* speech_type);
+                             SpeechType* speech_type) = 0;
 
   virtual int DecodeRedundantInternal(const uint8_t* encoded,
                                       size_t encoded_len,
@@ -115,8 +116,8 @@ class AudioDecoder {
                                       SpeechType* speech_type);
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_AUDIO_DECODER_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 72e4265e98..e99fc30995 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -9,31 +9,50 @@
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
-AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() {
-}
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
 
-AudioEncoder::EncodedInfo::~EncodedInfo() {
-}
-
-AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
-                                               const int16_t* audio,
-                                               size_t num_samples_per_channel,
-                                               size_t max_encoded_bytes,
-                                               uint8_t* encoded) {
-  CHECK_EQ(num_samples_per_channel,
-           static_cast<size_t>(SampleRateHz() / 100));
-  EncodedInfo info =
-      EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
-  CHECK_LE(info.encoded_bytes, max_encoded_bytes);
-  return info;
-}
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
 
 int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
+  EncodedInfo info =
+      EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
+  RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
+  return info;
+}
+
+bool AudioEncoder::SetFec(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::SetDtx(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::SetApplication(Application application) {
+  return false;
+}
+
+void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
+
+void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.h
index e8a2c6ea1e..a46b0e86a7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -14,6 +14,7 @@
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -23,18 +24,11 @@ namespace webrtc {
 class AudioEncoder {
  public:
   struct EncodedInfoLeaf {
-    EncodedInfoLeaf()
-        : encoded_bytes(0),
-          encoded_timestamp(0),
-          payload_type(0),
-          send_even_if_empty(false),
-          speech(true) {}
-
-    size_t encoded_bytes;
-    uint32_t encoded_timestamp;
-    int payload_type;
-    bool send_even_if_empty;
-    bool speech;
+    size_t encoded_bytes = 0;
+    uint32_t encoded_timestamp = 0;
+    int payload_type = 0;
+    bool send_even_if_empty = false;
+    bool speech = true;
   };
 
   // This is the main struct for auxiliary encoding information. Each encoded
@@ -54,26 +48,9 @@ class AudioEncoder {
     std::vector<EncodedInfoLeaf> redundant;
   };
 
-  virtual ~AudioEncoder() {}
+  virtual ~AudioEncoder() = default;
 
-  // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
-  // num_channels() samples). Multi-channel audio must be sample-interleaved.
-  // The encoder produces zero or more bytes of output in |encoded| and
-  // returns additional encoding information.
-  // The caller is responsible for making sure that |max_encoded_bytes| is
-  // not smaller than the number of bytes actually produced by the encoder.
-  EncodedInfo Encode(uint32_t rtp_timestamp,
-                     const int16_t* audio,
-                     size_t num_samples_per_channel,
-                     size_t max_encoded_bytes,
-                     uint8_t* encoded);
-
-  // Return the input sample rate in Hz and the number of input channels.
-  // These are constants set at instantiation time.
-  virtual int SampleRateHz() const = 0;
-  virtual int NumChannels() const = 0;
-
-  // Return the maximum number of bytes that can be produced by the encoder
+  // Returns the maximum number of bytes that can be produced by the encoder
   // at each Encode() call. The caller can use the return value to determine
   // the size of the buffer that needs to be allocated. This value is allowed
   // to depend on encoder parameters like bitrate, frame size etc., so if
@@ -81,8 +58,13 @@ class AudioEncoder {
   // that the buffer is large enough by calling MaxEncodedBytes() again.
   virtual size_t MaxEncodedBytes() const = 0;
 
-  // Returns the rate with which the RTP timestamps are updated. By default,
-  // this is the same as sample_rate_hz().
+  // Returns the input sample rate in Hz and the number of input channels.
+  // These are constants set at instantiation time.
+  virtual int SampleRateHz() const = 0;
+  virtual size_t NumChannels() const = 0;
+
+  // Returns the rate at which the RTP timestamps are updated. The default
+  // implementation returns SampleRateHz().
   virtual int RtpTimestampRateHz() const;
 
   // Returns the number of 10 ms frames the encoder will put in the next
@@ -90,27 +72,72 @@ class AudioEncoder {
   // the encoder may vary the number of 10 ms frames from packet to packet, but
   // it must decide the length of the next packet no later than when outputting
   // the preceding packet.
-  virtual int Num10MsFramesInNextPacket() const = 0;
+  virtual size_t Num10MsFramesInNextPacket() const = 0;
 
   // Returns the maximum value that can be returned by
   // Num10MsFramesInNextPacket().
-  virtual int Max10MsFramesInAPacket() const = 0;
+  virtual size_t Max10MsFramesInAPacket() const = 0;
 
-  // Changes the target bitrate. The implementation is free to alter this value,
-  // e.g., if the desired value is outside the valid range.
-  virtual void SetTargetBitrate(int bits_per_second) {}
+  // Returns the current target bitrate in bits/s. The value -1 means that the
+  // codec adapts the target automatically, and a current target cannot be
+  // provided.
+  virtual int GetTargetBitrate() const = 0;
 
-  // Tells the implementation what the projected packet loss rate is. The rate
-  // is in the range [0.0, 1.0]. This rate is typically used to adjust channel
-  // coding efforts, such as FEC.
-  virtual void SetProjectedPacketLossRate(double fraction) {}
+  // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
+  // NumChannels() samples). Multi-channel audio must be sample-interleaved.
+  // The encoder produces zero or more bytes of output in |encoded| and
+  // returns additional encoding information.
+  // The caller is responsible for making sure that |max_encoded_bytes| is
+  // not smaller than the number of bytes actually produced by the encoder.
+  // Encode() checks some preconditions, calls EncodeInternal() which does the
+  // actual work, and then checks some postconditions.
+  EncodedInfo Encode(uint32_t rtp_timestamp,
+                     rtc::ArrayView<const int16_t> audio,
+                     size_t max_encoded_bytes,
+                     uint8_t* encoded);
 
- protected:
   virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                                     const int16_t* audio,
+                                     rtc::ArrayView<const int16_t> audio,
                                      size_t max_encoded_bytes,
                                      uint8_t* encoded) = 0;
-};
 
+  // Resets the encoder to its starting state, discarding any input that has
+  // been fed to the encoder but not yet emitted in a packet.
+  virtual void Reset() = 0;
+
+  // Enables or disables codec-internal FEC (forward error correction). Returns
+  // true if the codec was able to comply. The default implementation returns
+  // true when asked to disable FEC and false when asked to enable it (meaning
+  // that FEC isn't supported).
+  virtual bool SetFec(bool enable);
+
+  // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
+  // able to comply. The default implementation returns true when asked to
+  // disable DTX and false when asked to enable it (meaning that DTX isn't
+  // supported).
+  virtual bool SetDtx(bool enable);
+
+  // Sets the application mode. Returns true if the codec was able to comply.
+  // The default implementation just returns false.
+  enum class Application { kSpeech, kAudio };
+  virtual bool SetApplication(Application application);
+
+  // Tells the encoder about the highest sample rate the decoder is expected to
+  // use when decoding the bitstream. The encoder would typically use this
+  // information to adjust the quality of the encoding. The default
+  // implementation does nothing.
+  virtual void SetMaxPlaybackRate(int frequency_hz);
+
+  // Tells the encoder what the projected packet loss rate is. The rate is in
+  // the range [0.0, 1.0]. The encoder would typically use this information to
+  // adjust channel coding efforts, such as FEC. The default implementation
+  // does nothing.
+  virtual void SetProjectedPacketLossRate(double fraction);
+
+  // Tells the encoder what average bitrate we'd like it to produce. The
+  // encoder is free to adjust or disregard the given bitrate (the default
+  // implementation does the latter).
+  virtual void SetTargetBitrate(int target_bps);
+};
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 58fd24f53e..180166c40c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 
 #include <algorithm>
 #include <limits>
@@ -19,18 +19,20 @@ namespace {
 
 const int kMaxFrameSizeMs = 60;
 
-}  // namespace
-
-AudioEncoderCng::Config::Config()
-    : num_channels(1),
-      payload_type(13),
-      speech_encoder(NULL),
-      vad_mode(Vad::kVadNormal),
-      sid_frame_interval_ms(100),
-      num_cng_coefficients(8),
-      vad(NULL) {
+rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
+    int sample_rate_hz,
+    int sid_frame_interval_ms,
+    int num_cng_coefficients) {
+  rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst;
+  RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(cng_inst.accept()));
+  RTC_CHECK_EQ(0,
+               WebRtcCng_InitEnc(cng_inst.get(), sample_rate_hz,
+                                 sid_frame_interval_ms, num_cng_coefficients));
+  return cng_inst;
 }
 
+}  // namespace
+
 bool AudioEncoderCng::Config::IsOk() const {
   if (num_channels != 1)
     return false;
@@ -38,7 +40,8 @@ bool AudioEncoderCng::Config::IsOk() const {
     return false;
   if (num_channels != speech_encoder->NumChannels())
     return false;
-  if (sid_frame_interval_ms < speech_encoder->Max10MsFramesInAPacket() * 10)
+  if (sid_frame_interval_ms <
+      static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
     return false;
   if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
       num_cng_coefficients <= 0)
@@ -50,39 +53,16 @@ AudioEncoderCng::AudioEncoderCng(const Config& config)
     : speech_encoder_(config.speech_encoder),
       cng_payload_type_(config.payload_type),
       num_cng_coefficients_(config.num_cng_coefficients),
-      first_timestamp_in_buffer_(0),
-      frames_in_buffer_(0),
+      sid_frame_interval_ms_(config.sid_frame_interval_ms),
       last_frame_active_(true),
-      vad_(new Vad(config.vad_mode)) {
-  if (config.vad) {
-    // Replace default Vad object with user-provided one.
-    vad_.reset(config.vad);
-  }
-  CHECK(config.IsOk()) << "Invalid configuration.";
-  CNG_enc_inst* cng_inst;
-  CHECK_EQ(WebRtcCng_CreateEnc(&cng_inst), 0) << "WebRtcCng_CreateEnc failed.";
-  cng_inst_.reset(cng_inst);  // Transfer ownership to scoped_ptr.
-  CHECK_EQ(WebRtcCng_InitEnc(cng_inst_.get(), SampleRateHz(),
-                             config.sid_frame_interval_ms,
-                             config.num_cng_coefficients),
-           0)
-      << "WebRtcCng_InitEnc failed";
+      vad_(config.vad ? rtc_make_scoped_ptr(config.vad)
+                      : CreateVad(config.vad_mode)) {
+  RTC_CHECK(config.IsOk()) << "Invalid configuration.";
+  cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
+                            num_cng_coefficients_);
 }
 
-AudioEncoderCng::~AudioEncoderCng() {
-}
-
-int AudioEncoderCng::SampleRateHz() const {
-  return speech_encoder_->SampleRateHz();
-}
-
-int AudioEncoderCng::RtpTimestampRateHz() const {
-  return speech_encoder_->RtpTimestampRateHz();
-}
-
-int AudioEncoderCng::NumChannels() const {
-  return 1;
-}
+AudioEncoderCng::~AudioEncoderCng() = default;
 
 size_t AudioEncoderCng::MaxEncodedBytes() const {
   const size_t max_encoded_bytes_active = speech_encoder_->MaxEncodedBytes();
@@ -91,60 +71,62 @@ size_t AudioEncoderCng::MaxEncodedBytes() const {
   return std::max(max_encoded_bytes_active, max_encoded_bytes_passive);
 }
 
-int AudioEncoderCng::Num10MsFramesInNextPacket() const {
+int AudioEncoderCng::SampleRateHz() const {
+  return speech_encoder_->SampleRateHz();
+}
+
+size_t AudioEncoderCng::NumChannels() const {
+  return 1;
+}
+
+int AudioEncoderCng::RtpTimestampRateHz() const {
+  return speech_encoder_->RtpTimestampRateHz();
+}
+
+size_t AudioEncoderCng::Num10MsFramesInNextPacket() const {
   return speech_encoder_->Num10MsFramesInNextPacket();
 }
 
-int AudioEncoderCng::Max10MsFramesInAPacket() const {
+size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
   return speech_encoder_->Max10MsFramesInAPacket();
 }
 
-void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
-  speech_encoder_->SetTargetBitrate(bits_per_second);
-}
-
-void AudioEncoderCng::SetProjectedPacketLossRate(double fraction) {
-  DCHECK_GE(fraction, 0.0);
-  DCHECK_LE(fraction, 1.0);
-  speech_encoder_->SetProjectedPacketLossRate(fraction);
+int AudioEncoderCng::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
 }
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  CHECK_GE(max_encoded_bytes, static_cast<size_t>(num_cng_coefficients_ + 1));
-  const int num_samples = SampleRateHz() / 100 * NumChannels();
-  if (speech_buffer_.empty()) {
-    CHECK_EQ(frames_in_buffer_, 0);
-    first_timestamp_in_buffer_ = rtp_timestamp;
-  }
-  for (int i = 0; i < num_samples; ++i) {
-    speech_buffer_.push_back(audio[i]);
-  }
-  ++frames_in_buffer_;
-  if (frames_in_buffer_ < speech_encoder_->Num10MsFramesInNextPacket()) {
+  RTC_CHECK_GE(max_encoded_bytes,
+               static_cast<size_t>(num_cng_coefficients_ + 1));
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  RTC_CHECK_EQ(speech_buffer_.size(),
+               rtp_timestamps_.size() * samples_per_10ms_frame);
+  rtp_timestamps_.push_back(rtp_timestamp);
+  RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
+  const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
+  if (rtp_timestamps_.size() < frames_to_encode) {
     return EncodedInfo();
   }
-  CHECK_LE(frames_in_buffer_ * 10, kMaxFrameSizeMs)
+  RTC_CHECK_LE(static_cast<int>(frames_to_encode * 10), kMaxFrameSizeMs)
       << "Frame size cannot be larger than " << kMaxFrameSizeMs
       << " ms when using VAD/CNG.";
-  const size_t samples_per_10ms_frame = 10 * SampleRateHz() / 1000;
-  CHECK_EQ(speech_buffer_.size(),
-           static_cast<size_t>(frames_in_buffer_) * samples_per_10ms_frame);
 
   // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
   // following split sizes:
   // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
   // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
-  int blocks_in_first_vad_call =
-      (frames_in_buffer_ > 3 ? 3 : frames_in_buffer_);
-  if (frames_in_buffer_ == 4)
+  size_t blocks_in_first_vad_call =
+      (frames_to_encode > 3 ? 3 : frames_to_encode);
+  if (frames_to_encode == 4)
     blocks_in_first_vad_call = 2;
-  const int blocks_in_second_vad_call =
-      frames_in_buffer_ - blocks_in_first_vad_call;
-  CHECK_GE(blocks_in_second_vad_call, 0);
+  RTC_CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
+  const size_t blocks_in_second_vad_call =
+      frames_to_encode - blocks_in_first_vad_call;
 
   // Check if all of the buffer is passive speech. Start with checking the first
   // block.
@@ -161,12 +143,12 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
   EncodedInfo info;
   switch (activity) {
     case Vad::kPassive: {
-      info = EncodePassive(max_encoded_bytes, encoded);
+      info = EncodePassive(frames_to_encode, max_encoded_bytes, encoded);
       last_frame_active_ = false;
       break;
     }
     case Vad::kActive: {
-      info = EncodeActive(max_encoded_bytes, encoded);
+      info = EncodeActive(frames_to_encode, max_encoded_bytes, encoded);
       last_frame_active_ = true;
       break;
     }
@@ -176,33 +158,76 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
     }
   }
 
-  speech_buffer_.clear();
-  frames_in_buffer_ = 0;
+  speech_buffer_.erase(
+      speech_buffer_.begin(),
+      speech_buffer_.begin() + frames_to_encode * samples_per_10ms_frame);
+  rtp_timestamps_.erase(rtp_timestamps_.begin(),
+                        rtp_timestamps_.begin() + frames_to_encode);
   return info;
 }
 
+void AudioEncoderCng::Reset() {
+  speech_encoder_->Reset();
+  speech_buffer_.clear();
+  rtp_timestamps_.clear();
+  last_frame_active_ = true;
+  vad_->Reset();
+  cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
+                            num_cng_coefficients_);
+}
+
+bool AudioEncoderCng::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCng::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCng::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+void AudioEncoderCng::SetProjectedPacketLossRate(double fraction) {
+  speech_encoder_->SetProjectedPacketLossRate(fraction);
+}
+
+void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
+  speech_encoder_->SetTargetBitrate(bits_per_second);
+}
+
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
+    size_t frames_to_encode,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   bool force_sid = last_frame_active_;
   bool output_produced = false;
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
-  CHECK_GE(max_encoded_bytes, frames_in_buffer_ * samples_per_10ms_frame);
+  RTC_CHECK_GE(max_encoded_bytes, frames_to_encode * samples_per_10ms_frame);
   AudioEncoder::EncodedInfo info;
-  for (int i = 0; i < frames_in_buffer_; ++i) {
-    int16_t encoded_bytes_tmp = 0;
-    CHECK_GE(WebRtcCng_Encode(cng_inst_.get(),
-                              &speech_buffer_[i * samples_per_10ms_frame],
-                              static_cast<int16_t>(samples_per_10ms_frame),
-                              encoded, &encoded_bytes_tmp, force_sid), 0);
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    // It's important not to pass &info.encoded_bytes directly to
+    // WebRtcCng_Encode(), since later loop iterations may return zero in that
+    // value, in which case we don't want to overwrite any value from an earlier
+    // iteration.
+    size_t encoded_bytes_tmp = 0;
+    RTC_CHECK_GE(WebRtcCng_Encode(cng_inst_.get(),
+                                  &speech_buffer_[i * samples_per_10ms_frame],
+                                  samples_per_10ms_frame, encoded,
+                                  &encoded_bytes_tmp, force_sid),
+                 0);
     if (encoded_bytes_tmp > 0) {
-      CHECK(!output_produced);
-      info.encoded_bytes = static_cast<size_t>(encoded_bytes_tmp);
+      RTC_CHECK(!output_produced);
+      info.encoded_bytes = encoded_bytes_tmp;
       output_produced = true;
       force_sid = false;
     }
   }
-  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.encoded_timestamp = rtp_timestamps_.front();
   info.payload_type = cng_payload_type_;
   info.send_even_if_empty = true;
   info.speech = false;
@@ -210,16 +235,23 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
 }
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
+    size_t frames_to_encode,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   AudioEncoder::EncodedInfo info;
-  for (int i = 0; i < frames_in_buffer_; ++i) {
-    info = speech_encoder_->Encode(
-        first_timestamp_in_buffer_, &speech_buffer_[i * samples_per_10ms_frame],
-        samples_per_10ms_frame, max_encoded_bytes, encoded);
-    if (i < frames_in_buffer_ - 1) {
-      CHECK_EQ(info.encoded_bytes, 0u) << "Encoder delivered data too early.";
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    info =
+        speech_encoder_->Encode(rtp_timestamps_.front(),
+                                rtc::ArrayView<const int16_t>(
+                                    &speech_buffer_[i * samples_per_10ms_frame],
+                                    samples_per_10ms_frame),
+                                max_encoded_bytes, encoded);
+    if (i + 1 == frames_to_encode) {
+      RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
+    } else {
+      RTC_CHECK_EQ(info.encoded_bytes, 0u)
+          << "Encoder delivered data too early.";
     }
   }
   return info;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
similarity index 52%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index daecd51ff3..87383e2ac5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -8,80 +8,88 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
 
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/include/vad.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 
 namespace webrtc {
 
+// Deleter for use with scoped_ptr.
+struct CngInstDeleter {
+  void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); }
+};
+
 class Vad;
 
 class AudioEncoderCng final : public AudioEncoder {
  public:
   struct Config {
-    Config();
     bool IsOk() const;
 
-    int num_channels;
-    int payload_type;
+    size_t num_channels = 1;
+    int payload_type = 13;
     // Caller keeps ownership of the AudioEncoder object.
-    AudioEncoder* speech_encoder;
-    Vad::Aggressiveness vad_mode;
-    int sid_frame_interval_ms;
-    int num_cng_coefficients;
+    AudioEncoder* speech_encoder = nullptr;
+    Vad::Aggressiveness vad_mode = Vad::kVadNormal;
+    int sid_frame_interval_ms = 100;
+    int num_cng_coefficients = 8;
     // The Vad pointer is mainly for testing. If a NULL pointer is passed, the
     // AudioEncoderCng creates (and destroys) a Vad object internally. If an
     // object is passed, the AudioEncoderCng assumes ownership of the Vad
     // object.
-    Vad* vad;
+    Vad* vad = nullptr;
   };
 
   explicit AudioEncoderCng(const Config& config);
-
   ~AudioEncoderCng() override;
 
-  int SampleRateHz() const override;
-  int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
-  void SetTargetBitrate(int bits_per_second) override;
-  void SetProjectedPacketLossRate(double fraction) override;
-
- protected:
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
+  void Reset() override;
+  bool SetFec(bool enable) override;
+  bool SetDtx(bool enable) override;
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
 
  private:
-  // Deleter for use with scoped_ptr. E.g., use as
-  //   rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
-  struct CngInstDeleter {
-    inline void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); }
-  };
-
-  EncodedInfo EncodePassive(size_t max_encoded_bytes, uint8_t* encoded);
-  EncodedInfo EncodeActive(size_t max_encoded_bytes, uint8_t* encoded);
+  EncodedInfo EncodePassive(size_t frames_to_encode,
+                            size_t max_encoded_bytes,
+                            uint8_t* encoded);
+  EncodedInfo EncodeActive(size_t frames_to_encode,
+                           size_t max_encoded_bytes,
+                           uint8_t* encoded);
   size_t SamplesPer10msFrame() const;
 
   AudioEncoder* speech_encoder_;
   const int cng_payload_type_;
   const int num_cng_coefficients_;
+  const int sid_frame_interval_ms_;
   std::vector<int16_t> speech_buffer_;
-  uint32_t first_timestamp_in_buffer_;
-  int frames_in_buffer_;
+  std::vector<uint32_t> rtp_timestamps_;
   bool last_frame_active_;
   rtc::scoped_ptr<Vad> vad_;
   rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index a31f0deb1d..feb3ed1f0a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -13,7 +13,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/mock/mock_vad.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 
 using ::testing::Return;
@@ -34,7 +34,7 @@ static const int kCngPayloadType = 18;
 class AudioEncoderCngTest : public ::testing::Test {
  protected:
   AudioEncoderCngTest()
-      : mock_vad_(new MockVad(Vad::kVadNormal)),
+      : mock_vad_(new MockVad),
         timestamp_(4711),
         num_audio_samples_10ms_(0),
         sample_rate_hz_(8000) {
@@ -59,14 +59,14 @@ class AudioEncoderCngTest : public ::testing::Test {
   void CreateCng() {
     // The config_ parameters may be changed by the TEST_Fs up until CreateCng()
     // is called, thus we cannot use the values until now.
-    num_audio_samples_10ms_ = 10 * sample_rate_hz_ / 1000;
+    num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000);
     ASSERT_LE(num_audio_samples_10ms_, kMaxNumSamples);
     EXPECT_CALL(mock_encoder_, SampleRateHz())
         .WillRepeatedly(Return(sample_rate_hz_));
     // Max10MsFramesInAPacket() is just used to verify that the SID frame period
     // is not too small. The return value does not matter that much, as long as
     // it is smaller than 10.
-    EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(1));
+    EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(1u));
     EXPECT_CALL(mock_encoder_, MaxEncodedBytes())
         .WillRepeatedly(Return(kMockMaxEncodedBytes));
     cng_.reset(new AudioEncoderCng(config_));
@@ -75,15 +75,32 @@ class AudioEncoderCngTest : public ::testing::Test {
 
   void Encode() {
     ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
-    encoded_info_ = cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_,
-                                 encoded_.size(), &encoded_[0]);
-    timestamp_ += num_audio_samples_10ms_;
+    encoded_info_ = cng_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+        encoded_.size(), &encoded_[0]);
+    timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
+  }
+
+  // Expect |num_calls| calls to the encoder, all successful. The last call
+  // claims to have encoded |kMockMaxEncodedBytes| bytes, and all the preceding
+  // ones 0 bytes.
+  void ExpectEncodeCalls(size_t num_calls) {
+    InSequence s;
+    AudioEncoder::EncodedInfo info;
+    for (size_t j = 0; j < num_calls - 1; ++j) {
+      EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
+          .WillOnce(Return(info));
+    }
+    info.encoded_bytes = kMockReturnEncodedBytes;
+    EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
+        .WillOnce(Return(info));
   }
 
   // Verifies that the cng_ object waits until it has collected
   // |blocks_per_frame| blocks of audio, and then dispatches all of them to
   // the underlying codec (speech or cng).
-  void CheckBlockGrouping(int blocks_per_frame, bool active_speech) {
+  void CheckBlockGrouping(size_t blocks_per_frame, bool active_speech) {
     EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
         .WillRepeatedly(Return(blocks_per_frame));
     CreateCng();
@@ -92,24 +109,12 @@ class AudioEncoderCngTest : public ::testing::Test {
 
     // Don't expect any calls to the encoder yet.
     EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).Times(0);
-    for (int i = 0; i < blocks_per_frame - 1; ++i) {
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
       Encode();
       EXPECT_EQ(0u, encoded_info_.encoded_bytes);
     }
-    if (active_speech) {
-      // Now expect |blocks_per_frame| calls to the encoder in sequence.
-      // Let the speech codec mock return true and set the number of encoded
-      // bytes to |kMockReturnEncodedBytes|.
-      InSequence s;
-      AudioEncoder::EncodedInfo info;
-      for (int j = 0; j < blocks_per_frame - 1; ++j) {
-        EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
-            .WillOnce(Return(info));
-      }
-      info.encoded_bytes = kMockReturnEncodedBytes;
-      EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
-          .WillOnce(Return(info));
-    }
+    if (active_speech)
+      ExpectEncodeCalls(blocks_per_frame);
     Encode();
     if (active_speech) {
       EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
@@ -124,14 +129,15 @@ class AudioEncoderCngTest : public ::testing::Test {
   void CheckVadInputSize(int input_frame_size_ms,
                          int expected_first_block_size_ms,
                          int expected_second_block_size_ms) {
-    const int blocks_per_frame = input_frame_size_ms / 10;
+    const size_t blocks_per_frame =
+        static_cast<size_t>(input_frame_size_ms / 10);
 
     EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
         .WillRepeatedly(Return(blocks_per_frame));
 
     // Expect nothing to happen before the last block is sent to cng_.
     EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)).Times(0);
-    for (int i = 0; i < blocks_per_frame - 1; ++i) {
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
       Encode();
     }
 
@@ -160,7 +166,7 @@ class AudioEncoderCngTest : public ::testing::Test {
                                Vad::Activity second_type) {
     // Set the speech encoder frame size to 60 ms, to ensure that the VAD will
     // be called twice.
-    const int blocks_per_frame = 6;
+    const size_t blocks_per_frame = 6;
     EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
         .WillRepeatedly(Return(blocks_per_frame));
     InSequence s;
@@ -172,7 +178,7 @@ class AudioEncoderCngTest : public ::testing::Test {
           .WillOnce(Return(second_type));
     }
     encoded_info_.payload_type = 0;
-    for (int i = 0; i < blocks_per_frame; ++i) {
+    for (size_t i = 0; i < blocks_per_frame; ++i) {
       Encode();
     }
     return encoded_info_.payload_type != kCngPayloadType;
@@ -196,8 +202,8 @@ TEST_F(AudioEncoderCngTest, CreateAndDestroy) {
 
 TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
   CreateCng();
-  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17));
-  EXPECT_EQ(17, cng_->Num10MsFramesInNextPacket());
+  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, cng_->Num10MsFramesInNextPacket());
 }
 
 TEST_F(AudioEncoderCngTest, CheckChangeBitratePropagation) {
@@ -214,7 +220,7 @@ TEST_F(AudioEncoderCngTest, CheckProjectedPacketLossRatePropagation) {
 
 TEST_F(AudioEncoderCngTest, EncodeCallsVad) {
   EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
-      .WillRepeatedly(Return(1));
+      .WillRepeatedly(Return(1U));
   CreateCng();
   EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
       .WillOnce(Return(Vad::kPassive));
@@ -246,7 +252,7 @@ TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksActiveSpeech) {
 }
 
 TEST_F(AudioEncoderCngTest, EncodePassive) {
-  const int kBlocksPerFrame = 3;
+  const size_t kBlocksPerFrame = 3;
   EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
       .WillRepeatedly(Return(kBlocksPerFrame));
   CreateCng();
@@ -255,7 +261,7 @@ TEST_F(AudioEncoderCngTest, EncodePassive) {
   // Expect no calls at all to the speech encoder mock.
   EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).Times(0);
   uint32_t expected_timestamp = timestamp_;
-  for (int i = 0; i < 100; ++i) {
+  for (size_t i = 0; i < 100; ++i) {
     Encode();
     // Check if it was time to call the cng encoder. This is done once every
     // |kBlocksPerFrame| calls.
@@ -283,23 +289,17 @@ TEST_F(AudioEncoderCngTest, MixedActivePassive) {
   CreateCng();
 
   // All of the frame is active speech.
-  EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
-      .Times(6)
-      .WillRepeatedly(Return(AudioEncoder::EncodedInfo()));
+  ExpectEncodeCalls(6);
   EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kActive));
   EXPECT_TRUE(encoded_info_.speech);
 
   // First half of the frame is active speech.
-  EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
-      .Times(6)
-      .WillRepeatedly(Return(AudioEncoder::EncodedInfo()));
+  ExpectEncodeCalls(6);
   EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kPassive));
   EXPECT_TRUE(encoded_info_.speech);
 
   // Second half of the frame is active speech.
-  EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _))
-      .Times(6)
-      .WillRepeatedly(Return(AudioEncoder::EncodedInfo()));
+  ExpectEncodeCalls(6);
   EXPECT_TRUE(CheckMixedActivePassive(Vad::kPassive, Vad::kActive));
   EXPECT_TRUE(encoded_info_.speech);
 
@@ -342,7 +342,7 @@ TEST_F(AudioEncoderCngTest, VadInputSize60Ms) {
 TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) {
   CreateCng();
   EXPECT_CALL(mock_encoder_, EncodeInternal(_, _, _, _)).Times(0);
-  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1));
+  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1U));
   EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
       .WillOnce(Return(Vad::kPassive));
   encoded_info_.payload_type = 0;
@@ -355,7 +355,7 @@ TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) {
 TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) {
   CreateCng();
   EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
-      .WillRepeatedly(Return(1));
+      .WillRepeatedly(Return(1U));
   // Start with encoding noise.
   EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
       .Times(2)
@@ -388,6 +388,14 @@ TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) {
             encoded_info_.encoded_bytes);
 }
 
+// Resetting the CNG should reset both the VAD and the encoder.
+TEST_F(AudioEncoderCngTest, Reset) {
+  CreateCng();
+  EXPECT_CALL(mock_encoder_, Reset()).Times(1);
+  EXPECT_CALL(*mock_vad_, Reset()).Times(1);
+  cng_->Reset();
+}
+
 #if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 // This test fixture tests various error conditions that makes the
@@ -446,7 +454,7 @@ TEST_F(AudioEncoderCngDeathTest, Stereo) {
 TEST_F(AudioEncoderCngDeathTest, EncoderFrameSizeTooLarge) {
   CreateCng();
   EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket())
-      .WillRepeatedly(Return(7));
+      .WillRepeatedly(Return(7U));
   for (int i = 0; i < 6; ++i)
     Encode();
   EXPECT_DEATH(Encode(),
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng.gypi
index af9fbd3af0..c020f4740d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng.gypi
@@ -9,29 +9,19 @@
 {
   'targets': [
     {
-      'target_name': 'CNG',
+      'target_name': 'cng',
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/audio_encoder_cng.h',
-        'include/webrtc_cng.h',
         'audio_encoder_cng.cc',
-        'webrtc_cng.c',
+        'audio_encoder_cng.h',
         'cng_helpfuns.c',
         'cng_helpfuns.h',
+        'webrtc_cng.c',
+        'webrtc_cng.h',
       ],
     },
   ], # targets
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
index 0d1c670290..1061dca69a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -99,7 +99,7 @@ TEST_F(CngTest, CngInitFail) {
 
 TEST_F(CngTest, CngEncode) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
@@ -151,7 +151,7 @@ TEST_F(CngTest, CngEncode) {
 // Encode Cng with too long input vector.
 TEST_F(CngTest, CngEncodeTooLong) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create and init encoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
@@ -170,7 +170,7 @@ TEST_F(CngTest, CngEncodeTooLong) {
 // Call encode without calling init.
 TEST_F(CngTest, CngEncodeNoInit) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
@@ -187,14 +187,14 @@ TEST_F(CngTest, CngEncodeNoInit) {
 // Update SID parameters, for both 9 and 16 parameters.
 TEST_F(CngTest, CngUpdateSid) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create and initialize encoder and decoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Run normal Encode and UpdateSid.
   EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
@@ -205,7 +205,7 @@ TEST_F(CngTest, CngUpdateSid) {
   // Reinit with new length.
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsHigh));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Expect 0 because of unstable parameters after switching length.
   EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
@@ -224,7 +224,7 @@ TEST_F(CngTest, CngUpdateSid) {
 // Update SID parameters, with wrong parameters or without calling decode.
 TEST_F(CngTest, CngUpdateSidErroneous) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create encoder and decoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
@@ -242,7 +242,7 @@ TEST_F(CngTest, CngUpdateSidErroneous) {
   EXPECT_EQ(6220, WebRtcCng_GetErrorCodeDec(cng_dec_inst_));
 
   // Initialize decoder.
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // First run with valid parameters, then with too many CNG parameters.
   // The function will operate correctly by only reading the maximum number of
@@ -261,14 +261,14 @@ TEST_F(CngTest, CngUpdateSidErroneous) {
 TEST_F(CngTest, CngGenerate) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
   int16_t out_data[640];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create and initialize encoder and decoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Normal Encode.
   EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
@@ -294,14 +294,14 @@ TEST_F(CngTest, CngGenerate) {
 // Test automatic SID.
 TEST_F(CngTest, CngAutoSid) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create and initialize encoder and decoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // Normal Encode, 100 msec, where no SID data should be generated.
   for (int i = 0; i < 10; i++) {
@@ -321,14 +321,14 @@ TEST_F(CngTest, CngAutoSid) {
 // Test automatic SID, with very short interval.
 TEST_F(CngTest, CngAutoSidShort) {
   uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t number_bytes;
+  size_t number_bytes;
 
   // Create and initialize encoder and decoder memory.
   EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
   EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
   EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidShortIntervalUpdate,
                                  kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_InitDec(cng_dec_inst_));
+  WebRtcCng_InitDec(cng_dec_inst_);
 
   // First call will never generate SID, unless forced to.
   EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index 9862f12537..8dddc5c717 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -35,8 +35,8 @@ typedef struct WebRtcCngDecoder_ {
 } WebRtcCngDecoder;
 
 typedef struct WebRtcCngEncoder_ {
-  int16_t enc_nrOfCoefs;
-  uint16_t enc_sampfreq;
+  size_t enc_nrOfCoefs;
+  int enc_sampfreq;
   int16_t enc_interval;
   int16_t enc_msSinceSID;
   int32_t enc_Energy;
@@ -142,8 +142,8 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst) {
  * Return value       :  0 - Ok
  *                      -1 - Error
  */
-int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
-                          int16_t quality) {
+int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
+                      int16_t quality) {
   int i;
   WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
   memset(inst, 0, sizeof(WebRtcCngEncoder));
@@ -169,7 +169,7 @@ int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
   return 0;
 }
 
-int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
+void WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
   int i;
 
   WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
@@ -188,8 +188,6 @@ int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
   inst->dec_used_reflCoefs[0] = 0;
   inst->dec_used_energy = 0;
   inst->initflag = 1;
-
-  return 0;
 }
 
 /****************************************************************************
@@ -227,9 +225,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst) {
  * Return value       :  0 - Ok
  *                      -1 - Error
  */
-int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
-                         int16_t nrOfSamples, uint8_t* SIDdata,
-                         int16_t* bytesOut, int16_t forceSID) {
+int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
+                     size_t nrOfSamples, uint8_t* SIDdata,
+                     size_t* bytesOut, int16_t forceSID) {
   WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
 
   int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
@@ -240,10 +238,11 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
   int16_t ReflBetaComp = 13107;  /* 0.4 in q15. */
   int32_t outEnergy;
   int outShifts;
-  int i, stab;
+  size_t i;
+  int stab;
   int acorrScale;
-  int index;
-  int16_t ind, factor;
+  size_t index;
+  size_t ind, factor;
   int32_t* bptr;
   int32_t blo, bhi;
   int16_t negate;
@@ -281,7 +280,7 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
       outShifts--;
     }
   }
-  outEnergy = WebRtcSpl_DivW32W16(outEnergy, factor);
+  outEnergy = WebRtcSpl_DivW32W16(outEnergy, (int16_t)factor);
 
   if (outEnergy > 1) {
     /* Create Hanning Window. */
@@ -370,7 +369,7 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
     }
     if ((i == 93) && (index == 0))
       index = 94;
-    SIDdata[0] = index;
+    SIDdata[0] = (uint8_t)index;
 
     /* Quantize coefficients with tweak for WebRtc implementation of RFC3389. */
     if (inst->enc_nrOfCoefs == WEBRTC_CNG_MAX_LPC_ORDER) {
@@ -388,10 +387,12 @@ int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
     inst->enc_msSinceSID = 0;
     *bytesOut = inst->enc_nrOfCoefs + 1;
 
-    inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
-    return inst->enc_nrOfCoefs + 1;
+    inst->enc_msSinceSID +=
+        (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
+    return (int)(inst->enc_nrOfCoefs + 1);
   } else {
-    inst->enc_msSinceSID += (1000 * nrOfSamples) / inst->enc_sampfreq;
+    inst->enc_msSinceSID +=
+        (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
     *bytesOut = 0;
     return 0;
   }
@@ -473,10 +474,10 @@ int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
  *                       -1 - Error
  */
 int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
-                           int16_t nrOfSamples, int16_t new_period) {
+                           size_t nrOfSamples, int16_t new_period) {
   WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
 
-  int i;
+  size_t i;
   int16_t excitation[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
   int16_t low[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
   int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
index b016f4017f..64bea1e26f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -9,8 +9,8 @@
  */
 
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
 
 #include <stddef.h>
 #include "webrtc/typedefs.h"
@@ -68,9 +68,9 @@ int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
  *                      -1 - Error
  */
 
-int16_t WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, uint16_t fs, int16_t interval,
-                          int16_t quality);
-int16_t WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
+int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
+                      int16_t quality);
+void WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
 
 /****************************************************************************
  * WebRtcCng_FreeEnc/Dec(...)
@@ -103,9 +103,9 @@ int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst);
  * Return value       :  0 - Ok
  *                      -1 - Error
  */
-int16_t WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
-                         int16_t nrOfSamples, uint8_t* SIDdata,
-                         int16_t* bytesOut, int16_t forceSID);
+int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
+                     size_t nrOfSamples, uint8_t* SIDdata,
+                     size_t* bytesOut, int16_t forceSID);
 
 /****************************************************************************
  * WebRtcCng_UpdateSid(...)
@@ -138,13 +138,13 @@ int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
  *                      -1 - Error
  */
 int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
-                           int16_t nrOfSamples, int16_t new_period);
+                           size_t nrOfSamples, int16_t new_period);
 
 /*****************************************************************************
  * WebRtcCng_GetErrorCodeEnc/Dec(...)
  *
  * This functions can be used to check the error code of a CNG instance. When
- * a function returns -1 a error code will be set for that instance. The 
+ * a function returns -1 a error code will be set for that instance. The
  * function below extract the code of the last error that occurred in the
  * specified instance.
  *
@@ -160,4 +160,4 @@ int16_t WebRtcCng_GetErrorCodeDec(CNG_dec_inst* cng_inst);
 }
 #endif
 
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
new file mode 100644
index 0000000000..9757b4a010
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
+
+namespace webrtc {
+
+void AudioDecoderPcmU::Reset() {}
+
+size_t AudioDecoderPcmU::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+void AudioDecoderPcmA::Reset() {}
+
+size_t AudioDecoderPcmA::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
new file mode 100644
index 0000000000..9dc3a6fd7a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+namespace webrtc {
+
+class AudioDecoderPcmU final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1u);
+  }
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
+};
+
+class AudioDecoderPcmA final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1u);
+  }
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 5c45fa5178..ff61db8e8d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -8,101 +8,126 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 #include <limits>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 
 namespace {
-int16_t NumSamplesPerFrame(int num_channels,
-                           int frame_size_ms,
-                           int sample_rate_hz) {
-  int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
-  CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
-      << "Frame size too large.";
-  return static_cast<int16_t>(samples_per_frame);
+
+template <typename T>
+typename T::Config CreateConfig(const CodecInst& codec_inst) {
+  typename T::Config config;
+  config.frame_size_ms = codec_inst.pacsize / 8;
+  config.num_channels = codec_inst.channels;
+  config.payload_type = codec_inst.pltype;
+  return config;
 }
+
 }  // namespace
 
+bool AudioEncoderPcm::Config::IsOk() const {
+  return (frame_size_ms % 10 == 0) && (num_channels >= 1);
+}
+
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     : sample_rate_hz_(sample_rate_hz),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
-      num_10ms_frames_per_packet_(config.frame_size_ms / 10),
-      full_frame_samples_(NumSamplesPerFrame(config.num_channels,
-                                             config.frame_size_ms,
-                                             sample_rate_hz_)),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      full_frame_samples_(
+          config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
       first_timestamp_in_buffer_(0) {
-  CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
-  CHECK_EQ(config.frame_size_ms % 10, 0)
+  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
+  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
       << "Frame size must be an integer multiple of 10 ms.";
   speech_buffer_.reserve(full_frame_samples_);
 }
 
-AudioEncoderPcm::~AudioEncoderPcm() {
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+size_t AudioEncoderPcm::MaxEncodedBytes() const {
+  return full_frame_samples_ * BytesPerSample();
 }
 
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
-int AudioEncoderPcm::NumChannels() const {
+size_t AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
-size_t AudioEncoderPcm::MaxEncodedBytes() const {
-  return full_frame_samples_;
-}
-
-int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderPcm::Max10MsFramesInAPacket() const {
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return static_cast<int>(
+      8 * BytesPerSample() * SampleRateHz() * NumChannels());
+}
+
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  const int num_samples = SampleRateHz() / 100 * NumChannels();
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
-  for (int i = 0; i < num_samples; ++i) {
-    speech_buffer_.push_back(audio[i]);
-  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
-  CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
-  CHECK_GE(max_encoded_bytes, full_frame_samples_);
-  int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
-  CHECK_GE(ret, 0);
-  speech_buffer_.clear();
+  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
+  RTC_CHECK_GE(max_encoded_bytes, full_frame_samples_);
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
-  info.encoded_bytes = static_cast<size_t>(ret);
+  info.encoded_bytes =
+      EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
+  speech_buffer_.clear();
   return info;
 }
 
-int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
-                                     size_t input_len,
-                                     uint8_t* encoded) {
-  return WebRtcG711_EncodeA(audio, static_cast<int16_t>(input_len), encoded);
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
 }
 
-int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
-                                     size_t input_len,
-                                     uint8_t* encoded) {
-  return WebRtcG711_EncodeU(audio, static_cast<int16_t>(input_len), encoded);
+AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
+    : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
+
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
+    : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
+
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
similarity index 55%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index 6e588ecfcd..b839488628 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
 
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
@@ -21,8 +22,10 @@ class AudioEncoderPcm : public AudioEncoder {
  public:
   struct Config {
    public:
+    bool IsOk() const;
+
     int frame_size_ms;
-    int num_channels;
+    size_t num_channels;
     int payload_type;
 
    protected:
@@ -32,35 +35,40 @@ class AudioEncoderPcm : public AudioEncoder {
 
   ~AudioEncoderPcm() override;
 
-  int SampleRateHz() const override;
-  int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             size_t max_encoded_bytes,
+                             uint8_t* encoded) override;
+  void Reset() override;
 
  protected:
   AudioEncoderPcm(const Config& config, int sample_rate_hz);
 
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
+  virtual size_t EncodeCall(const int16_t* audio,
+                            size_t input_len,
+                            uint8_t* encoded) = 0;
 
-  virtual int16_t EncodeCall(const int16_t* audio,
-                             size_t input_len,
-                             uint8_t* encoded) = 0;
+  virtual size_t BytesPerSample() const = 0;
 
  private:
   const int sample_rate_hz_;
-  const int num_channels_;
+  const size_t num_channels_;
   const int payload_type_;
-  const int num_10ms_frames_per_packet_;
+  const size_t num_10ms_frames_per_packet_;
   const size_t full_frame_samples_;
   std::vector<int16_t> speech_buffer_;
   uint32_t first_timestamp_in_buffer_;
 };
 
-class AudioEncoderPcmA : public AudioEncoderPcm {
+struct CodecInst;
+
+class AudioEncoderPcmA final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
     Config() : AudioEncoderPcm::Config(8) {}
@@ -68,17 +76,21 @@ class AudioEncoderPcmA : public AudioEncoderPcm {
 
   explicit AudioEncoderPcmA(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
+  explicit AudioEncoderPcmA(const CodecInst& codec_inst);
 
  protected:
-  int16_t EncodeCall(const int16_t* audio,
-                     size_t input_len,
-                     uint8_t* encoded) override;
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
 };
 
-class AudioEncoderPcmU : public AudioEncoderPcm {
+class AudioEncoderPcmU final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
     Config() : AudioEncoderPcm::Config(0) {}
@@ -86,15 +98,20 @@ class AudioEncoderPcmU : public AudioEncoderPcm {
 
   explicit AudioEncoderPcmU(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
+  explicit AudioEncoderPcmU(const CodecInst& codec_inst);
 
  protected:
-  int16_t EncodeCall(const int16_t* audio,
-                     size_t input_len,
-                     uint8_t* encoded) override;
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711.gypi
index 779f05339a..4b902809ea 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711.gypi
@@ -9,28 +9,20 @@
 {
   'targets': [
     {
-      'target_name': 'G711',
+      'target_name': 'g711',
       'type': 'static_library',
       'dependencies': [
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/g711_interface.h',
-        'include/audio_encoder_pcm.h',
+        'audio_decoder_pcm.cc',
+        'audio_decoder_pcm.h',
+        'audio_encoder_pcm.cc',
+        'audio_encoder_pcm.h',
         'g711_interface.c',
+        'g711_interface.h',
         'g711.c',
         'g711.h',
-        'audio_encoder_pcm.cc',
       ],
     },
   ], # targets
@@ -41,7 +33,7 @@
           'target_name': 'g711_test',
           'type': 'executable',
           'dependencies': [
-            'G711',
+            'g711',
           ],
           'sources': [
             'test/testG711.cc',
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
index 809a70e883..5b96a9c555 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
@@ -12,53 +12,46 @@
 #include "g711_interface.h"
 #include "webrtc/typedefs.h"
 
-int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
-                           int16_t len,
-                           uint8_t* encoded) {
-  int n;
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
   for (n = 0; n < len; n++)
     encoded[n] = linear_to_alaw(speechIn[n]);
   return len;
 }
 
-int16_t WebRtcG711_EncodeU(const int16_t* speechIn,
-                           int16_t len,
-                           uint8_t* encoded) {
-  int n;
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
   for (n = 0; n < len; n++)
     encoded[n] = linear_to_ulaw(speechIn[n]);
   return len;
 }
 
-int16_t WebRtcG711_DecodeA(const uint8_t* encoded,
-                           int16_t len,
-                           int16_t* decoded,
-                           int16_t* speechType) {
-  int n;
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
   for (n = 0; n < len; n++)
     decoded[n] = alaw_to_linear(encoded[n]);
   *speechType = 1;
   return len;
 }
 
-int16_t WebRtcG711_DecodeU(const uint8_t* encoded,
-                           int16_t len,
-                           int16_t* decoded,
-                           int16_t* speechType) {
-  int n;
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
   for (n = 0; n < len; n++)
     decoded[n] = ulaw_to_linear(encoded[n]);
   *speechType = 1;
   return len;
 }
 
-int WebRtcG711_DurationEst(const uint8_t* payload,
-                           int payload_length_bytes) {
-  (void) payload;
-  /* G.711 is one byte per sample, so we can just return the number of bytes. */
-  return payload_length_bytes;
-}
-
 int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
   strncpy(version, "2.0.0", lenBytes);
   return 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
similarity index 70%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
index 0b798a6b9d..00854bbb2c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -38,9 +38,9 @@ extern "C" {
  *                             Always equal to len input parameter.
  */
 
-int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
-                           int16_t len,
-                           uint8_t* encoded);
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
 
 /****************************************************************************
  * WebRtcG711_EncodeU(...)
@@ -59,9 +59,9 @@ int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
  *                             Always equal to len input parameter.
  */
 
-int16_t WebRtcG711_EncodeU(const int16_t* speechIn,
-                           int16_t len,
-                           uint8_t* encoded);
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
 
 /****************************************************************************
  * WebRtcG711_DecodeA(...)
@@ -82,10 +82,10 @@ int16_t WebRtcG711_EncodeU(const int16_t* speechIn,
  *                             -1 - Error
  */
 
-int16_t WebRtcG711_DecodeA(const uint8_t* encoded,
-                           int16_t len,
-                           int16_t* decoded,
-                           int16_t* speechType);
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
 
 /****************************************************************************
  * WebRtcG711_DecodeU(...)
@@ -106,27 +106,10 @@ int16_t WebRtcG711_DecodeA(const uint8_t* encoded,
  *                             -1 - Error
  */
 
-int16_t WebRtcG711_DecodeU(const uint8_t* encoded,
-                           int16_t len,
-                           int16_t* decoded,
-                           int16_t* speechType);
-
-/****************************************************************************
- * WebRtcG711_DurationEst(...)
- *
- * This function estimates the duration of a G711 packet in samples.
- *
- * Input:
- *      - payload            : Encoded data
- *      - payloadLengthBytes : Bytes in encoded vector
- *
- * Return value              : The duration of the packet in samples, which is
- *                             just payload_length_bytes, since G.711 uses one
- *                             byte per sample.
- */
-
-int WebRtcG711_DurationEst(const uint8_t* payload,
-                           int payload_length_bytes);
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
 
 /**********************************************************************
 * WebRtcG711_Version(...)
@@ -149,4 +132,4 @@ int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
 }
 #endif
 
-#endif /* MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_ */
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
index e891810ca9..5675b1f8b0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -17,25 +17,19 @@
 #include <string.h>
 
 /* include API */
-#include "g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 /* Runtime statistics */
 #include <time.h>
 #define CLOCKS_PER_SEC_G711 1000
 
 /* function for reading audio data from PCM file */
-int readframe(int16_t* data, FILE* inp, int length) {
-
-  short k, rlen, status = 0;
-
-  rlen = (short) fread(data, sizeof(int16_t), length, inp);
-  if (rlen < length) {
-    for (k = rlen; k < length; k++)
-      data[k] = 0;
-    status = 1;
-  }
-
-  return status;
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
 }
 
 int main(int argc, char* argv[]) {
@@ -43,18 +37,17 @@ int main(int argc, char* argv[]) {
   FILE* inp;
   FILE* outp;
   FILE* bitp = NULL;
-  int framecnt, endfile;
+  int framecnt;
+  bool endfile;
 
-  int16_t framelength = 80;
-
-  int err;
+  size_t framelength = 80;
 
   /* Runtime statistics */
   double starttime;
   double runtime;
   double length_file;
 
-  int16_t stream_len = 0;
+  size_t stream_len = 0;
   int16_t shortdata[480];
   int16_t decoded[480];
   uint8_t streamdata[1000];
@@ -85,7 +78,12 @@ int main(int argc, char* argv[]) {
   printf("-----------------------------------\n");
   printf("G.711 version: %s\n\n", versionNumber);
   /* Get frame length */
-  framelength = atoi(argv[1]);
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+      printf("  G.722: Invalid framelength %d.\n", framelength_int);
+      exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
 
   /* Get compression law */
   strcpy(law, argv[2]);
@@ -118,8 +116,8 @@ int main(int argc, char* argv[]) {
 
   /* Initialize encoder and decoder */
   framecnt = 0;
-  endfile = 0;
-  while (endfile == 0) {
+  endfile = false;
+  while (!endfile) {
     framecnt++;
     /* Read speech block */
     endfile = readframe(shortdata, inp, framelength);
@@ -131,36 +129,29 @@ int main(int argc, char* argv[]) {
       if (argc == 6) {
         /* Write bits to file */
         if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
-            static_cast<size_t>(stream_len)) {
+            stream_len) {
           return -1;
         }
       }
-      err = WebRtcG711_DecodeA(streamdata, stream_len, decoded,
-                               speechType);
+      WebRtcG711_DecodeA(streamdata, stream_len, decoded, speechType);
     } else if (!strcmp(law, "u")) {
       /* u-law encoding */
       stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata);
       if (argc == 6) {
         /* Write bits to file */
         if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
-            static_cast<size_t>(stream_len)) {
+            stream_len) {
           return -1;
         }
       }
-      err = WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType);
+      WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType);
     } else {
       printf("Wrong law mode\n");
       exit(1);
     }
-    if (stream_len < 0 || err < 0) {
-      /* exit if returned with error */
-      printf("Error in encoder/decoder\n");
-    } else {
-      /* Write coded speech to file */
-      if (fwrite(decoded, sizeof(short), framelength, outp) !=
-          static_cast<size_t>(framelength)) {
-        return -1;
-      }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
     }
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
new file mode 100644
index 0000000000..7676e90d9e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+
+namespace webrtc {
+
+AudioDecoderG722::AudioDecoderG722() {
+  WebRtcG722_CreateDecoder(&dec_state_);
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+AudioDecoderG722::~AudioDecoderG722() {
+  WebRtcG722_FreeDecoder(dec_state_);
+}
+
+bool AudioDecoderG722::HasDecodePlc() const {
+  return false;
+}
+
+int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 16000);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+void AudioDecoderG722::Reset() {
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // 1/2 encoded byte per sample per channel.
+  return static_cast<int>(2 * encoded_len / Channels());
+}
+
+size_t AudioDecoderG722::Channels() const {
+  return 1;
+}
+
+AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
+  WebRtcG722_CreateDecoder(&dec_state_left_);
+  WebRtcG722_CreateDecoder(&dec_state_right_);
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
+  WebRtcG722_FreeDecoder(dec_state_left_);
+  WebRtcG722_FreeDecoder(dec_state_right_);
+}
+
+int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
+                                           size_t encoded_len,
+                                           int sample_rate_hz,
+                                           int16_t* decoded,
+                                           SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 16000);
+  int16_t temp_type = 1;  // Default is speech.
+  // De-interleave the bit-stream into two separate payloads.
+  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
+  SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
+  // Decode left and right.
+  size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
+                                         encoded_len / 2, decoded, &temp_type);
+  size_t ret = WebRtcG722_Decode(
+      dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
+      encoded_len / 2, &decoded[decoded_len], &temp_type);
+  if (ret == decoded_len) {
+    ret += decoded_len;  // Return total number of samples.
+    // Interleave output.
+    for (size_t k = ret / 2; k < ret; k++) {
+      int16_t temp = decoded[k];
+      memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
+              (ret - k - 1) * sizeof(int16_t));
+      decoded[2 * k - ret + 1] = temp;
+    }
+  }
+  *speech_type = ConvertSpeechType(temp_type);
+  delete[] encoded_deinterleaved;
+  return static_cast<int>(ret);
+}
+
+size_t AudioDecoderG722Stereo::Channels() const {
+  return 2;
+}
+
+void AudioDecoderG722Stereo::Reset() {
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+// Split the stereo packet and place left and right channel after each other
+// in the output array.
+void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
+                                               size_t encoded_len,
+                                               uint8_t* encoded_deinterleaved) {
+  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
+  // where "lx" is 4 bits representing left sample number x, and "rx" right
+  // sample. Two samples fit in one byte, represented with |...|.
+  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
+    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
+    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
+    encoded_deinterleaved[i + 1] = right_byte;
+  }
+
+  // Move one byte representing right channel each loop, and place it at the
+  // end of the bytestream vector. After looping the data is reordered to:
+  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
+  // where N is the total number of samples.
+  for (size_t i = 0; i < encoded_len / 2; i++) {
+    uint8_t right_byte = encoded_deinterleaved[i + 1];
+    memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
+            encoded_len - i - 2);
+    encoded_deinterleaved[encoded_len - 1] = right_byte;
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
new file mode 100644
index 0000000000..7cc2ea9877
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+class AudioDecoderG722 final : public AudioDecoder {
+ public:
+  AudioDecoderG722();
+  ~AudioDecoderG722() override;
+  bool HasDecodePlc() const override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  G722DecInst* dec_state_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
+};
+
+class AudioDecoderG722Stereo final : public AudioDecoder {
+ public:
+  AudioDecoderG722Stereo();
+  ~AudioDecoderG722Stereo() override;
+  void Reset() override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  size_t Channels() const override;
+
+ private:
+  // Splits the stereo-interleaved payload in |encoded| into separate payloads
+  // for left and right channels. The separated payloads are written to
+  // |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
+  // The left channel starts at offset 0, while the right channel starts at
+  // offset encoded_len / 2 into |encoded_deinterleaved|.
+  void SplitStereoPacket(const uint8_t* encoded,
+                         size_t encoded_len,
+                         uint8_t* encoded_deinterleaved);
+
+  G722DecInst* dec_state_left_;
+  G722DecInst* dec_state_right_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 68e8e084a1..d7203b9da3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -8,89 +8,103 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 
 #include <limits>
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
 namespace {
 
-const int kSampleRateHz = 16000;
+const size_t kSampleRateHz = 16000;
+
+AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderG722::Config config;
+  config.num_channels = codec_inst.channels;
+  config.frame_size_ms = codec_inst.pacsize / 16;
+  config.payload_type = codec_inst.pltype;
+  return config;
+}
 
 }  // namespace
 
-AudioEncoderG722::EncoderState::EncoderState() {
-  CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
-  CHECK_EQ(0, WebRtcG722_EncoderInit(encoder));
-}
-
-AudioEncoderG722::EncoderState::~EncoderState() {
-  CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+bool AudioEncoderG722::Config::IsOk() const {
+  return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
+      (num_channels >= 1);
 }
 
 AudioEncoderG722::AudioEncoderG722(const Config& config)
     : num_channels_(config.num_channels),
       payload_type_(config.payload_type),
-      num_10ms_frames_per_packet_(config.frame_size_ms / 10),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
       num_10ms_frames_buffered_(0),
       first_timestamp_in_buffer_(0),
       encoders_(new EncoderState[num_channels_]),
-      interleave_buffer_(new uint8_t[2 * num_channels_]) {
-  CHECK_EQ(config.frame_size_ms % 10, 0)
-      << "Frame size must be an integer multiple of 10 ms.";
-  const int samples_per_channel =
+      interleave_buffer_(2 * num_channels_) {
+  RTC_CHECK(config.IsOk());
+  const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
-    encoders_[i].encoded_buffer.reset(new uint8_t[samples_per_channel / 2]);
+    encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
+  Reset();
 }
 
-AudioEncoderG722::~AudioEncoderG722() {}
+AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
+    : AudioEncoderG722(CreateConfig(codec_inst)) {}
+
+AudioEncoderG722::~AudioEncoderG722() = default;
+
+size_t AudioEncoderG722::MaxEncodedBytes() const {
+  return SamplesPerChannel() / 2 * num_channels_;
+}
 
 int AudioEncoderG722::SampleRateHz() const {
   return kSampleRateHz;
 }
 
+size_t AudioEncoderG722::NumChannels() const {
+  return num_channels_;
+}
+
 int AudioEncoderG722::RtpTimestampRateHz() const {
   // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
   // codec.
   return kSampleRateHz / 2;
 }
 
-int AudioEncoderG722::NumChannels() const {
-  return num_channels_;
-}
-
-size_t AudioEncoderG722::MaxEncodedBytes() const {
-  return static_cast<size_t>(SamplesPerChannel() / 2 * num_channels_);
-}
-
-int AudioEncoderG722::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderG722::Max10MsFramesInAPacket() const {
+size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
+int AudioEncoderG722::GetTargetBitrate() const {
+  // 4 bits/sample, 16000 samples/s/channel.
+  return static_cast<int>(64000 * NumChannels());
+}
+
 AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
+  RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
 
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Deinterleave samples and save them in each channel's buffer.
-  const int start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
-  for (int i = 0; i < kSampleRateHz / 100; ++i)
-    for (int j = 0; j < num_channels_; ++j)
+  const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
+  for (size_t i = 0; i < kSampleRateHz / 100; ++i)
+    for (size_t j = 0; j < num_channels_; ++j)
       encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
 
   // If we don't yet have enough samples for a packet, we're done for now.
@@ -99,29 +113,28 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   }
 
   // Encode each channel separately.
-  CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
-  const int samples_per_channel = SamplesPerChannel();
-  for (int i = 0; i < num_channels_; ++i) {
-    const int encoded = WebRtcG722_Encode(
+  const size_t samples_per_channel = SamplesPerChannel();
+  for (size_t i = 0; i < num_channels_; ++i) {
+    const size_t encoded = WebRtcG722_Encode(
         encoders_[i].encoder, encoders_[i].speech_buffer.get(),
-        samples_per_channel, encoders_[i].encoded_buffer.get());
-    CHECK_GE(encoded, 0);
-    CHECK_EQ(encoded, samples_per_channel / 2);
+        samples_per_channel, encoders_[i].encoded_buffer.data());
+    RTC_CHECK_EQ(encoded, samples_per_channel / 2);
   }
 
   // Interleave the encoded bytes of the different channels. Each separate
   // channel and the interleaved stream encodes two samples per byte, most
   // significant half first.
-  for (int i = 0; i < samples_per_channel / 2; ++i) {
-    for (int j = 0; j < num_channels_; ++j) {
-      uint8_t two_samples = encoders_[j].encoded_buffer[i];
-      interleave_buffer_[j] = two_samples >> 4;
-      interleave_buffer_[num_channels_ + j] = two_samples & 0xf;
+  for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+    for (size_t j = 0; j < num_channels_; ++j) {
+      uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+      interleave_buffer_.data()[j] = two_samples >> 4;
+      interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
     }
-    for (int j = 0; j < num_channels_; ++j)
-      encoded[i * num_channels_ + j] =
-          interleave_buffer_[2 * j] << 4 | interleave_buffer_[2 * j + 1];
+    for (size_t j = 0; j < num_channels_; ++j)
+      encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
+                                       interleave_buffer_.data()[2 * j + 1];
   }
   EncodedInfo info;
   info.encoded_bytes = samples_per_channel / 2 * num_channels_;
@@ -130,7 +143,21 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   return info;
 }
 
-int AudioEncoderG722::SamplesPerChannel() const {
+void AudioEncoderG722::Reset() {
+  num_10ms_frames_buffered_ = 0;
+  for (size_t i = 0; i < num_channels_; ++i)
+    RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
+}
+
+AudioEncoderG722::EncoderState::EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
+}
+
+AudioEncoderG722::EncoderState::~EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+}
+
+size_t AudioEncoderG722::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
similarity index 54%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index b1be6b952e..07d767e778 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -8,61 +8,66 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
-class AudioEncoderG722 : public AudioEncoder {
+struct CodecInst;
+
+class AudioEncoderG722 final : public AudioEncoder {
  public:
   struct Config {
-    Config() : payload_type(9), frame_size_ms(20), num_channels(1) {}
+    bool IsOk() const;
 
-    int payload_type;
-    int frame_size_ms;
-    int num_channels;
+    int payload_type = 9;
+    int frame_size_ms = 20;
+    size_t num_channels = 1;
   };
 
   explicit AudioEncoderG722(const Config& config);
+  explicit AudioEncoderG722(const CodecInst& codec_inst);
   ~AudioEncoderG722() override;
 
-  int SampleRateHz() const override;
-  int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
-
- protected:
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
+  void Reset() override;
 
  private:
   // The encoder state for one channel.
   struct EncoderState {
     G722EncInst* encoder;
     rtc::scoped_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
-    rtc::scoped_ptr<uint8_t[]> encoded_buffer;  // Already encoded.
+    rtc::Buffer encoded_buffer;                 // Already encoded.
     EncoderState();
     ~EncoderState();
   };
 
-  int SamplesPerChannel() const;
+  size_t SamplesPerChannel() const;
 
-  const int num_channels_;
+  const size_t num_channels_;
   const int payload_type_;
-  const int num_10ms_frames_per_packet_;
-  int num_10ms_frames_buffered_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   const rtc::scoped_ptr<EncoderState[]> encoders_;
-  const rtc::scoped_ptr<uint8_t[]> interleave_buffer_;
+  rtc::Buffer interleave_buffer_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722.gypi
index 38dac31231..756fabe345 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722.gypi
@@ -8,29 +8,21 @@
 {
   'targets': [
     {
-      'target_name': 'G722',
+      'target_name': 'g722',
       'type': 'static_library',
       'dependencies': [
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
+        'audio_decoder_g722.cc',
+        'audio_decoder_g722.h',
         'audio_encoder_g722.cc',
-        'include/audio_encoder_g722.h',
-        'include/g722_interface.h',
+        'audio_encoder_g722.h',
         'g722_interface.c',
-        'g722_encode.c',
+        'g722_interface.h',
         'g722_decode.c',
         'g722_enc_dec.h',
+        'g722_encode.c',
       ],
     },
   ], # targets
@@ -38,10 +30,10 @@
     ['include_tests==1', {
       'targets': [
         {
-          'target_name': 'G722Test',
+          'target_name': 'g722_test',
           'type': 'executable',
           'dependencies': [
-            'G722',
+            'g722',
           ],
           'sources': [
             'test/testG722.cc',
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_decode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
index ee0eb89618..952a7d037f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
@@ -157,11 +157,7 @@ static void block4(G722DecoderState *s, int band, int d)
 G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
                                           int rate,
                                           int options) {
-    if (s == NULL)
-    {
-        if ((s = (G722DecoderState *) malloc(sizeof(*s))) == NULL)
-            return NULL;
-    }
+    s = s ? s : malloc(sizeof(*s));
     memset(s, 0, sizeof(*s));
     if (rate == 48000)
         s->bits_per_sample = 6;
@@ -188,8 +184,8 @@ int WebRtc_g722_decode_release(G722DecoderState *s)
 }
 /*- End of function --------------------------------------------------------*/
 
-int WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
-                       const uint8_t g722_data[], int len)
+size_t WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
+                          const uint8_t g722_data[], size_t len)
 {
     static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
     static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
@@ -258,9 +254,9 @@ int WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
     int wd2;
     int wd3;
     int code;
-    int outlen;
+    size_t outlen;
     int i;
-    int j;
+    size_t j;
 
     outlen = 0;
     rhigh = 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
index 5cd1b2d30f..7db4895fa5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
@@ -139,19 +139,19 @@ G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
                                           int rate,
                                           int options);
 int WebRtc_g722_encode_release(G722EncoderState *s);
-int WebRtc_g722_encode(G722EncoderState *s,
-                       uint8_t g722_data[],
-                       const int16_t amp[],
-                       int len);
+size_t WebRtc_g722_encode(G722EncoderState *s,
+                          uint8_t g722_data[],
+                          const int16_t amp[],
+                          size_t len);
 
 G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
                                           int rate,
                                           int options);
 int WebRtc_g722_decode_release(G722DecoderState *s);
-int WebRtc_g722_decode(G722DecoderState *s,
-                       int16_t amp[],
-                       const uint8_t g722_data[],
-                       int len);
+size_t WebRtc_g722_decode(G722DecoderState *s,
+                          int16_t amp[],
+                          const uint8_t g722_data[],
+                          size_t len);
 
 #ifdef __cplusplus
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_encode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
index bed2d218b1..01ec127ca1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
@@ -202,8 +202,8 @@ int16_t limitValues (int16_t rl)
 }
 #endif
 
-int WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
-                       const int16_t amp[], int len)
+size_t WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
+                          const int16_t amp[], size_t len)
 {
     static const int q6[32] =
     {
@@ -275,11 +275,11 @@ int WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
     int eh;
     int mih;
     int i;
-    int j;
+    size_t j;
     /* Low and high band PCM from the QMF */
     int xlow;
     int xhigh;
-    int g722_bytes;
+    size_t g722_bytes;
     /* Even and odd tap accumulators */
     int sumeven;
     int sumodd;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
index d06c588d0d..4244d5c809 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -39,21 +39,21 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
     }
 }
 
-int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
 {
     // Free encoder memory
     return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
 }
 
-int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
-                          const int16_t* speechIn,
-                          int16_t len,
-                          uint8_t* encoded)
+size_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded)
 {
     unsigned char *codechar = (unsigned char*) encoded;
     // Encode the input speech vector
-    return WebRtc_g722_encode((G722EncoderState*) G722enc_inst,
-                       codechar, speechIn, len);
+    return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
+                              speechIn, len);
 }
 
 int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
@@ -66,35 +66,28 @@ int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
     }
 }
 
-int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
-{
-    // Create and/or reset the G.722 decoder
-    // Bitrate 64 kbps and wideband mode (2)
-    G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
-        (G722DecoderState*) G722dec_inst, 64000, 2);
-    if (G722dec_inst == NULL) {
-        return -1;
-    } else {
-        return 0;
-    }
+void WebRtcG722_DecoderInit(G722DecInst* inst) {
+  // Create and/or reset the G.722 decoder
+  // Bitrate 64 kbps and wideband mode (2)
+  WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
 }
 
-int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
 {
     // Free encoder memory
     return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
 }
 
-int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
-                          const uint8_t *encoded,
-                          int16_t len,
-                          int16_t *decoded,
-                          int16_t *speechType)
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                         const uint8_t *encoded,
+                         size_t len,
+                         int16_t *decoded,
+                         int16_t *speechType)
 {
     // Decode the G.722 encoder stream
     *speechType=G722_WEBRTC_SPEECH;
-    return WebRtc_g722_decode((G722DecoderState*) G722dec_inst,
-                              decoded, encoded, len);
+    return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
+                              encoded, len);
 }
 
 int16_t WebRtcG722_Version(char *versionStr, short len)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
similarity index 76%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
index 7fe11a7eb3..b411ef0e8e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -73,7 +73,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
  * Return value               :  0 - Ok
  *                              -1 - Error
  */
-int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
 
 
 
@@ -91,13 +91,13 @@ int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
  * Output:
  *        - encoded           : The encoded data vector
  *
- * Return value              : Length (in bytes) of coded data
+ * Return value               : Length (in bytes) of coded data
  */
 
-int16_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
-                          const int16_t* speechIn,
-                          int16_t len,
-                          uint8_t* encoded);
+size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
 
 
 /****************************************************************************
@@ -113,22 +113,16 @@ int16_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
  */
 int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
 
-
 /****************************************************************************
  * WebRtcG722_DecoderInit(...)
  *
- * This function initializes a G729 instance
+ * This function initializes a G722 instance
  *
  * Input:
- *     - G729_decinst_t    : G729 instance, i.e. the user that should receive
- *                           be initialized
- *
- * Return value            :  0 - Ok
- *                           -1 - Error
+ *     - inst      : G722 instance
  */
 
-int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
-
+void WebRtcG722_DecoderInit(G722DecInst* inst);
 
 /****************************************************************************
  * WebRtcG722_FreeDecoder(...)
@@ -142,7 +136,7 @@ int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
  *                              -1 - Error
  */
 
-int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
 
 
 /****************************************************************************
@@ -162,15 +156,14 @@ int16_t WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
  *      - speechType        : 1 normal, 2 CNG (Since G722 does not have its own
  *                            DTX/CNG scheme it should always return 1)
  *
- * Return value             : >0 - Samples in decoded vector
- *                            -1 - Error
+ * Return value             : Samples in decoded vector
  */
 
-int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
-                          const uint8_t* encoded,
-                          int16_t len,
-                          int16_t *decoded,
-                          int16_t *speechType);
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t *decoded,
+                         int16_t *speechType);
 
 /****************************************************************************
  * WebRtcG722_Version(...)
@@ -186,4 +179,4 @@ int16_t WebRtcG722_Version(char *versionStr, short len);
 #endif
 
 
-#endif /* MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_ */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
index 6d0c4322e6..c55a2eb357 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -18,7 +18,7 @@
 #include "webrtc/typedefs.h"
 
 /* include API */
-#include "g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 /* Runtime statistics */
 #include <time.h>
@@ -29,18 +29,13 @@ typedef struct WebRtcG722EncInst    G722EncInst;
 typedef struct WebRtcG722DecInst    G722DecInst;
 
 /* function for reading audio data from PCM file */
-int readframe(int16_t *data, FILE *inp, int length)
+bool readframe(int16_t *data, FILE *inp, size_t length)
 {
-    short k, rlen, status = 0;
-
-    rlen = (short)fread(data, sizeof(int16_t), length, inp);
-    if (rlen < length) {
-        for (k = rlen; k < length; k++)
-            data[k] = 0;
-        status = 1;
-    }
-
-    return status;
+    size_t rlen = fread(data, sizeof(int16_t), length, inp);
+    if (rlen >= length)
+      return false;
+    memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+    return true;
 }
 
 int main(int argc, char* argv[])
@@ -48,18 +43,18 @@ int main(int argc, char* argv[])
     char inname[60], outbit[40], outname[40];
     FILE *inp, *outbitp, *outp;
 
-    int framecnt, endfile;
-    int16_t framelength = 160;
+    int framecnt;
+    bool endfile;
+    size_t framelength = 160;
     G722EncInst *G722enc_inst;
     G722DecInst *G722dec_inst;
-    int err;
 
     /* Runtime statistics */
     double starttime;
     double runtime = 0;
     double length_file;
 
-    int16_t stream_len = 0;
+    size_t stream_len = 0;
     int16_t shortdata[960];
     int16_t decoded[960];
     uint8_t streamdata[80 * 6];
@@ -82,7 +77,12 @@ int main(int argc, char* argv[])
     }
 
     /* Get frame length */
-    framelength = atoi(argv[1]);
+    int framelength_int = atoi(argv[1]);
+    if (framelength_int < 0) {
+        printf("  G.722: Invalid framelength %d.\n", framelength_int);
+        exit(1);
+    }
+    framelength = static_cast<size_t>(framelength_int);
 
     /* Get Input and Output files */
     sscanf(argv[2], "%s", inname);
@@ -112,8 +112,8 @@ int main(int argc, char* argv[])
 
     /* Initialize encoder and decoder */
     framecnt = 0;
-    endfile = 0;
-    while (endfile == 0) {
+    endfile = false;
+    while (!endfile) {
         framecnt++;
 
         /* Read speech block */
@@ -124,26 +124,21 @@ int main(int argc, char* argv[])
 
         /* G.722 encoding + decoding */
         stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
-        err = WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
-                                speechType);
+        WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
+                          speechType);
 
         /* Stop clock after call to encoder and decoder */
         runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);
 
-        if (stream_len < 0 || err < 0) {
-            /* exit if returned with error */
-            printf("Error in encoder/decoder\n");
-        } else {
-          /* Write coded bits to file */
-          if (fwrite(streamdata, sizeof(short), stream_len/2,
-                     outbitp) != static_cast<size_t>(stream_len/2)) {
-            return -1;
-          }
-          /* Write coded speech to file */
-          if (fwrite(decoded, sizeof(short), framelength,
-                     outp) != static_cast<size_t>(framelength)) {
-            return -1;
-          }
+        /* Write coded bits to file */
+        if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
+            stream_len / 2) {
+          return -1;
+        }
+        /* Write coded speech to file */
+        if (fwrite(decoded, sizeof(short), framelength, outp) !=
+            framelength) {
+          return -1;
         }
     }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
index 75fc970dde..263749ad2a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -36,7 +36,7 @@ void WebRtcIlbcfix_AbsQuant(
     int16_t *weightDenum   /* (i) denominator of synthesis filter */
                             ) {
   int16_t *syntOut;
-  int16_t quantLen[2];
+  size_t quantLen[2];
 
   /* Stack based */
   int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
index 1a18a1d27f..4b76453446 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
@@ -21,9 +21,9 @@
 #include "sort_sq.h"
 
 void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
-                                int16_t *weightDenumIN, int16_t *quantLenIN,
+                                int16_t *weightDenumIN, size_t *quantLenIN,
                                 int16_t *idxVecIN ) {
-  int n, k1, k2;
+  size_t k1, k2;
   int16_t index;
   int32_t toQW32;
   int32_t toQ32;
@@ -33,11 +33,9 @@ void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
   int16_t *syntOut   = syntOutIN;
   int16_t *in_weighted  = in_weightedIN;
   int16_t *weightDenum  = weightDenumIN;
-  int16_t *quantLen  = quantLenIN;
+  size_t *quantLen  = quantLenIN;
   int16_t *idxVec   = idxVecIN;
 
-  n=0;
-
   for(k1=0;k1<2;k1++) {
     for(k2=0;k2<quantLen[k1];k2++){
 
@@ -81,7 +79,6 @@ void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
 
       *syntOut     = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32);
 
-      n++;
       syntOut++; in_weighted++;
     }
     /* Update perceptual weighting filter at subframe border */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
index 50c6ffeb59..c8bf67575f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
@@ -27,7 +27,7 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
-                                int16_t *weightDenumIN, int16_t *quantLenIN,
+                                int16_t *weightDenumIN, size_t *quantLenIN,
                                 int16_t *idxVecIN);
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
new file mode 100644
index 0000000000..9ae0e1a95e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
+
+namespace webrtc {
+
+AudioDecoderIlbc::AudioDecoderIlbc() {
+  WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+AudioDecoderIlbc::~AudioDecoderIlbc() {
+  WebRtcIlbcfix_DecoderFree(dec_state_);
+}
+
+bool AudioDecoderIlbc::HasDecodePlc() const {
+  return true;
+}
+
+int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
+}
+
+void AudioDecoderIlbc::Reset() {
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+size_t AudioDecoderIlbc::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
new file mode 100644
index 0000000000..e890635da0
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+namespace webrtc {
+
+class AudioDecoderIlbc final : public AudioDecoder {
+ public:
+  AudioDecoderIlbc();
+  ~AudioDecoderIlbc() override;
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  IlbcDecoderInstance* dec_state_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index fad531f6bb..ddd6dde31c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -8,12 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 
-#include <cstring>
+#include <algorithm>
 #include <limits>
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 namespace webrtc {
 
@@ -21,63 +22,88 @@ namespace {
 
 const int kSampleRateHz = 8000;
 
+AudioEncoderIlbc::Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderIlbc::Config config;
+  config.frame_size_ms = codec_inst.pacsize / 8;
+  config.payload_type = codec_inst.pltype;
+  return config;
+}
+
 }  // namespace
 
-AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
-    : payload_type_(config.payload_type),
-      num_10ms_frames_per_packet_(config.frame_size_ms / 10),
-      num_10ms_frames_buffered_(0) {
-  CHECK(config.frame_size_ms == 20 || config.frame_size_ms == 30 ||
-        config.frame_size_ms == 40 || config.frame_size_ms == 60)
-      << "Frame size must be 20, 30, 40, or 60 ms.";
-  DCHECK_LE(kSampleRateHz / 100 * num_10ms_frames_per_packet_,
-            kMaxSamplesPerPacket);
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
-  const int encoder_frame_size_ms = config.frame_size_ms > 30
-                                        ? config.frame_size_ms / 2
-                                        : config.frame_size_ms;
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+// static
+const size_t AudioEncoderIlbc::kMaxSamplesPerPacket;
+
+bool AudioEncoderIlbc::Config::IsOk() const {
+  return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
+          frame_size_ms == 60) &&
+      static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <=
+          kMaxSamplesPerPacket;
 }
 
+AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
+    : config_(config),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      encoder_(nullptr) {
+  Reset();
+}
+
+AudioEncoderIlbc::AudioEncoderIlbc(const CodecInst& codec_inst)
+    : AudioEncoderIlbc(CreateConfig(codec_inst)) {}
+
 AudioEncoderIlbc::~AudioEncoderIlbc() {
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
-}
-
-int AudioEncoderIlbc::SampleRateHz() const {
-  return kSampleRateHz;
-}
-
-int AudioEncoderIlbc::NumChannels() const {
-  return 1;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
 size_t AudioEncoderIlbc::MaxEncodedBytes() const {
   return RequiredOutputSizeBytes();
 }
 
-int AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
+int AudioEncoderIlbc::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderIlbc::NumChannels() const {
+  return 1;
+}
+
+size_t AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderIlbc::Max10MsFramesInAPacket() const {
+size_t AudioEncoderIlbc::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
+int AudioEncoderIlbc::GetTargetBitrate() const {
+  switch (num_10ms_frames_per_packet_) {
+    case 2: case 4:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 3: case 6:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      FATAL();
+  }
+}
+
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
+  RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Buffer input.
-  std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
-              audio,
-              kSampleRateHz / 100 * sizeof(audio[0]));
+  RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
 
   // If we don't yet have enough buffered input for a whole packet, we're done
   // for now.
@@ -86,22 +112,34 @@ AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
   }
 
   // Encode buffered input.
-  DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
   const int output_len = WebRtcIlbcfix_Encode(
       encoder_,
       input_buffer_,
       kSampleRateHz / 100 * num_10ms_frames_per_packet_,
       encoded);
-  CHECK_GE(output_len, 0);
+  RTC_CHECK_GE(output_len, 0);
   EncodedInfo info;
-  info.encoded_bytes = output_len;
-  DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
+  info.encoded_bytes = static_cast<size_t>(output_len);
+  RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
   info.encoded_timestamp = first_timestamp_in_buffer_;
-  info.payload_type = payload_type_;
+  info.payload_type = config_.payload_type;
   return info;
 }
 
+void AudioEncoderIlbc::Reset() {
+  if (encoder_)
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK(config_.IsOk());
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms = config_.frame_size_ms > 30
+                                        ? config_.frame_size_ms / 2
+                                        : config_.frame_size_ms;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
 size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
similarity index 55%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 91d17b4f85..102a274642 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -8,52 +8,56 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 namespace webrtc {
 
-class AudioEncoderIlbc : public AudioEncoder {
+struct CodecInst;
+
+class AudioEncoderIlbc final : public AudioEncoder {
  public:
   struct Config {
-    Config() : payload_type(102), frame_size_ms(30) {}
+    bool IsOk() const;
 
-    int payload_type;
-    int frame_size_ms;  // Valid values are 20, 30, 40, and 60 ms.
+    int payload_type = 102;
+    int frame_size_ms = 30;  // Valid values are 20, 30, 40, and 60 ms.
     // Note that frame size 40 ms produces encodings with two 20 ms frames in
     // them, and frame size 60 ms consists of two 30 ms frames.
   };
 
   explicit AudioEncoderIlbc(const Config& config);
+  explicit AudioEncoderIlbc(const CodecInst& codec_inst);
   ~AudioEncoderIlbc() override;
 
-  int SampleRateHz() const override;
-  int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
-
- protected:
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
+  void Reset() override;
 
  private:
   size_t RequiredOutputSizeBytes() const;
 
-  static const int kMaxSamplesPerPacket = 480;
-  const int payload_type_;
-  const int num_10ms_frames_per_packet_;
-  int num_10ms_frames_buffered_;
+  static const size_t kMaxSamplesPerPacket = 480;
+  const Config config_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   int16_t input_buffer_[kMaxSamplesPerPacket];
   IlbcEncoderInstance* encoder_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbc);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
index d8f8c93a88..1a3735fc3d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -28,14 +28,14 @@ void WebRtcIlbcfix_AugmentedCbCorr(
     int32_t *crossDot,  /* (o) The cross correlation between
                                  the target and the Augmented
                                  vector */
-    int16_t low,    /* (i) Lag to start from (typically
+    size_t low,    /* (i) Lag to start from (typically
                              20) */
-    int16_t high,   /* (i) Lag to end at (typically 39) */
-    int16_t scale)   /* (i) Scale factor to use for
+    size_t high,   /* (i) Lag to end at (typically 39) */
+    int scale)   /* (i) Scale factor to use for
                               the crossDot */
 {
-  int lagcount;
-  int16_t ilow;
+  size_t lagcount;
+  size_t ilow;
   int16_t *targetPtr;
   int32_t *crossDotPtr;
   int16_t *iSPtr=interpSamples;
@@ -46,7 +46,7 @@ void WebRtcIlbcfix_AugmentedCbCorr(
   crossDotPtr=crossDot;
   for (lagcount=low; lagcount<=high; lagcount++) {
 
-    ilow = (int16_t) (lagcount-4);
+    ilow = lagcount - 4;
 
     /* Compute dot product for the first (lagcount-4) samples */
     (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
index 533d0a49ec..c5c408880e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -33,10 +33,9 @@ void WebRtcIlbcfix_AugmentedCbCorr(
     int32_t *crossDot,  /* (o) The cross correlation between
                                            the target and the Augmented
                                            vector */
-    int16_t low,    /* (i) Lag to start from (typically
+    size_t low,    /* (i) Lag to start from (typically
                                                    20) */
-    int16_t high,   /* (i) Lag to end at (typically 39 */
-    int16_t scale);   /* (i) Scale factor to use for
-                                                   the crossDot */
+    size_t high,   /* (i) Lag to end at (typically 39 */
+    int scale);   /* (i) Scale factor to use for the crossDot */
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
index 9d11b83acc..cacf3ace28 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
@@ -29,10 +29,10 @@ void WebRtcIlbcfix_CbConstruct(
     int16_t *index,   /* (i) Codebook indices */
     int16_t *gain_index,  /* (i) Gain quantization indices */
     int16_t *mem,   /* (i) Buffer for codevector construction */
-    int16_t lMem,   /* (i) Length of buffer */
-    int16_t veclen   /* (i) Length of vector */
+    size_t lMem,   /* (i) Length of buffer */
+    size_t veclen   /* (i) Length of vector */
                                ){
-  int j;
+  size_t j;
   int16_t gain[CB_NSTAGES];
   /* Stack based */
   int16_t cbvec0[SUBL];
@@ -50,9 +50,9 @@ void WebRtcIlbcfix_CbConstruct(
   /* codebook vector construction and construction of total vector */
 
   /* Stack based */
-  WebRtcIlbcfix_GetCbVec(cbvec0, mem, index[0], lMem, veclen);
-  WebRtcIlbcfix_GetCbVec(cbvec1, mem, index[1], lMem, veclen);
-  WebRtcIlbcfix_GetCbVec(cbvec2, mem, index[2], lMem, veclen);
+  WebRtcIlbcfix_GetCbVec(cbvec0, mem, (size_t)index[0], lMem, veclen);
+  WebRtcIlbcfix_GetCbVec(cbvec1, mem, (size_t)index[1], lMem, veclen);
+  WebRtcIlbcfix_GetCbVec(cbvec2, mem, (size_t)index[2], lMem, veclen);
 
   gainPtr = &gain[0];
   for (j=0;j<veclen;j++) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
index 2e9080f401..b676ef97ad 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -30,8 +30,8 @@ void WebRtcIlbcfix_CbConstruct(
     int16_t *index,   /* (i) Codebook indices */
     int16_t *gain_index,  /* (i) Gain quantization indices */
     int16_t *mem,   /* (i) Buffer for codevector construction */
-    int16_t lMem,   /* (i) Length of buffer */
-    int16_t veclen   /* (i) Length of vector */
+    size_t lMem,   /* (i) Length of buffer */
+    size_t veclen   /* (i) Length of vector */
                                );
 
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
index f8a09334f2..6ad2f8eb7a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -27,15 +27,15 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CbMemEnergy(
-    int16_t range,
+    size_t range,
     int16_t *CB,   /* (i) The CB memory (1:st section) */
     int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
-    int16_t lMem,   /* (i) Length of the CB memory */
-    int16_t lTarget,   /* (i) Length of the target vector */
+    size_t lMem,   /* (i) Length of the CB memory */
+    size_t lTarget,   /* (i) Length of the target vector */
     int16_t *energyW16,  /* (o) Energy in the CB vectors */
     int16_t *energyShifts, /* (o) Shift value of the energy */
-    int16_t scale,   /* (i) The scaling of all energy values */
-    int16_t base_size  /* (i) Index to where the energy values should be stored */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
                                ) {
   int16_t *ppi, *ppo, *pp;
   int32_t energy, tmp32;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
index 1b50c0b4e2..6da2f43c0b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -20,15 +20,15 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
 
 void WebRtcIlbcfix_CbMemEnergy(
-    int16_t range,
+    size_t range,
     int16_t *CB,   /* (i) The CB memory (1:st section) */
     int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
-    int16_t lMem,   /* (i) Length of the CB memory */
-    int16_t lTarget,   /* (i) Length of the target vector */
+    size_t lMem,   /* (i) Length of the CB memory */
+    size_t lTarget,   /* (i) Length of the target vector */
     int16_t *energyW16,  /* (o) Energy in the CB vectors */
     int16_t *energyShifts, /* (o) Shift value of the energy */
-    int16_t scale,   /* (i) The scaling of all energy values */
-    int16_t base_size  /* (i) Index to where the energy values should be stored */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
                                );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
index 7e6daf921c..acd6b9c118 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -22,14 +22,15 @@
 void WebRtcIlbcfix_CbMemEnergyAugmentation(
     int16_t *interpSamples, /* (i) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t scale,   /* (i) The scaling of all energy values */
-    int16_t base_size,  /* (i) Index to where the energy values should be stored */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size,  /* (i) Index to where energy values should be stored */
     int16_t *energyW16,  /* (o) Energy in the CB vectors */
     int16_t *energyShifts /* (o) Shift value of the energy */
                                            ){
   int32_t energy, tmp32;
   int16_t *ppe, *pp, *interpSamplesPtr;
-  int16_t *CBmemPtr, lagcount;
+  int16_t *CBmemPtr;
+  size_t lagcount;
   int16_t *enPtr=&energyW16[base_size-20];
   int16_t *enShPtr=&energyShifts[base_size-20];
   int32_t nrjRecursive;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
index 6c181bd0e1..594ba5f01b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -22,8 +22,8 @@
 void WebRtcIlbcfix_CbMemEnergyAugmentation(
     int16_t *interpSamples, /* (i) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t scale,   /* (i) The scaling of all energy values */
-    int16_t base_size,  /* (i) Index to where the energy values should be stored */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size,  /* (i) Index to where energy values should be stored */
     int16_t *energyW16,  /* (o) Energy in the CB vectors */
     int16_t *energyShifts /* (o) Shift value of the energy */
                                            );
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
index e5d14246f2..f2415febc0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -23,16 +23,17 @@
  * sample and the last sample respectively */
 void WebRtcIlbcfix_CbMemEnergyCalc(
     int32_t energy,   /* (i) input start energy */
-    int16_t range,   /* (i) number of iterations */
+    size_t range,   /* (i) number of iterations */
     int16_t *ppi,   /* (i) input pointer 1 */
     int16_t *ppo,   /* (i) input pointer 2 */
     int16_t *energyW16,  /* (o) Energy in the CB vectors */
     int16_t *energyShifts, /* (o) Shift value of the energy */
-    int16_t scale,   /* (i) The scaling of all energy values */
-    int16_t base_size  /* (i) Index to where the energy values should be stored */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
                                    )
 {
-  int16_t j,shft;
+  size_t j;
+  int16_t shft;
   int32_t tmp;
   int16_t *eSh_ptr;
   int16_t *eW16_ptr;
@@ -41,7 +42,7 @@ void WebRtcIlbcfix_CbMemEnergyCalc(
   eSh_ptr  = &energyShifts[1+base_size];
   eW16_ptr = &energyW16[1+base_size];
 
-  for(j=0;j<range-1;j++) {
+  for (j = 0; j + 1 < range; j++) {
 
     /* Calculate next energy by a +/-
        operation on the edge samples */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
index c7e1e54178..2991869dd9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -21,13 +21,13 @@
 
 void WebRtcIlbcfix_CbMemEnergyCalc(
     int32_t energy,   /* (i) input start energy */
-    int16_t range,   /* (i) number of iterations */
+    size_t range,   /* (i) number of iterations */
     int16_t *ppi,   /* (i) input pointer 1 */
     int16_t *ppo,   /* (i) input pointer 2 */
     int16_t *energyW16,  /* (o) Energy in the CB vectors */
     int16_t *energyShifts, /* (o) Shift value of the energy */
-    int16_t scale,   /* (i) The scaling of all energy values */
-    int16_t base_size  /* (i) Index to where the energy values should be stored */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
                                    );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
index 4c6196bdf9..be94951449 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -40,27 +40,31 @@ void WebRtcIlbcfix_CbSearch(
     int16_t *gain_index, /* (o) Gain quantization indices */
     int16_t *intarget, /* (i) Target vector for encoding */
     int16_t *decResidual,/* (i) Decoded residual for codebook construction */
-    int16_t lMem,  /* (i) Length of buffer */
-    int16_t lTarget,  /* (i) Length of vector */
+    size_t lMem,  /* (i) Length of buffer */
+    size_t lTarget,  /* (i) Length of vector */
     int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
-    int16_t block  /* (i) the subblock number */
+    size_t block  /* (i) the subblock number */
                             ) {
-  int16_t i, j, stage, range;
-  int16_t *pp, scale, tmp;
+  size_t i, range;
+  int16_t ii, j, stage;
+  int16_t *pp;
+  int16_t tmp;
+  int scale;
   int16_t bits, temp1, temp2;
-  int16_t base_size;
+  size_t base_size;
   int32_t codedEner, targetEner;
   int16_t gains[CB_NSTAGES+1];
   int16_t *cb_vecPtr;
-  int16_t indexOffset, sInd, eInd;
+  size_t indexOffset, sInd, eInd;
   int32_t CritMax=0;
   int16_t shTotMax=WEBRTC_SPL_WORD16_MIN;
-  int16_t bestIndex=0;
+  size_t bestIndex=0;
   int16_t bestGain=0;
-  int16_t indexNew, CritNewSh;
+  size_t indexNew;
+  int16_t CritNewSh;
   int32_t CritNew;
   int32_t *cDotPtr;
-  int16_t noOfZeros;
+  size_t noOfZeros;
   int16_t *gainPtr;
   int32_t t32, tmpW32;
   int16_t *WebRtcIlbcfix_kGainSq5_ptr;
@@ -108,8 +112,8 @@ void WebRtcIlbcfix_CbSearch(
 
   /* Find the highest absolute value to calculate proper
      vector scale factor (so that it uses 12 bits) */
-  temp1 = WebRtcSpl_MaxAbsValueW16(buf, (int16_t)lMem);
-  temp2 = WebRtcSpl_MaxAbsValueW16(target, (int16_t)lTarget);
+  temp1 = WebRtcSpl_MaxAbsValueW16(buf, lMem);
+  temp2 = WebRtcSpl_MaxAbsValueW16(target, lTarget);
 
   if ((temp1>0)&&(temp2>0)) {
     temp1 = WEBRTC_SPL_MAX(temp1, temp2);
@@ -146,8 +150,9 @@ void WebRtcIlbcfix_CbSearch(
                                           scale, 20, energyW16, energyShifts);
 
     /* Compute the CB vectors' energies for the second cb section (filtered cb) */
-    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors,
-                                          scale, (int16_t)(base_size+20), energyW16, energyShifts);
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors, scale,
+                                          base_size + 20, energyW16,
+                                          energyShifts);
 
     /* Compute the CB vectors' energies and store them in the vector
      * energyW16. Also the corresponding shift values are stored. The
@@ -221,16 +226,13 @@ void WebRtcIlbcfix_CbSearch(
 
     /* Update the global best index and the corresponding gain */
     WebRtcIlbcfix_CbUpdateBestIndex(
-        CritNew, CritNewSh, (int16_t)(indexNew+indexOffset), cDot[indexNew+indexOffset],
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew+indexOffset],
         inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
         &CritMax, &shTotMax, &bestIndex, &bestGain);
 
-    sInd=bestIndex-(int16_t)(CB_RESRANGE>>1);
+    sInd = ((CB_RESRANGE >> 1) > bestIndex) ?
+        0 : (bestIndex - (CB_RESRANGE >> 1));
     eInd=sInd+CB_RESRANGE;
-    if (sInd<0) {
-      eInd-=sInd;
-      sInd=0;
-    }
     if (eInd>=range) {
       eInd=range-1;
       sInd=eInd-CB_RESRANGE;
@@ -241,24 +243,28 @@ void WebRtcIlbcfix_CbSearch(
     if (lTarget==SUBL) {
       i=sInd;
       if (sInd<20) {
-        WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors+lMem,
-                                      interpSamplesFilt, cDot,
-                                      (int16_t)(sInd+20), (int16_t)(WEBRTC_SPL_MIN(39, (eInd+20))), scale);
+        WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors + lMem,
+                                      interpSamplesFilt, cDot, sInd + 20,
+                                      WEBRTC_SPL_MIN(39, (eInd + 20)), scale);
         i=20;
+        cDotPtr = &cDot[20 - sInd];
+      } else {
+        cDotPtr = cDot;
       }
 
-      cDotPtr=&cDot[WEBRTC_SPL_MAX(0,(20-sInd))];
       cb_vecPtr = cbvectors+lMem-20-i;
 
       /* Calculate the cross correlations (main part of the filtered CB) */
-      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (int16_t)(eInd-i+1), scale, -1);
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - i + 1, scale, -1);
 
     } else {
       cDotPtr = cDot;
       cb_vecPtr = cbvectors+lMem-lTarget-sInd;
 
       /* Calculate the cross correlations (main part of the filtered CB) */
-      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (int16_t)(eInd-sInd+1), scale, -1);
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - sInd + 1, scale, -1);
 
     }
 
@@ -267,17 +273,17 @@ void WebRtcIlbcfix_CbSearch(
 
     /* Search for best index in this part of the vector */
     WebRtcIlbcfix_CbSearchCore(
-        cDot, (int16_t)(eInd-sInd+1), stage, inverseEnergy+indexOffset,
+        cDot, eInd-sInd+1, stage, inverseEnergy+indexOffset,
         inverseEnergyShifts+indexOffset, Crit,
         &indexNew, &CritNew, &CritNewSh);
 
     /* Update the global best index and the corresponding gain */
     WebRtcIlbcfix_CbUpdateBestIndex(
-        CritNew, CritNewSh, (int16_t)(indexNew+indexOffset), cDot[indexNew],
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew],
         inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
         &CritMax, &shTotMax, &bestIndex, &bestGain);
 
-    index[stage] = bestIndex;
+    index[stage] = (int16_t)bestIndex;
 
 
     bestGain = WebRtcIlbcfix_GainQuant(bestGain,
@@ -290,7 +296,7 @@ void WebRtcIlbcfix_CbSearch(
 
     if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
 
-      if(index[stage]<base_size) {
+      if((size_t)index[stage]<base_size) {
         pp=buf+lMem-lTarget-index[stage];
       } else {
         pp=cbvectors+lMem-lTarget-
@@ -299,16 +305,16 @@ void WebRtcIlbcfix_CbSearch(
 
     } else {
 
-      if (index[stage]<base_size) {
+      if ((size_t)index[stage]<base_size) {
         if (index[stage]>=20) {
           /* Adjust index and extract vector */
           index[stage]-=20;
           pp=buf+lMem-lTarget-index[stage];
         } else {
           /* Adjust index and extract vector */
-          index[stage]+=(base_size-20);
+          index[stage]+=(int16_t)(base_size-20);
 
-          WebRtcIlbcfix_CreateAugmentedVec((int16_t)(index[stage]-base_size+40),
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-base_size+40,
                                            buf+lMem, aug_vec);
           pp = aug_vec;
 
@@ -322,8 +328,8 @@ void WebRtcIlbcfix_CbSearch(
               index[stage]+base_size;
         } else {
           /* Adjust index and extract vector */
-          index[stage]+=(base_size-20);
-          WebRtcIlbcfix_CreateAugmentedVec((int16_t)(index[stage]-2*base_size+40),
+          index[stage]+=(int16_t)(base_size-20);
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-2*base_size+40,
                                            cbvectors+lMem, aug_vec);
           pp = aug_vec;
         }
@@ -333,7 +339,8 @@ void WebRtcIlbcfix_CbSearch(
     /* Subtract the best codebook vector, according
        to measure, from the target vector */
 
-    WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain), (int32_t)8192, (int16_t)14, (int)lTarget);
+    WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain),
+                                      (int32_t)8192, (int16_t)14, lTarget);
 
     /* record quantized gain */
     gains[stage+1] = bestGain;
@@ -373,7 +380,7 @@ void WebRtcIlbcfix_CbSearch(
   WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[j];
 
   /* targetEner and codedEner are in Q(-2*scale) */
-  for (i=gain_index[0];i<32;i++) {
+  for (ii=gain_index[0];ii<32;ii++) {
 
     /* Change the index if
        (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND
@@ -384,8 +391,8 @@ void WebRtcIlbcfix_CbSearch(
     t32 = t32 - targetEner;
     if (t32 < 0) {
       if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
-        j=i;
-        WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[i];
+        j=ii;
+        WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[ii];
       }
     }
     gainPtr++;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
index 2fe236f4c5..ed1580c09d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -26,10 +26,10 @@ void WebRtcIlbcfix_CbSearch(
     int16_t *gain_index, /* (o) Gain quantization indices */
     int16_t *intarget, /* (i) Target vector for encoding */
     int16_t *decResidual,/* (i) Decoded residual for codebook construction */
-    int16_t lMem,  /* (i) Length of buffer */
-    int16_t lTarget,  /* (i) Length of vector */
+    size_t lMem,  /* (i) Length of buffer */
+    size_t lTarget,  /* (i) Length of vector */
     int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
-    int16_t block  /* (i) the subblock number */
+    size_t block  /* (i) the subblock number */
                             );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
index 3deb08a75c..fafa39f69b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
@@ -21,13 +21,13 @@
 
 void WebRtcIlbcfix_CbSearchCore(
     int32_t *cDot,    /* (i) Cross Correlation */
-    int16_t range,    /* (i) Search range */
+    size_t range,    /* (i) Search range */
     int16_t stage,    /* (i) Stage of this search */
     int16_t *inverseEnergy,  /* (i) Inversed energy */
     int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
                                            with the offset 2*16-29 */
     int32_t *Crit,    /* (o) The criteria */
-    int16_t *bestIndex,   /* (o) Index that corresponds to
+    size_t *bestIndex,   /* (o) Index that corresponds to
                                                    maximum criteria (in this
                                                    vector) */
     int32_t *bestCrit,   /* (o) Value of critera for the
@@ -37,7 +37,7 @@ void WebRtcIlbcfix_CbSearchCore(
 {
   int32_t maxW32, tmp32;
   int16_t max, sh, tmp16;
-  int i;
+  size_t i;
   int32_t *cDotPtr;
   int16_t cDotSqW16;
   int16_t *inverseEnergyPtr;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
index e4f2e92028..9648cf29d3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -23,13 +23,13 @@
 
 void WebRtcIlbcfix_CbSearchCore(
     int32_t *cDot,    /* (i) Cross Correlation */
-    int16_t range,    /* (i) Search range */
+    size_t range,    /* (i) Search range */
     int16_t stage,    /* (i) Stage of this search */
     int16_t *inverseEnergy,  /* (i) Inversed energy */
     int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
                                           with the offset 2*16-29 */
     int32_t *Crit,    /* (o) The criteria */
-    int16_t *bestIndex,   /* (o) Index that corresponds to
+    size_t *bestIndex,   /* (o) Index that corresponds to
                                    maximum criteria (in this
                                    vector) */
     int32_t *bestCrit,   /* (o) Value of critera for the
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
index 6fdec27aba..fc27ea9f6c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
@@ -23,13 +23,13 @@
 void WebRtcIlbcfix_CbUpdateBestIndex(
     int32_t CritNew,    /* (i) New Potentially best Criteria */
     int16_t CritNewSh,   /* (i) Shift value of above Criteria */
-    int16_t IndexNew,   /* (i) Index of new Criteria */
+    size_t IndexNew,   /* (i) Index of new Criteria */
     int32_t cDotNew,    /* (i) Cross dot of new index */
     int16_t invEnergyNew,  /* (i) Inversed energy new index */
     int16_t energyShiftNew,  /* (i) Energy shifts of new index */
     int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
     int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
-    int16_t *bestIndex,   /* (i/o) Index that corresponds to
+    size_t *bestIndex,   /* (i/o) Index that corresponds to
                                                    maximum criteria */
     int16_t *bestGain)   /* (i/o) Gain in Q14 that corresponds
                                                    to maximum criteria */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
index e8519d4118..a20fa38b2e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -24,13 +24,13 @@
 void WebRtcIlbcfix_CbUpdateBestIndex(
     int32_t CritNew,    /* (i) New Potentially best Criteria */
     int16_t CritNewSh,   /* (i) Shift value of above Criteria */
-    int16_t IndexNew,   /* (i) Index of new Criteria */
+    size_t IndexNew,   /* (i) Index of new Criteria */
     int32_t cDotNew,    /* (i) Cross dot of new index */
     int16_t invEnergyNew,  /* (i) Inversed energy new index */
     int16_t energyShiftNew,  /* (i) Energy shifts of new index */
     int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
     int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
-    int16_t *bestIndex,   /* (i/o) Index that corresponds to
+    size_t *bestIndex,   /* (i/o) Index that corresponds to
                                    maximum criteria */
     int16_t *bestGain);   /* (i/o) Gain in Q14 that corresponds
                                    to maximum criteria */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
index a53e8a77f1..7653cb0c25 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
@@ -27,9 +27,9 @@ void WebRtcIlbcfix_CompCorr(
     int32_t *corr, /* (o) cross correlation */
     int32_t *ener, /* (o) energy */
     int16_t *buffer, /* (i) signal buffer */
-    int16_t lag,  /* (i) pitch lag */
-    int16_t bLen, /* (i) length of buffer */
-    int16_t sRange, /* (i) correlation search length */
+    size_t lag,  /* (i) pitch lag */
+    size_t bLen, /* (i) length of buffer */
+    size_t sRange, /* (i) correlation search length */
     int16_t scale /* (i) number of rightshifts to use */
                             ){
   int16_t *w16ptr;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
index 4ff80aac46..ab78c72b3e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -30,9 +30,9 @@ void WebRtcIlbcfix_CompCorr(
     int32_t *corr, /* (o) cross correlation */
     int32_t *ener, /* (o) energy */
     int16_t *buffer, /* (i) signal buffer */
-    int16_t lag,  /* (i) pitch lag */
-    int16_t bLen, /* (i) length of buffer */
-    int16_t sRange, /* (i) correlation search length */
+    size_t lag,  /* (i) pitch lag */
+    size_t bLen, /* (i) length of buffer */
+    size_t sRange, /* (i) correlation search length */
     int16_t scale /* (i) number of rightshifts to use */
                             );
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.c
index 1d384b750e..9e341942e6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.c
@@ -593,10 +593,10 @@ const int16_t WebRtcIlbcfix_kAlpha[4]={
 
 /* Ranges for search and filters at different subframes */
 
-const int16_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
+const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
   {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
 
-const int16_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
+const size_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
 
 /* Gain Quantization for the codebook gains of the 3 stages */
 
@@ -647,7 +647,7 @@ const int16_t WebRtcIlbcfix_kEnhWt[3] = {
   4800, 16384, 27968 /* Q16 */
 };
 
-const int16_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
+const size_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
   160, 480, 800, 1120, 1440, 1760, 2080, 2400  /* Q(-2) */
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.h
index ff6370e14c..7c4ad4d928 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/constants.h
@@ -61,8 +61,8 @@ extern const int16_t WebRtcIlbcfix_kFrgQuantMod[];
 
 /* Ranges for search and filters at different subframes */
 
-extern const int16_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
-extern const int16_t WebRtcIlbcfix_kFilterRange[];
+extern const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
+extern const size_t WebRtcIlbcfix_kFilterRange[];
 
 /* gain quantization tables */
 
@@ -81,7 +81,7 @@ extern const int16_t WebRtcIlbcfix_kAlpha[];
 
 extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1];
 extern const int16_t WebRtcIlbcfix_kEnhWt[];
-extern const int16_t WebRtcIlbcfix_kEnhPlocs[];
+extern const size_t WebRtcIlbcfix_kEnhPlocs[];
 
 /* PLC tables */
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
index 965cbe0d39..8ae28ac3b9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -25,12 +25,12 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CreateAugmentedVec(
-    int16_t index,  /* (i) Index for the augmented vector to be created */
+    size_t index,  /* (i) Index for the augmented vector to be created */
     int16_t *buffer,  /* (i) Pointer to the end of the codebook memory that
                                            is used for creation of the augmented codebook */
     int16_t *cbVec  /* (o) The construced codebook vector */
                                       ) {
-  int16_t ilow;
+  size_t ilow;
   int16_t *ppo, *ppi;
   int16_t cbVecTmp[4];
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
index e3c3c7b4bc..430dfe9b9d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -27,7 +27,7 @@
  *----------------------------------------------------------------*/
 
 void WebRtcIlbcfix_CreateAugmentedVec(
-    int16_t index,  /* (i) Index for the augmented vector to be created */
+    size_t index,  /* (i) Index for the augmented vector to be created */
     int16_t *buffer,  /* (i) Pointer to the end of the codebook memory that
                                            is used for creation of the augmented codebook */
     int16_t *cbVec  /* (o) The construced codebook vector */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode.c
index 3a2e5a2344..4c8497a568 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -44,7 +44,7 @@ void WebRtcIlbcfix_DecodeImpl(
     int16_t mode      /* (i) 0: bad packet, PLC,
                                                                    1: normal */
                            ) {
-  int i;
+  size_t i;
   int16_t order_plus_one;
 
   int16_t last_bit;
@@ -103,9 +103,10 @@ void WebRtcIlbcfix_DecodeImpl(
       WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst, decresidual, syntdenum);
 
       /* preparing the plc for a future loss! */
-      WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 0,
-                              decresidual, syntdenum + (LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
-                              (int16_t)(iLBCdec_inst->last_lag), iLBCdec_inst);
+      WebRtcIlbcfix_DoThePlc(
+          PLCresidual, PLClpc, 0, decresidual,
+          syntdenum + (LPC_FILTERORDER + 1) * (iLBCdec_inst->nsub - 1),
+          iLBCdec_inst->last_lag, iLBCdec_inst);
 
       /* Use the output from doThePLC */
       WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
@@ -120,8 +121,8 @@ void WebRtcIlbcfix_DecodeImpl(
 
     /* packet loss conceal */
 
-    WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 1,
-                            decresidual, syntdenum, (int16_t)(iLBCdec_inst->last_lag), iLBCdec_inst);
+    WebRtcIlbcfix_DoThePlc(PLCresidual, PLClpc, 1, decresidual, syntdenum,
+                           iLBCdec_inst->last_lag, iLBCdec_inst);
 
     WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
 
@@ -187,18 +188,18 @@ void WebRtcIlbcfix_DecodeImpl(
     WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
 
   } else { /* Enhancer not activated */
-    int16_t lag;
+    size_t lag;
 
     /* Find last lag (since the enhancer is not called to give this info) */
     lag = 20;
     if (iLBCdec_inst->mode==20) {
-      lag = (int16_t)WebRtcIlbcfix_XcorrCoef(
+      lag = WebRtcIlbcfix_XcorrCoef(
           &decresidual[iLBCdec_inst->blockl-60],
           &decresidual[iLBCdec_inst->blockl-60-lag],
           60,
           80, lag, -1);
     } else {
-      lag = (int16_t)WebRtcIlbcfix_XcorrCoef(
+      lag = WebRtcIlbcfix_XcorrCoef(
           &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
           &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
           ENH_BLOCKL,
@@ -206,7 +207,7 @@ void WebRtcIlbcfix_DecodeImpl(
     }
 
     /* Store lag (it is needed if next packet is lost) */
-    (*iLBCdec_inst).last_lag = (int)lag;
+    (*iLBCdec_inst).last_lag = lag;
 
     /* copy data and run synthesis filter */
     WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
index 169218aa85..b8a067e0f3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -41,8 +41,8 @@ void WebRtcIlbcfix_DecodeResidual(
     int16_t *syntdenum   /* (i) the decoded synthesis filter
                                   coefficients */
                                   ) {
-  int16_t meml_gotten, Nfor, Nback, diff, start_pos;
-  int16_t subcount, subframe;
+  size_t meml_gotten, diff, start_pos;
+  size_t subcount, subframe;
   int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
   int16_t *memVec = iLBCdec_inst->prevResidual;  /* Memory for codebook and filter state (reuse memory in state) */
   int16_t *mem = &memVec[CB_HALFFILTERLEN];   /* Memory for codebook */
@@ -66,7 +66,7 @@ void WebRtcIlbcfix_DecodeResidual(
 
     /* setup memory */
 
-    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCdec_inst->state_short_len));
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCdec_inst->state_short_len);
     WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
                           iLBCdec_inst->state_short_len);
 
@@ -76,8 +76,7 @@ void WebRtcIlbcfix_DecodeResidual(
         &decresidual[start_pos+iLBCdec_inst->state_short_len],
         iLBC_encbits->cb_index, iLBC_encbits->gain_index,
         mem+CB_MEML-ST_MEM_L_TBL,
-        ST_MEM_L_TBL, (int16_t)diff
-                              );
+        ST_MEM_L_TBL, diff);
 
   }
   else {/* put adaptive part in the beginning */
@@ -87,7 +86,7 @@ void WebRtcIlbcfix_DecodeResidual(
     meml_gotten = iLBCdec_inst->state_short_len;
     WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
                                   decresidual+start_pos, meml_gotten);
-    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
 
     /* construct decoded vector */
 
@@ -110,9 +109,7 @@ void WebRtcIlbcfix_DecodeResidual(
 
   /* forward prediction of subframes */
 
-  Nfor = iLBCdec_inst->nsub-iLBC_encbits->startIdx-1;
-
-  if( Nfor > 0 ) {
+  if (iLBCdec_inst->nsub > iLBC_encbits->startIdx + 1) {
 
     /* setup memory */
     WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
@@ -121,6 +118,7 @@ void WebRtcIlbcfix_DecodeResidual(
 
     /* loop over subframes to encode */
 
+    size_t Nfor = iLBCdec_inst->nsub - iLBC_encbits->startIdx - 1;
     for (subframe=0; subframe<Nfor; subframe++) {
 
       /* construct decoded vector */
@@ -143,9 +141,7 @@ void WebRtcIlbcfix_DecodeResidual(
 
   /* backward prediction of subframes */
 
-  Nback = iLBC_encbits->startIdx-1;
-
-  if( Nback > 0 ){
+  if (iLBC_encbits->startIdx > 1) {
 
     /* setup memory */
 
@@ -156,10 +152,11 @@ void WebRtcIlbcfix_DecodeResidual(
 
     WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
                                   decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
-    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
 
     /* loop over subframes to decode */
 
+    size_t Nback = iLBC_encbits->startIdx - 1;
     for (subframe=0; subframe<Nback; subframe++) {
 
       /* construct decoded vector */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
index fad8170652..06ab2e7190 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -34,7 +34,8 @@ void WebRtcIlbcfix_DecoderInterpolateLsp(
     IlbcDecoder *iLBCdec_inst
     /* (i) the decoder state structure */
                                           ){
-  int  i, pos, lp_length;
+  size_t i;
+  int pos, lp_length;
   int16_t  lp[LPC_FILTERORDER + 1], *lsfdeq2;
 
   lsfdeq2 = lsfdeq + length;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/defines.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/defines.h
index 2d37e52650..2faaea15fc 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/defines.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/defines.h
@@ -121,11 +121,11 @@ typedef struct iLBC_bits_t_ {
   int16_t lsf[LSF_NSPLIT*LPC_N_MAX];
   int16_t cb_index[CB_NSTAGES*(NASUB_MAX+1)];  /* First CB_NSTAGES values contains extra CB index */
   int16_t gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */
-  int16_t idxForMax;
+  size_t idxForMax;
   int16_t state_first;
   int16_t idxVec[STATE_SHORT_LEN_30MS];
   int16_t firstbits;
-  int16_t startIdx;
+  size_t startIdx;
 } iLBC_bits;
 
 /* type definition encoder instance */
@@ -135,12 +135,12 @@ typedef struct IlbcEncoder_ {
   int16_t mode;
 
   /* basic parameters for different frame sizes */
-  int16_t blockl;
-  int16_t nsub;
+  size_t blockl;
+  size_t nsub;
   int16_t nasub;
-  int16_t no_of_bytes, no_of_words;
+  size_t no_of_bytes, no_of_words;
   int16_t lpc_n;
-  int16_t state_short_len;
+  size_t state_short_len;
 
   /* analysis filter state */
   int16_t anaMem[LPC_FILTERORDER];
@@ -164,7 +164,7 @@ typedef struct IlbcEncoder_ {
   int16_t Nfor_flag;
   int16_t Nback_flag;
   int16_t start_pos;
-  int16_t diff;
+  size_t diff;
 #endif
 
 } IlbcEncoder;
@@ -176,12 +176,12 @@ typedef struct IlbcDecoder_ {
   int16_t mode;
 
   /* basic parameters for different frame sizes */
-  int16_t blockl;
-  int16_t nsub;
+  size_t blockl;
+  size_t nsub;
   int16_t nasub;
-  int16_t no_of_bytes, no_of_words;
+  size_t no_of_bytes, no_of_words;
   int16_t lpc_n;
-  int16_t state_short_len;
+  size_t state_short_len;
 
   /* synthesis filter state */
   int16_t syntMem[LPC_FILTERORDER];
@@ -190,14 +190,15 @@ typedef struct IlbcDecoder_ {
   int16_t lsfdeqold[LPC_FILTERORDER];
 
   /* pitch lag estimated in enhancer and used in PLC */
-  int last_lag;
+  size_t last_lag;
 
   /* PLC state information */
   int consPLICount, prev_enh_pl;
   int16_t perSquare;
 
   int16_t prevScale, prevPLI;
-  int16_t prevLag, prevLpc[LPC_FILTERORDER+1];
+  size_t prevLag;
+  int16_t prevLpc[LPC_FILTERORDER+1];
   int16_t prevResidual[NSUB_MAX*SUBL];
   int16_t seed;
 
@@ -212,7 +213,7 @@ typedef struct IlbcDecoder_ {
   /* enhancer state information */
   int use_enhancer;
   int16_t enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
-  int16_t enh_period[ENH_NBLOCKS_TOT];
+  size_t enh_period[ENH_NBLOCKS_TOT];
 
 } IlbcDecoder;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
index 2d7207522c..f74439ede6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -33,18 +33,19 @@ void WebRtcIlbcfix_DoThePlc(
                                                            0 - no PL, 1 = PL */
     int16_t *decresidual,  /* (i) decoded residual */
     int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
-    int16_t inlag,    /* (i) pitch lag */
+    size_t inlag,    /* (i) pitch lag */
     IlbcDecoder *iLBCdec_inst
     /* (i/o) decoder instance */
                             ){
-  int16_t i, pick;
+  size_t i;
   int32_t cross, ener, cross_comp, ener_comp = 0;
   int32_t measure, maxMeasure, energy;
   int16_t max, crossSquareMax, crossSquare;
-  int16_t j, lag, tmp1, tmp2, randlag;
+  size_t j, lag, randlag;
+  int16_t tmp1, tmp2;
   int16_t shift1, shift2, shift3, shiftMax;
   int16_t scale3;
-  int16_t corrLen;
+  size_t corrLen;
   int32_t tmpW32, tmp2W32;
   int16_t use_gain;
   int16_t tot_gain;
@@ -54,7 +55,7 @@ void WebRtcIlbcfix_DoThePlc(
   int32_t nom;
   int16_t denom;
   int16_t pitchfact;
-  int16_t use_lag;
+  size_t use_lag;
   int ind;
   int16_t randvec[BLOCKL_MAX];
 
@@ -70,7 +71,8 @@ void WebRtcIlbcfix_DoThePlc(
 
       /* Maximum 60 samples are correlated, preserve as high accuracy
          as possible without getting overflow */
-      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, (int16_t)iLBCdec_inst->blockl);
+      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual,
+                                     iLBCdec_inst->blockl);
       scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
       if (scale3 < 0) {
         scale3 = 0;
@@ -85,7 +87,7 @@ void WebRtcIlbcfix_DoThePlc(
       lag = inlag - 3;
 
       /* Guard against getting outside the frame */
-      corrLen = WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
+      corrLen = (size_t)WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
 
       WebRtcIlbcfix_CompCorr( &cross, &ener,
                               iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3);
@@ -233,23 +235,20 @@ void WebRtcIlbcfix_DoThePlc(
 
       /* noise component -  52 < randlagFIX < 117 */
       iLBCdec_inst->seed = (int16_t)(iLBCdec_inst->seed * 31821 + 13849);
-      randlag = 53 + (int16_t)(iLBCdec_inst->seed & 63);
-
-      pick = i - randlag;
-
-      if (pick < 0) {
-        randvec[i] = iLBCdec_inst->prevResidual[iLBCdec_inst->blockl+pick];
+      randlag = 53 + (iLBCdec_inst->seed & 63);
+      if (randlag > i) {
+        randvec[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - randlag];
       } else {
-        randvec[i] = iLBCdec_inst->prevResidual[pick];
+        randvec[i] = iLBCdec_inst->prevResidual[i - randlag];
       }
 
       /* pitch repeatition component */
-      pick = i - use_lag;
-
-      if (pick < 0) {
-        PLCresidual[i] = iLBCdec_inst->prevResidual[iLBCdec_inst->blockl+pick];
+      if (use_lag > i) {
+        PLCresidual[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - use_lag];
       } else {
-        PLCresidual[i] = PLCresidual[pick];
+        PLCresidual[i] = PLCresidual[i - use_lag];
       }
 
       /* Attinuate total gain for each 10 ms */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
index c55b81540c..38b8fdb7c0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -33,7 +33,7 @@ void WebRtcIlbcfix_DoThePlc(
                                                            0 - no PL, 1 = PL */
     int16_t *decresidual,  /* (i) decoded residual */
     int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
-    int16_t inlag,    /* (i) pitch lag */
+    size_t inlag,    /* (i) pitch lag */
     IlbcDecoder *iLBCdec_inst
     /* (i/o) decoder instance */
                             );
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/encode.c
index 1d46eff432..812ec8d6c7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/encode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -48,11 +48,11 @@ void WebRtcIlbcfix_EncodeImpl(
     IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
                                      state */
                           ){
-  int n, meml_gotten, Nfor, Nback;
-  int16_t diff, start_pos;
-  int index;
-  int subcount, subframe;
-  int16_t start_count, end_count;
+  size_t n, meml_gotten, Nfor;
+  size_t diff, start_pos;
+  size_t index;
+  size_t subcount, subframe;
+  size_t start_count, end_count;
   int16_t *residual;
   int32_t en1, en2;
   int16_t scale, max;
@@ -86,7 +86,7 @@ void WebRtcIlbcfix_EncodeImpl(
 #ifdef SPLIT_10MS
 
   WebRtcSpl_MemSetW16 (  (int16_t *) iLBCbits_inst, 0,
-                         (int16_t) (sizeof(iLBC_bits) / sizeof(int16_t))  );
+                         sizeof(iLBC_bits) / sizeof(int16_t)  );
 
   start_pos = iLBCenc_inst->start_pos;
   diff = iLBCenc_inst->diff;
@@ -193,7 +193,7 @@ void WebRtcIlbcfix_EncodeImpl(
 
       /* setup memory */
 
-      WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCenc_inst->state_short_len));
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
       WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
                             decresidual+start_pos, iLBCenc_inst->state_short_len);
 
@@ -224,7 +224,7 @@ void WebRtcIlbcfix_EncodeImpl(
 
       meml_gotten = iLBCenc_inst->state_short_len;
       WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
-      WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCenc_inst->state_short_len));
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
 
       /* encode subframes */
       WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
@@ -317,17 +317,17 @@ void WebRtcIlbcfix_EncodeImpl(
       if (iLBCenc_inst->section == 1)
       {
         start_count = 0;
-        end_count = WEBRTC_SPL_MIN (Nfor, 2);
+        end_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
       }
       if (iLBCenc_inst->section == 2)
       {
-        start_count = WEBRTC_SPL_MIN (Nfor, 2);
+        start_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
         end_count = Nfor;
       }
     }
 #else
     start_count = 0;
-    end_count = (int16_t)Nfor;
+    end_count = Nfor;
 #endif
 
     /* loop over subframes to encode */
@@ -341,7 +341,7 @@ void WebRtcIlbcfix_EncodeImpl(
                              &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
                              mem, MEM_LF_TBL, SUBL,
                              &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
-                             (int16_t)subcount);
+                             subcount);
 
       /* construct decoded vector */
 
@@ -379,15 +379,14 @@ void WebRtcIlbcfix_EncodeImpl(
 
   /* backward prediction of subframes */
 
-  Nback = iLBCbits_inst->startIdx-1;
-
-  if( Nback > 0 ){
+  if (iLBCbits_inst->startIdx > 1) {
 
     /* create reverse order vectors
        (The decresidual does not need to be copied since it is
        contained in the same vector as the residual)
     */
 
+    size_t Nback = iLBCbits_inst->startIdx - 1;
     WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL);
 
     /* setup memory */
@@ -398,7 +397,7 @@ void WebRtcIlbcfix_EncodeImpl(
     }
 
     WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
-    WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
 
 #ifdef SPLIT_10MS
     if (iLBCenc_inst->Nback_flag > 0)
@@ -425,17 +424,17 @@ void WebRtcIlbcfix_EncodeImpl(
       if (iLBCenc_inst->section == 1)
       {
         start_count = 0;
-        end_count = WEBRTC_SPL_MAX (2 - Nfor, 0);
+        end_count = (Nfor >= 2) ? 0 : (2 - NFor);
       }
       if (iLBCenc_inst->section == 2)
       {
-        start_count = WEBRTC_SPL_MAX (2 - Nfor, 0);
+        start_count = (Nfor >= 2) ? 0 : (2 - NFor);
         end_count = Nback;
       }
     }
 #else
     start_count = 0;
-    end_count = (int16_t)Nback;
+    end_count = Nback;
 #endif
 
     /* loop over subframes to encode */
@@ -448,7 +447,7 @@ void WebRtcIlbcfix_EncodeImpl(
                              iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
                              mem, MEM_LF_TBL, SUBL,
                              &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
-                             (int16_t)subcount);
+                             subcount);
 
       /* construct decoded vector */
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
index a6b1c758f9..b2bdcfffc3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
@@ -23,12 +23,12 @@
 void WebRtcIlbcfix_EnergyInverse(
     int16_t *energy,    /* (i/o) Energy and inverse
                                                            energy (in Q29) */
-    int noOfEnergies)  /* (i)   The length of the energy
+    size_t noOfEnergies)  /* (i)   The length of the energy
                                    vector */
 {
   int32_t Nom=(int32_t)0x1FFFFFFF;
   int16_t *energyPtr;
-  int i;
+  size_t i;
 
   /* Set the minimum energy value to 16384 to avoid overflow */
   energyPtr=energy;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
index 7bb67215fc..fe25094325 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -26,7 +26,7 @@
 void WebRtcIlbcfix_EnergyInverse(
     int16_t *energy,     /* (i/o) Energy and inverse
                                                                    energy (in Q29) */
-    int noOfEnergies);   /* (i)   The length of the energy
+    size_t noOfEnergies);   /* (i)   The length of the energy
                                    vector */
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c
index 38c3de379a..521d00441c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c
@@ -29,11 +29,11 @@
 void WebRtcIlbcfix_Enhancer(
     int16_t *odata,   /* (o) smoothed block, dimension blockl */
     int16_t *idata,   /* (i) data buffer used for enhancing */
-    int16_t idatal,   /* (i) dimension idata */
-    int16_t centerStartPos, /* (i) first sample current block within idata */
-    int16_t *period,   /* (i) pitch period array (pitch bward-in time) */
-    int16_t *plocs,   /* (i) locations where period array values valid */
-    int16_t periodl   /* (i) dimension of period and plocs */
+    size_t idatal,   /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    const size_t *plocs,   /* (i) locations where period array values valid */
+    size_t periodl   /* (i) dimension of period and plocs */
                             ){
   /* Stack based */
   int16_t surround[ENH_BLOCKL];
@@ -47,5 +47,5 @@ void WebRtcIlbcfix_Enhancer(
 
   /* compute the smoothed output from said second sequence */
 
-  WebRtcIlbcfix_Smooth(odata, idata+centerStartPos, surround);
+  WebRtcIlbcfix_Smooth(odata, idata + centerStartPos, surround);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h
index 83f48b0505..ed219fb1bb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -29,11 +29,11 @@
 void WebRtcIlbcfix_Enhancer(
     int16_t *odata,   /* (o) smoothed block, dimension blockl */
     int16_t *idata,   /* (i) data buffer used for enhancing */
-    int16_t idatal,   /* (i) dimension idata */
-    int16_t centerStartPos, /* (i) first sample current block within idata */
-    int16_t *period,   /* (i) pitch period array (pitch bward-in time) */
-    int16_t *plocs,   /* (i) locations where period array values valid */
-    int16_t periodl   /* (i) dimension of period and plocs */
+    size_t idatal,   /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    const size_t *plocs,   /* (i) locations where period array values valid */
+    size_t periodl   /* (i) dimension of period and plocs */
                             );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
index fde541455c..1c0fd42383 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -30,25 +30,29 @@
  * interface for enhancer
  *---------------------------------------------------------------*/
 
-int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
+size_t WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     int16_t *out,     /* (o) enhanced signal */
     int16_t *in,      /* (i) unenhanced signal */
     IlbcDecoder *iLBCdec_inst /* (i) buffers etc */
                                         ){
-  int iblock;
-  int lag=20, tlag=20;
-  int inLen=iLBCdec_inst->blockl+120;
-  int16_t scale, scale1, plc_blockl;
-  int16_t *enh_buf, *enh_period;
-  int32_t tmp1, tmp2, max, new_blocks;
+  size_t iblock;
+  size_t lag=20, tlag=20;
+  size_t inLen=iLBCdec_inst->blockl+120;
+  int16_t scale, scale1;
+  size_t plc_blockl;
+  int16_t *enh_buf;
+  size_t *enh_period;
+  int32_t tmp1, tmp2, max;
+  size_t new_blocks;
   int16_t *enh_bufPtr1;
-  int i, k;
+  size_t i;
+  size_t k;
   int16_t EnChange;
   int16_t SqrtEnChange;
   int16_t inc;
   int16_t win;
   int16_t *tmpW16ptr;
-  int16_t startPos;
+  size_t startPos;
   int16_t *plc_pred;
   int16_t *target, *regressor;
   int16_t max16;
@@ -56,8 +60,9 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
   int32_t ener;
   int16_t enerSh;
   int16_t corrSh;
-  int16_t ind, sh;
-  int16_t start, stop;
+  size_t ind;
+  int16_t sh;
+  size_t start, stop;
   /* Stack based */
   int16_t totsh[3];
   int16_t downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
@@ -65,7 +70,7 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
   int32_t corrmax[3];
   int16_t corr16[3];
   int16_t en16[3];
-  int16_t lagmax[3];
+  size_t lagmax[3];
 
   plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
                               downsampled are non overlapping */
@@ -96,11 +101,11 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
   memmove(enh_period, &enh_period[new_blocks],
           (ENH_NBLOCKS_TOT - new_blocks) * sizeof(*enh_period));
 
-  k=WebRtcSpl_DownsampleFast(
+  WebRtcSpl_DownsampleFast(
       enh_buf+ENH_BUFL-inLen,    /* Input samples */
-      (int16_t)(inLen+ENH_BUFL_FILTEROVERHEAD),
+      inLen + ENH_BUFL_FILTEROVERHEAD,
       downsampled,
-      (int16_t)(inLen / 2),
+      inLen / 2,
       (int16_t*)WebRtcIlbcfix_kLpFiltCoefs,  /* Coefficients in Q12 */
       FILTERORDER_DS_PLUS1,    /* Length of filter (order-1) */
       FACTOR_DS,
@@ -110,19 +115,17 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
   for(iblock = 0; iblock<new_blocks; iblock++){
 
     /* references */
-    i = 60 + iblock * ENH_BLOCKL_HALF;
-    target=downsampled+i;
-    regressor=downsampled+i-10;
+    target = downsampled + 60 + iblock * ENH_BLOCKL_HALF;
+    regressor = target - 10;
 
     /* scaling */
-    max16=WebRtcSpl_MaxAbsValueW16(&regressor[-50],
-                                   (int16_t)(ENH_BLOCKL_HALF+50-1));
+    max16 = WebRtcSpl_MaxAbsValueW16(&regressor[-50], ENH_BLOCKL_HALF + 50 - 1);
     shifts = WebRtcSpl_GetSizeInBits((uint32_t)(max16 * max16)) - 25;
     shifts = WEBRTC_SPL_MAX(0, shifts);
 
     /* compute cross correlation */
-    WebRtcSpl_CrossCorrelation(corr32, target, regressor,
-                               ENH_BLOCKL_HALF, 50, (int16_t)shifts, -1);
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
+                               shifts, -1);
 
     /* Find 3 highest correlations that should be compared for the
        highest (corr*corr)/ener */
@@ -130,11 +133,9 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     for (i=0;i<2;i++) {
       lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50);
       corrmax[i] = corr32[lagmax[i]];
-      start = lagmax[i] - 2;
-      stop = lagmax[i] + 2;
-      start = WEBRTC_SPL_MAX(0,  start);
-      stop  = WEBRTC_SPL_MIN(49, stop);
-      for (k=start; k<=stop; k++) {
+      start = WEBRTC_SPL_MAX(2, lagmax[i]) - 2;
+      stop = WEBRTC_SPL_MIN(47, lagmax[i]) + 2;
+      for (k = start; k <= stop; k++) {
         corr32[k] = 0;
       }
     }
@@ -144,8 +145,8 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     /* Calculate normalized corr^2 and ener */
     for (i=0;i<3;i++) {
       corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]);
-      ener = WebRtcSpl_DotProductWithScale(&regressor[-lagmax[i]],
-                                           &regressor[-lagmax[i]],
+      ener = WebRtcSpl_DotProductWithScale(regressor - lagmax[i],
+                                           regressor - lagmax[i],
                                            ENH_BLOCKL_HALF, shifts);
       enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
       corr16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
@@ -173,7 +174,7 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     lag = lagmax[ind] + 10;
 
     /* Store the estimated lag in the non-downsampled domain */
-    enh_period[ENH_NBLOCKS_TOT - new_blocks + iblock] = (int16_t)(lag * 8);
+    enh_period[ENH_NBLOCKS_TOT - new_blocks + iblock] = lag * 8;
 
     /* Store the estimated lag for backward PLC */
     if (iLBCdec_inst->prev_enh_pl==1) {
@@ -200,15 +201,15 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     regressor=in+tlag-1;
 
     /* scaling */
-    max16=WebRtcSpl_MaxAbsValueW16(regressor, (int16_t)(plc_blockl+3-1));
+    max16 = WebRtcSpl_MaxAbsValueW16(regressor, plc_blockl + 3 - 1);
     if (max16>5000)
       shifts=2;
     else
       shifts=0;
 
     /* compute cross correlation */
-    WebRtcSpl_CrossCorrelation(corr32, target, regressor,
-                               plc_blockl, 3, (int16_t)shifts, 1);
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
+                               1);
 
     /* find lag */
     lag=WebRtcSpl_MaxIndexW32(corr32, 3);
@@ -226,7 +227,7 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
             (plc_blockl-lag));
       }
     } else {
-      int pos;
+      size_t pos;
 
       pos = plc_blockl;
 
@@ -282,8 +283,8 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
 
 
         /* Multiply first part of vector with 2*SqrtEnChange */
-        WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange,
-                              (int16_t)(plc_blockl-16), 14);
+        WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange, plc_blockl-16,
+                              14);
 
         /* Calculate increase parameter for window part (16 last samples) */
         /* (1-2*SqrtEnChange)/16 in Q15 */
@@ -338,25 +339,25 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
           enh_bufPtr1,
           synt,
           &iLBCdec_inst->old_syntdenum[
-                                       (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
-                                       LPC_FILTERORDER+1, (int16_t)lag);
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
 
       WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
                             LPC_FILTERORDER);
       WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
                              iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
-                             (int16_t)lag);
+                             lag);
       WebRtcSpl_FilterARFastQ12(
           enh_bufPtr1, synt,
           &iLBCdec_inst->old_syntdenum[
-                                       (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
-                                       LPC_FILTERORDER+1, (int16_t)lag);
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
 
       WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
                             LPC_FILTERORDER);
       WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
                              iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
-                             (int16_t)lag);
+                             lag);
     }
   }
 
@@ -367,9 +368,9 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     WebRtcIlbcfix_Enhancer(out + iblock * ENH_BLOCKL,
                            enh_buf,
                            ENH_BUFL,
-                           (int16_t)(iblock * ENH_BLOCKL + startPos),
+                           iblock * ENH_BLOCKL + startPos,
                            enh_period,
-                           (int16_t*)WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
+                           WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
   }
 
   return (lag);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
index fa58b7a67f..61efd22604 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -25,7 +25,7 @@
  * interface for enhancer
  *---------------------------------------------------------------*/
 
-int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
+size_t WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
     int16_t *out,     /* (o) enhanced signal */
     int16_t *in,      /* (i) unenhanced signal */
     IlbcDecoder *iLBCdec_inst /* (i) buffers etc */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
index aa8170cb76..04d17a67ef 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
@@ -29,8 +29,8 @@ void WebRtcIlbcfix_FilteredCbVecs(
     int16_t *cbvectors, /* (o) Codebook vector for the higher section */
     int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
                                            second CB section */
-    int lMem,  /* (i) Length of codebook memory */
-    int16_t samples    /* (i) Number of samples to filter */
+    size_t lMem,  /* (i) Length of codebook memory */
+    size_t samples    /* (i) Number of samples to filter */
                                   ) {
 
   /* Set up the memory, start with zero state */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
index 99e89a0807..d23b25c1ac 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -31,8 +31,8 @@ void WebRtcIlbcfix_FilteredCbVecs(
     int16_t *cbvectors, /* (o) Codebook vector for the higher section */
     int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
                                            second CB section */
-    int lMem,  /* (i) Length of codebook memory */
-    int16_t samples    /* (i) Number of samples to filter */
+    size_t lMem,  /* (i) Length of codebook memory */
+    size_t samples    /* (i) Number of samples to filter */
                                   );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
index d124b6b7f7..48332808e4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -23,7 +23,7 @@
  *  Classification of subframes to localize start state
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_FrameClassify(
+size_t WebRtcIlbcfix_FrameClassify(
     /* (o) Index to the max-energy sub frame */
     IlbcEncoder *iLBCenc_inst,
     /* (i/o) the encoder state structure */
@@ -35,8 +35,8 @@ int16_t WebRtcIlbcfix_FrameClassify(
   int32_t *seqEnPtr;
   int32_t maxW32;
   int16_t scale1;
-  int16_t pos;
-  int n;
+  size_t pos;
+  size_t n;
 
   /*
     Calculate the energy of each of the 80 sample blocks
@@ -62,7 +62,7 @@ int16_t WebRtcIlbcfix_FrameClassify(
   }
 
   /* Scale to maximum 20 bits in order to allow for the 11 bit window */
-  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, (int16_t)(iLBCenc_inst->nsub-1));
+  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, iLBCenc_inst->nsub - 1);
   scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
   scale1 = WEBRTC_SPL_MAX(0, scale);
 
@@ -82,7 +82,7 @@ int16_t WebRtcIlbcfix_FrameClassify(
   }
 
   /* Extract the best choise of start state */
-  pos = WebRtcSpl_MaxIndexW32(ssqEn, (int16_t)(iLBCenc_inst->nsub-1)) + 1;
+  pos = WebRtcSpl_MaxIndexW32(ssqEn, iLBCenc_inst->nsub - 1) + 1;
 
   return(pos);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
index b32e2c87d5..99f7144782 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -19,7 +19,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
 
-int16_t WebRtcIlbcfix_FrameClassify(
+size_t WebRtcIlbcfix_FrameClassify(
     /* (o) Index to the max-energy sub frame */
     IlbcEncoder *iLBCenc_inst,
     /* (i/o) the encoder state structure */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
index cf05ce3310..d7c2e75553 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -27,12 +27,12 @@
 void WebRtcIlbcfix_GetCbVec(
     int16_t *cbvec,   /* (o) Constructed codebook vector */
     int16_t *mem,   /* (i) Codebook buffer */
-    int16_t index,   /* (i) Codebook index */
-    int16_t lMem,   /* (i) Length of codebook buffer */
-    int16_t cbveclen   /* (i) Codebook vector length */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,   /* (i) Length of codebook buffer */
+    size_t cbveclen   /* (i) Codebook vector length */
                             ){
-  int16_t k, base_size;
-  int16_t lag;
+  size_t k, base_size;
+  size_t lag;
   /* Stack based */
   int16_t tempbuff2[SUBL+5];
 
@@ -58,7 +58,7 @@ void WebRtcIlbcfix_GetCbVec(
 
     /* Calculate lag */
 
-    k = (int16_t)(2 * (index - (lMem - cbveclen + 1))) + cbveclen;
+    k = (2 * (index - (lMem - cbveclen + 1))) + cbveclen;
 
     lag = k / 2;
 
@@ -70,7 +70,7 @@ void WebRtcIlbcfix_GetCbVec(
 
   else {
 
-    int16_t memIndTest;
+    size_t memIndTest;
 
     /* first non-interpolated vectors */
 
@@ -100,7 +100,7 @@ void WebRtcIlbcfix_GetCbVec(
       /* do filtering */
       WebRtcSpl_FilterMAFastQ12(
           &mem[memIndTest+7], tempbuff2, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
-          CB_FILTERLEN, (int16_t)(cbveclen+5));
+          CB_FILTERLEN, cbveclen+5);
 
       /* Calculate lag index */
       lag = (cbveclen<<1)-20+index-base_size-lMem-1;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
index 1c5ac8f16e..07f67a2aa5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -22,9 +22,9 @@
 void WebRtcIlbcfix_GetCbVec(
     int16_t *cbvec,   /* (o) Constructed codebook vector */
     int16_t *mem,   /* (i) Codebook buffer */
-    int16_t index,   /* (i) Codebook index */
-    int16_t lMem,   /* (i) Length of codebook buffer */
-    int16_t cbveclen   /* (i) Codebook vector length */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,   /* (i) Length of codebook buffer */
+    size_t cbveclen   /* (i) Codebook vector length */
                             );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
index 480ed7c6cd..a98a96cdf1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
@@ -27,71 +27,68 @@
 
 void WebRtcIlbcfix_GetSyncSeq(
     int16_t *idata,   /* (i) original data */
-    int16_t idatal,   /* (i) dimension of data */
-    int16_t centerStartPos, /* (i) where current block starts */
-    int16_t *period,   /* (i) rough-pitch-period array       (Q-2) */
-    int16_t *plocs,   /* (i) where periods of period array are taken (Q-2) */
-    int16_t periodl,   /* (i) dimension period array */
-    int16_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    size_t idatal,   /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,   /* (i) dimension period array */
+    size_t hl,    /* (i) 2*hl+1 is the number of sequences */
     int16_t *surround  /* (i/o) The contribution from this sequence
                                 summed with earlier contributions */
                               ){
-  int16_t i,centerEndPos,q;
+  size_t i, centerEndPos, q;
   /* Stack based */
-  int16_t lagBlock[2*ENH_HL+1];
-  int16_t blockStartPos[2*ENH_HL+1]; /* Defines the position to search around (Q2) */
-  int16_t plocs2[ENH_PLOCSL];
+  size_t lagBlock[2 * ENH_HL + 1];
+  size_t blockStartPos[2 * ENH_HL + 1]; /* The position to search around (Q2) */
+  size_t plocs2[ENH_PLOCSL];
 
-  centerEndPos=centerStartPos+ENH_BLOCKL-1;
+  centerEndPos = centerStartPos + ENH_BLOCKL - 1;
 
   /* present (find predicted lag from this position) */
 
   WebRtcIlbcfix_NearestNeighbor(lagBlock + hl,
                                 plocs,
-                                (int16_t)(2 * (centerStartPos + centerEndPos)),
+                                2 * (centerStartPos + centerEndPos),
                                 periodl);
 
-  blockStartPos[hl] = (int16_t)(4 * centerStartPos);
+  blockStartPos[hl] = 4 * centerStartPos;
 
   /* past (find predicted position and perform a refined
      search to find the best sequence) */
 
-  for(q=hl-1;q>=0;q--) {
-    blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
+  for (q = hl; q > 0; q--) {
+    size_t qq = q - 1;
+    size_t period_q = period[lagBlock[q]];
+    /* Stop if this sequence would be outside the buffer; that means all
+       further-past sequences would also be outside the buffer. */
+    if (blockStartPos[q] < period_q + (4 * ENH_OVERHANG))
+      break;
+    blockStartPos[qq] = blockStartPos[q] - period_q;
 
-    WebRtcIlbcfix_NearestNeighbor(
-        lagBlock + q,
-        plocs,
-        (int16_t)(blockStartPos[q] + 4 * ENH_BLOCKL_HALF -
-            period[lagBlock[q + 1]]),
-        periodl);
+    size_t value = blockStartPos[qq] + 4 * ENH_BLOCKL_HALF;
+    value = (value > period_q) ? (value - period_q) : 0;
+    WebRtcIlbcfix_NearestNeighbor(lagBlock + qq, plocs, value, periodl);
 
-    if (blockStartPos[q] - 4 * ENH_OVERHANG >= 0) {
-
-      /* Find the best possible sequence in the 4 times upsampled
-         domain around blockStartPos+q */
-      WebRtcIlbcfix_Refiner(blockStartPos+q,idata,idatal,
-                            centerStartPos,blockStartPos[q],surround,WebRtcIlbcfix_kEnhWt[q]);
-
-    } else {
-      /* Don't add anything since this sequence would
-         be outside the buffer */
-    }
+    /* Find the best possible sequence in the 4 times upsampled
+        domain around blockStartPos+q */
+    WebRtcIlbcfix_Refiner(blockStartPos + qq, idata, idatal, centerStartPos,
+                          blockStartPos[qq], surround,
+                          WebRtcIlbcfix_kEnhWt[qq]);
   }
 
   /* future (find predicted position and perform a refined
      search to find the best sequence) */
 
-  for(i=0;i<periodl;i++) {
-    plocs2[i]=(plocs[i]-period[i]);
+  for (i = 0; i < periodl; i++) {
+    plocs2[i] = plocs[i] - period[i];
   }
 
-  for (q = hl + 1; q <= (int16_t)(2 * hl); q++) {
+  for (q = hl + 1; q <= (2 * hl); q++) {
 
     WebRtcIlbcfix_NearestNeighbor(
         lagBlock + q,
         plocs2,
-        (int16_t)(blockStartPos[q - 1] + 4 * ENH_BLOCKL_HALF),
+        blockStartPos[q - 1] + 4 * ENH_BLOCKL_HALF,
         periodl);
 
     blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
@@ -100,11 +97,11 @@ void WebRtcIlbcfix_GetSyncSeq(
 
       /* Find the best possible sequence in the 4 times upsampled
          domain around blockStartPos+q */
-      WebRtcIlbcfix_Refiner(blockStartPos+q, idata, idatal,
-                            centerStartPos,blockStartPos[q],surround,WebRtcIlbcfix_kEnhWt[2*hl-q]);
+      WebRtcIlbcfix_Refiner(blockStartPos + q, idata, idatal, centerStartPos,
+                            blockStartPos[q], surround,
+                            WebRtcIlbcfix_kEnhWt[2 * hl - q]);
 
-    }
-    else {
+    } else {
       /* Don't add anything since this sequence would
          be outside the buffer */
     }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
index f9b08b750e..0e3b207753 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -27,12 +27,12 @@
 
 void WebRtcIlbcfix_GetSyncSeq(
     int16_t *idata,   /* (i) original data */
-    int16_t idatal,   /* (i) dimension of data */
-    int16_t centerStartPos, /* (i) where current block starts */
-    int16_t *period,   /* (i) rough-pitch-period array       (Q-2) */
-    int16_t *plocs,   /* (i) where periods of period array are taken (Q-2) */
-    int16_t periodl,   /* (i) dimension period array */
-    int16_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    size_t idatal,   /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,   /* (i) dimension period array */
+    size_t hl,    /* (i) 2*hl+1 is the number of sequences */
     int16_t *surround  /* (i/o) The contribution from this sequence
                                 summed with earlier contributions */
                               );
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c
index 260591ede2..5d8a86050e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c
@@ -30,9 +30,9 @@ void WebRtcIlbcfix_HpInput(
     int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                                                                    yhi[n-2] ylow[n-2] */
     int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    int16_t len)      /* (i)   Number of samples to filter */
+    size_t len)      /* (i)   Number of samples to filter */
 {
-  int i;
+  size_t i;
   int32_t tmpW32;
   int32_t tmpW32b;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h
index a30f703351..acdfa91262 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -29,6 +29,6 @@ void WebRtcIlbcfix_HpInput(
     int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                                                                    yhi[n-2] ylow[n-2] */
     int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    int16_t len);     /* (i)   Number of samples to filter */
+    size_t len);     /* (i)   Number of samples to filter */
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
index 3abb427b9a..bd101bf30c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -30,9 +30,9 @@ void WebRtcIlbcfix_HpOutput(
     int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                                                                    yhi[n-2] ylow[n-2] */
     int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    int16_t len)      /* (i)   Number of samples to filter */
+    size_t len)      /* (i)   Number of samples to filter */
 {
-  int i;
+  size_t i;
   int32_t tmpW32;
   int32_t tmpW32b;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h
index 7937ba00cd..1840b68186 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -29,6 +29,6 @@ void WebRtcIlbcfix_HpOutput(
     int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
                               yhi[n-2] ylow[n-2] */
     int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
-    int16_t len);      /* (i)   Number of samples to filter */
+    size_t len);      /* (i)   Number of samples to filter */
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
index 88ad33b5af..6cd9a723fd 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -88,12 +88,12 @@ int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
   }
 }
 
-int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
-                             const int16_t* speechIn,
-                             int16_t len,
-                             uint8_t* encoded) {
-  int16_t pos = 0;
-  int16_t encpos = 0;
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded) {
+  size_t pos = 0;
+  size_t encpos = 0;
 
   if ((len != ((IlbcEncoder*)iLBCenc_inst)->blockl) &&
 #ifdef SPLIT_10MS
@@ -118,7 +118,7 @@ int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
 #endif
       encpos += ((IlbcEncoder*)iLBCenc_inst)->no_of_words;
     }
-    return (encpos*2);
+    return (int)(encpos*2);
   }
 }
 
@@ -131,23 +131,21 @@ int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
     return(-1);
   }
 }
-int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst) {
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) {
   WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
-  return(0);
 }
-int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst) {
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) {
   WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
-  return(0);
 }
 
 
-int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
-                             const uint8_t* encoded,
-                             int16_t len,
-                             int16_t* decoded,
-                             int16_t* speechType)
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType)
 {
-  int i=0;
+  size_t i=0;
   /* Allow for automatic switching between the frame sizes
      (although you do get some discontinuity) */
   if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
@@ -191,16 +189,16 @@ int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
   }
   /* iLBC does not support VAD/CNG yet */
   *speechType=1;
-  return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
-int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                 const uint8_t* encoded,
-                                 int16_t len,
-                                 int16_t* decoded,
-                                 int16_t* speechType)
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
 {
-  int i=0;
+  size_t i=0;
   if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
       (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
       (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
@@ -219,16 +217,16 @@ int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
   }
   /* iLBC does not support VAD/CNG yet */
   *speechType=1;
-  return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
-int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                 const uint8_t* encoded,
-                                 int16_t len,
-                                 int16_t* decoded,
-                                 int16_t* speechType)
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
 {
-  int i=0;
+  size_t i=0;
   if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
       (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
       (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
@@ -247,13 +245,13 @@ int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
   }
   /* iLBC does not support VAD/CNG yet */
   *speechType=1;
-  return(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
-int16_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
-                                int16_t* decoded,
-                                int16_t noOfLostFrames) {
-  int i;
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames) {
+  size_t i;
   uint16_t dummy;
 
   for (i=0;i<noOfLostFrames;i++) {
@@ -265,9 +263,9 @@ int16_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
   return (noOfLostFrames*((IlbcDecoder*)iLBCdec_inst)->blockl);
 }
 
-int16_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
-                               int16_t* decoded,
-                               int16_t noOfLostFrames) {
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames) {
   /* Two input parameters not used, but needed for function pointers in NetEQ */
   (void)(decoded = NULL);
   (void)(noOfLostFrames = 0);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi
index 5f6fed1a24..ffb0574588 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.gypi
@@ -9,28 +9,19 @@
 {
   'targets': [
     {
-      'target_name': 'iLBC',
+      'target_name': 'ilbc',
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        'interface',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'interface',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'interface/audio_encoder_ilbc.h',
-        'interface/ilbc.h',
         'abs_quant.c',
         'abs_quant_loop.c',
+        'audio_decoder_ilbc.cc',
+        'audio_decoder_ilbc.h',
         'audio_encoder_ilbc.cc',
+        'audio_encoder_ilbc.h',
         'augmented_cb_corr.c',
         'bw_expand.c',
         'cb_construct.c',
@@ -63,6 +54,7 @@
         'hp_input.c',
         'hp_output.c',
         'ilbc.c',
+        'ilbc.h',
         'index_conv_dec.c',
         'index_conv_enc.c',
         'init_decode.c',
@@ -168,21 +160,21 @@
         'window32_w32.h',
         'xcorr_coef.h',
      ], # sources
-    }, # iLBC
+    }, # ilbc
   ], # targets
   'conditions': [
     ['include_tests==1', {
       'targets': [
         {
-          'target_name': 'iLBCtest',
+          'target_name': 'ilbc_test',
           'type': 'executable',
           'dependencies': [
-            'iLBC',
+            'ilbc',
           ],
           'sources': [
             'test/iLBC_test.c',
           ],
-        }, # iLBCtest
+        }, # ilbc_test
       ], # targets
     }], # include_tests
   ], # conditions
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
similarity index 76%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
index b7e1735e5b..c021f5be52 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
@@ -15,8 +15,10 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_INTERFACE_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_INTERFACE_ILBC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+
+#include <stddef.h>
 
 /*
  * Define the fixpoint numeric formats
@@ -51,10 +53,10 @@ extern "C" {
    * memory location
    *
    * Input:
-   *      - XXX_xxxinst       : Pointer to created instance that should be
-   *                            assigned
-   *      - ILBCXXX_inst_Addr : Pointer to the desired memory space
-   *      - size              : The size that this structure occupies (in Word16)
+   *     - XXX_xxxinst       : Pointer to created instance that should be
+   *                           assigned
+   *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
+   *     - size              : The size that this structure occupies (in Word16)
    *
    * Return value             :  0 - Ok
    *                            -1 - Error
@@ -74,10 +76,10 @@ extern "C" {
    * These functions create a instance to the specified structure
    *
    * Input:
-   *      - XXX_inst          : Pointer to created instance that should be created
+   *      - XXX_inst        : Pointer to created instance that should be created
    *
-   * Return value             :  0 - Ok
-   *                            -1 - Error
+   * Return value           :  0 - Ok
+   *                          -1 - Error
    */
 
   int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst);
@@ -135,10 +137,10 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
-                               const int16_t *speechIn,
-                               int16_t len,
-                               uint8_t* encoded);
+  int WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
+                           const int16_t *speechIn,
+                           size_t len,
+                           uint8_t* encoded);
 
   /****************************************************************************
    * WebRtcIlbcfix_DecoderInit(...)
@@ -157,8 +159,8 @@ extern "C" {
 
   int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance *iLBCdec_inst,
                                     int16_t frameLen);
-  int16_t WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance *iLBCdec_inst);
-  int16_t WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance *iLBCdec_inst);
+  void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
+  void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
 
   /****************************************************************************
    * WebRtcIlbcfix_Decode(...)
@@ -180,21 +182,21 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+  int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                           const uint8_t* encoded,
+                           size_t len,
+                           int16_t* decoded,
+                           int16_t* speechType);
+  int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
                                const uint8_t* encoded,
-                               int16_t len,
+                               size_t len,
+                               int16_t* decoded,
+                               int16_t* speechType);
+  int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                               const uint8_t* encoded,
+                               size_t len,
                                int16_t* decoded,
                                int16_t* speechType);
-  int16_t WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                   const uint8_t* encoded,
-                                   int16_t len,
-                                   int16_t* decoded,
-                                   int16_t* speechType);
-  int16_t WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
-                                   const uint8_t* encoded,
-                                   int16_t len,
-                                   int16_t* decoded,
-                                   int16_t* speechType);
 
   /****************************************************************************
    * WebRtcIlbcfix_DecodePlc(...)
@@ -210,13 +212,12 @@ extern "C" {
    * Output:
    *      - decoded           : The "decoded" vector
    *
-   * Return value             : >0 - Samples in decoded PLC vector
-   *                            -1 - Error
+   * Return value             : Samples in decoded PLC vector
    */
 
-  int16_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance *iLBCdec_inst,
-                                  int16_t *decoded,
-                                  int16_t noOfLostFrames);
+  size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance *iLBCdec_inst,
+                                 int16_t *decoded,
+                                 size_t noOfLostFrames);
 
   /****************************************************************************
    * WebRtcIlbcfix_NetEqPlc(...)
@@ -232,13 +233,12 @@ extern "C" {
    * Output:
    *      - decoded           : The "decoded" vector (nothing in this case)
    *
-   * Return value             : >0 - Samples in decoded PLC vector
-   *                            -1 - Error
+   * Return value             : Samples in decoded PLC vector
    */
 
-  int16_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance *iLBCdec_inst,
-                                 int16_t *decoded,
-                                 int16_t noOfLostFrames);
+  size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance *iLBCdec_inst,
+                                int16_t *decoded,
+                                size_t noOfLostFrames);
 
   /****************************************************************************
    * WebRtcIlbcfix_version(...)
@@ -255,4 +255,4 @@ extern "C" {
 }
 #endif
 
-#endif
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
index d903ac7e82..1f92480d9f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -23,7 +23,7 @@
  *  Initiation of decoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
     IlbcDecoder *iLBCdec_inst,  /* (i/o) Decoder instance */
     int16_t mode,  /* (i) frame size mode */
     int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
@@ -92,5 +92,5 @@ int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
 
   iLBCdec_inst->prev_enh_pl = 0;
 
-  return (iLBCdec_inst->blockl);
+  return (int)(iLBCdec_inst->blockl);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
index 4871b5c1ac..cdd2192079 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -25,7 +25,7 @@
  *  Initiation of decoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
     IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
     int16_t mode,     /* (i) frame size mode */
     int use_enhancer           /* (i) 1 to use enhancer
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
index 1a2fa08923..f559d8441f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -23,7 +23,7 @@
  *  Initiation of encoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
     IlbcEncoder *iLBCenc_inst,  /* (i/o) Encoder instance */
     int16_t mode) {  /* (i) frame size mode */
   iLBCenc_inst->mode = mode;
@@ -67,5 +67,5 @@ int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
   iLBCenc_inst->section = 0;
 #endif
 
-  return (iLBCenc_inst->no_of_bytes);
+  return (int)(iLBCenc_inst->no_of_bytes);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
index 2eea27c8ed..7154661fbd 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -25,7 +25,7 @@
  *  Initiation of encoder instance.
  *---------------------------------------------------------------*/
 
-int16_t WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
     IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */
     int16_t mode     /* (i) frame size mode */
                                          );
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
index 4957142145..376dbbb668 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
@@ -22,7 +22,7 @@
 void WebRtcIlbcfix_InterpolateSamples(
     int16_t *interpSamples, /* (o) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t lMem    /* (i) Length of the CB memory */
+    size_t lMem    /* (i) Length of the CB memory */
                                       ) {
   int16_t *ppi, *ppo, i, j, temp1, temp2;
   int16_t *tmpPtr;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
index 586c27d354..7549d2c216 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -28,7 +28,7 @@
 void WebRtcIlbcfix_InterpolateSamples(
     int16_t *interpSamples, /* (o) The interpolated samples */
     int16_t *CBmem,   /* (i) The CB memory */
-    int16_t lMem    /* (i) Length of the CB memory */
+    size_t lMem    /* (i) Length of the CB memory */
                                       );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
index 048745a3a4..bd6ff561c2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -23,13 +23,15 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_MyCorr(
-    int32_t *corr,  /* (o) correlation of seq1 and seq2 */
-    int16_t *seq1,  /* (i) first sequence */
-    int16_t dim1,  /* (i) dimension first seq1 */
-    const int16_t *seq2, /* (i) second sequence */
-    int16_t dim2   /* (i) dimension seq2 */
+    int32_t* corr,  /* (o) correlation of seq1 and seq2 */
+    const int16_t* seq1,  /* (i) first sequence */
+    size_t dim1,  /* (i) dimension first seq1 */
+    const int16_t* seq2, /* (i) second sequence */
+    size_t dim2   /* (i) dimension seq2 */
                           ){
-  int16_t max, scale, loops;
+  int16_t max;
+  size_t loops;
+  int scale;
 
   /* Calculate correlation between the two sequences. Scale the
      result of the multiplcication to maximum 26 bits in order
@@ -37,7 +39,7 @@ void WebRtcIlbcfix_MyCorr(
   max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
   scale=WebRtcSpl_GetSizeInBits(max);
 
-  scale = (int16_t)(2 * scale - 26);
+  scale = 2 * scale - 26;
   if (scale<0) {
     scale=0;
   }
@@ -45,7 +47,7 @@ void WebRtcIlbcfix_MyCorr(
   loops=dim1-dim2+1;
 
   /* Calculate the cross correlations */
-  WebRtcSpl_CrossCorrelation(corr, (int16_t*)seq2, seq1, dim2, loops, scale, 1);
+  WebRtcSpl_CrossCorrelation(corr, seq2, seq1, dim2, loops, scale, 1);
 
   return;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h
index ee66998313..214946410e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -26,11 +26,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_MyCorr(
-    int32_t *corr,  /* (o) correlation of seq1 and seq2 */
-    int16_t *seq1,  /* (i) first sequence */
-    int16_t dim1,  /* (i) dimension first seq1 */
-    const int16_t *seq2, /* (i) second sequence */
-    int16_t dim2   /* (i) dimension seq2 */
+    int32_t* corr,  /* (o) correlation of seq1 and seq2 */
+    const int16_t* seq1,  /* (i) first sequence */
+    size_t dim1,  /* (i) dimension first seq1 */
+    const int16_t* seq2, /* (i) second sequence */
+    size_t dim2   /* (i) dimension seq2 */
                           );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
index 6329908851..2b58abc4f9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -18,29 +18,18 @@
 
 #include "defines.h"
 
-/*----------------------------------------------------------------*
- * Find index in array such that the array element with said
- * index is the element of said array closest to "value"
- * according to the squared-error criterion
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_NearestNeighbor(
-    int16_t *index, /* (o) index of array element closest to value */
-    int16_t *array, /* (i) data array (Q2) */
-    int16_t value, /* (i) value (Q2) */
-    int16_t arlength /* (i) dimension of data array (==8) */
-                                   ){
-  int i;
-  int16_t diff;
-  /* Stack based */
-  int32_t crit[8];
-
-  /* Calculate square distance */
-  for(i=0;i<arlength;i++){
-    diff=array[i]-value;
-    crit[i] = diff * diff;
+void WebRtcIlbcfix_NearestNeighbor(size_t* index,
+                                   const size_t* array,
+                                   size_t value,
+                                   size_t arlength) {
+  size_t i;
+  size_t min_diff = (size_t)-1;
+  for (i = 0; i < arlength; i++) {
+    const size_t diff =
+        (array[i] < value) ? (value - array[i]) : (array[i] - value);
+    if (diff < min_diff) {
+      *index = i;
+      min_diff = diff;
+    }
   }
-
-  /* Find the minimum square distance */
-  *index=WebRtcSpl_MinIndexW32(crit, (int16_t)arlength);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
index 0c03470084..7d7fb6fccc 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -24,14 +24,13 @@
 /*----------------------------------------------------------------*
  * Find index in array such that the array element with said
  * index is the element of said array closest to "value"
- * according to the squared-error criterion
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_NearestNeighbor(
-    int16_t *index, /* (o) index of array element closest to value */
-    int16_t *array, /* (i) data array (Q2) */
-    int16_t value, /* (i) value (Q2) */
-    int16_t arlength /* (i) dimension of data array (==8) */
+    size_t* index, /* (o) index of array element closest to value */
+    const size_t* array, /* (i) data array (Q2) */
+    size_t value, /* (i) value (Q2) */
+    size_t arlength /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
                                    );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
index ca99b3a912..3c1265e5d5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
@@ -30,17 +30,17 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Refiner(
-    int16_t *updStartPos, /* (o) updated start point (Q-2) */
+    size_t *updStartPos, /* (o) updated start point (Q-2) */
     int16_t *idata,   /* (i) original data buffer */
-    int16_t idatal,   /* (i) dimension of idata */
-    int16_t centerStartPos, /* (i) beginning center segment */
-    int16_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    size_t idatal,   /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
     int16_t *surround,  /* (i/o) The contribution from this sequence
                                            summed with earlier contributions */
     int16_t gain    /* (i) Gain to use for this sequence */
                            ){
-  int16_t estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
-  int16_t tloc,tloc2,i,st,en,fraction;
+  size_t estSegPosRounded, searchSegStartPos, searchSegEndPos, corrdim;
+  size_t tloc, tloc2, i;
 
   int32_t maxtemp, scalefact;
   int16_t *filtStatePtr, *polyPtr;
@@ -55,96 +55,86 @@ void WebRtcIlbcfix_Refiner(
 
   estSegPosRounded = (estSegPos - 2) >> 2;
 
-  searchSegStartPos=estSegPosRounded-ENH_SLOP;
+  searchSegStartPos =
+      (estSegPosRounded < ENH_SLOP) ? 0 : (estSegPosRounded - ENH_SLOP);
 
-  if (searchSegStartPos<0) {
-    searchSegStartPos=0;
+  searchSegEndPos = estSegPosRounded + ENH_SLOP;
+  if ((searchSegEndPos + ENH_BLOCKL) >= idatal) {
+    searchSegEndPos = idatal - ENH_BLOCKL - 1;
   }
-  searchSegEndPos=estSegPosRounded+ENH_SLOP;
 
-  if(searchSegEndPos+ENH_BLOCKL >= idatal) {
-    searchSegEndPos=idatal-ENH_BLOCKL-1;
-  }
-  corrdim=searchSegEndPos-searchSegStartPos+1;
+  corrdim = searchSegEndPos + 1 - searchSegStartPos;
 
   /* compute upsampled correlation and find
      location of max */
 
-  WebRtcIlbcfix_MyCorr(corrVecTemp,idata+searchSegStartPos,
-                       (int16_t)(corrdim+ENH_BLOCKL-1),idata+centerStartPos,ENH_BLOCKL);
+  WebRtcIlbcfix_MyCorr(corrVecTemp, idata + searchSegStartPos,
+                       corrdim + ENH_BLOCKL - 1, idata + centerStartPos,
+                       ENH_BLOCKL);
 
   /* Calculate the rescaling factor for the correlation in order to
      put the correlation in a int16_t vector instead */
-  maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, (int16_t)corrdim);
+  maxtemp = WebRtcSpl_MaxAbsValueW32(corrVecTemp, corrdim);
 
-  scalefact=WebRtcSpl_GetSizeInBits(maxtemp)-15;
+  scalefact = WebRtcSpl_GetSizeInBits(maxtemp) - 15;
 
-  if (scalefact>0) {
-    for (i=0;i<corrdim;i++) {
+  if (scalefact > 0) {
+    for (i = 0; i < corrdim; i++) {
       corrVec[i] = (int16_t)(corrVecTemp[i] >> scalefact);
     }
   } else {
-    for (i=0;i<corrdim;i++) {
-      corrVec[i]=(int16_t)corrVecTemp[i];
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)corrVecTemp[i];
     }
   }
   /* In order to guarantee that all values are initialized */
-  for (i=corrdim;i<ENH_CORRDIM;i++) {
-    corrVec[i]=0;
+  for (i = corrdim; i < ENH_CORRDIM; i++) {
+    corrVec[i] = 0;
   }
 
   /* Upsample the correlation */
-  WebRtcIlbcfix_EnhUpsample(corrVecUps,corrVec);
+  WebRtcIlbcfix_EnhUpsample(corrVecUps, corrVec);
 
   /* Find maximum */
-  tloc=WebRtcSpl_MaxIndexW32(corrVecUps, (int16_t) (ENH_UPS0*corrdim));
+  tloc = WebRtcSpl_MaxIndexW32(corrVecUps, ENH_UPS0 * corrdim);
 
   /* make vector can be upsampled without ever running outside
      bounds */
-  *updStartPos = (int16_t)(searchSegStartPos * 4) + tloc + 4;
+  *updStartPos = searchSegStartPos * 4 + tloc + 4;
 
   tloc2 = (tloc + 3) >> 2;
 
-  st=searchSegStartPos+tloc2-ENH_FL0;
-
   /* initialize the vector to be filtered, stuff with zeros
      when data is outside idata buffer */
-  if(st<0){
-    WebRtcSpl_MemSetW16(vect, 0, (int16_t)(-st));
-    WEBRTC_SPL_MEMCPY_W16(&vect[-st], idata, (ENH_VECTL+st));
-  }
-  else{
-    en=st+ENH_VECTL;
-
-    if(en>idatal){
-      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st],
-                            (ENH_VECTL-(en-idatal)));
-      WebRtcSpl_MemSetW16(&vect[ENH_VECTL-(en-idatal)], 0,
-                          (int16_t)(en-idatal));
-    }
-    else {
+  if (ENH_FL0 > (searchSegStartPos + tloc2)) {
+    const size_t st = ENH_FL0 - searchSegStartPos - tloc2;
+    WebRtcSpl_MemSetW16(vect, 0, st);
+    WEBRTC_SPL_MEMCPY_W16(&vect[st], idata, ENH_VECTL - st);
+  } else {
+    const size_t st = searchSegStartPos + tloc2 - ENH_FL0;
+    if ((st + ENH_VECTL) > idatal) {
+      const size_t en = st + ENH_VECTL - idatal;
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL - en);
+      WebRtcSpl_MemSetW16(&vect[ENH_VECTL - en], 0, en);
+    } else {
       WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
     }
   }
-  /* Calculate which of the 4 fractions to use */
-  fraction = (int16_t)(tloc2 * ENH_UPS0) - tloc;
 
   /* compute the segment (this is actually a convolution) */
-
   filtStatePtr = filt + 6;
-  polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[fraction];
-  for (i=0;i<7;i++) {
+  polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[tloc2 * ENH_UPS0 - tloc];
+  for (i = 0; i < 7; i++) {
     *filtStatePtr-- = *polyPtr++;
   }
 
-  WebRtcSpl_FilterMAFastQ12(
-      &vect[6], vect, filt,
-      ENH_FLO_MULT2_PLUS1, ENH_BLOCKL);
+  WebRtcSpl_FilterMAFastQ12(&vect[6], vect, filt, ENH_FLO_MULT2_PLUS1,
+                            ENH_BLOCKL);
 
-  /* Add the contribution from this vector (scaled with gain) to the total surround vector */
-  WebRtcSpl_AddAffineVectorToVector(
-      surround, vect, gain,
-      (int32_t)32768, 16, ENH_BLOCKL);
+  /* Add the contribution from this vector (scaled with gain) to the total
+     surround vector */
+  WebRtcSpl_AddAffineVectorToVector(surround, vect, gain, 32768, 16,
+                                    ENH_BLOCKL);
 
   return;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.h
index d13996152d..f8a2abc2d6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/refiner.h
@@ -30,11 +30,11 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_Refiner(
-    int16_t *updStartPos, /* (o) updated start point (Q-2) */
+    size_t *updStartPos, /* (o) updated start point (Q-2) */
     int16_t *idata,   /* (i) original data buffer */
-    int16_t idatal,   /* (i) dimension of idata */
-    int16_t centerStartPos, /* (i) beginning center segment */
-    int16_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    size_t idatal,   /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
     int16_t *surround,  /* (i/o) The contribution from this sequence
                                  summed with earlier contributions */
     int16_t gain    /* (i) Gain to use for this sequence */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
index d89770ec0e..e63dda8c8f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -42,7 +42,8 @@ void WebRtcIlbcfix_SimpleInterpolateLsf(
     IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                         ) {
-  int i, pos, lp_length;
+  size_t i;
+  int pos, lp_length;
 
   int16_t *lsf2, *lsfdeq2;
   /* Stack based */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
index dfc637bef4..72d80e0430 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -34,7 +34,7 @@ void WebRtcIlbcfix_SimpleLpcAnalysis(
                                      ) {
   int k;
   int scale;
-  int16_t is;
+  size_t is;
   int16_t stability;
   /* Stack based */
   int16_t A[LPC_FILTERORDER + 1];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
index 80b3e1b732..29fe91b87e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -24,14 +24,14 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateConstruct(
-    int16_t idxForMax,   /* (i) 6-bit index for the quantization of
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
                                            max amplitude */
     int16_t *idxVec,   /* (i) vector of quantization indexes */
     int16_t *syntDenum,  /* (i) synthesis filter denumerator */
     int16_t *Out_fix,  /* (o) the decoded state vector */
-    int16_t len    /* (i) length of a state vector */
+    size_t len    /* (i) length of a state vector */
                                   ) {
-  int k;
+  size_t k;
   int16_t maxVal;
   int16_t *tmp1, *tmp2, *tmp3;
   /* Stack based */
@@ -96,11 +96,11 @@ void WebRtcIlbcfix_StateConstruct(
   /* Run MA filter + AR filter */
   WebRtcSpl_FilterMAFastQ12(
       sampleVal, sampleMa,
-      numerator, LPC_FILTERORDER+1, (int16_t)(len + LPC_FILTERORDER));
+      numerator, LPC_FILTERORDER+1, len + LPC_FILTERORDER);
   WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
   WebRtcSpl_FilterARFastQ12(
       sampleMa, sampleAr,
-      syntDenum, LPC_FILTERORDER+1, (int16_t)(2*len));
+      syntDenum, LPC_FILTERORDER+1, 2 * len);
 
   tmp1 = &sampleAr[len-1];
   tmp2 = &sampleAr[2*len-1];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h
index 22d75e2444..26319193b8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -24,12 +24,12 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateConstruct(
-    int16_t idxForMax,   /* (i) 6-bit index for the quantization of
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
                                            max amplitude */
     int16_t *idxVec,   /* (i) vector of quantization indexes */
     int16_t *syntDenum,  /* (i) synthesis filter denumerator */
     int16_t *Out_fix,  /* (o) the decoded state vector */
-    int16_t len    /* (i) length of a state vector */
+    size_t len    /* (i) length of a state vector */
                                   );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_search.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
index 5d85a84b28..295c543d84 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
@@ -33,7 +33,7 @@ void WebRtcIlbcfix_StateSearch(
     int16_t *syntDenum,  /* (i) lpc synthesis filter */
     int16_t *weightDenum  /* (i) weighting filter denuminator */
                                ) {
-  int16_t k, index;
+  size_t k, index;
   int16_t maxVal;
   int16_t scale, shift;
   int32_t maxValsq;
@@ -64,14 +64,14 @@ void WebRtcIlbcfix_StateSearch(
 
   /* Run the Zero-Pole filter (Ciurcular convolution) */
   WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
-  WebRtcSpl_FilterMAFastQ12(
-      residualLong, sampleMa,
-      numerator, LPC_FILTERORDER+1, (int16_t)(iLBCenc_inst->state_short_len + LPC_FILTERORDER));
+  WebRtcSpl_FilterMAFastQ12(residualLong, sampleMa, numerator,
+                            LPC_FILTERORDER + 1,
+                            iLBCenc_inst->state_short_len + LPC_FILTERORDER);
   WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
 
   WebRtcSpl_FilterARFastQ12(
       sampleMa, sampleAr,
-      syntDenum, LPC_FILTERORDER+1, (int16_t)(2*iLBCenc_inst->state_short_len));
+      syntDenum, LPC_FILTERORDER+1, 2 * iLBCenc_inst->state_short_len);
 
   for(k=0;k<iLBCenc_inst->state_short_len;k++){
     sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
index 8bbac42b1c..b795e56ac4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
@@ -24,10 +24,10 @@
 
 void WebRtcIlbcfix_SwapBytes(
     const uint16_t* input,   /* (i) the sequence to swap */
-    int16_t wordLength,      /* (i) number or uint16_t to swap */
+    size_t wordLength,      /* (i) number or uint16_t to swap */
     uint16_t* output         /* (o) the swapped sequence */
                               ) {
-  int k;
+  size_t k;
   for (k = wordLength; k > 0; k--) {
     *output++ = (*input >> 8)|(*input << 8);
     input++;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
index a909b2cda4..a4484d621e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -27,7 +27,7 @@
 
 void WebRtcIlbcfix_SwapBytes(
     const uint16_t* input,   /* (i) the sequence to swap */
-    int16_t wordLength,      /* (i) number or uint16_t to swap */
+    size_t wordLength,      /* (i) number or uint16_t to swap */
     uint16_t* output         /* (o) the swapped sequence */
                               );
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
index 3daf186ce6..b440c7a45f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -19,7 +19,7 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include <string.h>
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 /*---------------------------------------------------------------*
  *  Main program to test iLBC encoding and decoding
@@ -47,11 +47,11 @@ int main(int argc, char* argv[])
   int16_t data[BLOCKL_MAX];
   uint8_t encoded_data[2 * ILBCNOOFWORDS_MAX];
   int16_t decoded_data[BLOCKL_MAX];
-  int len;
-  short pli, mode;
+  int len_int, mode;
+  short pli;
   int blockcount = 0;
   int packetlosscount = 0;
-  int frameLen;
+  size_t frameLen, len, len_i16s;
   int16_t speechType;
   IlbcEncoderInstance *Enc_Inst;
   IlbcDecoderInstance *Dec_Inst;
@@ -152,26 +152,29 @@ int main(int argc, char* argv[])
 
   WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
   WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
-  frameLen = mode*8;
+  frameLen = (size_t)(mode*8);
 
   /* loop over input blocks */
 
-  while (((int16_t)fread(data,sizeof(int16_t),frameLen,ifileid))==
-         frameLen) {
+  while (fread(data,sizeof(int16_t),frameLen,ifileid) == frameLen) {
 
     blockcount++;
 
     /* encoding */
 
     fprintf(stderr, "--- Encoding block %i --- ",blockcount);
-    len = WebRtcIlbcfix_Encode(Enc_Inst, data, (int16_t)frameLen, encoded_data);
+    len_int = WebRtcIlbcfix_Encode(Enc_Inst, data, frameLen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
     fprintf(stderr, "\r");
 
     /* write byte file */
 
-    if (fwrite(encoded_data, sizeof(int16_t),
-               ((len+1)/sizeof(int16_t)), efileid) !=
-        (size_t)(((len+1)/sizeof(int16_t)))) {
+    len_i16s = (len + 1) / sizeof(int16_t);
+    if (fwrite(encoded_data, sizeof(int16_t), len_i16s, efileid) != len_i16s) {
       return -1;
     }
 
@@ -200,8 +203,13 @@ int main(int argc, char* argv[])
 
     fprintf(stderr, "--- Decoding block %i --- ",blockcount);
     if (pli==1) {
-      len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
-                               (int16_t)len, decoded_data,&speechType);
+      len_int=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
+                                   len, decoded_data,&speechType);
+      if (len_int < 0) {
+        fprintf(stderr, "Error decoding\n");
+        exit(0);
+      }
+      len = (size_t)len_int;
     } else {
       len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
     }
@@ -209,8 +217,7 @@ int main(int argc, char* argv[])
 
     /* write output file */
 
-    if (fwrite(decoded_data, sizeof(int16_t), len,
-               ofileid) != (size_t)len) {
+    if (fwrite(decoded_data, sizeof(int16_t), len, ofileid) != len) {
       return -1;
     }
   }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
index 370bf9d8a6..7ffa4a7d0e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -21,7 +21,7 @@ iLBC_test.c
 #include <stdio.h>
 #include <string.h>
 #include <time.h>
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 //#define JUNK_DATA
 #ifdef JUNK_DATA
@@ -41,13 +41,15 @@ int main(int argc, char* argv[])
 {
   FILE *ifileid,*efileid,*ofileid, *chfileid;
   short encoded_data[55], data[240], speechType;
-  short len, mode, pli;
+  int len_int, mode;
+  short pli;
+  size_t len, readlen;
   int blockcount = 0;
 
   IlbcEncoderInstance *Enc_Inst;
   IlbcDecoderInstance *Dec_Inst;
 #ifdef JUNK_DATA
-  int i;
+  size_t i;
   FILE *seedfile;
   unsigned int random_seed = (unsigned int) time(NULL);//1196764538
 #endif
@@ -125,19 +127,21 @@ int main(int argc, char* argv[])
 
   /* loop over input blocks */
 #ifdef SPLIT_10MS
-  while(fread(data, sizeof(short), 80, ifileid) == 80) {
+  readlen = 80;
 #else
-  while((short)fread(data,sizeof(short),(mode<<3),ifileid)==(mode<<3)) {
+  readlen = (size_t)(mode << 3);
 #endif
+  while(fread(data, sizeof(short), readlen, ifileid) == readlen) {
     blockcount++;
 
     /* encoding */
     fprintf(stderr, "--- Encoding block %i --- ",blockcount);
-#ifdef SPLIT_10MS
-    len=WebRtcIlbcfix_Encode(Enc_Inst, data, 80, encoded_data);
-#else
-    len=WebRtcIlbcfix_Encode(Enc_Inst, data, (short)(mode<<3), encoded_data);
-#endif
+    len_int=WebRtcIlbcfix_Encode(Enc_Inst, data, readlen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
     fprintf(stderr, "\r");
 
 #ifdef JUNK_DATA
@@ -148,9 +152,7 @@ int main(int argc, char* argv[])
     /* write byte file */
     if(len != 0){ //len may be 0 in 10ms split case
       fwrite(encoded_data,1,len,efileid);
-    }
 
-    if(len != 0){ //len may be 0 in 10ms split case
       /* get channel data if provided */
       if (argc==6) {
         if (fread(&pli, sizeof(int16_t), 1, chfileid)) {
@@ -173,7 +175,13 @@ int main(int argc, char* argv[])
       /* decoding */
       fprintf(stderr, "--- Decoding block %i --- ",blockcount);
       if (pli==1) {
-        len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data, &speechType);
+        len_int = WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data,
+                                       &speechType);
+        if (len_int < 0) {
+          fprintf(stderr, "Error decoding\n");
+          exit(0);
+        }
+        len = (size_t)len_int;
       } else {
         len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
       }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
index 303ede3e63..5454948287 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -21,13 +21,13 @@
 #include <stdio.h>
 #include <string.h>
 
-#include "defines.h"
-#include "nit_encode.h"
-#include "encode.h"
-#include "init_decode.h"
-#include "decode.h"
-#include "constants.h"
-#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/defines.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/nit_encode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/encode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/decode.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/constants.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 #define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
index dbecc33abe..dc12a5a7c4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
@@ -26,9 +26,9 @@ void WebRtcIlbcfix_Window32W32(
     int32_t *z,    /* Output */
     int32_t *x,    /* Input (same domain as Output)*/
     const int32_t  *y,  /* Q31 Window */
-    int16_t N     /* length to process */
+    size_t N     /* length to process */
                                ) {
-  int16_t i;
+  size_t i;
   int16_t x_low, x_hi, y_low, y_hi;
   int16_t left_shifts;
   int32_t temp;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
index 4ee6fce54f..27ed1b6a33 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -29,7 +29,7 @@ void WebRtcIlbcfix_Window32W32(
     int32_t *z,    /* Output */
     int32_t *x,    /* Input (same domain as Output)*/
     const int32_t  *y,  /* Q31 Window */
-    int16_t N     /* length to process */
+    size_t N     /* length to process */
                                );
 
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
index 3490461d21..0d898c54a4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -23,16 +23,16 @@
  * crossCorr*crossCorr/(energy) criteria
  *---------------------------------------------------------------*/
 
-int WebRtcIlbcfix_XcorrCoef(
+size_t WebRtcIlbcfix_XcorrCoef(
     int16_t *target,  /* (i) first array */
     int16_t *regressor, /* (i) second array */
-    int16_t subl,  /* (i) dimension arrays */
-    int16_t searchLen, /* (i) the search lenght */
-    int16_t offset,  /* (i) samples offset between arrays */
+    size_t subl,  /* (i) dimension arrays */
+    size_t searchLen, /* (i) the search lenght */
+    size_t offset,  /* (i) samples offset between arrays */
     int16_t step   /* (i) +1 or -1 */
                             ){
-  int k;
-  int16_t maxlag;
+  size_t k;
+  size_t maxlag;
   int16_t pos;
   int16_t max;
   int16_t crossCorrScale, Energyscale;
@@ -55,13 +55,13 @@ int WebRtcIlbcfix_XcorrCoef(
 
   /* Find scale value and start position */
   if (step==1) {
-    max=WebRtcSpl_MaxAbsValueW16(regressor, (int16_t)(subl+searchLen-1));
+    max=WebRtcSpl_MaxAbsValueW16(regressor, subl + searchLen - 1);
     rp_beg = regressor;
-    rp_end = &regressor[subl];
+    rp_end = regressor + subl;
   } else { /* step==-1 */
-    max=WebRtcSpl_MaxAbsValueW16(&regressor[-searchLen], (int16_t)(subl+searchLen-1));
-    rp_beg = &regressor[-1];
-    rp_end = &regressor[subl-1];
+    max = WebRtcSpl_MaxAbsValueW16(regressor - searchLen, subl + searchLen - 1);
+    rp_beg = regressor - 1;
+    rp_end = regressor + subl - 1;
   }
 
   /* Introduce a scale factor on the Energy in int32_t in
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
index 1f4c58d934..9b81c0fe97 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -26,12 +26,12 @@
  * crossCorr*crossCorr/(energy) criteria
  *---------------------------------------------------------------*/
 
-int WebRtcIlbcfix_XcorrCoef(
+size_t WebRtcIlbcfix_XcorrCoef(
     int16_t *target,  /* (i) first array */
     int16_t *regressor, /* (i) second array */
-    int16_t subl,  /* (i) dimension arrays */
-    int16_t searchLen, /* (i) the search lenght */
-    int16_t offset,  /* (i) samples offset between arrays */
+    size_t subl,  /* (i) dimension arrays */
+    size_t searchLen, /* (i) the search lenght */
+    size_t offset,  /* (i) samples offset between arrays */
     int16_t step   /* (i) +1 or -1 */
                             );
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
new file mode 100644
index 0000000000..845af42479
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+
+namespace webrtc {
+
+template <typename T>
+class AudioDecoderIsacT final : public AudioDecoder {
+ public:
+  AudioDecoderIsacT();
+  explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
+  ~AudioDecoderIsacT() override;
+
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  int IncomingPacket(const uint8_t* payload,
+                     size_t payload_len,
+                     uint16_t rtp_sequence_number,
+                     uint32_t rtp_timestamp,
+                     uint32_t arrival_timestamp) override;
+  int ErrorCode() override;
+  size_t Channels() const override;
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  typename T::instance_type* isac_state_;
+  LockedIsacBandwidthInfo* bwinfo_;
+  int decoder_sample_rate_hz_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
new file mode 100644
index 0000000000..a986bc479d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT()
+    : AudioDecoderIsacT(nullptr) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
+    : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
+  RTC_CHECK_EQ(0, T::Create(&isac_state_));
+  T::DecoderInit(isac_state_);
+  if (bwinfo_) {
+    IsacBandwidthInfo bi;
+    T::GetBandwidthInfo(isac_state_, &bi);
+    bwinfo_->Set(bi);
+  }
+}
+
+template <typename T>
+AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
+  RTC_CHECK_EQ(0, T::Free(isac_state_));
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
+      << "Unsupported sample rate " << sample_rate_hz;
+  if (sample_rate_hz != decoder_sample_rate_hz_) {
+    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
+    decoder_sample_rate_hz_ = sample_rate_hz;
+  }
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+template <typename T>
+bool AudioDecoderIsacT<T>::HasDecodePlc() const {
+  return false;
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return T::DecodePlc(isac_state_, decoded, num_frames);
+}
+
+template <typename T>
+void AudioDecoderIsacT<T>::Reset() {
+  T::DecoderInit(isac_state_);
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
+                                         size_t payload_len,
+                                         uint16_t rtp_sequence_number,
+                                         uint32_t rtp_timestamp,
+                                         uint32_t arrival_timestamp) {
+  int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
+                                rtp_sequence_number, rtp_timestamp,
+                                arrival_timestamp);
+  if (bwinfo_) {
+    IsacBandwidthInfo bwinfo;
+    T::GetBandwidthInfo(isac_state_, &bwinfo);
+    bwinfo_->Set(bwinfo);
+  }
+  return ret;
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::ErrorCode() {
+  return T::GetErrorCode(isac_state_);
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index cda554876b..321dac3567 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -13,120 +13,85 @@
 
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
+struct CodecInst;
 
 template <typename T>
-class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
+class AudioEncoderIsacT final : public AudioEncoder {
  public:
-  // For constructing an encoder in instantaneous mode. Allowed combinations
-  // are
+  // Allowed combinations of sample rate, frame size, and bit rate are
   //  - 16000 Hz, 30 ms, 10000-32000 bps
   //  - 16000 Hz, 60 ms, 10000-32000 bps
   //  - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
-  //  - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
   struct Config {
-    Config();
     bool IsOk() const;
-    int payload_type;
-    int sample_rate_hz;
-    int frame_size_ms;
-    int bit_rate;  // Limit on the short-term average bit rate, in bits/second.
-    int max_bit_rate;
-    int max_payload_size_bytes;
+
+    LockedIsacBandwidthInfo* bwinfo = nullptr;
+
+    int payload_type = 103;
+    int sample_rate_hz = 16000;
+    int frame_size_ms = 30;
+    int bit_rate = kDefaultBitRate;  // Limit on the short-term average bit
+                                     // rate, in bits/s.
+    int max_payload_size_bytes = -1;
+    int max_bit_rate = -1;
+
+    // If true, the encoder will dynamically adjust frame size and bit rate;
+    // the configured values are then merely the starting point.
+    bool adaptive_mode = false;
+
+    // In adaptive mode, prevent adaptive changes to the frame size. (Not used
+    // in nonadaptive mode.)
+    bool enforce_frame_size = false;
   };
 
-  // For constructing an encoder in channel-adaptive mode. Allowed combinations
-  // are
-  //  - 16000 Hz, 30 ms, 10000-32000 bps
-  //  - 16000 Hz, 60 ms, 10000-32000 bps
-  //  - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
-  //  - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
-  struct ConfigAdaptive {
-    ConfigAdaptive();
-    bool IsOk() const;
-    int payload_type;
-    int sample_rate_hz;
-    int initial_frame_size_ms;
-    int initial_bit_rate;
-    int max_bit_rate;
-    bool enforce_frame_size;  // Prevent adaptive changes to the frame size?
-    int max_payload_size_bytes;
-  };
+  explicit AudioEncoderIsacT(const Config& config);
+  explicit AudioEncoderIsacT(const CodecInst& codec_inst,
+                             LockedIsacBandwidthInfo* bwinfo);
+  ~AudioEncoderIsacT() override;
 
-  explicit AudioEncoderDecoderIsacT(const Config& config);
-  explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config);
-  ~AudioEncoderDecoderIsacT() override;
-
-  // AudioEncoder public methods.
-  int SampleRateHz() const override;
-  int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
-
-  // AudioDecoder methods.
-  bool HasDecodePlc() const override;
-  int DecodePlc(int num_frames, int16_t* decoded) override;
-  int Init() override;
-  int IncomingPacket(const uint8_t* payload,
-                     size_t payload_len,
-                     uint16_t rtp_sequence_number,
-                     uint32_t rtp_timestamp,
-                     uint32_t arrival_timestamp) override;
-  int ErrorCode() override;
-  size_t Channels() const override { return 1; }
-
- protected:
-  // AudioEncoder protected method.
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
-
-  // AudioDecoder protected method.
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
+  void Reset() override;
 
  private:
   // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
   // STREAM_MAXW16_60MS for iSAC fix (60 ms).
   static const size_t kSufficientEncodeBufferSizeBytes = 400;
 
-  const int payload_type_;
+  static const int kDefaultBitRate = 32000;
 
-  // iSAC encoder/decoder state, guarded by a mutex to ensure that encode calls
-  // from one thread won't clash with decode calls from another thread.
-  // Note: PT_GUARDED_BY is disabled since it is not yet supported by clang.
-  const rtc::scoped_ptr<CriticalSectionWrapper> state_lock_;
-  typename T::instance_type* isac_state_
-      GUARDED_BY(state_lock_) /* PT_GUARDED_BY(lock_)*/;
+  // Recreate the iSAC encoder instance with the given settings, and save them.
+  void RecreateEncoderInstance(const Config& config);
 
-  int decoder_sample_rate_hz_ GUARDED_BY(state_lock_);
-
-  // Must be acquired before state_lock_.
-  const rtc::scoped_ptr<CriticalSectionWrapper> lock_;
+  Config config_;
+  typename T::instance_type* isac_state_ = nullptr;
+  LockedIsacBandwidthInfo* bwinfo_ = nullptr;
 
   // Have we accepted input but not yet emitted it in a packet?
-  bool packet_in_progress_ GUARDED_BY(lock_);
+  bool packet_in_progress_ = false;
 
   // Timestamp of the first input of the currently in-progress packet.
-  uint32_t packet_timestamp_ GUARDED_BY(lock_);
+  uint32_t packet_timestamp_;
 
   // Timestamp of the previously encoded packet.
-  uint32_t last_encoded_timestamp_ GUARDED_BY(lock_);
+  uint32_t last_encoded_timestamp_;
 
-  DISALLOW_COPY_AND_ASSIGN(AudioEncoderDecoderIsacT);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
 }  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index e81686a96c..d4438cc775 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -11,35 +11,36 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
-
-#include <algorithm>
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
-const int kIsacPayloadType = 103;
-const int kDefaultBitRate = 32000;
-
 template <typename T>
-AudioEncoderDecoderIsacT<T>::Config::Config()
-    : payload_type(kIsacPayloadType),
-      sample_rate_hz(16000),
-      frame_size_ms(30),
-      bit_rate(kDefaultBitRate),
-      max_bit_rate(-1),
-      max_payload_size_bytes(-1) {
+typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
+    const CodecInst& codec_inst,
+    LockedIsacBandwidthInfo* bwinfo) {
+  typename AudioEncoderIsacT<T>::Config config;
+  config.bwinfo = bwinfo;
+  config.payload_type = codec_inst.pltype;
+  config.sample_rate_hz = codec_inst.plfreq;
+  config.frame_size_ms =
+      rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
+  config.adaptive_mode = (codec_inst.rate == -1);
+  if (codec_inst.rate != -1)
+    config.bit_rate = codec_inst.rate;
+  return config;
 }
 
 template <typename T>
-bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const {
+bool AudioEncoderIsacT<T>::Config::IsOk() const {
   if (max_bit_rate < 32000 && max_bit_rate != -1)
     return false;
   if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
     return false;
+  if (adaptive_mode && !bwinfo)
+    return false;
   switch (sample_rate_hz) {
     case 16000:
       if (max_bit_rate > 53400)
@@ -47,164 +48,92 @@ bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const {
       if (max_payload_size_bytes > 400)
         return false;
       return (frame_size_ms == 30 || frame_size_ms == 60) &&
-             ((bit_rate >= 10000 && bit_rate <= 32000) || bit_rate == 0);
+             (bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 32000));
     case 32000:
-    case 48000:
       if (max_bit_rate > 160000)
         return false;
       if (max_payload_size_bytes > 600)
         return false;
       return T::has_swb &&
              (frame_size_ms == 30 &&
-              ((bit_rate >= 10000 && bit_rate <= 56000) || bit_rate == 0));
+              (bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 56000)));
     default:
       return false;
   }
 }
 
 template <typename T>
-AudioEncoderDecoderIsacT<T>::ConfigAdaptive::ConfigAdaptive()
-    : payload_type(kIsacPayloadType),
-      sample_rate_hz(16000),
-      initial_frame_size_ms(30),
-      initial_bit_rate(kDefaultBitRate),
-      max_bit_rate(-1),
-      enforce_frame_size(false),
-      max_payload_size_bytes(-1) {
+AudioEncoderIsacT<T>::AudioEncoderIsacT(const Config& config) {
+  RecreateEncoderInstance(config);
 }
 
 template <typename T>
-bool AudioEncoderDecoderIsacT<T>::ConfigAdaptive::IsOk() const {
-  if (max_bit_rate < 32000 && max_bit_rate != -1)
-    return false;
-  if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
-    return false;
-  switch (sample_rate_hz) {
-    case 16000:
-      if (max_bit_rate > 53400)
-        return false;
-      if (max_payload_size_bytes > 400)
-        return false;
-      return (initial_frame_size_ms == 30 || initial_frame_size_ms == 60) &&
-             initial_bit_rate >= 10000 && initial_bit_rate <= 32000;
-    case 32000:
-    case 48000:
-      if (max_bit_rate > 160000)
-        return false;
-      if (max_payload_size_bytes > 600)
-        return false;
-      return T::has_swb &&
-             (initial_frame_size_ms == 30 && initial_bit_rate >= 10000 &&
-              initial_bit_rate <= 56000);
-    default:
-      return false;
-  }
+AudioEncoderIsacT<T>::AudioEncoderIsacT(const CodecInst& codec_inst,
+                                        LockedIsacBandwidthInfo* bwinfo)
+    : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
+
+template <typename T>
+AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
+  RTC_CHECK_EQ(0, T::Free(isac_state_));
 }
 
 template <typename T>
-AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
-    : payload_type_(config.payload_type),
-      state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      decoder_sample_rate_hz_(0),
-      lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      packet_in_progress_(false) {
-  CHECK(config.IsOk());
-  CHECK_EQ(0, T::Create(&isac_state_));
-  CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
-  CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
-  CHECK_EQ(0, T::Control(isac_state_, config.bit_rate == 0 ? kDefaultBitRate
-                                                           : config.bit_rate,
-                         config.frame_size_ms));
-  // When config.sample_rate_hz is set to 48000 Hz (iSAC-fb), the decoder is
-  // still set to 32000 Hz, since there is no full-band mode in the decoder.
-  CHECK_EQ(0, T::SetDecSampRate(isac_state_,
-                                std::min(config.sample_rate_hz, 32000)));
-  if (config.max_payload_size_bytes != -1)
-    CHECK_EQ(0,
-             T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
-  if (config.max_bit_rate != -1)
-    CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
-}
-
-template <typename T>
-AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(
-    const ConfigAdaptive& config)
-    : payload_type_(config.payload_type),
-      state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      decoder_sample_rate_hz_(0),
-      lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      packet_in_progress_(false) {
-  CHECK(config.IsOk());
-  CHECK_EQ(0, T::Create(&isac_state_));
-  CHECK_EQ(0, T::EncoderInit(isac_state_, 0));
-  CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
-  CHECK_EQ(0, T::ControlBwe(isac_state_, config.initial_bit_rate,
-                            config.initial_frame_size_ms,
-                            config.enforce_frame_size));
-  CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
-  if (config.max_payload_size_bytes != -1)
-    CHECK_EQ(0,
-             T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
-  if (config.max_bit_rate != -1)
-    CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
-}
-
-template <typename T>
-AudioEncoderDecoderIsacT<T>::~AudioEncoderDecoderIsacT() {
-  CHECK_EQ(0, T::Free(isac_state_));
-}
-
-template <typename T>
-int AudioEncoderDecoderIsacT<T>::SampleRateHz() const {
-  CriticalSectionScoped cs(state_lock_.get());
-  return T::EncSampRate(isac_state_);
-}
-
-template <typename T>
-int AudioEncoderDecoderIsacT<T>::NumChannels() const {
-  return 1;
-}
-
-template <typename T>
-size_t AudioEncoderDecoderIsacT<T>::MaxEncodedBytes() const {
+size_t AudioEncoderIsacT<T>::MaxEncodedBytes() const {
   return kSufficientEncodeBufferSizeBytes;
 }
 
 template <typename T>
-int AudioEncoderDecoderIsacT<T>::Num10MsFramesInNextPacket() const {
-  CriticalSectionScoped cs(state_lock_.get());
-  const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
-  return rtc::CheckedDivExact(samples_in_next_packet,
-                              rtc::CheckedDivExact(SampleRateHz(), 100));
+int AudioEncoderIsacT<T>::SampleRateHz() const {
+  return T::EncSampRate(isac_state_);
 }
 
 template <typename T>
-int AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const {
+size_t AudioEncoderIsacT<T>::NumChannels() const {
+  return 1;
+}
+
+template <typename T>
+size_t AudioEncoderIsacT<T>::Num10MsFramesInNextPacket() const {
+  const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
+  return static_cast<size_t>(
+      rtc::CheckedDivExact(samples_in_next_packet,
+                           rtc::CheckedDivExact(SampleRateHz(), 100)));
+}
+
+template <typename T>
+size_t AudioEncoderIsacT<T>::Max10MsFramesInAPacket() const {
   return 6;  // iSAC puts at most 60 ms in a packet.
 }
 
 template <typename T>
-AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT<T>::EncodeInternal(
+int AudioEncoderIsacT<T>::GetTargetBitrate() const {
+  if (config_.adaptive_mode)
+    return -1;
+  return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
+}
+
+template <typename T>
+AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  CriticalSectionScoped cs_lock(lock_.get());
   if (!packet_in_progress_) {
     // Starting a new packet; remember the timestamp for later.
     packet_in_progress_ = true;
     packet_timestamp_ = rtp_timestamp;
   }
-  int r;
-  {
-    CriticalSectionScoped cs(state_lock_.get());
-    r = T::Encode(isac_state_, audio, encoded);
+  if (bwinfo_) {
+    IsacBandwidthInfo bwinfo = bwinfo_->Get();
+    T::SetBandwidthInfo(isac_state_, &bwinfo);
   }
-  CHECK_GE(r, 0);
+  int r = T::Encode(isac_state_, audio.data(), encoded);
+  RTC_CHECK_GE(r, 0) << "Encode failed (error code "
+                     << T::GetErrorCode(isac_state_) << ")";
 
   // T::Encode doesn't allow us to tell it the size of the output
   // buffer. All we can do is check for an overrun after the fact.
-  CHECK(static_cast<size_t>(r) <= max_encoded_bytes);
+  RTC_CHECK_LE(static_cast<size_t>(r), max_encoded_bytes);
 
   if (r == 0)
     return EncodedInfo();
@@ -215,68 +144,45 @@ AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT<T>::EncodeInternal(
   EncodedInfo info;
   info.encoded_bytes = r;
   info.encoded_timestamp = packet_timestamp_;
-  info.payload_type = payload_type_;
+  info.payload_type = config_.payload_type;
   return info;
 }
 
 template <typename T>
-int AudioEncoderDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
-                                                size_t encoded_len,
-                                                int sample_rate_hz,
-                                                int16_t* decoded,
-                                                SpeechType* speech_type) {
-  CriticalSectionScoped cs(state_lock_.get());
-  // We want to crate the illusion that iSAC supports 48000 Hz decoding, while
-  // in fact it outputs 32000 Hz. This is the iSAC fullband mode.
-  if (sample_rate_hz == 48000)
-    sample_rate_hz = 32000;
-  CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
-      << "Unsupported sample rate " << sample_rate_hz;
-  if (sample_rate_hz != decoder_sample_rate_hz_) {
-    CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
-    decoder_sample_rate_hz_ = sample_rate_hz;
+void AudioEncoderIsacT<T>::Reset() {
+  RecreateEncoderInstance(config_);
+}
+
+template <typename T>
+void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
+  RTC_CHECK(config.IsOk());
+  packet_in_progress_ = false;
+  bwinfo_ = config.bwinfo;
+  if (isac_state_)
+    RTC_CHECK_EQ(0, T::Free(isac_state_));
+  RTC_CHECK_EQ(0, T::Create(&isac_state_));
+  RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
+  RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
+  const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
+  if (config.adaptive_mode) {
+    RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
+                                  config.enforce_frame_size));
+  } else {
+    RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
   }
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret =
-      T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len),
-                        decoded, &temp_type);
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
+  if (config.max_payload_size_bytes != -1)
+    RTC_CHECK_EQ(
+        0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
+  if (config.max_bit_rate != -1)
+    RTC_CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
 
-template <typename T>
-bool AudioEncoderDecoderIsacT<T>::HasDecodePlc() const {
-  return false;
-}
+  // Set the decoder sample rate even though we just use the encoder. This
+  // doesn't appear to be necessary to produce a valid encoding, but without it
+  // we get an encoding that isn't bit-for-bit identical with what a combined
+  // encoder+decoder object produces.
+  RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
 
-template <typename T>
-int AudioEncoderDecoderIsacT<T>::DecodePlc(int num_frames, int16_t* decoded) {
-  CriticalSectionScoped cs(state_lock_.get());
-  return T::DecodePlc(isac_state_, decoded, num_frames);
-}
-
-template <typename T>
-int AudioEncoderDecoderIsacT<T>::Init() {
-  CriticalSectionScoped cs(state_lock_.get());
-  return T::DecoderInit(isac_state_);
-}
-
-template <typename T>
-int AudioEncoderDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
-                                                size_t payload_len,
-                                                uint16_t rtp_sequence_number,
-                                                uint32_t rtp_timestamp,
-                                                uint32_t arrival_timestamp) {
-  CriticalSectionScoped cs(state_lock_.get());
-  return T::UpdateBwEstimate(
-      isac_state_, payload, static_cast<int32_t>(payload_len),
-      rtp_sequence_number, rtp_timestamp, arrival_timestamp);
-}
-
-template <typename T>
-int AudioEncoderDecoderIsacT<T>::ErrorCode() {
-  CriticalSectionScoped cs(state_lock_.get());
-  return T::GetErrorCode(isac_state_);
+  config_ = config;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
new file mode 100644
index 0000000000..1e3f4c9a86
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+
+#include "webrtc/typedefs.h"
+
+typedef struct {
+  int in_use;
+  int32_t send_bw_avg;
+  int32_t send_max_delay_avg;
+  int16_t bottleneck_idx;
+  int16_t jitter_info;
+} IsacBandwidthInfo;
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h
new file mode 100644
index 0000000000..e78eb786ad
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
+
+namespace webrtc {
+
+using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h
new file mode 100644
index 0000000000..b97f04bbf2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
+
+namespace webrtc {
+
+using AudioEncoderIsacFix = AudioEncoderIsacT<IsacFix>;
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
index 961fd3fad5..7f277ca25c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_
 
-/*
- * Define the fixpoint numeric formats
- */
+#include <stddef.h>
+
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 #include "webrtc/typedefs.h"
 
 typedef struct {
@@ -128,9 +128,9 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
-                               const int16_t *speechIn,
-                               uint8_t* encoded);
+  int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+                           const int16_t *speechIn,
+                           uint8_t* encoded);
 
 
 
@@ -174,14 +174,9 @@ extern "C" {
    *
    * Input:
    *  - ISAC_main_inst : ISAC instance.
-   *
-   * Return value
-   *       :  0 - Ok
-   *         -1 - Error
    */
 
-  int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
-
+  void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct* ISAC_main_inst);
 
   /****************************************************************************
    * WebRtcIsacfix_UpdateBwEstimate1(...)
@@ -191,7 +186,7 @@ extern "C" {
    * Input:
    *      - ISAC_main_inst    : ISAC instance.
    *      - encoded           : encoded ISAC frame(s).
-   *      - packet_size       : size of the packet.
+   *      - packet_size       : size of the packet in bytes.
    *      - rtp_seq_number    : the RTP number of the packet.
    *      - arr_ts            : the arrival time of the packet (from NetEq)
    *                            in samples.
@@ -202,7 +197,7 @@ extern "C" {
 
   int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
                                           const uint8_t* encoded,
-                                          int32_t  packet_size,
+                                          size_t packet_size,
                                           uint16_t rtp_seq_number,
                                           uint32_t arr_ts);
 
@@ -214,7 +209,7 @@ extern "C" {
    * Input:
    *      - ISAC_main_inst    : ISAC instance.
    *      - encoded           : encoded ISAC frame(s).
-   *      - packet_size       : size of the packet.
+   *      - packet_size       : size of the packet in bytes.
    *      - rtp_seq_number    : the RTP number of the packet.
    *      - send_ts           : the send time of the packet from RTP header,
    *                            in samples.
@@ -227,10 +222,10 @@ extern "C" {
 
   int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
                                          const uint8_t* encoded,
-                                         int32_t          packet_size,
-                                         uint16_t         rtp_seq_number,
-                                         uint32_t         send_ts,
-                                         uint32_t         arr_ts);
+                                         size_t packet_size,
+                                         uint16_t rtp_seq_number,
+                                         uint32_t send_ts,
+                                         uint32_t arr_ts);
 
   /****************************************************************************
    * WebRtcIsacfix_Decode(...)
@@ -251,11 +246,11 @@ extern "C" {
    *                            -1 - Error
    */
 
-  int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                               const uint8_t* encoded,
-                               int16_t len,
-                               int16_t *decoded,
-                               int16_t *speechType);
+  int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
+                           const uint8_t* encoded,
+                           size_t len,
+                           int16_t *decoded,
+                           int16_t *speechType);
 
 
   /****************************************************************************
@@ -280,11 +275,11 @@ extern "C" {
    */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-  int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                 const uint16_t *encoded,
-                                 int16_t len,
-                                 int16_t *decoded,
-                                 int16_t *speechType);
+  int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
+                             const uint16_t *encoded,
+                             size_t len,
+                             int16_t *decoded,
+                             int16_t *speechType);
 #endif //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
 
@@ -305,14 +300,13 @@ extern "C" {
    * Output:
    *      - decoded           : The decoded vector
    *
-   * Return value             : >0 - number of samples in decoded PLC vector
-   *                            -1 - Error
+   * Return value             : Number of samples in decoded PLC vector
    */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-  int16_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                    int16_t *decoded,
-                                    int16_t noOfLostFrames);
+  size_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
+                                   int16_t *decoded,
+                                   size_t noOfLostFrames);
 #endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
 
 
@@ -334,13 +328,12 @@ extern "C" {
    * Output:
    *      - decoded           : The decoded vector
    *
-   * Return value             : >0 - number of samples in decoded PLC vector
-   *                            -1 - Error
+   * Return value             : Number of samples in decoded PLC vector
    */
 
-  int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
-                                  int16_t *decoded,
-                                  int16_t noOfLostFrames );
+  size_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
+                                 int16_t *decoded,
+                                 size_t noOfLostFrames );
 
 
   /****************************************************************************
@@ -358,8 +351,8 @@ extern "C" {
    */
 
   int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
-                                     int encoded_len_bytes,
-                                     int16_t* frameLength);
+                                     size_t encoded_len_bytes,
+                                     size_t* frameLength);
 
   /****************************************************************************
    * WebRtcIsacfix_Control(...)
@@ -378,10 +371,11 @@ extern "C" {
    */
 
   int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
-                                int16_t          rate,
-                                int16_t          framesize);
-
+                                int16_t rate,
+                                int framesize);
 
+  void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst,
+                                             int bottleneck_bits_per_second);
 
   /****************************************************************************
    * WebRtcIsacfix_ControlBwe(...)
@@ -407,7 +401,7 @@ extern "C" {
 
   int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
                                    int16_t rateBPS,
-                                   int16_t frameSizeMs,
+                                   int frameSizeMs,
                                    int16_t enforceFrameSize);
 
 
@@ -609,7 +603,7 @@ extern "C" {
    */
 
   int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
-                                    int encoded_len_bytes,
+                                    size_t encoded_len_bytes,
                                     int16_t* rateIndex);
 
 
@@ -626,6 +620,13 @@ extern "C" {
 
   int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
 
+  /* Fills in an IsacBandwidthInfo struct. */
+  void WebRtcIsacfix_GetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
+                                      IsacBandwidthInfo* bwinfo);
+
+  /* Uses the values from an IsacBandwidthInfo struct. */
+  void WebRtcIsacfix_SetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
+                                      const IsacBandwidthInfo* bwinfo);
 
 #if defined(__cplusplus)
 }
@@ -633,4 +634,4 @@ extern "C" {
 
 
 
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_ */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index 23048a5c38..808aeb7fd9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -226,10 +226,10 @@ int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
  * Return value             : number of bytes in the stream so far
  *                            -1 if error detected
  */
-int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
-                                        Bitstr_dec *streamData,
-                                        const int32_t *envQ8,
-                                        const int16_t lenData)
+int WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
+                                    Bitstr_dec *streamData,
+                                    const int32_t *envQ8,
+                                    const int16_t lenData)
 {
   uint32_t    W_lower;
   uint32_t    W_upper;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
index 584bc471f2..40bbb4cdaa 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
@@ -74,7 +74,7 @@ int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
  * Return value             : number of bytes in the stream so far
  *                            <0 if error detected
  */
-int16_t WebRtcIsacfix_DecLogisticMulti2(
+int WebRtcIsacfix_DecLogisticMulti2(
     int16_t *data,
     Bitstr_dec *streamData,
     const int32_t *env,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
index aecb0ec0ce..45eefb913b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
@@ -8,15 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
 
-#include <vector>
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
 
 namespace webrtc {
 
-MockNonlinearBeamformer::MockNonlinearBeamformer(
-    const std::vector<Point>& array_geometry)
-    : NonlinearBeamformer(array_geometry) {
-}
+// Explicit instantiation:
+template class AudioDecoderIsacT<IsacFix>;
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
index d0aea26e44..257a8b5597 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
@@ -8,16 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
 
 #include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
 
 namespace webrtc {
 
-const uint16_t IsacFix::kFixSampleRate;
-
-// Explicit instantiation of AudioEncoderDecoderIsacT<IsacFix>, a.k.a.
-// AudioEncoderDecoderIsacFix.
-template class AudioEncoderDecoderIsacT<IsacFix>;
+// Explicit instantiation:
+template class AudioEncoderIsacT<IsacFix>;
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index 67b18331b3..b074962eae 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@@ -19,6 +19,8 @@
  */
 
 #include "bandwidth_estimator.h"
+
+#include <assert.h>
 #include "settings.h"
 
 
@@ -116,6 +118,8 @@ int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bweStr)
   bweStr->maxBwInv              = kInvBandwidth[3];
   bweStr->minBwInv              = kInvBandwidth[2];
 
+  bweStr->external_bw_info.in_use = 0;
+
   return 0;
 }
 
@@ -144,7 +148,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
                                          const int16_t  frameSize,
                                          const uint32_t sendTime,
                                          const uint32_t arrivalTime,
-                                         const int16_t  pksize,
+                                         const size_t   pksize,
                                          const uint16_t Index)
 {
   uint16_t  weight = 0;
@@ -176,6 +180,8 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
 
   int16_t errCode;
 
+  assert(!bweStr->external_bw_info.in_use);
+
   /* UPDATE ESTIMATES FROM OTHER SIDE */
 
   /* The function also checks if Index has a valid value */
@@ -373,8 +379,8 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
 
         /* compute inverse receiving rate for last packet, in Q19 */
         numBytesInv = (uint16_t) WebRtcSpl_DivW32W16(
-            524288 + ((pksize + HEADER_SIZE) >> 1),
-            pksize + HEADER_SIZE);
+            (int32_t)(524288 + ((pksize + HEADER_SIZE) >> 1)),
+            (int16_t)(pksize + HEADER_SIZE));
 
         /* 8389 is  ~ 1/128000 in Q30 */
         byteSecondsPerBit = (uint32_t)(arrTimeDiff * 8389);
@@ -545,6 +551,8 @@ int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr,
 {
   uint16_t RateInd;
 
+  assert(!bweStr->external_bw_info.in_use);
+
   if ( (Index < 0) || (Index > 23) ) {
     return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
   }
@@ -616,6 +624,9 @@ uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bweStr)
   int32_t  tempMin;
   int32_t  tempMax;
 
+  if (bweStr->external_bw_info.in_use)
+    return bweStr->external_bw_info.bottleneck_idx;
+
   /* Get Rate Index */
 
   /* Get unquantized rate. Always returns 10000 <= rate <= 32000 */
@@ -721,6 +732,8 @@ uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
   int32_t   rec_jitter_short_term_abs_inv; /* Q18 */
   int32_t   temp;
 
+  assert(!bweStr->external_bw_info.in_use);
+
   /* Q18  rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
      2^18 then needs to be shifted 13 bits to 2^31 */
   rec_jitter_short_term_abs_inv = 0x80000000u / bweStr->recJitterShortTermAbs;
@@ -777,6 +790,8 @@ int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
 {
   int16_t recMaxDelay = (int16_t)(bweStr->recMaxDelay >> 15);
 
+  assert(!bweStr->external_bw_info.in_use);
+
   /* limit range of jitter estimate */
   if (recMaxDelay < MIN_ISAC_MD) {
     recMaxDelay = MIN_ISAC_MD;
@@ -787,42 +802,39 @@ int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
   return recMaxDelay;
 }
 
-/* get the bottle neck rate from here to far side, as estimated by far side */
-int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bweStr)
-{
-  int16_t send_bw;
-
-  send_bw = (int16_t) WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7);
-
-  /* limit range of bottle neck rate */
-  if (send_bw < MIN_ISAC_BW) {
-    send_bw = MIN_ISAC_BW;
-  } else if (send_bw > MAX_ISAC_BW) {
-    send_bw = MAX_ISAC_BW;
-  }
-
-  return send_bw;
+/* Clamp val to the closed interval [min,max]. */
+static int16_t clamp(int16_t val, int16_t min, int16_t max) {
+  assert(min <= max);
+  return val < min ? min : (val > max ? max : val);
 }
 
-
-
-/* Returns the max delay value from the other side in ms */
-int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bweStr)
-{
-  int16_t send_max_delay = (int16_t)(bweStr->sendMaxDelayAvg >> 9);
-
-  /* limit range of jitter estimate */
-  if (send_max_delay < MIN_ISAC_MD) {
-    send_max_delay = MIN_ISAC_MD;
-  } else if (send_max_delay > MAX_ISAC_MD) {
-    send_max_delay = MAX_ISAC_MD;
-  }
-
-  return send_max_delay;
+int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr* bweStr) {
+  return bweStr->external_bw_info.in_use
+             ? bweStr->external_bw_info.send_bw_avg
+             : clamp(bweStr->sendBwAvg >> 7, MIN_ISAC_BW, MAX_ISAC_BW);
 }
 
+int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bweStr) {
+  return bweStr->external_bw_info.in_use
+             ? bweStr->external_bw_info.send_max_delay_avg
+             : clamp(bweStr->sendMaxDelayAvg >> 9, MIN_ISAC_MD, MAX_ISAC_MD);
+}
 
+void WebRtcIsacfixBw_GetBandwidthInfo(BwEstimatorstr* bweStr,
+                                   IsacBandwidthInfo* bwinfo) {
+  assert(!bweStr->external_bw_info.in_use);
+  bwinfo->in_use = 1;
+  bwinfo->send_bw_avg = WebRtcIsacfix_GetUplinkBandwidth(bweStr);
+  bwinfo->send_max_delay_avg = WebRtcIsacfix_GetUplinkMaxDelay(bweStr);
+  bwinfo->bottleneck_idx = WebRtcIsacfix_GetDownlinkBwIndexImpl(bweStr);
+  bwinfo->jitter_info = 0;  // Not used.
+}
 
+void WebRtcIsacfixBw_SetBandwidthInfo(BwEstimatorstr* bweStr,
+                                   const IsacBandwidthInfo* bwinfo) {
+  memcpy(&bweStr->external_bw_info, bwinfo,
+         sizeof bweStr->external_bw_info);
+}
 
 /*
  * update long-term average bitrate and amount of data in buffer
@@ -1005,13 +1017,17 @@ int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples)
   /* find new SNR value */
   //consider BottleNeck to be in Q10 ( * 1 in Q10)
   switch(framesamples) {
+  // TODO(bjornv): The comments below confuses me. I don't know if there is a
+  // difference between frame lengths (in which case the implementation is
+  // wrong), or if it is frame length independent in which case we should
+  // correct the comment and simplify the implementation.
     case 480:
       /*s2nr = -1*(a_30 << 10) + ((b_30 * bottle_neck) >> 10);*/
-      s2nr = -22500 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+      s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10);
       break;
     case 960:
       /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/
-      s2nr = -22500 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
+      s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10);
       break;
     default:
       s2nr = -1; /* Error */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
index acd5dd7354..101ef62081 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
@@ -62,7 +62,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr       *bwest_str,
                                          const int16_t         frameSize,
                                          const uint32_t        send_ts,
                                          const uint32_t        arr_ts,
-                                         const int16_t         pksize,
+                                         const size_t          pksize,
                                          const uint16_t        Index);
 
 /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
@@ -95,6 +95,14 @@ int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str);
 /* Returns the max delay value from the other side in ms */
 int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str);
 
+/* Fills in an IsacExternalBandwidthInfo struct. */
+void WebRtcIsacfixBw_GetBandwidthInfo(BwEstimatorstr* bwest_str,
+                                      IsacBandwidthInfo* bwinfo);
+
+/* Uses the values from an IsacExternalBandwidthInfo struct. */
+void WebRtcIsacfixBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
+                                      const IsacBandwidthInfo* bwinfo);
+
 /*
  * update amount of data in bottle neck buffer and burst handling
  * returns minimum payload size (bytes)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
index 488ee2e76a..fdbb2fcb0d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -27,18 +27,18 @@ extern "C" {
 
 int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr* bwest_str,
                                     Bitstr_dec* streamdata,
-                                    int32_t packet_size,
+                                    size_t packet_size,
                                     uint16_t rtp_seq_number,
                                     uint32_t send_ts,
                                     uint32_t arr_ts);
 
-int16_t WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
-                                       IsacFixDecoderInstance* ISACdec_obj,
-                                       int16_t* current_framesamples);
+int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
+                             IsacFixDecoderInstance* ISACdec_obj,
+                             size_t* current_framesamples);
 
-int16_t WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
-                                          IsacFixDecoderInstance* ISACdec_obj,
-                                          int16_t* current_framesample );
+void WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
+                                 IsacFixDecoderInstance* ISACdec_obj,
+                                 size_t* current_framesample );
 
 int WebRtcIsacfix_EncodeImpl(int16_t* in,
                              IsacFixEncoderInstance* ISACenc_obj,
@@ -90,7 +90,7 @@ void WebRtcIsacfix_Spec2TimeC(int16_t* inreQ7,
                               int32_t* outre1Q16,
                               int32_t* outre2Q16);
 
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9,
                                  int16_t* inre2Q9,
                                  int16_t* outre,
@@ -141,7 +141,7 @@ void WebRtcIsacfix_FilterAndCombine2(int16_t* tempin_ch1,
 
 /* normalized lattice filters */
 
-void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
+void WebRtcIsacfix_NormLatticeFilterMa(size_t orderCoef,
                                        int32_t* stateGQ15,
                                        int16_t* lat_inQ0,
                                        int16_t* filt_coefQ15,
@@ -149,7 +149,7 @@ void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
                                        int16_t lo_hi,
                                        int16_t* lat_outQ9);
 
-void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
+void WebRtcIsacfix_NormLatticeFilterAr(size_t orderCoef,
                                        int16_t* stateGQ0,
                                        int32_t* lat_inQ25,
                                        int16_t* filt_coefQ15,
@@ -174,7 +174,7 @@ void WebRtcIsacfix_FilterMaLoopC(int16_t input0,
                                  int32_t* ptr1,
                                  int32_t* ptr2);
 
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 int WebRtcIsacfix_AutocorrNeon(int32_t* __restrict r,
                                const int16_t* __restrict x,
                                int16_t N,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
index 714a897cdb..e3de437a58 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -27,14 +27,14 @@
 
 
 
-int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
-                                 IsacFixDecoderInstance *ISACdec_obj,
-                                 int16_t       *current_framesamples)
+int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
+                             IsacFixDecoderInstance* ISACdec_obj,
+                             size_t* current_framesamples)
 {
   int k;
   int err;
   int16_t BWno;
-  int16_t len = 0;
+  int len = 0;
 
   int16_t model;
 
@@ -58,9 +58,9 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
   int16_t gainQ13;
 
 
-  int16_t frame_nb; /* counter */
-  int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */
-  static const int16_t kProcessedSamples = 480; /* 480 (for both 30, 60 ms) */
+  size_t frame_nb; /* counter */
+  size_t frame_mode; /* 0 for 30ms, 1 for 60ms */
+  static const size_t kProcessedSamples = 480; /* 480 (for both 30, 60 ms) */
 
   /* PLC */
   int16_t overlapWin[ 240 ];
@@ -130,14 +130,15 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
       ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX;    /* DECAY_RATE is in Q15 */
       ISACdec_obj->plcstr_obj.pitchCycles = 0;
 
-      PitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
+      PitchGains_Q12[0] = (int16_t)(PitchGains_Q12[0] * 700 >> 10);
 
       /* ---- Add-overlap ---- */
       WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP );
       for( k = 0; k < RECOVERY_OVERLAP; k++ )
         Vector_Word16_1[k] = WebRtcSpl_AddSatW16(
-            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
-            (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
+            (int16_t)(ISACdec_obj->plcstr_obj.overlapLP[k] *
+                overlapWin[RECOVERY_OVERLAP - k - 1] >> 14),
+            (int16_t)(Vector_Word16_1[k] * overlapWin[k] >> 14));
 
 
 
@@ -176,7 +177,7 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t       *signal_out16,
 
     /* reduce gain to compensate for pitch enhancer */
     /* gain = 1.0f - 0.45f * AvgPitchGain; */
-    tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(AvgPitchGain_Q12, 29, 0); // Q18
+    tmp32a = AvgPitchGain_Q12 * 29;  // Q18
     gainQ13 = (int16_t)((262144 - tmp32a) >> 5);  // Q18 -> Q13.
 
     for (k = 0; k < FRAMESAMPLES/2; k++)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
index b1f5d10a65..316f59a5e2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
@@ -26,13 +26,13 @@
 
 int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
                                     Bitstr_dec  *streamdata,
-                                    int32_t  packet_size,
+                                    size_t packet_size,
                                     uint16_t rtp_seq_number,
                                     uint32_t send_ts,
                                     uint32_t arr_ts)
 {
   int16_t index;
-  int16_t frame_samples;
+  size_t frame_samples;
   int err;
 
   /* decode framelength */
@@ -53,10 +53,10 @@ int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
   err = WebRtcIsacfix_UpdateUplinkBwImpl(
       bwest_str,
       rtp_seq_number,
-      frame_samples * 1000 / FS,
+      (int16_t)(frame_samples * 1000 / FS),
       send_ts,
       arr_ts,
-      (int16_t) packet_size,  /* in bytes */
+      packet_size,  /* in bytes */
       index);
 
   /* error check */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
index 4243ea3874..e907f2b6a6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
@@ -72,7 +72,7 @@ static int16_t plc_filterma_Fast(
     o >>= rshift;
 
     /* decay the output signal; this is specific to plc */
-    *Out++ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (int16_t)o, decay, 15); // ((o + (int32_t)2048) >> 12);
+    *Out++ = (int16_t)((int16_t)o * decay >> 15);
 
     /* change the decay */
     decay -= reduceDecay;
@@ -139,7 +139,7 @@ static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B )
          x*15 + (x*983)/(2^12); note that 983/2^12 = 0.23999     */
 
       /* we are sure that x is in the range of int16_t            */
-      x = (int16_t)(in * 15 + WEBRTC_SPL_MUL_16_16_RSFT(in, 983, 12));
+      x = (int16_t)(in * 15 + (in * 983 >> 12));
       /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
          be in correct domain and one more for the division by 2 */
       *B = (int16_t)((x * x + 0x00010000) >> 17);
@@ -157,7 +157,7 @@ static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B )
     {
       /* This is a mirror case of the above */
       in = 4300 - in;
-      x = (int16_t)(in * 15 + WEBRTC_SPL_MUL_16_16_RSFT(in, 983, 12));
+      x = (int16_t)(in * 15 + (in * 983 >> 12));
       /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
          be in correct domain and one more for the division by 2 */
       *A = (int16_t)((x * x + 0x00010000) >> 17);
@@ -175,10 +175,14 @@ static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B )
 
 
 
-static void LinearResampler( int16_t *in, int16_t *out, int16_t lenIn, int16_t lenOut )
+static void LinearResampler(int16_t* in,
+                            int16_t* out,
+                            size_t lenIn,
+                            size_t lenOut)
 {
-  int32_t n = (lenIn - 1) * RESAMP_RES;
-  int16_t resOut, i, j, relativePos, diff; /* */
+  size_t n = (lenIn - 1) * RESAMP_RES;
+  int16_t resOut, relativePos, diff; /* */
+  size_t i, j;
   uint16_t udiff;
 
   if( lenIn == lenOut )
@@ -187,7 +191,7 @@ static void LinearResampler( int16_t *in, int16_t *out, int16_t lenIn, int16_t l
     return;
   }
 
-  resOut = WebRtcSpl_DivW32W16ResW16( n, (int16_t)(lenOut-1) );
+  resOut = WebRtcSpl_DivW32W16ResW16( (int32_t)n, (int16_t)(lenOut-1) );
 
   out[0] = in[0];
   for( i = 1, j = 0, relativePos = 0; i < lenOut; i++ )
@@ -220,7 +224,7 @@ static void LinearResampler( int16_t *in, int16_t *out, int16_t lenIn, int16_t l
       else
       {
         diff = in[ j + 1 ] - in[ j ];
-        out[ i ] = in[ j ] + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( diff, relativePos, RESAMP_RES_BIT );
+        out[i] = in[j] + (int16_t)(diff * relativePos >> RESAMP_RES_BIT);
       }
     }
   }
@@ -230,12 +234,11 @@ static void LinearResampler( int16_t *in, int16_t *out, int16_t lenIn, int16_t l
 
 
 
-int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
-                                    IsacFixDecoderInstance *ISACdec_obj,
-                                    int16_t *current_framesamples )
+void WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
+                                 IsacFixDecoderInstance *ISACdec_obj,
+                                 size_t *current_framesamples )
 {
   int subframecnt;
-  int16_t len = 0;
 
   int16_t* Vector_Word16_1;
   int16_t  Vector_Word16_Extended_1[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
@@ -258,12 +261,14 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
   int16_t myDecayRate;
 
   /* ---------- PLC variables ------------ */
-  int16_t lag0, i, k, noiseIndex;
+  size_t lag0, i, k;
+  int16_t noiseIndex;
   int16_t stretchPitchLP[PITCH_MAX_LAG + 10], stretchPitchLP1[PITCH_MAX_LAG + 10];
 
   int32_t gain_lo_hiQ17[2*SUBFRAMES];
 
-  int16_t nLP, pLP, wNoisyLP, wPriodicLP, tmp16, minIdx;
+  int16_t nLP, pLP, wNoisyLP, wPriodicLP, tmp16;
+  size_t minIdx;
   int32_t nHP, pHP, wNoisyHP, wPriodicHP, corr, minCorr, maxCoeff;
   int16_t noise1, rshift;
 
@@ -298,7 +303,7 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
 
 
-  lag0 = ((ISACdec_obj->plcstr_obj.lastPitchLag_Q7 + 64) >> 7) + 1;
+  lag0 = (size_t)(((ISACdec_obj->plcstr_obj.lastPitchLag_Q7 + 64) >> 7) + 1);
 
 
   if( (ISACdec_obj->plcstr_obj).used != PLC_WAS_USED )
@@ -309,7 +314,7 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
         &((ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0]);
     minCorr = WEBRTC_SPL_WORD32_MAX;
 
-    if ( (FRAMESAMPLES_HALF - 2*lag0 - 10) > 0 )
+    if ((FRAMESAMPLES_HALF - 10) > 2 * lag0)
     {
       minIdx = 11;
       for( i = 0; i < 21; i++ )
@@ -447,14 +452,11 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
   /* inverse pitch filter */
 
   pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] =
-      ((ISACdec_obj->plcstr_obj).stretchLag<<7);
+      (int16_t)((ISACdec_obj->plcstr_obj).stretchLag<<7);
   pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12);
-  pitchGains_Q12[2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-      pitchGains_Q12[3], 1010, 10 );
-  pitchGains_Q12[1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-      pitchGains_Q12[2], 1010, 10 );
-  pitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-      pitchGains_Q12[1], 1010, 10 );
+  pitchGains_Q12[2] = (int16_t)(pitchGains_Q12[3] * 1010 >> 10);
+  pitchGains_Q12[1] = (int16_t)(pitchGains_Q12[2] * 1010 >> 10);
+  pitchGains_Q12[0] = (int16_t)(pitchGains_Q12[1] * 1010 >> 10);
 
 
   /* most of the time either B or A are zero so seperating */
@@ -527,9 +529,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
       for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
       {
         /* --- Lowpass                                               */
-        pLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-            stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
-            (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
+        pLP = (int16_t)(stretchPitchLP[ISACdec_obj->plcstr_obj.pitchIndex] *
+            ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15);
 
         /* --- Highpass                                              */
         pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
@@ -626,9 +627,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
         noise1 = (ISACdec_obj->plcstr_obj.seed >> 10) - 16;
 
-        nLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-            (int16_t)((noise1)*(ISACdec_obj->plcstr_obj).std),
-            (ISACdec_obj->plcstr_obj).decayCoeffNoise, 15 );
+        nLP = (int16_t)((int16_t)(noise1 * ISACdec_obj->plcstr_obj.std) *
+            ISACdec_obj->plcstr_obj.decayCoeffNoise >> 15);
 
         /* --- Highpass                                              */
         (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
@@ -646,9 +646,8 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
         /* ------ Periodic Vector ---                                */
         /* --- Lowpass                                               */
-        pLP = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-            stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
-            (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
+        pLP = (int16_t)(stretchPitchLP[ISACdec_obj->plcstr_obj.pitchIndex] *
+            ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15);
 
         /* --- Highpass                                              */
         pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15(
@@ -665,13 +664,11 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
         }
 
         /* ------ Weighting the noisy and periodic vectors -------   */
-        wNoisyLP = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT(
-            (ISACdec_obj->plcstr_obj).A, nLP, 15 ) );
+        wNoisyLP = (int16_t)(ISACdec_obj->plcstr_obj.A * nLP >> 15);
         wNoisyHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).A, (nHP) ) );
 
-        wPriodicLP = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT(
-            (ISACdec_obj->plcstr_obj).B, pLP, 15));
+        wPriodicLP = (int16_t)(ISACdec_obj->plcstr_obj.B * pLP >> 15);
         wPriodicHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15(
             (ISACdec_obj->plcstr_obj).B, pHP));
 
@@ -752,13 +749,13 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
   for( i = 0; i < RECOVERY_OVERLAP; i++ )
   {
-    (ISACdec_obj->plcstr_obj).overlapLP[i] = (int16_t)(
-        WEBRTC_SPL_MUL_16_16_RSFT(stretchPitchLP[k],
-                                  (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15) );
+    ISACdec_obj->plcstr_obj.overlapLP[i] = (int16_t)(
+        stretchPitchLP[k] * ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15);
     k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0;
   }
 
-  (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = (ISACdec_obj->plcstr_obj).stretchLag << 7;
+  (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 =
+      (int16_t)((ISACdec_obj->plcstr_obj).stretchLag << 7);
 
 
   /* --- Inverse Pitch Filter --- */
@@ -767,8 +764,7 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
   /* reduce gain to compensate for pitch enhancer */
   /* gain = 1.0f - 0.45f * AvgPitchGain; */
-  tmp32a = WEBRTC_SPL_MUL_16_16_RSFT((ISACdec_obj->plcstr_obj).AvgPitchGain_Q12,
-                                     29, 0); // Q18
+  tmp32a = ISACdec_obj->plcstr_obj.AvgPitchGain_Q12 * 29;  // Q18
   tmp32b = 262144 - tmp32a;  // Q18
   gainQ13 = (int16_t) (tmp32b >> 5); // Q13
 
@@ -806,6 +802,4 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
 
   (ISACdec_obj->plcstr_obj).used = PLC_WAS_USED;
   *current_framesamples = 480;
-
-  return len;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
index d3e7107b6e..757c0b85c8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -115,8 +115,9 @@ int WebRtcIsacfix_EncodeImpl(int16_t      *in,
 
     // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT
     // 901/1024 is 0.87988281250000
-    ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr((int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ISACenc_obj->BottleNeck, 901, 10),
-                                             ISACenc_obj->current_framesamples);
+    ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr(
+        (int16_t)(ISACenc_obj->BottleNeck * 901 >> 10),
+        ISACenc_obj->current_framesamples);
 
     /* encode frame length */
     status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj);
@@ -352,8 +353,8 @@ int WebRtcIsacfix_EncodeImpl(int16_t      *in,
     // scale FFT coefficients to reduce the bit-rate
     for(k = 0; k < FRAMESAMPLES_HALF; k++)
     {
-      LP16a[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(LP16a[k], scaleQ14[idx], 14);
-      LPandHP[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(LPandHP[k], scaleQ14[idx], 14);
+      LP16a[k] = (int16_t)(LP16a[k] * scaleQ14[idx] >> 14);
+      LPandHP[k] = (int16_t)(LPandHP[k] * scaleQ14[idx] >> 14);
     }
 
     // Save data for multiple packets memory
@@ -497,7 +498,7 @@ int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance  *ISACenc_obj,
 {
   int ii;
   int status;
-  int16_t BWno = BWnumber;
+  int16_t BWno = (int16_t)BWnumber;
   int stream_length = 0;
 
   int16_t model;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index c150e60692..2379ba5066 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -110,7 +110,7 @@ static int16_t CalcLogN(int32_t arg) {
   zeros=WebRtcSpl_NormU32(arg);
   frac = (int16_t)((uint32_t)((arg << zeros) & 0x7FFFFFFF) >> 23);
   log2 = (int16_t)(((31 - zeros) << 8) + frac);  // log2(x) in Q8
-  logN=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(log2,22713,15); //Q8*Q15 log(2) = 0.693147 = 22713 in Q15
+  logN = (int16_t)(log2 * 22713 >> 15);  // log(2) = 0.693147 = 22713 in Q15
   logN=logN+11; //Scalar compensation which minimizes the (log(x)-logN(x))^2 error over all x.
 
   return logN;
@@ -129,13 +129,12 @@ static int16_t CalcLogN(int32_t arg) {
 */
 
 static int32_t CalcExpN(int16_t x) {
-  int16_t ax, axINT, axFRAC;
+  int16_t axINT, axFRAC;
   int16_t exp16;
   int32_t exp;
+  int16_t ax = (int16_t)(x * 23637 >> 14);  // Q8
 
   if (x>=0) {
-    //  ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637-700, 14); //Q8
-    ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
     axINT = ax >> 8;  //Q0
     axFRAC = ax&0x00FF;
     exp16 = 1 << axINT;  // Q0
@@ -143,8 +142,6 @@ static int32_t CalcExpN(int16_t x) {
     exp = exp16 * axFRAC;  // Q0*Q8 = Q8
     exp <<= 9;  // Q17
   } else {
-    //  ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637+700, 14); //Q8
-    ax=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
     ax = -ax;
     axINT = 1 + (ax >> 8);  //Q0
     axFRAC = 0x00FF - (ax&0x00FF);
@@ -453,10 +450,10 @@ static void GenerateDitherQ7(int16_t *bufQ7,
  * function to decode the complex spectrum from the bitstream
  * returns the total number of bytes in the stream
  */
-int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
-                                 int16_t *frQ7,
-                                 int16_t *fiQ7,
-                                 int16_t AvgPitchGain_Q12)
+int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
+                             int16_t *frQ7,
+                             int16_t *fiQ7,
+                             int16_t AvgPitchGain_Q12)
 {
   int16_t  data[FRAMESAMPLES];
   int32_t  invARSpec2_Q16[FRAMESAMPLES/4];
@@ -464,7 +461,7 @@ int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
   int16_t  RCQ15[AR_ORDER];
   int16_t  gainQ10;
   int32_t  gain2_Q10;
-  int16_t  len;
+  int len;
   int          k;
 
   /* create dither signal */
@@ -679,16 +676,16 @@ static void Rc2LarFix(const int16_t *rcQ15, int32_t *larQ17, int16_t order) {
 
     if (rc<24956) {  //0.7615966 in Q15
       // (Q15*Q13)>>11 = Q17
-      larAbsQ17 = WEBRTC_SPL_MUL_16_16_RSFT(rc, 21512, 11);
+      larAbsQ17 = rc * 21512 >> 11;
     } else if (rc<30000) { //0.91552734375 in Q15
       // Q17 + (Q15*Q12)>>10 = Q17
-      larAbsQ17 = -465024 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 29837, 10);
+      larAbsQ17 = -465024 + (rc * 29837 >> 10);
     } else if (rc<32500) { //0.99182128906250 in Q15
       // Q17 + (Q15*Q10)>>8 = Q17
-      larAbsQ17 = -3324784 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 31863, 8);
+      larAbsQ17 = -3324784 + (rc * 31863 >> 8);
     } else  {
       // Q17 + (Q15*Q5)>>3 = Q17
-      larAbsQ17 = -88546020 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 21973, 3);
+      larAbsQ17 = -88546020 + (rc * 21973 >> 3);
     }
 
     if (rcQ15[k]>0) {
@@ -717,7 +714,7 @@ static void Lar2RcFix(const int32_t *larQ17, int16_t *rcQ15,  int16_t order) {
 
     if (larAbsQ11<4097) { //2.000012018559 in Q11
       // Q11*Q16>>12 = Q15
-      rc = WEBRTC_SPL_MUL_16_16_RSFT(larAbsQ11, 24957, 12);
+      rc = larAbsQ11 * 24957 >> 12;
     } else if (larAbsQ11<6393) { //3.121320351712 in Q11
       // (Q11*Q17 + Q13)>>13 = Q15
       rc = (larAbsQ11 * 17993 + 130738688) >> 13;
@@ -995,7 +992,8 @@ int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
       pos = LPC_SHAPE_ORDER * j;
       pos2 = LPC_SHAPE_ORDER * k;
       for (n=0; n<LPC_SHAPE_ORDER; n++) {
-        sumQQ += WEBRTC_SPL_MUL_16_16_RSFT(tmpcoeffs_sQ10[pos], WebRtcIsacfix_kT1ShapeQ15[model][pos2], 7); // (Q10*Q15)>>7 = Q18
+        sumQQ += tmpcoeffs_sQ10[pos] *
+            WebRtcIsacfix_kT1ShapeQ15[model][pos2] >> 7;  // (Q10*Q15)>>7 = Q18
         pos++;
         pos2++;
       }
@@ -1609,7 +1607,7 @@ int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12,
 
   /* get the approximate arcsine (almost linear)*/
   for (k=0; k<PITCH_SUBFRAMES; k++)
-    SQ15[k] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[k],33,2); //Q15
+    SQ15[k] = (int16_t)(PitchGains_Q12[k] * 33 >> 2);  // Q15
 
 
   /* find quantization index; only for the first three transform coefficients */
@@ -1618,7 +1616,7 @@ int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12,
     /*  transform */
     CQ17=0;
     for (j=0; j<PITCH_SUBFRAMES; j++) {
-      CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], SQ15[j],10); // Q17
+      CQ17 += WebRtcIsacfix_kTransform[k][j] * SQ15[j] >> 10;  // Q17
     }
 
     index[k] = (int16_t)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8)
@@ -1677,7 +1675,7 @@ int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
 
   int32_t meangainQ12;
   int32_t CQ11, CQ10,tmp32a,tmp32b;
-  int16_t shft,tmp16a,tmp16c;
+  int16_t shft;
 
   meangainQ12=0;
   for (k = 0; k < 4; k++)
@@ -1727,22 +1725,19 @@ int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
   CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11
   for (k=0; k<PITCH_SUBFRAMES; k++) {
     tmp32a =  WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11);
-    tmp16a = (int16_t)(tmp32a >> 5);
-    PitchLags_Q7[k] = tmp16a;
+    PitchLags_Q7[k] = (int16_t)(tmp32a >> 5);
   }
 
   CQ10 = mean_val2Q10[index[1]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[1][k], (int16_t) CQ10,10);
-    tmp16c = (int16_t)(tmp32b >> 5);
-    PitchLags_Q7[k] += tmp16c;
+    tmp32b = WebRtcIsacfix_kTransform[1][k] * (int16_t)CQ10 >> 10;
+    PitchLags_Q7[k] += (int16_t)(tmp32b >> 5);
   }
 
   CQ10 = mean_val4Q10[index[3]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[3][k], (int16_t) CQ10,10);
-    tmp16c = (int16_t)(tmp32b >> 5);
-    PitchLags_Q7[k] += tmp16c;
+    tmp32b = WebRtcIsacfix_kTransform[3][k] * (int16_t)CQ10 >> 10;
+    PitchLags_Q7[k] += (int16_t)(tmp32b >> 5);
   }
 
   return 0;
@@ -1763,7 +1758,7 @@ int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7,
   const int16_t *mean_val2Q10,*mean_val4Q10;
   const int16_t *lower_limit, *upper_limit;
   const uint16_t **cdf;
-  int16_t shft, tmp16a, tmp16b, tmp16c;
+  int16_t shft, tmp16b;
   int32_t tmp32b;
   int status = 0;
 
@@ -1809,7 +1804,7 @@ int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7,
     /*  transform */
     CQ17=0;
     for (j=0; j<PITCH_SUBFRAMES; j++)
-      CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], PitchLagsQ7[j],2); // Q17
+      CQ17 += WebRtcIsacfix_kTransform[k][j] * PitchLagsQ7[j] >> 2;  // Q17
 
     CQ17 = WEBRTC_SPL_SHIFT_W32(CQ17,shft); // Scale with StepSize
 
@@ -1834,22 +1829,19 @@ int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7,
 
   for (k=0; k<PITCH_SUBFRAMES; k++) {
     tmp32a =  WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11); // Q12
-    tmp16a = (int16_t)(tmp32a >> 5);  // Q7.
-    PitchLagsQ7[k] = tmp16a;
+    PitchLagsQ7[k] = (int16_t)(tmp32a >> 5);  // Q7.
   }
 
   CQ10 = mean_val2Q10[index[1]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[1][k], (int16_t) CQ10,10);
-    tmp16c = (int16_t)(tmp32b >> 5);  // Q7.
-    PitchLagsQ7[k] += tmp16c;
+    tmp32b = WebRtcIsacfix_kTransform[1][k] * (int16_t)CQ10 >> 10;
+    PitchLagsQ7[k] += (int16_t)(tmp32b >> 5);  // Q7.
   }
 
   CQ10 = mean_val4Q10[index[3]];
   for (k=0; k<PITCH_SUBFRAMES; k++) {
-    tmp32b =  (int32_t) WEBRTC_SPL_MUL_16_16_RSFT((int16_t) WebRtcIsacfix_kTransform[3][k], (int16_t) CQ10,10);
-    tmp16c = (int16_t)(tmp32b >> 5);  // Q7.
-    PitchLagsQ7[k] += tmp16c;
+    tmp32b = WebRtcIsacfix_kTransform[3][k] * (int16_t)CQ10 >> 10;
+    PitchLagsQ7[k] += (int16_t)(tmp32b >> 5);  // Q7.
   }
 
   /* entropy coding of quantization pitch lags */
@@ -1878,7 +1870,7 @@ const uint16_t kFrameLenInitIndex[1] = {1};
 
 
 int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
-                                 int16_t *framesamples)
+                                 size_t *framesamples)
 {
 
   int err;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
index 741646f01a..2c8c923cd3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
@@ -22,10 +22,10 @@
 #include "structs.h"
 
 /* decode complex spectrum (return number of bytes in stream) */
-int16_t WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
-                                 int16_t *frQ7,
-                                 int16_t *fiQ7,
-                                 int16_t AvgPitchGain_Q12);
+int WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
+                             int16_t *frQ7,
+                             int16_t *fiQ7,
+                             int16_t AvgPitchGain_Q12);
 
 /* encode complex spectrum */
 int WebRtcIsacfix_EncodeSpec(const int16_t *fr,
@@ -92,7 +92,7 @@ int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
                                  int16_t *PitchLagQ7);
 
 int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
-                                 int16_t *framelength);
+                                 size_t *framelength);
 
 
 int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
@@ -147,7 +147,7 @@ void WebRtcIsacfix_MatrixProduct2C(const int16_t matrix0[],
                                    const int matrix0_index_factor,
                                    const int matrix0_index_step);
 
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
                                       const int32_t matrix1[],
                                       int32_t matrix_product[],
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
index 2aa587fc0d..0e67e300ac 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
@@ -60,7 +60,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16C(
    int32_t *filter_state_ch1,
    int32_t *filter_state_ch2);
 
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
 void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
    int16_t *data_ch1,
    int16_t *data_ch2,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.S
deleted file mode 100644
index 0a43551ad0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.S
+++ /dev/null
@@ -1,270 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS.  All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ Contains a function for WebRtcIsacfix_AllpassFilter2FixDec16Neon()
-@ in iSAC codec, optimized for ARM Neon platform. Bit exact with function
-@ WebRtcIsacfix_AllpassFilter2FixDec16Neon() in filterbanks.c. Prototype
-@ C code is at end of this file.
-
-#include "webrtc/system_wrappers/interface/asm_defines.h"
-
-GLOBAL_FUNCTION WebRtcIsacfix_AllpassFilter2FixDec16Neon
-.align  2
-
-@void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
-@    int16_t *data_ch1,  // Input and output in channel 1, in Q0
-@    int16_t *data_ch2,  // Input and output in channel 2, in Q0
-@    const int16_t *factor_ch1,  // Scaling factor for channel 1, in Q15
-@    const int16_t *factor_ch2,  // Scaling factor for channel 2, in Q15
-@    const int length,           // Length of the data buffers
-@    int32_t *filter_state_ch1,  // Filter state for channel 1, in Q16
-@    int32_t *filter_state_ch2); // Filter state for channel 2, in Q16
-
-DEFINE_FUNCTION WebRtcIsacfix_AllpassFilter2FixDec16Neon
-  push {r4 - r7}
-
-  ldr r5, [sp, #24]           @ filter_state_ch2
-  ldr r6, [sp, #20]           @ filter_state_ch1
-
-  @ Initialize the Neon registers.
-  vld1.16 d0[0], [r0]!        @ data_ch1[0]
-  vld1.16 d0[2], [r1]!        @ data_ch2[0]
-  vld1.32 d30[0], [r2]        @ factor_ch1[0], factor_ch1[1]
-  vld1.32 d30[1], [r3]        @ factor_ch2[0], factor_ch2[1]
-  vld1.32 d16[0], [r6]!       @ filter_state_ch1[0]
-  vld1.32 d17[0], [r5]!       @ filter_state_ch2[0]
-  vneg.s16 d31, d30
-
-  ldr r3, [sp, #16]           @ length
-  mov r4, #4                  @ Post offset value for the loop
-  mov r2, #-2                 @ Post offset value for the loop
-  sub r3, #2                  @ Loop counter
-
-  @ Loop unrolling pre-processing.
-  vqdmull.s16 q1, d30, d0
-  vshll.s16 q0, d0, #16
-  vqadd.s32 q2, q1, q8
-  vshrn.i32 d6, q2, #16
-  vmull.s16 q1, d31, d6
-  vshl.s32 q1, #1
-  vqadd.s32 q8, q1, q0
-  vld1.32 d16[1], [r6]        @ filter_state_ch1[1]
-  vld1.32 d17[1], [r5]        @ filter_state_ch2[1]
-  sub r6, #4                  @ &filter_state_ch1[0]
-  sub r5, #4                  @ &filter_state_ch2[0]
-  vld1.16 d6[1], [r0], r2     @ data_ch1[1]
-  vld1.16 d6[3], [r1], r2     @ data_ch2[1]
-  vrev32.16 d0, d6
-
-FOR_LOOP:
-  vqdmull.s16 q1, d30, d0
-  vshll.s16 q0, d0, #16
-  vqadd.s32 q2, q1, q8
-  vshrn.i32 d4, q2, #16
-  vmull.s16 q1, d31, d4
-  vst1.16 d4[1], [r0], r4     @ Store data_ch1[n]
-  vst1.16 d4[3], [r1], r4     @ Store data_ch2[n]
-  vshl.s32 q1, #1
-  vld1.16 d4[1], [r0], r2     @ Load data_ch1[n + 2]
-  vld1.16 d4[3], [r1], r2     @ Load data_ch2[n + 2]
-  vqadd.s32 q8, q1, q0
-  vrev32.16 d0, d4
-  vqdmull.s16 q1, d30, d0
-  subs r3, #2
-  vqadd.s32 q2, q1, q8
-  vshrn.i32 d6, q2, #16
-  vmull.s16 q1, d31, d6
-  vshll.s16 q0, d0, #16
-  vst1.16 d6[1], [r0], r4     @ Store data_ch1[n + 1]
-  vst1.16 d6[3], [r1], r4     @ Store data_ch2[n + 1]
-  vshl.s32 q1, #1
-  vld1.16 d6[1], [r0], r2     @ Load data_ch1[n + 3]
-  vld1.16 d6[3], [r1], r2     @ Load data_ch2[n + 3]
-  vqadd.s32 q8, q1, q0
-  vrev32.16 d0, d6
-  bgt FOR_LOOP
-
-  @ Loop unrolling post-processing.
-  vqdmull.s16 q1, d30, d0
-  vshll.s16 q0, d0, #16
-  vqadd.s32 q2, q1, q8
-  vshrn.i32 d4, q2, #16
-  vmull.s16 q1, d31, d4
-  vst1.16 d4[1], [r0]!        @ Store data_ch1[n]
-  vst1.16 d4[3], [r1]!        @ Store data_ch2[n]
-  vshl.s32 q1, #1
-  vqadd.s32 q8, q1, q0
-  vrev32.16 d0, d4
-  vqdmull.s16 q1, d30, d0
-  vshll.s16 q0, d0, #16
-  vqadd.s32 q2, q1, q8
-  vshrn.i32 d6, q2, #16
-  vmull.s16 q1, d31, d6
-  vst1.16 d6[1], [r0]         @ Store data_ch1[n + 1]
-  vst1.16 d6[3], [r1]         @ Store data_ch2[n + 1]
-  vshl.s32 q1, #1
-  vst1.32 d16[0], [r6]!       @ Store filter_state_ch1[0]
-  vqadd.s32 q9, q1, q0
-  vst1.32 d17[0], [r5]!       @ Store filter_state_ch1[1]
-  vst1.32 d18[1], [r6]        @ Store filter_state_ch2[0]
-  vst1.32 d19[1], [r5]        @ Store filter_state_ch2[1]
-
-  pop {r4 - r7}
-  bx lr
-
-@void AllpassFilter2FixDec16BothChannels(
-@    int16_t *data_ch1,  // Input and output in channel 1, in Q0
-@    int16_t *data_ch2,  // Input and output in channel 2, in Q0
-@    const int16_t *factor_ch1,  // Scaling factor for channel 1, in Q15
-@    const int16_t *factor_ch2,  // Scaling factor for channel 2, in Q15
-@    const int length,  // Length of the data buffers
-@    int32_t *filter_state_ch1,  // Filter state for channel 1, in Q16
-@    int32_t *filter_state_ch2) {  // Filter state for channel 2, in Q16
-@  int n = 0;
-@  int32_t state0_ch1 = filter_state_ch1[0], state1_ch1 = filter_state_ch1[1];
-@  int32_t state0_ch2 = filter_state_ch2[0], state1_ch2 = filter_state_ch2[1];
-@  int16_t sample0_ch1 = 0, sample0_ch2 = 0;
-@  int16_t sample1_ch1 = 0, sample1_ch2  = 0;
-@  int32_t a0_ch1 = 0, a0_ch2 = 0;
-@  int32_t b0_ch1 = 0, b0_ch2 = 0;
-@
-@  int32_t a1_ch1 = 0, a1_ch2 = 0;
-@  int32_t b1_ch1 = 0, b1_ch2 = 0;
-@  int32_t b2_ch1  = 0, b2_ch2 = 0;
-@
-@  // Loop unrolling preprocessing.
-@
-@  sample0_ch1 = data_ch1[n];
-@  sample0_ch2 = data_ch2[n];
-@
-@  a0_ch1 = (factor_ch1[0] * sample0_ch1) << 1;
-@  a0_ch2 = (factor_ch2[0] * sample0_ch2) << 1;
-@
-@  b0_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state0_ch1);
-@  b0_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state0_ch2); //Q16+Q16=Q16
-@
-@  a0_ch1 = -factor_ch1[0] * (int16_t)(b0_ch1 >> 16);
-@  a0_ch2 = -factor_ch2[0] * (int16_t)(b0_ch2 >> 16);
-@
-@  state0_ch1 = WebRtcSpl_AddSatW32(a0_ch1 <<1, (uint32_t)sample0_ch1 << 16);
-@  state0_ch2 = WebRtcSpl_AddSatW32(a0_ch2 <<1, (uint32_t)sample0_ch2 << 16);
-@
-@  sample1_ch1 = data_ch1[n + 1];
-@  sample0_ch1 = (int16_t) (b0_ch1 >> 16); //Save as Q0
-@  sample1_ch2  = data_ch2[n + 1];
-@  sample0_ch2 = (int16_t) (b0_ch2 >> 16); //Save as Q0
-@
-@
-@  for (n = 0; n < length - 2; n += 2) {
-@    a1_ch1 = (factor_ch1[0] * sample1_ch1) << 1;
-@    a0_ch1 = (factor_ch1[1] * sample0_ch1) << 1;
-@    a1_ch2 = (factor_ch2[0] * sample1_ch2) << 1;
-@    a0_ch2 = (factor_ch2[1] * sample0_ch2) << 1;
-@
-@    b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state0_ch1);
-@    b0_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state1_ch1); //Q16+Q16=Q16
-@    b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state0_ch2); //Q16+Q16=Q16
-@    b0_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state1_ch2); //Q16+Q16=Q16
-@
-@    a1_ch1 = -factor_ch1[0] * (int16_t)(b1_ch1 >> 16);
-@    a0_ch1 = -factor_ch1[1] * (int16_t)(b0_ch1 >> 16);
-@    a1_ch2 = -factor_ch2[0] * (int16_t)(b1_ch2 >> 16);
-@    a0_ch2 = -factor_ch2[1] * (int16_t)(b0_ch2 >> 16);
-@
-@    state0_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1 <<16);
-@    state1_ch1 = WebRtcSpl_AddSatW32(a0_ch1<<1, (uint32_t)sample0_ch1 <<16);
-@    state0_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2 <<16);
-@    state1_ch2 = WebRtcSpl_AddSatW32(a0_ch2<<1, (uint32_t)sample0_ch2 <<16);
-@
-@    sample0_ch1 = data_ch1[n + 2];
-@    sample1_ch1 = (int16_t) (b1_ch1 >> 16); //Save as Q0
-@    sample0_ch2 = data_ch2[n + 2];
-@    sample1_ch2  = (int16_t) (b1_ch2 >> 16); //Save as Q0
-@
-@    a0_ch1 = (factor_ch1[0] * sample0_ch1) << 1;
-@    a1_ch1 = (factor_ch1[1] * sample1_ch1) << 1;
-@    a0_ch2 = (factor_ch2[0] * sample0_ch2) << 1;
-@    a1_ch2 = (factor_ch2[1] * sample1_ch2) << 1;
-@
-@    b2_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state0_ch1);
-@    b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state1_ch1); //Q16+Q16=Q16
-@    b2_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state0_ch2); //Q16+Q16=Q16
-@    b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state1_ch2); //Q16+Q16=Q16
-@
-@    a0_ch1 = -factor_ch1[0] * (int16_t)(b2_ch1 >> 16);
-@    a1_ch1 = -factor_ch1[1] * (int16_t)(b1_ch1 >> 16);
-@    a0_ch2 = -factor_ch2[0] * (int16_t)(b2_ch2 >> 16);
-@    a1_ch2 = -factor_ch2[1] * (int16_t)(b1_ch2 >> 16);
-@
-@    state0_ch1 = WebRtcSpl_AddSatW32(a0_ch1<<1, (uint32_t)sample0_ch1<<16);
-@    state1_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1<<16);
-@    state0_ch2 = WebRtcSpl_AddSatW32(a0_ch2<<1, (uint32_t)sample0_ch2<<16);
-@    state1_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2<<16);
-@
-@
-@    sample1_ch1 = data_ch1[n + 3];
-@    sample0_ch1 = (int16_t) (b2_ch1  >> 16); //Save as Q0
-@    sample1_ch2 = data_ch2[n + 3];
-@    sample0_ch2 = (int16_t) (b2_ch2 >> 16); //Save as Q0
-@
-@    data_ch1[n]     = (int16_t) (b0_ch1 >> 16); //Save as Q0
-@    data_ch1[n + 1] = (int16_t) (b1_ch1 >> 16); //Save as Q0
-@    data_ch2[n]     = (int16_t) (b0_ch2 >> 16);
-@    data_ch2[n + 1] = (int16_t) (b1_ch2 >> 16);
-@  }
-@
-@  // Loop unrolling post-processing.
-@
-@  a1_ch1 = (factor_ch1[0] * sample1_ch1) << 1;
-@  a0_ch1 = (factor_ch1[1] * sample0_ch1) << 1;
-@  a1_ch2 = (factor_ch2[0] * sample1_ch2) << 1;
-@  a0_ch2 = (factor_ch2[1] * sample0_ch2) << 1;
-@
-@  b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state0_ch1);
-@  b0_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state1_ch1);
-@  b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state0_ch2);
-@  b0_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state1_ch2);
-@
-@  a1_ch1 = -factor_ch1[0] * (int16_t)(b1_ch1 >> 16);
-@  a0_ch1 = -factor_ch1[1] * (int16_t)(b0_ch1 >> 16);
-@  a1_ch2 = -factor_ch2[0] * (int16_t)(b1_ch2 >> 16);
-@  a0_ch2 = -factor_ch2[1] * (int16_t)(b0_ch2 >> 16);
-@
-@  state0_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1 << 16);
-@  state1_ch1 = WebRtcSpl_AddSatW32(a0_ch1<<1, (uint32_t)sample0_ch1 << 16);
-@  state0_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2 << 16);
-@  state1_ch2 = WebRtcSpl_AddSatW32(a0_ch2<<1, (uint32_t)sample0_ch2 << 16);
-@
-@  data_ch1[n] = (int16_t) (b0_ch1 >> 16); //Save as Q0
-@  data_ch2[n] = (int16_t) (b0_ch2 >> 16);
-@
-@  sample1_ch1 = (int16_t) (b1_ch1 >> 16); //Save as Q0
-@  sample1_ch2  = (int16_t) (b1_ch2 >> 16); //Save as Q0
-@
-@  a1_ch1 = (factor_ch1[1] * sample1_ch1) << 1;
-@  a1_ch2 = (factor_ch2[1] * sample1_ch2) << 1;
-@
-@  b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state1_ch1); //Q16+Q16=Q16
-@  b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state1_ch2); //Q16+Q16=Q16
-@
-@  a1_ch1 = -factor_ch1[1] * (int16_t)(b1_ch1 >> 16);
-@  a1_ch2 = -factor_ch2[1] * (int16_t)(b1_ch2 >> 16);
-@
-@  state1_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1<<16);
-@  state1_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2<<16);
-@
-@  data_ch1[n + 1] = (int16_t) (b1_ch1 >> 16); //Save as Q0
-@  data_ch2[n + 1] = (int16_t) (b1_ch2 >> 16);
-@
-@  filter_state_ch1[0] = state0_ch1;
-@  filter_state_ch1[1] = state1_ch1;
-@  filter_state_ch2[0] = state0_ch2;
-@  filter_state_ch2[1] = state1_ch2;
-@}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
index 614e16982e..20f80aefec 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
@@ -29,12 +29,13 @@ void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
   int16x4_t factorv;
   int16x4_t datav;
   int32x4_t statev;
-  int32x2_t tmp;
 
   // Load factor_ch1 and factor_ch2.
-  tmp = vld1_dup_s32((int32_t*)factor_ch1);
-  tmp = vld1_lane_s32((int32_t*)factor_ch2, tmp, 1);
-  factorv = vreinterpret_s16_s32(tmp);
+  factorv = vld1_dup_s16(factor_ch1);
+  factorv = vld1_lane_s16(factor_ch1 + 1, factorv, 1);
+  factorv = vld1_lane_s16(factor_ch2, factorv, 2);
+  factorv = vld1_lane_s16(factor_ch2 + 1, factorv, 3);
+
   // Load filter_state_ch1[0] and filter_state_ch2[0].
   statev = vld1q_dup_s32(filter_state_ch1);
   statev = vld1q_lane_s32(filter_state_ch2, statev, 2);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
index 3276331e79..0ec115414b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
@@ -13,7 +13,7 @@
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 class FilterBanksTest : public testing::Test {
@@ -64,11 +64,11 @@ class FilterBanksTest : public testing::Test {
 
 TEST_F(FilterBanksTest, AllpassFilter2FixDec16Test) {
   CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16C);
-#ifdef WEBRTC_DETECT_ARM_NEON
+#ifdef WEBRTC_DETECT_NEON
   if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
     CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16Neon);
   }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16Neon);
 #endif
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.S
deleted file mode 100644
index 3c5ac646c7..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_neon.S
+++ /dev/null
@@ -1,145 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS.  All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-@ Reference code in filters.c. Output is bit-exact.
-
-#include "webrtc/system_wrappers/interface/asm_defines.h"
-
-GLOBAL_FUNCTION WebRtcIsacfix_AutocorrNeon
-.align  2
-
-@ int WebRtcIsacfix_AutocorrNeon(
-@     int32_t* __restrict r,
-@     const int16_t* __restrict x,
-@     int16_t N,
-@     int16_t order,
-@     int16_t* __restrict scale);
-
-DEFINE_FUNCTION WebRtcIsacfix_AutocorrNeon
-  push       {r3 - r12}
-
-  @ Constant initializations
-  mov        r4, #33
-  vmov.i32   d0, #0
-  vmov.i32   q8, #0
-  vmov.i32   d29, #0               @ Initialize (-scale).
-  vmov.u8    d30, #255             @ Initialize d30 as -1.
-  vmov.i32   d0[0], r4             @ d0: 00000033 (low), 00000000 (high)
-  vmov.i32   d25, #32
-
-  mov        r5, r1                @ x
-  mov        r6, r2                @ N
-
-@ Generate the first coefficient r0.
-LOOP_R0:
-  vld1.16    {d18}, [r5]!          @ x[]
-  subs       r6, r6, #4
-  vmull.s16  q9, d18, d18
-  vpadal.s32 q8, q9
-  bgt        LOOP_R0
-
-  vadd.i64   d16, d16, d17
-
-  @ Calculate scaling (the value of shifting).
-  vmov       d17, d16
-
-  @ Check overflow and determine the value for 'scale'.
-  @ vclz cannot deal with a 64-bit, so we have to do vclz on both the upper and
-  @ lower 32-bit words. Note that we don't care about the value of the upper
-  @ word in d17.
-
-  @ Check the case of 1 bit overflow. If it occurs store the results for
-  @ scale and r[0] in d17 and d29.
-
-  vshr.u64   d3, d16, #1
-  vclt.s32   d1, d16, #0           @ < 0 ?
-  vbit       d17, d3, d1           @ For r[0]
-  vbit       d29, d30, d1          @ -scale = -1
-
-  @ For the case of more than 1 bit overflow. If it occurs overwrite the
-  @ results for scale and r[0] in d17 and d29.
-  vclz.s32   d5, d16               @ Leading zeros of the two 32 bit words.
-  vshr.s64   d26, d5, #32          @ Keep only the upper 32 bits.
-  vsub.i64   d31, d26, d0          @ zeros - 33
-  vshl.i64   d27, d26, #32
-  vorr       d27, d26              @ Duplicate the high word with its low one.
-  vshl.u64   d2, d16, d31          @ Shift by (-scale).
-  vclt.s32   d1, d27, d25          @ < 32 ?
-  vbit       d17, d2, d1           @ For r[0]
-  vbit       d29, d31, d1          @ -scale
-
-  vst1.32    d17[0], [r0]!         @ r[0]
-  mov        r5, #1                @ outer loop counter
-
-@ Generate rest of the coefficients
-LOOP_R:
-  vmov.i32   q8, #0                @ Initialize the accumulation result.
-  vmov.i32   q9, #0                @ Initialize the accumulation result.
-  mov        r7, r1                @ &x[0]
-  add        r6, r7, r5, lsl #1    @ x[i]
-  sub        r12, r2, r5           @ N - i
-  lsr        r8, r12, #3           @ inner loop counter
-  sub        r12, r8, lsl #3       @ Leftover samples to be processed
-
-LOOP_8X_SAMPLES:                   @ Multiple of 8 samples
-  vld1.16    {d20, d21}, [r7]!     @ x[0, ...]
-  vld1.16    {d22, d23}, [r6]!     @ x[i, ...]
-  vmull.s16  q12, d20, d22
-  vmull.s16  q13, d21, d23
-  subs       r8, #1
-  vpadal.s32 q8, q12
-  vpadal.s32 q9, q13
-  bgt        LOOP_8X_SAMPLES
-
-  cmp r12, #4
-  blt REST_SAMPLES
-
-Four_SAMPLES:
-  vld1.16    d20, [r7]!
-  vld1.16    d22, [r6]!
-  vmull.s16  q12, d20, d22
-  vpadal.s32 q8, q12
-  sub r12, #4
-
-REST_SAMPLES:
-  mov        r8, #0                @ Initialize lower word of the accumulation.
-  mov        r4, #0                @ Initialize upper word of the accumulation.
-  cmp r12, #0
-  ble SUMUP
-
-LOOP_REST_SAMPLES:
-  ldrh       r9, [r7], #2          @ x[0, ...]
-  ldrh       r10, [r6], #2         @ x[i, ...]
-  smulbb     r11, r9, r10
-  adds       r8, r8, r11           @ lower word of the accumulation.
-  adc        r4, r4, r11, asr #31  @ upper word of the accumulation.
-  subs       r12, #1
-  bgt        LOOP_REST_SAMPLES
-
-@ Added the multiplication results together and do a shift.
-SUMUP:
-  vadd.i64   d16, d17
-  vadd.i64   d18, d19
-  vadd.i64   d18, d16
-  vmov       d17, r8, r4
-  vadd.i64   d18, d17
-  vshl.s64   d18, d29              @ Shift left by (-scale).
-  vst1.32    d18[0], [r0]!         @ r[i]
-
-  add        r5, #1
-  cmp        r5, r3
-  ble        LOOP_R
-
-  vneg.s32   d29, d29              @ Get value for 'scale'.
-  ldr        r2, [sp, #40]         @ &scale
-  add        r0, r3, #1            @ return (order + 1)
-  vst1.s16   d29[0], [r2]          @ Store 'scale'
-
-  pop        {r3 - r12}
-  bx         lr
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
index 4ea4dabc55..5cce1e9f0b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
@@ -9,7 +9,7 @@
  */
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 class FiltersTest : public testing::Test {
@@ -59,11 +59,11 @@ class FiltersTest : public testing::Test {
 
 TEST_F(FiltersTest, AutocorrFixTest) {
   FiltersTester(WebRtcIsacfix_AutocorrC);
-#ifdef WEBRTC_DETECT_ARM_NEON
+#ifdef WEBRTC_DETECT_NEON
   if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
     FiltersTester(WebRtcIsacfix_AutocorrNeon);
   }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   FiltersTester(WebRtcIsacfix_AutocorrNeon);
 #endif
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
similarity index 56%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
index d12c1678b8..d6385314a1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,27 +8,26 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 
 namespace webrtc {
 
-struct IsacFix {
-  typedef ISACFIX_MainStruct instance_type;
+class IsacFix {
+ public:
+  using instance_type = ISACFIX_MainStruct;
   static const bool has_swb = false;
-  static const uint16_t kFixSampleRate = 16000;
   static inline int16_t Control(instance_type* inst,
                                 int32_t rate,
-                                int16_t framesize) {
+                                int framesize) {
     return WebRtcIsacfix_Control(inst, rate, framesize);
   }
   static inline int16_t ControlBwe(instance_type* inst,
                                    int32_t rate_bps,
-                                   int16_t frame_size_ms,
+                                   int frame_size_ms,
                                    int16_t enforce_frame_size) {
     return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms,
                                     enforce_frame_size);
@@ -36,24 +35,24 @@ struct IsacFix {
   static inline int16_t Create(instance_type** inst) {
     return WebRtcIsacfix_Create(inst);
   }
-  static inline int16_t DecodeInternal(instance_type* inst,
-                                       const uint8_t* encoded,
-                                       int16_t len,
-                                       int16_t* decoded,
-                                       int16_t* speech_type) {
+  static inline int DecodeInternal(instance_type* inst,
+                                   const uint8_t* encoded,
+                                   size_t len,
+                                   int16_t* decoded,
+                                   int16_t* speech_type) {
     return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
   }
-  static inline int16_t DecodePlc(instance_type* inst,
-                                  int16_t* decoded,
-                                  int16_t num_lost_frames) {
+  static inline size_t DecodePlc(instance_type* inst,
+                                 int16_t* decoded,
+                                 size_t num_lost_frames) {
     return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
   }
-  static inline int16_t DecoderInit(instance_type* inst) {
-    return WebRtcIsacfix_DecoderInit(inst);
+  static inline void DecoderInit(instance_type* inst) {
+    WebRtcIsacfix_DecoderInit(inst);
   }
-  static inline int16_t Encode(instance_type* inst,
-                               const int16_t* speech_in,
-                               uint8_t* encoded) {
+  static inline int Encode(instance_type* inst,
+                           const int16_t* speech_in,
+                           uint8_t* encoded) {
     return WebRtcIsacfix_Encode(inst, speech_in, encoded);
   }
   static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
@@ -66,6 +65,10 @@ struct IsacFix {
   static inline int16_t Free(instance_type* inst) {
     return WebRtcIsacfix_Free(inst);
   }
+  static inline void GetBandwidthInfo(instance_type* inst,
+                                      IsacBandwidthInfo* bwinfo) {
+    WebRtcIsacfix_GetBandwidthInfo(inst, bwinfo);
+  }
   static inline int16_t GetErrorCode(instance_type* inst) {
     return WebRtcIsacfix_GetErrorCode(inst);
   }
@@ -73,20 +76,31 @@ struct IsacFix {
   static inline int16_t GetNewFrameLen(instance_type* inst) {
     return WebRtcIsacfix_GetNewFrameLen(inst);
   }
-
+  static inline void SetBandwidthInfo(instance_type* inst,
+                                      const IsacBandwidthInfo* bwinfo) {
+    WebRtcIsacfix_SetBandwidthInfo(inst, bwinfo);
+  }
   static inline int16_t SetDecSampRate(instance_type* inst,
                                        uint16_t sample_rate_hz) {
-    DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
     return 0;
   }
   static inline int16_t SetEncSampRate(instance_type* inst,
                                        uint16_t sample_rate_hz) {
-    DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
     return 0;
   }
+  static inline void SetEncSampRateInDecoder(instance_type* inst,
+                                             uint16_t sample_rate_hz) {
+    RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+  }
+  static inline void SetInitialBweBottleneck(instance_type* inst,
+                                             int bottleneck_bits_per_second) {
+    WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
+  }
   static inline int16_t UpdateBwEstimate(instance_type* inst,
                                          const uint8_t* encoded,
-                                         int32_t packet_size,
+                                         size_t packet_size,
                                          uint16_t rtp_seq_number,
                                          uint32_t send_ts,
                                          uint32_t arr_ts) {
@@ -100,9 +114,10 @@ struct IsacFix {
   static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
     return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate);
   }
+
+ private:
+  enum { kFixSampleRate = 16000 };
 };
 
-typedef AudioEncoderDecoderIsacT<IsacFix> AudioEncoderDecoderIsacFix;
-
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 922e0299ba..aba3aa0c0b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -15,7 +15,7 @@
  *
  */
 
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 
 #include <assert.h>
 #include <stdlib.h>
@@ -26,7 +26,7 @@
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
 // Declare function pointers.
 FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix;
@@ -38,7 +38,7 @@ MatrixProduct2 WebRtcIsacfix_MatrixProduct2;
 /* This method assumes that |stream_size_bytes| is in valid range,
  * i.e. >= 0 && <=  STREAM_MAXW16_60MS
  */
-static void InitializeDecoderBitstream(int stream_size_bytes,
+static void InitializeDecoderBitstream(size_t stream_size_bytes,
                                        Bitstr_dec* bitstream) {
   bitstream->W_upper = 0xFFFFFFFF;
   bitstream->streamval = 0;
@@ -72,10 +72,12 @@ int16_t WebRtcIsacfix_AssignSize(int *sizeinbytes) {
 
 int16_t WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst, void *ISACFIX_inst_Addr) {
   if (ISACFIX_inst_Addr!=NULL) {
-    *inst = (ISACFIX_MainStruct*)ISACFIX_inst_Addr;
-    (*(ISACFIX_SubStruct**)inst)->errorcode = 0;
-    (*(ISACFIX_SubStruct**)inst)->initflag = 0;
-    (*(ISACFIX_SubStruct**)inst)->ISACenc_obj.SaveEnc_ptr = NULL;
+    ISACFIX_SubStruct* self = ISACFIX_inst_Addr;
+    *inst = (ISACFIX_MainStruct*)self;
+    self->errorcode = 0;
+    self->initflag = 0;
+    self->ISACenc_obj.SaveEnc_ptr = NULL;
+    WebRtcIsacfix_InitBandwidthEstimator(&self->bwestimator_obj);
     return(0);
   } else {
     return(-1);
@@ -108,6 +110,7 @@ int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst)
     (*(ISACFIX_SubStruct**)ISAC_main_inst)->initflag = 0;
     (*(ISACFIX_SubStruct**)ISAC_main_inst)->ISACenc_obj.SaveEnc_ptr = NULL;
     WebRtcSpl_Init();
+    WebRtcIsacfix_InitBandwidthEstimator(&tempo->bwestimator_obj);
     return(0);
   } else {
     return(-1);
@@ -198,14 +201,12 @@ int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst)
  * This function initializes function pointers for ARM Neon platform.
  */
 
-#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON)
+#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 static void WebRtcIsacfix_InitNeon(void) {
   WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrNeon;
   WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopNeon;
   WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeNeon;
   WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecNeon;
-  WebRtcIsacfix_CalculateResidualEnergy =
-      WebRtcIsacfix_CalculateResidualEnergyNeon;
   WebRtcIsacfix_AllpassFilter2FixDec16 =
       WebRtcIsacfix_AllpassFilter2FixDec16Neon;
   WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1Neon;
@@ -240,6 +241,31 @@ static void WebRtcIsacfix_InitMIPS(void) {
 }
 #endif
 
+static void InitFunctionPointers(void) {
+  WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrC;
+  WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopC;
+  WebRtcIsacfix_CalculateResidualEnergy =
+      WebRtcIsacfix_CalculateResidualEnergyC;
+  WebRtcIsacfix_AllpassFilter2FixDec16 = WebRtcIsacfix_AllpassFilter2FixDec16C;
+  WebRtcIsacfix_HighpassFilterFixDec32 = WebRtcIsacfix_HighpassFilterFixDec32C;
+  WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecC;
+  WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeC;
+  WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1C;
+  WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2C;
+
+#ifdef WEBRTC_DETECT_NEON
+  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
+    WebRtcIsacfix_InitNeon();
+  }
+#elif defined(WEBRTC_HAS_NEON)
+  WebRtcIsacfix_InitNeon();
+#endif
+
+#if defined(MIPS32_LE)
+  WebRtcIsacfix_InitMIPS();
+#endif
+}
+
 /****************************************************************************
  * WebRtcIsacfix_EncoderInit(...)
  *
@@ -290,8 +316,6 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
   WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACenc_obj.pitchfiltstr_obj);
   WebRtcIsacfix_InitPitchAnalysis(&ISAC_inst->ISACenc_obj.pitchanalysisstr_obj);
 
-
-  WebRtcIsacfix_InitBandwidthEstimator(&ISAC_inst->bwestimator_obj);
   WebRtcIsacfix_InitRateModel(&ISAC_inst->ISACenc_obj.rate_data_obj);
 
 
@@ -318,29 +342,7 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
   WebRtcIsacfix_InitPostFilterbank(&ISAC_inst->ISACenc_obj.interpolatorstr_obj);
 #endif
 
-  // Initiaze function pointers.
-  WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrC;
-  WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopC;
-  WebRtcIsacfix_CalculateResidualEnergy =
-      WebRtcIsacfix_CalculateResidualEnergyC;
-  WebRtcIsacfix_AllpassFilter2FixDec16 = WebRtcIsacfix_AllpassFilter2FixDec16C;
-  WebRtcIsacfix_HighpassFilterFixDec32 = WebRtcIsacfix_HighpassFilterFixDec32C;
-  WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecC;
-  WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeC;
-  WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1C;
-  WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2C;
-
-#ifdef WEBRTC_DETECT_ARM_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    WebRtcIsacfix_InitNeon();
-  }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
-  WebRtcIsacfix_InitNeon();
-#endif
-
-#if defined(MIPS32_LE)
-  WebRtcIsacfix_InitMIPS();
-#endif
+  InitFunctionPointers();
 
   return statusInit;
 }
@@ -396,12 +398,12 @@ static void write_be16(const uint16_t* src, size_t nbytes, uint8_t* dest) {
  *                          : -1 - Error
  */
 
-int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
-                             const int16_t    *speechIn,
-                             uint8_t* encoded)
+int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
+                         const int16_t    *speechIn,
+                         uint8_t* encoded)
 {
   ISACFIX_SubStruct *ISAC_inst;
-  int16_t stream_len;
+  int stream_len;
 
   /* typecast pointer to rela structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -418,11 +420,12 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
                                         &ISAC_inst->bwestimator_obj,
                                         ISAC_inst->CodingMode);
   if (stream_len<0) {
-    ISAC_inst->errorcode = - stream_len;
+    ISAC_inst->errorcode = -(int16_t)stream_len;
     return -1;
   }
 
-  write_be16(ISAC_inst->ISACenc_obj.bitstr_obj.stream, stream_len, encoded);
+  write_be16(ISAC_inst->ISACenc_obj.bitstr_obj.stream, (size_t)stream_len,
+             encoded);
   return stream_len;
 
 }
@@ -565,16 +568,14 @@ int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
  *
  * Input:
  *      - ISAC_main_inst    : ISAC instance.
- *
- * Return value
- *                          :  0 - Ok
- *                            -1 - Error
  */
 
-int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
+void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
 {
   ISACFIX_SubStruct *ISAC_inst;
 
+  InitFunctionPointers();
+
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
 
@@ -592,8 +593,6 @@ int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
   WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACdec_obj.decimatorstr_obj);
 #endif
-
-  return 0;
 }
 
 
@@ -616,20 +615,20 @@ int16_t WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
 
 int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
                                         const uint8_t* encoded,
-                                        int32_t          packet_size,
-                                        uint16_t         rtp_seq_number,
-                                        uint32_t         arr_ts)
+                                        size_t packet_size,
+                                        uint16_t rtp_seq_number,
+                                        uint32_t arr_ts)
 {
   ISACFIX_SubStruct *ISAC_inst;
   Bitstr_dec streamdata;
   int16_t err;
-  const int kRequiredEncodedLenBytes = 10;
+  const size_t kRequiredEncodedLenBytes = 10;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
 
   /* Sanity check of packet length */
-  if (packet_size <= 0) {
+  if (packet_size == 0) {
     /* return error code if the packet length is null or less */
     ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
     return -1;
@@ -688,21 +687,21 @@ int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
 
 int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
                                        const uint8_t* encoded,
-                                       int32_t          packet_size,
-                                       uint16_t         rtp_seq_number,
-                                       uint32_t         send_ts,
-                                       uint32_t         arr_ts)
+                                       size_t packet_size,
+                                       uint16_t rtp_seq_number,
+                                       uint32_t send_ts,
+                                       uint32_t arr_ts)
 {
   ISACFIX_SubStruct *ISAC_inst;
   Bitstr_dec streamdata;
   int16_t err;
-  const int kRequiredEncodedLenBytes = 10;
+  const size_t kRequiredEncodedLenBytes = 10;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
 
   /* Sanity check of packet length */
-  if (packet_size <= 0) {
+  if (packet_size == 0) {
     /* return error code if the packet length is null  or less */
     ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
     return -1;
@@ -763,17 +762,18 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
  */
 
 
-int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
-                             const uint8_t* encoded,
-                             int16_t          len,
-                             int16_t          *decoded,
-                             int16_t     *speechType)
+int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* number of samples (480 or 960), output from decoder */
   /* that were actually used in the encoder/decoder (determined on the fly) */
-  int16_t     number_of_samples;
-  int16_t declen = 0;
+  size_t number_of_samples;
+  int declen_int = 0;
+  size_t declen;
 
   /* typecast pointer to real structure */
   ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
@@ -785,7 +785,7 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
   }
 
   /* Sanity check of packet length */
-  if (len <= 0) {
+  if (len == 0) {
     /* return error code if the packet length is null  or less */
     ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
     return -1;
@@ -802,32 +802,37 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
   /* added for NetEq purposes (VAD/DTX related) */
   *speechType=1;
 
-  declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
-
-  if (declen < 0) {
+  declen_int = WebRtcIsacfix_DecodeImpl(decoded, &ISAC_inst->ISACdec_obj,
+                                        &number_of_samples);
+  if (declen_int < 0) {
     /* Some error inside the decoder */
-    ISAC_inst->errorcode = -declen;
+    ISAC_inst->errorcode = -(int16_t)declen_int;
     memset(decoded, 0, sizeof(int16_t) * MAX_FRAMESAMPLES);
     return -1;
   }
+  declen = (size_t)declen_int;
 
   /* error check */
 
-  if (declen & 0x0001) {
-    if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
+  if (declen & 1) {
+    if (len != declen &&
+        len != declen +
+            ((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) & 0xFF)) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
       memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   } else {
-    if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
+    if (len != declen &&
+        len != declen +
+            ((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) >> 8)) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
       memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   }
 
-  return number_of_samples;
+  return (int)number_of_samples;
 }
 
 
@@ -856,17 +861,18 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
  */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
-                               const uint16_t   *encoded,
-                               int16_t          len,
-                               int16_t          *decoded,
-                               int16_t    *speechType)
+int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct* ISAC_main_inst,
+                           const uint16_t* encoded,
+                           size_t len,
+                           int16_t* decoded,
+                           int16_t* speechType)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* twice the number of samples (480 or 960), output from decoder */
   /* that were actually used in the encoder/decoder (determined on the fly) */
-  int16_t     number_of_samples;
-  int16_t declen = 0;
+  size_t number_of_samples;
+  int declen_int = 0;
+  size_t declen;
   int16_t dummy[FRAMESAMPLES/2];
 
 
@@ -879,7 +885,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
     return (-1);
   }
 
-  if (len <= 0) {
+  if (len == 0) {
     /* return error code if the packet length is null  or less */
     ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
     return -1;
@@ -896,25 +902,30 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
   /* added for NetEq purposes (VAD/DTX related) */
   *speechType=1;
 
-  declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
-
-  if (declen < 0) {
+  declen_int = WebRtcIsacfix_DecodeImpl(decoded, &ISAC_inst->ISACdec_obj,
+                                        &number_of_samples);
+  if (declen_int < 0) {
     /* Some error inside the decoder */
-    ISAC_inst->errorcode = -declen;
+    ISAC_inst->errorcode = -(int16_t)declen_int;
     memset(decoded, 0, sizeof(int16_t) * FRAMESAMPLES);
     return -1;
   }
+  declen = (size_t)declen_int;
 
   /* error check */
 
-  if (declen & 0x0001) {
-    if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
+  if (declen & 1) {
+    if (len != declen &&
+        len != declen +
+            ((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) & 0xFF)) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
       memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
     }
   } else {
-    if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
+    if (len != declen &&
+        len != declen +
+            ((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >>1]) >> 8)) {
       ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
       memset(decoded, 0, sizeof(int16_t) * number_of_samples);
       return -1;
@@ -928,7 +939,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
                                   dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
   }
 
-  return number_of_samples/2;
+  return (int)(number_of_samples / 2);
 }
 #endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
 
@@ -949,16 +960,15 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
  * Output:
  *      - decoded           : The decoded vector
  *
- * Return value             : >0 - number of samples in decoded PLC vector
- *                            -1 - Error
+ * Return value             : Number of samples in decoded PLC vector
  */
 
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-int16_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
-                                  int16_t          *decoded,
-                                  int16_t noOfLostFrames )
+size_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct* ISAC_main_inst,
+                                 int16_t* decoded,
+                                 size_t noOfLostFrames )
 {
-  int16_t no_of_samples, declen, k, ok;
+  size_t no_of_samples, declen, k;
   int16_t outframeNB[FRAMESAMPLES];
   int16_t outframeWB[FRAMESAMPLES];
   int16_t dummy[FRAMESAMPLES/2];
@@ -977,9 +987,8 @@ int16_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
   declen = 0;
   while( noOfLostFrames > 0 )
   {
-    ok = WebRtcIsacfix_DecodePlcImpl( outframeWB, &ISAC_inst->ISACdec_obj, &no_of_samples );
-    if(ok)
-      return -1;
+    WebRtcIsacfix_DecodePlcImpl(outframeWB, &ISAC_inst->ISACdec_obj,
+                                &no_of_samples);
 
     WebRtcIsacfix_SplitAndFilter2(outframeWB, &(outframeNB[k*240]), dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
 
@@ -1016,16 +1025,15 @@ int16_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
  * Output:
  *      - decoded           : The decoded vector
  *
- * Return value             : >0 - number of samples in decoded PLC vector
- *                            -1 - Error
+ * Return value             : Number of samples in decoded PLC vector
  */
 
-int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
-                                int16_t          *decoded,
-                                int16_t noOfLostFrames)
+size_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct* ISAC_main_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames)
 {
 
-  int16_t no_of_samples, declen, k, ok;
+  size_t no_of_samples, declen, k;
   int16_t outframe16[MAX_FRAMESAMPLES];
 
   ISACFIX_SubStruct *ISAC_inst;
@@ -1040,9 +1048,8 @@ int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
   declen = 0;
   while( noOfLostFrames > 0 )
   {
-    ok = WebRtcIsacfix_DecodePlcImpl( &(outframe16[k*480]), &ISAC_inst->ISACdec_obj, &no_of_samples );
-    if(ok)
-      return -1;
+    WebRtcIsacfix_DecodePlcImpl(&(outframe16[k*480]), &ISAC_inst->ISACdec_obj,
+                                &no_of_samples);
     declen += no_of_samples;
     noOfLostFrames--;
     k++;
@@ -1073,8 +1080,8 @@ int16_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
  */
 
 int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
-                              int16_t          rate,
-                              int16_t          framesize)
+                              int16_t rate,
+                              int framesize)
 {
   ISACFIX_SubStruct *ISAC_inst;
   /* typecast pointer to real structure */
@@ -1098,7 +1105,7 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
 
 
   if (framesize  == 30 || framesize == 60)
-    ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * framesize;
+    ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * framesize);
   else {
     ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
     return -1;
@@ -1107,6 +1114,13 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
   return 0;
 }
 
+void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst,
+                                           int bottleneck_bits_per_second) {
+  ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst;
+  assert(bottleneck_bits_per_second >= 10000 &&
+         bottleneck_bits_per_second <= 32000);
+  inst->bwestimator_obj.sendBwAvg = ((uint32_t)bottleneck_bits_per_second) << 7;
+}
 
 /****************************************************************************
  * WebRtcIsacfix_ControlBwe(...)
@@ -1133,7 +1147,7 @@ int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
 
 int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
                                  int16_t rateBPS,
-                                 int16_t frameSizeMs,
+                                 int frameSizeMs,
                                  int16_t enforceFrameSize)
 {
   ISACFIX_SubStruct *ISAC_inst;
@@ -1167,7 +1181,7 @@ int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
 
   /* Set initial framesize. If enforceFrameSize is set the frame size will not change */
   if ((frameSizeMs  == 30) || (frameSizeMs == 60)) {
-    ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * frameSizeMs;
+    ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * frameSizeMs);
   } else {
     ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
     return -1;
@@ -1254,12 +1268,12 @@ int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
  */
 
 int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
-                                   int encoded_len_bytes,
-                                   int16_t* frameLength)
+                                   size_t encoded_len_bytes,
+                                   size_t* frameLength)
 {
   Bitstr_dec streamdata;
   int16_t err;
-  const int kRequiredEncodedLenBytes = 10;
+  const size_t kRequiredEncodedLenBytes = 10;
 
   if (encoded_len_bytes < kRequiredEncodedLenBytes) {
     return -1;
@@ -1293,12 +1307,12 @@ int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
  */
 
 int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
-                                  int encoded_len_bytes,
+                                  size_t encoded_len_bytes,
                                   int16_t* rateIndex)
 {
   Bitstr_dec streamdata;
   int16_t err;
-  const int kRequiredEncodedLenBytes = 10;
+  const size_t kRequiredEncodedLenBytes = 10;
 
   if (encoded_len_bytes < kRequiredEncodedLenBytes) {
     return -1;
@@ -1309,7 +1323,8 @@ int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
   read_be16(encoded, kRequiredEncodedLenBytes, streamdata.stream);
 
   /* decode frame length, needed to get to the rateIndex in the bitstream */
-  err = WebRtcIsacfix_DecodeFrameLen(&streamdata, rateIndex);
+  size_t frameLength;
+  err = WebRtcIsacfix_DecodeFrameLen(&streamdata, &frameLength);
   if (err<0)  // error check
     return err;
 
@@ -1524,3 +1539,17 @@ void WebRtcIsacfix_version(char *version)
 {
   strcpy(version, "3.6.0");
 }
+
+void WebRtcIsacfix_GetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
+                                    IsacBandwidthInfo* bwinfo) {
+  ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst;
+  assert(inst->initflag & 1);  // Decoder initialized.
+  WebRtcIsacfixBw_GetBandwidthInfo(&inst->bwestimator_obj, bwinfo);
+}
+
+void WebRtcIsacfix_SetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
+                                    const IsacBandwidthInfo* bwinfo) {
+  ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst;
+  assert(inst->initflag & 2);  // Encoder initialized.
+  WebRtcIsacfixBw_SetBandwidthInfo(&inst->bwestimator_obj, bwinfo);
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
index cfd9a9eb73..22224a8071 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
@@ -43,7 +43,7 @@ void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0,
                                 int16_t* ar_f_Q0,
                                 int16_t* cth_Q15,
                                 int16_t* sth_Q15,
-                                int16_t order_coef);
+                                size_t order_coef);
 
 /* Inner loop used for function WebRtcIsacfix_NormLatticeFilterMa(). It does:
    for 0 <= n < HALF_SUBFRAMELEN - 1:
@@ -86,7 +86,7 @@ void WebRtcIsacfix_FilterMaLoopC(int16_t input0,  // Filter coefficient
 
 /* filter the signal using normalized lattice filter */
 /* MA filter */
-void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
+void WebRtcIsacfix_NormLatticeFilterMa(size_t orderCoef,
                                        int32_t *stateGQ15,
                                        int16_t *lat_inQ0,
                                        int16_t *filt_coefQ15,
@@ -97,9 +97,10 @@ void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
   int16_t sthQ15[MAX_AR_MODEL_ORDER];
   int16_t cthQ15[MAX_AR_MODEL_ORDER];
 
-  int u, i, k, n;
+  int u, n;
+  size_t i, k;
   int16_t temp2,temp3;
-  int16_t ord_1 = orderCoef+1;
+  size_t ord_1 = orderCoef+1;
   int32_t inv_cthQ16[MAX_AR_MODEL_ORDER];
 
   int32_t gain32, fQtmp;
@@ -210,7 +211,7 @@ void WebRtcIsacfix_NormLatticeFilterMa(int16_t orderCoef,
 
 /* ----------------AR filter-------------------------*/
 /* filter the signal using normalized lattice filter */
-void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
+void WebRtcIsacfix_NormLatticeFilterAr(size_t orderCoef,
                                        int16_t *stateGQ0,
                                        int32_t *lat_inQ25,
                                        int16_t *filt_coefQ15,
@@ -218,7 +219,8 @@ void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
                                        int16_t lo_hi,
                                        int16_t *lat_outQ0)
 {
-  int ii,n,k,i,u;
+  size_t ii, k, i;
+  int n, u;
   int16_t sthQ15[MAX_AR_MODEL_ORDER];
   int16_t cthQ15[MAX_AR_MODEL_ORDER];
   int32_t tmp32;
@@ -234,7 +236,7 @@ void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
   int16_t sh;
 
   int16_t temp2,temp3;
-  int16_t ord_1 = orderCoef+1;
+  size_t ord_1 = orderCoef+1;
 
   for (u=0;u<SUBFRAMES;u++)
   {
@@ -279,13 +281,16 @@ void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
       ARfQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
     }
 
-    for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
+    // Get the state of f & g for the first input, for all orders.
+    for (i = orderCoef; i > 0; i--)
     {
-      tmp32 = (cthQ15[i] * ARfQ0vec[0] - sthQ15[i] * stateGQ0[i] + 16384) >> 15;
+      tmp32 = (cthQ15[i - 1] * ARfQ0vec[0] - sthQ15[i - 1] * stateGQ0[i - 1] +
+               16384) >> 15;
       tmpAR = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
 
-      tmp32 = (sthQ15[i] * ARfQ0vec[0] + cthQ15[i] * stateGQ0[i] + 16384) >> 15;
-      ARgQ0vec[i+1] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
+      tmp32 = (sthQ15[i - 1] * ARfQ0vec[0] + cthQ15[i - 1] * stateGQ0[i - 1] +
+               16384) >> 15;
+      ARgQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
       ARfQ0vec[0] = tmpAR;
     }
     ARgQ0vec[0] = ARfQ0vec[0];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
index 35fd9ef748..36411df0ab 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
@@ -25,7 +25,7 @@
 @ r12: constant #16384
 @ r6, r7, r8, r10, r11: scratch
 
-#include "webrtc/system_wrappers/interface/asm_defines.h"
+#include "webrtc/system_wrappers/include/asm_defines.h"
 #include "settings.h"
 
 GLOBAL_FUNCTION WebRtcIsacfix_FilterArLoop
@@ -38,7 +38,7 @@ DEFINE_FUNCTION WebRtcIsacfix_FilterArLoop
   mov     r4, #HALF_SUBFRAMELEN
   sub     r4, #1                 @ Outer loop counter = HALF_SUBFRAMELEN - 1
 
-HALF_SUBFRAME_LOOP:  @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
+HALF_SUBFRAME_LOOP:  @ for (n = 0; n < HALF_SUBFRAMELEN - 1; n++)
 
   ldr     r9, [sp, #32]          @ Restore the inner loop counter to order_coef
   ldrh    r5, [r1]               @ tmpAR = ar_f_Q0[n+1]
@@ -46,21 +46,23 @@ HALF_SUBFRAME_LOOP:  @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
   add     r2, r9, asl #1         @ Restore r2 to &cth_Q15[order_coef]
   add     r3, r9, asl #1         @ Restore r3 to &sth_Q15[order_coef]
 
-ORDER_COEF_LOOP:  @ for(k = order_coef - 1 ; k >= 0; k--)
+ORDER_COEF_LOOP:  @ for (k = order_coef; k > 0; k--)
 
-  ldrh    r7, [r3, #-2]!         @ sth_Q15[k]
-  ldrh    r6, [r2, #-2]!         @ cth_Q15[k]
+  ldrh    r7, [r3, #-2]!         @ sth_Q15[k - 1]
+  ldrh    r6, [r2, #-2]!         @ cth_Q15[k - 1]
 
-  ldrh    r8, [r0, #-2]          @ ar_g_Q0[k]
-  smlabb  r11, r7, r5, r12       @ sth_Q15[k] * tmpAR + 16384
-  smlabb  r10, r6, r5, r12       @ cth_Q15[k] * tmpAR + 16384
-  smulbb  r7, r7, r8             @ sth_Q15[k] * ar_g_Q0[k]
-  smlabb  r11, r6, r8, r11       @ cth_Q15[k]*ar_g_Q0[k]+(sth_Q15[k]*tmpAR+16384)
+  ldrh    r8, [r0, #-2]          @ ar_g_Q0[k - 1]
+  smlabb  r11, r7, r5, r12       @ sth_Q15[k - 1] * tmpAR + 16384
+  smlabb  r10, r6, r5, r12       @ cth_Q15[k - 1] * tmpAR + 16384
+  smulbb  r7, r7, r8             @ sth_Q15[k - 1] * ar_g_Q0[k - 1]
+  smlabb  r11, r6, r8, r11       @ cth_Q15[k - 1] * ar_g_Q0[k - 1] +
+                                 @     (sth_Q15[k - 1] * tmpAR + 16384)
 
-  sub     r10, r10, r7           @ cth_Q15[k]*tmpAR+16384-(sth_Q15[k]*ar_g_Q0[k])
+  sub     r10, r10, r7           @ cth_Q15[k - 1] * tmpAR + 16384 -
+                                 @     (sth_Q15[k - 1] * ar_g_Q0[k - 1])
   ssat    r11, #16, r11, asr #15
   ssat    r5, #16, r10, asr #15
-  strh    r11, [r0], #-2         @ Output: ar_g_Q0[k+1]
+  strh    r11, [r0], #-2         @ Output: ar_g_Q0[k]
 
   subs    r9, #1
   bgt     ORDER_COEF_LOOP
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
index 8c53b0bf32..40c3bf8617 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
@@ -25,21 +25,23 @@ void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0,     // Input samples
                                 int16_t* ar_f_Q0,     // Input samples
                                 int16_t* cth_Q15,     // Filter coefficients
                                 int16_t* sth_Q15,     // Filter coefficients
-                                int16_t order_coef) { // order of the filter
+                                size_t order_coef) { // order of the filter
   int n = 0;
 
   for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) {
-    int k = 0;
+    size_t k = 0;
     int16_t tmpAR = 0;
     int32_t tmp32 = 0;
     int32_t tmp32_2 = 0;
 
     tmpAR = ar_f_Q0[n + 1];
-    for (k = order_coef - 1; k >= 0; k--) {
-      tmp32 = (cth_Q15[k] * tmpAR - sth_Q15[k] * ar_g_Q0[k] + 16384) >> 15;
-      tmp32_2 = (sth_Q15[k] * tmpAR + cth_Q15[k] * ar_g_Q0[k] + 16384) >> 15;
+    for (k = order_coef; k > 0; k--) {
+      tmp32 = (cth_Q15[k - 1] * tmpAR - sth_Q15[k - 1] * ar_g_Q0[k - 1] +
+               16384) >> 15;
+      tmp32_2 = (sth_Q15[k - 1] * tmpAR + cth_Q15[k - 1] * ar_g_Q0[k - 1] +
+                 16384) >> 15;
       tmpAR   = (int16_t)WebRtcSpl_SatW32ToW16(tmp32);
-      ar_g_Q0[k + 1] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32_2);
+      ar_g_Q0[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32_2);
     }
     ar_f_Q0[n + 1] = tmpAR;
     ar_g_Q0[0] = tmpAR;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
index c596922168..d488bfcb51 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <stddef.h>
+
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/typedefs.h"
 
@@ -17,11 +19,11 @@ void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0,     // Input samples
                                 int16_t* ar_f_Q0,     // Input samples
                                 int16_t* cth_Q15,     // Filter coefficients
                                 int16_t* sth_Q15,     // Filter coefficients
-                                int16_t order_coef) { // order of the filter
+                                size_t order_coef) { // order of the filter
   int n = 0;
 
   for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) {
-    int count = order_coef - 1;
+    int count = (int)(order_coef - 1);
     int offset;
 #if !defined(MIPS_DSP_R1_LE)
     int16_t* tmp_cth;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_neon.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_neon.S
deleted file mode 100644
index f31a32d9df..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_neon.S
+++ /dev/null
@@ -1,146 +0,0 @@
-@
-@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS.  All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ lattice_neon.s
-@
-@ Contains a function for the core loop in the normalized lattice MA 
-@ filter routine for iSAC codec, optimized for ARM Neon platform.
-@ void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0,
-@                                     int16_t input1,
-@                                     int32_t input2,
-@                                     int32_t* ptr0,
-@                                     int32_t* ptr1,
-@                                     int32_t* __restrict ptr2);
-@ It calculates
-@   *ptr2 = input2 * (*ptr2) + input0 * (*ptr0));
-@   *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
-@ in Q15 domain.
-@
-@ Reference code in lattice.c.
-@ Output is not bit-exact with the reference C code, due to the replacement
-@ of WEBRTC_SPL_MUL_16_32_RSFT15 and LATTICE_MUL_32_32_RSFT16 with Neon
-@ instructions, smulwb, and smull. Speech quality was not degraded by
-@ testing speech and tone vectors.
-
-#include "webrtc/system_wrappers/interface/asm_defines.h"
-#include "settings.h"
-
-GLOBAL_FUNCTION WebRtcIsacfix_FilterMaLoopNeon
-.align  2
-DEFINE_FUNCTION WebRtcIsacfix_FilterMaLoopNeon
-  push        {r4-r8}
-
-  vdup.32     d28, r0             @ Initialize Neon register with input0
-  vdup.32     d29, r1             @ Initialize Neon register with input1
-  vdup.32     d30, r2             @ Initialize Neon register with input2
-  ldr         r4, [sp, #20]       @ ptr1
-  ldr         r12, [sp, #24]      @ ptr2
-
-  @ Number of loop iterations after unrolling: r5 = (HALF_SUBFRAMELEN - 1) >> 2
-  @ Leftover samples after the loop, in r6:
-  @    r6 = (HALF_SUBFRAMELEN - 1) - (HALF_SUBFRAMELEN - 1) >> 2 << 2
-  mov         r6, #HALF_SUBFRAMELEN
-  sub         r6, #1
-  lsr         r5, r6, #2
-  sub         r6, r5, lsl #2
-
-  @ First r5 iterations in a loop.
-
-LOOP:
-  vld1.32     {d0, d1}, [r3]!     @ *ptr0
-
-  vmull.s32   q10, d0, d28        @ tmp32a = input0 * (*ptr0)
-  vmull.s32   q11, d1, d28        @ tmp32a = input0 * (*ptr0)
-  vmull.s32   q12, d0, d29        @ input1 * (*ptr0)
-  vmull.s32   q13, d1, d29        @ input1 * (*ptr0)
-                                  
-  vrshrn.i64  d4, q10, #15        
-  vrshrn.i64  d5, q11, #15        
-                                  
-  vld1.32     {d2, d3}, [r12]     @ *ptr2
-  vadd.i32    q3, q2, q1          @ tmp32b = *ptr2 + tmp32a
-                                  
-  vrshrn.i64  d0, q12, #15        
-                                  
-  vmull.s32   q10, d6, d30        @ input2 * (*ptr2 + tmp32b)
-  vmull.s32   q11, d7, d30        @ input2 * (*ptr2 + tmp32b)
-
-  vrshrn.i64  d16, q10, #16
-  vrshrn.i64  d17, q11, #16
-
-  vmull.s32   q10, d16, d28       @ input0 * (*ptr2)
-  vmull.s32   q11, d17, d28       @ input0 * (*ptr2)
-
-  vrshrn.i64  d1, q13, #15
-  vrshrn.i64  d18, q10, #15
-  vrshrn.i64  d19, q11, #15
-
-  vst1.32     {d16, d17}, [r12]!  @ *ptr2
-
-  vadd.i32    q9, q0, q9
-  subs        r5, #1
-  vst1.32     {d18, d19}, [r4]!   @ *ptr1
-
-  bgt         LOOP
-
-  @ Check how many samples still need to be processed.
-  subs        r6, #2
-  blt         LAST_SAMPLE
-
-  @ Process two more samples:
-  vld1.32     d0, [r3]!           @ *ptr0
-
-  vmull.s32   q11, d0, d28        @ tmp32a = input0 * (*ptr0)
-  vmull.s32   q13, d0, d29        @ input1 * (*ptr0)
-
-  vld1.32     d18, [r12]          @ *ptr2
-  vrshrn.i64  d4, q11, #15
-
-  vadd.i32    d7, d4, d18         @ tmp32b = *ptr2 + tmp32a
-  vmull.s32   q11, d7, d30        @ input2 * (*ptr2 + tmp32b)
-  vrshrn.i64  d16, q11, #16
-
-  vmull.s32   q11, d16, d28       @ input0 * (*ptr2)
-  vst1.32     d16, [r12]!         @ *ptr2
-
-  vrshrn.i64  d0, q13, #15
-  vrshrn.i64  d19, q11, #15
-  vadd.i32    d19, d0, d19
-
-  vst1.32     d19, [r4]!          @ *ptr1
-
-  @ If there's still one more sample, process it here.
-LAST_SAMPLE:
-  cmp         r6, #1
-  bne         END
-
-  @ *ptr2 = input2 * (*ptr2 + input0 * (*ptr0));
-  
-  ldr         r7, [r3]            @ *ptr0
-  ldr         r8, [r12]           @ *ptr2
-
-  smulwb      r5, r7, r0          @ tmp32a = *ptr0 * input0 >> 16
-  add         r8, r8, r5, lsl #1  @ tmp32b = *ptr2 + (tmp32a << 1)
-  smull       r5, r6, r8, r2      @ tmp32b * input2, in 64 bits
-  lsl         r6, #16
-  add         r6, r5, lsr #16     @ Only take the middle 32 bits
-  str         r6, [r12]           @ Output (*ptr2, as 32 bits)
-
-  @ *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
-
-  smulwb      r5, r7, r1          @ tmp32a = *ptr0 * input1 >> 16
-  smulwb      r6, r6, r0          @ tmp32b = *ptr2 * input0 >> 16
-  lsl         r5, r5, #1
-  add         r5, r6, lsl #1
-  str         r5, [r4]            @ Output (*ptr1)
-
-END:
-  pop         {r4-r8}
-  bx          lr
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
index 5ae951a9e3..97e4ce03a4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
@@ -393,7 +393,7 @@ void WebRtcIsacfix_GetVars(const int16_t *input, const int16_t *pitchGains_Q12,
   chng3 = WEBRTC_SPL_ABS_W16(nrgQlog[1]-nrgQlog[0]);
   chng4 = WEBRTC_SPL_ABS_W16(nrgQlog[0]-oldNrgQlog);
   tmp = chng1+chng2+chng3+chng4;
-  chngQ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp, kChngFactor, 10); /* Q12 */
+  chngQ = (int16_t)(tmp * kChngFactor >> 10);  /* Q12 */
   chngQ += 2926; /* + 1.0/1.4 in Q12 */
 
   /* Find average pitch gain */
@@ -403,10 +403,10 @@ void WebRtcIsacfix_GetVars(const int16_t *input, const int16_t *pitchGains_Q12,
     pgQ += pitchGains_Q12[k];
   }
 
-  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pgQ,11); /* pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 */
-  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pg3,13); /* Q17*Q14>>13 =>Q18  */
-  pg3 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(pg3, kMulPitchGain ,5); /* Q10  kMulPitchGain = -25 = -200 in Q-3. */
-
+  pg3 = (int16_t)(pgQ * pgQ >> 11);  // pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17
+  pg3 = (int16_t)(pgQ * pg3 >> 13);  /* Q14*Q17>>13 =>Q18  */
+  /* kMulPitchGain = -25 = -200 in Q-3. */
+  pg3 = (int16_t)(pg3 * kMulPitchGain >> 5);  // Q10
   tmp16=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/
   if (tmp16<0) {
     tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
@@ -580,9 +580,9 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
   snrq=snrQ10;
 
   /* SNR= C * 2 ^ (D * snrq) ; C=0.289, D=0.05*log2(10)=0.166 (~=172 in Q10)*/
-  tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(snrq, 172, 10); // Q10
+  tmp16 = (int16_t)(snrq * 172 >> 10);  // Q10
   tmp16b = exp2_Q10_T(tmp16); // Q10
-  snrq = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 285, 10); // Q10
+  snrq = (int16_t)(tmp16b * 285 >> 10);  // Q10
 
   /* change quallevel depending on pitch gains and level fluctuations */
   WebRtcIsacfix_GetVars(inLoQ0, pitchGains_Q12, &(maskdata->OldEnergy), &varscaleQ14);
@@ -595,12 +595,12 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
   aaQ14 = (int16_t)((22938 * (8192 + (varscaleQ14 >> 1)) + 32768) >> 16);
 
   /* Calculate tmp = (1.0 + aa*aa); in Q12 */
-  tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(aaQ14, aaQ14, 15); //Q14*Q14>>15 = Q13
+  tmp16 = (int16_t)(aaQ14 * aaQ14 >> 15);  // Q14*Q14>>15 = Q13
   tmpQQlo = 4096 + (tmp16 >> 1);  // Q12 + Q13>>1 = Q12.
 
   /* Calculate tmp = (1.0+aa) * (1.0+aa); */
   tmp16 = 8192 + (aaQ14 >> 1);  // 1+a in Q13.
-  tmpQQhi = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 14); //Q13*Q13>>14 = Q12
+  tmpQQhi = (int16_t)(tmp16 * tmp16 >> 14);  // Q13*Q13>>14 = Q12
 
   /* replace data in buffer by new look-ahead data */
   for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) {
@@ -613,19 +613,19 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
     for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
       maskdata->DataBufferLoQ0[pos1] = maskdata->DataBufferLoQ0[pos1 + UPDATE/2];
       maskdata->DataBufferHiQ0[pos1] = maskdata->DataBufferHiQ0[pos1 + UPDATE/2];
-      DataLoQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
-      DataHiQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+      DataLoQ6[pos1] = (int16_t)(maskdata->DataBufferLoQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
+      DataHiQ6[pos1] = (int16_t)(maskdata->DataBufferHiQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
     }
     pos2 = (int16_t)(k * UPDATE / 2);
     for (n = 0; n < UPDATE/2; n++, pos1++) {
       maskdata->DataBufferLoQ0[pos1] = inLoQ0[QLOOKAHEAD + pos2];
       maskdata->DataBufferHiQ0[pos1] = inHiQ0[pos2++];
-      DataLoQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
-      DataHiQ6[pos1] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
+      DataLoQ6[pos1] = (int16_t)(maskdata->DataBufferLoQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
+      DataHiQ6[pos1] = (int16_t)(maskdata->DataBufferHiQ0[pos1] *
+          kWindowAutocorr[pos1] >> 15);  // Q0*Q21>>15 = Q6
     }
 
     /* Get correlation coefficients */
@@ -868,14 +868,12 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
       /* add hearing threshold and compute the gain */
       /* lo_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
 
-
-      //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17);  // Q14
       tmp32a = varscaleQ14 >> 1;  // H_T_HQ19=65536 (16-17=-1)
       ssh = sh_lo >> 1;  // sqrt_nrg is in Qssh.
       sh = ssh - 14;
       tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
       tmp32c = sqrt_nrg + tmp32b;  // Qssh  (denominator)
-      tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0);  //Q24 (numerator)
+      tmp32a = varscaleQ14 * snrq;  // Q24 (numerator)
 
       sh = WebRtcSpl_NormW32(tmp32c);
       shft = 16 - sh;
@@ -918,14 +916,13 @@ void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
       /* add hearing threshold and compute the gain */
       /* hi_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
 
-      //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17);  // Q14
       tmp32a = varscaleQ14 >> 1;  // H_T_HQ19=65536 (16-17=-1)
 
       ssh = sh_hi >> 1;  // |sqrt_nrg| is in Qssh.
       sh = ssh - 14;
       tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
       tmp32c = sqrt_nrg + tmp32b;  // Qssh  (denominator)
-      tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0);  //Q24 (numerator)
+      tmp32a = varscaleQ14 * snrq;  // Q24 (numerator)
 
       sh = WebRtcSpl_NormW32(tmp32c);
       shft = 16 - sh;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
index 1270c1429b..aac927586c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
@@ -53,15 +53,6 @@ int32_t WebRtcIsacfix_CalculateResidualEnergyC(int lpc_order,
                                                int32_t* corr_coeffs,
                                                int* q_val_residual_energy);
 
-#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
-int32_t WebRtcIsacfix_CalculateResidualEnergyNeon(int lpc_order,
-                                                  int32_t q_val_corr,
-                                                  int q_val_polynomial,
-                                                  int16_t* a_polynomial,
-                                                  int32_t* corr_coeffs,
-                                                  int* q_val_residual_energy);
-#endif
-
 #if defined(MIPS_DSP_R2_LE)
 int32_t WebRtcIsacfix_CalculateResidualEnergyMIPS(int lpc_order,
                                                   int32_t q_val_corr,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_neon.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_neon.S
deleted file mode 100644
index a5955c27ab..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_neon.S
+++ /dev/null
@@ -1,173 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS.  All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ Contains a function for WebRtcIsacfix_CalculateResidualEnergyNeon() in
-@ iSAC codec, optimized for ARM Neon platform. Reference code in
-@ lpc_masking_model.c.
-
-#include "webrtc/system_wrappers/interface/asm_defines.h"
-
-GLOBAL_FUNCTION WebRtcIsacfix_CalculateResidualEnergyNeon
-.align  2
-
-@ int32_t WebRtcIsacfix_CalculateResidualEnergyNeon(int lpc_order,
-@                                                   int32_t q_val_corr,
-@                                                   int q_val_polynomial,
-@                                                   int16_t* a_polynomial,
-@                                                   int32_t* corr_coeffs,
-@                                                   int* q_val_residual_energy);
-DEFINE_FUNCTION WebRtcIsacfix_CalculateResidualEnergyNeon
-  push {r4-r11}
-
-  sub r13, r13, #16
-  str r1, [r13, #8]
-  str r2, [r13, #12]
-
-  mov r4, #1
-  vmov.s64 q11, #0            @ Initialize shift_internal.
-  vmov.s64 q13, #0            @ Initialize sum64.
-  vmov.s64 q10, #0
-  vmov.u8 d20[0], r4          @ Set q10 to 1.
-
-  cmp r0, #0
-  blt POST_LOOP_I
-
-  add r9, r3, r0, asl #1      @ &a_polynomial[lpc_order]
-  mov r6, #0                  @ Loop counter i.
-  ldr r11, [r13, #48]
-  sub r10, r0, #1
-  mov r7, r3                  @ &a_polynomial[0]
-  str r9, [r13, #4]
-
-LOOP_I:
-  ldr r2, [r11], #4            @ corr_coeffs[i]
-  vmov.s64 q15, #0            @ Initialize the sum64_tmp.
-  vdup.s32 d25, r2
-
-  cmp r0, r6                  @ Compare lpc_order to i.
-  movle r2, r6
-  ble POST_LOOP_J
-
-  mov r1, r6                  @ j = i;
-  mov r12, r7                  @ &a_polynomial[i]
-  mov r4, r3                  @ &a_polynomial[j - i]
-
-LOOP_J:
-  ldr r8, [r12], #4
-  ldr r5, [r4], #4
-  vmov.u32 d0[0], r8
-  vmov.u32 d1[0], r5
-  vmull.s16 q0, d0, d1
-  vmull.s32 q0, d0, d25
-  cmp r6, #0                  @ i == 0?
-  vshl.s64 q0, q11
-  beq SUM1
-  vshl.s64 q0, #1
-
-SUM1:
-  vqadd.s64 q14, q0, q15      @ Sum and test overflow.
-  add r1, r1, #2
-  bvc MOV1                    @ Skip the shift if there's no overflow.
-  vshr.s64 q0, #1
-  vshr.s64 q15, #1
-  vadd.s64 q14, q0, q15
-  vsub.s64 q11, q10
-
-MOV1:
-  cmp r0, r1                  @ Compare lpc_order to j.
-  vmov.s64 q15, q14
-  bgt LOOP_J
-
-  bic r1, r10, #1
-  add r2, r6, #2
-  add r2, r1, r2
-
-POST_LOOP_J:
-  vqadd.s64 q0, q13, q15      @ Sum and test overflow.
-  bvc MOV2                    @ Skip the shift if there's no overflow.
-  vshr.s64 q13, #1
-  vshr.s64 q15, #1
-  vadd.s64 q0, q13, q15
-  vsub.s64 q11, q10
-
-MOV2:
-  vmov.s64 q13, q0            @ update sum64.
-  cmp r2, r0
-  bne CHECK_LOOP_CONDITION
-
-  @ Last sample in the inner loop.
-  ldr r4, [r13, #4]
-  ldrsh r8, [r4]
-  ldrsh r12, [r9]
-  mul r8, r8, r12
-  vmov.s32 d0[0], r8
-  vmull.s32 q0, d0, d25
-  cmp r6, #0                  @ i == 0?
-  vshl.s64 q0, q11
-  beq SUM2
-  vshl.s64 q0, #1
-
-SUM2:
-  vqadd.s64 d1, d0, d26       @ Sum and test overflow.
-  bvc MOV3                    @ Skip the shift if there's no overflow.
-  vshr.s64 q13, #1
-  vshr.s64 d0, #1
-  vadd.s64 d1, d0, d26
-  vsub.s64 q11, q10
-
-MOV3:
-  vmov.s64 d26, d1            @ update sum64.
-
-CHECK_LOOP_CONDITION:
-  add r6, r6, #1
-  sub r9, r9, #2
-  cmp r0, r6                  @ Compare i to lpc_order.
-  sub r10, r10, #1
-  add r7, r7, #2
-  bge LOOP_I
-
-POST_LOOP_I:
-  mov r3, #0
-  vqadd.s64 d0, d26, d27      @ Sum and test overflow.
-  bvc GET_SHIFT_NORM          @ Skip the shift if there's no overflow.
-  vshr.s64 q13, #1
-  vadd.s64 d0, d26, d27
-  vsub.s64 q11, q10
-
-GET_SHIFT_NORM:
-  vcls.s32 d1, d0             @ Count leading extra sign bits.
-  vmov.32 r2, d1[1]           @ Store # of sign bits of only the 32 MSBs.
-  vmovl.s32 q1, d1
-  vshl.s64 d0, d3             @ d3 contains # of sign bits of the 32 MSBs.
-
-  vcls.s32 d1, d0             @ Count again the leading extra sign bits.
-  vmov.s32 r1, d1[1]          @ Store # of sign bits of only the 32 MSBs.
-  vmovl.s32 q1, d1
-  vshl.s64 d0, d3             @ d3 contains # of sign bits of the 32 MSBs.
-
-  vmov.s32 r0, d0[1]          @ residual_energy
-  vmov.s32 r3, d22[0]         @ shift_internal
-
-  @ Calculate the value for q_val_residual_energy.
-  ldr r4, [r13, #8]            @ q_val_corr
-  ldr r5, [r13, #12]           @ q_val_polynomial
-  sub r12, r4, #32
-  add r12, r12, r5, asl #1
-  add r1, r12, r1              @ add 1st part of shift_internal.
-  add r12, r1, r2              @ add 2nd part of shift_internal.
-  ldr r2, [r13, #52]
-  add r3, r12, r3              @ value for q_val_residual_energy.
-  str r3, [r2, #0]
-
-  add r13, r13, #16
-  pop {r4-r11}
-  bx  r14
-
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
index aaeff2c5c8..0984346019 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
@@ -10,7 +10,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 class LpcMaskingModelTest : public testing::Test {
@@ -58,11 +58,4 @@ class LpcMaskingModelTest : public testing::Test {
 
 TEST_F(LpcMaskingModelTest, CalculateResidualEnergyTest) {
   CalculateResidualEnergyTester(WebRtcIsacfix_CalculateResidualEnergyC);
-#ifdef WEBRTC_DETECT_ARM_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    CalculateResidualEnergyTester(WebRtcIsacfix_CalculateResidualEnergyNeon);
-  }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
-  CalculateResidualEnergyTester(WebRtcIsacfix_CalculateResidualEnergyNeon);
-#endif
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
index e6ca7987fd..3e7beede9f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
@@ -9,13 +9,8 @@
  */
 
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
-
-#ifdef WEBRTC_ARCH_ARM_NEON
-#include <arm_neon.h>
-#endif
-
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
 
 /* log2[0.2, 0.5, 0.98] in Q8 */
 static const int16_t kLogLagWinQ8[3] = {
@@ -212,7 +207,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
   /* bias towards pitch lag of previous frame */
   tmp32a = WebRtcIsacfix_Log2Q8((uint32_t) old_lagQ8) - 2304;
       // log2(0.5*oldlag) in Q8
-  tmp32b = WEBRTC_SPL_MUL_16_16_RSFT(oldgQ12,oldgQ12, 10); //Q12 & * 4.0;
+  tmp32b = oldgQ12 * oldgQ12 >> 10;  // Q12 & * 4.0;
   gain_bias16 = (int16_t) tmp32b;  //Q12
   if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12
 
@@ -222,12 +217,12 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
     if (crrvecQ8_1[k]>0) {
       tmp32b = WebRtcIsacfix_Log2Q8((uint32_t) (k + (PITCH_MIN_LAG/2-2)));
       tmp16a = (int16_t) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4
-      tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(tmp16a,tmp16a, 6); //Q10
+      tmp32c = tmp16a * tmp16a >> 6;  // Q10
       tmp16b = (int16_t) tmp32c; // Q10 & <8
-      tmp32d = WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 177 , 8); // mult with ln2 in Q8
+      tmp32d = tmp16b * 177 >> 8;  // mult with ln2 in Q8
       tmp16c = (int16_t) tmp32d; // Q10 & <4
       tmp16d = Exp2Q10((int16_t) -tmp16c); //Q10
-      tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(gain_bias16,tmp16d,13); // Q10  & * 0.5
+      tmp32c = gain_bias16 * tmp16d >> 13;  // Q10  & * 0.5
       bias16 = (int16_t) (1024 + tmp32c); // Q10
       tmp32b = WebRtcIsacfix_Log2Q8((uint32_t)bias16) - 2560;
           // Q10 in -> Q8 out with 10*2^8 offset
@@ -306,7 +301,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
       tmp32a= WebRtcIsacfix_Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8
       /* Bias towards short lags */
       /* log(pow(0.8, log(2.0 * *y )))/log(2.0) */
-      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -42, 8);
+      tmp32b = (int16_t)tmp32a * -42 >> 8;
       tmp32c= tmp32b + 256;
       *fyq += tmp32c;
       if (*fyq > corr_max32) {
@@ -330,7 +325,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
   {
     tmp32a = k << 7; // 0.5*k Q8
     tmp32b = tmp32a * 2 - ratq;  // Q8
-    tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32b, (int16_t) tmp32b, 8); // Q8
+    tmp32c = (int16_t)tmp32b * (int16_t)tmp32b >> 8;  // Q8
 
     tmp32b = tmp32c + (ratq >> 1);
         // (k-r)^2 + 0.5 * r  Q8
@@ -380,7 +375,7 @@ void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */
       /* Bias towards short lags */
       /* log(pow(0.8, log(2.0f * *y )))/log(2.0f) */
       tmp32a= WebRtcIsacfix_Log2Q8((uint32_t) *yq) - 2048; // offset 8*2^8
-      tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((int16_t) tmp32a, -82, 8);
+      tmp32b = (int16_t)tmp32a * -82 >> 8;
       tmp32c= tmp32b + 256;
       *fyq += tmp32c;
       if (*fyq > corr_max32) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
index da401e5f11..40f15c433c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
@@ -39,7 +39,7 @@ void WebRtcIsacfix_PitchFilter(int16_t *indatFix,
 
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
                                    int16_t gain,
-                                   int index,
+                                   size_t index,
                                    int16_t sign,
                                    int16_t* inputState,
                                    int16_t* outputBuff2,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
index 7ba7b69daa..18377dd370 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
@@ -10,12 +10,12 @@
 
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
 
-#ifdef WEBRTC_ARCH_ARM_NEON
+#ifdef WEBRTC_HAS_NEON
 #include <arm_neon.h>
 #endif
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
 
 extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x);
 
@@ -34,12 +34,8 @@ void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) {
   csum32 = 0;
   x = in + PITCH_MAX_LAG / 2 + 2;
   for (n = 0; n < PITCH_CORR_LEN2; n++) {
-    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[n],
-                                        (int16_t)in[n],
-                                        scaling);  // Q0
-    csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)x[n],
-                                        (int16_t)in[n],
-                                        scaling);  // Q0
+    ysum32 += in[n] * in[n] >> scaling;  // Q0
+    csum32 += x[n] * in[n] >> scaling;  // Q0
   }
   logcorQ8 += PITCH_LAG_SPAN2 - 1;
   lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32) >> 1; // Q8, sqrt(ysum)
@@ -57,13 +53,13 @@ void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) {
 
   for (k = 1; k < PITCH_LAG_SPAN2; k++) {
     inptr = &in[k];
-    ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[k - 1],
-                                        (int16_t)in[k - 1],
-                                        scaling);
-    ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[PITCH_CORR_LEN2 + k - 1],
-                                        (int16_t)in[PITCH_CORR_LEN2 + k - 1],
-                                        scaling);
-#ifdef WEBRTC_ARCH_ARM_NEON
+    ysum32 -= in[k - 1] * in[k - 1] >> scaling;
+    ysum32 += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1] >>
+        scaling;
+
+    // TODO(zhongwei.yao): Move this function into a separate NEON code file so
+    // that WEBRTC_DETECT_NEON could take advantage of it.
+#ifdef WEBRTC_HAS_NEON
     {
       int32_t vbuff[4];
       int32x4_t int_32x4_sum = vmovq_n_s32(0);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
index f5e7e7fb9d..bd26700058 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
 
 extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x);
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
index 06471beb33..65d099d36a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
@@ -13,7 +13,7 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
 
 // Number of segments in a pitch subframe.
 static const int kSegments = 5;
@@ -34,21 +34,8 @@ static const int16_t kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
   { 271, -743,  1570, -3320, 12963,  7301, -2292,  953, -325}
 };
 
-// Function prototype for pitch filtering.
-// TODO(Turaj): Add descriptions of input and output parameters.
-void WebRtcIsacfix_PitchFilterCore(int loopNumber,
-                                   int16_t gain,
-                                   int index,
-                                   int16_t sign,
-                                   int16_t* inputState,
-                                   int16_t* outputBuf2,
-                                   const int16_t* coefficient,
-                                   int16_t* inputBuf,
-                                   int16_t* outputBuf,
-                                   int* index2);
-
-static __inline int32_t CalcLrIntQ(int32_t fixVal,
-                                   int16_t qDomain) {
+static __inline size_t CalcLrIntQ(int16_t fixVal,
+                                  int16_t qDomain) {
   int32_t roundVal = 1 << (qDomain - 1);
 
   return (fixVal + roundVal) >> qDomain;
@@ -68,8 +55,7 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
   const int16_t Gain = 21299;     // 1.3 in Q14
   int16_t oldLagQ7;
   int16_t oldGainQ12, lagdeltaQ7, curLagQ7, gaindeltaQ12, curGainQ12;
-  int indW32 = 0, frcQQ = 0;
-  int32_t tmpW32;
+  size_t indW32 = 0, frcQQ = 0;
   const int16_t* fracoeffQQ = NULL;
 
   // Assumptions in ARM assembly for WebRtcIsacfix_PitchFilterCoreARM().
@@ -89,14 +75,12 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
 
     // Make output more periodic.
     for (k = 0; k < PITCH_SUBFRAMES; k++) {
-      gainsQ12[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-          gainsQ12[k], Gain, 14);
+      gainsQ12[k] = (int16_t)(gainsQ12[k] * Gain >> 14);
     }
   }
 
   // No interpolation if pitch lag step is big.
-  if ((WEBRTC_SPL_MUL_16_16_RSFT(lagsQ7[0], 3, 1) < oldLagQ7) ||
-      (lagsQ7[0] > WEBRTC_SPL_MUL_16_16_RSFT(oldLagQ7, 3, 1))) {
+  if (((lagsQ7[0] * 3 >> 1) < oldLagQ7) || (lagsQ7[0] > (oldLagQ7 * 3 >> 1))) {
     oldLagQ7 = lagsQ7[0];
     oldGainQ12 = gainsQ12[0];
   }
@@ -110,8 +94,7 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
                   lagdeltaQ7, kDivFactor, 15);
     curLagQ7 = oldLagQ7;
     gaindeltaQ12 = gainsQ12[k] - oldGainQ12;
-    gaindeltaQ12 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
-                    gaindeltaQ12, kDivFactor, 15);
+    gaindeltaQ12 = (int16_t)(gaindeltaQ12 * kDivFactor >> 15);
 
     curGainQ12 = oldGainQ12;
     oldLagQ7 = lagsQ7[k];
@@ -126,8 +109,7 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
       curGainQ12 += gaindeltaQ12;
       curLagQ7 += lagdeltaQ7;
       indW32 = CalcLrIntQ(curLagQ7, 7);
-      tmpW32 = (indW32 << 7) - curLagQ7;
-      frcQQ = (tmpW32 >> 4) + 4;
+      frcQQ = ((indW32 << 7) + 64 - curLagQ7) >> 4;
 
       if (frcQQ == PITCH_FRACS) {
         frcQQ = 0;
@@ -159,13 +141,15 @@ void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
                                     PitchFiltstr* pfp,
                                     int16_t* lagsQ7,
                                     int16_t* gainsQ12) {
-  int  k, n, m, ind, pos, pos3QQ;
+  int  k, n, m;
+  size_t ind, pos, pos3QQ;
 
   int16_t ubufQQ[PITCH_INTBUFFSIZE];
   int16_t oldLagQ7, lagdeltaQ7, curLagQ7;
   const int16_t* fracoeffQQ = NULL;
   int16_t scale;
-  int16_t cnt = 0, frcQQ, indW16 = 0, tmpW16;
+  int16_t cnt = 0, tmpW16;
+  size_t frcQQ, indW16 = 0;
   int32_t tmpW32, tmp2W32, csum1QQ, esumxQQ;
 
   // Set up buffer and states.
@@ -173,8 +157,7 @@ void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
   oldLagQ7 = pfp->oldlagQ7;
 
   // No interpolation if pitch lag step is big.
-  if ((WEBRTC_SPL_MUL_16_16_RSFT(lagsQ7[0], 3, 1) < oldLagQ7) ||
-      (lagsQ7[0] > WEBRTC_SPL_MUL_16_16_RSFT(oldLagQ7, 3, 1))) {
+  if (((lagsQ7[0] * 3 >> 1) < oldLagQ7) || (lagsQ7[0] > (oldLagQ7 * 3 >> 1))) {
     oldLagQ7 = lagsQ7[0];
   }
 
@@ -198,9 +181,8 @@ void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
     for (cnt = 0; cnt < kSegments; cnt++) {
       // Update parameters for each segment.
       curLagQ7 += lagdeltaQ7;
-      indW16 = (int16_t)CalcLrIntQ(curLagQ7, 7);
-      tmpW16 = (indW16 << 7) - curLagQ7;
-      frcQQ = (tmpW16 >> 4) + 4;
+      indW16 = CalcLrIntQ(curLagQ7, 7);
+      frcQQ = ((indW16 << 7) + 64 - curLagQ7) >> 4;
 
       if (frcQQ == PITCH_FRACS) {
         frcQQ = 0;
@@ -222,7 +204,7 @@ void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
 
         tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32);
         tmpW32 += 8192;
-        tmpW16 = (int16_t)(tmpW32 >> 14);
+        tmpW16 = tmpW32 >> 14;
         tmpW32 = tmpW16 * tmpW16;
 
         if ((tmp2W32 > 1073700000) || (csum1QQ > 1073700000) ||
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
index 57796b0e6e..bc18d44568 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
@@ -13,7 +13,7 @@
 @
 @ Output is bit-exact with the reference C code in pitch_filter.c.
 
-#include "webrtc/system_wrappers/interface/asm_defines.h"
+#include "webrtc/system_wrappers/include/asm_defines.h"
 #include "settings.h"
 
 GLOBAL_FUNCTION WebRtcIsacfix_PitchFilterCore
@@ -21,7 +21,7 @@ GLOBAL_FUNCTION WebRtcIsacfix_PitchFilterCore
 
 @ void WebRtcIsacfix_PitchFilterCore(int loopNumber,
 @                                    int16_t gain,
-@                                    int index,
+@                                    size_t index,
 @                                    int16_t sign,
 @                                    int16_t* inputState,
 @                                    int16_t* outputBuf2,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
index 5c956780e6..366eef034d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
@@ -18,7 +18,7 @@ static const int16_t kDampFilter[PITCH_DAMPORDER] = {
 
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
                                    int16_t gain,
-                                   int index,
+                                   size_t index,
                                    int16_t sign,
                                    int16_t* inputState,
                                    int16_t* outputBuf2,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c
index 8334f7eb18..0f390b8a4f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c
@@ -12,7 +12,7 @@
 
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
                                    int16_t gain,
-                                   int index,
+                                   size_t index,
                                    int16_t sign,
                                    int16_t* inputState,
                                    int16_t* outputBuf2,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
index 5f3e0c3036..278af7527d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
@@ -20,6 +20,7 @@
 
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/typedefs.h"
 
@@ -33,7 +34,7 @@ typedef struct Bitstreamstruct_dec {
   int16_t   full;             /* 0 - first byte in memory filled, second empty*/
   /* 1 - both bytes are empty (we just filled the previous memory */
 
-  int stream_size;  /* The size of stream. */
+  size_t stream_size;  /* The size of stream in bytes. */
 } Bitstr_dec;
 
 /* Bitstream struct for encoder */
@@ -177,8 +178,8 @@ typedef struct {
   int16_t pitchCycles;
   int16_t A;
   int16_t B;
-  int16_t pitchIndex;
-  int16_t stretchLag;
+  size_t pitchIndex;
+  size_t stretchLag;
   int16_t *prevPitchLP;                                  // [ FRAMESAMPLES/2 ]; saved 240
   int16_t seed;
 
@@ -245,9 +246,7 @@ typedef struct {
      bwe will assume the connection is over broadband network */
   int16_t   highSpeedSend;
 
-
-
-
+  IsacBandwidthInfo external_bw_info;
 } BwEstimatorstr;
 
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_mips.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_mips.c
index 656a77ede8..e5d35f2b73 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_mips.c
@@ -606,7 +606,7 @@ void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7,
   int32_t* outre2;
   int16_t* cosptr = (int16_t*)WebRtcIsacfix_kCosTab2;
   int16_t* sinptr = (int16_t*)WebRtcIsacfix_kSinTab2;
-  int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9, r10, max, max1;
+  int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9, max, max1;
 #if defined(MIPS_DSP_R1_LE)
   int32_t offset = FRAMESAMPLES - 4;
 #else  // #if defined(MIPS_DSP_R1_LE)
@@ -658,14 +658,14 @@ void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7,
     "subu           %[r9],        %[r9],        %[r8]         \n\t"
     "subu           %[r7],        %[r6],        %[r9]         \n\t"
     "addu           %[r6],        %[r6],        %[r9]         \n\t"
-    "sll            %[r10],       %[offset],    1             \n\t"
-    "addu           %[r10],       %[outre1],    %[r10]        \n\t"
     "sw             %[r7],        0(%[outre1])                \n\t"
     "absq_s.w       %[r7],        %[r7]                       \n\t"
-    "sw             %[r6],        4(%[r10])                   \n\t"
-    "absq_s.w       %[r6],        %[r6]                       \n\t"
     "slt            %[r8],        %[max],       %[r7]         \n\t"
     "movn           %[max],       %[r7],        %[r8]         \n\t"
+    "sll            %[r7],        %[offset],    1             \n\t"
+    "addu           %[r7],        %[outre1],    %[r7]         \n\t"
+    "sw             %[r6],        4(%[r7])                    \n\t"
+    "absq_s.w       %[r6],        %[r6]                       \n\t"
     "slt            %[r8],        %[max],       %[r6]         \n\t"
     "movn           %[max],       %[r6],        %[r8]         \n\t"
     "muleq_s.w.phl  %[r6],        %[r0],        %[r2]         \n\t"
@@ -682,10 +682,12 @@ void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7,
     "addu           %[r6],        %[r6],        %[r9]         \n\t"
     "sw             %[r7],        4(%[outre1])                \n\t"
     "absq_s.w       %[r7],        %[r7]                       \n\t"
-    "sw             %[r6],        0(%[r10])                   \n\t"
-    "absq_s.w       %[r6],        %[r6]                       \n\t"
     "slt            %[r8],        %[max],       %[r7]         \n\t"
     "movn           %[max],       %[r7],        %[r8]         \n\t"
+    "sll            %[r7],        %[offset],    1             \n\t"
+    "addu           %[r7],        %[outre1],    %[r7]         \n\t"
+    "sw             %[r6],        0(%[r7])                    \n\t"
+    "absq_s.w       %[r6],        %[r6]                       \n\t"
     "slt            %[r8],        %[max],       %[r6]         \n\t"
     "movn           %[max],       %[r6],        %[r8]         \n\t"
     "muleq_s.w.phr  %[r6],        %[r1],        %[r2]         \n\t"
@@ -702,14 +704,14 @@ void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7,
     "subu           %[r7],        %[r6],        %[r9]         \n\t"
     "addu           %[r6],        %[r9],        %[r6]         \n\t"
     "negu           %[r6],        %[r6]                       \n\t"
-    "sll            %[r10],       %[offset],    1             \n\t"
-    "addu           %[r10],       %[outre2],    %[r10]        \n\t"
     "sw             %[r7],        0(%[outre2])                \n\t"
     "absq_s.w       %[r7],        %[r7]                       \n\t"
-    "sw             %[r6],        4(%[r10])                   \n\t"
-    "absq_s.w       %[r6],        %[r6]                       \n\t"
     "slt            %[r8],        %[max],       %[r7]         \n\t"
     "movn           %[max],       %[r7],        %[r8]         \n\t"
+    "sll            %[r7],        %[offset],    1             \n\t"
+    "addu           %[r7],        %[outre2],    %[r7]         \n\t"
+    "sw             %[r6],        4(%[r7])                    \n\t"
+    "absq_s.w       %[r6],        %[r6]                       \n\t"
     "slt            %[r8],        %[max],       %[r6]         \n\t"
     "movn           %[max],       %[r6],        %[r8]         \n\t"
     "muleq_s.w.phl  %[r6],       %[r1],         %[r2]         \n\t"
@@ -728,10 +730,12 @@ void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7,
     "negu           %[r6],       %[r6]                        \n\t"
     "sw             %[r7],       4(%[outre2])                 \n\t"
     "absq_s.w       %[r7],       %[r7]                        \n\t"
-    "sw             %[r6],       0(%[r10])                    \n\t"
-    "absq_s.w       %[r6],       %[r6]                        \n\t"
     "slt            %[r8],       %[max],        %[r7]         \n\t"
     "movn           %[max],      %[r7],         %[r8]         \n\t"
+    "sll            %[r7],       %[offset],     1             \n\t"
+    "addu           %[r7],       %[outre2],     %[r7]         \n\t"
+    "sw             %[r6],       0(%[r7])                     \n\t"
+    "absq_s.w       %[r6],       %[r6]                        \n\t"
     "slt            %[r8],       %[max],        %[r6]         \n\t"
     "movn           %[max],      %[r6],         %[r8]         \n\t"
     "bgtz           %[k],        1b                           \n\t"
@@ -824,8 +828,8 @@ void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7,
       [offset] "+r" (offset), [k] "+r" (k), [r0] "=&r" (r0),
       [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3),
       [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6),
-      [r7] "=&r" (r7), [r10] "=&r" (r10),
-      [r8] "=&r" (r8), [r9] "=&r" (r9), [max] "=&r" (max)
+      [r7] "=&r" (r7), [r8] "=&r" (r8), [r9] "=&r" (r9),
+      [max] "=&r" (max)
     : [inreQ7] "r" (inreQ7), [inimQ7] "r" (inimQ7),
       [cosptr] "r" (cosptr), [sinptr] "r" (sinptr),
       [outre1Q16] "r" (outre1Q16), [outre2Q16] "r" (outre2Q16)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
deleted file mode 100644
index 07d6e56bc3..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
+++ /dev/null
@@ -1,645 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS.  All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-@ Reference code in transform.c. Bit not exact due to how rounding is
-@ done in C code and ARM instructions, but quality by assembly code is
-@ not worse.
-
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/system_wrappers/interface/asm_defines.h"
-
-GLOBAL_FUNCTION WebRtcIsacfix_Spec2TimeNeon
-GLOBAL_FUNCTION WebRtcIsacfix_Time2SpecNeon
-GLOBAL_LABEL WebRtcIsacfix_kSinTab1
-GLOBAL_LABEL WebRtcIsacfix_kCosTab1
-GLOBAL_LABEL WebRtcIsacfix_kSinTab2
-
-@ void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9,
-@                                  int16_t* inre2Q9,
-@                                  int16_t* outreQ7,
-@                                  int16_t* outimQ7);
-
-DEFINE_FUNCTION WebRtcIsacfix_Time2SpecNeon
-.align  2
-  push {r3-r11,lr}            @ need to push r4-r11, but push r3 too to keep
-                              @ stack 8-byte aligned
-  sub sp, sp, #(16 + FRAMESAMPLES * 4)
-
-  str r0, [sp]                @ inre1Q9
-  str r1, [sp, #4]            @ inre2Q9
-  str r2, [sp, #8]            @ outreQ7
-  str r3, [sp, #12]           @ outimQ7
-
-  mov r8, #(FRAMESAMPLES - 16)
-  add r12, r0, r8             @ &inreQ7[FRAMESAMPLES / 2 - 4]
-  add r11, r1, r8             @ &inimQ7[FRAMESAMPLES / 2 - 4]
-  add r4, sp, #16             @ tmpreQ16;
-  add r5, sp, #(16 + FRAMESAMPLES * 2)  @ tmpimQ16;
-
-  adr r9, WebRtcIsacfix_kCosTab1
-#if defined(__APPLE__) || defined(__clang__)
-  mov r6, #:lower16:(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
-#else
-  mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
-#endif
-  add r10, r9, r6             @ WebRtcIsacfix_kSinTab1
-
-  vmov.u32 q14, #0            @ Initialize the maximum values for tmpInIm.
-  vmov.u32 q15, #0            @ Initialize the maximum values for tmpInRe.
-  movw r6, #16921             @ 0.5 / sqrt(240) in Q19
-  lsl r6, #5                  @ Together with vqdmulh, net effect is ">> 26".
-  mov r8, #(FRAMESAMPLES / 2) @ loop counter
-  vdup.s32 q11, r6
-
-Time2Spec_TransformAndFindMax:
-@ Use ">> 26", instead of ">> 7", ">> 16" and then ">> 3" as in the C code.
-
-  subs r8, #8
-
-  vld1.16 {q0}, [r9, :64]!    @ WebRtcIsacfix_kCosTab1[]
-  vld1.16 {q2}, [r0]!         @ inre1Q9[]
-  vmull.s16 q8, d0, d4        @ WebRtcIsacfix_kCosTab1[k] * inre1Q9[k]
-  vld1.16 {q1}, [r10, :64]!   @ WebRtcIsacfix_kSinTab1[]
-  vmull.s16 q9, d1, d5        @ WebRtcIsacfix_kCosTab1[k] * inre1Q9[k]
-  vld1.16 {q3}, [r1]!         @ inre2Q9[]
-  vmlal.s16 q8, d2, d6        @ WebRtcIsacfix_kSinTab1[k] * inre2Q9[k]
-  vmlal.s16 q9, d3, d7        @ WebRtcIsacfix_kSinTab1[k] * inre2Q9[k]
-  vmull.s16 q12, d0, d6       @ WebRtcIsacfix_kCosTab1[k] * inre2Q9[k]
-  vmull.s16 q13, d1, d7       @ WebRtcIsacfix_kCosTab1[k] * inre2Q9[k]
-  vmlsl.s16 q12, d2, d4       @ WebRtcIsacfix_kSinTab1[k] * inre1Q9[k]
-  vmlsl.s16 q13, d3, d5       @ WebRtcIsacfix_kSinTab1[k] * inre1Q9[k]
-
-  vqdmulh.s32 q0, q8, q11     @ xrQ16 * factQ19
-  vqdmulh.s32 q1, q9, q11     @ xrQ16 * factQ19
-  vqdmulh.s32 q2, q12, q11    @ xrQ16 * factQ19
-  vqdmulh.s32 q3, q13, q11    @ xrQ16 * factQ19
-
-  @ Find the absolute maximum in the vectors and store them.
-  vabs.s32 q8, q0
-  vabs.s32 q9, q1
-  vabs.s32 q12, q2
-  vst1.32  {q0, q1}, [r4]!    @ tmpreQ16[k]
-  vabs.s32 q13, q3
-  vmax.u32 q14, q8            @ Use u32 so we don't lose the value 0x80000000.
-  vmax.u32 q15, q12
-  vst1.32  {q2, q3}, [r5]!    @ tmpimQ16[k]
-  vmax.u32 q15, q13
-  vmax.u32 q14, q9            @ Maximum for outre1Q16[].
-
-  bgt Time2Spec_TransformAndFindMax
-
-  @ Find the maximum value in the Neon registers
-  vmax.u32 d28, d29
-  vmax.u32 d30, d31
-  vpmax.u32 d28, d28, d28     @ Both 32 bits words hold the same value tmpInIm.
-  vpmax.u32 d30, d30, d30     @ Both 32 bits words hold the same value tmpInRe.
-  vmax.s32 d30, d28, d30      @ if (yrQ16 > xrQ16) {xrQ16 = yrQ16};
-
-  ldr r4, [sp]                @ inre1Q9
-  vcls.s32  d31, d30          @ sh = WebRtcSpl_NormW32(tmpInRe);
-  ldr r5, [sp, #4]            @ inre2Q9
-  vmov.i32  d30, #24
-  add r6, sp, #16             @ tmpreQ16;
-  vsub.s32  d31, d31, d30     @ sh = sh - 24;
-  add r7, sp, #(16 + FRAMESAMPLES * 2)  @ tmpimQ16;
-  vdup.s32  q8, d31[0]                  @ sh
-
-  mov r8, #(FRAMESAMPLES / 2) @ loop counter
-
-Time2Spec_PreFftShift:
-  subs r8, #16
-
-  vld1.32 {q0, q1}, [r6]!     @ tmpreQ16[]
-  vrshl.s32 q0, q0, q8
-  vld1.32 {q2, q3}, [r6]!     @ tmpreQ16[]
-  vrshl.s32 q1, q1, q8
-  vld1.32 {q10, q11}, [r7]!   @ tmpimQ16[]
-  vrshl.s32 q2, q2, q8
-  vld1.32 {q12, q13}, [r7]!   @ tmpimQ16[]
-  vrshl.s32 q3, q3, q8
-  vrshl.s32 q10, q10, q8
-  vrshl.s32 q11, q11, q8
-  vrshl.s32 q12, q12, q8
-  vrshl.s32 q13, q13, q8
-
-  vmovn.s32 d0, q0
-  vmovn.s32 d1, q1
-  vmovn.s32 d2, q2
-  vmovn.s32 d3, q3
-  vmovn.s32 d4, q10
-  vmovn.s32 d5, q11
-  vmovn.s32 d6, q12
-  vmovn.s32 d7, q13
-
-  vst1.16 {q0, q1}, [r4]!     @ inre1Q9[]
-  vst1.16 {q2, q3}, [r5]!     @ inre2Q9[]
-
-  bgt Time2Spec_PreFftShift
-
-  vmov.s32 r10, d16[0]        @ Store value of sh.
-  ldr r0, [sp]                @ inre1Q9
-  ldr r1, [sp, #4]            @ inre2Q9
-  mov r2, #-1
-  CALL_FUNCTION WebRtcIsacfix_FftRadix16Fastest
-
-  vdup.s32 q8, r10            @ sh
-  mov r8, #(FRAMESAMPLES - 8)
-  ldr r2, [sp, #8]            @ outreQ7
-  ldr r3, [sp, #12]           @ outimQ7
-  add r11, r2, r8             @ &outRe1Q16[FRAMESAMPLES / 2 - 4]
-  add r12, r3, r8             @ &outim2Q16[FRAMESAMPLES / 2 - 4]
-  ldr r6, [sp]                @ inre1Q9
-  ldr r7, [sp, #4]            @ inre2Q9
-  add r4, r6, r8              @ &inre1Q9[FRAMESAMPLES / 2 - 4]
-  add r5, r7, r8              @ &inre2Q9[FRAMESAMPLES / 2 - 4]
-  adr r10, WebRtcIsacfix_kSinTab2
-
-  add r9, r10, #(120*2 - 8)   @ &WebRtcIsacfix_kSinTab2[119 - 4]
-
-  vneg.s32 q15, q8            @ -sh
-  vmov.i32 q0, #23
-  vsub.s32 q15, q15, q0       @ -sh - 23
-
-  mov r8, #(FRAMESAMPLES / 4) @ loop counter
-
-  @ Pre-load variables.
-  vld1.16 {d2}, [r4]          @ inre1Q9[FRAMESAMPLES / 2 - 4 - i]
-  vld1.16 {d3}, [r5]          @ inre2Q9[FRAMESAMPLES / 2 - 4 - i]
-  vld1.16 {d0}, [r6]!         @ inre1Q9
-  vld1.16 {d1}, [r7]!         @ inre2Q9
-
-Time2Spec_PostFftTransform:
-@ By vshl, we effectively did "<< (-sh - 23)", instead of "<< (-sh)",
-@ ">> 14" and then ">> 9" as in the C code.
-
-  vld1.16 {d6}, [r9, :64]     @ kCosTab2[]
-  vneg.s16 d6, d6
-  vld1.16 {d7}, [r10, :64]!   @ WebRtcIsacfix_kSinTab2[]
-  vrev64.16 q1, q1            @ Reverse samples in 2nd half of xrQ16[].
-  vqadd.s16 d4, d0, d2        @ xrQ16
-  vqsub.s16 d5, d1, d3        @ xiQ16
-  vrev64.16 d6, d6
-
-  sub r9, #8                  @ Update pointers for kCosTab2[].
-  sub r4, #8                  @ Update pointers for inre1Q9[].
-  sub r5, #8                  @ Update pointers for inr22Q9[].
-  subs r8, #4                 @ Update loop counter.
-
-  vqadd.s16 d1, d1, d3        @ yrQ16
-  vqsub.s16 d0, d2, d0        @ yiQ16
-
-  vmull.s16 q12, d6, d4       @ kCosTab2[k] * xrQ16
-  vmlsl.s16 q12, d7, d5       @ WebRtcIsacfix_kSinTab2[k] * xiQ16
-  vmull.s16 q13, d7, d4       @ WebRtcIsacfix_kSinTab2[k] * xrQ16
-  vmlal.s16 q13, d6, d5       @ kCosTab2[k] * xiQ16
-  vmull.s16 q9, d7, d1        @ WebRtcIsacfix_kSinTab2[k] * yrQ16
-  vmlal.s16 q9, d6, d0        @ kCosTab2[k] * yiQ16
-  vmull.s16 q10, d7, d0       @ WebRtcIsacfix_kSinTab2[k] * yiQ16
-  vmlsl.s16 q10, d6, d1       @ kCosTab2[k] * yrQ16
-
-  vshl.s32 q12, q12, q15
-  vshl.s32 q13, q13, q15
-  vshl.s32 q9, q9, q15
-  vshl.s32 q10, q10, q15
-
-  vneg.s32 q8, q9
-  vld1.16 {d0}, [r6]!         @ inre1Q9
-  vmovn.s32 d24, q12
-  vld1.16 {d1}, [r7]!         @ inre2Q9
-  vmovn.s32 d25, q13
-  vld1.16 {d2}, [r4]          @ inre1Q9[FRAMESAMPLES / 2 - 4 - i]
-  vmovn.s32 d5, q10
-  vld1.16 {d3}, [r5]          @ inre2Q9[FRAMESAMPLES / 2 - 4 - i]
-  vmovn.s32 d4, q8
-  vst1.16  {d24}, [r2]!       @ outreQ7[k]
-  vrev64.16 q2, q2            @ Reverse the order of the samples.
-  vst1.16  {d25}, [r3]!       @ outimQ7[k]
-  vst1.16 {d4}, [r11]         @ outreQ7[FRAMESAMPLES / 2 - 1 - k]
-  vst1.16 {d5}, [r12]         @ outimQ7[FRAMESAMPLES / 2 - 1 - k]
-  sub r11, #8                 @ Update pointers for outreQ7[].
-  sub r12, #8                 @ Update pointers for outimQ7[].
-
-  bgt Time2Spec_PostFftTransform
-
-  add sp, sp, #(16 + FRAMESAMPLES * 4)
-  pop {r3-r11,pc}
-
-.align  8
-@ Cosine table 1 in Q14
-WebRtcIsacfix_kCosTab1:
-_WebRtcIsacfix_kCosTab1:  @ Label for iOS
-  .short 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315
-  .short 16294, 16270, 16244, 16214, 16182, 16147, 16110, 16069
-  .short 16026, 15980, 15931, 15880, 15826, 15769, 15709, 15647
-  .short 15582, 15515, 15444, 15371, 15296, 15218, 15137, 15053
-  .short 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295
-  .short 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380
-  .short 13255, 13128, 12998, 12867, 12733, 12597, 12458, 12318
-  .short 12176, 12031, 11885, 11736, 11585, 11433, 11278, 11121
-  .short 10963, 10803, 10641, 10477, 10311, 10143, 9974, 9803
-  .short 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377
-  .short 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859
-  .short 6664, 6467, 6270, 6071, 5872, 5671, 5469, 5266
-  .short 5063, 4859, 4653, 4447, 4240, 4033, 3825, 3616
-  .short 3406, 3196, 2986, 2775, 2563, 2351, 2139, 1926
-  .short 1713, 1499, 1285, 1072,  857,  643,  429,  214
-  .short 0, -214, -429, -643, -857, -1072, -1285, -1499
-  .short -1713, -1926, -2139, -2351, -2563, -2775, -2986, -3196
-  .short -3406, -3616, -3825, -4033, -4240, -4447, -4653, -4859
-  .short -5063, -5266, -5469, -5671, -5872, -6071, -6270, -6467
-  .short -6664, -6859, -7053, -7246, -7438, -7629, -7818, -8006
-  .short -8192, -8377, -8561, -8743, -8923, -9102, -9280, -9456
-  .short -9630, -9803, -9974, -10143, -10311, -10477, -10641, -10803
-  .short -10963, -11121, -11278, -11433, -11585, -11736, -11885, -12031
-  .short -12176, -12318, -12458, -12597, -12733, -12867, -12998, -13128
-  .short -13255, -13380, -13502, -13623, -13741, -13856, -13970, -14081
-  .short -14189, -14295, -14399, -14500, -14598, -14694, -14788, -14879
-  .short -14968, -15053, -15137, -15218, -15296, -15371, -15444, -15515
-  .short -15582, -15647, -15709, -15769, -15826, -15880, -15931, -15980
-  .short -16026, -16069, -16110, -16147, -16182, -16214, -16244, -16270
-  .short -16294, -16315, -16333, -16349, -16362, -16371, -16378, -16383
-
-.align  8
-@ Sine table 2 in Q14
-WebRtcIsacfix_kSinTab2:
-_WebRtcIsacfix_kSinTab2:  @ Label for iOS
-  .short 16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305
-  .short 16283, -16257, 16229, -16199, 16165, -16129, 16090, -16048
-  .short 16003, -15956, 15906, -15853, 15798, -15739, 15679, -15615
-  .short 15549, -15480, 15408, -15334, 15257, -15178, 15095, -15011
-  .short 14924, -14834, 14741, -14647, 14549, -14449, 14347, -14242
-  .short 14135, -14025, 13913, -13799, 13682, -13563, 13441, -13318
-  .short 13192, -13063, 12933, -12800, 12665, -12528, 12389, -12247
-  .short 12104, -11958, 11810, -11661, 11509, -11356, 11200, -11042
-  .short 10883, -10722, 10559, -10394, 10227, -10059, 9889, -9717
-  .short 9543, -9368, 9191, -9013, 8833, -8652, 8469, -8285
-  .short 8099, -7912, 7723, -7534, 7342, -7150, 6957, -6762
-  .short 6566, -6369, 6171, -5971, 5771, -5570, 5368, -5165
-  .short 4961, -4756, 4550, -4344, 4137, -3929, 3720, -3511
-  .short 3301, -3091, 2880, -2669, 2457, -2245, 2032, -1819
-  .short 1606, -1392, 1179, -965, 750, -536, 322, -107
-
-@ Table kCosTab2 was removed since its data is redundant with kSinTab2.
-
-.align  8
-@ Sine table 1 in Q14
-WebRtcIsacfix_kSinTab1:
-_WebRtcIsacfix_kSinTab1:  @ Label for iOS
-  .short 0, 214, 429, 643, 857, 1072, 1285, 1499
-  .short 1713, 1926, 2139, 2351, 2563, 2775, 2986, 3196
-  .short 3406, 3616, 3825, 4033, 4240, 4447, 4653, 4859
-  .short 5063, 5266, 5469, 5671, 5872, 6071, 6270, 6467
-  .short 6664, 6859, 7053, 7246, 7438, 7629, 7818, 8006
-  .short 8192, 8377, 8561, 8743, 8923, 9102, 9280, 9456
-  .short 9630, 9803, 9974, 10143, 10311, 10477, 10641, 10803
-  .short 10963, 11121, 11278, 11433, 11585, 11736, 11885, 12031
-  .short 12176, 12318, 12458, 12597, 12733, 12867, 12998, 13128
-  .short 13255, 13380, 13502, 13623, 13741, 13856, 13970, 14081
-  .short 14189, 14295, 14399, 14500, 14598, 14694, 14788, 14879
-  .short 14968, 15053, 15137, 15218, 15296, 15371, 15444, 15515
-  .short 15582, 15647, 15709, 15769, 15826, 15880, 15931, 15980
-  .short 16026, 16069, 16110, 16147, 16182, 16214, 16244, 16270
-  .short 16294, 16315, 16333, 16349, 16362, 16371, 16378, 16383
-  .short 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315
-  .short 16294, 16270, 16244, 16214, 16182, 16147, 16110, 16069
-  .short 16026, 15980, 15931, 15880, 15826, 15769, 15709, 15647
-  .short 15582, 15515, 15444, 15371, 15296, 15218, 15137, 15053
-  .short 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295
-  .short 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380
-  .short 13255, 13128, 12998, 12867, 12733, 12597, 12458, 12318
-  .short 12176, 12031, 11885, 11736, 11585, 11433, 11278, 11121
-  .short 10963, 10803, 10641, 10477, 10311, 10143, 9974, 9803
-  .short 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377
-  .short 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859
-  .short 6664, 6467, 6270, 6071, 5872, 5671, 5469, 5266
-  .short 5063, 4859, 4653, 4447, 4240, 4033, 3825, 3616
-  .short 3406, 3196, 2986, 2775, 2563, 2351, 2139, 1926
-  .short 1713, 1499, 1285, 1072, 857, 643, 429, 214
-
-@ void WebRtcIsacfix_Spec2TimeNeon(int16_t *inreQ7,
-@                                  int16_t *inimQ7,
-@                                  int32_t *outre1Q16,
-@                                  int32_t *outre2Q16);
-
-DEFINE_FUNCTION WebRtcIsacfix_Spec2TimeNeon
-.align  2
-  push {r3-r11,lr}            @ need to push r4-r11, but push r3 too to keep
-                              @ stack 8-byte aligned
-
-  sub sp, sp, #16
-  str r0, [sp]                @ inreQ7
-  str r1, [sp, #4]            @ inimQ7
-  str r2, [sp, #8]            @ outre1Q16
-  str r3, [sp, #12]           @ outre2Q16
-
-  mov r8, #(FRAMESAMPLES - 16)
-  add r12, r0, r8             @ &inreQ7[FRAMESAMPLES / 2 - 8]
-  add r11, r1, r8             @ &inimQ7[FRAMESAMPLES / 2 - 8]
-  add r4, r2, r8, lsl #1      @ &outRe1Q16[FRAMESAMPLES / 2 - 8]
-  add r6, r3, r8, lsl #1      @ &outRe2Q16[FRAMESAMPLES / 2 - 8]
-
-  mov r8, #(FRAMESAMPLES / 2) @ loop counter
-  adr r10, WebRtcIsacfix_kSinTab2
-  add r9, r10, #(120*2 - 16)  @ &WebRtcIsacfix_kSinTab2[119 - 8]
-
-  vpush {q4-q7}
-
-  mov r5, #-32
-  mov r7, #-16
-  vmov.u32 q6, #0             @ Initialize the maximum values for tmpInIm.
-  vmov.u32 q7, #0             @ Initialize the maximum values for tmpInRe.
-
-TransformAndFindMax:
-@ Use ">> 5", instead of "<< 9" and then ">> 14" as in the C code.
-@ Bit-exact.
-
-  subs r8, #16
-
-  vld1.16 {q0}, [r9, :64]     @ kCosTab2[]
-  sub r9, #16
-  vld1.16 {q2}, [r0]!         @ inreQ7[]
-  vneg.s16 q0, q0
-  vld1.16 {q3}, [r1]!         @ inimQ7[]
-  vrev64.16 d0, d0
-  vrev64.16 d1, d1
-  vld1.16 {q1}, [r10, :64]!   @ WebRtcIsacfix_kSinTab2[]
-  vswp d0, d1
-
-  vmull.s16 q8, d2, d6        @ WebRtcIsacfix_kSinTab2[k] * inimQ7[k]
-  vmull.s16 q9, d3, d7        @ WebRtcIsacfix_kSinTab2[k] * inimQ7[k]
-  vmlal.s16 q8, d0, d4        @ kCosTab2[k] * inreQ7[k]
-  vmlal.s16 q9, d1, d5        @ kCosTab2[k] * inreQ7[k]
-  vmull.s16 q12, d0, d6       @ kCosTab2[k] * inimQ7[k]
-  vmull.s16 q13, d1, d7       @ kCosTab2[k] * inimQ7[k]
-  vmlsl.s16 q12, d2, d4       @ WebRtcIsacfix_kSinTab2[k] * inreQ7[k]
-  vmlsl.s16 q13, d3, d5       @ WebRtcIsacfix_kSinTab2[k] * inreQ7[k]
-
-  vld1.16 {q2}, [r11], r7     @ inimQ7[FRAMESAMPLES / 2 - 8 + i]
-  vld1.16 {q3}, [r12], r7     @ inreQ7[FRAMESAMPLES / 2 - 8 + i]
-
-  vrev64.16 q2, q2            @ Reverse the order of the samples
-  vrev64.16 q3, q3            @ Reverse the order of the samples
-
-  vmull.s16 q14, d2, d5       @ WebRtcIsacfix_kSinTab2[k] * inimQ7[k]
-  vmull.s16 q15, d3, d4       @ WebRtcIsacfix_kSinTab2[k] * inimQ7[k]
-  vmlsl.s16 q14, d0, d7       @ q14 -= kCosTab2[k] * inreQ7[k]
-  vmlsl.s16 q15, d1, d6       @ q15 -= kCosTab2[k] * inreQ7[k]
-
-  vmull.s16 q10, d0, d5       @ kCosTab2[k] * inimQ7[]
-  vmull.s16 q11, d1, d4       @ kCosTab2[k] * inimQ7[]
-  vmlal.s16 q10, d2, d7       @ q10 += WebRtcIsacfix_kSinTab2[k] * inreQ7[]
-  vmlal.s16 q11, d3, d6       @ q11 += WebRtcIsacfix_kSinTab2[k] * inreQ7[]
-
-  vshr.s32 q8, q8, #5         @ xrQ16
-  vshr.s32 q9, q9, #5         @ xrQ16
-  vshr.s32 q12, q12, #5       @ xiQ16
-  vshr.s32 q13, q13, #5       @ xiQ16
-  vshr.s32 q14, q14, #5       @ yiQ16
-  vshr.s32 q15, q15, #5       @ yiQ16
-
-  vneg.s32 q10, q10
-  vneg.s32 q11, q11
-
-  @ xrQ16 - yiQ16
-  vsub.s32 q0, q8, q14
-  vsub.s32 q1, q9, q15
-
-  vshr.s32 q10, q10, #5       @ yrQ16
-  vshr.s32 q11, q11, #5       @ yrQ16
-
-  @ xrQ16 + yiQ16
-  vadd.s32 q3, q8, q14
-  vadd.s32 q2, q9, q15
-
-  @ yrQ16 + xiQ16
-  vadd.s32 q4, q10, q12
-  vadd.s32 q5, q11, q13
-
-  @ yrQ16 - xiQ16
-  vsub.s32 q8, q11, q13
-  vsub.s32 q9, q10, q12
-
-  @ Reverse the order of the samples
-  vrev64.32 q2, q2
-  vrev64.32 q3, q3
-  vrev64.32 q8, q8
-  vrev64.32 q9, q9
-  vswp d4, d5
-  vswp d6, d7
-
-  vst1.32  {q0, q1}, [r2]!    @ outre1Q16[k]
-  vswp d16, d17
-  vswp d18, d19
-  vst1.32  {q2, q3}, [r4], r5 @ outre1Q16[FRAMESAMPLES / 2 - 1 - k]
-
-  @ Find the absolute maximum in the vectors and store them in q6 and q7.
-  vabs.s32 q10, q0
-  vabs.s32 q14, q4
-  vabs.s32 q11, q1
-  vabs.s32 q15, q5
-  vabs.s32 q12, q2
-  vmax.u32 q6, q10            @ Use u32 so we don't lose the value 0x80000000.
-  vmax.u32 q7, q14            @ Maximum for outre2Q16[].
-  vabs.s32 q0, q8
-  vmax.u32 q6, q11            @ Maximum for outre1Q16[].
-  vmax.u32 q7, q15
-  vabs.s32 q13, q3
-  vmax.u32 q6, q12
-  vmax.u32 q7, q0
-  vabs.s32 q1, q9
-  vst1.32  {q4, q5}, [r3]!    @ outre2Q16[k]
-  vst1.32  {q8, q9}, [r6], r5 @ outre2Q16[FRAMESAMPLES / 2 - 1 - k]
-  vmax.u32 q6, q13
-  vmax.u32 q7, q1
-
-  bgt TransformAndFindMax
-
-  adr r10, WebRtcIsacfix_kSinTab1
-#if defined(__APPLE__) || defined(__clang__)
-  mov r2, #:lower16:(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
-#else
-  mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
-#endif
-
-  sub r11, r10, r2            @ WebRtcIsacfix_kCosTab1
-
-  @ Find the maximum value in the Neon registers
-  vmax.u32 d12, d13
-  vmax.u32 d14, d15
-  vpmax.u32 d12, d12, d12     @ Both 32 bits words hold the same value tmpInIm.
-  vpmax.u32 d14, d14, d14     @ Both 32 bits words hold the same value tmpInRe.
-  vmax.s32 d0, d12, d14       @ if (tmpInIm>tmpInRe) tmpInRe = tmpInIm;
-
-  vpop {q4-q7}
-
-  ldr r4, [sp]                @ inreQ7
-  vcls.s32  d1, d0            @ sh = WebRtcSpl_NormW32(tmpInRe);
-  ldr r5, [sp, #4]            @ inimQ7
-  vmov.i32  d0, #24           @ sh = sh-24;
-  ldr r6, [sp, #8]            @ outre1Q16
-  vsub.s32  d1, d1, d0
-  ldr r7, [sp, #12]           @ outre2Q16
-  vdup.s32  q8, d1[0]         @ sh
-
-  mov r8, #(FRAMESAMPLES / 2)
-
-PreFftShift:
-  subs r8, #16
-  vld1.32 {q0, q1}, [r6]!     @ outre1Q16[]
-  vld1.32 {q2, q3}, [r6]!     @ outre1Q16[]
-  vrshl.s32 q0, q0, q8
-  vrshl.s32 q1, q1, q8
-  vrshl.s32 q2, q2, q8
-  vrshl.s32 q3, q3, q8
-  vld1.32 {q10, q11}, [r7]!   @ outre2Q16[]
-  vld1.32 {q12, q13}, [r7]!   @ outre2Q16[]
-  vrshl.s32 q10, q10, q8
-  vrshl.s32 q11, q11, q8
-  vrshl.s32 q12, q12, q8
-  vrshl.s32 q13, q13, q8
-
-  vmovn.s32 d0, q0
-  vmovn.s32 d1, q1
-  vmovn.s32 d2, q2
-  vmovn.s32 d3, q3
-  vmovn.s32 d4, q10
-  vmovn.s32 d5, q11
-  vmovn.s32 d6, q12
-  vmovn.s32 d7, q13
-
-  vst1.16 {q0, q1}, [r4]!     @ inreQ7[]
-  vst1.16 {q2, q3}, [r5]!     @ inimQ7[]
-
-  bgt PreFftShift
-
-  vmov.s32 r8, d16[0]         @ Store value of sh.
-  ldr r0, [sp]                @ inreQ7
-  ldr r1, [sp, #4]            @ inimQ7
-  mov r2, #1
-  CALL_FUNCTION WebRtcIsacfix_FftRadix16Fastest
-
-  vdup.s32 q8, r8             @ sh
-  mov r9, r11                 @ WebRtcIsacfix_kCosTab1
-  ldr r4, [sp]                @ inreQ7
-  ldr r5, [sp, #4]            @ inimQ7
-  ldr r6, [sp, #8]            @ outre1Q16
-  ldr r7, [sp, #12]           @ outre2Q16
-  mov r8, #(FRAMESAMPLES / 2)
-  vneg.s32 q15, q8            @ -sh
-  movw r0, #273
-  lsl r0, #15                 @ Together with vqdmulh, net effect is ">> 16".
-  vdup.s32 q14, r0
-
-PostFftShiftDivide:
-  subs r8, #16
-
-  vld1.16 {q0, q1}, [r4]!     @ inreQ7
-  vmovl.s16 q10, d0
-  vmovl.s16 q11, d1
-  vld1.16 {q2, q3}, [r5]!     @ inimQ7
-  vmovl.s16 q8, d2
-  vmovl.s16 q9, d3
-
-  vshl.s32 q10, q10, q15
-  vshl.s32 q11, q11, q15
-  vshl.s32 q8, q8, q15
-  vshl.s32 q9, q9, q15
-
-  vqdmulh.s32 q10, q10, q14
-  vqdmulh.s32 q11, q11, q14
-  vqdmulh.s32 q8, q8, q14
-  vqdmulh.s32 q9, q9, q14
-
-  vmovl.s16 q0, d4
-  vmovl.s16 q1, d5
-  vmovl.s16 q2, d6
-  vmovl.s16 q3, d7
-
-  vshl.s32 q0, q0, q15
-  vshl.s32 q1, q1, q15
-  vshl.s32 q2, q2, q15
-  vshl.s32 q3, q3, q15
-
-  @ WEBRTC_SPL_MUL_16_32_RSFT16(273, outre2Q16[k])
-  vqdmulh.s32 q0, q0, q14
-  vqdmulh.s32 q1, q1, q14
-  vst1.32 {q10, q11}, [r6]!   @ outre1Q16[]
-  vqdmulh.s32 q2, q2, q14
-  vqdmulh.s32 q3, q3, q14
-  vst1.32 {q8, q9}, [r6]!     @ outre1Q16[]
-  vst1.32 {q0, q1}, [r7]!     @ outre2Q16[]
-  vst1.32 {q2, q3}, [r7]!     @ outre2Q16[]
-
-  bgt PostFftShiftDivide
-
-  mov r8, #(FRAMESAMPLES / 2)
-  ldr r2, [sp, #8]            @ outre1Q16
-  ldr r3, [sp, #12]           @ outre2Q16
-  movw r0, #31727
-  lsl r0, #16                 @ With vqdmulh and vrshrn, net effect is ">> 25".
-
-DemodulateAndSeparate:
-  subs r8, #8
-
-  vld1.16 {q0}, [r9, :64]!    @ WebRtcIsacfix_kCosTab1[]
-  vmovl.s16 q10, d0           @ WebRtcIsacfix_kCosTab1[]
-  vld1.16 {q1}, [r10, :64]!   @ WebRtcIsacfix_kSinTab1[]
-  vmovl.s16 q11, d1           @ WebRtcIsacfix_kCosTab1[]
-  vld1.32 {q2, q3}, [r2]      @ outre1Q16
-  vmovl.s16 q12, d2           @ WebRtcIsacfix_kSinTab1[]
-  vld1.32 {q14, q15}, [r3]    @ outre2Q16
-  vmovl.s16 q13, d3           @ WebRtcIsacfix_kSinTab1[]
-
-  vmull.s32 q0, d20, d4       @ WebRtcIsacfix_kCosTab1[k] * outre1Q16[k]
-  vmull.s32 q1, d21, d5       @ WebRtcIsacfix_kCosTab1[k] * outre1Q16[k]
-  vmull.s32 q8, d22, d6       @ WebRtcIsacfix_kCosTab1[k] * outre1Q16[k]
-  vmull.s32 q9, d23, d7       @ WebRtcIsacfix_kCosTab1[k] * outre1Q16[k]
-
-  vmlsl.s32 q0, d24, d28      @ += WebRtcIsacfix_kSinTab1[k] * outre2Q16[k]
-  vmlsl.s32 q1, d25, d29      @ += WebRtcIsacfix_kSinTab1[k] * outre2Q16[k]
-  vmlsl.s32 q8, d26, d30      @ += WebRtcIsacfix_kSinTab1[k] * outre2Q16[k]
-  vmlsl.s32 q9, d27, d31      @ += WebRtcIsacfix_kSinTab1[k] * outre2Q16[k]
-
-  vrshrn.s64 d0, q0, #10      @ xrQ16
-  vrshrn.s64 d1, q1, #10      @ xrQ16
-  vrshrn.s64 d2, q8, #10      @ xrQ16
-  vrshrn.s64 d3, q9, #10      @ xrQ16
-
-  vmull.s32 q8, d20, d28      @ WebRtcIsacfix_kCosTab1[k] * outre2Q16[k]
-  vmull.s32 q9, d21, d29      @ WebRtcIsacfix_kCosTab1[k] * outre2Q16[k]
-  vmull.s32 q14, d22, d30     @ WebRtcIsacfix_kCosTab1[k] * outre2Q16[k]
-  vmull.s32 q15, d23, d31     @ WebRtcIsacfix_kCosTab1[k] * outre2Q16[k]
-
-  vmlal.s32 q8, d24, d4       @ += WebRtcIsacfix_kSinTab1[k] * outre1Q16[k]
-  vmlal.s32 q9, d25, d5       @ += WebRtcIsacfix_kSinTab1[k] * outre1Q16[k]
-  vmlal.s32 q14, d26, d6      @ += WebRtcIsacfix_kSinTab1[k] * outre1Q16[k]
-  vmlal.s32 q15, d27, d7      @ += WebRtcIsacfix_kSinTab1[k] * outre1Q16[k]
-
-  vdup.s32 q11, r0            @ generic -> Neon doesn't cost extra cycles.
-
-  vrshrn.s64 d24, q8, #10     @ xiQ16
-  vrshrn.s64 d25, q9, #10     @ xiQ16
-  vqdmulh.s32 q0, q0, q11
-  vrshrn.s64 d26, q14, #10    @ xiQ16
-  vrshrn.s64 d27, q15, #10    @ xiQ16
-
-  @ WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xrQ16)
-  @ WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xiQ16)
-
-  vqdmulh.s32 q1, q1, q11
-  vqdmulh.s32 q2, q12, q11
-  vqdmulh.s32 q3, q13, q11
-
-  vst1.16 {q0, q1}, [r2]!     @ outre1Q16[]
-  vst1.16 {q2, q3}, [r3]!     @ outre2Q16[]
-
-  bgt DemodulateAndSeparate
-
-  add sp, sp, #16
-  pop {r3-r11,pc}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.c
index b4e95b2077..f0cbd5d075 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.c
@@ -432,10 +432,10 @@ static inline void PostShiftAndDivideAndDemodulateNeon(int16_t* inre,
     xi3 = vmlal_s32(xi3, vget_high_s32(tmpi_1), vget_high_s32(outre1_1));
 #endif
 
-    outre1_0 = vcombine_s32(vshrn_n_s64(xr0, 10), vshrn_n_s64(xr1, 10));
-    outre2_0 = vcombine_s32(vshrn_n_s64(xi0, 10), vshrn_n_s64(xi1, 10));
-    outre1_1 = vcombine_s32(vshrn_n_s64(xr2, 10), vshrn_n_s64(xr3, 10));
-    outre2_1 = vcombine_s32(vshrn_n_s64(xi2, 10), vshrn_n_s64(xi3, 10));
+    outre1_0 = vcombine_s32(vrshrn_n_s64(xr0, 10), vrshrn_n_s64(xr1, 10));
+    outre2_0 = vcombine_s32(vrshrn_n_s64(xi0, 10), vrshrn_n_s64(xi1, 10));
+    outre1_1 = vcombine_s32(vrshrn_n_s64(xr2, 10), vrshrn_n_s64(xr3, 10));
+    outre2_1 = vcombine_s32(vrshrn_n_s64(xi2, 10), vrshrn_n_s64(xi3, 10));
     outre1_0 = vqdmulhq_s32(outre1_0, fact);
     outre2_0 = vqdmulhq_s32(outre2_0, fact);
     outre1_1 = vqdmulhq_s32(outre1_1, fact);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
index ee96b8e357..8f89fb8f80 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
@@ -16,7 +16,6 @@
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/typedefs.h"
 
-#if !(defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON)
 /* Cosine table 1 in Q14. */
 const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = {
   16384,  16383,  16378,  16371,  16362,  16349,  16333,  16315,  16294,  16270,
@@ -90,7 +89,6 @@ const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4] = {
   4137,  -3929,  3720,  -3511,  3301,  -3091,  2880,  -2669,  2457,  -2245,
   2032,  -1819,  1606,  -1392,  1179,   -965,   750,   -536,   322,   -107
 };
-#endif
 
 #if defined(MIPS32_LE)
 /* Cosine table 2 in Q14. Used only on MIPS platforms. */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
index 855b5feafd..58d890011f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
@@ -9,10 +9,11 @@
  */
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
 static const int kSamples = FRAMESAMPLES/2;
-static int32_t spec2time_out_expected_1[kSamples] = {-3366470, -2285227,
+static const int32_t spec2time_out_expected_1[kSamples] = {
+  -3366470, -2285227,
   -3415765, -2310215, -3118030, -2222470, -3030254, -2192091, -3423170,
   -2216041, -3305541, -2171936, -3195767, -2095779, -3153304, -2157560,
   -3071167, -2032108, -3101190, -1972016, -3103824, -2089118, -3139811,
@@ -45,7 +46,8 @@ static int32_t spec2time_out_expected_1[kSamples] = {-3366470, -2285227,
   3315952, 2406651, 3344038, 2370199, 3368980, 2144361, 3305030, 2183803,
   3401450, 2523102, 3405463, 2452475, 3463355, 2421678, 3551968, 2431949,
   3477251, 2148125, 3244489, 2174090};
-static int32_t spec2time_out_expected_2[kSamples]= {1691694, -2499988, -2035547,
+static const int32_t spec2time_out_expected_2[kSamples] = {
+  1691694, -2499988, -2035547,
   1060469, 988634, -2044502, -306271, 2041000, 201454, -2289456, 93694,
   2129427, -369152, -1887834, 860796, 2089102, -929424, -1673956, 1395291,
   1785651, -1619673, -1380109, 1963449, 1093311, -2111007, -840456,
@@ -79,7 +81,8 @@ static int32_t spec2time_out_expected_2[kSamples]= {1691694, -2499988, -2035547,
   -1588643, 1754528, 816552, -2376303, -1099167, 1864999, 122477,
   -2422762, -400027, 1889228, -579916, -2490353, 287139, 2011318,
   -1176657, -2502978, 812896, 1116502, -1940211};
-static int16_t time2spec_out_expected_1[kSamples]= {20342, 23889, -10063, -9419,
+static const int16_t time2spec_out_expected_1[kSamples] = {
+  20342, 23889, -10063, -9419,
   3242, 7280, -2012, -5029, 332, 4478, -97, -3244, -891, 3117, 773, -2204,
   -1335, 2009, 1236, -1469, -1562, 1277, 1366, -815, -1619, 599, 1449, -177,
   -1507, 116, 1294, 263, -1338, -244, 1059, 553, -1045, -549, 829, 826,
@@ -99,7 +102,8 @@ static int16_t time2spec_out_expected_1[kSamples]= {20342, 23889, -10063, -9419,
   -562, 627, -550, 560, -606, 529, -584, 568, -503, 532, -463, 512, -440,
   399, -457, 437, -349, 278, -317, 257, -220, 163, -8, -61, 18, -161, 367,
   -1306};
-static int16_t time2spec_out_expected_2[kSamples]= {14283, -11552, -15335, 6626,
+static const int16_t time2spec_out_expected_2[kSamples] = {
+  14283, -11552, -15335, 6626,
   7554, -2150, -6309, 1307, 4523, -4, -3908, -314, 3001, 914, -2715, -1042,
   2094, 1272, -1715, -1399, 1263, 1508, -1021, -1534, 735, 1595, -439, -1447,
   155, 1433, 22, -1325, -268, 1205, 424, -1030, -608, 950, 643, -733, -787,
@@ -175,22 +179,22 @@ class TransformTest : public testing::Test {
 
 TEST_F(TransformTest, Time2SpecTest) {
   Time2SpecTester(WebRtcIsacfix_Time2SpecC);
-#ifdef WEBRTC_DETECT_ARM_NEON
+#ifdef WEBRTC_DETECT_NEON
   if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
     Time2SpecTester(WebRtcIsacfix_Time2SpecNeon);
   }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   Time2SpecTester(WebRtcIsacfix_Time2SpecNeon);
 #endif
 }
 
 TEST_F(TransformTest, Spec2TimeTest) {
   Spec2TimeTester(WebRtcIsacfix_Spec2TimeC);
-#ifdef WEBRTC_DETECT_ARM_NEON
+#ifdef WEBRTC_DETECT_NEON
   if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
     Spec2TimeTester(WebRtcIsacfix_Spec2TimeNeon);
   }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   Spec2TimeTester(WebRtcIsacfix_Spec2TimeNeon);
 #endif
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index bb76257569..32f36c5261 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
@@ -26,8 +26,8 @@ class IsacSpeedTest : public AudioCodecSpeedTest {
   void SetUp() override;
   void TearDown() override;
   virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             int max_bytes, int* encoded_bytes);
-  virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+                             size_t max_bytes, size_t* encoded_bytes);
+  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
                              int16_t* out_data);
   ISACFIX_MainStruct *ISACFIX_main_inst_;
 };
@@ -43,12 +43,12 @@ void IsacSpeedTest::SetUp() {
   AudioCodecSpeedTest::SetUp();
 
   // Check whether the allocated buffer for the bit stream is large enough.
-  EXPECT_GE(max_bytes_, STREAM_MAXW16_60MS);
+  EXPECT_GE(max_bytes_, static_cast<size_t>(STREAM_MAXW16_60MS));
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_));
   EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1));
-  EXPECT_EQ(0, WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_));
+  WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_);
   // Set bitrate and block length.
   EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_,
                                      block_duration_ms_));
@@ -61,35 +61,38 @@ void IsacSpeedTest::TearDown() {
 }
 
 float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                                  int max_bytes, int* encoded_bytes) {
+                                  size_t max_bytes, size_t* encoded_bytes) {
   // ISAC takes 10 ms everycall
   const int subblocks = block_duration_ms_ / 10;
   const int subblock_length = 10 * input_sampling_khz_;
-  int value;
+  int value = 0;
 
   clock_t clocks = clock();
   size_t pointer = 0;
   for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
     value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
                                  bit_stream);
+    if (idx == subblocks - 1)
+      EXPECT_GT(value, 0);
+    else
+      EXPECT_EQ(0, value);
   }
   clocks = clock() - clocks;
-  EXPECT_GT(value, 0);
-  assert(value <= max_bytes);
-  *encoded_bytes = value;
+  *encoded_bytes = static_cast<size_t>(value);
+  assert(*encoded_bytes <= max_bytes);
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
-float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
+                                  size_t encoded_bytes,
                                   int16_t* out_data) {
   int value;
   int16_t audio_type;
   clock_t clocks = clock();
-  value = WebRtcIsacfix_Decode(ISACFIX_main_inst_,
-                               bit_stream,
-                               encoded_bytes, out_data, &audio_type);
+  value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
+                               out_data, &audio_type);
   clocks = clock() - clocks;
-  EXPECT_EQ(output_length_sample_, value);
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index ba50b0c678..c61449a3ef 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -14,7 +14,7 @@
 #include <time.h>
 #include <ctype.h>
 
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 #include "webrtc/test/testsupport/perf_test.h"
 
 // TODO(kma): Clean up the code and change benchmarking the whole codec to
@@ -50,7 +50,7 @@ typedef struct {
 } BottleNeckModel;
 
 void get_arrival_time(int current_framesamples,   /* samples */
-                      int packet_size,            /* bytes */
+                      size_t packet_size,         /* bytes */
                       int bottleneck,             /* excluding headers; bits/s */
                       BottleNeckModel *BN_data)
 {
@@ -62,7 +62,8 @@ void get_arrival_time(int current_framesamples,   /* samples */
   /* everything in samples */
   BN_data->sample_count = BN_data->sample_count + current_framesamples;
 
-  BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+  BN_data->arrival_time += static_cast<uint32_t>(
+      ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate));
   BN_data->send_time += current_framesamples;
 
   if (BN_data->arrival_time < BN_data->sample_count)
@@ -98,22 +99,25 @@ int main(int argc, char* argv[])
   FILE *inp, *outp, *f_bn, *outbits;
   int endfile;
 
-  int i, errtype, h = 0, k, packetLossPercent = 0;
+  size_t i;
+  int errtype, h = 0, k, packetLossPercent = 0;
   int16_t CodingMode;
   int16_t bottleneck;
-  int16_t framesize = 30;           /* ms */
+  int framesize = 30;           /* ms */
   int cur_framesmpls, err = 0, lostPackets = 0;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
-  int16_t stream_len = 0;
-  int16_t framecnt, declen = 0;
+  int stream_len_int = 0;
+  size_t stream_len = 0;
+  int16_t framecnt;
+  int declen = 0;
   int16_t shortdata[FRAMESAMPLES_10ms];
   int16_t decoded[MAX_FRAMESAMPLES];
   uint16_t streamdata[500];
   int16_t speechType[1];
-  int16_t prevFrameSize = 1;
+  size_t prevFrameSize = 1;
   int16_t rateBPS = 0;
   int16_t fixedFL = 0;
   int16_t payloadSize = 0;
@@ -231,7 +235,7 @@ int main(int argc, char* argv[])
   CodingMode = 0;
   testNum = 0;
   testCE = 0;
-  for (i = 1; i < argc-2;i++) {
+  for (i = 1; i + 2 < static_cast<size_t>(argc); i++) {
     /* Instantaneous mode */
     if (!strcmp ("-I", argv[i])) {
       printf("\nInstantaneous BottleNeck\n");
@@ -535,12 +539,7 @@ int main(int argc, char* argv[])
         printf("\n\n Error in encoderinit: %d.\n\n", errtype);
       }
 
-      err = WebRtcIsacfix_DecoderInit(ISAC_main_inst);
-      /* Error check */
-      if (err < 0) {
-        errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
-        printf("\n\n Error in decoderinit: %d.\n\n", errtype);
-      }
+      WebRtcIsacfix_DecoderInit(ISAC_main_inst);
     }
 
 
@@ -563,19 +562,19 @@ int main(int argc, char* argv[])
           short bwe;
 
           /* Encode */
-          stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
-                                            shortdata,
-                                            (uint8_t*)streamdata);
+          stream_len_int = WebRtcIsacfix_Encode(ISAC_main_inst,
+                                                shortdata,
+                                                (uint8_t*)streamdata);
 
           /* If packet is ready, and CE testing, call the different API
              functions from the internal API. */
-          if (stream_len>0) {
+          if (stream_len_int>0) {
             if (testCE == 1) {
               err = WebRtcIsacfix_ReadBwIndex(
                   reinterpret_cast<const uint8_t*>(streamdata),
-                  stream_len,
+                  static_cast<size_t>(stream_len_int),
                   &bwe);
-              stream_len = WebRtcIsacfix_GetNewBitStream(
+              stream_len_int = WebRtcIsacfix_GetNewBitStream(
                   ISAC_main_inst,
                   bwe,
                   scale,
@@ -604,11 +603,11 @@ int main(int argc, char* argv[])
           }
         } else {
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-          stream_len = WebRtcIsacfix_EncodeNb(ISAC_main_inst,
-                                              shortdata,
-                                              streamdata);
+          stream_len_int = WebRtcIsacfix_EncodeNb(ISAC_main_inst,
+                                                  shortdata,
+                                                  streamdata);
 #else
-          stream_len = -1;
+          stream_len_int = -1;
 #endif
         }
       }
@@ -617,13 +616,14 @@ int main(int argc, char* argv[])
         break;
       }
 
-      if (stream_len < 0 || err < 0) {
+      if (stream_len_int < 0 || err < 0) {
         /* exit if returned with error */
         errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
         printf("\nError in encoder: %d.\n", errtype);
       } else {
-        if (fwrite(streamdata, sizeof(char),
-                   stream_len, outbits) != (size_t)stream_len) {
+        stream_len = static_cast<size_t>(stream_len_int);
+        if (fwrite(streamdata, sizeof(char), stream_len, outbits) !=
+            stream_len) {
           return -1;
         }
       }
@@ -729,12 +729,12 @@ int main(int argc, char* argv[])
       /* iSAC decoding */
       if( lostFrame && framecnt >  0) {
         if (nbTest !=2) {
-          declen = WebRtcIsacfix_DecodePlc(ISAC_main_inst,
-                                           decoded, prevFrameSize );
+          declen = static_cast<int>(
+              WebRtcIsacfix_DecodePlc(ISAC_main_inst, decoded, prevFrameSize));
         } else {
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-          declen = WebRtcIsacfix_DecodePlcNb(ISAC_main_inst, decoded,
-                                             prevFrameSize );
+          declen = static_cast<int>(WebRtcIsacfix_DecodePlcNb(
+              ISAC_main_inst, decoded, prevFrameSize));
 #else
           declen = -1;
 #endif
@@ -742,7 +742,7 @@ int main(int argc, char* argv[])
         lostPackets++;
       } else {
         if (nbTest !=2 ) {
-          short FL;
+          size_t FL;
           /* Call getFramelen, only used here for function test */
           err = WebRtcIsacfix_ReadFrameLen(
               reinterpret_cast<const uint8_t*>(streamdata), stream_len, &FL);
@@ -753,11 +753,11 @@ int main(int argc, char* argv[])
               decoded,
               speechType);
           /* Error check */
-          if (err<0 || declen<0 || FL!=declen) {
+          if (err < 0 || declen < 0 || FL != static_cast<size_t>(declen)) {
             errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
             printf("\nError in decode_B/or getFrameLen: %d.\n", errtype);
           }
-          prevFrameSize = declen/480;
+          prevFrameSize = static_cast<size_t>(declen/480);
 
         } else {
 #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
@@ -766,7 +766,7 @@ int main(int argc, char* argv[])
 #else
           declen = -1;
 #endif
-          prevFrameSize = declen/240;
+          prevFrameSize = static_cast<size_t>(declen / 240);
         }
       }
 
@@ -789,7 +789,7 @@ int main(int argc, char* argv[])
     framecnt++;
 
     totalsmpls += declen;
-    totalbits += 8 * stream_len;
+    totalbits += static_cast<int>(8 * stream_len);
 
     /* Error test number 10, garbage data */
     if (testNum == 10) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index 965f2bc1fe..ac0fa350c9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -21,678 +21,609 @@
 /* include API */
 #include "isac.h"
 #include "isacfix.h"
-
+#include "webrtc/base/format_macros.h"
 
 /* max number of samples per frame (= 60 ms frame) */
-#define MAX_FRAMESAMPLES				960
+#define MAX_FRAMESAMPLES 960
 /* number of samples per 10ms frame */
-#define FRAMESAMPLES_10ms				160
+#define FRAMESAMPLES_10ms 160
 /* sampling frequency (Hz) */
-#define FS								16000
-
-
+#define FS 16000
 
 /* Runtime statistics */
 #include <time.h>
-#define CLOCKS_PER_SEC  1000
-
-
-
-//	FILE *histfile, *ratefile;
+#define CLOCKS_PER_SEC 1000
 
+// FILE *histfile, *ratefile;
 
 /* function for reading audio data from PCM file */
-int readframe(int16_t *data, FILE *inp, int length) {
-	
-	short k, rlen, status = 0;
-		
-	rlen = fread(data, sizeof(int16_t), length, inp);
-	if (rlen < length) {
-		for (k = rlen; k < length; k++)
-			data[k] = 0;
-		status = 1;
-	}
-	
-	return status;
+int readframe(int16_t* data, FILE* inp, int length) {
+  short k, rlen, status = 0;
+
+  rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen < length) {
+    for (k = rlen; k < length; k++)
+      data[k] = 0;
+    status = 1;
+  }
+
+  return status;
 }
 
 typedef struct {
-	uint32_t send_time;            /* samples */
-	uint32_t arrival_time;            /* samples */
-	uint32_t sample_count;            /* samples */
-	uint16_t rtp_number;
+  uint32_t send_time;    /* samples */
+  uint32_t arrival_time; /* samples */
+  uint32_t sample_count; /* samples */
+  uint16_t rtp_number;
 } BottleNeckModel;
 
-void get_arrival_time(int current_framesamples,   /* samples */
-					  int packet_size,            /* bytes */
-					  int bottleneck,             /* excluding headers; bits/s */
-					  BottleNeckModel *BN_data)
-{
-	const int HeaderSize = 35; 
-	int HeaderRate;
+void get_arrival_time(int current_framesamples, /* samples */
+                      size_t packet_size,       /* bytes */
+                      int bottleneck,           /* excluding headers; bits/s */
+                      BottleNeckModel* BN_data) {
+  const int HeaderSize = 35;
+  int HeaderRate;
 
-	HeaderRate = HeaderSize * 8 * FS / current_framesamples;     /* bits/s */
+  HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
 
-	/* everything in samples */
-	BN_data->sample_count = BN_data->sample_count + current_framesamples;
+  /* everything in samples */
+  BN_data->sample_count = BN_data->sample_count + current_framesamples;
 
-	BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
-	BN_data->send_time += current_framesamples;
+  BN_data->arrival_time += (uint32_t)
+      (((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate));
+  BN_data->send_time += current_framesamples;
 
-	if (BN_data->arrival_time < BN_data->sample_count)
-		BN_data->arrival_time = BN_data->sample_count;
+  if (BN_data->arrival_time < BN_data->sample_count)
+    BN_data->arrival_time = BN_data->sample_count;
 
-	BN_data->rtp_number++;
+  BN_data->rtp_number++;
 }
 
+int main(int argc, char* argv[]) {
+  char inname[50], outname[50], bottleneck_file[50], bitfilename[60],
+      bitending[10] = "_bits.pcm";
+  FILE* inp, *outp, *f_bn, *bitsp;
+  int framecnt, endfile;
 
+  int i, j, errtype, plc = 0;
+  int16_t CodingMode;
+  int16_t bottleneck;
 
-int main(int argc, char* argv[])
-{
+  int framesize = 30; /* ms */
+  // int framesize = 60; /* To invoke cisco complexity case at frame 2252 */
 
-	char inname[50], outname[50], bottleneck_file[50], bitfilename[60], bitending[10]="_bits.pcm";	
-	FILE *inp, *outp, *f_bn, *bitsp;
-	int framecnt, endfile;
+  int cur_framesmpls, err;
 
+  /* Runtime statistics */
+  double starttime;
+  double runtime;
+  double length_file;
 
-	int i,j,errtype, plc=0;
-	int16_t CodingMode;
-	int16_t bottleneck;
+  size_t stream_len = 0;
+  int declen;
 
-	int16_t framesize = 30;           /* ms */
-    //int16_t framesize = 60; /* To invoke cisco complexity case at frame 2252 */
-	
-	int cur_framesmpls, err;
-	
-	/* Runtime statistics */
-	double starttime;
-	double runtime;
-	double length_file;
-	
-	int16_t stream_len = 0;
-	int16_t declen;
-	
-	int16_t shortdata[FRAMESAMPLES_10ms];
-	int16_t decoded[MAX_FRAMESAMPLES];
-	uint16_t streamdata[600];
-	int16_t	speechType[1];
-	
-//	int16_t	*iSACstruct;
+  int16_t shortdata[FRAMESAMPLES_10ms];
+  int16_t decoded[MAX_FRAMESAMPLES];
+  uint16_t streamdata[600];
+  int16_t speechType[1];
 
-	char version_number[20];
-	int mode=-1, tmp, nbTest=0; /*,sss;*/
+  // int16_t* iSACstruct;
 
-#ifdef _DEBUG
-	FILE *fy;
-	double kbps;
-	int totalbits =0;
-	int totalsmpls =0;
-#endif /* _DEBUG */
+  char version_number[20];
+  int mode = -1, tmp, nbTest = 0; /*,sss;*/
 
+#if !defined(NDEBUG)
+  FILE* fy;
+  double kbps;
+  size_t totalbits = 0;
+  int totalsmpls = 0;
+#endif
 
+  /* only one structure used for ISAC encoder */
+  ISAC_MainStruct* ISAC_main_inst;
+  ISACFIX_MainStruct* ISACFIX_main_inst;
 
+  BottleNeckModel BN_data;
+  f_bn = NULL;
 
-	/* only one structure used for ISAC encoder */
-	ISAC_MainStruct *ISAC_main_inst;
-	ISACFIX_MainStruct *ISACFIX_main_inst;
+#if !defined(NDEBUG)
+  fy = fopen("bit_rate.dat", "w");
+  fclose(fy);
+  fy = fopen("bytes_frames.dat", "w");
+  fclose(fy);
+#endif
 
-	BottleNeckModel       BN_data;
-	f_bn  = NULL;
+  // histfile = fopen("histo.dat", "ab");
+  // ratefile = fopen("rates.dat", "ab");
 
-#ifdef _DEBUG
-	fy = fopen("bit_rate.dat", "w");
-	fclose(fy);
-	fy = fopen("bytes_frames.dat", "w");
-	fclose(fy);
-#endif /* _DEBUG */
+  /* handling wrong input arguments in the command line */
+  if ((argc < 6) || (argc > 10)) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
 
+    printf("\n");
+    WebRtcIsacfix_version(version_number);
+    printf("iSAC version %s \n\n", version_number);
 
-//histfile = fopen("histo.dat", "ab");
-//ratefile = fopen("rates.dat", "ab");
+    printf("Usage:\n\n");
+    printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
+    printf("with:\n");
 
-	/* handling wrong input arguments in the command line */
-	if ((argc<6) || (argc>10))  {
-		printf("\n\nWrong number of arguments or flag values.\n\n");
+    printf("[-I]            : If -I option is specified, the coder will use\n");
+    printf("                  an instantaneous Bottleneck value. If not, it\n");
+    printf("                  will be an adaptive Bottleneck value.\n\n");
+    printf("bottleneck_value: The value of the bottleneck provided either\n");
+    printf("                  as a fixed value (e.g. 25000) or\n");
+    printf("                  read from a file (e.g. bottleneck.txt)\n\n");
+    printf("[-m] mode       : Mode (encoder - decoder):\n");
+    printf("                    0 - float - float\n");
+    printf("                    1 - float - fix\n");
+    printf("                    2 - fix - float\n");
+    printf("                    3 - fix - fix\n\n");
+    printf("[-PLC]          : Test PLC packetlosses\n\n");
+    printf("[-NB] num       : Test NB interfaces:\n");
+    printf("                    1 - encNB\n");
+    printf("                    2 - decNB\n\n");
+    printf("infile          : Normal speech input file\n\n");
+    printf("outfile         : Speech output file\n\n");
+    printf("Example usage:\n\n");
+    printf("./kenny.exe -I bottleneck.txt -m 1 speechIn.pcm speechOut.pcm\n\n");
+    exit(0);
+  }
 
-		printf("\n");
-		WebRtcIsacfix_version(version_number);
-		printf("iSAC version %s \n\n", version_number);
+  printf("--------------------START---------------------\n\n");
+  WebRtcIsac_version(version_number);
+  printf("iSAC FLOAT version %s \n", version_number);
+  WebRtcIsacfix_version(version_number);
+  printf("iSAC FIX version   %s \n\n", version_number);
 
-		printf("Usage:\n\n");
-		printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
-		printf("with:\n");
+  CodingMode = 0;
+  tmp = 1;
+  for (i = 1; i < argc; i++) {
+    if (!strcmp("-I", argv[i])) {
+      printf("\nInstantaneous BottleNeck\n");
+      CodingMode = 1;
+      i++;
+      tmp = 0;
+    }
 
-		printf("[-I]			:	if -I option is specified, the coder will use\n");
-		printf("				an instantaneous Bottleneck value. If not, it\n");
-		printf("				will be an adaptive Bottleneck value.\n\n");
-		printf("bottleneck_value	:	the value of the bottleneck provided either\n");
-		printf("				as a fixed value (e.g. 25000) or\n");
-		printf("				read from a file (e.g. bottleneck.txt)\n\n");
-		printf("[-m] mode		: Mode (encoder - decoder):\n");
-		printf("				:		0 - float - float \n");
-		printf("				:		1 - float - fix \n");
-		printf("				:		2 - fix - float \n");
-		printf("				:		3 - fix - fix \n");
-		printf("[-PLC]	 		:	Test PLC packetlosses\n");
-		printf("[-NB] num		:	Test NB interfaces, num=1 encNB, num=2 decNB\n");
-		printf("infile			:	Normal speech input file\n\n");
-		printf("outfile			:	Speech output file\n\n");
-		printf("Example usage:\n\n");
-		printf("./kenny.exe -I bottleneck.txt -m 1 speechIn.pcm speechOut.pcm\n\n");
-		exit(0);
+    if (!strcmp("-m", argv[i])) {
+      mode = atoi(argv[i + 1]);
+      i++;
+    }
 
-	} 
-	
-	
-	printf("--------------------START---------------------\n\n");
-	WebRtcIsac_version(version_number);
-	printf("iSAC FLOAT version %s \n", version_number);
-	WebRtcIsacfix_version(version_number);
-	printf("iSAC FIX version   %s \n\n", version_number);
+    if (!strcmp("-PLC", argv[i])) {
+      plc = 1;
+    }
 
-	CodingMode = 0;
-	tmp=1;
-	for (i = 1; i < argc;i++)
-	{
-		if (!strcmp ("-I", argv[i]))
-		{
-			printf("\nInstantaneous BottleNeck\n");
-			CodingMode = 1;
-			i++;
-			tmp=0;
-		} 
+    if (!strcmp("-NB", argv[i])) {
+      nbTest = atoi(argv[i + 1]);
+      i++;
+    }
+  }
 
-		if (!strcmp ("-m", argv[i])) {
-			mode=atoi(argv[i+1]);
-			i++;
-		}		
+  if (mode < 0) {
+    printf("\nError! Mode must be set: -m 0 \n");
+    exit(0);
+  }
 
-		if (!strcmp ("-PLC", argv[i]))
-		{
-			plc=1;
-		}
-		
-		if (!strcmp ("-NB", argv[i]))
-		{
-			nbTest = atoi(argv[i + 1]);
-			i++;
-		}
-			  
-	}
-	
-	if(mode<0) {
-		printf("\nError! Mode must be set: -m 0 \n");
-		exit(0);
-	}
-	
-	if (CodingMode == 0)
-	{
-		printf("\nAdaptive BottleNeck\n");
-	}
+  if (CodingMode == 0) {
+    printf("\nAdaptive BottleNeck\n");
+  }
 
+  /* Get Bottleneck value */
+  bottleneck = atoi(argv[2 - tmp]);
+  if (bottleneck == 0) {
+    sscanf(argv[2 - tmp], "%s", bottleneck_file);
+    f_bn = fopen(bottleneck_file, "rb");
+    if (f_bn == NULL) {
+      printf("No value provided for BottleNeck and cannot read file %s.\n",
+             bottleneck_file);
+      exit(0);
+    } else {
+      printf("reading bottleneck rates from file %s\n\n", bottleneck_file);
+      if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+        /* Set pointer to beginning of file */
+        fseek(f_bn, 0L, SEEK_SET);
+        fscanf(f_bn, "%d", &bottleneck);
+      }
 
+      /* Bottleneck is a cosine function
+       * Matlab code for writing the bottleneck file:
+       * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
+       * fid = fopen('bottleneck.txt', 'wb');
+       * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
+       */
+    }
+  } else {
+    printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
+  }
 
-	/* Get Bottleneck value */
-	bottleneck = atoi(argv[2-tmp]);
-	if (bottleneck == 0)
-	{
-		sscanf(argv[2-tmp], "%s", bottleneck_file);
-		f_bn = fopen(bottleneck_file, "rb");
-		if (f_bn  == NULL)
-		{
-			printf("No value provided for BottleNeck and cannot read file %s.\n", bottleneck_file);
-			exit(0);
-		}
-		else {
-			printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
-			if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
-					/* Set pointer to beginning of file */
-					fseek(f_bn, 0L, SEEK_SET);
-					fscanf(f_bn, "%d", &bottleneck);
-			}		
+  /* Get Input and Output files */
+  sscanf(argv[argc - 2], "%s", inname);
+  sscanf(argv[argc - 1], "%s", outname);
 
-			/*	Bottleneck is a cosine function 
-			*	Matlab code for writing the bottleneck file:
-			*	BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
-			*	fid = fopen('bottleneck.txt', 'wb');
-			*	fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
-			*/
-		}
-	}
-	else 
-	{
-		printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
-	}
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  iSAC: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  iSAC: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:%s\nOutput:%s\n", inname, outname);
 
+  i = 0;
+  while (outname[i] != '\0') {
+    bitfilename[i] = outname[i];
+    i++;
+  }
+  i -= 4;
+  for (j = 0; j < 9; j++, i++)
+    bitfilename[i] = bitending[j];
+  bitfilename[i] = '\0';
+  if ((bitsp = fopen(bitfilename, "wb")) == NULL) {
+    printf("  iSAC: Cannot read file %s.\n", bitfilename);
+    exit(1);
+  }
+  printf("Bitstream:%s\n\n", bitfilename);
 
+  starttime = clock() / (double)CLOCKS_PER_SEC; /* Runtime statistics */
 
-	/* Get Input and Output files */
-	sscanf(argv[argc-2], "%s", inname);
-	sscanf(argv[argc-1], "%s", outname);
-	
-	if ((inp = fopen(inname,"rb")) == NULL) {
-		printf("  iSAC: Cannot read file %s.\n", inname);
-		exit(1);
-	}
-	if ((outp = fopen(outname,"wb")) == NULL) {
-		printf("  iSAC: Cannot write file %s.\n", outname);
-		exit(1);
-	}
-	printf("\nInput:%s\nOutput:%s\n", inname, outname);
+  /* Initialize the ISAC and BN structs */
+  WebRtcIsac_create(&ISAC_main_inst);
+  WebRtcIsacfix_Create(&ISACFIX_main_inst);
 
-	i=0;
-	while (outname[i]!='\0') {
-		bitfilename[i]=outname[i];
-		i++;
-	}
-	i-=4;
-	for (j=0;j<9;j++, i++)
-		bitfilename[i]=bitending[j];
-	bitfilename[i]='\0';
-	if ((bitsp = fopen(bitfilename,"wb")) == NULL) {
-		printf("  iSAC: Cannot read file %s.\n", bitfilename);
-		exit(1);
-	}
-	printf("Bitstream:%s\n\n", bitfilename);
+  BN_data.send_time = 0;
+  BN_data.arrival_time = 0;
+  BN_data.sample_count = 0;
+  BN_data.rtp_number = 0;
 
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = 0;
 
-	
-	starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
+  if (mode == 0) { /* Encode using FLOAT, decode using FLOAT */
 
+    printf("Coding mode: Encode using FLOAT, decode using FLOAT \n\n");
 
-	/* Initialize the ISAC and BN structs */
-	WebRtcIsac_create(&ISAC_main_inst);
-/*	WebRtcIsacfix_AssignSize(&sss);
-	iSACstruct=malloc(sss);
-	WebRtcIsacfix_Assign(&ISACFIX_main_inst,iSACstruct);*/
-	WebRtcIsacfix_Create(&ISACFIX_main_inst);
-	
-	BN_data.send_time	  = 0;
-	BN_data.arrival_time  = 0;
-	BN_data.sample_count  = 0;
-	BN_data.rtp_number    = 0;
-	
-	/* Initialize encoder and decoder */
-    framecnt= 0;
-    endfile	= 0;
+    /* Init iSAC FLOAT */
+    WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+    WebRtcIsac_DecoderInit(ISAC_main_inst);
+    if (CodingMode == 1) {
+      err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        printf("\n\n Error in initialization: %d.\n\n", errtype);
+        // exit(EXIT_FAILURE);
+      }
+    }
 
-	if (mode==0) { /* Encode using FLOAT, decode using FLOAT */
+  } else if (mode == 1) { /* Encode using FLOAT, decode using FIX */
 
-		printf("Coding mode: Encode using FLOAT, decode using FLOAT \n\n");
+    printf("Coding mode: Encode using FLOAT, decode using FIX \n\n");
 
-		/* Init iSAC FLOAT */
-		WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
-		WebRtcIsac_DecoderInit(ISAC_main_inst);
-		if (CodingMode == 1) {
-			err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\n Error in initialization: %d.\n\n", errtype);
-			//	exit(EXIT_FAILURE);
-			}
-		}
-	
-	} else if (mode==1) { /* Encode using FLOAT, decode using FIX */
+    /* Init iSAC FLOAT */
+    WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+    WebRtcIsac_DecoderInit(ISAC_main_inst);
+    if (CodingMode == 1) {
+      err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        printf("\n\n Error in initialization: %d.\n\n", errtype);
+        // exit(EXIT_FAILURE);
+      }
+    }
 
-		printf("Coding mode: Encode using FLOAT, decode using FIX \n\n");
+    /* Init iSAC FIX */
+    WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
+    WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
+    if (CodingMode == 1) {
+      err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+        printf("\n\n Error in initialization: %d.\n\n", errtype);
+        // exit(EXIT_FAILURE);
+      }
+    }
+  } else if (mode == 2) { /* Encode using FIX, decode using FLOAT */
 
-		/* Init iSAC FLOAT */
-		WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
-		WebRtcIsac_DecoderInit(ISAC_main_inst);
-		if (CodingMode == 1) {
-			err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\n Error in initialization: %d.\n\n", errtype);
-			//	exit(EXIT_FAILURE);
-			}
-		}
+    printf("Coding mode: Encode using FIX, decode using FLOAT \n\n");
 
-		/* Init iSAC FIX */
-		WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
-		WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
-		if (CodingMode == 1) {
-			err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\n Error in initialization: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		}
-	} else if (mode==2) { /* Encode using FIX, decode using FLOAT */
+    /* Init iSAC FLOAT */
+    WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+    WebRtcIsac_DecoderInit(ISAC_main_inst);
+    if (CodingMode == 1) {
+      err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        printf("\n\n Error in initialization: %d.\n\n", errtype);
+        // exit(EXIT_FAILURE);
+      }
+    }
 
-		printf("Coding mode: Encode using FIX, decode using FLOAT \n\n");
+    /* Init iSAC FIX */
+    WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
+    WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
+    if (CodingMode == 1) {
+      err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+        printf("\n\n Error in initialization: %d.\n\n", errtype);
+        // exit(EXIT_FAILURE);
+      }
+    }
+  } else if (mode == 3) {
+    printf("Coding mode: Encode using FIX, decode using FIX \n\n");
 
-		/* Init iSAC FLOAT */
-		WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
-		WebRtcIsac_DecoderInit(ISAC_main_inst);
-		if (CodingMode == 1) {
-			err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\n Error in initialization: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		}
-		
-		/* Init iSAC FIX */
-		WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
-		WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
-		if (CodingMode == 1) {
-			err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\n Error in initialization: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		}
-	} else if (mode==3) {
+    WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
+    WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
+    if (CodingMode == 1) {
+      err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+        printf("\n\n Error in initialization: %d.\n\n", errtype);
+        // exit(EXIT_FAILURE);
+      }
+    }
 
-		printf("Coding mode: Encode using FIX, decode using FIX \n\n");
-
-		WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
-		WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
-		if (CodingMode == 1) {
-			err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\n Error in initialization: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		}
-
-	} else
-		printf("Mode must be value between 0 and 3\n");
-	*speechType = 1;
+  } else
+    printf("Mode must be value between 0 and 3\n");
+  *speechType = 1;
 
 //#define BI_TEST 1
 #ifdef BI_TEST
-    err = WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_main_inst, 300);
-    if (err < 0) {
-            /* exit if returned with error */
-            errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-            printf("\n\n Error in setMaxPayloadSize: %d.\n\n", errtype);
-            fclose(inp);
-            fclose(outp);
-            fclose(bitsp);
-            return(EXIT_FAILURE);
-    }
+  err = WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_main_inst, 300);
+  if (err < 0) {
+    /* exit if returned with error */
+    errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+    printf("\n\n Error in setMaxPayloadSize: %d.\n\n", errtype);
+    fclose(inp);
+    fclose(outp);
+    fclose(bitsp);
+    return (EXIT_FAILURE);
+  }
 #endif
 
+  while (endfile == 0) {
+    cur_framesmpls = 0;
+    while (1) {
+      int stream_len_int;
 
-    while (endfile == 0) {	
+      /* Read 10 ms speech block */
+      if (nbTest != 1)
+        endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
+      else
+        endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms / 2));
 
-		cur_framesmpls = 0;
-		while (1) {
-			/* Read 10 ms speech block */
-			if (nbTest != 1)
-				endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
-			else
-				endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms/2));
+      /* iSAC encoding */
 
-			/* iSAC encoding */
+      if (mode == 0 || mode == 1) {
+        stream_len_int =
+            WebRtcIsac_Encode(ISAC_main_inst, shortdata, (uint8_t*)streamdata);
+        if (stream_len_int < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          printf("\n\nError in encoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+      } else if (mode == 2 || mode == 3) {
+        /* iSAC encoding */
+        if (nbTest != 1) {
+          stream_len_int = WebRtcIsacfix_Encode(ISACFIX_main_inst, shortdata,
+                                                (uint8_t*)streamdata);
+        } else {
+          stream_len_int =
+              WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);
+        }
 
-			if (mode==0 || mode ==1) {
-                          stream_len = WebRtcIsac_Encode(ISAC_main_inst,
-                                                         shortdata,
-                                                         (uint8_t*)streamdata);
-				if (stream_len < 0) {
-					/* exit if returned with error */
-					errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-					printf("\n\nError in encoder: %d.\n\n", errtype);
-				//	exit(EXIT_FAILURE);
-				}
-			} else if (mode==2 || mode==3) {
-				/* iSAC encoding */
-				if (nbTest != 1)
-                                  stream_len = WebRtcIsacfix_Encode(
-                                      ISACFIX_main_inst,
-                                      shortdata,
-                                      (uint8_t*)streamdata);
-				else
-					stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);
-		
-				if (stream_len < 0) {
-					/* exit if returned with error */
-					errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-					printf("\n\nError in encoder: %d.\n\n", errtype);
-				//	exit(EXIT_FAILURE);
-				}
-			}			
+        if (stream_len_int < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+          printf("\n\nError in encoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+      }
+      stream_len = (size_t)stream_len_int;
 
-			cur_framesmpls += FRAMESAMPLES_10ms;
+      cur_framesmpls += FRAMESAMPLES_10ms;
 
-			/* read next bottleneck rate */
-			if (f_bn != NULL) {
-				if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
-					/* Set pointer to beginning of file */
-					fseek(f_bn, 0L, SEEK_SET);
-					fscanf(f_bn, "%d", &bottleneck);
-				}
-				if (CodingMode == 1) {
-					if (mode==0 || mode==1)
-					  WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
-					else if	(mode==2 || mode==3)
-						WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
-				}
-			}
+      /* read next bottleneck rate */
+      if (f_bn != NULL) {
+        if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+          /* Set pointer to beginning of file */
+          fseek(f_bn, 0L, SEEK_SET);
+          fscanf(f_bn, "%d", &bottleneck);
+        }
+        if (CodingMode == 1) {
+          if (mode == 0 || mode == 1)
+            WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+          else if (mode == 2 || mode == 3)
+            WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
+        }
+      }
 
-			/* exit encoder loop if the encoder returned a bitstream */
-			if (stream_len != 0) break;
-		}
-		
-		fwrite(streamdata, 1, stream_len, bitsp); /* NOTE! Writes bytes to file */
+      /* exit encoder loop if the encoder returned a bitstream */
+      if (stream_len != 0)
+        break;
+    }
 
-		/* simulate packet handling through NetEq and the modem */
-		get_arrival_time(cur_framesmpls, stream_len, bottleneck,
-						 &BN_data);
-//*****************************
-		if (1){
-		if (mode==0) {
-			err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
-									  streamdata,
-									  stream_len,
-									  BN_data.rtp_number,
-									  BN_data.arrival_time);
+    fwrite(streamdata, 1, stream_len, bitsp); /* NOTE! Writes bytes to file */
 
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-			/* iSAC decoding */
-			declen = WebRtcIsac_Decode(ISAC_main_inst,
-										  streamdata,
-									  	  stream_len,
-										  decoded,
-										  speechType);
-			if (declen <= 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		} else if (mode==1) {
+    /* simulate packet handling through NetEq and the modem */
+    get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data);
+    //*****************************
+    if (1) {
+      if (mode == 0) {
+        err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst, streamdata,
+                                          stream_len, BN_data.rtp_number,
+                                          BN_data.arrival_time);
 
-			err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
-									  streamdata,
-									  stream_len,
-									  BN_data.rtp_number,
-									  BN_data.arrival_time);
-			err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst,
-									  streamdata,
-									  stream_len,
-									  BN_data.rtp_number,
-									  BN_data.arrival_time);
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
+        if (err < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+        /* iSAC decoding */
+        declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata, stream_len,
+                                   decoded, speechType);
+        if (declen <= 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+      } else if (mode == 1) {
+        err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst, streamdata,
+                                          stream_len, BN_data.rtp_number,
+                                          BN_data.arrival_time);
+        err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst, streamdata,
+                                              stream_len, BN_data.rtp_number,
+                                              BN_data.arrival_time);
+        if (err < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
 
-			declen = WebRtcIsac_Decode(ISAC_main_inst,
-										  streamdata,
-									  	  stream_len,
-										  decoded,
-										  speechType);
+        declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata, stream_len,
+                                   decoded, speechType);
 
-			/* iSAC decoding */
-			if (plc && (framecnt+1)%10 == 0) {
-				if (nbTest !=2 )
-					declen = WebRtcIsacfix_DecodePlc( ISACFIX_main_inst, decoded, 1 );
-				else
-					declen = WebRtcIsacfix_DecodePlcNb( ISACFIX_main_inst, decoded, 1 );
-			} else {
-				if (nbTest !=2 )
-					declen = WebRtcIsacfix_Decode(ISACFIX_main_inst,
-												  streamdata,
-											  	  stream_len,
-												  decoded,
-												  speechType);
-				else
-					declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst,
-												  streamdata,
-											  	  stream_len,
-												  decoded,
-												  speechType);
-			}
-			
-			if (declen <= 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		} else if (mode==2) {
-			err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst,
-									  streamdata,
-									  stream_len,
-									  BN_data.rtp_number,
-									  BN_data.arrival_time);
+        /* iSAC decoding */
+        if (plc && (framecnt + 1) % 10 == 0) {
+          if (nbTest != 2) {
+            declen =
+                (int)WebRtcIsacfix_DecodePlc(ISACFIX_main_inst, decoded, 1);
+          } else {
+            declen =
+                (int)WebRtcIsacfix_DecodePlcNb(ISACFIX_main_inst, decoded, 1);
+          }
+        } else {
+          if (nbTest != 2)
+            declen = WebRtcIsacfix_Decode(ISACFIX_main_inst, streamdata,
+                                          stream_len, decoded, speechType);
+          else
+            declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst, streamdata,
+                                            stream_len, decoded, speechType);
+        }
 
-			err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
-									  streamdata,
-									  stream_len,
-									  BN_data.rtp_number,
-									  BN_data.arrival_time);
+        if (declen <= 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+      } else if (mode == 2) {
+        err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst, streamdata,
+                                              stream_len, BN_data.rtp_number,
+                                              BN_data.arrival_time);
 
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-			/* iSAC decoding */
-			declen = WebRtcIsac_Decode(ISAC_main_inst,
-										  streamdata,
-									  	  stream_len,
-										  decoded,
-										  speechType);
-			if (declen <= 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		}  else if (mode==3) {
-			err = WebRtcIsacfix_UpdateBwEstimate(ISACFIX_main_inst,
-									  streamdata,
-									  stream_len,
-									  BN_data.rtp_number,
-									  BN_data.send_time,
-									  BN_data.arrival_time);
+        err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst, streamdata,
+                                          stream_len, BN_data.rtp_number,
+                                          BN_data.arrival_time);
 
-			if (err < 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-			/* iSAC decoding */
-						
-			if (plc && (framecnt+1)%10 == 0) {
-				if (nbTest !=2 )
-					declen = WebRtcIsacfix_DecodePlc( ISACFIX_main_inst, decoded, 1 );
-				else
-					declen = WebRtcIsacfix_DecodePlcNb( ISACFIX_main_inst, decoded, 1 );
-			} else {
-				if (nbTest !=2 )
-					declen = WebRtcIsacfix_Decode(ISACFIX_main_inst,
-												  streamdata,
-											  	  stream_len,
-												  decoded,
-												  speechType);
-				else
-					declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst,
-												  streamdata,
-											  	  stream_len,
-												  decoded,
-												  speechType);
-			}
-			if (declen <= 0) {
-				/* exit if returned with error */
-				errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
-				printf("\n\nError in decoder: %d.\n\n", errtype);
-				//exit(EXIT_FAILURE);
-			}
-		}
+        if (err < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+        /* iSAC decoding */
+        declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata, stream_len,
+                                   decoded, speechType);
+        if (declen <= 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+      } else if (mode == 3) {
+        err = WebRtcIsacfix_UpdateBwEstimate(
+            ISACFIX_main_inst, streamdata, stream_len, BN_data.rtp_number,
+            BN_data.send_time, BN_data.arrival_time);
 
-		/* Write decoded speech frame to file */
-		fwrite(decoded, sizeof(int16_t), declen, outp);
-		}
+        if (err < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+        /* iSAC decoding */
 
-		fprintf(stderr,"  \rframe = %d", framecnt);
-		framecnt++;
+        if (plc && (framecnt + 1) % 10 == 0) {
+          if (nbTest != 2) {
+            declen =
+                (int)WebRtcIsacfix_DecodePlc(ISACFIX_main_inst, decoded, 1);
+          } else {
+            declen =
+                (int)WebRtcIsacfix_DecodePlcNb(ISACFIX_main_inst, decoded, 1);
+          }
+        } else {
+          if (nbTest != 2) {
+            declen = WebRtcIsacfix_Decode(ISACFIX_main_inst, streamdata,
+                                          stream_len, decoded, speechType);
+          } else {
+            declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst, streamdata,
+                                            stream_len, decoded, speechType);
+          }
+        }
+        if (declen <= 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
+          printf("\n\nError in decoder: %d.\n\n", errtype);
+          // exit(EXIT_FAILURE);
+        }
+      }
 
+      /* Write decoded speech frame to file */
+      fwrite(decoded, sizeof(int16_t), declen, outp);
+    }
 
+    fprintf(stderr, "  \rframe = %d", framecnt);
+    framecnt++;
 
-#ifdef _DEBUG
-		
-		totalsmpls += declen;
-		totalbits += 8 * stream_len;
-		kbps = ((double) FS) / ((double) cur_framesmpls) * 8.0 * stream_len / 1000.0;// kbits/s
-		fy = fopen("bit_rate.dat", "a");
-		fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
-		fclose(fy);
-		
-#endif /* _DEBUG */
-	
-	}
-	
-#ifdef _DEBUG
-	printf("\n\ntotal bits				= %d bits", totalbits);
-	printf("\nmeasured average bitrate		= %0.3f kbits/s", (double)totalbits *(FS/1000) / totalsmpls);
-	printf("\n");
-#endif /* _DEBUG */
-	
-	/* Runtime statistics */
-	runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
-	length_file = ((double)framecnt*(double)declen/FS);
-	printf("\n\nLength of speech file: %.1f s\n", length_file);
-	printf("Time to run iSAC:      %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
-	printf("---------------------END----------------------\n");
-	
-	fclose(inp);
-	fclose(outp);
-	
-	WebRtcIsac_Free(ISAC_main_inst);
-	WebRtcIsacfix_Free(ISACFIX_main_inst);
+#if !defined(NDEBUG)
 
-	
+    totalsmpls += declen;
+    totalbits += 8 * stream_len;
+    kbps = (double)FS / (double)cur_framesmpls * 8.0 * stream_len / 1000.0;
+    fy = fopen("bit_rate.dat", "a");
+    fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
+    fclose(fy);
 
-//	fclose(histfile);
-//	fclose(ratefile);
-	
-	return 0;
+#endif
+  }
 
-}	
+#if !defined(NDEBUG)
+  printf("\n\ntotal bits               = %" PRIuS " bits", totalbits);
+  printf("\nmeasured average bitrate = %0.3f kbits/s",
+         (double)totalbits * (FS / 1000) / totalsmpls);
+  printf("\n");
+#endif
 
+  /* Runtime statistics */
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC - starttime);
+  length_file = ((double)framecnt * (double)declen / FS);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run iSAC:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
 
+  fclose(inp);
+  fclose(outp);
+
+  WebRtcIsac_Free(ISAC_main_inst);
+  WebRtcIsacfix_Free(ISACFIX_main_inst);
+
+  // fclose(histfile);
+  // fclose(ratefile);
+
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac.gypi
index 43da73a118..55d7c31995 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac.gypi
@@ -9,31 +9,32 @@
 {
   'targets': [
     {
-      'target_name': 'iSAC',
+      'target_name': 'isac',
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         'audio_decoder_interface',
         'audio_encoder_interface',
+        'isac_common',
       ],
       'include_dirs': [
-        'main/interface',
+        'main/include',
         '<(webrtc_root)',
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          'main/interface',
+          'main/include',
           '<(webrtc_root)',
         ],
       },
       'sources': [
-        'audio_encoder_isac_t.h',
-        'audio_encoder_isac_t_impl.h',
-        'main/interface/audio_encoder_isac.h',
-        'main/interface/isac.h',
+        'main/include/audio_decoder_isac.h',
+        'main/include/audio_encoder_isac.h',
+        'main/include/isac.h',
         'main/source/arith_routines.c',
         'main/source/arith_routines_hist.c',
         'main/source/arith_routines_logist.c',
+        'main/source/audio_decoder_isac.cc',
         'main/source/audio_encoder_isac.cc',
         'main/source/bandwidth_estimator.c',
         'main/source/crc.c',
@@ -47,6 +48,7 @@
         'main/source/filterbank_tables.c',
         'main/source/intialize.c',
         'main/source/isac.c',
+        'main/source/isac_float_type.h',
         'main/source/filterbanks.c',
         'main/source/pitch_lag_tables.c',
         'main/source/lattice.c',
diff --git a/media/webrtc/trunk/webrtc/video_engine/video_engine.gyp b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_common.gypi
similarity index 51%
rename from media/webrtc/trunk/webrtc/video_engine/video_engine.gyp
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_common.gypi
index 6e72dd551b..135ecd27cc 100644
--- a/media/webrtc/trunk/webrtc/video_engine/video_engine.gyp
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_common.gypi
@@ -1,4 +1,4 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
@@ -7,18 +7,16 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
-  'includes': [
-    '../build/common.gypi',
-    './video_engine_core.gypi',
-  ],
-
-  'conditions': [
-    ['include_tests==1', {
-      'includes': [
-        'test/libvietest/libvietest.gypi',
-        'test/auto_test/vie_auto_test.gypi',
+  'targets': [
+    {
+      'target_name': 'isac_common',
+      'type': 'static_library',
+      'sources': [
+        'audio_encoder_isac_t.h',
+        'audio_encoder_isac_t_impl.h',
+        'locked_bandwidth_info.cc',
+        'locked_bandwidth_info.h',
       ],
-    }],
+    },
   ],
 }
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
index a43450e047..54cedb4e18 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isac_test.gypi
@@ -10,14 +10,14 @@
   'targets': [
     # simple kenny
     {
-      'target_name': 'iSACtest',
+      'target_name': 'isac_test',
       'type': 'executable',
       'dependencies': [
-        'iSAC',
+        'isac',
       ],
       'include_dirs': [
         './main/test',
-        './main/interface',
+        './main/include',
         './main/util',
         '<(webrtc_root)',
       ],
@@ -25,17 +25,30 @@
         './main/test/simpleKenny.c',
         './main/util/utility.c',
       ],
+      'conditions': [
+        ['OS=="win" and clang==1', {
+          'msvs_settings': {
+            'VCCLCompilerTool': {
+              'AdditionalOptions': [
+                # Disable warnings failing when compiling with Clang on Windows.
+                # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                '-Wno-format',
+              ],
+            },
+          },
+        }],
+      ],  # conditions.
     },
     # ReleaseTest-API
     {
-      'target_name': 'iSACAPITest',
+      'target_name': 'isac_api_test',
       'type': 'executable',
       'dependencies': [
-        'iSAC',
+        'isac',
       ],
       'include_dirs': [
         './main/test',
-        './main/interface',
+        './main/include',
         './main/util',
         '<(webrtc_root)',
       ],
@@ -46,14 +59,14 @@
     },
     # SwitchingSampRate
     {
-      'target_name': 'iSACSwitchSampRateTest',
+      'target_name': 'isac_switch_samprate_test',
       'type': 'executable',
       'dependencies': [
-        'iSAC',
+        'isac',
       ],
       'include_dirs': [
         './main/test',
-        './main/interface',
+        './main/include',
         '../../../../common_audio/signal_processing/include',
         './main/util',
         '<(webrtc_root)',
@@ -61,8 +74,7 @@
       'sources': [
         './main/test/SwitchingSampRate/SwitchingSampRate.cc',
         './main/util/utility.c',
-      ],    
+      ],
     },
-
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
index a6a7f97775..25c285c1af 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix.gypi
@@ -7,32 +7,36 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
+  'includes': [
+    '../../../../build/common.gypi',
+  ],
   'targets': [
     {
-      'target_name': 'iSACFix',
+      'target_name': 'isac_fix',
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+        'isac_common',
       ],
       'include_dirs': [
-        'fix/interface',
+        'fix/include',
         '<(webrtc_root)'
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          'fix/interface',
+          'fix/include',
           '<(webrtc_root)',
         ],
       },
       'sources': [
-        'audio_encoder_isac_t.h',
-        'audio_encoder_isac_t_impl.h',
-        'fix/interface/audio_encoder_isacfix.h',
-        'fix/interface/isacfix.h',
+        'fix/include/audio_decoder_isacfix.h',
+        'fix/include/audio_encoder_isacfix.h',
+        'fix/include/isacfix.h',
         'fix/source/arith_routines.c',
         'fix/source/arith_routines_hist.c',
         'fix/source/arith_routines_logist.c',
+        'fix/source/audio_decoder_isacfix.cc',
         'fix/source/audio_encoder_isacfix.cc',
         'fix/source/bandwidth_estimator.c',
         'fix/source/decode.c',
@@ -45,6 +49,7 @@
         'fix/source/filterbanks.c',
         'fix/source/filters.c',
         'fix/source/initialize.c',
+        'fix/source/isac_fix_type.h',
         'fix/source/isacfix.c',
         'fix/source/lattice.c',
         'fix/source/lattice_c.c',
@@ -75,11 +80,6 @@
         'fix/source/structs.h',
       ],
       'conditions': [
-        ['OS!="win"', {
-          'defines': [
-            'WEBRTC_LINUX',
-          ],
-        }],
         ['target_arch=="arm" and arm_version>=7', {
           'sources': [
             'fix/source/lattice_armv7.S',
@@ -89,13 +89,11 @@
             'fix/source/lattice_c.c',
             'fix/source/pitch_filter_c.c',
           ],
-          'conditions': [
-            ['arm_neon==1 or arm_neon_optional==1', {
-              'dependencies': [ 'isac_neon' ],
-            }],
-          ],
         }],
-        ['target_arch=="mipsel" and mips_arch_variant!="r6" and android_webview_build==0', {
+        ['build_with_neon==1', {
+          'dependencies': ['isac_neon', ],
+        }],
+        ['target_arch=="mipsel" and mips_arch_variant!="r6"', {
           'sources': [
             'fix/source/entropy_coding_mips.c',
             'fix/source/filters_mips.c',
@@ -128,7 +126,7 @@
     },
   ],
   'conditions': [
-    ['target_arch=="arm" and arm_version>=7', {
+    ['build_with_neon==1', {
       'targets': [
         {
           'target_name': 'isac_neon',
@@ -137,25 +135,12 @@
           'dependencies': [
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
           ],
-          'include_dirs': [
-            '<(webrtc_root)',
-          ],
           'sources': [
             'fix/source/entropy_coding_neon.c',
-            'fix/source/filterbanks_neon.S',
-            'fix/source/filters_neon.S',
-            'fix/source/lattice_neon.S',
-            'fix/source/lpc_masking_model_neon.S',
-            'fix/source/transform_neon.S',
-          ],
-          'conditions': [
-            # Disable LTO in isac_neon target due to compiler bug
-            ['use_lto==1', {
-              'cflags!': [
-                '-flto',
-                '-ffat-lto-objects',
-              ],
-            }],
+            'fix/source/filterbanks_neon.c',
+            'fix/source/filters_neon.c',
+            'fix/source/lattice_neon.c',
+            'fix/source/transform_neon.c',
           ],
         },
       ],
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi
index 419d302fd9..7d9bc99519 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi
@@ -10,15 +10,15 @@
   'targets': [
     # kenny
     {
-      'target_name': 'iSACFixtest',
+      'target_name': 'isac_fix_test',
       'type': 'executable',
       'dependencies': [
-        'iSACFix',
+        'isac_fix',
         '<(webrtc_root)/test/test.gyp:test_support',
       ],
       'include_dirs': [
         './fix/test',
-        './fix/interface',
+        './fix/include',
         '<(webrtc_root)',
       ],
       'sources': [
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_dummy_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
similarity index 57%
rename from media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_dummy_impl.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
index 491ccbe79c..78b415c4c9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_dummy_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
@@ -6,14 +6,17 @@
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
- *
  */
 
-// This file contains an implementation of empty webrtc VP9 encoder/decoder
-// factories so it is possible to build webrtc without linking with vp9.
-#include "webrtc/modules/video_coding/codecs/vp9/vp9_impl.h"
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 
 namespace webrtc {
-VP9Encoder* VP9Encoder::Create() { return nullptr; }
-VP9Decoder* VP9Decoder::Create() { return nullptr; }
+
+LockedIsacBandwidthInfo::LockedIsacBandwidthInfo()
+      : lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+  bwinfo_.in_use = 0;
 }
+
+LockedIsacBandwidthInfo::~LockedIsacBandwidthInfo() = default;
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
new file mode 100644
index 0000000000..bbb040de1d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+// An IsacBandwidthInfo that's safe to access from multiple threads because
+// it's protected by a mutex.
+class LockedIsacBandwidthInfo final {
+ public:
+  LockedIsacBandwidthInfo();
+  ~LockedIsacBandwidthInfo();
+
+  IsacBandwidthInfo Get() const {
+    CriticalSectionScoped cs(lock_.get());
+    return bwinfo_;
+  }
+
+  void Set(const IsacBandwidthInfo& bwinfo) {
+    CriticalSectionScoped cs(lock_.get());
+    bwinfo_ = bwinfo;
+  }
+
+ private:
+  const rtc::scoped_ptr<CriticalSectionWrapper> lock_;
+  IsacBandwidthInfo bwinfo_ GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h
new file mode 100644
index 0000000000..dcd4852a68
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
+
+namespace webrtc {
+
+using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h
new file mode 100644
index 0000000000..cc8665d6b7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
+
+namespace webrtc {
+
+using AudioEncoderIsac = AudioEncoderIsacT<IsacFloat>;
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
similarity index 94%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
index 6d0c32deb1..327e7f4b1f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
 
-/*
- * Define the fixed-point numeric formats
- */
+#include <stddef.h>
+
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 #include "webrtc/typedefs.h"
 
 typedef struct WebRtcISACStruct    ISACStruct;
@@ -144,7 +144,7 @@ extern "C" {
    *                            : -1 - Error
    */
 
-  int16_t WebRtcIsac_Encode(
+  int WebRtcIsac_Encode(
       ISACStruct*        ISAC_main_inst,
       const int16_t* speechIn,
       uint8_t* encoded);
@@ -157,15 +157,9 @@ extern "C" {
    *
    * Input:
    *        - ISAC_main_inst    : ISAC instance.
-   *
-   * Return value
-   *                            : 0 - Ok
-   *                             -1 - Error
    */
 
-  int16_t WebRtcIsac_DecoderInit(
-      ISACStruct* ISAC_main_inst);
-
+  void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst);
 
   /******************************************************************************
    * WebRtcIsac_UpdateBwEstimate(...)
@@ -188,7 +182,7 @@ extern "C" {
   int16_t WebRtcIsac_UpdateBwEstimate(
       ISACStruct*         ISAC_main_inst,
       const uint8_t* encoded,
-      int32_t         packet_size,
+      size_t         packet_size,
       uint16_t        rtp_seq_number,
       uint32_t        send_ts,
       uint32_t        arr_ts);
@@ -214,10 +208,10 @@ extern "C" {
    *                              -1 - Error.
    */
 
-  int16_t WebRtcIsac_Decode(
+  int WebRtcIsac_Decode(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
-      int16_t         len,
+      size_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
 
@@ -237,14 +231,13 @@ extern "C" {
    * Output:
    *        - decoded           : The decoded vector.
    *
-   * Return value               : >0 - number of samples in decoded PLC vector
-   *                              -1 - Error
+   * Return value               : Number of samples in decoded PLC vector
    */
 
-  int16_t WebRtcIsac_DecodePlc(
+  size_t WebRtcIsac_DecodePlc(
       ISACStruct*  ISAC_main_inst,
       int16_t* decoded,
-      int16_t  noOfLostFrames);
+      size_t  noOfLostFrames);
 
 
   /******************************************************************************
@@ -269,8 +262,10 @@ extern "C" {
   int16_t WebRtcIsac_Control(
       ISACStruct*   ISAC_main_inst,
       int32_t rate,
-      int16_t framesize);
+      int framesize);
 
+  void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
+                                          int bottleneck_bits_per_second);
 
   /******************************************************************************
    * WebRtcIsac_ControlBwe(...)
@@ -300,7 +295,7 @@ extern "C" {
   int16_t WebRtcIsac_ControlBwe(
       ISACStruct* ISAC_main_inst,
       int32_t rateBPS,
-      int16_t frameSizeMs,
+      int frameSizeMs,
       int16_t enforceFrameSize);
 
 
@@ -701,13 +696,24 @@ extern "C" {
    * Return value              : >0 - number of samples in decoded vector
    *                             -1 - Error
    */
-  int16_t WebRtcIsac_DecodeRcu(
+  int WebRtcIsac_DecodeRcu(
       ISACStruct*           ISAC_main_inst,
       const uint8_t* encoded,
-      int16_t         len,
+      size_t         len,
       int16_t*        decoded,
       int16_t*        speechType);
 
+  /* Fills in an IsacBandwidthInfo struct. |inst| should be a decoder. */
+  void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst, IsacBandwidthInfo* bwinfo);
+
+  /* Uses the values from an IsacBandwidthInfo struct. |inst| should be an
+     encoder. */
+  void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
+                                   const IsacBandwidthInfo* bwinfo);
+
+  /* If |inst| is a decoder but not an encoder: tell it what sample rate the
+     encoder is using, for bandwidth estimation purposes. */
+  void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
 
 #if defined(__cplusplus)
 }
@@ -715,4 +721,4 @@ extern "C" {
 
 
 
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/fake_stdin.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
similarity index 55%
rename from media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/fake_stdin.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
index c298fe2e09..8e0603e99e 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/fake_stdin.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -7,20 +7,14 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef FAKE_STDIN_H_
-#define FAKE_STDIN_H_
 
-#include <stdio.h>
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
 
-#include <string>
-
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
 
 namespace webrtc {
 
-// Creates a fake stdin-like FILE* for unit test usage.
-FILE* FakeStdin(const std::string& input);
+// Explicit instantiation:
+template class AudioDecoderIsacT<IsacFloat>;
 
 }  // namespace webrtc
-
-#endif  // FAKE_STDIN_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
index ba08603637..64b9815b80 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
@@ -8,14 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 
 #include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
 
 namespace webrtc {
 
-// Explicit instantiation of AudioEncoderDecoderIsacT<IsacFloat>, a.k.a.
-// AudioEncoderDecoderIsac.
-template class AudioEncoderDecoderIsacT<IsacFloat>;
+// Explicit instantiation:
+template class AudioEncoderIsacT<IsacFloat>;
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
new file mode 100644
index 0000000000..62080954f7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+
+namespace webrtc {
+
+namespace {
+
+void TestBadConfig(const AudioEncoderIsac::Config& config) {
+  EXPECT_FALSE(config.IsOk());
+}
+
+void TestGoodConfig(const AudioEncoderIsac::Config& config) {
+  EXPECT_TRUE(config.IsOk());
+  AudioEncoderIsac aei(config);
+}
+
+// Wrap subroutine calls that test things in this, so that the error messages
+// will be accompanied by stack traces that make it possible to tell which
+// subroutine invocation caused the failure.
+#define S(x) do { SCOPED_TRACE(#x); x; } while (0)
+
+}  // namespace
+
+TEST(AudioEncoderIsacTest, TestConfigBitrate) {
+  AudioEncoderIsac::Config config;
+
+  // The default value is some real, positive value.
+  EXPECT_GT(config.bit_rate, 1);
+  S(TestGoodConfig(config));
+
+  // 0 is another way to ask for the default value.
+  config.bit_rate = 0;
+  S(TestGoodConfig(config));
+
+  // Try some unreasonable values and watch them fail.
+  config.bit_rate = -1;
+  S(TestBadConfig(config));
+  config.bit_rate = 1;
+  S(TestBadConfig(config));
+  config.bit_rate = std::numeric_limits<int>::max();
+  S(TestBadConfig(config));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
index c4ceb59062..51da3f7c76 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
@@ -20,7 +20,9 @@
 #include "settings.h"
 #include "isac.h"
 
+#include <assert.h>
 #include <math.h>
+#include <string.h>
 
 /* array of quantization levels for bottle neck info; Matlab code: */
 /* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
@@ -119,6 +121,9 @@ int32_t WebRtcIsac_InitBandwidthEstimator(
   bwest_str->inWaitLatePkts = 0;
   bwest_str->senderTimestamp = 0;
   bwest_str->receiverTimestamp = 0;
+
+  bwest_str->external_bw_info.in_use = 0;
+
   return 0;
 }
 
@@ -132,12 +137,12 @@ int32_t WebRtcIsac_InitBandwidthEstimator(
 /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
 /* returns 0 if everything went fine, -1 otherwise                                                   */
 int16_t WebRtcIsac_UpdateBandwidthEstimator(
-    BwEstimatorstr *bwest_str,
+    BwEstimatorstr* bwest_str,
     const uint16_t rtp_number,
-    const int32_t  frame_length,
+    const int32_t frame_length,
     const uint32_t send_ts,
     const uint32_t arr_ts,
-    const int32_t  pksize
+    const size_t pksize
     /*,    const uint16_t Index*/)
 {
   float weight = 0.0f;
@@ -154,6 +159,7 @@ int16_t WebRtcIsac_UpdateBandwidthEstimator(
   int immediate_set = 0;
   int num_pkts_expected;
 
+  assert(!bwest_str->external_bw_info.in_use);
 
   // We have to adjust the header-rate if the first packet has a
   // frame-size different than the initialized value.
@@ -508,6 +514,8 @@ int16_t WebRtcIsac_UpdateUplinkBwImpl(
     int16_t               index,
     enum IsacSamplingRate encoderSamplingFreq)
 {
+  assert(!bwest_str->external_bw_info.in_use);
+
   if((index < 0) || (index > 23))
   {
     return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
@@ -564,6 +572,8 @@ int16_t WebRtcIsac_UpdateUplinkJitter(
     BwEstimatorstr*              bwest_str,
     int32_t                  index)
 {
+  assert(!bwest_str->external_bw_info.in_use);
+
   if((index < 0) || (index > 23))
   {
     return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
@@ -589,7 +599,7 @@ int16_t WebRtcIsac_UpdateUplinkJitter(
 
 // Returns the bandwidth/jitter estimation code (integer 0...23)
 // to put in the sending iSAC payload
-uint16_t
+void
 WebRtcIsac_GetDownlinkBwJitIndexImpl(
     BwEstimatorstr*           bwest_str,
     int16_t*              bottleneckIndex,
@@ -609,6 +619,12 @@ WebRtcIsac_GetDownlinkBwJitIndexImpl(
   int16_t maxInd;
   int16_t midInd;
 
+  if (bwest_str->external_bw_info.in_use) {
+    *bottleneckIndex = bwest_str->external_bw_info.bottleneck_idx;
+    *jitterInfo = bwest_str->external_bw_info.jitter_info;
+    return;
+  }
+
   /* Get Max Delay Bit */
   /* get unquantized max delay */
   MaxDelay = (float)WebRtcIsac_GetDownlinkMaxDelay(bwest_str);
@@ -684,8 +700,6 @@ WebRtcIsac_GetDownlinkBwJitIndexImpl(
 
   bwest_str->rec_bw_avg = (1 - weight) * bwest_str->rec_bw_avg + weight *
       (rate + bwest_str->rec_header_rate);
-
-  return 0;
 }
 
 
@@ -697,6 +711,8 @@ int32_t WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str)
   float   jitter_sign;
   float   bw_adjust;
 
+  assert(!bwest_str->external_bw_info.in_use);
+
   /* create a value between -1.0 and 1.0 indicating "average sign" of jitter */
   jitter_sign = bwest_str->rec_jitter_short_term /
       bwest_str->rec_jitter_short_term_abs;
@@ -725,6 +741,8 @@ WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
 {
   int32_t rec_max_delay;
 
+  assert(!bwest_str->external_bw_info.in_use);
+
   rec_max_delay = (int32_t)(bwest_str->rec_max_delay);
 
   /* limit range of jitter estimate */
@@ -739,48 +757,41 @@ WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
   return rec_max_delay;
 }
 
-/* get the bottle neck rate from here to far side, as estimated by far side */
-void
-WebRtcIsac_GetUplinkBandwidth(
-    const BwEstimatorstr* bwest_str,
-    int32_t*          bitRate)
-{
-  /* limit range of bottle neck rate */
-  if (bwest_str->send_bw_avg < MIN_ISAC_BW)
-  {
-    *bitRate = MIN_ISAC_BW;
-  }
-  else if (bwest_str->send_bw_avg > MAX_ISAC_BW)
-  {
-    *bitRate = MAX_ISAC_BW;
-  }
-  else
-  {
-    *bitRate = (int32_t)(bwest_str->send_bw_avg);
-  }
-  return;
+/* Clamp val to the closed interval [min,max]. */
+static int32_t clamp(int32_t val, int32_t min, int32_t max) {
+  assert(min <= max);
+  return val < min ? min : (val > max ? max : val);
 }
 
-/* Returns the max delay value from the other side in ms */
-int32_t
-WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str)
-{
-  int32_t send_max_delay;
-
-  send_max_delay = (int32_t)(bwest_str->send_max_delay_avg);
-
-  /* limit range of jitter estimate */
-  if (send_max_delay < MIN_ISAC_MD)
-  {
-    send_max_delay = MIN_ISAC_MD;
-  }
-  else if (send_max_delay > MAX_ISAC_MD)
-  {
-    send_max_delay = MAX_ISAC_MD;
-  }
-  return send_max_delay;
+int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str) {
+  return bwest_str->external_bw_info.in_use
+             ? bwest_str->external_bw_info.send_bw_avg
+             : clamp(bwest_str->send_bw_avg, MIN_ISAC_BW, MAX_ISAC_BW);
 }
 
+int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str) {
+  return bwest_str->external_bw_info.in_use
+             ? bwest_str->external_bw_info.send_max_delay_avg
+             : clamp(bwest_str->send_max_delay_avg, MIN_ISAC_MD, MAX_ISAC_MD);
+}
+
+void WebRtcIsacBw_GetBandwidthInfo(BwEstimatorstr* bwest_str,
+                                   enum IsacSamplingRate decoder_sample_rate_hz,
+                                   IsacBandwidthInfo* bwinfo) {
+  assert(!bwest_str->external_bw_info.in_use);
+  bwinfo->in_use = 1;
+  bwinfo->send_bw_avg = WebRtcIsac_GetUplinkBandwidth(bwest_str);
+  bwinfo->send_max_delay_avg = WebRtcIsac_GetUplinkMaxDelay(bwest_str);
+  WebRtcIsac_GetDownlinkBwJitIndexImpl(bwest_str, &bwinfo->bottleneck_idx,
+                                       &bwinfo->jitter_info,
+                                       decoder_sample_rate_hz);
+}
+
+void WebRtcIsacBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
+                                   const IsacBandwidthInfo* bwinfo) {
+  memcpy(&bwest_str->external_bw_info, bwinfo,
+         sizeof bwest_str->external_bw_info);
+}
 
 /*
  * update long-term average bitrate and amount of data in buffer
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
index edabdff5cf..0704337f7d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
@@ -90,12 +90,12 @@ extern "C" {
   /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
   /* returns 0 if everything went fine, -1 otherwise                                                   */
   int16_t WebRtcIsac_UpdateBandwidthEstimator(
-      BwEstimatorstr*    bwest_str,
+      BwEstimatorstr* bwest_str,
       const uint16_t rtp_number,
-      const int32_t  frame_length,
+      const int32_t frame_length,
       const uint32_t send_ts,
       const uint32_t arr_ts,
-      const int32_t  pksize);
+      const size_t pksize);
 
   /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
   int16_t WebRtcIsac_UpdateUplinkBwImpl(
@@ -104,10 +104,10 @@ extern "C" {
       enum IsacSamplingRate encoderSamplingFreq);
 
   /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
-  uint16_t WebRtcIsac_GetDownlinkBwJitIndexImpl(
-      BwEstimatorstr*           bwest_str,
-      int16_t*              bottleneckIndex,
-      int16_t*              jitterInfo,
+  void WebRtcIsac_GetDownlinkBwJitIndexImpl(
+      BwEstimatorstr* bwest_str,
+      int16_t* bottleneckIndex,
+      int16_t* jitterInfo,
       enum IsacSamplingRate decoderSamplingFreq);
 
   /* Returns the bandwidth estimation (in bps) */
@@ -119,14 +119,21 @@ extern "C" {
       const BwEstimatorstr *bwest_str);
 
   /* Returns the bandwidth that iSAC should send with in bps */
-  void WebRtcIsac_GetUplinkBandwidth(
-      const BwEstimatorstr* bwest_str,
-      int32_t*          bitRate);
+  int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str);
 
   /* Returns the max delay value from the other side in ms */
   int32_t WebRtcIsac_GetUplinkMaxDelay(
       const BwEstimatorstr *bwest_str);
 
+  /* Fills in an IsacExternalBandwidthInfo struct. */
+  void WebRtcIsacBw_GetBandwidthInfo(
+      BwEstimatorstr* bwest_str,
+      enum IsacSamplingRate decoder_sample_rate_hz,
+      IsacBandwidthInfo* bwinfo);
+
+  /* Uses the values from an IsacExternalBandwidthInfo struct. */
+  void WebRtcIsacBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
+                                     const IsacBandwidthInfo* bwinfo);
 
   /*
    * update amount of data in bottle neck buffer and burst handling
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
index b7319b9ec3..7ef64b55fe 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
@@ -25,22 +25,26 @@
 void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
 
 int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
-                                 int32_t packet_size,
+                                 size_t packet_size,
                                  uint16_t rtp_seq_number,
                                  uint32_t send_ts, uint32_t arr_ts,
                                  enum IsacSamplingRate encoderSampRate,
                                  enum IsacSamplingRate decoderSampRate);
 
-int WebRtcIsac_DecodeLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
+int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
+                        float* signal_out,
+                        ISACLBDecStruct* ISACdec_obj,
                         int16_t* current_framesamples,
                         int16_t isRCUPayload);
 
 int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
                            int16_t* current_framesamples);
 
-int WebRtcIsac_EncodeLb(float* in, ISACLBEncStruct* ISACencLB_obj,
-                        int16_t codingMode, int16_t
-                        bottleneckIndex);
+int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
+                        float* in,
+                        ISACLBEncStruct* ISACencLB_obj,
+                        int16_t codingMode,
+                        int16_t bottleneckIndex);
 
 int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
                                   Bitstr* ISACBitStr_obj, int BWnumber,
@@ -93,10 +97,11 @@ int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
  * Return value              : >0 number of decoded bytes.
  *                             <0 if an error occurred.
  */
-int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdec_obj,
+int WebRtcIsac_DecodeUb16(const TransformTables* transform_tables,
+                          float* signal_out,
+                          ISACUBDecStruct* ISACdec_obj,
                           int16_t isRCUPayload);
 
-
 /******************************************************************************
  * WebRtcIsac_DecodeUb12()
  *
@@ -112,10 +117,11 @@ int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdec_obj,
  * Return value              : >0 number of decoded bytes.
  *                             <0 if an error occurred.
  */
-int WebRtcIsac_DecodeUb12(float* signal_out, ISACUBDecStruct* ISACdec_obj,
+int WebRtcIsac_DecodeUb12(const TransformTables* transform_tables,
+                          float* signal_out,
+                          ISACUBDecStruct* ISACdec_obj,
                           int16_t isRCUPayload);
 
-
 /******************************************************************************
  * WebRtcIsac_EncodeUb16()
  *
@@ -131,10 +137,11 @@ int WebRtcIsac_DecodeUb12(float* signal_out, ISACUBDecStruct* ISACdec_obj,
  * Return value              : >0 number of encoded bytes.
  *                             <0 if an error occurred.
  */
-int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACenc_obj,
+int WebRtcIsac_EncodeUb16(const TransformTables* transform_tables,
+                          float* in,
+                          ISACUBEncStruct* ISACenc_obj,
                           int32_t jitterInfo);
 
-
 /******************************************************************************
  * WebRtcIsac_EncodeUb12()
  *
@@ -150,7 +157,9 @@ int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACenc_obj,
  * Return value              : >0 number of encoded bytes.
  *                             <0 if an error occurred.
  */
-int WebRtcIsac_EncodeUb12(float* in, ISACUBEncStruct* ISACenc_obj,
+int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables,
+                          float* in,
+                          ISACUBEncStruct* ISACenc_obj,
                           int32_t jitterInfo);
 
 /************************** initialization functions *************************/
@@ -168,25 +177,32 @@ void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State);
 
 /**************************** transform functions ****************************/
 
-void WebRtcIsac_InitTransform();
+void WebRtcIsac_InitTransform(TransformTables* tables);
 
-void WebRtcIsac_Time2Spec(double* inre1, double* inre2, int16_t* outre,
-                          int16_t* outim, FFTstr* fftstr_obj);
-
-void WebRtcIsac_Spec2time(double* inre, double* inim, double* outre1,
-                          double* outre2, FFTstr* fftstr_obj);
+void WebRtcIsac_Time2Spec(const TransformTables* tables,
+                          double* inre1,
+                          double* inre2,
+                          int16_t* outre,
+                          int16_t* outim,
+                          FFTstr* fftstr_obj);
 
+void WebRtcIsac_Spec2time(const TransformTables* tables,
+                          double* inre,
+                          double* inim,
+                          double* outre1,
+                          double* outre2,
+                          FFTstr* fftstr_obj);
 
 /******************************* filter functions ****************************/
 
-void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, int lengthInOut,
+void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, size_t lengthInOut,
                               int orderCoef);
 
-void WebRtcIsac_AllZeroFilter(double* In, double* Coef, int lengthInOut,
+void WebRtcIsac_AllZeroFilter(double* In, double* Coef, size_t lengthInOut,
                               int orderCoef, double* Out);
 
 void WebRtcIsac_ZeroPoleFilter(double* In, double* ZeroCoef, double* PoleCoef,
-                               int lengthInOut, int orderCoef, double* Out);
+                               size_t lengthInOut, int orderCoef, double* Out);
 
 
 /***************************** filterbank functions **************************/
@@ -212,6 +228,6 @@ void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
 
 void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);
 
-void WebRtcIsac_AutoCorr(double* r, const double* x, int N, int order);
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
index 06c15cb390..2419e24033 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
@@ -80,9 +80,9 @@ static const uint32_t kCrcTable[256] = {
  *                             -1 - Error
  */
 
-int16_t WebRtcIsac_GetCrc(const int16_t* bitstream,
-                          int16_t        len_bitstream_in_bytes,
-                          uint32_t*      crc)
+int WebRtcIsac_GetCrc(const int16_t* bitstream,
+                      int len_bitstream_in_bytes,
+                      uint32_t* crc)
 {
   uint8_t* bitstream_ptr_uw8;
   uint32_t crc_state;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
index 19d1bf31d5..09583dfc5c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
@@ -36,10 +36,10 @@
  *                   -1 - Error
  */
 
-int16_t WebRtcIsac_GetCrc(
+int WebRtcIsac_GetCrc(
     const int16_t* encoded,
-    int16_t        no_of_word8s,
-    uint32_t*      crc);
+    int no_of_word8s,
+    uint32_t* crc);
 
 
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c
index e23765bbbe..e925efba6e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c
@@ -35,7 +35,8 @@
  * function to decode the bitstream
  * returns the total number of bytes in the stream
  */
-int WebRtcIsac_DecodeLb(float* signal_out, ISACLBDecStruct* ISACdecLB_obj,
+int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
+                        float* signal_out, ISACLBDecStruct* ISACdecLB_obj,
                         int16_t* current_framesamples,
                         int16_t isRCUPayload) {
   int k;
@@ -122,7 +123,7 @@ int WebRtcIsac_DecodeLb(float* signal_out, ISACLBDecStruct* ISACdecLB_obj,
     }
 
     /* Inverse transform. */
-    WebRtcIsac_Spec2time(real_f, imag_f, LPw, HPw,
+    WebRtcIsac_Spec2time(transform_tables, real_f, imag_f, LPw, HPw,
                          &ISACdecLB_obj->fftstr_obj);
 
     /* Convert PitchGains back to float for pitchfilter_post */
@@ -181,7 +182,8 @@ int WebRtcIsac_DecodeLb(float* signal_out, ISACLBDecStruct* ISACdecLB_obj,
  * Contrary to lower-band, the upper-band (8-16 kHz) is not split in
  * frequency, but split to 12 sub-frames, i.e. twice as lower-band.
  */
-int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
+int WebRtcIsac_DecodeUb16(const TransformTables* transform_tables,
+                          float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
                           int16_t isRCUPayload) {
   int len, err;
 
@@ -218,7 +220,8 @@ int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
     }
   }
   /* Inverse transform. */
-  WebRtcIsac_Spec2time(real_f, imag_f, halfFrameFirst, halfFrameSecond,
+  WebRtcIsac_Spec2time(transform_tables,
+                       real_f, imag_f, halfFrameFirst, halfFrameSecond,
                        &ISACdecUB_obj->fftstr_obj);
 
   /* Perceptual post-filtering (using normalized lattice filter). */
@@ -245,8 +248,9 @@ int WebRtcIsac_DecodeUb16(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
  * reconstructed and 12-16 kHz replaced with zeros. Then two bands
  * are combined, to reconstruct the upperband 8-16 kHz.
  */
-int WebRtcIsac_DecodeUb12(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
-                      int16_t isRCUPayload) {
+int WebRtcIsac_DecodeUb12(const TransformTables* transform_tables,
+                          float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
+                          int16_t isRCUPayload) {
   int len, err;
 
   float LP_dec_float[FRAMESAMPLES_HALF];
@@ -284,7 +288,8 @@ int WebRtcIsac_DecodeUb12(float* signal_out, ISACUBDecStruct* ISACdecUB_obj,
     }
   }
   /* Inverse transform. */
-  WebRtcIsac_Spec2time(real_f, imag_f, LPw, HPw, &ISACdecUB_obj->fftstr_obj);
+  WebRtcIsac_Spec2time(transform_tables,
+                       real_f, imag_f, LPw, HPw, &ISACdecUB_obj->fftstr_obj);
   /* perceptual post-filtering (using normalized lattice filter) */
   WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
                                  ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
index 5abe2041f9..019cc89528 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
@@ -18,7 +18,7 @@ int
 WebRtcIsac_EstimateBandwidth(
     BwEstimatorstr*           bwest_str,
     Bitstr*                   streamdata,
-    int32_t               packet_size,
+    size_t                packet_size,
     uint16_t              rtp_seq_number,
     uint32_t              send_ts,
     uint32_t              arr_ts,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
index a089f72cdf..3f1912b6d3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
@@ -177,7 +177,8 @@ void WebRtcIsac_ResetBitstream(Bitstr* bit_stream) {
   bit_stream->streamval = 0;
 }
 
-int WebRtcIsac_EncodeLb(float* in, ISACLBEncStruct* ISACencLB_obj,
+int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
+                        float* in, ISACLBEncStruct* ISACencLB_obj,
                         int16_t codingMode,
                         int16_t bottleneckIndex) {
   int stream_length = 0;
@@ -382,7 +383,8 @@ int WebRtcIsac_EncodeLb(float* in, ISACLBEncStruct* ISACencLB_obj,
   WebRtcIsac_PitchfilterPre(LPw, LPw_pf, &ISACencLB_obj->pitchfiltstr_obj,
                             PitchLags, PitchGains);
   /* Transform */
-  WebRtcIsac_Time2Spec(LPw_pf, HPw, fre, fim, &ISACencLB_obj->fftstr_obj);
+  WebRtcIsac_Time2Spec(transform_tables,
+                       LPw_pf, HPw, fre, fim, &ISACencLB_obj->fftstr_obj);
 
   /* Save data for multiple packets memory. */
   my_index = ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF;
@@ -641,7 +643,8 @@ static int LimitPayloadUb(ISACUBEncStruct* ISACencUB_obj,
   return 0;
 }
 
-int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACencUB_obj,
+int WebRtcIsac_EncodeUb16(const TransformTables* transform_tables,
+                          float* in, ISACUBEncStruct* ISACencUB_obj,
                           int32_t jitterInfo) {
   int err;
   int k;
@@ -782,7 +785,8 @@ int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACencUB_obj,
       &percepFilterParams[(UB_LPC_ORDER + 1) + SUBFRAMES * (UB_LPC_ORDER + 1)],
       &LP_lookahead[FRAMESAMPLES_HALF]);
 
-  WebRtcIsac_Time2Spec(&LP_lookahead[0], &LP_lookahead[FRAMESAMPLES_HALF],
+  WebRtcIsac_Time2Spec(transform_tables,
+                       &LP_lookahead[0], &LP_lookahead[FRAMESAMPLES_HALF],
                        fre, fim, &ISACencUB_obj->fftstr_obj);
 
   /* Store FFT coefficients for multiple encoding. */
@@ -826,7 +830,8 @@ int WebRtcIsac_EncodeUb16(float* in, ISACUBEncStruct* ISACencUB_obj,
 }
 
 
-int WebRtcIsac_EncodeUb12(float* in, ISACUBEncStruct* ISACencUB_obj,
+int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables,
+                          float* in, ISACUBEncStruct* ISACencUB_obj,
                           int32_t jitterInfo) {
   int err;
   int k;
@@ -957,7 +962,8 @@ int WebRtcIsac_EncodeUb12(float* in, ISACUBEncStruct* ISACencUB_obj,
   memset(HPw, 0, sizeof(HPw));
 
   /* Transform */
-  WebRtcIsac_Time2Spec(LPw, HPw, fre, fim, &ISACencUB_obj->fftstr_obj);
+  WebRtcIsac_Time2Spec(transform_tables,
+                       LPw, HPw, fre, fim, &ISACencUB_obj->fftstr_obj);
 
   /* Store FFT coefficients for multiple encoding. */
   memcpy(ISACencUB_obj->SaveEnc_obj.realFFT, fre,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
index d59f7489ee..12a263d14c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
@@ -440,7 +440,7 @@ WebRtcIsac_CorrelateInterVec(
   int16_t rowCntr;
   int16_t colCntr;
   int16_t interVecDim;
-  double myVec[UB16_LPC_VEC_PER_FRAME];
+  double myVec[UB16_LPC_VEC_PER_FRAME] = {0.0};
   const double* interVecDecorrMat;
 
   switch(bandwidth)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
index 76a9e7530d..d47eb1fa66 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
@@ -19,12 +19,15 @@
 
 
 
-void WebRtcIsac_AllPoleFilter(double *InOut, double *Coef, int lengthInOut, int orderCoef){
-
+void WebRtcIsac_AllPoleFilter(double* InOut,
+                              double* Coef,
+                              size_t lengthInOut,
+                              int orderCoef) {
   /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
   double scal;
   double sum;
-  int n,k;
+  size_t n;
+  int k;
 
   //if (fabs(Coef[0]-1.0)<0.001) {
   if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
@@ -53,11 +56,15 @@ void WebRtcIsac_AllPoleFilter(double *InOut, double *Coef, int lengthInOut, int
 }
 
 
-void WebRtcIsac_AllZeroFilter(double *In, double *Coef, int lengthInOut, int orderCoef, double *Out){
-
+void WebRtcIsac_AllZeroFilter(double* In,
+                              double* Coef,
+                              size_t lengthInOut,
+                              int orderCoef,
+                              double* Out) {
   /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
 
-  int n, k;
+  size_t n;
+  int k;
   double tmp;
 
   for(n = 0; n < lengthInOut; n++)
@@ -74,9 +81,12 @@ void WebRtcIsac_AllZeroFilter(double *In, double *Coef, int lengthInOut, int ord
 }
 
 
-
-void WebRtcIsac_ZeroPoleFilter(double *In, double *ZeroCoef, double *PoleCoef, int lengthInOut, int orderCoef, double *Out){
-
+void WebRtcIsac_ZeroPoleFilter(double* In,
+                               double* ZeroCoef,
+                               double* PoleCoef,
+                               size_t lengthInOut,
+                               int orderCoef,
+                               double* Out) {
   /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
   /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
 
@@ -85,14 +95,8 @@ void WebRtcIsac_ZeroPoleFilter(double *In, double *ZeroCoef, double *PoleCoef, i
 }
 
 
-void WebRtcIsac_AutoCorr(
-    double *r,
-    const double *x,
-    int N,
-    int order
-                        )
-{
-  int  lag, n;
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
+  size_t  lag, n;
   double sum, prod;
   const double *x_lag;
 
@@ -112,8 +116,8 @@ void WebRtcIsac_AutoCorr(
 }
 
 
-void WebRtcIsac_BwExpand(double *out, double *in, double coef, short length) {
-  int i;
+void WebRtcIsac_BwExpand(double* out, double* in, double coef, size_t length) {
+  size_t i;
   double  chirp;
 
   chirp = coef;
@@ -125,8 +129,10 @@ void WebRtcIsac_BwExpand(double *out, double *in, double coef, short length) {
   }
 }
 
-void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata) {
-
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata) {
   double  tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
   double  corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
   double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
@@ -195,15 +201,13 @@ static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
 static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
 
 
-
-void WebRtcIsac_AllpassFilterForDec(double *InOut,
-                                   const double *APSectionFactors,
-                                   int lengthInOut,
-                                   double *FilterState)
-{
+void WebRtcIsac_AllpassFilterForDec(double* InOut,
+                                    const double* APSectionFactors,
+                                    size_t lengthInOut,
+                                    double* FilterState) {
   //This performs all-pass filtering--a series of first order all-pass sections are used
   //to filter the input in a cascade manner.
-  int n,j;
+  size_t n,j;
   double temp;
   for (j=0; j<ALLPASSSECTIONS; j++){
     for (n=0;n<lengthInOut;n+=2){
@@ -214,12 +218,11 @@ void WebRtcIsac_AllpassFilterForDec(double *InOut,
   }
 }
 
-void WebRtcIsac_DecimateAllpass(const double *in,
-                                double *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
-                                int N,                   /* number of input samples */
-                                double *out)             /* array of size N/2 */
-{
-  int n;
+void WebRtcIsac_DecimateAllpass(const double* in,
+                                double* state_in,
+                                size_t N,
+                                double* out) {
+  size_t n;
   double data_vec[PITCH_FRAME_LEN];
 
   /* copy input */
@@ -237,7 +240,6 @@ void WebRtcIsac_DecimateAllpass(const double *in,
 }
 
 
-
 /* create high-pass filter ocefficients
  * z = 0.998 * exp(j*2*pi*35/8000);
  * p = 0.94 * exp(j*2*pi*140/8000);
@@ -245,13 +247,13 @@ void WebRtcIsac_DecimateAllpass(const double *in,
  * HP_a = [1, -2*real(p), abs(p)^2]; */
 static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
 static const double b_coef[2] = {-1.99524591718270,  0.99600400000000};
-static const float a_coef_float[2] = { 1.86864659625574f, -0.88360000000000f};
-static const float b_coef_float[2] = {-1.99524591718270f,  0.99600400000000f};
 
 /* second order high-pass filter */
-void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N)
-{
-  int k;
+void WebRtcIsac_Highpass(const double* in,
+                         double* out,
+                         double* state,
+                         size_t N) {
+  size_t k;
 
   for (k=0; k<N; k++) {
     *out = *in + state[1];
@@ -259,14 +261,3 @@ void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N)
     state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
   }
 }
-
-void WebRtcIsac_Highpass_float(const float *in, double *out, double *state, int N)
-{
-  int k;
-
-  for (k=0; k<N; k++) {
-    *out = (double)*in + state[1];
-    state[1] = state[0] + b_coef_float[0] * *in + a_coef_float[0] * *out;
-    state[0] = b_coef_float[1] * (double)*in++ + a_coef_float[1] * *out++;
-  }
-}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
index 6df034d1c7..01e683c573 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
@@ -40,10 +40,6 @@ void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) {
   }
 
   maskdata->OldEnergy = 10.0;
-
-  /* fill tables for transforms */
-  WebRtcIsac_InitTransform();
-
   return;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index db78e6de2e..875e7ac521 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -15,8 +15,9 @@
  *
  */
 
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 
+#include <assert.h>
 #include <math.h>
 #include <stdio.h>
 #include <stdlib.h>
@@ -113,9 +114,8 @@ static void UpdateBottleneck(ISACMainStruct* instISAC) {
   if ((instISAC->codingMode == 0) &&
       (instISAC->instLB.ISACencLB_obj.buffer_index == 0) &&
       (instISAC->instLB.ISACencLB_obj.frame_nb == 0)) {
-    int32_t bottleneck;
-    WebRtcIsac_GetUplinkBandwidth(&(instISAC->bwestimator_obj),
-                                  &bottleneck);
+    int32_t bottleneck =
+        WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj);
 
     /* Adding hysteresis when increasing signal bandwidth. */
     if ((instISAC->bandwidthKHz == isac8kHz)
@@ -251,6 +251,8 @@ int16_t WebRtcIsac_Assign(ISACStruct** ISAC_main_inst,
     instISAC->decoderSamplingRateKHz = kIsacWideband;
     instISAC->bandwidthKHz           = isac8kHz;
     instISAC->in_sample_rate_hz = 16000;
+
+    WebRtcIsac_InitTransform(&instISAC->transform_tables);
     return 0;
   } else {
     return -1;
@@ -284,6 +286,8 @@ int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
       instISAC->encoderSamplingRateKHz = kIsacWideband;
       instISAC->decoderSamplingRateKHz = kIsacWideband;
       instISAC->in_sample_rate_hz = 16000;
+
+      WebRtcIsac_InitTransform(&instISAC->transform_tables);
       return 0;
     } else {
       return -1;
@@ -465,7 +469,6 @@ int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
       return -1;
     }
   }
-  memset(instISAC->state_in_resampler, 0, sizeof(instISAC->state_in_resampler));
   /* Initialization is successful, set the flag. */
   instISAC->initFlag |= BIT_MASK_ENC_INIT;
   return 0;
@@ -494,17 +497,17 @@ int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
  *                                 samples.
  *                            : -1 - Error
  */
-int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
-                          const int16_t* speechIn,
-                          uint8_t* encoded) {
+int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
+                      const int16_t* speechIn,
+                      uint8_t* encoded) {
   float inFrame[FRAMESAMPLES_10ms];
   int16_t speechInLB[FRAMESAMPLES_10ms];
   int16_t speechInUB[FRAMESAMPLES_10ms];
-  int16_t streamLenLB = 0;
-  int16_t streamLenUB = 0;
-  int16_t streamLen = 0;
-  int16_t k = 0;
-  int garbageLen = 0;
+  int streamLenLB = 0;
+  int streamLenUB = 0;
+  int streamLen = 0;
+  size_t k = 0;
+  uint8_t garbageLen = 0;
   int32_t bottleneck = 0;
   int16_t bottleneckIdx = 0;
   int16_t jitterInfo = 0;
@@ -512,8 +515,6 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   ISACLBStruct* instLB = &(instISAC->instLB);
   ISACUBStruct* instUB = &(instISAC->instUB);
-  const int16_t* speech_in_ptr = speechIn;
-  int16_t resampled_buff[FRAMESAMPLES_10ms * 2];
 
   /* Check if encoder initiated. */
   if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
@@ -522,37 +523,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
     return -1;
   }
 
-  if (instISAC->in_sample_rate_hz == 48000) {
-    /* Samples in 10 ms @ 48 kHz. */
-    const int kNumInputSamples = FRAMESAMPLES_10ms * 3;
-    /* Samples 10 ms @ 32 kHz. */
-    const int kNumOutputSamples = FRAMESAMPLES_10ms * 2;
-    /* Resampler divide the input into blocks of 3 samples, i.e.
-     * kNumInputSamples / 3. */
-    const int kNumResamplerBlocks = FRAMESAMPLES_10ms;
-    int32_t buffer32[FRAMESAMPLES_10ms * 3 + SIZE_RESAMPLER_STATE];
-
-    /* Restore last samples from the past to the beginning of the buffer
-     * and store the last samples of current frame for the next resampling. */
-    for (k = 0; k < SIZE_RESAMPLER_STATE; k++) {
-      buffer32[k] = instISAC->state_in_resampler[k];
-      instISAC->state_in_resampler[k] = speechIn[kNumInputSamples -
-                                                 SIZE_RESAMPLER_STATE + k];
-    }
-    for (k = 0; k < kNumInputSamples; k++) {
-      buffer32[SIZE_RESAMPLER_STATE + k] = speechIn[k];
-    }
-    /* Resampling 3 samples to 2. Function divides the input in
-     * |kNumResamplerBlocks| number of 3-sample groups, and output is
-     * |kNumResamplerBlocks| number of 2-sample groups. */
-    WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, kNumResamplerBlocks);
-    WebRtcSpl_VectorBitShiftW32ToW16(resampled_buff, kNumOutputSamples,
-                                     buffer32, 15);
-    speech_in_ptr = resampled_buff;
-  }
-
   if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
-    WebRtcSpl_AnalysisQMF(speech_in_ptr, SWBFRAMESAMPLES_10ms, speechInLB,
+    WebRtcSpl_AnalysisQMF(speechIn, SWBFRAMESAMPLES_10ms, speechInLB,
                           speechInUB, instISAC->analysisFBState1,
                           instISAC->analysisFBState2);
 
@@ -579,7 +551,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
   GetSendBandwidthInfo(instISAC, &bottleneckIdx, &jitterInfo);
 
   /* Encode lower-band. */
-  streamLenLB = WebRtcIsac_EncodeLb(inFrame, &instLB->ISACencLB_obj,
+  streamLenLB = WebRtcIsac_EncodeLb(&instISAC->transform_tables,
+                                    inFrame, &instLB->ISACencLB_obj,
                                     instISAC->codingMode, bottleneckIdx);
   if (streamLenLB < 0) {
     return -1;
@@ -601,17 +574,19 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
 
     /* Tell to upper-band the number of bytes used so far.
      * This is for payload limitation. */
-    instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
-                                         LEN_CHECK_SUM_WORD8;
+    instUB->ISACencUB_obj.numBytesUsed =
+        (int16_t)(streamLenLB + 1 + LEN_CHECK_SUM_WORD8);
     /* Encode upper-band. */
     switch (instISAC->bandwidthKHz) {
       case isac12kHz: {
-        streamLenUB = WebRtcIsac_EncodeUb12(inFrame, &instUB->ISACencUB_obj,
+        streamLenUB = WebRtcIsac_EncodeUb12(&instISAC->transform_tables,
+                                            inFrame, &instUB->ISACencUB_obj,
                                             jitterInfo);
         break;
       }
       case isac16kHz: {
-        streamLenUB = WebRtcIsac_EncodeUb16(inFrame, &instUB->ISACencUB_obj,
+        streamLenUB = WebRtcIsac_EncodeUb16(&instISAC->transform_tables,
+                                            inFrame, &instUB->ISACencUB_obj,
                                             jitterInfo);
         break;
       }
@@ -645,7 +620,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
     memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
     streamLen = streamLenLB;
     if (streamLenUB > 0) {
-      encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
+      encoded[streamLenLB] = (uint8_t)(streamLenUB + 1 + LEN_CHECK_SUM_WORD8);
       memcpy(&encoded[streamLenLB + 1],
              instUB->ISACencUB_obj.bitstr_obj.stream,
              streamLenUB);
@@ -663,7 +638,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
   }
 
   /* Add Garbage if required. */
-  WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj, &bottleneck);
+  bottleneck = WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj);
   if (instISAC->codingMode == 0) {
     int minBytes;
     int limit;
@@ -703,14 +678,15 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
     }
     minBytes = (minBytes > limit) ? limit : minBytes;
 
-    garbageLen = (minBytes > streamLen) ? (minBytes - streamLen) : 0;
+    garbageLen = (minBytes > streamLen) ? (uint8_t)(minBytes - streamLen) : 0;
 
     /* Save data for creation of multiple bit-streams. */
     /* If bit-stream too short then add garbage at the end. */
     if (garbageLen > 0) {
-      for (k = 0; k < garbageLen; k++) {
-        ptrGarbage[k] = (uint8_t)(rand() & 0xFF);
-      }
+      /* Overwrite the garbage area to avoid leaking possibly sensitive data
+         over the network. This also makes the output deterministic. */
+      memset(ptrGarbage, 0, garbageLen);
+
       /* For a correct length of the upper-band bit-stream together
        * with the garbage. Garbage is embeded in upper-band bit-stream.
        * That is the only way to preserve backward compatibility. */
@@ -742,7 +718,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
                       streamLenUB + garbageLen, &crc);
 #ifndef WEBRTC_ARCH_BIG_ENDIAN
     for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
-      encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = crc >> (24 - k * 8);
+      encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] =
+          (uint8_t)(crc >> (24 - k * 8));
     }
 #else
     memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, LEN_CHECK_SUM_WORD8);
@@ -915,12 +892,8 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct*  ISAC_main_inst,
  *
  * Input:
  *        - ISAC_main_inst    : ISAC instance.
- *
- * Return value
- *                            :  0 - Ok
- *                              -1 - Error
  */
-static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
+static void DecoderInitLb(ISACLBStruct* instISAC) {
   int i;
   /* Initialize stream vector to zero. */
   for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@@ -931,10 +904,9 @@ static int16_t DecoderInitLb(ISACLBStruct* instISAC) {
   WebRtcIsac_InitPostFilterbank(
     &instISAC->ISACdecLB_obj.postfiltbankstr_obj);
   WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj);
-  return 0;
 }
 
-static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
+static void DecoderInitUb(ISACUBStruct* instISAC) {
   int i;
   /* Init stream vector to zero */
   for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
@@ -944,24 +916,18 @@ static int16_t DecoderInitUb(ISACUBStruct* instISAC) {
   WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj);
   WebRtcIsac_InitPostFilterbank(
     &instISAC->ISACdecUB_obj.postfiltbankstr_obj);
-  return (0);
 }
 
-int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
+void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
-  if (DecoderInitLb(&instISAC->instLB) < 0) {
-    return -1;
-  }
+  DecoderInitLb(&instISAC->instLB);
   if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) {
     memset(instISAC->synthesisFBState1, 0,
            FB_STATE_SIZE_WORD32 * sizeof(int32_t));
     memset(instISAC->synthesisFBState2, 0,
            FB_STATE_SIZE_WORD32 * sizeof(int32_t));
-
-    if (DecoderInitUb(&(instISAC->instUB)) < 0) {
-      return -1;
-    }
+    DecoderInitUb(&(instISAC->instUB));
   }
   if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) {
     WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
@@ -970,7 +936,6 @@ int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
   }
   instISAC->initFlag |= BIT_MASK_DEC_INIT;
   instISAC->resetFlag_8kHz = 0;
-  return 0;
 }
 
 
@@ -997,7 +962,7 @@ int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
  */
 int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
                                     const uint8_t* encoded,
-                                    int32_t packet_size,
+                                    size_t packet_size,
                                     uint16_t rtp_seq_number,
                                     uint32_t send_ts,
                                     uint32_t arr_ts) {
@@ -1045,12 +1010,12 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
   return 0;
 }
 
-static int16_t Decode(ISACStruct* ISAC_main_inst,
-                      const uint8_t* encoded,
-                      int16_t lenEncodedBytes,
-                      int16_t* decoded,
-                      int16_t* speechType,
-                      int16_t isRCUPayload) {
+static int Decode(ISACStruct* ISAC_main_inst,
+                  const uint8_t* encoded,
+                  size_t lenEncodedBytes,
+                  int16_t* decoded,
+                  int16_t* speechType,
+                  int16_t isRCUPayload) {
   /* Number of samples (480 or 960), output from decoder
      that were actually used in the encoder/decoder
      (determined on the fly). */
@@ -1060,13 +1025,14 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
   float outFrame[MAX_FRAMESAMPLES];
   int16_t outFrameLB[MAX_FRAMESAMPLES];
   int16_t outFrameUB[MAX_FRAMESAMPLES];
-  int16_t numDecodedBytesLB;
-  int16_t numDecodedBytesUB;
-  int16_t lenEncodedLBBytes;
+  int numDecodedBytesLBint;
+  size_t numDecodedBytesLB;
+  int numDecodedBytesUB;
+  size_t lenEncodedLBBytes;
   int16_t validChecksum = 1;
   int16_t k;
   uint16_t numLayer;
-  int16_t totSizeBytes;
+  size_t totSizeBytes;
   int16_t err;
 
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
@@ -1080,7 +1046,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
     return -1;
   }
 
-  if (lenEncodedBytes <= 0) {
+  if (lenEncodedBytes == 0) {
     /* return error code if the packet length is null. */
     instISAC->errorCode = ISAC_EMPTY_PACKET;
     return -1;
@@ -1106,10 +1072,12 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
   /* Regardless of that the current codec is setup to work in
    * wideband or super-wideband, the decoding of the lower-band
    * has to be performed. */
-  numDecodedBytesLB = WebRtcIsac_DecodeLb(outFrame, decInstLB,
-                                          &numSamplesLB, isRCUPayload);
-
-  if ((numDecodedBytesLB < 0) || (numDecodedBytesLB > lenEncodedLBBytes) ||
+  numDecodedBytesLBint = WebRtcIsac_DecodeLb(&instISAC->transform_tables,
+                                             outFrame, decInstLB,
+                                             &numSamplesLB, isRCUPayload);
+  numDecodedBytesLB = (size_t)numDecodedBytesLBint;
+  if ((numDecodedBytesLBint < 0) ||
+      (numDecodedBytesLB > lenEncodedLBBytes) ||
       (numSamplesLB > MAX_FRAMESAMPLES)) {
     instISAC->errorCode = ISAC_LENGTH_MISMATCH;
     return -1;
@@ -1249,8 +1217,8 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
 
         switch (bandwidthKHz) {
           case isac12kHz: {
-            numDecodedBytesUB = WebRtcIsac_DecodeUb12(outFrame, decInstUB,
-                                                      isRCUPayload);
+            numDecodedBytesUB = WebRtcIsac_DecodeUb12(
+                &instISAC->transform_tables, outFrame, decInstUB, isRCUPayload);
 
             /* Hang-over for transient alleviation -
              * wait two frames to add the upper band going up from 8 kHz. */
@@ -1277,8 +1245,8 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
             break;
           }
           case isac16kHz: {
-            numDecodedBytesUB = WebRtcIsac_DecodeUb16(outFrame, decInstUB,
-                                                      isRCUPayload);
+            numDecodedBytesUB = WebRtcIsac_DecodeUb16(
+                &instISAC->transform_tables, outFrame, decInstUB, isRCUPayload);
             break;
           }
           default:
@@ -1350,11 +1318,11 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
  *                              -1 - Error
  */
 
-int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
-                          const uint8_t* encoded,
-                          int16_t lenEncodedBytes,
-                          int16_t* decoded,
-                          int16_t* speechType) {
+int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
+                      const uint8_t* encoded,
+                      size_t lenEncodedBytes,
+                      int16_t* decoded,
+                      int16_t* speechType) {
   int16_t isRCUPayload = 0;
   return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
                 speechType, isRCUPayload);
@@ -1382,11 +1350,11 @@ int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
 
 
 
-int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
-                             const uint8_t* encoded,
-                             int16_t lenEncodedBytes,
-                             int16_t* decoded,
-                             int16_t* speechType) {
+int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
+                         const uint8_t* encoded,
+                         size_t lenEncodedBytes,
+                         int16_t* decoded,
+                         int16_t* speechType) {
   int16_t isRCUPayload = 1;
   return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
                 speechType, isRCUPayload);
@@ -1407,13 +1375,12 @@ int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
  * Output:
  *        - decoded           : The decoded vector
  *
- * Return value               : >0 - number of samples in decoded PLC vector
- *                              -1 - Error
+ * Return value               : Number of samples in decoded PLC vector
  */
-int16_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst,
-                             int16_t* decoded,
-                             int16_t noOfLostFrames) {
-  int16_t numSamples = 0;
+size_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst,
+                            int16_t* decoded,
+                            size_t noOfLostFrames) {
+  size_t numSamples = 0;
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
 
   /* Limit number of frames to two = 60 millisecond.
@@ -1485,7 +1452,7 @@ static int16_t ControlUb(ISACUBStruct* instISAC, double rate) {
 
 int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
                            int32_t bottleneckBPS,
-                           int16_t frameSize) {
+                           int frameSize) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   int16_t status;
   double rateLB;
@@ -1526,7 +1493,7 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
     return -1;
   }
 
-  status = ControlLb(&instISAC->instLB, rateLB, frameSize);
+  status = ControlLb(&instISAC->instLB, rateLB, (int16_t)frameSize);
   if (status < 0) {
     instISAC->errorCode = -status;
     return -1;
@@ -1569,6 +1536,13 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
   return 0;
 }
 
+void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
+                                        int bottleneck_bits_per_second) {
+  ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
+  assert(bottleneck_bits_per_second >= 10000 &&
+         bottleneck_bits_per_second <= 32000);
+  instISAC->bwestimator_obj.send_bw_avg = (float)bottleneck_bits_per_second;
+}
 
 /****************************************************************************
  * WebRtcIsac_ControlBwe(...)
@@ -1594,7 +1568,7 @@ int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
  */
 int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
                               int32_t bottleneckBPS,
-                              int16_t frameSizeMs,
+                              int frameSizeMs,
                               int16_t enforceFrameSize) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   enum ISACBandwidth bandwidth;
@@ -1641,8 +1615,8 @@ int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
    *  will not change */
   if (frameSizeMs != 0) {
     if ((frameSizeMs  == 30) || (frameSizeMs == 60)) {
-      instISAC->instLB.ISACencLB_obj.new_framelength = (FS / 1000) *
-          frameSizeMs;
+      instISAC->instLB.ISACencLB_obj.new_framelength =
+          (int16_t)((FS / 1000) * frameSizeMs);
     } else {
       instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
       return -1;
@@ -1846,10 +1820,8 @@ int16_t WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst) {
   /* Return new frame length. */
   if (instISAC->in_sample_rate_hz == 16000)
     return (instISAC->instLB.ISACencLB_obj.new_framelength);
-  else if (instISAC->in_sample_rate_hz == 32000)
+  else  /* 32000 Hz */
     return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 2);
-  else
-    return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 3);
 }
 
 
@@ -2199,17 +2171,10 @@ void WebRtcIsac_version(char* version) {
  * and the bottleneck remain unchanged by this call, however, the maximum rate
  * and maximum payload-size will be reset to their default values.
  *
- * NOTE:
- * The maximum internal sampling rate is 32 kHz. If the encoder sample rate is
- * set to 48 kHz the input is expected to be at 48 kHz but will be resampled to
- * 32 kHz before any further processing.
- * This mode is created for compatibility with full-band codecs if iSAC is used
- * in dual-streaming. See SetDecSampleRate() for sampling rates at the decoder.
- *
  * Input:
  *        - ISAC_main_inst    : iSAC instance
- *        - sample_rate_hz    : sampling rate in Hertz, valid values are 16000,
- *                              32000 and 48000.
+ *        - sample_rate_hz    : sampling rate in Hertz, valid values are 16000
+ *                              and 32000.
  *
  * Return value               : 0 if successful
  *                             -1 if failed.
@@ -2219,8 +2184,7 @@ int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   enum IsacSamplingRate encoder_operational_rate;
 
-  if ((sample_rate_hz != 16000) && (sample_rate_hz != 32000) &&
-      (sample_rate_hz != 48000)) {
+  if ((sample_rate_hz != 16000) && (sample_rate_hz != 32000)) {
     /* Sampling Frequency is not supported. */
     instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
     return -1;
@@ -2335,9 +2299,7 @@ int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
       memset(instISAC->synthesisFBState2, 0,
              FB_STATE_SIZE_WORD32 * sizeof(int32_t));
 
-      if (DecoderInitUb(&(instISAC->instUB)) < 0) {
-        return -1;
-      }
+      DecoderInitUb(&instISAC->instUB);
   }
   instISAC->decoderSamplingRateKHz = decoder_operational_rate;
   return 0;
@@ -2375,3 +2337,27 @@ uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   return instISAC->decoderSamplingRateKHz == kIsacWideband ? 16000 : 32000;
 }
+
+void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst,
+                                 IsacBandwidthInfo* bwinfo) {
+  ISACMainStruct* instISAC = (ISACMainStruct*)inst;
+  assert(instISAC->initFlag & BIT_MASK_DEC_INIT);
+  WebRtcIsacBw_GetBandwidthInfo(&instISAC->bwestimator_obj,
+                                instISAC->decoderSamplingRateKHz, bwinfo);
+}
+
+void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
+                                 const IsacBandwidthInfo* bwinfo) {
+  ISACMainStruct* instISAC = (ISACMainStruct*)inst;
+  assert(instISAC->initFlag & BIT_MASK_ENC_INIT);
+  WebRtcIsacBw_SetBandwidthInfo(&instISAC->bwestimator_obj, bwinfo);
+}
+
+void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst,
+                                        int sample_rate_hz) {
+  ISACMainStruct* instISAC = (ISACMainStruct*)inst;
+  assert(instISAC->initFlag & BIT_MASK_DEC_INIT);
+  assert(!(instISAC->initFlag & BIT_MASK_ENC_INIT));
+  assert(sample_rate_hz == 16000 || sample_rate_hz == 32000);
+  instISAC->encoderSamplingRateKHz = sample_rate_hz / 1000;
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
similarity index 59%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
index 7d8ac7951b..e150d39261 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,26 +8,24 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
 
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 
 namespace webrtc {
 
 struct IsacFloat {
-  typedef ISACStruct instance_type;
+  using instance_type = ISACStruct;
   static const bool has_swb = true;
   static inline int16_t Control(instance_type* inst,
                                 int32_t rate,
-                                int16_t framesize) {
+                                int framesize) {
     return WebRtcIsac_Control(inst, rate, framesize);
   }
   static inline int16_t ControlBwe(instance_type* inst,
                                    int32_t rate_bps,
-                                   int16_t frame_size_ms,
+                                   int frame_size_ms,
                                    int16_t enforce_frame_size) {
     return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
                                  enforce_frame_size);
@@ -35,25 +33,25 @@ struct IsacFloat {
   static inline int16_t Create(instance_type** inst) {
     return WebRtcIsac_Create(inst);
   }
-  static inline int16_t DecodeInternal(instance_type* inst,
-                                       const uint8_t* encoded,
-                                       int16_t len,
-                                       int16_t* decoded,
-                                       int16_t* speech_type) {
+  static inline int DecodeInternal(instance_type* inst,
+                                   const uint8_t* encoded,
+                                   size_t len,
+                                   int16_t* decoded,
+                                   int16_t* speech_type) {
     return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
   }
-  static inline int16_t DecodePlc(instance_type* inst,
-                                  int16_t* decoded,
-                                  int16_t num_lost_frames) {
+  static inline size_t DecodePlc(instance_type* inst,
+                                 int16_t* decoded,
+                                 size_t num_lost_frames) {
     return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
   }
 
-  static inline int16_t DecoderInit(instance_type* inst) {
-    return WebRtcIsac_DecoderInit(inst);
+  static inline void DecoderInit(instance_type* inst) {
+    WebRtcIsac_DecoderInit(inst);
   }
-  static inline int16_t Encode(instance_type* inst,
-                               const int16_t* speech_in,
-                               uint8_t* encoded) {
+  static inline int Encode(instance_type* inst,
+                           const int16_t* speech_in,
+                           uint8_t* encoded) {
     return WebRtcIsac_Encode(inst, speech_in, encoded);
   }
   static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
@@ -66,6 +64,10 @@ struct IsacFloat {
   static inline int16_t Free(instance_type* inst) {
     return WebRtcIsac_Free(inst);
   }
+  static inline void GetBandwidthInfo(instance_type* inst,
+                                      IsacBandwidthInfo* bwinfo) {
+    WebRtcIsac_GetBandwidthInfo(inst, bwinfo);
+  }
   static inline int16_t GetErrorCode(instance_type* inst) {
     return WebRtcIsac_GetErrorCode(inst);
   }
@@ -73,7 +75,10 @@ struct IsacFloat {
   static inline int16_t GetNewFrameLen(instance_type* inst) {
     return WebRtcIsac_GetNewFrameLen(inst);
   }
-
+  static inline void SetBandwidthInfo(instance_type* inst,
+                                      const IsacBandwidthInfo* bwinfo) {
+    WebRtcIsac_SetBandwidthInfo(inst, bwinfo);
+  }
   static inline int16_t SetDecSampRate(instance_type* inst,
                                        uint16_t sample_rate_hz) {
     return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz);
@@ -82,9 +87,17 @@ struct IsacFloat {
                                        uint16_t sample_rate_hz) {
     return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
   }
+  static inline void SetEncSampRateInDecoder(instance_type* inst,
+                                             uint16_t sample_rate_hz) {
+    WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
+  }
+  static inline void SetInitialBweBottleneck(instance_type* inst,
+                                             int bottleneck_bits_per_second) {
+    WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
+  }
   static inline int16_t UpdateBwEstimate(instance_type* inst,
                                          const uint8_t* encoded,
-                                         int32_t packet_size,
+                                         size_t packet_size,
                                          uint16_t rtp_seq_number,
                                          uint32_t send_ts,
                                          uint32_t arr_ts) {
@@ -100,7 +113,5 @@ struct IsacFloat {
   }
 };
 
-typedef AudioEncoderDecoderIsacT<IsacFloat> AudioEncoderDecoderIsac;
-
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
index 8b93e65ba7..6991f90316 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
@@ -10,7 +10,7 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 struct WebRtcISACStruct;
@@ -79,7 +79,7 @@ TEST_F(IsacTest, IsacUpdateBWE) {
   WebRtcIsac_EncoderInit(isac_codec_, 0);
   WebRtcIsac_DecoderInit(isac_codec_);
 
-  int16_t encoded_bytes;
+  int encoded_bytes;
 
   // Test with call with a small packet (sync packet).
   EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_small_, 7, 1,
@@ -97,10 +97,12 @@ TEST_F(IsacTest, IsacUpdateBWE) {
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
   EXPECT_EQ(0, encoded_bytes);
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_GT(encoded_bytes, 0);
 
   // Call to update bandwidth estimator with real data.
   EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_,
-                                           encoded_bytes, 1, 12345, 56789));
+                                           static_cast<size_t>(encoded_bytes),
+                                           1, 12345, 56789));
 
   // Free memory.
   EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
index 5198ebfab8..60fc25b98b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
@@ -75,11 +75,11 @@ static const double kLpcCorrWindow[WINLEN] = {
   0.00155690, 0.00124918, 0.00094895, 0.00066112, 0.00039320, 0.00015881
 };
 
-double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order)
+double WebRtcIsac_LevDurb(double *a, double *k, double *r, size_t order)
 {
 
-  double  sum, alpha;
-  int     m, m_h, i;
+  double sum, alpha;
+  size_t m, m_h, i;
   alpha = 0; //warning -DH
   a[0] = 1.0;
   if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
index 866c76d8fd..8dfe383802 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
@@ -21,7 +21,7 @@
 #include "settings.h"
 #include "structs.h"
 
-double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order);
+double WebRtcIsac_LevDurb(double *a, double *k, double *r, size_t order);
 
 void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
                        double *oldEnergy, double *varscale);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
index f5d93564be..6fb02b378f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
@@ -61,11 +61,15 @@ void WebRtcIsac_PitchfilterPre_gains(double *indat,
 
 void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata);
 
-void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N);
+void WebRtcIsac_Highpass(const double *in,
+                         double *out,
+                         double *state,
+                         size_t N);
 
 void WebRtcIsac_DecimateAllpass(const double *in,
-                                double *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
-                                int N,                   /* number of input samples */
-                                double *out);            /* array of size N/2 */
+                                double *state_in,  /* array of size:
+                                                    *     2*ALLPASSSECTIONS+1 */
+                                size_t N,          /* number of input samples */
+                                double *out);      /* array of size N/2 */
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
index 5562c35ad4..31a80653fe 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
@@ -168,8 +168,6 @@ enum IsacSamplingRate {kIsacWideband = 16,  kIsacSuperWideband = 32};
 #define RCU_TRANSCODING_SCALE_UB                0.50f
 #define RCU_TRANSCODING_SCALE_UB_INVERSE        2.0f
 
-#define SIZE_RESAMPLER_STATE  6
-
 /* Define Error codes */
 /* 6000 General */
 #define ISAC_MEMORY_ALLOCATION_FAILED    6010
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
index eb85cf34da..a2cdca2c14 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -18,7 +18,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
 
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/settings.h"
 #include "webrtc/typedefs.h"
 
@@ -223,6 +224,8 @@ typedef struct {
   uint16_t                 numConsecLatePkts;
   float                        consecLatency;
   int16_t                  inWaitLatePkts;
+
+  IsacBandwidthInfo external_bw_info;
 } BwEstimatorstr;
 
 
@@ -428,6 +431,16 @@ typedef struct {
   uint8_t  stream[3];
 } transcode_obj;
 
+typedef struct {
+  // TODO(kwiberg): The size of these tables could be reduced by storing floats
+  // instead of doubles, and by making use of the identity cos(x) =
+  // sin(x+pi/2). They could also be made global constants that we fill in at
+  // compile time.
+  double costab1[FRAMESAMPLES_HALF];
+  double sintab1[FRAMESAMPLES_HALF];
+  double costab2[FRAMESAMPLES_QUARTER];
+  double sintab2[FRAMESAMPLES_QUARTER];
+} TransformTables;
 
 typedef struct {
   // lower-band codec instance
@@ -471,12 +484,12 @@ typedef struct {
   int16_t               maxRateBytesPer30Ms;
   // Maximum allowed payload-size, measured in Bytes.
   int16_t               maxPayloadSizeBytes;
-  /* The expected sampling rate of the input signal. Valid values are 16000,
-   * 32000 and 48000. This is not the operation sampling rate of the codec.
-   * Input signals at 48 kHz are resampled to 32 kHz, then encoded. */
+  /* The expected sampling rate of the input signal. Valid values are 16000
+   * and 32000. This is not the operation sampling rate of the codec. */
   uint16_t in_sample_rate_hz;
-  /* State for the input-resampler. It is only used for 48 kHz input signals. */
-  int16_t state_in_resampler[SIZE_RESAMPLER_STATE];
+
+  // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
+  TransformTables transform_tables;
 } ISACMainStruct;
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
index ea6b579093..8992897f45 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
@@ -14,41 +14,33 @@
 #include "os_specific_inline.h"
 #include <math.h>
 
-static double costab1[FRAMESAMPLES_HALF];
-static double sintab1[FRAMESAMPLES_HALF];
-static double costab2[FRAMESAMPLES_QUARTER];
-static double sintab2[FRAMESAMPLES_QUARTER];
-
-void WebRtcIsac_InitTransform()
-{
+void WebRtcIsac_InitTransform(TransformTables* tables) {
   int k;
   double fact, phase;
 
   fact = PI / (FRAMESAMPLES_HALF);
   phase = 0.0;
   for (k = 0; k < FRAMESAMPLES_HALF; k++) {
-    costab1[k] = cos(phase);
-    sintab1[k] = sin(phase);
+    tables->costab1[k] = cos(phase);
+    tables->sintab1[k] = sin(phase);
     phase += fact;
   }
 
   fact = PI * ((double) (FRAMESAMPLES_HALF - 1)) / ((double) FRAMESAMPLES_HALF);
   phase = 0.5 * fact;
   for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
-    costab2[k] = cos(phase);
-    sintab2[k] = sin(phase);
+    tables->costab2[k] = cos(phase);
+    tables->sintab2[k] = sin(phase);
     phase += fact;
   }
 }
 
-
-void WebRtcIsac_Time2Spec(double *inre1,
-                         double *inre2,
-                         int16_t *outreQ7,
-                         int16_t *outimQ7,
-                         FFTstr *fftstr_obj)
-{
-
+void WebRtcIsac_Time2Spec(const TransformTables* tables,
+                          double* inre1,
+                          double* inre2,
+                          int16_t* outreQ7,
+                          int16_t* outimQ7,
+                          FFTstr* fftstr_obj) {
   int k;
   int dims[1];
   double tmp1r, tmp1i, xr, xi, yr, yi, fact;
@@ -61,8 +53,8 @@ void WebRtcIsac_Time2Spec(double *inre1,
   /* Multiply with complex exponentials and combine into one complex vector */
   fact = 0.5 / sqrt(FRAMESAMPLES_HALF);
   for (k = 0; k < FRAMESAMPLES_HALF; k++) {
-    tmp1r = costab1[k];
-    tmp1i = sintab1[k];
+    tmp1r = tables->costab1[k];
+    tmp1i = tables->sintab1[k];
     tmpre[k] = (inre1[k] * tmp1r + inre2[k] * tmp1i) * fact;
     tmpim[k] = (inre2[k] * tmp1r - inre1[k] * tmp1i) * fact;
   }
@@ -78,8 +70,8 @@ void WebRtcIsac_Time2Spec(double *inre1,
     xi = tmpim[k] - tmpim[FRAMESAMPLES_HALF - 1 - k];
     yr = tmpim[k] + tmpim[FRAMESAMPLES_HALF - 1 - k];
 
-    tmp1r = costab2[k];
-    tmp1i = sintab2[k];
+    tmp1r = tables->costab2[k];
+    tmp1i = tables->sintab2[k];
     outreQ7[k] = (int16_t)WebRtcIsac_lrint((xr * tmp1r - xi * tmp1i) * 128.0);
     outimQ7[k] = (int16_t)WebRtcIsac_lrint((xr * tmp1i + xi * tmp1r) * 128.0);
     outreQ7[FRAMESAMPLES_HALF - 1 - k] = (int16_t)WebRtcIsac_lrint((-yr * tmp1i - yi * tmp1r) * 128.0);
@@ -87,10 +79,12 @@ void WebRtcIsac_Time2Spec(double *inre1,
   }
 }
 
-
-void WebRtcIsac_Spec2time(double *inre, double *inim, double *outre1, double *outre2, FFTstr *fftstr_obj)
-{
-
+void WebRtcIsac_Spec2time(const TransformTables* tables,
+                          double* inre,
+                          double* inim,
+                          double* outre1,
+                          double* outre2,
+                          FFTstr* fftstr_obj) {
   int k;
   double tmp1r, tmp1i, xr, xi, yr, yi, fact;
 
@@ -100,8 +94,8 @@ void WebRtcIsac_Spec2time(double *inre, double *inim, double *outre1, double *ou
 
   for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
     /* Move zero in time to beginning of frames */
-    tmp1r = costab2[k];
-    tmp1i = sintab2[k];
+    tmp1r = tables->costab2[k];
+    tmp1i = tables->sintab2[k];
     xr = inre[k] * tmp1r + inim[k] * tmp1i;
     xi = inim[k] * tmp1r - inre[k] * tmp1i;
     yr = -inim[FRAMESAMPLES_HALF - 1 - k] * tmp1r - inre[FRAMESAMPLES_HALF - 1 - k] * tmp1i;
@@ -122,8 +116,8 @@ void WebRtcIsac_Spec2time(double *inre, double *inim, double *outre1, double *ou
   /* Demodulate and separate */
   fact = sqrt(FRAMESAMPLES_HALF);
   for (k = 0; k < FRAMESAMPLES_HALF; k++) {
-    tmp1r = costab1[k];
-    tmp1i = sintab1[k];
+    tmp1r = tables->costab1[k];
+    tmp1i = tables->sintab1[k];
     xr = (outre1[k] * tmp1r - outre2[k] * tmp1i) * fact;
     outre2[k] = (outre2[k] * tmp1r + outre1[k] * tmp1i) * fact;
     outre1[k] = xr;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index ffdcc0c1a7..4cef8f7b3b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -21,1075 +21,927 @@
 /* include API */
 #include "isac.h"
 #include "utility.h"
+#include "webrtc/base/format_macros.h"
 
 /* Defines */
-#define SEED_FILE "randseed.txt"  /* Used when running decoder on garbage data  */
-#define MAX_FRAMESAMPLES     960  /* max number of samples per frame
-                                      (= 60 ms frame & 16 kHz) or
-                                      (= 30 ms frame & 32 kHz)                  */
-#define FRAMESAMPLES_10ms	 160   /* number of samples per 10ms frame          */
+#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
+#define MAX_FRAMESAMPLES 960     /* max number of samples per frame
+                                    (= 60 ms frame & 16 kHz) or
+                                    (= 30 ms frame & 32 kHz) */
+#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
 #define SWBFRAMESAMPLES_10ms 320
-//#define FS		        	16000 /* sampling frequency (Hz) */
+//#define FS 16000 /* sampling frequency (Hz) */
 
 #ifdef WIN32
-#define CLOCKS_PER_SEC      1000  /* Runtime statistics */
+#define CLOCKS_PER_SEC 1000 /* Runtime statistics */
 #endif
 
-
-
-
 using namespace std;
 
-int main(int argc, char* argv[])
-{
+int main(int argc, char* argv[]) {
+  char inname[100], outname[100], bottleneck_file[100], vadfile[100];
+  FILE* inp, *outp, * f_bn = NULL, * vadp = NULL, *bandwidthp;
+  int framecnt, endfile;
 
-    char inname[100], outname[100], bottleneck_file[100], vadfile[100];
-	FILE *inp, *outp, *f_bn = NULL, *vadp = NULL, *bandwidthp;
-	int framecnt, endfile;
+  size_t i;
+  int errtype, VADusage = 0, packetLossPercent = 0;
+  int16_t CodingMode;
+  int32_t bottleneck = 0;
+  int framesize = 30; /* ms */
+  int cur_framesmpls, err;
 
-	int i, errtype, VADusage = 0, packetLossPercent = 0;
-	int16_t CodingMode;
-	int32_t bottleneck = 0;
-	int16_t framesize = 30;           /* ms */
-	int cur_framesmpls, err;
+  /* Runtime statistics */
+  double starttime, runtime, length_file;
 
-	/* Runtime statistics */
-	double starttime, runtime, length_file;
+  size_t stream_len = 0;
+  int declen = 0, declenTC = 0;
+  bool lostFrame = false;
 
-	int16_t stream_len = 0;
-	int16_t declen = 0, lostFrame = 0, declenTC = 0;
+  int16_t shortdata[SWBFRAMESAMPLES_10ms];
+  int16_t vaddata[SWBFRAMESAMPLES_10ms * 3];
+  int16_t decoded[MAX_FRAMESAMPLES << 1];
+  int16_t decodedTC[MAX_FRAMESAMPLES << 1];
+  uint16_t streamdata[500];
+  int16_t speechType[1];
+  int16_t rateBPS = 0;
+  int16_t fixedFL = 0;
+  int16_t payloadSize = 0;
+  int32_t payloadRate = 0;
+  int setControlBWE = 0;
+  short FL, testNum;
+  char version_number[20];
+  FILE* plFile;
+  int32_t sendBN;
 
-	int16_t shortdata[SWBFRAMESAMPLES_10ms];
-	int16_t vaddata[SWBFRAMESAMPLES_10ms*3];
-	int16_t decoded[MAX_FRAMESAMPLES << 1];
-	int16_t decodedTC[MAX_FRAMESAMPLES << 1];
-	uint16_t streamdata[500];
-	int16_t	speechType[1];
-    int16_t rateBPS = 0;
-    int16_t fixedFL = 0;
-    int16_t payloadSize = 0;
-    int32_t payloadRate = 0;
-    int setControlBWE = 0;
-    short FL, testNum;
-	char version_number[20];
-    FILE  *plFile;
-    int32_t sendBN;
+#if !defined(NDEBUG)
+  FILE* fy;
+  double kbps;
+#endif
+  size_t totalbits = 0;
+  int totalsmpls = 0;
 
-#ifdef _DEBUG
-	FILE *fy;
-	double kbps;
-#endif /* _DEBUG */
-	int totalbits =0;
-	int totalsmpls =0;
+  /* If use GNS file */
+  FILE* fp_gns = NULL;
+  char gns_file[100];
+  size_t maxStreamLen30 = 0;
+  size_t maxStreamLen60 = 0;
+  short sampFreqKHz = 32;
+  short samplesIn10Ms;
+  short useAssign = 0;
+  // FILE logFile;
+  bool doTransCoding = false;
+  int32_t rateTransCoding = 0;
+  uint8_t streamDataTransCoding[1200];
+  size_t streamLenTransCoding = 0;
+  FILE* transCodingFile = NULL;
+  FILE* transcodingBitstream = NULL;
+  size_t numTransCodingBytes = 0;
 
-    /* If use GNS file */
-    FILE *fp_gns = NULL;
-	char gns_file[100];
-    short maxStreamLen30 = 0;
-    short maxStreamLen60 = 0;
-    short sampFreqKHz = 32;
-    short samplesIn10Ms;
-    short useAssign = 0;
-    //FILE logFile;
-    bool doTransCoding = false;
-    int32_t rateTransCoding = 0;
-    uint8_t streamDataTransCoding[1200];
-    int16_t streamLenTransCoding = 0;
-    FILE* transCodingFile = NULL;
-    FILE* transcodingBitstream = NULL;
-    uint32_t numTransCodingBytes = 0;
+  /* only one structure used for ISAC encoder */
+  ISACStruct* ISAC_main_inst = NULL;
+  ISACStruct* decoderTransCoding = NULL;
 
-	/* only one structure used for ISAC encoder */
-	ISACStruct* ISAC_main_inst = NULL;
-    ISACStruct* decoderTransCoding = NULL;
+  BottleNeckModel BN_data;
 
-	BottleNeckModel       BN_data;
+#if !defined(NDEBUG)
+  fy = fopen("bit_rate.dat", "w");
+  fclose(fy);
+  fy = fopen("bytes_frames.dat", "w");
+  fclose(fy);
+#endif
 
-#ifdef _DEBUG
-	fy = fopen("bit_rate.dat", "w");
-	fclose(fy);
-	fy = fopen("bytes_frames.dat", "w");
-	fclose(fy);
-#endif /* _DEBUG */
+  /* Handling wrong input arguments in the command line */
+  if ((argc < 3) || (argc > 17)) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
 
-	/* Handling wrong input arguments in the command line */
-	if((argc<3) || (argc>17))  {
-		printf("\n\nWrong number of arguments or flag values.\n\n");
-
-        printf("\n");
-        WebRtcIsac_version(version_number);
-        printf("iSAC-swb version %s \n\n", version_number);
-
-        printf("Usage:\n\n");
-        printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
-        printf("with:\n");
-        printf("[-FS num]       :   sampling frequency in kHz, valid values are 16 & 32,\n");
-        printf("                    with 16 as default.\n");
-        printf("[-I]            :   if -I option is specified, the coder will use\n");
-        printf("                    an instantaneous Bottleneck value. If not, it\n");
-        printf("                    will be an adaptive Bottleneck value.\n\n");
-        printf("[-assign]       :   Use Assign API.\n");
-        printf("[-B num]        :   the value of the bottleneck provided either\n");
-        printf("                    as a fixed value in bits/sec (e.g. 25000) or\n");
-        printf("                    read from a file (e.g. bottleneck.txt)\n\n");
-        printf("[-INITRATE num] :   Set a new value for initial rate. Note! Only used in \n");
-        printf("                    adaptive mode.\n\n");
-        printf("[-FL num]       :   Set (initial) frame length in msec. Valid length are \n");
-        printf("                    30 and 60 msec.\n\n");
-        printf("[-FIXED_FL]     :   Frame length will be fixed to initial value.\n\n");
-        printf("[-MAX num]      :   Set the limit for the payload size of iSAC in bytes. \n");
-        printf("                    Minimum 100 maximum 400.\n\n");
-        printf("[-MAXRATE num]  :   Set the maxrate for iSAC in bits per second. \n");
-        printf("                    Minimum 32000, maximum 53400.\n\n");
-        printf("[-F num]        :   if -F option is specified, the test function\n");
-        printf("                    will run the iSAC API fault scenario specified by the\n");
-        printf("                    supplied number.\n");
-        printf("                    F 1 - Call encoder prior to init encoder call\n");
-        printf("                    F 2 - Call decoder prior to init decoder call\n");
-        printf("                    F 3 - Call decoder prior to encoder call\n");
-        printf("                    F 4 - Call decoder with a too short coded sequence\n");
-        printf("                    F 5 - Call decoder with a too long coded sequence\n");
-        printf("                    F 6 - Call decoder with random bit stream\n");
-        printf("                    F 7 - Call init encoder/decoder at random during a call\n");
-        printf("                    F 8 - Call encoder/decoder without having allocated memory \n");
-        printf("                          for encoder/decoder instance\n");
-        printf("                    F 9 - Call decodeB without calling decodeA\n");
-        printf("                    F 10 - Call decodeB with garbage data\n");
-        printf("[-PL num]       :   if -PL option is specified \n");
-        printf("[-T rate file]  :   test trans-coding with target bottleneck 'rate' bits/sec\n");
-        printf("                    the output file is written to 'file'\n");
-        printf("[-LOOP num]     :   number of times to repeat coding the input file for stress testing\n");
-        //printf("[-CE num]       :   Test of APIs used by Conference Engine.\n");
-        //printf("                    CE 1 - getNewBitstream, getBWE \n");
-        //printf("                    (CE 2 - RESERVED for transcoding)\n");
-        //printf("                    CE 3 - getSendBWE, setSendBWE.  \n\n");
-        //printf("-L filename     :   write the logging info into file (appending)\n");
-        printf("infile          :   Normal speech input file\n\n");
-        printf("outfile         :   Speech output file\n\n");
-    	exit(0);
-	}
-
-    /* Print version number */
-    printf("-------------------------------------------------\n");
+    printf("\n");
     WebRtcIsac_version(version_number);
-    printf("iSAC version %s \n\n", version_number);
+    printf("iSAC-swb version %s \n\n", version_number);
 
-    /* Loop over all command line arguments */
-	CodingMode = 0;
-	testNum = 0;
-    useAssign = 0;
-    //logFile = NULL;
-    char transCodingFileName[500];
-    int16_t totFileLoop = 0;
-    int16_t numFileLoop = 0;
-	for (i = 1; i < argc-2;i++)
-    {
-        if(!strcmp("-LOOP", argv[i]))
-        {
-            i++;
-            totFileLoop = (int16_t)atol(argv[i]);
-            if(totFileLoop <= 0)
-            {
-                fprintf(stderr, "Invalid number of runs for the given input file, %d.", totFileLoop);
-                exit(0);
-            }
-        }
+    printf("Usage:\n\n");
+    printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
+    printf("with:\n");
+    printf("[-FS num]       : sampling frequency in kHz, valid values are\n");
+    printf("                  16 & 32, with 16 as default.\n");
+    printf("[-I]            : if -I option is specified, the coder will use\n");
+    printf("                  an instantaneous Bottleneck value. If not, it\n");
+    printf("                  will be an adaptive Bottleneck value.\n");
+    printf("[-assign]       : Use Assign API.\n");
+    printf("[-B num]        : the value of the bottleneck provided either\n");
+    printf("                  as a fixed value in bits/sec (e.g. 25000) or\n");
+    printf("                  read from a file (e.g. bottleneck.txt)\n");
+    printf("[-INITRATE num] : Set a new value for initial rate. Note! Only\n");
+    printf("                  used in adaptive mode.\n");
+    printf("[-FL num]       : Set (initial) frame length in msec. Valid\n");
+    printf("                  lengths are 30 and 60 msec.\n");
+    printf("[-FIXED_FL]     : Frame length will be fixed to initial value.\n");
+    printf("[-MAX num]      : Set the limit for the payload size of iSAC\n");
+    printf("                  in bytes. Minimum 100 maximum 400.\n");
+    printf("[-MAXRATE num]  : Set the maxrate for iSAC in bits per second.\n");
+    printf("                  Minimum 32000, maximum 53400.\n");
+    printf("[-F num]        : if -F option is specified, the test function\n");
+    printf("                  will run the iSAC API fault scenario\n");
+    printf("                  specified by the supplied number.\n");
+    printf("                  F 1 - Call encoder prior to init encoder call\n");
+    printf("                  F 2 - Call decoder prior to init decoder call\n");
+    printf("                  F 3 - Call decoder prior to encoder call\n");
+    printf("                  F 4 - Call decoder with a too short coded\n");
+    printf("                        sequence\n");
+    printf("                  F 5 - Call decoder with a too long coded\n");
+    printf("                        sequence\n");
+    printf("                  F 6 - Call decoder with random bit stream\n");
+    printf("                  F 7 - Call init encoder/decoder at random\n");
+    printf("                        during a call\n");
+    printf("                  F 8 - Call encoder/decoder without having\n");
+    printf("                        allocated memory for encoder/decoder\n");
+    printf("                        instance\n");
+    printf("                  F 9 - Call decodeB without calling decodeA\n");
+    printf("                  F 10 - Call decodeB with garbage data\n");
+    printf("[-PL num]       : if -PL option is specified \n");
+    printf("[-T rate file]  : test trans-coding with target bottleneck\n");
+    printf("                  'rate' bits/sec\n");
+    printf("                  the output file is written to 'file'\n");
+    printf("[-LOOP num]     : number of times to repeat coding the input\n");
+    printf("                  file for stress testing\n");
+    // printf("[-CE num]       : Test of APIs used by Conference Engine.\n");
+    // printf("                  CE 1 - getNewBitstream, getBWE \n");
+    // printf("                  (CE 2 - RESERVED for transcoding)\n");
+    // printf("                  CE 3 - getSendBWE, setSendBWE.  \n");
+    // printf("-L filename     : write the logging info into file
+    // (appending)\n");
+    printf("infile          :   Normal speech input file\n");
+    printf("outfile         :   Speech output file\n");
+    exit(0);
+  }
 
-        if(!strcmp("-T", argv[i]))
-        {
-            doTransCoding = true;
-            i++;
-            rateTransCoding = atoi(argv[i]);
-            i++;
-            strcpy(transCodingFileName, argv[i]);
-        }
+  /* Print version number */
+  printf("-------------------------------------------------\n");
+  WebRtcIsac_version(version_number);
+  printf("iSAC version %s \n\n", version_number);
 
-        /*Should we use assign API*/
-        if(!strcmp("-assign", argv[i]))
-        {
-            useAssign = 1;
-        }
-
-        /* Set Sampling Rate */
-        if(!strcmp("-FS", argv[i]))
-        {
-            i++;
-            sampFreqKHz = atoi(argv[i]);
-        }
-
-        /* Instantaneous mode */
-		if(!strcmp ("-I", argv[i]))
-        {
-			printf("Instantaneous BottleNeck\n");
-			CodingMode = 1;
-		}
-
-        /* Set (initial) bottleneck value */
-        if(!strcmp ("-INITRATE", argv[i]))	{
-			rateBPS = atoi(argv[i + 1]);
-            setControlBWE = 1;
-            if((rateBPS < 10000) || (rateBPS > 32000))
-            {
-				printf("\n%d is not a initial rate. Valid values are in the range 10000 to 32000.\n", rateBPS);
-				exit(0);
-            }
-			printf("New initial rate: %d\n", rateBPS);
-			i++;
-		}
-
-        /* Set (initial) framelength */
-        if(!strcmp ("-FL", argv[i]))	{
-			framesize = atoi(argv[i + 1]);
-            if((framesize != 30) && (framesize != 60))
-            {
-				printf("\n%d is not a valid frame length. Valid length are 30 and 60 msec.\n", framesize);
-				exit(0);
-            }
-            setControlBWE = 1;
-			printf("Frame Length: %d\n", framesize);
-			i++;
-		}
-
-        /* Fixed frame length */
-        if(!strcmp ("-FIXED_FL", argv[i]))
-        {
-			fixedFL = 1;
-            setControlBWE = 1;
-			printf("Fixed Frame Length\n");
-		}
-
-        /* Set maximum allowed payload size in bytes */
-        if(!strcmp ("-MAX", argv[i]))	{
-			payloadSize = atoi(argv[i + 1]);
-            printf("Maximum Payload Size: %d\n", payloadSize);
-			i++;
-		}
-
-        /* Set maximum rate in bytes */
-        if(!strcmp ("-MAXRATE", argv[i]))	{
-			payloadRate = atoi(argv[i + 1]);
-            printf("Maximum Rate in kbps: %d\n", payloadRate);
-			i++;
-		}
-
-        /* Test of fault scenarious */
-        if(!strcmp ("-F", argv[i]))
-        {
-			testNum = atoi(argv[i + 1]);
-			printf("Fault test: %d\n", testNum);
-			if(testNum < 1 || testNum > 10)
-            {
-				printf("\n%d is not a valid Fault Scenario number. Valid Fault Scenarios are numbered 1-10.\n", testNum);
-				exit(0);
-			}
-			i++;
-		}
-
-        /* Packet loss test */
-		if(!strcmp ("-PL", argv[i]))
-        {
-			if( isdigit( *argv[i+1] ) )
-            {
-				packetLossPercent = atoi( argv[i+1] );
-				if( (packetLossPercent < 0) | (packetLossPercent > 100) )
-                {
-					printf( "\nInvalid packet loss perentage \n" );
-					exit( 0 );
-				}
-                if( packetLossPercent > 0 )
-                {
-					printf( "Simulating %d %% of independent packet loss\n", packetLossPercent );
-                }
-                else
-                {
-					printf( "\nNo Packet Loss Is Simulated \n" );
-                }
-            }
-            else
-            {
-				plFile = fopen( argv[i+1], "rb" );
-				if( plFile == NULL )
-                {
-					printf( "\n couldn't open the frameloss file: %s\n", argv[i+1] );
-					exit( 0 );
-				}
-				printf( "Simulating packet loss through the given channel file: %s\n", argv[i+1] );
-			}
-			i++;
-		}
-
-        /* Random packetlosses */
-		if(!strcmp ("-rnd", argv[i]))
-        {
-			srand((unsigned int)time(NULL) );
-			printf( "Random pattern in lossed packets \n" );
-		}
-
-        /* Use gns file */
-		if(!strcmp ("-G", argv[i]))
-        {
-			sscanf(argv[i + 1], "%s", gns_file);
-			fp_gns = fopen(gns_file, "rb");
-			if(fp_gns  == NULL)
-            {
-				printf("Cannot read file %s.\n", gns_file);
-				exit(0);
-			}
-			i++;
-		}
-
-
-        // make it with '-B'
-        /* Get Bottleneck value */
-        if(!strcmp("-B", argv[i]))
-        {
-            i++;
-            bottleneck = atoi(argv[i]);
-            if(bottleneck == 0)
-            {
-                sscanf(argv[i], "%s", bottleneck_file);
-                f_bn = fopen(bottleneck_file, "rb");
-                if(f_bn  == NULL)
-                {
-                    printf("Error No value provided for BottleNeck and cannot read file %s.\n", bottleneck_file);
-                    exit(0);
-                }
-                else
-                {
-                    printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
-                    if(fscanf(f_bn, "%d", &bottleneck) == EOF)
-                    {
-                        /* Set pointer to beginning of file */
-                        fseek(f_bn, 0L, SEEK_SET);
-                        if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
-                            exit(0);
-                        }
-                    }
-
-                    /*	Bottleneck is a cosine function
-                    *	Matlab code for writing the bottleneck file:
-                    *	BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
-                    *	fid = fopen('bottleneck.txt', 'wb');
-                    *	fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
-                    */
-                }
-            }
-            else
-            {
-                printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
-            }
-        }
-        /* Run Conference Engine APIs */
-        //     Do not test it in the first release
-        //
-        //     if(!strcmp ("-CE", argv[i]))
-        //     {
-        //         testCE = atoi(argv[i + 1]);
-        //         if(testCE==1)
-        //         {
-        //             i++;
-        //             scale = (float)atof( argv[i+1] );
-        //         }
-        //         else if(testCE == 2)
-        //         {
-        //             printf("\nCE-test 2 (transcoding) not implemented.\n");
-        //             exit(0);
-        //         }
-        //         else if(testCE < 1 || testCE > 3)
-        //         {
-        //             printf("\n%d is not a valid CE-test number. Valid CE tests are 1-3.\n", testCE);
-        //             exit(0);
-        //         }
-        //         printf("CE-test number: %d\n", testCE);
-        //         i++;
-        //     }
+  /* Loop over all command line arguments */
+  CodingMode = 0;
+  testNum = 0;
+  useAssign = 0;
+  // logFile = NULL;
+  char transCodingFileName[500];
+  int16_t totFileLoop = 0;
+  int16_t numFileLoop = 0;
+  for (i = 1; i + 2 < static_cast<size_t>(argc); i++) {
+    if (!strcmp("-LOOP", argv[i])) {
+      i++;
+      totFileLoop = (int16_t)atol(argv[i]);
+      if (totFileLoop <= 0) {
+        fprintf(stderr, "Invalid number of runs for the given input file, %d.",
+                totFileLoop);
+        exit(0);
+      }
     }
 
-	if(CodingMode == 0)
-	{
-		printf("\nAdaptive BottleNeck\n");
-	}
+    if (!strcmp("-T", argv[i])) {
+      doTransCoding = true;
+      i++;
+      rateTransCoding = atoi(argv[i]);
+      i++;
+      strcpy(transCodingFileName, argv[i]);
+    }
 
-    switch(sampFreqKHz)
-    {
-    case 16:
-        {
-            printf("iSAC Wideband.\n");
-            samplesIn10Ms = FRAMESAMPLES_10ms;
-            break;
+    /*Should we use assign API*/
+    if (!strcmp("-assign", argv[i])) {
+      useAssign = 1;
+    }
+
+    /* Set Sampling Rate */
+    if (!strcmp("-FS", argv[i])) {
+      i++;
+      sampFreqKHz = atoi(argv[i]);
+    }
+
+    /* Instantaneous mode */
+    if (!strcmp("-I", argv[i])) {
+      printf("Instantaneous BottleNeck\n");
+      CodingMode = 1;
+    }
+
+    /* Set (initial) bottleneck value */
+    if (!strcmp("-INITRATE", argv[i])) {
+      rateBPS = atoi(argv[i + 1]);
+      setControlBWE = 1;
+      if ((rateBPS < 10000) || (rateBPS > 32000)) {
+        printf("\n%d is not a initial rate. Valid values are in the range "
+               "10000 to 32000.\n", rateBPS);
+        exit(0);
+      }
+      printf("New initial rate: %d\n", rateBPS);
+      i++;
+    }
+
+    /* Set (initial) framelength */
+    if (!strcmp("-FL", argv[i])) {
+      framesize = atoi(argv[i + 1]);
+      if ((framesize != 30) && (framesize != 60)) {
+        printf("\n%d is not a valid frame length. Valid length are 30 and 60 "
+               "msec.\n", framesize);
+        exit(0);
+      }
+      setControlBWE = 1;
+      printf("Frame Length: %d\n", framesize);
+      i++;
+    }
+
+    /* Fixed frame length */
+    if (!strcmp("-FIXED_FL", argv[i])) {
+      fixedFL = 1;
+      setControlBWE = 1;
+      printf("Fixed Frame Length\n");
+    }
+
+    /* Set maximum allowed payload size in bytes */
+    if (!strcmp("-MAX", argv[i])) {
+      payloadSize = atoi(argv[i + 1]);
+      printf("Maximum Payload Size: %d\n", payloadSize);
+      i++;
+    }
+
+    /* Set maximum rate in bytes */
+    if (!strcmp("-MAXRATE", argv[i])) {
+      payloadRate = atoi(argv[i + 1]);
+      printf("Maximum Rate in kbps: %d\n", payloadRate);
+      i++;
+    }
+
+    /* Test of fault scenarious */
+    if (!strcmp("-F", argv[i])) {
+      testNum = atoi(argv[i + 1]);
+      printf("Fault test: %d\n", testNum);
+      if (testNum < 1 || testNum > 10) {
+        printf("\n%d is not a valid Fault Scenario number. Valid Fault "
+               "Scenarios are numbered 1-10.\n", testNum);
+        exit(0);
+      }
+      i++;
+    }
+
+    /* Packet loss test */
+    if (!strcmp("-PL", argv[i])) {
+      if (isdigit(*argv[i + 1])) {
+        packetLossPercent = atoi(argv[i + 1]);
+        if ((packetLossPercent < 0) | (packetLossPercent > 100)) {
+          printf("\nInvalid packet loss perentage \n");
+          exit(0);
         }
-    case 32:
-        {
-            printf("iSAC Supper-Wideband.\n");
-            samplesIn10Ms = SWBFRAMESAMPLES_10ms;
-            break;
+        if (packetLossPercent > 0) {
+          printf("Simulating %d %% of independent packet loss\n",
+                 packetLossPercent);
+        } else {
+          printf("\nNo Packet Loss Is Simulated \n");
         }
+      } else {
+        plFile = fopen(argv[i + 1], "rb");
+        if (plFile == NULL) {
+          printf("\n couldn't open the frameloss file: %s\n", argv[i + 1]);
+          exit(0);
+        }
+        printf("Simulating packet loss through the given channel file: %s\n",
+               argv[i + 1]);
+      }
+      i++;
+    }
+
+    /* Random packetlosses */
+    if (!strcmp("-rnd", argv[i])) {
+      srand((unsigned int)time(NULL));
+      printf("Random pattern in lossed packets \n");
+    }
+
+    /* Use gns file */
+    if (!strcmp("-G", argv[i])) {
+      sscanf(argv[i + 1], "%s", gns_file);
+      fp_gns = fopen(gns_file, "rb");
+      if (fp_gns == NULL) {
+        printf("Cannot read file %s.\n", gns_file);
+        exit(0);
+      }
+      i++;
+    }
+
+    // make it with '-B'
+    /* Get Bottleneck value */
+    if (!strcmp("-B", argv[i])) {
+      i++;
+      bottleneck = atoi(argv[i]);
+      if (bottleneck == 0) {
+        sscanf(argv[i], "%s", bottleneck_file);
+        f_bn = fopen(bottleneck_file, "rb");
+        if (f_bn == NULL) {
+          printf("Error No value provided for BottleNeck and cannot read file "
+                 "%s.\n", bottleneck_file);
+          exit(0);
+        } else {
+          printf("reading bottleneck rates from file %s\n\n", bottleneck_file);
+          if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+            /* Set pointer to beginning of file */
+            fseek(f_bn, 0L, SEEK_SET);
+            if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+              exit(0);
+            }
+          }
+
+          /* Bottleneck is a cosine function
+           * Matlab code for writing the bottleneck file:
+           * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
+           * fid = fopen('bottleneck.txt', 'wb');
+           * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
+           */
+        }
+      } else {
+        printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
+      }
+    }
+    /* Run Conference Engine APIs */
+    //     Do not test it in the first release
+    //
+    //     if(!strcmp ("-CE", argv[i]))
+    //     {
+    //         testCE = atoi(argv[i + 1]);
+    //         if(testCE==1)
+    //         {
+    //             i++;
+    //             scale = (float)atof( argv[i+1] );
+    //         }
+    //         else if(testCE == 2)
+    //         {
+    //             printf("\nCE-test 2 (transcoding) not implemented.\n");
+    //             exit(0);
+    //         }
+    //         else if(testCE < 1 || testCE > 3)
+    //         {
+    //             printf("\n%d is not a valid CE-test number. Valid CE tests
+    //             are 1-3.\n", testCE);
+    //             exit(0);
+    //         }
+    //         printf("CE-test number: %d\n", testCE);
+    //         i++;
+    //     }
+  }
+
+  if (CodingMode == 0) {
+    printf("\nAdaptive BottleNeck\n");
+  }
+
+  switch (sampFreqKHz) {
+    case 16: {
+      printf("iSAC Wideband.\n");
+      samplesIn10Ms = FRAMESAMPLES_10ms;
+      break;
+    }
+    case 32: {
+      printf("iSAC Supper-Wideband.\n");
+      samplesIn10Ms = SWBFRAMESAMPLES_10ms;
+      break;
+    }
     default:
-            printf("Unsupported sampling frequency %d kHz", sampFreqKHz);
-            exit(0);
+      printf("Unsupported sampling frequency %d kHz", sampFreqKHz);
+      exit(0);
+  }
+
+  /* Get Input and Output files */
+  sscanf(argv[argc - 2], "%s", inname);
+  sscanf(argv[argc - 1], "%s", outname);
+  printf("\nInput file: %s\n", inname);
+  printf("Output file: %s\n\n", outname);
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  Error iSAC Cannot read file %s.\n", inname);
+    cout << flush;
+    exit(1);
+  }
+
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  Error iSAC Cannot write file %s.\n", outname);
+    cout << flush;
+    getc(stdin);
+    exit(1);
+  }
+  if (VADusage) {
+    if ((vadp = fopen(vadfile, "rb")) == NULL) {
+      printf("  Error iSAC Cannot read file %s.\n", vadfile);
+      cout << flush;
+      exit(1);
     }
+  }
 
+  if ((bandwidthp = fopen("bwe.pcm", "wb")) == NULL) {
+    printf("  Error iSAC Cannot read file %s.\n", "bwe.pcm");
+    cout << flush;
+    exit(1);
+  }
 
+  starttime = clock() / (double)CLOCKS_PER_SEC; /* Runtime statistics */
 
+  /* Initialize the ISAC and BN structs */
+  if (testNum != 8) {
+    if (!useAssign) {
+      err = WebRtcIsac_Create(&ISAC_main_inst);
+      WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
+      WebRtcIsac_SetDecSampRate(ISAC_main_inst,
+                                sampFreqKHz >= 32 ? 32000 : 16000);
+    } else {
+      /* Test the Assign functions */
+      int sss;
+      void* ppp;
+      err = WebRtcIsac_AssignSize(&sss);
+      ppp = malloc(sss);
+      err = WebRtcIsac_Assign(&ISAC_main_inst, ppp);
+      WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
+      WebRtcIsac_SetDecSampRate(ISAC_main_inst,
+                                sampFreqKHz >= 32 ? 32000 : 16000);
+    }
+    /* Error check */
+    if (err < 0) {
+      printf("\n\n Error in create.\n\n");
+      cout << flush;
+      exit(EXIT_FAILURE);
+    }
+  }
+  BN_data.arrival_time = 0;
+  BN_data.sample_count = 0;
+  BN_data.rtp_number = 0;
 
-	/* Get Input and Output files */
-	sscanf(argv[argc-2], "%s", inname);
-	sscanf(argv[argc-1], "%s", outname);
-    printf("\nInput file: %s\n", inname);
-    printf("Output file: %s\n\n", outname);
-	if((inp = fopen(inname,"rb")) == NULL)
-    {
-		printf("  Error iSAC Cannot read file %s.\n", inname);
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = 0;
+
+  if (doTransCoding) {
+    WebRtcIsac_Create(&decoderTransCoding);
+    WebRtcIsac_SetEncSampRate(decoderTransCoding, sampFreqKHz * 1000);
+    WebRtcIsac_SetDecSampRate(decoderTransCoding,
+                              sampFreqKHz >= 32 ? 32000 : 16000);
+    WebRtcIsac_DecoderInit(decoderTransCoding);
+    transCodingFile = fopen(transCodingFileName, "wb");
+    if (transCodingFile == NULL) {
+      printf("Could not open %s to output trans-coding.\n",
+             transCodingFileName);
+      exit(0);
+    }
+    strcat(transCodingFileName, ".bit");
+    transcodingBitstream = fopen(transCodingFileName, "wb");
+    if (transcodingBitstream == NULL) {
+      printf("Could not open %s to write the bit-stream of transcoder.\n",
+             transCodingFileName);
+      exit(0);
+    }
+  }
+
+  if (testNum != 1) {
+    if (WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode) < 0) {
+      printf("Error could not initialize the encoder \n");
+      cout << flush;
+      return 0;
+    }
+  }
+  if (testNum != 2)
+    WebRtcIsac_DecoderInit(ISAC_main_inst);
+  if (CodingMode == 1) {
+    err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+    if (err < 0) {
+      /* exit if returned with error */
+      errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+      printf("\n\n Error in initialization (control): %d.\n\n", errtype);
+      cout << flush;
+      if (testNum == 0) {
+        exit(EXIT_FAILURE);
+      }
+    }
+  }
+
+  if ((setControlBWE) && (CodingMode == 0)) {
+    err = WebRtcIsac_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
+    if (err < 0) {
+      /* exit if returned with error */
+      errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+
+      printf("\n\n Error in Control BWE: %d.\n\n", errtype);
+      cout << flush;
+      exit(EXIT_FAILURE);
+    }
+  }
+
+  if (payloadSize != 0) {
+    err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
+    if (err < 0) {
+      /* exit if returned with error */
+      errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+      printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype);
+      cout << flush;
+      exit(EXIT_FAILURE);
+    }
+  }
+  if (payloadRate != 0) {
+    err = WebRtcIsac_SetMaxRate(ISAC_main_inst, payloadRate);
+    if (err < 0) {
+      /* exit if returned with error */
+      errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+      printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype);
+      cout << flush;
+      exit(EXIT_FAILURE);
+    }
+  }
+
+  *speechType = 1;
+
+  cout << "\n" << flush;
+
+  length_file = 0;
+  int16_t bnIdxTC = 0;
+  int16_t jitterInfoTC = 0;
+  while (endfile == 0) {
+    /* Call init functions at random, fault test number 7 */
+    if (testNum == 7 && (rand() % 2 == 0)) {
+      err = WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
+      /* Error check */
+      if (err < 0) {
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        printf("\n\n Error in encoderinit: %d.\n\n", errtype);
         cout << flush;
-		exit(1);
-	}
+      }
 
-	if((outp = fopen(outname,"wb")) == NULL)
-    {
-		printf("  Error iSAC Cannot write file %s.\n", outname);
-        cout << flush;
-        getc(stdin);
-		exit(1);
-	}
-	if(VADusage)
-    {
-		if((vadp = fopen(vadfile,"rb")) == NULL)
-        {
-			printf("  Error iSAC Cannot read file %s.\n", vadfile);
-            cout << flush;
-			exit(1);
-		}
-	}
-
-    if((bandwidthp = fopen("bwe.pcm","wb")) == NULL)
-    {
-            printf("  Error iSAC Cannot read file %s.\n", "bwe.pcm");
-            cout << flush;
-            exit(1);
+      WebRtcIsac_DecoderInit(ISAC_main_inst);
     }
 
+    cur_framesmpls = 0;
+    while (1) {
+      int stream_len_int = 0;
 
-	starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
+      /* Read 10 ms speech block */
+      endfile = readframe(shortdata, inp, samplesIn10Ms);
 
-    /* Initialize the ISAC and BN structs */
-    if(testNum != 8)
-    {
-        if(!useAssign)
-        {
-            err =WebRtcIsac_Create(&ISAC_main_inst);
-            WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
-            WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ?
-                32000 : 16000);
+      if (endfile) {
+        numFileLoop++;
+        if (numFileLoop < totFileLoop) {
+          rewind(inp);
+          framecnt = 0;
+          fprintf(stderr, "\n");
+          endfile = readframe(shortdata, inp, samplesIn10Ms);
         }
-        else
-        {
-            /* Test the Assign functions */
-            int sss;
-            void *ppp;
-            err = WebRtcIsac_AssignSize(&sss);
-            ppp = malloc(sss);
-            err = WebRtcIsac_Assign(&ISAC_main_inst, ppp);
-            WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
-            WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ?
-                32000 : 16000);
-        }
-        /* Error check */
-        if(err < 0)
-        {
-            printf("\n\n Error in create.\n\n");
-            cout << flush;
-            exit(EXIT_FAILURE);
-        }
-    }
-	BN_data.arrival_time  = 0;
-	BN_data.sample_count  = 0;
-	BN_data.rtp_number    = 0;
+      }
 
-	/* Initialize encoder and decoder */
-    framecnt= 0;
-    endfile	= 0;
+      if (testNum == 7) {
+        srand((unsigned int)time(NULL));
+      }
 
-    if(doTransCoding)
-    {
-        WebRtcIsac_Create(&decoderTransCoding);
-        WebRtcIsac_SetEncSampRate(decoderTransCoding, sampFreqKHz * 1000);
-        WebRtcIsac_SetDecSampRate(decoderTransCoding, sampFreqKHz >= 32 ?
-            32000 : 16000);
-        WebRtcIsac_DecoderInit(decoderTransCoding);
-        transCodingFile = fopen(transCodingFileName, "wb");
-        if(transCodingFile == NULL)
-        {
-            printf("Could not open %s to output trans-coding.\n", transCodingFileName);
-            exit(0);
-        }
-        strcat(transCodingFileName, ".bit");
-        transcodingBitstream = fopen(transCodingFileName, "wb");
-        if(transcodingBitstream == NULL)
-        {
-            printf("Could not open %s to write the bit-stream of transcoder.\n", transCodingFileName);
-            exit(0);
-        }
-    }
+      /* iSAC encoding */
+      if (!(testNum == 3 && framecnt == 0)) {
+        stream_len_int =
+            WebRtcIsac_Encode(ISAC_main_inst, shortdata, (uint8_t*)streamdata);
+        if ((payloadSize != 0) && (stream_len_int > payloadSize)) {
+          if (testNum == 0) {
+            printf("\n\n");
+          }
 
-    if(testNum != 1)
-    {
-		if(WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode) < 0)
-        {
-            printf("Error could not initialize the encoder \n");
-            cout << flush;
-            return 0;
+          printf("\nError: Streamsize out of range %d\n",
+                 stream_len_int - payloadSize);
+          cout << flush;
         }
-	}
-    if(testNum != 2)
-    {
-        if(WebRtcIsac_DecoderInit(ISAC_main_inst) < 0)
-        {
-            printf("Error could not initialize the decoder \n");
-            cout << flush;
-            return 0;
-        }
-	}
-	if(CodingMode == 1)
-    {
-        err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
-        if(err < 0)
-        {
-            /* exit if returned with error */
-            errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-            printf("\n\n Error in initialization (control): %d.\n\n", errtype);
-            cout << flush;
-            if(testNum == 0)
-            {
-                exit(EXIT_FAILURE);
+
+        WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN);
+
+        if (stream_len_int > 0) {
+          if (doTransCoding) {
+            int16_t indexStream;
+            uint8_t auxUW8;
+
+            /******************** Main Transcoding stream ********************/
+            WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC,
+                                          &jitterInfoTC);
+            int streamLenTransCoding_int = WebRtcIsac_GetNewBitStream(
+                ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
+                streamDataTransCoding, false);
+            if (streamLenTransCoding_int < 0) {
+              fprintf(stderr, "Error in trans-coding\n");
+              exit(0);
             }
-        }
-	}
-
-    if((setControlBWE) && (CodingMode == 0))
-    {
-        err = WebRtcIsac_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
-        if(err < 0)
-        {
-            /* exit if returned with error */
-            errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-
-            printf("\n\n Error in Control BWE: %d.\n\n", errtype);
-            cout << flush;
-            exit(EXIT_FAILURE);
-        }
-    }
-
-    if(payloadSize != 0)
-    {
-        err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
-        if(err < 0)
-        {
-            /* exit if returned with error */
-            errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-            printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype);
-            cout << flush;
-            exit(EXIT_FAILURE);
-        }
-    }
-    if(payloadRate != 0)
-    {
-        err = WebRtcIsac_SetMaxRate(ISAC_main_inst, payloadRate);
-        if(err < 0)
-        {
-            /* exit if returned with error */
-            errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-            printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype);
-            cout << flush;
-            exit(EXIT_FAILURE);
-        }
-    }
-
-	*speechType = 1;
-
-    cout << "\n" << flush;
-
-    length_file = 0;
-    int16_t bnIdxTC = 0;
-    int16_t jitterInfoTC = 0;
-    while (endfile == 0)
-    {
-        /* Call init functions at random, fault test number 7 */
-		if(testNum == 7 && (rand()%2 == 0))
-        {
-            err = WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
-            /* Error check */
-            if(err < 0)
-            {
-                errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-                printf("\n\n Error in encoderinit: %d.\n\n", errtype);
-                cout << flush;
+            streamLenTransCoding =
+                static_cast<size_t>(streamLenTransCoding_int);
+            auxUW8 = (uint8_t)(((streamLenTransCoding & 0xFF00) >> 8) & 0x00FF);
+            if (fwrite(&auxUW8, sizeof(uint8_t), 1, transcodingBitstream) !=
+                1) {
+              return -1;
             }
 
-            err = WebRtcIsac_DecoderInit(ISAC_main_inst);
-            /* Error check */
-            if(err < 0)
-            {
-                errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-                printf("\n\n Error in decoderinit: %d.\n\n", errtype);
-                cout << flush;
-            }
-        }
-
-		cur_framesmpls = 0;
-		while (1)
-        {
-            /* Read 10 ms speech block */
-            endfile = readframe(shortdata, inp, samplesIn10Ms);
-
-            if(endfile)
-            {
-                numFileLoop++;
-                if(numFileLoop < totFileLoop)
-                {
-                    rewind(inp);
-                    framecnt = 0;
-                    fprintf(stderr, "\n");
-                    endfile = readframe(shortdata, inp, samplesIn10Ms);
-                }
+            auxUW8 = (uint8_t)(streamLenTransCoding & 0x00FF);
+            if (fwrite(&auxUW8, sizeof(uint8_t), 1, transcodingBitstream) !=
+                1) {
+              return -1;
             }
 
-            if(testNum == 7)
-            {
-		    	srand((unsigned int)time(NULL));
-		    }
-
-            /* iSAC encoding */
-            if(!(testNum == 3 && framecnt == 0))
-            {
-                stream_len = WebRtcIsac_Encode(ISAC_main_inst,
-                                               shortdata,
-                                               (uint8_t*)streamdata);
-                if((payloadSize != 0) && (stream_len > payloadSize))
-                {
-                    if(testNum == 0)
-                    {
-                        printf("\n\n");
-                    }
-
-                    printf("\nError: Streamsize out of range %d\n", stream_len - payloadSize);
-                    cout << flush;
-                }
-
-                WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN);
-
-                if(stream_len>0)
-                {
-                    if(doTransCoding)
-                    {
-                        int16_t indexStream;
-                        uint8_t auxUW8;
-
-                        /************************* Main Transcoding stream *******************************/
-                        WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC, &jitterInfoTC);
-                        streamLenTransCoding = WebRtcIsac_GetNewBitStream(
-                            ISAC_main_inst,
-                            bnIdxTC,
-                            jitterInfoTC,
-                            rateTransCoding,
-                            streamDataTransCoding,
-                            false);
-                        if(streamLenTransCoding < 0)
-                        {
-                            fprintf(stderr, "Error in trans-coding\n");
-                            exit(0);
-                        }
-                        auxUW8 = (uint8_t)(((streamLenTransCoding & 0xFF00) >> 8) &  0x00FF);
-                        if (fwrite(&auxUW8, sizeof(uint8_t), 1,
-                                   transcodingBitstream) != 1) {
-                          return -1;
-                        }
-
-                        auxUW8 = (uint8_t)(streamLenTransCoding & 0x00FF);
-                        if (fwrite(&auxUW8, sizeof(uint8_t),
-                                   1, transcodingBitstream) != 1) {
-                          return -1;
-                        }
-
-                        if (fwrite(streamDataTransCoding,
-                                   sizeof(uint8_t),
-                                   streamLenTransCoding,
-                                   transcodingBitstream) !=
-                            static_cast<size_t>(streamLenTransCoding)) {
-                          return -1;
-                        }
-
-                        WebRtcIsac_ReadBwIndex(streamDataTransCoding,
-                                               &indexStream);
-                        if (indexStream != bnIdxTC) {
-                            fprintf(stderr, "Error in inserting Bandwidth index into transcoding stream.\n");
-                            exit(0);
-                        }
-                        numTransCodingBytes += streamLenTransCoding;
-                    }
-                }
-            }
-            else
-            {
-                break;
+            if (fwrite(streamDataTransCoding, sizeof(uint8_t),
+                       streamLenTransCoding, transcodingBitstream) !=
+                streamLenTransCoding) {
+              return -1;
             }
 
-			if(stream_len < 0)
-            {
-				/* exit if returned with error */
-				errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-                printf("\n\nError in encoder: %d.\n\n", errtype);
-                cout << flush;
-			}
-			cur_framesmpls += samplesIn10Ms;
-			/* exit encoder loop if the encoder returned a bitstream */
-			if(stream_len != 0) break;
-		}
-
-        /* read next bottleneck rate */
-        if(f_bn != NULL)
-        {
-            if(fscanf(f_bn, "%d", &bottleneck) == EOF)
-            {
-                /* Set pointer to beginning of file */
-                fseek(f_bn, 0L, SEEK_SET);
-                if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
-                    exit(0);
-                }
+            WebRtcIsac_ReadBwIndex(streamDataTransCoding, &indexStream);
+            if (indexStream != bnIdxTC) {
+              fprintf(stderr,
+                      "Error in inserting Bandwidth index into transcoding "
+                      "stream.\n");
+              exit(0);
             }
-            if(CodingMode == 1)
-            {
-                WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
-            }
-        }
-
-        length_file += cur_framesmpls;
-        if(cur_framesmpls == (3 * samplesIn10Ms))
-        {
-            maxStreamLen30 = (stream_len > maxStreamLen30)? stream_len:maxStreamLen30;
-        }
-        else
-        {
-            maxStreamLen60 = (stream_len > maxStreamLen60)? stream_len:maxStreamLen60;
-        }
-
-        if(!lostFrame)
-        {
-            lostFrame = ((rand()%100) < packetLossPercent);
-        }
-        else
-        {
-            lostFrame = 0;
-        }
-
-        // RED.
-        if(lostFrame)
-        {
-            stream_len = WebRtcIsac_GetRedPayload(
-                ISAC_main_inst, reinterpret_cast<uint8_t*>(streamdata));
-
-            if(doTransCoding)
-            {
-                streamLenTransCoding = WebRtcIsac_GetNewBitStream(
-                    ISAC_main_inst,
-                    bnIdxTC,
-                    jitterInfoTC,
-                    rateTransCoding,
-                    streamDataTransCoding,
-                    true);
-                if(streamLenTransCoding < 0)
-                {
-                    fprintf(stderr, "Error in RED trans-coding\n");
-                    exit(0);
-                }
-            }
-        }
-
-        /* make coded sequence to short be inreasing */
-		/* the length the decoder expects */
-		if(testNum == 4)
-        {
-			stream_len += 10;
-		}
-
-		/* make coded sequence to long be decreasing */
-		/* the length the decoder expects */
-		if(testNum == 5)
-        {
-			stream_len -= 10;
-		}
-
-        if(testNum == 6)
-        {
-			srand((unsigned int)time(NULL));
-            for(i = 0; i < stream_len; i++)
-            {
-				streamdata[i] = rand();
-            }
-		}
-
-        if(VADusage){
-            readframe(vaddata, vadp, samplesIn10Ms*3);
-        }
-
-		/* simulate packet handling through NetEq and the modem */
-		if(!(testNum == 3 && framecnt == 0))
-        {
-            get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data,
-                sampFreqKHz*1000, sampFreqKHz*1000);
-        }
-
-		if(VADusage && (framecnt>10 && vaddata[0]==0))
-        {
-			BN_data.rtp_number--;
-		}
-        else
-        {
-            /* Error test number 10, garbage data */
-            if(testNum == 10)
-            {
-                /* Test to run decoder with garbage data */
-                for(i = 0; i < stream_len; i++)
-                {
-                    streamdata[i] = (short) (streamdata[i]) + (short) rand();
-                }
-            }
-
-            if(testNum != 9)
-            {
-              err = WebRtcIsac_UpdateBwEstimate(
-                  ISAC_main_inst,
-                  reinterpret_cast<const uint8_t*>(streamdata),
-                  stream_len,
-                  BN_data.rtp_number,
-                  BN_data.sample_count,
-                  BN_data.arrival_time);
-
-                if(err < 0)
-                {
-                    /* exit if returned with error */
-                    errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-                    if(testNum == 0)
-                    {
-                        printf("\n\n");
-                    }
-
-                    printf("Error: in decoder: %d.", errtype);
-                    cout << flush;
-                    if(testNum == 0)
-                    {
-                        printf("\n\n");
-                    }
-
-                }
-            }
-
-            /* Call getFramelen, only used here for function test */
-            err = WebRtcIsac_ReadFrameLen(
-                ISAC_main_inst,
-                reinterpret_cast<const uint8_t*>(streamdata),
-                &FL);
-            if(err < 0)
-            {
-                /* exit if returned with error */
-                errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-                if(testNum == 0)
-                {
-                    printf("\n\n");
-                }
-                printf("    Error: in getFrameLen %d.", errtype);
-                cout << flush;
-                if(testNum == 0)
-                {
-                    printf("\n\n");
-                }
-            }
-
-            // iSAC decoding
-
-            if(lostFrame)
-            {
-                declen = WebRtcIsac_DecodeRcu(
-                    ISAC_main_inst,
-                    reinterpret_cast<const uint8_t*>(streamdata),
-                    stream_len,
-                    decoded,
-                    speechType);
-
-                if(doTransCoding)
-                {
-                    declenTC = WebRtcIsac_DecodeRcu(
-                        decoderTransCoding,
-                        streamDataTransCoding,
-                        streamLenTransCoding,
-                        decodedTC,
-                        speechType);
-                }
-            }
-            else
-            {
-                declen = WebRtcIsac_Decode(
-                    ISAC_main_inst,
-                    reinterpret_cast<const uint8_t*>(streamdata),
-                    stream_len,
-                    decoded,
-                    speechType);
-                if(doTransCoding)
-                {
-                    declenTC = WebRtcIsac_Decode(
-                        decoderTransCoding,
-                        streamDataTransCoding,
-                        streamLenTransCoding,
-                        decodedTC,
-                        speechType);
-                }
-            }
-
-            if(declen < 0)
-            {
-                /* exit if returned with error */
-                errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-                if(testNum == 0)
-                {
-                    printf("\n\n");
-                }
-                printf("    Error: in decoder %d.", errtype);
-                cout << flush;
-                if(testNum == 0)
-                {
-                    printf("\n\n");
-                }
-            }
-
-            if(declenTC < 0)
-            {
-                if(testNum == 0)
-                {
-                    printf("\n\n");
-                }
-                printf("    Error: in decoding the transcoded stream");
-                cout << flush;
-                if(testNum == 0)
-                {
-                    printf("\n\n");
-                }
-
-            }
-        }
-        /* Write decoded speech frame to file */
-        if((declen > 0) && (numFileLoop == 0))
-        {
-          if (fwrite(decoded, sizeof(int16_t), declen,
-                     outp) != static_cast<size_t>(declen)) {
-            return -1;
+            numTransCodingBytes += streamLenTransCoding;
           }
         }
+      } else {
+        break;
+      }
 
-        if((declenTC > 0) && (numFileLoop == 0))
-        {
-          if (fwrite(decodedTC, sizeof(int16_t), declen,
-                     transCodingFile) != static_cast<size_t>(declen)) {
-            return -1;
+      if (stream_len_int < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        fprintf(stderr, "Error in encoder: %d.\n", errtype);
+        cout << flush;
+        exit(0);
+      }
+      stream_len = static_cast<size_t>(stream_len_int);
+
+      cur_framesmpls += samplesIn10Ms;
+      /* exit encoder loop if the encoder returned a bitstream */
+      if (stream_len != 0)
+        break;
+    }
+
+    /* read next bottleneck rate */
+    if (f_bn != NULL) {
+      if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+        /* Set pointer to beginning of file */
+        fseek(f_bn, 0L, SEEK_SET);
+        if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+          exit(0);
+        }
+      }
+      if (CodingMode == 1) {
+        WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
+      }
+    }
+
+    length_file += cur_framesmpls;
+    if (cur_framesmpls == (3 * samplesIn10Ms)) {
+      maxStreamLen30 =
+          (stream_len > maxStreamLen30) ? stream_len : maxStreamLen30;
+    } else {
+      maxStreamLen60 =
+          (stream_len > maxStreamLen60) ? stream_len : maxStreamLen60;
+    }
+
+    if (!lostFrame) {
+      lostFrame = ((rand() % 100) < packetLossPercent);
+    } else {
+      lostFrame = false;
+    }
+
+    // RED.
+    if (lostFrame) {
+      int stream_len_int = WebRtcIsac_GetRedPayload(
+          ISAC_main_inst, reinterpret_cast<uint8_t*>(streamdata));
+      if (stream_len_int < 0) {
+        fprintf(stderr, "Error getting RED payload\n");
+        exit(0);
+      }
+      stream_len = static_cast<size_t>(stream_len_int);
+
+      if (doTransCoding) {
+        int streamLenTransCoding_int = WebRtcIsac_GetNewBitStream(
+            ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
+            streamDataTransCoding, true);
+        if (streamLenTransCoding_int < 0) {
+          fprintf(stderr, "Error in RED trans-coding\n");
+          exit(0);
+        }
+        streamLenTransCoding =
+            static_cast<size_t>(streamLenTransCoding_int);
+      }
+    }
+
+    /* make coded sequence to short be inreasing */
+    /* the length the decoder expects */
+    if (testNum == 4) {
+      stream_len += 10;
+    }
+
+    /* make coded sequence to long be decreasing */
+    /* the length the decoder expects */
+    if (testNum == 5) {
+      stream_len -= 10;
+    }
+
+    if (testNum == 6) {
+      srand((unsigned int)time(NULL));
+      for (i = 0; i < stream_len; i++) {
+        streamdata[i] = rand();
+      }
+    }
+
+    if (VADusage) {
+      readframe(vaddata, vadp, samplesIn10Ms * 3);
+    }
+
+    /* simulate packet handling through NetEq and the modem */
+    if (!(testNum == 3 && framecnt == 0)) {
+      get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data,
+                       sampFreqKHz * 1000, sampFreqKHz * 1000);
+    }
+
+    if (VADusage && (framecnt > 10 && vaddata[0] == 0)) {
+      BN_data.rtp_number--;
+    } else {
+      /* Error test number 10, garbage data */
+      if (testNum == 10) {
+        /* Test to run decoder with garbage data */
+        for (i = 0; i < stream_len; i++) {
+          streamdata[i] = (short)(streamdata[i]) + (short)rand();
+        }
+      }
+
+      if (testNum != 9) {
+        err = WebRtcIsac_UpdateBwEstimate(
+            ISAC_main_inst, reinterpret_cast<const uint8_t*>(streamdata),
+            stream_len, BN_data.rtp_number, BN_data.sample_count,
+            BN_data.arrival_time);
+
+        if (err < 0) {
+          /* exit if returned with error */
+          errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          if (testNum == 0) {
+            printf("\n\n");
+          }
+
+          printf("Error: in decoder: %d.", errtype);
+          cout << flush;
+          if (testNum == 0) {
+            printf("\n\n");
           }
         }
+      }
 
+      /* Call getFramelen, only used here for function test */
+      err = WebRtcIsac_ReadFrameLen(
+          ISAC_main_inst, reinterpret_cast<const uint8_t*>(streamdata), &FL);
+      if (err < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        if (testNum == 0) {
+          printf("\n\n");
+        }
+        printf("    Error: in getFrameLen %d.", errtype);
+        cout << flush;
+        if (testNum == 0) {
+          printf("\n\n");
+        }
+      }
 
-		fprintf(stderr, "\rframe = %5d  ", framecnt);
-        fflush(stderr);
-		framecnt++;
+      // iSAC decoding
 
-        /* Error test number 10, garbage data */
-        //if(testNum == 10)
-        //{
-        //    /* Test to run decoder with garbage data */
-        //    if( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL )
-        //    {
-        //        fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
-        //    }
-        //    else
-        //    {
-        //        fprintf(seedfile, "ok\n\n");
-        //        fclose(seedfile);
-        //    }
-        //}
-        /* Error test number 10, garbage data */
-        //if(testNum == 10)
-        //{
-        //    /* Test to run decoder with garbage data */
-        //    for ( i = 0; i < stream_len; i++)
-        //    {
-        //        streamdata[i] = (short) (streamdata[i] + (short) rand());
-        //    }
-        //}
+      if (lostFrame) {
+        declen = WebRtcIsac_DecodeRcu(
+            ISAC_main_inst, reinterpret_cast<const uint8_t*>(streamdata),
+            stream_len, decoded, speechType);
 
+        if (doTransCoding) {
+          declenTC =
+              WebRtcIsac_DecodeRcu(decoderTransCoding, streamDataTransCoding,
+                                   streamLenTransCoding, decodedTC, speechType);
+        }
+      } else {
+        declen = WebRtcIsac_Decode(ISAC_main_inst,
+                                   reinterpret_cast<const uint8_t*>(streamdata),
+                                   stream_len, decoded, speechType);
+        if (doTransCoding) {
+          declenTC =
+              WebRtcIsac_Decode(decoderTransCoding, streamDataTransCoding,
+                                streamLenTransCoding, decodedTC, speechType);
+        }
+      }
 
-		totalsmpls += declen;
-		totalbits += 8 * stream_len;
-#ifdef _DEBUG
-        kbps = ((double) sampFreqKHz * 1000.) / ((double) cur_framesmpls) * 8.0 * stream_len / 1000.0;// kbits/s
-		fy = fopen("bit_rate.dat", "a");
-		fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
-		fclose(fy);
+      if (declen < 0) {
+        /* exit if returned with error */
+        errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        if (testNum == 0) {
+          printf("\n\n");
+        }
+        printf("    Error: in decoder %d.", errtype);
+        cout << flush;
+        if (testNum == 0) {
+          printf("\n\n");
+        }
+      }
 
-#endif /* _DEBUG */
-
-	}
-	printf("\n");
-	printf("total bits               = %d bits\n", totalbits);
-	printf("measured average bitrate = %0.3f kbits/s\n",
-        (double)totalbits *(sampFreqKHz) / totalsmpls);
-    if(doTransCoding)
-    {
-        printf("Transcoding average bit-rate = %0.3f kbps\n",
-            (double)numTransCodingBytes * 8.0 *(sampFreqKHz) / totalsmpls);
-        fclose(transCodingFile);
+      if (declenTC < 0) {
+        if (testNum == 0) {
+          printf("\n\n");
+        }
+        printf("    Error: in decoding the transcoded stream");
+        cout << flush;
+        if (testNum == 0) {
+          printf("\n\n");
+        }
+      }
     }
-	printf("\n");
-
-	/* Runtime statistics */
-	runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
-	length_file = length_file /(sampFreqKHz * 1000.);
-
-    printf("\n\nLength of speech file: %.1f s\n", length_file);
-	printf("Time to run iSAC:      %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
-
-    if(maxStreamLen30 != 0)
-    {
-        printf("Maximum payload size 30ms Frames %d bytes (%0.3f kbps)\n",
-            maxStreamLen30,
-            maxStreamLen30 * 8 / 30.);
-    }
-    if(maxStreamLen60 != 0)
-    {
-        printf("Maximum payload size 60ms Frames %d bytes (%0.3f kbps)\n",
-            maxStreamLen60,
-            maxStreamLen60 * 8 / 60.);
-    }
-    //fprintf(stderr, "\n");
-
-	fprintf(stderr, "   %.1f s", length_file);
-    fprintf(stderr, "   %0.1f kbps", (double)totalbits *(sampFreqKHz) / totalsmpls);
-    if(maxStreamLen30 != 0)
-    {
-        fprintf(stderr, "   plmax-30ms %d bytes (%0.0f kbps)",
-            maxStreamLen30,
-            maxStreamLen30 * 8 / 30.);
-    }
-    if(maxStreamLen60 != 0)
-    {
-        fprintf(stderr, "   plmax-60ms %d bytes (%0.0f kbps)",
-            maxStreamLen60,
-            maxStreamLen60 * 8 / 60.);
-    }
-    if(doTransCoding)
-    {
-        fprintf(stderr, "  transcoding rate %.0f kbps",
-            (double)numTransCodingBytes * 8.0 *(sampFreqKHz) / totalsmpls);
+    /* Write decoded speech frame to file */
+    if ((declen > 0) && (numFileLoop == 0)) {
+      if (fwrite(decoded, sizeof(int16_t), declen, outp) !=
+          static_cast<size_t>(declen)) {
+        return -1;
+      }
     }
 
-    fclose(inp);
-	fclose(outp);
-	WebRtcIsac_Free(ISAC_main_inst);
+    if ((declenTC > 0) && (numFileLoop == 0)) {
+      if (fwrite(decodedTC, sizeof(int16_t), declen, transCodingFile) !=
+          static_cast<size_t>(declen)) {
+        return -1;
+      }
+    }
 
+    fprintf(stderr, "\rframe = %5d  ", framecnt);
+    fflush(stderr);
+    framecnt++;
 
-	exit(0);
+    /* Error test number 10, garbage data */
+    // if (testNum == 10)
+    // {
+    //   /* Test to run decoder with garbage data */
+    //   if ((seedfile = fopen(SEED_FILE, "a+t")) == NULL) {
+    //     fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+    //   } else {
+    //     fprintf(seedfile, "ok\n\n");
+    //     fclose(seedfile);
+    //   }
+    // }
+    /* Error test number 10, garbage data */
+    // if (testNum == 10) {
+    //   /* Test to run decoder with garbage data */
+    //   for (i = 0; i < stream_len; i++) {
+    //     streamdata[i] = (short) (streamdata[i] + (short) rand());
+    //   }
+    // }
+
+    totalsmpls += declen;
+    totalbits += 8 * stream_len;
+#if !defined(NDEBUG)
+    kbps = ((double)sampFreqKHz * 1000.) / ((double)cur_framesmpls) * 8.0 *
+           stream_len / 1000.0;  // kbits/s
+    fy = fopen("bit_rate.dat", "a");
+    fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
+    fclose(fy);
+
+#endif
+  }
+  printf("\n");
+  printf("total bits               = %" PRIuS " bits\n", totalbits);
+  printf("measured average bitrate = %0.3f kbits/s\n",
+         (double)totalbits * (sampFreqKHz) / totalsmpls);
+  if (doTransCoding) {
+    printf("Transcoding average bit-rate = %0.3f kbps\n",
+           (double)numTransCodingBytes * 8.0 * (sampFreqKHz) / totalsmpls);
+    fclose(transCodingFile);
+  }
+  printf("\n");
+
+  /* Runtime statistics */
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC - starttime);
+  length_file = length_file / (sampFreqKHz * 1000.);
+
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run iSAC:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+
+  if (maxStreamLen30 != 0) {
+    printf("Maximum payload size 30ms Frames %" PRIuS " bytes (%0.3f kbps)\n",
+           maxStreamLen30, maxStreamLen30 * 8 / 30.);
+  }
+  if (maxStreamLen60 != 0) {
+    printf("Maximum payload size 60ms Frames %" PRIuS " bytes (%0.3f kbps)\n",
+           maxStreamLen60, maxStreamLen60 * 8 / 60.);
+  }
+  // fprintf(stderr, "\n");
+
+  fprintf(stderr, "   %.1f s", length_file);
+  fprintf(stderr, "   %0.1f kbps",
+          (double)totalbits * (sampFreqKHz) / totalsmpls);
+  if (maxStreamLen30 != 0) {
+    fprintf(stderr, "   plmax-30ms %" PRIuS " bytes (%0.0f kbps)",
+            maxStreamLen30, maxStreamLen30 * 8 / 30.);
+  }
+  if (maxStreamLen60 != 0) {
+    fprintf(stderr, "   plmax-60ms %" PRIuS " bytes (%0.0f kbps)",
+            maxStreamLen60, maxStreamLen60 * 8 / 60.);
+  }
+  if (doTransCoding) {
+    fprintf(stderr, "  transcoding rate %.0f kbps",
+            (double)numTransCodingBytes * 8.0 * (sampFreqKHz) / totalsmpls);
+  }
+
+  fclose(inp);
+  fclose(outp);
+  WebRtcIsac_Free(ISAC_main_inst);
+
+  exit(0);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
index 6ec818ee76..a53e7bd0b5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -51,9 +51,9 @@ int main(int argc, char* argv[])
 
   short clientCntr;
 
-  unsigned int lenEncodedInBytes[MAX_NUM_CLIENTS];
+  size_t lenEncodedInBytes[MAX_NUM_CLIENTS];
   unsigned int lenAudioIn10ms[MAX_NUM_CLIENTS];
-  unsigned int lenEncodedInBytesTmp[MAX_NUM_CLIENTS];
+  size_t lenEncodedInBytesTmp[MAX_NUM_CLIENTS];
   unsigned int lenAudioIn10msTmp[MAX_NUM_CLIENTS];
   BottleNeckModel* packetData[MAX_NUM_CLIENTS];
 
@@ -166,13 +166,7 @@ int main(int argc, char* argv[])
       return -1;
     }
 
-    // Initialize Decoder
-    if(WebRtcIsac_DecoderInit(codecInstance[clientCntr]) < 0)
-    {
-      printf("Could not initialize decoder of client %d\n",
-             clientCntr + 1);
-      return -1;
-    }
+    WebRtcIsac_DecoderInit(codecInstance[clientCntr]);
 
     // setup Rate if in Instantaneous mode
     if(codingMode != 0)
@@ -189,9 +183,9 @@ int main(int argc, char* argv[])
   }
 
 
-  short streamLen;
+  size_t streamLen;
   short numSamplesRead;
-  short lenDecodedAudio;
+  size_t lenDecodedAudio;
   short senderIdx;
   short receiverIdx;
 
@@ -282,11 +276,11 @@ int main(int argc, char* argv[])
       // Encode
 
 
-      streamLen = WebRtcIsac_Encode(codecInstance[senderIdx],
-                                    audioBuff10ms,
-                                    (uint8_t*)bitStream);
+      int streamLen_int = WebRtcIsac_Encode(codecInstance[senderIdx],
+                                            audioBuff10ms,
+                                            (uint8_t*)bitStream);
       int16_t ggg;
-      if (streamLen > 0) {
+      if (streamLen_int > 0) {
         if ((WebRtcIsac_ReadFrameLen(
                 codecInstance[receiverIdx],
                 reinterpret_cast<const uint8_t*>(bitStream),
@@ -295,11 +289,12 @@ int main(int argc, char* argv[])
       }
 
       // Sanity check
-      if(streamLen < 0)
+      if(streamLen_int < 0)
       {
         printf(" Encoder error in client %d \n", senderIdx + 1);
         return -1;
       }
+      streamLen = static_cast<size_t>(streamLen_int);
 
 
       if(streamLen > 0)
@@ -423,18 +418,18 @@ int main(int argc, char* argv[])
         }
         /**/
         // Decode
-        lenDecodedAudio = WebRtcIsac_Decode(
+        int lenDecodedAudio_int = WebRtcIsac_Decode(
             codecInstance[receiverIdx],
             reinterpret_cast<const uint8_t*>(bitStream),
             streamLen,
             audioBuff60ms,
             speechType);
-        if(lenDecodedAudio < 0)
+        if(lenDecodedAudio_int < 0)
         {
           printf(" Decoder error in client %d \n", receiverIdx + 1);
           return -1;
         }
-
+        lenDecodedAudio = static_cast<size_t>(lenDecodedAudio_int);
 
         if(encoderSampRate[senderIdx] == 16000)
         {
@@ -442,7 +437,7 @@ int main(int argc, char* argv[])
                                 resamplerState[receiverIdx]);
           if (fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
                      outFile[receiverIdx]) !=
-              static_cast<size_t>(lenDecodedAudio << 1)) {
+              lenDecodedAudio << 1) {
             return -1;
           }
         }
@@ -450,7 +445,7 @@ int main(int argc, char* argv[])
         {
           if (fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
                      outFile[receiverIdx]) !=
-              static_cast<size_t>(lenDecodedAudio)) {
+              lenDecodedAudio) {
             return -1;
           }
         }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index d10b4addbd..e8116ffdf8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -17,7 +17,7 @@
 
 #ifdef WIN32
 #include "windows.h"
-#define CLOCKS_PER_SEC  1000
+#define CLOCKS_PER_SEC 1000
 #endif
 
 #include <ctype.h>
@@ -26,634 +26,551 @@
 /* include API */
 #include "isac.h"
 #include "utility.h"
+#include "webrtc/base/format_macros.h"
 //#include "commonDefs.h"
 
 /* max number of samples per frame (= 60 ms frame) */
-#define MAX_FRAMESAMPLES_SWB                1920
+#define MAX_FRAMESAMPLES_SWB 1920
 /* number of samples per 10ms frame */
-#define FRAMESAMPLES_SWB_10ms               320
-#define FRAMESAMPLES_WB_10ms                160
+#define FRAMESAMPLES_SWB_10ms 320
+#define FRAMESAMPLES_WB_10ms 160
 
 /* sampling frequency (Hz) */
-#define FS_SWB                               32000
-#define FS_WB                                16000
+#define FS_SWB 32000
+#define FS_WB 16000
 
 //#define CHANGE_OUTPUT_NAME
 
 #ifdef HAVE_DEBUG_INFO
-    #include "debugUtility.h"
-    debugStruct debugInfo;
+#include "debugUtility.h"
+debugStruct debugInfo;
 #endif
 
 unsigned long framecnt = 0;
 
-int main(int argc, char* argv[])
-{
-    //--- File IO ----
-    FILE* inp;
-    FILE* outp;
-    char inname[500];
-    char outname[500];
+int main(int argc, char* argv[]) {
+  //--- File IO ----
+  FILE* inp;
+  FILE* outp;
+  char inname[500];
+  char outname[500];
 
-    /* Runtime statistics */
-    double        rate;
-    double        rateRCU;
-    unsigned long totalbits = 0;
-    unsigned long totalBitsRCU = 0;
-    unsigned long totalsmpls =0;
+  /* Runtime statistics */
+  double rate;
+  double rateRCU;
+  size_t totalbits = 0;
+  unsigned long totalBitsRCU = 0;
+  unsigned long totalsmpls = 0;
 
-    int32_t   bottleneck = 39;
-    int16_t   frameSize = 30;           /* ms */
-    int16_t   codingMode = 1;
-    int16_t   shortdata[FRAMESAMPLES_SWB_10ms];
-    int16_t   decoded[MAX_FRAMESAMPLES_SWB];
-    //uint16_t  streamdata[1000];
-    int16_t   speechType[1];
-    int16_t   payloadLimit;
-    int32_t   rateLimit;
-    ISACStruct*   ISAC_main_inst;
+  int32_t bottleneck = 39;
+  int frameSize = 30; /* ms */
+  int16_t codingMode = 1;
+  int16_t shortdata[FRAMESAMPLES_SWB_10ms];
+  int16_t decoded[MAX_FRAMESAMPLES_SWB];
+  // uint16_t  streamdata[1000];
+  int16_t speechType[1];
+  int16_t payloadLimit;
+  int32_t rateLimit;
+  ISACStruct* ISAC_main_inst;
 
-    int16_t   stream_len = 0;
-    int16_t   declen = 0;
-    int16_t   err;
-    int16_t   cur_framesmpls;
-    int           endfile;
+  size_t stream_len = 0;
+  int declen = 0;
+  int16_t err;
+  int cur_framesmpls;
+  int endfile;
 #ifdef WIN32
-    double        length_file;
-    double        runtime;
-    char          outDrive[10];
-    char          outPath[500];
-    char          outPrefix[500];
-    char          outSuffix[500];
-    char          bitrateFileName[500];
-    FILE*         bitrateFile;
-    double        starttime;
-    double        rateLB = 0;
-    double        rateUB = 0;
+  double length_file;
+  double runtime;
+  char outDrive[10];
+  char outPath[500];
+  char outPrefix[500];
+  char outSuffix[500];
+  char bitrateFileName[500];
+  FILE* bitrateFile;
+  double starttime;
+  double rateLB = 0;
+  double rateUB = 0;
 #endif
-    FILE*         histFile;
-    FILE*         averageFile;
-    int           sampFreqKHz;
-    int           samplesIn10Ms;
-    int16_t   maxStreamLen = 0;
-    char          histFileName[500];
-    char          averageFileName[500];
-    unsigned int  hist[600];
-    unsigned int  tmpSumStreamLen = 0;
-    unsigned int  packetCntr = 0;
-    unsigned int  lostPacketCntr = 0;
-    uint8_t  payload[1200];
-    uint8_t  payloadRCU[1200];
-    uint16_t  packetLossPercent = 0;
-    int16_t   rcuStreamLen = 0;
-	int onlyEncode;
-	int onlyDecode;
+  FILE* histFile;
+  FILE* averageFile;
+  int sampFreqKHz;
+  int samplesIn10Ms;
+  size_t maxStreamLen = 0;
+  char histFileName[500];
+  char averageFileName[500];
+  unsigned int hist[600];
+  double tmpSumStreamLen = 0;
+  unsigned int packetCntr = 0;
+  unsigned int lostPacketCntr = 0;
+  uint8_t payload[1200];
+  uint8_t payloadRCU[1200];
+  uint16_t packetLossPercent = 0;
+  int16_t rcuStreamLen = 0;
+  int onlyEncode;
+  int onlyDecode;
 
+  BottleNeckModel packetData;
+  packetData.arrival_time = 0;
+  packetData.sample_count = 0;
+  packetData.rtp_number = 0;
+  memset(hist, 0, sizeof(hist));
 
-    BottleNeckModel packetData;
-	packetData.arrival_time  = 0;
-	packetData.sample_count  = 0;
-	packetData.rtp_number    = 0;
-    memset(hist, 0, sizeof(hist));
+  /* handling wrong input arguments in the command line */
+  if (argc < 5) {
+    int size;
+    WebRtcIsac_AssignSize(&size);
 
-    /* handling wrong input arguments in the command line */
-    if(argc < 5)
-    {
-		int size;
-		WebRtcIsac_AssignSize(&size);
+    printf("\n\nWrong number of arguments or flag values.\n\n");
 
-        printf("\n\nWrong number of arguments or flag values.\n\n");
+    printf("Usage:\n\n");
+    printf("%s infile outfile -bn bottleneck [options]\n\n", argv[0]);
+    printf("with:\n");
+    printf("-I.............. indicates encoding in instantaneous mode.\n");
+    printf("-bn bottleneck.. the value of the bottleneck in bit/sec, e.g.\n");
+    printf("                 39742, in instantaneous (channel-independent)\n");
+    printf("                 mode.\n\n");
+    printf("infile.......... Normal speech input file\n\n");
+    printf("outfile......... Speech output file\n\n");
+    printf("OPTIONS\n");
+    printf("-------\n");
+    printf("-fs sampFreq.... sampling frequency of codec 16 or 32 (default)\n");
+    printf("                 kHz.\n");
+    printf("-plim payloadLim payload limit in bytes, default is the maximum\n");
+    printf("                 possible.\n");
+    printf("-rlim rateLim... rate limit in bits/sec, default is the maximum\n");
+    printf("                 possible.\n");
+    printf("-h file......... record histogram and *append* to 'file'.\n");
+    printf("-ave file....... record average rate of 3 sec intervales and\n");
+    printf("                 *append* to 'file'.\n");
+    printf("-ploss.......... packet-loss percentage.\n");
+    printf("-enc............ do only encoding and store the bit-stream\n");
+    printf("-dec............ the input file is a bit-stream, decode it.\n\n");
+    printf("Example usage:\n\n");
+    printf("%s speechIn.pcm speechOut.pcm -B 40000 -fs 32\n\n", argv[0]);
+    printf("structure size %d bytes\n", size);
 
-        printf("Usage:\n\n");
-        printf("%s infile outfile -bn bottelneck [options] \n\n", argv[0]);
-        printf("with:\n");
-        printf("-I................... indicates encoding in instantaneous mode.\n");
-        printf("-bn bottleneck....... the value of the bottleneck in bit/sec, e.g. 39742,\n");
-		printf("                      in instantaneous (channel-independent) mode.\n\n");
-        printf("infile............... Normal speech input file\n\n");
-        printf("outfile.............. Speech output file\n\n");
-        printf("OPTIONS\n");
-        printf("-------\n");
-        printf("-fs sampFreq......... sampling frequency of codec 16 or 32 (default) kHz.\n");
-        printf("-plim payloadLim..... payload limit in bytes,\n");
-        printf("                      default is the maximum possible.\n");
-        printf("-rlim rateLim........ rate limit in bits/sec, \n");
-        printf("                      default is the maimum possible.\n");
-        printf("-h file.............. record histogram and *append* to 'file'.\n");
-        printf("-ave file............ record average rate of 3 sec intervales and *append* to 'file'.\n");
-        printf("-ploss............... packet-loss percentage.\n");
-		printf("-enc................. do only encoding and store the bit-stream\n");
-		printf("-dec................. the input file is a bit-stream, decode it.\n");
+    exit(0);
+  }
 
-        printf("\n");
-        printf("Example usage:\n\n");
-        printf("%s speechIn.pcm speechOut.pcm -B 40000 -fs 32 \n\n", argv[0]);
+  /* Get Bottleneck value */
+  bottleneck = readParamInt(argc, argv, "-bn", 50000);
+  fprintf(stderr, "\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
 
-		printf("structure size %d bytes\n", size);
-
-        exit(0);
+  /* Get Input and Output files */
+  sscanf(argv[1], "%s", inname);
+  sscanf(argv[2], "%s", outname);
+  codingMode = readSwitch(argc, argv, "-I");
+  sampFreqKHz = (int16_t)readParamInt(argc, argv, "-fs", 32);
+  if (readParamString(argc, argv, "-h", histFileName, 500) > 0) {
+    histFile = fopen(histFileName, "a");
+    if (histFile == NULL) {
+      printf("cannot open hist file %s", histFileName);
+      exit(0);
     }
+  } else {
+    // NO recording of hitstogram
+    histFile = NULL;
+  }
 
+  packetLossPercent = readParamInt(argc, argv, "-ploss", 0);
 
-
-    /* Get Bottleneck value */
-    bottleneck = readParamInt(argc, argv, "-bn", 50000);
-    fprintf(stderr,"\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
-
-    /* Get Input and Output files */
-    sscanf(argv[1], "%s", inname);
-    sscanf(argv[2], "%s", outname);
-    codingMode = readSwitch(argc, argv, "-I");
-    sampFreqKHz = (int16_t)readParamInt(argc, argv, "-fs", 32);
-    if(readParamString(argc, argv, "-h", histFileName, 500) > 0)
-    {
-        histFile = fopen(histFileName, "a");
-        if(histFile == NULL)
-        {
-            printf("cannot open hist file %s", histFileName);
-            exit(0);
-        }
+  if (readParamString(argc, argv, "-ave", averageFileName, 500) > 0) {
+    averageFile = fopen(averageFileName, "a");
+    if (averageFile == NULL) {
+      printf("cannot open file to write rate %s", averageFileName);
+      exit(0);
     }
-    else
-    {
-        // NO recording of hitstogram
-        histFile = NULL;
+  } else {
+    averageFile = NULL;
+  }
+
+  onlyEncode = readSwitch(argc, argv, "-enc");
+  onlyDecode = readSwitch(argc, argv, "-dec");
+
+  switch (sampFreqKHz) {
+    case 16: {
+      samplesIn10Ms = 160;
+      break;
     }
-
-
-    packetLossPercent = readParamInt(argc, argv, "-ploss", 0);
-
-    if(readParamString(argc, argv, "-ave", averageFileName, 500) > 0)
-    {
-        averageFile = fopen(averageFileName, "a");
-        if(averageFile == NULL)
-        {
-            printf("cannot open file to write rate %s", averageFileName);
-            exit(0);
-        }
+    case 32: {
+      samplesIn10Ms = 320;
+      break;
     }
-    else
-    {
-        averageFile = NULL;
-    }
-
-	onlyEncode = readSwitch(argc, argv, "-enc");
-	onlyDecode = readSwitch(argc, argv, "-dec");
-
-
-    switch(sampFreqKHz)
-    {
-    case 16:
-        {
-            samplesIn10Ms = 160;
-            break;
-        }
-    case 32:
-        {
-            samplesIn10Ms = 320;
-            break;
-        }
     default:
-        printf("A sampling frequency of %d kHz is not supported,\
-valid values are 8 and 16.\n", sampFreqKHz);
-        exit(-1);
-    }
-    payloadLimit = (int16_t)readParamInt(argc, argv, "-plim", 400);
-    rateLimit = readParamInt(argc, argv, "-rlim", 106800);
+      printf("A sampling frequency of %d kHz is not supported, valid values are"
+             " 8 and 16.\n", sampFreqKHz);
+      exit(-1);
+  }
+  payloadLimit = (int16_t)readParamInt(argc, argv, "-plim", 400);
+  rateLimit = readParamInt(argc, argv, "-rlim", 106800);
 
-    if ((inp = fopen(inname,"rb")) == NULL) {
-        printf("  iSAC: Cannot read file %s.\n", inname);
-        exit(1);
-    }
-    if ((outp = fopen(outname,"wb")) == NULL) {
-        printf("  iSAC: Cannot write file %s.\n", outname);
-        exit(1);
-    }
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  iSAC: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  iSAC: Cannot write file %s.\n", outname);
+    exit(1);
+  }
 
 #ifdef WIN32
-    _splitpath(outname, outDrive, outPath, outPrefix, outSuffix);
-    _makepath(bitrateFileName, outDrive, outPath, "bitrate", ".txt");
+  _splitpath(outname, outDrive, outPath, outPrefix, outSuffix);
+  _makepath(bitrateFileName, outDrive, outPath, "bitrate", ".txt");
 
-    bitrateFile = fopen(bitrateFileName, "a");
-    fprintf(bitrateFile, "%  %%s  \n", inname);
+  bitrateFile = fopen(bitrateFileName, "a");
+  fprintf(bitrateFile, "%  %%s  \n", inname);
 #endif
 
-    printf("\n");
-    printf("Input.................... %s\n", inname);
-    printf("Output................... %s\n", outname);
-    printf("Encoding Mode............ %s\n",
-        (codingMode == 1)? "Channel-Independent":"Channel-Adaptive");
-    printf("Bottleneck............... %d bits/sec\n", bottleneck);
-    printf("Packet-loss Percentage... %d\n", packetLossPercent);
-    printf("\n");
+  printf("\n");
+  printf("Input.................... %s\n", inname);
+  printf("Output................... %s\n", outname);
+  printf("Encoding Mode............ %s\n",
+         (codingMode == 1) ? "Channel-Independent" : "Channel-Adaptive");
+  printf("Bottleneck............... %d bits/sec\n", bottleneck);
+  printf("Packet-loss Percentage... %d\n", packetLossPercent);
+  printf("\n");
 
 #ifdef WIN32
-    starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
+  starttime = clock() / (double)CLOCKS_PER_SEC; /* Runtime statistics */
 #endif
 
-    /* Initialize the ISAC and BN structs */
-    err = WebRtcIsac_Create(&ISAC_main_inst);
+  /* Initialize the ISAC and BN structs */
+  err = WebRtcIsac_Create(&ISAC_main_inst);
 
-    WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
-    WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ? 32000 :
-        16000);
-    /* Error check */
-    if (err < 0) {
-        fprintf(stderr,"\n\n Error in create.\n\n");
+  WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000);
+  WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ? 32000 : 16000);
+  /* Error check */
+  if (err < 0) {
+    fprintf(stderr, "\n\n Error in create.\n\n");
+    exit(EXIT_FAILURE);
+  }
+
+  framecnt = 0;
+  endfile = 0;
+
+  /* Initialize encoder and decoder */
+  if (WebRtcIsac_EncoderInit(ISAC_main_inst, codingMode) < 0) {
+    printf("cannot initialize encoder\n");
+    return -1;
+  }
+  WebRtcIsac_DecoderInit(ISAC_main_inst);
+
+  // {
+  //   int32_t b1, b2;
+  //   FILE* fileID = fopen("GetBNTest.txt", "w");
+  //   b2 = 32100;
+  //   while (b2 <= 52000) {
+  //     WebRtcIsac_Control(ISAC_main_inst, b2, frameSize);
+  //     WebRtcIsac_GetUplinkBw(ISAC_main_inst, &b1);
+  //     fprintf(fileID, "%5d %5d\n", b2, b1);
+  //     b2 += 10;
+  //   }
+  // }
+
+  if (codingMode == 1) {
+    if (WebRtcIsac_Control(ISAC_main_inst, bottleneck, frameSize) < 0) {
+      printf("cannot set bottleneck\n");
+      return -1;
+    }
+  } else {
+    if (WebRtcIsac_ControlBwe(ISAC_main_inst, 15000, 30, 1) < 0) {
+      printf("cannot configure BWE\n");
+      return -1;
+    }
+  }
+
+  if (WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadLimit) < 0) {
+    printf("cannot set maximum payload size %d.\n", payloadLimit);
+    return -1;
+  }
+
+  if (rateLimit < 106800) {
+    if (WebRtcIsac_SetMaxRate(ISAC_main_inst, rateLimit) < 0) {
+      printf("cannot set the maximum rate %d.\n", rateLimit);
+      return -1;
+    }
+  }
+
+  //=====================================
+  //#ifdef HAVE_DEBUG_INFO
+  //    if(setupDebugStruct(&debugInfo) < 0)
+  //    {
+  //        exit(1);
+  //    }
+  //#endif
+
+  while (endfile == 0) {
+    fprintf(stderr, "  \rframe = %7li", framecnt);
+
+    //============== Readind from the file and encoding =================
+    cur_framesmpls = 0;
+    stream_len = 0;
+
+    if (onlyDecode) {
+      uint8_t auxUW8;
+      if (fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1) {
+        break;
+      }
+      stream_len = auxUW8 << 8;
+      if (fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1) {
+        break;
+      }
+      stream_len |= auxUW8;
+      if (fread(payload, 1, stream_len, inp) < stream_len) {
+        printf("last payload is corrupted\n");
+        break;
+      }
+    } else {
+      while (stream_len == 0) {
+        int stream_len_int;
+
+        // Read 10 ms speech block
+        endfile = readframe(shortdata, inp, samplesIn10Ms);
+        if (endfile) {
+          break;
+        }
+        cur_framesmpls += samplesIn10Ms;
+
+        //-------- iSAC encoding ---------
+        stream_len_int = WebRtcIsac_Encode(ISAC_main_inst, shortdata, payload);
+
+        if (stream_len_int < 0) {
+          // exit if returned with error
+          // errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+          fprintf(stderr, "\nError in encoder\n");
+          getc(stdin);
+          exit(EXIT_FAILURE);
+        }
+        stream_len = (size_t)stream_len_int;
+      }
+      //===================================================================
+      if (endfile) {
+        break;
+      }
+
+      rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, payloadRCU);
+      if (rcuStreamLen < 0) {
+        fprintf(stderr, "\nError getting RED payload\n");
+        getc(stdin);
         exit(EXIT_FAILURE);
-    }
+      }
 
-    framecnt = 0;
-    endfile     = 0;
-
-    /* Initialize encoder and decoder */
-    if(WebRtcIsac_EncoderInit(ISAC_main_inst, codingMode) < 0)
-    {
-        printf("cannot initialize encoder\n");
+      get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
+                       sampFreqKHz * 1000, sampFreqKHz * 1000);
+      if (WebRtcIsac_UpdateBwEstimate(
+              ISAC_main_inst, payload, stream_len, packetData.rtp_number,
+              packetData.sample_count, packetData.arrival_time) < 0) {
+        printf(" BWE Error at client\n");
         return -1;
+      }
     }
-    if(WebRtcIsac_DecoderInit(ISAC_main_inst) < 0)
-    {
-        printf("cannot initialize decoder\n");
+
+    if (endfile) {
+      break;
+    }
+
+    maxStreamLen = (stream_len > maxStreamLen) ? stream_len : maxStreamLen;
+    packetCntr++;
+
+    hist[stream_len]++;
+    if (averageFile != NULL) {
+      tmpSumStreamLen += stream_len;
+      if (packetCntr == 100) {
+        // kbps
+        fprintf(averageFile, "%8.3f ",
+                tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
+        packetCntr = 0;
+        tmpSumStreamLen = 0;
+      }
+    }
+
+    if (onlyEncode) {
+      uint8_t auxUW8;
+      auxUW8 = (uint8_t)(((stream_len & 0x7F00) >> 8) & 0xFF);
+      if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) {
         return -1;
-    }
+      }
 
-    //{
-    //    int32_t b1, b2;
-    //    FILE* fileID = fopen("GetBNTest.txt", "w");
-    //    b2 = 32100;
-    //    while(b2 <= 52000)
-    //    {
-    //        WebRtcIsac_Control(ISAC_main_inst, b2, frameSize);
-    //        WebRtcIsac_GetUplinkBw(ISAC_main_inst, &b1);
-    //        fprintf(fileID, "%5d %5d\n", b2, b1);
-    //        b2 += 10;
-    //    }
-    //}
-
-    if(codingMode == 1)
-    {
-        if(WebRtcIsac_Control(ISAC_main_inst, bottleneck, frameSize) < 0)
-        {
-            printf("cannot set bottleneck\n");
-            return -1;
-        }
-    }
-    else
-    {
-        if(WebRtcIsac_ControlBwe(ISAC_main_inst, 15000, 30, 1) < 0)
-        {
-            printf("cannot configure BWE\n");
-            return -1;
-        }
-    }
-
-    if(WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadLimit) < 0)
-    {
-        printf("cannot set maximum payload size %d.\n", payloadLimit);
+      auxUW8 = (uint8_t)(stream_len & 0xFF);
+      if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) {
         return -1;
+      }
+      if (fwrite(payload, 1, stream_len, outp) != stream_len) {
+        return -1;
+      }
+    } else {
+      //======================= iSAC decoding ===========================
+
+      if ((rand() % 100) < packetLossPercent) {
+        declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, payloadRCU,
+                                      (size_t)rcuStreamLen, decoded,
+                                      speechType);
+        lostPacketCntr++;
+      } else {
+        declen = WebRtcIsac_Decode(ISAC_main_inst, payload, stream_len, decoded,
+                                   speechType);
+      }
+      if (declen <= 0) {
+        // errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
+        fprintf(stderr, "\nError in decoder.\n");
+        getc(stdin);
+        exit(1);
+      }
+
+      // Write decoded speech frame to file
+      if (fwrite(decoded, sizeof(int16_t), declen, outp) != (size_t)declen) {
+        return -1;
+      }
+      cur_framesmpls = declen;
     }
-
-    if (rateLimit < 106800) {
-        if(WebRtcIsac_SetMaxRate(ISAC_main_inst, rateLimit) < 0)
-        {
-            printf("cannot set the maximum rate %d.\n", rateLimit);
-            return -1;
-        }
+    // Update Statistics
+    framecnt++;
+    totalsmpls += cur_framesmpls;
+    if (stream_len > 0) {
+      totalbits += 8 * stream_len;
     }
-
-    //=====================================
-//#ifdef HAVE_DEBUG_INFO
-//    if(setupDebugStruct(&debugInfo) < 0)
-//    {
-//        exit(1);
-//    }
-//#endif
-
-    while (endfile == 0)
-    {
-        fprintf(stderr,"  \rframe = %7li", framecnt);
-
-        //============== Readind from the file and encoding =================
-        cur_framesmpls = 0;
-        stream_len = 0;
-
-
-		if(onlyDecode)
-		{
-			uint8_t auxUW8;
-                        size_t auxSizet;
-			if(fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1)
-			{
-				break;
-			}
-			stream_len = ((uint8_t)auxUW8) << 8;
-			if(fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1)
-			{
-				break;
-			}
-			stream_len |= (uint16_t)auxUW8;
-                        auxSizet = (size_t)stream_len;
-                        if(fread(payload, 1, auxSizet, inp) < auxSizet)
-			{
-				printf("last payload is corrupted\n");
-				break;
-			}
-		}
-		else
-		{
-			while(stream_len == 0)
-			{
-				// Read 10 ms speech block
-				endfile = readframe(shortdata, inp, samplesIn10Ms);
-				if(endfile)
-				{
-					break;
-				}
-				cur_framesmpls += samplesIn10Ms;
-
-				//-------- iSAC encoding ---------
-                                stream_len = WebRtcIsac_Encode(
-                                        ISAC_main_inst,
-                                        shortdata,
-                                        payload);
-
-				if(stream_len < 0)
-				{
-					// exit if returned with error
-					//errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-					fprintf(stderr,"\nError in encoder\n");
-					getc(stdin);
-					exit(EXIT_FAILURE);
-				}
-
-
-			}
-			//===================================================================
-			if(endfile)
-			{
-				break;
-			}
-
-                        rcuStreamLen = WebRtcIsac_GetRedPayload(
-                            ISAC_main_inst, payloadRCU);
-
-			get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
-				sampFreqKHz * 1000, sampFreqKHz * 1000);
-			if(WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
-                                                       payload,
-                                                       stream_len,
-                                                       packetData.rtp_number,
-                                                       packetData.sample_count,
-                                                       packetData.arrival_time)
-                           < 0)
-			{
-				printf(" BWE Error at client\n");
-				return -1;
-			}
-		}
-
-        if(endfile)
-        {
-            break;
-        }
-
-        maxStreamLen = (stream_len > maxStreamLen)? stream_len:maxStreamLen;
-        packetCntr++;
-
-        hist[stream_len]++;
-        if(averageFile != NULL)
-        {
-            tmpSumStreamLen += stream_len;
-            if(packetCntr == 100)
-            {
-                // kbps
-                fprintf(averageFile, "%8.3f ", (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
-                packetCntr = 0;
-                tmpSumStreamLen = 0;
-            }
-        }
-
-		if(onlyEncode)
-		{
-                  uint8_t auxUW8;
-                  auxUW8 = (uint8_t)(((stream_len & 0x7F00) >> 8) & 0xFF);
-                  if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) {
-                    return -1;
-                  }
-
-                  auxUW8 = (uint8_t)(stream_len & 0xFF);
-                  if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) {
-                    return -1;
-                  }
-                  if (fwrite(payload, 1, stream_len,
-                             outp) != (size_t)stream_len) {
-                    return -1;
-                  }
-		}
-		else
-		{
-
-			//======================= iSAC decoding ===========================
-
-			if((rand() % 100) < packetLossPercent)
-			{
-                                declen = WebRtcIsac_DecodeRcu(
-                                    ISAC_main_inst,
-                                    payloadRCU,
-                                    rcuStreamLen,
-                                    decoded,
-                                    speechType);
-				lostPacketCntr++;
-			}
-			else
-			{
-                                declen = WebRtcIsac_Decode(
-                                    ISAC_main_inst,
-                                    payload,
-                                    stream_len,
-                                    decoded,
-                                    speechType);
-                        }
-                        if(declen <= 0)
-			{
-				//errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-				fprintf(stderr,"\nError in decoder.\n");
-				getc(stdin);
-				exit(1);
-			}
-
-			// Write decoded speech frame to file
-                        if (fwrite(decoded, sizeof(int16_t),
-                                   declen, outp) != (size_t)declen) {
-                          return -1;
-                        }
-			cur_framesmpls = declen;
-		}
-        // Update Statistics
-        framecnt++;
-        totalsmpls += cur_framesmpls;
-        if(stream_len > 0)
-        {
-            totalbits += 8 * stream_len;
-        }
-        if(rcuStreamLen > 0)
-        {
-            totalBitsRCU += 8 * rcuStreamLen;
-        }
+    if (rcuStreamLen > 0) {
+      totalBitsRCU += 8 * rcuStreamLen;
     }
+  }
 
-    rate =    ((double)totalbits    * (sampFreqKHz)) / (double)totalsmpls;
-    rateRCU = ((double)totalBitsRCU * (sampFreqKHz)) / (double)totalsmpls;
+  rate = ((double)totalbits * (sampFreqKHz)) / (double)totalsmpls;
+  rateRCU = ((double)totalBitsRCU * (sampFreqKHz)) / (double)totalsmpls;
 
-    printf("\n\n");
-    printf("Sampling Rate......................... %d kHz\n", sampFreqKHz);
-    printf("Payload Limit......................... %d bytes \n", payloadLimit);
-    printf("Rate Limit............................ %d bits/sec \n", rateLimit);
+  printf("\n\n");
+  printf("Sampling Rate............... %d kHz\n", sampFreqKHz);
+  printf("Payload Limit............... %d bytes \n", payloadLimit);
+  printf("Rate Limit.................. %d bits/sec \n", rateLimit);
 
 #ifdef WIN32
 #ifdef HAVE_DEBUG_INFO
-    rateLB = ((double)debugInfo.lbBytes * 8. *
-              (sampFreqKHz)) / (double)totalsmpls;
-    rateUB = ((double)debugInfo.ubBytes * 8. *
-              (sampFreqKHz)) / (double)totalsmpls;
+  rateLB =
+      ((double)debugInfo.lbBytes * 8. * (sampFreqKHz)) / (double)totalsmpls;
+  rateUB =
+      ((double)debugInfo.ubBytes * 8. * (sampFreqKHz)) / (double)totalsmpls;
 #endif
 
-    fprintf(bitrateFile, "%d  %10u     %d     %6.3f  %6.3f    %6.3f\n",
-        sampFreqKHz,
-        framecnt,
-        bottleneck,
-        rateLB,
-        rateUB,
-        rate);
-    fclose(bitrateFile);
-#endif   // WIN32
+  fprintf(bitrateFile, "%d  %10u     %d     %6.3f  %6.3f    %6.3f\n",
+          sampFreqKHz, framecnt, bottleneck, rateLB, rateUB, rate);
+  fclose(bitrateFile);
+#endif  // WIN32
 
-    printf("\n");
-    printf("Measured bit-rate..................... %0.3f kbps\n", rate);
-    printf("Measured RCU bit-ratre................ %0.3f kbps\n", rateRCU);
-    printf("Maximum bit-rate/payloadsize.......... %0.3f / %d\n",
-        maxStreamLen * 8 / 0.03, maxStreamLen);
-    printf("Measured packet-loss.................. %0.1f%% \n",
-        100.0f * (float)lostPacketCntr / (float)packetCntr);
+  printf("\n");
+  printf("Measured bit-rate........... %0.3f kbps\n", rate);
+  printf("Measured RCU bit-ratre...... %0.3f kbps\n", rateRCU);
+  printf("Maximum bit-rate/payloadsize %0.3f / %" PRIuS "\n",
+         maxStreamLen * 8 / 0.03, maxStreamLen);
+  printf("Measured packet-loss........ %0.1f%% \n",
+         100.0f * (float)lostPacketCntr / (float)packetCntr);
 
-//#ifdef HAVE_DEBUG_INFO
-//    printf("Measured lower-band bit-rate.......... %0.3f kbps (%.0f%%)\n",
-//        rateLB, (double)(rateLB) * 100. /(double)(rate));
-//    printf("Measured upper-band bit-rate.......... %0.3f kbps (%.0f%%)\n",
-//        rateUB, (double)(rateUB) * 100. /(double)(rate));
-//
-//    printf("Maximum payload lower-band............ %d bytes (%0.3f kbps)\n",
-//        debugInfo.maxPayloadLB, debugInfo.maxPayloadLB * 8.0 / 0.03);
-//    printf("Maximum payload upper-band............ %d bytes (%0.3f kbps)\n",
-//        debugInfo.maxPayloadUB, debugInfo.maxPayloadUB * 8.0 / 0.03);
-//#endif
+  // #ifdef HAVE_DEBUG_INFO
+  // printf("Measured lower-band bit-rate %0.3f kbps (%.0f%%)\n",
+  //        rateLB, (double)(rateLB) * 100. /(double)(rate));
+  // printf("Measured upper-band bit-rate %0.3f kbps (%.0f%%)\n",
+  //        rateUB, (double)(rateUB) * 100. /(double)(rate));
+  //
+  // printf("Maximum payload lower-band.. %d bytes (%0.3f kbps)\n",
+  //        debugInfo.maxPayloadLB, debugInfo.maxPayloadLB * 8.0 / 0.03);
+  // printf("Maximum payload upper-band.. %d bytes (%0.3f kbps)\n",
+  //        debugInfo.maxPayloadUB, debugInfo.maxPayloadUB * 8.0 / 0.03);
+  // #endif
 
-    printf("\n");
+  printf("\n");
 
-    /* Runtime statistics */
+/* Runtime statistics */
 #ifdef WIN32
-    runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
-    length_file = ((double)framecnt*(double)declen/(sampFreqKHz*1000));
-    printf("Length of speech file................ %.1f s\n", length_file);
-    printf("Time to run iSAC..................... %.2f s (%.2f %% of realtime)\n\n",
-        runtime, (100*runtime/length_file));
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC - starttime);
+  length_file = ((double)framecnt * (double)declen / (sampFreqKHz * 1000));
+  printf("Length of speech file....... %.1f s\n", length_file);
+  printf("Time to run iSAC............ %.2f s (%.2f %% of realtime)\n\n",
+         runtime, (100 * runtime / length_file));
 #endif
-    printf("\n\n_______________________________________________\n");
+  printf("\n\n_______________________________________________\n");
 
-    if(histFile != NULL)
-    {
-        int n;
-        for(n = 0; n < 600; n++)
-        {
-            fprintf(histFile, "%6d ", hist[n]);
-        }
-        fprintf(histFile, "\n");
-        fclose(histFile);
+  if (histFile != NULL) {
+    int n;
+    for (n = 0; n < 600; n++) {
+      fprintf(histFile, "%6d ", hist[n]);
     }
-    if(averageFile != NULL)
-    {
-        if(packetCntr > 0)
-        {
-            fprintf(averageFile, "%8.3f ", (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
-        }
-        fprintf(averageFile, "\n");
-        fclose(averageFile);
+    fprintf(histFile, "\n");
+    fclose(histFile);
+  }
+  if (averageFile != NULL) {
+    if (packetCntr > 0) {
+      fprintf(averageFile, "%8.3f ",
+              tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
     }
+    fprintf(averageFile, "\n");
+    fclose(averageFile);
+  }
 
-    fclose(inp);
-    fclose(outp);
-
-    WebRtcIsac_Free(ISAC_main_inst);
+  fclose(inp);
+  fclose(outp);
 
+  WebRtcIsac_Free(ISAC_main_inst);
 
 #ifdef CHANGE_OUTPUT_NAME
-    {
-        char* p;
-        char myExt[50];
-        char bitRateStr[10];
-        char newOutName[500];
-        strcpy(newOutName, outname);
+  {
+    char* p;
+    char myExt[50];
+    char bitRateStr[10];
+    char newOutName[500];
+    strcpy(newOutName, outname);
 
-        myExt[0] = '\0';
-        p = strchr(newOutName, '.');
-        if(p != NULL)
-        {
-            strcpy(myExt, p);
-            *p = '_';
-            p++;
-            *p = '\0';
-        }
-        else
-        {
-            strcat(newOutName, "_");
-        }
-        sprintf(bitRateStr, "%0.0fkbps", rate);
-        strcat(newOutName, bitRateStr);
-        strcat(newOutName, myExt);
-        rename(outname, newOutName);
+    myExt[0] = '\0';
+    p = strchr(newOutName, '.');
+    if (p != NULL) {
+      strcpy(myExt, p);
+      *p = '_';
+      p++;
+      *p = '\0';
+    } else {
+      strcat(newOutName, "_");
     }
+    sprintf(bitRateStr, "%0.0fkbps", rate);
+    strcat(newOutName, bitRateStr);
+    strcat(newOutName, myExt);
+    rename(outname, newOutName);
+  }
 #endif
-    exit(0);
+  exit(0);
 }
 
-
 #ifdef HAVE_DEBUG_INFO
-int setupDebugStruct(debugStruct* str)
-{
-    str->prevPacketLost = 0;
-    str->currPacketLost = 0;
+int setupDebugStruct(debugStruct* str) {
+  str->prevPacketLost = 0;
+  str->currPacketLost = 0;
 
-    OPEN_FILE_WB(str->res0to4FilePtr,     "Res0to4.dat");
-    OPEN_FILE_WB(str->res4to8FilePtr,     "Res4to8.dat");
-    OPEN_FILE_WB(str->res8to12FilePtr,    "Res8to12.dat");
-    OPEN_FILE_WB(str->res8to16FilePtr,    "Res8to16.dat");
+  OPEN_FILE_WB(str->res0to4FilePtr, "Res0to4.dat");
+  OPEN_FILE_WB(str->res4to8FilePtr, "Res4to8.dat");
+  OPEN_FILE_WB(str->res8to12FilePtr, "Res8to12.dat");
+  OPEN_FILE_WB(str->res8to16FilePtr, "Res8to16.dat");
 
-    OPEN_FILE_WB(str->res0to4DecFilePtr,  "Res0to4Dec.dat");
-    OPEN_FILE_WB(str->res4to8DecFilePtr,  "Res4to8Dec.dat");
-    OPEN_FILE_WB(str->res8to12DecFilePtr, "Res8to12Dec.dat");
-    OPEN_FILE_WB(str->res8to16DecFilePtr, "Res8to16Dec.dat");
+  OPEN_FILE_WB(str->res0to4DecFilePtr, "Res0to4Dec.dat");
+  OPEN_FILE_WB(str->res4to8DecFilePtr, "Res4to8Dec.dat");
+  OPEN_FILE_WB(str->res8to12DecFilePtr, "Res8to12Dec.dat");
+  OPEN_FILE_WB(str->res8to16DecFilePtr, "Res8to16Dec.dat");
 
-    OPEN_FILE_WB(str->in0to4FilePtr,      "in0to4.dat");
-    OPEN_FILE_WB(str->in4to8FilePtr,      "in4to8.dat");
-    OPEN_FILE_WB(str->in8to12FilePtr,     "in8to12.dat");
-    OPEN_FILE_WB(str->in8to16FilePtr,     "in8to16.dat");
+  OPEN_FILE_WB(str->in0to4FilePtr, "in0to4.dat");
+  OPEN_FILE_WB(str->in4to8FilePtr, "in4to8.dat");
+  OPEN_FILE_WB(str->in8to12FilePtr, "in8to12.dat");
+  OPEN_FILE_WB(str->in8to16FilePtr, "in8to16.dat");
 
-    OPEN_FILE_WB(str->out0to4FilePtr,     "out0to4.dat");
-    OPEN_FILE_WB(str->out4to8FilePtr,     "out4to8.dat");
-    OPEN_FILE_WB(str->out8to12FilePtr,    "out8to12.dat");
-    OPEN_FILE_WB(str->out8to16FilePtr,    "out8to16.dat");
-    OPEN_FILE_WB(str->fftFilePtr,         "riFFT.dat");
-    OPEN_FILE_WB(str->fftDecFilePtr,      "riFFTDec.dat");
+  OPEN_FILE_WB(str->out0to4FilePtr, "out0to4.dat");
+  OPEN_FILE_WB(str->out4to8FilePtr, "out4to8.dat");
+  OPEN_FILE_WB(str->out8to12FilePtr, "out8to12.dat");
+  OPEN_FILE_WB(str->out8to16FilePtr, "out8to16.dat");
+  OPEN_FILE_WB(str->fftFilePtr, "riFFT.dat");
+  OPEN_FILE_WB(str->fftDecFilePtr, "riFFTDec.dat");
 
-    OPEN_FILE_WB(str->arrivalTime,        NULL/*"ArivalTime.dat"*/);
-    str->lastArrivalTime = 0;
+  OPEN_FILE_WB(str->arrivalTime, NULL /*"ArivalTime.dat"*/);
+  str->lastArrivalTime = 0;
 
-    str->maxPayloadLB = 0;
-    str->maxPayloadUB = 0;
-    str->lbBytes = 0;
-    str->ubBytes = 0;
+  str->maxPayloadLB = 0;
+  str->maxPayloadUB = 0;
+  str->lbBytes = 0;
+  str->ubBytes = 0;
 
-    return 0;
+  return 0;
 };
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c
index 0a2256a036..d9c4332123 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c
@@ -135,7 +135,7 @@ readParamString(
 void
 get_arrival_time(
     int              current_framesamples,   /* samples */
-    int              packet_size,            /* bytes */
+    size_t           packet_size,            /* bytes */
     int              bottleneck,             /* excluding headers; bits/s */
     BottleNeckModel* BN_data,
     short            senderSampFreqHz,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
index f9fba94315..1bb6d295b4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
@@ -99,7 +99,7 @@ extern "C" {
 
   void get_arrival_time(
       int              current_framesamples,   /* samples */
-      int              packet_size,            /* bytes */
+      size_t           packet_size,            /* bytes */
       int              bottleneck,             /* excluding headers; bits/s */
       BottleNeckModel* BN_data,
       short            senderSampFreqHz,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/unittest.cc
new file mode 100644
index 0000000000..890397e587
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/isac/unittest.cc
@@ -0,0 +1,254 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <numeric>
+#include <sstream>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kIsacNumberOfSamples = 32 * 60;  // 60 ms at 32 kHz
+
+std::vector<int16_t> LoadSpeechData() {
+  webrtc::test::InputAudioFile input_file(
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
+  std::vector<int16_t> speech_data(kIsacNumberOfSamples);
+  input_file.Read(kIsacNumberOfSamples, speech_data.data());
+  return speech_data;
+}
+
+template <typename T>
+IsacBandwidthInfo GetBwInfo(typename T::instance_type* inst) {
+  IsacBandwidthInfo bi;
+  T::GetBandwidthInfo(inst, &bi);
+  EXPECT_TRUE(bi.in_use);
+  return bi;
+}
+
+// Encodes one packet. Returns the packet duration in milliseconds.
+template <typename T>
+int EncodePacket(typename T::instance_type* inst,
+                 const IsacBandwidthInfo* bi,
+                 const int16_t* speech_data,
+                 rtc::Buffer* output) {
+  output->SetSize(1000);
+  for (int duration_ms = 10;; duration_ms += 10) {
+    if (bi)
+      T::SetBandwidthInfo(inst, bi);
+    int encoded_bytes = T::Encode(inst, speech_data, output->data());
+    if (encoded_bytes > 0 || duration_ms >= 60) {
+      EXPECT_GT(encoded_bytes, 0);
+      EXPECT_LE(static_cast<size_t>(encoded_bytes), output->size());
+      output->SetSize(encoded_bytes);
+      return duration_ms;
+    }
+  }
+}
+
+template <typename T>
+std::vector<int16_t> DecodePacket(typename T::instance_type* inst,
+                                  const rtc::Buffer& encoded) {
+  std::vector<int16_t> decoded(kIsacNumberOfSamples);
+  int16_t speech_type;
+  int nsamples = T::DecodeInternal(inst, encoded.data(), encoded.size(),
+                                   &decoded.front(), &speech_type);
+  EXPECT_GT(nsamples, 0);
+  EXPECT_LE(static_cast<size_t>(nsamples), decoded.size());
+  decoded.resize(nsamples);
+  return decoded;
+}
+
+class BoundedCapacityChannel final {
+ public:
+  BoundedCapacityChannel(int sample_rate_hz, int rate_bits_per_second)
+      : current_time_rtp_(0),
+        channel_rate_bytes_per_sample_(rate_bits_per_second /
+                                       (8.0 * sample_rate_hz)) {}
+
+  // Simulate sending the given number of bytes at the given RTP time. Returns
+  // the new current RTP time after the sending is done.
+  int Send(int send_time_rtp, int nbytes) {
+    current_time_rtp_ = std::max(current_time_rtp_, send_time_rtp) +
+                        nbytes / channel_rate_bytes_per_sample_;
+    return current_time_rtp_;
+  }
+
+ private:
+  int current_time_rtp_;
+  // The somewhat strange unit for channel rate, bytes per sample, is because
+  // RTP time is measured in samples:
+  const double channel_rate_bytes_per_sample_;
+};
+
+// Test that the iSAC encoder produces identical output whether or not we use a
+// conjoined encoder+decoder pair or a separate encoder and decoder that
+// communicate BW estimation info explicitly.
+template <typename T, bool adaptive>
+void TestGetSetBandwidthInfo(const int16_t* speech_data,
+                             int rate_bits_per_second,
+                             int sample_rate_hz,
+                             int frame_size_ms) {
+  const int bit_rate = 32000;
+
+  // Conjoined encoder/decoder pair:
+  typename T::instance_type* encdec;
+  ASSERT_EQ(0, T::Create(&encdec));
+  ASSERT_EQ(0, T::EncoderInit(encdec, adaptive ? 0 : 1));
+  T::DecoderInit(encdec);
+  ASSERT_EQ(0, T::SetEncSampRate(encdec, sample_rate_hz));
+  if (adaptive)
+    ASSERT_EQ(0, T::ControlBwe(encdec, bit_rate, frame_size_ms, false));
+  else
+    ASSERT_EQ(0, T::Control(encdec, bit_rate, frame_size_ms));
+
+  // Disjoint encoder/decoder pair:
+  typename T::instance_type* enc;
+  ASSERT_EQ(0, T::Create(&enc));
+  ASSERT_EQ(0, T::EncoderInit(enc, adaptive ? 0 : 1));
+  ASSERT_EQ(0, T::SetEncSampRate(enc, sample_rate_hz));
+  if (adaptive)
+    ASSERT_EQ(0, T::ControlBwe(enc, bit_rate, frame_size_ms, false));
+  else
+    ASSERT_EQ(0, T::Control(enc, bit_rate, frame_size_ms));
+  typename T::instance_type* dec;
+  ASSERT_EQ(0, T::Create(&dec));
+  T::DecoderInit(dec);
+  T::SetInitialBweBottleneck(dec, bit_rate);
+  T::SetEncSampRateInDecoder(dec, sample_rate_hz);
+
+  // 0. Get initial BW info from decoder.
+  auto bi = GetBwInfo<T>(dec);
+
+  BoundedCapacityChannel channel1(sample_rate_hz, rate_bits_per_second),
+      channel2(sample_rate_hz, rate_bits_per_second);
+
+  int elapsed_time_ms = 0;
+  for (int i = 0; elapsed_time_ms < 10000; ++i) {
+    std::ostringstream ss;
+    ss << " i = " << i;
+    SCOPED_TRACE(ss.str());
+
+    // 1. Encode 3 * 10 ms or 6 * 10 ms. The separate encoder is given the BW
+    // info before each encode call.
+    rtc::Buffer bitstream1, bitstream2;
+    int duration1_ms =
+        EncodePacket<T>(encdec, nullptr, speech_data, &bitstream1);
+    int duration2_ms = EncodePacket<T>(enc, &bi, speech_data, &bitstream2);
+    EXPECT_EQ(duration1_ms, duration2_ms);
+    if (adaptive)
+      EXPECT_TRUE(duration1_ms == 30 || duration1_ms == 60);
+    else
+      EXPECT_EQ(frame_size_ms, duration1_ms);
+    ASSERT_EQ(bitstream1.size(), bitstream2.size());
+    EXPECT_EQ(bitstream1, bitstream2);
+
+    // 2. Deliver the encoded data to the decoders.
+    const int send_time = elapsed_time_ms * (sample_rate_hz / 1000);
+    EXPECT_EQ(0, T::UpdateBwEstimate(
+                     encdec, bitstream1.data(), bitstream1.size(), i, send_time,
+                     channel1.Send(send_time, bitstream1.size())));
+    EXPECT_EQ(0, T::UpdateBwEstimate(
+                     dec, bitstream2.data(), bitstream2.size(), i, send_time,
+                     channel2.Send(send_time, bitstream2.size())));
+
+    // 3. Decode, and get new BW info from the separate decoder.
+    ASSERT_EQ(0, T::SetDecSampRate(encdec, sample_rate_hz));
+    ASSERT_EQ(0, T::SetDecSampRate(dec, sample_rate_hz));
+    auto decoded1 = DecodePacket<T>(encdec, bitstream1);
+    auto decoded2 = DecodePacket<T>(dec, bitstream2);
+    EXPECT_EQ(decoded1, decoded2);
+    bi = GetBwInfo<T>(dec);
+
+    elapsed_time_ms += duration1_ms;
+  }
+
+  EXPECT_EQ(0, T::Free(encdec));
+  EXPECT_EQ(0, T::Free(enc));
+  EXPECT_EQ(0, T::Free(dec));
+}
+
+enum class IsacType { Fix, Float };
+
+std::ostream& operator<<(std::ostream& os, IsacType t) {
+  os << (t == IsacType::Fix ? "fix" : "float");
+  return os;
+}
+
+struct IsacTestParam {
+  IsacType isac_type;
+  bool adaptive;
+  int channel_rate_bits_per_second;
+  int sample_rate_hz;
+  int frame_size_ms;
+
+  friend std::ostream& operator<<(std::ostream& os, const IsacTestParam& itp) {
+    os << '{' << itp.isac_type << ','
+       << (itp.adaptive ? "adaptive" : "nonadaptive") << ','
+       << itp.channel_rate_bits_per_second << ',' << itp.sample_rate_hz << ','
+       << itp.frame_size_ms << '}';
+    return os;
+  }
+};
+
+class IsacCommonTest : public testing::TestWithParam<IsacTestParam> {};
+
+}  // namespace
+
+TEST_P(IsacCommonTest, GetSetBandwidthInfo) {
+  auto p = GetParam();
+  auto test_fun = [p] {
+    if (p.isac_type == IsacType::Fix) {
+      if (p.adaptive)
+        return TestGetSetBandwidthInfo<IsacFix, true>;
+      else
+        return TestGetSetBandwidthInfo<IsacFix, false>;
+    } else {
+      if (p.adaptive)
+        return TestGetSetBandwidthInfo<IsacFloat, true>;
+      else
+        return TestGetSetBandwidthInfo<IsacFloat, false>;
+    }
+  }();
+  test_fun(LoadSpeechData().data(), p.channel_rate_bits_per_second,
+           p.sample_rate_hz, p.frame_size_ms);
+}
+
+std::vector<IsacTestParam> TestCases() {
+  static const IsacType types[] = {IsacType::Fix, IsacType::Float};
+  static const bool adaptives[] = {true, false};
+  static const int channel_rates[] = {12000, 15000, 19000, 22000};
+  static const int sample_rates[] = {16000, 32000};
+  static const int frame_sizes[] = {30, 60};
+  std::vector<IsacTestParam> cases;
+  for (IsacType type : types)
+    for (bool adaptive : adaptives)
+      for (int channel_rate : channel_rates)
+        for (int sample_rate : sample_rates)
+          if (!(type == IsacType::Fix && sample_rate == 32000))
+            for (int frame_size : frame_sizes)
+              if (!(sample_rate == 32000 && frame_size == 60))
+                cases.push_back(
+                    {type, adaptive, channel_rate, sample_rate, frame_size});
+  return cases;
+}
+
+INSTANTIATE_TEST_CASE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 25fd7a837d..66adde4be1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -17,23 +17,33 @@
 
 namespace webrtc {
 
-class MockAudioEncoder : public AudioEncoder {
+class MockAudioEncoder final : public AudioEncoder {
  public:
-  virtual ~MockAudioEncoder() { Die(); }
+  ~MockAudioEncoder() override { Die(); }
   MOCK_METHOD0(Die, void());
-  MOCK_CONST_METHOD0(SampleRateHz, int());
-  MOCK_CONST_METHOD0(NumChannels, int());
+  MOCK_METHOD1(Mark, void(std::string desc));
   MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
-  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, int());
-  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, int());
-  MOCK_METHOD1(SetTargetBitrate, void(int));
-  MOCK_METHOD1(SetProjectedPacketLossRate, void(double));
+  MOCK_CONST_METHOD0(SampleRateHz, int());
+  MOCK_CONST_METHOD0(NumChannels, size_t());
+  MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
+  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
+  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
+  MOCK_CONST_METHOD0(GetTargetBitrate, int());
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD4(EncodeInternal,
                EncodedInfo(uint32_t timestamp,
-                           const int16_t* audio,
+                           rtc::ArrayView<const int16_t> audio,
                            size_t max_encoded_bytes,
                            uint8_t* encoded));
+  MOCK_METHOD0(Reset, void());
+  MOCK_METHOD1(SetFec, bool(bool enable));
+  MOCK_METHOD1(SetDtx, bool(bool enable));
+  MOCK_METHOD1(SetApplication, bool(Application application));
+  MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz));
+  MOCK_METHOD1(SetProjectedPacketLossRate, void(double fraction));
+  MOCK_METHOD1(SetTargetBitrate, void(int target_bps));
+  MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
+  MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000..f64e811afe
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
+    : channels_(num_channels) {
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
+  WebRtcOpus_DecoderCreate(&dec_state_, channels_);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpus::~AudioDecoderOpus() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
+                                              size_t encoded_len,
+                                              int sample_rate_hz,
+                                              int16_t* decoded,
+                                              SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpus::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
+                                              size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len);
+}
+
+bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
+                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+size_t AudioDecoderOpus::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
new file mode 100644
index 0000000000..af32a84512
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class AudioDecoderOpus final : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpus(size_t num_channels);
+  ~AudioDecoderOpus() override;
+
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 941e635b17..707d6c2488 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -8,55 +8,74 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
 namespace webrtc {
 
 namespace {
 
+const int kSampleRateHz = 48000;
 const int kMinBitrateBps = 500;
 const int kMaxBitrateBps = 512000;
 
-// TODO(tlegrand): Remove this code when we have proper APIs to set the
-// complexity at a higher level.
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
-// If we are on Android, iOS and/or ARM, use a lower complexity setting as
-// default, to save encoder complexity.
-const int kDefaultComplexity = 5;
-#else
-const int kDefaultComplexity = 9;
-#endif
-
-// We always encode at 48 kHz.
-const int kSampleRateHz = 48000;
-
-int16_t ClampInt16(size_t x) {
-  return static_cast<int16_t>(
-      std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())));
+AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderOpus::Config config;
+  config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
+  config.num_channels = codec_inst.channels;
+  config.bitrate_bps = codec_inst.rate;
+  config.payload_type = codec_inst.pltype;
+  config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
+                                                : AudioEncoderOpus::kAudio;
+  return config;
 }
 
-int16_t CastInt16(size_t x) {
-  DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()));
-  return static_cast<int16_t>(x);
+// Optimize the loss rate to configure Opus. Basically, optimized loss rate is
+// the input loss rate rounded down to various levels, because a robustly good
+// audio quality is achieved by lowering the packet loss down.
+// Additionally, to prevent toggling, margins are used, i.e., when jumping to
+// a loss rate from below, a higher threshold is used than jumping to the same
+// level from above.
+double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) {
+  RTC_DCHECK_GE(new_loss_rate, 0.0);
+  RTC_DCHECK_LE(new_loss_rate, 1.0);
+  RTC_DCHECK_GE(old_loss_rate, 0.0);
+  RTC_DCHECK_LE(old_loss_rate, 1.0);
+  const double kPacketLossRate20 = 0.20;
+  const double kPacketLossRate10 = 0.10;
+  const double kPacketLossRate5 = 0.05;
+  const double kPacketLossRate1 = 0.01;
+  const double kLossRate20Margin = 0.02;
+  const double kLossRate10Margin = 0.01;
+  const double kLossRate5Margin = 0.01;
+  if (new_loss_rate >=
+      kPacketLossRate20 +
+          kLossRate20Margin *
+              (kPacketLossRate20 - old_loss_rate > 0 ? 1 : -1)) {
+    return kPacketLossRate20;
+  } else if (new_loss_rate >=
+             kPacketLossRate10 +
+                 kLossRate10Margin *
+                     (kPacketLossRate10 - old_loss_rate > 0 ? 1 : -1)) {
+    return kPacketLossRate10;
+  } else if (new_loss_rate >=
+             kPacketLossRate5 +
+                 kLossRate5Margin *
+                     (kPacketLossRate5 - old_loss_rate > 0 ? 1 : -1)) {
+    return kPacketLossRate5;
+  } else if (new_loss_rate >= kPacketLossRate1) {
+    return kPacketLossRate1;
+  } else {
+    return 0.0;
+  }
 }
 
 }  // namespace
 
-AudioEncoderOpus::Config::Config()
-    : frame_size_ms(20),
-      num_channels(1),
-      payload_type(120),
-      application(kVoip),
-      bitrate_bps(64000),
-      fec_enabled(false),
-      max_playback_rate_hz(48000),
-      complexity(kDefaultComplexity),
-      dtx_enabled(false) {
-}
-
 bool AudioEncoderOpus::Config::IsOk() const {
   if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
     return false;
@@ -66,153 +85,172 @@ bool AudioEncoderOpus::Config::IsOk() const {
     return false;
   if (complexity < 0 || complexity > 10)
     return false;
-  if (dtx_enabled && application != kVoip)
-    return false;
   return true;
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const Config& config)
-    : num_10ms_frames_per_packet_(
-          rtc::CheckedDivExact(config.frame_size_ms, 10)),
-      num_channels_(config.num_channels),
-      payload_type_(config.payload_type),
-      application_(config.application),
-      dtx_enabled_(config.dtx_enabled),
-      samples_per_10ms_frame_(rtc::CheckedDivExact(kSampleRateHz, 100) *
-                              num_channels_),
-      packet_loss_rate_(0.0) {
-  CHECK(config.IsOk());
-  input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
-  CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_, application_));
-  SetTargetBitrate(config.bitrate_bps);
-  if (config.fec_enabled) {
-    CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
-  } else {
-    CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
-  }
-  CHECK_EQ(0,
-           WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
-  CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
-  if (config.dtx_enabled) {
-    CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
-  } else {
-    CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
-  }
+    : packet_loss_rate_(0.0), inst_(nullptr) {
+  RTC_CHECK(RecreateEncoderInstance(config));
 }
 
+AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
+    : AudioEncoderOpus(CreateConfig(codec_inst)) {}
+
 AudioEncoderOpus::~AudioEncoderOpus() {
-  CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+size_t AudioEncoderOpus::MaxEncodedBytes() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
 }
 
 int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-int AudioEncoderOpus::NumChannels() const {
-  return num_channels_;
+size_t AudioEncoderOpus::NumChannels() const {
+  return config_.num_channels;
 }
 
-size_t AudioEncoderOpus::MaxEncodedBytes() const {
-  // Calculate the number of bytes we expect the encoder to produce,
-  // then multiply by two to give a wide margin for error.
-  int frame_size_ms = num_10ms_frames_per_packet_ * 10;
-  int bytes_per_millisecond = bitrate_bps_ / (1000 * 8) + 1;
-  size_t approx_encoded_bytes =
-      static_cast<size_t>(frame_size_ms * bytes_per_millisecond);
-  return 2 * approx_encoded_bytes;
+size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
 }
 
-int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
-  return num_10ms_frames_per_packet_;
+size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
 }
 
-int AudioEncoderOpus::Max10MsFramesInAPacket() const {
-  return num_10ms_frames_per_packet_;
-}
-
-void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
-  bitrate_bps_ = std::max(std::min(bits_per_second, kMaxBitrateBps),
-                          kMinBitrateBps);
-  CHECK_EQ(WebRtcOpus_SetBitRate(inst_, bitrate_bps_), 0);
-}
-
-void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
-  DCHECK_GE(fraction, 0.0);
-  DCHECK_LE(fraction, 1.0);
-  // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
-  // the input loss rate rounded down to various levels, because a robustly good
-  // audio quality is achieved by lowering the packet loss down.
-  // Additionally, to prevent toggling, margins are used, i.e., when jumping to
-  // a loss rate from below, a higher threshold is used than jumping to the same
-  // level from above.
-  const double kPacketLossRate20 = 0.20;
-  const double kPacketLossRate10 = 0.10;
-  const double kPacketLossRate5 = 0.05;
-  const double kPacketLossRate1 = 0.01;
-  const double kLossRate20Margin = 0.02;
-  const double kLossRate10Margin = 0.01;
-  const double kLossRate5Margin = 0.01;
-  double opt_loss_rate;
-  if (fraction >=
-      kPacketLossRate20 +
-          kLossRate20Margin *
-              (kPacketLossRate20 - packet_loss_rate_ > 0 ? 1 : -1)) {
-    opt_loss_rate = kPacketLossRate20;
-  } else if (fraction >=
-             kPacketLossRate10 +
-                 kLossRate10Margin *
-                     (kPacketLossRate10 - packet_loss_rate_ > 0 ? 1 : -1)) {
-    opt_loss_rate = kPacketLossRate10;
-  } else if (fraction >=
-             kPacketLossRate5 +
-                 kLossRate5Margin *
-                     (kPacketLossRate5 - packet_loss_rate_ > 0 ? 1 : -1)) {
-    opt_loss_rate = kPacketLossRate5;
-  } else if (fraction >= kPacketLossRate1) {
-    opt_loss_rate = kPacketLossRate1;
-  } else {
-    opt_loss_rate = 0;
-  }
-
-  if (packet_loss_rate_ != opt_loss_rate) {
-    // Ask the encoder to change the target packet loss rate.
-    CHECK_EQ(WebRtcOpus_SetPacketLossRate(
-                 inst_, static_cast<int32_t>(opt_loss_rate * 100 + .5)),
-             0);
-    packet_loss_rate_ = opt_loss_rate;
-  }
+int AudioEncoderOpus::GetTargetBitrate() const {
+  return config_.bitrate_bps;
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
-  input_buffer_.insert(input_buffer_.end(), audio,
-                       audio + samples_per_10ms_frame_);
-  if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
-                              samples_per_10ms_frame_)) {
+  RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
-  CHECK_EQ(input_buffer_.size(),
-           static_cast<size_t>(num_10ms_frames_per_packet_) *
-           samples_per_10ms_frame_);
-  int16_t r = WebRtcOpus_Encode(
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+  int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
-      rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
-                           static_cast<int16_t>(num_channels_)),
-      ClampInt16(max_encoded_bytes), encoded);
-  CHECK_GE(r, 0);  // Fails only if fed invalid data.
+      rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+      rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
+  RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
-  info.encoded_bytes = r;
+  info.encoded_bytes = static_cast<size_t>(status);
   info.encoded_timestamp = first_timestamp_in_buffer_;
-  info.payload_type = payload_type_;
+  info.payload_type = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
-  info.speech = r > 0;
+  info.speech = (status > 0);
   return info;
 }
 
+void AudioEncoderOpus::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+bool AudioEncoderOpus::SetFec(bool enable) {
+  auto conf = config_;
+  conf.fec_enabled = enable;
+  return RecreateEncoderInstance(conf);
+}
+
+bool AudioEncoderOpus::SetDtx(bool enable) {
+  auto conf = config_;
+  conf.dtx_enabled = enable;
+  return RecreateEncoderInstance(conf);
+}
+
+bool AudioEncoderOpus::SetApplication(Application application) {
+  auto conf = config_;
+  switch (application) {
+    case Application::kSpeech:
+      conf.application = AudioEncoderOpus::kVoip;
+      break;
+    case Application::kAudio:
+      conf.application = AudioEncoderOpus::kAudio;
+      break;
+  }
+  return RecreateEncoderInstance(conf);
+}
+
+void AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) {
+  auto conf = config_;
+  conf.max_playback_rate_hz = frequency_hz;
+  RTC_CHECK(RecreateEncoderInstance(conf));
+}
+
+void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) {
+  double opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
+  if (packet_loss_rate_ != opt_loss_rate) {
+    packet_loss_rate_ = opt_loss_rate;
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetPacketLossRate(
+               inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  }
+}
+
+void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
+  config_.bitrate_bps =
+      std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps);
+  RTC_DCHECK(config_.IsOk());
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps));
+}
+
+size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+
+size_t AudioEncoderOpus::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
+  if (!config.IsOk())
+    return false;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, config.num_channels,
+                                           config.application));
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.bitrate_bps));
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  RTC_CHECK_EQ(0,
+               WebRtcOpus_SetPacketLossRate(
+                   inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  config_ = config;
+  return true;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
new file mode 100644
index 0000000000..59c8f796ee
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
+namespace webrtc {
+
+struct CodecInst;
+
+class AudioEncoderOpus final : public AudioEncoder {
+ public:
+  enum ApplicationMode {
+    kVoip = 0,
+    kAudio = 1,
+  };
+
+  struct Config {
+    bool IsOk() const;
+    int frame_size_ms = 20;
+    size_t num_channels = 1;
+    int payload_type = 120;
+    ApplicationMode application = kVoip;
+    int bitrate_bps = 64000;
+    bool fec_enabled = false;
+    int max_playback_rate_hz = 48000;
+    int complexity = kDefaultComplexity;
+    bool dtx_enabled = false;
+
+   private:
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
+    // If we are on Android, iOS and/or ARM, use a lower complexity setting as
+    // default, to save encoder complexity.
+    static const int kDefaultComplexity = 5;
+#else
+    static const int kDefaultComplexity = 9;
+#endif
+  };
+
+  explicit AudioEncoderOpus(const Config& config);
+  explicit AudioEncoderOpus(const CodecInst& codec_inst);
+  ~AudioEncoderOpus() override;
+
+  size_t MaxEncodedBytes() const override;
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             size_t max_encoded_bytes,
+                             uint8_t* encoded) override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
+  // being inactive. During that, it still sends 2 packets (one for content, one
+  // for signaling) about every 400 ms.
+  bool SetDtx(bool enable) override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
+
+  // Getters for testing.
+  double packet_loss_rate() const { return packet_loss_rate_; }
+  ApplicationMode application() const { return config_.application; }
+  bool dtx_enabled() const { return config_.dtx_enabled; }
+
+ private:
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  bool RecreateEncoderInstance(const Config& config);
+
+  Config config_;
+  double packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 33afa5fcc5..441e807b4f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -9,72 +9,144 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 namespace webrtc {
 
+namespace {
+const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000};
+}  // namespace
+
 class AudioEncoderOpusTest : public ::testing::Test {
  protected:
-  // The constructor simply creates an Opus encoder with default configuration.
-  AudioEncoderOpusTest()
-      : opus_(new AudioEncoderOpus(AudioEncoderOpus::Config())) {}
-
-  // Repeatedly sets packet loss rates in the range [from, to], increasing by
-  // 0.01 in each step. The function verifies that the actual loss rate is
-  // |expected_return|.
-  void TestSetPacketLossRate(double from, double to, double expected_return) {
-    ASSERT_TRUE(opus_);
-    for (double loss = from; loss <= to;
-         (to >= from) ? loss += 0.01 : loss -= 0.01) {
-      opus_->SetProjectedPacketLossRate(loss);
-      EXPECT_DOUBLE_EQ(expected_return, opus_->packet_loss_rate());
-    }
+  void CreateCodec(int num_channels) {
+    codec_inst_.channels = num_channels;
+    encoder_.reset(new AudioEncoderOpus(codec_inst_));
+    auto expected_app =
+        num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
+    EXPECT_EQ(expected_app, encoder_->application());
   }
 
-  rtc::scoped_ptr<AudioEncoderOpus> opus_;
+  CodecInst codec_inst_ = kOpusSettings;
+  rtc::scoped_ptr<AudioEncoderOpus> encoder_;
 };
 
+TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {
+  CreateCodec(1);
+}
+
+TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  CreateCodec(2);
+}
+
+TEST_F(AudioEncoderOpusTest, ChangeApplicationMode) {
+  CreateCodec(2);
+  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+}
+
+TEST_F(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
+  CreateCodec(2);
+
+  // Trigger a reset.
+  encoder_->Reset();
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
+
+  // Now change to kVoip.
+  EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+
+  // Trigger a reset again.
+  encoder_->Reset();
+  // Verify that the mode is still kVoip.
+  EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
+}
+
+TEST_F(AudioEncoderOpusTest, ToggleDtx) {
+  CreateCodec(2);
+  // Enable DTX
+  EXPECT_TRUE(encoder_->SetDtx(true));
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
+  // Turn off DTX.
+  EXPECT_TRUE(encoder_->SetDtx(false));
+}
+
+TEST_F(AudioEncoderOpusTest, SetBitrate) {
+  CreateCodec(1);
+  // Constants are replicated from audio_encoder_opus.cc.
+  const int kMinBitrateBps = 500;
+  const int kMaxBitrateBps = 512000;
+  // Set a too low bitrate.
+  encoder_->SetTargetBitrate(kMinBitrateBps - 1);
+  EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
+  // Set a too high bitrate.
+  encoder_->SetTargetBitrate(kMaxBitrateBps + 1);
+  EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
+  // Set the minimum rate.
+  encoder_->SetTargetBitrate(kMinBitrateBps);
+  EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
+  // Set the maximum rate.
+  encoder_->SetTargetBitrate(kMaxBitrateBps);
+  EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
+  // Set rates from 1000 up to 32000 bps.
+  for (int rate = 1000; rate <= 32000; rate += 1000) {
+    encoder_->SetTargetBitrate(rate);
+    EXPECT_EQ(rate, encoder_->GetTargetBitrate());
+  }
+}
+
 namespace {
-// These constants correspond to those used in
-// AudioEncoderOpus::SetProjectedPacketLossRate.
-const double kPacketLossRate20 = 0.20;
-const double kPacketLossRate10 = 0.10;
-const double kPacketLossRate5 = 0.05;
-const double kPacketLossRate1 = 0.01;
-const double kLossRate20Margin = 0.02;
-const double kLossRate10Margin = 0.01;
-const double kLossRate5Margin = 0.01;
+
+// Returns a vector with the n evenly-spaced numbers a, a + (b - a)/(n - 1),
+// ..., b.
+std::vector<double> IntervalSteps(double a, double b, size_t n) {
+  RTC_DCHECK_GT(n, 1u);
+  const double step = (b - a) / (n - 1);
+  std::vector<double> points;
+  for (size_t i = 0; i < n; ++i)
+    points.push_back(a + i * step);
+  return points;
+}
+
+// Sets the packet loss rate to each number in the vector in turn, and verifies
+// that the loss rate as reported by the encoder is |expected_return| for all
+// of them.
+void TestSetPacketLossRate(AudioEncoderOpus* encoder,
+                           const std::vector<double>& losses,
+                           double expected_return) {
+  for (double loss : losses) {
+    encoder->SetProjectedPacketLossRate(loss);
+    EXPECT_DOUBLE_EQ(expected_return, encoder->packet_loss_rate());
+  }
+}
+
 }  // namespace
 
 TEST_F(AudioEncoderOpusTest, PacketLossRateOptimized) {
+  CreateCodec(1);
+  auto I = [](double a, double b) { return IntervalSteps(a, b, 10); };
+  const double eps = 1e-15;
+
   // Note that the order of the following calls is critical.
-  TestSetPacketLossRate(0.0, 0.0, 0.0);
-  TestSetPacketLossRate(kPacketLossRate1,
-                        kPacketLossRate5 + kLossRate5Margin - 0.01,
-                        kPacketLossRate1);
-  TestSetPacketLossRate(kPacketLossRate5 + kLossRate5Margin,
-                        kPacketLossRate10 + kLossRate10Margin - 0.01,
-                        kPacketLossRate5);
-  TestSetPacketLossRate(kPacketLossRate10 + kLossRate10Margin,
-                        kPacketLossRate20 + kLossRate20Margin - 0.01,
-                        kPacketLossRate10);
-  TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
-                        1.0,
-                        kPacketLossRate20);
-  TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
-                        kPacketLossRate20 - kLossRate20Margin,
-                        kPacketLossRate20);
-  TestSetPacketLossRate(kPacketLossRate20 - kLossRate20Margin - 0.01,
-                        kPacketLossRate10 - kLossRate10Margin,
-                        kPacketLossRate10);
-  TestSetPacketLossRate(kPacketLossRate10 - kLossRate10Margin - 0.01,
-                        kPacketLossRate5 - kLossRate5Margin,
-                        kPacketLossRate5);
-  TestSetPacketLossRate(kPacketLossRate5 - kLossRate5Margin - 0.01,
-                        kPacketLossRate1,
-                        kPacketLossRate1);
-  TestSetPacketLossRate(0.0, 0.0, 0.0);
+
+  // clang-format off
+  TestSetPacketLossRate(encoder_.get(), I(0.00      , 0.01 - eps), 0.00);
+  TestSetPacketLossRate(encoder_.get(), I(0.01 + eps, 0.06 - eps), 0.01);
+  TestSetPacketLossRate(encoder_.get(), I(0.06 + eps, 0.11 - eps), 0.05);
+  TestSetPacketLossRate(encoder_.get(), I(0.11 + eps, 0.22 - eps), 0.10);
+  TestSetPacketLossRate(encoder_.get(), I(0.22 + eps, 1.00      ), 0.20);
+
+  TestSetPacketLossRate(encoder_.get(), I(1.00      , 0.18 + eps), 0.20);
+  TestSetPacketLossRate(encoder_.get(), I(0.18 - eps, 0.09 + eps), 0.10);
+  TestSetPacketLossRate(encoder_.get(), I(0.09 - eps, 0.04 + eps), 0.05);
+  TestSetPacketLossRate(encoder_.get(), I(0.04 - eps, 0.01 + eps), 0.01);
+  TestSetPacketLossRate(encoder_.get(), I(0.01 - eps, 0.00      ), 0.00);
+  // clang-format on
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
deleted file mode 100644
index bd76b4900a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
-
-#include <vector>
-
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-
-namespace webrtc {
-
-// NOTE: This class has neither ThreadChecker, nor locks. The owner of an
-// AudioEncoderOpus object must ensure that it is not accessed concurrently.
-
-class AudioEncoderOpus final : public AudioEncoder {
- public:
-  enum ApplicationMode {
-    kVoip = 0,
-    kAudio = 1,
-  };
-
-  struct Config {
-    Config();
-    bool IsOk() const;
-    int frame_size_ms;
-    int num_channels;
-    int payload_type;
-    ApplicationMode application;
-    int bitrate_bps;
-    bool fec_enabled;
-    int max_playback_rate_hz;
-    int complexity;
-    bool dtx_enabled;
-  };
-
-  explicit AudioEncoderOpus(const Config& config);
-  ~AudioEncoderOpus() override;
-
-  int SampleRateHz() const override;
-  int NumChannels() const override;
-  size_t MaxEncodedBytes() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
-  void SetTargetBitrate(int bits_per_second) override;
-  void SetProjectedPacketLossRate(double fraction) override;
-
-  double packet_loss_rate() const { return packet_loss_rate_; }
-  ApplicationMode application() const { return application_; }
-  bool dtx_enabled() const { return dtx_enabled_; }
-
- protected:
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
-
- private:
-  const int num_10ms_frames_per_packet_;
-  const int num_channels_;
-  const int payload_type_;
-  const ApplicationMode application_;
-  int bitrate_bps_;
-  const bool dtx_enabled_;
-  const int samples_per_10ms_frame_;
-  std::vector<int16_t> input_buffer_;
-  OpusEncInst* inst_;
-  uint32_t first_timestamp_in_buffer_;
-  double packet_loss_rate_;
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus.gypi
index e28f283d91..de793529b0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus.gypi
@@ -32,7 +32,7 @@
           'conditions': [
             ['build_with_mozilla==1', {
               # Mozilla provides its own build of the opus library.
-              'include_dirs': [
+             'include_dirs': [
                 '/media/libopus/include',
                 '/media/libopus/src',
                 '/media/libopus/celt',
@@ -51,18 +51,17 @@
       'dependencies': [
         'audio_encoder_interface',
       ],
-      'include_dirs': [
-        '<(webrtc_root)',
-      ],
       'defines': [
         'OPUS_COMPLEXITY=<(opus_complexity)'
       ],
       'sources': [
+        'audio_decoder_opus.cc',
+        'audio_decoder_opus.h',
         'audio_encoder_opus.cc',
-        'interface/audio_encoder_opus.h',
-        'interface/opus_interface.h',
+        'audio_encoder_opus.h',
         'opus_inst.h',
         'opus_interface.c',
+        'opus_interface.h',
       ],
     },
   ],
@@ -78,9 +77,6 @@
             '<(webrtc_root)/test/test.gyp:test_support_main',
             '<(DEPTH)/testing/gtest.gyp:gtest',
           ],
-          'include_dirs': [
-            '<(webrtc_root)',
-          ],
           'sources': [
             'opus_fec_test.cc',
           ],
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index a30b1cb903..4f9f7ff7bb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -9,8 +9,9 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::string;
@@ -21,7 +22,7 @@ using ::testing::TestWithParam;
 namespace webrtc {
 
 // Define coding parameter as <channels, bit_rate, filename, extension>.
-typedef tuple<int, int, string, string> coding_param;
+typedef tuple<size_t, int, string, string> coding_param;
 typedef struct mode mode;
 
 struct mode {
@@ -45,15 +46,15 @@ class OpusFecTest : public TestWithParam<coding_param> {
 
   int block_duration_ms_;
   int sampling_khz_;
-  int block_length_sample_;
+  size_t block_length_sample_;
 
-  int channels_;
+  size_t channels_;
   int bit_rate_;
 
   size_t data_pointer_;
   size_t loop_length_samples_;
-  int max_bytes_;
-  int encoded_bytes_;
+  size_t max_bytes_;
+  size_t encoded_bytes_;
 
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
@@ -68,7 +69,7 @@ class OpusFecTest : public TestWithParam<coding_param> {
 void OpusFecTest::SetUp() {
   channels_ = get<0>(GetParam());
   bit_rate_ = get<1>(GetParam());
-  printf("Coding %d channel signal at %d bps.\n", channels_, bit_rate_);
+  printf("Coding %" PRIuS " channel signal at %d bps.\n", channels_, bit_rate_);
 
   in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
 
@@ -122,7 +123,8 @@ void OpusFecTest::TearDown() {
 OpusFecTest::OpusFecTest()
     : block_duration_ms_(kOpusBlockDurationMs),
       sampling_khz_(kOpusSamplingKhz),
-      block_length_sample_(block_duration_ms_ * sampling_khz_),
+      block_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
       data_pointer_(0),
       max_bytes_(0),
       encoded_bytes_(0),
@@ -131,18 +133,18 @@ OpusFecTest::OpusFecTest()
 }
 
 void OpusFecTest::EncodeABlock() {
-  int16_t value = WebRtcOpus_Encode(opus_encoder_,
-                                    &in_data_[data_pointer_],
-                                    block_length_sample_,
-                                    max_bytes_, &bit_stream_[0]);
+  int value = WebRtcOpus_Encode(opus_encoder_,
+                                &in_data_[data_pointer_],
+                                block_length_sample_,
+                                max_bytes_, &bit_stream_[0]);
   EXPECT_GT(value, 0);
 
-  encoded_bytes_ = value;
+  encoded_bytes_ = static_cast<size_t>(value);
 }
 
 void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
   int16_t audio_type;
-  int16_t value_1 = 0, value_2 = 0;
+  int value_1 = 0, value_2 = 0;
 
   if (lost_previous) {
     // Decode previous frame.
@@ -154,14 +156,14 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
     } else {
       value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1);
     }
-    EXPECT_EQ(block_length_sample_, value_1);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
   }
 
   if (!lost_current) {
     // Decode current frame.
     value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
                                 &out_data_[value_1 * channels_], &audio_type);
-    EXPECT_EQ(block_length_sample_, value_2);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
   }
 }
 
@@ -196,7 +198,7 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
       EncodeABlock();
 
       // Check if payload has FEC.
-      int16_t fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+      int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
 
       // If FEC is disabled or the target packet loss rate is set to 0, there
       // should be no FEC in the bit stream.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_inst.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
index 373db392a6..8d032baf35 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -11,17 +11,26 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
 
+#include <stddef.h>
+
 #include "opus.h"
 
 struct WebRtcOpusEncInst {
   OpusEncoder* encoder;
+  size_t channels;
   int in_dtx_mode;
+  // When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
+  // to break long zero segment so as to prevent DTX from going wrong. We use
+  // one counter for each channel. After each encoding, |zero_counts| contain
+  // the remaining zeros from the last frame.
+  // TODO(minyue): remove this when Opus gets an internal fix to DTX.
+  size_t* zero_counts;
 };
 
 struct WebRtcOpusDecInst {
   OpusDecoder* decoder;
   int prev_decoded_samples;
-  int channels;
+  size_t channels;
   int in_dtx_mode;
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 1330e2bc17..3b842a0554 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 
+#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
@@ -29,48 +30,61 @@ enum {
 
   /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
   kWebRtcOpusDefaultFrameSize = 960,
+
+  // Maximum number of consecutive zeros, beyond or equal to which DTX can fail.
+  kZeroBreakCount = 157,
+
+#if defined(OPUS_FIXED_POINT)
+  kZeroBreakValue = 10,
+#else
+  kZeroBreakValue = 1,
+#endif
 };
 
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
-                                 int32_t channels,
+                                 size_t channels,
                                  int32_t application) {
-  OpusEncInst* state;
-  if (inst != NULL) {
-    state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
-    if (state) {
-      int opus_app;
-      switch (application) {
-        case 0: {
-          opus_app = OPUS_APPLICATION_VOIP;
-          break;
-        }
-        case 1: {
-          opus_app = OPUS_APPLICATION_AUDIO;
-          break;
-        }
-        default: {
-          free(state);
-          return -1;
-        }
-      }
+  int opus_app;
+  if (!inst)
+    return -1;
 
-      int error;
-      state->encoder = opus_encoder_create(48000, channels, opus_app,
-                                           &error);
-      state->in_dtx_mode = 0;
-      if (error == OPUS_OK && state->encoder != NULL) {
-        *inst = state;
-        return 0;
-      }
-      free(state);
-    }
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
   }
-  return -1;
+
+  OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
+  assert(state);
+
+  // Allocate zero counters.
+  state->zero_counts = calloc(channels, sizeof(size_t));
+  assert(state->zero_counts);
+
+  int error;
+  state->encoder = opus_encoder_create(48000, (int)channels, opus_app,
+                                       &error);
+  if (error != OPUS_OK || !state->encoder) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+
+  *inst = state;
+  return 0;
 }
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   if (inst) {
     opus_encoder_destroy(inst->encoder);
+    free(inst->zero_counts);
     free(inst);
     return 0;
   } else {
@@ -78,22 +92,51 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   }
 }
 
-int16_t WebRtcOpus_Encode(OpusEncInst* inst,
-                          const int16_t* audio_in,
-                          int16_t samples,
-                          int16_t length_encoded_buffer,
-                          uint8_t* encoded) {
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded) {
   int res;
+  size_t i;
+  size_t c;
+
+  int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs];
 
   if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
     return -1;
   }
 
+  const size_t channels = inst->channels;
+  int use_buffer = 0;
+
+  // Break long consecutive zeros by forcing a "1" every |kZeroBreakCount|
+  // samples.
+  if (inst->in_dtx_mode) {
+    for (i = 0; i < samples; ++i) {
+      for (c = 0; c < channels; ++c) {
+        if (audio_in[i * channels + c] == 0) {
+          ++inst->zero_counts[c];
+          if (inst->zero_counts[c] == kZeroBreakCount) {
+            if (!use_buffer) {
+              memcpy(buffer, audio_in, samples * channels * sizeof(int16_t));
+              use_buffer = 1;
+            }
+            buffer[i * channels + c] = kZeroBreakValue;
+            inst->zero_counts[c] = 0;
+          }
+        } else {
+          inst->zero_counts[c] = 0;
+        }
+      }
+    }
+  }
+
   res = opus_encode(inst->encoder,
-                    (const opus_int16*)audio_in,
-                    samples,
+                    use_buffer ? buffer : audio_in,
+                    (int)samples,
                     encoded,
-                    length_encoded_buffer);
+                    (opus_int32)length_encoded_buffer);
 
   if (res == 1) {
     // Indicates DTX since the packet has nothing but a header. In principle,
@@ -171,15 +214,29 @@ int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
 }
 
 int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
-  if (inst) {
-    return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
-  } else {
+  if (!inst) {
     return -1;
   }
+
+  // To prevent Opus from entering CELT-only mode by forcing signal type to
+  // voice to make sure that DTX behaves correctly. Currently, DTX does not
+  // last long during a pure silence, if the signal type is not forced.
+  // TODO(minyue): Remove the signal type forcing when Opus DTX works properly
+  // without it.
+  int ret = opus_encoder_ctl(inst->encoder,
+                             OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+  if (ret != OPUS_OK)
+    return ret;
+
+  return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
 }
 
 int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
   if (inst) {
+    int ret = opus_encoder_ctl(inst->encoder,
+                               OPUS_SET_SIGNAL(OPUS_AUTO));
+    if (ret != OPUS_OK)
+      return ret;
     return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
   } else {
     return -1;
@@ -194,7 +251,7 @@ int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
   }
 }
 
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
   int error;
   OpusDecInst* state;
 
@@ -206,7 +263,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
     }
 
     /* Create new memory, always at 48000 Hz. */
-    state->decoder = opus_decoder_create(48000, channels, &error);
+    state->decoder = opus_decoder_create(48000, (int)channels, &error);
     if (error == OPUS_OK && state->decoder != NULL) {
       /* Creation of memory all ok. */
       state->channels = channels;
@@ -235,21 +292,17 @@ int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
   }
 }
 
-int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
   return inst->channels;
 }
 
-int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
-  int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
-  if (error == OPUS_OK) {
-    inst->in_dtx_mode = 0;
-    return 0;
-  }
-  return -1;
+void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+  opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+  inst->in_dtx_mode = 0;
 }
 
 /* For decoder to determine if it is to output speech or comfort noise. */
-static int16_t DetermineAudioType(OpusDecInst* inst, int16_t encoded_bytes) {
+static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
   // Audio type becomes comfort noise if |encoded_byte| is 1 and keeps
   // to be so if the following |encoded_byte| are 0 or 1.
   if (encoded_bytes == 0 && inst->in_dtx_mode) {
@@ -267,9 +320,9 @@ static int16_t DetermineAudioType(OpusDecInst* inst, int16_t encoded_bytes) {
  * is set to the number of samples needed for PLC in case of losses.
  * It is up to the caller to make sure the value is correct. */
 static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
-                        int16_t encoded_bytes, int frame_size,
+                        size_t encoded_bytes, int frame_size,
                         int16_t* decoded, int16_t* audio_type, int decode_fec) {
-  int res = opus_decode(inst->decoder, encoded, encoded_bytes,
+  int res = opus_decode(inst->decoder, encoded, (opus_int32)encoded_bytes,
                         (opus_int16*)decoded, frame_size, decode_fec);
 
   if (res <= 0)
@@ -280,9 +333,9 @@ static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
   return res;
 }
 
-int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
-                          int16_t encoded_bytes, int16_t* decoded,
-                          int16_t* audio_type) {
+int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
+                      size_t encoded_bytes, int16_t* decoded,
+                      int16_t* audio_type) {
   int decoded_samples;
 
   if (encoded_bytes == 0) {
@@ -307,8 +360,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
   return decoded_samples;
 }
 
-int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
-                             int16_t number_of_lost_frames) {
+int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                         int number_of_lost_frames) {
   int16_t audio_type = 0;
   int decoded_samples;
   int plc_samples;
@@ -328,9 +381,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
   return decoded_samples;
 }
 
-int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
-                             int16_t encoded_bytes, int16_t* decoded,
-                             int16_t* audio_type) {
+int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
+                         size_t encoded_bytes, int16_t* decoded,
+                         int16_t* audio_type) {
   int decoded_samples;
   int fec_samples;
 
@@ -351,9 +404,15 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
 
 int WebRtcOpus_DurationEst(OpusDecInst* inst,
                            const uint8_t* payload,
-                           int payload_length_bytes) {
+                           size_t payload_length_bytes) {
+  if (payload_length_bytes == 0) {
+    // WebRtcOpus_Decode calls PLC when payload length is zero. So we return
+    // PLC duration correspondingly.
+    return WebRtcOpus_PlcDuration(inst);
+  }
+
   int frames, samples;
-  frames = opus_packet_get_nb_frames(payload, payload_length_bytes);
+  frames = opus_packet_get_nb_frames(payload, (opus_int32)payload_length_bytes);
   if (frames < 0) {
     /* Invalid payload data. */
     return 0;
@@ -366,8 +425,17 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
   return samples;
 }
 
+int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
+  /* The number of samples we ask for is |number_of_lost_frames| times
+   * |prev_decoded_samples_|. Limit the number of samples to maximum
+   * |kWebRtcOpusMaxFrameSizePerChannel|. */
+  const int plc_samples = inst->prev_decoded_samples;
+  return (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
+      plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
+}
+
 int WebRtcOpus_FecDurationEst(const uint8_t* payload,
-                              int payload_length_bytes) {
+                              size_t payload_length_bytes) {
   int samples;
   if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
     return 0;
@@ -382,13 +450,13 @@ int WebRtcOpus_FecDurationEst(const uint8_t* payload,
 }
 
 int WebRtcOpus_PacketHasFec(const uint8_t* payload,
-                            int payload_length_bytes) {
+                            size_t payload_length_bytes) {
   int frames, channels, payload_length_ms;
   int n;
   opus_int16 frame_sizes[48];
   const unsigned char *frame_data[48];
 
-  if (payload == NULL || payload_length_bytes <= 0)
+  if (payload == NULL || payload_length_bytes == 0)
     return 0;
 
   /* In CELT_ONLY mode, packets should not have FEC. */
@@ -421,8 +489,8 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload,
   }
 
   /* The following is to parse the LBRR flags. */
-  if (opus_packet_parse(payload, payload_length_bytes, NULL, frame_data,
-                        frame_sizes, NULL) < 0) {
+  if (opus_packet_parse(payload, (opus_int32)payload_length_bytes, NULL,
+                        frame_data, frame_sizes, NULL) < 0) {
     return 0;
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
similarity index 85%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
index 27009a86af..754b49c808 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+
+#include <stddef.h>
 
 #include "webrtc/typedefs.h"
 
@@ -41,7 +43,7 @@ typedef struct WebRtcOpusDecInst OpusDecInst;
  *                            -1 - Error
  */
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
-                                 int32_t channels,
+                                 size_t channels,
                                  int32_t application);
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
@@ -64,11 +66,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
  * Return value                 : >=0 - Length (in bytes) of coded data
  *                                -1 - Error
  */
-int16_t WebRtcOpus_Encode(OpusEncInst* inst,
-                          const int16_t* audio_in,
-                          int16_t samples,
-                          int16_t length_encoded_buffer,
-                          uint8_t* encoded);
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded);
 
 /****************************************************************************
  * WebRtcOpus_SetBitRate(...)
@@ -193,7 +195,7 @@ int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
  */
 int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
 
-int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels);
 int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
 
 /****************************************************************************
@@ -201,7 +203,7 @@ int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
  *
  * This function returns the number of channels created for Opus decoder.
  */
-int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst);
 
 /****************************************************************************
  * WebRtcOpus_DecoderInit(...)
@@ -210,11 +212,8 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
  *
  * Input:
  *      - inst               : Decoder context
- *
- * Return value              :  0 - Success
- *                             -1 - Error
  */
-int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
+void WebRtcOpus_DecoderInit(OpusDecInst* inst);
 
 /****************************************************************************
  * WebRtcOpus_Decode(...)
@@ -236,9 +235,9 @@ int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
  * Return value              : >0 - Samples per channel in decoded vector
  *                             -1 - Error
  */
-int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
-                          int16_t encoded_bytes, int16_t* decoded,
-                          int16_t* audio_type);
+int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
+                      size_t encoded_bytes, int16_t* decoded,
+                      int16_t* audio_type);
 
 /****************************************************************************
  * WebRtcOpus_DecodePlc(...)
@@ -254,8 +253,8 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
  * Return value                   : >0 - number of samples in decoded PLC vector
  *                                  -1 - Error
  */
-int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
-                             int16_t number_of_lost_frames);
+int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                         int number_of_lost_frames);
 
 /****************************************************************************
  * WebRtcOpus_DecodeFec(...)
@@ -275,9 +274,9 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
  *                              0 - No FEC data in the packet
  *                             -1 - Error
  */
-int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
-                             int16_t encoded_bytes, int16_t* decoded,
-                             int16_t* audio_type);
+int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
+                         size_t encoded_bytes, int16_t* decoded,
+                         int16_t* audio_type);
 
 /****************************************************************************
  * WebRtcOpus_DurationEst(...)
@@ -288,11 +287,26 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
  *        - payload              : Encoded data pointer
  *        - payload_length_bytes : Bytes of encoded data
  *
- * Return value                  : The duration of the packet, in samples.
+ * Return value                  : The duration of the packet, in samples per
+ *                                 channel.
  */
 int WebRtcOpus_DurationEst(OpusDecInst* inst,
                            const uint8_t* payload,
-                           int payload_length_bytes);
+                           size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PlcDuration(...)
+ *
+ * This function calculates the duration of a frame returned by packet loss
+ * concealment (PLC).
+ *
+ * Input:
+ *        - inst                 : Decoder context
+ *
+ * Return value                  : The duration of a frame returned by PLC, in
+ *                                 samples per channel.
+ */
+int WebRtcOpus_PlcDuration(OpusDecInst* inst);
 
 /* TODO(minyue): Check whether it is needed to add a decoder context to the
  * arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells
@@ -308,11 +322,11 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
  *        - payload_length_bytes : Bytes of encoded data
  *
  * Return value                  : >0 - The duration of the FEC data in the
- *                                 packet in samples.
+ *                                 packet in samples per channel.
  *                                  0 - No FEC data in the packet.
  */
 int WebRtcOpus_FecDurationEst(const uint8_t* payload,
-                              int payload_length_bytes);
+                              size_t payload_length_bytes);
 
 /****************************************************************************
  * WebRtcOpus_PacketHasFec(...)
@@ -326,10 +340,10 @@ int WebRtcOpus_FecDurationEst(const uint8_t* payload,
  *                                 1 - the packet contains FEC.
  */
 int WebRtcOpus_PacketHasFec(const uint8_t* payload,
-                            int payload_length_bytes);
+                            size_t payload_length_bytes);
 
 #ifdef __cplusplus
 }  // extern "C"
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
index b39de499a7..4d1aa42c89 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 using ::std::string;
@@ -24,8 +24,8 @@ class OpusSpeedTest : public AudioCodecSpeedTest {
   void SetUp() override;
   void TearDown() override;
   virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             int max_bytes, int* encoded_bytes);
-  virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+                             size_t max_bytes, size_t* encoded_bytes);
+  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
                              int16_t* out_data);
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
@@ -58,26 +58,26 @@ void OpusSpeedTest::TearDown() {
 }
 
 float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                                  int max_bytes, int* encoded_bytes) {
+                                  size_t max_bytes, size_t* encoded_bytes) {
   clock_t clocks = clock();
   int value = WebRtcOpus_Encode(opus_encoder_, in_data,
                                 input_length_sample_, max_bytes,
                                 bit_stream);
   clocks = clock() - clocks;
   EXPECT_GT(value, 0);
-  *encoded_bytes = value;
+  *encoded_bytes = static_cast<size_t>(value);
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
 float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
-                                  int encoded_bytes, int16_t* out_data) {
+                                  size_t encoded_bytes, int16_t* out_data) {
   int value;
   int16_t audio_type;
   clock_t clocks = clock();
   value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
                             &audio_type);
   clocks = clock() - clocks;
-  EXPECT_EQ(output_length_sample_, value);
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index 00c88a5e7c..c82b184b38 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -10,7 +10,8 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -25,27 +26,28 @@ using ::testing::Combine;
 // Maximum number of bytes in output bitstream.
 const size_t kMaxBytes = 1000;
 // Sample rate of Opus.
-const int kOpusRateKhz = 48;
+const size_t kOpusRateKhz = 48;
 // Number of samples-per-channel in a 20 ms frame, sampled at 48 kHz.
-const int kOpus20msFrameSamples = kOpusRateKhz * 20;
+const size_t kOpus20msFrameSamples = kOpusRateKhz * 20;
 // Number of samples-per-channel in a 10 ms frame, sampled at 48 kHz.
-const int kOpus10msFrameSamples = kOpusRateKhz * 10;
+const size_t kOpus10msFrameSamples = kOpusRateKhz * 10;
 
 class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
  protected:
   OpusTest();
 
-  void TestDtxEffect(bool dtx);
+  void TestDtxEffect(bool dtx, int block_length_ms);
 
   // Prepare |speech_data_| for encoding, read from a hard-coded file.
   // After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
   // block of |block_length_ms| milliseconds. The data is looped every
   // |loop_length_ms| milliseconds.
-  void PrepareSpeechData(int channel, int block_length_ms, int loop_length_ms);
+  void PrepareSpeechData(size_t channel,
+                         int block_length_ms,
+                         int loop_length_ms);
 
   int EncodeDecode(WebRtcOpusEncInst* encoder,
-                   const int16_t* input_audio,
-                   const int input_samples,
+                   rtc::ArrayView<const int16_t> input_audio,
                    WebRtcOpusDecInst* decoder,
                    int16_t* output_audio,
                    int16_t* audio_type);
@@ -53,13 +55,16 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
   void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
                           opus_int32 expect, int32_t set);
 
+  void CheckAudioBounded(const int16_t* audio, size_t samples, size_t channels,
+                         uint16_t bound) const;
+
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
 
   AudioLoop speech_data_;
   uint8_t bitstream_[kMaxBytes];
-  int encoded_bytes_;
-  int channels_;
+  size_t encoded_bytes_;
+  size_t channels_;
   int application_;
 };
 
@@ -67,11 +72,11 @@ OpusTest::OpusTest()
     : opus_encoder_(NULL),
       opus_decoder_(NULL),
       encoded_bytes_(0),
-      channels_(::testing::get<0>(GetParam())),
+      channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
       application_(::testing::get<1>(GetParam())) {
 }
 
-void OpusTest::PrepareSpeechData(int channel, int block_length_ms,
+void OpusTest::PrepareSpeechData(size_t channel, int block_length_ms,
                                  int loop_length_ms) {
   const std::string file_name =
         webrtc::test::ResourcePath((channel == 1) ?
@@ -95,25 +100,40 @@ void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
   EXPECT_EQ(expect, bandwidth);
 }
 
+void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples,
+                                 size_t channels, uint16_t bound) const {
+  for (size_t i = 0; i < samples; ++i) {
+    for (size_t c = 0; c < channels; ++c) {
+      ASSERT_GE(audio[i * channels + c], -bound);
+      ASSERT_LE(audio[i * channels + c], bound);
+    }
+  }
+}
+
 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
-                           const int16_t* input_audio,
-                           const int input_samples,
+                           rtc::ArrayView<const int16_t> input_audio,
                            WebRtcOpusDecInst* decoder,
                            int16_t* output_audio,
                            int16_t* audio_type) {
-  encoded_bytes_ = WebRtcOpus_Encode(encoder,
-                                    input_audio,
-                                    input_samples, kMaxBytes,
-                                    bitstream_);
-  return WebRtcOpus_Decode(decoder, bitstream_,
-                           encoded_bytes_, output_audio,
-                           audio_type);
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      encoder, input_audio.data(),
+      rtc::CheckedDivExact(input_audio.size(), channels_),
+      kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
+  int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
+  int act_len = WebRtcOpus_Decode(decoder, bitstream_,
+                                  encoded_bytes_, output_audio,
+                                  audio_type);
+  EXPECT_EQ(est_len, act_len);
+  return act_len;
 }
 
 // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
 // they should not. This test is signal dependent.
-void OpusTest::TestDtxEffect(bool dtx) {
-  PrepareSpeechData(channels_, 20, 2000);
+void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
+  PrepareSpeechData(channels_, block_length_ms, 2000);
+  const size_t samples = kOpusRateKhz * block_length_ms;
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
@@ -126,25 +146,24 @@ void OpusTest::TestDtxEffect(bool dtx) {
                                      channels_ == 1 ? 32000 : 64000));
 
   // Set input audio as silence.
-  int16_t* silence = new int16_t[kOpus20msFrameSamples * channels_];
-  memset(silence, 0, sizeof(int16_t) * kOpus20msFrameSamples * channels_);
+  std::vector<int16_t> silence(samples * channels_, 0);
 
   // Setting DTX.
   EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
       WebRtcOpus_DisableDtx(opus_encoder_));
 
   int16_t audio_type;
-  int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
+  int16_t* output_data_decode = new int16_t[samples * channels_];
 
   for (int i = 0; i < 100; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples,
-              EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
-                           kOpus20msFrameSamples, opus_decoder_,
-                           output_data_decode, &audio_type));
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+                  output_data_decode, &audio_type)));
     // If not DTX, it should never enter DTX mode. If DTX, we do not care since
     // whether it enters DTX depends on the signal type.
     if (!dtx) {
-      EXPECT_GT(encoded_bytes_, 1);
+      EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(0, audio_type);  // Speech.
@@ -154,16 +173,16 @@ void OpusTest::TestDtxEffect(bool dtx) {
   // We input some silent segments. In DTX mode, the encoder will stop sending.
   // However, DTX may happen after a while.
   for (int i = 0; i < 30; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples,
-              EncodeDecode(opus_encoder_, silence,
-                           kOpus20msFrameSamples, opus_decoder_,
-                           output_data_decode, &audio_type));
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     if (!dtx) {
-      EXPECT_GT(encoded_bytes_, 1);
+      EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(0, audio_type);  // Speech.
-    } else if (1 == encoded_bytes_) {
+    } else if (encoded_bytes_ == 1) {
       EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(2, audio_type);  // Comfort noise.
@@ -171,62 +190,98 @@ void OpusTest::TestDtxEffect(bool dtx) {
     }
   }
 
-  // DTX mode is maintained 400 ms.
-  for (int i = 0; i < 19; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples,
-              EncodeDecode(opus_encoder_, silence,
-                           kOpus20msFrameSamples, opus_decoder_,
-                           output_data_decode, &audio_type));
+  // When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
+  // one with an arbitrary size and the other of 1-byte, then stops sending for
+  // a certain number of frames.
+
+  // |max_dtx_frames| is the maximum number of frames Opus can stay in DTX.
+  const int max_dtx_frames = 400 / block_length_ms + 1;
+
+  // We run |kRunTimeMs| milliseconds of pure silence.
+  const int kRunTimeMs = 2000;
+
+  // We check that, after a |kCheckTimeMs| milliseconds (given that the CNG in
+  // Opus needs time to adapt), the absolute values of DTX decoded signal are
+  // bounded by |kOutputValueBound|.
+  const int kCheckTimeMs = 1500;
+
+#if defined(OPUS_FIXED_POINT)
+  const uint16_t kOutputValueBound = 20;
+#else
+  const uint16_t kOutputValueBound = 2;
+#endif
+
+  int time = 0;
+  while (time < kRunTimeMs) {
+    // DTX mode is maintained for maximum |max_dtx_frames| frames.
+    int i = 0;
+    for (; i < max_dtx_frames; ++i) {
+      time += block_length_ms;
+      EXPECT_EQ(samples,
+                static_cast<size_t>(EncodeDecode(
+                    opus_encoder_, silence, opus_decoder_, output_data_decode,
+                    &audio_type)));
+      if (dtx) {
+        if (encoded_bytes_ > 1)
+          break;
+        EXPECT_EQ(0U, encoded_bytes_)  // Send 0 byte.
+            << "Opus should have entered DTX mode.";
+        EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(2, audio_type);  // Comfort noise.
+        if (time >= kCheckTimeMs) {
+          CheckAudioBounded(output_data_decode, samples, channels_,
+                            kOutputValueBound);
+        }
+      } else {
+        EXPECT_GT(encoded_bytes_, 1U);
+        EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(0, audio_type);  // Speech.
+      }
+    }
+
     if (dtx) {
-      EXPECT_EQ(0, encoded_bytes_)  // Send 0 byte.
-          << "Opus should have entered DTX mode.";
+      // With DTX, Opus must stop transmission for some time.
+      EXPECT_GT(i, 1);
+    }
+
+    // We expect a normal payload.
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+
+    // Enters DTX again immediately.
+    time += block_length_ms;
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
+    if (dtx) {
+      EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
       EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(2, audio_type);  // Comfort noise.
+      if (time >= kCheckTimeMs) {
+        CheckAudioBounded(output_data_decode, samples, channels_,
+                          kOutputValueBound);
+      }
     } else {
-      EXPECT_GT(encoded_bytes_, 1);
+      EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(0, audio_type);  // Speech.
     }
   }
 
-  // Quit DTX after 400 ms
-  EXPECT_EQ(kOpus20msFrameSamples,
-            EncodeDecode(opus_encoder_, silence,
-                         kOpus20msFrameSamples, opus_decoder_,
-                         output_data_decode, &audio_type));
-
-  EXPECT_GT(encoded_bytes_, 1);
-  EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
-  EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
-  EXPECT_EQ(0, audio_type);  // Speech.
-
-  // Enters DTX again immediately.
-  EXPECT_EQ(kOpus20msFrameSamples,
-            EncodeDecode(opus_encoder_, silence,
-                         kOpus20msFrameSamples, opus_decoder_,
-                         output_data_decode, &audio_type));
-  if (dtx) {
-    EXPECT_EQ(1, encoded_bytes_);  // Send 1 byte.
-    EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
-    EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
-    EXPECT_EQ(2, audio_type);  // Comfort noise.
-  } else {
-    EXPECT_GT(encoded_bytes_, 1);
-    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
-    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
-    EXPECT_EQ(0, audio_type);  // Speech.
-  }
-
   silence[0] = 10000;
   if (dtx) {
     // Verify that encoder/decoder can jump out from DTX mode.
-    EXPECT_EQ(kOpus20msFrameSamples,
-              EncodeDecode(opus_encoder_, silence,
-                           kOpus20msFrameSamples, opus_decoder_,
-                           output_data_decode, &audio_type));
-    EXPECT_GT(encoded_bytes_, 1);
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
+    EXPECT_GT(encoded_bytes_, 1U);
     EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
     EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
     EXPECT_EQ(0, audio_type);  // Speech.
@@ -234,7 +289,6 @@ void OpusTest::TestDtxEffect(bool dtx) {
 
   // Free memory.
   delete[] output_data_decode;
-  delete[] silence;
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
@@ -304,9 +358,9 @@ TEST_P(OpusTest, OpusEncodeDecode) {
   int16_t audio_type;
   int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
-                         kOpus20msFrameSamples, opus_decoder_,
-                         output_data_decode, &audio_type));
+            static_cast<size_t>(
+                EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                             opus_decoder_, output_data_decode, &audio_type)));
 
   // Free memory.
   delete[] output_data_decode;
@@ -363,16 +417,16 @@ TEST_P(OpusTest, OpusDecodeInit) {
   int16_t audio_type;
   int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
-                         kOpus20msFrameSamples, opus_decoder_,
-                         output_data_decode, &audio_type));
+            static_cast<size_t>(
+                EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                             opus_decoder_, output_data_decode, &audio_type)));
 
-  EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_decoder_));
+  WebRtcOpus_DecoderInit(opus_decoder_);
 
   EXPECT_EQ(kOpus20msFrameSamples,
-            WebRtcOpus_Decode(opus_decoder_, bitstream_,
-                              encoded_bytes_, output_data_decode,
-                              &audio_type));
+            static_cast<size_t>(WebRtcOpus_Decode(
+                opus_decoder_, bitstream_, encoded_bytes_, output_data_decode,
+                &audio_type)));
 
   // Free memory.
   delete[] output_data_decode;
@@ -432,15 +486,15 @@ TEST_P(OpusTest, OpusEnableDisableDtx) {
 }
 
 TEST_P(OpusTest, OpusDtxOff) {
-  TestDtxEffect(false);
+  TestDtxEffect(false, 10);
+  TestDtxEffect(false, 20);
+  TestDtxEffect(false, 40);
 }
 
 TEST_P(OpusTest, OpusDtxOn) {
-  if (application_ == 1) {
-    // We do not check DTX under OPUS_APPLICATION_AUDIO mode.
-    return;
-  }
-  TestDtxEffect(true);
+  TestDtxEffect(true, 10);
+  TestDtxEffect(true, 20);
+  TestDtxEffect(true, 40);
 }
 
 TEST_P(OpusTest, OpusSetPacketLossRate) {
@@ -505,14 +559,15 @@ TEST_P(OpusTest, OpusDecodePlc) {
   int16_t audio_type;
   int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
-                         kOpus20msFrameSamples, opus_decoder_,
-                         output_data_decode, &audio_type));
+            static_cast<size_t>(
+                EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                             opus_decoder_, output_data_decode, &audio_type)));
 
   // Call decoder PLC.
   int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_];
   EXPECT_EQ(kOpus20msFrameSamples,
-            WebRtcOpus_DecodePlc(opus_decoder_, plc_buffer, 1));
+            static_cast<size_t>(WebRtcOpus_DecodePlc(
+                opus_decoder_, plc_buffer, 1)));
 
   // Free memory.
   delete[] plc_buffer;
@@ -532,22 +587,28 @@ TEST_P(OpusTest, OpusDurationEstimation) {
   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
 
   // 10 ms. We use only first 10 ms of a 20 ms block.
-  encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
-                                     speech_data_.GetNextBlock(),
-                                     kOpus10msFrameSamples, kMaxBytes,
-                                     bitstream_);
+  auto speech_block = speech_data_.GetNextBlock();
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), 2 * channels_),
+      kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
   EXPECT_EQ(kOpus10msFrameSamples,
-            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
-                                   encoded_bytes_));
+            static_cast<size_t>(WebRtcOpus_DurationEst(
+                opus_decoder_, bitstream_,
+                static_cast<size_t>(encoded_bytes_int))));
 
   // 20 ms
-  encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
-                                     speech_data_.GetNextBlock(),
-                                     kOpus20msFrameSamples, kMaxBytes,
-                                     bitstream_);
+  speech_block = speech_data_.GetNextBlock();
+  encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), channels_),
+      kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
   EXPECT_EQ(kOpus20msFrameSamples,
-            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
-                                   encoded_bytes_));
+            static_cast<size_t>(WebRtcOpus_DurationEst(
+                opus_decoder_, bitstream_,
+                static_cast<size_t>(encoded_bytes_int))));
 
   // Free memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
@@ -580,21 +641,24 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
   OpusRepacketizer* rp = opus_repacketizer_create();
 
   for (int idx = 0; idx < kPackets; idx++) {
-    encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
-                                       speech_data_.GetNextBlock(),
-                                       kOpus20msFrameSamples, kMaxBytes,
-                                       bitstream_);
+    auto speech_block = speech_data_.GetNextBlock();
+    encoded_bytes_ =
+        WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                          rtc::CheckedDivExact(speech_block.size(), channels_),
+                          kMaxBytes, bitstream_);
     EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
   }
 
   encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
 
   EXPECT_EQ(kOpus20msFrameSamples * kPackets,
-            WebRtcOpus_DurationEst(opus_decoder_, bitstream_, encoded_bytes_));
+            static_cast<size_t>(WebRtcOpus_DurationEst(
+                opus_decoder_, bitstream_, encoded_bytes_)));
 
   EXPECT_EQ(kOpus20msFrameSamples * kPackets,
-            WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
-                              output_data_decode.get(), &audio_type));
+            static_cast<size_t>(WebRtcOpus_Decode(
+                opus_decoder_, bitstream_, encoded_bytes_,
+                output_data_decode.get(), &audio_type)));
 
   // Free memory.
   opus_repacketizer_destroy(rp);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
new file mode 100644
index 0000000000..834c070073
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+
+namespace webrtc {
+
+AudioDecoderPcm16B::AudioDecoderPcm16B(size_t num_channels)
+    : num_channels_(num_channels) {
+  RTC_DCHECK_GE(num_channels, 1u);
+}
+
+void AudioDecoderPcm16B::Reset() {}
+
+size_t AudioDecoderPcm16B::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
+                                       size_t encoded_len,
+                                       int sample_rate_hz,
+                                       int16_t* decoded,
+                                       SpeechType* speech_type) {
+  RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+             sample_rate_hz == 32000 || sample_rate_hz == 48000)
+      << "Unsupported sample rate " << sample_rate_hz;
+  size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
+  *speech_type = ConvertSpeechType(1);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
+                                       size_t encoded_len) const {
+  // Two encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / (2 * Channels()));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
new file mode 100644
index 0000000000..692cb94282
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+namespace webrtc {
+
+class AudioDecoderPcm16B final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcm16B(size_t num_channels);
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index f761922201..f4d4022302 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -8,15 +8,44 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 
 namespace webrtc {
 
-int16_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
-                                       size_t input_len,
-                                       uint8_t* encoded) {
-  return WebRtcPcm16b_Encode(audio, static_cast<int16_t>(input_len), encoded);
+size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
+                                      size_t input_len,
+                                      uint8_t* encoded) {
+  return WebRtcPcm16b_Encode(audio, input_len, encoded);
 }
 
+size_t AudioEncoderPcm16B::BytesPerSample() const {
+  return 2;
+}
+
+namespace {
+AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderPcm16B::Config config;
+  config.num_channels = codec_inst.channels;
+  config.sample_rate_hz = codec_inst.plfreq;
+  config.frame_size_ms = rtc::CheckedDivExact(
+      codec_inst.pacsize, rtc::CheckedDivExact(config.sample_rate_hz, 1000));
+  config.payload_type = codec_inst.pltype;
+  return config;
+}
+}  // namespace
+
+bool AudioEncoderPcm16B::Config::IsOk() const {
+  if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) &&
+      (sample_rate_hz != 32000) && (sample_rate_hz != 48000))
+    return false;
+  return AudioEncoderPcm::Config::IsOk();
+}
+
+AudioEncoderPcm16B::AudioEncoderPcm16B(const CodecInst& codec_inst)
+    : AudioEncoderPcm16B(CreateConfig(codec_inst)) {}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
similarity index 51%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 99ecd249c1..68ca2da77e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -8,30 +8,41 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_ENCODER_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_ENCODER_PCM16B_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 namespace webrtc {
 
-class AudioEncoderPcm16B : public AudioEncoderPcm {
+struct CodecInst;
+
+class AudioEncoderPcm16B final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
    public:
     Config() : AudioEncoderPcm::Config(107), sample_rate_hz(8000) {}
+    bool IsOk() const;
 
     int sample_rate_hz;
   };
 
   explicit AudioEncoderPcm16B(const Config& config)
       : AudioEncoderPcm(config, config.sample_rate_hz) {}
+  explicit AudioEncoderPcm16B(const CodecInst& codec_inst);
 
  protected:
-  int16_t EncodeCall(const int16_t* audio,
-                     size_t input_len,
-                     uint8_t* encoded) override;
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_INCLUDE_AUDIO_ENCODER_PCM16B_H_
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
index b6de0b5e67..120c79052b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -12,10 +12,10 @@
 
 #include "webrtc/typedefs.h"
 
-int16_t WebRtcPcm16b_Encode(const int16_t* speech,
-                            int16_t len,
-                            uint8_t* encoded) {
-  int i;
+size_t WebRtcPcm16b_Encode(const int16_t* speech,
+                           size_t len,
+                           uint8_t* encoded) {
+  size_t i;
   for (i = 0; i < len; ++i) {
     uint16_t s = speech[i];
     encoded[2 * i] = s >> 8;
@@ -24,10 +24,10 @@ int16_t WebRtcPcm16b_Encode(const int16_t* speech,
   return 2 * len;
 }
 
-int16_t WebRtcPcm16b_Decode(const uint8_t* encoded,
-                            int16_t len,
-                            int16_t* speech) {
-  int i;
+size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
+                           size_t len,
+                           int16_t* speech) {
+  size_t i;
   for (i = 0; i < len / 2; ++i)
     speech[i] = encoded[2 * i] << 8 | encoded[2 * i + 1];
   return len / 2;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
index 44b4335407..d0dd21bb60 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
@@ -9,27 +9,19 @@
 {
   'targets': [
     {
-      'target_name': 'PCM16B',
+      'target_name': 'pcm16b',
       'type': 'static_library',
       'dependencies': [
         'audio_encoder_interface',
-        'G711',
+        'g711',
       ],
-      'include_dirs': [
-        'include',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'include',
-          '<(webrtc_root)',
-        ],
-      },
       'sources': [
-        'include/audio_encoder_pcm16b.h',
-        'include/pcm16b.h',
+        'audio_decoder_pcm16b.cc',
+        'audio_decoder_pcm16b.h',
         'audio_encoder_pcm16b.cc',
+        'audio_encoder_pcm16b.h',
         'pcm16b.c',
+        'pcm16b.h',
       ],
     },
   ], # targets
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
similarity index 76%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
index 1cdf92dbf8..f96e741c46 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
@@ -8,12 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INTERFACE_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INTERFACE_PCM16B_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
 /*
  * Define the fixpoint numeric formats
  */
 
+#include <stddef.h>
+
 #include "webrtc/typedefs.h"
 
 #ifdef __cplusplus
@@ -36,9 +38,9 @@ extern "C" {
  *                                Always equal to twice the len input parameter.
  */
 
-int16_t WebRtcPcm16b_Encode(const int16_t* speech,
-                            int16_t len,
-                            uint8_t* encoded);
+size_t WebRtcPcm16b_Encode(const int16_t* speech,
+                           size_t len,
+                           uint8_t* encoded);
 
 /****************************************************************************
  * WebRtcPcm16b_Decode(...)
@@ -55,12 +57,12 @@ int16_t WebRtcPcm16b_Encode(const int16_t* speech,
  * Returned value               : Samples in speech
  */
 
-int16_t WebRtcPcm16b_Decode(const uint8_t* encoded,
-                            int16_t len,
-                            int16_t* speech);
+size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
+                           size_t len,
+                           int16_t* speech);
 
 #ifdef __cplusplus
 }
 #endif
 
-#endif /* PCM16B */
+#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 86f1158d9d..7ef1ce096b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -18,81 +18,67 @@ namespace webrtc {
 
 AudioEncoderCopyRed::AudioEncoderCopyRed(const Config& config)
     : speech_encoder_(config.speech_encoder),
-      red_payload_type_(config.payload_type),
-      secondary_allocated_(0) {
-  CHECK(speech_encoder_) << "Speech encoder not provided.";
+      red_payload_type_(config.payload_type) {
+  RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
 }
 
-AudioEncoderCopyRed::~AudioEncoderCopyRed() {
+AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
+
+size_t AudioEncoderCopyRed::MaxEncodedBytes() const {
+  return 2 * speech_encoder_->MaxEncodedBytes();
 }
 
 int AudioEncoderCopyRed::SampleRateHz() const {
   return speech_encoder_->SampleRateHz();
 }
 
+size_t AudioEncoderCopyRed::NumChannels() const {
+  return speech_encoder_->NumChannels();
+}
+
 int AudioEncoderCopyRed::RtpTimestampRateHz() const {
   return speech_encoder_->RtpTimestampRateHz();
 }
 
-int AudioEncoderCopyRed::NumChannels() const {
-  return speech_encoder_->NumChannels();
-}
-
-size_t AudioEncoderCopyRed::MaxEncodedBytes() const {
-  return 2 * speech_encoder_->MaxEncodedBytes();
-}
-
-int AudioEncoderCopyRed::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderCopyRed::Num10MsFramesInNextPacket() const {
   return speech_encoder_->Num10MsFramesInNextPacket();
 }
 
-int AudioEncoderCopyRed::Max10MsFramesInAPacket() const {
+size_t AudioEncoderCopyRed::Max10MsFramesInAPacket() const {
   return speech_encoder_->Max10MsFramesInAPacket();
 }
 
-void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) {
-  speech_encoder_->SetTargetBitrate(bits_per_second);
-}
-
-void AudioEncoderCopyRed::SetProjectedPacketLossRate(double fraction) {
-  DCHECK_GE(fraction, 0.0);
-  DCHECK_LE(fraction, 1.0);
-  speech_encoder_->SetProjectedPacketLossRate(fraction);
+int AudioEncoderCopyRed::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
 }
 
 AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  EncodedInfo info = speech_encoder_->Encode(
-      rtp_timestamp, audio, static_cast<size_t>(SampleRateHz() / 100),
-      max_encoded_bytes, encoded);
-  CHECK_GE(max_encoded_bytes,
-           info.encoded_bytes + secondary_info_.encoded_bytes);
-  CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
+  EncodedInfo info =
+      speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
+  RTC_CHECK_GE(max_encoded_bytes,
+               info.encoded_bytes + secondary_info_.encoded_bytes);
+  RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
 
   if (info.encoded_bytes > 0) {
     // |info| will be implicitly cast to an EncodedInfoLeaf struct, effectively
     // discarding the (empty) vector of redundant information. This is
     // intentional.
     info.redundant.push_back(info);
-    DCHECK_EQ(info.redundant.size(), 1u);
+    RTC_DCHECK_EQ(info.redundant.size(), 1u);
     if (secondary_info_.encoded_bytes > 0) {
-      memcpy(&encoded[info.encoded_bytes], secondary_encoded_.get(),
+      memcpy(&encoded[info.encoded_bytes], secondary_encoded_.data(),
              secondary_info_.encoded_bytes);
       info.redundant.push_back(secondary_info_);
-      DCHECK_EQ(info.redundant.size(), 2u);
+      RTC_DCHECK_EQ(info.redundant.size(), 2u);
     }
     // Save primary to secondary.
-    if (secondary_allocated_ < info.encoded_bytes) {
-      secondary_encoded_.reset(new uint8_t[info.encoded_bytes]);
-      secondary_allocated_ = info.encoded_bytes;
-    }
-    CHECK(secondary_encoded_);
-    memcpy(secondary_encoded_.get(), encoded, info.encoded_bytes);
+    secondary_encoded_.SetData(encoded, info.encoded_bytes);
     secondary_info_ = info;
-    DCHECK_EQ(info.speech, info.redundant[0].speech);
+    RTC_DCHECK_EQ(info.speech, info.redundant[0].speech);
   }
   // Update main EncodedInfo.
   info.payload_type = red_payload_type_;
@@ -104,4 +90,34 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal(
   return info;
 }
 
+void AudioEncoderCopyRed::Reset() {
+  speech_encoder_->Reset();
+  secondary_encoded_.Clear();
+  secondary_info_.encoded_bytes = 0;
+}
+
+bool AudioEncoderCopyRed::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCopyRed::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCopyRed::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+void AudioEncoderCopyRed::SetProjectedPacketLossRate(double fraction) {
+  speech_encoder_->SetProjectedPacketLossRate(fraction);
+}
+
+void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) {
+  speech_encoder_->SetTargetBitrate(bits_per_second);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index fd92d52457..2f53765389 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
@@ -22,7 +23,7 @@ namespace webrtc {
 // underlying AudioEncoder object that performs the actual encodings. The
 // current class will gather the two latest encodings from the underlying codec
 // into one packet.
-class AudioEncoderCopyRed : public AudioEncoder {
+class AudioEncoderCopyRed final : public AudioEncoder {
  public:
   struct Config {
    public:
@@ -35,28 +36,33 @@ class AudioEncoderCopyRed : public AudioEncoder {
 
   ~AudioEncoderCopyRed() override;
 
-  int SampleRateHz() const override;
-  int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
-  void SetTargetBitrate(int bits_per_second) override;
-  void SetProjectedPacketLossRate(double fraction) override;
-
- protected:
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
+                             rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
+  void Reset() override;
+  bool SetFec(bool enable) override;
+  bool SetDtx(bool enable) override;
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
 
  private:
   AudioEncoder* speech_encoder_;
   int red_payload_type_;
-  rtc::scoped_ptr<uint8_t[]> secondary_encoded_;
-  size_t secondary_allocated_;
+  rtc::Buffer secondary_encoded_;
   EncodedInfoLeaf secondary_info_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCopyRed);
 };
 
 }  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 4debdfab8d..22601b6597 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -42,7 +42,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
     config.speech_encoder = &mock_encoder_;
     red_.reset(new AudioEncoderCopyRed(config));
     memset(audio_, 0, sizeof(audio_));
-    EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+    EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
     EXPECT_CALL(mock_encoder_, SampleRateHz())
         .WillRepeatedly(Return(sample_rate_hz_));
     EXPECT_CALL(mock_encoder_, MaxEncodedBytes())
@@ -60,8 +60,10 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
 
   void Encode() {
     ASSERT_TRUE(red_.get() != NULL);
-    encoded_info_ = red_->Encode(timestamp_, audio_, num_audio_samples_10ms,
-                                 encoded_.size(), &encoded_[0]);
+    encoded_info_ = red_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+        encoded_.size(), &encoded_[0]);
     timestamp_ += num_audio_samples_10ms;
   }
 
@@ -83,12 +85,12 @@ class MockEncodeHelper {
   }
 
   AudioEncoder::EncodedInfo Encode(uint32_t timestamp,
-                                   const int16_t* audio,
+                                   rtc::ArrayView<const int16_t> audio,
                                    size_t max_encoded_bytes,
                                    uint8_t* encoded) {
     if (write_payload_) {
-      CHECK(encoded);
-      CHECK_LE(info_.encoded_bytes, max_encoded_bytes);
+      RTC_CHECK(encoded);
+      RTC_CHECK_LE(info_.encoded_bytes, max_encoded_bytes);
       memcpy(encoded, payload_, info_.encoded_bytes);
     }
     return info_;
@@ -108,18 +110,18 @@ TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
-  EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->NumChannels());
+  EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->NumChannels());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
-  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->Num10MsFramesInNextPacket());
+  EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
-  EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->Max10MsFramesInAPacket());
+  EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckSetBitratePropagation) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index c7cafdff9b..3dc665482a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::tr1::get;
@@ -23,8 +24,10 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
     : block_duration_ms_(block_duration_ms),
       input_sampling_khz_(input_sampling_khz),
       output_sampling_khz_(output_sampling_khz),
-      input_length_sample_(block_duration_ms_ * input_sampling_khz_),
-      output_length_sample_(block_duration_ms_ * output_sampling_khz_),
+      input_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
+      output_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
       data_pointer_(0),
       loop_length_samples_(0),
       max_bytes_(0),
@@ -97,7 +100,7 @@ void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
   size_t time_now_ms = 0;
   float time_ms;
 
-  printf("Coding %d kHz-sampled %d-channel audio at %d bps ...\n",
+  printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
          input_sampling_khz_, channels_, bit_rate_);
 
   while (time_now_ms < audio_duration_sec * 1000) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index 35ac69e8ab..fb7b3e5b1e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -20,7 +20,8 @@ namespace webrtc {
 
 // Define coding parameter as
 // <channels, bit_rate, file_name, extension, if_save_output>.
-typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param;
+typedef std::tr1::tuple<size_t, int, std::string, std::string, bool>
+    coding_param;
 
 class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
  protected:
@@ -36,14 +37,14 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   // 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
   // 4. return the cost of time (in millisecond) spent on actual encoding.
   virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             int max_bytes, int* encoded_bytes) = 0;
+                             size_t max_bytes, size_t* encoded_bytes) = 0;
 
   // DecodeABlock(...) does the following:
   // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
   // (in bytes),
   // 2. save the decoded audio in |out_data|,
   // 3. return the cost of time (in millisecond) spent on actual decoding.
-  virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
                              int16_t* out_data) = 0;
 
   // Encoding and decode an audio of |audio_duration| (in seconds) and
@@ -55,10 +56,10 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   int output_sampling_khz_;
 
   // Number of samples-per-channel in a frame.
-  int input_length_sample_;
+  size_t input_length_sample_;
 
   // Expected output number of samples-per-channel in a frame.
-  int output_length_sample_;
+  size_t output_length_sample_;
 
   rtc::scoped_ptr<int16_t[]> in_data_;
   rtc::scoped_ptr<int16_t[]> out_data_;
@@ -67,14 +68,14 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   rtc::scoped_ptr<uint8_t[]> bit_stream_;
 
   // Maximum number of bytes in output bitstream for a frame of audio.
-  int max_bytes_;
+  size_t max_bytes_;
 
-  int encoded_bytes_;
+  size_t encoded_bytes_;
   float encoding_time_ms_;
   float decoding_time_ms_;
   FILE* out_file_;
 
-  int channels_;
+  size_t channels_;
 
   // Bit rate is in bit-per-second.
   int bit_rate_;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/media/webrtc/trunk/webrtc/modules/audio_coding/include/audio_coding_module.h
similarity index 56%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/include/audio_coding_module.h
index 796444e78b..9e7991f22f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -8,17 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 
+#include <string>
 #include <vector>
 
+#include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -26,6 +27,8 @@ namespace webrtc {
 // forward declarations
 struct CodecInst;
 struct WebRtcRTPHeader;
+class AudioDecoder;
+class AudioEncoder;
 class AudioFrame;
 class RTPFragmentationHeader;
 
@@ -44,15 +47,6 @@ class AudioPacketizationCallback {
                            const RTPFragmentationHeader* fragmentation) = 0;
 };
 
-// Callback class used for inband Dtmf detection
-class AudioCodingFeedback {
- public:
-  virtual ~AudioCodingFeedback() {}
-
-  virtual int32_t IncomingDtmf(const uint8_t digit_dtmf,
-                               const bool end) = 0;
-};
-
 // Callback class used for reporting VAD decision
 class ACMVADCallback {
  public:
@@ -61,29 +55,17 @@ class ACMVADCallback {
   virtual int32_t InFrameType(FrameType frame_type) = 0;
 };
 
-// Callback class used for reporting receiver statistics
-class ACMVQMonCallback {
- public:
-  virtual ~ACMVQMonCallback() {}
-
-  virtual int32_t NetEqStatistics(
-      const int32_t id,  // current ACM id
-      const uint16_t MIUsValid,  // valid voice duration in ms
-      const uint16_t MIUsReplaced,  // concealed voice duration in ms
-      const uint8_t eventFlags,  // concealed voice flags
-      const uint16_t delayMS) = 0;  // average delay in ms
-};
-
 class AudioCodingModule {
  protected:
   AudioCodingModule() {}
 
  public:
   struct Config {
-    Config()
-        : id(0),
-          neteq_config(),
-          clock(Clock::GetRealTimeClock()) {}
+    Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
+      // Post-decode VAD is disabled by default in NetEq, however, Audio
+      // Conference Mixer relies on VAD decisions and fails without them.
+      neteq_config.enable_post_decode_vad = true;
+    }
 
     int id;
     NetEq::Config neteq_config;
@@ -99,7 +81,8 @@ class AudioCodingModule {
   //
   static AudioCodingModule* Create(int id);
   static AudioCodingModule* Create(int id, Clock* clock);
-  virtual ~AudioCodingModule() {};
+  static AudioCodingModule* Create(const Config& config);
+  virtual ~AudioCodingModule() = default;
 
   ///////////////////////////////////////////////////////////////////////////
   //   Utility functions
@@ -151,7 +134,7 @@ class AudioCodingModule {
   //    0 if succeeded.
   //
   static int Codec(const char* payload_name, CodecInst* codec,
-                       int sampling_freq_hz, int channels);
+                   int sampling_freq_hz, size_t channels);
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t Codec()
@@ -170,7 +153,7 @@ class AudioCodingModule {
   //   -1 if the codec is not found.
   //
   static int Codec(const char* payload_name, int sampling_freq_hz,
-                             int channels);
+                   size_t channels);
 
   ///////////////////////////////////////////////////////////////////////////
   // bool IsCodecValid()
@@ -190,17 +173,6 @@ class AudioCodingModule {
   //   Sender
   //
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t ResetEncoder()
-  // This API resets the states of encoder. All the encoder settings, such as
-  // send-codec or VAD/DTX, will be preserved.
-  //
-  // Return value:
-  //   -1 if failed to initialize,
-  //    0 if succeeded.
-  //
-  virtual int32_t ResetEncoder() = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   // int32_t RegisterSendCodec()
   // Registers a codec, specified by |send_codec|, as sending codec.
@@ -230,18 +202,19 @@ class AudioCodingModule {
   //
   virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0;
 
+  // Registers |external_speech_encoder| as encoder. The new encoder will
+  // replace any previously registered speech encoder (internal or external).
+  virtual void RegisterExternalSendCodec(
+      AudioEncoder* external_speech_encoder) = 0;
+
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendCodec()
   // Get parameters for the codec currently registered as send codec.
   //
-  // Output:
-  //   -current_send_codec          : parameters of the send codec.
-  //
   // Return value:
-  //   -1 if failed to get send codec,
-  //    0 if succeeded.
+  //   The send codec, or nothing if we don't have one
   //
-  virtual int32_t SendCodec(CodecInst* current_send_codec) const = 0;
+  virtual rtc::Optional<CodecInst> SendCodec() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t SendFrequency()
@@ -254,33 +227,10 @@ class AudioCodingModule {
   virtual int32_t SendFrequency() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // int32_t Bitrate()
-  // Get encoding bit-rate in bits per second.
-  //
-  // Return value:
-  //   positive; encoding rate in bits/sec,
-  //   -1 if an error is happened.
-  //
-  virtual int32_t SendBitrate() const = 0;
+  // Sets the bitrate to the specified value in bits/sec. If the value is not
+  // supported by the codec, it will choose another appropriate value.
+  virtual void SetBitRate(int bitrate_bps) = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetReceivedEstimatedBandwidth()
-  // Set available bandwidth [bits/sec] of the up-link channel.
-  // This information is used for traffic shaping, and is currently only
-  // supported if iSAC is the send codec.
-  //
-  // Input:
-  //   -bw                 : bandwidth in bits/sec estimated for
-  //                         up-link.
-  // Return value
-  //   -1 if error occurred in setting the bandwidth,
-  //    0 bandwidth is set successfully.
-  //
-  // TODO(henrik.lundin) Unused. Remove?
-  virtual int32_t SetReceivedEstimatedBandwidth(
-      const int32_t bw) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
   // int32_t RegisterTransportCallback()
   // Register a transport callback which will be called to deliver
   // the encoded buffers whenever Process() is called and a
@@ -452,39 +402,6 @@ class AudioCodingModule {
   virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
                             ACMVADMode* vad_mode) const = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t ReplaceInternalDTXWithWebRtc()
-  // Used to replace codec internal DTX scheme with WebRtc.
-  //
-  // Input:
-  //   -use_webrtc_dtx     : if false (default) the codec built-in DTX/VAD
-  //                         scheme is used, otherwise the internal DTX is
-  //                         replaced with WebRtc DTX/VAD.
-  //
-  // Return value:
-  //   -1 if failed to replace codec internal DTX with WebRtc,
-  //    0 if succeeded.
-  //
-  virtual int32_t ReplaceInternalDTXWithWebRtc(
-      const bool use_webrtc_dtx = false) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t IsInternalDTXReplacedWithWebRtc()
-  // Get status if the codec internal DTX is replaced with WebRtc DTX.
-  // This should always be true if codec does not have an internal DTX.
-  //
-  // Output:
-  //   -uses_webrtc_dtx    : is set to true if the codec internal DTX is
-  //                         replaced with WebRtc DTX/VAD, otherwise it is set
-  //                         to false.
-  //
-  // Return value:
-  //   -1 if failed to determine if codec internal DTX is replaced with WebRtc,
-  //    0 if succeeded.
-  //
-  virtual int32_t IsInternalDTXReplacedWithWebRtc(
-      bool* uses_webrtc_dtx) = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   // int32_t RegisterVADCallback()
   // Call this method to register a callback function which is called
@@ -520,17 +437,6 @@ class AudioCodingModule {
   //
   virtual int32_t InitializeReceiver() = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t ResetDecoder()
-  // This API resets the states of decoders. ACM will not lose any
-  // decoder-related settings, such as registered codecs.
-  //
-  // Return value:
-  //   -1 if failed to initialize,
-  //    0 if succeeded.
-  //
-  virtual int32_t ResetDecoder() = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   // int32_t ReceiveFrequency()
   // Get sampling frequency of the last received payload.
@@ -564,8 +470,16 @@ class AudioCodingModule {
   //   -1 if failed to register the codec
   //    0 if the codec registered successfully.
   //
-  virtual int32_t RegisterReceiveCodec(
-      const CodecInst& receive_codec) = 0;
+  virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
+
+  // Registers an external decoder. The name is only used to provide information
+  // back to the caller about the decoder. Hence, the name is arbitrary, and may
+  // be empty.
+  virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
+                                           AudioDecoder* external_decoder,
+                                           int sample_rate_hz,
+                                           int num_channels,
+                                           const std::string& name) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t UnregisterReceiveCodec()
@@ -680,31 +594,6 @@ class AudioCodingModule {
   //
   virtual int LeastRequiredDelayMs() const = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetDtmfPlayoutStatus()
-  // Configure DTMF playout, i.e. whether out-of-band
-  // DTMF tones are played or not.
-  //
-  // Input:
-  //   -enable             : if true to enable playout out-of-band DTMF tones,
-  //                         false to disable.
-  //
-  // Return value:
-  //   -1 if the method fails, e.g. DTMF playout is not supported.
-  //    0 if the status is set successfully.
-  //
-  virtual int32_t SetDtmfPlayoutStatus(const bool enable) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // bool DtmfPlayoutStatus()
-  // Get Dtmf playout status.
-  //
-  // Return value:
-  //   true if out-of-band Dtmf tones are played,
-  //   false if playout of Dtmf tones is disabled.
-  //
-  virtual bool DtmfPlayoutStatus() const = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded
@@ -721,56 +610,6 @@ class AudioCodingModule {
   // TODO(tlegrand): Change function to return the timestamp.
   virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t DecoderEstimatedBandwidth()
-  // Get the estimate of the Bandwidth, in bits/second, based on the incoming
-  // stream. This API is useful in one-way communication scenarios, where
-  // the bandwidth information is sent in an out-of-band fashion.
-  // Currently only supported if iSAC is registered as a receiver.
-  //
-  // Return value:
-  //   >0 bandwidth in bits/second.
-  //   -1 if failed to get a bandwidth estimate.
-  //
-  virtual int32_t DecoderEstimatedBandwidth() const = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetPlayoutMode()
-  // Call this API to set the playout mode. Playout mode could be optimized
-  // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is
-  // optimized to deliver highest audio quality while maintaining a minimum
-  // delay. In FAX mode, NetEQ is optimized to have few delay changes as
-  // possible and maintain a constant delay, perhaps large relative to voice
-  // mode, to avoid PLC. In streaming mode, we tolerate a little more delay
-  // to achieve better jitter robustness.
-  //
-  // Input:
-  //   -mode               : playout mode. Possible inputs are:
-  //                         "voice",
-  //                         "fax" and
-  //                         "streaming".
-  //
-  // Return value:
-  //   -1 if failed to set the mode,
-  //    0 if succeeding.
-  //
-  virtual int32_t SetPlayoutMode(const AudioPlayoutMode mode) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // AudioPlayoutMode PlayoutMode()
-  // Get playout mode, i.e. whether it is speech, FAX or streaming. See
-  // audio_coding_module_typedefs.h for definition of AudioPlayoutMode.
-  //
-  // Return value:
-  //   voice:       is for voice output,
-  //   fax:         a mode that is optimized for receiving FAX signals.
-  //                In this mode NetEq tries to maintain a constant high
-  //                delay to avoid PLC if possible.
-  //   streaming:   a mode that is suitable for streaming. In this mode we
-  //                accept longer delay to improve jitter robustness.
-  //
-  virtual AudioPlayoutMode PlayoutMode() const = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutData10Ms(
   // Get 10 milliseconds of raw audio data for playout, at the given sampling
@@ -799,89 +638,20 @@ class AudioCodingModule {
   //
 
   ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetISACMaxRate()
-  // Set the maximum instantaneous rate of iSAC. For a payload of B bits
-  // with a frame-size of T sec the instantaneous rate is B/T bits per
-  // second. Therefore, (B/T < |max_rate_bps|) and
-  // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads,
-  // c.f SetISACMaxPayloadSize().
-  //
-  // Input:
-  //   -max_rate_bps       : maximum instantaneous bit-rate given in bits/sec.
-  //
-  // Return value:
-  //   -1 if failed to set the maximum rate.
-  //    0 if the maximum rate is set successfully.
-  //
-  virtual int SetISACMaxRate(int max_rate_bps) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetISACMaxPayloadSize()
-  // Set the maximum payload size of iSAC packets. No iSAC payload,
-  // regardless of its frame-size, may exceed the given limit. For
-  // an iSAC payload of size B bits and frame-size T seconds we have;
-  // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f.
-  // SetISACMaxRate().
-  //
-  // Input:
-  //   -max_payload_len_bytes : maximum payload size in bytes.
-  //
-  // Return value:
-  //   -1 if failed to set the maximum  payload-size.
-  //    0 if the given length is set successfully.
-  //
-  virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t ConfigISACBandwidthEstimator()
-  // Call this function to configure the bandwidth estimator of ISAC.
-  // During the adaptation of bit-rate, iSAC automatically adjusts the
-  // frame-size (either 30 or 60 ms) to save on RTP header. The initial
-  // frame-size can be specified by the first argument. The configuration also
-  // regards the initial estimate of bandwidths. The estimator starts from
-  // this point and converges to the actual bottleneck. This is given by the
-  // second parameter. Furthermore, it is also possible to control the
-  // adaptation of frame-size. This is specified by the last parameter.
-  //
-  // Input:
-  //   -init_frame_size_ms : initial frame-size in milliseconds. For iSAC-wb
-  //                         30 ms and 60 ms (default) are acceptable values,
-  //                         and for iSAC-swb 30 ms is the only acceptable
-  //                         value. Zero indicates default value.
-  //   -init_rate_bps      : initial estimate of the bandwidth. Values
-  //                         between 10000 and 58000 are acceptable.
-  //   -enforce_srame_size : if true, the frame-size will not be adapted.
-  //
-  // Return value:
-  //   -1 if failed to configure the bandwidth estimator,
-  //    0 if the configuration was successfully applied.
-  //
-  virtual int32_t ConfigISACBandwidthEstimator(
-      int init_frame_size_ms,
-      int init_rate_bps,
-      bool enforce_frame_size = false) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int SetOpusApplication(OpusApplicationMode application,
-  //                        bool disable_dtx_if_needed)
+  // int SetOpusApplication()
   // Sets the intended application if current send codec is Opus. Opus uses this
   // to optimize the encoding for applications like VOIP and music. Currently,
-  // two modes are supported: kVoip and kAudio. kAudio is only allowed when Opus
-  // DTX is switched off. If DTX is on, and |application| == kAudio, a failure
-  // will be triggered unless |disable_dtx_if_needed| == true, for which, the
-  // DTX will be forced off.
+  // two modes are supported: kVoip and kAudio.
   //
   // Input:
   //   - application            : intended application.
-  //   - disable_dtx_if_needed  : whether to force Opus DTX to stop.
   //
   // Return value:
   //   -1 if current send codec is not Opus or error occurred in setting the
   //      Opus application mode.
   //    0 if the Opus application mode is successfully set.
   //
-  virtual int SetOpusApplication(OpusApplicationMode application,
-                                 bool force_dtx) = 0;
+  virtual int SetOpusApplication(OpusApplicationMode application) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int SetOpusMaxPlaybackRate()
@@ -900,18 +670,15 @@ class AudioCodingModule {
   virtual int SetOpusMaxPlaybackRate(int frequency_hz) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // EnableOpusDtx(bool force_voip)
-  // Enable the DTX, if current send codec is Opus. Currently, DTX can only be
-  // enabled when the application mode is kVoip. If |force_voip| == true,
-  // the application mode will be forced to kVoip. Otherwise, a failure will be
-  // triggered if current application mode is kAudio.
-  // Input:
-  //   - force_application    : whether to force application mode to kVoip.
+  // EnableOpusDtx()
+  // Enable the DTX, if current send codec is Opus.
+  //
   // Return value:
   //   -1 if current send codec is not Opus or error occurred in enabling the
   //      Opus DTX.
-  //    0 if Opus DTX is enabled successfully..
-  virtual int EnableOpusDtx(bool force_application) = 0;
+  //    0 if Opus DTX is enabled successfully.
+  //
+  virtual int EnableOpusDtx() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int DisableOpusDtx()
@@ -942,23 +709,6 @@ class AudioCodingModule {
   virtual int32_t GetNetworkStatistics(
       NetworkStatistics* network_statistics) = 0;
 
-  //
-  // Set an initial delay for playout.
-  // An initial delay yields ACM playout silence until equivalent of |delay_ms|
-  // audio payload is accumulated in NetEq jitter. Thereafter, ACM pulls audio
-  // from NetEq in its regular fashion, and the given delay is maintained
-  // through out the call, unless channel conditions yield to a higher jitter
-  // buffer delay.
-  //
-  // Input:
-  //   -delay_ms           : delay in milliseconds.
-  //
-  // Return values:
-  //   -1 if failed to set the delay.
-  //    0 if delay is set successfully.
-  //
-  virtual int SetInitialPlayoutDelay(int delay_ms) = 0;
-
   //
   // Enable NACK and set the maximum size of the NACK list. If NACK is already
   // enable then the maximum NACK list size is modified accordingly.
@@ -991,186 +741,6 @@ class AudioCodingModule {
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
-class AudioEncoder;
-class ReceiverInfo;
-
-class AudioCoding {
- public:
-  struct Config {
-    Config()
-        : neteq_config(),
-          clock(Clock::GetRealTimeClock()),
-          transport(NULL),
-          vad_callback(NULL),
-          play_dtmf(true),
-          initial_playout_delay_ms(0),
-          playout_channels(1),
-          playout_frequency_hz(32000) {}
-
-    AudioCodingModule::Config ToOldConfig() const {
-      AudioCodingModule::Config old_config;
-      old_config.id = 0;
-      old_config.neteq_config = neteq_config;
-      old_config.clock = clock;
-      return old_config;
-    }
-
-    NetEq::Config neteq_config;
-    Clock* clock;
-    AudioPacketizationCallback* transport;
-    ACMVADCallback* vad_callback;
-    bool play_dtmf;
-    int initial_playout_delay_ms;
-    int playout_channels;
-    int playout_frequency_hz;
-  };
-
-  static AudioCoding* Create(const Config& config);
-  virtual ~AudioCoding() {};
-
-  // Registers a codec, specified by |send_codec|, as sending codec.
-  // This API can be called multiple times. The last codec registered overwrites
-  // the previous ones. Returns true if successful, false if not.
-  //
-  // Note: If a stereo codec is registered as send codec, VAD/DTX will
-  // automatically be turned off, since it is not supported for stereo sending.
-  virtual bool RegisterSendCodec(AudioEncoder* send_codec) = 0;
-
-  // Temporary solution to be used during refactoring:
-  // |encoder_type| should be from the anonymous enum in acm2::ACMCodecDB.
-  virtual bool RegisterSendCodec(int encoder_type,
-                                 uint8_t payload_type,
-                                 int frame_size_samples = 0) = 0;
-
-  // Returns the encoder object currently in use. This is the same as the
-  // codec that was registered in the latest call to RegisterSendCodec().
-  virtual const AudioEncoder* GetSenderInfo() const = 0;
-
-  // Temporary solution to be used during refactoring.
-  virtual const CodecInst* GetSenderCodecInst() = 0;
-
-  // Adds 10 ms of raw (PCM) audio data to the encoder. If the sampling
-  // frequency of the audio does not match the sampling frequency of the
-  // current encoder, ACM will resample the audio.
-  //
-  // Return value:
-  //      0   successfully added the frame.
-  //     -1   some error occurred and data is not added.
-  //   < -1   to add the frame to the buffer n samples had to be
-  //          overwritten, -n is the return value in this case.
-  // TODO(henrik.lundin): Make a better design for the return values. This one
-  // is just a copy of the old API.
-  virtual int Add10MsAudio(const AudioFrame& audio_frame) = 0;
-
-  // Returns a combined info about the currently used decoder(s).
-  virtual const ReceiverInfo* GetReceiverInfo() const = 0;
-
-  // Registers a codec, specified by |receive_codec|, as receiving codec.
-  // This API can be called multiple times. If registering with a payload type
-  // that was already registered in a previous call, the latest call will
-  // override previous calls. Returns true if successful, false if not.
-  virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) = 0;
-
-  // Temporary solution:
-  // |decoder_type| should be from the anonymous enum in acm2::ACMCodecDB.
-  virtual bool RegisterReceiveCodec(int decoder_type, uint8_t payload_type) = 0;
-
-  // The following two methods both inserts a new packet to the receiver.
-  // InsertPacket takes an RTP header input in |rtp_info|, while InsertPayload
-  // only requires a payload type and a timestamp. The latter assumes that the
-  // payloads come in the right order, and without any losses. In both cases,
-  // |incoming_payload| contains the RTP payload after the RTP header. Return
-  // true if successful, false if not.
-  virtual bool InsertPacket(const uint8_t* incoming_payload,
-                            size_t payload_len_bytes,
-                            const WebRtcRTPHeader& rtp_info) = 0;
-
-  // TODO(henrik.lundin): Remove this method?
-  virtual bool InsertPayload(const uint8_t* incoming_payload,
-                             size_t payload_len_byte,
-                             uint8_t payload_type,
-                             uint32_t timestamp) = 0;
-
-  // These two methods set a minimum and maximum jitter buffer delay in
-  // milliseconds. The pupose is mainly to adjust the delay to synchronize
-  // audio and video. The preferred jitter buffer size, computed by NetEq based
-  // on the current channel conditions, is clamped from below and above by these
-  // two methods. The given delay limits must be non-negative, less than
-  // 10000 ms, and the minimum must be strictly smaller than the maximum.
-  // Further, the maximum must be at lest one frame duration. If these
-  // conditions are not met, false is returned. Giving the value 0 effectively
-  // unsets the minimum or maximum delay limits.
-  // Note that calling these methods is optional. If not called, NetEq will
-  // determine the optimal buffer size based on the network conditions.
-  virtual bool SetMinimumPlayoutDelay(int time_ms) = 0;
-
-  virtual bool SetMaximumPlayoutDelay(int time_ms) = 0;
-
-  // Returns the current value of the jitter buffer's preferred latency. This
-  // is computed based on inter-arrival times and playout mode of NetEq. The
-  // actual target delay is this value clamped from below and above by the
-  // values specified through SetMinimumPlayoutDelay() and
-  // SetMaximumPlayoutDelay(), respectively, if provided.
-  // TODO(henrik.lundin) Rename to PreferredDelayMs?
-  virtual int LeastRequiredDelayMs() const = 0;
-
-  // The send timestamp of an RTP packet is associated with the decoded
-  // audio of the packet in question. This function returns the timestamp of
-  // the latest audio delivered by Get10MsAudio(). Returns false if no timestamp
-  // can be provided, true otherwise.
-  virtual bool PlayoutTimestamp(uint32_t* timestamp) = 0;
-
-  // Delivers 10 ms of audio in |audio_frame|. Returns true if successful,
-  // false otherwise.
-  virtual bool Get10MsAudio(AudioFrame* audio_frame) = 0;
-
-  // Returns the network statistics. Note that the internal statistics of NetEq
-  // are reset by this call. Returns true if successful, false otherwise.
-  virtual bool GetNetworkStatistics(NetworkStatistics* network_statistics) = 0;
-
-  // Enables NACK and sets the maximum size of the NACK list. If NACK is already
-  // enabled then the maximum NACK list size is modified accordingly. Returns
-  // true if successful, false otherwise.
-  //
-  // If the sequence number of last received packet is N, the sequence numbers
-  // of NACK list are in the range of [N - |max_nack_list_size|, N).
-  //
-  // |max_nack_list_size| should be positive and less than or equal to
-  // |Nack::kNackListSizeLimit|.
-  virtual bool EnableNack(size_t max_nack_list_size) = 0;
-
-  // Disables NACK.
-  virtual void DisableNack() = 0;
-
-
-  // Temporary solution to be used during refactoring.
-  // If DTX is enabled and the codec does not have internal DTX/VAD
-  // WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
-  //
-  // If DTX is disabled but VAD is enabled no DTX packets are sent,
-  // regardless of whether the codec has internal DTX/VAD or not. In this
-  // case, WebRtc VAD is running to label frames as active/in-active.
-  //
-  // NOTE! VAD/DTX is not supported when sending stereo.
-  //
-  // Return true if successful, false otherwise.
-  virtual bool SetVad(bool enable_dtx,
-                      bool enable_vad,
-                      ACMVADMode vad_mode) = 0;
-
-  // Returns a list of packets to request retransmission of.
-  // |round_trip_time_ms| is an estimate of the round-trip-time (in
-  // milliseconds). Missing packets which will be decoded sooner than the
-  // round-trip-time (with respect to the time this API is called) will not be
-  // included in the list.
-  // |round_trip_time_ms| must be non-negative.
-  virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
-
-  // Returns the timing statistics for calls to Get10MsAudio.
-  virtual void GetDecodingCallStatistics(
-      AudioDecodingCallStats* call_stats) const = 0;
-};
-
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
new file mode 100644
index 0000000000..280d6bffa2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+
+#include <map>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMVADMode
+// An enumerator for aggressiveness of VAD
+// -VADNormal                : least aggressive mode.
+// -VADLowBitrate            : more aggressive than "VADNormal" to save on
+//                             bit-rate.
+// -VADAggr                  : an aggressive mode.
+// -VADVeryAggr              : the most agressive mode.
+//
+enum ACMVADMode {
+  VADNormal = 0,
+  VADLowBitrate = 1,
+  VADAggr = 2,
+  VADVeryAggr = 3
+};
+
+///////////////////////////////////////////////////////////////////////////
+//
+// Enumeration of Opus mode for intended application.
+//
+// kVoip              : optimized for voice signals.
+// kAudio             : optimized for non-voice signals like music.
+//
+enum OpusApplicationMode {
+ kVoip = 0,
+ kAudio = 1,
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_coding/main/OWNERS
deleted file mode 100644
index 83880d21dc..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-tina.legrand@webrtc.org
-turaj@webrtc.org
-jan.skoglund@webrtc.org
-henrik.lundin@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
deleted file mode 100644
index 7710046d24..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ /dev/null
@@ -1,556 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file generates databases with information about all supported audio
- * codecs.
- */
-
-// TODO(tlegrand): Change constant input pointers in all functions to constant
-// references, where appropriate.
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-
-#include <assert.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-// Not yet used payload-types.
-// 83,  82,  81, 80, 79,  78,  77,  76,  75,  74,  73,  72,  71,  70,  69, 68,
-// 67, 66, 65
-
-const CodecInst ACMCodecDB::database_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-  {103, "ISAC", 16000, kIsacPacSize480, 1, kIsacWbDefaultRate},
-# if (defined(WEBRTC_CODEC_ISAC))
-  {104, "ISAC", 32000, kIsacPacSize960, 1, kIsacSwbDefaultRate},
-  {105, "ISAC", 48000, kIsacPacSize1440, 1, kIsacSwbDefaultRate},
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-  // Mono
-  {107, "L16", 8000, 80, 1, 128000},
-  {108, "L16", 16000, 160, 1, 256000},
-  {109, "L16", 32000, 320, 1, 512000},
-  // Stereo
-  {111, "L16", 8000, 80, 2, 128000},
-  {112, "L16", 16000, 160, 2, 256000},
-  {113, "L16", 32000, 320, 2, 512000},
-#endif
-  // G.711, PCM mu-law and A-law.
-  // Mono
-  {0, "PCMU", 8000, 160, 1, 64000},
-  {8, "PCMA", 8000, 160, 1, 64000},
-  // Stereo
-  {110, "PCMU", 8000, 160, 2, 64000},
-  {118, "PCMA", 8000, 160, 2, 64000},
-#ifdef WEBRTC_CODEC_ILBC
-  {102, "ILBC", 8000, 240, 1, 13300},
-#endif
-#ifdef WEBRTC_CODEC_G722
-  // Mono
-  {9, "G722", 16000, 320, 1, 64000},
-  // Stereo
-  {119, "G722", 16000, 320, 2, 64000},
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-  // Opus internally supports 48, 24, 16, 12, 8 kHz.
-  // Mono and stereo.
-  {120, "opus", 48000, 960, 2, 64000},
-#endif
-  // Comfort noise for four different sampling frequencies.
-  {13, "CN", 8000, 240, 1, 0},
-  {98, "CN", 16000, 480, 1, 0},
-  {99, "CN", 32000, 960, 1, 0},
-#ifdef ENABLE_48000_HZ
-  {100, "CN", 48000, 1440, 1, 0},
-#endif
-#ifdef WEBRTC_CODEC_AVT
-  {106, "telephone-event", 8000, 240, 1, 0},
-#endif
-#ifdef WEBRTC_CODEC_RED
-  {127, "red", 8000, 0, 1, 0},
-#endif
-  // To prevent compile errors due to trailing commas.
-  {-1, "Null", -1, -1, -1, -1}
-};
-
-// Create database with all codec settings at compile time.
-// Each entry needs the following parameters in the given order:
-// Number of allowed packet sizes, a vector with the allowed packet sizes,
-// Basic block samples, max number of channels that are supported.
-const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-    {2, {kIsacPacSize480, kIsacPacSize960}, 0, 1, true},
-# if (defined(WEBRTC_CODEC_ISAC))
-    {1, {kIsacPacSize960}, 0, 1, true},
-    {1, {kIsacPacSize1440}, 0, 1, true},
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-    // Mono
-    {4, {80, 160, 240, 320}, 0, 2, false},
-    {4, {160, 320, 480, 640}, 0, 2, false},
-    {2, {320, 640}, 0, 2, false},
-    // Stereo
-    {4, {80, 160, 240, 320}, 0, 2, false},
-    {4, {160, 320, 480, 640}, 0, 2, false},
-    {2, {320, 640}, 0, 2},
-#endif
-    // G.711, PCM mu-law and A-law.
-    // Mono
-    {6, {80, 160, 240, 320, 400, 480}, 0, 2, false},
-    {6, {80, 160, 240, 320, 400, 480}, 0, 2, false},
-    // Stereo
-    {6, {80, 160, 240, 320, 400, 480}, 0, 2, false},
-    {6, {80, 160, 240, 320, 400, 480}, 0, 2, false},
-#ifdef WEBRTC_CODEC_ILBC
-    {4, {160, 240, 320, 480}, 0, 1, false},
-#endif
-#ifdef WEBRTC_CODEC_G722
-    // Mono
-    {6, {160, 320, 480, 640, 800, 960}, 0, 2, false},
-    // Stereo
-    {6, {160, 320, 480, 640, 800, 960}, 0, 2, false},
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-    // Opus supports frames shorter than 10ms,
-    // but it doesn't help us to use them.
-    // Mono and stereo.
-    {4, {480, 960, 1920, 2880}, 0, 2, false},
-#endif
-    // Comfort noise for three different sampling frequencies.
-    {1, {240}, 240, 1, false},
-    {1, {480}, 480, 1, false},
-    {1, {960}, 960, 1, false},
-#ifdef ENABLE_48000_HZ
-    {1, {1440}, 1440, 1, false},
-#endif
-#ifdef WEBRTC_CODEC_AVT
-    {1, {240}, 240, 1, false},
-#endif
-#ifdef WEBRTC_CODEC_RED
-    {1, {0}, 0, 1, false},
-#endif
-    // To prevent compile errors due to trailing commas.
-    {-1, {-1}, -1, -1, false}
-};
-
-// Create a database of all NetEQ decoders at compile time.
-const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-    kDecoderISAC,
-# if (defined(WEBRTC_CODEC_ISAC))
-    kDecoderISACswb,
-    kDecoderISACfb,
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-    // Mono
-    kDecoderPCM16B,
-    kDecoderPCM16Bwb,
-    kDecoderPCM16Bswb32kHz,
-    // Stereo
-    kDecoderPCM16B_2ch,
-    kDecoderPCM16Bwb_2ch,
-    kDecoderPCM16Bswb32kHz_2ch,
-#endif
-    // G.711, PCM mu-las and A-law.
-    // Mono
-    kDecoderPCMu,
-    kDecoderPCMa,
-    // Stereo
-    kDecoderPCMu_2ch,
-    kDecoderPCMa_2ch,
-#ifdef WEBRTC_CODEC_ILBC
-    kDecoderILBC,
-#endif
-#ifdef WEBRTC_CODEC_G722
-    // Mono
-    kDecoderG722,
-    // Stereo
-    kDecoderG722_2ch,
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-    // Mono and stereo.
-    kDecoderOpus,
-#endif
-    // Comfort noise for three different sampling frequencies.
-    kDecoderCNGnb,
-    kDecoderCNGwb,
-    kDecoderCNGswb32kHz
-#ifdef ENABLE_48000_HZ
-    , kDecoderCNGswb48kHz
-#endif
-#ifdef WEBRTC_CODEC_AVT
-    , kDecoderAVT
-#endif
-#ifdef WEBRTC_CODEC_RED
-    , kDecoderRED
-#endif
-};
-
-// Get codec information from database.
-// TODO(tlegrand): replace memcpy with a pointer to the data base memory.
-int ACMCodecDB::Codec(int codec_id, CodecInst* codec_inst) {
-  // Error check to see that codec_id is not out of bounds.
-  if ((codec_id < 0) || (codec_id >= kNumCodecs)) {
-    return -1;
-  }
-
-  // Copy database information for the codec to the output.
-  memcpy(codec_inst, &database_[codec_id], sizeof(CodecInst));
-
-  return 0;
-}
-
-// Enumerator for error codes when asking for codec database id.
-enum {
-  kInvalidCodec = -10,
-  kInvalidPayloadtype = -30,
-  kInvalidPacketSize = -40,
-  kInvalidRate = -50
-};
-
-// Gets the codec id number from the database. If there is some mismatch in
-// the codec settings, the function will return an error code.
-// NOTE! The first mismatch found will generate the return value.
-int ACMCodecDB::CodecNumber(const CodecInst& codec_inst, int* mirror_id) {
-  // Look for a matching codec in the database.
-  int codec_id = CodecId(codec_inst);
-
-  // Checks if we found a matching codec.
-  if (codec_id == -1) {
-    return kInvalidCodec;
-  }
-
-  // Checks the validity of payload type
-  if (!ValidPayloadType(codec_inst.pltype)) {
-    return kInvalidPayloadtype;
-  }
-
-  // Comfort Noise is special case, packet-size & rate is not checked.
-  if (STR_CASE_CMP(database_[codec_id].plname, "CN") == 0) {
-    *mirror_id = codec_id;
-    return codec_id;
-  }
-
-  // RED is special case, packet-size & rate is not checked.
-  if (STR_CASE_CMP(database_[codec_id].plname, "red") == 0) {
-    *mirror_id = codec_id;
-    return codec_id;
-  }
-
-  // Checks the validity of packet size.
-  if (codec_settings_[codec_id].num_packet_sizes > 0) {
-    bool packet_size_ok = false;
-    int i;
-    int packet_size_samples;
-    for (i = 0; i < codec_settings_[codec_id].num_packet_sizes; i++) {
-      packet_size_samples =
-          codec_settings_[codec_id].packet_sizes_samples[i];
-      if (codec_inst.pacsize == packet_size_samples) {
-        packet_size_ok = true;
-        break;
-      }
-    }
-
-    if (!packet_size_ok) {
-      return kInvalidPacketSize;
-    }
-  }
-
-  if (codec_inst.pacsize < 1) {
-    return kInvalidPacketSize;
-  }
-
-  // Check the validity of rate. Codecs with multiple rates have their own
-  // function for this.
-  *mirror_id = codec_id;
-  if (STR_CASE_CMP("isac", codec_inst.plname) == 0) {
-    if (IsISACRateValid(codec_inst.rate)) {
-      // Set mirrorID to iSAC WB which is only created once to be used both for
-      // iSAC WB and SWB, because they need to share struct.
-      *mirror_id = kISAC;
-      return  codec_id;
-    } else {
-      return kInvalidRate;
-    }
-  } else if (STR_CASE_CMP("ilbc", codec_inst.plname) == 0) {
-    return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize)
-        ? codec_id : kInvalidRate;
-  } else if (STR_CASE_CMP("amr", codec_inst.plname) == 0) {
-    return IsAMRRateValid(codec_inst.rate)
-        ? codec_id : kInvalidRate;
-  } else if (STR_CASE_CMP("amr-wb", codec_inst.plname) == 0) {
-    return IsAMRwbRateValid(codec_inst.rate)
-        ? codec_id : kInvalidRate;
-  } else if (STR_CASE_CMP("g7291", codec_inst.plname) == 0) {
-    return IsG7291RateValid(codec_inst.rate)
-        ? codec_id : kInvalidRate;
-  } else if (STR_CASE_CMP("opus", codec_inst.plname) == 0) {
-    return IsOpusRateValid(codec_inst.rate)
-        ? codec_id : kInvalidRate;
-  } else if (STR_CASE_CMP("speex", codec_inst.plname) == 0) {
-    return IsSpeexRateValid(codec_inst.rate)
-        ? codec_id : kInvalidRate;
-  }
-
-  return IsRateValid(codec_id, codec_inst.rate) ?
-      codec_id : kInvalidRate;
-}
-
-// Looks for a matching payload name, frequency, and channels in the
-// codec list. Need to check all three since some codecs have several codec
-// entries with different frequencies and/or channels.
-// Does not check other codec settings, such as payload type and packet size.
-// Returns the id of the codec, or -1 if no match is found.
-int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
-  return (CodecId(codec_inst.plname, codec_inst.plfreq,
-                  codec_inst.channels));
-}
-
-int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) {
-  for (int id = 0; id < kNumCodecs; id++) {
-    bool name_match = false;
-    bool frequency_match = false;
-    bool channels_match = false;
-
-    // Payload name, sampling frequency and number of channels need to match.
-    // NOTE! If |frequency| is -1, the frequency is not applicable, and is
-    // always treated as true, like for RED.
-    name_match = (STR_CASE_CMP(database_[id].plname, payload_name) == 0);
-    frequency_match = (frequency == database_[id].plfreq) || (frequency == -1);
-    // The number of channels must match for all codecs but Opus.
-    if (STR_CASE_CMP(payload_name, "opus") != 0) {
-      channels_match = (channels == database_[id].channels);
-    } else {
-      // For opus we just check that number of channels is valid.
-      channels_match = (channels == 1 || channels == 2);
-    }
-
-    if (name_match && frequency_match && channels_match) {
-      // We have found a matching codec in the list.
-      return id;
-    }
-  }
-
-  // We didn't find a matching codec.
-  return -1;
-}
-// Gets codec id number, and mirror id, from database for the receiver.
-int ACMCodecDB::ReceiverCodecNumber(const CodecInst& codec_inst,
-                                    int* mirror_id) {
-  // Look for a matching codec in the database.
-  int codec_id = CodecId(codec_inst);
-
-  // Set |mirror_id| to |codec_id|, except for iSAC. In case of iSAC we always
-  // set |mirror_id| to iSAC WB (kISAC) which is only created once to be used
-  // both for iSAC WB and SWB, because they need to share struct.
-  if (STR_CASE_CMP(codec_inst.plname, "ISAC") != 0) {
-    *mirror_id = codec_id;
-  } else {
-    *mirror_id = kISAC;
-  }
-
-  return codec_id;
-}
-
-// Returns the codec sampling frequency for codec with id = "codec_id" in
-// database.
-int ACMCodecDB::CodecFreq(int codec_id) {
-  // Error check to see that codec_id is not out of bounds.
-  if (codec_id < 0 || codec_id >= kNumCodecs) {
-    return -1;
-  }
-
-  return database_[codec_id].plfreq;
-}
-
-// Returns the codec's basic coding block size in samples.
-int ACMCodecDB::BasicCodingBlock(int codec_id) {
-  // Error check to see that codec_id is not out of bounds.
-  if (codec_id < 0 || codec_id >= kNumCodecs) {
-    return -1;
-  }
-
-  return codec_settings_[codec_id].basic_block_samples;
-}
-
-// Returns the NetEQ decoder database.
-const NetEqDecoder* ACMCodecDB::NetEQDecoders() {
-  return neteq_decoders_;
-}
-
-// Gets mirror id. The Id is used for codecs sharing struct for settings that
-// need different payload types.
-int ACMCodecDB::MirrorID(int codec_id) {
-  if (STR_CASE_CMP(database_[codec_id].plname, "isac") == 0) {
-    return kISAC;
-  } else {
-    return codec_id;
-  }
-}
-
-// Creates memory/instance for storing codec state.
-ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst& codec_inst,
-                                                 int cng_pt_nb,
-                                                 int cng_pt_wb,
-                                                 int cng_pt_swb,
-                                                 int cng_pt_fb,
-                                                 bool enable_red,
-                                                 int red_payload_type) {
-  // All we have support for right now.
-  if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-    return new ACMGenericCodec(codec_inst, cng_pt_nb, cng_pt_wb, cng_pt_swb,
-                               cng_pt_fb, enable_red, red_payload_type);
-#endif
-  } else if (!STR_CASE_CMP(codec_inst.plname, "PCMU") ||
-             !STR_CASE_CMP(codec_inst.plname, "PCMA")) {
-    return new ACMGenericCodec(codec_inst, cng_pt_nb, cng_pt_wb, cng_pt_swb,
-                               cng_pt_fb, enable_red, red_payload_type);
-  } else if (!STR_CASE_CMP(codec_inst.plname, "ILBC")) {
-#ifdef WEBRTC_CODEC_ILBC
-    return new ACMGenericCodec(codec_inst, cng_pt_nb, cng_pt_wb, cng_pt_swb,
-                               cng_pt_fb, enable_red, red_payload_type);
-#endif
-  } else if (!STR_CASE_CMP(codec_inst.plname, "G722")) {
-#ifdef WEBRTC_CODEC_G722
-    return new ACMGenericCodec(codec_inst, cng_pt_nb, cng_pt_wb, cng_pt_swb,
-                               cng_pt_fb, enable_red, red_payload_type);
-#endif
-  } else if (!STR_CASE_CMP(codec_inst.plname, "opus")) {
-#ifdef WEBRTC_CODEC_OPUS
-    return new ACMGenericCodec(codec_inst, cng_pt_nb, cng_pt_wb, cng_pt_swb,
-                               cng_pt_fb, enable_red, red_payload_type);
-#endif
-  } else if (!STR_CASE_CMP(codec_inst.plname, "L16")) {
-#ifdef WEBRTC_CODEC_PCM16
-    return new ACMGenericCodec(codec_inst, cng_pt_nb, cng_pt_wb, cng_pt_swb,
-                               cng_pt_fb, enable_red, red_payload_type);
-#endif
-  }
-  return NULL;
-}
-
-// Checks if the bitrate is valid for the codec.
-bool ACMCodecDB::IsRateValid(int codec_id, int rate) {
-  return database_[codec_id].rate == rate;
-}
-
-// Checks if the bitrate is valid for iSAC.
-bool ACMCodecDB::IsISACRateValid(int rate) {
-  return (rate == -1) || ((rate <= 56000) && (rate >= 10000));
-}
-
-// Checks if the bitrate is valid for iLBC.
-bool ACMCodecDB::IsILBCRateValid(int rate, int frame_size_samples) {
-  if (((frame_size_samples == 240) || (frame_size_samples == 480)) &&
-      (rate == 13300)) {
-    return true;
-  } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
-      (rate == 15200)) {
-    return true;
-  } else {
-    return false;
-  }
-}
-
-// Check if the bitrate is valid for the GSM-AMR.
-bool ACMCodecDB::IsAMRRateValid(int rate) {
-  switch (rate) {
-    case 4750:
-    case 5150:
-    case 5900:
-    case 6700:
-    case 7400:
-    case 7950:
-    case 10200:
-    case 12200: {
-      return true;
-    }
-    default: {
-      return false;
-    }
-  }
-}
-
-// Check if the bitrate is valid for GSM-AMR-WB.
-bool ACMCodecDB::IsAMRwbRateValid(int rate) {
-  switch (rate) {
-    case 7000:
-    case 9000:
-    case 12000:
-    case 14000:
-    case 16000:
-    case 18000:
-    case 20000:
-    case 23000:
-    case 24000: {
-      return true;
-    }
-    default: {
-      return false;
-    }
-  }
-}
-
-// Check if the bitrate is valid for G.729.1.
-bool ACMCodecDB::IsG7291RateValid(int rate) {
-  switch (rate) {
-    case 8000:
-    case 12000:
-    case 14000:
-    case 16000:
-    case 18000:
-    case 20000:
-    case 22000:
-    case 24000:
-    case 26000:
-    case 28000:
-    case 30000:
-    case 32000: {
-      return true;
-    }
-    default: {
-      return false;
-    }
-  }
-}
-
-// Checks if the bitrate is valid for Speex.
-bool ACMCodecDB::IsSpeexRateValid(int rate) {
-  return rate > 2000;
-}
-
-// Checks if the bitrate is valid for Opus.
-bool ACMCodecDB::IsOpusRateValid(int rate) {
-  return (rate >= 6000) && (rate <= 510000);
-}
-
-// Checks if the payload type is in the valid range.
-bool ACMCodecDB::ValidPayloadType(int payload_type) {
-  return (payload_type >= 0) && (payload_type <= 127);
-}
-
-bool ACMCodecDB::OwnsDecoder(int codec_id) {
-  assert(codec_id >= 0 && codec_id < ACMCodecDB::kNumCodecs);
-  return ACMCodecDB::codec_settings_[codec_id].owns_decoder;
-}
-
-}  // namespace acm2
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
deleted file mode 100644
index ea7eb23ac0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
+++ /dev/null
@@ -1,298 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file generates databases with information about all supported audio
- * codecs.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-// TODO(tlegrand): replace class ACMCodecDB with a namespace.
-class ACMCodecDB {
- public:
-  // Enum with array indexes for the supported codecs. NOTE! The order MUST
-  // be the same as when creating the database in acm_codec_database.cc.
-  enum {
-    kNone = -1
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-    , kISAC
-# if (defined(WEBRTC_CODEC_ISAC))
-    , kISACSWB
-    , kISACFB
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-    // Mono
-    , kPCM16B
-    , kPCM16Bwb
-    , kPCM16Bswb32kHz
-    // Stereo
-    , kPCM16B_2ch
-    , kPCM16Bwb_2ch
-    , kPCM16Bswb32kHz_2ch
-#endif
-    // Mono
-    , kPCMU
-    , kPCMA
-    // Stereo
-    , kPCMU_2ch
-    , kPCMA_2ch
-#ifdef WEBRTC_CODEC_ILBC
-    , kILBC
-#endif
-#ifdef WEBRTC_CODEC_G722
-    // Mono
-    , kG722
-    // Stereo
-    , kG722_2ch
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-    // Mono and stereo
-    , kOpus
-#endif
-    , kCNNB
-    , kCNWB
-    , kCNSWB
-#ifdef ENABLE_48000_HZ
-    , kCNFB
-#endif
-#ifdef WEBRTC_CODEC_AVT
-    , kAVT
-#endif
-#ifdef WEBRTC_CODEC_RED
-    , kRED
-#endif
-    , kNumCodecs
-  };
-
-  // Set unsupported codecs to -1
-#ifndef WEBRTC_CODEC_ISAC
-  enum {kISACSWB = -1};
-  enum {kISACFB = -1};
-# ifndef WEBRTC_CODEC_ISACFX
-  enum {kISAC = -1};
-# endif
-#endif
-#ifndef WEBRTC_CODEC_PCM16
-  // Mono
-  enum {kPCM16B = -1};
-  enum {kPCM16Bwb = -1};
-  enum {kPCM16Bswb32kHz = -1};
-  // Stereo
-  enum {kPCM16B_2ch = -1};
-  enum {kPCM16Bwb_2ch = -1};
-  enum {kPCM16Bswb32kHz_2ch = -1};
-#endif
-  // 48 kHz not supported, always set to -1.
-  enum {kPCM16Bswb48kHz = -1};
-#ifndef WEBRTC_CODEC_ILBC
-  enum {kILBC = -1};
-#endif
-#ifndef WEBRTC_CODEC_G722
-  // Mono
-  enum {kG722 = -1};
-  // Stereo
-  enum {kG722_2ch = -1};
-#endif
-#ifndef WEBRTC_CODEC_OPUS
-  // Mono and stereo
-  enum {kOpus = -1};
-#endif
-#ifndef WEBRTC_CODEC_AVT
-  enum {kAVT = -1};
-#endif
-#ifndef WEBRTC_CODEC_RED
-  enum {kRED = -1};
-#endif
-#ifndef ENABLE_48000_HZ
-  enum { kCNFB = -1 };
-#endif
-
-  // kMaxNumCodecs - Maximum number of codecs that can be activated in one
-  //                 build.
-  // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec.
-  // These might need to be increased if adding a new codec to the database
-  static const int kMaxNumCodecs =  50;
-  static const int kMaxNumPacketSize = 6;
-
-  // Codec specific settings
-  //
-  // num_packet_sizes     - number of allowed packet sizes.
-  // packet_sizes_samples - list of the allowed packet sizes.
-  // basic_block_samples  - assigned a value different from 0 if the codec
-  //                        requires to be fed with a specific number of samples
-  //                        that can be different from packet size.
-  // channel_support      - number of channels supported to encode;
-  //                        1 = mono, 2 = stereo, etc.
-  // owns_decoder         - if true, it means that the codec should own the
-  //                        decoder instance. In this case, the codec should
-  //                        implement ACMGenericCodec::Decoder(), which returns
-  //                        a pointer to AudioDecoder. This pointer is injected
-  //                        into NetEq when this codec is registered as receive
-  //                        codec.
-  struct CodecSettings {
-    int num_packet_sizes;
-    int packet_sizes_samples[kMaxNumPacketSize];
-    int basic_block_samples;
-    int channel_support;
-    bool owns_decoder;
-  };
-
-  // Gets codec information from database at the position in database given by
-  // [codec_id].
-  // Input:
-  //   [codec_id] - number that specifies at what position in the database to
-  //                get the information.
-  // Output:
-  //   [codec_inst] - filled with information about the codec.
-  // Return:
-  //   0 if successful, otherwise -1.
-  static int Codec(int codec_id, CodecInst* codec_inst);
-
-  // Returns codec id and mirror id from database, given the information
-  // received in the input [codec_inst]. Mirror id is a number that tells
-  // where to find the codec's memory (instance). The number is either the
-  // same as codec id (most common), or a number pointing at a different
-  // entry in the database, if the codec has several entries with different
-  // payload types. This is used for codecs that must share one struct even if
-  // the payload type differs.
-  // One example is the codec iSAC which has the same struct for both 16 and
-  // 32 khz, but they have different entries in the database. Let's say the
-  // function is called with iSAC 32kHz. The function will return 1 as that is
-  // the entry in the data base, and [mirror_id] = 0, as that is the entry for
-  // iSAC 16 kHz, which holds the shared memory.
-  // Input:
-  //   [codec_inst] - Information about the codec for which we require the
-  //                  database id.
-  // Output:
-  //   [mirror_id] - mirror id, which most often is the same as the return
-  //                 value, see above.
-  //   [err_message] - if present, in the event of a mismatch found between the
-  //                   input and the database, a descriptive error message is
-  //                   written here.
-  //   [err_message] - if present, the length of error message is returned here.
-  // Return:
-  //   codec id if successful, otherwise < 0.
-  static int CodecNumber(const CodecInst& codec_inst, int* mirror_id,
-                         char* err_message, int max_message_len_byte);
-  static int CodecNumber(const CodecInst& codec_inst, int* mirror_id);
-  static int CodecId(const CodecInst& codec_inst);
-  static int CodecId(const char* payload_name, int frequency, int channels);
-  static int ReceiverCodecNumber(const CodecInst& codec_inst, int* mirror_id);
-
-  // Returns the codec sampling frequency for codec with id = "codec_id" in
-  // database.
-  // TODO(tlegrand): Check if function is needed, or if we can change
-  // to access database directly.
-  // Input:
-  //   [codec_id] - number that specifies at what position in the database to
-  //                get the information.
-  // Return:
-  //   codec sampling frequency if successful, otherwise -1.
-  static int CodecFreq(int codec_id);
-
-  // Return the codec's basic coding block size in samples.
-  // TODO(tlegrand): Check if function is needed, or if we can change
-  // to access database directly.
-  // Input:
-  //   [codec_id] - number that specifies at what position in the database to
-  //                get the information.
-  // Return:
-  //   codec basic block size if successful, otherwise -1.
-  static int BasicCodingBlock(int codec_id);
-
-  // Returns the NetEQ decoder database.
-  static const NetEqDecoder* NetEQDecoders();
-
-  // Returns mirror id, which is a number that tells where to find the codec's
-  // memory (instance). It is either the same as codec id (most common), or a
-  // number pointing at a different entry in the database, if the codec have
-  // several entries with different payload types. This is used for codecs that
-  // must share struct even if the payload type differs.
-  // TODO(tlegrand): Check if function is needed, or if we can change
-  // to access database directly.
-  // Input:
-  //   [codec_id] - number that specifies codec's position in the database.
-  // Return:
-  //   Mirror id on success, otherwise -1.
-  static int MirrorID(int codec_id);
-
-  // Creates a codec wrapper containing an AudioEncoder object (or an
-  // ACMGenericCodec subclass during the refactoring time). The type of
-  // AudioEncoder is decided by looking at the information in |codec_inst|.
-  // The |cng_pt_*| parameters should contain the RTP payload type used for each
-  // type of comfort noise; if not used (or not know when this function is
-  // called), -1 can be set. The parameter |enable_red| indicates that RED
-  // is enabled, and that |red_payload_type| should be used as RTP payload type
-  // for RED encodings.
-  static ACMGenericCodec* CreateCodecInstance(const CodecInst& codec_inst,
-                                              int cng_pt_nb,
-                                              int cng_pt_wb,
-                                              int cng_pt_swb,
-                                              int cng_pt_fb,
-                                              bool enable_red,
-                                              int red_payload_type);
-
-  // Specifies if the codec specified by |codec_id| MUST own its own decoder.
-  // This is the case for codecs which *should* share a single codec instance
-  // between encoder and decoder. Or for codecs which ACM should have control
-  // over the decoder. For instance iSAC is such a codec that encoder and
-  // decoder share the same codec instance.
-  static bool OwnsDecoder(int codec_id);
-
-  // Checks if the bitrate is valid for the codec.
-  // Input:
-  //   [codec_id] - number that specifies codec's position in the database.
-  //   [rate] - bitrate to check.
-  //   [frame_size_samples] - (used for iLBC) specifies which frame size to go
-  //                          with the rate.
-  static bool IsRateValid(int codec_id, int rate);
-  static bool IsISACRateValid(int rate);
-  static bool IsILBCRateValid(int rate, int frame_size_samples);
-  static bool IsAMRRateValid(int rate);
-  static bool IsAMRwbRateValid(int rate);
-  static bool IsG7291RateValid(int rate);
-  static bool IsSpeexRateValid(int rate);
-  static bool IsOpusRateValid(int rate);
-
-  // Check if the payload type is valid, meaning that it is in the valid range
-  // of 0 to 127.
-  // Input:
-  //   [payload_type] - payload type.
-  static bool ValidPayloadType(int payload_type);
-
-  // Databases with information about the supported codecs
-  // database_ - stored information about all codecs: payload type, name,
-  //             sampling frequency, packet size in samples, default channel
-  //             support, and default rate.
-  // codec_settings_ - stored codec settings: number of allowed packet sizes,
-  //                   a vector with the allowed packet sizes, basic block
-  //                   samples, and max number of channels that are supported.
-  // neteq_decoders_ - list of supported decoders in NetEQ.
-  static const CodecInst database_[kMaxNumCodecs];
-  static const CodecSettings codec_settings_[kMaxNumCodecs];
-  static const NetEqDecoder neteq_decoders_[kMaxNumCodecs];
-};
-
-}  // namespace acm2
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
deleted file mode 100644
index 85a287e126..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
-
-#include <string.h>
-
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/typedefs.h"
-
-// Checks for enabled codecs, we prevent enabling codecs which are not
-// compatible.
-#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
-#error iSAC and iSACFX codecs cannot be enabled at the same time
-#endif
-
-
-namespace webrtc {
-
-// 60 ms is the maximum block size we support. An extra 20 ms is considered
-// for safety if process() method is not called when it should be, i.e. we
-// accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples.
-#define AUDIO_BUFFER_SIZE_W16 7680
-
-// There is one timestamp per each 10 ms of audio
-// the audio buffer, at max, may contain 32 blocks of 10ms
-// audio if the sampling frequency is 8000 Hz (80 samples per block).
-// Therefore, The size of the buffer where we keep timestamps
-// is defined as follows
-#define TIMESTAMP_BUFFER_SIZE_W32  (AUDIO_BUFFER_SIZE_W16/80)
-
-// The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
-#define MAX_PAYLOAD_SIZE_BYTE   7680
-
-// General codec specific defines
-const int kIsacWbDefaultRate = 32000;
-const int kIsacSwbDefaultRate = 56000;
-const int kIsacPacSize480 = 480;
-const int kIsacPacSize960 = 960;
-const int kIsacPacSize1440 = 1440;
-
-// A structure which contains codec parameters. For instance, used when
-// initializing encoder and decoder.
-//
-//   codec_inst: c.f. common_types.h
-//   enable_dtx: set true to enable DTX. If codec does not have
-//               internal DTX, this will enable VAD.
-//   enable_vad: set true to enable VAD.
-//   vad_mode: VAD mode, c.f. audio_coding_module_typedefs.h
-//             for possible values.
-struct WebRtcACMCodecParams {
-  CodecInst codec_inst;
-  bool enable_dtx;
-  bool enable_vad;
-  ACMVADMode vad_mode;
-};
-
-// TODO(turajs): Remove when ACM1 is removed.
-struct WebRtcACMAudioBuff {
-  int16_t in_audio[AUDIO_BUFFER_SIZE_W16];
-  int16_t in_audio_ix_read;
-  int16_t in_audio_ix_write;
-  uint32_t in_timestamp[TIMESTAMP_BUFFER_SIZE_W32];
-  int16_t in_timestamp_ix_write;
-  uint32_t last_timestamp;
-  uint32_t last_in_timestamp;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
deleted file mode 100644
index 6d7b09528b..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ /dev/null
@@ -1,549 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
-
-#include <assert.h>
-#include <string.h>
-#include <algorithm>
-#include <utility>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/common_audio/vad/include/webrtc_vad.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-namespace {
-static const int kInvalidPayloadType = 255;
-
-std::map<int, int>::iterator FindSampleRateInMap(std::map<int, int>* cng_pt_map,
-                                                 int sample_rate_hz) {
-  return find_if(cng_pt_map->begin(), cng_pt_map->end(),
-                 [sample_rate_hz](decltype(*cng_pt_map->begin()) p) {
-                   return p.second == sample_rate_hz;
-                 });
-}
-
-void SetPtInMap(std::map<int, int>* pt_map,
-                int sample_rate_hz,
-                int payload_type) {
-  if (payload_type == kInvalidPayloadType)
-    return;
-  CHECK_GE(payload_type, 0);
-  CHECK_LT(payload_type, 128);
-  auto pt_iter = FindSampleRateInMap(pt_map, sample_rate_hz);
-  if (pt_iter != pt_map->end()) {
-    // Remove item in map with sample_rate_hz.
-    pt_map->erase(pt_iter);
-  }
-  (*pt_map)[payload_type] = sample_rate_hz;
-}
-}  // namespace
-
-namespace acm2 {
-
-// Enum for CNG
-enum {
-  kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
-  kNewCNGNumLPCParams = 8
-};
-
-// Interval for sending new CNG parameters (SID frames) is 100 msec.
-enum {
-  kCngSidIntervalMsec = 100
-};
-
-// We set some of the variables to invalid values as a check point
-// if a proper initialization has happened. Another approach is
-// to initialize to a default codec that we are sure is always included.
-ACMGenericCodec::ACMGenericCodec(const CodecInst& codec_inst,
-                                 int cng_pt_nb,
-                                 int cng_pt_wb,
-                                 int cng_pt_swb,
-                                 int cng_pt_fb,
-                                 bool enable_red,
-                                 int red_pt_nb)
-    : has_internal_fec_(false),
-      copy_red_enabled_(enable_red),
-      encoder_(NULL),
-      bitrate_bps_(0),
-      fec_enabled_(false),
-      loss_rate_(0),
-      max_playback_rate_hz_(48000),
-      max_payload_size_bytes_(-1),
-      max_rate_bps_(-1),
-      opus_dtx_enabled_(false),
-      is_opus_(false),
-      is_isac_(false),
-      opus_application_set_(false) {
-  acm_codec_params_.codec_inst = codec_inst;
-  acm_codec_params_.enable_dtx = false;
-  acm_codec_params_.enable_vad = false;
-  acm_codec_params_.vad_mode = VADNormal;
-  SetPtInMap(&red_pt_, 8000, red_pt_nb);
-  SetPtInMap(&cng_pt_, 8000, cng_pt_nb);
-  SetPtInMap(&cng_pt_, 16000, cng_pt_wb);
-  SetPtInMap(&cng_pt_, 32000, cng_pt_swb);
-  SetPtInMap(&cng_pt_, 48000, cng_pt_fb);
-  ResetAudioEncoder();
-  CHECK(encoder_);
-}
-
-ACMGenericCodec::~ACMGenericCodec() {
-}
-
-AudioDecoderProxy::AudioDecoderProxy()
-    : decoder_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      decoder_(nullptr) {
-}
-
-void AudioDecoderProxy::SetDecoder(AudioDecoder* decoder) {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  decoder_ = decoder;
-  CHECK_EQ(decoder_->Init(), 0);
-}
-
-bool AudioDecoderProxy::IsSet() const {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return (decoder_ != nullptr);
-}
-
-int AudioDecoderProxy::Decode(const uint8_t* encoded,
-                              size_t encoded_len,
-                              int sample_rate_hz,
-                              size_t max_decoded_bytes,
-                              int16_t* decoded,
-                              SpeechType* speech_type) {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
-                          max_decoded_bytes, decoded, speech_type);
-}
-
-int AudioDecoderProxy::DecodeRedundant(const uint8_t* encoded,
-                                       size_t encoded_len,
-                                       int sample_rate_hz,
-                                       size_t max_decoded_bytes,
-                                       int16_t* decoded,
-                                       SpeechType* speech_type) {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->DecodeRedundant(encoded, encoded_len, sample_rate_hz,
-                                   max_decoded_bytes, decoded, speech_type);
-}
-
-bool AudioDecoderProxy::HasDecodePlc() const {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->HasDecodePlc();
-}
-
-int AudioDecoderProxy::DecodePlc(int num_frames, int16_t* decoded) {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->DecodePlc(num_frames, decoded);
-}
-
-int AudioDecoderProxy::Init() {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->Init();
-}
-
-int AudioDecoderProxy::IncomingPacket(const uint8_t* payload,
-                                      size_t payload_len,
-                                      uint16_t rtp_sequence_number,
-                                      uint32_t rtp_timestamp,
-                                      uint32_t arrival_timestamp) {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->IncomingPacket(payload, payload_len, rtp_sequence_number,
-                                  rtp_timestamp, arrival_timestamp);
-}
-
-int AudioDecoderProxy::ErrorCode() {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->ErrorCode();
-}
-
-int AudioDecoderProxy::PacketDuration(const uint8_t* encoded,
-                                      size_t encoded_len) const {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->PacketDuration(encoded, encoded_len);
-}
-
-int AudioDecoderProxy::PacketDurationRedundant(const uint8_t* encoded,
-                                               size_t encoded_len) const {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->PacketDurationRedundant(encoded, encoded_len);
-}
-
-bool AudioDecoderProxy::PacketHasFec(const uint8_t* encoded,
-                                     size_t encoded_len) const {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->PacketHasFec(encoded, encoded_len);
-}
-
-CNG_dec_inst* AudioDecoderProxy::CngDecoderInstance() {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->CngDecoderInstance();
-}
-
-size_t AudioDecoderProxy::Channels() const {
-  CriticalSectionScoped decoder_lock(decoder_lock_.get());
-  return decoder_->Channels();
-}
-
-int16_t ACMGenericCodec::EncoderParams(WebRtcACMCodecParams* enc_params) const {
-  *enc_params = acm_codec_params_;
-  return 0;
-}
-
-int16_t ACMGenericCodec::InitEncoder(WebRtcACMCodecParams* codec_params,
-                                     bool force_initialization) {
-  bitrate_bps_ = 0;
-  loss_rate_ = 0;
-  opus_dtx_enabled_ = false;
-  acm_codec_params_ = *codec_params;
-  if (force_initialization)
-    opus_application_set_ = false;
-  opus_application_ = GetOpusApplication(codec_params->codec_inst.channels,
-                                         opus_dtx_enabled_);
-  opus_application_set_ = true;
-  ResetAudioEncoder();
-  return 0;
-}
-
-void ACMGenericCodec::ResetAudioEncoder() {
-  const CodecInst& codec_inst = acm_codec_params_.codec_inst;
-  if (!STR_CASE_CMP(codec_inst.plname, "PCMU")) {
-    AudioEncoderPcmU::Config config;
-    config.num_channels = codec_inst.channels;
-    config.frame_size_ms = codec_inst.pacsize / 8;
-    config.payload_type = codec_inst.pltype;
-    audio_encoder_.reset(new AudioEncoderPcmU(config));
-  } else if (!STR_CASE_CMP(codec_inst.plname, "PCMA")) {
-    AudioEncoderPcmA::Config config;
-    config.num_channels = codec_inst.channels;
-    config.frame_size_ms = codec_inst.pacsize / 8;
-    config.payload_type = codec_inst.pltype;
-    audio_encoder_.reset(new AudioEncoderPcmA(config));
-#ifdef WEBRTC_CODEC_PCM16
-  } else if (!STR_CASE_CMP(codec_inst.plname, "L16")) {
-    AudioEncoderPcm16B::Config config;
-    config.num_channels = codec_inst.channels;
-    config.sample_rate_hz = codec_inst.plfreq;
-    config.frame_size_ms = codec_inst.pacsize / (config.sample_rate_hz / 1000);
-    config.payload_type = codec_inst.pltype;
-    audio_encoder_.reset(new AudioEncoderPcm16B(config));
-#endif
-#ifdef WEBRTC_CODEC_ILBC
-  } else if (!STR_CASE_CMP(codec_inst.plname, "ILBC")) {
-    AudioEncoderIlbc::Config config;
-    config.frame_size_ms = codec_inst.pacsize / 8;
-    config.payload_type = codec_inst.pltype;
-    audio_encoder_.reset(new AudioEncoderIlbc(config));
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-  } else if (!STR_CASE_CMP(codec_inst.plname, "opus")) {
-    is_opus_ = true;
-    has_internal_fec_ = true;
-    AudioEncoderOpus::Config config;
-    config.frame_size_ms = codec_inst.pacsize / 48;
-    config.num_channels = codec_inst.channels;
-    config.fec_enabled = fec_enabled_;
-    config.bitrate_bps = codec_inst.rate;
-    config.max_playback_rate_hz = max_playback_rate_hz_;
-    config.dtx_enabled = opus_dtx_enabled_;
-    config.payload_type = codec_inst.pltype;
-    switch (GetOpusApplication(config.num_channels, config.dtx_enabled)) {
-      case kVoip:
-        config.application = AudioEncoderOpus::ApplicationMode::kVoip;
-        break;
-      case kAudio:
-        config.application = AudioEncoderOpus::ApplicationMode::kAudio;
-        break;
-    }
-    audio_encoder_.reset(new AudioEncoderOpus(config));
-#endif
-#ifdef WEBRTC_CODEC_G722
-  } else if (!STR_CASE_CMP(codec_inst.plname, "G722")) {
-    AudioEncoderG722::Config config;
-    config.num_channels = codec_inst.channels;
-    config.frame_size_ms = codec_inst.pacsize / 16;
-    config.payload_type = codec_inst.pltype;
-    audio_encoder_.reset(new AudioEncoderG722(config));
-#endif
-#ifdef WEBRTC_CODEC_ISACFX
-  } else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
-    DCHECK_EQ(codec_inst.plfreq, 16000);
-    is_isac_ = true;
-    AudioEncoderDecoderIsacFix* enc_dec;
-    if (codec_inst.rate == -1) {
-      // Adaptive mode.
-      AudioEncoderDecoderIsacFix::ConfigAdaptive config;
-      config.payload_type = codec_inst.pltype;
-      enc_dec = new AudioEncoderDecoderIsacFix(config);
-    } else {
-      // Channel independent mode.
-      AudioEncoderDecoderIsacFix::Config config;
-      config.bit_rate = codec_inst.rate;
-      config.frame_size_ms = codec_inst.pacsize / 16;
-      config.payload_type = codec_inst.pltype;
-      enc_dec = new AudioEncoderDecoderIsacFix(config);
-    }
-    decoder_proxy_.SetDecoder(enc_dec);
-    audio_encoder_.reset(enc_dec);
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-  } else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
-    is_isac_ = true;
-    AudioEncoderDecoderIsac* enc_dec;
-    if (codec_inst.rate == -1) {
-      // Adaptive mode.
-      AudioEncoderDecoderIsac::ConfigAdaptive config;
-      config.sample_rate_hz = codec_inst.plfreq;
-      config.initial_frame_size_ms = rtc::CheckedDivExact(
-          1000 * codec_inst.pacsize, config.sample_rate_hz);
-      config.max_payload_size_bytes = max_payload_size_bytes_;
-      config.max_bit_rate = max_rate_bps_;
-      config.payload_type = codec_inst.pltype;
-      enc_dec = new AudioEncoderDecoderIsac(config);
-    } else {
-      // Channel independent mode.
-      AudioEncoderDecoderIsac::Config config;
-      config.sample_rate_hz = codec_inst.plfreq;
-      config.bit_rate = codec_inst.rate;
-      config.frame_size_ms = rtc::CheckedDivExact(1000 * codec_inst.pacsize,
-                                                  config.sample_rate_hz);
-      config.max_payload_size_bytes = max_payload_size_bytes_;
-      config.max_bit_rate = max_rate_bps_;
-      config.payload_type = codec_inst.pltype;
-      enc_dec = new AudioEncoderDecoderIsac(config);
-    }
-    decoder_proxy_.SetDecoder(enc_dec);
-    audio_encoder_.reset(enc_dec);
-#endif
-  } else {
-    FATAL();
-  }
-  if (bitrate_bps_ != 0)
-    audio_encoder_->SetTargetBitrate(bitrate_bps_);
-  audio_encoder_->SetProjectedPacketLossRate(loss_rate_ / 100.0);
-  encoder_ = audio_encoder_.get();
-
-  // Attach RED if needed.
-  auto pt_iter =
-      FindSampleRateInMap(&red_pt_, audio_encoder_->SampleRateHz());
-  if (copy_red_enabled_ && pt_iter != red_pt_.end()) {
-    CHECK_NE(pt_iter->first, kInvalidPayloadType);
-    AudioEncoderCopyRed::Config config;
-    config.payload_type = pt_iter->first;
-    config.speech_encoder = encoder_;
-    red_encoder_.reset(new AudioEncoderCopyRed(config));
-    encoder_ = red_encoder_.get();
-  } else {
-    red_encoder_.reset();
-    copy_red_enabled_ = false;
-  }
-
-  // Attach CNG if needed.
-  // Reverse-lookup from sample rate to complete key-value pair.
-  pt_iter =
-      FindSampleRateInMap(&cng_pt_, audio_encoder_->SampleRateHz());
-  if (acm_codec_params_.enable_dtx && pt_iter != cng_pt_.end()) {
-    AudioEncoderCng::Config config;
-    config.num_channels = acm_codec_params_.codec_inst.channels;
-    config.payload_type = pt_iter->first;
-    config.speech_encoder = encoder_;
-    switch (acm_codec_params_.vad_mode) {
-      case VADNormal:
-        config.vad_mode = Vad::kVadNormal;
-        break;
-      case VADLowBitrate:
-        config.vad_mode = Vad::kVadLowBitrate;
-        break;
-      case VADAggr:
-        config.vad_mode = Vad::kVadAggressive;
-        break;
-      case VADVeryAggr:
-        config.vad_mode = Vad::kVadVeryAggressive;
-        break;
-      default:
-        FATAL();
-    }
-    cng_encoder_.reset(new AudioEncoderCng(config));
-    encoder_ = cng_encoder_.get();
-  } else {
-    cng_encoder_.reset();
-  }
-}
-
-OpusApplicationMode ACMGenericCodec::GetOpusApplication(
-    int num_channels, bool enable_dtx) const {
-  if (opus_application_set_)
-    return opus_application_;
-  return num_channels == 1 || enable_dtx ? kVoip : kAudio;
-}
-
-int16_t ACMGenericCodec::SetBitRate(const int32_t bitrate_bps) {
-  encoder_->SetTargetBitrate(bitrate_bps);
-  bitrate_bps_ = bitrate_bps;
-  return 0;
-}
-
-int16_t ACMGenericCodec::SetVAD(bool* enable_dtx,
-                                bool* enable_vad,
-                                ACMVADMode* mode) {
-  if (is_opus_) {
-    *enable_dtx = false;
-    *enable_vad = false;
-    return 0;
-  }
-  // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting and
-  // the |enable_vad| is set equal to |enable_dtx|.
-  // The case when VAD is enabled but DTX is disabled may result in a
-  // kPassiveNormalEncoded frame type, but this is not a case that VoE
-  // distinguishes from the cases where DTX is in fact used. In the case where
-  // DTX is enabled but VAD is disabled, the comment in the ACM interface states
-  // that VAD will be enabled anyway.
-  DCHECK_EQ(*enable_dtx, *enable_vad);
-  *enable_vad = *enable_dtx;
-  acm_codec_params_.enable_dtx = *enable_dtx;
-  acm_codec_params_.enable_vad = *enable_vad;
-  acm_codec_params_.vad_mode = *mode;
-  if (acm_codec_params_.enable_dtx && !cng_encoder_) {
-    ResetAudioEncoder();
-  } else if (!acm_codec_params_.enable_dtx && cng_encoder_) {
-    cng_encoder_.reset();
-    encoder_ = audio_encoder_.get();
-  }
-  return 0;
-}
-
-void ACMGenericCodec::SetCngPt(int sample_rate_hz, int payload_type) {
-  SetPtInMap(&cng_pt_, sample_rate_hz, payload_type);
-  ResetAudioEncoder();
-}
-
-void ACMGenericCodec::SetRedPt(int sample_rate_hz, int payload_type) {
-  SetPtInMap(&red_pt_, sample_rate_hz, payload_type);
-  ResetAudioEncoder();
-}
-
-int32_t ACMGenericCodec::SetISACMaxPayloadSize(
-    const uint16_t max_payload_len_bytes) {
-  if (!is_isac_)
-    return -1;  // Needed for tests to pass.
-  max_payload_size_bytes_ = max_payload_len_bytes;
-  ResetAudioEncoder();
-  return 0;
-}
-
-int32_t ACMGenericCodec::SetISACMaxRate(const uint32_t max_rate_bps) {
-  if (!is_isac_)
-    return -1;  // Needed for tests to pass.
-  max_rate_bps_ = max_rate_bps;
-  ResetAudioEncoder();
-  return 0;
-}
-
-int ACMGenericCodec::SetOpusMaxPlaybackRate(int frequency_hz) {
-  if (!is_opus_)
-    return -1;  // Needed for tests to pass.
-  max_playback_rate_hz_ = frequency_hz;
-  ResetAudioEncoder();
-  return 0;
-}
-
-AudioDecoder* ACMGenericCodec::Decoder() {
-  return decoder_proxy_.IsSet() ? &decoder_proxy_ : nullptr;
-}
-
-int ACMGenericCodec::EnableOpusDtx(bool force_voip) {
-  if (!is_opus_)
-    return -1;  // Needed for tests to pass.
-  if (!force_voip &&
-      GetOpusApplication(encoder_->NumChannels(), true) != kVoip) {
-      // Opus DTX can only be enabled when application mode is KVoip.
-      return -1;
-  }
-  opus_application_ = kVoip;
-  opus_application_set_ = true;
-  opus_dtx_enabled_ = true;
-  ResetAudioEncoder();
-  return 0;
-}
-
-int ACMGenericCodec::DisableOpusDtx() {
-  if (!is_opus_)
-    return -1;  // Needed for tests to pass.
-  opus_dtx_enabled_ = false;
-  ResetAudioEncoder();
-  return 0;
-}
-
-int ACMGenericCodec::SetFEC(bool enable_fec) {
-  if (!HasInternalFEC())
-    return enable_fec ? -1 : 0;
-  if (fec_enabled_ != enable_fec) {
-    fec_enabled_ = enable_fec;
-    ResetAudioEncoder();
-  }
-  return 0;
-}
-
-int ACMGenericCodec::SetOpusApplication(OpusApplicationMode application,
-                                        bool disable_dtx_if_needed) {
-  if (opus_dtx_enabled_ && application == kAudio) {
-    if (disable_dtx_if_needed) {
-      opus_dtx_enabled_ = false;
-    } else {
-      // Opus can only be set to kAudio when DTX is off.
-      return -1;
-    }
-  }
-  opus_application_ = application;
-  opus_application_set_ = true;
-  ResetAudioEncoder();
-  return 0;
-}
-
-int ACMGenericCodec::SetPacketLossRate(int loss_rate) {
-  encoder_->SetProjectedPacketLossRate(loss_rate / 100.0);
-  loss_rate_ = loss_rate;
-  return 0;
-}
-
-int ACMGenericCodec::SetCopyRed(bool enable) {
-  copy_red_enabled_ = enable;
-  ResetAudioEncoder();
-  return copy_red_enabled_ == enable ? 0 : -1;
-}
-
-AudioEncoder* ACMGenericCodec::GetAudioEncoder() {
-  return encoder_;
-}
-
-const AudioEncoder* ACMGenericCodec::GetAudioEncoder() const {
-  return encoder_;
-}
-
-}  // namespace acm2
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
deleted file mode 100644
index d491dfd40a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
+++ /dev/null
@@ -1,452 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
-
-#include <map>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-#define MAX_FRAME_SIZE_10MSEC 6
-
-// forward declaration
-struct WebRtcVadInst;
-struct WebRtcCngEncInst;
-
-namespace webrtc {
-
-struct WebRtcACMCodecParams;
-struct CodecInst;
-class CriticalSectionWrapper;
-
-namespace acm2 {
-
-// forward declaration
-class AcmReceiver;
-
-// Proxy for AudioDecoder
-class AudioDecoderProxy final : public AudioDecoder {
- public:
-  AudioDecoderProxy();
-  void SetDecoder(AudioDecoder* decoder);
-  bool IsSet() const;
-  int Decode(const uint8_t* encoded,
-             size_t encoded_len,
-             int sample_rate_hz,
-             size_t max_decoded_bytes,
-             int16_t* decoded,
-             SpeechType* speech_type) override;
-  int DecodeRedundant(const uint8_t* encoded,
-                      size_t encoded_len,
-                      int sample_rate_hz,
-                      size_t max_decoded_bytes,
-                      int16_t* decoded,
-                      SpeechType* speech_type) override;
-  bool HasDecodePlc() const override;
-  int DecodePlc(int num_frames, int16_t* decoded) override;
-  int Init() override;
-  int IncomingPacket(const uint8_t* payload,
-                     size_t payload_len,
-                     uint16_t rtp_sequence_number,
-                     uint32_t rtp_timestamp,
-                     uint32_t arrival_timestamp) override;
-  int ErrorCode() override;
-  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
-  int PacketDurationRedundant(const uint8_t* encoded,
-                              size_t encoded_len) const override;
-  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
-  CNG_dec_inst* CngDecoderInstance() override;
-  size_t Channels() const override;
-
- private:
-  rtc::scoped_ptr<CriticalSectionWrapper> decoder_lock_;
-  AudioDecoder* decoder_ GUARDED_BY(decoder_lock_);
-};
-
-class ACMGenericCodec {
- public:
-  ACMGenericCodec(const CodecInst& codec_inst,
-                  int cng_pt_nb,
-                  int cng_pt_wb,
-                  int cng_pt_swb,
-                  int cng_pt_fb,
-                  bool enable_red,
-                  int red_pt_nb);
-  ~ACMGenericCodec();
-
-  ///////////////////////////////////////////////////////////////////////////
-  // ACMGenericCodec* CreateInstance();
-  // The function will be used for FEC. It is not implemented yet.
-  //
-  ACMGenericCodec* CreateInstance();
-
-  ///////////////////////////////////////////////////////////////////////////
-  // bool EncoderInitialized();
-  //
-  // Return value:
-  //   True if the encoder is successfully initialized,
-  //   false otherwise.
-  //
-  bool EncoderInitialized();
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int16_t EncoderParams()
-  // It is called to get encoder parameters. It will call
-  // EncoderParamsSafe() in turn.
-  //
-  // Output:
-  //   -enc_params         : a buffer where the encoder parameters is
-  //                         written to. If the encoder is not
-  //                         initialized this buffer is filled with
-  //                         invalid values
-  // Return value:
-  //   -1 if the encoder is not initialized,
-  //    0 otherwise.
-  //
-  int16_t EncoderParams(WebRtcACMCodecParams* enc_params) const;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int16_t InitEncoder(...)
-  // This function is called to initialize the encoder with the given
-  // parameters.
-  //
-  // Input:
-  //   -codec_params        : parameters of encoder.
-  //   -force_initialization: if false the initialization is invoked only if
-  //                          the encoder is not initialized. If true the
-  //                          encoder is forced to (re)initialize.
-  //
-  // Return value:
-  //   0 if could initialize successfully,
-  //  -1 if failed to initialize.
-  //
-  //
-  int16_t InitEncoder(WebRtcACMCodecParams* codec_params,
-                      bool force_initialization);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // uint32_t NoMissedSamples()
-  // This function returns the number of samples which are overwritten in
-  // the audio buffer. The audio samples are overwritten if the input audio
-  // buffer is full, but Add10MsData() is called. (We might remove this
-  // function if it is not used)
-  //
-  // Return Value:
-  //   Number of samples which are overwritten.
-  //
-  uint32_t NoMissedSamples() const;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // void ResetNoMissedSamples()
-  // This function resets the number of overwritten samples to zero.
-  // (We might remove this function if we remove NoMissedSamples())
-  //
-  void ResetNoMissedSamples();
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int16_t SetBitRate()
-  // The function is called to set the encoding rate.
-  //
-  // Input:
-  //   -bitrate_bps        : encoding rate in bits per second
-  //
-  // Return value:
-  //   -1 if failed to set the rate, due to invalid input or given
-  //      codec is not rate-adjustable.
-  //    0 if the rate is adjusted successfully
-  //
-  int16_t SetBitRate(const int32_t bitrate_bps);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // uint32_t EarliestTimestamp()
-  // Returns the timestamp of the first 10 ms in audio buffer. This is used
-  // to identify if a synchronization of two encoders is required.
-  //
-  // Return value:
-  //   timestamp of the first 10 ms audio in the audio buffer.
-  //
-  uint32_t EarliestTimestamp() const;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int16_t SetVAD()
-  // This is called to set VAD & DTX. If the codec has internal DTX, it will
-  // be used. If DTX is enabled and the codec does not have internal DTX,
-  // WebRtc-VAD will be used to decide if the frame is active. If DTX is
-  // disabled but VAD is enabled, the audio is passed through VAD to label it
-  // as active or passive, but the frame is  encoded normally. However the
-  // bit-stream is labeled properly so that ACM::Process() can use this
-  // information. In case of failure, the previous states of the VAD & DTX
-  // are kept.
-  //
-  // Inputs/Output:
-  //   -enable_dtx         : if true DTX will be enabled otherwise the DTX is
-  //                         disabled. If codec has internal DTX that will be
-  //                         used, otherwise WebRtc-CNG is used. In the latter
-  //                         case VAD is automatically activated.
-  //   -enable_vad         : if true WebRtc-VAD is enabled, otherwise VAD is
-  //                         disabled, except for the case that DTX is enabled
-  //                         but codec doesn't have internal DTX. In this case
-  //                         VAD is enabled regardless of the value of
-  //                         |enable_vad|.
-  //   -mode               : this specifies the aggressiveness of VAD.
-  //
-  // Return value
-  //   -1 if failed to set DTX & VAD as specified,
-  //    0 if succeeded.
-  //
-  int16_t SetVAD(bool* enable_dtx, bool* enable_vad, ACMVADMode* mode);
-
-  // Registers comfort noise at |sample_rate_hz| to use |payload_type|.
-  void SetCngPt(int sample_rate_hz, int payload_type);
-
-  // Registers RED at |sample_rate_hz| to use |payload_type|.
-  void SetRedPt(int sample_rate_hz, int payload_type);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // UpdateEncoderSampFreq()
-  // Call this function to update the encoder sampling frequency. This
-  // is for codecs where one payload-name supports several encoder sampling
-  // frequencies. Otherwise, to change the sampling frequency we need to
-  // register new codec. ACM will consider that as registration of a new
-  // codec, not a change in parameter. For iSAC, switching from WB to SWB
-  // is treated as a change in parameter. Therefore, we need this function.
-  //
-  // Input:
-  //   -samp_freq_hz        : encoder sampling frequency.
-  //
-  // Return value:
-  //   -1 if failed, or if this is meaningless for the given codec.
-  //    0 if succeeded.
-  //
-  int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // EncoderSampFreq()
-  // Get the sampling frequency that the encoder (WebRtc wrapper) expects.
-  //
-  // Output:
-  //   -samp_freq_hz       : sampling frequency, in Hertz, which the encoder
-  //                         should be fed with.
-  //
-  // Return value:
-  //   -1 if failed to output sampling rate.
-  //    0 if the sample rate is returned successfully.
-  //
-  int16_t EncoderSampFreq(uint16_t* samp_freq_hz);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // SetISACMaxPayloadSize()
-  // Set the maximum payload size of iSAC packets. No iSAC payload,
-  // regardless of its frame-size, may exceed the given limit. For
-  // an iSAC payload of size B bits and frame-size T sec we have;
-  // (B < max_payload_len_bytes * 8) and (B/T < max_rate_bit_per_sec), c.f.
-  // SetISACMaxRate().
-  //
-  // Input:
-  //   -max_payload_len_bytes : maximum payload size in bytes.
-  //
-  // Return value:
-  //   -1 if failed to set the maximum  payload-size.
-  //    0 if the given length is set successfully.
-  //
-  int32_t SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // SetISACMaxRate()
-  // Set the maximum instantaneous rate of iSAC. For a payload of B bits
-  // with a frame-size of T sec the instantaneous rate is B/T bits per
-  // second. Therefore, (B/T < max_rate_bit_per_sec) and
-  // (B < max_payload_len_bytes * 8) are always satisfied for iSAC payloads,
-  // c.f SetISACMaxPayloadSize().
-  //
-  // Input:
-  //   -max_rate_bps       : maximum instantaneous bit-rate given in bits/sec.
-  //
-  // Return value:
-  //   -1 if failed to set the maximum rate.
-  //    0 if the maximum rate is set successfully.
-  //
-  int32_t SetISACMaxRate(const uint32_t max_rate_bps);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int SetOpusApplication(OpusApplicationMode application,
-  //                        bool disable_dtx_if_needed)
-  // Sets the intended application for the Opus encoder. Opus uses this to
-  // optimize the encoding for applications like VOIP and music. Currently, two
-  // modes are supported: kVoip and kAudio. kAudio is only allowed when Opus
-  // DTX is switched off. If DTX is on, and |application| == kAudio, a failure
-  // will be triggered unless |disable_dtx_if_needed| == true, for which, the
-  // DTX will be forced off.
-  //
-  // Input:
-  //   - application            : intended application.
-  //   - disable_dtx_if_needed  : whether to force Opus DTX to stop when needed.
-  //
-  // Return value:
-  //   -1 if failed or on codecs other than Opus.
-  //    0 if succeeded.
-  //
-  int SetOpusApplication(OpusApplicationMode application,
-                         bool disable_dtx_if_needed);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int SetOpusMaxPlaybackRate()
-  // Sets maximum playback rate the receiver will render, if the codec is Opus.
-  // This is to tell Opus that it is enough to code the input audio up to a
-  // bandwidth. Opus can take this information to optimize the bit rate and
-  // increase the computation efficiency.
-  //
-  // Input:
-  //   -frequency_hz      : maximum playback rate in Hz.
-  //
-  // Return value:
-  //   -1 if failed or on codecs other than Opus.
-  //    0 if succeeded.
-  //
-  int SetOpusMaxPlaybackRate(int /* frequency_hz */);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // EnableOpusDtx(bool force_voip)
-  // Enable the DTX, if the codec is Opus. Currently, DTX can only be enabled
-  // when the application mode is kVoip. If |force_voip| == true, the
-  // application mode will be forced to kVoip. Otherwise, a failure will be
-  // triggered if current application mode is kAudio.
-  // Input:
-  //   - force_voip  : whether to force application mode to kVoip.
-  // Return value:
-  //   -1 if failed or on codecs other than Opus.
-  //    0 if succeeded.
-  //
-  int EnableOpusDtx(bool force_voip);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // DisbleOpusDtx()
-  // Disable the DTX, if the codec is Opus.
-  // Return value:
-  //   -1 if failed or on codecs other than Opus.
-  //    0 if succeeded.
-  //
-  int DisableOpusDtx();
-
-  ///////////////////////////////////////////////////////////////////////////
-  // HasFrameToEncode()
-  // Returns true if there is enough audio buffered for encoding, such that
-  // calling Encode() will return a payload.
-  //
-  bool HasFrameToEncode() const;
-
-  // Returns a pointer to the AudioDecoder part of a joint encoder-decoder
-  // object, if it exists. Otherwise, nullptr is returned.
-  AudioDecoder* Decoder();
-
-  ///////////////////////////////////////////////////////////////////////////
-  // bool HasInternalFEC()
-  // Used to check if the codec has internal FEC.
-  //
-  // Return value:
-  //   true if the codec has an internal FEC, e.g. Opus.
-  //   false otherwise.
-  //
-  bool HasInternalFEC() const {
-    return has_internal_fec_;
-  }
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int SetFEC();
-  // Sets the codec internal FEC. No effects on codecs that do not provide
-  // internal FEC.
-  //
-  // Input:
-  //   -enable_fec         : if true FEC will be enabled otherwise the FEC is
-  //                         disabled.
-  //
-  // Return value:
-  //   -1 if failed,
-  //    0 if succeeded.
-  //
-  int SetFEC(bool enable_fec);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int SetPacketLossRate()
-  // Sets expected packet loss rate for encoding. Some encoders provide packet
-  // loss gnostic encoding to make stream less sensitive to packet losses,
-  // through e.g., FEC. No effects on codecs that do not provide such encoding.
-  //
-  // Input:
-  //   -loss_rate          : expected packet loss rate (0 -- 100 inclusive).
-  //
-  // Return value:
-  //   -1 if failed,
-  //    0 if succeeded or packet loss rate is ignored.
-  //
-  int SetPacketLossRate(int /* loss_rate */);
-
-  ///////////////////////////////////////////////////////////////////////////
-  // int SetCopyRed()
-  // Enable or disable copy RED. It fails if there is no RED payload that
-  // matches the codec, e.g., sample rate differs.
-  //
-  // Return value:
-  //   -1 if failed,
-  //    0 if succeeded.
-  int SetCopyRed(bool enable);
-
-  AudioEncoder* GetAudioEncoder();
-
-  const AudioEncoder* GetAudioEncoder() const;
-
- private:
-  bool has_internal_fec_;
-
-  bool copy_red_enabled_;
-
-  void ResetAudioEncoder();
-
-  OpusApplicationMode GetOpusApplication(int num_channels,
-                                         bool enable_dtx) const;
-
-  rtc::scoped_ptr<AudioEncoder> audio_encoder_;
-  rtc::scoped_ptr<AudioEncoder> cng_encoder_;
-  rtc::scoped_ptr<AudioEncoder> red_encoder_;
-  AudioEncoder* encoder_;
-  AudioDecoderProxy decoder_proxy_;
-  WebRtcACMCodecParams acm_codec_params_;
-  int bitrate_bps_;
-  bool fec_enabled_;
-  int loss_rate_;
-  int max_playback_rate_hz_;
-  int max_payload_size_bytes_;
-  int max_rate_bps_;
-  bool opus_dtx_enabled_;
-  bool is_opus_;
-  bool is_isac_;
-  // Map from payload type to CNG sample rate (Hz).
-  std::map<int, int> cng_pt_;
-  // Map from payload type to RED sample rate (Hz).
-  std::map<int, int> red_pt_;
-  OpusApplicationMode opus_application_;
-  bool opus_application_set_;
-};
-
-}  // namespace acm2
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec_opus_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec_opus_test.cc
deleted file mode 100644
index de5cc855a8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec_opus_test.cc
+++ /dev/null
@@ -1,145 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
-
-namespace webrtc {
-namespace acm2 {
-
-#ifdef WEBRTC_CODEC_OPUS
-namespace {
-const CodecInst kDefaultOpusCodecInst = {105, "opus", 48000, 960, 1, 32000};
-const int kCngPt = 255;  // Not using CNG in this test.
-const int kRedPt = 255;  // Not using RED in this test.
-}  // namespace
-
-class AcmGenericCodecOpusTest : public ::testing::Test {
- protected:
-  AcmGenericCodecOpusTest() {
-    acm_codec_params_ = {kDefaultOpusCodecInst, false, false, VADNormal};
-  }
-
-  void CreateCodec() {
-    codec_wrapper_.reset(new ACMGenericCodec(
-        acm_codec_params_.codec_inst, kCngPt, kCngPt, kCngPt, kCngPt,
-        false /* enable RED */, kRedPt));
-    ASSERT_TRUE(codec_wrapper_);
-    ASSERT_EQ(0, codec_wrapper_->InitEncoder(&acm_codec_params_, true));
-  }
-
-  const AudioEncoderOpus* GetAudioEncoderOpus() {
-    const AudioEncoderOpus* ptr = static_cast<const AudioEncoderOpus*>(
-        codec_wrapper_->GetAudioEncoder());
-    EXPECT_NE(nullptr, ptr);
-    return ptr;
-  }
-  WebRtcACMCodecParams acm_codec_params_;
-  rtc::scoped_ptr<ACMGenericCodec> codec_wrapper_;
-};
-
-TEST_F(AcmGenericCodecOpusTest, DefaultApplicationModeMono) {
-  acm_codec_params_.codec_inst.channels = 1;
-  CreateCodec();
-  EXPECT_EQ(AudioEncoderOpus::kVoip, GetAudioEncoderOpus()->application());
-}
-
-TEST_F(AcmGenericCodecOpusTest, DefaultApplicationModeStereo) {
-  acm_codec_params_.codec_inst.channels = 2;
-  CreateCodec();
-  EXPECT_EQ(AudioEncoderOpus::kAudio, GetAudioEncoderOpus()->application());
-}
-
-TEST_F(AcmGenericCodecOpusTest, ChangeApplicationMode) {
-  // Create a stereo encoder.
-  acm_codec_params_.codec_inst.channels = 2;
-  CreateCodec();
-  // Verify that the mode is kAudio.
-  const AudioEncoderOpus* opus_ptr = GetAudioEncoderOpus();
-  EXPECT_EQ(AudioEncoderOpus::kAudio, opus_ptr->application());
-
-  // Change mode.
-  EXPECT_EQ(0, codec_wrapper_->SetOpusApplication(kVoip, false));
-  // Verify that the AudioEncoder object was changed.
-  EXPECT_NE(opus_ptr, GetAudioEncoderOpus());
-  EXPECT_EQ(AudioEncoderOpus::kVoip, GetAudioEncoderOpus()->application());
-}
-
-TEST_F(AcmGenericCodecOpusTest, ResetWontChangeApplicationMode) {
-  // Create a stereo encoder.
-  acm_codec_params_.codec_inst.channels = 2;
-  CreateCodec();
-  const AudioEncoderOpus* opus_ptr = GetAudioEncoderOpus();
-  // Verify that the mode is kAudio.
-  EXPECT_EQ(AudioEncoderOpus::kAudio, opus_ptr->application());
-
-  // Trigger a reset.
-  ASSERT_EQ(0, codec_wrapper_->InitEncoder(&acm_codec_params_, false));
-  // Verify that the AudioEncoder object changed.
-  EXPECT_NE(opus_ptr, GetAudioEncoderOpus());
-  // Verify that the mode is still kAudio.
-  EXPECT_EQ(AudioEncoderOpus::kAudio, GetAudioEncoderOpus()->application());
-
-  // Now change to kVoip.
-  EXPECT_EQ(0, codec_wrapper_->SetOpusApplication(kVoip, false));
-  EXPECT_EQ(AudioEncoderOpus::kVoip, GetAudioEncoderOpus()->application());
-
-  opus_ptr = GetAudioEncoderOpus();
-  // Trigger a reset again.
-  ASSERT_EQ(0, codec_wrapper_->InitEncoder(&acm_codec_params_, false));
-  // Verify that the AudioEncoder object changed.
-  EXPECT_NE(opus_ptr, GetAudioEncoderOpus());
-  // Verify that the mode is still kVoip.
-  EXPECT_EQ(AudioEncoderOpus::kVoip, GetAudioEncoderOpus()->application());
-}
-
-TEST_F(AcmGenericCodecOpusTest, ToggleDtx) {
-  // Create a stereo encoder.
-  acm_codec_params_.codec_inst.channels = 2;
-  CreateCodec();
-  // Verify that the mode is still kAudio.
-  EXPECT_EQ(AudioEncoderOpus::kAudio, GetAudioEncoderOpus()->application());
-
-  // DTX is not allowed in audio mode, if mode forcing flag is false.
-  EXPECT_EQ(-1, codec_wrapper_->EnableOpusDtx(false));
-  EXPECT_EQ(AudioEncoderOpus::kAudio, GetAudioEncoderOpus()->application());
-
-  // DTX will be on, if mode forcing flag is true. Then application mode is
-  // switched to kVoip.
-  EXPECT_EQ(0, codec_wrapper_->EnableOpusDtx(true));
-  EXPECT_EQ(AudioEncoderOpus::kVoip, GetAudioEncoderOpus()->application());
-
-  // Audio mode is not allowed when DTX is on, and DTX forcing flag is false.
-  EXPECT_EQ(-1, codec_wrapper_->SetOpusApplication(kAudio, false));
-  EXPECT_TRUE(GetAudioEncoderOpus()->dtx_enabled());
-
-  // Audio mode will be set, if DTX forcing flag is true. Then DTX is switched
-  // off.
-  EXPECT_EQ(0, codec_wrapper_->SetOpusApplication(kAudio, true));
-  EXPECT_FALSE(GetAudioEncoderOpus()->dtx_enabled());
-
-  // Now we set VOIP mode. The DTX forcing flag has no effect.
-  EXPECT_EQ(0, codec_wrapper_->SetOpusApplication(kVoip, true));
-  EXPECT_FALSE(GetAudioEncoderOpus()->dtx_enabled());
-
-  // In VOIP mode, we can enable DTX with mode forcing flag being false.
-  EXPECT_EQ(0, codec_wrapper_->EnableOpusDtx(false));
-
-  // Turn off DTX.
-  EXPECT_EQ(0, codec_wrapper_->DisableOpusDtx());
-
-  // When DTX is off, we can set Audio mode with DTX forcing flag being false.
-  EXPECT_EQ(0, codec_wrapper_->SetOpusApplication(kAudio, false));
-}
-#endif  // WEBRTC_CODEC_OPUS
-
-}  // namespace acm2
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec_test.cc
deleted file mode 100644
index 3f2e9e6269..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_generic_codec_test.cc
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/safe_conversions.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
-
-namespace webrtc {
-namespace acm2 {
-
-namespace {
-const int kDataLengthSamples = 80;
-const int kPacketSizeSamples = 2 * kDataLengthSamples;
-const int16_t kZeroData[kDataLengthSamples] = {0};
-const CodecInst kDefaultCodecInst =
-    {0, "pcmu", 8000, kPacketSizeSamples, 1, 64000};
-const int kCngPt = 13;
-const int kNoCngPt = 255;
-const int kRedPt = 255;  // Not using RED in this test.
-}  // namespace
-
-class AcmGenericCodecTest : public ::testing::Test {
- protected:
-  AcmGenericCodecTest() : timestamp_(0) {
-    acm_codec_params_ = {kDefaultCodecInst, true, true, VADNormal};
-  }
-
-  void CreateCodec() {
-    codec_.reset(new ACMGenericCodec(acm_codec_params_.codec_inst, kCngPt,
-                                     kNoCngPt, kNoCngPt, kNoCngPt,
-                                     false /* enable RED */, kRedPt));
-    ASSERT_TRUE(codec_);
-    ASSERT_EQ(0, codec_->InitEncoder(&acm_codec_params_, true));
-  }
-
-  void EncodeAndVerify(size_t expected_out_length,
-                       uint32_t expected_timestamp,
-                       int expected_payload_type,
-                       int expected_send_even_if_empty) {
-    uint8_t out[kPacketSizeSamples];
-    AudioEncoder::EncodedInfo encoded_info;
-    encoded_info = codec_->GetAudioEncoder()->Encode(
-        timestamp_, kZeroData, kDataLengthSamples, kPacketSizeSamples, out);
-    timestamp_ += kDataLengthSamples;
-    EXPECT_TRUE(encoded_info.redundant.empty());
-    EXPECT_EQ(expected_out_length, encoded_info.encoded_bytes);
-    EXPECT_EQ(expected_timestamp, encoded_info.encoded_timestamp);
-    if (expected_payload_type >= 0)
-      EXPECT_EQ(expected_payload_type, encoded_info.payload_type);
-    if (expected_send_even_if_empty >= 0)
-      EXPECT_EQ(static_cast<bool>(expected_send_even_if_empty),
-                encoded_info.send_even_if_empty);
-  }
-
-  WebRtcACMCodecParams acm_codec_params_;
-  rtc::scoped_ptr<ACMGenericCodec> codec_;
-  uint32_t timestamp_;
-};
-
-// This test verifies that CNG frames are delivered as expected. Since the frame
-// size is set to 20 ms, we expect the first encode call to produce no output
-// (which is signaled as 0 bytes output of type kNoEncoding). The next encode
-// call should produce one SID frame of 9 bytes. The third call should not
-// result in any output (just like the first one). The fourth and final encode
-// call should produce an "empty frame", which is like no output, but with
-// AudioEncoder::EncodedInfo::send_even_if_empty set to true. (The reason to
-// produce an empty frame is to drive sending of DTMF packets in the RTP/RTCP
-// module.)
-TEST_F(AcmGenericCodecTest, VerifyCngFrames) {
-  CreateCodec();
-  uint32_t expected_timestamp = timestamp_;
-  // Verify no frame.
-  {
-    SCOPED_TRACE("First encoding");
-    EncodeAndVerify(0, expected_timestamp, -1, -1);
-  }
-
-  // Verify SID frame delivered.
-  {
-    SCOPED_TRACE("Second encoding");
-    EncodeAndVerify(9, expected_timestamp, kCngPt, 1);
-  }
-
-  // Verify no frame.
-  {
-    SCOPED_TRACE("Third encoding");
-    EncodeAndVerify(0, expected_timestamp, -1, -1);
-  }
-
-  // Verify NoEncoding.
-  expected_timestamp += 2 * kDataLengthSamples;
-  {
-    SCOPED_TRACE("Fourth encoding");
-    EncodeAndVerify(0, expected_timestamp, kCngPt, 1);
-  }
-}
-
-}  // namespace acm2
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
deleted file mode 100644
index e74ce2270c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
-
-#include <assert.h>
-#include <stdio.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-
-namespace webrtc {
-namespace test {
-
-AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
-                               AudioSink* audio_sink,
-                               int output_freq_hz,
-                               NumOutputChannels exptected_output_channels)
-    : clock_(0),
-      packet_source_(packet_source),
-      audio_sink_(audio_sink),
-      output_freq_hz_(output_freq_hz),
-      exptected_output_channels_(exptected_output_channels) {
-  webrtc::AudioCoding::Config config;
-  config.clock = &clock_;
-  config.playout_frequency_hz = output_freq_hz_;
-  acm_.reset(webrtc::AudioCoding::Create(config));
-}
-
-void AcmReceiveTest::RegisterDefaultCodecs() {
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
-#ifndef WEBRTC_ANDROID
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 105));
-#endif
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 107));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 108));
-  ASSERT_TRUE(
-      acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 109));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B_2ch, 111));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb_2ch, 112));
-  ASSERT_TRUE(
-      acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 113));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127));
-}
-
-void AcmReceiveTest::RegisterNetEqTestCodecs() {
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
-#ifndef WEBRTC_ANDROID
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 124));
-#endif
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 93));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 94));
-  ASSERT_TRUE(
-      acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 95));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 117));
-}
-
-void AcmReceiveTest::Run() {
-  for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
-       packet.reset(packet_source_->NextPacket())) {
-    // Pull audio until time to insert packet.
-    while (clock_.TimeInMilliseconds() < packet->time_ms()) {
-      AudioFrame output_frame;
-      EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
-      EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
-      const int samples_per_block = output_freq_hz_ * 10 / 1000;
-      EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
-      if (exptected_output_channels_ != kArbitraryChannels) {
-        if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
-          // Don't check number of channels for PLC output, since each test run
-          // usually starts with a short period of mono PLC before decoding the
-          // first packet.
-        } else {
-          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
-        }
-      }
-      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
-      clock_.AdvanceTimeMilliseconds(10);
-    }
-
-    // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
-    WebRtcRTPHeader header;
-    header.header = packet->header();
-    header.frameType = kAudioFrameSpeech;
-    memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
-    EXPECT_TRUE(acm_->InsertPacket(packet->payload(),
-                                   packet->payload_length_bytes(),
-                                   header))
-        << "Failure when inserting packet:" << std::endl
-        << "  PT = " << static_cast<int>(header.header.payloadType) << std::endl
-        << "  TS = " << header.header.timestamp << std::endl
-        << "  SN = " << header.header.sequenceNumber;
-  }
-}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h
deleted file mode 100644
index 552a7486a8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-
-namespace webrtc {
-class AudioCoding;
-struct CodecInst;
-
-namespace test {
-class AudioSink;
-class PacketSource;
-
-class AcmReceiveTest {
- public:
-  enum NumOutputChannels {
-    kArbitraryChannels = 0,
-    kMonoOutput = 1,
-    kStereoOutput = 2
-  };
-
-  AcmReceiveTest(
-      PacketSource* packet_source,
-      AudioSink* audio_sink,
-      int output_freq_hz,
-      NumOutputChannels exptected_output_channels);
-  virtual ~AcmReceiveTest() {}
-
-  // Registers the codecs with default parameters from ACM.
-  void RegisterDefaultCodecs();
-
-  // Registers codecs with payload types matching the pre-encoded NetEq test
-  // files.
-  void RegisterNetEqTestCodecs();
-
-  // Runs the test and returns true if successful.
-  void Run();
-
- private:
-  SimulatedClock clock_;
-  rtc::scoped_ptr<AudioCoding> acm_;
-  PacketSource* packet_source_;
-  AudioSink* audio_sink_;
-  const int output_freq_hz_;
-  NumOutputChannels exptected_output_channels_;
-
-  DISALLOW_COPY_AND_ASSIGN(AcmReceiveTest);
-};
-
-}  // namespace test
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
deleted file mode 100644
index b6333ec1b0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ /dev/null
@@ -1,842 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
-
-#include <stdlib.h>  // malloc
-
-#include <algorithm>  // sort
-#include <vector>
-
-#include "webrtc/base/format_macros.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/main/acm2/nack.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-namespace {
-
-const int kNackThresholdPackets = 2;
-
-// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
-// before the call to this function.
-void SetAudioFrameActivityAndType(bool vad_enabled,
-                                  NetEqOutputType type,
-                                  AudioFrame* audio_frame) {
-  if (vad_enabled) {
-    switch (type) {
-      case kOutputNormal: {
-        audio_frame->vad_activity_ = AudioFrame::kVadActive;
-        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
-        break;
-      }
-      case kOutputVADPassive: {
-        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
-        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
-        break;
-      }
-      case kOutputCNG: {
-        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
-        audio_frame->speech_type_ = AudioFrame::kCNG;
-        break;
-      }
-      case kOutputPLC: {
-        // Don't change |audio_frame->vad_activity_|, it should be the same as
-        // |previous_audio_activity_|.
-        audio_frame->speech_type_ = AudioFrame::kPLC;
-        break;
-      }
-      case kOutputPLCtoCNG: {
-        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
-        audio_frame->speech_type_ = AudioFrame::kPLCCNG;
-        break;
-      }
-      default:
-        assert(false);
-    }
-  } else {
-    // Always return kVadUnknown when receive VAD is inactive
-    audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
-    switch (type) {
-      case kOutputNormal: {
-        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
-        break;
-      }
-      case kOutputCNG: {
-        audio_frame->speech_type_ = AudioFrame::kCNG;
-        break;
-      }
-      case kOutputPLC: {
-        audio_frame->speech_type_ = AudioFrame::kPLC;
-        break;
-      }
-      case kOutputPLCtoCNG: {
-        audio_frame->speech_type_ = AudioFrame::kPLCCNG;
-        break;
-      }
-      case kOutputVADPassive: {
-        // Normally, we should no get any VAD decision if post-decoding VAD is
-        // not active. However, if post-decoding VAD has been active then
-        // disabled, we might be here for couple of frames.
-        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
-        LOG_F(LS_WARNING) << "Post-decoding VAD is disabled but output is "
-            << "labeled VAD-passive";
-        break;
-      }
-      default:
-        assert(false);
-    }
-  }
-}
-
-// Is the given codec a CNG codec?
-bool IsCng(int codec_id) {
-  return (codec_id == ACMCodecDB::kCNNB || codec_id == ACMCodecDB::kCNWB ||
-      codec_id == ACMCodecDB::kCNSWB || codec_id == ACMCodecDB::kCNFB);
-}
-
-}  // namespace
-
-AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      id_(config.id),
-      last_audio_decoder_(nullptr),
-      previous_audio_activity_(AudioFrame::kVadPassive),
-      current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
-      audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
-      last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
-      nack_(),
-      nack_enabled_(false),
-      neteq_(NetEq::Create(config.neteq_config)),
-      vad_enabled_(true),
-      clock_(config.clock),
-      resampled_last_output_frame_(true),
-      av_sync_(false),
-      initial_delay_manager_(),
-      missing_packets_sync_stream_(),
-      late_packets_sync_stream_() {
-  assert(clock_);
-
-  // Make sure we are on the same page as NetEq. Post-decode VAD is disabled by
-  // default in NetEq4, however, Audio Conference Mixer relies on VAD decision
-  // and fails if VAD decision is not provided.
-  if (vad_enabled_)
-    neteq_->EnableVad();
-  else
-    neteq_->DisableVad();
-
-  memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
-  memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
-}
-
-AcmReceiver::~AcmReceiver() {
-  delete neteq_;
-}
-
-int AcmReceiver::SetMinimumDelay(int delay_ms) {
-  if (neteq_->SetMinimumDelay(delay_ms))
-    return 0;
-  LOG_FERR1(LS_ERROR, "AcmReceiver::SetExtraDelay", delay_ms);
-  return -1;
-}
-
-int AcmReceiver::SetInitialDelay(int delay_ms) {
-  if (delay_ms < 0 || delay_ms > 10000) {
-    return -1;
-  }
-  CriticalSectionScoped lock(crit_sect_.get());
-
-  if (delay_ms == 0) {
-    av_sync_ = false;
-    initial_delay_manager_.reset();
-    missing_packets_sync_stream_.reset();
-    late_packets_sync_stream_.reset();
-    neteq_->SetMinimumDelay(0);
-    return 0;
-  }
-
-  if (av_sync_ && initial_delay_manager_->PacketBuffered()) {
-    // Too late for this API. Only works before a call is started.
-    return -1;
-  }
-
-  // Most of places NetEq calls are not within AcmReceiver's critical section to
-  // improve performance. Here, this call has to be placed before the following
-  // block, therefore, we keep it inside critical section. Otherwise, we have to
-  // release |neteq_crit_sect_| and acquire it again, which seems an overkill.
-  if (!neteq_->SetMinimumDelay(delay_ms))
-    return -1;
-
-  const int kLatePacketThreshold = 5;
-  av_sync_ = true;
-  initial_delay_manager_.reset(new InitialDelayManager(delay_ms,
-                                                       kLatePacketThreshold));
-  missing_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
-  late_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
-  return 0;
-}
-
-int AcmReceiver::SetMaximumDelay(int delay_ms) {
-  if (neteq_->SetMaximumDelay(delay_ms))
-    return 0;
-  LOG_FERR1(LS_ERROR, "AcmReceiver::SetExtraDelay", delay_ms);
-  return -1;
-}
-
-int AcmReceiver::LeastRequiredDelayMs() const {
-  return neteq_->LeastRequiredDelayMs();
-}
-
-int AcmReceiver::current_sample_rate_hz() const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  return current_sample_rate_hz_;
-}
-
-// TODO(turajs): use one set of enumerators, e.g. the one defined in
-// common_types.h
-// TODO(henrik.lundin): This method is not used any longer. The call hierarchy
-// stops in voe::Channel::SetNetEQPlayoutMode(). Remove it.
-void AcmReceiver::SetPlayoutMode(AudioPlayoutMode mode) {
-  enum NetEqPlayoutMode playout_mode = kPlayoutOn;
-  switch (mode) {
-    case voice:
-      playout_mode = kPlayoutOn;
-      break;
-    case fax:  // No change to background noise mode.
-      playout_mode = kPlayoutFax;
-      break;
-    case streaming:
-      playout_mode = kPlayoutStreaming;
-      break;
-    case off:
-      playout_mode = kPlayoutOff;
-      break;
-  }
-  neteq_->SetPlayoutMode(playout_mode);
-}
-
-AudioPlayoutMode AcmReceiver::PlayoutMode() const {
-  AudioPlayoutMode acm_mode = voice;
-  NetEqPlayoutMode mode = neteq_->PlayoutMode();
-  switch (mode) {
-    case kPlayoutOn:
-      acm_mode = voice;
-      break;
-    case kPlayoutOff:
-      acm_mode = off;
-      break;
-    case kPlayoutFax:
-      acm_mode = fax;
-      break;
-    case kPlayoutStreaming:
-      acm_mode = streaming;
-      break;
-    default:
-      assert(false);
-  }
-  return acm_mode;
-}
-
-int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
-                              const uint8_t* incoming_payload,
-                              size_t length_payload) {
-  uint32_t receive_timestamp = 0;
-  InitialDelayManager::PacketType packet_type =
-      InitialDelayManager::kUndefinedPacket;
-  bool new_codec = false;
-  const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
-
-  {
-    CriticalSectionScoped lock(crit_sect_.get());
-
-    const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload);
-    if (!decoder) {
-      LOG_F(LS_ERROR) << "Payload-type "
-                      << static_cast<int>(header->payloadType)
-                      << " is not registered.";
-      return -1;
-    }
-    const int sample_rate_hz = ACMCodecDB::CodecFreq(decoder->acm_codec_id);
-    receive_timestamp = NowInTimestamp(sample_rate_hz);
-
-    if (IsCng(decoder->acm_codec_id)) {
-      // If this is a CNG while the audio codec is not mono skip pushing in
-      // packets into NetEq.
-      if (last_audio_decoder_ && last_audio_decoder_->channels > 1)
-        return 0;
-      packet_type = InitialDelayManager::kCngPacket;
-    } else if (decoder->acm_codec_id == ACMCodecDB::kAVT) {
-      packet_type = InitialDelayManager::kAvtPacket;
-    } else {
-      if (decoder != last_audio_decoder_) {
-        // This is either the first audio packet or send codec is changed.
-        // Therefore, either NetEq buffer is empty or will be flushed when this
-        // packet is inserted.
-        new_codec = true;
-
-        // Updating NACK'sampling rate is required, either first packet is
-        // received or codec is changed. Furthermore, reset is required if codec
-        // is changed (NetEq flushes its buffer so NACK should reset its list).
-        if (nack_enabled_) {
-          assert(nack_.get());
-          nack_->Reset();
-          nack_->UpdateSampleRate(sample_rate_hz);
-        }
-        last_audio_decoder_ = decoder;
-      }
-      packet_type = InitialDelayManager::kAudioPacket;
-    }
-
-    if (nack_enabled_) {
-      assert(nack_.get());
-      nack_->UpdateLastReceivedPacket(header->sequenceNumber,
-                                      header->timestamp);
-    }
-
-    if (av_sync_) {
-      assert(initial_delay_manager_.get());
-      assert(missing_packets_sync_stream_.get());
-      // This updates |initial_delay_manager_| and specifies an stream of
-      // sync-packets, if required to be inserted. We insert the sync-packets
-      // when AcmReceiver lock is released and |decoder_lock_| is acquired.
-      initial_delay_manager_->UpdateLastReceivedPacket(
-          rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
-          missing_packets_sync_stream_.get());
-    }
-  }  // |crit_sect_| is released.
-
-  // If |missing_packets_sync_stream_| is allocated then we are in AV-sync and
-  // we may need to insert sync-packets. We don't check |av_sync_| as we are
-  // outside AcmReceiver's critical section.
-  if (missing_packets_sync_stream_.get()) {
-    InsertStreamOfSyncPackets(missing_packets_sync_stream_.get());
-  }
-
-  if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
-                           receive_timestamp) < 0) {
-    LOG_FERR1(LS_ERROR, "AcmReceiver::InsertPacket",
-              static_cast<int>(header->payloadType))
-        << " Failed to insert packet";
-    return -1;
-  }
-  return 0;
-}
-
-int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
-  enum NetEqOutputType type;
-  int samples_per_channel;
-  int num_channels;
-  bool return_silence = false;
-
-  {
-    // Accessing members, take the lock.
-    CriticalSectionScoped lock(crit_sect_.get());
-
-    if (av_sync_) {
-      assert(initial_delay_manager_.get());
-      assert(late_packets_sync_stream_.get());
-      return_silence = GetSilence(desired_freq_hz, audio_frame);
-      uint32_t timestamp_now = NowInTimestamp(current_sample_rate_hz_);
-      initial_delay_manager_->LatePackets(timestamp_now,
-                                          late_packets_sync_stream_.get());
-    }
-  }
-
-  // If |late_packets_sync_stream_| is allocated then we have been in AV-sync
-  // mode and we might have to insert sync-packets.
-  if (late_packets_sync_stream_.get()) {
-    InsertStreamOfSyncPackets(late_packets_sync_stream_.get());
-    if (return_silence)  // Silence generated, don't pull from NetEq.
-      return 0;
-  }
-
-  // Accessing members, take the lock.
-  CriticalSectionScoped lock(crit_sect_.get());
-
-  // Always write the output to |audio_buffer_| first.
-  if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
-                       audio_buffer_.get(),
-                       &samples_per_channel,
-                       &num_channels,
-                       &type) != NetEq::kOK) {
-    LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
-    return -1;
-  }
-
-  // Update NACK.
-  int decoded_sequence_num = 0;
-  uint32_t decoded_timestamp = 0;
-  bool update_nack = nack_enabled_ &&  // Update NACK only if it is enabled.
-      neteq_->DecodedRtpInfo(&decoded_sequence_num, &decoded_timestamp);
-  if (update_nack) {
-    assert(nack_.get());
-    nack_->UpdateLastDecodedPacket(decoded_sequence_num, decoded_timestamp);
-  }
-
-  // NetEq always returns 10 ms of audio.
-  current_sample_rate_hz_ = samples_per_channel * 100;
-
-  // Update if resampling is required.
-  bool need_resampling = (desired_freq_hz != -1) &&
-      (current_sample_rate_hz_ != desired_freq_hz);
-
-  if (need_resampling && !resampled_last_output_frame_) {
-    // Prime the resampler with the last frame.
-    int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
-    samples_per_channel =
-        resampler_.Resample10Msec(last_audio_buffer_.get(),
-                                  current_sample_rate_hz_,
-                                  desired_freq_hz,
-                                  num_channels,
-                                  AudioFrame::kMaxDataSizeSamples,
-                                  temp_output);
-    if (samples_per_channel < 0) {
-      LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio")
-          << "Resampling last_audio_buffer_ failed.";
-      return -1;
-    }
-  }
-
-  // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
-  // through resampling, or through straight memcpy.
-  // TODO(henrik.lundin) Glitches in the output may appear if the output rate
-  // from NetEq changes. See WebRTC issue 3923.
-  if (need_resampling) {
-    samples_per_channel =
-        resampler_.Resample10Msec(audio_buffer_.get(),
-                                  current_sample_rate_hz_,
-                                  desired_freq_hz,
-                                  num_channels,
-                                  AudioFrame::kMaxDataSizeSamples,
-                                  audio_frame->data_);
-    if (samples_per_channel < 0) {
-      LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio")
-          << "Resampling audio_buffer_ failed.";
-      return -1;
-    }
-    resampled_last_output_frame_ = true;
-  } else {
-    resampled_last_output_frame_ = false;
-    // We might end up here ONLY if codec is changed.
-    memcpy(audio_frame->data_,
-           audio_buffer_.get(),
-           samples_per_channel * num_channels * sizeof(int16_t));
-  }
-
-  // Swap buffers, so that the current audio is stored in |last_audio_buffer_|
-  // for next time.
-  audio_buffer_.swap(last_audio_buffer_);
-
-  audio_frame->num_channels_ = num_channels;
-  audio_frame->samples_per_channel_ = samples_per_channel;
-  audio_frame->sample_rate_hz_ = samples_per_channel * 100;
-
-  // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
-  audio_frame->vad_activity_ = previous_audio_activity_;
-  SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
-  previous_audio_activity_ = audio_frame->vad_activity_;
-  call_stats_.DecodedByNetEq(audio_frame->speech_type_);
-
-  // Computes the RTP timestamp of the first sample in |audio_frame| from
-  // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
-  // |audio_frame|.
-  uint32_t playout_timestamp = 0;
-  if (GetPlayoutTimestamp(&playout_timestamp)) {
-    audio_frame->timestamp_ =
-        playout_timestamp - audio_frame->samples_per_channel_;
-  } else {
-    // Remain 0 until we have a valid |playout_timestamp|.
-    audio_frame->timestamp_ = 0;
-  }
-
-  return 0;
-}
-
-int32_t AcmReceiver::AddCodec(int acm_codec_id,
-                              uint8_t payload_type,
-                              int channels,
-                              AudioDecoder* audio_decoder) {
-  assert(acm_codec_id >= 0);
-  NetEqDecoder neteq_decoder = ACMCodecDB::neteq_decoders_[acm_codec_id];
-
-  // Make sure the right decoder is registered for Opus.
-  if (neteq_decoder == kDecoderOpus && channels == 2) {
-    neteq_decoder = kDecoderOpus_2ch;
-  }
-
-  CriticalSectionScoped lock(crit_sect_.get());
-
-  // The corresponding NetEq decoder ID.
-  // If this codec has been registered before.
-  auto it = decoders_.find(payload_type);
-  if (it != decoders_.end()) {
-    const Decoder& decoder = it->second;
-    if (decoder.acm_codec_id == acm_codec_id && decoder.channels == channels) {
-      // Re-registering the same codec. Do nothing and return.
-      return 0;
-    }
-
-    // Changing codec or number of channels. First unregister the old codec,
-    // then register the new one.
-    if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
-      LOG_F(LS_ERROR) << "Cannot remove payload "
-                      << static_cast<int>(payload_type);
-      return -1;
-    }
-
-    decoders_.erase(it);
-  }
-
-  int ret_val;
-  if (!audio_decoder) {
-    ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
-  } else {
-    ret_val = neteq_->RegisterExternalDecoder(
-        audio_decoder, neteq_decoder, payload_type);
-  }
-  if (ret_val != NetEq::kOK) {
-    LOG_FERR3(LS_ERROR, "AcmReceiver::AddCodec", acm_codec_id,
-              static_cast<int>(payload_type), channels);
-    return -1;
-  }
-
-  Decoder decoder;
-  decoder.acm_codec_id = acm_codec_id;
-  decoder.payload_type = payload_type;
-  decoder.channels = channels;
-  decoders_[payload_type] = decoder;
-  return 0;
-}
-
-void AcmReceiver::EnableVad() {
-  neteq_->EnableVad();
-  CriticalSectionScoped lock(crit_sect_.get());
-  vad_enabled_ = true;
-}
-
-void AcmReceiver::DisableVad() {
-  neteq_->DisableVad();
-  CriticalSectionScoped lock(crit_sect_.get());
-  vad_enabled_ = false;
-}
-
-void AcmReceiver::FlushBuffers() {
-  neteq_->FlushBuffers();
-}
-
-// If failed in removing one of the codecs, this method continues to remove as
-// many as it can.
-int AcmReceiver::RemoveAllCodecs() {
-  int ret_val = 0;
-  CriticalSectionScoped lock(crit_sect_.get());
-  for (auto it = decoders_.begin(); it != decoders_.end(); ) {
-    auto cur = it;
-    ++it;  // it will be valid even if we erase cur
-    if (neteq_->RemovePayloadType(cur->second.payload_type) == 0) {
-      decoders_.erase(cur);
-    } else {
-      LOG_F(LS_ERROR) << "Cannot remove payload "
-                      << static_cast<int>(cur->second.payload_type);
-      ret_val = -1;
-    }
-  }
-
-  // No codec is registered, invalidate last audio decoder.
-  last_audio_decoder_ = nullptr;
-  return ret_val;
-}
-
-int AcmReceiver::RemoveCodec(uint8_t payload_type) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  auto it = decoders_.find(payload_type);
-  if (it == decoders_.end()) {  // Such a payload-type is not registered.
-    return 0;
-  }
-  if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
-    LOG_FERR1(LS_ERROR, "AcmReceiver::RemoveCodec",
-              static_cast<int>(payload_type));
-    return -1;
-  }
-  if (last_audio_decoder_ == &it->second)
-    last_audio_decoder_ = nullptr;
-  decoders_.erase(it);
-  return 0;
-}
-
-void AcmReceiver::set_id(int id) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  id_ = id;
-}
-
-bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
-  if (av_sync_) {
-    assert(initial_delay_manager_.get());
-    if (initial_delay_manager_->buffering()) {
-      return initial_delay_manager_->GetPlayoutTimestamp(timestamp);
-    }
-  }
-  return neteq_->GetPlayoutTimestamp(timestamp);
-}
-
-int AcmReceiver::last_audio_codec_id() const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  return last_audio_decoder_ ? last_audio_decoder_->acm_codec_id : -1;
-}
-
-int AcmReceiver::RedPayloadType() const {
-  if (ACMCodecDB::kRED >= 0) {  // This ensures that RED is defined in WebRTC.
-    CriticalSectionScoped lock(crit_sect_.get());
-    for (const auto& decoder_pair : decoders_) {
-      const Decoder& decoder = decoder_pair.second;
-      if (decoder.acm_codec_id == ACMCodecDB::kRED)
-        return decoder.payload_type;
-    }
-  }
-  LOG_F(LS_WARNING) << "RED is not registered.";
-  return -1;
-}
-
-int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  if (!last_audio_decoder_) {
-    return -1;
-  }
-  memcpy(codec, &ACMCodecDB::database_[last_audio_decoder_->acm_codec_id],
-         sizeof(CodecInst));
-  codec->pltype = last_audio_decoder_->payload_type;
-  codec->channels = last_audio_decoder_->channels;
-  return 0;
-}
-
-void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
-  NetEqNetworkStatistics neteq_stat;
-  // NetEq function always returns zero, so we don't check the return value.
-  neteq_->NetworkStatistics(&neteq_stat);
-
-  acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
-  acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
-  acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
-  acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
-  acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
-  acm_stat->currentExpandRate = neteq_stat.expand_rate;
-  acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
-  acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
-  acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
-  acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
-  acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
-  acm_stat->addedSamples = neteq_stat.added_zero_samples;
-
-  std::vector<int> waiting_times;
-  neteq_->WaitingTimes(&waiting_times);
-  size_t size = waiting_times.size();
-  if (size == 0) {
-    acm_stat->meanWaitingTimeMs = -1;
-    acm_stat->medianWaitingTimeMs = -1;
-    acm_stat->minWaitingTimeMs = -1;
-    acm_stat->maxWaitingTimeMs = -1;
-  } else {
-    std::sort(waiting_times.begin(), waiting_times.end());
-    if ((size & 0x1) == 0) {
-      acm_stat->medianWaitingTimeMs = (waiting_times[size / 2 - 1] +
-          waiting_times[size / 2]) / 2;
-    } else {
-      acm_stat->medianWaitingTimeMs = waiting_times[size / 2];
-    }
-    acm_stat->minWaitingTimeMs = waiting_times.front();
-    acm_stat->maxWaitingTimeMs = waiting_times.back();
-    double sum = 0;
-    for (size_t i = 0; i < size; ++i) {
-      sum += waiting_times[i];
-    }
-    acm_stat->meanWaitingTimeMs = static_cast<int>(sum / size);
-  }
-}
-
-int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
-                                      CodecInst* codec) const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  auto it = decoders_.find(payload_type);
-  if (it == decoders_.end()) {
-    LOG_FERR1(LS_ERROR, "AcmReceiver::DecoderByPayloadType",
-              static_cast<int>(payload_type));
-    return -1;
-  }
-  const Decoder& decoder = it->second;
-  memcpy(codec, &ACMCodecDB::database_[decoder.acm_codec_id],
-         sizeof(CodecInst));
-  codec->pltype = decoder.payload_type;
-  codec->channels = decoder.channels;
-  return 0;
-}
-
-int AcmReceiver::EnableNack(size_t max_nack_list_size) {
-  // Don't do anything if |max_nack_list_size| is out of range.
-  if (max_nack_list_size == 0 || max_nack_list_size > Nack::kNackListSizeLimit)
-    return -1;
-
-  CriticalSectionScoped lock(crit_sect_.get());
-  if (!nack_enabled_) {
-    nack_.reset(Nack::Create(kNackThresholdPackets));
-    nack_enabled_ = true;
-
-    // Sampling rate might need to be updated if we change from disable to
-    // enable. Do it if the receive codec is valid.
-    if (last_audio_decoder_) {
-      nack_->UpdateSampleRate(
-          ACMCodecDB::database_[last_audio_decoder_->acm_codec_id].plfreq);
-    }
-  }
-  return nack_->SetMaxNackListSize(max_nack_list_size);
-}
-
-void AcmReceiver::DisableNack() {
-  CriticalSectionScoped lock(crit_sect_.get());
-  nack_.reset();  // Memory is released.
-  nack_enabled_ = false;
-}
-
-std::vector<uint16_t> AcmReceiver::GetNackList(
-    int64_t round_trip_time_ms) const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  if (round_trip_time_ms < 0) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
-                 "GetNackList: round trip time cannot be negative."
-                 " round_trip_time_ms=%" PRId64, round_trip_time_ms);
-  }
-  if (nack_enabled_ && round_trip_time_ms >= 0) {
-    assert(nack_.get());
-    return nack_->GetNackList(round_trip_time_ms);
-  }
-  std::vector<uint16_t> empty_list;
-  return empty_list;
-}
-
-void AcmReceiver::ResetInitialDelay() {
-  {
-    CriticalSectionScoped lock(crit_sect_.get());
-    av_sync_ = false;
-    initial_delay_manager_.reset(NULL);
-    missing_packets_sync_stream_.reset(NULL);
-    late_packets_sync_stream_.reset(NULL);
-  }
-  neteq_->SetMinimumDelay(0);
-  // TODO(turajs): Should NetEq Buffer be flushed?
-}
-
-// This function is called within critical section, no need to acquire a lock.
-bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
-  assert(av_sync_);
-  assert(initial_delay_manager_.get());
-  if (!initial_delay_manager_->buffering()) {
-    return false;
-  }
-
-  // We stop accumulating packets, if the number of packets or the total size
-  // exceeds a threshold.
-  int num_packets;
-  int max_num_packets;
-  const float kBufferingThresholdScale = 0.9f;
-  neteq_->PacketBufferStatistics(&num_packets, &max_num_packets);
-  if (num_packets > max_num_packets * kBufferingThresholdScale) {
-    initial_delay_manager_->DisableBuffering();
-    return false;
-  }
-
-  // Update statistics.
-  call_stats_.DecodedBySilenceGenerator();
-
-  // Set the values if already got a packet, otherwise set to default values.
-  if (last_audio_decoder_) {
-    current_sample_rate_hz_ =
-        ACMCodecDB::database_[last_audio_decoder_->acm_codec_id].plfreq;
-    frame->num_channels_ = last_audio_decoder_->channels;
-  } else {
-    frame->num_channels_ = 1;
-  }
-
-  // Set the audio frame's sampling frequency.
-  if (desired_sample_rate_hz > 0) {
-    frame->sample_rate_hz_ = desired_sample_rate_hz;
-  } else {
-    frame->sample_rate_hz_ = current_sample_rate_hz_;
-  }
-
-  frame->samples_per_channel_ = frame->sample_rate_hz_ / 100;  // Always 10 ms.
-  frame->speech_type_ = AudioFrame::kCNG;
-  frame->vad_activity_ = AudioFrame::kVadPassive;
-  int samples = frame->samples_per_channel_ * frame->num_channels_;
-  memset(frame->data_, 0, samples * sizeof(int16_t));
-  return true;
-}
-
-const AcmReceiver::Decoder* AcmReceiver::RtpHeaderToDecoder(
-    const RTPHeader& rtp_header,
-    const uint8_t* payload) const {
-  auto it = decoders_.find(rtp_header.payloadType);
-  if (ACMCodecDB::kRED >= 0 &&  // This ensures that RED is defined in WebRTC.
-      it != decoders_.end() && ACMCodecDB::kRED == it->second.acm_codec_id) {
-    // This is a RED packet, get the payload of the audio codec.
-    it = decoders_.find(payload[0] & 0x7F);
-  }
-
-  // Check if the payload is registered.
-  return it != decoders_.end() ? &it->second : nullptr;
-}
-
-uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
-  // Down-cast the time to (32-6)-bit since we only care about
-  // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
-  // We masked 6 most significant bits of 32-bit so there is no overflow in
-  // the conversion from milliseconds to timestamp.
-  const uint32_t now_in_ms = static_cast<uint32_t>(
-      clock_->TimeInMilliseconds() & 0x03ffffff);
-  return static_cast<uint32_t>(
-      (decoder_sampling_rate / 1000) * now_in_ms);
-}
-
-// This function only interacts with |neteq_|, therefore, it does not have to
-// be within critical section of AcmReceiver. It is inserting packets
-// into NetEq, so we call it when |decode_lock_| is acquired. However, this is
-// not essential as sync-packets do not interact with codecs (especially BWE).
-void AcmReceiver::InsertStreamOfSyncPackets(
-    InitialDelayManager::SyncStream* sync_stream) {
-  assert(sync_stream);
-  assert(av_sync_);
-  for (int n = 0; n < sync_stream->num_sync_packets; ++n) {
-    neteq_->InsertSyncPacket(sync_stream->rtp_info,
-                             sync_stream->receive_timestamp);
-    ++sync_stream->rtp_info.header.sequenceNumber;
-    sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step;
-    sync_stream->receive_timestamp += sync_stream->timestamp_step;
-  }
-}
-
-void AcmReceiver::GetDecodingCallStatistics(
-    AudioDecodingCallStats* stats) const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  *stats = call_stats_.GetDecodingStatistics();
-}
-
-}  // namespace acm2
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
deleted file mode 100644
index 5ec39c64c9..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ /dev/null
@@ -1,373 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
-
-#include <algorithm>  // std::min
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/test/test_suite.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-namespace acm2 {
-namespace {
-
-bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) {
-    if (strcmp(codec_a.plname, codec_b.plname) != 0 ||
-        codec_a.plfreq != codec_b.plfreq ||
-        codec_a.pltype != codec_b.pltype ||
-        codec_b.channels != codec_a.channels)
-      return false;
-    return true;
-}
-
-}  // namespace
-
-class AcmReceiverTest : public AudioPacketizationCallback,
-                        public ::testing::Test {
- protected:
-  AcmReceiverTest()
-      : timestamp_(0),
-        packet_sent_(false),
-        last_packet_send_timestamp_(timestamp_),
-        last_frame_type_(kFrameEmpty) {
-    AudioCoding::Config config;
-    config.transport = this;
-    acm_.reset(new AudioCodingImpl(config));
-    receiver_.reset(new AcmReceiver(config.ToOldConfig()));
-  }
-
-  ~AcmReceiverTest() {}
-
-  void SetUp() override {
-    ASSERT_TRUE(receiver_.get() != NULL);
-    ASSERT_TRUE(acm_.get() != NULL);
-    for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) {
-      ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n]));
-    }
-
-    rtp_header_.header.sequenceNumber = 0;
-    rtp_header_.header.timestamp = 0;
-    rtp_header_.header.markerBit = false;
-    rtp_header_.header.ssrc = 0x12345678;  // Arbitrary.
-    rtp_header_.header.numCSRCs = 0;
-    rtp_header_.header.payloadType = 0;
-    rtp_header_.frameType = kAudioFrameSpeech;
-    rtp_header_.type.Audio.isCNG = false;
-  }
-
-  void TearDown() override {}
-
-  void InsertOnePacketOfSilence(int codec_id) {
-    CodecInst codec;
-    ACMCodecDB::Codec(codec_id, &codec);
-    if (timestamp_ == 0) {  // This is the first time inserting audio.
-      ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype));
-    } else {
-      const CodecInst* current_codec = acm_->GetSenderCodecInst();
-      ASSERT_TRUE(current_codec);
-      if (!CodecsEqual(codec, *current_codec))
-        ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype));
-    }
-    AudioFrame frame;
-    // Frame setup according to the codec.
-    frame.sample_rate_hz_ = codec.plfreq;
-    frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
-    frame.num_channels_ = codec.channels;
-    memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ *
-           sizeof(int16_t));
-    int num_bytes = 0;
-    packet_sent_ = false;
-    last_packet_send_timestamp_ = timestamp_;
-    while (num_bytes == 0) {
-      frame.timestamp_ = timestamp_;
-      timestamp_ += frame.samples_per_channel_;
-      num_bytes = acm_->Add10MsAudio(frame);
-      ASSERT_GE(num_bytes, 0);
-    }
-    ASSERT_TRUE(packet_sent_);  // Sanity check.
-  }
-
-  // Last element of id should be negative.
-  void AddSetOfCodecs(const int* id) {
-    int n = 0;
-    while (id[n] >= 0) {
-      ASSERT_EQ(0, receiver_->AddCodec(id[n], codecs_[id[n]].pltype,
-                                       codecs_[id[n]].channels, NULL));
-      ++n;
-    }
-  }
-
-  int32_t SendData(FrameType frame_type,
-                   uint8_t payload_type,
-                   uint32_t timestamp,
-                   const uint8_t* payload_data,
-                   size_t payload_len_bytes,
-                   const RTPFragmentationHeader* fragmentation) override {
-    if (frame_type == kFrameEmpty)
-      return 0;
-
-    rtp_header_.header.payloadType = payload_type;
-    rtp_header_.frameType = frame_type;
-    if (frame_type == kAudioFrameSpeech)
-      rtp_header_.type.Audio.isCNG = false;
-    else
-      rtp_header_.type.Audio.isCNG = true;
-    rtp_header_.header.timestamp = timestamp;
-
-    int ret_val = receiver_->InsertPacket(rtp_header_, payload_data,
-                                          payload_len_bytes);
-    if (ret_val < 0) {
-      assert(false);
-      return -1;
-    }
-    rtp_header_.header.sequenceNumber++;
-    packet_sent_ = true;
-    last_frame_type_ = frame_type;
-    return 0;
-  }
-
-  rtc::scoped_ptr<AcmReceiver> receiver_;
-  CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
-  rtc::scoped_ptr<AudioCoding> acm_;
-  WebRtcRTPHeader rtp_header_;
-  uint32_t timestamp_;
-  bool packet_sent_;  // Set when SendData is called reset when inserting audio.
-  uint32_t last_packet_send_timestamp_;
-  FrameType last_frame_type_;
-};
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(AddCodecGetCodec)) {
-  // Add codec.
-  for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) {
-    if (n & 0x1)  // Just add codecs with odd index.
-      EXPECT_EQ(0, receiver_->AddCodec(n, codecs_[n].pltype,
-                                       codecs_[n].channels, NULL));
-  }
-  // Get codec and compare.
-  for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) {
-    CodecInst my_codec;
-    if (n & 0x1) {
-      // Codecs with odd index should match the reference.
-      EXPECT_EQ(0, receiver_->DecoderByPayloadType(codecs_[n].pltype,
-                                                   &my_codec));
-      EXPECT_TRUE(CodecsEqual(codecs_[n], my_codec));
-    } else {
-      // Codecs with even index are not registered.
-      EXPECT_EQ(-1, receiver_->DecoderByPayloadType(codecs_[n].pltype,
-                                                    &my_codec));
-    }
-  }
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) {
-  const int codec_id = ACMCodecDB::kPCMA;
-  CodecInst ref_codec1;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &ref_codec1));
-  CodecInst ref_codec2 = ref_codec1;
-  ++ref_codec2.pltype;
-  CodecInst test_codec;
-
-  // Register the same codec with different payload types.
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec1.pltype,
-                                   ref_codec1.channels, NULL));
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec2.pltype,
-                                   ref_codec2.channels, NULL));
-
-  // Both payload types should exist.
-  EXPECT_EQ(0, receiver_->DecoderByPayloadType(ref_codec1.pltype, &test_codec));
-  EXPECT_EQ(true, CodecsEqual(ref_codec1, test_codec));
-  EXPECT_EQ(0, receiver_->DecoderByPayloadType(ref_codec2.pltype, &test_codec));
-  EXPECT_EQ(true, CodecsEqual(ref_codec2, test_codec));
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(AddCodecChangeCodecId)) {
-  const int codec_id1 = ACMCodecDB::kPCMU;
-  CodecInst ref_codec1;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id1, &ref_codec1));
-  const int codec_id2 = ACMCodecDB::kPCMA;
-  CodecInst ref_codec2;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id2, &ref_codec2));
-  ref_codec2.pltype = ref_codec1.pltype;
-  CodecInst test_codec;
-
-  // Register the same payload type with different codec ID.
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id1, ref_codec1.pltype,
-                                   ref_codec1.channels, NULL));
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id2, ref_codec2.pltype,
-                                   ref_codec2.channels, NULL));
-
-  // Make sure that the last codec is used.
-  EXPECT_EQ(0, receiver_->DecoderByPayloadType(ref_codec2.pltype, &test_codec));
-  EXPECT_EQ(true, CodecsEqual(ref_codec2, test_codec));
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) {
-  CodecInst codec;
-  const int codec_id = ACMCodecDB::kPCMA;
-  EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &codec));
-  const int payload_type = codec.pltype;
-  EXPECT_EQ(0, receiver_->AddCodec(codec_id, codec.pltype,
-                                   codec.channels, NULL));
-
-  // Remove non-existing codec should not fail. ACM1 legacy.
-  EXPECT_EQ(0, receiver_->RemoveCodec(payload_type + 1));
-
-  // Remove an existing codec.
-  EXPECT_EQ(0, receiver_->RemoveCodec(payload_type));
-
-  // Ask for the removed codec, must fail.
-  EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &codec));
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(SampleRate)) {
-  const int kCodecId[] = {
-      ACMCodecDB::kISAC, ACMCodecDB::kISACSWB, ACMCodecDB::kISACFB,
-      -1  // Terminator.
-  };
-  AddSetOfCodecs(kCodecId);
-
-  AudioFrame frame;
-  const int kOutSampleRateHz = 8000;  // Different than codec sample rate.
-  int n = 0;
-  while (kCodecId[n] >= 0) {
-    const int num_10ms_frames = codecs_[kCodecId[n]].pacsize /
-        (codecs_[kCodecId[n]].plfreq / 100);
-    InsertOnePacketOfSilence(kCodecId[n]);
-    for (int k = 0; k < num_10ms_frames; ++k) {
-      EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
-    }
-    EXPECT_EQ(std::min(32000, codecs_[kCodecId[n]].plfreq),
-              receiver_->current_sample_rate_hz());
-    ++n;
-  }
-}
-
-// Verify that the playout mode is set correctly.
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PlayoutMode)) {
-  receiver_->SetPlayoutMode(voice);
-  EXPECT_EQ(voice, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(streaming);
-  EXPECT_EQ(streaming, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(fax);
-  EXPECT_EQ(fax, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(off);
-  EXPECT_EQ(off, receiver_->PlayoutMode());
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PostdecodingVad)) {
-  receiver_->EnableVad();
-  EXPECT_TRUE(receiver_->vad_enabled());
-
-  const int id = ACMCodecDB::kPCM16Bwb;
-  ASSERT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels,
-                                   NULL));
-  const int kNumPackets = 5;
-  const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100);
-  AudioFrame frame;
-  for (int n = 0; n < kNumPackets; ++n) {
-    InsertOnePacketOfSilence(id);
-    for (int k = 0; k < num_10ms_frames; ++k)
-      ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
-  }
-  EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
-
-  receiver_->DisableVad();
-  EXPECT_FALSE(receiver_->vad_enabled());
-
-  for (int n = 0; n < kNumPackets; ++n) {
-    InsertOnePacketOfSilence(id);
-    for (int k = 0; k < num_10ms_frames; ++k)
-      ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
-  }
-  EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) {
-  const int kCodecId[] = {
-      ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB,
-      ACMCodecDB::kPCM16Bswb32kHz,
-      -1  // Terminator.
-  };
-  AddSetOfCodecs(kCodecId);
-
-  const int kCngId[] = {  // Not including full-band.
-      ACMCodecDB::kCNNB, ACMCodecDB::kCNWB, ACMCodecDB::kCNSWB,
-      -1  // Terminator.
-  };
-  AddSetOfCodecs(kCngId);
-
-  // Register CNG at sender side.
-  int n = 0;
-  while (kCngId[n] > 0) {
-    ASSERT_TRUE(acm_->RegisterSendCodec(kCngId[n], codecs_[kCngId[n]].pltype));
-    ++n;
-  }
-
-  CodecInst codec;
-  // No audio payload is received.
-  EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
-
-  // Start with sending DTX.
-  ASSERT_TRUE(acm_->SetVad(true, true, VADVeryAggr));
-  packet_sent_ = false;
-  InsertOnePacketOfSilence(kCodecId[0]);  // Enough to test with one codec.
-  ASSERT_TRUE(packet_sent_);
-  EXPECT_EQ(kAudioFrameCN, last_frame_type_);
-
-  // Has received, only, DTX. Last Audio codec is undefined.
-  EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
-  EXPECT_EQ(-1, receiver_->last_audio_codec_id());
-
-  n = 0;
-  while (kCodecId[n] >= 0) {  // Loop over codecs.
-    // Set DTX off to send audio payload.
-    acm_->SetVad(false, false, VADAggr);
-    packet_sent_ = false;
-    InsertOnePacketOfSilence(kCodecId[n]);
-
-    // Sanity check if Actually an audio payload received, and it should be
-    // of type "speech."
-    ASSERT_TRUE(packet_sent_);
-    ASSERT_EQ(kAudioFrameSpeech, last_frame_type_);
-    EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
-
-    // Set VAD on to send DTX. Then check if the "Last Audio codec" returns
-    // the expected codec.
-    acm_->SetVad(true, true, VADAggr);
-
-    // Do as many encoding until a DTX is sent.
-    while (last_frame_type_ != kAudioFrameCN) {
-      packet_sent_ = false;
-      InsertOnePacketOfSilence(kCodecId[n]);
-      ASSERT_TRUE(packet_sent_);
-    }
-    EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
-    EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
-    EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec));
-    ++n;
-  }
-}
-
-}  // namespace acm2
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
deleted file mode 100644
index 56830a4ea6..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <string.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-
-namespace webrtc {
-namespace test {
-
-AcmSendTest::AcmSendTest(InputAudioFile* audio_source,
-                         int source_rate_hz,
-                         int test_duration_ms)
-    : clock_(0),
-      audio_source_(audio_source),
-      source_rate_hz_(source_rate_hz),
-      input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
-      codec_registered_(false),
-      test_duration_ms_(test_duration_ms),
-      frame_type_(kAudioFrameSpeech),
-      payload_type_(0),
-      timestamp_(0),
-      sequence_number_(0) {
-  webrtc::AudioCoding::Config config;
-  config.clock = &clock_;
-  config.transport = this;
-  acm_.reset(webrtc::AudioCoding::Create(config));
-  input_frame_.sample_rate_hz_ = source_rate_hz_;
-  input_frame_.num_channels_ = 1;
-  input_frame_.samples_per_channel_ = input_block_size_samples_;
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
-}
-
-bool AcmSendTest::RegisterCodec(int codec_type,
-                                int channels,
-                                int payload_type,
-                                int frame_size_samples) {
-  codec_registered_ =
-      acm_->RegisterSendCodec(codec_type, payload_type, frame_size_samples);
-  input_frame_.num_channels_ = channels;
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
-  return codec_registered_;
-}
-
-Packet* AcmSendTest::NextPacket() {
-  assert(codec_registered_);
-  if (filter_.test(static_cast<size_t>(payload_type_))) {
-    // This payload type should be filtered out. Since the payload type is the
-    // same throughout the whole test run, no packet at all will be delivered.
-    // We can just as well signal that the test is over by returning NULL.
-    return NULL;
-  }
-  // Insert audio and process until one packet is produced.
-  while (clock_.TimeInMilliseconds() < test_duration_ms_) {
-    clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
-    CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_));
-    if (input_frame_.num_channels_ > 1) {
-      InputAudioFile::DuplicateInterleaved(input_frame_.data_,
-                                           input_block_size_samples_,
-                                           input_frame_.num_channels_,
-                                           input_frame_.data_);
-    }
-    int32_t encoded_bytes = acm_->Add10MsAudio(input_frame_);
-    EXPECT_GE(encoded_bytes, 0);
-    input_frame_.timestamp_ += input_block_size_samples_;
-    if (encoded_bytes > 0) {
-      // Encoded packet received.
-      return CreatePacket();
-    }
-  }
-  // Test ended.
-  return NULL;
-}
-
-// This method receives the callback from ACM when a new packet is produced.
-int32_t AcmSendTest::SendData(FrameType frame_type,
-                              uint8_t payload_type,
-                              uint32_t timestamp,
-                              const uint8_t* payload_data,
-                              size_t payload_len_bytes,
-                              const RTPFragmentationHeader* fragmentation) {
-  // Store the packet locally.
-  frame_type_ = frame_type;
-  payload_type_ = payload_type;
-  timestamp_ = timestamp;
-  last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
-  assert(last_payload_vec_.size() == payload_len_bytes);
-  return 0;
-}
-
-Packet* AcmSendTest::CreatePacket() {
-  const size_t kRtpHeaderSize = 12;
-  size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize;
-  uint8_t* packet_memory = new uint8_t[allocated_bytes];
-  // Populate the header bytes.
-  packet_memory[0] = 0x80;
-  packet_memory[1] = static_cast<uint8_t>(payload_type_);
-  packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
-  packet_memory[3] = (sequence_number_) & 0xFF;
-  packet_memory[4] = (timestamp_ >> 24) & 0xFF;
-  packet_memory[5] = (timestamp_ >> 16) & 0xFF;
-  packet_memory[6] = (timestamp_ >> 8) & 0xFF;
-  packet_memory[7] = timestamp_ & 0xFF;
-  // Set SSRC to 0x12345678.
-  packet_memory[8] = 0x12;
-  packet_memory[9] = 0x34;
-  packet_memory[10] = 0x56;
-  packet_memory[11] = 0x78;
-
-  ++sequence_number_;
-
-  // Copy the payload data.
-  memcpy(packet_memory + kRtpHeaderSize,
-         &last_payload_vec_[0],
-         last_payload_vec_.size());
-  Packet* packet =
-      new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
-  assert(packet);
-  assert(packet->valid_header());
-  return packet;
-}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
deleted file mode 100644
index 4c4db5bd13..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
-
-#include <vector>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-
-namespace webrtc {
-
-namespace test {
-class InputAudioFile;
-class Packet;
-
-class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
- public:
-  AcmSendTest(InputAudioFile* audio_source,
-              int source_rate_hz,
-              int test_duration_ms);
-  virtual ~AcmSendTest() {}
-
-  // Registers the send codec. Returns true on success, false otherwise.
-  bool RegisterCodec(int codec_type,
-                     int channels,
-                     int payload_type,
-                     int frame_size_samples);
-
-  // Returns the next encoded packet. Returns NULL if the test duration was
-  // exceeded. Ownership of the packet is handed over to the caller.
-  // Inherited from PacketSource.
-  Packet* NextPacket() override;
-
-  // Inherited from AudioPacketizationCallback.
-  int32_t SendData(FrameType frame_type,
-                   uint8_t payload_type,
-                   uint32_t timestamp,
-                   const uint8_t* payload_data,
-                   size_t payload_len_bytes,
-                   const RTPFragmentationHeader* fragmentation) override;
-
- private:
-  static const int kBlockSizeMs = 10;
-
-  // Creates a Packet object from the last packet produced by ACM (and received
-  // through the SendData method as a callback). Ownership of the new Packet
-  // object is transferred to the caller.
-  Packet* CreatePacket();
-
-  SimulatedClock clock_;
-  rtc::scoped_ptr<AudioCoding> acm_;
-  InputAudioFile* audio_source_;
-  int source_rate_hz_;
-  const int input_block_size_samples_;
-  AudioFrame input_frame_;
-  bool codec_registered_;
-  int test_duration_ms_;
-  // The following member variables are set whenever SendData() is called.
-  FrameType frame_type_;
-  int payload_type_;
-  uint32_t timestamp_;
-  uint16_t sequence_number_;
-  std::vector<uint8_t> last_payload_vec_;
-
-  DISALLOW_COPY_AND_ASSIGN(AcmSendTest);
-};
-
-}  // namespace test
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
deleted file mode 100644
index 9b61d33480..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-// Create module
-AudioCodingModule* AudioCodingModule::Create(int id) {
-  return Create(id, Clock::GetRealTimeClock());
-}
-
-AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
-  AudioCodingModule::Config config;
-  config.id = id;
-  config.clock = clock;
-  return new acm2::AudioCodingModuleImpl(config);
-}
-
-// Get number of supported codecs
-int AudioCodingModule::NumberOfCodecs() {
-  return acm2::ACMCodecDB::kNumCodecs;
-}
-
-// Get supported codec parameters with id
-int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
-  // Get the codec settings for the codec with the given list ID
-  return acm2::ACMCodecDB::Codec(list_id, codec);
-}
-
-// Get supported codec parameters with name, frequency and number of channels.
-int AudioCodingModule::Codec(const char* payload_name,
-                             CodecInst* codec,
-                             int sampling_freq_hz,
-                             int channels) {
-  int codec_id;
-
-  // Get the id of the codec from the database.
-  codec_id = acm2::ACMCodecDB::CodecId(
-      payload_name, sampling_freq_hz, channels);
-  if (codec_id < 0) {
-    // We couldn't find a matching codec, set the parameters to unacceptable
-    // values and return.
-    codec->plname[0] = '\0';
-    codec->pltype = -1;
-    codec->pacsize = 0;
-    codec->rate = 0;
-    codec->plfreq = 0;
-    return -1;
-  }
-
-  // Get default codec settings.
-  acm2::ACMCodecDB::Codec(codec_id, codec);
-
-  // Keep the number of channels from the function call. For most codecs it
-  // will be the same value as in default codec settings, but not for all.
-  codec->channels = channels;
-
-  return 0;
-}
-
-// Get supported codec Index with name, frequency and number of channels.
-int AudioCodingModule::Codec(const char* payload_name,
-                             int sampling_freq_hz,
-                             int channels) {
-  return acm2::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
-}
-
-// Checks the validity of the parameters of the given codec
-bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
-  int mirror_id;
-
-  int codec_number = acm2::ACMCodecDB::CodecNumber(codec, &mirror_id);
-
-  if (codec_number < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
-                 "Invalid codec setting");
-    return false;
-  } else {
-    return true;
-  }
-}
-
-AudioCoding* AudioCoding::Create(const Config& config) {
-  return new AudioCodingImpl(config);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
deleted file mode 100644
index b4b4873719..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ /dev/null
@@ -1,177 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'variables': {
-    'audio_coding_dependencies': [
-      'CNG',
-      'red',
-      '<(webrtc_root)/common.gyp:webrtc_common',
-      '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-    ],
-    'audio_coding_defines': [],
-    'conditions': [
-      ['include_opus==1', {
-        'audio_coding_dependencies': ['webrtc_opus',],
-        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
-      }],
-      ['include_g711==1', {
-        'audio_coding_dependencies': ['G711',],
-        'audio_coding_defines': ['WEBRTC_CODEC_G711',],
-      }],
-      ['include_g722==1', {
-        'audio_coding_dependencies': ['G722',],
-        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
-      }],
-      ['include_ilbc==1', {
-        'audio_coding_dependencies': ['iLBC',],
-        'audio_coding_defines': ['WEBRTC_CODEC_ILBC',],
-      }],
-      ['include_isac==1', {
-        'audio_coding_dependencies': ['iSAC', 'iSACFix',],
-#        'audio_coding_defines': ['WEBRTC_CODEC_ISAC', 'WEBRTC_CODEC_ISACFX',],
-      }],
-      ['include_pcm16b==1', {
-        'audio_coding_dependencies': ['PCM16B',],
-        'audio_coding_defines': ['WEBRTC_CODEC_PCM16',],
-      }],
-    ],
-  },
-  'targets': [
-    {
-      'target_name': 'audio_coding_module',
-      'type': 'static_library',
-      'defines': [
-        '<@(audio_coding_defines)',
-      ],
-      'dependencies': [
-        '<@(audio_coding_dependencies)',
-        '<(webrtc_root)/common.gyp:webrtc_common',
-        'neteq',
-      ],
-      'include_dirs': [
-        '../interface',
-        '../../../interface',
-        '<(webrtc_root)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-          '../../../interface',
-          '<(webrtc_root)',
-        ],
-      },
-      'sources': [
-        '../interface/audio_coding_module.h',
-        '../interface/audio_coding_module_typedefs.h',
-        'acm_codec_database.cc',
-        'acm_codec_database.h',
-        'acm_common_defs.h',
-        'acm_generic_codec.cc',
-        'acm_generic_codec.h',
-        'acm_receiver.cc',
-        'acm_receiver.h',
-        'acm_resampler.cc',
-        'acm_resampler.h',
-        'audio_coding_module.cc',
-        'audio_coding_module_impl.cc',
-        'audio_coding_module_impl.h',
-        'call_statistics.cc',
-        'call_statistics.h',
-        'codec_manager.cc',
-        'codec_manager.h',
-        'initial_delay_manager.cc',
-        'initial_delay_manager.h',
-        'nack.cc',
-        'nack.h',
-      ],
-    },
-  ],
-  'conditions': [
-    ['include_tests==1', {
-      'targets': [
-        {
-          'target_name': 'acm_receive_test',
-          'type': 'static_library',
-          'defines': [
-            '<@(audio_coding_defines)',
-          ],
-          'dependencies': [
-            '<@(audio_coding_dependencies)',
-            'audio_coding_module',
-            'neteq_unittest_tools',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-          ],
-          'sources': [
-            'acm_receive_test.cc',
-            'acm_receive_test.h',
-            'acm_receive_test_oldapi.cc',
-            'acm_receive_test_oldapi.h',
-          ],
-        }, # acm_receive_test
-        {
-          'target_name': 'acm_send_test',
-          'type': 'static_library',
-          'defines': [
-            '<@(audio_coding_defines)',
-          ],
-          'dependencies': [
-            '<@(audio_coding_dependencies)',
-            'audio_coding_module',
-            'neteq_unittest_tools',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-          ],
-          'sources': [
-            'acm_send_test.cc',
-            'acm_send_test.h',
-            'acm_send_test_oldapi.cc',
-            'acm_send_test_oldapi.h',
-          ],
-        }, # acm_send_test
-        {
-          'target_name': 'delay_test',
-          'type': 'executable',
-          'dependencies': [
-            'audio_coding_module',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/test/test.gyp:test_support',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-          ],
-          'sources': [
-             '../test/delay_test.cc',
-             '../test/Channel.cc',
-             '../test/PCMFile.cc',
-             '../test/utility.cc',
-           ],
-        }, # delay_test
-        {
-          'target_name': 'insert_packet_with_timing',
-          'type': 'executable',
-          'dependencies': [
-            'audio_coding_module',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/test/test.gyp:test_support',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-          ],
-          'sources': [
-             '../test/insert_packet_with_timing.cc',
-             '../test/Channel.cc',
-             '../test/PCMFile.cc',
-           ],
-        }, # delay_test
-      ],
-    }],
-  ],
-}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
deleted file mode 100644
index 0fac8acf4c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ /dev/null
@@ -1,1211 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-
-#include <assert.h>
-#include <stdlib.h>
-#include <vector>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/safe_conversions.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-enum {
-  kACMToneEnd = 999
-};
-
-// Maximum number of bytes in one packet (PCM16B, 20 ms packets, stereo).
-enum {
-  kMaxPacketSize = 2560
-};
-
-// Maximum number of payloads that can be packed in one RED packet. For
-// regular RED, we only pack two payloads. In case of dual-streaming, in worst
-// case we might pack 3 payloads in one RED packet.
-enum {
-  kNumRedFragmentationVectors = 2,
-  kMaxNumFragmentationVectors = 3
-};
-
-// If packet N is arrived all packets prior to N - |kNackThresholdPackets| which
-// are not received are considered as lost, and appear in NACK list.
-enum {
-  kNackThresholdPackets = 2
-};
-
-namespace {
-
-// TODO(turajs): the same functionality is used in NetEq. If both classes
-// need them, make it a static function in ACMCodecDB.
-bool IsCodecRED(const CodecInst* codec) {
-  return (STR_CASE_CMP(codec->plname, "RED") == 0);
-}
-
-bool IsCodecRED(int index) {
-  return (IsCodecRED(&ACMCodecDB::database_[index]));
-}
-
-bool IsCodecCN(const CodecInst* codec) {
-  return (STR_CASE_CMP(codec->plname, "CN") == 0);
-}
-
-bool IsCodecCN(int index) {
-  return (IsCodecCN(&ACMCodecDB::database_[index]));
-}
-
-// Stereo-to-mono can be used as in-place.
-int DownMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
-  if (length_out_buff < frame.samples_per_channel_) {
-    return -1;
-  }
-  for (int n = 0; n < frame.samples_per_channel_; ++n)
-    out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
-  return 0;
-}
-
-// Mono-to-stereo can be used as in-place.
-int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
-  if (length_out_buff < frame.samples_per_channel_) {
-    return -1;
-  }
-  for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) {
-    out_buff[2 * n + 1] = frame.data_[n];
-    out_buff[2 * n] = frame.data_[n];
-  }
-  return 0;
-}
-
-void ConvertEncodedInfoToFragmentationHeader(
-    const AudioEncoder::EncodedInfo& info,
-    RTPFragmentationHeader* frag) {
-  if (info.redundant.empty()) {
-    frag->fragmentationVectorSize = 0;
-    return;
-  }
-
-  frag->VerifyAndAllocateFragmentationHeader(
-      static_cast<uint16_t>(info.redundant.size()));
-  frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size());
-  size_t offset = 0;
-  for (size_t i = 0; i < info.redundant.size(); ++i) {
-    frag->fragmentationOffset[i] = offset;
-    offset += info.redundant[i].encoded_bytes;
-    frag->fragmentationLength[i] = info.redundant[i].encoded_bytes;
-    frag->fragmentationTimeDiff[i] = rtc::checked_cast<uint16_t>(
-        info.encoded_timestamp - info.redundant[i].encoded_timestamp);
-    frag->fragmentationPlType[i] = info.redundant[i].payload_type;
-  }
-}
-}  // namespace
-
-AudioCodingModuleImpl::AudioCodingModuleImpl(
-    const AudioCodingModule::Config& config)
-    : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      id_(config.id),
-      expected_codec_ts_(0xD87F3F9F),
-      expected_in_ts_(0xD87F3F9F),
-      receiver_(config),
-      codec_manager_(this),
-      previous_pltype_(255),
-      aux_rtp_header_(NULL),
-      receiver_initialized_(false),
-      first_10ms_data_(false),
-      first_frame_(true),
-      callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      packetization_callback_(NULL),
-      vad_callback_(NULL) {
-  if (InitializeReceiverSafe() < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Cannot initialize receiver");
-  }
-  WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
-}
-
-AudioCodingModuleImpl::~AudioCodingModuleImpl() {
-  if (aux_rtp_header_ != NULL) {
-    delete aux_rtp_header_;
-    aux_rtp_header_ = NULL;
-  }
-
-  delete callback_crit_sect_;
-  callback_crit_sect_ = NULL;
-
-  delete acm_crit_sect_;
-  acm_crit_sect_ = NULL;
-  WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_,
-               "Destroyed");
-}
-
-int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
-  uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE];  // Make room for 1 RED payload.
-  AudioEncoder::EncodedInfo encoded_info;
-  uint8_t previous_pltype;
-
-  // Keep the scope of the ACM critical section limited.
-  {
-    CriticalSectionScoped lock(acm_crit_sect_);
-    // Check if there is an encoder before.
-    if (!HaveValidEncoder("Process")) {
-      return -1;
-    }
-
-    AudioEncoder* audio_encoder =
-        codec_manager_.current_encoder()->GetAudioEncoder();
-    // Scale the timestamp to the codec's RTP timestamp rate.
-    uint32_t rtp_timestamp =
-        first_frame_ ? input_data.input_timestamp
-                     : last_rtp_timestamp_ +
-                           rtc::CheckedDivExact(
-                               input_data.input_timestamp - last_timestamp_,
-                               static_cast<uint32_t>(rtc::CheckedDivExact(
-                                   audio_encoder->SampleRateHz(),
-                                   audio_encoder->RtpTimestampRateHz())));
-    last_timestamp_ = input_data.input_timestamp;
-    last_rtp_timestamp_ = rtp_timestamp;
-    first_frame_ = false;
-
-    encoded_info = audio_encoder->Encode(rtp_timestamp, input_data.audio,
-                                         input_data.length_per_channel,
-                                         sizeof(stream), stream);
-    if (encoded_info.encoded_bytes == 0 && !encoded_info.send_even_if_empty) {
-      // Not enough data.
-      return 0;
-    }
-    previous_pltype = previous_pltype_;  // Read it while we have the critsect.
-  }
-
-  RTPFragmentationHeader my_fragmentation;
-  ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
-  FrameType frame_type;
-  if (encoded_info.encoded_bytes == 0 && encoded_info.send_even_if_empty) {
-    frame_type = kFrameEmpty;
-    encoded_info.payload_type = previous_pltype;
-  } else {
-    DCHECK_GT(encoded_info.encoded_bytes, 0u);
-    frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
-  }
-
-  {
-    CriticalSectionScoped lock(callback_crit_sect_);
-    if (packetization_callback_) {
-      packetization_callback_->SendData(
-          frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
-          stream, encoded_info.encoded_bytes,
-          my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
-                                                       : nullptr);
-    }
-
-    if (vad_callback_) {
-      // Callback with VAD decision.
-      vad_callback_->InFrameType(frame_type);
-    }
-  }
-  {
-    CriticalSectionScoped lock(acm_crit_sect_);
-    previous_pltype_ = encoded_info.payload_type;
-  }
-  return static_cast<int32_t>(encoded_info.encoded_bytes);
-}
-
-/////////////////////////////////////////
-//   Sender
-//
-
-// TODO(henrik.lundin): Remove this method; only used in tests.
-int AudioCodingModuleImpl::ResetEncoder() {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (!HaveValidEncoder("ResetEncoder")) {
-    return -1;
-  }
-  return 0;
-}
-
-// Can be called multiple times for Codec, CNG, RED.
-int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return codec_manager_.RegisterSendCodec(send_codec);
-}
-
-// Get current send codec.
-int AudioCodingModuleImpl::SendCodec(CodecInst* current_codec) const {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return codec_manager_.SendCodec(current_codec);
-}
-
-// Get current send frequency.
-int AudioCodingModuleImpl::SendFrequency() const {
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-               "SendFrequency()");
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  if (!codec_manager_.current_encoder()) {
-    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-                 "SendFrequency Failed, no codec is registered");
-    return -1;
-  }
-
-  return codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz();
-}
-
-// Get encode bitrate.
-// Adaptive rate codecs return their current encode target rate, while other
-// codecs return there longterm avarage or their fixed rate.
-// TODO(henrik.lundin): Remove; not used.
-int AudioCodingModuleImpl::SendBitrate() const {
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  if (!codec_manager_.current_encoder()) {
-    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-                 "SendBitrate Failed, no codec is registered");
-    return -1;
-  }
-
-  WebRtcACMCodecParams encoder_param;
-  codec_manager_.current_encoder()->EncoderParams(&encoder_param);
-
-  return encoder_param.codec_inst.rate;
-}
-
-// Set available bandwidth, inform the encoder about the estimated bandwidth
-// received from the remote party.
-// TODO(henrik.lundin): Remove; not used.
-int AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(int bw) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  FATAL() << "Dead code?";
-  return -1;
-//  return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw);
-}
-
-// Register a transport callback which will be called to deliver
-// the encoded buffers.
-int AudioCodingModuleImpl::RegisterTransportCallback(
-    AudioPacketizationCallback* transport) {
-  CriticalSectionScoped lock(callback_crit_sect_);
-  packetization_callback_ = transport;
-  return 0;
-}
-
-// Add 10MS of raw (PCM) audio data to the encoder.
-int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
-  InputData input_data;
-  int r = Add10MsDataInternal(audio_frame, &input_data);
-  return r < 0 ? r : Encode(input_data);
-}
-
-int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
-                                               InputData* input_data) {
-  if (audio_frame.samples_per_channel_ <= 0) {
-    assert(false);
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Cannot Add 10 ms audio, payload length is negative or "
-                 "zero");
-    return -1;
-  }
-
-  if (audio_frame.sample_rate_hz_ > 48000) {
-    assert(false);
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Cannot Add 10 ms audio, input frequency not valid");
-    return -1;
-  }
-
-  // If the length and frequency matches. We currently just support raw PCM.
-  if ((audio_frame.sample_rate_hz_ / 100)
-      != audio_frame.samples_per_channel_) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Cannot Add 10 ms audio, input frequency and length doesn't"
-                 " match");
-    return -1;
-  }
-
-  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Cannot Add 10 ms audio, invalid number of channels.");
-    return -1;
-  }
-
-  CriticalSectionScoped lock(acm_crit_sect_);
-  // Do we have a codec registered?
-  if (!HaveValidEncoder("Add10MsData")) {
-    return -1;
-  }
-
-  const AudioFrame* ptr_frame;
-  // Perform a resampling, also down-mix if it is required and can be
-  // performed before resampling (a down mix prior to resampling will take
-  // place if both primary and secondary encoders are mono and input is in
-  // stereo).
-  if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
-    return -1;
-  }
-
-  // Check whether we need an up-mix or down-mix?
-  bool remix =
-      ptr_frame->num_channels_ !=
-      codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels();
-
-  if (remix) {
-    if (ptr_frame->num_channels_ == 1) {
-      if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
-        return -1;
-    } else {
-      if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
-        return -1;
-    }
-  }
-
-  // When adding data to encoders this pointer is pointing to an audio buffer
-  // with correct number of channels.
-  const int16_t* ptr_audio = ptr_frame->data_;
-
-  // For pushing data to primary, point the |ptr_audio| to correct buffer.
-  if (codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels() !=
-      ptr_frame->num_channels_)
-    ptr_audio = input_data->buffer;
-
-  input_data->input_timestamp = ptr_frame->timestamp_;
-  input_data->audio = ptr_audio;
-  input_data->length_per_channel = ptr_frame->samples_per_channel_;
-  input_data->audio_channel =
-      codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels();
-
-  return 0;
-}
-
-// Perform a resampling and down-mix if required. We down-mix only if
-// encoder is mono and input is stereo. In case of dual-streaming, both
-// encoders has to be mono for down-mix to take place.
-// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
-// is required, |*ptr_out| points to |in_frame|.
-int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
-                                               const AudioFrame** ptr_out) {
-  bool resample =
-      (in_frame.sample_rate_hz_ !=
-       codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz());
-
-  // This variable is true if primary codec and secondary codec (if exists)
-  // are both mono and input is stereo.
-  bool down_mix =
-      (in_frame.num_channels_ == 2) &&
-      (codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels() == 1);
-
-  if (!first_10ms_data_) {
-    expected_in_ts_ = in_frame.timestamp_;
-    expected_codec_ts_ = in_frame.timestamp_;
-    first_10ms_data_ = true;
-  } else if (in_frame.timestamp_ != expected_in_ts_) {
-    // TODO(turajs): Do we need a warning here.
-    expected_codec_ts_ +=
-        (in_frame.timestamp_ - expected_in_ts_) *
-        static_cast<uint32_t>(
-            (static_cast<double>(codec_manager_.current_encoder()
-                                     ->GetAudioEncoder()
-                                     ->SampleRateHz()) /
-             static_cast<double>(in_frame.sample_rate_hz_)));
-    expected_in_ts_ = in_frame.timestamp_;
-  }
-
-
-  if (!down_mix && !resample) {
-    // No pre-processing is required.
-    expected_in_ts_ += in_frame.samples_per_channel_;
-    expected_codec_ts_ += in_frame.samples_per_channel_;
-    *ptr_out = &in_frame;
-    return 0;
-  }
-
-  *ptr_out = &preprocess_frame_;
-  preprocess_frame_.num_channels_ = in_frame.num_channels_;
-  int16_t audio[WEBRTC_10MS_PCM_AUDIO];
-  const int16_t* src_ptr_audio = in_frame.data_;
-  int16_t* dest_ptr_audio = preprocess_frame_.data_;
-  if (down_mix) {
-    // If a resampling is required the output of a down-mix is written into a
-    // local buffer, otherwise, it will be written to the output frame.
-    if (resample)
-      dest_ptr_audio = audio;
-    if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
-      return -1;
-    preprocess_frame_.num_channels_ = 1;
-    // Set the input of the resampler is the down-mixed signal.
-    src_ptr_audio = audio;
-  }
-
-  preprocess_frame_.timestamp_ = expected_codec_ts_;
-  preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
-  preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
-  // If it is required, we have to do a resampling.
-  if (resample) {
-    // The result of the resampler is written to output frame.
-    dest_ptr_audio = preprocess_frame_.data_;
-
-    preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec(
-        src_ptr_audio, in_frame.sample_rate_hz_,
-        codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz(),
-        preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
-        dest_ptr_audio);
-
-    if (preprocess_frame_.samples_per_channel_ < 0) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                   "Cannot add 10 ms audio, resampling failed");
-      return -1;
-    }
-    preprocess_frame_.sample_rate_hz_ =
-        codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz();
-  }
-
-  expected_codec_ts_ += preprocess_frame_.samples_per_channel_;
-  expected_in_ts_ += in_frame.samples_per_channel_;
-
-  return 0;
-}
-
-/////////////////////////////////////////
-//   (RED) Redundant Coding
-//
-
-bool AudioCodingModuleImpl::REDStatus() const {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return codec_manager_.red_enabled();
-}
-
-// Configure RED status i.e on/off.
-int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-#ifdef WEBRTC_CODEC_RED
-  return codec_manager_.SetCopyRed(enable_red) ? 0 : -1;
-#else
-  WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
-               "  WEBRTC_CODEC_RED is undefined");
-  return -1;
-#endif
-}
-
-/////////////////////////////////////////
-//   (FEC) Forward Error Correction (codec internal)
-//
-
-bool AudioCodingModuleImpl::CodecFEC() const {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return codec_manager_.codec_fec_enabled();
-}
-
-int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return codec_manager_.SetCodecFEC(enable_codec_fec);
-}
-
-int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (HaveValidEncoder("SetPacketLossRate") &&
-      codec_manager_.current_encoder()->SetPacketLossRate(loss_rate) < 0) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                   "Set packet loss rate failed.");
-    return -1;
-  }
-  return 0;
-}
-
-/////////////////////////////////////////
-//   (VAD) Voice Activity Detection
-//
-int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
-                                  bool enable_vad,
-                                  ACMVADMode mode) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return codec_manager_.SetVAD(enable_dtx, enable_vad, mode);
-}
-
-// Get VAD/DTX settings.
-int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
-                               ACMVADMode* mode) const {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  codec_manager_.VAD(dtx_enabled, vad_enabled, mode);
-  return 0;
-}
-
-/////////////////////////////////////////
-//   Receiver
-//
-
-int AudioCodingModuleImpl::InitializeReceiver() {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return InitializeReceiverSafe();
-}
-
-// Initialize receiver, resets codec database etc.
-int AudioCodingModuleImpl::InitializeReceiverSafe() {
-  // If the receiver is already initialized then we want to destroy any
-  // existing decoders. After a call to this function, we should have a clean
-  // start-up.
-  if (receiver_initialized_) {
-    if (receiver_.RemoveAllCodecs() < 0)
-      return -1;
-  }
-  receiver_.set_id(id_);
-  receiver_.ResetInitialDelay();
-  receiver_.SetMinimumDelay(0);
-  receiver_.SetMaximumDelay(0);
-  receiver_.FlushBuffers();
-
-  // Register RED and CN.
-  for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) {
-    if (IsCodecRED(i) || IsCodecCN(i)) {
-      uint8_t pl_type = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
-      if (receiver_.AddCodec(i, pl_type, 1, NULL) < 0) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                     "Cannot register master codec.");
-        return -1;
-      }
-    }
-  }
-  receiver_initialized_ = true;
-  return 0;
-}
-
-// TODO(turajs): If NetEq opens an API for reseting the state of decoders then
-// implement this method. Otherwise it should be removed. I might be that by
-// removing and registering a decoder we can achieve the effect of resetting.
-// Reset the decoder state.
-// TODO(henrik.lundin): Remove; only used in one test, and does nothing.
-int AudioCodingModuleImpl::ResetDecoder() {
-  return 0;
-}
-
-// Get current receive frequency.
-int AudioCodingModuleImpl::ReceiveFrequency() const {
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-               "ReceiveFrequency()");
-
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  int codec_id = receiver_.last_audio_codec_id();
-
-  return codec_id < 0 ? receiver_.current_sample_rate_hz() :
-                        ACMCodecDB::database_[codec_id].plfreq;
-}
-
-// Get current playout frequency.
-int AudioCodingModuleImpl::PlayoutFrequency() const {
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-               "PlayoutFrequency()");
-
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  return receiver_.current_sample_rate_hz();
-}
-
-// Register possible receive codecs, can be called multiple times,
-// for codecs, CNG (NB, WB and SWB), DTMF, RED.
-int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  DCHECK(receiver_initialized_);
-  return codec_manager_.RegisterReceiveCodec(codec);
-}
-
-// Get current received codec.
-int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  return receiver_.LastAudioCodec(current_codec);
-}
-
-int AudioCodingModuleImpl::RegisterDecoder(int acm_codec_id,
-                                           uint8_t payload_type,
-                                           int channels,
-                                           AudioDecoder* audio_decoder) {
-  return receiver_.AddCodec(acm_codec_id, payload_type, channels,
-                            audio_decoder);
-}
-
-// Incoming packet from network parsed and ready for decode.
-int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
-                                          const size_t payload_length,
-                                          const WebRtcRTPHeader& rtp_header) {
-  return receiver_.InsertPacket(rtp_header, incoming_payload, payload_length);
-}
-
-// Minimum playout delay (Used for lip-sync).
-int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
-  if ((time_ms < 0) || (time_ms > 10000)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Delay must be in the range of 0-1000 milliseconds.");
-    return -1;
-  }
-  return receiver_.SetMinimumDelay(time_ms);
-}
-
-int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
-  if ((time_ms < 0) || (time_ms > 10000)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Delay must be in the range of 0-1000 milliseconds.");
-    return -1;
-  }
-  return receiver_.SetMaximumDelay(time_ms);
-}
-
-// Estimate the Bandwidth based on the incoming stream, needed for one way
-// audio where the RTCP send the BW estimate.
-// This is also done in the RTP module.
-int AudioCodingModuleImpl::DecoderEstimatedBandwidth() const {
-  // We can estimate far-end to near-end bandwidth if the iSAC are sent. Check
-  // if the last received packets were iSAC packet then retrieve the bandwidth.
-  int last_audio_codec_id = receiver_.last_audio_codec_id();
-  if (last_audio_codec_id >= 0 &&
-      STR_CASE_CMP("ISAC", ACMCodecDB::database_[last_audio_codec_id].plname)) {
-    CriticalSectionScoped lock(acm_crit_sect_);
-    FATAL() << "Dead code?";
-//    return codecs_[last_audio_codec_id]->GetEstimatedBandwidth();
-  }
-  return -1;
-}
-
-// Set playout mode for: voice, fax, streaming or off.
-int AudioCodingModuleImpl::SetPlayoutMode(AudioPlayoutMode mode) {
-  receiver_.SetPlayoutMode(mode);
-  return 0;  // TODO(turajs): return value is for backward compatibility.
-}
-
-// Get playout mode voice, fax, streaming or off.
-AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
-  return receiver_.PlayoutMode();
-}
-
-// Get 10 milliseconds of raw audio data to play out.
-// Automatic resample to the requested frequency.
-int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
-                                           AudioFrame* audio_frame) {
-  // GetAudio always returns 10 ms, at the requested sample rate.
-  if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "PlayoutData failed, RecOut Failed");
-    return -1;
-  }
-
-  audio_frame->id_ = id_;
-  return 0;
-}
-
-/////////////////////////////////////////
-//   Statistics
-//
-
-// TODO(turajs) change the return value to void. Also change the corresponding
-// NetEq function.
-int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
-  receiver_.GetNetworkStatistics(statistics);
-  return 0;
-}
-
-int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
-  WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
-               "RegisterVADCallback()");
-  CriticalSectionScoped lock(callback_crit_sect_);
-  vad_callback_ = vad_callback;
-  return 0;
-}
-
-// TODO(tlegrand): Modify this function to work for stereo, and add tests.
-int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
-                                           size_t payload_length,
-                                           uint8_t payload_type,
-                                           uint32_t timestamp) {
-  // We are not acquiring any lock when interacting with |aux_rtp_header_| no
-  // other method uses this member variable.
-  if (aux_rtp_header_ == NULL) {
-    // This is the first time that we are using |dummy_rtp_header_|
-    // so we have to create it.
-    aux_rtp_header_ = new WebRtcRTPHeader;
-    aux_rtp_header_->header.payloadType = payload_type;
-    // Don't matter in this case.
-    aux_rtp_header_->header.ssrc = 0;
-    aux_rtp_header_->header.markerBit = false;
-    // Start with random numbers.
-    aux_rtp_header_->header.sequenceNumber = 0x1234;  // Arbitrary.
-    aux_rtp_header_->type.Audio.channel = 1;
-  }
-
-  aux_rtp_header_->header.timestamp = timestamp;
-  IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
-  // Get ready for the next payload.
-  aux_rtp_header_->header.sequenceNumber++;
-  return 0;
-}
-
-int AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) {
-    WEBRTC_TRACE(
-        webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-        "Cannot replace codec internal DTX when no send codec is registered.");
-    return -1;
-  }
-
-  FATAL() << "Dead code?";
-//  int res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX(
-//      use_webrtc_dtx);
-  // Check if VAD is turned on, or if there is any error.
-//  if (res == 1) {
-//    vad_enabled_ = true;
-//  } else if (res < 0) {
-//    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-//                 "Failed to set ReplaceInternalDTXWithWebRtc(%d)",
-//                 use_webrtc_dtx);
-//    return res;
-//  }
-
-  return 0;
-}
-
-int AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
-    bool* uses_webrtc_dtx) {
-  *uses_webrtc_dtx = true;
-  return 0;
-}
-
-// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
-int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  if (!HaveValidEncoder("SetISACMaxRate")) {
-    return -1;
-  }
-
-  return codec_manager_.current_encoder()->SetISACMaxRate(max_bit_per_sec);
-}
-
-// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
-int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
-    return -1;
-  }
-
-  return codec_manager_.current_encoder()->SetISACMaxPayloadSize(
-      max_size_bytes);
-}
-
-// TODO(henrik.lundin): Remove? Only used in tests.
-int AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
-    int frame_size_ms,
-    int rate_bit_per_sec,
-    bool enforce_frame_size) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-
-  if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
-    return -1;
-  }
-
-  FATAL() << "Dead code?";
-  return -1;
-//  return codecs_[current_send_codec_idx_]->ConfigISACBandwidthEstimator(
-//      frame_size_ms, rate_bit_per_sec, enforce_frame_size);
-}
-
-int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application,
-                                              bool disable_dtx_if_needed) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (!HaveValidEncoder("SetOpusApplication")) {
-    return -1;
-  }
-  return codec_manager_.current_encoder()->SetOpusApplication(
-      application, disable_dtx_if_needed);
-}
-
-// Informs Opus encoder of the maximum playback rate the receiver will render.
-int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
-    return -1;
-  }
-  return codec_manager_.current_encoder()->SetOpusMaxPlaybackRate(frequency_hz);
-}
-
-int AudioCodingModuleImpl::EnableOpusDtx(bool force_voip) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (!HaveValidEncoder("EnableOpusDtx")) {
-    return -1;
-  }
-  return codec_manager_.current_encoder()->EnableOpusDtx(force_voip);
-}
-
-int AudioCodingModuleImpl::DisableOpusDtx() {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (!HaveValidEncoder("DisableOpusDtx")) {
-    return -1;
-  }
-  return codec_manager_.current_encoder()->DisableOpusDtx();
-}
-
-int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
-  return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1;
-}
-
-bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
-  if (!codec_manager_.current_encoder()) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "%s failed: No send codec is registered.", caller_name);
-    return false;
-  }
-  return true;
-}
-
-int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
-  return receiver_.RemoveCodec(payload_type);
-}
-
-// TODO(turajs): correct the type of |length_bytes| when it is corrected in
-// GenericCodec.
-int AudioCodingModuleImpl::REDPayloadISAC(int isac_rate,
-                                          int isac_bw_estimate,
-                                          uint8_t* payload,
-                                          int16_t* length_bytes) {
-  CriticalSectionScoped lock(acm_crit_sect_);
-  if (!HaveValidEncoder("EncodeData")) {
-    return -1;
-  }
-  FATAL() << "Dead code?";
-  return -1;
-//  int status;
-//  status = codecs_[current_send_codec_idx_]->REDPayloadISAC(isac_rate,
-//                                                            isac_bw_estimate,
-//                                                            payload,
-//                                                            length_bytes);
-//  return status;
-}
-
-int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
-  {
-    CriticalSectionScoped lock(acm_crit_sect_);
-    // Initialize receiver, if it is not initialized. Otherwise, initial delay
-    // is reset upon initialization of the receiver.
-    if (!receiver_initialized_)
-      InitializeReceiverSafe();
-  }
-  return receiver_.SetInitialDelay(delay_ms);
-}
-
-int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
-  return receiver_.EnableNack(max_nack_list_size);
-}
-
-void AudioCodingModuleImpl::DisableNack() {
-  receiver_.DisableNack();
-}
-
-std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
-    int64_t round_trip_time_ms) const {
-  return receiver_.GetNackList(round_trip_time_ms);
-}
-
-int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
-  return receiver_.LeastRequiredDelayMs();
-}
-
-void AudioCodingModuleImpl::GetDecodingCallStatistics(
-      AudioDecodingCallStats* call_stats) const {
-  receiver_.GetDecodingCallStatistics(call_stats);
-}
-
-}  // namespace acm2
-
-bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::RegisterSendCodec(int encoder_type,
-                                        uint8_t payload_type,
-                                        int frame_size_samples) {
-  std::string codec_name;
-  int sample_rate_hz;
-  int channels;
-  if (!MapCodecTypeToParameters(
-          encoder_type, &codec_name, &sample_rate_hz, &channels)) {
-    return false;
-  }
-  webrtc::CodecInst codec;
-  AudioCodingModule::Codec(
-      codec_name.c_str(), &codec, sample_rate_hz, channels);
-  codec.pltype = payload_type;
-  if (frame_size_samples > 0) {
-    codec.pacsize = frame_size_samples;
-  }
-  return acm_old_->RegisterSendCodec(codec) == 0;
-}
-
-const AudioEncoder* AudioCodingImpl::GetSenderInfo() const {
-  FATAL() << "Not implemented yet.";
-  return NULL;
-}
-
-const CodecInst* AudioCodingImpl::GetSenderCodecInst() {
-  if (acm_old_->SendCodec(&current_send_codec_) != 0) {
-    return NULL;
-  }
-  return &current_send_codec_;
-}
-
-int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) {
-  acm2::AudioCodingModuleImpl::InputData input_data;
-  if (acm_old_->Add10MsDataInternal(audio_frame, &input_data) != 0)
-    return -1;
-  return acm_old_->Encode(input_data);
-}
-
-const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const {
-  FATAL() << "Not implemented yet.";
-  return NULL;
-}
-
-bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::RegisterReceiveCodec(int decoder_type,
-                                           uint8_t payload_type) {
-  std::string codec_name;
-  int sample_rate_hz;
-  int channels;
-  if (!MapCodecTypeToParameters(
-          decoder_type, &codec_name, &sample_rate_hz, &channels)) {
-    return false;
-  }
-  webrtc::CodecInst codec;
-  AudioCodingModule::Codec(
-      codec_name.c_str(), &codec, sample_rate_hz, channels);
-  codec.pltype = payload_type;
-  return acm_old_->RegisterReceiveCodec(codec) == 0;
-}
-
-bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload,
-                                   size_t payload_len_bytes,
-                                   const WebRtcRTPHeader& rtp_info) {
-  return acm_old_->IncomingPacket(
-             incoming_payload, payload_len_bytes, rtp_info) == 0;
-}
-
-bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload,
-                                    size_t payload_len_byte,
-                                    uint8_t payload_type,
-                                    uint32_t timestamp) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-int AudioCodingImpl::LeastRequiredDelayMs() const {
-  FATAL() << "Not implemented yet.";
-  return -1;
-}
-
-bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) {
-  return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0;
-}
-
-bool AudioCodingImpl::GetNetworkStatistics(
-    NetworkStatistics* network_statistics) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-void AudioCodingImpl::DisableNack() {
-  // A bug in the linker of Visual Studio 2013 Update 3 prevent us from using
-  // FATAL() here, if we do so then the linker hang when the WPO is turned on.
-  // TODO(sebmarchand): Re-evaluate this when we upgrade the toolchain.
-}
-
-bool AudioCodingImpl::SetVad(bool enable_dtx,
-                             bool enable_vad,
-                             ACMVADMode vad_mode) {
-  return acm_old_->SetVAD(enable_dtx, enable_vad, vad_mode) == 0;
-}
-
-std::vector<uint16_t> AudioCodingImpl::GetNackList(
-    int round_trip_time_ms) const {
-  return acm_old_->GetNackList(round_trip_time_ms);
-}
-
-void AudioCodingImpl::GetDecodingCallStatistics(
-    AudioDecodingCallStats* call_stats) const {
-  acm_old_->GetDecodingCallStatistics(call_stats);
-}
-
-bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type,
-                                               std::string* codec_name,
-                                               int* sample_rate_hz,
-                                               int* channels) {
-  switch (codec_type) {
-#ifdef WEBRTC_CODEC_PCM16
-    case acm2::ACMCodecDB::kPCM16B:
-      *codec_name = "L16";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kPCM16Bwb:
-      *codec_name = "L16";
-      *sample_rate_hz = 16000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kPCM16Bswb32kHz:
-      *codec_name = "L16";
-      *sample_rate_hz = 32000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kPCM16B_2ch:
-      *codec_name = "L16";
-      *sample_rate_hz = 8000;
-      *channels = 2;
-      break;
-    case acm2::ACMCodecDB::kPCM16Bwb_2ch:
-      *codec_name = "L16";
-      *sample_rate_hz = 16000;
-      *channels = 2;
-      break;
-    case acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch:
-      *codec_name = "L16";
-      *sample_rate_hz = 32000;
-      *channels = 2;
-      break;
-#endif
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-    case acm2::ACMCodecDB::kISAC:
-      *codec_name = "ISAC";
-      *sample_rate_hz = 16000;
-      *channels = 1;
-      break;
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-    case acm2::ACMCodecDB::kISACSWB:
-      *codec_name = "ISAC";
-      *sample_rate_hz = 32000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kISACFB:
-      *codec_name = "ISAC";
-      *sample_rate_hz = 48000;
-      *channels = 1;
-      break;
-#endif
-#ifdef WEBRTC_CODEC_ILBC
-    case acm2::ACMCodecDB::kILBC:
-      *codec_name = "ILBC";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-#endif
-    case acm2::ACMCodecDB::kPCMA:
-      *codec_name = "PCMA";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kPCMA_2ch:
-      *codec_name = "PCMA";
-      *sample_rate_hz = 8000;
-      *channels = 2;
-      break;
-    case acm2::ACMCodecDB::kPCMU:
-      *codec_name = "PCMU";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kPCMU_2ch:
-      *codec_name = "PCMU";
-      *sample_rate_hz = 8000;
-      *channels = 2;
-      break;
-#ifdef WEBRTC_CODEC_G722
-    case acm2::ACMCodecDB::kG722:
-      *codec_name = "G722";
-      *sample_rate_hz = 16000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kG722_2ch:
-      *codec_name = "G722";
-      *sample_rate_hz = 16000;
-      *channels = 2;
-      break;
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-    case acm2::ACMCodecDB::kOpus:
-      *codec_name = "opus";
-      *sample_rate_hz = 48000;
-      *channels = 2;
-      break;
-#endif
-    case acm2::ACMCodecDB::kCNNB:
-      *codec_name = "CN";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kCNWB:
-      *codec_name = "CN";
-      *sample_rate_hz = 16000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kCNSWB:
-      *codec_name = "CN";
-      *sample_rate_hz = 32000;
-      *channels = 1;
-      break;
-    case acm2::ACMCodecDB::kRED:
-      *codec_name = "red";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-#ifdef WEBRTC_CODEC_AVT
-    case acm2::ACMCodecDB::kAVT:
-      *codec_name = "telephone-event";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-#endif
-    default:
-      FATAL() << "Codec type " << codec_type << " not supported.";
-  }
-  return true;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
deleted file mode 100644
index 283595ee41..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ /dev/null
@@ -1,966 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/md5digest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-const int kSampleRateHz = 16000;
-const int kNumSamples10ms = kSampleRateHz / 100;
-const int kFrameSizeMs = 10;  // Multiple of 10.
-const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
-const size_t kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
-const uint8_t kPayloadType = 111;
-
-class RtpUtility {
- public:
-  RtpUtility(int samples_per_packet, uint8_t payload_type)
-      : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
-
-  virtual ~RtpUtility() {}
-
-  void Populate(WebRtcRTPHeader* rtp_header) {
-    rtp_header->header.sequenceNumber = 0xABCD;
-    rtp_header->header.timestamp = 0xABCDEF01;
-    rtp_header->header.payloadType = payload_type_;
-    rtp_header->header.markerBit = false;
-    rtp_header->header.ssrc = 0x1234;
-    rtp_header->header.numCSRCs = 0;
-    rtp_header->frameType = kAudioFrameSpeech;
-
-    rtp_header->header.payload_type_frequency = kSampleRateHz;
-    rtp_header->type.Audio.channel = 1;
-    rtp_header->type.Audio.isCNG = false;
-  }
-
-  void Forward(WebRtcRTPHeader* rtp_header) {
-    ++rtp_header->header.sequenceNumber;
-    rtp_header->header.timestamp += samples_per_packet_;
-  }
-
- private:
-  int samples_per_packet_;
-  uint8_t payload_type_;
-};
-
-class PacketizationCallbackStub : public AudioPacketizationCallback {
- public:
-  PacketizationCallbackStub()
-      : num_calls_(0),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
-
-  int32_t SendData(FrameType frame_type,
-                   uint8_t payload_type,
-                   uint32_t timestamp,
-                   const uint8_t* payload_data,
-                   size_t payload_len_bytes,
-                   const RTPFragmentationHeader* fragmentation) override {
-    CriticalSectionScoped lock(crit_sect_.get());
-    ++num_calls_;
-    last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
-    return 0;
-  }
-
-  int num_calls() const {
-    CriticalSectionScoped lock(crit_sect_.get());
-    return num_calls_;
-  }
-
-  int last_payload_len_bytes() const {
-    CriticalSectionScoped lock(crit_sect_.get());
-    return last_payload_vec_.size();
-  }
-
-  void SwapBuffers(std::vector<uint8_t>* payload) {
-    CriticalSectionScoped lock(crit_sect_.get());
-    last_payload_vec_.swap(*payload);
-  }
-
- private:
-  int num_calls_ GUARDED_BY(crit_sect_);
-  std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-};
-
-class AudioCodingModuleTest : public ::testing::Test {
- protected:
-  AudioCodingModuleTest()
-      : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) {
-    config_.transport = &packet_cb_;
-  }
-
-  ~AudioCodingModuleTest() {}
-
-  void TearDown() override {}
-
-  void SetUp() override {
-    rtp_utility_->Populate(&rtp_header_);
-
-    input_frame_.sample_rate_hz_ = kSampleRateHz;
-    input_frame_.num_channels_ = 1;
-    input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000;  // 10 ms.
-    static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
-                  "audio frame too small");
-    memset(input_frame_.data_,
-           0,
-           input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0]));
-  }
-
-  void CreateAcm() {
-    acm_.reset(AudioCoding::Create(config_));
-    ASSERT_TRUE(acm_.get() != NULL);
-    RegisterCodec();
-  }
-
-  virtual void RegisterCodec() {
-    // Register L16 codec in ACM.
-    int codec_type = acm2::ACMCodecDB::kNone;
-    switch (kSampleRateHz) {
-      case 8000:
-        codec_type = acm2::ACMCodecDB::kPCM16B;
-        break;
-      case 16000:
-        codec_type = acm2::ACMCodecDB::kPCM16Bwb;
-        break;
-      case 32000:
-        codec_type = acm2::ACMCodecDB::kPCM16Bswb32kHz;
-        break;
-      default:
-        FATAL() << "Sample rate not supported in this test.";
-    }
-    ASSERT_TRUE(acm_->RegisterSendCodec(codec_type, kPayloadType));
-    ASSERT_TRUE(acm_->RegisterReceiveCodec(codec_type, kPayloadType));
-  }
-
-  virtual void InsertPacketAndPullAudio() {
-    InsertPacket();
-    PullAudio();
-  }
-
-  virtual void InsertPacket() {
-    const uint8_t kPayload[kPayloadSizeBytes] = {0};
-    ASSERT_TRUE(acm_->InsertPacket(kPayload, kPayloadSizeBytes, rtp_header_));
-    rtp_utility_->Forward(&rtp_header_);
-  }
-
-  virtual void PullAudio() {
-    AudioFrame audio_frame;
-    ASSERT_TRUE(acm_->Get10MsAudio(&audio_frame));
-  }
-
-  virtual void InsertAudio() {
-    int encoded_bytes = acm_->Add10MsAudio(input_frame_);
-    ASSERT_GE(encoded_bytes, 0);
-    input_frame_.timestamp_ += kNumSamples10ms;
-  }
-
-  AudioCoding::Config config_;
-  rtc::scoped_ptr<RtpUtility> rtp_utility_;
-  rtc::scoped_ptr<AudioCoding> acm_;
-  PacketizationCallbackStub packet_cb_;
-  WebRtcRTPHeader rtp_header_;
-  AudioFrame input_frame_;
-};
-
-// Check if the statistics are initialized correctly. Before any call to ACM
-// all fields have to be zero.
-TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) {
-  CreateAcm();
-  AudioDecodingCallStats stats;
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(0, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
-// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
-// should result in generating silence, check the associated field.
-TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
-  const int kInitialDelay = 100;
-  config_.initial_playout_delay_ms = kInitialDelay;
-  CreateAcm();
-  AudioDecodingCallStats stats;
-
-  int num_calls = 0;
-  for (int time_ms = 0; time_ms < kInitialDelay;
-       time_ms += kFrameSizeMs, ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
-// Insert some packets and pull audio. Check statistics are valid. Then,
-// simulate packet loss and check if PLC and PLC-to-CNG statistics are
-// correctly updated.
-TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) {
-  CreateAcm();
-  AudioDecodingCallStats stats;
-  const int kNumNormalCalls = 10;
-
-  for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
-  EXPECT_EQ(0, stats.calls_to_silence_generator);
-  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-
-  const int kNumPlc = 3;
-  const int kNumPlcCng = 5;
-
-  // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
-  for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
-    PullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
-  EXPECT_EQ(0, stats.calls_to_silence_generator);
-  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(kNumPlc, stats.decoded_plc);
-  EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
-}
-
-TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {
-  CreateAcm();
-  AudioFrame audio_frame;
-  const int kSampleRateHz = 32000;
-  EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame));
-  EXPECT_EQ(0u, audio_frame.timestamp_);
-  EXPECT_GT(audio_frame.num_channels_, 0);
-  EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
-  EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
-}
-
-// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz
-// codec, while the derive class AcmIsacMtTest is using iSAC.
-class AudioCodingModuleMtTest : public AudioCodingModuleTest {
- protected:
-  static const int kNumPackets = 500;
-  static const int kNumPullCalls = 500;
-
-  AudioCodingModuleMtTest()
-      : AudioCodingModuleTest(),
-        send_thread_(ThreadWrapper::CreateThread(CbSendThread, this, "send")),
-        insert_packet_thread_(ThreadWrapper::CreateThread(
-            CbInsertPacketThread, this, "insert_packet")),
-        pull_audio_thread_(ThreadWrapper::CreateThread(
-            CbPullAudioThread, this, "pull_audio")),
-        test_complete_(EventWrapper::Create()),
-        send_count_(0),
-        insert_packet_count_(0),
-        pull_audio_count_(0),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-        next_insert_packet_time_ms_(0),
-        fake_clock_(new SimulatedClock(0)) {
-    config_.clock = fake_clock_.get();
-  }
-
-  void SetUp() override {
-    AudioCodingModuleTest::SetUp();
-    CreateAcm();
-    StartThreads();
-  }
-
-  void StartThreads() {
-    ASSERT_TRUE(send_thread_->Start());
-    send_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(insert_packet_thread_->Start());
-    insert_packet_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(pull_audio_thread_->Start());
-    pull_audio_thread_->SetPriority(kRealtimePriority);
-  }
-
-  void TearDown() override {
-    AudioCodingModuleTest::TearDown();
-    pull_audio_thread_->Stop();
-    send_thread_->Stop();
-    insert_packet_thread_->Stop();
-  }
-
-  EventTypeWrapper RunTest() {
-    return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
-  }
-
-  virtual bool TestDone() {
-    if (packet_cb_.num_calls() > kNumPackets) {
-      CriticalSectionScoped lock(crit_sect_.get());
-      if (pull_audio_count_ > kNumPullCalls) {
-        // Both conditions for completion are met. End the test.
-        return true;
-      }
-    }
-    return false;
-  }
-
-  static bool CbSendThread(void* context) {
-    return reinterpret_cast<AudioCodingModuleMtTest*>(context)->CbSendImpl();
-  }
-
-  // The send thread doesn't have to care about the current simulated time,
-  // since only the AcmReceiver is using the clock.
-  bool CbSendImpl() {
-    SleepMs(1);
-    if (HasFatalFailure()) {
-      // End the test early if a fatal failure (ASSERT_*) has occurred.
-      test_complete_->Set();
-    }
-    ++send_count_;
-    InsertAudio();
-    if (TestDone()) {
-      test_complete_->Set();
-    }
-    return true;
-  }
-
-  static bool CbInsertPacketThread(void* context) {
-    return reinterpret_cast<AudioCodingModuleMtTest*>(context)
-        ->CbInsertPacketImpl();
-  }
-
-  bool CbInsertPacketImpl() {
-    SleepMs(1);
-    {
-      CriticalSectionScoped lock(crit_sect_.get());
-      if (fake_clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
-        return true;
-      }
-      next_insert_packet_time_ms_ += 10;
-    }
-    // Now we're not holding the crit sect when calling ACM.
-    ++insert_packet_count_;
-    InsertPacket();
-    return true;
-  }
-
-  static bool CbPullAudioThread(void* context) {
-    return reinterpret_cast<AudioCodingModuleMtTest*>(context)
-        ->CbPullAudioImpl();
-  }
-
-  bool CbPullAudioImpl() {
-    SleepMs(1);
-    {
-      CriticalSectionScoped lock(crit_sect_.get());
-      // Don't let the insert thread fall behind.
-      if (next_insert_packet_time_ms_ < fake_clock_->TimeInMilliseconds()) {
-        return true;
-      }
-      ++pull_audio_count_;
-    }
-    // Now we're not holding the crit sect when calling ACM.
-    PullAudio();
-    fake_clock_->AdvanceTimeMilliseconds(10);
-    return true;
-  }
-
-  rtc::scoped_ptr<ThreadWrapper> send_thread_;
-  rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
-  rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
-  const rtc::scoped_ptr<EventWrapper> test_complete_;
-  int send_count_;
-  int insert_packet_count_;
-  int pull_audio_count_ GUARDED_BY(crit_sect_);
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
-  rtc::scoped_ptr<SimulatedClock> fake_clock_;
-};
-
-TEST_F(AudioCodingModuleMtTest, DoTest) {
-  EXPECT_EQ(kEventSignaled, RunTest());
-}
-
-// This is a multi-threaded ACM test using iSAC. The test encodes audio
-// from a PCM file. The most recent encoded frame is used as input to the
-// receiving part. Depending on timing, it may happen that the same RTP packet
-// is inserted into the receiver multiple times, but this is a valid use-case,
-// and simplifies the test code a lot.
-class AcmIsacMtTest : public AudioCodingModuleMtTest {
- protected:
-  static const int kNumPackets = 500;
-  static const int kNumPullCalls = 500;
-
-  AcmIsacMtTest()
-      : AudioCodingModuleMtTest(),
-        last_packet_number_(0) {}
-
-  ~AcmIsacMtTest() {}
-
-  void SetUp() override {
-    AudioCodingModuleTest::SetUp();
-    CreateAcm();
-
-    // Set up input audio source to read from specified file, loop after 5
-    // seconds, and deliver blocks of 10 ms.
-    const std::string input_file_name =
-        webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
-    audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
-
-    // Generate one packet to have something to insert.
-    int loop_counter = 0;
-    while (packet_cb_.last_payload_len_bytes() == 0) {
-      InsertAudio();
-      ASSERT_LT(loop_counter++, 10);
-    }
-    // Set |last_packet_number_| to one less that |num_calls| so that the packet
-    // will be fetched in the next InsertPacket() call.
-    last_packet_number_ = packet_cb_.num_calls() - 1;
-
-    StartThreads();
-  }
-
-  void RegisterCodec() override {
-    static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
-
-    // Register iSAC codec in ACM, effectively unregistering the PCM16B codec
-    // registered in AudioCodingModuleTest::SetUp();
-    ASSERT_TRUE(acm_->RegisterSendCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
-    ASSERT_TRUE(
-        acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
-  }
-
-  void InsertPacket() override {
-    int num_calls = packet_cb_.num_calls();  // Store locally for thread safety.
-    if (num_calls > last_packet_number_) {
-      // Get the new payload out from the callback handler.
-      // Note that since we swap buffers here instead of directly inserting
-      // a pointer to the data in |packet_cb_|, we avoid locking the callback
-      // for the duration of the IncomingPacket() call.
-      packet_cb_.SwapBuffers(&last_payload_vec_);
-      ASSERT_GT(last_payload_vec_.size(), 0u);
-      rtp_utility_->Forward(&rtp_header_);
-      last_packet_number_ = num_calls;
-    }
-    ASSERT_GT(last_payload_vec_.size(), 0u);
-    ASSERT_TRUE(acm_->InsertPacket(
-        &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
-  }
-
-  void InsertAudio() override {
-    memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
-    AudioCodingModuleTest::InsertAudio();
-  }
-
-  // This method is the same as AudioCodingModuleMtTest::TestDone(), but here
-  // it is using the constants defined in this class (i.e., shorter test run).
-  bool TestDone() override {
-    if (packet_cb_.num_calls() > kNumPackets) {
-      CriticalSectionScoped lock(crit_sect_.get());
-      if (pull_audio_count_ > kNumPullCalls) {
-        // Both conditions for completion are met. End the test.
-        return true;
-      }
-    }
-    return false;
-  }
-
-  int last_packet_number_;
-  std::vector<uint8_t> last_payload_vec_;
-  test::AudioLoop audio_loop_;
-};
-
-TEST_F(AcmIsacMtTest, DoTest) {
-  EXPECT_EQ(kEventSignaled, RunTest());
-}
-
-class AcmReceiverBitExactness : public ::testing::Test {
- public:
-  static std::string PlatformChecksum(std::string win64,
-                                      std::string android,
-                                      std::string others) {
-#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
-    return win64;
-#elif defined(WEBRTC_ANDROID)
-    return android;
-#else
-    return others;
-#endif
-  }
-
- protected:
-  void Run(int output_freq_hz, const std::string& checksum_ref) {
-    const std::string input_file_name =
-        webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
-    rtc::scoped_ptr<test::RtpFileSource> packet_source(
-        test::RtpFileSource::Create(input_file_name));
-#ifdef WEBRTC_ANDROID
-    // Filter out iLBC and iSAC-swb since they are not supported on Android.
-    packet_source->FilterOutPayloadType(102);  // iLBC.
-    packet_source->FilterOutPayloadType(104);  // iSAC-swb.
-#endif
-
-    test::AudioChecksum checksum;
-    const std::string output_file_name =
-        webrtc::test::OutputPath() +
-        ::testing::UnitTest::GetInstance()
-            ->current_test_info()
-            ->test_case_name() +
-        "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
-        "_output.pcm";
-    test::OutputAudioFile output_file(output_file_name);
-    test::AudioSinkFork output(&checksum, &output_file);
-
-    test::AcmReceiveTest test(packet_source.get(), &output, output_freq_hz,
-                              test::AcmReceiveTest::kArbitraryChannels);
-    ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
-    test.Run();
-
-    std::string checksum_string = checksum.Finish();
-    EXPECT_EQ(checksum_ref, checksum_string);
-  }
-};
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_8kHzOutput DISABLED_8kHzOutput
-#else
-#define MAYBE_8kHzOutput 8kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, MAYBE_8kHzOutput) {
-  Run(8000,
-      PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
-                       "adc92e173f908f93b96ba5844209815a",
-                       "908002dc01fc4eb1d2be24eb1d3f354b"));
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_16kHzOutput DISABLED_16kHzOutput
-#else
-#define MAYBE_16kHzOutput 16kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, MAYBE_16kHzOutput) {
-  Run(16000,
-      PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
-                       "8cffa6abcb3e18e33b9d857666dff66a",
-                       "a909560b5ca49fa472b17b7b277195e9"));
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_32kHzOutput DISABLED_32kHzOutput
-#else
-#define MAYBE_32kHzOutput 32kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, MAYBE_32kHzOutput) {
-  Run(32000,
-      PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
-                       "3e126fe894720c3f85edadcc91964ba5",
-                       "441aab4b347fb3db4e9244337aca8d8e"));
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_48kHzOutput DISABLED_48kHzOutput
-#else
-#define MAYBE_48kHzOutput 48kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, MAYBE_48kHzOutput) {
-  Run(48000,
-      PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
-                       "0155665e93067c4e89256b944dd11999",
-                       "4ee2730fa1daae755e8a8fd3abd779ec"));
-}
-
-// This test verifies bit exactness for the send-side of ACM. The test setup is
-// a chain of three different test classes:
-//
-// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
-//
-// The receiver side is driving the test by requesting new packets from
-// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
-// packet from test::AcmSendTest::NextPacket, which inserts audio from the
-// input file until one packet is produced. (The input file loops indefinitely.)
-// Before passing the packet to the receiver, this test class verifies the
-// packet header and updates a payload checksum with the new payload. The
-// decoded output from the receiver is also verified with a (separate) checksum.
-class AcmSenderBitExactness : public ::testing::Test,
-                              public test::PacketSource {
- protected:
-  static const int kTestDurationMs = 1000;
-
-  AcmSenderBitExactness()
-      : frame_size_rtp_timestamps_(0),
-        packet_count_(0),
-        payload_type_(0),
-        last_sequence_number_(0),
-        last_timestamp_(0) {}
-
-  // Sets up the test::AcmSendTest object. Returns true on success, otherwise
-  // false.
-  bool SetUpSender() {
-    const std::string input_file_name =
-        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-    // Note that |audio_source_| will loop forever. The test duration is set
-    // explicitly by |kTestDurationMs|.
-    audio_source_.reset(new test::InputAudioFile(input_file_name));
-    static const int kSourceRateHz = 32000;
-    send_test_.reset(new test::AcmSendTest(
-        audio_source_.get(), kSourceRateHz, kTestDurationMs));
-    return send_test_.get() != NULL;
-  }
-
-  // Registers a send codec in the test::AcmSendTest object. Returns true on
-  // success, false on failure.
-  bool RegisterSendCodec(int codec_type,
-                         int channels,
-                         int payload_type,
-                         int frame_size_samples,
-                         int frame_size_rtp_timestamps) {
-    payload_type_ = payload_type;
-    frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
-    return send_test_->RegisterCodec(
-        codec_type, channels, payload_type, frame_size_samples);
-  }
-
-  // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
-  // before calling this method.
-  void Run(const std::string& audio_checksum_ref,
-           const std::string& payload_checksum_ref,
-           int expected_packets,
-           test::AcmReceiveTest::NumOutputChannels expected_channels) {
-    // Set up the receiver used to decode the packets and verify the decoded
-    // output.
-    test::AudioChecksum audio_checksum;
-    const std::string output_file_name =
-        webrtc::test::OutputPath() +
-        ::testing::UnitTest::GetInstance()
-            ->current_test_info()
-            ->test_case_name() +
-        "_" +
-        ::testing::UnitTest::GetInstance()->current_test_info()->name() +
-        "_output.pcm";
-    test::OutputAudioFile output_file(output_file_name);
-    // Have the output audio sent both to file and to the checksum calculator.
-    test::AudioSinkFork output(&audio_checksum, &output_file);
-    const int kOutputFreqHz = 8000;
-    test::AcmReceiveTest receive_test(
-        this, &output, kOutputFreqHz, expected_channels);
-    ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
-
-    // This is where the actual test is executed.
-    receive_test.Run();
-
-    // Extract and verify the audio checksum.
-    std::string checksum_string = audio_checksum.Finish();
-    EXPECT_EQ(audio_checksum_ref, checksum_string);
-
-    // Extract and verify the payload checksum.
-    char checksum_result[rtc::Md5Digest::kSize];
-    payload_checksum_.Finish(checksum_result, rtc::Md5Digest::kSize);
-    checksum_string = rtc::hex_encode(checksum_result, rtc::Md5Digest::kSize);
-    EXPECT_EQ(payload_checksum_ref, checksum_string);
-
-    // Verify number of packets produced.
-    EXPECT_EQ(expected_packets, packet_count_);
-  }
-
-  // Returns a pointer to the next packet. Returns NULL if the source is
-  // depleted (i.e., the test duration is exceeded), or if an error occurred.
-  // Inherited from test::PacketSource.
-  test::Packet* NextPacket() override {
-    // Get the next packet from AcmSendTest. Ownership of |packet| is
-    // transferred to this method.
-    test::Packet* packet = send_test_->NextPacket();
-    if (!packet)
-      return NULL;
-
-    VerifyPacket(packet);
-    // TODO(henrik.lundin) Save the packet to file as well.
-
-    // Pass it on to the caller. The caller becomes the owner of |packet|.
-    return packet;
-  }
-
-  // Verifies the packet.
-  void VerifyPacket(const test::Packet* packet) {
-    EXPECT_TRUE(packet->valid_header());
-    // (We can check the header fields even if valid_header() is false.)
-    EXPECT_EQ(payload_type_, packet->header().payloadType);
-    if (packet_count_ > 0) {
-      // This is not the first packet.
-      uint16_t sequence_number_diff =
-          packet->header().sequenceNumber - last_sequence_number_;
-      EXPECT_EQ(1, sequence_number_diff);
-      uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_;
-      EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff);
-    }
-    ++packet_count_;
-    last_sequence_number_ = packet->header().sequenceNumber;
-    last_timestamp_ = packet->header().timestamp;
-    // Update the checksum.
-    payload_checksum_.Update(packet->payload(), packet->payload_length_bytes());
-  }
-
-  void SetUpTest(int codec_type,
-                 int channels,
-                 int payload_type,
-                 int codec_frame_size_samples,
-                 int codec_frame_size_rtp_timestamps) {
-    ASSERT_TRUE(SetUpSender());
-    ASSERT_TRUE(RegisterSendCodec(codec_type,
-                                  channels,
-                                  payload_type,
-                                  codec_frame_size_samples,
-                                  codec_frame_size_rtp_timestamps));
-  }
-
-  rtc::scoped_ptr<test::AcmSendTest> send_test_;
-  rtc::scoped_ptr<test::InputAudioFile> audio_source_;
-  uint32_t frame_size_rtp_timestamps_;
-  int packet_count_;
-  uint8_t payload_type_;
-  uint16_t last_sequence_number_;
-  uint32_t last_timestamp_;
-  rtc::Md5Digest payload_checksum_;
-};
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_IsacWb30ms DISABLED_IsacWb30ms
-#else
-#define MAYBE_IsacWb30ms IsacWb30ms
-#endif
-TEST_F(AcmSenderBitExactness, MAYBE_IsacWb30ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 480, 480));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "c7e5bdadfa2871df95639fcc297cf23d",
-          "0499ca260390769b3172136faad925b9",
-          "0b58f9eeee43d5891f5f6c75e77984a3"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "d42cb5195463da26c8129bbfe73a22e6",
-          "83de248aea9c3c2bd680b6952401b4ca",
-          "3c79f16f34218271f3dca4e2b1dfe1bb"),
-      33,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_IsacWb60ms DISABLED_IsacWb60ms
-#else
-#define MAYBE_IsacWb60ms IsacWb60ms
-#endif
-TEST_F(AcmSenderBitExactness, MAYBE_IsacWb60ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 960, 960));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "14d63c5f08127d280e722e3191b73bdd",
-          "8da003e16c5371af2dc2be79a50f9076",
-          "1ad29139a04782a33daad8c2b9b35875"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "ebe04a819d3a9d83a83a17f271e1139a",
-          "97aeef98553b5a4b5a68f8b716e8eaf0",
-          "9e0a0ab743ad987b55b8e14802769c56"),
-      16,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "2b3c387d06f00b7b7aad4c9be56fb83d",
-          "",
-          "5683b58da0fbf2063c7adc2e6bfb3fb8"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "bcc2041e7744c7ebd9f701866856849c",
-          "",
-          "ce86106a93419aefb063097108ec94ab"),
-      33, test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCM16B, 1, 107, 80, 80));
-  Run("de4a98e1406f8b798d99cd0704e862e2",
-      "c1edd36339ce0326cc4550041ad719a0",
-      100,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_16000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bwb, 1, 108, 160, 160));
-  Run("ae646d7b68384a1269cc080dd4501916",
-      "ad786526383178b08d80d6eee06e9bad",
-      100,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_32000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz, 1, 109, 320, 320));
-  Run("7fe325e8fbaf755e3c5df0b11a4774fb",
-      "5ef82ea885e922263606c6fdbc49f651",
-      100,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_stereo_8000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16B_2ch, 2, 111, 80, 80));
-  Run("fb263b74e7ac3de915474d77e4744ceb",
-      "62ce5adb0d4965d0a52ec98ae7f98974",
-      100,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_stereo_16000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bwb_2ch, 2, 112, 160, 160));
-  Run("d09e9239553649d7ac93e19d304281fd",
-      "41ca8edac4b8c71cd54fd9f25ec14870",
-      100,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_stereo_32000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 2, 113, 320, 320));
-  Run("5f025d4f390982cc26b3d92fe02e3044",
-      "50e58502fb04421bf5b857dda4c96879",
-      100,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcmu_20ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMU, 1, 0, 160, 160));
-  Run("81a9d4c0bb72e9becc43aef124c981e9",
-      "8f9b8750bd80fe26b6cbf6659b89f0f9",
-      50,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcma_20ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMA, 1, 8, 160, 160));
-  Run("39611f798969053925a49dc06d08de29",
-      "6ad745e55aa48981bfc790d0eeef2dd1",
-      50,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcmu_stereo_20ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCMU_2ch, 2, 110, 160, 160));
-  Run("437bec032fdc5cbaa0d5175430af7b18",
-      "60b6f25e8d1e74cb679cfe756dd9bca5",
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCMA_2ch, 2, 118, 160, 160));
-  Run("a5c6d83c5b7cedbeff734238220a4b0c",
-      "92b282c83efd20e7eeef52ba40842cf7",
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kILBC, 1, 102, 240, 240));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "7b6ec10910debd9af08011d3ed5249f7",
-          "android_audio",
-          "7b6ec10910debd9af08011d3ed5249f7"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "cfae2e9f6aba96e145f2bcdd5050ce78",
-          "android_payload",
-          "cfae2e9f6aba96e145f2bcdd5050ce78"),
-      33,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kG722, 1, 9, 320, 160));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "7d759436f2533582950d148b5161a36c",
-          "android_audio",
-          "7d759436f2533582950d148b5161a36c"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "fc68a87e1380614e658087cb35d5ca10",
-          "android_payload",
-          "fc68a87e1380614e658087cb35d5ca10"),
-      50,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kG722_2ch, 2, 119, 320, 160));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "7190ee718ab3d80eca181e5f7140c210",
-          "android_audio",
-          "7190ee718ab3d80eca181e5f7140c210"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "66516152eeaa1e650ad94ff85f668dac",
-          "android_payload",
-          "66516152eeaa1e650ad94ff85f668dac"),
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
-#else
-#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
-#endif
-TEST_F(AcmSenderBitExactness, MAYBE_Opus_stereo_20ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kOpus, 2, 120, 960, 960));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "855041f2490b887302bce9d544731849",
-          "1e1a0fce893fef2d66886a7f09e2ebce",
-          "855041f2490b887302bce9d544731849"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "d781cce1ab986b618d0da87226cdde30",
-          "1a1fe04dd12e755949987c8d729fb3e0",
-          "d781cce1ab986b618d0da87226cdde30"),
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
deleted file mode 100644
index c3e340e0e2..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
+++ /dev/null
@@ -1,612 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-
-namespace webrtc {
-namespace acm2 {
-
-namespace {
-bool IsCodecRED(const CodecInst* codec) {
-  return (STR_CASE_CMP(codec->plname, "RED") == 0);
-}
-
-bool IsCodecRED(int index) {
-  return (IsCodecRED(&ACMCodecDB::database_[index]));
-}
-
-bool IsCodecCN(const CodecInst* codec) {
-  return (STR_CASE_CMP(codec->plname, "CN") == 0);
-}
-
-bool IsCodecCN(int index) {
-  return (IsCodecCN(&ACMCodecDB::database_[index]));
-}
-
-// Check if the given codec is a valid to be registered as send codec.
-int IsValidSendCodec(const CodecInst& send_codec,
-                     bool is_primary_encoder,
-                     int* mirror_id) {
-  int dummy_id = 0;
-  if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "Wrong number of channels (%d, only mono and stereo are "
-                 "supported) for %s encoder",
-                 send_codec.channels,
-                 is_primary_encoder ? "primary" : "secondary");
-    return -1;
-  }
-
-  int codec_id = ACMCodecDB::CodecNumber(send_codec, mirror_id);
-  if (codec_id < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "Invalid codec setting for the send codec.");
-    return -1;
-  }
-
-  // TODO(tlegrand): Remove this check. Already taken care of in
-  // ACMCodecDB::CodecNumber().
-  // Check if the payload-type is valid
-  if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "Invalid payload-type %d for %s.", send_codec.pltype,
-                 send_codec.plname);
-    return -1;
-  }
-
-  // Telephone-event cannot be a send codec.
-  if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "telephone-event cannot be a send codec");
-    *mirror_id = -1;
-    return -1;
-  }
-
-  if (ACMCodecDB::codec_settings_[codec_id].channel_support <
-      send_codec.channels) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                 "%d number of channels not supportedn for %s.",
-                 send_codec.channels, send_codec.plname);
-    *mirror_id = -1;
-    return -1;
-  }
-
-  if (!is_primary_encoder) {
-    // If registering the secondary encoder, then RED and CN are not valid
-    // choices as encoder.
-    if (IsCodecRED(&send_codec)) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "RED cannot be secondary codec");
-      *mirror_id = -1;
-      return -1;
-    }
-
-    if (IsCodecCN(&send_codec)) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "DTX cannot be secondary codec");
-      *mirror_id = -1;
-      return -1;
-    }
-  }
-  return codec_id;
-}
-
-const CodecInst kEmptyCodecInst = {-1, "noCodecRegistered", 0, 0, 0, 0};
-}  // namespace
-
-CodecManager::CodecManager(AudioCodingModuleImpl* acm)
-    : acm_(acm),
-      cng_nb_pltype_(255),
-      cng_wb_pltype_(255),
-      cng_swb_pltype_(255),
-      cng_fb_pltype_(255),
-      red_nb_pltype_(255),
-      stereo_send_(false),
-      vad_enabled_(false),
-      dtx_enabled_(false),
-      vad_mode_(VADNormal),
-      current_encoder_(nullptr),
-      send_codec_inst_(kEmptyCodecInst),
-      red_enabled_(false),
-      codec_fec_enabled_(false) {
-  for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) {
-    codecs_[i] = nullptr;
-    mirror_codec_idx_[i] = -1;
-  }
-
-  // Register the default payload type for RED and for CNG at sampling rates of
-  // 8, 16, 32 and 48 kHz.
-  for (int i = (ACMCodecDB::kNumCodecs - 1); i >= 0; i--) {
-    if (IsCodecRED(i) && ACMCodecDB::database_[i].plfreq == 8000) {
-      red_nb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
-    } else if (IsCodecCN(i)) {
-      if (ACMCodecDB::database_[i].plfreq == 8000) {
-        cng_nb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
-      } else if (ACMCodecDB::database_[i].plfreq == 16000) {
-        cng_wb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
-      } else if (ACMCodecDB::database_[i].plfreq == 32000) {
-        cng_swb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
-      } else if (ACMCodecDB::database_[i].plfreq == 48000) {
-        cng_fb_pltype_ = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype);
-      }
-    }
-  }
-  thread_checker_.DetachFromThread();
-}
-
-CodecManager::~CodecManager() {
-  for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) {
-    if (codecs_[i] != NULL) {
-      // Mirror index holds the address of the codec memory.
-      assert(mirror_codec_idx_[i] > -1);
-      if (codecs_[mirror_codec_idx_[i]] != NULL) {
-        delete codecs_[mirror_codec_idx_[i]];
-        codecs_[mirror_codec_idx_[i]] = NULL;
-      }
-
-      codecs_[i] = NULL;
-    }
-  }
-}
-
-int CodecManager::RegisterSendCodec(const CodecInst& send_codec) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  int mirror_id;
-  int codec_id = IsValidSendCodec(send_codec, true, &mirror_id);
-
-  // Check for reported errors from function IsValidSendCodec().
-  if (codec_id < 0) {
-    return -1;
-  }
-
-  int dummy_id = 0;
-  // RED can be registered with other payload type. If not registered a default
-  // payload type is used.
-  if (IsCodecRED(&send_codec)) {
-    // TODO(tlegrand): Remove this check. Already taken care of in
-    // ACMCodecDB::CodecNumber().
-    // Check if the payload-type is valid
-    if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "Invalid payload-type %d for %s.", send_codec.pltype,
-                   send_codec.plname);
-      return -1;
-    }
-    // Set RED payload type.
-    if (send_codec.plfreq == 8000) {
-      red_nb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-    } else {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                   "RegisterSendCodec() failed, invalid frequency for RED "
-                   "registration");
-      return -1;
-    }
-    SetRedPayloadType(send_codec.plfreq, send_codec.pltype);
-    return 0;
-  }
-
-  // CNG can be registered with other payload type. If not registered the
-  // default payload types from codec database will be used.
-  if (IsCodecCN(&send_codec)) {
-    // CNG is registered.
-    switch (send_codec.plfreq) {
-      case 8000: {
-        cng_nb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        break;
-      }
-      case 16000: {
-        cng_wb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        break;
-      }
-      case 32000: {
-        cng_swb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        break;
-      }
-      case 48000: {
-        cng_fb_pltype_ = static_cast<uint8_t>(send_codec.pltype);
-        break;
-      }
-      default: {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "RegisterSendCodec() failed, invalid frequency for CNG "
-                     "registration");
-        return -1;
-      }
-    }
-    SetCngPayloadType(send_codec.plfreq, send_codec.pltype);
-    return 0;
-  }
-
-  // Set Stereo, and make sure VAD and DTX is turned off.
-  if (send_codec.channels == 2) {
-    stereo_send_ = true;
-    if (vad_enabled_ || dtx_enabled_) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, dummy_id,
-                   "VAD/DTX is turned off, not supported when sending stereo.");
-    }
-    vad_enabled_ = false;
-    dtx_enabled_ = false;
-  } else {
-    stereo_send_ = false;
-  }
-
-  // Check if the codec is already registered as send codec.
-  bool is_send_codec;
-  if (current_encoder_) {
-    int send_codec_mirror_id;
-    int send_codec_id =
-        ACMCodecDB::CodecNumber(send_codec_inst_, &send_codec_mirror_id);
-    assert(send_codec_id >= 0);
-    is_send_codec =
-        (send_codec_id == codec_id) || (mirror_id == send_codec_mirror_id);
-  } else {
-    is_send_codec = false;
-  }
-
-  // If new codec, or new settings, register.
-  if (!is_send_codec) {
-    if (!codecs_[mirror_id]) {
-      codecs_[mirror_id] = ACMCodecDB::CreateCodecInstance(
-          send_codec, cng_nb_pltype_, cng_wb_pltype_, cng_swb_pltype_,
-          cng_fb_pltype_, red_enabled_, red_nb_pltype_);
-      if (!codecs_[mirror_id]) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Cannot Create the codec");
-        return -1;
-      }
-      mirror_codec_idx_[mirror_id] = mirror_id;
-    }
-
-    if (mirror_id != codec_id) {
-      codecs_[codec_id] = codecs_[mirror_id];
-      mirror_codec_idx_[codec_id] = mirror_id;
-    }
-
-    ACMGenericCodec* codec_ptr = codecs_[codec_id];
-    WebRtcACMCodecParams codec_params;
-
-    memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst));
-    codec_params.enable_vad = vad_enabled_;
-    codec_params.enable_dtx = dtx_enabled_;
-    codec_params.vad_mode = vad_mode_;
-    // Force initialization.
-    if (codec_ptr->InitEncoder(&codec_params, true) < 0) {
-      // Could not initialize the encoder.
-
-      // Check if already have a registered codec.
-      // Depending on that different messages are logged.
-      if (!current_encoder_) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Cannot Initialize the encoder No Encoder is registered");
-      } else {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Cannot Initialize the encoder, continue encoding with "
-                     "the previously registered codec");
-      }
-      return -1;
-    }
-
-    // Update states.
-    dtx_enabled_ = codec_params.enable_dtx;
-    vad_enabled_ = codec_params.enable_vad;
-    vad_mode_ = codec_params.vad_mode;
-
-    // Everything is fine so we can replace the previous codec with this one.
-    if (current_encoder_) {
-      // If we change codec we start fresh with RED.
-      // This is not strictly required by the standard.
-
-      if (codec_ptr->SetCopyRed(red_enabled_) < 0) {
-        // We tried to preserve the old red status, if failed, it means the
-        // red status has to be flipped.
-        red_enabled_ = !red_enabled_;
-      }
-
-      codec_ptr->SetVAD(&dtx_enabled_, &vad_enabled_, &vad_mode_);
-
-      if (!codec_ptr->HasInternalFEC()) {
-        codec_fec_enabled_ = false;
-      } else {
-        if (codec_ptr->SetFEC(codec_fec_enabled_) < 0) {
-          WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                       "Cannot set codec FEC");
-          return -1;
-        }
-      }
-    }
-
-    current_encoder_ = codecs_[codec_id];
-    DCHECK(current_encoder_);
-    memcpy(&send_codec_inst_, &send_codec, sizeof(CodecInst));
-    return 0;
-  } else {
-    // If codec is the same as already registered check if any parameters
-    // has changed compared to the current values.
-    // If any parameter is valid then apply it and record.
-    bool force_init = false;
-
-    if (mirror_id != codec_id) {
-      codecs_[codec_id] = codecs_[mirror_id];
-      mirror_codec_idx_[codec_id] = mirror_id;
-    }
-
-    // Check the payload type.
-    if (send_codec.pltype != send_codec_inst_.pltype) {
-      // At this point check if the given payload type is valid.
-      // Record it later when the sampling frequency is changed
-      // successfully.
-      if (!ACMCodecDB::ValidPayloadType(send_codec.pltype)) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Out of range payload type");
-        return -1;
-      }
-    }
-
-    // If there is a codec that ONE instance of codec supports multiple
-    // sampling frequencies, then we need to take care of it here.
-    // one such a codec is iSAC. Both WB and SWB are encoded and decoded
-    // with one iSAC instance. Therefore, we need to update the encoder
-    // frequency if required.
-    if (send_codec_inst_.plfreq != send_codec.plfreq) {
-      force_init = true;
-    }
-
-    // If packet size or number of channels has changed, we need to
-    // re-initialize the encoder.
-    if (send_codec_inst_.pacsize != send_codec.pacsize) {
-      force_init = true;
-    }
-    if (send_codec_inst_.channels != send_codec.channels) {
-      force_init = true;
-    }
-
-    if (force_init) {
-      WebRtcACMCodecParams codec_params;
-
-      memcpy(&(codec_params.codec_inst), &send_codec, sizeof(CodecInst));
-      codec_params.enable_vad = vad_enabled_;
-      codec_params.enable_dtx = dtx_enabled_;
-      codec_params.vad_mode = vad_mode_;
-
-      // Force initialization.
-      if (current_encoder_->InitEncoder(&codec_params, true) < 0) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Could not change the codec packet-size.");
-        return -1;
-      }
-
-      send_codec_inst_.plfreq = send_codec.plfreq;
-      send_codec_inst_.pacsize = send_codec.pacsize;
-      send_codec_inst_.channels = send_codec.channels;
-    }
-
-    // If the change of sampling frequency has been successful then
-    // we store the payload-type.
-    send_codec_inst_.pltype = send_codec.pltype;
-
-    // Check if a change in Rate is required.
-    if (send_codec.rate != send_codec_inst_.rate) {
-      if (codecs_[codec_id]->SetBitRate(send_codec.rate) < 0) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Could not change the codec rate.");
-        return -1;
-      }
-      send_codec_inst_.rate = send_codec.rate;
-    }
-
-    if (!codecs_[codec_id]->HasInternalFEC()) {
-      codec_fec_enabled_ = false;
-    } else {
-      if (codecs_[codec_id]->SetFEC(codec_fec_enabled_) < 0) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
-                     "Cannot set codec FEC");
-        return -1;
-      }
-    }
-
-    return 0;
-  }
-}
-
-int CodecManager::SendCodec(CodecInst* current_codec) const {
-  int dummy_id = 0;
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, dummy_id,
-               "SendCodec()");
-
-  if (!current_encoder_) {
-    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, dummy_id,
-                 "SendCodec Failed, no codec is registered");
-    return -1;
-  }
-  WebRtcACMCodecParams encoder_param;
-  current_encoder_->EncoderParams(&encoder_param);
-  encoder_param.codec_inst.pltype = send_codec_inst_.pltype;
-  memcpy(current_codec, &(encoder_param.codec_inst), sizeof(CodecInst));
-
-  return 0;
-}
-
-// Register possible receive codecs, can be called multiple times,
-// for codecs, CNG (NB, WB and SWB), DTMF, RED.
-int CodecManager::RegisterReceiveCodec(const CodecInst& codec) {
-  if (codec.channels > 2 || codec.channels < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "Unsupported number of channels, %d.", codec.channels);
-    return -1;
-  }
-
-  int mirror_id;
-  int codec_id = ACMCodecDB::ReceiverCodecNumber(codec, &mirror_id);
-
-  if (codec_id < 0 || codec_id >= ACMCodecDB::kNumCodecs) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "Wrong codec params to be registered as receive codec");
-    return -1;
-  }
-
-  // Check if the payload-type is valid.
-  if (!ACMCodecDB::ValidPayloadType(codec.pltype)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "Invalid payload-type %d for %s.", codec.pltype, codec.plname);
-    return -1;
-  }
-
-  AudioDecoder* decoder = NULL;
-  // Get |decoder| associated with |codec|. |decoder| can be NULL if |codec|
-  // does not own its decoder.
-  if (GetAudioDecoder(codec, codec_id, mirror_id, &decoder) < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "Wrong codec params to be registered as receive codec");
-    return -1;
-  }
-  uint8_t payload_type = static_cast<uint8_t>(codec.pltype);
-  return acm_->RegisterDecoder(codec_id, payload_type, codec.channels, decoder);
-}
-
-bool CodecManager::SetCopyRed(bool enable) {
-  if (enable && codec_fec_enabled_) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
-                 "Codec internal FEC and RED cannot be co-enabled.");
-    return false;
-  }
-  if (current_encoder_ && current_encoder_->SetCopyRed(enable) < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "SetCopyRed failed");
-    return false;
-  }
-  red_enabled_ = enable;
-  return true;
-}
-
-int CodecManager::SetVAD(bool enable_dtx, bool enable_vad, ACMVADMode mode) {
-  // Sanity check of the mode.
-  if ((mode != VADNormal) && (mode != VADLowBitrate) && (mode != VADAggr) &&
-      (mode != VADVeryAggr)) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "Invalid VAD Mode %d, no change is made to VAD/DTX status",
-                 mode);
-    return -1;
-  }
-
-  // Check that the send codec is mono. We don't support VAD/DTX for stereo
-  // sending.
-  if ((enable_dtx || enable_vad) && stereo_send_) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "VAD/DTX not supported for stereo sending");
-    dtx_enabled_ = false;
-    vad_enabled_ = false;
-    vad_mode_ = mode;
-    return -1;
-  }
-
-  // Store VAD/DTX settings. Values can be changed in the call to "SetVAD"
-  // below.
-  dtx_enabled_ = enable_dtx;
-  vad_enabled_ = enable_vad;
-  vad_mode_ = mode;
-
-  // If a send codec is registered, set VAD/DTX for the codec.
-  if (current_encoder_ &&
-      current_encoder_->SetVAD(&dtx_enabled_, &vad_enabled_, &vad_mode_) < 0) {
-    // SetVAD failed.
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "SetVAD failed");
-    vad_enabled_ = false;
-    dtx_enabled_ = false;
-    return -1;
-  }
-  return 0;
-}
-
-void CodecManager::VAD(bool* dtx_enabled,
-                       bool* vad_enabled,
-                       ACMVADMode* mode) const {
-  *dtx_enabled = dtx_enabled_;
-  *vad_enabled = vad_enabled_;
-  *mode = vad_mode_;
-}
-
-int CodecManager::SetCodecFEC(bool enable_codec_fec) {
-  if (enable_codec_fec == true && red_enabled_ == true) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
-                 "Codec internal FEC and RED cannot be co-enabled.");
-    return -1;
-  }
-
-  // Set codec FEC.
-  if (current_encoder_ && current_encoder_->SetFEC(enable_codec_fec) < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                 "Set codec internal FEC failed.");
-    return -1;
-  }
-  codec_fec_enabled_ = enable_codec_fec;
-  return 0;
-}
-
-void CodecManager::SetCngPayloadType(int sample_rate_hz, int payload_type) {
-  for (auto* codec : codecs_) {
-    if (codec) {
-      codec->SetCngPt(sample_rate_hz, payload_type);
-    }
-  }
-}
-
-void CodecManager::SetRedPayloadType(int sample_rate_hz, int payload_type) {
-  for (auto* codec : codecs_) {
-    if (codec) {
-      codec->SetRedPt(sample_rate_hz, payload_type);
-    }
-  }
-}
-
-int CodecManager::GetAudioDecoder(const CodecInst& codec,
-                                  int codec_id,
-                                  int mirror_id,
-                                  AudioDecoder** decoder) {
-  if (ACMCodecDB::OwnsDecoder(codec_id)) {
-    // This codec has to own its own decoder. Therefore, it should create the
-    // corresponding AudioDecoder class and insert it into NetEq. If the codec
-    // does not exist create it.
-    //
-    // TODO(turajs): this part of the code is common with RegisterSendCodec(),
-    //               make a method for it.
-    if (codecs_[mirror_id] == NULL) {
-      codecs_[mirror_id] = ACMCodecDB::CreateCodecInstance(
-          codec, cng_nb_pltype_, cng_wb_pltype_, cng_swb_pltype_,
-          cng_fb_pltype_, red_enabled_, red_nb_pltype_);
-      if (codecs_[mirror_id] == NULL) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-                     "Cannot Create the codec");
-        return -1;
-      }
-      mirror_codec_idx_[mirror_id] = mirror_id;
-    }
-
-    if (mirror_id != codec_id) {
-      codecs_[codec_id] = codecs_[mirror_id];
-      mirror_codec_idx_[codec_id] = mirror_id;
-    }
-    *decoder = codecs_[codec_id]->Decoder();
-    if (!*decoder) {
-      assert(false);
-      return -1;
-    }
-  } else {
-    *decoder = NULL;
-  }
-
-  return 0;
-}
-
-}  // namespace acm2
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.h
deleted file mode 100644
index 8d5350d71e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class AudioDecoder;
-
-namespace acm2 {
-
-class ACMGenericCodec;
-class AudioCodingModuleImpl;
-
-class CodecManager final {
- public:
-  explicit CodecManager(AudioCodingModuleImpl* acm);
-  ~CodecManager();
-
-  int RegisterSendCodec(const CodecInst& send_codec);
-
-  int SendCodec(CodecInst* current_codec) const;
-
-  int RegisterReceiveCodec(const CodecInst& receive_codec);
-
-  bool SetCopyRed(bool enable);
-
-  int SetVAD(bool enable_dtx, bool enable_vad, ACMVADMode mode);
-
-  void VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const;
-
-  int SetCodecFEC(bool enable_codec_fec);
-
-  bool stereo_send() const { return stereo_send_; }
-
-  bool red_enabled() const { return red_enabled_; }
-
-  bool codec_fec_enabled() const { return codec_fec_enabled_; }
-
-  ACMGenericCodec* current_encoder() { return current_encoder_; }
-
-  const ACMGenericCodec* current_encoder() const { return current_encoder_; }
-
- private:
-  void SetCngPayloadType(int sample_rate_hz, int payload_type);
-
-  void SetRedPayloadType(int sample_rate_hz, int payload_type);
-
-  // Get a pointer to AudioDecoder of the given codec. For some codecs, e.g.
-  // iSAC, encoding and decoding have to be performed on a shared
-  // codec-instance. By calling this method, we get the codec-instance that ACM
-  // owns, then pass that to NetEq. This way, we perform both encoding and
-  // decoding on the same codec-instance. Furthermore, ACM would have control
-  // over decoder functionality if required. If |codec| does not share an
-  // instance between encoder and decoder, the |*decoder| is set NULL.
-  // The field ACMCodecDB::CodecSettings.owns_decoder indicates that if a
-  // codec owns the decoder-instance. For such codecs |*decoder| should be a
-  // valid pointer, otherwise it will be NULL.
-  int GetAudioDecoder(const CodecInst& codec,
-                      int codec_id,
-                      int mirror_id,
-                      AudioDecoder** decoder);
-
-  AudioCodingModuleImpl* acm_;
-  rtc::ThreadChecker thread_checker_;
-  uint8_t cng_nb_pltype_;
-  uint8_t cng_wb_pltype_;
-  uint8_t cng_swb_pltype_;
-  uint8_t cng_fb_pltype_;
-  uint8_t red_nb_pltype_;
-  bool stereo_send_;
-  bool vad_enabled_;
-  bool dtx_enabled_;
-  ACMVADMode vad_mode_;
-  ACMGenericCodec* current_encoder_;
-  CodecInst send_codec_inst_;
-  bool red_enabled_;
-  bool codec_fec_enabled_;
-  ACMGenericCodec* codecs_[ACMCodecDB::kMaxNumCodecs];
-  int mirror_codec_idx_[ACMCodecDB::kMaxNumCodecs];
-
-  DISALLOW_COPY_AND_ASSIGN(CodecManager);
-};
-
-}  // namespace acm2
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
deleted file mode 100644
index ee7a2f1340..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
+++ /dev/null
@@ -1,175 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-
-#include <map>
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-///////////////////////////////////////////////////////////////////////////
-// enum AudioPlayoutMode
-// An enumerator for different playout modes.
-//
-// -voice       : This is the standard mode for VoIP calls. The trade-off
-//                between low delay and jitter robustness is optimized
-//                for high-quality two-way communication.
-//                NetEQs packet loss concealment and signal processing
-//                capabilities are fully employed.
-// -fax         : The fax mode is optimized for decodability of fax signals
-//                rather than for perceived audio quality. When this mode
-//                is selected, NetEQ will do as few delay changes as possible,
-//                trying to maintain a high and constant delay. Meanwhile,
-//                the packet loss concealment efforts are reduced.
-//
-// -streaming   : In the case of one-way communication such as passive
-//                conference participant, a webinar, or a streaming application,
-//                this mode can be used to improve the jitter robustness at
-//                the cost of increased delay.
-// -off         : Turns off most of NetEQ's features. Stuffs zeros for lost
-//                packets and during buffer increases.
-//
-enum AudioPlayoutMode {
-  voice = 0,
-  fax = 1,
-  streaming = 2,
-  off = 3,
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMSpeechType
-// An enumerator for possible labels of a decoded frame.
-//
-// -normal      : a normal speech frame. If VAD is enabled on the
-//                incoming stream this label indicate that the
-//                frame is active.
-// -PLC         : a PLC frame. The corresponding packet was lost
-//                and this frame generated by PLC techniques.
-// -CNG         : the frame is comfort noise. This happens if VAD
-//                is enabled at the sender and we have received
-//                SID.
-// -PLCCNG      : PLC will fade to comfort noise if the duration
-//                of PLC is long. This labels such a case.
-// -VADPassive  : the VAD at the receiver recognizes this frame as
-//                passive.
-//
-enum ACMSpeechType {
-  normal = 0,
-  PLC = 1,
-  CNG = 2,
-  PLCCNG = 3,
-  VADPassive = 4
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMVADMode
-// An enumerator for aggressiveness of VAD
-// -VADNormal                : least aggressive mode.
-// -VADLowBitrate            : more aggressive than "VADNormal" to save on
-//                             bit-rate.
-// -VADAggr                  : an aggressive mode.
-// -VADVeryAggr              : the most agressive mode.
-//
-enum ACMVADMode {
-  VADNormal = 0,
-  VADLowBitrate = 1,
-  VADAggr = 2,
-  VADVeryAggr = 3
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMCountries
-// An enumerator for countries, used when enabling CPT for a specific country.
-//
-enum ACMCountries {
-  ACMDisableCountryDetection = -1, // disable CPT detection
-  ACMUSA = 0,
-  ACMJapan,
-  ACMCanada,
-  ACMFrance,
-  ACMGermany,
-  ACMAustria,
-  ACMBelgium,
-  ACMUK,
-  ACMCzech,
-  ACMDenmark,
-  ACMFinland,
-  ACMGreece,
-  ACMHungary,
-  ACMIceland,
-  ACMIreland,
-  ACMItaly,
-  ACMLuxembourg,
-  ACMMexico,
-  ACMNorway,
-  ACMPoland,
-  ACMPortugal,
-  ACMSpain,
-  ACMSweden,
-  ACMTurkey,
-  ACMChina,
-  ACMHongkong,
-  ACMTaiwan,
-  ACMKorea,
-  ACMSingapore,
-  ACMNonStandard1
-// non-standard countries
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMAMRPackingFormat
-// An enumerator for different bit-packing format of AMR codec according to
-// RFC 3267.
-//
-// -AMRUndefined           : undefined.
-// -AMRBandwidthEfficient  : bandwidth-efficient mode.
-// -AMROctetAlligned       : Octet-alligned mode.
-// -AMRFileStorage         : file-storage mode.
-//
-enum ACMAMRPackingFormat {
-  AMRUndefined = -1,
-  AMRBandwidthEfficient = 0,
-  AMROctetAlligned = 1,
-  AMRFileStorage = 2
-};
-
-///////////////////////////////////////////////////////////////////////////
-//
-// Enumeration of background noise mode a mapping from NetEQ interface.
-//
-// -On                  : default "normal" behavior with eternal noise
-// -Fade                : noise fades to zero after some time
-// -Off                 : background noise is always zero
-//
-enum ACMBackgroundNoiseMode {
-  On,
-  Fade,
-  Off
-};
-
-///////////////////////////////////////////////////////////////////////////
-//
-// Enumeration of Opus mode for intended application.
-//
-// kVoip              : optimized for voice signals.
-// kAudio             : optimized for non-voice signals like music.
-//
-enum OpusApplicationMode {
- kVoip = 0,
- kAudio = 1,
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
deleted file mode 100644
index 3d0d312f50..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ /dev/null
@@ -1,175 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-
-#include <assert.h>
-#include <math.h>
-
-#include <iostream>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-namespace {
-
-double FrameRms(AudioFrame& frame) {
-  int samples = frame.num_channels_ * frame.samples_per_channel_;
-  double rms = 0;
-  for (int n = 0; n < samples; ++n)
-    rms += frame.data_[n] * frame.data_[n];
-  rms /= samples;
-  rms = sqrt(rms);
-  return rms;
-}
-
-}
-
-class InitialPlayoutDelayTest : public ::testing::Test {
- protected:
-  InitialPlayoutDelayTest()
-      : acm_a_(AudioCodingModule::Create(0)),
-        acm_b_(AudioCodingModule::Create(1)),
-        channel_a2b_(NULL) {}
-
-  ~InitialPlayoutDelayTest() {
-    if (channel_a2b_ != NULL) {
-      delete channel_a2b_;
-      channel_a2b_ = NULL;
-    }
-  }
-
-  void SetUp() {
-    ASSERT_TRUE(acm_a_.get() != NULL);
-    ASSERT_TRUE(acm_b_.get() != NULL);
-
-    EXPECT_EQ(0, acm_b_->InitializeReceiver());
-    EXPECT_EQ(0, acm_a_->InitializeReceiver());
-
-    // Register all L16 codecs in receiver.
-    CodecInst codec;
-    const int kFsHz[3] = { 8000, 16000, 32000 };
-    const int kChannels[2] = { 1, 2 };
-    for (int n = 0; n < 3; ++n) {
-      for (int k = 0; k < 2; ++k) {
-        AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
-        acm_b_->RegisterReceiveCodec(codec);
-      }
-    }
-
-    // Create and connect the channel
-    channel_a2b_ = new Channel;
-    acm_a_->RegisterTransportCallback(channel_a2b_);
-    channel_a2b_->RegisterReceiverACM(acm_b_.get());
-  }
-
-  void NbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 1);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbMono() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 1);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
-  void NbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 8000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void WbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 16000, 2);
-    codec.pacsize = codec.plfreq * 30 / 1000;  // 30 ms packets.
-    Run(codec, 1000);
-  }
-
-  void SwbStereo() {
-    CodecInst codec;
-    AudioCodingModule::Codec("L16", &codec, 32000, 2);
-    codec.pacsize = codec.plfreq * 10 / 1000;  // 10 ms packets.
-    Run(codec, 400);  // Memory constraints limit the buffer at <500 ms.
-  }
-
- private:
-  void Run(CodecInst codec, int initial_delay_ms) {
-    AudioFrame in_audio_frame;
-    AudioFrame out_audio_frame;
-    int num_frames = 0;
-    const int kAmp = 10000;
-    in_audio_frame.sample_rate_hz_ = codec.plfreq;
-    in_audio_frame.num_channels_ = codec.channels;
-    in_audio_frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
-    int samples = in_audio_frame.num_channels_ *
-        in_audio_frame.samples_per_channel_;
-    for (int n = 0; n < samples; ++n) {
-      in_audio_frame.data_[n] = kAmp;
-    }
-
-    uint32_t timestamp = 0;
-    double rms = 0;
-    ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
-    acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
-    while (rms < kAmp / 2) {
-      in_audio_frame.timestamp_ = timestamp;
-      timestamp += in_audio_frame.samples_per_channel_;
-      ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
-      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
-      rms = FrameRms(out_audio_frame);
-      ++num_frames;
-    }
-
-    ASSERT_GE(num_frames * 10, initial_delay_ms);
-    ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
-  }
-
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
-  Channel* channel_a2b_;
-};
-
-TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
-
-TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
-
-TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/OWNERS
index 072e754998..bbffda7e49 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/OWNERS
@@ -1,8 +1,3 @@
-henrik.lundin@webrtc.org
-tina.legrand@webrtc.org
-turaj@webrtc.org
-minyue@webrtc.org
-
 per-file *.isolate=kjellander@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.cc
index 6acd778a23..1c36fa8c61 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.cc
@@ -14,44 +14,57 @@
 
 namespace webrtc {
 
-Accelerate::ReturnCodes Accelerate::Process(
-    const int16_t* input,
-    size_t input_length,
-    AudioMultiVector* output,
-    int16_t* length_change_samples) {
+Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
+                                            size_t input_length,
+                                            bool fast_accelerate,
+                                            AudioMultiVector* output,
+                                            size_t* length_change_samples) {
   // Input length must be (almost) 30 ms.
-  static const int k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
-  if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ <
-      (2 * k15ms - 1) * fs_mult_) {
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
     // Length of input data too short to do accelerate. Simply move all data
     // from input to output.
     output->PushBackInterleaved(input, input_length);
     return kError;
   }
-  return TimeStretch::Process(input, input_length, output,
+  return TimeStretch::Process(input, input_length, fast_accelerate, output,
                               length_change_samples);
 }
 
 void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
                                                int16_t* best_correlation,
-                                               int* /*peak_index*/) const {
+                                               size_t* /*peak_index*/) const {
   // When the signal does not contain any active speech, the correlation does
   // not matter. Simply set it to zero.
   *best_correlation = 0;
 }
 
 Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
-    const int16_t* input, size_t input_length, size_t peak_index,
-    int16_t best_correlation, bool active_speech,
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool fast_mode,
     AudioMultiVector* output) const {
   // Check for strong correlation or passive speech.
-  if ((best_correlation > kCorrelationThreshold) || !active_speech) {
+  // Use 8192 (0.5 in Q14) in fast mode.
+  const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
+  if ((best_correlation > correlation_threshold) || !active_speech) {
     // Do accelerate operation by overlap add.
 
     // Pre-calculate common multiplication with |fs_mult_|.
     // 120 corresponds to 15 ms.
     size_t fs_mult_120 = fs_mult_ * 120;
 
+    if (fast_mode) {
+      // Fit as many multiples of |peak_index| as possible in fs_mult_120.
+      // TODO(henrik.lundin) Consider finding multiple correlation peaks and
+      // pick the one with the longest correlation lag in this case.
+      peak_index = (fs_mult_120 / peak_index) * peak_index;
+    }
+
     assert(fs_mult_120 >= peak_index);  // Should be handled in Process().
     // Copy first part; 0 to 15 ms.
     output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.h
index 6e3aa4634f..f66bc8ed34 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/accelerate.h
@@ -34,24 +34,25 @@ class Accelerate : public TimeStretch {
       : TimeStretch(sample_rate_hz, num_channels, background_noise) {
   }
 
-  virtual ~Accelerate() {}
-
   // This method performs the actual Accelerate operation. The samples are
   // read from |input|, of length |input_length| elements, and are written to
   // |output|. The number of samples removed through time-stretching is
   // is provided in the output |length_change_samples|. The method returns
-  // the outcome of the operation as an enumerator value.
+  // the outcome of the operation as an enumerator value. If |fast_accelerate|
+  // is true, the algorithm will relax the requirements on finding strong
+  // correlations, and may remove multiple pitch periods if possible.
   ReturnCodes Process(const int16_t* input,
                       size_t input_length,
+                      bool fast_accelerate,
                       AudioMultiVector* output,
-                      int16_t* length_change_samples);
+                      size_t* length_change_samples);
 
  protected:
   // Sets the parameters |best_correlation| and |peak_index| to suitable
   // values when the signal contains no active speech.
   void SetParametersForPassiveSpeech(size_t len,
                                      int16_t* best_correlation,
-                                     int* peak_index) const override;
+                                     size_t* peak_index) const override;
 
   // Checks the criteria for performing the time-stretching operation and,
   // if possible, performs the time-stretching.
@@ -60,10 +61,11 @@ class Accelerate : public TimeStretch {
                                       size_t peak_index,
                                       int16_t best_correlation,
                                       bool active_speech,
+                                      bool fast_mode,
                                       AudioMultiVector* output) const override;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(Accelerate);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Accelerate);
 };
 
 struct AccelerateFactory {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc
index 78e0ae76ed..2200c9a092 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc
@@ -68,4 +68,8 @@ bool AudioClassifier::Analysis(const int16_t* input,
   return is_music_;
 }
 
+bool AudioClassifier::is_music() const {
+  return is_music_;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.h
index 2812ea25c7..b32f9d5f8f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.h
@@ -37,7 +37,7 @@ class AudioClassifier {
   bool Analysis(const int16_t* input, int input_length, int channels);
 
   // Gets the current classification : true = music, false = speech.
-  virtual bool is_music() const { return is_music_; }
+  virtual bool is_music() const;
 
   // Gets the current music probability.
   float music_probability() const { return music_probability_; }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index ce24e4a7e5..d800cc7dbe 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -11,383 +11,107 @@
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 
 #include <assert.h>
-#include <string.h>  // memmove
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
 #endif
 #ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 #endif
 #ifdef WEBRTC_CODEC_ISACFX
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
 #endif
 #ifdef WEBRTC_CODEC_ISAC
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 #endif
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 
 namespace webrtc {
 
-// PCMu
-int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
-                                     size_t encoded_len,
-                                     int sample_rate_hz,
-                                     int16_t* decoded,
-                                     SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 8000);
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len),
-                                   decoded, &temp_type);
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
-                                     size_t encoded_len) const {
-  // One encoded byte per sample per channel.
-  return static_cast<int>(encoded_len / Channels());
-}
-
-// PCMa
-int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
-                                     size_t encoded_len,
-                                     int sample_rate_hz,
-                                     int16_t* decoded,
-                                     SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 8000);
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len),
-                                   decoded, &temp_type);
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
-                                     size_t encoded_len) const {
-  // One encoded byte per sample per channel.
-  return static_cast<int>(encoded_len / Channels());
-}
-
-// PCM16B
-#ifdef WEBRTC_CODEC_PCM16
-AudioDecoderPcm16B::AudioDecoderPcm16B() {}
-
-int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
-                                       size_t encoded_len,
-                                       int sample_rate_hz,
-                                       int16_t* decoded,
-                                       SpeechType* speech_type) {
-  DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
-         sample_rate_hz == 32000 || sample_rate_hz == 48000)
-      << "Unsupported sample rate " << sample_rate_hz;
-  int16_t ret =
-      WebRtcPcm16b_Decode(encoded, static_cast<int16_t>(encoded_len), decoded);
-  *speech_type = ConvertSpeechType(1);
-  return ret;
-}
-
-int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
-                                       size_t encoded_len) const {
-  // Two encoded byte per sample per channel.
-  return static_cast<int>(encoded_len / (2 * Channels()));
-}
-
-AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels)
-    : channels_(num_channels) {
-  DCHECK(num_channels > 0);
-}
-#endif
-
-// iLBC
-#ifdef WEBRTC_CODEC_ILBC
-AudioDecoderIlbc::AudioDecoderIlbc() {
-  WebRtcIlbcfix_DecoderCreate(&dec_state_);
-}
-
-AudioDecoderIlbc::~AudioDecoderIlbc() {
-  WebRtcIlbcfix_DecoderFree(dec_state_);
-}
-
-int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
-                                     size_t encoded_len,
-                                     int sample_rate_hz,
-                                     int16_t* decoded,
-                                     SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 8000);
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
-                                     static_cast<int16_t>(encoded_len), decoded,
-                                     &temp_type);
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
-  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
-}
-
-int AudioDecoderIlbc::Init() {
-  return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
-}
-#endif
-
-// G.722
-#ifdef WEBRTC_CODEC_G722
-AudioDecoderG722::AudioDecoderG722() {
-  WebRtcG722_CreateDecoder(&dec_state_);
-}
-
-AudioDecoderG722::~AudioDecoderG722() {
-  WebRtcG722_FreeDecoder(dec_state_);
-}
-
-int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
-                                     size_t encoded_len,
-                                     int sample_rate_hz,
-                                     int16_t* decoded,
-                                     SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 16000);
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret =
-      WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len),
-                        decoded, &temp_type);
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-int AudioDecoderG722::Init() {
-  return WebRtcG722_DecoderInit(dec_state_);
-}
-
-int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
-                                     size_t encoded_len) const {
-  // 1/2 encoded byte per sample per channel.
-  return static_cast<int>(2 * encoded_len / Channels());
-}
-
-AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
-  WebRtcG722_CreateDecoder(&dec_state_left_);
-  WebRtcG722_CreateDecoder(&dec_state_right_);
-}
-
-AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
-  WebRtcG722_FreeDecoder(dec_state_left_);
-  WebRtcG722_FreeDecoder(dec_state_right_);
-}
-
-int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
-                                           size_t encoded_len,
-                                           int sample_rate_hz,
-                                           int16_t* decoded,
-                                           SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 16000);
-  int16_t temp_type = 1;  // Default is speech.
-  // De-interleave the bit-stream into two separate payloads.
-  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
-  SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
-  // Decode left and right.
-  int16_t ret = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
-                                  static_cast<int16_t>(encoded_len / 2),
-                                  decoded, &temp_type);
-  if (ret >= 0) {
-    int decoded_len = ret;
-    ret = WebRtcG722_Decode(dec_state_right_,
-                            &encoded_deinterleaved[encoded_len / 2],
-                            static_cast<int16_t>(encoded_len / 2),
-                            &decoded[decoded_len], &temp_type);
-    if (ret == decoded_len) {
-      decoded_len += ret;
-      // Interleave output.
-      for (int k = decoded_len / 2; k < decoded_len; k++) {
-          int16_t temp = decoded[k];
-          memmove(&decoded[2 * k - decoded_len + 2],
-                  &decoded[2 * k - decoded_len + 1],
-                  (decoded_len - k - 1) * sizeof(int16_t));
-          decoded[2 * k - decoded_len + 1] = temp;
-      }
-      ret = decoded_len;  // Return total number of samples.
-    }
-  }
-  *speech_type = ConvertSpeechType(temp_type);
-  delete [] encoded_deinterleaved;
-  return ret;
-}
-
-int AudioDecoderG722Stereo::Init() {
-  int r = WebRtcG722_DecoderInit(dec_state_left_);
-  if (r != 0)
-    return r;
-  return WebRtcG722_DecoderInit(dec_state_right_);
-}
-
-// Split the stereo packet and place left and right channel after each other
-// in the output array.
-void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
-                                               size_t encoded_len,
-                                               uint8_t* encoded_deinterleaved) {
-  assert(encoded);
-  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
-  // where "lx" is 4 bits representing left sample number x, and "rx" right
-  // sample. Two samples fit in one byte, represented with |...|.
-  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
-    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
-    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
-    encoded_deinterleaved[i + 1] = right_byte;
-  }
-
-  // Move one byte representing right channel each loop, and place it at the
-  // end of the bytestream vector. After looping the data is reordered to:
-  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
-  // where N is the total number of samples.
-  for (size_t i = 0; i < encoded_len / 2; i++) {
-    uint8_t right_byte = encoded_deinterleaved[i + 1];
-    memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
-            encoded_len - i - 2);
-    encoded_deinterleaved[encoded_len - 1] = right_byte;
-  }
-}
-#endif
-
-// Opus
-#ifdef WEBRTC_CODEC_OPUS
-AudioDecoderOpus::AudioDecoderOpus(int num_channels) : channels_(num_channels) {
-  DCHECK(num_channels == 1 || num_channels == 2);
-  WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
-}
-
-AudioDecoderOpus::~AudioDecoderOpus() {
-  WebRtcOpus_DecoderFree(dec_state_);
-}
-
-int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
-                                     size_t encoded_len,
-                                     int sample_rate_hz,
-                                     int16_t* decoded,
-                                     SpeechType* speech_type) {
-  DCHECK_EQ(sample_rate_hz, 48000);
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
-                                  static_cast<int16_t>(encoded_len), decoded,
-                                  &temp_type);
-  if (ret > 0)
-    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
-                                              size_t encoded_len,
-                                              int sample_rate_hz,
-                                              int16_t* decoded,
-                                              SpeechType* speech_type) {
-  if (!PacketHasFec(encoded, encoded_len)) {
-    // This packet is a RED packet.
-    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
-                          speech_type);
-  }
-
-  DCHECK_EQ(sample_rate_hz, 48000);
-  int16_t temp_type = 1;  // Default is speech.
-  int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
-                                     static_cast<int16_t>(encoded_len), decoded,
-                                     &temp_type);
-  if (ret > 0)
-    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
-  *speech_type = ConvertSpeechType(temp_type);
-  return ret;
-}
-
-int AudioDecoderOpus::Init() {
-  return WebRtcOpus_DecoderInit(dec_state_);
-}
-
-int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
-                                     size_t encoded_len) const {
-  return WebRtcOpus_DurationEst(dec_state_,
-                                encoded, static_cast<int>(encoded_len));
-}
-
-int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
-                                              size_t encoded_len) const {
-  if (!PacketHasFec(encoded, encoded_len)) {
-    // This packet is a RED packet.
-    return PacketDuration(encoded, encoded_len);
-  }
-
-  return WebRtcOpus_FecDurationEst(encoded, static_cast<int>(encoded_len));
-}
-
-bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
-                                    size_t encoded_len) const {
-  int fec;
-  fec = WebRtcOpus_PacketHasFec(encoded, static_cast<int>(encoded_len));
-  return (fec == 1);
-}
-#endif
-
 AudioDecoderCng::AudioDecoderCng() {
-  CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
+  RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
+  WebRtcCng_InitDec(dec_state_);
 }
 
 AudioDecoderCng::~AudioDecoderCng() {
   WebRtcCng_FreeDec(dec_state_);
 }
 
-int AudioDecoderCng::Init() {
-  return WebRtcCng_InitDec(dec_state_);
+void AudioDecoderCng::Reset() {
+  WebRtcCng_InitDec(dec_state_);
+}
+
+int AudioDecoderCng::IncomingPacket(const uint8_t* payload,
+                                    size_t payload_len,
+                                    uint16_t rtp_sequence_number,
+                                    uint32_t rtp_timestamp,
+                                    uint32_t arrival_timestamp) {
+  return -1;
+}
+
+CNG_dec_inst* AudioDecoderCng::CngDecoderInstance() {
+  return dec_state_;
+}
+
+size_t AudioDecoderCng::Channels() const {
+  return 1;
+}
+
+int AudioDecoderCng::DecodeInternal(const uint8_t* encoded,
+                                    size_t encoded_len,
+                                    int sample_rate_hz,
+                                    int16_t* decoded,
+                                    SpeechType* speech_type) {
+  return -1;
 }
 
 bool CodecSupported(NetEqDecoder codec_type) {
   switch (codec_type) {
-    case kDecoderPCMu:
-    case kDecoderPCMa:
-    case kDecoderPCMu_2ch:
-    case kDecoderPCMa_2ch:
+    case NetEqDecoder::kDecoderPCMu:
+    case NetEqDecoder::kDecoderPCMa:
+    case NetEqDecoder::kDecoderPCMu_2ch:
+    case NetEqDecoder::kDecoderPCMa_2ch:
 #ifdef WEBRTC_CODEC_ILBC
-    case kDecoderILBC:
+    case NetEqDecoder::kDecoderILBC:
 #endif
 #if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
-    case kDecoderISAC:
+    case NetEqDecoder::kDecoderISAC:
 #endif
 #ifdef WEBRTC_CODEC_ISAC
-    case kDecoderISACswb:
-    case kDecoderISACfb:
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-    case kDecoderPCM16B:
-    case kDecoderPCM16Bwb:
-    case kDecoderPCM16Bswb32kHz:
-    case kDecoderPCM16Bswb48kHz:
-    case kDecoderPCM16B_2ch:
-    case kDecoderPCM16Bwb_2ch:
-    case kDecoderPCM16Bswb32kHz_2ch:
-    case kDecoderPCM16Bswb48kHz_2ch:
-    case kDecoderPCM16B_5ch:
+    case NetEqDecoder::kDecoderISACswb:
 #endif
+    case NetEqDecoder::kDecoderPCM16B:
+    case NetEqDecoder::kDecoderPCM16Bwb:
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz:
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz:
+    case NetEqDecoder::kDecoderPCM16B_2ch:
+    case NetEqDecoder::kDecoderPCM16Bwb_2ch:
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
+    case NetEqDecoder::kDecoderPCM16B_5ch:
 #ifdef WEBRTC_CODEC_G722
-    case kDecoderG722:
-    case kDecoderG722_2ch:
+    case NetEqDecoder::kDecoderG722:
+    case NetEqDecoder::kDecoderG722_2ch:
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-    case kDecoderOpus:
-    case kDecoderOpus_2ch:
+    case NetEqDecoder::kDecoderOpus:
+    case NetEqDecoder::kDecoderOpus_2ch:
 #endif
-    case kDecoderRED:
-    case kDecoderAVT:
-    case kDecoderCNGnb:
-    case kDecoderCNGwb:
-    case kDecoderCNGswb32kHz:
-    case kDecoderCNGswb48kHz:
-    case kDecoderArbitrary: {
+    case NetEqDecoder::kDecoderRED:
+    case NetEqDecoder::kDecoderAVT:
+    case NetEqDecoder::kDecoderCNGnb:
+    case NetEqDecoder::kDecoderCNGwb:
+    case NetEqDecoder::kDecoderCNGswb32kHz:
+    case NetEqDecoder::kDecoderCNGswb48kHz:
+    case NetEqDecoder::kDecoderArbitrary: {
       return true;
     }
     default: {
@@ -398,59 +122,50 @@ bool CodecSupported(NetEqDecoder codec_type) {
 
 int CodecSampleRateHz(NetEqDecoder codec_type) {
   switch (codec_type) {
-    case kDecoderPCMu:
-    case kDecoderPCMa:
-    case kDecoderPCMu_2ch:
-    case kDecoderPCMa_2ch:
+    case NetEqDecoder::kDecoderPCMu:
+    case NetEqDecoder::kDecoderPCMa:
+    case NetEqDecoder::kDecoderPCMu_2ch:
+    case NetEqDecoder::kDecoderPCMa_2ch:
 #ifdef WEBRTC_CODEC_ILBC
-    case kDecoderILBC:
+    case NetEqDecoder::kDecoderILBC:
 #endif
-#ifdef WEBRTC_CODEC_PCM16
-    case kDecoderPCM16B:
-    case kDecoderPCM16B_2ch:
-    case kDecoderPCM16B_5ch:
-#endif
-    case kDecoderCNGnb: {
+    case NetEqDecoder::kDecoderPCM16B:
+    case NetEqDecoder::kDecoderPCM16B_2ch:
+    case NetEqDecoder::kDecoderPCM16B_5ch:
+    case NetEqDecoder::kDecoderCNGnb: {
       return 8000;
     }
 #if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
-    case kDecoderISAC:
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-    case kDecoderPCM16Bwb:
-    case kDecoderPCM16Bwb_2ch:
+    case NetEqDecoder::kDecoderISAC:
 #endif
+    case NetEqDecoder::kDecoderPCM16Bwb:
+    case NetEqDecoder::kDecoderPCM16Bwb_2ch:
 #ifdef WEBRTC_CODEC_G722
-    case kDecoderG722:
-    case kDecoderG722_2ch:
+    case NetEqDecoder::kDecoderG722:
+    case NetEqDecoder::kDecoderG722_2ch:
 #endif
-    case kDecoderCNGwb: {
+    case NetEqDecoder::kDecoderCNGwb: {
       return 16000;
     }
 #ifdef WEBRTC_CODEC_ISAC
-    case kDecoderISACswb:
-    case kDecoderISACfb:
+    case NetEqDecoder::kDecoderISACswb:
 #endif
-#ifdef WEBRTC_CODEC_PCM16
-    case kDecoderPCM16Bswb32kHz:
-    case kDecoderPCM16Bswb32kHz_2ch:
-#endif
-    case kDecoderCNGswb32kHz: {
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz:
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
+    case NetEqDecoder::kDecoderCNGswb32kHz: {
       return 32000;
     }
-#ifdef WEBRTC_CODEC_PCM16
-    case kDecoderPCM16Bswb48kHz:
-    case kDecoderPCM16Bswb48kHz_2ch: {
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz:
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch: {
       return 48000;
     }
-#endif
 #ifdef WEBRTC_CODEC_OPUS
-    case kDecoderOpus:
-    case kDecoderOpus_2ch: {
+    case NetEqDecoder::kDecoderOpus:
+    case NetEqDecoder::kDecoderOpus_2ch: {
       return 48000;
     }
 #endif
-    case kDecoderCNGswb48kHz: {
+    case NetEqDecoder::kDecoderCNGswb48kHz: {
       // TODO(tlegrand): Remove limitation once ACM has full 48 kHz support.
       return 32000;
     }
@@ -465,70 +180,58 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
     return NULL;
   }
   switch (codec_type) {
-    case kDecoderPCMu:
-      return new AudioDecoderPcmU;
-    case kDecoderPCMa:
-      return new AudioDecoderPcmA;
-    case kDecoderPCMu_2ch:
-      return new AudioDecoderPcmUMultiCh(2);
-    case kDecoderPCMa_2ch:
-      return new AudioDecoderPcmAMultiCh(2);
+    case NetEqDecoder::kDecoderPCMu:
+      return new AudioDecoderPcmU(1);
+    case NetEqDecoder::kDecoderPCMa:
+      return new AudioDecoderPcmA(1);
+    case NetEqDecoder::kDecoderPCMu_2ch:
+      return new AudioDecoderPcmU(2);
+    case NetEqDecoder::kDecoderPCMa_2ch:
+      return new AudioDecoderPcmA(2);
 #ifdef WEBRTC_CODEC_ILBC
-    case kDecoderILBC:
+    case NetEqDecoder::kDecoderILBC:
       return new AudioDecoderIlbc;
 #endif
 #if defined(WEBRTC_CODEC_ISACFX)
-    case kDecoderISAC: {
-      AudioEncoderDecoderIsacFix::Config config;
-      return new AudioEncoderDecoderIsacFix(config);
-    }
+    case NetEqDecoder::kDecoderISAC:
+      return new AudioDecoderIsacFix();
 #elif defined(WEBRTC_CODEC_ISAC)
-    case kDecoderISAC: {
-      AudioEncoderDecoderIsac::Config config;
-      config.sample_rate_hz = 16000;
-      return new AudioEncoderDecoderIsac(config);
-    }
-    case kDecoderISACswb:
-    case kDecoderISACfb: {
-      AudioEncoderDecoderIsac::Config config;
-      config.sample_rate_hz = 32000;
-      return new AudioEncoderDecoderIsac(config);
-    }
-#endif
-#ifdef WEBRTC_CODEC_PCM16
-    case kDecoderPCM16B:
-    case kDecoderPCM16Bwb:
-    case kDecoderPCM16Bswb32kHz:
-    case kDecoderPCM16Bswb48kHz:
-      return new AudioDecoderPcm16B;
-    case kDecoderPCM16B_2ch:
-    case kDecoderPCM16Bwb_2ch:
-    case kDecoderPCM16Bswb32kHz_2ch:
-    case kDecoderPCM16Bswb48kHz_2ch:
-      return new AudioDecoderPcm16BMultiCh(2);
-    case kDecoderPCM16B_5ch:
-      return new AudioDecoderPcm16BMultiCh(5);
+    case NetEqDecoder::kDecoderISAC:
+    case NetEqDecoder::kDecoderISACswb:
+      return new AudioDecoderIsac();
 #endif
+    case NetEqDecoder::kDecoderPCM16B:
+    case NetEqDecoder::kDecoderPCM16Bwb:
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz:
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz:
+      return new AudioDecoderPcm16B(1);
+    case NetEqDecoder::kDecoderPCM16B_2ch:
+    case NetEqDecoder::kDecoderPCM16Bwb_2ch:
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
+      return new AudioDecoderPcm16B(2);
+    case NetEqDecoder::kDecoderPCM16B_5ch:
+      return new AudioDecoderPcm16B(5);
 #ifdef WEBRTC_CODEC_G722
-    case kDecoderG722:
+    case NetEqDecoder::kDecoderG722:
       return new AudioDecoderG722;
-    case kDecoderG722_2ch:
+    case NetEqDecoder::kDecoderG722_2ch:
       return new AudioDecoderG722Stereo;
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-    case kDecoderOpus:
+    case NetEqDecoder::kDecoderOpus:
       return new AudioDecoderOpus(1);
-    case kDecoderOpus_2ch:
+    case NetEqDecoder::kDecoderOpus_2ch:
       return new AudioDecoderOpus(2);
 #endif
-    case kDecoderCNGnb:
-    case kDecoderCNGwb:
-    case kDecoderCNGswb32kHz:
-    case kDecoderCNGswb48kHz:
+    case NetEqDecoder::kDecoderCNGnb:
+    case NetEqDecoder::kDecoderCNGwb:
+    case NetEqDecoder::kDecoderCNGswb32kHz:
+    case NetEqDecoder::kDecoderCNGswb48kHz:
       return new AudioDecoderCng;
-    case kDecoderRED:
-    case kDecoderAVT:
-    case kDecoderArbitrary:
+    case NetEqDecoder::kDecoderRED:
+    case NetEqDecoder::kDecoderAVT:
+    case NetEqDecoder::kDecoderArbitrary:
     default: {
       return NULL;
     }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index 5f9c35be55..bc8bdd9626 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -13,231 +13,18 @@
 
 #include <assert.h>
 
-#ifndef AUDIO_DECODER_UNITTEST
-// If this is compiled as a part of the audio_deoder_unittest, the codec
-// selection is made in the gypi file instead of in engine_configurations.h.
 #include "webrtc/engine_configurations.h"
-#endif
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #endif
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-class AudioDecoderPcmU : public AudioDecoder {
- public:
-  AudioDecoderPcmU() {}
-  virtual int Init() { return 0; }
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-  size_t Channels() const override { return 1; }
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
-};
-
-class AudioDecoderPcmA : public AudioDecoder {
- public:
-  AudioDecoderPcmA() {}
-  virtual int Init() { return 0; }
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-  size_t Channels() const override { return 1; }
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
-};
-
-class AudioDecoderPcmUMultiCh : public AudioDecoderPcmU {
- public:
-  explicit AudioDecoderPcmUMultiCh(size_t channels)
-      : AudioDecoderPcmU(), channels_(channels) {
-    assert(channels > 0);
-  }
-  size_t Channels() const override { return channels_; }
-
- private:
-  const size_t channels_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmUMultiCh);
-};
-
-class AudioDecoderPcmAMultiCh : public AudioDecoderPcmA {
- public:
-  explicit AudioDecoderPcmAMultiCh(size_t channels)
-      : AudioDecoderPcmA(), channels_(channels) {
-    assert(channels > 0);
-  }
-  size_t Channels() const override { return channels_; }
-
- private:
-  const size_t channels_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmAMultiCh);
-};
-
-#ifdef WEBRTC_CODEC_PCM16
-// This class handles all four types (i.e., sample rates) of PCM16B codecs.
-// The type is specified in the constructor parameter |type|.
-class AudioDecoderPcm16B : public AudioDecoder {
- public:
-  AudioDecoderPcm16B();
-  virtual int Init() { return 0; }
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-  size_t Channels() const override { return 1; }
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
-};
-
-// This class handles all four types (i.e., sample rates) of PCM16B codecs.
-// The type is specified in the constructor parameter |type|, and the number
-// of channels is derived from the type.
-class AudioDecoderPcm16BMultiCh : public AudioDecoderPcm16B {
- public:
-  explicit AudioDecoderPcm16BMultiCh(int num_channels);
-  size_t Channels() const override { return channels_; }
-
- private:
-  const size_t channels_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16BMultiCh);
-};
-#endif
-
-#ifdef WEBRTC_CODEC_ILBC
-class AudioDecoderIlbc : public AudioDecoder {
- public:
-  AudioDecoderIlbc();
-  virtual ~AudioDecoderIlbc();
-  virtual bool HasDecodePlc() const { return true; }
-  virtual int DecodePlc(int num_frames, int16_t* decoded);
-  virtual int Init();
-  size_t Channels() const override { return 1; }
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  IlbcDecoderInstance* dec_state_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
-};
-#endif
-
-#ifdef WEBRTC_CODEC_G722
-class AudioDecoderG722 : public AudioDecoder {
- public:
-  AudioDecoderG722();
-  virtual ~AudioDecoderG722();
-  virtual bool HasDecodePlc() const { return false; }
-  virtual int Init();
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-  size_t Channels() const override { return 1; }
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  G722DecInst* dec_state_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
-};
-
-class AudioDecoderG722Stereo : public AudioDecoder {
- public:
-  AudioDecoderG722Stereo();
-  virtual ~AudioDecoderG722Stereo();
-  virtual int Init();
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-  size_t Channels() const override { return 2; }
-
- private:
-  // Splits the stereo-interleaved payload in |encoded| into separate payloads
-  // for left and right channels. The separated payloads are written to
-  // |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
-  // The left channel starts at offset 0, while the right channel starts at
-  // offset encoded_len / 2 into |encoded_deinterleaved|.
-  void SplitStereoPacket(const uint8_t* encoded, size_t encoded_len,
-                         uint8_t* encoded_deinterleaved);
-
-  G722DecInst* dec_state_left_;
-  G722DecInst* dec_state_right_;
-
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
-};
-#endif
-
-#ifdef WEBRTC_CODEC_OPUS
-class AudioDecoderOpus : public AudioDecoder {
- public:
-  explicit AudioDecoderOpus(int num_channels);
-  virtual ~AudioDecoderOpus();
-
-  virtual int Init();
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-  virtual int PacketDurationRedundant(const uint8_t* encoded,
-                                      size_t encoded_len) const;
-  virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
-  size_t Channels() const override { return channels_; }
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-  int DecodeRedundantInternal(const uint8_t* encoded,
-                              size_t encoded_len,
-                              int sample_rate_hz,
-                              int16_t* decoded,
-                              SpeechType* speech_type) override;
-
- private:
-  OpusDecInst* dec_state_;
-  const size_t channels_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
-};
-#endif
-
 // AudioDecoderCng is a special type of AudioDecoder. It inherits from
 // AudioDecoder just to fit in the DecoderDatabase. None of the class methods
 // should be used, except constructor, destructor, and accessors.
@@ -247,61 +34,30 @@ class AudioDecoderOpus : public AudioDecoder {
 class AudioDecoderCng : public AudioDecoder {
  public:
   explicit AudioDecoderCng();
-  virtual ~AudioDecoderCng();
-  virtual int Init();
-  virtual int IncomingPacket(const uint8_t* payload,
-                             size_t payload_len,
-                             uint16_t rtp_sequence_number,
-                             uint32_t rtp_timestamp,
-                             uint32_t arrival_timestamp) { return -1; }
+  ~AudioDecoderCng() override;
+  void Reset() override;
+  int IncomingPacket(const uint8_t* payload,
+                     size_t payload_len,
+                     uint16_t rtp_sequence_number,
+                     uint32_t rtp_timestamp,
+                     uint32_t arrival_timestamp) override;
 
-  CNG_dec_inst* CngDecoderInstance() override { return dec_state_; }
-  size_t Channels() const override { return 1; }
+  CNG_dec_inst* CngDecoderInstance() override;
+  size_t Channels() const override;
 
  protected:
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
                      int16_t* decoded,
-                     SpeechType* speech_type) override  {
-    return -1;
-  }
+                     SpeechType* speech_type) override;
 
  private:
   CNG_dec_inst* dec_state_;
-  DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
 };
 
-enum NetEqDecoder {
-  kDecoderPCMu,
-  kDecoderPCMa,
-  kDecoderPCMu_2ch,
-  kDecoderPCMa_2ch,
-  kDecoderILBC,
-  kDecoderISAC,
-  kDecoderISACswb,
-  kDecoderISACfb,
-  kDecoderPCM16B,
-  kDecoderPCM16Bwb,
-  kDecoderPCM16Bswb32kHz,
-  kDecoderPCM16Bswb48kHz,
-  kDecoderPCM16B_2ch,
-  kDecoderPCM16Bwb_2ch,
-  kDecoderPCM16Bswb32kHz_2ch,
-  kDecoderPCM16Bswb48kHz_2ch,
-  kDecoderPCM16B_5ch,
-  kDecoderG722,
-  kDecoderG722_2ch,
-  kDecoderRED,
-  kDecoderAVT,
-  kDecoderCNGnb,
-  kDecoderCNGwb,
-  kDecoderCNGswb32kHz,
-  kDecoderCNGswb48kHz,
-  kDecoderArbitrary,
-  kDecoderOpus,
-  kDecoderOpus_2ch,
-};
+using NetEqDecoder = acm2::RentACodec::NetEqDecoder;
 
 // Returns true if |codec_type| is supported.
 bool CodecSupported(NetEqDecoder codec_type);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 728caefcfe..599929e78d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -18,15 +18,22 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -137,11 +144,11 @@ class AudioDecoderTest : public ::testing::Test {
                           uint8_t* output) {
     encoded_info_.encoded_bytes = 0;
     const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;
-    CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
-             input_len_samples);
+    RTC_CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
+                 input_len_samples);
     rtc::scoped_ptr<int16_t[]> interleaved_input(
         new int16_t[channels_ * samples_per_10ms]);
-    for (int i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
+    for (size_t i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
       EXPECT_EQ(0u, encoded_info_.encoded_bytes);
 
       // Duplicate the mono input signal to however many channels the test
@@ -151,7 +158,10 @@ class AudioDecoderTest : public ::testing::Test {
                                                  interleaved_input.get());
 
       encoded_info_ = audio_encoder_->Encode(
-          0, interleaved_input.get(), audio_encoder_->SampleRateHz() / 100,
+          0, rtc::ArrayView<const int16_t>(interleaved_input.get(),
+                                           audio_encoder_->NumChannels() *
+                                               audio_encoder_->SampleRateHz() /
+                                               100),
           data_length_ * 2, output);
     }
     EXPECT_EQ(payload_type_, encoded_info_.payload_type);
@@ -171,7 +181,6 @@ class AudioDecoderTest : public ::testing::Test {
     size_t processed_samples = 0u;
     encoded_bytes_ = 0u;
     InitEncoder();
-    EXPECT_EQ(0, decoder_->Init());
     std::vector<int16_t> input;
     std::vector<int16_t> decoded;
     while (processed_samples + frame_size_ <= data_length_) {
@@ -220,7 +229,7 @@ class AudioDecoderTest : public ::testing::Test {
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     size_t dec_len;
     AudioDecoder::SpeechType speech_type1, speech_type2;
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
     dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                frame_size_ * channels_ * sizeof(int16_t),
@@ -228,7 +237,7 @@ class AudioDecoderTest : public ::testing::Test {
     ASSERT_LE(dec_len, frame_size_ * channels_);
     EXPECT_EQ(frame_size_ * channels_, dec_len);
     // Re-init decoder and decode again.
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
     dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                frame_size_ * channels_ * sizeof(int16_t),
@@ -249,7 +258,7 @@ class AudioDecoderTest : public ::testing::Test {
         input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     AudioDecoder::SpeechType speech_type;
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
     size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                       frame_size_ * channels_ * sizeof(int16_t),
@@ -280,7 +289,7 @@ class AudioDecoderPcmUTest : public AudioDecoderTest {
   AudioDecoderPcmUTest() : AudioDecoderTest() {
     frame_size_ = 160;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcmU;
+    decoder_ = new AudioDecoderPcmU(1);
     AudioEncoderPcmU::Config config;
     config.frame_size_ms = static_cast<int>(frame_size_ / 8);
     config.payload_type = payload_type_;
@@ -293,7 +302,7 @@ class AudioDecoderPcmATest : public AudioDecoderTest {
   AudioDecoderPcmATest() : AudioDecoderTest() {
     frame_size_ = 160;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcmA;
+    decoder_ = new AudioDecoderPcmA(1);
     AudioEncoderPcmA::Config config;
     config.frame_size_ms = static_cast<int>(frame_size_ / 8);
     config.payload_type = payload_type_;
@@ -307,7 +316,7 @@ class AudioDecoderPcm16BTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 16000;
     frame_size_ = 20 * codec_input_rate_hz_ / 1000;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcm16B;
+    decoder_ = new AudioDecoderPcm16B(1);
     assert(decoder_);
     AudioEncoderPcm16B::Config config;
     config.sample_rate_hz = codec_input_rate_hz_;
@@ -341,14 +350,14 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
         input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
     size_t enc_len = EncodeFrame(input.get(), frame_size_, encoded_);
     AudioDecoder::SpeechType speech_type;
-    EXPECT_EQ(0, decoder_->Init());
+    decoder_->Reset();
     rtc::scoped_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
     size_t dec_len = decoder_->Decode(encoded_, enc_len, codec_input_rate_hz_,
                                       frame_size_ * channels_ * sizeof(int16_t),
                                       output.get(), &speech_type);
     EXPECT_EQ(frame_size_, dec_len);
     // Simply call DecodePlc and verify that we get 0 as return value.
-    EXPECT_EQ(0, decoder_->DecodePlc(1, output.get()));
+    EXPECT_EQ(0U, decoder_->DecodePlc(1, output.get()));
   }
 };
 
@@ -358,16 +367,14 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 16000;
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
-    AudioEncoderDecoderIsac::Config config;
+    AudioEncoderIsac::Config config;
     config.payload_type = payload_type_;
     config.sample_rate_hz = codec_input_rate_hz_;
+    config.adaptive_mode = false;
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
-
-    // We need to create separate AudioEncoderDecoderIsac objects for encoding
-    // and decoding, because the test class destructor destroys them both.
-    audio_encoder_.reset(new AudioEncoderDecoderIsac(config));
-    decoder_ = new AudioEncoderDecoderIsac(config);
+    audio_encoder_.reset(new AudioEncoderIsac(config));
+    decoder_ = new AudioDecoderIsac();
   }
 };
 
@@ -377,16 +384,14 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 32000;
     frame_size_ = 960;
     data_length_ = 10 * frame_size_;
-    AudioEncoderDecoderIsac::Config config;
+    AudioEncoderIsac::Config config;
     config.payload_type = payload_type_;
     config.sample_rate_hz = codec_input_rate_hz_;
+    config.adaptive_mode = false;
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
-
-    // We need to create separate AudioEncoderDecoderIsac objects for encoding
-    // and decoding, because the test class destructor destroys them both.
-    audio_encoder_.reset(new AudioEncoderDecoderIsac(config));
-    decoder_ = new AudioEncoderDecoderIsac(config);
+    audio_encoder_.reset(new AudioEncoderIsac(config));
+    decoder_ = new AudioDecoderIsac();
   }
 };
 
@@ -396,17 +401,14 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 16000;
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
-    AudioEncoderDecoderIsacFix::Config config;
+    AudioEncoderIsacFix::Config config;
     config.payload_type = payload_type_;
     config.sample_rate_hz = codec_input_rate_hz_;
+    config.adaptive_mode = false;
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
-
-    // We need to create separate AudioEncoderDecoderIsacFix objects for
-    // encoding and decoding, because the test class destructor destroys them
-    // both.
-    audio_encoder_.reset(new AudioEncoderDecoderIsacFix(config));
-    decoder_ = new AudioEncoderDecoderIsacFix(config);
+    audio_encoder_.reset(new AudioEncoderIsacFix(config));
+    decoder_ = new AudioDecoderIsacFix();
   }
 };
 
@@ -476,77 +478,102 @@ class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
 TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
   int tolerance = 251;
   double mse = 1734.0;
-  EXPECT_TRUE(CodecSupported(kDecoderPCMu));
   EncodeDecodeTest(data_length_, tolerance, mse);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+namespace {
+int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
+  audio_encoder->SetTargetBitrate(rate);
+  return audio_encoder->GetTargetBitrate();
+}
+void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
+                                               int fixed_rate) {
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, 32000));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate - 1));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate + 1));
+}
+}  // namespace
+
+TEST_F(AudioDecoderPcmUTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
 TEST_F(AudioDecoderPcmATest, EncodeDecode) {
   int tolerance = 308;
   double mse = 1931.0;
-  EXPECT_TRUE(CodecSupported(kDecoderPCMa));
   EncodeDecodeTest(data_length_, tolerance, mse);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+TEST_F(AudioDecoderPcmATest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
 TEST_F(AudioDecoderPcm16BTest, EncodeDecode) {
   int tolerance = 0;
   double mse = 0.0;
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16B));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bwb));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb32kHz));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb48kHz));
   EncodeDecodeTest(2 * data_length_, tolerance, mse);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+TEST_F(AudioDecoderPcm16BTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(),
+                                            codec_input_rate_hz_ * 16);
+}
+
 TEST_F(AudioDecoderIlbcTest, EncodeDecode) {
   int tolerance = 6808;
   double mse = 2.13e6;
   int delay = 80;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderILBC));
   EncodeDecodeTest(500, tolerance, mse, delay);
   ReInitTest();
   EXPECT_TRUE(decoder_->HasDecodePlc());
   DecodePlcTest();
 }
 
+TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333);
+}
+
 TEST_F(AudioDecoderIsacFloatTest, EncodeDecode) {
   int tolerance = 3399;
   double mse = 434951.0;
   int delay = 48;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderISAC));
   EncodeDecodeTest(0, tolerance, mse, delay);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+TEST_F(AudioDecoderIsacFloatTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
+}
+
 TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
   int tolerance = 19757;
   double mse = 8.18e6;
   int delay = 160;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderISACswb));
   EncodeDecodeTest(0, tolerance, mse, delay);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4198
-#if defined(WEBRTC_ANDROID) && defined(__aarch64__)
-#define MAYBE_EncodeDecode DISABLED_EncodeDecode
-#else
-#define MAYBE_EncodeDecode EncodeDecode
-#endif
-TEST_F(AudioDecoderIsacFixTest, MAYBE_EncodeDecode) {
+TEST_F(AudioDecoderIsacSwbTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
+}
+
+TEST_F(AudioDecoderIsacFixTest, EncodeDecode) {
   int tolerance = 11034;
   double mse = 3.46e6;
   int delay = 54;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderISAC));
-#ifdef WEBRTC_ANDROID
+#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
   static const int kEncodedBytes = 685;
+#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
+  static const int kEncodedBytes = 673;
 #else
   static const int kEncodedBytes = 671;
 #endif
@@ -555,18 +582,21 @@ TEST_F(AudioDecoderIsacFixTest, MAYBE_EncodeDecode) {
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+TEST_F(AudioDecoderIsacFixTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
+}
+
 TEST_F(AudioDecoderG722Test, EncodeDecode) {
   int tolerance = 6176;
   double mse = 238630.0;
   int delay = 22;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderG722));
   EncodeDecodeTest(data_length_ / 2, tolerance, mse, delay);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
-TEST_F(AudioDecoderG722StereoTest, CreateAndDestroy) {
-  EXPECT_TRUE(CodecSupported(kDecoderG722_2ch));
+TEST_F(AudioDecoderG722Test, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
 }
 
 TEST_F(AudioDecoderG722StereoTest, EncodeDecode) {
@@ -574,94 +604,148 @@ TEST_F(AudioDecoderG722StereoTest, EncodeDecode) {
   int channel_diff_tolerance = 0;
   double mse = 238630.0;
   int delay = 22;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderG722_2ch));
   EncodeDecodeTest(data_length_, tolerance, mse, delay, channel_diff_tolerance);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+TEST_F(AudioDecoderG722StereoTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 128000);
+}
+
 TEST_F(AudioDecoderOpusTest, EncodeDecode) {
   int tolerance = 6176;
   double mse = 238630.0;
   int delay = 22;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderOpus));
   EncodeDecodeTest(0, tolerance, mse, delay);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+namespace {
+void TestOpusSetTargetBitrates(AudioEncoder* audio_encoder) {
+  EXPECT_EQ(500, SetAndGetTargetBitrate(audio_encoder, 499));
+  EXPECT_EQ(500, SetAndGetTargetBitrate(audio_encoder, 500));
+  EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder, 32000));
+  EXPECT_EQ(512000, SetAndGetTargetBitrate(audio_encoder, 512000));
+  EXPECT_EQ(512000, SetAndGetTargetBitrate(audio_encoder, 513000));
+}
+}  // namespace
+
+TEST_F(AudioDecoderOpusTest, SetTargetBitrate) {
+  TestOpusSetTargetBitrates(audio_encoder_.get());
+}
+
 TEST_F(AudioDecoderOpusStereoTest, EncodeDecode) {
   int tolerance = 6176;
   int channel_diff_tolerance = 0;
   double mse = 238630.0;
   int delay = 22;  // Delay from input to output.
-  EXPECT_TRUE(CodecSupported(kDecoderOpus_2ch));
   EncodeDecodeTest(0, tolerance, mse, delay, channel_diff_tolerance);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
 }
 
+TEST_F(AudioDecoderOpusStereoTest, SetTargetBitrate) {
+  TestOpusSetTargetBitrates(audio_encoder_.get());
+}
+
+namespace {
+#ifdef WEBRTC_CODEC_ILBC
+const bool has_ilbc = true;
+#else
+const bool has_ilbc = false;
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+const bool has_isac = true;
+#else
+const bool has_isac = false;
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+const bool has_isac_swb = true;
+#else
+const bool has_isac_swb = false;
+#endif
+#ifdef WEBRTC_CODEC_G722
+const bool has_g722 = true;
+#else
+const bool has_g722 = false;
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+const bool has_opus = true;
+#else
+const bool has_opus = false;
+#endif
+}  // namespace
+
 TEST(AudioDecoder, CodecSampleRateHz) {
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCMu));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCMa));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCMu_2ch));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCMa_2ch));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderILBC));
-  EXPECT_EQ(16000, CodecSampleRateHz(kDecoderISAC));
-  EXPECT_EQ(32000, CodecSampleRateHz(kDecoderISACswb));
-  EXPECT_EQ(32000, CodecSampleRateHz(kDecoderISACfb));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCM16B));
-  EXPECT_EQ(16000, CodecSampleRateHz(kDecoderPCM16Bwb));
-  EXPECT_EQ(32000, CodecSampleRateHz(kDecoderPCM16Bswb32kHz));
-  EXPECT_EQ(48000, CodecSampleRateHz(kDecoderPCM16Bswb48kHz));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCM16B_2ch));
-  EXPECT_EQ(16000, CodecSampleRateHz(kDecoderPCM16Bwb_2ch));
-  EXPECT_EQ(32000, CodecSampleRateHz(kDecoderPCM16Bswb32kHz_2ch));
-  EXPECT_EQ(48000, CodecSampleRateHz(kDecoderPCM16Bswb48kHz_2ch));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCM16B_5ch));
-  EXPECT_EQ(16000, CodecSampleRateHz(kDecoderG722));
-  EXPECT_EQ(16000, CodecSampleRateHz(kDecoderG722_2ch));
-  EXPECT_EQ(-1, CodecSampleRateHz(kDecoderRED));
-  EXPECT_EQ(-1, CodecSampleRateHz(kDecoderAVT));
-  EXPECT_EQ(8000, CodecSampleRateHz(kDecoderCNGnb));
-  EXPECT_EQ(16000, CodecSampleRateHz(kDecoderCNGwb));
-  EXPECT_EQ(32000, CodecSampleRateHz(kDecoderCNGswb32kHz));
-  EXPECT_EQ(48000, CodecSampleRateHz(kDecoderOpus));
-  EXPECT_EQ(48000, CodecSampleRateHz(kDecoderOpus_2ch));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMu));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMa));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMu_2ch));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMa_2ch));
+  EXPECT_EQ(has_ilbc ? 8000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderILBC));
+  EXPECT_EQ(has_isac ? 16000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderISAC));
+  EXPECT_EQ(has_isac_swb ? 32000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderISACswb));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16B));
+  EXPECT_EQ(16000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bwb));
+  EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb32kHz));
+  EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb48kHz));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16B_2ch));
+  EXPECT_EQ(16000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bwb_2ch));
+  EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch));
+  EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16B_5ch));
+  EXPECT_EQ(has_g722 ? 16000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderG722));
+  EXPECT_EQ(has_g722 ? 16000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderG722_2ch));
+  EXPECT_EQ(-1, CodecSampleRateHz(NetEqDecoder::kDecoderRED));
+  EXPECT_EQ(-1, CodecSampleRateHz(NetEqDecoder::kDecoderAVT));
+  EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGnb));
+  EXPECT_EQ(16000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGwb));
+  EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGswb32kHz));
+  EXPECT_EQ(has_opus ? 48000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderOpus));
+  EXPECT_EQ(has_opus ? 48000 : -1,
+            CodecSampleRateHz(NetEqDecoder::kDecoderOpus_2ch));
+  EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderOpus));
+  EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderOpus_2ch));
   // TODO(tlegrand): Change 32000 to 48000 below once ACM has 48 kHz support.
-  EXPECT_EQ(32000, CodecSampleRateHz(kDecoderCNGswb48kHz));
-  EXPECT_EQ(-1, CodecSampleRateHz(kDecoderArbitrary));
+  EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGswb48kHz));
+  EXPECT_EQ(-1, CodecSampleRateHz(NetEqDecoder::kDecoderArbitrary));
 }
 
 TEST(AudioDecoder, CodecSupported) {
-  EXPECT_TRUE(CodecSupported(kDecoderPCMu));
-  EXPECT_TRUE(CodecSupported(kDecoderPCMa));
-  EXPECT_TRUE(CodecSupported(kDecoderPCMu_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderPCMa_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderILBC));
-  EXPECT_TRUE(CodecSupported(kDecoderISAC));
-  EXPECT_TRUE(CodecSupported(kDecoderISACswb));
-  EXPECT_TRUE(CodecSupported(kDecoderISACfb));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16B));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bwb));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb32kHz));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb48kHz));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16B_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bwb_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb32kHz_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb48kHz_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderPCM16B_5ch));
-  EXPECT_TRUE(CodecSupported(kDecoderG722));
-  EXPECT_TRUE(CodecSupported(kDecoderG722_2ch));
-  EXPECT_TRUE(CodecSupported(kDecoderRED));
-  EXPECT_TRUE(CodecSupported(kDecoderAVT));
-  EXPECT_TRUE(CodecSupported(kDecoderCNGnb));
-  EXPECT_TRUE(CodecSupported(kDecoderCNGwb));
-  EXPECT_TRUE(CodecSupported(kDecoderCNGswb32kHz));
-  EXPECT_TRUE(CodecSupported(kDecoderCNGswb48kHz));
-  EXPECT_TRUE(CodecSupported(kDecoderArbitrary));
-  EXPECT_TRUE(CodecSupported(kDecoderOpus));
-  EXPECT_TRUE(CodecSupported(kDecoderOpus_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMu));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMa));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMu_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMa_2ch));
+  EXPECT_EQ(has_ilbc, CodecSupported(NetEqDecoder::kDecoderILBC));
+  EXPECT_EQ(has_isac, CodecSupported(NetEqDecoder::kDecoderISAC));
+  EXPECT_EQ(has_isac_swb, CodecSupported(NetEqDecoder::kDecoderISACswb));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16B));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bwb));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb32kHz));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb48kHz));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16B_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bwb_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16B_5ch));
+  EXPECT_EQ(has_g722, CodecSupported(NetEqDecoder::kDecoderG722));
+  EXPECT_EQ(has_g722, CodecSupported(NetEqDecoder::kDecoderG722_2ch));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderRED));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderAVT));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGnb));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGwb));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb32kHz));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb48kHz));
+  EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderArbitrary));
+  EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus));
+  EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus_2ch));
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
index b19eef98db..1381895085 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -183,6 +183,10 @@ void AudioMultiVector::CrossFade(const AudioMultiVector& append_this,
   }
 }
 
+size_t AudioMultiVector::Channels() const {
+  return num_channels_;
+}
+
 size_t AudioMultiVector::Size() const {
   assert(channels_[0]);
   return channels_[0]->Size();
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
index 27f377e8e0..1c28648816 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
@@ -106,7 +106,7 @@ class AudioMultiVector {
                          size_t fade_length);
 
   // Returns the number of channels.
-  virtual size_t Channels() const { return num_channels_; }
+  virtual size_t Channels() const;
 
   // Returns the number of elements per channel in this AudioMultiVector.
   virtual size_t Size() const;
@@ -132,7 +132,7 @@ class AudioMultiVector {
   size_t num_channels_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(AudioMultiVector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioMultiVector);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc
index d0f1aca55e..fa16481b69 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc
@@ -18,6 +18,21 @@
 
 namespace webrtc {
 
+AudioVector::AudioVector()
+    : array_(new int16_t[kDefaultInitialSize]),
+      first_free_ix_(0),
+      capacity_(kDefaultInitialSize) {
+}
+
+AudioVector::AudioVector(size_t initial_size)
+    : array_(new int16_t[initial_size]),
+      first_free_ix_(initial_size),
+      capacity_(initial_size) {
+  memset(array_.get(), 0, initial_size * sizeof(int16_t));
+}
+
+AudioVector::~AudioVector() = default;
+
 void AudioVector::Clear() {
   first_free_ix_ = 0;
 }
@@ -145,6 +160,16 @@ void AudioVector::CrossFade(const AudioVector& append_this,
     PushBack(&append_this[fade_length], samples_to_push_back);
 }
 
+// Returns the number of elements in this AudioVector.
+size_t AudioVector::Size() const {
+  return first_free_ix_;
+}
+
+// Returns true if this AudioVector is empty.
+bool AudioVector::Empty() const {
+  return first_free_ix_ == 0;
+}
+
 const int16_t& AudioVector::operator[](size_t index) const {
   return array_[index];
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.h
index 28e53ee65c..e046e38277 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_vector.h
@@ -22,20 +22,12 @@ namespace webrtc {
 class AudioVector {
  public:
   // Creates an empty AudioVector.
-  AudioVector()
-      : array_(new int16_t[kDefaultInitialSize]),
-        first_free_ix_(0),
-        capacity_(kDefaultInitialSize) {}
+  AudioVector();
 
   // Creates an AudioVector with an initial size.
-  explicit AudioVector(size_t initial_size)
-      : array_(new int16_t[initial_size]),
-        first_free_ix_(initial_size),
-        capacity_(initial_size) {
-    memset(array_.get(), 0, initial_size * sizeof(int16_t));
-  }
+  explicit AudioVector(size_t initial_size);
 
-  virtual ~AudioVector() {}
+  virtual ~AudioVector();
 
   // Deletes all values and make the vector empty.
   virtual void Clear();
@@ -94,10 +86,10 @@ class AudioVector {
   virtual void CrossFade(const AudioVector& append_this, size_t fade_length);
 
   // Returns the number of elements in this AudioVector.
-  virtual size_t Size() const { return first_free_ix_; }
+  virtual size_t Size() const;
 
   // Returns true if this AudioVector is empty.
-  virtual bool Empty() const { return (first_free_ix_ == 0); }
+  virtual bool Empty() const;
 
   // Accesses and modifies an element of AudioVector.
   const int16_t& operator[](size_t index) const;
@@ -113,7 +105,7 @@ class AudioVector {
                           // Note that this index may point outside of array_.
   size_t capacity_;  // Allocated number of samples in the array.
 
-  DISALLOW_COPY_AND_ASSIGN(AudioVector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioVector);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.cc
index 4fbc84c5a3..7e7a6325e9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.cc
@@ -21,6 +21,9 @@
 
 namespace webrtc {
 
+// static
+const size_t BackgroundNoise::kMaxLpcOrder;
+
 BackgroundNoise::BackgroundNoise(size_t num_channels)
     : num_channels_(num_channels),
       channel_parameters_(new ChannelParameters[num_channels_]),
@@ -150,7 +153,7 @@ const int16_t* BackgroundNoise::FilterState(size_t channel) const {
 void BackgroundNoise::SetFilterState(size_t channel, const int16_t* input,
                                      size_t length) {
   assert(channel < num_channels_);
-  length = std::min(length, static_cast<size_t>(kMaxLpcOrder));
+  length = std::min(length, kMaxLpcOrder);
   memcpy(channel_parameters_[channel].filter_state, input,
          length * sizeof(int16_t));
 }
@@ -165,7 +168,7 @@ int16_t BackgroundNoise::ScaleShift(size_t channel) const {
 }
 
 int32_t BackgroundNoise::CalculateAutoCorrelation(
-    const int16_t* signal, int length, int32_t* auto_correlation) const {
+    const int16_t* signal, size_t length, int32_t* auto_correlation) const {
   int16_t signal_max = WebRtcSpl_MaxAbsValueW16(signal, length);
   int correlation_scale = kLogVecLen -
       WebRtcSpl_NormW32(signal_max * signal_max);
@@ -239,19 +242,19 @@ void BackgroundNoise::SaveParameters(size_t channel,
   parameters.low_energy_update_threshold = 0;
 
   // Normalize residual_energy to 29 or 30 bits before sqrt.
-  int norm_shift = WebRtcSpl_NormW32(residual_energy) - 1;
+  int16_t norm_shift = WebRtcSpl_NormW32(residual_energy) - 1;
   if (norm_shift & 0x1) {
     norm_shift -= 1;  // Even number of shifts required.
   }
-  assert(norm_shift >= 0);  // Should always be positive.
-  residual_energy = residual_energy << norm_shift;
+  residual_energy = WEBRTC_SPL_SHIFT_W32(residual_energy, norm_shift);
 
   // Calculate scale and shift factor.
-  parameters.scale = WebRtcSpl_SqrtFloor(residual_energy);
+  parameters.scale = static_cast<int16_t>(WebRtcSpl_SqrtFloor(residual_energy));
   // Add 13 to the |scale_shift_|, since the random numbers table is in
   // Q13.
   // TODO(hlundin): Move the "13" to where the |scale_shift_| is used?
-  parameters.scale_shift = 13 + ((kLogResidualLength + norm_shift) / 2);
+  parameters.scale_shift =
+      static_cast<int16_t>(13 + ((kLogResidualLength + norm_shift) / 2));
 
   initialized_ = true;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.h
index fd4e6a565a..976c55874b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/background_noise.h
@@ -16,7 +16,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,7 +29,7 @@ class BackgroundNoise {
  public:
   // TODO(hlundin): For 48 kHz support, increase kMaxLpcOrder to 10.
   // Will work anyway, but probably sound a little worse.
-  static const int kMaxLpcOrder = 8;  // 32000 / 8000 + 4.
+  static const size_t kMaxLpcOrder = 8;  // 32000 / 8000 + 4.
 
   explicit BackgroundNoise(size_t num_channels);
   virtual ~BackgroundNoise();
@@ -76,10 +76,10 @@ class BackgroundNoise {
 
  private:
   static const int kThresholdIncrement = 229;  // 0.0035 in Q16.
-  static const int kVecLen = 256;
+  static const size_t kVecLen = 256;
   static const int kLogVecLen = 8;  // log2(kVecLen).
-  static const int kResidualLength = 64;
-  static const int kLogResidualLength = 6;  // log2(kResidualLength)
+  static const size_t kResidualLength = 64;
+  static const int16_t kLogResidualLength = 6;  // log2(kResidualLength)
 
   struct ChannelParameters {
     // Constructor.
@@ -112,7 +112,7 @@ class BackgroundNoise {
   };
 
   int32_t CalculateAutoCorrelation(const int16_t* signal,
-                                   int length,
+                                   size_t length,
                                    int32_t* auto_correlation) const;
 
   // Increments the energy threshold by a factor 1 + |kThresholdIncrement|.
@@ -130,7 +130,7 @@ class BackgroundNoise {
   bool initialized_;
   NetEq::BackgroundNoiseMode mode_;
 
-  DISALLOW_COPY_AND_ASSIGN(BackgroundNoise);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BackgroundNoise);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
index 0388b19502..905479178d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -23,16 +23,16 @@ void BufferLevelFilter::Reset() {
   level_factor_ = 253;
 }
 
-void BufferLevelFilter::Update(int buffer_size_packets,
+void BufferLevelFilter::Update(size_t buffer_size_packets,
                                int time_stretched_samples,
-                               int packet_len_samples) {
+                               size_t packet_len_samples) {
   // Filter:
   // |filtered_current_level_| = |level_factor_| * |filtered_current_level_| +
   //                            (1 - |level_factor_|) * |buffer_size_packets|
   // |level_factor_| and |filtered_current_level_| are in Q8.
   // |buffer_size_packets| is in Q0.
   filtered_current_level_ = ((level_factor_ * filtered_current_level_) >> 8) +
-      ((256 - level_factor_) * buffer_size_packets);
+      ((256 - level_factor_) * static_cast<int>(buffer_size_packets));
 
   // Account for time-scale operations (accelerate and pre-emptive expand).
   if (time_stretched_samples && packet_len_samples > 0) {
@@ -42,7 +42,7 @@ void BufferLevelFilter::Update(int buffer_size_packets,
     // Make sure that the filtered value remains non-negative.
     filtered_current_level_ = std::max(0,
         filtered_current_level_ -
-        (time_stretched_samples << 8) / packet_len_samples);
+        (time_stretched_samples << 8) / static_cast<int>(packet_len_samples));
   }
 }
 
@@ -57,4 +57,9 @@ void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) {
     level_factor_ = 254;
   }
 }
+
+int BufferLevelFilter::filtered_current_level() const {
+  return filtered_current_level_;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
index 48f7f564c9..030870653c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
 
+#include <stddef.h>
+
 #include "webrtc/base/constructormagic.h"
 
 namespace webrtc {
@@ -26,21 +28,21 @@ class BufferLevelFilter {
   // corresponding number of packets, and is subtracted from the filtered
   // value (thus bypassing the filter operation). |packet_len_samples| is the
   // number of audio samples carried in each incoming packet.
-  virtual void Update(int buffer_size_packets, int time_stretched_samples,
-                      int packet_len_samples);
+  virtual void Update(size_t buffer_size_packets, int time_stretched_samples,
+                      size_t packet_len_samples);
 
   // Set the current target buffer level (obtained from
   // DelayManager::base_target_level()). Used to select the appropriate
   // filter coefficient.
   virtual void SetTargetBufferLevel(int target_buffer_level);
 
-  virtual int filtered_current_level() const { return filtered_current_level_; }
+  virtual int filtered_current_level() const;
 
  private:
   int level_factor_;  // Filter factor for the buffer level filter in Q8.
   int filtered_current_level_;  // Filtered current buffer level in Q8.
 
-  DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.cc
index 54b0a28e52..a5b08469be 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.cc
@@ -12,8 +12,9 @@
 
 #include <assert.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
@@ -44,6 +45,7 @@ int ComfortNoise::UpdateParameters(Packet* packet) {
   delete packet;
   if (ret < 0) {
     internal_error_code_ = WebRtcCng_GetErrorCodeDec(cng_inst);
+    LOG(LS_ERROR) << "WebRtcCng_UpdateSid produced " << internal_error_code_;
     return kInternalError;
   }
   return kOK;
@@ -56,6 +58,7 @@ int ComfortNoise::Generate(size_t requested_length,
          fs_hz_ == 48000);
   // Not adapted for multi-channel yet.
   if (output->Channels() != 1) {
+    LOG(LS_ERROR) << "No multi-channel support";
     return kMultiChannelNotSupported;
   }
 
@@ -70,17 +73,18 @@ int ComfortNoise::Generate(size_t requested_length,
   // Get the decoder from the database.
   AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
   if (!cng_decoder) {
+    LOG(LS_ERROR) << "Unknwown payload type";
     return kUnknownPayloadType;
   }
   CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
   // The expression &(*output)[0][0] is a pointer to the first element in
   // the first channel.
-  if (WebRtcCng_Generate(cng_inst, &(*output)[0][0],
-                         static_cast<int16_t>(number_of_samples),
+  if (WebRtcCng_Generate(cng_inst, &(*output)[0][0], number_of_samples,
                          new_period) < 0) {
     // Error returned.
     output->Zeros(requested_length);
     internal_error_code_ = WebRtcCng_GetErrorCodeDec(cng_inst);
+    LOG(LS_ERROR) << "WebRtcCng_Generate produced " << internal_error_code_;
     return kInternalError;
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.h
index d465596245..1fc2258663 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/comfort_noise.h
@@ -66,7 +66,7 @@ class ComfortNoise {
   DecoderDatabase* decoder_database_;
   SyncBuffer* sync_buffer_;
   int internal_error_code_;
-  DISALLOW_COPY_AND_ASSIGN(ComfortNoise);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ComfortNoise);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.cc
index 5fb054c785..39bb4662c7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -19,12 +19,12 @@
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
 DecisionLogic* DecisionLogic::Create(int fs_hz,
-                                     int output_size_samples,
+                                     size_t output_size_samples,
                                      NetEqPlayoutMode playout_mode,
                                      DecoderDatabase* decoder_database,
                                      const PacketBuffer& packet_buffer,
@@ -56,7 +56,7 @@ DecisionLogic* DecisionLogic::Create(int fs_hz,
 }
 
 DecisionLogic::DecisionLogic(int fs_hz,
-                             int output_size_samples,
+                             size_t output_size_samples,
                              NetEqPlayoutMode playout_mode,
                              DecoderDatabase* decoder_database,
                              const PacketBuffer& packet_buffer,
@@ -95,7 +95,7 @@ void DecisionLogic::SoftReset() {
   timescale_hold_off_ = kMinTimescaleInterval;
 }
 
-void DecisionLogic::SetSampleRate(int fs_hz, int output_size_samples) {
+void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
   // TODO(hlundin): Change to an enumerator and skip assert.
   assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz ==  32000 || fs_hz == 48000);
   fs_mult_ = fs_hz / 8000;
@@ -104,7 +104,7 @@ void DecisionLogic::SetSampleRate(int fs_hz, int output_size_samples) {
 
 Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
                                       const Expand& expand,
-                                      int decoder_frame_length,
+                                      size_t decoder_frame_length,
                                       const RTPHeader* packet_header,
                                       Modes prev_mode,
                                       bool play_dtmf, bool* reset_decoder) {
@@ -123,14 +123,11 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
     }
   }
 
-  const int samples_left = static_cast<int>(
-      sync_buffer.FutureLength() - expand.overlap_length());
-  const int cur_size_samples =
+  const size_t samples_left =
+      sync_buffer.FutureLength() - expand.overlap_length();
+  const size_t cur_size_samples =
       samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
                                                        decoder_frame_length);
-  LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
-      " packets * " << decoder_frame_length << " samples/packet + " <<
-      samples_left << " samples in sync buffer = " << cur_size_samples;
 
   prev_time_scale_ = prev_time_scale_ &&
       (prev_mode == kModeAccelerateSuccess ||
@@ -153,9 +150,10 @@ void DecisionLogic::ExpandDecision(Operations operation) {
   }
 }
 
-void DecisionLogic::FilterBufferLevel(int buffer_size_samples,
+void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
                                       Modes prev_mode) {
-  const int elapsed_time_ms = output_size_samples_ / (8 * fs_mult_);
+  const int elapsed_time_ms =
+      static_cast<int>(output_size_samples_ / (8 * fs_mult_));
   delay_manager_->UpdateCounters(elapsed_time_ms);
 
   // Do not update buffer history if currently playing CNG since it will bias
@@ -164,7 +162,7 @@ void DecisionLogic::FilterBufferLevel(int buffer_size_samples,
     buffer_level_filter_->SetTargetBufferLevel(
         delay_manager_->base_target_level());
 
-    int buffer_size_packets = 0;
+    size_t buffer_size_packets = 0;
     if (packet_length_samples_ > 0) {
       // Calculate size in packets.
       buffer_size_packets = buffer_size_samples / packet_length_samples_;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.h
index 672ce939d4..72121b7aac 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -34,7 +34,7 @@ class DecisionLogic {
   // Static factory function which creates different types of objects depending
   // on the |playout_mode|.
   static DecisionLogic* Create(int fs_hz,
-                               int output_size_samples,
+                               size_t output_size_samples,
                                NetEqPlayoutMode playout_mode,
                                DecoderDatabase* decoder_database,
                                const PacketBuffer& packet_buffer,
@@ -43,7 +43,7 @@ class DecisionLogic {
 
   // Constructor.
   DecisionLogic(int fs_hz,
-                int output_size_samples,
+                size_t output_size_samples,
                 NetEqPlayoutMode playout_mode,
                 DecoderDatabase* decoder_database,
                 const PacketBuffer& packet_buffer,
@@ -60,7 +60,7 @@ class DecisionLogic {
   void SoftReset();
 
   // Sets the sample rate and the output block size.
-  void SetSampleRate(int fs_hz, int output_size_samples);
+  void SetSampleRate(int fs_hz, size_t output_size_samples);
 
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
@@ -75,7 +75,7 @@ class DecisionLogic {
   // return value.
   Operations GetDecision(const SyncBuffer& sync_buffer,
                          const Expand& expand,
-                         int decoder_frame_length,
+                         size_t decoder_frame_length,
                          const RTPHeader* packet_header,
                          Modes prev_mode,
                          bool play_dtmf,
@@ -101,12 +101,12 @@ class DecisionLogic {
 
   // Accessors and mutators.
   void set_sample_memory(int32_t value) { sample_memory_ = value; }
-  int generated_noise_samples() const { return generated_noise_samples_; }
-  void set_generated_noise_samples(int value) {
+  size_t generated_noise_samples() const { return generated_noise_samples_; }
+  void set_generated_noise_samples(size_t value) {
     generated_noise_samples_ = value;
   }
-  int packet_length_samples() const { return packet_length_samples_; }
-  void set_packet_length_samples(int value) {
+  size_t packet_length_samples() const { return packet_length_samples_; }
+  void set_packet_length_samples(size_t value) {
     packet_length_samples_ = value;
   }
   void set_prev_time_scale(bool value) { prev_time_scale_ = value; }
@@ -134,7 +134,7 @@ class DecisionLogic {
   // Should be implemented by derived classes.
   virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
                                             const Expand& expand,
-                                            int decoder_frame_length,
+                                            size_t decoder_frame_length,
                                             const RTPHeader* packet_header,
                                             Modes prev_mode,
                                             bool play_dtmf,
@@ -142,18 +142,18 @@ class DecisionLogic {
 
   // Updates the |buffer_level_filter_| with the current buffer level
   // |buffer_size_packets|.
-  void FilterBufferLevel(int buffer_size_packets, Modes prev_mode);
+  void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
 
   DecoderDatabase* decoder_database_;
   const PacketBuffer& packet_buffer_;
   DelayManager* delay_manager_;
   BufferLevelFilter* buffer_level_filter_;
   int fs_mult_;
-  int output_size_samples_;
+  size_t output_size_samples_;
   CngState cng_state_;  // Remember if comfort noise is interrupted by other
                         // event (e.g., DTMF).
-  int generated_noise_samples_;
-  int packet_length_samples_;
+  size_t generated_noise_samples_;
+  size_t packet_length_samples_;
   int sample_memory_;
   bool prev_time_scale_;
   int timescale_hold_off_;
@@ -161,7 +161,7 @@ class DecisionLogic {
   const NetEqPlayoutMode playout_mode_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
index 08a4c4cb64..ddea64425f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
@@ -22,7 +22,7 @@ namespace webrtc {
 Operations DecisionLogicFax::GetDecisionSpecialized(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
index 97c481db8b..204dcc168a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
@@ -23,7 +23,7 @@ class DecisionLogicFax : public DecisionLogic {
  public:
   // Constructor.
   DecisionLogicFax(int fs_hz,
-                   int output_size_samples,
+                   size_t output_size_samples,
                    NetEqPlayoutMode playout_mode,
                    DecoderDatabase* decoder_database,
                    const PacketBuffer& packet_buffer,
@@ -34,9 +34,6 @@ class DecisionLogicFax : public DecisionLogic {
                       buffer_level_filter) {
   }
 
-  // Destructor.
-  virtual ~DecisionLogicFax() {}
-
  protected:
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
@@ -49,14 +46,14 @@ class DecisionLogicFax : public DecisionLogic {
   // remain true if it was true before the call).
   Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
                                     const Expand& expand,
-                                    int decoder_frame_length,
+                                    size_t decoder_frame_length,
                                     const RTPHeader* packet_header,
                                     Modes prev_mode,
                                     bool play_dtmf,
                                     bool* reset_decoder) override;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index f2382845b0..0252d1cdfa 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -20,14 +20,14 @@
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 Operations DecisionLogicNormal::GetDecisionSpecialized(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
@@ -67,7 +67,8 @@ Operations DecisionLogicNormal::GetDecisionSpecialized(
     return kNormal;
   }
 
-  const uint32_t five_seconds_samples = 5 * 8000 * fs_mult_;
+  const uint32_t five_seconds_samples =
+      static_cast<uint32_t>(5 * 8000 * fs_mult_);
   // Check if the required packet is available.
   if (target_timestamp == available_timestamp) {
     return ExpectedPacketAvailable(prev_mode, play_dtmf);
@@ -87,10 +88,11 @@ Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
                                              uint32_t target_timestamp,
                                              uint32_t available_timestamp) {
   // Signed difference between target and available timestamp.
-  int32_t timestamp_diff = (generated_noise_samples_ + target_timestamp) -
-      available_timestamp;
-  int32_t optimal_level_samp =
-      (delay_manager_->TargetLevel() * packet_length_samples_) >> 8;
+  int32_t timestamp_diff = static_cast<int32_t>(
+      static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) -
+      available_timestamp);
+  int32_t optimal_level_samp = static_cast<int32_t>(
+      (delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
   int32_t excess_waiting_time_samp = -timestamp_diff - optimal_level_samp;
 
   if (excess_waiting_time_samp > optimal_level_samp / 2) {
@@ -132,15 +134,13 @@ Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
     // Check criterion for time-stretching.
     int low_limit, high_limit;
     delay_manager_->BufferLimits(&low_limit, &high_limit);
-    if ((buffer_level_filter_->filtered_current_level() >= high_limit &&
-        TimescaleAllowed()) ||
-        buffer_level_filter_->filtered_current_level() >= high_limit << 2) {
-      // Buffer level higher than limit and time-scaling allowed,
-      // or buffer level really high.
-      return kAccelerate;
-    } else if ((buffer_level_filter_->filtered_current_level() < low_limit)
-        && TimescaleAllowed()) {
-      return kPreemptiveExpand;
+    if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
+      return kFastAccelerate;
+    if (TimescaleAllowed()) {
+      if (buffer_level_filter_->filtered_current_level() >= high_limit)
+        return kAccelerate;
+      if (buffer_level_filter_->filtered_current_level() < low_limit)
+        return kPreemptiveExpand;
     }
   }
   return kNormal;
@@ -149,7 +149,7 @@ Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
 Operations DecisionLogicNormal::FuturePacketAvailable(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     Modes prev_mode,
     uint32_t target_timestamp,
     uint32_t available_timestamp,
@@ -172,9 +172,9 @@ Operations DecisionLogicNormal::FuturePacketAvailable(
     }
   }
 
-  const int samples_left = static_cast<int>(sync_buffer.FutureLength() -
-      expand.overlap_length());
-  const int cur_size_samples = samples_left +
+  const size_t samples_left =
+      sync_buffer.FutureLength() - expand.overlap_length();
+  const size_t cur_size_samples = samples_left +
       packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
 
   // If previous was comfort noise, then no merge is needed.
@@ -184,11 +184,11 @@ Operations DecisionLogicNormal::FuturePacketAvailable(
     // safety precaution), but make sure that the number of samples in buffer
     // is no higher than 4 times the optimal level. (Note that TargetLevel()
     // is in Q8.)
-    int32_t timestamp_diff = (generated_noise_samples_ + target_timestamp) -
-        available_timestamp;
-    if (timestamp_diff >= 0 ||
+    if (static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) >=
+            available_timestamp ||
         cur_size_samples >
-        4 * ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8)) {
+            ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
+            4) {
       // Time to play this new packet.
       return kNormal;
     } else {
@@ -205,7 +205,8 @@ Operations DecisionLogicNormal::FuturePacketAvailable(
   // fs_mult_ * 8 = fs / 1000.)
   if (prev_mode == kModeExpand ||
       (decoder_frame_length < output_size_samples_ &&
-       cur_size_samples > kAllowMergeWithoutExpandMs * fs_mult_ * 8)) {
+       cur_size_samples >
+           static_cast<size_t>(kAllowMergeWithoutExpandMs * fs_mult_ * 8))) {
     return kMerge;
   } else if (play_dtmf) {
     // Play DTMF instead of expand.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
index a339d160f2..7465906a38 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
@@ -23,7 +23,7 @@ class DecisionLogicNormal : public DecisionLogic {
  public:
   // Constructor.
   DecisionLogicNormal(int fs_hz,
-                      int output_size_samples,
+                      size_t output_size_samples,
                       NetEqPlayoutMode playout_mode,
                       DecoderDatabase* decoder_database,
                       const PacketBuffer& packet_buffer,
@@ -34,9 +34,6 @@ class DecisionLogicNormal : public DecisionLogic {
                       buffer_level_filter) {
   }
 
-  // Destructor.
-  virtual ~DecisionLogicNormal() {}
-
  protected:
   static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
   static const int kReinitAfterExpands = 100;
@@ -51,19 +48,21 @@ class DecisionLogicNormal : public DecisionLogic {
   // should be set to true. The output variable |reset_decoder| will be set to
   // true if a reset is required; otherwise it is left unchanged (i.e., it can
   // remain true if it was true before the call).
-  virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
-                                            const Expand& expand,
-                                            int decoder_frame_length,
-                                            const RTPHeader* packet_header,
-                                            Modes prev_mode, bool play_dtmf,
-                                            bool* reset_decoder);
+  Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
+                                    const Expand& expand,
+                                    size_t decoder_frame_length,
+                                    const RTPHeader* packet_header,
+                                    Modes prev_mode,
+                                    bool play_dtmf,
+                                    bool* reset_decoder) override;
 
   // Returns the operation to do given that the expected packet is not
   // available, but a packet further into the future is at hand.
   virtual Operations FuturePacketAvailable(
       const SyncBuffer& sync_buffer,
       const Expand& expand,
-      int decoder_frame_length, Modes prev_mode,
+      size_t decoder_frame_length,
+      Modes prev_mode,
       uint32_t target_timestamp,
       uint32_t available_timestamp,
       bool play_dtmf);
@@ -100,7 +99,7 @@ class DecisionLogicNormal : public DecisionLogic {
   // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
   bool MaxWaitForPacket() const;
 
-  DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.cc
index b9097b0873..92d4bab1e4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -13,6 +13,8 @@
 #include <assert.h>
 #include <utility>  // pair
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 namespace webrtc {
@@ -37,17 +39,17 @@ void DecoderDatabase::Reset() {
 }
 
 int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
-                                     NetEqDecoder codec_type) {
+                                     NetEqDecoder codec_type,
+                                     const std::string& name) {
   if (rtp_payload_type > 0x7F) {
     return kInvalidRtpPayloadType;
   }
   if (!CodecSupported(codec_type)) {
     return kCodecNotSupported;
   }
-  int fs_hz = CodecSampleRateHz(codec_type);
-  std::pair<DecoderMap::iterator, bool> ret;
-  DecoderInfo info(codec_type, fs_hz, NULL, false);
-  ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
+  const int fs_hz = CodecSampleRateHz(codec_type);
+  DecoderInfo info(codec_type, name, fs_hz, NULL, false);
+  auto ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
   if (ret.second == false) {
     // Database already contains a decoder with type |rtp_payload_type|.
     return kDecoderExists;
@@ -57,6 +59,7 @@ int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
 
 int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
                                     NetEqDecoder codec_type,
+                                    const std::string& codec_name,
                                     int fs_hz,
                                     AudioDecoder* decoder) {
   if (rtp_payload_type > 0x7F) {
@@ -71,9 +74,8 @@ int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
   if (!decoder) {
     return kInvalidPointer;
   }
-  decoder->Init();
   std::pair<DecoderMap::iterator, bool> ret;
-  DecoderInfo info(codec_type, fs_hz, decoder, true);
+  DecoderInfo info(codec_type, codec_name, fs_hz, decoder, true);
   ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
   if (ret.second == false) {
     // Database already contains a decoder with type |rtp_payload_type|.
@@ -135,7 +137,6 @@ AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) {
     AudioDecoder* decoder = CreateAudioDecoder(info->codec_type);
     assert(decoder);  // Should not be able to have an unsupported codec here.
     info->decoder = decoder;
-    info->decoder->Init();
   }
   return info->decoder;
 }
@@ -151,10 +152,10 @@ bool DecoderDatabase::IsType(uint8_t rtp_payload_type,
 }
 
 bool DecoderDatabase::IsComfortNoise(uint8_t rtp_payload_type) const {
-  if (IsType(rtp_payload_type, kDecoderCNGnb) ||
-      IsType(rtp_payload_type, kDecoderCNGwb) ||
-      IsType(rtp_payload_type, kDecoderCNGswb32kHz) ||
-      IsType(rtp_payload_type, kDecoderCNGswb48kHz)) {
+  if (IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGnb) ||
+      IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGwb) ||
+      IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGswb32kHz) ||
+      IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGswb48kHz)) {
     return true;
   } else {
     return false;
@@ -162,11 +163,11 @@ bool DecoderDatabase::IsComfortNoise(uint8_t rtp_payload_type) const {
 }
 
 bool DecoderDatabase::IsDtmf(uint8_t rtp_payload_type) const {
-  return IsType(rtp_payload_type, kDecoderAVT);
+  return IsType(rtp_payload_type, NetEqDecoder::kDecoderAVT);
 }
 
 bool DecoderDatabase::IsRed(uint8_t rtp_payload_type) const {
-  return IsType(rtp_payload_type, kDecoderRED);
+  return IsType(rtp_payload_type, NetEqDecoder::kDecoderRED);
 }
 
 int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
@@ -249,6 +250,8 @@ int DecoderDatabase::CheckPayloadTypes(const PacketList& packet_list) const {
   for (it = packet_list.begin(); it != packet_list.end(); ++it) {
     if (decoders_.find((*it)->header.payloadType) == decoders_.end()) {
       // Payload type is not found.
+      LOG(LS_WARNING) << "CheckPayloadTypes: unknown RTP payload type "
+                      << static_cast<int>((*it)->header.payloadType);
       return kDecoderNotFound;
     }
   }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.h
index 1dbc685c37..f34904fda8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -12,8 +12,10 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
 
 #include <map>
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"  // NULL
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
@@ -35,26 +37,28 @@ class DecoderDatabase {
 
   // Struct used to store decoder info in the database.
   struct DecoderInfo {
-    // Constructors.
-    DecoderInfo()
-        : codec_type(kDecoderArbitrary),
-          fs_hz(8000),
-          decoder(NULL),
-          external(false) {
-    }
+    DecoderInfo() = default;
     DecoderInfo(NetEqDecoder ct, int fs, AudioDecoder* dec, bool ext)
+        : DecoderInfo(ct, "", fs, dec, ext) {}
+    DecoderInfo(NetEqDecoder ct,
+                const std::string& nm,
+                int fs,
+                AudioDecoder* dec,
+                bool ext)
         : codec_type(ct),
+          name(nm),
           fs_hz(fs),
+          rtp_sample_rate_hz(fs),
           decoder(dec),
-          external(ext) {
-    }
-    // Destructor. (Defined in decoder_database.cc.)
+          external(ext) {}
     ~DecoderInfo();
 
-    NetEqDecoder codec_type;
-    int fs_hz;
-    AudioDecoder* decoder;
-    bool external;
+    NetEqDecoder codec_type = NetEqDecoder::kDecoderArbitrary;
+    std::string name;
+    int fs_hz = 8000;
+    int rtp_sample_rate_hz = 8000;
+    AudioDecoder* decoder = nullptr;
+    bool external = false;
   };
 
   // Maximum value for 8 bits, and an invalid RTP payload type (since it is
@@ -76,16 +80,21 @@ class DecoderDatabase {
   // using InsertExternal().
   virtual void Reset();
 
-  // Registers |rtp_payload_type| as a decoder of type |codec_type|. Returns
-  // kOK on success; otherwise an error code.
+  // Registers |rtp_payload_type| as a decoder of type |codec_type|. The |name|
+  // is only used to populate the name field in the DecoderInfo struct in the
+  // database, and can be arbitrary (including empty). Returns kOK on success;
+  // otherwise an error code.
   virtual int RegisterPayload(uint8_t rtp_payload_type,
-                              NetEqDecoder codec_type);
+                              NetEqDecoder codec_type,
+                              const std::string& name);
 
   // Registers an externally created AudioDecoder object, and associates it
   // as a decoder of type |codec_type| with |rtp_payload_type|.
   virtual int InsertExternal(uint8_t rtp_payload_type,
                              NetEqDecoder codec_type,
-                             int fs_hz, AudioDecoder* decoder);
+                             const std::string& codec_name,
+                             int fs_hz,
+                             AudioDecoder* decoder);
 
   // Removes the entry for |rtp_payload_type| from the database.
   // Returns kDecoderNotFound or kOK depending on the outcome of the operation.
@@ -147,7 +156,7 @@ class DecoderDatabase {
   int active_decoder_;
   int active_cng_decoder_;
 
-  DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
index 1e4e58af3c..85aaef1143 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -19,7 +19,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -32,8 +31,10 @@ TEST(DecoderDatabase, CreateAndDestroy) {
 TEST(DecoderDatabase, InsertAndRemove) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, kDecoderPCMu));
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu, kCodecName));
   EXPECT_EQ(1, db.Size());
   EXPECT_FALSE(db.Empty());
   EXPECT_EQ(DecoderDatabase::kOK, db.Remove(kPayloadType));
@@ -44,14 +45,17 @@ TEST(DecoderDatabase, InsertAndRemove) {
 TEST(DecoderDatabase, GetDecoderInfo) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, kDecoderPCMu));
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu, kCodecName));
   const DecoderDatabase::DecoderInfo* info;
   info = db.GetDecoderInfo(kPayloadType);
   ASSERT_TRUE(info != NULL);
-  EXPECT_EQ(kDecoderPCMu, info->codec_type);
+  EXPECT_EQ(NetEqDecoder::kDecoderPCMu, info->codec_type);
   EXPECT_EQ(NULL, info->decoder);
   EXPECT_EQ(8000, info->fs_hz);
+  EXPECT_EQ(kCodecName, info->name);
   EXPECT_FALSE(info->external);
   info = db.GetDecoderInfo(kPayloadType + 1);  // Other payload type.
   EXPECT_TRUE(info == NULL);  // Should not be found.
@@ -60,19 +64,24 @@ TEST(DecoderDatabase, GetDecoderInfo) {
 TEST(DecoderDatabase, GetRtpPayloadType) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, kDecoderPCMu));
-  EXPECT_EQ(kPayloadType, db.GetRtpPayloadType(kDecoderPCMu));
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCMu, kCodecName));
+  EXPECT_EQ(kPayloadType, db.GetRtpPayloadType(NetEqDecoder::kDecoderPCMu));
   const uint8_t expected_value = DecoderDatabase::kRtpPayloadTypeError;
   EXPECT_EQ(expected_value,
-            db.GetRtpPayloadType(kDecoderISAC));  // iSAC is not registered.
+            db.GetRtpPayloadType(
+                NetEqDecoder::kDecoderISAC));  // iSAC is not registered.
 }
 
 TEST(DecoderDatabase, GetDecoder) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadType, kDecoderPCM16B));
+            db.RegisterPayload(kPayloadType, NetEqDecoder::kDecoderPCM16B,
+                               kCodecName));
   AudioDecoder* dec = db.GetDecoder(kPayloadType);
   ASSERT_TRUE(dec != NULL);
 }
@@ -85,14 +94,18 @@ TEST(DecoderDatabase, TypeTests) {
   const uint8_t kPayloadTypeRed = 101;
   const uint8_t kPayloadNotUsed = 102;
   // Load into database.
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypePcmU, NetEqDecoder::kDecoderPCMu, "pcmu"));
   EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypePcmU, kDecoderPCMu));
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypeCng, kDecoderCNGnb));
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypeDtmf, kDecoderAVT));
-  EXPECT_EQ(DecoderDatabase::kOK,
-            db.RegisterPayload(kPayloadTypeRed, kDecoderRED));
+            db.RegisterPayload(kPayloadTypeCng, NetEqDecoder::kDecoderCNGnb,
+                               "cng-nb"));
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypeDtmf, NetEqDecoder::kDecoderAVT, "avt"));
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypeRed, NetEqDecoder::kDecoderRED, "red"));
   EXPECT_EQ(4, db.Size());
   // Test.
   EXPECT_FALSE(db.IsComfortNoise(kPayloadNotUsed));
@@ -101,8 +114,8 @@ TEST(DecoderDatabase, TypeTests) {
   EXPECT_FALSE(db.IsComfortNoise(kPayloadTypePcmU));
   EXPECT_FALSE(db.IsDtmf(kPayloadTypePcmU));
   EXPECT_FALSE(db.IsRed(kPayloadTypePcmU));
-  EXPECT_FALSE(db.IsType(kPayloadTypePcmU, kDecoderISAC));
-  EXPECT_TRUE(db.IsType(kPayloadTypePcmU, kDecoderPCMu));
+  EXPECT_FALSE(db.IsType(kPayloadTypePcmU, NetEqDecoder::kDecoderISAC));
+  EXPECT_TRUE(db.IsType(kPayloadTypePcmU, NetEqDecoder::kDecoderPCMu));
   EXPECT_TRUE(db.IsComfortNoise(kPayloadTypeCng));
   EXPECT_TRUE(db.IsDtmf(kPayloadTypeDtmf));
   EXPECT_TRUE(db.IsRed(kPayloadTypeRed));
@@ -111,11 +124,12 @@ TEST(DecoderDatabase, TypeTests) {
 TEST(DecoderDatabase, ExternalDecoder) {
   DecoderDatabase db;
   const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   MockAudioDecoder decoder;
   // Load into database.
   EXPECT_EQ(DecoderDatabase::kOK,
-            db.InsertExternal(kPayloadType, kDecoderPCMu, 8000,
-                               &decoder));
+            db.InsertExternal(kPayloadType, NetEqDecoder::kDecoderPCMu,
+                              kCodecName, 8000, &decoder));
   EXPECT_EQ(1, db.Size());
   // Get decoder and make sure we get the external one.
   EXPECT_EQ(&decoder, db.GetDecoder(kPayloadType));
@@ -123,7 +137,8 @@ TEST(DecoderDatabase, ExternalDecoder) {
   const DecoderDatabase::DecoderInfo* info;
   info = db.GetDecoderInfo(kPayloadType);
   ASSERT_TRUE(info != NULL);
-  EXPECT_EQ(kDecoderPCMu, info->codec_type);
+  EXPECT_EQ(NetEqDecoder::kDecoderPCMu, info->codec_type);
+  EXPECT_EQ(kCodecName, info->name);
   EXPECT_EQ(&decoder, info->decoder);
   EXPECT_EQ(8000, info->fs_hz);
   EXPECT_TRUE(info->external);
@@ -143,8 +158,9 @@ TEST(DecoderDatabase, CheckPayloadTypes) {
   // matter for the test).
   const int kNumPayloads = 10;
   for (uint8_t payload_type = 0; payload_type < kNumPayloads; ++payload_type) {
-    EXPECT_EQ(DecoderDatabase::kOK,
-              db.RegisterPayload(payload_type, kDecoderArbitrary));
+    EXPECT_EQ(
+        DecoderDatabase::kOK,
+        db.RegisterPayload(payload_type, NetEqDecoder::kDecoderArbitrary, ""));
   }
   PacketList packet_list;
   for (int i = 0; i < kNumPayloads + 1; ++i) {
@@ -172,13 +188,22 @@ TEST(DecoderDatabase, CheckPayloadTypes) {
   }
 }
 
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define IF_ISAC(x) x
+#else
+#define IF_ISAC(x) DISABLED_##x
+#endif
+
 // Test the methods for setting and getting active speech and CNG decoders.
-TEST(DecoderDatabase, ActiveDecoders) {
+TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
   DecoderDatabase db;
   // Load payload types.
-  ASSERT_EQ(DecoderDatabase::kOK, db.RegisterPayload(0, kDecoderPCMu));
-  ASSERT_EQ(DecoderDatabase::kOK, db.RegisterPayload(103, kDecoderISAC));
-  ASSERT_EQ(DecoderDatabase::kOK, db.RegisterPayload(13, kDecoderCNGnb));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(0, NetEqDecoder::kDecoderPCMu, "pcmu"));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(103, NetEqDecoder::kDecoderISAC, "isac"));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(13, NetEqDecoder::kDecoderCNGnb, "cng-nb"));
   // Verify that no decoders are active from the start.
   EXPECT_EQ(NULL, db.GetActiveDecoder());
   EXPECT_EQ(NULL, db.GetActiveCngDecoder());
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/defines.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/defines.h
index 33d1bd9c3f..3ed6b61889 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/defines.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/defines.h
@@ -18,6 +18,7 @@ enum Operations {
   kMerge,
   kExpand,
   kAccelerate,
+  kFastAccelerate,
   kPreemptiveExpand,
   kRfc3389Cng,
   kRfc3389CngNoPacket,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.cc
index a935561eff..806d02b8de 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -17,12 +17,12 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
-DelayManager::DelayManager(int max_packets_in_buffer,
+DelayManager::DelayManager(size_t max_packets_in_buffer,
                            DelayPeakDetector* peak_detector)
     : first_packet_received_(false),
       max_packets_in_buffer_(max_packets_in_buffer),
@@ -239,7 +239,8 @@ void DelayManager::LimitTargetLevel() {
   }
 
   // Shift to Q8, then 75%.;
-  int max_buffer_packets_q8 = (3 * (max_packets_in_buffer_ << 8)) / 4;
+  int max_buffer_packets_q8 =
+      static_cast<int>((3 * (max_packets_in_buffer_ << 8)) / 4);
   target_level_ = std::min(target_level_, max_buffer_packets_q8);
 
   // Sanity check, at least 1 packet (in Q8).
@@ -372,11 +373,11 @@ int DelayManager::TargetLevel() const {
 }
 
 void DelayManager::LastDecoderType(NetEqDecoder decoder_type) {
-  if (decoder_type == kDecoderAVT ||
-      decoder_type == kDecoderCNGnb ||
-      decoder_type == kDecoderCNGwb ||
-      decoder_type == kDecoderCNGswb32kHz ||
-      decoder_type == kDecoderCNGswb48kHz) {
+  if (decoder_type == NetEqDecoder::kDecoderAVT ||
+      decoder_type == NetEqDecoder::kDecoderCNGnb ||
+      decoder_type == NetEqDecoder::kDecoderCNGwb ||
+      decoder_type == NetEqDecoder::kDecoderCNGswb32kHz ||
+      decoder_type == NetEqDecoder::kDecoderCNGswb48kHz) {
     last_pack_cng_or_dtmf_ = 1;
   } else if (last_pack_cng_or_dtmf_ != 0) {
     last_pack_cng_or_dtmf_ = -1;
@@ -389,7 +390,8 @@ bool DelayManager::SetMinimumDelay(int delay_ms) {
   // |max_packets_in_buffer_|.
   if ((maximum_delay_ms_ > 0 && delay_ms > maximum_delay_ms_) ||
       (packet_len_ms_ > 0 &&
-          delay_ms > 3 * max_packets_in_buffer_ * packet_len_ms_ / 4)) {
+       delay_ms >
+           static_cast<int>(3 * max_packets_in_buffer_ * packet_len_ms_ / 4))) {
     return false;
   }
   minimum_delay_ms_ = delay_ms;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.h
index 33c4a40a6a..785fced15d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager.h
@@ -32,7 +32,7 @@ class DelayManager {
   // buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
   // is the number of packet slots in the buffer). Supply a PeakDetector
   // object to the DelayManager.
-  DelayManager(int max_packets_in_buffer, DelayPeakDetector* peak_detector);
+  DelayManager(size_t max_packets_in_buffer, DelayPeakDetector* peak_detector);
 
   virtual ~DelayManager();
 
@@ -132,7 +132,7 @@ class DelayManager {
   void LimitTargetLevel();
 
   bool first_packet_received_;
-  const int max_packets_in_buffer_;  // Capacity of the packet buffer.
+  const size_t max_packets_in_buffer_;  // Capacity of the packet buffer.
   IATVector iat_vector_;  // Histogram of inter-arrival times.
   int iat_factor_;  // Forgetting factor for updating the IAT histogram (Q15).
   int packet_iat_count_ms_;  // Milliseconds elapsed since last packet.
@@ -157,7 +157,7 @@ class DelayManager {
   DelayPeakDetector& peak_detector_;
   int last_pack_cng_or_dtmf_;
 
-  DISALLOW_COPY_AND_ASSIGN(DelayManager);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
index 30e7647bc4..f231c3da30 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -88,8 +88,8 @@ TEST_F(DelayManagerTest, VectorInitialization) {
   const DelayManager::IATVector& vec = dm_->iat_vector();
   double sum = 0.0;
   for (size_t i = 0; i < vec.size(); i++) {
-    EXPECT_NEAR(ldexp(pow(0.5, static_cast<int>(i + 1)), 30), vec[i], 65536);
-    // Tolerance 65536 in Q30 corresponds to a delta of approximately 0.00006.
+    EXPECT_NEAR(ldexp(pow(0.5, static_cast<int>(i + 1)), 30), vec[i], 65537);
+    // Tolerance 65537 in Q30 corresponds to a delta of approximately 0.00006.
     sum += vec[i];
   }
   EXPECT_EQ(1 << 30, static_cast<int>(sum));  // Should be 1 in Q30.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
index 5996d7d197..712c7788ac 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
@@ -21,6 +21,8 @@ namespace webrtc {
 // peak-mode is engaged and the DelayManager asks the DelayPeakDetector for
 // the worst peak height.
 
+DelayPeakDetector::~DelayPeakDetector() = default;
+
 DelayPeakDetector::DelayPeakDetector()
   : peak_found_(false),
     peak_detection_threshold_(0),
@@ -40,6 +42,10 @@ void DelayPeakDetector::SetPacketAudioLength(int length_ms) {
   }
 }
 
+bool DelayPeakDetector::peak_found() {
+  return peak_found_;
+}
+
 int DelayPeakDetector::MaxPeakHeight() const {
   int max_height = -1;  // Returns -1 for an empty history.
   std::list<Peak>::const_iterator it;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
index 8bf6aba8b5..69433b4524 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
@@ -22,7 +22,7 @@ namespace webrtc {
 class DelayPeakDetector {
  public:
   DelayPeakDetector();
-  virtual ~DelayPeakDetector() {}
+  virtual ~DelayPeakDetector();
   virtual void Reset();
 
   // Notifies the DelayPeakDetector of how much audio data is carried in each
@@ -31,7 +31,7 @@ class DelayPeakDetector {
 
   // Returns true if peak-mode is active. That is, delay peaks were observed
   // recently.
-  virtual bool peak_found() { return peak_found_; }
+  virtual bool peak_found();
 
   // Calculates and returns the maximum delay peak height. Returns -1 if no
   // delay peaks have been observed recently. The unit is number of packets.
@@ -69,7 +69,7 @@ class DelayPeakDetector {
   int peak_detection_threshold_;
   int peak_period_counter_ms_;
 
-  DISALLOW_COPY_AND_ASSIGN(DelayPeakDetector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DelayPeakDetector);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.cc
index 7451ae26f8..4188914c86 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -99,13 +99,13 @@ int DspHelper::RampSignal(AudioMultiVector* signal,
   return end_factor;
 }
 
-void DspHelper::PeakDetection(int16_t* data, int data_length,
-                              int num_peaks, int fs_mult,
-                              int* peak_index, int16_t* peak_value) {
-  int16_t min_index = 0;
-  int16_t max_index = 0;
+void DspHelper::PeakDetection(int16_t* data, size_t data_length,
+                              size_t num_peaks, int fs_mult,
+                              size_t* peak_index, int16_t* peak_value) {
+  size_t min_index = 0;
+  size_t max_index = 0;
 
-  for (int i = 0; i <= num_peaks - 1; i++) {
+  for (size_t i = 0; i <= num_peaks - 1; i++) {
     if (num_peaks == 1) {
       // Single peak.  The parabola fit assumes that an extra point is
       // available; worst case it gets a zero on the high end of the signal.
@@ -117,7 +117,7 @@ void DspHelper::PeakDetection(int16_t* data, int data_length,
     peak_index[i] = WebRtcSpl_MaxIndexW16(data, data_length - 1);
 
     if (i != num_peaks - 1) {
-      min_index = std::max(0, peak_index[i] - 2);
+      min_index = (peak_index[i] > 2) ? (peak_index[i] - 2) : 0;
       max_index = std::min(data_length - 1, peak_index[i] + 2);
     }
 
@@ -148,7 +148,7 @@ void DspHelper::PeakDetection(int16_t* data, int data_length,
 }
 
 void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
-                             int* peak_index, int16_t* peak_value) {
+                             size_t* peak_index, int16_t* peak_value) {
   uint16_t fit_index[13];
   if (fs_mult == 1) {
     fit_index[0] = 0;
@@ -235,16 +235,16 @@ void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
   }
 }
 
-int DspHelper::MinDistortion(const int16_t* signal, int min_lag,
-                             int max_lag, int length,
-                             int32_t* distortion_value) {
-  int best_index = -1;
+size_t DspHelper::MinDistortion(const int16_t* signal, size_t min_lag,
+                                size_t max_lag, size_t length,
+                                int32_t* distortion_value) {
+  size_t best_index = 0;
   int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
-  for (int i = min_lag; i <= max_lag; i++) {
+  for (size_t i = min_lag; i <= max_lag; i++) {
     int32_t sum_diff = 0;
     const int16_t* data1 = signal;
     const int16_t* data2 = signal - i;
-    for (int j = 0; j < length; j++) {
+    for (size_t j = 0; j < length; j++) {
       sum_diff += WEBRTC_SPL_ABS_W32(data1[j] - data2[j]);
     }
     // Compare with previous minimum.
@@ -272,7 +272,7 @@ void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2,
 }
 
 void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
-                             int16_t* factor, int16_t increment,
+                             int16_t* factor, int increment,
                              int16_t* output) {
   uint16_t factor_16b = *factor;
   int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
@@ -284,7 +284,7 @@ void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
   *factor = factor_16b;
 }
 
-void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
+void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
   int32_t factor = (16384 << 6) + 32;
   for (size_t i = 0; i < length; i++) {
     signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
@@ -293,15 +293,15 @@ void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
 }
 
 int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
-                                int output_length, int input_rate_hz,
+                                size_t output_length, int input_rate_hz,
                                 bool compensate_delay, int16_t* output) {
   // Set filter parameters depending on input frequency.
   // NOTE: The phase delay values are wrong compared to the true phase delay
   // of the filters. However, the error is preserved (through the +1 term) for
   // consistency.
   const int16_t* filter_coefficients;  // Filter coefficients.
-  int16_t filter_length;  // Number of coefficients.
-  int16_t filter_delay;  // Phase delay in samples.
+  size_t filter_length;  // Number of coefficients.
+  size_t filter_delay;  // Phase delay in samples.
   int16_t factor;  // Conversion rate (inFsHz / 8000).
   switch (input_rate_hz) {
     case 8000: {
@@ -345,9 +345,8 @@ int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
 
   // Returns -1 if input signal is too short; 0 otherwise.
   return WebRtcSpl_DownsampleFast(
-      &input[filter_length - 1], static_cast<int>(input_length) -
-      (filter_length - 1), output, output_length, filter_coefficients,
-      filter_length, factor, filter_delay);
+      &input[filter_length - 1], input_length - filter_length + 1, output,
+      output_length, filter_coefficients, filter_length, factor, filter_delay);
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.h
index af4f4d6c88..269c2eb0f2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dsp_helper.h
@@ -78,9 +78,9 @@ class DspHelper {
   // locations and values are written to the arrays |peak_index| and
   // |peak_value|, respectively. Both arrays must hold at least |num_peaks|
   // elements.
-  static void PeakDetection(int16_t* data, int data_length,
-                            int num_peaks, int fs_mult,
-                            int* peak_index, int16_t* peak_value);
+  static void PeakDetection(int16_t* data, size_t data_length,
+                            size_t num_peaks, int fs_mult,
+                            size_t* peak_index, int16_t* peak_value);
 
   // Estimates the height and location of a maximum. The three values in the
   // array |signal_points| are used as basis for a parabolic fit, which is then
@@ -89,14 +89,15 @@ class DspHelper {
   // |peak_index| and |peak_value| is given in the full sample rate, as
   // indicated by the sample rate multiplier |fs_mult|.
   static void ParabolicFit(int16_t* signal_points, int fs_mult,
-                           int* peak_index, int16_t* peak_value);
+                           size_t* peak_index, int16_t* peak_value);
 
   // Calculates the sum-abs-diff for |signal| when compared to a displaced
   // version of itself. Returns the displacement lag that results in the minimum
   // distortion. The resulting distortion is written to |distortion_value|.
   // The values of |min_lag| and |max_lag| are boundaries for the search.
-  static int MinDistortion(const int16_t* signal, int min_lag,
-                           int max_lag, int length, int32_t* distortion_value);
+  static size_t MinDistortion(const int16_t* signal, size_t min_lag,
+                           size_t max_lag, size_t length,
+                           int32_t* distortion_value);
 
   // Mixes |length| samples from |input1| and |input2| together and writes the
   // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and
@@ -110,11 +111,11 @@ class DspHelper {
   // sample and increases the gain by |increment| (Q20) for each sample. The
   // result is written to |output|. |length| samples are processed.
   static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
-                           int16_t increment, int16_t* output);
+                           int increment, int16_t* output);
 
   // Starts at unity gain and gradually fades out |signal|. For each sample,
   // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
-  static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
+  static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
 
   // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
   // has |input_length| samples, and the method will write |output_length|
@@ -122,14 +123,14 @@ class DspHelper {
   // filters if |compensate_delay| is true. Returns -1 if the input is too short
   // to produce |output_length| samples, otherwise 0.
   static int DownsampleTo4kHz(const int16_t* input, size_t input_length,
-                              int output_length, int input_rate_hz,
+                              size_t output_length, int input_rate_hz,
                               bool compensate_delay, int16_t* output);
 
  private:
   // Table of constants used in method DspHelper::ParabolicFit().
   static const int16_t kParabolaCoefficients[17][3];
 
-  DISALLOW_COPY_AND_ASSIGN(DspHelper);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
index b07d561993..779d1d340b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
@@ -13,6 +13,9 @@
 #include <assert.h>
 #include <algorithm>  // max
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
 // longer required, this #define should be removed (and the code that it
 // enables).
@@ -20,6 +23,16 @@
 
 namespace webrtc {
 
+DtmfBuffer::DtmfBuffer(int fs_hz) {
+  SetSampleRate(fs_hz);
+}
+
+DtmfBuffer::~DtmfBuffer() = default;
+
+void DtmfBuffer::Flush() {
+  buffer_.clear();
+}
+
 // The ParseEvent method parses 4 bytes from |payload| according to this format
 // from RFC 4733:
 //
@@ -57,10 +70,10 @@ int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
                            const uint8_t* payload,
                            size_t payload_length_bytes,
                            DtmfEvent* event) {
-  if (!payload || !event) {
-    return kInvalidPointer;
-  }
+  RTC_CHECK(payload);
+  RTC_CHECK(event);
   if (payload_length_bytes < 4) {
+    LOG(LS_WARNING) << "ParseEvent payload too short";
     return kPayloadTooShort;
   }
 
@@ -88,6 +101,7 @@ int DtmfBuffer::InsertEvent(const DtmfEvent& event) {
   if (event.event_no < 0 || event.event_no > 15 ||
       event.volume < 0 || event.volume > 36 ||
       event.duration <= 0 || event.duration > 65535) {
+    LOG(LS_WARNING) << "InsertEvent invalid parameters";
     return kInvalidEventParameters;
   }
   DtmfList::iterator it = buffer_.begin();
@@ -173,6 +187,14 @@ bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
   return false;
 }
 
+size_t DtmfBuffer::Length() const {
+  return buffer_.size();
+}
+
+bool DtmfBuffer::Empty() const {
+  return buffer_.empty();
+}
+
 int DtmfBuffer::SetSampleRate(int fs_hz) {
   if (fs_hz != 8000 &&
       fs_hz != 16000 &&
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
index 5da3a16a2d..1f415ce81f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
@@ -55,14 +55,12 @@ class DtmfBuffer {
   };
 
   // Set up the buffer for use at sample rate |fs_hz|.
-  explicit DtmfBuffer(int fs_hz) {
-    SetSampleRate(fs_hz);
-  }
+  explicit DtmfBuffer(int fs_hz);
 
-  virtual ~DtmfBuffer() {}
+  virtual ~DtmfBuffer();
 
   // Flushes the buffer.
-  virtual void Flush() { buffer_.clear(); }
+  virtual void Flush();
 
   // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
   // and write the parsed information into the struct |event|. Input variable
@@ -82,9 +80,9 @@ class DtmfBuffer {
   virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
 
   // Number of events in the buffer.
-  virtual size_t Length() const { return buffer_.size(); }
+  virtual size_t Length() const;
 
-  virtual bool Empty() const { return buffer_.empty(); }
+  virtual bool Empty() const;
 
   // Set a new sample rate.
   virtual int SetSampleRate(int fs_hz);
@@ -109,7 +107,7 @@ class DtmfBuffer {
 
   DtmfList buffer_;
 
-  DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
index 4824b2baca..dad4e76627 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
@@ -77,12 +77,6 @@ TEST(DtmfBuffer, ParseEvent) {
   EXPECT_EQ(timestamp, event.timestamp);
   EXPECT_EQ(volume, event.volume);
 
-  EXPECT_EQ(DtmfBuffer::kInvalidPointer,
-            DtmfBuffer::ParseEvent(timestamp, NULL, 4, &event));
-
-  EXPECT_EQ(DtmfBuffer::kInvalidPointer,
-            DtmfBuffer::ParseEvent(timestamp, payload_ptr, 4, NULL));
-
   EXPECT_EQ(DtmfBuffer::kPayloadTooShort,
             DtmfBuffer::ParseEvent(timestamp, payload_ptr, 3, &event));
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
index 3429bcdadc..f4d5190c61 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
@@ -149,18 +149,18 @@ void DtmfToneGenerator::Reset() {
 }
 
 // Generate num_samples of DTMF signal and write to |output|.
-int DtmfToneGenerator::Generate(int num_samples,
+int DtmfToneGenerator::Generate(size_t num_samples,
                                 AudioMultiVector* output) {
   if (!initialized_) {
     return kNotInitialized;
   }
 
-  if (num_samples < 0 || !output) {
+  if (!output) {
     return kParameterError;
   }
 
   output->AssertSize(num_samples);
-  for (int i = 0; i < num_samples; ++i) {
+  for (size_t i = 0; i < num_samples; ++i) {
     // Use recursion formula y[n] = a * y[n - 1] - y[n - 2].
     int16_t temp_val_low = ((coeff1_ * sample_history1_[1] + 8192) >> 14)
         - sample_history1_[0];
@@ -186,7 +186,11 @@ int DtmfToneGenerator::Generate(int num_samples,
     output->CopyChannel(0, channel);
   }
 
-  return num_samples;
+  return static_cast<int>(num_samples);
+}
+
+bool DtmfToneGenerator::initialized() const {
+  return initialized_;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
index 232eba4cb0..36d902ad3f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
@@ -30,8 +30,8 @@ class DtmfToneGenerator {
   virtual ~DtmfToneGenerator() {}
   virtual int Init(int fs, int event, int attenuation);
   virtual void Reset();
-  virtual int Generate(int num_samples, AudioMultiVector* output);
-  virtual bool initialized() const { return initialized_; }
+  virtual int Generate(size_t num_samples, AudioMultiVector* output);
+  virtual bool initialized() const;
 
  private:
   static const int kCoeff1[4][16];  // 1st oscillator model coefficient table.
@@ -48,7 +48,7 @@ class DtmfToneGenerator {
   int16_t sample_history1_[2];  // Last 2 samples for the 1st oscillator.
   int16_t sample_history2_[2];  // Last 2 samples for the 2nd oscillator.
 
-  DISALLOW_COPY_AND_ASSIGN(DtmfToneGenerator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DtmfToneGenerator);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
index ccd7fa606f..a55e6c9028 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
@@ -171,8 +171,6 @@ TEST(DtmfToneGenerator, TestErrors) {
   // Initialize with valid parameters.
   ASSERT_EQ(0, tone_gen.Init(fs, event, attenuation));
   EXPECT_TRUE(tone_gen.initialized());
-  // Negative number of samples.
-  EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Generate(-1, &signal));
   // NULL pointer to destination.
   EXPECT_EQ(DtmfToneGenerator::kParameterError,
             tone_gen.Generate(kNumSamples, NULL));
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.cc
index d13c2cdf8d..ef7af46597 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.cc
@@ -16,14 +16,45 @@
 #include <algorithm>  // min, max
 #include <limits>  // numeric_limits<T>
 
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 namespace webrtc {
 
+Expand::Expand(BackgroundNoise* background_noise,
+               SyncBuffer* sync_buffer,
+               RandomVector* random_vector,
+               StatisticsCalculator* statistics,
+               int fs,
+               size_t num_channels)
+    : random_vector_(random_vector),
+      sync_buffer_(sync_buffer),
+      first_expand_(true),
+      fs_hz_(fs),
+      num_channels_(num_channels),
+      consecutive_expands_(0),
+      background_noise_(background_noise),
+      statistics_(statistics),
+      overlap_length_(5 * fs / 8000),
+      lag_index_direction_(0),
+      current_lag_index_(0),
+      stop_muting_(false),
+      expand_duration_samples_(0),
+      channel_parameters_(new ChannelParameters[num_channels_]) {
+  assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
+  assert(fs <= static_cast<int>(kMaxSampleRate));  // Should not be possible.
+  assert(num_channels_ > 0);
+  memset(expand_lags_, 0, sizeof(expand_lags_));
+  Reset();
+}
+
+Expand::~Expand() = default;
+
 void Expand::Reset() {
   first_expand_ = true;
   consecutive_expands_ = 0;
@@ -41,7 +72,7 @@ int Expand::Process(AudioMultiVector* output) {
   int16_t temp_data[kTempDataSize];  // TODO(hlundin) Remove this.
   int16_t* voiced_vector_storage = temp_data;
   int16_t* voiced_vector = &voiced_vector_storage[overlap_length_];
-  static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
   int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
   int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
   int16_t* noise_vector = unvoiced_array_memory + kNoiseLpcOrder;
@@ -52,10 +83,11 @@ int Expand::Process(AudioMultiVector* output) {
     // Perform initial setup if this is the first expansion since last reset.
     AnalyzeSignal(random_vector);
     first_expand_ = false;
+    expand_duration_samples_ = 0;
   } else {
     // This is not the first expansion, parameters are already estimated.
     // Extract a noise segment.
-    int16_t rand_length = max_lag_;
+    size_t rand_length = max_lag_;
     // This only applies to SWB where length could be larger than 256.
     assert(rand_length <= kMaxSampleRate / 8000 * 120 + 30);
     GenerateRandomVector(2, rand_length, random_vector);
@@ -87,7 +119,7 @@ int Expand::Process(AudioMultiVector* output) {
       WebRtcSpl_ScaleAndAddVectorsWithRound(
           &parameters.expand_vector0[expansion_vector_position], 3,
           &parameters.expand_vector1[expansion_vector_position], 1, 2,
-          voiced_vector_storage, static_cast<int>(temp_length));
+          voiced_vector_storage, temp_length);
     } else if (current_lag_index_ == 2) {
       // Mix 1/2 of expand_vector0 with 1/2 of expand_vector1.
       assert(expansion_vector_position + temp_length <=
@@ -97,7 +129,7 @@ int Expand::Process(AudioMultiVector* output) {
       WebRtcSpl_ScaleAndAddVectorsWithRound(
           &parameters.expand_vector0[expansion_vector_position], 1,
           &parameters.expand_vector1[expansion_vector_position], 1, 1,
-          voiced_vector_storage, static_cast<int>(temp_length));
+          voiced_vector_storage, temp_length);
     }
 
     // Get tapering window parameters. Values are in Q15.
@@ -164,10 +196,10 @@ int Expand::Process(AudioMultiVector* output) {
     WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector,
                                     parameters.ar_gain, add_constant,
                                     parameters.ar_gain_scale,
-                                    static_cast<int>(current_lag));
+                                    current_lag);
     WebRtcSpl_FilterARFastQ12(scaled_random_vector, unvoiced_vector,
                               parameters.ar_filter, kUnvoicedLpcOrder + 1,
-                              static_cast<int>(current_lag));
+                              current_lag);
     memcpy(parameters.ar_filter_state,
            &(unvoiced_vector[current_lag - kUnvoicedLpcOrder]),
            sizeof(int16_t) * kUnvoicedLpcOrder);
@@ -180,7 +212,8 @@ int Expand::Process(AudioMultiVector* output) {
     //  (>= 31 .. <= 63) * fs_mult  => go from 1 to 0 in about 16 ms;
     //   >= 64 * fs_mult            => go from 1 to 0 in about 32 ms.
     // temp_shift = getbits(max_lag_) - 5.
-    int temp_shift = (31 - WebRtcSpl_NormW32(max_lag_)) - 5;
+    int temp_shift =
+        (31 - WebRtcSpl_NormW32(rtc::checked_cast<int32_t>(max_lag_))) - 5;
     int16_t mix_factor_increment = 256 >> temp_shift;
     if (stop_muting_) {
       mix_factor_increment = 0;
@@ -188,24 +221,24 @@ int Expand::Process(AudioMultiVector* output) {
 
     // Create combined signal by shifting in more and more of unvoiced part.
     temp_shift = 8 - temp_shift;  // = getbits(mix_factor_increment).
-    size_t temp_lenght = (parameters.current_voice_mix_factor -
+    size_t temp_length = (parameters.current_voice_mix_factor -
         parameters.voice_mix_factor) >> temp_shift;
-    temp_lenght = std::min(temp_lenght, current_lag);
-    DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_lenght,
+    temp_length = std::min(temp_length, current_lag);
+    DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_length,
                          &parameters.current_voice_mix_factor,
                          mix_factor_increment, temp_data);
 
     // End of cross-fading period was reached before end of expanded signal
     // path. Mix the rest with a fixed mixing factor.
-    if (temp_lenght < current_lag) {
+    if (temp_length < current_lag) {
       if (mix_factor_increment != 0) {
         parameters.current_voice_mix_factor = parameters.voice_mix_factor;
       }
-      int temp_scale = 16384 - parameters.current_voice_mix_factor;
+      int16_t temp_scale = 16384 - parameters.current_voice_mix_factor;
       WebRtcSpl_ScaleAndAddVectorsWithRound(
-          voiced_vector + temp_lenght, parameters.current_voice_mix_factor,
-          unvoiced_vector + temp_lenght, temp_scale, 14,
-          temp_data + temp_lenght, static_cast<int>(current_lag - temp_lenght));
+          voiced_vector + temp_length, parameters.current_voice_mix_factor,
+          unvoiced_vector + temp_length, temp_scale, 14,
+          temp_data + temp_length, current_lag - temp_length);
     }
 
     // Select muting slope depending on how many consecutive expands we have
@@ -213,14 +246,12 @@ int Expand::Process(AudioMultiVector* output) {
     if (consecutive_expands_ == 3) {
       // Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
       // mute_slope = 0.0010 / fs_mult in Q20.
-      parameters.mute_slope = std::max(parameters.mute_slope,
-                                       static_cast<int16_t>(1049 / fs_mult));
+      parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
     }
     if (consecutive_expands_ == 7) {
       // Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
       // mute_slope = 0.0020 / fs_mult in Q20.
-      parameters.mute_slope = std::max(parameters.mute_slope,
-                                       static_cast<int16_t>(2097 / fs_mult));
+      parameters.mute_slope = std::max(parameters.mute_slope, 2097 / fs_mult);
     }
 
     // Mute segment according to slope value.
@@ -228,7 +259,7 @@ int Expand::Process(AudioMultiVector* output) {
       // Mute to the previous level, then continue with the muting.
       WebRtcSpl_AffineTransformVector(temp_data, temp_data,
                                       parameters.mute_factor, 8192,
-                                      14, static_cast<int>(current_lag));
+                                      14, current_lag);
 
       if (!stop_muting_) {
         DspHelper::MuteSignal(temp_data, parameters.mute_slope, current_lag);
@@ -274,6 +305,10 @@ int Expand::Process(AudioMultiVector* output) {
   // Increase call number and cap it.
   consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ?
       kMaxConsecutiveExpands : consecutive_expands_ + 1;
+  expand_duration_samples_ += output->Size();
+  // Clamp the duration counter at 2 seconds.
+  expand_duration_samples_ =
+      std::min(expand_duration_samples_, rtc::checked_cast<size_t>(fs_hz_ * 2));
   return 0;
 }
 
@@ -281,6 +316,8 @@ void Expand::SetParametersForNormalAfterExpand() {
   current_lag_index_ = 0;
   lag_index_direction_ = 0;
   stop_muting_ = true;  // Do not mute signal any more.
+  statistics_->LogDelayedPacketOutageEvent(
+      rtc::checked_cast<int>(expand_duration_samples_) / (fs_hz_ / 1000));
 }
 
 void Expand::SetParametersForMergeAfterExpand() {
@@ -289,6 +326,10 @@ void Expand::SetParametersForMergeAfterExpand() {
   stop_muting_ = true;
 }
 
+size_t Expand::overlap_length() const {
+  return overlap_length_;
+}
+
 void Expand::InitializeForAnExpandPeriod() {
   lag_index_direction_ = 1;
   current_lag_index_ = -1;
@@ -311,26 +352,26 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   int32_t auto_correlation[kUnvoicedLpcOrder + 1];
   int16_t reflection_coeff[kUnvoicedLpcOrder];
   int16_t correlation_vector[kMaxSampleRate / 8000 * 102];
-  int best_correlation_index[kNumCorrelationCandidates];
+  size_t best_correlation_index[kNumCorrelationCandidates];
   int16_t best_correlation[kNumCorrelationCandidates];
-  int16_t best_distortion_index[kNumCorrelationCandidates];
+  size_t best_distortion_index[kNumCorrelationCandidates];
   int16_t best_distortion[kNumCorrelationCandidates];
   int32_t correlation_vector2[(99 * kMaxSampleRate / 8000) + 1];
   int32_t best_distortion_w32[kNumCorrelationCandidates];
-  static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
   int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
   int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
 
   int fs_mult = fs_hz_ / 8000;
 
   // Pre-calculate common multiplications with fs_mult.
-  int fs_mult_4 = fs_mult * 4;
-  int fs_mult_20 = fs_mult * 20;
-  int fs_mult_120 = fs_mult * 120;
-  int fs_mult_dist_len = fs_mult * kDistortionLength;
-  int fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength;
+  size_t fs_mult_4 = static_cast<size_t>(fs_mult * 4);
+  size_t fs_mult_20 = static_cast<size_t>(fs_mult * 20);
+  size_t fs_mult_120 = static_cast<size_t>(fs_mult * 120);
+  size_t fs_mult_dist_len = fs_mult * kDistortionLength;
+  size_t fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength;
 
-  const size_t signal_length = 256 * fs_mult;
+  const size_t signal_length = static_cast<size_t>(256 * fs_mult);
   const int16_t* audio_history =
       &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length];
 
@@ -338,8 +379,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   InitializeForAnExpandPeriod();
 
   // Calculate correlation in downsampled domain (4 kHz sample rate).
-  int16_t correlation_scale;
-  int correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
+  int correlation_scale;
+  size_t correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
   // If it is decided to break bit-exactness |correlation_length| should be
   // initialized to the return value of Correlation().
   Correlation(audio_history, signal_length, correlation_vector,
@@ -358,11 +399,11 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
 
   // Calculate distortion around the |kNumCorrelationCandidates| best lags.
   int distortion_scale = 0;
-  for (int i = 0; i < kNumCorrelationCandidates; i++) {
-    int16_t min_index = std::max(fs_mult_20,
-                                 best_correlation_index[i] - fs_mult_4);
-    int16_t max_index = std::min(fs_mult_120 - 1,
-                                 best_correlation_index[i] + fs_mult_4);
+  for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
+    size_t min_index = std::max(fs_mult_20,
+                                best_correlation_index[i] - fs_mult_4);
+    size_t max_index = std::min(fs_mult_120 - 1,
+                                best_correlation_index[i] + fs_mult_4);
     best_distortion_index[i] = DspHelper::MinDistortion(
         &(audio_history[signal_length - fs_mult_dist_len]), min_index,
         max_index, fs_mult_dist_len, &best_distortion_w32[i]);
@@ -376,8 +417,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   // Find the maximizing index |i| of the cost function
   // f[i] = best_correlation[i] / best_distortion[i].
   int32_t best_ratio = std::numeric_limits<int32_t>::min();
-  int best_index = -1;
-  for (int i = 0; i < kNumCorrelationCandidates; ++i) {
+  size_t best_index = std::numeric_limits<size_t>::max();
+  for (size_t i = 0; i < kNumCorrelationCandidates; ++i) {
     int32_t ratio;
     if (best_distortion[i] > 0) {
       ratio = (best_correlation[i] << 16) / best_distortion[i];
@@ -392,20 +433,20 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     }
   }
 
-  int distortion_lag = best_distortion_index[best_index];
-  int correlation_lag = best_correlation_index[best_index];
+  size_t distortion_lag = best_distortion_index[best_index];
+  size_t correlation_lag = best_correlation_index[best_index];
   max_lag_ = std::max(distortion_lag, correlation_lag);
 
   // Calculate the exact best correlation in the range between
   // |correlation_lag| and |distortion_lag|.
-  correlation_length = distortion_lag + 10;
-  correlation_length = std::min(correlation_length, fs_mult_120);
-  correlation_length = std::max(correlation_length, 60 * fs_mult);
+  correlation_length =
+      std::max(std::min(distortion_lag + 10, fs_mult_120),
+               static_cast<size_t>(60 * fs_mult));
 
-  int start_index = std::min(distortion_lag, correlation_lag);
-  int correlation_lags = WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag))
-      + 1;
-  assert(correlation_lags <= 99 * fs_mult + 1);  // Cannot be larger.
+  size_t start_index = std::min(distortion_lag, correlation_lag);
+  size_t correlation_lags = static_cast<size_t>(
+      WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag)) + 1);
+  assert(correlation_lags <= static_cast<size_t>(99 * fs_mult + 1));
 
   for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
     ChannelParameters& parameters = channel_parameters_[channel_ix];
@@ -414,9 +455,9 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
         &audio_history[signal_length - correlation_length - start_index
                        - correlation_lags],
                        correlation_length + start_index + correlation_lags - 1);
-    correlation_scale = ((31 - WebRtcSpl_NormW32(signal_max * signal_max))
-        + (31 - WebRtcSpl_NormW32(correlation_length))) - 31;
-    correlation_scale = std::max(static_cast<int16_t>(0), correlation_scale);
+    correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
+        (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
+    correlation_scale = std::max(0, correlation_scale);
 
     // Calculate the correlation, store in |correlation_vector2|.
     WebRtcSpl_CrossCorrelation(
@@ -443,7 +484,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
 
     // Calculate the correlation coefficient between the two portions of the
     // signal.
-    int16_t corr_coefficient;
+    int32_t corr_coefficient;
     if ((energy1 > 0) && (energy2 > 0)) {
       int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
       int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
@@ -452,24 +493,24 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
         // If sum is odd, add 1 to make it even.
         energy1_scale += 1;
       }
-      int16_t scaled_energy1 = energy1 >> energy1_scale;
-      int16_t scaled_energy2 = energy2 >> energy2_scale;
-      int16_t sqrt_energy_product = WebRtcSpl_SqrtFloor(
-          scaled_energy1 * scaled_energy2);
+      int32_t scaled_energy1 = energy1 >> energy1_scale;
+      int32_t scaled_energy2 = energy2 >> energy2_scale;
+      int16_t sqrt_energy_product = static_cast<int16_t>(
+          WebRtcSpl_SqrtFloor(scaled_energy1 * scaled_energy2));
       // Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
       int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
       max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
       corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
                                              sqrt_energy_product);
-      corr_coefficient = std::min(static_cast<int16_t>(16384),
-                                  corr_coefficient);  // Cap at 1.0 in Q14.
+      // Cap at 1.0 in Q14.
+      corr_coefficient = std::min(16384, corr_coefficient);
     } else {
       corr_coefficient = 0;
     }
 
     // Extract the two vectors expand_vector0 and expand_vector1 from
     // |audio_history|.
-    int16_t expansion_length = static_cast<int16_t>(max_lag_ + overlap_length_);
+    size_t expansion_length = max_lag_ + overlap_length_;
     const int16_t* vector1 = &(audio_history[signal_length - expansion_length]);
     const int16_t* vector2 = vector1 - distortion_lag;
     // Normalize the second vector to the same energy as the first.
@@ -478,25 +519,25 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     energy2 = WebRtcSpl_DotProductWithScale(vector2, vector2, expansion_length,
                                             correlation_scale);
     // Confirm that amplitude ratio sqrt(energy1 / energy2) is within 0.5 - 2.0,
-    // i.e., energy1 / energy1 is within 0.25 - 4.
+    // i.e., energy1 / energy2 is within 0.25 - 4.
     int16_t amplitude_ratio;
     if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
       // Energy constraint fulfilled. Use both vectors and scale them
       // accordingly.
-      int16_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
-      int16_t scaled_energy1 = scaled_energy2 - 13;
+      int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+      int32_t scaled_energy1 = scaled_energy2 - 13;
       // Calculate scaled_energy1 / scaled_energy2 in Q13.
       int32_t energy_ratio = WebRtcSpl_DivW32W16(
           WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
-          energy2 >> scaled_energy2);
+          static_cast<int16_t>(energy2 >> scaled_energy2));
       // Calculate sqrt ratio in Q13 (sqrt of en1/en2 in Q26).
-      amplitude_ratio = WebRtcSpl_SqrtFloor(energy_ratio << 13);
+      amplitude_ratio =
+          static_cast<int16_t>(WebRtcSpl_SqrtFloor(energy_ratio << 13));
       // Copy the two vectors and give them the same energy.
       parameters.expand_vector0.Clear();
       parameters.expand_vector0.PushBack(vector1, expansion_length);
       parameters.expand_vector1.Clear();
-      if (parameters.expand_vector1.Size() <
-          static_cast<size_t>(expansion_length)) {
+      if (parameters.expand_vector1.Size() < expansion_length) {
         parameters.expand_vector1.Extend(
             expansion_length - parameters.expand_vector1.Size());
       }
@@ -521,9 +562,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     }
 
     // Set the 3 lag values.
-    int lag_difference = distortion_lag - correlation_lag;
-    if (lag_difference == 0) {
-      // |distortion_lag| and |correlation_lag| are equal.
+    if (distortion_lag == correlation_lag) {
       expand_lags_[0] = distortion_lag;
       expand_lags_[1] = distortion_lag;
       expand_lags_[2] = distortion_lag;
@@ -535,7 +574,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
       // Second lag is the average of the two.
       expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
       // Third lag is the average again, but rounding towards |correlation_lag|.
-      if (lag_difference > 0) {
+      if (distortion_lag > correlation_lag) {
         expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
       } else {
         expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2;
@@ -589,7 +628,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
 
     if (channel_ix == 0) {
       // Extract a noise segment.
-      int16_t noise_length;
+      size_t noise_length;
       if (distortion_lag < 40) {
         noise_length = 2 * distortion_lag + 30;
       } else {
@@ -641,7 +680,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     // even, which is suitable for the sqrt.
     unvoiced_scale += ((unvoiced_scale & 0x1) ^ 0x1);
     unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
-    int32_t unvoiced_gain = WebRtcSpl_SqrtFloor(unvoiced_energy);
+    int16_t unvoiced_gain =
+        static_cast<int16_t>(WebRtcSpl_SqrtFloor(unvoiced_energy));
     parameters.ar_gain_scale = 13
         + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
     parameters.ar_gain = unvoiced_gain;
@@ -654,7 +694,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     //   voice_mix_factor = 0;
     if (corr_coefficient > 7875) {
       int16_t x1, x2, x3;
-      x1 = corr_coefficient;  // |corr_coefficient| is in Q14.
+      // |corr_coefficient| is in Q14.
+      x1 = static_cast<int16_t>(corr_coefficient);
       x2 = (x1 * x1) >> 14;   // Shift 14 to keep result in Q14.
       x3 = (x1 * x2) >> 14;
       static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
@@ -662,9 +703,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
       temp_sum += kCoefficients[1] * x1;
       temp_sum += kCoefficients[2] * x2;
       temp_sum += kCoefficients[3] * x3;
-      parameters.voice_mix_factor = temp_sum / 4096;
-      parameters.voice_mix_factor = std::min(parameters.voice_mix_factor,
-                                             static_cast<int16_t>(16384));
+      parameters.voice_mix_factor =
+          static_cast<int16_t>(std::min(temp_sum / 4096, 16384));
       parameters.voice_mix_factor = std::max(parameters.voice_mix_factor,
                                              static_cast<int16_t>(0));
     } else {
@@ -682,8 +722,9 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
       // the division.
       // Shift the denominator from Q13 to Q5 before the division. The result of
       // the division will then be in Q20.
-      int16_t temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
-                                               (distortion_lag * slope) >> 8);
+      int temp_ratio = WebRtcSpl_DivW32W16(
+          (slope - 8192) << 12,
+          static_cast<int16_t>((distortion_lag * slope) >> 8));
       if (slope > 14746) {
         // slope > 1.8.
         // Divide by 2, with proper rounding.
@@ -696,14 +737,13 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     } else {
       // Calculate (1 - slope) / distortion_lag.
       // Shift |slope| by 7 to Q20 before the division. The result is in Q20.
-      parameters.mute_slope = WebRtcSpl_DivW32W16((8192 - slope) << 7,
-                                                   distortion_lag);
+      parameters.mute_slope = WebRtcSpl_DivW32W16(
+          (8192 - slope) << 7, static_cast<int16_t>(distortion_lag));
       if (parameters.voice_mix_factor <= 13107) {
         // Make sure the mute factor decreases from 1.0 to 0.9 in no more than
         // 6.25 ms.
         // mute_slope >= 0.005 / fs_mult in Q20.
-        parameters.mute_slope = std::max(static_cast<int16_t>(5243 / fs_mult),
-                                         parameters.mute_slope);
+        parameters.mute_slope = std::max(5243 / fs_mult, parameters.mute_slope);
       } else if (slope > 8028) {
         parameters.mute_slope = 0;
       }
@@ -712,11 +752,25 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   }
 }
 
-int16_t Expand::Correlation(const int16_t* input, size_t input_length,
-                            int16_t* output, int16_t* output_scale) const {
+Expand::ChannelParameters::ChannelParameters()
+    : mute_factor(16384),
+      ar_gain(0),
+      ar_gain_scale(0),
+      voice_mix_factor(0),
+      current_voice_mix_factor(0),
+      onset(false),
+      mute_slope(0) {
+  memset(ar_filter, 0, sizeof(ar_filter));
+  memset(ar_filter_state, 0, sizeof(ar_filter_state));
+}
+
+void Expand::Correlation(const int16_t* input,
+                         size_t input_length,
+                         int16_t* output,
+                         int* output_scale) const {
   // Set parameters depending on sample rate.
   const int16_t* filter_coefficients;
-  int16_t num_coefficients;
+  size_t num_coefficients;
   int16_t downsampling_factor;
   if (fs_hz_ == 8000) {
     num_coefficients = 3;
@@ -738,14 +792,14 @@ int16_t Expand::Correlation(const int16_t* input, size_t input_length,
 
   // Correlate from lag 10 to lag 60 in downsampled domain.
   // (Corresponds to 20-120 for narrow-band, 40-240 for wide-band, and so on.)
-  static const int kCorrelationStartLag = 10;
-  static const int kNumCorrelationLags = 54;
-  static const int kCorrelationLength = 60;
+  static const size_t kCorrelationStartLag = 10;
+  static const size_t kNumCorrelationLags = 54;
+  static const size_t kCorrelationLength = 60;
   // Downsample to 4 kHz sample rate.
-  static const int kDownsampledLength = kCorrelationStartLag
+  static const size_t kDownsampledLength = kCorrelationStartLag
       + kNumCorrelationLags + kCorrelationLength;
   int16_t downsampled_input[kDownsampledLength];
-  static const int kFilterDelay = 0;
+  static const size_t kFilterDelay = 0;
   WebRtcSpl_DownsampleFast(
       input + input_length - kDownsampledLength * downsampling_factor,
       kDownsampledLength * downsampling_factor, downsampled_input,
@@ -771,12 +825,12 @@ int16_t Expand::Correlation(const int16_t* input, size_t input_length,
   // Normalize and move data from 32-bit to 16-bit vector.
   int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
                                                      kNumCorrelationLags);
-  int16_t norm_shift2 = std::max(18 - WebRtcSpl_NormW32(max_correlation), 0);
+  int16_t norm_shift2 = static_cast<int16_t>(
+      std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
   WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
                                    norm_shift2);
   // Total scale factor (right shifts) of correlation value.
   *output_scale = 2 * norm_shift + kCorrelationShift + norm_shift2;
-  return kNumCorrelationLags;
 }
 
 void Expand::UpdateLagIndex() {
@@ -793,22 +847,23 @@ void Expand::UpdateLagIndex() {
 Expand* ExpandFactory::Create(BackgroundNoise* background_noise,
                               SyncBuffer* sync_buffer,
                               RandomVector* random_vector,
+                              StatisticsCalculator* statistics,
                               int fs,
                               size_t num_channels) const {
-  return new Expand(background_noise, sync_buffer, random_vector, fs,
-                    num_channels);
+  return new Expand(background_noise, sync_buffer, random_vector, statistics,
+                    fs, num_channels);
 }
 
 // TODO(turajs): This can be moved to BackgroundNoise class.
 void Expand::GenerateBackgroundNoise(int16_t* random_vector,
                                      size_t channel,
-                                     int16_t mute_slope,
+                                     int mute_slope,
                                      bool too_many_expands,
                                      size_t num_noise_samples,
                                      int16_t* buffer) {
-  static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
   int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
-  assert(static_cast<size_t>(kMaxSampleRate / 8000 * 125) >= num_noise_samples);
+  assert(num_noise_samples <= (kMaxSampleRate / 8000 * 125));
   int16_t* noise_samples = &buffer[kNoiseLpcOrder];
   if (background_noise_->initialized()) {
     // Use background noise parameters.
@@ -826,12 +881,12 @@ void Expand::GenerateBackgroundNoise(int16_t* random_vector,
         scaled_random_vector, random_vector,
         background_noise_->Scale(channel), dc_offset,
         background_noise_->ScaleShift(channel),
-        static_cast<int>(num_noise_samples));
+        num_noise_samples);
 
     WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples,
                               background_noise_->Filter(channel),
                               kNoiseLpcOrder + 1,
-                              static_cast<int>(num_noise_samples));
+                              num_noise_samples);
 
     background_noise_->SetFilterState(
         channel,
@@ -845,7 +900,7 @@ void Expand::GenerateBackgroundNoise(int16_t* random_vector,
         bgn_mute_factor > 0) {
       // Fade BGN to zero.
       // Calculate muting slope, approximately -2^18 / fs_hz.
-      int16_t mute_slope;
+      int mute_slope;
       if (fs_hz_ == 8000) {
         mute_slope = -32;
       } else if (fs_hz_ == 16000) {
@@ -878,7 +933,7 @@ void Expand::GenerateBackgroundNoise(int16_t* random_vector,
         // kBgnFade has reached 0.
         WebRtcSpl_AffineTransformVector(noise_samples, noise_samples,
                                         bgn_mute_factor, 8192, 14,
-                                        static_cast<int>(num_noise_samples));
+                                        num_noise_samples);
       }
     }
     // Update mute_factor in BackgroundNoise class.
@@ -889,7 +944,7 @@ void Expand::GenerateBackgroundNoise(int16_t* random_vector,
   }
 }
 
-void Expand::GenerateRandomVector(int seed_increment,
+void Expand::GenerateRandomVector(int16_t seed_increment,
                                   size_t length,
                                   int16_t* random_vector) {
   // TODO(turajs): According to hlundin The loop should not be needed. Should be
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.h
index 7b41114917..25c8c21bdb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand.h
@@ -23,6 +23,7 @@ namespace webrtc {
 // Forward declarations.
 class BackgroundNoise;
 class RandomVector;
+class StatisticsCalculator;
 class SyncBuffer;
 
 // This class handles extrapolation of audio data from the sync_buffer to
@@ -34,28 +35,11 @@ class Expand {
   Expand(BackgroundNoise* background_noise,
          SyncBuffer* sync_buffer,
          RandomVector* random_vector,
+         StatisticsCalculator* statistics,
          int fs,
-         size_t num_channels)
-      : random_vector_(random_vector),
-        sync_buffer_(sync_buffer),
-        first_expand_(true),
-        fs_hz_(fs),
-        num_channels_(num_channels),
-        consecutive_expands_(0),
-        background_noise_(background_noise),
-        overlap_length_(5 * fs / 8000),
-        lag_index_direction_(0),
-        current_lag_index_(0),
-        stop_muting_(false),
-        channel_parameters_(new ChannelParameters[num_channels_]) {
-    assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
-    assert(fs <= kMaxSampleRate);  // Should not be possible.
-    assert(num_channels_ > 0);
-    memset(expand_lags_, 0, sizeof(expand_lags_));
-    Reset();
-  }
+         size_t num_channels);
 
-  virtual ~Expand() {}
+  virtual ~Expand();
 
   // Resets the object.
   virtual void Reset();
@@ -72,12 +56,6 @@ class Expand {
   // a period of expands.
   virtual void SetParametersForMergeAfterExpand();
 
-  // Sets the mute factor for |channel| to |value|.
-  void SetMuteFactor(int16_t value, size_t channel) {
-    assert(channel < num_channels_);
-    channel_parameters_[channel].mute_factor = value;
-  }
-
   // Returns the mute factor for |channel|.
   int16_t MuteFactor(size_t channel) {
     assert(channel < num_channels_);
@@ -85,18 +63,18 @@ class Expand {
   }
 
   // Accessors and mutators.
-  virtual size_t overlap_length() const { return overlap_length_; }
-  int16_t max_lag() const { return max_lag_; }
+  virtual size_t overlap_length() const;
+  size_t max_lag() const { return max_lag_; }
 
  protected:
   static const int kMaxConsecutiveExpands = 200;
-  void GenerateRandomVector(int seed_increment,
+  void GenerateRandomVector(int16_t seed_increment,
                             size_t length,
                             int16_t* random_vector);
 
   void GenerateBackgroundNoise(int16_t* random_vector,
                                size_t channel,
-                               int16_t mute_slope,
+                               int mute_slope,
                                bool too_many_expands,
                                size_t num_noise_samples,
                                int16_t* buffer);
@@ -110,34 +88,23 @@ class Expand {
   // necessary to produce concealment data.
   void AnalyzeSignal(int16_t* random_vector);
 
-  RandomVector* random_vector_;
-  SyncBuffer* sync_buffer_;
+  RandomVector* const random_vector_;
+  SyncBuffer* const sync_buffer_;
   bool first_expand_;
   const int fs_hz_;
   const size_t num_channels_;
   int consecutive_expands_;
 
  private:
-  static const int kUnvoicedLpcOrder = 6;
-  static const int kNumCorrelationCandidates = 3;
-  static const int kDistortionLength = 20;
-  static const int kLpcAnalysisLength = 160;
-  static const int kMaxSampleRate = 48000;
+  static const size_t kUnvoicedLpcOrder = 6;
+  static const size_t kNumCorrelationCandidates = 3;
+  static const size_t kDistortionLength = 20;
+  static const size_t kLpcAnalysisLength = 160;
+  static const size_t kMaxSampleRate = 48000;
   static const int kNumLags = 3;
 
   struct ChannelParameters {
-    // Constructor.
-    ChannelParameters()
-        : mute_factor(16384),
-          ar_gain(0),
-          ar_gain_scale(0),
-          voice_mix_factor(0),
-          current_voice_mix_factor(0),
-          onset(false),
-          mute_slope(0) {
-      memset(ar_filter, 0, sizeof(ar_filter));
-      memset(ar_filter_state, 0, sizeof(ar_filter_state));
-    }
+    ChannelParameters();
     int16_t mute_factor;
     int16_t ar_filter[kUnvoicedLpcOrder + 1];
     int16_t ar_filter_state[kUnvoicedLpcOrder];
@@ -148,28 +115,32 @@ class Expand {
     AudioVector expand_vector0;
     AudioVector expand_vector1;
     bool onset;
-    int16_t mute_slope; /* Q20 */
+    int mute_slope; /* Q20 */
   };
 
   // Calculate the auto-correlation of |input|, with length |input_length|
   // samples. The correlation is calculated from a downsampled version of
   // |input|, and is written to |output|. The scale factor is written to
-  // |output_scale|. Returns the length of the correlation vector.
-  int16_t Correlation(const int16_t* input, size_t input_length,
-                      int16_t* output, int16_t* output_scale) const;
+  // |output_scale|.
+  void Correlation(const int16_t* input,
+                   size_t input_length,
+                   int16_t* output,
+                   int* output_scale) const;
 
   void UpdateLagIndex();
 
-  BackgroundNoise* background_noise_;
+  BackgroundNoise* const background_noise_;
+  StatisticsCalculator* const statistics_;
   const size_t overlap_length_;
-  int16_t max_lag_;
+  size_t max_lag_;
   size_t expand_lags_[kNumLags];
   int lag_index_direction_;
   int current_lag_index_;
   bool stop_muting_;
+  size_t expand_duration_samples_;
   rtc::scoped_ptr<ChannelParameters[]> channel_parameters_;
 
-  DISALLOW_COPY_AND_ASSIGN(Expand);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Expand);
 };
 
 struct ExpandFactory {
@@ -179,6 +150,7 @@ struct ExpandFactory {
   virtual Expand* Create(BackgroundNoise* background_noise,
                          SyncBuffer* sync_buffer,
                          RandomVector* random_vector,
+                         StatisticsCalculator* statistics,
                          int fs,
                          size_t num_channels) const;
 };
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand_unittest.cc
index 68b4f60f15..1441704102 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/expand_unittest.cc
@@ -13,9 +13,14 @@
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
@@ -25,7 +30,8 @@ TEST(Expand, CreateAndDestroy) {
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(1, 1000);
   RandomVector random_vector;
-  Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
 }
 
 TEST(Expand, CreateUsingFactory) {
@@ -34,13 +40,135 @@ TEST(Expand, CreateUsingFactory) {
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(1, 1000);
   RandomVector random_vector;
+  StatisticsCalculator statistics;
   ExpandFactory expand_factory;
-  Expand* expand =
-      expand_factory.Create(&bgn, &sync_buffer, &random_vector, fs, channels);
+  Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector,
+                                         &statistics, fs, channels);
   EXPECT_TRUE(expand != NULL);
   delete expand;
 }
 
+namespace {
+class FakeStatisticsCalculator : public StatisticsCalculator {
+ public:
+  void LogDelayedPacketOutageEvent(int outage_duration_ms) override {
+    last_outage_duration_ms_ = outage_duration_ms;
+  }
+
+  int last_outage_duration_ms() const { return last_outage_duration_ms_; }
+
+ private:
+  int last_outage_duration_ms_ = 0;
+};
+
+// This is the same size that is given to the SyncBuffer object in NetEq.
+const size_t kNetEqSyncBufferLengthMs = 720;
+}  // namespace
+
+class ExpandTest : public ::testing::Test {
+ protected:
+  ExpandTest()
+      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+                    32000),
+        test_sample_rate_hz_(32000),
+        num_channels_(1),
+        background_noise_(num_channels_),
+        sync_buffer_(num_channels_,
+                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
+        expand_(&background_noise_,
+                &sync_buffer_,
+                &random_vector_,
+                &statistics_,
+                test_sample_rate_hz_,
+                num_channels_) {
+    WebRtcSpl_Init();
+    input_file_.set_output_rate_hz(test_sample_rate_hz_);
+  }
+
+  void SetUp() override {
+    // Fast-forward the input file until there is speech (about 1.1 second into
+    // the file).
+    const size_t speech_start_samples =
+        static_cast<size_t>(test_sample_rate_hz_ * 1.1f);
+    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
+
+    // Pre-load the sync buffer with speech data.
+    ASSERT_TRUE(
+        input_file_.Read(sync_buffer_.Size(), &sync_buffer_.Channel(0)[0]));
+    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
+  }
+
+  test::ResampleInputAudioFile input_file_;
+  int test_sample_rate_hz_;
+  size_t num_channels_;
+  BackgroundNoise background_noise_;
+  SyncBuffer sync_buffer_;
+  RandomVector random_vector_;
+  FakeStatisticsCalculator statistics_;
+  Expand expand_;
+};
+
+// This test calls the expand object to produce concealment data a few times,
+// and then ends by calling SetParametersForNormalAfterExpand. This simulates
+// the situation where the packet next up for decoding was just delayed, not
+// lost.
+TEST_F(ExpandTest, DelayedPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert |sum_output_len_samples| to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
+                                   (test_sample_rate_hz_ / 1000)),
+            statistics_.last_outage_duration_ms());
+}
+
+// This test is similar to DelayedPacketOutage, but ends by calling
+// SetParametersForMergeAfterExpand. This simulates the situation where the
+// packet next up for decoding was actually lost (or at least a later packet
+// arrived before it).
+TEST_F(ExpandTest, LostPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+  }
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+}
+
+// This test is similar to the DelayedPacketOutage test above, but with the
+// difference that Expand::Reset() is called after 5 calls to Expand::Process().
+// This should reset the statistics, and will in the end lead to an outage of
+// 5 periods instead of 10.
+TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    if (i == 5) {
+      expand_.Reset();
+      sum_output_len_samples = 0;
+    }
+    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert |sum_output_len_samples| to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
+                                   (test_sample_rate_hz_ / 1000)),
+            statistics_.last_outage_duration_ms());
+}
+
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/include/neteq.h
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/neteq/include/neteq.h
index 5d1bbfce9d..1322223970 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
 
 #include <string.h>  // Provide access to size_t.
 
-#include <vector>
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/common_types.h"
@@ -44,7 +44,13 @@ struct NetEqNetworkStatistics {
                                     // decoding (in Q14).
   int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
                            // (positive or negative).
-  int added_zero_samples;  // Number of zero samples added in "off" mode.
+  size_t added_zero_samples;  // Number of zero samples added in "off" mode.
+  // Statistics for packet waiting times, i.e., the time between a packet
+  // arrives until it is decoded.
+  int mean_waiting_time_ms;
+  int median_waiting_time_ms;
+  int min_waiting_time_ms;
+  int max_waiting_time_ms;
 };
 
 enum NetEqOutputType {
@@ -75,18 +81,24 @@ class NetEq {
     Config()
         : sample_rate_hz(16000),
           enable_audio_classifier(false),
+          enable_post_decode_vad(false),
           max_packets_in_buffer(50),
           // |max_delay_ms| has the same effect as calling SetMaximumDelay().
           max_delay_ms(2000),
           background_noise_mode(kBgnOff),
-          playout_mode(kPlayoutOn) {}
+          playout_mode(kPlayoutOn),
+          enable_fast_accelerate(false) {}
 
-    int sample_rate_hz;  // Initial vale. Will change with input data.
+    std::string ToString() const;
+
+    int sample_rate_hz;  // Initial value. Will change with input data.
     bool enable_audio_classifier;
-    int max_packets_in_buffer;
+    bool enable_post_decode_vad;
+    size_t max_packets_in_buffer;
     int max_delay_ms;
     BackgroundNoiseMode background_noise_mode;
     NetEqPlayoutMode playout_mode;
+    bool enable_fast_accelerate;
   };
 
   enum ReturnCodes {
@@ -135,8 +147,7 @@ class NetEq {
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                           const uint8_t* payload,
-                           size_t length_bytes,
+                           rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp) = 0;
 
   // Inserts a sync-packet into packet queue. Sync-packets are decoded to
@@ -160,21 +171,29 @@ class NetEq {
   // The speech type is written to |type|, if |type| is not NULL.
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(size_t max_length, int16_t* output_audio,
-                       int* samples_per_channel, int* num_channels,
+                       size_t* samples_per_channel, size_t* num_channels,
                        NetEqOutputType* type) = 0;
 
-  // Associates |rtp_payload_type| with |codec| and stores the information in
-  // the codec database. Returns 0 on success, -1 on failure.
-  virtual int RegisterPayloadType(enum NetEqDecoder codec,
+  // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
+  // information in the codec database. Returns 0 on success, -1 on failure.
+  // The name is only used to provide information back to the caller about the
+  // decoders. Hence, the name is arbitrary, and may be empty.
+  virtual int RegisterPayloadType(NetEqDecoder codec,
+                                  const std::string& codec_name,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
   // decoder database. The decoder implements a decoder of type |codec| and
-  // associates it with |rtp_payload_type|. Returns kOK on success,
-  // kFail on failure.
+  // associates it with |rtp_payload_type| and |codec_name|. The decoder will
+  // produce samples at the rate |sample_rate_hz|. Returns kOK on success, kFail
+  // on failure.
+  // The name is only used to provide information back to the caller about the
+  // decoders. Hence, the name is arbitrary, and may be empty.
   virtual int RegisterExternalDecoder(AudioDecoder* decoder,
-                                      enum NetEqDecoder codec,
-                                      uint8_t rtp_payload_type) = 0;
+                                      NetEqDecoder codec,
+                                      const std::string& codec_name,
+                                      uint8_t rtp_payload_type,
+                                      int sample_rate_hz) = 0;
 
   // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
   // -1 on failure.
@@ -204,8 +223,8 @@ class NetEq {
   // Not implemented.
   virtual int TargetDelay() = 0;
 
-  // Not implemented.
-  virtual int CurrentDelay() = 0;
+  // Returns the current total delay (packet buffer and sync buffer) in ms.
+  virtual int CurrentDelayMs() const = 0;
 
   // Sets the playout mode to |mode|.
   // Deprecated. Set the mode in the Config struct passed to the constructor.
@@ -221,11 +240,6 @@ class NetEq {
   // after the call.
   virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
 
-  // Writes the last packet waiting times (in ms) to |waiting_times|. The number
-  // of values written is no more than 100, but may be smaller if the interface
-  // is polled again before 100 packets has arrived.
-  virtual void WaitingTimes(std::vector<int>* waiting_times) = 0;
-
   // Writes the current RTCP statistics to |stats|. The statistics are reset
   // and a new report period is started with the call.
   virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
@@ -244,6 +258,11 @@ class NetEq {
   // Returns true if the RTP timestamp is valid, otherwise false.
   virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
 
+  // Returns the sample rate in Hz of the audio produced in the last GetAudio
+  // call. If GetAudio has not been called yet, the configured sample rate
+  // (Config::sample_rate_hz) is returned.
+  virtual int last_output_sample_rate_hz() const = 0;
+
   // Not implemented.
   virtual int SetTargetNumberOfChannels() = 0;
 
@@ -266,17 +285,24 @@ class NetEq {
   virtual void PacketBufferStatistics(int* current_num_packets,
                                       int* max_num_packets) const = 0;
 
-  // Get sequence number and timestamp of the latest RTP.
-  // This method is to facilitate NACK.
-  virtual int DecodedRtpInfo(int* sequence_number,
-                             uint32_t* timestamp) const = 0;
+  // Enables NACK and sets the maximum size of the NACK list, which should be
+  // positive and no larger than Nack::kNackListSizeLimit. If NACK is already
+  // enabled then the maximum NACK list size is modified accordingly.
+  virtual void EnableNack(size_t max_nack_list_size) = 0;
+
+  virtual void DisableNack() = 0;
+
+  // Returns a list of RTP sequence numbers corresponding to packets to be
+  // retransmitted, given an estimate of the round-trip time in milliseconds.
+  virtual std::vector<uint16_t> GetNackList(
+      int64_t round_trip_time_ms) const = 0;
 
  protected:
   NetEq() {}
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(NetEq);
+  RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.cc
index bc22000ed3..b6fb2d8a26 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.cc
@@ -24,18 +24,32 @@
 
 namespace webrtc {
 
-int Merge::Process(int16_t* input, size_t input_length,
-                   int16_t* external_mute_factor_array,
-                   AudioMultiVector* output) {
+Merge::Merge(int fs_hz,
+             size_t num_channels,
+             Expand* expand,
+             SyncBuffer* sync_buffer)
+    : fs_hz_(fs_hz),
+      num_channels_(num_channels),
+      fs_mult_(fs_hz_ / 8000),
+      timestamps_per_call_(static_cast<size_t>(fs_hz_ / 100)),
+      expand_(expand),
+      sync_buffer_(sync_buffer),
+      expanded_(num_channels_) {
+  assert(num_channels_ > 0);
+}
+
+size_t Merge::Process(int16_t* input, size_t input_length,
+                      int16_t* external_mute_factor_array,
+                      AudioMultiVector* output) {
   // TODO(hlundin): Change to an enumerator and skip assert.
   assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ ==  32000 ||
          fs_hz_ == 48000);
   assert(fs_hz_ <= kMaxSampleRate);  // Should not be possible.
 
-  int old_length;
-  int expand_period;
+  size_t old_length;
+  size_t expand_period;
   // Get expansion data to overlap and mix with.
-  int expanded_length = GetExpandedSignal(&old_length, &expand_period);
+  size_t expanded_length = GetExpandedSignal(&old_length, &expand_period);
 
   // Transfer input signal to an AudioMultiVector.
   AudioMultiVector input_vector(num_channels_);
@@ -43,7 +57,7 @@ int Merge::Process(int16_t* input, size_t input_length,
   size_t input_length_per_channel = input_vector.Size();
   assert(input_length_per_channel == input_length / num_channels_);
 
-  int16_t best_correlation_index = 0;
+  size_t best_correlation_index = 0;
   size_t output_length = 0;
 
   for (size_t channel = 0; channel < num_channels_; ++channel) {
@@ -51,8 +65,8 @@ int Merge::Process(int16_t* input, size_t input_length,
     int16_t* expanded_channel = &expanded_[channel][0];
     int16_t expanded_max, input_max;
     int16_t new_mute_factor = SignalScaling(
-        input_channel, static_cast<int>(input_length_per_channel),
-        expanded_channel, &expanded_max, &input_max);
+        input_channel, input_length_per_channel, expanded_channel,
+        &expanded_max, &input_max);
 
     // Adjust muting factor (product of "main" muting factor and expand muting
     // factor).
@@ -70,13 +84,13 @@ int Merge::Process(int16_t* input, size_t input_length,
       // Downsample, correlate, and find strongest correlation period for the
       // master (i.e., first) channel only.
       // Downsample to 4kHz sample rate.
-      Downsample(input_channel, static_cast<int>(input_length_per_channel),
-                 expanded_channel, expanded_length);
+      Downsample(input_channel, input_length_per_channel, expanded_channel,
+                 expanded_length);
 
       // Calculate the lag of the strongest correlation period.
       best_correlation_index = CorrelateAndPeakSearch(
           expanded_max, input_max, old_length,
-          static_cast<int>(input_length_per_channel), expand_period);
+          input_length_per_channel, expand_period);
     }
 
     static const int kTempDataSize = 3600;
@@ -85,19 +99,20 @@ int Merge::Process(int16_t* input, size_t input_length,
 
     // Mute the new decoded data if needed (and unmute it linearly).
     // This is the overlapping part of expanded_signal.
-    int interpolation_length = std::min(
+    size_t interpolation_length = std::min(
         kMaxCorrelationLength * fs_mult_,
         expanded_length - best_correlation_index);
     interpolation_length = std::min(interpolation_length,
-                                    static_cast<int>(input_length_per_channel));
+                                    input_length_per_channel);
     if (*external_mute_factor < 16384) {
       // Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
       // and so on.
       int increment = 4194 / fs_mult_;
-      *external_mute_factor = DspHelper::RampSignal(input_channel,
-                                                    interpolation_length,
-                                                    *external_mute_factor,
-                                                    increment);
+      *external_mute_factor =
+          static_cast<int16_t>(DspHelper::RampSignal(input_channel,
+                                                     interpolation_length,
+                                                     *external_mute_factor,
+                                                     increment));
       DspHelper::UnmuteSignal(&input_channel[interpolation_length],
                               input_length_per_channel - interpolation_length,
                               external_mute_factor, increment,
@@ -111,7 +126,8 @@ int Merge::Process(int16_t* input, size_t input_length,
     }
 
     // Do overlap and mix linearly.
-    int increment = 16384 / (interpolation_length + 1);  // In Q14.
+    int16_t increment =
+        static_cast<int16_t>(16384 / (interpolation_length + 1));  // In Q14.
     int16_t mute_factor = 16384 - increment;
     memmove(temp_data, expanded_channel,
             sizeof(int16_t) * best_correlation_index);
@@ -137,14 +153,14 @@ int Merge::Process(int16_t* input, size_t input_length,
 
   // Return new added length. |old_length| samples were borrowed from
   // |sync_buffer_|.
-  return static_cast<int>(output_length) - old_length;
+  return output_length - old_length;
 }
 
-int Merge::GetExpandedSignal(int* old_length, int* expand_period) {
+size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
   // Check how much data that is left since earlier.
-  *old_length = static_cast<int>(sync_buffer_->FutureLength());
+  *old_length = sync_buffer_->FutureLength();
   // Should never be less than overlap_length.
-  assert(*old_length >= static_cast<int>(expand_->overlap_length()));
+  assert(*old_length >= expand_->overlap_length());
   // Generate data to merge the overlap with using expand.
   expand_->SetParametersForMergeAfterExpand();
 
@@ -155,44 +171,44 @@ int Merge::GetExpandedSignal(int* old_length, int* expand_period) {
     // but shift them towards the end of the buffer. This is ok, since all of
     // the buffer will be expand data anyway, so as long as the beginning is
     // left untouched, we're fine.
-    int16_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
+    size_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
     sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index());
     *old_length = 210 * kMaxSampleRate / 8000;
     // This is the truncated length.
   }
   // This assert should always be true thanks to the if statement above.
-  assert(210 * kMaxSampleRate / 8000 - *old_length >= 0);
+  assert(210 * kMaxSampleRate / 8000 >= *old_length);
 
   AudioMultiVector expanded_temp(num_channels_);
   expand_->Process(&expanded_temp);
-  *expand_period = static_cast<int>(expanded_temp.Size());  // Samples per
-                                                            // channel.
+  *expand_period = expanded_temp.Size();  // Samples per channel.
 
   expanded_.Clear();
   // Copy what is left since earlier into the expanded vector.
   expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index());
-  assert(expanded_.Size() == static_cast<size_t>(*old_length));
+  assert(expanded_.Size() == *old_length);
   assert(expanded_temp.Size() > 0);
   // Do "ugly" copy and paste from the expanded in order to generate more data
   // to correlate (but not interpolate) with.
-  const int required_length = (120 + 80 + 2) * fs_mult_;
-  if (expanded_.Size() < static_cast<size_t>(required_length)) {
-    while (expanded_.Size() < static_cast<size_t>(required_length)) {
+  const size_t required_length = static_cast<size_t>((120 + 80 + 2) * fs_mult_);
+  if (expanded_.Size() < required_length) {
+    while (expanded_.Size() < required_length) {
       // Append one more pitch period each time.
       expanded_.PushBack(expanded_temp);
     }
     // Trim the length to exactly |required_length|.
     expanded_.PopBack(expanded_.Size() - required_length);
   }
-  assert(expanded_.Size() >= static_cast<size_t>(required_length));
+  assert(expanded_.Size() >= required_length);
   return required_length;
 }
 
-int16_t Merge::SignalScaling(const int16_t* input, int input_length,
+int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
                              const int16_t* expanded_signal,
                              int16_t* expanded_max, int16_t* input_max) const {
   // Adjust muting factor if new vector is more or less of the BGN energy.
-  const int mod_input_length = std::min(64 * fs_mult_, input_length);
+  const size_t mod_input_length =
+      std::min(static_cast<size_t>(64 * fs_mult_), input_length);
   *expanded_max = WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
   *input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
 
@@ -232,7 +248,8 @@ int16_t Merge::SignalScaling(const int16_t* input, int input_length,
     // energy_expanded / energy_input is in Q14.
     energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
     // Calculate sqrt(energy_expanded / energy_input) in Q14.
-    mute_factor = WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14);
+    mute_factor = static_cast<int16_t>(
+        WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
   } else {
     // Set to 1 (in Q14) when |expanded| has higher energy than |input|.
     mute_factor = 16384;
@@ -243,13 +260,13 @@ int16_t Merge::SignalScaling(const int16_t* input, int input_length,
 
 // TODO(hlundin): There are some parameter values in this method that seem
 // strange. Compare with Expand::Correlation.
-void Merge::Downsample(const int16_t* input, int input_length,
-                       const int16_t* expanded_signal, int expanded_length) {
+void Merge::Downsample(const int16_t* input, size_t input_length,
+                       const int16_t* expanded_signal, size_t expanded_length) {
   const int16_t* filter_coefficients;
-  int num_coefficients;
+  size_t num_coefficients;
   int decimation_factor = fs_hz_ / 4000;
-  static const int kCompensateDelay = 0;
-  int length_limit = fs_hz_ / 100;  // 10 ms in samples.
+  static const size_t kCompensateDelay = 0;
+  size_t length_limit = static_cast<size_t>(fs_hz_ / 100);  // 10 ms in samples.
   if (fs_hz_ == 8000) {
     filter_coefficients = DspHelper::kDownsample8kHzTbl;
     num_coefficients = 3;
@@ -263,7 +280,7 @@ void Merge::Downsample(const int16_t* input, int input_length,
     filter_coefficients = DspHelper::kDownsample48kHzTbl;
     num_coefficients = 7;
   }
-  int signal_offset = num_coefficients - 1;
+  size_t signal_offset = num_coefficients - 1;
   WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset],
                            expanded_length - signal_offset,
                            expanded_downsampled_, kExpandDownsampLength,
@@ -271,10 +288,10 @@ void Merge::Downsample(const int16_t* input, int input_length,
                            decimation_factor, kCompensateDelay);
   if (input_length <= length_limit) {
     // Not quite long enough, so we have to cheat a bit.
-    int16_t temp_len = input_length - signal_offset;
+    size_t temp_len = input_length - signal_offset;
     // TODO(hlundin): Should |downsamp_temp_len| be corrected for round-off
     // errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
-    int16_t downsamp_temp_len = temp_len / decimation_factor;
+    size_t downsamp_temp_len = temp_len / decimation_factor;
     WebRtcSpl_DownsampleFast(&input[signal_offset], temp_len,
                              input_downsampled_, downsamp_temp_len,
                              filter_coefficients, num_coefficients,
@@ -290,14 +307,14 @@ void Merge::Downsample(const int16_t* input, int input_length,
   }
 }
 
-int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
-                                      int start_position, int input_length,
-                                      int expand_period) const {
+size_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
+                                     size_t start_position, size_t input_length,
+                                     size_t expand_period) const {
   // Calculate correlation without any normalization.
-  const int max_corr_length = kMaxCorrelationLength;
-  int stop_position_downsamp = std::min(
-      max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
-  int16_t correlation_shift = 0;
+  const size_t max_corr_length = kMaxCorrelationLength;
+  size_t stop_position_downsamp =
+      std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
+  int correlation_shift = 0;
   if (expanded_max * input_max > 26843546) {
     correlation_shift = 3;
   }
@@ -308,15 +325,15 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
                              stop_position_downsamp, correlation_shift, 1);
 
   // Normalize correlation to 14 bits and copy to a 16-bit array.
-  const int pad_length = static_cast<int>(expand_->overlap_length() - 1);
-  const int correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
+  const size_t pad_length = expand_->overlap_length() - 1;
+  const size_t correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
   rtc::scoped_ptr<int16_t[]> correlation16(
       new int16_t[correlation_buffer_size]);
   memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
   int16_t* correlation_ptr = &correlation16[pad_length];
   int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
                                                      stop_position_downsamp);
-  int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
+  int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
   WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
                                    correlation, norm_shift);
 
@@ -325,21 +342,20 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   // (1) w16_bestIndex + input_length <
   //     timestamps_per_call_ + expand_->overlap_length();
   // (2) w16_bestIndex + input_length < start_position.
-  int start_index = timestamps_per_call_ +
-      static_cast<int>(expand_->overlap_length());
+  size_t start_index = timestamps_per_call_ + expand_->overlap_length();
   start_index = std::max(start_position, start_index);
-  start_index = std::max(start_index - input_length, 0);
+  start_index = (input_length > start_index) ? 0 : (start_index - input_length);
   // Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
-  int start_index_downsamp = start_index / (fs_mult_ * 2);
+  size_t start_index_downsamp = start_index / (fs_mult_ * 2);
 
   // Calculate a modified |stop_position_downsamp| to account for the increased
   // start index |start_index_downsamp| and the effective array length.
-  int modified_stop_pos =
+  size_t modified_stop_pos =
       std::min(stop_position_downsamp,
                kMaxCorrelationLength + pad_length - start_index_downsamp);
-  int best_correlation_index;
+  size_t best_correlation_index;
   int16_t best_correlation;
-  static const int kNumCorrelationCandidates = 1;
+  static const size_t kNumCorrelationCandidates = 1;
   DspHelper::PeakDetection(&correlation_ptr[start_index_downsamp],
                            modified_stop_pos, kNumCorrelationCandidates,
                            fs_mult_, &best_correlation_index,
@@ -350,17 +366,17 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   // Ensure that underrun does not occur for 10ms case => we have to get at
   // least 10ms + overlap . (This should never happen thanks to the above
   // modification of peak-finding starting point.)
-  while ((best_correlation_index + input_length) <
-      static_cast<int>(timestamps_per_call_ + expand_->overlap_length()) ||
-      best_correlation_index + input_length < start_position) {
+  while (((best_correlation_index + input_length) <
+          (timestamps_per_call_ + expand_->overlap_length())) ||
+         ((best_correlation_index + input_length) < start_position)) {
     assert(false);  // Should never happen.
     best_correlation_index += expand_period;  // Jump one lag ahead.
   }
   return best_correlation_index;
 }
 
-int Merge::RequiredFutureSamples() {
-  return static_cast<int>(fs_hz_ / 100 * num_channels_);  // 10 ms.
+size_t Merge::RequiredFutureSamples() {
+  return fs_hz_ / 100 * num_channels_;  // 10 ms.
 }
 
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.h
index 1bf0483dfe..a168502c27 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge.h
@@ -33,17 +33,10 @@ class SyncBuffer;
 // what the Merge class does.
 class Merge {
  public:
-  Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
-      : fs_hz_(fs_hz),
-        num_channels_(num_channels),
-        fs_mult_(fs_hz_ / 8000),
-        timestamps_per_call_(fs_hz_ / 100),
-        expand_(expand),
-        sync_buffer_(sync_buffer),
-        expanded_(num_channels_) {
-    assert(num_channels_ > 0);
-  }
-
+  Merge(int fs_hz,
+        size_t num_channels,
+        Expand* expand,
+        SyncBuffer* sync_buffer);
   virtual ~Merge() {}
 
   // The main method to produce the audio data. The decoded data is supplied in
@@ -53,11 +46,11 @@ class Merge {
   // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
   // will be used to scale the audio, and is updated in the process. The array
   // must have |num_channels_| elements.
-  virtual int Process(int16_t* input, size_t input_length,
-                      int16_t* external_mute_factor_array,
-                      AudioMultiVector* output);
+  virtual size_t Process(int16_t* input, size_t input_length,
+                         int16_t* external_mute_factor_array,
+                         AudioMultiVector* output);
 
-  virtual int RequiredFutureSamples();
+  virtual size_t RequiredFutureSamples();
 
  protected:
   const int fs_hz_;
@@ -65,45 +58,45 @@ class Merge {
 
  private:
   static const int kMaxSampleRate = 48000;
-  static const int kExpandDownsampLength = 100;
-  static const int kInputDownsampLength = 40;
-  static const int kMaxCorrelationLength = 60;
+  static const size_t kExpandDownsampLength = 100;
+  static const size_t kInputDownsampLength = 40;
+  static const size_t kMaxCorrelationLength = 60;
 
   // Calls |expand_| to get more expansion data to merge with. The data is
   // written to |expanded_signal_|. Returns the length of the expanded data,
   // while |expand_period| will be the number of samples in one expansion period
   // (typically one pitch period). The value of |old_length| will be the number
   // of samples that were taken from the |sync_buffer_|.
-  int GetExpandedSignal(int* old_length, int* expand_period);
+  size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
 
   // Analyzes |input| and |expanded_signal| to find maximum values. Returns
   // a muting factor (Q14) to be used on the new data.
-  int16_t SignalScaling(const int16_t* input, int input_length,
+  int16_t SignalScaling(const int16_t* input, size_t input_length,
                         const int16_t* expanded_signal,
                         int16_t* expanded_max, int16_t* input_max) const;
 
   // Downsamples |input| (|input_length| samples) and |expanded_signal| to
   // 4 kHz sample rate. The downsampled signals are written to
   // |input_downsampled_| and |expanded_downsampled_|, respectively.
-  void Downsample(const int16_t* input, int input_length,
-                  const int16_t* expanded_signal, int expanded_length);
+  void Downsample(const int16_t* input, size_t input_length,
+                  const int16_t* expanded_signal, size_t expanded_length);
 
   // Calculates cross-correlation between |input_downsampled_| and
   // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
   // lag is returned.
-  int16_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
-                                 int start_position, int input_length,
-                                 int expand_period) const;
+  size_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
+                                size_t start_position, size_t input_length,
+                                size_t expand_period) const;
 
   const int fs_mult_;  // fs_hz_ / 8000.
-  const int timestamps_per_call_;
+  const size_t timestamps_per_call_;
   Expand* expand_;
   SyncBuffer* sync_buffer_;
   int16_t expanded_downsampled_[kExpandDownsampLength];
   int16_t input_downsampled_[kInputDownsampLength];
   AudioMultiVector expanded_;
 
-  DISALLOW_COPY_AND_ASSIGN(Merge);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge_unittest.cc
index bdcbbb8a9b..ddb0e16ddf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/merge_unittest.cc
@@ -18,6 +18,7 @@
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 namespace webrtc {
@@ -28,7 +29,8 @@ TEST(Merge, CreateAndDestroy) {
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(1, 1000);
   RandomVector random_vector;
-  Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
   Merge merge(fs, channels, &expand, &sync_buffer);
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
index 93261ab607..c1cc09cb5e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -22,15 +22,15 @@ class MockAudioDecoder : public AudioDecoder {
   MockAudioDecoder() {}
   virtual ~MockAudioDecoder() { Die(); }
   MOCK_METHOD0(Die, void());
-  MOCK_METHOD6(
-      Decode,
-      int(const uint8_t*, size_t, int, size_t, int16_t*, SpeechType*));
+  MOCK_METHOD5(DecodeInternal,
+               int(const uint8_t*, size_t, int, int16_t*, SpeechType*));
   MOCK_CONST_METHOD0(HasDecodePlc, bool());
-  MOCK_METHOD2(DecodePlc, int(int, int16_t*));
-  MOCK_METHOD0(Init, int());
+  MOCK_METHOD2(DecodePlc, size_t(size_t, int16_t*));
+  MOCK_METHOD0(Reset, void());
   MOCK_METHOD5(IncomingPacket, int(const uint8_t*, size_t, uint16_t, uint32_t,
                                    uint32_t));
   MOCK_METHOD0(ErrorCode, int());
+  MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
   MOCK_CONST_METHOD0(Channels, size_t());
   MOCK_CONST_METHOD0(codec_type, NetEqDecoder());
   MOCK_METHOD1(CodecSupported, bool(NetEqDecoder));
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
index ebc6acda99..82dee2a345 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -25,8 +25,8 @@ class MockBufferLevelFilter : public BufferLevelFilter {
   MOCK_METHOD0(Reset,
       void());
   MOCK_METHOD3(Update,
-      void(int buffer_size_packets, int time_stretched_samples,
-           int packet_len_samples));
+      void(size_t buffer_size_packets, int time_stretched_samples,
+           size_t packet_len_samples));
   MOCK_METHOD1(SetTargetBufferLevel,
       void(int target_buffer_level));
   MOCK_CONST_METHOD0(filtered_current_level,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
index d127c5d810..1b4a3c9da5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
 
+#include <string>
+
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
@@ -27,10 +29,12 @@ class MockDecoderDatabase : public DecoderDatabase {
       int());
   MOCK_METHOD0(Reset,
       void());
-  MOCK_METHOD2(RegisterPayload,
-      int(uint8_t rtp_payload_type, NetEqDecoder codec_type));
-  MOCK_METHOD4(InsertExternal,
-      int(uint8_t rtp_payload_type, NetEqDecoder codec_type, int fs_hz,
+  MOCK_METHOD3(RegisterPayload,
+      int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
+          const std::string& name));
+  MOCK_METHOD5(InsertExternal,
+      int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
+          const std::string& codec_name, int fs_hz,
           AudioDecoder* decoder));
   MOCK_METHOD1(Remove,
       int(uint8_t rtp_payload_type));
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
index 1d2dc8ea3d..6fb85854d7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -19,7 +19,8 @@ namespace webrtc {
 
 class MockDelayManager : public DelayManager {
  public:
-  MockDelayManager(int max_packets_in_buffer, DelayPeakDetector* peak_detector)
+  MockDelayManager(size_t max_packets_in_buffer,
+                   DelayPeakDetector* peak_detector)
       : DelayManager(max_packets_in_buffer, peak_detector) {}
   virtual ~MockDelayManager() { Die(); }
   MOCK_METHOD0(Die, void());
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
index 881e9005bb..a1c370e180 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
@@ -26,7 +26,7 @@ class MockDtmfToneGenerator : public DtmfToneGenerator {
   MOCK_METHOD0(Reset,
       void());
   MOCK_METHOD2(Generate,
-      int(int num_samples, AudioMultiVector* output));
+      int(size_t num_samples, AudioMultiVector* output));
   MOCK_CONST_METHOD0(initialized,
       bool());
 };
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_expand.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_expand.h
index 45e3239f61..f5ca077531 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_expand.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_expand.h
@@ -22,10 +22,15 @@ class MockExpand : public Expand {
   MockExpand(BackgroundNoise* background_noise,
              SyncBuffer* sync_buffer,
              RandomVector* random_vector,
+             StatisticsCalculator* statistics,
              int fs,
              size_t num_channels)
-      : Expand(background_noise, sync_buffer, random_vector, fs, num_channels) {
-  }
+      : Expand(background_noise,
+               sync_buffer,
+               random_vector,
+               statistics,
+               fs,
+               num_channels) {}
   virtual ~MockExpand() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD0(Reset,
@@ -46,10 +51,11 @@ namespace webrtc {
 
 class MockExpandFactory : public ExpandFactory {
  public:
-  MOCK_CONST_METHOD5(Create,
+  MOCK_CONST_METHOD6(Create,
                      Expand*(BackgroundNoise* background_noise,
                              SyncBuffer* sync_buffer,
                              RandomVector* random_vector,
+                             StatisticsCalculator* statistics,
                              int fs,
                              size_t num_channels));
 };
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
index d8c88561a2..42c17ae054 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
@@ -15,7 +15,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -28,23 +28,21 @@ using ::testing::Invoke;
 class ExternalPcm16B : public AudioDecoder {
  public:
   ExternalPcm16B() {}
-  virtual int Init() { return 0; }
+  void Reset() override {}
 
- protected:
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
                      int16_t* decoded,
                      SpeechType* speech_type) override {
-    int16_t ret = WebRtcPcm16b_Decode(
-        encoded, static_cast<int16_t>(encoded_len), decoded);
+    size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
     *speech_type = ConvertSpeechType(1);
-    return ret;
+    return static_cast<int>(ret);
   }
   size_t Channels() const override { return 1; }
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(ExternalPcm16B);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ExternalPcm16B);
 };
 
 // Create a mock of ExternalPcm16B which delegates all calls to the real object.
@@ -53,14 +51,14 @@ class MockExternalPcm16B : public ExternalPcm16B {
  public:
   MockExternalPcm16B() {
     // By default, all calls are delegated to the real object.
-    ON_CALL(*this, Decode(_, _, _, _, _, _))
-        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Decode));
+    ON_CALL(*this, DecodeInternal(_, _, _, _, _))
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodeInternal));
     ON_CALL(*this, HasDecodePlc())
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::HasDecodePlc));
     ON_CALL(*this, DecodePlc(_, _))
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodePlc));
-    ON_CALL(*this, Init())
-        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Init));
+    ON_CALL(*this, Reset())
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Reset));
     ON_CALL(*this, IncomingPacket(_, _, _, _, _))
         .WillByDefault(Invoke(&real_, &ExternalPcm16B::IncomingPacket));
     ON_CALL(*this, ErrorCode())
@@ -69,19 +67,17 @@ class MockExternalPcm16B : public ExternalPcm16B {
   virtual ~MockExternalPcm16B() { Die(); }
 
   MOCK_METHOD0(Die, void());
-  MOCK_METHOD6(Decode,
+  MOCK_METHOD5(DecodeInternal,
                int(const uint8_t* encoded,
                    size_t encoded_len,
                    int sample_rate_hz,
-                   size_t max_decoded_bytes,
                    int16_t* decoded,
                    SpeechType* speech_type));
   MOCK_CONST_METHOD0(HasDecodePlc,
       bool());
   MOCK_METHOD2(DecodePlc,
-      int(int num_frames, int16_t* decoded));
-  MOCK_METHOD0(Init,
-      int());
+      size_t(size_t num_frames, int16_t* decoded));
+  MOCK_METHOD0(Reset, void());
   MOCK_METHOD5(IncomingPacket,
       int(const uint8_t* payload, size_t payload_len,
           uint16_t rtp_sequence_number, uint32_t rtp_timestamp,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index 0eb7edc9c5..97e54d83a5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -41,7 +41,7 @@ class MockPacketBuffer : public PacketBuffer {
   MOCK_CONST_METHOD0(NextRtpHeader,
       const RTPHeader*());
   MOCK_METHOD1(GetNextPacket,
-      Packet*(int* discard_count));
+      Packet*(size_t* discard_count));
   MOCK_METHOD0(DiscardNextPacket,
       int());
   MOCK_METHOD2(DiscardOldPackets,
@@ -49,7 +49,7 @@ class MockPacketBuffer : public PacketBuffer {
   MOCK_METHOD1(DiscardAllOldPackets,
       int(uint32_t timestamp_limit));
   MOCK_CONST_METHOD0(NumPacketsInBuffer,
-      int());
+      size_t());
   MOCK_METHOD1(IncrementWaitingTimes,
       void(int));
   MOCK_CONST_METHOD0(current_memory_bytes,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc
similarity index 81%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc
index 4324cd2e79..011914b3d9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc
@@ -8,19 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/nack.h"
+#include "webrtc/modules/audio_coding/neteq/nack.h"
 
 #include <assert.h>  // For assert.
 
 #include <algorithm>  // For std::max.
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
-
-namespace acm2 {
-
 namespace {
 
 const int kDefaultSampleRateKhz = 48;
@@ -40,6 +38,8 @@ Nack::Nack(int nack_threshold_packets)
       samples_per_packet_(sample_rate_khz_ * kDefaultPacketSizeMs),
       max_nack_list_size_(kNackListSizeLimit) {}
 
+Nack::~Nack() = default;
+
 Nack* Nack::Create(int nack_threshold_packets) {
   return new Nack(nack_threshold_packets);
 }
@@ -87,10 +87,10 @@ void Nack::UpdateLastReceivedPacket(uint16_t sequence_number,
 
 void Nack::UpdateSamplesPerPacket(uint16_t sequence_number_current_received_rtp,
                                   uint32_t timestamp_current_received_rtp) {
-  uint32_t timestamp_increase = timestamp_current_received_rtp -
-      timestamp_last_received_rtp_;
-  uint16_t sequence_num_increase = sequence_number_current_received_rtp -
-      sequence_num_last_received_rtp_;
+  uint32_t timestamp_increase =
+      timestamp_current_received_rtp - timestamp_last_received_rtp_;
+  uint16_t sequence_num_increase =
+      sequence_number_current_received_rtp - sequence_num_last_received_rtp_;
 
   samples_per_packet_ = timestamp_increase / sequence_num_increase;
 }
@@ -106,9 +106,9 @@ void Nack::UpdateList(uint16_t sequence_number_current_received_rtp) {
 
 void Nack::ChangeFromLateToMissing(
     uint16_t sequence_number_current_received_rtp) {
-  NackList::const_iterator lower_bound = nack_list_.lower_bound(
-      static_cast<uint16_t>(sequence_number_current_received_rtp -
-                            nack_threshold_packets_));
+  NackList::const_iterator lower_bound =
+      nack_list_.lower_bound(static_cast<uint16_t>(
+          sequence_number_current_received_rtp - nack_threshold_packets_));
 
   for (NackList::iterator it = nack_list_.begin(); it != lower_bound; ++it)
     it->second.is_missing = true;
@@ -120,16 +120,17 @@ uint32_t Nack::EstimateTimestamp(uint16_t sequence_num) {
 }
 
 void Nack::AddToList(uint16_t sequence_number_current_received_rtp) {
-  assert(!any_rtp_decoded_ || IsNewerSequenceNumber(
-      sequence_number_current_received_rtp, sequence_num_last_decoded_rtp_));
+  assert(!any_rtp_decoded_ ||
+         IsNewerSequenceNumber(sequence_number_current_received_rtp,
+                               sequence_num_last_decoded_rtp_));
 
   // Packets with sequence numbers older than |upper_bound_missing| are
   // considered missing, and the rest are considered late.
-  uint16_t upper_bound_missing = sequence_number_current_received_rtp -
-      nack_threshold_packets_;
+  uint16_t upper_bound_missing =
+      sequence_number_current_received_rtp - nack_threshold_packets_;
 
   for (uint16_t n = sequence_num_last_received_rtp_ + 1;
-      IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) {
+       IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) {
     bool is_missing = IsNewerSequenceNumber(upper_bound_missing, n);
     uint32_t timestamp = EstimateTimestamp(n);
     NackElement nack_element(TimeToPlay(timestamp), timestamp, is_missing);
@@ -139,7 +140,7 @@ void Nack::AddToList(uint16_t sequence_number_current_received_rtp) {
 
 void Nack::UpdateEstimatedPlayoutTimeBy10ms() {
   while (!nack_list_.empty() &&
-      nack_list_.begin()->second.time_to_play_ms <= 10)
+         nack_list_.begin()->second.time_to_play_ms <= 10)
     nack_list_.erase(nack_list_.begin());
 
   for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); ++it)
@@ -155,12 +156,12 @@ void Nack::UpdateLastDecodedPacket(uint16_t sequence_number,
     // Packets in the list with sequence numbers less than the
     // sequence number of the decoded RTP should be removed from the lists.
     // They will be discarded by the jitter buffer if they arrive.
-    nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(
-        sequence_num_last_decoded_rtp_));
+    nack_list_.erase(nack_list_.begin(),
+                     nack_list_.upper_bound(sequence_num_last_decoded_rtp_));
 
     // Update estimated time-to-play.
     for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end();
-        ++it)
+         ++it)
       it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp);
   } else {
     assert(sequence_number == sequence_num_last_decoded_rtp_);
@@ -193,17 +194,19 @@ void Nack::Reset() {
   samples_per_packet_ = sample_rate_khz_ * kDefaultPacketSizeMs;
 }
 
-int Nack::SetMaxNackListSize(size_t max_nack_list_size) {
-  if (max_nack_list_size == 0 || max_nack_list_size > kNackListSizeLimit)
-    return -1;
+void Nack::SetMaxNackListSize(size_t max_nack_list_size) {
+  RTC_CHECK_GT(max_nack_list_size, 0u);
+  // Ugly hack to get around the problem of passing static consts by reference.
+  const size_t kNackListSizeLimitLocal = Nack::kNackListSizeLimit;
+  RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal);
+
   max_nack_list_size_ = max_nack_list_size;
   LimitNackListSize();
-  return 0;
 }
 
 void Nack::LimitNackListSize() {
   uint16_t limit = sequence_num_last_received_rtp_ -
-      static_cast<uint16_t>(max_nack_list_size_) - 1;
+                   static_cast<uint16_t>(max_nack_list_size_) - 1;
   nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(limit));
 }
 
@@ -214,9 +217,10 @@ int64_t Nack::TimeToPlay(uint32_t timestamp) const {
 
 // We don't erase elements with time-to-play shorter than round-trip-time.
 std::vector<uint16_t> Nack::GetNackList(int64_t round_trip_time_ms) const {
+  RTC_DCHECK_GE(round_trip_time_ms, 0);
   std::vector<uint16_t> sequence_numbers;
   for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
-      ++it) {
+       ++it) {
     if (it->second.is_missing &&
         it->second.time_to_play_ms > round_trip_time_ms)
       sequence_numbers.push_back(it->first);
@@ -224,6 +228,4 @@ std::vector<uint16_t> Nack::GetNackList(int64_t round_trip_time_ms) const {
   return sequence_numbers;
 }
 
-}  // namespace acm2
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h
similarity index 94%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h
index 4224c99f79..17fef46464 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_
 
 #include <vector>
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 
 //
@@ -49,8 +49,6 @@
 //
 namespace webrtc {
 
-namespace acm2 {
-
 class Nack {
  public:
   // A limit for the size of the NACK list.
@@ -59,14 +57,14 @@ class Nack {
   // Factory method.
   static Nack* Create(int nack_threshold_packets);
 
-  ~Nack() {}
+  ~Nack();
 
   // Set a maximum for the size of the NACK list. If the last received packet
   // has sequence number of N, then NACK list will not contain any element
   // with sequence number earlier than N - |max_nack_list_size|.
   //
   // The largest maximum size is defined by |kNackListSizeLimit|
-  int SetMaxNackListSize(size_t max_nack_list_size);
+  void SetMaxNackListSize(size_t max_nack_list_size);
 
   // Set the sampling rate.
   //
@@ -124,8 +122,8 @@ class Nack {
 
   class NackListCompare {
    public:
-    bool operator() (uint16_t sequence_number_old,
-                     uint16_t sequence_number_new) const {
+    bool operator()(uint16_t sequence_number_old,
+                    uint16_t sequence_number_new) const {
       return IsNewerSequenceNumber(sequence_number_new, sequence_number_old);
     }
   };
@@ -206,8 +204,6 @@ class Nack {
   size_t max_nack_list_size_;
 };
 
-}  // namespace acm2
-
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc
similarity index 90%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc
index c880e32808..53b19dc50f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/acm2/nack.h"
+#include "webrtc/modules/audio_coding/neteq/nack.h"
 
 #include <stdint.h>
 
@@ -17,12 +17,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 
 namespace webrtc {
-
-namespace acm2 {
-
 namespace {
 
 const int kNackThreshold = 3;
@@ -90,8 +87,9 @@ TEST(NackTest, NoNackIfReorderWithinNackThreshold) {
 
   // Push in reverse order
   while (num_late_packets > 0) {
-    nack->UpdateLastReceivedPacket(seq_num + num_late_packets, timestamp +
-                            num_late_packets * kTimestampIncrement);
+    nack->UpdateLastReceivedPacket(
+        seq_num + num_late_packets,
+        timestamp + num_late_packets * kTimestampIncrement);
     nack_list = nack->GetNackList(kShortRoundTripTimeMs);
     EXPECT_TRUE(nack_list.empty());
     num_late_packets--;
@@ -99,9 +97,9 @@ TEST(NackTest, NoNackIfReorderWithinNackThreshold) {
 }
 
 TEST(NackTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
-  const uint16_t kSequenceNumberLostPackets[] = { 2, 3, 4, 5, 6, 7, 8, 9 };
+  const uint16_t kSequenceNumberLostPackets[] = {2, 3, 4, 5, 6, 7, 8, 9};
   static const int kNumAllLostPackets = sizeof(kSequenceNumberLostPackets) /
-      sizeof(kSequenceNumberLostPackets[0]);
+                                        sizeof(kSequenceNumberLostPackets[0]);
 
   for (int k = 0; k < 2; k++) {  // Two iteration with/without wrap around.
     rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
@@ -109,8 +107,9 @@ TEST(NackTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
 
     uint16_t sequence_num_lost_packets[kNumAllLostPackets];
     for (int n = 0; n < kNumAllLostPackets; n++) {
-      sequence_num_lost_packets[n] = kSequenceNumberLostPackets[n] + k *
-          65531;  // Have wrap around in sequence numbers for |k == 1|.
+      sequence_num_lost_packets[n] =
+          kSequenceNumberLostPackets[n] +
+          k * 65531;  // Have wrap around in sequence numbers for |k == 1|.
     }
     uint16_t seq_num = sequence_num_lost_packets[0] - 1;
 
@@ -147,9 +146,9 @@ TEST(NackTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
 }
 
 TEST(NackTest, ArrivedPacketsAreRemovedFromNackList) {
-  const uint16_t kSequenceNumberLostPackets[] = { 2, 3, 4, 5, 6, 7, 8, 9 };
+  const uint16_t kSequenceNumberLostPackets[] = {2, 3, 4, 5, 6, 7, 8, 9};
   static const int kNumAllLostPackets = sizeof(kSequenceNumberLostPackets) /
-      sizeof(kSequenceNumberLostPackets[0]);
+                                        sizeof(kSequenceNumberLostPackets[0]);
 
   for (int k = 0; k < 2; ++k) {  // Two iteration with/without wrap around.
     rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
@@ -157,8 +156,8 @@ TEST(NackTest, ArrivedPacketsAreRemovedFromNackList) {
 
     uint16_t sequence_num_lost_packets[kNumAllLostPackets];
     for (int n = 0; n < kNumAllLostPackets; ++n) {
-      sequence_num_lost_packets[n] = kSequenceNumberLostPackets[n] + k *
-          65531;  // Wrap around for |k == 1|.
+      sequence_num_lost_packets[n] = kSequenceNumberLostPackets[n] +
+                                     k * 65531;  // Wrap around for |k == 1|.
     }
 
     uint16_t seq_num = sequence_num_lost_packets[0] - 1;
@@ -208,11 +207,10 @@ TEST(NackTest, ArrivedPacketsAreRemovedFromNackList) {
 // Assess if estimation of timestamps and time-to-play is correct. Introduce all
 // combinations that timestamps and sequence numbers might have wrap around.
 TEST(NackTest, EstimateTimestampAndTimeToPlay) {
-  const uint16_t kLostPackets[] = { 2, 3, 4, 5, 6, 7, 8, 9, 10,
-      11, 12, 13, 14, 15 };
-  static const int kNumAllLostPackets = sizeof(kLostPackets) /
-      sizeof(kLostPackets[0]);
-
+  const uint16_t kLostPackets[] = {2, 3,  4,  5,  6,  7,  8,
+                                   9, 10, 11, 12, 13, 14, 15};
+  static const int kNumAllLostPackets =
+      sizeof(kLostPackets) / sizeof(kLostPackets[0]);
 
   for (int k = 0; k < 4; ++k) {
     rtc::scoped_ptr<Nack> nack(Nack::Create(kNackThreshold));
@@ -222,14 +220,14 @@ TEST(NackTest, EstimateTimestampAndTimeToPlay) {
     int seq_num_offset = (k < 2) ? 0 : 65531;
 
     // Timestamp wrap around if |k| is 1 or 3.
-    uint32_t timestamp_offset = (k & 0x1) ?
-        static_cast<uint32_t>(0xffffffff) - 6 : 0;
+    uint32_t timestamp_offset =
+        (k & 0x1) ? static_cast<uint32_t>(0xffffffff) - 6 : 0;
 
     uint32_t timestamp_lost_packets[kNumAllLostPackets];
     uint16_t seq_num_lost_packets[kNumAllLostPackets];
     for (int n = 0; n < kNumAllLostPackets; ++n) {
-      timestamp_lost_packets[n] = timestamp_offset + kLostPackets[n] *
-          kTimestampIncrement;
+      timestamp_lost_packets[n] =
+          timestamp_offset + kLostPackets[n] * kTimestampIncrement;
       seq_num_lost_packets[n] = seq_num_offset + kLostPackets[n];
     }
 
@@ -248,8 +246,8 @@ TEST(NackTest, EstimateTimestampAndTimeToPlay) {
 
     // A packet after the last one which is supposed to be lost.
     seq_num = seq_num_lost_packets[kNumAllLostPackets - 1] + 1;
-    timestamp = timestamp_lost_packets[kNumAllLostPackets - 1] +
-        kTimestampIncrement;
+    timestamp =
+        timestamp_lost_packets[kNumAllLostPackets - 1] + kTimestampIncrement;
     nack->UpdateLastReceivedPacket(seq_num, timestamp);
 
     Nack::NackList nack_list = nack->GetNackList();
@@ -292,16 +290,16 @@ TEST(NackTest, MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
     // Two consecutive packets to have a correct estimate of timestamp increase.
     uint16_t seq_num = 0;
     nack->UpdateLastReceivedPacket(seq_num_offset + seq_num,
-      seq_num * kTimestampIncrement);
+                                   seq_num * kTimestampIncrement);
     seq_num++;
     nack->UpdateLastReceivedPacket(seq_num_offset + seq_num,
-      seq_num * kTimestampIncrement);
+                                   seq_num * kTimestampIncrement);
 
     // Skip 10 packets (larger than NACK threshold).
     const int kNumLostPackets = 10;
     seq_num += kNumLostPackets + 1;
     nack->UpdateLastReceivedPacket(seq_num_offset + seq_num,
-      seq_num * kTimestampIncrement);
+                                   seq_num * kTimestampIncrement);
 
     const size_t kExpectedListSize = kNumLostPackets - kNackThreshold;
     std::vector<uint16_t> nack_list = nack->GetNackList(kShortRoundTripTimeMs);
@@ -319,7 +317,7 @@ TEST(NackTest, MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
 
     // Decoding of the last received packet.
     nack->UpdateLastDecodedPacket(seq_num + seq_num_offset,
-      seq_num * kTimestampIncrement);
+                                  seq_num * kTimestampIncrement);
     nack_list = nack->GetNackList(kShortRoundTripTimeMs);
     EXPECT_TRUE(nack_list.empty());
 
@@ -329,7 +327,7 @@ TEST(NackTest, MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
     for (int n = 0; n < kNackThreshold + 10; ++n) {
       seq_num++;
       nack->UpdateLastReceivedPacket(seq_num_offset + seq_num,
-       seq_num * kTimestampIncrement);
+                                     seq_num * kTimestampIncrement);
       nack_list = nack->GetNackList(kShortRoundTripTimeMs);
       EXPECT_TRUE(nack_list.empty());
     }
@@ -481,6 +479,4 @@ TEST(NackTest, RoudTripTimeIsApplied) {
   EXPECT_EQ(5, nack_list[1]);
 }
 
-}  // namespace acm2
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.cc
index 420165b3f7..c31dbdc1a3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.cc
@@ -8,7 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+
+#include <sstream>
 
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
@@ -26,6 +28,19 @@
 
 namespace webrtc {
 
+std::string NetEq::Config::ToString() const {
+  std::stringstream ss;
+  ss << "sample_rate_hz=" << sample_rate_hz << ", enable_audio_classifier="
+     << (enable_audio_classifier ? "true" : "false")
+     << ", enable_post_decode_vad="
+     << (enable_post_decode_vad ? "true" : "false")
+     << ", max_packets_in_buffer=" << max_packets_in_buffer
+     << ", background_noise_mode=" << background_noise_mode
+     << ", playout_mode=" << playout_mode
+     << ", enable_fast_accelerate=" << enable_fast_accelerate;
+  return ss.str();
+}
+
 // Creates all classes needed and inject them into a new NetEqImpl object.
 // Return the new object.
 NetEq* NetEq::Create(const NetEq::Config& config) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi
index 82c56a7620..15c5a06582 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -9,36 +9,28 @@
 {
   'variables': {
     'codecs': [
-      'G711',
-      'PCM16B',
-      'CNG',
+      'cng',
+      'g711',
+      'pcm16b',
     ],
     'neteq_defines': [],
     'conditions': [
+      ['include_opus==1', {
+        'codecs': ['webrtc_opus',],
+        'neteq_defines': ['WEBRTC_CODEC_OPUS',],
+      }],
       ['include_g722==1', {
-        'neteq_dependencies': ['G722'],
+        'codecs': ['g722',],
         'neteq_defines': ['WEBRTC_CODEC_G722',],
       }],
       ['include_ilbc==1', {
-        'neteq_dependencies': ['iLBC'],
+        'codecs': ['ilbc',],
         'neteq_defines': ['WEBRTC_CODEC_ILBC',],
       }],
       ['include_isac==1', {
-        'neteq_dependencies': ['iSAC', 'iSACFix',],
+        'codecs': ['isac', 'isac_fix',],
         'neteq_defines': ['WEBRTC_CODEC_ISAC', 'WEBRTC_CODEC_ISACFIX',],
       }],
-      ['include_opus==1', {
-        'codecs': ['webrtc_opus'],
-        'neteq_dependencies': ['webrtc_opus'],
-        'neteq_defines': ['WEBRTC_CODEC_OPUS',],
-        'conditions': [
-          ['build_with_mozilla==0', {
-            'neteq_dependencies': [
-              '<(DEPTH)/third_party/opus/opus.gyp:opus',
-	    ],
-	  }],
- 	],
-      }],
     ],
     'neteq_dependencies': [
       '<@(codecs)',
@@ -101,7 +93,7 @@
         }],
       ],
       'sources': [
-        'interface/neteq.h',
+        'include/neteq.h',
         'accelerate.cc',
         'accelerate.h',
         'audio_classifier.cc',
@@ -141,6 +133,8 @@
         'expand.h',
         'merge.cc',
         'merge.h',
+        'nack.h',
+        'nack.cc',
         'neteq_impl.cc',
         'neteq_impl.h',
         'neteq.cc',
@@ -178,20 +172,17 @@
           'type': '<(gtest_target_type)',
           'dependencies': [
             '<@(codecs)',
+            'g722',
+            'ilbc',
+            'isac',
+            'isac_fix',
             'audio_decoder_interface',
             'neteq_unittest_tools',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
-# FIX for include_isac/etc
           'defines': [
-            'AUDIO_DECODER_UNITTEST',
-            'WEBRTC_CODEC_G722',
-            'WEBRTC_CODEC_ILBC',
-            'WEBRTC_CODEC_ISACFX',
-            'WEBRTC_CODEC_ISAC',
-            'WEBRTC_CODEC_PCM16',
             '<@(neteq_defines)',
           ],
           'sources': [
@@ -259,23 +250,6 @@
             },
           ],
         }],
-        ['test_isolation_mode != "noop"', {
-          'targets': [
-            {
-              'target_name': 'audio_decoder_unittests_run',
-              'type': 'none',
-              'dependencies': [
-                'audio_decoder_unittests',
-              ],
-              'includes': [
-                '../../../build/isolate.gypi',
-              ],
-              'sources': [
-                'audio_decoder_unittests.isolate',
-              ],
-            },
-          ],
-        }],
       ],
     }], # include_tests
   ], # conditions
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index f0158b972c..c03fbb7347 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -40,8 +40,6 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
         payload_size_bytes_(0),
         last_send_time_(0),
         last_arrival_time_(0) {
-    // Init() will trigger external_decoder_->Init().
-    EXPECT_CALL(*external_decoder_, Init());
     // NetEq is not allowed to delete the external decoder (hence Times(0)).
     EXPECT_CALL(*external_decoder_, Die()).Times(0);
     Init();
@@ -73,7 +71,7 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
       return -1;
     }
     payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
-                                              encoded_);;
+                                              encoded_);
 
     int next_send_time = rtp_generator_->GetRtpHeader(
         kPayloadType, frame_size_samples_, &rtp_header_);
@@ -100,14 +98,16 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
       next_arrival_time = GetArrivalTime(next_send_time);
     } while (Lost());  // If lost, immediately read the next packet.
 
-    EXPECT_CALL(*external_decoder_,
-                Decode(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _, _))
+    EXPECT_CALL(
+        *external_decoder_,
+        DecodeInternal(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _))
         .Times(NumExpectedDecodeCalls(num_loops));
 
     uint32_t time_now = 0;
     for (int k = 0; k < num_loops; ++k) {
       while (time_now >= next_arrival_time) {
-        InsertPacket(rtp_header_, encoded_, payload_size_bytes_,
+        InsertPacket(rtp_header_, rtc::ArrayView<const uint8_t>(
+                                      encoded_, payload_size_bytes_),
                      next_arrival_time);
         // Get next input packet.
         do {
@@ -126,17 +126,14 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
     }
   }
 
-  void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                    size_t payload_size_bytes,
+  void InsertPacket(WebRtcRTPHeader rtp_header,
+                    rtc::ArrayView<const uint8_t> payload,
                     uint32_t receive_timestamp) override {
-    EXPECT_CALL(*external_decoder_,
-                IncomingPacket(_,
-                               payload_size_bytes,
-                               rtp_header.header.sequenceNumber,
-                               rtp_header.header.timestamp,
-                               receive_timestamp));
+    EXPECT_CALL(
+        *external_decoder_,
+        IncomingPacket(_, payload.size(), rtp_header.header.sequenceNumber,
+                       rtp_header.header.timestamp, receive_timestamp));
     NetEqExternalDecoderTest::InsertPacket(rtp_header, payload,
-                                           payload_size_bytes,
                                            receive_timestamp);
   }
 
@@ -169,27 +166,29 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
 class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
                                            public ::testing::Test {
  protected:
-  static const int kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
+  static const size_t kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
 
   NetEqExternalVsInternalDecoderTest()
-      : NetEqExternalDecoderUnitTest(kDecoderPCM16Bswb32kHz,
+      : NetEqExternalDecoderUnitTest(NetEqDecoder::kDecoderPCM16Bswb32kHz,
                                      new MockExternalPcm16B),
-        sample_rate_hz_(CodecSampleRateHz(kDecoderPCM16Bswb32kHz)) {
+        sample_rate_hz_(
+            CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb32kHz)) {
     NetEq::Config config;
-    config.sample_rate_hz = CodecSampleRateHz(kDecoderPCM16Bswb32kHz);
+    config.sample_rate_hz =
+        CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb32kHz);
     neteq_internal_.reset(NetEq::Create(config));
   }
 
   void SetUp() override {
-    ASSERT_EQ(NetEq::kOK,
-              neteq_internal_->RegisterPayloadType(kDecoderPCM16Bswb32kHz,
-                                                   kPayloadType));
+    ASSERT_EQ(NetEq::kOK, neteq_internal_->RegisterPayloadType(
+                              NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                              "pcm16-swb32", kPayloadType));
   }
 
   void GetAndVerifyOutput() override {
     NetEqOutputType output_type;
-    int samples_per_channel;
-    int num_channels;
+    size_t samples_per_channel;
+    size_t num_channels;
     // Get audio from internal decoder instance.
     EXPECT_EQ(NetEq::kOK,
               neteq_internal_->GetAudio(kMaxBlockSize,
@@ -197,30 +196,28 @@ class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
                                         &samples_per_channel,
                                         &num_channels,
                                         &output_type));
-    EXPECT_EQ(1, num_channels);
-    EXPECT_EQ(kOutputLengthMs * sample_rate_hz_ / 1000, samples_per_channel);
+    EXPECT_EQ(1u, num_channels);
+    EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
+              samples_per_channel);
 
     // Get audio from external decoder instance.
     samples_per_channel = GetOutputAudio(kMaxBlockSize, output_, &output_type);
 
-    for (int i = 0; i < samples_per_channel; ++i) {
+    for (size_t i = 0; i < samples_per_channel; ++i) {
       ASSERT_EQ(output_[i], output_internal_[i]) <<
           "Diff in sample " << i << ".";
     }
   }
 
-  void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                    size_t payload_size_bytes,
+  void InsertPacket(WebRtcRTPHeader rtp_header,
+                    rtc::ArrayView<const uint8_t> payload,
                     uint32_t receive_timestamp) override {
     // Insert packet in internal decoder.
-    ASSERT_EQ(
-        NetEq::kOK,
-        neteq_internal_->InsertPacket(
-            rtp_header, payload, payload_size_bytes, receive_timestamp));
+    ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload,
+                                                        receive_timestamp));
 
     // Insert packet in external decoder instance.
     NetEqExternalDecoderUnitTest::InsertPacket(rtp_header, payload,
-                                               payload_size_bytes,
                                                receive_timestamp);
   }
 
@@ -240,7 +237,7 @@ TEST_F(NetEqExternalVsInternalDecoderTest, RunTest) {
 class LargeTimestampJumpTest : public NetEqExternalDecoderUnitTest,
                                public ::testing::Test {
  protected:
-  static const int kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
+  static const size_t kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
 
   enum TestStates {
     kInitialPhase,
@@ -251,7 +248,7 @@ class LargeTimestampJumpTest : public NetEqExternalDecoderUnitTest,
   };
 
   LargeTimestampJumpTest()
-      : NetEqExternalDecoderUnitTest(kDecoderPCM16B,
+      : NetEqExternalDecoderUnitTest(NetEqDecoder::kDecoderPCM16B,
                                      new MockExternalPcm16B),
         test_state_(kInitialPhase) {
     EXPECT_CALL(*external_decoder(), HasDecodePlc())
@@ -293,7 +290,7 @@ class LargeTimestampJumpTest : public NetEqExternalDecoderUnitTest,
   }
 
   void GetAndVerifyOutput() override {
-    int num_samples;
+    size_t num_samples;
     NetEqOutputType output_type;
     num_samples = GetOutputAudio(kMaxBlockSize, output_, &output_type);
     UpdateState(output_type);
@@ -303,7 +300,7 @@ class LargeTimestampJumpTest : public NetEqExternalDecoderUnitTest,
       return;
     }
 
-    for (int i = 0; i < num_samples; ++i) {
+    for (size_t i = 0; i < num_samples; ++i) {
       if (output_[i] != 0)
         return;
     }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 32c66295ed..6c07da46f0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -15,6 +15,10 @@
 
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
@@ -30,6 +34,7 @@
 #include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/merge.h"
+#include "webrtc/modules/audio_coding/neteq/nack.h"
 #include "webrtc/modules/audio_coding/neteq/normal.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
@@ -38,9 +43,8 @@
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
 // longer required, this #define should be removed (and the code that it
@@ -84,7 +88,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
       new_codec_(false),
       timestamp_(0),
       reset_decoder_(false),
-      current_rtp_payload_type_(0xFF),  // Invalid RTP payload type.
+      current_rtp_payload_type_(0xFF),      // Invalid RTP payload type.
       current_cng_rtp_payload_type_(0xFF),  // Invalid RTP payload type.
       ssrc_(0),
       first_packet_(true),
@@ -92,43 +96,40 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
       decoder_error_code_(0),
       background_noise_mode_(config.background_noise_mode),
       playout_mode_(config.playout_mode),
-      decoded_packet_sequence_number_(-1),
-      decoded_packet_timestamp_(0) {
+      enable_fast_accelerate_(config.enable_fast_accelerate),
+      nack_enabled_(false) {
+  LOG(LS_INFO) << "NetEq config: " << config.ToString();
   int fs = config.sample_rate_hz;
   if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
     LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. " <<
         "Changing to 8000 Hz.";
     fs = 8000;
   }
-  LOG(LS_VERBOSE) << "Create NetEqImpl object with fs = " << fs << ".";
   fs_hz_ = fs;
   fs_mult_ = fs / 8000;
-  output_size_samples_ = kOutputSizeMs * 8 * fs_mult_;
+  last_output_sample_rate_hz_ = fs;
+  output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
   decoder_frame_length_ = 3 * output_size_samples_;
   WebRtcSpl_Init();
   if (create_components) {
     SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
   }
+  RTC_DCHECK(!vad_->enabled());
+  if (config.enable_post_decode_vad) {
+    vad_->Enable();
+  }
 }
 
-NetEqImpl::~NetEqImpl() {
-  LOG(LS_INFO) << "Deleting NetEqImpl object.";
-}
+NetEqImpl::~NetEqImpl() = default;
 
 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
-                            const uint8_t* payload,
-                            size_t length_bytes,
+                            rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp) {
+  TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
-      ", sn=" << rtp_header.header.sequenceNumber <<
-      ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
-      ", ssrc=" << rtp_header.header.ssrc <<
-      ", len=" << length_bytes;
-  int error = InsertPacketInternal(rtp_header, payload, length_bytes,
-                                   receive_timestamp, false);
+  int error =
+      InsertPacketInternal(rtp_header, payload, receive_timestamp, false);
   if (error != 0) {
-    LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
     error_code_ = error;
     return kFail;
   }
@@ -138,18 +139,11 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                 uint32_t receive_timestamp) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket-Sync: ts="
-      << rtp_header.header.timestamp <<
-      ", sn=" << rtp_header.header.sequenceNumber <<
-      ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
-      ", ssrc=" << rtp_header.header.ssrc;
-
   const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
-  int error = InsertPacketInternal(
-      rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
+  int error =
+      InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
 
   if (error != 0) {
-    LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
     error_code_ = error;
     return kFail;
   }
@@ -157,33 +151,38 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
 }
 
 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
-                        int* samples_per_channel, int* num_channels,
+                        size_t* samples_per_channel, size_t* num_channels,
                         NetEqOutputType* type) {
+  TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "GetAudio";
   int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
                                num_channels);
-  LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
-      " samples/channel for " << *num_channels << " channel(s)";
   if (error != 0) {
-    LOG_FERR1(LS_WARNING, GetAudioInternal, error);
     error_code_ = error;
     return kFail;
   }
   if (type) {
     *type = LastOutputType();
   }
+  last_output_sample_rate_hz_ =
+      rtc::checked_cast<int>(*samples_per_channel * 100);
+  RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
+             last_output_sample_rate_hz_ == 16000 ||
+             last_output_sample_rate_hz_ == 32000 ||
+             last_output_sample_rate_hz_ == 48000)
+      << "Unexpected sample rate " << last_output_sample_rate_hz_;
   return kOK;
 }
 
-int NetEqImpl::RegisterPayloadType(enum NetEqDecoder codec,
+int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
+                                   const std::string& name,
                                    uint8_t rtp_payload_type) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG_API2(static_cast<int>(rtp_payload_type), codec);
-  int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec);
+  LOG(LS_VERBOSE) << "RegisterPayloadType "
+                  << static_cast<int>(rtp_payload_type) << " "
+                  << static_cast<int>(codec);
+  int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec, name);
   if (ret != DecoderDatabase::kOK) {
-    LOG_FERR2(LS_WARNING, RegisterPayload, static_cast<int>(rtp_payload_type),
-              codec);
     switch (ret) {
       case DecoderDatabase::kInvalidRtpPayloadType:
         error_code_ = kInvalidRtpPayloadType;
@@ -203,21 +202,22 @@ int NetEqImpl::RegisterPayloadType(enum NetEqDecoder codec,
 }
 
 int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
-                                       enum NetEqDecoder codec,
-                                       uint8_t rtp_payload_type) {
+                                       NetEqDecoder codec,
+                                       const std::string& codec_name,
+                                       uint8_t rtp_payload_type,
+                                       int sample_rate_hz) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG_API2(static_cast<int>(rtp_payload_type), codec);
+  LOG(LS_VERBOSE) << "RegisterExternalDecoder "
+                  << static_cast<int>(rtp_payload_type) << " "
+                  << static_cast<int>(codec);
   if (!decoder) {
     LOG(LS_ERROR) << "Cannot register external decoder with NULL pointer";
     assert(false);
     return kFail;
   }
-  const int sample_rate_hz = CodecSampleRateHz(codec);
-  int ret = decoder_database_->InsertExternal(rtp_payload_type, codec,
-                                              sample_rate_hz, decoder);
+  int ret = decoder_database_->InsertExternal(
+      rtp_payload_type, codec, codec_name, sample_rate_hz, decoder);
   if (ret != DecoderDatabase::kOK) {
-    LOG_FERR2(LS_WARNING, InsertExternal, static_cast<int>(rtp_payload_type),
-              codec);
     switch (ret) {
       case DecoderDatabase::kInvalidRtpPayloadType:
         error_code_ = kInvalidRtpPayloadType;
@@ -244,7 +244,6 @@ int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
 
 int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG_API1(static_cast<int>(rtp_payload_type));
   int ret = decoder_database_->Remove(rtp_payload_type);
   if (ret == DecoderDatabase::kOK) {
     return kOK;
@@ -253,7 +252,6 @@ int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
   } else {
     error_code_ = kOtherError;
   }
-  LOG_FERR1(LS_WARNING, Remove, static_cast<int>(rtp_payload_type));
   return kFail;
 }
 
@@ -281,6 +279,30 @@ int NetEqImpl::LeastRequiredDelayMs() const {
   return delay_manager_->least_required_delay_ms();
 }
 
+int NetEqImpl::SetTargetDelay() {
+  return kNotImplemented;
+}
+
+int NetEqImpl::TargetDelay() {
+  return kNotImplemented;
+}
+
+int NetEqImpl::CurrentDelayMs() const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  if (fs_hz_ == 0)
+    return 0;
+  // Sum up the samples in the packet buffer with the future length of the sync
+  // buffer, and divide the sum by the sample rate.
+  const size_t delay_samples =
+      packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
+                                         decoder_frame_length_) +
+      sync_buffer_->FutureLength();
+  // The division below will truncate.
+  const int delay_ms =
+      static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
+  return delay_ms;
+}
+
 // Deprecated.
 // TODO(henrik.lundin) Delete.
 void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
@@ -301,9 +323,10 @@ NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
 int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
   CriticalSectionScoped lock(crit_sect_.get());
   assert(decoder_database_.get());
-  const int total_samples_in_buffers = packet_buffer_->NumSamplesInBuffer(
-      decoder_database_.get(), decoder_frame_length_) +
-          static_cast<int>(sync_buffer_->FutureLength());
+  const size_t total_samples_in_buffers =
+      packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
+                                         decoder_frame_length_) +
+      sync_buffer_->FutureLength();
   assert(delay_manager_.get());
   assert(decision_logic_.get());
   stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
@@ -312,11 +335,6 @@ int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
   return 0;
 }
 
-void NetEqImpl::WaitingTimes(std::vector<int>* waiting_times) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  stats_.WaitingTimes(waiting_times);
-}
-
 void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
   CriticalSectionScoped lock(crit_sect_.get());
   if (stats) {
@@ -354,6 +372,19 @@ bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) {
   return true;
 }
 
+int NetEqImpl::last_output_sample_rate_hz() const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  return last_output_sample_rate_hz_;
+}
+
+int NetEqImpl::SetTargetNumberOfChannels() {
+  return kNotImplemented;
+}
+
+int NetEqImpl::SetTargetSampleRate() {
+  return kNotImplemented;
+}
+
 int NetEqImpl::LastError() const {
   CriticalSectionScoped lock(crit_sect_.get());
   return error_code_;
@@ -366,7 +397,7 @@ int NetEqImpl::LastDecoderError() {
 
 void NetEqImpl::FlushBuffers() {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG_API0();
+  LOG(LS_VERBOSE) << "FlushBuffers";
   packet_buffer_->Flush();
   assert(sync_buffer_.get());
   assert(expand_.get());
@@ -383,13 +414,30 @@ void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
   packet_buffer_->BufferStat(current_num_packets, max_num_packets);
 }
 
-int NetEqImpl::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const {
+void NetEqImpl::EnableNack(size_t max_nack_list_size) {
   CriticalSectionScoped lock(crit_sect_.get());
-  if (decoded_packet_sequence_number_ < 0)
-    return -1;
-  *sequence_number = decoded_packet_sequence_number_;
-  *timestamp = decoded_packet_timestamp_;
-  return 0;
+  if (!nack_enabled_) {
+    const int kNackThresholdPackets = 2;
+    nack_.reset(Nack::Create(kNackThresholdPackets));
+    nack_enabled_ = true;
+    nack_->UpdateSampleRate(fs_hz_);
+  }
+  nack_->SetMaxNackListSize(max_nack_list_size);
+}
+
+void NetEqImpl::DisableNack() {
+  CriticalSectionScoped lock(crit_sect_.get());
+  nack_.reset();
+  nack_enabled_ = false;
+}
+
+std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
+  CriticalSectionScoped lock(crit_sect_.get());
+  if (!nack_enabled_) {
+    return std::vector<uint16_t>();
+  }
+  RTC_DCHECK(nack_.get());
+  return nack_->GetNackList(round_trip_time_ms);
 }
 
 const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
@@ -400,12 +448,11 @@ const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
 // Methods below this line are private.
 
 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
-                                    const uint8_t* payload,
-                                    size_t length_bytes,
+                                    rtc::ArrayView<const uint8_t> payload,
                                     uint32_t receive_timestamp,
                                     bool is_sync_packet) {
-  if (!payload) {
-    LOG_F(LS_ERROR) << "payload == NULL";
+  if (payload.empty()) {
+    LOG_F(LS_ERROR) << "payload is empty";
     return kInvalidPointer;
   }
   // Sanity checks for sync-packets.
@@ -441,7 +488,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     packet->header.timestamp = rtp_header.header.timestamp;
     packet->header.ssrc = rtp_header.header.ssrc;
     packet->header.numCSRCs = 0;
-    packet->payload_length = length_bytes;
+    packet->payload_length = payload.size();
     packet->primary = true;
     packet->waiting_time = 0;
     packet->payload = new uint8_t[packet->payload_length];
@@ -449,8 +496,8 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     if (!packet->payload) {
       LOG_F(LS_ERROR) << "Payload pointer is NULL.";
     }
-    assert(payload);  // Already checked above.
-    memcpy(packet->payload, payload, packet->payload_length);
+    assert(!payload.empty());  // Already checked above.
+    memcpy(packet->payload, payload.data(), packet->payload_length);
     // Insert packet in a packet list.
     packet_list.push_back(packet);
     // Save main payloads header for later.
@@ -494,7 +541,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   if (decoder_database_->IsRed(main_header.payloadType)) {
     assert(!is_sync_packet);  // We had a sanity check for this.
     if (payload_splitter_->SplitRed(&packet_list) != PayloadSplitter::kOK) {
-      LOG_FERR1(LS_WARNING, SplitRed, packet_list.size());
       PacketBuffer::DeleteAllPackets(&packet_list);
       return kRedundancySplitError;
     }
@@ -509,7 +555,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   // Check payload types.
   if (decoder_database_->CheckPayloadTypes(packet_list) ==
       DecoderDatabase::kDecoderNotFound) {
-    LOG_FERR1(LS_WARNING, CheckPayloadTypes, packet_list.size());
     PacketBuffer::DeleteAllPackets(&packet_list);
     return kUnknownRtpPayloadType;
   }
@@ -533,13 +578,10 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
           current_packet->payload_length,
           &event);
       if (ret != DtmfBuffer::kOK) {
-        LOG_FERR2(LS_WARNING, ParseEvent, ret,
-                  current_packet->payload_length);
         PacketBuffer::DeleteAllPackets(&packet_list);
         return kDtmfParsingError;
       }
       if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) {
-        LOG_FERR0(LS_WARNING, InsertEvent);
         PacketBuffer::DeleteAllPackets(&packet_list);
         return kDtmfInsertError;
       }
@@ -555,7 +597,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   // Check for FEC in packets, and separate payloads into several packets.
   int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get());
   if (ret != PayloadSplitter::kOK) {
-    LOG_FERR1(LS_WARNING, SplitFec, packet_list.size());
     PacketBuffer::DeleteAllPackets(&packet_list);
     switch (ret) {
       case PayloadSplitter::kUnknownPayloadType:
@@ -570,7 +611,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   // sync-packets.
   ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_);
   if (ret != PayloadSplitter::kOK) {
-    LOG_FERR1(LS_WARNING, SplitAudio, packet_list.size());
     PacketBuffer::DeleteAllPackets(&packet_list);
     switch (ret) {
       case PayloadSplitter::kUnknownPayloadType:
@@ -596,8 +636,18 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
                             receive_timestamp);
   }
 
+  if (nack_enabled_) {
+    RTC_DCHECK(nack_);
+    if (update_sample_rate_and_channels) {
+      nack_->Reset();
+    }
+    nack_->UpdateLastReceivedPacket(packet_list.front()->header.sequenceNumber,
+                                    packet_list.front()->header.timestamp);
+  }
+
   // Insert packets in buffer.
-  int temp_bufsize = packet_buffer_->NumPacketsInBuffer();
+  const size_t buffer_length_before_insert =
+      packet_buffer_->NumPacketsInBuffer();
   ret = packet_buffer_->InsertPacketList(
       &packet_list,
       *decoder_database_,
@@ -607,9 +657,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     // Reset DSP timestamp etc. if packet buffer flushed.
     new_codec_ = true;
     update_sample_rate_and_channels = true;
-    LOG_F(LS_WARNING) << "Packet buffer flushed";
   } else if (ret != PacketBuffer::kOK) {
-    LOG_FERR1(LS_WARNING, InsertPacketList, packet_list.size());
     PacketBuffer::DeleteAllPackets(&packet_list);
     return kOtherError;
   }
@@ -644,8 +692,14 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
         decoder_database_->GetDecoderInfo(payload_type);
     assert(decoder_info);
     if (decoder_info->fs_hz != fs_hz_ ||
-        decoder->Channels() != algorithm_buffer_->Channels())
+        decoder->Channels() != algorithm_buffer_->Channels()) {
       SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
+    }
+    if (nack_enabled_) {
+      RTC_DCHECK(nack_);
+      // Update the sample rate even if the rate is not new, because of Reset().
+      nack_->UpdateSampleRate(fs_hz_);
+    }
   }
 
   // TODO(hlundin): Move this code to DelayManager class.
@@ -655,13 +709,18 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   delay_manager_->LastDecoderType(dec_info->codec_type);
   if (delay_manager_->last_pack_cng_or_dtmf() == 0) {
     // Calculate the total speech length carried in each packet.
-    temp_bufsize = packet_buffer_->NumPacketsInBuffer() - temp_bufsize;
-    temp_bufsize *= decoder_frame_length_;
+    const size_t buffer_length_after_insert =
+        packet_buffer_->NumPacketsInBuffer();
 
-    if ((temp_bufsize > 0) &&
-        (temp_bufsize != decision_logic_->packet_length_samples())) {
-      decision_logic_->set_packet_length_samples(temp_bufsize);
-      delay_manager_->SetPacketAudioLength((1000 * temp_bufsize) / fs_hz_);
+    if (buffer_length_after_insert > buffer_length_before_insert) {
+      const size_t packet_length_samples =
+          (buffer_length_after_insert - buffer_length_before_insert) *
+          decoder_frame_length_;
+      if (packet_length_samples != decision_logic_->packet_length_samples()) {
+        decision_logic_->set_packet_length_samples(packet_length_samples);
+        delay_manager_->SetPacketAudioLength(
+            rtc::checked_cast<int>((1000 * packet_length_samples) / fs_hz_));
+      }
     }
 
     // Update statistics.
@@ -682,8 +741,10 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   return 0;
 }
 
-int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
-                                int* samples_per_channel, int* num_channels) {
+int NetEqImpl::GetAudioInternal(size_t max_length,
+                                int16_t* output,
+                                size_t* samples_per_channel,
+                                size_t* num_channels) {
   PacketList packet_list;
   DtmfEvent dtmf_event;
   Operations operation;
@@ -691,13 +752,9 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
   int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
                                  &play_dtmf);
   if (return_value != 0) {
-    LOG_FERR1(LS_WARNING, GetDecision, return_value);
-    assert(false);
     last_mode_ = kModeError;
     return return_value;
   }
-  LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
-      " and " << packet_list.size() << " packet(s)";
 
   AudioDecoder::SpeechType speech_type;
   int length = 0;
@@ -707,7 +764,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
   assert(vad_.get());
   bool sid_frame_available =
       (operation == kRfc3389Cng && !packet_list.empty());
-  vad_->Update(decoded_buffer_.get(), length, speech_type,
+  vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
                sid_frame_available, fs_hz_);
 
   algorithm_buffer_->Clear();
@@ -724,9 +781,12 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
       return_value = DoExpand(play_dtmf);
       break;
     }
-    case kAccelerate: {
+    case kAccelerate:
+    case kFastAccelerate: {
+      const bool fast_accelerate =
+          enable_fast_accelerate_ && (operation == kFastAccelerate);
       return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
-                                  play_dtmf);
+                                  play_dtmf, fast_accelerate);
       break;
     }
     case kPreemptiveExpand: {
@@ -743,7 +803,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
       // This handles the case when there is no transmission and the decoder
       // should produce internal comfort noise.
       // TODO(hlundin): Write test for codec-internal CNG.
-      DoCodecInternalCng();
+      DoCodecInternalCng(decoded_buffer_.get(), length);
       break;
     }
     case kDtmf: {
@@ -763,7 +823,8 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
     }
     case kAudioRepetitionIncreaseTimestamp: {
       // TODO(hlundin): Write test for this.
-      sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
+      sync_buffer_->IncreaseEndTimestamp(
+          static_cast<uint32_t>(output_size_samples_));
       // Skipping break on purpose. Execution should move on into the
       // next case.
       FALLTHROUGH();
@@ -778,7 +839,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
       break;
     }
     case kUndefined: {
-      LOG_F(LS_ERROR) << "Invalid operation kUndefined.";
+      LOG(LS_ERROR) << "Invalid operation kUndefined.";
       assert(false);  // This should not happen.
       last_mode_ = kModeError;
       return kInvalidOperation;
@@ -802,18 +863,26 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
     LOG(LS_WARNING) << "Output array is too short. " << max_length << " < " <<
         output_size_samples_ << " * " << sync_buffer_->Channels();
     num_output_samples = max_length;
-    num_output_samples_per_channel = static_cast<int>(
-        max_length / sync_buffer_->Channels());
+    num_output_samples_per_channel = max_length / sync_buffer_->Channels();
   }
-  int samples_from_sync = static_cast<int>(
+  const size_t samples_from_sync =
       sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
-                                            output));
-  *num_channels = static_cast<int>(sync_buffer_->Channels());
-  LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
-      " insert " << algorithm_buffer_->Size() << " samples, extract " <<
-      samples_from_sync << " samples";
+                                            output);
+  *num_channels = sync_buffer_->Channels();
+  if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
+    // The sync buffer should always contain |overlap_length| samples, but now
+    // too many samples have been extracted. Reinstall the |overlap_length|
+    // lookahead by moving the index.
+    const size_t missing_lookahead_samples =
+        expand_->overlap_length() - sync_buffer_->FutureLength();
+    RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
+    sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                                 missing_lookahead_samples);
+  }
   if (samples_from_sync != output_size_samples_) {
-    LOG_F(LS_ERROR) << "samples_from_sync != output_size_samples_";
+    LOG(LS_ERROR) << "samples_from_sync (" << samples_from_sync
+                  << ") != output_size_samples_ (" << output_size_samples_
+                  << ")";
     // TODO(minyue): treatment of under-run, filling zeros
     memset(output, 0, num_output_samples * sizeof(int16_t));
     *samples_per_channel = output_size_samples_;
@@ -822,7 +891,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
   *samples_per_channel = output_size_samples_;
 
   // Should always have overlap samples left in the |sync_buffer_|.
-  assert(sync_buffer_->FutureLength() >= expand_->overlap_length());
+  RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
 
   if (play_dtmf) {
     return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), output);
@@ -856,7 +925,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
     }
   } else {
     // Use dead reckoning to estimate the |playout_timestamp_|.
-    playout_timestamp_ += output_size_samples_;
+    playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
   }
 
   if (decode_return_value) return decode_return_value;
@@ -911,13 +980,15 @@ int NetEqImpl::GetDecision(Operations* operation,
       last_mode_ == kModePreemptiveExpandSuccess ||
       last_mode_ == kModePreemptiveExpandLowEnergy) {
     // Subtract (samples_left + output_size_samples_) from sampleMemory.
-    decision_logic_->AddSampleMemory(-(samples_left + output_size_samples_));
+    decision_logic_->AddSampleMemory(
+        -(samples_left + rtc::checked_cast<int>(output_size_samples_)));
   }
 
   // Check if it is time to play a DTMF event.
-  if (dtmf_buffer_->GetEvent(end_timestamp +
-                             decision_logic_->generated_noise_samples(),
-                             dtmf_event)) {
+  if (dtmf_buffer_->GetEvent(
+      static_cast<uint32_t>(
+          end_timestamp + decision_logic_->generated_noise_samples()),
+      dtmf_event)) {
     *play_dtmf = true;
   }
 
@@ -935,9 +1006,10 @@ int NetEqImpl::GetDecision(Operations* operation,
   // Check if we already have enough samples in the |sync_buffer_|. If so,
   // change decision to normal, unless the decision was merge, accelerate, or
   // preemptive expand.
-  if (samples_left >= output_size_samples_ &&
+  if (samples_left >= rtc::checked_cast<int>(output_size_samples_) &&
       *operation != kMerge &&
       *operation != kAccelerate &&
+      *operation != kFastAccelerate &&
       *operation != kPreemptiveExpand) {
     *operation = kNormal;
     return 0;
@@ -952,9 +1024,8 @@ int NetEqImpl::GetDecision(Operations* operation,
     if (*play_dtmf && !header) {
       timestamp_ = dtmf_event->timestamp;
     } else {
-      assert(header);
       if (!header) {
-        LOG_F(LS_ERROR) << "Packet missing where it shouldn't.";
+        LOG(LS_ERROR) << "Packet missing where it shouldn't.";
         return -1;
       }
       timestamp_ = header->timestamp;
@@ -985,10 +1056,10 @@ int NetEqImpl::GetDecision(Operations* operation,
     stats_.ResetMcu();
   }
 
-  int required_samples = output_size_samples_;
-  const int samples_10_ms = 80 * fs_mult_;
-  const int samples_20_ms = 2 * samples_10_ms;
-  const int samples_30_ms = 3 * samples_10_ms;
+  size_t required_samples = output_size_samples_;
+  const size_t samples_10_ms = static_cast<size_t>(80 * fs_mult_);
+  const size_t samples_20_ms = 2 * samples_10_ms;
+  const size_t samples_30_ms = 3 * samples_10_ms;
 
   switch (*operation) {
     case kExpand: {
@@ -1006,26 +1077,28 @@ int NetEqImpl::GetDecision(Operations* operation,
       if (decision_logic_->generated_noise_samples() > 0 &&
           last_mode_ != kModeDtmf) {
         // Make a jump in timestamp due to the recently played comfort noise.
-        uint32_t timestamp_jump = decision_logic_->generated_noise_samples();
+        uint32_t timestamp_jump =
+            static_cast<uint32_t>(decision_logic_->generated_noise_samples());
         sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
         timestamp_ += timestamp_jump;
       }
       decision_logic_->set_generated_noise_samples(0);
       return 0;
     }
-    case kAccelerate: {
-      // In order to do a accelerate we need at least 30 ms of audio data.
-      if (samples_left >= samples_30_ms) {
+    case kAccelerate:
+    case kFastAccelerate: {
+      // In order to do an accelerate we need at least 30 ms of audio data.
+      if (samples_left >= static_cast<int>(samples_30_ms)) {
         // Already have enough data, so we do not need to extract any more.
         decision_logic_->set_sample_memory(samples_left);
         decision_logic_->set_prev_time_scale(true);
         return 0;
-      } else if (samples_left >= samples_10_ms &&
+      } else if (samples_left >= static_cast<int>(samples_10_ms) &&
           decoder_frame_length_ >= samples_30_ms) {
         // Avoid decoding more data as it might overflow the playout buffer.
         *operation = kNormal;
         return 0;
-      } else if (samples_left < samples_20_ms &&
+      } else if (samples_left < static_cast<int>(samples_20_ms) &&
           decoder_frame_length_ < samples_30_ms) {
         // Build up decoded data by decoding at least 20 ms of audio data. Do
         // not perform accelerate yet, but wait until we only need to do one
@@ -1043,8 +1116,8 @@ int NetEqImpl::GetDecision(Operations* operation,
     case kPreemptiveExpand: {
       // In order to do a preemptive expand we need at least 30 ms of decoded
       // audio data.
-      if ((samples_left >= samples_30_ms) ||
-          (samples_left >= samples_10_ms &&
+      if ((samples_left >= static_cast<int>(samples_30_ms)) ||
+          (samples_left >= static_cast<int>(samples_10_ms) &&
               decoder_frame_length_ >= samples_30_ms)) {
         // Already have enough data, so we do not need to extract any more.
         // Or, avoid decoding more data as it might overflow the playout buffer.
@@ -1053,7 +1126,7 @@ int NetEqImpl::GetDecision(Operations* operation,
         decision_logic_->set_prev_time_scale(true);
         return 0;
       }
-      if (samples_left < samples_20_ms &&
+      if (samples_left < static_cast<int>(samples_20_ms) &&
           decoder_frame_length_ < samples_30_ms) {
         // Build up decoded data by decoding at least 20 ms of audio data.
         // Still try to perform preemptive expand.
@@ -1098,20 +1171,19 @@ int NetEqImpl::GetDecision(Operations* operation,
 
     extracted_samples = ExtractPackets(required_samples, packet_list);
     if (extracted_samples < 0) {
-      LOG_F(LS_WARNING) << "Failed to extract packets from buffer.";
       return kPacketBufferCorruption;
     }
   }
 
-  if (*operation == kAccelerate ||
+  if (*operation == kAccelerate || *operation == kFastAccelerate ||
       *operation == kPreemptiveExpand) {
     decision_logic_->set_sample_memory(samples_left + extracted_samples);
     decision_logic_->set_prev_time_scale(true);
   }
 
-  if (*operation == kAccelerate) {
+  if (*operation == kAccelerate || *operation == kFastAccelerate) {
     // Check that we have enough data (30ms) to do accelerate.
-    if (extracted_samples + samples_left < samples_30_ms) {
+    if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) {
       // TODO(hlundin): Write test for this.
       // Not enough, do normal operation instead.
       *operation = kNormal;
@@ -1126,7 +1198,11 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
                       int* decoded_length,
                       AudioDecoder::SpeechType* speech_type) {
   *speech_type = AudioDecoder::kSpeech;
-  AudioDecoder* decoder = NULL;
+
+  // When packet_list is empty, we may be in kCodecInternalCng mode, and for
+  // that we use current active decoder.
+  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
+
   if (!packet_list->empty()) {
     const Packet* packet = packet_list->front();
     uint8_t payload_type = packet->header.payloadType;
@@ -1134,7 +1210,8 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
       decoder = decoder_database_->GetDecoder(payload_type);
       assert(decoder);
       if (!decoder) {
-        LOG_FERR1(LS_WARNING, GetDecoder, static_cast<int>(payload_type));
+        LOG(LS_WARNING) << "Unknown payload type "
+                        << static_cast<int>(payload_type);
         PacketBuffer::DeleteAllPackets(packet_list);
         return kDecoderNotFound;
       }
@@ -1146,7 +1223,8 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
             ->GetDecoderInfo(payload_type);
         assert(decoder_info);
         if (!decoder_info) {
-          LOG_FERR1(LS_WARNING, GetDecoderInfo, static_cast<int>(payload_type));
+          LOG(LS_WARNING) << "Unknown payload type "
+                          << static_cast<int>(payload_type);
           PacketBuffer::DeleteAllPackets(packet_list);
           return kDecoderNotFound;
         }
@@ -1165,15 +1243,14 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
 
   if (reset_decoder_) {
     // TODO(hlundin): Write test for this.
-    // Reset decoder.
-    if (decoder) {
-      decoder->Init();
-    }
+    if (decoder)
+      decoder->Reset();
+
     // Reset comfort noise decoder.
     AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
-    if (cng_decoder) {
-      cng_decoder->Init();
-    }
+    if (cng_decoder)
+      cng_decoder->Reset();
+
     reset_decoder_ = false;
   }
 
@@ -1193,13 +1270,20 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
     decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]);
   }
 
-  int return_value = DecodeLoop(packet_list, operation, decoder,
-                                decoded_length, speech_type);
+  int return_value;
+  if (*operation == kCodecInternalCng) {
+    RTC_DCHECK(packet_list->empty());
+    return_value = DecodeCng(decoder, decoded_length, speech_type);
+  } else {
+    return_value = DecodeLoop(packet_list, *operation, decoder,
+                              decoded_length, speech_type);
+  }
 
   if (*decoded_length < 0) {
     // Error returned from the decoder.
     *decoded_length = 0;
-    sync_buffer_->IncreaseEndTimestamp(decoder_frame_length_);
+    sync_buffer_->IncreaseEndTimestamp(
+        static_cast<uint32_t>(decoder_frame_length_));
     int error_code = 0;
     if (decoder)
       error_code = decoder->ErrorCode();
@@ -1207,11 +1291,12 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
       // Got some error code from the decoder.
       decoder_error_code_ = error_code;
       return_value = kDecoderErrorCode;
+      LOG(LS_WARNING) << "Decoder returned error code: " << error_code;
     } else {
       // Decoder does not implement error codes. Return generic error.
       return_value = kOtherDecoderError;
+      LOG(LS_WARNING) << "Decoder error (no error code)";
     }
-    LOG_FERR2(LS_WARNING, DecodeLoop, error_code, packet_list->size());
     *operation = kExpand;  // Do expansion to get data instead.
   }
   if (*speech_type != AudioDecoder::kComfortNoise) {
@@ -1226,13 +1311,44 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
   return return_value;
 }
 
-int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
+int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
+                         AudioDecoder::SpeechType* speech_type) {
+  if (!decoder) {
+    // This happens when active decoder is not defined.
+    *decoded_length = -1;
+    return 0;
+  }
+
+  while (*decoded_length < rtc::checked_cast<int>(output_size_samples_)) {
+    const int length = decoder->Decode(
+            nullptr, 0, fs_hz_,
+            (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
+            &decoded_buffer_[*decoded_length], speech_type);
+    if (length > 0) {
+      *decoded_length += length;
+    } else {
+      // Error.
+      LOG(LS_WARNING) << "Failed to decode CNG";
+      *decoded_length = -1;
+      break;
+    }
+    if (*decoded_length > static_cast<int>(decoded_buffer_length_)) {
+      // Guard against overflow.
+      LOG(LS_WARNING) << "Decoded too much CNG.";
+      return kDecodedTooMuch;
+    }
+  }
+  return 0;
+}
+
+int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
                           AudioDecoder* decoder, int* decoded_length,
                           AudioDecoder::SpeechType* speech_type) {
   Packet* packet = NULL;
   if (!packet_list->empty()) {
     packet = packet_list->front();
   }
+
   // Do decoding.
   while (packet &&
       !decoder_database_->IsComfortNoise(packet->header.payloadType)) {
@@ -1241,41 +1357,25 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
     // number decoder channels.
     assert(sync_buffer_->Channels() == decoder->Channels());
     assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels());
-    assert(*operation == kNormal || *operation == kAccelerate ||
-           *operation == kMerge || *operation == kPreemptiveExpand);
+    assert(operation == kNormal || operation == kAccelerate ||
+           operation == kFastAccelerate || operation == kMerge ||
+           operation == kPreemptiveExpand);
     packet_list->pop_front();
     size_t payload_length = packet->payload_length;
-    int16_t decode_length;
+    int decode_length;
     if (packet->sync_packet) {
       // Decode to silence with the same frame size as the last decode.
-      LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
-          " ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       memset(&decoded_buffer_[*decoded_length], 0,
              decoder_frame_length_ * decoder->Channels() *
                  sizeof(decoded_buffer_[0]));
-      decode_length = decoder_frame_length_;
+      decode_length = rtc::checked_cast<int>(decoder_frame_length_);
     } else if (!packet->primary) {
       // This is a redundant payload; call the special decoder method.
-      LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
-          " ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       decode_length = decoder->DecodeRedundant(
           packet->payload, packet->payload_length, fs_hz_,
           (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
           &decoded_buffer_[*decoded_length], speech_type);
     } else {
-      LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       decode_length =
           decoder->Decode(
               packet->payload, packet->payload_length, fs_hz_,
@@ -1290,20 +1390,17 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
       *decoded_length += decode_length;
       // Update |decoder_frame_length_| with number of samples per channel.
       decoder_frame_length_ =
-          decode_length / static_cast<int>(decoder->Channels());
-      LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples ("
-                      << decoder->Channels() << " channel(s) -> "
-                      << decoder_frame_length_ << " samples per channel)";
+          static_cast<size_t>(decode_length) / decoder->Channels();
     } else if (decode_length < 0) {
       // Error.
-      LOG_FERR2(LS_WARNING, Decode, decode_length, payload_length);
+      LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length;
       *decoded_length = -1;
       PacketBuffer::DeleteAllPackets(packet_list);
       break;
     }
     if (*decoded_length > static_cast<int>(decoded_buffer_length_)) {
       // Guard against overflow.
-      LOG_F(LS_WARNING) << "Decoded too much.";
+      LOG(LS_WARNING) << "Decoded too much.";
       PacketBuffer::DeleteAllPackets(packet_list);
       return kDecodedTooMuch;
     }
@@ -1349,11 +1446,11 @@ void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
                         AudioDecoder::SpeechType speech_type, bool play_dtmf) {
   assert(mute_factor_array_.get());
   assert(merge_.get());
-  int new_length = merge_->Process(decoded_buffer, decoded_length,
-                                   mute_factor_array_.get(),
-                                   algorithm_buffer_.get());
-  int expand_length_correction = new_length -
-      static_cast<int>(decoded_length / algorithm_buffer_->Channels());
+  size_t new_length = merge_->Process(decoded_buffer, decoded_length,
+                                      mute_factor_array_.get(),
+                                      algorithm_buffer_.get());
+  size_t expand_length_correction = new_length -
+      decoded_length / algorithm_buffer_->Channels();
 
   // Update in-call and post-call statistics.
   if (expand_->MuteFactor(0) == 0) {
@@ -1377,10 +1474,10 @@ void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
 
 int NetEqImpl::DoExpand(bool play_dtmf) {
   while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
-      static_cast<size_t>(output_size_samples_)) {
+      output_size_samples_) {
     algorithm_buffer_->Clear();
     int return_value = expand_->Process(algorithm_buffer_.get());
-    int length = static_cast<int>(algorithm_buffer_->Size());
+    size_t length = algorithm_buffer_->Size();
 
     // Update in-call and post-call statistics.
     if (expand_->MuteFactor(0) == 0) {
@@ -1406,10 +1503,13 @@ int NetEqImpl::DoExpand(bool play_dtmf) {
   return 0;
 }
 
-int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
+int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
+                            size_t decoded_length,
                             AudioDecoder::SpeechType speech_type,
-                            bool play_dtmf) {
-  const size_t required_samples = 240 * fs_mult_;  // Must have 30 ms.
+                            bool play_dtmf,
+                            bool fast_accelerate) {
+  const size_t required_samples =
+      static_cast<size_t>(240 * fs_mult_);  // Must have 30 ms.
   size_t borrowed_samples_per_channel = 0;
   size_t num_channels = algorithm_buffer_->Channels();
   size_t decoded_length_per_channel = decoded_length / num_channels;
@@ -1425,10 +1525,10 @@ int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
     decoded_length = required_samples * num_channels;
   }
 
-  int16_t samples_removed;
-  Accelerate::ReturnCodes return_code = accelerate_->Process(
-      decoded_buffer, decoded_length, algorithm_buffer_.get(),
-      &samples_removed);
+  size_t samples_removed;
+  Accelerate::ReturnCodes return_code =
+      accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate,
+                           algorithm_buffer_.get(), &samples_removed);
   stats_.AcceleratedSamples(samples_removed);
   switch (return_code) {
     case Accelerate::kSuccess:
@@ -1482,20 +1582,20 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
                                   size_t decoded_length,
                                   AudioDecoder::SpeechType speech_type,
                                   bool play_dtmf) {
-  const size_t required_samples = 240 * fs_mult_;  // Must have 30 ms.
+  const size_t required_samples =
+      static_cast<size_t>(240 * fs_mult_);  // Must have 30 ms.
   size_t num_channels = algorithm_buffer_->Channels();
-  int borrowed_samples_per_channel = 0;
-  int old_borrowed_samples_per_channel = 0;
+  size_t borrowed_samples_per_channel = 0;
+  size_t old_borrowed_samples_per_channel = 0;
   size_t decoded_length_per_channel = decoded_length / num_channels;
   if (decoded_length_per_channel < required_samples) {
     // Must move data from the |sync_buffer_| in order to get 30 ms.
-    borrowed_samples_per_channel = static_cast<int>(required_samples -
-        decoded_length_per_channel);
+    borrowed_samples_per_channel =
+        required_samples - decoded_length_per_channel;
     // Calculate how many of these were already played out.
-    old_borrowed_samples_per_channel = static_cast<int>(
-        borrowed_samples_per_channel - sync_buffer_->FutureLength());
-    old_borrowed_samples_per_channel = std::max(
-        0, old_borrowed_samples_per_channel);
+    old_borrowed_samples_per_channel =
+        (borrowed_samples_per_channel > sync_buffer_->FutureLength()) ?
+        (borrowed_samples_per_channel - sync_buffer_->FutureLength()) : 0;
     memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
             decoded_buffer,
             sizeof(int16_t) * decoded_length);
@@ -1504,9 +1604,9 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
     decoded_length = required_samples * num_channels;
   }
 
-  int16_t samples_added;
+  size_t samples_added;
   PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process(
-      decoded_buffer, static_cast<int>(decoded_length),
+      decoded_buffer, decoded_length,
       old_borrowed_samples_per_channel,
       algorithm_buffer_.get(), &samples_added);
   stats_.PreemptiveExpandedSamples(samples_added);
@@ -1559,16 +1659,16 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
       // Clearly wrong, but will maintain bit-exactness with legacy.
       if (fs_hz_ == 8000) {
         packet->header.payloadType =
-            decoder_database_->GetRtpPayloadType(kDecoderCNGnb);
+            decoder_database_->GetRtpPayloadType(NetEqDecoder::kDecoderCNGnb);
       } else if (fs_hz_ == 16000) {
         packet->header.payloadType =
-            decoder_database_->GetRtpPayloadType(kDecoderCNGwb);
+            decoder_database_->GetRtpPayloadType(NetEqDecoder::kDecoderCNGwb);
       } else if (fs_hz_ == 32000) {
-        packet->header.payloadType =
-            decoder_database_->GetRtpPayloadType(kDecoderCNGswb32kHz);
+        packet->header.payloadType = decoder_database_->GetRtpPayloadType(
+            NetEqDecoder::kDecoderCNGswb32kHz);
       } else if (fs_hz_ == 48000) {
-        packet->header.payloadType =
-            decoder_database_->GetRtpPayloadType(kDecoderCNGswb48kHz);
+        packet->header.payloadType = decoder_database_->GetRtpPayloadType(
+            NetEqDecoder::kDecoderCNGswb48kHz);
       }
       assert(decoder_database_->IsComfortNoise(packet->header.payloadType));
 #else
@@ -1579,7 +1679,6 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
     // UpdateParameters() deletes |packet|.
     if (comfort_noise_->UpdateParameters(packet) ==
         ComfortNoise::kInternalError) {
-      LOG_FERR0(LS_WARNING, UpdateParameters);
       algorithm_buffer_->Zeros(output_size_samples_);
       return -comfort_noise_->internal_error_code();
     }
@@ -1592,31 +1691,20 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
     dtmf_tone_generator_->Reset();
   }
   if (cn_return == ComfortNoise::kInternalError) {
-    LOG_FERR1(LS_WARNING, comfort_noise_->Generate, cn_return);
     decoder_error_code_ = comfort_noise_->internal_error_code();
     return kComfortNoiseErrorCode;
   } else if (cn_return == ComfortNoise::kUnknownPayloadType) {
-    LOG_FERR1(LS_WARNING, comfort_noise_->Generate, cn_return);
     return kUnknownRtpPayloadType;
   }
   return 0;
 }
 
-void NetEqImpl::DoCodecInternalCng() {
-  int length = 0;
-  // TODO(hlundin): Will probably need a longer buffer for multi-channel.
-  int16_t decoded_buffer[kMaxFrameSize];
-  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
-  if (decoder) {
-    const uint8_t* dummy_payload = NULL;
-    AudioDecoder::SpeechType speech_type;
-    length = decoder->Decode(
-        dummy_payload, 0, fs_hz_, kMaxFrameSize * sizeof(int16_t),
-        decoded_buffer, &speech_type);
-  }
-  assert(mute_factor_array_.get());
-  normal_->Process(decoded_buffer, length, last_mode_, mute_factor_array_.get(),
-                   algorithm_buffer_.get());
+void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer,
+                                   size_t decoded_length) {
+  RTC_DCHECK(normal_.get());
+  RTC_DCHECK(mute_factor_array_.get());
+  normal_->Process(decoded_buffer, decoded_length, last_mode_,
+                   mute_factor_array_.get(), algorithm_buffer_.get());
   last_mode_ = kModeCodecInternalCng;
   expand_->Reset();
 }
@@ -1691,7 +1779,8 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
   //    algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
   //  }
 
-  sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
+  sync_buffer_->IncreaseEndTimestamp(
+      static_cast<uint32_t>(output_size_samples_));
   expand_->Reset();
   last_mode_ = kModeDtmf;
 
@@ -1702,17 +1791,14 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
 
 void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
   AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
-  int length;
+  size_t length;
   if (decoder && decoder->HasDecodePlc()) {
     // Use the decoder's packet-loss concealment.
     // TODO(hlundin): Will probably need a longer buffer for multi-channel.
     int16_t decoded_buffer[kMaxFrameSize];
     length = decoder->DecodePlc(1, decoded_buffer);
-    if (length > 0) {
+    if (length > 0)
       algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
-    } else {
-      length = 0;
-    }
   } else {
     // Do simple zero-stuffing.
     length = output_size_samples_;
@@ -1721,7 +1807,7 @@ void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
     stats_.AddZeros(length);
   }
   if (increase_timestamp) {
-    sync_buffer_->IncreaseEndTimestamp(length);
+    sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
   }
   expand_->Reset();
 }
@@ -1729,14 +1815,14 @@ void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
 int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels,
                            int16_t* output) const {
   size_t out_index = 0;
-  int overdub_length = output_size_samples_;  // Default value.
+  size_t overdub_length = output_size_samples_;  // Default value.
 
   if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) {
     // Special operation for transition from "DTMF only" to "DTMF overdub".
     out_index = std::min(
         sync_buffer_->dtmf_index() - sync_buffer_->next_index(),
-        static_cast<size_t>(output_size_samples_));
-    overdub_length = output_size_samples_ - static_cast<int>(out_index);
+        output_size_samples_);
+    overdub_length = output_size_samples_ - out_index;
   }
 
   AudioMultiVector dtmf_output(num_channels);
@@ -1748,13 +1834,14 @@ int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels,
   if (dtmf_return_value == 0) {
     dtmf_return_value = dtmf_tone_generator_->Generate(overdub_length,
                                                        &dtmf_output);
-    assert((size_t) overdub_length == dtmf_output.Size());
+    assert(overdub_length == dtmf_output.Size());
   }
   dtmf_output.ReadInterleaved(overdub_length, &output[out_index]);
   return dtmf_return_value < 0 ? dtmf_return_value : 0;
 }
 
-int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
+int NetEqImpl::ExtractPackets(size_t required_samples,
+                              PacketList* packet_list) {
   bool first_packet = true;
   uint8_t prev_payload_type = 0;
   uint32_t prev_timestamp = 0;
@@ -1764,6 +1851,7 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
   const RTPHeader* header = packet_buffer_->NextRtpHeader();
   assert(header);
   if (!header) {
+    LOG(LS_ERROR) << "Packet buffer unexpectedly empty.";
     return -1;
   }
   uint32_t first_timestamp = header->timestamp;
@@ -1772,13 +1860,12 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
   // Packet extraction loop.
   do {
     timestamp_ = header->timestamp;
-    int discard_count = 0;
+    size_t discard_count = 0;
     Packet* packet = packet_buffer_->GetNextPacket(&discard_count);
     // |header| may be invalid after the |packet_buffer_| operation.
     header = NULL;
     if (!packet) {
-      LOG_FERR1(LS_ERROR, GetNextPacket, discard_count) <<
-          "Should always be able to extract a packet here";
+      LOG(LS_ERROR) << "Should always be able to extract a packet here";
       assert(false);  // Should always be able to extract a packet here.
       return -1;
     }
@@ -1790,9 +1877,14 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
 
     if (first_packet) {
       first_packet = false;
-      decoded_packet_sequence_number_ = prev_sequence_number =
-          packet->header.sequenceNumber;
-      decoded_packet_timestamp_ = prev_timestamp = packet->header.timestamp;
+      if (nack_enabled_) {
+        RTC_DCHECK(nack_);
+        // TODO(henrik.lundin): Should we update this for all decoded packets?
+        nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber,
+                                       packet->header.timestamp);
+      }
+      prev_sequence_number = packet->header.sequenceNumber;
+      prev_timestamp = packet->header.timestamp;
       prev_payload_type = packet->header.payloadType;
     }
 
@@ -1802,7 +1894,7 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
         packet->header.payloadType);
     if (decoder) {
       if (packet->sync_packet) {
-        packet_duration = decoder_frame_length_;
+        packet_duration = rtc::checked_cast<int>(decoder_frame_length_);
       } else {
         if (packet->primary) {
           packet_duration = decoder->PacketDuration(packet->payload,
@@ -1814,15 +1906,14 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
         }
       }
     } else {
-      LOG_FERR1(LS_WARNING, GetDecoder,
-                static_cast<int>(packet->header.payloadType))
-          << "Could not find a decoder for a packet about to be extracted.";
+      LOG(LS_WARNING) << "Unknown payload type "
+                      << static_cast<int>(packet->header.payloadType);
       assert(false);
     }
     if (packet_duration <= 0) {
       // Decoder did not return a packet duration. Assume that the packet
       // contains the same number of samples as the previous one.
-      packet_duration = decoder_frame_length_;
+      packet_duration = rtc::checked_cast<int>(decoder_frame_length_);
     }
     extracted_samples = packet->header.timestamp - first_timestamp +
         packet_duration;
@@ -1832,7 +1923,7 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
     next_packet_available = false;
     if (header && prev_payload_type == header->payloadType) {
       int16_t seq_no_diff = header->sequenceNumber - prev_sequence_number;
-      int32_t ts_diff = header->timestamp - prev_timestamp;
+      size_t ts_diff = header->timestamp - prev_timestamp;
       if (seq_no_diff == 1 ||
           (seq_no_diff == 0 && ts_diff == decoder_frame_length_)) {
         // The next sequence number is available, or the next part of a packet
@@ -1841,7 +1932,8 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
       }
       prev_sequence_number = header->sequenceNumber;
     }
-  } while (extracted_samples < required_samples && next_packet_available);
+  } while (extracted_samples < rtc::checked_cast<int>(required_samples) &&
+           next_packet_available);
 
   if (extracted_samples > 0) {
     // Delete old packets only when we are going to decode something. Otherwise,
@@ -1858,19 +1950,19 @@ void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) {
   // Delete objects and create new ones.
   expand_.reset(expand_factory_->Create(background_noise_.get(),
                                         sync_buffer_.get(), &random_vector_,
-                                        fs_hz, channels));
+                                        &stats_, fs_hz, channels));
   merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
 }
 
 void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
-  LOG_API2(fs_hz, channels);
+  LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " " << channels;
   // TODO(hlundin): Change to an enumerator and skip assert.
   assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz ==  32000 || fs_hz == 48000);
   assert(channels > 0);
 
   fs_hz_ = fs_hz;
   fs_mult_ = fs_hz / 8000;
-  output_size_samples_ = kOutputSizeMs * 8 * fs_mult_;
+  output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
   decoder_frame_length_ = 3 * output_size_samples_;  // Initialize to 30ms.
 
   last_mode_ = kModeNormal;
@@ -1881,11 +1973,9 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
     mute_factor_array_[i] = 16384;  // 1.0 in Q14.
   }
 
-  // Reset comfort noise decoder, if there is one active.
   AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
-  if (cng_decoder) {
-    cng_decoder->Init();
-  }
+  if (cng_decoder)
+    cng_decoder->Reset();
 
   // Reinit post-decode VAD with new sample rate.
   assert(vad_.get());  // Cannot be NULL here.
@@ -1915,9 +2005,7 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
   accelerate_.reset(
       accelerate_factory_->Create(fs_hz, channels, *background_noise_));
   preemptive_expand_.reset(preemptive_expand_factory_->Create(
-      fs_hz, channels,
-      *background_noise_,
-      static_cast<int>(expand_->overlap_length())));
+      fs_hz, channels, *background_noise_, expand_->overlap_length()));
 
   // Delete ComfortNoise object and create a new one.
   comfort_noise_.reset(new ComfortNoise(fs_hz, decoder_database_.get(),
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.h
index ac4689bbc4..940deadd2f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -11,14 +11,14 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
 
-#include <vector>
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"  // Declare PacketList.
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
 #include "webrtc/modules/audio_coding/neteq/rtcp.h"
@@ -41,6 +41,7 @@ class DtmfBuffer;
 class DtmfToneGenerator;
 class Expand;
 class Merge;
+class Nack;
 class Normal;
 class PacketBuffer;
 class PayloadSplitter;
@@ -73,15 +74,14 @@ class NetEqImpl : public webrtc::NetEq {
             PreemptiveExpandFactory* preemptive_expand_factory,
             bool create_components = true);
 
-  virtual ~NetEqImpl();
+  ~NetEqImpl() override;
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* payload,
-                   size_t length_bytes,
+                   rtc::ArrayView<const uint8_t> payload,
                    uint32_t receive_timestamp) override;
 
   // Inserts a sync-packet into packet queue. Sync-packets are decoded to
@@ -106,22 +106,19 @@ class NetEqImpl : public webrtc::NetEq {
   // Returns kOK on success, or kFail in case of an error.
   int GetAudio(size_t max_length,
                int16_t* output_audio,
-               int* samples_per_channel,
-               int* num_channels,
+               size_t* samples_per_channel,
+               size_t* num_channels,
                NetEqOutputType* type) override;
 
-  // Associates |rtp_payload_type| with |codec| and stores the information in
-  // the codec database. Returns kOK on success, kFail on failure.
-  int RegisterPayloadType(enum NetEqDecoder codec,
+  int RegisterPayloadType(NetEqDecoder codec,
+                          const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
 
-  // Provides an externally created decoder object |decoder| to insert in the
-  // decoder database. The decoder implements a decoder of type |codec| and
-  // associates it with |rtp_payload_type|. Returns kOK on success, kFail on
-  // failure.
   int RegisterExternalDecoder(AudioDecoder* decoder,
-                              enum NetEqDecoder codec,
-                              uint8_t rtp_payload_type) override;
+                              NetEqDecoder codec,
+                              const std::string& codec_name,
+                              uint8_t rtp_payload_type,
+                              int sample_rate_hz) override;
 
   // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
   // -1 on failure.
@@ -133,11 +130,11 @@ class NetEqImpl : public webrtc::NetEq {
 
   int LeastRequiredDelayMs() const override;
 
-  int SetTargetDelay() override { return kNotImplemented; }
+  int SetTargetDelay() override;
 
-  int TargetDelay() override { return kNotImplemented; }
+  int TargetDelay() override;
 
-  int CurrentDelay() override { return kNotImplemented; }
+  int CurrentDelayMs() const override;
 
   // Sets the playout mode to |mode|.
   // Deprecated.
@@ -153,11 +150,6 @@ class NetEqImpl : public webrtc::NetEq {
   // after the call.
   int NetworkStatistics(NetEqNetworkStatistics* stats) override;
 
-  // Writes the last packet waiting times (in ms) to |waiting_times|. The number
-  // of values written is no more than 100, but may be smaller if the interface
-  // is polled again before 100 packets has arrived.
-  void WaitingTimes(std::vector<int>* waiting_times) override;
-
   // Writes the current RTCP statistics to |stats|. The statistics are reset
   // and a new report period is started with the call.
   void GetRtcpStatistics(RtcpStatistics* stats) override;
@@ -174,9 +166,11 @@ class NetEqImpl : public webrtc::NetEq {
 
   bool GetPlayoutTimestamp(uint32_t* timestamp) override;
 
-  int SetTargetNumberOfChannels() override { return kNotImplemented; }
+  int last_output_sample_rate_hz() const override;
 
-  int SetTargetSampleRate() override { return kNotImplemented; }
+  int SetTargetNumberOfChannels() override;
+
+  int SetTargetSampleRate() override;
 
   // Returns the error code for the last occurred error. If no error has
   // occurred, 0 is returned.
@@ -193,25 +187,26 @@ class NetEqImpl : public webrtc::NetEq {
   void PacketBufferStatistics(int* current_num_packets,
                               int* max_num_packets) const override;
 
-  // Get sequence number and timestamp of the latest RTP.
-  // This method is to facilitate NACK.
-  int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const override;
+  void EnableNack(size_t max_nack_list_size) override;
+
+  void DisableNack() override;
+
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
 
   // This accessor method is only intended for testing purposes.
   const SyncBuffer* sync_buffer_for_test() const;
 
  protected:
   static const int kOutputSizeMs = 10;
-  static const int kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
+  static const size_t kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
-  static const int kSyncBufferSize = 2 * kMaxFrameSize;
+  static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
 
   // Inserts a new packet into NetEq. This is used by the InsertPacket method
   // above. Returns 0 on success, otherwise an error code.
   // TODO(hlundin): Merge this with InsertPacket above?
   int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
-                           const uint8_t* payload,
-                           size_t length_bytes,
+                           rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp,
                            bool is_sync_packet)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
@@ -224,8 +219,9 @@ class NetEqImpl : public webrtc::NetEq {
   // Returns 0 on success, otherwise an error code.
   int GetAudioInternal(size_t max_length,
                        int16_t* output,
-                       int* samples_per_channel,
-                       int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+                       size_t* samples_per_channel,
+                       size_t* num_channels)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to
   // do is written to |operation|. Packets to decode are written to
@@ -249,9 +245,14 @@ class NetEqImpl : public webrtc::NetEq {
              AudioDecoder::SpeechType* speech_type)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
+  // Sub-method to Decode(). Performs codec internal CNG.
+  int DecodeCng(AudioDecoder* decoder, int* decoded_length,
+                AudioDecoder::SpeechType* speech_type)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
   // Sub-method to Decode(). Performs the actual decoding.
   int DecodeLoop(PacketList* packet_list,
-                 Operations* operation,
+                 const Operations& operation,
                  AudioDecoder* decoder,
                  int* decoded_length,
                  AudioDecoder::SpeechType* speech_type)
@@ -277,7 +278,8 @@ class NetEqImpl : public webrtc::NetEq {
   int DoAccelerate(int16_t* decoded_buffer,
                    size_t decoded_length,
                    AudioDecoder::SpeechType speech_type,
-                   bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+                   bool play_dtmf,
+                   bool fast_accelerate) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Sub-method which calls the PreemptiveExpand class to perform the
   // preemtive expand operation.
@@ -295,7 +297,8 @@ class NetEqImpl : public webrtc::NetEq {
 
   // Calls the audio decoder to generate codec-internal comfort noise when
   // no packet was received.
-  void DoCodecInternalCng() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Calls the DtmfToneGenerator class to generate DTMF tones.
   int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
@@ -316,7 +319,7 @@ class NetEqImpl : public webrtc::NetEq {
   // |required_samples| samples. The packets are inserted into |packet_list|.
   // Returns the number of samples that the packets in the list will produce, or
   // -1 in case of an error.
-  int ExtractPackets(int required_samples, PacketList* packet_list)
+  int ExtractPackets(size_t required_samples, PacketList* packet_list)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Resets various variables and objects to new values based on the sample rate
@@ -373,8 +376,9 @@ class NetEqImpl : public webrtc::NetEq {
   StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
   int fs_hz_ GUARDED_BY(crit_sect_);
   int fs_mult_ GUARDED_BY(crit_sect_);
-  int output_size_samples_ GUARDED_BY(crit_sect_);
-  int decoder_frame_length_ GUARDED_BY(crit_sect_);
+  int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
+  size_t output_size_samples_ GUARDED_BY(crit_sect_);
+  size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
   Modes last_mode_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
   size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
@@ -391,19 +395,12 @@ class NetEqImpl : public webrtc::NetEq {
   int decoder_error_code_ GUARDED_BY(crit_sect_);
   const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
   NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
-
-  // These values are used by NACK module to estimate time-to-play of
-  // a missing packet. Occasionally, NetEq might decide to decode more
-  // than one packet. Therefore, these values store sequence number and
-  // timestamp of the first packet pulled from the packet buffer. In
-  // such cases, these values do not exactly represent the sequence number
-  // or timestamp associated with a 10ms audio pulled from NetEq. NACK
-  // module is designed to compensate for this.
-  int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
-  uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
+  bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
+  rtc::scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
+  bool nack_enabled_ GUARDED_BY(crit_sect_);
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
+  RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 3823d96751..f734883635 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
@@ -238,10 +239,11 @@ TEST(NetEq, CreateAndDestroy) {
 TEST_F(NetEqImplTest, RegisterPayloadType) {
   CreateInstance();
   uint8_t rtp_payload_type = 0;
-  NetEqDecoder codec_type = kDecoderPCMu;
+  NetEqDecoder codec_type = NetEqDecoder::kDecoderPCMu;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_CALL(*mock_decoder_database_,
-              RegisterPayload(rtp_payload_type, codec_type));
-  neteq_->RegisterPayloadType(codec_type, rtp_payload_type);
+              RegisterPayload(rtp_payload_type, codec_type, kCodecName));
+  neteq_->RegisterPayloadType(codec_type, kCodecName, rtp_payload_type);
 }
 
 TEST_F(NetEqImplTest, RemovePayloadType) {
@@ -299,7 +301,7 @@ TEST_F(NetEqImplTest, InsertPacket) {
   EXPECT_CALL(*mock_decoder_database_, IsComfortNoise(kPayloadType))
       .WillRepeatedly(Return(false));  // This is not CNG.
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderPCMu;
+  info.codec_type = NetEqDecoder::kDecoderPCMu;
   EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(&info));
 
@@ -333,7 +335,8 @@ TEST_F(NetEqImplTest, InsertPacket) {
     // All expectations within this block must be called in this specific order.
     InSequence sequence;  // Dummy variable.
     // Expectations when the first packet is inserted.
-    EXPECT_CALL(*mock_delay_manager_, LastDecoderType(kDecoderPCMu))
+    EXPECT_CALL(*mock_delay_manager_,
+                LastDecoderType(NetEqDecoder::kDecoderPCMu))
         .Times(1);
     EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
         .Times(2)
@@ -342,7 +345,8 @@ TEST_F(NetEqImplTest, InsertPacket) {
         .Times(1);
     EXPECT_CALL(*mock_delay_manager_, ResetPacketIatCount()).Times(1);
     // Expectations when the second packet is inserted. Slightly different.
-    EXPECT_CALL(*mock_delay_manager_, LastDecoderType(kDecoderPCMu))
+    EXPECT_CALL(*mock_delay_manager_,
+                LastDecoderType(NetEqDecoder::kDecoderPCMu))
         .Times(1);
     EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
         .WillOnce(Return(0));
@@ -356,13 +360,12 @@ TEST_F(NetEqImplTest, InsertPacket) {
       .WillRepeatedly(Return(PayloadSplitter::kOK));
 
   // Insert first packet.
-  neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime);
+  neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime);
 
   // Insert second packet.
   rtp_header.header.timestamp += 160;
   rtp_header.header.sequenceNumber += 1;
-  neteq_->InsertPacket(rtp_header, payload, kPayloadLength,
-                       kFirstReceiveTime + 155);
+  neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155);
 }
 
 TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
@@ -380,14 +383,13 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
   rtp_header.header.timestamp = 0x12345678;
   rtp_header.header.ssrc = 0x87654321;
 
-  EXPECT_EQ(NetEq::kOK,
-            neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
+                            NetEqDecoder::kDecoderPCM16B, "", kPayloadType));
 
   // Insert packets. The buffer should not flush.
-  for (int i = 1; i <= config_.max_packets_in_buffer; ++i) {
+  for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(
-                  rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
     rtp_header.header.timestamp += kPayloadLengthSamples;
     rtp_header.header.sequenceNumber += 1;
     EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
@@ -396,9 +398,8 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
   // Insert one more packet and make sure the buffer got flushed. That is, it
   // should only hold one single packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
-  EXPECT_EQ(1, packet_buffer_->NumPacketsInBuffer());
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
   const RTPHeader* test_header = packet_buffer_->NextRtpHeader();
   EXPECT_EQ(rtp_header.header.timestamp, test_header->timestamp);
   EXPECT_EQ(rtp_header.header.sequenceNumber, test_header->sequenceNumber);
@@ -413,7 +414,8 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateHz = 8000;
-  const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000;  // 10 ms.
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
   const size_t kPayloadLengthBytes = kPayloadLengthSamples;
   uint8_t payload[kPayloadLengthBytes] = {0};
   WebRtcRTPHeader rtp_header;
@@ -430,12 +432,11 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
     CountingSamplesDecoder() : next_value_(1) {}
 
     // Produce as many samples as input bytes (|encoded_len|).
-    int Decode(const uint8_t* encoded,
-               size_t encoded_len,
-               int /* sample_rate_hz */,
-               size_t /* max_decoded_bytes */,
-               int16_t* decoded,
-               SpeechType* speech_type) override {
+    int DecodeInternal(const uint8_t* encoded,
+                       size_t encoded_len,
+                       int /* sample_rate_hz */,
+                       int16_t* decoded,
+                       SpeechType* speech_type) override {
       for (size_t i = 0; i < encoded_len; ++i) {
         decoded[i] = next_value_++;
       }
@@ -443,10 +444,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
       return encoded_len;
     }
 
-    virtual int Init() {
-      next_value_ = 1;
-      return 0;
-    }
+    void Reset() override { next_value_ = 1; }
 
     size_t Channels() const override { return 1; }
 
@@ -456,31 +454,31 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
     int16_t next_value_;
   } decoder_;
 
-  EXPECT_EQ(NetEq::kOK,
-            neteq_->RegisterExternalDecoder(
-                &decoder_, kDecoderPCM16B, kPayloadType));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &decoder_, NetEqDecoder::kDecoderPCM16B,
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Pull audio once.
-  const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
-  int samples_per_channel;
-  int num_channels;
+  size_t samples_per_channel;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Start with a simple check that the fake decoder is behaving as expected.
-  EXPECT_EQ(kPayloadLengthSamples, decoder_.next_value() - 1);
+  EXPECT_EQ(kPayloadLengthSamples,
+            static_cast<size_t>(decoder_.next_value() - 1));
 
   // The value of the last of the output samples is the same as the number of
   // samples played from the decoded packet. Thus, this number + the RTP
@@ -500,7 +498,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   // Check that the number of samples still to play from the sync buffer add
   // up with what was already played out.
   EXPECT_EQ(kPayloadLengthSamples - output[samples_per_channel - 1],
-            static_cast<int>(sync_buffer->FutureLength()));
+            sync_buffer->FutureLength());
 }
 
 TEST_F(NetEqImplTest, ReorderedPacket) {
@@ -510,7 +508,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateHz = 8000;
-  const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000;  // 10 ms.
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
   const size_t kPayloadLengthBytes = kPayloadLengthSamples;
   uint8_t payload[kPayloadLengthBytes] = {0};
   WebRtcRTPHeader rtp_header;
@@ -521,40 +520,39 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
 
   // Create a mock decoder object.
   MockAudioDecoder mock_decoder;
-  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
   int16_t dummy_output[kPayloadLengthSamples] = {0};
   // The below expectation will make the mock decoder write
   // |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
-  EXPECT_CALL(mock_decoder,
-              Decode(Pointee(0), kPayloadLengthBytes, kSampleRateHz, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
-  EXPECT_EQ(NetEq::kOK,
-            neteq_->RegisterExternalDecoder(
-                &mock_decoder, kDecoderPCM16B, kPayloadType));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &mock_decoder, NetEqDecoder::kDecoderPCM16B,
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Pull audio once.
-  const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
-  int samples_per_channel;
-  int num_channels;
+  size_t samples_per_channel;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Insert two more packets. The first one is out of order, and is already too
@@ -563,22 +561,20 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   rtp_header.header.timestamp -= kPayloadLengthSamples;
   payload[0] = 1;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   rtp_header.header.sequenceNumber += 2;
   rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
   payload[0] = 2;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Expect only the second packet to be decoded (the one with "2" as the first
   // payload byte).
-  EXPECT_CALL(mock_decoder,
-              Decode(Pointee(2), kPayloadLengthBytes, kSampleRateHz, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
   // Pull audio once.
@@ -587,7 +583,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Now check the packet buffer, and make sure it is empty, since the
@@ -606,7 +602,8 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateHz = 8000;
-  const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000;  // 10 ms.
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
   const size_t kPayloadLengthBytes = kPayloadLengthSamples;
   uint8_t payload[kPayloadLengthBytes] = {0};
   WebRtcRTPHeader rtp_header;
@@ -618,46 +615,44 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
   // Insert one packet. Note that we have not registered any payload type, so
   // this packet will be rejected.
   EXPECT_EQ(NetEq::kFail,
-            neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                 kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 
   // Pull audio once.
-  const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
-  int samples_per_channel;
-  int num_channels;
+  size_t samples_per_channel;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   ASSERT_LE(samples_per_channel, kMaxOutputSize);
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputPLC, type);
 
   // Register the payload type.
-  EXPECT_EQ(NetEq::kOK,
-            neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
+                            NetEqDecoder::kDecoderPCM16B, "", kPayloadType));
 
   // Insert 10 packets.
-  for (int i = 0; i < 10; ++i) {
+  for (size_t i = 0; i < 10; ++i) {
     rtp_header.header.sequenceNumber++;
     rtp_header.header.timestamp += kPayloadLengthSamples;
     EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
-                                   kReceiveTime));
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
     EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
   }
 
   // Pull audio repeatedly and make sure we get normal output, that is not PLC.
-  for (int i = 0; i < 3; ++i) {
+  for (size_t i = 0; i < 3; ++i) {
     EXPECT_EQ(NetEq::kOK,
               neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                                &num_channels, &type));
     ASSERT_LE(samples_per_channel, kMaxOutputSize);
     EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(kOutputNormal, type)
         << "NetEq did not decode the packets as expected.";
   }
@@ -672,8 +667,9 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateKhz = 48;
-  const int kPayloadLengthSamples = 20 * kSampleRateKhz;  // 20 ms.
-  const int kPayloadLengthBytes = 10;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(20 * kSampleRateKhz);  // 20 ms.
+  const size_t kPayloadLengthBytes = 10;
   uint8_t payload[kPayloadLengthBytes] = {0};
   int16_t dummy_output[kPayloadLengthSamples] = {0};
 
@@ -685,61 +681,60 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
 
   // Create a mock decoder object.
   MockAudioDecoder mock_decoder;
-  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
 
   // Pointee(x) verifies that first byte of the payload equals x, this makes it
   // possible to verify that the correct payload is fed to Decode().
-  EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes,
-                                   kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
+                                           kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_CALL(mock_decoder, Decode(Pointee(1), kPayloadLengthBytes,
-                                   kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(1), kPayloadLengthBytes,
+                                           kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kComfortNoise),
+                      SetArgPointee<4>(AudioDecoder::kComfortNoise),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_CALL(mock_decoder, Decode(IsNull(), 0, kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kComfortNoise),
+                      SetArgPointee<4>(AudioDecoder::kComfortNoise),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes,
-                                   kSampleRateKhz * 1000, _, _, _))
-      .WillOnce(DoAll(SetArrayArgument<4>(dummy_output,
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
+                                           kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
                                           dummy_output + kPayloadLengthSamples),
-                      SetArgPointee<5>(AudioDecoder::kSpeech),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
-  EXPECT_EQ(NetEq::kOK,
-            neteq_->RegisterExternalDecoder(
-                &mock_decoder, kDecoderOpus, kPayloadType));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &mock_decoder, NetEqDecoder::kDecoderOpus,
+                            "dummy name", kPayloadType, kSampleRateKhz * 1000));
 
   // Insert one packet (decoder will return speech).
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Insert second packet (decoder will return CNG).
   payload[0] = 1;
   rtp_header.header.sequenceNumber++;
   rtp_header.header.timestamp += kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
-  const int kMaxOutputSize = 10 * kSampleRateKhz;
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
   int16_t output[kMaxOutputSize];
-  int samples_per_channel;
-  int num_channels;
+  size_t samples_per_channel;
+  size_t num_channels;
   uint32_t timestamp;
   uint32_t last_timestamp;
   NetEqOutputType type;
@@ -762,9 +757,9 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
                              &num_channels, &type));
   EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&last_timestamp));
 
-  for (int i = 1; i < 6; ++i) {
+  for (size_t i = 1; i < 6; ++i) {
     ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(expected_type[i - 1], type);
     EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&timestamp));
     EXPECT_EQ(NetEq::kOK,
@@ -780,12 +775,11 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
   rtp_header.header.sequenceNumber += 2;
   rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
-  for (int i = 6; i < 8; ++i) {
+  for (size_t i = 6; i < 8; ++i) {
     ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(expected_type[i - 1], type);
     EXPECT_EQ(NetEq::kOK,
               neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
@@ -805,16 +799,17 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   UseNoMocks();
   CreateInstance();
   static const size_t kNetEqMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
+  static const size_t kChannels = 2;
 
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateHz = 8000;
-  const int kChannles = 1;
 
-  const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000;  // 10 ms.
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
   const size_t kPayloadLengthBytes = 1;
   uint8_t payload[kPayloadLengthBytes]= {0};
-  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
   WebRtcRTPHeader rtp_header;
   rtp_header.header.payloadType = kPayloadType;
   rtp_header.header.sequenceNumber = 0x1234;
@@ -823,13 +818,11 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
 
   class MockAudioDecoder : public AudioDecoder {
    public:
-    int Init() override {
-      return 0;
-    }
+    void Reset() override {}
     MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
     MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
                                      SpeechType*));
-    size_t Channels() const override { return 1; }
+    size_t Channels() const override { return kChannels; }
   } decoder_;
 
   const uint8_t kFirstPayloadValue = 1;
@@ -838,7 +831,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   EXPECT_CALL(decoder_, PacketDuration(Pointee(kFirstPayloadValue),
                                        kPayloadLengthBytes))
     .Times(AtLeast(1))
-    .WillRepeatedly(Return(kNetEqMaxFrameSize * kChannles + 1));
+    .WillRepeatedly(Return(kNetEqMaxFrameSize + 1));
 
   EXPECT_CALL(decoder_,
               DecodeInternal(Pointee(kFirstPayloadValue), _, _, _, _))
@@ -849,24 +842,25 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
                                        kSampleRateHz, _, _))
       .Times(1)
       .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
-                                          dummy_output + kPayloadLengthSamples),
+                                          dummy_output +
+                                          kPayloadLengthSamples * kChannels),
                       SetArgPointee<4>(AudioDecoder::kSpeech),
-                      Return(kPayloadLengthSamples)));
+                      Return(static_cast<int>(
+                          kPayloadLengthSamples * kChannels))));
 
   EXPECT_CALL(decoder_, PacketDuration(Pointee(kSecondPayloadValue),
                                        kPayloadLengthBytes))
     .Times(AtLeast(1))
-    .WillRepeatedly(Return(kNetEqMaxFrameSize * kChannles));
+    .WillRepeatedly(Return(kNetEqMaxFrameSize));
 
-  EXPECT_EQ(NetEq::kOK,
-            neteq_->RegisterExternalDecoder(
-                &decoder_, kDecoderPCM16B, kPayloadType));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &decoder_, NetEqDecoder::kDecoderPCM16B,
+                            "dummy name", kPayloadType, kSampleRateHz));
 
   // Insert one packet.
   payload[0] = kFirstPayloadValue;  // This will make Decode() fail.
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   // Insert another packet.
   payload[0] = kSecondPayloadValue;  // This will make Decode() successful.
@@ -875,27 +869,374 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   // the second packet get decoded.
   rtp_header.header.timestamp += 3 * kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
-  const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
+  const size_t kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels;
   int16_t output[kMaxOutputSize];
-  int samples_per_channel;
-  int num_channels;
+  size_t samples_per_channel;
+  size_t num_channels;
   NetEqOutputType type;
 
   EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(kMaxOutputSize, output,
                                            &samples_per_channel, &num_channels,
                                            &type));
   EXPECT_EQ(NetEq::kOtherDecoderError, neteq_->LastError());
-  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(kChannles, num_channels);
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
+  EXPECT_EQ(kChannels, num_channels);
 
   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(kMaxOutputSize, output,
                                          &samples_per_channel, &num_channels,
                                          &type));
-  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(kChannles, num_channels);
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
+  EXPECT_EQ(kChannels, num_channels);
 }
 
-}  // namespace webrtc
+// This test inserts packets until the buffer is flushed. After that, it asks
+// NetEq for the network statistics. The purpose of the test is to make sure
+// that even though the buffer size increment is negative (which it becomes when
+// the packet causing a flush is inserted), the packet length stored in the
+// decision logic remains valid.
+TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
+  UseNoMocks();
+  CreateInstance();
+
+  const size_t kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
+                            NetEqDecoder::kDecoderPCM16B, "", kPayloadType));
+
+  // Insert packets until the buffer flushes.
+  for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
+    EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+    EXPECT_EQ(NetEq::kOK,
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+    rtp_header.header.timestamp +=
+        rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
+    ++rtp_header.header.sequenceNumber;
+  }
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+
+  // Ask for network statistics. This should not crash.
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
+}
+
+TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
+      .WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(kPayloadLengthSamples));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // |kPayloadLengthSamples| - 5 zeros to the output array, and mark it as
+  // speech. That is, the decoded length is 5 samples shorter than the expected.
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+      .WillOnce(
+          DoAll(SetArrayArgument<3>(dummy_output,
+                                    dummy_output + kPayloadLengthSamples - 5),
+                SetArgPointee<4>(AudioDecoder::kSpeech),
+                Return(kPayloadLengthSamples - 5)));
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &mock_decoder, NetEqDecoder::kDecoderPCM16B,
+                            "dummy name", kPayloadType, kSampleRateHz));
+
+  // Insert one packet.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+
+  EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  int16_t output[kMaxOutputSize];
+  size_t samples_per_channel;
+  size_t num_channels;
+  NetEqOutputType type;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  ASSERT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  EXPECT_EQ(kOutputNormal, type);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test checks the behavior of NetEq when audio decoder fails.
+TEST_F(NetEqImplTest, DecodingError) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
+  const int kSampleRateHz = 8000;
+  const int kDecoderErrorCode = -97;  // Any negative number.
+
+  // We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
+  const size_t kFrameLengthSamples =
+      static_cast<size_t>(5 * kSampleRateHz / 1000);
+
+  const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+
+  uint8_t payload[kPayloadLengthBytes] = {0};
+
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
+      .WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(kFrameLengthSamples));
+  EXPECT_CALL(mock_decoder, ErrorCode())
+      .WillOnce(Return(kDecoderErrorCode));
+  EXPECT_CALL(mock_decoder, HasDecodePlc())
+      .WillOnce(Return(false));
+  int16_t dummy_output[kFrameLengthSamples] = {0};
+
+  {
+    InSequence sequence;  // Dummy variable.
+    // Mock decoder works normally the first time.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(3)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
+                  Return(kFrameLengthSamples)))
+        .RetiresOnSaturation();
+
+    // Then mock decoder fails. A common reason for failure can be buffer being
+    // too short
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .WillOnce(Return(-1))
+        .RetiresOnSaturation();
+
+    // Mock decoder finally returns to normal.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
+                  Return(kFrameLengthSamples)));
+  }
+
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &mock_decoder, NetEqDecoder::kDecoderPCM16B,
+                            "dummy name", kPayloadType, kSampleRateHz));
+
+  // Insert packets.
+  for (int i = 0; i < 6; ++i) {
+    rtp_header.header.sequenceNumber += 1;
+    rtp_header.header.timestamp += kFrameLengthSamples;
+    EXPECT_EQ(NetEq::kOK,
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+  }
+
+  // Pull audio.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  int16_t output[kMaxOutputSize];
+  size_t samples_per_channel;
+  size_t num_channels;
+  NetEqOutputType type;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  EXPECT_EQ(kOutputNormal, type);
+
+  // Pull audio again. Decoder fails.
+  EXPECT_EQ(NetEq::kFail,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
+  EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  // TODO(minyue): should NetEq better give kOutputPLC, since it is actually an
+  // expansion.
+  EXPECT_EQ(kOutputNormal, type);
+
+  // Pull audio again, should continue an expansion.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  EXPECT_EQ(kOutputPLC, type);
+
+  // Pull audio again, should behave normal.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  EXPECT_EQ(kOutputNormal, type);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test checks the behavior of NetEq when audio decoder fails during CNG.
+TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
+  const int kSampleRateHz = 8000;
+  const int kDecoderErrorCode = -97;  // Any negative number.
+
+  // We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
+  const size_t kFrameLengthSamples =
+      static_cast<size_t>(5 * kSampleRateHz / 1000);
+
+  const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+
+  uint8_t payload[kPayloadLengthBytes] = {0};
+
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
+      .WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(kFrameLengthSamples));
+  EXPECT_CALL(mock_decoder, ErrorCode())
+      .WillOnce(Return(kDecoderErrorCode));
+  int16_t dummy_output[kFrameLengthSamples] = {0};
+
+  {
+    InSequence sequence;  // Dummy variable.
+    // Mock decoder works normally the first 2 times.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                  Return(kFrameLengthSamples)))
+        .RetiresOnSaturation();
+
+    // Then mock decoder fails. A common reason for failure can be buffer being
+    // too short
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
+        .WillOnce(Return(-1))
+        .RetiresOnSaturation();
+
+    // Mock decoder finally returns to normal.
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                  Return(kFrameLengthSamples)));
+  }
+
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                            &mock_decoder, NetEqDecoder::kDecoderPCM16B,
+                            "dummy name", kPayloadType, kSampleRateHz));
+
+  // Insert 2 packets. This will make netEq into codec internal CNG mode.
+  for (int i = 0; i < 2; ++i) {
+    rtp_header.header.sequenceNumber += 1;
+    rtp_header.header.timestamp += kFrameLengthSamples;
+    EXPECT_EQ(NetEq::kOK,
+              neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+  }
+
+  // Pull audio.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  int16_t output[kMaxOutputSize];
+  size_t samples_per_channel;
+  size_t num_channels;
+  NetEqOutputType type;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  EXPECT_EQ(kOutputCNG, type);
+
+  // Pull audio again. Decoder fails.
+  EXPECT_EQ(NetEq::kFail,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
+  EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  // TODO(minyue): should NetEq better give kOutputPLC, since it is actually an
+  // expansion.
+  EXPECT_EQ(kOutputCNG, type);
+
+  // Pull audio again, should resume codec CNG.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  EXPECT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1u, num_channels);
+  EXPECT_EQ(kOutputCNG, type);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// Tests that the return value from last_output_sample_rate_hz() is equal to the
+// configured inital sample rate.
+TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
+  UseNoMocks();
+  config_.sample_rate_hz = 48000;
+  CreateInstance();
+  EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+}
+
+}// namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e1a0f69dfa..34ca9ea856 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -10,7 +10,6 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 
@@ -21,19 +20,17 @@ using ::testing::_;
 using ::testing::SetArgPointee;
 using ::testing::Return;
 
-
-class MockAudioDecoderOpus : public AudioDecoderOpus {
+class MockAudioDecoder final : public AudioDecoder {
  public:
   static const int kPacketDuration = 960;  // 48 kHz * 20 ms
 
-  explicit MockAudioDecoderOpus(int num_channels)
-      : AudioDecoderOpus(num_channels),
-        fec_enabled_(false) {
+  explicit MockAudioDecoder(size_t num_channels)
+      : num_channels_(num_channels), fec_enabled_(false) {
   }
-  virtual ~MockAudioDecoderOpus() { Die(); }
+  ~MockAudioDecoder() override { Die(); }
   MOCK_METHOD0(Die, void());
 
-  MOCK_METHOD0(Init, int());
+  MOCK_METHOD0(Reset, void());
 
   int PacketDuration(const uint8_t* encoded,
                      size_t encoded_len) const override {
@@ -49,6 +46,8 @@ class MockAudioDecoderOpus : public AudioDecoderOpus {
     return fec_enabled_;
   }
 
+  size_t Channels() const override { return num_channels_; }
+
   void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; }
 
   bool fec_enabled() const { return fec_enabled_; }
@@ -75,13 +74,14 @@ class MockAudioDecoderOpus : public AudioDecoderOpus {
   }
 
  private:
+  const size_t num_channels_;
   bool fec_enabled_;
 };
 
 class NetEqNetworkStatsTest : public NetEqExternalDecoderTest {
  public:
   static const int kPayloadSizeByte = 30;
-  static const int kFrameSizeMs = 20;  // frame size of Opus
+  static const int kFrameSizeMs = 20;
   static const int kMaxOutputSize = 960;  // 10 ms * 48 kHz * 2 channels.
 
 enum logic {
@@ -108,7 +108,7 @@ struct NetEqNetworkStatsCheck {
 };
 
   NetEqNetworkStatsTest(NetEqDecoder codec,
-                        MockAudioDecoderOpus* decoder)
+                        MockAudioDecoder* decoder)
       : NetEqExternalDecoderTest(codec, decoder),
         external_decoder_(decoder),
         samples_per_ms_(CodecSampleRateHz(codec) / 1000),
@@ -170,6 +170,9 @@ struct NetEqNetworkStatsCheck {
     CHECK_NETEQ_NETWORK_STATS(added_zero_samples);
 
 #undef CHECK_NETEQ_NETWORK_STATS
+
+    // Compare with CurrentDelay, which should be identical.
+    EXPECT_EQ(stats.current_buffer_size_ms, neteq()->CurrentDelayMs());
   }
 
   void RunTest(int num_loops, NetEqNetworkStatsCheck expects) {
@@ -188,8 +191,7 @@ struct NetEqNetworkStatsCheck {
                                                       frame_size_samples_,
                                                       &rtp_header_);
         if (!Lost(next_send_time)) {
-          InsertPacket(rtp_header_, payload_, kPayloadSizeByte,
-                       next_send_time);
+          InsertPacket(rtp_header_, payload_, next_send_time);
         }
       }
       GetOutputAudio(kMaxOutputSize, output_, &output_type);
@@ -224,7 +226,7 @@ struct NetEqNetworkStatsCheck {
     expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 1065;
     RunTest(50, expects);
 
-    // Next we enable Opus FEC.
+    // Next we enable FEC.
     external_decoder_->set_fec_enabled(true);
     // If FEC fills in the lost packets, no packet loss will be counted.
     expects.stats_ref.packet_loss_rate = 0;
@@ -258,7 +260,7 @@ struct NetEqNetworkStatsCheck {
   }
 
  private:
-  MockAudioDecoderOpus* external_decoder_;
+  MockAudioDecoder* external_decoder_;
   const int samples_per_ms_;
   const size_t frame_size_samples_;
   rtc::scoped_ptr<test::RtpGenerator> rtp_generator_;
@@ -269,26 +271,23 @@ struct NetEqNetworkStatsCheck {
   int16_t output_[kMaxOutputSize];
 };
 
-TEST(NetEqNetworkStatsTest, OpusDecodeFec) {
-  MockAudioDecoderOpus decoder(1);
-  EXPECT_CALL(decoder, Init());
-  NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
+TEST(NetEqNetworkStatsTest, DecodeFec) {
+  MockAudioDecoder decoder(1);
+  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
-TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
-  MockAudioDecoderOpus decoder(2);
-  EXPECT_CALL(decoder, Init());
-  NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
+TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
+  MockAudioDecoder decoder(2);
+  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
-  MockAudioDecoderOpus decoder(1);
-  EXPECT_CALL(decoder, Init());
-  NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
+  MockAudioDecoder decoder(1);
+  NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index ea88f24a17..d3f59ec668 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -16,19 +16,18 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
 struct TestParameters {
   int frame_size;
   int sample_rate;
-  int num_channels;
+  size_t num_channels;
 };
 
 // This is a parameterized test. The test parameters are supplied through a
@@ -43,7 +42,7 @@ struct TestParameters {
 class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
  protected:
   static const int kTimeStepMs = 10;
-  static const int kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
+  static const size_t kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
   static const uint8_t kPayloadTypeMono = 95;
   static const uint8_t kPayloadTypeMulti = 96;
 
@@ -52,7 +51,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
         sample_rate_hz_(GetParam().sample_rate),
         samples_per_ms_(sample_rate_hz_ / 1000),
         frame_size_ms_(GetParam().frame_size),
-        frame_size_samples_(frame_size_ms_ * samples_per_ms_),
+        frame_size_samples_(
+            static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
         output_size_samples_(10 * samples_per_ms_),
         rtp_generator_mono_(samples_per_ms_),
         rtp_generator_(samples_per_ms_),
@@ -90,35 +90,35 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
     NetEqDecoder multi_decoder;
     switch (sample_rate_hz_) {
       case 8000:
-        mono_decoder = kDecoderPCM16B;
+        mono_decoder = NetEqDecoder::kDecoderPCM16B;
         if (num_channels_ == 2) {
-          multi_decoder = kDecoderPCM16B_2ch;
+          multi_decoder = NetEqDecoder::kDecoderPCM16B_2ch;
         } else if (num_channels_ == 5) {
-          multi_decoder = kDecoderPCM16B_5ch;
+          multi_decoder = NetEqDecoder::kDecoderPCM16B_5ch;
         } else {
           FAIL() << "Only 2 and 5 channels supported for 8000 Hz.";
         }
         break;
       case 16000:
-        mono_decoder = kDecoderPCM16Bwb;
+        mono_decoder = NetEqDecoder::kDecoderPCM16Bwb;
         if (num_channels_ == 2) {
-          multi_decoder = kDecoderPCM16Bwb_2ch;
+          multi_decoder = NetEqDecoder::kDecoderPCM16Bwb_2ch;
         } else {
           FAIL() << "More than 2 channels is not supported for 16000 Hz.";
         }
         break;
       case 32000:
-        mono_decoder = kDecoderPCM16Bswb32kHz;
+        mono_decoder = NetEqDecoder::kDecoderPCM16Bswb32kHz;
         if (num_channels_ == 2) {
-          multi_decoder = kDecoderPCM16Bswb32kHz_2ch;
+          multi_decoder = NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch;
         } else {
           FAIL() << "More than 2 channels is not supported for 32000 Hz.";
         }
         break;
       case 48000:
-        mono_decoder = kDecoderPCM16Bswb48kHz;
+        mono_decoder = NetEqDecoder::kDecoderPCM16Bswb48kHz;
         if (num_channels_ == 2) {
-          multi_decoder = kDecoderPCM16Bswb48kHz_2ch;
+          multi_decoder = NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch;
         } else {
           FAIL() << "More than 2 channels is not supported for 48000 Hz.";
         }
@@ -126,11 +126,10 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       default:
         FAIL() << "We shouldn't get here.";
     }
+    ASSERT_EQ(NetEq::kOK, neteq_mono_->RegisterPayloadType(mono_decoder, "mono",
+                                                           kPayloadTypeMono));
     ASSERT_EQ(NetEq::kOK,
-              neteq_mono_->RegisterPayloadType(mono_decoder,
-                                               kPayloadTypeMono));
-    ASSERT_EQ(NetEq::kOK,
-              neteq_->RegisterPayloadType(multi_decoder,
+              neteq_->RegisterPayloadType(multi_decoder, "multi-channel",
                                           kPayloadTypeMulti));
   }
 
@@ -164,7 +163,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
 
   void VerifyOutput(size_t num_samples) {
     for (size_t i = 0; i < num_samples; ++i) {
-      for (int j = 0; j < num_channels_; ++j) {
+      for (size_t j = 0; j < num_channels_; ++j) {
         ASSERT_EQ(output_[i], output_multi_channel_[i * num_channels_ + j]) <<
             "Diff in sample " << i << ", channel " << j << ".";
       }
@@ -195,14 +194,16 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       while (time_now >= next_arrival_time) {
         // Insert packet in mono instance.
         ASSERT_EQ(NetEq::kOK,
-                  neteq_mono_->InsertPacket(rtp_header_mono_, encoded_,
-                                            payload_size_bytes_,
+                  neteq_mono_->InsertPacket(rtp_header_mono_,
+                                            rtc::ArrayView<const uint8_t>(
+                                                encoded_, payload_size_bytes_),
                                             next_arrival_time));
         // Insert packet in multi-channel instance.
-        ASSERT_EQ(NetEq::kOK,
-                  neteq_->InsertPacket(rtp_header_, encoded_multi_channel_,
-                                       multi_payload_size_bytes_,
-                                       next_arrival_time));
+        ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
+                                  rtp_header_, rtc::ArrayView<const uint8_t>(
+                                                   encoded_multi_channel_,
+                                                   multi_payload_size_bytes_),
+                                  next_arrival_time));
         // Get next input packets (mono and multi-channel).
         do {
           next_send_time = GetNewPackets();
@@ -212,13 +213,13 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
       }
       NetEqOutputType output_type;
       // Get audio from mono instance.
-      int samples_per_channel;
-      int num_channels;
+      size_t samples_per_channel;
+      size_t num_channels;
       EXPECT_EQ(NetEq::kOK,
                 neteq_mono_->GetAudio(kMaxBlockSize, output_,
                                       &samples_per_channel, &num_channels,
                                       &output_type));
-      EXPECT_EQ(1, num_channels);
+      EXPECT_EQ(1u, num_channels);
       EXPECT_EQ(output_size_samples_, samples_per_channel);
       // Get audio from multi-channel instance.
       ASSERT_EQ(NetEq::kOK,
@@ -238,12 +239,12 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
     }
   }
 
-  const int num_channels_;
+  const size_t num_channels_;
   const int sample_rate_hz_;
   const int samples_per_ms_;
   const int frame_size_ms_;
-  const int frame_size_samples_;
-  const int output_size_samples_;
+  const size_t frame_size_samples_;
+  const size_t output_size_samples_;
   NetEq* neteq_mono_;
   NetEq* neteq_;
   test::RtpGenerator rtp_generator_mono_;
@@ -256,8 +257,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
   int16_t* output_multi_channel_;
   WebRtcRTPHeader rtp_header_mono_;
   WebRtcRTPHeader rtp_header_;
-  int payload_size_bytes_;
-  int multi_payload_size_bytes_;
+  size_t payload_size_bytes_;
+  size_t multi_payload_size_bytes_;
   int last_send_time_;
   int last_arrival_time_;
   rtc::scoped_ptr<test::InputAudioFile> input_file_;
@@ -274,7 +275,12 @@ class NetEqStereoTestNoJitter : public NetEqStereoTest {
   }
 };
 
-TEST_P(NetEqStereoTestNoJitter, DISABLED_ON_ANDROID(RunTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_RunTest DISABLED_RunTest
+#else
+#define MAYBE_RunTest RunTest
+#endif
+TEST_P(NetEqStereoTestNoJitter, MAYBE_RunTest) {
   RunTest(8);
 }
 
@@ -299,7 +305,7 @@ class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
   double drift_factor;
 };
 
-TEST_P(NetEqStereoTestPositiveDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
   RunTest(100);
 }
 
@@ -312,7 +318,7 @@ class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
   }
 };
 
-TEST_P(NetEqStereoTestNegativeDrift, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
   RunTest(100);
 }
 
@@ -340,7 +346,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestDelays, DISABLED_ON_ANDROID(RunTest)) {
+TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
   RunTest(1000);
 }
 
@@ -359,7 +365,10 @@ class NetEqStereoTestLosses : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestLosses, DISABLED_ON_ANDROID(RunTest)) {
+// TODO(pbos): Enable on non-Android, this went failing while being accidentally
+// disabled on all platforms and not just Android.
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5387
+TEST_P(NetEqStereoTestLosses, DISABLED_RunTest) {
   RunTest(100);
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
index 4dd392cd82..f02d3deee9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
@@ -7,37 +7,71 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
+  'conditions': [
+    ['enable_protobuf==1', {
+      'targets': [
+        {
+          'target_name': 'rtc_event_log_source',
+          'type': 'static_library',
+          'dependencies': [
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_proto',
+          ],
+          'sources': [
+            'tools/rtc_event_log_source.h',
+            'tools/rtc_event_log_source.cc',
+          ],
+        },
+        {
+          'target_name': 'neteq_rtpplay',
+          'type': 'executable',
+          'dependencies': [
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            '<(webrtc_root)/test/test.gyp:test_support_main',
+            'rtc_event_log_source',
+            'neteq',
+            'neteq_unittest_tools',
+            'pcm16b',
+          ],
+          'sources': [
+            'tools/neteq_rtpplay.cc',
+          ],
+          'defines': [
+          ],
+        }, # neteq_rtpplay
+        {
+          'target_name': 'neteq_unittest_proto',
+          'type': 'static_library',
+          'sources': [
+            'neteq_unittest.proto',
+          ],
+          'variables': {
+            'proto_in_dir': '.',
+            # Workaround to protect against gyp's pathname relativization when
+            # this file is included by modules.gyp.
+            'proto_out_protected': 'webrtc/audio_coding/neteq',
+            'proto_out_dir': '<(proto_out_protected)',
+          },
+          'includes': ['../../../build/protoc.gypi',],
+        },
+      ],
+    }],
+  ],
   'targets': [
-    {
-      'target_name': 'neteq_rtpplay',
-      'type': 'executable',
-      'dependencies': [
-        'neteq',
-        'neteq_unittest_tools',
-        'PCM16B',
-        '<(webrtc_root)/test/test.gyp:test_support_main',
-        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-      ],
-      'sources': [
-        'tools/neteq_rtpplay.cc',
-      ],
-      'defines': [
-      ],
-    }, # neteq_rtpplay
-
     {
       'target_name': 'RTPencode',
       'type': 'executable',
       'dependencies': [
         # TODO(hlundin): Make RTPencode use ACM to encode files.
-        'neteq_test_tools',# Test helpers
-        'G711',
-        'G722',
-        'PCM16B',
-        'iLBC',
-        'iSAC',
-        'CNG',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+        'cng',
+        'g711',
+        'g722',
+        'ilbc',
+        'isac',
+        'neteq_test_tools',  # Test helpers
+        'pcm16b',
+        'webrtc_opus',
       ],
       'defines': [
         'CODEC_ILBC',
@@ -54,9 +88,10 @@
         'CODEC_CNGCODEC32',
         'CODEC_ATEVENT_DECODE',
         'CODEC_RED',
+        'CODEC_OPUS',
       ],
       'include_dirs': [
-        'interface',
+        'include',
         'test',
         '<(webrtc_root)',
       ],
@@ -84,10 +119,10 @@
       'target_name': 'rtp_analyze',
       'type': 'executable',
       'dependencies': [
-        'neteq_unittest_tools',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+        'neteq_unittest_tools',
       ],
       'sources': [
         'tools/rtp_analyze.cc',
@@ -98,8 +133,8 @@
       'target_name': 'RTPchange',
       'type': 'executable',
       'dependencies': [
-        'neteq_test_tools',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        'neteq_test_tools',
       ],
       'sources': [
        'test/RTPchange.cc',
@@ -110,8 +145,8 @@
       'target_name': 'RTPtimeshift',
       'type': 'executable',
       'dependencies': [
-       'neteq_test_tools',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        'neteq_test_tools',
       ],
       'sources': [
         'test/RTPtimeshift.cc',
@@ -134,8 +169,8 @@
       'target_name': 'rtp_to_text',
       'type': 'executable',
       'dependencies': [
-        'neteq_test_tools',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+        'neteq_test_tools',
       ],
       'sources': [
         'test/rtp_to_text.cc',
@@ -158,11 +193,11 @@
       'target_name': 'neteq_test_support',
       'type': 'static_library',
       'dependencies': [
-        'neteq',
-        'PCM16B',
-        'neteq_unittest_tools',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+        'neteq',
+        'neteq_unittest_tools',
+        'pcm16b',
       ],
       'sources': [
         'tools/neteq_external_decoder_test.cc',
@@ -178,10 +213,10 @@
       'target_name': 'neteq_speed_test',
       'type': 'executable',
       'dependencies': [
-        'neteq',
-        'neteq_test_support',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/test/test.gyp:test_support_main',
+        'neteq',
+        'neteq_test_support',
       ],
       'sources': [
         'test/neteq_speed_test.cc',
@@ -192,12 +227,12 @@
       'target_name': 'neteq_opus_quality_test',
       'type': 'executable',
       'dependencies': [
-        'neteq',
-        'neteq_test_support',
-        'webrtc_opus',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/test/test.gyp:test_support_main',
+        'neteq',
+        'neteq_test_support',
+        'webrtc_opus',
       ],
       'sources': [
         'test/neteq_opus_quality_test.cc',
@@ -208,18 +243,50 @@
       'target_name': 'neteq_isac_quality_test',
       'type': 'executable',
       'dependencies': [
-        'neteq',
-        'neteq_test_support',
-        'iSACFix',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/test/test.gyp:test_support_main',
+        'isac_fix',
+        'neteq',
+        'neteq_test_support',
       ],
       'sources': [
         'test/neteq_isac_quality_test.cc',
       ],
     },
 
+    {
+      'target_name': 'neteq_pcmu_quality_test',
+      'type': 'executable',
+      'dependencies': [
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+        '<(webrtc_root)/test/test.gyp:test_support_main',
+        'g711',
+        'neteq',
+        'neteq_test_support',
+      ],
+      'sources': [
+        'test/neteq_pcmu_quality_test.cc',
+      ],
+    },
+
+    {
+      'target_name': 'neteq_ilbc_quality_test',
+      'type': 'executable',
+      'dependencies': [
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+        '<(webrtc_root)/test/test.gyp:test_support_main',
+        'neteq',
+        'neteq_test_support',
+        'ilbc',
+      ],
+      'sources': [
+        'test/neteq_ilbc_quality_test.cc',
+      ],
+    },
+
     {
      'target_name': 'neteq_test_tools',
       # Collection of useful functions used in other tests.
@@ -229,18 +296,18 @@
         'neteq_dummy_rtp%': 0,
       },
       'dependencies': [
-        'G711',
-        'G722',
-        'PCM16B',
-        'iLBC',
-        'iSAC',
-        'CNG',
-        '<(webrtc_root)/common.gyp:webrtc_common',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        'cng',
+        'g711',
+        'g722',
+        'ilbc',
+        'isac',
+        'pcm16b',
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          'interface',
+          'include',
           'test',
           '<(webrtc_root)',
         ],
@@ -248,7 +315,7 @@
       'defines': [
       ],
       'include_dirs': [
-        'interface',
+        'include',
         'test',
         '<(webrtc_root)',
       ],
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index b3d6f25f89..8d52c615da 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -12,7 +12,7 @@
  * This file includes unit tests for NetEQ.
  */
 
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 #include <math.h>
 #include <stdlib.h>
@@ -28,29 +28,91 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "webrtc/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+#endif
+
 DEFINE_bool(gen_ref, false, "Generate reference files.");
 
-namespace webrtc {
+namespace {
 
-static bool IsAllZero(const int16_t* buf, int buf_length) {
+bool IsAllZero(const int16_t* buf, size_t buf_length) {
   bool all_zero = true;
-  for (int n = 0; n < buf_length && all_zero; ++n)
+  for (size_t n = 0; n < buf_length && all_zero; ++n)
     all_zero = buf[n] == 0;
   return all_zero;
 }
 
-static bool IsAllNonZero(const int16_t* buf, int buf_length) {
+bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
   bool all_non_zero = true;
-  for (int n = 0; n < buf_length && all_non_zero; ++n)
+  for (size_t n = 0; n < buf_length && all_non_zero; ++n)
     all_non_zero = buf[n] != 0;
   return all_non_zero;
 }
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
+             webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
+  stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
+  stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
+  stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
+  stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
+  stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
+  stats->set_expand_rate(stats_raw.expand_rate);
+  stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
+  stats->set_preemptive_rate(stats_raw.preemptive_rate);
+  stats->set_accelerate_rate(stats_raw.accelerate_rate);
+  stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+  stats->set_clockdrift_ppm(stats_raw.clockdrift_ppm);
+  stats->set_added_zero_samples(stats_raw.added_zero_samples);
+  stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
+  stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
+  stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
+  stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
+}
+
+void Convert(const webrtc::RtcpStatistics& stats_raw,
+             webrtc::neteq_unittest::RtcpStatistics* stats) {
+  stats->set_fraction_lost(stats_raw.fraction_lost);
+  stats->set_cumulative_lost(stats_raw.cumulative_lost);
+  stats->set_extended_max_sequence_number(
+      stats_raw.extended_max_sequence_number);
+  stats->set_jitter(stats_raw.jitter);
+}
+
+void WriteMessage(FILE* file, const std::string& message) {
+  int32_t size = message.length();
+  ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  if (size <= 0)
+    return;
+  ASSERT_EQ(static_cast<size_t>(size),
+            fwrite(message.data(), sizeof(char), size, file));
+}
+
+void ReadMessage(FILE* file, std::string* message) {
+  int32_t size;
+  ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file));
+  if (size <= 0)
+    return;
+  rtc::scoped_ptr<char[]> buffer(new char[size]);
+  ASSERT_EQ(static_cast<size_t>(size),
+            fread(buffer.get(), sizeof(char), size, file));
+  message->assign(buffer.get(), size);
+}
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+
+}  // namespace
+
+namespace webrtc {
+
 class RefFiles {
  public:
   RefFiles(const std::string& input_file, const std::string& output_file);
@@ -128,91 +190,84 @@ void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
   }
 }
 
-void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats) {
-  if (output_fp_) {
-    ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
-                         output_fp_));
-  }
+void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!output_fp_)
+    return;
+  neteq_unittest::NetEqNetworkStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  WriteMessage(output_fp_, stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
 void RefFiles::ReadFromFileAndCompare(
     const NetEqNetworkStatistics& stats) {
-  // TODO(minyue): Update resource/audio_coding/neteq_network_stats.dat and
-  // resource/audio_coding/neteq_network_stats_win32.dat.
-  struct NetEqNetworkStatisticsOld {
-    uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
-    uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
-    uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
-                                  // jitter; 0 otherwise.
-    uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
-    uint16_t packet_discard_rate;  // Late loss rate in Q14.
-    uint16_t expand_rate;  // Fraction (of original stream) of synthesized
-                           // audio inserted through expansion (in Q14).
-    uint16_t preemptive_rate;  // Fraction of data inserted through pre-emptive
-                               // expansion (in Q14).
-    uint16_t accelerate_rate;  // Fraction of data removed through acceleration
-                               // (in Q14).
-    int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
-                             // (positive or negative).
-    int added_zero_samples;  // Number of zero samples added in "off" mode.
-  };
-  if (input_fp_) {
-    // Read from ref file.
-    size_t stat_size = sizeof(NetEqNetworkStatisticsOld);
-    NetEqNetworkStatisticsOld ref_stats;
-    ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
-    // Compare
-    ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms);
-    ASSERT_EQ(stats.preferred_buffer_size_ms,
-              ref_stats.preferred_buffer_size_ms);
-    ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found);
-    ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate);
-    ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate);
-    ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate);
-    ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
-    ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
-    ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
-    ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples);
-    ASSERT_EQ(stats.secondary_decoded_rate, 0);
-    ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
-  }
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!input_fp_)
+    return;
+
+  std::string stats_string;
+  ReadMessage(input_fp_, &stats_string);
+  neteq_unittest::NetEqNetworkStatistics ref_stats;
+  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
+
+  // Compare
+  ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms());
+  ASSERT_EQ(stats.preferred_buffer_size_ms,
+            ref_stats.preferred_buffer_size_ms());
+  ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found());
+  ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate());
+  ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate());
+  ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate());
+  ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate());
+  ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate());
+  ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm());
+  ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples());
+  ASSERT_EQ(stats.secondary_decoded_rate, ref_stats.secondary_decoded_rate());
+  ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate());
+#else
+  FAIL() << "Reading from reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
-void RefFiles::WriteToFile(const RtcpStatistics& stats) {
-  if (output_fp_) {
-    ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
-                         output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
-                         sizeof(stats.cumulative_lost), 1, output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
-                         sizeof(stats.extended_max_sequence_number), 1,
-                         output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
-                         output_fp_));
-  }
+void RefFiles::WriteToFile(const RtcpStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!output_fp_)
+    return;
+  neteq_unittest::RtcpStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  WriteMessage(output_fp_, stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
-void RefFiles::ReadFromFileAndCompare(
-    const RtcpStatistics& stats) {
-  if (input_fp_) {
-    // Read from ref file.
-    RtcpStatistics ref_stats;
-    ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
-                        sizeof(ref_stats.fraction_lost), 1, input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
-                        sizeof(ref_stats.cumulative_lost), 1, input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
-                        sizeof(ref_stats.extended_max_sequence_number), 1,
-                        input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
-                        input_fp_));
-    // Compare
-    ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
-    ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
-    ASSERT_EQ(ref_stats.extended_max_sequence_number,
-              stats.extended_max_sequence_number);
-    ASSERT_EQ(ref_stats.jitter, stats.jitter);
-  }
+void RefFiles::ReadFromFileAndCompare(const RtcpStatistics& stats) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  if (!input_fp_)
+    return;
+  std::string stats_string;
+  ReadMessage(input_fp_, &stats_string);
+  neteq_unittest::RtcpStatistics ref_stats;
+  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
+
+  // Compare
+  ASSERT_EQ(stats.fraction_lost, ref_stats.fraction_lost());
+  ASSERT_EQ(stats.cumulative_lost, ref_stats.cumulative_lost());
+  ASSERT_EQ(stats.extended_max_sequence_number,
+            ref_stats.extended_max_sequence_number());
+  ASSERT_EQ(stats.jitter, ref_stats.jitter());
+#else
+  FAIL() << "Reading from reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
 class NetEqDecodingTest : public ::testing::Test {
@@ -220,10 +275,11 @@ class NetEqDecodingTest : public ::testing::Test {
   // NetEQ must be polled for data once every 10 ms. Thus, neither of the
   // constants below can be changed.
   static const int kTimeStepMs = 10;
-  static const int kBlockSize8kHz = kTimeStepMs * 8;
-  static const int kBlockSize16kHz = kTimeStepMs * 16;
-  static const int kBlockSize32kHz = kTimeStepMs * 32;
-  static const size_t kMaxBlockSize = kBlockSize32kHz;
+  static const size_t kBlockSize8kHz = kTimeStepMs * 8;
+  static const size_t kBlockSize16kHz = kTimeStepMs * 16;
+  static const size_t kBlockSize32kHz = kTimeStepMs * 32;
+  static const size_t kBlockSize48kHz = kTimeStepMs * 48;
+  static const size_t kMaxBlockSize = kBlockSize48kHz;
   static const int kInitSampleRateHz = 8000;
 
   NetEqDecodingTest();
@@ -232,11 +288,13 @@ class NetEqDecodingTest : public ::testing::Test {
   void SelectDecoders(NetEqDecoder* used_codec);
   void LoadDecoders();
   void OpenInputFile(const std::string &rtp_file);
-  void Process(int* out_len);
+  void Process(size_t* out_len);
+
   void DecodeAndCompare(const std::string& rtp_file,
                         const std::string& ref_file,
                         const std::string& stat_ref_file,
                         const std::string& rtcp_ref_file);
+
   static void PopulateRtpInfo(int frame_index,
                               int timestamp,
                               WebRtcRTPHeader* rtp_info);
@@ -272,9 +330,9 @@ class NetEqDecodingTest : public ::testing::Test {
 
 // Allocating the static const so that it can be passed by reference.
 const int NetEqDecodingTest::kTimeStepMs;
-const int NetEqDecodingTest::kBlockSize8kHz;
-const int NetEqDecodingTest::kBlockSize16kHz;
-const int NetEqDecodingTest::kBlockSize32kHz;
+const size_t NetEqDecodingTest::kBlockSize8kHz;
+const size_t NetEqDecodingTest::kBlockSize16kHz;
+const size_t NetEqDecodingTest::kBlockSize32kHz;
 const size_t NetEqDecodingTest::kMaxBlockSize;
 const int NetEqDecodingTest::kInitSampleRateHz;
 
@@ -303,47 +361,63 @@ void NetEqDecodingTest::TearDown() {
 
 void NetEqDecodingTest::LoadDecoders() {
   // Load PCMu.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMu, "pcmu", 0));
   // Load PCMa.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
-#ifndef WEBRTC_ANDROID
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMa, "pcma", 8));
+#ifdef WEBRTC_CODEC_ILBC
   // Load iLBC.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
-#endif  // WEBRTC_ANDROID
+  ASSERT_EQ(
+      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderILBC, "ilbc", 102));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
   // Load iSAC.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
-#ifndef WEBRTC_ANDROID
+  ASSERT_EQ(
+      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac", 103));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
   // Load iSAC SWB.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
-  // Load iSAC FB.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
-#endif  // WEBRTC_ANDROID
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISACswb,
+                                           "isac-swb", 104));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderOpus,
+                                           "opus", 111));
+#endif
   // Load PCM16B nb.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16B, 93));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16B,
+                                           "pcm16-nb", 93));
   // Load PCM16B wb.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb, 94));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
+                                           "pcm16-wb", 94));
   // Load PCM16B swb32.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bswb32kHz, 95));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                                           "pcm16-swb32", 95));
   // Load CNG 8 kHz.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, 13));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
+                                           "cng-nb", 13));
   // Load CNG 16 kHz.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, 98));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
+                                           "cng-wb", 98));
 }
 
 void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
   rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
 }
 
-void NetEqDecodingTest::Process(int* out_len) {
+void NetEqDecodingTest::Process(size_t* out_len) {
   // Check if time to receive.
   while (packet_ && sim_clock_ >= packet_->time_ms()) {
     if (packet_->payload_length_bytes() > 0) {
       WebRtcRTPHeader rtp_header;
       packet_->ConvertHeader(&rtp_header);
       ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_header, packet_->payload(),
-                       packet_->payload_length_bytes(),
-                       packet_->time_ms() * (output_sample_rate_ / 1000)));
+                       rtp_header,
+                       rtc::ArrayView<const uint8_t>(
+                           packet_->payload(), packet_->payload_length_bytes()),
+                       static_cast<uint32_t>(packet_->time_ms() *
+                                             (output_sample_rate_ / 1000))));
     }
     // Get next packet.
     packet_.reset(rtp_source_->NextPacket());
@@ -351,13 +425,15 @@ void NetEqDecodingTest::Process(int* out_len) {
 
   // Get audio from NetEq.
   NetEqOutputType type;
-  int num_channels;
+  size_t num_channels;
   ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
                                 &num_channels, &type));
   ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
               (*out_len == kBlockSize16kHz) ||
-              (*out_len == kBlockSize32kHz));
-  output_sample_rate_ = *out_len / 10 * 1000;
+              (*out_len == kBlockSize32kHz) ||
+              (*out_len == kBlockSize48kHz));
+  output_sample_rate_ = static_cast<int>(*out_len / 10 * 1000);
+  EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
 
   // Increase time.
   sim_clock_ += kTimeStepMs;
@@ -393,7 +469,7 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
     std::ostringstream ss;
     ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
-    int out_len = 0;
+    size_t out_len = 0;
     ASSERT_NO_FATAL_FAILURE(Process(&out_len));
     ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len));
 
@@ -404,6 +480,8 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
       ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
       ASSERT_NO_FATAL_FAILURE(
           network_stat_files.ProcessReference(network_stats));
+      // Compare with CurrentDelay, which should be identical.
+      EXPECT_EQ(network_stats.current_buffer_size_ms, neteq_->CurrentDelayMs());
 
       // Process RTCPstat.
       RtcpStatistics rtcp_stats;
@@ -437,9 +515,17 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) {
-  const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
-      "resources/audio_coding/neteq_universal_new.rtp";
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
+    (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#define MAYBE_TestBitExactness TestBitExactness
+#else
+#define MAYBE_TestBitExactness DISABLED_TestBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
@@ -467,6 +553,34 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) {
   }
 }
 
+#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
+    defined(WEBRTC_CODEC_OPUS)
+#define MAYBE_TestOpusBitExactness TestOpusBitExactness
+#else
+#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
+  const std::string input_ref_file =
+      webrtc::test::ResourcePath("audio_coding/neteq4_opus_ref", "pcm");
+  const std::string network_stat_ref_file =
+      webrtc::test::ResourcePath("audio_coding/neteq4_opus_network_stats",
+                                 "dat");
+  const std::string rtcp_stat_ref_file =
+      webrtc::test::ResourcePath("audio_coding/neteq4_opus_rtcp_stats", "dat");
+
+  if (FLAGS_gen_ref) {
+    DecodeAndCompare(input_rtp_file, "", "", "");
+  } else {
+    DecodeAndCompare(input_rtp_file,
+                     input_ref_file,
+                     network_stat_ref_file,
+                     rtcp_stat_ref_file);
+  }
+}
+
 // Use fax mode to avoid time-scaling. This is to simplify the testing of
 // packet waiting times in the packet buffer.
 class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
@@ -482,68 +596,42 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   for (size_t i = 0; i < num_frames; ++i) {
-    uint16_t payload[kSamples] = {0};
+    const uint8_t payload[kPayloadBytes] = {0};
     WebRtcRTPHeader rtp_info;
     rtp_info.header.sequenceNumber = i;
     rtp_info.header.timestamp = i * kSamples;
     rtp_info.header.ssrc = 0x1234;  // Just an arbitrary SSRC.
     rtp_info.header.payloadType = 94;  // PCM16b WB codec.
     rtp_info.header.markerBit = 0;
-    ASSERT_EQ(0, neteq_->InsertPacket(
-        rtp_info,
-        reinterpret_cast<uint8_t*>(payload),
-        kPayloadBytes, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   }
   // Pull out all data.
   for (size_t i = 0; i < num_frames; ++i) {
-    int out_len;
-    int num_channels;
+    size_t out_len;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
 
-  std::vector<int> waiting_times;
-  neteq_->WaitingTimes(&waiting_times);
-  EXPECT_EQ(num_frames, waiting_times.size());
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
   // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
   // spacing (per definition), we expect the delay to increase with 10 ms for
-  // each packet.
-  for (size_t i = 0; i < waiting_times.size(); ++i) {
-    EXPECT_EQ(static_cast<int>(i + 1) * 10, waiting_times[i]);
-  }
+  // each packet. Thus, we are calculating the statistics for a series from 10
+  // to 300, in steps of 10 ms.
+  EXPECT_EQ(155, stats.mean_waiting_time_ms);
+  EXPECT_EQ(155, stats.median_waiting_time_ms);
+  EXPECT_EQ(10, stats.min_waiting_time_ms);
+  EXPECT_EQ(300, stats.max_waiting_time_ms);
 
   // Check statistics again and make sure it's been reset.
-  neteq_->WaitingTimes(&waiting_times);
-  int len = waiting_times.size();
-  EXPECT_EQ(0, len);
-
-  // Process > 100 frames, and make sure that that we get statistics
-  // only for 100 frames. Note the new SSRC, causing NetEQ to reset.
-  num_frames = 110;
-  for (size_t i = 0; i < num_frames; ++i) {
-    uint16_t payload[kSamples] = {0};
-    WebRtcRTPHeader rtp_info;
-    rtp_info.header.sequenceNumber = i;
-    rtp_info.header.timestamp = i * kSamples;
-    rtp_info.header.ssrc = 0x1235;  // Just an arbitrary SSRC.
-    rtp_info.header.payloadType = 94;  // PCM16b WB codec.
-    rtp_info.header.markerBit = 0;
-    ASSERT_EQ(0, neteq_->InsertPacket(
-        rtp_info,
-        reinterpret_cast<uint8_t*>(payload),
-        kPayloadBytes, 0));
-    int out_len;
-    int num_channels;
-    NetEqOutputType type;
-    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
-                                  &num_channels, &type));
-    ASSERT_EQ(kBlockSize16kHz, out_len);
-  }
-
-  neteq_->WaitingTimes(&waiting_times);
-  EXPECT_EQ(100u, waiting_times.size());
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  EXPECT_EQ(-1, stats.mean_waiting_time_ms);
+  EXPECT_EQ(-1, stats.median_waiting_time_ms);
+  EXPECT_EQ(-1, stats.min_waiting_time_ms);
+  EXPECT_EQ(-1, stats.max_waiting_time_ms);
 }
 
 TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
@@ -559,13 +647,13 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
-    int out_len;
-    int num_channels;
+    size_t out_len;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
@@ -590,13 +678,13 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
-    int out_len;
-    int num_channels;
+    size_t out_len;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
@@ -620,8 +708,8 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
   const size_t kPayloadBytes = kSamples * 2;
   double next_input_time_ms = 0.0;
   double t_ms;
-  int out_len;
-  int num_channels;
+  size_t out_len;
+  size_t num_channels;
   NetEqOutputType type;
 
   // Insert speech for 5 seconds.
@@ -633,7 +721,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
@@ -659,7 +747,9 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(
+                       rtp_info,
+                       rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
@@ -706,7 +796,9 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(
+                       rtp_info,
+                       rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += kCngPeriodMs * drift_factor;
@@ -722,7 +814,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += kFrameSizeMs * drift_factor;
@@ -833,26 +925,31 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 1;  // Not registered as a decoder.
-  EXPECT_EQ(NetEq::kFail,
-            neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_DecoderError DISABLED_DecoderError
+#else
+#define MAYBE_DecoderError DecoderError
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
     out_data_[i] = 1;
   }
-  int num_channels;
-  int samples_per_channel;
+  size_t num_channels;
+  size_t samples_per_channel;
   EXPECT_EQ(NetEq::kFail,
             neteq_->GetAudio(kMaxBlockSize, out_data_,
                              &samples_per_channel, &num_channels, &type));
@@ -876,6 +973,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
     EXPECT_EQ(1, out_data_[i]);
   }
 }
+#endif
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   NetEqOutputType type;
@@ -884,8 +982,8 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
     out_data_[i] = 1;
   }
-  int num_channels;
-  int samples_per_channel;
+  size_t num_channels;
+  size_t samples_per_channel;
   EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
                                 &samples_per_channel,
                                 &num_channels, &type));
@@ -898,6 +996,8 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(0, out_data_[i]);
   }
+  // Verify that the sample rate did not change from the initial configuration.
+  EXPECT_EQ(config_.sample_rate_hz, neteq_->last_output_sample_rate_hz());
 }
 
 class NetEqBgnTest : public NetEqDecodingTest {
@@ -906,7 +1006,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
                              bool should_be_faded) = 0;
 
   void CheckBgn(int sampling_rate_hz) {
-    int16_t expected_samples_per_channel = 0;
+    size_t expected_samples_per_channel = 0;
     uint8_t payload_type = 0xFF;  // Invalid.
     if (sampling_rate_hz == 8000) {
       expected_samples_per_channel = kBlockSize8kHz;
@@ -930,7 +1030,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
     ASSERT_TRUE(input.Init(
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
         10 * sampling_rate_hz,  // Max 10 seconds loop length.
-        static_cast<size_t>(expected_samples_per_channel)));
+        expected_samples_per_channel));
 
     // Payload of 10 ms of PCM16 32 kHz.
     uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
@@ -938,28 +1038,29 @@ class NetEqBgnTest : public NetEqDecodingTest {
     PopulateRtpInfo(0, 0, &rtp_info);
     rtp_info.header.payloadType = payload_type;
 
-    int number_channels = 0;
-    int samples_per_channel = 0;
+    size_t number_channels = 0;
+    size_t samples_per_channel = 0;
 
     uint32_t receive_timestamp = 0;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
-      int16_t enc_len_bytes = WebRtcPcm16b_Encode(
-          input.GetNextBlock(), expected_samples_per_channel, payload);
+      auto block = input.GetNextBlock();
+      ASSERT_EQ(expected_samples_per_channel, block.size());
+      size_t enc_len_bytes =
+          WebRtcPcm16b_Encode(block.data(), block.size(), payload);
       ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
 
       number_channels = 0;
       samples_per_channel = 0;
-      ASSERT_EQ(0,
-                neteq_->InsertPacket(rtp_info, payload,
-                                     static_cast<size_t>(enc_len_bytes),
-                                     receive_timestamp));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, enc_len_bytes),
+                                        receive_timestamp));
       ASSERT_EQ(0,
                 neteq_->GetAudio(kBlockSize32kHz,
                                  output,
                                  &samples_per_channel,
                                  &number_channels,
                                  &type));
-      ASSERT_EQ(1, number_channels);
+      ASSERT_EQ(1u, number_channels);
       ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
       ASSERT_EQ(kOutputNormal, type);
 
@@ -981,7 +1082,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
                                &samples_per_channel,
                                &number_channels,
                                &type));
-    ASSERT_EQ(1, number_channels);
+    ASSERT_EQ(1u, number_channels);
     ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
 
     // To be able to test the fading of background noise we need at lease to
@@ -1002,12 +1103,12 @@ class NetEqBgnTest : public NetEqDecodingTest {
                                  &samples_per_channel,
                                  &number_channels,
                                  &type));
-      ASSERT_EQ(1, number_channels);
+      ASSERT_EQ(1u, number_channels);
       ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
       if (type == kOutputPLCtoCNG) {
         plc_to_cng = true;
         double sum_squared = 0;
-        for (int k = 0; k < number_channels * samples_per_channel; ++k)
+        for (size_t k = 0; k < number_channels * samples_per_channel; ++k)
           sum_squared += output[k] * output[k];
         TestCondition(sum_squared, n > kFadingThreshold);
       } else {
@@ -1070,6 +1171,7 @@ TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(32000);
 }
 
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
 TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
@@ -1085,17 +1187,22 @@ TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   uint8_t kIsacPayloadType = 9;  // Payload type 8 is already registered.
 
   // Register decoders.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb,
-                                           kPcm16WbPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, kCngNbPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, kCngWbPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb32kHz,
-                                           kCngSwb32PayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb48kHz,
-                                           kCngSwb48PayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderAVT, kAvtPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderRED, kRedPayloadType));
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, kIsacPayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
+                                           "pcm16-wb", kPcm16WbPayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
+                                           "cng-nb", kCngNbPayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
+                                           "cng-wb", kCngWbPayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb32kHz,
+                                           "cng-swb32", kCngSwb32PayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb48kHz,
+                                           "cng-swb48", kCngSwb48PayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderAVT, "avt",
+                                           kAvtPayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderRED, "red",
+                                           kRedPayloadType));
+  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac",
+                                           kIsacPayloadType));
 
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = kPcm16WbPayloadType;
@@ -1106,8 +1213,7 @@ TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   // Payload length of 10 ms PCM16 16 kHz.
   const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
   uint8_t payload[kPayloadBytes] = {0};
-  ASSERT_EQ(0, neteq_->InsertPacket(
-      rtp_info, payload, kPayloadBytes, receive_timestamp));
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
 
   // Next packet. Last packet contained 10 ms audio.
   rtp_info.header.sequenceNumber++;
@@ -1145,6 +1251,7 @@ TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   --rtp_info.header.ssrc;
   EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 }
+#endif
 
 // First insert several noise like packets, then sync-packets. Decoding all
 // packets should not produce error, statistics should not show any packet loss
@@ -1165,17 +1272,16 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   NetEqOutputType output_type;
-  int num_channels;
-  int samples_per_channel;
+  size_t num_channels;
+  size_t samples_per_channel;
   uint32_t receive_timestamp = 0;
   for (int n = 0; n < 100; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
 
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
@@ -1193,7 +1299,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     if (n > algorithmic_frame_delay) {
       EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
     }
@@ -1205,8 +1311,7 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // We insert regular packets, if sync packet are not correctly buffered then
   // network statistics would show some packet loss.
   for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
@@ -1243,18 +1348,17 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   NetEqOutputType output_type;
-  int num_channels;
-  int samples_per_channel;
+  size_t num_channels;
+  size_t samples_per_channel;
   uint32_t receive_timestamp = 0;
   int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
   for (int n = 0; n < algorithmic_frame_delay; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
@@ -1281,8 +1385,7 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
 
   // Insert.
   for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
@@ -1294,7 +1397,7 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
   }
 }
@@ -1312,8 +1415,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
   const size_t kPayloadBytes = kSamples * sizeof(int16_t);
   double next_input_time_ms = 0.0;
   int16_t decoded[kBlockSize16kHz];
-  int num_channels;
-  int samples_per_channel;
+  size_t num_channels;
+  size_t samples_per_channel;
   NetEqOutputType output_type;
   uint32_t receive_timestamp = 0;
 
@@ -1334,8 +1437,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
       if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
         // This sequence number was not in the set to drop. Insert it.
         ASSERT_EQ(0,
-                  neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                       receive_timestamp));
+                  neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
         ++packets_inserted;
       }
       NetEqNetworkStatistics network_stats;
@@ -1366,7 +1468,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
 
     // Expect delay (in samples) to be less than 2 packets.
     EXPECT_LE(timestamp - PlayoutTimestamp(),
@@ -1416,14 +1518,14 @@ void NetEqDecodingTest::DuplicateCng() {
       algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
   // Insert three speech packets. Three are needed to get the frame length
   // correct.
-  int out_len;
-  int num_channels;
+  size_t out_len;
+  size_t num_channels;
   NetEqOutputType type;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
@@ -1442,7 +1544,9 @@ void NetEqDecodingTest::DuplicateCng() {
   size_t payload_len;
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   // This is the first time this CNG packet is inserted.
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+  ASSERT_EQ(
+      0, neteq_->InsertPacket(
+             rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio once and make sure CNG is played.
   ASSERT_EQ(0,
@@ -1454,7 +1558,9 @@ void NetEqDecodingTest::DuplicateCng() {
 
   // Insert the same CNG packet again. Note that at this point it is old, since
   // we have already decoded the first copy of it.
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+  ASSERT_EQ(
+      0, neteq_->InsertPacket(
+             rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
   // we have already pulled out CNG once.
@@ -1472,7 +1578,7 @@ void NetEqDecodingTest::DuplicateCng() {
   ++seq_no;
   timestamp += kCngPeriodSamples;
   PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
 
   // Pull audio once and verify that the output is speech again.
   ASSERT_EQ(0,
@@ -1507,14 +1613,16 @@ TEST_F(NetEqDecodingTest, CngFirst) {
   WebRtcRTPHeader rtp_info;
 
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-  ASSERT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+  ASSERT_EQ(
+      NetEq::kOK,
+      neteq_->InsertPacket(
+          rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
   ++seq_no;
   timestamp += kCngPeriodSamples;
 
   // Pull audio once and make sure CNG is played.
-  int out_len;
-  int num_channels;
+  size_t out_len;
+  size_t num_channels;
   NetEqOutputType type;
   ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                 &num_channels, &type));
@@ -1524,7 +1632,7 @@ TEST_F(NetEqDecodingTest, CngFirst) {
   // Insert some speech packets.
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.proto b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.proto
new file mode 100644
index 0000000000..4b59848eb2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.proto
@@ -0,0 +1,29 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.neteq_unittest;
+
+message NetEqNetworkStatistics {
+  optional uint32 current_buffer_size_ms = 1;
+  optional uint32 preferred_buffer_size_ms = 2;
+  optional uint32 jitter_peaks_found = 3;
+  optional uint32 packet_loss_rate = 4;
+  optional uint32 packet_discard_rate = 5;
+  optional uint32 expand_rate = 6;
+  optional uint32 speech_expand_rate = 7;
+  optional uint32 preemptive_rate = 8;
+  optional uint32 accelerate_rate = 9;
+  optional uint32 secondary_decoded_rate = 10;
+  optional int32 clockdrift_ppm = 11;
+  optional uint64 added_zero_samples = 12;
+  optional int32 mean_waiting_time_ms = 13;
+  optional int32 median_waiting_time_ms = 14;
+  optional int32 min_waiting_time_ms = 15;
+  optional int32 max_waiting_time_ms = 16;
+}
+
+message RtcpStatistics {
+  optional uint32 fraction_lost = 1;
+  optional uint32 cumulative_lost = 2;
+  optional uint32 extended_max_sequence_number = 3;
+  optional uint32 jitter = 4;
+}
\ No newline at end of file
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.cc
index fd4c46a55f..1b888f70d1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.cc
@@ -16,7 +16,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
@@ -45,7 +45,7 @@ int Normal::Process(const int16_t* input,
   output->PushBackInterleaved(input, length);
   int16_t* signal = &(*output)[0][0];
 
-  const unsigned fs_mult = fs_hz_ / 8000;
+  const int fs_mult = fs_hz_ / 8000;
   assert(fs_mult > 0);
   // fs_shift = log2(fs_mult), rounded down.
   // Note that |fs_shift| is not "exact" for 48 kHz.
@@ -73,18 +73,20 @@ int Normal::Process(const int16_t* input,
       int16_t* signal = &(*output)[channel_ix][0];
       size_t length_per_channel = length / output->Channels();
       // Find largest absolute value in new data.
-      int16_t decoded_max = WebRtcSpl_MaxAbsValueW16(
-        signal,  static_cast<int>(length_per_channel));
+      int16_t decoded_max =
+          WebRtcSpl_MaxAbsValueW16(signal, length_per_channel);
       // Adjust muting factor if needed (to BGN level).
-      int energy_length = std::min(static_cast<int>(fs_mult * 64),
-                                   static_cast<int>(length_per_channel));
+      size_t energy_length =
+          std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
       int scaling = 6 + fs_shift
           - WebRtcSpl_NormW32(decoded_max * decoded_max);
       scaling = std::max(scaling, 0);  // |scaling| should always be >= 0.
       int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
                                                      energy_length, scaling);
-      if ((energy_length >> scaling) > 0) {
-        energy = energy / (energy_length >> scaling);
+      int32_t scaled_energy_length =
+          static_cast<int32_t>(energy_length >> scaling);
+      if (scaled_energy_length > 0) {
+        energy = energy / scaled_energy_length;
       } else {
         energy = 0;
       }
@@ -97,18 +99,19 @@ int Normal::Process(const int16_t* input,
         // We want background_noise_.energy() / energy in Q14.
         int32_t bgn_energy =
             background_noise_.Energy(channel_ix) << (scaling+14);
-        int16_t energy_scaled = energy << scaling;
-        int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
-        mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(ratio) << 14);
+        int16_t energy_scaled = static_cast<int16_t>(energy << scaling);
+        int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
+        mute_factor = WebRtcSpl_SqrtFloor(ratio << 14);
       } else {
         mute_factor = 16384;  // 1.0 in Q14.
       }
       if (mute_factor > external_mute_factor_array[channel_ix]) {
-        external_mute_factor_array[channel_ix] = std::min(mute_factor, 16384);
+        external_mute_factor_array[channel_ix] =
+            static_cast<int16_t>(std::min(mute_factor, 16384));
       }
 
       // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-      int16_t increment = 64 / fs_mult;
+      int increment = 64 / fs_mult;
       for (size_t i = 0; i < length_per_channel; i++) {
         // Scale with mute factor.
         assert(channel_ix < output->Channels());
@@ -116,10 +119,11 @@ int Normal::Process(const int16_t* input,
         int32_t scaled_signal = (*output)[channel_ix][i] *
             external_mute_factor_array[channel_ix];
         // Shift 14 with proper rounding.
-        (*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
+        (*output)[channel_ix][i] =
+            static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
-        external_mute_factor_array[channel_ix] =
-            std::min(external_mute_factor_array[channel_ix] + increment, 16384);
+        external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
+            external_mute_factor_array[channel_ix] + increment, 16384));
       }
 
       // Interpolate the expanded data into the new vector.
@@ -127,20 +131,20 @@ int Normal::Process(const int16_t* input,
       assert(fs_shift < 3);  // Will always be 0, 1, or, 2.
       increment = 4 >> fs_shift;
       int fraction = increment;
-      for (size_t i = 0; i < 8 * fs_mult; i++) {
+      for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
         // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
         // now for legacy bit-exactness.
         assert(channel_ix < output->Channels());
         assert(i < output->Size());
         (*output)[channel_ix][i] =
-            (fraction * (*output)[channel_ix][i] +
-                (32 - fraction) * expanded[channel_ix][i] + 8) >> 5;
+            static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
+                (32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
         fraction += increment;
       }
     }
   } else if (last_mode == kModeRfc3389Cng) {
     assert(output->Channels() == 1);  // Not adapted for multi-channel yet.
-    static const int kCngLength = 32;
+    static const size_t kCngLength = 32;
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
@@ -163,7 +167,7 @@ int Normal::Process(const int16_t* input,
     assert(fs_shift < 3);  // Will always be 0, 1, or, 2.
     int16_t increment = 4 >> fs_shift;
     int16_t fraction = increment;
-    for (size_t i = 0; i < 8 * fs_mult; i++) {
+    for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
       // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
       // for legacy bit-exactness.
       signal[i] =
@@ -174,7 +178,7 @@ int Normal::Process(const int16_t* input,
     // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
     // still ramping up from previous muting.
     // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-    int16_t increment = 64 / fs_mult;
+    int increment = 64 / fs_mult;
     size_t length_per_channel = length / output->Channels();
     for (size_t i = 0; i < length_per_channel; i++) {
       for (size_t channel_ix = 0; channel_ix < output->Channels();
@@ -185,10 +189,11 @@ int Normal::Process(const int16_t* input,
         int32_t scaled_signal = (*output)[channel_ix][i] *
             external_mute_factor_array[channel_ix];
         // Shift 14 with proper rounding.
-        (*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
+        (*output)[channel_ix][i] =
+            static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
-        external_mute_factor_array[channel_ix] =
-            std::min(16384, external_mute_factor_array[channel_ix] + increment);
+        external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
+            16384, external_mute_factor_array[channel_ix] + increment));
       }
     }
   }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.h
index aa24b528af..23887f5134 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal.h
@@ -61,7 +61,7 @@ class Normal {
   const BackgroundNoise& background_noise_;
   Expand* expand_;
 
-  DISALLOW_COPY_AND_ASSIGN(Normal);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Normal);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal_unittest.cc
index 796409b25d..1ac32f46a7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -23,6 +23,7 @@
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 using ::testing::_;
@@ -36,7 +37,8 @@ TEST(Normal, CreateAndDestroy) {
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(1, 1000);
   RandomVector random_vector;
-  Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
   Normal normal(fs, &db, bgn, &expand);
   EXPECT_CALL(db, Die());  // Called when |db| goes out of scope.
 }
@@ -49,7 +51,9 @@ TEST(Normal, AvoidDivideByZero) {
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(1, 1000);
   RandomVector random_vector;
-  MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
   Normal normal(fs, &db, bgn, &expand);
 
   int16_t input[1000] = {0};
@@ -93,7 +97,9 @@ TEST(Normal, InputLengthAndChannelsDoNotMatch) {
   BackgroundNoise bgn(channels);
   SyncBuffer sync_buffer(channels, 1000);
   RandomVector random_vector;
-  MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
   Normal normal(fs, &db, bgn, &expand);
 
   int16_t input[1000] = {0};
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet.h
index 723ed8b0a3..64b325e027 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet.h
@@ -13,7 +13,7 @@
 
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.cc
index b0c939b8b9..c89de12318 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -16,6 +16,7 @@
 
 #include <algorithm>  // find_if()
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 
@@ -49,11 +50,16 @@ void PacketBuffer::Flush() {
   DeleteAllPackets(&buffer_);
 }
 
+bool PacketBuffer::Empty() const {
+  return buffer_.empty();
+}
+
 int PacketBuffer::InsertPacket(Packet* packet) {
   if (!packet || !packet->payload) {
     if (packet) {
       delete packet;
     }
+    LOG(LS_WARNING) << "InsertPacket invalid packet";
     return kInvalidPacket;
   }
 
@@ -62,6 +68,7 @@ int PacketBuffer::InsertPacket(Packet* packet) {
   if (buffer_.size() >= max_number_of_packets_) {
     // Buffer is full. Flush it.
     Flush();
+    LOG(LS_WARNING) << "Packet buffer flushed";
     return_val = kFlushed;
   }
 
@@ -174,7 +181,7 @@ const RTPHeader* PacketBuffer::NextRtpHeader() const {
   return const_cast<const RTPHeader*>(&(buffer_.front()->header));
 }
 
-Packet* PacketBuffer::GetNextPacket(int* discard_count) {
+Packet* PacketBuffer::GetNextPacket(size_t* discard_count) {
   if (Empty()) {
     // Buffer is empty.
     return NULL;
@@ -187,7 +194,7 @@ Packet* PacketBuffer::GetNextPacket(int* discard_count) {
 
   // Discard other packets with the same timestamp. These are duplicates or
   // redundant payloads that should not be used.
-  int discards = 0;
+  size_t discards = 0;
 
   while (!Empty() &&
       buffer_.front()->header.timestamp == packet->header.timestamp) {
@@ -229,25 +236,29 @@ int PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
   return 0;
 }
 
-int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
-                                     int last_decoded_length) const {
+int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) {
+  return DiscardOldPackets(timestamp_limit, 0);
+}
+
+size_t PacketBuffer::NumPacketsInBuffer() const {
+  return buffer_.size();
+}
+
+size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
+                                        size_t last_decoded_length) const {
   PacketList::const_iterator it;
-  int num_samples = 0;
-  int last_duration = last_decoded_length;
+  size_t num_samples = 0;
+  size_t last_duration = last_decoded_length;
   for (it = buffer_.begin(); it != buffer_.end(); ++it) {
     Packet* packet = (*it);
     AudioDecoder* decoder =
         decoder_database->GetDecoder(packet->header.payloadType);
-    if (decoder) {
-      int duration;
-      if (packet->sync_packet) {
-        duration = last_duration;
-      } else if (packet->primary) {
-        duration =
-            decoder->PacketDuration(packet->payload, packet->payload_length);
-      } else {
+    if (decoder && !packet->sync_packet) {
+      if (!packet->primary) {
         continue;
       }
+      int duration =
+          decoder->PacketDuration(packet->payload, packet->payload_length);
       if (duration >= 0) {
         last_duration = duration;  // Save the most up-to-date (valid) duration.
       }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.h
index b9a1618944..03c11e61b6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -43,7 +43,7 @@ class PacketBuffer {
   virtual void Flush();
 
   // Returns true for an empty buffer.
-  virtual bool Empty() const { return buffer_.empty(); }
+  virtual bool Empty() const;
 
   // Inserts |packet| into the buffer. The buffer will take over ownership of
   // the packet object.
@@ -88,7 +88,7 @@ class PacketBuffer {
   // Subsequent packets with the same timestamp as the one extracted will be
   // discarded and properly deleted. The number of discarded packets will be
   // written to the output variable |discard_count|.
-  virtual Packet* GetNextPacket(int* discard_count);
+  virtual Packet* GetNextPacket(size_t* discard_count);
 
   // Discards the first packet in the buffer. The packet is deleted.
   // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
@@ -105,20 +105,16 @@ class PacketBuffer {
                                 uint32_t horizon_samples);
 
   // Discards all packets that are (strictly) older than timestamp_limit.
-  virtual int DiscardAllOldPackets(uint32_t timestamp_limit) {
-    return DiscardOldPackets(timestamp_limit, 0);
-  }
+  virtual int DiscardAllOldPackets(uint32_t timestamp_limit);
 
   // Returns the number of packets in the buffer, including duplicates and
   // redundant packets.
-  virtual int NumPacketsInBuffer() const {
-    return static_cast<int>(buffer_.size());
-  }
+  virtual size_t NumPacketsInBuffer() const;
 
   // Returns the number of samples in the buffer, including samples carried in
   // duplicate and redundant packets.
-  virtual int NumSamplesInBuffer(DecoderDatabase* decoder_database,
-                                 int last_decoded_length) const;
+  virtual size_t NumSamplesInBuffer(DecoderDatabase* decoder_database,
+                                    size_t last_decoded_length) const;
 
   // Increase the waiting time counter for every packet in the buffer by |inc|.
   // The default value for |inc| is 1.
@@ -152,7 +148,7 @@ class PacketBuffer {
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
-  DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
index dc8b68c32c..435b6c848d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -97,7 +97,7 @@ TEST(PacketBuffer, InsertPacket) {
   EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
   EXPECT_EQ(4711u, next_ts);
   EXPECT_FALSE(buffer.Empty());
-  EXPECT_EQ(1, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
   const RTPHeader* hdr = buffer.NextRtpHeader();
   EXPECT_EQ(&(packet->header), hdr);  // Compare pointer addresses.
 
@@ -116,12 +116,12 @@ TEST(PacketBuffer, FlushBuffer) {
     Packet* packet = gen.NextPacket(payload_len);
     EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(packet));
   }
-  EXPECT_EQ(10, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
   EXPECT_FALSE(buffer.Empty());
 
   buffer.Flush();
   // Buffer should delete the payloads itself.
-  EXPECT_EQ(0, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(0u, buffer.NumPacketsInBuffer());
   EXPECT_TRUE(buffer.Empty());
 }
 
@@ -137,7 +137,7 @@ TEST(PacketBuffer, OverfillBuffer) {
     Packet* packet = gen.NextPacket(payload_len);
     EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(packet));
   }
-  EXPECT_EQ(10, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
   uint32_t next_ts;
   EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
   EXPECT_EQ(0u, next_ts);  // Expect first inserted packet to be first in line.
@@ -145,7 +145,7 @@ TEST(PacketBuffer, OverfillBuffer) {
   // Insert 11th packet; should flush the buffer and insert it after flushing.
   Packet* packet = gen.NextPacket(payload_len);
   EXPECT_EQ(PacketBuffer::kFlushed, buffer.InsertPacket(packet));
-  EXPECT_EQ(1, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
   EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
   // Expect last inserted packet to be first in line.
   EXPECT_EQ(packet->header.timestamp, next_ts);
@@ -179,7 +179,7 @@ TEST(PacketBuffer, InsertPacketList) {
                                                        &current_pt,
                                                        &current_cng_pt));
   EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
-  EXPECT_EQ(10, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
   EXPECT_EQ(0, current_pt);  // Current payload type changed to 0.
   EXPECT_EQ(0xFF, current_cng_pt);  // CNG payload type not changed.
 
@@ -220,7 +220,7 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) {
                                                             &current_pt,
                                                             &current_cng_pt));
   EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
-  EXPECT_EQ(1, buffer.NumPacketsInBuffer());  // Only the last packet.
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());  // Only the last packet.
   EXPECT_EQ(1, current_pt);  // Current payload type changed to 0.
   EXPECT_EQ(0xFF, current_cng_pt);  // CNG payload type not changed.
 
@@ -256,7 +256,7 @@ TEST(PacketBuffer, ExtractOrderRedundancy) {
     {0x0006, 0x0000001E, 1, false, -1},
   };
 
-  const int kExpectPacketsInBuffer = 9;
+  const size_t kExpectPacketsInBuffer = 9;
 
   std::vector<Packet*> expect_order(kExpectPacketsInBuffer);
 
@@ -277,10 +277,10 @@ TEST(PacketBuffer, ExtractOrderRedundancy) {
 
   EXPECT_EQ(kExpectPacketsInBuffer, buffer.NumPacketsInBuffer());
 
-  int drop_count;
-  for (int i = 0; i < kExpectPacketsInBuffer; ++i) {
+  size_t drop_count;
+  for (size_t i = 0; i < kExpectPacketsInBuffer; ++i) {
     Packet* packet = buffer.GetNextPacket(&drop_count);
-    EXPECT_EQ(0, drop_count);
+    EXPECT_EQ(0u, drop_count);
     EXPECT_EQ(packet, expect_order[i]);  // Compare pointer addresses.
     delete[] packet->payload;
     delete packet;
@@ -302,7 +302,7 @@ TEST(PacketBuffer, DiscardPackets) {
     Packet* packet = gen.NextPacket(payload_len);
     buffer.InsertPacket(packet);
   }
-  EXPECT_EQ(10, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
 
   // Discard them one by one and make sure that the right packets are at the
   // front of the buffer.
@@ -350,7 +350,7 @@ TEST(PacketBuffer, Reordering) {
                                                        decoder_database,
                                                        &current_pt,
                                                        &current_cng_pt));
-  EXPECT_EQ(10, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
 
   // Extract them and make sure that come out in the right order.
   uint32_t current_ts = start_ts;
@@ -425,7 +425,7 @@ TEST(PacketBuffer, Failures) {
                                      &current_pt,
                                      &current_cng_pt));
   EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
-  EXPECT_EQ(1, buffer->NumPacketsInBuffer());
+  EXPECT_EQ(1u, buffer->NumPacketsInBuffer());
   delete buffer;
   EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
 }
@@ -531,9 +531,14 @@ void TestIsObsoleteTimestamp(uint32_t limit_timestamp) {
   // 1 sample ahead is not old.
   EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
       limit_timestamp + 1, limit_timestamp, kZeroHorizon));
-  // 2^31 samples ahead is not old.
+  // If |t1-t2|=2^31 and t1>t2, t2 is older than t1 but not the opposite.
+  uint32_t other_timestamp = limit_timestamp + (1 << 31);
+  uint32_t lowest_timestamp = std::min(limit_timestamp, other_timestamp);
+  uint32_t highest_timestamp = std::max(limit_timestamp, other_timestamp);
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
+      lowest_timestamp, highest_timestamp, kZeroHorizon));
   EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
-      limit_timestamp + (1 << 31), limit_timestamp, kZeroHorizon));
+      highest_timestamp, lowest_timestamp, kZeroHorizon));
 
   // Fixed horizon at 10 samples.
   static const uint32_t kHorizon = 10;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.cc
index c19375b726..8530718134 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.cc
@@ -12,6 +12,7 @@
 
 #include <assert.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 
 namespace webrtc {
@@ -88,6 +89,7 @@ int PayloadSplitter::SplitRed(PacketList* packet_list) {
         // The block lengths in the RED headers do not match the overall packet
         // length. Something is corrupt. Discard this and the remaining
         // payloads from this packet.
+        LOG(LS_WARNING) << "SplitRed length mismatch";
         while (new_it != new_packets.end()) {
           // Payload should not have been allocated yet.
           assert(!(*new_it)->payload);
@@ -130,6 +132,7 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
     const DecoderDatabase::DecoderInfo* info =
         decoder_database->GetDecoderInfo(payload_type);
     if (!info) {
+      LOG(LS_WARNING) << "SplitFec unknown payload type";
       return kUnknownPayloadType;
     }
     // No splitting for a sync-packet.
@@ -149,8 +152,8 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
     }
 
     switch (info->codec_type) {
-      case kDecoderOpus:
-      case kDecoderOpus_2ch: {
+      case NetEqDecoder::kDecoderOpus:
+      case NetEqDecoder::kDecoderOpus_2ch: {
         // The main payload of this packet should be decoded as a primary
         // payload, even if it comes as a secondary payload in a RED packet.
         packet->primary = true;
@@ -171,6 +174,7 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
         break;
       }
       default: {
+        LOG(LS_WARNING) << "SplitFec wrong payload type";
         return kFecSplitError;
       }
     }
@@ -222,6 +226,7 @@ int PayloadSplitter::SplitAudio(PacketList* packet_list,
     const DecoderDatabase::DecoderInfo* info =
         decoder_database.GetDecoderInfo(packet->header.payloadType);
     if (!info) {
+      LOG(LS_WARNING) << "SplitAudio unknown payload type";
       return kUnknownPayloadType;
     }
     // No splitting for a sync-packet.
@@ -231,72 +236,73 @@ int PayloadSplitter::SplitAudio(PacketList* packet_list,
     }
     PacketList new_packets;
     switch (info->codec_type) {
-      case kDecoderPCMu:
-      case kDecoderPCMa: {
+      case NetEqDecoder::kDecoderPCMu:
+      case NetEqDecoder::kDecoderPCMa: {
         // 8 bytes per ms; 8 timestamps per ms.
         SplitBySamples(packet, 8, 8, &new_packets);
         break;
       }
-      case kDecoderPCMu_2ch:
-      case kDecoderPCMa_2ch: {
+      case NetEqDecoder::kDecoderPCMu_2ch:
+      case NetEqDecoder::kDecoderPCMa_2ch: {
         // 2 * 8 bytes per ms; 8 timestamps per ms.
         SplitBySamples(packet, 2 * 8, 8, &new_packets);
         break;
       }
-      case kDecoderG722: {
+      case NetEqDecoder::kDecoderG722: {
         // 8 bytes per ms; 16 timestamps per ms.
         SplitBySamples(packet, 8, 16, &new_packets);
         break;
       }
-      case kDecoderPCM16B: {
+      case NetEqDecoder::kDecoderPCM16B: {
         // 16 bytes per ms; 8 timestamps per ms.
         SplitBySamples(packet, 16, 8, &new_packets);
         break;
       }
-      case kDecoderPCM16Bwb: {
+      case NetEqDecoder::kDecoderPCM16Bwb: {
         // 32 bytes per ms; 16 timestamps per ms.
         SplitBySamples(packet, 32, 16, &new_packets);
         break;
       }
-      case kDecoderPCM16Bswb32kHz: {
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz: {
         // 64 bytes per ms; 32 timestamps per ms.
         SplitBySamples(packet, 64, 32, &new_packets);
         break;
       }
-      case kDecoderPCM16Bswb48kHz: {
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz: {
         // 96 bytes per ms; 48 timestamps per ms.
         SplitBySamples(packet, 96, 48, &new_packets);
         break;
       }
-      case kDecoderPCM16B_2ch: {
+      case NetEqDecoder::kDecoderPCM16B_2ch: {
         // 2 * 16 bytes per ms; 8 timestamps per ms.
         SplitBySamples(packet, 2 * 16, 8, &new_packets);
         break;
       }
-      case kDecoderPCM16Bwb_2ch: {
+      case NetEqDecoder::kDecoderPCM16Bwb_2ch: {
         // 2 * 32 bytes per ms; 16 timestamps per ms.
         SplitBySamples(packet, 2 * 32, 16, &new_packets);
         break;
       }
-      case kDecoderPCM16Bswb32kHz_2ch: {
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch: {
         // 2 * 64 bytes per ms; 32 timestamps per ms.
         SplitBySamples(packet, 2 * 64, 32, &new_packets);
         break;
       }
-      case kDecoderPCM16Bswb48kHz_2ch: {
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch: {
         // 2 * 96 bytes per ms; 48 timestamps per ms.
         SplitBySamples(packet, 2 * 96, 48, &new_packets);
         break;
       }
-      case kDecoderPCM16B_5ch: {
+      case NetEqDecoder::kDecoderPCM16B_5ch: {
         // 5 * 16 bytes per ms; 8 timestamps per ms.
         SplitBySamples(packet, 5 * 16, 8, &new_packets);
         break;
       }
-      case kDecoderILBC: {
+      case NetEqDecoder::kDecoderILBC: {
         size_t bytes_per_frame;
         int timestamps_per_frame;
         if (packet->payload_length >= 950) {
+          LOG(LS_WARNING) << "SplitAudio too large iLBC payload";
           return kTooLargePayload;
         }
         if (packet->payload_length % 38 == 0) {
@@ -308,6 +314,7 @@ int PayloadSplitter::SplitAudio(PacketList* packet_list,
           bytes_per_frame = 50;
           timestamps_per_frame = 240;
         } else {
+          LOG(LS_WARNING) << "SplitAudio invalid iLBC payload";
           return kFrameSplitError;
         }
         int ret = SplitByFrames(packet, bytes_per_frame, timestamps_per_frame,
@@ -402,6 +409,7 @@ int PayloadSplitter::SplitByFrames(const Packet* packet,
                                    uint32_t timestamps_per_frame,
                                    PacketList* new_packets) {
   if (packet->payload_length % bytes_per_frame != 0) {
+    LOG(LS_WARNING) << "SplitByFrames length mismatch";
     return kFrameSplitError;
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.h
index 6023d4e007..b0c4b5fe5c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter.h
@@ -83,7 +83,7 @@ class PayloadSplitter {
                             uint32_t timestamps_per_frame,
                             PacketList* new_packets);
 
-  DISALLOW_COPY_AND_ASSIGN(PayloadSplitter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PayloadSplitter);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
index 305e526bac..07c4bac0b6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
@@ -310,10 +310,10 @@ TEST(RedPayloadSplitter, CheckRedPayloads) {
   // easier to just register the payload types and let the actual implementation
   // do its job.
   DecoderDatabase decoder_database;
-  decoder_database.RegisterPayload(0, kDecoderCNGnb);
-  decoder_database.RegisterPayload(1, kDecoderPCMu);
-  decoder_database.RegisterPayload(2, kDecoderAVT);
-  decoder_database.RegisterPayload(3, kDecoderILBC);
+  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderCNGnb, "cng-nb");
+  decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
+  decoder_database.RegisterPayload(2, NetEqDecoder::kDecoderAVT, "avt");
+  decoder_database.RegisterPayload(3, NetEqDecoder::kDecoderILBC, "ilbc");
 
   PayloadSplitter splitter;
   splitter.CheckRedPayloads(&packet_list, decoder_database);
@@ -372,27 +372,33 @@ TEST(AudioPayloadSplitter, NonSplittable) {
   // codec types.
   // Use scoped pointers to avoid having to delete them later.
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info0(
-      new DecoderDatabase::DecoderInfo(kDecoderISAC, 16000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderISAC, 16000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
       .WillRepeatedly(Return(info0.get()));
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info1(
-      new DecoderDatabase::DecoderInfo(kDecoderISACswb, 32000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderISACswb, 32000,
+                                       NULL, false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(1))
       .WillRepeatedly(Return(info1.get()));
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info2(
-      new DecoderDatabase::DecoderInfo(kDecoderRED, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderRED, 8000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(2))
       .WillRepeatedly(Return(info2.get()));
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info3(
-      new DecoderDatabase::DecoderInfo(kDecoderAVT, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderAVT, 8000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(3))
       .WillRepeatedly(Return(info3.get()));
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info4(
-      new DecoderDatabase::DecoderInfo(kDecoderCNGnb, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderCNGnb, 8000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(4))
       .WillRepeatedly(Return(info4.get()));
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info5(
-      new DecoderDatabase::DecoderInfo(kDecoderArbitrary, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderArbitrary, 8000,
+                                       NULL, false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(5))
       .WillRepeatedly(Return(info5.get()));
 
@@ -452,53 +458,53 @@ class SplitBySamplesTest : public ::testing::TestWithParam<NetEqDecoder> {
   virtual void SetUp() {
     decoder_type_ = GetParam();
     switch (decoder_type_) {
-      case kDecoderPCMu:
-      case kDecoderPCMa:
+      case NetEqDecoder::kDecoderPCMu:
+      case NetEqDecoder::kDecoderPCMa:
         bytes_per_ms_ = 8;
         samples_per_ms_ = 8;
         break;
-      case kDecoderPCMu_2ch:
-      case kDecoderPCMa_2ch:
+      case NetEqDecoder::kDecoderPCMu_2ch:
+      case NetEqDecoder::kDecoderPCMa_2ch:
         bytes_per_ms_ = 2 * 8;
         samples_per_ms_ = 8;
         break;
-      case kDecoderG722:
+      case NetEqDecoder::kDecoderG722:
         bytes_per_ms_ = 8;
         samples_per_ms_ = 16;
         break;
-      case kDecoderPCM16B:
+      case NetEqDecoder::kDecoderPCM16B:
         bytes_per_ms_ = 16;
         samples_per_ms_ = 8;
         break;
-      case kDecoderPCM16Bwb:
+      case NetEqDecoder::kDecoderPCM16Bwb:
         bytes_per_ms_ = 32;
         samples_per_ms_ = 16;
         break;
-      case kDecoderPCM16Bswb32kHz:
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz:
         bytes_per_ms_ = 64;
         samples_per_ms_ = 32;
         break;
-      case kDecoderPCM16Bswb48kHz:
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz:
         bytes_per_ms_ = 96;
         samples_per_ms_ = 48;
         break;
-      case kDecoderPCM16B_2ch:
+      case NetEqDecoder::kDecoderPCM16B_2ch:
         bytes_per_ms_ = 2 * 16;
         samples_per_ms_ = 8;
         break;
-      case kDecoderPCM16Bwb_2ch:
+      case NetEqDecoder::kDecoderPCM16Bwb_2ch:
         bytes_per_ms_ = 2 * 32;
         samples_per_ms_ = 16;
         break;
-      case kDecoderPCM16Bswb32kHz_2ch:
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
         bytes_per_ms_ = 2 * 64;
         samples_per_ms_ = 32;
         break;
-      case kDecoderPCM16Bswb48kHz_2ch:
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
         bytes_per_ms_ = 2 * 96;
         samples_per_ms_ = 48;
         break;
-      case kDecoderPCM16B_5ch:
+      case NetEqDecoder::kDecoderPCM16B_5ch:
         bytes_per_ms_ = 5 * 16;
         samples_per_ms_ = 8;
         break;
@@ -569,14 +575,22 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
 }
 
 INSTANTIATE_TEST_CASE_P(
-    PayloadSplitter, SplitBySamplesTest,
-    ::testing::Values(kDecoderPCMu, kDecoderPCMa, kDecoderPCMu_2ch,
-                      kDecoderPCMa_2ch, kDecoderG722, kDecoderPCM16B,
-                      kDecoderPCM16Bwb, kDecoderPCM16Bswb32kHz,
-                      kDecoderPCM16Bswb48kHz, kDecoderPCM16B_2ch,
-                      kDecoderPCM16Bwb_2ch, kDecoderPCM16Bswb32kHz_2ch,
-                      kDecoderPCM16Bswb48kHz_2ch, kDecoderPCM16B_5ch));
-
+    PayloadSplitter,
+    SplitBySamplesTest,
+    ::testing::Values(NetEqDecoder::kDecoderPCMu,
+                      NetEqDecoder::kDecoderPCMa,
+                      NetEqDecoder::kDecoderPCMu_2ch,
+                      NetEqDecoder::kDecoderPCMa_2ch,
+                      NetEqDecoder::kDecoderG722,
+                      NetEqDecoder::kDecoderPCM16B,
+                      NetEqDecoder::kDecoderPCM16Bwb,
+                      NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                      NetEqDecoder::kDecoderPCM16Bswb48kHz,
+                      NetEqDecoder::kDecoderPCM16B_2ch,
+                      NetEqDecoder::kDecoderPCM16Bwb_2ch,
+                      NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch,
+                      NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch,
+                      NetEqDecoder::kDecoderPCM16B_5ch));
 
 class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
  protected:
@@ -609,7 +623,8 @@ TEST_P(SplitIlbcTest, NumFrames) {
   // codec types.
   // Use scoped pointers to avoid having to delete them later.
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(kDecoderILBC, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, 8000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -672,7 +687,8 @@ TEST(IlbcPayloadSplitter, TooLargePayload) {
 
   MockDecoderDatabase decoder_database;
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(kDecoderILBC, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, 8000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -703,7 +719,8 @@ TEST(IlbcPayloadSplitter, UnevenPayload) {
 
   MockDecoderDatabase decoder_database;
   rtc::scoped_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(kDecoderILBC, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, 8000, NULL,
+                                       false));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -728,8 +745,8 @@ TEST(FecPayloadSplitter, MixedPayload) {
   PacketList packet_list;
   DecoderDatabase decoder_database;
 
-  decoder_database.RegisterPayload(0, kDecoderOpus);
-  decoder_database.RegisterPayload(1, kDecoderPCMu);
+  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus, "opus");
+  decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
 
   Packet* packet = CreatePacket(0, 10, 0xFF, true);
   packet_list.push_back(packet);
@@ -785,7 +802,7 @@ TEST(FecPayloadSplitter, EmbedFecInRed) {
 
   const int kTimestampOffset = 20 * 48;  // 20 ms * 48 kHz.
   uint8_t payload_types[] = {0, 0};
-  decoder_database.RegisterPayload(0, kDecoderOpus);
+  decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus, "opus");
   Packet* packet = CreateRedPayload(2, payload_types, kTimestampOffset, true);
   packet_list.push_back(packet);
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
index 7ae7f97abc..714073ad10 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
@@ -20,7 +20,8 @@ PostDecodeVad::~PostDecodeVad() {
 void PostDecodeVad::Enable() {
   if (!vad_instance_) {
     // Create the instance.
-    if (WebRtcVad_Create(&vad_instance_) != 0) {
+    vad_instance_ = WebRtcVad_Create();
+    if (vad_instance_ == nullptr) {
       // Failed to create instance.
       Disable();
       return;
@@ -44,7 +45,7 @@ void PostDecodeVad::Init() {
   }
 }
 
-void PostDecodeVad::Update(int16_t* signal, int length,
+void PostDecodeVad::Update(int16_t* signal, size_t length,
                            AudioDecoder::SpeechType speech_type,
                            bool sid_frame,
                            int fs_hz) {
@@ -67,12 +68,13 @@ void PostDecodeVad::Update(int16_t* signal, int length,
   }
 
   if (length > 0 && running_) {
-    int vad_sample_index = 0;
+    size_t vad_sample_index = 0;
     active_speech_ = false;
     // Loop through frame sizes 30, 20, and 10 ms.
     for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
         vad_frame_size_ms -= 10) {
-      int vad_frame_size_samples = vad_frame_size_ms * fs_hz / 1000;
+      size_t vad_frame_size_samples =
+          static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
       while (length - vad_sample_index >= vad_frame_size_samples) {
         int vad_return = WebRtcVad_Process(
             vad_instance_, fs_hz, &signal[vad_sample_index],
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.h
index fa276aa41b..7bf5ad1383 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/post_decode_vad.h
@@ -46,7 +46,7 @@ class PostDecodeVad {
 
   // Updates post-decode VAD with the audio data in |signal| having |length|
   // samples. The data is of type |speech_type|, at the sample rate |fs_hz|.
-  void Update(int16_t* signal, int length,
+  void Update(int16_t* signal, size_t length,
               AudioDecoder::SpeechType speech_type, bool sid_frame, int fs_hz);
 
   // Accessors.
@@ -65,7 +65,7 @@ class PostDecodeVad {
   int sid_interval_counter_;
   ::VadInst* vad_instance_;
 
-  DISALLOW_COPY_AND_ASSIGN(PostDecodeVad);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PostDecodeVad);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
index b2dc3e60cb..f51a5bdbc5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
@@ -18,14 +18,14 @@ namespace webrtc {
 
 PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
     const int16_t* input,
-    int input_length,
-    int old_data_length,
+    size_t input_length,
+    size_t old_data_length,
     AudioMultiVector* output,
-    int16_t* length_change_samples) {
+    size_t* length_change_samples) {
   old_data_length_per_channel_ = old_data_length;
   // Input length must be (almost) 30 ms.
   // Also, the new part must be at least |overlap_samples_| elements.
-  static const int k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
   if (num_channels_ == 0 ||
       input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
       old_data_length >= input_length / num_channels_ - overlap_samples_) {
@@ -34,13 +34,14 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
     output->PushBackInterleaved(input, input_length);
     return kError;
   }
-  return TimeStretch::Process(input, input_length, output,
+  const bool kFastMode = false;  // Fast mode is not available for PE Expand.
+  return TimeStretch::Process(input, input_length, kFastMode, output,
                               length_change_samples);
 }
 
 void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
                                                      int16_t* best_correlation,
-                                                     int* peak_index) const {
+                                                     size_t* peak_index) const {
   // When the signal does not contain any active speech, the correlation does
   // not matter. Simply set it to zero.
   *best_correlation = 0;
@@ -50,17 +51,20 @@ void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
   // the new data.
   // but we must ensure that best_correlation is not larger than the new data.
   *peak_index = std::min(*peak_index,
-                         static_cast<int>(len - old_data_length_per_channel_));
+                         len - old_data_length_per_channel_);
 }
 
 PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
-    const int16_t *input, size_t input_length, size_t peak_index,
-    int16_t best_correlation, bool active_speech,
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool /*fast_mode*/,
     AudioMultiVector* output) const {
   // Pre-calculate common multiplication with |fs_mult_|.
   // 120 corresponds to 15 ms.
-  int fs_mult_120 = fs_mult_ * 120;
-  assert(old_data_length_per_channel_ >= 0);  // Make sure it's been set.
+  size_t fs_mult_120 = static_cast<size_t>(fs_mult_ * 120);
   // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
   // or passive speech.
   if (((best_correlation > kCorrelationThreshold) &&
@@ -102,7 +106,7 @@ PreemptiveExpand* PreemptiveExpandFactory::Create(
     int sample_rate_hz,
     size_t num_channels,
     const BackgroundNoise& background_noise,
-    int overlap_samples) const {
+    size_t overlap_samples) const {
   return new PreemptiveExpand(
       sample_rate_hz, num_channels, background_noise, overlap_samples);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.h
index 1aa6133014..c4c236080c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/preemptive_expand.h
@@ -32,44 +32,45 @@ class PreemptiveExpand : public TimeStretch {
   PreemptiveExpand(int sample_rate_hz,
                    size_t num_channels,
                    const BackgroundNoise& background_noise,
-                   int overlap_samples)
+                   size_t overlap_samples)
       : TimeStretch(sample_rate_hz, num_channels, background_noise),
-        old_data_length_per_channel_(-1),
+        old_data_length_per_channel_(0),
         overlap_samples_(overlap_samples) {
   }
 
-  virtual ~PreemptiveExpand() {}
-
   // This method performs the actual PreemptiveExpand operation. The samples are
   // read from |input|, of length |input_length| elements, and are written to
   // |output|. The number of samples added through time-stretching is
   // is provided in the output |length_change_samples|. The method returns
   // the outcome of the operation as an enumerator value.
   ReturnCodes Process(const int16_t *pw16_decoded,
-                      int len,
-                      int old_data_len,
+                      size_t len,
+                      size_t old_data_len,
                       AudioMultiVector* output,
-                      int16_t* length_change_samples);
+                      size_t* length_change_samples);
 
  protected:
   // Sets the parameters |best_correlation| and |peak_index| to suitable
   // values when the signal contains no active speech.
-  virtual void SetParametersForPassiveSpeech(size_t len,
-                                             int16_t* w16_bestCorr,
-                                             int* w16_bestIndex) const;
+  void SetParametersForPassiveSpeech(size_t input_length,
+                                     int16_t* best_correlation,
+                                     size_t* peak_index) const override;
 
   // Checks the criteria for performing the time-stretching operation and,
   // if possible, performs the time-stretching.
-  virtual ReturnCodes CheckCriteriaAndStretch(
-      const int16_t *pw16_decoded, size_t len, size_t w16_bestIndex,
-      int16_t w16_bestCorr, bool w16_VAD,
-      AudioMultiVector* output) const;
+  ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
+                                      size_t input_length,
+                                      size_t peak_index,
+                                      int16_t best_correlation,
+                                      bool active_speech,
+                                      bool /*fast_mode*/,
+                                      AudioMultiVector* output) const override;
 
  private:
-  int old_data_length_per_channel_;
-  int overlap_samples_;
+  size_t old_data_length_per_channel_;
+  size_t overlap_samples_;
 
-  DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PreemptiveExpand);
 };
 
 struct PreemptiveExpandFactory {
@@ -80,7 +81,7 @@ struct PreemptiveExpandFactory {
       int sample_rate_hz,
       size_t num_channels,
       const BackgroundNoise& background_noise,
-      int overlap_samples) const;
+      size_t overlap_samples) const;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector.h
index 767dc48eee..61651e57c2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/random_vector.h
@@ -21,7 +21,7 @@ namespace webrtc {
 // This class generates pseudo-random samples.
 class RandomVector {
  public:
-  static const int kRandomTableSize = 256;
+  static const size_t kRandomTableSize = 256;
   static const int16_t kRandomTable[kRandomTableSize];
 
   RandomVector()
@@ -43,7 +43,7 @@ class RandomVector {
   uint32_t seed_;
   int16_t seed_increment_;
 
-  DISALLOW_COPY_AND_ASSIGN(RandomVector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RandomVector);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.cc
index cf8e0280bb..7ef40bc814 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.cc
@@ -15,7 +15,7 @@
 #include <algorithm>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.h
index 2a765efa58..eacb328328 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/rtcp.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -51,7 +51,7 @@ class Rtcp {
   uint32_t jitter_;  // Current jitter value.
   int32_t transit_;  // Clock difference for previous packet.
 
-  DISALLOW_COPY_AND_ASSIGN(Rtcp);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index 14e93859b8..8f873762c5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -12,12 +12,97 @@
 
 #include <assert.h>
 #include <string.h>  // memset
+#include <algorithm>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
+// Allocating the static const so that it can be passed by reference to
+// RTC_DCHECK.
+const size_t StatisticsCalculator::kLenWaitingTimes;
+
+StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
+    const std::string& uma_name,
+    int report_interval_ms,
+    int max_value)
+    : uma_name_(uma_name),
+      report_interval_ms_(report_interval_ms),
+      max_value_(max_value),
+      timer_(0) {
+}
+
+StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;
+
+void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
+  timer_ += step_ms;
+  if (timer_ < report_interval_ms_) {
+    return;
+  }
+  LogToUma(Metric());
+  Reset();
+  timer_ -= report_interval_ms_;
+  RTC_DCHECK_GE(timer_, 0);
+}
+
+void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
+  RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
+}
+
+StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
+    const std::string& uma_name,
+    int report_interval_ms,
+    int max_value)
+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {
+}
+
+StatisticsCalculator::PeriodicUmaCount::~PeriodicUmaCount() {
+  // Log the count for the current (incomplete) interval.
+  LogToUma(Metric());
+}
+
+void StatisticsCalculator::PeriodicUmaCount::RegisterSample() {
+  ++counter_;
+}
+
+int StatisticsCalculator::PeriodicUmaCount::Metric() const {
+  return counter_;
+}
+
+void StatisticsCalculator::PeriodicUmaCount::Reset() {
+  counter_ = 0;
+}
+
+StatisticsCalculator::PeriodicUmaAverage::PeriodicUmaAverage(
+    const std::string& uma_name,
+    int report_interval_ms,
+    int max_value)
+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {
+}
+
+StatisticsCalculator::PeriodicUmaAverage::~PeriodicUmaAverage() {
+  // Log the average for the current (incomplete) interval.
+  LogToUma(Metric());
+}
+
+void StatisticsCalculator::PeriodicUmaAverage::RegisterSample(int value) {
+  sum_ += value;
+  ++counter_;
+}
+
+int StatisticsCalculator::PeriodicUmaAverage::Metric() const {
+  return static_cast<int>(sum_ / counter_);
+}
+
+void StatisticsCalculator::PeriodicUmaAverage::Reset() {
+  sum_ = 0.0;
+  counter_ = 0;
+}
+
 StatisticsCalculator::StatisticsCalculator()
     : preemptive_samples_(0),
       accelerate_samples_(0),
@@ -27,12 +112,18 @@ StatisticsCalculator::StatisticsCalculator()
       discarded_packets_(0),
       lost_timestamps_(0),
       timestamps_since_last_report_(0),
-      len_waiting_times_(0),
-      next_waiting_time_index_(0),
-      secondary_decoded_samples_(0) {
-  memset(waiting_times_, 0, kLenWaitingTimes * sizeof(waiting_times_[0]));
+      secondary_decoded_samples_(0),
+      delayed_packet_outage_counter_(
+          "WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
+          60000,  // 60 seconds report interval.
+          100),
+      excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
+                           60000,  // 60 seconds report interval.
+                           1000) {
 }
 
+StatisticsCalculator::~StatisticsCalculator() = default;
+
 void StatisticsCalculator::Reset() {
   preemptive_samples_ = 0;
   accelerate_samples_ = 0;
@@ -40,6 +131,7 @@ void StatisticsCalculator::Reset() {
   expanded_speech_samples_ = 0;
   expanded_noise_samples_ = 0;
   secondary_decoded_samples_ = 0;
+  waiting_times_.clear();
 }
 
 void StatisticsCalculator::ResetMcu() {
@@ -48,42 +140,40 @@ void StatisticsCalculator::ResetMcu() {
   timestamps_since_last_report_ = 0;
 }
 
-void StatisticsCalculator::ResetWaitingTimeStatistics() {
-  memset(waiting_times_, 0, kLenWaitingTimes * sizeof(waiting_times_[0]));
-  len_waiting_times_ = 0;
-  next_waiting_time_index_ = 0;
-}
-
-void StatisticsCalculator::ExpandedVoiceSamples(int num_samples) {
+void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples) {
   expanded_speech_samples_ += num_samples;
 }
 
-void StatisticsCalculator::ExpandedNoiseSamples(int num_samples) {
+void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples) {
   expanded_noise_samples_ += num_samples;
 }
 
-void StatisticsCalculator::PreemptiveExpandedSamples(int num_samples) {
+void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
   preemptive_samples_ += num_samples;
 }
 
-void StatisticsCalculator::AcceleratedSamples(int num_samples) {
+void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
   accelerate_samples_ += num_samples;
 }
 
-void StatisticsCalculator::AddZeros(int num_samples) {
+void StatisticsCalculator::AddZeros(size_t num_samples) {
   added_zero_samples_ += num_samples;
 }
 
-void StatisticsCalculator::PacketsDiscarded(int num_packets) {
+void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
   discarded_packets_ += num_packets;
 }
 
-void StatisticsCalculator::LostSamples(int num_samples) {
+void StatisticsCalculator::LostSamples(size_t num_samples) {
   lost_timestamps_ += num_samples;
 }
 
-void StatisticsCalculator::IncreaseCounter(int num_samples, int fs_hz) {
-  timestamps_since_last_report_ += num_samples;
+void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
+  const int time_step_ms =
+      rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
+  delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
+  excess_buffer_delay_.AdvanceClock(time_step_ms);
+  timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
   if (timestamps_since_last_report_ >
       static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
     lost_timestamps_ = 0;
@@ -96,22 +186,27 @@ void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
   secondary_decoded_samples_ += num_samples;
 }
 
+void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
+  RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.DelayedPacketOutageEventMs",
+                              outage_duration_ms, 1 /* min */, 2000 /* max */,
+                              100 /* bucket count */);
+  delayed_packet_outage_counter_.RegisterSample();
+}
+
 void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
-  assert(next_waiting_time_index_ < kLenWaitingTimes);
-  waiting_times_[next_waiting_time_index_] = waiting_time_ms;
-  next_waiting_time_index_++;
-  if (next_waiting_time_index_ >= kLenWaitingTimes) {
-    next_waiting_time_index_ = 0;
-  }
-  if (len_waiting_times_ < kLenWaitingTimes) {
-    len_waiting_times_++;
+  excess_buffer_delay_.RegisterSample(waiting_time_ms);
+  RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
+  if (waiting_times_.size() == kLenWaitingTimes) {
+    // Erase first value.
+    waiting_times_.pop_front();
   }
+  waiting_times_.push_back(waiting_time_ms);
 }
 
 void StatisticsCalculator::GetNetworkStatistics(
     int fs_hz,
-    int num_samples_in_buffers,
-    int samples_per_packet,
+    size_t num_samples_in_buffers,
+    size_t samples_per_packet,
     const DelayManager& delay_manager,
     const DecisionLogic& decision_logic,
     NetEqNetworkStatistics *stats) {
@@ -121,9 +216,10 @@ void StatisticsCalculator::GetNetworkStatistics(
   }
 
   stats->added_zero_samples = added_zero_samples_;
-  stats->current_buffer_size_ms = num_samples_in_buffers * 1000 / fs_hz;
-  const int ms_per_packet = decision_logic.packet_length_samples() /
-      (fs_hz / 1000);
+  stats->current_buffer_size_ms =
+      static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
+  const int ms_per_packet = rtc::checked_cast<int>(
+      decision_logic.packet_length_samples() / (fs_hz / 1000));
   stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) *
       ms_per_packet;
   stats->jitter_peaks_found = delay_manager.PeakFound();
@@ -132,7 +228,7 @@ void StatisticsCalculator::GetNetworkStatistics(
   stats->packet_loss_rate =
       CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
 
-  const unsigned discarded_samples = discarded_packets_ * samples_per_packet;
+  const size_t discarded_samples = discarded_packets_ * samples_per_packet;
   stats->packet_discard_rate =
       CalculateQ14Ratio(discarded_samples, timestamps_since_last_report_);
 
@@ -148,33 +244,49 @@ void StatisticsCalculator::GetNetworkStatistics(
 
   stats->speech_expand_rate =
       CalculateQ14Ratio(expanded_speech_samples_,
-      timestamps_since_last_report_);
+                        timestamps_since_last_report_);
 
   stats->secondary_decoded_rate =
       CalculateQ14Ratio(secondary_decoded_samples_,
                         timestamps_since_last_report_);
 
+  if (waiting_times_.size() == 0) {
+    stats->mean_waiting_time_ms = -1;
+    stats->median_waiting_time_ms = -1;
+    stats->min_waiting_time_ms = -1;
+    stats->max_waiting_time_ms = -1;
+  } else {
+    std::sort(waiting_times_.begin(), waiting_times_.end());
+    // Find mid-point elements. If the size is odd, the two values
+    // |middle_left| and |middle_right| will both be the one middle element; if
+    // the size is even, they will be the the two neighboring elements at the
+    // middle of the list.
+    const int middle_left = waiting_times_[(waiting_times_.size() - 1) / 2];
+    const int middle_right = waiting_times_[waiting_times_.size() / 2];
+    // Calculate the average of the two. (Works also for odd sizes.)
+    stats->median_waiting_time_ms = (middle_left + middle_right) / 2;
+    stats->min_waiting_time_ms = waiting_times_.front();
+    stats->max_waiting_time_ms = waiting_times_.back();
+    double sum = 0;
+    for (auto time : waiting_times_) {
+      sum += time;
+    }
+    stats->mean_waiting_time_ms = static_cast<int>(sum / waiting_times_.size());
+  }
+
   // Reset counters.
   ResetMcu();
   Reset();
 }
 
-void StatisticsCalculator::WaitingTimes(std::vector<int>* waiting_times) {
-  if (!waiting_times) {
-    return;
-  }
-  waiting_times->assign(waiting_times_, waiting_times_ + len_waiting_times_);
-  ResetWaitingTimeStatistics();
-}
-
-int StatisticsCalculator::CalculateQ14Ratio(uint32_t numerator,
-                                            uint32_t denominator) {
+uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
+                                                 uint32_t denominator) {
   if (numerator == 0) {
     return 0;
   } else if (numerator < denominator) {
     // Ratio must be smaller than 1 in Q14.
     assert((numerator << 14) / denominator < (1 << 14));
-    return (numerator << 14) / denominator;
+    return static_cast<uint16_t>((numerator << 14) / denominator);
   } else {
     // Will not produce a ratio larger than 1, since this is probably an error.
     return 1 << 14;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.h
index cd4d8677de..b2df865f63 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/statistics_calculator.h
@@ -11,10 +11,11 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
 
-#include <vector>
+#include <deque>
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -28,7 +29,7 @@ class StatisticsCalculator {
  public:
   StatisticsCalculator();
 
-  virtual ~StatisticsCalculator() {}
+  virtual ~StatisticsCalculator();
 
   // Resets most of the counters.
   void Reset();
@@ -36,36 +37,34 @@ class StatisticsCalculator {
   // Resets the counters that are not handled by Reset().
   void ResetMcu();
 
-  // Resets the waiting time statistics.
-  void ResetWaitingTimeStatistics();
-
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced other than just noise samples.
-  void ExpandedVoiceSamples(int num_samples);
+  void ExpandedVoiceSamples(size_t num_samples);
 
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced only noise samples.
-  void ExpandedNoiseSamples(int num_samples);
+  void ExpandedNoiseSamples(size_t num_samples);
 
   // Reports that |num_samples| samples were produced through preemptive
   // expansion.
-  void PreemptiveExpandedSamples(int num_samples);
+  void PreemptiveExpandedSamples(size_t num_samples);
 
   // Reports that |num_samples| samples were removed through accelerate.
-  void AcceleratedSamples(int num_samples);
+  void AcceleratedSamples(size_t num_samples);
 
   // Reports that |num_samples| zeros were inserted into the output.
-  void AddZeros(int num_samples);
+  void AddZeros(size_t num_samples);
 
   // Reports that |num_packets| packets were discarded.
-  void PacketsDiscarded(int num_packets);
+  void PacketsDiscarded(size_t num_packets);
 
   // Reports that |num_samples| were lost.
-  void LostSamples(int num_samples);
+  void LostSamples(size_t num_samples);
 
   // Increases the report interval counter with |num_samples| at a sample rate
-  // of |fs_hz|.
-  void IncreaseCounter(int num_samples, int fs_hz);
+  // of |fs_hz|. This is how the StatisticsCalculator gets notified that current
+  // time is increasing.
+  void IncreaseCounter(size_t num_samples, int fs_hz);
 
   // Stores new packet waiting time in waiting time statistics.
   void StoreWaitingTime(int waiting_time_ms);
@@ -73,40 +72,95 @@ class StatisticsCalculator {
   // Reports that |num_samples| samples were decoded from secondary packets.
   void SecondaryDecodedSamples(int num_samples);
 
+  // Logs a delayed packet outage event of |outage_duration_ms|. A delayed
+  // packet outage event is defined as an expand period caused not by an actual
+  // packet loss, but by a delayed packet.
+  virtual void LogDelayedPacketOutageEvent(int outage_duration_ms);
+
   // Returns the current network statistics in |stats|. The current sample rate
   // is |fs_hz|, the total number of samples in packet buffer and sync buffer
   // yet to play out is |num_samples_in_buffers|, and the number of samples per
   // packet is |samples_per_packet|.
   void GetNetworkStatistics(int fs_hz,
-                            int num_samples_in_buffers,
-                            int samples_per_packet,
+                            size_t num_samples_in_buffers,
+                            size_t samples_per_packet,
                             const DelayManager& delay_manager,
                             const DecisionLogic& decision_logic,
                             NetEqNetworkStatistics *stats);
 
-  void WaitingTimes(std::vector<int>* waiting_times);
-
  private:
   static const int kMaxReportPeriod = 60;  // Seconds before auto-reset.
-  static const int kLenWaitingTimes = 100;
+  static const size_t kLenWaitingTimes = 100;
+
+  class PeriodicUmaLogger {
+   public:
+    PeriodicUmaLogger(const std::string& uma_name,
+                      int report_interval_ms,
+                      int max_value);
+    virtual ~PeriodicUmaLogger();
+    void AdvanceClock(int step_ms);
+
+   protected:
+    void LogToUma(int value) const;
+    virtual int Metric() const = 0;
+    virtual void Reset() = 0;
+
+    const std::string uma_name_;
+    const int report_interval_ms_;
+    const int max_value_;
+    int timer_ = 0;
+  };
+
+  class PeriodicUmaCount final : public PeriodicUmaLogger {
+   public:
+    PeriodicUmaCount(const std::string& uma_name,
+                     int report_interval_ms,
+                     int max_value);
+    ~PeriodicUmaCount() override;
+    void RegisterSample();
+
+   protected:
+    int Metric() const override;
+    void Reset() override;
+
+   private:
+    int counter_ = 0;
+  };
+
+  class PeriodicUmaAverage final : public PeriodicUmaLogger {
+   public:
+    PeriodicUmaAverage(const std::string& uma_name,
+                       int report_interval_ms,
+                       int max_value);
+    ~PeriodicUmaAverage() override;
+    void RegisterSample(int value);
+
+   protected:
+    int Metric() const override;
+    void Reset() override;
+
+   private:
+    double sum_ = 0.0;
+    int counter_ = 0;
+  };
 
   // Calculates numerator / denominator, and returns the value in Q14.
-  static int CalculateQ14Ratio(uint32_t numerator, uint32_t denominator);
+  static uint16_t CalculateQ14Ratio(size_t numerator, uint32_t denominator);
 
-  uint32_t preemptive_samples_;
-  uint32_t accelerate_samples_;
-  int added_zero_samples_;
-  uint32_t expanded_speech_samples_;
-  uint32_t expanded_noise_samples_;
-  int discarded_packets_;
-  uint32_t lost_timestamps_;
+  size_t preemptive_samples_;
+  size_t accelerate_samples_;
+  size_t added_zero_samples_;
+  size_t expanded_speech_samples_;
+  size_t expanded_noise_samples_;
+  size_t discarded_packets_;
+  size_t lost_timestamps_;
   uint32_t timestamps_since_last_report_;
-  int waiting_times_[kLenWaitingTimes];  // Used as a circular buffer.
-  int len_waiting_times_;
-  int next_waiting_time_index_;
+  std::deque<int> waiting_times_;
   uint32_t secondary_decoded_samples_;
+  PeriodicUmaCount delayed_packet_outage_counter_;
+  PeriodicUmaAverage excess_buffer_delay_;
 
-  DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer.h
index 59bd4d87e2..38e7887794 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -25,8 +25,6 @@ class SyncBuffer : public AudioMultiVector {
         end_timestamp_(0),
         dtmf_index_(0) {}
 
-  virtual ~SyncBuffer() {}
-
   // Returns the number of samples yet to play out form the buffer.
   size_t FutureLength() const;
 
@@ -34,7 +32,7 @@ class SyncBuffer : public AudioMultiVector {
   // the same number of samples from the beginning of the SyncBuffer, to
   // maintain a constant buffer size. The |next_index_| is updated to reflect
   // the move of the beginning of "future" data.
-  void PushBack(const AudioMultiVector& append_this);
+  void PushBack(const AudioMultiVector& append_this) override;
 
   // Adds |length| zeros to the beginning of each channel. Removes
   // the same number of samples from the end of the SyncBuffer, to
@@ -94,7 +92,7 @@ class SyncBuffer : public AudioMultiVector {
   uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
   size_t dtmf_index_;  // Index to the first non-DTMF sample in the buffer.
 
-  DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 3fbce8be5c..56ed72fcee 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -14,7 +14,7 @@
 #include <map>
 #include <stdio.h>
 #include "webrtc/typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 enum stereoModes {
     stereoModeMono,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
index c46a3daece..aba525b162 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
@@ -39,7 +39,7 @@
 #define NETEQ_CODEC_G722_1_16_PT		108
 #define NETEQ_CODEC_G722_1_24_PT		109
 #define NETEQ_CODEC_G722_1_32_PT		110
-#define NETEQ_CODEC_SC3_PT				111
+#define NETEQ_CODEC_OPUS_PT             111
 #define NETEQ_CODEC_AMR_PT				112
 #define NETEQ_CODEC_GSMEFR_PT			113
 //#define NETEQ_CODEC_ILBCRCU_PT			114
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 4e779b49b0..45586ee111 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-//TODO(hlundin): Reformat file to meet style guide.
+// TODO(hlundin): Reformat file to meet style guide.
 
 /* header includes */
 #include <stdio.h>
@@ -23,10 +23,14 @@
 
 #include <assert.h>
 
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
 #include "webrtc/typedefs.h"
+
 // needed for NetEqDecoder
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 /************************/
 /* Define payload types */
@@ -34,23 +38,29 @@
 
 #include "PayloadTypes.h"
 
-
+namespace {
+const size_t kRtpDataSize = 8000;
+}
 
 /*********************/
 /* Misc. definitions */
 /*********************/
 
 #define STOPSENDTIME 3000
-#define RESTARTSENDTIME 0 //162500
+#define RESTARTSENDTIME 0  // 162500
 #define FIRSTLINELEN 40
-#define CHECK_NOT_NULL(a) if((a)==0){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
+#define CHECK_NOT_NULL(a)                                                \
+  if ((a) == 0) {                                                        \
+    printf("\n %s \n line: %d \nerror at %s\n", __FILE__, __LINE__, #a); \
+    return (-1);                                                         \
+  }
 
 //#define MULTIPLE_SAME_TIMESTAMP
 #define REPEAT_PACKET_DISTANCE 17
 #define REPEAT_PACKET_COUNT 1  // number of extra packets to send
 
 //#define INSERT_OLD_PACKETS
-#define OLD_PACKET 5 // how many seconds too old should the packet be?
+#define OLD_PACKET 5  // how many seconds too old should the packet be?
 
 //#define TIMESTAMP_WRAPAROUND
 
@@ -64,8 +74,8 @@
 #define DTMF_DURATION 500
 
 #define STEREO_MODE_FRAME 0
-#define STEREO_MODE_SAMPLE_1 1 //1 octet per sample
-#define STEREO_MODE_SAMPLE_2 2 //2 octets per sample
+#define STEREO_MODE_SAMPLE_1 1  // 1 octet per sample
+#define STEREO_MODE_SAMPLE_2 2  // 2 octets per sample
 
 /*************************/
 /* Function declarations */
@@ -74,27 +84,27 @@
 void NetEQTest_GetCodec_and_PT(char* name,
                                webrtc::NetEqDecoder* codec,
                                int* PT,
-                               int frameLen,
+                               size_t frameLen,
                                int* fs,
                                int* bitrate,
                                int* useRed);
 int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
-                          int enc_frameSize,
+                          size_t enc_frameSize,
                           int bitrate,
                           int sampfreq,
                           int vad,
-                          int numChannels);
+                          size_t numChannels);
 void defineCodecs(webrtc::NetEqDecoder* usedCodec, int* noOfCodecs);
-int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels);
-int NetEQTest_encode(int coder,
-                     int16_t* indata,
-                     int frameLen,
-                     unsigned char* encoded,
-                     int sampleRate,
-                     int* vad,
-                     int useVAD,
-                     int bitrate,
-                     int numChannels);
+int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels);
+size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
+                        int16_t* indata,
+                        size_t frameLen,
+                        unsigned char* encoded,
+                        int sampleRate,
+                        int* vad,
+                        int useVAD,
+                        int bitrate,
+                        size_t numChannels);
 void makeRTPheader(unsigned char* rtp_data,
                    int payloadType,
                    int seqNo,
@@ -107,13 +117,13 @@ int makeRedundantHeader(unsigned char* rtp_data,
                         uint16_t* blockLen,
                         int seqNo,
                         uint32_t ssrc);
-int makeDTMFpayload(unsigned char* payload_data,
-                    int Event,
-                    int End,
-                    int Volume,
-                    int Duration);
-void stereoDeInterleave(int16_t* audioSamples, int numSamples);
-void stereoInterleave(unsigned char* data, int dataLen, int stride);
+size_t makeDTMFpayload(unsigned char* payload_data,
+                       int Event,
+                       int End,
+                       int Volume,
+                       int Duration);
+void stereoDeInterleave(int16_t* audioSamples, size_t numSamples);
+void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
 
 /*********************/
 /* Codec definitions */
@@ -121,702 +131,706 @@ void stereoInterleave(unsigned char* data, int dataLen, int stride);
 
 #include "webrtc_vad.h"
 
-#if ((defined CODEC_PCM16B)||(defined NETEQ_ARBITRARY_CODEC))
-	#include "pcm16b.h"
+#if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #endif
 #ifdef CODEC_G711
-	#include "g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 #endif
 #ifdef CODEC_G729
-	#include "G729Interface.h"
+#include "G729Interface.h"
 #endif
 #ifdef CODEC_G729_1
-	#include "G729_1Interface.h"
+#include "G729_1Interface.h"
 #endif
 #ifdef CODEC_AMR
-	#include "AMRInterface.h"
-	#include "AMRCreation.h"
+#include "AMRInterface.h"
+#include "AMRCreation.h"
 #endif
 #ifdef CODEC_AMRWB
-	#include "AMRWBInterface.h"
-	#include "AMRWBCreation.h"
+#include "AMRWBInterface.h"
+#include "AMRWBCreation.h"
 #endif
 #ifdef CODEC_ILBC
-	#include "ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 #endif
-#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB) 
-	#include "isac.h"
+#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 #endif
 #ifdef NETEQ_ISACFIX_CODEC
-	#include "isacfix.h"
-	#ifdef CODEC_ISAC
-		#error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp
-	#endif
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#ifdef CODEC_ISAC
+#error Cannot have both ISAC and ISACfix defined. Please de-select one.
+#endif
 #endif
 #ifdef CODEC_G722
-	#include "g722_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #endif
 #ifdef CODEC_G722_1_24
-	#include "G722_1Interface.h"
+#include "G722_1Interface.h"
 #endif
 #ifdef CODEC_G722_1_32
-	#include "G722_1Interface.h"
+#include "G722_1Interface.h"
 #endif
 #ifdef CODEC_G722_1_16
-	#include "G722_1Interface.h"
+#include "G722_1Interface.h"
 #endif
 #ifdef CODEC_G722_1C_24
-	#include "G722_1Interface.h"
+#include "G722_1Interface.h"
 #endif
 #ifdef CODEC_G722_1C_32
-	#include "G722_1Interface.h"
+#include "G722_1Interface.h"
 #endif
 #ifdef CODEC_G722_1C_48
-	#include "G722_1Interface.h"
+#include "G722_1Interface.h"
 #endif
 #ifdef CODEC_G726
-    #include "G726Creation.h"
-    #include "G726Interface.h"
+#include "G726Creation.h"
+#include "G726Interface.h"
 #endif
 #ifdef CODEC_GSMFR
-	#include "GSMFRInterface.h"
-	#include "GSMFRCreation.h"
+#include "GSMFRInterface.h"
+#include "GSMFRCreation.h"
 #endif
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
-    defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-  #include "webrtc_cng.h"
+     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #endif
-#if ((defined CODEC_SPEEX_8)||(defined CODEC_SPEEX_16))
-	#include "SpeexInterface.h"
+#if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
+#include "SpeexInterface.h"
+#endif
+#ifdef CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #endif
 
 /***********************************/
 /* Global codec instance variables */
 /***********************************/
 
-WebRtcVadInst *VAD_inst[2];
+WebRtcVadInst* VAD_inst[2];
 
 #ifdef CODEC_G722
-    G722EncInst *g722EncState[2];
+G722EncInst* g722EncState[2];
 #endif
 
 #ifdef CODEC_G722_1_24
-	G722_1_24_encinst_t *G722_1_24enc_inst[2];
+G722_1_24_encinst_t* G722_1_24enc_inst[2];
 #endif
 #ifdef CODEC_G722_1_32
-	G722_1_32_encinst_t *G722_1_32enc_inst[2];
+G722_1_32_encinst_t* G722_1_32enc_inst[2];
 #endif
 #ifdef CODEC_G722_1_16
-	G722_1_16_encinst_t *G722_1_16enc_inst[2];
+G722_1_16_encinst_t* G722_1_16enc_inst[2];
 #endif
 #ifdef CODEC_G722_1C_24
-	G722_1C_24_encinst_t *G722_1C_24enc_inst[2];
+G722_1C_24_encinst_t* G722_1C_24enc_inst[2];
 #endif
 #ifdef CODEC_G722_1C_32
-	G722_1C_32_encinst_t *G722_1C_32enc_inst[2];
+G722_1C_32_encinst_t* G722_1C_32enc_inst[2];
 #endif
 #ifdef CODEC_G722_1C_48
-	G722_1C_48_encinst_t *G722_1C_48enc_inst[2];
+G722_1C_48_encinst_t* G722_1C_48enc_inst[2];
 #endif
 #ifdef CODEC_G726
-    G726_encinst_t *G726enc_inst[2];
+G726_encinst_t* G726enc_inst[2];
 #endif
 #ifdef CODEC_G729
-	G729_encinst_t *G729enc_inst[2];
+G729_encinst_t* G729enc_inst[2];
 #endif
 #ifdef CODEC_G729_1
-	G729_1_inst_t *G729_1_inst[2];
+G729_1_inst_t* G729_1_inst[2];
 #endif
 #ifdef CODEC_AMR
-	AMR_encinst_t *AMRenc_inst[2];
-	int16_t		  AMR_bitrate;
+AMR_encinst_t* AMRenc_inst[2];
+int16_t AMR_bitrate;
 #endif
 #ifdef CODEC_AMRWB
-	AMRWB_encinst_t *AMRWBenc_inst[2];
-	int16_t		  AMRWB_bitrate;
+AMRWB_encinst_t* AMRWBenc_inst[2];
+int16_t AMRWB_bitrate;
 #endif
 #ifdef CODEC_ILBC
-	IlbcEncoderInstance *iLBCenc_inst[2];
+IlbcEncoderInstance* iLBCenc_inst[2];
 #endif
 #ifdef CODEC_ISAC
-	ISACStruct *ISAC_inst[2];
+ISACStruct* ISAC_inst[2];
 #endif
 #ifdef NETEQ_ISACFIX_CODEC
-	ISACFIX_MainStruct *ISAC_inst[2];
+ISACFIX_MainStruct* ISAC_inst[2];
 #endif
 #ifdef CODEC_ISAC_SWB
-	ISACStruct *ISACSWB_inst[2];
+ISACStruct* ISACSWB_inst[2];
 #endif
 #ifdef CODEC_GSMFR
-	GSMFR_encinst_t *GSMFRenc_inst[2];
+GSMFR_encinst_t* GSMFRenc_inst[2];
 #endif
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
-    defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-	CNG_enc_inst *CNGenc_inst[2];
+     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+CNG_enc_inst* CNGenc_inst[2];
 #endif
 #ifdef CODEC_SPEEX_8
-	SPEEX_encinst_t *SPEEX8enc_inst[2];
+SPEEX_encinst_t* SPEEX8enc_inst[2];
 #endif
 #ifdef CODEC_SPEEX_16
-	SPEEX_encinst_t *SPEEX16enc_inst[2];
+SPEEX_encinst_t* SPEEX16enc_inst[2];
+#endif
+#ifdef CODEC_OPUS
+OpusEncInst* opus_inst[2];
 #endif
 
-int main(int argc, char* argv[])
-{
-    int packet_size, fs;
-    webrtc::NetEqDecoder usedCodec;
-    int payloadType;
-    int bitrate = 0;
-    int useVAD, vad;
-    int useRed=0;
-    int len, enc_len;
-    int16_t org_data[4000];
-    unsigned char rtp_data[8000];
-    int16_t seqNo=0xFFF;
-    uint32_t ssrc=1235412312;
-    uint32_t timestamp=0xAC1245;
-    uint16_t length, plen;
-    uint32_t offset;
-    double sendtime = 0;
-    int red_PT[2] = {0};
-    uint32_t red_TS[2] = {0};
-    uint16_t red_len[2] = {0};
-    int RTPheaderLen=12;
-    uint8_t red_data[8000];
+int main(int argc, char* argv[]) {
+  size_t packet_size;
+  int fs;
+  webrtc::NetEqDecoder usedCodec;
+  int payloadType;
+  int bitrate = 0;
+  int useVAD, vad;
+  int useRed = 0;
+  size_t len, enc_len;
+  int16_t org_data[4000];
+  unsigned char rtp_data[kRtpDataSize];
+  int16_t seqNo = 0xFFF;
+  uint32_t ssrc = 1235412312;
+  uint32_t timestamp = 0xAC1245;
+  uint16_t length, plen;
+  uint32_t offset;
+  double sendtime = 0;
+  int red_PT[2] = {0};
+  uint32_t red_TS[2] = {0};
+  uint16_t red_len[2] = {0};
+  size_t RTPheaderLen = 12;
+  uint8_t red_data[kRtpDataSize];
 #ifdef INSERT_OLD_PACKETS
-    uint16_t old_length, old_plen;
-    int old_enc_len;
-    int first_old_packet=1;
-    unsigned char old_rtp_data[8000];
-    int packet_age=0;
+  uint16_t old_length, old_plen;
+  size_t old_enc_len;
+  int first_old_packet = 1;
+  unsigned char old_rtp_data[kRtpDataSize];
+  size_t packet_age = 0;
 #endif
 #ifdef INSERT_DTMF_PACKETS
-    int NTone = 1;
-    int DTMFfirst = 1;
-    uint32_t DTMFtimestamp;
-    bool dtmfSent = false;
+  int NTone = 1;
+  int DTMFfirst = 1;
+  uint32_t DTMFtimestamp;
+  bool dtmfSent = false;
 #endif
-    bool usingStereo = false;
-    int stereoMode = 0;
-    int numChannels = 1;
+  bool usingStereo = false;
+  size_t stereoMode = 0;
+  size_t numChannels = 1;
 
-	/* check number of parameters */
-	if ((argc != 6) && (argc != 7)) {
-		/* print help text and exit */
-		printf("Application to encode speech into an RTP stream.\n");
-		printf("The program reads a PCM file and encodes is using the specified codec.\n");
-		printf("The coded speech is packetized in RTP packest and written to the output file.\n");
-		printf("The format of the RTP stream file is simlilar to that of rtpplay,\n");
-		printf("but with the receive time euqal to 0 for all packets.\n");
-		printf("Usage:\n\n");
-		printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
-		printf("where:\n");
+  /* check number of parameters */
+  if ((argc != 6) && (argc != 7)) {
+    /* print help text and exit */
+    printf("Application to encode speech into an RTP stream.\n");
+    printf("The program reads a PCM file and encodes is using the specified "
+           "codec.\n");
+    printf("The coded speech is packetized in RTP packest and written to the "
+           "output file.\n");
+    printf("The format of the RTP stream file is simlilar to that of "
+           "rtpplay,\n");
+    printf("but with the receive time euqal to 0 for all packets.\n");
+    printf("Usage:\n\n");
+    printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
+    printf("where:\n");
 
-		printf("PCMfile      : PCM speech input file\n\n");
+    printf("PCMfile      : PCM speech input file\n\n");
 
-		printf("RTPfile      : RTP stream output file\n\n");
+    printf("RTPfile      : RTP stream output file\n\n");
 
-		printf("frameLen     : 80...960...  Number of samples per packet (limit depends on codec)\n\n");
+    printf("frameLen     : 80...960...  Number of samples per packet (limit "
+           "depends on codec)\n\n");
 
-		printf("codecName\n");
+    printf("codecName\n");
 #ifdef CODEC_PCM16B
-		printf("             : pcm16b       16 bit PCM (8kHz)\n");
+    printf("             : pcm16b       16 bit PCM (8kHz)\n");
 #endif
 #ifdef CODEC_PCM16B_WB
-		printf("             : pcm16b_wb   16 bit PCM (16kHz)\n");
+    printf("             : pcm16b_wb   16 bit PCM (16kHz)\n");
 #endif
 #ifdef CODEC_PCM16B_32KHZ
-		printf("             : pcm16b_swb32 16 bit PCM (32kHz)\n");
+    printf("             : pcm16b_swb32 16 bit PCM (32kHz)\n");
 #endif
 #ifdef CODEC_PCM16B_48KHZ
-		printf("             : pcm16b_swb48 16 bit PCM (48kHz)\n");
+    printf("             : pcm16b_swb48 16 bit PCM (48kHz)\n");
 #endif
 #ifdef CODEC_G711
-		printf("             : pcma         g711 A-law (8kHz)\n");
+    printf("             : pcma         g711 A-law (8kHz)\n");
 #endif
 #ifdef CODEC_G711
-		printf("             : pcmu         g711 u-law (8kHz)\n");
+    printf("             : pcmu         g711 u-law (8kHz)\n");
 #endif
 #ifdef CODEC_G729
-		printf("             : g729         G729 (8kHz and 8kbps) CELP (One-Three frame(s)/packet)\n");
+    printf("             : g729         G729 (8kHz and 8kbps) CELP (One-Three "
+           "frame(s)/packet)\n");
 #endif
 #ifdef CODEC_G729_1
-		printf("             : g729.1       G729.1 (16kHz) variable rate (8--32 kbps)\n");
+    printf("             : g729.1       G729.1 (16kHz) variable rate (8--32 "
+           "kbps)\n");
 #endif
 #ifdef CODEC_G722_1_16
-		printf("             : g722.1_16    G722.1 coder (16kHz) (g722.1 with 16kbps)\n");
+    printf("             : g722.1_16    G722.1 coder (16kHz) (g722.1 with "
+           "16kbps)\n");
 #endif
 #ifdef CODEC_G722_1_24
-		printf("             : g722.1_24    G722.1 coder (16kHz) (the 24kbps version)\n");
+    printf("             : g722.1_24    G722.1 coder (16kHz) (the 24kbps "
+           "version)\n");
 #endif
 #ifdef CODEC_G722_1_32
-		printf("             : g722.1_32    G722.1 coder (16kHz) (the 32kbps version)\n");
+    printf("             : g722.1_32    G722.1 coder (16kHz) (the 32kbps "
+           "version)\n");
 #endif
 #ifdef CODEC_G722_1C_24
-		printf("             : g722.1C_24    G722.1 C coder (32kHz) (the 24kbps version)\n");
+    printf("             : g722.1C_24    G722.1 C coder (32kHz) (the 24kbps "
+           "version)\n");
 #endif
 #ifdef CODEC_G722_1C_32
-		printf("             : g722.1C_32    G722.1 C coder (32kHz) (the 32kbps version)\n");
+    printf("             : g722.1C_32    G722.1 C coder (32kHz) (the 32kbps "
+           "version)\n");
 #endif
 #ifdef CODEC_G722_1C_48
-		printf("             : g722.1C_48    G722.1 C coder (32kHz) (the 48kbps)\n");
+    printf("             : g722.1C_48    G722.1 C coder (32kHz) (the 48kbps "
+           "version)\n");
 #endif
 
 #ifdef CODEC_G726
-        printf("             : g726_16      G726 coder (8kHz) 16kbps\n");
-        printf("             : g726_24      G726 coder (8kHz) 24kbps\n");
-        printf("             : g726_32      G726 coder (8kHz) 32kbps\n");
-        printf("             : g726_40      G726 coder (8kHz) 40kbps\n");
+    printf("             : g726_16      G726 coder (8kHz) 16kbps\n");
+    printf("             : g726_24      G726 coder (8kHz) 24kbps\n");
+    printf("             : g726_32      G726 coder (8kHz) 32kbps\n");
+    printf("             : g726_40      G726 coder (8kHz) 40kbps\n");
 #endif
 #ifdef CODEC_AMR
-		printf("             : AMRXk        Adaptive Multi Rate CELP codec (8kHz)\n");
-		printf("                            X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 or 12.2\n");
+    printf("             : AMRXk        Adaptive Multi Rate CELP codec "
+           "(8kHz)\n");
+    printf("                            X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, "
+           "10.2 or 12.2\n");
 #endif
 #ifdef CODEC_AMRWB
-		printf("             : AMRwbXk      Adaptive Multi Rate Wideband CELP codec (16kHz)\n");
-		printf("                            X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n");
+    printf("             : AMRwbXk      Adaptive Multi Rate Wideband CELP "
+           "codec (16kHz)\n");
+    printf("                            X = 7, 9, 12, 14, 16, 18, 20, 23 or "
+           "24\n");
 #endif
 #ifdef CODEC_ILBC
-		printf("             : ilbc         iLBC codec (8kHz and 13.8kbps)\n");
+    printf("             : ilbc         iLBC codec (8kHz and 13.8kbps)\n");
 #endif
 #ifdef CODEC_ISAC
-		printf("             : isac         iSAC (16kHz and 32.0 kbps). To set rate specify a rate parameter as last parameter\n");
+    printf("             : isac         iSAC (16kHz and 32.0 kbps). To set "
+           "rate specify a rate parameter as last parameter\n");
 #endif
 #ifdef CODEC_ISAC_SWB
-		printf("             : isacswb       iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
+    printf("             : isacswb       iSAC SWB (32kHz and 32.0-52.0 kbps). "
+           "To set rate specify a rate parameter as last parameter\n");
 #endif
 #ifdef CODEC_GSMFR
-		printf("             : gsmfr        GSM FR codec (8kHz and 13kbps)\n");
+    printf("             : gsmfr        GSM FR codec (8kHz and 13kbps)\n");
 #endif
 #ifdef CODEC_G722
-		printf("             : g722         g722 coder (16kHz) (the 64kbps version)\n");
+    printf("             : g722         g722 coder (16kHz) (the 64kbps "
+           "version)\n");
 #endif
 #ifdef CODEC_SPEEX_8
-		printf("             : speex8       speex coder (8 kHz)\n");
+    printf("             : speex8       speex coder (8 kHz)\n");
 #endif
 #ifdef CODEC_SPEEX_16
-		printf("             : speex16      speex coder (16 kHz)\n");
+    printf("             : speex16      speex coder (16 kHz)\n");
 #endif
 #ifdef CODEC_RED
 #ifdef CODEC_G711
-		printf("             : red_pcm      Redundancy RTP packet with 2*G711A frames\n");
+    printf("             : red_pcm      Redundancy RTP packet with 2*G711A "
+           "frames\n");
 #endif
 #ifdef CODEC_ISAC
-		printf("             : red_isac     Redundancy RTP packet with 2*iSAC frames\n");
+    printf("             : red_isac     Redundancy RTP packet with 2*iSAC "
+           "frames\n");
 #endif
+#endif  // CODEC_RED
+#ifdef CODEC_OPUS
+    printf("             : opus         Opus codec with FEC (48kHz, 32kbps, FEC"
+        " on and tuned for 5%% packet losses)\n");
 #endif
-        printf("\n");
+    printf("\n");
 
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
-    defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-		printf("useVAD       : 0 Voice Activity Detection is switched off\n");
-		printf("             : 1 Voice Activity Detection is switched on\n\n");
+     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+    printf("useVAD       : 0 Voice Activity Detection is switched off\n");
+    printf("             : 1 Voice Activity Detection is switched on\n\n");
 #else
-		printf("useVAD       : 0 Voice Activity Detection switched off (on not supported)\n\n");
+    printf("useVAD       : 0 Voice Activity Detection switched off (on not "
+           "supported)\n\n");
 #endif
-		printf("bitrate      : Codec bitrate in bps (only applies to vbr codecs)\n\n");
+    printf("bitrate      : Codec bitrate in bps (only applies to vbr "
+           "codecs)\n\n");
 
-		return(0);
-	}
+    return (0);
+  }
 
-	FILE* in_file=fopen(argv[1],"rb");
-	CHECK_NOT_NULL(in_file);
-	printf("Input file: %s\n",argv[1]);
-	FILE* out_file=fopen(argv[2],"wb");
-	CHECK_NOT_NULL(out_file);
-	printf("Output file: %s\n\n",argv[2]);
-	packet_size=atoi(argv[3]);
-	CHECK_NOT_NULL(packet_size);
-	printf("Packet size: %i\n",packet_size);
+  FILE* in_file = fopen(argv[1], "rb");
+  CHECK_NOT_NULL(in_file);
+  printf("Input file: %s\n", argv[1]);
+  FILE* out_file = fopen(argv[2], "wb");
+  CHECK_NOT_NULL(out_file);
+  printf("Output file: %s\n\n", argv[2]);
+  int packet_size_int = atoi(argv[3]);
+  if (packet_size_int <= 0) {
+     printf("Packet size %d must be positive", packet_size_int);
+     return -1;
+  }
+  printf("Packet size: %d\n", packet_size_int);
+  packet_size = static_cast<size_t>(packet_size_int);
 
-    // check for stereo
-    if(argv[4][strlen(argv[4])-1] == '*') {
-        // use stereo
-        usingStereo = true;
-        numChannels = 2;
-        argv[4][strlen(argv[4])-1] = '\0';
-    }
+  // check for stereo
+  if (argv[4][strlen(argv[4]) - 1] == '*') {
+    // use stereo
+    usingStereo = true;
+    numChannels = 2;
+    argv[4][strlen(argv[4]) - 1] = '\0';
+  }
 
-	NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed);
+  NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs,
+                            &bitrate, &useRed);
 
-    if(useRed) {
-        RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant payload, except last one which is 1 byte */
-    }
+  if (useRed) {
+    RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant
+                                  payload, except last one which is 1 byte */
+  }
 
-	useVAD=atoi(argv[5]);
+  useVAD = atoi(argv[5]);
 #if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
-    defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-	if (useVAD!=0) {
-		printf("Error: this simulation does not support VAD/DTX/CNG\n");
-	}
+      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+  if (useVAD != 0) {
+    printf("Error: this simulation does not support VAD/DTX/CNG\n");
+  }
 #endif
-	
-    // check stereo type
-    if(usingStereo)
-    {
-        switch(usedCodec) 
-        {
-            // sample based codecs 
-        case webrtc::kDecoderPCMu:
-        case webrtc::kDecoderPCMa:
-        case webrtc::kDecoderG722:
-            {
-                // 1 octet per sample
-                stereoMode = STEREO_MODE_SAMPLE_1;
-                break;
-            }
-        case webrtc::kDecoderPCM16B:
-        case webrtc::kDecoderPCM16Bwb:
-        case webrtc::kDecoderPCM16Bswb32kHz:
-        case webrtc::kDecoderPCM16Bswb48kHz:
-            {
-                // 2 octets per sample
-                stereoMode = STEREO_MODE_SAMPLE_2;
-                break;
-            }
 
-            // fixed-rate frame codecs (with internal VAD)
-        default:
-            {
-                printf("Cannot use codec %s as stereo codec\n", argv[4]);
-                exit(0);
-            }
-        }
-    }
+  // check stereo type
+  if (usingStereo) {
+    switch (usedCodec) {
+      // sample based codecs
+      case webrtc::NetEqDecoder::kDecoderPCMu:
+      case webrtc::NetEqDecoder::kDecoderPCMa:
+      case webrtc::NetEqDecoder::kDecoderG722: {
+        // 1 octet per sample
+        stereoMode = STEREO_MODE_SAMPLE_1;
+        break;
+      }
+      case webrtc::NetEqDecoder::kDecoderPCM16B:
+      case webrtc::NetEqDecoder::kDecoderPCM16Bwb:
+      case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz:
+      case webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz: {
+        // 2 octets per sample
+        stereoMode = STEREO_MODE_SAMPLE_2;
+        break;
+      }
 
-	if ((usedCodec == webrtc::kDecoderISAC) || (usedCodec == webrtc::kDecoderISACswb))
-    {
-        if (argc != 7)
-        {
-            if (usedCodec == webrtc::kDecoderISAC)
-            {
-                bitrate = 32000;
-                printf(
-                    "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n");
-            }
-            else // (usedCodec==webrtc::kDecoderISACswb)
-            {
-                bitrate = 56000;
-                printf(
-                    "Running iSAC at default bitrate of 56000 bps (to specify explicitly add the bps as last parameter)\n");
-            }
-        }
-        else
-        {
-            bitrate = atoi(argv[6]);
-            if (usedCodec == webrtc::kDecoderISAC)
-            {
-                if ((bitrate < 10000) || (bitrate > 32000))
-                {
-                    printf(
-                        "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is invalid)\n",
-                        bitrate);
-                    exit(0);
-                }
-                printf("Running iSAC at bitrate of %i bps\n", bitrate);
-            }
-            else // (usedCodec==webrtc::kDecoderISACswb)
-            {
-                if ((bitrate < 32000) || (bitrate > 56000))
-                {
-                    printf(
-                        "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
-                        bitrate);
-                    exit(0);
-                }
-            }
-        }
+      // fixed-rate frame codecs (with internal VAD)
+      default: {
+        printf("Cannot use codec %s as stereo codec\n", argv[4]);
+        exit(0);
+      }
     }
-    else
-    {
-        if (argc == 7)
-        {
-            printf(
-                "Error: Bitrate parameter can only be specified for iSAC, G.723, and G.729.1\n");
-            exit(0);
-        }
-    }
-	
-    if(useRed) {
-        printf("Redundancy engaged. ");
-    }
-	printf("Used codec: %i\n",usedCodec);
-	printf("Payload type: %i\n",payloadType);
-	
-	NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels);
+  }
 
-	/* write file header */
-	//fprintf(out_file, "#!RTPencode%s\n", "1.0");
-	fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
-	uint32_t dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
-        if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
-          return -1;
+  if ((usedCodec == webrtc::NetEqDecoder::kDecoderISAC) ||
+      (usedCodec == webrtc::NetEqDecoder::kDecoderISACswb)) {
+    if (argc != 7) {
+      if (usedCodec == webrtc::NetEqDecoder::kDecoderISAC) {
+        bitrate = 32000;
+        printf("Running iSAC at default bitrate of 32000 bps (to specify "
+               "explicitly add the bps as last parameter)\n");
+      } else  // (usedCodec==webrtc::kDecoderISACswb)
+      {
+        bitrate = 56000;
+        printf("Running iSAC at default bitrate of 56000 bps (to specify "
+               "explicitly add the bps as last parameter)\n");
+      }
+    } else {
+      bitrate = atoi(argv[6]);
+      if (usedCodec == webrtc::NetEqDecoder::kDecoderISAC) {
+        if ((bitrate < 10000) || (bitrate > 32000)) {
+          printf("Error: iSAC bitrate must be between 10000 and 32000 bps (%i "
+                 "is invalid)\n", bitrate);
+          exit(0);
         }
-        if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
-          return -1;
-        }
-        if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
-          return -1;
-        }
-        if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
-          return -1;
-        }
-        if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
-          return -1;
+        printf("Running iSAC at bitrate of %i bps\n", bitrate);
+      } else  // (usedCodec==webrtc::kDecoderISACswb)
+      {
+        if ((bitrate < 32000) || (bitrate > 56000)) {
+          printf("Error: iSAC SWB bitrate must be between 32000 and 56000 bps "
+                 "(%i is invalid)\n", bitrate);
+          exit(0);
         }
+      }
+    }
+  } else {
+    if (argc == 7) {
+      printf("Error: Bitrate parameter can only be specified for iSAC, G.723, "
+             "and G.729.1\n");
+      exit(0);
+    }
+  }
+
+  if (useRed) {
+    printf("Redundancy engaged. ");
+  }
+  printf("Used codec: %i\n", static_cast<int>(usedCodec));
+  printf("Payload type: %i\n", payloadType);
+
+  NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD,
+                        numChannels);
+
+  /* write file header */
+  // fprintf(out_file, "#!RTPencode%s\n", "1.0");
+  fprintf(out_file, "#!rtpplay%s \n",
+          "1.0");               // this is the string that rtpplay needs
+  uint32_t dummy_variable = 0;  // should be converted to network endian format,
+                                // but does not matter when 0
+  if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
+    return -1;
+  }
+  if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
+    return -1;
+  }
+  if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
+    return -1;
+  }
+  if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
+    return -1;
+  }
+  if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
+    return -1;
+  }
 
 #ifdef TIMESTAMP_WRAPAROUND
-	timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */
+  timestamp = 0xFFFFFFFF - fs * 10; /* should give wrap-around in 10 seconds */
 #endif
 #if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
-	srand(RANDOM_SEED);
+  srand(RANDOM_SEED);
 #endif
 
-    /* if redundancy is used, the first redundant payload is zero length */
-    red_len[0] = 0;
+  /* if redundancy is used, the first redundant payload is zero length */
+  red_len[0] = 0;
 
-	/* read first frame */
-	len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
+  /* read first frame */
+  len = fread(org_data, 2, packet_size * numChannels, in_file) / numChannels;
 
-    /* de-interleave if stereo */
-    if ( usingStereo )
-    {
-        stereoDeInterleave(org_data, len * numChannels);
-    }
-
-	while (len==packet_size) {
+  /* de-interleave if stereo */
+  if (usingStereo) {
+    stereoDeInterleave(org_data, len * numChannels);
+  }
 
+  while (len == packet_size) {
 #ifdef INSERT_DTMF_PACKETS
-        dtmfSent = false;
+    dtmfSent = false;
 
-        if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) {
-            if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) {
-                // tone has not ended
-                if (DTMFfirst==1) {
-                    DTMFtimestamp = timestamp; // save this timestamp
-                    DTMFfirst=0;
-                }
-                makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
-                enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len);
-            }
-            else {
-                // tone has ended
-                makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
-                enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4, DTMF_DURATION*(fs/1000));
-                NTone++;
-                DTMFfirst=1;
-            }
-
-            /* write RTP packet to file */
-            length = htons(12 + enc_len + 8);
-            plen = htons(12 + enc_len);
-            offset = (uint32_t) sendtime; //(timestamp/(fs/1000));
-            offset = htonl(offset);
-            if (fwrite(&length, 2, 1, out_file) != 1) {
-              return -1;
-            }
-            if (fwrite(&plen, 2, 1, out_file) != 1) {
-              return -1;
-            }
-            if (fwrite(&offset, 4, 1, out_file) != 1) {
-              return -1;
-            }
-            if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
-              return -1;
-            }
-
-            dtmfSent = true;
+    if (sendtime >= NTone * DTMF_PACKET_INTERVAL) {
+      if (sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION) {
+        // tone has not ended
+        if (DTMFfirst == 1) {
+          DTMFtimestamp = timestamp;  // save this timestamp
+          DTMFfirst = 0;
         }
+        makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
+        enc_len = makeDTMFpayload(
+            &rtp_data[12], NTone % 12, 0, 4,
+            (int)(sendtime - NTone * DTMF_PACKET_INTERVAL) * (fs / 1000) + len);
+      } else {
+        // tone has ended
+        makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
+        enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4,
+                                  DTMF_DURATION * (fs / 1000));
+        NTone++;
+        DTMFfirst = 1;
+      }
+
+      /* write RTP packet to file */
+      length = htons(static_cast<unsigned short>(12 + enc_len + 8));
+      plen = htons(static_cast<unsigned short>(12 + enc_len));
+      offset = (uint32_t)sendtime;  //(timestamp/(fs/1000));
+      offset = htonl(offset);
+      if (fwrite(&length, 2, 1, out_file) != 1) {
+        return -1;
+      }
+      if (fwrite(&plen, 2, 1, out_file) != 1) {
+        return -1;
+      }
+      if (fwrite(&offset, 4, 1, out_file) != 1) {
+        return -1;
+      }
+      if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
+        return -1;
+      }
+
+      dtmfSent = true;
+    }
 #endif
 
 #ifdef NO_DTMF_OVERDUB
-        /* If DTMF is sent, we should not send any speech packets during the same time */
-        if (dtmfSent) {
-            enc_len = 0;
-        }
-        else {
+    /* If DTMF is sent, we should not send any speech packets during the same
+     * time */
+    if (dtmfSent) {
+      enc_len = 0;
+    } else {
 #endif
-		/* encode frame */
-		enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels);
-		if (enc_len==-1) {
-			printf("Error encoding frame\n");
-			exit(0);
-		}
+      /* encode frame */
+      enc_len =
+          NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12], fs,
+                           &vad, useVAD, bitrate, numChannels);
 
-        if ( usingStereo &&
-            stereoMode != STEREO_MODE_FRAME &&
-            vad == 1 )
-        {
-            // interleave the encoded payload for sample-based codecs (not for CNG)
-            stereoInterleave(&rtp_data[12], enc_len, stereoMode);
-        }
+      if (usingStereo && stereoMode != STEREO_MODE_FRAME && vad == 1) {
+        // interleave the encoded payload for sample-based codecs (not for CNG)
+        stereoInterleave(&rtp_data[12], enc_len, stereoMode);
+      }
 #ifdef NO_DTMF_OVERDUB
-        }
+    }
 #endif
-		
-		if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
-            if(useRed) {
-                if(red_len[0] > 0) {
-                    memmove(&rtp_data[RTPheaderLen+red_len[0]], &rtp_data[12], enc_len);
-                    memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
 
-                    red_len[1] = enc_len;
-                    red_TS[1] = timestamp;
-                    if(vad)
-                        red_PT[1] = payloadType;
-                    else
-                        red_PT[1] = NETEQ_CODEC_CN_PT;
+    if (enc_len > 0 &&
+        (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
+      if (useRed) {
+        if (red_len[0] > 0) {
+          memmove(&rtp_data[RTPheaderLen + red_len[0]], &rtp_data[12], enc_len);
+          memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
 
-                    makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
+          red_len[1] = static_cast<uint16_t>(enc_len);
+          red_TS[1] = timestamp;
+          if (vad)
+            red_PT[1] = payloadType;
+          else
+            red_PT[1] = NETEQ_CODEC_CN_PT;
 
+          makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++,
+                              ssrc);
 
-                    enc_len += red_len[0] + RTPheaderLen - 12;
-                }
-                else { // do not use redundancy payload for this packet, i.e., only last payload
-                    memmove(&rtp_data[RTPheaderLen-4], &rtp_data[12], enc_len);
-                    //memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
+          enc_len += red_len[0] + RTPheaderLen - 12;
+        } else {  // do not use redundancy payload for this packet, i.e., only
+                  // last payload
+          memmove(&rtp_data[RTPheaderLen - 4], &rtp_data[12], enc_len);
+          // memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
 
-                    red_len[1] = enc_len;
-                    red_TS[1] = timestamp;
-                    if(vad)
-                        red_PT[1] = payloadType;
-                    else
-                        red_PT[1] = NETEQ_CODEC_CN_PT;
+          red_len[1] = static_cast<uint16_t>(enc_len);
+          red_TS[1] = timestamp;
+          if (vad)
+            red_PT[1] = payloadType;
+          else
+            red_PT[1] = NETEQ_CODEC_CN_PT;
 
-                    makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
+          makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++,
+                              ssrc);
 
-
-                    enc_len += red_len[0] + RTPheaderLen - 4 - 12; // 4 is length of redundancy header (not used)
-                }
-            }
-            else {
-                
-                /* make RTP header */
-                if (vad) // regular speech data
-                    makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc);
-                else // CNG data
-                    makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc);
-                
-            }
+          enc_len += red_len[0] + RTPheaderLen - 4 -
+                     12;  // 4 is length of redundancy header (not used)
+        }
+      } else {
+        /* make RTP header */
+        if (vad)  // regular speech data
+          makeRTPheader(rtp_data, payloadType, seqNo++, timestamp, ssrc);
+        else  // CNG data
+          makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++, timestamp, ssrc);
+      }
 #ifdef MULTIPLE_SAME_TIMESTAMP
-			int mult_pack=0;
-			do {
-#endif //MULTIPLE_SAME_TIMESTAMP
-			/* write RTP packet to file */
-                          length = htons(12 + enc_len + 8);
-                          plen = htons(12 + enc_len);
-                          offset = (uint32_t) sendtime;
-                          //(timestamp/(fs/1000));
-                          offset = htonl(offset);
-                          if (fwrite(&length, 2, 1, out_file) != 1) {
-                            return -1;
-                          }
-                          if (fwrite(&plen, 2, 1, out_file) != 1) {
-                            return -1;
-                          }
-                          if (fwrite(&offset, 4, 1, out_file) != 1) {
-                            return -1;
-                          }
+      int mult_pack = 0;
+      do {
+#endif  // MULTIPLE_SAME_TIMESTAMP
+        /* write RTP packet to file */
+        length = htons(static_cast<unsigned short>(12 + enc_len + 8));
+        plen = htons(static_cast<unsigned short>(12 + enc_len));
+        offset = (uint32_t)sendtime;
+        //(timestamp/(fs/1000));
+        offset = htonl(offset);
+        if (fwrite(&length, 2, 1, out_file) != 1) {
+          return -1;
+        }
+        if (fwrite(&plen, 2, 1, out_file) != 1) {
+          return -1;
+        }
+        if (fwrite(&offset, 4, 1, out_file) != 1) {
+          return -1;
+        }
 #ifdef RANDOM_DATA
-			for (int k=0; k<12+enc_len; k++) {
-				rtp_data[k] = rand() + rand();
-			}
+        for (size_t k = 0; k < 12 + enc_len; k++) {
+          rtp_data[k] = rand() + rand();
+        }
 #endif
 #ifdef RANDOM_PAYLOAD_DATA
-			for (int k=12; k<12+enc_len; k++) {
-				rtp_data[k] = rand() + rand();
-			}
+        for (size_t k = 12; k < 12 + enc_len; k++) {
+          rtp_data[k] = rand() + rand();
+        }
 #endif
-                        if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
-                          return -1;
-                        }
+        if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
+          return -1;
+        }
 #ifdef MULTIPLE_SAME_TIMESTAMP
-			} while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) );
-#endif //MULTIPLE_SAME_TIMESTAMP
+      } while ((seqNo % REPEAT_PACKET_DISTANCE == 0) &&
+               (mult_pack++ < REPEAT_PACKET_COUNT));
+#endif  // MULTIPLE_SAME_TIMESTAMP
 
 #ifdef INSERT_OLD_PACKETS
-			if (packet_age >= OLD_PACKET*fs) {
-				if (!first_old_packet) {
-                                  // send the old packet
-                                  if (fwrite(&old_length, 2, 1,
-                                             out_file) != 1) {
-                                    return -1;
-                                  }
-                                  if (fwrite(&old_plen, 2, 1,
-                                             out_file) != 1) {
-                                    return -1;
-                                  }
-                                  if (fwrite(&offset, 4, 1,
-                                             out_file) != 1) {
-                                    return -1;
-                                  }
-                                  if (fwrite(old_rtp_data, 12 + old_enc_len,
-                                             1, out_file) != 1) {
-                                    return -1;
-                                  }
-				}
-				// store current packet as old
-				old_length=length;
-				old_plen=plen;
-				memcpy(old_rtp_data,rtp_data,12+enc_len);
-				old_enc_len=enc_len;
-				first_old_packet=0;
-				packet_age=0;
-
-			}
-			packet_age += packet_size;
-#endif
-			
-            if(useRed) {
-                /* move data to redundancy store */
-#ifdef CODEC_ISAC
-                if(usedCodec==webrtc::kDecoderISAC)
-                {
-                    assert(!usingStereo); // Cannot handle stereo yet
-                    red_len[0] =
-                        WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data);
-                }
-                else
-                {
-#endif
-                    memcpy(red_data, &rtp_data[RTPheaderLen+red_len[0]], enc_len);
-                    red_len[0]=red_len[1];
-#ifdef CODEC_ISAC
-                }
-#endif
-                red_TS[0]=red_TS[1];
-                red_PT[0]=red_PT[1];
-            }
-            
-		}
-
-		/* read next frame */
-        len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
-        /* de-interleave if stereo */
-        if ( usingStereo )
-        {
-            stereoDeInterleave(org_data, len * numChannels);
+      if (packet_age >= OLD_PACKET * fs) {
+        if (!first_old_packet) {
+          // send the old packet
+          if (fwrite(&old_length, 2, 1, out_file) != 1) {
+            return -1;
+          }
+          if (fwrite(&old_plen, 2, 1, out_file) != 1) {
+            return -1;
+          }
+          if (fwrite(&offset, 4, 1, out_file) != 1) {
+            return -1;
+          }
+          if (fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file) != 1) {
+            return -1;
+          }
         }
+        // store current packet as old
+        old_length = length;
+        old_plen = plen;
+        memcpy(old_rtp_data, rtp_data, 12 + enc_len);
+        old_enc_len = enc_len;
+        first_old_packet = 0;
+        packet_age = 0;
+      }
+      packet_age += packet_size;
+#endif
 
-        if (payloadType==NETEQ_CODEC_G722_PT)
-            timestamp+=len>>1;
-        else
-            timestamp+=len;
+      if (useRed) {
+/* move data to redundancy store */
+#ifdef CODEC_ISAC
+        if (usedCodec == webrtc::NetEqDecoder::kDecoderISAC) {
+          assert(!usingStereo);  // Cannot handle stereo yet
+          red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data);
+        } else {
+#endif
+          memcpy(red_data, &rtp_data[RTPheaderLen + red_len[0]], enc_len);
+          red_len[0] = red_len[1];
+#ifdef CODEC_ISAC
+        }
+#endif
+        red_TS[0] = red_TS[1];
+        red_PT[0] = red_PT[1];
+      }
+    }
 
-		sendtime += (double) len/(fs/1000);
-	}
-	
-	NetEQTest_free_coders(usedCodec, numChannels);
-	fclose(in_file);
-	fclose(out_file);
-    printf("Done!\n");
+    /* read next frame */
+    len = fread(org_data, 2, packet_size * numChannels, in_file) / numChannels;
+    /* de-interleave if stereo */
+    if (usingStereo) {
+      stereoDeInterleave(org_data, len * numChannels);
+    }
 
-	return(0);
+    if (payloadType == NETEQ_CODEC_G722_PT)
+      timestamp += len >> 1;
+    else
+      timestamp += len;
+
+    sendtime += (double)len / (fs / 1000);
+  }
+
+  NetEQTest_free_coders(usedCodec, numChannels);
+  fclose(in_file);
+  fclose(out_file);
+  printf("Done!\n");
+
+  return (0);
 }
 
-
-
-
 /****************/
 /* Subfunctions */
 /****************/
@@ -824,1002 +838,1050 @@ int main(int argc, char* argv[])
 void NetEQTest_GetCodec_and_PT(char* name,
                                webrtc::NetEqDecoder* codec,
                                int* PT,
-                               int frameLen,
+                               size_t frameLen,
                                int* fs,
                                int* bitrate,
                                int* useRed) {
+  *bitrate = 0; /* Default bitrate setting */
+  *useRed = 0;  /* Default no redundancy */
 
-	*bitrate = 0; /* Default bitrate setting */
-    *useRed = 0; /* Default no redundancy */
-
-	if(!strcmp(name,"pcmu")){
-		*codec=webrtc::kDecoderPCMu;
-		*PT=NETEQ_CODEC_PCMU_PT;
-		*fs=8000;
-	}
-	else if(!strcmp(name,"pcma")){
-		*codec=webrtc::kDecoderPCMa;
-		*PT=NETEQ_CODEC_PCMA_PT;
-		*fs=8000;
-	}
-	else if(!strcmp(name,"pcm16b")){
-		*codec=webrtc::kDecoderPCM16B;
-		*PT=NETEQ_CODEC_PCM16B_PT;
-		*fs=8000;
-	}
-	else if(!strcmp(name,"pcm16b_wb")){
-		*codec=webrtc::kDecoderPCM16Bwb;
-		*PT=NETEQ_CODEC_PCM16B_WB_PT;
-		*fs=16000;
-	}
-	else if(!strcmp(name,"pcm16b_swb32")){
-		*codec=webrtc::kDecoderPCM16Bswb32kHz;
-		*PT=NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
-		*fs=32000;
-	}
-	else if(!strcmp(name,"pcm16b_swb48")){
-		*codec=webrtc::kDecoderPCM16Bswb48kHz;
-		*PT=NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
-		*fs=48000;
-	}
-	else if(!strcmp(name,"g722")){
-		*codec=webrtc::kDecoderG722;
-		*PT=NETEQ_CODEC_G722_PT;
-		*fs=16000;
-	}
-	else if((!strcmp(name,"ilbc"))&&((frameLen%240==0)||(frameLen%160==0))){
-		*fs=8000;
-		*codec=webrtc::kDecoderILBC;
-		*PT=NETEQ_CODEC_ILBC_PT;
-	}
-	else if(!strcmp(name,"isac")){
-		*fs=16000;
-		*codec=webrtc::kDecoderISAC;
-		*PT=NETEQ_CODEC_ISAC_PT;
-	}
-    else if(!strcmp(name,"isacswb")){
-		*fs=32000;
-		*codec=webrtc::kDecoderISACswb;
-		*PT=NETEQ_CODEC_ISACSWB_PT;
-	}
-    else if(!strcmp(name,"red_pcm")){
-		*codec=webrtc::kDecoderPCMa;
-		*PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
-		*fs=8000;
-        *useRed = 1;
-	} else if(!strcmp(name,"red_isac")){
-		*codec=webrtc::kDecoderISAC;
-		*PT=NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
-		*fs=16000;
-        *useRed = 1;
-    } else {
-		printf("Error: Not a supported codec (%s)\n", name);
-		exit(0);
-	}
-
+  if (!strcmp(name, "pcmu")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCMu;
+    *PT = NETEQ_CODEC_PCMU_PT;
+    *fs = 8000;
+  } else if (!strcmp(name, "pcma")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCMa;
+    *PT = NETEQ_CODEC_PCMA_PT;
+    *fs = 8000;
+  } else if (!strcmp(name, "pcm16b")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCM16B;
+    *PT = NETEQ_CODEC_PCM16B_PT;
+    *fs = 8000;
+  } else if (!strcmp(name, "pcm16b_wb")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCM16Bwb;
+    *PT = NETEQ_CODEC_PCM16B_WB_PT;
+    *fs = 16000;
+  } else if (!strcmp(name, "pcm16b_swb32")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz;
+    *PT = NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
+    *fs = 32000;
+  } else if (!strcmp(name, "pcm16b_swb48")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz;
+    *PT = NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
+    *fs = 48000;
+  } else if (!strcmp(name, "g722")) {
+    *codec = webrtc::NetEqDecoder::kDecoderG722;
+    *PT = NETEQ_CODEC_G722_PT;
+    *fs = 16000;
+  } else if ((!strcmp(name, "ilbc")) &&
+             ((frameLen % 240 == 0) || (frameLen % 160 == 0))) {
+    *fs = 8000;
+    *codec = webrtc::NetEqDecoder::kDecoderILBC;
+    *PT = NETEQ_CODEC_ILBC_PT;
+  } else if (!strcmp(name, "isac")) {
+    *fs = 16000;
+    *codec = webrtc::NetEqDecoder::kDecoderISAC;
+    *PT = NETEQ_CODEC_ISAC_PT;
+  } else if (!strcmp(name, "isacswb")) {
+    *fs = 32000;
+    *codec = webrtc::NetEqDecoder::kDecoderISACswb;
+    *PT = NETEQ_CODEC_ISACSWB_PT;
+  } else if (!strcmp(name, "red_pcm")) {
+    *codec = webrtc::NetEqDecoder::kDecoderPCMa;
+    *PT = NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
+    *fs = 8000;
+    *useRed = 1;
+  } else if (!strcmp(name, "red_isac")) {
+    *codec = webrtc::NetEqDecoder::kDecoderISAC;
+    *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
+    *fs = 16000;
+    *useRed = 1;
+  } else if (!strcmp(name, "opus")) {
+    *codec = webrtc::NetEqDecoder::kDecoderOpus;
+    *PT = NETEQ_CODEC_OPUS_PT; /* this will be the PT for the sub-headers */
+    *fs = 48000;
+  } else {
+    printf("Error: Not a supported codec (%s)\n", name);
+    exit(0);
+  }
 }
 
+int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
+                          size_t enc_frameSize,
+                          int bitrate,
+                          int sampfreq,
+                          int vad,
+                          size_t numChannels) {
+  int ok = 0;
 
-
-
-int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels){
-	
-	int ok=0;
-	
-    for (int k = 0; k < numChannels; k++) 
-    {
-        ok=WebRtcVad_Create(&VAD_inst[k]);
-        if (ok!=0) {
-            printf("Error: Couldn't allocate memory for VAD instance\n");
-            exit(0);
-        }
-        ok=WebRtcVad_Init(VAD_inst[k]);
-        if (ok==-1) {
-            printf("Error: Initialization of VAD struct failed\n");	
-            exit(0); 
-        }
-
+  for (size_t k = 0; k < numChannels; k++) {
+    VAD_inst[k] = WebRtcVad_Create();
+    if (!VAD_inst[k]) {
+      printf("Error: Couldn't allocate memory for VAD instance\n");
+      exit(0);
+    }
+    ok = WebRtcVad_Init(VAD_inst[k]);
+    if (ok == -1) {
+      printf("Error: Initialization of VAD struct failed\n");
+      exit(0);
+    }
 
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
-    defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-        ok=WebRtcCng_CreateEnc(&CNGenc_inst[k]);
-        if (ok!=0) {
-            printf("Error: Couldn't allocate memory for CNG encoding instance\n");
-            exit(0);
-        }
-        if(sampfreq <= 16000) {
-            ok=WebRtcCng_InitEnc(CNGenc_inst[k],sampfreq, 200, 5);
-            if (ok==-1) {
-                printf("Error: Initialization of CNG struct failed. Error code %d\n", 
-                    WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));	
-                exit(0); 
-            }
-        }
+     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+    ok = WebRtcCng_CreateEnc(&CNGenc_inst[k]);
+    if (ok != 0) {
+      printf("Error: Couldn't allocate memory for CNG encoding instance\n");
+      exit(0);
+    }
+    if (sampfreq <= 16000) {
+      ok = WebRtcCng_InitEnc(CNGenc_inst[k], sampfreq, 200, 5);
+      if (ok == -1) {
+        printf("Error: Initialization of CNG struct failed. Error code %d\n",
+               WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
+        exit(0);
+      }
+    }
 #endif
 
-        switch (coder) {
+    switch (coder) {
 #ifdef CODEC_PCM16B
-    case webrtc::kDecoderPCM16B :
+      case webrtc::NetEqDecoder::kDecoderPCM16B:
 #endif
 #ifdef CODEC_PCM16B_WB
-    case webrtc::kDecoderPCM16Bwb :
+      case webrtc::NetEqDecoder::kDecoderPCM16Bwb:
 #endif
 #ifdef CODEC_PCM16B_32KHZ
-    case webrtc::kDecoderPCM16Bswb32kHz :
+      case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz:
 #endif
 #ifdef CODEC_PCM16B_48KHZ
-    case webrtc::kDecoderPCM16Bswb48kHz :
+      case webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz:
 #endif
 #ifdef CODEC_G711
-    case webrtc::kDecoderPCMu :
-    case webrtc::kDecoderPCMa :
+      case webrtc::NetEqDecoder::kDecoderPCMu:
+      case webrtc::NetEqDecoder::kDecoderPCMa:
 #endif
         // do nothing
         break;
 #ifdef CODEC_G729
-    case webrtc::kDecoderG729:
-        if (sampfreq==8000) {
-            if ((enc_frameSize==80)||(enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==400)||(enc_frameSize==480)) {
-                ok=WebRtcG729_CreateEnc(&G729enc_inst[k]);
-                if (ok!=0) {
-                    printf("Error: Couldn't allocate memory for G729 encoding instance\n");
-                    exit(0);
-                }
-            } else {
-                printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 ms!!\n\n");
-                exit(0);
+      case webrtc::kDecoderG729:
+        if (sampfreq == 8000) {
+          if ((enc_frameSize == 80) || (enc_frameSize == 160) ||
+              (enc_frameSize == 240) || (enc_frameSize == 320) ||
+              (enc_frameSize == 400) || (enc_frameSize == 480)) {
+            ok = WebRtcG729_CreateEnc(&G729enc_inst[k]);
+            if (ok != 0) {
+              printf("Error: Couldn't allocate memory for G729 encoding "
+                     "instance\n");
+              exit(0);
             }
-            WebRtcG729_EncoderInit(G729enc_inst[k], vad);
-            if ((vad==1)&&(enc_frameSize!=80)) {
-                printf("\nError - This simulation only supports VAD for G729 at 10ms packets (not %dms)\n", (enc_frameSize>>3));
-            }
-        } else {
-            printf("\nError - g729 is only developed for 8kHz \n");
+          } else {
+            printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 "
+                   "ms!!\n\n");
             exit(0);
+          }
+          WebRtcG729_EncoderInit(G729enc_inst[k], vad);
+          if ((vad == 1) && (enc_frameSize != 80)) {
+            printf("\nError - This simulation only supports VAD for G729 at "
+                   "10ms packets (not %" PRIuS "ms)\n", (enc_frameSize >> 3));
+          }
+        } else {
+          printf("\nError - g729 is only developed for 8kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G729_1
-    case webrtc::kDecoderG729_1:
-        if (sampfreq==16000) {
-            if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)
-                ) {
-                    ok=WebRtcG7291_Create(&G729_1_inst[k]);
-                    if (ok!=0) {
-                        printf("Error: Couldn't allocate memory for G.729.1 codec instance\n");
-                        exit(0);
-                    }
-                } else {
-                    printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
-                    exit(0);
-                }
-                if (!(((bitrate >= 12000) && (bitrate <= 32000) && (bitrate%2000 == 0)) || (bitrate == 8000))) {
-                    /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
-                    printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in steps of 2000 bps\n");
-                    exit(0);
-                }
-                WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/, 0 /*flagG729mode*/);
-        } else {
-            printf("\nError - G.729.1 input is always 16 kHz \n");
+      case webrtc::kDecoderG729_1:
+        if (sampfreq == 16000) {
+          if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
+              (enc_frameSize == 960)) {
+            ok = WebRtcG7291_Create(&G729_1_inst[k]);
+            if (ok != 0) {
+              printf("Error: Couldn't allocate memory for G.729.1 codec "
+                     "instance\n");
+              exit(0);
+            }
+          } else {
+            printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
             exit(0);
+          }
+          if (!(((bitrate >= 12000) && (bitrate <= 32000) &&
+                 (bitrate % 2000 == 0)) ||
+                (bitrate == 8000))) {
+            /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
+            printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in "
+                   "steps of 2000 bps\n");
+            exit(0);
+          }
+          WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/,
+                                  0 /*flagG729mode*/);
+        } else {
+          printf("\nError - G.729.1 input is always 16 kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_SPEEX_8
-    case webrtc::kDecoderSPEEX_8 :
-        if (sampfreq==8000) {
-            if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
-                ok=WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
-                if (ok!=0) {
-                    printf("Error: Couldn't allocate memory for Speex encoding instance\n");
-                    exit(0);
-                }
-            } else {
-                printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
-                exit(0);
+      case webrtc::kDecoderSPEEX_8:
+        if (sampfreq == 8000) {
+          if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
+              (enc_frameSize == 480)) {
+            ok = WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
+            if (ok != 0) {
+              printf("Error: Couldn't allocate memory for Speex encoding "
+                     "instance\n");
+              exit(0);
             }
-            if ((vad==1)&&(enc_frameSize!=160)) {
-                printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>3));
-                vad=0;
-            }
-            ok=WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
-            if (ok!=0) exit(0);
+          } else {
+            printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
+            exit(0);
+          }
+          if ((vad == 1) && (enc_frameSize != 160)) {
+            printf("\nError - This simulation only supports VAD for Speex at "
+                   "20ms packets (not %" PRIuS "ms)\n",
+                (enc_frameSize >> 3));
+            vad = 0;
+          }
+          ok = WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0 /*vbr*/,
+                                       3 /*complexity*/, vad);
+          if (ok != 0)
+            exit(0);
         } else {
-            printf("\nError - Speex8 called with sample frequency other than 8 kHz.\n\n");
+          printf("\nError - Speex8 called with sample frequency other than 8 "
+                 "kHz.\n\n");
         }
         break;
 #endif
 #ifdef CODEC_SPEEX_16
-    case webrtc::kDecoderSPEEX_16 :
-        if (sampfreq==16000) {
-            if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)) {
-                ok=WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
-                if (ok!=0) {
-                    printf("Error: Couldn't allocate memory for Speex encoding instance\n");
-                    exit(0);
-                }
-            } else {
-                printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
-                exit(0);
+      case webrtc::kDecoderSPEEX_16:
+        if (sampfreq == 16000) {
+          if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
+              (enc_frameSize == 960)) {
+            ok = WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
+            if (ok != 0) {
+              printf("Error: Couldn't allocate memory for Speex encoding "
+                     "instance\n");
+              exit(0);
             }
-            if ((vad==1)&&(enc_frameSize!=320)) {
-                printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>4));
-                vad=0;
-            }
-            ok=WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
-            if (ok!=0) exit(0);
+          } else {
+            printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
+            exit(0);
+          }
+          if ((vad == 1) && (enc_frameSize != 320)) {
+            printf("\nError - This simulation only supports VAD for Speex at "
+                   "20ms packets (not %" PRIuS "ms)\n",
+                (enc_frameSize >> 4));
+            vad = 0;
+          }
+          ok = WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0 /*vbr*/,
+                                       3 /*complexity*/, vad);
+          if (ok != 0)
+            exit(0);
         } else {
-            printf("\nError - Speex16 called with sample frequency other than 16 kHz.\n\n");
+          printf("\nError - Speex16 called with sample frequency other than 16 "
+                 "kHz.\n\n");
         }
         break;
 #endif
 
 #ifdef CODEC_G722_1_16
-    case webrtc::kDecoderG722_1_16 :
-        if (sampfreq==16000) {
-            ok=WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for G.722.1 instance\n");
-                exit(0);
-            }
-            if (enc_frameSize==320) {				
-            } else {
-                printf("\nError: G722.1 only supports 20 ms!!\n\n");
-                exit(0);
-            }
-            WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
-        } else {
-            printf("\nError - G722.1 is only developed for 16kHz \n");
+      case webrtc::kDecoderG722_1_16:
+        if (sampfreq == 16000) {
+          ok = WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for G.722.1 instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 320) {
+          } else {
+            printf("\nError: G722.1 only supports 20 ms!!\n\n");
+            exit(0);
+          }
+          WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
+        } else {
+          printf("\nError - G722.1 is only developed for 16kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G722_1_24
-    case webrtc::kDecoderG722_1_24 :
-        if (sampfreq==16000) {
-            ok=WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for G.722.1 instance\n");
-                exit(0);
-            }
-            if (enc_frameSize==320) {
-            } else {
-                printf("\nError: G722.1 only supports 20 ms!!\n\n");
-                exit(0);
-            }
-            WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
-        } else {
-            printf("\nError - G722.1 is only developed for 16kHz \n");
+      case webrtc::kDecoderG722_1_24:
+        if (sampfreq == 16000) {
+          ok = WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for G.722.1 instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 320) {
+          } else {
+            printf("\nError: G722.1 only supports 20 ms!!\n\n");
+            exit(0);
+          }
+          WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
+        } else {
+          printf("\nError - G722.1 is only developed for 16kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G722_1_32
-    case webrtc::kDecoderG722_1_32 :
-        if (sampfreq==16000) {
-            ok=WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for G.722.1 instance\n");
-                exit(0);
-            }
-            if (enc_frameSize==320) {
-            } else {
-                printf("\nError: G722.1 only supports 20 ms!!\n\n");
-                exit(0);
-            }
-            WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
-        } else {
-            printf("\nError - G722.1 is only developed for 16kHz \n");
+      case webrtc::kDecoderG722_1_32:
+        if (sampfreq == 16000) {
+          ok = WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for G.722.1 instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 320) {
+          } else {
+            printf("\nError: G722.1 only supports 20 ms!!\n\n");
+            exit(0);
+          }
+          WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
+        } else {
+          printf("\nError - G722.1 is only developed for 16kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G722_1C_24
-    case webrtc::kDecoderG722_1C_24 :
-        if (sampfreq==32000) {
-            ok=WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for G.722.1C instance\n");
-                exit(0);
-            }
-            if (enc_frameSize==640) {
-            } else {
-                printf("\nError: G722.1 C only supports 20 ms!!\n\n");
-                exit(0);
-            }
-            WebRtcG7221C_EncoderInit24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
-        } else {
-            printf("\nError - G722.1 C is only developed for 32kHz \n");
+      case webrtc::kDecoderG722_1C_24:
+        if (sampfreq == 32000) {
+          ok = WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for G.722.1C instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 640) {
+          } else {
+            printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+            exit(0);
+          }
+          WebRtcG7221C_EncoderInit24(
+              (G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
+        } else {
+          printf("\nError - G722.1 C is only developed for 32kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G722_1C_32
-    case webrtc::kDecoderG722_1C_32 :
-        if (sampfreq==32000) {
-            ok=WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for G.722.1C instance\n");
-                exit(0);
-            }
-            if (enc_frameSize==640) {
-            } else {
-                printf("\nError: G722.1 C only supports 20 ms!!\n\n");
-                exit(0);
-            }
-            WebRtcG7221C_EncoderInit32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
-        } else {
-            printf("\nError - G722.1 C is only developed for 32kHz \n");
+      case webrtc::kDecoderG722_1C_32:
+        if (sampfreq == 32000) {
+          ok = WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for G.722.1C instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 640) {
+          } else {
+            printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+            exit(0);
+          }
+          WebRtcG7221C_EncoderInit32(
+              (G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
+        } else {
+          printf("\nError - G722.1 C is only developed for 32kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G722_1C_48
-    case webrtc::kDecoderG722_1C_48 :
-        if (sampfreq==32000) {
-            ok=WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for G.722.1C instance\n");
-                exit(0);
-            }
-            if (enc_frameSize==640) {
-            } else {
-                printf("\nError: G722.1 C only supports 20 ms!!\n\n");
-                exit(0);
-            }
-            WebRtcG7221C_EncoderInit48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
-        } else {
-            printf("\nError - G722.1 C is only developed for 32kHz \n");
+      case webrtc::kDecoderG722_1C_48:
+        if (sampfreq == 32000) {
+          ok = WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for G.722.1C instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 640) {
+          } else {
+            printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+            exit(0);
+          }
+          WebRtcG7221C_EncoderInit48(
+              (G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
+        } else {
+          printf("\nError - G722.1 C is only developed for 32kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_G722
-    case webrtc::kDecoderG722 :
-        if (sampfreq==16000) {
-            if (enc_frameSize%2==0) {				
-            } else {
-                printf("\nError - g722 frames must have an even number of enc_frameSize\n");
-                exit(0);
-            }
-            WebRtcG722_CreateEncoder(&g722EncState[k]);
-            WebRtcG722_EncoderInit(g722EncState[k]);
-        } else {
-            printf("\nError - g722 is only developed for 16kHz \n");
+      case webrtc::NetEqDecoder::kDecoderG722:
+        if (sampfreq == 16000) {
+          if (enc_frameSize % 2 == 0) {
+          } else {
+            printf(
+                "\nError - g722 frames must have an even number of "
+                "enc_frameSize\n");
             exit(0);
+          }
+          WebRtcG722_CreateEncoder(&g722EncState[k]);
+          WebRtcG722_EncoderInit(g722EncState[k]);
+        } else {
+          printf("\nError - g722 is only developed for 16kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_AMR
-    case webrtc::kDecoderAMR :
-        if (sampfreq==8000) {
-            ok=WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for AMR encoding instance\n");
-                exit(0);
-            }if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {				
-            } else {
-                printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
-                exit(0);
-            }
-            WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
-            WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
-            AMR_bitrate = bitrate;
-        } else {
-            printf("\nError - AMR is only developed for 8kHz \n");
+      case webrtc::kDecoderAMR:
+        if (sampfreq == 8000) {
+          ok = WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
+          if (ok != 0) {
+            printf(
+                "Error: Couldn't allocate memory for AMR encoding instance\n");
             exit(0);
+          }
+          if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
+              (enc_frameSize == 480)) {
+          } else {
+            printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
+            exit(0);
+          }
+          WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
+          WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
+          AMR_bitrate = bitrate;
+        } else {
+          printf("\nError - AMR is only developed for 8kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_AMRWB
-    case webrtc::kDecoderAMRWB :
-        if (sampfreq==16000) {
-            ok=WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for AMRWB encoding instance\n");
-                exit(0);
-            }
-            if (((enc_frameSize/320)<0)||((enc_frameSize/320)>3)||((enc_frameSize%320)!=0)) {
-                printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
-                exit(0);
-            }
-            WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
-            if (bitrate==7000) {
-                AMRWB_bitrate = AMRWB_MODE_7k;
-            } else if (bitrate==9000) {
-                AMRWB_bitrate = AMRWB_MODE_9k;
-            } else if (bitrate==12000) {
-                AMRWB_bitrate = AMRWB_MODE_12k;
-            } else if (bitrate==14000) {
-                AMRWB_bitrate = AMRWB_MODE_14k;
-            } else if (bitrate==16000) {
-                AMRWB_bitrate = AMRWB_MODE_16k;
-            } else if (bitrate==18000) {
-                AMRWB_bitrate = AMRWB_MODE_18k;
-            } else if (bitrate==20000) {
-                AMRWB_bitrate = AMRWB_MODE_20k;
-            } else if (bitrate==23000) {
-                AMRWB_bitrate = AMRWB_MODE_23k;
-            } else if (bitrate==24000) {
-                AMRWB_bitrate = AMRWB_MODE_24k;
-            }
-            WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
+      case webrtc::kDecoderAMRWB:
+        if (sampfreq == 16000) {
+          ok = WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for AMRWB encoding "
+                   "instance\n");
+            exit(0);
+          }
+          if (((enc_frameSize / 320) > 3) || ((enc_frameSize % 320) != 0)) {
+            printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
+            exit(0);
+          }
+          WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
+          if (bitrate == 7000) {
+            AMRWB_bitrate = AMRWB_MODE_7k;
+          } else if (bitrate == 9000) {
+            AMRWB_bitrate = AMRWB_MODE_9k;
+          } else if (bitrate == 12000) {
+            AMRWB_bitrate = AMRWB_MODE_12k;
+          } else if (bitrate == 14000) {
+            AMRWB_bitrate = AMRWB_MODE_14k;
+          } else if (bitrate == 16000) {
+            AMRWB_bitrate = AMRWB_MODE_16k;
+          } else if (bitrate == 18000) {
+            AMRWB_bitrate = AMRWB_MODE_18k;
+          } else if (bitrate == 20000) {
+            AMRWB_bitrate = AMRWB_MODE_20k;
+          } else if (bitrate == 23000) {
+            AMRWB_bitrate = AMRWB_MODE_23k;
+          } else if (bitrate == 24000) {
+            AMRWB_bitrate = AMRWB_MODE_24k;
+          }
+          WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
 
         } else {
-            printf("\nError - AMRwb is only developed for 16kHz \n");
-            exit(0);
+          printf("\nError - AMRwb is only developed for 16kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_ILBC
-    case webrtc::kDecoderILBC :
-        if (sampfreq==8000) {
-            ok=WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for iLBC encoding instance\n");
-                exit(0);
-            }
-            if ((enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==480)) {				
-            } else {
-                printf("\nError - iLBC only supports 160, 240, 320 and 480 enc_frameSize (20, 30, 40 and 60 ms)\n");
-                exit(0);
-            }
-            if ((enc_frameSize==160)||(enc_frameSize==320)) {
-                /* 20 ms version */
-                WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
-            } else {
-                /* 30 ms version */
-                WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
-            }
-        } else {
-            printf("\nError - iLBC is only developed for 8kHz \n");
+      case webrtc::NetEqDecoder::kDecoderILBC:
+        if (sampfreq == 8000) {
+          ok = WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for iLBC encoding "
+                   "instance\n");
             exit(0);
+          }
+          if ((enc_frameSize == 160) || (enc_frameSize == 240) ||
+              (enc_frameSize == 320) || (enc_frameSize == 480)) {
+          } else {
+            printf("\nError - iLBC only supports 160, 240, 320 and 480 "
+                   "enc_frameSize (20, 30, 40 and 60 ms)\n");
+            exit(0);
+          }
+          if ((enc_frameSize == 160) || (enc_frameSize == 320)) {
+            /* 20 ms version */
+            WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
+          } else {
+            /* 30 ms version */
+            WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
+          }
+        } else {
+          printf("\nError - iLBC is only developed for 8kHz \n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_ISAC
-    case webrtc::kDecoderISAC:
-        if (sampfreq==16000) {
-            ok=WebRtcIsac_Create(&ISAC_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for iSAC instance\n");
-                exit(0);
-            }if ((enc_frameSize==480)||(enc_frameSize==960)) {
-            } else {
-                printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
-                exit(0);
-            }
-            WebRtcIsac_EncoderInit(ISAC_inst[k],1);
-            if ((bitrate<10000)||(bitrate>32000)) {
-                printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
-                exit(0);
-            }
-            WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
-        } else {
-            printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
+      case webrtc::NetEqDecoder::kDecoderISAC:
+        if (sampfreq == 16000) {
+          ok = WebRtcIsac_Create(&ISAC_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for iSAC instance\n");
             exit(0);
+          }
+          if ((enc_frameSize == 480) || (enc_frameSize == 960)) {
+          } else {
+            printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
+            exit(0);
+          }
+          WebRtcIsac_EncoderInit(ISAC_inst[k], 1);
+          if ((bitrate < 10000) || (bitrate > 32000)) {
+            printf("\nError - iSAC bitrate has to be between 10000 and 32000 "
+                   "bps (not %i)\n",
+                bitrate);
+            exit(0);
+          }
+          WebRtcIsac_Control(ISAC_inst[k], bitrate,
+                             static_cast<int>(enc_frameSize >> 4));
+        } else {
+          printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or "
+                 "60 ms)\n");
+          exit(0);
         }
         break;
 #endif
 #ifdef NETEQ_ISACFIX_CODEC
-    case webrtc::kDecoderISAC:
-        if (sampfreq==16000) {
-            ok=WebRtcIsacfix_Create(&ISAC_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for iSAC instance\n");
-                exit(0);
-            }if ((enc_frameSize==480)||(enc_frameSize==960)) {
-            } else {
-                printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
-                exit(0);
-            }
-            WebRtcIsacfix_EncoderInit(ISAC_inst[k],1);
-            if ((bitrate<10000)||(bitrate>32000)) {
-                printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
-                exit(0);
-            }
-            WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
-        } else {
-            printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
+      case webrtc::kDecoderISAC:
+        if (sampfreq == 16000) {
+          ok = WebRtcIsacfix_Create(&ISAC_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for iSAC instance\n");
             exit(0);
+          }
+          if ((enc_frameSize == 480) || (enc_frameSize == 960)) {
+          } else {
+            printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
+            exit(0);
+          }
+          WebRtcIsacfix_EncoderInit(ISAC_inst[k], 1);
+          if ((bitrate < 10000) || (bitrate > 32000)) {
+            printf("\nError - iSAC bitrate has to be between 10000 and 32000 "
+                   "bps (not %i)\n", bitrate);
+            exit(0);
+          }
+          WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize >> 4);
+        } else {
+          printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or "
+                 "60 ms)\n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_ISAC_SWB
-    case webrtc::kDecoderISACswb:
-        if (sampfreq==32000) {
-            ok=WebRtcIsac_Create(&ISACSWB_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
-                exit(0);
-            }if (enc_frameSize==960) {
-            } else {
-                printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
-                exit(0);
-            }
-            ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
-            if (ok!=0) {
-                printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
-                exit(0);
-            }
-            WebRtcIsac_EncoderInit(ISACSWB_inst[k],1);
-            if ((bitrate<32000)||(bitrate>56000)) {
-                printf("\nError - iSAC SWB bitrate has to be between 32000 and 56000 bps (not %i)\n", bitrate);
-                exit(0);
-            }
-            WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize>>5);
-        } else {
-            printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n");
+      case webrtc::NetEqDecoder::kDecoderISACswb:
+        if (sampfreq == 32000) {
+          ok = WebRtcIsac_Create(&ISACSWB_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
             exit(0);
+          }
+          if (enc_frameSize == 960) {
+          } else {
+            printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
+            exit(0);
+          }
+          ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
+          if (ok != 0) {
+            printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
+            exit(0);
+          }
+          WebRtcIsac_EncoderInit(ISACSWB_inst[k], 1);
+          if ((bitrate < 32000) || (bitrate > 56000)) {
+            printf("\nError - iSAC SWB bitrate has to be between 32000 and "
+                   "56000 bps (not %i)\n", bitrate);
+            exit(0);
+          }
+          WebRtcIsac_Control(ISACSWB_inst[k], bitrate,
+                             static_cast<int>(enc_frameSize >> 5));
+        } else {
+          printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 "
+                 "ms)\n");
+          exit(0);
         }
         break;
 #endif
 #ifdef CODEC_GSMFR
-    case webrtc::kDecoderGSMFR:
-        if (sampfreq==8000) {
-            ok=WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
-            if (ok!=0) {
-                printf("Error: Couldn't allocate memory for GSM FR encoding instance\n");
-                exit(0);
-            }
-            if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {			
-            } else {
-                printf("\nError - GSM FR must have a multiple of 160 enc_frameSize\n");
-                exit(0);
-            }
-            WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
-        } else {
-            printf("\nError - GSM FR is only developed for 8kHz \n");
+      case webrtc::kDecoderGSMFR:
+        if (sampfreq == 8000) {
+          ok = WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
+          if (ok != 0) {
+            printf("Error: Couldn't allocate memory for GSM FR encoding "
+                   "instance\n");
             exit(0);
+          }
+          if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
+              (enc_frameSize == 480)) {
+          } else {
+            printf("\nError - GSM FR must have a multiple of 160 "
+                   "enc_frameSize\n");
+            exit(0);
+          }
+          WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
+        } else {
+          printf("\nError - GSM FR is only developed for 8kHz \n");
+          exit(0);
         }
         break;
 #endif
-    default :
+#ifdef CODEC_OPUS
+      case webrtc::NetEqDecoder::kDecoderOpus:
+        ok = WebRtcOpus_EncoderCreate(&opus_inst[k], 1, 0);
+        if (ok != 0) {
+          printf("Error: Couldn't allocate memory for Opus encoding "
+                 "instance\n");
+          exit(0);
+        }
+        WebRtcOpus_EnableFec(opus_inst[k]);
+        WebRtcOpus_SetPacketLossRate(opus_inst[k], 5);
+        break;
+#endif
+      default:
         printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
         exit(0);
         break;
-        }
+    }
+    if (ok != 0) {
+      return (ok);
+    }
+  }  // end for
 
-        if (ok != 0) {
-            return(ok);
-        }
-    }  // end for
+  return (0);
+}
 
-    return(0);
-}			
-
-
-
-
-int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels) {
-
-    for (int k = 0; k < numChannels; k++)
-    {
-        WebRtcVad_Free(VAD_inst[k]);
+int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
+  for (size_t k = 0; k < numChannels; k++) {
+    WebRtcVad_Free(VAD_inst[k]);
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
-    defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-        WebRtcCng_FreeEnc(CNGenc_inst[k]);
+     defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+    WebRtcCng_FreeEnc(CNGenc_inst[k]);
 #endif
 
-        switch (coder) 
-        {
+    switch (coder) {
 #ifdef CODEC_PCM16B
-        case webrtc::kDecoderPCM16B :
+      case webrtc::NetEqDecoder::kDecoderPCM16B:
 #endif
 #ifdef CODEC_PCM16B_WB
-        case webrtc::kDecoderPCM16Bwb :
+      case webrtc::NetEqDecoder::kDecoderPCM16Bwb:
 #endif
 #ifdef CODEC_PCM16B_32KHZ
-        case webrtc::kDecoderPCM16Bswb32kHz :
+      case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz:
 #endif
 #ifdef CODEC_PCM16B_48KHZ
-        case webrtc::kDecoderPCM16Bswb48kHz :
+      case webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz:
 #endif
 #ifdef CODEC_G711
-        case webrtc::kDecoderPCMu :
-        case webrtc::kDecoderPCMa :
+      case webrtc::NetEqDecoder::kDecoderPCMu:
+      case webrtc::NetEqDecoder::kDecoderPCMa:
 #endif
-            // do nothing
-            break;
+        // do nothing
+        break;
 #ifdef CODEC_G729
-        case webrtc::kDecoderG729:
-            WebRtcG729_FreeEnc(G729enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG729:
+        WebRtcG729_FreeEnc(G729enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G729_1
-        case webrtc::kDecoderG729_1:
-            WebRtcG7291_Free(G729_1_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG729_1:
+        WebRtcG7291_Free(G729_1_inst[k]);
+        break;
 #endif
 #ifdef CODEC_SPEEX_8
-        case webrtc::kDecoderSPEEX_8 :
-            WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderSPEEX_8:
+        WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_SPEEX_16
-        case webrtc::kDecoderSPEEX_16 :
-            WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderSPEEX_16:
+        WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
+        break;
 #endif
 
 #ifdef CODEC_G722_1_16
-        case webrtc::kDecoderG722_1_16 :
-            WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722_1_16:
+        WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G722_1_24
-        case webrtc::kDecoderG722_1_24 :
-            WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722_1_24:
+        WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G722_1_32
-        case webrtc::kDecoderG722_1_32 :
-            WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722_1_32:
+        WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G722_1C_24
-        case webrtc::kDecoderG722_1C_24 :
-            WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722_1C_24:
+        WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G722_1C_32
-        case webrtc::kDecoderG722_1C_32 :
-            WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722_1C_32:
+        WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G722_1C_48
-        case webrtc::kDecoderG722_1C_48 :
-            WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722_1C_48:
+        WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_G722
-        case webrtc::kDecoderG722 :
-            WebRtcG722_FreeEncoder(g722EncState[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderG722:
+        WebRtcG722_FreeEncoder(g722EncState[k]);
+        break;
 #endif
 #ifdef CODEC_AMR
-        case webrtc::kDecoderAMR :
-            WebRtcAmr_FreeEnc(AMRenc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderAMR:
+        WebRtcAmr_FreeEnc(AMRenc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_AMRWB
-        case webrtc::kDecoderAMRWB :
-            WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderAMRWB:
+        WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_ILBC
-        case webrtc::kDecoderILBC :
-            WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderILBC:
+        WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
+        break;
 #endif
 #ifdef CODEC_ISAC
-        case webrtc::kDecoderISAC:
-            WebRtcIsac_Free(ISAC_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderISAC:
+        WebRtcIsac_Free(ISAC_inst[k]);
+        break;
 #endif
 #ifdef NETEQ_ISACFIX_CODEC
-        case webrtc::kDecoderISAC:
-            WebRtcIsacfix_Free(ISAC_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderISAC:
+        WebRtcIsacfix_Free(ISAC_inst[k]);
+        break;
 #endif
 #ifdef CODEC_ISAC_SWB
-        case webrtc::kDecoderISACswb:
-            WebRtcIsac_Free(ISACSWB_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderISACswb:
+        WebRtcIsac_Free(ISACSWB_inst[k]);
+        break;
 #endif
 #ifdef CODEC_GSMFR
-        case webrtc::kDecoderGSMFR:
-            WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
-            break;
+      case webrtc::NetEqDecoder::kDecoderGSMFR:
+        WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
+        break;
 #endif
-        default :
-            printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
-            exit(0);
-            break;
+#ifdef CODEC_OPUS
+      case webrtc::NetEqDecoder::kDecoderOpus:
+        WebRtcOpus_EncoderFree(opus_inst[k]);
+        break;
+#endif
+      default:
+        printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
+        exit(0);
+        break;
+    }
+  }
+
+  return (0);
+}
+
+size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
+                        int16_t* indata,
+                        size_t frameLen,
+                        unsigned char* encoded,
+                        int sampleRate,
+                        int* vad,
+                        int useVAD,
+                        int bitrate,
+                        size_t numChannels) {
+  size_t cdlen = 0;
+  int16_t* tempdata;
+  static int first_cng = 1;
+  size_t tempLen;
+  *vad = 1;
+
+  // check VAD first
+  if (useVAD) {
+    *vad = 0;
+
+    size_t sampleRate_10 = static_cast<size_t>(10 * sampleRate / 1000);
+    size_t sampleRate_20 = static_cast<size_t>(20 * sampleRate / 1000);
+    size_t sampleRate_30 = static_cast<size_t>(30 * sampleRate / 1000);
+    for (size_t k = 0; k < numChannels; k++) {
+      tempLen = frameLen;
+      tempdata = &indata[k * frameLen];
+      int localVad = 0;
+      /* Partition the signal and test each chunk for VAD.
+         All chunks must be VAD=0 to produce a total VAD=0. */
+      while (tempLen >= sampleRate_10) {
+        if ((tempLen % sampleRate_30) == 0) {  // tempLen is multiple of 30ms
+          localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
+                                        sampleRate_30);
+          tempdata += sampleRate_30;
+          tempLen -= sampleRate_30;
+        } else if (tempLen >= sampleRate_20) {  // tempLen >= 20ms
+          localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
+                                        sampleRate_20);
+          tempdata += sampleRate_20;
+          tempLen -= sampleRate_20;
+        } else {  // use 10ms
+          localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
+                                        sampleRate_10);
+          tempdata += sampleRate_10;
+          tempLen -= sampleRate_10;
         }
+      }
+
+      // aggregate all VAD decisions over all channels
+      *vad |= localVad;
     }
 
-	return(0);
-}
+    if (!*vad) {
+      // all channels are silent
+      cdlen = 0;
+      for (size_t k = 0; k < numChannels; k++) {
+        WebRtcCng_Encode(CNGenc_inst[k], &indata[k * frameLen],
+                         (frameLen <= 640 ? frameLen : 640) /* max 640 */,
+                         encoded, &tempLen, first_cng);
+        encoded += tempLen;
+        cdlen += tempLen;
+      }
+      *vad = 0;
+      first_cng = 0;
+      return (cdlen);
+    }
+  }
 
+  // loop over all channels
+  size_t totalLen = 0;
 
-
-
-
-
-int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , 
-						  int * vad, int useVAD, int bitrate, int numChannels){
-
-	short cdlen = 0;
-	int16_t *tempdata;
-	static int first_cng=1;
-	int16_t tempLen;
-
-	*vad =1;
-
-    // check VAD first
-	if(useVAD)
-    {
-        *vad = 0;
-
-        for (int k = 0; k < numChannels; k++)
-        {
-            tempLen = frameLen;
-            tempdata = &indata[k*frameLen];
-            int localVad=0;
-            /* Partition the signal and test each chunk for VAD.
-            All chunks must be VAD=0 to produce a total VAD=0. */
-            while (tempLen >= 10*sampleRate/1000) {
-                if ((tempLen % 30*sampleRate/1000) == 0) { // tempLen is multiple of 30ms
-                    localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 30*sampleRate/1000);
-                    tempdata += 30*sampleRate/1000;
-                    tempLen -= 30*sampleRate/1000;
-                }
-                else if (tempLen >= 20*sampleRate/1000) { // tempLen >= 20ms
-                    localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 20*sampleRate/1000);
-                    tempdata += 20*sampleRate/1000;
-                    tempLen -= 20*sampleRate/1000;
-                }
-                else { // use 10ms
-                    localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 10*sampleRate/1000);
-                    tempdata += 10*sampleRate/1000;
-                    tempLen -= 10*sampleRate/1000;
-                }
-            }
-
-            // aggregate all VAD decisions over all channels
-            *vad |= localVad;
-        }
-
-        if(!*vad){
-            // all channels are silent
-            cdlen = 0;
-            for (int k = 0; k < numChannels; k++)
-            {
-                WebRtcCng_Encode(CNGenc_inst[k],&indata[k*frameLen], (frameLen <= 640 ? frameLen : 640) /* max 640 */,
-                    encoded,&tempLen,first_cng);
-                encoded += tempLen;
-                cdlen += tempLen;
-            }
-            *vad=0;
-            first_cng=0;
-            return(cdlen);
-        }
-	}
-
-
-    // loop over all channels
-    int totalLen = 0;
-
-    for (int k = 0; k < numChannels; k++)
-    {
-        /* Encode with the selected coder type */
-        if (coder==webrtc::kDecoderPCMu) { /*g711 u-law */
+  for (size_t k = 0; k < numChannels; k++) {
+    /* Encode with the selected coder type */
+    if (coder == webrtc::NetEqDecoder::kDecoderPCMu) { /*g711 u-law */
 #ifdef CODEC_G711
-            cdlen = WebRtcG711_EncodeU(indata, frameLen, encoded);
+      cdlen = WebRtcG711_EncodeU(indata, frameLen, encoded);
 #endif
-        }  
-        else if (coder==webrtc::kDecoderPCMa) { /*g711 A-law */
+    } else if (coder == webrtc::NetEqDecoder::kDecoderPCMa) { /*g711 A-law */
 #ifdef CODEC_G711
-            cdlen = WebRtcG711_EncodeA(indata, frameLen, encoded);
-        }
+      cdlen = WebRtcG711_EncodeA(indata, frameLen, encoded);
+    }
 #endif
 #ifdef CODEC_PCM16B
-        else if ((coder==webrtc::kDecoderPCM16B)||(coder==webrtc::kDecoderPCM16Bwb)||
-            (coder==webrtc::kDecoderPCM16Bswb32kHz)||(coder==webrtc::kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
-                cdlen = WebRtcPcm16b_Encode(indata, frameLen, encoded);
-            }
+    else if ((coder == webrtc::NetEqDecoder::kDecoderPCM16B) ||
+             (coder == webrtc::NetEqDecoder::kDecoderPCM16Bwb) ||
+             (coder == webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz) ||
+             (coder == webrtc::NetEqDecoder::
+                           kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz,
+                                                        32kHz or 48kHz) */
+      cdlen = WebRtcPcm16b_Encode(indata, frameLen, encoded);
+    }
 #endif
 #ifdef CODEC_G722
-        else if (coder==webrtc::kDecoderG722) { /*g722 */
-            cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded);
-            assert(cdlen == frameLen>>1);
-        }
+    else if (coder == webrtc::NetEqDecoder::kDecoderG722) { /*g722 */
+      cdlen = WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded);
+      assert(cdlen == frameLen >> 1);
+    }
 #endif
 #ifdef CODEC_ILBC
-        else if (coder==webrtc::kDecoderILBC) { /*iLBC */
-            cdlen = WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,
-                                         frameLen, encoded);
-        }
+    else if (coder == webrtc::NetEqDecoder::kDecoderILBC) { /*iLBC */
+      cdlen = static_cast<size_t>(std::max(
+          WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata, frameLen, encoded), 0));
+    }
 #endif
-#if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
-        else if (coder==webrtc::kDecoderISAC) { /*iSAC */
-            int noOfCalls=0;
-            cdlen=0;
-            while (cdlen<=0) {
+#if (defined(CODEC_ISAC) || \
+     defined(NETEQ_ISACFIX_CODEC))            // TODO(hlundin): remove all
+                                              // NETEQ_ISACFIX_CODEC
+    else if (coder == webrtc::NetEqDecoder::kDecoderISAC) { /*iSAC */
+      int noOfCalls = 0;
+      int res = 0;
+      while (res <= 0) {
 #ifdef CODEC_ISAC /* floating point */
-                cdlen = WebRtcIsac_Encode(ISAC_inst[k],
-                                          &indata[noOfCalls * 160],
-                                          encoded);
+        res =
+            WebRtcIsac_Encode(ISAC_inst[k], &indata[noOfCalls * 160], encoded);
 #else /* fixed point */
-                cdlen = WebRtcIsacfix_Encode(ISAC_inst[k],
-                                             &indata[noOfCalls * 160],
-                                             encoded);
+        res = WebRtcIsacfix_Encode(ISAC_inst[k], &indata[noOfCalls * 160],
+                                   encoded);
 #endif
-                noOfCalls++;
-            }
-        }
+        noOfCalls++;
+      }
+      cdlen = static_cast<size_t>(res);
+    }
 #endif
 #ifdef CODEC_ISAC_SWB
-        else if (coder==webrtc::kDecoderISACswb) { /* iSAC SWB */
-            int noOfCalls=0;
-            cdlen=0;
-            while (cdlen<=0) {
-                cdlen = WebRtcIsac_Encode(ISACSWB_inst[k],
-                                          &indata[noOfCalls * 320],
-                                          encoded);
-                noOfCalls++;
-            }
-        }
+    else if (coder == webrtc::NetEqDecoder::kDecoderISACswb) { /* iSAC SWB */
+      int noOfCalls = 0;
+      int res = 0;
+      while (res <= 0) {
+        res = WebRtcIsac_Encode(ISACSWB_inst[k], &indata[noOfCalls * 320],
+                                encoded);
+        noOfCalls++;
+      }
+      cdlen = static_cast<size_t>(res);
+    }
 #endif
-        indata += frameLen;
-        encoded += cdlen;
-        totalLen += cdlen;
+#ifdef CODEC_OPUS
+    cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12,
+                              encoded);
+    RTC_CHECK_GT(cdlen, 0u);
+#endif
+    indata += frameLen;
+    encoded += cdlen;
+    totalLen += cdlen;
 
-    }  // end for
+  }  // end for
 
-	first_cng=1;
-	return(totalLen);
+  first_cng = 1;
+  return (totalLen);
 }
 
-
-
 void makeRTPheader(unsigned char* rtp_data,
                    int payloadType,
                    int seqNo,
                    uint32_t timestamp,
                    uint32_t ssrc) {
-    rtp_data[0] = 0x80;
-    rtp_data[1] = payloadType & 0xFF;
-    rtp_data[2] = (seqNo >> 8) & 0xFF;
-    rtp_data[3] = seqNo & 0xFF;
-    rtp_data[4] = timestamp >> 24;
-    rtp_data[5] = (timestamp >> 16) & 0xFF;
-    rtp_data[6] = (timestamp >> 8) & 0xFF;
-    rtp_data[7] = timestamp & 0xFF;
-    rtp_data[8] = ssrc >> 24;
-    rtp_data[9] = (ssrc >> 16) & 0xFF;
-    rtp_data[10] = (ssrc >> 8) & 0xFF;
-    rtp_data[11] = ssrc & 0xFF;
+  rtp_data[0] = 0x80;
+  rtp_data[1] = payloadType & 0xFF;
+  rtp_data[2] = (seqNo >> 8) & 0xFF;
+  rtp_data[3] = seqNo & 0xFF;
+  rtp_data[4] = timestamp >> 24;
+  rtp_data[5] = (timestamp >> 16) & 0xFF;
+  rtp_data[6] = (timestamp >> 8) & 0xFF;
+  rtp_data[7] = timestamp & 0xFF;
+  rtp_data[8] = ssrc >> 24;
+  rtp_data[9] = (ssrc >> 16) & 0xFF;
+  rtp_data[10] = (ssrc >> 8) & 0xFF;
+  rtp_data[11] = ssrc & 0xFF;
 }
 
-
 int makeRedundantHeader(unsigned char* rtp_data,
                         int* payloadType,
                         int numPayloads,
                         uint32_t* timestamp,
                         uint16_t* blockLen,
                         int seqNo,
-                        uint32_t ssrc)
-{
-    int i;
-    unsigned char* rtpPointer;
-    uint16_t offset;
+                        uint32_t ssrc) {
+  int i;
+  unsigned char* rtpPointer;
+  uint16_t offset;
 
-    /* first create "standard" RTP header */
-    makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1],
-                  ssrc);
+  /* first create "standard" RTP header */
+  makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads - 1],
+                ssrc);
 
-    rtpPointer = &rtp_data[12];
+  rtpPointer = &rtp_data[12];
 
-    /* add one sub-header for each redundant payload (not the primary) */
-    for (i = 0; i < numPayloads - 1; i++) {
-        if (blockLen[i] > 0) {
-            offset = static_cast<uint16_t>(
-                timestamp[numPayloads - 1] - timestamp[i]);
+  /* add one sub-header for each redundant payload (not the primary) */
+  for (i = 0; i < numPayloads - 1; i++) {
+    if (blockLen[i] > 0) {
+      offset = static_cast<uint16_t>(timestamp[numPayloads - 1] - timestamp[i]);
 
-            // Byte |0|       |1       2     |  3       |
-            // Bit  |0|1234567|01234567012345|6701234567|
-            //      |F|payload|   timestamp  |   block  |
-            //      | |  type |    offset    |  length  |
-            rtpPointer[0] = (payloadType[i] & 0x7F) | 0x80;
-            rtpPointer[1] = (offset >> 6) & 0xFF;
-            rtpPointer[2] =
-                ((offset & 0x3F) << 2) | ((blockLen[i] >> 8) & 0x03);
-            rtpPointer[3] = blockLen[i] & 0xFF;
+      // Byte |0|       |1       2     |  3       |
+      // Bit  |0|1234567|01234567012345|6701234567|
+      //      |F|payload|   timestamp  |   block  |
+      //      | |  type |    offset    |  length  |
+      rtpPointer[0] = (payloadType[i] & 0x7F) | 0x80;
+      rtpPointer[1] = (offset >> 6) & 0xFF;
+      rtpPointer[2] = ((offset & 0x3F) << 2) | ((blockLen[i] >> 8) & 0x03);
+      rtpPointer[3] = blockLen[i] & 0xFF;
 
-            rtpPointer += 4;
-        }
+      rtpPointer += 4;
     }
+  }
 
-    // Bit  |0|1234567|
-    //      |0|payload|
-    //      | |  type |
-    rtpPointer[0] = payloadType[numPayloads - 1] & 0x7F;
-    ++rtpPointer;
+  // Bit  |0|1234567|
+  //      |0|payload|
+  //      | |  type |
+  rtpPointer[0] = payloadType[numPayloads - 1] & 0x7F;
+  ++rtpPointer;
 
-    return rtpPointer - rtp_data;  // length of header in bytes
+  return rtpPointer - rtp_data;  // length of header in bytes
 }
 
-
-
-int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) {
-	unsigned char E,R,V;
-	R=0;
-	V=(unsigned char)Volume;
-	if (End==0) {
-		E = 0x00;
-	} else {
-		E = 0x80;
-	}
-	payload_data[0]=(unsigned char)Event;
-	payload_data[1]=(unsigned char)(E|R|V);
-	//Duration equals 8 times time_ms, default is 8000 Hz.
-	payload_data[2]=(unsigned char)((Duration>>8)&0xFF);
-	payload_data[3]=(unsigned char)(Duration&0xFF);
-	return(4);
+size_t makeDTMFpayload(unsigned char* payload_data,
+                       int Event,
+                       int End,
+                       int Volume,
+                       int Duration) {
+  unsigned char E, R, V;
+  R = 0;
+  V = (unsigned char)Volume;
+  if (End == 0) {
+    E = 0x00;
+  } else {
+    E = 0x80;
+  }
+  payload_data[0] = (unsigned char)Event;
+  payload_data[1] = (unsigned char)(E | R | V);
+  // Duration equals 8 times time_ms, default is 8000 Hz.
+  payload_data[2] = (unsigned char)((Duration >> 8) & 0xFF);
+  payload_data[3] = (unsigned char)(Duration & 0xFF);
+  return (4);
 }
 
-void stereoDeInterleave(int16_t* audioSamples, int numSamples)
-{
+void stereoDeInterleave(int16_t* audioSamples, size_t numSamples) {
+  int16_t* tempVec;
+  int16_t* readPtr, *writeL, *writeR;
 
-    int16_t *tempVec;
-    int16_t *readPtr, *writeL, *writeR;
+  if (numSamples == 0)
+    return;
 
-    if (numSamples <= 0)
-        return;
+  tempVec = (int16_t*)malloc(sizeof(int16_t) * numSamples);
+  if (tempVec == NULL) {
+    printf("Error allocating memory\n");
+    exit(0);
+  }
 
-    tempVec = (int16_t *) malloc(sizeof(int16_t) * numSamples);
-    if (tempVec == NULL) {
-        printf("Error allocating memory\n");
-        exit(0);
-    }
+  memcpy(tempVec, audioSamples, numSamples * sizeof(int16_t));
 
-    memcpy(tempVec, audioSamples, numSamples*sizeof(int16_t));
+  writeL = audioSamples;
+  writeR = &audioSamples[numSamples / 2];
+  readPtr = tempVec;
 
-    writeL = audioSamples;
-    writeR = &audioSamples[numSamples/2];
-    readPtr = tempVec;
-
-    for (int k = 0; k < numSamples; k += 2)
-    {
-        *writeL = *readPtr;
-        readPtr++;
-        *writeR = *readPtr;
-        readPtr++;
-        writeL++;
-        writeR++;
-    }
-
-    free(tempVec);
+  for (size_t k = 0; k < numSamples; k += 2) {
+    *writeL = *readPtr;
+    readPtr++;
+    *writeR = *readPtr;
+    readPtr++;
+    writeL++;
+    writeR++;
+  }
 
+  free(tempVec);
 }
 
+void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride) {
+  unsigned char* ptrL, *ptrR;
+  unsigned char temp[10];
 
-void stereoInterleave(unsigned char* data, int dataLen, int stride)
-{
+  if (stride > 10) {
+    exit(0);
+  }
 
-    unsigned char *ptrL, *ptrR;
-    unsigned char temp[10];
+  if (dataLen % 1 != 0) {
+    // must be even number of samples
+    printf("Error: cannot interleave odd sample number\n");
+    exit(0);
+  }
 
-    if (stride > 10)
-    {
-        exit(0);
-    }
+  ptrL = data + stride;
+  ptrR = &data[dataLen / 2];
 
-    if (dataLen%1 != 0)
-    {
-        // must be even number of samples
-        printf("Error: cannot interleave odd sample number\n");
-        exit(0);
-    }
+  while (ptrL < ptrR) {
+    // copy from right pointer to temp
+    memcpy(temp, ptrR, stride);
 
-    ptrL = data + stride;
-    ptrR = &data[dataLen/2];
+    // shift data between pointers
+    memmove(ptrL + stride, ptrL, ptrR - ptrL);
 
-    while (ptrL < ptrR) {
-        // copy from right pointer to temp
-        memcpy(temp, ptrR, stride);
-
-        // shift data between pointers
-        memmove(ptrL + stride, ptrL, ptrR - ptrL);
-
-        // copy from temp to left pointer
-        memcpy(ptrL, temp, stride);
-
-        // advance pointers
-        ptrL += stride*2;
-        ptrR += stride;
-    }
+    // copy from temp to left pointer
+    memcpy(ptrL, temp, stride);
 
+    // advance pointers
+    ptrL += stride * 2;
+    ptrR += stride;
+  }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPtimeshift.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPtimeshift.cc
index f27819d03c..f45a97af2f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPtimeshift.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/RTPtimeshift.cc
@@ -8,92 +8,79 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <algorithm>
 #include <stdio.h>
+#include <algorithm>
 #include <vector>
 
-#include "NETEQTEST_RTPpacket.h"
 #include "testing/gtest/include/gtest/gtest.h"
-
-/*********************/
-/* Misc. definitions */
-/*********************/
+#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
 
 #define FIRSTLINELEN 40
 
+int main(int argc, char* argv[]) {
+  if (argc < 4 || argc > 6) {
+    printf(
+        "Usage: RTPtimeshift in.rtp out.rtp newStartTS [newStartSN "
+        "[newStartArrTime]]\n");
+    exit(1);
+  }
 
-int main(int argc, char* argv[])
-{
-    if(argc < 4 || argc > 6)
-    {
-        printf("Usage: RTPtimeshift in.rtp out.rtp newStartTS [newStartSN [newStartArrTime]]\n");
-        exit(1);
+  FILE* inFile = fopen(argv[1], "rb");
+  if (!inFile) {
+    printf("Cannot open input file %s\n", argv[1]);
+    return (-1);
+  }
+  printf("Input RTP file: %s\n", argv[1]);
+
+  FILE* outFile = fopen(argv[2], "wb");
+  if (!outFile) {
+    printf("Cannot open output file %s\n", argv[2]);
+    return (-1);
+  }
+  printf("Output RTP file: %s\n\n", argv[2]);
+
+  // Read file header and write directly to output file.
+  const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+  char firstline[FIRSTLINELEN];
+  EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
+  EXPECT_GT(fputs(firstline, outFile), 0);
+  EXPECT_EQ(kRtpDumpHeaderSize,
+            fread(firstline, 1, kRtpDumpHeaderSize, inFile));
+  EXPECT_EQ(kRtpDumpHeaderSize,
+            fwrite(firstline, 1, kRtpDumpHeaderSize, outFile));
+  NETEQTEST_RTPpacket packet;
+  int packLen = packet.readFromFile(inFile);
+  if (packLen < 0) {
+    exit(1);
+  }
+
+  // Get new start TS and start SeqNo from arguments.
+  uint32_t TSdiff = atoi(argv[3]) - packet.timeStamp();
+  uint16_t SNdiff = 0;
+  uint32_t ATdiff = 0;
+  if (argc > 4) {
+    int startSN = atoi(argv[4]);
+    if (startSN >= 0)
+      SNdiff = startSN - packet.sequenceNumber();
+    if (argc > 5) {
+      int startTS = atoi(argv[5]);
+      if (startTS >= 0)
+        ATdiff = startTS - packet.time();
     }
+  }
 
-	FILE *inFile=fopen(argv[1],"rb");
-	if (!inFile)
-    {
-        printf("Cannot open input file %s\n", argv[1]);
-        return(-1);
-    }
-    printf("Input RTP file: %s\n",argv[1]);
+  while (packLen >= 0) {
+    packet.setTimeStamp(packet.timeStamp() + TSdiff);
+    packet.setSequenceNumber(packet.sequenceNumber() + SNdiff);
+    packet.setTime(packet.time() + ATdiff);
 
-	FILE *outFile=fopen(argv[2],"wb");
-	if (!outFile)
-    {
-        printf("Cannot open output file %s\n", argv[2]);
-        return(-1);
-    }
-	printf("Output RTP file: %s\n\n",argv[2]);
+    packet.writeToFile(outFile);
 
-    // read file header and write directly to output file
-	const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
-	char firstline[FIRSTLINELEN];
-	EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
-	EXPECT_GT(fputs(firstline, outFile), 0);
-	EXPECT_EQ(kRtpDumpHeaderSize,
-	          fread(firstline, 1, kRtpDumpHeaderSize, inFile));
-	EXPECT_EQ(kRtpDumpHeaderSize,
-	          fwrite(firstline, 1, kRtpDumpHeaderSize, outFile));
-	NETEQTEST_RTPpacket packet;
-	int packLen = packet.readFromFile(inFile);
-	if (packLen < 0)
-	{
-	    exit(1);
-	}
+    packLen = packet.readFromFile(inFile);
+  }
 
-    // get new start TS and start SeqNo from arguments
-	uint32_t TSdiff = atoi(argv[3]) - packet.timeStamp();
-	uint16_t SNdiff = 0;
-	uint32_t ATdiff = 0;
-    if (argc > 4)
-    {
-        int startSN = atoi(argv[4]);
-        if (startSN >= 0)
-            SNdiff = startSN - packet.sequenceNumber();
-        if (argc > 5)
-        {
-            int startTS = atoi(argv[5]);
-            if (startTS >= 0)
-                ATdiff = startTS - packet.time();
-        }
-    }
+  fclose(inFile);
+  fclose(outFile);
 
-    while (packLen >= 0)
-    {
-
-        packet.setTimeStamp(packet.timeStamp() + TSdiff);
-        packet.setSequenceNumber(packet.sequenceNumber() + SNdiff);
-        packet.setTime(packet.time() + ATdiff);
-
-        packet.writeToFile(outFile);
-
-        packLen = packet.readFromFile(inFile);
-
-    }
-
-    fclose(inFile);
-    fclose(outFile);
-
-    return 0;
+  return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
new file mode 100644
index 0000000000..0c09e92b4d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using google::RegisterFlagValidator;
+using google::ParseCommandLineFlags;
+using std::string;
+using testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+
+// Define switch for frame size.
+static bool ValidateFrameSize(const char* flagname, int32_t value) {
+  if (value == 20 || value == 30 || value == 40 || value == 60)
+    return true;
+  printf("Invalid frame size, should be 20, 30, 40, or 60 ms.");
+  return false;
+}
+
+DEFINE_int32(frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+static const bool frame_size_dummy =
+    RegisterFlagValidator(&FLAGS_frame_size_ms, &ValidateFrameSize);
+
+}  // namespace
+
+class NetEqIlbcQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqIlbcQualityTest()
+      : NetEqQualityTest(FLAGS_frame_size_ms,
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         NetEqDecoder::kDecoderILBC) {}
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
+    AudioEncoderIlbc::Config config;
+    config.frame_size_ms = FLAGS_frame_size_ms;
+    encoder_.reset(new AudioEncoderIlbc(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  uint8_t* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              max_bytes, payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  rtc::scoped_ptr<AudioEncoderIlbc> encoder_;
+};
+
+TEST_F(NetEqIlbcQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index 7abf5a1fa7..4ccebb3e66 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -8,9 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
-#include "webrtc/test/testsupport/fileutils.h"
 
 using google::RegisterFlagValidator;
 using google::ParseCommandLineFlags;
@@ -19,47 +18,12 @@ using testing::InitGoogleTest;
 
 namespace webrtc {
 namespace test {
-
+namespace {
 static const int kIsacBlockDurationMs = 30;
 static const int kIsacInputSamplingKhz = 16;
 static const int kIsacOutputSamplingKhz = 16;
 
-// Define switch for input file name.
-static bool ValidateInFilename(const char* flagname, const string& value) {
-  FILE* fid = fopen(value.c_str(), "rb");
-  if (fid != NULL) {
-    fclose(fid);
-    return true;
-  }
-  printf("Invalid input filename.");
-  return false;
-}
-
-DEFINE_string(in_filename,
-              ResourcePath("audio_coding/speech_mono_16kHz", "pcm"),
-              "Filename for input audio (should be 16 kHz sampled mono).");
-
-static const bool in_filename_dummy =
-    RegisterFlagValidator(&FLAGS_in_filename, &ValidateInFilename);
-
-// Define switch for output file name.
-static bool ValidateOutFilename(const char* flagname, const string& value) {
-  FILE* fid = fopen(value.c_str(), "wb");
-  if (fid != NULL) {
-    fclose(fid);
-    return true;
-  }
-  printf("Invalid output filename.");
-  return false;
-}
-
-DEFINE_string(out_filename, OutputPath() + "neteq4_isac_quality_test.pcm",
-              "Name of output audio file.");
-
-static const bool out_filename_dummy =
-    RegisterFlagValidator(&FLAGS_out_filename, &ValidateOutFilename);
-
-// Define switch for bir rate.
+// Define switch for bit rate.
 static bool ValidateBitRate(const char* flagname, int32_t value) {
   if (value >= 10 && value <= 32)
     return true;
@@ -72,43 +36,30 @@ DEFINE_int32(bit_rate_kbps, 32, "Target bit rate (kbps).");
 static const bool bit_rate_dummy =
     RegisterFlagValidator(&FLAGS_bit_rate_kbps, &ValidateBitRate);
 
-// Define switch for runtime.
-static bool ValidateRuntime(const char* flagname, int32_t value) {
-  if (value > 0)
-    return true;
-  printf("Invalid runtime, should be greater than 0.");
-  return false;
-}
-
-DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");
-
-static const bool runtime_dummy =
-    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
+}  // namespace
 
 class NetEqIsacQualityTest : public NetEqQualityTest {
  protected:
   NetEqIsacQualityTest();
   void SetUp() override;
   void TearDown() override;
-  virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
-                          uint8_t* payload, int max_bytes);
+  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
+                          uint8_t* payload, size_t max_bytes);
  private:
   ISACFIX_MainStruct* isac_encoder_;
   int bit_rate_kbps_;
 };
 
 NetEqIsacQualityTest::NetEqIsacQualityTest()
-    : NetEqQualityTest(kIsacBlockDurationMs, kIsacInputSamplingKhz,
+    : NetEqQualityTest(kIsacBlockDurationMs,
+                       kIsacInputSamplingKhz,
                        kIsacOutputSamplingKhz,
-                       kDecoderISAC,
-                       1,
-                       FLAGS_in_filename,
-                       FLAGS_out_filename),
+                       NetEqDecoder::kDecoderISAC),
       isac_encoder_(NULL),
-      bit_rate_kbps_(FLAGS_bit_rate_kbps) {
-}
+      bit_rate_kbps_(FLAGS_bit_rate_kbps) {}
 
 void NetEqIsacQualityTest::SetUp() {
+  ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
   // Create encoder memory.
   WebRtcIsacfix_Create(&isac_encoder_);
   ASSERT_TRUE(isac_encoder_ != NULL);
@@ -126,8 +77,8 @@ void NetEqIsacQualityTest::TearDown() {
 }
 
 int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
-                                      int block_size_samples,
-                                      uint8_t* payload, int max_bytes) {
+                                      size_t block_size_samples,
+                                      uint8_t* payload, size_t max_bytes) {
   // ISAC takes 10 ms for every call.
   const int subblocks = kIsacBlockDurationMs / 10;
   const int subblock_length = 10 * kIsacInputSamplingKhz;
@@ -145,7 +96,7 @@ int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
 }
 
 TEST_F(NetEqIsacQualityTest, Test) {
-  Simulate(FLAGS_runtime_ms);
+  Simulate();
 }
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
index 3edf89cf52..5ab55ba9e8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
-#include "webrtc/test/testsupport/fileutils.h"
 
 using google::RegisterFlagValidator;
 using google::ParseCommandLineFlags;
@@ -20,58 +19,11 @@ using testing::InitGoogleTest;
 
 namespace webrtc {
 namespace test {
+namespace {
 
 static const int kOpusBlockDurationMs = 20;
 static const int kOpusSamplingKhz = 48;
 
-// Define switch for input file name.
-static bool ValidateInFilename(const char* flagname, const string& value) {
-  FILE* fid = fopen(value.c_str(), "rb");
-  if (fid != NULL) {
-    fclose(fid);
-    return true;
-  }
-  printf("Invalid input filename.");
-  return false;
-}
-
-DEFINE_string(in_filename,
-              ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"),
-              "Filename for input audio (should be 48 kHz sampled raw data).");
-
-static const bool in_filename_dummy =
-    RegisterFlagValidator(&FLAGS_in_filename, &ValidateInFilename);
-
-// Define switch for output file name.
-static bool ValidateOutFilename(const char* flagname, const string& value) {
-  FILE* fid = fopen(value.c_str(), "wb");
-  if (fid != NULL) {
-    fclose(fid);
-    return true;
-  }
-  printf("Invalid output filename.");
-  return false;
-}
-
-DEFINE_string(out_filename, OutputPath() + "neteq_opus_quality_test.pcm",
-              "Name of output audio file.");
-
-static const bool out_filename_dummy =
-    RegisterFlagValidator(&FLAGS_out_filename, &ValidateOutFilename);
-
-// Define switch for channels.
-static bool ValidateChannels(const char* flagname, int32_t value) {
-  if (value == 1 || value == 2)
-    return true;
-  printf("Invalid number of channels, should be either 1 or 2.");
-  return false;
-}
-
-DEFINE_int32(channels, 1, "Number of channels in input audio.");
-
-static const bool channels_dummy =
-    RegisterFlagValidator(&FLAGS_channels, &ValidateChannels);
-
 // Define switch for bit rate.
 static bool ValidateBitRate(const char* flagname, int32_t value) {
   if (value >= 6 && value <= 510)
@@ -85,6 +37,37 @@ DEFINE_int32(bit_rate_kbps, 32, "Target bit rate (kbps).");
 static const bool bit_rate_dummy =
     RegisterFlagValidator(&FLAGS_bit_rate_kbps, &ValidateBitRate);
 
+// Define switch for complexity.
+static bool ValidateComplexity(const char* flagname, int32_t value) {
+  if (value >= -1 && value <= 10)
+    return true;
+  printf("Invalid complexity setting, should be between 0 and 10.");
+  return false;
+}
+
+DEFINE_int32(complexity, 10, "Complexity: 0 ~ 10 -- defined as in Opus"
+    "specification.");
+
+static const bool complexity_dummy =
+    RegisterFlagValidator(&FLAGS_complexity, &ValidateComplexity);
+
+// Define switch for maxplaybackrate
+DEFINE_int32(maxplaybackrate, 48000, "Maximum playback rate (Hz).");
+
+// Define switch for application mode.
+static bool ValidateApplication(const char* flagname, int32_t value) {
+  if (value != 0 && value != 1) {
+    printf("Invalid application mode, should be 0 or 1.");
+    return false;
+  }
+  return true;
+}
+
+DEFINE_int32(application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
+
+static const bool application_dummy =
+    RegisterFlagValidator(&FLAGS_application, &ValidateApplication);
+
 // Define switch for reported packet loss rate.
 static bool ValidatePacketLossRate(const char* flagname, int32_t value) {
   if (value >= 0 && value <= 100)
@@ -98,21 +81,9 @@ DEFINE_int32(reported_loss_rate, 10, "Reported percentile of packet loss.");
 static const bool reported_loss_rate_dummy =
     RegisterFlagValidator(&FLAGS_reported_loss_rate, &ValidatePacketLossRate);
 
-// Define switch for runtime.
-static bool ValidateRuntime(const char* flagname, int32_t value) {
-  if (value > 0)
-    return true;
-  printf("Invalid runtime, should be greater than 0.");
-  return false;
-}
+DEFINE_bool(fec, false, "Enable FEC for encoding (-nofec to disable).");
 
-DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");
-static const bool runtime_dummy =
-    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
-
-DEFINE_bool(fec, true, "Whether to enable FEC for encoding.");
-
-DEFINE_bool(dtx, true, "Whether to enable DTX for encoding.");
+DEFINE_bool(dtx, false, "Enable DTX for encoding (-nodtx to disable).");
 
 // Define switch for number of sub packets to repacketize.
 static bool ValidateSubPackets(const char* flagname, int32_t value) {
@@ -125,49 +96,55 @@ DEFINE_int32(sub_packets, 1, "Number of sub packets to repacketize.");
 static const bool sub_packets_dummy =
     RegisterFlagValidator(&FLAGS_sub_packets, &ValidateSubPackets);
 
+}  // namepsace
+
 class NetEqOpusQualityTest : public NetEqQualityTest {
  protected:
   NetEqOpusQualityTest();
   void SetUp() override;
   void TearDown() override;
-  virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
-                          uint8_t* payload, int max_bytes);
+  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
+                          uint8_t* payload, size_t max_bytes);
  private:
   WebRtcOpusEncInst* opus_encoder_;
   OpusRepacketizer* repacketizer_;
-  int sub_block_size_samples_;
-  int channels_;
+  size_t sub_block_size_samples_;
   int bit_rate_kbps_;
   bool fec_;
   bool dtx_;
+  int complexity_;
+  int maxplaybackrate_;
   int target_loss_rate_;
   int sub_packets_;
+  int application_;
 };
 
 NetEqOpusQualityTest::NetEqOpusQualityTest()
     : NetEqQualityTest(kOpusBlockDurationMs * FLAGS_sub_packets,
                        kOpusSamplingKhz,
                        kOpusSamplingKhz,
-                       (FLAGS_channels == 1) ? kDecoderOpus : kDecoderOpus_2ch,
-                       FLAGS_channels,
-                       FLAGS_in_filename,
-                       FLAGS_out_filename),
+                       NetEqDecoder::kDecoderOpus),
       opus_encoder_(NULL),
       repacketizer_(NULL),
-      sub_block_size_samples_(kOpusBlockDurationMs * kOpusSamplingKhz),
-      channels_(FLAGS_channels),
+      sub_block_size_samples_(
+          static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
       bit_rate_kbps_(FLAGS_bit_rate_kbps),
       fec_(FLAGS_fec),
       dtx_(FLAGS_dtx),
+      complexity_(FLAGS_complexity),
+      maxplaybackrate_(FLAGS_maxplaybackrate),
       target_loss_rate_(FLAGS_reported_loss_rate),
       sub_packets_(FLAGS_sub_packets) {
+  // Redefine decoder type if input is stereo.
+  if (channels_ > 1) {
+    decoder_type_ = NetEqDecoder::kDecoderOpus_2ch;
+  }
+  application_ = FLAGS_application;
 }
 
 void NetEqOpusQualityTest::SetUp() {
-  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
-  int app = channels_ == 1 ? 0 : 1;
   // Create encoder memory.
-  WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app);
+  WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_);
   ASSERT_TRUE(opus_encoder_);
 
   // Create repacketizer.
@@ -182,6 +159,8 @@ void NetEqOpusQualityTest::SetUp() {
   if (dtx_) {
     EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
   }
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
   EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_,
                                             target_loss_rate_));
   NetEqQualityTest::SetUp();
@@ -195,8 +174,8 @@ void NetEqOpusQualityTest::TearDown() {
 }
 
 int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
-                                      int block_size_samples,
-                                      uint8_t* payload, int max_bytes) {
+                                      size_t block_size_samples,
+                                      uint8_t* payload, size_t max_bytes) {
   EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
   int16_t* pointer = in_data;
   int value;
@@ -204,6 +183,9 @@ int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
   for (int idx = 0; idx < sub_packets_; idx++) {
     value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
                               max_bytes, payload);
+    Log() << "Encoded a frame with Opus mode "
+          << (value == 0 ? 0 : payload[0] >> 3)
+          << std::endl;
     if (OPUS_OK != opus_repacketizer_cat(repacketizer_, payload, value)) {
       opus_repacketizer_init(repacketizer_);
       // If the repacketization fails, we discard this frame.
@@ -211,13 +193,14 @@ int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
     }
     pointer += sub_block_size_samples_ * channels_;
   }
-  value = opus_repacketizer_out(repacketizer_, payload, max_bytes);
+  value = opus_repacketizer_out(repacketizer_, payload,
+                                static_cast<opus_int32>(max_bytes));
   EXPECT_GE(value, 0);
   return value;
 }
 
 TEST_F(NetEqOpusQualityTest, Test) {
-  Simulate(FLAGS_runtime_ms);
+  Simulate();
 }
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
new file mode 100644
index 0000000000..ac478ab5ac
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using google::RegisterFlagValidator;
+using google::ParseCommandLineFlags;
+using std::string;
+using testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+
+// Define switch for frame size.
+static bool ValidateFrameSize(const char* flagname, int32_t value) {
+  if (value >= 10 && value <= 60 && (value % 10) == 0)
+    return true;
+  printf("Invalid frame size, should be 10, 20, ..., 60 ms.");
+  return false;
+}
+
+DEFINE_int32(frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+static const bool frame_size_dummy =
+    RegisterFlagValidator(&FLAGS_frame_size_ms, &ValidateFrameSize);
+
+}  // namespace
+
+class NetEqPcmuQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcmuQualityTest()
+      : NetEqQualityTest(FLAGS_frame_size_ms,
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         NetEqDecoder::kDecoderPCMu) {}
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = FLAGS_frame_size_ms;
+    encoder_.reset(new AudioEncoderPcmU(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  uint8_t* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              max_bytes, payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  rtc::scoped_ptr<AudioEncoderPcmU> encoder_;
+};
+
+TEST_F(NetEqPcmuQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/rtp_to_text.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/rtp_to_text.cc
index aa3f28f361..572b9d3b70 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/rtp_to_text.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/rtp_to_text.cc
@@ -17,7 +17,7 @@
  *
  */
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 #include "NETEQTEST_DummyRTPpacket.h"
 #include "NETEQTEST_RTPpacket.h"
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.cc
index 02305c83e5..6ae81e6e96 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -12,6 +12,7 @@
 
 #include <algorithm>  // min, max
 
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
@@ -19,14 +20,14 @@
 
 namespace webrtc {
 
-TimeStretch::ReturnCodes TimeStretch::Process(
-    const int16_t* input,
-    size_t input_len,
-    AudioMultiVector* output,
-    int16_t* length_change_samples) {
-
+TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
+                                              size_t input_len,
+                                              bool fast_mode,
+                                              AudioMultiVector* output,
+                                              size_t* length_change_samples) {
   // Pre-calculate common multiplication with |fs_mult_|.
-  int fs_mult_120 = fs_mult_ * 120;  // Corresponds to 15 ms.
+  size_t fs_mult_120 =
+      static_cast<size_t>(fs_mult_ * 120);  // Corresponds to 15 ms.
 
   const int16_t* signal;
   rtc::scoped_ptr<int16_t[]> signal_array;
@@ -49,8 +50,7 @@ TimeStretch::ReturnCodes TimeStretch::Process(
   }
 
   // Find maximum absolute value of input signal.
-  max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal,
-                                              static_cast<int>(signal_len));
+  max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal, signal_len);
 
   // Downsample to 4 kHz sample rate and calculate auto-correlation.
   DspHelper::DownsampleTo4kHz(signal, signal_len, kDownsampledLen,
@@ -59,13 +59,12 @@ TimeStretch::ReturnCodes TimeStretch::Process(
   AutoCorrelation();
 
   // Find the strongest correlation peak.
-  static const int kNumPeaks = 1;
-  int peak_index;
+  static const size_t kNumPeaks = 1;
+  size_t peak_index;
   int16_t peak_value;
   DspHelper::PeakDetection(auto_correlation_, kCorrelationLen, kNumPeaks,
                            fs_mult_, &peak_index, &peak_value);
   // Assert that |peak_index| stays within boundaries.
-  assert(peak_index >= 0);
   assert(peak_index <= (2 * kCorrelationLen - 1) * fs_mult_);
 
   // Compensate peak_index for displaced starting position. The displacement
@@ -74,13 +73,13 @@ TimeStretch::ReturnCodes TimeStretch::Process(
   // multiplication by fs_mult_ * 2.
   peak_index += kMinLag * fs_mult_ * 2;
   // Assert that |peak_index| stays within boundaries.
-  assert(peak_index >= 20 * fs_mult_);
+  assert(peak_index >= static_cast<size_t>(20 * fs_mult_));
   assert(peak_index <= 20 * fs_mult_ + (2 * kCorrelationLen - 1) * fs_mult_);
 
   // Calculate scaling to ensure that |peak_index| samples can be square-summed
   // without overflowing.
   int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) -
-      WebRtcSpl_NormW32(peak_index);
+      WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));
   scaling = std::max(0, scaling);
 
   // |vec1| starts at 15 ms minus one pitch period.
@@ -140,8 +139,9 @@ TimeStretch::ReturnCodes TimeStretch::Process(
 
 
   // Check accelerate criteria and stretch the signal.
-  ReturnCodes return_value = CheckCriteriaAndStretch(
-      input, input_len, peak_index, best_correlation, active_speech, output);
+  ReturnCodes return_value =
+      CheckCriteriaAndStretch(input, input_len, peak_index, best_correlation,
+                              active_speech, fast_mode, output);
   switch (return_value) {
     case kSuccess:
       *length_change_samples = peak_index;
@@ -177,7 +177,7 @@ void TimeStretch::AutoCorrelation() {
 }
 
 bool TimeStretch::SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
-                                  int peak_index, int scaling) const {
+                                  size_t peak_index, int scaling) const {
   // Check if the signal seems to be active speech or not (simple VAD).
   // If (vec1_energy + vec2_energy) / (2 * peak_index) <=
   // 8 * background_noise_energy, then we say that the signal contains no
@@ -197,7 +197,8 @@ bool TimeStretch::SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
   int right_scale = 16 - WebRtcSpl_NormW32(right_side);
   right_scale = std::max(0, right_scale);
   left_side = left_side >> right_scale;
-  right_side = peak_index * (right_side >> right_scale);
+  right_side =
+      rtc::checked_cast<int32_t>(peak_index) * (right_side >> right_scale);
 
   // Scale |left_side| properly before comparing with |right_side|.
   // (|scaling| is the scale factor before energy calculation, thus the scale
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.h
index 9396d8ff51..00a141508b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch.h
@@ -39,7 +39,7 @@ class TimeStretch {
               const BackgroundNoise& background_noise)
       : sample_rate_hz_(sample_rate_hz),
         fs_mult_(sample_rate_hz / 8000),
-        num_channels_(static_cast<int>(num_channels)),
+        num_channels_(num_channels),
         master_channel_(0),  // First channel is master.
         background_noise_(background_noise),
         max_input_value_(0) {
@@ -48,7 +48,7 @@ class TimeStretch {
            sample_rate_hz_ == 32000 ||
            sample_rate_hz_ == 48000);
     assert(num_channels_ > 0);
-    assert(static_cast<int>(master_channel_) < num_channels_);
+    assert(master_channel_ < num_channels_);
     memset(auto_correlation_, 0, sizeof(auto_correlation_));
   }
 
@@ -58,8 +58,9 @@ class TimeStretch {
   // PreemptiveExpand.
   ReturnCodes Process(const int16_t* input,
                       size_t input_len,
+                      bool fast_mode,
                       AudioMultiVector* output,
-                      int16_t* length_change_samples);
+                      size_t* length_change_samples);
 
  protected:
   // Sets the parameters |best_correlation| and |peak_index| to suitable
@@ -67,26 +68,30 @@ class TimeStretch {
   // implemented by the sub-classes.
   virtual void SetParametersForPassiveSpeech(size_t input_length,
                                              int16_t* best_correlation,
-                                             int* peak_index) const = 0;
+                                             size_t* peak_index) const = 0;
 
   // Checks the criteria for performing the time-stretching operation and,
   // if possible, performs the time-stretching. This method must be implemented
   // by the sub-classes.
   virtual ReturnCodes CheckCriteriaAndStretch(
-      const int16_t* input, size_t input_length, size_t peak_index,
-      int16_t best_correlation, bool active_speech,
+      const int16_t* input,
+      size_t input_length,
+      size_t peak_index,
+      int16_t best_correlation,
+      bool active_speech,
+      bool fast_mode,
       AudioMultiVector* output) const = 0;
 
-  static const int kCorrelationLen = 50;
-  static const int kLogCorrelationLen = 6;  // >= log2(kCorrelationLen).
-  static const int kMinLag = 10;
-  static const int kMaxLag = 60;
-  static const int kDownsampledLen = kCorrelationLen + kMaxLag;
+  static const size_t kCorrelationLen = 50;
+  static const size_t kLogCorrelationLen = 6;  // >= log2(kCorrelationLen).
+  static const size_t kMinLag = 10;
+  static const size_t kMaxLag = 60;
+  static const size_t kDownsampledLen = kCorrelationLen + kMaxLag;
   static const int kCorrelationThreshold = 14746;  // 0.9 in Q14.
 
   const int sample_rate_hz_;
   const int fs_mult_;  // Sample rate multiplier = sample_rate_hz_ / 8000.
-  const int num_channels_;
+  const size_t num_channels_;
   const size_t master_channel_;
   const BackgroundNoise& background_noise_;
   int16_t max_input_value_;
@@ -102,9 +107,9 @@ class TimeStretch {
 
   // Performs a simple voice-activity detection based on the input parameters.
   bool SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
-                       int peak_index, int scaling) const;
+                       size_t peak_index, int scaling) const;
 
-  DISALLOW_COPY_AND_ASSIGN(TimeStretch);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
index 3d1e06936a..0769fd34b7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
@@ -13,14 +13,24 @@
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 
+#include <map>
+
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
+namespace {
+const size_t kNumChannels = 1;
+}
+
 TEST(TimeStretch, CreateAndDestroy) {
   const int kSampleRate = 8000;
-  const size_t kNumChannels = 1;
   const int kOverlapSamples = 5 * kSampleRate / 8000;
   BackgroundNoise bgn(kNumChannels);
   Accelerate accelerate(kSampleRate, kNumChannels, bgn);
@@ -30,7 +40,6 @@ TEST(TimeStretch, CreateAndDestroy) {
 
 TEST(TimeStretch, CreateUsingFactory) {
   const int kSampleRate = 8000;
-  const size_t kNumChannels = 1;
   const int kOverlapSamples = 5 * kSampleRate / 8000;
   BackgroundNoise bgn(kNumChannels);
 
@@ -47,6 +56,72 @@ TEST(TimeStretch, CreateUsingFactory) {
   delete preemptive_expand;
 }
 
-// TODO(hlundin): Write more tests.
+class TimeStretchTest : public ::testing::Test {
+ protected:
+  TimeStretchTest()
+      : input_file_(new test::InputAudioFile(
+            test::ResourcePath("audio_coding/testfile32kHz", "pcm"))),
+        sample_rate_hz_(32000),
+        block_size_(30 * sample_rate_hz_ / 1000),  // 30 ms
+        audio_(new int16_t[block_size_]),
+        background_noise_(kNumChannels) {
+    WebRtcSpl_Init();
+  }
+
+  const int16_t* Next30Ms() {
+    RTC_CHECK(input_file_->Read(block_size_, audio_.get()));
+    return audio_.get();
+  }
+
+  // Returns the total length change (in samples) that the accelerate operation
+  // resulted in during the run.
+  size_t TestAccelerate(size_t loops, bool fast_mode) {
+    Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_);
+    size_t total_length_change = 0;
+    for (size_t i = 0; i < loops; ++i) {
+      AudioMultiVector output(kNumChannels);
+      size_t length_change;
+      UpdateReturnStats(accelerate.Process(Next30Ms(), block_size_, fast_mode,
+                                           &output, &length_change));
+      total_length_change += length_change;
+    }
+    return total_length_change;
+  }
+
+  void UpdateReturnStats(TimeStretch::ReturnCodes ret) {
+    switch (ret) {
+      case TimeStretch::kSuccess:
+      case TimeStretch::kSuccessLowEnergy:
+      case TimeStretch::kNoStretch:
+        ++return_stats_[ret];
+        break;
+      case TimeStretch::kError:
+        FAIL() << "Process returned an error";
+    }
+  }
+
+  rtc::scoped_ptr<test::InputAudioFile> input_file_;
+  const int sample_rate_hz_;
+  const size_t block_size_;
+  rtc::scoped_ptr<int16_t[]> audio_;
+  std::map<TimeStretch::ReturnCodes, int> return_stats_;
+  BackgroundNoise background_noise_;
+};
+
+TEST_F(TimeStretchTest, Accelerate) {
+  // TestAccelerate returns the total length change in samples.
+  EXPECT_EQ(15268U, TestAccelerate(100, false));
+  EXPECT_EQ(9, return_stats_[TimeStretch::kSuccess]);
+  EXPECT_EQ(58, return_stats_[TimeStretch::kSuccessLowEnergy]);
+  EXPECT_EQ(33, return_stats_[TimeStretch::kNoStretch]);
+}
+
+TEST_F(TimeStretchTest, AccelerateFastMode) {
+  // TestAccelerate returns the total length change in samples.
+  EXPECT_EQ(21400U, TestAccelerate(100, true));
+  EXPECT_EQ(31, return_stats_[TimeStretch::kSuccess]);
+  EXPECT_EQ(58, return_stats_[TimeStretch::kSuccessLowEnergy]);
+  EXPECT_EQ(11, return_stats_[TimeStretch::kNoStretch]);
+}
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
index 11d5a20dd6..c1abdc30f5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -12,10 +12,14 @@
 
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
+void TimestampScaler::Reset() {
+  first_packet_received_ = false;
+}
+
 void TimestampScaler::ToInternal(Packet* packet) {
   if (!packet) {
     return;
@@ -40,28 +44,19 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
     return external_timestamp;
   }
   switch (info->codec_type) {
-    case kDecoderG722:
-    case kDecoderG722_2ch: {
+    case NetEqDecoder::kDecoderG722:
+    case NetEqDecoder::kDecoderG722_2ch: {
       // Use timestamp scaling with factor 2 (two output samples per RTP
       // timestamp).
       numerator_ = 2;
       denominator_ = 1;
       break;
     }
-    case kDecoderISACfb:
-    case kDecoderCNGswb48kHz: {
-      // Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP
-      // timestamps run on 48 kHz).
-      // TODO(tlegrand): Remove scaling for kDecoderCNGswb48kHz once ACM has
-      // full 48 kHz support.
-      numerator_ = 2;
-      denominator_ = 3;
-      break;
-    }
-    case kDecoderAVT:
-    case kDecoderCNGnb:
-    case kDecoderCNGwb:
-    case kDecoderCNGswb32kHz: {
+    case NetEqDecoder::kDecoderAVT:
+    case NetEqDecoder::kDecoderCNGnb:
+    case NetEqDecoder::kDecoderCNGwb:
+    case NetEqDecoder::kDecoderCNGswb32kHz:
+    case NetEqDecoder::kDecoderCNGswb48kHz: {
       // Do not change the timestamp scaling settings for DTMF or CNG.
       break;
     }
@@ -84,8 +79,6 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
     assert(denominator_ > 0);  // Should not be possible.
     external_ref_ = external_timestamp;
     internal_ref_ += (external_diff * numerator_) / denominator_;
-    LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
-        " -> " << internal_ref_;
     return internal_ref_;
   } else {
     // No scaling.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
index 59b8cc7d1d..9129d843bf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
@@ -36,7 +36,7 @@ class TimestampScaler {
   virtual ~TimestampScaler() {}
 
   // Start over.
-  virtual void Reset() { first_packet_received_ = false; }
+  virtual void Reset();
 
   // Scale the timestamp in |packet| from external to internal.
   virtual void ToInternal(Packet* packet);
@@ -61,7 +61,7 @@ class TimestampScaler {
   uint32_t internal_ref_;
   const DecoderDatabase& decoder_database_;
 
-  DISALLOW_COPY_AND_ASSIGN(TimestampScaler);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TimestampScaler);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index 4b6d940aeb..b1cb45d201 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -24,7 +24,8 @@ namespace webrtc {
 TEST(TimestampScaler, TestNoScaling) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderPCMu;  // Does not use scaled timestamps.
+  info.codec_type =
+      NetEqDecoder::kDecoderPCMu;  // Does not use scaled timestamps.
   static const uint8_t kRtpPayloadType = 0;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -44,7 +45,8 @@ TEST(TimestampScaler, TestNoScaling) {
 TEST(TimestampScaler, TestNoScalingLargeStep) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderPCMu;  // Does not use scaled timestamps.
+  info.codec_type =
+      NetEqDecoder::kDecoderPCMu;  // Does not use scaled timestamps.
   static const uint8_t kRtpPayloadType = 0;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -69,7 +71,7 @@ TEST(TimestampScaler, TestNoScalingLargeStep) {
 TEST(TimestampScaler, TestG722) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderG722;  // Uses a factor 2 scaling.
+  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -93,7 +95,7 @@ TEST(TimestampScaler, TestG722) {
 TEST(TimestampScaler, TestG722LargeStep) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderG722;  // Uses a factor 2 scaling.
+  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -121,8 +123,9 @@ TEST(TimestampScaler, TestG722LargeStep) {
 TEST(TimestampScaler, TestG722WithCng) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info_g722, info_cng;
-  info_g722.codec_type = kDecoderG722;  // Uses a factor 2 scaling.
-  info_cng.codec_type = kDecoderCNGwb;
+  info_g722.codec_type =
+      NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
+  info_cng.codec_type = NetEqDecoder::kDecoderCNGwb;
   static const uint8_t kRtpPayloadTypeG722 = 17;
   static const uint8_t kRtpPayloadTypeCng = 13;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
@@ -162,7 +165,8 @@ TEST(TimestampScaler, TestG722WithCng) {
 TEST(TimestampScaler, TestG722Packet) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderG722;  // Does uses a factor 2 scaling.
+  info.codec_type =
+      NetEqDecoder::kDecoderG722;  // Does uses a factor 2 scaling.
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -190,7 +194,7 @@ TEST(TimestampScaler, TestG722Packet) {
 TEST(TimestampScaler, TestG722PacketList) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderG722;  // Uses a factor 2 scaling.
+  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -219,7 +223,7 @@ TEST(TimestampScaler, TestG722PacketList) {
 TEST(TimestampScaler, TestG722Reset) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderG722;  // Uses a factor 2 scaling.
+  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -259,7 +263,7 @@ TEST(TimestampScaler, TestG722Reset) {
 TEST(TimestampScaler, TestOpusLargeStep) {
   MockDecoderDatabase db;
   DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderOpus;
+  info.codec_type = NetEqDecoder::kDecoderOpus;
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -283,34 +287,6 @@ TEST(TimestampScaler, TestOpusLargeStep) {
   EXPECT_CALL(db, Die());  // Called when database object is deleted.
 }
 
-TEST(TimestampScaler, TestIsacFbLargeStep) {
-  MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = kDecoderISACfb;
-  static const uint8_t kRtpPayloadType = 17;
-  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
-      .WillRepeatedly(Return(&info));
-
-  TimestampScaler scaler(db);
-  // Test both sides of the timestamp wrap-around.
-  static const uint32_t kStep = 960;
-  uint32_t external_timestamp = 0;
-  // |external_timestamp| will be a large positive value.
-  external_timestamp = external_timestamp - 5 * kStep;
-  uint32_t internal_timestamp = external_timestamp;
-  for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
-    // Scale to internal timestamp.
-    EXPECT_EQ(internal_timestamp,
-              scaler.ToInternal(external_timestamp, kRtpPayloadType));
-    // Scale back.
-    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
-    // Internal timestamp should be incremented with two-thirds the step.
-    internal_timestamp += 2 * kStep / 3;
-  }
-
-  EXPECT_CALL(db, Die());  // Called when database object is deleted.
-}
-
 TEST(TimestampScaler, Failures) {
   static const uint8_t kRtpPayloadType = 17;
   MockDecoderDatabase db;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
index b4a6a817b4..a302cff908 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -51,7 +51,7 @@ class AudioChecksum : public AudioSink {
   char checksum_result_[rtc::Md5Digest::kSize];
   bool finished_;
 
-  DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
index 2d2a7e3dd4..eed95753f0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -43,13 +43,14 @@ bool AudioLoop::Init(const std::string file_name,
   return true;
 }
 
-const int16_t* AudioLoop::GetNextBlock() {
+rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
   // Check that the AudioLoop is initialized.
-  if (block_length_samples_ == 0) return NULL;
+  if (block_length_samples_ == 0)
+    return rtc::ArrayView<const int16_t>();
 
   const int16_t* output_ptr = &audio_array_[next_index_];
   next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
-  return output_ptr;
+  return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
 }
 
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
index 87ff688738..14e20f68ac 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -13,6 +13,7 @@
 
 #include <string>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
@@ -40,10 +41,9 @@ class AudioLoop {
   bool Init(const std::string file_name, size_t max_loop_length_samples,
             size_t block_length_samples);
 
-  // Returns a pointer to the next block of audio. The number given as
-  // |block_length_samples| to the Init() function determines how many samples
-  // that can be safely read from the pointer.
-  const int16_t* GetNextBlock();
+  // Returns a (pointer,size) pair for the next block of audio. The size is
+  // equal to the |block_length_samples| Init() argument.
+  rtc::ArrayView<const int16_t> GetNextBlock();
 
  private:
   size_t next_index_;
@@ -51,7 +51,7 @@ class AudioLoop {
   size_t block_length_samples_;
   rtc::scoped_ptr<int16_t[]> audio_array_;
 
-  DISALLOW_COPY_AND_ASSIGN(AudioLoop);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioLoop);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
index b7b3ed1115..489a8b2ad8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -38,7 +38,7 @@ class AudioSink {
   }
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(AudioSink);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
 };
 
 // Forks the output audio to two AudioSink objects.
@@ -56,7 +56,7 @@ class AudioSinkFork : public AudioSink {
   AudioSink* left_sink_;
   AudioSink* right_sink_;
 
-  DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
index 65c4e9dc82..5a9f79f877 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -13,7 +13,7 @@
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 
 namespace webrtc {
@@ -31,12 +31,12 @@ ConstantPcmPacketSource::ConstantPcmPacketSource(size_t payload_len_samples,
       seq_number_(0),
       timestamp_(0),
       payload_ssrc_(0xABCD1234) {
-  int encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_);
-  CHECK_EQ(encoded_len, 2);
+  size_t encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_);
+  RTC_CHECK_EQ(2U, encoded_len);
 }
 
 Packet* ConstantPcmPacketSource::NextPacket() {
-  CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
+  RTC_CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
   uint8_t* packet_memory = new uint8_t[packet_len_bytes_];
   // Fill the payload part of the packet memory with the pre-encoded value.
   for (unsigned i = 0; i < 2 * payload_len_samples_; ++i)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
index b780fbfac1..6972303541 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -49,7 +49,7 @@ class ConstantPcmPacketSource : public PacketSource {
   uint32_t timestamp_;
   const uint32_t payload_ssrc_;
 
-  DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ConstantPcmPacketSource);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
index 6bbb3286e4..76f31096db 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 
+#include "webrtc/base/checks.h"
+
 namespace webrtc {
 namespace test {
 
@@ -37,6 +39,27 @@ bool InputAudioFile::Read(size_t samples, int16_t* destination) {
   return true;
 }
 
+bool InputAudioFile::Seek(int samples) {
+  if (!fp_) {
+    return false;
+  }
+  // Find file boundaries.
+  const long current_pos = ftell(fp_);
+  RTC_CHECK_NE(EOF, current_pos)
+      << "Error returned when getting file position.";
+  RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END));  // Move to end of file.
+  const long file_size = ftell(fp_);
+  RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
+  // Find new position.
+  long new_pos = current_pos + sizeof(int16_t) * samples;  // Samples to bytes.
+  RTC_CHECK_GE(new_pos, 0)
+      << "Trying to move to before the beginning of the file";
+  new_pos = new_pos % file_size;  // Wrap around the end of the file.
+  // Move to new position relative to the beginning of the file.
+  RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
+  return true;
+}
+
 void InputAudioFile::DuplicateInterleaved(const int16_t* source, size_t samples,
                                           size_t channels,
                                           int16_t* destination) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
index f546119be2..a6e12db24d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -32,7 +32,13 @@ class InputAudioFile {
   // if the read was successful, otherwise false. If the file end is reached,
   // the file is rewound and reading continues from the beginning.
   // The output |destination| must have the capacity to hold |samples| elements.
-  bool Read(size_t samples, int16_t* destination);
+  virtual bool Read(size_t samples, int16_t* destination);
+
+  // Fast-forwards (|samples| > 0) or -backwards (|samples| < 0) the file by the
+  // indicated number of samples. Just like Read(), Seek() starts over at the
+  // beginning of the file if the end is reached. However, seeking backwards
+  // past the beginning of the file is not possible.
+  virtual bool Seek(int samples);
 
   // Creates a multi-channel signal from a mono signal. Each sample is repeated
   // |channels| times to create an interleaved multi-channel signal where all
@@ -44,7 +50,7 @@ class InputAudioFile {
 
  private:
   FILE* fp_;
-  DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
+  RTC_DISALLOW_COPY_AND_ASSIGN(InputAudioFile);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 3eb4a29805..694b9ed153 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -12,6 +12,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 
 namespace webrtc {
 namespace test {
@@ -21,34 +22,33 @@ NetEqExternalDecoderTest::NetEqExternalDecoderTest(NetEqDecoder codec,
     : codec_(codec),
       decoder_(decoder),
       sample_rate_hz_(CodecSampleRateHz(codec_)),
-      channels_(static_cast<int>(decoder_->Channels())) {
+      channels_(decoder_->Channels()) {
   NetEq::Config config;
   config.sample_rate_hz = sample_rate_hz_;
   neteq_.reset(NetEq::Create(config));
-  printf("%d\n", channels_);
+  printf("%" PRIuS "\n", channels_);
 }
 
 void NetEqExternalDecoderTest::Init() {
   ASSERT_EQ(NetEq::kOK,
-            neteq_->RegisterExternalDecoder(decoder_, codec_, kPayloadType));
+            neteq_->RegisterExternalDecoder(decoder_, codec_, name_,
+                                            kPayloadType, sample_rate_hz_));
 }
 
-void NetEqExternalDecoderTest::InsertPacket(WebRtcRTPHeader rtp_header,
-                                            const uint8_t* payload,
-                                            size_t payload_size_bytes,
-                                            uint32_t receive_timestamp) {
-  ASSERT_EQ(
-      NetEq::kOK,
-      neteq_->InsertPacket(
-          rtp_header, payload, payload_size_bytes, receive_timestamp));
+void NetEqExternalDecoderTest::InsertPacket(
+    WebRtcRTPHeader rtp_header,
+    rtc::ArrayView<const uint8_t> payload,
+    uint32_t receive_timestamp) {
+  ASSERT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(rtp_header, payload, receive_timestamp));
 }
 
-int NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
-                                             int16_t* output,
-                                             NetEqOutputType* output_type) {
+size_t NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
+                                                int16_t* output,
+                                                NetEqOutputType* output_type) {
   // Get audio from regular instance.
-  int samples_per_channel;
-  int num_channels;
+  size_t samples_per_channel;
+  size_t num_channels;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(max_length,
                              output,
@@ -56,7 +56,9 @@ int NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
                              &num_channels,
                              output_type));
   EXPECT_EQ(channels_, num_channels);
-  EXPECT_EQ(kOutputLengthMs * sample_rate_hz_ / 1000, samples_per_channel);
+  EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
+            samples_per_channel);
+  EXPECT_EQ(sample_rate_hz_, neteq_->last_output_sample_rate_hz());
   return samples_per_channel;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 0d4d2f9037..d7b01fe33a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -11,10 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
 
+#include <string>
+
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
@@ -36,22 +38,23 @@ class NetEqExternalDecoderTest {
   // |payload_size_bytes| bytes. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
-  virtual void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
-                            size_t payload_size_bytes,
+  virtual void InsertPacket(WebRtcRTPHeader rtp_header,
+                            rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp);
 
   // Get 10 ms of audio data. The data is written to |output|, which can hold
   // (at least) |max_length| elements. Returns number of samples.
-  int GetOutputAudio(size_t max_length, int16_t* output,
-                     NetEqOutputType* output_type);
+  size_t GetOutputAudio(size_t max_length, int16_t* output,
+                        NetEqOutputType* output_type);
 
   NetEq* neteq() { return neteq_.get(); }
 
  private:
   NetEqDecoder codec_;
+  std::string name_ = "dummy name";
   AudioDecoder* decoder_;
   int sample_rate_hz_;
-  int channels_;
+  size_t channels_;
   rtc::scoped_ptr<NetEq> neteq_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 080b99bf03..7d1f9f9798 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -10,11 +10,11 @@
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -32,7 +32,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
   const int kSampRateHz = 32000;
-  const webrtc::NetEqDecoder kDecoderType = webrtc::kDecoderPCM16Bswb32kHz;
+  const webrtc::NetEqDecoder kDecoderType =
+      webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz;
+  const std::string kDecoderName = "pcm16-swb32";
   const int kPayloadType = 95;
 
   // Initialize NetEq instance.
@@ -40,7 +42,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   config.sample_rate_hz = kSampRateHz;
   NetEq* neteq = NetEq::Create(config);
   // Register decoder in |neteq|.
-  if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0)
+  if (neteq->RegisterPayloadType(kDecoderType, kDecoderName, kPayloadType) != 0)
     return -1;
 
   // Set up AudioLoop object.
@@ -61,12 +63,13 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   bool drift_flipped = false;
   int32_t packet_input_time_ms =
       rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
-  const int16_t* input_samples = audio_loop.GetNextBlock();
-  if (!input_samples) exit(1);
+  auto input_samples = audio_loop.GetNextBlock();
+  if (input_samples.empty())
+    exit(1);
   uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
-  size_t payload_len =
-      WebRtcPcm16b_Encode(input_samples, kInputBlockSizeSamples, input_payload);
-  assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
+  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                           input_samples.size(), input_payload);
+  RTC_CHECK_EQ(sizeof(input_payload), payload_len);
 
   // Main loop.
   webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
@@ -80,9 +83,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
       }
       if (!lost) {
         // Insert packet.
-        int error = neteq->InsertPacket(
-            rtp_header, input_payload, payload_len,
-            packet_input_time_ms * kSampRateHz / 1000);
+        int error =
+            neteq->InsertPacket(rtp_header, input_payload,
+                                packet_input_time_ms * kSampRateHz / 1000);
         if (error != NetEq::kOK)
           return -1;
       }
@@ -92,28 +95,28 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
                                                   kInputBlockSizeSamples,
                                                   &rtp_header);
       input_samples = audio_loop.GetNextBlock();
-      if (!input_samples) return -1;
-      payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
-                                        kInputBlockSizeSamples,
-                                        input_payload);
+      if (input_samples.empty())
+        return -1;
+      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                        input_samples.size(), input_payload);
       assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
     }
 
     // Get output audio, but don't do anything with it.
     static const int kMaxChannels = 1;
-    static const int kMaxSamplesPerMs = 48000 / 1000;
+    static const size_t kMaxSamplesPerMs = 48000 / 1000;
     static const int kOutputBlockSizeMs = 10;
-    static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
-        kMaxChannels;
+    static const size_t kOutDataLen =
+        kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
     int16_t out_data[kOutDataLen];
-    int num_channels;
-    int samples_per_channel;
+    size_t num_channels;
+    size_t samples_per_channel;
     int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                 &num_channels, NULL);
     if (error != NetEq::kOK)
       return -1;
 
-    assert(samples_per_channel == kSampRateHz * 10 / 1000);
+    assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000));
 
     time_now_ms += kOutputBlockSizeMs;
     if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 832885573c..9c64e0fb48 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -10,7 +10,14 @@
 
 #include <math.h>
 #include <stdio.h>
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using std::string;
 
 namespace webrtc {
 namespace test {
@@ -20,6 +27,75 @@ const int kOutputSizeMs = 10;
 const int kInitSeed = 0x12345678;
 const int kPacketLossTimeUnitMs = 10;
 
+// Common validator for file names.
+static bool ValidateFilename(const string& value, bool write) {
+  FILE* fid = write ? fopen(value.c_str(), "wb") : fopen(value.c_str(), "rb");
+  if (fid == nullptr)
+    return false;
+  fclose(fid);
+  return true;
+}
+
+// Define switch for input file name.
+static bool ValidateInFilename(const char* flagname, const string& value) {
+  if (!ValidateFilename(value, false)) {
+    printf("Invalid input filename.");
+    return false;
+  }
+  return true;
+}
+
+DEFINE_string(
+    in_filename,
+    ResourcePath("audio_coding/speech_mono_16kHz", "pcm"),
+    "Filename for input audio (specify sample rate with --input_sample_rate ,"
+    "and channels with --channels).");
+
+static const bool in_filename_dummy =
+    RegisterFlagValidator(&FLAGS_in_filename, &ValidateInFilename);
+
+// Define switch for sample rate.
+static bool ValidateSampleRate(const char* flagname, int32_t value) {
+  if (value == 8000 || value == 16000 || value == 32000 || value == 48000)
+    return true;
+  printf("Invalid sample rate should be 8000, 16000, 32000 or 48000 Hz.");
+  return false;
+}
+
+DEFINE_int32(input_sample_rate, 16000, "Sample rate of input file in Hz.");
+
+static const bool sample_rate_dummy =
+    RegisterFlagValidator(&FLAGS_input_sample_rate, &ValidateSampleRate);
+
+// Define switch for channels.
+static bool ValidateChannels(const char* flagname, int32_t value) {
+  if (value == 1)
+    return true;
+  printf("Invalid number of channels, current support only 1.");
+  return false;
+}
+
+DEFINE_int32(channels, 1, "Number of channels in input audio.");
+
+static const bool channels_dummy =
+    RegisterFlagValidator(&FLAGS_channels, &ValidateChannels);
+
+// Define switch for output file name.
+static bool ValidateOutFilename(const char* flagname, const string& value) {
+  if (!ValidateFilename(value, true)) {
+    printf("Invalid output filename.");
+    return false;
+  }
+  return true;
+}
+
+DEFINE_string(out_filename,
+              OutputPath() + "neteq_quality_test_out.pcm",
+              "Name of output audio file.");
+
+static const bool out_filename_dummy =
+    RegisterFlagValidator(&FLAGS_out_filename, &ValidateOutFilename);
+
 // Define switch for packet loss rate.
 static bool ValidatePacketLossRate(const char* /* flag_name */, int32_t value) {
   if (value >= 0 && value <= 100)
@@ -28,6 +104,19 @@ static bool ValidatePacketLossRate(const char* /* flag_name */, int32_t value) {
   return false;
 }
 
+// Define switch for runtime.
+static bool ValidateRuntime(const char* flagname, int32_t value) {
+  if (value > 0)
+    return true;
+  printf("Invalid runtime, should be greater than 0.");
+  return false;
+}
+
+DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");
+
+static const bool runtime_dummy =
+    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
+
 DEFINE_int32(packet_loss_rate, 10, "Percentile of packet loss.");
 
 static const bool packet_loss_rate_dummy =
@@ -119,32 +208,42 @@ static double ProbTrans00Solver(int units, double loss_rate,
 NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
                                    int in_sampling_khz,
                                    int out_sampling_khz,
-                                   enum NetEqDecoder decoder_type,
-                                   int channels,
-                                   std::string in_filename,
-                                   std::string out_filename)
-    : decoded_time_ms_(0),
+                                   NetEqDecoder decoder_type)
+    : decoder_type_(decoder_type),
+      channels_(static_cast<size_t>(FLAGS_channels)),
+      decoded_time_ms_(0),
       decodable_time_ms_(0),
       drift_factor_(FLAGS_drift_factor),
       packet_loss_rate_(FLAGS_packet_loss_rate),
       block_duration_ms_(block_duration_ms),
       in_sampling_khz_(in_sampling_khz),
       out_sampling_khz_(out_sampling_khz),
-      decoder_type_(decoder_type),
-      channels_(channels),
-      in_filename_(in_filename),
-      out_filename_(out_filename),
-      log_filename_(out_filename + ".log"),
-      in_size_samples_(in_sampling_khz_ * block_duration_ms_),
-      out_size_samples_(out_sampling_khz_ * kOutputSizeMs),
+      in_size_samples_(
+          static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
+      out_size_samples_(static_cast<size_t>(out_sampling_khz_ * kOutputSizeMs)),
       payload_size_bytes_(0),
       max_payload_bytes_(0),
-      in_file_(new InputAudioFile(in_filename_)),
-      out_file_(NULL),
-      log_file_(NULL),
-      rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0,
-                                      decodable_time_ms_)),
+      in_file_(new ResampleInputAudioFile(FLAGS_in_filename,
+                                          FLAGS_input_sample_rate,
+                                          in_sampling_khz * 1000)),
+      rtp_generator_(
+          new RtpGenerator(in_sampling_khz_, 0, 0, decodable_time_ms_)),
       total_payload_size_bytes_(0) {
+  const std::string out_filename = FLAGS_out_filename;
+  const std::string log_filename = out_filename + ".log";
+  log_file_.open(log_filename.c_str(), std::ofstream::out);
+  RTC_CHECK(log_file_.is_open());
+
+  if (out_filename.size() >= 4 &&
+      out_filename.substr(out_filename.size() - 4) == ".wav") {
+    // Open a wav file.
+    output_.reset(
+        new webrtc::test::OutputWavFile(out_filename, 1000 * out_sampling_khz));
+  } else {
+    // Open a pcm file.
+    output_.reset(new webrtc::test::OutputAudioFile(out_filename));
+  }
+
   NetEq::Config config;
   config.sample_rate_hz = out_sampling_khz_ * 1000;
   neteq_.reset(NetEq::Create(config));
@@ -154,6 +253,10 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
   out_data_.reset(new int16_t[out_size_samples_ * channels_]);
 }
 
+NetEqQualityTest::~NetEqQualityTest() {
+  log_file_.close();
+}
+
 bool NoLoss::Lost() {
   return false;
 }
@@ -189,10 +292,8 @@ bool GilbertElliotLoss::Lost() {
 }
 
 void NetEqQualityTest::SetUp() {
-  out_file_ = fopen(out_filename_.c_str(), "wb");
-  log_file_ = fopen(log_filename_.c_str(), "wt");
-  ASSERT_TRUE(out_file_ != NULL);
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType));
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(decoder_type_, "noname", kPayloadType));
   rtp_generator_->set_drift_factor(drift_factor_);
 
   int units = block_duration_ms_ / kPacketLossTimeUnitMs;
@@ -245,8 +346,8 @@ void NetEqQualityTest::SetUp() {
   srand(kInitSeed);
 }
 
-void NetEqQualityTest::TearDown() {
-  fclose(out_file_);
+std::ofstream& NetEqQualityTest::Log() {
+  return log_file_;
 }
 
 bool NetEqQualityTest::PacketLost() {
@@ -270,25 +371,31 @@ int NetEqQualityTest::Transmit() {
   int packet_input_time_ms =
       rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
                                    &rtp_header_);
+  Log() << "Packet of size "
+        << payload_size_bytes_
+        << " bytes, for frame at "
+        << packet_input_time_ms
+        << " ms ";
   if (payload_size_bytes_ > 0) {
-    fprintf(log_file_, "Packet at %d ms", packet_input_time_ms);
     if (!PacketLost()) {
-      int ret = neteq_->InsertPacket(rtp_header_, &payload_[0],
-                                     payload_size_bytes_,
-                                     packet_input_time_ms * in_sampling_khz_);
+      int ret = neteq_->InsertPacket(
+          rtp_header_,
+          rtc::ArrayView<const uint8_t>(payload_.get(), payload_size_bytes_),
+          packet_input_time_ms * in_sampling_khz_);
       if (ret != NetEq::kOK)
         return -1;
-      fprintf(log_file_, " OK.\n");
+      Log() << "was sent.";
     } else {
-      fprintf(log_file_, " Lost.\n");
+      Log() << "was lost.";
     }
   }
+  Log() << std::endl;
   return packet_input_time_ms;
 }
 
 int NetEqQualityTest::DecodeBlock() {
-  int channels;
-  int samples;
+  size_t channels;
+  size_t samples;
   int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
                              &samples, &channels, NULL);
 
@@ -296,16 +403,16 @@ int NetEqQualityTest::DecodeBlock() {
     return -1;
   } else {
     assert(channels == channels_);
-    assert(samples == kOutputSizeMs * out_sampling_khz_);
-    fwrite(&out_data_[0], sizeof(int16_t), samples * channels, out_file_);
-    return samples;
+    assert(samples == static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
+    RTC_CHECK(output_->WriteArray(out_data_.get(), samples * channels));
+    return static_cast<int>(samples);
   }
 }
 
-void NetEqQualityTest::Simulate(int end_time_ms) {
+void NetEqQualityTest::Simulate() {
   int audio_size_samples;
 
-  while (decoded_time_ms_ < end_time_ms) {
+  while (decoded_time_ms_ < FLAGS_runtime_ms) {
     // Assume 10 packets in packets buffer.
     while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) {
       ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
@@ -320,7 +427,10 @@ void NetEqQualityTest::Simulate(int end_time_ms) {
       decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
     }
   }
-  fprintf(log_file_, "%f", 8.0f * total_payload_size_bytes_ / end_time_ms);
+  Log() << "Average bit rate was "
+        << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
+        << " kbps"
+        << std::endl;
 }
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index cb40b1cc82..c2b2effee2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -11,11 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
 
+#include <fstream>
 #include <gflags/gflags.h>
-#include <string>
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/typedefs.h"
@@ -65,20 +66,18 @@ class NetEqQualityTest : public ::testing::Test {
   NetEqQualityTest(int block_duration_ms,
                    int in_sampling_khz,
                    int out_sampling_khz,
-                   enum NetEqDecoder decoder_type,
-                   int channels,
-                   std::string in_filename,
-                   std::string out_filename);
+                   NetEqDecoder decoder_type);
+  virtual ~NetEqQualityTest();
+
   void SetUp() override;
-  void TearDown() override;
 
   // EncodeBlock(...) does the following:
   // 1. encodes a block of audio, saved in |in_data| and has a length of
   // |block_size_samples| (samples per channel),
   // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
   // 3. returns the length of the payload (in bytes),
-  virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
-                          uint8_t* payload, int max_bytes) = 0;
+  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
+                          uint8_t* payload, size_t max_bytes) = 0;
 
   // PacketLost(...) determines weather a packet sent at an indicated time gets
   // lost or not.
@@ -93,10 +92,14 @@ class NetEqQualityTest : public ::testing::Test {
   // |neteq_|.
   int Transmit();
 
-  // Simulate(...) runs encoding / transmitting / decoding up to |end_time_ms|
-  // (miliseconds), the resulted audio is stored in the file with the name of
-  // |out_filename_|.
-  void Simulate(int end_time_ms);
+  // Runs encoding / transmitting / decoding.
+  void Simulate();
+
+  // Write to log file. Usage Log() << ...
+  std::ofstream& Log();
+
+  NetEqDecoder decoder_type_;
+  const size_t channels_;
 
  private:
   int decoded_time_ms_;
@@ -106,24 +109,19 @@ class NetEqQualityTest : public ::testing::Test {
   const int block_duration_ms_;
   const int in_sampling_khz_;
   const int out_sampling_khz_;
-  const enum NetEqDecoder decoder_type_;
-  const int channels_;
-  const std::string in_filename_;
-  const std::string out_filename_;
-  const std::string log_filename_;
 
   // Number of samples per channel in a frame.
-  const int in_size_samples_;
+  const size_t in_size_samples_;
 
   // Expected output number of samples per channel in a frame.
-  const int out_size_samples_;
+  const size_t out_size_samples_;
 
   size_t payload_size_bytes_;
-  int max_payload_bytes_;
+  size_t max_payload_bytes_;
 
   rtc::scoped_ptr<InputAudioFile> in_file_;
-  FILE* out_file_;
-  FILE* log_file_;
+  rtc::scoped_ptr<AudioSink> output_;
+  std::ofstream log_file_;
 
   rtc::scoped_ptr<RtpGenerator> rtp_generator_;
   rtc::scoped_ptr<NetEq> neteq_;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 11dd20a8f9..3d79e5b5a2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -19,20 +19,24 @@
 
 #include <algorithm>
 #include <iostream>
+#include <limits>
 #include <string>
 
 #include "google/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -143,39 +147,39 @@ const bool hex_ssrc_dummy =
 // Maps a codec type to a printable name string.
 std::string CodecName(webrtc::NetEqDecoder codec) {
   switch (codec) {
-    case webrtc::kDecoderPCMu:
+    case webrtc::NetEqDecoder::kDecoderPCMu:
       return "PCM-u";
-    case webrtc::kDecoderPCMa:
+    case webrtc::NetEqDecoder::kDecoderPCMa:
       return "PCM-a";
-    case webrtc::kDecoderILBC:
+    case webrtc::NetEqDecoder::kDecoderILBC:
       return "iLBC";
-    case webrtc::kDecoderISAC:
+    case webrtc::NetEqDecoder::kDecoderISAC:
       return "iSAC";
-    case webrtc::kDecoderISACswb:
+    case webrtc::NetEqDecoder::kDecoderISACswb:
       return "iSAC-swb (32 kHz)";
-    case webrtc::kDecoderOpus:
+    case webrtc::NetEqDecoder::kDecoderOpus:
       return "Opus";
-    case webrtc::kDecoderPCM16B:
+    case webrtc::NetEqDecoder::kDecoderPCM16B:
       return "PCM16b-nb (8 kHz)";
-    case webrtc::kDecoderPCM16Bwb:
+    case webrtc::NetEqDecoder::kDecoderPCM16Bwb:
       return "PCM16b-wb (16 kHz)";
-    case webrtc::kDecoderPCM16Bswb32kHz:
+    case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz:
       return "PCM16b-swb32 (32 kHz)";
-    case webrtc::kDecoderPCM16Bswb48kHz:
+    case webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz:
       return "PCM16b-swb48 (48 kHz)";
-    case webrtc::kDecoderG722:
+    case webrtc::NetEqDecoder::kDecoderG722:
       return "G.722";
-    case webrtc::kDecoderRED:
+    case webrtc::NetEqDecoder::kDecoderRED:
       return "redundant audio (RED)";
-    case webrtc::kDecoderAVT:
+    case webrtc::NetEqDecoder::kDecoderAVT:
       return "AVT/DTMF";
-    case webrtc::kDecoderCNGnb:
+    case webrtc::NetEqDecoder::kDecoderCNGnb:
       return "comfort noise (8 kHz)";
-    case webrtc::kDecoderCNGwb:
+    case webrtc::NetEqDecoder::kDecoderCNGwb:
       return "comfort noise (16 kHz)";
-    case webrtc::kDecoderCNGswb32kHz:
+    case webrtc::NetEqDecoder::kDecoderCNGswb32kHz:
       return "comfort noise (32 kHz)";
-    case webrtc::kDecoderCNGswb48kHz:
+    case webrtc::NetEqDecoder::kDecoderCNGswb48kHz:
       return "comfort noise (48 kHz)";
     default:
       assert(false);
@@ -185,8 +189,9 @@ std::string CodecName(webrtc::NetEqDecoder codec) {
 
 void RegisterPayloadType(NetEq* neteq,
                          webrtc::NetEqDecoder codec,
+                         const std::string& name,
                          google::int32 flag) {
-  if (neteq->RegisterPayloadType(codec, static_cast<uint8_t>(flag))) {
+  if (neteq->RegisterPayloadType(codec, name, static_cast<uint8_t>(flag))) {
     std::cerr << "Cannot register payload type " << flag << " as "
               << CodecName(codec) << std::endl;
     exit(1);
@@ -196,25 +201,40 @@ void RegisterPayloadType(NetEq* neteq,
 // Registers all decoders in |neteq|.
 void RegisterPayloadTypes(NetEq* neteq) {
   assert(neteq);
-  RegisterPayloadType(neteq, webrtc::kDecoderPCMu, FLAGS_pcmu);
-  RegisterPayloadType(neteq, webrtc::kDecoderPCMa, FLAGS_pcma);
-  RegisterPayloadType(neteq, webrtc::kDecoderILBC, FLAGS_ilbc);
-  RegisterPayloadType(neteq, webrtc::kDecoderISAC, FLAGS_isac);
-  RegisterPayloadType(neteq, webrtc::kDecoderISACswb, FLAGS_isac_swb);
-  RegisterPayloadType(neteq, webrtc::kDecoderOpus, FLAGS_opus);
-  RegisterPayloadType(neteq, webrtc::kDecoderPCM16B, FLAGS_pcm16b);
-  RegisterPayloadType(neteq, webrtc::kDecoderPCM16Bwb, FLAGS_pcm16b_wb);
-  RegisterPayloadType(neteq, webrtc::kDecoderPCM16Bswb32kHz,
-                      FLAGS_pcm16b_swb32);
-  RegisterPayloadType(neteq, webrtc::kDecoderPCM16Bswb48kHz,
-                      FLAGS_pcm16b_swb48);
-  RegisterPayloadType(neteq, webrtc::kDecoderG722, FLAGS_g722);
-  RegisterPayloadType(neteq, webrtc::kDecoderAVT, FLAGS_avt);
-  RegisterPayloadType(neteq, webrtc::kDecoderRED, FLAGS_red);
-  RegisterPayloadType(neteq, webrtc::kDecoderCNGnb, FLAGS_cn_nb);
-  RegisterPayloadType(neteq, webrtc::kDecoderCNGwb, FLAGS_cn_wb);
-  RegisterPayloadType(neteq, webrtc::kDecoderCNGswb32kHz, FLAGS_cn_swb32);
-  RegisterPayloadType(neteq, webrtc::kDecoderCNGswb48kHz, FLAGS_cn_swb48);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCMu, "pcmu",
+                      FLAGS_pcmu);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCMa, "pcma",
+                      FLAGS_pcma);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderILBC, "ilbc",
+                      FLAGS_ilbc);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderISAC, "isac",
+                      FLAGS_isac);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderISACswb, "isac-swb",
+                      FLAGS_isac_swb);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderOpus, "opus",
+                      FLAGS_opus);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16B, "pcm16-nb",
+                      FLAGS_pcm16b);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb",
+                      FLAGS_pcm16b_wb);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                      "pcm16-swb32", FLAGS_pcm16b_swb32);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz,
+                      "pcm16-swb48", FLAGS_pcm16b_swb48);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderG722, "g722",
+                      FLAGS_g722);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderAVT, "avt",
+                      FLAGS_avt);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderRED, "red",
+                      FLAGS_red);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGnb, "cng-nb",
+                      FLAGS_cn_nb);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGwb, "cng-wb",
+                      FLAGS_cn_wb);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGswb32kHz,
+                      "cng-swb32", FLAGS_cn_swb32);
+  RegisterPayloadType(neteq, webrtc::NetEqDecoder::kDecoderCNGswb48kHz,
+                      "cng-swb48", FLAGS_cn_swb48);
 }
 
 void PrintCodecMappingEntry(webrtc::NetEqDecoder codec, google::int32 flag) {
@@ -222,23 +242,28 @@ void PrintCodecMappingEntry(webrtc::NetEqDecoder codec, google::int32 flag) {
 }
 
 void PrintCodecMapping() {
-  PrintCodecMappingEntry(webrtc::kDecoderPCMu, FLAGS_pcmu);
-  PrintCodecMappingEntry(webrtc::kDecoderPCMa, FLAGS_pcma);
-  PrintCodecMappingEntry(webrtc::kDecoderILBC, FLAGS_ilbc);
-  PrintCodecMappingEntry(webrtc::kDecoderISAC, FLAGS_isac);
-  PrintCodecMappingEntry(webrtc::kDecoderISACswb, FLAGS_isac_swb);
-  PrintCodecMappingEntry(webrtc::kDecoderOpus, FLAGS_opus);
-  PrintCodecMappingEntry(webrtc::kDecoderPCM16B, FLAGS_pcm16b);
-  PrintCodecMappingEntry(webrtc::kDecoderPCM16Bwb, FLAGS_pcm16b_wb);
-  PrintCodecMappingEntry(webrtc::kDecoderPCM16Bswb32kHz, FLAGS_pcm16b_swb32);
-  PrintCodecMappingEntry(webrtc::kDecoderPCM16Bswb48kHz, FLAGS_pcm16b_swb48);
-  PrintCodecMappingEntry(webrtc::kDecoderG722, FLAGS_g722);
-  PrintCodecMappingEntry(webrtc::kDecoderAVT, FLAGS_avt);
-  PrintCodecMappingEntry(webrtc::kDecoderRED, FLAGS_red);
-  PrintCodecMappingEntry(webrtc::kDecoderCNGnb, FLAGS_cn_nb);
-  PrintCodecMappingEntry(webrtc::kDecoderCNGwb, FLAGS_cn_wb);
-  PrintCodecMappingEntry(webrtc::kDecoderCNGswb32kHz, FLAGS_cn_swb32);
-  PrintCodecMappingEntry(webrtc::kDecoderCNGswb48kHz, FLAGS_cn_swb48);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderPCMu, FLAGS_pcmu);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderPCMa, FLAGS_pcma);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderILBC, FLAGS_ilbc);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderISAC, FLAGS_isac);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderISACswb, FLAGS_isac_swb);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderOpus, FLAGS_opus);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderPCM16B, FLAGS_pcm16b);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderPCM16Bwb,
+                         FLAGS_pcm16b_wb);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                         FLAGS_pcm16b_swb32);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz,
+                         FLAGS_pcm16b_swb48);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderG722, FLAGS_g722);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderAVT, FLAGS_avt);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderRED, FLAGS_red);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderCNGnb, FLAGS_cn_nb);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderCNGwb, FLAGS_cn_wb);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderCNGswb32kHz,
+                         FLAGS_cn_swb32);
+  PrintCodecMappingEntry(webrtc::NetEqDecoder::kDecoderCNGswb48kHz,
+                         FLAGS_cn_swb48);
 }
 
 bool IsComfortNoise(uint8_t payload_type) {
@@ -324,7 +349,7 @@ size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file,
     // Encode it as PCM16.
     assert((*payload).get());
     payload_len = WebRtcPcm16b_Encode((*replacement_audio).get(),
-                                      static_cast<int16_t>(*frame_size_samples),
+                                      *frame_size_samples,
                                       (*payload).get());
     assert(payload_len == 2 * *frame_size_samples);
     // Change payload type to PCM16.
@@ -358,7 +383,7 @@ size_t ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file,
 
 int main(int argc, char* argv[]) {
   static const int kMaxChannels = 5;
-  static const int kMaxSamplesPerMs = 48000 / 1000;
+  static const size_t kMaxSamplesPerMs = 48000 / 1000;
   static const int kOutputBlockSizeMs = 10;
 
   std::string program_name = argv[0];
@@ -384,14 +409,25 @@ int main(int argc, char* argv[]) {
   }
 
   printf("Input file: %s\n", argv[1]);
-  rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
-      webrtc::test::RtpFileSource::Create(argv[1]));
+
+  bool is_rtp_dump = false;
+  rtc::scoped_ptr<webrtc::test::PacketSource> file_source;
+  webrtc::test::RtcEventLogSource* event_log_source = nullptr;
+  if (webrtc::test::RtpFileSource::ValidRtpDump(argv[1]) ||
+      webrtc::test::RtpFileSource::ValidPcap(argv[1])) {
+    is_rtp_dump = true;
+    file_source.reset(webrtc::test::RtpFileSource::Create(argv[1]));
+  } else {
+    event_log_source = webrtc::test::RtcEventLogSource::Create(argv[1]);
+    file_source.reset(event_log_source);
+  }
+
   assert(file_source.get());
 
   // Check if an SSRC value was provided.
   if (!FLAGS_ssrc.empty()) {
     uint32_t ssrc;
-    CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
+    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed.";
     file_source->SelectSsrc(ssrc);
   }
 
@@ -413,7 +449,12 @@ int main(int argc, char* argv[]) {
     webrtc::Trace::ReturnTrace();
     return 0;
   }
-  bool packet_available = true;
+  if (packet->payload_length_bytes() == 0 && !replace_payload) {
+    std::cerr << "Warning: input file contains header-only packets, but no "
+              << "replacement file is specified." << std::endl;
+    webrtc::Trace::ReturnTrace();
+    return -1;
+  }
 
   // Check the sample rate.
   int sample_rate_hz = CodecSampleRate(packet->header().payloadType);
@@ -475,17 +516,29 @@ int main(int argc, char* argv[]) {
 
   // This is the main simulation loop.
   // Set the simulation clock to start immediately with the first packet.
-  int start_time_ms = packet->time_ms();
-  int time_now_ms = packet->time_ms();
-  int next_input_time_ms = time_now_ms;
-  int next_output_time_ms = time_now_ms;
+  int64_t start_time_ms = rtc::checked_cast<int64_t>(packet->time_ms());
+  int64_t time_now_ms = start_time_ms;
+  int64_t next_input_time_ms = time_now_ms;
+  int64_t next_output_time_ms = time_now_ms;
   if (time_now_ms % kOutputBlockSizeMs != 0) {
     // Make sure that next_output_time_ms is rounded up to the next multiple
     // of kOutputBlockSizeMs. (Legacy bit-exactness.)
     next_output_time_ms +=
         kOutputBlockSizeMs - time_now_ms % kOutputBlockSizeMs;
   }
-  while (packet_available) {
+
+  bool packet_available = true;
+  bool output_event_available = true;
+  if (!is_rtp_dump) {
+    next_output_time_ms = event_log_source->NextAudioOutputEventMs();
+    if (next_output_time_ms == std::numeric_limits<int64_t>::max())
+      output_event_available = false;
+    start_time_ms = time_now_ms =
+        std::min(next_input_time_ms, next_output_time_ms);
+  }
+  while (packet_available || output_event_available) {
+    // Advance time to next event.
+    time_now_ms = std::min(next_input_time_ms, next_output_time_ms);
     // Check if it is time to insert packet.
     while (time_now_ms >= next_input_time_ms && packet_available) {
       assert(packet->virtual_payload_length_bytes() > 0);
@@ -504,11 +557,9 @@ int main(int argc, char* argv[]) {
                                      next_packet.get());
         payload_ptr = payload.get();
       }
-      int error =
-          neteq->InsertPacket(rtp_header,
-                              payload_ptr,
-                              payload_len,
-                              packet->time_ms() * sample_rate_hz / 1000);
+      int error = neteq->InsertPacket(
+          rtp_header, rtc::ArrayView<const uint8_t>(payload_ptr, payload_len),
+          static_cast<uint32_t>(packet->time_ms() * sample_rate_hz / 1000));
       if (error != NetEq::kOK) {
         if (neteq->LastError() == NetEq::kUnknownRtpPayloadType) {
           std::cerr << "RTP Payload type "
@@ -534,29 +585,32 @@ int main(int argc, char* argv[]) {
       webrtc::test::Packet* temp_packet = file_source->NextPacket();
       if (temp_packet) {
         packet.reset(temp_packet);
+        if (replace_payload) {
+          // At this point |packet| contains the packet *after* |next_packet|.
+          // Swap Packet objects between |packet| and |next_packet|.
+          packet.swap(next_packet);
+          // Swap the status indicators unless they're already the same.
+          if (packet_available != next_packet_available) {
+            packet_available = !packet_available;
+            next_packet_available = !next_packet_available;
+          }
+        }
+        next_input_time_ms = rtc::checked_cast<int64_t>(packet->time_ms());
       } else {
+        // Set next input time to the maximum value of int64_t to prevent the
+        // time_now_ms from becoming stuck at the final value.
+        next_input_time_ms = std::numeric_limits<int64_t>::max();
         packet_available = false;
       }
-      if (replace_payload) {
-        // At this point |packet| contains the packet *after* |next_packet|.
-        // Swap Packet objects between |packet| and |next_packet|.
-        packet.swap(next_packet);
-        // Swap the status indicators unless they're already the same.
-        if (packet_available != next_packet_available) {
-          packet_available = !packet_available;
-          next_packet_available = !next_packet_available;
-        }
-      }
-      next_input_time_ms = packet->time_ms();
     }
 
     // Check if it is time to get output audio.
-    if (time_now_ms >= next_output_time_ms) {
-      static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
-          kMaxChannels;
+    while (time_now_ms >= next_output_time_ms && output_event_available) {
+      static const size_t kOutDataLen =
+          kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
       int16_t out_data[kOutDataLen];
-      int num_channels;
-      int samples_per_channel;
+      size_t num_channels;
+      size_t samples_per_channel;
       int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                    &num_channels, NULL);
       if (error != NetEq::kOK) {
@@ -564,7 +618,8 @@ int main(int argc, char* argv[]) {
             neteq->LastError() << std::endl;
       } else {
         // Calculate sample rate from output size.
-        sample_rate_hz = 1000 * samples_per_channel / kOutputBlockSizeMs;
+        sample_rate_hz = rtc::checked_cast<int>(
+            1000 * samples_per_channel / kOutputBlockSizeMs);
       }
 
       // Write to file.
@@ -575,14 +630,20 @@ int main(int argc, char* argv[]) {
         webrtc::Trace::ReturnTrace();
         exit(1);
       }
-      next_output_time_ms += kOutputBlockSizeMs;
+      if (is_rtp_dump) {
+        next_output_time_ms += kOutputBlockSizeMs;
+        if (!packet_available)
+          output_event_available = false;
+      } else {
+        next_output_time_ms = event_log_source->NextAudioOutputEventMs();
+        if (next_output_time_ms == std::numeric_limits<int64_t>::max())
+          output_event_available = false;
+      }
     }
-    // Advance time to next event.
-    time_now_ms = std::min(next_input_time_ms, next_output_time_ms);
   }
-
   printf("Simulation done\n");
-  printf("Produced %i ms of audio\n", time_now_ms - start_time_ms);
+  printf("Produced %i ms of audio\n",
+         static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
index ff30f673d5..a9142a63c1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -42,7 +42,7 @@ class OutputAudioFile : public AudioSink {
  private:
   FILE* out_file_;
 
-  DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
index 1b1ed42829..c36c7da983 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -35,7 +35,7 @@ class OutputWavFile : public AudioSink {
  private:
   WavWriter wav_writer_;
 
-  DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.cc
index b8b27afdec..2b2fcc286e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -12,8 +12,8 @@
 
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
 namespace test {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.h
index a4e48d8953..8e43633423 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet.h
@@ -114,7 +114,7 @@ class Packet {
   double time_ms_;     // Used to denote a packet's arrival time.
   bool valid_header_;  // Set by the RtpHeaderParser.
 
-  DISALLOW_COPY_AND_ASSIGN(Packet);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet_source.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet_source.h
index 968400c215..804a94dc49 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet_source.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/packet_source.h
@@ -46,7 +46,7 @@ class PacketSource {
   uint32_t ssrc_;  // The selected SSRC. All other SSRCs will be discarded.
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(PacketSource);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
index 74d593e6ac..7a0bb1a6af 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
@@ -20,22 +20,28 @@ bool ResampleInputAudioFile::Read(size_t samples,
                                   int output_rate_hz,
                                   int16_t* destination) {
   const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
-  CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
+  RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
       << "Frame size and sample rates don't add up to an integer.";
   rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
   if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
     return false;
   resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
-  int output_length = 0;
-  CHECK_EQ(resampler_.Push(temp_destination.get(),
-                           static_cast<int>(samples_to_read),
-                           destination,
-                           static_cast<int>(samples),
-                           output_length),
-           0);
-  CHECK_EQ(static_cast<int>(samples), output_length);
+  size_t output_length = 0;
+  RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
+                               destination, samples, output_length),
+               0);
+  RTC_CHECK_EQ(samples, output_length);
   return true;
 }
 
+bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
+  RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
+  return Read(samples, output_rate_hz_, destination);
+}
+
+void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
+  output_rate_hz_ = rate_hz;
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
index 8c028005cb..c0af3546b0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
@@ -25,14 +25,25 @@ namespace test {
 class ResampleInputAudioFile : public InputAudioFile {
  public:
   ResampleInputAudioFile(const std::string file_name, int file_rate_hz)
-      : InputAudioFile(file_name), file_rate_hz_(file_rate_hz) {}
+      : InputAudioFile(file_name),
+        file_rate_hz_(file_rate_hz),
+        output_rate_hz_(-1) {}
+  ResampleInputAudioFile(const std::string file_name,
+                         int file_rate_hz,
+                         int output_rate_hz)
+      : InputAudioFile(file_name),
+        file_rate_hz_(file_rate_hz),
+        output_rate_hz_(output_rate_hz) {}
 
   bool Read(size_t samples, int output_rate_hz, int16_t* destination);
+  bool Read(size_t samples, int16_t* destination) override;
+  void set_output_rate_hz(int rate_hz);
 
  private:
   const int file_rate_hz_;
+  int output_rate_hz_;
   Resampler resampler_;
-  DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
new file mode 100644
index 0000000000..dad72eaecd
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+
+#include <assert.h>
+#include <string.h>
+#include <iostream>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) {
+  if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT)
+    return nullptr;
+  if (!event.has_timestamp_us() || !event.has_rtp_packet())
+    return nullptr;
+  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+  if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
+      !rtp_packet.has_incoming() || !rtp_packet.incoming() ||
+      !rtp_packet.has_packet_length() || rtp_packet.packet_length() == 0 ||
+      !rtp_packet.has_header() || rtp_packet.header().size() == 0 ||
+      rtp_packet.packet_length() < rtp_packet.header().size())
+    return nullptr;
+  return &rtp_packet;
+}
+
+const rtclog::AudioPlayoutEvent* GetAudioPlayoutEvent(
+    const rtclog::Event& event) {
+  if (!event.has_type() || event.type() != rtclog::Event::AUDIO_PLAYOUT_EVENT)
+    return nullptr;
+  if (!event.has_timestamp_us() || !event.has_audio_playout_event())
+    return nullptr;
+  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
+  if (!playout_event.has_local_ssrc())
+    return nullptr;
+  return &playout_event;
+}
+
+}  // namespace
+
+RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
+  RtcEventLogSource* source = new RtcEventLogSource();
+  RTC_CHECK(source->OpenFile(file_name));
+  return source;
+}
+
+RtcEventLogSource::~RtcEventLogSource() {}
+
+bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
+                                                   uint8_t id) {
+  RTC_CHECK(parser_.get());
+  return parser_->RegisterRtpHeaderExtension(type, id);
+}
+
+Packet* RtcEventLogSource::NextPacket() {
+  while (rtp_packet_index_ < event_log_->stream_size()) {
+    const rtclog::Event& event = event_log_->stream(rtp_packet_index_);
+    const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event);
+    rtp_packet_index_++;
+    if (rtp_packet) {
+      uint8_t* packet_header = new uint8_t[rtp_packet->header().size()];
+      memcpy(packet_header, rtp_packet->header().data(),
+             rtp_packet->header().size());
+      Packet* packet = new Packet(packet_header, rtp_packet->header().size(),
+                                  rtp_packet->packet_length(),
+                                  event.timestamp_us() / 1000, *parser_.get());
+      if (packet->valid_header()) {
+        // Check if the packet should not be filtered out.
+        if (!filter_.test(packet->header().payloadType) &&
+            !(use_ssrc_filter_ && packet->header().ssrc != ssrc_))
+          return packet;
+      } else {
+        std::cout << "Warning: Packet with index " << (rtp_packet_index_ - 1)
+                  << " has an invalid header and will be ignored." << std::endl;
+      }
+      // The packet has either an invalid header or needs to be filtered out, so
+      // it can be deleted.
+      delete packet;
+    }
+  }
+  return nullptr;
+}
+
+int64_t RtcEventLogSource::NextAudioOutputEventMs() {
+  while (audio_output_index_ < event_log_->stream_size()) {
+    const rtclog::Event& event = event_log_->stream(audio_output_index_);
+    const rtclog::AudioPlayoutEvent* playout_event =
+        GetAudioPlayoutEvent(event);
+    audio_output_index_++;
+    if (playout_event)
+      return event.timestamp_us() / 1000;
+  }
+  return std::numeric_limits<int64_t>::max();
+}
+
+RtcEventLogSource::RtcEventLogSource()
+    : PacketSource(), parser_(RtpHeaderParser::Create()) {}
+
+bool RtcEventLogSource::OpenFile(const std::string& file_name) {
+  event_log_.reset(new rtclog::EventStream());
+  return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
new file mode 100644
index 0000000000..90d5931224
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtpHeaderParser;
+
+namespace rtclog {
+class EventStream;
+}  // namespace rtclog
+
+namespace test {
+
+class Packet;
+
+class RtcEventLogSource : public PacketSource {
+ public:
+  // Creates an RtcEventLogSource reading from |file_name|. If the file cannot
+  // be opened, or has the wrong format, NULL will be returned.
+  static RtcEventLogSource* Create(const std::string& file_name);
+
+  virtual ~RtcEventLogSource();
+
+  // Registers an RTP header extension and binds it to |id|.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
+
+  // Returns a pointer to the next packet. Returns NULL if end of file was
+  // reached.
+  Packet* NextPacket() override;
+
+  // Returns the timestamp of the next audio output event, in milliseconds. The
+  // maximum value of int64_t is returned if there are no more audio output
+  // events available.
+  int64_t NextAudioOutputEventMs();
+
+ private:
+  RtcEventLogSource();
+
+  bool OpenFile(const std::string& file_name);
+
+  int rtp_packet_index_ = 0;
+  int audio_output_index_ = 0;
+
+  rtc::scoped_ptr<rtclog::EventStream> event_log_;
+  rtc::scoped_ptr<RtpHeaderParser> parser_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
index d062b386f6..78f0497fff 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -38,6 +38,9 @@ static const bool red_dummy =
 DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
 static const bool audio_level_dummy =
     google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
+DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
+static const bool abs_send_time_dummy =
+    google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
 
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
@@ -63,13 +66,19 @@ int main(int argc, char* argv[]) {
   rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
       webrtc::test::RtpFileSource::Create(argv[1]));
   assert(file_source.get());
-  // Set RTP extension ID.
+  // Set RTP extension IDs.
   bool print_audio_level = false;
   if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
     print_audio_level = true;
     file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
                                             FLAGS_audio_level);
   }
+  bool print_abs_send_time = false;
+  if (!google::GetCommandLineFlagInfoOrDie("abs_send_time").is_default) {
+    print_abs_send_time = true;
+    file_source->RegisterRtpHeaderExtension(
+        webrtc::kRtpExtensionAbsoluteSendTime, FLAGS_abs_send_time);
+  }
 
   FILE* out_file;
   if (argc == 3) {
@@ -88,8 +97,13 @@ int main(int argc, char* argv[]) {
   if (print_audio_level) {
     fprintf(out_file, " AuLvl (V)");
   }
+  if (print_abs_send_time) {
+    fprintf(out_file, " AbsSendTime");
+  }
   fprintf(out_file, "\n");
 
+  uint32_t max_abs_send_time = 0;
+  int cycles = -1;
   rtc::scoped_ptr<webrtc::test::Packet> packet;
   while (true) {
     packet.reset(file_source->NextPacket());
@@ -97,22 +111,51 @@ int main(int argc, char* argv[]) {
       // End of file reached.
       break;
     }
-    // Write packet data to file.
+    // Write packet data to file. Use virtual_packet_length_bytes so that the
+    // correct packet sizes are printed also for RTP header-only dumps.
     fprintf(out_file,
             "%5u %10u %10u %5i %5i %2i %#08X",
             packet->header().sequenceNumber,
             packet->header().timestamp,
             static_cast<unsigned int>(packet->time_ms()),
-            static_cast<int>(packet->packet_length_bytes()),
+            static_cast<int>(packet->virtual_packet_length_bytes()),
             packet->header().payloadType,
             packet->header().markerBit,
             packet->header().ssrc);
     if (print_audio_level && packet->header().extension.hasAudioLevel) {
-      // |audioLevel| consists of one bit for "V" and then 7 bits level.
       fprintf(out_file,
               " %5u (%1i)",
-              packet->header().extension.audioLevel & 0x7F,
-              (packet->header().extension.audioLevel & 0x80) == 0 ? 0 : 1);
+              packet->header().extension.audioLevel,
+              packet->header().extension.voiceActivity);
+    }
+    if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
+      if (cycles == -1) {
+        // Initialize.
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+        cycles = 0;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
+      // 32 bits (unsigned). Calculate the difference between this packet's
+      // send time and the maximum observed. Cast to signed 32-bit to get the
+      // desired wrap-around behavior.
+      if (static_cast<int32_t>(
+              (packet->header().extension.absoluteSendTime << 8) -
+              (max_abs_send_time << 8)) >= 0) {
+        // The difference is non-negative, meaning that this packet is newer
+        // than the previously observed maximum absolute send time.
+        if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
+          // Wrap detected.
+          cycles++;
+        }
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
+      // to floating point representation.
+      double send_time_seconds =
+          static_cast<double>(packet->header().extension.absoluteSendTime) /
+              262144 +
+          64.0 * cycles;
+      fprintf(out_file, " %11f", send_time_seconds);
     }
     fprintf(out_file, "\n");
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index f5d323ecf6..b7a3109c01 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -20,7 +20,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 namespace webrtc {
@@ -28,10 +28,22 @@ namespace test {
 
 RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
   RtpFileSource* source = new RtpFileSource();
-  CHECK(source->OpenFile(file_name));
+  RTC_CHECK(source->OpenFile(file_name));
   return source;
 }
 
+bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
+  rtc::scoped_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  return !!temp_file;
+}
+
+bool RtpFileSource::ValidPcap(const std::string& file_name) {
+  rtc::scoped_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  return !!temp_file;
+}
+
 RtpFileSource::~RtpFileSource() {
 }
 
@@ -47,7 +59,7 @@ Packet* RtpFileSource::NextPacket() {
     if (!rtp_reader_->NextPacket(&temp_packet)) {
       return NULL;
     }
-    if (temp_packet.length == 0) {
+    if (temp_packet.original_length == 0) {
       // May be an RTCP packet.
       // Read the next one.
       continue;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index d711685950..2febf68b91 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -18,7 +18,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -34,6 +34,10 @@ class RtpFileSource : public PacketSource {
   // opened, or has the wrong format, NULL will be returned.
   static RtpFileSource* Create(const std::string& file_name);
 
+  // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
+  static bool ValidRtpDump(const std::string& file_name);
+  static bool ValidPcap(const std::string& file_name);
+
   virtual ~RtpFileSource();
 
   // Registers an RTP header extension and binds it to |id|.
@@ -55,7 +59,7 @@ class RtpFileSource : public PacketSource {
   rtc::scoped_ptr<RtpFileReader> rtp_reader_;
   rtc::scoped_ptr<RtpHeaderParser> parser_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
index e09f6e4ca1..53371be8f6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -54,7 +54,7 @@ class RtpGenerator {
   double drift_factor_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
 };
 
 class TimestampJumpRtpGenerator : public RtpGenerator {
@@ -75,7 +75,7 @@ class TimestampJumpRtpGenerator : public RtpGenerator {
  private:
   uint32_t jump_from_timestamp_;
   uint32_t jump_to_timestamp_;
-  DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
index f7490de551..f2b87a5b95 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/tools/rtpcat.cc
@@ -28,18 +28,18 @@ int main(int argc, char* argv[]) {
 
   scoped_ptr<RtpFileWriter> output(
       RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1]));
-  CHECK(output.get() != NULL) << "Cannot open output file.";
+  RTC_CHECK(output.get() != NULL) << "Cannot open output file.";
   printf("Output RTP file: %s\n", argv[argc - 1]);
 
   for (int i = 1; i < argc - 1; i++) {
     scoped_ptr<RtpFileReader> input(
         RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
-    CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
+    RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
     printf("Input RTP file: %s\n", argv[i]);
 
     webrtc::test::RtpPacket packet;
     while (input->NextPacket(&packet))
-      CHECK(output->WritePacket(&packet));
+      RTC_CHECK(output->WritePacket(&packet));
   }
   return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ACMTest.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/ACMTest.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ACMTest.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/ACMTest.h
index f73961f5e5..d7e87d34ba 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ACMTest.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/ACMTest.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
 
 class ACMTest {
  public:
@@ -18,4 +18,4 @@ class ACMTest {
   virtual void Perform() = 0;
 };
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.cc
similarity index 81%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.cc
index e2ff938cab..bf04d7c825 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
 
 #include <ctype.h>
 #include <stdio.h>
@@ -20,15 +20,15 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -36,12 +36,6 @@ namespace webrtc {
 #define TEST_DURATION_SEC 600
 #define NUMBER_OF_SENDER_TESTS 6
 #define MAX_FILE_NAME_LENGTH_BYTE 500
-#define CHECK_THREAD_NULLITY(myThread, S)                                      \
-  if(myThread != NULL) {                                                       \
-    (myThread)->Start();                                                       \
-  } else {                                                                     \
-    ADD_FAILURE() << S;                                                        \
-  }
 
 void APITest::Wait(uint32_t waitLengthMs) {
   if (_randomTest) {
@@ -86,7 +80,6 @@ APITest::APITest(const Config& config)
       _dotMoveDirectionA(1),
       _dotPositionB(39),
       _dotMoveDirectionB(-1),
-      _dtmfCallback(NULL),
       _vadCallbackA(NULL),
       _vadCallbackB(NULL),
       _apiTestRWLock(*RWLockWrapper::CreateRWLock()),
@@ -125,7 +118,6 @@ APITest::~APITest() {
   _inFileB.Close();
   _outFileB.Close();
 
-  DELETE_POINTER(_dtmfCallback);
   DELETE_POINTER(_vadCallbackA);
   DELETE_POINTER(_vadCallbackB);
 
@@ -247,14 +239,14 @@ int16_t APITest::SetUp() {
 
   //--- EVENT TIMERS
   // A
-  _pullEventA = EventWrapper::Create();
-  _pushEventA = EventWrapper::Create();
-  _processEventA = EventWrapper::Create();
+  _pullEventA = EventTimerWrapper::Create();
+  _pushEventA = EventTimerWrapper::Create();
+  _processEventA = EventTimerWrapper::Create();
   _apiEventA = EventWrapper::Create();
   // B
-  _pullEventB = EventWrapper::Create();
-  _pushEventB = EventWrapper::Create();
-  _processEventB = EventWrapper::Create();
+  _pullEventB = EventTimerWrapper::Create();
+  _pushEventB = EventTimerWrapper::Create();
+  _processEventB = EventTimerWrapper::Create();
   _apiEventB = EventWrapper::Create();
 
   //--- I/O params
@@ -290,9 +282,6 @@ int16_t APITest::SetUp() {
     }
   }
 
-#ifdef WEBRTC_DTMF_DETECTION
-  _dtmfCallback = new DTMFDetector;
-#endif
   _vadCallbackA = new VADCallback;
   _vadCallbackB = new VADCallback;
 
@@ -429,7 +418,7 @@ void APITest::RunTest(char thread) {
   {
     WriteLockScoped cs(_apiTestRWLock);
     if (thread == 'A') {
-      _testNumA = (_testNumB + 1 + (rand() % 4)) % 5;
+      _testNumA = (_testNumB + 1 + (rand() % 3)) % 4;
       testNum = _testNumA;
 
       _movingDot[_dotPositionA] = ' ';
@@ -442,7 +431,7 @@ void APITest::RunTest(char thread) {
       _dotPositionA += _dotMoveDirectionA;
       _movingDot[_dotPositionA] = (_dotMoveDirectionA > 0) ? '>' : '<';
     } else {
-      _testNumB = (_testNumA + 1 + (rand() % 4)) % 5;
+      _testNumB = (_testNumA + 1 + (rand() % 3)) % 4;
       testNum = _testNumB;
 
       _movingDot[_dotPositionB] = ' ';
@@ -464,18 +453,15 @@ void APITest::RunTest(char thread) {
       ChangeCodec('A');
       break;
     case 1:
-      TestPlayout('B');
-      break;
-    case 2:
       if (!_randomTest) {
         fprintf(stdout, "\nTesting Delay ...\n");
       }
       TestDelay('A');
       break;
-    case 3:
+    case 2:
       TestSendVAD('A');
       break;
-    case 4:
+    case 3:
       TestRegisteration('A');
       break;
     default:
@@ -498,7 +484,6 @@ bool APITest::APIRunA() {
   } else {
     CurrentCodec('A');
     ChangeCodec('A');
-    TestPlayout('B');
     if (_codecCntrA == 0) {
       fprintf(stdout, "\nTesting Delay ...\n");
       TestDelay('A');
@@ -531,38 +516,34 @@ void APITest::Perform() {
   //--- THREADS
   // A
   // PUSH
-  rtc::scoped_ptr<ThreadWrapper> myPushAudioThreadA =
-      ThreadWrapper::CreateThread(PushAudioThreadA, this, "PushAudioThreadA");
-  CHECK_THREAD_NULLITY(myPushAudioThreadA, "Unable to start A::PUSH thread");
+  rtc::PlatformThread myPushAudioThreadA(PushAudioThreadA, this,
+                                         "PushAudioThreadA");
+  myPushAudioThreadA.Start();
   // PULL
-  rtc::scoped_ptr<ThreadWrapper> myPullAudioThreadA =
-      ThreadWrapper::CreateThread(PullAudioThreadA, this, "PullAudioThreadA");
-  CHECK_THREAD_NULLITY(myPullAudioThreadA, "Unable to start A::PULL thread");
+  rtc::PlatformThread myPullAudioThreadA(PullAudioThreadA, this,
+                                         "PullAudioThreadA");
+  myPullAudioThreadA.Start();
   // Process
-  rtc::scoped_ptr<ThreadWrapper> myProcessThreadA = ThreadWrapper::CreateThread(
-      ProcessThreadA, this, "ProcessThreadA");
-  CHECK_THREAD_NULLITY(myProcessThreadA, "Unable to start A::Process thread");
+  rtc::PlatformThread myProcessThreadA(ProcessThreadA, this, "ProcessThreadA");
+  myProcessThreadA.Start();
   // API
-  rtc::scoped_ptr<ThreadWrapper> myAPIThreadA = ThreadWrapper::CreateThread(
-      APIThreadA, this, "APIThreadA");
-  CHECK_THREAD_NULLITY(myAPIThreadA, "Unable to start A::API thread");
+  rtc::PlatformThread myAPIThreadA(APIThreadA, this, "APIThreadA");
+  myAPIThreadA.Start();
   // B
   // PUSH
-  rtc::scoped_ptr<ThreadWrapper> myPushAudioThreadB =
-      ThreadWrapper::CreateThread(PushAudioThreadB, this, "PushAudioThreadB");
-  CHECK_THREAD_NULLITY(myPushAudioThreadB, "Unable to start B::PUSH thread");
+  rtc::PlatformThread myPushAudioThreadB(PushAudioThreadB, this,
+                                         "PushAudioThreadB");
+  myPushAudioThreadB.Start();
   // PULL
-  rtc::scoped_ptr<ThreadWrapper> myPullAudioThreadB =
-      ThreadWrapper::CreateThread(PullAudioThreadB, this, "PullAudioThreadB");
-  CHECK_THREAD_NULLITY(myPullAudioThreadB, "Unable to start B::PULL thread");
+  rtc::PlatformThread myPullAudioThreadB(PullAudioThreadB, this,
+                                         "PullAudioThreadB");
+  myPullAudioThreadB.Start();
   // Process
-  rtc::scoped_ptr<ThreadWrapper> myProcessThreadB = ThreadWrapper::CreateThread(
-      ProcessThreadB, this, "ProcessThreadB");
-  CHECK_THREAD_NULLITY(myProcessThreadB, "Unable to start B::Process thread");
+  rtc::PlatformThread myProcessThreadB(ProcessThreadB, this, "ProcessThreadB");
+  myProcessThreadB.Start();
   // API
-  rtc::scoped_ptr<ThreadWrapper> myAPIThreadB = ThreadWrapper::CreateThread(
-      APIThreadB, this, "APIThreadB");
-  CHECK_THREAD_NULLITY(myAPIThreadB, "Unable to start B::API thread");
+  rtc::PlatformThread myAPIThreadB(APIThreadB, this, "APIThreadB");
+  myAPIThreadB.Start();
 
   //_apiEventA->StartTimer(true, 5000);
   //_apiEventB->StartTimer(true, 5000);
@@ -596,15 +577,15 @@ void APITest::Perform() {
   //(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000));
   delete completeEvent;
 
-  myPushAudioThreadA->Stop();
-  myPullAudioThreadA->Stop();
-  myProcessThreadA->Stop();
-  myAPIThreadA->Stop();
+  myPushAudioThreadA.Stop();
+  myPullAudioThreadA.Stop();
+  myProcessThreadA.Stop();
+  myAPIThreadA.Stop();
 
-  myPushAudioThreadB->Stop();
-  myPullAudioThreadB->Stop();
-  myProcessThreadB->Stop();
-  myAPIThreadB->Stop();
+  myPushAudioThreadB.Stop();
+  myPullAudioThreadB.Stop();
+  myProcessThreadB.Stop();
+  myAPIThreadB.Stop();
 }
 
 void APITest::CheckVADStatus(char side) {
@@ -682,7 +663,7 @@ void APITest::TestDelay(char side) {
   AudioCodingModule* myACM;
   Channel* myChannel;
   int32_t* myMinDelay;
-  EventWrapper* myEvent = EventWrapper::Create();
+  EventTimerWrapper* myEvent = EventTimerWrapper::Create();
 
   uint32_t inTimestamp = 0;
   uint32_t outTimestamp = 0;
@@ -832,9 +813,11 @@ void APITest::TestRegisteration(char sendSide) {
       exit(-1);
   }
 
-  CodecInst myCodec;
-  if (sendACM->SendCodec(&myCodec) < 0) {
-    AudioCodingModule::Codec(_codecCntrA, &myCodec);
+  auto myCodec = sendACM->SendCodec();
+  if (!myCodec) {
+    CodecInst ci;
+    AudioCodingModule::Codec(_codecCntrA, &ci);
+    myCodec = rtc::Optional<CodecInst>(ci);
   }
 
   if (!_randomTest) {
@@ -846,12 +829,12 @@ void APITest::TestRegisteration(char sendSide) {
     *thereIsDecoder = false;
   }
   //myEvent->Wait(20);
-  CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+  CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
   Wait(1000);
 
-  int currentPayload = myCodec.pltype;
+  int currentPayload = myCodec->pltype;
 
-  if (!FixedPayloadTypeCodec(myCodec.plname)) {
+  if (!FixedPayloadTypeCodec(myCodec->plname)) {
     int32_t i;
     for (i = 0; i < 32; i++) {
       if (!_payloadUsed[i]) {
@@ -859,9 +842,9 @@ void APITest::TestRegisteration(char sendSide) {
           fprintf(stdout,
                   "Register receive codec with new Payload, AUDIO BACK.\n");
         }
-        //myCodec.pltype = i + 96;
-        //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
-        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
+        //myCodec->pltype = i + 96;
+        //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
         //myEvent->Wait(20);
         //{
         //    WriteLockScoped wl(_apiTestRWLock);
@@ -877,17 +860,17 @@ void APITest::TestRegisteration(char sendSide) {
         //    *thereIsDecoder = false;
         //}
         //myEvent->Wait(20);
-        //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
+        //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
         Wait(1000);
 
-        myCodec.pltype = currentPayload;
+        myCodec->pltype = currentPayload;
         if (!_randomTest) {
           fprintf(stdout,
                   "Register receive codec with default Payload, AUDIO BACK.\n");
           fflush (stdout);
         }
-        CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
-        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
+        CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
         myEvent->Wait(20);
         {
           WriteLockScoped wl(_apiTestRWLock);
@@ -899,7 +882,7 @@ void APITest::TestRegisteration(char sendSide) {
       }
     }
     if (i == 32) {
-      CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+      CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
       {
         WriteLockScoped wl(_apiTestRWLock);
         *thereIsDecoder = true;
@@ -911,9 +894,9 @@ void APITest::TestRegisteration(char sendSide) {
               "Register receive codec with fixed Payload, AUDIO BACK.\n");
       fflush (stdout);
     }
-    CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
-    //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
-    //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
+    CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+    //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
+    //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
     myEvent->Wait(20);
     {
       WriteLockScoped wl(_apiTestRWLock);
@@ -927,67 +910,6 @@ void APITest::TestRegisteration(char sendSide) {
   }
 }
 
-// Playout Mode, background noise mode.
-// Receiver Frequency, playout frequency.
-void APITest::TestPlayout(char receiveSide) {
-  AudioCodingModule* receiveACM;
-  AudioPlayoutMode* playoutMode = NULL;
-  switch (receiveSide) {
-    case 'A': {
-      receiveACM = _acmA.get();
-      playoutMode = &_playoutModeA;
-      break;
-    }
-    case 'B': {
-      receiveACM = _acmB.get();
-      playoutMode = &_playoutModeB;
-      break;
-    }
-    default:
-      receiveACM = _acmA.get();
-  }
-
-  int32_t receiveFreqHz = receiveACM->ReceiveFrequency();
-  int32_t playoutFreqHz = receiveACM->PlayoutFrequency();
-
-  CHECK_ERROR_MT(receiveFreqHz);
-  CHECK_ERROR_MT(playoutFreqHz);
-
-
-  char playoutString[25];
-  switch (*playoutMode) {
-    case voice: {
-      *playoutMode = fax;
-      strncpy(playoutString, "FAX", 25);
-      break;
-    }
-    case fax: {
-      *playoutMode = streaming;
-      strncpy(playoutString, "Streaming", 25);
-      break;
-    }
-    case streaming: {
-      *playoutMode = voice;
-      strncpy(playoutString, "Voice", 25);
-      break;
-    }
-    default:
-      *playoutMode = voice;
-      strncpy(playoutString, "Voice", 25);
-  }
-  CHECK_ERROR_MT(receiveACM->SetPlayoutMode(*playoutMode));
-  playoutString[24] = '\0';
-
-  if (!_randomTest) {
-    fprintf(stdout, "\n");
-    fprintf(stdout, "In Side %c\n", receiveSide);
-    fprintf(stdout, "---------------------------------\n");
-    fprintf(stdout, "Receive Frequency....... %d Hz\n", receiveFreqHz);
-    fprintf(stdout, "Playout Frequency....... %d Hz\n", playoutFreqHz);
-    fprintf(stdout, "Audio Playout Mode...... %s\n", playoutString);
-  }
-}
-
 void APITest::TestSendVAD(char side) {
   if (_randomTest) {
     return;
@@ -1071,22 +993,17 @@ void APITest::TestSendVAD(char side) {
 }
 
 void APITest::CurrentCodec(char side) {
-  CodecInst myCodec;
-  if (side == 'A') {
-    _acmA->SendCodec(&myCodec);
-  } else {
-    _acmB->SendCodec(&myCodec);
-  }
+  auto myCodec = (side == 'A' ? _acmA : _acmB)->SendCodec();
 
   if (!_randomTest) {
     fprintf(stdout, "\n\n");
     fprintf(stdout, "Send codec in Side A\n");
     fprintf(stdout, "----------------------------\n");
-    fprintf(stdout, "Name................. %s\n", myCodec.plname);
-    fprintf(stdout, "Sampling Frequency... %d\n", myCodec.plfreq);
-    fprintf(stdout, "Rate................. %d\n", myCodec.rate);
-    fprintf(stdout, "Payload-type......... %d\n", myCodec.pltype);
-    fprintf(stdout, "Packet-size.......... %d\n", myCodec.pacsize);
+    fprintf(stdout, "Name................. %s\n", myCodec->plname);
+    fprintf(stdout, "Sampling Frequency... %d\n", myCodec->plfreq);
+    fprintf(stdout, "Rate................. %d\n", myCodec->rate);
+    fprintf(stdout, "Payload-type......... %d\n", myCodec->pltype);
+    fprintf(stdout, "Packet-size.......... %d\n", myCodec->pacsize);
   }
 
   Wait(100);
@@ -1129,7 +1046,6 @@ void APITest::ChangeCodec(char side) {
     myChannel = _channel_B2A;
   }
 
-  myACM->ResetEncoder();
   Wait(100);
 
   // Register the next codec
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.h
index 7ad51a6c17..a1937c2b00 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/APITest.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/APITest.h
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 namespace webrtc {
 
@@ -109,14 +109,14 @@ class APITest : public ACMTest {
   bool _writeToFile;
   //--- Events
   // A
-  EventWrapper* _pullEventA;      // pulling data from ACM
-  EventWrapper* _pushEventA;      // pushing data to ACM
-  EventWrapper* _processEventA;   // process
+  EventTimerWrapper* _pullEventA;      // pulling data from ACM
+  EventTimerWrapper* _pushEventA;      // pushing data to ACM
+  EventTimerWrapper* _processEventA;   // process
   EventWrapper* _apiEventA;       // API calls
   // B
-  EventWrapper* _pullEventB;      // pulling data from ACM
-  EventWrapper* _pushEventB;      // pushing data to ACM
-  EventWrapper* _processEventB;   // process
+  EventTimerWrapper* _pullEventB;      // pulling data from ACM
+  EventTimerWrapper* _pushEventB;      // pushing data to ACM
+  EventTimerWrapper* _processEventB;   // process
   EventWrapper* _apiEventB;       // API calls
 
   // keep track of the codec in either side.
@@ -141,9 +141,6 @@ class APITest : public ACMTest {
   int32_t _minDelayB;
   bool _payloadUsed[32];
 
-  AudioPlayoutMode _playoutModeA;
-  AudioPlayoutMode _playoutModeB;
-
   bool _verbose;
 
   int _dotPositionA;
@@ -153,7 +150,6 @@ class APITest : public ACMTest {
 
   char _movingDot[41];
 
-  DTMFDetector* _dtmfCallback;
   VADCallback* _vadCallbackA;
   VADCallback* _vadCallbackB;
   RWLockWrapper& _apiTestRWLock;
@@ -164,4 +160,4 @@ class APITest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.cc
similarity index 98%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.cc
index 779718dd50..31521fe1e3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
 
 #include <assert.h>
 #include <iostream>
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -42,7 +42,7 @@ int32_t Channel::SendData(FrameType frameType,
   } else {
     rtpInfo.type.Audio.isCNG = false;
   }
-  if (frameType == kFrameEmpty) {
+  if (frameType == kEmptyFrame) {
     // When frame is empty, we should not transmit it. The frame size of the
     // next non-empty frame will be based on the previous frame size.
     _useLastFrameSize = _lastFrameSizeSample > 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.h
index c4ad7d120f..b047aa9909 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/Channel.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
 
 #include <stdio.h>
 
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -127,4 +127,4 @@ class Channel : public AudioPacketizationCallback {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
similarity index 93%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index 8394bc03ab..ba3c8d9ad2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 #include <sstream>
 #include <stdio.h>
@@ -17,10 +17,10 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -52,7 +52,7 @@ Sender::Sender()
 }
 
 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                   std::string in_file_name, int sample_rate, int channels) {
+                   std::string in_file_name, int sample_rate, size_t channels) {
   struct CodecInst sendCodec;
   int noOfCodecs = acm->NumberOfCodecs();
   int codecNo;
@@ -63,6 +63,10 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
   if (channels == 2) {
     _pcmFile.ReadStereo(true);
   }
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _pcmFile.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  _pcmFile.FastForward(100);
 
   // Set the codec for the current test.
   if ((testMode == 0) || (testMode == 1)) {
@@ -119,7 +123,7 @@ Receiver::Receiver()
 }
 
 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-                     std::string out_file_name, int channels) {
+                     std::string out_file_name, size_t channels) {
   struct CodecInst recvCodec = CodecInst();
   int noOfCodecs;
   EXPECT_EQ(0, acm->InitializeReceiver());
@@ -339,8 +343,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
   _sender.codeId = codeId;
 
   _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
-  struct CodecInst sendCodecInst;
-  if (acm->SendCodec(&sendCodecInst) >= 0) {
+  if (acm->SendCodec()) {
     _sender.Run();
   }
   _sender.Teardown();
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
similarity index 82%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
index 44fb0b2d76..f9a9a5bb52 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
 
 #include <stdio.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/RTPFile.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/RTPFile.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -48,7 +48,7 @@ class Sender {
  public:
   Sender();
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string in_file_name, int sample_rate, int channels);
+             std::string in_file_name, int sample_rate, size_t channels);
   void Teardown();
   void Run();
   bool Add10MsData();
@@ -71,7 +71,7 @@ class Receiver {
   Receiver();
   virtual ~Receiver() {};
   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
-             std::string out_file_name, int channels);
+             std::string out_file_name, size_t channels);
   void Teardown();
   void Run();
   virtual bool IncomingPacket();
@@ -120,4 +120,4 @@ class EncodeDecodeTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.cc
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.cc
index 4f46098a7b..9289d73baa 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "PCMFile.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 
 #include <ctype.h>
 #include <stdio.h>
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -137,6 +137,9 @@ int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) {
   audio_frame.num_channels_ = channels;
   audio_frame.timestamp_ = timestamp_;
   timestamp_ += samples_10ms_;
+  ++blocks_read_;
+  if (num_10ms_blocks_to_read_ && blocks_read_ >= *num_10ms_blocks_to_read_)
+    end_of_file_ = true;
   return samples_10ms_;
 }
 
@@ -150,8 +153,7 @@ void PCMFile::Write10MsData(AudioFrame& audio_frame) {
       }
     } else {
       int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
-      int k;
-      for (k = 0; k < audio_frame.samples_per_channel_; k++) {
+      for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) {
         stereo_audio[k << 1] = audio_frame.data_[k];
         stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
       }
@@ -173,7 +175,7 @@ void PCMFile::Write10MsData(AudioFrame& audio_frame) {
   }
 }
 
-void PCMFile::Write10MsData(int16_t* playout_buffer, uint16_t length_smpls) {
+void PCMFile::Write10MsData(int16_t* playout_buffer, size_t length_smpls) {
   if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
       length_smpls) {
     return;
@@ -183,11 +185,21 @@ void PCMFile::Write10MsData(int16_t* playout_buffer, uint16_t length_smpls) {
 void PCMFile::Close() {
   fclose(pcm_file_);
   pcm_file_ = NULL;
+  blocks_read_ = 0;
+}
+
+void PCMFile::FastForward(int num_10ms_blocks) {
+  const int channels = read_stereo_ ? 2 : 1;
+  long num_bytes_to_move =
+      num_10ms_blocks * sizeof(int16_t) * samples_10ms_ * channels;
+  int error = fseek(pcm_file_, num_bytes_to_move, SEEK_CUR);
+  RTC_DCHECK_EQ(error, 0);
 }
 
 void PCMFile::Rewind() {
   rewind(pcm_file_);
   end_of_file_ = false;
+  blocks_read_ = 0;
 }
 
 bool PCMFile::Rewinded() {
@@ -202,4 +214,8 @@ void PCMFile::ReadStereo(bool is_stereo) {
   read_stereo_ = is_stereo;
 }
 
+void PCMFile::SetNum10MsBlocksToRead(int value) {
+  num_10ms_blocks_to_read_ = rtc::Optional<int>(value);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.h
similarity index 63%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.h
index c4487b8133..840933a1bd 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PCMFile.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PCMFile.h
@@ -8,15 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
 
 #include <stdio.h>
 #include <stdlib.h>
 
 #include <string>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -36,7 +37,7 @@ class PCMFile {
 
   int32_t Read10MsData(AudioFrame& audio_frame);
 
-  void Write10MsData(int16_t *playout_buffer, uint16_t length_smpls);
+  void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
   void Write10MsData(AudioFrame& audio_frame);
 
   uint16_t PayloadLength10Ms() const;
@@ -45,12 +46,21 @@ class PCMFile {
   bool EndOfFile() const {
     return end_of_file_;
   }
+  // Moves forward the specified number of 10 ms blocks. If a limit has been set
+  // with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
+  // limit.
+  void FastForward(int num_10ms_blocks);
   void Rewind();
   static int16_t ChooseFile(std::string* file_name, int16_t max_len,
                             uint16_t* frequency_hz);
   bool Rewinded();
   void SaveStereo(bool is_stereo = true);
   void ReadStereo(bool is_stereo = true);
+  // If set, the reading will stop after the specified number of blocks have
+  // been read. When that has happened, EndOfFile() will return true. Calling
+  // Rewind() will reset the counter and start over.
+  void SetNum10MsBlocksToRead(int value);
+
  private:
   FILE* pcm_file_;
   uint16_t samples_10ms_;
@@ -61,8 +71,10 @@ class PCMFile {
   uint32_t timestamp_;
   bool read_stereo_;
   bool save_stereo_;
+  rtc::Optional<int> num_10ms_blocks_to_read_;
+  int blocks_read_ = 0;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.cc
similarity index 97%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.cc
index f19d491d2d..ad3e83403e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common.h"
@@ -143,8 +143,7 @@ void PacketLossTest::Perform() {
 
   sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
                  expected_loss_rate_);
-  struct CodecInst sendCodecInst;
-  if (acm->SendCodec(&sendCodecInst) >= 0) {
+  if (acm->SendCodec()) {
     sender_->Run();
   }
   sender_->Teardown();
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.h
index d25dea264f..f3570ae1ca 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/PacketLossTest.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/PacketLossTest.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 
 #include <string>
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 namespace webrtc {
 
@@ -64,4 +64,4 @@ class PacketLossTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.cc
similarity index 99%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.cc
index 4e81943de4..60777178c6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.cc
@@ -21,7 +21,7 @@
 
 #include "audio_coding_module.h"
 #include "engine_configurations.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 // TODO(tlegrand): Consider removing usage of gtest.
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.h
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.h
index 346440b1b3..696d41ebd2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/RTPFile.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
 
 #include <stdio.h>
 #include <queue>
 
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -123,4 +123,4 @@ class RTPFile : public RTPStream {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.cc
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.cc
index b28c510a56..c9f8080826 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.cc
@@ -14,9 +14,9 @@
 #include <math.h>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/SpatialAudio.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/test/SpatialAudio.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -159,13 +159,13 @@ void SpatialAudio::EncodeDecode(const double leftPanning,
 
   while (!_inFile.EndOfFile()) {
     _inFile.Read10MsData(audioFrame);
-    for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
+    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
       audioFrame.data_[n] = (int16_t) floor(
           audioFrame.data_[n] * leftPanning + 0.5);
     }
     CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
 
-    for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
+    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
       audioFrame.data_[n] = (int16_t) floor(
           audioFrame.data_[n] * rightToLeftRatio + 0.5);
     }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.h
similarity index 66%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.h
index f5e127f8a4..3548cc98eb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/SpatialAudio.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/SpatialAudio.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 
@@ -44,4 +44,4 @@ class SpatialAudio : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.cc
similarity index 96%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.cc
index b1badb6363..bacfd37188 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
 
 #include <cstdio>
 #include <limits>
@@ -18,10 +18,10 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -74,7 +74,7 @@ int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
   } else {
     rtp_info.type.Audio.isCNG = false;
   }
-  if (frame_type == kFrameEmpty) {
+  if (frame_type == kEmptyFrame) {
     // Skip this frame.
     return 0;
   }
@@ -223,7 +223,6 @@ void TestAllCodecs::Perform() {
   Run(channel_a_to_b_);
   outfile_b_.Close();
 #endif
-#ifdef WEBRTC_CODEC_PCM16
   if (test_mode_ != 0) {
     printf("===============================================================\n");
   }
@@ -263,7 +262,6 @@ void TestAllCodecs::Perform() {
   RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
   Run(channel_a_to_b_);
   outfile_b_.Close();
-#endif
   if (test_mode_ != 0) {
     printf("===============================================================\n");
   }
@@ -339,9 +337,6 @@ void TestAllCodecs::Perform() {
 #ifndef WEBRTC_CODEC_ISACFX
     printf("   ISAC fix\n");
 #endif
-#ifndef WEBRTC_CODEC_PCM16
-    printf("   PCM16\n");
-#endif
 
     printf("\nTo complete the test, listen to the %d number of output files.\n",
            test_count_);
@@ -427,8 +422,12 @@ void TestAllCodecs::Run(TestPack* channel) {
   uint32_t timestamp_diff;
   channel->reset_payload_size();
   int error_count = 0;
-
   int counter = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  infile_a_.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  infile_a_.FastForward(100);
+
   while (!infile_a_.EndOfFile()) {
     // Add 10 msec to ACM.
     infile_a_.Read10MsData(audio_frame);
@@ -482,8 +481,7 @@ void TestAllCodecs::OpenOutFile(int test_number) {
 
 void TestAllCodecs::DisplaySendReceiveCodec() {
   CodecInst my_codec_param;
-  acm_a_->SendCodec(&my_codec_param);
-  printf("%s -> ", my_codec_param.plname);
+  printf("%s -> ", acm_a_->SendCodec()->plname);
   acm_b_->ReceiveCodec(&my_codec_param);
   printf("%s\n", my_codec_param.plname);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.h
similarity index 85%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.h
index 1cdc0cba98..e79bd69faa 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestAllCodecs.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -81,4 +81,4 @@ class TestAllCodecs : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.cc
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.cc
index 6027a4d0e1..a1bdc04e53 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.cc
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
 
 #include <assert.h>
 
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 #ifdef SUPPORT_RED_WB
@@ -37,11 +37,15 @@ namespace webrtc {
 namespace {
   const char kNameL16[] = "L16";
   const char kNamePCMU[] = "PCMU";
+  const char kNameCN[] = "CN";
+  const char kNameRED[] = "RED";
+
+  // These three are only used by code #ifdeffed on WEBRTC_CODEC_G722.
+#ifdef WEBRTC_CODEC_G722
   const char kNameISAC[] = "ISAC";
   const char kNameG722[] = "G722";
   const char kNameOPUS[] = "opus";
-  const char kNameCN[] = "CN";
-  const char kNameRED[] = "RED";
+#endif
 }
 
 TestRedFec::TestRedFec()
@@ -82,10 +86,6 @@ void TestRedFec::Perform() {
   _acmA->RegisterTransportCallback(_channelA2B);
   _channelA2B->RegisterReceiverACM(_acmB.get());
 
-#ifndef WEBRTC_CODEC_PCM16
-  EXPECT_TRUE(false) << "PCM16 needs to be activated to run this test\n");
-  return;
-#endif
   EXPECT_EQ(0, RegisterSendCodec('A', kNameL16, 8000));
   EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 8000));
   EXPECT_EQ(0, RegisterSendCodec('A', kNameRED));
@@ -108,7 +108,7 @@ void TestRedFec::Perform() {
   EXPECT_TRUE(false);
   printf("G722 needs to be activated to run this test\n");
   return;
-#endif
+#else
   EXPECT_EQ(0, RegisterSendCodec('A', kNameG722, 16000));
   EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
 
@@ -412,6 +412,8 @@ void TestRedFec::Perform() {
   EXPECT_FALSE(_acmA->REDStatus());
   EXPECT_EQ(0, _acmA->SetCodecFEC(false));
   EXPECT_FALSE(_acmA->CodecFEC());
+
+#endif  // defined(WEBRTC_CODEC_G722)
 }
 
 int32_t TestRedFec::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {
@@ -451,6 +453,10 @@ int16_t TestRedFec::RegisterSendCodec(char side, const char* codecName,
 void TestRedFec::Run() {
   AudioFrame audioFrame;
   int32_t outFreqHzB = _outFileB.SamplingFrequency();
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _inFileA.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  _inFileA.FastForward(100);
 
   while (!_inFileA.EndOfFile()) {
     EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.h
similarity index 78%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.h
index ac0b6cdfc7..6343d8e374 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestRedFec.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestRedFec.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
 
 #include <string>
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 
 namespace webrtc {
 
@@ -48,4 +48,4 @@ class TestRedFec : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.cc
similarity index 96%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.cc
index 72ae25ea18..9bf560d323 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
 
 #include <assert.h>
 
@@ -17,9 +17,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -58,7 +58,7 @@ int32_t TestPackStereo::SendData(const FrameType frame_type,
   rtp_info.header.sequenceNumber = seq_no_++;
   rtp_info.header.payloadType = payload_type;
   rtp_info.header.timestamp = timestamp;
-  if (frame_type == kFrameEmpty) {
+  if (frame_type == kEmptyFrame) {
     // Skip this frame
     return 0;
   }
@@ -118,11 +118,9 @@ TestStereo::TestStereo(int test_mode)
 #ifdef WEBRTC_CODEC_G722
       , g722_pltype_(0)
 #endif
-#ifdef WEBRTC_CODEC_PCM16
       , l16_8khz_pltype_(-1)
       , l16_16khz_pltype_(-1)
       , l16_32khz_pltype_(-1)
-#endif
 #ifdef PCMA_AND_PCMU
       , pcma_pltype_(-1)
       , pcmu_pltype_(-1)
@@ -247,7 +245,6 @@ void TestStereo::Perform() {
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
 #endif
-#ifdef WEBRTC_CODEC_PCM16
   if (test_mode_ != 0) {
     printf("===========================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -306,7 +303,6 @@ void TestStereo::Perform() {
       l16_32khz_pltype_);
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
-#endif
 #ifdef PCMA_AND_PCMU
   if (test_mode_ != 0) {
     printf("===========================================================\n");
@@ -343,14 +339,6 @@ void TestStereo::Perform() {
   EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
   EXPECT_FALSE(dtx);
   EXPECT_FALSE(vad);
-  EXPECT_EQ(-1, acm_a_->SetVAD(true, false, VADNormal));
-  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
-  EXPECT_FALSE(dtx);
-  EXPECT_FALSE(vad);
-  EXPECT_EQ(-1, acm_a_->SetVAD(false, true, VADNormal));
-  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
-  EXPECT_FALSE(dtx);
-  EXPECT_FALSE(vad);
   EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
   EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
   EXPECT_FALSE(dtx);
@@ -443,7 +431,6 @@ void TestStereo::Perform() {
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
 #endif
-#ifdef WEBRTC_CODEC_PCM16
   if (test_mode_ != 0) {
     printf("===============================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -478,7 +465,6 @@ void TestStereo::Perform() {
       l16_32khz_pltype_);
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
-#endif
 #ifdef PCMA_AND_PCMU
   if (test_mode_ != 0) {
     printf("===============================================================\n");
@@ -546,7 +532,6 @@ void TestStereo::Perform() {
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
 #endif
-#ifdef WEBRTC_CODEC_PCM16
   if (test_mode_ != 0) {
     printf("===============================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -580,7 +565,6 @@ void TestStereo::Perform() {
       l16_32khz_pltype_);
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
-#endif
 #ifdef PCMA_AND_PCMU
   if (test_mode_ != 0) {
     printf("===============================================================\n");
@@ -670,9 +654,7 @@ void TestStereo::Perform() {
 #ifdef WEBRTC_CODEC_G722
     printf("   G.722\n");
 #endif
-#ifdef WEBRTC_CODEC_PCM16
     printf("   PCM16\n");
-#endif
     printf("   G.711\n");
 #ifdef WEBRTC_CODEC_OPUS
     printf("   Opus\n");
@@ -753,6 +735,12 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
   int error_count = 0;
   int variable_bytes = 0;
   int variable_packets = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  in_file_mono_->SetNum10MsBlocksToRead(50);
+  in_file_stereo_->SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_->FastForward(100);
+  in_file_mono_->FastForward(100);
 
   while (1) {
     // Simulate packet loss by setting |packet_loss_| to "true" in
@@ -818,7 +806,7 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
   // such as Opus.
   if (variable_packets > 0) {
     variable_bytes /= variable_packets;
-    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
+    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
   }
 
   if (in_file_mono_->EndOfFile()) {
@@ -841,14 +829,15 @@ void TestStereo::OpenOutFile(int16_t test_number) {
 }
 
 void TestStereo::DisplaySendReceiveCodec() {
-  CodecInst my_codec_param;
-  acm_a_->SendCodec(&my_codec_param);
+  auto send_codec = acm_a_->SendCodec();
   if (test_mode_ != 0) {
-    printf("%s -> ", my_codec_param.plname);
+    ASSERT_TRUE(send_codec);
+    printf("%s -> ", send_codec->plname);
   }
-  acm_b_->ReceiveCodec(&my_codec_param);
+  CodecInst receive_codec;
+  acm_b_->ReceiveCodec(&receive_codec);
   if (test_mode_ != 0) {
-    printf("%s\n", my_codec_param.plname);
+    printf("%s\n", receive_codec.plname);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.h
index c6412c7946..4526be6960 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestStereo.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
 
 #include <math.h>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
 
 #define PCMA_AND_PCMU
 
@@ -100,11 +100,9 @@ class TestStereo : public ACMTest {
 #ifdef WEBRTC_CODEC_G722
   int g722_pltype_;
 #endif
-#ifdef WEBRTC_CODEC_PCM16
   int l16_8khz_pltype_;
   int l16_16khz_pltype_;
   int l16_32khz_pltype_;
-#endif
 #ifdef PCMA_AND_PCMU
   int pcma_pltype_;
   int pcmu_pltype_;
@@ -116,4 +114,4 @@ class TestStereo : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.cc
similarity index 85%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.cc
index e544ae31c7..229dc2d474 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
 
 #include <string>
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -44,7 +44,7 @@ int32_t ActivityMonitor::InFrameType(FrameType frame_type) {
 
 void ActivityMonitor::PrintStatistics() {
   printf("\n");
-  printf("kFrameEmpty       %u\n", counter_[kFrameEmpty]);
+  printf("kEmptyFrame       %u\n", counter_[kEmptyFrame]);
   printf("kAudioFrameSpeech %u\n", counter_[kAudioFrameSpeech]);
   printf("kAudioFrameCN     %u\n", counter_[kAudioFrameCN]);
   printf("kVideoFrameKey    %u\n", counter_[kVideoFrameKey]);
@@ -87,6 +87,11 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
   PCMFile in_file;
   in_file.Open(in_filename, frequency, "rb");
   in_file.ReadStereo(channels > 1);
+  // Set test length to 1000 ms (100 blocks of 10 ms each).
+  in_file.SetNum10MsBlocksToRead(100);
+  // Fast-forward both files 500 ms (50 blocks). The first second of the file is
+  // silence, but we want to keep half of that to test silence periods.
+  in_file.FastForward(50);
 
   PCMFile out_file;
   if (append) {
@@ -137,7 +142,6 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
 TestWebRtcVadDtx::TestWebRtcVadDtx()
     : vad_enabled_(false),
       dtx_enabled_(false),
-      use_webrtc_dtx_(false),
       output_file_num_(0) {
 }
 
@@ -191,7 +195,7 @@ void TestWebRtcVadDtx::RunTestCases() {
 
 // Set the expectation and run the test.
 void TestWebRtcVadDtx::Test(bool new_outfile) {
-  int expects[] = {-1, 1, use_webrtc_dtx_, 0, 0};
+  int expects[] = {-1, 1, dtx_enabled_, 0, 0};
   if (new_outfile) {
     output_file_num_++;
   }
@@ -210,26 +214,19 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
   EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
   EXPECT_EQ(0, acm_send_->VAD(&dtx_enabled_, &vad_enabled_, &mode));
 
-  CodecInst codec_param;
-  acm_send_->SendCodec(&codec_param);
-  if (STR_CASE_CMP(codec_param.plname, "opus") == 0) {
+  auto codec_param = acm_send_->SendCodec();
+  ASSERT_TRUE(codec_param);
+  if (STR_CASE_CMP(codec_param->plname, "opus") == 0) {
     // If send codec is Opus, WebRTC VAD/DTX cannot be used.
     enable_dtx = enable_vad = false;
   }
 
   EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
 
-  bool replaced = false;
-  acm_send_->IsInternalDTXReplacedWithWebRtc(&replaced);
-
-  use_webrtc_dtx_ = dtx_enabled_ && replaced;
-
-  if (use_webrtc_dtx_) {
+  if (dtx_enabled_) {
     EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
-  }
-
-  if (!dtx_enabled_ || !use_webrtc_dtx_) {
-    // Using no DTX or codec Internal DTX should not affect setting of VAD.
+  } else {
+    // Using no DTX should not affect setting of VAD.
     EXPECT_EQ(enable_vad, vad_enabled_);
   }
 }
@@ -237,10 +234,10 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
 // Following is the implementation of TestOpusDtx.
 void TestOpusDtx::Perform() {
 #ifdef WEBRTC_CODEC_ISAC
-  // If we set other codec than Opus, DTX cannot be toggled.
+  // If we set other codec than Opus, DTX cannot be switched on.
   RegisterCodec(kIsacWb);
-  EXPECT_EQ(-1, acm_send_->EnableOpusDtx(false));
-  EXPECT_EQ(-1, acm_send_->DisableOpusDtx());
+  EXPECT_EQ(-1, acm_send_->EnableOpusDtx());
+  EXPECT_EQ(0, acm_send_->DisableOpusDtx());
 #endif
 
 #ifdef WEBRTC_CODEC_OPUS
@@ -255,8 +252,8 @@ void TestOpusDtx::Perform() {
   Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
       32000, 1, out_filename, false, expects);
 
-  EXPECT_EQ(0, acm_send_->EnableOpusDtx(false));
-  expects[kFrameEmpty] = 1;
+  EXPECT_EQ(0, acm_send_->EnableOpusDtx());
+  expects[kEmptyFrame] = 1;
   Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
       32000, 1, out_filename, true, expects);
 
@@ -264,16 +261,13 @@ void TestOpusDtx::Perform() {
   out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
   RegisterCodec(kOpusStereo);
   EXPECT_EQ(0, acm_send_->DisableOpusDtx());
-  expects[kFrameEmpty] = 0;
+  expects[kEmptyFrame] = 0;
   Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
       32000, 2, out_filename, false, expects);
 
-  // Opus should be now in kAudio mode. Opus DTX should not be set without
-  // forcing kVoip mode.
-  EXPECT_EQ(-1, acm_send_->EnableOpusDtx(false));
-  EXPECT_EQ(0, acm_send_->EnableOpusDtx(true));
+  EXPECT_EQ(0, acm_send_->EnableOpusDtx());
 
-  expects[kFrameEmpty] = 1;
+  expects[kEmptyFrame] = 1;
   Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
       32000, 2, out_filename, true, expects);
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.h
similarity index 83%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.h
index b664a9b4d4..1e7f0ef4d7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TestVADDTX.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TestVADDTX.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
 
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
 
 namespace webrtc {
 
@@ -29,7 +29,7 @@ class ActivityMonitor : public ACMVADCallback {
   void ResetStatistics();
   void GetStatistics(uint32_t* stats);
  private:
-  // 0 - kFrameEmpty
+  // 0 - kEmptyFrame
   // 1 - kAudioFrameSpeech
   // 2 - kAudioFrameCN
   // 3 - kVideoFrameKey (not used by audio)
@@ -60,7 +60,7 @@ class TestVadDtx : public ACMTest {
   // 0  : there have been no packets of type |x|,
   // 1  : there have been packets of type |x|,
   // with |x| indicates the following packet types
-  // 0 - kFrameEmpty
+  // 0 - kEmptyFrame
   // 1 - kAudioFrameSpeech
   // 2 - kAudioFrameCN
   // 3 - kVideoFrameKey (not used by audio)
@@ -88,7 +88,6 @@ class TestWebRtcVadDtx final : public TestVadDtx {
 
   bool vad_enabled_;
   bool dtx_enabled_;
-  bool use_webrtc_dtx_;
   int output_file_num_;
 };
 
@@ -100,4 +99,4 @@ class TestOpusDtx final : public TestVadDtx {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Tester.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/Tester.cc
similarity index 68%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Tester.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/Tester.cc
index 22510f3416..a27f0bc58b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/Tester.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/Tester.cc
@@ -13,20 +13,19 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/APITest.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
-#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
-#include "webrtc/modules/audio_coding/main/test/opus_test.h"
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
-#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
-#include "webrtc/modules/audio_coding/main/test/TestRedFec.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
-#include "webrtc/modules/audio_coding/main/test/TwoWayCommunication.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using webrtc::Trace;
 
@@ -42,7 +41,11 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
+#else
+TEST(AudioCodingModuleTest, TestEncodeDecode) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_encodedecode_trace.txt").c_str());
@@ -50,31 +53,54 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestRedFec)) {
+#if defined(WEBRTC_CODEC_RED)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestRedFec) {
+#else
+TEST(AudioCodingModuleTest, TestRedFec) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_fec_trace.txt").c_str());
   webrtc::TestRedFec().Perform();
   Trace::ReturnTrace();
 }
+#endif
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
+#else
+TEST(AudioCodingModuleTest, TestIsac) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_isac_trace.txt").c_str());
   webrtc::ISACTest(ACM_TEST_MODE).Perform();
   Trace::ReturnTrace();
 }
+#endif
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunication)) {
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
+#else
+TEST(AudioCodingModuleTest, TwoWayCommunication) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_twowaycom_trace.txt").c_str());
   webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
   Trace::ReturnTrace();
 }
+#endif
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
+#else
+TEST(AudioCodingModuleTest, TestStereo) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_stereo_trace.txt").c_str());
@@ -82,7 +108,11 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
   Trace::ReturnTrace();
 }
 
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestWebRtcVadDtx)) {
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestWebRtcVadDtx) {
+#else
+TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
+#endif
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
       "acm_vaddtx_trace.txt").c_str());
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.cc
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.cc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.h
similarity index 82%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.h
index ef9609a267..0793eb0c0c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TimedTrace.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TimedTrace.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TIMED_TRACE_H
-#define TIMED_TRACE_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -33,4 +33,4 @@ class TimedTrace {
 
 };
 
-#endif
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
similarity index 82%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
index 1014fc9d0a..56e136bd34 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
@@ -21,9 +21,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -32,10 +32,16 @@ namespace webrtc {
 
 TwoWayCommunication::TwoWayCommunication(int testMode)
     : _acmA(AudioCodingModule::Create(1)),
-      _acmB(AudioCodingModule::Create(2)),
       _acmRefA(AudioCodingModule::Create(3)),
-      _acmRefB(AudioCodingModule::Create(4)),
-      _testMode(testMode) {}
+      _testMode(testMode) {
+  AudioCodingModule::Config config;
+  // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
+  config.neteq_config.playout_mode = kPlayoutFax;
+  config.id = 2;
+  _acmB.reset(AudioCodingModule::Create(config));
+  config.id = 4;
+  _acmRefB.reset(AudioCodingModule::Create(config));
+}
 
 TwoWayCommunication::~TwoWayCommunication() {
   delete _channel_A2B;
@@ -96,11 +102,6 @@ void TwoWayCommunication::SetUp() {
   //--- Set A codecs
   EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
   EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
-#ifdef WEBRTC_DTMF_DETECTION
-  _dtmfDetectorA = new(DTMFDetector);
-  EXPECT_GT(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA, ACMUSA),
-            -1);
-#endif
   //--- Set ref-A codecs
   EXPECT_EQ(0, _acmRefA->RegisterSendCodec(codecInst_A));
   EXPECT_EQ(0, _acmRefA->RegisterReceiveCodec(codecInst_B));
@@ -108,11 +109,6 @@ void TwoWayCommunication::SetUp() {
   //--- Set B codecs
   EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
   EXPECT_EQ(0, _acmB->RegisterReceiveCodec(codecInst_A));
-#ifdef WEBRTC_DTMF_DETECTION
-  _dtmfDetectorB = new(DTMFDetector);
-  EXPECT_GT(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB, ACMUSA),
-            -1);
-#endif
 
   //--- Set ref-B codecs
   EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
@@ -169,11 +165,6 @@ void TwoWayCommunication::SetUp() {
   _channelRef_B2A = new Channel;
   _acmRefB->RegisterTransportCallback(_channelRef_B2A);
   _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
-
-  // The clicks will be more obvious when we
-  // are in FAX mode.
-  EXPECT_EQ(_acmB->SetPlayoutMode(fax), 0);
-  EXPECT_EQ(_acmRefB->SetPlayoutMode(fax), 0);
 }
 
 void TwoWayCommunication::SetUpAutotest() {
@@ -188,10 +179,6 @@ void TwoWayCommunication::SetUpAutotest() {
   //--- Set A codecs
   EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
   EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
-#ifdef WEBRTC_DTMF_DETECTION
-  _dtmfDetectorA = new(DTMFDetector);
-  EXPECT_EQ(0, _acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA, ACMUSA));
-#endif
 
   //--- Set ref-A codecs
   EXPECT_GT(_acmRefA->RegisterSendCodec(codecInst_A), -1);
@@ -200,10 +187,6 @@ void TwoWayCommunication::SetUpAutotest() {
   //--- Set B codecs
   EXPECT_GT(_acmB->RegisterSendCodec(codecInst_B), -1);
   EXPECT_GT(_acmB->RegisterReceiveCodec(codecInst_A), -1);
-#ifdef WEBRTC_DTMF_DETECTION
-  _dtmfDetectorB = new(DTMFDetector);
-  EXPECT_EQ(0, _acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB, ACMUSA));
-#endif
 
   //--- Set ref-B codecs
   EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
@@ -251,11 +234,6 @@ void TwoWayCommunication::SetUpAutotest() {
   _channelRef_B2A = new Channel;
   _acmRefB->RegisterTransportCallback(_channelRef_B2A);
   _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
-
-  // The clicks will be more obvious when we
-  // are in FAX mode.
-  EXPECT_EQ(0, _acmB->SetPlayoutMode(fax));
-  EXPECT_EQ(0, _acmRefB->SetPlayoutMode(fax));
 }
 
 void TwoWayCommunication::Perform() {
@@ -272,15 +250,13 @@ void TwoWayCommunication::Perform() {
 
   AudioFrame audioFrame;
 
-  CodecInst codecInst_B;
-  CodecInst dummy;
-
-  EXPECT_EQ(0, _acmB->SendCodec(&codecInst_B));
+  auto codecInst_B = _acmB->SendCodec();
+  ASSERT_TRUE(codecInst_B);
 
   // In the following loop we tests that the code can handle misuse of the APIs.
   // In the middle of a session with data flowing between two sides, called A
-  // and B, APIs will be called, like ResetEncoder(), and the code should
-  // continue to run, and be able to recover.
+  // and B, APIs will be called, and the code should continue to run, and be
+  // able to recover.
   while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
     msecPassed += 10;
     EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
@@ -305,24 +281,17 @@ void TwoWayCommunication::Perform() {
       msecPassed = 0;
       secPassed++;
     }
-    // Call RestEncoder for ACM on side A, and InitializeSender for ACM on
-    // side B.
-    if (((secPassed % 5) == 4) && (msecPassed == 0)) {
-      EXPECT_EQ(0, _acmA->ResetEncoder());
-    }
     // Re-register send codec on side B.
     if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
-      EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
-      EXPECT_EQ(0, _acmB->SendCodec(&dummy));
+      EXPECT_EQ(0, _acmB->RegisterSendCodec(*codecInst_B));
+      EXPECT_TRUE(_acmB->SendCodec());
     }
-    // Reset decoder on side B, and initialize receiver on side A.
-    if (((secPassed % 7) == 6) && (msecPassed == 0)) {
-      EXPECT_EQ(0, _acmB->ResetDecoder());
+    // Initialize receiver on side A.
+    if (((secPassed % 7) == 6) && (msecPassed == 0))
       EXPECT_EQ(0, _acmA->InitializeReceiver());
-    }
     // Re-register codec on side A.
     if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
-      EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
+      EXPECT_EQ(0, _acmA->RegisterReceiveCodec(*codecInst_B));
     }
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.h
similarity index 69%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.h
index e591bbabe9..77639935da 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/TwoWayCommunication.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 
 namespace webrtc {
 
@@ -57,4 +57,4 @@ class TwoWayCommunication : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/delay_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/delay_test.cc
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/delay_test.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/delay_test.cc
index 0f9e8b07f5..a8c137f501 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/delay_test.cc
@@ -19,13 +19,13 @@
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 DEFINE_string(codec, "isac", "Codec Name");
@@ -33,7 +33,6 @@ DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
 DEFINE_int32(num_channels, 1, "Number of Channels.");
 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
 DEFINE_int32(delay, 0, "Delay in millisecond.");
-DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
@@ -89,10 +88,6 @@ class DelayTest {
         "Couldn't initialize receiver.\n";
     ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
         "Couldn't initialize receiver.\n";
-    if (FLAGS_init_delay > 0) {
-      ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
-          "Failed to set initial delay.\n";
-    }
 
     if (FLAGS_delay > 0) {
       ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
@@ -172,7 +167,7 @@ class DelayTest {
   void OpenOutFile(const char* output_id) {
     std::stringstream file_stream;
     file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
-        << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
+        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
     out_file_b_.Open(file_name.c_str(), 32000, "wb");
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.cc
similarity index 77%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.cc
index 26236f2caf..9f223fb81f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
 
 #include <ctype.h>
 #include <stdio.h>
@@ -23,11 +23,11 @@
 #include <time.h>
 #endif
 
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -35,8 +35,6 @@ namespace webrtc {
 void SetISACConfigDefault(ACMTestISACConfig& isacConfig) {
   isacConfig.currentRateBitPerSec = 0;
   isacConfig.currentFrameSizeMsec = 0;
-  isacConfig.maxRateBitPerSec = 0;
-  isacConfig.maxPayloadSizeByte = 0;
   isacConfig.encodingMode = -1;
   isacConfig.initRateBitPerSec = 0;
   isacConfig.initFrameSizeInMsec = 0;
@@ -49,40 +47,24 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
 
   if ((isacConfig.currentRateBitPerSec != 0)
       || (isacConfig.currentFrameSizeMsec != 0)) {
-    CodecInst sendCodec;
-    EXPECT_EQ(0, acm->SendCodec(&sendCodec));
+    auto sendCodec = acm->SendCodec();
+    EXPECT_TRUE(sendCodec);
     if (isacConfig.currentRateBitPerSec < 0) {
       // Register iSAC in adaptive (channel-dependent) mode.
-      sendCodec.rate = -1;
-      EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+      sendCodec->rate = -1;
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
     } else {
       if (isacConfig.currentRateBitPerSec != 0) {
-        sendCodec.rate = isacConfig.currentRateBitPerSec;
+        sendCodec->rate = isacConfig.currentRateBitPerSec;
       }
       if (isacConfig.currentFrameSizeMsec != 0) {
-        sendCodec.pacsize = isacConfig.currentFrameSizeMsec
-            * (sendCodec.plfreq / 1000);
+        sendCodec->pacsize = isacConfig.currentFrameSizeMsec
+            * (sendCodec->plfreq / 1000);
       }
-      EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
     }
   }
 
-  if (isacConfig.maxRateBitPerSec > 0) {
-    // Set max rate.
-    EXPECT_EQ(0, acm->SetISACMaxRate(isacConfig.maxRateBitPerSec));
-  }
-  if (isacConfig.maxPayloadSizeByte > 0) {
-    // Set max payload size.
-    EXPECT_EQ(0, acm->SetISACMaxPayloadSize(isacConfig.maxPayloadSizeByte));
-  }
-  if ((isacConfig.initFrameSizeInMsec != 0)
-      || (isacConfig.initRateBitPerSec != 0)) {
-    EXPECT_EQ(0, acm->ConfigISACBandwidthEstimator(
-        static_cast<uint8_t>(isacConfig.initFrameSizeInMsec),
-        static_cast<uint16_t>(isacConfig.initRateBitPerSec),
-        isacConfig.enforceFrameSize));
-  }
-
   return 0;
 }
 
@@ -135,6 +117,10 @@ void ISACTest::Setup() {
   EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
 
   _inFileA.Open(file_name_swb_, 32000, "rb");
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _inFileA.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  _inFileA.FastForward(100);
   std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
   std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
   _outFileA.Open(fileNameA, 32000, "wb");
@@ -200,41 +186,6 @@ void ISACTest::Perform() {
   testNr++;
   EncodeDecode(testNr, wbISACConfig, swbISACConfig);
 
-  int user_input;
-  if ((_testMode == 0) || (_testMode == 1)) {
-    swbISACConfig.maxPayloadSizeByte = static_cast<uint16_t>(200);
-    wbISACConfig.maxPayloadSizeByte = static_cast<uint16_t>(200);
-  } else {
-    printf("Enter the max payload-size for side A: ");
-    CHECK_ERROR(scanf("%d", &user_input));
-    swbISACConfig.maxPayloadSizeByte = (uint16_t) user_input;
-    printf("Enter the max payload-size for side B: ");
-    CHECK_ERROR(scanf("%d", &user_input));
-    wbISACConfig.maxPayloadSizeByte = (uint16_t) user_input;
-  }
-  testNr++;
-  EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
-  _acmA->ResetEncoder();
-  _acmB->ResetEncoder();
-  SetISACConfigDefault(wbISACConfig);
-  SetISACConfigDefault(swbISACConfig);
-
-  if ((_testMode == 0) || (_testMode == 1)) {
-    swbISACConfig.maxRateBitPerSec = static_cast<uint32_t>(48000);
-    wbISACConfig.maxRateBitPerSec = static_cast<uint32_t>(48000);
-  } else {
-    printf("Enter the max rate for side A: ");
-    CHECK_ERROR(scanf("%d", &user_input));
-    swbISACConfig.maxRateBitPerSec = (uint32_t) user_input;
-    printf("Enter the max rate for side B: ");
-    CHECK_ERROR(scanf("%d", &user_input));
-    wbISACConfig.maxRateBitPerSec = (uint32_t) user_input;
-  }
-
-  testNr++;
-  EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
   testNr++;
   if (_testMode == 0) {
     SwitchingSamplingRate(testNr, 4);
@@ -291,8 +242,7 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
   _channel_B2A->ResetStats();
 
   char currentTime[500];
-  CodecInst sendCodec;
-  EventWrapper* myEvent = EventWrapper::Create();
+  EventTimerWrapper* myEvent = EventTimerWrapper::Create();
   EXPECT_TRUE(myEvent->StartTimer(true, 10));
   while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
     Run10ms();
@@ -301,8 +251,8 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
 
     if ((adaptiveMode) && (_testMode != 0)) {
       myEvent->Wait(5000);
-      EXPECT_EQ(0, _acmA->SendCodec(&sendCodec));
-      EXPECT_EQ(0, _acmB->SendCodec(&sendCodec));
+      EXPECT_TRUE(_acmA->SendCodec());
+      EXPECT_TRUE(_acmB->SendCodec());
     }
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.h
index f4223f7512..c5bb515437 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/iSACTest.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/iSACTest.h
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
 
 #include <string.h>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
 
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 #define NO_OF_CLIENTS             15
@@ -29,8 +29,6 @@ namespace webrtc {
 struct ACMTestISACConfig {
   int32_t currentRateBitPerSec;
   int16_t currentFrameSizeMsec;
-  uint32_t maxRateBitPerSec;
-  int16_t maxPayloadSizeByte;
   int16_t encodingMode;
   uint32_t initRateBitPerSec;
   int16_t initFrameSizeInMsec;
@@ -78,4 +76,4 @@ class ISACTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
index 7331696c20..481df55ffd 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -14,11 +14,11 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 // Codec.
@@ -42,7 +42,6 @@ DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
 DEFINE_string(delay, "", "Log for delay.");
 
 // Other setups
-DEFINE_int32(init_delay, 0, "Initial delay.");
 DEFINE_bool(verbose, false, "Verbosity.");
 DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
 
@@ -122,9 +121,6 @@ class InsertPacketWithTiming {
         << " Hz." << std::endl;
 
     // Other setups
-    if (FLAGS_init_delay > 0)
-      EXPECT_EQ(0, receive_acm_->SetInitialPlayoutDelay(FLAGS_init_delay));
-
     if (FLAGS_loss_rate > 0)
       loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
     else
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.cc
similarity index 85%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.cc
index 09301df51c..104b5e587b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/main/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
 
 #include <assert.h>
 
@@ -17,12 +17,11 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -63,7 +62,7 @@ void OpusTest::Perform() {
   return;
 #else
   uint16_t frequency_hz;
-  int audio_channels;
+  size_t audio_channels;
   int16_t test_cntr = 0;
 
   // Open both mono and stereo test files in 32 kHz.
@@ -84,8 +83,8 @@ void OpusTest::Perform() {
   // Create Opus decoders for mono and stereo for stand-alone testing of Opus.
   ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
   ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
-  ASSERT_GT(WebRtcOpus_DecoderInit(opus_mono_decoder_), -1);
-  ASSERT_GT(WebRtcOpus_DecoderInit(opus_stereo_decoder_), -1);
+  WebRtcOpus_DecoderInit(opus_mono_decoder_);
+  WebRtcOpus_DecoderInit(opus_stereo_decoder_);
 
   ASSERT_TRUE(acm_receiver_.get() != NULL);
   EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
@@ -206,17 +205,17 @@ void OpusTest::Perform() {
 #endif
 }
 
-void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
-                   int frame_length, int percent_loss) {
+void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
+                   size_t frame_length, int percent_loss) {
   AudioFrame audio_frame;
   int32_t out_freq_hz_b = out_file_.SamplingFrequency();
-  const int kBufferSizeSamples = 480 * 12 * 2;  // Can hold 120 ms stereo audio.
+  const size_t kBufferSizeSamples = 480 * 12 * 2;  // 120 ms stereo audio.
   int16_t audio[kBufferSizeSamples];
   int16_t out_audio[kBufferSizeSamples];
   int16_t audio_type;
-  int written_samples = 0;
-  int read_samples = 0;
-  int decoded_samples = 0;
+  size_t written_samples = 0;
+  size_t read_samples = 0;
+  size_t decoded_samples = 0;
   bool first_packet = true;
   uint32_t start_time_stamp = 0;
 
@@ -236,8 +235,12 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
                                         kOpusComplexity5));
 #endif
 
-  // Make sure the runtime is less than 60 seconds to pass Android test.
-  for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) {
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_.FastForward(100);
+  in_file_mono_.FastForward(100);
+
+  // Limit the runtime to 1000 blocks of 10 ms each.
+  for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
     bool lost_packet = false;
 
     // Get 10 msec of audio.
@@ -265,25 +268,19 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
 
     // Sometimes we need to loop over the audio vector to produce the right
     // number of packets.
-    int loop_encode = (written_samples - read_samples) /
+    size_t loop_encode = (written_samples - read_samples) /
         (channels * frame_length);
 
     if (loop_encode > 0) {
-      const int kMaxBytes = 1000;  // Maximum number of bytes for one packet.
-      int16_t bitstream_len_byte;
+      const size_t kMaxBytes = 1000;  // Maximum number of bytes for one packet.
+      size_t bitstream_len_byte;
       uint8_t bitstream[kMaxBytes];
-      for (int i = 0; i < loop_encode; i++) {
-        if (channels == 1) {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_mono_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GT(bitstream_len_byte, -1);
-        } else {
-          bitstream_len_byte = WebRtcOpus_Encode(
-              opus_stereo_encoder_, &audio[read_samples],
-              frame_length, kMaxBytes, bitstream);
-          ASSERT_GT(bitstream_len_byte, -1);
-        }
+      for (size_t i = 0; i < loop_encode; i++) {
+        int bitstream_len_byte_int = WebRtcOpus_Encode(
+            (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
+            &audio[read_samples], frame_length, kMaxBytes, bitstream);
+        ASSERT_GE(bitstream_len_byte_int, 0);
+        bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
 
         // Simulate packet loss by setting |packet_loss_| to "true" in
         // |percent_loss| percent of the loops.
@@ -329,7 +326,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
           first_packet = false;
           start_time_stamp = rtp_timestamp_;
         }
-        rtp_timestamp_ += frame_length;
+        rtp_timestamp_ += static_cast<uint32_t>(frame_length);
         read_samples += frame_length * channels;
       }
       if (read_samples == written_samples) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.h
similarity index 63%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.h
index 4c3d8c160e..93c9ffb263 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/opus_test.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/opus_test.h
@@ -8,17 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
 
 #include <math.h>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
 
 namespace webrtc {
 
@@ -30,7 +31,10 @@ class OpusTest : public ACMTest {
   void Perform();
 
  private:
-  void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
+  void Run(TestPackStereo* channel,
+           size_t channels,
+           int bitrate,
+           size_t frame_length,
            int percent_loss = 0);
 
   void OpenOutFile(int test_number);
@@ -43,7 +47,7 @@ class OpusTest : public ACMTest {
   PCMFile out_file_standalone_;
   int counter_;
   uint8_t payload_type_;
-  int rtp_timestamp_;
+  uint32_t rtp_timestamp_;
   acm2::ACMResampler resampler_;
   WebRtcOpusEncInst* opus_mono_encoder_;
   WebRtcOpusEncInst* opus_stereo_encoder_;
@@ -53,4 +57,4 @@ class OpusTest : public ACMTest {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/target_delay_unittest.cc
similarity index 81%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index f1c43829bc..195e9d8145 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -11,13 +11,12 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -154,7 +153,7 @@ class TargetDelayTest : public ::testing::Test {
       ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
       // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
       ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
-      ASSERT_EQ(1, frame.num_channels_);
+      ASSERT_EQ(1u, frame.num_channels_);
       ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
     }
   }
@@ -199,23 +198,50 @@ class TargetDelayTest : public ::testing::Test {
   uint8_t payload_[kPayloadLenBytes];
 };
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutOfRangeInput DISABLED_OutOfRangeInput
+#else
+#define MAYBE_OutOfRangeInput OutOfRangeInput
+#endif
+TEST_F(TargetDelayTest, MAYBE_OutOfRangeInput) {
   OutOfRangeInput();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NoTargetDelayBufferSizeChanges \
+  DISABLED_NoTargetDelayBufferSizeChanges
+#else
+#define MAYBE_NoTargetDelayBufferSizeChanges NoTargetDelayBufferSizeChanges
+#endif
+TEST_F(TargetDelayTest, MAYBE_NoTargetDelayBufferSizeChanges) {
   NoTargetDelayBufferSizeChanges();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_WithTargetDelayBufferNotChanging \
+  DISABLED_WithTargetDelayBufferNotChanging
+#else
+#define MAYBE_WithTargetDelayBufferNotChanging WithTargetDelayBufferNotChanging
+#endif
+TEST_F(TargetDelayTest, MAYBE_WithTargetDelayBufferNotChanging) {
   WithTargetDelayBufferNotChanging();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_RequiredDelayAtCorrectRange DISABLED_RequiredDelayAtCorrectRange
+#else
+#define MAYBE_RequiredDelayAtCorrectRange RequiredDelayAtCorrectRange
+#endif
+TEST_F(TargetDelayTest, MAYBE_RequiredDelayAtCorrectRange) {
   RequiredDelayAtCorrectRange();
 }
 
-TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
+#else
+#define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
+#endif
+TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
   TargetDelayBufferMinMax();
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.cc b/media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.cc
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.cc
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.cc
index e4e6dd4a35..89368bce51 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.cc
@@ -18,8 +18,8 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 
 #define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
 
@@ -279,32 +279,6 @@ bool FixedPayloadTypeCodec(const char* payloadName) {
   return false;
 }
 
-DTMFDetector::DTMFDetector() {
-  for (int16_t n = 0; n < 1000; n++) {
-    _toneCntr[n] = 0;
-  }
-}
-
-DTMFDetector::~DTMFDetector() {
-}
-
-int32_t DTMFDetector::IncomingDtmf(const uint8_t digitDtmf,
-                                   const bool /* toneEnded */) {
-  fprintf(stdout, "%d-", digitDtmf);
-  _toneCntr[digitDtmf]++;
-  return 0;
-}
-
-void DTMFDetector::PrintDetectedDigits() {
-  for (int16_t n = 0; n < 1000; n++) {
-    if (_toneCntr[n] > 0) {
-      fprintf(stdout, "%d %u  msec, \n", n, _toneCntr[n] * 10);
-    }
-  }
-  fprintf(stdout, "\n");
-  return;
-}
-
 void VADCallback::Reset() {
   memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
 }
@@ -314,7 +288,7 @@ VADCallback::VADCallback() {
 }
 
 void VADCallback::PrintFrameTypes() {
-  printf("kFrameEmpty......... %d\n", _numFrameTypes[kFrameEmpty]);
+  printf("kEmptyFrame......... %d\n", _numFrameTypes[kEmptyFrame]);
   printf("kAudioFrameSpeech... %d\n", _numFrameTypes[kAudioFrameSpeech]);
   printf("kAudioFrameCN....... %d\n", _numFrameTypes[kAudioFrameCN]);
   printf("kVideoFrameKey...... %d\n", _numFrameTypes[kVideoFrameKey]);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.h b/media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.h
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.h
rename to media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.h
index eccb68f6d1..23869be7ed 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/utility.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/test/utility.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 
 namespace webrtc {
 
@@ -115,19 +115,6 @@ void PrintCodecs();
 
 bool FixedPayloadTypeCodec(const char* payloadName);
 
-class DTMFDetector : public AudioCodingFeedback {
- public:
-  DTMFDetector();
-  ~DTMFDetector();
-  // used for inband DTMF detection
-  int32_t IncomingDtmf(const uint8_t digitDtmf, const bool toneEnded);
-  void PrintDetectedDigits();
-
- private:
-  uint32_t _toneCntr[1000];
-
-};
-
 class VADCallback : public ACMVADCallback {
  public:
   VADCallback();
@@ -149,4 +136,4 @@ void UseNewAcm(webrtc::Config* config);
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/BUILD.gn b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/BUILD.gn
index 31f2e9affd..36391c7abc 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -9,21 +9,19 @@
 config("audio_conference_mixer_config") {
   visibility = [ ":*" ]  # Only targets in this file can depend on this.
   include_dirs = [
-    "interface",
-    "../interface",
+    "include",
+    "../include",
   ]
 }
 
 source_set("audio_conference_mixer") {
   sources = [
-    "interface/audio_conference_mixer.h",
-    "interface/audio_conference_mixer_defines.h",
+    "include/audio_conference_mixer.h",
+    "include/audio_conference_mixer_defines.h",
     "source/audio_conference_mixer_impl.cc",
     "source/audio_conference_mixer_impl.h",
     "source/audio_frame_manipulator.cc",
     "source/audio_frame_manipulator.h",
-    "source/level_indicator.cc",
-    "source/level_indicator.h",
     "source/memory_pool.h",
     "source/memory_pool_posix.h",
     "source/memory_pool_win.h",
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/OWNERS
index 34bc7389ba..ea2062a9a0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/OWNERS
@@ -1,3 +1,8 @@
-andrew@webrtc.org
+minyue@webrtc.org
+
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
 
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
index 51ee6891d5..9d7179504c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
@@ -17,12 +17,10 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'interface/audio_conference_mixer.h',
-        'interface/audio_conference_mixer_defines.h',
+        'include/audio_conference_mixer.h',
+        'include/audio_conference_mixer_defines.h',
         'source/audio_frame_manipulator.cc',
         'source/audio_frame_manipulator.h',
-        'source/level_indicator.cc',
-        'source/level_indicator.h',
         'source/memory_pool.h',
         'source/memory_pool_posix.h',
         'source/memory_pool_win.h',
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
similarity index 58%
rename from media/webrtc/trunk/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
rename to media/webrtc/trunk/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
index b9be6c649d..7370442704 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
@@ -8,16 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
 
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioMixerOutputReceiver;
-class AudioMixerStatusReceiver;
 class MixerParticipant;
 class Trace;
 
@@ -45,31 +44,25 @@ public:
 
     // Register/unregister a callback class for receiving the mixed audio.
     virtual int32_t RegisterMixedStreamCallback(
-        AudioMixerOutputReceiver& receiver) = 0;
+        AudioMixerOutputReceiver* receiver) = 0;
     virtual int32_t UnRegisterMixedStreamCallback() = 0;
 
-    // Register/unregister a callback class for receiving status information.
-    virtual int32_t RegisterMixerStatusCallback(
-        AudioMixerStatusReceiver& mixerStatusCallback,
-        const uint32_t amountOf10MsBetweenCallbacks) = 0;
-    virtual int32_t UnRegisterMixerStatusCallback() = 0;
-
     // Add/remove participants as candidates for mixing.
-    virtual int32_t SetMixabilityStatus(MixerParticipant& participant,
+    virtual int32_t SetMixabilityStatus(MixerParticipant* participant,
                                         bool mixable) = 0;
-    // mixable is set to true if a participant is a candidate for mixing.
-    virtual int32_t MixabilityStatus(MixerParticipant& participant,
-                                     bool& mixable) = 0;
+    // Returns true if a participant is a candidate for mixing.
+    virtual bool MixabilityStatus(
+        const MixerParticipant& participant) const = 0;
 
     // Inform the mixer that the participant should always be mixed and not
     // count toward the number of mixed participants. Note that a participant
     // must have been added to the mixer (by calling SetMixabilityStatus())
     // before this function can be successfully called.
-    virtual int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
-                                                 const bool mixable) = 0;
-    // mixable is set to true if the participant is mixed anonymously.
-    virtual int32_t AnonymousMixabilityStatus(MixerParticipant& participant,
-                                              bool& mixable) = 0;
+    virtual int32_t SetAnonymousMixabilityStatus(
+        MixerParticipant* participant, bool mixable) = 0;
+    // Returns true if the participant is mixed anonymously.
+    virtual bool AnonymousMixabilityStatus(
+        const MixerParticipant& participant) const = 0;
 
     // Set the minimum sampling frequency at which to mix. The mixing algorithm
     // may still choose to mix at a higher samling frequency to avoid
@@ -81,4 +74,4 @@ protected:
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
new file mode 100644
index 0000000000..5d58f42435
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class MixHistory;
+
+// A callback class that all mixer participants must inherit from/implement.
+class MixerParticipant
+{
+public:
+    // The implementation of this function should update audioFrame with new
+    // audio every time it's called.
+    //
+    // If it returns -1, the frame will not be added to the mix.
+    virtual int32_t GetAudioFrame(int32_t id,
+                                  AudioFrame* audioFrame) = 0;
+
+    // Returns true if the participant was mixed this mix iteration.
+    bool IsMixed() const;
+
+    // This function specifies the sampling frequency needed for the AudioFrame
+    // for future GetAudioFrame(..) calls.
+    virtual int32_t NeededFrequency(int32_t id) const = 0;
+
+    MixHistory* _mixHistory;
+protected:
+    MixerParticipant();
+    virtual ~MixerParticipant();
+};
+
+class AudioMixerOutputReceiver
+{
+public:
+    // This callback function provides the mixed audio for this mix iteration.
+    // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
+    // size according to the size parameter.
+    virtual void NewMixedAudio(const int32_t id,
+                               const AudioFrame& generalAudioFrame,
+                               const AudioFrame** uniqueAudioFrames,
+                               const uint32_t size) = 0;
+protected:
+    AudioMixerOutputReceiver() {}
+    virtual ~AudioMixerOutputReceiver() {}
+};
+}  // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
deleted file mode 100644
index 663be182dd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class MixHistory;
-
-// A callback class that all mixer participants must inherit from/implement.
-class MixerParticipant
-{
-public:
-    // The implementation of this function should update audioFrame with new
-    // audio every time it's called.
-    //
-    // If it returns -1, the frame will not be added to the mix.
-    virtual int32_t GetAudioFrame(const int32_t id, AudioFrame& audioFrame) = 0;
-
-    // mixed will be set to true if the participant was mixed this mix iteration
-    int32_t IsMixed(bool& mixed) const;
-
-    // This function specifies the sampling frequency needed for the AudioFrame
-    // for future GetAudioFrame(..) calls.
-    virtual int32_t NeededFrequency(const int32_t id) = 0;
-
-    MixHistory* _mixHistory;
-protected:
-    MixerParticipant();
-    virtual ~MixerParticipant();
-};
-
-// Container struct for participant statistics.
-struct ParticipantStatistics
-{
-    int32_t participant;
-    int32_t level;
-};
-
-class AudioMixerStatusReceiver
-{
-public:
-    // Callback function that provides an array of ParticipantStatistics for the
-    // participants that were mixed last mix iteration.
-    virtual void MixedParticipants(
-        const int32_t id,
-        const ParticipantStatistics* participantStatistics,
-        const uint32_t size) = 0;
-    // Callback function that provides an array of the ParticipantStatistics for
-    // the participants that had a positiv VAD last mix iteration.
-    virtual void VADPositiveParticipants(
-        const int32_t id,
-        const ParticipantStatistics* participantStatistics,
-        const uint32_t size) = 0;
-    // Callback function that provides the audio level of the mixed audio frame
-    // from the last mix iteration.
-    virtual void MixedAudioLevel(
-        const int32_t  id,
-        const uint32_t level) = 0;
-protected:
-    AudioMixerStatusReceiver() {}
-    virtual ~AudioMixerStatusReceiver() {}
-};
-
-class AudioMixerOutputReceiver
-{
-public:
-    // This callback function provides the mixed audio for this mix iteration.
-    // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
-    // size according to the size parameter.
-    virtual void NewMixedAudio(const int32_t id,
-                               const AudioFrame& generalAudioFrame,
-                               const AudioFrame** uniqueAudioFrames,
-                               const uint32_t size) = 0;
-protected:
-    AudioMixerOutputReceiver() {}
-    virtual ~AudioMixerOutputReceiver() {}
-};
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 3ee2a08634..afb060f46d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace {
@@ -50,8 +50,8 @@ void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
 }
 
 // Return the max number of channels from a |list| composed of AudioFrames.
-int MaxNumChannels(const AudioFrameList* list) {
-  int max_num_channels = 1;
+size_t MaxNumChannels(const AudioFrameList* list) {
+  size_t max_num_channels = 1;
   for (AudioFrameList::const_iterator iter = list->begin();
        iter != list->end();
        ++iter) {
@@ -60,12 +60,6 @@ int MaxNumChannels(const AudioFrameList* list) {
   return max_num_channels;
 }
 
-void SetParticipantStatistics(ParticipantStatistics* stats,
-                              const AudioFrame& frame) {
-    stats->participant = frame.id_;
-    stats->level = 0;  // TODO(andrew): to what should this be set?
-}
-
 }  // namespace
 
 MixerParticipant::MixerParticipant()
@@ -76,8 +70,8 @@ MixerParticipant::~MixerParticipant() {
     delete _mixHistory;
 }
 
-int32_t MixerParticipant::IsMixed(bool& mixed) const {
-    return _mixHistory->IsMixed(mixed);
+bool MixerParticipant::IsMixed() const {
+    return _mixHistory->IsMixed();
 }
 
 MixHistory::MixHistory()
@@ -87,15 +81,14 @@ MixHistory::MixHistory()
 MixHistory::~MixHistory() {
 }
 
-int32_t MixHistory::IsMixed(bool& mixed) const {
-    mixed = _isMixed;
-    return 0;
+bool MixHistory::IsMixed() const {
+    return _isMixed;
 }
 
-int32_t MixHistory::WasMixed(bool& wasMixed) const {
+bool MixHistory::WasMixed() const {
     // Was mixed is the same as is mixed depending on perspective. This function
     // is for the perspective of AudioConferenceMixerImpl.
-    return IsMixed(wasMixed);
+    return IsMixed();
 }
 
 int32_t MixHistory::SetIsMixed(const bool mixed) {
@@ -117,17 +110,9 @@ AudioConferenceMixer* AudioConferenceMixer::Create(int id) {
 }
 
 AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id)
-    : _scratchParticipantsToMixAmount(0),
-      _scratchMixedParticipants(),
-      _scratchVadPositiveParticipantsAmount(0),
-      _scratchVadPositiveParticipants(),
-      _id(id),
+    : _id(id),
       _minimumMixingFreq(kLowestPossible),
       _mixReceiver(NULL),
-      _mixerStatusCallback(NULL),
-      _amountOf10MsBetweenCallbacks(1),
-      _amountOf10MsUntilNextCallback(0),
-      _mixerStatusCb(false),
       _outputFrequency(kDefaultFrequency),
       _sampleSize(0),
       _audioFramePool(NULL),
@@ -137,7 +122,6 @@ AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id)
       use_limiter_(true),
       _timeStamp(0),
       _timeScheduler(kProcessPeriodicityInMs),
-      _mixedAudioLevel(),
       _processCalls(0) {}
 
 bool AudioConferenceMixerImpl::Init() {
@@ -271,11 +255,10 @@ int32_t AudioConferenceMixerImpl::Process() {
         }
 
         UpdateToMix(&mixList, &rampOutList, &mixedParticipantsMap,
-                    remainingParticipantsAllowedToMix);
+                    &remainingParticipantsAllowedToMix);
 
         GetAdditionalAudio(&additionalFramesList);
         UpdateMixedStatus(mixedParticipantsMap);
-        _scratchParticipantsToMixAmount = mixedParticipantsMap.size();
     }
 
     // Get an AudioFrame for mixing from the memory pool.
@@ -287,9 +270,7 @@ int32_t AudioConferenceMixerImpl::Process() {
         return -1;
     }
 
-    bool timeForMixerCallback = false;
     int retval = 0;
-    int32_t audioLevel = 0;
     {
         CriticalSectionScoped cs(_crit.get());
 
@@ -297,7 +278,7 @@ int32_t AudioConferenceMixerImpl::Process() {
         //                with an API instead of dynamically.
 
         // Find the max channels over all mixing lists.
-        const int num_mixed_channels = std::max(MaxNumChannels(&mixList),
+        const size_t num_mixed_channels = std::max(MaxNumChannels(&mixList),
             std::max(MaxNumChannels(&additionalFramesList),
                      MaxNumChannels(&rampOutList)));
 
@@ -305,16 +286,17 @@ int32_t AudioConferenceMixerImpl::Process() {
                                 AudioFrame::kNormalSpeech,
                                 AudioFrame::kVadPassive, num_mixed_channels);
 
-        _timeStamp += _sampleSize;
+        _timeStamp += static_cast<uint32_t>(_sampleSize);
 
         // We only use the limiter if it supports the output sample rate and
         // we're actually mixing multiple streams.
-        use_limiter_ = _numMixedParticipants > 1 &&
-                       _outputFrequency <= kAudioProcMaxNativeSampleRateHz;
+        use_limiter_ =
+            _numMixedParticipants > 1 &&
+            _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
 
-        MixFromList(*mixedAudio, &mixList);
-        MixAnonomouslyFromList(*mixedAudio, &additionalFramesList);
-        MixAnonomouslyFromList(*mixedAudio, &rampOutList);
+        MixFromList(mixedAudio, mixList);
+        MixAnonomouslyFromList(mixedAudio, additionalFramesList);
+        MixAnonomouslyFromList(mixedAudio, rampOutList);
 
         if(mixedAudio->samples_per_channel_ == 0) {
             // Nothing was mixed, set the audio samples to silence.
@@ -322,21 +304,9 @@ int32_t AudioConferenceMixerImpl::Process() {
             mixedAudio->Mute();
         } else {
             // Only call the limiter if we have something to mix.
-            if(!LimitMixedAudio(*mixedAudio))
+            if(!LimitMixedAudio(mixedAudio))
                 retval = -1;
         }
-
-        _mixedAudioLevel.ComputeLevel(mixedAudio->data_,_sampleSize);
-        audioLevel = _mixedAudioLevel.GetLevel();
-
-        if(_mixerStatusCb) {
-            _scratchVadPositiveParticipantsAmount = 0;
-            UpdateVADPositiveParticipants(&mixList);
-            if(_amountOf10MsUntilNextCallback-- == 0) {
-                _amountOf10MsUntilNextCallback = _amountOf10MsBetweenCallbacks;
-                timeForMixerCallback = true;
-            }
-        }
     }
 
     {
@@ -349,20 +319,6 @@ int32_t AudioConferenceMixerImpl::Process() {
                 dummy,
                 0);
         }
-
-        if((_mixerStatusCallback != NULL) &&
-            timeForMixerCallback) {
-            _mixerStatusCallback->MixedParticipants(
-                _id,
-                _scratchMixedParticipants,
-                static_cast<uint32_t>(_scratchParticipantsToMixAmount));
-
-            _mixerStatusCallback->VADPositiveParticipants(
-                _id,
-                _scratchVadPositiveParticipants,
-                _scratchVadPositiveParticipantsAmount);
-            _mixerStatusCallback->MixedAudioLevel(_id,audioLevel);
-        }
     }
 
     // Reclaim all outstanding memory.
@@ -378,12 +334,12 @@ int32_t AudioConferenceMixerImpl::Process() {
 }
 
 int32_t AudioConferenceMixerImpl::RegisterMixedStreamCallback(
-    AudioMixerOutputReceiver& mixReceiver) {
+    AudioMixerOutputReceiver* mixReceiver) {
     CriticalSectionScoped cs(_cbCrit.get());
     if(_mixReceiver != NULL) {
         return -1;
     }
-    _mixReceiver = &mixReceiver;
+    _mixReceiver = mixReceiver;
     return 0;
 }
 
@@ -397,11 +353,12 @@ int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() {
 }
 
 int32_t AudioConferenceMixerImpl::SetOutputFrequency(
-    const Frequency frequency) {
+    const Frequency& frequency) {
     CriticalSectionScoped cs(_crit.get());
 
     _outputFrequency = frequency;
-    _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000;
+    _sampleSize =
+        static_cast<size_t>((_outputFrequency*kProcessPeriodicityInMs) / 1000);
 
     return 0;
 }
@@ -412,56 +369,8 @@ AudioConferenceMixerImpl::OutputFrequency() const {
     return _outputFrequency;
 }
 
-int32_t AudioConferenceMixerImpl::RegisterMixerStatusCallback(
-    AudioMixerStatusReceiver& mixerStatusCallback,
-    const uint32_t amountOf10MsBetweenCallbacks) {
-    if(amountOf10MsBetweenCallbacks == 0) {
-        WEBRTC_TRACE(
-            kTraceWarning,
-            kTraceAudioMixerServer,
-            _id,
-            "amountOf10MsBetweenCallbacks(%d) needs to be larger than 0");
-        return -1;
-    }
-    {
-        CriticalSectionScoped cs(_cbCrit.get());
-        if(_mixerStatusCallback != NULL) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
-                         "Mixer status callback already registered");
-            return -1;
-        }
-        _mixerStatusCallback = &mixerStatusCallback;
-    }
-    {
-        CriticalSectionScoped cs(_crit.get());
-        _amountOf10MsBetweenCallbacks  = amountOf10MsBetweenCallbacks;
-        _amountOf10MsUntilNextCallback = 0;
-        _mixerStatusCb                 = true;
-    }
-    return 0;
-}
-
-int32_t AudioConferenceMixerImpl::UnRegisterMixerStatusCallback() {
-    {
-        CriticalSectionScoped cs(_crit.get());
-        if(!_mixerStatusCb)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
-                         "Mixer status callback not registered");
-            return -1;
-        }
-        _mixerStatusCb = false;
-    }
-    {
-        CriticalSectionScoped cs(_cbCrit.get());
-        _mixerStatusCallback = NULL;
-    }
-    return 0;
-}
-
 int32_t AudioConferenceMixerImpl::SetMixabilityStatus(
-    MixerParticipant& participant,
-    bool mixable) {
+    MixerParticipant* participant, bool mixable) {
     if (!mixable) {
         // Anonymous participants are in a separate list. Make sure that the
         // participant is in the _participantList if it is being mixed.
@@ -471,7 +380,7 @@ int32_t AudioConferenceMixerImpl::SetMixabilityStatus(
     {
         CriticalSectionScoped cs(_cbCrit.get());
         const bool isMixed =
-            IsParticipantInList(participant, &_participantList);
+            IsParticipantInList(*participant, _participantList);
         // API must be called with a new state.
         if(!(mixable ^ isMixed)) {
             WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
@@ -508,18 +417,16 @@ int32_t AudioConferenceMixerImpl::SetMixabilityStatus(
     return 0;
 }
 
-int32_t AudioConferenceMixerImpl::MixabilityStatus(
-    MixerParticipant& participant,
-    bool& mixable) {
+bool AudioConferenceMixerImpl::MixabilityStatus(
+    const MixerParticipant& participant) const {
     CriticalSectionScoped cs(_cbCrit.get());
-    mixable = IsParticipantInList(participant, &_participantList);
-    return 0;
+    return IsParticipantInList(participant, _participantList);
 }
 
 int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
-    MixerParticipant& participant, const bool anonymous) {
+    MixerParticipant* participant, bool anonymous) {
     CriticalSectionScoped cs(_cbCrit.get());
-    if(IsParticipantInList(participant, &_additionalParticipantList)) {
+    if(IsParticipantInList(*participant, _additionalParticipantList)) {
         if(anonymous) {
             return 0;
         }
@@ -551,12 +458,10 @@ int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
         0 : -1;
 }
 
-int32_t AudioConferenceMixerImpl::AnonymousMixabilityStatus(
-    MixerParticipant& participant, bool& mixable) {
+bool AudioConferenceMixerImpl::AnonymousMixabilityStatus(
+    const MixerParticipant& participant) const {
     CriticalSectionScoped cs(_cbCrit.get());
-    mixable = IsParticipantInList(participant,
-                                  &_additionalParticipantList);
-    return 0;
+    return IsParticipantInList(participant, _additionalParticipantList);
 }
 
 int32_t AudioConferenceMixerImpl::SetMinimumMixingFrequency(
@@ -583,11 +488,11 @@ int32_t AudioConferenceMixerImpl::SetMinimumMixingFrequency(
 
 // Check all AudioFrames that are to be mixed. The highest sampling frequency
 // found is the lowest that can be used without losing information.
-int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() {
+int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() const {
     const int participantListFrequency =
-        GetLowestMixingFrequencyFromList(&_participantList);
+        GetLowestMixingFrequencyFromList(_participantList);
     const int anonymousListFrequency =
-        GetLowestMixingFrequencyFromList(&_additionalParticipantList);
+        GetLowestMixingFrequencyFromList(_additionalParticipantList);
     const int highestFreq =
         (participantListFrequency > anonymousListFrequency) ?
             participantListFrequency : anonymousListFrequency;
@@ -601,10 +506,10 @@ int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() {
 }
 
 int32_t AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList(
-    MixerParticipantList* mixList) {
+    const MixerParticipantList& mixList) const {
     int32_t highestFreq = 8000;
-    for (MixerParticipantList::iterator iter = mixList->begin();
-         iter != mixList->end();
+    for (MixerParticipantList::const_iterator iter = mixList.begin();
+         iter != mixList.end();
          ++iter) {
         const int32_t neededFrequency = (*iter)->NeededFrequency(_id);
         if(neededFrequency > highestFreq) {
@@ -618,28 +523,28 @@ void AudioConferenceMixerImpl::UpdateToMix(
     AudioFrameList* mixList,
     AudioFrameList* rampOutList,
     std::map<int, MixerParticipant*>* mixParticipantList,
-    size_t& maxAudioFrameCounter) {
+    size_t* maxAudioFrameCounter) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "UpdateToMix(mixList,rampOutList,mixParticipantList,%d)",
-                 maxAudioFrameCounter);
+                 *maxAudioFrameCounter);
     const size_t mixListStartSize = mixList->size();
     AudioFrameList activeList;
     // Struct needed by the passive lists to keep track of which AudioFrame
     // belongs to which MixerParticipant.
     ParticipantFramePairList passiveWasNotMixedList;
     ParticipantFramePairList passiveWasMixedList;
-    for (MixerParticipantList::iterator participant = _participantList.begin();
-         participant != _participantList.end();
+    for (MixerParticipantList::const_iterator participant =
+        _participantList.begin(); participant != _participantList.end();
          ++participant) {
         // Stop keeping track of passive participants if there are already
         // enough participants available (they wont be mixed anyway).
-        bool mustAddToPassiveList = (maxAudioFrameCounter >
+        bool mustAddToPassiveList = (*maxAudioFrameCounter >
                                     (activeList.size() +
                                      passiveWasMixedList.size() +
                                      passiveWasNotMixedList.size()));
 
         bool wasMixed = false;
-        (*participant)->_mixHistory->WasMixed(wasMixed);
+        wasMixed = (*participant)->_mixHistory->WasMixed();
         AudioFrame* audioFrame = NULL;
         if(_audioFramePool->PopMemory(audioFrame) == -1) {
             WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
@@ -649,7 +554,7 @@ void AudioConferenceMixerImpl::UpdateToMix(
         }
         audioFrame->sample_rate_hz_ = _outputFrequency;
 
-        if((*participant)->GetAudioFrame(_id,*audioFrame) != 0) {
+        if((*participant)->GetAudioFrame(_id, audioFrame) != 0) {
             WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
                          "failed to GetAudioFrame() from participant");
             _audioFramePool->PushMemory(audioFrame);
@@ -674,7 +579,7 @@ void AudioConferenceMixerImpl::UpdateToMix(
                 RampIn(*audioFrame);
             }
 
-            if(activeList.size() >= maxAudioFrameCounter) {
+            if(activeList.size() >= *maxAudioFrameCounter) {
                 // There are already more active participants than should be
                 // mixed. Only keep the ones with the highest energy.
                 AudioFrameList::iterator replaceItem;
@@ -696,14 +601,14 @@ void AudioConferenceMixerImpl::UpdateToMix(
                     AudioFrame* replaceFrame = *replaceItem;
 
                     bool replaceWasMixed = false;
-                    std::map<int, MixerParticipant*>::iterator it =
+                    std::map<int, MixerParticipant*>::const_iterator it =
                         mixParticipantList->find(replaceFrame->id_);
 
                     // When a frame is pushed to |activeList| it is also pushed
                     // to mixParticipantList with the frame's id. This means
                     // that the Find call above should never fail.
                     assert(it != mixParticipantList->end());
-                    it->second->_mixHistory->WasMixed(replaceWasMixed);
+                    replaceWasMixed = it->second->_mixHistory->WasMixed();
 
                     mixParticipantList->erase(replaceFrame->id_);
                     activeList.erase(replaceItem);
@@ -754,10 +659,10 @@ void AudioConferenceMixerImpl::UpdateToMix(
             }
         }
     }
-    assert(activeList.size() <= maxAudioFrameCounter);
+    assert(activeList.size() <= *maxAudioFrameCounter);
     // At this point it is known which participants should be mixed. Transfer
     // this information to this functions output parameters.
-    for (AudioFrameList::iterator iter = activeList.begin();
+    for (AudioFrameList::const_iterator iter = activeList.begin();
          iter != activeList.end();
          ++iter) {
         mixList->push_back(*iter);
@@ -766,10 +671,10 @@ void AudioConferenceMixerImpl::UpdateToMix(
     // Always mix a constant number of AudioFrames. If there aren't enough
     // active participants mix passive ones. Starting with those that was mixed
     // last iteration.
-    for (ParticipantFramePairList::iterator iter = passiveWasMixedList.begin();
-         iter != passiveWasMixedList.end();
+    for (ParticipantFramePairList::const_iterator
+        iter = passiveWasMixedList.begin(); iter != passiveWasMixedList.end();
          ++iter) {
-        if(mixList->size() < maxAudioFrameCounter + mixListStartSize) {
+        if(mixList->size() < *maxAudioFrameCounter + mixListStartSize) {
             mixList->push_back((*iter)->audioFrame);
             (*mixParticipantList)[(*iter)->audioFrame->id_] =
                 (*iter)->participant;
@@ -781,11 +686,11 @@ void AudioConferenceMixerImpl::UpdateToMix(
         delete *iter;
     }
     // And finally the ones that have not been mixed for a while.
-    for (ParticipantFramePairList::iterator iter =
+    for (ParticipantFramePairList::const_iterator iter =
              passiveWasNotMixedList.begin();
          iter != passiveWasNotMixedList.end();
          ++iter) {
-        if(mixList->size() <  maxAudioFrameCounter + mixListStartSize) {
+        if(mixList->size() <  *maxAudioFrameCounter + mixListStartSize) {
           mixList->push_back((*iter)->audioFrame);
             (*mixParticipantList)[(*iter)->audioFrame->id_] =
                 (*iter)->participant;
@@ -796,12 +701,12 @@ void AudioConferenceMixerImpl::UpdateToMix(
         }
         delete *iter;
     }
-    assert(maxAudioFrameCounter + mixListStartSize >= mixList->size());
-    maxAudioFrameCounter += mixListStartSize - mixList->size();
+    assert(*maxAudioFrameCounter + mixListStartSize >= mixList->size());
+    *maxAudioFrameCounter += mixListStartSize - mixList->size();
 }
 
 void AudioConferenceMixerImpl::GetAdditionalAudio(
-    AudioFrameList* additionalFramesList) {
+    AudioFrameList* additionalFramesList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "GetAdditionalAudio(additionalFramesList)");
     // The GetAudioFrame() callback may result in the participant being removed
@@ -813,7 +718,7 @@ void AudioConferenceMixerImpl::GetAdditionalAudio(
                                      _additionalParticipantList.begin(),
                                      _additionalParticipantList.end());
 
-    for (MixerParticipantList::iterator participant =
+    for (MixerParticipantList::const_iterator participant =
              additionalParticipantList.begin();
          participant != additionalParticipantList.end();
          ++participant) {
@@ -825,7 +730,7 @@ void AudioConferenceMixerImpl::GetAdditionalAudio(
             return;
         }
         audioFrame->sample_rate_hz_ = _outputFrequency;
-        if((*participant)->GetAudioFrame(_id, *audioFrame) != 0) {
+        if((*participant)->GetAudioFrame(_id, audioFrame) != 0) {
             WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
                          "failed to GetAudioFrame() from participant");
             _audioFramePool->PushMemory(audioFrame);
@@ -841,18 +746,19 @@ void AudioConferenceMixerImpl::GetAdditionalAudio(
 }
 
 void AudioConferenceMixerImpl::UpdateMixedStatus(
-    std::map<int, MixerParticipant*>& mixedParticipantsMap) {
+    const std::map<int, MixerParticipant*>& mixedParticipantsMap) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "UpdateMixedStatus(mixedParticipantsMap)");
     assert(mixedParticipantsMap.size() <= kMaximumAmountOfMixedParticipants);
 
     // Loop through all participants. If they are in the mix map they
     // were mixed.
-    for (MixerParticipantList::iterator participant = _participantList.begin();
-         participant != _participantList.end();
+    for (MixerParticipantList::const_iterator
+        participant =_participantList.begin();
+        participant != _participantList.end();
          ++participant) {
         bool isMixed = false;
-        for (std::map<int, MixerParticipant*>::iterator it =
+        for (std::map<int, MixerParticipant*>::const_iterator it =
                  mixedParticipantsMap.begin();
              it != mixedParticipantsMap.end();
              ++it) {
@@ -866,7 +772,7 @@ void AudioConferenceMixerImpl::UpdateMixedStatus(
 }
 
 void AudioConferenceMixerImpl::ClearAudioFrameList(
-    AudioFrameList* audioFrameList) {
+    AudioFrameList* audioFrameList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "ClearAudioFrameList(audioFrameList)");
     for (AudioFrameList::iterator iter = audioFrameList->begin();
@@ -878,33 +784,24 @@ void AudioConferenceMixerImpl::ClearAudioFrameList(
 }
 
 void AudioConferenceMixerImpl::UpdateVADPositiveParticipants(
-    AudioFrameList* mixList) {
+    AudioFrameList* mixList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "UpdateVADPositiveParticipants(mixList)");
 
-    for (AudioFrameList::iterator iter = mixList->begin();
+    for (AudioFrameList::const_iterator iter = mixList->begin();
          iter != mixList->end();
          ++iter) {
         CalculateEnergy(**iter);
-        if((*iter)->vad_activity_ == AudioFrame::kVadActive) {
-            _scratchVadPositiveParticipants[
-                _scratchVadPositiveParticipantsAmount].participant =
-                (*iter)->id_;
-            // TODO(andrew): to what should this be set?
-            _scratchVadPositiveParticipants[
-                _scratchVadPositiveParticipantsAmount].level = 0;
-            _scratchVadPositiveParticipantsAmount++;
-        }
     }
 }
 
 bool AudioConferenceMixerImpl::IsParticipantInList(
-    MixerParticipant& participant,
-    MixerParticipantList* participantList) const {
+    const MixerParticipant& participant,
+    const MixerParticipantList& participantList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "IsParticipantInList(participant,participantList)");
-    for (MixerParticipantList::const_iterator iter = participantList->begin();
-         iter != participantList->end();
+    for (MixerParticipantList::const_iterator iter = participantList.begin();
+         iter != participantList.end();
          ++iter) {
         if(&participant == *iter) {
             return true;
@@ -914,28 +811,28 @@ bool AudioConferenceMixerImpl::IsParticipantInList(
 }
 
 bool AudioConferenceMixerImpl::AddParticipantToList(
-    MixerParticipant& participant,
-    MixerParticipantList* participantList) {
+    MixerParticipant* participant,
+    MixerParticipantList* participantList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "AddParticipantToList(participant, participantList)");
-    participantList->push_back(&participant);
+    participantList->push_back(participant);
     // Make sure that the mixed status is correct for new MixerParticipant.
-    participant._mixHistory->ResetMixedStatus();
+    participant->_mixHistory->ResetMixedStatus();
     return true;
 }
 
 bool AudioConferenceMixerImpl::RemoveParticipantFromList(
-    MixerParticipant& participant,
-    MixerParticipantList* participantList) {
+    MixerParticipant* participant,
+    MixerParticipantList* participantList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "RemoveParticipantFromList(participant, participantList)");
     for (MixerParticipantList::iterator iter = participantList->begin();
          iter != participantList->end();
          ++iter) {
-        if(*iter == &participant) {
+        if(*iter == participant) {
             participantList->erase(iter);
             // Participant is no longer mixed, reset to default.
-            participant._mixHistory->ResetMixedStatus();
+            participant->_mixHistory->ResetMixedStatus();
             return true;
         }
     }
@@ -943,26 +840,26 @@ bool AudioConferenceMixerImpl::RemoveParticipantFromList(
 }
 
 int32_t AudioConferenceMixerImpl::MixFromList(
-    AudioFrame& mixedAudio,
-    const AudioFrameList* audioFrameList) {
+    AudioFrame* mixedAudio,
+    const AudioFrameList& audioFrameList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "MixFromList(mixedAudio, audioFrameList)");
-    if(audioFrameList->empty()) return 0;
+    if(audioFrameList.empty()) return 0;
 
     uint32_t position = 0;
 
     if (_numMixedParticipants == 1) {
-      mixedAudio.timestamp_ = audioFrameList->front()->timestamp_;
-      mixedAudio.elapsed_time_ms_ = audioFrameList->front()->elapsed_time_ms_;
+      mixedAudio->timestamp_ = audioFrameList.front()->timestamp_;
+      mixedAudio->elapsed_time_ms_ = audioFrameList.front()->elapsed_time_ms_;
     } else {
       // TODO(wu): Issue 3390.
       // Audio frame timestamp is only supported in one channel case.
-      mixedAudio.timestamp_ = 0;
-      mixedAudio.elapsed_time_ms_ = -1;
+      mixedAudio->timestamp_ = 0;
+      mixedAudio->elapsed_time_ms_ = -1;
     }
 
-    for (AudioFrameList::const_iterator iter = audioFrameList->begin();
-         iter != audioFrameList->end();
+    for (AudioFrameList::const_iterator iter = audioFrameList.begin();
+         iter != audioFrameList.end();
          ++iter) {
         if(position >= kMaximumAmountOfMixedParticipants) {
             WEBRTC_TRACE(
@@ -975,10 +872,7 @@ int32_t AudioConferenceMixerImpl::MixFromList(
             assert(false);
             position = 0;
         }
-        MixFrames(&mixedAudio, (*iter), use_limiter_);
-
-        SetParticipantStatistics(&_scratchMixedParticipants[position],
-                                 **iter);
+        MixFrames(mixedAudio, (*iter), use_limiter_);
 
         position++;
     }
@@ -988,28 +882,28 @@ int32_t AudioConferenceMixerImpl::MixFromList(
 
 // TODO(andrew): consolidate this function with MixFromList.
 int32_t AudioConferenceMixerImpl::MixAnonomouslyFromList(
-    AudioFrame& mixedAudio,
-    const AudioFrameList* audioFrameList) {
+    AudioFrame* mixedAudio,
+    const AudioFrameList& audioFrameList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "MixAnonomouslyFromList(mixedAudio, audioFrameList)");
 
-    if(audioFrameList->empty()) return 0;
+    if(audioFrameList.empty()) return 0;
 
-    for (AudioFrameList::const_iterator iter = audioFrameList->begin();
-         iter != audioFrameList->end();
+    for (AudioFrameList::const_iterator iter = audioFrameList.begin();
+         iter != audioFrameList.end();
          ++iter) {
-        MixFrames(&mixedAudio, *iter, use_limiter_);
+        MixFrames(mixedAudio, *iter, use_limiter_);
     }
     return 0;
 }
 
-bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) {
+bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame* mixedAudio) const {
     if (!use_limiter_) {
       return true;
     }
 
     // Smoothly limit the mixed frame.
-    const int error = _limiter->ProcessStream(&mixedAudio);
+    const int error = _limiter->ProcessStream(mixedAudio);
 
     // And now we can safely restore the level. This procedure results in
     // some loss of resolution, deemed acceptable.
@@ -1021,7 +915,7 @@ bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) {
     //
     // Instead we double the frame (with addition since left-shifting a
     // negative value is undefined).
-    mixedAudio += mixedAudio;
+    *mixedAudio += *mixedAudio;
 
     if(error != _limiter->kNoError) {
         WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
index b1a812a113..2466112769 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
@@ -16,11 +16,10 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioProcessing;
@@ -36,15 +35,15 @@ public:
     MixHistory();
     ~MixHistory();
 
-    // MixerParticipant function
-    int32_t IsMixed(bool& mixed) const;
+    // Returns true if the participant is being mixed.
+    bool IsMixed() const;
 
-    // Sets wasMixed to true if the participant was mixed previous mix
+    // Returns true if the participant was mixed previous mix
     // iteration.
-    int32_t WasMixed(bool& wasMixed) const;
+    bool WasMixed() const;
 
     // Updates the mixed status.
-    int32_t SetIsMixed(const bool mixed);
+    int32_t SetIsMixed(bool mixed);
 
     void ResetMixedStatus();
 private:
@@ -69,32 +68,26 @@ public:
 
     // AudioConferenceMixer functions
     int32_t RegisterMixedStreamCallback(
-        AudioMixerOutputReceiver& mixReceiver) override;
+        AudioMixerOutputReceiver* mixReceiver) override;
     int32_t UnRegisterMixedStreamCallback() override;
-    int32_t RegisterMixerStatusCallback(
-        AudioMixerStatusReceiver& mixerStatusCallback,
-        const uint32_t amountOf10MsBetweenCallbacks) override;
-    int32_t UnRegisterMixerStatusCallback() override;
-    int32_t SetMixabilityStatus(MixerParticipant& participant,
+    int32_t SetMixabilityStatus(MixerParticipant* participant,
                                 bool mixable) override;
-    int32_t MixabilityStatus(MixerParticipant& participant,
-                             bool& mixable) override;
+    bool MixabilityStatus(const MixerParticipant& participant) const override;
     int32_t SetMinimumMixingFrequency(Frequency freq) override;
-    int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
-                                         const bool mixable) override;
-    int32_t AnonymousMixabilityStatus(MixerParticipant& participant,
-                                      bool& mixable) override;
+    int32_t SetAnonymousMixabilityStatus(
+        MixerParticipant* participant, bool mixable) override;
+    bool AnonymousMixabilityStatus(
+        const MixerParticipant& participant) const override;
 
 private:
     enum{DEFAULT_AUDIO_FRAME_POOLSIZE = 50};
 
     // Set/get mix frequency
-    int32_t SetOutputFrequency(const Frequency frequency);
+    int32_t SetOutputFrequency(const Frequency& frequency);
     Frequency OutputFrequency() const;
 
     // Fills mixList with the AudioFrames pointers that should be used when
-    // mixing. Fills mixParticipantList with ParticipantStatistics for the
-    // participants who's AudioFrames are inside mixList.
+    // mixing.
     // maxAudioFrameCounter both input and output specifies how many more
     // AudioFrames that are allowed to be mixed.
     // rampOutList contain AudioFrames corresponding to an audio stream that
@@ -104,65 +97,54 @@ private:
         AudioFrameList* mixList,
         AudioFrameList* rampOutList,
         std::map<int, MixerParticipant*>* mixParticipantList,
-        size_t& maxAudioFrameCounter);
+        size_t* maxAudioFrameCounter) const;
 
     // Return the lowest mixing frequency that can be used without having to
     // downsample any audio.
-    int32_t GetLowestMixingFrequency();
-    int32_t GetLowestMixingFrequencyFromList(MixerParticipantList* mixList);
+    int32_t GetLowestMixingFrequency() const;
+    int32_t GetLowestMixingFrequencyFromList(
+        const MixerParticipantList& mixList) const;
 
     // Return the AudioFrames that should be mixed anonymously.
-    void GetAdditionalAudio(AudioFrameList* additionalFramesList);
+    void GetAdditionalAudio(AudioFrameList* additionalFramesList) const;
 
     // Update the MixHistory of all MixerParticipants. mixedParticipantsList
     // should contain a map of MixerParticipants that have been mixed.
     void UpdateMixedStatus(
-        std::map<int, MixerParticipant*>& mixedParticipantsList);
+        const std::map<int, MixerParticipant*>& mixedParticipantsList) const;
 
     // Clears audioFrameList and reclaims all memory associated with it.
-    void ClearAudioFrameList(AudioFrameList* audioFrameList);
+    void ClearAudioFrameList(AudioFrameList* audioFrameList) const;
 
     // Update the list of MixerParticipants who have a positive VAD. mixList
     // should be a list of AudioFrames
-    void UpdateVADPositiveParticipants(
-        AudioFrameList* mixList);
+    void UpdateVADPositiveParticipants(AudioFrameList* mixList) const;
 
     // This function returns true if it finds the MixerParticipant in the
     // specified list of MixerParticipants.
-    bool IsParticipantInList(
-        MixerParticipant& participant,
-        MixerParticipantList* participantList) const;
+    bool IsParticipantInList(const MixerParticipant& participant,
+                             const MixerParticipantList& participantList) const;
 
     // Add/remove the MixerParticipant to the specified
     // MixerParticipant list.
     bool AddParticipantToList(
-        MixerParticipant& participant,
-        MixerParticipantList* participantList);
+        MixerParticipant* participant,
+        MixerParticipantList* participantList) const;
     bool RemoveParticipantFromList(
-        MixerParticipant& removeParticipant,
-        MixerParticipantList* participantList);
+        MixerParticipant* removeParticipant,
+        MixerParticipantList* participantList) const;
 
     // Mix the AudioFrames stored in audioFrameList into mixedAudio.
-    int32_t MixFromList(
-        AudioFrame& mixedAudio,
-        const AudioFrameList* audioFrameList);
+    int32_t MixFromList(AudioFrame* mixedAudio,
+                        const AudioFrameList& audioFrameList) const;
+
     // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
     // record will be kept of this mix (e.g. the corresponding MixerParticipants
     // will not be marked as IsMixed()
-    int32_t MixAnonomouslyFromList(AudioFrame& mixedAudio,
-                                   const AudioFrameList* audioFrameList);
+    int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
+                                   const AudioFrameList& audioFrameList) const;
 
-    bool LimitMixedAudio(AudioFrame& mixedAudio);
-
-    // Scratch memory
-    // Note that the scratch memory may only be touched in the scope of
-    // Process().
-    size_t         _scratchParticipantsToMixAmount;
-    ParticipantStatistics  _scratchMixedParticipants[
-        kMaximumAmountOfMixedParticipants];
-    uint32_t         _scratchVadPositiveParticipantsAmount;
-    ParticipantStatistics  _scratchVadPositiveParticipants[
-        kMaximumAmountOfMixedParticipants];
+    bool LimitMixedAudio(AudioFrame* mixedAudio) const;
 
     rtc::scoped_ptr<CriticalSectionWrapper> _crit;
     rtc::scoped_ptr<CriticalSectionWrapper> _cbCrit;
@@ -174,14 +156,9 @@ private:
     // Mix result callback
     AudioMixerOutputReceiver* _mixReceiver;
 
-    AudioMixerStatusReceiver* _mixerStatusCallback;
-    uint32_t _amountOf10MsBetweenCallbacks;
-    uint32_t _amountOf10MsUntilNextCallback;
-    bool _mixerStatusCb;
-
     // The current sample frequency and sample size when mixing.
     Frequency _outputFrequency;
-    uint16_t _sampleSize;
+    size_t _sampleSize;
 
     // Memory pool to avoid allocating/deallocating AudioFrames
     MemoryPool<AudioFrame>* _audioFramePool;
@@ -201,9 +178,6 @@ private:
     // Metronome class.
     TimeScheduler _timeScheduler;
 
-    // Smooth level indicator.
-    LevelIndicator _mixedAudioLevel;
-
     // Counter keeping track of concurrent calls to process.
     // Note: should never be higher than 1 or lower than 0.
     int16_t _processCalls;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
index 3dce5c8bea..9c5d3b939d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace {
@@ -35,14 +35,14 @@ const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
                            0.8608f, 0.8734f, 0.8861f, 0.8987f,
                            0.9114f, 0.9241f, 0.9367f, 0.9494f,
                            0.9620f, 0.9747f, 0.9873f, 1.0000f};
-const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
+const size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
 }  // namespace
 
 namespace webrtc {
 void CalculateEnergy(AudioFrame& audioFrame)
 {
     audioFrame.energy_ = 0;
-    for(int position = 0; position < audioFrame.samples_per_channel_;
+    for(size_t position = 0; position < audioFrame.samples_per_channel_;
         position++)
     {
         // TODO(andrew): this can easily overflow.
@@ -54,7 +54,7 @@ void CalculateEnergy(AudioFrame& audioFrame)
 void RampIn(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
-    for(int i = 0; i < rampSize; i++)
+    for(size_t i = 0; i < rampSize; i++)
     {
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] *
                                                    audioFrame.data_[i]);
@@ -64,9 +64,9 @@ void RampIn(AudioFrame& audioFrame)
 void RampOut(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
-    for(int i = 0; i < rampSize; i++)
+    for(size_t i = 0; i < rampSize; i++)
     {
-        const int rampPos = rampSize - 1 - i;
+        const size_t rampPos = rampSize - 1 - i;
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] *
                                                    audioFrame.data_[i]);
     }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/level_indicator.cc b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/level_indicator.cc
deleted file mode 100644
index 3c573d41ae..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/level_indicator.cc
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_conference_mixer/source/level_indicator.h"
-
-namespace webrtc {
-// Array for adding smothing to level changes (ad-hoc).
-const uint32_t perm[] =
-    {0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9};
-
-LevelIndicator::LevelIndicator()
-    : _max(0),
-      _count(0),
-      _currentLevel(0)
-{
-}
-
-LevelIndicator::~LevelIndicator()
-{
-}
-
-// Level is based on the highest absolute value for all samples.
-void LevelIndicator::ComputeLevel(const int16_t* speech,
-                                  const uint16_t nrOfSamples)
-{
-    int32_t min = 0;
-    for(uint32_t i = 0; i < nrOfSamples; i++)
-    {
-        if(_max < speech[i])
-        {
-            _max = speech[i];
-        }
-        if(min > speech[i])
-        {
-            min = speech[i];
-        }
-    }
-
-    // Absolute max value.
-    if(-min > _max)
-    {
-        _max = -min;
-    }
-
-    if(_count == TICKS_BEFORE_CALCULATION)
-    {
-        // Highest sample value maps directly to a level.
-        int32_t position = _max / 1000;
-        if ((position == 0) &&
-            (_max > 250))
-        {
-            position = 1;
-        }
-        _currentLevel = perm[position];
-        // The max value is decayed and stored so that it can be reused to slow
-        // down decreases in level.
-        _max = _max >> 1;
-        _count = 0;
-    } else {
-        _count++;
-    }
-}
-
-int32_t LevelIndicator::GetLevel()
-{
-    return _currentLevel;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/level_indicator.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/level_indicator.h
deleted file mode 100644
index b0e87ffa71..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/level_indicator.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class LevelIndicator
-{
-public:
-    enum{TICKS_BEFORE_CALCULATION = 10};
-
-    LevelIndicator();
-    ~LevelIndicator();
-
-    // Updates the level.
-    void ComputeLevel(const int16_t* speech,
-                      const uint16_t nrOfSamples);
-
-    int32_t GetLevel();
-private:
-    int32_t  _max;
-    uint32_t _count;
-    uint32_t _currentLevel;
-};
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h
index 04e7cd5225..12aae3f278 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h
@@ -14,7 +14,7 @@
 #include <assert.h>
 #include <list>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h
index 772a123ab7..3ec9187492 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h
@@ -14,8 +14,8 @@
 #include <assert.h>
 #include <windows.h>
 
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
index 4d75bc55bb..19f5bd8848 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 TimeScheduler::TimeScheduler(const int64_t periodicityInMs)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.h b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
index 5152b2dadb..09d0caa66a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
@@ -15,7 +15,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
 #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
 
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
new file mode 100644
index 0000000000..293bfa0db9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+
+namespace webrtc {
+
+using testing::_;
+using testing::AtLeast;
+using testing::Invoke;
+using testing::Return;
+
+class MockAudioMixerOutputReceiver : public AudioMixerOutputReceiver {
+ public:
+  MOCK_METHOD4(NewMixedAudio, void(const int32_t id,
+                                   const AudioFrame& general_audio_frame,
+                                   const AudioFrame** unique_audio_frames,
+                                   const uint32_t size));
+};
+
+class MockMixerParticipant : public MixerParticipant {
+ public:
+  MockMixerParticipant() {
+    ON_CALL(*this, GetAudioFrame(_, _))
+        .WillByDefault(Invoke(this, &MockMixerParticipant::FakeAudioFrame));
+  }
+  MOCK_METHOD2(GetAudioFrame,
+               int32_t(const int32_t id, AudioFrame* audio_frame));
+  MOCK_CONST_METHOD1(NeededFrequency, int32_t(const int32_t id));
+  AudioFrame* fake_frame() { return &fake_frame_; }
+
+ private:
+  AudioFrame fake_frame_;
+  int32_t FakeAudioFrame(const int32_t id, AudioFrame* audio_frame) {
+    audio_frame->CopyFrom(fake_frame_);
+    return 0;
+  }
+};
+
+TEST(AudioConferenceMixer, AnonymousAndNamed) {
+  const int kId = 1;
+  // Should not matter even if partipants are more than
+  // kMaximumAmountOfMixedParticipants.
+  const int kNamed =
+      AudioConferenceMixer::kMaximumAmountOfMixedParticipants + 1;
+  const int kAnonymous =
+      AudioConferenceMixer::kMaximumAmountOfMixedParticipants + 1;
+
+  rtc::scoped_ptr<AudioConferenceMixer> mixer(
+      AudioConferenceMixer::Create(kId));
+
+  MockMixerParticipant named[kNamed];
+  MockMixerParticipant anonymous[kAnonymous];
+
+  for (int i = 0; i < kNamed; ++i) {
+    EXPECT_EQ(0, mixer->SetMixabilityStatus(&named[i], true));
+    EXPECT_TRUE(mixer->MixabilityStatus(named[i]));
+  }
+
+  for (int i = 0; i < kAnonymous; ++i) {
+    // Participant must be registered before turning it into anonymous.
+    EXPECT_EQ(-1, mixer->SetAnonymousMixabilityStatus(&anonymous[i], true));
+    EXPECT_EQ(0, mixer->SetMixabilityStatus(&anonymous[i], true));
+    EXPECT_TRUE(mixer->MixabilityStatus(anonymous[i]));
+    EXPECT_FALSE(mixer->AnonymousMixabilityStatus(anonymous[i]));
+
+    EXPECT_EQ(0, mixer->SetAnonymousMixabilityStatus(&anonymous[i], true));
+    EXPECT_TRUE(mixer->AnonymousMixabilityStatus(anonymous[i]));
+
+    // Anonymous participants do not show status by MixabilityStatus.
+    EXPECT_FALSE(mixer->MixabilityStatus(anonymous[i]));
+  }
+
+  for (int i = 0; i < kNamed; ++i) {
+    EXPECT_EQ(0, mixer->SetMixabilityStatus(&named[i], false));
+    EXPECT_FALSE(mixer->MixabilityStatus(named[i]));
+  }
+
+  for (int i = 0; i < kAnonymous - 1; i++) {
+    EXPECT_EQ(0, mixer->SetAnonymousMixabilityStatus(&anonymous[i], false));
+    EXPECT_FALSE(mixer->AnonymousMixabilityStatus(anonymous[i]));
+
+    // SetAnonymousMixabilityStatus(anonymous, false) moves anonymous to the
+    // named group.
+    EXPECT_TRUE(mixer->MixabilityStatus(anonymous[i]));
+  }
+
+  // SetMixabilityStatus(anonymous, false) will remove anonymous from both
+  // anonymous and named groups.
+  EXPECT_EQ(0, mixer->SetMixabilityStatus(&anonymous[kAnonymous - 1], false));
+  EXPECT_FALSE(mixer->AnonymousMixabilityStatus(anonymous[kAnonymous - 1]));
+  EXPECT_FALSE(mixer->MixabilityStatus(anonymous[kAnonymous - 1]));
+}
+
+TEST(AudioConferenceMixer, LargestEnergyVadActiveMixed) {
+  const int kId = 1;
+  const int kParticipants =
+      AudioConferenceMixer::kMaximumAmountOfMixedParticipants + 3;
+  const int kSampleRateHz = 32000;
+
+  rtc::scoped_ptr<AudioConferenceMixer> mixer(
+      AudioConferenceMixer::Create(kId));
+
+  MockAudioMixerOutputReceiver output_receiver;
+  EXPECT_EQ(0, mixer->RegisterMixedStreamCallback(&output_receiver));
+
+  MockMixerParticipant participants[kParticipants];
+
+  for (int i = 0; i < kParticipants; ++i) {
+    participants[i].fake_frame()->id_ = i;
+    participants[i].fake_frame()->sample_rate_hz_ = kSampleRateHz;
+    participants[i].fake_frame()->speech_type_ =  AudioFrame::kNormalSpeech;
+    participants[i].fake_frame()->vad_activity_ = AudioFrame::kVadActive;
+    participants[i].fake_frame()->num_channels_ = 1;
+
+    // Frame duration 10ms.
+    participants[i].fake_frame()->samples_per_channel_ = kSampleRateHz / 100;
+
+    // We set the 80-th sample value since the first 80 samples may be
+    // modified by a ramped-in window.
+    participants[i].fake_frame()->data_[80] = i;
+
+    EXPECT_EQ(0, mixer->SetMixabilityStatus(&participants[i], true));
+    EXPECT_CALL(participants[i], GetAudioFrame(_, _))
+        .Times(AtLeast(1));
+    EXPECT_CALL(participants[i], NeededFrequency(_))
+        .WillRepeatedly(Return(kSampleRateHz));
+  }
+
+  // Last participant gives audio frame with passive VAD, although it has the
+  // largest energy.
+  participants[kParticipants - 1].fake_frame()->vad_activity_ =
+      AudioFrame::kVadPassive;
+
+  EXPECT_CALL(output_receiver, NewMixedAudio(_, _, _, _))
+      .Times(AtLeast(1));
+
+  EXPECT_EQ(0, mixer->Process());
+
+  for (int i = 0; i < kParticipants; ++i) {
+    bool is_mixed = participants[i].IsMixed();
+    if (i == kParticipants - 1 || i < kParticipants - 1 -
+        AudioConferenceMixer::kMaximumAmountOfMixedParticipants) {
+      EXPECT_FALSE(is_mixed) << "Mixing status of Participant #"
+                             << i << " wrong.";
+    } else {
+      EXPECT_TRUE(is_mixed) << "Mixing status of Participant #"
+                            << i << " wrong.";
+    }
+  }
+
+  EXPECT_EQ(0, mixer->UnRegisterMixedStreamCallback());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate b/media/webrtc/trunk/webrtc/modules/audio_decoder_unittests.isolate
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate
rename to media/webrtc/trunk/webrtc/modules/audio_decoder_unittests.isolate
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn b/media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn
index 91d031eede..5897176845 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn
@@ -10,7 +10,7 @@ import("../../build/webrtc.gni")
 
 config("audio_device_config") {
   include_dirs = [
-    "../interface",
+    "../include",
     "include",
     "dummy",  # Contains dummy audio device implementations.
   ]
@@ -18,23 +18,19 @@ config("audio_device_config") {
 
 source_set("audio_device") {
   sources = [
-    "include/audio_device.h",
-    "include/audio_device_defines.h",
     "audio_device_buffer.cc",
     "audio_device_buffer.h",
+    "audio_device_config.h",
     "audio_device_generic.cc",
     "audio_device_generic.h",
-    "audio_device_utility.cc",
-    "audio_device_utility.h",
-    "audio_device_impl.cc",
-    "audio_device_impl.h",
-    "audio_device_config.h",
     "dummy/audio_device_dummy.cc",
     "dummy/audio_device_dummy.h",
-    "dummy/audio_device_utility_dummy.cc",
-    "dummy/audio_device_utility_dummy.h",
     "dummy/file_audio_device.cc",
     "dummy/file_audio_device.h",
+    "fine_audio_buffer.cc",
+    "fine_audio_buffer.h",
+    "include/audio_device.h",
+    "include/audio_device_defines.h",
   ]
 
   include_dirs = []
@@ -53,71 +49,51 @@ source_set("audio_device") {
   if (is_android) {
     include_dirs += [ "android" ]
   }
+  defines = []
+  cflags = []
   if (rtc_include_internal_audio_device) {
+    defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
     sources += [
-      "linux/alsasymboltable_linux.cc",
-      "linux/alsasymboltable_linux.h",
-      "linux/audio_device_alsa_linux.cc",
-      "linux/audio_device_alsa_linux.h",
-      "linux/audio_device_utility_linux.cc",
-      "linux/audio_device_utility_linux.h",
-      "linux/audio_mixer_manager_alsa_linux.cc",
-      "linux/audio_mixer_manager_alsa_linux.h",
-      "linux/latebindingsymboltable_linux.cc",
-      "linux/latebindingsymboltable_linux.h",
-      "ios/audio_device_ios.mm",
-      "ios/audio_device_ios.h",
-      "ios/audio_device_utility_ios.cc",
-      "ios/audio_device_utility_ios.h",
-      "mac/audio_device_mac.cc",
-      "mac/audio_device_mac.h",
-      "mac/audio_device_utility_mac.cc",
-      "mac/audio_device_utility_mac.h",
-      "mac/audio_mixer_manager_mac.cc",
-      "mac/audio_mixer_manager_mac.h",
-      "mac/portaudio/pa_memorybarrier.h",
-      "mac/portaudio/pa_ringbuffer.c",
-      "mac/portaudio/pa_ringbuffer.h",
-      "win/audio_device_core_win.cc",
-      "win/audio_device_core_win.h",
-      "win/audio_device_wave_win.cc",
-      "win/audio_device_wave_win.h",
-      "win/audio_device_utility_win.cc",
-      "win/audio_device_utility_win.h",
-      "win/audio_mixer_manager_win.cc",
-      "win/audio_mixer_manager_win.h",
-      "android/audio_device_template.h",
-      "android/audio_device_utility_android.cc",
-      "android/audio_device_utility_android.h",
-      "android/audio_manager.cc",
-      "android/audio_manager.h",
-      "android/audio_manager_jni.cc",
-      "android/audio_manager_jni.h",
-      "android/audio_record_jni.cc",
-      "android/audio_record_jni.h",
-      "android/audio_track_jni.cc",
-      "android/audio_track_jni.h",
-      "android/fine_audio_buffer.cc",
-      "android/fine_audio_buffer.h",
-      "android/low_latency_event_posix.cc",
-      "android/low_latency_event.h",
-      "android/opensles_common.cc",
-      "android/opensles_common.h",
-      "android/opensles_input.cc",
-      "android/opensles_input.h",
-      "android/opensles_output.cc",
-      "android/opensles_output.h",
-      "android/single_rw_fifo.cc",
-      "android/single_rw_fifo.h",
+      "audio_device_impl.cc",
+      "audio_device_impl.h",
     ]
+    if (is_android) {
+      sources += [
+        "android/audio_device_template.h",
+        "android/audio_manager.cc",
+        "android/audio_manager.h",
+        "android/audio_record_jni.cc",
+        "android/audio_record_jni.h",
+        "android/audio_track_jni.cc",
+        "android/audio_track_jni.h",
+        "android/build_info.cc",
+        "android/build_info.h",
+        "android/opensles_common.cc",
+        "android/opensles_common.h",
+        "android/opensles_player.cc",
+        "android/opensles_player.h",
+      ]
+      libs = [
+        "log",
+        "OpenSLES",
+      ]
+    }
     if (is_linux) {
+      sources += [
+        "linux/alsasymboltable_linux.cc",
+        "linux/alsasymboltable_linux.h",
+        "linux/audio_device_alsa_linux.cc",
+        "linux/audio_device_alsa_linux.h",
+        "linux/audio_mixer_manager_alsa_linux.cc",
+        "linux/audio_mixer_manager_alsa_linux.h",
+        "linux/latebindingsymboltable_linux.cc",
+        "linux/latebindingsymboltable_linux.h",
+      ]
       defines += [ "LINUX_ALSA" ]
-
       libs = [
         "dl",
         "X11",
       ]
-
       if (rtc_include_pulse_audio) {
         sources += [
           "linux/audio_device_pulse_linux.cc",
@@ -127,26 +103,47 @@ source_set("audio_device") {
           "linux/pulseaudiosymboltable_linux.cc",
           "linux/pulseaudiosymboltable_linux.h",
         ]
-
         defines += [ "LINUX_PULSE" ]
       }
     }
     if (is_mac) {
+      sources += [
+        "mac/audio_device_mac.cc",
+        "mac/audio_device_mac.h",
+        "mac/audio_mixer_manager_mac.cc",
+        "mac/audio_mixer_manager_mac.h",
+        "mac/portaudio/pa_memorybarrier.h",
+        "mac/portaudio/pa_ringbuffer.c",
+        "mac/portaudio/pa_ringbuffer.h",
+      ]
       libs = [
         "AudioToolbox.framework",
         "CoreAudio.framework",
       ]
     }
     if (is_ios) {
+      sources += [
+        "ios/audio_device_ios.h",
+        "ios/audio_device_ios.mm",
+        "ios/audio_device_not_implemented_ios.mm",
+      ]
       cflags += [ "-fobjc-arc" ]  # CLANG_ENABLE_OBJC_ARC = YES.
-
       libs = [
         "AudioToolbox.framework",
         "AVFoundation.framework",
         "Foundation.framework",
+        "UIKit.framework",
       ]
     }
     if (is_win) {
+      sources += [
+        "win/audio_device_core_win.cc",
+        "win/audio_device_core_win.h",
+        "win/audio_device_wave_win.cc",
+        "win/audio_device_wave_win.h",
+        "win/audio_mixer_manager_win.cc",
+        "win/audio_mixer_manager_win.h",
+      ]
       libs = [
         # Required for the built-in WASAPI AEC.
         "dmoguids.lib",
@@ -170,7 +167,7 @@ source_set("audio_device") {
   configs += [ "../..:common_config" ]
   public_configs = [
     "../..:common_inherited_config",
-     ":audio_device_config",
+    ":audio_device_config",
   ]
 
   if (is_clang) {
@@ -187,5 +184,3 @@ source_set("audio_device") {
     "../utility",
   ]
 }
-
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_device/OWNERS
index bb11a4ec0e..12d67c035b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/OWNERS
@@ -4,8 +4,6 @@ niklas.enbom@webrtc.org
 tkchin@webrtc.org
 xians@webrtc.org
 
-per-file *.isolate=kjellander@webrtc.org
-
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
 per-file *.gyp=*
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_common.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_common.h
index 447f59587b..4eecae4b70 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_common.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_common.h
@@ -13,23 +13,19 @@
 
 namespace webrtc {
 
-enum {
-  kDefaultSampleRate = 44100,
-  kNumChannels = 1,
-  kDefaultBufSizeInSamples = kDefaultSampleRate * 10 / 1000,
-  // Number of bytes per audio frame.
-  // Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame]
-  kBytesPerFrame = kNumChannels * (16 / 8),
-};
-
-class PlayoutDelayProvider {
- public:
-  virtual int PlayoutDelayMs() = 0;
-
- protected:
-  PlayoutDelayProvider() {}
-  virtual ~PlayoutDelayProvider() {}
-};
+const int kDefaultSampleRate = 44100;
+const int kNumChannels = 1;
+// Number of bytes per audio frame.
+// Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame]
+const size_t kBytesPerFrame = kNumChannels * (16 / 8);
+// Delay estimates for the two different supported modes. These values are based
+// on real-time round-trip delay estimates on a large set of devices and they
+// are lower bounds since the filter length is 128 ms, so the AEC works for
+// delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most
+// cases, the lowest delay estimate will not be utilized since devices that
+// support low-latency output audio often supports HW AEC as well.
+const int kLowLatencyModeDelayEstimateInMilliseconds = 50;
+const int kHighLatencyModeDelayEstimateInMilliseconds = 150;
 
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_template.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_template.h
index f401a65b00..e206191401 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_template.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_template.h
@@ -11,36 +11,41 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
 
+#include <android/log.h>
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+#define TAG "AudioDeviceTemplate"
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
 
 namespace webrtc {
 
 // InputType/OutputType can be any class that implements the capturing/rendering
 // part of the AudioDeviceGeneric API.
+// Construction and destruction must be done on one and the same thread. Each
+// internal implementation of InputType and OutputType will RTC_DCHECK if that
+// is not the case. All implemented methods must also be called on the same
+// thread. See comments in each InputType/OutputType class for more info.
+// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
+// and ClearAndroidAudioDeviceObjects) from a different thread but both will
+// RTC_CHECK that the calling thread is attached to a Java VM.
+
 template <class InputType, class OutputType>
 class AudioDeviceTemplate : public AudioDeviceGeneric {
  public:
-  static void SetAndroidAudioDeviceObjects(void* javaVM,
-                                           void* context) {
-    AudioManager::SetAndroidAudioDeviceObjects(javaVM, context);
-    OutputType::SetAndroidAudioDeviceObjects(javaVM, context);
-    InputType::SetAndroidAudioDeviceObjects(javaVM, context);
-  }
-
-  static void ClearAndroidAudioDeviceObjects() {
-    OutputType::ClearAndroidAudioDeviceObjects();
-    InputType::ClearAndroidAudioDeviceObjects();
-    AudioManager::ClearAndroidAudioDeviceObjects();
-  }
-
-  // TODO(henrika): remove id.
-  explicit AudioDeviceTemplate(const int32_t id)
-      : audio_manager_(),
-        output_(&audio_manager_),
-        input_(&output_, &audio_manager_) {
+  AudioDeviceTemplate(AudioDeviceModule::AudioLayer audio_layer,
+                      AudioManager* audio_manager)
+      : audio_layer_(audio_layer),
+        audio_manager_(audio_manager),
+        output_(audio_manager_),
+        input_(audio_manager_),
+        initialized_(false) {
+    RTC_CHECK(audio_manager);
+    audio_manager_->SetActiveAudioLayer(audio_layer);
   }
 
   virtual ~AudioDeviceTemplate() {
@@ -48,20 +53,41 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
 
   int32_t ActiveAudioLayer(
       AudioDeviceModule::AudioLayer& audioLayer) const override {
-    audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+    audioLayer = audio_layer_;
     return 0;
-  };
+  }
 
   int32_t Init() override {
-    return audio_manager_.Init() | output_.Init() | input_.Init();
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(!initialized_);
+    if (!audio_manager_->Init())
+      return -1;
+    if (output_.Init() != 0) {
+      audio_manager_->Close();
+      return -1;
+    }
+    if (input_.Init() != 0) {
+      output_.Terminate();
+      audio_manager_->Close();
+      return -1;
+    }
+    initialized_ = true;
+    return 0;
   }
 
   int32_t Terminate() override {
-    return output_.Terminate() | input_.Terminate() | audio_manager_.Close();
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    int32_t err = input_.Terminate();
+    err |= output_.Terminate();
+    err |= !audio_manager_->Close();
+    initialized_ = false;
+    RTC_DCHECK_EQ(err, 0);
+    return err;
   }
 
   bool Initialized() const override {
-    return true;
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    return initialized_;
   }
 
   int16_t PlayoutDevices() override {
@@ -138,11 +164,18 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
   }
 
   int32_t StartPlayout() override {
+    if (!audio_manager_->IsCommunicationModeEnabled()) {
+      ALOGW("The application should use MODE_IN_COMMUNICATION audio mode!");
+    }
     return output_.StartPlayout();
   }
 
   int32_t StopPlayout() override {
-    return output_.StopPlayout();
+    // Avoid using audio manger (JNI/Java cost) if playout was inactive.
+    if (!Playing())
+      return 0;
+    int32_t err = output_.StopPlayout();
+    return err;
   }
 
   bool Playing() const override {
@@ -150,11 +183,18 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
   }
 
   int32_t StartRecording() override {
+    if (!audio_manager_->IsCommunicationModeEnabled()) {
+      ALOGW("The application should use MODE_IN_COMMUNICATION audio mode!");
+    }
     return input_.StartRecording();
   }
 
   int32_t StopRecording() override {
-    return input_.StopRecording();
+    // Avoid using audio manger (JNI/Java cost) if recording was inactive.
+    if (!Recording())
+      return 0;
+    int32_t err = input_.StopRecording();
+    return err;
   }
 
   bool Recording() const override {
@@ -344,12 +384,18 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
     return -1;
   }
 
-  int32_t PlayoutDelay(uint16_t& delayMS) const override {
-    return output_.PlayoutDelay(delayMS);
+  int32_t PlayoutDelay(uint16_t& delay_ms) const override {
+    // Best guess we can do is to use half of the estimated total delay.
+    delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
+    RTC_DCHECK_GT(delay_ms, 0);
+    return 0;
   }
 
-  int32_t RecordingDelay(uint16_t& delayMS) const override {
-    return input_.RecordingDelay(delayMS);
+  int32_t RecordingDelay(uint16_t& delay_ms) const override {
+    // Best guess we can do is to use half of the estimated total delay.
+    delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
+    RTC_DCHECK_GT(delay_ms, 0);
+    return 0;
   }
 
   int32_t CPULoad(uint16_t& load) const override {
@@ -402,18 +448,57 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
     return -1;
   }
 
+  // Returns true if the device both supports built in AEC and the device
+  // is not blacklisted.
   bool BuiltInAECIsAvailable() const override {
-    return input_.BuiltInAECIsAvailable();
+    return audio_manager_->IsAcousticEchoCancelerSupported();
   }
 
   int32_t EnableBuiltInAEC(bool enable) override {
+    RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
     return input_.EnableBuiltInAEC(enable);
   }
 
+  // Returns true if the device both supports built in AGC and the device
+  // is not blacklisted.
+  bool BuiltInAGCIsAvailable() const override {
+    return audio_manager_->IsAutomaticGainControlSupported();
+  }
+
+  int32_t EnableBuiltInAGC(bool enable) override {
+    RTC_CHECK(BuiltInAGCIsAvailable()) << "HW AGC is not available";
+    return input_.EnableBuiltInAGC(enable);
+  }
+
+  // Returns true if the device both supports built in NS and the device
+  // is not blacklisted.
+  bool BuiltInNSIsAvailable() const override {
+    return audio_manager_->IsNoiseSuppressorSupported();
+  }
+
+  int32_t EnableBuiltInNS(bool enable) override {
+    RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
+    return input_.EnableBuiltInNS(enable);
+  }
+
  private:
-  AudioManager audio_manager_;
+  rtc::ThreadChecker thread_checker_;
+
+  // Local copy of the audio layer set during construction of the
+  // AudioDeviceModuleImpl instance. Read only value.
+  const AudioDeviceModule::AudioLayer audio_layer_;
+
+  // Non-owning raw pointer to AudioManager instance given to use at
+  // construction. The real object is owned by AudioDeviceModuleImpl and the
+  // life time is the same as that of the AudioDeviceModuleImpl, hence there
+  // is no risk of reading a NULL pointer at any time in this class.
+  AudioManager* const audio_manager_;
+
   OutputType output_;
+
   InputType input_;
+
+  bool initialized_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_unittest.cc
index c79b9256de..768047df51 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -8,20 +8,29 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
+#include <limits>
 #include <list>
 #include <numeric>
+#include <string>
+#include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/audio_device/android/audio_common.h"
+#include "webrtc/modules/audio_device/android/audio_manager.h"
+#include "webrtc/modules/audio_device/android/build_info.h"
 #include "webrtc/modules/audio_device/android/ensure_initialized.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using std::cout;
@@ -45,42 +54,31 @@ using ::testing::TestWithParam;
 
 namespace webrtc {
 
-// Perform all tests for the different audio layers listed in this array.
-// See the INSTANTIATE_TEST_CASE_P statement for details.
-// TODO(henrika): the test framework supports both Java and OpenSL ES based
-// audio backends but there are currently some issues (crashes) in the
-// OpenSL ES implementation, hence it is not added to kAudioLayers yet.
-static const AudioDeviceModule::AudioLayer kAudioLayers[] = {
-    AudioDeviceModule::kAndroidJavaAudio
-    /*, AudioDeviceModule::kAndroidOpenSLESAudio */};
 // Number of callbacks (input or output) the tests waits for before we set
 // an event indicating that the test was OK.
-static const int kNumCallbacks = 10;
+static const size_t kNumCallbacks = 10;
 // Max amount of time we wait for an event to be set while counting callbacks.
 static const int kTestTimeOutInMilliseconds = 10 * 1000;
 // Average number of audio callbacks per second assuming 10ms packet size.
-static const int kNumCallbacksPerSecond = 100;
+static const size_t kNumCallbacksPerSecond = 100;
 // Play out a test file during this time (unit is in seconds).
 static const int kFilePlayTimeInSec = 5;
-// Fixed value for the recording delay using Java based audio backend.
-// TODO(henrika): harmonize with OpenSL ES and look for possible improvements.
-static const uint32_t kFixedRecordingDelay = 100;
-static const int kBitsPerSample = 16;
-static const int kBytesPerSample = kBitsPerSample / 8;
+static const size_t kBitsPerSample = 16;
+static const size_t kBytesPerSample = kBitsPerSample / 8;
 // Run the full-duplex test during this time (unit is in seconds).
 // Note that first |kNumIgnoreFirstCallbacks| are ignored.
 static const int kFullDuplexTimeInSec = 5;
 // Wait for the callback sequence to stabilize by ignoring this amount of the
 // initial callbacks (avoids initial FIFO access).
 // Only used in the RunPlayoutAndRecordingInFullDuplex test.
-static const int kNumIgnoreFirstCallbacks = 50;
+static const size_t kNumIgnoreFirstCallbacks = 50;
 // Sets the number of impulses per second in the latency test.
 static const int kImpulseFrequencyInHz = 1;
 // Length of round-trip latency measurements. Number of transmitted impulses
 // is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
 static const int kMeasureLatencyTimeInSec = 11;
 // Utilized in round-trip latency measurements to avoid capturing noise samples.
-static const int kImpulseThreshold = 500;
+static const int kImpulseThreshold = 1000;
 static const char kTag[] = "[..........] ";
 
 enum TransportType {
@@ -88,27 +86,13 @@ enum TransportType {
   kRecording = 0x2,
 };
 
-// Simple helper struct for device specific audio parameters.
-struct AudioParameters {
-  int playout_frames_per_buffer() const {
-    return playout_sample_rate / 100;  // WebRTC uses 10 ms as buffer size.
-  }
-  int recording_frames_per_buffer() const {
-    return recording_sample_rate / 100;
-  }
-  int playout_sample_rate;
-  int recording_sample_rate;
-  int playout_channels;
-  int recording_channels;
-};
-
 // Interface for processing the audio stream. Real implementations can e.g.
 // run audio in loopback, read audio from a file or perform latency
 // measurements.
 class AudioStreamInterface {
  public:
-  virtual void Write(const void* source, int num_frames) = 0;
-  virtual void Read(void* destination, int num_frames) = 0;
+  virtual void Write(const void* source, size_t num_frames) = 0;
+  virtual void Read(void* destination, size_t num_frames) = 0;
  protected:
   virtual ~AudioStreamInterface() {}
 };
@@ -118,7 +102,7 @@ class AudioStreamInterface {
 class FileAudioStream : public AudioStreamInterface {
  public:
   FileAudioStream(
-      int num_callbacks, const std::string& file_name, int sample_rate)
+      size_t num_callbacks, const std::string& file_name, int sample_rate)
       : file_size_in_bytes_(0),
         sample_rate_(sample_rate),
         file_pos_(0) {
@@ -126,23 +110,23 @@ class FileAudioStream : public AudioStreamInterface {
     sample_rate_ = sample_rate;
     EXPECT_GE(file_size_in_callbacks(), num_callbacks)
         << "Size of test file is not large enough to last during the test.";
-    const int num_16bit_samples =
+    const size_t num_16bit_samples =
         test::GetFileSize(file_name) / kBytesPerSample;
     file_.reset(new int16_t[num_16bit_samples]);
     FILE* audio_file = fopen(file_name.c_str(), "rb");
     EXPECT_NE(audio_file, nullptr);
-    int num_samples_read = fread(
+    size_t num_samples_read = fread(
         file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
     EXPECT_EQ(num_samples_read, num_16bit_samples);
     fclose(audio_file);
   }
 
   // AudioStreamInterface::Write() is not implemented.
-  virtual void Write(const void* source, int num_frames) override {}
+  void Write(const void* source, size_t num_frames) override {}
 
   // Read samples from file stored in memory (at construction) and copy
   // |num_frames| (<=> 10ms) to the |destination| byte buffer.
-  virtual void Read(void* destination, int num_frames) override {
+  void Read(void* destination, size_t num_frames) override {
     memcpy(destination,
            static_cast<int16_t*> (&file_[file_pos_]),
            num_frames * sizeof(int16_t));
@@ -150,17 +134,18 @@ class FileAudioStream : public AudioStreamInterface {
   }
 
   int file_size_in_seconds() const {
-    return (file_size_in_bytes_ / (kBytesPerSample * sample_rate_));
+    return static_cast<int>(
+        file_size_in_bytes_ / (kBytesPerSample * sample_rate_));
   }
-  int file_size_in_callbacks() const {
+  size_t file_size_in_callbacks() const {
     return file_size_in_seconds() * kNumCallbacksPerSecond;
   }
 
  private:
-  int file_size_in_bytes_;
+  size_t file_size_in_bytes_;
   int sample_rate_;
   rtc::scoped_ptr<int16_t[]> file_;
-  int file_pos_;
+  size_t file_pos_;
 };
 
 // Simple first in first out (FIFO) class that wraps a list of 16-bit audio
@@ -173,7 +158,7 @@ class FileAudioStream : public AudioStreamInterface {
 // since both sides (playout and recording) are driven by its own thread.
 class FifoAudioStream : public AudioStreamInterface {
  public:
-  explicit FifoAudioStream(int frames_per_buffer)
+  explicit FifoAudioStream(size_t frames_per_buffer)
       : frames_per_buffer_(frames_per_buffer),
         bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
         fifo_(new AudioBufferList),
@@ -185,13 +170,12 @@ class FifoAudioStream : public AudioStreamInterface {
 
   ~FifoAudioStream() {
     Flush();
-    PRINTD("[%4.3f]\n", average_size());
   }
 
   // Allocate new memory, copy |num_frames| samples from |source| into memory
   // and add pointer to the memory location to end of the list.
   // Increases the size of the FIFO by one element.
-  virtual void Write(const void* source, int num_frames) override {
+  void Write(const void* source, size_t num_frames) override {
     ASSERT_EQ(num_frames, frames_per_buffer_);
     PRINTD("+");
     if (write_count_++ < kNumIgnoreFirstCallbacks) {
@@ -203,10 +187,10 @@ class FifoAudioStream : public AudioStreamInterface {
            bytes_per_buffer_);
     rtc::CritScope lock(&lock_);
     fifo_->push_back(memory);
-    const int size = fifo_->size();
+    const size_t size = fifo_->size();
     if (size > largest_size_) {
       largest_size_ = size;
-      PRINTD("(%d)", largest_size_);
+      PRINTD("(%" PRIuS ")", largest_size_);
     }
     total_written_elements_ += size;
   }
@@ -214,7 +198,7 @@ class FifoAudioStream : public AudioStreamInterface {
   // Read pointer to data buffer from front of list, copy |num_frames| of stored
   // data into |destination| and delete the utilized memory allocation.
   // Decreases the size of the FIFO by one element.
-  virtual void Read(void* destination, int num_frames) override {
+  void Read(void* destination, size_t num_frames) override {
     ASSERT_EQ(num_frames, frames_per_buffer_);
     PRINTD("-");
     rtc::CritScope lock(&lock_);
@@ -230,15 +214,15 @@ class FifoAudioStream : public AudioStreamInterface {
     }
   }
 
-  int size() const {
+  size_t size() const {
     return fifo_->size();
   }
 
-  int largest_size() const {
+  size_t largest_size() const {
     return largest_size_;
   }
 
-  int average_size() const {
+  size_t average_size() const {
     return (total_written_elements_ == 0) ? 0.0 : 0.5 + static_cast<float> (
       total_written_elements_) / (write_count_ - kNumIgnoreFirstCallbacks);
   }
@@ -253,12 +237,12 @@ class FifoAudioStream : public AudioStreamInterface {
 
   using AudioBufferList = std::list<int16_t*>;
   rtc::CriticalSection lock_;
-  const int frames_per_buffer_;
-  const int bytes_per_buffer_;
+  const size_t frames_per_buffer_;
+  const size_t bytes_per_buffer_;
   rtc::scoped_ptr<AudioBufferList> fifo_;
-  int largest_size_;
-  int total_written_elements_;
-  int write_count_;
+  size_t largest_size_;
+  size_t total_written_elements_;
+  size_t write_count_;
 };
 
 // Inserts periodic impulses and measures the latency between the time of
@@ -267,7 +251,7 @@ class FifoAudioStream : public AudioStreamInterface {
 // See http://source.android.com/devices/audio/loopback.html for details.
 class LatencyMeasuringAudioStream : public AudioStreamInterface {
  public:
-  explicit LatencyMeasuringAudioStream(int frames_per_buffer)
+  explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
       : clock_(Clock::GetRealTimeClock()),
         frames_per_buffer_(frames_per_buffer),
         bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
@@ -277,7 +261,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
   }
 
   // Insert periodic impulses in first two samples of |destination|.
-  virtual void Read(void* destination, int num_frames) override {
+  void Read(void* destination, size_t num_frames) override {
     ASSERT_EQ(num_frames, frames_per_buffer_);
     if (play_count_ == 0) {
       PRINT("[");
@@ -291,15 +275,15 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
       PRINT(".");
       const int16_t impulse = std::numeric_limits<int16_t>::max();
       int16_t* ptr16 = static_cast<int16_t*> (destination);
-      for (int i = 0; i < 2; ++i) {
-        *ptr16++ = impulse;
+      for (size_t i = 0; i < 2; ++i) {
+        ptr16[i] = impulse;
       }
     }
   }
 
   // Detect received impulses in |source|, derive time between transmission and
   // detection and add the calculated delay to list of latencies.
-  virtual void Write(const void* source, int num_frames) override {
+  void Write(const void* source, size_t num_frames) override {
     ASSERT_EQ(num_frames, frames_per_buffer_);
     rec_count_++;
     if (pulse_time_ == 0) {
@@ -333,7 +317,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
     }
   }
 
-  int num_latency_values() const {
+  size_t num_latency_values() const {
     return latencies_.size();
   }
 
@@ -368,15 +352,15 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
   }
 
   int IndexToMilliseconds(double index) const {
-    return 10.0 * (index / frames_per_buffer_) + 0.5;
+    return static_cast<int>(10.0 * (index / frames_per_buffer_) + 0.5);
   }
 
  private:
   Clock* clock_;
-  const int frames_per_buffer_;
-  const int bytes_per_buffer_;
-  int play_count_;
-  int rec_count_;
+  const size_t frames_per_buffer_;
+  const size_t bytes_per_buffer_;
+  size_t play_count_;
+  size_t rec_count_;
   int64_t pulse_time_;
   std::vector<int> latencies_;
 };
@@ -397,9 +381,9 @@ class MockAudioTransport : public AudioTransport {
 
   MOCK_METHOD10(RecordedDataIsAvailable,
                 int32_t(const void* audioSamples,
-                        const uint32_t nSamples,
-                        const uint8_t nBytesPerSample,
-                        const uint8_t nChannels,
+                        const size_t nSamples,
+                        const size_t nBytesPerSample,
+                        const size_t nChannels,
                         const uint32_t samplesPerSec,
                         const uint32_t totalDelayMS,
                         const int32_t clockDrift,
@@ -407,12 +391,12 @@ class MockAudioTransport : public AudioTransport {
                         const bool keyPressed,
                         uint32_t& newMicLevel));
   MOCK_METHOD8(NeedMorePlayData,
-               int32_t(const uint32_t nSamples,
-                       const uint8_t nBytesPerSample,
-                       const uint8_t nChannels,
+               int32_t(const size_t nSamples,
+                       const size_t nBytesPerSample,
+                       const size_t nChannels,
                        const uint32_t samplesPerSec,
                        void* audioSamples,
-                       uint32_t& nSamplesOut,
+                       size_t& nSamplesOut,
                        int64_t* elapsed_time_ms,
                        int64_t* ntp_time_ms));
 
@@ -437,9 +421,9 @@ class MockAudioTransport : public AudioTransport {
   }
 
   int32_t RealRecordedDataIsAvailable(const void* audioSamples,
-                                      const uint32_t nSamples,
-                                      const uint8_t nBytesPerSample,
-                                      const uint8_t nChannels,
+                                      const size_t nSamples,
+                                      const size_t nBytesPerSample,
+                                      const size_t nChannels,
                                       const uint32_t samplesPerSec,
                                       const uint32_t totalDelayMS,
                                       const int32_t clockDrift,
@@ -459,12 +443,12 @@ class MockAudioTransport : public AudioTransport {
     return 0;
   }
 
-  int32_t RealNeedMorePlayData(const uint32_t nSamples,
-                               const uint8_t nBytesPerSample,
-                               const uint8_t nChannels,
+  int32_t RealNeedMorePlayData(const size_t nSamples,
+                               const size_t nBytesPerSample,
+                               const size_t nChannels,
                                const uint32_t samplesPerSec,
                                void* audioSamples,
-                               uint32_t& nSamplesOut,
+                               size_t& nSamplesOut,
                                int64_t* elapsed_time_ms,
                                int64_t* ntp_time_ms) {
     EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
@@ -502,17 +486,16 @@ class MockAudioTransport : public AudioTransport {
 
  private:
   EventWrapper* test_is_done_;
-  int num_callbacks_;
+  size_t num_callbacks_;
   int type_;
-  int play_count_;
-  int rec_count_;
+  size_t play_count_;
+  size_t rec_count_;
   AudioStreamInterface* audio_stream_;
   rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
 };
 
-// AudioDeviceTest is a value-parameterized test.
-class AudioDeviceTest
-    : public testing::TestWithParam<AudioDeviceModule::AudioLayer> {
+// AudioDeviceTest test fixture.
+class AudioDeviceTest : public ::testing::Test {
  protected:
   AudioDeviceTest()
       : test_is_done_(EventWrapper::Create()) {
@@ -520,54 +503,67 @@ class AudioDeviceTest
     // can do calls between C++ and Java. Initializes both Java and OpenSL ES
     // implementations.
     webrtc::audiodevicemodule::EnsureInitialized();
-    // Creates an audio device based on the test parameter. See
-    // INSTANTIATE_TEST_CASE_P() for details.
-    audio_device_ = CreateAudioDevice();
+    // Creates an audio device using a default audio layer.
+    audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
     EXPECT_NE(audio_device_.get(), nullptr);
     EXPECT_EQ(0, audio_device_->Init());
-    CacheAudioParameters();
+    playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
+    record_parameters_ = audio_manager()->GetRecordAudioParameters();
+    build_info_.reset(new BuildInfo());
   }
   virtual ~AudioDeviceTest() {
     EXPECT_EQ(0, audio_device_->Terminate());
   }
 
   int playout_sample_rate() const {
-    return parameters_.playout_sample_rate;
+    return playout_parameters_.sample_rate();
   }
-  int recording_sample_rate() const {
-    return parameters_.recording_sample_rate;
+  int record_sample_rate() const {
+    return record_parameters_.sample_rate();
   }
-  int playout_channels() const {
-    return parameters_.playout_channels;
+  size_t playout_channels() const {
+    return playout_parameters_.channels();
   }
-  int recording_channels() const {
-    return parameters_.playout_channels;
+  size_t record_channels() const {
+    return record_parameters_.channels();
   }
-  int playout_frames_per_buffer() const {
-    return parameters_.playout_frames_per_buffer();
+  size_t playout_frames_per_10ms_buffer() const {
+    return playout_parameters_.frames_per_10ms_buffer();
   }
-  int recording_frames_per_buffer() const {
-    return parameters_.recording_frames_per_buffer();
+  size_t record_frames_per_10ms_buffer() const {
+    return record_parameters_.frames_per_10ms_buffer();
   }
 
-  scoped_refptr<AudioDeviceModule> audio_device() const {
+  int total_delay_ms() const {
+    return audio_manager()->GetDelayEstimateInMilliseconds();
+  }
+
+  rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
     return audio_device_;
   }
 
-  scoped_refptr<AudioDeviceModule> CreateAudioDevice() {
-    scoped_refptr<AudioDeviceModule> module(
-        AudioDeviceModuleImpl::Create(0, GetParam()));
-    return module;
+  AudioDeviceModuleImpl* audio_device_impl() const {
+    return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
   }
 
-  void CacheAudioParameters() {
-    AudioDeviceBuffer* audio_buffer =
-        static_cast<AudioDeviceModuleImpl*> (
-            audio_device_.get())->GetAudioDeviceBuffer();
-    parameters_.playout_sample_rate = audio_buffer->PlayoutSampleRate();
-    parameters_.recording_sample_rate = audio_buffer->RecordingSampleRate();
-    parameters_.playout_channels = audio_buffer->PlayoutChannels();
-    parameters_.recording_channels = audio_buffer->RecordingChannels();
+  AudioManager* audio_manager() const {
+    return audio_device_impl()->GetAndroidAudioManagerForTest();
+  }
+
+  AudioManager* GetAudioManager(AudioDeviceModule* adm) const {
+    return static_cast<AudioDeviceModuleImpl*>(adm)->
+        GetAndroidAudioManagerForTest();
+  }
+
+  AudioDeviceBuffer* audio_device_buffer() const {
+    return audio_device_impl()->GetAudioDeviceBuffer();
+  }
+
+  rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
+      AudioDeviceModule::AudioLayer audio_layer) {
+    rtc::scoped_refptr<AudioDeviceModule> module(
+        AudioDeviceModuleImpl::Create(0, audio_layer));
+    return module;
   }
 
   // Returns file name relative to the resource root given a sample rate.
@@ -582,22 +578,69 @@ class AudioDeviceTest
     EXPECT_TRUE(test::FileExists(file_name));
 #ifdef ENABLE_PRINTF
     PRINT("file name: %s\n", file_name.c_str());
-    const int bytes = test::GetFileSize(file_name);
-    PRINT("file size: %d [bytes]\n", bytes);
-    PRINT("file size: %d [samples]\n", bytes / kBytesPerSample);
-    const int seconds = bytes / (sample_rate * kBytesPerSample);
+    const size_t bytes = test::GetFileSize(file_name);
+    PRINT("file size: %" PRIuS " [bytes]\n", bytes);
+    PRINT("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
+    const int seconds =
+        static_cast<int>(bytes / (sample_rate * kBytesPerSample));
     PRINT("file size: %d [secs]\n", seconds);
-    PRINT("file size: %d [callbacks]\n", seconds * kNumCallbacksPerSecond);
+    PRINT("file size: %" PRIuS " [callbacks]\n",
+          seconds * kNumCallbacksPerSecond);
 #endif
     return file_name;
   }
 
+  AudioDeviceModule::AudioLayer GetActiveAudioLayer() const {
+    AudioDeviceModule::AudioLayer audio_layer;
+    EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
+    return audio_layer;
+  }
+
+  int TestDelayOnAudioLayer(
+      const AudioDeviceModule::AudioLayer& layer_to_test) {
+    rtc::scoped_refptr<AudioDeviceModule> audio_device;
+    audio_device = CreateAudioDevice(layer_to_test);
+    EXPECT_NE(audio_device.get(), nullptr);
+    AudioManager* audio_manager = GetAudioManager(audio_device.get());
+    EXPECT_NE(audio_manager, nullptr);
+    return audio_manager->GetDelayEstimateInMilliseconds();
+  }
+
+  AudioDeviceModule::AudioLayer TestActiveAudioLayer(
+      const AudioDeviceModule::AudioLayer& layer_to_test) {
+    rtc::scoped_refptr<AudioDeviceModule> audio_device;
+    audio_device = CreateAudioDevice(layer_to_test);
+    EXPECT_NE(audio_device.get(), nullptr);
+    AudioDeviceModule::AudioLayer active;
+    EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active));
+    return active;
+  }
+
+  bool DisableTestForThisDevice(const std::string& model) {
+    return (build_info_->GetDeviceModel() == model);
+  }
+
+  // Volume control is currently only supported for the Java output audio layer.
+  // For OpenSL ES, the internal stream volume is always on max level and there
+  // is no need for this test to set it to max.
+  bool AudioLayerSupportsVolumeControl() const {
+    return GetActiveAudioLayer() == AudioDeviceModule::kAndroidJavaAudio;
+  }
+
   void SetMaxPlayoutVolume() {
+    if (!AudioLayerSupportsVolumeControl())
+      return;
     uint32_t max_volume;
     EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
     EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
   }
 
+  void DisableBuiltInAECIfAvailable() {
+    if (audio_device()->BuiltInAECIsAvailable()) {
+      EXPECT_EQ(0, audio_device()->EnableBuiltInAEC(false));
+    }
+  }
+
   void StartPlayout() {
     EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
     EXPECT_FALSE(audio_device()->Playing());
@@ -610,6 +653,7 @@ class AudioDeviceTest
   void StopPlayout() {
     EXPECT_EQ(0, audio_device()->StopPlayout());
     EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
   }
 
   void StartRecording() {
@@ -645,65 +689,131 @@ class AudioDeviceTest
   }
 
   rtc::scoped_ptr<EventWrapper> test_is_done_;
-  scoped_refptr<AudioDeviceModule> audio_device_;
-  AudioParameters parameters_;
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+  rtc::scoped_ptr<BuildInfo> build_info_;
 };
 
-TEST_P(AudioDeviceTest, ConstructDestruct) {
+TEST_F(AudioDeviceTest, ConstructDestruct) {
   // Using the test fixture to create and destruct the audio device module.
 }
 
-// Create an audio device instance and print out the native audio parameters.
-TEST_P(AudioDeviceTest, AudioParameters) {
-  EXPECT_NE(0, playout_sample_rate());
-  PRINT("%splayout_sample_rate: %d\n", kTag, playout_sample_rate());
-  EXPECT_NE(0, recording_sample_rate());
-  PRINT("%srecording_sample_rate: %d\n", kTag, recording_sample_rate());
-  EXPECT_NE(0, playout_channels());
-  PRINT("%splayout_channels: %d\n", kTag, playout_channels());
-  EXPECT_NE(0, recording_channels());
-  PRINT("%srecording_channels: %d\n", kTag, recording_channels());
+// We always ask for a default audio layer when the ADM is constructed. But the
+// ADM will then internally set the best suitable combination of audio layers,
+// for input and output based on if low-latency output audio in combination
+// with OpenSL ES is supported or not. This test ensures that the correct
+// selection is done.
+TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) {
+  const AudioDeviceModule::AudioLayer audio_layer = GetActiveAudioLayer();
+  bool low_latency_output = audio_manager()->IsLowLatencyPlayoutSupported();
+  AudioDeviceModule::AudioLayer expected_audio_layer = low_latency_output ?
+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio :
+      AudioDeviceModule::kAndroidJavaAudio;
+  EXPECT_EQ(expected_audio_layer, audio_layer);
 }
 
-TEST_P(AudioDeviceTest, InitTerminate) {
+// Verify that it is possible to explicitly create the two types of supported
+// ADMs. These two tests overrides the default selection of native audio layer
+// by ignoring if the device supports low-latency output or not.
+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
+  AudioDeviceModule::AudioLayer active_layer = TestActiveAudioLayer(
+      expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidJavaAudio;
+  AudioDeviceModule::AudioLayer active_layer = TestActiveAudioLayer(
+      expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+// The Android ADM supports two different delay reporting modes. One for the
+// low-latency output path (in combination with OpenSL ES), and one for the
+// high-latency output path (Java backends in both directions). These two tests
+// verifies that the audio manager reports correct delay estimate given the
+// selected audio layer. Note that, this delay estimate will only be utilized
+// if the HW AEC is disabled.
+TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) {
+  EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds,
+            TestDelayOnAudioLayer(AudioDeviceModule::kAndroidJavaAudio));
+}
+
+TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
+  EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds,
+            TestDelayOnAudioLayer(
+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
+}
+
+// Ensure that the ADM internal audio device buffer is configured to use the
+// correct set of parameters.
+TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) {
+  EXPECT_EQ(playout_parameters_.sample_rate(),
+            audio_device_buffer()->PlayoutSampleRate());
+  EXPECT_EQ(record_parameters_.sample_rate(),
+            audio_device_buffer()->RecordingSampleRate());
+  EXPECT_EQ(playout_parameters_.channels(),
+            audio_device_buffer()->PlayoutChannels());
+  EXPECT_EQ(record_parameters_.channels(),
+            audio_device_buffer()->RecordingChannels());
+}
+
+
+TEST_F(AudioDeviceTest, InitTerminate) {
   // Initialization is part of the test fixture.
   EXPECT_TRUE(audio_device()->Initialized());
   EXPECT_EQ(0, audio_device()->Terminate());
   EXPECT_FALSE(audio_device()->Initialized());
 }
 
-TEST_P(AudioDeviceTest, Devices) {
+TEST_F(AudioDeviceTest, Devices) {
   // Device enumeration is not supported. Verify fixed values only.
   EXPECT_EQ(1, audio_device()->PlayoutDevices());
   EXPECT_EQ(1, audio_device()->RecordingDevices());
 }
 
-TEST_P(AudioDeviceTest, BuiltInAECIsAvailable) {
-  PRINT("%sBuiltInAECIsAvailable: %s\n",
-      kTag, audio_device()->BuiltInAECIsAvailable() ? "true" : "false");
-}
-
-TEST_P(AudioDeviceTest, SpeakerVolumeShouldBeAvailable) {
+TEST_F(AudioDeviceTest, SpeakerVolumeShouldBeAvailable) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
   bool available;
   EXPECT_EQ(0, audio_device()->SpeakerVolumeIsAvailable(&available));
   EXPECT_TRUE(available);
 }
 
-TEST_P(AudioDeviceTest, MaxSpeakerVolumeIsPositive) {
+TEST_F(AudioDeviceTest, MaxSpeakerVolumeIsPositive) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
+  StartPlayout();
   EXPECT_GT(GetMaxSpeakerVolume(), 0);
+  StopPlayout();
 }
 
-TEST_P(AudioDeviceTest, MinSpeakerVolumeIsZero) {
+TEST_F(AudioDeviceTest, MinSpeakerVolumeIsZero) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
   EXPECT_EQ(GetMinSpeakerVolume(), 0);
 }
 
-TEST_P(AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) {
+TEST_F(AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
   const int default_volume = GetSpeakerVolume();
   EXPECT_GE(default_volume, GetMinSpeakerVolume());
   EXPECT_LE(default_volume, GetMaxSpeakerVolume());
 }
 
-TEST_P(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) {
+TEST_F(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
   const int default_volume = GetSpeakerVolume();
   const int max_volume = GetMaxSpeakerVolume();
   EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
@@ -712,18 +822,43 @@ TEST_P(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) {
   EXPECT_EQ(0, audio_device()->SetSpeakerVolume(default_volume));
 }
 
-// Tests that playout can be initiated, started and stopped.
-TEST_P(AudioDeviceTest, StartStopPlayout) {
+// Tests that playout can be initiated, started and stopped. No audio callback
+// is registered in this test.
+// Flaky on our trybots makes this test unusable.
+// https://code.google.com/p/webrtc/issues/detail?id=5046
+TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) {
+  StartPlayout();
+  StopPlayout();
   StartPlayout();
   StopPlayout();
 }
 
+// Tests that recording can be initiated, started and stopped. No audio callback
+// is registered in this test.
+TEST_F(AudioDeviceTest, StartStopRecording) {
+  StartRecording();
+  StopRecording();
+  StartRecording();
+  StopRecording();
+}
+
+// Verify that calling StopPlayout() will leave us in an uninitialized state
+// which will require a new call to InitPlayout(). This test does not call
+// StartPlayout() while being uninitialized since doing so will hit a
+// RTC_DCHECK.
+TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_EQ(0, audio_device()->StartPlayout());
+  EXPECT_EQ(0, audio_device()->StopPlayout());
+  EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+}
+
 // Start playout and verify that the native audio layer starts asking for real
 // audio samples to play out using the NeedMorePlayData callback.
-TEST_P(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
+TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
   MockAudioTransport mock(kPlayout);
   mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
-  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_buffer(),
+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
                                      kBytesPerSample,
                                      playout_channels(),
                                      playout_sample_rate(),
@@ -738,15 +873,15 @@ TEST_P(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
 
 // Start recording and verify that the native audio layer starts feeding real
 // audio samples via the RecordedDataIsAvailable callback.
-TEST_P(AudioDeviceTest, StartRecordingVerifyCallbacks) {
+TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
   MockAudioTransport mock(kRecording);
   mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
   EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(),
-                                            recording_frames_per_buffer(),
+                                            record_frames_per_10ms_buffer(),
                                             kBytesPerSample,
-                                            recording_channels(),
-                                            recording_sample_rate(),
-                                            kFixedRecordingDelay,
+                                            record_channels(),
+                                            record_sample_rate(),
+                                            total_delay_ms(),
                                             0,
                                             0,
                                             false,
@@ -762,10 +897,10 @@ TEST_P(AudioDeviceTest, StartRecordingVerifyCallbacks) {
 
 // Start playout and recording (full-duplex audio) and verify that audio is
 // active in both directions.
-TEST_P(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
+TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
   MockAudioTransport mock(kPlayout | kRecording);
   mock.HandleCallbacks(test_is_done_.get(), nullptr,  kNumCallbacks);
-  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_buffer(),
+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
                                      kBytesPerSample,
                                      playout_channels(),
                                      playout_sample_rate(),
@@ -773,11 +908,11 @@ TEST_P(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
                                      _, _, _))
       .Times(AtLeast(kNumCallbacks));
   EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(),
-                                            recording_frames_per_buffer(),
+                                            record_frames_per_10ms_buffer(),
                                             kBytesPerSample,
-                                            recording_channels(),
-                                            recording_sample_rate(),
-                                            Gt(kFixedRecordingDelay),
+                                            record_channels(),
+                                            record_sample_rate(),
+                                            total_delay_ms(),
                                             0,
                                             0,
                                             false,
@@ -794,9 +929,9 @@ TEST_P(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
 // Start playout and read audio from an external PCM file when the audio layer
 // asks for data to play out. Real audio is played out in this test but it does
 // not contain any explicit verification that the audio quality is perfect.
-TEST_P(AudioDeviceTest, RunPlayoutWithFileAsSource) {
+TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
   // TODO(henrika): extend test when mono output is supported.
-  EXPECT_EQ(1, playout_channels());
+  EXPECT_EQ(1u, playout_channels());
   NiceMock<MockAudioTransport> mock(kPlayout);
   const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
   std::string file_name = GetFileName(playout_sample_rate());
@@ -805,7 +940,7 @@ TEST_P(AudioDeviceTest, RunPlayoutWithFileAsSource) {
   mock.HandleCallbacks(test_is_done_.get(),
                        file_audio_stream.get(),
                        num_callbacks);
-  SetMaxPlayoutVolume();
+  // SetMaxPlayoutVolume();
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
   StartPlayout();
   test_is_done_->Wait(kTestTimeOutInMilliseconds);
@@ -825,12 +960,12 @@ TEST_P(AudioDeviceTest, RunPlayoutWithFileAsSource) {
 // recording side and decreased by the playout side.
 // TODO(henrika): tune the final test parameters after running tests on several
 // different devices.
-TEST_P(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
-  EXPECT_EQ(recording_channels(), playout_channels());
-  EXPECT_EQ(recording_sample_rate(), playout_sample_rate());
+TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
+  EXPECT_EQ(record_channels(), playout_channels());
+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
   rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream(
-      new FifoAudioStream(playout_frames_per_buffer()));
+      new FifoAudioStream(playout_frames_per_10ms_buffer()));
   mock.HandleCallbacks(test_is_done_.get(),
                        fifo_audio_stream.get(),
                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
@@ -842,8 +977,8 @@ TEST_P(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
                                1000 * kFullDuplexTimeInSec));
   StopPlayout();
   StopRecording();
-  EXPECT_LE(fifo_audio_stream->average_size(), 10);
-  EXPECT_LE(fifo_audio_stream->largest_size(), 20);
+  EXPECT_LE(fifo_audio_stream->average_size(), 10u);
+  EXPECT_LE(fifo_audio_stream->largest_size(), 20u);
 }
 
 // Measures loopback latency and reports the min, max and average values for
@@ -855,17 +990,18 @@ TEST_P(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
 // - Store time differences in a vector and calculate min, max and average.
 // This test requires a special hardware called Audio Loopback Dongle.
 // See http://source.android.com/devices/audio/loopback.html for details.
-TEST_P(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
-  EXPECT_EQ(recording_channels(), playout_channels());
-  EXPECT_EQ(recording_sample_rate(), playout_sample_rate());
+TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
+  EXPECT_EQ(record_channels(), playout_channels());
+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
   rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
-      new LatencyMeasuringAudioStream(playout_frames_per_buffer()));
+      new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
   mock.HandleCallbacks(test_is_done_.get(),
                        latency_audio_stream.get(),
                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
   SetMaxPlayoutVolume();
+  DisableBuiltInAECIfAvailable();
   StartRecording();
   StartPlayout();
   test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds,
@@ -874,11 +1010,9 @@ TEST_P(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
-            kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1);
+            static_cast<size_t>(
+                kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
-INSTANTIATE_TEST_CASE_P(AudioDeviceTest, AudioDeviceTest,
-  ::testing::ValuesIn(kAudioLayers));
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_utility_android.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_utility_android.cc
deleted file mode 100644
index f8c26624cd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_utility_android.cc
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "webrtc/modules/audio_device/android/audio_device_utility_android.h"
-
-namespace webrtc {
-
-AudioDeviceUtilityAndroid::AudioDeviceUtilityAndroid(const int32_t id) {}
-
-AudioDeviceUtilityAndroid::~AudioDeviceUtilityAndroid() {}
-
-int32_t AudioDeviceUtilityAndroid::Init() {
-  return 0;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_utility_android.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_utility_android.h
deleted file mode 100644
index 1c1ce1ca6b..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_utility_android.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- *  Android audio device utility interface
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
-
-#include <jni.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-namespace webrtc {
-
-// TODO(henrika): this utility class is not used but I would like to keep this
-// file for the other helper methods which are unique for Android.
-class AudioDeviceUtilityAndroid: public AudioDeviceUtility {
- public:
-    AudioDeviceUtilityAndroid(const int32_t id);
-    ~AudioDeviceUtilityAndroid();
-
-    virtual int32_t Init();
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.cc
index 5424ba3dee..6f28e77a85 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.cc
@@ -9,14 +9,21 @@
  */
 
 #include "webrtc/modules/audio_device/android/audio_manager.h"
+#if !defined(MOZ_WIDGET_GONK)
 #include "AndroidJNIWrapper.h"
+#endif
+
+#include <utility>
 
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#if !defined(MOZ_WIDGET_GONK)
+#include "webrtc/modules/utility/include/helpers_android.h"
+#endif
 
 #define TAG "AudioManager"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
@@ -27,150 +34,215 @@
 
 namespace webrtc {
 
-static JavaVM* g_jvm = NULL;
-static jobject g_context = NULL;
-static jclass g_audio_manager_class = NULL;
-
-void AudioManager::SetAndroidAudioDeviceObjects(void* jvm, void* context) {
-  ALOGD("SetAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-
-  CHECK(jvm);
-  CHECK(context);
-
-  g_jvm = reinterpret_cast<JavaVM*>(jvm);
-  JNIEnv* jni = GetEnv(g_jvm);
-  CHECK(jni) << "AttachCurrentThread must be called on this tread";
-
-  if (!g_context) {
-    g_context = NewGlobalRef(jni, reinterpret_cast<jobject>(context));
-  }
-
-  if (!g_audio_manager_class) {
-    g_audio_manager_class = jsjni_GetGlobalClassRef(
-                                "org/webrtc/voiceengine/WebRtcAudioManager");
-    DCHECK(g_audio_manager_class);
-  }
-  // Register native methods with the WebRtcAudioManager class. These methods
-  // are declared private native in WebRtcAudioManager.java.
-  JNINativeMethod native_methods[] = {
-      {"nativeCacheAudioParameters", "(IIJ)V",
-       reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
-  jni->RegisterNatives(g_audio_manager_class,
-                       native_methods, arraysize(native_methods));
-  CHECK_EXCEPTION(jni) << "Error during RegisterNatives";
+// AudioManager::JavaAudioManager implementation
+AudioManager::JavaAudioManager::JavaAudioManager(
+    NativeRegistration* native_reg,
+    rtc::scoped_ptr<GlobalRef> audio_manager)
+    : audio_manager_(std::move(audio_manager)),
+      init_(native_reg->GetMethodId("init", "()Z")),
+      dispose_(native_reg->GetMethodId("dispose", "()V")),
+      is_communication_mode_enabled_(
+          native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
+      is_device_blacklisted_for_open_sles_usage_(
+          native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
+                                  "()Z")) {
+  ALOGD("JavaAudioManager::ctor%s", GetThreadInfo().c_str());
 }
 
-void AudioManager::ClearAndroidAudioDeviceObjects() {
-  ALOGD("ClearAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-  JNIEnv* jni = GetEnv(g_jvm);
-  CHECK(jni) << "AttachCurrentThread must be called on this tread";
-  jni->UnregisterNatives(g_audio_manager_class);
-  CHECK_EXCEPTION(jni) << "Error during UnregisterNatives";
-  DeleteGlobalRef(jni, g_audio_manager_class);
-  g_audio_manager_class = NULL;
-  DeleteGlobalRef(jni, g_context);
-  g_context = NULL;
-  g_jvm = NULL;
+AudioManager::JavaAudioManager::~JavaAudioManager() {
+  ALOGD("JavaAudioManager::dtor%s", GetThreadInfo().c_str());
 }
 
+bool AudioManager::JavaAudioManager::Init() {
+  return audio_manager_->CallBooleanMethod(init_);
+}
+
+void AudioManager::JavaAudioManager::Close() {
+  audio_manager_->CallVoidMethod(dispose_);
+}
+
+bool AudioManager::JavaAudioManager::IsCommunicationModeEnabled() {
+  return audio_manager_->CallBooleanMethod(is_communication_mode_enabled_);
+}
+
+bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
+  return audio_manager_->CallBooleanMethod(
+      is_device_blacklisted_for_open_sles_usage_);
+}
+
+// AudioManager implementation
 AudioManager::AudioManager()
-    : initialized_(false) {
-  j_audio_manager_ = NULL;
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
+      initialized_(false),
+      hardware_aec_(false),
+      hardware_agc_(false),
+      hardware_ns_(false),
+      low_latency_playout_(false),
+      delay_estimate_in_milliseconds_(0) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
-  CHECK(HasDeviceObjects());
-  CreateJavaInstance();
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheAudioParameters",
+       "(IIZZZZIIJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioManager",
+      native_methods, arraysize(native_methods));
+  j_audio_manager_.reset(new JavaAudioManager(
+      j_native_registration_.get(),
+      j_native_registration_->NewObject(
+          "<init>", "(Landroid/content/Context;J)V",
+          JVM::GetInstance()->context(), PointerTojlong(this))));
 }
 
 AudioManager::~AudioManager() {
+#if !defined(MOZ_WIDGET_GONK)
   ALOGD("~dtor%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+#endif
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Close();
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jni->DeleteGlobalRef(j_audio_manager_);
-  j_audio_manager_ = NULL;
-  DCHECK(!initialized_);
+}
+
+void AudioManager::SetActiveAudioLayer(
+    AudioDeviceModule::AudioLayer audio_layer) {
+  ALOGD("SetActiveAudioLayer(%d)%s", audio_layer, GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!initialized_);
+  // Store the currenttly utilized audio layer.
+  audio_layer_ = audio_layer;
+  // The delay estimate can take one of two fixed values depending on if the
+  // device supports low-latency output or not. However, it is also possible
+  // that the user explicitly selects the high-latency audio path, hence we use
+  // the selected |audio_layer| here to set the delay estimate.
+  delay_estimate_in_milliseconds_ =
+      (audio_layer == AudioDeviceModule::kAndroidJavaAudio) ?
+      kHighLatencyModeDelayEstimateInMilliseconds :
+      kLowLatencyModeDelayEstimateInMilliseconds;
+  ALOGD("delay_estimate_in_milliseconds: %d", delay_estimate_in_milliseconds_);
 }
 
 bool AudioManager::Init() {
+#if !defined(MOZ_WIDGET_GONK)
   ALOGD("Init%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!initialized_);
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID initID = GetMethodID(jni, g_audio_manager_class, "init", "()Z");
-  jboolean res = jni->CallBooleanMethod(j_audio_manager_, initID);
-  CHECK_EXCEPTION(jni);
-  if (!res) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
+  if (!j_audio_manager_->Init()) {
     ALOGE("init failed!");
     return false;
   }
+#endif
   initialized_ = true;
   return true;
 }
 
 bool AudioManager::Close() {
+#if !defined(MOZ_WIDGET_GONK)
   ALOGD("Close%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (!initialized_)
     return true;
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID disposeID = GetMethodID(
-      jni, g_audio_manager_class, "dispose", "()V");
-  jni->CallVoidMethod(j_audio_manager_, disposeID);
-  CHECK_EXCEPTION(jni);
+  j_audio_manager_->Close();
+#endif
   initialized_ = false;
   return true;
 }
 
-void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env, jobject obj,
-    jint sample_rate, jint channels, jlong nativeAudioManager) {
-  webrtc::AudioManager* this_object =
-      reinterpret_cast<webrtc::AudioManager*> (nativeAudioManager);
-  this_object->OnCacheAudioParameters(env, sample_rate, channels);
+bool AudioManager::IsCommunicationModeEnabled() const {
+  ALOGD("IsCommunicationModeEnabled()");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_manager_->IsCommunicationModeEnabled();
 }
 
-void AudioManager::OnCacheAudioParameters(
-    JNIEnv* env, jint sample_rate, jint channels) {
+bool AudioManager::IsAcousticEchoCancelerSupported() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return hardware_aec_;
+}
+
+bool AudioManager::IsAutomaticGainControlSupported() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return hardware_agc_;
+}
+
+bool AudioManager::IsNoiseSuppressorSupported() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return hardware_ns_;
+}
+
+bool AudioManager::IsLowLatencyPlayoutSupported() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ALOGD("IsLowLatencyPlayoutSupported()");
+  // Some devices are blacklisted for usage of OpenSL ES even if they report
+  // that low-latency playout is supported. See b/21485703 for details.
+  return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage() ?
+      false : low_latency_playout_;
+}
+
+int AudioManager::GetDelayEstimateInMilliseconds() const {
+  return delay_estimate_in_milliseconds_;
+}
+
+#if !defined(MOZ_WIDGET_GONK)
+void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
+                                                jobject obj,
+                                                jint sample_rate,
+                                                jint channels,
+                                                jboolean hardware_aec,
+                                                jboolean hardware_agc,
+                                                jboolean hardware_ns,
+                                                jboolean low_latency_output,
+                                                jint output_buffer_size,
+                                                jint input_buffer_size,
+                                                jlong native_audio_manager) {
+  webrtc::AudioManager* this_object =
+      reinterpret_cast<webrtc::AudioManager*>(native_audio_manager);
+  this_object->OnCacheAudioParameters(
+      env, sample_rate, channels, hardware_aec, hardware_agc, hardware_ns,
+      low_latency_output, output_buffer_size, input_buffer_size);
+}
+
+void AudioManager::OnCacheAudioParameters(JNIEnv* env,
+                                          jint sample_rate,
+                                          jint channels,
+                                          jboolean hardware_aec,
+                                          jboolean hardware_agc,
+                                          jboolean hardware_ns,
+                                          jboolean low_latency_output,
+                                          jint output_buffer_size,
+                                          jint input_buffer_size) {
   ALOGD("OnCacheAudioParameters%s", GetThreadInfo().c_str());
+  ALOGD("hardware_aec: %d", hardware_aec);
+  ALOGD("hardware_agc: %d", hardware_agc);
+  ALOGD("hardware_ns: %d", hardware_ns);
+  ALOGD("low_latency_output: %d", low_latency_output);
   ALOGD("sample_rate: %d", sample_rate);
   ALOGD("channels: %d", channels);
-  DCHECK(thread_checker_.CalledOnValidThread());
-  // TODO(henrika): add support stereo output.
-  playout_parameters_.reset(sample_rate, channels);
-  record_parameters_.reset(sample_rate, channels);
+  ALOGD("output_buffer_size: %d", output_buffer_size);
+  ALOGD("input_buffer_size: %d", input_buffer_size);
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  hardware_aec_ = hardware_aec;
+  hardware_agc_ = hardware_agc;
+  hardware_ns_ = hardware_ns;
+  low_latency_playout_ = low_latency_output;
+  // TODO(henrika): add support for stereo output.
+  playout_parameters_.reset(sample_rate, static_cast<size_t>(channels),
+                            static_cast<size_t>(output_buffer_size));
+  record_parameters_.reset(sample_rate, static_cast<size_t>(channels),
+                           static_cast<size_t>(input_buffer_size));
 }
-
-AudioParameters AudioManager::GetPlayoutAudioParameters() const {
-  CHECK(playout_parameters_.is_valid());
+#endif
+  
+const AudioParameters& AudioManager::GetPlayoutAudioParameters() {
+  RTC_CHECK(playout_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return playout_parameters_;
 }
 
-AudioParameters AudioManager::GetRecordAudioParameters() const {
-  CHECK(record_parameters_.is_valid());
+const AudioParameters& AudioManager::GetRecordAudioParameters() {
+  RTC_CHECK(record_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return record_parameters_;
 }
 
-bool AudioManager::HasDeviceObjects() {
-  return (g_jvm && g_context && g_audio_manager_class);
-}
-
-void AudioManager::CreateJavaInstance() {
-  ALOGD("CreateJavaInstance");
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID constructorID = GetMethodID(
-      jni, g_audio_manager_class, "<init>", "(Landroid/content/Context;J)V");
-  j_audio_manager_ = jni->NewObject(g_audio_manager_class,
-                                    constructorID,
-                                    g_context,
-                                    reinterpret_cast<intptr_t>(this));
-  CHECK_EXCEPTION(jni) << "Error during NewObject";
-  CHECK(j_audio_manager_);
-  j_audio_manager_ = jni->NewGlobalRef(j_audio_manager_);
-  CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
-  CHECK(j_audio_manager_);
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.h
index a6c712e11e..d1722caf01 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager.h
@@ -11,116 +11,158 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
 
+#if !defined(MOZ_WIDGET_GONK)
 #include <jni.h>
+#endif
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
+#include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#if !defined(MOZ_WIDGET_GONK)
+#include "webrtc/modules/utility/include/helpers_android.h"
+#endif
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
-class AudioParameters {
- public:
-  enum { kBitsPerSample = 16 };
-  AudioParameters()
-      : sample_rate_(0),
-        channels_(0),
-        frames_per_buffer_(0),
-        bits_per_sample_(kBitsPerSample) {}
-  AudioParameters(int sample_rate, int channels)
-      : sample_rate_(sample_rate),
-        channels_(channels),
-        frames_per_buffer_(sample_rate / 100),
-        bits_per_sample_(kBitsPerSample) {}
-  void reset(int sample_rate, int channels) {
-    sample_rate_ = sample_rate;
-    channels_ = channels;
-    // WebRTC uses a fixed buffer size equal to 10ms.
-    frames_per_buffer_ = (sample_rate / 100);
-  }
-  int sample_rate() const { return sample_rate_; }
-  int channels() const { return channels_; }
-  int frames_per_buffer() const { return frames_per_buffer_; }
-  bool is_valid() const {
-    return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
-  }
-  int GetBytesPerFrame() const { return channels_ * bits_per_sample_ / 8; }
-  int GetBytesPerBuffer() const {
-    return frames_per_buffer_ * GetBytesPerFrame();
-  }
-
- private:
-  int sample_rate_;
-  int channels_;
-  int frames_per_buffer_;
-  const int bits_per_sample_;
-};
-
 // Implements support for functions in the WebRTC audio stack for Android that
 // relies on the AudioManager in android.media. It also populates an
 // AudioParameter structure with native audio parameters detected at
 // construction. This class does not make any audio-related modifications
 // unless Init() is called. Caching audio parameters makes no changes but only
 // reads data from the Java side.
-// TODO(henrika): expand this class when adding support for low-latency
-// OpenSL ES. Currently, it only contains very basic functionality.
 class AudioManager {
  public:
-  // Use the invocation API to allow the native application to use the JNI
-  // interface pointer to access VM features. |jvm| denotes the Java VM and
-  // |context| corresponds to android.content.Context in Java.
-  // This method also sets a global jclass object, |g_audio_manager_class| for
-  // the "org/webrtc/voiceengine/WebRtcAudioManager"-class.
-  static void SetAndroidAudioDeviceObjects(void* jvm, void* context);
-  // Always call this method after the object has been destructed. It deletes
-  // existing global references and enables garbage collection.
-  static void ClearAndroidAudioDeviceObjects();
+  // Wraps the Java specific parts of the AudioManager into one helper class.
+  // Stores method IDs for all supported methods at construction and then
+  // allows calls like JavaAudioManager::Close() while hiding the Java/JNI
+  // parts that are associated with this call.
+  class JavaAudioManager {
+   public:
+    JavaAudioManager(NativeRegistration* native_registration,
+                     rtc::scoped_ptr<GlobalRef> audio_manager);
+    ~JavaAudioManager();
+
+    bool Init();
+    void Close();
+    bool IsCommunicationModeEnabled();
+    bool IsDeviceBlacklistedForOpenSLESUsage();
+
+   private:
+    rtc::scoped_ptr<GlobalRef> audio_manager_;
+    jmethodID init_;
+    jmethodID dispose_;
+    jmethodID is_communication_mode_enabled_;
+    jmethodID is_device_blacklisted_for_open_sles_usage_;
+  };
 
   AudioManager();
   ~AudioManager();
 
-  // Initializes the audio manager (changes mode to MODE_IN_COMMUNICATION,
-  // request audio focus etc.).
-  // It is possible to use this class without calling Init() if the calling
-  // application prefers to set up the audio environment on its own instead.
+  // Sets the currently active audio layer combination. Must be called before
+  // Init().
+  void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer);
+
+  // Initializes the audio manager and stores the current audio mode.
   bool Init();
   // Revert any setting done by Init().
   bool Close();
 
-  // Native audio parameters stored during construction.
-  AudioParameters GetPlayoutAudioParameters() const;
-  AudioParameters GetRecordAudioParameters() const;
+  // Returns true if current audio mode is AudioManager.MODE_IN_COMMUNICATION.
+  bool IsCommunicationModeEnabled() const;
 
-  bool initialized() const { return initialized_; }
+  // Native audio parameters stored during construction.
+  const AudioParameters& GetPlayoutAudioParameters();
+  const AudioParameters& GetRecordAudioParameters();
+
+  // Returns true if the device supports built-in audio effects for AEC, AGC
+  // and NS. Some devices can also be blacklisted for use in combination with
+  // platform effects and these devices will return false.
+  // Can currently only be used in combination with a Java based audio backend
+  // for the recoring side (i.e. using the android.media.AudioRecord API).
+  bool IsAcousticEchoCancelerSupported() const;
+  bool IsAutomaticGainControlSupported() const;
+  bool IsNoiseSuppressorSupported() const;
+
+  // Returns true if the device supports the low-latency audio paths in
+  // combination with OpenSL ES.
+  bool IsLowLatencyPlayoutSupported() const;
+
+  // Returns the estimated total delay of this device. Unit is in milliseconds.
+  // The vaule is set once at construction and never changes after that.
+  // Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and
+  // webrtc::kHighLatencyModeDelayEstimateInMilliseconds.
+  int GetDelayEstimateInMilliseconds() const;
 
  private:
   // Called from Java side so we can cache the native audio parameters.
   // This method will be called by the WebRtcAudioManager constructor, i.e.
   // on the same thread that this object is created on.
-  static void JNICALL CacheAudioParameters(JNIEnv* env, jobject obj,
-      jint sample_rate, jint channels, jlong nativeAudioManager);
-  void OnCacheAudioParameters(JNIEnv* env, jint sample_rate, jint channels);
-  
-  // Returns true if SetAndroidAudioDeviceObjects() has been called
-  // successfully.
-  bool HasDeviceObjects();
-
-  // Called from the constructor. Defines the |j_audio_manager_| member.
-  void CreateJavaInstance();
+#if !defined(MOZ_WIDGET_GONK)
+  static void JNICALL CacheAudioParameters(JNIEnv* env,
+                                           jobject obj,
+                                           jint sample_rate,
+                                           jint channels,
+                                           jboolean hardware_aec,
+                                           jboolean hardware_agc,
+                                           jboolean hardware_ns,
+                                           jboolean low_latency_output,
+                                           jint output_buffer_size,
+                                           jint input_buffer_size,
+                                           jlong native_audio_manager);
+  void OnCacheAudioParameters(JNIEnv* env,
+                              jint sample_rate,
+                              jint channels,
+                              jboolean hardware_aec,
+                              jboolean hardware_agc,
+                              jboolean hardware_ns,
+                              jboolean low_latency_output,
+                              jint output_buffer_size,
+                              jint input_buffer_size);
+#endif
 
   // Stores thread ID in the constructor.
   // We can then use ThreadChecker::CalledOnValidThread() to ensure that
   // other methods are called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
-  // The Java WebRtcAudioManager instance.
-  jobject j_audio_manager_;
+  // Calls AttachCurrentThread() if this thread is not attached at construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  AttachCurrentThreadIfNeeded attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  rtc::scoped_ptr<NativeRegistration> j_native_registration_;
+
+#if !defined(MOZ_WIDGET_GONK)
+  // Wraps the Java specific parts of the AudioManager.
+  rtc::scoped_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
+#endif
+
+  AudioDeviceModule::AudioLayer audio_layer_;
 
   // Set to true by Init() and false by Close().
   bool initialized_;
 
+  // True if device supports hardware (or built-in) AEC.
+  bool hardware_aec_;
+  // True if device supports hardware (or built-in) AGC.
+  bool hardware_agc_;
+  // True if device supports hardware (or built-in) NS.
+  bool hardware_ns_;
+
+  // True if device supports the low-latency OpenSL ES audio path.
+  bool low_latency_playout_;
+
+  // The delay estimate can take one of two fixed values depending on if the
+  // device supports low-latency output or not.
+  int delay_estimate_in_milliseconds_;
+
   // Contains native parameters (e.g. sample rate, channel configuration).
   // Set at construction in OnCacheAudioParameters() which is called from
   // Java on the same thread as this object is created on.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.cc
deleted file mode 100644
index 83bd6ee0ac..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.cc
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/android/audio_manager_jni.h"
-
-#include <android/log.h>
-#include <assert.h>
-
-#include "AndroidJNIWrapper.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-#define TAG "AudioManagerJni"
-#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
-
-namespace webrtc {
-
-static JavaVM* g_jvm_ = NULL;
-static JNIEnv* g_jni_env_ = NULL;
-static jobject g_context_ = NULL;
-static jclass g_audio_manager_class_ = NULL;
-static jobject g_audio_manager_ = NULL;
-
-AudioManagerJni::AudioManagerJni()
-    : low_latency_supported_(false),
-      native_output_sample_rate_(0),
-      native_buffer_size_(0) {
-  if (!HasDeviceObjects()) {
-    assert(false);
-  }
-  AttachThreadScoped ats(g_jvm_);
-  JNIEnv* env = ats.env();
-  assert(env && "Unsupported JNI version!");
-  CreateInstance(env);
-  // Pre-store device specific values.
-  SetLowLatencySupported(env);
-  SetNativeOutputSampleRate(env);
-  SetNativeFrameSize(env);
-}
-
-void AudioManagerJni::SetAndroidAudioDeviceObjects(void* jvm, void* context) {
-  ALOGD("SetAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-
-  assert(jvm);
-  assert(context);
-
-  // Store global Java VM variables to be accessed by API calls.
-  g_jvm_ = reinterpret_cast<JavaVM*>(jvm);
-  g_jni_env_ = GetEnv(g_jvm_);
-
-  if (!g_context_) {
-    g_context_ = g_jni_env_->NewGlobalRef(reinterpret_cast<jobject>(context));
-  }
-
-  if (!g_audio_manager_class_) {
-    // Create a global reference such that the class object is not recycled by
-    // the garbage collector.
-    g_audio_manager_class_ = jsjni_GetGlobalClassRef(
-                                 "org/webrtc/voiceengine/AudioManagerAndroid");
-    DCHECK(g_audio_manager_class_);
-  }
-}
-
-void AudioManagerJni::ClearAndroidAudioDeviceObjects() {
-  ALOGD("ClearAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-  g_jni_env_->DeleteGlobalRef(g_audio_manager_class_);
-  g_audio_manager_class_ = NULL;
-  g_jni_env_->DeleteGlobalRef(g_context_);
-  g_context_ = NULL;
-  g_jni_env_->DeleteGlobalRef(g_audio_manager_);
-  g_audio_manager_ = NULL;
-  g_jni_env_ = NULL;
-  g_jvm_ = NULL;
-}
-
-void AudioManagerJni::SetLowLatencySupported(JNIEnv* env) {
-  jmethodID id = LookUpMethodId(env, "isAudioLowLatencySupported", "()Z");
-  low_latency_supported_ = env->CallBooleanMethod(g_audio_manager_, id);
-}
-
-void AudioManagerJni::SetNativeOutputSampleRate(JNIEnv* env) {
-  jmethodID id = LookUpMethodId(env, "getNativeOutputSampleRate", "()I");
-  native_output_sample_rate_ = env->CallIntMethod(g_audio_manager_, id);
-}
-
-void AudioManagerJni::SetNativeFrameSize(JNIEnv* env) {
-  jmethodID id = LookUpMethodId(env,
-                                "getAudioLowLatencyOutputFrameSize", "()I");
-  native_buffer_size_ = env->CallIntMethod(g_audio_manager_, id);
-}
-
-bool AudioManagerJni::HasDeviceObjects() {
-  return g_jvm_ && g_jni_env_ && g_context_ && g_audio_manager_class_;
-}
-
-jmethodID AudioManagerJni::LookUpMethodId(JNIEnv* env,
-                                          const char* method_name,
-                                          const char* method_signature) {
-  jmethodID ret_val = env->GetMethodID(g_audio_manager_class_, method_name,
-                                       method_signature);
-  assert(ret_val);
-  return ret_val;
-}
-
-void AudioManagerJni::CreateInstance(JNIEnv* env) {
-  // Get the method ID for the constructor taking Context.
-  jmethodID id = LookUpMethodId(env, "<init>", "(Landroid/content/Context;)V");
-  g_audio_manager_ = env->NewObject(g_audio_manager_class_, id, g_context_);
-  // Create a global reference so that the instance is accessible until no
-  // longer needed.
-  g_audio_manager_ = env->NewGlobalRef(g_audio_manager_);
-  assert(g_audio_manager_);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.h
deleted file mode 100644
index 5df2490ead..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Android APIs used to access Java functionality needed to enable low latency
-// audio.
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_JNI_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_JNI_H_
-
-#include <jni.h>
-
-namespace webrtc {
-
-class AudioManagerJni {
- public:
-  AudioManagerJni();
-  ~AudioManagerJni() {}
-
-  // SetAndroidAudioDeviceObjects must only be called once unless there has
-  // been a successive call to ClearAndroidAudioDeviceObjects. For each
-  // call to ClearAndroidAudioDeviceObjects, SetAndroidAudioDeviceObjects may be
-  // called once.
-  // This function must be called by a Java thread as calling it from a thread
-  // created by the native application will prevent FindClass from working. See
-  // http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
-  // for more details.
-  // It has to be called for this class' APIs to be successful. Calling
-  // ClearAndroidAudioDeviceObjects will prevent this class' APIs to be called
-  // successfully if SetAndroidAudioDeviceObjects is not called after it.
-  static void SetAndroidAudioDeviceObjects(void* jvm, void* context);
-  // This function must be called when the AudioManagerJni class is no
-  // longer needed. It frees up the global references acquired in
-  // SetAndroidAudioDeviceObjects.
-  static void ClearAndroidAudioDeviceObjects();
-
-  bool low_latency_supported() const { return low_latency_supported_; }
-  int native_output_sample_rate() const { return native_output_sample_rate_; }
-  int native_buffer_size() const { return native_buffer_size_; }
-
- private:
-  bool HasDeviceObjects();
-
-  // Following functions assume that the calling thread has been attached.
-  void SetLowLatencySupported(JNIEnv* env);
-  void SetNativeOutputSampleRate(JNIEnv* env);
-  void SetNativeFrameSize(JNIEnv* env);
-
-  jmethodID LookUpMethodId(JNIEnv* env, const char* method_name,
-                           const char* method_signature);
-
-  void CreateInstance(JNIEnv* env);
-
-  // Whether or not low latency audio is supported, the native output sample
-  // rate and the audio buffer size do not change. I.e the values might as well
-  // just be cached when initializing.
-  bool low_latency_supported_;
-  int native_output_sample_rate_;
-  int native_buffer_size_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_JNI_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_unittest.cc
new file mode 100644
index 0000000000..ddae73067a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_unittest.cc
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_device/android/build_info.h"
+#include "webrtc/modules/audio_device/android/audio_manager.h"
+#include "webrtc/modules/audio_device/android/ensure_initialized.h"
+
+#define PRINT(...) fprintf(stderr, __VA_ARGS__);
+
+namespace webrtc {
+
+static const char kTag[] = "  ";
+
+class AudioManagerTest : public ::testing::Test {
+ protected:
+  AudioManagerTest() {
+    // One-time initialization of JVM and application context. Ensures that we
+    // can do calls between C++ and Java.
+    webrtc::audiodevicemodule::EnsureInitialized();
+    audio_manager_.reset(new AudioManager());
+    SetActiveAudioLayer();
+    playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
+    record_parameters_ = audio_manager()->GetRecordAudioParameters();
+  }
+
+  AudioManager* audio_manager() const { return audio_manager_.get(); }
+
+  // A valid audio layer must always be set before calling Init(), hence we
+  // might as well make it a part of the test fixture.
+  void SetActiveAudioLayer() {
+    EXPECT_EQ(0, audio_manager()->GetDelayEstimateInMilliseconds());
+    audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio);
+    EXPECT_NE(0, audio_manager()->GetDelayEstimateInMilliseconds());
+  }
+
+  rtc::scoped_ptr<AudioManager> audio_manager_;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+};
+
+TEST_F(AudioManagerTest, ConstructDestruct) {
+}
+
+TEST_F(AudioManagerTest, InitClose) {
+  EXPECT_TRUE(audio_manager()->Init());
+  EXPECT_TRUE(audio_manager()->Close());
+}
+
+TEST_F(AudioManagerTest, IsAcousticEchoCancelerSupported) {
+  PRINT("%sAcoustic Echo Canceler support: %s\n", kTag,
+        audio_manager()->IsAcousticEchoCancelerSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsAutomaticGainControlSupported) {
+  PRINT("%sAutomatic Gain Control support: %s\n", kTag,
+        audio_manager()->IsAutomaticGainControlSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsNoiseSuppressorSupported) {
+  PRINT("%sNoise Suppressor support: %s\n", kTag,
+        audio_manager()->IsNoiseSuppressorSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsLowLatencyPlayoutSupported) {
+  PRINT("%sLow latency output support: %s\n", kTag,
+        audio_manager()->IsLowLatencyPlayoutSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
+  const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
+  PRINT("PLAYOUT:\n");
+  PRINT("%saudio layer: %s\n", kTag,
+        low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
+  PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
+  PRINT("%schannels: %" PRIuS "\n", kTag, playout_parameters_.channels());
+  PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
+        playout_parameters_.frames_per_buffer(),
+        playout_parameters_.GetBufferSizeInMilliseconds());
+  PRINT("RECORD: \n");
+  PRINT("%saudio layer: %s\n", kTag, "Java/JNI based AudioRecord");
+  PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
+  PRINT("%schannels: %" PRIuS "\n", kTag, record_parameters_.channels());
+  PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
+        record_parameters_.frames_per_buffer(),
+        record_parameters_.GetBufferSizeInMilliseconds());
+}
+
+// Add device-specific information to the test for logging purposes.
+TEST_F(AudioManagerTest, ShowDeviceInfo) {
+  BuildInfo build_info;
+  PRINT("%smodel: %s\n", kTag, build_info.GetDeviceModel().c_str());
+  PRINT("%sbrand: %s\n", kTag, build_info.GetBrand().c_str());
+  PRINT("%smanufacturer: %s\n",
+        kTag, build_info.GetDeviceManufacturer().c_str());
+}
+
+// Add Android build information to the test for logging purposes.
+TEST_F(AudioManagerTest, ShowBuildInfo) {
+  BuildInfo build_info;
+  PRINT("%sbuild release: %s\n", kTag, build_info.GetBuildRelease().c_str());
+  PRINT("%sbuild id: %s\n", kTag, build_info.GetAndroidBuildId().c_str());
+  PRINT("%sbuild type: %s\n", kTag, build_info.GetBuildType().c_str());
+  PRINT("%sSDK version: %s\n", kTag, build_info.GetSdkVersion().c_str());
+}
+
+// Basic test of the AudioParameters class using default construction where
+// all members are set to zero.
+TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) {
+  AudioParameters params;
+  EXPECT_FALSE(params.is_valid());
+  EXPECT_EQ(0, params.sample_rate());
+  EXPECT_EQ(0U, params.channels());
+  EXPECT_EQ(0U, params.frames_per_buffer());
+  EXPECT_EQ(0U, params.frames_per_10ms_buffer());
+  EXPECT_EQ(0U, params.GetBytesPerFrame());
+  EXPECT_EQ(0U, params.GetBytesPerBuffer());
+  EXPECT_EQ(0U, params.GetBytesPer10msBuffer());
+  EXPECT_EQ(0.0f, params.GetBufferSizeInMilliseconds());
+}
+
+// Basic test of the AudioParameters class using non default construction.
+TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
+  const int kSampleRate = 48000;
+  const size_t kChannels = 1;
+  const size_t kFramesPerBuffer = 480;
+  const size_t kFramesPer10msBuffer = 480;
+  const size_t kBytesPerFrame = 2;
+  const float kBufferSizeInMs = 10.0f;
+  AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer);
+  EXPECT_TRUE(params.is_valid());
+  EXPECT_EQ(kSampleRate, params.sample_rate());
+  EXPECT_EQ(kChannels, params.channels());
+  EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer());
+  EXPECT_EQ(static_cast<size_t>(kSampleRate / 100),
+            params.frames_per_10ms_buffer());
+  EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
+  EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
+  EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
+            params.GetBytesPer10msBuffer());
+  EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
+}
+
+}  // namespace webrtc
+
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.cc
index 4d5d150c62..440b472a88 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.cc
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "AndroidJNIWrapper.h"
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 
+#include <utility>
+
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 
 #define TAG "AudioRecordJni"
@@ -26,78 +28,79 @@
 
 namespace webrtc {
 
-// We are unable to obtain exact measurements of the hardware delay on Android.
-// Instead, a lower bound (based on measurements) is used.
-// TODO(henrika): is it possible to improve this?
-static const int kHardwareDelayInMilliseconds = 100;
+// AudioRecordJni::JavaAudioRecord implementation.
+AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
+    NativeRegistration* native_reg,
+    rtc::scoped_ptr<GlobalRef> audio_record)
+    : audio_record_(std::move(audio_record)),
+      init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
+      start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
+      stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
+      enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_agc_(native_reg->GetMethodId("enableBuiltInAGC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
 
-static JavaVM* g_jvm = NULL;
-static jobject g_context = NULL;
-static jclass g_audio_record_class = NULL;
+AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
 
-void AudioRecordJni::SetAndroidAudioDeviceObjects(void* jvm, void* context) {
-  ALOGD("SetAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-
-  CHECK(jvm);
-  CHECK(context);
-
-  g_jvm = reinterpret_cast<JavaVM*>(jvm);
-  JNIEnv* jni = GetEnv(g_jvm);
-  CHECK(jni) << "AttachCurrentThread must be called on this tread";
-
-  if (!g_context) {
-    // Protect context from being deleted during garbage collection.
-    g_context = NewGlobalRef(jni, reinterpret_cast<jobject>(context));
-  }
-
-  if (!g_audio_record_class) {
-    g_audio_record_class = jsjni_GetGlobalClassRef(
-                               "org/webrtc/voiceengine/WebRtcAudioRecord");
-    DCHECK(g_audio_record_class);
-  }
-
-  // Register native methods with the WebRtcAudioRecord class. These methods
-  // are declared private native in WebRtcAudioRecord.java.
-  JNINativeMethod native_methods[] = {
-      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
-          reinterpret_cast<void*>(
-       &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
-      {"nativeDataIsRecorded", "(IJ)V",
-          reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
-  jni->RegisterNatives(g_audio_record_class,
-                       native_methods, arraysize(native_methods));
-  CHECK_EXCEPTION(jni) << "Error during RegisterNatives";
+int AudioRecordJni::JavaAudioRecord::InitRecording(
+    int sample_rate, size_t channels) {
+  return audio_record_->CallIntMethod(init_recording_,
+                                      static_cast<jint>(sample_rate),
+                                      static_cast<jint>(channels));
 }
 
-void AudioRecordJni::ClearAndroidAudioDeviceObjects() {
-  ALOGD("ClearAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-  JNIEnv* jni = GetEnv(g_jvm);
-  CHECK(jni) << "AttachCurrentThread must be called on this tread";
-  jni->UnregisterNatives(g_audio_record_class);
-  CHECK_EXCEPTION(jni) << "Error during UnregisterNatives";
-  DeleteGlobalRef(jni, g_audio_record_class);
-  g_audio_record_class = NULL;
-  DeleteGlobalRef(jni, g_context);
-  g_context = NULL;
-  g_jvm = NULL;
+bool AudioRecordJni::JavaAudioRecord::StartRecording() {
+  return audio_record_->CallBooleanMethod(start_recording_);
 }
 
-AudioRecordJni::AudioRecordJni(
-    PlayoutDelayProvider* delay_provider, AudioManager* audio_manager)
-    : delay_provider_(delay_provider),
+bool AudioRecordJni::JavaAudioRecord::StopRecording() {
+  return audio_record_->CallBooleanMethod(stop_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_aec_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAGC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_agc_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_ns_,
+                                          static_cast<jboolean>(enable));
+}
+
+// AudioRecordJni implementation.
+AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_manager_(audio_manager),
       audio_parameters_(audio_manager->GetRecordAudioParameters()),
-      j_audio_record_(NULL),
-      direct_buffer_address_(NULL),
+      total_delay_in_milliseconds_(0),
+      direct_buffer_address_(nullptr),
       direct_buffer_capacity_in_bytes_(0),
       frames_per_buffer_(0),
       initialized_(false),
       recording_(false),
-      audio_device_buffer_(NULL),
-      playout_delay_in_milliseconds_(0) {
+      audio_device_buffer_(nullptr) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
-  DCHECK(audio_parameters_.is_valid());
-  CHECK(HasDeviceObjects());
-  CreateJavaInstance();
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+      reinterpret_cast<void*>(
+          &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
+      {"nativeDataIsRecorded", "(IJ)V",
+      reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioRecord",
+      native_methods, arraysize(native_methods));
+  j_audio_record_.reset(new JavaAudioRecord(
+      j_native_registration_.get(),
+      j_native_registration_->NewObject(
+          "<init>", "(Landroid/content/Context;J)V",
+          JVM::GetInstance()->context(), PointerTojlong(this))));
   // Detach from this thread since we want to use the checker to verify calls
   // from the Java based audio thread.
   thread_checker_java_.DetachFromThread();
@@ -105,71 +108,49 @@ AudioRecordJni::AudioRecordJni(
 
 AudioRecordJni::~AudioRecordJni() {
   ALOGD("~dtor%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Terminate();
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jni->DeleteGlobalRef(j_audio_record_);
-  j_audio_record_ = NULL;
 }
 
 int32_t AudioRecordJni::Init() {
   ALOGD("Init%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return 0;
 }
 
 int32_t AudioRecordJni::Terminate() {
   ALOGD("Terminate%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   StopRecording();
   return 0;
 }
 
 int32_t AudioRecordJni::InitRecording() {
   ALOGD("InitRecording%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!initialized_);
-  DCHECK(!recording_);
-  if (initialized_ || recording_) {
-    return -1;
-  }
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID initRecordingID = GetMethodID(
-      jni, g_audio_record_class, "InitRecording", "(II)I");
-  jint frames_per_buffer = jni->CallIntMethod(
-      j_audio_record_, initRecordingID, audio_parameters_.sample_rate(),
-      audio_parameters_.channels());
-  CHECK_EXCEPTION(jni);
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  int frames_per_buffer = j_audio_record_->InitRecording(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
   if (frames_per_buffer < 0) {
     ALOGE("InitRecording failed!");
     return -1;
   }
-  frames_per_buffer_ = frames_per_buffer;
-  ALOGD("frames_per_buffer: %d", frames_per_buffer_);
-  CHECK_EQ(direct_buffer_capacity_in_bytes_,
-           frames_per_buffer_ * kBytesPerFrame);
-  CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_buffer());
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
+               frames_per_buffer_ * kBytesPerFrame);
+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
   initialized_ = true;
   return 0;
 }
 
 int32_t AudioRecordJni::StartRecording() {
   ALOGD("StartRecording%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(initialized_);
-  DCHECK(!recording_);
-  if (!initialized_ || recording_) {
-    return -1;
-  }
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID startRecordingID = GetMethodID(
-      jni, g_audio_record_class, "StartRecording", "()Z");
-  jboolean res = jni->CallBooleanMethod(j_audio_record_, startRecordingID);
-  CHECK_EXCEPTION(jni);
-  if (!res) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!recording_);
+  if (!j_audio_record_->StartRecording()) {
     ALOGE("StartRecording failed!");
     return -1;
   }
@@ -179,75 +160,56 @@ int32_t AudioRecordJni::StartRecording() {
 
 int32_t AudioRecordJni::StopRecording() {
   ALOGD("StopRecording%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (!initialized_ || !recording_) {
     return 0;
   }
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID stopRecordingID = GetMethodID(
-      jni, g_audio_record_class, "StopRecording", "()Z");
-  jboolean res = jni->CallBooleanMethod(j_audio_record_, stopRecordingID);
-  CHECK_EXCEPTION(jni);
-  if (!res) {
+  if (!j_audio_record_->StopRecording()) {
     ALOGE("StopRecording failed!");
     return -1;
   }
-  // If we don't detach here, we will hit a DCHECK in OnDataIsRecorded() next
-  // time StartRecording() is called since it will create a new Java thread.
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
   thread_checker_java_.DetachFromThread();
   initialized_ = false;
   recording_ = false;
-  return 0;
-}
-
-int32_t AudioRecordJni::RecordingDelay(uint16_t& delayMS) const {  // NOLINT
-  // TODO(henrika): is it possible to improve this estimate?
-  delayMS = kHardwareDelayInMilliseconds;
+  direct_buffer_address_= nullptr;
   return 0;
 }
 
 void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   ALOGD("AttachAudioBuffer");
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   audio_device_buffer_ = audioBuffer;
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetRecordingChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetRecordingChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetRecordingChannels(channels);
-}
-
-bool AudioRecordJni::BuiltInAECIsAvailable() const {
-  ALOGD("BuiltInAECIsAvailable%s", GetThreadInfo().c_str());
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID builtInAECIsAvailable = jni->GetStaticMethodID(
-      g_audio_record_class, "BuiltInAECIsAvailable", "()Z");
-  CHECK_EXCEPTION(jni);
-  CHECK(builtInAECIsAvailable);
-  jboolean hw_aec = jni->CallStaticBooleanMethod(g_audio_record_class,
-                                                 builtInAECIsAvailable);
-  CHECK_EXCEPTION(jni);
-  return hw_aec;
+  total_delay_in_milliseconds_ =
+      audio_manager_->GetDelayEstimateInMilliseconds();
+  RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
+  ALOGD("total_delay_in_milliseconds: %d", total_delay_in_milliseconds_);
 }
 
 int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
   ALOGD("EnableBuiltInAEC%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID enableBuiltInAEC = GetMethodID(
-      jni, g_audio_record_class, "EnableBuiltInAEC", "(Z)Z");
-  jboolean res = jni->CallBooleanMethod(
-      j_audio_record_, enableBuiltInAEC, enable);
-  CHECK_EXCEPTION(jni);
-  if (!res) {
-    ALOGE("EnableBuiltInAEC failed!");
-    return -1;
-  }
-  return 0;
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
+}
+
+int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
+  ALOGD("EnableBuiltInAGC%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_record_->EnableBuiltInAGC(enable) ? 0 : -1;
+}
+
+int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
+  ALOGD("EnableBuiltInNS%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
 }
 
 void JNICALL AudioRecordJni::CacheDirectBufferAddress(
@@ -260,12 +222,13 @@ void JNICALL AudioRecordJni::CacheDirectBufferAddress(
 void AudioRecordJni::OnCacheDirectBufferAddress(
     JNIEnv* env, jobject byte_buffer) {
   ALOGD("OnCacheDirectBufferAddress");
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!direct_buffer_address_);
   direct_buffer_address_ =
       env->GetDirectBufferAddress(byte_buffer);
   jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
   ALOGD("direct buffer capacity: %lld", capacity);
-  direct_buffer_capacity_in_bytes_ = static_cast<int> (capacity);
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
 }
 
 void JNICALL AudioRecordJni::DataIsRecorded(
@@ -278,46 +241,24 @@ void JNICALL AudioRecordJni::DataIsRecorded(
 // This method is called on a high-priority thread from Java. The name of
 // the thread is 'AudioRecordThread'.
 void AudioRecordJni::OnDataIsRecorded(int length) {
-  DCHECK(thread_checker_java_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_java_.CalledOnValidThread());
   if (!audio_device_buffer_) {
     ALOGE("AttachAudioBuffer has not been called!");
     return;
   }
-  if (playout_delay_in_milliseconds_ == 0) {
-    playout_delay_in_milliseconds_ = delay_provider_->PlayoutDelayMs();
-    ALOGD("cached playout delay: %d", playout_delay_in_milliseconds_);
-  }
   audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
                                           frames_per_buffer_);
-  audio_device_buffer_->SetVQEData(playout_delay_in_milliseconds_,
-                                   kHardwareDelayInMilliseconds,
-                                   0 /* clockDrift */);
-  if (audio_device_buffer_->DeliverRecordedData() == 1) {
+  // We provide one (combined) fixed delay estimate for the APM and use the
+  // |playDelayMs| parameter only. Components like the AEC only sees the sum
+  // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
+  audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_,
+                                   0,   // recDelayMs
+                                   0);  // clockDrift
+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
     ALOGE("AudioDeviceBuffer::DeliverRecordedData failed!");
   }
 }
 
-bool AudioRecordJni::HasDeviceObjects() {
-  return (g_jvm && g_context && g_audio_record_class);
-}
-
-void AudioRecordJni::CreateJavaInstance() {
-  ALOGD("CreateJavaInstance");
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID constructorID = GetMethodID(
-      jni, g_audio_record_class, "<init>", "(Landroid/content/Context;J)V");
-  j_audio_record_ = jni->NewObject(g_audio_record_class,
-                                   constructorID,
-                                   g_context,
-                                   reinterpret_cast<intptr_t>(this));
-  CHECK_EXCEPTION(jni) << "Error during NewObject";
-  CHECK(j_audio_record_);
-  j_audio_record_ = jni->NewGlobalRef(j_audio_record_);
-  CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
-  CHECK(j_audio_record_);
-}
-
 int32_t AudioRecordJni::RecordingDeviceName(uint16_t index,
                                             char name[kAdmMaxDeviceNameSize],
                                             char guid[kAdmMaxGuidSize]) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.h
index 87556aee7b..8d431d3398 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -17,12 +17,11 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
-class PlayoutDelayProvider;
-
 // Implements 16-bit mono PCM audio input support for Android using the Java
 // AudioRecord interface. Most of the work is done by its Java counterpart in
 // WebRtcAudioRecord.java. This class is created and lives on a thread in
@@ -36,30 +35,38 @@ class PlayoutDelayProvider;
 //
 // An instance must be created and destroyed on one and the same thread.
 // All public methods must also be called on the same thread. A thread checker
-// will DCHECK if any method is called on an invalid thread.
-// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
-// and ClearAndroidAudioDeviceObjects) from a different thread but both will
-// CHECK that the calling thread is attached to a Java VM.
+// will RTC_DCHECK if any method is called on an invalid thread.
 //
-// All methods use AttachThreadScoped to attach to a Java VM if needed and then
-// detach when method goes out of scope. We do so because this class does not
-// own the thread is is created and called on and other objects on the same
-// thread might put us in a detached state at any time.
+// This class uses AttachCurrentThreadIfNeeded to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
 class AudioRecordJni {
  public:
-  // Use the invocation API to allow the native application to use the JNI
-  // interface pointer to access VM features.
-  // |jvm| denotes the Java VM and |context| corresponds to
-  // android.content.Context in Java.
-  // This method also sets a global jclass object, |g_audio_record_class| for
-  // the "org/webrtc/voiceengine/WebRtcAudioRecord"-class.
-  static void SetAndroidAudioDeviceObjects(void* jvm, void* context);
-  // Always call this method after the object has been destructed. It deletes
-  // existing global references and enables garbage collection.
-  static void ClearAndroidAudioDeviceObjects();
+  // Wraps the Java specific parts of the AudioRecordJni into one helper class.
+  class JavaAudioRecord {
+   public:
+    JavaAudioRecord(NativeRegistration* native_registration,
+                   rtc::scoped_ptr<GlobalRef> audio_track);
+    ~JavaAudioRecord();
 
-  AudioRecordJni(
-      PlayoutDelayProvider* delay_provider, AudioManager* audio_manager);
+    int InitRecording(int sample_rate, size_t channels);
+    bool StartRecording();
+    bool StopRecording();
+    bool EnableBuiltInAEC(bool enable);
+    bool EnableBuiltInAGC(bool enable);
+    bool EnableBuiltInNS(bool enable);
+
+   private:
+    rtc::scoped_ptr<GlobalRef> audio_record_;
+    jmethodID init_recording_;
+    jmethodID start_recording_;
+    jmethodID stop_recording_;
+    jmethodID enable_built_in_aec_;
+    jmethodID enable_built_in_agc_;
+    jmethodID enable_built_in_ns_;
+  };
+
+  explicit AudioRecordJni(AudioManager* audio_manager);
   ~AudioRecordJni();
 
   int32_t Init();
@@ -69,15 +76,14 @@ class AudioRecordJni {
   bool RecordingIsInitialized() const { return initialized_; }
 
   int32_t StartRecording();
-  int32_t StopRecording ();
+  int32_t StopRecording();
   bool Recording() const { return recording_; }
 
-  int32_t RecordingDelay(uint16_t& delayMS) const;
-
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
 
-  bool BuiltInAECIsAvailable() const;
   int32_t EnableBuiltInAEC(bool enable);
+  int32_t EnableBuiltInAGC(bool enable);
+  int32_t EnableBuiltInNS(bool enable);
   int32_t RecordingDeviceName(uint16_t index,
                               char name[kAdmMaxDeviceNameSize],
                               char guid[kAdmMaxGuidSize]);
@@ -102,47 +108,49 @@ class AudioRecordJni {
     JNIEnv* env, jobject obj, jint length, jlong nativeAudioRecord);
   void OnDataIsRecorded(int length);
 
-  // Returns true if SetAndroidAudioDeviceObjects() has been called
-  // successfully.
-  bool HasDeviceObjects();
-
-  // Called from the constructor. Defines the |j_audio_record_| member.
-  void CreateJavaInstance();
-
   // Stores thread ID in constructor.
-  // We can then use ThreadChecker::CalledOnValidThread() to ensure that
-  // other methods are called from the same thread.
-  // Currently only does DCHECK(thread_checker_.CalledOnValidThread()).
   rtc::ThreadChecker thread_checker_;
 
   // Stores thread ID in first call to OnDataIsRecorded() from high-priority
   // thread in Java. Detached during construction of this object.
   rtc::ThreadChecker thread_checker_java_;
 
-  // Returns the current playout delay.
-  // TODO(henrika): this value is currently fixed since initial tests have
-  // shown that the estimated delay varies very little over time. It might be
-  // possible to make improvements in this area.
-  PlayoutDelayProvider* delay_provider_;
+  // Calls AttachCurrentThread() if this thread is not attached at construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  AttachCurrentThreadIfNeeded attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  rtc::scoped_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioRecordJni class.
+  rtc::scoped_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
+
+  // Raw pointer to the audio manger.
+  const AudioManager* audio_manager_;
 
   // Contains audio parameters provided to this class at construction by the
   // AudioManager.
   const AudioParameters audio_parameters_;
 
-  // The Java WebRtcAudioRecord instance.
-  jobject j_audio_record_;
+  // Delay estimate of the total round-trip delay (input + output).
+  // Fixed value set once in AttachAudioBuffer() and it can take one out of two
+  // possible values. See audio_common.h for details.
+  int total_delay_in_milliseconds_;
 
   // Cached copy of address to direct audio buffer owned by |j_audio_record_|.
   void* direct_buffer_address_;
 
   // Number of bytes in the direct audio buffer owned by |j_audio_record_|.
-  int direct_buffer_capacity_in_bytes_;
+  size_t direct_buffer_capacity_in_bytes_;
 
   // Number audio frames per audio buffer. Each audio frame corresponds to
   // one sample of PCM mono data at 16 bits per sample. Hence, each audio
   // frame contains 2 bytes (given that the Java layer only supports mono).
   // Example: 480 for 48000 Hz or 441 for 44100 Hz.
-  int frames_per_buffer_;
+  size_t frames_per_buffer_;
 
   bool initialized_;
 
@@ -151,9 +159,6 @@ class AudioRecordJni {
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
   // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
   AudioDeviceBuffer* audio_device_buffer_;
-
-  // Contains a delay estimate from the playout side given by |delay_provider_|.
-  int playout_delay_in_milliseconds_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.cc
index e81ace3f5d..acf6bb2f54 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.cc
@@ -10,12 +10,14 @@
 
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "AndroidJNIWrapper.h"
+
+#include <utility>
 
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 
 #define TAG "AudioTrackJni"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
@@ -26,73 +28,72 @@
 
 namespace webrtc {
 
-static JavaVM* g_jvm = NULL;
-static jobject g_context = NULL;
-static jclass g_audio_track_class = NULL;
+// AudioTrackJni::JavaAudioTrack implementation.
+AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
+    NativeRegistration* native_reg,
+    rtc::scoped_ptr<GlobalRef> audio_track)
+    : audio_track_(std::move(audio_track)),
+      init_playout_(native_reg->GetMethodId("initPlayout", "(II)V")),
+      start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
+      stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
+      set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
+      get_stream_max_volume_(
+          native_reg->GetMethodId("getStreamMaxVolume", "()I")),
+      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {}
 
-void AudioTrackJni::SetAndroidAudioDeviceObjects(void* jvm, void* context) {
-  ALOGD("SetAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
+AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
 
-  CHECK(jvm);
-  CHECK(context);
-
-  g_jvm = reinterpret_cast<JavaVM*>(jvm);
-  JNIEnv* jni = GetEnv(g_jvm);
-  CHECK(jni) << "AttachCurrentThread must be called on this tread";
-
-  if (!g_context) {
-    g_context = NewGlobalRef(jni, reinterpret_cast<jobject>(context));
-  }
-
-  if (!g_audio_track_class) {
-    g_audio_track_class = jsjni_GetGlobalClassRef(
-                              "org/webrtc/voiceengine/WebRtcAudioTrack");
-    DCHECK(g_audio_track_class);
-  }
-
-  // Register native methods with the WebRtcAudioTrack class. These methods
-  // are declared private native in WebRtcAudioTrack.java.
-  JNINativeMethod native_methods[] = {
-      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
-          reinterpret_cast<void*>(
-       &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
-      {"nativeGetPlayoutData", "(IJ)V",
-          reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
-  jni->RegisterNatives(g_audio_track_class,
-                       native_methods, arraysize(native_methods));
-  CHECK_EXCEPTION(jni) << "Error during RegisterNatives";
+void AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
+  audio_track_->CallVoidMethod(init_playout_, sample_rate, channels);
 }
 
-// TODO(henrika): figure out if it is required to call this method? If so,
-// ensure that is is always called as part of the destruction phase.
-void AudioTrackJni::ClearAndroidAudioDeviceObjects() {
-  ALOGD("ClearAndroidAudioDeviceObjects%s", GetThreadInfo().c_str());
-  JNIEnv* jni = GetEnv(g_jvm);
-  CHECK(jni) << "AttachCurrentThread must be called on this tread";
-  jni->UnregisterNatives(g_audio_track_class);
-  CHECK_EXCEPTION(jni) << "Error during UnregisterNatives";
-  DeleteGlobalRef(jni, g_audio_track_class);
-  g_audio_track_class = NULL;
-  DeleteGlobalRef(jni, g_context);
-  g_context = NULL;
-  g_jvm = NULL;
+bool AudioTrackJni::JavaAudioTrack::StartPlayout() {
+  return audio_track_->CallBooleanMethod(start_playout_);
+}
+
+bool AudioTrackJni::JavaAudioTrack::StopPlayout() {
+  return audio_track_->CallBooleanMethod(stop_playout_);
+}
+
+bool AudioTrackJni::JavaAudioTrack::SetStreamVolume(int volume) {
+  return audio_track_->CallBooleanMethod(set_stream_volume_, volume);
+}
+
+int AudioTrackJni::JavaAudioTrack::GetStreamMaxVolume() {
+  return audio_track_->CallIntMethod(get_stream_max_volume_);
+}
+
+int AudioTrackJni::JavaAudioTrack::GetStreamVolume() {
+  return audio_track_->CallIntMethod(get_stream_volume_);
 }
 
 // TODO(henrika): possible extend usage of AudioManager and add it as member.
 AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
-    : audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
-      j_audio_track_(NULL),
-      direct_buffer_address_(NULL),
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      direct_buffer_address_(nullptr),
       direct_buffer_capacity_in_bytes_(0),
       frames_per_buffer_(0),
       initialized_(false),
       playing_(false),
-      audio_device_buffer_(NULL),
-      delay_in_milliseconds_(0) {
+      audio_device_buffer_(nullptr) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
-  DCHECK(audio_parameters_.is_valid());
-  CHECK(HasDeviceObjects());
-  CreateJavaInstance();
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+      reinterpret_cast<void*>(
+          &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
+      {"nativeGetPlayoutData", "(IJ)V",
+      reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioTrack",
+      native_methods, arraysize(native_methods));
+  j_audio_track_.reset(new JavaAudioTrack(
+      j_native_registration_.get(),
+      j_native_registration_->NewObject(
+          "<init>", "(Landroid/content/Context;J)V",
+          JVM::GetInstance()->context(), PointerTojlong(this))));
   // Detach from this thread since we want to use the checker to verify calls
   // from the Java based audio thread.
   thread_checker_java_.DetachFromThread();
@@ -100,68 +101,40 @@ AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
 
 AudioTrackJni::~AudioTrackJni() {
   ALOGD("~dtor%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Terminate();
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jni->DeleteGlobalRef(j_audio_track_);
-  j_audio_track_ = NULL;
 }
 
 int32_t AudioTrackJni::Init() {
   ALOGD("Init%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return 0;
 }
 
 int32_t AudioTrackJni::Terminate() {
   ALOGD("Terminate%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   StopPlayout();
   return 0;
 }
 
 int32_t AudioTrackJni::InitPlayout() {
   ALOGD("InitPlayout%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!initialized_);
-  DCHECK(!playing_);
-  if (initialized_ || playing_) {
-    return -1;
-  }
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID initPlayoutID = GetMethodID(
-      jni, g_audio_track_class, "InitPlayout", "(II)I");
-  jint delay_in_milliseconds = jni->CallIntMethod(
-      j_audio_track_, initPlayoutID, audio_parameters_.sample_rate(),
-      audio_parameters_.channels());
-  CHECK_EXCEPTION(jni);
-  if (delay_in_milliseconds < 0) {
-    ALOGE("InitPlayout failed!");
-    return -1;
-  }
-  delay_in_milliseconds_ = delay_in_milliseconds;
-  ALOGD("delay_in_milliseconds: %d", delay_in_milliseconds);
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  j_audio_track_->InitPlayout(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
   initialized_ = true;
   return 0;
 }
 
 int32_t AudioTrackJni::StartPlayout() {
   ALOGD("StartPlayout%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(initialized_);
-  DCHECK(!playing_);
-  if (!initialized_ || playing_) {
-    return -1;
-  }
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID startPlayoutID = GetMethodID(
-      jni, g_audio_track_class, "StartPlayout", "()Z");
-  jboolean res = jni->CallBooleanMethod(j_audio_track_, startPlayoutID);
-  CHECK_EXCEPTION(jni);
-  if (!res) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!playing_);
+  if (!j_audio_track_->StartPlayout()) {
     ALOGE("StartPlayout failed!");
     return -1;
   }
@@ -171,25 +144,21 @@ int32_t AudioTrackJni::StartPlayout() {
 
 int32_t AudioTrackJni::StopPlayout() {
   ALOGD("StopPlayout%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (!initialized_ || !playing_) {
     return 0;
   }
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID stopPlayoutID = GetMethodID(
-      jni, g_audio_track_class, "StopPlayout", "()Z");
-  jboolean res = jni->CallBooleanMethod(j_audio_track_, stopPlayoutID);
-  CHECK_EXCEPTION(jni);
-  if (!res) {
+  if (!j_audio_track_->StopPlayout()) {
     ALOGE("StopPlayout failed!");
     return -1;
   }
-  // If we don't detach here, we will hit a DCHECK in OnDataIsRecorded() next
-  // time StartRecording() is called since it will create a new Java thread.
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
   thread_checker_java_.DetachFromThread();
   initialized_ = false;
   playing_ = false;
+  direct_buffer_address_ = nullptr;
   return 0;
 }
 
@@ -200,77 +169,44 @@ int AudioTrackJni::SpeakerVolumeIsAvailable(bool& available) {
 
 int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
   ALOGD("SetSpeakerVolume(%d)%s", volume, GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID setStreamVolume = GetMethodID(
-      jni, g_audio_track_class, "SetStreamVolume", "(I)Z");
-  jboolean res = jni->CallBooleanMethod(
-      j_audio_track_, setStreamVolume, volume);
-  CHECK_EXCEPTION(jni);
-  return res ? 0 : -1;
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_track_->SetStreamVolume(volume) ? 0 : -1;
 }
 
 int AudioTrackJni::MaxSpeakerVolume(uint32_t& max_volume) const {
   ALOGD("MaxSpeakerVolume%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID getStreamMaxVolume = GetMethodID(
-      jni, g_audio_track_class, "GetStreamMaxVolume", "()I");
-  jint max_vol = jni->CallIntMethod(j_audio_track_, getStreamMaxVolume);
-  CHECK_EXCEPTION(jni);
-  max_volume = max_vol;
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  max_volume = j_audio_track_->GetStreamMaxVolume();
   return 0;
 }
 
 int AudioTrackJni::MinSpeakerVolume(uint32_t& min_volume) const {
   ALOGD("MaxSpeakerVolume%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   min_volume = 0;
   return 0;
 }
 
 int AudioTrackJni::SpeakerVolume(uint32_t& volume) const {
   ALOGD("SpeakerVolume%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID getStreamVolume = GetMethodID(
-      jni, g_audio_track_class, "GetStreamVolume", "()I");
-  jint stream_volume = jni->CallIntMethod(j_audio_track_, getStreamVolume);
-  CHECK_EXCEPTION(jni);
-  volume = stream_volume;
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  volume = j_audio_track_->GetStreamVolume();
   return 0;
 }
 
 // TODO(henrika): possibly add stereo support.
 void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   ALOGD("AttachAudioBuffer%s", GetThreadInfo().c_str());
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   audio_device_buffer_ = audioBuffer;
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetPlayoutChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetPlayoutChannels(channels);
 }
 
-int32_t AudioTrackJni::PlayoutDelay(uint16_t& delayMS) const {
-  // No need for thread check or locking since we set |delay_in_milliseconds_|
-  // only once  (on the creating thread) during initialization.
-  delayMS = delay_in_milliseconds_;
-  return 0;
-}
-
-int AudioTrackJni::PlayoutDelayMs() {
-  // This method can be called from the Java based AudioRecordThread but we
-  // don't need locking since it is only set once (on the main thread) during
-  // initialization.
-  return delay_in_milliseconds_;
-}
-
 void JNICALL AudioTrackJni::CacheDirectBufferAddress(
     JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioTrack) {
   webrtc::AudioTrackJni* this_object =
@@ -281,28 +217,29 @@ void JNICALL AudioTrackJni::CacheDirectBufferAddress(
 void AudioTrackJni::OnCacheDirectBufferAddress(
     JNIEnv* env, jobject byte_buffer) {
   ALOGD("OnCacheDirectBufferAddress");
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!direct_buffer_address_);
   direct_buffer_address_ =
       env->GetDirectBufferAddress(byte_buffer);
   jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
   ALOGD("direct buffer capacity: %lld", capacity);
-  direct_buffer_capacity_in_bytes_ = static_cast<int> (capacity);
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
   frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / kBytesPerFrame;
-  ALOGD("frames_per_buffer: %d", frames_per_buffer_);
+  ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
 }
 
 void JNICALL AudioTrackJni::GetPlayoutData(
   JNIEnv* env, jobject obj, jint length, jlong nativeAudioTrack) {
   webrtc::AudioTrackJni* this_object =
       reinterpret_cast<webrtc::AudioTrackJni*> (nativeAudioTrack);
-  this_object->OnGetPlayoutData(length);
+  this_object->OnGetPlayoutData(static_cast<size_t>(length));
 }
 
 // This method is called on a high-priority thread from Java. The name of
 // the thread is 'AudioRecordTrack'.
-void AudioTrackJni::OnGetPlayoutData(int length) {
-  DCHECK(thread_checker_java_.CalledOnValidThread());
-  DCHECK_EQ(frames_per_buffer_, length / kBytesPerFrame);
+void AudioTrackJni::OnGetPlayoutData(size_t length) {
+  RTC_DCHECK(thread_checker_java_.CalledOnValidThread());
+  RTC_DCHECK_EQ(frames_per_buffer_, length / kBytesPerFrame);
   if (!audio_device_buffer_) {
     ALOGE("AttachAudioBuffer has not been called!");
     return;
@@ -313,32 +250,11 @@ void AudioTrackJni::OnGetPlayoutData(int length) {
     ALOGE("AudioDeviceBuffer::RequestPlayoutData failed!");
     return;
   }
-  DCHECK_EQ(samples, frames_per_buffer_);
+  RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_);
   // Copy decoded data into common byte buffer to ensure that it can be
   // written to the Java based audio track.
   samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
-  DCHECK_EQ(length, kBytesPerFrame * samples);
-}
-
-bool AudioTrackJni::HasDeviceObjects() {
-  return (g_jvm && g_context && g_audio_track_class);
-}
-
-void AudioTrackJni::CreateJavaInstance() {
-  ALOGD("CreateJavaInstance");
-  AttachThreadScoped ats(g_jvm);
-  JNIEnv* jni = ats.env();
-  jmethodID constructorID = GetMethodID(
-      jni, g_audio_track_class, "<init>", "(Landroid/content/Context;J)V");
-  j_audio_track_ = jni->NewObject(g_audio_track_class,
-                                  constructorID,
-                                  g_context,
-                                  reinterpret_cast<intptr_t>(this));
-  CHECK_EXCEPTION(jni) << "Error during NewObject";
-  CHECK(j_audio_track_);
-  j_audio_track_ = jni->NewGlobalRef(j_audio_track_);
-  CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
-  CHECK(j_audio_track_);
+  RTC_DCHECK_EQ(length, kBytesPerFrame * samples);
 }
 
 int32_t AudioTrackJni::PlayoutDeviceName(uint16_t index,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.h
index 44b751f2af..c322ac7d5b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/audio_track_jni.h
@@ -18,7 +18,8 @@
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 
@@ -30,29 +31,38 @@ namespace webrtc {
 //
 // An instance must be created and destroyed on one and the same thread.
 // All public methods must also be called on the same thread. A thread checker
-// will DCHECK if any method is called on an invalid thread.
-// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
-// and ClearAndroidAudioDeviceObjects) from a different thread but both will
-// CHECK that the calling thread is attached to a Java VM.
+// will RTC_DCHECK if any method is called on an invalid thread.
 //
-// All methods use AttachThreadScoped to attach to a Java VM if needed and then
-// detach when method goes out of scope. We do so because this class does not
-// own the thread is is created and called on and other objects on the same
-// thread might put us in a detached state at any time.
-class AudioTrackJni : public PlayoutDelayProvider {
+// This class uses AttachCurrentThreadIfNeeded to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioTrackJni {
  public:
-  // Use the invocation API to allow the native application to use the JNI
-  // interface pointer to access VM features.
-  // |jvm| denotes the Java VM and |context| corresponds to
-  // android.content.Context in Java.
-  // This method also sets a global jclass object, |g_audio_track_class| for
-  // the "org/webrtc/voiceengine/WebRtcAudioTrack"-class.
-  static void SetAndroidAudioDeviceObjects(void* jvm, void* context);
-  // Always call this method after the object has been destructed. It deletes
-  // existing global references and enables garbage collection.
-  static void ClearAndroidAudioDeviceObjects();
+  // Wraps the Java specific parts of the AudioTrackJni into one helper class.
+  class JavaAudioTrack {
+   public:
+    JavaAudioTrack(NativeRegistration* native_registration,
+                   rtc::scoped_ptr<GlobalRef> audio_track);
+    ~JavaAudioTrack();
 
-  AudioTrackJni(AudioManager* audio_manager);
+    void InitPlayout(int sample_rate, int channels);
+    bool StartPlayout();
+    bool StopPlayout();
+    bool SetStreamVolume(int volume);
+    int GetStreamMaxVolume();
+    int GetStreamVolume();
+
+   private:
+    rtc::scoped_ptr<GlobalRef> audio_track_;
+    jmethodID init_playout_;
+    jmethodID start_playout_;
+    jmethodID stop_playout_;
+    jmethodID set_stream_volume_;
+    jmethodID get_stream_max_volume_;
+    jmethodID get_stream_volume_;
+  };
+
+  explicit AudioTrackJni(AudioManager* audio_manager);
   ~AudioTrackJni();
 
   int32_t Init();
@@ -71,17 +81,12 @@ class AudioTrackJni : public PlayoutDelayProvider {
   int MaxSpeakerVolume(uint32_t& max_volume) const;
   int MinSpeakerVolume(uint32_t& min_volume) const;
 
-  int32_t PlayoutDelay(uint16_t& delayMS) const;
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
 
   int32_t PlayoutDeviceName(uint16_t index,
                             char name[kAdmMaxDeviceNameSize],
                             char guid[kAdmMaxGuidSize]);
 
- protected:
-  // PlayoutDelayProvider implementation.
-  virtual int PlayoutDelayMs();
-
  private:
   // Called from Java side so we can cache the address of the Java-manged
   // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
@@ -98,42 +103,43 @@ class AudioTrackJni : public PlayoutDelayProvider {
   // the thread is 'AudioTrackThread'.
   static void JNICALL GetPlayoutData(
     JNIEnv* env, jobject obj, jint length, jlong nativeAudioTrack);
-  void OnGetPlayoutData(int length);
-
-  // Returns true if SetAndroidAudioDeviceObjects() has been called
-  // successfully.
-  bool HasDeviceObjects();
-
-  // Called from the constructor. Defines the |j_audio_track_| member.
-  void CreateJavaInstance();
+  void OnGetPlayoutData(size_t length);
 
   // Stores thread ID in constructor.
-  // We can then use ThreadChecker::CalledOnValidThread() to ensure that
-  // other methods are called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
   // Stores thread ID in first call to OnGetPlayoutData() from high-priority
   // thread in Java. Detached during construction of this object.
   rtc::ThreadChecker thread_checker_java_;
 
+  // Calls AttachCurrentThread() if this thread is not attached at construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  AttachCurrentThreadIfNeeded attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  rtc::scoped_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioTrackJni class.
+  rtc::scoped_ptr<AudioTrackJni::JavaAudioTrack> j_audio_track_;
+
   // Contains audio parameters provided to this class at construction by the
   // AudioManager.
   const AudioParameters audio_parameters_;
 
-  // The Java WebRtcAudioTrack instance.
-  jobject j_audio_track_;
-
   // Cached copy of address to direct audio buffer owned by |j_audio_track_|.
   void* direct_buffer_address_;
 
   // Number of bytes in the direct audio buffer owned by |j_audio_track_|.
-  int direct_buffer_capacity_in_bytes_;
+  size_t direct_buffer_capacity_in_bytes_;
 
   // Number of audio frames per audio buffer. Each audio frame corresponds to
   // one sample of PCM mono data at 16 bits per sample. Hence, each audio
   // frame contains 2 bytes (given that the Java layer only supports mono).
   // Example: 480 for 48000 Hz or 441 for 44100 Hz.
-  int frames_per_buffer_;
+  size_t frames_per_buffer_;
 
   bool initialized_;
 
@@ -144,12 +150,6 @@ class AudioTrackJni : public PlayoutDelayProvider {
   // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
   // and therefore outlives this object.
   AudioDeviceBuffer* audio_device_buffer_;
-
-  // Estimated playout delay caused by buffering in the Java based audio track.
-  // We are using a fixed value here since measurements have shown that the
-  // variations are very small (~10ms) and it is not worth the extra complexity
-  // to update this estimate on a continuous basis.
-  int delay_in_milliseconds_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/build_info.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/build_info.cc
new file mode 100644
index 0000000000..6289697073
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/build_info.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_device/android/build_info.h"
+
+#include "webrtc/modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+BuildInfo::BuildInfo()
+    : j_environment_(JVM::GetInstance()->environment()),
+      j_build_info_(JVM::GetInstance()->GetClass(
+          "org/webrtc/voiceengine/BuildInfo")) {
+}
+
+std::string BuildInfo::GetStringFromJava(const char* name) {
+  jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;");
+  jstring j_string = static_cast<jstring>(
+      j_build_info_.CallStaticObjectMethod(id));
+  return j_environment_->JavaToStdString(j_string);
+}
+
+std::string BuildInfo::GetDeviceModel() {
+  return GetStringFromJava("getDeviceModel");
+}
+
+std::string BuildInfo::GetBrand() {
+  return GetStringFromJava("getBrand");
+}
+
+std::string BuildInfo::GetDeviceManufacturer() {
+  return GetStringFromJava("getDeviceManufacturer");
+}
+
+std::string BuildInfo::GetAndroidBuildId() {
+  return GetStringFromJava("getAndroidBuildId");
+}
+
+std::string BuildInfo::GetBuildType() {
+  return GetStringFromJava("getBuildType");
+}
+
+std::string BuildInfo::GetBuildRelease() {
+  return GetStringFromJava("getBuildRelease");
+}
+
+std::string BuildInfo::GetSdkVersion() {
+  return GetStringFromJava("getSdkVersion");
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/build_info.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/build_info.h
new file mode 100644
index 0000000000..1490fa0772
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/build_info.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
+
+#include <jni.h>
+#include <string>
+
+#include "webrtc/modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Utility class used to query the Java class (org/webrtc/voiceengine/BuildInfo)
+// for device and Android build information.
+// The calling thread is attached to the JVM at construction if needed and a
+// valid Java environment object is also created.
+// All Get methods must be called on the creating thread. If not, the code will
+// hit RTC_DCHECKs when calling JNIEnvironment::JavaToStdString().
+class BuildInfo {
+ public:
+  BuildInfo();
+  ~BuildInfo() {}
+
+  // End-user-visible name for the end product (e.g. "Nexus 6").
+  std::string GetDeviceModel();
+  // Consumer-visible brand (e.g. "google").
+  std::string GetBrand();
+  // Manufacturer of the product/hardware (e.g. "motorola").
+  std::string GetDeviceManufacturer();
+  // Android build ID (e.g. LMY47D).
+  std::string GetAndroidBuildId();
+  // The type of build (e.g. "user" or "eng").
+  std::string GetBuildType();
+  // The user-visible version string (e.g. "5.1").
+  std::string GetBuildRelease();
+  // The user-visible SDK version of the framework (e.g. 21).
+  std::string GetSdkVersion();
+
+ private:
+  // Helper method which calls a static getter method with |name| and returns
+  // a string from Java.
+  std::string GetStringFromJava(const char* name);
+
+  // Ensures that this class can access a valid JNI interface pointer even
+  // if the creating thread was not attached to the JVM.
+  AttachCurrentThreadIfNeeded attach_thread_if_needed_;
+
+  // Provides access to the JNIEnv interface pointer and the JavaToStdString()
+  // method which is used to translate Java strings to std strings.
+  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+
+  // Holds the jclass object and provides access to CallStaticObjectMethod().
+  // Used by GetStringFromJava() during construction only.
+  JavaClass j_build_info_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/ensure_initialized.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/ensure_initialized.cc
index b07c04a0cf..b63aec1f27 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/ensure_initialized.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/ensure_initialized.cc
@@ -12,13 +12,14 @@
 
 #include <pthread.h>
 
+// Note: this dependency is dangerous since it reaches into Chromium's base.
+// There's a risk of e.g. macro clashes. This file may only be used in tests.
+#include "base/android/context_utils.h"
 #include "base/android/jni_android.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_device/android/audio_device_template.h"
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "webrtc/modules/audio_device/android/opensles_input.h"
-#include "webrtc/modules/audio_device/android/opensles_output.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
 
 namespace webrtc {
 namespace audiodevicemodule {
@@ -26,25 +27,18 @@ namespace audiodevicemodule {
 static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
 
 void EnsureInitializedOnce() {
-  CHECK(::base::android::IsVMInitialized());
+  RTC_CHECK(::base::android::IsVMInitialized());
   JNIEnv* jni = ::base::android::AttachCurrentThread();
   JavaVM* jvm = NULL;
-  CHECK_EQ(0, jni->GetJavaVM(&jvm));
+  RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
   jobject context = ::base::android::GetApplicationContext();
 
-  // Provide JVM and context to Java and OpenSL ES implementations.
-  using AudioDeviceJava = AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>;
-  AudioDeviceJava::SetAndroidAudioDeviceObjects(jvm, context);
-
-  // TODO(henrika): enable OpenSL ES when it has been refactored to avoid
-  // crashes.
-  // using AudioDeviceOpenSLES =
-  //    AudioDeviceTemplate<OpenSlesInput, OpenSlesOutput>;
-  // AudioDeviceOpenSLES::SetAndroidAudioDeviceObjects(jvm, context);
+  // Initialize the Java environment (currently only used by the audio manager).
+  webrtc::JVM::Initialize(jvm, context);
 }
 
 void EnsureInitialized() {
-  CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
+  RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
 }
 
 }  // namespace audiodevicemodule
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer.cc
deleted file mode 100644
index ee5667991d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer.cc
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/android/fine_audio_buffer.h"
-
-#include <memory.h>
-#include <stdio.h>
-#include <algorithm>
-
-#include "webrtc/modules/audio_device/audio_device_buffer.h"
-
-namespace webrtc {
-
-FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                                 int desired_frame_size_bytes,
-                                 int sample_rate)
-    : device_buffer_(device_buffer),
-      desired_frame_size_bytes_(desired_frame_size_bytes),
-      sample_rate_(sample_rate),
-      samples_per_10_ms_(sample_rate_ * 10 / 1000),
-      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
-      cached_buffer_start_(0),
-      cached_bytes_(0) {
-  cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
-}
-
-FineAudioBuffer::~FineAudioBuffer() {
-}
-
-int FineAudioBuffer::RequiredBufferSizeBytes() {
-  // It is possible that we store the desired frame size - 1 samples. Since new
-  // audio frames are pulled in chunks of 10ms we will need a buffer that can
-  // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
-  return desired_frame_size_bytes_ + bytes_per_10_ms_;
-}
-
-void FineAudioBuffer::GetBufferData(int8_t* buffer) {
-  if (desired_frame_size_bytes_ <= cached_bytes_) {
-    memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_],
-           desired_frame_size_bytes_);
-    cached_buffer_start_ += desired_frame_size_bytes_;
-    cached_bytes_ -= desired_frame_size_bytes_;
-    assert(cached_buffer_start_ + cached_bytes_ < bytes_per_10_ms_);
-    return;
-  }
-  memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_], cached_bytes_);
-  // Push another n*10ms of audio to |buffer|. n > 1 if
-  // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
-  // write the audio after the cached bytes copied earlier.
-  int8_t* unwritten_buffer = &buffer[cached_bytes_];
-  int bytes_left = desired_frame_size_bytes_ - cached_bytes_;
-  // Ceiling of integer division: 1 + ((x - 1) / y)
-  int number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
-  for (int i = 0; i < number_of_requests; ++i) {
-    device_buffer_->RequestPlayoutData(samples_per_10_ms_);
-    int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
-    if (num_out != samples_per_10_ms_) {
-      assert(num_out == 0);
-      cached_bytes_ = 0;
-      return;
-    }
-    unwritten_buffer += bytes_per_10_ms_;
-    assert(bytes_left >= 0);
-    bytes_left -= bytes_per_10_ms_;
-  }
-  assert(bytes_left <= 0);
-  // Put the samples that were written to |buffer| but are not used in the
-  // cache.
-  int cache_location = desired_frame_size_bytes_;
-  int8_t* cache_ptr = &buffer[cache_location];
-  cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
-      (desired_frame_size_bytes_ - cached_bytes_);
-  // If cached_bytes_ is larger than the cache buffer, uninitialized memory
-  // will be read.
-  assert(cached_bytes_ <= bytes_per_10_ms_);
-  assert(-bytes_left == cached_bytes_);
-  cached_buffer_start_ = 0;
-  memcpy(cache_buffer_.get(), cache_ptr, cached_bytes_);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer.h
deleted file mode 100644
index 812fe1fec6..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class AudioDeviceBuffer;
-
-// FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
-// corresponding to 10ms of data. It then allows for this data to be pulled in
-// a finer or coarser granularity. I.e. interacting with this class instead of
-// directly with the AudioDeviceBuffer one can ask for any number of audio data
-// samples.
-class FineAudioBuffer {
- public:
-  // |device_buffer| is a buffer that provides 10ms of audio data.
-  // |desired_frame_size_bytes| is the number of bytes of audio data
-  // (not samples) |GetBufferData| should return on success.
-  // |sample_rate| is the sample rate of the audio data. This is needed because
-  // |device_buffer| delivers 10ms of data. Given the sample rate the number
-  // of samples can be calculated.
-  FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                  int desired_frame_size_bytes,
-                  int sample_rate);
-  ~FineAudioBuffer();
-
-  // Returns the required size of |buffer| when calling GetBufferData. If the
-  // buffer is smaller memory trampling will happen.
-  // |desired_frame_size_bytes| and |samples_rate| are as described in the
-  // constructor.
-  int RequiredBufferSizeBytes();
-
-  // |buffer| must be of equal or greater size than what is returned by
-  // RequiredBufferSize. This is to avoid unnecessary memcpy.
-  void GetBufferData(int8_t* buffer);
-
- private:
-  // Device buffer that provides 10ms chunks of data.
-  AudioDeviceBuffer* device_buffer_;
-  int desired_frame_size_bytes_;  // Number of bytes delivered per GetBufferData
-  int sample_rate_;
-  int samples_per_10_ms_;
-  // Convenience parameter to avoid converting from samples
-  int bytes_per_10_ms_;
-
-  // Storage for samples that are not yet asked for.
-  rtc::scoped_ptr<int8_t[]> cache_buffer_;
-  int cached_buffer_start_;  // Location of first unread sample.
-  int cached_bytes_;  // Number of bytes stored in cache.
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java
deleted file mode 100644
index c80207e28a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// The functions in this file are called from native code. They can still be
-// accessed even though they are declared private.
-
-package org.webrtc.voiceengine;
-
-import android.content.Context;
-import android.content.pm.PackageManager;
-import android.media.AudioManager;
-import android.util.Log;
-
-import java.lang.reflect.Field;
-import java.lang.reflect.Method;
-
-import org.mozilla.gecko.annotation.WebRTCJNITarget;
-
-@WebRTCJNITarget
-class AudioManagerAndroid {
-  // Most of Google lead devices use 44.1K as the default sampling rate, 44.1K
-  // is also widely used on other android devices.
-  private static final int DEFAULT_SAMPLING_RATE = 44100;
-  // Randomly picked up frame size which is close to return value on N4.
-  // Return this default value when
-  // getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER) fails.
-  private static final int DEFAULT_FRAMES_PER_BUFFER = 256;
-
-  private int mNativeOutputSampleRate;
-  private boolean mAudioLowLatencySupported;
-  private int mAudioLowLatencyOutputFrameSize;
-
-
-  @SuppressWarnings("unused")
-  private AudioManagerAndroid(Context context) {
-    AudioManager audioManager = (AudioManager)
-        context.getSystemService(Context.AUDIO_SERVICE);
-
-    mNativeOutputSampleRate = DEFAULT_SAMPLING_RATE;
-    mAudioLowLatencyOutputFrameSize = DEFAULT_FRAMES_PER_BUFFER;
-    mAudioLowLatencySupported = context.getPackageManager().hasSystemFeature(
-      PackageManager.FEATURE_AUDIO_LOW_LATENCY);
-    if (android.os.Build.VERSION.SDK_INT >=
-        17 /*android.os.Build.VERSION_CODES.JELLY_BEAN_MR1*/) {
-      try {
-        Method getProperty = AudioManager.class.getMethod("getProperty", String.class);
-        Field sampleRateField = AudioManager.class.getField("PROPERTY_OUTPUT_SAMPLE_RATE");
-        Field framePerBufferField = AudioManager.class.getField("PROPERTY_OUTPUT_FRAMES_PER_BUFFER");
-        String sampleRateKey = (String)sampleRateField.get(null);
-        String framePerBufferKey = (String)framePerBufferField.get(null);
-        String sampleRateString = (String)getProperty.invoke(audioManager, sampleRateKey);
-        if (sampleRateString != null) {
-          mNativeOutputSampleRate = Integer.parseInt(sampleRateString);
-        }
-        String framesPerBuffer = (String)getProperty.invoke(audioManager, sampleRateKey);
-        if (framesPerBuffer != null) {
-          mAudioLowLatencyOutputFrameSize = Integer.parseInt(framesPerBuffer);
-        }
-      } catch (Exception ex) {
-        Log.w("WebRTC", "error getting low latency params", ex);
-      }
-    }
-  }
-
-    @SuppressWarnings("unused")
-    private int getNativeOutputSampleRate() {
-      return mNativeOutputSampleRate;
-    }
-
-    @SuppressWarnings("unused")
-    private boolean isAudioLowLatencySupported() {
-        return mAudioLowLatencySupported;
-    }
-
-    @SuppressWarnings("unused")
-    private int getAudioLowLatencyOutputFrameSize() {
-        return mAudioLowLatencyOutputFrameSize;
-    }
-}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java
new file mode 100644
index 0000000000..6ba1c666ec
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.os.Build;
+import android.util.Log;
+
+import org.mozilla.gecko.annotation.WebRTCJNITarget;
+
+@WebRTCJNITarget
+public final class BuildInfo {
+  public static String getDevice() {
+    return Build.DEVICE;
+  }
+
+  public static String getDeviceModel() {
+    return Build.MODEL;
+  }
+
+  public static String getProduct() {
+    return Build.PRODUCT;
+  }
+
+  public static String getBrand() {
+    return Build.BRAND;
+  }
+
+  public static String getDeviceManufacturer() {
+    return Build.MANUFACTURER;
+  }
+
+  public static String getAndroidBuildId() {
+    return Build.ID;
+  }
+
+  public static String getBuildType() {
+    return Build.TYPE;
+  }
+
+  public static String getBuildRelease() {
+    return Build.VERSION.RELEASE;
+  }
+
+  public static String getSdkVersion() {
+    return Integer.toString(Build.VERSION.SDK_INT);
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
new file mode 100644
index 0000000000..14adc9fc56
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
@@ -0,0 +1,394 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.annotation.TargetApi;
+import android.media.audiofx.AcousticEchoCanceler;
+import android.media.audiofx.AudioEffect;
+import android.media.audiofx.AudioEffect.Descriptor;
+import android.media.audiofx.AutomaticGainControl;
+import android.media.audiofx.NoiseSuppressor;
+import android.os.Build;
+
+import android.util.Log;
+
+import java.util.List;
+
+import java.util.UUID;
+
+// This class wraps control of three different platform effects. Supported
+// effects are: AcousticEchoCanceler (AEC), AutomaticGainControl (AGC) and
+// NoiseSuppressor (NS). Calling enable() will active all effects that are
+// supported by the device if the corresponding |shouldEnableXXX| member is set.
+class WebRtcAudioEffects {
+  private static final boolean DEBUG = false;
+
+  private static final String TAG = "WebRtcAudioEffects";
+
+  // UUIDs for Software Audio Effects that we want to avoid using.
+  // The implementor field will be set to "The Android Open Source Project".
+  private static final UUID AOSP_ACOUSTIC_ECHO_CANCELER =
+      UUID.fromString("bb392ec0-8d4d-11e0-a896-0002a5d5c51b");
+  private static final UUID AOSP_AUTOMATIC_GAIN_CONTROL =
+      UUID.fromString("aa8130e0-66fc-11e0-bad0-0002a5d5c51b");
+  private static final UUID AOSP_NOISE_SUPPRESSOR =
+      UUID.fromString("c06c8400-8e06-11e0-9cb6-0002a5d5c51b");
+
+  // Static Boolean objects used to avoid expensive queries more than once.
+  // The first result is cached in these members and then reused if needed.
+  // Each member is null until it has been evaluated/set for the first time.
+  private static Boolean canUseAcousticEchoCanceler = null;
+  private static Boolean canUseAutomaticGainControl = null;
+  private static Boolean canUseNoiseSuppressor = null;
+
+  // Contains the audio effect objects. Created in enable() and destroyed
+  // in release().
+  private AcousticEchoCanceler aec = null;
+  private AutomaticGainControl agc = null;
+  private NoiseSuppressor ns = null;
+
+  // Affects the final state given to the setEnabled() method on each effect.
+  // The default state is set to "disabled" but each effect can also be enabled
+  // by calling setAEC(), setAGC() and setNS().
+  // To enable an effect, both the shouldEnableXXX member and the static
+  // canUseXXX() must be true.
+  private boolean shouldEnableAec = false;
+  private boolean shouldEnableAgc = false;
+  private boolean shouldEnableNs = false;
+
+  // Checks if the device implements Acoustic Echo Cancellation (AEC).
+  // Returns true if the device implements AEC, false otherwise.
+  public static boolean isAcousticEchoCancelerSupported() {
+    return WebRtcAudioUtils.runningOnJellyBeanOrHigher()
+        && AcousticEchoCanceler.isAvailable();
+  }
+
+  // Checks if the device implements Automatic Gain Control (AGC).
+  // Returns true if the device implements AGC, false otherwise.
+  public static boolean isAutomaticGainControlSupported() {
+    return WebRtcAudioUtils.runningOnJellyBeanOrHigher()
+        && AutomaticGainControl.isAvailable();
+  }
+
+  // Checks if the device implements Noise Suppression (NS).
+  // Returns true if the device implements NS, false otherwise.
+  public static boolean isNoiseSuppressorSupported() {
+    return WebRtcAudioUtils.runningOnJellyBeanOrHigher()
+        && NoiseSuppressor.isAvailable();
+  }
+
+  // Returns true if the device is blacklisted for HW AEC usage.
+  public static boolean isAcousticEchoCancelerBlacklisted() {
+    List<String> blackListedModels =
+        WebRtcAudioUtils.getBlackListedModelsForAecUsage();
+    boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
+    if (isBlacklisted) {
+      Log.w(TAG, Build.MODEL + " is blacklisted for HW AEC usage!");
+    }
+    return isBlacklisted;
+  }
+
+  // Returns true if the device is blacklisted for HW AGC usage.
+  public static boolean isAutomaticGainControlBlacklisted() {
+   List<String> blackListedModels =
+        WebRtcAudioUtils.getBlackListedModelsForAgcUsage();
+    boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
+    if (isBlacklisted) {
+      Log.w(TAG, Build.MODEL + " is blacklisted for HW AGC usage!");
+    }
+    return isBlacklisted;
+  }
+
+  // Returns true if the device is blacklisted for HW NS usage.
+  public static boolean isNoiseSuppressorBlacklisted() {
+    List<String> blackListedModels =
+        WebRtcAudioUtils.getBlackListedModelsForNsUsage();
+    boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
+    if (isBlacklisted) {
+      Log.w(TAG, Build.MODEL + " is blacklisted for HW NS usage!");
+    }
+    return isBlacklisted;
+  }
+
+  // Returns true if the platform AEC should be excluded based on its UUID.
+  // AudioEffect.queryEffects() can throw IllegalStateException.
+  @TargetApi(18)
+  private static boolean isAcousticEchoCancelerExcludedByUUID() {
+    for (Descriptor d : AudioEffect.queryEffects()) {
+      if (d.type.equals(AudioEffect.EFFECT_TYPE_AEC) &&
+          d.uuid.equals(AOSP_ACOUSTIC_ECHO_CANCELER)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // Returns true if the platform AGC should be excluded based on its UUID.
+  // AudioEffect.queryEffects() can throw IllegalStateException.
+  @TargetApi(18)
+  private static boolean isAutomaticGainControlExcludedByUUID() {
+    for (Descriptor d : AudioEffect.queryEffects()) {
+      if (d.type.equals(AudioEffect.EFFECT_TYPE_AGC) &&
+          d.uuid.equals(AOSP_AUTOMATIC_GAIN_CONTROL)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // Returns true if the platform NS should be excluded based on its UUID.
+  // AudioEffect.queryEffects() can throw IllegalStateException.
+  @TargetApi(18)
+  private static boolean isNoiseSuppressorExcludedByUUID() {
+    for (Descriptor d : AudioEffect.queryEffects()) {
+      if (d.type.equals(AudioEffect.EFFECT_TYPE_NS) &&
+          d.uuid.equals(AOSP_NOISE_SUPPRESSOR)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // Returns true if all conditions for supporting the HW AEC are fulfilled.
+  // It will not be possible to enable the HW AEC if this method returns false.
+  public static boolean canUseAcousticEchoCanceler() {
+    if (canUseAcousticEchoCanceler == null) {
+      canUseAcousticEchoCanceler = new Boolean(
+          isAcousticEchoCancelerSupported()
+          && !WebRtcAudioUtils.useWebRtcBasedAcousticEchoCanceler()
+          && !isAcousticEchoCancelerBlacklisted()
+          && !isAcousticEchoCancelerExcludedByUUID());
+      Log.d(TAG, "canUseAcousticEchoCanceler: "
+          + canUseAcousticEchoCanceler);
+    }
+    return canUseAcousticEchoCanceler;
+  }
+
+  // Returns true if all conditions for supporting the HW AGC are fulfilled.
+  // It will not be possible to enable the HW AGC if this method returns false.
+  public static boolean canUseAutomaticGainControl() {
+    if (canUseAutomaticGainControl == null) {
+      canUseAutomaticGainControl = new Boolean(
+          isAutomaticGainControlSupported()
+          && !WebRtcAudioUtils.useWebRtcBasedAutomaticGainControl()
+          && !isAutomaticGainControlBlacklisted()
+          && !isAutomaticGainControlExcludedByUUID());
+      Log.d(TAG, "canUseAutomaticGainControl: "
+          + canUseAutomaticGainControl);
+    }
+    return canUseAutomaticGainControl;
+  }
+
+  // Returns true if all conditions for supporting the HW NS are fulfilled.
+  // It will not be possible to enable the HW NS if this method returns false.
+  public static boolean canUseNoiseSuppressor() {
+    if (canUseNoiseSuppressor == null) {
+      canUseNoiseSuppressor = new Boolean(
+          isNoiseSuppressorSupported()
+          && !WebRtcAudioUtils.useWebRtcBasedNoiseSuppressor()
+          && !isNoiseSuppressorBlacklisted()
+          && !isNoiseSuppressorExcludedByUUID());
+      Log.d(TAG, "canUseNoiseSuppressor: " + canUseNoiseSuppressor);
+    }
+    return canUseNoiseSuppressor;
+  }
+
+  static WebRtcAudioEffects create() {
+    // Return null if VoIP effects (AEC, AGC and NS) are not supported.
+    if (!WebRtcAudioUtils.runningOnJellyBeanOrHigher()) {
+      Log.w(TAG, "API level 16 or higher is required!");
+      return null;
+    }
+    return new WebRtcAudioEffects();
+  }
+
+  private WebRtcAudioEffects() {
+    Log.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+  }
+
+  // Call this method to enable or disable the platform AEC. It modifies
+  // |shouldEnableAec| which is used in enable() where the actual state
+  // of the AEC effect is modified. Returns true if HW AEC is supported and
+  // false otherwise.
+  public boolean setAEC(boolean enable) {
+    Log.d(TAG, "setAEC(" + enable + ")");
+    if (!canUseAcousticEchoCanceler()) {
+      Log.w(TAG, "Platform AEC is not supported");
+      shouldEnableAec = false;
+      return false;
+    }
+    if (aec != null && (enable != shouldEnableAec)) {
+      Log.e(TAG, "Platform AEC state can't be modified while recording");
+      return false;
+    }
+    shouldEnableAec = enable;
+    return true;
+  }
+
+  // Call this method to enable or disable the platform AGC. It modifies
+  // |shouldEnableAgc| which is used in enable() where the actual state
+  // of the AGC effect is modified. Returns true if HW AGC is supported and
+  // false otherwise.
+  public boolean setAGC(boolean enable) {
+    Log.d(TAG, "setAGC(" + enable + ")");
+    if (!canUseAutomaticGainControl()) {
+      Log.w(TAG, "Platform AGC is not supported");
+      shouldEnableAgc = false;
+      return false;
+    }
+    if (agc != null && (enable != shouldEnableAgc)) {
+      Log.e(TAG, "Platform AGC state can't be modified while recording");
+      return false;
+    }
+    shouldEnableAgc = enable;
+    return true;
+  }
+
+  // Call this method to enable or disable the platform NS. It modifies
+  // |shouldEnableNs| which is used in enable() where the actual state
+  // of the NS effect is modified. Returns true if HW NS is supported and
+  // false otherwise.
+  public boolean setNS(boolean enable) {
+    Log.d(TAG, "setNS(" + enable + ")");
+    if (!canUseNoiseSuppressor()) {
+      Log.w(TAG, "Platform NS is not supported");
+      shouldEnableNs = false;
+      return false;
+    }
+    if (ns != null && (enable != shouldEnableNs)) {
+      Log.e(TAG, "Platform NS state can't be modified while recording");
+      return false;
+    }
+    shouldEnableNs = enable;
+    return true;
+  }
+
+  public void enable(int audioSession) {
+    Log.d(TAG, "enable(audioSession=" + audioSession + ")");
+    assertTrue(aec == null);
+    assertTrue(agc == null);
+    assertTrue(ns == null);
+
+    // Add logging of supported effects but filter out "VoIP effects", i.e.,
+    // AEC, AEC and NS.
+    for (Descriptor d : AudioEffect.queryEffects()) {
+      if (effectTypeIsVoIP(d.type) || DEBUG) {
+        Log.d(TAG, "name: " + d.name + ", "
+            + "mode: " + d.connectMode + ", "
+            + "implementor: " + d.implementor + ", "
+            + "UUID: " + d.uuid);
+      }
+    }
+
+    if (isAcousticEchoCancelerSupported()) {
+      // Create an AcousticEchoCanceler and attach it to the AudioRecord on
+      // the specified audio session.
+      aec = AcousticEchoCanceler.create(audioSession);
+      if (aec != null) {
+        boolean enabled = aec.getEnabled();
+        boolean enable = shouldEnableAec && canUseAcousticEchoCanceler();
+        if (aec.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Log.e(TAG, "Failed to set the AcousticEchoCanceler state");
+        }
+        Log.d(TAG, "AcousticEchoCanceler: was "
+            + (enabled ? "enabled" : "disabled")
+            + ", enable: " + enable + ", is now: "
+            + (aec.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Log.e(TAG, "Failed to create the AcousticEchoCanceler instance");
+      }
+    }
+
+    if (isAutomaticGainControlSupported()) {
+      // Create an AutomaticGainControl and attach it to the AudioRecord on
+      // the specified audio session.
+      agc = AutomaticGainControl.create(audioSession);
+      if (agc != null) {
+        boolean enabled = agc.getEnabled();
+        boolean enable = shouldEnableAgc && canUseAutomaticGainControl();
+        if (agc.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Log.e(TAG, "Failed to set the AutomaticGainControl state");
+        }
+        Log.d(TAG, "AutomaticGainControl: was "
+            + (enabled ? "enabled" : "disabled")
+            + ", enable: " + enable + ", is now: "
+            + (agc.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Log.e(TAG, "Failed to create the AutomaticGainControl instance");
+      }
+    }
+
+    if (isNoiseSuppressorSupported()) {
+      // Create an NoiseSuppressor and attach it to the AudioRecord on the
+      // specified audio session.
+      ns = NoiseSuppressor.create(audioSession);
+      if (ns != null) {
+        boolean enabled = ns.getEnabled();
+        boolean enable = shouldEnableNs && canUseNoiseSuppressor();
+        if (ns.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Log.e(TAG, "Failed to set the NoiseSuppressor state");
+        }
+        Log.d(TAG, "NoiseSuppressor: was "
+            + (enabled ? "enabled" : "disabled")
+            + ", enable: " + enable + ", is now: "
+            + (ns.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Log.e(TAG, "Failed to create the NoiseSuppressor instance");
+      }
+    }
+  }
+
+  // Releases all native audio effect resources. It is a good practice to
+  // release the effect engine when not in use as control can be returned
+  // to other applications or the native resources released.
+  public void release() {
+    Log.d(TAG, "release");
+    if (aec != null) {
+      aec.release();
+      aec = null;
+    }
+    if (agc != null) {
+      agc.release();
+      agc = null;
+    }
+    if (ns != null) {
+      ns.release();
+      ns = null;
+    }
+  }
+
+  // Returns true for effect types in |type| that are of "VoIP" types:
+  // Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
+  // Noise Suppressor (NS). Note that, an extra check for support is needed
+  // in each comparison since some devices includes effects in the
+  // AudioEffect.Descriptor array that are actually not available on the device.
+  // As an example: Samsung Galaxy S6 includes an AGC in the descriptor but
+  // AutomaticGainControl.isAvailable() returns false.
+  @TargetApi(18)
+  private boolean effectTypeIsVoIP(UUID type) {
+    if (!WebRtcAudioUtils.runningOnJellyBeanMR2OrHigher())
+      return false;
+
+    return (AudioEffect.EFFECT_TYPE_AEC.equals(type)
+        && isAcousticEchoCancelerSupported())
+        || (AudioEffect.EFFECT_TYPE_AGC.equals(type)
+        && isAutomaticGainControlSupported())
+        || (AudioEffect.EFFECT_TYPE_NS.equals(type)
+        && isNoiseSuppressorSupported());
+  }
+
+  // Helper method which throws an exception when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
index d66ae96496..8666480b8e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
@@ -10,11 +10,19 @@
 
 package org.webrtc.voiceengine;
 
+import android.annotation.TargetApi;
 import android.content.Context;
 import android.content.pm.PackageManager;
+import android.media.AudioFormat;
 import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioTrack;
+import android.os.Build;
+
 import android.util.Log;
 
+import java.lang.Math;
+
 // WebRtcAudioManager handles tasks that uses android.media.AudioManager.
 // At construction, storeAudioParameters() is called and it retrieves
 // fundamental audio parameters like native sample rate and number of channels.
@@ -30,17 +38,41 @@ import android.util.Log;
 import org.mozilla.gecko.annotation.WebRTCJNITarget;
 
 @WebRTCJNITarget
-class WebRtcAudioManager {
+public class WebRtcAudioManager {
   private static final boolean DEBUG = false;
 
   private static final String TAG = "WebRtcAudioManager";
 
-   // Use 44.1kHz as the default sampling rate.
-  private static final int SAMPLE_RATE_HZ = 44100;
+  private static boolean blacklistDeviceForOpenSLESUsage = false;
+  private static boolean blacklistDeviceForOpenSLESUsageIsOverridden = false;
+
+  // Call this method to override the deault list of blacklisted devices
+  // specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
+  // Allows an app to take control over which devices to exlude from using
+  // the OpenSL ES audio output path
+  public static synchronized void setBlacklistDeviceForOpenSLESUsage(
+      boolean enable) {
+    blacklistDeviceForOpenSLESUsageIsOverridden = true;
+    blacklistDeviceForOpenSLESUsage = enable;
+  }
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  private static final int BITS_PER_SAMPLE = 16;
+
+  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
 
   // TODO(henrika): add stereo support for playout.
   private static final int CHANNELS = 1;
 
+  // List of possible audio modes.
+  private static final String[] AUDIO_MODES = new String[] {
+      "MODE_NORMAL",
+      "MODE_RINGTONE",
+      "MODE_IN_CALL",
+      "MODE_IN_COMMUNICATION",
+  };
+
   private final long nativeAudioManager;
   private final Context context;
   private final AudioManager audioManager;
@@ -48,11 +80,18 @@ class WebRtcAudioManager {
   private boolean initialized = false;
   private int nativeSampleRate;
   private int nativeChannels;
-  private int savedAudioMode = AudioManager.MODE_INVALID;
-  private boolean savedIsSpeakerPhoneOn = false;
+
+  private boolean hardwareAEC;
+  private boolean hardwareAGC;
+  private boolean hardwareNS;
+  private boolean lowLatencyOutput;
+  private int sampleRate;
+  private int channels;
+  private int outputBufferSize;
+  private int inputBufferSize;
 
   WebRtcAudioManager(Context context, long nativeAudioManager) {
-    Logd("ctor" + WebRtcAudioUtils.getThreadInfo());
+    Log.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
     this.context = context;
     this.nativeAudioManager = nativeAudioManager;
     audioManager = (AudioManager) context.getSystemService(
@@ -61,92 +100,197 @@ class WebRtcAudioManager {
       WebRtcAudioUtils.logDeviceInfo(TAG);
     }
     storeAudioParameters();
-    // TODO(henrika): add stereo support for playout side.
     nativeCacheAudioParameters(
-      nativeSampleRate, nativeChannels, nativeAudioManager);
+        sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS,
+        lowLatencyOutput, outputBufferSize, inputBufferSize,
+        nativeAudioManager);
   }
 
   private boolean init() {
-    Logd("init" + WebRtcAudioUtils.getThreadInfo());
+    Log.d(TAG, "init" + WebRtcAudioUtils.getThreadInfo());
     if (initialized) {
       return true;
     }
-
-    // Store current audio state so we can restore it when close() is called.
-    savedAudioMode = audioManager.getMode();
-    savedIsSpeakerPhoneOn = audioManager.isSpeakerphoneOn();
-
-    // Switch to COMMUNICATION mode for best possible VoIP performance.
-    audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION);
-
-    if (DEBUG) {
-      Logd("savedAudioMode: " + savedAudioMode);
-      Logd("savedIsSpeakerPhoneOn: " + savedIsSpeakerPhoneOn);
-      Logd("hasEarpiece: " + hasEarpiece());
-    }
-
+    Log.d(TAG, "audio mode is: " + AUDIO_MODES[audioManager.getMode()]);
     initialized = true;
     return true;
   }
 
   private void dispose() {
-    Logd("dispose" + WebRtcAudioUtils.getThreadInfo());
+    Log.d(TAG, "dispose" + WebRtcAudioUtils.getThreadInfo());
     if (!initialized) {
       return;
     }
-    // Restore previously stored audio states.
-    setSpeakerphoneOn(savedIsSpeakerPhoneOn);
-    audioManager.setMode(savedAudioMode);
+  }
+
+  private boolean isCommunicationModeEnabled() {
+    return (audioManager.getMode() == AudioManager.MODE_IN_COMMUNICATION);
+  }
+
+  private boolean isDeviceBlacklistedForOpenSLESUsage() {
+    boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden ?
+        blacklistDeviceForOpenSLESUsage :
+        WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
+    if (blacklisted) {
+      Log.e(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
+    }
+    return blacklisted;
   }
 
   private void storeAudioParameters() {
     // Only mono is supported currently (in both directions).
     // TODO(henrika): add support for stereo playout.
-    nativeChannels = CHANNELS;
-    // Get native sample rate and store it in |nativeSampleRate|.
-    // Most common rates are 44100 and 48000 Hz.
-    if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
-      nativeSampleRate = SAMPLE_RATE_HZ;
-    } else {
-      String sampleRateString = audioManager.getProperty(
-          AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
-      nativeSampleRate = (sampleRateString == null) ?
-          SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
-    }
-    Logd("nativeSampleRate: " + nativeSampleRate);
-    Logd("nativeChannels: " + nativeChannels);
+    channels = CHANNELS;
+    sampleRate = getNativeOutputSampleRate();
+    hardwareAEC = isAcousticEchoCancelerSupported();
+    hardwareAGC = isAutomaticGainControlSupported();
+    hardwareNS = isNoiseSuppressorSupported();
+    lowLatencyOutput = isLowLatencyOutputSupported();
+    outputBufferSize = lowLatencyOutput ?
+        getLowLatencyOutputFramesPerBuffer() :
+        getMinOutputFrameSize(sampleRate, channels);
+    // TODO(henrika): add support for low-latency input.
+    inputBufferSize = getMinInputFrameSize(sampleRate, channels);
   }
 
-  /** Sets the speaker phone mode. */
-  private void setSpeakerphoneOn(boolean on) {
-    boolean wasOn = audioManager.isSpeakerphoneOn();
-    if (wasOn == on) {
-      return;
-    }
-    audioManager.setSpeakerphoneOn(on);
-  }
-
-  /** Gets the current earpiece state. */
+  // Gets the current earpiece state.
   private boolean hasEarpiece() {
     return context.getPackageManager().hasSystemFeature(
         PackageManager.FEATURE_TELEPHONY);
   }
 
-  /** Helper method which throws an exception  when an assertion has failed. */
+  // Returns true if low-latency audio output is supported.
+  private boolean isLowLatencyOutputSupported() {
+    return isOpenSLESSupported() &&
+        context.getPackageManager().hasSystemFeature(
+            PackageManager.FEATURE_AUDIO_LOW_LATENCY);
+  }
+
+  // Returns true if low-latency audio input is supported.
+  public boolean isLowLatencyInputSupported() {
+    // TODO(henrika): investigate if some sort of device list is needed here
+    // as well. The NDK doc states that: "As of API level 21, lower latency
+    // audio input is supported on select devices. To take advantage of this
+    // feature, first confirm that lower latency output is available".
+    return WebRtcAudioUtils.runningOnLollipopOrHigher() &&
+        isLowLatencyOutputSupported();
+  }
+
+  // Returns the native output sample rate for this device's output stream.
+  private int getNativeOutputSampleRate() {
+    // Override this if we're running on an old emulator image which only
+    // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE.
+    if (WebRtcAudioUtils.runningOnEmulator()) {
+      Log.d(TAG, "Running emulator, overriding sample rate to 8 kHz.");
+      return 8000;
+    }
+    // Default can be overriden by WebRtcAudioUtils.setDefaultSampleRateHz().
+    // If so, use that value and return here.
+    if (WebRtcAudioUtils.isDefaultSampleRateOverridden()) {
+      Log.d(TAG, "Default sample rate is overriden to " +
+          WebRtcAudioUtils.getDefaultSampleRateHz() + " Hz");
+      return WebRtcAudioUtils.getDefaultSampleRateHz();
+    }
+    // No overrides available. Deliver best possible estimate based on default
+    // Android AudioManager APIs.
+    final int sampleRateHz;
+    if (WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
+      sampleRateHz = getSampleRateOnJellyBeanMR10OrHigher();
+    } else {
+      sampleRateHz = WebRtcAudioUtils.getDefaultSampleRateHz();
+    }
+    Log.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
+    return sampleRateHz;
+  }
+
+  @TargetApi(17)
+  private int getSampleRateOnJellyBeanMR10OrHigher() {
+    String sampleRateString = audioManager.getProperty(
+        AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+    return (sampleRateString == null)
+        ? WebRtcAudioUtils.getDefaultSampleRateHz()
+        : Integer.parseInt(sampleRateString);
+  }
+
+  // Returns the native output buffer size for low-latency output streams.
+  @TargetApi(17)
+  private int getLowLatencyOutputFramesPerBuffer() {
+    assertTrue(isLowLatencyOutputSupported());
+    if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
+      return DEFAULT_FRAME_PER_BUFFER;
+    }
+    String framesPerBuffer = audioManager.getProperty(
+        AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+    return framesPerBuffer == null ?
+        DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
+  }
+
+  // Returns true if the device supports an audio effect (AEC, AGC or NS).
+  // Four conditions must be fulfilled if functions are to return true:
+  // 1) the platform must support the built-in (HW) effect,
+  // 2) explicit use (override) of a WebRTC based version must not be set,
+  // 3) the device must not be blacklisted for use of the effect, and
+  // 4) the UUID of the effect must be approved (some UUIDs can be excluded).
+  private static boolean isAcousticEchoCancelerSupported() {
+    return WebRtcAudioEffects.canUseAcousticEchoCanceler();
+  }
+  private static boolean isAutomaticGainControlSupported() {
+    return WebRtcAudioEffects.canUseAutomaticGainControl();
+  }
+  private static boolean isNoiseSuppressorSupported() {
+    return WebRtcAudioEffects.canUseNoiseSuppressor();
+  }
+
+  // Returns the minimum output buffer size for Java based audio (AudioTrack).
+  // This size can also be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency output.
+  private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig;
+    if (numChannels == 1) {
+      channelConfig = AudioFormat.CHANNEL_OUT_MONO;
+    } else if (numChannels == 2) {
+      channelConfig = AudioFormat.CHANNEL_OUT_STEREO;
+    } else {
+      return -1;
+    }
+    return AudioTrack.getMinBufferSize(
+        sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT) /
+        bytesPerFrame;
+  }
+
+  // Returns the native input buffer size for input streams.
+  private int getLowLatencyInputFramesPerBuffer() {
+    assertTrue(isLowLatencyInputSupported());
+    return getLowLatencyOutputFramesPerBuffer();
+  }
+
+  // Returns the minimum input buffer size for Java based audio (AudioRecord).
+  // This size can calso be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency input.
+  private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    assertTrue(numChannels == CHANNELS);
+    return AudioRecord.getMinBufferSize(sampleRateInHz,
+        AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT) /
+        bytesPerFrame;
+  }
+
+  // Returns true if OpenSL ES audio is supported.
+  private static boolean isOpenSLESSupported() {
+    // Check for API level 9 or higher, to confirm use of OpenSL ES.
+    return WebRtcAudioUtils.runningOnGingerBreadOrHigher();
+  }
+
+  // Helper method which throws an exception  when an assertion has failed.
   private static void assertTrue(boolean condition) {
     if (!condition) {
       throw new AssertionError("Expected condition to be true");
     }
   }
 
-  private static void Logd(String msg) {
-    Log.d(TAG, msg);
-  }
-
-  private static void Loge(String msg) {
-    Log.e(TAG, msg);
-  }
-
   private native void nativeCacheAudioParameters(
-      int sampleRate, int channels, long nativeAudioManager);
+    int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC,
+    boolean hardwareNS, boolean lowLatencyOutput, int outputBufferSize,
+    int inputBufferSize, long nativeAudioManager);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
index 12c5e42cdb..ca28eb5536 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
@@ -16,14 +16,12 @@ import java.util.concurrent.TimeUnit;
 
 import android.content.Context;
 import android.media.AudioFormat;
-import android.media.audiofx.AcousticEchoCanceler;
-import android.media.audiofx.AudioEffect;
-import android.media.audiofx.AudioEffect.Descriptor;
 import android.media.AudioRecord;
 import android.media.MediaRecorder.AudioSource;
 import android.os.Build;
 import android.os.Process;
 import android.os.SystemClock;
+
 import android.util.Log;
 
 import org.mozilla.gecko.annotation.WebRTCJNITarget;
@@ -44,17 +42,21 @@ class  WebRtcAudioRecord {
   // Average number of callbacks per second.
   private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
 
+  // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
+  // buffer size). The extra space is allocated to guard against glitches under
+  // high load.
+  private static final int BUFFER_SIZE_FACTOR = 2;
+
   private final long nativeAudioRecord;
   private final Context context;
 
+  private WebRtcAudioEffects effects = null;
+
   private ByteBuffer byteBuffer;
 
   private AudioRecord audioRecord;
   private AudioRecordThread audioThread = null;
 
-  private AcousticEchoCanceler aec = null;
-  private boolean useBuiltInAEC = false;
-
   /**
    * Audio thread which keeps calling ByteBuffer.read() waiting for audio
    * to be recorded. Feeds recorded data to the native counterpart as a
@@ -71,14 +73,7 @@ class  WebRtcAudioRecord {
     @Override
     public void run() {
       Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
-      Logd("AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
-
-      try {
-        audioRecord.startRecording();
-      } catch (IllegalStateException e) {
-          Loge("AudioRecord.startRecording failed: " + e.getMessage());
-        return;
-      }
+      Log.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
       assertTrue(audioRecord.getRecordingState()
           == AudioRecord.RECORDSTATE_RECORDING);
 
@@ -88,7 +83,7 @@ class  WebRtcAudioRecord {
         if (bytesRead == byteBuffer.capacity()) {
           nativeDataIsRecorded(bytesRead, nativeAudioRecord);
         } else {
-          Loge("AudioRecord.read failed: " + bytesRead);
+          Log.e(TAG,"AudioRecord.read failed: " + bytesRead);
           if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
             keepAlive = false;
           }
@@ -98,14 +93,14 @@ class  WebRtcAudioRecord {
           long durationInMs =
               TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime));
           lastTime = nowTime;
-          Logd("bytesRead[" + durationInMs + "] " + bytesRead);
+          Log.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead);
         }
       }
 
       try {
         audioRecord.stop();
       } catch (IllegalStateException e) {
-        Loge("AudioRecord.stop failed: " + e.getMessage());
+        Log.e(TAG,"AudioRecord.stop failed: " + e.getMessage());
       }
     }
 
@@ -122,52 +117,58 @@ class  WebRtcAudioRecord {
   }
 
   WebRtcAudioRecord(Context context, long nativeAudioRecord) {
-    Logd("ctor" + WebRtcAudioUtils.getThreadInfo());
+    Log.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
     this.context = context;
     this.nativeAudioRecord = nativeAudioRecord;
     if (DEBUG) {
       WebRtcAudioUtils.logDeviceInfo(TAG);
     }
+    effects = WebRtcAudioEffects.create();
   }
 
-  public static boolean BuiltInAECIsAvailable() {
-    // AcousticEchoCanceler was added in API level 16 (Jelly Bean).
-    if (!WebRtcAudioUtils.runningOnJellyBeanOrHigher()) {
+  private boolean enableBuiltInAEC(boolean enable) {
+    Log.d(TAG, "enableBuiltInAEC(" + enable + ')');
+    if (effects == null) {
+      Log.e(TAG,"Built-in AEC is not supported on this platform");
       return false;
     }
-    // TODO(henrika): add black-list based on device name. We could also
-    // use uuid to exclude devices but that would require a session ID from
-    // an existing AudioRecord object.
-    return AcousticEchoCanceler.isAvailable();
+    return effects.setAEC(enable);
   }
 
-  private boolean EnableBuiltInAEC(boolean enable) {
-    Logd("EnableBuiltInAEC(" + enable + ')');
-    // AcousticEchoCanceler was added in API level 16 (Jelly Bean).
-    if (!WebRtcAudioUtils.runningOnJellyBeanOrHigher()) {
+  private boolean enableBuiltInAGC(boolean enable) {
+    Log.d(TAG, "enableBuiltInAGC(" + enable + ')');
+    if (effects == null) {
+      Log.e(TAG,"Built-in AGC is not supported on this platform");
       return false;
     }
-    // Store the AEC state.
-    useBuiltInAEC = enable;
-    // Set AEC state if AEC has already been created.
-    if (aec != null) {
-      int ret = aec.setEnabled(enable);
-      if (ret != AudioEffect.SUCCESS) {
-        Loge("AcousticEchoCanceler.setEnabled failed");
-        return false;
-      }
-      Logd("AcousticEchoCanceler.getEnabled: " + aec.getEnabled());
-    }
-    return true;
+    return effects.setAGC(enable);
   }
 
-  private int InitRecording(int sampleRate, int channels) {
-    Logd("InitRecording(sampleRate=" + sampleRate + ", channels=" +
+  private boolean enableBuiltInNS(boolean enable) {
+    Log.d(TAG, "enableBuiltInNS(" + enable + ')');
+    if (effects == null) {
+      Log.e(TAG,"Built-in NS is not supported on this platform");
+      return false;
+    }
+    return effects.setNS(enable);
+  }
+
+  private int initRecording(int sampleRate, int channels) {
+    Log.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" +
         channels + ")");
+    if (!WebRtcAudioUtils.hasPermission(
+        context, android.Manifest.permission.RECORD_AUDIO)) {
+      Log.e(TAG,"RECORD_AUDIO permission is missing");
+      return -1;
+    }
+    if (audioRecord != null) {
+      Log.e(TAG,"InitRecording() called twice without StopRecording()");
+      return -1;
+    }
     final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
     final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
     byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
-    Logd("byteBuffer.capacity: " + byteBuffer.capacity());
+    Log.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
     // Rather than passing the ByteBuffer with every callback (requiring
     // the potentially expensive GetDirectBufferAddress) we simply have the
     // the native class cache the address to the memory once.
@@ -176,21 +177,23 @@ class  WebRtcAudioRecord {
     // Get the minimum buffer size required for the successful creation of
     // an AudioRecord object, in byte units.
     // Note that this size doesn't guarantee a smooth recording under load.
-    // TODO(henrika): Do we need to make this larger to avoid underruns?
     int minBufferSize = AudioRecord.getMinBufferSize(
           sampleRate,
           AudioFormat.CHANNEL_IN_MONO,
           AudioFormat.ENCODING_PCM_16BIT);
-    Logd("AudioRecord.getMinBufferSize: " + minBufferSize);
-
-    if (aec != null) {
-      aec.release();
-      aec = null;
+    if (minBufferSize == AudioRecord.ERROR
+        || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
+      Log.e(TAG, "AudioRecord.getMinBufferSize failed: " + minBufferSize);
+      return -1;
     }
-    assertTrue(audioRecord == null);
+    Log.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);
 
-    int bufferSizeInBytes = Math.max(byteBuffer.capacity(), minBufferSize);
-    Logd("bufferSizeInBytes: " + bufferSizeInBytes);
+    // Use a larger buffer size than the minimum required when creating the
+    // AudioRecord instance to ensure smooth recording under load. It has been
+    // verified that it does not increase the actual recording latency.
+    int bufferSizeInBytes =
+        Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
+    Log.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
 
     int audioSource = AudioSource.VOICE_COMMUNICATION;
     if (android.os.Build.VERSION.SDK_INT < 11) {
@@ -203,80 +206,77 @@ class  WebRtcAudioRecord {
                                     AudioFormat.CHANNEL_IN_MONO,
                                     AudioFormat.ENCODING_PCM_16BIT,
                                     bufferSizeInBytes);
-
     } catch (IllegalArgumentException e) {
-      Logd(e.getMessage());
+      Log.e(TAG,e.getMessage());
       return -1;
     }
-    assertTrue(audioRecord.getState() == AudioRecord.STATE_INITIALIZED);
+    if (audioRecord == null ||
+        audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
+      Log.e(TAG,"Failed to create a new AudioRecord instance");
+      return -1;
+    }
+    Log.d(TAG, "AudioRecord "
+        + "session ID: " + audioRecord.getAudioSessionId() + ", "
+        + "audio format: " + audioRecord.getAudioFormat() + ", "
+        + "channels: " + audioRecord.getChannelCount() + ", "
+        + "sample rate: " + audioRecord.getSampleRate());
+    if (effects != null) {
+      effects.enable(audioRecord.getAudioSessionId());
+    }
+    // TODO(phoglund): put back audioRecord.getBufferSizeInFrames when
+    // all known downstream users supports M.
+    // if (WebRtcAudioUtils.runningOnMOrHigher()) {
+      // Returns the frame count of the native AudioRecord buffer. This is
+      // greater than or equal to the bufferSizeInBytes converted to frame
+      // units. The native frame count may be enlarged to accommodate the
+      // requirements of the source on creation or if the AudioRecord is
+      // subsequently rerouted.
 
-    Logd("AudioRecord " +
-          "session ID: " + audioRecord.getAudioSessionId() + ", " +
-          "audio format: " + audioRecord.getAudioFormat() + ", " +
-          "channels: " + audioRecord.getChannelCount() + ", " +
-          "sample rate: " + audioRecord.getSampleRate());
-    Logd("AcousticEchoCanceler.isAvailable: " + BuiltInAECIsAvailable());
-    if (!BuiltInAECIsAvailable()) {
-      return framesPerBuffer;
-    }
-
-    aec = AcousticEchoCanceler.create(audioRecord.getAudioSessionId());
-    if (aec == null) {
-      Loge("AcousticEchoCanceler.create failed");
-      return -1;
-    }
-    int ret = aec.setEnabled(useBuiltInAEC);
-    if (ret != AudioEffect.SUCCESS) {
-      Loge("AcousticEchoCanceler.setEnabled failed");
-      return -1;
-    }
-    Descriptor descriptor = aec.getDescriptor();
-    Logd("AcousticEchoCanceler " +
-          "name: " + descriptor.name + ", " +
-          "implementor: " + descriptor.implementor + ", " +
-          "uuid: " + descriptor.uuid);
-    Logd("AcousticEchoCanceler.getEnabled: " + aec.getEnabled());
+      // Log.d(TAG, "bufferSizeInFrames: "
+      //     + audioRecord.getBufferSizeInFrames());
+    //}
     return framesPerBuffer;
   }
 
-  private boolean StartRecording() {
-    Logd("StartRecording");
+  private boolean startRecording() {
+    Log.d(TAG, "startRecording");
     assertTrue(audioRecord != null);
     assertTrue(audioThread == null);
+    try {
+      audioRecord.startRecording();
+    } catch (IllegalStateException e) {
+      Log.e(TAG,"AudioRecord.startRecording failed: " + e.getMessage());
+      return false;
+    }
+    if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
+      Log.e(TAG,"AudioRecord.startRecording failed");
+      return false;
+    }
     audioThread = new AudioRecordThread("AudioRecordJavaThread");
     audioThread.start();
     return true;
   }
 
-  private boolean StopRecording() {
-    Logd("StopRecording");
+  private boolean stopRecording() {
+    Log.d(TAG, "stopRecording");
     assertTrue(audioThread != null);
     audioThread.joinThread();
     audioThread = null;
-    if (aec != null) {
-      aec.release();
-      aec = null;
+    if (effects != null) {
+      effects.release();
     }
     audioRecord.release();
     audioRecord = null;
     return true;
   }
 
-  /** Helper method which throws an exception  when an assertion has failed. */
+  // Helper method which throws an exception  when an assertion has failed.
   private static void assertTrue(boolean condition) {
     if (!condition) {
       throw new AssertionError("Expected condition to be true");
     }
   }
 
-  private static void Logd(String msg) {
-    Log.d(TAG, msg);
-  }
-
-  private static void Loge(String msg) {
-    Log.e(TAG, msg);
-  }
-
   private native void nativeCacheDirectBufferAddress(
       ByteBuffer byteBuffer, long nativeAudioRecord);
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
index be8cd3bfef..a28c374dea 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
@@ -13,11 +13,13 @@ package org.webrtc.voiceengine;
 import java.lang.Thread;
 import java.nio.ByteBuffer;
 
+import android.annotation.TargetApi;
 import android.content.Context;
 import android.media.AudioFormat;
 import android.media.AudioManager;
 import android.media.AudioTrack;
 import android.os.Process;
+
 import android.util.Log;
 
 import org.mozilla.gecko.annotation.WebRTCJNITarget;
@@ -63,7 +65,7 @@ class WebRtcAudioTrack {
     @Override
     public void run() {
       Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
-      Logd("AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());
+      Log.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());
 
       try {
         // In MODE_STREAM mode we can optionally prime the output buffer by
@@ -73,7 +75,7 @@ class WebRtcAudioTrack {
         audioTrack.play();
         assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
       } catch (IllegalStateException e) {
-          Loge("AudioTrack.play failed: " + e.getMessage());
+          Log.e(TAG, "AudioTrack.play failed: " + e.getMessage());
         return;
       }
 
@@ -92,16 +94,12 @@ class WebRtcAudioTrack {
         assertTrue(sizeInBytes <= byteBuffer.remaining());
         int bytesWritten = 0;
         if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
-          bytesWritten = audioTrack.write(byteBuffer,
-                                          sizeInBytes,
-                                          AudioTrack.WRITE_BLOCKING);
+          bytesWritten = writeOnLollipop(audioTrack, byteBuffer, sizeInBytes);
         } else {
-          bytesWritten = audioTrack.write(byteBuffer.array(),
-                                          byteBuffer.arrayOffset(),
-                                          sizeInBytes);
+          bytesWritten = writePreLollipop(audioTrack, byteBuffer, sizeInBytes);
         }
         if (bytesWritten != sizeInBytes) {
-          Loge("AudioTrack.write failed: " + bytesWritten);
+          Log.e(TAG, "AudioTrack.write failed: " + bytesWritten);
           if (bytesWritten == AudioTrack.ERROR_INVALID_OPERATION) {
             keepAlive = false;
           }
@@ -119,12 +117,21 @@ class WebRtcAudioTrack {
       try {
         audioTrack.stop();
       } catch (IllegalStateException e) {
-        Loge("AudioTrack.stop failed: " + e.getMessage());
+        Log.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
       }
       assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_STOPPED);
       audioTrack.flush();
     }
 
+    @TargetApi(21)
+    private int writeOnLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
+      return audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
+    }
+
+    private int writePreLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
+      return audioTrack.write(byteBuffer.array(), byteBuffer.arrayOffset(), sizeInBytes);
+    }
+
     public void joinThread() {
       keepAlive = false;
       while (isAlive()) {
@@ -138,7 +145,7 @@ class WebRtcAudioTrack {
   }
 
   WebRtcAudioTrack(Context context, long nativeAudioTrack) {
-    Logd("ctor" + WebRtcAudioUtils.getThreadInfo());
+    Log.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
     this.context = context;
     this.nativeAudioTrack = nativeAudioTrack;
     audioManager = (AudioManager) context.getSystemService(
@@ -148,13 +155,13 @@ class WebRtcAudioTrack {
     }
   }
 
-  private int InitPlayout(int sampleRate, int channels) {
-    Logd("InitPlayout(sampleRate=" + sampleRate + ", channels=" +
-         channels + ")");
+  private void initPlayout(int sampleRate, int channels) {
+    Log.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels="
+        + channels + ")");
     final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
     byteBuffer = ByteBuffer.allocateDirect(
         bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
-    Logd("byteBuffer.capacity: " + byteBuffer.capacity());
+    Log.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
     // Rather than passing the ByteBuffer with every callback (requiring
     // the potentially expensive GetDirectBufferAddress) we simply have the
     // the native class cache the address to the memory once.
@@ -168,7 +175,7 @@ class WebRtcAudioTrack {
         sampleRate,
         AudioFormat.CHANNEL_OUT_MONO,
         AudioFormat.ENCODING_PCM_16BIT);
-    Logd("AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
+    Log.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
     assertTrue(audioTrack == null);
 
     // For the streaming mode, data must be written to the audio sink in
@@ -186,21 +193,16 @@ class WebRtcAudioTrack {
                                   minBufferSizeInBytes,
                                   AudioTrack.MODE_STREAM);
     } catch (IllegalArgumentException e) {
-      Logd(e.getMessage());
-      return -1;
+      Log.d(TAG, e.getMessage());
+      return;
     }
     assertTrue(audioTrack.getState() == AudioTrack.STATE_INITIALIZED);
     assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_STOPPED);
     assertTrue(audioTrack.getStreamType() == AudioManager.STREAM_VOICE_CALL);
-
-    // Return a delay estimate in milliseconds given the minimum buffer size.
-    // TODO(henrika): improve estimate and use real measurements of total
-    // latency instead. We can most likely ignore this value.
-    return (1000 * (minBufferSizeInBytes / bytesPerFrame) / sampleRate);
   }
 
-  private boolean StartPlayout() {
-    Logd("StartPlayout");
+  private boolean startPlayout() {
+    Log.d(TAG, "startPlayout");
     assertTrue(audioTrack != null);
     assertTrue(audioThread == null);
     audioThread = new AudioTrackThread("AudioTrackJavaThread");
@@ -208,8 +210,8 @@ class WebRtcAudioTrack {
     return true;
   }
 
-  private boolean StopPlayout() {
-    Logd("StopPlayout");
+  private boolean stopPlayout() {
+    Log.d(TAG, "stopPlayout");
     assertTrue(audioThread != null);
     audioThread.joinThread();
     audioThread = null;
@@ -221,29 +223,34 @@ class WebRtcAudioTrack {
   }
 
   /** Get max possible volume index for a phone call audio stream. */
-  private int GetStreamMaxVolume() {
-    Logd("GetStreamMaxVolume");
+  private int getStreamMaxVolume() {
+    Log.d(TAG, "getStreamMaxVolume");
     assertTrue(audioManager != null);
     return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
   }
 
   /** Set current volume level for a phone call audio stream. */
-  private boolean SetStreamVolume(int volume) {
-    Logd("SetStreamVolume(" + volume + ")");
+  private boolean setStreamVolume(int volume) {
+    Log.d(TAG, "setStreamVolume(" + volume + ")");
     assertTrue(audioManager != null);
-    if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
-      if (audioManager.isVolumeFixed()) {
-        Loge("The device implements a fixed volume policy.");
-        return false;
-      }
+    if (isVolumeFixed()) {
+      Log.e(TAG, "The device implements a fixed volume policy.");
+      return false;
     }
     audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
     return true;
   }
 
+  @TargetApi(21)
+  private boolean isVolumeFixed() {
+    if (!WebRtcAudioUtils.runningOnLollipopOrHigher())
+      return false;
+    return audioManager.isVolumeFixed();
+  }
+
   /** Get current volume level for a phone call audio stream. */
-  private int GetStreamVolume() {
-    Logd("GetStreamVolume");
+  private int getStreamVolume() {
+    Log.d(TAG, "getStreamVolume");
     assertTrue(audioManager != null);
     return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
   }
@@ -255,14 +262,6 @@ class WebRtcAudioTrack {
     }
   }
 
-  private static void Logd(String msg) {
-    Log.d(TAG, msg);
-  }
-
-  private static void Loge(String msg) {
-    Log.e(TAG, msg);
-  }
-
   private native void nativeCacheDirectBufferAddress(
       ByteBuffer byteBuffer, long nativeAudioRecord);
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
index 69d41e7094..369834158c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
@@ -10,35 +10,176 @@
 
 package org.webrtc.voiceengine;
 
-import java.lang.Thread;
-
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.media.audiofx.AcousticEchoCanceler;
+import android.media.audiofx.AudioEffect;
+import android.media.audiofx.AudioEffect.Descriptor;
 import android.media.AudioManager;
 import android.os.Build;
+import android.os.Process;
+
 import android.util.Log;
 
+import java.lang.Thread;
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.List;
+
 public final class WebRtcAudioUtils {
-  // Use 44.1kHz as the default sampling rate.
-  private static final int SAMPLE_RATE_HZ = 44100;
+  private static final String TAG = "WebRtcAudioUtils";
+
+  // List of devices where we have seen issues (e.g. bad audio quality) using
+  // the low latency output mode in combination with OpenSL ES.
+  // The device name is given by Build.MODEL.
+  private static final String[] BLACKLISTED_OPEN_SL_ES_MODELS = new String[] {
+    // This list is currently empty ;-)
+  };
+
+  // List of devices where it has been verified that the built-in effect
+  // bad and where it makes sense to avoid using it and instead rely on the
+  // native WebRTC version instead. The device name is given by Build.MODEL.
+  private static final String[] BLACKLISTED_AEC_MODELS = new String[] {
+      "Nexus 5",
+      "D6503",      // Sony Xperia Z2 D6503
+      "ONE A2005",  // OnePlus 2
+  };
+  private static final String[] BLACKLISTED_AGC_MODELS = new String[] {
+      "Nexus 10",
+      "Nexus 9",
+  };
+  private static final String[] BLACKLISTED_NS_MODELS = new String[] {
+      "Nexus 10",
+      "Nexus 9",
+      "Nexus 5",
+      "ONE A2005",  // OnePlus 2
+  };
+
+  // Use 16kHz as the default sample rate. A higher sample rate might prevent
+  // us from supporting communication mode on some older (e.g. ICS) devices.
+  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
+  private static int defaultSampleRateHz = DEFAULT_SAMPLE_RATE_HZ;
+  // Set to true if setDefaultSampleRateHz() has been called.
+  private static boolean isDefaultSampleRateOverridden = false;
+
+  // By default, utilize hardware based audio effects when available.
+  private static boolean useWebRtcBasedAcousticEchoCanceler = false;
+  private static boolean useWebRtcBasedAutomaticGainControl = false;
+  private static boolean useWebRtcBasedNoiseSuppressor = false;
+
+  // Call these methods if any hardware based effect shall be replaced by a
+  // software based version provided by the WebRTC stack instead.
+  public static synchronized void setWebRtcBasedAcousticEchoCanceler(
+      boolean enable) {
+    useWebRtcBasedAcousticEchoCanceler = enable;
+  }
+  public static synchronized void setWebRtcBasedAutomaticGainControl(
+      boolean enable) {
+    useWebRtcBasedAutomaticGainControl = enable;
+  }
+  public static synchronized void setWebRtcBasedNoiseSuppressor(
+      boolean enable) {
+    useWebRtcBasedNoiseSuppressor = enable;
+  }
+
+  public static synchronized boolean useWebRtcBasedAcousticEchoCanceler() {
+    if (useWebRtcBasedAcousticEchoCanceler) {
+      Log.w(TAG, "Overriding default behavior; now using WebRTC AEC!");
+    }
+    return useWebRtcBasedAcousticEchoCanceler;
+  }
+  public static synchronized boolean useWebRtcBasedAutomaticGainControl() {
+    if (useWebRtcBasedAutomaticGainControl) {
+      Log.w(TAG, "Overriding default behavior; now using WebRTC AGC!");
+    }
+    return useWebRtcBasedAutomaticGainControl;
+  }
+  public static synchronized boolean useWebRtcBasedNoiseSuppressor() {
+    if (useWebRtcBasedNoiseSuppressor) {
+      Log.w(TAG, "Overriding default behavior; now using WebRTC NS!");
+    }
+    return useWebRtcBasedNoiseSuppressor;
+  }
+
+  // Call this method if the default handling of querying the native sample
+  // rate shall be overridden. Can be useful on some devices where the
+  // available Android APIs are known to return invalid results.
+  public static synchronized void setDefaultSampleRateHz(int sampleRateHz) {
+    isDefaultSampleRateOverridden = true;
+    defaultSampleRateHz = sampleRateHz;
+  }
+
+  public static synchronized boolean isDefaultSampleRateOverridden() {
+    return isDefaultSampleRateOverridden;
+  }
+
+  public static synchronized int getDefaultSampleRateHz() {
+    return defaultSampleRateHz;
+  }
+
+  public static List<String> getBlackListedModelsForAecUsage() {
+    return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_AEC_MODELS);
+  }
+
+  public static List<String> getBlackListedModelsForAgcUsage() {
+    return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_AGC_MODELS);
+  }
+
+  public static List<String> getBlackListedModelsForNsUsage() {
+    return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_NS_MODELS);
+  }
+
+  public static boolean runningOnGingerBreadOrHigher() {
+    // November 2010: Android 2.3, API Level 9.
+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.GINGERBREAD;
+  }
 
   public static boolean runningOnJellyBeanOrHigher() {
+    // June 2012: Android 4.1. API Level 16.
     return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN;
   }
 
   public static boolean runningOnJellyBeanMR1OrHigher() {
+    // November 2012: Android 4.2. API Level 17.
     return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN_MR1;
   }
 
+  public static boolean runningOnJellyBeanMR2OrHigher() {
+    // July 24, 2013: Android 4.3. API Level 18.
+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN_MR2;
+  }
+
   public static boolean runningOnLollipopOrHigher() {
+    // API Level 21.
     return Build.VERSION.SDK_INT >= Build.VERSION_CODES.LOLLIPOP;
   }
 
-  /** Helper method for building a string of thread information.*/
+  // TODO(phoglund): enable when all downstream users use M.
+  // public static boolean runningOnMOrHigher() {
+    // API Level 23.
+    // return Build.VERSION.SDK_INT >= Build.VERSION_CODES.M;
+  //}
+
+  // Helper method for building a string of thread information.
   public static String getThreadInfo() {
     return "@[name=" + Thread.currentThread().getName()
         + ", id=" + Thread.currentThread().getId() + "]";
   }
 
-  /** Information about the current build, taken from system properties. */
+  // Returns true if we're running on emulator.
+  public static boolean runningOnEmulator() {
+    return Build.HARDWARE.equals("goldfish") &&
+        Build.BRAND.startsWith("generic_");
+  }
+
+  // Returns true if the device is blacklisted for OpenSL ES usage.
+  public static boolean deviceIsBlacklistedForOpenSLESUsage() {
+    List<String> blackListedModels =
+        Arrays.asList(BLACKLISTED_OPEN_SL_ES_MODELS);
+    return blackListedModels.contains(Build.MODEL);
+  }
+
+  // Information about the current build, taken from system properties.
   public static void logDeviceInfo(String tag) {
     Log.d(tag, "Android SDK: " + Build.VERSION.SDK_INT + ", "
         + "Release: " + Build.VERSION.RELEASE + ", "
@@ -50,4 +191,12 @@ public final class WebRtcAudioUtils {
         + "Model: " + Build.MODEL + ", "
         + "Product: " + Build.PRODUCT);
   }
+
+  // Checks if the process has as specified permission or not.
+  public static boolean hasPermission(Context context, String permission) {
+    return context.checkPermission(
+        permission,
+        Process.myPid(),
+        Process.myUid()) == PackageManager.PERMISSION_GRANTED;
+    }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event.h
deleted file mode 100644
index a19483d56c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_LOW_LATENCY_EVENT_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_LOW_LATENCY_EVENT_H_
-
-#include <errno.h>
-#include <limits.h>
-#include <fcntl.h>
-#include <stdio.h>
-#include <sys/ioctl.h>
-#include <sys/socket.h>
-#include <sys/types.h>
-
-namespace webrtc {
-
-// Implementation of event for single waiter, single signal threads. Event
-// is sticky.
-class LowLatencyEvent {
- public:
-  LowLatencyEvent();
-  ~LowLatencyEvent();
-
-  // Readies the event. Must be called before signaling or waiting for event.
-  // Returns true on success.
-  bool Start();
-  // Shuts down the event and releases threads calling WaitOnEvent. Once
-  // stopped SignalEvent and WaitOnEvent will have no effect. Start can be
-  // called to re-enable the event.
-  // Returns true on success.
-  bool Stop();
-
-  // Releases thread calling WaitOnEvent in a sticky fashion.
-  void SignalEvent(int event_id, int event_msg);
-  // Waits until SignalEvent or Stop is called.
-  void WaitOnEvent(int* event_id, int* event_msg);
-
- private:
-  typedef int Handle;
-  static const Handle kInvalidHandle;
-  static const int kReadHandle;
-  static const int kWriteHandle;
-
-  // Closes the handle. Returns true on success.
-  static bool Close(Handle* handle);
-
-  // SignalEvent and WaitOnEvent are actually read/write to file descriptors.
-  // Write is signal.
-  void WriteFd(int message_id, int message);
-  // Read is wait.
-  void ReadFd(int* message_id, int* message);
-
-  Handle handles_[2];
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_LOW_LATENCY_EVENT_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event_posix.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event_posix.cc
deleted file mode 100644
index f25b030d04..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event_posix.cc
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/android/low_latency_event.h"
-
-#include <assert.h>
-
-#define HANDLE_EINTR(x) ({                \
-      typeof(x) eintr_wrapper_result;     \
-      do {                                                        \
-        eintr_wrapper_result = (x);                               \
-      } while (eintr_wrapper_result == -1 && errno == EINTR);     \
-      eintr_wrapper_result;                                       \
-    })
-
-#define IGNORE_EINTR(x) ({                \
-      typeof(x) eintr_wrapper_result;     \
-      do {                                                        \
-        eintr_wrapper_result = (x);                               \
-        if (eintr_wrapper_result == -1 && errno == EINTR) {       \
-          eintr_wrapper_result = 0;                               \
-        }                                                         \
-      } while (0);                                                \
-      eintr_wrapper_result;                                       \
-    })
-
-namespace webrtc {
-
-const LowLatencyEvent::Handle LowLatencyEvent::kInvalidHandle = -1;
-const int LowLatencyEvent::kReadHandle = 0;
-const int LowLatencyEvent::kWriteHandle = 1;
-
-LowLatencyEvent::LowLatencyEvent() {
-  handles_[kReadHandle] = kInvalidHandle;
-  handles_[kWriteHandle] = kInvalidHandle;
-}
-
-LowLatencyEvent::~LowLatencyEvent() {
-  Stop();
-}
-
-bool LowLatencyEvent::Start() {
-  assert(handles_[kReadHandle] == kInvalidHandle);
-  assert(handles_[kWriteHandle] == kInvalidHandle);
-
-  return socketpair(AF_UNIX, SOCK_STREAM, 0, handles_) == 0;
-}
-
-bool LowLatencyEvent::Stop() {
-  bool ret = Close(&handles_[kReadHandle]) && Close(&handles_[kWriteHandle]);
-  handles_[kReadHandle] = kInvalidHandle;
-  handles_[kWriteHandle] = kInvalidHandle;
-  return ret;
-}
-
-void LowLatencyEvent::SignalEvent(int event_id, int event_msg) {
-  WriteFd(event_id, event_msg);
-}
-
-void LowLatencyEvent::WaitOnEvent(int* event_id, int* event_msg) {
-  ReadFd(event_id, event_msg);
-}
-
-bool LowLatencyEvent::Close(Handle* handle) {
-  if (*handle == kInvalidHandle) {
-    return false;
-  }
-  int retval = IGNORE_EINTR(close(*handle));
-  *handle = kInvalidHandle;
-  return retval == 0;
-}
-
-void LowLatencyEvent::WriteFd(int message_id, int message) {
-  char buffer[sizeof(message_id) + sizeof(message)];
-  size_t bytes = sizeof(buffer);
-  memcpy(buffer, &message_id, sizeof(message_id));
-  memcpy(&buffer[sizeof(message_id)], &message, sizeof(message));
-  ssize_t bytes_written = HANDLE_EINTR(write(handles_[kWriteHandle], buffer,
-                                             bytes));
-  if (bytes_written != static_cast<ssize_t>(bytes)) {
-    assert(false);
-  }
-}
-
-void LowLatencyEvent::ReadFd(int* message_id, int* message) {
-  char buffer[sizeof(message_id) + sizeof(message)];
-  size_t bytes = sizeof(buffer);
-  ssize_t bytes_read = HANDLE_EINTR(read(handles_[kReadHandle], buffer, bytes));
-  if (bytes_read == 0) {
-    *message_id = 0;
-    *message = 0;
-    return;
-  } else if (bytes_read == static_cast<ssize_t>(bytes)) {
-    memcpy(message_id, buffer, sizeof(*message_id));
-    memcpy(message, &buffer[sizeof(*message_id)], sizeof(*message));
-  } else {
-    assert(false);
-  }
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event_unittest.cc
deleted file mode 100644
index 2138f1f860..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/low_latency_event_unittest.cc
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/android/low_latency_event.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-namespace webrtc {
-
-static const int kEventMsg = 1;
-
-class LowLatencyEventTest : public testing::Test {
- public:
-  LowLatencyEventTest()
-      : process_thread_(ThreadWrapper::CreateThread(
-            CbThread, this, "test_thread")),
-        terminated_(false),
-        iteration_count_(0),
-        allowed_iterations_(0) {
-    EXPECT_TRUE(event_.Start());
-    Start();
-  }
-  ~LowLatencyEventTest() {
-    EXPECT_GE(allowed_iterations_, 1);
-    EXPECT_GE(iteration_count_, 1);
-    Stop();
-  }
-
-  void AllowOneIteration() {
-    ++allowed_iterations_;
-    event_.SignalEvent(allowed_iterations_, kEventMsg);
-  }
-
- private:
-  void Start() {
-    EXPECT_TRUE(process_thread_->Start());
-    process_thread_->SetPriority(kRealtimePriority);
-  }
-  void Stop() {
-    terminated_ = true;
-    event_.Stop();
-    process_thread_->Stop();
-  }
-
-  static bool CbThread(void* context) {
-    return reinterpret_cast<LowLatencyEventTest*>(context)->CbThreadImpl();
-  }
-  bool CbThreadImpl() {
-    int allowed_iterations;
-    int message;
-    ++iteration_count_;
-    event_.WaitOnEvent(&allowed_iterations, &message);
-    EXPECT_EQ(iteration_count_, allowed_iterations);
-    EXPECT_EQ(message, kEventMsg);
-    return !terminated_;
-  }
-
-  LowLatencyEvent event_;
-
-  rtc::scoped_ptr<ThreadWrapper> process_thread_;
-  bool terminated_;
-  int iteration_count_;
-  int allowed_iterations_;
-};
-
-
-TEST_F(LowLatencyEventTest, TriggerEvent) {
-  for (int i = 0; i < 3; ++i) {
-    AllowOneIteration();
-  }
-}
-
-// Events trigger in less than 3ms. Wait for 3 ms to ensure there are no
-// spurious wakeups.
-TEST_F(LowLatencyEventTest, NoTriggerEvent) {
-  SleepMs(3);
-  // If there were spurious wakeups either the wakeups would have triggered a
-  // failure as we haven't allowed an iteration yet. Or the wakeup happened
-  // to signal 0, 0 in which case the mismatch will be discovered when allowing
-  // an iteration to happen.
-  AllowOneIteration();
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.cc
index 9a16f7071c..da1e25483c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.cc
@@ -16,7 +16,7 @@
 
 using webrtc::kNumChannels;
 
-namespace webrtc_opensl {
+namespace webrtc {
 
 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) {
   SLDataFormat_PCM configuration;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.h
index daa51a2868..a4487b095c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_common.h
@@ -13,10 +13,42 @@
 
 #include <SLES/OpenSLES.h>
 
-namespace webrtc_opensl {
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
 
 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate);
 
+// Helper class for using SLObjectItf interfaces.
+template <typename SLType, typename SLDerefType>
+class ScopedSLObject {
+ public:
+  ScopedSLObject() : obj_(nullptr) {}
+
+  ~ScopedSLObject() { Reset(); }
+
+  SLType* Receive() {
+    RTC_DCHECK(!obj_);
+    return &obj_;
+  }
+
+  SLDerefType operator->() { return *obj_; }
+
+  SLType Get() const { return obj_; }
+
+  void Reset() {
+    if (obj_) {
+      (*obj_)->Destroy(obj_);
+      obj_ = nullptr;
+    }
+  }
+
+ private:
+  SLType obj_;
+};
+
+typedef ScopedSLObject<SLObjectItf, const SLObjectItf_*> ScopedSLObjectItf;
+
 }  // namespace webrtc_opensl
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.cc
deleted file mode 100644
index e01cf85a07..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.cc
+++ /dev/null
@@ -1,710 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/android/opensles_input.h"
-
-#include <assert.h>
-#include <dlfcn.h>
-
-#include "OpenSLESProvider.h"
-#include "webrtc/modules/audio_device/android/audio_common.h"
-#include "webrtc/modules/audio_device/android/opensles_common.h"
-#include "webrtc/modules/audio_device/android/single_rw_fifo.h"
-#include "webrtc/modules/audio_device/audio_device_buffer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-#include <media/AudioSystem.h>
-#include <audio_effects/effect_aec.h>
-#include <audio_effects/effect_ns.h>
-#include <utils/Errors.h>
-#endif
-
-#define VOID_RETURN
-#define OPENSL_RETURN_ON_FAILURE(op, ret_val)                    \
-  do {                                                           \
-    SLresult err = (op);                                         \
-    if (err != SL_RESULT_SUCCESS) {                              \
-      assert(false);                                             \
-      return ret_val;                                            \
-    }                                                            \
-  } while (0)
-
-static const SLEngineOption kOption[] = {
-  { SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE) },
-};
-
-enum {
-  kNoOverrun,
-  kOverrun,
-};
-
-namespace webrtc {
-
-OpenSlesInput::OpenSlesInput(
-    PlayoutDelayProvider* delay_provider, AudioManager* audio_manager)
-    : delay_provider_(delay_provider),
-      initialized_(false),
-      mic_initialized_(false),
-      rec_initialized_(false),
-      crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      recording_(false),
-      num_fifo_buffers_needed_(0),
-      number_overruns_(0),
-      sles_engine_(NULL),
-      sles_engine_itf_(NULL),
-      sles_recorder_(NULL),
-      sles_recorder_itf_(NULL),
-      sles_recorder_sbq_itf_(NULL),
-      audio_buffer_(NULL),
-      active_queue_(0),
-      rec_sampling_rate_(0),
-      agc_enabled_(false),
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-      aec_(NULL),
-      ns_(NULL),
-#endif
-      recording_delay_(0),
-      opensles_lib_(NULL) {
-}
-
-OpenSlesInput::~OpenSlesInput() {
-}
-
-int32_t OpenSlesInput::SetAndroidAudioDeviceObjects(void* javaVM,
-                                                    void* context) {
-#if !defined(WEBRTC_GONK)
-  AudioManagerJni::SetAndroidAudioDeviceObjects(javaVM, context);
-#endif
-  return 0;
-}
-
-void OpenSlesInput::ClearAndroidAudioDeviceObjects() {
-#if !defined(WEBRTC_GONK)
-  AudioManagerJni::ClearAndroidAudioDeviceObjects();
-#endif
-}
-
-int32_t OpenSlesInput::Init() {
-  assert(!initialized_);
-
-  /* Try to dynamically open the OpenSLES library */
-  opensles_lib_ = dlopen("libOpenSLES.so", RTLD_LAZY);
-  if (!opensles_lib_) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0,
-                   "  failed to dlopen OpenSLES library");
-      return -1;
-  }
-
-  f_slCreateEngine = (slCreateEngine_t)dlsym(opensles_lib_, "slCreateEngine");
-  SL_IID_ENGINE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ENGINE");
-  SL_IID_BUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_BUFFERQUEUE");
-  SL_IID_ANDROIDCONFIGURATION_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDCONFIGURATION");
-  SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE");
-  SL_IID_RECORD_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_RECORD");
-
-  if (!f_slCreateEngine ||
-      !SL_IID_ENGINE_ ||
-      !SL_IID_BUFFERQUEUE_ ||
-      !SL_IID_ANDROIDCONFIGURATION_ ||
-      !SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ ||
-      !SL_IID_RECORD_) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0,
-                   "  failed to find OpenSLES function");
-      return -1;
-  }
-
-  // Set up OpenSL engine.
-#ifndef MOZILLA_INTERNAL_API
-  OPENSL_RETURN_ON_FAILURE(f_slCreateEngine(&sles_engine_, 1, kOption, 0,
-                                            NULL, NULL),
-                           -1);
-#else
-  OPENSL_RETURN_ON_FAILURE(mozilla_get_sles_engine(&sles_engine_, 1, kOption), -1);
-#endif
-#ifndef MOZILLA_INTERNAL_API
-  OPENSL_RETURN_ON_FAILURE((*sles_engine_)->Realize(sles_engine_,
-                                                    SL_BOOLEAN_FALSE),
-                           -1);
-#else
-  OPENSL_RETURN_ON_FAILURE(mozilla_realize_sles_engine(sles_engine_), -1);
-#endif
-  OPENSL_RETURN_ON_FAILURE((*sles_engine_)->GetInterface(sles_engine_,
-                                                         SL_IID_ENGINE_,
-                                                         &sles_engine_itf_),
-                           -1);
-
-  if (InitSampleRate() != 0) {
-    return -1;
-  }
-  AllocateBuffers();
-  initialized_ = true;
-  return 0;
-}
-
-int32_t OpenSlesInput::Terminate() {
-  // It is assumed that the caller has stopped recording before terminating.
-  assert(!recording_);
-#ifndef MOZILLA_INTERNAL_API
-  (*sles_engine_)->Destroy(sles_engine_);
-#else
-  mozilla_destroy_sles_engine(&sles_engine_);
-#endif
-  initialized_ = false;
-  mic_initialized_ = false;
-  rec_initialized_ = false;
-  dlclose(opensles_lib_);
-  return 0;
-}
-
-int32_t OpenSlesInput::RecordingDeviceName(uint16_t index,
-                                           char name[kAdmMaxDeviceNameSize],
-                                           char guid[kAdmMaxGuidSize]) {
-  assert(index == 0);
-  // Empty strings.
-  name[0] = '\0';
-  guid[0] = '\0';
-  return 0;
-}
-
-int32_t OpenSlesInput::SetRecordingDevice(uint16_t index) {
-  assert(index == 0);
-  return 0;
-}
-
-int32_t OpenSlesInput::RecordingIsAvailable(bool& available) {  // NOLINT
-  available = true;
-  return 0;
-}
-
-int32_t OpenSlesInput::InitRecording() {
-  assert(initialized_);
-  rec_initialized_ = true;
-  return 0;
-}
-
-int32_t OpenSlesInput::StartRecording() {
-  assert(rec_initialized_);
-  assert(!recording_);
-  if (!CreateAudioRecorder()) {
-    return -1;
-  }
-  // Setup to receive buffer queue event callbacks.
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_sbq_itf_)->RegisterCallback(
-          sles_recorder_sbq_itf_,
-          RecorderSimpleBufferQueueCallback,
-          this),
-      -1);
-
-  if (!EnqueueAllBuffers()) {
-    return -1;
-  }
-
-  {
-    // To prevent the compiler from e.g. optimizing the code to
-    // recording_ = StartCbThreads() which wouldn't have been thread safe.
-    CriticalSectionScoped lock(crit_sect_.get());
-    recording_ = true;
-  }
-  if (!StartCbThreads()) {
-    recording_ = false;
-    return -1;
-  }
-  return 0;
-}
-
-int32_t OpenSlesInput::StopRecording() {
-  StopCbThreads();
-  DestroyAudioRecorder();
-  recording_ = false;
-  return 0;
-}
-
-int32_t OpenSlesInput::SetAGC(bool enable) {
-  agc_enabled_ = enable;
-  return 0;
-}
-
-int32_t OpenSlesInput::InitMicrophone() {
-  assert(initialized_);
-  assert(!recording_);
-  mic_initialized_ = true;
-  return 0;
-}
-
-int32_t OpenSlesInput::MicrophoneVolumeIsAvailable(bool& available) {  // NOLINT
-  available = false;
-  return 0;
-}
-
-int32_t OpenSlesInput::MinMicrophoneVolume(
-    uint32_t& minVolume) const {  // NOLINT
-  minVolume = 0;
-  return 0;
-}
-
-int32_t OpenSlesInput::MicrophoneVolumeStepSize(
-    uint16_t& stepSize) const {
-  stepSize = 1;
-  return 0;
-}
-
-int32_t OpenSlesInput::MicrophoneMuteIsAvailable(bool& available) {  // NOLINT
-  available = false;  // Mic mute not supported on Android
-  return 0;
-}
-
-int32_t OpenSlesInput::MicrophoneBoostIsAvailable(bool& available) {  // NOLINT
-  available = false;  // Mic boost not supported on Android.
-  return 0;
-}
-
-int32_t OpenSlesInput::SetMicrophoneBoost(bool enable) {
-  assert(false);
-  return -1;  // Not supported
-}
-
-int32_t OpenSlesInput::MicrophoneBoost(bool& enabled) const {  // NOLINT
-  assert(false);
-  return -1;  // Not supported
-}
-
-int32_t OpenSlesInput::StereoRecordingIsAvailable(bool& available) {  // NOLINT
-  available = false;  // Stereo recording not supported on Android.
-  return 0;
-}
-
-int32_t OpenSlesInput::SetStereoRecording(bool enable) {  // NOLINT
-  if (enable) {
-    return -1;
-  } else {
-    return 0;
-  }
-}
-
-int32_t OpenSlesInput::StereoRecording(bool& enabled) const {  // NOLINT
-  enabled = false;
-  return 0;
-}
-
-int32_t OpenSlesInput::RecordingDelay(uint16_t& delayMS) const {  // NOLINT
-  delayMS = recording_delay_;
-  return 0;
-}
-
-void OpenSlesInput::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
-  audio_buffer_ = audioBuffer;
-}
-
-int OpenSlesInput::InitSampleRate() {
-  UpdateSampleRate();
-  audio_buffer_->SetRecordingSampleRate(rec_sampling_rate_);
-  audio_buffer_->SetRecordingChannels(kNumChannels);
-  UpdateRecordingDelay();
-  return 0;
-}
-
-int OpenSlesInput::buffer_size_samples() const {
-  // Since there is no low latency recording, use buffer size corresponding to
-  // 10ms of data since that's the framesize WebRTC uses. Getting any other
-  // size would require patching together buffers somewhere before passing them
-  // to WebRTC.
-  return rec_sampling_rate_ * 10 / 1000;
-}
-
-int OpenSlesInput::buffer_size_bytes() const {
-  return buffer_size_samples() * kNumChannels * sizeof(int16_t);
-}
-
-void OpenSlesInput::UpdateRecordingDelay() {
-  // TODO(hellner): Add accurate delay estimate.
-  // On average half the current buffer will have been filled with audio.
-  int outstanding_samples =
-      (TotalBuffersUsed() - 0.5) * buffer_size_samples();
-  recording_delay_ = outstanding_samples / (rec_sampling_rate_ / 1000);
-}
-
-void OpenSlesInput::UpdateSampleRate() {
-#if !defined(WEBRTC_GONK)
-  rec_sampling_rate_ = audio_manager_.low_latency_supported() ?
-      audio_manager_.native_output_sample_rate() : kDefaultSampleRate;
-#else
-  rec_sampling_rate_ = kDefaultSampleRate;
-#endif
-}
-
-void OpenSlesInput::CalculateNumFifoBuffersNeeded() {
-  // Buffer size is 10ms of data.
-  num_fifo_buffers_needed_ = kNum10MsToBuffer;
-}
-
-void OpenSlesInput::AllocateBuffers() {
-  // Allocate FIFO to handle passing buffers between processing and OpenSL
-  // threads.
-  CalculateNumFifoBuffersNeeded();
-  assert(num_fifo_buffers_needed_ > 0);
-  fifo_.reset(new SingleRwFifo(num_fifo_buffers_needed_));
-
-  // Allocate the memory area to be used.
-  rec_buf_.reset(new rtc::scoped_ptr<int8_t[]>[TotalBuffersUsed()]);
-  for (int i = 0; i < TotalBuffersUsed(); ++i) {
-    rec_buf_[i].reset(new int8_t[buffer_size_bytes()]);
-  }
-}
-
-int OpenSlesInput::TotalBuffersUsed() const {
-  return num_fifo_buffers_needed_ + kNumOpenSlBuffers;
-}
-
-bool OpenSlesInput::EnqueueAllBuffers() {
-  active_queue_ = 0;
-  number_overruns_ = 0;
-  for (int i = 0; i < kNumOpenSlBuffers; ++i) {
-    memset(rec_buf_[i].get(), 0, buffer_size_bytes());
-    OPENSL_RETURN_ON_FAILURE(
-        (*sles_recorder_sbq_itf_)->Enqueue(
-            sles_recorder_sbq_itf_,
-            reinterpret_cast<void*>(rec_buf_[i].get()),
-            buffer_size_bytes()),
-        false);
-  }
-  // In case of underrun the fifo will be at capacity. In case of first enqueue
-  // no audio can have been returned yet meaning fifo must be empty. Any other
-  // values are unexpected.
-  assert(fifo_->size() == fifo_->capacity() ||
-         fifo_->size() == 0);
-  // OpenSL recording has been stopped. I.e. only this thread is touching
-  // |fifo_|.
-  while (fifo_->size() != 0) {
-    // Clear the fifo.
-    fifo_->Pop();
-  }
-  return true;
-}
-
-void OpenSlesInput::SetupVoiceMode() {
-  SLAndroidConfigurationItf configItf;
-  SLresult res = (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_ANDROIDCONFIGURATION_,
-                                                 (void*)&configItf);
-  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL GetInterface: %d", res);
-
-  if (res == SL_RESULT_SUCCESS) {
-    SLuint32 voiceMode = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
-    SLuint32 voiceSize = sizeof(voiceMode);
-
-    res = (*configItf)->SetConfiguration(configItf,
-                                         SL_ANDROID_KEY_RECORDING_PRESET,
-                                         &voiceMode, voiceSize);
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL Set Voice mode res: %d", res);
-  }
-}
-
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-bool OpenSlesInput::CheckPlatformAEC() {
-  effect_descriptor_t fxDesc;
-  uint32_t numFx;
-
-  if (android::AudioEffect::queryNumberEffects(&numFx) != android::NO_ERROR) {
-    return false;
-  }
-
-  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "Platform has %d effects", numFx);
-
-  for (uint32_t i = 0; i < numFx; i++) {
-    if (android::AudioEffect::queryEffect(i, &fxDesc) != android::NO_ERROR) {
-      continue;
-    }
-    if (memcmp(&fxDesc.type, FX_IID_AEC, sizeof(fxDesc.type)) == 0) {
-      return true;
-    }
-  }
-
-  return false;
-}
-
-void OpenSlesInput::SetupAECAndNS() {
-  bool hasAec = CheckPlatformAEC();
-  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "Platform has AEC: %d", hasAec);
-  // This code should not have been enabled if this fails, because it means the
-  // software AEC has will have been disabled as well. If you hit this, you need
-  // to fix your B2G config or fix the hardware AEC on your device.
-  assert(hasAec);
-
-  SLAndroidConfigurationItf configItf;
-  SLresult res = (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_ANDROIDCONFIGURATION_,
-                                                 (void*)&configItf);
-  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL GetInterface: %d", res);
-
-  if (res == SL_RESULT_SUCCESS) {
-    SLuint32 sessionId = 0;
-    SLuint32 idSize = sizeof(sessionId);
-    res = (*configItf)->GetConfiguration(configItf,
-                                         SL_ANDROID_KEY_RECORDING_SESSION_ID,
-                                         &idSize, &sessionId);
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL Get sessionId res: %d", res);
-
-    if (res == SL_RESULT_SUCCESS && idSize == sizeof(sessionId)) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL sessionId: %d", sessionId);
-
-      aec_ = new android::AudioEffect(FX_IID_AEC, NULL, 0, 0, 0, sessionId, 0);
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL aec: %p", aec_);
-
-      if (aec_) {
-        android::status_t status = aec_->initCheck();
-        if (status == android::NO_ERROR || status == android::ALREADY_EXISTS) {
-          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL aec enabled");
-          aec_->setEnabled(true);
-        } else {
-          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL aec disabled: %d", status);
-          delete aec_;
-          aec_ = NULL;
-        }
-      }
-
-      ns_ = new android::AudioEffect(FX_IID_NS, NULL, 0, 0, 0, sessionId, 0);
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL ns: %p", ns_);
-
-      if (ns_) {
-        android::status_t status = ns_->initCheck();
-        if (status == android::NO_ERROR || status == android::ALREADY_EXISTS) {
-          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL ns enabled");
-          ns_->setEnabled(true);
-        } else {
-          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, 0, "OpenSL ns disabled: %d", status);
-          delete ns_;
-          ns_ = NULL;
-        }
-      }
-    }
-  }
-}
-#endif
-
-bool OpenSlesInput::CreateAudioRecorder() {
-  if (!event_.Start()) {
-    assert(false);
-    return false;
-  }
-  SLDataLocator_IODevice micLocator = {
-    SL_DATALOCATOR_IODEVICE, SL_IODEVICE_AUDIOINPUT,
-    SL_DEFAULTDEVICEID_AUDIOINPUT, NULL };
-  SLDataSource audio_source = { &micLocator, NULL };
-
-  SLDataLocator_AndroidSimpleBufferQueue simple_buf_queue = {
-    SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
-    static_cast<SLuint32>(TotalBuffersUsed())
-  };
-  SLDataFormat_PCM configuration =
-      webrtc_opensl::CreatePcmConfiguration(rec_sampling_rate_);
-  SLDataSink audio_sink = { &simple_buf_queue, &configuration };
-
-  // Interfaces for recording android audio data and Android are needed.
-  // Note the interfaces still need to be initialized. This only tells OpenSl
-  // that the interfaces will be needed at some point.
-  const SLInterfaceID id[kNumInterfaces] = {
-    SL_IID_ANDROIDSIMPLEBUFFERQUEUE_, SL_IID_ANDROIDCONFIGURATION_ };
-  const SLboolean req[kNumInterfaces] = {
-    SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_engine_itf_)->CreateAudioRecorder(sles_engine_itf_,
-                                               &sles_recorder_,
-                                               &audio_source,
-                                               &audio_sink,
-                                               kNumInterfaces,
-                                               id,
-                                               req),
-      false);
-
-  SLAndroidConfigurationItf recorder_config;
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_)->GetInterface(sles_recorder_,
-                                      SL_IID_ANDROIDCONFIGURATION_,
-                                      &recorder_config),
-      false);
-
-  SetupVoiceMode();
-
-  // Realize the recorder in synchronous mode.
-  OPENSL_RETURN_ON_FAILURE((*sles_recorder_)->Realize(sles_recorder_,
-                                                      SL_BOOLEAN_FALSE),
-                           false);
-
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-  SetupAECAndNS();
-#endif
-
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_RECORD_,
-                                      static_cast<void*>(&sles_recorder_itf_)),
-      false);
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_)->GetInterface(
-          sles_recorder_,
-          SL_IID_ANDROIDSIMPLEBUFFERQUEUE_,
-          static_cast<void*>(&sles_recorder_sbq_itf_)),
-      false);
-  return true;
-}
-
-void OpenSlesInput::DestroyAudioRecorder() {
-  event_.Stop();
-
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-  delete aec_;
-  delete ns_;
-  aec_ = NULL;
-  ns_ = NULL;
-#endif
-
-  if (sles_recorder_sbq_itf_) {
-    // Release all buffers currently queued up.
-    OPENSL_RETURN_ON_FAILURE(
-        (*sles_recorder_sbq_itf_)->Clear(sles_recorder_sbq_itf_),
-        VOID_RETURN);
-    sles_recorder_sbq_itf_ = NULL;
-  }
-  sles_recorder_itf_ = NULL;
-
-  if (sles_recorder_) {
-    (*sles_recorder_)->Destroy(sles_recorder_);
-    sles_recorder_ = NULL;
-  }
-}
-
-bool OpenSlesInput::HandleOverrun(int event_id, int event_msg) {
-  if (!recording_) {
-    return false;
-  }
-  if (event_id == kNoOverrun) {
-    return false;
-  }
-  assert(event_id == kOverrun);
-  assert(event_msg > 0);
-  // Wait for all enqueued buffers be flushed.
-  if (event_msg != kNumOpenSlBuffers) {
-    return true;
-  }
-  // All buffers passed to OpenSL have been flushed. Restart the audio from
-  // scratch.
-  // No need to check sles_recorder_itf_ as recording_ would be false before it
-  // is set to NULL.
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_itf_)->SetRecordState(sles_recorder_itf_,
-                                            SL_RECORDSTATE_STOPPED),
-      true);
-  EnqueueAllBuffers();
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_itf_)->SetRecordState(sles_recorder_itf_,
-                                            SL_RECORDSTATE_RECORDING),
-      true);
-  return true;
-}
-
-void OpenSlesInput::RecorderSimpleBufferQueueCallback(
-    SLAndroidSimpleBufferQueueItf queue_itf,
-    void* context) {
-  OpenSlesInput* audio_device = reinterpret_cast<OpenSlesInput*>(context);
-  audio_device->RecorderSimpleBufferQueueCallbackHandler(queue_itf);
-}
-
-void OpenSlesInput::RecorderSimpleBufferQueueCallbackHandler(
-    SLAndroidSimpleBufferQueueItf queue_itf) {
-  if (fifo_->size() >= fifo_->capacity() || number_overruns_ > 0) {
-    ++number_overruns_;
-    event_.SignalEvent(kOverrun, number_overruns_);
-    return;
-  }
-  int8_t* audio = rec_buf_[active_queue_].get();
-  // There is at least one spot available in the fifo.
-  fifo_->Push(audio);
-  active_queue_ = (active_queue_ + 1) % TotalBuffersUsed();
-  event_.SignalEvent(kNoOverrun, 0);
-  // active_queue_ is indexing the next buffer to record to. Since the current
-  // buffer has been recorded it means that the buffer index
-  // kNumOpenSlBuffers - 1 past |active_queue_| contains the next free buffer.
-  // Since |fifo_| wasn't at capacity, at least one buffer is free to be used.
-  int next_free_buffer =
-      (active_queue_ + kNumOpenSlBuffers - 1) % TotalBuffersUsed();
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_sbq_itf_)->Enqueue(
-          sles_recorder_sbq_itf_,
-          reinterpret_cast<void*>(rec_buf_[next_free_buffer].get()),
-          buffer_size_bytes()),
-      VOID_RETURN);
-}
-
-bool OpenSlesInput::StartCbThreads() {
-  rec_thread_ = ThreadWrapper::CreateThread(CbThread, this,
-                                            "opensl_rec_thread");
-  assert(rec_thread_.get());
-  if (!rec_thread_->Start()) {
-    assert(false);
-    return false;
-  }
-  rec_thread_->SetPriority(kRealtimePriority);
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_itf_)->SetRecordState(sles_recorder_itf_,
-                                            SL_RECORDSTATE_RECORDING),
-      false);
-  return true;
-}
-
-void OpenSlesInput::StopCbThreads() {
-  {
-    CriticalSectionScoped lock(crit_sect_.get());
-    recording_ = false;
-  }
-  if (sles_recorder_itf_) {
-    OPENSL_RETURN_ON_FAILURE(
-        (*sles_recorder_itf_)->SetRecordState(sles_recorder_itf_,
-                                              SL_RECORDSTATE_STOPPED),
-        VOID_RETURN);
-  }
-  if (rec_thread_.get() == NULL) {
-    return;
-  }
-  event_.Stop();
-  if (rec_thread_->Stop()) {
-    rec_thread_.reset();
-  } else {
-    assert(false);
-  }
-}
-
-bool OpenSlesInput::CbThread(void* context) {
-  return reinterpret_cast<OpenSlesInput*>(context)->CbThreadImpl();
-}
-
-bool OpenSlesInput::CbThreadImpl() {
-  int event_id;
-  int event_msg;
-  // event_ must not be waited on while a lock has been taken.
-  event_.WaitOnEvent(&event_id, &event_msg);
-
-  CriticalSectionScoped lock(crit_sect_.get());
-  if (HandleOverrun(event_id, event_msg)) {
-    return recording_;
-  }
-  // If the fifo_ has audio data process it.
-  while (fifo_->size() > 0 && recording_) {
-    int8_t* audio = fifo_->Pop();
-    audio_buffer_->SetRecordedBuffer(audio, buffer_size_samples());
-    audio_buffer_->SetVQEData(delay_provider_ ?
-                              delay_provider_->PlayoutDelayMs() : 0,
-                              recording_delay_, 0);
-    audio_buffer_->DeliverRecordedData();
-  }
-  return recording_;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.h
deleted file mode 100644
index d89b71907d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.h
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_INPUT_H_
-
-#include <SLES/OpenSLES.h>
-#include <SLES/OpenSLES_Android.h>
-#include <SLES/OpenSLES_AndroidConfiguration.h>
-
-#include "webrtc/base/scoped_ptr.h"
-// Not defined in the android version we use to build with
-#define SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION ((SLuint32) 0x00000004)
-
-#if !defined(WEBRTC_GONK)
-#include "webrtc/modules/audio_device/android/audio_manager_jni.h"
-#else
-#include "media/AudioEffect.h"
-#endif
-#include "webrtc/modules/audio_device/android/low_latency_event.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/modules/audio_device/include/audio_device_defines.h"
-
-namespace webrtc {
-
-class AudioDeviceBuffer;
-class AudioManager;
-class CriticalSectionWrapper;
-class PlayoutDelayProvider;
-class SingleRwFifo;
-class ThreadWrapper;
-
-// OpenSL implementation that facilitate capturing PCM data from an android
-// device's microphone.
-// This class is Thread-compatible. I.e. Given an instance of this class, calls
-// to non-const methods require exclusive access to the object.
-class OpenSlesInput {
- public:
-  OpenSlesInput(
-      PlayoutDelayProvider* delay_provider, AudioManager* audio_manager);
-  ~OpenSlesInput();
-
-  static int32_t SetAndroidAudioDeviceObjects(void* javaVM,
-                                              void* context);
-  static void ClearAndroidAudioDeviceObjects();
-
-  // Main initializaton and termination
-  int32_t Init();
-  int32_t Terminate();
-  bool Initialized() const { return initialized_; }
-
-  // Device enumeration
-  int16_t RecordingDevices() { return 1; }
-  int32_t RecordingDeviceName(uint16_t index,
-                              char name[kAdmMaxDeviceNameSize],
-                              char guid[kAdmMaxGuidSize]);
-
-  // Device selection
-  int32_t SetRecordingDevice(uint16_t index);
-  int32_t SetRecordingDevice(
-      AudioDeviceModule::WindowsDeviceType device) { return -1; }
-
-  // No-op
-  int32_t SetRecordingSampleRate(uint32_t sample_rate_hz) { return 0; }
-
-  // Audio transport initialization
-  int32_t RecordingIsAvailable(bool& available);  // NOLINT
-  int32_t InitRecording();
-  bool RecordingIsInitialized() const { return rec_initialized_; }
-
-  // Audio transport control
-  int32_t StartRecording();
-  int32_t StopRecording();
-  bool Recording() const { return recording_; }
-
-  // Microphone Automatic Gain Control (AGC)
-  int32_t SetAGC(bool enable);
-  bool AGC() const { return agc_enabled_; }
-
-  // Audio mixer initialization
-  int32_t InitMicrophone();
-  bool MicrophoneIsInitialized() const { return mic_initialized_; }
-
-  // Microphone volume controls
-  int32_t MicrophoneVolumeIsAvailable(bool& available);  // NOLINT
-  // TODO(leozwang): Add microphone volume control when OpenSL APIs
-  // are available.
-  int32_t SetMicrophoneVolume(uint32_t volume) { return 0; }
-  int32_t MicrophoneVolume(uint32_t& volume) const { return -1; }  // NOLINT
-  int32_t MaxMicrophoneVolume(
-      uint32_t& maxVolume) const { return 0; }  // NOLINT
-  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;  // NOLINT
-  int32_t MicrophoneVolumeStepSize(
-      uint16_t& stepSize) const;  // NOLINT
-
-  // Microphone mute control
-  int32_t MicrophoneMuteIsAvailable(bool& available);  // NOLINT
-  int32_t SetMicrophoneMute(bool enable) { return -1; }
-  int32_t MicrophoneMute(bool& enabled) const { return -1; }  // NOLINT
-
-  // Microphone boost control
-  int32_t MicrophoneBoostIsAvailable(bool& available);  // NOLINT
-  int32_t SetMicrophoneBoost(bool enable);
-  int32_t MicrophoneBoost(bool& enabled) const;  // NOLINT
-
-  // Stereo support
-  int32_t StereoRecordingIsAvailable(bool& available);  // NOLINT
-  int32_t SetStereoRecording(bool enable);
-  int32_t StereoRecording(bool& enabled) const;  // NOLINT
-
-  // Delay information and control
-  int32_t RecordingDelay(uint16_t& delayMS) const;  // NOLINT
-
-  bool RecordingWarning() const { return false; }
-  bool RecordingError() const  { return false; }
-  void ClearRecordingWarning() {}
-  void ClearRecordingError() {}
-
-  // Attach audio buffer
-  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-  // Built-in AEC is only supported in combination with Java/AudioRecord.
-  bool BuiltInAECIsAvailable() const { return false; }
-  int32_t EnableBuiltInAEC(bool enable) { return -1; }
-
- private:
-  enum {
-    kNumInterfaces = 2,
-    // Keep as few OpenSL buffers as possible to avoid wasting memory. 2 is
-    // minimum for playout. Keep 2 for recording as well.
-    kNumOpenSlBuffers = 2,
-    kNum10MsToBuffer = 8,
-  };
-
-  int InitSampleRate();
-  int buffer_size_samples() const;
-  int buffer_size_bytes() const;
-  void UpdateRecordingDelay();
-  void UpdateSampleRate();
-  void CalculateNumFifoBuffersNeeded();
-  void AllocateBuffers();
-  int TotalBuffersUsed() const;
-  bool EnqueueAllBuffers();
-  // This function also configures the audio recorder, e.g. sample rate to use
-  // etc, so it should be called when starting recording.
-  bool CreateAudioRecorder();
-  void DestroyAudioRecorder();
-  void SetupVoiceMode();
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-  void SetupAECAndNS();
-  bool CheckPlatformAEC();
-#endif
-
-  // When overrun happens there will be more frames received from OpenSL than
-  // the desired number of buffers. It is possible to expand the number of
-  // buffers as you go but that would greatly increase the complexity of this
-  // code. HandleOverrun gracefully handles the scenario by restarting playout,
-  // throwing away all pending audio data. This will sound like a click. This
-  // is also logged to identify these types of clicks.
-  // This function returns true if there has been overrun. Further processing
-  // of audio data should be avoided until this function returns false again.
-  // The function needs to be protected by |crit_sect_|.
-  bool HandleOverrun(int event_id, int event_msg);
-
-  static void RecorderSimpleBufferQueueCallback(
-      SLAndroidSimpleBufferQueueItf queueItf,
-      void* pContext);
-  // This function must not take any locks or do any heavy work. It is a
-  // requirement for the OpenSL implementation to work as intended. The reason
-  // for this is that taking locks exposes the OpenSL thread to the risk of
-  // priority inversion.
-  void RecorderSimpleBufferQueueCallbackHandler(
-      SLAndroidSimpleBufferQueueItf queueItf);
-
-  bool StartCbThreads();
-  void StopCbThreads();
-  static bool CbThread(void* context);
-  // This function must be protected against data race with threads calling this
-  // class' public functions. It is a requirement for this class to be
-  // Thread-compatible.
-  bool CbThreadImpl();
-
-  PlayoutDelayProvider* delay_provider_;
-
-#if !defined(WEBRTC_GONK)
-  // Java API handle
-  AudioManagerJni audio_manager_;
-#endif
-
-  // TODO(henrika): improve this area
-  // PlayoutDelayProvider* delay_provider_;
-
-  bool initialized_;
-  bool mic_initialized_;
-  bool rec_initialized_;
-
-  // Members that are read/write accessed concurrently by the process thread and
-  // threads calling public functions of this class.
-  rtc::scoped_ptr<ThreadWrapper> rec_thread_;  // Processing thread
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  // This member controls the starting and stopping of recording audio to the
-  // the device.
-  bool recording_;
-
-  // Only one thread, T1, may push and only one thread, T2, may pull. T1 may or
-  // may not be the same thread as T2. T2 is the process thread and T1 is the
-  // OpenSL thread.
-  rtc::scoped_ptr<SingleRwFifo> fifo_;
-  int num_fifo_buffers_needed_;
-  LowLatencyEvent event_;
-  int number_overruns_;
-
-  // OpenSL handles
-  SLObjectItf sles_engine_;
-  SLEngineItf sles_engine_itf_;
-  SLObjectItf sles_recorder_;
-  SLRecordItf sles_recorder_itf_;
-  SLAndroidSimpleBufferQueueItf sles_recorder_sbq_itf_;
-
-  // Audio buffers
-  AudioDeviceBuffer* audio_buffer_;
-  // Holds all allocated memory such that it is deallocated properly.
-  rtc::scoped_ptr<rtc::scoped_ptr<int8_t[]>[]> rec_buf_;
-  // Index in |rec_buf_| pointing to the audio buffer that will be ready the
-  // next time RecorderSimpleBufferQueueCallbackHandler is invoked.
-  // Ready means buffer contains audio data from the device.
-  int active_queue_;
-
-  // Audio settings
-  uint32_t rec_sampling_rate_;
-  bool agc_enabled_;
-
-#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
-  android::AudioEffect* aec_;
-  android::AudioEffect* ns_;
-#endif
-  // Audio status
-  uint16_t recording_delay_;
-
-  // dlopen for OpenSLES
-  void *opensles_lib_;
-  typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
-                                       SLuint32,
-                                       const SLEngineOption *,
-                                       SLuint32,
-                                       const SLInterfaceID *,
-                                       const SLboolean *);
-  slCreateEngine_t f_slCreateEngine;
-  SLInterfaceID SL_IID_ENGINE_;
-  SLInterfaceID SL_IID_BUFFERQUEUE_;
-  SLInterfaceID SL_IID_ANDROIDCONFIGURATION_;
-  SLInterfaceID SL_IID_ANDROIDSIMPLEBUFFERQUEUE_;
-  SLInterfaceID SL_IID_RECORD_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_INPUT_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.cc
deleted file mode 100644
index 8d573df8fe..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.cc
+++ /dev/null
@@ -1,628 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifdef WEBRTC_ANDROID_OPENSLES_OUTPUT
-
-#include "webrtc/modules/audio_device/android/opensles_output.h"
-
-#include <assert.h>
-#include <dlfcn.h>
-
-#include "OpenSLESProvider.h"
-#include "webrtc/modules/audio_device/android/opensles_common.h"
-#include "webrtc/modules/audio_device/android/fine_audio_buffer.h"
-#include "webrtc/modules/audio_device/android/single_rw_fifo.h"
-#include "webrtc/modules/audio_device/audio_device_buffer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-#define VOID_RETURN
-#define OPENSL_RETURN_ON_FAILURE(op, ret_val)                    \
-  do {                                                           \
-    SLresult err = (op);                                         \
-    if (err != SL_RESULT_SUCCESS) {                              \
-      assert(false);                                             \
-      return ret_val;                                            \
-    }                                                            \
-  } while (0)
-
-static const SLEngineOption kOption[] = {
-  { SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE) },
-};
-
-enum {
-  kNoUnderrun,
-  kUnderrun,
-};
-
-namespace webrtc {
-
-OpenSlesOutput::OpenSlesOutput(AudioManager* audio_manager)
-    : initialized_(false),
-      speaker_initialized_(false),
-      play_initialized_(false),
-      crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      playing_(false),
-      num_fifo_buffers_needed_(0),
-      number_underruns_(0),
-      sles_engine_(NULL),
-      sles_engine_itf_(NULL),
-      sles_player_(NULL),
-      sles_player_itf_(NULL),
-      sles_player_sbq_itf_(NULL),
-      sles_output_mixer_(NULL),
-      audio_buffer_(NULL),
-      active_queue_(0),
-      speaker_sampling_rate_(kDefaultSampleRate),
-      buffer_size_samples_(0),
-      buffer_size_bytes_(0),
-      playout_delay_(0),
-      opensles_lib_(NULL) {
-}
-
-OpenSlesOutput::~OpenSlesOutput() {
-}
-
-int32_t OpenSlesOutput::SetAndroidAudioDeviceObjects(void* javaVM,
-                                                     void* context) {
-  AudioManagerJni::SetAndroidAudioDeviceObjects(javaVM, context);
-  return 0;
-}
-
-void OpenSlesOutput::ClearAndroidAudioDeviceObjects() {
-  AudioManagerJni::ClearAndroidAudioDeviceObjects();
-}
-
-int32_t OpenSlesOutput::Init() {
-  assert(!initialized_);
-
-  /* Try to dynamically open the OpenSLES library */
-  opensles_lib_ = dlopen("libOpenSLES.so", RTLD_LAZY);
-  if (!opensles_lib_) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-                   "  failed to dlopen OpenSLES library");
-      return -1;
-  }
-
-  f_slCreateEngine = (slCreateEngine_t)dlsym(opensles_lib_, "slCreateEngine");
-  SL_IID_ENGINE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ENGINE");
-  SL_IID_BUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_BUFFERQUEUE");
-  SL_IID_ANDROIDCONFIGURATION_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDCONFIGURATION");
-  SL_IID_PLAY_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_PLAY");
-  SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE");
-  SL_IID_VOLUME_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_VOLUME");
-
-  if (!f_slCreateEngine ||
-      !SL_IID_ENGINE_ ||
-      !SL_IID_BUFFERQUEUE_ ||
-      !SL_IID_ANDROIDCONFIGURATION_ ||
-      !SL_IID_PLAY_ ||
-      !SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ ||
-      !SL_IID_VOLUME_) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-                   "  failed to find OpenSLES function");
-      return -1;
-  }
-
-  // Set up OpenSl engine.
-#ifndef MOZILLA_INTERNAL_API
-  OPENSL_RETURN_ON_FAILURE(f_slCreateEngine(&sles_engine_, 1, kOption, 0,
-                                            NULL, NULL),
-                           -1);
-#else
-  OPENSL_RETURN_ON_FAILURE(mozilla_get_sles_engine(&sles_engine_, 1, kOption), -1);
-#endif
-#ifndef MOZILLA_INTERNAL_API
-  OPENSL_RETURN_ON_FAILURE((*sles_engine_)->Realize(sles_engine_,
-                                                    SL_BOOLEAN_FALSE),
-                           -1);
-#else
-  OPENSL_RETURN_ON_FAILURE(mozilla_realize_sles_engine(sles_engine_), -1);
-#endif
-  OPENSL_RETURN_ON_FAILURE((*sles_engine_)->GetInterface(sles_engine_,
-                                                         SL_IID_ENGINE_,
-                                                         &sles_engine_itf_),
-                           -1);
-  // Set up OpenSl output mix.
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_engine_itf_)->CreateOutputMix(sles_engine_itf_,
-                                           &sles_output_mixer_,
-                                           0,
-                                           NULL,
-                                           NULL),
-      -1);
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_output_mixer_)->Realize(sles_output_mixer_,
-                                     SL_BOOLEAN_FALSE),
-      -1);
-
-  if (!InitSampleRate()) {
-    return -1;
-  }
-  AllocateBuffers();
-  initialized_ = true;
-  return 0;
-}
-
-int32_t OpenSlesOutput::Terminate() {
-  // It is assumed that the caller has stopped recording before terminating.
-  assert(!playing_);
-  (*sles_output_mixer_)->Destroy(sles_output_mixer_);
-#ifndef MOZILLA_INTERNAL_API
-  (*sles_engine_)->Destroy(sles_engine_);
-#else
-  mozilla_destroy_sles_engine(&sles_engine_);
-#endif
-  initialized_ = false;
-  speaker_initialized_ = false;
-  play_initialized_ = false;
-  dlclose(opensles_lib_);
-  return 0;
-}
-
-int32_t OpenSlesOutput::PlayoutDeviceName(uint16_t index,
-                                          char name[kAdmMaxDeviceNameSize],
-                                          char guid[kAdmMaxGuidSize]) {
-  assert(index == 0);
-  // Empty strings.
-  name[0] = '\0';
-  guid[0] = '\0';
-  return 0;
-}
-
-int32_t OpenSlesOutput::SetPlayoutDevice(uint16_t index) {
-  assert(index == 0);
-  return 0;
-}
-
-int32_t OpenSlesOutput::PlayoutIsAvailable(bool& available) {  // NOLINT
-  available = true;
-  return 0;
-}
-
-int32_t OpenSlesOutput::InitPlayout() {
-  assert(initialized_);
-  play_initialized_ = true;
-  return 0;
-}
-
-int32_t OpenSlesOutput::StartPlayout() {
-  assert(play_initialized_);
-  assert(!playing_);
-  if (!CreateAudioPlayer()) {
-    return -1;
-  }
-
-  // Register callback to receive enqueued buffers.
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_sbq_itf_)->RegisterCallback(sles_player_sbq_itf_,
-                                                PlayerSimpleBufferQueueCallback,
-                                                this),
-      -1);
-  if (!EnqueueAllBuffers()) {
-    return -1;
-  }
-
-  {
-    // To prevent the compiler from e.g. optimizing the code to
-    // playing_ = StartCbThreads() which wouldn't have been thread safe.
-    CriticalSectionScoped lock(crit_sect_.get());
-    playing_ = true;
-  }
-  if (!StartCbThreads()) {
-    playing_ = false;
-  }
-  return 0;
-}
-
-int32_t OpenSlesOutput::StopPlayout() {
-  StopCbThreads();
-  DestroyAudioPlayer();
-  playing_ = false;
-  return 0;
-}
-
-int32_t OpenSlesOutput::InitSpeaker() {
-  assert(!playing_);
-  speaker_initialized_ = true;
-  return 0;
-}
-
-int32_t OpenSlesOutput::SpeakerVolumeIsAvailable(bool& available) {  // NOLINT
-  available = true;
-  return 0;
-}
-
-int32_t OpenSlesOutput::SetSpeakerVolume(uint32_t volume) {
-  assert(speaker_initialized_);
-  assert(initialized_);
-  // TODO(hellner): implement.
-  return 0;
-}
-
-int32_t OpenSlesOutput::MaxSpeakerVolume(uint32_t& maxVolume) const {  // NOLINT
-  assert(speaker_initialized_);
-  assert(initialized_);
-  // TODO(hellner): implement.
-  maxVolume = 0;
-  return 0;
-}
-
-int32_t OpenSlesOutput::MinSpeakerVolume(uint32_t& minVolume) const {  // NOLINT
-  assert(speaker_initialized_);
-  assert(initialized_);
-  // TODO(hellner): implement.
-  minVolume = 0;
-  return 0;
-}
-
-int32_t OpenSlesOutput::SpeakerVolumeStepSize(
-    uint16_t& stepSize) const {  // NOLINT
-  assert(speaker_initialized_);
-  stepSize = 1;
-  return 0;
-}
-
-int32_t OpenSlesOutput::SpeakerMuteIsAvailable(bool& available) {  // NOLINT
-  available = false;
-  return 0;
-}
-
-int32_t OpenSlesOutput::StereoPlayoutIsAvailable(bool& available) {  // NOLINT
-  available = false;
-  return 0;
-}
-
-int32_t OpenSlesOutput::SetStereoPlayout(bool enable) {
-  if (enable) {
-    assert(false);
-    return -1;
-  }
-  return 0;
-}
-
-int32_t OpenSlesOutput::StereoPlayout(bool& enabled) const {  // NOLINT
-  enabled = kNumChannels == 2;
-  return 0;
-}
-
-int32_t OpenSlesOutput::PlayoutBuffer(
-    AudioDeviceModule::BufferType& type,  // NOLINT
-    uint16_t& sizeMS) const {  // NOLINT
-  type = AudioDeviceModule::kAdaptiveBufferSize;
-  sizeMS = playout_delay_;
-  return 0;
-}
-
-int32_t OpenSlesOutput::PlayoutDelay(uint16_t& delayMS) const {  // NOLINT
-  delayMS = playout_delay_;
-  return 0;
-}
-
-void OpenSlesOutput::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
-  audio_buffer_ = audioBuffer;
-}
-
-int32_t OpenSlesOutput::SetLoudspeakerStatus(bool enable) {
-  return 0;
-}
-
-int32_t OpenSlesOutput::GetLoudspeakerStatus(bool& enabled) const {  // NOLINT
-  enabled = true;
-  return 0;
-}
-
-int OpenSlesOutput::PlayoutDelayMs() {
-  return playout_delay_;
-}
-
-bool OpenSlesOutput::InitSampleRate() {
-  if (!SetLowLatency()) {
-    speaker_sampling_rate_ = kDefaultSampleRate;
-    // Default is to use 10ms buffers.
-    buffer_size_samples_ = speaker_sampling_rate_ * 10 / 1000;
-  }
-  if (audio_buffer_->SetPlayoutSampleRate(speaker_sampling_rate_) < 0) {
-    return false;
-  }
-  if (audio_buffer_->SetPlayoutChannels(kNumChannels) < 0) {
-    return false;
-  }
-  UpdatePlayoutDelay();
-  return true;
-}
-
-void OpenSlesOutput::UpdatePlayoutDelay() {
-  // TODO(hellner): Add accurate delay estimate.
-  // On average half the current buffer will have been played out.
-  int outstanding_samples = (TotalBuffersUsed() - 0.5) * buffer_size_samples_;
-  playout_delay_ = outstanding_samples / (speaker_sampling_rate_ / 1000);
-}
-
-bool OpenSlesOutput::SetLowLatency() {
-#if !defined(WEBRTC_GONK)
-  if (!audio_manager_.low_latency_supported()) {
-    return false;
-  }
-  buffer_size_samples_ = audio_manager_.native_buffer_size();
-  assert(buffer_size_samples_ > 0);
-  speaker_sampling_rate_ = audio_manager_.native_output_sample_rate();
-  assert(speaker_sampling_rate_ > 0);
-  return true;
-#else
-  return false;
-#endif
-}
-
-void OpenSlesOutput::CalculateNumFifoBuffersNeeded() {
-  int number_of_bytes_needed =
-      (speaker_sampling_rate_ * kNumChannels * sizeof(int16_t)) * 10 / 1000;
-
-  // Ceiling of integer division: 1 + ((x - 1) / y)
-  int buffers_per_10_ms =
-      1 + ((number_of_bytes_needed - 1) / buffer_size_bytes_);
-  // |num_fifo_buffers_needed_| is a multiple of 10ms of buffered up audio.
-  num_fifo_buffers_needed_ = kNum10MsToBuffer * buffers_per_10_ms;
-}
-
-void OpenSlesOutput::AllocateBuffers() {
-  // Allocate fine buffer to provide frames of the desired size.
-  buffer_size_bytes_ = buffer_size_samples_ * kNumChannels * sizeof(int16_t);
-  fine_buffer_.reset(new FineAudioBuffer(audio_buffer_, buffer_size_bytes_,
-                                         speaker_sampling_rate_));
-
-  // Allocate FIFO to handle passing buffers between processing and OpenSl
-  // threads.
-  CalculateNumFifoBuffersNeeded();  // Needs |buffer_size_bytes_| to be known
-  assert(num_fifo_buffers_needed_ > 0);
-  fifo_.reset(new SingleRwFifo(num_fifo_buffers_needed_));
-
-  // Allocate the memory area to be used.
-  play_buf_.reset(new rtc::scoped_ptr<int8_t[]>[TotalBuffersUsed()]);
-  int required_buffer_size = fine_buffer_->RequiredBufferSizeBytes();
-  for (int i = 0; i < TotalBuffersUsed(); ++i) {
-    play_buf_[i].reset(new int8_t[required_buffer_size]);
-  }
-}
-
-int OpenSlesOutput::TotalBuffersUsed() const {
-  return num_fifo_buffers_needed_ + kNumOpenSlBuffers;
-}
-
-bool OpenSlesOutput::EnqueueAllBuffers() {
-  active_queue_ = 0;
-  number_underruns_ = 0;
-  for (int i = 0; i < kNumOpenSlBuffers; ++i) {
-    memset(play_buf_[i].get(), 0, buffer_size_bytes_);
-    OPENSL_RETURN_ON_FAILURE(
-        (*sles_player_sbq_itf_)->Enqueue(
-            sles_player_sbq_itf_,
-            reinterpret_cast<void*>(play_buf_[i].get()),
-            buffer_size_bytes_),
-        false);
-  }
-  // OpenSL playing has been stopped. I.e. only this thread is touching
-  // |fifo_|.
-  while (fifo_->size() != 0) {
-    // Underrun might have happened when pushing new buffers to the FIFO.
-    fifo_->Pop();
-  }
-  for (int i = kNumOpenSlBuffers; i < TotalBuffersUsed(); ++i) {
-    memset(play_buf_[i].get(), 0, buffer_size_bytes_);
-    fifo_->Push(play_buf_[i].get());
-  }
-  return true;
-}
-
-bool OpenSlesOutput::CreateAudioPlayer() {
-  if (!event_.Start()) {
-    assert(false);
-    return false;
-  }
-  SLDataLocator_AndroidSimpleBufferQueue simple_buf_queue = {
-    SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
-    static_cast<SLuint32>(kNumOpenSlBuffers)
-  };
-  SLDataFormat_PCM configuration =
-      webrtc_opensl::CreatePcmConfiguration(speaker_sampling_rate_);
-  SLDataSource audio_source = { &simple_buf_queue, &configuration };
-
-  SLDataLocator_OutputMix locator_outputmix;
-  // Setup the data sink structure.
-  locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
-  locator_outputmix.outputMix = sles_output_mixer_;
-  SLDataSink audio_sink = { &locator_outputmix, NULL };
-
-  // Interfaces for streaming audio data, setting volume and Android are needed.
-  // Note the interfaces still need to be initialized. This only tells OpenSl
-  // that the interfaces will be needed at some point.
-  SLInterfaceID ids[kNumInterfaces] = {
-    SL_IID_BUFFERQUEUE_, SL_IID_VOLUME_, SL_IID_ANDROIDCONFIGURATION_ };
-  SLboolean req[kNumInterfaces] = {
-    SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_engine_itf_)->CreateAudioPlayer(sles_engine_itf_, &sles_player_,
-                                             &audio_source, &audio_sink,
-                                             kNumInterfaces, ids, req),
-      false);
-
-  SLAndroidConfigurationItf player_config;
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_)->GetInterface(sles_player_,
-                                    SL_IID_ANDROIDCONFIGURATION_,
-                                    &player_config),
-      false);
-
-  // Set audio player configuration to SL_ANDROID_STREAM_VOICE which corresponds
-  // to android.media.AudioManager.STREAM_VOICE_CALL.
-  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
-  OPENSL_RETURN_ON_FAILURE(
-      (*player_config)->SetConfiguration(player_config,
-                                         SL_ANDROID_KEY_STREAM_TYPE,
-                                         &stream_type,
-                                         sizeof(SLint32)),
-      false);
-
-  // Realize the player in synchronous mode.
-  OPENSL_RETURN_ON_FAILURE((*sles_player_)->Realize(sles_player_,
-                                                    SL_BOOLEAN_FALSE),
-                           false);
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_)->GetInterface(sles_player_, SL_IID_PLAY_,
-                                    &sles_player_itf_),
-      false);
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_)->GetInterface(sles_player_, SL_IID_BUFFERQUEUE_,
-                                    &sles_player_sbq_itf_),
-      false);
-  return true;
-}
-
-void OpenSlesOutput::DestroyAudioPlayer() {
-  SLAndroidSimpleBufferQueueItf sles_player_sbq_itf = sles_player_sbq_itf_;
-  {
-    CriticalSectionScoped lock(crit_sect_.get());
-    sles_player_sbq_itf_ = NULL;
-    sles_player_itf_ = NULL;
-  }
-  event_.Stop();
-  if (sles_player_sbq_itf) {
-    // Release all buffers currently queued up.
-    OPENSL_RETURN_ON_FAILURE(
-        (*sles_player_sbq_itf)->Clear(sles_player_sbq_itf),
-        VOID_RETURN);
-  }
-
-  if (sles_player_) {
-    (*sles_player_)->Destroy(sles_player_);
-    sles_player_ = NULL;
-  }
-}
-
-bool OpenSlesOutput::HandleUnderrun(int event_id, int event_msg) {
-  if (!playing_) {
-    return false;
-  }
-  if (event_id == kNoUnderrun) {
-    return false;
-  }
-  assert(event_id == kUnderrun);
-  assert(event_msg > 0);
-  // Wait for all enqueued buffers to be flushed.
-  if (event_msg != kNumOpenSlBuffers) {
-    return true;
-  }
-  // All buffers have been flushed. Restart the audio from scratch.
-  // No need to check sles_player_itf_ as playing_ would be false before it is
-  // set to NULL.
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_itf_)->SetPlayState(sles_player_itf_,
-                                        SL_PLAYSTATE_STOPPED),
-      true);
-  EnqueueAllBuffers();
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_itf_)->SetPlayState(sles_player_itf_,
-                                        SL_PLAYSTATE_PLAYING),
-      true);
-  return true;
-}
-
-void OpenSlesOutput::PlayerSimpleBufferQueueCallback(
-    SLAndroidSimpleBufferQueueItf sles_player_sbq_itf,
-    void* p_context) {
-  OpenSlesOutput* audio_device = reinterpret_cast<OpenSlesOutput*>(p_context);
-  audio_device->PlayerSimpleBufferQueueCallbackHandler(sles_player_sbq_itf);
-}
-
-void OpenSlesOutput::PlayerSimpleBufferQueueCallbackHandler(
-    SLAndroidSimpleBufferQueueItf sles_player_sbq_itf) {
-  if (fifo_->size() <= 0 || number_underruns_ > 0) {
-    ++number_underruns_;
-    event_.SignalEvent(kUnderrun, number_underruns_);
-    return;
-  }
-  int8_t* audio = fifo_->Pop();
-  if (audio)
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_sbq_itf)->Enqueue(sles_player_sbq_itf,
-                                      audio,
-                                      buffer_size_bytes_),
-      VOID_RETURN);
-  event_.SignalEvent(kNoUnderrun, 0);
-}
-
-bool OpenSlesOutput::StartCbThreads() {
-  play_thread_ = ThreadWrapper::CreateThread(CbThread, this,
-                                             "opensl_play_thread");
-  assert(play_thread_.get());
-  OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_itf_)->SetPlayState(sles_player_itf_,
-                                        SL_PLAYSTATE_PLAYING),
-      false);
-
-  if (!play_thread_->Start()) {
-    assert(false);
-    return false;
-  }
-  play_thread_->SetPriority(kRealtimePriority);
-  return true;
-}
-
-void OpenSlesOutput::StopCbThreads() {
-  {
-    CriticalSectionScoped lock(crit_sect_.get());
-    playing_ = false;
-  }
-  if (sles_player_itf_) {
-    OPENSL_RETURN_ON_FAILURE(
-        (*sles_player_itf_)->SetPlayState(sles_player_itf_,
-                                          SL_PLAYSTATE_STOPPED),
-        VOID_RETURN);
-  }
-  if (play_thread_.get() == NULL) {
-    return;
-  }
-  event_.Stop();
-  if (play_thread_->Stop()) {
-    play_thread_.reset();
-  } else {
-    assert(false);
-  }
-}
-
-bool OpenSlesOutput::CbThread(void* context) {
-  return reinterpret_cast<OpenSlesOutput*>(context)->CbThreadImpl();
-}
-
-bool OpenSlesOutput::CbThreadImpl() {
-  assert(fine_buffer_.get() != NULL);
-  int event_id;
-  int event_msg;
-  // event_ must not be waited on while a lock has been taken.
-  event_.WaitOnEvent(&event_id, &event_msg);
-
-  CriticalSectionScoped lock(crit_sect_.get());
-  if (HandleUnderrun(event_id, event_msg)) {
-    return playing_;
-  }
-  // if fifo_ is not full it means next item in memory must be free.
-  while (fifo_->size() < num_fifo_buffers_needed_ && playing_) {
-    int8_t* audio = play_buf_[active_queue_].get();
-    fine_buffer_->GetBufferData(audio);
-    fifo_->Push(audio);
-    active_queue_ = (active_queue_ + 1) % TotalBuffersUsed();
-  }
-  return playing_;
-}
-
-}  // namespace webrtc
-
-#endif
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.h
deleted file mode 100644
index 148e483da8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.h
+++ /dev/null
@@ -1,459 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_OUTPUT_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_OUTPUT_H_
-
-#include <SLES/OpenSLES.h>
-#include <SLES/OpenSLES_Android.h>
-#include <SLES/OpenSLES_AndroidConfiguration.h>
-
-#include "webrtc/base/scoped_ptr.h"
-#if !defined(WEBRTC_GONK)
-#include "webrtc/modules/audio_device/android/audio_manager.h"
-#include "webrtc/modules/audio_device/android/audio_manager_jni.h"
-#endif
-#include "webrtc/modules/audio_device/android/low_latency_event.h"
-#include "webrtc/modules/audio_device/android/audio_common.h"
-#include "webrtc/modules/audio_device/include/audio_device_defines.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-namespace webrtc {
-
-class AudioDeviceBuffer;
-class CriticalSectionWrapper;
-class FineAudioBuffer;
-class SingleRwFifo;
-class ThreadWrapper;
-
-#if defined(WEBRTC_ANDROID_OPENSLES_OUTPUT)
-// allow us to replace it with a dummy
-
-// OpenSL implementation that facilitate playing PCM data to an android device.
-// This class is Thread-compatible. I.e. Given an instance of this class, calls
-// to non-const methods require exclusive access to the object.
-class OpenSlesOutput : public PlayoutDelayProvider {
- public:
-  // TODO(henrika): use this new audio manager instead of old.
-  explicit OpenSlesOutput(AudioManager* audio_manager);
-  virtual ~OpenSlesOutput();
-
-  static int32_t SetAndroidAudioDeviceObjects(void* javaVM,
-                                              void* context);
-  static void ClearAndroidAudioDeviceObjects();
-
-  // Main initializaton and termination
-  int32_t Init();
-  int32_t Terminate();
-  bool Initialized() const { return initialized_; }
-
-  // Device enumeration
-  int16_t PlayoutDevices() { return 1; }
-
-  int32_t PlayoutDeviceName(uint16_t index,
-                            char name[kAdmMaxDeviceNameSize],
-                            char guid[kAdmMaxGuidSize]);
-
-  // Device selection
-  int32_t SetPlayoutDevice(uint16_t index);
-  int32_t SetPlayoutDevice(
-      AudioDeviceModule::WindowsDeviceType device) { return 0; }
-
-  // No-op
-  int32_t SetPlayoutSampleRate(uint32_t sample_rate_hz) { return 0; }
-
-  // Audio transport initialization
-  int32_t PlayoutIsAvailable(bool& available);  // NOLINT
-  int32_t InitPlayout();
-  bool PlayoutIsInitialized() const { return play_initialized_; }
-
-  // Audio transport control
-  int32_t StartPlayout();
-  int32_t StopPlayout();
-  bool Playing() const { return playing_; }
-
-  // Audio mixer initialization
-  int32_t InitSpeaker();
-  bool SpeakerIsInitialized() const { return speaker_initialized_; }
-
-  // Speaker volume controls
-  int32_t SpeakerVolumeIsAvailable(bool& available);  // NOLINT
-  int32_t SetSpeakerVolume(uint32_t volume);
-  int32_t SpeakerVolume(uint32_t& volume) const { return 0; }  // NOLINT
-  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;  // NOLINT
-  int32_t MinSpeakerVolume(uint32_t& minVolume) const;  // NOLINT
-  int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;  // NOLINT
-
-  // Speaker mute control
-  int32_t SpeakerMuteIsAvailable(bool& available);  // NOLINT
-  int32_t SetSpeakerMute(bool enable) { return -1; }
-  int32_t SpeakerMute(bool& enabled) const { return -1; }  // NOLINT
-
-
-  // Stereo support
-  int32_t StereoPlayoutIsAvailable(bool& available);  // NOLINT
-  int32_t SetStereoPlayout(bool enable);
-  int32_t StereoPlayout(bool& enabled) const;  // NOLINT
-
-  // Delay information and control
-  int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                                   uint16_t sizeMS) { return -1; }
-  int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,  // NOLINT
-                        uint16_t& sizeMS) const;
-  int32_t PlayoutDelay(uint16_t& delayMS) const;  // NOLINT
-
-
-  // Error and warning information
-  bool PlayoutWarning() const { return false; }
-  bool PlayoutError() const { return false; }
-  void ClearPlayoutWarning() {}
-  void ClearPlayoutError() {}
-
-  // Attach audio buffer
-  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-  // Speaker audio routing
-  int32_t SetLoudspeakerStatus(bool enable);
-  int32_t GetLoudspeakerStatus(bool& enable) const;  // NOLINT
-
- protected:
-  virtual int PlayoutDelayMs();
-
- private:
-  enum {
-    kNumInterfaces = 3,
-    // TODO(xians): Reduce the numbers of buffers to improve the latency.
-    //              Currently 30ms worth of buffers are needed due to audio
-    //              pipeline processing jitter. Note: kNumOpenSlBuffers must
-    //              not be changed.
-    // According to the opensles documentation in the ndk:
-    // The lower output latency path is used only if the application requests a
-    // buffer count of 2 or more. Use minimum number of buffers to keep delay
-    // as low as possible.
-    kNumOpenSlBuffers = 2,
-    // NetEq delivers frames on a 10ms basis. This means that every 10ms there
-    // will be a time consuming task. Keeping 10ms worth of buffers will ensure
-    // that there is 10ms to perform the time consuming task without running
-    // into underflow.
-    // In addition to the 10ms that needs to be stored for NetEq processing
-    // there will be jitter in audio pipe line due to the acquisition of locks.
-    // Note: The buffers in the OpenSL queue do not count towards the 10ms of
-    // frames needed since OpenSL needs to have them ready for playout.
-    kNum10MsToBuffer = 6,
-  };
-
-  bool InitSampleRate();
-  bool SetLowLatency();
-  void UpdatePlayoutDelay();
-  // It might be possible to dynamically add or remove buffers based on how
-  // close to depletion the fifo is. Few buffers means low delay. Too few
-  // buffers will cause underrun. Dynamically changing the number of buffer
-  // will greatly increase code complexity.
-  void CalculateNumFifoBuffersNeeded();
-  void AllocateBuffers();
-  int TotalBuffersUsed() const;
-  bool EnqueueAllBuffers();
-  // This function also configures the audio player, e.g. sample rate to use
-  // etc, so it should be called when starting playout.
-  bool CreateAudioPlayer();
-  void DestroyAudioPlayer();
-
-  // When underrun happens there won't be a new frame ready for playout that
-  // can be retrieved yet. Since the OpenSL thread must return ASAP there will
-  // be one less queue available to OpenSL. This function handles this case
-  // gracefully by restarting the audio, pushing silent frames to OpenSL for
-  // playout. This will sound like a click. Underruns are also logged to
-  // make it possible to identify these types of audio artifacts.
-  // This function returns true if there has been underrun. Further processing
-  // of audio data should be avoided until this function returns false again.
-  // The function needs to be protected by |crit_sect_|.
-  bool HandleUnderrun(int event_id, int event_msg);
-
-  static void PlayerSimpleBufferQueueCallback(
-      SLAndroidSimpleBufferQueueItf queueItf,
-      void* pContext);
-  // This function must not take any locks or do any heavy work. It is a
-  // requirement for the OpenSL implementation to work as intended. The reason
-  // for this is that taking locks exposes the OpenSL thread to the risk of
-  // priority inversion.
-  void PlayerSimpleBufferQueueCallbackHandler(
-      SLAndroidSimpleBufferQueueItf queueItf);
-
-  bool StartCbThreads();
-  void StopCbThreads();
-  static bool CbThread(void* context);
-  // This function must be protected against data race with threads calling this
-  // class' public functions. It is a requirement for this class to be
-  // Thread-compatible.
-  bool CbThreadImpl();
-
-#if !defined(WEBRTC_GONK)
-  // Java API handle
-  AudioManagerJni audio_manager_;
-#endif
-
-  bool initialized_;
-  bool speaker_initialized_;
-  bool play_initialized_;
-
-  // Members that are read/write accessed concurrently by the process thread and
-  // threads calling public functions of this class.
-  rtc::scoped_ptr<ThreadWrapper> play_thread_;  // Processing thread
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  // This member controls the starting and stopping of playing audio to the
-  // the device.
-  bool playing_;
-
-  // Only one thread, T1, may push and only one thread, T2, may pull. T1 may or
-  // may not be the same thread as T2. T1 is the process thread and T2 is the
-  // OpenSL thread.
-  rtc::scoped_ptr<SingleRwFifo> fifo_;
-  int num_fifo_buffers_needed_;
-  LowLatencyEvent event_;
-  int number_underruns_;
-
-  // OpenSL handles
-  SLObjectItf sles_engine_;
-  SLEngineItf sles_engine_itf_;
-  SLObjectItf sles_player_;
-  SLPlayItf sles_player_itf_;
-  SLAndroidSimpleBufferQueueItf sles_player_sbq_itf_;
-  SLObjectItf sles_output_mixer_;
-
-  // Audio buffers
-  AudioDeviceBuffer* audio_buffer_;
-  rtc::scoped_ptr<FineAudioBuffer> fine_buffer_;
-  rtc::scoped_ptr<rtc::scoped_ptr<int8_t[]>[]> play_buf_;
-  // Index in |rec_buf_| pointing to the audio buffer that will be ready the
-  // next time PlayerSimpleBufferQueueCallbackHandler is invoked.
-  // Ready means buffer is ready to be played out to device.
-  int active_queue_;
-
-  // Audio settings
-  uint32_t speaker_sampling_rate_;
-  int buffer_size_samples_;
-  int buffer_size_bytes_;
-
-  // Audio status
-  uint16_t playout_delay_;
-
-  // dlopen for OpenSLES
-  void *opensles_lib_;
-  typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
-                                       SLuint32,
-                                       const SLEngineOption *,
-                                       SLuint32,
-                                       const SLInterfaceID *,
-                                       const SLboolean *);
-  slCreateEngine_t f_slCreateEngine;
-  SLInterfaceID SL_IID_ENGINE_;
-  SLInterfaceID SL_IID_BUFFERQUEUE_;
-  SLInterfaceID SL_IID_ANDROIDCONFIGURATION_;
-  SLInterfaceID SL_IID_PLAY_;
-  SLInterfaceID SL_IID_ANDROIDSIMPLEBUFFERQUEUE_;
-  SLInterfaceID SL_IID_VOLUME_;
-};
-
-#else
-
-// Dummy OpenSlesOutput
-class OpenSlesOutput : public PlayoutDelayProvider {
- public:
-  explicit OpenSlesOutput(AudioManager* audio_manager) :
-    initialized_(false), speaker_initialized_(false),
-    play_initialized_(false), playing_(false)
-  {}
-  virtual ~OpenSlesOutput() {}
-
-  static int32_t SetAndroidAudioDeviceObjects(void* javaVM,
-                                              void* context) { return 0; }
-  static void ClearAndroidAudioDeviceObjects() {}
-
-  // Main initializaton and termination
-  int32_t Init() { initialized_ = true; return 0; }
-  int32_t Terminate() { initialized_ = false; return 0; }
-  bool Initialized() const { return initialized_; }
-
-  // Device enumeration
-  int16_t PlayoutDevices() { return 1; }
-
-  int32_t PlayoutDeviceName(uint16_t index,
-                            char name[kAdmMaxDeviceNameSize],
-                            char guid[kAdmMaxGuidSize])
-  {
-    assert(index == 0);
-    // Empty strings.
-    name[0] = '\0';
-    guid[0] = '\0';
-    return 0;
-  }
-
-  // Device selection
-  int32_t SetPlayoutDevice(uint16_t index)
-  {
-    assert(index == 0);
-    return 0;
-  }
-  int32_t SetPlayoutDevice(
-      AudioDeviceModule::WindowsDeviceType device) { return 0; }
-
-  // No-op
-  int32_t SetPlayoutSampleRate(uint32_t sample_rate_hz) { return 0; }
-
-  // Audio transport initialization
-  int32_t PlayoutIsAvailable(bool& available)  // NOLINT
-  {
-    available = true;
-    return 0;
-  }
-  int32_t InitPlayout()
-  {
-    assert(initialized_);
-    play_initialized_ = true;
-    return 0;
-  }
-  bool PlayoutIsInitialized() const { return play_initialized_; }
-
-  // Audio transport control
-  int32_t StartPlayout()
-  {
-    assert(play_initialized_);
-    assert(!playing_);
-    playing_ = true;
-    return 0;
-  }
-
-  int32_t StopPlayout()
-  {
-    playing_ = false;
-    return 0;
-  }
-
-  bool Playing() const { return playing_; }
-
-  // Audio mixer initialization
-  int32_t SpeakerIsAvailable(bool& available)  // NOLINT
-  {
-    available = true;
-    return 0;
-  }
-  int32_t InitSpeaker()
-  {
-    assert(!playing_);
-    speaker_initialized_ = true;
-    return 0;
-  }
-  bool SpeakerIsInitialized() const { return speaker_initialized_; }
-
-  // Speaker volume controls
-  int32_t SpeakerVolumeIsAvailable(bool& available)  // NOLINT
-  {
-    available = true;
-    return 0;
-  }
-  int32_t SetSpeakerVolume(uint32_t volume)
-  {
-    assert(speaker_initialized_);
-    assert(initialized_);
-    return 0;
-  }
-  int32_t SpeakerVolume(uint32_t& volume) const { return 0; }  // NOLINT
-  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const  // NOLINT
-  {
-    assert(speaker_initialized_);
-    assert(initialized_);
-    maxVolume = 0;
-    return 0;
-  }
-  int32_t MinSpeakerVolume(uint32_t& minVolume) const  // NOLINT
-  {
-    assert(speaker_initialized_);
-    assert(initialized_);
-    minVolume = 0;
-    return 0;
-  }
-  int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const  // NOLINT
-  {
-    assert(speaker_initialized_);
-    assert(initialized_);
-    stepSize = 0;
-    return 0;
-  }
-
-  // Speaker mute control
-  int32_t SpeakerMuteIsAvailable(bool& available)  // NOLINT
-  {
-    available = true;
-    return 0;
-  }
-  int32_t SetSpeakerMute(bool enable) { return -1; }
-  int32_t SpeakerMute(bool& enabled) const { return -1; }  // NOLINT
-
-
-  // Stereo support
-  int32_t StereoPlayoutIsAvailable(bool& available)  // NOLINT
-  {
-    available = true;
-    return 0;
-  }
-  int32_t SetStereoPlayout(bool enable)
-  {
-    return 0;
-  }
-  int32_t StereoPlayout(bool& enabled) const  // NOLINT
-  {
-    enabled = kNumChannels == 2;
-    return 0;
-  }
-
-  // Delay information and control
-  int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                                   uint16_t sizeMS) { return -1; }
-  int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,  // NOLINT
-                        uint16_t& sizeMS) const
-  {
-    type = AudioDeviceModule::kAdaptiveBufferSize;
-    sizeMS = 40;
-    return 0;
-  }
-  int32_t PlayoutDelay(uint16_t& delayMS) const  // NOLINT
-  {
-    delayMS = 0;
-    return 0;
-  }
-
-
-  // Error and warning information
-  bool PlayoutWarning() const { return false; }
-  bool PlayoutError() const { return false; }
-  void ClearPlayoutWarning() {}
-  void ClearPlayoutError() {}
-
-  // Attach audio buffer
-  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
-
-  // Speaker audio routing
-  int32_t SetLoudspeakerStatus(bool enable) { return 0; }
-  int32_t GetLoudspeakerStatus(bool& enable) const { enable = true; return 0; }  // NOLINT
-
- protected:
-  virtual int PlayoutDelayMs() { return 40; }
-
- private:
-  bool initialized_;
-  bool speaker_initialized_;
-  bool play_initialized_;
-  bool playing_;
-};
-#endif
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_OUTPUT_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_player.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_player.cc
new file mode 100644
index 0000000000..f89552f626
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_player.cc
@@ -0,0 +1,509 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_device/android/opensles_player.h"
+
+#include <android/log.h>
+#include <dlfcn.h>
+
+#ifdef MOZILLA_INTERNAL_API
+#include "OpenSLESProvider.h"
+#endif
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/modules/audio_device/android/audio_manager.h"
+#include "webrtc/modules/audio_device/fine_audio_buffer.h"
+
+#define TAG "OpenSLESPlayer"
+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
+
+#define RETURN_ON_ERROR(op, ...)        \
+  do {                                  \
+    SLresult err = (op);                \
+    if (err != SL_RESULT_SUCCESS) {     \
+      ALOGE("%s failed: %ld", #op, err); \
+      return __VA_ARGS__;               \
+    }                                   \
+  } while (0)
+
+namespace webrtc {
+
+OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
+    : audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      audio_device_buffer_(NULL),
+      initialized_(false),
+      playing_(false),
+      bytes_per_buffer_(0),
+      buffer_index_(0),
+      engine_object_(nullptr),
+      engine_(nullptr),
+      player_(nullptr),
+      simple_buffer_queue_(nullptr),
+      volume_(nullptr),
+      last_play_time_(0) {
+  ALOGD("ctor%s", GetThreadInfo().c_str());
+  // Use native audio output parameters provided by the audio manager and
+  // define the PCM format structure.
+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
+                                       audio_parameters_.sample_rate(),
+                                       audio_parameters_.bits_per_sample());
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the internal  audio thread.
+  thread_checker_opensles_.DetachFromThread();
+}
+
+OpenSLESPlayer::~OpenSLESPlayer() {
+  ALOGD("dtor%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  Terminate();
+  DestroyAudioPlayer();
+  DestroyMix();
+  DestroyEngine();
+  RTC_DCHECK(!engine_object_);
+  RTC_DCHECK(!engine_);
+  RTC_DCHECK(!output_mix_.Get());
+  RTC_DCHECK(!player_);
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_DCHECK(!volume_);
+}
+
+int OpenSLESPlayer::Init() {
+  ALOGD("Init%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
+  /* Try to dynamically open the OpenSLES library */
+  opensles_lib_ = dlopen("libOpenSLES.so", RTLD_LAZY);
+  if (!opensles_lib_) {
+    ALOGE("failed to dlopen OpenSLES library");
+    return -1;
+  }
+
+  slCreateEngine_ = (slCreateEngine_t)dlsym(opensles_lib_, "slCreateEngine");
+  SL_IID_ENGINE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ENGINE");
+  SL_IID_ANDROIDCONFIGURATION_ =
+    *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDCONFIGURATION");
+  SL_IID_BUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_BUFFERQUEUE");
+  SL_IID_VOLUME_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_VOLUME");
+  SL_IID_PLAY_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_PLAY");
+
+  if (!slCreateEngine ||
+      !SL_IID_ENGINE_ ||
+      !SL_IID_ANDROIDCONFIGURATION_ ||
+      !SL_IID_BUFFERQUEUE_ ||
+      !SL_IID_VOLUME_ ||
+      !SL_IID_PLAY_) {
+    ALOGE("failed to links to SLES library");
+    return -1;
+  }
+
+  return 0;
+}
+
+int OpenSLESPlayer::Terminate() {
+  ALOGD("Terminate%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  StopPlayout();
+  return 0;
+}
+
+int OpenSLESPlayer::InitPlayout() {
+  ALOGD("InitPlayout%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  CreateEngine();
+  CreateMix();
+  initialized_ = true;
+  buffer_index_ = 0;
+  last_play_time_ = rtc::Time();
+  return 0;
+}
+
+int OpenSLESPlayer::StartPlayout() {
+  ALOGD("StartPlayout%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!playing_);
+  // The number of lower latency audio players is limited, hence we create the
+  // audio player in Start() and destroy it in Stop().
+  CreateAudioPlayer();
+  // Fill up audio buffers to avoid initial glitch and to ensure that playback
+  // starts when mode is later changed to SL_PLAYSTATE_PLAYING.
+  // TODO(henrika): we can save some delay by only making one call to
+  // EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    EnqueuePlayoutData();
+  }
+  // Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
+  // For a player object, when the object is in the SL_PLAYSTATE_PLAYING
+  // state, adding buffers will implicitly start playback.
+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING), -1);
+  playing_ = (GetPlayState() == SL_PLAYSTATE_PLAYING);
+  RTC_DCHECK(playing_);
+  return 0;
+}
+
+int OpenSLESPlayer::StopPlayout() {
+  ALOGD("StopPlayout%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!initialized_ || !playing_) {
+    return 0;
+  }
+  // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
+  // Clear the buffer queue to flush out any remaining data.
+  RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
+#ifndef NDEBUG
+  // Verify that the buffer queue is in fact cleared as it should.
+  SLAndroidSimpleBufferQueueState buffer_queue_state;
+  (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
+  RTC_DCHECK_EQ(0u, buffer_queue_state.count);
+  RTC_DCHECK_EQ(0u, buffer_queue_state.index);
+#endif
+  // The number of lower latency audio players is limited, hence we create the
+  // audio player in Start() and destroy it in Stop().
+  DestroyAudioPlayer();
+  thread_checker_opensles_.DetachFromThread();
+  initialized_ = false;
+  playing_ = false;
+  return 0;
+}
+
+int OpenSLESPlayer::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int OpenSLESPlayer::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int OpenSLESPlayer::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int OpenSLESPlayer::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  ALOGD("AttachAudioBuffer");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
+  audio_device_buffer_->SetPlayoutChannels(channels);
+  RTC_CHECK(audio_device_buffer_);
+  AllocateDataBuffers();
+}
+
+SLDataFormat_PCM OpenSLESPlayer::CreatePCMConfiguration(
+    size_t channels,
+    int sample_rate,
+    size_t bits_per_sample) {
+  ALOGD("CreatePCMConfiguration");
+  RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
+  SLDataFormat_PCM format;
+  format.formatType = SL_DATAFORMAT_PCM;
+  format.numChannels = static_cast<SLuint32>(channels);
+  // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
+  switch (sample_rate) {
+    case 8000:
+      format.samplesPerSec = SL_SAMPLINGRATE_8;
+      break;
+    case 16000:
+      format.samplesPerSec = SL_SAMPLINGRATE_16;
+      break;
+    case 22050:
+      format.samplesPerSec = SL_SAMPLINGRATE_22_05;
+      break;
+    case 32000:
+      format.samplesPerSec = SL_SAMPLINGRATE_32;
+      break;
+    case 44100:
+      format.samplesPerSec = SL_SAMPLINGRATE_44_1;
+      break;
+    case 48000:
+      format.samplesPerSec = SL_SAMPLINGRATE_48;
+      break;
+    default:
+      RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;
+  }
+  format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.endianness = SL_BYTEORDER_LITTLEENDIAN;
+  if (format.numChannels == 1)
+    format.channelMask = SL_SPEAKER_FRONT_CENTER;
+  else if (format.numChannels == 2)
+    format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+  else
+    RTC_CHECK(false) << "Unsupported number of channels: "
+                     << format.numChannels;
+  return format;
+}
+
+void OpenSLESPlayer::AllocateDataBuffers() {
+  ALOGD("AllocateDataBuffers");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_CHECK(audio_device_buffer_);
+  // Don't use the lowest possible size as native buffer size. Instead,
+  // use 10ms to better match the frame size that WebRTC uses. It will result
+  // in a reduced risk for audio glitches and also in a more "clean" sequence
+  // of callbacks from the OpenSL ES thread in to WebRTC when asking for audio
+  // to render.
+  ALOGD("lowest possible buffer size: %" PRIuS,
+      audio_parameters_.GetBytesPerBuffer());
+  bytes_per_buffer_ = audio_parameters_.GetBytesPerFrame() *
+      audio_parameters_.frames_per_10ms_buffer();
+  RTC_DCHECK_GE(bytes_per_buffer_, audio_parameters_.GetBytesPerBuffer());
+  ALOGD("native buffer size: %" PRIuS, bytes_per_buffer_);
+  // Create a modified audio buffer class which allows us to ask for any number
+  // of samples (and not only multiple of 10ms) to match the native OpenSL ES
+  // buffer size.
+  fine_buffer_.reset(new FineAudioBuffer(audio_device_buffer_,
+                                         bytes_per_buffer_,
+                                         audio_parameters_.sample_rate()));
+  // Each buffer must be of this size to avoid unnecessary memcpy while caching
+  // data between successive callbacks.
+  const size_t required_buffer_size =
+      fine_buffer_->RequiredPlayoutBufferSizeBytes();
+  ALOGD("required buffer size: %" PRIuS, required_buffer_size);
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    audio_buffers_[i].reset(new SLint8[required_buffer_size]);
+  }
+}
+
+bool OpenSLESPlayer::CreateEngine() {
+  ALOGD("CreateEngine");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (engine_object_) {
+    return true;
+  }
+  RTC_DCHECK(!engine_);
+  const SLEngineOption option[] = {
+    {SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}};
+
+#ifndef MOZILLA_INTERNAL_API
+  RETURN_ON_ERROR(slCreateEngine_(&engine_object_, 1, option, 0, NULL, NULL),
+                  false);
+  RETURN_ON_ERROR((*engine_object_)->Realize(engine_object_, SL_BOOLEAN_FALSE), false);
+#else
+  RETURN_ON_ERROR(mozilla_get_sles_engine(&engine_object_, 1, option),
+                  false);
+  RETURN_ON_ERROR(mozilla_realize_sles_engine(engine_object_), false);
+#endif
+  RETURN_ON_ERROR(
+    (*engine_object_)->GetInterface(engine_object_, SL_IID_ENGINE_, &engine_),
+    false);
+
+  return true;
+}
+
+void OpenSLESPlayer::DestroyEngine() {
+  ALOGD("DestroyEngine");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!engine_object_)
+    return;
+  engine_ = nullptr;
+#ifndef MOZILLA_INTERNAL_API
+  (*engine_object_)->Destroy(engine_object_);
+#else
+  mozilla_destroy_sles_engine(&engine_object_);
+#endif
+}
+
+bool OpenSLESPlayer::CreateMix() {
+  ALOGD("CreateMix");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(engine_);
+  if (output_mix_.Get())
+    return true;
+
+  // Create the ouput mix on the engine object. No interfaces will be used.
+  RETURN_ON_ERROR((*engine_)->CreateOutputMix(engine_, output_mix_.Receive(), 0,
+                                              NULL, NULL),
+                  false);
+  RETURN_ON_ERROR(output_mix_->Realize(output_mix_.Get(), SL_BOOLEAN_FALSE),
+                  false);
+  return true;
+}
+
+void OpenSLESPlayer::DestroyMix() {
+  ALOGD("DestroyMix");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!output_mix_.Get())
+    return;
+  output_mix_.Reset();
+}
+
+bool OpenSLESPlayer::CreateAudioPlayer() {
+  ALOGD("CreateAudioPlayer");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(engine_object_);
+  RTC_DCHECK(output_mix_.Get());
+  if (player_object_.Get())
+    return true;
+  RTC_DCHECK(!player_);
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_DCHECK(!volume_);
+
+  // source: Android Simple Buffer Queue Data Locator is source.
+  SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
+  SLDataSource audio_source = {&simple_buffer_queue, &pcm_format_};
+
+  // sink: OutputMix-based data is sink.
+  SLDataLocator_OutputMix locator_output_mix = {SL_DATALOCATOR_OUTPUTMIX,
+                                                output_mix_.Get()};
+  SLDataSink audio_sink = {&locator_output_mix, NULL};
+
+  // Define interfaces that we indend to use and realize.
+  const SLInterfaceID interface_ids[] = {
+      SL_IID_ANDROIDCONFIGURATION_, SL_IID_BUFFERQUEUE_, SL_IID_VOLUME_};
+  const SLboolean interface_required[] = {
+      SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+
+  // Create the audio player on the engine interface.
+  RETURN_ON_ERROR(
+      (*engine_)->CreateAudioPlayer(
+          engine_, player_object_.Receive(), &audio_source, &audio_sink,
+          arraysize(interface_ids), interface_ids, interface_required),
+      false);
+
+  // Use the Android configuration interface to set platform-specific
+  // parameters. Should be done before player is realized.
+  SLAndroidConfigurationItf player_config;
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(),
+                                   SL_IID_ANDROIDCONFIGURATION_, &player_config),
+      false);
+  // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
+  // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
+  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
+  RETURN_ON_ERROR(
+      (*player_config)
+          ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
+                             &stream_type, sizeof(SLint32)),
+      false);
+
+  // Realize the audio player object after configuration has been set.
+  RETURN_ON_ERROR(
+      player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
+
+  // Get the SLPlayItf interface on the audio player.
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(), SL_IID_PLAY_, &player_),
+      false);
+
+  // Get the SLAndroidSimpleBufferQueueItf interface on the audio player.
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(), SL_IID_BUFFERQUEUE_,
+                                   &simple_buffer_queue_),
+      false);
+
+  // Register callback method for the Android Simple Buffer Queue interface.
+  // This method will be called when the native audio layer needs audio data.
+  RETURN_ON_ERROR((*simple_buffer_queue_)
+                      ->RegisterCallback(simple_buffer_queue_,
+                                         SimpleBufferQueueCallback, this),
+                  false);
+
+  // Get the SLVolumeItf interface on the audio player.
+  RETURN_ON_ERROR(player_object_->GetInterface(player_object_.Get(),
+                                               SL_IID_VOLUME_, &volume_),
+                  false);
+
+  // TODO(henrika): might not be required to set volume to max here since it
+  // seems to be default on most devices. Might be required for unit tests.
+  // RETURN_ON_ERROR((*volume_)->SetVolumeLevel(volume_, 0), false);
+  return true;
+}
+
+void OpenSLESPlayer::DestroyAudioPlayer() {
+  ALOGD("DestroyAudioPlayer");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!player_object_.Get())
+    return;
+  player_object_.Reset();
+  player_ = nullptr;
+  simple_buffer_queue_ = nullptr;
+  volume_ = nullptr;
+}
+
+// static
+void OpenSLESPlayer::SimpleBufferQueueCallback(
+    SLAndroidSimpleBufferQueueItf caller,
+    void* context) {
+  OpenSLESPlayer* stream = reinterpret_cast<OpenSLESPlayer*>(context);
+  stream->FillBufferQueue();
+}
+
+void OpenSLESPlayer::FillBufferQueue() {
+  RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
+  SLuint32 state = GetPlayState();
+  if (state != SL_PLAYSTATE_PLAYING) {
+    ALOGW("Buffer callback in non-playing state!");
+    return;
+  }
+  EnqueuePlayoutData();
+}
+
+void OpenSLESPlayer::EnqueuePlayoutData() {
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 100ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_play_time_;
+  if (diff > 100) {
+    ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
+  }
+  last_play_time_ = current_time;
+  // Read audio data from the WebRTC source using the FineAudioBuffer object
+  // to adjust for differences in buffer size between WebRTC (10ms) and native
+  // OpenSL ES.
+  SLint8* audio_ptr = audio_buffers_[buffer_index_].get();
+  fine_buffer_->GetPlayoutData(audio_ptr);
+  // Enqueue the decoded audio buffer for playback.
+  SLresult err =
+      (*simple_buffer_queue_)
+          ->Enqueue(simple_buffer_queue_, audio_ptr, bytes_per_buffer_);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("Enqueue failed: %ld", err);
+  }
+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
+}
+
+SLuint32 OpenSLESPlayer::GetPlayState() const {
+  RTC_DCHECK(player_);
+  SLuint32 state;
+  SLresult err = (*player_)->GetPlayState(player_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetPlayState failed: %ld", err);
+  }
+  return state;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_player.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_player.h
new file mode 100644
index 0000000000..c193c138b7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_player.h
@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <SLES/OpenSLES_AndroidConfiguration.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_device/android/audio_common.h"
+#include "webrtc/modules/audio_device/android/audio_manager.h"
+#include "webrtc/modules/audio_device/android/opensles_common.h"
+#include "webrtc/modules/audio_device/include/audio_device_defines.h"
+#include "webrtc/modules/audio_device/audio_device_generic.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+// Implements 16-bit mono PCM audio output support for Android using the
+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Decoded audio
+// buffers are requested on a dedicated internal thread managed by the OpenSL
+// ES layer.
+//
+// The existing design forces the user to call InitPlayout() after Stoplayout()
+// to be able to call StartPlayout() again. This is inline with how the Java-
+// based implementation works.
+//
+// OpenSL ES is a native C API which have no Dalvik-related overhead such as
+// garbage collection pauses and it supports reduced audio output latency.
+// If the device doesn't claim this feature but supports API level 9 (Android
+// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
+// the output latency may be higher.
+class OpenSLESPlayer {
+ public:
+  // The lower output latency path is used only if the application requests a
+  // buffer count of 2 or more, and a buffer size and sample rate that are
+  // compatible with the device's native output configuration provided via the
+  // audio manager at construction.
+  static const int kNumOfOpenSLESBuffers = 4;
+
+  // There is no need for this class to use JNI.
+  static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* context) {
+    return 0;
+  }
+  static void ClearAndroidAudioDeviceObjects() {}
+
+  explicit OpenSLESPlayer(AudioManager* audio_manager);
+  ~OpenSLESPlayer();
+
+  int Init();
+  int Terminate();
+
+  int InitPlayout();
+  bool PlayoutIsInitialized() const { return initialized_; }
+
+  int StartPlayout();
+  int StopPlayout();
+  bool Playing() const { return playing_; }
+
+  int SpeakerVolumeIsAvailable(bool& available);
+  int SetSpeakerVolume(uint32_t volume);
+  int SpeakerVolume(uint32_t& volume) const;
+  int MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int MinSpeakerVolume(uint32_t& minVolume) const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  // These callback methods are called when data is required for playout.
+  // They are both called from an internal "OpenSL ES thread" which is not
+  // attached to the Dalvik VM.
+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
+                                        void* context);
+  void FillBufferQueue();
+  // Reads audio data in PCM format using the AudioDeviceBuffer.
+  // Can be called both on the main thread (during Start()) and from the
+  // internal audio thread while output streaming is active.
+  void EnqueuePlayoutData();
+
+  // Configures the SL_DATAFORMAT_PCM structure.
+  SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
+                                          int sample_rate,
+                                          size_t bits_per_sample);
+
+  // Allocate memory for audio buffers which will be used to render audio
+  // via the SLAndroidSimpleBufferQueueItf interface.
+  void AllocateDataBuffers();
+
+  // Creates/destroys the main engine object and the SLEngineItf interface.
+  bool CreateEngine();
+  void DestroyEngine();
+
+  // Creates/destroys the output mix object.
+  bool CreateMix();
+  void DestroyMix();
+
+  // Creates/destroys the audio player and the simple-buffer object.
+  // Also creates the volume object.
+  bool CreateAudioPlayer();
+  void DestroyAudioPlayer();
+
+  SLuint32 GetPlayState() const;
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  rtc::ThreadChecker thread_checker_;
+
+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
+  // non-application thread which is not attached to the Dalvik JVM.
+  // Detached during construction of this object.
+  rtc::ThreadChecker thread_checker_opensles_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  bool initialized_;
+  bool playing_;
+
+  // PCM-type format definition.
+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
+  // 32-bit float representation is needed.
+  SLDataFormat_PCM pcm_format_;
+
+  // Number of bytes per audio buffer in each |audio_buffers_[i]|.
+  // Typical sizes are 480 or 512 bytes corresponding to native output buffer
+  // sizes of 240 or 256 audio frames respectively.
+  size_t bytes_per_buffer_;
+
+  // Queue of audio buffers to be used by the player object for rendering
+  // audio. They will be used in a Round-robin way and the size of each buffer
+  // is given by FineAudioBuffer::RequiredBufferSizeBytes().
+  rtc::scoped_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples.
+  // Example: native buffer size is 240 audio frames at 48kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 240
+  // in each callback (one every 5ms). This class can then ask for 240 and the
+  // FineAudioBuffer will ask WebRTC for new data only every second callback
+  // and also cach non-utilized audio.
+  rtc::scoped_ptr<FineAudioBuffer> fine_buffer_;
+
+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
+  int buffer_index_;
+
+  // The engine object which provides the SLEngineItf interface.
+  // Created by the global Open SL ES constructor slCreateEngine().
+  SLObjectItf engine_object_;
+
+  // This interface exposes creation methods for all the OpenSL ES object types.
+  // It is the OpenSL ES API entry point.
+  SLEngineItf engine_;
+
+  // Output mix object to be used by the player object.
+  webrtc::ScopedSLObjectItf output_mix_;
+
+  // The audio player media object plays out audio to the speakers. It also
+  // supports volume control.
+  webrtc::ScopedSLObjectItf player_object_;
+
+  // This interface is supported on the audio player and it controls the state
+  // of the audio player.
+  SLPlayItf player_;
+
+  // The Android Simple Buffer Queue interface is supported on the audio player
+  // and it provides methods to send audio data from the source to the audio
+  // player for rendering.
+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
+
+  // This interface exposes controls for manipulating the object’s audio volume
+  // properties. This interface is supported on the Audio Player object.
+  SLVolumeItf volume_;
+
+  // Last time the OpenSL ES layer asked for audio data to play out.
+  uint32_t last_play_time_;
+
+  void *opensles_lib_;
+  typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
+                                       SLuint32,
+                                       const SLEngineOption *,
+                                       SLuint32,
+                                       const SLInterfaceID *,
+                                       const SLboolean *);
+  slCreateEngine_t slCreateEngine_;
+  SLInterfaceID SL_IID_ENGINE_;
+  SLInterfaceID SL_IID_ANDROIDCONFIGURATION_;
+  SLInterfaceID SL_IID_BUFFERQUEUE_;
+  SLInterfaceID SL_IID_VOLUME_;
+  SLInterfaceID SL_IID_PLAY_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.cc
deleted file mode 100644
index 967630a7e1..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.cc
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if defined(_MSC_VER)
-#include <windows.h>
-#endif
-
-#include "webrtc/modules/audio_device/android/single_rw_fifo.h"
-
-#include <assert.h>
-
-static int UpdatePos(int pos, int capacity) {
-  return (pos + 1) % capacity;
-}
-
-namespace webrtc {
-
-namespace subtle {
-
-// Start with compiler support, then processor-specific hacks
-#if defined(__GNUC__) || defined(__clang__)
-// Available on GCC and clang - others?
-inline void MemoryBarrier() {
-  __sync_synchronize();
-}
-
-#elif defined(_MSC_VER)
-inline void MemoryBarrier() {
-  ::MemoryBarrier();
-}
-
-#elif defined(__aarch64__)
-// From http://http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_arm64_gcc.h
-inline void MemoryBarrier() {
-  __asm__ __volatile__ ("dmb ish" ::: "memory");
-}
-
-#elif defined(__ARMEL__)
-// From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_arm_gcc.h
-inline void MemoryBarrier() {
-  // Note: This is a function call, which is also an implicit compiler barrier.
-  typedef void (*KernelMemoryBarrierFunc)();
-  ((KernelMemoryBarrierFunc)0xffff0fa0)();
-}
-
-#elif defined(__x86_64__) || defined (__i386__)
-// From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_x86_gcc.h
-// mfence exists on x64 and x86 platforms containing SSE2.
-// x86 platforms that don't have SSE2 will crash with SIGILL.
-// If this code needs to run on such platforms in the future,
-// add runtime CPU detection here.
-inline void MemoryBarrier() {
-  __asm__ __volatile__("mfence" : : : "memory");
-}
-
-#elif defined(__MIPSEL__)
-// From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_mips_gcc.h
-inline void MemoryBarrier() {
-  __asm__ __volatile__("sync" : : : "memory");
-}
-
-#else
-#error Add an implementation of MemoryBarrier() for this platform!
-#endif
-
-}  // namespace subtle
-
-SingleRwFifo::SingleRwFifo(int capacity)
-    : capacity_(capacity),
-      size_(0),
-      read_pos_(0),
-      write_pos_(0) {
-  queue_.reset(new int8_t*[capacity_]);
-}
-
-SingleRwFifo::~SingleRwFifo() {
-}
-
-void SingleRwFifo::Push(int8_t* mem) {
-  assert(mem);
-
-  // Ensure that there is space for the new data in the FIFO.
-  // Note there is only one writer meaning that the other thread is guaranteed
-  // only to decrease the size.
-  const int free_slots = capacity() - size();
-  if (free_slots <= 0) {
-    // Size can be queried outside of the Push function. The caller is assumed
-    // to ensure that Push will be successful before calling it.
-    assert(false);
-    return;
-  }
-  queue_[write_pos_] = mem;
-  // Memory barrier ensures that |size_| is updated after the size has changed.
-  subtle::MemoryBarrier();
-  ++size_;
-  write_pos_ = UpdatePos(write_pos_, capacity());
-}
-
-int8_t* SingleRwFifo::Pop() {
-  int8_t* ret_val = NULL;
-  if (size() <= 0) {
-    // Size can be queried outside of the Pop function. The caller is assumed
-    // to ensure that Pop will be successfull before calling it.
-    assert(false);
-    return ret_val;
-  }
-  ret_val = queue_[read_pos_];
-  // Memory barrier ensures that |size_| is updated after the size has changed.
-  subtle::MemoryBarrier();
-  --size_;
-  read_pos_ = UpdatePos(read_pos_, capacity());
-  return ret_val;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.h b/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.h
deleted file mode 100644
index e51ea5ae95..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_SINGLE_RW_FIFO_H_
-#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_SINGLE_RW_FIFO_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Implements a lock-free FIFO losely based on
-// http://src.chromium.org/viewvc/chrome/trunk/src/media/base/audio_fifo.cc
-// Note that this class assumes there is one producer (writer) and one
-// consumer (reader) thread.
-class SingleRwFifo {
- public:
-  explicit SingleRwFifo(int capacity);
-  ~SingleRwFifo();
-
-  void Push(int8_t* mem);
-  int8_t* Pop();
-
-  void Clear();
-
-  int size() { return size_.Value(); }
-  int capacity() const { return capacity_; }
-
- private:
-  rtc::scoped_ptr<int8_t* []> queue_;
-  int capacity_;
-
-  Atomic32 size_;
-
-  int read_pos_;
-  int write_pos_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_SINGLE_RW_FIFO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc
deleted file mode 100644
index b53c9e427e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/single_rw_fifo_unittest.cc
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/android/single_rw_fifo.h"
-
-#include <list>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace webrtc {
-
-class SingleRwFifoTest : public testing::Test {
- public:
-  enum {
-    // Uninteresting as it does not affect test
-    kBufferSize = 8,
-    kCapacity = 6,
-  };
-
-  SingleRwFifoTest() : fifo_(kCapacity), pushed_(0), available_(0) {
-  }
-  virtual ~SingleRwFifoTest() {}
-
-  void SetUp() {
-    for (int8_t i = 0; i < kCapacity; ++i) {
-      // Create memory area.
-      buffer_[i].reset(new int8_t[kBufferSize]);
-      // Set the first byte in the buffer to the order in which it was created
-      // this allows us to e.g. check that the buffers don't re-arrange.
-      buffer_[i][0] = i;
-      // Queue used by test.
-      memory_queue_.push_back(buffer_[i].get());
-    }
-    available_ = kCapacity;
-    VerifySizes();
-  }
-
-  void Push(int number_of_buffers) {
-    for (int8_t i = 0; i < number_of_buffers; ++i) {
-      int8_t* data = memory_queue_.front();
-      memory_queue_.pop_front();
-      fifo_.Push(data);
-      --available_;
-      ++pushed_;
-    }
-    VerifySizes();
-    VerifyOrdering();
-  }
-  void Pop(int number_of_buffers) {
-    for (int8_t i = 0; i < number_of_buffers; ++i) {
-      int8_t* data = fifo_.Pop();
-      memory_queue_.push_back(data);
-      ++available_;
-      --pushed_;
-    }
-    VerifySizes();
-    VerifyOrdering();
-  }
-
-  void VerifyOrdering() const {
-    std::list<int8_t*>::const_iterator iter = memory_queue_.begin();
-    if (iter == memory_queue_.end()) {
-      return;
-    }
-    int8_t previous_index = DataToElementIndex(*iter);
-    ++iter;
-    for (; iter != memory_queue_.end(); ++iter) {
-      int8_t current_index = DataToElementIndex(*iter);
-      EXPECT_EQ(current_index, ++previous_index % kCapacity);
-    }
-  }
-
-  void VerifySizes() {
-    EXPECT_EQ(available_, static_cast<int>(memory_queue_.size()));
-    EXPECT_EQ(pushed_, fifo_.size());
-  }
-
-  int8_t DataToElementIndex(int8_t* data) const {
-    return data[0];
-  }
-
- protected:
-  SingleRwFifo fifo_;
-  // Memory area for proper de-allocation.
-  rtc::scoped_ptr<int8_t[]> buffer_[kCapacity];
-  std::list<int8_t*> memory_queue_;
-
-  int pushed_;
-  int available_;
-
- private:
-  DISALLOW_COPY_AND_ASSIGN(SingleRwFifoTest);
-};
-
-TEST_F(SingleRwFifoTest, Construct) {
-  // All verifications are done in SetUp.
-}
-
-TEST_F(SingleRwFifoTest, Push) {
-  Push(kCapacity);
-}
-
-TEST_F(SingleRwFifoTest, Pop) {
-  // Push all available.
-  Push(available_);
-
-  // Test border cases:
-  // At capacity
-  Pop(1);
-  Push(1);
-
-  // At minimal capacity
-  Pop(pushed_);
-  Push(1);
-  Pop(1);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi
index 495ff23b7f..b4bc3d95d0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi
@@ -20,13 +20,13 @@
       ],
       'include_dirs': [
         '.',
-        '../interface',
+        '../include',
         'include',
         'dummy',  # Contains dummy audio device implementations.
       ],
       'direct_dependent_settings': {
         'include_dirs': [
-          '../interface',
+          '../include',
           'include',
         ],
       },
@@ -38,17 +38,13 @@
         'audio_device_buffer.h',
         'audio_device_generic.cc',
         'audio_device_generic.h',
-        'audio_device_utility.cc',
-        'audio_device_utility.h',
-        'audio_device_impl.cc',
-        'audio_device_impl.h',
         'audio_device_config.h',
         'dummy/audio_device_dummy.cc',
         'dummy/audio_device_dummy.h',
-        'dummy/audio_device_utility_dummy.cc',
-        'dummy/audio_device_utility_dummy.h',
         'dummy/file_audio_device.cc',
         'dummy/file_audio_device.h',
+        'fine_audio_buffer.cc',
+        'fine_audio_buffer.h',
       ],
       'conditions': [
         ['build_with_mozilla==1', {
@@ -111,84 +107,41 @@
         }],
         ['include_internal_audio_device==1', {
           'sources': [
-            'linux/audio_device_utility_linux.cc',
-            'linux/audio_device_utility_linux.h',
-            'linux/latebindingsymboltable_linux.cc',
-            'linux/latebindingsymboltable_linux.h',
-            'ios/audio_device_ios.mm',
-            'ios/audio_device_ios.h',
-            'ios/audio_device_utility_ios.cc',
-            'ios/audio_device_utility_ios.h',
-            'mac/audio_device_mac.cc',
-            'mac/audio_device_mac.h',
-            'mac/audio_device_utility_mac.cc',
-            'mac/audio_device_utility_mac.h',
-            'mac/audio_mixer_manager_mac.cc',
-            'mac/audio_mixer_manager_mac.h',
-            'mac/portaudio/pa_memorybarrier.h',
-            'mac/portaudio/pa_ringbuffer.c',
-            'mac/portaudio/pa_ringbuffer.h',
-            'win/audio_device_core_win.cc',
-            'win/audio_device_core_win.h',
-            'win/audio_device_wave_win.cc',
-            'win/audio_device_wave_win.h',
-            'win/audio_device_utility_win.cc',
-            'win/audio_device_utility_win.h',
-            'win/audio_mixer_manager_win.cc',
-            'win/audio_mixer_manager_win.h',
+            'audio_device_impl.cc',
+            'audio_device_impl.h',
             # used externally for getUserMedia
             'opensl/single_rw_fifo.cc',
             'opensl/single_rw_fifo.h',
-            'android/audio_device_template.h',
-            'android/audio_manager.cc',
-            'android/audio_manager.h',
-            'android/audio_manager_jni.cc',
-            'android/audio_manager_jni.h',
-            'android/audio_record_jni.cc',
-            'android/audio_record_jni.h',
-            'android/audio_track_jni.cc',
-            'android/audio_track_jni.h',
           ],
           'conditions': [
             ['OS=="android"', {
+               'sources': [
+                'android/audio_device_template.h',
+                'android/audio_manager.cc',
+                'android/audio_manager.h',
+                'android/audio_record_jni.cc',
+                'android/audio_record_jni.h',
+                'android/audio_track_jni.cc',
+                'android/audio_track_jni.h',
+                'android/build_info.cc',
+                'android/build_info.h',
+                'android/opensles_common.cc',
+                'android/opensles_common.h',
+                'android/opensles_player.cc',
+                'android/opensles_player.h',
+              ],
               'link_settings': {
                 'libraries': [
                   '-llog',
                   '-lOpenSLES',
                 ],
               },
-              'conditions': [
-                ['enable_android_opensl==1', {
-                  'sources': [
-                    'opensl/fine_audio_buffer.cc',
-                    'opensl/fine_audio_buffer.h',
-                    'opensl/low_latency_event_posix.cc',
-                    'opensl/low_latency_event.h',
-                    'opensl/opensles_common.cc',
-                    'opensl/opensles_common.h',
-                    'opensl/opensles_input.cc',
-                    'opensl/opensles_input.h',
-                    'opensl/opensles_output.h',
-                    'shared/audio_device_utility_shared.cc',
-                    'shared/audio_device_utility_shared.h',
-                  ],
-                }, {
-                  'sources': [
-                    'shared/audio_device_utility_shared.cc',
-                    'shared/audio_device_utility_shared.h',
-                  ],
-                }],
-                ['enable_android_opensl_output==1', {
-                  'sources': [
-                    'opensl/opensles_output.cc'
-                  ],
-                  'defines': [
-                    'WEBRTC_ANDROID_OPENSLES_OUTPUT',
-                  ],
-                }],
-              ],
             }],
             ['OS=="linux"', {
+              'sources': [
+                'linux/latebindingsymboltable_linux.cc',
+                'linux/latebindingsymboltable_linux.h',
+              ],
               'link_settings': {
                 'libraries': [
                   '-ldl','-lX11',
@@ -241,6 +194,15 @@
               ],
             }],
             ['OS=="mac"', {
+              'sources': [
+                'mac/audio_device_mac.cc',
+                'mac/audio_device_mac.h',
+                'mac/audio_mixer_manager_mac.cc',
+                'mac/audio_mixer_manager_mac.h',
+                'mac/portaudio/pa_memorybarrier.h',
+                'mac/portaudio/pa_ringbuffer.c',
+                'mac/portaudio/pa_ringbuffer.h',
+              ],
               'link_settings': {
                 'libraries': [
                   '$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework',
@@ -249,6 +211,11 @@
               },
             }],
             ['OS=="ios"', {
+              'sources': [
+                'ios/audio_device_ios.h',
+                'ios/audio_device_ios.mm',
+                'ios/audio_device_not_implemented_ios.mm',
+              ],
               'xcode_settings': {
                 'CLANG_ENABLE_OBJC_ARC': 'YES',
               },
@@ -258,11 +225,20 @@
                     '-framework AudioToolbox',
                     '-framework AVFoundation',
                     '-framework Foundation',
+                    '-framework UIKit',
                   ],
                 },
               },
             }],
             ['OS=="win"', {
+              'sources': [
+                'win/audio_device_core_win.cc',
+                'win/audio_device_core_win.h',
+                'win/audio_device_wave_win.cc',
+                'win/audio_device_wave_win.h',
+                'win/audio_mixer_manager_win.cc',
+                'win/audio_mixer_manager_win.h',
+              ],
               'link_settings': {
                 'libraries': [
                   # Required for the built-in WASAPI AEC.
@@ -273,18 +249,40 @@
                 ],
               },
             }],
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-bool-conversion',
+                    '-Wno-delete-non-virtual-dtor',
+                    '-Wno-logical-op-parentheses',
+                    '-Wno-microsoft-extra-qualification',
+                    '-Wno-microsoft-goto',
+                    '-Wno-missing-braces',
+                    '-Wno-parentheses-equality',
+                    '-Wno-reorder',
+                    '-Wno-shift-overflow',
+                    '-Wno-tautological-compare',
+                    '-Wno-unused-private-field',
+                  ],
+                },
+              },
+            }],
           ], # conditions
         }], # include_internal_audio_device==1
       ], # conditions
     },
   ],
   'conditions': [
-    ['include_tests==1', {
+    # Does not compile on iOS: webrtc:4755.
+    ['include_tests==1 and OS!="ios"', {
       'targets': [
         {
           'target_name': 'audio_device_tests',
-         'type': 'executable',
-         'dependencies': [
+          'type': 'executable',
+          'dependencies': [
             'audio_device',
             'webrtc_utility',
             '<(webrtc_root)/test/test.gyp:test_support_main',
@@ -315,49 +313,6 @@
           ],
         },
       ], # targets
-      'conditions': [
-        ['test_isolation_mode != "noop"', {
-          'targets': [
-            {
-              'target_name': 'audio_device_tests_run',
-              'type': 'none',
-              'dependencies': [
-                'audio_device_tests',
-              ],
-              'includes': [
-                '../../build/isolate.gypi',
-              ],
-              'sources': [
-                'audio_device_tests.isolate',
-              ],
-            },
-          ],
-        }],
-        ['OS=="android"', {
-          'targets': [
-            {
-              'target_name': 'audio_device_unittest',
-              'type': 'executable',
-              'dependencies': [
-                'audio_device',
-                'webrtc_utility',
-                '<(DEPTH)/testing/gmock.gyp:gmock',
-                '<(DEPTH)/testing/gtest.gyp:gtest',
-                '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-                '<(webrtc_root)/test/test.gyp:test_support_main',
-              ],
-              'sources': [
-                'android/audio_manager.cc',
-                'android/audio_manager.h',
-                'android/fine_audio_buffer_unittest.cc',
-                'android/low_latency_event_unittest.cc',
-                'android/single_rw_fifo_unittest.cc',
-                'mock/mock_audio_device_buffer.h',
-              ],
-            },
-          ],
-        }],
-      ],
-    }], # include_tests
+    }], # include_tests==1 and OS!=ios
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.cc
index 42fdaad22c..48ae88ee90 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -13,11 +13,11 @@
 #include <assert.h>
 #include <string.h>
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -131,8 +131,6 @@ int32_t AudioDeviceBuffer::InitRecording()
 
 int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingSampleRate(fsHz=%u)", fsHz);
-
     CriticalSectionScoped lock(&_critSect);
     _recSampleRate = fsHz;
     return 0;
@@ -144,8 +142,6 @@ int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz)
 
 int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutSampleRate(fsHz=%u)", fsHz);
-
     CriticalSectionScoped lock(&_critSect);
     _playSampleRate = fsHz;
     return 0;
@@ -173,10 +169,8 @@ int32_t AudioDeviceBuffer::PlayoutSampleRate() const
 //  SetRecordingChannels
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingChannels(channels=%u)", channels);
-
     CriticalSectionScoped lock(&_critSect);
     _recChannels = channels;
     _recBytesPerSample = 2*channels;  // 16 bits per sample in mono, 32 bits in stereo
@@ -187,10 +181,8 @@ int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
 //  SetPlayoutChannels
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutChannels(channels=%u)", channels);
-
     CriticalSectionScoped lock(&_critSect);
     _playChannels = channels;
     // 16 bits per sample in mono, 32 bits in stereo
@@ -247,7 +239,7 @@ int32_t AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& chan
 //  RecordingChannels
 // ----------------------------------------------------------------------------
 
-uint8_t AudioDeviceBuffer::RecordingChannels() const
+size_t AudioDeviceBuffer::RecordingChannels() const
 {
     return _recChannels;
 }
@@ -256,7 +248,7 @@ uint8_t AudioDeviceBuffer::RecordingChannels() const
 //  PlayoutChannels
 // ----------------------------------------------------------------------------
 
-uint8_t AudioDeviceBuffer::PlayoutChannels() const
+size_t AudioDeviceBuffer::PlayoutChannels() const
 {
     return _playChannels;
 }
@@ -389,7 +381,7 @@ int32_t AudioDeviceBuffer::StopOutputFileRecording()
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
-                                             uint32_t nSamples)
+                                             size_t nSamples)
 {
     CriticalSectionScoped lock(&_critSect);
 
@@ -407,12 +399,6 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
         return -1;
     }
 
-    if (nSamples != _recSamples)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples);
-        return -1;
-    }
-
     if (_recChannel == AudioDeviceModule::kChannelBoth)
     {
         // (default) copy the complete input buffer to the local buffer
@@ -429,7 +415,7 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
         }
 
         // exctract left or right channel from input buffer to the local buffer
-        for (uint32_t i = 0; i < _recSamples; i++)
+        for (size_t i = 0; i < _recSamples; i++)
         {
             *ptr16Out = *ptr16In;
             ptr16Out++;
@@ -497,11 +483,11 @@ int32_t AudioDeviceBuffer::DeliverRecordedData()
 //  RequestPlayoutData
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
+int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples)
 {
     uint32_t playSampleRate = 0;
-    uint8_t playBytesPerSample = 0;
-    uint8_t playChannels = 0;
+    size_t playBytesPerSample = 0;
+    size_t playChannels = 0;
     {
         CriticalSectionScoped lock(&_critSect);
 
@@ -535,7 +521,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
         }
     }
 
-    uint32_t nSamplesOut(0);
+    size_t nSamplesOut(0);
 
     CriticalSectionScoped lock(&_critSectCb);
 
@@ -564,7 +550,7 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
         }
     }
 
-    return nSamplesOut;
+    return static_cast<int32_t>(nSamplesOut);
 }
 
 // ----------------------------------------------------------------------------
@@ -577,8 +563,9 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
 
     if (_playSize > kMaxBufferSizeBytes)
     {
-       WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "_playSize %i exceeds "
-       "kMaxBufferSizeBytes in AudioDeviceBuffer::GetPlayoutData", _playSize);
+       WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
+                    "_playSize %" PRIuS " exceeds kMaxBufferSizeBytes in "
+                    "AudioDeviceBuffer::GetPlayoutData", _playSize);
        assert(false);
        return -1;
     }
@@ -591,7 +578,7 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
         _playFile.Write(&_playBuffer[0], _playSize);
     }
 
-    return _playSamples;
+    return static_cast<int32_t>(_playSamples);
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.h
index a89927f711..1095971040 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_buffer.h
@@ -12,14 +12,14 @@
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
 
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
 
 const uint32_t kPulsePeriodMs = 1000;
-const uint32_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
+const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
 
 class AudioDeviceObserver;
 
@@ -40,17 +40,17 @@ public:
     int32_t RecordingSampleRate() const;
     int32_t PlayoutSampleRate() const;
 
-    virtual int32_t SetRecordingChannels(uint8_t channels);
-    virtual int32_t SetPlayoutChannels(uint8_t channels);
-    uint8_t RecordingChannels() const;
-    uint8_t PlayoutChannels() const;
+    virtual int32_t SetRecordingChannels(size_t channels);
+    virtual int32_t SetPlayoutChannels(size_t channels);
+    size_t RecordingChannels() const;
+    size_t PlayoutChannels() const;
     int32_t SetRecordingChannel(
         const AudioDeviceModule::ChannelType channel);
     int32_t RecordingChannel(
         AudioDeviceModule::ChannelType& channel) const;
 
     virtual int32_t SetRecordedBuffer(const void* audioBuffer,
-                                      uint32_t nSamples);
+                                      size_t nSamples);
     int32_t SetCurrentMicLevel(uint32_t level);
     virtual void SetVQEData(int playDelayMS,
                             int recDelayMS,
@@ -58,7 +58,7 @@ public:
     virtual int32_t DeliverRecordedData();
     uint32_t NewMicLevel() const;
 
-    virtual int32_t RequestPlayoutData(uint32_t nSamples);
+    virtual int32_t RequestPlayoutData(size_t nSamples);
     virtual int32_t GetPlayoutData(void* audioBuffer);
 
     int32_t StartInputFileRecording(
@@ -80,29 +80,29 @@ private:
     uint32_t                  _recSampleRate;
     uint32_t                  _playSampleRate;
 
-    uint8_t                   _recChannels;
-    uint8_t                   _playChannels;
+    size_t                   _recChannels;
+    size_t                   _playChannels;
 
     // selected recording channel (left/right/both)
     AudioDeviceModule::ChannelType _recChannel;
 
     // 2 or 4 depending on mono or stereo
-    uint8_t                   _recBytesPerSample;
-    uint8_t                   _playBytesPerSample;
+    size_t                   _recBytesPerSample;
+    size_t                   _playBytesPerSample;
 
     // 10ms in stereo @ 96kHz
     int8_t                          _recBuffer[kMaxBufferSizeBytes];
 
     // one sample <=> 2 or 4 bytes
-    uint32_t                  _recSamples;
-    uint32_t                  _recSize;           // in bytes
+    size_t                    _recSamples;
+    size_t                    _recSize;           // in bytes
 
     // 10ms in stereo @ 96kHz
     int8_t                          _playBuffer[kMaxBufferSizeBytes];
 
     // one sample <=> 2 or 4 bytes
-    uint32_t                  _playSamples;
-    uint32_t                  _playSize;          // in bytes
+    size_t                    _playSamples;
+    size_t                    _playSize;          // in bytes
 
     FileWrapper&                    _recFile;
     FileWrapper&                    _playFile;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.cc b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.cc
index 958abbf4d2..501faba7cf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.cc
@@ -9,73 +9,88 @@
  */
 
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
 int32_t AudioDeviceGeneric::SetRecordingSampleRate(
-    const uint32_t samplesPerSec)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Set recording sample rate not supported on this platform");
-    return -1;
+    const uint32_t samplesPerSec) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
-int32_t AudioDeviceGeneric::SetPlayoutSampleRate(
-    const uint32_t samplesPerSec)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Set playout sample rate not supported on this platform");
-    return -1;
+int32_t AudioDeviceGeneric::SetPlayoutSampleRate(const uint32_t samplesPerSec) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
-int32_t AudioDeviceGeneric::SetLoudspeakerStatus(bool enable)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Set loudspeaker status not supported on this platform");
-    return -1;
+int32_t AudioDeviceGeneric::SetLoudspeakerStatus(bool enable) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
-int32_t AudioDeviceGeneric::GetLoudspeakerStatus(bool& enable) const
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Get loudspeaker status not supported on this platform");
-    return -1;
+int32_t AudioDeviceGeneric::GetLoudspeakerStatus(bool& enable) const {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
-int32_t AudioDeviceGeneric::ResetAudioDevice()
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Reset audio device not supported on this platform");
-    return -1;
+int32_t AudioDeviceGeneric::ResetAudioDevice() {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
 int32_t AudioDeviceGeneric::SoundDeviceControl(unsigned int par1,
-    unsigned int par2, unsigned int par3, unsigned int par4)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Sound device control not supported on this platform");
-    return -1;
+                                               unsigned int par2,
+                                               unsigned int par3,
+                                               unsigned int par4) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
 bool AudioDeviceGeneric::BuiltInAECIsAvailable() const {
-  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-      "Built-in AEC not supported on this platform");
+  LOG_F(LS_ERROR) << "Not supported on this platform";
   return false;
 }
 
-int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Built-in AEC not supported on this platform");
-    return -1;
+int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
-bool AudioDeviceGeneric::BuiltInAECIsEnabled() const
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-        "Windows AEC not supported on this platform");
-    return false;
+bool AudioDeviceGeneric::BuiltInAECIsEnabled() const {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+bool AudioDeviceGeneric::BuiltInAGCIsAvailable() const {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInAGC(bool enable) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+bool AudioDeviceGeneric::BuiltInNSIsAvailable() const {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInNS(bool enable) {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+int AudioDeviceGeneric::GetPlayoutAudioParameters(
+    AudioParameters* params) const {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+int AudioDeviceGeneric::GetRecordAudioParameters(
+    AudioParameters* params) const {
+  LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.h b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.h
index 800cc395a8..c76ea52428 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_generic.h
@@ -16,168 +16,164 @@
 
 namespace webrtc {
 
-class AudioDeviceGeneric
-{
+class AudioDeviceGeneric {
  public:
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const = 0;
 
-	// Retrieve the currently utilized audio layer
-	virtual int32_t ActiveAudioLayer(
-        AudioDeviceModule::AudioLayer& audioLayer) const = 0;
+  // Main initializaton and termination
+  virtual int32_t Init() = 0;
+  virtual int32_t Terminate() = 0;
+  virtual bool Initialized() const = 0;
 
-	// Main initializaton and termination
-    virtual int32_t Init() = 0;
-    virtual int32_t Terminate() = 0;
-	virtual bool Initialized() const = 0;
+  // Device enumeration
+  virtual int16_t PlayoutDevices() = 0;
+  virtual int16_t RecordingDevices() = 0;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) = 0;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) = 0;
 
-	// Device enumeration
-	virtual int16_t PlayoutDevices() = 0;
-	virtual int16_t RecordingDevices() = 0;
-	virtual int32_t PlayoutDeviceName(
-        uint16_t index,
-        char name[kAdmMaxDeviceNameSize],
-        char guid[kAdmMaxGuidSize]) = 0;
-    virtual int32_t RecordingDeviceName(
-        uint16_t index,
-        char name[kAdmMaxDeviceNameSize],
-        char guid[kAdmMaxGuidSize]) = 0;
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+  virtual int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) = 0;
+  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) = 0;
 
-	// Device selection
-	virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
-	virtual int32_t SetPlayoutDevice(
-        AudioDeviceModule::WindowsDeviceType device) = 0;
-    virtual int32_t SetRecordingDevice(uint16_t index) = 0;
-	virtual int32_t SetRecordingDevice(
-        AudioDeviceModule::WindowsDeviceType device) = 0;
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available) = 0;
+  virtual int32_t InitPlayout() = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int32_t RecordingIsAvailable(bool& available) = 0;
+  virtual int32_t InitRecording() = 0;
+  virtual bool RecordingIsInitialized() const = 0;
 
-	// Audio transport initialization
-    virtual int32_t PlayoutIsAvailable(bool& available) = 0;
-    virtual int32_t InitPlayout() = 0;
-    virtual bool PlayoutIsInitialized() const = 0;
-    virtual int32_t RecordingIsAvailable(bool& available) = 0;
-    virtual int32_t InitRecording() = 0;
-    virtual bool RecordingIsInitialized() const = 0;
+  // Audio transport control
+  virtual int32_t StartPlayout() = 0;
+  virtual int32_t StopPlayout() = 0;
+  virtual bool Playing() const = 0;
+  virtual int32_t StartRecording() = 0;
+  virtual int32_t StopRecording() = 0;
+  virtual bool Recording() const = 0;
 
-	// Audio transport control
-    virtual int32_t StartPlayout() = 0;
-    virtual int32_t StopPlayout() = 0;
-    virtual bool Playing() const = 0;
-	virtual int32_t StartRecording() = 0;
-    virtual int32_t StopRecording() = 0;
-    virtual bool Recording() const = 0;
+  // Microphone Automatic Gain Control (AGC)
+  virtual int32_t SetAGC(bool enable) = 0;
+  virtual bool AGC() const = 0;
 
-    // Microphone Automatic Gain Control (AGC)
-    virtual int32_t SetAGC(bool enable) = 0;
-    virtual bool AGC() const = 0;
+  // Volume control based on the Windows Wave API (Windows only)
+  virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
+                                   uint16_t volumeRight) = 0;
+  virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
+                                uint16_t& volumeRight) const = 0;
 
-    // Volume control based on the Windows Wave API (Windows only)
-    virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
-                                     uint16_t volumeRight) = 0;
-    virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
-                                  uint16_t& volumeRight) const = 0;
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() = 0;
+  virtual bool SpeakerIsInitialized() const = 0;
+  virtual int32_t InitMicrophone() = 0;
+  virtual bool MicrophoneIsInitialized() const = 0;
 
-	// Audio mixer initialization
-    virtual int32_t InitSpeaker() = 0;
-    virtual bool SpeakerIsInitialized() const = 0;
-    virtual int32_t InitMicrophone() = 0;
-    virtual bool MicrophoneIsInitialized() const = 0;
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available) = 0;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+  virtual int32_t SpeakerVolume(uint32_t& volume) const = 0;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const = 0;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const = 0;
+  virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const = 0;
 
-    // Speaker volume controls
-	virtual int32_t SpeakerVolumeIsAvailable(bool& available) = 0;
-    virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
-    virtual int32_t SpeakerVolume(uint32_t& volume) const = 0;
-    virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const = 0;
-    virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const = 0;
-    virtual int32_t SpeakerVolumeStepSize(
-        uint16_t& stepSize) const = 0;
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const = 0;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const = 0;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const = 0;
+  virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const = 0;
 
-    // Microphone volume controls
-	virtual int32_t MicrophoneVolumeIsAvailable(bool& available) = 0;
-    virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
-    virtual int32_t MicrophoneVolume(uint32_t& volume) const = 0;
-    virtual int32_t MaxMicrophoneVolume(
-        uint32_t& maxVolume) const = 0;
-    virtual int32_t MinMicrophoneVolume(
-        uint32_t& minVolume) const = 0;
-    virtual int32_t MicrophoneVolumeStepSize(
-        uint16_t& stepSize) const = 0;
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available) = 0;
+  virtual int32_t SetSpeakerMute(bool enable) = 0;
+  virtual int32_t SpeakerMute(bool& enabled) const = 0;
 
-    // Speaker mute control
-    virtual int32_t SpeakerMuteIsAvailable(bool& available) = 0;
-    virtual int32_t SetSpeakerMute(bool enable) = 0;
-    virtual int32_t SpeakerMute(bool& enabled) const = 0;
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneMute(bool enable) = 0;
+  virtual int32_t MicrophoneMute(bool& enabled) const = 0;
 
-	// Microphone mute control
-    virtual int32_t MicrophoneMuteIsAvailable(bool& available) = 0;
-    virtual int32_t SetMicrophoneMute(bool enable) = 0;
-    virtual int32_t MicrophoneMute(bool& enabled) const = 0;
+  // Microphone boost control
+  virtual int32_t MicrophoneBoostIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneBoost(bool enable) = 0;
+  virtual int32_t MicrophoneBoost(bool& enabled) const = 0;
 
-    // Microphone boost control
-    virtual int32_t MicrophoneBoostIsAvailable(bool& available) = 0;
-	virtual int32_t SetMicrophoneBoost(bool enable) = 0;
-    virtual int32_t MicrophoneBoost(bool& enabled) const = 0;
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available) = 0;
+  virtual int32_t SetStereoPlayout(bool enable) = 0;
+  virtual int32_t StereoPlayout(bool& enabled) const = 0;
+  virtual int32_t StereoRecordingIsAvailable(bool& available) = 0;
+  virtual int32_t SetStereoRecording(bool enable) = 0;
+  virtual int32_t StereoRecording(bool& enabled) const = 0;
 
-    // Stereo support
-    virtual int32_t StereoPlayoutIsAvailable(bool& available) = 0;
-	virtual int32_t SetStereoPlayout(bool enable) = 0;
-    virtual int32_t StereoPlayout(bool& enabled) const = 0;
-    virtual int32_t StereoRecordingIsAvailable(bool& available) = 0;
-    virtual int32_t SetStereoRecording(bool enable) = 0;
-    virtual int32_t StereoRecording(bool& enabled) const = 0;
+  // Delay information and control
+  virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+                                   uint16_t sizeMS = 0) = 0;
+  virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                uint16_t& sizeMS) const = 0;
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const = 0;
+  virtual int32_t RecordingDelay(uint16_t& delayMS) const = 0;
 
-    // Delay information and control
-	virtual int32_t SetPlayoutBuffer(
-        const AudioDeviceModule::BufferType type,
-        uint16_t sizeMS = 0) = 0;
-    virtual int32_t PlayoutBuffer(
-        AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const = 0;
-    virtual int32_t PlayoutDelay(uint16_t& delayMS) const = 0;
-	virtual int32_t RecordingDelay(uint16_t& delayMS) const = 0;
+  // CPU load
+  virtual int32_t CPULoad(uint16_t& load) const = 0;
 
-    // CPU load
-    virtual int32_t CPULoad(uint16_t& load) const = 0;
+  // Native sample rate controls (samples/sec)
+  virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec);
+  virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec);
 
-    // Native sample rate controls (samples/sec)
-	virtual int32_t SetRecordingSampleRate(
-        const uint32_t samplesPerSec);
-	virtual int32_t SetPlayoutSampleRate(
-        const uint32_t samplesPerSec);
+  // Speaker audio routing (for mobile devices)
+  virtual int32_t SetLoudspeakerStatus(bool enable);
+  virtual int32_t GetLoudspeakerStatus(bool& enable) const;
 
-    // Speaker audio routing (for mobile devices)
-    virtual int32_t SetLoudspeakerStatus(bool enable);
-    virtual int32_t GetLoudspeakerStatus(bool& enable) const;
+  // Reset Audio Device (for mobile devices)
+  virtual int32_t ResetAudioDevice();
 
-    // Reset Audio Device (for mobile devices)
-    virtual int32_t ResetAudioDevice();
+  // Sound Audio Device control (for WinCE only)
+  virtual int32_t SoundDeviceControl(unsigned int par1 = 0,
+                                     unsigned int par2 = 0,
+                                     unsigned int par3 = 0,
+                                     unsigned int par4 = 0);
 
-    // Sound Audio Device control (for WinCE only)
-    virtual int32_t SoundDeviceControl(unsigned int par1 = 0,
-                                       unsigned int par2 = 0,
-                                       unsigned int par3 = 0,
-                                       unsigned int par4 = 0);
+  // Android only
+  virtual bool BuiltInAECIsAvailable() const;
+  virtual bool BuiltInAGCIsAvailable() const;
+  virtual bool BuiltInNSIsAvailable() const;
 
-    // Android only
-    virtual bool BuiltInAECIsAvailable() const;
+  // Windows Core Audio and Android only.
+  virtual int32_t EnableBuiltInAEC(bool enable);
+  virtual int32_t EnableBuiltInAGC(bool enable);
+  virtual int32_t EnableBuiltInNS(bool enable);
 
-    // Windows Core Audio and Android only.
-    virtual int32_t EnableBuiltInAEC(bool enable);
+  // Windows Core Audio only.
+  virtual bool BuiltInAECIsEnabled() const;
 
-    // Windows Core Audio only.
-    virtual bool BuiltInAECIsEnabled() const;
+  // iOS only.
+  // TODO(henrika): add Android support.
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const;
+  virtual int GetRecordAudioParameters(AudioParameters* params) const;
 
-public:
-    virtual bool PlayoutWarning() const = 0;
-    virtual bool PlayoutError() const = 0;
-    virtual bool RecordingWarning() const = 0;
-    virtual bool RecordingError() const = 0;
-    virtual void ClearPlayoutWarning() = 0;
-    virtual void ClearPlayoutError() = 0;
-    virtual void ClearRecordingWarning() = 0;
-    virtual void ClearRecordingError() = 0;
+  virtual bool PlayoutWarning() const = 0;
+  virtual bool PlayoutError() const = 0;
+  virtual bool RecordingWarning() const = 0;
+  virtual bool RecordingError() const = 0;
+  virtual void ClearPlayoutWarning() = 0;
+  virtual void ClearPlayoutError() = 0;
+  virtual void ClearRecordingWarning() = 0;
+  virtual void ClearRecordingError() = 0;
 
-public:
-    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
 
-    virtual ~AudioDeviceGeneric() {}
+  virtual ~AudioDeviceGeneric() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc
index b701ff9289..097d919f87 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc
@@ -11,7 +11,8 @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 #include <assert.h>
 #include <string.h>
@@ -19,40 +20,31 @@
 #if defined(WEBRTC_DUMMY_AUDIO_BUILD)
 // do not include platform specific headers
 #elif defined(_WIN32)
-    #include "audio_device_utility_win.h"
     #include "audio_device_wave_win.h"
  #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
     #include "audio_device_core_win.h"
  #endif
 #elif defined(WEBRTC_ANDROID_OPENSLES)
-// ANDROID and GONK
-    #include <stdlib.h>
-    #include <dlfcn.h>
-    #include "audio_device_utility_android.h"
-    #include "webrtc/modules/audio_device/android/audio_device_template.h"
-#if !defined(WEBRTC_GONK)
-// GONK only supports opensles; android can use that or jni
-    #include "webrtc/modules/audio_device/android/audio_record_jni.h"
-    #include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#endif
-    #include "webrtc/modules/audio_device/android/opensles_input.h"
-    #include "webrtc/modules/audio_device/android/opensles_output.h"
+#include <stdlib.h>
+#include <dlfcn.h>
+#include "webrtc/modules/audio_device/android/audio_device_template.h"
+#include "webrtc/modules/audio_device/android/audio_manager.h"
+#include "webrtc/modules/audio_device/android/audio_record_jni.h"
+#include "webrtc/modules/audio_device/android/audio_track_jni.h"
+#include "webrtc/modules/audio_device/android/opensles_player.h"
 #elif defined(WEBRTC_AUDIO_SNDIO)
-    #include "audio_device_utility_sndio.h"
-    #include "audio_device_sndio.h"
+#include "audio_device_utility_sndio.h"
+#include "audio_device_sndio.h"
 #elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
-    #include "audio_device_utility_linux.h"
  #if defined(LINUX_ALSA)
-    #include "audio_device_alsa_linux.h"
+   #include "audio_device_alsa_linux.h"
  #endif
- #if defined(LINUX_PULSE)
+#if defined(LINUX_PULSE)
     #include "audio_device_pulse_linux.h"
- #endif
+#endif
 #elif defined(WEBRTC_IOS)
-    #include "audio_device_utility_ios.h"
     #include "audio_device_ios.h"
 #elif defined(WEBRTC_MAC)
-    #include "audio_device_utility_mac.h"
     #include "audio_device_mac.h"
 #endif
 
@@ -61,10 +53,9 @@
 #endif
 
 #include "webrtc/modules/audio_device/dummy/audio_device_dummy.h"
-#include "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #define CHECK_INITIALIZED()         \
 {                                   \
@@ -88,7 +79,6 @@ AudioDeviceModule* CreateAudioDeviceModule(
   return AudioDeviceModuleImpl::Create(id, audioLayer);
 }
 
-
 // ============================================================================
 //                                   Static methods
 // ============================================================================
@@ -144,11 +134,10 @@ AudioDeviceModuleImpl::AudioDeviceModuleImpl(const int32_t id, const AudioLayer
     _critSectEventCb(*CriticalSectionWrapper::CreateCriticalSection()),
     _critSectAudioCb(*CriticalSectionWrapper::CreateCriticalSection()),
     _ptrCbAudioDeviceObserver(NULL),
-    _ptrAudioDeviceUtility(NULL),
     _ptrAudioDevice(NULL),
     _id(id),
     _platformAudioLayer(audioLayer),
-    _lastProcessTime(AudioDeviceUtility::GetTimeInMS()),
+    _lastProcessTime(TickTime::MillisecondTimestamp()),
     _platformType(kPlatformNotSupported),
     _initialized(false),
     _lastError(kAdmErrNone)
@@ -211,24 +200,14 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
     AudioDeviceGeneric* ptrAudioDevice(NULL);
-    AudioDeviceUtility* ptrAudioDeviceUtility(NULL);
 
 #if defined(WEBRTC_DUMMY_AUDIO_BUILD)
     ptrAudioDevice = new AudioDeviceDummy(Id());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Dummy Audio APIs will be utilized");
-
-    if (ptrAudioDevice != NULL)
-    {
-        ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
-    }
 #elif defined(WEBRTC_DUMMY_FILE_DEVICES)
     ptrAudioDevice = FileAudioDeviceFactory::CreateFileAudioDevice(Id());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "Will use file-playing dummy device.");
-    if (ptrAudioDevice != NULL)
-    {
-        ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
-    }
 #else
     AudioLayer audioLayer(PlatformAudioLayer());
 
@@ -272,69 +251,58 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         }
     }
 #endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
-    if (ptrAudioDevice != NULL)
-    {
-        // Create the Windows implementation of the Device Utility.
-        // This class is independent of the selected audio layer
-        // for Windows.
-        //
-        ptrAudioDeviceUtility = new AudioDeviceUtilityWindows(Id());
-    }
 #endif  // #if defined(_WIN32)
 
-    // Create the *Android OpenSLES* implementation of the Audio Device
-    //
-#if defined(WEBRTC_ANDROID) || defined (WEBRTC_GONK)
+#if defined(WEBRTC_ANDROID)
+    // Create an Android audio manager.
+    _audioManagerAndroid.reset(new AudioManager());
+    // Select best possible combination of audio layers.
     if (audioLayer == kPlatformDefaultAudio) {
-    // AudioRecordJni provides hardware AEC and OpenSlesOutput low latency.
-#if defined (WEBRTC_ANDROID_OPENSLES)
-      // Android and Gonk
-      // Check if the OpenSLES library is available before going further.
-      void* opensles_lib = dlopen("libOpenSLES.so", RTLD_LAZY);
-      if (opensles_lib) {
-        // That worked, close for now and proceed normally.
-        dlclose(opensles_lib);
-        if (audioLayer == kPlatformDefaultAudio)
-          {
-            // Create *Android OpenSLES Audio* implementation
-            ptrAudioDevice = new AudioDeviceTemplate<OpenSlesInput, OpenSlesOutput>(Id());
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Android OpenSLES Audio APIs will be utilized");
-          }
+      // Use Java-based audio in both directions when low-latency output
+      // is not supported.
+      audioLayer =  kAndroidJavaAudio;
+      //Check to see if low latency output is allowed for this device.
+      if (_audioManagerAndroid->IsLowLatencyPlayoutSupported()) {
+        // Always use OpenSL ES for output on devices that supports the
+        // low-latency output audio path.
+        // Check if the OpenSLES library is available before going further.
+        void* opensles_lib = dlopen("libOpenSLES.so", RTLD_LAZY);
+        if (opensles_lib) {
+          // That worked, close for now and proceed normally.
+          dlclose(opensles_lib);
+          audioLayer = kAndroidJavaInputAndOpenSLESOutputAudio;
+        }
       }
-#endif // defined (WEBRTC_ANDROID_OPENSLES)
-#if !defined(WEBRTC_GONK)
-      // Fall back to this case if on Android 2.2/OpenSLES not available.
-      if (ptrAudioDevice == NULL) {
-        // Create the *Android Java* implementation of the Audio Device
-        if (audioLayer == kPlatformDefaultAudio)
-          {
-            // Create *Android JNI Audio* implementation
-            ptrAudioDevice = new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(Id());
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Android JNI Audio APIs will be utilized");
-          }
-      }
-#endif // !defined (WEBRTC_GONK)
     }
 
-    if (ptrAudioDevice != NULL) {
-        // Create the Android implementation of the Device Utility.
-        ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
+    AudioManager* audio_manager = _audioManagerAndroid.get();
+    if (audioLayer == kAndroidJavaAudio) {
+      // Java audio for both input and output audio.
+      ptrAudioDevice = new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
+          audioLayer, audio_manager);
+    } else if (audioLayer == kAndroidJavaInputAndOpenSLESOutputAudio) {
+      // Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
+      // This combination provides low-latency output audio and at the same
+      // time support for HW AEC using the AudioRecord Java API.
+      ptrAudioDevice = new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
+          audioLayer, audio_manager);
+    } else {
+      // Invalid audio layer.
+      ptrAudioDevice = NULL;
     }
-
+    // END #if defined(WEBRTC_ANDROID)
 #elif defined(WEBRTC_AUDIO_SNDIO)
     ptrAudioDevice = new AudioDeviceSndio(Id());
     if (ptrAudioDevice != NULL)
     {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "attempting to use the Sndio audio API...");
-        _platformAudioLayer = kSndioAudio;
-        // Create the sndio implementation of the Device Utility.
-        ptrAudioDeviceUtility = new AudioDeviceUtilitySndio(Id());
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "attempting to use the Sndio audio API...");
+      _platformAudioLayer = kSndioAudio;
+      // Create the sndio implementation of the Device Utility.
+      ptrAudioDeviceUtility = new AudioDeviceUtilitySndio(Id());
     }
-
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     // Create the *Linux* implementation of the Audio Device
     //
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     if ((audioLayer == kLinuxPulseAudio) || (audioLayer == kPlatformDefaultAudio))
     {
 #if defined(LINUX_PULSE)
@@ -373,15 +341,6 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Linux ALSA APIs will be utilized");
 #endif
     }
-
-    if (ptrAudioDevice != NULL)
-    {
-        // Create the Linux implementation of the Device Utility.
-        // This class is independent of the selected audio layer
-        // for Linux.
-        //
-        ptrAudioDeviceUtility = new AudioDeviceUtilityLinux(Id());
-    }
 #endif  // #if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
 
     // Create the *iPhone* implementation of the Audio Device
@@ -390,15 +349,9 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
     if (audioLayer == kPlatformDefaultAudio)
     {
         // Create iOS Audio Device implementation.
-        ptrAudioDevice = new AudioDeviceIOS(Id());
+      ptrAudioDevice = new AudioDeviceIOS();
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "iPhone Audio APIs will be utilized");
     }
-
-    if (ptrAudioDevice != NULL)
-    {
-        // Create iOS Device Utility implementation.
-        ptrAudioDeviceUtility = new AudioDeviceUtilityIOS(Id());
-    }
     // END #if defined(WEBRTC_IOS)
 
     // Create the *Mac* implementation of the Audio Device
@@ -410,12 +363,6 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         ptrAudioDevice = new AudioDeviceMac(Id());
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Mac OS X Audio APIs will be utilized");
     }
-
-    if (ptrAudioDevice != NULL)
-    {
-        // Create the Mac implementation of the Device Utility.
-        ptrAudioDeviceUtility = new AudioDeviceUtilityMac(Id());
-    }
 #endif  // WEBRTC_MAC
 
     // Create the *Dummy* implementation of the Audio Device
@@ -427,11 +374,6 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         assert(!ptrAudioDevice);
         ptrAudioDevice = new AudioDeviceDummy(Id());
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Dummy Audio APIs will be utilized");
-
-        if (ptrAudioDevice != NULL)
-        {
-            ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
-        }
     }
 #endif  // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
 
@@ -441,16 +383,9 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         return -1;
     }
 
-    if (ptrAudioDeviceUtility == NULL)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "unable to create the platform specific audio device utility");
-        return -1;
-    }
-
     // Store valid output pointers
     //
     _ptrAudioDevice = ptrAudioDevice;
-    _ptrAudioDeviceUtility = ptrAudioDeviceUtility;
 
     return 0;
 }
@@ -486,12 +421,6 @@ AudioDeviceModuleImpl::~AudioDeviceModuleImpl()
         _ptrAudioDevice = NULL;
     }
 
-    if (_ptrAudioDeviceUtility)
-    {
-        delete _ptrAudioDeviceUtility;
-        _ptrAudioDeviceUtility = NULL;
-    }
-
     delete &_critSect;
     delete &_critSectEventCb;
     delete &_critSectAudioCb;
@@ -510,9 +439,9 @@ AudioDeviceModuleImpl::~AudioDeviceModuleImpl()
 
 int64_t AudioDeviceModuleImpl::TimeUntilNextProcess()
 {
-    uint32_t now = AudioDeviceUtility::GetTimeInMS();
-    int32_t deltaProcess = kAdmMaxIdleTimeProcess - (now - _lastProcessTime);
-    return (deltaProcess);
+    int64_t now = TickTime::MillisecondTimestamp();
+    int64_t deltaProcess = kAdmMaxIdleTimeProcess - (now - _lastProcessTime);
+    return deltaProcess;
 }
 
 // ----------------------------------------------------------------------------
@@ -525,7 +454,7 @@ int64_t AudioDeviceModuleImpl::TimeUntilNextProcess()
 int32_t AudioDeviceModuleImpl::Process()
 {
 
-    _lastProcessTime = AudioDeviceUtility::GetTimeInMS();
+    _lastProcessTime = TickTime::MillisecondTimestamp();
 
     // kPlayoutWarning
     if (_ptrAudioDevice->PlayoutWarning())
@@ -586,40 +515,13 @@ int32_t AudioDeviceModuleImpl::Process()
 //  ActiveAudioLayer
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const
-{
-
-    AudioLayer activeAudio;
-
-    if (_ptrAudioDevice->ActiveAudioLayer(activeAudio) == -1)
-    {
-        return -1;
-    }
-
-    *audioLayer = activeAudio;
-
-    if (*audioLayer == AudioDeviceModule::kWindowsWaveAudio)
-    {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsWaveAudio");
-    }
-    else if (*audioLayer == AudioDeviceModule::kWindowsCoreAudio)
-    {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsCoreAudio");
-    }
-    else if (*audioLayer == AudioDeviceModule::kLinuxAlsaAudio)
-    {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kLinuxAlsaAudio");
-    }
-    else if (*audioLayer == AudioDeviceModule::kSndioAudio)
-    {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kSndioAudio");
-    }
-    else
-    {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: NOT_SUPPORTED");
-    }
-
-    return 0;
+int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
+  AudioLayer activeAudio;
+  if (_ptrAudioDevice->ActiveAudioLayer(activeAudio) == -1) {
+    return -1;
+  }
+  *audioLayer = activeAudio;
+  return 0;
 }
 
 // ----------------------------------------------------------------------------
@@ -641,14 +543,9 @@ int32_t AudioDeviceModuleImpl::Init()
     if (_initialized)
         return 0;
 
-    if (!_ptrAudioDeviceUtility)
-        return -1;
-
     if (!_ptrAudioDevice)
         return -1;
 
-    _ptrAudioDeviceUtility->Init();
-
     if (_ptrAudioDevice->Init() == -1)
     {
         return -1;
@@ -1996,29 +1893,17 @@ int32_t AudioDeviceModuleImpl::SetLoudspeakerStatus(bool enable)
 //  GetLoudspeakerStatus
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceModuleImpl::GetLoudspeakerStatus(bool* enabled) const
-{
-    CHECK_INITIALIZED();
-
-    if (_ptrAudioDevice->GetLoudspeakerStatus(*enabled) != 0)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable)
-{
+int32_t AudioDeviceModuleImpl::GetLoudspeakerStatus(bool* enabled) const {
   CHECK_INITIALIZED();
-  return _ptrAudioDevice->EnableBuiltInAEC(enable);
+  if (_ptrAudioDevice->GetLoudspeakerStatus(*enabled) != 0) {
+    return -1;
+  }
+  return 0;
 }
 
-bool AudioDeviceModuleImpl::BuiltInAECIsEnabled() const
-{
-    CHECK_INITIALIZED_BOOL();
-
-    return _ptrAudioDevice->BuiltInAECIsEnabled();
+bool AudioDeviceModuleImpl::BuiltInAECIsEnabled() const {
+  CHECK_INITIALIZED_BOOL();
+  return _ptrAudioDevice->BuiltInAECIsEnabled();
 }
 
 bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
@@ -2026,6 +1911,41 @@ bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
   return _ptrAudioDevice->BuiltInAECIsAvailable();
 }
 
+int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) {
+  CHECK_INITIALIZED();
+  return _ptrAudioDevice->EnableBuiltInAEC(enable);
+}
+
+bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const {
+  CHECK_INITIALIZED_BOOL();
+  return _ptrAudioDevice->BuiltInAGCIsAvailable();
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) {
+  CHECK_INITIALIZED();
+  return _ptrAudioDevice->EnableBuiltInAGC(enable);
+}
+
+bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const {
+  CHECK_INITIALIZED_BOOL();
+  return _ptrAudioDevice->BuiltInNSIsAvailable();
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) {
+  CHECK_INITIALIZED();
+  return _ptrAudioDevice->EnableBuiltInNS(enable);
+}
+
+int AudioDeviceModuleImpl::GetPlayoutAudioParameters(
+    AudioParameters* params) const {
+  return _ptrAudioDevice->GetPlayoutAudioParameters(params);
+}
+
+int AudioDeviceModuleImpl::GetRecordAudioParameters(
+    AudioParameters* params) const {
+  return _ptrAudioDevice->GetRecordAudioParameters(params);
+}
+
 // ============================================================================
 //                                 Private Methods
 // ============================================================================
@@ -2045,39 +1965,6 @@ AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const
 
 AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer() const
 {
-
-    switch (_platformAudioLayer)
-    {
-    case kPlatformDefaultAudio:
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                     "output: kPlatformDefaultAudio");
-        break;
-    case kWindowsWaveAudio:
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                     "output: kWindowsWaveAudio");
-        break;
-    case kWindowsCoreAudio:
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                     "output: kWindowsCoreAudio");
-        break;
-    case kLinuxAlsaAudio:
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                     "output: kLinuxAlsaAudio");
-        break;
-    case kSndioAudio:
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                     "output: kSndioAudio");
-        break;
-    case kDummyAudio:
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                     "output: kDummyAudio");
-        break;
-    default:
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "output: INVALID");
-        break;
-    }
-
     return _platformAudioLayer;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.h b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.h
index e95af70f78..e43a8c491f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.h
@@ -11,217 +11,224 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
 
+#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
 class AudioDeviceGeneric;
-class AudioDeviceUtility;
+class AudioManager;
 class CriticalSectionWrapper;
 
-class AudioDeviceModuleImpl : public AudioDeviceModule
-{
-public:
-    enum PlatformType
-    {
-        kPlatformNotSupported = 0,
-        kPlatformWin32 = 1,
-        kPlatformWinCe = 2,
-        kPlatformLinux = 3,
-        kPlatformMac = 4,
-        kPlatformAndroid = 5,
-        kPlatformIOS = 6,
-        kPlatformSndio = 7
-    };
+class AudioDeviceModuleImpl : public AudioDeviceModule {
+ public:
+  enum PlatformType {
+    kPlatformNotSupported = 0,
+    kPlatformWin32 = 1,
+    kPlatformWinCe = 2,
+    kPlatformLinux = 3,
+    kPlatformMac = 4,
+    kPlatformAndroid = 5,
+    kPlatformIOS = 6,
+    kPlatformSndio = 7
+  };
 
-    int32_t CheckPlatform();
-    int32_t CreatePlatformSpecificObjects();
-    int32_t AttachAudioBuffer();
+  int32_t CheckPlatform();
+  int32_t CreatePlatformSpecificObjects();
+  int32_t AttachAudioBuffer();
 
-    AudioDeviceModuleImpl(const int32_t id, const AudioLayer audioLayer);
-    virtual ~AudioDeviceModuleImpl();
+  AudioDeviceModuleImpl(const int32_t id, const AudioLayer audioLayer);
+  virtual ~AudioDeviceModuleImpl();
 
-public: // RefCountedModule
- int64_t TimeUntilNextProcess() override;
- int32_t Process() override;
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
 
-public:
-    // Factory methods (resource allocation/deallocation)
-    static AudioDeviceModule* Create(
-        const int32_t id,
-        const AudioLayer audioLayer = kPlatformDefaultAudio);
+  // Factory methods (resource allocation/deallocation)
+  static AudioDeviceModule* Create(
+      const int32_t id,
+      const AudioLayer audioLayer = kPlatformDefaultAudio);
 
-    // Retrieve the currently utilized audio layer
-    int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
 
-    // Error handling
-    ErrorCode LastError() const override;
-    int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) override;
+  // Error handling
+  ErrorCode LastError() const override;
+  int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) override;
 
-    // Full-duplex transportation of PCM audio
-    int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
+  // Full-duplex transportation of PCM audio
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
 
-    // Main initializaton and termination
-    int32_t Init() override;
-    int32_t Terminate() override;
-    bool Initialized() const override;
+  // Main initializaton and termination
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
 
-    // Device enumeration
-    int16_t PlayoutDevices() override;
-    int16_t RecordingDevices() override;
-    int32_t PlayoutDeviceName(uint16_t index,
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
                               char name[kAdmMaxDeviceNameSize],
                               char guid[kAdmMaxGuidSize]) override;
-    int32_t RecordingDeviceName(uint16_t index,
-                                char name[kAdmMaxDeviceNameSize],
-                                char guid[kAdmMaxGuidSize]) override;
 
-    // Device selection
-    int32_t SetPlayoutDevice(uint16_t index) override;
-    int32_t SetPlayoutDevice(WindowsDeviceType device) override;
-    int32_t SetRecordingDevice(uint16_t index) override;
-    int32_t SetRecordingDevice(WindowsDeviceType device) override;
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
 
-    // Audio transport initialization
-    int32_t PlayoutIsAvailable(bool* available) override;
-    int32_t InitPlayout() override;
-    bool PlayoutIsInitialized() const override;
-    int32_t RecordingIsAvailable(bool* available) override;
-    int32_t InitRecording() override;
-    bool RecordingIsInitialized() const override;
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool* available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
 
-    // Audio transport control
-    int32_t StartPlayout() override;
-    int32_t StopPlayout() override;
-    bool Playing() const override;
-    int32_t StartRecording() override;
-    int32_t StopRecording() override;
-    bool Recording() const override;
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
 
-    // Microphone Automatic Gain Control (AGC)
-    int32_t SetAGC(bool enable) override;
-    bool AGC() const override;
+  // Microphone Automatic Gain Control (AGC)
+  int32_t SetAGC(bool enable) override;
+  bool AGC() const override;
 
-    // Volume control based on the Windows Wave API (Windows only)
-    int32_t SetWaveOutVolume(uint16_t volumeLeft,
-                             uint16_t volumeRight) override;
-    int32_t WaveOutVolume(uint16_t* volumeLeft,
-                          uint16_t* volumeRight) const override;
+  // Volume control based on the Windows Wave API (Windows only)
+  int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight) override;
+  int32_t WaveOutVolume(uint16_t* volumeLeft,
+                        uint16_t* volumeRight) const override;
 
-    // Audio mixer initialization
-    int32_t InitSpeaker() override;
-    bool SpeakerIsInitialized() const override;
-    int32_t InitMicrophone() override;
-    bool MicrophoneIsInitialized() const override;
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
 
-    // Speaker volume controls
-    int32_t SpeakerVolumeIsAvailable(bool* available) override;
-    int32_t SetSpeakerVolume(uint32_t volume) override;
-    int32_t SpeakerVolume(uint32_t* volume) const override;
-    int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
-    int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
-    int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const override;
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t* volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+  int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const override;
 
-    // Microphone volume controls
-    int32_t MicrophoneVolumeIsAvailable(bool* available) override;
-    int32_t SetMicrophoneVolume(uint32_t volume) override;
-    int32_t MicrophoneVolume(uint32_t* volume) const override;
-    int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
-    int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
-    int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const override;
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+  int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const override;
 
-    // Speaker mute control
-    int32_t SpeakerMuteIsAvailable(bool* available) override;
-    int32_t SetSpeakerMute(bool enable) override;
-    int32_t SpeakerMute(bool* enabled) const override;
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool* available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool* enabled) const override;
 
-    // Microphone mute control
-    int32_t MicrophoneMuteIsAvailable(bool* available) override;
-    int32_t SetMicrophoneMute(bool enable) override;
-    int32_t MicrophoneMute(bool* enabled) const override;
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool* enabled) const override;
 
-    // Microphone boost control
-    int32_t MicrophoneBoostIsAvailable(bool* available) override;
-    int32_t SetMicrophoneBoost(bool enable) override;
-    int32_t MicrophoneBoost(bool* enabled) const override;
+  // Microphone boost control
+  int32_t MicrophoneBoostIsAvailable(bool* available) override;
+  int32_t SetMicrophoneBoost(bool enable) override;
+  int32_t MicrophoneBoost(bool* enabled) const override;
 
-    // Stereo support
-    int32_t StereoPlayoutIsAvailable(bool* available) const override;
-    int32_t SetStereoPlayout(bool enable) override;
-    int32_t StereoPlayout(bool* enabled) const override;
-    int32_t StereoRecordingIsAvailable(bool* available) const override;
-    int32_t SetStereoRecording(bool enable) override;
-    int32_t StereoRecording(bool* enabled) const override;
-    int32_t SetRecordingChannel(const ChannelType channel) override;
-    int32_t RecordingChannel(ChannelType* channel) const override;
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool* enabled) const override;
+  int32_t SetRecordingChannel(const ChannelType channel) override;
+  int32_t RecordingChannel(ChannelType* channel) const override;
 
-    // Delay information and control
-    int32_t SetPlayoutBuffer(const BufferType type,
-                             uint16_t sizeMS = 0) override;
-    int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const override;
-    int32_t PlayoutDelay(uint16_t* delayMS) const override;
-    int32_t RecordingDelay(uint16_t* delayMS) const override;
+  // Delay information and control
+  int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0) override;
+  int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const override;
+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
+  int32_t RecordingDelay(uint16_t* delayMS) const override;
 
-    // CPU load
-    int32_t CPULoad(uint16_t* load) const override;
+  // CPU load
+  int32_t CPULoad(uint16_t* load) const override;
 
-    // Recording of raw PCM data
-    int32_t StartRawOutputFileRecording(
-        const char pcmFileNameUTF8[kAdmMaxFileNameSize]) override;
-    int32_t StopRawOutputFileRecording() override;
-    int32_t StartRawInputFileRecording(
-        const char pcmFileNameUTF8[kAdmMaxFileNameSize]) override;
-    int32_t StopRawInputFileRecording() override;
+  // Recording of raw PCM data
+  int32_t StartRawOutputFileRecording(
+      const char pcmFileNameUTF8[kAdmMaxFileNameSize]) override;
+  int32_t StopRawOutputFileRecording() override;
+  int32_t StartRawInputFileRecording(
+      const char pcmFileNameUTF8[kAdmMaxFileNameSize]) override;
+  int32_t StopRawInputFileRecording() override;
 
-    // Native sample rate controls (samples/sec)
-    int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override;
-    int32_t RecordingSampleRate(uint32_t* samplesPerSec) const override;
-    int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override;
-    int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const override;
+  // Native sample rate controls (samples/sec)
+  int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override;
+  int32_t RecordingSampleRate(uint32_t* samplesPerSec) const override;
+  int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override;
+  int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const override;
 
-    // Mobile device specific functions
-    int32_t ResetAudioDevice() override;
-    int32_t SetLoudspeakerStatus(bool enable) override;
-    int32_t GetLoudspeakerStatus(bool* enabled) const override;
+  // Mobile device specific functions
+  int32_t ResetAudioDevice() override;
+  int32_t SetLoudspeakerStatus(bool enable) override;
+  int32_t GetLoudspeakerStatus(bool* enabled) const override;
 
-    bool BuiltInAECIsAvailable() const override;
+  bool BuiltInAECIsEnabled() const override;
+  bool BuiltInAECIsAvailable() const override;
+  int32_t EnableBuiltInAEC(bool enable) override;
+  bool BuiltInAGCIsAvailable() const override;
+  int32_t EnableBuiltInAGC(bool enable) override;
+  bool BuiltInNSIsAvailable() const override;
+  int32_t EnableBuiltInNS(bool enable) override;
 
-    int32_t EnableBuiltInAEC(bool enable) override;
-    bool BuiltInAECIsEnabled() const override;
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
 
-public:
-    int32_t Id() {return _id;}
+  int32_t Id() { return _id; }
+#if defined(WEBRTC_ANDROID)
+  // Only use this acccessor for test purposes on Android.
+  AudioManager* GetAndroidAudioManagerForTest() {
+    return _audioManagerAndroid.get();
+  }
+#endif
+  AudioDeviceBuffer* GetAudioDeviceBuffer() { return &_audioDeviceBuffer; }
 
-    AudioDeviceBuffer* GetAudioDeviceBuffer() {
-        return &_audioDeviceBuffer;
-    }
+ private:
+  PlatformType Platform() const;
+  AudioLayer PlatformAudioLayer() const;
 
-private:
-    PlatformType Platform() const;
-    AudioLayer PlatformAudioLayer() const;
+  CriticalSectionWrapper& _critSect;
+  CriticalSectionWrapper& _critSectEventCb;
+  CriticalSectionWrapper& _critSectAudioCb;
 
-private:
-    CriticalSectionWrapper&     _critSect;
-    CriticalSectionWrapper&     _critSectEventCb;
-    CriticalSectionWrapper&     _critSectAudioCb;
+  AudioDeviceObserver* _ptrCbAudioDeviceObserver;
 
-    AudioDeviceObserver*        _ptrCbAudioDeviceObserver;
+  AudioDeviceGeneric* _ptrAudioDevice;
 
-    AudioDeviceUtility*         _ptrAudioDeviceUtility;
-    AudioDeviceGeneric*         _ptrAudioDevice;
-
-    AudioDeviceBuffer           _audioDeviceBuffer;
-
-    int32_t               _id;
-    AudioLayer                  _platformAudioLayer;
-    uint32_t              _lastProcessTime;
-    PlatformType                _platformType;
-    bool                        _initialized;
-    mutable ErrorCode           _lastError;
+  AudioDeviceBuffer _audioDeviceBuffer;
+#if defined(WEBRTC_ANDROID)
+  rtc::scoped_ptr<AudioManager> _audioManagerAndroid;
+#endif
+  int32_t _id;
+  AudioLayer _platformAudioLayer;
+  int64_t _lastProcessTime;
+  PlatformType _platformType;
+  bool _initialized;
+  mutable ErrorCode _lastError;
 };
 
 }  // namespace webrtc
 
+#endif  // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
 #endif  // WEBRTC_MODULES_INTERFACE_AUDIO_DEVICE_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.cc b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.cc
deleted file mode 100644
index 0e54468da9..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-
-#if defined(_WIN32)
-
-// ============================================================================
-//                                     Windows
-// ============================================================================
-
-#include <windows.h>
-#include <conio.h>
-#include <ctype.h>
-#include <stdio.h>
-#include <mmsystem.h>
-
-namespace webrtc
-{
-
-void AudioDeviceUtility::WaitForKey()
-{
-	_getch();
-}
-
-uint32_t AudioDeviceUtility::GetTimeInMS()
-{
-	return timeGetTime();
-}
-
-bool AudioDeviceUtility::StringCompare(
-    const char* str1 , const char* str2,
-    const uint32_t length)
-{
-	return ((_strnicmp(str1, str2, length) == 0) ? true : false);
-}
-
-}  // namespace webrtc
-
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-
-// ============================================================================
-//                                 Linux & Mac
-// ============================================================================
-
-#include <stdio.h>      // getchar
-#include <string.h>     // strncasecmp
-#include <sys/time.h>   // gettimeofday
-#include <termios.h>    // tcgetattr
-#include <time.h>       // gettimeofday
-
-#include <unistd.h>
-
-namespace webrtc
-{
-
-void AudioDeviceUtility::WaitForKey()
-{
-
-    struct termios oldt, newt;
-
-    tcgetattr( STDIN_FILENO, &oldt );
-
-    // we don't want getchar to echo!
-
-    newt = oldt;
-    newt.c_lflag &= ~( ICANON | ECHO );
-    tcsetattr( STDIN_FILENO, TCSANOW, &newt );
-
-    // catch any newline that's hanging around...
-
-    // you'll have to hit enter twice if you
-
-    // choose enter out of all available keys
-
-    if (getc(stdin) == '\n')
-    {
-        getc(stdin);
-    }
-
-    tcsetattr( STDIN_FILENO, TCSANOW, &oldt );
-}
-
-uint32_t AudioDeviceUtility::GetTimeInMS()
-{
-    struct timeval tv;
-    struct timezone tz;
-    uint32_t val;
-
-    gettimeofday(&tv, &tz);
-    val = (uint32_t)(tv.tv_sec*1000 + tv.tv_usec/1000);
-    return val;
-}
-
-bool AudioDeviceUtility::StringCompare(
-    const char* str1 , const char* str2, const uint32_t length)
-{
-    return (strncasecmp(str1, str2, length) == 0)?true: false;
-}
-
-}  // namespace webrtc
-
-#endif  // defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.h b/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.h
deleted file mode 100644
index ebe06d1fca..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class AudioDeviceUtility
-{
- public:
-  static uint32_t GetTimeInMS();
-  static void WaitForKey();
-  static bool StringCompare(const char* str1,
-                            const char* str2,
-                            const uint32_t length);
-  virtual int32_t Init() = 0;
-
-  virtual ~AudioDeviceUtility() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h
deleted file mode 100644
index 90aa6c28da..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
-
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityDummy: public AudioDeviceUtility
-{
-public:
-    AudioDeviceUtilityDummy(const int32_t id) {}
-    virtual ~AudioDeviceUtilityDummy() {}
-
-    int32_t Init() override;
-};
-}  // namespace webrtc
-
-#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_DUMMY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.cc b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.cc
index 82569e8fcf..aac0962a50 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -7,21 +7,20 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <iostream>
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 
 namespace webrtc {
 
-int kRecordingFixedSampleRate = 48000;
-int kRecordingNumChannels = 2;
-int kPlayoutFixedSampleRate = 48000;
-int kPlayoutNumChannels = 2;
-int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100
-                         * kPlayoutNumChannels * 2;
-int kRecordingBufferSize = kRecordingFixedSampleRate / 100
-                           * kRecordingNumChannels * 2;
+const int kRecordingFixedSampleRate = 48000;
+const size_t kRecordingNumChannels = 2;
+const int kPlayoutFixedSampleRate = 48000;
+const size_t kPlayoutNumChannels = 2;
+const size_t kPlayoutBufferSize =
+    kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2;
+const size_t kRecordingBufferSize =
+    kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
 
 FileAudioDevice::FileAudioDevice(const int32_t id,
                                  const char* inputFilename,
@@ -172,7 +171,7 @@ int32_t FileAudioDevice::InitRecording() {
     return -1;
   }
 
-  _recordingFramesIn10MS = kRecordingFixedSampleRate/100;
+  _recordingFramesIn10MS = static_cast<size_t>(kRecordingFixedSampleRate / 100);
 
   if (_ptrAudioBuffer) {
     _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
@@ -190,14 +189,12 @@ int32_t FileAudioDevice::StartPlayout() {
       return 0;
   }
 
-  _playoutFramesIn10MS = kPlayoutFixedSampleRate/100;
+  _playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
   _playing = true;
   _playoutFramesLeft = 0;
 
   if (!_playoutBuffer) {
-      _playoutBuffer = new int8_t[2 *
-                                  kPlayoutNumChannels *
-                                  kPlayoutFixedSampleRate/100];
+      _playoutBuffer = new int8_t[kPlayoutBufferSize];
   }
   if (!_playoutBuffer) {
     _playing = false;
@@ -214,17 +211,10 @@ int32_t FileAudioDevice::StartPlayout() {
     return -1;
   }
 
-  const char* threadName = "webrtc_audio_module_play_thread";
-  _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
-                                               threadName);
-  if (!_ptrThreadPlay->Start()) {
-      _ptrThreadPlay.reset();
-      _playing = false;
-      delete [] _playoutBuffer;
-      _playoutBuffer = NULL;
-      return -1;
-  }
-  _ptrThreadPlay->SetPriority(kRealtimePriority);
+  _ptrThreadPlay.reset(new rtc::PlatformThread(
+      PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+  _ptrThreadPlay->Start();
+  _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
   return 0;
 }
 
@@ -277,17 +267,11 @@ int32_t FileAudioDevice::StartRecording() {
     return -1;
   }
 
-  const char* threadName = "webrtc_audio_module_capture_thread";
-  _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this, threadName);
+  _ptrThreadRec.reset(new rtc::PlatformThread(
+      RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
-  if (!_ptrThreadRec->Start()) {
-      _ptrThreadRec.reset();
-      _recording = false;
-      delete [] _recordingBuffer;
-      _recordingBuffer = NULL;
-      return -1;
-  }
-  _ptrThreadRec->SetPriority(kRealtimePriority);
+  _ptrThreadRec->Start();
+  _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
   return 0;
 }
@@ -514,7 +498,12 @@ bool FileAudioDevice::PlayThreadProcess()
     }
     _playoutFramesLeft = 0;
     _critSect.Leave();
-    SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
+
+    uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
+    if(deltaTimeMillis < 10) {
+      SleepMs(10 - deltaTimeMillis);
+    }
+
     return true;
 }
 
@@ -544,7 +533,12 @@ bool FileAudioDevice::RecThreadProcess()
     }
 
     _critSect.Leave();
-    SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
+
+    uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
+    if(deltaTimeMillis < 10) {
+      SleepMs(10 - deltaTimeMillis);
+    }
+
     return true;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.h b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.h
index ffc8adc016..77179409ea 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device.h
@@ -16,13 +16,16 @@
 #include <string>
 
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace rtc {
+class PlatformThread;
+}  // namespace rtc
 
 namespace webrtc {
 class EventWrapper;
-class ThreadWrapper;
 
 // This is a fake audio device which plays audio from a file as its microphone
 // and plays out into a file.
@@ -174,12 +177,13 @@ class FileAudioDevice : public AudioDeviceGeneric {
   uint32_t _playoutFramesLeft;
   CriticalSectionWrapper& _critSect;
 
-  uint32_t _recordingBufferSizeIn10MS;
-  uint32_t _recordingFramesIn10MS;
-  uint32_t _playoutFramesIn10MS;
+  size_t _recordingBufferSizeIn10MS;
+  size_t _recordingFramesIn10MS;
+  size_t _playoutFramesIn10MS;
 
-  rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
-  rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
+  // TODO(pbos): Make plain members instead of pointers and stop resetting them.
+  rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
+  rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
   bool _playing;
   bool _recording;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device_factory.h b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device_factory.h
index 96cab67835..250b7f63a8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device_factory.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/dummy/file_audio_device_factory.h
@@ -31,7 +31,7 @@ class FileAudioDeviceFactory {
                                 const char* outputAudioFilename);
 
  private:
-  static const uint32_t MAX_FILENAME_LEN = 256;
+  static const uint32_t MAX_FILENAME_LEN = 512;
   static bool _isConfigured;
   static char _inputAudioFilename[MAX_FILENAME_LEN];
   static char _outputAudioFilename[MAX_FILENAME_LEN];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer.cc
new file mode 100644
index 0000000000..7fffdd14fb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer.cc
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_device/fine_audio_buffer.h"
+
+#include <memory.h>
+#include <stdio.h>
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_device/audio_device_buffer.h"
+
+namespace webrtc {
+
+FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
+                                 size_t desired_frame_size_bytes,
+                                 int sample_rate)
+    : device_buffer_(device_buffer),
+      desired_frame_size_bytes_(desired_frame_size_bytes),
+      sample_rate_(sample_rate),
+      samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
+      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
+      playout_cached_buffer_start_(0),
+      playout_cached_bytes_(0),
+      // Allocate extra space on the recording side to reduce the number of
+      // memmove() calls.
+      required_record_buffer_size_bytes_(
+          5 * (desired_frame_size_bytes + bytes_per_10_ms_)),
+      record_cached_bytes_(0),
+      record_read_pos_(0),
+      record_write_pos_(0) {
+  playout_cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
+  record_cache_buffer_.reset(new int8_t[required_record_buffer_size_bytes_]);
+  memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
+}
+
+FineAudioBuffer::~FineAudioBuffer() {}
+
+size_t FineAudioBuffer::RequiredPlayoutBufferSizeBytes() {
+  // It is possible that we store the desired frame size - 1 samples. Since new
+  // audio frames are pulled in chunks of 10ms we will need a buffer that can
+  // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
+  return desired_frame_size_bytes_ + bytes_per_10_ms_;
+}
+
+void FineAudioBuffer::ResetPlayout() {
+  playout_cached_buffer_start_ = 0;
+  playout_cached_bytes_ = 0;
+  memset(playout_cache_buffer_.get(), 0, bytes_per_10_ms_);
+}
+
+void FineAudioBuffer::ResetRecord() {
+  record_cached_bytes_ = 0;
+  record_read_pos_ = 0;
+  record_write_pos_ = 0;
+  memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
+}
+
+void FineAudioBuffer::GetPlayoutData(int8_t* buffer) {
+  if (desired_frame_size_bytes_ <= playout_cached_bytes_) {
+    memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
+           desired_frame_size_bytes_);
+    playout_cached_buffer_start_ += desired_frame_size_bytes_;
+    playout_cached_bytes_ -= desired_frame_size_bytes_;
+    RTC_CHECK_LT(playout_cached_buffer_start_ + playout_cached_bytes_,
+                 bytes_per_10_ms_);
+    return;
+  }
+  memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
+         playout_cached_bytes_);
+  // Push another n*10ms of audio to |buffer|. n > 1 if
+  // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
+  // write the audio after the cached bytes copied earlier.
+  int8_t* unwritten_buffer = &buffer[playout_cached_bytes_];
+  int bytes_left =
+      static_cast<int>(desired_frame_size_bytes_ - playout_cached_bytes_);
+  // Ceiling of integer division: 1 + ((x - 1) / y)
+  size_t number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
+  for (size_t i = 0; i < number_of_requests; ++i) {
+    device_buffer_->RequestPlayoutData(samples_per_10_ms_);
+    int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
+    if (static_cast<size_t>(num_out) != samples_per_10_ms_) {
+      RTC_CHECK_EQ(num_out, 0);
+      playout_cached_bytes_ = 0;
+      return;
+    }
+    unwritten_buffer += bytes_per_10_ms_;
+    RTC_CHECK_GE(bytes_left, 0);
+    bytes_left -= static_cast<int>(bytes_per_10_ms_);
+  }
+  RTC_CHECK_LE(bytes_left, 0);
+  // Put the samples that were written to |buffer| but are not used in the
+  // cache.
+  size_t cache_location = desired_frame_size_bytes_;
+  int8_t* cache_ptr = &buffer[cache_location];
+  playout_cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
+                          (desired_frame_size_bytes_ - playout_cached_bytes_);
+  // If playout_cached_bytes_ is larger than the cache buffer, uninitialized
+  // memory will be read.
+  RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
+  RTC_CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_);
+  playout_cached_buffer_start_ = 0;
+  memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_);
+}
+
+void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
+                                          size_t size_in_bytes,
+                                          int playout_delay_ms,
+                                          int record_delay_ms) {
+  // Check if the temporary buffer can store the incoming buffer. If not,
+  // move the remaining (old) bytes to the beginning of the temporary buffer
+  // and start adding new samples after the old samples.
+  if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) {
+    if (record_cached_bytes_ > 0) {
+      memmove(record_cache_buffer_.get(),
+              record_cache_buffer_.get() + record_read_pos_,
+              record_cached_bytes_);
+    }
+    record_write_pos_ = record_cached_bytes_;
+    record_read_pos_ = 0;
+  }
+  // Add recorded samples to a temporary buffer.
+  memcpy(record_cache_buffer_.get() + record_write_pos_, buffer, size_in_bytes);
+  record_write_pos_ += size_in_bytes;
+  record_cached_bytes_ += size_in_bytes;
+  // Consume samples in temporary buffer in chunks of 10ms until there is not
+  // enough data left. The number of remaining bytes in the cache is given by
+  // |record_cached_bytes_| after this while loop is done.
+  while (record_cached_bytes_ >= bytes_per_10_ms_) {
+    device_buffer_->SetRecordedBuffer(
+        record_cache_buffer_.get() + record_read_pos_, samples_per_10_ms_);
+    device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
+    device_buffer_->DeliverRecordedData();
+    // Read next chunk of 10ms data.
+    record_read_pos_ += bytes_per_10_ms_;
+    // Reduce number of cached bytes with the consumed amount.
+    record_cached_bytes_ -= bytes_per_10_ms_;
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer.h
new file mode 100644
index 0000000000..4ab5cd268c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer.h
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+
+// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data
+// corresponding to 10ms of data. It then allows for this data to be pulled in
+// a finer or coarser granularity. I.e. interacting with this class instead of
+// directly with the AudioDeviceBuffer one can ask for any number of audio data
+// samples. This class also ensures that audio data can be delivered to the ADB
+// in 10ms chunks when the size of the provided audio buffers differs from 10ms.
+// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
+// accumulated 10ms worth of data to the ADB every second call.
+class FineAudioBuffer {
+ public:
+  // |device_buffer| is a buffer that provides 10ms of audio data.
+  // |desired_frame_size_bytes| is the number of bytes of audio data
+  // GetPlayoutData() should return on success. It is also the required size of
+  // each recorded buffer used in DeliverRecordedData() calls.
+  // |sample_rate| is the sample rate of the audio data. This is needed because
+  // |device_buffer| delivers 10ms of data. Given the sample rate the number
+  // of samples can be calculated.
+  FineAudioBuffer(AudioDeviceBuffer* device_buffer,
+                  size_t desired_frame_size_bytes,
+                  int sample_rate);
+  ~FineAudioBuffer();
+
+  // Returns the required size of |buffer| when calling GetPlayoutData(). If
+  // the buffer is smaller memory trampling will happen.
+  size_t RequiredPlayoutBufferSizeBytes();
+
+  // Clears buffers and counters dealing with playour and/or recording.
+  void ResetPlayout();
+  void ResetRecord();
+
+  // |buffer| must be of equal or greater size than what is returned by
+  // RequiredBufferSize(). This is to avoid unnecessary memcpy.
+  void GetPlayoutData(int8_t* buffer);
+
+  // Consumes the audio data in |buffer| and sends it to the WebRTC layer in
+  // chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
+  // |record_delay_ms| are given to the AEC in the audio processing module.
+  // They can be fixed values on most platforms and they are ignored if an
+  // external (hardware/built-in) AEC is used.
+  // The size of |buffer| is given by |size_in_bytes| and must be equal to
+  // |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the
+  // case.
+  // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
+  // 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
+  // cache. Call #3 restarts the scheme above.
+  void DeliverRecordedData(const int8_t* buffer,
+                           size_t size_in_bytes,
+                           int playout_delay_ms,
+                           int record_delay_ms);
+
+ private:
+  // Device buffer that works with 10ms chunks of data both for playout and
+  // for recording. I.e., the WebRTC side will always be asked for audio to be
+  // played out in 10ms chunks and recorded audio will be sent to WebRTC in
+  // 10ms chunks as well. This pointer is owned by the constructor of this
+  // class and the owner must ensure that the pointer is valid during the life-
+  // time of this object.
+  AudioDeviceBuffer* const device_buffer_;
+  // Number of bytes delivered by GetPlayoutData() call and provided to
+  // DeliverRecordedData().
+  const size_t desired_frame_size_bytes_;
+  // Sample rate in Hertz.
+  const int sample_rate_;
+  // Number of audio samples per 10ms.
+  const size_t samples_per_10_ms_;
+  // Number of audio bytes per 10ms.
+  const size_t bytes_per_10_ms_;
+  // Storage for output samples that are not yet asked for.
+  rtc::scoped_ptr<int8_t[]> playout_cache_buffer_;
+  // Location of first unread output sample.
+  size_t playout_cached_buffer_start_;
+  // Number of bytes stored in output (contain samples to be played out) cache.
+  size_t playout_cached_bytes_;
+  // Storage for input samples that are about to be delivered to the WebRTC
+  // ADB or remains from the last successful delivery of a 10ms audio buffer.
+  rtc::scoped_ptr<int8_t[]> record_cache_buffer_;
+  // Required (max) size in bytes of the |record_cache_buffer_|.
+  const size_t required_record_buffer_size_bytes_;
+  // Number of bytes in input (contains recorded samples) cache.
+  size_t record_cached_bytes_;
+  // Read and write pointers used in the buffering scheme on the recording side.
+  size_t record_read_pos_;
+  size_t record_write_pos_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
similarity index 59%
rename from media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
index 4cff883129..6666364c9e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/android/fine_audio_buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_device/android/fine_audio_buffer.h"
+#include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
 #include <limits.h>
 #include <memory>
@@ -19,6 +19,7 @@
 #include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
 
 using ::testing::_;
+using ::testing::AtLeast;
 using ::testing::InSequence;
 using ::testing::Return;
 
@@ -40,10 +41,10 @@ bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
   return true;
 }
 
-// This function replaces GetPlayoutData when it's called (which is done
-// implicitly when calling GetBufferData). It writes the sequence
-// 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a buffer of
-// different size than the one VerifyBuffer verifies.
+// This function replaces the real AudioDeviceBuffer::GetPlayoutData when it's
+// called (which is done implicitly when calling GetBufferData). It writes the
+// sequence 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a
+// buffer of different size than the one VerifyBuffer verifies.
 // |iteration| is the number of calls made to UpdateBuffer prior to this call.
 // |samples_per_10_ms| is the number of samples that should be written to the
 // buffer (|arg0|).
@@ -57,10 +58,33 @@ ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
   return samples_per_10_ms;
 }
 
+// Writes a periodic ramp pattern to the supplied |buffer|. See UpdateBuffer()
+// for details.
+void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
+  int start_value = (iteration * size) % SCHAR_MAX;
+  for (int i = 0; i < size; ++i) {
+    buffer[i] = (i + start_value) % SCHAR_MAX;
+  }
+}
+
+// Action macro which verifies that the recorded 10ms chunk of audio data
+// (in |arg0|) contains the correct reference values even if they have been
+// supplied using a buffer size that is smaller or larger than 10ms.
+// See VerifyBuffer() for details.
+ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
+  const int8_t* buffer = static_cast<const int8_t*>(arg0);
+  int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
+  int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
+  for (int i = 0; i < bytes_per_10_ms; ++i) {
+    EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
+  }
+  return 0;
+}
+
 void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
   const int kSamplesPer10Ms = sample_rate * 10 / 1000;
-  const int kFrameSizeBytes = frame_size_in_samples *
-      static_cast<int>(sizeof(int16_t));
+  const int kFrameSizeBytes =
+      frame_size_in_samples * static_cast<int>(sizeof(int16_t));
   const int kNumberOfFrames = 5;
   // Ceiling of integer division: 1 + ((x - 1) / y)
   const int kNumberOfUpdateBufferCalls =
@@ -77,15 +101,32 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
           .RetiresOnSaturation();
     }
   }
+  {
+    InSequence s;
+    for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
+      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
+          .WillOnce(VerifyInputBuffer(j, kSamplesPer10Ms))
+          .RetiresOnSaturation();
+    }
+  }
+  EXPECT_CALL(audio_device_buffer, SetVQEData(_, _, _))
+      .Times(kNumberOfUpdateBufferCalls - 1);
+  EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
+      .Times(kNumberOfUpdateBufferCalls - 1)
+      .WillRepeatedly(Return(kSamplesPer10Ms));
+
   FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
                               sample_rate);
 
   rtc::scoped_ptr<int8_t[]> out_buffer;
-  out_buffer.reset(
-      new int8_t[fine_buffer.RequiredBufferSizeBytes()]);
+  out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
+  rtc::scoped_ptr<int8_t[]> in_buffer;
+  in_buffer.reset(new int8_t[kFrameSizeBytes]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
-    fine_buffer.GetBufferData(out_buffer.get());
+    fine_buffer.GetPlayoutData(out_buffer.get());
     EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
+    UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
+    fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device.h b/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device.h
index ee592d9a7d..ae394b1b29 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device.h
@@ -12,7 +12,7 @@
 #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
 
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 
 namespace webrtc {
 
@@ -30,7 +30,7 @@ class AudioDeviceModule : public RefCountedModule {
     kLinuxAlsaAudio = 3,
     kLinuxPulseAudio = 4,
     kAndroidJavaAudio = 5,
-    kAndroidOpenSLESAudio = 6,
+    kAndroidJavaInputAndOpenSLESOutputAudio = 6,
     kSndioAudio = 7,
     kDummyAudio = 8
   };
@@ -188,23 +188,28 @@ class AudioDeviceModule : public RefCountedModule {
   // Only supported on Android.
   // TODO(henrika): Make pure virtual after updating Chromium.
   virtual bool BuiltInAECIsAvailable() const { return false; }
+  virtual bool BuiltInAGCIsAvailable() const { return false; }
+  virtual bool BuiltInNSIsAvailable() const { return false; }
 
-  // Enables the built-in AEC. Only supported on Windows and Android.
-  //
-  // For usage on Windows (requires Core Audio):
-  // Must be called before InitRecording(). When enabled:
-  // 1. StartPlayout() must be called before StartRecording().
-  // 2. StopRecording() should be called before StopPlayout().
-  //    The reverse order may cause garbage audio to be rendered or the
-  //    capture side to halt until StopRecording() is called.
+  // Enables the built-in audio effects. Only supported on Android.
   // TODO(henrika): Make pure virtual after updating Chromium.
   virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
-
+  virtual int32_t EnableBuiltInAGC(bool enable) { return -1; }
+  virtual int32_t EnableBuiltInNS(bool enable) { return -1; }
   // Don't use.
   virtual bool BuiltInAECIsEnabled() const { return false; }
 
+  // Only supported on iOS.
+  // TODO(henrika): Make pure virtual after updating Chromium.
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const {
+    return -1;
+  }
+  virtual int GetRecordAudioParameters(AudioParameters* params) const {
+    return -1;
+  }
+
  protected:
-  virtual ~AudioDeviceModule() {};
+  virtual ~AudioDeviceModule() {}
 };
 
 AudioDeviceModule* CreateAudioDeviceModule(
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device_defines.h b/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device_defines.h
index 56a584ef9e..b847729f05 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/include/audio_device_defines.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+
+#include <stddef.h>
 
 #include "webrtc/typedefs.h"
 
@@ -26,115 +28,183 @@ static const int kAdmMaxPlayoutBufferSizeMs = 250;
 //  AudioDeviceObserver
 // ----------------------------------------------------------------------------
 
-class AudioDeviceObserver
-{
-public:
-    enum ErrorCode
-    {
-        kRecordingError = 0,
-        kPlayoutError = 1
-    };
-    enum WarningCode
-    {
-        kRecordingWarning = 0,
-        kPlayoutWarning = 1
-    };
+class AudioDeviceObserver {
+ public:
+  enum ErrorCode { kRecordingError = 0, kPlayoutError = 1 };
+  enum WarningCode { kRecordingWarning = 0, kPlayoutWarning = 1 };
 
-    virtual void OnErrorIsReported(const ErrorCode error) = 0;
-    virtual void OnWarningIsReported(const WarningCode warning) = 0;
+  virtual void OnErrorIsReported(const ErrorCode error) = 0;
+  virtual void OnWarningIsReported(const WarningCode warning) = 0;
 
-protected:
-    virtual ~AudioDeviceObserver() {}
+ protected:
+  virtual ~AudioDeviceObserver() {}
 };
 
 // ----------------------------------------------------------------------------
 //  AudioTransport
 // ----------------------------------------------------------------------------
 
-class AudioTransport
-{
-public:
-    virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                            const uint32_t nSamples,
-                                            const uint8_t nBytesPerSample,
-                                            const uint8_t nChannels,
-                                            const uint32_t samplesPerSec,
-                                            const uint32_t totalDelayMS,
-                                            const int32_t clockDrift,
-                                            const uint32_t currentMicLevel,
-                                            const bool keyPressed,
-                                            uint32_t& newMicLevel) = 0;
+class AudioTransport {
+ public:
+  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                          const size_t nSamples,
+                                          const size_t nBytesPerSample,
+                                          const size_t nChannels,
+                                          const uint32_t samplesPerSec,
+                                          const uint32_t totalDelayMS,
+                                          const int32_t clockDrift,
+                                          const uint32_t currentMicLevel,
+                                          const bool keyPressed,
+                                          uint32_t& newMicLevel) = 0;
 
-    virtual int32_t NeedMorePlayData(const uint32_t nSamples,
-                                     const uint8_t nBytesPerSample,
-                                     const uint8_t nChannels,
-                                     const uint32_t samplesPerSec,
-                                     void* audioSamples,
-                                     uint32_t& nSamplesOut,
-                                     int64_t* elapsed_time_ms,
-                                     int64_t* ntp_time_ms) = 0;
+  virtual int32_t NeedMorePlayData(const size_t nSamples,
+                                   const size_t nBytesPerSample,
+                                   const size_t nChannels,
+                                   const uint32_t samplesPerSec,
+                                   void* audioSamples,
+                                   size_t& nSamplesOut,
+                                   int64_t* elapsed_time_ms,
+                                   int64_t* ntp_time_ms) = 0;
 
-    // Method to pass captured data directly and unmixed to network channels.
-    // |channel_ids| contains a list of VoE channels which are the
-    // sinks to the capture data. |audio_delay_milliseconds| is the sum of
-    // recording delay and playout delay of the hardware. |current_volume| is
-    // in the range of [0, 255], representing the current microphone analog
-    // volume. |key_pressed| is used by the typing detection.
-    // |need_audio_processing| specify if the data needs to be processed by APM.
-    // Currently WebRtc supports only one APM, and Chrome will make sure only
-    // one stream goes through APM. When |need_audio_processing| is false, the
-    // values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
-    // will be ignored.
-    // The return value is the new microphone volume, in the range of |0, 255].
-    // When the volume does not need to be updated, it returns 0.
-    // TODO(xians): Remove this interface after Chrome and Libjingle switches
-    // to OnData().
-    virtual int OnDataAvailable(const int voe_channels[],
-                                int number_of_voe_channels,
-                                const int16_t* audio_data,
-                                int sample_rate,
-                                int number_of_channels,
-                                int number_of_frames,
-                                int audio_delay_milliseconds,
-                                int current_volume,
-                                bool key_pressed,
-                                bool need_audio_processing) { return 0; }
+  // Method to pass captured data directly and unmixed to network channels.
+  // |channel_ids| contains a list of VoE channels which are the
+  // sinks to the capture data. |audio_delay_milliseconds| is the sum of
+  // recording delay and playout delay of the hardware. |current_volume| is
+  // in the range of [0, 255], representing the current microphone analog
+  // volume. |key_pressed| is used by the typing detection.
+  // |need_audio_processing| specify if the data needs to be processed by APM.
+  // Currently WebRtc supports only one APM, and Chrome will make sure only
+  // one stream goes through APM. When |need_audio_processing| is false, the
+  // values of |audio_delay_milliseconds|, |current_volume| and |key_pressed|
+  // will be ignored.
+  // The return value is the new microphone volume, in the range of |0, 255].
+  // When the volume does not need to be updated, it returns 0.
+  // TODO(xians): Remove this interface after Chrome and Libjingle switches
+  // to OnData().
+  virtual int OnDataAvailable(const int voe_channels[],
+                              size_t number_of_voe_channels,
+                              const int16_t* audio_data,
+                              int sample_rate,
+                              size_t number_of_channels,
+                              size_t number_of_frames,
+                              int audio_delay_milliseconds,
+                              int current_volume,
+                              bool key_pressed,
+                              bool need_audio_processing) {
+    return 0;
+  }
 
-    // Method to pass the captured audio data to the specific VoE channel.
-    // |voe_channel| is the id of the VoE channel which is the sink to the
-    // capture data.
-    // TODO(xians): Remove this interface after Libjingle switches to
-    // PushCaptureData().
-    virtual void OnData(int voe_channel, const void* audio_data,
-                        int bits_per_sample, int sample_rate,
-                        int number_of_channels,
-                        int number_of_frames) {}
+  // Method to pass the captured audio data to the specific VoE channel.
+  // |voe_channel| is the id of the VoE channel which is the sink to the
+  // capture data.
+  // TODO(xians): Remove this interface after Libjingle switches to
+  // PushCaptureData().
+  virtual void OnData(int voe_channel,
+                      const void* audio_data,
+                      int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames) {}
 
-    // Method to push the captured audio data to the specific VoE channel.
-    // The data will not undergo audio processing.
-    // |voe_channel| is the id of the VoE channel which is the sink to the
-    // capture data.
-    // TODO(xians): Make the interface pure virtual after Libjingle
-    // has its implementation.
-    virtual void PushCaptureData(int voe_channel, const void* audio_data,
-                                 int bits_per_sample, int sample_rate,
-                                 int number_of_channels,
-                                 int number_of_frames) {}
+  // Method to push the captured audio data to the specific VoE channel.
+  // The data will not undergo audio processing.
+  // |voe_channel| is the id of the VoE channel which is the sink to the
+  // capture data.
+  // TODO(xians): Make the interface pure virtual after Libjingle
+  // has its implementation.
+  virtual void PushCaptureData(int voe_channel,
+                               const void* audio_data,
+                               int bits_per_sample,
+                               int sample_rate,
+                               size_t number_of_channels,
+                               size_t number_of_frames) {}
 
-    // Method to pull mixed render audio data from all active VoE channels.
-    // The data will not be passed as reference for audio processing internally.
-    // TODO(xians): Support getting the unmixed render data from specific VoE
-    // channel.
-    virtual void PullRenderData(int bits_per_sample, int sample_rate,
-                                int number_of_channels, int number_of_frames,
-                                void* audio_data,
-                                int64_t* elapsed_time_ms,
-                                int64_t* ntp_time_ms) {}
+  // Method to pull mixed render audio data from all active VoE channels.
+  // The data will not be passed as reference for audio processing internally.
+  // TODO(xians): Support getting the unmixed render data from specific VoE
+  // channel.
+  virtual void PullRenderData(int bits_per_sample,
+                              int sample_rate,
+                              size_t number_of_channels,
+                              size_t number_of_frames,
+                              void* audio_data,
+                              int64_t* elapsed_time_ms,
+                              int64_t* ntp_time_ms) {}
 
-protected:
-    virtual ~AudioTransport() {}
+ protected:
+  virtual ~AudioTransport() {}
+};
+
+// Helper class for storage of fundamental audio parameters such as sample rate,
+// number of channels, native buffer size etc.
+// Note that one audio frame can contain more than one channel sample and each
+// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
+// stereo contains 2 * (16/8) = 4 bytes of data.
+class AudioParameters {
+ public:
+  // This implementation does only support 16-bit PCM samples.
+  static const size_t kBitsPerSample = 16;
+  AudioParameters()
+      : sample_rate_(0),
+        channels_(0),
+        frames_per_buffer_(0),
+        frames_per_10ms_buffer_(0) {}
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
+      : sample_rate_(sample_rate),
+        channels_(channels),
+        frames_per_buffer_(frames_per_buffer),
+        frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
+    sample_rate_ = sample_rate;
+    channels_ = channels;
+    frames_per_buffer_ = frames_per_buffer;
+    frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
+  }
+  size_t bits_per_sample() const { return kBitsPerSample; }
+  void reset(int sample_rate, size_t channels, double ms_per_buffer) {
+    reset(sample_rate, channels,
+          static_cast<size_t>(sample_rate * ms_per_buffer + 0.5));
+  }
+  void reset(int sample_rate, size_t channels) {
+    reset(sample_rate, channels, static_cast<size_t>(0));
+  }
+  int sample_rate() const { return sample_rate_; }
+  size_t channels() const { return channels_; }
+  size_t frames_per_buffer() const { return frames_per_buffer_; }
+  size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
+  size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
+  size_t GetBytesPerBuffer() const {
+    return frames_per_buffer_ * GetBytesPerFrame();
+  }
+  // The WebRTC audio device buffer (ADB) only requires that the sample rate
+  // and number of channels are configured. Hence, to be "valid", only these
+  // two attributes must be set.
+  bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
+  // Most platforms also require that a native buffer size is defined.
+  // An audio parameter instance is considered to be "complete" if it is both
+  // "valid" (can be used by the ADB) and also has a native frame size.
+  bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
+  size_t GetBytesPer10msBuffer() const {
+    return frames_per_10ms_buffer_ * GetBytesPerFrame();
+  }
+  double GetBufferSizeInMilliseconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return frames_per_buffer_ / (sample_rate_ / 1000.0);
+  }
+  double GetBufferSizeInSeconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return static_cast<double>(frames_per_buffer_) / (sample_rate_);
+  }
+
+ private:
+  int sample_rate_;
+  size_t channels_;
+  size_t frames_per_buffer_;
+  size_t frames_per_10ms_buffer_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/include/fake_audio_device.h b/media/webrtc/trunk/webrtc/modules/audio_device/include/fake_audio_device.h
index 8b7e87c619..8b154794c7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/include/fake_audio_device.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/include/fake_audio_device.h
@@ -16,8 +16,8 @@ class FakeAudioDeviceModule : public AudioDeviceModule {
  public:
   FakeAudioDeviceModule() {}
   virtual ~FakeAudioDeviceModule() {}
-  virtual int32_t AddRef() { return 0; }
-  virtual int32_t Release() { return 0; }
+  virtual int32_t AddRef() const { return 0; }
+  virtual int32_t Release() const { return 0; }
   virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) {
     return 0;
   }
@@ -147,6 +147,10 @@ class FakeAudioDeviceModule : public AudioDeviceModule {
   virtual bool BuiltInAECIsAvailable() const { return false; }
   virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
   virtual bool BuiltInAECIsEnabled() const { return false; }
+  virtual bool BuiltInAGCIsAvailable() const { return false; }
+  virtual int32_t EnableBuiltInAGC(bool enable) { return -1; }
+  virtual bool BuiltInNSIsAvailable() const { return false; }
+  virtual int32_t EnableBuiltInNS(bool enable) { return -1; }
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.h b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.h
index 70ab4f8e2a..c4eb0d6f64 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.h
@@ -8,263 +8,291 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 
 #include <AudioUnit/AudioUnit.h>
 
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 namespace webrtc {
-const uint32_t N_REC_SAMPLES_PER_SEC = 44100;
-const uint32_t N_PLAY_SAMPLES_PER_SEC = 44100;
 
-const uint32_t N_REC_CHANNELS = 1;  // default is mono recording
-const uint32_t N_PLAY_CHANNELS = 1;  // default is mono playout
-const uint32_t N_DEVICE_CHANNELS = 8;
-
-const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
-const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
-
-// Number of 10 ms recording blocks in recording buffer
-const uint16_t N_REC_BUFFERS = 20;
+class FineAudioBuffer;
 
+// Implements full duplex 16-bit mono PCM audio support for iOS using a
+// Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
+// supports audio echo cancellation. It also adds automatic gain control,
+// adjustment of voice-processing quality and muting.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All supported public methods must also be called on the same thread.
+// A thread checker will RTC_DCHECK if any supported method is called on an
+// invalid thread.
+//
+// Recorded audio will be delivered on a real-time internal I/O thread in the
+// audio unit. The audio unit will also ask for audio data to play out on this
+// same thread.
 class AudioDeviceIOS : public AudioDeviceGeneric {
  public:
-  AudioDeviceIOS(const int32_t id);
+  AudioDeviceIOS();
   ~AudioDeviceIOS();
 
-  // Retrieve the currently utilized audio layer
-  virtual int32_t ActiveAudioLayer(
-      AudioDeviceModule::AudioLayer& audioLayer) const;
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
-  // Main initializaton and termination
-  virtual int32_t Init();
-  virtual int32_t Terminate();
-  virtual bool Initialized() const;
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override { return initialized_; }
 
-  // Device enumeration
-  virtual int16_t PlayoutDevices();
-  virtual int16_t RecordingDevices();
-  virtual int32_t PlayoutDeviceName(uint16_t index,
-                                    char name[kAdmMaxDeviceNameSize],
-                                    char guid[kAdmMaxGuidSize]);
-  virtual int32_t RecordingDeviceName(uint16_t index,
-                                      char name[kAdmMaxDeviceNameSize],
-                                      char guid[kAdmMaxGuidSize]);
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override { return play_is_initialized_; }
 
-  // Device selection
-  virtual int32_t SetPlayoutDevice(uint16_t index);
-  virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
-  virtual int32_t SetRecordingDevice(uint16_t index);
-  virtual int32_t SetRecordingDevice(
-      AudioDeviceModule::WindowsDeviceType device);
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override { return rec_is_initialized_; }
 
-  // Audio transport initialization
-  virtual int32_t PlayoutIsAvailable(bool& available);
-  virtual int32_t InitPlayout();
-  virtual bool PlayoutIsInitialized() const;
-  virtual int32_t RecordingIsAvailable(bool& available);
-  virtual int32_t InitRecording();
-  virtual bool RecordingIsInitialized() const;
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override { return playing_; }
 
-  // Audio transport control
-  virtual int32_t StartPlayout();
-  virtual int32_t StopPlayout();
-  virtual bool Playing() const;
-  virtual int32_t StartRecording();
-  virtual int32_t StopRecording();
-  virtual bool Recording() const;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override { return recording_; }
 
-  // Microphone Automatic Gain Control (AGC)
-  virtual int32_t SetAGC(bool enable);
-  virtual bool AGC() const;
+  int32_t SetLoudspeakerStatus(bool enable) override;
+  int32_t GetLoudspeakerStatus(bool& enabled) const override;
 
-  // Volume control based on the Windows Wave API (Windows only)
-  virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
-  virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
-                                uint16_t& volumeRight) const;
+  // These methods returns hard-coded delay values and not dynamic delay
+  // estimates. The reason is that iOS supports a built-in AEC and the WebRTC
+  // AEC will always be disabled in the Libjingle layer to avoid running two
+  // AEC implementations at the same time. And, it saves resources to avoid
+  // updating these delay values continuously.
+  // TODO(henrika): it would be possible to mark these two methods as not
+  // implemented since they are only called for A/V-sync purposes today and
+  // A/V-sync is not supported on iOS. However, we avoid adding error messages
+  // the log by using these dummy implementations instead.
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+  int32_t RecordingDelay(uint16_t& delayMS) const override;
 
-  // Audio mixer initialization
-  virtual int32_t InitSpeaker();
-  virtual bool SpeakerIsInitialized() const;
-  virtual int32_t InitMicrophone();
-  virtual bool MicrophoneIsInitialized() const;
+  // Native audio parameters stored during construction.
+  // These methods are unique for the iOS implementation.
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
 
-  // Speaker volume controls
-  virtual int32_t SpeakerVolumeIsAvailable(bool& available);
-  virtual int32_t SetSpeakerVolume(uint32_t volume);
-  virtual int32_t SpeakerVolume(uint32_t& volume) const;
-  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
-  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
-  virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
+  // These methods are currently not fully implemented on iOS:
 
-  // Microphone volume controls
-  virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
-  virtual int32_t SetMicrophoneVolume(uint32_t volume);
-  virtual int32_t MicrophoneVolume(uint32_t& volume) const;
-  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
-  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
-  virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
-
-  // Microphone mute control
-  virtual int32_t MicrophoneMuteIsAvailable(bool& available);
-  virtual int32_t SetMicrophoneMute(bool enable);
-  virtual int32_t MicrophoneMute(bool& enabled) const;
-
-  // Speaker mute control
-  virtual int32_t SpeakerMuteIsAvailable(bool& available);
-  virtual int32_t SetSpeakerMute(bool enable);
-  virtual int32_t SpeakerMute(bool& enabled) const;
-
-  // Microphone boost control
-  virtual int32_t MicrophoneBoostIsAvailable(bool& available);
-  virtual int32_t SetMicrophoneBoost(bool enable);
-  virtual int32_t MicrophoneBoost(bool& enabled) const;
-
-  // Stereo support
-  virtual int32_t StereoPlayoutIsAvailable(bool& available);
-  virtual int32_t SetStereoPlayout(bool enable);
-  virtual int32_t StereoPlayout(bool& enabled) const;
-  virtual int32_t StereoRecordingIsAvailable(bool& available);
-  virtual int32_t SetStereoRecording(bool enable);
-  virtual int32_t StereoRecording(bool& enabled) const;
-
-  // Delay information and control
-  virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                                   uint16_t sizeMS);
-  virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
-                                uint16_t& sizeMS) const;
-  virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
-  virtual int32_t RecordingDelay(uint16_t& delayMS) const;
-
-  // CPU load
-  virtual int32_t CPULoad(uint16_t& load) const;
-
- public:
-  virtual bool PlayoutWarning() const;
-  virtual bool PlayoutError() const;
-  virtual bool RecordingWarning() const;
-  virtual bool RecordingError() const;
-  virtual void ClearPlayoutWarning();
-  virtual void ClearPlayoutError();
-  virtual void ClearRecordingWarning();
-  virtual void ClearRecordingError();
-
- public:
-  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-  // Reset Audio Device (for mobile devices only)
-  virtual int32_t ResetAudioDevice();
-
-  // enable or disable loud speaker (for iphone only)
-  virtual int32_t SetLoudspeakerStatus(bool enable);
-  virtual int32_t GetLoudspeakerStatus(bool& enabled) const;
+  // See audio_device_not_implemented.cc for trivial implementations.
+  int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                        uint16_t& sizeMS) const override;
+  int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+  int32_t ResetAudioDevice() override;
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t SetAGC(bool enable) override;
+  bool AGC() const override;
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight) override;
+  int32_t WaveOutVolume(uint16_t& volumeLeft,
+                        uint16_t& volumeRight) const override;
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+  int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const override;
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+  int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const override;
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+  int32_t MicrophoneBoostIsAvailable(bool& available) override;
+  int32_t SetMicrophoneBoost(bool enable) override;
+  int32_t MicrophoneBoost(bool& enabled) const override;
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+  int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+                           uint16_t sizeMS) override;
+  int32_t CPULoad(uint16_t& load) const override;
+  bool PlayoutWarning() const override;
+  bool PlayoutError() const override;
+  bool RecordingWarning() const override;
+  bool RecordingError() const override;
+  void ClearPlayoutWarning() override {}
+  void ClearPlayoutError() override {}
+  void ClearRecordingWarning() override {}
+  void ClearRecordingError() override {}
 
  private:
-  void Lock() {
-    _critSect.Enter();
-  }
+  // Uses current |playout_parameters_| and |record_parameters_| to inform the
+  // audio device buffer (ADB) about our internal audio parameters.
+  void UpdateAudioDeviceBuffer();
 
-  void UnLock() {
-    _critSect.Leave();
-  }
+  // Registers observers for the AVAudioSessionRouteChangeNotification and
+  // AVAudioSessionInterruptionNotification notifications.
+  void RegisterNotificationObservers();
+  void UnregisterNotificationObservers();
 
-  int32_t Id() {
-    return _id;
-  }
+  // Since the preferred audio parameters are only hints to the OS, the actual
+  // values may be different once the AVAudioSession has been activated.
+  // This method asks for the current hardware parameters and takes actions
+  // if they should differ from what we have asked for initially. It also
+  // defines |playout_parameters_| and |record_parameters_|.
+  void SetupAudioBuffersForActiveAudioSession();
 
-  // Init and shutdown
-  int32_t InitPlayOrRecord();
-  int32_t ShutdownPlayOrRecord();
+  // Creates a Voice-Processing I/O unit and configures it for full-duplex
+  // audio. The selected stream format is selected to avoid internal resampling
+  // and to match the 10ms callback rate for WebRTC as well as possible.
+  // This method also initializes the created audio unit.
+  bool SetupAndInitializeVoiceProcessingAudioUnit();
 
-  void UpdateRecordingDelay();
-  void UpdatePlayoutDelay();
+  // Restarts active audio streams using a new sample rate. Required when e.g.
+  // a BT headset is enabled or disabled.
+  bool RestartAudioUnitWithNewFormat(float sample_rate);
 
-  static OSStatus RecordProcess(void *inRefCon,
-                                AudioUnitRenderActionFlags *ioActionFlags,
-                                const AudioTimeStamp *timeStamp,
-                                UInt32 inBusNumber,
-                                UInt32 inNumberFrames,
-                                AudioBufferList *ioData);
+  // Activates our audio session, creates and initializes the voice-processing
+  // audio unit and verifies that we got the preferred native audio parameters.
+  bool InitPlayOrRecord();
 
-  static OSStatus PlayoutProcess(void *inRefCon,
-                                 AudioUnitRenderActionFlags *ioActionFlags,
-                                 const AudioTimeStamp *timeStamp,
-                                 UInt32 inBusNumber,
-                                 UInt32 inNumberFrames,
-                                 AudioBufferList *ioData);
+  // Closes and deletes the voice-processing I/O unit.
+  void ShutdownPlayOrRecord();
 
-  OSStatus RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags,
-                             const AudioTimeStamp *timeStamp,
-                             uint32_t inBusNumber,
-                             uint32_t inNumberFrames);
+  // Helper method for destroying the existing audio unit.
+  void DisposeAudioUnit();
 
-  OSStatus PlayoutProcessImpl(uint32_t inNumberFrames,
-                              AudioBufferList *ioData);
+  // Callback function called on a real-time priority I/O thread from the audio
+  // unit. This method is used to signal that recorded audio is available.
+  static OSStatus RecordedDataIsAvailable(
+      void* in_ref_con,
+      AudioUnitRenderActionFlags* io_action_flags,
+      const AudioTimeStamp* time_stamp,
+      UInt32 in_bus_number,
+      UInt32 in_number_frames,
+      AudioBufferList* io_data);
+  OSStatus OnRecordedDataIsAvailable(
+      AudioUnitRenderActionFlags* io_action_flags,
+      const AudioTimeStamp* time_stamp,
+      UInt32 in_bus_number,
+      UInt32 in_number_frames);
 
-  static bool RunCapture(void* ptrThis);
-  bool CaptureWorkerThread();
+  // Callback function called on a real-time priority I/O thread from the audio
+  // unit. This method is used to provide audio samples to the audio unit.
+  static OSStatus GetPlayoutData(void* in_ref_con,
+                                 AudioUnitRenderActionFlags* io_action_flags,
+                                 const AudioTimeStamp* time_stamp,
+                                 UInt32 in_bus_number,
+                                 UInt32 in_number_frames,
+                                 AudioBufferList* io_data);
+  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
+                            UInt32 in_number_frames,
+                            AudioBufferList* io_data);
 
- private:
-  AudioDeviceBuffer* _ptrAudioBuffer;
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  rtc::ThreadChecker thread_checker_;
 
-  CriticalSectionWrapper& _critSect;
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
+  // and therefore outlives this object.
+  AudioDeviceBuffer* audio_device_buffer_;
 
-  rtc::scoped_ptr<ThreadWrapper> _captureWorkerThread;
+  // Contains audio parameters (sample rate, #channels, buffer size etc.) for
+  // the playout and recording sides. These structure is set in two steps:
+  // first, native sample rate and #channels are defined in Init(). Next, the
+  // audio session is activated and we verify that the preferred parameters
+  // were granted by the OS. At this stage it is also possible to add a third
+  // component to the parameters; the native I/O buffer duration.
+  // A RTC_CHECK will be hit if we for some reason fail to open an audio session
+  // using the specified parameters.
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
 
-  int32_t _id;
+  // The Voice-Processing I/O unit has the same characteristics as the
+  // Remote I/O unit (supports full duplex low-latency audio input and output)
+  // and adds AEC for for two-way duplex communication. It also adds AGC,
+  // adjustment of voice-processing quality, and muting. Hence, ideal for
+  // VoIP applications.
+  AudioUnit vpio_unit_;
 
-  AudioUnit _auVoiceProcessing;
-  void* _audioInterruptionObserver;
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples. Is also supports a similar scheme for the recording
+  // side.
+  // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
+  // in each callback (one every 8ms). This class can then ask for 128 and the
+  // FineAudioBuffer will ask WebRTC for new data only when needed and also
+  // cache non-utilized audio between callbacks. On the recording side, iOS
+  // can provide audio data frames of size 128 and these are accumulated until
+  // enough data to supply one 10ms call exists. This 10ms chunk is then sent
+  // to WebRTC and the remaining part is stored.
+  rtc::scoped_ptr<FineAudioBuffer> fine_audio_buffer_;
 
- private:
-  bool _initialized;
-  bool _isShutDown;
-  bool _recording;
-  bool _playing;
-  bool _recIsInitialized;
-  bool _playIsInitialized;
+  // Extra audio buffer to be used by the playout side for rendering audio.
+  // The buffer size is given by FineAudioBuffer::RequiredBufferSizeBytes().
+  rtc::scoped_ptr<SInt8[]> playout_audio_buffer_;
 
-  bool _recordingDeviceIsSpecified;
-  bool _playoutDeviceIsSpecified;
-  bool _micIsInitialized;
-  bool _speakerIsInitialized;
+  // Provides a mechanism for encapsulating one or more buffers of audio data.
+  // Only used on the recording side.
+  AudioBufferList audio_record_buffer_list_;
 
-  bool _AGC;
+  // Temporary storage for recorded data. AudioUnitRender() renders into this
+  // array as soon as a frame of the desired buffer size has been recorded.
+  rtc::scoped_ptr<SInt8[]> record_audio_buffer_;
 
-  // The sampling rate to use with Audio Device Buffer
-  uint32_t _adbSampFreq;
+  // Set to 1 when recording is active and 0 otherwise.
+  volatile int recording_;
 
-  // Delay calculation
-  uint32_t _recordingDelay;
-  uint32_t _playoutDelay;
-  uint32_t _playoutDelayMeasurementCounter;
-  uint32_t _recordingDelayHWAndOS;
-  uint32_t _recordingDelayMeasurementCounter;
+  // Set to 1 when playout is active and 0 otherwise.
+  volatile int playing_;
 
-  // Errors and warnings count
-  uint16_t _playWarning;
-  uint16_t _playError;
-  uint16_t _recWarning;
-  uint16_t _recError;
+  // Set to true after successful call to Init(), false otherwise.
+  bool initialized_;
 
-  // Playout buffer, needed for 44.0 / 44.1 kHz mismatch
-  int16_t _playoutBuffer[ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
-  uint32_t  _playoutBufferUsed;  // How much is filled
+  // Set to true after successful call to InitRecording(), false otherwise.
+  bool rec_is_initialized_;
 
-  // Recording buffers
-  int16_t _recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES];
-  uint32_t _recordingLength[N_REC_BUFFERS];
-  uint32_t _recordingSeqNumber[N_REC_BUFFERS];
-  uint32_t _recordingCurrentSeq;
+  // Set to true after successful call to InitPlayout(), false otherwise.
+  bool play_is_initialized_;
 
-  // Current total size all data in buffers, used for delay estimate
-  uint32_t _recordingBufferTotalSize;
+  // Audio interruption observer instance.
+  void* audio_interruption_observer_;
+  void* route_change_observer_;
+
+  // Contains the audio data format specification for a stream of audio.
+  AudioStreamBasicDescription application_format_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.mm b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.mm
index 178ce3e0a9..f6dee5b3cf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -8,1899 +8,1105 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
+
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_device/fine_audio_buffer.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
 
 namespace webrtc {
-AudioDeviceIOS::AudioDeviceIOS(const int32_t id)
-    :
-    _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
-    _auVoiceProcessing(NULL),
-    _audioInterruptionObserver(NULL),
-    _initialized(false),
-    _isShutDown(false),
-    _recording(false),
-    _playing(false),
-    _recIsInitialized(false),
-    _playIsInitialized(false),
-    _recordingDeviceIsSpecified(false),
-    _playoutDeviceIsSpecified(false),
-    _micIsInitialized(false),
-    _speakerIsInitialized(false),
-    _AGC(false),
-    _adbSampFreq(0),
-    _recordingDelay(0),
-    _playoutDelay(0),
-    _playoutDelayMeasurementCounter(9999),
-    _recordingDelayHWAndOS(0),
-    _recordingDelayMeasurementCounter(9999),
-    _playWarning(0),
-    _playError(0),
-    _recWarning(0),
-    _recError(0),
-    _playoutBufferUsed(0),
-    _recordingCurrentSeq(0),
-    _recordingBufferTotalSize(0) {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
 
-    memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-    memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
-    memset(_recordingLength, 0, sizeof(_recordingLength));
-    memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
+// Protects |g_audio_session_users|.
+static rtc::GlobalLockPod g_lock;
+
+// Counts number of users (=instances of this object) who needs an active
+// audio session. This variable is used to ensure that we only activate an audio
+// session for the first user and deactivate it for the last.
+// Member is static to ensure that the value is counted for all instances
+// and not per instance.
+static int g_audio_session_users GUARDED_BY(g_lock) = 0;
+
+#define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
+
+#define LOG_AND_RETURN_IF_ERROR(error, message) \
+  do {                                          \
+    OSStatus err = error;                       \
+    if (err) {                                  \
+      LOG(LS_ERROR) << message << ": " << err;  \
+      return false;                             \
+    }                                           \
+  } while (0)
+
+#define LOG_IF_ERROR(error, message)           \
+  do {                                         \
+    OSStatus err = error;                      \
+    if (err) {                                 \
+      LOG(LS_ERROR) << message << ": " << err; \
+    }                                          \
+  } while (0)
+
+// Preferred hardware sample rate (unit is in Hertz). The client sample rate
+// will be set to this value as well to avoid resampling the the audio unit's
+// format converter. Note that, some devices, e.g. BT headsets, only supports
+// 8000Hz as native sample rate.
+const double kPreferredSampleRate = 48000.0;
+// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
+// size used by WebRTC. The exact actual size will differ between devices.
+// Example: using 48kHz on iPhone 6 results in a native buffer size of
+// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
+// take care of any buffering required to convert between native buffers and
+// buffers used by WebRTC. It is beneficial for the performance if the native
+// size is as close to 10ms as possible since it results in "clean" callback
+// sequence without bursts of callbacks back to back.
+const double kPreferredIOBufferDuration = 0.01;
+// Try to use mono to save resources. Also avoids channel format conversion
+// in the I/O audio unit. Initial tests have shown that it is possible to use
+// mono natively for built-in microphones and for BT headsets but not for
+// wired headsets. Wired headsets only support stereo as native channel format
+// but it is a low cost operation to do a format conversion to mono in the
+// audio unit. Hence, we will not hit a RTC_CHECK in
+// VerifyAudioParametersForActiveAudioSession() for a mismatch between the
+// preferred number of channels and the actual number of channels.
+const int kPreferredNumberOfChannels = 1;
+// Number of bytes per audio sample for 16-bit signed integer representation.
+const UInt32 kBytesPerSample = 2;
+// Hardcoded delay estimates based on real measurements.
+// TODO(henrika): these value is not used in combination with built-in AEC.
+// Can most likely be removed.
+const UInt16 kFixedPlayoutDelayEstimate = 30;
+const UInt16 kFixedRecordDelayEstimate = 30;
+// Calls to AudioUnitInitialize() can fail if called back-to-back on different
+// ADM instances. A fall-back solution is to allow multiple sequential calls
+// with as small delay between each. This factor sets the max number of allowed
+// initialization attempts.
+const int kMaxNumberOfAudioUnitInitializeAttempts = 5;
+
+
+using ios::CheckAndLogError;
+
+// Verifies that the current audio session supports input audio and that the
+// required category and mode are enabled.
+static bool VerifyAudioSession(AVAudioSession* session) {
+  LOG(LS_INFO) << "VerifyAudioSession";
+  // Ensure that the device currently supports audio input.
+  if (!session.isInputAvailable) {
+    LOG(LS_ERROR) << "No audio input path is available!";
+    return false;
+  }
+
+  // Ensure that the required category and mode are actually activated.
+  if (![session.category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) {
+    LOG(LS_ERROR)
+        << "Failed to set category to AVAudioSessionCategoryPlayAndRecord";
+    return false;
+  }
+  if (![session.mode isEqualToString:AVAudioSessionModeVoiceChat]) {
+    LOG(LS_ERROR) << "Failed to set mode to AVAudioSessionModeVoiceChat";
+    return false;
+  }
+  return true;
+}
+
+// Activates an audio session suitable for full duplex VoIP sessions when
+// |activate| is true. Also sets the preferred sample rate and IO buffer
+// duration. Deactivates an active audio session if |activate| is set to false.
+static bool ActivateAudioSession(AVAudioSession* session, bool activate)
+    EXCLUSIVE_LOCKS_REQUIRED(g_lock) {
+  LOG(LS_INFO) << "ActivateAudioSession(" << activate << ")";
+  @autoreleasepool {
+    NSError* error = nil;
+    BOOL success = NO;
+
+    if (!activate) {
+      // Deactivate the audio session using an extra option and then return.
+      // AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation is used to
+      // ensure that other audio sessions that were interrupted by our session
+      // can return to their active state. It is recommended for VoIP apps to
+      // use this option.
+      success = [session
+            setActive:NO
+          withOptions:AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation
+                error:&error];
+      return CheckAndLogError(success, error);
+    }
+
+    // Go ahead and active our own audio session since |activate| is true.
+    // Use a category which supports simultaneous recording and playback.
+    // By default, using this category implies that our app’s audio is
+    // nonmixable, hence activating the session will interrupt any other
+    // audio sessions which are also nonmixable.
+    if (session.category != AVAudioSessionCategoryPlayAndRecord) {
+      error = nil;
+      success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
+                         withOptions:AVAudioSessionCategoryOptionAllowBluetooth
+                               error:&error];
+      RTC_DCHECK(CheckAndLogError(success, error));
+    }
+
+    // Specify mode for two-way voice communication (e.g. VoIP).
+    if (session.mode != AVAudioSessionModeVoiceChat) {
+      error = nil;
+      success = [session setMode:AVAudioSessionModeVoiceChat error:&error];
+      RTC_DCHECK(CheckAndLogError(success, error));
+    }
+
+    // Set the session's sample rate or the hardware sample rate.
+    // It is essential that we use the same sample rate as stream format
+    // to ensure that the I/O unit does not have to do sample rate conversion.
+    error = nil;
+    success =
+        [session setPreferredSampleRate:kPreferredSampleRate error:&error];
+    RTC_DCHECK(CheckAndLogError(success, error));
+
+    // Set the preferred audio I/O buffer duration, in seconds.
+    error = nil;
+    success = [session setPreferredIOBufferDuration:kPreferredIOBufferDuration
+                                              error:&error];
+    RTC_DCHECK(CheckAndLogError(success, error));
+
+    // Activate the audio session. Activation can fail if another active audio
+    // session (e.g. phone call) has higher priority than ours.
+    error = nil;
+    success = [session setActive:YES error:&error];
+    if (!CheckAndLogError(success, error)) {
+      return false;
+    }
+
+    // Ensure that the active audio session has the correct category and mode.
+    if (!VerifyAudioSession(session)) {
+      LOG(LS_ERROR) << "Failed to verify audio session category and mode";
+      return false;
+    }
+
+    // Try to set the preferred number of hardware audio channels. These calls
+    // must be done after setting the audio session’s category and mode and
+    // activating the session.
+    // We try to use mono in both directions to save resources and format
+    // conversions in the audio unit. Some devices does only support stereo;
+    // e.g. wired headset on iPhone 6.
+    // TODO(henrika): add support for stereo if needed.
+    error = nil;
+    success =
+        [session setPreferredInputNumberOfChannels:kPreferredNumberOfChannels
+                                             error:&error];
+    RTC_DCHECK(CheckAndLogError(success, error));
+    error = nil;
+    success =
+        [session setPreferredOutputNumberOfChannels:kPreferredNumberOfChannels
+                                              error:&error];
+    RTC_DCHECK(CheckAndLogError(success, error));
+    return true;
+  }
+}
+
+// An application can create more than one ADM and start audio streaming
+// for all of them. It is essential that we only activate the app's audio
+// session once (for the first one) and deactivate it once (for the last).
+static bool ActivateAudioSession() {
+  LOGI() << "ActivateAudioSession";
+  rtc::GlobalLockScope ls(&g_lock);
+  if (g_audio_session_users == 0) {
+    // The system provides an audio session object upon launch of an
+    // application. However, we must initialize the session in order to
+    // handle interruptions. Implicit initialization occurs when obtaining
+    // a reference to the AVAudioSession object.
+    AVAudioSession* session = [AVAudioSession sharedInstance];
+    // Try to activate the audio session and ask for a set of preferred audio
+    // parameters.
+    if (!ActivateAudioSession(session, true)) {
+      LOG(LS_ERROR) << "Failed to activate the audio session";
+      return false;
+    }
+    LOG(LS_INFO) << "The audio session is now activated";
+  }
+  ++g_audio_session_users;
+  LOG(LS_INFO) << "Number of audio session users: " << g_audio_session_users;
+  return true;
+}
+
+// If more than one object is using the audio session, ensure that only the
+// last object deactivates. Apple recommends: "activate your audio session
+// only as needed and deactivate it when you are not using audio".
+static bool DeactivateAudioSession() {
+  LOGI() << "DeactivateAudioSession";
+  rtc::GlobalLockScope ls(&g_lock);
+  if (g_audio_session_users == 1) {
+    AVAudioSession* session = [AVAudioSession sharedInstance];
+    if (!ActivateAudioSession(session, false)) {
+      LOG(LS_ERROR) << "Failed to deactivate the audio session";
+      return false;
+    }
+    LOG(LS_INFO) << "Our audio session is now deactivated";
+  }
+  --g_audio_session_users;
+  LOG(LS_INFO) << "Number of audio session users: " << g_audio_session_users;
+  return true;
+}
+
+#if !defined(NDEBUG)
+// Helper method for printing out an AudioStreamBasicDescription structure.
+static void LogABSD(AudioStreamBasicDescription absd) {
+  char formatIDString[5];
+  UInt32 formatID = CFSwapInt32HostToBig(absd.mFormatID);
+  bcopy(&formatID, formatIDString, 4);
+  formatIDString[4] = '\0';
+  LOG(LS_INFO) << "LogABSD";
+  LOG(LS_INFO) << " sample rate: " << absd.mSampleRate;
+  LOG(LS_INFO) << " format ID: " << formatIDString;
+  LOG(LS_INFO) << " format flags: " << std::hex << absd.mFormatFlags;
+  LOG(LS_INFO) << " bytes per packet: " << absd.mBytesPerPacket;
+  LOG(LS_INFO) << " frames per packet: " << absd.mFramesPerPacket;
+  LOG(LS_INFO) << " bytes per frame: " << absd.mBytesPerFrame;
+  LOG(LS_INFO) << " channels per packet: " << absd.mChannelsPerFrame;
+  LOG(LS_INFO) << " bits per channel: " << absd.mBitsPerChannel;
+  LOG(LS_INFO) << " reserved: " << absd.mReserved;
+}
+
+// Helper method that logs essential device information strings.
+static void LogDeviceInfo() {
+  LOG(LS_INFO) << "LogDeviceInfo";
+  @autoreleasepool {
+    LOG(LS_INFO) << " system name: " << ios::GetSystemName();
+    LOG(LS_INFO) << " system version: " << ios::GetSystemVersion();
+    LOG(LS_INFO) << " device type: " << ios::GetDeviceType();
+    LOG(LS_INFO) << " device name: " << ios::GetDeviceName();
+  }
+}
+#endif  // !defined(NDEBUG)
+
+AudioDeviceIOS::AudioDeviceIOS()
+    : audio_device_buffer_(nullptr),
+      vpio_unit_(nullptr),
+      recording_(0),
+      playing_(0),
+      initialized_(false),
+      rec_is_initialized_(false),
+      play_is_initialized_(false),
+      audio_interruption_observer_(nullptr),
+      route_change_observer_(nullptr) {
+  LOGI() << "ctor" << ios::GetCurrentThreadDescription();
 }
 
 AudioDeviceIOS::~AudioDeviceIOS() {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
-
-    Terminate();
-
-    delete &_critSect;
+  LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  Terminate();
 }
 
-
-// ============================================================================
-//                                     API
-// ============================================================================
-
 void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    _ptrAudioBuffer = audioBuffer;
-
-    // inform the AudioBuffer about default settings for this implementation
-    _ptrAudioBuffer->SetRecordingSampleRate(ENGINE_REC_BUF_SIZE_IN_SAMPLES);
-    _ptrAudioBuffer->SetPlayoutSampleRate(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
-    _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
-    _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-int32_t AudioDeviceIOS::ActiveAudioLayer(
-    AudioDeviceModule::AudioLayer& audioLayer) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
-    return 0;
+  LOGI() << "AttachAudioBuffer";
+  RTC_DCHECK(audioBuffer);
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  audio_device_buffer_ = audioBuffer;
 }
 
 int32_t AudioDeviceIOS::Init() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_initialized) {
-        return 0;
-    }
-
-    _isShutDown = false;
-
-    // Create and start capture thread
-    if (!_captureWorkerThread) {
-        _captureWorkerThread = ThreadWrapper::CreateThread(
-            RunCapture, this, "CaptureWorkerThread");
-        bool res = _captureWorkerThread->Start();
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                     _id, "CaptureWorkerThread started (res=%d)", res);
-        _captureWorkerThread->SetPriority(kRealtimePriority);
-    } else {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                     _id, "Thread already created");
-    }
-    _playWarning = 0;
-    _playError = 0;
-    _recWarning = 0;
-    _recError = 0;
-
-    _initialized = true;
-
+  LOGI() << "Init";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (initialized_) {
     return 0;
+  }
+#if !defined(NDEBUG)
+  LogDeviceInfo();
+#endif
+  // Store the preferred sample rate and preferred number of channels already
+  // here. They have not been set and confirmed yet since ActivateAudioSession()
+  // is not called until audio is about to start. However, it makes sense to
+  // store the parameters now and then verify at a later stage.
+  playout_parameters_.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
+  record_parameters_.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
+  // Ensure that the audio device buffer (ADB) knows about the internal audio
+  // parameters. Note that, even if we are unable to get a mono audio session,
+  // we will always tell the I/O audio unit to do a channel format conversion
+  // to guarantee mono on the "input side" of the audio unit.
+  UpdateAudioDeviceBuffer();
+  initialized_ = true;
+  return 0;
 }
 
 int32_t AudioDeviceIOS::Terminate() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    if (!_initialized) {
-        return 0;
-    }
-
-
-    // Stop capture thread
-    if (_captureWorkerThread) {
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                     _id, "Stopping CaptureWorkerThread");
-        bool res = _captureWorkerThread->Stop();
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                     _id, "CaptureWorkerThread stopped (res=%d)", res);
-        _captureWorkerThread.reset();
-    }
-
-    // Shut down Audio Unit
-    ShutdownPlayOrRecord();
-
-    _isShutDown = true;
-    _initialized = false;
-    _speakerIsInitialized = false;
-    _micIsInitialized = false;
-    _playoutDeviceIsSpecified = false;
-    _recordingDeviceIsSpecified = false;
+  LOGI() << "Terminate";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!initialized_) {
     return 0;
-}
-
-bool AudioDeviceIOS::Initialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return (_initialized);
-}
-
-int32_t AudioDeviceIOS::InitSpeaker() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Not initialized");
-        return -1;
+  }
+  StopPlayout();
+  StopRecording();
+  initialized_ = false;
+  {
+    rtc::GlobalLockScope ls(&g_lock);
+    if (g_audio_session_users != 0) {
+      LOG(LS_WARNING) << "Object is destructed with an active audio session";
     }
-
-    if (_playing) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Cannot init speaker when playing");
-        return -1;
-    }
-
-    if (!_playoutDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Playout device is not specified");
-        return -1;
-    }
-
-    // Do nothing
-    _speakerIsInitialized = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::InitMicrophone() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Not initialized");
-        return -1;
-    }
-
-    if (_recording) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Cannot init mic when recording");
-        return -1;
-    }
-
-    if (!_recordingDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Recording device is not specified");
-        return -1;
-    }
-
-    // Do nothing
-
-    _micIsInitialized = true;
-
-    return 0;
-}
-
-bool AudioDeviceIOS::SpeakerIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return _speakerIsInitialized;
-}
-
-bool AudioDeviceIOS::MicrophoneIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return _micIsInitialized;
-}
-
-int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Speaker volume not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetSpeakerVolume(volume=%u)", volume);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::SetWaveOutVolume(uint16_t volumeLeft,
-                                     uint16_t volumeRight) {
-    WEBRTC_TRACE(
-        kTraceModuleCall,
-        kTraceAudioDevice,
-        _id,
-        "AudioDeviceIOS::SetWaveOutVolume(volumeLeft=%u, volumeRight=%u)",
-        volumeLeft, volumeRight);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-
-    return -1;
-}
-
-int32_t
-AudioDeviceIOS::WaveOutVolume(uint16_t& /*volumeLeft*/,
-                                 uint16_t& /*volumeRight*/) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MinSpeakerVolume(
-    uint32_t& minVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::SpeakerVolumeStepSize(uint16_t& stepSize) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Speaker mute not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Mic mute not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneBoostIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Mic boost not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneBoost(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetMicrophoneBoost(enable=%u)", enable);
-
-    if (!_micIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Microphone not initialized");
-        return -1;
-    }
-
-    if (enable) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  SetMicrophoneBoost cannot be enabled on this platform");
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::MicrophoneBoost(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    if (!_micIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Microphone not initialized");
-        return -1;
-    }
-
-    enabled = false;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Stereo recording not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetStereoRecording(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetStereoRecording(enable=%u)", enable);
-
-    if (enable) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Stereo recording is not supported on this platform");
-        return -1;
-    }
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    enabled = false;
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Stereo playout not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetStereoPlayout(enable=%u)", enable);
-
-    if (enable) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Stereo playout is not supported on this platform");
-        return -1;
-    }
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    enabled = false;
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetAGC(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetAGC(enable=%d)", enable);
-
-    _AGC = enable;
-
-    return 0;
-}
-
-bool AudioDeviceIOS::AGC() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    return _AGC;
-}
-
-int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Mic volume not supported on IOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetMicrophoneVolume(volume=%u)", volume);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MicrophoneVolumeStepSize(
-                                            uint16_t& stepSize) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int16_t AudioDeviceIOS::PlayoutDevices() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    return (int16_t)1;
-}
-
-int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetPlayoutDevice(index=%u)", index);
-
-    if (_playIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Playout already initialized");
-        return -1;
-    }
-
-    if (index !=0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  SetPlayoutDevice invalid index");
-        return -1;
-    }
-    _playoutDeviceIsSpecified = true;
-
-    return 0;
-}
-
-int32_t
-    AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) {
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::PlayoutDeviceName(uint16_t index,
-                                         char name[kAdmMaxDeviceNameSize],
-                                         char guid[kAdmMaxGuidSize]) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::PlayoutDeviceName(index=%u)", index);
-
-    if (index != 0) {
-        return -1;
-    }
-    // return empty strings
-    memset(name, 0, kAdmMaxDeviceNameSize);
-    if (guid != NULL) {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
-
-    return 0;
-}
-
-int32_t
-    AudioDeviceIOS::RecordingDeviceName(uint16_t index,
-                                           char name[kAdmMaxDeviceNameSize],
-                                           char guid[kAdmMaxGuidSize]) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::RecordingDeviceName(index=%u)", index);
-
-    if (index != 0) {
-        return -1;
-    }
-    // return empty strings
-    memset(name, 0, kAdmMaxDeviceNameSize);
-    if (guid != NULL) {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
-
-    return 0;
-}
-
-int16_t AudioDeviceIOS::RecordingDevices() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    return (int16_t)1;
-}
-
-int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetRecordingDevice(index=%u)", index);
-
-    if (_recIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Recording already initialized");
-        return -1;
-    }
-
-    if (index !=0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  SetRecordingDevice invalid index");
-        return -1;
-    }
-
-    _recordingDeviceIsSpecified = true;
-
-    return 0;
-}
-
-int32_t
-    AudioDeviceIOS::SetRecordingDevice(
-                                        AudioDeviceModule::WindowsDeviceType) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
-}
-
-// ----------------------------------------------------------------------------
-//  SetLoudspeakerStatus
-//
-//  Change the default receiver playout route to speaker.
-//
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) {
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetLoudspeakerStatus(enable=%d)", enable);
-
-    AVAudioSession* session = [AVAudioSession sharedInstance];
-    NSString* category = session.category;
-    AVAudioSessionCategoryOptions options = session.categoryOptions;
-    // Respect old category options if category is
-    // AVAudioSessionCategoryPlayAndRecord. Otherwise reset it since old options
-    // might not be valid for this category.
-    if ([category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) {
-      if (enable) {
-        options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
-      } else {
-        options &= ~AVAudioSessionCategoryOptionDefaultToSpeaker;
-      }
-    } else {
-      options = AVAudioSessionCategoryOptionDefaultToSpeaker;
-    }
-
-    NSError* error = nil;
-    [session setCategory:AVAudioSessionCategoryPlayAndRecord
-             withOptions:options
-                   error:&error];
-    if (error != nil) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                   "Error changing default output route ");
-      return -1;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::GetLoudspeakerStatus(bool &enabled) const {
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetLoudspeakerStatus(enabled=?)");
-
-    AVAudioSession* session = [AVAudioSession sharedInstance];
-    AVAudioSessionCategoryOptions options = session.categoryOptions;
-    enabled = options & AVAudioSessionCategoryOptionDefaultToSpeaker;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    available = false;
-
-    // Try to initialize the playout side
-    int32_t res = InitPlayout();
-
-    // Cancel effect of initialization
-    StopPlayout();
-
-    if (res != -1) {
-        available = true;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    available = false;
-
-    // Try to initialize the recording side
-    int32_t res = InitRecording();
-
-    // Cancel effect of initialization
-    StopRecording();
-
-    if (res != -1) {
-        available = true;
-    }
-
-    return 0;
+    RTC_DCHECK_GE(g_audio_session_users, 0);
+  }
+  return 0;
 }
 
 int32_t AudioDeviceIOS::InitPlayout() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "  Not initialized");
-        return -1;
+  LOGI() << "InitPlayout";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!play_is_initialized_);
+  RTC_DCHECK(!playing_);
+  if (!rec_is_initialized_) {
+    if (!InitPlayOrRecord()) {
+      LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitPlayout!";
+      return -1;
     }
-
-    if (_playing) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Playout already started");
-        return -1;
-    }
-
-    if (_playIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout already initialized");
-        return 0;
-    }
-
-    if (!_playoutDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Playout device is not specified");
-        return -1;
-    }
-
-    // Initialize the speaker
-    if (InitSpeaker() == -1) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitSpeaker() failed");
-    }
-
-    _playIsInitialized = true;
-
-    if (!_recIsInitialized) {
-        // Audio init
-        if (InitPlayOrRecord() == -1) {
-            // todo: Handle error
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  InitPlayOrRecord() failed");
-        }
-    } else {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-        "  Recording already initialized - InitPlayOrRecord() not called");
-    }
-
-    return 0;
-}
-
-bool AudioDeviceIOS::PlayoutIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-    return (_playIsInitialized);
+  }
+  play_is_initialized_ = true;
+  return 0;
 }
 
 int32_t AudioDeviceIOS::InitRecording() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Not initialized");
-        return -1;
+  LOGI() << "InitRecording";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!rec_is_initialized_);
+  RTC_DCHECK(!recording_);
+  if (!play_is_initialized_) {
+    if (!InitPlayOrRecord()) {
+      LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitRecording!";
+      return -1;
     }
-
-    if (_recording) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Recording already started");
-        return -1;
-    }
-
-    if (_recIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Recording already initialized");
-        return 0;
-    }
-
-    if (!_recordingDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Recording device is not specified");
-        return -1;
-    }
-
-    // Initialize the microphone
-    if (InitMicrophone() == -1) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitMicrophone() failed");
-    }
-
-    _recIsInitialized = true;
-
-    if (!_playIsInitialized) {
-        // Audio init
-        if (InitPlayOrRecord() == -1) {
-            // todo: Handle error
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  InitPlayOrRecord() failed");
-        }
-    } else {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout already initialized - InitPlayOrRecord() " \
-                     "not called");
-    }
-
-    return 0;
-}
-
-bool AudioDeviceIOS::RecordingIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-    return (_recIsInitialized);
-}
-
-int32_t AudioDeviceIOS::StartRecording() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Recording not initialized");
-        return -1;
-    }
-
-    if (_recording) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Recording already started");
-        return 0;
-    }
-
-    // Reset recording buffer
-    memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
-    memset(_recordingLength, 0, sizeof(_recordingLength));
-    memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
-    _recordingCurrentSeq = 0;
-    _recordingBufferTotalSize = 0;
-    _recordingDelay = 0;
-    _recordingDelayHWAndOS = 0;
-    // Make sure first call to update delay function will update delay
-    _recordingDelayMeasurementCounter = 9999;
-    _recWarning = 0;
-    _recError = 0;
-
-    if (!_playing) {
-        // Start Audio Unit
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "  Starting Audio Unit");
-        OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                         "  Error starting Audio Unit (result=%d)", result);
-            return -1;
-        }
-    }
-
-    _recording = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StopRecording() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Recording is not initialized");
-        return 0;
-    }
-
-    _recording = false;
-
-    if (!_playing) {
-        // Both playout and recording has stopped, shutdown the device
-        ShutdownPlayOrRecord();
-    }
-
-    _recIsInitialized = false;
-    _micIsInitialized = false;
-
-    return 0;
-}
-
-bool AudioDeviceIOS::Recording() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-    return (_recording);
+  }
+  rec_is_initialized_ = true;
+  return 0;
 }
 
 int32_t AudioDeviceIOS::StartPlayout() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    // This lock is (among other things) needed to avoid concurrency issues
-    // with capture thread
-    // shutting down Audio Unit
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Playout not initialized");
-        return -1;
+  LOGI() << "StartPlayout";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(play_is_initialized_);
+  RTC_DCHECK(!playing_);
+  fine_audio_buffer_->ResetPlayout();
+  if (!recording_) {
+    OSStatus result = AudioOutputUnitStart(vpio_unit_);
+    if (result != noErr) {
+      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed for StartPlayout: "
+                      << result;
+      return -1;
     }
-
-    if (_playing) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playing already started");
-        return 0;
-    }
-
-    // Reset playout buffer
-    memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-    _playoutBufferUsed = 0;
-    _playoutDelay = 0;
-    // Make sure first call to update delay function will update delay
-    _playoutDelayMeasurementCounter = 9999;
-    _playWarning = 0;
-    _playError = 0;
-
-    if (!_recording) {
-        // Start Audio Unit
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "  Starting Audio Unit");
-        OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                         "  Error starting Audio Unit (result=%d)", result);
-            return -1;
-        }
-    }
-
-    _playing = true;
-
-    return 0;
+    LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
+  }
+  rtc::AtomicOps::ReleaseStore(&playing_, 1);
+  return 0;
 }
 
 int32_t AudioDeviceIOS::StopPlayout() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout is not initialized");
-        return 0;
-    }
-
-    _playing = false;
-
-    if (!_recording) {
-        // Both playout and recording has stopped, signal shutdown the device
-        ShutdownPlayOrRecord();
-    }
-
-    _playIsInitialized = false;
-    _speakerIsInitialized = false;
-
+  LOGI() << "StopPlayout";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!play_is_initialized_ || !playing_) {
     return 0;
+  }
+  if (!recording_) {
+    ShutdownPlayOrRecord();
+  }
+  play_is_initialized_ = false;
+  rtc::AtomicOps::ReleaseStore(&playing_, 0);
+  return 0;
 }
 
-bool AudioDeviceIOS::Playing() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return (_playing);
+int32_t AudioDeviceIOS::StartRecording() {
+  LOGI() << "StartRecording";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(rec_is_initialized_);
+  RTC_DCHECK(!recording_);
+  fine_audio_buffer_->ResetRecord();
+  if (!playing_) {
+    OSStatus result = AudioOutputUnitStart(vpio_unit_);
+    if (result != noErr) {
+      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed for StartRecording: "
+                      << result;
+      return -1;
+    }
+    LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
+  }
+  rtc::AtomicOps::ReleaseStore(&recording_, 1);
+  return 0;
 }
 
-// ----------------------------------------------------------------------------
-//  ResetAudioDevice
-//
-//  Disable playout and recording, signal to capture thread to shutdown,
-//  and set enable states after shutdown to same as current.
-//  In capture thread audio device will be shutdown, then started again.
-// ----------------------------------------------------------------------------
-int32_t AudioDeviceIOS::ResetAudioDevice() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized && !_recIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout or recording not initialized, doing nothing");
-        return 0;  // Nothing to reset
-    }
-
-    // Store the states we have before stopping to restart below
-    bool initPlay = _playIsInitialized;
-    bool play = _playing;
-    bool initRec = _recIsInitialized;
-    bool rec = _recording;
-
-    int res(0);
-
-    // Stop playout and recording
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  Stopping playout and recording");
-    res += StopPlayout();
-    res += StopRecording();
-
-    // Restart
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  Restarting playout and recording (%d, %d, %d, %d)",
-                 initPlay, play, initRec, rec);
-    if (initPlay) res += InitPlayout();
-    if (initRec)  res += InitRecording();
-    if (play)     res += StartPlayout();
-    if (rec)      res += StartRecording();
-
-    if (0 != res) {
-        // Logging is done in init/start/stop calls above
-        return -1;
-    }
-
+int32_t AudioDeviceIOS::StopRecording() {
+  LOGI() << "StopRecording";
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!rec_is_initialized_ || !recording_) {
     return 0;
+  }
+  if (!playing_) {
+    ShutdownPlayOrRecord();
+  }
+  rec_is_initialized_ = false;
+  rtc::AtomicOps::ReleaseStore(&recording_, 0);
+  return 0;
+}
+
+// Change the default receiver playout route to speaker.
+int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) {
+  LOGI() << "SetLoudspeakerStatus(" << enable << ")";
+
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  NSString* category = session.category;
+  AVAudioSessionCategoryOptions options = session.categoryOptions;
+  // Respect old category options if category is
+  // AVAudioSessionCategoryPlayAndRecord. Otherwise reset it since old options
+  // might not be valid for this category.
+  if ([category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) {
+    if (enable) {
+      options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
+    } else {
+      options &= ~AVAudioSessionCategoryOptionDefaultToSpeaker;
+    }
+  } else {
+    options = AVAudioSessionCategoryOptionDefaultToSpeaker;
+  }
+  NSError* error = nil;
+  BOOL success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
+                          withOptions:options
+                                error:&error];
+  ios::CheckAndLogError(success, error);
+  return (error == nil) ? 0 : -1;
+}
+
+int32_t AudioDeviceIOS::GetLoudspeakerStatus(bool& enabled) const {
+  LOGI() << "GetLoudspeakerStatus";
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  AVAudioSessionCategoryOptions options = session.categoryOptions;
+  enabled = options & AVAudioSessionCategoryOptionDefaultToSpeaker;
+  return 0;
 }
 
 int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
-    delayMS = _playoutDelay;
-    return 0;
+  delayMS = kFixedPlayoutDelayEstimate;
+  return 0;
 }
 
 int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const {
-    delayMS = _recordingDelay;
-    return 0;
+  delayMS = kFixedRecordDelayEstimate;
+  return 0;
 }
 
-int32_t
-    AudioDeviceIOS::SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                                     uint16_t sizeMS) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetPlayoutBuffer(type=%u, sizeMS=%u)",
-                 type, sizeMS);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
+int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
+  LOGI() << "GetPlayoutAudioParameters";
+  RTC_DCHECK(playout_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  *params = playout_parameters_;
+  return 0;
 }
 
-int32_t
-    AudioDeviceIOS::PlayoutBuffer(AudioDeviceModule::BufferType& type,
-                                     uint16_t& sizeMS) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    type = AudioDeviceModule::kAdaptiveBufferSize;
-
-    sizeMS = _playoutDelay;
-
-    return 0;
+int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
+  LOGI() << "GetRecordAudioParameters";
+  RTC_DCHECK(record_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  *params = record_parameters_;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::CPULoad(uint16_t& /*load*/) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
+void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
+  LOGI() << "UpdateAudioDevicebuffer";
+  // AttachAudioBuffer() is called at construction by the main class but check
+  // just in case.
+  RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
+  // Inform the audio device buffer (ADB) about the new audio format.
+  audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
+  audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
+  audio_device_buffer_->SetRecordingSampleRate(
+      record_parameters_.sample_rate());
+  audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
 }
 
-bool AudioDeviceIOS::PlayoutWarning() const {
-    return (_playWarning > 0);
-}
-
-bool AudioDeviceIOS::PlayoutError() const {
-    return (_playError > 0);
-}
-
-bool AudioDeviceIOS::RecordingWarning() const {
-    return (_recWarning > 0);
-}
-
-bool AudioDeviceIOS::RecordingError() const {
-    return (_recError > 0);
-}
-
-void AudioDeviceIOS::ClearPlayoutWarning() {
-    _playWarning = 0;
-}
-
-void AudioDeviceIOS::ClearPlayoutError() {
-    _playError = 0;
-}
-
-void AudioDeviceIOS::ClearRecordingWarning() {
-    _recWarning = 0;
-}
-
-void AudioDeviceIOS::ClearRecordingError() {
-    _recError = 0;
-}
-
-// ============================================================================
-//                                 Private Methods
-// ============================================================================
-
-int32_t AudioDeviceIOS::InitPlayOrRecord() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    OSStatus result = -1;
-
-    // Check if already initialized
-    if (NULL != _auVoiceProcessing) {
-        // We already have initialized before and created any of the audio unit,
-        // check that all exist
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Already initialized");
-        // todo: Call AudioUnitReset() here and empty all buffers?
-        return 0;
+void AudioDeviceIOS::RegisterNotificationObservers() {
+  LOGI() << "RegisterNotificationObservers";
+  // This code block will be called when AVAudioSessionInterruptionNotification
+  // is observed.
+  void (^interrupt_block)(NSNotification*) = ^(NSNotification* notification) {
+    NSNumber* type_number =
+        notification.userInfo[AVAudioSessionInterruptionTypeKey];
+    AVAudioSessionInterruptionType type =
+        (AVAudioSessionInterruptionType)type_number.unsignedIntegerValue;
+    LOG(LS_INFO) << "Audio session interruption:";
+    switch (type) {
+      case AVAudioSessionInterruptionTypeBegan:
+        // The system has deactivated our audio session.
+        // Stop the active audio unit.
+        LOG(LS_INFO) << " Began => stopping the audio unit";
+        LOG_IF_ERROR(AudioOutputUnitStop(vpio_unit_),
+                     "Failed to stop the the Voice-Processing I/O unit");
+        break;
+      case AVAudioSessionInterruptionTypeEnded:
+        // The interruption has ended. Restart the audio session and start the
+        // initialized audio unit again.
+        LOG(LS_INFO) << " Ended => restarting audio session and audio unit";
+        NSError* error = nil;
+        BOOL success = NO;
+        AVAudioSession* session = [AVAudioSession sharedInstance];
+        success = [session setActive:YES error:&error];
+        if (CheckAndLogError(success, error)) {
+          LOG_IF_ERROR(AudioOutputUnitStart(vpio_unit_),
+                       "Failed to start the the Voice-Processing I/O unit");
+        }
+        break;
     }
+  };
 
-    // Create Voice Processing Audio Unit
-    AudioComponentDescription desc;
-    AudioComponent comp;
+  // This code block will be called when AVAudioSessionRouteChangeNotification
+  // is observed.
+  void (^route_change_block)(NSNotification*) =
+      ^(NSNotification* notification) {
+        // Get reason for current route change.
+        NSNumber* reason_number =
+            notification.userInfo[AVAudioSessionRouteChangeReasonKey];
+        AVAudioSessionRouteChangeReason reason =
+            (AVAudioSessionRouteChangeReason)reason_number.unsignedIntegerValue;
+        bool valid_route_change = true;
+        LOG(LS_INFO) << "Route change:";
+        switch (reason) {
+          case AVAudioSessionRouteChangeReasonUnknown:
+            LOG(LS_INFO) << " ReasonUnknown";
+            break;
+          case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
+            LOG(LS_INFO) << " NewDeviceAvailable";
+            break;
+          case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
+            LOG(LS_INFO) << " OldDeviceUnavailable";
+            break;
+          case AVAudioSessionRouteChangeReasonCategoryChange:
+            // It turns out that we see this notification (at least in iOS 9.2)
+            // when making a switch from a BT device to e.g. Speaker using the
+            // iOS Control Center and that we therefore must check if the sample
+            // rate has changed. And if so is the case, restart the audio unit.
+            LOG(LS_INFO) << " CategoryChange";
+            LOG(LS_INFO) << " New category: " << ios::GetAudioSessionCategory();
+            break;
+          case AVAudioSessionRouteChangeReasonOverride:
+            LOG(LS_INFO) << " Override";
+            break;
+          case AVAudioSessionRouteChangeReasonWakeFromSleep:
+            LOG(LS_INFO) << " WakeFromSleep";
+            break;
+          case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
+            LOG(LS_INFO) << " NoSuitableRouteForCategory";
+            break;
+          case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
+            // The set of input and output ports has not changed, but their
+            // configuration has, e.g., a port’s selected data source has
+            // changed. Ignore this type of route change since we are focusing
+            // on detecting headset changes.
+            LOG(LS_INFO) << " RouteConfigurationChange (ignored)";
+            valid_route_change = false;
+            break;
+        }
 
-    desc.componentType = kAudioUnitType_Output;
-    desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
-    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-    desc.componentFlags = 0;
-    desc.componentFlagsMask = 0;
+        if (valid_route_change) {
+          // Log previous route configuration.
+          AVAudioSessionRouteDescription* prev_route =
+              notification.userInfo[AVAudioSessionRouteChangePreviousRouteKey];
+          LOG(LS_INFO) << "Previous route:";
+          LOG(LS_INFO) << ios::StdStringFromNSString(
+              [NSString stringWithFormat:@"%@", prev_route]);
 
-    comp = AudioComponentFindNext(NULL, &desc);
-    if (NULL == comp) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not find audio component for Audio Unit");
-        return -1;
-    }
-
-    result = AudioComponentInstanceNew(comp, &_auVoiceProcessing);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not create Audio Unit instance (result=%d)",
-                     result);
-        return -1;
-    }
-
-    // Set preferred hardware sample rate to 16 kHz
-    NSError* error = nil;
-    AVAudioSession* session = [AVAudioSession sharedInstance];
-    Float64 preferredSampleRate(16000.0);
-    [session setPreferredSampleRate:preferredSampleRate
-                              error:&error];
-    if (error != nil) {
-        const char* errorString = [[error localizedDescription] UTF8String];
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Could not set preferred sample rate: %s", errorString);
-    }
-    error = nil;
-    [session setMode:AVAudioSessionModeVoiceChat
-               error:&error];
-    if (error != nil) {
-        const char* errorString = [[error localizedDescription] UTF8String];
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Could not set mode: %s", errorString);
-    }
-    error = nil;
-    [session setCategory:AVAudioSessionCategoryPlayAndRecord
-                   error:&error];
-    if (error != nil) {
-        const char* errorString = [[error localizedDescription] UTF8String];
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Could not set category: %s", errorString);
-    }
-
-    //////////////////////
-    // Setup Voice Processing Audio Unit
-
-    // Note: For Signal Processing AU element 0 is output bus, element 1 is
-    //       input bus for global scope element is irrelevant (always use
-    //       element 0)
-
-    // Enable IO on both elements
-
-    // todo: Below we just log and continue upon error. We might want
-    //       to close AU and return error for some cases.
-    // todo: Log info about setup.
-
-    UInt32 enableIO = 1;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioOutputUnitProperty_EnableIO,
-                                  kAudioUnitScope_Input,
-                                  1,  // input bus
-                                  &enableIO,
-                                  sizeof(enableIO));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not enable IO on input (result=%d)", result);
-    }
-
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioOutputUnitProperty_EnableIO,
-                                  kAudioUnitScope_Output,
-                                  0,   // output bus
-                                  &enableIO,
-                                  sizeof(enableIO));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not enable IO on output (result=%d)", result);
-    }
-
-    // Disable AU buffer allocation for the recorder, we allocate our own
-    UInt32 flag = 0;
-    result = AudioUnitSetProperty(
-        _auVoiceProcessing, kAudioUnitProperty_ShouldAllocateBuffer,
-        kAudioUnitScope_Output,  1, &flag, sizeof(flag));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Could not disable AU buffer allocation (result=%d)",
-                     result);
-        // Should work anyway
-    }
-
-    // Set recording callback
-    AURenderCallbackStruct auCbS;
-    memset(&auCbS, 0, sizeof(auCbS));
-    auCbS.inputProc = RecordProcess;
-    auCbS.inputProcRefCon = this;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioOutputUnitProperty_SetInputCallback,
-                                  kAudioUnitScope_Global, 1,
-                                  &auCbS, sizeof(auCbS));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set record callback for Audio Unit (result=%d)",
-            result);
-    }
-
-    // Set playout callback
-    memset(&auCbS, 0, sizeof(auCbS));
-    auCbS.inputProc = PlayoutProcess;
-    auCbS.inputProcRefCon = this;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_SetRenderCallback,
-                                  kAudioUnitScope_Global, 0,
-                                  &auCbS, sizeof(auCbS));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set play callback for Audio Unit (result=%d)",
-            result);
-    }
-
-    // Get stream format for out/0
-    AudioStreamBasicDescription playoutDesc;
-    UInt32 size = sizeof(playoutDesc);
-    result = AudioUnitGetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Output, 0, &playoutDesc,
-                                  &size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not get stream format Audio Unit out/0 (result=%d)",
-            result);
-    }
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  Audio Unit playout opened in sampling rate %f",
-                 playoutDesc.mSampleRate);
-
-    playoutDesc.mSampleRate = preferredSampleRate;
-
-    // Store the sampling frequency to use towards the Audio Device Buffer
-    // todo: Add 48 kHz (increase buffer sizes). Other fs?
-    if ((playoutDesc.mSampleRate > 44090.0)
-        && (playoutDesc.mSampleRate < 44110.0)) {
-        _adbSampFreq = 44100;
-    } else if ((playoutDesc.mSampleRate > 15990.0)
-               && (playoutDesc.mSampleRate < 16010.0)) {
-        _adbSampFreq = 16000;
-    } else if ((playoutDesc.mSampleRate > 7990.0)
-               && (playoutDesc.mSampleRate < 8010.0)) {
-        _adbSampFreq = 8000;
-    } else {
-        _adbSampFreq = 0;
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Audio Unit out/0 opened in unknown sampling rate (%f)",
-            playoutDesc.mSampleRate);
-        // todo: We should bail out here.
-    }
-
-    // Set the audio device buffer sampling rate,
-    // we assume we get the same for play and record
-    if (_ptrAudioBuffer->SetRecordingSampleRate(_adbSampFreq) < 0) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set audio device buffer recording sampling rate (%d)",
-            _adbSampFreq);
-    }
-
-    if (_ptrAudioBuffer->SetPlayoutSampleRate(_adbSampFreq) < 0) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set audio device buffer playout sampling rate (%d)",
-            _adbSampFreq);
-    }
-
-    // Set stream format for in/0  (use same sampling frequency as for out/0)
-    playoutDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-                               | kLinearPCMFormatFlagIsPacked
-                               | kLinearPCMFormatFlagIsNonInterleaved;
-    playoutDesc.mBytesPerPacket = 2;
-    playoutDesc.mFramesPerPacket = 1;
-    playoutDesc.mBytesPerFrame = 2;
-    playoutDesc.mChannelsPerFrame = 1;
-    playoutDesc.mBitsPerChannel = 16;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Input, 0, &playoutDesc, size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set stream format Audio Unit in/0 (result=%d)",
-            result);
-    }
-
-    // Get stream format for in/1
-    AudioStreamBasicDescription recordingDesc;
-    size = sizeof(recordingDesc);
-    result = AudioUnitGetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Input, 1, &recordingDesc,
-                                  &size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not get stream format Audio Unit in/1 (result=%d)",
-            result);
-    }
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  Audio Unit recording opened in sampling rate %f",
-                 recordingDesc.mSampleRate);
-
-    recordingDesc.mSampleRate = preferredSampleRate;
-
-    // Set stream format for out/1 (use same sampling frequency as for in/1)
-    recordingDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-                                 | kLinearPCMFormatFlagIsPacked
-                                 | kLinearPCMFormatFlagIsNonInterleaved;
-
-    recordingDesc.mBytesPerPacket = 2;
-    recordingDesc.mFramesPerPacket = 1;
-    recordingDesc.mBytesPerFrame = 2;
-    recordingDesc.mChannelsPerFrame = 1;
-    recordingDesc.mBitsPerChannel = 16;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Output, 1, &recordingDesc,
-                                  size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set stream format Audio Unit out/1 (result=%d)",
-            result);
-    }
-
-    // Initialize here already to be able to get/set stream properties.
-    result = AudioUnitInitialize(_auVoiceProcessing);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not init Audio Unit (result=%d)", result);
-    }
-
-    // Get hardware sample rate for logging (see if we get what we asked for)
-    double sampleRate = session.sampleRate;
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  Current HW sample rate is %f, ADB sample rate is %d",
-                 sampleRate, _adbSampFreq);
-
-    // Listen to audio interruptions.
-    NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
-    id observer =
-        [center addObserverForName:AVAudioSessionInterruptionNotification
-                            object:nil
-                             queue:[NSOperationQueue mainQueue]
-                        usingBlock:^(NSNotification* notification) {
-          NSNumber* typeNumber =
-              [notification userInfo][AVAudioSessionInterruptionTypeKey];
-          AVAudioSessionInterruptionType type =
-              (AVAudioSessionInterruptionType)[typeNumber unsignedIntegerValue];
-          switch (type) {
-            case AVAudioSessionInterruptionTypeBegan:
-              // At this point our audio session has been deactivated and the
-              // audio unit render callbacks no longer occur. Nothing to do.
-              break;
-            case AVAudioSessionInterruptionTypeEnded: {
-              NSError* error = nil;
-              AVAudioSession* session = [AVAudioSession sharedInstance];
-              [session setActive:YES
-                           error:&error];
-              if (error != nil) {
-                  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                               "Error activating audio session");
-              }
-              // Post interruption the audio unit render callbacks don't
-              // automatically continue, so we restart the unit manually here.
-              AudioOutputUnitStop(_auVoiceProcessing);
-              AudioOutputUnitStart(_auVoiceProcessing);
-              break;
+          // Only restart audio for a valid route change and if the
+          // session sample rate has changed.
+          AVAudioSession* session = [AVAudioSession sharedInstance];
+          const double session_sample_rate = session.sampleRate;
+          LOG(LS_INFO) << "session sample rate: " << session_sample_rate;
+          if (playout_parameters_.sample_rate() != session_sample_rate) {
+            if (!RestartAudioUnitWithNewFormat(session_sample_rate)) {
+              LOG(LS_ERROR) << "Audio restart failed";
             }
           }
-        }];
-    // Increment refcount on observer using ARC bridge. Instance variable is a
-    // void* instead of an id because header is included in other pure C++
-    // files.
-    _audioInterruptionObserver = (__bridge_retained void*)observer;
+        }
+      };
 
-    // Activate audio session.
-    error = nil;
-    [session setActive:YES
-                 error:&error];
-    if (error != nil) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "Error activating audio session");
-    }
+  // Get the default notification center of the current process.
+  NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
 
-    return 0;
+  // Add AVAudioSessionInterruptionNotification observer.
+  id interruption_observer =
+      [center addObserverForName:AVAudioSessionInterruptionNotification
+                          object:nil
+                           queue:[NSOperationQueue mainQueue]
+                      usingBlock:interrupt_block];
+  // Add AVAudioSessionRouteChangeNotification observer.
+  id route_change_observer =
+      [center addObserverForName:AVAudioSessionRouteChangeNotification
+                          object:nil
+                           queue:[NSOperationQueue mainQueue]
+                      usingBlock:route_change_block];
+
+  // Increment refcount on observers using ARC bridge. Instance variable is a
+  // void* instead of an id because header is included in other pure C++
+  // files.
+  audio_interruption_observer_ = (__bridge_retained void*)interruption_observer;
+  route_change_observer_ = (__bridge_retained void*)route_change_observer;
 }
 
-int32_t AudioDeviceIOS::ShutdownPlayOrRecord() {
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    if (_audioInterruptionObserver != NULL) {
-        NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
-        // Transfer ownership of observer back to ARC, which will dealloc the
-        // observer once it exits this scope.
-        id observer = (__bridge_transfer id)_audioInterruptionObserver;
-        [center removeObserver:observer];
-        _audioInterruptionObserver = NULL;
-    }
-
-    // Close and delete AU
-    OSStatus result = -1;
-    if (NULL != _auVoiceProcessing) {
-        result = AudioOutputUnitStop(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                "  Error stopping Audio Unit (result=%d)", result);
-        }
-        result = AudioComponentInstanceDispose(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                "  Error disposing Audio Unit (result=%d)", result);
-        }
-        _auVoiceProcessing = NULL;
-    }
-
-    return 0;
+void AudioDeviceIOS::UnregisterNotificationObservers() {
+  LOGI() << "UnregisterNotificationObservers";
+  // Transfer ownership of observer back to ARC, which will deallocate the
+  // observer once it exits this scope.
+  NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
+  if (audio_interruption_observer_ != nullptr) {
+    id observer = (__bridge_transfer id)audio_interruption_observer_;
+    [center removeObserver:observer];
+    audio_interruption_observer_ = nullptr;
+  }
+  if (route_change_observer_ != nullptr) {
+    id observer = (__bridge_transfer id)route_change_observer_;
+    [center removeObserver:observer];
+    route_change_observer_ = nullptr;
+  }
 }
 
-// ============================================================================
-//                                  Thread Methods
-// ============================================================================
+void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
+  LOGI() << "SetupAudioBuffersForActiveAudioSession";
+  // Verify the current values once the audio session has been activated.
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  LOG(LS_INFO) << " sample rate: " << session.sampleRate;
+  LOG(LS_INFO) << " IO buffer duration: " << session.IOBufferDuration;
+  LOG(LS_INFO) << " output channels: " << session.outputNumberOfChannels;
+  LOG(LS_INFO) << " input channels: " << session.inputNumberOfChannels;
+  LOG(LS_INFO) << " output latency: " << session.outputLatency;
+  LOG(LS_INFO) << " input latency: " << session.inputLatency;
 
-OSStatus
-    AudioDeviceIOS::RecordProcess(void *inRefCon,
-                                  AudioUnitRenderActionFlags *ioActionFlags,
-                                  const AudioTimeStamp *inTimeStamp,
-                                  UInt32 inBusNumber,
-                                  UInt32 inNumberFrames,
-                                  AudioBufferList *ioData) {
-    AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
+  // Log a warning message for the case when we are unable to set the preferred
+  // hardware sample rate but continue and use the non-ideal sample rate after
+  // reinitializing the audio parameters. Most BT headsets only support 8kHz or
+  // 16kHz.
+  if (session.sampleRate != kPreferredSampleRate) {
+    LOG(LS_WARNING) << "Unable to set the preferred sample rate";
+  }
 
-    return ptrThis->RecordProcessImpl(ioActionFlags,
-                                      inTimeStamp,
-                                      inBusNumber,
-                                      inNumberFrames);
+  // At this stage, we also know the exact IO buffer duration and can add
+  // that info to the existing audio parameters where it is converted into
+  // number of audio frames.
+  // Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
+  // Hence, 128 is the size we expect to see in upcoming render callbacks.
+  playout_parameters_.reset(session.sampleRate, playout_parameters_.channels(),
+                            session.IOBufferDuration);
+  RTC_DCHECK(playout_parameters_.is_complete());
+  record_parameters_.reset(session.sampleRate, record_parameters_.channels(),
+                           session.IOBufferDuration);
+  RTC_DCHECK(record_parameters_.is_complete());
+  LOG(LS_INFO) << " frames per I/O buffer: "
+               << playout_parameters_.frames_per_buffer();
+  LOG(LS_INFO) << " bytes per I/O buffer: "
+               << playout_parameters_.GetBytesPerBuffer();
+  RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(),
+                record_parameters_.GetBytesPerBuffer());
+
+  // Update the ADB parameters since the sample rate might have changed.
+  UpdateAudioDeviceBuffer();
+
+  // Create a modified audio buffer class which allows us to ask for,
+  // or deliver, any number of samples (and not only multiple of 10ms) to match
+  // the native audio unit buffer size.
+  RTC_DCHECK(audio_device_buffer_);
+  fine_audio_buffer_.reset(new FineAudioBuffer(
+      audio_device_buffer_, playout_parameters_.GetBytesPerBuffer(),
+      playout_parameters_.sample_rate()));
+
+  // The extra/temporary playoutbuffer must be of this size to avoid
+  // unnecessary memcpy while caching data between successive callbacks.
+  const int required_playout_buffer_size =
+      fine_audio_buffer_->RequiredPlayoutBufferSizeBytes();
+  LOG(LS_INFO) << " required playout buffer size: "
+               << required_playout_buffer_size;
+  playout_audio_buffer_.reset(new SInt8[required_playout_buffer_size]);
+
+  // Allocate AudioBuffers to be used as storage for the received audio.
+  // The AudioBufferList structure works as a placeholder for the
+  // AudioBuffer structure, which holds a pointer to the actual data buffer
+  // in |record_audio_buffer_|. Recorded audio will be rendered into this memory
+  // at each input callback when calling AudioUnitRender().
+  const int data_byte_size = record_parameters_.GetBytesPerBuffer();
+  record_audio_buffer_.reset(new SInt8[data_byte_size]);
+  audio_record_buffer_list_.mNumberBuffers = 1;
+  AudioBuffer* audio_buffer = &audio_record_buffer_list_.mBuffers[0];
+  audio_buffer->mNumberChannels = record_parameters_.channels();
+  audio_buffer->mDataByteSize = data_byte_size;
+  audio_buffer->mData = record_audio_buffer_.get();
 }
 
+bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
+  LOGI() << "SetupAndInitializeVoiceProcessingAudioUnit";
+  RTC_DCHECK(!vpio_unit_) << "VoiceProcessingIO audio unit already exists";
+  // Create an audio component description to identify the Voice-Processing
+  // I/O audio unit.
+  AudioComponentDescription vpio_unit_description;
+  vpio_unit_description.componentType = kAudioUnitType_Output;
+  vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
+  vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
+  vpio_unit_description.componentFlags = 0;
+  vpio_unit_description.componentFlagsMask = 0;
 
-OSStatus
-    AudioDeviceIOS::RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags,
-                                      const AudioTimeStamp *inTimeStamp,
-                                      uint32_t inBusNumber,
-                                      uint32_t inNumberFrames) {
-    // Setup some basic stuff
-    // Use temp buffer not to lock up recording buffer more than necessary
-    // todo: Make dataTmp a member variable with static size that holds
-    //       max possible frames?
-    int16_t* dataTmp = new int16_t[inNumberFrames];
-    memset(dataTmp, 0, 2*inNumberFrames);
+  // Obtain an audio unit instance given the description.
+  AudioComponent found_vpio_unit_ref =
+      AudioComponentFindNext(nullptr, &vpio_unit_description);
 
-    AudioBufferList abList;
-    abList.mNumberBuffers = 1;
-    abList.mBuffers[0].mData = dataTmp;
-    abList.mBuffers[0].mDataByteSize = 2*inNumberFrames;  // 2 bytes/sample
-    abList.mBuffers[0].mNumberChannels = 1;
+  // Create a Voice-Processing IO audio unit.
+  OSStatus result = noErr;
+  result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
+  if (result != noErr) {
+    vpio_unit_ = nullptr;
+    LOG(LS_ERROR) << "AudioComponentInstanceNew failed: " << result;
+    return false;
+  }
 
-    // Get data from mic
-    OSStatus res = AudioUnitRender(_auVoiceProcessing,
-                                   ioActionFlags, inTimeStamp,
-                                   inBusNumber, inNumberFrames, &abList);
-    if (res != 0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Error getting rec data, error = %d", res);
+  // A VP I/O unit's bus 1 connects to input hardware (microphone). Enable
+  // input on the input scope of the input element.
+  AudioUnitElement input_bus = 1;
+  UInt32 enable_input = 1;
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Input, input_bus, &enable_input,
+                                sizeof(enable_input));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to enable input on input scope of input element: "
+                  << result;
+    return false;
+  }
 
-        if (_recWarning > 0) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  Pending rec warning exists");
-        }
-        _recWarning = 1;
+  // A VP I/O unit's bus 0 connects to output hardware (speaker). Enable
+  // output on the output scope of the output element.
+  AudioUnitElement output_bus = 0;
+  UInt32 enable_output = 1;
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Output, output_bus,
+                                &enable_output, sizeof(enable_output));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR)
+        << "Failed to enable output on output scope of output element: "
+        << result;
+    return false;
+  }
 
-        delete [] dataTmp;
-        return 0;
+  // Set the application formats for input and output:
+  // - use same format in both directions
+  // - avoid resampling in the I/O unit by using the hardware sample rate
+  // - linear PCM => noncompressed audio data format with one frame per packet
+  // - no need to specify interleaving since only mono is supported
+  AudioStreamBasicDescription application_format = {0};
+  UInt32 size = sizeof(application_format);
+  RTC_DCHECK_EQ(playout_parameters_.sample_rate(),
+                record_parameters_.sample_rate());
+  RTC_DCHECK_EQ(1, kPreferredNumberOfChannels);
+  application_format.mSampleRate = playout_parameters_.sample_rate();
+  application_format.mFormatID = kAudioFormatLinearPCM;
+  application_format.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
+  application_format.mBytesPerPacket = kBytesPerSample;
+  application_format.mFramesPerPacket = 1;  // uncompressed
+  application_format.mBytesPerFrame = kBytesPerSample;
+  application_format.mChannelsPerFrame = kPreferredNumberOfChannels;
+  application_format.mBitsPerChannel = 8 * kBytesPerSample;
+  // Store the new format.
+  application_format_ = application_format;
+#if !defined(NDEBUG)
+  LogABSD(application_format_);
+#endif
+
+  // Set the application format on the output scope of the input element/bus.
+  result = AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                                kAudioUnitScope_Output, input_bus,
+                                &application_format, size);
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR)
+        << "Failed to set application format on output scope of input bus: "
+        << result;
+    return false;
+  }
+
+  // Set the application format on the input scope of the output element/bus.
+  result = AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                                kAudioUnitScope_Input, output_bus,
+                                &application_format, size);
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR)
+        << "Failed to set application format on input scope of output bus: "
+        << result;
+    return false;
+  }
+
+  // Specify the callback function that provides audio samples to the audio
+  // unit.
+  AURenderCallbackStruct render_callback;
+  render_callback.inputProc = GetPlayoutData;
+  render_callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
+      output_bus, &render_callback, sizeof(render_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to specify the render callback on the output bus: "
+                  << result;
+    return false;
+  }
+
+  // Disable AU buffer allocation for the recorder, we allocate our own.
+  // TODO(henrika): not sure that it actually saves resource to make this call.
+  UInt32 flag = 0;
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
+      kAudioUnitScope_Output, input_bus, &flag, sizeof(flag));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to disable buffer allocation on the input bus: "
+                  << result;
+  }
+
+  // Specify the callback to be called by the I/O thread to us when input audio
+  // is available. The recorded samples can then be obtained by calling the
+  // AudioUnitRender() method.
+  AURenderCallbackStruct input_callback;
+  input_callback.inputProc = RecordedDataIsAvailable;
+  input_callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(vpio_unit_,
+                                kAudioOutputUnitProperty_SetInputCallback,
+                                kAudioUnitScope_Global, input_bus,
+                                &input_callback, sizeof(input_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    LOG(LS_ERROR) << "Failed to specify the input callback on the input bus: "
+                  << result;
+  }
+
+  // Initialize the Voice-Processing I/O unit instance.
+  // Calls to AudioUnitInitialize() can fail if called back-to-back on
+  // different ADM instances. The error message in this case is -66635 which is
+  // undocumented. Tests have shown that calling AudioUnitInitialize a second
+  // time, after a short sleep, avoids this issue.
+  // See webrtc:5166 for details.
+  int failed_initalize_attempts = 0;
+  result = AudioUnitInitialize(vpio_unit_);
+  while (result != noErr) {
+    LOG(LS_ERROR) << "Failed to initialize the Voice-Processing I/O unit: "
+                  << result;
+    ++failed_initalize_attempts;
+    if (failed_initalize_attempts == kMaxNumberOfAudioUnitInitializeAttempts) {
+      // Max number of initialization attempts exceeded, hence abort.
+      LOG(LS_WARNING) << "Too many initialization attempts";
+      DisposeAudioUnit();
+      return false;
     }
-
-    if (_recording) {
-        // Insert all data in temp buffer into recording buffers
-        // There is zero or one buffer partially full at any given time,
-        // all others are full or empty
-        // Full means filled with noSamp10ms samples.
-
-        const unsigned int noSamp10ms = _adbSampFreq / 100;
-        unsigned int dataPos = 0;
-        uint16_t bufPos = 0;
-        int16_t insertPos = -1;
-        unsigned int nCopy = 0;  // Number of samples to copy
-
-        while (dataPos < inNumberFrames) {
-            // Loop over all recording buffers or
-            // until we find the partially full buffer
-            // First choice is to insert into partially full buffer,
-            // second choice is to insert into empty buffer
-            bufPos = 0;
-            insertPos = -1;
-            nCopy = 0;
-            while (bufPos < N_REC_BUFFERS) {
-                if ((_recordingLength[bufPos] > 0)
-                    && (_recordingLength[bufPos] < noSamp10ms)) {
-                    // Found the partially full buffer
-                    insertPos = static_cast<int16_t>(bufPos);
-                    // Don't need to search more, quit loop
-                    bufPos = N_REC_BUFFERS;
-                } else if ((-1 == insertPos)
-                           && (0 == _recordingLength[bufPos])) {
-                    // Found an empty buffer
-                    insertPos = static_cast<int16_t>(bufPos);
-                }
-                ++bufPos;
-            }
-
-            // Insert data into buffer
-            if (insertPos > -1) {
-                // We found a non-full buffer, copy data to it
-                unsigned int dataToCopy = inNumberFrames - dataPos;
-                unsigned int currentRecLen = _recordingLength[insertPos];
-                unsigned int roomInBuffer = noSamp10ms - currentRecLen;
-                nCopy = (dataToCopy < roomInBuffer ? dataToCopy : roomInBuffer);
-
-                memcpy(&_recordingBuffer[insertPos][currentRecLen],
-                       &dataTmp[dataPos], nCopy*sizeof(int16_t));
-                if (0 == currentRecLen) {
-                    _recordingSeqNumber[insertPos] = _recordingCurrentSeq;
-                    ++_recordingCurrentSeq;
-                }
-                _recordingBufferTotalSize += nCopy;
-                // Has to be done last to avoid interrupt problems
-                // between threads
-                _recordingLength[insertPos] += nCopy;
-                dataPos += nCopy;
-            } else {
-                // Didn't find a non-full buffer
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             "  Could not insert into recording buffer");
-                if (_recWarning > 0) {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  Pending rec warning exists");
-                }
-                _recWarning = 1;
-                dataPos = inNumberFrames;  // Don't try to insert more
-            }
-        }
-    }
-
-    delete [] dataTmp;
-
-    return 0;
+    LOG(LS_INFO) << "pause 100ms and try audio unit initialization again...";
+    [NSThread sleepForTimeInterval:0.1f];
+    result = AudioUnitInitialize(vpio_unit_);
+  }
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now initialized";
+  return true;
 }
 
-OSStatus
-    AudioDeviceIOS::PlayoutProcess(void *inRefCon,
-                                   AudioUnitRenderActionFlags *ioActionFlags,
-                                   const AudioTimeStamp *inTimeStamp,
-                                   UInt32 inBusNumber,
-                                   UInt32 inNumberFrames,
-                                   AudioBufferList *ioData) {
-    AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
+bool AudioDeviceIOS::RestartAudioUnitWithNewFormat(float sample_rate) {
+  LOGI() << "RestartAudioUnitWithNewFormat(sample_rate=" << sample_rate << ")";
+  // Stop the active audio unit.
+  LOG_AND_RETURN_IF_ERROR(AudioOutputUnitStop(vpio_unit_),
+                          "Failed to stop the the Voice-Processing I/O unit");
 
-    return ptrThis->PlayoutProcessImpl(inNumberFrames, ioData);
+  // The stream format is about to be changed and it requires that we first
+  // uninitialize it to deallocate its resources.
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitUninitialize(vpio_unit_),
+      "Failed to uninitialize the the Voice-Processing I/O unit");
+
+  // Allocate new buffers given the new stream format.
+  SetupAudioBuffersForActiveAudioSession();
+
+  // Update the existing application format using the new sample rate.
+  application_format_.mSampleRate = playout_parameters_.sample_rate();
+  UInt32 size = sizeof(application_format_);
+  AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                       kAudioUnitScope_Output, 1, &application_format_, size);
+  AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                       kAudioUnitScope_Input, 0, &application_format_, size);
+
+  // Prepare the audio unit to render audio again.
+  LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(vpio_unit_),
+                          "Failed to initialize the Voice-Processing I/O unit");
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now reinitialized";
+
+  // Start rendering audio using the new format.
+  LOG_AND_RETURN_IF_ERROR(AudioOutputUnitStart(vpio_unit_),
+                          "Failed to start the Voice-Processing I/O unit");
+  LOG(LS_INFO) << "Voice-Processing I/O unit is now restarted";
+  return true;
 }
 
-OSStatus
-    AudioDeviceIOS::PlayoutProcessImpl(uint32_t inNumberFrames,
-                                       AudioBufferList *ioData) {
-    // Setup some basic stuff
-//    assert(sizeof(short) == 2); // Assumption for implementation
+bool AudioDeviceIOS::InitPlayOrRecord() {
+  LOGI() << "InitPlayOrRecord";
+  // Activate the audio session if not already activated.
+  if (!ActivateAudioSession()) {
+    return false;
+  }
 
-    int16_t* data =
-        static_cast<int16_t*>(ioData->mBuffers[0].mData);
-    unsigned int dataSizeBytes = ioData->mBuffers[0].mDataByteSize;
-    unsigned int dataSize = dataSizeBytes/2;  // Number of samples
-        if (dataSize != inNumberFrames) {  // Should always be the same
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "dataSize (%u) != inNumberFrames (%u)",
-                     dataSize, (unsigned int)inNumberFrames);
-        if (_playWarning > 0) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  Pending play warning exists");
-        }
-        _playWarning = 1;
-    }
-    memset(data, 0, dataSizeBytes);  // Start with empty buffer
+  // Ensure that the active audio session has the correct category and mode.
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  if (!VerifyAudioSession(session)) {
+    DeactivateAudioSession();
+    LOG(LS_ERROR) << "Failed to verify audio session category and mode";
+    return false;
+  }
 
+  // Start observing audio session interruptions and route changes.
+  RegisterNotificationObservers();
 
-    // Get playout data from Audio Device Buffer
+  // Ensure that we got what what we asked for in our active audio session.
+  SetupAudioBuffersForActiveAudioSession();
 
-    if (_playing) {
-        unsigned int noSamp10ms = _adbSampFreq / 100;
-        // todo: Member variable and allocate when samp freq is determined
-        int16_t* dataTmp = new int16_t[noSamp10ms];
-        memset(dataTmp, 0, 2*noSamp10ms);
-        unsigned int dataPos = 0;
-        int noSamplesOut = 0;
-        unsigned int nCopy = 0;
-
-        // First insert data from playout buffer if any
-        if (_playoutBufferUsed > 0) {
-            nCopy = (dataSize < _playoutBufferUsed) ?
-                    dataSize : _playoutBufferUsed;
-            if (nCopy != _playoutBufferUsed) {
-                // todo: If dataSize < _playoutBufferUsed
-                //       (should normally never be)
-                //       we must move the remaining data
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             "nCopy (%u) != _playoutBufferUsed (%u)",
-                             nCopy, _playoutBufferUsed);
-                if (_playWarning > 0) {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  Pending play warning exists");
-                }
-                _playWarning = 1;
-            }
-            memcpy(data, _playoutBuffer, 2*nCopy);
-            dataPos = nCopy;
-            memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-            _playoutBufferUsed = 0;
-        }
-
-        // Now get the rest from Audio Device Buffer
-        while (dataPos < dataSize) {
-            // Update playout delay
-            UpdatePlayoutDelay();
-
-            // Ask for new PCM data to be played out using the AudioDeviceBuffer
-            noSamplesOut = _ptrAudioBuffer->RequestPlayoutData(noSamp10ms);
-
-            // Get data from Audio Device Buffer
-            noSamplesOut =
-                _ptrAudioBuffer->GetPlayoutData(
-                    reinterpret_cast<int8_t*>(dataTmp));
-            // Cast OK since only equality comparison
-            if (noSamp10ms != (unsigned int)noSamplesOut) {
-                // Should never happen
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             "noSamp10ms (%u) != noSamplesOut (%d)",
-                             noSamp10ms, noSamplesOut);
-
-                if (_playWarning > 0) {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  Pending play warning exists");
-                }
-                _playWarning = 1;
-            }
-
-            // Insert as much as fits in data buffer
-            nCopy = (dataSize-dataPos) > noSamp10ms ?
-                    noSamp10ms : (dataSize-dataPos);
-            memcpy(&data[dataPos], dataTmp, 2*nCopy);
-
-            // Save rest in playout buffer if any
-            if (nCopy < noSamp10ms) {
-                memcpy(_playoutBuffer, &dataTmp[nCopy], 2*(noSamp10ms-nCopy));
-                _playoutBufferUsed = noSamp10ms - nCopy;
-            }
-
-            // Update loop/index counter, if we copied less than noSamp10ms
-            // samples we shall quit loop anyway
-            dataPos += noSamp10ms;
-        }
-
-        delete [] dataTmp;
-    }
-
-    return 0;
+  // Create, setup and initialize a new Voice-Processing I/O unit.
+  if (!SetupAndInitializeVoiceProcessingAudioUnit()) {
+    // Reduce usage count for the audio session and possibly deactivate it if
+    // this object is the only user.
+    DeactivateAudioSession();
+    return false;
+  }
+  return true;
 }
 
-void AudioDeviceIOS::UpdatePlayoutDelay() {
-    ++_playoutDelayMeasurementCounter;
-
-    if (_playoutDelayMeasurementCounter >= 100) {
-        // Update HW and OS delay every second, unlikely to change
-
-        // Since this is eventually rounded to integral ms, add 0.5ms
-        // here to get round-to-nearest-int behavior instead of
-        // truncation.
-        double totalDelaySeconds = 0.0005;
-
-        // HW output latency
-        AVAudioSession* session = [AVAudioSession sharedInstance];
-        double latency = session.outputLatency;
-        assert(latency >= 0);
-        totalDelaySeconds += latency;
-
-        // HW buffer duration
-        double ioBufferDuration = session.IOBufferDuration;
-        assert(ioBufferDuration >= 0);
-        totalDelaySeconds += ioBufferDuration;
-
-        // AU latency
-        Float64 f64(0);
-        UInt32 size = sizeof(f64);
-        OSStatus result = AudioUnitGetProperty(
-            _auVoiceProcessing, kAudioUnitProperty_Latency,
-            kAudioUnitScope_Global, 0, &f64, &size);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "error AU latency (result=%d)", result);
-        }
-        assert(f64 >= 0);
-        totalDelaySeconds += f64;
-
-        // To ms
-        _playoutDelay = static_cast<uint32_t>(totalDelaySeconds / 1000);
-
-        // Reset counter
-        _playoutDelayMeasurementCounter = 0;
+void AudioDeviceIOS::ShutdownPlayOrRecord() {
+  LOGI() << "ShutdownPlayOrRecord";
+  // Close and delete the voice-processing I/O unit.
+  OSStatus result = -1;
+  if (nullptr != vpio_unit_) {
+    result = AudioOutputUnitStop(vpio_unit_);
+    if (result != noErr) {
+      LOG_F(LS_ERROR) << "AudioOutputUnitStop failed: " << result;
     }
+    result = AudioUnitUninitialize(vpio_unit_);
+    if (result != noErr) {
+      LOG_F(LS_ERROR) << "AudioUnitUninitialize failed: " << result;
+    }
+    DisposeAudioUnit();
+  }
 
-    // todo: Add playout buffer?
+  // Remove audio session notification observers.
+  UnregisterNotificationObservers();
+
+  // All I/O should be stopped or paused prior to deactivating the audio
+  // session, hence we deactivate as last action.
+  DeactivateAudioSession();
 }
 
-void AudioDeviceIOS::UpdateRecordingDelay() {
-    ++_recordingDelayMeasurementCounter;
-
-    if (_recordingDelayMeasurementCounter >= 100) {
-        // Update HW and OS delay every second, unlikely to change
-
-        // Since this is eventually rounded to integral ms, add 0.5ms
-        // here to get round-to-nearest-int behavior instead of
-        // truncation.
-        double totalDelaySeconds = 0.0005;
-
-        // HW input latency
-        AVAudioSession* session = [AVAudioSession sharedInstance];
-        double latency = session.inputLatency;
-        assert(latency >= 0);
-        totalDelaySeconds += latency;
-
-        // HW buffer duration
-        double ioBufferDuration = session.IOBufferDuration;
-        assert(ioBufferDuration >= 0);
-        totalDelaySeconds += ioBufferDuration;
-
-        // AU latency
-        Float64 f64(0);
-        UInt32 size = sizeof(f64);
-        OSStatus result = AudioUnitGetProperty(
-             _auVoiceProcessing, kAudioUnitProperty_Latency,
-             kAudioUnitScope_Global, 0, &f64, &size);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "error AU latency (result=%d)", result);
-        }
-        assert(f64 >= 0);
-        totalDelaySeconds += f64;
-
-        // To ms
-        _recordingDelayHWAndOS =
-            static_cast<uint32_t>(totalDelaySeconds / 1000);
-
-        // Reset counter
-        _recordingDelayMeasurementCounter = 0;
-    }
-
-    _recordingDelay = _recordingDelayHWAndOS;
-
-    // ADB recording buffer size, update every time
-    // Don't count the one next 10 ms to be sent, then convert samples => ms
-    const uint32_t noSamp10ms = _adbSampFreq / 100;
-    if (_recordingBufferTotalSize > noSamp10ms) {
-        _recordingDelay +=
-            (_recordingBufferTotalSize - noSamp10ms) / (_adbSampFreq / 1000);
-    }
+void AudioDeviceIOS::DisposeAudioUnit() {
+  if (nullptr == vpio_unit_)
+    return;
+  OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
+  if (result != noErr) {
+    LOG(LS_ERROR) << "AudioComponentInstanceDispose failed:" << result;
+  }
+  vpio_unit_ = nullptr;
 }
 
-bool AudioDeviceIOS::RunCapture(void* ptrThis) {
-    return static_cast<AudioDeviceIOS*>(ptrThis)->CaptureWorkerThread();
+OSStatus AudioDeviceIOS::RecordedDataIsAvailable(
+    void* in_ref_con,
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* in_time_stamp,
+    UInt32 in_bus_number,
+    UInt32 in_number_frames,
+    AudioBufferList* io_data) {
+  RTC_DCHECK_EQ(1u, in_bus_number);
+  RTC_DCHECK(
+      !io_data);  // no buffer should be allocated for input at this stage
+  AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(in_ref_con);
+  return audio_device_ios->OnRecordedDataIsAvailable(
+      io_action_flags, in_time_stamp, in_bus_number, in_number_frames);
 }
 
-bool AudioDeviceIOS::CaptureWorkerThread() {
-    if (_recording) {
-        int bufPos = 0;
-        unsigned int lowestSeq = 0;
-        int lowestSeqBufPos = 0;
-        bool foundBuf = true;
-        const unsigned int noSamp10ms = _adbSampFreq / 100;
+OSStatus AudioDeviceIOS::OnRecordedDataIsAvailable(
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* in_time_stamp,
+    UInt32 in_bus_number,
+    UInt32 in_number_frames) {
+  OSStatus result = noErr;
+  // Simply return if recording is not enabled.
+  if (!rtc::AtomicOps::AcquireLoad(&recording_))
+    return result;
+  if (in_number_frames != record_parameters_.frames_per_buffer()) {
+    // We have seen short bursts (1-2 frames) where |in_number_frames| changes.
+    // Add a log to keep track of longer sequences if that should ever happen.
+    // Also return since calling AudioUnitRender in this state will only result
+    // in kAudio_ParamError (-50) anyhow.
+    LOG(LS_WARNING) << "in_number_frames (" << in_number_frames
+                    << ") != " << record_parameters_.frames_per_buffer();
+    return noErr;
+  }
+  // Obtain the recorded audio samples by initiating a rendering cycle.
+  // Since it happens on the input bus, the |io_data| parameter is a reference
+  // to the preallocated audio buffer list that the audio unit renders into.
+  // TODO(henrika): should error handling be improved?
+  AudioBufferList* io_data = &audio_record_buffer_list_;
+  result = AudioUnitRender(vpio_unit_, io_action_flags, in_time_stamp,
+                           in_bus_number, in_number_frames, io_data);
+  if (result != noErr) {
+    LOG_F(LS_ERROR) << "AudioUnitRender failed: " << result;
+    return result;
+  }
+  // Get a pointer to the recorded audio and send it to the WebRTC ADB.
+  // Use the FineAudioBuffer instance to convert between native buffer size
+  // and the 10ms buffer size used by WebRTC.
+  const UInt32 data_size_in_bytes = io_data->mBuffers[0].mDataByteSize;
+  RTC_CHECK_EQ(data_size_in_bytes / kBytesPerSample, in_number_frames);
+  SInt8* data = static_cast<SInt8*>(io_data->mBuffers[0].mData);
+  fine_audio_buffer_->DeliverRecordedData(data, data_size_in_bytes,
+                                          kFixedPlayoutDelayEstimate,
+                                          kFixedRecordDelayEstimate);
+  return noErr;
+}
 
-        while (foundBuf) {
-            // Check if we have any buffer with data to insert
-            // into the Audio Device Buffer,
-            // and find the one with the lowest seq number
-            foundBuf = false;
-            for (bufPos = 0; bufPos < N_REC_BUFFERS; ++bufPos) {
-                if (noSamp10ms == _recordingLength[bufPos]) {
-                    if (!foundBuf) {
-                        lowestSeq = _recordingSeqNumber[bufPos];
-                        lowestSeqBufPos = bufPos;
-                        foundBuf = true;
-                    } else if (_recordingSeqNumber[bufPos] < lowestSeq) {
-                        lowestSeq = _recordingSeqNumber[bufPos];
-                        lowestSeqBufPos = bufPos;
-                    }
-                }
-            }  // for
+OSStatus AudioDeviceIOS::GetPlayoutData(
+    void* in_ref_con,
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* in_time_stamp,
+    UInt32 in_bus_number,
+    UInt32 in_number_frames,
+    AudioBufferList* io_data) {
+  RTC_DCHECK_EQ(0u, in_bus_number);
+  RTC_DCHECK(io_data);
+  AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(in_ref_con);
+  return audio_device_ios->OnGetPlayoutData(io_action_flags, in_number_frames,
+                                            io_data);
+}
 
-            // Insert data into the Audio Device Buffer if found any
-            if (foundBuf) {
-                // Update recording delay
-                UpdateRecordingDelay();
-
-                // Set the recorded buffer
-                _ptrAudioBuffer->SetRecordedBuffer(
-                    reinterpret_cast<int8_t*>(
-                        _recordingBuffer[lowestSeqBufPos]),
-                        _recordingLength[lowestSeqBufPos]);
-
-                // Don't need to set the current mic level in ADB since we only
-                // support digital AGC,
-                // and besides we cannot get or set the IOS mic level anyway.
-
-                // Set VQE info, use clockdrift == 0
-                _ptrAudioBuffer->SetVQEData(_playoutDelay, _recordingDelay, 0);
-
-                // Deliver recorded samples at specified sample rate, mic level
-                // etc. to the observer using callback
-                _ptrAudioBuffer->DeliverRecordedData();
-
-                // Make buffer available
-                _recordingSeqNumber[lowestSeqBufPos] = 0;
-                _recordingBufferTotalSize -= _recordingLength[lowestSeqBufPos];
-                // Must be done last to avoid interrupt problems between threads
-                _recordingLength[lowestSeqBufPos] = 0;
-            }
-        }  // while (foundBuf)
-    }  // if (_recording)
-
-    {
-        // Normal case
-        // Sleep thread (5ms) to let other threads get to work
-        // todo: Is 5 ms optimal? Sleep shorter if inserted into the Audio
-        //       Device Buffer?
-        timespec t;
-        t.tv_sec = 0;
-        t.tv_nsec = 5*1000*1000;
-        nanosleep(&t, NULL);
-    }
-
-    return true;
+OSStatus AudioDeviceIOS::OnGetPlayoutData(
+    AudioUnitRenderActionFlags* io_action_flags,
+    UInt32 in_number_frames,
+    AudioBufferList* io_data) {
+  // Verify 16-bit, noninterleaved mono PCM signal format.
+  RTC_DCHECK_EQ(1u, io_data->mNumberBuffers);
+  RTC_DCHECK_EQ(1u, io_data->mBuffers[0].mNumberChannels);
+  // Get pointer to internal audio buffer to which new audio data shall be
+  // written.
+  const UInt32 dataSizeInBytes = io_data->mBuffers[0].mDataByteSize;
+  RTC_CHECK_EQ(dataSizeInBytes / kBytesPerSample, in_number_frames);
+  SInt8* destination = static_cast<SInt8*>(io_data->mBuffers[0].mData);
+  // Produce silence and give audio unit a hint about it if playout is not
+  // activated.
+  if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
+    *io_action_flags |= kAudioUnitRenderAction_OutputIsSilence;
+    memset(destination, 0, dataSizeInBytes);
+    return noErr;
+  }
+  // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
+  // the native I/O audio unit) to a preallocated intermediate buffer and
+  // copy the result to the audio buffer in the |io_data| destination.
+  SInt8* source = playout_audio_buffer_.get();
+  fine_audio_buffer_->GetPlayoutData(source);
+  memcpy(destination, source, dataSizeInBytes);
+  return noErr;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_not_implemented_ios.mm b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_not_implemented_ios.mm
new file mode 100644
index 0000000000..acfc30d7f3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_not_implemented_ios.mm
@@ -0,0 +1,292 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_device/ios/audio_device_ios.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+int32_t AudioDeviceIOS::PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                      uint16_t& sizeMS) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::ResetAudioDevice() {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int16_t AudioDeviceIOS::PlayoutDevices() {
+  // TODO(henrika): improve.
+  LOG_F(LS_WARNING) << "Not implemented";
+  return (int16_t)1;
+}
+
+int16_t AudioDeviceIOS::RecordingDevices() {
+  // TODO(henrika): improve.
+  LOG_F(LS_WARNING) << "Not implemented";
+  return (int16_t)1;
+}
+
+int32_t AudioDeviceIOS::InitSpeaker() {
+  return 0;
+}
+
+bool AudioDeviceIOS::SpeakerIsInitialized() const {
+  return true;
+}
+
+int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SetWaveOutVolume(uint16_t, uint16_t) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::WaveOutVolume(uint16_t&, uint16_t&) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MinSpeakerVolume(uint32_t& minVolume) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
+  LOG_F(LS_WARNING) << "Not implemented";
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+bool AudioDeviceIOS::PlayoutWarning() const {
+  return false;
+}
+
+bool AudioDeviceIOS::PlayoutError() const {
+  return false;
+}
+
+bool AudioDeviceIOS::RecordingWarning() const {
+  return false;
+}
+
+bool AudioDeviceIOS::RecordingError() const {
+  return false;
+}
+
+int32_t AudioDeviceIOS::InitMicrophone() {
+  return 0;
+}
+
+bool AudioDeviceIOS::MicrophoneIsInitialized() const {
+  return true;
+}
+
+int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneBoostIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetMicrophoneBoost(bool enable) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneBoost(bool& enabled) const {
+  enabled = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetStereoRecording(bool enable) {
+  LOG_F(LS_WARNING) << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const {
+  enabled = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
+  LOG_F(LS_WARNING) << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
+  enabled = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetAGC(bool enable) {
+  if (enable) {
+    RTC_NOTREACHED() << "Should never be called";
+  }
+  return -1;
+}
+
+bool AudioDeviceIOS::AGC() const {
+  return false;
+}
+
+int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneVolumeStepSize(uint16_t& stepSize) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::PlayoutDeviceName(uint16_t index,
+                                          char name[kAdmMaxDeviceNameSize],
+                                          char guid[kAdmMaxGuidSize]) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::RecordingDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
+  LOG_F(LS_WARNING) << "Not implemented";
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetPlayoutBuffer(
+    const AudioDeviceModule::BufferType type,
+    uint16_t sizeMS) {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::CPULoad(uint16_t&) const {
+  RTC_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
new file mode 100644
index 0000000000..076a67430d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
@@ -0,0 +1,847 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <limits>
+#include <list>
+#include <numeric>
+#include <string>
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/audio_device/audio_device_impl.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/audio_device/ios/audio_device_ios.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using std::cout;
+using std::endl;
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Gt;
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::NotNull;
+using ::testing::Return;
+
+// #define ENABLE_DEBUG_PRINTF
+#ifdef ENABLE_DEBUG_PRINTF
+#define PRINTD(...) fprintf(stderr, __VA_ARGS__);
+#else
+#define PRINTD(...) ((void)0)
+#endif
+#define PRINT(...) fprintf(stderr, __VA_ARGS__);
+
+namespace webrtc {
+
+// Number of callbacks (input or output) the tests waits for before we set
+// an event indicating that the test was OK.
+static const size_t kNumCallbacks = 10;
+// Max amount of time we wait for an event to be set while counting callbacks.
+static const int kTestTimeOutInMilliseconds = 10 * 1000;
+// Number of bits per PCM audio sample.
+static const size_t kBitsPerSample = 16;
+// Number of bytes per PCM audio sample.
+static const size_t kBytesPerSample = kBitsPerSample / 8;
+// Average number of audio callbacks per second assuming 10ms packet size.
+static const size_t kNumCallbacksPerSecond = 100;
+// Play out a test file during this time (unit is in seconds).
+static const int kFilePlayTimeInSec = 15;
+// Run the full-duplex test during this time (unit is in seconds).
+// Note that first |kNumIgnoreFirstCallbacks| are ignored.
+static const int kFullDuplexTimeInSec = 10;
+// Wait for the callback sequence to stabilize by ignoring this amount of the
+// initial callbacks (avoids initial FIFO access).
+// Only used in the RunPlayoutAndRecordingInFullDuplex test.
+static const size_t kNumIgnoreFirstCallbacks = 50;
+// Sets the number of impulses per second in the latency test.
+// TODO(henrika): fine tune this setting for iOS.
+static const int kImpulseFrequencyInHz = 1;
+// Length of round-trip latency measurements. Number of transmitted impulses
+// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
+// TODO(henrika): fine tune this setting for iOS.
+static const int kMeasureLatencyTimeInSec = 5;
+// Utilized in round-trip latency measurements to avoid capturing noise samples.
+// TODO(henrika): fine tune this setting for iOS.
+static const int kImpulseThreshold = 50;
+static const char kTag[] = "[..........] ";
+
+enum TransportType {
+  kPlayout = 0x1,
+  kRecording = 0x2,
+};
+
+// Interface for processing the audio stream. Real implementations can e.g.
+// run audio in loopback, read audio from a file or perform latency
+// measurements.
+class AudioStreamInterface {
+ public:
+  virtual void Write(const void* source, size_t num_frames) = 0;
+  virtual void Read(void* destination, size_t num_frames) = 0;
+
+ protected:
+  virtual ~AudioStreamInterface() {}
+};
+
+// Reads audio samples from a PCM file where the file is stored in memory at
+// construction.
+class FileAudioStream : public AudioStreamInterface {
+ public:
+  FileAudioStream(size_t num_callbacks,
+                  const std::string& file_name,
+                  int sample_rate)
+      : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) {
+    file_size_in_bytes_ = test::GetFileSize(file_name);
+    sample_rate_ = sample_rate;
+    EXPECT_GE(file_size_in_callbacks(), num_callbacks)
+        << "Size of test file is not large enough to last during the test.";
+    const size_t num_16bit_samples =
+        test::GetFileSize(file_name) / kBytesPerSample;
+    file_.reset(new int16_t[num_16bit_samples]);
+    FILE* audio_file = fopen(file_name.c_str(), "rb");
+    EXPECT_NE(audio_file, nullptr);
+    size_t num_samples_read =
+        fread(file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
+    EXPECT_EQ(num_samples_read, num_16bit_samples);
+    fclose(audio_file);
+  }
+
+  // AudioStreamInterface::Write() is not implemented.
+  void Write(const void* source, size_t num_frames) override {}
+
+  // Read samples from file stored in memory (at construction) and copy
+  // |num_frames| (<=> 10ms) to the |destination| byte buffer.
+  void Read(void* destination, size_t num_frames) override {
+    memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
+           num_frames * sizeof(int16_t));
+    file_pos_ += num_frames;
+  }
+
+  int file_size_in_seconds() const {
+    return static_cast<int>(
+        file_size_in_bytes_ / (kBytesPerSample * sample_rate_));
+  }
+  size_t file_size_in_callbacks() const {
+    return file_size_in_seconds() * kNumCallbacksPerSecond;
+  }
+
+ private:
+  size_t file_size_in_bytes_;
+  int sample_rate_;
+  rtc::scoped_ptr<int16_t[]> file_;
+  size_t file_pos_;
+};
+
+// Simple first in first out (FIFO) class that wraps a list of 16-bit audio
+// buffers of fixed size and allows Write and Read operations. The idea is to
+// store recorded audio buffers (using Write) and then read (using Read) these
+// stored buffers with as short delay as possible when the audio layer needs
+// data to play out. The number of buffers in the FIFO will stabilize under
+// normal conditions since there will be a balance between Write and Read calls.
+// The container is a std::list container and access is protected with a lock
+// since both sides (playout and recording) are driven by its own thread.
+class FifoAudioStream : public AudioStreamInterface {
+ public:
+  explicit FifoAudioStream(size_t frames_per_buffer)
+      : frames_per_buffer_(frames_per_buffer),
+        bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
+        fifo_(new AudioBufferList),
+        largest_size_(0),
+        total_written_elements_(0),
+        write_count_(0) {
+    EXPECT_NE(fifo_.get(), nullptr);
+  }
+
+  ~FifoAudioStream() { Flush(); }
+
+  // Allocate new memory, copy |num_frames| samples from |source| into memory
+  // and add pointer to the memory location to end of the list.
+  // Increases the size of the FIFO by one element.
+  void Write(const void* source, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    PRINTD("+");
+    if (write_count_++ < kNumIgnoreFirstCallbacks) {
+      return;
+    }
+    int16_t* memory = new int16_t[frames_per_buffer_];
+    memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_);
+    rtc::CritScope lock(&lock_);
+    fifo_->push_back(memory);
+    const size_t size = fifo_->size();
+    if (size > largest_size_) {
+      largest_size_ = size;
+      PRINTD("(%" PRIuS ")", largest_size_);
+    }
+    total_written_elements_ += size;
+  }
+
+  // Read pointer to data buffer from front of list, copy |num_frames| of stored
+  // data into |destination| and delete the utilized memory allocation.
+  // Decreases the size of the FIFO by one element.
+  void Read(void* destination, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    PRINTD("-");
+    rtc::CritScope lock(&lock_);
+    if (fifo_->empty()) {
+      memset(destination, 0, bytes_per_buffer_);
+    } else {
+      int16_t* memory = fifo_->front();
+      fifo_->pop_front();
+      memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
+      delete memory;
+    }
+  }
+
+  size_t size() const { return fifo_->size(); }
+
+  size_t largest_size() const { return largest_size_; }
+
+  size_t average_size() const {
+    return (total_written_elements_ == 0)
+               ? 0.0
+               : 0.5 +
+                     static_cast<float>(total_written_elements_) /
+                         (write_count_ - kNumIgnoreFirstCallbacks);
+  }
+
+ private:
+  void Flush() {
+    for (auto it = fifo_->begin(); it != fifo_->end(); ++it) {
+      delete *it;
+    }
+    fifo_->clear();
+  }
+
+  using AudioBufferList = std::list<int16_t*>;
+  rtc::CriticalSection lock_;
+  const size_t frames_per_buffer_;
+  const size_t bytes_per_buffer_;
+  rtc::scoped_ptr<AudioBufferList> fifo_;
+  size_t largest_size_;
+  size_t total_written_elements_;
+  size_t write_count_;
+};
+
+// Inserts periodic impulses and measures the latency between the time of
+// transmission and time of receiving the same impulse.
+// Usage requires a special hardware called Audio Loopback Dongle.
+// See http://source.android.com/devices/audio/loopback.html for details.
+class LatencyMeasuringAudioStream : public AudioStreamInterface {
+ public:
+  explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
+      : clock_(Clock::GetRealTimeClock()),
+        frames_per_buffer_(frames_per_buffer),
+        bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
+        play_count_(0),
+        rec_count_(0),
+        pulse_time_(0) {}
+
+  // Insert periodic impulses in first two samples of |destination|.
+  void Read(void* destination, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    if (play_count_ == 0) {
+      PRINT("[");
+    }
+    play_count_++;
+    memset(destination, 0, bytes_per_buffer_);
+    if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
+      if (pulse_time_ == 0) {
+        pulse_time_ = clock_->TimeInMilliseconds();
+      }
+      PRINT(".");
+      const int16_t impulse = std::numeric_limits<int16_t>::max();
+      int16_t* ptr16 = static_cast<int16_t*>(destination);
+      for (size_t i = 0; i < 2; ++i) {
+        ptr16[i] = impulse;
+      }
+    }
+  }
+
+  // Detect received impulses in |source|, derive time between transmission and
+  // detection and add the calculated delay to list of latencies.
+  void Write(const void* source, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    rec_count_++;
+    if (pulse_time_ == 0) {
+      // Avoid detection of new impulse response until a new impulse has
+      // been transmitted (sets |pulse_time_| to value larger than zero).
+      return;
+    }
+    const int16_t* ptr16 = static_cast<const int16_t*>(source);
+    std::vector<int16_t> vec(ptr16, ptr16 + num_frames);
+    // Find max value in the audio buffer.
+    int max = *std::max_element(vec.begin(), vec.end());
+    // Find index (element position in vector) of the max element.
+    int index_of_max =
+        std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
+    if (max > kImpulseThreshold) {
+      PRINTD("(%d,%d)", max, index_of_max);
+      int64_t now_time = clock_->TimeInMilliseconds();
+      int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
+      PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
+      PRINTD("[%d]", extra_delay);
+      // Total latency is the difference between transmit time and detection
+      // tome plus the extra delay within the buffer in which we detected the
+      // received impulse. It is transmitted at sample 0 but can be received
+      // at sample N where N > 0. The term |extra_delay| accounts for N and it
+      // is a value between 0 and 10ms.
+      latencies_.push_back(now_time - pulse_time_ + extra_delay);
+      pulse_time_ = 0;
+    } else {
+      PRINTD("-");
+    }
+  }
+
+  size_t num_latency_values() const { return latencies_.size(); }
+
+  int min_latency() const {
+    if (latencies_.empty())
+      return 0;
+    return *std::min_element(latencies_.begin(), latencies_.end());
+  }
+
+  int max_latency() const {
+    if (latencies_.empty())
+      return 0;
+    return *std::max_element(latencies_.begin(), latencies_.end());
+  }
+
+  int average_latency() const {
+    if (latencies_.empty())
+      return 0;
+    return 0.5 +
+           static_cast<double>(
+               std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
+               latencies_.size();
+  }
+
+  void PrintResults() const {
+    PRINT("] ");
+    for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
+      PRINT("%d ", *it);
+    }
+    PRINT("\n");
+    PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, min_latency(),
+          max_latency(), average_latency());
+  }
+
+  int IndexToMilliseconds(double index) const {
+    return 10.0 * (index / frames_per_buffer_) + 0.5;
+  }
+
+ private:
+  Clock* clock_;
+  const size_t frames_per_buffer_;
+  const size_t bytes_per_buffer_;
+  size_t play_count_;
+  size_t rec_count_;
+  int64_t pulse_time_;
+  std::vector<int> latencies_;
+};
+// Mocks the AudioTransport object and proxies actions for the two callbacks
+// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
+// of AudioStreamInterface.
+class MockAudioTransport : public AudioTransport {
+ public:
+  explicit MockAudioTransport(int type)
+      : num_callbacks_(0),
+        type_(type),
+        play_count_(0),
+        rec_count_(0),
+        audio_stream_(nullptr) {}
+
+  virtual ~MockAudioTransport() {}
+
+  MOCK_METHOD10(RecordedDataIsAvailable,
+                int32_t(const void* audioSamples,
+                        const size_t nSamples,
+                        const size_t nBytesPerSample,
+                        const size_t nChannels,
+                        const uint32_t samplesPerSec,
+                        const uint32_t totalDelayMS,
+                        const int32_t clockDrift,
+                        const uint32_t currentMicLevel,
+                        const bool keyPressed,
+                        uint32_t& newMicLevel));
+  MOCK_METHOD8(NeedMorePlayData,
+               int32_t(const size_t nSamples,
+                       const size_t nBytesPerSample,
+                       const size_t nChannels,
+                       const uint32_t samplesPerSec,
+                       void* audioSamples,
+                       size_t& nSamplesOut,
+                       int64_t* elapsed_time_ms,
+                       int64_t* ntp_time_ms));
+
+  // Set default actions of the mock object. We are delegating to fake
+  // implementations (of AudioStreamInterface) here.
+  void HandleCallbacks(EventWrapper* test_is_done,
+                       AudioStreamInterface* audio_stream,
+                       size_t num_callbacks) {
+    test_is_done_ = test_is_done;
+    audio_stream_ = audio_stream;
+    num_callbacks_ = num_callbacks;
+    if (play_mode()) {
+      ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
+          .WillByDefault(
+              Invoke(this, &MockAudioTransport::RealNeedMorePlayData));
+    }
+    if (rec_mode()) {
+      ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
+          .WillByDefault(
+              Invoke(this, &MockAudioTransport::RealRecordedDataIsAvailable));
+    }
+  }
+
+  int32_t RealRecordedDataIsAvailable(const void* audioSamples,
+                                      const size_t nSamples,
+                                      const size_t nBytesPerSample,
+                                      const size_t nChannels,
+                                      const uint32_t samplesPerSec,
+                                      const uint32_t totalDelayMS,
+                                      const int32_t clockDrift,
+                                      const uint32_t currentMicLevel,
+                                      const bool keyPressed,
+                                      uint32_t& newMicLevel) {
+    EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
+    rec_count_++;
+    // Process the recorded audio stream if an AudioStreamInterface
+    // implementation exists.
+    if (audio_stream_) {
+      audio_stream_->Write(audioSamples, nSamples);
+    }
+    if (ReceivedEnoughCallbacks()) {
+      if (test_is_done_) {
+        test_is_done_->Set();
+      }
+    }
+    return 0;
+  }
+
+  int32_t RealNeedMorePlayData(const size_t nSamples,
+                               const size_t nBytesPerSample,
+                               const size_t nChannels,
+                               const uint32_t samplesPerSec,
+                               void* audioSamples,
+                               size_t& nSamplesOut,
+                               int64_t* elapsed_time_ms,
+                               int64_t* ntp_time_ms) {
+    EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
+    play_count_++;
+    nSamplesOut = nSamples;
+    // Read (possibly processed) audio stream samples to be played out if an
+    // AudioStreamInterface implementation exists.
+    if (audio_stream_) {
+      audio_stream_->Read(audioSamples, nSamples);
+    }
+    if (ReceivedEnoughCallbacks()) {
+      if (test_is_done_) {
+        test_is_done_->Set();
+      }
+    }
+    return 0;
+  }
+
+  bool ReceivedEnoughCallbacks() {
+    bool recording_done = false;
+    if (rec_mode())
+      recording_done = rec_count_ >= num_callbacks_;
+    else
+      recording_done = true;
+
+    bool playout_done = false;
+    if (play_mode())
+      playout_done = play_count_ >= num_callbacks_;
+    else
+      playout_done = true;
+
+    return recording_done && playout_done;
+  }
+
+  bool play_mode() const { return type_ & kPlayout; }
+  bool rec_mode() const { return type_ & kRecording; }
+
+ private:
+  EventWrapper* test_is_done_;
+  size_t num_callbacks_;
+  int type_;
+  size_t play_count_;
+  size_t rec_count_;
+  AudioStreamInterface* audio_stream_;
+};
+
+// AudioDeviceTest test fixture.
+class AudioDeviceTest : public ::testing::Test {
+ protected:
+  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
+    old_sev_ = rtc::LogMessage::GetLogToDebug();
+    // Set suitable logging level here. Change to rtc::LS_INFO for more verbose
+    // output. See webrtc/base/logging.h for complete list of options.
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+    // Add extra logging fields here (timestamps and thread id).
+    // rtc::LogMessage::LogTimestamps();
+    rtc::LogMessage::LogThreads();
+    // Creates an audio device using a default audio layer.
+    audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
+    EXPECT_NE(audio_device_.get(), nullptr);
+    EXPECT_EQ(0, audio_device_->Init());
+    EXPECT_EQ(0,
+              audio_device()->GetPlayoutAudioParameters(&playout_parameters_));
+    EXPECT_EQ(0, audio_device()->GetRecordAudioParameters(&record_parameters_));
+  }
+  virtual ~AudioDeviceTest() {
+    EXPECT_EQ(0, audio_device_->Terminate());
+    rtc::LogMessage::LogToDebug(old_sev_);
+  }
+
+  int playout_sample_rate() const { return playout_parameters_.sample_rate(); }
+  int record_sample_rate() const { return record_parameters_.sample_rate(); }
+  int playout_channels() const { return playout_parameters_.channels(); }
+  int record_channels() const { return record_parameters_.channels(); }
+  size_t playout_frames_per_10ms_buffer() const {
+    return playout_parameters_.frames_per_10ms_buffer();
+  }
+  size_t record_frames_per_10ms_buffer() const {
+    return record_parameters_.frames_per_10ms_buffer();
+  }
+
+  rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
+    return audio_device_;
+  }
+
+  AudioDeviceModuleImpl* audio_device_impl() const {
+    return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
+  }
+
+  AudioDeviceBuffer* audio_device_buffer() const {
+    return audio_device_impl()->GetAudioDeviceBuffer();
+  }
+
+  rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
+      AudioDeviceModule::AudioLayer audio_layer) {
+    rtc::scoped_refptr<AudioDeviceModule> module(
+        AudioDeviceModuleImpl::Create(0, audio_layer));
+    return module;
+  }
+
+  // Returns file name relative to the resource root given a sample rate.
+  std::string GetFileName(int sample_rate) {
+    EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100 ||
+                sample_rate == 16000);
+    char fname[64];
+    snprintf(fname, sizeof(fname), "audio_device/audio_short%d",
+             sample_rate / 1000);
+    std::string file_name(webrtc::test::ResourcePath(fname, "pcm"));
+    EXPECT_TRUE(test::FileExists(file_name));
+#ifdef ENABLE_DEBUG_PRINTF
+    PRINTD("file name: %s\n", file_name.c_str());
+    const size_t bytes = test::GetFileSize(file_name);
+    PRINTD("file size: %" PRIuS " [bytes]\n", bytes);
+    PRINTD("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
+    const int seconds =
+        static_cast<int>(bytes / (sample_rate * kBytesPerSample));
+    PRINTD("file size: %d [secs]\n", seconds);
+    PRINTD("file size: %" PRIuS " [callbacks]\n",
+           seconds * kNumCallbacksPerSecond);
+#endif
+    return file_name;
+  }
+
+  void StartPlayout() {
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+    EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_EQ(0, audio_device()->InitPlayout());
+    EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+    EXPECT_EQ(0, audio_device()->StartPlayout());
+    EXPECT_TRUE(audio_device()->Playing());
+  }
+
+  void StopPlayout() {
+    EXPECT_EQ(0, audio_device()->StopPlayout());
+    EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+  }
+
+  void StartRecording() {
+    EXPECT_FALSE(audio_device()->RecordingIsInitialized());
+    EXPECT_FALSE(audio_device()->Recording());
+    EXPECT_EQ(0, audio_device()->InitRecording());
+    EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+    EXPECT_EQ(0, audio_device()->StartRecording());
+    EXPECT_TRUE(audio_device()->Recording());
+  }
+
+  void StopRecording() {
+    EXPECT_EQ(0, audio_device()->StopRecording());
+    EXPECT_FALSE(audio_device()->Recording());
+  }
+
+  rtc::scoped_ptr<EventWrapper> test_is_done_;
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+  rtc::LoggingSeverity old_sev_;
+};
+
+TEST_F(AudioDeviceTest, ConstructDestruct) {
+  // Using the test fixture to create and destruct the audio device module.
+}
+
+TEST_F(AudioDeviceTest, InitTerminate) {
+  // Initialization is part of the test fixture.
+  EXPECT_TRUE(audio_device()->Initialized());
+  EXPECT_EQ(0, audio_device()->Terminate());
+  EXPECT_FALSE(audio_device()->Initialized());
+}
+
+// Tests that playout can be initiated, started and stopped. No audio callback
+// is registered in this test.
+TEST_F(AudioDeviceTest, StartStopPlayout) {
+  StartPlayout();
+  StopPlayout();
+  StartPlayout();
+  StopPlayout();
+}
+
+// Tests that recording can be initiated, started and stopped. No audio callback
+// is registered in this test.
+TEST_F(AudioDeviceTest, StartStopRecording) {
+  StartRecording();
+  StopRecording();
+  StartRecording();
+  StopRecording();
+}
+
+// Verify that calling StopPlayout() will leave us in an uninitialized state
+// which will require a new call to InitPlayout(). This test does not call
+// StartPlayout() while being uninitialized since doing so will hit a
+// RTC_DCHECK.
+TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_EQ(0, audio_device()->StartPlayout());
+  EXPECT_EQ(0, audio_device()->StopPlayout());
+  EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+}
+
+// Verify that we can create two ADMs and start playing on the second ADM.
+// Only the first active instance shall activate an audio session and the
+// last active instance shall deactivate the audio session. The test does not
+// explicitly verify correct audio session calls but instead focuses on
+// ensuring that audio starts for both ADMs.
+TEST_F(AudioDeviceTest, StartPlayoutOnTwoInstances) {
+  // Create and initialize a second/extra ADM instance. The default ADM is
+  // created by the test harness.
+  rtc::scoped_refptr<AudioDeviceModule> second_audio_device =
+      CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
+  EXPECT_NE(second_audio_device.get(), nullptr);
+  EXPECT_EQ(0, second_audio_device->Init());
+
+  // Start playout for the default ADM but don't wait here. Instead use the
+  // upcoming second stream for that. We set the same expectation on number
+  // of callbacks as for the second stream.
+  NiceMock<MockAudioTransport> mock(kPlayout);
+  mock.HandleCallbacks(nullptr, nullptr, 0);
+  EXPECT_CALL(
+      mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
+                             playout_channels(), playout_sample_rate(),
+                             NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+
+  // Initialize playout for the second ADM. If all is OK, the second ADM shall
+  // reuse the audio session activated when the first ADM started playing.
+  // This call will also ensure that we avoid a problem related to initializing
+  // two different audio unit instances back to back (see webrtc:5166 for
+  // details).
+  EXPECT_EQ(0, second_audio_device->InitPlayout());
+  EXPECT_TRUE(second_audio_device->PlayoutIsInitialized());
+
+  // Start playout for the second ADM and verify that it starts as intended.
+  // Passing this test ensures that initialization of the second audio unit
+  // has been done successfully and that there is no conflict with the already
+  // playing first ADM.
+  MockAudioTransport mock2(kPlayout);
+  mock2.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
+  EXPECT_CALL(
+      mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
+                              playout_channels(), playout_sample_rate(),
+                              NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2));
+  EXPECT_EQ(0, second_audio_device->StartPlayout());
+  EXPECT_TRUE(second_audio_device->Playing());
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  EXPECT_EQ(0, second_audio_device->StopPlayout());
+  EXPECT_FALSE(second_audio_device->Playing());
+  EXPECT_FALSE(second_audio_device->PlayoutIsInitialized());
+
+  EXPECT_EQ(0, second_audio_device->Terminate());
+}
+
+// Start playout and verify that the native audio layer starts asking for real
+// audio samples to play out using the NeedMorePlayData callback.
+TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
+  MockAudioTransport mock(kPlayout);
+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
+                                     kBytesPerSample, playout_channels(),
+                                     playout_sample_rate(), NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  StopPlayout();
+}
+
+// Start recording and verify that the native audio layer starts feeding real
+// audio samples via the RecordedDataIsAvailable callback.
+TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
+  MockAudioTransport mock(kRecording);
+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock,
+              RecordedDataIsAvailable(
+                  NotNull(), record_frames_per_10ms_buffer(), kBytesPerSample,
+                  record_channels(), record_sample_rate(),
+                  _,  // TODO(henrika): fix delay
+                  0, 0, false, _)).Times(AtLeast(kNumCallbacks));
+
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  StopRecording();
+}
+
+// Start playout and recording (full-duplex audio) and verify that audio is
+// active in both directions.
+TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
+  MockAudioTransport mock(kPlayout | kRecording);
+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
+                                     kBytesPerSample, playout_channels(),
+                                     playout_sample_rate(), NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_CALL(mock,
+              RecordedDataIsAvailable(
+                  NotNull(), record_frames_per_10ms_buffer(), kBytesPerSample,
+                  record_channels(), record_sample_rate(),
+                  _,  // TODO(henrika): fix delay
+                  0, 0, false, _)).Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  StartRecording();
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  StopRecording();
+  StopPlayout();
+}
+
+// Start playout and read audio from an external PCM file when the audio layer
+// asks for data to play out. Real audio is played out in this test but it does
+// not contain any explicit verification that the audio quality is perfect.
+TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
+  // TODO(henrika): extend test when mono output is supported.
+  EXPECT_EQ(1, playout_channels());
+  NiceMock<MockAudioTransport> mock(kPlayout);
+  const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
+  std::string file_name = GetFileName(playout_sample_rate());
+  rtc::scoped_ptr<FileAudioStream> file_audio_stream(
+      new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
+  mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
+                       num_callbacks);
+  // SetMaxPlayoutVolume();
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  StopPlayout();
+}
+
+TEST_F(AudioDeviceTest, Devices) {
+  // Device enumeration is not supported. Verify fixed values only.
+  EXPECT_EQ(1, audio_device()->PlayoutDevices());
+  EXPECT_EQ(1, audio_device()->RecordingDevices());
+}
+
+// Start playout and recording and store recorded data in an intermediate FIFO
+// buffer from which the playout side then reads its samples in the same order
+// as they were stored. Under ideal circumstances, a callback sequence would
+// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
+// means 'packet played'. Under such conditions, the FIFO would only contain
+// one packet on average. However, under more realistic conditions, the size
+// of the FIFO will vary more due to an unbalance between the two sides.
+// This test tries to verify that the device maintains a balanced callback-
+// sequence by running in loopback for ten seconds while measuring the size
+// (max and average) of the FIFO. The size of the FIFO is increased by the
+// recording side and decreased by the playout side.
+// TODO(henrika): tune the final test parameters after running tests on several
+// different devices.
+TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
+  EXPECT_EQ(record_channels(), playout_channels());
+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
+  NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
+  rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream(
+      new FifoAudioStream(playout_frames_per_10ms_buffer()));
+  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
+                       kFullDuplexTimeInSec * kNumCallbacksPerSecond);
+  // SetMaxPlayoutVolume();
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  StartPlayout();
+  test_is_done_->Wait(
+      std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
+  StopPlayout();
+  StopRecording();
+  EXPECT_LE(fifo_audio_stream->average_size(), 10u);
+  EXPECT_LE(fifo_audio_stream->largest_size(), 20u);
+}
+
+// Measures loopback latency and reports the min, max and average values for
+// a full duplex audio session.
+// The latency is measured like so:
+// - Insert impulses periodically on the output side.
+// - Detect the impulses on the input side.
+// - Measure the time difference between the transmit time and receive time.
+// - Store time differences in a vector and calculate min, max and average.
+// This test requires a special hardware called Audio Loopback Dongle.
+// See http://source.android.com/devices/audio/loopback.html for details.
+TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
+  EXPECT_EQ(record_channels(), playout_channels());
+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
+  NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
+  rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
+      new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
+  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
+                       kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  // SetMaxPlayoutVolume();
+  // DisableBuiltInAECIfAvailable();
+  StartRecording();
+  StartPlayout();
+  test_is_done_->Wait(
+      std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
+  StopPlayout();
+  StopRecording();
+  // Verify that the correct number of transmitted impulses are detected.
+  EXPECT_EQ(latency_audio_stream->num_latency_values(),
+            static_cast<size_t>(
+                kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
+  latency_audio_stream->PrintResults();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_utility_ios.cc b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_utility_ios.cc
deleted file mode 100644
index 336281757a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_utility_ios.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/ios/audio_device_utility_ios.h"
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-AudioDeviceUtilityIOS::AudioDeviceUtilityIOS(const int32_t id)
-:
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
-    _lastError(AudioDeviceModule::kAdmErrNone) {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
-}
-
-AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS() {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
-    {
-        CriticalSectionScoped lock(&_critSect);
-    }
-    delete &_critSect;
-}
-
-int32_t AudioDeviceUtilityIOS::Init() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                 "  OS info: %s", "iOS");
-
-    return 0;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_utility_ios.h b/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_utility_ios.h
deleted file mode 100644
index 16948685d2..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_utility_ios.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H
-
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityIOS: public AudioDeviceUtility {
- public:
-    AudioDeviceUtilityIOS(const int32_t id);
-    AudioDeviceUtilityIOS();
-    virtual ~AudioDeviceUtilityIOS();
-
-    virtual int32_t Init();
-
- private:
-    CriticalSectionWrapper& _critSect;
-    int32_t _id;
-    AudioDeviceModule::ErrorCode _lastError;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
index 7bc70e5cc5..0028d6fa3f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -11,13 +11,12 @@
 #include <assert.h>
 
 #include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
 #include "webrtc/modules/audio_device/linux/audio_device_alsa_linux.h"
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/trace.h"
+ 
 #include "Latency.h"
 
 #define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
@@ -25,6 +24,7 @@
 #define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
                                                  reinterpret_cast<uint64_t>(x), frames)
 
+
 webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
 
 // Accesses ALSA functions through our late-binding symbol table instead of
@@ -216,7 +216,7 @@ int32_t AudioDeviceLinuxALSA::Terminate()
     // RECORDING
     if (_ptrThreadRec)
     {
-        ThreadWrapper* tmpThread = _ptrThreadRec.release();
+        rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
         _critSect.Leave();
 
         tmpThread->Stop();
@@ -228,7 +228,7 @@ int32_t AudioDeviceLinuxALSA::Terminate()
     // PLAYOUT
     if (_ptrThreadPlay)
     {
-        ThreadWrapper* tmpThread = _ptrThreadPlay.release();
+        rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
         _critSect.Leave();
 
         tmpThread->Stop();
@@ -1373,22 +1373,12 @@ int32_t AudioDeviceLinuxALSA::StartRecording()
         return -1;
     }
     // RECORDING
-    const char* threadName = "webrtc_audio_module_capture_thread";
     _firstRecord = true;
-    _ptrThreadRec = ThreadWrapper::CreateThread(
-        RecThreadFunc, this, threadName);
+    _ptrThreadRec.reset(new rtc::PlatformThread(
+        RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
-    if (!_ptrThreadRec->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the rec audio thread");
-        _recording = false;
-        _ptrThreadRec.reset();
-        delete [] _recordingBuffer;
-        _recordingBuffer = NULL;
-        return -1;
-    }
-    _ptrThreadRec->SetPriority(kRealtimePriority);
+    _ptrThreadRec->Start();
+    _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
     errVal = LATE(snd_pcm_prepare)(_handleRecord);
     if (errVal < 0)
@@ -1528,9 +1518,9 @@ int32_t AudioDeviceLinuxALSA::StartPlayout()
     }
 
     // PLAYOUT
-    const char* threadName = "webrtc_audio_module_play_thread";
-    _ptrThreadPlay =  ThreadWrapper::CreateThread(PlayThreadFunc, this,
-                                                  threadName);
+    _ptrThreadPlay.reset(new rtc::PlatformThread(
+        PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+
     int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
     if (errVal < 0)
     {
@@ -1541,17 +1531,8 @@ int32_t AudioDeviceLinuxALSA::StartPlayout()
         // if snd_pcm_open fails will return -1
     }
 
-    if (!_ptrThreadPlay->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the play audio thread");
-        _playing = false;
-        _ptrThreadPlay.reset();
-        delete [] _playoutBuffer;
-        _playoutBuffer = NULL;
-        return -1;
-    }
-    _ptrThreadPlay->SetPriority(kRealtimePriority);
+    _ptrThreadPlay->Start();
+    _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
 
     return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
index ad92560020..a08de541fd 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #if defined(USE_X11)
 #include <X11/Xlib.h>
@@ -187,8 +187,10 @@ private:
 
     CriticalSectionWrapper& _critSect;
 
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
+    // TODO(pbos): Make plain members and start/stop instead of resetting these
+    // pointers. A thread can be reused.
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
     int32_t _id;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
index 7e4b17aaff..6b9a439cb1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -10,12 +10,13 @@
 
 #include <assert.h>
 
+#include "webrtc/base/checks.h"
+
 #include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
 #include "webrtc/modules/audio_device/linux/audio_device_pulse_linux.h"
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
 
@@ -28,10 +29,6 @@ webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
 namespace webrtc
 {
 
-// ============================================================================
-//                              Static Methods
-// ============================================================================
-
 AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const int32_t id) :
     _ptrAudioBuffer(NULL),
     _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
@@ -109,7 +106,7 @@ AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
                  "%s destroyed", __FUNCTION__);
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     Terminate();
 
     if (_recBuffer)
@@ -142,8 +139,7 @@ AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
 
 void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
 {
-
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
     _ptrAudioBuffer = audioBuffer;
 
@@ -169,9 +165,7 @@ int32_t AudioDeviceLinuxPulse::ActiveAudioLayer(
 
 int32_t AudioDeviceLinuxPulse::Init()
 {
-
-    CriticalSectionScoped lock(&_critSect);
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_initialized)
     {
         return 0;
@@ -208,33 +202,17 @@ int32_t AudioDeviceLinuxPulse::Init()
 #endif
 
     // RECORDING
-    const char* threadName = "webrtc_audio_module_rec_thread";
-    _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
-                                                threadName);
-    if (!_ptrThreadRec->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the rec audio thread");
+    _ptrThreadRec.reset(new rtc::PlatformThread(
+        RecThreadFunc, this, "webrtc_audio_module_rec_thread"));
 
-        _ptrThreadRec.reset();
-        return -1;
-    }
-
-    _ptrThreadRec->SetPriority(kRealtimePriority);
+    _ptrThreadRec->Start();
+    _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
     // PLAYOUT
-    threadName = "webrtc_audio_module_play_thread";
-    _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
-                                                 threadName);
-    if (!_ptrThreadPlay->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "  failed to start the play audio thread");
-
-        _ptrThreadPlay.reset();
-        return -1;
-    }
-    _ptrThreadPlay->SetPriority(kRealtimePriority);
+    _ptrThreadPlay.reset(new rtc::PlatformThread(
+        PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+    _ptrThreadPlay->Start();
+    _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
 
     _initialized = true;
 
@@ -243,40 +221,32 @@ int32_t AudioDeviceLinuxPulse::Init()
 
 int32_t AudioDeviceLinuxPulse::Terminate()
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (!_initialized)
     {
         return 0;
     }
 
-    Lock();
-
     _mixerManager.Close();
 
     // RECORDING
     if (_ptrThreadRec)
     {
-        ThreadWrapper* tmpThread = _ptrThreadRec.release();
-        UnLock();
+        rtc::PlatformThread* tmpThread = _ptrThreadRec.release();
 
         _timeEventRec.Set();
         tmpThread->Stop();
         delete tmpThread;
-        // Lock again since we need to protect _ptrThreadPlay.
-        Lock();
     }
 
     // PLAYOUT
     if (_ptrThreadPlay)
     {
-        ThreadWrapper* tmpThread = _ptrThreadPlay.release();
-        _critSect.Leave();
+        rtc::PlatformThread* tmpThread = _ptrThreadPlay.release();
 
         _timeEventPlay.Set();
         tmpThread->Stop();
         delete tmpThread;
-    } else {
-      UnLock();
     }
 
     // Terminate PulseAudio
@@ -304,13 +274,13 @@ int32_t AudioDeviceLinuxPulse::Terminate()
 
 bool AudioDeviceLinuxPulse::Initialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_initialized);
 }
 
 int32_t AudioDeviceLinuxPulse::InitSpeaker()
 {
-
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
     if (_playing)
     {
@@ -354,9 +324,7 @@ int32_t AudioDeviceLinuxPulse::InitSpeaker()
 
 int32_t AudioDeviceLinuxPulse::InitMicrophone()
 {
-
-    CriticalSectionScoped lock(&_critSect);
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_recording)
     {
         return -1;
@@ -399,17 +367,19 @@ int32_t AudioDeviceLinuxPulse::InitMicrophone()
 
 bool AudioDeviceLinuxPulse::SpeakerIsInitialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_mixerManager.SpeakerIsInitialized());
 }
 
 bool AudioDeviceLinuxPulse::MicrophoneIsInitialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_mixerManager.MicrophoneIsInitialized());
 }
 
 int32_t AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
     // Make an attempt to open up the
@@ -422,7 +392,7 @@ int32_t AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
         return 0;
     }
 
-    // Given that InitSpeaker was successful, we know that a volume control exists
+    // Given that InitSpeaker was successful, we know volume control exists.
     available = true;
 
     // Close the initialized output mixer
@@ -436,6 +406,7 @@ int32_t AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::SetSpeakerVolume(uint32_t volume)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (!_playing) {
       // Only update the volume if it's been set while we weren't playing.
       update_speaker_volume_at_startup_ = true;
@@ -445,7 +416,7 @@ int32_t AudioDeviceLinuxPulse::SetSpeakerVolume(uint32_t volume)
 
 int32_t AudioDeviceLinuxPulse::SpeakerVolume(uint32_t& volume) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     uint32_t level(0);
 
     if (_mixerManager.SpeakerVolume(level) == -1)
@@ -481,7 +452,7 @@ int32_t AudioDeviceLinuxPulse::WaveOutVolume(
 int32_t AudioDeviceLinuxPulse::MaxSpeakerVolume(
     uint32_t& maxVolume) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     uint32_t maxVol(0);
 
     if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
@@ -497,7 +468,7 @@ int32_t AudioDeviceLinuxPulse::MaxSpeakerVolume(
 int32_t AudioDeviceLinuxPulse::MinSpeakerVolume(
     uint32_t& minVolume) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     uint32_t minVol(0);
 
     if (_mixerManager.MinSpeakerVolume(minVol) == -1)
@@ -513,7 +484,7 @@ int32_t AudioDeviceLinuxPulse::MinSpeakerVolume(
 int32_t AudioDeviceLinuxPulse::SpeakerVolumeStepSize(
     uint16_t& stepSize) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     uint16_t delta(0);
 
     if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
@@ -528,7 +499,7 @@ int32_t AudioDeviceLinuxPulse::SpeakerVolumeStepSize(
 
 int32_t AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool isAvailable(false);
     bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
@@ -560,13 +531,13 @@ int32_t AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::SetSpeakerMute(bool enable)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_mixerManager.SetSpeakerMute(enable));
 }
 
 int32_t AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool muted(0);
     if (_mixerManager.SpeakerMute(muted) == -1)
     {
@@ -579,7 +550,7 @@ int32_t AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const
 
 int32_t AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool isAvailable(false);
     bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
@@ -588,9 +559,9 @@ int32_t AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available)
     //
     if (!wasInitialized && InitMicrophone() == -1)
     {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
+        // If we end up here it means that the selected microphone has no
+        // volume control, hence it is safe to state that there is no
+        // boost control already at this stage.
         available = false;
         return 0;
     }
@@ -612,13 +583,13 @@ int32_t AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::SetMicrophoneMute(bool enable)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_mixerManager.SetMicrophoneMute(enable));
 }
 
 int32_t AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool muted(0);
     if (_mixerManager.MicrophoneMute(muted) == -1)
     {
@@ -631,7 +602,7 @@ int32_t AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const
 
 int32_t AudioDeviceLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool isAvailable(false);
     bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
@@ -640,9 +611,9 @@ int32_t AudioDeviceLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
     //
     if (!wasInitialized && InitMicrophone() == -1)
     {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
+        // If we end up here it means that the selected microphone has no
+        // volume control, hence it is safe to state that there is no
+        // boost control already at this stage.
         available = false;
         return 0;
     }
@@ -662,13 +633,13 @@ int32_t AudioDeviceLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::SetMicrophoneBoost(bool enable)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_mixerManager.SetMicrophoneBoost(enable));
 }
 
 int32_t AudioDeviceLinuxPulse::MicrophoneBoost(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool onOff(0);
 
     if (_mixerManager.MicrophoneBoost(onOff) == -1)
@@ -683,7 +654,7 @@ int32_t AudioDeviceLinuxPulse::MicrophoneBoost(bool& enabled) const
 
 int32_t AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_recChannels == 2 && _recording) {
       available = true;
       return 0;
@@ -717,7 +688,7 @@ int32_t AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::SetStereoRecording(bool enable)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (enable)
         _recChannels = 2;
     else
@@ -728,7 +699,7 @@ int32_t AudioDeviceLinuxPulse::SetStereoRecording(bool enable)
 
 int32_t AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_recChannels == 2)
         enabled = true;
     else
@@ -739,7 +710,7 @@ int32_t AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const
 
 int32_t AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_playChannels == 2 && _playing) {
       available = true;
       return 0;
@@ -772,7 +743,7 @@ int32_t AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::SetStereoPlayout(bool enable)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (enable)
         _playChannels = 2;
     else
@@ -783,7 +754,7 @@ int32_t AudioDeviceLinuxPulse::SetStereoPlayout(bool enable)
 
 int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_playChannels == 2)
         enabled = true;
     else
@@ -794,7 +765,7 @@ int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const
 
 int32_t AudioDeviceLinuxPulse::SetAGC(bool enable)
 {
-
+    CriticalSectionScoped lock(&_critSect);
     _AGC = enable;
 
     return 0;
@@ -802,28 +773,28 @@ int32_t AudioDeviceLinuxPulse::SetAGC(bool enable)
 
 bool AudioDeviceLinuxPulse::AGC() const
 {
-
+    CriticalSectionScoped lock(&_critSect);
     return _AGC;
 }
 
 int32_t AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(
     bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
     // Make an attempt to open up the
     // input mixer corresponding to the currently selected output device.
     if (!wasInitialized && InitMicrophone() == -1)
     {
-        // If we end up here it means that the selected microphone has no volume
-        // control.
+        // If we end up here it means that the selected microphone has no
+        // volume control.
         available = false;
         return 0;
     }
 
     // Given that InitMicrophone was successful, we know that a volume control
-    // exists
+    // exists.
     available = true;
 
     // Close the initialized input mixer
@@ -837,7 +808,6 @@ int32_t AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(
 
 int32_t AudioDeviceLinuxPulse::SetMicrophoneVolume(uint32_t volume)
 {
-
     return (_mixerManager.SetMicrophoneVolume(volume));
 }
 
@@ -894,7 +864,7 @@ int32_t AudioDeviceLinuxPulse::MinMicrophoneVolume(
 int32_t AudioDeviceLinuxPulse::MicrophoneVolumeStepSize(
     uint16_t& stepSize) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     uint16_t delta(0);
 
     if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
@@ -909,7 +879,6 @@ int32_t AudioDeviceLinuxPulse::MicrophoneVolumeStepSize(
 
 int16_t AudioDeviceLinuxPulse::PlayoutDevices()
 {
-
     PaLock();
 
     pa_operation* paOperation = NULL;
@@ -929,7 +898,7 @@ int16_t AudioDeviceLinuxPulse::PlayoutDevices()
 
 int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(uint16_t index)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_playIsInitialized)
     {
         return -1;
@@ -966,7 +935,7 @@ int32_t AudioDeviceLinuxPulse::PlayoutDeviceName(
     char name[kAdmMaxDeviceNameSize],
     char guid[kAdmMaxGuidSize])
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     const uint16_t nDevices = PlayoutDevices();
 
     if ((index > (nDevices - 1)) || (name == NULL))
@@ -1008,7 +977,7 @@ int32_t AudioDeviceLinuxPulse::RecordingDeviceName(
     char name[kAdmMaxDeviceNameSize],
     char guid[kAdmMaxGuidSize])
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     const uint16_t nDevices(RecordingDevices());
 
     if ((index > (nDevices - 1)) || (name == NULL))
@@ -1047,7 +1016,6 @@ int32_t AudioDeviceLinuxPulse::RecordingDeviceName(
 
 int16_t AudioDeviceLinuxPulse::RecordingDevices()
 {
-
     PaLock();
 
     pa_operation* paOperation = NULL;
@@ -1067,7 +1035,7 @@ int16_t AudioDeviceLinuxPulse::RecordingDevices()
 
 int32_t AudioDeviceLinuxPulse::SetRecordingDevice(uint16_t index)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_recIsInitialized)
     {
         return -1;
@@ -1101,7 +1069,7 @@ int32_t AudioDeviceLinuxPulse::SetRecordingDevice(
 
 int32_t AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     available = false;
 
     // Try to initialize the playout side
@@ -1120,7 +1088,7 @@ int32_t AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     available = false;
 
     // Try to initialize the playout side
@@ -1139,8 +1107,7 @@ int32_t AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available)
 
 int32_t AudioDeviceLinuxPulse::InitPlayout()
 {
-
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
     if (_playing)
     {
@@ -1193,7 +1160,8 @@ int32_t AudioDeviceLinuxPulse::InitPlayout()
     }
 
     WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  stream state %d\n", LATE(pa_stream_get_state)(_playStream));
+                 "  stream state %d\n",
+                 LATE(pa_stream_get_state)(_playStream));
 
     // Set stream flags
     _playStreamFlags = (pa_stream_flags_t) (PA_STREAM_AUTO_TIMING_UPDATE
@@ -1203,9 +1171,9 @@ int32_t AudioDeviceLinuxPulse::InitPlayout()
     {
         // If configuring a specific latency then we want to specify
         // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
-        // automatically to reach that target latency. However, that flag doesn't
-        // exist in Ubuntu 8.04 and many people still use that, so we have to check
-        // the protocol version of libpulse.
+        // automatically to reach that target latency. However, that flag
+        // doesn't exist in Ubuntu 8.04 and many people still use that,
+        // so we have to check the protocol version of libpulse.
         if (LATE(pa_context_get_protocol_version)(_paContext)
             >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
         {
@@ -1222,16 +1190,18 @@ int32_t AudioDeviceLinuxPulse::InitPlayout()
         }
 
         size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
-        uint32_t latency = bytesPerSec
-            * WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS / WEBRTC_PA_MSECS_PER_SEC;
+        uint32_t latency = bytesPerSec *
+                           WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS /
+                           WEBRTC_PA_MSECS_PER_SEC;
 
         // Set the play buffer attributes
         _playBufferAttr.maxlength = latency; // num bytes stored in the buffer
         _playBufferAttr.tlength = latency; // target fill level of play buffer
         // minimum free num bytes before server request more data
         _playBufferAttr.minreq = latency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
-        _playBufferAttr.prebuf = _playBufferAttr.tlength
-            - _playBufferAttr.minreq; // prebuffer tlength before starting playout
+        // prebuffer tlength before starting playout
+        _playBufferAttr.prebuf = _playBufferAttr.tlength -
+                                 _playBufferAttr.minreq;
 
         _configuredLatencyPlay = latency;
     }
@@ -1246,7 +1216,8 @@ int32_t AudioDeviceLinuxPulse::InitPlayout()
                                            PaStreamUnderflowCallback, this);
 
     // Set the state callback function for the stream
-    LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this);
+    LATE(pa_stream_set_state_callback)(_playStream,
+                                       PaStreamStateCallback, this);
 
     // Mark playout side as initialized
     _playIsInitialized = true;
@@ -1258,8 +1229,7 @@ int32_t AudioDeviceLinuxPulse::InitPlayout()
 
 int32_t AudioDeviceLinuxPulse::InitRecording()
 {
-
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
     if (_recording)
     {
@@ -1317,9 +1287,9 @@ int32_t AudioDeviceLinuxPulse::InitRecording()
 
         // If configuring a specific latency then we want to specify
         // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
-        // automatically to reach that target latency. However, that flag doesn't
-        // exist in Ubuntu 8.04 and many people still use that, so we have to check
-        // the protocol version of libpulse.
+        // automatically to reach that target latency. However, that flag
+        // doesn't exist in Ubuntu 8.04 and many people still use that,
+        //  so we have to check the protocol version of libpulse.
         if (LATE(pa_context_get_protocol_version)(_paContext)
             >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
         {
@@ -1354,11 +1324,14 @@ int32_t AudioDeviceLinuxPulse::InitRecording()
     _recBuffer = new int8_t[_recordBufferSize];
 
     // Enable overflow callback
-    LATE(pa_stream_set_overflow_callback)(_recStream, PaStreamOverflowCallback,
+    LATE(pa_stream_set_overflow_callback)(_recStream,
+                                          PaStreamOverflowCallback,
                                           this);
 
     // Set the state callback function for the stream
-    LATE(pa_stream_set_state_callback)(_recStream, PaStreamStateCallback, this);
+    LATE(pa_stream_set_state_callback)(_recStream,
+                                       PaStreamStateCallback,
+                                       this);
 
     // Mark recording side as initialized
     _recIsInitialized = true;
@@ -1368,7 +1341,7 @@ int32_t AudioDeviceLinuxPulse::InitRecording()
 
 int32_t AudioDeviceLinuxPulse::StartRecording()
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (!_recIsInitialized)
     {
         return -1;
@@ -1379,10 +1352,10 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
         return 0;
     }
 
-    // set state to ensure that the recording starts from the audio thread
+    // Set state to ensure that the recording starts from the audio thread.
     _startRec = true;
 
-    // the audio thread will signal when recording has started
+    // The audio thread will signal when recording has started.
     _timeEventRec.Set();
     if (kEventTimeout == _recStartEvent.Wait(10000))
     {
@@ -1400,7 +1373,8 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
         CriticalSectionScoped lock(&_critSect);
         if (_recording)
         {
-            // the recording state is set by the audio thread after recording has started
+            // The recording state is set by the audio thread after recording
+            // has started.
         } else
         {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -1414,7 +1388,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
 
 int32_t AudioDeviceLinuxPulse::StopRecording()
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     CriticalSectionScoped lock(&_critSect);
 
     if (!_recIsInitialized)
@@ -1477,22 +1451,26 @@ int32_t AudioDeviceLinuxPulse::StopRecording()
 
 bool AudioDeviceLinuxPulse::RecordingIsInitialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_recIsInitialized);
 }
 
 bool AudioDeviceLinuxPulse::Recording() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_recording);
 }
 
 bool AudioDeviceLinuxPulse::PlayoutIsInitialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_playIsInitialized);
 }
 
 int32_t AudioDeviceLinuxPulse::StartPlayout()
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
     if (!_playIsInitialized)
     {
         return -1;
@@ -1503,13 +1481,16 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
         return 0;
     }
 
-    // set state to ensure that playout starts from the audio thread
-    _startPlay = true;
+    // Set state to ensure that playout starts from the audio thread.
+    {
+        CriticalSectionScoped lock(&_critSect);
+        _startPlay = true;
+    }
 
     // Both |_startPlay| and |_playing| needs protction since they are also
     // accessed on the playout thread.
 
-    // the audio thread will signal when playout has started
+    // The audio thread will signal when playout has started.
     _timeEventPlay.Set();
     if (kEventTimeout == _playStartEvent.Wait(10000))
     {
@@ -1527,7 +1508,8 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
         CriticalSectionScoped lock(&_critSect);
         if (_playing)
         {
-            // the playing state is set by the audio thread after playout has started
+            // The playing state is set by the audio thread after playout
+            // has started.
         } else
         {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -1541,7 +1523,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
 
 int32_t AudioDeviceLinuxPulse::StopPlayout()
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     CriticalSectionScoped lock(&_critSect);
 
     if (!_playIsInitialized)
@@ -1613,14 +1595,14 @@ int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const
 
 int32_t AudioDeviceLinuxPulse::RecordingDelay(uint16_t& delayMS) const
 {
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     delayMS = (uint16_t) _sndCardRecDelay;
     return 0;
 }
 
 bool AudioDeviceLinuxPulse::Playing() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     return (_playing);
 }
 
@@ -1628,7 +1610,7 @@ int32_t AudioDeviceLinuxPulse::SetPlayoutBuffer(
     const AudioDeviceModule::BufferType type,
     uint16_t sizeMS)
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (type != AudioDeviceModule::kFixedBufferSize)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -1646,7 +1628,7 @@ int32_t AudioDeviceLinuxPulse::PlayoutBuffer(
     AudioDeviceModule::BufferType& type,
     uint16_t& sizeMS) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     type = _playBufType;
     sizeMS = _playBufDelayFixed;
 
@@ -1715,8 +1697,8 @@ void AudioDeviceLinuxPulse::ClearRecordingError()
 
 void AudioDeviceLinuxPulse::PaContextStateCallback(pa_context *c, void *pThis)
 {
-    static_cast<AudioDeviceLinuxPulse*> (pThis)->PaContextStateCallbackHandler(
-        c);
+    static_cast<AudioDeviceLinuxPulse*> (pThis)->
+        PaContextStateCallbackHandler(c);
 }
 
 // ----------------------------------------------------------------------------
@@ -1743,12 +1725,14 @@ void AudioDeviceLinuxPulse::PaServerInfoCallback(pa_context */*c*/,
                                                  const pa_server_info *i,
                                                  void *pThis)
 {
-    static_cast<AudioDeviceLinuxPulse*> (pThis)->PaServerInfoCallbackHandler(i);
+    static_cast<AudioDeviceLinuxPulse*> (pThis)->
+        PaServerInfoCallbackHandler(i);
 }
 
 void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream *p, void *pThis)
 {
-    static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamStateCallbackHandler(p);
+    static_cast<AudioDeviceLinuxPulse*> (pThis)->
+        PaStreamStateCallbackHandler(p);
 }
 
 void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context *c)
@@ -1856,7 +1840,8 @@ void AudioDeviceLinuxPulse::PaSourceInfoCallbackHandler(
     }
 }
 
-void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(const pa_server_info *i)
+void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(
+    const pa_server_info *i)
 {
     // Use PA native sampling rate
     sample_rate_hz_ = i->sample_spec.rate;
@@ -1922,7 +1907,8 @@ int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion()
 
     // get the server info and update deviceName
     paOperation = LATE(pa_context_get_server_info)(_paContext,
-                                                   PaServerInfoCallback, this);
+                                                   PaServerInfoCallback,
+                                                   this);
 
     WaitForOperationCompletion(paOperation);
 
@@ -1942,7 +1928,8 @@ int32_t AudioDeviceLinuxPulse::InitSamplingFrequency()
 
     // Get the server info and update sample_rate_hz_
     paOperation = LATE(pa_context_get_server_info)(_paContext,
-                                                   PaServerInfoCallback, this);
+                                                   PaServerInfoCallback,
+                                                   this);
 
     WaitForOperationCompletion(paOperation);
 
@@ -1989,7 +1976,8 @@ int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
 
     // Get the server info and update deviceName
     paOperation = LATE(pa_context_get_server_info)(_paContext,
-                                                   PaServerInfoCallback, this);
+                                                   PaServerInfoCallback,
+                                                   this);
 
     WaitForOperationCompletion(paOperation);
 
@@ -2006,7 +1994,8 @@ int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
         paOperation
             = LATE(pa_context_get_sink_info_by_name)(_paContext,
                                                      (char *) tmpName,
-                                                     PaSinkInfoCallback, this);
+                                                     PaSinkInfoCallback,
+                                                     this);
     }
 
     WaitForOperationCompletion(paOperation);
@@ -2108,7 +2097,9 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio()
 
     // Connect the context to a server (default)
     _paStateChanged = false;
-    retVal = LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN,
+    retVal = LATE(pa_context_connect)(_paContext,
+                                      NULL,
+                                      PA_CONTEXT_NOAUTOSPAWN,
                                       NULL);
 
     if (retVal != PA_OK)
@@ -2158,7 +2149,8 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio()
     if (CheckPulseAudioVersion() < 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  PulseAudio version %s not supported", _paServerVersion);
+                     "  PulseAudio version %s not supported",
+                     _paServerVersion);
         return -1;
     }
 
@@ -2166,7 +2158,8 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio()
     if (InitSamplingFrequency() < 0 || sample_rate_hz_ == 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  failed to initialize sampling frequency, set to %d Hz",
+                     "  failed to initialize sampling frequency,"
+                     " set to %d Hz",
                      sample_rate_hz_);
         return -1;
     }
@@ -2259,9 +2252,9 @@ void AudioDeviceLinuxPulse::EnableWriteCallback()
         _tempBufferSpace = LATE(pa_stream_writable_size)(_playStream);
         if (_tempBufferSpace > 0)
         {
-            // Yup, there is already space available, so if we register a write
-            // callback then it will not receive any event. So dispatch one ourself
-            // instead
+            // Yup, there is already space available, so if we register a
+            // write callback then it will not receive any event. So dispatch
+            // one ourself instead.
             _timeEventPlay.Set();
             return;
         }
@@ -2298,7 +2291,8 @@ void AudioDeviceLinuxPulse::PaStreamWriteCallbackHandler(size_t bufferSpace)
 void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream */*unused*/,
                                                       void *pThis)
 {
-    static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamUnderflowCallbackHandler();
+    static_cast<AudioDeviceLinuxPulse*> (pThis)->
+        PaStreamUnderflowCallbackHandler();
 }
 
 void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler()
@@ -2308,8 +2302,8 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler()
 
     if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
     {
-        // We didn't configure a pa_buffer_attr before, so switching to one now
-        // would be questionable.
+        // We didn't configure a pa_buffer_attr before, so switching to
+        // one now would be questionable.
         return;
     }
 
@@ -2324,8 +2318,9 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler()
     }
 
     size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
-    uint32_t newLatency = _configuredLatencyPlay + bytesPerSec
-        * WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS / WEBRTC_PA_MSECS_PER_SEC;
+    uint32_t newLatency = _configuredLatencyPlay + bytesPerSec *
+                          WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS /
+                          WEBRTC_PA_MSECS_PER_SEC;
 
     // Set the play buffer attributes
     _playBufferAttr.maxlength = newLatency;
@@ -2352,7 +2347,9 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler()
 
 void AudioDeviceLinuxPulse::EnableReadCallback()
 {
-    LATE(pa_stream_set_read_callback)(_recStream, &PaStreamReadCallback, this);
+    LATE(pa_stream_set_read_callback)(_recStream,
+                                      &PaStreamReadCallback,
+                                      this);
 }
 
 void AudioDeviceLinuxPulse::DisableReadCallback()
@@ -2364,15 +2361,17 @@ void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream */*unused1*/,
                                                  size_t /*unused2*/,
                                                  void *pThis)
 {
-    static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamReadCallbackHandler();
+    static_cast<AudioDeviceLinuxPulse*> (pThis)->
+        PaStreamReadCallbackHandler();
 }
 
 void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler()
 {
     // We get the data pointer and size now in order to save one Lock/Unlock
-    // in the worker thread
-    if (LATE(pa_stream_peek)(_recStream, &_tempSampleData, &_tempSampleDataSize)
-        != 0)
+    // in the worker thread.
+    if (LATE(pa_stream_peek)(_recStream,
+                             &_tempSampleData,
+                             &_tempSampleDataSize) != 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                      "  Can't read data!");
@@ -2393,7 +2392,7 @@ void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler()
 
     // Since we consume the data asynchronously on a different thread, we have
     // to temporarily disable the read callback or else Pulse will call it
-    // continuously until we consume the data. We re-enable it below
+    // continuously until we consume the data. We re-enable it below.
     DisableReadCallback();
     _timeEventRec.Set();
 }
@@ -2401,7 +2400,8 @@ void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler()
 void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream */*unused*/,
                                                      void *pThis)
 {
-    static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamOverflowCallbackHandler();
+    static_cast<AudioDeviceLinuxPulse*> (pThis)->
+        PaStreamOverflowCallbackHandler();
 }
 
 void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler()
@@ -2428,23 +2428,24 @@ int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream *stream)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                      "  Can't query latency");
-        // We'd rather continue playout/capture with an incorrect delay than stop
-        // it altogether, so return a valid value.
+        // We'd rather continue playout/capture with an incorrect delay than
+        // stop it altogether, so return a valid value.
         return 0;
     }
 
     if (negative)
     {
         WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "  warning: pa_stream_get_latency reported negative delay");
+                     "  warning: pa_stream_get_latency reported negative "
+                     "delay");
 
         // The delay can be negative for monitoring streams if the captured
-        // samples haven't been played yet. In such a case, "latency" contains the
-        // magnitude, so we must negate it to get the real value.
+        // samples haven't been played yet. In such a case, "latency"
+        // contains the magnitude, so we must negate it to get the real value.
         int32_t tmpLatency = (int32_t) -latency;
         if (tmpLatency < 0)
         {
-            // Make sure that we don't use a negative delay
+            // Make sure that we don't use a negative delay.
             tmpLatency = 0;
         }
 
@@ -2455,13 +2456,14 @@ int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream *stream)
     }
 }
 
-int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
-                                                size_t bufferSize)
+int32_t AudioDeviceLinuxPulse::ReadRecordedData(
+    const void* bufferData,
+    size_t bufferSize)  EXCLUSIVE_LOCKS_REQUIRED(_critSect)
 {
     size_t size = bufferSize;
     uint32_t numRecSamples = _recordBufferSize / (2 * _recChannels);
 
-    // Account for the peeked data and the used data
+    // Account for the peeked data and the used data.
     uint32_t recDelay = (uint32_t) ((LatencyUsecs(_recStream)
         / 1000) + 10 * ((size + _recordBufferUsed) / _recordBufferSize));
 
@@ -2469,13 +2471,13 @@ int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
 
     if (_playStream)
     {
-        // Get the playout delay
+        // Get the playout delay.
         _sndCardPlayDelay = (uint32_t) (LatencyUsecs(_playStream) / 1000);
     }
 
     if (_recordBufferUsed > 0)
     {
-        // Have to copy to the buffer until it is full
+        // Have to copy to the buffer until it is full.
         size_t copy = _recordBufferSize - _recordBufferUsed;
         if (size < copy)
         {
@@ -2489,36 +2491,37 @@ int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
 
         if (_recordBufferUsed != _recordBufferSize)
         {
-            // Not enough data yet to pass to VoE
+            // Not enough data yet to pass to VoE.
             return 0;
         }
 
-        // Provide data to VoiceEngine
+        // Provide data to VoiceEngine.
         if (ProcessRecordedData(_recBuffer, numRecSamples, recDelay) == -1)
         {
-            // We have stopped recording
+            // We have stopped recording.
             return -1;
         }
 
         _recordBufferUsed = 0;
     }
 
-    // Now process full 10ms sample sets directly from the input
+    // Now process full 10ms sample sets directly from the input.
     while (size >= _recordBufferSize)
     {
-        // Provide data to VoiceEngine
+        // Provide data to VoiceEngine.
         if (ProcessRecordedData(
             static_cast<int8_t *> (const_cast<void *> (bufferData)),
             numRecSamples, recDelay) == -1)
         {
-            // We have stopped recording
+            // We have stopped recording.
             return -1;
         }
 
-        bufferData = static_cast<const char *> (bufferData) + _recordBufferSize;
+        bufferData = static_cast<const char *> (bufferData) +
+                     _recordBufferSize;
         size -= _recordBufferSize;
 
-        // We have consumed 10ms of data
+        // We have consumed 10ms of data.
         recDelay -= 10;
     }
 
@@ -2555,9 +2558,9 @@ int32_t AudioDeviceLinuxPulse::ProcessRecordedData(
     const uint32_t clockDrift(0);
     // TODO(andrew): this is a temporary hack, to avoid non-causal far- and
     // near-end signals at the AEC for PulseAudio. I think the system delay is
-    // being correctly calculated here, but for legacy reasons we add +10 ms to
-    // the value in the AEC. The real fix will be part of a larger investigation
-    // into managing system delay in the AEC.
+    // being correctly calculated here, but for legacy reasons we add +10 ms
+    // to the value in the AEC. The real fix will be part of a larger
+    // investigation into managing system delay in the AEC.
     if (recDelay > 10)
         recDelay -= 10;
     else
@@ -2565,12 +2568,12 @@ int32_t AudioDeviceLinuxPulse::ProcessRecordedData(
     _ptrAudioBuffer->SetVQEData(_sndCardPlayDelay, recDelay, clockDrift);
     _ptrAudioBuffer->SetTypingStatus(KeyPressed());
     // Deliver recorded samples at specified sample rate,
-    // mic level etc. to the observer using callback
+    // mic level etc. to the observer using callback.
     UnLock();
     _ptrAudioBuffer->DeliverRecordedData();
     Lock();
 
-    // We have been unlocked - check the flag again
+    // We have been unlocked - check the flag again.
     if (!_recording)
     {
         return -1;
@@ -2625,7 +2628,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
             return true;
     }
 
-    Lock();
+    CriticalSectionScoped lock(&_critSect);
 
     if (_startPlay)
     {
@@ -2718,7 +2721,6 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
         _playing = true;
         _playStartEvent.Set();
 
-        UnLock();
         return true;
     }
 
@@ -2742,10 +2744,10 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
 
             PaLock();
             if (LATE(pa_stream_write)(
-                                      _playStream,
-                                      (void *) &_playBuffer[_playbackBufferUnused],
-                                      write, NULL, (int64_t) 0,
-                                      PA_SEEK_RELATIVE) != PA_OK)
+                _playStream,
+                (void *) &_playBuffer[_playbackBufferUnused],
+                write, NULL, (int64_t) 0,
+                PA_SEEK_RELATIVE) != PA_OK)
             {
                 _writeErrors++;
                 if (_writeErrors > 10)
@@ -2756,7 +2758,8 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
                                      kTraceUtility, _id,
                                      "  pending playout error exists");
                     }
-                    _playError = 1; // Triggers callback from module process thread
+                    // Triggers callback from module process thread.
+                    _playError = 1;
                     WEBRTC_TRACE(
                                  kTraceError,
                                  kTraceUtility,
@@ -2775,11 +2778,12 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
         }
 
         uint32_t numPlaySamples = _playbackBufferSize / (2 * _playChannels);
-        if (_tempBufferSpace > 0) // Might have been reduced to zero by the above
+        // Might have been reduced to zero by the above.
+        if (_tempBufferSpace > 0)
         {
-            // Ask for new PCM data to be played out using the AudioDeviceBuffer
-            // ensure that this callback is executed without taking the
-            // audio-thread lock
+            // Ask for new PCM data to be played out using the
+            // AudioDeviceBuffer ensure that this callback is executed
+            // without taking the audio-thread lock.
             UnLock();
             WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
                          "  requesting data");
@@ -2787,10 +2791,9 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
                 _ptrAudioBuffer->RequestPlayoutData(numPlaySamples);
             Lock();
 
-            // We have been unlocked - check the flag again
+            // We have been unlocked - check the flag again.
             if (!_playing)
             {
-                UnLock();
                 return true;
             }
 
@@ -2824,7 +2827,8 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
                                      kTraceUtility, _id,
                                      "  pending playout error exists");
                     }
-                    _playError = 1; // triggers callback from module process thread
+                     // Triggers callback from module process thread.
+                    _playError = 1;
                     WEBRTC_TRACE(
                                  kTraceError,
                                  kTraceUtility,
@@ -2848,7 +2852,6 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
 
     }  // _playing
 
-    UnLock();
     return true;
 }
 
@@ -2866,7 +2869,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
             return true;
     }
 
-    Lock();
+    CriticalSectionScoped lock(&_critSect);
 
     if (_startRec)
     {
@@ -2890,10 +2893,10 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
                      "  connecting stream");
 
         // Connect the stream to a source
-        if (LATE(pa_stream_connect_record)(_recStream, _recDeviceName,
-                                           &_recBufferAttr,
-                                           (pa_stream_flags_t) _recStreamFlags)
-            != PA_OK)
+        if (LATE(pa_stream_connect_record)(_recStream,
+            _recDeviceName,
+            &_recBufferAttr,
+            (pa_stream_flags_t) _recStreamFlags) != PA_OK)
         {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "  failed to connect rec stream, err=%d",
@@ -2928,7 +2931,6 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
         _recording = true;
         _recStartEvent.Set();
 
-        UnLock();
         return true;
     }
 
@@ -2937,7 +2939,6 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
         // Read data and provide it to VoiceEngine
         if (ReadRecordedData(_tempSampleData, _tempSampleDataSize) == -1)
         {
-            UnLock();
             return true;
         }
 
@@ -2983,7 +2984,6 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
             // Read data and provide it to VoiceEngine
             if (ReadRecordedData(sampleData, sampleDataSize) == -1)
             {
-                UnLock();
                 return true;
             }
             PaLock();
@@ -2996,11 +2996,11 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
 
     }  // _recording
 
-    UnLock();
     return true;
 }
 
 bool AudioDeviceLinuxPulse::KeyPressed() const{
+
 #ifdef USE_X11
   char szKey[32];
   unsigned int i = 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
index 4324c5ea8c..15ae21ab45 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -11,10 +11,11 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #ifdef USE_X11
 #include <X11/Xlib.h>
@@ -206,18 +207,16 @@ public:
     // CPU load
     int32_t CPULoad(uint16_t& load) const override;
 
-public:
- bool PlayoutWarning() const override;
- bool PlayoutError() const override;
- bool RecordingWarning() const override;
- bool RecordingError() const override;
- void ClearPlayoutWarning() override;
- void ClearPlayoutError() override;
- void ClearRecordingWarning() override;
- void ClearRecordingError() override;
+    bool PlayoutWarning() const override;
+    bool PlayoutError() const override;
+    bool RecordingWarning() const override;
+    bool RecordingError() const override;
+    void ClearPlayoutWarning() override;
+    void ClearPlayoutError() override;
+    void ClearRecordingWarning() override;
+    void ClearRecordingError() override;
 
-public:
- void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
 private:
     void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) {
@@ -229,10 +228,8 @@ private:
     void WaitForOperationCompletion(pa_operation* paOperation) const;
     void WaitForSuccess(pa_operation* paOperation) const;
 
-private:
     bool KeyPressed() const;
 
-private:
     static void PaContextStateCallback(pa_context *c, void *pThis);
     static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
                                    int eol, void *pThis);
@@ -281,7 +278,6 @@ private:
     bool RecThreadProcess();
     bool PlayThreadProcess();
 
-private:
     AudioDeviceBuffer* _ptrAudioBuffer;
 
     CriticalSectionWrapper& _critSect;
@@ -290,8 +286,9 @@ private:
     EventWrapper& _recStartEvent;
     EventWrapper& _playStartEvent;
 
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
+    // TODO(pbos): Remove scoped_ptr and use directly without resetting.
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
     int32_t _id;
 
     AudioMixerManagerLinuxPulse _mixerManager;
@@ -307,7 +304,12 @@ private:
 
     AudioDeviceModule::BufferType _playBufType;
 
-private:
+    // Stores thread ID in constructor.
+    // We can then use ThreadChecker::CalledOnValidThread() to ensure that
+    // other methods are called from the same thread.
+    // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
+    rtc::ThreadChecker thread_checker_;
+
     bool _initialized;
     bool _recording;
     bool _playing;
@@ -320,7 +322,6 @@ private:
     bool _AGC;
     bool update_speaker_volume_at_startup_;
 
-private:
     uint16_t _playBufDelayFixed; // fixed playback delay
 
     uint32_t _sndCardPlayDelay;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_utility_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_utility_linux.cc
deleted file mode 100644
index fdd7d14167..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_utility_linux.cc
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/linux/audio_device_utility_linux.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc
-{
-
-AudioDeviceUtilityLinux::AudioDeviceUtilityLinux(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
-}
-
-AudioDeviceUtilityLinux::~AudioDeviceUtilityLinux()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
-    {
-        CriticalSectionScoped lock(&_critSect);
-
-        // free stuff here...
-    }
-
-    delete &_critSect;
-}
-
-// ============================================================================
-//                                     API
-// ============================================================================
-
-
-int32_t AudioDeviceUtilityLinux::Init()
-{
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                 "  OS info: %s", "Linux");
-
-    return 0;
-}
-
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_utility_linux.h b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_utility_linux.h
deleted file mode 100644
index f29e211ac9..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_utility_linux.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
-
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityLinux: public AudioDeviceUtility
-{
-public:
-    AudioDeviceUtilityLinux(const int32_t id);
-    virtual ~AudioDeviceUtilityLinux();
-
-    int32_t Init() override;
-
-private:
-    CriticalSectionWrapper& _critSect;
-    int32_t _id;
-};
-
-}  // namespace webrtc
-
-#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_LINUX_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
index df9dcabea5..29620eb043 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
@@ -11,7 +11,7 @@
 #include <assert.h>
 
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 extern webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
index 21618e2f83..b8be8c1b70 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/linux/alsasymboltable_linux.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 #include <alsa/asoundlib.h>
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
index 1ee94cd42f..f73a399c9d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
@@ -11,23 +11,37 @@
 #include <assert.h>
 
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/base/checks.h"
 
 extern webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
 
 // Accesses Pulse functions through our late-binding symbol table instead of
-// directly. This way we don't have to link to libpulse, which means our binary
-// will work on systems that don't have it.
+// directly. This way we don't have to link to libpulse, which means our
+// binary will work on systems that don't have it.
 #define LATE(sym) \
-  LATESYM_GET(webrtc_adm_linux_pulse::PulseAudioSymbolTable, &PaSymbolTable, sym)
+  LATESYM_GET(webrtc_adm_linux_pulse::PulseAudioSymbolTable, \
+              &PaSymbolTable, sym)
 
 namespace webrtc
 {
 
-enum { kMaxRetryOnFailure = 2 };
+class AutoPulseLock {
+ public:
+  explicit AutoPulseLock(pa_threaded_mainloop* pa_mainloop)
+      : pa_mainloop_(pa_mainloop) {
+    LATE(pa_threaded_mainloop_lock)(pa_mainloop_);
+  }
+
+  ~AutoPulseLock() {
+    LATE(pa_threaded_mainloop_unlock)(pa_mainloop_);
+  }
+
+ private:
+  pa_threaded_mainloop* const pa_mainloop_;
+};
 
 AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _id(id),
     _paOutputDeviceIndex(-1),
     _paInputDeviceIndex(-1),
@@ -41,8 +55,7 @@ AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse(const int32_t id) :
     _paSpeakerMute(false),
     _paSpeakerVolume(PA_VOLUME_NORM),
     _paChannels(0),
-    _paObjectsSet(false),
-    _callbackValues(false)
+    _paObjectsSet(false)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
                  "%s constructed", __FUNCTION__);
@@ -50,27 +63,25 @@ AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse(const int32_t id) :
 
 AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse()
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
                  "%s destructed", __FUNCTION__);
 
     Close();
-
-    delete &_critSect;
 }
 
-// ============================================================================
+// ===========================================================================
 //                                    PUBLIC METHODS
-// ============================================================================
+// ===========================================================================
 
 int32_t AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
     pa_threaded_mainloop* mainloop,
     pa_context* context)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
-
     if (!mainloop || !context)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -90,11 +101,10 @@ int32_t AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
 
 int32_t AudioMixerManagerLinuxPulse::Close()
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
-
     CloseSpeaker();
     CloseMicrophone();
 
@@ -108,11 +118,10 @@ int32_t AudioMixerManagerLinuxPulse::Close()
 
 int32_t AudioMixerManagerLinuxPulse::CloseSpeaker()
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
-
     // Reset the index to -1
     _paOutputDeviceIndex = -1;
     _paPlayStream = NULL;
@@ -122,11 +131,10 @@ int32_t AudioMixerManagerLinuxPulse::CloseSpeaker()
 
 int32_t AudioMixerManagerLinuxPulse::CloseMicrophone()
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
-
     // Reset the index to -1
     _paInputDeviceIndex = -1;
     _paRecStream = NULL;
@@ -136,20 +144,20 @@ int32_t AudioMixerManagerLinuxPulse::CloseMicrophone()
 
 int32_t AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)");
 
-    CriticalSectionScoped lock(&_critSect);
     _paPlayStream = playStream;
     return 0;
 }
 
 int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::SetRecStream(recStream)");
 
-    CriticalSectionScoped lock(&_critSect);
     _paRecStream = recStream;
     return 0;
 }
@@ -157,12 +165,11 @@ int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream)
 int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(
     uint16_t deviceIndex)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex=%d)",
                  deviceIndex);
 
-    CriticalSectionScoped lock(&_critSect);
-
     // No point in opening the speaker
     // if PA objects have not been set
     if (!_paObjectsSet)
@@ -185,11 +192,10 @@ int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(
 int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(
     uint16_t deviceIndex)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex=%d)",
-                 deviceIndex);
-
-    CriticalSectionScoped lock(&_critSect);
+                 "AudioMixerManagerLinuxPulse::OpenMicrophone"
+                 "(deviceIndex=%d)", deviceIndex);
 
     // No point in opening the microphone
     // if PA objects have not been set
@@ -212,6 +218,7 @@ int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(
 
 bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
@@ -220,6 +227,7 @@ bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const
 
 bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
@@ -229,12 +237,11 @@ bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const
 int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(
     uint32_t volume)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume=%u)",
                  volume);
 
-    CriticalSectionScoped lock(&_critSect);
-
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -248,7 +255,7 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(
         != PA_STREAM_UNCONNECTED))
     {
         // We can only really set the volume if we have a connected stream
-        PaLock();
+        AutoPulseLock auto_lock(_paMainloop);
 
         // Get the number of channels from the sample specification
         const pa_sample_spec *spec =
@@ -257,7 +264,6 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(
         {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "  could not get sample specification");
-            PaUnLock();
             return -1;
         }
 
@@ -278,8 +284,6 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(
 
         // Don't need to wait for the completion
         LATE(pa_operation_unref)(paOperation);
-
-        PaUnLock();
     } else
     {
         // We have not created a stream or it's not connected to the sink
@@ -302,7 +306,6 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(
 int32_t
 AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const
 {
-
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -317,15 +320,16 @@ AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const
         if (!GetSinkInputInfo())
           return -1;
 
+        AutoPulseLock auto_lock(_paMainloop);
         volume = static_cast<uint32_t> (_paVolume);
-        ResetCallbackVariables();
     } else
     {
+        AutoPulseLock auto_lock(_paMainloop);
         volume = _paSpeakerVolume;
     }
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerLinuxPulse::SpeakerVolume() => vol=%i",
+                 "\tAudioMixerManagerLinuxPulse::SpeakerVolume() => vol=%i",
                  volume);
 
     return 0;
@@ -368,7 +372,7 @@ AudioMixerManagerLinuxPulse::MinSpeakerVolume(uint32_t& minVolume) const
 int32_t
 AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize(uint16_t& stepSize) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -381,11 +385,8 @@ AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize(uint16_t& stepSize) const
     stepSize = 1;
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize() => "
-                 "size=%i, stepSize");
-
-    // Reset members modified by callback
-    ResetCallbackVariables();
+                 "\tAudioMixerManagerLinuxPulse::SpeakerVolumeStepSize() => "
+                 "size=%i", stepSize);
 
     return 0;
 }
@@ -393,6 +394,7 @@ AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize(uint16_t& stepSize) const
 int32_t
 AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -409,6 +411,7 @@ AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
 int32_t
 AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -424,12 +427,11 @@ AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available)
 
 int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable=%u)",
                  enable);
 
-    CriticalSectionScoped lock(&_critSect);
-
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -443,7 +445,7 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable)
         != PA_STREAM_UNCONNECTED))
     {
         // We can only really mute if we have a connected stream
-        PaLock();
+        AutoPulseLock auto_lock(_paMainloop);
 
         pa_operation* paOperation = NULL;
         paOperation = LATE(pa_context_set_sink_input_mute)(
@@ -459,8 +461,6 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable)
 
         // Don't need to wait for the completion
         LATE(pa_operation_unref)(paOperation);
-
-        PaUnLock();
     } else
     {
         // We have not created a stream or it's not connected to the sink
@@ -497,7 +497,6 @@ int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const
           return -1;
 
         enabled = static_cast<bool> (_paMute);
-        ResetCallbackVariables();
     } else
     {
         enabled = _paSpeakerMute;
@@ -513,6 +512,7 @@ int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const
 int32_t
 AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paOutputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -522,33 +522,31 @@ AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available)
 
     uint32_t deviceIndex = (uint32_t) _paOutputDeviceIndex;
 
-    PaLock();
-
-    // Get the actual stream device index if we have a connected stream
-    // The device used by the stream can be changed
-    // during the call
-    if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
-        != PA_STREAM_UNCONNECTED))
     {
-        deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
-    }
+        AutoPulseLock auto_lock(_paMainloop);
 
-    PaUnLock();
+        // Get the actual stream device index if we have a connected stream
+        // The device used by the stream can be changed
+        // during the call
+        if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
+            != PA_STREAM_UNCONNECTED))
+        {
+            deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
+        }
+    }
 
     if (!GetSinkInfoByIndex(deviceIndex))
       return -1;
 
     available = static_cast<bool> (_paChannels == 2);
 
-    // Reset members modified by callback
-    ResetCallbackVariables();
-
     return 0;
 }
 
 int32_t
 AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -558,7 +556,7 @@ AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
 
     uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
 
-    PaLock();
+    AutoPulseLock auto_lock(_paMainloop);
 
     // Get the actual stream device index if we have a connected stream
     // The device used by the stream can be changed
@@ -570,7 +568,6 @@ AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
     }
 
     pa_operation* paOperation = NULL;
-    ResetCallbackVariables();
 
     // Get info for this source
     // We want to know if the actual device can record in stereo
@@ -580,31 +577,20 @@ AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
         (void*) this);
 
     WaitForOperationCompletion(paOperation);
-    PaUnLock();
-
-    if (!_callbackValues)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Error getting number of input channels: %d",
-                     LATE(pa_context_errno)(_paContext));
-        return -1;
-    }
 
     available = static_cast<bool> (_paChannels == 2);
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
+                 " AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
                  " => available=%i, available");
 
-    // Reset members modified by callback
-    ResetCallbackVariables();
-
     return 0;
 }
 
 int32_t AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
     bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -620,12 +606,11 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
 
 int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable=%u)",
                  enable);
 
-    CriticalSectionScoped lock(&_critSect);
-
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -635,11 +620,10 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
 
     bool setFailed(false);
     pa_operation* paOperation = NULL;
-    ResetCallbackVariables();
 
     uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
 
-    PaLock();
+    AutoPulseLock auto_lock(_paMainloop);
 
     // Get the actual stream device index if we have a connected stream
     // The device used by the stream can be changed
@@ -664,11 +648,6 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
     // Don't need to wait for this to complete.
     LATE(pa_operation_unref)(paOperation);
 
-    PaUnLock();
-
-    // Reset variables altered by callback
-    ResetCallbackVariables();
-
     if (setFailed)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -682,7 +661,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
 
 int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -692,30 +671,26 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const
 
     uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
 
-    PaLock();
-
-    // Get the actual stream device index if we have a connected stream
-    // The device used by the stream can be changed
-    // during the call
-    if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
-        != PA_STREAM_UNCONNECTED))
     {
-        deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+        AutoPulseLock auto_lock(_paMainloop);
+        // Get the actual stream device index if we have a connected stream
+        // The device used by the stream can be changed
+        // during the call
+        if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+            != PA_STREAM_UNCONNECTED))
+        {
+            deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+        }
     }
 
-    PaUnLock();
-
     if (!GetSourceInfoByIndex(deviceIndex))
       return -1;
 
     enabled = static_cast<bool> (_paMute);
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerLinuxPulse::MicrophoneMute() =>"
-                 " enabled=%i, enabled");
-
-    // Reset members modified by callback
-    ResetCallbackVariables();
+                 "\tAudioMixerManagerLinuxPulse::MicrophoneMute() =>"
+                 " enabled=%i", enabled);
 
     return 0;
 }
@@ -723,6 +698,7 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const
 int32_t
 AudioMixerManagerLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -740,12 +716,11 @@ AudioMixerManagerLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
 
 int32_t AudioMixerManagerLinuxPulse::SetMicrophoneBoost(bool enable)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxPulse::SetMicrophoneBoost(enable=%u)",
                  enable);
 
-    CriticalSectionScoped lock(&_critSect);
-
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -753,7 +728,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneBoost(bool enable)
         return -1;
     }
 
-    // Ensure that the selected microphone destination has a valid boost control
+    // Ensure the selected microphone destination has a valid boost control
     bool available(false);
     MicrophoneBoostIsAvailable(available);
     if (!available)
@@ -770,7 +745,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneBoost(bool enable)
 
 int32_t AudioMixerManagerLinuxPulse::MicrophoneBoost(bool& enabled) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -787,6 +762,7 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneBoost(bool& enabled) const
 int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
     bool& available)
 {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -804,10 +780,8 @@ int32_t
 AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=%u)",
-                 volume);
-
-    CriticalSectionScoped lock(&_critSect);
+                 "AudioMixerManagerLinuxPulse::SetMicrophoneVolume"
+                 "(volume=%u)", volume);
 
     if (_paInputDeviceIndex == -1)
     {
@@ -816,22 +790,20 @@ AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
         return -1;
     }
 
-    // Unlike output streams, input streams have no concept of a stream volume,
-    // only a device volume. So we have to change the volume of the device
-    // itself.
+    // Unlike output streams, input streams have no concept of a stream
+    // volume, only a device volume. So we have to change the volume of the
+    // device itself.
 
     // The device may have a different number of channels than the stream and
-    // their mapping may be different, so we don't want to use the channel count
-    // from our sample spec. We could use PA_CHANNELS_MAX to cover our bases,
-    // and the server allows that even if the device's channel count is lower,
-    // but some buggy PA clients don't like that (the pavucontrol on Hardy dies
-    // in an assert if the channel count is different). So instead we look up
-    // the actual number of channels that the device has.
-
+    // their mapping may be different, so we don't want to use the channel
+    // count from our sample spec. We could use PA_CHANNELS_MAX to cover our
+    // bases, and the server allows that even if the device's channel count
+    // is lower, but some buggy PA clients don't like that (the pavucontrol
+    // on Hardy dies in an assert if the channel count is different). So
+    // instead we look up the actual number of channels that the device has.
+    AutoPulseLock auto_lock(_paMainloop);
     uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
 
-    PaLock();
-
     // Get the actual stream device index if we have a connected stream
     // The device used by the stream can be changed
     // during the call
@@ -843,7 +815,6 @@ AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
 
     bool setFailed(false);
     pa_operation* paOperation = NULL;
-    ResetCallbackVariables();
 
     // Get the number of channels for this source
     paOperation
@@ -853,18 +824,7 @@ AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
 
     WaitForOperationCompletion(paOperation);
 
-    if (!_callbackValues)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Error getting input channels: %d",
-                     LATE(pa_context_errno)(_paContext));
-        PaUnLock();
-        return -1;
-    }
-
     uint8_t channels = _paChannels;
-    ResetCallbackVariables();
-
     pa_cvolume cVolumes;
     LATE(pa_cvolume_set)(&cVolumes, channels, volume);
 
@@ -872,7 +832,8 @@ AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
     paOperation
         = LATE(pa_context_set_source_volume_by_index)(_paContext, deviceIndex,
                                                       &cVolumes,
-                                                      PaSetVolumeCallback, NULL);
+                                                      PaSetVolumeCallback,
+                                                      NULL);
 
     if (!paOperation)
     {
@@ -882,11 +843,6 @@ AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume)
     // Don't need to wait for this to complete.
     LATE(pa_operation_unref)(paOperation);
 
-    PaUnLock();
-
-    // Reset variables altered by callback
-    ResetCallbackVariables();
-
     if (setFailed)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -911,29 +867,28 @@ AudioMixerManagerLinuxPulse::MicrophoneVolume(uint32_t& volume) const
 
     uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
 
-    PaLock();
-
-    // Get the actual stream device index if we have a connected stream
-    // The device used by the stream can be changed
-    // during the call
-    if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
-        != PA_STREAM_UNCONNECTED))
     {
-        deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+      AutoPulseLock auto_lock(_paMainloop);
+      // Get the actual stream device index if we have a connected stream.
+      // The device used by the stream can be changed during the call.
+      if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
+          != PA_STREAM_UNCONNECTED))
+      {
+          deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+      }
     }
 
-    PaUnLock();
-
     if (!GetSourceInfoByIndex(deviceIndex))
-      return -1;
+        return -1;
 
-    volume = static_cast<uint32_t> (_paVolume);
+    {
+        AutoPulseLock auto_lock(_paMainloop);
+        volume = static_cast<uint32_t> (_paVolume);
+    }
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol=%i, volume");
-
-    // Reset members modified by callback
-    ResetCallbackVariables();
+                 "     AudioMixerManagerLinuxPulse::MicrophoneVolume()"
+                 " => vol=%i, volume");
 
     return 0;
 }
@@ -976,7 +931,7 @@ AudioMixerManagerLinuxPulse::MinMicrophoneVolume(uint32_t& minVolume) const
 int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize(
     uint16_t& stepSize) const
 {
-
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
     if (_paInputDeviceIndex == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -986,7 +941,7 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize(
 
     uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
 
-    PaLock();
+    AutoPulseLock auto_lock(_paMainloop);
 
     // Get the actual stream device index if we have a connected stream
     // The device used by the stream can be changed
@@ -998,7 +953,6 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize(
     }
 
     pa_operation* paOperation = NULL;
-    ResetCallbackVariables();
 
     // Get info for this source
     paOperation
@@ -1008,60 +962,55 @@ int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize(
 
     WaitForOperationCompletion(paOperation);
 
-    PaUnLock();
-
-    if (!_callbackValues)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Error getting step size: %d",
-                     LATE(pa_context_errno)(_paContext));
-        return -1;
-    }
-
     stepSize = static_cast<uint16_t> ((PA_VOLUME_NORM + 1) / _paVolSteps);
 
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize()"
-                 " => size=%i, stepSize");
-
-    // Reset members modified by callback
-    ResetCallbackVariables();
+                 "\tAudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize()"
+                 " => size=%i", stepSize);
 
     return 0;
 }
 
-// ============================================================================
+// ===========================================================================
 //                                 Private Methods
-// ============================================================================
+// ===========================================================================
 
-void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
-                                                     const pa_sink_info *i,
-                                                     int eol, void *pThis)
+void
+AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
+                                                const pa_sink_info *i,
+                                                int eol,
+                                                void *pThis)
 {
-    static_cast<AudioMixerManagerLinuxPulse*> (pThis)-> PaSinkInfoCallbackHandler(
-        i, eol);
+    static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
+        PaSinkInfoCallbackHandler(i, eol);
 }
 
-void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
+void
+AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
     pa_context */*c*/,
     const pa_sink_input_info *i,
-    int eol, void *pThis)
+    int eol,
+    void *pThis)
 {
     static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
         PaSinkInputInfoCallbackHandler(i, eol);
 }
 
 
-void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
-                                                       const pa_source_info *i,
-                                                       int eol, void *pThis)
+void
+AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
+                                                  const pa_source_info *i,
+                                                  int eol,
+                                                  void *pThis)
 {
     static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
         PaSourceInfoCallbackHandler(i, eol);
 }
 
-void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context * c,
-                                                      int success, void */*pThis*/)
+void
+AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context * c,
+                                                 int success,
+                                                 void */*pThis*/)
 {
     if (!success)
     {
@@ -1081,7 +1030,6 @@ void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
         return;
     }
 
-    _callbackValues = true;
     _paChannels = i->channel_map.channels; // Get number of channels
     pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
     for (int j = 0; j < _paChannels; ++j)
@@ -1111,7 +1059,6 @@ void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
         return;
     }
 
-    _callbackValues = true;
     _paChannels = i->channel_map.channels; // Get number of channels
     pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
     for (int j = 0; j < _paChannels; ++j)
@@ -1136,7 +1083,6 @@ void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
         return;
     }
 
-    _callbackValues = true;
     _paChannels = i->channel_map.channels; // Get number of channels
     pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
     for (int j = 0; j < _paChannels; ++j)
@@ -1155,15 +1101,6 @@ void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
     _paVolSteps = PA_VOLUME_NORM + 1;
 }
 
-void AudioMixerManagerLinuxPulse::ResetCallbackVariables() const
-{
-    _paVolume = 0;
-    _paMute = 0;
-    _paVolSteps = 0;
-    _paChannels = 0;
-    _callbackValues = false;
-}
-
 void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
     pa_operation* paOperation) const
 {
@@ -1175,92 +1112,42 @@ void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
     LATE(pa_operation_unref)(paOperation);
 }
 
-void AudioMixerManagerLinuxPulse::PaLock() const
-{
-    LATE(pa_threaded_mainloop_lock)(_paMainloop);
-}
-
-void AudioMixerManagerLinuxPulse::PaUnLock() const
-{
-    LATE(pa_threaded_mainloop_unlock)(_paMainloop);
-}
-
 bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
   pa_operation* paOperation = NULL;
-  ResetCallbackVariables();
 
-  PaLock();
-  for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
-       retries ++) {
-    // Get info for this stream (sink input).
-    paOperation = LATE(pa_context_get_sink_input_info)(
-        _paContext,
-        LATE(pa_stream_get_index)(_paPlayStream),
-        PaSinkInputInfoCallback,
-        (void*) this);
-
-    WaitForOperationCompletion(paOperation);
-  }
-  PaUnLock();
-
-  if (!_callbackValues) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "GetSinkInputInfo failed to get volume info : %d",
-                 LATE(pa_context_errno)(_paContext));
-    return false;
-  }
+  AutoPulseLock auto_lock(_paMainloop);
+  // Get info for this stream (sink input).
+  paOperation = LATE(pa_context_get_sink_input_info)(
+      _paContext,
+      LATE(pa_stream_get_index)(_paPlayStream),
+      PaSinkInputInfoCallback,
+      (void*) this);
 
+  WaitForOperationCompletion(paOperation);
   return true;
 }
 
 bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(
     int device_index) const {
   pa_operation* paOperation = NULL;
-  ResetCallbackVariables();
 
-  PaLock();
-  for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
-       retries ++) {
-    paOperation = LATE(pa_context_get_sink_info_by_index)(_paContext,
-        device_index, PaSinkInfoCallback, (void*) this);
-
-    WaitForOperationCompletion(paOperation);
-  }
-  PaUnLock();
-
-  if (!_callbackValues) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "GetSinkInfoByIndex failed to get volume info: %d",
-                 LATE(pa_context_errno)(_paContext));
-    return false;
-  }
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_sink_info_by_index)(_paContext,
+      device_index, PaSinkInfoCallback, (void*) this);
 
+  WaitForOperationCompletion(paOperation);
   return true;
 }
 
 bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(
     int device_index) const {
   pa_operation* paOperation = NULL;
-  ResetCallbackVariables();
 
-  PaLock();
-  for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
-       retries ++) {
+  AutoPulseLock auto_lock(_paMainloop);
   paOperation  = LATE(pa_context_get_source_info_by_index)(
       _paContext, device_index, PaSourceInfoCallback, (void*) this);
 
   WaitForOperationCompletion(paOperation);
-  }
-
-  PaUnLock();
-
-  if (!_callbackValues) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "GetSourceInfoByIndex error: %d",
-                 LATE(pa_context_errno)(_paContext));
-    return false;
-  }
-
   return true;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
index 9296c76488..025fdc5868 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
@@ -13,8 +13,9 @@
 
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/base/thread_checker.h"
 
 #include <pulse/pulseaudio.h>
 #include <stdint.h>
@@ -82,17 +83,13 @@ private:
     void PaSinkInputInfoCallbackHandler(const pa_sink_input_info *i, int eol);
     void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
 
-    void ResetCallbackVariables() const;
     void WaitForOperationCompletion(pa_operation* paOperation) const;
-    void PaLock() const;
-    void PaUnLock() const;
 
     bool GetSinkInputInfo() const;
     bool GetSinkInfoByIndex(int device_index)const ;
     bool GetSourceInfoByIndex(int device_index) const;
 
 private:
-    CriticalSectionWrapper& _critSect;
     int32_t _id;
     int16_t _paOutputDeviceIndex;
     int16_t _paInputDeviceIndex;
@@ -110,7 +107,12 @@ private:
     mutable uint32_t _paSpeakerVolume;
     mutable uint8_t _paChannels;
     bool _paObjectsSet;
-    mutable bool _callbackValues;
+
+    // Stores thread ID in constructor.
+    // We can then use ThreadChecker::CalledOnValidThread() to ensure that
+    // other methods are called from the same thread.
+    // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
+    rtc::ThreadChecker thread_checker_;
 };
 
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h b/media/webrtc/trunk/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
index 2daf9945c8..131d473da5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
@@ -16,7 +16,7 @@
 #include <string.h>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 // This file provides macros for creating "symbol table" classes to simplify the
 // dynamic loading of symbols from DLLs. Currently the implementation only
@@ -119,7 +119,7 @@ class LateBindingSymbolTable {
   bool undefined_symbols_;
   void *symbols_[SYMBOL_TABLE_SIZE];
 
-  DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
 };
 
 // This macro must be invoked in a header to declare a symbol table class.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.cc b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.cc
index d0577c8a7f..49bfd136e5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.cc
@@ -10,3234 +10,2765 @@
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
 #include "webrtc/modules/audio_device/mac/audio_device_mac.h"
 #include "webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <ApplicationServices/ApplicationServices.h>
-#include <libkern/OSAtomic.h>   // OSAtomicCompareAndSwap()
-#include <mach/mach.h>          // mach_task_self()
-#include <sys/sysctl.h>         // sysctlbyname()
-
+#include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
+#include <mach/mach.h>         // mach_task_self()
+#include <sys/sysctl.h>        // sysctlbyname()
 #ifdef MOZILLA_INTERNAL_API
 #include <OSXRunLoopSingleton.h>
 #endif
 
-namespace webrtc
-{
+namespace webrtc {
 
-#define WEBRTC_CA_RETURN_ON_ERR(expr)                                   \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,               \
-                "Error in " #expr, (const char *)&err);                 \
-            return -1;                                                  \
-        }                                                               \
-    } while(0)
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                  \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+      return -1;                                                       \
+    }                                                                  \
+  } while (0)
 
-#define WEBRTC_CA_LOG_ERR(expr)                                         \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,               \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
+#define WEBRTC_CA_LOG_ERR(expr)                                        \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+    }                                                                  \
+  } while (0)
 
-#define WEBRTC_CA_LOG_WARN(expr)                                        \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceWarning, kTraceAudioDevice, _id,             \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
+#define WEBRTC_CA_LOG_WARN(expr)                                         \
+  do {                                                                   \
+    err = expr;                                                          \
+    if (err != noErr) {                                                  \
+      logCAMsg(kTraceWarning, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                     \
+    }                                                                    \
+  } while (0)
 
-enum
-{
-    MaxNumberDevices = 64
-};
+enum { MaxNumberDevices = 64 };
 
-void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue)
-{
-    while (1)
-    {
-        int32_t oldValue = *theValue;
-        if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue)
-            == true)
-        {
-            return;
-        }
+void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue) {
+  while (1) {
+    int32_t oldValue = *theValue;
+    if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue) == true) {
+      return;
     }
+  }
 }
 
-int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue)
-{
-    while (1)
-    {
-        int32_t value = *theValue;
-        if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true)
-        {
-            return value;
-        }
+int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue) {
+  while (1) {
+    int32_t value = *theValue;
+    if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true) {
+      return value;
     }
+  }
 }
 
 // CoreAudio errors are best interpreted as four character strings.
 void AudioDeviceMac::logCAMsg(const TraceLevel level,
                               const TraceModule module,
-                              const int32_t id, const char *msg,
-                              const char *err)
-{
-    DCHECK(msg != NULL);
-    DCHECK(err != NULL);
+                              const int32_t id,
+                              const char* msg,
+                              const char* err) {
+  RTC_DCHECK(msg != NULL);
+  RTC_DCHECK(err != NULL);
 
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+  WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
-    // We need to flip the characters in this case.
-    WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
-        + 2, err + 1, err);
+  // We need to flip the characters in this case.
+  WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err + 2,
+               err + 1, err);
 #endif
 }
 
-AudioDeviceMac::AudioDeviceMac(const int32_t id) :
-    _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _stopEventRec(*EventWrapper::Create()),
-    _stopEvent(*EventWrapper::Create()),
-    _id(id),
-    _mixerManager(id),
-    _inputDeviceIndex(0),
-    _outputDeviceIndex(0),
-    _inputDeviceID(kAudioObjectUnknown),
-    _outputDeviceID(kAudioObjectUnknown),
-    _inputDeviceIsSpecified(false),
-    _outputDeviceIsSpecified(false),
-    _recChannels(N_REC_CHANNELS),
-    _playChannels(N_PLAY_CHANNELS),
-    _captureBufData(NULL),
-    _renderBufData(NULL),
-    _playBufType(AudioDeviceModule::kFixedBufferSize),
-    _initialized(false),
-    _isShutDown(false),
-    _recording(false),
-    _playing(false),
-    _recIsInitialized(false),
-    _playIsInitialized(false),
-    _AGC(false),
-    _renderDeviceIsAlive(1),
-    _captureDeviceIsAlive(1),
-    _twoDevices(true),
-    _doStop(false),
-    _doStopRec(false),
-    _macBookPro(false),
-    _macBookProPanRight(false),
-    _captureLatencyUs(0),
-    _renderLatencyUs(0),
-    _captureDelayUs(0),
-    _renderDelayUs(0),
-    _renderDelayOffsetSamples(0),
-    _playBufDelayFixed(20),
-    _playWarning(0),
-    _playError(0),
-    _recWarning(0),
-    _recError(0),
-    _paCaptureBuffer(NULL),
-    _paRenderBuffer(NULL),
-    _captureBufSizeSamples(0),
-    _renderBufSizeSamples(0),
-    prev_key_state_()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
+AudioDeviceMac::AudioDeviceMac(const int32_t id)
+    : _ptrAudioBuffer(NULL),
+      _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _stopEventRec(*EventWrapper::Create()),
+      _stopEvent(*EventWrapper::Create()),
+      _id(id),
+      _mixerManager(id),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      _recChannels(N_REC_CHANNELS),
+      _playChannels(N_PLAY_CHANNELS),
+      _captureBufData(NULL),
+      _renderBufData(NULL),
+      _playBufType(AudioDeviceModule::kFixedBufferSize),
+      _initialized(false),
+      _isShutDown(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _AGC(false),
+      _renderDeviceIsAlive(1),
+      _captureDeviceIsAlive(1),
+      _twoDevices(true),
+      _doStop(false),
+      _doStopRec(false),
+      _macBookPro(false),
+      _macBookProPanRight(false),
+      _captureLatencyUs(0),
+      _renderLatencyUs(0),
+      _captureDelayUs(0),
+      _renderDelayUs(0),
+      _renderDelayOffsetSamples(0),
+      _playBufDelayFixed(20),
+      _playWarning(0),
+      _playError(0),
+      _recWarning(0),
+      _recError(0),
+      _paCaptureBuffer(NULL),
+      _paRenderBuffer(NULL),
+      _captureBufSizeSamples(0),
+      _renderBufSizeSamples(0),
+      prev_key_state_(),
+      get_mic_volume_counter_ms_(0) {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
 
-    DCHECK(&_stopEvent != NULL);
-    DCHECK(&_stopEventRec != NULL);
+  RTC_DCHECK(&_stopEvent != NULL);
+  RTC_DCHECK(&_stopEventRec != NULL);
 
-    memset(_renderConvertData, 0, sizeof(_renderConvertData));
-    memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+  memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
 }
 
+AudioDeviceMac::~AudioDeviceMac() {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
+               __FUNCTION__);
 
-AudioDeviceMac::~AudioDeviceMac()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
+  if (!_isShutDown) {
+    Terminate();
+  }
 
-    if (!_isShutDown)
-    {
-        Terminate();
-    }
+  RTC_DCHECK(!capture_worker_thread_.get());
+  RTC_DCHECK(!render_worker_thread_.get());
 
-    DCHECK(!capture_worker_thread_.get());
-    DCHECK(!render_worker_thread_.get());
+  if (_paRenderBuffer) {
+    delete _paRenderBuffer;
+    _paRenderBuffer = NULL;
+  }
 
-    if (_paRenderBuffer)
-    {
-        delete _paRenderBuffer;
-        _paRenderBuffer = NULL;
-    }
+  if (_paCaptureBuffer) {
+    delete _paCaptureBuffer;
+    _paCaptureBuffer = NULL;
+  }
 
-    if (_paCaptureBuffer)
-    {
-        delete _paCaptureBuffer;
-        _paCaptureBuffer = NULL;
-    }
+  if (_renderBufData) {
+    delete[] _renderBufData;
+    _renderBufData = NULL;
+  }
 
-    if (_renderBufData)
-    {
-        delete[] _renderBufData;
-        _renderBufData = NULL;
-    }
+  if (_captureBufData) {
+    delete[] _captureBufData;
+    _captureBufData = NULL;
+  }
 
-    if (_captureBufData)
-    {
-        delete[] _captureBufData;
-        _captureBufData = NULL;
-    }
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_destroy() error: %d", kernErr);
+  }
 
-    kern_return_t kernErr = KERN_SUCCESS;
-    kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_destroy() error: %d", kernErr);
-    }
+  kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_destroy() error: %d", kernErr);
+  }
 
-    kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_destroy() error: %d", kernErr);
-    }
-
-    delete &_stopEvent;
-    delete &_stopEventRec;
-    delete &_critSect;
+  delete &_stopEvent;
+  delete &_stopEventRec;
+  delete &_critSect;
 }
 
 // ============================================================================
 //                                     API
 // ============================================================================
 
-void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
+void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  _ptrAudioBuffer = audioBuffer;
 
-    _ptrAudioBuffer = audioBuffer;
-
-    // inform the AudioBuffer about default settings for this implementation
-    _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
-    _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
-    _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+  // inform the AudioBuffer about default settings for this implementation
+  _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+  _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
 }
 
 int32_t AudioDeviceMac::ActiveAudioLayer(
-    AudioDeviceModule::AudioLayer& audioLayer) const
-{
-    audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
-    return 0;
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+  return 0;
 }
 
-int32_t AudioDeviceMac::Init()
-{
+int32_t AudioDeviceMac::Init() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (_initialized) {
+    return 0;
+  }
 
-    if (_initialized)
-    {
-        return 0;
+  OSStatus err = noErr;
+
+  _isShutDown = false;
+
+  // PortAudio ring buffers require an elementCount which is a power of two.
+  if (_renderBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
     }
+    _renderBufSizeSamples = powerOfTwo;
+    _renderBufData = new SInt16[_renderBufSizeSamples];
+  }
 
-    OSStatus err = noErr;
-
-    _isShutDown = false;
-
-    // PortAudio ring buffers require an elementCount which is a power of two.
-    if (_renderBufData == NULL)
-    {
-        UInt32 powerOfTwo = 1;
-        while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES)
-        {
-            powerOfTwo <<= 1;
-        }
-        _renderBufSizeSamples = powerOfTwo;
-        _renderBufData = new SInt16[_renderBufSizeSamples];
+  if (_paRenderBuffer == NULL) {
+    _paRenderBuffer = new PaUtilRingBuffer;
+    PaRingBufferSize bufSize = -1;
+    bufSize = PaUtil_InitializeRingBuffer(
+        _paRenderBuffer, sizeof(SInt16), _renderBufSizeSamples, _renderBufData);
+    if (bufSize == -1) {
+      WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                   " PaUtil_InitializeRingBuffer() error");
+      return -1;
     }
+  }
 
-    if (_paRenderBuffer == NULL)
-    {
-        _paRenderBuffer = new PaUtilRingBuffer;
-        PaRingBufferSize bufSize = -1;
-        bufSize = PaUtil_InitializeRingBuffer(_paRenderBuffer, sizeof(SInt16),
-                                              _renderBufSizeSamples,
-                                              _renderBufData);
-        if (bufSize == -1)
-        {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
-                         _id, " PaUtil_InitializeRingBuffer() error");
-            return -1;
-        }
+  if (_captureBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
     }
+    _captureBufSizeSamples = powerOfTwo;
+    _captureBufData = new Float32[_captureBufSizeSamples];
+  }
 
-    if (_captureBufData == NULL)
-    {
-        UInt32 powerOfTwo = 1;
-        while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES)
-        {
-            powerOfTwo <<= 1;
-        }
-        _captureBufSizeSamples = powerOfTwo;
-        _captureBufData = new Float32[_captureBufSizeSamples];
+  if (_paCaptureBuffer == NULL) {
+    _paCaptureBuffer = new PaUtilRingBuffer;
+    PaRingBufferSize bufSize = -1;
+    bufSize =
+        PaUtil_InitializeRingBuffer(_paCaptureBuffer, sizeof(Float32),
+                                    _captureBufSizeSamples, _captureBufData);
+    if (bufSize == -1) {
+      WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                   " PaUtil_InitializeRingBuffer() error");
+      return -1;
     }
+  }
 
-    if (_paCaptureBuffer == NULL)
-    {
-        _paCaptureBuffer = new PaUtilRingBuffer;
-        PaRingBufferSize bufSize = -1;
-        bufSize = PaUtil_InitializeRingBuffer(_paCaptureBuffer,
-                                              sizeof(Float32),
-                                              _captureBufSizeSamples,
-                                              _captureBufData);
-        if (bufSize == -1)
-        {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
-                         _id, " PaUtil_InitializeRingBuffer() error");
-            return -1;
-        }
-    }
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                 " semaphore_create() error: %d", kernErr);
+    return -1;
+  }
 
-    kern_return_t kernErr = KERN_SUCCESS;
-    kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
-                               SYNC_POLICY_FIFO, 0);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     " semaphore_create() error: %d", kernErr);
-        return -1;
-    }
+  kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                 " semaphore_create() error: %d", kernErr);
+    return -1;
+  }
 
-    kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
-                               SYNC_POLICY_FIFO, 0);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     " semaphore_create() error: %d", kernErr);
-        return -1;
-    }
-
-    // Setting RunLoop to NULL here instructs HAL to manage its own thread for
-    // notifications. This was the default behaviour on OS X 10.5 and earlier,
-    // but now must be explicitly specified. HAL would otherwise try to use the
-    // main thread to issue notifications.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyRunLoop,
-            kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
+  // Setting RunLoop to NULL here instructs HAL to manage its own thread for
+  // notifications. This was the default behaviour on OS X 10.5 and earlier,
+  // but now must be explicitly specified. HAL would otherwise try to use the
+  // main thread to issue notifications.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyRunLoop, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
 #ifdef MOZILLA_INTERNAL_API
-    mozilla_set_coreaudio_notification_runloop_if_needed();
+  mozilla_set_coreaudio_notification_runloop_if_needed();
 #else
-    CFRunLoopRef runLoop = NULL;
-    UInt32 size = sizeof(CFRunLoopRef);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, size, &runLoop));
+  CFRunLoopRef runLoop = NULL;
+  UInt32 size = sizeof(CFRunLoopRef);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, size, &runLoop));
+
+  // Listen for any device changes.
+  propertyAddress.mSelector = kAudioHardwarePropertyDevices;
+  WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
 #endif
 
-    // Listen for any device changes.
-    propertyAddress.mSelector = kAudioHardwarePropertyDevices;
-    WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(kAudioObjectSystemObject,
-            &propertyAddress, &objectListenerProc, this));
+  // Determine if this is a MacBook Pro
+  _macBookPro = false;
+  _macBookProPanRight = false;
+  char buf[128];
+  size_t length = sizeof(buf);
+  memset(buf, 0, length);
 
-    // Determine if this is a MacBook Pro
-    _macBookPro = false;
-    _macBookProPanRight = false;
-    char buf[128];
-    size_t length = sizeof(buf);
-    memset(buf, 0, length);
-
-    int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
-    if (intErr != 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Error in sysctlbyname(): %d", err);
-    } else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Hardware model: %s", buf);
-        if (strncmp(buf, "MacBookPro", 10) == 0)
-        {
-            _macBookPro = true;
-        }
+  int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
+  if (intErr != 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Error in sysctlbyname(): %d", err);
+  } else {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Hardware model: %s",
+                 buf);
+    if (strncmp(buf, "MacBookPro", 10) == 0) {
+      _macBookPro = true;
     }
+  }
 
-    _playWarning = 0;
-    _playError = 0;
-    _recWarning = 0;
-    _recError = 0;
+  _playWarning = 0;
+  _playError = 0;
+  _recWarning = 0;
+  _recError = 0;
 
-    _initialized = true;
+  get_mic_volume_counter_ms_ = 0;
 
+  _initialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::Terminate() {
+  if (!_initialized) {
     return 0;
+  }
+
+  if (_recording) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Recording must be stopped");
+    return -1;
+  }
+
+  if (_playing) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Playback must be stopped");
+    return -1;
+  }
+
+  _critSect.Enter();
+
+  _mixerManager.Close();
+
+  OSStatus err = noErr;
+  int retVal = 0;
+
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
+
+  err = AudioHardwareUnload();
+  if (err != noErr) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Error in AudioHardwareUnload()", (const char*)&err);
+    retVal = -1;
+  }
+
+  _isShutDown = true;
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  _critSect.Leave();
+
+  return retVal;
 }
 
-int32_t AudioDeviceMac::Terminate()
-{
-
-    if (!_initialized)
-    {
-        return 0;
-    }
-
-    if (_recording)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Recording must be stopped");
-        return -1;
-    }
-
-    if (_playing)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Playback must be stopped");
-        return -1;
-    }
-
-    _critSect.Enter();
-
-    _mixerManager.Close();
-
-    OSStatus err = noErr;
-    int retVal = 0;
-
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
-            &propertyAddress, &objectListenerProc, this));
-
-    err = AudioHardwareUnload();
-    if (err != noErr)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Error in AudioHardwareUnload()", (const char*) &err);
-        retVal = -1;
-    }
-
-    _isShutDown = true;
-    _initialized = false;
-    _outputDeviceIsSpecified = false;
-    _inputDeviceIsSpecified = false;
-
-    _critSect.Leave();
-
-    return retVal;
+bool AudioDeviceMac::Initialized() const {
+  return (_initialized);
 }
 
-bool AudioDeviceMac::Initialized() const
-{
-    return (_initialized);
-}
-
-int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        available = false;
-        return 0;
-    }
-
-    // Given that InitSpeaker was successful, we know that a valid speaker
-    // exists.
-    available = true;
-
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
+int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    available = false;
     return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a valid speaker
+  // exists.
+  available = true;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitSpeaker()
-{
+int32_t AudioDeviceMac::InitSpeaker() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (_playing) {
+    return -1;
+  }
 
-    if (_playing)
-    {
-        return -1;
-    }
+  if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1) {
+    return -1;
+  }
 
-    if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1)
-    {
-        return -1;
-    }
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
 
-    if (_inputDeviceID == _outputDeviceID)
-    {
-        _twoDevices = false;
-    } else
-    {
-        _twoDevices = true;
-    }
+  if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1) {
+    return -1;
+  }
 
-    if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1)
-    {
-        return -1;
-    }
+  return 0;
+}
 
+int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    available = false;
     return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a valid microphone
+  // exists.
+  available = true;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::InitMicrophone() {
+  CriticalSectionScoped lock(&_critSect);
 
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+  if (_recording) {
+    return -1;
+  }
 
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        available = false;
-        return 0;
-    }
+  if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1) {
+    return -1;
+  }
 
-    // Given that InitMicrophone was successful, we know that a valid microphone
-    // exists.
-    available = true;
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+  if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1) {
+    return -1;
+  }
 
+  return 0;
+}
+
+bool AudioDeviceMac::SpeakerIsInitialized() const {
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceMac::MicrophoneIsInitialized() const {
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
     return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a volume control exists
+  //
+  available = true;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitMicrophone()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_recording)
-    {
-        return -1;
-    }
-
-    if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1)
-    {
-        return -1;
-    }
-
-    if (_inputDeviceID == _outputDeviceID)
-    {
-        _twoDevices = false;
-    } else
-    {
-        _twoDevices = true;
-    }
-
-    if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1)
-    {
-        return -1;
-    }
-
-    return 0;
+int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume) {
+  return (_mixerManager.SetSpeakerVolume(volume));
 }
 
-bool AudioDeviceMac::SpeakerIsInitialized() const
-{
-    return (_mixerManager.SpeakerIsInitialized());
-}
+int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const {
+  uint32_t level(0);
 
-bool AudioDeviceMac::MicrophoneIsInitialized() const
-{
-    return (_mixerManager.MicrophoneIsInitialized());
-}
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
 
-int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // If we end up here it means that the selected speaker has no volume
-        // control.
-        available = false;
-        return 0;
-    }
-
-    // Given that InitSpeaker was successful, we know that a volume control exists
-    //
-    available = true;
-
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume)
-{
-
-    return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const
-{
-
-    uint32_t level(0);
-
-    if (_mixerManager.SpeakerVolume(level) == -1)
-    {
-        return -1;
-    }
-
-    volume = level;
-    return 0;
+  volume = level;
+  return 0;
 }
 
 int32_t AudioDeviceMac::SetWaveOutVolume(uint16_t volumeLeft,
-                                         uint16_t volumeRight)
-{
+                                         uint16_t volumeRight) {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+  return -1;
+}
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
+int32_t AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
+                                      uint16_t& /*volumeRight*/) const {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+  return -1;
+}
+
+int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
     return -1;
+  }
+
+  maxVolume = maxVol;
+  return 0;
 }
 
-int32_t
-AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
-                              uint16_t& /*volumeRight*/) const
-{
+int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
     return -1;
+  }
+
+  minVolume = minVol;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
+int32_t AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  uint16_t delta(0);
 
-    uint32_t maxVol(0);
+  if (_mixerManager.SpeakerVolumeStepSize(delta) == -1) {
+    return -1;
+  }
 
-    if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
-    {
-        return -1;
-    }
+  stepSize = delta;
+  return 0;
+}
 
-    maxVolume = maxVol;
+int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
     return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  //
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const
-{
+int32_t AudioDeviceMac::SetSpeakerMute(bool enable) {
+  return (_mixerManager.SetSpeakerMute(enable));
+}
 
-    uint32_t minVol(0);
+int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const {
+  bool muted(0);
 
-    if (_mixerManager.MinSpeakerVolume(minVol) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
 
-    minVolume = minVol;
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
     return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
 }
 
-int32_t
-AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const
-{
+int32_t AudioDeviceMac::SetMicrophoneMute(bool enable) {
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
 
-    uint16_t delta(0);
+int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const {
+  bool muted(0);
 
-    if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
-    {
-        return -1;
-    }
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
 
-    stepSize = delta;
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Enumerate all avaliable microphone and make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
     return 0;
+  }
+
+  // Check if the selected microphone has a boost control
+  //
+  _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable) {
+  return (_mixerManager.SetMicrophoneBoost(enable));
+}
 
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const {
+  bool onOff(0);
 
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // If we end up here it means that the selected speaker has no volume
-        // control, hence it is safe to state that there is no mute control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
+  if (_mixerManager.MicrophoneBoost(onOff) == -1) {
+    return -1;
+  }
 
-    // Check if the selected speaker has a mute control
-    //
-    _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+  enabled = onOff;
+  return 0;
+}
 
-    available = isAvailable;
-
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
+int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
 
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // Cannot open the specified device
+    available = false;
     return 0;
+  }
+
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoRecordingIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetSpeakerMute(bool enable)
-{
-    return (_mixerManager.SetSpeakerMute(enable));
+int32_t AudioDeviceMac::SetStereoRecording(bool enable) {
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const
-{
+int32_t AudioDeviceMac::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
 
-    bool muted(0);
+  return 0;
+}
 
-    if (_mixerManager.SpeakerMute(muted) == -1)
-    {
-        return -1;
-    }
+int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
 
-    enabled = muted;
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // Cannot open the specified device
+    available = false;
     return 0;
+  }
+
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::SetStereoPlayout(bool enable) {
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
 
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+  return 0;
+}
 
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected input device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
+int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
 
-    // Check if the selected microphone has a mute control
-    //
-    _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
-    available = isAvailable;
+  return 0;
+}
 
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
+int32_t AudioDeviceMac::SetAGC(bool enable) {
+  _AGC = enable;
 
+  return 0;
+}
+
+bool AudioDeviceMac::AGC() const {
+  return _AGC;
+}
+
+int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control.
+    available = false;
     return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists
+  //
+  available = true;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetMicrophoneMute(bool enable)
-{
-    return (_mixerManager.SetMicrophoneMute(enable));
+int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
 }
 
-int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const
-{
+int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
 
-    bool muted(0);
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  failed to retrive current microphone level");
+    return -1;
+  }
 
-    if (_mixerManager.MicrophoneMute(muted) == -1)
-    {
-        return -1;
-    }
-
-    enabled = muted;
-    return 0;
+  volume = level;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available)
-{
+int32_t AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
 
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
 
-    // Enumerate all avaliable microphone and make an attempt to open up the
-    // input mixer corresponding to the currently selected input device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone has a boost control
-    //
-    _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
+  maxVolume = maxVol;
+  return 0;
 }
 
-int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable)
-{
+int32_t AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
 
-    return (_mixerManager.SetMicrophoneBoost(enable));
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+  return 0;
 }
 
-int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const
-{
+int32_t AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const {
+  uint16_t delta(0);
 
-    bool onOff(0);
+  if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1) {
+    return -1;
+  }
 
-    if (_mixerManager.MicrophoneBoost(onOff) == -1)
-    {
-        return -1;
-    }
-
-    enabled = onOff;
-    return 0;
+  stepSize = delta;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // Cannot open the specified device
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone can record stereo
-    //
-    _mixerManager.StereoRecordingIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
+int16_t AudioDeviceMac::PlayoutDevices() {
+  AudioDeviceID playDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
+                          MaxNumberDevices);
 }
 
-int32_t AudioDeviceMac::SetStereoRecording(bool enable)
-{
+int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index) {
+  CriticalSectionScoped lock(&_critSect);
 
-    if (enable)
-        _recChannels = 2;
-    else
-        _recChannels = 1;
+  if (_playIsInitialized) {
+    return -1;
+  }
 
-    return 0;
-}
+  AudioDeviceID playDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
+                                       playDevices, MaxNumberDevices);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  number of availiable waveform-audio output devices is %u",
+               nDevices);
 
-int32_t AudioDeviceMac::StereoRecording(bool& enabled) const
-{
+  if (index > (nDevices - 1)) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  device index is out of range [0,%u]", (nDevices - 1));
+    return -1;
+  }
 
-    if (_recChannels == 2)
-        enabled = true;
-    else
-        enabled = false;
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
 
-    return 0;
-}
-
-int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // Cannot open the specified device
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone can record stereo
-    //
-    _mixerManager.StereoPlayoutIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetStereoPlayout(bool enable)
-{
-
-    if (enable)
-        _playChannels = 2;
-    else
-        _playChannels = 1;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const
-{
-
-    if (_playChannels == 2)
-        enabled = true;
-    else
-        enabled = false;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetAGC(bool enable)
-{
-
-    _AGC = enable;
-
-    return 0;
-}
-
-bool AudioDeviceMac::AGC() const
-{
-
-    return _AGC;
-}
-
-int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control.
-        available = false;
-        return 0;
-    }
-
-    // Given that InitMicrophone was successful, we know that a volume control
-    // exists
-    //
-    available = true;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume)
-{
-
-    return (_mixerManager.SetMicrophoneVolume(volume));
-}
-
-int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const
-{
-
-    uint32_t level(0);
-
-    if (_mixerManager.MicrophoneVolume(level) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  failed to retrive current microphone level");
-        return -1;
-    }
-
-    volume = level;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
-
-    uint32_t maxVol(0);
-
-    if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
-    {
-        return -1;
-    }
-
-    maxVolume = maxVol;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const
-{
-
-    uint32_t minVol(0);
-
-    if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
-    {
-        return -1;
-    }
-
-    minVolume = minVol;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
-{
-
-    uint16_t delta(0);
-
-    if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
-    {
-        return -1;
-    }
-
-    stepSize = delta;
-    return 0;
-}
-
-int16_t AudioDeviceMac::PlayoutDevices()
-{
-
-    AudioDeviceID playDevices[MaxNumberDevices];
-    return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
-                            MaxNumberDevices);
-}
-
-int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index)
-{
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_playIsInitialized)
-    {
-        return -1;
-    }
-
-    AudioDeviceID playDevices[MaxNumberDevices];
-    uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
-                                         playDevices, MaxNumberDevices);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  number of availiable waveform-audio output devices is %u",
-                 nDevices);
-
-    if (index > (nDevices - 1))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  device index is out of range [0,%u]", (nDevices - 1));
-        return -1;
-    }
-
-    _outputDeviceIndex = index;
-    _outputDeviceIsSpecified = true;
-
-    return 0;
+  return 0;
 }
 
 int32_t AudioDeviceMac::SetPlayoutDevice(
-    AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+               "WindowsDeviceType not supported");
+  return -1;
+}
+
+int32_t AudioDeviceMac::PlayoutDeviceName(uint16_t index,
+                                          char name[kAdmMaxDeviceNameSize],
+                                          char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(PlayoutDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
     return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
 }
 
-int32_t AudioDeviceMac::PlayoutDeviceName(
-    uint16_t index,
-    char name[kAdmMaxDeviceNameSize],
-    char guid[kAdmMaxGuidSize])
-{
+int32_t AudioDeviceMac::RecordingDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(RecordingDevices());
 
-    const uint16_t nDevices(PlayoutDevices());
-
-    if ((index > (nDevices - 1)) || (name == NULL))
-    {
-        return -1;
-    }
-
-    memset(name, 0, kAdmMaxDeviceNameSize);
-
-    if (guid != NULL)
-    {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
-
-    return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
-}
-
-int32_t AudioDeviceMac::RecordingDeviceName(
-    uint16_t index,
-    char name[kAdmMaxDeviceNameSize],
-    char guid[kAdmMaxGuidSize])
-{
-
-    const uint16_t nDevices(RecordingDevices());
-
-    if ((index > (nDevices - 1)) || (name == NULL))
-    {
-        return -1;
-    }
-
-    memset(name, 0, kAdmMaxDeviceNameSize);
-
-    if (guid != NULL)
-    {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
-
-    return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
-}
-
-int16_t AudioDeviceMac::RecordingDevices()
-{
-
-    AudioDeviceID recDevices[MaxNumberDevices];
-    return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
-                            MaxNumberDevices);
-}
-
-int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index)
-{
-
-    if (_recIsInitialized)
-    {
-        return -1;
-    }
-
-    AudioDeviceID recDevices[MaxNumberDevices];
-    uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
-                                         recDevices, MaxNumberDevices);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  number of availiable waveform-audio input devices is %u",
-                 nDevices);
-
-    if (index > (nDevices - 1))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  device index is out of range [0,%u]", (nDevices - 1));
-        return -1;
-    }
-
-    _inputDeviceIndex = index;
-    _inputDeviceIsSpecified = true;
-
-    return 0;
-}
-
-
-int32_t
-AudioDeviceMac::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
+  if ((index > (nDevices - 1)) || (name == NULL)) {
     return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
 }
 
-int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available)
-{
+int16_t AudioDeviceMac::RecordingDevices() {
+  AudioDeviceID recDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
+                          MaxNumberDevices);
+}
 
-    available = true;
+int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
 
-    // Try to initialize the playout side
-    if (InitPlayout() == -1)
-    {
-        available = false;
-    }
+  AudioDeviceID recDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
+                                       recDevices, MaxNumberDevices);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  number of availiable waveform-audio input devices is %u",
+               nDevices);
 
-    // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
-    // We must actually start playout here in order to have the IOProc
-    // deleted by calling StopPlayout().
-    if (StartPlayout() == -1)
-    {
-        available = false;
-    }
+  if (index > (nDevices - 1)) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  device index is out of range [0,%u]", (nDevices - 1));
+    return -1;
+  }
 
-    // Cancel effect of initialization
-    if (StopPlayout() == -1)
-    {
-        available = false;
-    }
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
 
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+               "WindowsDeviceType not supported");
+  return -1;
+}
+
+int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available) {
+  available = true;
+
+  // Try to initialize the playout side
+  if (InitPlayout() == -1) {
+    available = false;
+  }
+
+  // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
+  // We must actually start playout here in order to have the IOProc
+  // deleted by calling StopPlayout().
+  if (StartPlayout() == -1) {
+    available = false;
+  }
+
+  // Cancel effect of initialization
+  if (StopPlayout() == -1) {
+    available = false;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::RecordingIsAvailable(bool& available) {
+  available = true;
+
+  // Try to initialize the recording side
+  if (InitRecording() == -1) {
+    available = false;
+  }
+
+  // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
+  // We must actually start recording here in order to have the IOProc
+  // deleted by calling StopRecording().
+  if (StartRecording() == -1) {
+    available = false;
+  }
+
+  // Cancel effect of initialization
+  if (StopRecording() == -1) {
+    available = false;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitPlayout() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
     return 0;
-}
+  }
 
-int32_t AudioDeviceMac::RecordingIsAvailable(bool& available)
-{
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeaker() == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  InitSpeaker() failed");
+  }
 
-    available = true;
-
-    // Try to initialize the recording side
-    if (InitRecording() == -1)
-    {
-        available = false;
+  if (!MicrophoneIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (MicrophoneIsAvailable(available) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "  MicrophoneIsAvailable() failed");
     }
+  }
 
-    // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
-    // We must actually start recording here in order to have the IOProc
-    // deleted by calling StopRecording().
-    if (StartRecording() == -1)
-    {
-        available = false;
+  PaUtil_FlushRingBuffer(_paRenderBuffer);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  _renderDelayOffsetSamples = 0;
+  _renderDelayUs = 0;
+  _renderLatencyUs = 0;
+  _renderDeviceIsAlive = 1;
+  _doStop = false;
+
+  // The internal microphone of a MacBook Pro is located under the left speaker
+  // grille. When the internal speakers are in use, we want to fully stereo
+  // pan to the right.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  if (_macBookPro) {
+    _macBookProPanRight = false;
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      UInt32 dataSource = 0;
+      size = sizeof(dataSource);
+      WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, &size, &dataSource));
+
+      if (dataSource == 'ispk') {
+        _macBookProPanRight = true;
+        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                     "MacBook Pro using internal speakers; stereo"
+                     " panning right");
+      } else {
+        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                     "MacBook Pro not using internal speakers");
+      }
+
+      // Add a listener to determine if the status changes.
+      WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
     }
+  }
 
-    // Cancel effect of initialization
-    if (StopRecording() == -1)
-    {
-        available = false;
-    }
+  // Get current stream description
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
+  size = sizeof(_outStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &_outStreamFormat));
 
-    return 0;
-}
+  if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable output stream format -> mFormatID",
+             (const char*)&_outStreamFormat.mFormatID);
+    return -1;
+  }
 
-int32_t AudioDeviceMac::InitPlayout()
-{
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_playing)
-    {
-        return -1;
-    }
-
-    if (!_outputDeviceIsSpecified)
-    {
-        return -1;
-    }
-
-    if (_playIsInitialized)
-    {
-        return 0;
-    }
-
-    // Initialize the speaker (devices might have been added or removed)
-    if (InitSpeaker() == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitSpeaker() failed");
-    }
-
-    if (!MicrophoneIsInitialized())
-    {
-        // Make this call to check if we are using
-        // one or two devices (_twoDevices)
-        bool available = false;
-        if (MicrophoneIsAvailable(available) == -1)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  MicrophoneIsAvailable() failed");
-        }
-    }
-
-    PaUtil_FlushRingBuffer(_paRenderBuffer);
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    _renderDelayOffsetSamples = 0;
-    _renderDelayUs = 0;
-    _renderLatencyUs = 0;
-    _renderDeviceIsAlive = 1;
-    _doStop = false;
-
-    // The internal microphone of a MacBook Pro is located under the left speaker
-    // grille. When the internal speakers are in use, we want to fully stereo
-    // pan to the right.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyDataSource,
-                kAudioDevicePropertyScopeOutput, 0 };
-    if (_macBookPro)
-    {
-        _macBookProPanRight = false;
-        Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                     &propertyAddress);
-        if (hasProperty)
-        {
-            UInt32 dataSource = 0;
-            size = sizeof(dataSource);
-            WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, &size, &dataSource));
-
-            if (dataSource == 'ispk')
-            {
-                _macBookProPanRight = true;
-                WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
-                             _id,
-                             "MacBook Pro using internal speakers; stereo"
-                             " panning right");
-            } else
-            {
-                WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
-                             _id, "MacBook Pro not using internal speakers");
-            }
-
-            // Add a listener to determine if the status changes.
-            WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
-                    &propertyAddress, &objectListenerProc, this));
-        }
-    }
-
-    // Get current stream description
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
-    size = sizeof(_outStreamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &_outStreamFormat));
-
-    if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable output stream format -> mFormatID",
-                 (const char *) &_outStreamFormat.mFormatID);
-        return -1;
-    }
-
-    if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Too many channels on output device (mChannelsPerFrame = %d)",
-            _outStreamFormat.mChannelsPerFrame);
-        return -1;
-    }
-
-    if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Non-interleaved audio data is not supported.",
-                     "AudioHardware streams should not have this format.");
-        return -1;
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "Ouput stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mSampleRate = %f, mChannelsPerFrame = %u",
-                 _outStreamFormat.mSampleRate,
+  if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on output device (mChannelsPerFrame = %d)",
                  _outStreamFormat.mChannelsPerFrame);
+    return -1;
+  }
+
+  if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Non-interleaved audio data is not supported.",
+                 "AudioHardware streams should not have this format.");
+    return -1;
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Ouput stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mSampleRate = %f, mChannelsPerFrame = %u",
+               _outStreamFormat.mSampleRate,
+               _outStreamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerPacket = %u, mFramesPerPacket = %u",
+               _outStreamFormat.mBytesPerPacket,
+               _outStreamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerFrame = %u, mBitsPerChannel = %u",
+               _outStreamFormat.mBytesPerFrame,
+               _outStreamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "mFormatFlags = %u",
+               _outStreamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&_outStreamFormat.mFormatID);
+
+  // Our preferred format to work with.
+  if (_outStreamFormat.mChannelsPerFrame < 2) {
+    // Disable stereo playout when we only have one channel on the device.
+    _playChannels = 1;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 _outStreamFormat.mBytesPerPacket,
-                 _outStreamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 _outStreamFormat.mBytesPerFrame,
-                 _outStreamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mFormatFlags = %u",
-                 _outStreamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &_outStreamFormat.mFormatID);
+                 "Stereo playout unavailable on this device");
+  }
+  WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
 
-    // Our preferred format to work with.
-    if (_outStreamFormat.mChannelsPerFrame < 2)
-    {
-        // Disable stereo playout when we only have one channel on the device.
-        _playChannels = 1;
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Stereo playout unavailable on this device");
-    }
-    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+  // Listen for format changes.
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    // Listen for format changes.
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
-                                                           &propertyAddress,
-                                                           &objectListenerProc,
-                                                           this));
+  // Listen for processor overloads.
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    // Listen for processor overloads.
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
-                                                      &propertyAddress,
-                                                      &objectListenerProc,
-                                                      this));
+  if (_twoDevices || !_recIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _outputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
 
-    if (_twoDevices || !_recIsInitialized)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
-                                                          deviceIOProc,
-                                                          this,
-                                                          &_deviceIOProcID));
-    }
+  _playIsInitialized = true;
 
-    _playIsInitialized = true;
-
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::InitRecording()
-{
+int32_t AudioDeviceMac::InitRecording() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (_recording) {
+    return -1;
+  }
 
-    if (_recording)
-    {
-        return -1;
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophone() == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  InitMicrophone() failed");
+  }
+
+  if (!SpeakerIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (SpeakerIsAvailable(available) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "  SpeakerIsAvailable() failed");
     }
+  }
 
-    if (!_inputDeviceIsSpecified)
-    {
-        return -1;
-    }
+  OSStatus err = noErr;
+  UInt32 size = 0;
 
-    if (_recIsInitialized)
-    {
-        return 0;
-    }
+  PaUtil_FlushRingBuffer(_paCaptureBuffer);
 
-    // Initialize the microphone (devices might have been added or removed)
-    if (InitMicrophone() == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitMicrophone() failed");
-    }
+  _captureDelayUs = 0;
+  _captureLatencyUs = 0;
+  _captureDeviceIsAlive = 1;
+  _doStopRec = false;
 
-    if (!SpeakerIsInitialized())
-    {
-        // Make this call to check if we are using
-        // one or two devices (_twoDevices)
-        bool available = false;
-        if (SpeakerIsAvailable(available) == -1)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  SpeakerIsAvailable() failed");
-        }
-    }
+  // Get current stream description
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
+  size = sizeof(_inStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &_inStreamFormat));
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
+  if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable input stream format -> mFormatID",
+             (const char*)&_inStreamFormat.mFormatID);
+    return -1;
+  }
 
-    PaUtil_FlushRingBuffer(_paCaptureBuffer);
+  if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on input device (mChannelsPerFrame = %d)",
+                 _inStreamFormat.mChannelsPerFrame);
+    return -1;
+  }
 
-    _captureDelayUs = 0;
-    _captureLatencyUs = 0;
-    _captureDeviceIsAlive = 1;
-    _doStopRec = false;
+  const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
+                                    _inStreamFormat.mSampleRate / 100 *
+                                    N_BLOCKS_IO;
+  if (io_block_size_samples > _captureBufSizeSamples) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Input IO block size (%d) is larger than ring buffer (%u)",
+                 io_block_size_samples, _captureBufSizeSamples);
+    return -1;
+  }
 
-    // Get current stream description
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyStreamFormat,
-                kAudioDevicePropertyScopeInput, 0 };
-    memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
-    size = sizeof(_inStreamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &_inStreamFormat));
-
-    if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable input stream format -> mFormatID",
-                 (const char *) &_inStreamFormat.mFormatID);
-        return -1;
-    }
-
-    if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Too many channels on input device (mChannelsPerFrame = %d)",
-            _inStreamFormat.mChannelsPerFrame);
-        return -1;
-    }
-
-    const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
-        _inStreamFormat.mSampleRate / 100 * N_BLOCKS_IO;
-    if (io_block_size_samples > _captureBufSizeSamples)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Input IO block size (%d) is larger than ring buffer (%u)",
-            io_block_size_samples, _captureBufSizeSamples);
-        return -1;
-    }
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Input stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mSampleRate = %f, mChannelsPerFrame = %u",
+               _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mBytesPerPacket = %u, mFramesPerPacket = %u",
+               _inStreamFormat.mBytesPerPacket,
+               _inStreamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mBytesPerFrame = %u, mBitsPerChannel = %u",
+               _inStreamFormat.mBytesPerFrame, _inStreamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " mFormatFlags = %u",
+               _inStreamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&_inStreamFormat.mFormatID);
 
+  // Our preferred format to work with
+  if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+    _inDesiredFormat.mChannelsPerFrame = 2;
+  } else {
+    // Disable stereo recording when we only have one channel on the device.
+    _inDesiredFormat.mChannelsPerFrame = 1;
+    _recChannels = 1;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " Input stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mSampleRate = %f, mChannelsPerFrame = %u",
-                 _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 _inStreamFormat.mBytesPerPacket,
-                 _inStreamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 _inStreamFormat.mBytesPerFrame,
-                 _inStreamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mFormatFlags = %u",
-                 _inStreamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &_inStreamFormat.mFormatID);
+                 "Stereo recording unavailable on this device");
+  }
 
-    // Our preferred format to work with
-    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
-    {
-        _inDesiredFormat.mChannelsPerFrame = 2;
-    } else
-    {
-        // Disable stereo recording when we only have one channel on the device.
-        _inDesiredFormat.mChannelsPerFrame = 1;
-        _recChannels = 1;
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Stereo recording unavailable on this device");
-    }
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  }
 
-    if (_ptrAudioBuffer)
-    {
-        // Update audio buffer with the selected parameters
-        _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-        _ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
-    }
+  _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
+  _inDesiredFormat.mBytesPerPacket =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mFramesPerPacket = 1;
+  _inDesiredFormat.mBytesPerFrame =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
 
-    _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
-    _inDesiredFormat.mBytesPerPacket = _inDesiredFormat.mChannelsPerFrame
-        * sizeof(SInt16);
-    _inDesiredFormat.mFramesPerPacket = 1;
-    _inDesiredFormat.mBytesPerFrame = _inDesiredFormat.mChannelsPerFrame
-        * sizeof(SInt16);
-    _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
-
-    _inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-        | kLinearPCMFormatFlagIsPacked;
+  _inDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+  _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
-    _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+  _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
 
-    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
-            &_captureConverter));
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
+                                            &_captureConverter));
 
-    // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
-    // TODO(xians): investigate this block.
-    UInt32 bufByteCount = (UInt32)((_inStreamFormat.mSampleRate / 1000.0)
-        * 10.0 * N_BLOCKS_IO * _inStreamFormat.mChannelsPerFrame
-        * sizeof(Float32));
-    if (_inStreamFormat.mFramesPerPacket != 0)
-    {
-        if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0)
-        {
-            bufByteCount = ((UInt32)(bufByteCount
-                / _inStreamFormat.mFramesPerPacket) + 1)
-                * _inStreamFormat.mFramesPerPacket;
-        }
+  // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
+  // TODO(xians): investigate this block.
+  UInt32 bufByteCount =
+      (UInt32)((_inStreamFormat.mSampleRate / 1000.0) * 10.0 * N_BLOCKS_IO *
+               _inStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_inStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount =
+          ((UInt32)(bufByteCount / _inStreamFormat.mFramesPerPacket) + 1) *
+          _inStreamFormat.mFramesPerPacket;
     }
+  }
 
-    // Ensure the buffer size is within the acceptable range provided by the device.
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
-    AudioValueRange range;
-    size = sizeof(range);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &range));
-    if (range.mMinimum > bufByteCount)
-    {
-        bufByteCount = range.mMinimum;
-    } else if (range.mMaximum < bufByteCount)
-    {
-        bufByteCount = range.mMaximum;
-    }
+  // Ensure the buffer size is within the acceptable range provided by the
+  // device.
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
 
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
-    size = sizeof(bufByteCount);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, size, &bufByteCount));
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
 
-    // Get capture device latency
-    propertyAddress.mSelector = kAudioDevicePropertyLatency;
-    UInt32 latency = 0;
-    size = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &latency));
-    _captureLatencyUs = (UInt32)((1.0e6 * latency)
-        / _inStreamFormat.mSampleRate);
+  // Get capture device latency
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs = (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
 
-    // Get capture stream latency
-    propertyAddress.mSelector = kAudioDevicePropertyStreams;
-    AudioStreamID stream = 0;
-    size = sizeof(AudioStreamID);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &stream));
-    propertyAddress.mSelector = kAudioStreamPropertyLatency;
-    size = sizeof(UInt32);
-    latency = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &latency));
-    _captureLatencyUs += (UInt32)((1.0e6 * latency)
-        / _inStreamFormat.mSampleRate);
+  // Get capture stream latency
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs +=
+      (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
 
-    // Listen for format changes
-    // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
+  // Listen for format changes
+  // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    // Listen for processor overloads
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
+  // Listen for processor overloads
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
 
-    if (_twoDevices)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
-                inDeviceIOProc, this, &_inDeviceIOProcID));
-    } else if (!_playIsInitialized)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
-                deviceIOProc, this, &_deviceIOProcID));
-    }
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, inDeviceIOProc, this, &_inDeviceIOProcID));
+  } else if (!_playIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
 
-    // Mark recording side as initialized
-    _recIsInitialized = true;
+  // Mark recording side as initialized
+  _recIsInitialized = true;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::StartRecording()
-{
+int32_t AudioDeviceMac::StartRecording() {
+  CriticalSectionScoped lock(&_critSect);
 
-    CriticalSectionScoped lock(&_critSect);
+  if (!_recIsInitialized) {
+    return -1;
+  }
 
-    if (!_recIsInitialized)
-    {
-        return -1;
+  if (_recording) {
+    return 0;
+  }
+
+  if (!_initialized) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Recording worker thread has not been started");
+    return -1;
+  }
+
+  RTC_DCHECK(!capture_worker_thread_.get());
+  capture_worker_thread_.reset(
+      new rtc::PlatformThread(RunCapture, this, "CaptureWorkerThread"));
+  RTC_DCHECK(capture_worker_thread_.get());
+  capture_worker_thread_->Start();
+  capture_worker_thread_->SetPriority(rtc::kRealtimePriority);
+
+  OSStatus err = noErr;
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(
+        AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
+  } else if (!_playing) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
+  }
+
+  _recording = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StopRecording() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  OSStatus err = noErr;
+
+  // Stop device
+  int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
+  if (_twoDevices) {
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStopRec = true;  // Signal to io proc to stop audio device
+      _critSect.Leave();  // Cannot be under lock, risk of deadlock
+      if (kEventTimeout == _stopEventRec.Wait(2000)) {
+        CriticalSectionScoped critScoped(&_critSect);
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     " Timed out stopping the capture IOProc. "
+                     "We may have failed to detect a device removal.");
+
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      }
+      _critSect.Enter();
+      _doStopRec = false;
+      WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " Recording stopped");
     }
+  } else {
+    // We signal a stop for a shared device even when rendering has
+    // not yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking |_recording|) before accessing
+    // resources we free below (e.g. the capture converter).
+    //
+    // In the case of a shared devcie, the IOProc will verify
+    // rendering has ended before stopping itself.
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStop = true;     // Signal to io proc to stop audio device
+      _critSect.Leave();  // Cannot be under lock, risk of deadlock
+      if (kEventTimeout == _stopEvent.Wait(2000)) {
+        CriticalSectionScoped critScoped(&_critSect);
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     " Timed out stopping the shared IOProc. "
+                     "We may have failed to detect a device removal.");
 
-    if (_recording)
-    {
-        return 0;
+        // We assume rendering on a shared device has stopped as well if
+        // the IOProc times out.
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+      }
+      _critSect.Enter();
+      _doStop = false;
+      WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                   " Recording stopped (shared)");
     }
+  }
 
-    if (!_initialized)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Recording worker thread has not been started");
-        return -1;
-    }
+  // Setting this signal will allow the worker thread to be stopped.
+  AtomicSet32(&_captureDeviceIsAlive, 0);
 
-    DCHECK(!capture_worker_thread_.get());
-    capture_worker_thread_ =
-        ThreadWrapper::CreateThread(RunCapture, this, "CaptureWorkerThread");
-    DCHECK(capture_worker_thread_.get());
-    capture_worker_thread_->Start();
-    capture_worker_thread_->SetPriority(kRealtimePriority);
+  if (capture_worker_thread_.get()) {
+    _critSect.Leave();
+    capture_worker_thread_->Stop();
+    capture_worker_thread_.reset();
+    _critSect.Enter();
+  }
 
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
+
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  _recIsInitialized = false;
+  _recording = false;
+
+  return 0;
+}
+
+bool AudioDeviceMac::RecordingIsInitialized() const {
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceMac::Recording() const {
+  return (_recording);
+}
+
+bool AudioDeviceMac::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceMac::StartPlayout() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  RTC_DCHECK(!render_worker_thread_.get());
+  render_worker_thread_.reset(
+      new rtc::PlatformThread(RunRender, this, "RenderWorkerThread"));
+  render_worker_thread_->Start();
+  render_worker_thread_->SetPriority(rtc::kRealtimePriority);
+
+  if (_twoDevices || !_recording) {
     OSStatus err = noErr;
-    if (_twoDevices)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
-    } else if (!_playing)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
-    }
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
+  }
+  _playing = true;
 
-    _recording = true;
+  return 0;
+}
 
+int32_t AudioDeviceMac::StopPlayout() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_playIsInitialized) {
     return 0;
-}
+  }
 
-int32_t AudioDeviceMac::StopRecording()
-{
+  OSStatus err = noErr;
 
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized)
-    {
-        return 0;
-    }
-
-    OSStatus err = noErr;
-
-    // Stop device
-    int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
-    if (_twoDevices)
-    {
-        if (_recording && captureDeviceIsAlive == 1)
-        {
-            _recording = false;
-            _doStopRec = true; // Signal to io proc to stop audio device
-            _critSect.Leave(); // Cannot be under lock, risk of deadlock
-            if (kEventTimeout == _stopEventRec.Wait(2000))
-            {
-                CriticalSectionScoped critScoped(&_critSect);
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             " Timed out stopping the capture IOProc. "
-                             "We may have failed to detect a device removal.");
-
-                WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID,
-                                                   _inDeviceIOProcID));
-                WEBRTC_CA_LOG_WARN(
-                    AudioDeviceDestroyIOProcID(_inputDeviceID,
-                                               _inDeviceIOProcID));
-            }
-            _critSect.Enter();
-            _doStopRec = false;
-            WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                         " Recording stopped");
-        }
-    }
-    else
-    {
-        // We signal a stop for a shared device even when rendering has
-        // not yet ended. This is to ensure the IOProc will return early as
-        // intended (by checking |_recording|) before accessing
-        // resources we free below (e.g. the capture converter).
-        //
-        // In the case of a shared devcie, the IOProc will verify
-        // rendering has ended before stopping itself.
-        if (_recording && captureDeviceIsAlive == 1)
-        {
-            _recording = false;
-            _doStop = true; // Signal to io proc to stop audio device
-            _critSect.Leave(); // Cannot be under lock, risk of deadlock
-            if (kEventTimeout == _stopEvent.Wait(2000))
-            {
-                CriticalSectionScoped critScoped(&_critSect);
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             " Timed out stopping the shared IOProc. "
-                             "We may have failed to detect a device removal.");
-
-                // We assume rendering on a shared device has stopped as well if
-                // the IOProc times out.
-                WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
-                                                   _deviceIOProcID));
-                WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                              _deviceIOProcID));
-            }
-            _critSect.Enter();
-            _doStop = false;
-            WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                         " Recording stopped (shared)");
-        }
-    }
-
-    // Setting this signal will allow the worker thread to be stopped.
-    AtomicSet32(&_captureDeviceIsAlive, 0);
-
-    if (capture_worker_thread_.get()) {
-        _critSect.Leave();
-        capture_worker_thread_->Stop();
-        capture_worker_thread_.reset();
-        _critSect.Enter();
-    }
-
-    WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
-
-    // Remove listeners.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyStreamFormat,
-                kAudioDevicePropertyScopeInput, 0 };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    _recIsInitialized = false;
-    _recording = false;
-
-    return 0;
-}
-
-bool AudioDeviceMac::RecordingIsInitialized() const
-{
-    return (_recIsInitialized);
-}
-
-bool AudioDeviceMac::Recording() const
-{
-    return (_recording);
-}
-
-bool AudioDeviceMac::PlayoutIsInitialized() const
-{
-    return (_playIsInitialized);
-}
-
-int32_t AudioDeviceMac::StartPlayout()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized)
-    {
-        return -1;
-    }
-
-    if (_playing)
-    {
-        return 0;
-    }
-
-    DCHECK(!render_worker_thread_.get());
-    render_worker_thread_ =
-        ThreadWrapper::CreateThread(RunRender, this, "RenderWorkerThread");
-    render_worker_thread_->Start();
-    render_worker_thread_->SetPriority(kRealtimePriority);
-
-    if (_twoDevices || !_recording)
-    {
-        OSStatus err = noErr;
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
-    }
-    _playing = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StopPlayout()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized)
-    {
-        return 0;
-    }
-
-    OSStatus err = noErr;
-
-    int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
-    if (_playing && renderDeviceIsAlive == 1)
-    {
-        // We signal a stop for a shared device even when capturing has not
-        // yet ended. This is to ensure the IOProc will return early as
-        // intended (by checking |_playing|) before accessing resources we
-        // free below (e.g. the render converter).
-        //
-        // In the case of a shared device, the IOProc will verify capturing
-        // has ended before stopping itself.
-        _playing = false;
-        _doStop = true; // Signal to io proc to stop audio device
-        _critSect.Leave(); // Cannot be under lock, risk of deadlock
-        if (kEventTimeout == _stopEvent.Wait(2000))
-        {
-            CriticalSectionScoped critScoped(&_critSect);
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Timed out stopping the render IOProc. "
-                         "We may have failed to detect a device removal.");
-
-            // We assume capturing on a shared device has stopped as well if the
-            // IOProc times out.
-            WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
-                                               _deviceIOProcID));
-            WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                          _deviceIOProcID));
-        }
-        _critSect.Enter();
-        _doStop = false;
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "Playout stopped");
-    }
-
-    // Setting this signal will allow the worker thread to be stopped.
-    AtomicSet32(&_renderDeviceIsAlive, 0);
-    if (render_worker_thread_.get()) {
-        _critSect.Leave();
-        render_worker_thread_->Stop();
-        render_worker_thread_.reset();
-        _critSect.Enter();
-    }
-
-    WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
-
-    // Remove listeners.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput,
-            0 };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    if (_macBookPro)
-    {
-        Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                     &propertyAddress);
-        if (hasProperty)
-        {
-            propertyAddress.mSelector = kAudioDevicePropertyDataSource;
-            WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-                    &propertyAddress, &objectListenerProc, this));
-        }
-    }
-
-    _playIsInitialized = false;
+  int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
+  if (_playing && renderDeviceIsAlive == 1) {
+    // We signal a stop for a shared device even when capturing has not
+    // yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking |_playing|) before accessing resources we
+    // free below (e.g. the render converter).
+    //
+    // In the case of a shared device, the IOProc will verify capturing
+    // has ended before stopping itself.
     _playing = false;
+    _doStop = true;     // Signal to io proc to stop audio device
+    _critSect.Leave();  // Cannot be under lock, risk of deadlock
+    if (kEventTimeout == _stopEvent.Wait(2000)) {
+      CriticalSectionScoped critScoped(&_critSect);
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Timed out stopping the render IOProc. "
+                   "We may have failed to detect a device removal.");
 
-    return 0;
+      // We assume capturing on a shared device has stopped as well if the
+      // IOProc times out.
+      WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+    }
+    _critSect.Enter();
+    _doStop = false;
+    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Playout stopped");
+  }
+
+  // Setting this signal will allow the worker thread to be stopped.
+  AtomicSet32(&_renderDeviceIsAlive, 0);
+  if (render_worker_thread_.get()) {
+    _critSect.Leave();
+    render_worker_thread_->Stop();
+    render_worker_thread_.reset();
+    _critSect.Enter();
+  }
+
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
+
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_macBookPro) {
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      propertyAddress.mSelector = kAudioDevicePropertyDataSource;
+      WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+    }
+  }
+
+  _playIsInitialized = false;
+  _playing = false;
+
+  return 0;
 }
 
-int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const
-{
-    int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-    delayMS = static_cast<uint16_t> (1e-3 * (renderDelayUs + _renderLatencyUs) +
-                                     0.5);
-    return 0;
+int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const {
+  int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+  return 0;
 }
 
-int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const
-{
-    int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-    delayMS = static_cast<uint16_t> (1e-3 * (captureDelayUs +
-                                             _captureLatencyUs) + 0.5);
-    return 0;
+int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const {
+  int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+  return 0;
 }
 
-bool AudioDeviceMac::Playing() const
-{
-    return (_playing);
+bool AudioDeviceMac::Playing() const {
+  return (_playing);
 }
 
 int32_t AudioDeviceMac::SetPlayoutBuffer(
     const AudioDeviceModule::BufferType type,
-    uint16_t sizeMS)
-{
+    uint16_t sizeMS) {
+  if (type != AudioDeviceModule::kFixedBufferSize) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Adaptive buffer size not supported on this platform");
+    return -1;
+  }
 
-    if (type != AudioDeviceModule::kFixedBufferSize)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Adaptive buffer size not supported on this platform");
-        return -1;
-    }
-
-    _playBufType = type;
-    _playBufDelayFixed = sizeMS;
-    return 0;
+  _playBufType = type;
+  _playBufDelayFixed = sizeMS;
+  return 0;
 }
 
-int32_t AudioDeviceMac::PlayoutBuffer(
-    AudioDeviceModule::BufferType& type,
-    uint16_t& sizeMS) const
-{
+int32_t AudioDeviceMac::PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                      uint16_t& sizeMS) const {
+  type = _playBufType;
+  sizeMS = _playBufDelayFixed;
 
-    type = _playBufType;
-    sizeMS = _playBufDelayFixed;
-
-    return 0;
+  return 0;
 }
 
 // Not implemented for Mac.
-int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const
-{
+int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-
-    return -1;
+  return -1;
 }
 
-bool AudioDeviceMac::PlayoutWarning() const
-{
-    return (_playWarning > 0);
+bool AudioDeviceMac::PlayoutWarning() const {
+  return (_playWarning > 0);
 }
 
-bool AudioDeviceMac::PlayoutError() const
-{
-    return (_playError > 0);
+bool AudioDeviceMac::PlayoutError() const {
+  return (_playError > 0);
 }
 
-bool AudioDeviceMac::RecordingWarning() const
-{
-    return (_recWarning > 0);
+bool AudioDeviceMac::RecordingWarning() const {
+  return (_recWarning > 0);
 }
 
-bool AudioDeviceMac::RecordingError() const
-{
-    return (_recError > 0);
+bool AudioDeviceMac::RecordingError() const {
+  return (_recError > 0);
 }
 
-void AudioDeviceMac::ClearPlayoutWarning()
-{
-    _playWarning = 0;
+void AudioDeviceMac::ClearPlayoutWarning() {
+  _playWarning = 0;
 }
 
-void AudioDeviceMac::ClearPlayoutError()
-{
-    _playError = 0;
+void AudioDeviceMac::ClearPlayoutError() {
+  _playError = 0;
 }
 
-void AudioDeviceMac::ClearRecordingWarning()
-{
-    _recWarning = 0;
+void AudioDeviceMac::ClearRecordingWarning() {
+  _recWarning = 0;
 }
 
-void AudioDeviceMac::ClearRecordingError()
-{
-    _recError = 0;
+void AudioDeviceMac::ClearRecordingError() {
+  _recError = 0;
 }
 
 // ============================================================================
 //                                 Private Methods
 // ============================================================================
 
-int32_t
-AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
-                                 AudioDeviceID scopedDeviceIds[],
-                                 const uint32_t deviceListLength)
-{
-    OSStatus err = noErr;
+int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
+                                         AudioDeviceID scopedDeviceIds[],
+                                         const uint32_t deviceListLength) {
+  OSStatus err = noErr;
 
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    UInt32 size = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, &size));
-    if (size == 0)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "No devices");
-        return 0;
-    }
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  UInt32 size = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size));
+  if (size == 0) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "No devices");
+    return 0;
+  }
 
-    AudioDeviceID* deviceIds = (AudioDeviceID*) malloc(size);
-    UInt32 numberDevices = size / sizeof(AudioDeviceID);
-    AudioBufferList* bufferList = NULL;
-    UInt32 numberScopedDevices = 0;
+  AudioDeviceID* deviceIds = (AudioDeviceID*)malloc(size);
+  UInt32 numberDevices = size / sizeof(AudioDeviceID);
+  AudioBufferList* bufferList = NULL;
+  UInt32 numberScopedDevices = 0;
 
-    // First check if there is a default device and list it
-    UInt32 hardwareProperty = 0;
-    if (scope == kAudioDevicePropertyScopeOutput)
-    {
-        hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
-    } else
-    {
-        hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
-    }
+  // First check if there is a default device and list it
+  UInt32 hardwareProperty = 0;
+  if (scope == kAudioDevicePropertyScopeOutput) {
+    hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+  } else {
+    hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+  }
 
-    AudioObjectPropertyAddress
-        propertyAddressDefault = { hardwareProperty,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
+  AudioObjectPropertyAddress propertyAddressDefault = {
+      hardwareProperty, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
 
-    AudioDeviceID usedID;
-    UInt32 uintSize = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-            &propertyAddressDefault, 0, NULL, &uintSize, &usedID));
-    if (usedID != kAudioDeviceUnknown)
-    {
-        scopedDeviceIds[numberScopedDevices] = usedID;
-        numberScopedDevices++;
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "GetNumberDevices(): Default device unknown");
-    }
+  AudioDeviceID usedID;
+  UInt32 uintSize = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+                                                     &propertyAddressDefault, 0,
+                                                     NULL, &uintSize, &usedID));
+  if (usedID != kAudioDeviceUnknown) {
+    scopedDeviceIds[numberScopedDevices] = usedID;
+    numberScopedDevices++;
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "GetNumberDevices(): Default device unknown");
+  }
 
-    // Then list the rest of the devices
-    bool listOK = true;
+  // Then list the rest of the devices
+  bool listOK = true;
 
-    WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, &size, deviceIds));
-    if (err != noErr)
-    {
+  WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds));
+  if (err != noErr) {
+    listOK = false;
+  } else {
+    propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
+    propertyAddress.mScope = scope;
+    propertyAddress.mElement = 0;
+    for (UInt32 i = 0; i < numberDevices; i++) {
+      // Check for input channels
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(
+          deviceIds[i], &propertyAddress, 0, NULL, &size));
+      if (err == kAudioHardwareBadDeviceError) {
+        // This device doesn't actually exist; continue iterating.
+        continue;
+      } else if (err != noErr) {
         listOK = false;
-    } else
-    {
-        propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
-        propertyAddress.mScope = scope;
-        propertyAddress.mElement = 0;
-        for (UInt32 i = 0; i < numberDevices; i++)
-        {
-            // Check for input channels
-            WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(deviceIds[i],
-                    &propertyAddress, 0, NULL, &size));
-            if (err == kAudioHardwareBadDeviceError)
-            {
-                // This device doesn't actually exist; continue iterating.
-                continue;
-            } else if (err != noErr)
-            {
-                listOK = false;
-                break;
-            }
+        break;
+      }
 
-            bufferList = (AudioBufferList*) malloc(size);
-            WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(deviceIds[i],
-                    &propertyAddress, 0, NULL, &size, bufferList));
-            if (err != noErr)
-            {
-                listOK = false;
-                break;
-            }
+      bufferList = (AudioBufferList*)malloc(size);
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+          deviceIds[i], &propertyAddress, 0, NULL, &size, bufferList));
+      if (err != noErr) {
+        listOK = false;
+        break;
+      }
 
-            if (bufferList->mNumberBuffers > 0)
-            {
-                if (numberScopedDevices >= deviceListLength)
-                {
-                    WEBRTC_TRACE(kTraceError,
-                                 kTraceAudioDevice, _id,
-                                 "Device list is not long enough");
-                    listOK = false;
-                    break;
-                }
-
-                scopedDeviceIds[numberScopedDevices] = deviceIds[i];
-                numberScopedDevices++;
-            }
-
-            free(bufferList);
-            bufferList = NULL;
-        }  // for
-    }
-
-    if (!listOK)
-    {
-        if (deviceIds)
-        {
-            free(deviceIds);
-            deviceIds = NULL;
+      if (bufferList->mNumberBuffers > 0) {
+        if (numberScopedDevices >= deviceListLength) {
+          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                       "Device list is not long enough");
+          listOK = false;
+          break;
         }
 
-        if (bufferList)
-        {
-            free(bufferList);
-            bufferList = NULL;
-        }
+        scopedDeviceIds[numberScopedDevices] = deviceIds[i];
+        numberScopedDevices++;
+      }
 
-        return -1;
+      free(bufferList);
+      bufferList = NULL;
+    }  // for
+  }
+
+  if (!listOK) {
+    if (deviceIds) {
+      free(deviceIds);
+      deviceIds = NULL;
     }
 
-    // Happy ending
-    if (deviceIds)
-    {
-        free(deviceIds);
-        deviceIds = NULL;
+    if (bufferList) {
+      free(bufferList);
+      bufferList = NULL;
     }
 
-    return numberScopedDevices;
+    return -1;
+  }
+
+  // Happy ending
+  if (deviceIds) {
+    free(deviceIds);
+    deviceIds = NULL;
+  }
+
+  return numberScopedDevices;
 }
 
-int32_t
-AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
-                              const uint16_t index,
-                              char* name)
-{
-    OSStatus err = noErr;
-    UInt32 len = kAdmMaxDeviceNameSize;
-    AudioDeviceID deviceIds[MaxNumberDevices];
+int32_t AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
+                                      const uint16_t index,
+                                      char* name) {
+  OSStatus err = noErr;
+  UInt32 len = kAdmMaxDeviceNameSize;
+  AudioDeviceID deviceIds[MaxNumberDevices];
 
-    int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
-    if (numberDevices < 0)
-    {
-        return -1;
-    } else if (numberDevices == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "No devices");
-        return -1;
+  int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "No devices");
+    return -1;
+  }
+
+  // If the number is below the number of devices, assume it's "WEBRTC ID"
+  // otherwise assume it's a CoreAudio ID
+  AudioDeviceID usedID;
+
+  // Check if there is a default device
+  bool isDefaultDevice = false;
+  if (index == 0) {
+    UInt32 hardwareProperty = 0;
+    if (scope == kAudioDevicePropertyScopeOutput) {
+      hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+    } else {
+      hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
     }
-
-    // If the number is below the number of devices, assume it's "WEBRTC ID"
-    // otherwise assume it's a CoreAudio ID
-    AudioDeviceID usedID;
-
-    // Check if there is a default device
-    bool isDefaultDevice = false;
-    if (index == 0)
-    {
-        UInt32 hardwareProperty = 0;
-        if (scope == kAudioDevicePropertyScopeOutput)
-        {
-            hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
-        } else
-        {
-            hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
-        }
-        AudioObjectPropertyAddress propertyAddress = { hardwareProperty,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-        UInt32 size = sizeof(UInt32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                &propertyAddress, 0, NULL, &size, &usedID));
-        if (usedID == kAudioDeviceUnknown)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "GetDeviceName(): Default device unknown");
-        } else
-        {
-            isDefaultDevice = true;
-        }
-    }
-
     AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyDeviceName, scope, 0 };
+        hardwareProperty, kAudioObjectPropertyScopeGlobal,
+        kAudioObjectPropertyElementMaster};
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &usedID));
+    if (usedID == kAudioDeviceUnknown) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "GetDeviceName(): Default device unknown");
+    } else {
+      isDefaultDevice = true;
+    }
+  }
 
-    if (isDefaultDevice)
-    {
-        char devName[len];
+  AudioObjectPropertyAddress propertyAddress = {kAudioDevicePropertyDeviceName,
+                                                scope, 0};
 
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
-                &propertyAddress, 0, NULL, &len, devName));
+  if (isDefaultDevice) {
+    char devName[len];
 
-        sprintf(name, "default (%s)", devName);
-    } else
-    {
-        if (index < numberDevices)
-        {
-            usedID = deviceIds[index];
-        } else
-        {
-            usedID = index;
-        }
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID, &propertyAddress,
+                                                       0, NULL, &len, devName));
 
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
-                &propertyAddress, 0, NULL, &len, name));
+    sprintf(name, "default (%s)", devName);
+  } else {
+    if (index < numberDevices) {
+      usedID = deviceIds[index];
+    } else {
+      usedID = index;
     }
 
-    return 0;
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID, &propertyAddress,
+                                                       0, NULL, &len, name));
+  }
+
+  return 0;
 }
 
 int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
                                    AudioDeviceID& deviceId,
-                                   const bool isInput)
-{
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    AudioObjectPropertyScope deviceScope;
-    AudioObjectPropertySelector defaultDeviceSelector;
-    AudioDeviceID deviceIds[MaxNumberDevices];
+                                   const bool isInput) {
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  AudioObjectPropertyScope deviceScope;
+  AudioObjectPropertySelector defaultDeviceSelector;
+  AudioDeviceID deviceIds[MaxNumberDevices];
 
-    if (isInput)
-    {
-        deviceScope = kAudioDevicePropertyScopeInput;
-        defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
-    } else
-    {
-        deviceScope = kAudioDevicePropertyScopeOutput;
-        defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  if (isInput) {
+    deviceScope = kAudioDevicePropertyScopeInput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
+  } else {
+    deviceScope = kAudioDevicePropertyScopeOutput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  }
+
+  AudioObjectPropertyAddress propertyAddress = {
+      defaultDeviceSelector, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+
+  // Get the actual device IDs
+  int numberDevices =
+      GetNumberDevices(deviceScope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "InitDevice(): No devices");
+    return -1;
+  }
+
+  bool isDefaultDevice = false;
+  deviceId = kAudioDeviceUnknown;
+  if (userDeviceIndex == 0) {
+    // Try to use default system device
+    size = sizeof(AudioDeviceID);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &deviceId));
+    if (deviceId == kAudioDeviceUnknown) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " No default device exists");
+    } else {
+      isDefaultDevice = true;
     }
+  }
 
-    AudioObjectPropertyAddress
-        propertyAddress = { defaultDeviceSelector,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
+  if (!isDefaultDevice) {
+    deviceId = deviceIds[userDeviceIndex];
+  }
 
-    // Get the actual device IDs
-    int numberDevices = GetNumberDevices(deviceScope, deviceIds,
-                                         MaxNumberDevices);
-    if (numberDevices < 0)
-    {
-        return -1;
-    } else if (numberDevices == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "InitDevice(): No devices");
-        return -1;
-    }
+  // Obtain device name and manufacturer for logging.
+  // Also use this as a test to ensure a user-set device ID is valid.
+  char devName[128];
+  char devManf[128];
+  memset(devName, 0, sizeof(devName));
+  memset(devManf, 0, sizeof(devManf));
 
-    bool isDefaultDevice = false;
-    deviceId = kAudioDeviceUnknown;
-    if (userDeviceIndex == 0)
-    {
-        // Try to use default system device
-        size = sizeof(AudioDeviceID);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                &propertyAddress, 0, NULL, &size, &deviceId));
-        if (deviceId == kAudioDeviceUnknown)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " No default device exists");
-        } else
-        {
-            isDefaultDevice = true;
-        }
-    }
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
+  propertyAddress.mScope = deviceScope;
+  propertyAddress.mElement = 0;
+  size = sizeof(devName);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devName));
 
-    if (!isDefaultDevice)
-    {
-        deviceId = deviceIds[userDeviceIndex];
-    }
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
+  size = sizeof(devManf);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devManf));
 
-    // Obtain device name and manufacturer for logging.
-    // Also use this as a test to ensure a user-set device ID is valid.
-    char devName[128];
-    char devManf[128];
-    memset(devName, 0, sizeof(devName));
-    memset(devManf, 0, sizeof(devManf));
+  if (isInput) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Input device: %s %s",
+                 devManf, devName);
+  } else {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Output device: %s %s",
+                 devManf, devName);
+  }
 
-    propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
-    propertyAddress.mScope = deviceScope;
-    propertyAddress.mElement = 0;
-    size = sizeof(devName);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
-            &propertyAddress, 0, NULL, &size, devName));
-
-    propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
-    size = sizeof(devManf);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
-            &propertyAddress, 0, NULL, &size, devManf));
-
-    if (isInput)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Input device: %s %s", devManf, devName);
-    } else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Output device: %s %s", devManf, devName);
-    }
-
-    return 0;
+  return 0;
 }
 
-OSStatus AudioDeviceMac::SetDesiredPlayoutFormat()
-{
-    // Our preferred format to work with.
-    _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
-    _outDesiredFormat.mChannelsPerFrame = _playChannels;
+OSStatus AudioDeviceMac::SetDesiredPlayoutFormat() {
+  // Our preferred format to work with.
+  _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
+  _outDesiredFormat.mChannelsPerFrame = _playChannels;
 
-    if (_ptrAudioBuffer)
-    {
-        // Update audio buffer with the selected parameters.
-        _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
-        _ptrAudioBuffer->SetPlayoutChannels((uint8_t) _playChannels);
-    }
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters.
+    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  }
 
-    _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT *
-        ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+  _renderDelayOffsetSamples = _renderBufSizeSamples -
+                              N_BUFFERS_OUT * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES *
+                                  _outDesiredFormat.mChannelsPerFrame;
 
-    _outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame *
-                                        sizeof(SInt16);
-    // In uncompressed audio, a packet is one frame.
-    _outDesiredFormat.mFramesPerPacket = 1;
-    _outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame *
-                                       sizeof(SInt16);
-    _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+  _outDesiredFormat.mBytesPerPacket =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  // In uncompressed audio, a packet is one frame.
+  _outDesiredFormat.mFramesPerPacket = 1;
+  _outDesiredFormat.mBytesPerFrame =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
 
-    _outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
-                                     kLinearPCMFormatFlagIsPacked;
+  _outDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+  _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
-    _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+  _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
 
-    OSStatus err = noErr;
-    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat,
-                                              &_outStreamFormat,
-                                              &_renderConverter));
+  OSStatus err = noErr;
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(
+      &_outDesiredFormat, &_outStreamFormat, &_renderConverter));
 
-    // Try to set buffer size to desired value (_playBufDelayFixed).
-    UInt32 bufByteCount = static_cast<UInt32> ((_outStreamFormat.mSampleRate /
-                          1000.0) *
-                          _playBufDelayFixed *
-                          _outStreamFormat.mChannelsPerFrame *
-                          sizeof(Float32));
-    if (_outStreamFormat.mFramesPerPacket != 0)
-    {
-        if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
-        {
-            bufByteCount = (static_cast<UInt32> (bufByteCount /
-                           _outStreamFormat.mFramesPerPacket) + 1) *
-                           _outStreamFormat.mFramesPerPacket;
-        }
+  // Try to set buffer size to desired value (_playBufDelayFixed).
+  UInt32 bufByteCount = static_cast<UInt32>(
+      (_outStreamFormat.mSampleRate / 1000.0) * _playBufDelayFixed *
+      _outStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_outStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount = (static_cast<UInt32>(bufByteCount /
+                                          _outStreamFormat.mFramesPerPacket) +
+                      1) *
+                     _outStreamFormat.mFramesPerPacket;
     }
+  }
 
-    // Ensure the buffer size is within the range provided by the device.
-    AudioObjectPropertyAddress propertyAddress =
-        {kAudioDevicePropertyDataSource,
-         kAudioDevicePropertyScopeOutput,
-         0};
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
-    AudioValueRange range;
-    UInt32 size = sizeof(range);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &range));
-    if (range.mMinimum > bufByteCount)
-    {
-        bufByteCount = range.mMinimum;
-    } else if (range.mMaximum < bufByteCount)
-    {
-        bufByteCount = range.mMaximum;
-    }
+  // Ensure the buffer size is within the range provided by the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  UInt32 size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
 
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
-    size = sizeof(bufByteCount);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       size,
-                                                       &bufByteCount));
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
 
-    // Get render device latency.
-    propertyAddress.mSelector = kAudioDevicePropertyLatency;
-    UInt32 latency = 0;
-    size = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &latency));
-    _renderLatencyUs = static_cast<uint32_t> ((1.0e6 * latency) /
-                       _outStreamFormat.mSampleRate);
+  // Get render device latency.
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs =
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
 
-    // Get render stream latency.
-    propertyAddress.mSelector = kAudioDevicePropertyStreams;
-    AudioStreamID stream = 0;
-    size = sizeof(AudioStreamID);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &stream));
-    propertyAddress.mSelector = kAudioStreamPropertyLatency;
-    size = sizeof(UInt32);
-    latency = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &latency));
-    _renderLatencyUs += static_cast<uint32_t> ((1.0e6 * latency) /
-                        _outStreamFormat.mSampleRate);
+  // Get render stream latency.
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs +=
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  initial playout status: _renderDelayOffsetSamples=%d,"
-                 " _renderDelayUs=%d, _renderLatencyUs=%d",
-                 _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
-    return 0;
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  initial playout status: _renderDelayOffsetSamples=%d,"
+               " _renderDelayUs=%d, _renderLatencyUs=%d",
+               _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
+  return 0;
 }
 
 OSStatus AudioDeviceMac::objectListenerProc(
     AudioObjectID objectId,
     UInt32 numberAddresses,
     const AudioObjectPropertyAddress addresses[],
-    void* clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    DCHECK(ptrThis != NULL);
+    void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
+  ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
 
-    // AudioObjectPropertyListenerProc functions are supposed to return 0
-    return 0;
+  // AudioObjectPropertyListenerProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::implObjectListenerProc(
     const AudioObjectID objectId,
     const UInt32 numberAddresses,
-    const AudioObjectPropertyAddress addresses[])
-{
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "AudioDeviceMac::implObjectListenerProc()");
+    const AudioObjectPropertyAddress addresses[]) {
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+               "AudioDeviceMac::implObjectListenerProc()");
 
-    for (UInt32 i = 0; i < numberAddresses; i++)
-    {
-        if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
-        {
-            HandleDeviceChange();
-        } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat)
-        {
-            HandleStreamFormatChange(objectId, addresses[i]);
-        } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource)
-        {
-            HandleDataSourceChange(objectId, addresses[i]);
-        } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload)
-        {
-            HandleProcessorOverload(addresses[i]);
-        }
+  for (UInt32 i = 0; i < numberAddresses; i++) {
+    if (addresses[i].mSelector == kAudioHardwarePropertyDevices) {
+      HandleDeviceChange();
+    } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat) {
+      HandleStreamFormatChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource) {
+      HandleDataSourceChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload) {
+      HandleProcessorOverload(addresses[i]);
     }
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioDeviceMac::HandleDeviceChange()
-{
-    OSStatus err = noErr;
+int32_t AudioDeviceMac::HandleDeviceChange() {
+  OSStatus err = noErr;
 
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "kAudioHardwarePropertyDevices");
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+               "kAudioHardwarePropertyDevices");
 
-    // A device has changed. Check if our registered devices have been removed.
-    // Ensure the devices have been initialized, meaning the IDs are valid.
-    if (MicrophoneIsInitialized())
-    {
-        AudioObjectPropertyAddress propertyAddress = {
-                kAudioDevicePropertyDeviceIsAlive,
-                kAudioDevicePropertyScopeInput, 0 };
-        UInt32 deviceIsAlive = 1;
-        UInt32 size = sizeof(UInt32);
-        err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0,
-                                         NULL, &size, &deviceIsAlive);
+  // A device has changed. Check if our registered devices have been removed.
+  // Ensure the devices have been initialized, meaning the IDs are valid.
+  if (MicrophoneIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeInput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
 
-        if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "Capture device is not alive (probably removed)");
-            AtomicSet32(&_captureDeviceIsAlive, 0);
-            _mixerManager.CloseMicrophone();
-            if (_recError == 1)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                             _id, "  pending recording error exists");
-            }
-            _recError = 1; // triggers callback from module process thread
-        } else if (err != noErr)
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioDeviceGetProperty()", (const char*) &err);
-            return -1;
-        }
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "Capture device is not alive (probably removed)");
+      AtomicSet32(&_captureDeviceIsAlive, 0);
+      _mixerManager.CloseMicrophone();
+      if (_recError == 1) {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     "  pending recording error exists");
+      }
+      _recError = 1;  // triggers callback from module process thread
+    } else if (err != noErr) {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioDeviceGetProperty()", (const char*)&err);
+      return -1;
     }
+  }
 
-    if (SpeakerIsInitialized())
-    {
-        AudioObjectPropertyAddress propertyAddress = {
-                kAudioDevicePropertyDeviceIsAlive,
-                kAudioDevicePropertyScopeOutput, 0 };
-        UInt32 deviceIsAlive = 1;
-        UInt32 size = sizeof(UInt32);
-        err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0,
-                                         NULL, &size, &deviceIsAlive);
+  if (SpeakerIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeOutput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
 
-        if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "Render device is not alive (probably removed)");
-            AtomicSet32(&_renderDeviceIsAlive, 0);
-            _mixerManager.CloseSpeaker();
-            if (_playError == 1)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                             _id, "  pending playout error exists");
-            }
-            _playError = 1; // triggers callback from module process thread
-        } else if (err != noErr)
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioDeviceGetProperty()", (const char*) &err);
-            return -1;
-        }
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "Render device is not alive (probably removed)");
+      AtomicSet32(&_renderDeviceIsAlive, 0);
+      _mixerManager.CloseSpeaker();
+      if (_playError == 1) {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     "  pending playout error exists");
+      }
+      _playError = 1;  // triggers callback from module process thread
+    } else if (err != noErr) {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioDeviceGetProperty()", (const char*)&err);
+      return -1;
     }
+  }
 
-    return 0;
+  return 0;
 }
 
 int32_t AudioDeviceMac::HandleStreamFormatChange(
     const AudioObjectID objectId,
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    OSStatus err = noErr;
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
 
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "Stream format changed");
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Stream format changed");
 
-    if (objectId != _inputDeviceID && objectId != _outputDeviceID)
-    {
-        return 0;
-    }
-
-    // Get the new device format
-    AudioStreamBasicDescription streamFormat;
-    UInt32 size = sizeof(streamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
-
-    if (streamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable input stream format -> mFormatID",
-                 (const char *) &streamFormat.mFormatID);
-        return -1;
-    }
-
-    if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Too many channels on device (mChannelsPerFrame = %d)",
-                     streamFormat.mChannelsPerFrame);
-        return -1;
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "Stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mSampleRate = %f, mChannelsPerFrame = %u",
-                 streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mFormatFlags = %u",
-                 streamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &streamFormat.mFormatID);
-
-    if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
-    {
-        const int io_block_size_samples = streamFormat.mChannelsPerFrame *
-            streamFormat.mSampleRate / 100 * N_BLOCKS_IO;
-        if (io_block_size_samples > _captureBufSizeSamples)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                "Input IO block size (%d) is larger than ring buffer (%u)",
-                io_block_size_samples, _captureBufSizeSamples);
-            return -1;
-
-        }
-
-        memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
-
-        if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
-        {
-            _inDesiredFormat.mChannelsPerFrame = 2;
-        } else
-        {
-            // Disable stereo recording when we only have one channel on the device.
-            _inDesiredFormat.mChannelsPerFrame = 1;
-            _recChannels = 1;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Stereo recording unavailable on this device");
-        }
-
-        if (_ptrAudioBuffer)
-        {
-            // Update audio buffer with the selected parameters
-            _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-            _ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
-        }
-
-        // Recreate the converter with the new format
-        // TODO(xians): make this thread safe
-        WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
-
-        WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
-                &_captureConverter));
-    } else
-    {
-        memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
-
-        // Our preferred format to work with
-        if (_outStreamFormat.mChannelsPerFrame < 2)
-        {
-            _playChannels = 1;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Stereo playout unavailable on this device");
-        }
-        WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
-    }
+  if (objectId != _inputDeviceID && objectId != _outputDeviceID) {
     return 0;
+  }
+
+  // Get the new device format
+  AudioStreamBasicDescription streamFormat;
+  UInt32 size = sizeof(streamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      objectId, &propertyAddress, 0, NULL, &size, &streamFormat));
+
+  if (streamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable input stream format -> mFormatID",
+             (const char*)&streamFormat.mFormatID);
+    return -1;
+  }
+
+  if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on device (mChannelsPerFrame = %d)",
+                 streamFormat.mChannelsPerFrame);
+    return -1;
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mSampleRate = %f, mChannelsPerFrame = %u",
+               streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerPacket = %u, mFramesPerPacket = %u",
+               streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerFrame = %u, mBitsPerChannel = %u",
+               streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "mFormatFlags = %u",
+               streamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&streamFormat.mFormatID);
+
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    const int io_block_size_samples = streamFormat.mChannelsPerFrame *
+                                      streamFormat.mSampleRate / 100 *
+                                      N_BLOCKS_IO;
+    if (io_block_size_samples > _captureBufSizeSamples) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   "Input IO block size (%d) is larger than ring buffer (%u)",
+                   io_block_size_samples, _captureBufSizeSamples);
+      return -1;
+    }
+
+    memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+      _inDesiredFormat.mChannelsPerFrame = 2;
+    } else {
+      // Disable stereo recording when we only have one channel on the device.
+      _inDesiredFormat.mChannelsPerFrame = 1;
+      _recChannels = 1;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "Stereo recording unavailable on this device");
+    }
+
+    if (_ptrAudioBuffer) {
+      // Update audio buffer with the selected parameters
+      _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+      _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+    }
+
+    // Recreate the converter with the new format
+    // TODO(xians): make this thread safe
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
+                                              &_captureConverter));
+  } else {
+    memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    // Our preferred format to work with
+    if (_outStreamFormat.mChannelsPerFrame < 2) {
+      _playChannels = 1;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "Stereo playout unavailable on this device");
+    }
+    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+  }
+  return 0;
 }
 
 int32_t AudioDeviceMac::HandleDataSourceChange(
     const AudioObjectID objectId,
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    OSStatus err = noErr;
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
 
-    if (_macBookPro && propertyAddress.mScope
-        == kAudioDevicePropertyScopeOutput)
-    {
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "Data source changed");
+  if (_macBookPro &&
+      propertyAddress.mScope == kAudioDevicePropertyScopeOutput) {
+    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Data source changed");
 
-        _macBookProPanRight = false;
-        UInt32 dataSource = 0;
-        UInt32 size = sizeof(UInt32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
-                &propertyAddress, 0, NULL, &size, &dataSource));
-        if (dataSource == 'ispk')
-        {
-            _macBookProPanRight = true;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "MacBook Pro using internal speakers; stereo panning right");
-        } else
-        {
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "MacBook Pro not using internal speakers");
-        }
+    _macBookProPanRight = false;
+    UInt32 dataSource = 0;
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        objectId, &propertyAddress, 0, NULL, &size, &dataSource));
+    if (dataSource == 'ispk') {
+      _macBookProPanRight = true;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "MacBook Pro using internal speakers; stereo panning right");
+    } else {
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "MacBook Pro not using internal speakers");
     }
+  }
 
-    return 0;
+  return 0;
 }
 int32_t AudioDeviceMac::HandleProcessorOverload(
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    // TODO(xians): we probably want to notify the user in some way of the
-    // overload. However, the Windows interpretations of these errors seem to
-    // be more severe than what ProcessorOverload is thrown for.
-    //
-    // We don't log the notification, as it's sent from the HAL's IO thread. We
-    // don't want to slow it down even further.
-    if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
-    {
-        //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
-        // overload");
-        //_callback->ProblemIsReported(
-        // SndCardStreamObserver::ERecordingProblem);
-    } else
-    {
-        //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-        // "Render processor overload");
-        //_callback->ProblemIsReported(
-        // SndCardStreamObserver::EPlaybackProblem);
-    }
+    const AudioObjectPropertyAddress propertyAddress) {
+  // TODO(xians): we probably want to notify the user in some way of the
+  // overload. However, the Windows interpretations of these errors seem to
+  // be more severe than what ProcessorOverload is thrown for.
+  //
+  // We don't log the notification, as it's sent from the HAL's IO thread. We
+  // don't want to slow it down even further.
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    // WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
+    // overload");
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::ERecordingProblem);
+  } else {
+    // WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+    // "Render processor overload");
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::EPlaybackProblem);
+  }
 
-    return 0;
+  return 0;
 }
 
 // ============================================================================
 //                                  Thread Methods
 // ============================================================================
 
-OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID,
+                                      const AudioTimeStamp*,
                                       const AudioBufferList* inputData,
                                       const AudioTimeStamp* inputTime,
                                       AudioBufferList* outputData,
                                       const AudioTimeStamp* outputTime,
-                                      void *clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    DCHECK(ptrThis != NULL);
+                                      void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
+  ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
 
-    // AudioDeviceIOProc functions are supposed to return 0
-    return 0;
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
-                                          UInt32 *numberDataPackets,
-                                          AudioBufferList *data,
-                                          AudioStreamPacketDescription **,
-                                          void *userData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) userData;
-    DCHECK(ptrThis != NULL);
+                                          UInt32* numberDataPackets,
+                                          AudioBufferList* data,
+                                          AudioStreamPacketDescription**,
+                                          void* userData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)userData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    return ptrThis->implOutConverterProc(numberDataPackets, data);
+  return ptrThis->implOutConverterProc(numberDataPackets, data);
 }
 
-OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID,
+                                        const AudioTimeStamp*,
                                         const AudioBufferList* inputData,
                                         const AudioTimeStamp* inputTime,
                                         AudioBufferList*,
-                                        const AudioTimeStamp*, void* clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    DCHECK(ptrThis != NULL);
+                                        const AudioTimeStamp*,
+                                        void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
 
-    ptrThis->implInDeviceIOProc(inputData, inputTime);
+  ptrThis->implInDeviceIOProc(inputData, inputTime);
 
-    // AudioDeviceIOProc functions are supposed to return 0
-    return 0;
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::inConverterProc(
     AudioConverterRef,
-    UInt32 *numberDataPackets,
-    AudioBufferList *data,
-    AudioStreamPacketDescription ** /*dataPacketDescription*/,
-    void *userData)
-{
-    AudioDeviceMac *ptrThis = static_cast<AudioDeviceMac*> (userData);
-    DCHECK(ptrThis != NULL);
+    UInt32* numberDataPackets,
+    AudioBufferList* data,
+    AudioStreamPacketDescription** /*dataPacketDescription*/,
+    void* userData) {
+  AudioDeviceMac* ptrThis = static_cast<AudioDeviceMac*>(userData);
+  RTC_DCHECK(ptrThis != NULL);
 
-    return ptrThis->implInConverterProc(numberDataPackets, data);
+  return ptrThis->implInConverterProc(numberDataPackets, data);
 }
 
-OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList *inputData,
-                                          const AudioTimeStamp *inputTime,
-                                          AudioBufferList *outputData,
-                                          const AudioTimeStamp *outputTime)
-{
-    OSStatus err = noErr;
-    UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
-    UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList* inputData,
+                                          const AudioTimeStamp* inputTime,
+                                          AudioBufferList* outputData,
+                                          const AudioTimeStamp* outputTime) {
+  OSStatus err = noErr;
+  UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
 
-    if (!_twoDevices && _recording)
-    {
-        implInDeviceIOProc(inputData, inputTime);
-    }
+  if (!_twoDevices && _recording) {
+    implInDeviceIOProc(inputData, inputTime);
+  }
 
-    // Check if we should close down audio device
-    // Double-checked locking optimization to remove locking overhead
-    if (_doStop)
-    {
-        _critSect.Enter();
-        if (_doStop)
-        {
-            if (_twoDevices || (!_recording && !_playing))
-            {
-               // In the case of a shared device, the single driving ioProc
-               // is stopped here
-               WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID,
-                                                 _deviceIOProcID));
-               WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                             _deviceIOProcID));
-               if (err == noErr)
-               {
-                  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                               _id, " Playout or shared device stopped");
-               }
-            }
-
-            _doStop = false;
-            _stopEvent.Set();
-            _critSect.Leave();
-            return 0;
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStop) {
+    _critSect.Enter();
+    if (_doStop) {
+      if (_twoDevices || (!_recording && !_playing)) {
+        // In the case of a shared device, the single driving ioProc
+        // is stopped here
+        WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+        if (err == noErr) {
+          WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                       " Playout or shared device stopped");
         }
-        _critSect.Leave();
+      }
+
+      _doStop = false;
+      _stopEvent.Set();
+      _critSect.Leave();
+      return 0;
     }
+    _critSect.Leave();
+  }
 
-    if (!_playing)
-    {
-        // This can be the case when a shared device is capturing but not
-        // rendering. We allow the checks above before returning to avoid a
-        // timeout when capturing is stopped.
-        return 0;
-    }
-
-    DCHECK(_outStreamFormat.mBytesPerFrame != 0);
-    UInt32 size = outputData->mBuffers->mDataByteSize
-        / _outStreamFormat.mBytesPerFrame;
-
-    // TODO(xians): signal an error somehow?
-    err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
-                                          this, &size, outputData, NULL);
-    if (err != noErr)
-    {
-        if (err == 1)
-        {
-            // This is our own error.
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " Error in AudioConverterFillComplexBuffer()");
-            return 1;
-        } else
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioConverterFillComplexBuffer()",
-                     (const char *) &err);
-            return 1;
-        }
-    }
-
-    PaRingBufferSize bufSizeSamples =
-        PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
-
-    int32_t renderDelayUs = static_cast<int32_t> (1e-3 * (outputTimeNs - nowNs)
-        + 0.5);
-    renderDelayUs += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
-        / _outDesiredFormat.mChannelsPerFrame / _outDesiredFormat.mSampleRate
-        + 0.5);
-
-    AtomicSet32(&_renderDelayUs, renderDelayUs);
-
+  if (!_playing) {
+    // This can be the case when a shared device is capturing but not
+    // rendering. We allow the checks above before returning to avoid a
+    // timeout when capturing is stopped.
     return 0;
+  }
+
+  RTC_DCHECK(_outStreamFormat.mBytesPerFrame != 0);
+  UInt32 size =
+      outputData->mBuffers->mDataByteSize / _outStreamFormat.mBytesPerFrame;
+
+  // TODO(xians): signal an error somehow?
+  err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
+                                        this, &size, outputData, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " Error in AudioConverterFillComplexBuffer()");
+      return 1;
+    } else {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+      return 1;
+    }
+  }
+
+  PaRingBufferSize bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
+
+  int32_t renderDelayUs =
+      static_cast<int32_t>(1e-3 * (outputTimeNs - nowNs) + 0.5);
+  renderDelayUs += static_cast<int32_t>(
+      (1.0e6 * bufSizeSamples) / _outDesiredFormat.mChannelsPerFrame /
+          _outDesiredFormat.mSampleRate +
+      0.5);
+
+  AtomicSet32(&_renderDelayUs, renderDelayUs);
+
+  return 0;
 }
 
-OSStatus AudioDeviceMac::implOutConverterProc(UInt32 *numberDataPackets,
-                                              AudioBufferList *data)
-{
-    DCHECK(data->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = *numberDataPackets
-        * _outDesiredFormat.mChannelsPerFrame;
+OSStatus AudioDeviceMac::implOutConverterProc(UInt32* numberDataPackets,
+                                              AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  PaRingBufferSize numSamples =
+      *numberDataPackets * _outDesiredFormat.mChannelsPerFrame;
 
-    data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
-    // Always give the converter as much as it wants, zero padding as required.
-    data->mBuffers->mDataByteSize = *numberDataPackets
-        * _outDesiredFormat.mBytesPerPacket;
-    data->mBuffers->mData = _renderConvertData;
-    memset(_renderConvertData, 0, sizeof(_renderConvertData));
+  data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
+  // Always give the converter as much as it wants, zero padding as required.
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _outDesiredFormat.mBytesPerPacket;
+  data->mBuffers->mData = _renderConvertData;
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
 
-    PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
+  PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
 
-    kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_signal_all() error: %d", kernErr);
+  kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_signal_all() error: %d", kernErr);
+    return 1;
+  }
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList* inputData,
+                                            const AudioTimeStamp* inputTime) {
+  OSStatus err = noErr;
+  UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStopRec) {
+    _critSect.Enter();
+    if (_doStopRec) {
+      // This will be signalled only when a shared device is not in use.
+      WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      if (err == noErr) {
+        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                     " Recording device stopped");
+      }
+
+      _doStopRec = false;
+      _stopEventRec.Set();
+      _critSect.Leave();
+      return 0;
+    }
+    _critSect.Leave();
+  }
+
+  if (!_recording) {
+    // Allow above checks to avoid a timeout on stopping capture.
+    return 0;
+  }
+
+  PaRingBufferSize bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
+
+  int32_t captureDelayUs =
+      static_cast<int32_t>(1e-3 * (nowNs - inputTimeNs) + 0.5);
+  captureDelayUs += static_cast<int32_t>((1.0e6 * bufSizeSamples) /
+                                             _inStreamFormat.mChannelsPerFrame /
+                                             _inStreamFormat.mSampleRate +
+                                         0.5);
+
+  AtomicSet32(&_captureDelayUs, captureDelayUs);
+
+  RTC_DCHECK(inputData->mNumberBuffers == 1);
+  PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize *
+                                _inStreamFormat.mChannelsPerFrame /
+                                _inStreamFormat.mBytesPerPacket;
+  PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
+                         numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_signal_all() error: %d", kernErr);
+  }
+
+  return err;
+}
+
+OSStatus AudioDeviceMac::implInConverterProc(UInt32* numberDataPackets,
+                                             AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  PaRingBufferSize numSamples =
+      *numberDataPackets * _inStreamFormat.mChannelsPerFrame;
+
+  while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
+      if (signal == 0) {
+        // The capture device is no longer alive; stop the worker thread.
+        *numberDataPackets = 0;
         return 1;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " semaphore_wait() error: %d", kernErr);
     }
+  }
 
-    return 0;
+  // Pass the read pointer directly to the converter to avoid a memcpy.
+  void* dummyPtr;
+  PaRingBufferSize dummySize;
+  PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
+                                  &data->mBuffers->mData, &numSamples,
+                                  &dummyPtr, &dummySize);
+  PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
+
+  data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
+  *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _inStreamFormat.mBytesPerPacket;
+
+  return 0;
 }
 
-OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList *inputData,
-                                            const AudioTimeStamp *inputTime)
-{
-    OSStatus err = noErr;
-    UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
-    UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
-    // Check if we should close down audio device
-    // Double-checked locking optimization to remove locking overhead
-    if (_doStopRec)
-    {
-        _critSect.Enter();
-        if (_doStopRec)
-        {
-            // This will be signalled only when a shared device is not in use.
-            WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
-            WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_inputDeviceID,
-                                                          _inDeviceIOProcID));
-            if (err == noErr)
-            {
-                WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                             _id, " Recording device stopped");
-            }
-
-            _doStopRec = false;
-            _stopEventRec.Set();
-            _critSect.Leave();
-            return 0;
-        }
-        _critSect.Leave();
-    }
-
-    if (!_recording)
-    {
-        // Allow above checks to avoid a timeout on stopping capture.
-        return 0;
-    }
-
-    PaRingBufferSize bufSizeSamples =
-        PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
-
-    int32_t captureDelayUs = static_cast<int32_t> (1e-3 * (nowNs - inputTimeNs)
-        + 0.5);
-    captureDelayUs
-        += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
-            / _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mSampleRate
-            + 0.5);
-
-    AtomicSet32(&_captureDelayUs, captureDelayUs);
-
-    DCHECK(inputData->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize
-        * _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mBytesPerPacket;
-    PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
-                           numSamples);
-
-    kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_signal_all() error: %d", kernErr);
-    }
-
-    return err;
+bool AudioDeviceMac::RunRender(void* ptrThis) {
+  return static_cast<AudioDeviceMac*>(ptrThis)->RenderWorkerThread();
 }
 
-OSStatus AudioDeviceMac::implInConverterProc(UInt32 *numberDataPackets,
-                                             AudioBufferList *data)
-{
-    DCHECK(data->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = *numberDataPackets
-        * _inStreamFormat.mChannelsPerFrame;
+bool AudioDeviceMac::RenderWorkerThread() {
+  PaRingBufferSize numSamples =
+      ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+  while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer) -
+             _renderDelayOffsetSamples <
+         numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
 
-    while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples)
-    {
-        mach_timespec_t timeout;
-        timeout.tv_sec = 0;
-        timeout.tv_nsec = TIMER_PERIOD_MS;
-
-        kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
-        if (kernErr == KERN_OPERATION_TIMED_OUT)
-        {
-            int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
-            if (signal == 0)
-            {
-                // The capture device is no longer alive; stop the worker thread.
-                *numberDataPackets = 0;
-                return 1;
-            }
-        } else if (kernErr != KERN_SUCCESS)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " semaphore_wait() error: %d", kernErr);
-        }
-    }
-
-    // Pass the read pointer directly to the converter to avoid a memcpy.
-    void* dummyPtr;
-    PaRingBufferSize dummySize;
-    PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
-                                    &data->mBuffers->mData, &numSamples,
-                                    &dummyPtr, &dummySize);
-    PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
-
-    data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
-    *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
-    data->mBuffers->mDataByteSize = *numberDataPackets
-        * _inStreamFormat.mBytesPerPacket;
-
-    return 0;
-}
-
-bool AudioDeviceMac::RunRender(void* ptrThis)
-{
-    return static_cast<AudioDeviceMac*> (ptrThis)->RenderWorkerThread();
-}
-
-bool AudioDeviceMac::RenderWorkerThread()
-{
-    PaRingBufferSize numSamples = ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
-        * _outDesiredFormat.mChannelsPerFrame;
-    while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer)
-        - _renderDelayOffsetSamples < numSamples)
-    {
-        mach_timespec_t timeout;
-        timeout.tv_sec = 0;
-        timeout.tv_nsec = TIMER_PERIOD_MS;
-
-        kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
-        if (kernErr == KERN_OPERATION_TIMED_OUT)
-        {
-            int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
-            if (signal == 0)
-            {
-                // The render device is no longer alive; stop the worker thread.
-                return false;
-            }
-        } else if (kernErr != KERN_SUCCESS)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " semaphore_timedwait() error: %d", kernErr);
-        }
-    }
-
-    int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
-
-    if (!_ptrAudioBuffer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  capture AudioBuffer is invalid");
+    kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
+      if (signal == 0) {
+        // The render device is no longer alive; stop the worker thread.
         return false;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " semaphore_timedwait() error: %d", kernErr);
     }
+  }
 
-    // Ask for new PCM data to be played out using the AudioDeviceBuffer.
-    uint32_t nSamples =
-        _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+  int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
 
-    nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
-    if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  invalid number of output samples(%d)", nSamples);
+  if (!_ptrAudioBuffer) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  capture AudioBuffer is invalid");
+    return false;
+  }
+
+  // Ask for new PCM data to be played out using the AudioDeviceBuffer.
+  uint32_t nSamples =
+      _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+
+  nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+  if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  invalid number of output samples(%d)", nSamples);
+  }
+
+  uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
+
+  SInt16* pPlayBuffer = (SInt16*)&playBuffer;
+  if (_macBookProPanRight && (_playChannels == 2)) {
+    // Mix entirely into the right channel and zero the left channel.
+    SInt32 sampleInt32 = 0;
+    for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx += 2) {
+      sampleInt32 = pPlayBuffer[sampleIdx];
+      sampleInt32 += pPlayBuffer[sampleIdx + 1];
+      sampleInt32 /= 2;
+
+      if (sampleInt32 > 32767) {
+        sampleInt32 = 32767;
+      } else if (sampleInt32 < -32768) {
+        sampleInt32 = -32768;
+      }
+
+      pPlayBuffer[sampleIdx] = 0;
+      pPlayBuffer[sampleIdx + 1] = static_cast<SInt16>(sampleInt32);
     }
+  }
 
-    uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
+  PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
 
-    SInt16 *pPlayBuffer = (SInt16 *) &playBuffer;
-    if (_macBookProPanRight && (_playChannels == 2))
-    {
-        // Mix entirely into the right channel and zero the left channel.
-        SInt32 sampleInt32 = 0;
-        for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx
-            += 2)
-        {
-            sampleInt32 = pPlayBuffer[sampleIdx];
-            sampleInt32 += pPlayBuffer[sampleIdx + 1];
-            sampleInt32 /= 2;
-
-            if (sampleInt32 > 32767)
-            {
-                sampleInt32 = 32767;
-            } else if (sampleInt32 < -32768)
-            {
-                sampleInt32 = -32768;
-            }
-
-            pPlayBuffer[sampleIdx] = 0;
-            pPlayBuffer[sampleIdx + 1] = static_cast<SInt16> (sampleInt32);
-        }
-    }
-
-    PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
-
-    return true;
+  return true;
 }
 
-bool AudioDeviceMac::RunCapture(void* ptrThis)
-{
-    return static_cast<AudioDeviceMac*> (ptrThis)->CaptureWorkerThread();
+bool AudioDeviceMac::RunCapture(void* ptrThis) {
+  return static_cast<AudioDeviceMac*>(ptrThis)->CaptureWorkerThread();
 }
 
-bool AudioDeviceMac::CaptureWorkerThread()
-{
-    OSStatus err = noErr;
-    UInt32 noRecSamples = ENGINE_REC_BUF_SIZE_IN_SAMPLES
-        * _inDesiredFormat.mChannelsPerFrame;
-    SInt16 recordBuffer[noRecSamples];
-    UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
+bool AudioDeviceMac::CaptureWorkerThread() {
+  OSStatus err = noErr;
+  UInt32 noRecSamples =
+      ENGINE_REC_BUF_SIZE_IN_SAMPLES * _inDesiredFormat.mChannelsPerFrame;
+  SInt16 recordBuffer[noRecSamples];
+  UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
 
-    AudioBufferList engineBuffer;
-    engineBuffer.mNumberBuffers = 1; // Interleaved channels.
-    engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
-    engineBuffer.mBuffers->mDataByteSize = _inDesiredFormat.mBytesPerPacket
-        * noRecSamples;
-    engineBuffer.mBuffers->mData = recordBuffer;
+  AudioBufferList engineBuffer;
+  engineBuffer.mNumberBuffers = 1;  // Interleaved channels.
+  engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
+  engineBuffer.mBuffers->mDataByteSize =
+      _inDesiredFormat.mBytesPerPacket * noRecSamples;
+  engineBuffer.mBuffers->mData = recordBuffer;
 
-    err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
-                                          this, &size, &engineBuffer, NULL);
-    if (err != noErr)
-    {
-        if (err == 1)
-        {
-            // This is our own error.
-            return false;
-        } else
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioConverterFillComplexBuffer()",
-                     (const char *) &err);
-            return false;
-        }
+  err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
+                                        this, &size, &engineBuffer, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      return false;
+    } else {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+      return false;
+    }
+  }
+
+  // TODO(xians): what if the returned size is incorrect?
+  if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES) {
+    uint32_t currentMicLevel(0);
+    uint32_t newMicLevel(0);
+    int32_t msecOnPlaySide;
+    int32_t msecOnRecordSide;
+
+    int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+    int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+
+    msecOnPlaySide =
+        static_cast<int32_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+    msecOnRecordSide =
+        static_cast<int32_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+
+    if (!_ptrAudioBuffer) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   "  capture AudioBuffer is invalid");
+      return false;
     }
 
-    // TODO(xians): what if the returned size is incorrect?
-    if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES)
-    {
-        uint32_t currentMicLevel(0);
-        uint32_t newMicLevel(0);
-        int32_t msecOnPlaySide;
-        int32_t msecOnRecordSide;
+    // store the recorded buffer (no action will be taken if the
+    // #recorded samples is not a full buffer)
+    _ptrAudioBuffer->SetRecordedBuffer((int8_t*)&recordBuffer, (uint32_t)size);
 
-        int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-        int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-
-        msecOnPlaySide = static_cast<int32_t> (1e-3 * (renderDelayUs +
-                                                       _renderLatencyUs) + 0.5);
-        msecOnRecordSide = static_cast<int32_t> (1e-3 * (captureDelayUs +
-                                                         _captureLatencyUs) +
-                                                 0.5);
-
-        if (!_ptrAudioBuffer)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "  capture AudioBuffer is invalid");
-            return false;
-        }
-
-        // store the recorded buffer (no action will be taken if the
-        // #recorded samples is not a full buffer)
-        _ptrAudioBuffer->SetRecordedBuffer((int8_t*) &recordBuffer,
-                                           (uint32_t) size);
-
-        if (AGC())
-        {
-            // store current mic level in the audio buffer if AGC is enabled
-            if (MicrophoneVolume(currentMicLevel) == 0)
-            {
-                // this call does not affect the actual microphone volume
-                _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
-            }
-        }
-
-        _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
-
-        _ptrAudioBuffer->SetTypingStatus(KeyPressed());
-
-        // deliver recorded samples at specified sample rate, mic level etc.
-        // to the observer using callback
-        _ptrAudioBuffer->DeliverRecordedData();
-
-        if (AGC())
-        {
-            newMicLevel = _ptrAudioBuffer->NewMicLevel();
-            if (newMicLevel != 0)
-            {
-                // The VQE will only deliver non-zero microphone levels when
-                // a change is needed.
-                // Set this new mic level (received from the observer as return
-                // value in the callback).
-                WEBRTC_TRACE(kTraceStream, kTraceAudioDevice,
-                             _id, "  AGC change of volume: old=%u => new=%u",
-                             currentMicLevel, newMicLevel);
-                if (SetMicrophoneVolume(newMicLevel) == -1)
-                {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  the required modification of the microphone "
-                                 "volume failed");
-                }
-            }
+    if (AGC()) {
+      // Use mod to ensure we check the volume on the first pass.
+      if (get_mic_volume_counter_ms_ % kGetMicVolumeIntervalMs == 0) {
+        get_mic_volume_counter_ms_ = 0;
+        // store current mic level in the audio buffer if AGC is enabled
+        if (MicrophoneVolume(currentMicLevel) == 0) {
+          // this call does not affect the actual microphone volume
+          _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
         }
+      }
+      get_mic_volume_counter_ms_ += kBufferSizeMs;
     }
 
-    return true;
+    _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
+
+    _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+    // deliver recorded samples at specified sample rate, mic level etc.
+    // to the observer using callback
+    _ptrAudioBuffer->DeliverRecordedData();
+
+    if (AGC()) {
+      newMicLevel = _ptrAudioBuffer->NewMicLevel();
+      if (newMicLevel != 0) {
+        // The VQE will only deliver non-zero microphone levels when
+        // a change is needed.
+        // Set this new mic level (received from the observer as return
+        // value in the callback).
+        WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+                     "  AGC change of volume: old=%u => new=%u",
+                     currentMicLevel, newMicLevel);
+        if (SetMicrophoneVolume(newMicLevel) == -1) {
+          WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                       "  the required modification of the microphone "
+                       "volume failed");
+        }
+      }
+    }
+  }
+
+  return true;
 }
 
 bool AudioDeviceMac::KeyPressed() {
   bool key_down = false;
   // Loop through all Mac virtual key constant values.
-  for (unsigned int key_index = 0;
-                    key_index < arraysize(prev_key_state_);
-                    ++key_index) {
-    bool keyState = CGEventSourceKeyState(
-                             kCGEventSourceStateHIDSystemState,
-                             key_index);
+  for (unsigned int key_index = 0; key_index < arraysize(prev_key_state_);
+       ++key_index) {
+    bool keyState =
+        CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, key_index);
     // A false -> true change in keymap means a key is pressed.
     key_down |= (keyState && !prev_key_state_[key_index]);
     // Save current state.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.h b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.h
index 9029a01701..ca3a51997d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_mac.h
@@ -15,7 +15,7 @@
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #include <AudioToolbox/AudioConverter.h>
 #include <CoreAudio/CoreAudio.h>
@@ -23,350 +23,359 @@
 
 struct PaUtilRingBuffer;
 
-namespace webrtc
-{
+namespace rtc {
+class PlatformThread;
+}  // namespace rtc
+
+namespace webrtc {
 class EventWrapper;
-class ThreadWrapper;
 
 const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
 const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
 
-const uint32_t N_REC_CHANNELS = 1; // default is mono recording
-const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
+const uint32_t N_REC_CHANNELS = 1;   // default is mono recording
+const uint32_t N_PLAY_CHANNELS = 2;  // default is stereo playout
 const uint32_t N_DEVICE_CHANNELS = 64;
 
-const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
-const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
+const int kBufferSizeMs = 10;
+
+const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES =
+    N_REC_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
+const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES =
+    N_PLAY_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
 
 const int N_BLOCKS_IO = 2;
-const int N_BUFFERS_IN = 2;  // Must be at least N_BLOCKS_IO.
+const int N_BUFFERS_IN = 2;   // Must be at least N_BLOCKS_IO.
 const int N_BUFFERS_OUT = 3;  // Must be at least N_BLOCKS_IO.
 
-const uint32_t TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
+const uint32_t TIMER_PERIOD_MS = 2 * 10 * N_BLOCKS_IO * 1000000;
 
 const uint32_t REC_BUF_SIZE_IN_SAMPLES =
     ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
 const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
     ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
 
-class AudioDeviceMac: public AudioDeviceGeneric
-{
-public:
-    AudioDeviceMac(const int32_t id);
-    ~AudioDeviceMac();
+const int kGetMicVolumeIntervalMs = 1000;
 
-    // Retrieve the currently utilized audio layer
-    virtual int32_t
-        ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
+class AudioDeviceMac : public AudioDeviceGeneric {
+ public:
+  AudioDeviceMac(const int32_t id);
+  ~AudioDeviceMac();
 
-    // Main initializaton and termination
-    virtual int32_t Init();
-    virtual int32_t Terminate();
-    virtual bool Initialized() const;
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const;
 
-    // Device enumeration
-    virtual int16_t PlayoutDevices();
-    virtual int16_t RecordingDevices();
-    virtual int32_t PlayoutDeviceName(
-        uint16_t index,
-        char name[kAdmMaxDeviceNameSize],
-        char guid[kAdmMaxGuidSize]);
-    virtual int32_t RecordingDeviceName(
-        uint16_t index,
-        char name[kAdmMaxDeviceNameSize],
-        char guid[kAdmMaxGuidSize]);
+  // Main initializaton and termination
+  virtual int32_t Init();
+  virtual int32_t Terminate();
+  virtual bool Initialized() const;
 
-    // Device selection
-    virtual int32_t SetPlayoutDevice(uint16_t index);
-    virtual int32_t SetPlayoutDevice(
-        AudioDeviceModule::WindowsDeviceType device);
-    virtual int32_t SetRecordingDevice(uint16_t index);
-    virtual int32_t SetRecordingDevice(
-        AudioDeviceModule::WindowsDeviceType device);
+  // Device enumeration
+  virtual int16_t PlayoutDevices();
+  virtual int16_t RecordingDevices();
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]);
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]);
 
-    // Audio transport initialization
-    virtual int32_t PlayoutIsAvailable(bool& available);
-    virtual int32_t InitPlayout();
-    virtual bool PlayoutIsInitialized() const;
-    virtual int32_t RecordingIsAvailable(bool& available);
-    virtual int32_t InitRecording();
-    virtual bool RecordingIsInitialized() const;
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index);
+  virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+  virtual int32_t SetRecordingDevice(uint16_t index);
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device);
 
-    // Audio transport control
-    virtual int32_t StartPlayout();
-    virtual int32_t StopPlayout();
-    virtual bool Playing() const;
-    virtual int32_t StartRecording();
-    virtual int32_t StopRecording();
-    virtual bool Recording() const;
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available);
+  virtual int32_t InitPlayout();
+  virtual bool PlayoutIsInitialized() const;
+  virtual int32_t RecordingIsAvailable(bool& available);
+  virtual int32_t InitRecording();
+  virtual bool RecordingIsInitialized() const;
 
-    // Microphone Automatic Gain Control (AGC)
-    virtual int32_t SetAGC(bool enable);
-    virtual bool AGC() const;
+  // Audio transport control
+  virtual int32_t StartPlayout();
+  virtual int32_t StopPlayout();
+  virtual bool Playing() const;
+  virtual int32_t StartRecording();
+  virtual int32_t StopRecording();
+  virtual bool Recording() const;
 
-    // Volume control based on the Windows Wave API (Windows only)
-    virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
-    virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
-                                  uint16_t& volumeRight) const;
+  // Microphone Automatic Gain Control (AGC)
+  virtual int32_t SetAGC(bool enable);
+  virtual bool AGC() const;
 
-    // Audio mixer initialization
-    virtual int32_t InitSpeaker();
-    virtual bool SpeakerIsInitialized() const;
-    virtual int32_t InitMicrophone();
-    virtual bool MicrophoneIsInitialized() const;
+  // Volume control based on the Windows Wave API (Windows only)
+  virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
+  virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
+                                uint16_t& volumeRight) const;
 
-    // Speaker volume controls
-    virtual int32_t SpeakerVolumeIsAvailable(bool& available);
-    virtual int32_t SetSpeakerVolume(uint32_t volume);
-    virtual int32_t SpeakerVolume(uint32_t& volume) const;
-    virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
-    virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
-    virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker();
+  virtual bool SpeakerIsInitialized() const;
+  virtual int32_t InitMicrophone();
+  virtual bool MicrophoneIsInitialized() const;
 
-    // Microphone volume controls
-    virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
-    virtual int32_t SetMicrophoneVolume(uint32_t volume);
-    virtual int32_t MicrophoneVolume(uint32_t& volume) const;
-    virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
-    virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
-    virtual int32_t
-        MicrophoneVolumeStepSize(uint16_t& stepSize) const;
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available);
+  virtual int32_t SetSpeakerVolume(uint32_t volume);
+  virtual int32_t SpeakerVolume(uint32_t& volume) const;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
 
-    // Microphone mute control
-    virtual int32_t MicrophoneMuteIsAvailable(bool& available);
-    virtual int32_t SetMicrophoneMute(bool enable);
-    virtual int32_t MicrophoneMute(bool& enabled) const;
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
+  virtual int32_t SetMicrophoneVolume(uint32_t volume);
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
 
-    // Speaker mute control
-    virtual int32_t SpeakerMuteIsAvailable(bool& available);
-    virtual int32_t SetSpeakerMute(bool enable);
-    virtual int32_t SpeakerMute(bool& enabled) const;
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available);
+  virtual int32_t SetMicrophoneMute(bool enable);
+  virtual int32_t MicrophoneMute(bool& enabled) const;
 
-    // Microphone boost control
-    virtual int32_t MicrophoneBoostIsAvailable(bool& available);
-    virtual int32_t SetMicrophoneBoost(bool enable);
-    virtual int32_t MicrophoneBoost(bool& enabled) const;
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available);
+  virtual int32_t SetSpeakerMute(bool enable);
+  virtual int32_t SpeakerMute(bool& enabled) const;
 
-    // Stereo support
-    virtual int32_t StereoPlayoutIsAvailable(bool& available);
-    virtual int32_t SetStereoPlayout(bool enable);
-    virtual int32_t StereoPlayout(bool& enabled) const;
-    virtual int32_t StereoRecordingIsAvailable(bool& available);
-    virtual int32_t SetStereoRecording(bool enable);
-    virtual int32_t StereoRecording(bool& enabled) const;
+  // Microphone boost control
+  virtual int32_t MicrophoneBoostIsAvailable(bool& available);
+  virtual int32_t SetMicrophoneBoost(bool enable);
+  virtual int32_t MicrophoneBoost(bool& enabled) const;
 
-    // Delay information and control
-    virtual int32_t
-        SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                         uint16_t sizeMS);
-    virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
-                                  uint16_t& sizeMS) const;
-    virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
-    virtual int32_t RecordingDelay(uint16_t& delayMS) const;
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available);
+  virtual int32_t SetStereoPlayout(bool enable);
+  virtual int32_t StereoPlayout(bool& enabled) const;
+  virtual int32_t StereoRecordingIsAvailable(bool& available);
+  virtual int32_t SetStereoRecording(bool enable);
+  virtual int32_t StereoRecording(bool& enabled) const;
 
-    // CPU load
-    virtual int32_t CPULoad(uint16_t& load) const;
+  // Delay information and control
+  virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+                                   uint16_t sizeMS);
+  virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                uint16_t& sizeMS) const;
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
+  virtual int32_t RecordingDelay(uint16_t& delayMS) const;
 
-    virtual bool PlayoutWarning() const;
-    virtual bool PlayoutError() const;
-    virtual bool RecordingWarning() const;
-    virtual bool RecordingError() const;
-    virtual void ClearPlayoutWarning();
-    virtual void ClearPlayoutError();
-    virtual void ClearRecordingWarning();
-    virtual void ClearRecordingError();
+  // CPU load
+  virtual int32_t CPULoad(uint16_t& load) const;
 
-    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+  virtual bool PlayoutWarning() const;
+  virtual bool PlayoutError() const;
+  virtual bool RecordingWarning() const;
+  virtual bool RecordingError() const;
+  virtual void ClearPlayoutWarning();
+  virtual void ClearPlayoutError();
+  virtual void ClearRecordingWarning();
+  virtual void ClearRecordingError();
 
-private:
-    virtual int32_t MicrophoneIsAvailable(bool& available);
-    virtual int32_t SpeakerIsAvailable(bool& available);
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
 
-    static void AtomicSet32(int32_t* theValue, int32_t newValue);
-    static int32_t AtomicGet32(int32_t* theValue);
+ private:
+  virtual int32_t MicrophoneIsAvailable(bool& available);
+  virtual int32_t SpeakerIsAvailable(bool& available);
 
-    static void logCAMsg(const TraceLevel level,
-                         const TraceModule module,
-                         const int32_t id, const char *msg,
-                         const char *err);
+  static void AtomicSet32(int32_t* theValue, int32_t newValue);
+  static int32_t AtomicGet32(int32_t* theValue);
 
-    int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
-                             AudioDeviceID scopedDeviceIds[],
-                             const uint32_t deviceListLength);
+  static void logCAMsg(const TraceLevel level,
+                       const TraceModule module,
+                       const int32_t id,
+                       const char* msg,
+                       const char* err);
 
-    int32_t GetDeviceName(const AudioObjectPropertyScope scope,
-                          const uint16_t index, char* name);
+  int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
+                           AudioDeviceID scopedDeviceIds[],
+                           const uint32_t deviceListLength);
 
-    int32_t InitDevice(uint16_t userDeviceIndex,
-                       AudioDeviceID& deviceId, bool isInput);
+  int32_t GetDeviceName(const AudioObjectPropertyScope scope,
+                        const uint16_t index,
+                        char* name);
 
-    // Always work with our preferred playout format inside VoE.
-    // Then convert the output to the OS setting using an AudioConverter.
-    OSStatus SetDesiredPlayoutFormat();
+  int32_t InitDevice(uint16_t userDeviceIndex,
+                     AudioDeviceID& deviceId,
+                     bool isInput);
 
-    static OSStatus
-        objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
-                           const AudioObjectPropertyAddress addresses[],
-                           void* clientData);
+  // Always work with our preferred playout format inside VoE.
+  // Then convert the output to the OS setting using an AudioConverter.
+  OSStatus SetDesiredPlayoutFormat();
 
-    OSStatus
-        implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
-                               const AudioObjectPropertyAddress addresses[]);
+  static OSStatus objectListenerProc(
+      AudioObjectID objectId,
+      UInt32 numberAddresses,
+      const AudioObjectPropertyAddress addresses[],
+      void* clientData);
 
-    int32_t HandleDeviceChange();
+  OSStatus implObjectListenerProc(AudioObjectID objectId,
+                                  UInt32 numberAddresses,
+                                  const AudioObjectPropertyAddress addresses[]);
 
-    int32_t
-        HandleStreamFormatChange(AudioObjectID objectId,
+  int32_t HandleDeviceChange();
+
+  int32_t HandleStreamFormatChange(AudioObjectID objectId,
+                                   AudioObjectPropertyAddress propertyAddress);
+
+  int32_t HandleDataSourceChange(AudioObjectID objectId,
                                  AudioObjectPropertyAddress propertyAddress);
 
-    int32_t
-        HandleDataSourceChange(AudioObjectID objectId,
-                               AudioObjectPropertyAddress propertyAddress);
+  int32_t HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
 
-    int32_t
-        HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
+  static OSStatus deviceIOProc(AudioDeviceID device,
+                               const AudioTimeStamp* now,
+                               const AudioBufferList* inputData,
+                               const AudioTimeStamp* inputTime,
+                               AudioBufferList* outputData,
+                               const AudioTimeStamp* outputTime,
+                               void* clientData);
 
-    static OSStatus deviceIOProc(AudioDeviceID device,
-                                 const AudioTimeStamp *now,
-                                 const AudioBufferList *inputData,
-                                 const AudioTimeStamp *inputTime,
-                                 AudioBufferList *outputData,
+  static OSStatus outConverterProc(
+      AudioConverterRef audioConverter,
+      UInt32* numberDataPackets,
+      AudioBufferList* data,
+      AudioStreamPacketDescription** dataPacketDescription,
+      void* userData);
+
+  static OSStatus inDeviceIOProc(AudioDeviceID device,
+                                 const AudioTimeStamp* now,
+                                 const AudioBufferList* inputData,
+                                 const AudioTimeStamp* inputTime,
+                                 AudioBufferList* outputData,
                                  const AudioTimeStamp* outputTime,
-                                 void *clientData);
+                                 void* clientData);
 
-    static OSStatus
-        outConverterProc(AudioConverterRef audioConverter,
-                         UInt32 *numberDataPackets, AudioBufferList *data,
-                         AudioStreamPacketDescription **dataPacketDescription,
-                         void *userData);
+  static OSStatus inConverterProc(
+      AudioConverterRef audioConverter,
+      UInt32* numberDataPackets,
+      AudioBufferList* data,
+      AudioStreamPacketDescription** dataPacketDescription,
+      void* inUserData);
 
-    static OSStatus inDeviceIOProc(AudioDeviceID device,
-                                   const AudioTimeStamp *now,
-                                   const AudioBufferList *inputData,
-                                   const AudioTimeStamp *inputTime,
-                                   AudioBufferList *outputData,
-                                   const AudioTimeStamp *outputTime,
-                                   void *clientData);
+  OSStatus implDeviceIOProc(const AudioBufferList* inputData,
+                            const AudioTimeStamp* inputTime,
+                            AudioBufferList* outputData,
+                            const AudioTimeStamp* outputTime);
 
-    static OSStatus
-        inConverterProc(AudioConverterRef audioConverter,
-                        UInt32 *numberDataPackets, AudioBufferList *data,
-                        AudioStreamPacketDescription **dataPacketDescription,
-                        void *inUserData);
+  OSStatus implOutConverterProc(UInt32* numberDataPackets,
+                                AudioBufferList* data);
 
-    OSStatus implDeviceIOProc(const AudioBufferList *inputData,
-                              const AudioTimeStamp *inputTime,
-                              AudioBufferList *outputData,
-                              const AudioTimeStamp *outputTime);
+  OSStatus implInDeviceIOProc(const AudioBufferList* inputData,
+                              const AudioTimeStamp* inputTime);
 
-    OSStatus implOutConverterProc(UInt32 *numberDataPackets,
-                                  AudioBufferList *data);
+  OSStatus implInConverterProc(UInt32* numberDataPackets,
+                               AudioBufferList* data);
 
-    OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
-                                const AudioTimeStamp *inputTime);
+  static bool RunCapture(void*);
+  static bool RunRender(void*);
+  bool CaptureWorkerThread();
+  bool RenderWorkerThread();
 
-    OSStatus implInConverterProc(UInt32 *numberDataPackets,
-                                 AudioBufferList *data);
+  bool KeyPressed();
 
-    static bool RunCapture(void*);
-    static bool RunRender(void*);
-    bool CaptureWorkerThread();
-    bool RenderWorkerThread();
+  AudioDeviceBuffer* _ptrAudioBuffer;
 
-    bool KeyPressed();
+  CriticalSectionWrapper& _critSect;
 
-    AudioDeviceBuffer* _ptrAudioBuffer;
+  EventWrapper& _stopEventRec;
+  EventWrapper& _stopEvent;
 
-    CriticalSectionWrapper& _critSect;
+  // TODO(pbos): Replace with direct members, just start/stop, no need to
+  // recreate the thread.
+  // Only valid/running between calls to StartRecording and StopRecording.
+  rtc::scoped_ptr<rtc::PlatformThread> capture_worker_thread_;
 
-    EventWrapper& _stopEventRec;
-    EventWrapper& _stopEvent;
+  // Only valid/running between calls to StartPlayout and StopPlayout.
+  rtc::scoped_ptr<rtc::PlatformThread> render_worker_thread_;
 
-    // Only valid/running between calls to StartRecording and StopRecording.
-    rtc::scoped_ptr<ThreadWrapper> capture_worker_thread_;
+  int32_t _id;
 
-    // Only valid/running between calls to StartPlayout and StopPlayout.
-    rtc::scoped_ptr<ThreadWrapper> render_worker_thread_;
+  AudioMixerManagerMac _mixerManager;
 
-    int32_t _id;
-
-    AudioMixerManagerMac _mixerManager;
-
-    uint16_t _inputDeviceIndex;
-    uint16_t _outputDeviceIndex;
-    AudioDeviceID _inputDeviceID;
-    AudioDeviceID _outputDeviceID;
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  AudioDeviceID _inputDeviceID;
+  AudioDeviceID _outputDeviceID;
 #if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
-    AudioDeviceIOProcID _inDeviceIOProcID;
-    AudioDeviceIOProcID _deviceIOProcID;
+  AudioDeviceIOProcID _inDeviceIOProcID;
+  AudioDeviceIOProcID _deviceIOProcID;
 #endif
-    bool _inputDeviceIsSpecified;
-    bool _outputDeviceIsSpecified;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
 
-    uint8_t _recChannels;
-    uint8_t _playChannels;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
 
-    Float32* _captureBufData;
-    SInt16* _renderBufData;
+  Float32* _captureBufData;
+  SInt16* _renderBufData;
 
-    SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
+  SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
 
-    AudioDeviceModule::BufferType _playBufType;
+  AudioDeviceModule::BufferType _playBufType;
 
-    bool _initialized;
-    bool _isShutDown;
-    bool _recording;
-    bool _playing;
-    bool _recIsInitialized;
-    bool _playIsInitialized;
-    bool _AGC;
+  bool _initialized;
+  bool _isShutDown;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+  bool _AGC;
 
-    // Atomically set varaibles
-    int32_t _renderDeviceIsAlive;
-    int32_t _captureDeviceIsAlive;
+  // Atomically set varaibles
+  int32_t _renderDeviceIsAlive;
+  int32_t _captureDeviceIsAlive;
 
-    bool _twoDevices;
-    bool _doStop; // For play if not shared device or play+rec if shared device
-    bool _doStopRec; // For rec if not shared device
-    bool _macBookPro;
-    bool _macBookProPanRight;
+  bool _twoDevices;
+  bool _doStop;  // For play if not shared device or play+rec if shared device
+  bool _doStopRec;  // For rec if not shared device
+  bool _macBookPro;
+  bool _macBookProPanRight;
 
-    AudioConverterRef _captureConverter;
-    AudioConverterRef _renderConverter;
+  AudioConverterRef _captureConverter;
+  AudioConverterRef _renderConverter;
 
-    AudioStreamBasicDescription _outStreamFormat;
-    AudioStreamBasicDescription _outDesiredFormat;
-    AudioStreamBasicDescription _inStreamFormat;
-    AudioStreamBasicDescription _inDesiredFormat;
+  AudioStreamBasicDescription _outStreamFormat;
+  AudioStreamBasicDescription _outDesiredFormat;
+  AudioStreamBasicDescription _inStreamFormat;
+  AudioStreamBasicDescription _inDesiredFormat;
 
-    uint32_t _captureLatencyUs;
-    uint32_t _renderLatencyUs;
+  uint32_t _captureLatencyUs;
+  uint32_t _renderLatencyUs;
 
-    // Atomically set variables
-    mutable int32_t _captureDelayUs;
-    mutable int32_t _renderDelayUs;
+  // Atomically set variables
+  mutable int32_t _captureDelayUs;
+  mutable int32_t _renderDelayUs;
 
-    int32_t _renderDelayOffsetSamples;
+  int32_t _renderDelayOffsetSamples;
 
-    uint16_t _playBufDelayFixed; // fixed playback delay
+  uint16_t _playBufDelayFixed;  // fixed playback delay
 
-    uint16_t _playWarning;
-    uint16_t _playError;
-    uint16_t _recWarning;
-    uint16_t _recError;
+  uint16_t _playWarning;
+  uint16_t _playError;
+  uint16_t _recWarning;
+  uint16_t _recError;
 
-    PaUtilRingBuffer* _paCaptureBuffer;
-    PaUtilRingBuffer* _paRenderBuffer;
+  PaUtilRingBuffer* _paCaptureBuffer;
+  PaUtilRingBuffer* _paRenderBuffer;
 
-    semaphore_t _renderSemaphore;
-    semaphore_t _captureSemaphore;
+  semaphore_t _renderSemaphore;
+  semaphore_t _captureSemaphore;
 
-    int _captureBufSizeSamples;
-    int _renderBufSizeSamples;
+  int _captureBufSizeSamples;
+  int _renderBufSizeSamples;
 
-    // Typing detection
-    // 0x5c is key "9", after that comes function keys.
-    bool prev_key_state_[0x5d];
+  // Typing detection
+  // 0x5c is key "9", after that comes function keys.
+  bool prev_key_state_[0x5d];
+
+  int get_mic_volume_counter_ms_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_utility_mac.cc b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_utility_mac.cc
deleted file mode 100644
index ab1e15019e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_utility_mac.cc
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/mac/audio_device_utility_mac.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc
-{
-
-AudioDeviceUtilityMac::AudioDeviceUtilityMac(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
-}
-
-// ----------------------------------------------------------------------------
-//  AudioDeviceUtilityMac() - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceUtilityMac::~AudioDeviceUtilityMac()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
-    {
-        CriticalSectionScoped lock(&_critSect);
-
-        // free stuff here...
-    }
-
-    delete &_critSect;
-}
-
-int32_t AudioDeviceUtilityMac::Init()
-{
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
-                 "  OS info: %s", "OS X");
-
-    return 0;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_utility_mac.h b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_utility_mac.h
deleted file mode 100644
index 71e07a4ab1..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_device_utility_mac.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
-
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityMac: public AudioDeviceUtility
-{
-public:
-    AudioDeviceUtilityMac(const int32_t id);
-    ~AudioDeviceUtilityMac();
-
-    virtual int32_t Init();
-
-private:
-    CriticalSectionWrapper& _critSect;
-    int32_t _id;
-};
-
-}  // namespace webrtc
-
-#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_UTILITY_MAC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
index 952dc11d8b..e7e0754695 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
@@ -9,1136 +9,991 @@
  */
 
 #include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
-#include <unistd.h>             // getpid()
+#include <unistd.h>  // getpid()
 
 namespace webrtc {
-	
-#define WEBRTC_CA_RETURN_ON_ERR(expr)                                     \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,    \
-                "Error in " #expr, (const char *)&err);                 \
-            return -1;                                                  \
-        }                                                               \
-    } while(0)
 
-#define WEBRTC_CA_LOG_ERR(expr)                                           \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,    \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                  \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+      return -1;                                                       \
+    }                                                                  \
+  } while (0)
 
-#define WEBRTC_CA_LOG_WARN(expr)                                           \
-    do {                                                                 \
-        err = expr;                                                      \
-        if (err != noErr) {                                              \
-            logCAMsg(kTraceWarning, kTraceAudioDevice, _id,  \
-                "Error in " #expr, (const char *)&err);                  \
-        }                                                                \
-    } while(0)
+#define WEBRTC_CA_LOG_ERR(expr)                                        \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+    }                                                                  \
+  } while (0)
 
-AudioMixerManagerMac::AudioMixerManagerMac(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
-    _inputDeviceID(kAudioObjectUnknown),
-    _outputDeviceID(kAudioObjectUnknown),
-    _noInputChannels(0),
-    _noOutputChannels(0)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s constructed", __FUNCTION__);
+#define WEBRTC_CA_LOG_WARN(expr)                                         \
+  do {                                                                   \
+    err = expr;                                                          \
+    if (err != noErr) {                                                  \
+      logCAMsg(kTraceWarning, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                     \
+    }                                                                    \
+  } while (0)
+
+AudioMixerManagerMac::AudioMixerManagerMac(const int32_t id)
+    : _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _id(id),
+      _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _noInputChannels(0),
+      _noOutputChannels(0) {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s constructed",
+               __FUNCTION__);
 }
 
-AudioMixerManagerMac::~AudioMixerManagerMac()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destructed", __FUNCTION__);
+AudioMixerManagerMac::~AudioMixerManagerMac() {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destructed",
+               __FUNCTION__);
 
-    Close();
+  Close();
 
-    delete &_critSect;
+  delete &_critSect;
 }
 
 // ============================================================================
 //	                                PUBLIC METHODS
 // ============================================================================
 
-int32_t AudioMixerManagerMac::Close()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+int32_t AudioMixerManagerMac::Close() {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    CloseSpeaker();
-    CloseMicrophone();
-
-    return 0;
+  CloseSpeaker();
+  CloseMicrophone();
 
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::CloseSpeaker()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+int32_t AudioMixerManagerMac::CloseSpeaker() {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    _outputDeviceID = kAudioObjectUnknown;
-    _noOutputChannels = 0;
+  _outputDeviceID = kAudioObjectUnknown;
+  _noOutputChannels = 0;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::CloseMicrophone()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
+int32_t AudioMixerManagerMac::CloseMicrophone() {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    _inputDeviceID = kAudioObjectUnknown;
-    _noInputChannels = 0;
+  _inputDeviceID = kAudioObjectUnknown;
+  _noInputChannels = 0;
 
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID)
-{
+int32_t AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::OpenSpeaker(id=%d)", deviceID);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  pid_t hogPid = -1;
+
+  _outputDeviceID = deviceID;
+
+  // Check which process, if any, has hogged the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyHogMode, kAudioDevicePropertyScopeOutput, 0};
+
+  size = sizeof(hogPid);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &hogPid));
+
+  if (hogPid == -1) {
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::OpenSpeaker(id=%d)", deviceID);
+                 " No process has hogged the input device");
+  }
+  // getpid() is apparently "always successful"
+  else if (hogPid == getpid()) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 " Our process has hogged the input device");
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Another process (pid = %d) has hogged the input device",
+                 static_cast<int>(hogPid));
 
-    CriticalSectionScoped lock(&_critSect);
+    return -1;
+  }
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    pid_t hogPid = -1;
+  // get number of channels from stream format
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
 
-    _outputDeviceID = deviceID;
+  // Get the stream format, to be able to read the number of channels.
+  AudioStreamBasicDescription streamFormat;
+  size = sizeof(AudioStreamBasicDescription);
+  memset(&streamFormat, 0, size);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &streamFormat));
 
-    // Check which process, if any, has hogged the device. 
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
-            kAudioDevicePropertyScopeOutput, 0 };
+  _noOutputChannels = streamFormat.mChannelsPerFrame;
 
-    size = sizeof(hogPid);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &hogPid));
+  return 0;
+}
 
-    if (hogPid == -1)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " No process has hogged the input device");
-    }
-    // getpid() is apparently "always successful"
-    else if (hogPid == getpid())
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Our process has hogged the input device");
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Another process (pid = %d) has hogged the input device",
-                     static_cast<int> (hogPid));
+int32_t AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::OpenMicrophone(id=%d)", deviceID);
 
-        return -1;
-    }
+  CriticalSectionScoped lock(&_critSect);
 
-    // get number of channels from stream format
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  pid_t hogPid = -1;
 
-    // Get the stream format, to be able to read the number of channels.
-    AudioStreamBasicDescription streamFormat;
-    size = sizeof(AudioStreamBasicDescription);
-    memset(&streamFormat, 0, size);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
+  _inputDeviceID = deviceID;
 
-    _noOutputChannels = streamFormat.mChannelsPerFrame;
+  // Check which process, if any, has hogged the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyHogMode, kAudioDevicePropertyScopeInput, 0};
+  size = sizeof(hogPid);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &hogPid));
+  if (hogPid == -1) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 " No process has hogged the input device");
+  }
+  // getpid() is apparently "always successful"
+  else if (hogPid == getpid()) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 " Our process has hogged the input device");
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Another process (pid = %d) has hogged the input device",
+                 static_cast<int>(hogPid));
+
+    return -1;
+  }
+
+  // get number of channels from stream format
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+
+  // Get the stream format, to be able to read the number of channels.
+  AudioStreamBasicDescription streamFormat;
+  size = sizeof(AudioStreamBasicDescription);
+  memset(&streamFormat, 0, size);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &streamFormat));
+
+  _noInputChannels = streamFormat.mChannelsPerFrame;
+
+  return 0;
+}
+
+bool AudioMixerManagerMac::SpeakerIsInitialized() const {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+  return (_outputDeviceID != kAudioObjectUnknown);
+}
+
+bool AudioMixerManagerMac::MicrophoneIsInitialized() const {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+  return (_inputDeviceID != kAudioObjectUnknown);
+}
+
+int32_t AudioMixerManagerMac::SetSpeakerVolume(uint32_t volume) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetSpeakerVolume(volume=%u)", volume);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  bool success = false;
+
+  // volume range is 0.0 - 1.0, convert from 0 -255
+  const Float32 vol = (Float32)(volume / 255.0);
+
+  assert(vol <= 1.0 && vol >= 0.0);
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, size, &vol));
 
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::OpenMicrophone(id=%d)", deviceID);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    pid_t hogPid = -1;
-
-    _inputDeviceID = deviceID;
-
-    // Check which process, if any, has hogged the device. 
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
-            kAudioDevicePropertyScopeInput, 0 };
-    size = sizeof(hogPid);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &hogPid));
-    if (hogPid == -1)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " No process has hogged the input device");
-    }
-    // getpid() is apparently "always successful"
-    else if (hogPid == getpid())
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Our process has hogged the input device");
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Another process (pid = %d) has hogged the input device",
-                     static_cast<int> (hogPid));
-
-        return -1;
-    }
-
-    // get number of channels from stream format
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-
-    // Get the stream format, to be able to read the number of channels.
-    AudioStreamBasicDescription streamFormat;
-    size = sizeof(AudioStreamBasicDescription);
-    memset(&streamFormat, 0, size);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
-
-    _noInputChannels = streamFormat.mChannelsPerFrame;
-
-    return 0;
-}
-
-bool AudioMixerManagerMac::SpeakerIsInitialized() const
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
-
-    return (_outputDeviceID != kAudioObjectUnknown);
-}
-
-bool AudioMixerManagerMac::MicrophoneIsInitialized() const
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
-                 __FUNCTION__);
-
-    return (_inputDeviceID != kAudioObjectUnknown);
-}
-
-int32_t AudioMixerManagerMac::SetSpeakerVolume(uint32_t volume)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetSpeakerVolume(volume=%u)", volume);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    bool success = false;
-
-    // volume range is 0.0 - 1.0, convert from 0 -255
-    const Float32 vol = (Float32)(volume / 255.0);
-
-    assert(vol <= 1.0 && vol >= 0.0);
-
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
-            0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(vol);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, size, &vol));
-
-        return 0;
+    if (err == noErr && isSettable) {
+      size = sizeof(vol);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, size, &vol));
     }
+    success = true;
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(vol);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, size, &vol));
-        }
-        success = true;
-    }
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set a volume on any output channel");
+    return -1;
+  }
 
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set a volume on any output channel");
-        return -1;
-    }
-
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SpeakerVolume(uint32_t& volume) const
-{
+int32_t AudioMixerManagerMac::SpeakerVolume(uint32_t& volume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  Float32 channelVol = 0;
+  Float32 vol = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &vol));
+
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(vol * 255 + 0.5);
+  } else {
+    // Otherwise get the average volume across channels.
+    vol = 0;
+    for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+      channelVol = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelVol);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _outputDeviceID, &propertyAddress, 0, NULL, &size, &channelVol));
+
+        vol += channelVol;
+        channels++;
+      }
     }
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    Float32 channelVol = 0;
-    Float32 vol = 0;
-
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
-            0 };
-    Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(vol);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, &size, &vol));
-
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> (vol * 255 + 0.5);
-    } else
-    {
-        // Otherwise get the average volume across channels.
-        vol = 0;
-        for (UInt32 i = 1; i <= _noOutputChannels; i++)
-        {
-            channelVol = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelVol);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelVol));
-
-                vol += channelVol;
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get a volume on any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> (255 * vol / channels + 0.5);
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get a volume on any channel");
+      return -1;
     }
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::SpeakerVolume() => vol=%i", vol);
+    assert(channels > 0);
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(255 * vol / channels + 0.5);
+  }
 
-    return 0;
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "     AudioMixerManagerMac::SpeakerVolume() => vol=%i", vol);
+
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
+int32_t AudioMixerManagerMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  maxVolume = 255;
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    maxVolume = 255;
-
-    return 0;
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MinSpeakerVolume(uint32_t& minVolume) const
-{
+int32_t AudioMixerManagerMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  minVolume = 0;
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    minVolume = 0;
-
-    return 0;
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::SpeakerVolumeStepSize(uint16_t& stepSize) const
-{
+int32_t AudioMixerManagerMac::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  stepSize = 1;
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    stepSize = 1;
-
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available)
-{
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+int32_t AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    OSStatus err = noErr;
+  OSStatus err = noErr;
 
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
-            0 };
-    Boolean isSettable = false;
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    available = true;
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Volume cannot be set for output channel %d, err=%d", i,
+                   err);
+      return -1;
     }
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Volume cannot be set for output channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
-
-    available = true;
-    return 0;
+  available = true;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available)
-{
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+int32_t AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    OSStatus err = noErr;
+  OSStatus err = noErr;
 
-    // Does the capture device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeOutput, 0 };
-    Boolean isSettable = false;
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    available = true;
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Mute cannot be set for output channel %d, err=%d", i, err);
+      return -1;
     }
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Mute cannot be set for output channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
-
-    available = true;
-    return 0;
+  available = true;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SetSpeakerMute(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetSpeakerMute(enable=%u)", enable);
+int32_t AudioMixerManagerMac::SetSpeakerMute(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetSpeakerMute(enable=%u)", enable);
 
-    CriticalSectionScoped lock(&_critSect);
+  CriticalSectionScoped lock(&_critSect);
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    UInt32 mute = enable ? 1 : 0;
-    bool success = false;
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  UInt32 mute = enable ? 1 : 0;
+  bool success = false;
 
-    // Does the render device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeOutput, 0 };
-    Boolean isSettable = false;
+  // Does the render device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(mute);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(mute);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, size, &mute));
-
-        return 0;
+    if (err == noErr && isSettable) {
+      size = sizeof(mute);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, size, &mute));
     }
+    success = true;
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noOutputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(mute);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, size, &mute));
-        }
-        success = true;
-    }
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set mute on any input channel");
+    return -1;
+  }
 
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set mute on any input channel");
-        return -1;
-    }
-
-    return 0;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::SpeakerMute(bool& enabled) const
-{
+int32_t AudioMixerManagerMac::SpeakerMute(bool& enabled) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  UInt32 channelMuted = 0;
+  UInt32 muted = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(muted);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &muted));
+
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  } else {
+    // Otherwise check if all channels are muted.
+    for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+      muted = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelMuted);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _outputDeviceID, &propertyAddress, 0, NULL, &size, &channelMuted));
+
+        muted = (muted && channelMuted);
+        channels++;
+      }
     }
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    UInt32 channelMuted = 0;
-    UInt32 muted = 0;
-
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeOutput, 0 };
-    Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(muted);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                &propertyAddress, 0, NULL, &size, &muted));
-
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    } else
-    {
-        // Otherwise check if all channels are muted.
-        for (UInt32 i = 1; i <= _noOutputChannels; i++)
-        {
-            muted = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelMuted);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelMuted));
-
-                muted = (muted && channelMuted);
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get mute for any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get mute for any channel");
+      return -1;
     }
 
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::SpeakerMute() => enabled=%d, enabled");
+    assert(channels > 0);
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  }
 
-    return 0;
+  WEBRTC_TRACE(
+      kTraceInfo, kTraceAudioDevice, _id,
+      "     AudioMixerManagerMac::SpeakerMute() => enabled=%d, enabled");
+
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available)
-{
-    if (_outputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+int32_t AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    available = (_noOutputChannels == 2);
-    return 0;
+  available = (_noOutputChannels == 2);
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+int32_t AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    available = (_noInputChannels == 2);
-    return 0;
+  available = (_noInputChannels == 2);
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+int32_t AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    OSStatus err = noErr;
-
-    // Does the capture device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
-    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                        &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
-    }
-
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Mute cannot be set for output channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
-    }
+  OSStatus err = noErr;
 
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
     available = true;
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::SetMicrophoneMute(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetMicrophoneMute(enable=%u)", enable);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    UInt32 mute = enable ? 1 : 0;
-    bool success = false;
-
-    // Does the capture device have a master mute control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(mute);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, size, &mute));
-
-        return 0;
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Mute cannot be set for output channel %d, err=%d", i, err);
+      return -1;
     }
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(mute);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                    &propertyAddress, 0, NULL, size, &mute));
-        }
-        success = true;
-    }
-
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set mute on any input channel");
-        return -1;
-    }
-
-    return 0;
+  available = true;
+  return 0;
 }
 
-int32_t AudioMixerManagerMac::MicrophoneMute(bool& enabled) const
-{
+int32_t AudioMixerManagerMac::SetMicrophoneMute(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetMicrophoneMute(enable=%u)", enable);
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  CriticalSectionScoped lock(&_critSect);
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    UInt32 channelMuted = 0;
-    UInt32 muted = 0;
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
-            kAudioDevicePropertyScopeInput, 0 };
-    Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(muted);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, &size, &muted));
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  UInt32 mute = enable ? 1 : 0;
+  bool success = false;
 
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    } else
-    {
-        // Otherwise check if all channels are muted.
-        for (UInt32 i = 1; i <= _noInputChannels; i++)
-        {
-            muted = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelMuted);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelMuted));
-
-                muted = (muted && channelMuted);
-                channels++;
-            }
-        }
-
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get mute for any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // 1 means muted
-        enabled = static_cast<bool> (muted);
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::MicrophoneMute() => enabled=%d",
-                 enabled);
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(mute);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, size, &mute));
 
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::MicrophoneBoostIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    available = false; // No AudioObjectPropertySelector value for Mic Boost
-
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::SetMicrophoneBoost(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetMicrophoneBoost(enable=%u)", enable);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    // Ensure that the selected microphone has a valid boost control.
-    bool available(false);
-    MicrophoneBoostIsAvailable(available);
-    if (!available)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  it is not possible to enable microphone boost");
-        return -1;
-    }
-
-    // It is assumed that the call above fails!
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneBoost(bool& enabled) const
-{
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    // Microphone boost cannot be enabled on this platform!
-    enabled = false;
-
-    return 0;
-}
-
-int32_t AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available)
-{
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyVolumeScalar,
-                kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        available = true;
-        return 0;
+    if (err == noErr && isSettable) {
+      size = sizeof(mute);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _inputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set mute on any input channel");
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneMute(bool& enabled) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  UInt32 channelMuted = 0;
+  UInt32 muted = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(muted);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, &size, &muted));
+
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  } else {
+    // Otherwise check if all channels are muted.
+    for (UInt32 i = 1; i <= _noInputChannels; i++) {
+      muted = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelMuted);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _inputDeviceID, &propertyAddress, 0, NULL, &size, &channelMuted));
+
+        muted = (muted && channelMuted);
+        channels++;
+      }
     }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err != noErr || !isSettable)
-        {
-            available = false;
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Volume cannot be set for input channel %d, err=%d",
-                         i, err);
-            return -1;
-        }
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get mute for any channel");
+      return -1;
     }
 
+    assert(channels > 0);
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "     AudioMixerManagerMac::MicrophoneMute() => enabled=%d",
+               enabled);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneBoostIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  available = false;  // No AudioObjectPropertySelector value for Mic Boost
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetMicrophoneBoost(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetMicrophoneBoost(enable=%u)", enable);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // Ensure that the selected microphone has a valid boost control.
+  bool available(false);
+  MicrophoneBoostIsAvailable(available);
+  if (!available) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  it is not possible to enable microphone boost");
+    return -1;
+  }
+
+  // It is assumed that the call above fails!
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneBoost(bool& enabled) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // Microphone boost cannot be enabled on this platform!
+  enabled = false;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
     available = true;
     return 0;
-}
+  }
 
-int32_t AudioMixerManagerMac::SetMicrophoneVolume(uint32_t volume)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioMixerManagerMac::SetMicrophoneVolume(volume=%u)", volume);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    bool success = false;
-
-    // volume range is 0.0 - 1.0, convert from 0 - 255
-    const Float32 vol = (Float32)(volume / 255.0);
-
-    assert(vol <= 1.0 && vol >= 0.0);
-
-    // Does the capture device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyVolumeScalar,
-                kAudioDevicePropertyScopeInput, 0 };
-    Boolean isSettable = false;
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
     err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
                                         &isSettable);
-    if (err == noErr && isSettable)
-    {
-        size = sizeof(vol);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, size, &vol));
-
-        return 0;
+    if (err != noErr || !isSettable) {
+      available = false;
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Volume cannot be set for input channel %d, err=%d", i,
+                   err);
+      return -1;
     }
+  }
 
-    // Otherwise try to set each channel.
-    for (UInt32 i = 1; i <= _noInputChannels; i++)
-    {
-        propertyAddress.mElement = i;
-        isSettable = false;
-        err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
-                                            &isSettable);
-        if (err == noErr && isSettable)
-        {
-            size = sizeof(vol);
-            WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-                    &propertyAddress, 0, NULL, size, &vol));
-        }
-        success = true;
-    }
-
-    if (!success)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Unable to set a level on any input channel");
-        return -1;
-    }
-
-    return 0;
+  available = true;
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MicrophoneVolume(uint32_t& volume) const
-{
+int32_t AudioMixerManagerMac::SetMicrophoneVolume(uint32_t volume) {
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "AudioMixerManagerMac::SetMicrophoneVolume(volume=%u)", volume);
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  CriticalSectionScoped lock(&_critSect);
 
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    unsigned int channels = 0;
-    Float32 channelVol = 0;
-    Float32 volFloat32 = 0;
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    // Does the device have a master volume control?
-    // If so, use it exclusively.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyVolumeScalar,
-                kAudioDevicePropertyScopeInput, 0 };
-    Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-    if (hasProperty)
-    {
-        size = sizeof(volFloat32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                &propertyAddress, 0, NULL, &size, &volFloat32));
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  bool success = false;
 
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> (volFloat32 * 255 + 0.5);
-    } else
-    {
-        // Otherwise get the average volume across channels.
-        volFloat32 = 0;
-        for (UInt32 i = 1; i <= _noInputChannels; i++)
-        {
-            channelVol = 0;
-            propertyAddress.mElement = i;
-            hasProperty = AudioObjectHasProperty(_inputDeviceID,
-                                                 &propertyAddress);
-            if (hasProperty)
-            {
-                size = sizeof(channelVol);
-                WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-                        &propertyAddress, 0, NULL, &size, &channelVol));
+  // volume range is 0.0 - 1.0, convert from 0 - 255
+  const Float32 vol = (Float32)(volume / 255.0);
 
-                volFloat32 += channelVol;
-                channels++;
-            }
-        }
+  assert(vol <= 1.0 && vol >= 0.0);
 
-        if (channels == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Unable to get a level on any channel");
-            return -1;
-        }
-
-        assert(channels > 0);
-        // vol 0.0 to 1.0 -> convert to 0 - 255
-        volume = static_cast<uint32_t> 
-            (255 * volFloat32 / channels + 0.5);
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "     AudioMixerManagerMac::MicrophoneVolume() => vol=%u",
-                 volume);
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, size, &vol));
 
     return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err == noErr && isSettable) {
+      size = sizeof(vol);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _inputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 " Unable to set a level on any input channel");
+    return -1;
+  }
+
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
+int32_t AudioMixerManagerMac::MicrophoneVolume(uint32_t& volume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  Float32 channelVol = 0;
+  Float32 volFloat32 = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(volFloat32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, &size, &volFloat32));
+
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(volFloat32 * 255 + 0.5);
+  } else {
+    // Otherwise get the average volume across channels.
+    volFloat32 = 0;
+    for (UInt32 i = 1; i <= _noInputChannels; i++) {
+      channelVol = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelVol);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _inputDeviceID, &propertyAddress, 0, NULL, &size, &channelVol));
+
+        volFloat32 += channelVol;
+        channels++;
+      }
     }
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 255
-    maxVolume = 255;
+    if (channels == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Unable to get a level on any channel");
+      return -1;
+    }
 
-    return 0;
+    assert(channels > 0);
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(255 * volFloat32 / channels + 0.5);
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "     AudioMixerManagerMac::MicrophoneVolume() => vol=%u",
+               volume);
+
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MinMicrophoneVolume(uint32_t& minVolume) const
-{
+int32_t AudioMixerManagerMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  maxVolume = 255;
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 10
-    minVolume = 0;
-
-    return 0;
+  return 0;
 }
 
-int32_t
-AudioMixerManagerMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
-{
+int32_t AudioMixerManagerMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
 
-    if (_inputDeviceID == kAudioObjectUnknown)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  device ID has not been set");
-        return -1;
-    }
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 10
+  minVolume = 0;
 
-    // volume range is 0.0 to 1.0
-    // we convert that to 0 - 10
-    stepSize = 1;
+  return 0;
+}
 
-    return 0;
+int32_t AudioMixerManagerMac::MicrophoneVolumeStepSize(
+    uint16_t& stepSize) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  device ID has not been set");
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 10
+  stepSize = 1;
+
+  return 0;
 }
 
 // ============================================================================
@@ -1148,18 +1003,18 @@ AudioMixerManagerMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
 // CoreAudio errors are best interpreted as four character strings.
 void AudioMixerManagerMac::logCAMsg(const TraceLevel level,
                                     const TraceModule module,
-                                    const int32_t id, const char *msg,
-                                    const char *err)
-{
-    assert(msg != NULL);
-    assert(err != NULL);
+                                    const int32_t id,
+                                    const char* msg,
+                                    const char* err) {
+  assert(msg != NULL);
+  assert(err != NULL);
 
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+  WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
-    // We need to flip the characters in this case.
-    WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
-        + 2, err + 1, err);
+  // We need to flip the characters in this case.
+  WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err + 2,
+               err + 1, err);
 #endif
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
index 711d36d150..9cbfe2deb4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
@@ -12,69 +12,68 @@
 #define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
 
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 #include <CoreAudio/CoreAudio.h>
 
 namespace webrtc {
-	
-class AudioMixerManagerMac
-{
-public:
-    int32_t OpenSpeaker(AudioDeviceID deviceID);
-    int32_t OpenMicrophone(AudioDeviceID deviceID);
-    int32_t SetSpeakerVolume(uint32_t volume);
-    int32_t SpeakerVolume(uint32_t& volume) const;
-    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
-    int32_t MinSpeakerVolume(uint32_t& minVolume) const;
-    int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
-    int32_t SpeakerVolumeIsAvailable(bool& available);
-    int32_t SpeakerMuteIsAvailable(bool& available);
-    int32_t SetSpeakerMute(bool enable);
-    int32_t SpeakerMute(bool& enabled) const;
-    int32_t StereoPlayoutIsAvailable(bool& available);
-    int32_t StereoRecordingIsAvailable(bool& available);
-    int32_t MicrophoneMuteIsAvailable(bool& available);
-    int32_t SetMicrophoneMute(bool enable);
-    int32_t MicrophoneMute(bool& enabled) const;
-    int32_t MicrophoneBoostIsAvailable(bool& available);
-    int32_t SetMicrophoneBoost(bool enable);
-    int32_t MicrophoneBoost(bool& enabled) const;
-    int32_t MicrophoneVolumeIsAvailable(bool& available);
-    int32_t SetMicrophoneVolume(uint32_t volume);
-    int32_t MicrophoneVolume(uint32_t& volume) const;
-    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
-    int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
-    int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
-    int32_t Close();
-    int32_t CloseSpeaker();
-    int32_t CloseMicrophone();
-    bool SpeakerIsInitialized() const;
-    bool MicrophoneIsInitialized() const;
 
-public:
-    AudioMixerManagerMac(const int32_t id);
-    ~AudioMixerManagerMac();
+class AudioMixerManagerMac {
+ public:
+  int32_t OpenSpeaker(AudioDeviceID deviceID);
+  int32_t OpenMicrophone(AudioDeviceID deviceID);
+  int32_t SetSpeakerVolume(uint32_t volume);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t StereoPlayoutIsAvailable(bool& available);
+  int32_t StereoRecordingIsAvailable(bool& available);
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneBoostIsAvailable(bool& available);
+  int32_t SetMicrophoneBoost(bool enable);
+  int32_t MicrophoneBoost(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
+  int32_t Close();
+  int32_t CloseSpeaker();
+  int32_t CloseMicrophone();
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
 
-private:
-    static void logCAMsg(const TraceLevel level,
-                         const TraceModule module,
-                         const int32_t id, const char *msg,
-                         const char *err);
+ public:
+  AudioMixerManagerMac(const int32_t id);
+  ~AudioMixerManagerMac();
 
-private:
-    CriticalSectionWrapper& _critSect;
-    int32_t _id;
+ private:
+  static void logCAMsg(const TraceLevel level,
+                       const TraceModule module,
+                       const int32_t id,
+                       const char* msg,
+                       const char* err);
 
-    AudioDeviceID _inputDeviceID;
-    AudioDeviceID _outputDeviceID;
+ private:
+  CriticalSectionWrapper& _critSect;
+  int32_t _id;
 
-    uint16_t _noInputChannels;
-    uint16_t _noOutputChannels;
+  AudioDeviceID _inputDeviceID;
+  AudioDeviceID _outputDeviceID;
 
+  uint16_t _noInputChannels;
+  uint16_t _noOutputChannels;
 };
-	
+
 }  // namespace webrtc
 
 #endif  // AUDIO_MIXER_MAC_H
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/main/interface/audio_device.h b/media/webrtc/trunk/webrtc/modules/audio_device/main/interface/audio_device.h
deleted file mode 100644
index 71f16b685a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/main/interface/audio_device.h
+++ /dev/null
@@ -1,16 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
-#define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
-
-#include "webrtc/modules/audio_device/include/audio_device.h"
-
-#endif  // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/main/source/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_device/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/main/source/audio_device.gypi b/media/webrtc/trunk/webrtc/modules/audio_device/main/source/audio_device.gypi
deleted file mode 100644
index 126eb98ac1..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/main/source/audio_device.gypi
+++ /dev/null
@@ -1,14 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../audio_device.gypi',
-  ],
-}
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/mock_audio_device_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_device/mock_audio_device_buffer.h
index b9e66f7d1c..07c9e2912e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/mock_audio_device_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/mock_audio_device_buffer.h
@@ -20,9 +20,13 @@ class MockAudioDeviceBuffer : public AudioDeviceBuffer {
  public:
   MockAudioDeviceBuffer() {}
   virtual ~MockAudioDeviceBuffer() {}
-
-  MOCK_METHOD1(RequestPlayoutData, int32_t(uint32_t nSamples));
+  MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples));
   MOCK_METHOD1(GetPlayoutData, int32_t(void* audioBuffer));
+  MOCK_METHOD2(SetRecordedBuffer,
+               int32_t(const void* audioBuffer, size_t nSamples));
+  MOCK_METHOD3(SetVQEData,
+               void(int playDelayMS, int recDelayMS, int clockDrift));
+  MOCK_METHOD0(DeliverRecordedData, int32_t());
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.cc
deleted file mode 100644
index f7ce95b17d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/audio_device_opensles_android.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.h
deleted file mode 100644
index 480149e86d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/audio_device_opensles_android.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_manager_jni.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_manager_jni.h
deleted file mode 100644
index 961fa23cb0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_manager_jni.h
+++ /dev/null
@@ -1,6 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/audio_manager_jni.h"
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.cc
deleted file mode 100644
index 31351f4e4c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/fine_audio_buffer.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.h
deleted file mode 100644
index 2512b71ee3..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/fine_audio_buffer.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.cc
deleted file mode 100644
index b529566521..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/low_latency_event_posix.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.h
deleted file mode 100644
index 21e4e87ed0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/low_latency_event_posix.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.cc
deleted file mode 100644
index ea7a4f92be..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/opensles_common.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.h
deleted file mode 100644
index 33b8e5c43f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/opensles_common.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.cc
deleted file mode 100644
index 48ae8813aa..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/opensles_input.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.h
deleted file mode 100644
index f42cdea837..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/opensles_input.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.cc
deleted file mode 100644
index 270feb7b53..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/opensles_output.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.h
deleted file mode 100644
index d8c426eed2..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/opensles_output.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc
index c3927ff0c9..3621bcf33f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc
@@ -1,5 +1,122 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
 
-#include "../android/single_rw_fifo.cc"
+#if defined(_MSC_VER)
+#include <windows.h>
+#endif
+
+#include "single_rw_fifo.h"
+
+#include <assert.h>
+
+static int UpdatePos(int pos, int capacity) {
+  return (pos + 1) % capacity;
+}
+
+namespace webrtc {
+
+namespace subtle {
+
+// Start with compiler support, then processor-specific hacks
+#if defined(__GNUC__) || defined(__clang__)
+// Available on GCC and clang - others?
+inline void MemoryBarrier() {
+  __sync_synchronize();
+}
+
+#elif defined(_MSC_VER)
+inline void MemoryBarrier() {
+  ::MemoryBarrier();
+}
+
+#elif defined(__aarch64__)
+// From http://http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_arm64_gcc.h
+inline void MemoryBarrier() {
+  __asm__ __volatile__ ("dmb ish" ::: "memory");
+}
+
+#elif defined(__ARMEL__)
+// From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_arm_gcc.h
+inline void MemoryBarrier() {
+  // Note: This is a function call, which is also an implicit compiler barrier.
+  typedef void (*KernelMemoryBarrierFunc)();
+  ((KernelMemoryBarrierFunc)0xffff0fa0)();
+}
+
+#elif defined(__x86_64__) || defined (__i386__)
+// From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_x86_gcc.h
+// mfence exists on x64 and x86 platforms containing SSE2.
+// x86 platforms that don't have SSE2 will crash with SIGILL.
+// If this code needs to run on such platforms in the future,
+// add runtime CPU detection here.
+inline void MemoryBarrier() {
+  __asm__ __volatile__("mfence" : : : "memory");
+}
+
+#elif defined(__MIPSEL__)
+// From http://src.chromium.org/viewvc/chrome/trunk/src/base/atomicops_internals_mips_gcc.h
+inline void MemoryBarrier() {
+  __asm__ __volatile__("sync" : : : "memory");
+}
+
+#else
+#error Add an implementation of MemoryBarrier() for this platform!
+#endif
+
+}  // namespace subtle
+
+SingleRwFifo::SingleRwFifo(int capacity)
+    : capacity_(capacity),
+      size_(0),
+      read_pos_(0),
+      write_pos_(0) {
+  queue_.reset(new int8_t*[capacity_]);
+}
+
+SingleRwFifo::~SingleRwFifo() {
+}
+
+void SingleRwFifo::Push(int8_t* mem) {
+  assert(mem);
+
+  // Ensure that there is space for the new data in the FIFO.
+  // Note there is only one writer meaning that the other thread is guaranteed
+  // only to decrease the size.
+  const int free_slots = capacity() - size();
+  if (free_slots <= 0) {
+    // Size can be queried outside of the Push function. The caller is assumed
+    // to ensure that Push will be successful before calling it.
+    assert(false);
+    return;
+  }
+  queue_[write_pos_] = mem;
+  // Memory barrier ensures that |size_| is updated after the size has changed.
+  subtle::MemoryBarrier();
+  ++size_;
+  write_pos_ = UpdatePos(write_pos_, capacity());
+}
+
+int8_t* SingleRwFifo::Pop() {
+  int8_t* ret_val = NULL;
+  if (size() <= 0) {
+    // Size can be queried outside of the Pop function. The caller is assumed
+    // to ensure that Pop will be successfull before calling it.
+    assert(false);
+    return ret_val;
+  }
+  ret_val = queue_[read_pos_];
+  // Memory barrier ensures that |size_| is updated after the size has changed.
+  subtle::MemoryBarrier();
+  --size_;
+  read_pos_ = UpdatePos(read_pos_, capacity());
+  return ret_val;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.h b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.h
index 378be4dad0..ecd5a79110 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.h
@@ -1,5 +1,49 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
 
-#include "../android/single_rw_fifo.h"
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_SINGLE_RW_FIFO_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_SINGLE_RW_FIFO_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Implements a lock-free FIFO losely based on
+// http://src.chromium.org/viewvc/chrome/trunk/src/media/base/audio_fifo.cc
+// Note that this class assumes there is one producer (writer) and one
+// consumer (reader) thread.
+class SingleRwFifo {
+ public:
+  explicit SingleRwFifo(int capacity);
+  ~SingleRwFifo();
+
+  void Push(int8_t* mem);
+  int8_t* Pop();
+
+  void Clear();
+
+  int size() { return size_.Value(); }
+  int capacity() const { return capacity_; }
+
+ private:
+  rtc::scoped_ptr<int8_t* []> queue_;
+  int capacity_;
+
+  Atomic32 size_;
+
+  int read_pos_;
+  int write_pos_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_SINGLE_RW_FIFO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.cc b/media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.cc
deleted file mode 100644
index ea40bbfe6c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.cc
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/audio_device_utility_android.cc"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.h b/media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.h
deleted file mode 100644
index 38eeebee7e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.h
+++ /dev/null
@@ -1,5 +0,0 @@
-/* This Source Code Form is subject to the terms of the Mozilla Public
- * License, v. 2.0. If a copy of the MPL was not distributed with this file,
- * You can obtain one at http://mozilla.org/MPL/2.0/. */
-
-#include "../android/audio_device_utility_android.h"
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
index 227b97f741..a8ff5c69e9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.cc
@@ -15,9 +15,9 @@
 #include "webrtc/modules/audio_device/audio_device_utility.h"
 #include "webrtc/modules/audio_device/sndio/audio_device_sndio.h"
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include "Latency.h"
 
@@ -699,9 +699,9 @@ int32_t AudioDeviceSndio::StartRecording()
         return 0;
     }
 
-    _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc,
-                                                this,
-                                                threadName);
+    _ptrThreadRec = new rtc::PlatformThread(RecThreadFunc,
+                                            this,
+                                            threadName);
     if (_ptrThreadRec == NULL)
     {
         WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
@@ -796,9 +796,9 @@ int32_t AudioDeviceSndio::StartPlayout()
         return 0;
     }
 
-    _ptrThreadPlay =  ThreadWrapper::CreateThread(PlayThreadFunc,
-                                                  this,
-                                                  threadName);
+    _ptrThreadPlay = new rtc::PlatformThread(PlayThreadFunc,
+                                             this,
+                                             threadName);
     if (_ptrThreadPlay == NULL)
     {
         WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.h b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.h
index c95aef0c69..22961ad804 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_sndio.h
@@ -14,8 +14,8 @@
 #include <sndio.h>
 
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/base/platform_thread.h"
 
 namespace webrtc
 {
@@ -169,8 +169,8 @@ private:
 
     CriticalSectionWrapper& _critSect;
 
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
-    rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
+    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
     int32_t _id;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_utility_sndio.cc b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_utility_sndio.cc
index f0e9f3b65c..d2fbfab6f3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_utility_sndio.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/sndio/audio_device_utility_sndio.cc
@@ -9,8 +9,8 @@
  */
 
 #include "webrtc/modules/audio_device/sndio/audio_device_utility_sndio.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc
 {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc b/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc
index 77405f4fb3..0098525fc6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc
@@ -19,8 +19,7 @@
 
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 
 // Helper functions
 #if defined(ANDROID)
@@ -83,17 +82,16 @@ class AudioTransportAPI: public AudioTransport {
 
   ~AudioTransportAPI() {}
 
-  virtual int32_t RecordedDataIsAvailable(
-      const void* audioSamples,
-      const uint32_t nSamples,
-      const uint8_t nBytesPerSample,
-      const uint8_t nChannels,
-      const uint32_t sampleRate,
-      const uint32_t totalDelay,
-      const int32_t clockSkew,
-      const uint32_t currentMicLevel,
-      const bool keyPressed,
-      uint32_t& newMicLevel) {
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  const size_t nSamples,
+                                  const size_t nBytesPerSample,
+                                  const size_t nChannels,
+                                  const uint32_t sampleRate,
+                                  const uint32_t totalDelay,
+                                  const int32_t clockSkew,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override {
     rec_count_++;
     if (rec_count_ % 100 == 0) {
       if (nChannels == 1) {
@@ -110,15 +108,14 @@ class AudioTransportAPI: public AudioTransport {
     return 0;
   }
 
-  virtual int32_t NeedMorePlayData(
-      const uint32_t nSamples,
-      const uint8_t nBytesPerSample,
-      const uint8_t nChannels,
-      const uint32_t sampleRate,
-      void* audioSamples,
-      uint32_t& nSamplesOut,
-      int64_t* elapsed_time_ms,
-      int64_t* ntp_time_ms) {
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t sampleRate,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
     play_count_++;
     if (play_count_ % 100 == 0) {
       if (nChannels == 1) {
@@ -131,29 +128,6 @@ class AudioTransportAPI: public AudioTransport {
     return 0;
   }
 
-  virtual int OnDataAvailable(const int voe_channels[],
-                              int number_of_voe_channels,
-                              const int16_t* audio_data,
-                              int sample_rate,
-                              int number_of_channels,
-                              int number_of_frames,
-                              int audio_delay_milliseconds,
-                              int current_volume,
-                              bool key_pressed,
-                              bool need_audio_processing) {
-    return 0;
-  }
-
-  virtual void PushCaptureData(int voe_channel, const void* audio_data,
-                               int bits_per_sample, int sample_rate,
-                               int number_of_channels,
-                               int number_of_frames) {}
-
-  virtual void PullRenderData(int bits_per_sample, int sample_rate,
-                              int number_of_channels, int number_of_frames,
-                              void* audio_data,
-                              int64_t* elapsed_time_ms,
-                              int64_t* ntp_time_ms) {}
  private:
   uint32_t rec_count_;
   uint32_t play_count_;
@@ -166,7 +140,7 @@ class AudioDeviceAPITest: public testing::Test {
   virtual ~AudioDeviceAPITest() {}
 
   static void SetUpTestCase() {
-    process_thread_ = ProcessThread::Create();
+    process_thread_ = ProcessThread::Create("ProcessThread");
     process_thread_->Start();
 
     // Windows:
@@ -1049,9 +1023,15 @@ TEST_F(AudioDeviceAPITest, MicrophoneVolumeIsAvailable) {
 // MicrophoneVolume
 // MaxMicrophoneVolume
 // MinMicrophoneVolume
-// NOTE: Disabled on mac due to issue 257.
-#ifndef WEBRTC_MAC
-TEST_F(AudioDeviceAPITest, MicrophoneVolumeTests) {
+
+// Disabled on Mac and Linux,
+// see https://bugs.chromium.org/p/webrtc/issues/detail?id=5414
+#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+#define MAYBE_MicrophoneVolumeTests DISABLED_MicrophoneVolumeTests
+#else
+#define MAYBE_MicrophoneVolumeTests MicrophoneVolumeTests
+#endif
+TEST_F(AudioDeviceAPITest, MAYBE_MicrophoneVolumeTests) {
   uint32_t vol(0);
   uint32_t volume(0);
   uint32_t maxVolume(0);
@@ -1144,7 +1124,6 @@ TEST_F(AudioDeviceAPITest, MicrophoneVolumeTests) {
     EXPECT_EQ(0, audio_device_->SetMicrophoneVolume(maxVolume/10));
   }
 }
-#endif  // !WEBRTC_MAC
 
 TEST_F(AudioDeviceAPITest, SpeakerMuteIsAvailable) {
   bool available;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_defines.h b/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_defines.h
index 479861b125..cc8e3e3aef 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_defines.h
@@ -13,8 +13,8 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #ifdef _WIN32
 #define MACRO_DEFAULT_DEVICE AudioDeviceModule::kDefaultDevice
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.cc
index 3c5f88505c..0e5a9179ff 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.cc
@@ -13,9 +13,15 @@
 #include <stdio.h>
 #include <string.h>
 
+#if defined(_WIN32)
+#include <conio.h>
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#include <termios.h>    // tcgetattr
+#endif
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_device/test/func_test_manager.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 #include "webrtc/modules/audio_device/audio_device_config.h"
@@ -37,6 +43,21 @@ const char* RecordedMicrophoneBoostFile =
 const char* RecordedMicrophoneAGCFile = "recorded_microphone_AGC_mono_48.pcm";
 const char* RecordedSpeakerFile = "recorded_speaker_48.pcm";
 
+#if defined(WEBRTC_IOS) || defined(ANDROID)
+#define USE_SLEEP_AS_PAUSE
+#else
+//#define USE_SLEEP_AS_PAUSE
+#endif
+
+// Sets the default pause time if using sleep as pause
+#define DEFAULT_PAUSE_TIME 5000
+
+#if defined(USE_SLEEP_AS_PAUSE)
+#define PAUSE(a) SleepMs(a);
+#else
+#define PAUSE(a) WaitForKey();
+#endif
+
 // Helper functions
 #if !defined(WEBRTC_IOS)
 char* GetFilename(char* filename)
@@ -57,6 +78,35 @@ const char* GetResource(const char* resource)
 }
 #endif
 
+#if !defined(USE_SLEEP_AS_PAUSE)
+static void WaitForKey() {
+#if defined(_WIN32)
+    _getch();
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+    struct termios oldt, newt;
+
+    tcgetattr( STDIN_FILENO, &oldt );
+
+    // we don't want getchar to echo!
+
+    newt = oldt;
+    newt.c_lflag &= ~( ICANON | ECHO );
+    tcsetattr( STDIN_FILENO, TCSANOW, &newt );
+
+    // catch any newline that's hanging around...
+    // you'll have to hit enter twice if you
+    // choose enter out of all available keys
+
+    if (getc(stdin) == '\n')
+    {
+        getc(stdin);
+    }
+
+    tcsetattr( STDIN_FILENO, TCSANOW, &oldt );
+#endif  // defined(_WIN32)
+}
+#endif  // !defined(USE_SLEEP_AS_PAUSE)
+
 namespace webrtc
 {
 
@@ -142,9 +192,9 @@ void AudioTransportImpl::SetFullDuplex(bool enable)
 
 int32_t AudioTransportImpl::RecordedDataIsAvailable(
     const void* audioSamples,
-    const uint32_t nSamples,
-    const uint8_t nBytesPerSample,
-    const uint8_t nChannels,
+    const size_t nSamples,
+    const size_t nBytesPerSample,
+    const size_t nChannels,
     const uint32_t samplesPerSec,
     const uint32_t totalDelayMS,
     const int32_t clockDrift,
@@ -156,7 +206,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
     {
         AudioPacket* packet = new AudioPacket();
         memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
-        packet->nSamples = (uint16_t) nSamples;
+        packet->nSamples = nSamples;
         packet->nBytesPerSample = nBytesPerSample;
         packet->nChannels = nChannels;
         packet->samplesPerSec = samplesPerSec;
@@ -287,12 +337,12 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
 
 
 int32_t AudioTransportImpl::NeedMorePlayData(
-    const uint32_t nSamples,
-    const uint8_t nBytesPerSample,
-    const uint8_t nChannels,
+    const size_t nSamples,
+    const size_t nBytesPerSample,
+    const size_t nChannels,
     const uint32_t samplesPerSec,
     void* audioSamples,
-    uint32_t& nSamplesOut,
+    size_t& nSamplesOut,
     int64_t* elapsed_time_ms,
     int64_t* ntp_time_ms)
 {
@@ -309,16 +359,15 @@ int32_t AudioTransportImpl::NeedMorePlayData(
             if (packet)
             {
                 int ret(0);
-                int lenOut(0);
+                size_t lenOut(0);
                 int16_t tmpBuf_96kHz[80 * 12];
                 int16_t* ptr16In = NULL;
                 int16_t* ptr16Out = NULL;
 
-                const uint16_t nSamplesIn = packet->nSamples;
-                const uint8_t nChannelsIn = packet->nChannels;
+                const size_t nSamplesIn = packet->nSamples;
+                const size_t nChannelsIn = packet->nChannels;
                 const uint32_t samplesPerSecIn = packet->samplesPerSec;
-                const uint16_t nBytesPerSampleIn =
-                    packet->nBytesPerSample;
+                const size_t nBytesPerSampleIn = packet->nBytesPerSample;
 
                 int32_t fsInHz(samplesPerSecIn);
                 int32_t fsOutHz(samplesPerSec);
@@ -333,21 +382,20 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                         {
                             _resampler.Push(
                                 (const int16_t*) packet->dataBuffer,
-                                2 * nSamplesIn,
-                                (int16_t*) audioSamples, 2
-                                * nSamples, lenOut);
+                                2 * nSamplesIn, (int16_t*) audioSamples,
+                                2 * nSamples, lenOut);
                         } else
                         {
                             _resampler.Push(
                                 (const int16_t*) packet->dataBuffer,
-                                2 * nSamplesIn, tmpBuf_96kHz, 2
-                                * nSamples, lenOut);
+                                2 * nSamplesIn, tmpBuf_96kHz, 2 * nSamples,
+                                lenOut);
 
                             ptr16In = &tmpBuf_96kHz[0];
                             ptr16Out = (int16_t*) audioSamples;
 
                             // do stereo -> mono
-                            for (unsigned int i = 0; i < nSamples; i++)
+                            for (size_t i = 0; i < nSamples; i++)
                             {
                                 *ptr16Out = *ptr16In; // use left channel
                                 ptr16Out++;
@@ -355,7 +403,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                                 ptr16In++;
                             }
                         }
-                        assert(2*nSamples == (uint32_t)lenOut);
+                        assert(2*nSamples == lenOut);
                     } else
                     {
                         if (_playCount % 100 == 0)
@@ -373,22 +421,19 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                         if (nChannels == 1)
                         {
                             _resampler.Push(
-                                (const int16_t*) packet->dataBuffer,
-                                nSamplesIn,
-                                (int16_t*) audioSamples,
-                                nSamples, lenOut);
+                                (const int16_t*) packet->dataBuffer, nSamplesIn,
+                                (int16_t*) audioSamples, nSamples, lenOut);
                         } else
                         {
                             _resampler.Push(
-                                (const int16_t*) packet->dataBuffer,
-                                nSamplesIn, tmpBuf_96kHz, nSamples,
-                                lenOut);
+                                (const int16_t*) packet->dataBuffer, nSamplesIn,
+                                tmpBuf_96kHz, nSamples, lenOut);
 
                             ptr16In = &tmpBuf_96kHz[0];
                             ptr16Out = (int16_t*) audioSamples;
 
                             // do mono -> stereo
-                            for (unsigned int i = 0; i < nSamples; i++)
+                            for (size_t i = 0; i < nSamples; i++)
                             {
                                 *ptr16Out = *ptr16In; // left
                                 ptr16Out++;
@@ -397,7 +442,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                                 ptr16In++;
                             }
                         }
-                        assert(nSamples == (uint32_t)lenOut);
+                        assert(nSamples == lenOut);
                     } else
                     {
                         if (_playCount % 100 == 0)
@@ -416,8 +461,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
         int16_t fileBuf[480];
 
         // read mono-file
-        int32_t len = _playFile.Read((int8_t*) fileBuf, 2
-            * nSamples);
+        int32_t len = _playFile.Read((int8_t*) fileBuf, 2 * nSamples);
         if (len != 2 * (int32_t) nSamples)
         {
             _playFile.Rewind();
@@ -433,7 +477,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(
             // mono sample from file is duplicated and sent to left and right
             // channels
             int16_t* audio16 = (int16_t*) audioSamples;
-            for (unsigned int i = 0; i < nSamples; i++)
+            for (size_t i = 0; i < nSamples; i++)
             {
                 (*audio16) = fileBuf[i]; // left
                 audio16++;
@@ -523,32 +567,6 @@ int32_t AudioTransportImpl::NeedMorePlayData(
     return 0;
 }
 
-int AudioTransportImpl::OnDataAvailable(const int voe_channels[],
-                                        int number_of_voe_channels,
-                                        const int16_t* audio_data,
-                                        int sample_rate,
-                                        int number_of_channels,
-                                        int number_of_frames,
-                                        int audio_delay_milliseconds,
-                                        int current_volume,
-                                        bool key_pressed,
-                                        bool need_audio_processing) {
-  return 0;
-}
-
-void AudioTransportImpl::PushCaptureData(int voe_channel,
-                                         const void* audio_data,
-                                         int bits_per_sample, int sample_rate,
-                                         int number_of_channels,
-                                         int number_of_frames) {}
-
-void AudioTransportImpl::PullRenderData(int bits_per_sample, int sample_rate,
-                                        int number_of_channels,
-                                        int number_of_frames,
-                                        void* audio_data,
-                                        int64_t* elapsed_time_ms,
-                                        int64_t* ntp_time_ms) {}
-
 FuncTestManager::FuncTestManager() :
     _audioDevice(NULL),
     _audioEventObserver(NULL),
@@ -570,7 +588,8 @@ FuncTestManager::~FuncTestManager()
 
 int32_t FuncTestManager::Init()
 {
-    EXPECT_TRUE((_processThread = ProcessThread::Create()) != NULL);
+  EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) !=
+              NULL);
     if (_processThread == NULL)
     {
         return -1;
@@ -635,7 +654,7 @@ int32_t FuncTestManager::Close()
         _audioDevice = NULL;
     }
 
-    // return the ThreadWrapper (singleton)
+    // return the PlatformThread (singleton)
     Trace::ReturnTrace();
 
     // PRINT_TEST_RESULTS;
@@ -810,7 +829,8 @@ int32_t FuncTestManager::TestAudioLayerSelection()
         // ==================================================
         // Next, try to make fresh start with new audio layer
 
-        EXPECT_TRUE((_processThread = ProcessThread::Create()) != NULL);
+        EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) !=
+                    NULL);
         if (_processThread == NULL)
         {
             return -1;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.h b/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.h
index 3c80cdc4a8..b7cc81cc1a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.h
@@ -11,31 +11,15 @@
 #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
 #define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
 
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-
 #include <list>
 #include <string>
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/test/audio_device_test_defines.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
-#if defined(WEBRTC_IOS) || defined(ANDROID)
-#define USE_SLEEP_AS_PAUSE
-#else
-//#define USE_SLEEP_AS_PAUSE
-#endif
-
-// Sets the default pause time if using sleep as pause
-#define DEFAULT_PAUSE_TIME 5000
-
-#if defined(USE_SLEEP_AS_PAUSE)
-#define PAUSE(a) SleepMs(a);
-#else
-#define PAUSE(a) AudioDeviceUtility::WaitForKey();
-#endif
 
 #define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio
 //#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio
@@ -63,9 +47,9 @@ enum TestType
 struct AudioPacket
 {
     uint8_t dataBuffer[4 * 960];
-    uint16_t nSamples;
-    uint16_t nBytesPerSample;
-    uint8_t nChannels;
+    size_t nSamples;
+    size_t nBytesPerSample;
+    size_t nChannels;
     uint32_t samplesPerSec;
 };
 
@@ -101,48 +85,25 @@ public:
 class AudioTransportImpl: public AudioTransport
 {
 public:
-    virtual int32_t
-        RecordedDataIsAvailable(const void* audioSamples,
-                                const uint32_t nSamples,
-                                const uint8_t nBytesPerSample,
-                                const uint8_t nChannels,
-                                const uint32_t samplesPerSec,
-                                const uint32_t totalDelayMS,
-                                const int32_t clockDrift,
-                                const uint32_t currentMicLevel,
-                                const bool keyPressed,
-                                uint32_t& newMicLevel);
+    int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                    const size_t nSamples,
+                                    const size_t nBytesPerSample,
+                                    const size_t nChannels,
+                                    const uint32_t samplesPerSec,
+                                    const uint32_t totalDelayMS,
+                                    const int32_t clockDrift,
+                                    const uint32_t currentMicLevel,
+                                    const bool keyPressed,
+                                    uint32_t& newMicLevel) override;
 
-    virtual int32_t NeedMorePlayData(const uint32_t nSamples,
-                                     const uint8_t nBytesPerSample,
-                                     const uint8_t nChannels,
-                                     const uint32_t samplesPerSec,
-                                     void* audioSamples,
-                                     uint32_t& nSamplesOut,
-                                     int64_t* elapsed_time_ms,
-                                     int64_t* ntp_time_ms);
-
-    virtual int OnDataAvailable(const int voe_channels[],
-                                int number_of_voe_channels,
-                                const int16_t* audio_data,
-                                int sample_rate,
-                                int number_of_channels,
-                                int number_of_frames,
-                                int audio_delay_milliseconds,
-                                int current_volume,
-                                bool key_pressed,
-                                bool need_audio_processing);
-
-    virtual void PushCaptureData(int voe_channel, const void* audio_data,
-                                 int bits_per_sample, int sample_rate,
-                                 int number_of_channels,
-                                 int number_of_frames);
-
-    virtual void PullRenderData(int bits_per_sample, int sample_rate,
-                                int number_of_channels, int number_of_frames,
-                                void* audio_data,
-                                int64_t* elapsed_time_ms,
-                                int64_t* ntp_time_ms);
+    int32_t NeedMorePlayData(const size_t nSamples,
+                             const size_t nBytesPerSample,
+                             const size_t nChannels,
+                             const uint32_t samplesPerSec,
+                             void* audioSamples,
+                             size_t& nSamplesOut,
+                             int64_t* elapsed_time_ms,
+                             int64_t* ntp_time_ms) override;
 
     AudioTransportImpl(AudioDeviceModule* audioDevice);
     ~AudioTransportImpl();
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc
index 6a851dac69..61f9516880 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc
@@ -35,9 +35,9 @@
 #include <strsafe.h>
 #include <uuids.h>
 
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 // Macro that calls a COM method returning HRESULT value.
 #define EXIT_ON_ERROR(hres)    do { if (FAILED(hres)) goto Exit; } while(0)
@@ -3392,7 +3392,7 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
       return 1;
     }
 
-    _SetThreadName(0, "webrtc_core_audio_render_thread");
+    rtc::SetCurrentThreadName("webrtc_core_audio_render_thread");
 
     // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority.
     //
@@ -3669,7 +3669,7 @@ DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority()
 {
     _hMmTask = NULL;
 
-    _SetThreadName(0, "webrtc_core_audio_capture_thread");
+    rtc::SetCurrentThreadName("webrtc_core_audio_capture_thread");
 
     // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
     // priority.
@@ -4208,7 +4208,7 @@ int AudioDeviceWindowsCore::SetDMOProperties()
     HRESULT hr = S_OK;
     assert(_dmo != NULL);
 
-    scoped_refptr<IPropertyStore> ps;
+    rtc::scoped_refptr<IPropertyStore> ps;
     {
         IPropertyStore* ptrPS = NULL;
         hr = _dmo->QueryInterface(IID_IPropertyStore,
@@ -4641,7 +4641,7 @@ int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
     for (UINT i = 0; i < count; i++)
     {
         memset(szDeviceID, 0, sizeof(szDeviceID));
-        scoped_refptr<IMMDevice> device;
+        rtc::scoped_refptr<IMMDevice> device;
         {
             IMMDevice* ptrDevice = NULL;
             hr = collection->Item(i, &ptrDevice);
@@ -5072,30 +5072,6 @@ void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const
     WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
 }
 
-// ----------------------------------------------------------------------------
-//  _SetThreadName
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::_SetThreadName(DWORD dwThreadID, LPCSTR szThreadName)
-{
-    // See http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx for details on the code
-    // in this function. Name of article is "Setting a Thread Name (Unmanaged)".
-
-    THREADNAME_INFO info;
-    info.dwType = 0x1000;
-    info.szName = szThreadName;
-    info.dwThreadID = dwThreadID;
-    info.dwFlags = 0;
-
-    __try
-    {
-        RaiseException( 0x406D1388, 0, sizeof(info)/sizeof(DWORD), (ULONG_PTR *)&info );
-    }
-    __except (EXCEPTION_CONTINUE_EXECUTION)
-    {
-    }
-}
-
 // ----------------------------------------------------------------------------
 //  WideToUTF8
 // ----------------------------------------------------------------------------
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.h b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.h
index 4d30928c5e..5c94cfdd01 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_core_win.h
@@ -24,8 +24,8 @@
 #include <endpointvolume.h>
 #include <mediaobj.h>        // IMediaObject
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
 #pragma comment( lib, "avrt.lib" )
@@ -235,7 +235,6 @@ private:    // thread functions
     static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
     DWORD DoSetCaptureVolumeThread();
 
-    void _SetThreadName(DWORD dwThreadID, LPCSTR szThreadName);
     void _Lock() { _critSect.Enter(); };
     void _UnLock() { _critSect.Leave(); };
 
@@ -297,8 +296,8 @@ private:  // WASAPI
     ISimpleAudioVolume*                     _ptrRenderSimpleVolume;
 
     // DirectX Media Object (DMO) for the built-in AEC.
-    scoped_refptr<IMediaObject>             _dmo;
-    scoped_refptr<IMediaBuffer>             _mediaBuffer;
+    rtc::scoped_refptr<IMediaObject> _dmo;
+    rtc::scoped_refptr<IMediaBuffer> _mediaBuffer;
     bool                                    _builtInAecEnabled;
 
     HANDLE                                  _hRenderSamplesReadyEvent;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_utility_win.cc b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_utility_win.cc
deleted file mode 100644
index 9cfd6bea6c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_utility_win.cc
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_device/win/audio_device_utility_win.h"
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-#include <windows.h>
-#include <tchar.h>
-#include <strsafe.h>
-
-#define STRING_MAX_SIZE 256
-
-typedef void (WINAPI *PGNSI)(LPSYSTEM_INFO);
-typedef BOOL (WINAPI *PGPI)(DWORD, DWORD, DWORD, DWORD, PDWORD);
-
-namespace webrtc
-{
-
-// ============================================================================
-//                            Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-//  AudioDeviceUtilityWindows() - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceUtilityWindows::AudioDeviceUtilityWindows(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
-    _lastError(AudioDeviceModule::kAdmErrNone)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
-}
-
-// ----------------------------------------------------------------------------
-//  AudioDeviceUtilityWindows() - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceUtilityWindows::~AudioDeviceUtilityWindows()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
-    {
-        CriticalSectionScoped lock(&_critSect);
-
-        // free stuff here...
-    }
-
-    delete &_critSect;
-}
-
-// ============================================================================
-//                                     API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-//  Init()
-// ----------------------------------------------------------------------------
-
-int32_t AudioDeviceUtilityWindows::Init()
-{
-
-    TCHAR szOS[STRING_MAX_SIZE];
-
-    if (GetOSDisplayString(szOS))
-    {
-#ifdef _UNICODE
-        char os[STRING_MAX_SIZE];
-        if (WideCharToMultiByte(CP_UTF8, 0, szOS, -1, os, STRING_MAX_SIZE, NULL, NULL) == 0)
-        {
-            strncpy(os, "Could not get OS info", STRING_MAX_SIZE);
-        }
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "  OS info: %s", os);
-#else
-        WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "  OS info: %s", szOS);
-#endif
-    }
-
-    return 0;
-}
-
-// ============================================================================
-//                                 Private Methods
-// ============================================================================
-
-BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
-{
-    OSVERSIONINFOEX osvi;
-    SYSTEM_INFO si;
-    PGNSI pGNSI;
-    BOOL bOsVersionInfoEx;
-
-    ZeroMemory(&si, sizeof(SYSTEM_INFO));
-    ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
-
-    osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
-
-    // Retrieve information about the current operating system
-    //
-    bOsVersionInfoEx = GetVersionEx((OSVERSIONINFO *) &osvi);
-    if (!bOsVersionInfoEx)
-        return FALSE;
-
-    // Parse our OS version string
-    //
-    if (VER_PLATFORM_WIN32_NT == osvi.dwPlatformId && osvi.dwMajorVersion > 4)
-    {
-        StringCchCopy(pszOS, STRING_MAX_SIZE, TEXT("Microsoft "));
-
-        // Test for the specific product
-        //
-        //  Operating system	    Version number
-        //  --------------------------------------
-        //  Windows 7	            6.1
-        //  Windows Server 2008 R2	6.1
-        //  Windows Server 2008	    6.0
-        //  Windows Vista	        6.0
-        //  - - - - - - - - - - - - - - - - - - -
-        //  Windows Server 2003 R2	5.2
-        //  Windows Server 2003	    5.2
-        //  Windows XP	            5.1
-        //  Windows 2000	        5.0
-        //
-        //  see http://msdn.microsoft.com/en-us/library/ms724832(VS.85).aspx for details
-        //
-        if (osvi.dwMajorVersion == 6)
-        {
-            if (osvi.dwMinorVersion == 0)
-            {
-                // Windows Vista or Server 2008
-                if (osvi.wProductType == VER_NT_WORKSTATION)
-                    StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Vista "));
-                else
-                    StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 " ));
-            }
-
-            if (osvi.dwMinorVersion == 1)
-            {
-                // Windows 7 or Server 2008 R2
-                if (osvi.wProductType == VER_NT_WORKSTATION)
-                    StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 7 "));
-                else
-                    StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 R2 " ));
-            }
-        }
-
-        if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 2)
-        {
-            StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2003"));
-        }
-
-        if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 1)
-        {
-            StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows XP "));
-            if (osvi.wSuiteMask & VER_SUITE_PERSONAL)
-                StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Home Edition" ));
-            else
-                StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
-        }
-
-        if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 0)
-        {
-            StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 2000 "));
-
-            if (osvi.wProductType == VER_NT_WORKSTATION )
-            {
-                StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
-            }
-            else
-            {
-                if (osvi.wSuiteMask & VER_SUITE_DATACENTER)
-                    StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Datacenter Server" ));
-                else if (osvi.wSuiteMask & VER_SUITE_ENTERPRISE)
-                    StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Advanced Server" ));
-                else StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Server" ));
-            }
-        }
-
-        // Include service pack (if any)
-        //
-        if (_tcslen(osvi.szCSDVersion) > 0)
-        {
-            StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(" "));
-            StringCchCat(pszOS, STRING_MAX_SIZE, osvi.szCSDVersion);
-        }
-
-        TCHAR buf[80];
-
-        // Include build number
-        //
-        StringCchPrintf( buf, 80, TEXT(" (build %d)"), osvi.dwBuildNumber);
-        StringCchCat(pszOS, STRING_MAX_SIZE, buf);
-
-        // Call GetNativeSystemInfo if supported or GetSystemInfo otherwise
-        //
-        pGNSI = (PGNSI) GetProcAddress(GetModuleHandle(TEXT("kernel32.dll")), "GetNativeSystemInfo");
-        if (NULL != pGNSI)
-            pGNSI(&si);
-        else
-            GetSystemInfo(&si);
-
-        // Add 64-bit or 32-bit for OS versions "later than" Vista
-        //
-        if (osvi.dwMajorVersion >= 6)
-        {
-            if ((si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_AMD64) ||
-                (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_IA64))
-                StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( ", 64-bit" ));
-            else if (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_INTEL )
-                StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(", 32-bit"));
-        }
-
-        return TRUE;
-    }
-    else
-    {
-        return FALSE;
-   }
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_utility_win.h b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_utility_win.h
deleted file mode 100644
index 9f836eef1e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_utility_win.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
-
-#include "webrtc/modules/audio_device/audio_device_utility.h"
-#include "webrtc/modules/audio_device/include/audio_device.h"
-#include <windows.h>
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityWindows : public AudioDeviceUtility
-{
-public:
-    AudioDeviceUtilityWindows(const int32_t id);
-    ~AudioDeviceUtilityWindows();
-
-    virtual int32_t Init();
-
-private:
-    BOOL GetOSDisplayString(LPTSTR pszOS);
-
-private:
-    CriticalSectionWrapper&         _critSect;
-    int32_t                         _id;
-    AudioDeviceModule::ErrorCode    _lastError;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.cc b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.cc
index bcea3176e9..6f4d7df397 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.cc
@@ -9,11 +9,11 @@
  */
 
 #include "webrtc/modules/audio_device/audio_device_config.h"
-#include "webrtc/modules/audio_device/audio_device_utility.h"
 #include "webrtc/modules/audio_device/win/audio_device_wave_win.h"
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <windows.h>
 #include <objbase.h>    // CoTaskMemAlloc, CoTaskMemFree
@@ -47,7 +47,7 @@ namespace webrtc {
 AudioDeviceWindowsWave::AudioDeviceWindowsWave(const int32_t id) :
     _ptrAudioBuffer(NULL),
     _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _timeEvent(*EventWrapper::Create()),
+    _timeEvent(*EventTimerWrapper::Create()),
     _recStartEvent(*EventWrapper::Create()),
     _playStartEvent(*EventWrapper::Create()),
     _hGetCaptureVolumeThread(NULL),
@@ -206,7 +206,7 @@ int32_t AudioDeviceWindowsWave::Init()
         return 0;
     }
 
-    const uint32_t nowTime(AudioDeviceUtility::GetTimeInMS());
+    const uint32_t nowTime(TickTime::MillisecondTimestamp());
 
     _recordedBytes = 0;
     _prevRecByteCheckTime = nowTime;
@@ -228,15 +228,9 @@ int32_t AudioDeviceWindowsWave::Init()
     }
 
     const char* threadName = "webrtc_audio_module_thread";
-    _ptrThread = ThreadWrapper::CreateThread(ThreadFunc, this, threadName);
-    if (!_ptrThread->Start())
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "failed to start the audio thread");
-        _ptrThread.reset();
-        return -1;
-    }
-    _ptrThread->SetPriority(kRealtimePriority);
+    _ptrThread.reset(new rtc::PlatformThread(ThreadFunc, this, threadName));
+    _ptrThread->Start();
+    _ptrThread->SetPriority(rtc::kRealtimePriority);
 
     const bool periodic(true);
     if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS))
@@ -250,12 +244,8 @@ int32_t AudioDeviceWindowsWave::Init()
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "periodic timer (dT=%d) is now active", TIMER_PERIOD_MS);
 
-    _hGetCaptureVolumeThread = CreateThread(NULL,
-                                            0,
-                                            GetCaptureVolumeThread,
-                                            this,
-                                            0,
-                                            NULL);
+    _hGetCaptureVolumeThread =
+        CreateThread(NULL, 0, GetCaptureVolumeThread, this, 0, NULL);
     if (_hGetCaptureVolumeThread == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -265,12 +255,8 @@ int32_t AudioDeviceWindowsWave::Init()
 
     SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
 
-    _hSetCaptureVolumeThread = CreateThread(NULL,
-                                            0,
-                                            SetCaptureVolumeThread,
-                                            this,
-                                            0,
-                                            NULL);
+    _hSetCaptureVolumeThread =
+        CreateThread(NULL, 0, SetCaptureVolumeThread, this, 0, NULL);
     if (_hSetCaptureVolumeThread == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -303,7 +289,7 @@ int32_t AudioDeviceWindowsWave::Terminate()
 
     if (_ptrThread)
     {
-        ThreadWrapper* tmpThread = _ptrThread.release();
+        rtc::PlatformThread* tmpThread = _ptrThread.release();
         _critSect.Leave();
 
         _timeEvent.Set();
@@ -3052,7 +3038,7 @@ bool AudioDeviceWindowsWave::ThreadProcess()
         return true;
     }
 
-    time = AudioDeviceUtility::GetTimeInMS();
+    time = TickTime::MillisecondTimestamp();
 
     if (_startPlay)
     {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.h b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.h
index d59f9cb6ad..a1cfc6acbf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_device_wave_win.h
@@ -11,13 +11,14 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/win/audio_mixer_manager_win.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 #pragma comment( lib, "winmm.lib" )
 
 namespace webrtc {
+class EventTimerWrapper;
 class EventWrapper;
 
 const uint32_t TIMER_PERIOD_MS = 2;
@@ -211,7 +212,7 @@ private:
     AudioDeviceBuffer*                      _ptrAudioBuffer;
 
     CriticalSectionWrapper&                 _critSect;
-    EventWrapper&                           _timeEvent;
+    EventTimerWrapper&                      _timeEvent;
     EventWrapper&                           _recStartEvent;
     EventWrapper&                           _playStartEvent;
 
@@ -221,7 +222,8 @@ private:
     HANDLE                                  _hShutdownSetVolumeEvent;
     HANDLE                                  _hSetCaptureVolumeEvent;
 
-    rtc::scoped_ptr<ThreadWrapper>          _ptrThread;
+    // TODO(pbos): Remove scoped_ptr usage and use PlatformThread directly
+    rtc::scoped_ptr<rtc::PlatformThread>    _ptrThread;
 
     CriticalSectionWrapper&                 _critSectCb;
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc
index 4d6e7bb9a6..368b54c746 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/audio_device/win/audio_mixer_manager_win.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <assert.h>      // assert()
 #include <strsafe.h>    // StringCchCopy(), StringCchCat(), StringCchPrintf()
@@ -195,7 +195,9 @@ int32_t AudioMixerManager::EnumerateSpeakers()
     for (mixId = 0; mixId < nDevices; mixId++)
     {
         // get capabilities for the specified mixer ID
-        GetCapabilities(mixId, caps);
+        if (!GetCapabilities(mixId, caps))
+            continue;
+
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[mixerID=%d] %s: ", mixId, WideToUTF8(caps.szPname));
         // scan all avaliable destinations for this mixer
         for (destId = 0; destId < caps.cDestinations; destId++)
@@ -280,7 +282,9 @@ int32_t AudioMixerManager::EnumerateMicrophones()
     for (mixId = 0; mixId < nDevices; mixId++)
     {
         // get capabilities for the specified mixer ID
-        GetCapabilities(mixId, caps);
+        if (!GetCapabilities(mixId, caps))
+            continue;
+
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[mixerID=%d] %s: ", mixId, WideToUTF8(caps.szPname));
         // scan all avaliable destinations for this mixer
         for (destId = 0; destId < caps.cDestinations; destId++)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.h b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.h
index 52b46e0201..1e0ab47ad4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/win/audio_mixer_manager_win.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/typedefs.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include <Windows.h>
 #include <mmsystem.h>
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/audio_device_tests.isolate b/media/webrtc/trunk/webrtc/modules/audio_device_tests.isolate
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/audio_device/audio_device_tests.isolate
rename to media/webrtc/trunk/webrtc/modules/audio_device_tests.isolate
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/BUILD.gn b/media/webrtc/trunk/webrtc/modules/audio_processing/BUILD.gn
index 016c684063..9d91911bc2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/BUILD.gn
@@ -30,25 +30,17 @@ source_set("audio_processing") {
     "aec/aec_resampler.c",
     "aec/aec_resampler.h",
     "aec/echo_cancellation.c",
+    "aec/echo_cancellation.h",
     "aec/echo_cancellation_internal.h",
-    "aec/include/echo_cancellation.h",
     "aecm/aecm_core.c",
     "aecm/aecm_core.h",
     "aecm/echo_control_mobile.c",
-    "aecm/include/echo_control_mobile.h",
+    "aecm/echo_control_mobile.h",
     "agc/agc.cc",
     "agc/agc.h",
-    "agc/agc_audio_proc.cc",
-    "agc/agc_audio_proc.h",
-    "agc/agc_audio_proc_internal.h",
     "agc/agc_manager_direct.cc",
     "agc/agc_manager_direct.h",
-    "agc/circular_buffer.cc",
-    "agc/circular_buffer.h",
-    "agc/common.h",
     "agc/gain_map_internal.h",
-    "agc/gmm.cc",
-    "agc/gmm.h",
     "agc/histogram.cc",
     "agc/histogram.h",
     "agc/legacy/analog_agc.c",
@@ -56,22 +48,14 @@ source_set("audio_processing") {
     "agc/legacy/digital_agc.c",
     "agc/legacy/digital_agc.h",
     "agc/legacy/gain_control.h",
-    "agc/noise_gmm_tables.h",
-    "agc/pitch_based_vad.cc",
-    "agc/pitch_based_vad.h",
-    "agc/pitch_internal.cc",
-    "agc/pitch_internal.h",
-    "agc/pole_zero_filter.cc",
-    "agc/pole_zero_filter.h",
-    "agc/standalone_vad.cc",
-    "agc/standalone_vad.h",
     "agc/utility.cc",
     "agc/utility.h",
-    "agc/voice_gmm_tables.h",
     "audio_buffer.cc",
     "audio_buffer.h",
     "audio_processing_impl.cc",
     "audio_processing_impl.h",
+    "beamformer/array_util.cc",
+    "beamformer/array_util.h",
     "beamformer/beamformer.h",
     "beamformer/complex_matrix.h",
     "beamformer/covariance_matrix_generator.cc",
@@ -89,8 +73,15 @@ source_set("audio_processing") {
     "high_pass_filter_impl.cc",
     "high_pass_filter_impl.h",
     "include/audio_processing.h",
+    "intelligibility/intelligibility_enhancer.cc",
+    "intelligibility/intelligibility_enhancer.h",
+    "intelligibility/intelligibility_utils.cc",
+    "intelligibility/intelligibility_utils.h",
     "level_estimator_impl.cc",
     "level_estimator_impl.h",
+    "logging/aec_logging.h",
+    "logging/aec_logging_file_handling.cc",
+    "logging/aec_logging_file_handling.h",
     "noise_suppression_impl.cc",
     "noise_suppression_impl.h",
     "processing_component.cc",
@@ -99,6 +90,8 @@ source_set("audio_processing") {
     "rms_level.h",
     "splitting_filter.cc",
     "splitting_filter.h",
+    "three_band_filter_bank.cc",
+    "three_band_filter_bank.h",
     "transient/common.h",
     "transient/daubechies_8_wavelet_coeffs.h",
     "transient/dyadic_decimator.h",
@@ -119,6 +112,26 @@ source_set("audio_processing") {
     "utility/delay_estimator_internal.h",
     "utility/delay_estimator_wrapper.c",
     "utility/delay_estimator_wrapper.h",
+    "vad/common.h",
+    "vad/gmm.cc",
+    "vad/gmm.h",
+    "vad/noise_gmm_tables.h",
+    "vad/pitch_based_vad.cc",
+    "vad/pitch_based_vad.h",
+    "vad/pitch_internal.cc",
+    "vad/pitch_internal.h",
+    "vad/pole_zero_filter.cc",
+    "vad/pole_zero_filter.h",
+    "vad/standalone_vad.cc",
+    "vad/standalone_vad.h",
+    "vad/vad_audio_proc.cc",
+    "vad/vad_audio_proc.h",
+    "vad/vad_audio_proc_internal.h",
+    "vad/vad_circular_buffer.cc",
+    "vad/vad_circular_buffer.h",
+    "vad/voice_activity_detector.cc",
+    "vad/voice_activity_detector.h",
+    "vad/voice_gmm_tables.h",
     "voice_detection_impl.cc",
     "voice_detection_impl.h",
   ]
@@ -127,7 +140,10 @@ source_set("audio_processing") {
   public_configs = [ "../..:common_inherited_config" ]
 
   defines = []
-  deps = [ "../..:webrtc_common" ]
+  deps = [
+    "../..:webrtc_common",
+    "../audio_coding:isac",
+  ]
 
   if (aec_debug_dump) {
     defines += [ "WEBRTC_AEC_DEBUG_DUMP" ]
@@ -145,8 +161,8 @@ source_set("audio_processing") {
   if (rtc_prefer_fixed_point) {
     defines += [ "WEBRTC_NS_FIXED" ]
     sources += [
-      "ns/include/noise_suppression_x.h",
       "ns/noise_suppression_x.c",
+      "ns/noise_suppression_x.h",
       "ns/nsx_core.c",
       "ns/nsx_core.h",
       "ns/nsx_defines.h",
@@ -160,8 +176,8 @@ source_set("audio_processing") {
     defines += [ "WEBRTC_NS_FLOAT" ]
     sources += [
       "ns/defines.h",
-      "ns/include/noise_suppression.h",
       "ns/noise_suppression.c",
+      "ns/noise_suppression.h",
       "ns/ns_core.c",
       "ns/ns_core.h",
       "ns/windows_private.h",
@@ -172,7 +188,7 @@ source_set("audio_processing") {
     deps += [ ":audio_processing_sse2" ]
   }
 
-  if (rtc_build_armv7_neon || current_cpu == "arm64") {
+  if (rtc_build_with_neon) {
     deps += [ ":audio_processing_neon" ]
   }
 
@@ -210,7 +226,9 @@ source_set("audio_processing") {
 
 if (rtc_enable_protobuf) {
   proto_library("audioproc_debug_proto") {
-    sources = [ "debug.proto" ]
+    sources = [
+      "debug.proto",
+    ]
 
     proto_out_dir = "webrtc/audio_processing"
   }
@@ -223,14 +241,16 @@ if (current_cpu == "x86" || current_cpu == "x64") {
       "aec/aec_rdft_sse2.c",
     ]
 
-    cflags = [ "-msse2" ]
+    if (is_posix) {
+      cflags = [ "-msse2" ]
+    }
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
   }
 }
 
-if (rtc_build_armv7_neon || current_cpu == "arm64") {
+if (rtc_build_with_neon) {
   source_set("audio_processing_neon") {
     sources = [
       "aec/aec_core_neon.c",
@@ -239,30 +259,28 @@ if (rtc_build_armv7_neon || current_cpu == "arm64") {
       "ns/nsx_core_neon.c",
     ]
 
-    configs += [ "../..:common_config" ]
-    public_configs = [ "../..:common_inherited_config" ]
-
-    deps = [ "../../common_audio" ]
-
-    # Enable compilation for the ARM v7 Neon instruction set. This is needed
-    # since //build/config/arm.gni only enables Neon for iOS, not Android.
-    # This provides the same functionality as webrtc/build/arm_neon.gypi.
-    # TODO(kjellander): Investigate if this can be moved into webrtc.gni or
-    # //build/config/arm.gni instead, to reduce code duplication.
-    # Remove the -mfpu=vfpv3-d16 cflag.
-    configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
-
-    # "-mfpu=neon" is not requried for arm64 in GCC.
     if (current_cpu != "arm64") {
-       cflags = [ "-mfpu=neon" ]
+      # Enable compilation for the NEON instruction set. This is needed
+      # since //build/config/arm.gni only enables NEON for iOS, not Android.
+      # This provides the same functionality as webrtc/build/arm_neon.gypi.
+      configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
     }
 
-    # Disable LTO in audio_processing_neon target due to compiler bug.
+    # Disable LTO on NEON targets due to compiler bug.
+    # TODO(fdegans): Enable this. See crbug.com/408997.
     if (rtc_use_lto) {
       cflags -= [
         "-flto",
         "-ffat-lto-objects",
       ]
     }
+
+    configs += [ "../..:common_config" ]
+    public_configs = [ "../..:common_inherited_config" ]
+
+    deps = [
+      "../../common_audio",
+    ]
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/OWNERS b/media/webrtc/trunk/webrtc/modules/audio_processing/OWNERS
index 41a82af20d..d14f7f8614 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/OWNERS
@@ -1,6 +1,6 @@
 aluebs@webrtc.org
-andrew@webrtc.org
-bjornv@webrtc.org
+henrik.lundin@webrtc.org
+peah@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c
index 1722ba82ec..c712f46ae8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c
@@ -30,8 +30,9 @@
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
 #include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
+#include "webrtc/modules/audio_processing/logging/aec_logging.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 extern int AECDebug();
@@ -49,7 +50,6 @@ static const int countLen = 50;
 static const int kDelayMetricsAggregationWindow = 1250;  // 5 seconds at 16 kHz.
 
 // Quantities to control H band scaling for SWB input
-static const int flagHbandCn = 1;  // flag for adding comfort noise in H band
 static const float cnScaleHband =
     (float)0.4;  // scale for comfort noise in H band
 // Initial bin for averaging nlp gain in low band
@@ -109,7 +109,7 @@ ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65] = {
 static const float kDelayQualityThresholdMax = 0.07f;
 static const float kDelayQualityThresholdMin = 0.01f;
 static const int kInitialShiftOffset = 5;
-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_GONK)
+#if !defined(WEBRTC_ANDROID)
 static const int kDelayCorrectionStart = 1500;  // 10 ms chunks
 #endif
 
@@ -140,6 +140,9 @@ WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
 WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
 WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
 WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
+WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
+WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
+WebRtcAecWindowData WebRtcAec_WindowData;
 
 __inline static float MulRe(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bRe - aIm * bIm;
@@ -156,40 +159,49 @@ static int CmpFloat(const void* a, const void* b) {
   return (*da > *db) - (*da < *db);
 }
 
-static void FilterFar(AecCore* aec, float yf[2][PART_LEN1]) {
+static void FilterFar(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
   int i;
-  for (i = 0; i < aec->num_partitions; i++) {
+  for (i = 0; i < num_partitions; i++) {
     int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= aec->num_partitions * (PART_LEN1);
+    if (i + x_fft_buf_block_pos >= num_partitions) {
+      xPos -= num_partitions * (PART_LEN1);
     }
 
     for (j = 0; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
+      y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
+      y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
     }
   }
 }
 
-static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1]) {
-  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-  const float error_threshold = aec->extended_filter_enabled
+static void ScaleErrorSignal(int extended_filter_enabled,
+                             float normal_mu,
+                             float normal_error_threshold,
+                             float x_pow[PART_LEN1],
+                             float ef[2][PART_LEN1]) {
+  const float mu = extended_filter_enabled ? kExtendedMu : normal_mu;
+  const float error_threshold = extended_filter_enabled
                                     ? kExtendedErrorThreshold
-                                    : aec->normal_error_threshold;
+                                    : normal_error_threshold;
   int i;
   float abs_ef;
   for (i = 0; i < (PART_LEN1); i++) {
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
+    ef[0][i] /= (x_pow[i] + 1e-10f);
+    ef[1][i] /= (x_pow[i] + 1e-10f);
     abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
     if (abs_ef > error_threshold) {
@@ -204,59 +216,40 @@ static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1]) {
   }
 }
 
-// Time-unconstrined filter adaptation.
-// TODO(andrew): consider for a low-complexity mode.
-// static void FilterAdaptationUnconstrained(AecCore* aec, float *fft,
-//                                          float ef[2][PART_LEN1]) {
-//  int i, j;
-//  for (i = 0; i < aec->num_partitions; i++) {
-//    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
-//    int pos;
-//    // Check for wrap
-//    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-//      xPos -= aec->num_partitions * PART_LEN1;
-//    }
-//
-//    pos = i * PART_LEN1;
-//
-//    for (j = 0; j < PART_LEN1; j++) {
-//      aec->wfBuf[0][pos + j] += MulRe(aec->xfBuf[0][xPos + j],
-//                                      -aec->xfBuf[1][xPos + j],
-//                                      ef[0][j], ef[1][j]);
-//      aec->wfBuf[1][pos + j] += MulIm(aec->xfBuf[0][xPos + j],
-//                                      -aec->xfBuf[1][xPos + j],
-//                                      ef[0][j], ef[1][j]);
-//    }
-//  }
-//}
 
-static void FilterAdaptation(AecCore* aec, float* fft, float ef[2][PART_LEN1]) {
+static void FilterAdaptation(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
   int i, j;
-  for (i = 0; i < aec->num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * (PART_LEN1);
+  float fft[PART_LEN2];
+  for (i = 0; i < num_partitions; i++) {
+    int xPos = (i + x_fft_buf_block_pos) * (PART_LEN1);
     int pos;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= aec->num_partitions * PART_LEN1;
+    if (i + x_fft_buf_block_pos >= num_partitions) {
+      xPos -= num_partitions * PART_LEN1;
     }
 
     pos = i * PART_LEN1;
 
     for (j = 0; j < PART_LEN; j++) {
 
-      fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
-                         -aec->xfBuf[1][xPos + j],
-                         ef[0][j],
-                         ef[1][j]);
-      fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
-                             -aec->xfBuf[1][xPos + j],
-                             ef[0][j],
-                             ef[1][j]);
+      fft[2 * j] = MulRe(x_fft_buf[0][xPos + j],
+                         -x_fft_buf[1][xPos + j],
+                         e_fft[0][j],
+                         e_fft[1][j]);
+      fft[2 * j + 1] = MulIm(x_fft_buf[0][xPos + j],
+                             -x_fft_buf[1][xPos + j],
+                             e_fft[0][j],
+                             e_fft[1][j]);
     }
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN],
-                   ef[1][PART_LEN]);
+    fft[1] = MulRe(x_fft_buf[0][xPos + PART_LEN],
+                   -x_fft_buf[1][xPos + PART_LEN],
+                   e_fft[0][PART_LEN],
+                   e_fft[1][PART_LEN]);
 
     aec_rdft_inverse_128(fft);
     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
@@ -270,12 +263,12 @@ static void FilterAdaptation(AecCore* aec, float* fft, float ef[2][PART_LEN1]) {
     }
     aec_rdft_forward_128(fft);
 
-    aec->wfBuf[0][pos] += fft[0];
-    aec->wfBuf[0][pos + PART_LEN] += fft[1];
+    h_fft_buf[0][pos] += fft[0];
+    h_fft_buf[0][pos + PART_LEN] += fft[1];
 
     for (j = 1; j < PART_LEN; j++) {
-      aec->wfBuf[0][pos + j] += fft[2 * j];
-      aec->wfBuf[1][pos + j] += fft[2 * j + 1];
+      h_fft_buf[0][pos + j] += fft[2 * j];
+      h_fft_buf[1][pos + j] += fft[2 * j + 1];
     }
   }
 }
@@ -339,12 +332,13 @@ const float WebRtcAec_kMinFarendPSD = 15;
 //  - sde : cross-PSD of near-end and residual echo
 //  - sxd : cross-PSD of near-end and far-end
 //
-// In addition to updating the PSDs, also the filter diverge state is determined
-// upon actions are taken.
+// In addition to updating the PSDs, also the filter diverge state is
+// determined.
 static void SmoothedPSD(AecCore* aec,
                         float efw[2][PART_LEN1],
                         float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1]) {
+                        float xfw[2][PART_LEN1],
+                        int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh = aec->extended_filter_enabled
       ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
@@ -385,15 +379,12 @@ static void SmoothedPSD(AecCore* aec,
     seSum += aec->se[i];
   }
 
-  // Divergent filter safeguard.
+  // Divergent filter safeguard update.
   aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
 
-  if (aec->divergeState)
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-
-  // Reset if error is significantly larger than nearend (13 dB).
-  if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+  // Signal extreme filter divergence if the error is significantly larger
+  // than the nearend (13 dB).
+  *extreme_filter_divergence = (seSum > (19.95f * sdSum));
 }
 
 // Window time domain data to be used by the fft.
@@ -422,32 +413,15 @@ __inline static void StoreAsComplex(const float* data,
 
 static void SubbandCoherence(AecCore* aec,
                              float efw[2][PART_LEN1],
+                             float dfw[2][PART_LEN1],
                              float xfw[2][PART_LEN1],
                              float* fft,
                              float* cohde,
-                             float* cohxd) {
-  float dfw[2][PART_LEN1];
+                             float* cohxd,
+                             int* extreme_filter_divergence) {
   int i;
 
-  if (aec->delayEstCtr == 0)
-    aec->delayIdx = PartitionDelay(aec);
-
-  // Use delayed far.
-  memcpy(xfw,
-         aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  // Windowed near fft
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed error fft
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  SmoothedPSD(aec, efw, dfw, xfw);
+  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
 
   // Subband coherence
   for (i = 0; i < PART_LEN1; i++) {
@@ -463,23 +437,23 @@ static void SubbandCoherence(AecCore* aec,
 static void GetHighbandGain(const float* lambda, float* nlpGainHband) {
   int i;
 
-  nlpGainHband[0] = (float)0.0;
+  *nlpGainHband = (float)0.0;
   for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
-    nlpGainHband[0] += lambda[i];
+    *nlpGainHband += lambda[i];
   }
-  nlpGainHband[0] /= (float)(PART_LEN1 - 1 - freqAvgIc);
+  *nlpGainHband /= (float)(PART_LEN1 - 1 - freqAvgIc);
 }
 
 static void ComfortNoise(AecCore* aec,
                          float efw[2][PART_LEN1],
-                         complex_t* comfortNoiseHband,
+                         float comfortNoiseHband[2][PART_LEN1],
                          const float* noisePow,
                          const float* lambda) {
   int i, num;
   float rand[PART_LEN];
   float noise, noiseAvg, tmp, tmpAvg;
   int16_t randW16[PART_LEN];
-  complex_t u[PART_LEN1];
+  float u[2][PART_LEN1];
 
   const float pi2 = 6.28318530717959f;
 
@@ -491,22 +465,22 @@ static void ComfortNoise(AecCore* aec,
 
   // Reject LF noise
   u[0][0] = 0;
-  u[0][1] = 0;
+  u[1][0] = 0;
   for (i = 1; i < PART_LEN1; i++) {
     tmp = pi2 * rand[i - 1];
 
     noise = sqrtf(noisePow[i]);
-    u[i][0] = noise * cosf(tmp);
-    u[i][1] = -noise * sinf(tmp);
+    u[0][i] = noise * cosf(tmp);
+    u[1][i] = -noise * sinf(tmp);
   }
-  u[PART_LEN][1] = 0;
+  u[1][PART_LEN] = 0;
 
   for (i = 0; i < PART_LEN1; i++) {
     // This is the proper weighting to match the background noise power
     tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
     // tmp = 1 - lambda[i];
-    efw[0][i] += tmp * u[i][0];
-    efw[1][i] += tmp * u[i][1];
+    efw[0][i] += tmp * u[0][i];
+    efw[1][i] += tmp * u[1][i];
   }
 
   // For H band comfort noise
@@ -514,7 +488,7 @@ static void ComfortNoise(AecCore* aec,
   noiseAvg = 0.0;
   tmpAvg = 0.0;
   num = 0;
-  if (aec->num_bands > 1 && flagHbandCn == 1) {
+  if (aec->num_bands > 1) {
 
     // average noise scale
     // average over second half of freq spectrum (i.e., 4->8khz)
@@ -539,21 +513,24 @@ static void ComfortNoise(AecCore* aec,
     // TODO: we should probably have a new random vector here.
     // Reject LF noise
     u[0][0] = 0;
-    u[0][1] = 0;
+    u[1][0] = 0;
     for (i = 1; i < PART_LEN1; i++) {
       tmp = pi2 * rand[i - 1];
 
       // Use average noise for H band
-      u[i][0] = noiseAvg * (float)cos(tmp);
-      u[i][1] = -noiseAvg * (float)sin(tmp);
+      u[0][i] = noiseAvg * (float)cos(tmp);
+      u[1][i] = -noiseAvg * (float)sin(tmp);
     }
-    u[PART_LEN][1] = 0;
+    u[1][PART_LEN] = 0;
 
     for (i = 0; i < PART_LEN1; i++) {
       // Use average NLP weight for H band
-      comfortNoiseHband[i][0] = tmpAvg * u[i][0];
-      comfortNoiseHband[i][1] = tmpAvg * u[i][1];
+      comfortNoiseHband[0][i] = tmpAvg * u[0][i];
+      comfortNoiseHband[1][i] = tmpAvg * u[1][i];
     }
+  } else {
+    memset(comfortNoiseHband, 0,
+           2 * PART_LEN1 * sizeof(comfortNoiseHband[0][0]));
   }
 }
 
@@ -842,21 +819,29 @@ static void UpdateDelayMetrics(AecCore* self) {
   return;
 }
 
-static void TimeToFrequency(float time_data[PART_LEN2],
-                            float freq_data[2][PART_LEN1],
-                            int window) {
-  int i = 0;
-
-  // TODO(bjornv): Should we have a different function/wrapper for windowed FFT?
-  if (window) {
-    for (i = 0; i < PART_LEN; i++) {
-      time_data[i] *= WebRtcAec_sqrtHanning[i];
-      time_data[PART_LEN + i] *= WebRtcAec_sqrtHanning[PART_LEN - i];
-    }
+static void ScaledInverseFft(float freq_data[2][PART_LEN1],
+                             float time_data[PART_LEN2],
+                             float scale,
+                             int conjugate) {
+  int i;
+  const float normalization = scale / ((float)PART_LEN2);
+  const float sign = (conjugate ? -1 : 1);
+  time_data[0] = freq_data[0][0] * normalization;
+  time_data[1] = freq_data[0][PART_LEN] * normalization;
+  for (i = 1; i < PART_LEN; i++) {
+    time_data[2 * i] = freq_data[0][i] * normalization;
+    time_data[2 * i + 1] = sign * freq_data[1][i] * normalization;
   }
+  aec_rdft_inverse_128(time_data);
+}
 
+
+static void Fft(float time_data[PART_LEN2],
+                float freq_data[2][PART_LEN1]) {
+  int i;
   aec_rdft_forward_128(time_data);
-  // Reorder.
+
+  // Reorder fft output data.
   freq_data[1][0] = 0;
   freq_data[1][PART_LEN] = 0;
   freq_data[0][0] = time_data[0];
@@ -867,19 +852,12 @@ static void TimeToFrequency(float time_data[PART_LEN2],
   }
 }
 
-static int MoveFarReadPtrWithoutSystemDelayUpdate(AecCore* self, int elements) {
-  WebRtc_MoveReadPtr(self->far_buf_windowed, elements);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  WebRtc_MoveReadPtr(self->far_time_buf, elements);
-#endif
-  return WebRtc_MoveReadPtr(self->far_buf, elements);
-}
 
 static int SignalBasedDelayCorrection(AecCore* self) {
   int delay_correction = 0;
   int last_delay = -2;
   assert(self != NULL);
-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_GONK)
+#if !defined(WEBRTC_ANDROID)
   // On desktops, turn on correction after |kDelayCorrectionStart| frames.  This
   // is to let the delay estimation get a chance to converge.  Also, if the
   // playout audio volume is low (or even muted) the delay estimation can return
@@ -914,7 +892,7 @@ static int SignalBasedDelayCorrection(AecCore* self) {
     const int upper_bound = self->num_partitions * 3 / 4;
     const int do_correction = delay <= lower_bound || delay > upper_bound;
     if (do_correction == 1) {
-      int available_read = (int)WebRtc_available_read(self->far_buf);
+      int available_read = (int)WebRtc_available_read(self->far_time_buf);
       // With |shift_offset| we gradually rely on the delay estimates.  For
       // positive delays we reduce the correction by |shift_offset| to lower the
       // risk of pushing the AEC into a non causal state.  For negative delays
@@ -948,42 +926,6 @@ static int SignalBasedDelayCorrection(AecCore* self) {
 }
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
-// Open a new Wav file for writing. If it was already open with a different
-// sample frequency, close it first.
-static void ReopenWav(rtc_WavWriter** wav_file,
-                      const char* name,
-                      int seq1,
-                      int seq2,
-                      int sample_rate) {
-  int written /*UNUSED*/;
-  char path[1024];
-  char *filename;
-  if (*wav_file) {
-    if (rtc_WavSampleRate(*wav_file) == sample_rate)
-      return;
-    rtc_WavClose(*wav_file);
-    *wav_file = NULL;
-  }
-  AECDebugFilenameBase(path, sizeof(path));
-  filename = path + strlen(path);
-  if (filename > path) {
-#ifdef WEBRTC_WIN
-    if (*(filename-1) != '\\') {
-      *filename++ = '\\';
-    }
-#else
-    if (*(filename-1) != '/') {
-      *filename++ = '/';
-    }
-#endif
-  }
-  written = snprintf(filename, sizeof(path) - (filename-path), "%s%d-%d.wav",
-                     name, seq1, seq2);
-  assert(written >= 0);  // no output error
-  assert(filename+written < path + sizeof(path)-1);  // buffer was large enough
-  *wav_file = rtc_WavOpen(path, sample_rate, 1);
-}
-
 static void
 OpenCoreDebugFiles(AecCore* aec, int *aec_instance_count)
 {
@@ -992,47 +934,134 @@ OpenCoreDebugFiles(AecCore* aec, int *aec_instance_count)
     if (!aec->farFile)
     {
       int process_rate = aec->sampFreq > 16000 ? 16000 : aec->sampFreq;
-      ReopenWav(&aec->farFile, "aec_far",
-                aec->instance_index, aec->debug_dump_count, process_rate);
-      ReopenWav(&aec->nearFile, "aec_near",
-                aec->instance_index, aec->debug_dump_count, process_rate);
-      ReopenWav(&aec->outFile, "aec_out",
-                aec->instance_index, aec->debug_dump_count, process_rate);
-      ReopenWav(&aec->outLinearFile, "aec_out_linear",
-                aec->instance_index, aec->debug_dump_count, process_rate);
-      ReopenWav(&aec->e_fft_file, "aec_fft",
-                aec->instance_index, aec->debug_dump_count, process_rate);
+      RTC_AEC_DEBUG_WAV_REOPEN("aec_far", aec->instance_index, aec->debug_dump_count,
+                               process_rate, &aec->farFile);
+      RTC_AEC_DEBUG_WAV_REOPEN("aec_near", aec->instance_index, aec->debug_dump_count,
+                               process_rate, &aec->nearFile);
+      RTC_AEC_DEBUG_WAV_REOPEN("aec_out", aec->instance_index, aec->debug_dump_count,
+                               process_rate, &aec->outFile);
+      RTC_AEC_DEBUG_WAV_REOPEN("aec_out_linear", aec->instance_index, aec->debug_dump_count,
+                               process_rate, &aec->outLinearFile);
+      RTC_AEC_DEBUG_RAW_CLOSE(aec->e_fft_file);
+      RTC_AEC_DEBUG_RAW_OPEN("aec_fft", aec->instance_index, aec->debug_dump_count,
+                             &aec->e_fft_file);
       ++aec->debug_dump_count;
     }
   } else {
     if (aec->farFile) {
-      rtc_WavClose(aec->farFile);
+      RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile);
     }
     if (aec->nearFile) {
-      rtc_WavClose(aec->nearFile);
+      RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile);
     }
     if (aec->outFile) {
-      rtc_WavClose(aec->outFile);
+      RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile);
     }
     if (aec->outLinearFile) {
-      rtc_WavClose(aec->outLinearFile);
+      RTC_AEC_DEBUG_WAV_CLOSE(aec->outLinearFile);
     }
     if (aec->e_fft_file) {
-      rtc_WavClose(aec->e_fft_file);
+      RTC_AEC_DEBUG_RAW_CLOSE(aec->e_fft_file);
     }
-    aec->outLinearFile = aec->outFile = aec->nearFile = aec->farFile = aec->e_fft_file = NULL;
+    aec->outLinearFile = aec->outFile = aec->nearFile = aec->farFile = NULL;
+    aec->e_fft_file = NULL;
     aec->debugWritten = 0;
   }
 }
 #endif
 
-static void NonLinearProcessing(AecCore* aec,
-                                float* output,
-                                float* const* outputH) {
-  float efw[2][PART_LEN1], xfw[2][PART_LEN1];
-  complex_t comfortNoiseHband[PART_LEN1];
+static void EchoSubtraction(
+    AecCore* aec,
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    int metrics_mode,
+    int extended_filter_enabled,
+    float normal_mu,
+    float normal_error_threshold,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float* const y,
+    float x_pow[PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    PowerLevel* linout_level,
+    float echo_subtractor_output[PART_LEN]) {
+  float s_fft[2][PART_LEN1];
+  float e_extended[PART_LEN2];
+  float s_extended[PART_LEN2];
+  float *s;
+  float e[PART_LEN];
+  float e_fft[2][PART_LEN1];
+  int i;
+  memset(s_fft, 0, sizeof(s_fft));
+
+  // Conditionally reset the echo subtraction filter if the filter has diverged
+  // significantly.
+  if (!aec->extended_filter_enabled &&
+      aec->extreme_filter_divergence) {
+    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+    aec->extreme_filter_divergence = 0;
+  }
+
+  // Produce echo estimate s_fft.
+  WebRtcAec_FilterFar(num_partitions,
+                      x_fft_buf_block_pos,
+                      x_fft_buf,
+                      h_fft_buf,
+                      s_fft);
+
+  // Compute the time-domain echo estimate s.
+  ScaledInverseFft(s_fft, s_extended, 2.0f, 0);
+  s = &s_extended[PART_LEN];
+
+  // Compute the time-domain echo prediction error.
+  for (i = 0; i < PART_LEN; ++i) {
+    e[i] = y[i] - s[i];
+  }
+
+  // Compute the frequency domain echo prediction error.
+  memset(e_extended, 0, sizeof(float) * PART_LEN);
+  memcpy(e_extended + PART_LEN, e, sizeof(float) * PART_LEN);
+  Fft(e_extended, e_fft);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+  if (aec->e_fft_file) {
+    RTC_AEC_DEBUG_RAW_WRITE(aec->e_fft_file,
+                            &e_fft[0][0],
+                            sizeof(e_fft[0][0]) * PART_LEN1 * 2);
+  }
+#endif
+
+  if (metrics_mode == 1) {
+    // Note that the first PART_LEN samples in fft (before transformation) are
+    // zero. Hence, the scaling by two in UpdateLevel() should not be
+    // performed. That scaling is taken care of in UpdateMetrics() instead.
+    UpdateLevel(linout_level, e_fft);
+  }
+
+  // Scale error signal inversely with far power.
+  WebRtcAec_ScaleErrorSignal(extended_filter_enabled,
+                             normal_mu,
+                             normal_error_threshold,
+                             x_pow,
+                             e_fft);
+  WebRtcAec_FilterAdaptation(num_partitions,
+                             x_fft_buf_block_pos,
+                             x_fft_buf,
+                             e_fft,
+                             h_fft_buf);
+  memcpy(echo_subtractor_output, e, sizeof(float) * PART_LEN);
+}
+
+
+static void EchoSuppression(AecCore* aec,
+                            float farend[PART_LEN2],
+                            float* echo_subtractor_output,
+                            float* output,
+                            float* const* outputH) {
+  float efw[2][PART_LEN1];
+  float xfw[2][PART_LEN1];
+  float dfw[2][PART_LEN1];
+  float comfortNoiseHband[2][PART_LEN1];
   float fft[PART_LEN2];
-  float scale, dtmp;
   float nlpGainHband;
   int i;
   size_t j;
@@ -1056,27 +1085,51 @@ static void NonLinearProcessing(AecCore* aec,
 
   float* xfw_ptr = NULL;
 
-  aec->delayEstCtr++;
-  if (aec->delayEstCtr == delayEstInterval) {
-    aec->delayEstCtr = 0;
-  }
+  // Update eBuf with echo subtractor output.
+  memcpy(aec->eBuf + PART_LEN,
+         echo_subtractor_output,
+         sizeof(float) * PART_LEN);
 
-  // initialize comfort noise for H band
-  memset(comfortNoiseHband, 0, sizeof(comfortNoiseHband));
-  nlpGainHband = (float)0.0;
-  dtmp = (float)0.0;
+  // Analysis filter banks for the echo suppressor.
+  // Windowed near-end ffts.
+  WindowData(fft, aec->dBuf);
+  aec_rdft_forward_128(fft);
+  StoreAsComplex(fft, dfw);
+
+  // Windowed echo suppressor output ffts.
+  WindowData(fft, aec->eBuf);
+  aec_rdft_forward_128(fft);
+  StoreAsComplex(fft, efw);
 
-  // We should always have at least one element stored in |far_buf|.
-  assert(WebRtc_available_read(aec->far_buf_windowed) > 0);
   // NLP
-  WebRtc_ReadBuffer(aec->far_buf_windowed, (void**)&xfw_ptr, &xfw[0][0], 1);
 
-  // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of
-  // |xfwBuf|.
+  // Convert far-end partition to the frequency domain with windowing.
+  WindowData(fft, farend);
+  Fft(fft, xfw);
+  xfw_ptr = &xfw[0][0];
+
   // Buffer far.
   memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
 
-  WebRtcAec_SubbandCoherence(aec, efw, xfw, fft, cohde, cohxd);
+  aec->delayEstCtr++;
+  if (aec->delayEstCtr == delayEstInterval) {
+    aec->delayEstCtr = 0;
+    aec->delayIdx = WebRtcAec_PartitionDelay(aec);
+  }
+
+  // Use delayed far.
+  memcpy(xfw,
+         aec->xfwBuf + aec->delayIdx * PART_LEN1,
+         sizeof(xfw[0][0]) * 2 * PART_LEN1);
+
+  WebRtcAec_SubbandCoherence(aec, efw, dfw, xfw, fft, cohde, cohxd,
+                             &aec->extreme_filter_divergence);
+
+  // Select the microphone signal as output if the filter is deemed to have
+  // diverged.
+  if (aec->divergeState) {
+    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
+  }
 
   hNlXdAvg = 0;
   for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
@@ -1182,67 +1235,51 @@ static void NonLinearProcessing(AecCore* aec,
     // scaling only in UpdateMetrics().
     UpdateLevel(&aec->nlpoutlevel, efw);
   }
+
   // Inverse error fft.
-  fft[0] = efw[0][0];
-  fft[1] = efw[0][PART_LEN];
-  for (i = 1; i < PART_LEN; i++) {
-    fft[2 * i] = efw[0][i];
-    // Sign change required by Ooura fft.
-    fft[2 * i + 1] = -efw[1][i];
-  }
-  aec_rdft_inverse_128(fft);
+  ScaledInverseFft(efw, fft, 2.0f, 1);
 
   // Overlap and add to obtain output.
-  scale = 2.0f / PART_LEN2;
   for (i = 0; i < PART_LEN; i++) {
-    fft[i] *= scale;  // fft scaling
-    fft[i] = fft[i] * WebRtcAec_sqrtHanning[i] + aec->outBuf[i];
-
-    fft[PART_LEN + i] *= scale;  // fft scaling
-    aec->outBuf[i] = fft[PART_LEN + i] * WebRtcAec_sqrtHanning[PART_LEN - i];
+    output[i] = (fft[i] * WebRtcAec_sqrtHanning[i] +
+                 aec->outBuf[i] * WebRtcAec_sqrtHanning[PART_LEN - i]);
 
     // Saturate output to keep it in the allowed range.
     output[i] = WEBRTC_SPL_SAT(
-        WEBRTC_SPL_WORD16_MAX, fft[i], WEBRTC_SPL_WORD16_MIN);
+        WEBRTC_SPL_WORD16_MAX, output[i], WEBRTC_SPL_WORD16_MIN);
   }
+  memcpy(aec->outBuf, &fft[PART_LEN], PART_LEN * sizeof(aec->outBuf[0]));
 
   // For H band
   if (aec->num_bands > 1) {
-
     // H band gain
     // average nlp over low band: average over second half of freq spectrum
     // (4->8khz)
     GetHighbandGain(hNl, &nlpGainHband);
 
     // Inverse comfort_noise
-    if (flagHbandCn == 1) {
-      fft[0] = comfortNoiseHband[0][0];
-      fft[1] = comfortNoiseHband[PART_LEN][0];
-      for (i = 1; i < PART_LEN; i++) {
-        fft[2 * i] = comfortNoiseHband[i][0];
-        fft[2 * i + 1] = comfortNoiseHband[i][1];
-      }
-      aec_rdft_inverse_128(fft);
-      scale = 2.0f / PART_LEN2;
-    }
+    ScaledInverseFft(comfortNoiseHband, fft, 2.0f, 0);
 
     // compute gain factor
     for (j = 0; j < aec->num_bands - 1; ++j) {
       for (i = 0; i < PART_LEN; i++) {
-        dtmp = aec->dBufH[j][i];
-        dtmp = dtmp * nlpGainHband;  // for variable gain
-
-        // add some comfort noise where Hband is attenuated
-        if (flagHbandCn == 1 && j == 0) {
-          fft[i] *= scale;  // fft scaling
-          dtmp += cnScaleHband * fft[i];
-        }
-
-        // Saturate output to keep it in the allowed range.
-        outputH[j][i] = WEBRTC_SPL_SAT(
-            WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN);
+        outputH[j][i] = aec->dBufH[j][i] * nlpGainHband;
       }
     }
+
+    // Add some comfort noise where Hband is attenuated.
+    for (i = 0; i < PART_LEN; i++) {
+      outputH[0][i] += cnScaleHband * fft[i];
+    }
+
+    // Saturate output to keep it in the allowed range.
+    for (j = 0; j < aec->num_bands - 1; ++j) {
+      for (i = 0; i < PART_LEN; i++) {
+        outputH[j][i] = WEBRTC_SPL_SAT(
+            WEBRTC_SPL_WORD16_MAX, outputH[j][i], WEBRTC_SPL_WORD16_MIN);
+      }
+    }
+
   }
 
   // Copy the current block to the old position.
@@ -1261,11 +1298,9 @@ static void NonLinearProcessing(AecCore* aec,
 
 static void ProcessBlock(AecCore* aec) {
   size_t i;
-  float y[PART_LEN], e[PART_LEN];
-  float scale;
 
   float fft[PART_LEN2];
-  float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
+  float xf[2][PART_LEN1];
   float df[2][PART_LEN1];
   float far_spectrum = 0.0f;
   float near_spectrum = 0.0f;
@@ -1282,15 +1317,18 @@ static void ProcessBlock(AecCore* aec) {
 
   float nearend[PART_LEN];
   float* nearend_ptr = NULL;
+  float farend[PART_LEN2];
+  float* farend_ptr = NULL;
+  float echo_subtractor_output[PART_LEN];
   float output[PART_LEN];
   float outputH[NUM_HIGH_BANDS_MAX][PART_LEN];
   float* outputH_ptr[NUM_HIGH_BANDS_MAX];
+  float* xf_ptr = NULL;
+
   for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
     outputH_ptr[i] = outputH[i];
   }
 
-  float* xf_ptr = NULL;
-
   // Concatenate old and new nearend blocks.
   for (i = 0; i < aec->num_bands - 1; ++i) {
     WebRtc_ReadBuffer(aec->nearFrBufH[i],
@@ -1302,17 +1340,18 @@ static void ProcessBlock(AecCore* aec) {
   WebRtc_ReadBuffer(aec->nearFrBuf, (void**)&nearend_ptr, nearend, PART_LEN);
   memcpy(aec->dBuf + PART_LEN, nearend_ptr, sizeof(nearend));
 
-  // ---------- Ooura fft ----------
+  // We should always have at least one element stored in |far_buf|.
+  assert(WebRtc_available_read(aec->far_time_buf) > 0);
+  WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1);
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   {
-    float farend[PART_LEN];
-    float* farend_ptr = NULL;
-    WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1);
+    // TODO(minyue): |farend_ptr| starts from buffered samples. This will be
+    // modified when |aec->far_time_buf| is revised.
     OpenCoreDebugFiles(aec, &webrtc_aec_instance_count);
     if (aec->farFile) {
-      rtc_WavWriteSamples(aec->farFile, farend_ptr, PART_LEN);
-      rtc_WavWriteSamples(aec->nearFile, nearend_ptr, PART_LEN);
+      RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, &farend_ptr[PART_LEN], PART_LEN);
+      RTC_AEC_DEBUG_WAV_WRITE(aec->nearFile, nearend_ptr, PART_LEN);
       aec->debugWritten += sizeof(int16_t) * PART_LEN;
       if (aec->debugWritten >= AECDebugMaxSize()) {
         AECDebugEnable(0);
@@ -1321,13 +1360,14 @@ static void ProcessBlock(AecCore* aec) {
   }
 #endif
 
-  // We should always have at least one element stored in |far_buf|.
-  assert(WebRtc_available_read(aec->far_buf) > 0);
-  WebRtc_ReadBuffer(aec->far_buf, (void**)&xf_ptr, &xf[0][0], 1);
+  // Convert far-end signal to the frequency domain.
+  memcpy(fft, farend_ptr, sizeof(float) * PART_LEN2);
+  Fft(fft, xf);
+  xf_ptr = &xf[0][0];
 
   // Near fft
   memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
-  TimeToFrequency(fft, df, 0);
+  Fft(fft, df);
 
   // Power smoothing
   for (i = 0; i < PART_LEN1; i++) {
@@ -1405,63 +1445,24 @@ static void ProcessBlock(AecCore* aec) {
          &xf_ptr[PART_LEN1],
          sizeof(float) * PART_LEN1);
 
-  memset(yf, 0, sizeof(yf));
+  // Perform echo subtraction.
+  EchoSubtraction(aec,
+                  aec->num_partitions,
+                  aec->xfBufBlockPos,
+                  aec->metricsMode,
+                  aec->extended_filter_enabled,
+                  aec->normal_mu,
+                  aec->normal_error_threshold,
+                  aec->xfBuf,
+                  nearend_ptr,
+                  aec->xPow,
+                  aec->wfBuf,
+                  &aec->linoutlevel,
+                  echo_subtractor_output);
 
-  // Filter far
-  WebRtcAec_FilterFar(aec, yf);
 
-  // Inverse fft to obtain echo estimate and error.
-  fft[0] = yf[0][0];
-  fft[1] = yf[0][PART_LEN];
-  for (i = 1; i < PART_LEN; i++) {
-    fft[2 * i] = yf[0][i];
-    fft[2 * i + 1] = yf[1][i];
-  }
-  aec_rdft_inverse_128(fft);
-
-  scale = 2.0f / PART_LEN2;
-  for (i = 0; i < PART_LEN; i++) {
-    y[i] = fft[PART_LEN + i] * scale;  // fft scaling
-  }
-
-  for (i = 0; i < PART_LEN; i++) {
-    e[i] = nearend_ptr[i] - y[i];
-  }
-
-  // Error fft
-  memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
-  memset(fft, 0, sizeof(float) * PART_LEN);
-  memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
-  // TODO(bjornv): Change to use TimeToFrequency().
-  aec_rdft_forward_128(fft);
-
-  ef[1][0] = 0;
-  ef[1][PART_LEN] = 0;
-  ef[0][0] = fft[0];
-  ef[0][PART_LEN] = fft[1];
-  for (i = 1; i < PART_LEN; i++) {
-    ef[0][i] = fft[2 * i];
-    ef[1][i] = fft[2 * i + 1];
-  }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  if (aec->e_fft_file) {
-    rtc_WavWriteSamples(aec->e_fft_file, &ef[0][0],
-                        sizeof(ef[0][0]) * PART_LEN1 * 2);
-  }
-#endif
-
-  if (aec->metricsMode == 1) {
-    // Note that the first PART_LEN samples in fft (before transformation) are
-    // zero. Hence, the scaling by two in UpdateLevel() should not be
-    // performed. That scaling is taken care of in UpdateMetrics() instead.
-    UpdateLevel(&aec->linoutlevel, ef);
-  }
-
-  // Scale error signal inversely with far power.
-  WebRtcAec_ScaleErrorSignal(aec, ef);
-  WebRtcAec_FilterAdaptation(aec, fft, ef);
-  NonLinearProcessing(aec, output, outputH_ptr);
+  // Perform echo suppression.
+  EchoSuppression(aec, farend_ptr, echo_subtractor_output, output, outputH_ptr);
 
   if (aec->metricsMode == 1) {
     // Update power levels and echo metrics
@@ -1480,8 +1481,8 @@ static void ProcessBlock(AecCore* aec) {
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   OpenCoreDebugFiles(aec, &webrtc_aec_instance_count);
   if (aec->outLinearFile) {
-    rtc_WavWriteSamples(aec->outLinearFile, e, PART_LEN);
-    rtc_WavWriteSamples(aec->outFile, output, PART_LEN);
+    RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, echo_subtractor_output, PART_LEN);
+    RTC_AEC_DEBUG_WAV_WRITE(aec->outFile, output, PART_LEN);
   }
 #endif
 }
@@ -1493,9 +1494,6 @@ AecCore* WebRtcAec_CreateAec() {
     return NULL;
   }
 
-  // set the mem with 0 in order to prevent garbage data
-  memset(aec, 0, sizeof(*aec));
-
   aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
   if (!aec->nearFrBuf) {
     WebRtcAec_FreeAec(aec);
@@ -1524,30 +1522,23 @@ AecCore* WebRtcAec_CreateAec() {
   }
 
   // Create far-end buffers.
-  aec->far_buf =
-      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
-  if (!aec->far_buf) {
-    WebRtcAec_FreeAec(aec);
-    return NULL;
-  }
-  aec->far_buf_windowed =
-      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
-  if (!aec->far_buf_windowed) {
-    WebRtcAec_FreeAec(aec);
-    return NULL;
-  }
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  aec->instance_index = webrtc_aec_instance_count;
+  // For bit exactness with legacy code, each element in |far_time_buf| is
+  // supposed to contain |PART_LEN2| samples with an overlap of |PART_LEN|
+  // samples from the last frame.
+  // TODO(minyue): reduce |far_time_buf| to non-overlapped |PART_LEN| samples.
   aec->far_time_buf =
-      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN);
+      WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN2);
   if (!aec->far_time_buf) {
     WebRtcAec_FreeAec(aec);
     return NULL;
   }
-  aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = aec->e_fft_file = NULL;
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+  aec->instance_index = webrtc_aec_instance_count;
+  aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL;
+  aec->e_fft_file = NULL;
   aec->debug_dump_count = 0;
   aec->debugWritten = 0;
-  OpenCoreDebugFiles(aec, &webrtc_aec_instance_count);
 #endif
   aec->delay_estimator_farend =
       WebRtc_CreateDelayEstimatorFarend(PART_LEN1, kHistorySizeBlocks);
@@ -1563,7 +1554,7 @@ AecCore* WebRtcAec_CreateAec() {
     WebRtcAec_FreeAec(aec);
     return NULL;
   }
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_GONK)
+#ifdef WEBRTC_ANDROID
   aec->delay_agnostic_enabled = 1;  // DA-AEC enabled by default.
   // DA-AEC assumes the system is causal from the beginning and will self adjust
   // the lookahead when shifting is required.
@@ -1584,6 +1575,10 @@ AecCore* WebRtcAec_CreateAec() {
     WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
     WebRtcAec_ComfortNoise = ComfortNoise;
     WebRtcAec_SubbandCoherence = SubbandCoherence;
+    WebRtcAec_StoreAsComplex = StoreAsComplex;
+    WebRtcAec_PartitionDelay = PartitionDelay;
+    WebRtcAec_WindowData = WindowData;
+
 
 #if defined(WEBRTC_ARCH_X86_FAMILY)
     if (WebRtc_GetCPUInfo(kSSE2)) {
@@ -1595,9 +1590,9 @@ AecCore* WebRtcAec_CreateAec() {
     WebRtcAec_InitAec_mips();
 #endif
 
-#if defined(WEBRTC_ARCH_ARM_NEON)
+#if defined(WEBRTC_HAS_NEON)
     WebRtcAec_InitAec_neon();
-#elif defined(WEBRTC_DETECT_ARM_NEON)
+#elif defined(WEBRTC_DETECT_NEON)
     if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
       WebRtcAec_InitAec_neon();
     }
@@ -1622,19 +1617,18 @@ void WebRtcAec_FreeAec(AecCore* aec) {
     WebRtc_FreeBuffer(aec->outFrBufH[i]);
   }
 
-  WebRtc_FreeBuffer(aec->far_buf);
-  WebRtc_FreeBuffer(aec->far_buf_windowed);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
   WebRtc_FreeBuffer(aec->far_time_buf);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
   if (aec->farFile) {
-    // we don't let one be open and not the others
-    rtc_WavClose(aec->farFile);
-    rtc_WavClose(aec->nearFile);
-    rtc_WavClose(aec->outFile);
-    rtc_WavClose(aec->outLinearFile);
-    rtc_WavClose(aec->e_fft_file);
+    RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile);
+    RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile);
+    RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile);
+    RTC_AEC_DEBUG_WAV_CLOSE(aec->outLinearFile);
+    RTC_AEC_DEBUG_RAW_CLOSE(aec->e_fft_file);
   }
 #endif
+
   WebRtc_FreeDelayEstimator(aec->delay_estimator);
   WebRtc_FreeDelayEstimatorFarend(aec->delay_estimator_farend);
 
@@ -1664,12 +1658,12 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   }
 
   // Initialize far-end buffers.
-  WebRtc_InitBuffer(aec->far_buf);
-  WebRtc_InitBuffer(aec->far_buf_windowed);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
   WebRtc_InitBuffer(aec->far_time_buf);
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
   aec->instance_index = webrtc_aec_instance_count;
   OpenCoreDebugFiles(aec, &webrtc_aec_instance_count);
+  ++aec->debug_dump_count;
 #endif
   aec->system_delay = 0;
 
@@ -1783,6 +1777,8 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   aec->seed = 777;
   aec->delayEstCtr = 0;
 
+  aec->extreme_filter_divergence = 0;
+
   // Metrics disabled by default
   aec->metricsMode = 0;
   InitMetrics(aec);
@@ -1790,27 +1786,22 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   return 0;
 }
 
-void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) {
-  float fft[PART_LEN2];
-  float xf[2][PART_LEN1];
 
+// For bit exactness with a legacy code, |farend| is supposed to contain
+// |PART_LEN2| samples with an overlap of |PART_LEN| samples from the last
+// frame.
+// TODO(minyue): reduce |farend| to non-overlapped |PART_LEN| samples.
+void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) {
   // Check if the buffer is full, and in that case flush the oldest data.
-  if (WebRtc_available_write(aec->far_buf) < 1) {
+  if (WebRtc_available_write(aec->far_time_buf) < 1) {
     WebRtcAec_MoveFarReadPtr(aec, 1);
   }
-  // Convert far-end partition to the frequency domain without windowing.
-  memcpy(fft, farend, sizeof(float) * PART_LEN2);
-  TimeToFrequency(fft, xf, 0);
-  WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
 
-  // Convert far-end partition to the frequency domain with windowing.
-  memcpy(fft, farend, sizeof(float) * PART_LEN2);
-  TimeToFrequency(fft, xf, 1);
-  WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
+  WebRtc_WriteBuffer(aec->far_time_buf, farend, 1);
 }
 
 int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) {
-  int elements_moved = MoveFarReadPtrWithoutSystemDelayUpdate(aec, elements);
+  int elements_moved = WebRtc_MoveReadPtr(aec->far_time_buf, elements);
   aec->system_delay -= elements_moved * PART_LEN;
   return elements_moved;
 }
@@ -1884,14 +1875,14 @@ void WebRtcAec_ProcessFrames(AecCore* aec,
       // rounding, like -16.
       int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
       int moved_elements =
-          MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
+          WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
       aec->knownDelay -= moved_elements * PART_LEN;
     } else {
       // 2 b) Apply signal based delay correction.
       int move_elements = SignalBasedDelayCorrection(aec);
       int moved_elements =
-          MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
-      int far_near_buffer_diff = WebRtc_available_read(aec->far_buf) -
+          WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+      int far_near_buffer_diff = WebRtc_available_read(aec->far_time_buf) -
           WebRtc_available_read(aec->nearFrBuf) / PART_LEN;
       WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
       WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
@@ -1970,10 +1961,6 @@ void WebRtcAec_GetEchoStats(AecCore* self,
   *a_nlp = self->aNlp;
 }
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-void* WebRtcAec_far_time_buf(AecCore* self) { return self->far_time_buf; }
-#endif
-
 void WebRtcAec_SetConfigCore(AecCore* self,
                              int nlp_mode,
                              int metrics_mode,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.h
index 8d29533992..241f077524 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.h
@@ -60,7 +60,7 @@ void WebRtcAec_InitAec_SSE2(void);
 #if defined(MIPS_FPU_LE)
 void WebRtcAec_InitAec_mips(void);
 #endif
-#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
+#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 void WebRtcAec_InitAec_neon(void);
 #endif
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_internal.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_internal.h
index 6045e00e04..85f9fe43be 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -95,8 +95,8 @@ struct AecCore {
 
   int xfBufBlockPos;
 
-  RingBuffer* far_buf;
-  RingBuffer* far_buf_windowed;
+  RingBuffer* far_time_buf;
+
   int system_delay;  // Current system delay buffered in AEC.
 
   int mult;  // sampling frequency multiple
@@ -152,6 +152,10 @@ struct AecCore {
   // Runtime selection of number of filter partitions.
   int num_partitions;
 
+  // Flag that extreme filter divergence has been detected by the Echo
+  // Suppressor.
+  int extreme_filter_divergence;
+
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   // Sequence number of this AEC instance, so that different instances can
   // choose different dump file names.
@@ -161,23 +165,34 @@ struct AecCore {
   // each time.
   int debug_dump_count;
 
-  RingBuffer* far_time_buf;
   rtc_WavWriter* farFile;
   rtc_WavWriter* nearFile;
   rtc_WavWriter* outFile;
   rtc_WavWriter* outLinearFile;
-  rtc_WavWriter* e_fft_file;
+  FILE* e_fft_file;
   uint32_t debugWritten;
 #endif
 };
 
-typedef void (*WebRtcAecFilterFar)(AecCore* aec, float yf[2][PART_LEN1]);
+typedef void (*WebRtcAecFilterFar)(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]);
 extern WebRtcAecFilterFar WebRtcAec_FilterFar;
-typedef void (*WebRtcAecScaleErrorSignal)(AecCore* aec, float ef[2][PART_LEN1]);
-extern WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
-typedef void (*WebRtcAecFilterAdaptation)(AecCore* aec,
-                                          float* fft,
+typedef void (*WebRtcAecScaleErrorSignal)(int extended_filter_enabled,
+                                          float normal_mu,
+                                          float normal_error_threshold,
+                                          float x_pow[PART_LEN1],
                                           float ef[2][PART_LEN1]);
+extern WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
+typedef void (*WebRtcAecFilterAdaptation)(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
 extern WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
 typedef void (*WebRtcAecOverdriveAndSuppress)(AecCore* aec,
                                               float hNl[PART_LEN1],
@@ -187,17 +202,29 @@ extern WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
 
 typedef void (*WebRtcAecComfortNoise)(AecCore* aec,
                                       float efw[2][PART_LEN1],
-                                      complex_t* comfortNoiseHband,
+                                      float comfortNoiseHband[2][PART_LEN1],
                                       const float* noisePow,
                                       const float* lambda);
 extern WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
 
 typedef void (*WebRtcAecSubBandCoherence)(AecCore* aec,
                                           float efw[2][PART_LEN1],
+                                          float dfw[2][PART_LEN1],
                                           float xfw[2][PART_LEN1],
                                           float* fft,
                                           float* cohde,
-                                          float* cohxd);
+                                          float* cohxd,
+                                          int* extreme_filter_divergence);
 extern WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
 
+typedef int (*WebRtcAecPartitionDelay)(const AecCore* aec);
+extern WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
+
+typedef void (*WebRtcAecStoreAsComplex)(const float* data,
+                                        float data_complex[2][PART_LEN1]);
+extern WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
+
+typedef void (*WebRtcAecWindowData)(float* x_windowed, const float* x);
+extern WebRtcAecWindowData WebRtcAec_WindowData;
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_mips.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_mips.c
index bb33087aee..035a4b76af 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_mips.c
@@ -20,13 +20,12 @@
 #include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
 
-static const int flagHbandCn = 1; // flag for adding comfort noise in H band
 extern const float WebRtcAec_weightCurve[65];
 extern const float WebRtcAec_overDriveCurve[65];
 
 void WebRtcAec_ComfortNoise_mips(AecCore* aec,
                                  float efw[2][PART_LEN1],
-                                 complex_t* comfortNoiseHband,
+                                 float comfortNoiseHband[2][PART_LEN1],
                                  const float* noisePow,
                                  const float* lambda) {
   int i, num;
@@ -274,7 +273,7 @@ void WebRtcAec_ComfortNoise_mips(AecCore* aec,
   noiseAvg = 0.0;
   tmpAvg = 0.0;
   num = 0;
-  if ((aec->sampFreq == 32000 || aec->sampFreq == 48000) && flagHbandCn == 1) {
+  if (aec->num_bands > 1) {
     for (i = 0; i < PART_LEN; i++) {
       rand[i] = ((float)randW16[i]) / 32768;
     }
@@ -314,27 +313,35 @@ void WebRtcAec_ComfortNoise_mips(AecCore* aec,
 
     for (i = 0; i < PART_LEN1; i++) {
       // Use average NLP weight for H band
-      comfortNoiseHband[i][0] = tmpAvg * u[i][0];
-      comfortNoiseHband[i][1] = tmpAvg * u[i][1];
+      comfortNoiseHband[0][i] = tmpAvg * u[i][0];
+      comfortNoiseHband[1][i] = tmpAvg * u[i][1];
     }
+  } else {
+    memset(comfortNoiseHband, 0,
+           2 * PART_LEN1 * sizeof(comfortNoiseHband[0][0]));
   }
 }
 
-void WebRtcAec_FilterFar_mips(AecCore* aec, float yf[2][PART_LEN1]) {
+void WebRtcAec_FilterFar_mips(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
   int i;
-  for (i = 0; i < aec->num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+  for (i = 0; i < num_partitions; i++) {
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >=  aec->num_partitions) {
-      xPos -=  aec->num_partitions * (PART_LEN1);
+    if (i + x_fft_buf_block_pos >=  num_partitions) {
+      xPos -=  num_partitions * (PART_LEN1);
     }
-    float* yf0 = yf[0];
-    float* yf1 = yf[1];
-    float* aRe = aec->xfBuf[0] + xPos;
-    float* aIm = aec->xfBuf[1] + xPos;
-    float* bRe = aec->wfBuf[0] + pos;
-    float* bIm = aec->wfBuf[1] + pos;
+    float* yf0 = y_fft[0];
+    float* yf1 = y_fft[1];
+    float* aRe = x_fft_buf[0] + xPos;
+    float* aIm = x_fft_buf[1] + xPos;
+    float* bRe = h_fft_buf[0] + pos;
+    float* bIm = h_fft_buf[1] + pos;
     float f0, f1, f2, f3, f4, f5, f6, f7, f8, f9, f10, f11, f12, f13;
     int len = PART_LEN1 >> 1;
 
@@ -432,23 +439,27 @@ void WebRtcAec_FilterFar_mips(AecCore* aec, float yf[2][PART_LEN1]) {
   }
 }
 
-void WebRtcAec_FilterAdaptation_mips(AecCore* aec,
-                                     float* fft,
-                                     float ef[2][PART_LEN1]) {
+void WebRtcAec_FilterAdaptation_mips(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
+  float fft[PART_LEN2];
   int i;
-  for (i = 0; i < aec->num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+  for (i = 0; i < num_partitions; i++) {
+    int xPos = (i + x_fft_buf_block_pos)*(PART_LEN1);
     int pos;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= aec->num_partitions * PART_LEN1;
+    if (i + x_fft_buf_block_pos >= num_partitions) {
+      xPos -= num_partitions * PART_LEN1;
     }
 
     pos = i * PART_LEN1;
-    float* aRe = aec->xfBuf[0] + xPos;
-    float* aIm = aec->xfBuf[1] + xPos;
-    float* bRe = ef[0];
-    float* bIm = ef[1];
+    float* aRe = x_fft_buf[0] + xPos;
+    float* aIm = x_fft_buf[1] + xPos;
+    float* bRe = e_fft[0];
+    float* bIm = e_fft[1];
     float* fft_tmp;
 
     float f0, f1, f2, f3, f4, f5, f6 ,f7, f8, f9, f10, f11, f12;
@@ -573,8 +584,8 @@ void WebRtcAec_FilterAdaptation_mips(AecCore* aec,
       );
     }
     aec_rdft_forward_128(fft);
-    aRe = aec->wfBuf[0] + pos;
-    aIm = aec->wfBuf[1] + pos;
+    aRe = h_fft_buf[0] + pos;
+    aIm = h_fft_buf[1] + pos;
     __asm __volatile (
       ".set     push                                    \n\t"
       ".set     noreorder                               \n\t"
@@ -699,15 +710,18 @@ void WebRtcAec_OverdriveAndSuppress_mips(AecCore* aec,
   }
 }
 
-void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
-  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-  const float error_threshold = aec->extended_filter_enabled
+void WebRtcAec_ScaleErrorSignal_mips(int extended_filter_enabled,
+                                     float normal_mu,
+                                     float normal_error_threshold,
+                                     float x_pow[PART_LEN1],
+                                     float ef[2][PART_LEN1]) {
+  const float mu = extended_filter_enabled ? kExtendedMu : normal_mu;
+  const float error_threshold = extended_filter_enabled
                                     ? kExtendedErrorThreshold
-                                    : aec->normal_error_threshold;
+                                    : normal_error_threshold;
   int len = (PART_LEN1);
   float* ef0 = ef[0];
   float* ef1 = ef[1];
-  float* xPow = aec->xPow;
   float fac1 = 1e-10f;
   float err_th2 = error_threshold * error_threshold;
   float f0, f1, f2;
@@ -719,7 +733,7 @@ void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
     ".set       push                                   \n\t"
     ".set       noreorder                              \n\t"
    "1:                                                 \n\t"
-    "lwc1       %[f0],     0(%[xPow])                  \n\t"
+    "lwc1       %[f0],     0(%[x_pow])                 \n\t"
     "lwc1       %[f1],     0(%[ef0])                   \n\t"
     "lwc1       %[f2],     0(%[ef1])                   \n\t"
     "add.s      %[f0],     %[f0],       %[fac1]        \n\t"
@@ -747,7 +761,7 @@ void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
     "swc1       %[f1],     0(%[ef0])                   \n\t"
     "swc1       %[f2],     0(%[ef1])                   \n\t"
     "addiu      %[len],    %[len],      -1             \n\t"
-    "addiu      %[xPow],   %[xPow],     4              \n\t"
+    "addiu      %[x_pow],  %[x_pow],    4              \n\t"
     "addiu      %[ef0],    %[ef0],      4              \n\t"
     "bgtz       %[len],    1b                          \n\t"
     " addiu     %[ef1],    %[ef1],      4              \n\t"
@@ -756,7 +770,7 @@ void WebRtcAec_ScaleErrorSignal_mips(AecCore* aec, float ef[2][PART_LEN1]) {
 #if !defined(MIPS32_R2_LE)
       [f3] "=&f" (f3),
 #endif
-      [xPow] "+r" (xPow), [ef0] "+r" (ef0), [ef1] "+r" (ef1),
+      [x_pow] "+r" (x_pow), [ef0] "+r" (ef0), [ef1] "+r" (ef1),
       [len] "+r" (len)
     : [fac1] "f" (fac1), [err_th2] "f" (err_th2), [mu] "f" (mu),
       [err_th] "f" (error_threshold)
@@ -771,4 +785,3 @@ void WebRtcAec_InitAec_mips(void) {
   WebRtcAec_ComfortNoise = WebRtcAec_ComfortNoise_mips;
   WebRtcAec_OverdriveAndSuppress = WebRtcAec_OverdriveAndSuppress_mips;
 }
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.c
index f8d0b2419b..7898ab2543 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.c
@@ -34,51 +34,55 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bIm + aIm * bRe;
 }
 
-static void FilterFarNEON(AecCore* aec, float yf[2][PART_LEN1]) {
+static void FilterFarNEON(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
   int i;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
     int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * PART_LEN1;
     }
 
     // vectorized code (four at once)
     for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const float32x4_t xfBuf_re = vld1q_f32(&aec->xfBuf[0][xPos + j]);
-      const float32x4_t xfBuf_im = vld1q_f32(&aec->xfBuf[1][xPos + j]);
-      const float32x4_t wfBuf_re = vld1q_f32(&aec->wfBuf[0][pos + j]);
-      const float32x4_t wfBuf_im = vld1q_f32(&aec->wfBuf[1][pos + j]);
-      const float32x4_t yf_re = vld1q_f32(&yf[0][j]);
-      const float32x4_t yf_im = vld1q_f32(&yf[1][j]);
-      const float32x4_t a = vmulq_f32(xfBuf_re, wfBuf_re);
-      const float32x4_t e = vmlsq_f32(a, xfBuf_im, wfBuf_im);
-      const float32x4_t c = vmulq_f32(xfBuf_re, wfBuf_im);
-      const float32x4_t f = vmlaq_f32(c, xfBuf_im, wfBuf_re);
-      const float32x4_t g = vaddq_f32(yf_re, e);
-      const float32x4_t h = vaddq_f32(yf_im, f);
-      vst1q_f32(&yf[0][j], g);
-      vst1q_f32(&yf[1][j], h);
+      const float32x4_t x_fft_buf_re = vld1q_f32(&x_fft_buf[0][xPos + j]);
+      const float32x4_t x_fft_buf_im = vld1q_f32(&x_fft_buf[1][xPos + j]);
+      const float32x4_t h_fft_buf_re = vld1q_f32(&h_fft_buf[0][pos + j]);
+      const float32x4_t h_fft_buf_im = vld1q_f32(&h_fft_buf[1][pos + j]);
+      const float32x4_t y_fft_re = vld1q_f32(&y_fft[0][j]);
+      const float32x4_t y_fft_im = vld1q_f32(&y_fft[1][j]);
+      const float32x4_t a = vmulq_f32(x_fft_buf_re, h_fft_buf_re);
+      const float32x4_t e = vmlsq_f32(a, x_fft_buf_im, h_fft_buf_im);
+      const float32x4_t c = vmulq_f32(x_fft_buf_re, h_fft_buf_im);
+      const float32x4_t f = vmlaq_f32(c, x_fft_buf_im, h_fft_buf_re);
+      const float32x4_t g = vaddq_f32(y_fft_re, e);
+      const float32x4_t h = vaddq_f32(y_fft_im, f);
+      vst1q_f32(&y_fft[0][j], g);
+      vst1q_f32(&y_fft[1][j], h);
     }
     // scalar code for the remaining items.
     for (; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
+      y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
+      y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
     }
   }
 }
 
 // ARM64's arm_neon.h has already defined vdivq_f32 vsqrtq_f32.
-#if !defined (WEBRTC_ARCH_ARM64_NEON)
+#if !defined (WEBRTC_ARCH_ARM64)
 static float32x4_t vdivq_f32(float32x4_t a, float32x4_t b) {
   int i;
   float32x4_t x = vrecpeq_f32(b);
@@ -120,22 +124,26 @@ static float32x4_t vsqrtq_f32(float32x4_t s) {
   // sqrt(s) = s * 1/sqrt(s)
   return vmulq_f32(s, x);;
 }
-#endif  // WEBRTC_ARCH_ARM64_NEON
+#endif  // WEBRTC_ARCH_ARM64
 
-static void ScaleErrorSignalNEON(AecCore* aec, float ef[2][PART_LEN1]) {
-  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-  const float error_threshold = aec->extended_filter_enabled ?
-      kExtendedErrorThreshold : aec->normal_error_threshold;
+static void ScaleErrorSignalNEON(int extended_filter_enabled,
+                                 float normal_mu,
+                                 float normal_error_threshold,
+                                 float x_pow[PART_LEN1],
+                                 float ef[2][PART_LEN1]) {
+  const float mu = extended_filter_enabled ? kExtendedMu : normal_mu;
+  const float error_threshold = extended_filter_enabled ?
+      kExtendedErrorThreshold : normal_error_threshold;
   const float32x4_t k1e_10f = vdupq_n_f32(1e-10f);
   const float32x4_t kMu = vmovq_n_f32(mu);
   const float32x4_t kThresh = vmovq_n_f32(error_threshold);
   int i;
   // vectorized code (four at once)
   for (i = 0; i + 3 < PART_LEN1; i += 4) {
-    const float32x4_t xPow = vld1q_f32(&aec->xPow[i]);
+    const float32x4_t x_pow_local = vld1q_f32(&x_pow[i]);
     const float32x4_t ef_re_base = vld1q_f32(&ef[0][i]);
     const float32x4_t ef_im_base = vld1q_f32(&ef[1][i]);
-    const float32x4_t xPowPlus = vaddq_f32(xPow, k1e_10f);
+    const float32x4_t xPowPlus = vaddq_f32(x_pow_local, k1e_10f);
     float32x4_t ef_re = vdivq_f32(ef_re_base, xPowPlus);
     float32x4_t ef_im = vdivq_f32(ef_im_base, xPowPlus);
     const float32x4_t ef_re2 = vmulq_f32(ef_re, ef_re);
@@ -162,8 +170,8 @@ static void ScaleErrorSignalNEON(AecCore* aec, float ef[2][PART_LEN1]) {
   // scalar code for the remaining items.
   for (; i < PART_LEN1; i++) {
     float abs_ef;
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
+    ef[0][i] /= (x_pow[i] + 1e-10f);
+    ef[1][i] /= (x_pow[i] + 1e-10f);
     abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
     if (abs_ef > error_threshold) {
@@ -178,34 +186,37 @@ static void ScaleErrorSignalNEON(AecCore* aec, float ef[2][PART_LEN1]) {
   }
 }
 
-static void FilterAdaptationNEON(AecCore* aec,
-                                 float* fft,
-                                 float ef[2][PART_LEN1]) {
+static void FilterAdaptationNEON(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
+  float fft[PART_LEN2];
   int i;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     int j;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * PART_LEN1;
     }
 
     // Process the whole array...
     for (j = 0; j < PART_LEN; j += 4) {
-      // Load xfBuf and ef.
-      const float32x4_t xfBuf_re = vld1q_f32(&aec->xfBuf[0][xPos + j]);
-      const float32x4_t xfBuf_im = vld1q_f32(&aec->xfBuf[1][xPos + j]);
-      const float32x4_t ef_re = vld1q_f32(&ef[0][j]);
-      const float32x4_t ef_im = vld1q_f32(&ef[1][j]);
-      // Calculate the product of conjugate(xfBuf) by ef.
+      // Load x_fft_buf and e_fft.
+      const float32x4_t x_fft_buf_re = vld1q_f32(&x_fft_buf[0][xPos + j]);
+      const float32x4_t x_fft_buf_im = vld1q_f32(&x_fft_buf[1][xPos + j]);
+      const float32x4_t e_fft_re = vld1q_f32(&e_fft[0][j]);
+      const float32x4_t e_fft_im = vld1q_f32(&e_fft[1][j]);
+      // Calculate the product of conjugate(x_fft_buf) by e_fft.
       //   re(conjugate(a) * b) = aRe * bRe + aIm * bIm
       //   im(conjugate(a) * b)=  aRe * bIm - aIm * bRe
-      const float32x4_t a = vmulq_f32(xfBuf_re, ef_re);
-      const float32x4_t e = vmlaq_f32(a, xfBuf_im, ef_im);
-      const float32x4_t c = vmulq_f32(xfBuf_re, ef_im);
-      const float32x4_t f = vmlsq_f32(c, xfBuf_im, ef_re);
+      const float32x4_t a = vmulq_f32(x_fft_buf_re, e_fft_re);
+      const float32x4_t e = vmlaq_f32(a, x_fft_buf_im, e_fft_im);
+      const float32x4_t c = vmulq_f32(x_fft_buf_re, e_fft_im);
+      const float32x4_t f = vmlsq_f32(c, x_fft_buf_im, e_fft_re);
       // Interleave real and imaginary parts.
       const float32x4x2_t g_n_h = vzipq_f32(e, f);
       // Store
@@ -213,10 +224,10 @@ static void FilterAdaptationNEON(AecCore* aec,
       vst1q_f32(&fft[2 * j + 4], g_n_h.val[1]);
     }
     // ... and fixup the first imaginary entry.
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN],
-                   ef[1][PART_LEN]);
+    fft[1] = MulRe(x_fft_buf[0][xPos + PART_LEN],
+                   -x_fft_buf[1][xPos + PART_LEN],
+                   e_fft[0][PART_LEN],
+                   e_fft[1][PART_LEN]);
 
     aec_rdft_inverse_128(fft);
     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
@@ -234,21 +245,21 @@ static void FilterAdaptationNEON(AecCore* aec,
     aec_rdft_forward_128(fft);
 
     {
-      const float wt1 = aec->wfBuf[1][pos];
-      aec->wfBuf[0][pos + PART_LEN] += fft[1];
+      const float wt1 = h_fft_buf[1][pos];
+      h_fft_buf[0][pos + PART_LEN] += fft[1];
       for (j = 0; j < PART_LEN; j += 4) {
-        float32x4_t wtBuf_re = vld1q_f32(&aec->wfBuf[0][pos + j]);
-        float32x4_t wtBuf_im = vld1q_f32(&aec->wfBuf[1][pos + j]);
+        float32x4_t wtBuf_re = vld1q_f32(&h_fft_buf[0][pos + j]);
+        float32x4_t wtBuf_im = vld1q_f32(&h_fft_buf[1][pos + j]);
         const float32x4_t fft0 = vld1q_f32(&fft[2 * j + 0]);
         const float32x4_t fft4 = vld1q_f32(&fft[2 * j + 4]);
         const float32x4x2_t fft_re_im = vuzpq_f32(fft0, fft4);
         wtBuf_re = vaddq_f32(wtBuf_re, fft_re_im.val[0]);
         wtBuf_im = vaddq_f32(wtBuf_im, fft_re_im.val[1]);
 
-        vst1q_f32(&aec->wfBuf[0][pos + j], wtBuf_re);
-        vst1q_f32(&aec->wfBuf[1][pos + j], wtBuf_im);
+        vst1q_f32(&h_fft_buf[0][pos + j], wtBuf_re);
+        vst1q_f32(&h_fft_buf[1][pos + j], wtBuf_im);
       }
-      aec->wfBuf[1][pos] = wt1;
+      h_fft_buf[1][pos] = wt1;
     }
   }
 }
@@ -442,7 +453,7 @@ static void OverdriveAndSuppressNEON(AecCore* aec,
   }
 }
 
-static int PartitionDelay(const AecCore* aec) {
+static int PartitionDelayNEON(const AecCore* aec) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -499,7 +510,8 @@ static int PartitionDelay(const AecCore* aec) {
 static void SmoothedPSD(AecCore* aec,
                         float efw[2][PART_LEN1],
                         float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1]) {
+                        float xfw[2][PART_LEN1],
+                        int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh = aec->extended_filter_enabled
       ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
@@ -615,19 +627,16 @@ static void SmoothedPSD(AecCore* aec,
     seSum += aec->se[i];
   }
 
-  // Divergent filter safeguard.
+  // Divergent filter safeguard update.
   aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
 
-  if (aec->divergeState)
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-
-  // Reset if error is significantly larger than nearend (13 dB).
-  if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+  // Signal extreme filter divergence if the error is significantly larger
+  // than the nearend (13 dB).
+  *extreme_filter_divergence = (seSum > (19.95f * sdSum));
 }
 
 // Window time domain data to be used by the fft.
-__inline static void WindowData(float* x_windowed, const float* x) {
+static void WindowDataNEON(float* x_windowed, const float* x) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const float32x4_t vec_Buf1 = vld1q_f32(&x[i]);
@@ -648,8 +657,8 @@ __inline static void WindowData(float* x_windowed, const float* x) {
 }
 
 // Puts fft output data into a complex valued array.
-__inline static void StoreAsComplex(const float* data,
-                                    float data_complex[2][PART_LEN1]) {
+static void StoreAsComplexNEON(const float* data,
+                               float data_complex[2][PART_LEN1]) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const float32x4x2_t vec_data = vld2q_f32(&data[2 * i]);
@@ -665,32 +674,15 @@ __inline static void StoreAsComplex(const float* data,
 
 static void SubbandCoherenceNEON(AecCore* aec,
                                  float efw[2][PART_LEN1],
+                                 float dfw[2][PART_LEN1],
                                  float xfw[2][PART_LEN1],
                                  float* fft,
                                  float* cohde,
-                                 float* cohxd) {
-  float dfw[2][PART_LEN1];
+                                 float* cohxd,
+                                 int* extreme_filter_divergence) {
   int i;
 
-  if (aec->delayEstCtr == 0)
-    aec->delayIdx = PartitionDelay(aec);
-
-  // Use delayed far.
-  memcpy(xfw,
-         aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  // Windowed near fft
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed error fft
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  SmoothedPSD(aec, efw, dfw, xfw);
+  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
 
   {
     const float32x4_t vec_1eminus10 =  vdupq_n_f32(1e-10f);
@@ -732,5 +724,7 @@ void WebRtcAec_InitAec_neon(void) {
   WebRtcAec_FilterAdaptation = FilterAdaptationNEON;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressNEON;
   WebRtcAec_SubbandCoherence = SubbandCoherenceNEON;
+  WebRtcAec_StoreAsComplex = StoreAsComplexNEON;
+  WebRtcAec_PartitionDelay = PartitionDelayNEON;
+  WebRtcAec_WindowData = WindowDataNEON;
 }
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_sse2.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_sse2.c
index b1bffcbb9f..f897a4c0c7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_sse2.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_sse2.c
@@ -29,67 +29,76 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bIm + aIm * bRe;
 }
 
-static void FilterFarSSE2(AecCore* aec, float yf[2][PART_LEN1]) {
+static void FilterFarSSE2(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]) {
+
   int i;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
     int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+    int xPos = (i + x_fft_buf_block_pos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * (PART_LEN1);
     }
 
     // vectorized code (four at once)
     for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 wfBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-      const __m128 wfBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
-      const __m128 yf_re = _mm_loadu_ps(&yf[0][j]);
-      const __m128 yf_im = _mm_loadu_ps(&yf[1][j]);
-      const __m128 a = _mm_mul_ps(xfBuf_re, wfBuf_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, wfBuf_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, wfBuf_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, wfBuf_re);
+      const __m128 x_fft_buf_re = _mm_loadu_ps(&x_fft_buf[0][xPos + j]);
+      const __m128 x_fft_buf_im = _mm_loadu_ps(&x_fft_buf[1][xPos + j]);
+      const __m128 h_fft_buf_re = _mm_loadu_ps(&h_fft_buf[0][pos + j]);
+      const __m128 h_fft_buf_im = _mm_loadu_ps(&h_fft_buf[1][pos + j]);
+      const __m128 y_fft_re = _mm_loadu_ps(&y_fft[0][j]);
+      const __m128 y_fft_im = _mm_loadu_ps(&y_fft[1][j]);
+      const __m128 a = _mm_mul_ps(x_fft_buf_re, h_fft_buf_re);
+      const __m128 b = _mm_mul_ps(x_fft_buf_im, h_fft_buf_im);
+      const __m128 c = _mm_mul_ps(x_fft_buf_re, h_fft_buf_im);
+      const __m128 d = _mm_mul_ps(x_fft_buf_im, h_fft_buf_re);
       const __m128 e = _mm_sub_ps(a, b);
       const __m128 f = _mm_add_ps(c, d);
-      const __m128 g = _mm_add_ps(yf_re, e);
-      const __m128 h = _mm_add_ps(yf_im, f);
-      _mm_storeu_ps(&yf[0][j], g);
-      _mm_storeu_ps(&yf[1][j], h);
+      const __m128 g = _mm_add_ps(y_fft_re, e);
+      const __m128 h = _mm_add_ps(y_fft_im, f);
+      _mm_storeu_ps(&y_fft[0][j], g);
+      _mm_storeu_ps(&y_fft[1][j], h);
     }
     // scalar code for the remaining items.
     for (; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
-                        aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][pos + j],
-                        aec->wfBuf[1][pos + j]);
+      y_fft[0][j] += MulRe(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
+      y_fft[1][j] += MulIm(x_fft_buf[0][xPos + j],
+                           x_fft_buf[1][xPos + j],
+                           h_fft_buf[0][pos + j],
+                           h_fft_buf[1][pos + j]);
     }
   }
 }
 
-static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1]) {
+static void ScaleErrorSignalSSE2(int extended_filter_enabled,
+                                 float normal_mu,
+                                 float normal_error_threshold,
+                                 float x_pow[PART_LEN1],
+                                 float ef[2][PART_LEN1]) {
   const __m128 k1e_10f = _mm_set1_ps(1e-10f);
-  const __m128 kMu = aec->extended_filter_enabled ? _mm_set1_ps(kExtendedMu)
-                                                  : _mm_set1_ps(aec->normal_mu);
-  const __m128 kThresh = aec->extended_filter_enabled
+  const __m128 kMu = extended_filter_enabled ? _mm_set1_ps(kExtendedMu)
+      : _mm_set1_ps(normal_mu);
+  const __m128 kThresh = extended_filter_enabled
                              ? _mm_set1_ps(kExtendedErrorThreshold)
-                             : _mm_set1_ps(aec->normal_error_threshold);
+                             : _mm_set1_ps(normal_error_threshold);
 
   int i;
   // vectorized code (four at once)
   for (i = 0; i + 3 < PART_LEN1; i += 4) {
-    const __m128 xPow = _mm_loadu_ps(&aec->xPow[i]);
+    const __m128 x_pow_local = _mm_loadu_ps(&x_pow[i]);
     const __m128 ef_re_base = _mm_loadu_ps(&ef[0][i]);
     const __m128 ef_im_base = _mm_loadu_ps(&ef[1][i]);
 
-    const __m128 xPowPlus = _mm_add_ps(xPow, k1e_10f);
+    const __m128 xPowPlus = _mm_add_ps(x_pow_local, k1e_10f);
     __m128 ef_re = _mm_div_ps(ef_re_base, xPowPlus);
     __m128 ef_im = _mm_div_ps(ef_im_base, xPowPlus);
     const __m128 ef_re2 = _mm_mul_ps(ef_re, ef_re);
@@ -116,14 +125,14 @@ static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1]) {
   // scalar code for the remaining items.
   {
     const float mu =
-        aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
-    const float error_threshold = aec->extended_filter_enabled
+        extended_filter_enabled ? kExtendedMu : normal_mu;
+    const float error_threshold = extended_filter_enabled
                                       ? kExtendedErrorThreshold
-                                      : aec->normal_error_threshold;
+                                      : normal_error_threshold;
     for (; i < (PART_LEN1); i++) {
       float abs_ef;
-      ef[0][i] /= (aec->xPow[i] + 1e-10f);
-      ef[1][i] /= (aec->xPow[i] + 1e-10f);
+      ef[0][i] /= (x_pow[i] + 1e-10f);
+      ef[1][i] /= (x_pow[i] + 1e-10f);
       abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
       if (abs_ef > error_threshold) {
@@ -139,33 +148,36 @@ static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1]) {
   }
 }
 
-static void FilterAdaptationSSE2(AecCore* aec,
-                                 float* fft,
-                                 float ef[2][PART_LEN1]) {
+static void FilterAdaptationSSE2(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
+  float fft[PART_LEN2];
   int i, j;
-  const int num_partitions = aec->num_partitions;
   for (i = 0; i < num_partitions; i++) {
-    int xPos = (i + aec->xfBufBlockPos) * (PART_LEN1);
+    int xPos = (i + x_fft_buf_block_pos) * (PART_LEN1);
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= num_partitions) {
+    if (i + x_fft_buf_block_pos >= num_partitions) {
       xPos -= num_partitions * PART_LEN1;
     }
 
     // Process the whole array...
     for (j = 0; j < PART_LEN; j += 4) {
-      // Load xfBuf and ef.
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 ef_re = _mm_loadu_ps(&ef[0][j]);
-      const __m128 ef_im = _mm_loadu_ps(&ef[1][j]);
-      // Calculate the product of conjugate(xfBuf) by ef.
+      // Load x_fft_buf and e_fft.
+      const __m128 x_fft_buf_re = _mm_loadu_ps(&x_fft_buf[0][xPos + j]);
+      const __m128 x_fft_buf_im = _mm_loadu_ps(&x_fft_buf[1][xPos + j]);
+      const __m128 e_fft_re = _mm_loadu_ps(&e_fft[0][j]);
+      const __m128 e_fft_im = _mm_loadu_ps(&e_fft[1][j]);
+      // Calculate the product of conjugate(x_fft_buf) by e_fft.
       //   re(conjugate(a) * b) = aRe * bRe + aIm * bIm
       //   im(conjugate(a) * b)=  aRe * bIm - aIm * bRe
-      const __m128 a = _mm_mul_ps(xfBuf_re, ef_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, ef_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, ef_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, ef_re);
+      const __m128 a = _mm_mul_ps(x_fft_buf_re, e_fft_re);
+      const __m128 b = _mm_mul_ps(x_fft_buf_im, e_fft_im);
+      const __m128 c = _mm_mul_ps(x_fft_buf_re, e_fft_im);
+      const __m128 d = _mm_mul_ps(x_fft_buf_im, e_fft_re);
       const __m128 e = _mm_add_ps(a, b);
       const __m128 f = _mm_sub_ps(c, d);
       // Interleave real and imaginary parts.
@@ -176,10 +188,10 @@ static void FilterAdaptationSSE2(AecCore* aec,
       _mm_storeu_ps(&fft[2 * j + 4], h);
     }
     // ... and fixup the first imaginary entry.
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN],
-                   ef[1][PART_LEN]);
+    fft[1] = MulRe(x_fft_buf[0][xPos + PART_LEN],
+                   -x_fft_buf[1][xPos + PART_LEN],
+                   e_fft[0][PART_LEN],
+                   e_fft[1][PART_LEN]);
 
     aec_rdft_inverse_128(fft);
     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
@@ -197,11 +209,11 @@ static void FilterAdaptationSSE2(AecCore* aec,
     aec_rdft_forward_128(fft);
 
     {
-      float wt1 = aec->wfBuf[1][pos];
-      aec->wfBuf[0][pos + PART_LEN] += fft[1];
+      float wt1 = h_fft_buf[1][pos];
+      h_fft_buf[0][pos + PART_LEN] += fft[1];
       for (j = 0; j < PART_LEN; j += 4) {
-        __m128 wtBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-        __m128 wtBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
+        __m128 wtBuf_re = _mm_loadu_ps(&h_fft_buf[0][pos + j]);
+        __m128 wtBuf_im = _mm_loadu_ps(&h_fft_buf[1][pos + j]);
         const __m128 fft0 = _mm_loadu_ps(&fft[2 * j + 0]);
         const __m128 fft4 = _mm_loadu_ps(&fft[2 * j + 4]);
         const __m128 fft_re =
@@ -210,10 +222,10 @@ static void FilterAdaptationSSE2(AecCore* aec,
             _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(3, 1, 3, 1));
         wtBuf_re = _mm_add_ps(wtBuf_re, fft_re);
         wtBuf_im = _mm_add_ps(wtBuf_im, fft_im);
-        _mm_storeu_ps(&aec->wfBuf[0][pos + j], wtBuf_re);
-        _mm_storeu_ps(&aec->wfBuf[1][pos + j], wtBuf_im);
+        _mm_storeu_ps(&h_fft_buf[0][pos + j], wtBuf_re);
+        _mm_storeu_ps(&h_fft_buf[1][pos + j], wtBuf_im);
       }
-      aec->wfBuf[1][pos] = wt1;
+      h_fft_buf[1][pos] = wt1;
     }
   }
 }
@@ -427,7 +439,8 @@ __inline static void _mm_add_ps_4x1(__m128 sum, float *dst) {
   sum = _mm_add_ps(sum, _mm_shuffle_ps(sum, sum, _MM_SHUFFLE(1, 1, 1, 1)));
   _mm_store_ss(dst, sum);
 }
-static int PartitionDelay(const AecCore* aec) {
+
+static int PartitionDelaySSE2(const AecCore* aec) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -476,7 +489,8 @@ static int PartitionDelay(const AecCore* aec) {
 static void SmoothedPSD(AecCore* aec,
                         float efw[2][PART_LEN1],
                         float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1]) {
+                        float xfw[2][PART_LEN1],
+                        int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh = aec->extended_filter_enabled
       ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
@@ -595,19 +609,16 @@ static void SmoothedPSD(AecCore* aec,
     seSum += aec->se[i];
   }
 
-  // Divergent filter safeguard.
+  // Divergent filter safeguard update.
   aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
 
-  if (aec->divergeState)
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-
-  // Reset if error is significantly larger than nearend (13 dB).
-  if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+  // Signal extreme filter divergence if the error is significantly larger
+  // than the nearend (13 dB).
+  *extreme_filter_divergence = (seSum > (19.95f * sdSum));
 }
 
 // Window time domain data to be used by the fft.
-__inline static void WindowData(float* x_windowed, const float* x) {
+static void WindowDataSSE2(float* x_windowed, const float* x) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const __m128 vec_Buf1 = _mm_loadu_ps(&x[i]);
@@ -627,8 +638,8 @@ __inline static void WindowData(float* x_windowed, const float* x) {
 }
 
 // Puts fft output data into a complex valued array.
-__inline static void StoreAsComplex(const float* data,
-                                    float data_complex[2][PART_LEN1]) {
+static void StoreAsComplexSSE2(const float* data,
+                               float data_complex[2][PART_LEN1]) {
   int i;
   for (i = 0; i < PART_LEN; i += 4) {
     const __m128 vec_fft0 = _mm_loadu_ps(&data[2 * i]);
@@ -649,32 +660,15 @@ __inline static void StoreAsComplex(const float* data,
 
 static void SubbandCoherenceSSE2(AecCore* aec,
                                  float efw[2][PART_LEN1],
+                                 float dfw[2][PART_LEN1],
                                  float xfw[2][PART_LEN1],
                                  float* fft,
                                  float* cohde,
-                                 float* cohxd) {
-  float dfw[2][PART_LEN1];
+                                 float* cohxd,
+                                 int* extreme_filter_divergence) {
   int i;
 
-  if (aec->delayEstCtr == 0)
-    aec->delayIdx = PartitionDelay(aec);
-
-  // Use delayed far.
-  memcpy(xfw,
-         aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  // Windowed near fft
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed error fft
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  SmoothedPSD(aec, efw, dfw, xfw);
+  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
 
   {
     const __m128 vec_1eminus10 =  _mm_set1_ps(1e-10f);
@@ -728,4 +722,7 @@ void WebRtcAec_InitAec_SSE2(void) {
   WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
   WebRtcAec_SubbandCoherence = SubbandCoherenceSSE2;
+  WebRtcAec_StoreAsComplex = StoreAsComplexSSE2;
+  WebRtcAec_PartitionDelay = PartitionDelaySSE2;
+  WebRtcAec_WindowData = WindowDataSSE2;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.c
index 015617c937..03efc103ea 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.c
@@ -23,7 +23,7 @@
 
 #include <math.h>
 
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 // These tables used to be computed at run-time. For example, refer to:
@@ -579,9 +579,9 @@ void aec_rdft_init(void) {
 #if defined(MIPS_FPU_LE)
   aec_rdft_init_mips();
 #endif
-#if defined(WEBRTC_ARCH_ARM_NEON)
+#if defined(WEBRTC_HAS_NEON)
   aec_rdft_init_neon();
-#elif defined(WEBRTC_DETECT_ARM_NEON)
+#elif defined(WEBRTC_DETECT_NEON)
   if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
     aec_rdft_init_neon();
   }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.h
index 22f5dd1e89..18eb7a5c3f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft.h
@@ -54,7 +54,7 @@ void aec_rdft_inverse_128(float* a);
 #if defined(MIPS_FPU_LE)
 void aec_rdft_init_mips(void);
 #endif
-#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
+#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 void aec_rdft_init_neon(void);
 #endif
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.c
index 99c39efa88..01f1e137b6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.c
@@ -102,7 +102,7 @@ void WebRtcAec_ResampleLinear(void* resampInst,
     mm++;
 
     tnew = be * mm + obj->position;
-    tn = (int)tnew;
+    tn = (size_t)tnew;
   }
 
   *size_out = mm;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c
index 69233c877a..d5fa516c88 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -11,7 +11,7 @@
 /*
  * Contains the API functions for the AEC.
  */
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 
 #include <math.h>
 #ifdef WEBRTC_AEC_DEBUG_DUMP
@@ -261,7 +261,7 @@ int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) {
   }
 
   return 0;
-  }
+}
 
 // Returns any error that is caused when buffering the
 // far-end signal.
@@ -296,9 +296,9 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
   int32_t error_code = WebRtcAec_GetBufferFarendError(aecInst, farend,
                                                       nrOfSamples);
 
-  if (error_code != 0)
+  if (error_code != 0) {
     return error_code;
-
+  }
 
   if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
     // Resample and get a new number of samples
@@ -318,7 +318,8 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
   // Write the time-domain data to |far_pre_buf|.
   WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, newNrOfSamples);
 
-  // Transform to frequency domain if we have enough data.
+  // TODO(minyue): reduce to |PART_LEN| samples for each buffering, when
+  // WebRtcAec_BufferFarendPartition() is changed to take |PART_LEN| samples.
   while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
     // We have enough data to pass to the FFT, hence read PART_LEN2 samples.
     {
@@ -326,10 +327,6 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
       float tmp[PART_LEN2];
       WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**)&ptmp, tmp, PART_LEN2);
       WebRtcAec_BufferFarendPartition(aecpc->aec, ptmp);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-      WebRtc_WriteBuffer(
-          WebRtcAec_far_time_buf(aecpc->aec), &ptmp[PART_LEN], 1);
-#endif
     }
 
     // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
@@ -710,7 +707,7 @@ static int ProcessNormal(Aec* aecpc,
     }
   } else {
     // AEC is enabled.
-      EstBufDelayNormal(aecpc);
+    EstBufDelayNormal(aecpc);
 
     // Call the AEC.
     // TODO(bjornv): Re-structure such that we don't have to pass
@@ -968,3 +965,4 @@ OpenDebugFiles(Aec* aecpc,
 }
 
 #endif
+
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/include/echo_cancellation.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.h
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.h
index 235323c129..de84b2e6d1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
 
 #include <stddef.h>
 
@@ -91,7 +91,7 @@ void WebRtcAec_Free(void* aecInst);
  *
  * Outputs                      Description
  * -------------------------------------------------------------------
- * int32_t      return          0: OK
+ * int32_t        return        0: OK
  *                             -1: error
  */
 int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq);
@@ -176,7 +176,7 @@ int32_t WebRtcAec_Process(void* aecInst,
  *
  * Outputs                      Description
  * -------------------------------------------------------------------
- * int            return         0: OK
+ * int            return        0: OK
  *                              12000-12050: error code
  */
 int WebRtcAec_set_config(void* handle, AecConfig config);
@@ -192,7 +192,7 @@ int WebRtcAec_set_config(void* handle, AecConfig config);
  * -------------------------------------------------------------------
  * int*           status        0: Almost certainly nearend single-talk
  *                              1: Might not be neared single-talk
- * int            return         0: OK
+ * int            return        0: OK
  *                              12000-12050: error code
  */
 int WebRtcAec_get_echo_status(void* handle, int* status);
@@ -208,7 +208,7 @@ int WebRtcAec_get_echo_status(void* handle, int* status);
  * -------------------------------------------------------------------
  * AecMetrics*    metrics       Struct which will be filled out with the
  *                              current echo metrics.
- * int            return         0: OK
+ * int            return        0: OK
  *                              12000-12050: error code
  */
 int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics);
@@ -227,7 +227,7 @@ int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics);
  * float*  fraction_poor_delays Fraction of the delay estimates that may
  *                              cause the AEC to perform poorly.
  *
- * int     return                0: OK
+ * int            return        0: OK
  *                              12000-12050: error code
  */
 int WebRtcAec_GetDelayMetrics(void* handle,
@@ -248,4 +248,4 @@ struct AecCore* WebRtcAec_aec_core(void* handle);
 #ifdef __cplusplus
 }
 #endif
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
index e87219f33d..95a6cf3324 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
@@ -57,6 +57,8 @@ typedef struct {
 
   RingBuffer* far_pre_buf;  // Time domain far-end pre-buffer.
 
+  int lastError;
+
   int farend_started;
 
   AecCore* aec;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
index 469efd3247..42db082ff9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
@@ -10,7 +10,7 @@
 
 // TODO(bjornv): Make this a comprehensive test.
 
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 
 #include <stdlib.h>
 #include <time.h>
@@ -20,22 +20,20 @@ extern "C" {
 }
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
-TEST(EchoCancellationTest, CreateAndFreeHandlesErrors) {
-  EXPECT_EQ(-1, WebRtcAec_Create(NULL));
-  void* handle = NULL;
-  ASSERT_EQ(0, WebRtcAec_Create(&handle));
-  EXPECT_TRUE(handle != NULL);
-  EXPECT_EQ(-1, WebRtcAec_Free(NULL));
-  EXPECT_EQ(0, WebRtcAec_Free(handle));
+TEST(EchoCancellationTest, CreateAndFreeHasExpectedBehavior) {
+  void* handle = WebRtcAec_Create();
+  ASSERT_TRUE(handle);
+  WebRtcAec_Free(nullptr);
+  WebRtcAec_Free(handle);
 }
 
 TEST(EchoCancellationTest, ApplyAecCoreHandle) {
-  void* handle = NULL;
-  ASSERT_EQ(0, WebRtcAec_Create(&handle));
-  EXPECT_TRUE(handle != NULL);
+  void* handle = WebRtcAec_Create();
+  ASSERT_TRUE(handle);
   EXPECT_TRUE(WebRtcAec_aec_core(NULL) == NULL);
   AecCore* aec_core = WebRtcAec_aec_core(handle);
   EXPECT_TRUE(aec_core != NULL);
@@ -44,7 +42,7 @@ TEST(EchoCancellationTest, ApplyAecCoreHandle) {
   int delay = 111;
   WebRtcAec_SetSystemDelay(aec_core, delay);
   EXPECT_EQ(delay, WebRtcAec_system_delay(aec_core));
-  EXPECT_EQ(0, WebRtcAec_Free(handle));
+  WebRtcAec_Free(handle);
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index 654ae54c7d..567118d828 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -13,8 +13,7 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/typedefs.h"
 
 namespace {
@@ -33,18 +32,18 @@ class SystemDelayTest : public ::testing::Test {
   void RenderAndCapture(int device_buffer_ms);
 
   // Fills up the far-end buffer with respect to the default device buffer size.
-  int BufferFillUp();
+  size_t BufferFillUp();
 
   // Runs and verifies the behavior in a stable startup procedure.
   void RunStableStartup();
 
   // Maps buffer size in ms into samples, taking the unprocessed frame into
   // account.
-  int MapBufferSizeToSamples(int size_in_ms);
+  int MapBufferSizeToSamples(int size_in_ms, bool extended_filter);
 
   void* handle_;
   Aec* self_;
-  int samples_per_frame_;
+  size_t samples_per_frame_;
   // Dummy input/output speech data.
   static const int kSamplesPerChunk = 160;
   float far_[kSamplesPerChunk];
@@ -67,13 +66,14 @@ SystemDelayTest::SystemDelayTest()
 }
 
 void SystemDelayTest::SetUp() {
-  ASSERT_EQ(0, WebRtcAec_Create(&handle_));
+  handle_ = WebRtcAec_Create();
+  ASSERT_TRUE(handle_);
   self_ = reinterpret_cast<Aec*>(handle_);
 }
 
 void SystemDelayTest::TearDown() {
   // Free AEC
-  ASSERT_EQ(0, WebRtcAec_Free(handle_));
+  WebRtcAec_Free(handle_);
   handle_ = NULL;
 }
 
@@ -98,9 +98,10 @@ static const int kMaxConvergenceMs = 500;
 void SystemDelayTest::Init(int sample_rate_hz) {
   // Initialize AEC
   EXPECT_EQ(0, WebRtcAec_Init(handle_, sample_rate_hz, 48000));
+  EXPECT_EQ(0, WebRtcAec_system_delay(self_->aec));
 
   // One frame equals 10 ms of data.
-  samples_per_frame_ = sample_rate_hz / 100;
+  samples_per_frame_ = static_cast<size_t>(sample_rate_hz / 100);
 }
 
 void SystemDelayTest::RenderAndCapture(int device_buffer_ms) {
@@ -115,15 +116,16 @@ void SystemDelayTest::RenderAndCapture(int device_buffer_ms) {
                               0));
 }
 
-int SystemDelayTest::BufferFillUp() {
+size_t SystemDelayTest::BufferFillUp() {
   // To make sure we have a full buffer when we verify stability we first fill
   // up the far-end buffer with the same amount as we will report in through
   // Process().
-  int buffer_size = 0;
+  size_t buffer_size = 0;
   for (int i = 0; i < kDeviceBufMs / 10; i++) {
     EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
     buffer_size += samples_per_frame_;
-    EXPECT_EQ(buffer_size, WebRtcAec_system_delay(self_->aec));
+    EXPECT_EQ(static_cast<int>(buffer_size),
+              WebRtcAec_system_delay(self_->aec));
   }
   return buffer_size;
 }
@@ -132,27 +134,41 @@ void SystemDelayTest::RunStableStartup() {
   // To make sure we have a full buffer when we verify stability we first fill
   // up the far-end buffer with the same amount as we will report in through
   // Process().
-  int buffer_size = BufferFillUp();
-  // A stable device should be accepted and put in a regular process mode within
-  // |kStableConvergenceMs|.
-  int process_time_ms = 0;
-  for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
+  size_t buffer_size = BufferFillUp();
+
+  if (WebRtcAec_delay_agnostic_enabled(self_->aec) == 1) {
+    // In extended_filter mode we set the buffer size after the first processed
+    // 10 ms chunk. Hence, we don't need to wait for the reported system delay
+    // values to become stable.
     RenderAndCapture(kDeviceBufMs);
     buffer_size += samples_per_frame_;
-    if (self_->startup_phase == 0) {
-      // We have left the startup phase.
-      break;
+    EXPECT_EQ(0, self_->startup_phase);
+  } else {
+    // A stable device should be accepted and put in a regular process mode
+    // within |kStableConvergenceMs|.
+    int process_time_ms = 0;
+    for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
+      RenderAndCapture(kDeviceBufMs);
+      buffer_size += samples_per_frame_;
+      if (self_->startup_phase == 0) {
+        // We have left the startup phase.
+        break;
+      }
     }
+    // Verify convergence time.
+    EXPECT_GT(kStableConvergenceMs, process_time_ms);
   }
-  // Verify convergence time.
-  EXPECT_GT(kStableConvergenceMs, process_time_ms);
   // Verify that the buffer has been flushed.
-  EXPECT_GE(buffer_size, WebRtcAec_system_delay(self_->aec));
+  EXPECT_GE(static_cast<int>(buffer_size),
+            WebRtcAec_system_delay(self_->aec));
 }
 
-int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms) {
-  // The extra 10 ms corresponds to the unprocessed frame.
-  return (size_in_ms + 10) * samples_per_frame_ / 10;
+  int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms,
+                                              bool extended_filter) {
+  // If extended_filter is disabled we add an extra 10 ms for the unprocessed
+  // frame. That is simply how the algorithm is constructed.
+  return static_cast<int>(
+      (size_in_ms + (extended_filter ? 0 : 10)) * samples_per_frame_ / 10);
 }
 
 // The tests should meet basic requirements and not be adjusted to what is
@@ -179,14 +195,24 @@ int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms) {
 TEST_F(SystemDelayTest, CorrectIncreaseWhenBufferFarend) {
   // When we add data to the AEC buffer the internal system delay should be
   // incremented with the same amount as the size of data.
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-
-    // Loop through a couple of calls to make sure the system delay increments
-    // correctly.
-    for (int j = 1; j <= 5; j++) {
-      EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
-      EXPECT_EQ(j * samples_per_frame_, WebRtcAec_system_delay(self_->aec));
+  // This process should be independent of DA-AEC and extended_filter mode.
+  for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
+    WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+    EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
+    for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+      WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+      EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+      for (size_t i = 0; i < kNumSampleRates; i++) {
+        Init(kSampleRateHz[i]);
+        // Loop through a couple of calls to make sure the system delay
+        // increments correctly.
+        for (int j = 1; j <= 5; j++) {
+          EXPECT_EQ(0,
+                    WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
+          EXPECT_EQ(static_cast<int>(j * samples_per_frame_),
+                    WebRtcAec_system_delay(self_->aec));
+        }
+      }
     }
   }
 }
@@ -197,21 +223,43 @@ TEST_F(SystemDelayTest, CorrectIncreaseWhenBufferFarend) {
 TEST_F(SystemDelayTest, CorrectDelayAfterStableStartup) {
   // We run the system in a stable startup. After that we verify that the system
   // delay meets the requirements.
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-    RunStableStartup();
+  // This process should be independent of DA-AEC and extended_filter mode.
+  for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
+    WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+    EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
+    for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+      WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+      EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+      for (size_t i = 0; i < kNumSampleRates; i++) {
+        Init(kSampleRateHz[i]);
+        RunStableStartup();
 
-    // Verify system delay with respect to requirements, i.e., the
-    // |system_delay| is in the interval [75%, 100%] of what's reported on the
-    // average.
-    int average_reported_delay = kDeviceBufMs * samples_per_frame_ / 10;
-    EXPECT_GE(average_reported_delay, WebRtcAec_system_delay(self_->aec));
-    EXPECT_LE(average_reported_delay * 3 / 4,
-              WebRtcAec_system_delay(self_->aec));
+        // Verify system delay with respect to requirements, i.e., the
+        // |system_delay| is in the interval [75%, 100%] of what's reported on
+        // the average.
+        // In extended_filter mode we target 50% and measure after one processed
+        // 10 ms chunk.
+        int average_reported_delay =
+            static_cast<int>(kDeviceBufMs * samples_per_frame_ / 10);
+        EXPECT_GE(average_reported_delay, WebRtcAec_system_delay(self_->aec));
+        int lower_bound = WebRtcAec_extended_filter_enabled(self_->aec)
+                              ? average_reported_delay / 2 - samples_per_frame_
+                              : average_reported_delay * 3 / 4;
+        EXPECT_LE(lower_bound, WebRtcAec_system_delay(self_->aec));
+      }
+    }
   }
 }
 
 TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) {
+  // This test does not apply in extended_filter mode, since we only use the
+  // the first 10 ms chunk to determine a reasonable buffer size. Neither does
+  // it apply if DA-AEC is on because that overrides the startup procedure.
+  WebRtcAec_enable_extended_filter(self_->aec, 0);
+  EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(self_->aec));
+  WebRtcAec_enable_delay_agnostic(self_->aec, 0);
+  EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(self_->aec));
+
   // In an unstable system we would start processing after |kMaxConvergenceMs|.
   // On the last frame the AEC buffer is adjusted to 60% of the last reported
   // device buffer size.
@@ -223,7 +271,7 @@ TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) {
     // To make sure we have a full buffer when we verify stability we first fill
     // up the far-end buffer with the same amount as we will report in on the
     // average through Process().
-    int buffer_size = BufferFillUp();
+    size_t buffer_size = BufferFillUp();
 
     int buffer_offset_ms = 25;
     int reported_delay_ms = 0;
@@ -241,26 +289,32 @@ TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) {
     // Verify convergence time.
     EXPECT_GE(kMaxConvergenceMs, process_time_ms);
     // Verify that the buffer has been flushed.
-    EXPECT_GE(buffer_size, WebRtcAec_system_delay(self_->aec));
+    EXPECT_GE(static_cast<int>(buffer_size),
+              WebRtcAec_system_delay(self_->aec));
 
     // Verify system delay with respect to requirements, i.e., the
     // |system_delay| is in the interval [60%, 100%] of what's last reported.
-    EXPECT_GE(reported_delay_ms * samples_per_frame_ / 10,
-              WebRtcAec_system_delay(self_->aec));
-    EXPECT_LE(reported_delay_ms * samples_per_frame_ / 10 * 3 / 5,
+    EXPECT_GE(static_cast<int>(reported_delay_ms * samples_per_frame_ / 10),
               WebRtcAec_system_delay(self_->aec));
+    EXPECT_LE(
+        static_cast<int>(reported_delay_ms * samples_per_frame_ / 10 * 3 / 5),
+        WebRtcAec_system_delay(self_->aec));
   }
 }
 
-TEST_F(SystemDelayTest,
-       DISABLED_ON_ANDROID(CorrectDelayAfterStableBufferBuildUp)) {
+TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) {
+  // This test does not apply in extended_filter mode, since we only use the
+  // the first 10 ms chunk to determine a reasonable buffer size. Neither does
+  // it apply if DA-AEC is on because that overrides the startup procedure.
+  WebRtcAec_enable_extended_filter(self_->aec, 0);
+  EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(self_->aec));
+  WebRtcAec_enable_delay_agnostic(self_->aec, 0);
+  EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(self_->aec));
+
   // In this test we start by establishing the device buffer size during stable
   // conditions, but with an empty internal far-end buffer. Once that is done we
   // verify that the system delay is increased correctly until we have reach an
   // internal buffer size of 75% of what's been reported.
-
-  // This test assumes the reported delays are used.
-  WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(handle_), 1);
   for (size_t i = 0; i < kNumSampleRates; i++) {
     Init(kSampleRateHz[i]);
 
@@ -283,8 +337,8 @@ TEST_F(SystemDelayTest,
 
     // We now have established the required buffer size. Let us verify that we
     // fill up before leaving the startup phase for normal processing.
-    int buffer_size = 0;
-    int target_buffer_size = kDeviceBufMs * samples_per_frame_ / 10 * 3 / 4;
+    size_t buffer_size = 0;
+    size_t target_buffer_size = kDeviceBufMs * samples_per_frame_ / 10 * 3 / 4;
     process_time_ms = 0;
     for (; process_time_ms <= kMaxConvergenceMs; process_time_ms += 10) {
       RenderAndCapture(kDeviceBufMs);
@@ -297,7 +351,8 @@ TEST_F(SystemDelayTest,
     // Verify convergence time.
     EXPECT_GT(kMaxConvergenceMs, process_time_ms);
     // Verify that the buffer has reached the desired size.
-    EXPECT_LE(target_buffer_size, WebRtcAec_system_delay(self_->aec));
+    EXPECT_LE(static_cast<int>(target_buffer_size),
+              WebRtcAec_system_delay(self_->aec));
 
     // Verify normal behavior (system delay is kept constant) after startup by
     // running a couple of calls to BufferFarend() and Process().
@@ -314,62 +369,73 @@ TEST_F(SystemDelayTest, CorrectDelayWhenBufferUnderrun) {
   // WebRtcAec_Process() we will finally run out of data, but should
   // automatically stuff the buffer. We verify this behavior by checking if the
   // system delay goes negative.
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-    RunStableStartup();
+  // This process should be independent of DA-AEC and extended_filter mode.
+  for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
+    WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+    EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
+    for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+      WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+      EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+      for (size_t i = 0; i < kNumSampleRates; i++) {
+        Init(kSampleRateHz[i]);
+        RunStableStartup();
 
-    // The AEC has now left the Startup phase. We now have at most
-    // |kStableConvergenceMs| in the buffer. Keep on calling Process() until
-    // we run out of data and verify that the system delay is non-negative.
-    for (int j = 0; j <= kStableConvergenceMs; j += 10) {
-      EXPECT_EQ(0,
-                WebRtcAec_Process(handle_,
-                                  &near_ptr_,
-                                  1,
-                                  &out_ptr_,
-                                  samples_per_frame_,
-                                  kDeviceBufMs,
-                                  0));
-      EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
+        // The AEC has now left the Startup phase. We now have at most
+        // |kStableConvergenceMs| in the buffer. Keep on calling Process() until
+        // we run out of data and verify that the system delay is non-negative.
+        for (int j = 0; j <= kStableConvergenceMs; j += 10) {
+          EXPECT_EQ(0, WebRtcAec_Process(handle_, &near_ptr_, 1, &out_ptr_,
+                                         samples_per_frame_, kDeviceBufMs, 0));
+          EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
+        }
+      }
     }
   }
 }
 
-TEST_F(SystemDelayTest, DISABLED_ON_ANDROID(CorrectDelayDuringDrift)) {
+TEST_F(SystemDelayTest, CorrectDelayDuringDrift) {
   // This drift test should verify that the system delay is never exceeding the
   // device buffer. The drift is simulated by decreasing the reported device
   // buffer size by 1 ms every 100 ms. If the device buffer size goes below 30
   // ms we jump (add) 10 ms to give a repeated pattern.
 
-  // This test assumes the reported delays are used.
-  WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(handle_), 1);
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-    RunStableStartup();
+  // This process should be independent of DA-AEC and extended_filter mode.
+  for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
+    WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+    EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
+    for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+      WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+      EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+      for (size_t i = 0; i < kNumSampleRates; i++) {
+        Init(kSampleRateHz[i]);
+        RunStableStartup();
 
-    // We have now left the startup phase and proceed with normal processing.
-    int jump = 0;
-    for (int j = 0; j < 1000; j++) {
-      // Drift = -1 ms per 100 ms of data.
-      int device_buf_ms = kDeviceBufMs - (j / 10) + jump;
-      int device_buf = MapBufferSizeToSamples(device_buf_ms);
+        // We have left the startup phase and proceed with normal processing.
+        int jump = 0;
+        for (int j = 0; j < 1000; j++) {
+          // Drift = -1 ms per 100 ms of data.
+          int device_buf_ms = kDeviceBufMs - (j / 10) + jump;
+          int device_buf = MapBufferSizeToSamples(device_buf_ms,
+                                                  extended_filter == 1);
 
-      if (device_buf_ms < 30) {
-        // Add 10 ms data, taking affect next frame.
-        jump += 10;
+          if (device_buf_ms < 30) {
+            // Add 10 ms data, taking affect next frame.
+            jump += 10;
+          }
+          RenderAndCapture(device_buf_ms);
+
+          // Verify that the system delay does not exceed the device buffer.
+          EXPECT_GE(device_buf, WebRtcAec_system_delay(self_->aec));
+
+          // Verify that the system delay is non-negative.
+          EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
+        }
       }
-      RenderAndCapture(device_buf_ms);
-
-      // Verify that the system delay does not exceed the device buffer.
-      EXPECT_GE(device_buf, WebRtcAec_system_delay(self_->aec));
-
-      // Verify that the system delay is non-negative.
-      EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
     }
   }
 }
 
-TEST_F(SystemDelayTest, DISABLED_ON_ANDROID(ShouldRecoverAfterGlitch)) {
+TEST_F(SystemDelayTest, ShouldRecoverAfterGlitch) {
   // This glitch test should verify that the system delay recovers if there is
   // a glitch in data. The data glitch is constructed as 200 ms of buffering
   // after which the stable procedure continues. The glitch is never reported by
@@ -377,79 +443,100 @@ TEST_F(SystemDelayTest, DISABLED_ON_ANDROID(ShouldRecoverAfterGlitch)) {
   // The system is said to be in a non-causal state if the difference between
   // the device buffer and system delay is less than a block (64 samples).
 
-  // This test assumes the reported delays are used.
-  WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(handle_), 1);
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-    RunStableStartup();
-    int device_buf = MapBufferSizeToSamples(kDeviceBufMs);
-    // Glitch state.
-    for (int j = 0; j < 20; j++) {
-      EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
-      // No need to verify system delay, since that is done in a separate test.
-    }
-    // Verify that we are in a non-causal state, i.e.,
-    // |system_delay| > |device_buf|.
-    EXPECT_LT(device_buf, WebRtcAec_system_delay(self_->aec));
-
-    // Recover state. Should recover at least 4 ms of data per 10 ms, hence a
-    // glitch of 200 ms will take at most 200 * 10 / 4 = 500 ms to recover from.
-    bool non_causal = true;  // We are currently in a non-causal state.
-    for (int j = 0; j < 50; j++) {
-      int system_delay_before = WebRtcAec_system_delay(self_->aec);
-      RenderAndCapture(kDeviceBufMs);
-      int system_delay_after = WebRtcAec_system_delay(self_->aec);
-
-      // We have recovered if |device_buf| - |system_delay_after| >= 64 (one
-      // block). During recovery |system_delay_after| < |system_delay_before|,
-      // otherwise they are equal.
-      if (non_causal) {
-        EXPECT_LT(system_delay_after, system_delay_before);
-        if (device_buf - system_delay_after >= 64) {
-          non_causal = false;
+  // This process should be independent of DA-AEC and extended_filter mode.
+  for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
+    WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+    EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
+    for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+      WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+      EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+      for (size_t i = 0; i < kNumSampleRates; i++) {
+        Init(kSampleRateHz[i]);
+        RunStableStartup();
+        int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
+                                                extended_filter == 1);
+        // Glitch state.
+        for (int j = 0; j < 20; j++) {
+          EXPECT_EQ(0,
+                    WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
+          // No need to verify system delay, since that is done in a separate
+          // test.
         }
-      } else {
-        EXPECT_EQ(system_delay_before, system_delay_after);
+        // Verify that we are in a non-causal state, i.e.,
+        // |system_delay| > |device_buf|.
+        EXPECT_LT(device_buf, WebRtcAec_system_delay(self_->aec));
+
+        // Recover state. Should recover at least 4 ms of data per 10 ms, hence
+        // a glitch of 200 ms will take at most 200 * 10 / 4 = 500 ms to recover
+        // from.
+        bool non_causal = true;  // We are currently in a non-causal state.
+        for (int j = 0; j < 50; j++) {
+          int system_delay_before = WebRtcAec_system_delay(self_->aec);
+          RenderAndCapture(kDeviceBufMs);
+          int system_delay_after = WebRtcAec_system_delay(self_->aec);
+          // We have recovered if
+          // |device_buf| - |system_delay_after| >= PART_LEN (1 block).
+          // During recovery, |system_delay_after| < |system_delay_before|,
+          // otherwise they are equal.
+          if (non_causal) {
+            EXPECT_LT(system_delay_after, system_delay_before);
+            if (device_buf - system_delay_after >= PART_LEN) {
+              non_causal = false;
+            }
+          } else {
+            EXPECT_EQ(system_delay_before, system_delay_after);
+          }
+          // Verify that the system delay is non-negative.
+          EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
+        }
+        // Check that we have recovered.
+        EXPECT_FALSE(non_causal);
       }
-      // Verify that the system delay is non-negative.
-      EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
     }
-    // Check that we have recovered.
-    EXPECT_FALSE(non_causal);
   }
 }
 
 TEST_F(SystemDelayTest, UnaffectedWhenSpuriousDeviceBufferValues) {
-  // This spurious device buffer data test aims at verifying that the system
-  // delay is unaffected by large outliers.
-  // The system is said to be in a non-causal state if the difference between
-  // the device buffer and system delay is less than a block (64 samples).
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-    RunStableStartup();
-    int device_buf = MapBufferSizeToSamples(kDeviceBufMs);
+  // This test does not apply in extended_filter mode, since we only use the
+  // the first 10 ms chunk to determine a reasonable buffer size.
+  const int extended_filter = 0;
+  WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+  EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
 
-    // Normal state. We are currently not in a non-causal state.
-    bool non_causal = false;
+  // Should be DA-AEC independent.
+  for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+    WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+    EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+    // This spurious device buffer data test aims at verifying that the system
+    // delay is unaffected by large outliers.
+    // The system is said to be in a non-causal state if the difference between
+    // the device buffer and system delay is less than a block (64 samples).
+    for (size_t i = 0; i < kNumSampleRates; i++) {
+      Init(kSampleRateHz[i]);
+      RunStableStartup();
+      int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
+                                              extended_filter == 1);
 
-    // Run 1 s and replace device buffer size with 500 ms every 100 ms.
-    for (int j = 0; j < 100; j++) {
-      int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
-      int device_buf_ms = kDeviceBufMs;
-      if (j % 10 == 0) {
-        device_buf_ms = 500;
+      // Normal state. We are currently not in a non-causal state.
+      bool non_causal = false;
+
+      // Run 1 s and replace device buffer size with 500 ms every 100 ms.
+      for (int j = 0; j < 100; j++) {
+        int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
+        int device_buf_ms = j % 10 == 0 ? 500 : kDeviceBufMs;
+        RenderAndCapture(device_buf_ms);
+
+        // Check for non-causality.
+        if (device_buf - WebRtcAec_system_delay(self_->aec) < PART_LEN) {
+          non_causal = true;
+        }
+        EXPECT_FALSE(non_causal);
+        EXPECT_EQ(system_delay_before_calls,
+                  WebRtcAec_system_delay(self_->aec));
+
+        // Verify that the system delay is non-negative.
+        EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
       }
-      RenderAndCapture(device_buf_ms);
-
-      // Check for non-causality.
-      if (device_buf - WebRtcAec_system_delay(self_->aec) < 64) {
-        non_causal = true;
-      }
-      EXPECT_FALSE(non_causal);
-      EXPECT_EQ(system_delay_before_calls, WebRtcAec_system_delay(self_->aec));
-
-      // Verify that the system delay is non-negative.
-      EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
     }
   }
 }
@@ -460,36 +547,54 @@ TEST_F(SystemDelayTest, CorrectImpactWhenTogglingDeviceBufferValues) {
   // The test is constructed such that every other device buffer value is zero
   // and then 2 * |kDeviceBufMs|, hence the size is constant on the average. The
   // zero values will force us into a non-causal state and thereby lowering the
-  // system delay until we basically runs out of data. Once that happens the
+  // system delay until we basically run out of data. Once that happens the
   // buffer will be stuffed.
   // TODO(bjornv): This test will have a better impact if we verified that the
-  // delay estimate goes up when the system delay goes done to meet the average
+  // delay estimate goes up when the system delay goes down to meet the average
   // device buffer size.
-  for (size_t i = 0; i < kNumSampleRates; i++) {
-    Init(kSampleRateHz[i]);
-    RunStableStartup();
-    int device_buf = MapBufferSizeToSamples(kDeviceBufMs);
 
-    // Normal state. We are currently not in a non-causal state.
-    bool non_causal = false;
+  // This test does not apply if DA-AEC is enabled and extended_filter mode
+  // disabled.
+  for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
+    WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
+    EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
+    for (int da_aec = 0; da_aec <= 1; ++da_aec) {
+      WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
+      EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
+      if (extended_filter == 0 && da_aec == 1) {
+        continue;
+      }
+      for (size_t i = 0; i < kNumSampleRates; i++) {
+        Init(kSampleRateHz[i]);
+        RunStableStartup();
+        const int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
+                                                      extended_filter == 1);
 
-    // Loop through 100 frames (both render and capture), which equals 1 s of
-    // data. Every odd frame we set the device buffer size to 2 * |kDeviceBufMs|
-    // and even frames we set the device buffer size to zero.
-    for (int j = 0; j < 100; j++) {
-      int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
-      int device_buf_ms = 2 * (j % 2) * kDeviceBufMs;
-      RenderAndCapture(device_buf_ms);
+        // Normal state. We are currently not in a non-causal state.
+        bool non_causal = false;
 
-      // Check for non-causality, compared with the average device buffer size.
-      non_causal |= (device_buf - WebRtcAec_system_delay(self_->aec) < 64);
-      EXPECT_GE(system_delay_before_calls, WebRtcAec_system_delay(self_->aec));
+        // Loop through 100 frames (both render and capture), which equals 1 s
+        // of data. Every odd frame we set the device buffer size to
+        // 2 * |kDeviceBufMs| and even frames we set the device buffer size to
+        // zero.
+        for (int j = 0; j < 100; j++) {
+          int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
+          int device_buf_ms = 2 * (j % 2) * kDeviceBufMs;
+          RenderAndCapture(device_buf_ms);
 
-      // Verify that the system delay is non-negative.
-      EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
+          // Check for non-causality, compared with the average device buffer
+          // size.
+          non_causal |= (device_buf - WebRtcAec_system_delay(self_->aec) < 64);
+          EXPECT_GE(system_delay_before_calls,
+                    WebRtcAec_system_delay(self_->aec));
+
+          // Verify that the system delay is non-negative.
+          EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
+        }
+        // Verify we are not in a non-causal state.
+        EXPECT_FALSE(non_causal);
+      }
     }
-    // Verify we are not in a non-causal state.
-    EXPECT_FALSE(non_causal);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.c
index e35fad43cd..6bf1cf7f3e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -16,10 +16,10 @@
 
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 #ifdef AEC_DEBUG
@@ -361,8 +361,7 @@ static void ResetAdaptiveChannelC(AecmCore* aecm) {
 }
 
 // Initialize function pointers for ARM Neon platform.
-#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON || \
-     defined WEBRTC_ARCH_ARM64_NEON)
+#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
 static void WebRtcAecm_InitNeon(void)
 {
   WebRtcAecm_StoreAdaptiveChannel = WebRtcAecm_StoreAdaptiveChannelNeon;
@@ -509,13 +508,13 @@ int WebRtcAecm_InitCore(AecmCore* const aecm, int samplingFreq) {
     WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
     WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
 
-#ifdef WEBRTC_DETECT_ARM_NEON
+#ifdef WEBRTC_DETECT_NEON
     uint64_t features = WebRtc_GetCPUFeaturesARM();
     if ((features & kCPUFeatureNEON) != 0)
     {
       WebRtcAecm_InitNeon();
     }
-#elif defined(WEBRTC_ARCH_ARM_NEON) || defined(WEBRTC_ARCH_ARM64_NEON)
+#elif defined(WEBRTC_HAS_NEON)
     WebRtcAecm_InitNeon();
 #endif
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.h
index d679c92145..b52bb62d2d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.h
@@ -400,8 +400,7 @@ extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
 // For the above function pointers, functions for generic platforms are declared
 // and defined as static in file aecm_core.c, while those for ARM Neon platforms
 // are declared below and defined in file aecm_core_neon.c.
-#if (defined WEBRTC_DETECT_ARM_NEON) || defined (WEBRTC_ARCH_ARM_NEON) || \
-     defined (WEBRTC_ARCH_ARM64_NEON)
+#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
                                        const uint16_t* far_spectrum,
                                        int32_t* echo_est,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.c
index 3a06bb6690..3a8fafa4ec 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.c
@@ -16,14 +16,14 @@
 
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
 // Square root of Hanning window in Q14.
-#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
+#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
 // Table is defined in an ARM assembly file.
 extern const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END;
 #else
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
index 3c2343a892..3ca9982ebf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
@@ -12,7 +12,7 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
 
 static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
index 40c145a17e..1751fcf7ad 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
@@ -32,7 +32,7 @@ const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
 };
 
 static inline void AddLanes(uint32_t* ptr, uint32x4_t v) {
-#if defined(__aarch64__)
+#if defined(WEBRTC_ARCH_ARM64)
   *(ptr) = vaddvq_u32(v);
 #else
   uint32x2_t tmp_v;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
index f59cd125f2..91e6f0e80c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 
 #ifdef AEC_DEBUG
 #include <stdio.h>
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
index d7ba3b7820..b45ff59907 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
 
 #include <stdlib.h>
 
@@ -49,7 +49,7 @@ void* WebRtcAecm_Create();
 /*
  * This function releases the memory allocated by WebRtcAecm_Create()
  *
- * Inputs                           Description
+ * Inputs                       Description
  * -------------------------------------------------------------------
  * void*    aecmInst            Pointer to the AECM instance
  */
@@ -60,12 +60,12 @@ void WebRtcAecm_Free(void* aecmInst);
  *
  * Inputs                       Description
  * -------------------------------------------------------------------
- * void*    aecmInst            Pointer to the AECM instance
+ * void*          aecmInst      Pointer to the AECM instance
  * int32_t        sampFreq      Sampling frequency of data
  *
  * Outputs                      Description
  * -------------------------------------------------------------------
- * int32_t  return              0: OK
+ * int32_t        return        0: OK
  *                              1200-12004,12100: error/warning
  */
 int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq);
@@ -206,4 +206,4 @@ size_t WebRtcAecm_echo_path_size_bytes();
 #ifdef __cplusplus
 }
 #endif
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.cc
index 6041435bd9..fc78f07ebb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.cc
@@ -14,21 +14,17 @@
 #include <cstdlib>
 
 #include <algorithm>
+#include <vector>
 
-#include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/agc/histogram.h"
-#include "webrtc/modules/audio_processing/agc/pitch_based_vad.h"
-#include "webrtc/modules/audio_processing/agc/standalone_vad.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace {
 
 const int kDefaultLevelDbfs = -18;
-const double kDefaultVoiceValue = 1.0;
 const int kNumAnalysisFrames = 100;
 const double kActivityThreshold = 0.3;
 
@@ -36,80 +32,31 @@ const double kActivityThreshold = 0.3;
 
 Agc::Agc()
     : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
-      last_voice_probability_(kDefaultVoiceValue),
       target_level_dbfs_(kDefaultLevelDbfs),
-      standalone_vad_enabled_(true),
       histogram_(Histogram::Create(kNumAnalysisFrames)),
-      inactive_histogram_(Histogram::Create()),
-      audio_processing_(new AgcAudioProc()),
-      pitch_based_vad_(new PitchBasedVad()),
-      standalone_vad_(StandaloneVad::Create()),
-      // Initialize to the most common resampling situation.
-      resampler_(new Resampler(32000, kSampleRateHz, 1)) {
+      inactive_histogram_(Histogram::Create()) {
   }
 
 Agc::~Agc() {}
 
-float Agc::AnalyzePreproc(const int16_t* audio, int length) {
+float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
   assert(length > 0);
-  int num_clipped = 0;
-  for (int i = 0; i < length; ++i) {
+  size_t num_clipped = 0;
+  for (size_t i = 0; i < length; ++i) {
     if (audio[i] == 32767 || audio[i] == -32768)
       ++num_clipped;
   }
   return 1.0f * num_clipped / length;
 }
 
-int Agc::Process(const int16_t* audio, int length, int sample_rate_hz) {
-  assert(length == sample_rate_hz / 100);
-  if (sample_rate_hz > 32000) {
-    return -1;
-  }
-  // Resample to the required rate.
-  int16_t resampled[kLength10Ms];
-  const int16_t* resampled_ptr = audio;
-  if (sample_rate_hz != kSampleRateHz) {
-    if (resampler_->ResetIfNeeded(sample_rate_hz, kSampleRateHz, 1) != 0) {
-      return -1;
-    }
-    resampler_->Push(audio, length, resampled, kLength10Ms, length);
-    resampled_ptr = resampled;
-  }
-  assert(length == kLength10Ms);
-
-  if (standalone_vad_enabled_) {
-    if (standalone_vad_->AddAudio(resampled_ptr, length) != 0)
-      return -1;
-  }
-
-  AudioFeatures features;
-  audio_processing_->ExtractFeatures(resampled_ptr, length, &features);
-  if (features.num_frames > 0) {
-    if (features.silence) {
-      // The other features are invalid, so update the histogram with an
-      // arbitrary low value.
-      for (int n = 0; n < features.num_frames; ++n)
-        histogram_->Update(features.rms[n], 0.01);
-      return 0;
-    }
-
-    // Initialize to 0.5 which is a neutral value for combining probabilities,
-    // in case the standalone-VAD is not enabled.
-    double p_combined[] = {0.5, 0.5, 0.5, 0.5};
-    static_assert(sizeof(p_combined) / sizeof(p_combined[0]) == kMaxNumFrames,
-                  "combined probability incorrect size");
-    if (standalone_vad_enabled_) {
-      if (standalone_vad_->GetActivity(p_combined, kMaxNumFrames) < 0)
-        return -1;
-    }
-    // If any other VAD is enabled it must be combined before calling the
-    // pitch-based VAD.
-    if (pitch_based_vad_->VoicingProbability(features, p_combined) < 0)
-      return -1;
-    for (int n = 0; n < features.num_frames; n++) {
-      histogram_->Update(features.rms[n], p_combined[n]);
-      last_voice_probability_ = p_combined[n];
-    }
+int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
+  vad_.ProcessChunk(audio, length, sample_rate_hz);
+  const std::vector<double>& rms = vad_.chunkwise_rms();
+  const std::vector<double>& probabilities =
+      vad_.chunkwise_voice_probabilities();
+  RTC_DCHECK_EQ(rms.size(), probabilities.size());
+  for (size_t i = 0; i < rms.size(); ++i) {
+    histogram_->Update(rms[i], probabilities[i]);
   }
   return 0;
 }
@@ -151,8 +98,4 @@ int Agc::set_target_level_dbfs(int level) {
   return 0;
 }
 
-void Agc::EnableStandaloneVad(bool enable) {
-  standalone_vad_enabled_ = enable;
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.h
index 1ecdab1166..08c287f820 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.h
@@ -12,16 +12,13 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioFrame;
-class AgcAudioProc;
 class Histogram;
-class PitchBasedVad;
-class Resampler;
-class StandaloneVad;
 
 class Agc {
  public:
@@ -30,10 +27,10 @@ class Agc {
 
   // Returns the proportion of samples in the buffer which are at full-scale
   // (and presumably clipped).
-  virtual float AnalyzePreproc(const int16_t* audio, int length);
+  virtual float AnalyzePreproc(const int16_t* audio, size_t length);
   // |audio| must be mono; in a multi-channel stream, provide the first (usually
   // left) channel.
-  virtual int Process(const int16_t* audio, int length, int sample_rate_hz);
+  virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
 
   // Retrieves the difference between the target RMS level and the current
   // signal RMS level in dB. Returns true if an update is available and false
@@ -44,24 +41,16 @@ class Agc {
   virtual int set_target_level_dbfs(int level);
   virtual int target_level_dbfs() const { return target_level_dbfs_; }
 
-  virtual void EnableStandaloneVad(bool enable);
-  virtual bool standalone_vad_enabled() const {
-    return standalone_vad_enabled_;
+  virtual float voice_probability() const {
+    return vad_.last_voice_probability();
   }
 
-  virtual double voice_probability() const { return last_voice_probability_; }
-
  private:
   double target_level_loudness_;
-  double last_voice_probability_;
   int target_level_dbfs_;
-  bool standalone_vad_enabled_;
   rtc::scoped_ptr<Histogram> histogram_;
   rtc::scoped_ptr<Histogram> inactive_histogram_;
-  rtc::scoped_ptr<AgcAudioProc> audio_processing_;
-  rtc::scoped_ptr<PitchBasedVad> pitch_based_vad_;
-  rtc::scoped_ptr<StandaloneVad> standalone_vad_;
-  rtc::scoped_ptr<Resampler> resampler_;
+  VoiceActivityDetector vad_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_internal.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_internal.h
deleted file mode 100644
index f3b7fd1e93..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_internal.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_INTERNAL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_INTERNAL_H_
-
-namespace webrtc {
-
-// These values should match MATLAB counterparts for unit-tests to pass.
-static const double kCorrWeight[] = {
-  1.000000, 0.985000, 0.970225, 0.955672, 0.941337, 0.927217, 0.913308,
-  0.899609, 0.886115, 0.872823, 0.859730, 0.846834, 0.834132, 0.821620,
-  0.809296, 0.797156, 0.785199
-};
-
-static const double kLpcAnalWin[] = {
-  0.00000000, 0.01314436, 0.02628645, 0.03942400, 0.05255473, 0.06567639,
-  0.07878670, 0.09188339, 0.10496421, 0.11802689, 0.13106918, 0.14408883,
-  0.15708358, 0.17005118, 0.18298941, 0.19589602, 0.20876878, 0.22160547,
-  0.23440387, 0.24716177, 0.25987696, 0.27254725, 0.28517045, 0.29774438,
-  0.31026687, 0.32273574, 0.33514885, 0.34750406, 0.35979922, 0.37203222,
-  0.38420093, 0.39630327, 0.40833713, 0.42030043, 0.43219112, 0.44400713,
-  0.45574642, 0.46740697, 0.47898676, 0.49048379, 0.50189608, 0.51322164,
-  0.52445853, 0.53560481, 0.54665854, 0.55761782, 0.56848075, 0.57924546,
-  0.58991008, 0.60047278, 0.61093173, 0.62128512, 0.63153117, 0.64166810,
-  0.65169416, 0.66160761, 0.67140676, 0.68108990, 0.69065536, 0.70010148,
-  0.70942664, 0.71862923, 0.72770765, 0.73666033, 0.74548573, 0.75418233,
-  0.76274862, 0.77118312, 0.77948437, 0.78765094, 0.79568142, 0.80357442,
-  0.81132858, 0.81894256, 0.82641504, 0.83374472, 0.84093036, 0.84797069,
-  0.85486451, 0.86161063, 0.86820787, 0.87465511, 0.88095122, 0.88709512,
-  0.89308574, 0.89892206, 0.90460306, 0.91012776, 0.91549520, 0.92070447,
-  0.92575465, 0.93064488, 0.93537432, 0.93994213, 0.94434755, 0.94858979,
-  0.95266814, 0.95658189, 0.96033035, 0.96391289, 0.96732888, 0.97057773,
-  0.97365889, 0.97657181, 0.97931600, 0.98189099, 0.98429632, 0.98653158,
-  0.98859639, 0.99049038, 0.99221324, 0.99376466, 0.99514438, 0.99635215,
-  0.99738778, 0.99825107, 0.99894188, 0.99946010, 0.99980562, 0.99997840,
-  0.99997840, 0.99980562, 0.99946010, 0.99894188, 0.99825107, 0.99738778,
-  0.99635215, 0.99514438, 0.99376466, 0.99221324, 0.99049038, 0.98859639,
-  0.98653158, 0.98429632, 0.98189099, 0.97931600, 0.97657181, 0.97365889,
-  0.97057773, 0.96732888, 0.96391289, 0.96033035, 0.95658189, 0.95266814,
-  0.94858979, 0.94434755, 0.93994213, 0.93537432, 0.93064488, 0.92575465,
-  0.92070447, 0.91549520, 0.91012776, 0.90460306, 0.89892206, 0.89308574,
-  0.88709512, 0.88095122, 0.87465511, 0.86820787, 0.86161063, 0.85486451,
-  0.84797069, 0.84093036, 0.83374472, 0.82641504, 0.81894256, 0.81132858,
-  0.80357442, 0.79568142, 0.78765094, 0.77948437, 0.77118312, 0.76274862,
-  0.75418233, 0.74548573, 0.73666033, 0.72770765, 0.71862923, 0.70942664,
-  0.70010148, 0.69065536, 0.68108990, 0.67140676, 0.66160761, 0.65169416,
-  0.64166810, 0.63153117, 0.62128512, 0.61093173, 0.60047278, 0.58991008,
-  0.57924546, 0.56848075, 0.55761782, 0.54665854, 0.53560481, 0.52445853,
-  0.51322164, 0.50189608, 0.49048379, 0.47898676, 0.46740697, 0.45574642,
-  0.44400713, 0.43219112, 0.42030043, 0.40833713, 0.39630327, 0.38420093,
-  0.37203222, 0.35979922, 0.34750406, 0.33514885, 0.32273574, 0.31026687,
-  0.29774438, 0.28517045, 0.27254725, 0.25987696, 0.24716177, 0.23440387,
-  0.22160547, 0.20876878, 0.19589602, 0.18298941, 0.17005118, 0.15708358,
-  0.14408883, 0.13106918, 0.11802689, 0.10496421, 0.09188339, 0.07878670,
-  0.06567639, 0.05255473, 0.03942400, 0.02628645, 0.01314436, 0.00000000
-};
-
-static const int kFilterOrder = 2;
-static const float kCoeffNumerator[kFilterOrder + 1] = {0.974827f,  -1.949650f,
-    0.974827f};
-static const float kCoeffDenominator[kFilterOrder + 1] = {1.0f, -1.971999f,
-    0.972457f};
-
-static_assert(kFilterOrder + 1 ==
-                  sizeof(kCoeffNumerator) / sizeof(kCoeffNumerator[0]),
-              "numerator coefficients incorrect size");
-static_assert(kFilterOrder + 1 ==
-                  sizeof(kCoeffDenominator) / sizeof(kCoeffDenominator[0]),
-              "denominator coefficients incorrect size");
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AUDIO_PROCESSING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
index 24fbd56eb2..e56984a1b1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
@@ -19,8 +19,8 @@
 
 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -48,7 +48,6 @@ const float kCompressionGainStep = 0.05f;
 const int kMaxMicLevel = 255;
 static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
 const int kMinMicLevel = 12;
-const int kMinInitMicLevel = 85;
 
 // Prevent very large microphone level changes.
 const int kMaxResidualGainChange = 15;
@@ -57,6 +56,10 @@ const int kMaxResidualGainChange = 15;
 // restrictions from clipping events.
 const int kSurplusCompressionGain = 6;
 
+int ClampLevel(int mic_level) {
+  return std::min(std::max(kMinMicLevel, mic_level), kMaxMicLevel);
+}
+
 int LevelFromGainError(int gain_error, int level) {
   assert(level >= 0 && level <= kMaxMicLevel);
   if (gain_error == 0) {
@@ -92,7 +95,7 @@ class DebugFile {
   ~DebugFile() {
     fclose(file_);
   }
-  void Write(const int16_t* data, int length_samples) {
+  void Write(const int16_t* data, size_t length_samples) {
     fwrite(data, 1, length_samples * sizeof(int16_t), file_);
   }
  private:
@@ -103,13 +106,14 @@ class DebugFile {
   }
   ~DebugFile() {
   }
-  void Write(const int16_t* data, int length_samples) {
+  void Write(const int16_t* data, size_t length_samples) {
   }
 #endif  // WEBRTC_AGC_DEBUG_DUMP
 };
 
 AgcManagerDirect::AgcManagerDirect(GainControl* gctrl,
-                                   VolumeCallbacks* volume_callbacks)
+                                   VolumeCallbacks* volume_callbacks,
+                                   int startup_min_level)
     : agc_(new Agc()),
       gctrl_(gctrl),
       volume_callbacks_(volume_callbacks),
@@ -123,13 +127,15 @@ AgcManagerDirect::AgcManagerDirect(GainControl* gctrl,
       capture_muted_(false),
       check_volume_on_next_process_(true),  // Check at startup.
       startup_(true),
+      startup_min_level_(ClampLevel(startup_min_level)),
       file_preproc_(new DebugFile("agc_preproc.pcm")),
       file_postproc_(new DebugFile("agc_postproc.pcm")) {
 }
 
 AgcManagerDirect::AgcManagerDirect(Agc* agc,
                                    GainControl* gctrl,
-                                   VolumeCallbacks* volume_callbacks)
+                                   VolumeCallbacks* volume_callbacks,
+                                   int startup_min_level)
     : agc_(agc),
       gctrl_(gctrl),
       volume_callbacks_(volume_callbacks),
@@ -143,6 +149,7 @@ AgcManagerDirect::AgcManagerDirect(Agc* agc,
       capture_muted_(false),
       check_volume_on_next_process_(true),  // Check at startup.
       startup_(true),
+      startup_min_level_(ClampLevel(startup_min_level)),
       file_preproc_(new DebugFile("agc_preproc.pcm")),
       file_postproc_(new DebugFile("agc_postproc.pcm")) {
 }
@@ -161,19 +168,19 @@ int AgcManagerDirect::Initialize() {
   // example, what happens when we change devices.
 
   if (gctrl_->set_mode(GainControl::kFixedDigital) != 0) {
-    LOG_FERR1(LS_ERROR, set_mode, GainControl::kFixedDigital);
+    LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
     return -1;
   }
   if (gctrl_->set_target_level_dbfs(2) != 0) {
-    LOG_FERR1(LS_ERROR, set_target_level_dbfs, 2);
+    LOG(LS_ERROR) << "set_target_level_dbfs(2) failed.";
     return -1;
   }
   if (gctrl_->set_compression_gain_db(kDefaultCompressionGain) != 0) {
-    LOG_FERR1(LS_ERROR, set_compression_gain_db, kDefaultCompressionGain);
+    LOG(LS_ERROR) << "set_compression_gain_db(kDefaultCompressionGain) failed.";
     return -1;
   }
   if (gctrl_->enable_limiter(true) != 0) {
-    LOG_FERR1(LS_ERROR, enable_limiter, true);
+    LOG(LS_ERROR) << "enable_limiter(true) failed.";
     return -1;
   }
   return 0;
@@ -181,8 +188,8 @@ int AgcManagerDirect::Initialize() {
 
 void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
                                          int num_channels,
-                                         int samples_per_channel) {
-  int length = num_channels * samples_per_channel;
+                                         size_t samples_per_channel) {
+  size_t length = num_channels * samples_per_channel;
   if (capture_muted_) {
     return;
   }
@@ -223,7 +230,7 @@ void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
 }
 
 void AgcManagerDirect::Process(const int16_t* audio,
-                               int length,
+                               size_t length,
                                int sample_rate_hz) {
   if (capture_muted_) {
     return;
@@ -237,7 +244,7 @@ void AgcManagerDirect::Process(const int16_t* audio,
   }
 
   if (agc_->Process(audio, length, sample_rate_hz) != 0) {
-    LOG_FERR0(LS_ERROR, Agc::Process);
+    LOG(LS_ERROR) << "Agc::Process failed";
     assert(false);
   }
 
@@ -314,7 +321,7 @@ void AgcManagerDirect::SetCaptureMuted(bool muted) {
 }
 
 float AgcManagerDirect::voice_probability() {
-  return static_cast<float>(agc_->voice_probability());
+  return agc_->voice_probability();
 }
 
 int AgcManagerDirect::CheckVolumeAndReset() {
@@ -336,7 +343,7 @@ int AgcManagerDirect::CheckVolumeAndReset() {
   }
   LOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
 
-  int minLevel = startup_ ? kMinInitMicLevel : kMinMicLevel;
+  int minLevel = startup_ ? startup_min_level_ : kMinMicLevel;
   if (level < minLevel) {
     level = minLevel;
     LOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
@@ -427,7 +434,8 @@ void AgcManagerDirect::UpdateCompressor() {
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
-      LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_);
+      LOG(LS_ERROR) << "set_compression_gain_db(" << compression_
+                    << ") failed.";
     }
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.h
index 05b770183c..6edb0f7bf1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct.h
@@ -21,9 +21,9 @@ class DebugFile;
 class GainControl;
 
 // Callbacks that need to be injected into AgcManagerDirect to read and control
-// the volume values. They have different behavior if they are called from
-// AgcManager or AudioProcessing. This is done to remove the VoiceEngine
-// dependency in AgcManagerDirect.
+// the volume values. This is done to remove the VoiceEngine dependency in
+// AgcManagerDirect.
+// TODO(aluebs): Remove VolumeCallbacks.
 class VolumeCallbacks {
  public:
   virtual ~VolumeCallbacks() {}
@@ -33,28 +33,41 @@ class VolumeCallbacks {
 
 // Direct interface to use AGC to set volume and compression values.
 // AudioProcessing uses this interface directly to integrate the callback-less
-// AGC. AgcManager delegates most of its calls here. See agc_manager.h for
-// undocumented methods.
+// AGC.
 //
 // This class is not thread-safe.
-class AgcManagerDirect {
+class AgcManagerDirect final {
  public:
   // AgcManagerDirect will configure GainControl internally. The user is
   // responsible for processing the audio using it after the call to Process.
-  AgcManagerDirect(GainControl* gctrl, VolumeCallbacks* volume_callbacks);
+  // The operating range of startup_min_level is [12, 255] and any input value
+  // outside that range will be clamped.
+  AgcManagerDirect(GainControl* gctrl,
+                   VolumeCallbacks* volume_callbacks,
+                   int startup_min_level);
   // Dependency injection for testing. Don't delete |agc| as the memory is owned
   // by the manager.
   AgcManagerDirect(Agc* agc,
                    GainControl* gctrl,
-                   VolumeCallbacks* volume_callbacks);
+                   VolumeCallbacks* volume_callbacks,
+                   int startup_min_level);
   ~AgcManagerDirect();
 
   int Initialize();
   void AnalyzePreProcess(int16_t* audio,
                          int num_channels,
-                         int samples_per_channel);
-  void Process(const int16_t* audio, int length, int sample_rate_hz);
+                         size_t samples_per_channel);
+  void Process(const int16_t* audio, size_t length, int sample_rate_hz);
 
+  // Call when the capture stream has been muted/unmuted. This causes the
+  // manager to disregard all incoming audio; chances are good it's background
+  // noise to which we'd like to avoid adapting.
+  void SetCaptureMuted(bool muted);
+  bool capture_muted() { return capture_muted_; }
+
+  float voice_probability();
+
+ private:
   // Sets a new microphone level, after first checking that it hasn't been
   // updated by the user, in which case no action is taken.
   void SetLevel(int new_level);
@@ -64,12 +77,6 @@ class AgcManagerDirect {
   // |kClippedLevelMin|.
   void SetMaxLevel(int level);
 
-  void SetCaptureMuted(bool muted);
-  bool capture_muted() { return capture_muted_; }
-
-  float voice_probability();
-
- private:
   int CheckVolumeAndReset();
   void UpdateGain();
   void UpdateCompressor();
@@ -88,9 +95,12 @@ class AgcManagerDirect {
   bool capture_muted_;
   bool check_volume_on_next_process_;
   bool startup_;
+  int startup_min_level_;
 
   rtc::scoped_ptr<DebugFile> file_preproc_;
   rtc::scoped_ptr<DebugFile> file_postproc_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
new file mode 100644
index 0000000000..ce4db59139
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
@@ -0,0 +1,686 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_processing/agc/mock_agc.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+
+using ::testing::_;
+using ::testing::DoAll;
+using ::testing::Eq;
+using ::testing::Mock;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+using ::testing::SetArgReferee;
+
+namespace webrtc {
+namespace {
+
+const int kSampleRateHz = 32000;
+const int kNumChannels = 1;
+const int kSamplesPerChannel = kSampleRateHz / 100;
+const int kInitialVolume = 128;
+const float kAboveClippedThreshold = 0.2f;
+
+class TestVolumeCallbacks : public VolumeCallbacks {
+ public:
+  TestVolumeCallbacks() : volume_(0) {}
+  void SetMicVolume(int volume) override { volume_ = volume; }
+  int GetMicVolume() override { return volume_; }
+
+ private:
+  int volume_;
+};
+
+}  // namespace
+
+class AgcManagerDirectTest : public ::testing::Test {
+ protected:
+  AgcManagerDirectTest()
+      : agc_(new MockAgc), manager_(agc_, &gctrl_, &volume_, kInitialVolume) {
+    ExpectInitialize();
+    manager_.Initialize();
+  }
+
+  void FirstProcess() {
+    EXPECT_CALL(*agc_, Reset());
+    EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
+    CallProcess(1);
+  }
+
+  void SetVolumeAndProcess(int volume) {
+    volume_.SetMicVolume(volume);
+    FirstProcess();
+  }
+
+  void ExpectCheckVolumeAndReset(int volume) {
+    volume_.SetMicVolume(volume);
+    EXPECT_CALL(*agc_, Reset());
+  }
+
+  void ExpectInitialize() {
+    EXPECT_CALL(gctrl_, set_mode(GainControl::kFixedDigital));
+    EXPECT_CALL(gctrl_, set_target_level_dbfs(2));
+    EXPECT_CALL(gctrl_, set_compression_gain_db(7));
+    EXPECT_CALL(gctrl_, enable_limiter(true));
+  }
+
+  void CallProcess(int num_calls) {
+    for (int i = 0; i < num_calls; ++i) {
+      EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return(0));
+      manager_.Process(nullptr, kSamplesPerChannel, kSampleRateHz);
+    }
+  }
+
+  void CallPreProc(int num_calls) {
+    for (int i = 0; i < num_calls; ++i) {
+      manager_.AnalyzePreProcess(nullptr, kNumChannels, kSamplesPerChannel);
+    }
+  }
+
+  MockAgc* agc_;
+  MockGainControl gctrl_;
+  TestVolumeCallbacks volume_;
+  AgcManagerDirect manager_;
+  test::TraceToStderr trace_to_stderr;
+};
+
+TEST_F(AgcManagerDirectTest, StartupMinVolumeConfigurationIsRespected) {
+  FirstProcess();
+  EXPECT_EQ(kInitialVolume, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, MicVolumeResponseToRmsError) {
+  FirstProcess();
+
+  // Compressor default; no residual error.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
+  CallProcess(1);
+
+  // Inside the compressor's window; no change of volume.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)));
+  CallProcess(1);
+
+  // Above the compressor's window; volume should be increased.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(130, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(168, volume_.GetMicVolume());
+
+  // Inside the compressor's window; no change of volume.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
+  CallProcess(1);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)));
+  CallProcess(1);
+
+  // Below the compressor's window; volume should be decreased.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(167, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(163, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-9), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(129, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, MicVolumeIsLimited) {
+  FirstProcess();
+
+  // Maximum upwards change is limited.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(183, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(243, volume_.GetMicVolume());
+
+  // Won't go higher than the maximum.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(255, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(254, volume_.GetMicVolume());
+
+  // Maximum downwards change is limited.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(194, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(137, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(88, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(54, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(33, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(18, volume_.GetMicVolume());
+
+  // Won't go lower than the minimum.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(12, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, CompressorStepsTowardsTarget) {
+  FirstProcess();
+
+  // Compressor default; no call to set_compression_gain_db.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(20);
+
+  // Moves slowly upwards.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  CallProcess(1);
+
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  CallProcess(1);
+
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(20);
+
+  // Moves slowly downward, then reverses before reaching the original target.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  CallProcess(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  CallProcess(1);
+
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(20);
+}
+
+TEST_F(AgcManagerDirectTest, CompressorErrorIsDeemphasized) {
+  FirstProcess();
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillRepeatedly(Return(false));
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  CallProcess(1);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(20);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillRepeatedly(Return(false));
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(7)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
+  CallProcess(1);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(_)).Times(0);
+  CallProcess(20);
+}
+
+TEST_F(AgcManagerDirectTest, CompressorReachesMaximum) {
+  FirstProcess();
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillRepeatedly(Return(false));
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
+  CallProcess(1);
+}
+
+TEST_F(AgcManagerDirectTest, CompressorReachesMinimum) {
+  FirstProcess();
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillRepeatedly(Return(false));
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(5)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(4)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(3)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(2)).WillOnce(Return(0));
+  CallProcess(1);
+}
+
+TEST_F(AgcManagerDirectTest, NoActionWhileMuted) {
+  manager_.SetCaptureMuted(true);
+  manager_.Process(nullptr, kSamplesPerChannel, kSampleRateHz);
+}
+
+TEST_F(AgcManagerDirectTest, UnmutingChecksVolumeWithoutRaising) {
+  FirstProcess();
+
+  manager_.SetCaptureMuted(true);
+  manager_.SetCaptureMuted(false);
+  ExpectCheckVolumeAndReset(127);
+  // SetMicVolume should not be called.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
+  CallProcess(1);
+  EXPECT_EQ(127, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, UnmutingRaisesTooLowVolume) {
+  FirstProcess();
+
+  manager_.SetCaptureMuted(true);
+  manager_.SetCaptureMuted(false);
+  ExpectCheckVolumeAndReset(11);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
+  CallProcess(1);
+  EXPECT_EQ(12, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ManualLevelChangeResultsInNoSetMicCall) {
+  FirstProcess();
+
+  // Change outside of compressor's range, which would normally trigger a call
+  // to SetMicVolume.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  // GetMicVolume returns a value outside of the quantization slack, indicating
+  // a manual volume change.
+  volume_.SetMicVolume(154);
+  // SetMicVolume should not be called.
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallProcess(1);
+  EXPECT_EQ(154, volume_.GetMicVolume());
+
+  // Do the same thing, except downwards now.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  volume_.SetMicVolume(100);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallProcess(1);
+  EXPECT_EQ(100, volume_.GetMicVolume());
+
+  // And finally verify the AGC continues working without a manual change.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(99, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, RecoveryAfterManualLevelChangeFromMax) {
+  FirstProcess();
+
+  // Force the mic up to max volume. Takes a few steps due to the residual
+  // gain limitation.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(183, volume_.GetMicVolume());
+  CallProcess(1);
+  EXPECT_EQ(243, volume_.GetMicVolume());
+  CallProcess(1);
+  EXPECT_EQ(255, volume_.GetMicVolume());
+
+  // Manual change does not result in SetMicVolume call.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  volume_.SetMicVolume(50);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallProcess(1);
+  EXPECT_EQ(50, volume_.GetMicVolume());
+
+  // Continues working as usual afterwards.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(69, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, RecoveryAfterManualLevelChangeBelowMin) {
+  FirstProcess();
+
+  // Manual change below min.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  // Don't set to zero, which will cause AGC to take no action.
+  volume_.SetMicVolume(1);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallProcess(1);
+  EXPECT_EQ(1, volume_.GetMicVolume());
+
+  // Continues working as usual afterwards.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(2, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(11, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(18, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, NoClippingHasNoImpact) {
+  FirstProcess();
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillRepeatedly(Return(0));
+  CallPreProc(100);
+  EXPECT_EQ(128, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ClippingUnderThresholdHasNoImpact) {
+  FirstProcess();
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.099));
+  CallPreProc(1);
+  EXPECT_EQ(128, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ClippingLowersVolume) {
+  SetVolumeAndProcess(255);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.101));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, WaitingPeriodBetweenClippingChecks) {
+  SetVolumeAndProcess(255);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillRepeatedly(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  CallPreProc(300);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(225, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ClippingLoweringIsLimited) {
+  SetVolumeAndProcess(180);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(170, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillRepeatedly(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  CallPreProc(1000);
+  EXPECT_EQ(170, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ClippingMaxIsRespectedWhenEqualToLevel) {
+  SetVolumeAndProcess(255);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(10);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ClippingMaxIsRespectedWhenHigherThanLevel) {
+  SetVolumeAndProcess(200);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(185, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+  CallProcess(10);
+  EXPECT_EQ(240, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, MaxCompressionIsIncreasedAfterClipping) {
+  SetVolumeAndProcess(210);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(195, volume_.GetMicVolume());
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillRepeatedly(Return(false));
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(13)).WillOnce(Return(0));
+  CallProcess(1);
+
+  // Continue clipping until we hit the maximum surplus compression.
+  CallPreProc(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(180, volume_.GetMicVolume());
+
+  CallPreProc(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(170, volume_.GetMicVolume());
+
+  // Current level is now at the minimum, but the maximum allowed level still
+  // has more to decrease.
+  CallPreProc(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  CallPreProc(1);
+
+  CallPreProc(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  CallPreProc(1);
+
+  CallPreProc(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  CallPreProc(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillRepeatedly(Return(false));
+  CallProcess(19);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(14)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(15)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(16)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(17)).WillOnce(Return(0));
+  CallProcess(20);
+  EXPECT_CALL(gctrl_, set_compression_gain_db(18)).WillOnce(Return(0));
+  CallProcess(1);
+}
+
+TEST_F(AgcManagerDirectTest, UserCanRaiseVolumeAfterClipping) {
+  SetVolumeAndProcess(225);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallPreProc(1);
+  EXPECT_EQ(210, volume_.GetMicVolume());
+
+  // High enough error to trigger a volume check.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(14), Return(true)));
+  // User changed the volume.
+  volume_.SetMicVolume(250);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  CallProcess(1);
+  EXPECT_EQ(250, volume_.GetMicVolume());
+
+  // Move down...
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-10), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(210, volume_.GetMicVolume());
+  // And back up to the new max established by the user.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(40), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(250, volume_.GetMicVolume());
+  // Will not move above new maximum.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  CallProcess(1);
+  EXPECT_EQ(250, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, ClippingDoesNotPullLowVolumeBackUp) {
+  SetVolumeAndProcess(80);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  int initial_volume = volume_.GetMicVolume();
+  CallPreProc(1);
+  EXPECT_EQ(initial_volume, volume_.GetMicVolume());
+}
+
+TEST_F(AgcManagerDirectTest, TakesNoActionOnZeroMicVolume) {
+  FirstProcess();
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  volume_.SetMicVolume(0);
+  CallProcess(10);
+  EXPECT_EQ(0, volume_.GetMicVolume());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_unittest.cc
index 66a8a2b1b3..25b99d8773 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_unittest.cc
@@ -13,7 +13,7 @@
 #include "gmock/gmock.h"
 #include "gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/tools/agc/test_utils.h"
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/histogram.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/histogram.cc
index 1d3035fe12..5c66727a9f 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/histogram.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/histogram.cc
@@ -13,7 +13,7 @@
 #include <cmath>
 #include <cstring>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/analog_agc.c b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
index 17e06fe139..3a1dc9d5ce 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
@@ -41,7 +41,7 @@ static const int16_t kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17
 
 static const int16_t kMuteGuardTimeMs = 8000;
 static const int16_t kInitCheck = 42;
-static const int16_t kNumSubframes = 10;
+static const size_t kNumSubframes = 10;
 
 /* Default settings if config is not used */
 #define AGC_DEFAULT_TARGET_LEVEL 3
@@ -112,13 +112,14 @@ static const int32_t kTargetLevelTable[64] = {134209536, 106606424, 84680493, 67
         6726, 5343, 4244, 3371, 2678, 2127, 1690, 1342, 1066, 847, 673, 534, 424, 337, 268,
         213, 169, 134, 107, 85, 67};
 
-int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, int16_t num_bands,
-                     int16_t samples)
+int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, size_t num_bands,
+                     size_t samples)
 {
     int32_t nrg, max_nrg, sample, tmp32;
     int32_t *ptr;
     uint16_t targetGainIdx, gain;
-    int16_t i, n, L, tmp16, tmp_speech[16];
+    size_t i;
+    int16_t n, L, tmp16, tmp_speech[16];
     LegacyAgc* stt;
     stt = (LegacyAgc*)state;
 
@@ -164,7 +165,7 @@ int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, int16_t num_bands,
 
         for (i = 0; i < samples; i++)
         {
-            int j;
+            size_t j;
             for (j = 0; j < num_bands; ++j)
             {
                 sample = (in_mic[j][i] * gain) >> 12;
@@ -249,47 +250,48 @@ int WebRtcAgc_AddMic(void *state, int16_t* const* in_mic, int16_t num_bands,
     return 0;
 }
 
-int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
-{
-  LegacyAgc* stt;
-  stt = (LegacyAgc*)state;
+int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, size_t samples) {
+  LegacyAgc* stt = (LegacyAgc*)state;
 
-    if (stt == NULL)
-    {
-        return -1;
-    }
+    int err =  WebRtcAgc_GetAddFarendError(state, samples);
 
-    if (stt->fs == 8000)
-    {
-        if (samples != 80)
-        {
-            return -1;
-        }
-    } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000)
-    {
-        if (samples != 160)
-        {
-            return -1;
-        }
-    } else
-    {
-        return -1;
-    }
+    if (err != 0)
+      return err;
 
     return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
 }
 
+int WebRtcAgc_GetAddFarendError(void *state, size_t samples) {
+  LegacyAgc* stt;
+  stt = (LegacyAgc*)state;
+
+  if (stt == NULL)
+    return -1;
+
+  if (stt->fs == 8000) {
+    if (samples != 80)
+      return -1;
+  } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000) {
+    if (samples != 160)
+      return -1;
+  } else {
+    return -1;
+  }
+
+  return 0;
+}
+
 int WebRtcAgc_VirtualMic(void *agcInst, int16_t* const* in_near,
-                         int16_t num_bands, int16_t samples, int32_t micLevelIn,
+                         size_t num_bands, size_t samples, int32_t micLevelIn,
                          int32_t *micLevelOut)
 {
     int32_t tmpFlt, micLevelTmp, gainIdx;
     uint16_t gain;
-    int16_t ii, j;
+    size_t ii, j;
     LegacyAgc* stt;
 
     uint32_t nrg;
-    int16_t sampleCntr;
+    size_t sampleCntr;
     uint32_t frameNrg = 0;
     uint32_t frameNrgLimit = 5500;
     int16_t numZeroCrossing = 0;
@@ -1132,7 +1134,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
 }
 
 int WebRtcAgc_Process(void *agcInst, const int16_t* const* in_near,
-                      int16_t num_bands, int16_t samples,
+                      size_t num_bands, size_t samples,
                       int16_t* const* out, int32_t inMicLevel,
                       int32_t *outMicLevel, int16_t echo,
                       uint8_t *saturationWarning)
@@ -1313,46 +1315,31 @@ int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config) {
     return 0;
 }
 
-int WebRtcAgc_Create(void **agcInst)
-{
-  LegacyAgc* stt;
-    if (agcInst == NULL)
-    {
-        return -1;
-    }
-    stt = (LegacyAgc*)malloc(sizeof(LegacyAgc));
-
-    *agcInst = stt;
-    if (stt == NULL)
-    {
-        return -1;
-    }
+void* WebRtcAgc_Create() {
+  LegacyAgc* stt = malloc(sizeof(LegacyAgc));
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
-    stt->fpt = fopen("./agc_test_log.txt", "wt");
-    stt->agcLog = fopen("./agc_debug_log.txt", "wt");
-    stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
+  stt->fpt = fopen("./agc_test_log.txt", "wt");
+  stt->agcLog = fopen("./agc_debug_log.txt", "wt");
+  stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
 #endif
 
-    stt->initFlag = 0;
-    stt->lastError = 0;
+  stt->initFlag = 0;
+  stt->lastError = 0;
 
-    return 0;
+  return stt;
 }
 
-int WebRtcAgc_Free(void *state)
-{
+void WebRtcAgc_Free(void *state) {
   LegacyAgc* stt;
 
   stt = (LegacyAgc*)state;
 #ifdef WEBRTC_AGC_DEBUG_DUMP
-    fclose(stt->fpt);
-    fclose(stt->agcLog);
-    fclose(stt->digitalAgc.logFile);
+  fclose(stt->fpt);
+  fclose(stt->agcLog);
+  fclose(stt->digitalAgc.logFile);
 #endif
-    free(stt);
-
-    return 0;
+  free(stt);
 }
 
 /* minLevel     - Minimum volume level
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.c b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
index 4619b88ae5..aeafb65c78 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
@@ -283,7 +283,7 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc* stt, int16_t agcMode) {
 
 int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt,
                                      const int16_t* in_far,
-                                     int16_t nrSamples) {
+                                     size_t nrSamples) {
     assert(stt != NULL);
     // VAD for far end
     WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
@@ -293,7 +293,7 @@ int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt,
 
 int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt,
                                  const int16_t* const* in_near,
-                                 int16_t num_bands,
+                                 size_t num_bands,
                                  int16_t* const* out,
                                  uint32_t FS,
                                  int16_t lowlevelSignal) {
@@ -310,8 +310,9 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt,
     int16_t zeros = 0, zeros_fast, frac = 0;
     int16_t decay;
     int16_t gate, gain_adj;
-    int16_t k, n, i;
-    int16_t L, L2; // samples/subframe
+    int16_t k;
+    size_t n, i, L;
+    int16_t L2; // samples/subframe
 
     // determine number of samples per ms
     if (FS == 8000)
@@ -632,7 +633,7 @@ void WebRtcAgc_InitVad(AgcVad* state) {
 
 int16_t WebRtcAgc_ProcessVad(AgcVad* state,      // (i) VAD state
                              const int16_t* in,  // (i) Speech signal
-                             int16_t nrSamples)  // (i) number of samples
+                             size_t nrSamples)  // (i) number of samples
 {
     int32_t out, nrg, tmp32, tmp32b;
     uint16_t tmpU16;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
index b8314d9891..819844d774 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
@@ -56,20 +56,20 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
 
 int32_t WebRtcAgc_ProcessDigital(DigitalAgc* digitalAgcInst,
                                  const int16_t* const* inNear,
-                                 int16_t num_bands,
+                                 size_t num_bands,
                                  int16_t* const* out,
                                  uint32_t FS,
                                  int16_t lowLevelSignal);
 
 int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
                                      const int16_t* inFar,
-                                     int16_t nrSamples);
+                                     size_t nrSamples);
 
 void WebRtcAgc_InitVad(AgcVad* vadInst);
 
 int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,     // (i) VAD state
                              const int16_t* in,   // (i) Speech signal
-                             int16_t nrSamples);  // (i) number of samples
+                             size_t nrSamples);  // (i) number of samples
 
 int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
                                      int16_t compressionGaindB, // Q0 (in dB)
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/gain_control.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/gain_control.h
index cf1e4f1fc3..db942fe5ec 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/gain_control.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/legacy/gain_control.h
@@ -49,6 +49,20 @@ extern "C"
 {
 #endif
 
+/*
+ * This function analyses the number of samples passed to
+ * farend and produces any error code that could arise.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - samples           : Number of samples in input vector.
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error.
+ */
+int WebRtcAgc_GetAddFarendError(void* state, size_t samples);
+
 /*
  * This function processes a 10 ms frame of far-end speech to determine
  * if there is active speech. The length of the input speech vector must be
@@ -66,7 +80,7 @@ extern "C"
  */
 int WebRtcAgc_AddFarend(void* agcInst,
                         const int16_t* inFar,
-                        int16_t samples);
+                        size_t samples);
 
 /*
  * This function processes a 10 ms frame of microphone speech to determine
@@ -90,8 +104,8 @@ int WebRtcAgc_AddFarend(void* agcInst,
  */
 int WebRtcAgc_AddMic(void* agcInst,
                      int16_t* const* inMic,
-                     int16_t num_bands,
-                     int16_t samples);
+                     size_t num_bands,
+                     size_t samples);
 
 /*
  * This function replaces the analog microphone with a virtual one.
@@ -118,8 +132,8 @@ int WebRtcAgc_AddMic(void* agcInst,
  */
 int WebRtcAgc_VirtualMic(void* agcInst,
                          int16_t* const* inMic,
-                         int16_t num_bands,
-                         int16_t samples,
+                         size_t num_bands,
+                         size_t samples,
                          int32_t micLevelIn,
                          int32_t* micLevelOut);
 
@@ -159,8 +173,8 @@ int WebRtcAgc_VirtualMic(void* agcInst,
  */
 int WebRtcAgc_Process(void* agcInst,
                       const int16_t* const* inNear,
-                      int16_t num_bands,
-                      int16_t samples,
+                      size_t num_bands,
+                      size_t samples,
                       int16_t* const* out,
                       int32_t inMicLevel,
                       int32_t* outMicLevel,
@@ -200,24 +214,18 @@ int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig config);
 int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config);
 
 /*
- * This function creates an AGC instance, which will contain the state
- * information for one (duplex) channel.
- *
- * Return value             : AGC instance if successful
- *                          : 0 (i.e., a NULL pointer) if unsuccessful
+ * This function creates and returns an AGC instance, which will contain the
+ * state information for one (duplex) channel.
  */
-int WebRtcAgc_Create(void **agcInst);
+void* WebRtcAgc_Create();
 
 /*
  * This function frees the AGC instance created at the beginning.
  *
  * Input:
  *      - agcInst           : AGC instance.
- *
- * Return value             :  0 - Ok
- *                            -1 - Error
  */
-int WebRtcAgc_Free(void *agcInst);
+void WebRtcAgc_Free(void* agcInst);
 
 /*
  * This function initializes an AGC instance.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/mock_agc.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/mock_agc.h
index 1c36a055ec..e362200d86 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/mock_agc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/mock_agc.h
@@ -14,14 +14,14 @@
 #include "webrtc/modules/audio_processing/agc/agc.h"
 
 #include "gmock/gmock.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 class MockAgc : public Agc {
  public:
-  MOCK_METHOD2(AnalyzePreproc, float(const int16_t* audio, int length));
-  MOCK_METHOD3(Process, int(const int16_t* audio, int length,
+  MOCK_METHOD2(AnalyzePreproc, float(const int16_t* audio, size_t length));
+  MOCK_METHOD3(Process, int(const int16_t* audio, size_t length,
                             int sample_rate_hz));
   MOCK_METHOD1(GetRmsErrorDb, bool(int* error));
   MOCK_METHOD0(Reset, void());
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/noise_gmm_tables.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/noise_gmm_tables.h
deleted file mode 100644
index 779fd8c368..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/noise_gmm_tables.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// GMM tables for inactive segments. Generated by MakeGmmTables.m.
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_NOISE_GMM_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_NOISE_GMM_TABLES_H_
-
-static const int kNoiseGmmNumMixtures = 12;
-static const int kNoiseGmmDim = 3;
-
-static const double kNoiseGmmCovarInverse[kNoiseGmmNumMixtures]
-                                          [kNoiseGmmDim][kNoiseGmmDim] = {
-  {{ 7.36219567592941e+00,  4.83060785179861e-03,  1.23335151497610e-02},
-   { 4.83060785179861e-03,  1.65289507047817e-04, -2.41490588169997e-04},
-   { 1.23335151497610e-02, -2.41490588169997e-04,  6.59472060689382e-03}},
-  {{ 8.70265239309140e+00, -5.30636201431086e-04,  5.44014966585347e-03},
-   {-5.30636201431086e-04,  3.11095453521008e-04, -1.86287206836035e-04},
-   { 5.44014966585347e-03, -1.86287206836035e-04,  6.29493388790744e-04}},
-  {{ 4.53467851955055e+00, -3.92977536695197e-03, -2.46521420693317e-03},
-   {-3.92977536695197e-03,  4.94650752632750e-05, -1.08587438501826e-05},
-   {-2.46521420693317e-03, -1.08587438501826e-05,  9.28793975422261e-05}},
-  {{ 9.26817997114275e-01, -4.03976069276753e-04, -3.56441427392165e-03},
-   {-4.03976069276753e-04,  2.51976251631430e-06,  1.46914206734572e-07},
-   {-3.56441427392165e-03,  1.46914206734572e-07,  8.19914567685373e-05}},
-  {{ 7.61715986787441e+00, -1.54889041216888e-04,  2.41756280071656e-02},
-   {-1.54889041216888e-04,  3.50282550461672e-07, -6.27251196972490e-06},
-   { 2.41756280071656e-02, -6.27251196972490e-06,  1.45061847649872e-02}},
-  {{ 8.31193642663158e+00, -3.84070508164323e-04, -3.09750630821876e-02},
-   {-3.84070508164323e-04,  3.80433432277336e-07, -1.14321142836636e-06},
-   {-3.09750630821876e-02, -1.14321142836636e-06,  8.35091486289997e-04}},
-  {{ 9.67283151270894e-01,  5.82465812445039e-05, -3.18350798617053e-03},
-   { 5.82465812445039e-05,  2.23762672000318e-07, -7.74196587408623e-07},
-   {-3.18350798617053e-03, -7.74196587408623e-07,  3.85120938338325e-04}},
-  {{ 8.28066236985388e+00,  5.87634508319763e-05,  6.99303090891743e-03},
-   { 5.87634508319763e-05,  2.93746018618058e-07,  3.40843332882272e-07},
-   { 6.99303090891743e-03,  3.40843332882272e-07,  1.99379171190344e-04}},
-  {{ 6.07488998675646e+00, -1.11494526618473e-02,  5.10013111123381e-03},
-   {-1.11494526618473e-02,  6.99238879921751e-04,  5.36718550370870e-05},
-   { 5.10013111123381e-03,  5.36718550370870e-05,  5.26909853276753e-04}},
-  {{ 6.90492021419175e+00,  4.20639355257863e-04, -2.38612752336481e-03},
-   { 4.20639355257863e-04,  3.31246767338153e-06, -2.42052288150859e-08},
-   {-2.38612752336481e-03, -2.42052288150859e-08,  4.46608368363412e-04}},
-  {{ 1.31069150869715e+01, -1.73718583865670e-04, -1.97591814508578e-02},
-   {-1.73718583865670e-04,  2.80451716300124e-07,  9.96570755379865e-07},
-   {-1.97591814508578e-02,  9.96570755379865e-07,  2.41361900868847e-03}},
-  {{ 4.69566344239814e+00, -2.61077567563690e-04,  5.26359000761433e-03},
-   {-2.61077567563690e-04,  1.82420859823767e-06, -7.83645887541601e-07},
-   { 5.26359000761433e-03, -7.83645887541601e-07,  1.33586288288802e-02}}};
-
-static const double kNoiseGmmMean[kNoiseGmmNumMixtures][kNoiseGmmDim] = {
-  {-2.01386094766163e+00,  1.69702162045397e+02,  7.41715804872181e+01},
-  {-1.94684591777290e+00,  1.42398396732668e+02,  1.64186321157831e+02},
-  {-2.29319297562437e+00,  3.86415425589868e+02,  2.13452215267125e+02},
-  {-3.25487177070268e+00,  1.08668712553616e+03,  2.33119949467419e+02},
-  {-2.13159632447467e+00,  4.83821702557717e+03,  6.86786166673740e+01},
-  {-2.26171410780526e+00,  4.79420193982422e+03,  1.53222513286450e+02},
-  {-3.32166740703185e+00,  4.35161135834358e+03,  1.33206448431316e+02},
-  {-2.19290322814343e+00,  3.98325506609408e+03,  2.13249167359934e+02},
-  {-2.02898459255404e+00,  7.37039893155007e+03,  1.12518527491926e+02},
-  {-2.26150236399500e+00,  1.54896745196145e+03,  1.49717357868579e+02},
-  {-2.00417668301790e+00,  3.82434760310304e+03,  1.07438913004312e+02},
-  {-2.30193040814533e+00,  1.43953696546439e+03,  7.04085275122649e+01}};
-
-static const double kNoiseGmmWeights[kNoiseGmmNumMixtures] = {
-  -1.09422832086193e+01, -1.10847897513425e+01, -1.36767587732187e+01,
-  -1.79789356118641e+01, -1.42830169160894e+01, -1.56500228061379e+01,
-  -1.83124990950113e+01, -1.69979436177477e+01, -1.12329424387828e+01,
-  -1.41311785780639e+01, -1.47171861448585e+01, -1.35963362781839e+01};
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_NOISE_GMM_TABLES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/voice_gmm_tables.h b/media/webrtc/trunk/webrtc/modules/audio_processing/agc/voice_gmm_tables.h
deleted file mode 100644
index 9a490a47e0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/voice_gmm_tables.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// GMM tables for active segments. Generated by MakeGmmTables.m.
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_VOICE_GMM_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_VOICE_GMM_TABLES_H_
-
-static const int kVoiceGmmNumMixtures = 12;
-static const int kVoiceGmmDim = 3;
-
-static const double kVoiceGmmCovarInverse[kVoiceGmmNumMixtures]
-                                          [kVoiceGmmDim][kVoiceGmmDim] = {
-  {{ 1.83673825579513e+00, -8.09791637570095e-04,  4.60106414365986e-03},
-   {-8.09791637570095e-04,  8.89351738394608e-04, -9.80188953277734e-04},
-   { 4.60106414365986e-03, -9.80188953277734e-04,  1.38706060206582e-03}},
-  {{ 6.76228912850703e+01, -1.98893120119660e-02, -3.53548357253551e-03},
-   {-1.98893120119660e-02,  3.96216858500530e-05, -4.08492938394097e-05},
-   {-3.53548357253551e-03, -4.08492938394097e-05,  9.31864352856416e-04}},
-  {{ 9.98612435944558e+00, -5.27880954316893e-03, -6.30342541619017e-03},
-   {-5.27880954316893e-03,  4.54359480225226e-05,  6.30804591626044e-05},
-   {-6.30342541619017e-03,  6.30804591626044e-05,  5.36466441382942e-04}},
-  {{ 3.39917474216349e+01, -1.56213579433191e-03, -4.01459014990225e-02},
-   {-1.56213579433191e-03,  6.40415424897724e-05,  6.20076342427833e-05},
-   {-4.01459014990225e-02,  6.20076342427833e-05,  3.51199070103063e-03}},
-  {{ 1.34545062271428e+01, -7.94513610147144e-03, -5.34401019341728e-02},
-   {-7.94513610147144e-03,  1.16511820098649e-04,  4.66063702069293e-05},
-   {-5.34401019341728e-02,  4.66063702069293e-05,  2.72354323774163e-03}},
-  {{ 1.08557844314806e+02, -1.54885805673668e-02, -1.88029692674851e-02},
-   {-1.54885805673668e-02,  1.16404042786406e-04,  6.45579292702802e-06},
-   {-1.88029692674851e-02,  6.45579292702802e-06,  4.32330478391416e-04}},
-  {{ 8.22940066541450e+01, -1.15903110231303e-02, -4.92166764865343e-02},
-   {-1.15903110231303e-02,  7.42510742165261e-05,  3.73007314191290e-06},
-   {-4.92166764865343e-02,  3.73007314191290e-06,  3.64005221593244e-03}},
-  {{ 2.31133605685660e+00, -7.83261568950254e-04,  7.45744012346313e-04},
-   {-7.83261568950254e-04,  1.29460648214142e-05, -2.22774455093730e-06},
-   { 7.45744012346313e-04, -2.22774455093730e-06,  1.05117294093010e-04}},
-  {{ 3.78767849189611e+02,  1.57759761011568e-03, -2.08551217988774e-02},
-   { 1.57759761011568e-03,  4.76066236886865e-05, -2.33977412299324e-05},
-   {-2.08551217988774e-02, -2.33977412299324e-05,  5.24261005371196e-04}},
-  {{ 6.98580096506135e-01, -5.13850255217378e-04, -4.01124551717056e-04},
-   {-5.13850255217378e-04,  1.40501021984840e-06, -2.09496928716569e-06},
-   {-4.01124551717056e-04, -2.09496928716569e-06,  2.82879357740037e-04}},
-  {{ 2.62770945162399e+00, -2.31825753241430e-03, -5.30447217466318e-03},
-   {-2.31825753241430e-03,  4.59108572227649e-05,  7.67631886355405e-05},
-   {-5.30447217466318e-03,  7.67631886355405e-05,  2.28521601674098e-03}},
-  {{ 1.89940391362152e+02, -4.23280856852379e-03, -2.70608873541399e-02},
-   {-4.23280856852379e-03,  6.77547582742563e-05,  2.69154203800467e-05},
-   {-2.70608873541399e-02,  2.69154203800467e-05,  3.88574543373470e-03}}};
-
-static const double kVoiceGmmMean[kVoiceGmmNumMixtures][kVoiceGmmDim] = {
-  {-2.15020241646536e+00,  4.97079062999877e+02,  4.77078119504505e+02},
-  {-8.92097680029190e-01,  5.92064964199921e+02,  1.81045145941059e+02},
-  {-1.29435784144398e+00,  4.98450293410611e+02,  1.71991263804064e+02},
-  {-1.03925228397884e+00,  4.99511274321571e+02,  1.05838336539105e+02},
-  {-1.29229047206129e+00,  4.15026762566707e+02,  1.12861119017125e+02},
-  {-7.88748114599810e-01,  4.48739336688113e+02,  1.89784216956337e+02},
-  {-8.77777402332642e-01,  4.86620285054533e+02,  1.13477708016491e+02},
-  {-2.06465957063057e+00,  6.33385049870607e+02,  2.32758546796149e+02},
-  {-6.98893789231685e-01,  5.93622051503385e+02,  1.92536982473203e+02},
-  {-2.55901217508894e+00,  1.55914919756205e+03,  1.39769980835570e+02},
-  {-1.92070024165837e+00,  4.87983940444185e+02,  1.02745468128289e+02},
-  {-7.29187507662854e-01,  5.22717685022855e+02,  1.16377942283991e+02}};
-
-static const double kVoiceGmmWeights[kVoiceGmmNumMixtures] = {
-  -1.39789694361035e+01, -1.19527720202104e+01, -1.32396317929055e+01,
-  -1.09436815209238e+01, -1.13440027478149e+01, -1.12200721834504e+01,
-  -1.02537324043693e+01, -1.60789861938302e+01, -1.03394494048344e+01,
-  -1.83207938586818e+01, -1.31186044948288e+01, -9.52479998673554e+00};
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_VOICE_GMM_TABLES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.cc
index e7419440e7..ff64267e8c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 
+#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/channel_buffer.h"
@@ -18,58 +19,35 @@
 namespace webrtc {
 namespace {
 
-bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
-  switch (layout) {
-    case AudioProcessing::kMono:
-    case AudioProcessing::kStereo:
-      return false;
-    case AudioProcessing::kMonoAndKeyboard:
-    case AudioProcessing::kStereoAndKeyboard:
-      return true;
+const size_t kSamplesPer16kHzChannel = 160;
+const size_t kSamplesPer32kHzChannel = 320;
+const size_t kSamplesPer48kHzChannel = 480;
+
+int KeyboardChannelIndex(const StreamConfig& stream_config) {
+  if (!stream_config.has_keyboard()) {
+    assert(false);
+    return 0;
   }
-  assert(false);
-  return false;
+
+  return stream_config.num_channels();
 }
 
-int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) {
-  switch (layout) {
-    case AudioProcessing::kMono:
-    case AudioProcessing::kStereo:
-      assert(false);
-      return -1;
-    case AudioProcessing::kMonoAndKeyboard:
-      return 1;
-    case AudioProcessing::kStereoAndKeyboard:
-      return 2;
-  }
-  assert(false);
-  return -1;
-}
-
-template <typename T>
-void StereoToMono(const T* left, const T* right, T* out,
-                  int num_frames) {
-  for (int i = 0; i < num_frames; ++i)
-    out[i] = (left[i] + right[i]) / 2;
-}
-
-int NumBandsFromSamplesPerChannel(int num_frames) {
-  int num_bands = 1;
+size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
+  size_t num_bands = 1;
   if (num_frames == kSamplesPer32kHzChannel ||
       num_frames == kSamplesPer48kHzChannel) {
-    num_bands = rtc::CheckedDivExact(num_frames,
-                                     static_cast<int>(kSamplesPer16kHzChannel));
+    num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel);
   }
   return num_bands;
 }
 
 }  // namespace
 
-AudioBuffer::AudioBuffer(int input_num_frames,
-                         int num_input_channels,
-                         int process_num_frames,
-                         int num_process_channels,
-                         int output_num_frames)
+AudioBuffer::AudioBuffer(size_t input_num_frames,
+                         size_t num_input_channels,
+                         size_t process_num_frames,
+                         size_t num_process_channels,
+                         size_t output_num_frames)
   : input_num_frames_(input_num_frames),
     num_input_channels_(num_input_channels),
     proc_num_frames_(process_num_frames),
@@ -77,8 +55,7 @@ AudioBuffer::AudioBuffer(int input_num_frames,
     output_num_frames_(output_num_frames),
     num_channels_(num_process_channels),
     num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
-    num_split_frames_(rtc::CheckedDivExact(
-        proc_num_frames_, num_bands_)),
+    num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
@@ -87,14 +64,9 @@ AudioBuffer::AudioBuffer(int input_num_frames,
   assert(input_num_frames_ > 0);
   assert(proc_num_frames_ > 0);
   assert(output_num_frames_ > 0);
-  assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
+  assert(num_input_channels_ > 0);
   assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_);
 
-  if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
-    input_buffer_.reset(new ChannelBuffer<float>(input_num_frames_,
-                                                 num_proc_channels_));
-  }
-
   if (input_num_frames_ != proc_num_frames_ ||
       output_num_frames_ != proc_num_frames_) {
     // Create an intermediate buffer for resampling.
@@ -102,7 +74,7 @@ AudioBuffer::AudioBuffer(int input_num_frames,
                                                    num_proc_channels_));
 
     if (input_num_frames_ != proc_num_frames_) {
-      for (int i = 0; i < num_proc_channels_; ++i) {
+      for (size_t i = 0; i < num_proc_channels_; ++i) {
         input_resamplers_.push_back(
             new PushSincResampler(input_num_frames_,
                                   proc_num_frames_));
@@ -110,7 +82,7 @@ AudioBuffer::AudioBuffer(int input_num_frames,
     }
 
     if (output_num_frames_ != proc_num_frames_) {
-      for (int i = 0; i < num_proc_channels_; ++i) {
+      for (size_t i = 0; i < num_proc_channels_; ++i) {
         output_resamplers_.push_back(
             new PushSincResampler(proc_num_frames_,
                                   output_num_frames_));
@@ -122,36 +94,43 @@ AudioBuffer::AudioBuffer(int input_num_frames,
     split_data_.reset(new IFChannelBuffer(proc_num_frames_,
                                           num_proc_channels_,
                                           num_bands_));
-    splitting_filter_.reset(new SplittingFilter(num_proc_channels_));
+    splitting_filter_.reset(new SplittingFilter(num_proc_channels_,
+                                                num_bands_,
+                                                proc_num_frames_));
   }
 }
 
 AudioBuffer::~AudioBuffer() {}
 
 void AudioBuffer::CopyFrom(const float* const* data,
-                           int num_frames,
-                           AudioProcessing::ChannelLayout layout) {
-  assert(num_frames == input_num_frames_);
-  assert(ChannelsFromLayout(layout) == num_input_channels_);
+                           const StreamConfig& stream_config) {
+  assert(stream_config.num_frames() == input_num_frames_);
+  assert(stream_config.num_channels() == num_input_channels_);
   InitForNewData();
+  // Initialized lazily because there's a different condition in
+  // DeinterleaveFrom.
+  const bool need_to_downmix =
+      num_input_channels_ > 1 && num_proc_channels_ == 1;
+  if (need_to_downmix && !input_buffer_) {
+    input_buffer_.reset(
+        new IFChannelBuffer(input_num_frames_, num_proc_channels_));
+  }
 
-  if (HasKeyboardChannel(layout)) {
-    keyboard_data_ = data[KeyboardChannelIndex(layout)];
+  if (stream_config.has_keyboard()) {
+    keyboard_data_ = data[KeyboardChannelIndex(stream_config)];
   }
 
   // Downmix.
   const float* const* data_ptr = data;
-  if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
-    StereoToMono(data[0],
-                 data[1],
-                 input_buffer_->channels()[0],
-                 input_num_frames_);
-    data_ptr = input_buffer_->channels();
+  if (need_to_downmix) {
+    DownmixToMono<float, float>(data, input_num_frames_, num_input_channels_,
+                                input_buffer_->fbuf()->channels()[0]);
+    data_ptr = input_buffer_->fbuf_const()->channels();
   }
 
   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_proc_channels_; ++i) {
+    for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(data_ptr[i],
                                      input_num_frames_,
                                      process_buffer_->channels()[i],
@@ -161,18 +140,17 @@ void AudioBuffer::CopyFrom(const float* const* data,
   }
 
   // Convert to the S16 range.
-  for (int i = 0; i < num_proc_channels_; ++i) {
+  for (size_t i = 0; i < num_proc_channels_; ++i) {
     FloatToFloatS16(data_ptr[i],
                     proc_num_frames_,
                     data_->fbuf()->channels()[i]);
   }
 }
 
-void AudioBuffer::CopyTo(int num_frames,
-                         AudioProcessing::ChannelLayout layout,
+void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                          float* const* data) {
-  assert(num_frames == output_num_frames_);
-  assert(ChannelsFromLayout(layout) == num_channels_);
+  assert(stream_config.num_frames() == output_num_frames_);
+  assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);
 
   // Convert to the float range.
   float* const* data_ptr = data;
@@ -180,7 +158,7 @@ void AudioBuffer::CopyTo(int num_frames,
     // Convert to an intermediate buffer for subsequent resampling.
     data_ptr = process_buffer_->channels();
   }
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     FloatS16ToFloat(data_->fbuf()->channels()[i],
                     proc_num_frames_,
                     data_ptr[i]);
@@ -188,13 +166,18 @@ void AudioBuffer::CopyTo(int num_frames,
 
   // Resample.
   if (output_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_channels_; ++i) {
+    for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(data_ptr[i],
                                       proc_num_frames_,
                                       data[i],
                                       output_num_frames_);
     }
   }
+
+  // Upmix.
+  for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
+    memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
+  }
 }
 
 void AudioBuffer::InitForNewData() {
@@ -214,13 +197,13 @@ int16_t* const* AudioBuffer::channels() {
   return data_->ibuf()->channels();
 }
 
-const int16_t* const* AudioBuffer::split_bands_const(int channel) const {
+const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
   return split_data_.get() ?
          split_data_->ibuf_const()->bands(channel) :
          data_->ibuf_const()->bands(channel);
 }
 
-int16_t* const* AudioBuffer::split_bands(int channel) {
+int16_t* const* AudioBuffer::split_bands(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->ibuf()->bands(channel) :
@@ -271,13 +254,13 @@ float* const* AudioBuffer::channels_f() {
   return data_->fbuf()->channels();
 }
 
-const float* const* AudioBuffer::split_bands_const_f(int channel) const {
+const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
   return split_data_.get() ?
          split_data_->fbuf_const()->bands(channel) :
          data_->fbuf_const()->bands(channel);
 }
 
-float* const* AudioBuffer::split_bands_f(int channel) {
+float* const* AudioBuffer::split_bands_f(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->fbuf()->bands(channel) :
@@ -320,9 +303,6 @@ const ChannelBuffer<float>* AudioBuffer::split_data_f() const {
 }
 
 const int16_t* AudioBuffer::mixed_low_pass_data() {
-  // Currently only mixing stereo to mono is supported.
-  assert(num_proc_channels_ == 1 || num_proc_channels_ == 2);
-
   if (num_proc_channels_ == 1) {
     return split_bands_const(0)[kBand0To8kHz];
   }
@@ -332,10 +312,10 @@ const int16_t* AudioBuffer::mixed_low_pass_data() {
       mixed_low_pass_channels_.reset(
           new ChannelBuffer<int16_t>(num_split_frames_, 1));
     }
-    StereoToMono(split_bands_const(0)[kBand0To8kHz],
-                 split_bands_const(1)[kBand0To8kHz],
-                 mixed_low_pass_channels_->channels()[0],
-                 num_split_frames_);
+
+    DownmixToMono<int16_t, int32_t>(split_channels_const(kBand0To8kHz),
+                                    num_split_frames_, num_channels_,
+                                    mixed_low_pass_channels_->channels()[0]);
     mixed_low_pass_valid_ = true;
   }
   return mixed_low_pass_channels_->channels()[0];
@@ -361,78 +341,102 @@ AudioFrame::VADActivity AudioBuffer::activity() const {
   return activity_;
 }
 
-int AudioBuffer::num_channels() const {
+size_t AudioBuffer::num_channels() const {
   return num_channels_;
 }
 
-void AudioBuffer::set_num_channels(int num_channels) {
+void AudioBuffer::set_num_channels(size_t num_channels) {
   num_channels_ = num_channels;
 }
 
-int AudioBuffer::num_frames() const {
+size_t AudioBuffer::num_frames() const {
   return proc_num_frames_;
 }
 
-int AudioBuffer::num_frames_per_band() const {
+size_t AudioBuffer::num_frames_per_band() const {
   return num_split_frames_;
 }
 
-int AudioBuffer::num_keyboard_frames() const {
+size_t AudioBuffer::num_keyboard_frames() const {
   // We don't resample the keyboard channel.
   return input_num_frames_;
 }
 
-int AudioBuffer::num_bands() const {
+size_t AudioBuffer::num_bands() const {
   return num_bands_;
 }
 
-// TODO(andrew): Do deinterleaving and mixing in one step?
+// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
 void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
-  assert(proc_num_frames_ == input_num_frames_);
   assert(frame->num_channels_ == num_input_channels_);
-  assert(frame->samples_per_channel_ ==  proc_num_frames_);
+  assert(frame->samples_per_channel_ == input_num_frames_);
   InitForNewData();
+  // Initialized lazily because there's a different condition in CopyFrom.
+  if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
+    input_buffer_.reset(
+        new IFChannelBuffer(input_num_frames_, num_proc_channels_));
+  }
   activity_ = frame->vad_activity_;
 
-  if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
-    // Downmix directly; no explicit deinterleaving needed.
-    int16_t* downmixed = data_->ibuf()->channels()[0];
-    for (int i = 0; i < input_num_frames_; ++i) {
-      downmixed[i] = (frame->data_[i * 2] + frame->data_[i * 2 + 1]) / 2;
-    }
+  int16_t* const* deinterleaved;
+  if (input_num_frames_ == proc_num_frames_) {
+    deinterleaved = data_->ibuf()->channels();
+  } else {
+    deinterleaved = input_buffer_->ibuf()->channels();
+  }
+  if (num_proc_channels_ == 1) {
+    // Downmix and deinterleave simultaneously.
+    DownmixInterleavedToMono(frame->data_, input_num_frames_,
+                             num_input_channels_, deinterleaved[0]);
   } else {
     assert(num_proc_channels_ == num_input_channels_);
-    int16_t* interleaved = frame->data_;
-    for (int i = 0; i < num_proc_channels_; ++i) {
-      int16_t* deinterleaved = data_->ibuf()->channels()[i];
-      int interleaved_idx = i;
-      for (int j = 0; j < proc_num_frames_; ++j) {
-        deinterleaved[j] = interleaved[interleaved_idx];
-        interleaved_idx += num_proc_channels_;
-      }
+    Deinterleave(frame->data_,
+                 input_num_frames_,
+                 num_proc_channels_,
+                 deinterleaved);
+  }
+
+  // Resample.
+  if (input_num_frames_ != proc_num_frames_) {
+    for (size_t i = 0; i < num_proc_channels_; ++i) {
+      input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
+                                     input_num_frames_,
+                                     data_->fbuf()->channels()[i],
+                                     proc_num_frames_);
     }
   }
 }
 
-void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
-  assert(proc_num_frames_ == output_num_frames_);
-  assert(num_channels_ == num_input_channels_);
-  assert(frame->num_channels_ == num_channels_);
-  assert(frame->samples_per_channel_ == proc_num_frames_);
+void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
   frame->vad_activity_ = activity_;
-
   if (!data_changed) {
     return;
   }
 
-  int16_t* interleaved = frame->data_;
-  for (int i = 0; i < num_channels_; i++) {
-    int16_t* deinterleaved = data_->ibuf()->channels()[i];
-    int interleaved_idx = i;
-    for (int j = 0; j < proc_num_frames_; j++) {
-      interleaved[interleaved_idx] = deinterleaved[j];
-      interleaved_idx += num_channels_;
+  assert(frame->num_channels_ == num_channels_ || num_channels_ == 1);
+  assert(frame->samples_per_channel_ == output_num_frames_);
+
+  // Resample if necessary.
+  IFChannelBuffer* data_ptr = data_.get();
+  if (proc_num_frames_ != output_num_frames_) {
+    if (!output_buffer_) {
+      output_buffer_.reset(
+          new IFChannelBuffer(output_num_frames_, num_channels_));
     }
+    for (size_t i = 0; i < num_channels_; ++i) {
+      output_resamplers_[i]->Resample(
+          data_->fbuf()->channels()[i], proc_num_frames_,
+          output_buffer_->fbuf()->channels()[i], output_num_frames_);
+    }
+    data_ptr = output_buffer_.get();
+  }
+
+  if (frame->num_channels_ == num_channels_) {
+    Interleave(data_ptr->ibuf()->channels(), proc_num_frames_, num_channels_,
+               frame->data_);
+  } else {
+    UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], proc_num_frames_,
+                           frame->num_channels_, frame->data_);
   }
 }
 
@@ -444,7 +448,7 @@ void AudioBuffer::CopyLowPassToReference() {
         new ChannelBuffer<int16_t>(num_split_frames_,
                                    num_proc_channels_));
   }
-  for (int i = 0; i < num_proc_channels_; i++) {
+  for (size_t i = 0; i < num_proc_channels_; i++) {
     memcpy(low_pass_reference_channels_->channels()[i],
            split_bands_const(i)[kBand0To8kHz],
            low_pass_reference_channels_->num_frames_per_band() *
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.h
index eb45fb2a7a..ff12ca2d95 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_buffer.h
@@ -11,15 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
 
-#include <vector>
-
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -36,19 +33,19 @@ enum Band {
 class AudioBuffer {
  public:
   // TODO(ajm): Switch to take ChannelLayouts.
-  AudioBuffer(int input_num_frames,
-              int num_input_channels,
-              int process_num_frames,
-              int num_process_channels,
-              int output_num_frames);
+  AudioBuffer(size_t input_num_frames,
+              size_t num_input_channels,
+              size_t process_num_frames,
+              size_t num_process_channels,
+              size_t output_num_frames);
   virtual ~AudioBuffer();
 
-  int num_channels() const;
-  void set_num_channels(int num_channels);
-  int num_frames() const;
-  int num_frames_per_band() const;
-  int num_keyboard_frames() const;
-  int num_bands() const;
+  size_t num_channels() const;
+  void set_num_channels(size_t num_channels);
+  size_t num_frames() const;
+  size_t num_frames_per_band() const;
+  size_t num_keyboard_frames() const;
+  size_t num_bands() const;
 
   // Returns a pointer array to the full-band channels.
   // Usage:
@@ -68,10 +65,10 @@ class AudioBuffer {
   // 0 <= channel < |num_proc_channels_|
   // 0 <= band < |num_bands_|
   // 0 <= sample < |num_split_frames_|
-  int16_t* const* split_bands(int channel);
-  const int16_t* const* split_bands_const(int channel) const;
-  float* const* split_bands_f(int channel);
-  const float* const* split_bands_const_f(int channel) const;
+  int16_t* const* split_bands(size_t channel);
+  const int16_t* const* split_bands_const(size_t channel) const;
+  float* const* split_bands_f(size_t channel);
+  const float* const* split_bands_const_f(size_t channel) const;
 
   // Returns a pointer array to the channels for a specific band.
   // Usage:
@@ -112,15 +109,11 @@ class AudioBuffer {
   void DeinterleaveFrom(AudioFrame* audioFrame);
   // If |data_changed| is false, only the non-audio data members will be copied
   // to |frame|.
-  void InterleaveTo(AudioFrame* frame, bool data_changed) const;
+  void InterleaveTo(AudioFrame* frame, bool data_changed);
 
   // Use for float deinterleaved data.
-  void CopyFrom(const float* const* data,
-                int num_frames,
-                AudioProcessing::ChannelLayout layout);
-  void CopyTo(int num_frames,
-              AudioProcessing::ChannelLayout layout,
-              float* const* data);
+  void CopyFrom(const float* const* data, const StreamConfig& stream_config);
+  void CopyTo(const StreamConfig& stream_config, float* const* data);
   void CopyLowPassToReference();
 
   // Splits the signal into different bands.
@@ -134,20 +127,20 @@ class AudioBuffer {
 
   // The audio is passed into DeinterleaveFrom() or CopyFrom() with input
   // format (samples per channel and number of channels).
-  const int input_num_frames_;
-  const int num_input_channels_;
+  const size_t input_num_frames_;
+  const size_t num_input_channels_;
   // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
   // format.
-  const int proc_num_frames_;
-  const int num_proc_channels_;
+  const size_t proc_num_frames_;
+  const size_t num_proc_channels_;
   // The audio is returned by InterleaveTo() and CopyTo() with output samples
   // per channels and the current number of channels. This last one can be
   // changed at any time using set_num_channels().
-  const int output_num_frames_;
-  int num_channels_;
+  const size_t output_num_frames_;
+  size_t num_channels_;
 
-  int num_bands_;
-  int num_split_frames_;
+  size_t num_bands_;
+  size_t num_split_frames_;
   bool mixed_low_pass_valid_;
   bool reference_copied_;
   AudioFrame::VADActivity activity_;
@@ -158,7 +151,8 @@ class AudioBuffer {
   rtc::scoped_ptr<SplittingFilter> splitting_filter_;
   rtc::scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
-  rtc::scoped_ptr<ChannelBuffer<float> > input_buffer_;
+  rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
+  rtc::scoped_ptr<IFChannelBuffer> output_buffer_;
   rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
   ScopedVector<PushSincResampler> input_resamplers_;
   ScopedVector<PushSincResampler> output_resamplers_;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing.gypi b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing.gypi
index 22bbbebc75..ee72fbbe58 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing.gypi
@@ -7,6 +7,9 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
+  'includes': [
+    '../../build/common.gypi',
+  ],
   'variables': {
     'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets',
   },
@@ -28,7 +31,7 @@
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-#        '<(webrtc_root)/modules/modules.gyp:iSAC',
+#        '<(webrtc_root)/modules/modules.gyp:isac',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
@@ -41,24 +44,16 @@
         'aec/aec_resampler.h',
         'aec/echo_cancellation.c',
         'aec/echo_cancellation_internal.h',
-        'aec/include/echo_cancellation.h',
+        'aec/echo_cancellation.h',
         'aecm/aecm_core.c',
         'aecm/aecm_core.h',
         'aecm/echo_control_mobile.c',
-        'aecm/include/echo_control_mobile.h',
+        'aecm/echo_control_mobile.h',
         'agc/agc.cc',
         'agc/agc.h',
-        'agc/agc_audio_proc.cc',
-        'agc/agc_audio_proc.h',
-        'agc/agc_audio_proc_internal.h',
         'agc/agc_manager_direct.cc',
         'agc/agc_manager_direct.h',
-        'agc/circular_buffer.cc',
-        'agc/circular_buffer.h',
-        'agc/common.h',
         'agc/gain_map_internal.h',
-        'agc/gmm.cc',
-        'agc/gmm.h',
         'agc/histogram.cc',
         'agc/histogram.h',
         'agc/legacy/analog_agc.c',
@@ -66,22 +61,14 @@
         'agc/legacy/digital_agc.c',
         'agc/legacy/digital_agc.h',
         'agc/legacy/gain_control.h',
-        'agc/noise_gmm_tables.h',
-        'agc/pitch_based_vad.cc',
-        'agc/pitch_based_vad.h',
-        'agc/pitch_internal.cc',
-        'agc/pitch_internal.h',
-        'agc/pole_zero_filter.cc',
-        'agc/pole_zero_filter.h',
-        'agc/standalone_vad.cc',
-        'agc/standalone_vad.h',
         'agc/utility.cc',
         'agc/utility.h',
-        'agc/voice_gmm_tables.h',
         'audio_buffer.cc',
         'audio_buffer.h',
         'audio_processing_impl.cc',
         'audio_processing_impl.h',
+        'beamformer/array_util.cc',
+        'beamformer/array_util.h',
         'beamformer/beamformer.h',
         'beamformer/complex_matrix.h',
         'beamformer/covariance_matrix_generator.cc',
@@ -99,8 +86,15 @@
         'high_pass_filter_impl.cc',
         'high_pass_filter_impl.h',
         'include/audio_processing.h',
+        'intelligibility/intelligibility_enhancer.cc',
+        'intelligibility/intelligibility_enhancer.h',
+        'intelligibility/intelligibility_utils.cc',
+        'intelligibility/intelligibility_utils.h',
         'level_estimator_impl.cc',
         'level_estimator_impl.h',
+        'logging/aec_logging.h',
+        'logging/aec_logging_file_handling.cc',
+        'logging/aec_logging_file_handling.h',
         'noise_suppression_impl.cc',
         'noise_suppression_impl.h',
         'processing_component.cc',
@@ -109,6 +103,8 @@
         'rms_level.h',
         'splitting_filter.cc',
         'splitting_filter.h',
+        'three_band_filter_bank.cc',
+        'three_band_filter_bank.h',
         'transient/common.h',
         'transient/daubechies_8_wavelet_coeffs.h',
         'transient/dyadic_decimator.h',
@@ -129,6 +125,26 @@
         'utility/delay_estimator_internal.h',
         'utility/delay_estimator_wrapper.c',
         'utility/delay_estimator_wrapper.h',
+        'vad/common.h',
+        'vad/gmm.cc',
+        'vad/gmm.h',
+        'vad/noise_gmm_tables.h',
+        'vad/pitch_based_vad.cc',
+        'vad/pitch_based_vad.h',
+        'vad/pitch_internal.cc',
+        'vad/pitch_internal.h',
+        'vad/pole_zero_filter.cc',
+        'vad/pole_zero_filter.h',
+        'vad/standalone_vad.cc',
+        'vad/standalone_vad.h',
+        'vad/vad_audio_proc.cc',
+        'vad/vad_audio_proc.h',
+        'vad/vad_audio_proc_internal.h',
+        'vad/vad_circular_buffer.cc',
+        'vad/vad_circular_buffer.h',
+        'vad/voice_activity_detector.cc',
+        'vad/voice_activity_detector.h',
+        'vad/voice_gmm_tables.h',
         'voice_detection_impl.cc',
         'voice_detection_impl.h',
       ],
@@ -149,14 +165,14 @@
         ['prefer_fixed_point==1', {
           'defines': ['WEBRTC_NS_FIXED'],
           'sources': [
-            'ns/include/noise_suppression_x.h',
+            'ns/noise_suppression_x.h',
             'ns/noise_suppression_x.c',
             'ns/nsx_core.c',
             'ns/nsx_core.h',
             'ns/nsx_defines.h',
           ],
           'conditions': [
-            ['target_arch=="mipsel" and mips_arch_variant!="r6" and android_webview_build==0', {
+            ['target_arch=="mipsel" and mips_arch_variant!="r6"', {
               'sources': [
                 'ns/nsx_core_mips.c',
               ],
@@ -170,7 +186,7 @@
           'defines': ['WEBRTC_NS_FLOAT'],
           'sources': [
             'ns/defines.h',
-            'ns/include/noise_suppression.h',
+            'ns/noise_suppression.h',
             'ns/noise_suppression.c',
             'ns/ns_core.c',
             'ns/ns_core.h',
@@ -180,10 +196,10 @@
         ['target_arch=="ia32" or target_arch=="x64"', {
           'dependencies': ['audio_processing_sse2',],
         }],
-        ['(target_arch=="arm" and arm_version>=7) or target_arch=="arm64"', {
+        ['build_with_neon==1', {
           'dependencies': ['audio_processing_neon',],
         }],
-        ['target_arch=="mipsel" and mips_arch_variant!="r6" and android_webview_build==0', {
+        ['target_arch=="mipsel" and mips_arch_variant!="r6"', {
           'sources': [
             'aecm/aecm_core_mips.c',
           ],
@@ -232,19 +248,19 @@
             'aec/aec_core_sse2.c',
             'aec/aec_rdft_sse2.c',
           ],
-          'cflags': ['-msse2',],
           'conditions': [
-            [ 'os_posix == 1', {
+            ['os_posix==1', {
+              'cflags': [ '-msse2', ],
               'cflags_mozilla': ['-msse2',],
+              'xcode_settings': {
+                'OTHER_CFLAGS': [ '-msse2', ],
+              },
             }],
           ],
-          'xcode_settings': {
-            'OTHER_CFLAGS': ['-msse2',],
-          },
         },
       ],
     }],
-    ['(target_arch=="arm" and arm_version>=7) or target_arch=="arm64"', {
+    ['build_with_neon==1', {
       'targets': [{
         'target_name': 'audio_processing_neon',
         'type': 'static_library',
@@ -258,15 +274,6 @@
           'aecm/aecm_core_neon.c',
           'ns/nsx_core_neon.c',
         ],
-        'conditions': [
-          # Disable LTO in audio_processing_neon target due to compiler bug
-          ['use_lto==1', {
-            'cflags!': [
-              '-flto',
-              '-ffat-lto-objects',
-            ],
-          }],
-        ],
       }],
     }],
   ],
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.cc
index 1074b61828..744309c774 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -11,11 +11,18 @@
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
 #include <assert.h>
+#include <algorithm>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/platform_file.h"
-#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+}
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
@@ -24,15 +31,16 @@
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
@@ -43,15 +51,31 @@
 #endif
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
-#define RETURN_ON_ERR(expr)  \
-  do {                       \
-    int err = (expr);        \
-    if (err != kNoError) {   \
-      return err;            \
-    }                        \
+#define RETURN_ON_ERR(expr) \
+  do {                      \
+    int err = (expr);       \
+    if (err != kNoError) {  \
+      return err;           \
+    }                       \
   } while (0)
 
 namespace webrtc {
+namespace {
+
+static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
+  switch (layout) {
+    case AudioProcessing::kMono:
+    case AudioProcessing::kStereo:
+      return false;
+    case AudioProcessing::kMonoAndKeyboard:
+    case AudioProcessing::kStereoAndKeyboard:
+      return true;
+  }
+
+  assert(false);
+  return false;
+}
+}  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
 static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
@@ -70,9 +94,7 @@ static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
 class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
  public:
   explicit GainControlForNewAgc(GainControlImpl* gain_control)
-      : real_gain_control_(gain_control),
-        volume_(0) {
-  }
+      : real_gain_control_(gain_control), volume_(0) {}
 
   // GainControl implementation.
   int Enable(bool enable) override {
@@ -124,6 +146,46 @@ class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
   int volume_;
 };
 
+struct AudioProcessingImpl::ApmPublicSubmodules {
+  ApmPublicSubmodules()
+      : echo_cancellation(nullptr),
+        echo_control_mobile(nullptr),
+        gain_control(nullptr) {}
+  // Accessed externally of APM without any lock acquired.
+  EchoCancellationImpl* echo_cancellation;
+  EchoControlMobileImpl* echo_control_mobile;
+  GainControlImpl* gain_control;
+  rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
+  rtc::scoped_ptr<LevelEstimatorImpl> level_estimator;
+  rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
+  rtc::scoped_ptr<VoiceDetectionImpl> voice_detection;
+  rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
+
+  // Accessed internally from both render and capture.
+  rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
+  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
+};
+
+struct AudioProcessingImpl::ApmPrivateSubmodules {
+  explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
+      : beamformer(beamformer) {}
+  // Accessed internally from capture or during initialization
+  std::list<ProcessingComponent*> component_list;
+  rtc::scoped_ptr<Beamformer<float>> beamformer;
+  rtc::scoped_ptr<AgcManagerDirect> agc_manager;
+};
+
+const int AudioProcessing::kNativeSampleRatesHz[] = {
+    AudioProcessing::kSampleRate8kHz,
+    AudioProcessing::kSampleRate16kHz,
+    AudioProcessing::kSampleRate32kHz,
+    AudioProcessing::kSampleRate48kHz};
+const size_t AudioProcessing::kNumNativeSampleRates =
+    arraysize(AudioProcessing::kNativeSampleRatesHz);
+const int AudioProcessing::kMaxNativeSampleRateHz = AudioProcessing::
+    kNativeSampleRatesHz[AudioProcessing::kNumNativeSampleRates - 1];
+const int AudioProcessing::kMaxAECMSampleRateHz = kSampleRate16kHz;
+
 AudioProcessing* AudioProcessing::Create() {
   Config config;
   return Create(config, nullptr);
@@ -138,7 +200,7 @@ AudioProcessing* AudioProcessing::Create(const Config& config,
   AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
   if (apm->Initialize() != kNoError) {
     delete apm;
-    apm = NULL;
+    apm = nullptr;
   }
 
   return apm;
@@ -149,156 +211,193 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config)
 
 AudioProcessingImpl::AudioProcessingImpl(const Config& config,
                                          Beamformer<float>* beamformer)
-    : echo_cancellation_(NULL),
-      echo_control_mobile_(NULL),
-      gain_control_(NULL),
-      high_pass_filter_(NULL),
-      level_estimator_(NULL),
-      noise_suppression_(NULL),
-      voice_detection_(NULL),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-      debug_file_(FileWrapper::Create()),
-      event_msg_(new audioproc::Event()),
-#endif
-      fwd_in_format_(kSampleRate16kHz, 1),
-      fwd_proc_format_(kSampleRate16kHz),
-      fwd_out_format_(kSampleRate16kHz, 1),
-      rev_in_format_(kSampleRate16kHz, 1),
-      rev_proc_format_(kSampleRate16kHz, 1),
-      split_rate_(kSampleRate16kHz),
-      stream_delay_ms_(0),
-      delay_offset_ms_(0),
-      was_stream_delay_set_(false),
-      output_will_be_muted_(false),
-      key_pressed_(false),
+    : public_submodules_(new ApmPublicSubmodules()),
+      private_submodules_(new ApmPrivateSubmodules(beamformer)),
+      constants_(config.Get<ExperimentalAgc>().startup_min_volume,
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-      use_new_agc_(false),
+                 false,
 #else
-      use_new_agc_(config.Get<ExperimentalAgc>().enabled),
+                 config.Get<ExperimentalAgc>().enabled,
 #endif
-      transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
-      beamformer_enabled_(config.Get<Beamforming>().enabled),
-      beamformer_(beamformer),
-      array_geometry_(config.Get<Beamforming>().array_geometry),
-      supports_48kHz_(config.Get<AudioProcessing48kHzSupport>().enabled) {
-  echo_cancellation_ = new EchoCancellationImpl(this, crit_);
-  component_list_.push_back(echo_cancellation_);
+                 config.Get<Intelligibility>().enabled),
 
-  echo_control_mobile_ = new EchoControlMobileImpl(this, crit_);
-  component_list_.push_back(echo_control_mobile_);
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+      capture_(false,
+#else
+      capture_(config.Get<ExperimentalNs>().enabled,
+#endif
+               config.Get<Beamforming>().array_geometry,
+               config.Get<Beamforming>().target_direction),
+      capture_nonlocked_(config.Get<Beamforming>().enabled)
+{
+  {
+    rtc::CritScope cs_render(&crit_render_);
+    rtc::CritScope cs_capture(&crit_capture_);
 
-  gain_control_ = new GainControlImpl(this, crit_);
-  component_list_.push_back(gain_control_);
+    public_submodules_->echo_cancellation =
+        new EchoCancellationImpl(this, &crit_render_, &crit_capture_);
+    public_submodules_->echo_control_mobile =
+        new EchoControlMobileImpl(this, &crit_render_, &crit_capture_);
+    public_submodules_->gain_control =
+        new GainControlImpl(this, &crit_capture_, &crit_capture_);
+    public_submodules_->high_pass_filter.reset(
+        new HighPassFilterImpl(&crit_capture_));
+    public_submodules_->level_estimator.reset(
+        new LevelEstimatorImpl(&crit_capture_));
+    public_submodules_->noise_suppression.reset(
+        new NoiseSuppressionImpl(&crit_capture_));
+    public_submodules_->voice_detection.reset(
+        new VoiceDetectionImpl(&crit_capture_));
+    public_submodules_->gain_control_for_new_agc.reset(
+        new GainControlForNewAgc(public_submodules_->gain_control));
 
-  high_pass_filter_ = new HighPassFilterImpl(this, crit_);
-  component_list_.push_back(high_pass_filter_);
-
-  level_estimator_ = new LevelEstimatorImpl(this, crit_);
-  component_list_.push_back(level_estimator_);
-
-  noise_suppression_ = new NoiseSuppressionImpl(this, crit_);
-  component_list_.push_back(noise_suppression_);
-
-  voice_detection_ = new VoiceDetectionImpl(this, crit_);
-  component_list_.push_back(voice_detection_);
-
-  gain_control_for_new_agc_.reset(new GainControlForNewAgc(gain_control_));
+    private_submodules_->component_list.push_back(
+        public_submodules_->echo_cancellation);
+    private_submodules_->component_list.push_back(
+        public_submodules_->echo_control_mobile);
+    private_submodules_->component_list.push_back(
+        public_submodules_->gain_control);
+  }
 
   SetExtraOptions(config);
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
-  {
-    CriticalSectionScoped crit_scoped(crit_);
-    // Depends on gain_control_ and gain_control_for_new_agc_.
-    agc_manager_.reset();
-    // Depends on gain_control_.
-    gain_control_for_new_agc_.reset();
-    while (!component_list_.empty()) {
-      ProcessingComponent* component = component_list_.front();
-      component->Destroy();
-      delete component;
-      component_list_.pop_front();
-    }
+  // Depends on gain_control_ and
+  // public_submodules_->gain_control_for_new_agc.
+  private_submodules_->agc_manager.reset();
+  // Depends on gain_control_.
+  public_submodules_->gain_control_for_new_agc.reset();
+  while (!private_submodules_->component_list.empty()) {
+    ProcessingComponent* component =
+        private_submodules_->component_list.front();
+    component->Destroy();
+    delete component;
+    private_submodules_->component_list.pop_front();
+  }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-    if (debug_file_->Open()) {
-      debug_file_->CloseFile();
-    }
-#endif
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.debug_file->CloseFile();
   }
-  delete crit_;
-  crit_ = NULL;
+#endif
 }
 
 int AudioProcessingImpl::Initialize() {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   return InitializeLocked();
 }
 
-int AudioProcessingImpl::set_sample_rate_hz(int rate) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return InitializeLocked(rate,
-                          rate,
-                          rev_in_format_.rate(),
-                          fwd_in_format_.num_channels(),
-                          fwd_out_format_.num_channels(),
-                          rev_in_format_.num_channels());
-}
-
 int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
                                     int output_sample_rate_hz,
                                     int reverse_sample_rate_hz,
                                     ChannelLayout input_layout,
                                     ChannelLayout output_layout,
                                     ChannelLayout reverse_layout) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return InitializeLocked(input_sample_rate_hz,
-                          output_sample_rate_hz,
-                          reverse_sample_rate_hz,
-                          ChannelsFromLayout(input_layout),
-                          ChannelsFromLayout(output_layout),
-                          ChannelsFromLayout(reverse_layout));
+  const ProcessingConfig processing_config = {
+      {{input_sample_rate_hz,
+        ChannelsFromLayout(input_layout),
+        LayoutHasKeyboard(input_layout)},
+       {output_sample_rate_hz,
+        ChannelsFromLayout(output_layout),
+        LayoutHasKeyboard(output_layout)},
+       {reverse_sample_rate_hz,
+        ChannelsFromLayout(reverse_layout),
+        LayoutHasKeyboard(reverse_layout)},
+       {reverse_sample_rate_hz,
+        ChannelsFromLayout(reverse_layout),
+        LayoutHasKeyboard(reverse_layout)}}};
+
+  return Initialize(processing_config);
+}
+
+int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
+  // Run in a single-threaded manner during initialization.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  return InitializeLocked(processing_config);
+}
+
+int AudioProcessingImpl::MaybeInitializeRender(
+    const ProcessingConfig& processing_config) {
+  return MaybeInitialize(processing_config);
+}
+
+int AudioProcessingImpl::MaybeInitializeCapture(
+    const ProcessingConfig& processing_config) {
+  return MaybeInitialize(processing_config);
+}
+
+// Calls InitializeLocked() if any of the audio parameters have changed from
+// their current values (needs to be called while holding the crit_render_lock).
+int AudioProcessingImpl::MaybeInitialize(
+    const ProcessingConfig& processing_config) {
+  // Called from both threads. Thread check is therefore not possible.
+  if (processing_config == formats_.api_format) {
+    return kNoError;
+  }
+
+  rtc::CritScope cs_capture(&crit_capture_);
+  return InitializeLocked(processing_config);
 }
 
 int AudioProcessingImpl::InitializeLocked() {
-  const int fwd_audio_buffer_channels = beamformer_enabled_ ?
-                                        fwd_in_format_.num_channels() :
-                                        fwd_out_format_.num_channels();
-  render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(),
-                                      rev_in_format_.num_channels(),
-                                      rev_proc_format_.samples_per_channel(),
-                                      rev_proc_format_.num_channels(),
-                                      rev_proc_format_.samples_per_channel()));
-  capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(),
-                                       fwd_in_format_.num_channels(),
-                                       fwd_proc_format_.samples_per_channel(),
-                                       fwd_audio_buffer_channels,
-                                       fwd_out_format_.samples_per_channel()));
+  const int fwd_audio_buffer_channels =
+      capture_nonlocked_.beamformer_enabled
+          ? formats_.api_format.input_stream().num_channels()
+          : formats_.api_format.output_stream().num_channels();
+  const int rev_audio_buffer_out_num_frames =
+      formats_.api_format.reverse_output_stream().num_frames() == 0
+          ? formats_.rev_proc_format.num_frames()
+          : formats_.api_format.reverse_output_stream().num_frames();
+  if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
+    render_.render_audio.reset(new AudioBuffer(
+        formats_.api_format.reverse_input_stream().num_frames(),
+        formats_.api_format.reverse_input_stream().num_channels(),
+        formats_.rev_proc_format.num_frames(),
+        formats_.rev_proc_format.num_channels(),
+        rev_audio_buffer_out_num_frames));
+    if (rev_conversion_needed()) {
+      render_.render_converter = AudioConverter::Create(
+          formats_.api_format.reverse_input_stream().num_channels(),
+          formats_.api_format.reverse_input_stream().num_frames(),
+          formats_.api_format.reverse_output_stream().num_channels(),
+          formats_.api_format.reverse_output_stream().num_frames());
+    } else {
+      render_.render_converter.reset(nullptr);
+    }
+  } else {
+    render_.render_audio.reset(nullptr);
+    render_.render_converter.reset(nullptr);
+  }
+  capture_.capture_audio.reset(
+      new AudioBuffer(formats_.api_format.input_stream().num_frames(),
+                      formats_.api_format.input_stream().num_channels(),
+                      capture_nonlocked_.fwd_proc_format.num_frames(),
+                      fwd_audio_buffer_channels,
+                      formats_.api_format.output_stream().num_frames()));
 
   // Initialize all components.
-  for (auto item : component_list_) {
+  for (auto item : private_submodules_->component_list) {
     int err = item->Initialize();
     if (err != kNoError) {
       return err;
     }
   }
 
-  int err = InitializeExperimentalAgc();
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = InitializeTransient();
-  if (err != kNoError) {
-    return err;
-  }
-
+  InitializeExperimentalAgc();
+  InitializeTransient();
   InitializeBeamformer();
+  InitializeIntelligibility();
+  InitializeHighPassFilter();
+  InitializeNoiseSuppression();
+  InitializeLevelEstimator();
+  InitializeVoiceDetection();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
+  if (debug_dump_.debug_file->Open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
@@ -309,62 +408,57 @@ int AudioProcessingImpl::InitializeLocked() {
   return kNoError;
 }
 
-int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
-                                          int output_sample_rate_hz,
-                                          int reverse_sample_rate_hz,
-                                          int num_input_channels,
-                                          int num_output_channels,
-                                          int num_reverse_channels) {
-  if (input_sample_rate_hz <= 0 ||
-      output_sample_rate_hz <= 0 ||
-      reverse_sample_rate_hz <= 0) {
-    return kBadSampleRateError;
+int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
+  for (const auto& stream : config.streams) {
+    if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
+      return kBadSampleRateError;
+    }
   }
-  if (num_output_channels > num_input_channels) {
-    return kBadNumberChannelsError;
-  }
-  // Only mono and stereo supported currently.
-  if (num_input_channels > 2 || num_input_channels < 1 ||
-      num_output_channels > 2 || num_output_channels < 1 ||
-      num_reverse_channels > 2 || num_reverse_channels < 1) {
-    return kBadNumberChannelsError;
-  }
-  if (beamformer_enabled_ &&
-      (static_cast<size_t>(num_input_channels) != array_geometry_.size() ||
-       num_output_channels > 1)) {
+
+  const size_t num_in_channels = config.input_stream().num_channels();
+  const size_t num_out_channels = config.output_stream().num_channels();
+
+  // Need at least one input channel.
+  // Need either one output channel or as many outputs as there are inputs.
+  if (num_in_channels == 0 ||
+      !(num_out_channels == 1 || num_out_channels == num_in_channels)) {
     return kBadNumberChannelsError;
   }
 
-  fwd_in_format_.set(input_sample_rate_hz, num_input_channels);
-  fwd_out_format_.set(output_sample_rate_hz, num_output_channels);
-  rev_in_format_.set(reverse_sample_rate_hz, num_reverse_channels);
+  if (capture_nonlocked_.beamformer_enabled &&
+      num_in_channels != capture_.array_geometry.size()) {
+    return kBadNumberChannelsError;
+  }
+
+  formats_.api_format = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
-  int min_proc_rate = std::min(fwd_in_format_.rate(), fwd_out_format_.rate());
+  const int min_proc_rate =
+      std::min(formats_.api_format.input_stream().sample_rate_hz(),
+               formats_.api_format.output_stream().sample_rate_hz());
   int fwd_proc_rate;
-  if (supports_48kHz_ && min_proc_rate > kSampleRate32kHz) {
-    fwd_proc_rate = kSampleRate48kHz;
-  } else if (min_proc_rate > kSampleRate16kHz) {
-    fwd_proc_rate = kSampleRate32kHz;
-  } else if (min_proc_rate > kSampleRate8kHz) {
-    fwd_proc_rate = kSampleRate16kHz;
-  } else {
-    fwd_proc_rate = kSampleRate8kHz;
+  for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
+    fwd_proc_rate = kNativeSampleRatesHz[i];
+    if (fwd_proc_rate >= min_proc_rate) {
+      break;
+    }
   }
   // ...with one exception.
-  if (echo_control_mobile_->is_enabled() && min_proc_rate > kSampleRate16kHz) {
-    fwd_proc_rate = kSampleRate16kHz;
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
+      min_proc_rate > kMaxAECMSampleRateHz) {
+    fwd_proc_rate = kMaxAECMSampleRateHz;
   }
 
-  fwd_proc_format_.set(fwd_proc_rate);
+  capture_nonlocked_.fwd_proc_format = StreamConfig(fwd_proc_rate);
 
   // We normally process the reverse stream at 16 kHz. Unless...
   int rev_proc_rate = kSampleRate16kHz;
-  if (fwd_proc_format_.rate() == kSampleRate8kHz) {
+  if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
     // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    if (rev_in_format_.rate() == kSampleRate32kHz) {
+    if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
+        kSampleRate32kHz) {
       // ...or the input is at 32 kHz, in which case we use the splitting
       // filter rather than the resampler.
       rev_proc_rate = kSampleRate32kHz;
@@ -373,144 +467,187 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
 
   // Always downmix the reverse stream to mono for analysis. This has been
   // demonstrated to work well for AEC in most practical scenarios.
-  rev_proc_format_.set(rev_proc_rate, 1);
+  formats_.rev_proc_format = StreamConfig(rev_proc_rate, 1);
 
-  if (fwd_proc_format_.rate() == kSampleRate32kHz ||
-      fwd_proc_format_.rate() == kSampleRate48kHz) {
-    split_rate_ = kSampleRate16kHz;
+  if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate32kHz ||
+      capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) {
+    capture_nonlocked_.split_rate = kSampleRate16kHz;
   } else {
-    split_rate_ = fwd_proc_format_.rate();
+    capture_nonlocked_.split_rate =
+        capture_nonlocked_.fwd_proc_format.sample_rate_hz();
   }
 
   return InitializeLocked();
 }
 
-// Calls InitializeLocked() if any of the audio parameters have changed from
-// their current values.
-int AudioProcessingImpl::MaybeInitializeLocked(int input_sample_rate_hz,
-                                               int output_sample_rate_hz,
-                                               int reverse_sample_rate_hz,
-                                               int num_input_channels,
-                                               int num_output_channels,
-                                               int num_reverse_channels) {
-  if (input_sample_rate_hz == fwd_in_format_.rate() &&
-      output_sample_rate_hz == fwd_out_format_.rate() &&
-      reverse_sample_rate_hz == rev_in_format_.rate() &&
-      num_input_channels == fwd_in_format_.num_channels() &&
-      num_output_channels == fwd_out_format_.num_channels() &&
-      num_reverse_channels == rev_in_format_.num_channels()) {
-    return kNoError;
-  }
-  return InitializeLocked(input_sample_rate_hz,
-                          output_sample_rate_hz,
-                          reverse_sample_rate_hz,
-                          num_input_channels,
-                          num_output_channels,
-                          num_reverse_channels);
-}
-
 void AudioProcessingImpl::SetExtraOptions(const Config& config) {
-  CriticalSectionScoped crit_scoped(crit_);
-  for (auto item : component_list_) {
+  // Run in a single-threaded manner when setting the extra options.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  for (auto item : private_submodules_->component_list) {
     item->SetExtraOptions(config);
   }
 
-  if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) {
-    transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled;
+  if (capture_.transient_suppressor_enabled !=
+      config.Get<ExperimentalNs>().enabled) {
+    capture_.transient_suppressor_enabled =
+        config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
+
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+  if (capture_nonlocked_.beamformer_enabled !=
+          config.Get<Beamforming>().enabled) {
+    capture_nonlocked_.beamformer_enabled = config.Get<Beamforming>().enabled;
+    if (config.Get<Beamforming>().array_geometry.size() > 1) {
+      capture_.array_geometry = config.Get<Beamforming>().array_geometry;
+    }
+    capture_.target_direction = config.Get<Beamforming>().target_direction;
+    InitializeBeamformer();
+  }
+#endif  // WEBRTC_ANDROID_PLATFORM_BUILD
 }
 
 int AudioProcessingImpl::input_sample_rate_hz() const {
-  CriticalSectionScoped crit_scoped(crit_);
-  return fwd_in_format_.rate();
-}
-
-int AudioProcessingImpl::sample_rate_hz() const {
-  CriticalSectionScoped crit_scoped(crit_);
-  return fwd_in_format_.rate();
+  // Accessed from outside APM, hence a lock is needed.
+  rtc::CritScope cs(&crit_capture_);
+  return formats_.api_format.input_stream().sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_sample_rate_hz() const {
-  return fwd_proc_format_.rate();
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
-  return split_rate_;
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.split_rate;
 }
 
-int AudioProcessingImpl::num_reverse_channels() const {
-  return rev_proc_format_.num_channels();
+size_t AudioProcessingImpl::num_reverse_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.rev_proc_format.num_channels();
 }
 
-int AudioProcessingImpl::num_input_channels() const {
-  return fwd_in_format_.num_channels();
+size_t AudioProcessingImpl::num_input_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.api_format.input_stream().num_channels();
 }
 
-int AudioProcessingImpl::num_output_channels() const {
-  return fwd_out_format_.num_channels();
+size_t AudioProcessingImpl::num_proc_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.beamformer_enabled ? 1 : num_output_channels();
+}
+
+size_t AudioProcessingImpl::num_output_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.api_format.output_stream().num_channels();
 }
 
 void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
-  CriticalSectionScoped lock(crit_);
-  output_will_be_muted_ = muted;
-  if (agc_manager_.get()) {
-    agc_manager_->SetCaptureMuted(output_will_be_muted_);
+  rtc::CritScope cs(&crit_capture_);
+  capture_.output_will_be_muted = muted;
+  if (private_submodules_->agc_manager.get()) {
+    private_submodules_->agc_manager->SetCaptureMuted(
+        capture_.output_will_be_muted);
   }
 }
 
-bool AudioProcessingImpl::output_will_be_muted() const {
-  CriticalSectionScoped lock(crit_);
-  return output_will_be_muted_;
-}
 
 int AudioProcessingImpl::ProcessStream(const float* const* src,
-                                       int samples_per_channel,
+                                       size_t samples_per_channel,
                                        int input_sample_rate_hz,
                                        ChannelLayout input_layout,
                                        int output_sample_rate_hz,
                                        ChannelLayout output_layout,
                                        float* const* dest) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (!src || !dest) {
-    return kNullPointerError;
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_ChannelLayout");
+  StreamConfig input_stream;
+  StreamConfig output_stream;
+  {
+    // Access the formats_.api_format.input_stream beneath the capture lock.
+    // The lock must be released as it is later required in the call
+    // to ProcessStream(,,,);
+    rtc::CritScope cs(&crit_capture_);
+    input_stream = formats_.api_format.input_stream();
+    output_stream = formats_.api_format.output_stream();
   }
 
-  RETURN_ON_ERR(MaybeInitializeLocked(input_sample_rate_hz,
-                                      output_sample_rate_hz,
-                                      rev_in_format_.rate(),
-                                      ChannelsFromLayout(input_layout),
-                                      ChannelsFromLayout(output_layout),
-                                      rev_in_format_.num_channels()));
-  if (samples_per_channel != fwd_in_format_.samples_per_channel()) {
+  input_stream.set_sample_rate_hz(input_sample_rate_hz);
+  input_stream.set_num_channels(ChannelsFromLayout(input_layout));
+  input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
+  output_stream.set_sample_rate_hz(output_sample_rate_hz);
+  output_stream.set_num_channels(ChannelsFromLayout(output_layout));
+  output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
+
+  if (samples_per_channel != input_stream.num_frames()) {
     return kBadDataLengthError;
   }
+  return ProcessStream(src, input_stream, output_stream, dest);
+}
+
+int AudioProcessingImpl::ProcessStream(const float* const* src,
+                                       const StreamConfig& input_config,
+                                       const StreamConfig& output_config,
+                                       float* const* dest) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
+  ProcessingConfig processing_config;
+  {
+    // Acquire the capture lock in order to safely call the function
+    // that retrieves the render side data. This function accesses apm
+    // getters that need the capture lock held when being called.
+    rtc::CritScope cs_capture(&crit_capture_);
+    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    public_submodules_->echo_control_mobile->ReadQueuedRenderData();
+    public_submodules_->gain_control->ReadQueuedRenderData();
+
+    if (!src || !dest) {
+      return kNullPointerError;
+    }
+
+    processing_config = formats_.api_format;
+  }
+
+  processing_config.input_stream() = input_config;
+  processing_config.output_stream() = output_config;
+
+  {
+    // Do conditional reinitialization.
+    rtc::CritScope cs_render(&crit_render_);
+    RETURN_ON_ERR(MaybeInitializeCapture(processing_config));
+  }
+  rtc::CritScope cs_capture(&crit_capture_);
+  assert(processing_config.input_stream().num_frames() ==
+         formats_.api_format.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    RETURN_ON_ERR(WriteConfigMessage(false));
+
+    debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * fwd_in_format_.samples_per_channel();
-    for (int i = 0; i < fwd_in_format_.num_channels(); ++i)
+        sizeof(float) * formats_.api_format.input_stream().num_frames();
+    for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
+         ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
-  capture_audio_->CopyFrom(src, samples_per_channel, input_layout);
+  capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->CopyTo(fwd_out_format_.samples_per_channel(),
-                         output_layout,
-                         dest);
+  capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
-        sizeof(float) * fwd_out_format_.samples_per_channel();
-    for (int i = 0; i < fwd_out_format_.num_channels(); ++i)
+        sizeof(float) * formats_.api_format.output_stream().num_frames();
+    for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
+         ++i)
       msg->add_output_channel(dest[i], channel_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
@@ -518,7 +655,20 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(crit_);
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
+  {
+    // Acquire the capture lock in order to safely call the function
+    // that retrieves the render side data. This function accesses apm
+    // getters that need the capture lock held when being called.
+    // The lock needs to be released as
+    // public_submodules_->echo_control_mobile->is_enabled() aquires this lock
+    // as well.
+    rtc::CritScope cs_capture(&crit_capture_);
+    public_submodules_->echo_cancellation->ReadQueuedRenderData();
+    public_submodules_->echo_control_mobile->ReadQueuedRenderData();
+    public_submodules_->gain_control->ReadQueuedRenderData();
+  }
+
   if (!frame) {
     return kNullPointerError;
   }
@@ -529,70 +679,89 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
       frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
-  if (echo_control_mobile_->is_enabled() &&
-      frame->sample_rate_hz_ > kSampleRate16kHz) {
+
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
+      frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
     LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
     return kUnsupportedComponentError;
   }
 
-  // TODO(ajm): The input and output rates and channels are currently
-  // constrained to be identical in the int16 interface.
-  RETURN_ON_ERR(MaybeInitializeLocked(frame->sample_rate_hz_,
-                                      frame->sample_rate_hz_,
-                                      rev_in_format_.rate(),
-                                      frame->num_channels_,
-                                      frame->num_channels_,
-                                      rev_in_format_.num_channels()));
-  if (frame->samples_per_channel_ != fwd_in_format_.samples_per_channel()) {
+  ProcessingConfig processing_config;
+  {
+    // Aquire lock for the access of api_format.
+    // The lock is released immediately due to the conditional
+    // reinitialization.
+    rtc::CritScope cs_capture(&crit_capture_);
+    // TODO(ajm): The input and output rates and channels are currently
+    // constrained to be identical in the int16 interface.
+    processing_config = formats_.api_format;
+  }
+  processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
+  processing_config.input_stream().set_num_channels(frame->num_channels_);
+  processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
+  processing_config.output_stream().set_num_channels(frame->num_channels_);
+
+  {
+    // Do conditional reinitialization.
+    rtc::CritScope cs_render(&crit_render_);
+    RETURN_ON_ERR(MaybeInitializeCapture(processing_config));
+  }
+  rtc::CritScope cs_capture(&crit_capture_);
+  if (frame->samples_per_channel_ !=
+      formats_.api_format.input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t data_size = sizeof(int16_t) *
-                             frame->samples_per_channel_ *
-                             frame->num_channels_;
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
+    const size_t data_size =
+        sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_input_data(frame->data_, data_size);
   }
 #endif
 
-  capture_audio_->DeinterleaveFrom(frame);
+  capture_.capture_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessStreamLocked());
-  capture_audio_->InterleaveTo(frame, output_copy_needed(is_data_processed()));
+  capture_.capture_audio->InterleaveTo(frame,
+                                       output_copy_needed(is_data_processed()));
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t data_size = sizeof(int16_t) *
-                             frame->samples_per_channel_ *
-                             frame->num_channels_;
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
+    const size_t data_size =
+        sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_data(frame->data_, data_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
-
 int AudioProcessingImpl::ProcessStreamLocked() {
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    msg->set_delay(stream_delay_ms_);
-    msg->set_drift(echo_cancellation_->stream_drift_samples());
+  if (debug_dump_.debug_file->Open()) {
+    audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
+    msg->set_delay(capture_nonlocked_.stream_delay_ms);
+    msg->set_drift(
+        public_submodules_->echo_cancellation->stream_drift_samples());
     msg->set_level(gain_control()->stream_analog_level());
-    msg->set_keypress(key_pressed_);
+    msg->set_keypress(capture_.key_pressed);
   }
 #endif
 
-  AudioBuffer* ca = capture_audio_.get();  // For brevity.
-  if (use_new_agc_ && gain_control_->is_enabled()) {
-    agc_manager_->AnalyzePreProcess(ca->channels()[0],
-                                    ca->num_channels(),
-                                    fwd_proc_format_.samples_per_channel());
+  MaybeUpdateHistograms();
+
+  AudioBuffer* ca = capture_.capture_audio.get();  // For brevity.
+
+  if (constants_.use_new_agc &&
+      public_submodules_->gain_control->is_enabled()) {
+    private_submodules_->agc_manager->AnalyzePreProcess(
+        ca->channels()[0], ca->num_channels(),
+        capture_nonlocked_.fwd_proc_format.num_frames());
   }
 
   bool data_processed = is_data_processed();
@@ -600,31 +769,41 @@ int AudioProcessingImpl::ProcessStreamLocked() {
     ca->SplitIntoFrequencyBands();
   }
 
-  if (beamformer_enabled_) {
-    beamformer_->ProcessChunk(*ca->split_data_f(), ca->split_data_f());
+  if (constants_.intelligibility_enabled) {
+    public_submodules_->intelligibility_enhancer->AnalyzeCaptureAudio(
+        ca->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
+        ca->num_channels());
+  }
+
+  if (capture_nonlocked_.beamformer_enabled) {
+    private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
+                                                  ca->split_data_f());
     ca->set_num_channels(1);
   }
 
-  RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
-  RETURN_ON_ERR(noise_suppression_->AnalyzeCaptureAudio(ca));
-  RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca));
+  public_submodules_->high_pass_filter->ProcessCaptureAudio(ca);
+  RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca));
+  public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca);
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca));
 
-  if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) {
+  if (public_submodules_->echo_control_mobile->is_enabled() &&
+      public_submodules_->noise_suppression->is_enabled()) {
     ca->CopyLowPassToReference();
   }
-  RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
-  RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
+  public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
+  RETURN_ON_ERR(
+      public_submodules_->echo_control_mobile->ProcessCaptureAudio(ca));
+  public_submodules_->voice_detection->ProcessCaptureAudio(ca);
 
-  if (use_new_agc_ &&
-      gain_control_->is_enabled() &&
-      (!beamformer_enabled_ || beamformer_->is_target_present())) {
-    agc_manager_->Process(ca->split_bands_const(0)[kBand0To8kHz],
-                          ca->num_frames_per_band(),
-                          split_rate_);
+  if (constants_.use_new_agc &&
+      public_submodules_->gain_control->is_enabled() &&
+      (!capture_nonlocked_.beamformer_enabled ||
+       private_submodules_->beamformer->is_target_present())) {
+    private_submodules_->agc_manager->Process(
+        ca->split_bands_const(0)[kBand0To8kHz], ca->num_frames_per_band(),
+        capture_nonlocked_.split_rate);
   }
-  RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
+  RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(ca));
 
   if (synthesis_needed(data_processed)) {
     ca->MergeFrequencyBands();
@@ -632,67 +811,120 @@ int AudioProcessingImpl::ProcessStreamLocked() {
 
   // TODO(aluebs): Investigate if the transient suppression placement should be
   // before or after the AGC.
-  if (transient_suppressor_enabled_) {
+  if (capture_.transient_suppressor_enabled) {
     float voice_probability =
-        agc_manager_.get() ? agc_manager_->voice_probability() : 1.f;
+        private_submodules_->agc_manager.get()
+            ? private_submodules_->agc_manager->voice_probability()
+            : 1.f;
 
-    transient_suppressor_->Suppress(ca->channels_f()[0],
-                                    ca->num_frames(),
-                                    ca->num_channels(),
-                                    ca->split_bands_const_f(0)[kBand0To8kHz],
-                                    ca->num_frames_per_band(),
-                                    ca->keyboard_data(),
-                                    ca->num_keyboard_frames(),
-                                    voice_probability,
-                                    key_pressed_);
+    public_submodules_->transient_suppressor->Suppress(
+        ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
+        ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
+        ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
+        capture_.key_pressed);
   }
 
   // The level estimator operates on the recombined data.
-  RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
+  public_submodules_->level_estimator->ProcessStream(ca);
 
-  was_stream_delay_set_ = false;
+  capture_.was_stream_delay_set = false;
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
-                                              int samples_per_channel,
-                                              int sample_rate_hz,
+                                              size_t samples_per_channel,
+                                              int rev_sample_rate_hz,
                                               ChannelLayout layout) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (data == NULL) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout");
+  rtc::CritScope cs(&crit_render_);
+  const StreamConfig reverse_config = {
+      rev_sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
+  };
+  if (samples_per_channel != reverse_config.num_frames()) {
+    return kBadDataLengthError;
+  }
+  return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
+}
+
+int AudioProcessingImpl::ProcessReverseStream(
+    const float* const* src,
+    const StreamConfig& reverse_input_config,
+    const StreamConfig& reverse_output_config,
+    float* const* dest) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig");
+  rtc::CritScope cs(&crit_render_);
+  RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, reverse_input_config,
+                                           reverse_output_config));
+  if (is_rev_processed()) {
+    render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(),
+                                 dest);
+  } else if (render_check_rev_conversion_needed()) {
+    render_.render_converter->Convert(src, reverse_input_config.num_samples(),
+                                      dest,
+                                      reverse_output_config.num_samples());
+  } else {
+    CopyAudioIfNeeded(src, reverse_input_config.num_frames(),
+                      reverse_input_config.num_channels(), dest);
+  }
+
+  return kNoError;
+}
+
+int AudioProcessingImpl::AnalyzeReverseStreamLocked(
+    const float* const* src,
+    const StreamConfig& reverse_input_config,
+    const StreamConfig& reverse_output_config) {
+  if (src == nullptr) {
     return kNullPointerError;
   }
 
-  const int num_channels = ChannelsFromLayout(layout);
-  RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
-                                      fwd_out_format_.rate(),
-                                      sample_rate_hz,
-                                      fwd_in_format_.num_channels(),
-                                      fwd_out_format_.num_channels(),
-                                      num_channels));
-  if (samples_per_channel != rev_in_format_.samples_per_channel()) {
-    return kBadDataLengthError;
+  if (reverse_input_config.num_channels() == 0) {
+    return kBadNumberChannelsError;
   }
 
+  ProcessingConfig processing_config = formats_.api_format;
+  processing_config.reverse_input_stream() = reverse_input_config;
+  processing_config.reverse_output_stream() = reverse_output_config;
+
+  RETURN_ON_ERR(MaybeInitializeRender(processing_config));
+  assert(reverse_input_config.num_frames() ==
+         formats_.api_format.reverse_input_stream().num_frames());
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
+    audioproc::ReverseStream* msg =
+        debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t channel_size =
-        sizeof(float) * rev_in_format_.samples_per_channel();
-    for (int i = 0; i < num_channels; ++i)
-      msg->add_channel(data[i], channel_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+        sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
+    for (size_t i = 0;
+         i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
+      msg->add_channel(src[i], channel_size);
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.render));
   }
 #endif
 
-  render_audio_->CopyFrom(data, samples_per_channel, layout);
-  return AnalyzeReverseStreamLocked();
+  render_.render_audio->CopyFrom(src,
+                                 formats_.api_format.reverse_input_stream());
+  return ProcessReverseStreamLocked();
+}
+
+int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
+  RETURN_ON_ERR(AnalyzeReverseStream(frame));
+  rtc::CritScope cs(&crit_render_);
+  if (is_rev_processed()) {
+    render_.render_audio->InterleaveTo(frame, true);
+  }
+
+  return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (frame == NULL) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_AudioFrame");
+  rtc::CritScope cs(&crit_render_);
+  if (frame == nullptr) {
     return kNullPointerError;
   }
   // Must be a native rate.
@@ -703,55 +935,83 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
     return kBadSampleRateError;
   }
   // This interface does not tolerate different forward and reverse rates.
-  if (frame->sample_rate_hz_ != fwd_in_format_.rate()) {
+  if (frame->sample_rate_hz_ !=
+      formats_.api_format.input_stream().sample_rate_hz()) {
     return kBadSampleRateError;
   }
 
-  RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
-                                      fwd_out_format_.rate(),
-                                      frame->sample_rate_hz_,
-                                      fwd_in_format_.num_channels(),
-                                      fwd_in_format_.num_channels(),
-                                      frame->num_channels_));
-  if (frame->samples_per_channel_ != rev_in_format_.samples_per_channel()) {
+  if (frame->num_channels_ <= 0) {
+    return kBadNumberChannelsError;
+  }
+
+  ProcessingConfig processing_config = formats_.api_format;
+  processing_config.reverse_input_stream().set_sample_rate_hz(
+      frame->sample_rate_hz_);
+  processing_config.reverse_input_stream().set_num_channels(
+      frame->num_channels_);
+  processing_config.reverse_output_stream().set_sample_rate_hz(
+      frame->sample_rate_hz_);
+  processing_config.reverse_output_stream().set_num_channels(
+      frame->num_channels_);
+
+  RETURN_ON_ERR(MaybeInitializeRender(processing_config));
+  if (frame->samples_per_channel_ !=
+      formats_.api_format.reverse_input_stream().num_frames()) {
     return kBadDataLengthError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
-    const size_t data_size = sizeof(int16_t) *
-                             frame->samples_per_channel_ *
-                             frame->num_channels_;
+  if (debug_dump_.debug_file->Open()) {
+    debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
+    audioproc::ReverseStream* msg =
+        debug_dump_.render.event_msg->mutable_reverse_stream();
+    const size_t data_size =
+        sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_data(frame->data_, data_size);
-    RETURN_ON_ERR(WriteMessageToDebugFile());
+    RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                          &crit_debug_, &debug_dump_.render));
   }
 #endif
-
-  render_audio_->DeinterleaveFrom(frame);
-  return AnalyzeReverseStreamLocked();
+  render_.render_audio->DeinterleaveFrom(frame);
+  return ProcessReverseStreamLocked();
 }
 
-int AudioProcessingImpl::AnalyzeReverseStreamLocked() {
-  AudioBuffer* ra = render_audio_.get();  // For brevity.
-  if (rev_proc_format_.rate() == kSampleRate32kHz) {
+int AudioProcessingImpl::ProcessReverseStreamLocked() {
+  AudioBuffer* ra = render_.render_audio.get();  // For brevity.
+  if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz) {
     ra->SplitIntoFrequencyBands();
   }
 
-  RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra));
-  RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra));
-  if (!use_new_agc_) {
-    RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra));
+  if (constants_.intelligibility_enabled) {
+    // Currently run in single-threaded mode when the intelligibility
+    // enhancer is activated.
+    // TODO(peah): Fix to be properly multi-threaded.
+    rtc::CritScope cs(&crit_capture_);
+    public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
+        ra->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
+        ra->num_channels());
+  }
+
+  RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessRenderAudio(ra));
+  RETURN_ON_ERR(
+      public_submodules_->echo_control_mobile->ProcessRenderAudio(ra));
+  if (!constants_.use_new_agc) {
+    RETURN_ON_ERR(public_submodules_->gain_control->ProcessRenderAudio(ra));
+  }
+
+  if (formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
+      is_rev_processed()) {
+    ra->MergeFrequencyBands();
   }
 
   return kNoError;
 }
 
 int AudioProcessingImpl::set_stream_delay_ms(int delay) {
+  rtc::CritScope cs(&crit_capture_);
   Error retval = kNoError;
-  was_stream_delay_set_ = true;
-  delay += delay_offset_ms_;
+  capture_.was_stream_delay_set = true;
+  delay += capture_.delay_offset_ms;
 
   if (delay < 0) {
     delay = 0;
@@ -764,61 +1024,61 @@ int AudioProcessingImpl::set_stream_delay_ms(int delay) {
     retval = kBadStreamParameterWarning;
   }
 
-  stream_delay_ms_ = delay;
+  capture_nonlocked_.stream_delay_ms = delay;
   return retval;
 }
 
 int AudioProcessingImpl::stream_delay_ms() const {
-  return stream_delay_ms_;
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.stream_delay_ms;
 }
 
 bool AudioProcessingImpl::was_stream_delay_set() const {
-  return was_stream_delay_set_;
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_.was_stream_delay_set;
 }
 
 void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
-  key_pressed_ = key_pressed;
-}
-
-bool AudioProcessingImpl::stream_key_pressed() const {
-  return key_pressed_;
+  rtc::CritScope cs(&crit_capture_);
+  capture_.key_pressed = key_pressed;
 }
 
 void AudioProcessingImpl::set_delay_offset_ms(int offset) {
-  CriticalSectionScoped crit_scoped(crit_);
-  delay_offset_ms_ = offset;
+  rtc::CritScope cs(&crit_capture_);
+  capture_.delay_offset_ms = offset;
 }
 
 int AudioProcessingImpl::delay_offset_ms() const {
-  return delay_offset_ms_;
+  rtc::CritScope cs(&crit_capture_);
+  return capture_.delay_offset_ms;
 }
 
 int AudioProcessingImpl::StartDebugRecording(
     const char filename[AudioProcessing::kMaxFilenameSize]) {
-  CriticalSectionScoped crit_scoped(crit_);
-  assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  static_assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize, "");
 
-  if (filename == NULL) {
+  if (filename == nullptr) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
-  if (debug_file_->OpenFile(filename, false) == -1) {
-    debug_file_->CloseFile();
+  if (debug_dump_.debug_file->OpenFile(filename, false) == -1) {
+    debug_dump_.debug_file->CloseFile();
     return kFileError;
   }
 
-  int err = WriteInitMessage();
-  if (err != kNoError) {
-    return err;
-  }
+  RETURN_ON_ERR(WriteConfigMessage(true));
+  RETURN_ON_ERR(WriteInitMessage());
   return kNoError;
 #else
   return kUnsupportedFunctionError;
@@ -826,28 +1086,28 @@ int AudioProcessingImpl::StartDebugRecording(
 }
 
 int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
 
-  if (handle == NULL) {
+  if (handle == nullptr) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
 
-  if (debug_file_->OpenFromFileHandle(handle, true, false) == -1) {
+  if (debug_dump_.debug_file->OpenFromFileHandle(handle, true, false) == -1) {
     return kFileError;
   }
 
-  int err = WriteInitMessage();
-  if (err != kNoError) {
-    return err;
-  }
+  RETURN_ON_ERR(WriteConfigMessage(true));
+  RETURN_ON_ERR(WriteInitMessage());
   return kNoError;
 #else
   return kUnsupportedFunctionError;
@@ -856,17 +1116,22 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
 
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
   return StartDebugRecording(stream);
 }
 
 int AudioProcessingImpl::StopDebugRecording() {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
+  if (debug_dump_.debug_file->Open()) {
+    if (debug_dump_.debug_file->CloseFile() == -1) {
       return kFileError;
     }
   }
@@ -877,58 +1142,87 @@ int AudioProcessingImpl::StopDebugRecording() {
 }
 
 EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
-  return echo_cancellation_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->echo_cancellation;
 }
 
 EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
-  return echo_control_mobile_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->echo_control_mobile;
 }
 
 GainControl* AudioProcessingImpl::gain_control() const {
-  if (use_new_agc_) {
-    return gain_control_for_new_agc_.get();
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  if (constants_.use_new_agc) {
+    return public_submodules_->gain_control_for_new_agc.get();
   }
-  return gain_control_;
+  return public_submodules_->gain_control;
 }
 
 HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
-  return high_pass_filter_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->high_pass_filter.get();
 }
 
 LevelEstimator* AudioProcessingImpl::level_estimator() const {
-  return level_estimator_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->level_estimator.get();
 }
 
 NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
-  return noise_suppression_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->noise_suppression.get();
 }
 
 VoiceDetection* AudioProcessingImpl::voice_detection() const {
-  return voice_detection_;
+  // Adding a lock here has no effect as it allows any access to the submodule
+  // from the returned pointer.
+  return public_submodules_->voice_detection.get();
 }
 
 bool AudioProcessingImpl::is_data_processed() const {
-  if (beamformer_enabled_) {
+  if (capture_nonlocked_.beamformer_enabled) {
     return true;
   }
 
   int enabled_count = 0;
-  for (auto item : component_list_) {
+  for (auto item : private_submodules_->component_list) {
     if (item->is_component_enabled()) {
       enabled_count++;
     }
   }
+  if (public_submodules_->high_pass_filter->is_enabled()) {
+    enabled_count++;
+  }
+  if (public_submodules_->noise_suppression->is_enabled()) {
+    enabled_count++;
+  }
+  if (public_submodules_->level_estimator->is_enabled()) {
+    enabled_count++;
+  }
+  if (public_submodules_->voice_detection->is_enabled()) {
+    enabled_count++;
+  }
 
-  // Data is unchanged if no components are enabled, or if only level_estimator_
-  // or voice_detection_ is enabled.
+  // Data is unchanged if no components are enabled, or if only
+  // public_submodules_->level_estimator
+  // or public_submodules_->voice_detection is enabled.
   if (enabled_count == 0) {
     return false;
   } else if (enabled_count == 1) {
-    if (level_estimator_->is_enabled() || voice_detection_->is_enabled()) {
+    if (public_submodules_->level_estimator->is_enabled() ||
+        public_submodules_->voice_detection->is_enabled()) {
       return false;
     }
   } else if (enabled_count == 2) {
-    if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
+    if (public_submodules_->level_estimator->is_enabled() &&
+        public_submodules_->voice_detection->is_enabled()) {
       return false;
     }
   }
@@ -937,104 +1231,296 @@ bool AudioProcessingImpl::is_data_processed() const {
 
 bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
   // Check if we've upmixed or downmixed the audio.
-  return ((fwd_out_format_.num_channels() != fwd_in_format_.num_channels()) ||
-          is_data_processed || transient_suppressor_enabled_);
+  return ((formats_.api_format.output_stream().num_channels() !=
+           formats_.api_format.input_stream().num_channels()) ||
+          is_data_processed || capture_.transient_suppressor_enabled);
 }
 
 bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
-  return (is_data_processed && (fwd_proc_format_.rate() == kSampleRate32kHz ||
-          fwd_proc_format_.rate() == kSampleRate48kHz));
+  return (is_data_processed &&
+          (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+               kSampleRate32kHz ||
+           capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+               kSampleRate48kHz));
 }
 
 bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
-  if (!is_data_processed && !voice_detection_->is_enabled() &&
-      !transient_suppressor_enabled_) {
-    // Only level_estimator_ is enabled.
+  if (!is_data_processed &&
+      !public_submodules_->voice_detection->is_enabled() &&
+      !capture_.transient_suppressor_enabled) {
+    // Only public_submodules_->level_estimator is enabled.
     return false;
-  } else if (fwd_proc_format_.rate() == kSampleRate32kHz ||
-             fwd_proc_format_.rate() == kSampleRate48kHz) {
-    // Something besides level_estimator_ is enabled, and we have super-wb.
+  } else if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+                 kSampleRate32kHz ||
+             capture_nonlocked_.fwd_proc_format.sample_rate_hz() ==
+                 kSampleRate48kHz) {
+    // Something besides public_submodules_->level_estimator is enabled, and we
+    // have super-wb.
     return true;
   }
   return false;
 }
 
-int AudioProcessingImpl::InitializeExperimentalAgc() {
-  if (use_new_agc_) {
-    if (!agc_manager_.get()) {
-      agc_manager_.reset(
-          new AgcManagerDirect(gain_control_, gain_control_for_new_agc_.get()));
-    }
-    agc_manager_->Initialize();
-    agc_manager_->SetCaptureMuted(output_will_be_muted_);
-  }
-  return kNoError;
+bool AudioProcessingImpl::is_rev_processed() const {
+  return constants_.intelligibility_enabled &&
+         public_submodules_->intelligibility_enhancer->active();
 }
 
-int AudioProcessingImpl::InitializeTransient() {
-  if (transient_suppressor_enabled_) {
-    if (!transient_suppressor_.get()) {
-      transient_suppressor_.reset(new TransientSuppressor());
+bool AudioProcessingImpl::render_check_rev_conversion_needed() const {
+  return rev_conversion_needed();
+}
+
+bool AudioProcessingImpl::rev_conversion_needed() const {
+  return (formats_.api_format.reverse_input_stream() !=
+          formats_.api_format.reverse_output_stream());
+}
+
+void AudioProcessingImpl::InitializeExperimentalAgc() {
+  if (constants_.use_new_agc) {
+    if (!private_submodules_->agc_manager.get()) {
+      private_submodules_->agc_manager.reset(new AgcManagerDirect(
+          public_submodules_->gain_control,
+          public_submodules_->gain_control_for_new_agc.get(),
+          constants_.agc_startup_min_volume));
     }
-    transient_suppressor_->Initialize(fwd_proc_format_.rate(),
-                                      split_rate_,
-                                      fwd_out_format_.num_channels());
+    private_submodules_->agc_manager->Initialize();
+    private_submodules_->agc_manager->SetCaptureMuted(
+        capture_.output_will_be_muted);
+  }
+}
+
+void AudioProcessingImpl::InitializeTransient() {
+  if (capture_.transient_suppressor_enabled) {
+    if (!public_submodules_->transient_suppressor.get()) {
+      public_submodules_->transient_suppressor.reset(new TransientSuppressor());
+    }
+    public_submodules_->transient_suppressor->Initialize(
+        capture_nonlocked_.fwd_proc_format.sample_rate_hz(),
+        capture_nonlocked_.split_rate,
+        num_proc_channels());
   }
-  return kNoError;
 }
 
 void AudioProcessingImpl::InitializeBeamformer() {
-  if (beamformer_enabled_) {
-    if (!beamformer_) {
-      beamformer_.reset(new NonlinearBeamformer(array_geometry_));
+  if (capture_nonlocked_.beamformer_enabled) {
+    if (!private_submodules_->beamformer) {
+      private_submodules_->beamformer.reset(new NonlinearBeamformer(
+          capture_.array_geometry, capture_.target_direction));
     }
-    beamformer_->Initialize(kChunkSizeMs, split_rate_);
+    private_submodules_->beamformer->Initialize(kChunkSizeMs,
+                                                capture_nonlocked_.split_rate);
   }
 }
 
+void AudioProcessingImpl::InitializeIntelligibility() {
+  if (constants_.intelligibility_enabled) {
+    IntelligibilityEnhancer::Config config;
+    config.sample_rate_hz = capture_nonlocked_.split_rate;
+    config.num_capture_channels = capture_.capture_audio->num_channels();
+    config.num_render_channels = render_.render_audio->num_channels();
+    public_submodules_->intelligibility_enhancer.reset(
+        new IntelligibilityEnhancer(config));
+  }
+}
+
+void AudioProcessingImpl::InitializeHighPassFilter() {
+  public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
+                                                   proc_sample_rate_hz());
+}
+
+void AudioProcessingImpl::InitializeNoiseSuppression() {
+  public_submodules_->noise_suppression->Initialize(num_proc_channels(),
+                                                    proc_sample_rate_hz());
+}
+
+void AudioProcessingImpl::InitializeLevelEstimator() {
+  public_submodules_->level_estimator->Initialize();
+}
+
+void AudioProcessingImpl::InitializeVoiceDetection() {
+  public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
+}
+
+void AudioProcessingImpl::MaybeUpdateHistograms() {
+  static const int kMinDiffDelayMs = 60;
+
+  if (echo_cancellation()->is_enabled()) {
+    // Activate delay_jumps_ counters if we know echo_cancellation is runnning.
+    // If a stream has echo we know that the echo_cancellation is in process.
+    if (capture_.stream_delay_jumps == -1 &&
+        echo_cancellation()->stream_has_echo()) {
+      capture_.stream_delay_jumps = 0;
+    }
+    if (capture_.aec_system_delay_jumps == -1 &&
+        echo_cancellation()->stream_has_echo()) {
+      capture_.aec_system_delay_jumps = 0;
+    }
+
+    // Detect a jump in platform reported system delay and log the difference.
+    const int diff_stream_delay_ms =
+        capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
+    if (diff_stream_delay_ms > kMinDiffDelayMs &&
+        capture_.last_stream_delay_ms != 0) {
+      RTC_HISTOGRAM_COUNTS_SPARSE(
+          "WebRTC.Audio.PlatformReportedStreamDelayJump", diff_stream_delay_ms,
+          kMinDiffDelayMs, 1000, 100);
+      if (capture_.stream_delay_jumps == -1) {
+        capture_.stream_delay_jumps = 0;  // Activate counter if needed.
+      }
+      capture_.stream_delay_jumps++;
+    }
+    capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
+
+    // Detect a jump in AEC system delay and log the difference.
+    const int frames_per_ms =
+        rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
+    const int aec_system_delay_ms =
+        WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
+    const int diff_aec_system_delay_ms =
+        aec_system_delay_ms - capture_.last_aec_system_delay_ms;
+    if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
+        capture_.last_aec_system_delay_ms != 0) {
+      RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.AecSystemDelayJump",
+                                  diff_aec_system_delay_ms, kMinDiffDelayMs,
+                                  1000, 100);
+      if (capture_.aec_system_delay_jumps == -1) {
+        capture_.aec_system_delay_jumps = 0;  // Activate counter if needed.
+      }
+      capture_.aec_system_delay_jumps++;
+    }
+    capture_.last_aec_system_delay_ms = aec_system_delay_ms;
+  }
+}
+
+void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+
+  if (capture_.stream_delay_jumps > -1) {
+    RTC_HISTOGRAM_ENUMERATION_SPARSE(
+        "WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
+        capture_.stream_delay_jumps, 51);
+  }
+  capture_.stream_delay_jumps = -1;
+  capture_.last_stream_delay_ms = 0;
+
+  if (capture_.aec_system_delay_jumps > -1) {
+    RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Audio.NumOfAecSystemDelayJumps",
+                                     capture_.aec_system_delay_jumps, 51);
+  }
+  capture_.aec_system_delay_jumps = -1;
+  capture_.last_aec_system_delay_ms = 0;
+}
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-int AudioProcessingImpl::WriteMessageToDebugFile() {
-  int32_t size = event_msg_->ByteSize();
+int AudioProcessingImpl::WriteMessageToDebugFile(
+    FileWrapper* debug_file,
+    rtc::CriticalSection* crit_debug,
+    ApmDebugDumpThreadState* debug_state) {
+  int32_t size = debug_state->event_msg->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
   }
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)
-  // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
-  //            pretty safe in assuming little-endian.
+// TODO(ajm): Use little-endian "on the wire". For the moment, we can be
+//            pretty safe in assuming little-endian.
 #endif
 
-  if (!event_msg_->SerializeToString(&event_str_)) {
+  if (!debug_state->event_msg->SerializeToString(&debug_state->event_str)) {
     return kUnspecifiedError;
   }
 
-  // Write message preceded by its size.
-  if (!debug_file_->Write(&size, sizeof(int32_t))) {
-    return kFileError;
-  }
-  if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
-    return kFileError;
+  {
+    // Ensure atomic writes of the message.
+    rtc::CritScope cs_capture(crit_debug);
+    // Write message preceded by its size.
+    if (!debug_file->Write(&size, sizeof(int32_t))) {
+      return kFileError;
+    }
+    if (!debug_file->Write(debug_state->event_str.data(),
+                           debug_state->event_str.length())) {
+      return kFileError;
+    }
   }
 
-  event_msg_->Clear();
+  debug_state->event_msg->Clear();
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
-  event_msg_->set_type(audioproc::Event::INIT);
-  audioproc::Init* msg = event_msg_->mutable_init();
-  msg->set_sample_rate(fwd_in_format_.rate());
-  msg->set_num_input_channels(fwd_in_format_.num_channels());
-  msg->set_num_output_channels(fwd_out_format_.num_channels());
-  msg->set_num_reverse_channels(rev_in_format_.num_channels());
-  msg->set_reverse_sample_rate(rev_in_format_.rate());
-  msg->set_output_sample_rate(fwd_out_format_.rate());
+  debug_dump_.capture.event_msg->set_type(audioproc::Event::INIT);
+  audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
+  msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
 
-  int err = WriteMessageToDebugFile();
-  if (err != kNoError) {
-    return err;
+  msg->set_num_input_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.input_stream().num_channels()));
+  msg->set_num_output_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.output_stream().num_channels()));
+  msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.reverse_input_stream().num_channels()));
+  msg->set_reverse_sample_rate(
+      formats_.api_format.reverse_input_stream().sample_rate_hz());
+  msg->set_output_sample_rate(
+      formats_.api_format.output_stream().sample_rate_hz());
+  // TODO(ekmeyerson): Add reverse output fields to
+  // debug_dump_.capture.event_msg.
+
+  RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &crit_debug_, &debug_dump_.capture));
+  return kNoError;
+}
+
+int AudioProcessingImpl::WriteConfigMessage(bool forced) {
+  audioproc::Config config;
+
+  config.set_aec_enabled(public_submodules_->echo_cancellation->is_enabled());
+  config.set_aec_delay_agnostic_enabled(
+      public_submodules_->echo_cancellation->is_delay_agnostic_enabled());
+  config.set_aec_drift_compensation_enabled(
+      public_submodules_->echo_cancellation->is_drift_compensation_enabled());
+  config.set_aec_extended_filter_enabled(
+      public_submodules_->echo_cancellation->is_extended_filter_enabled());
+  config.set_aec_suppression_level(static_cast<int>(
+      public_submodules_->echo_cancellation->suppression_level()));
+
+  config.set_aecm_enabled(
+      public_submodules_->echo_control_mobile->is_enabled());
+  config.set_aecm_comfort_noise_enabled(
+      public_submodules_->echo_control_mobile->is_comfort_noise_enabled());
+  config.set_aecm_routing_mode(static_cast<int>(
+      public_submodules_->echo_control_mobile->routing_mode()));
+
+  config.set_agc_enabled(public_submodules_->gain_control->is_enabled());
+  config.set_agc_mode(
+      static_cast<int>(public_submodules_->gain_control->mode()));
+  config.set_agc_limiter_enabled(
+      public_submodules_->gain_control->is_limiter_enabled());
+  config.set_noise_robust_agc_enabled(constants_.use_new_agc);
+
+  config.set_hpf_enabled(public_submodules_->high_pass_filter->is_enabled());
+
+  config.set_ns_enabled(public_submodules_->noise_suppression->is_enabled());
+  config.set_ns_level(
+      static_cast<int>(public_submodules_->noise_suppression->level()));
+
+  config.set_transient_suppression_enabled(
+      capture_.transient_suppressor_enabled);
+
+  std::string serialized_config = config.SerializeAsString();
+  if (!forced &&
+      debug_dump_.capture.last_serialized_config == serialized_config) {
+    return kNoError;
   }
 
+  debug_dump_.capture.last_serialized_config = serialized_config;
+
+  debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
+  debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
+
+  RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
+                                        &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.h
index 500f08e82f..b310896903 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -13,87 +13,40 @@
 
 #include <list>
 #include <string>
+#include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 namespace webrtc {
 
 class AgcManagerDirect;
-class AudioBuffer;
+class AudioConverter;
 
 template<typename T>
 class Beamformer;
 
-class CriticalSectionWrapper;
-class EchoCancellationImpl;
-class EchoControlMobileImpl;
-class FileWrapper;
-class GainControlImpl;
-class GainControlForNewAgc;
-class HighPassFilterImpl;
-class LevelEstimatorImpl;
-class NoiseSuppressionImpl;
-class ProcessingComponent;
-class TransientSuppressor;
-class VoiceDetectionImpl;
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-namespace audioproc {
-
-class Event;
-
-}  // namespace audioproc
-#endif
-
-class AudioRate {
- public:
-  explicit AudioRate(int sample_rate_hz)
-      : rate_(sample_rate_hz),
-        samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {}
-  virtual ~AudioRate() {}
-
-  void set(int rate) {
-    rate_ = rate;
-    samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
-  }
-
-  int rate() const { return rate_; }
-  int samples_per_channel() const { return samples_per_channel_; }
-
- private:
-  int rate_;
-  int samples_per_channel_;
-};
-
-class AudioFormat : public AudioRate {
- public:
-  AudioFormat(int sample_rate_hz, int num_channels)
-      : AudioRate(sample_rate_hz),
-        num_channels_(num_channels) {}
-  virtual ~AudioFormat() {}
-
-  void set(int rate, int num_channels) {
-    AudioRate::set(rate);
-    num_channels_ = num_channels;
-  }
-
-  int num_channels() const { return num_channels_; }
-
- private:
-  int num_channels_;
-};
-
 class AudioProcessingImpl : public AudioProcessing {
  public:
+  // Methods forcing APM to run in a single-threaded manner.
+  // Acquires both the render and capture locks.
   explicit AudioProcessingImpl(const Config& config);
-
   // AudioProcessingImpl takes ownership of beamformer.
   AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
   virtual ~AudioProcessingImpl();
-
-  // AudioProcessing methods.
   int Initialize() override;
   int Initialize(int input_sample_rate_hz,
                  int output_sample_rate_hz,
@@ -101,41 +54,66 @@ class AudioProcessingImpl : public AudioProcessing {
                  ChannelLayout input_layout,
                  ChannelLayout output_layout,
                  ChannelLayout reverse_layout) override;
+  int Initialize(const ProcessingConfig& processing_config) override;
   void SetExtraOptions(const Config& config) override;
-  int set_sample_rate_hz(int rate) override;
-  int input_sample_rate_hz() const override;
-  int sample_rate_hz() const override;
-  int proc_sample_rate_hz() const override;
-  int proc_split_sample_rate_hz() const override;
-  int num_input_channels() const override;
-  int num_output_channels() const override;
-  int num_reverse_channels() const override;
-  void set_output_will_be_muted(bool muted) override;
-  bool output_will_be_muted() const override;
+  void UpdateHistogramsOnCallEnd() override;
+  int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
+  int StartDebugRecording(FILE* handle) override;
+  int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
+  int StopDebugRecording() override;
+
+  // Capture-side exclusive methods possibly running APM in a
+  // multi-threaded manner. Acquire the capture lock.
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
-                    int samples_per_channel,
+                    size_t samples_per_channel,
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
                     int output_sample_rate_hz,
                     ChannelLayout output_layout,
                     float* const* dest) override;
-  int AnalyzeReverseStream(AudioFrame* frame) override;
-  int AnalyzeReverseStream(const float* const* data,
-                           int samples_per_channel,
-                           int sample_rate_hz,
-                           ChannelLayout layout) override;
+  int ProcessStream(const float* const* src,
+                    const StreamConfig& input_config,
+                    const StreamConfig& output_config,
+                    float* const* dest) override;
+  void set_output_will_be_muted(bool muted) override;
   int set_stream_delay_ms(int delay) override;
-  int stream_delay_ms() const override;
-  bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
-  bool stream_key_pressed() const override;
-  int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
-  int StartDebugRecording(FILE* handle) override;
-  int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
-  int StopDebugRecording() override;
+  int input_sample_rate_hz() const override;
+
+  // Render-side exclusive methods possibly running APM in a
+  // multi-threaded manner. Acquire the render lock.
+  int AnalyzeReverseStream(AudioFrame* frame) override;
+  int ProcessReverseStream(AudioFrame* frame) override;
+  int AnalyzeReverseStream(const float* const* data,
+                           size_t samples_per_channel,
+                           int sample_rate_hz,
+                           ChannelLayout layout) override;
+  int ProcessReverseStream(const float* const* src,
+                           const StreamConfig& reverse_input_config,
+                           const StreamConfig& reverse_output_config,
+                           float* const* dest) override;
+
+  // Methods only accessed from APM submodules or
+  // from AudioProcessing tests in a single-threaded manner.
+  // Hence there is no need for locks in these.
+  int proc_sample_rate_hz() const override;
+  int proc_split_sample_rate_hz() const override;
+  size_t num_input_channels() const override;
+  size_t num_proc_channels() const override;
+  size_t num_output_channels() const override;
+  size_t num_reverse_channels() const override;
+  int stream_delay_ms() const override;
+  bool was_stream_delay_set() const override
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Methods returning pointers to APM submodules.
+  // No locks are aquired in those, as those locks
+  // would offer no protection (the submodules are
+  // created only once in a single-treaded manner
+  // during APM creation).
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
@@ -146,85 +124,216 @@ class AudioProcessingImpl : public AudioProcessing {
 
  protected:
   // Overridden in a mock.
-  virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  virtual int InitializeLocked()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
  private:
-  int InitializeLocked(int input_sample_rate_hz,
-                       int output_sample_rate_hz,
-                       int reverse_sample_rate_hz,
-                       int num_input_channels,
-                       int num_output_channels,
-                       int num_reverse_channels)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int MaybeInitializeLocked(int input_sample_rate_hz,
-                            int output_sample_rate_hz,
-                            int reverse_sample_rate_hz,
-                            int num_input_channels,
-                            int num_output_channels,
-                            int num_reverse_channels)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  struct ApmPublicSubmodules;
+  struct ApmPrivateSubmodules;
 
-  bool is_data_processed() const;
-  bool output_copy_needed(bool is_data_processed) const;
-  bool synthesis_needed(bool is_data_processed) const;
-  bool analysis_needed(bool is_data_processed) const;
-  int InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  // State for the debug dump.
+  struct ApmDebugDumpThreadState {
+    ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
+    rtc::scoped_ptr<audioproc::Event> event_msg;  // Protobuf message.
+    std::string event_str;  // Memory for protobuf serialization.
 
-  EchoCancellationImpl* echo_cancellation_;
-  EchoControlMobileImpl* echo_control_mobile_;
-  GainControlImpl* gain_control_;
-  HighPassFilterImpl* high_pass_filter_;
-  LevelEstimatorImpl* level_estimator_;
-  NoiseSuppressionImpl* noise_suppression_;
-  VoiceDetectionImpl* voice_detection_;
-  rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
+    // Serialized string of last saved APM configuration.
+    std::string last_serialized_config;
+  };
 
-  std::list<ProcessingComponent*> component_list_;
-  CriticalSectionWrapper* crit_;
-  rtc::scoped_ptr<AudioBuffer> render_audio_;
-  rtc::scoped_ptr<AudioBuffer> capture_audio_;
+  struct ApmDebugDumpState {
+    ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
+    rtc::scoped_ptr<FileWrapper> debug_file;
+    ApmDebugDumpThreadState render;
+    ApmDebugDumpThreadState capture;
+  };
+#endif
+
+  // Method for modifying the formats struct that are called from both
+  // the render and capture threads. The check for whether modifications
+  // are needed is done while holding the render lock only, thereby avoiding
+  // that the capture thread blocks the render thread.
+  // The struct is modified in a single-threaded manner by holding both the
+  // render and capture locks.
+  int MaybeInitialize(const ProcessingConfig& config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  int MaybeInitializeRender(const ProcessingConfig& processing_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  int MaybeInitializeCapture(const ProcessingConfig& processing_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  // Method for checking for the need of conversion. Accesses the formats
+  // structs in a read manner but the requirement for the render lock to be held
+  // was added as it currently anyway is always called in that manner.
+  bool rev_conversion_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  bool render_check_rev_conversion_needed() const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+  // Methods requiring APM running in a single-threaded manner.
+  // Are called with both the render and capture locks already
+  // acquired.
+  void InitializeExperimentalAgc()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeTransient()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeBeamformer()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeIntelligibility()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void InitializeHighPassFilter()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeNoiseSuppression()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeLevelEstimator()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeVoiceDetection()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  int InitializeLocked(const ProcessingConfig& config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+
+  // Capture-side exclusive methods possibly running APM in a multi-threaded
+  // manner that are called with the render lock already acquired.
+  int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool output_copy_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool synthesis_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  bool analysis_needed(bool is_data_processed) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Render-side exclusive methods possibly running APM in a multi-threaded
+  // manner that are called with the render lock already acquired.
+  // TODO(ekm): Remove once all clients updated to new interface.
+  int AnalyzeReverseStreamLocked(const float* const* src,
+                                 const StreamConfig& input_config,
+                                 const StreamConfig& output_config)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+  int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
+// Debug dump methods that are internal and called without locks.
+// TODO(peah): Make thread safe.
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
-  int WriteMessageToDebugFile();
-  int WriteInitMessage();
-  rtc::scoped_ptr<FileWrapper> debug_file_;
-  rtc::scoped_ptr<audioproc::Event> event_msg_;  // Protobuf message.
-  std::string event_str_;  // Memory for protobuf serialization.
+  static int WriteMessageToDebugFile(FileWrapper* debug_file,
+                                     rtc::CriticalSection* crit_debug,
+                                     ApmDebugDumpThreadState* debug_state);
+  int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+
+  // Writes Config message. If not |forced|, only writes the current config if
+  // it is different from the last saved one; if |forced|, writes the config
+  // regardless of the last saved.
+  int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+
+  // Critical section.
+  mutable rtc::CriticalSection crit_debug_;
+
+  // Debug dump state.
+  ApmDebugDumpState debug_dump_;
 #endif
 
-  AudioFormat fwd_in_format_;
-  // This one is an AudioRate, because the forward processing number of channels
-  // is mutable and is tracked by the capture_audio_.
-  AudioRate fwd_proc_format_;
-  AudioFormat fwd_out_format_;
-  AudioFormat rev_in_format_;
-  AudioFormat rev_proc_format_;
-  int split_rate_;
+  // Critical sections.
+  mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  mutable rtc::CriticalSection crit_capture_;
 
-  int stream_delay_ms_;
-  int delay_offset_ms_;
-  bool was_stream_delay_set_;
+  // Structs containing the pointers to the submodules.
+  rtc::scoped_ptr<ApmPublicSubmodules> public_submodules_;
+  rtc::scoped_ptr<ApmPrivateSubmodules> private_submodules_
+      GUARDED_BY(crit_capture_);
 
-  bool output_will_be_muted_ GUARDED_BY(crit_);
+  // State that is written to while holding both the render and capture locks
+  // but can be read without any lock being held.
+  // As this is only accessed internally of APM, and all internal methods in APM
+  // either are holding the render or capture locks, this construct is safe as
+  // it is not possible to read the variables while writing them.
+  struct ApmFormatState {
+    ApmFormatState()
+        :  // Format of processing streams at input/output call sites.
+          api_format({{{kSampleRate16kHz, 1, false},
+                       {kSampleRate16kHz, 1, false},
+                       {kSampleRate16kHz, 1, false},
+                       {kSampleRate16kHz, 1, false}}}),
+          rev_proc_format(kSampleRate16kHz, 1) {}
+    ProcessingConfig api_format;
+    StreamConfig rev_proc_format;
+  } formats_;
 
-  bool key_pressed_;
+  // APM constants.
+  const struct ApmConstants {
+    ApmConstants(int agc_startup_min_volume,
+                 bool use_new_agc,
+                 bool intelligibility_enabled)
+        :  // Format of processing streams at input/output call sites.
+          agc_startup_min_volume(agc_startup_min_volume),
+          use_new_agc(use_new_agc),
+          intelligibility_enabled(intelligibility_enabled) {}
+    int agc_startup_min_volume;
+    bool use_new_agc;
+    bool intelligibility_enabled;
+  } constants_;
 
-  // Only set through the constructor's Config parameter.
-  const bool use_new_agc_;
-  rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
+  struct ApmCaptureState {
+    ApmCaptureState(bool transient_suppressor_enabled,
+                    const std::vector<Point>& array_geometry,
+                    SphericalPointf target_direction)
+        : aec_system_delay_jumps(-1),
+          delay_offset_ms(0),
+          was_stream_delay_set(false),
+          last_stream_delay_ms(0),
+          last_aec_system_delay_ms(0),
+          stream_delay_jumps(-1),
+          output_will_be_muted(false),
+          key_pressed(false),
+          transient_suppressor_enabled(transient_suppressor_enabled),
+          array_geometry(array_geometry),
+          target_direction(target_direction),
+          fwd_proc_format(kSampleRate16kHz),
+          split_rate(kSampleRate16kHz) {}
+    int aec_system_delay_jumps;
+    int delay_offset_ms;
+    bool was_stream_delay_set;
+    int last_stream_delay_ms;
+    int last_aec_system_delay_ms;
+    int stream_delay_jumps;
+    bool output_will_be_muted;
+    bool key_pressed;
+    bool transient_suppressor_enabled;
+    std::vector<Point> array_geometry;
+    SphericalPointf target_direction;
+    rtc::scoped_ptr<AudioBuffer> capture_audio;
+    // Only the rate and samples fields of fwd_proc_format_ are used because the
+    // forward processing number of channels is mutable and is tracked by the
+    // capture_audio_.
+    StreamConfig fwd_proc_format;
+    int split_rate;
+  } capture_ GUARDED_BY(crit_capture_);
 
-  bool transient_suppressor_enabled_;
-  rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
-  const bool beamformer_enabled_;
-  rtc::scoped_ptr<Beamformer<float>> beamformer_;
-  const std::vector<Point> array_geometry_;
+  struct ApmCaptureNonLockedState {
+    ApmCaptureNonLockedState(bool beamformer_enabled)
+        : fwd_proc_format(kSampleRate16kHz),
+          split_rate(kSampleRate16kHz),
+          stream_delay_ms(0),
+          beamformer_enabled(beamformer_enabled) {}
+    // Only the rate and samples fields of fwd_proc_format_ are used because the
+    // forward processing number of channels is mutable and is tracked by the
+    // capture_audio_.
+    StreamConfig fwd_proc_format;
+    int split_rate;
+    int stream_delay_ms;
+    bool beamformer_enabled;
+  } capture_nonlocked_;
 
-  const bool supports_48kHz_;
+  struct ApmRenderState {
+    rtc::scoped_ptr<AudioConverter> render_converter;
+    rtc::scoped_ptr<AudioBuffer> render_audio;
+  } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
new file mode 100644
index 0000000000..e1e6a310a5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -0,0 +1,1133 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/random.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+namespace {
+
+class AudioProcessingImplLockTest;
+
+// Type of the render thread APM API call to use in the test.
+enum class RenderApiImpl {
+  ProcessReverseStreamImpl1,
+  ProcessReverseStreamImpl2,
+  AnalyzeReverseStreamImpl1,
+  AnalyzeReverseStreamImpl2
+};
+
+// Type of the capture thread APM API call to use in the test.
+enum class CaptureApiImpl {
+  ProcessStreamImpl1,
+  ProcessStreamImpl2,
+  ProcessStreamImpl3
+};
+
+// The runtime parameter setting scheme to use in the test.
+enum class RuntimeParameterSettingScheme {
+  SparseStreamMetadataChangeScheme,
+  ExtremeStreamMetadataChangeScheme,
+  FixedMonoStreamMetadataScheme,
+  FixedStereoStreamMetadataScheme
+};
+
+// Variant of echo canceller settings to use in the test.
+enum class AecType {
+  BasicWebRtcAecSettings,
+  AecTurnedOff,
+  BasicWebRtcAecSettingsWithExtentedFilter,
+  BasicWebRtcAecSettingsWithDelayAgnosticAec,
+  BasicWebRtcAecSettingsWithAecMobile
+};
+
+// Thread-safe random number generator wrapper.
+class RandomGenerator {
+ public:
+  RandomGenerator() : rand_gen_(42U) {}
+
+  int RandInt(int min, int max) {
+    rtc::CritScope cs(&crit_);
+    return rand_gen_.Rand(min, max);
+  }
+
+  int RandInt(int max) {
+    rtc::CritScope cs(&crit_);
+    return rand_gen_.Rand(max);
+  }
+
+  float RandFloat() {
+    rtc::CritScope cs(&crit_);
+    return rand_gen_.Rand<float>();
+  }
+
+ private:
+  rtc::CriticalSection crit_;
+  Random rand_gen_ GUARDED_BY(crit_);
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(int max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+  }
+
+  AudioFrame frame;
+  std::vector<float*> output_frame;
+  std::vector<float> output_frame_channels;
+  AudioProcessing::ChannelLayout output_channel_layout =
+      AudioProcessing::ChannelLayout::kMono;
+  int input_sample_rate_hz = 16000;
+  int input_number_of_channels = -1;
+  std::vector<float*> input_frame;
+  std::vector<float> input_framechannels;
+  AudioProcessing::ChannelLayout input_channel_layout =
+      AudioProcessing::ChannelLayout::kMono;
+  int output_sample_rate_hz = 16000;
+  int output_number_of_channels = -1;
+  StreamConfig input_stream_config;
+  StreamConfig output_stream_config;
+  int input_samples_per_channel = -1;
+  int output_samples_per_channel = -1;
+};
+
+// The configuration for the test.
+struct TestConfig {
+  // Test case generator for the test configurations to use in the brief tests.
+  static std::vector<TestConfig> GenerateBriefTestConfigs() {
+    std::vector<TestConfig> test_configs;
+    AecType aec_types[] = {AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec,
+                           AecType::BasicWebRtcAecSettingsWithAecMobile};
+    for (auto aec_type : aec_types) {
+      TestConfig test_config;
+      test_config.aec_type = aec_type;
+
+      test_config.min_number_of_calls = 300;
+
+      // Perform tests only with the extreme runtime parameter setting scheme.
+      test_config.runtime_parameter_setting_scheme =
+          RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme;
+
+      // Only test 16 kHz for this test suite.
+      test_config.initial_sample_rate_hz = 16000;
+
+      // Create test config for the second processing API function set.
+      test_config.render_api_function =
+          RenderApiImpl::ProcessReverseStreamImpl2;
+      test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl2;
+
+      // Create test config for the first processing API function set.
+      test_configs.push_back(test_config);
+      test_config.render_api_function =
+          RenderApiImpl::AnalyzeReverseStreamImpl2;
+      test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl3;
+      test_configs.push_back(test_config);
+    }
+
+    // Return the created test configurations.
+    return test_configs;
+  }
+
+  // Test case generator for the test configurations to use in the extensive
+  // tests.
+  static std::vector<TestConfig> GenerateExtensiveTestConfigs() {
+    // Lambda functions for the test config generation.
+    auto add_processing_apis = [](TestConfig test_config) {
+      struct AllowedApiCallCombinations {
+        RenderApiImpl render_api;
+        CaptureApiImpl capture_api;
+      };
+
+      const AllowedApiCallCombinations api_calls[] = {
+          {RenderApiImpl::ProcessReverseStreamImpl1,
+           CaptureApiImpl::ProcessStreamImpl1},
+          {RenderApiImpl::AnalyzeReverseStreamImpl1,
+           CaptureApiImpl::ProcessStreamImpl1},
+          {RenderApiImpl::ProcessReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl2},
+          {RenderApiImpl::ProcessReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl3},
+          {RenderApiImpl::AnalyzeReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl2},
+          {RenderApiImpl::AnalyzeReverseStreamImpl2,
+           CaptureApiImpl::ProcessStreamImpl3}};
+      std::vector<TestConfig> out;
+      for (auto api_call : api_calls) {
+        test_config.render_api_function = api_call.render_api;
+        test_config.capture_api_function = api_call.capture_api;
+        out.push_back(test_config);
+      }
+      return out;
+    };
+
+    auto add_aec_settings = [](const std::vector<TestConfig>& in) {
+      std::vector<TestConfig> out;
+      AecType aec_types[] = {
+          AecType::BasicWebRtcAecSettings, AecType::AecTurnedOff,
+          AecType::BasicWebRtcAecSettingsWithExtentedFilter,
+          AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec,
+          AecType::BasicWebRtcAecSettingsWithAecMobile};
+      for (auto test_config : in) {
+        for (auto aec_type : aec_types) {
+          test_config.aec_type = aec_type;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    auto add_settings_scheme = [](const std::vector<TestConfig>& in) {
+      std::vector<TestConfig> out;
+      RuntimeParameterSettingScheme schemes[] = {
+          RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme,
+          RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme,
+          RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme,
+          RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme};
+
+      for (auto test_config : in) {
+        for (auto scheme : schemes) {
+          test_config.runtime_parameter_setting_scheme = scheme;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    auto add_sample_rates = [](const std::vector<TestConfig>& in) {
+      const int sample_rates[] = {8000, 16000, 32000, 48000};
+
+      std::vector<TestConfig> out;
+      for (auto test_config : in) {
+        auto available_rates =
+            (test_config.aec_type ==
+                     AecType::BasicWebRtcAecSettingsWithAecMobile
+                 ? rtc::ArrayView<const int>(sample_rates, 2)
+                 : rtc::ArrayView<const int>(sample_rates));
+
+        for (auto rate : available_rates) {
+          test_config.initial_sample_rate_hz = rate;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    // Generate test configurations of the relevant combinations of the
+    // parameters to
+    // test.
+    TestConfig test_config;
+    test_config.min_number_of_calls = 10000;
+    return add_sample_rates(add_settings_scheme(
+        add_aec_settings(add_processing_apis(test_config))));
+  }
+
+  RenderApiImpl render_api_function = RenderApiImpl::ProcessReverseStreamImpl2;
+  CaptureApiImpl capture_api_function = CaptureApiImpl::ProcessStreamImpl2;
+  RuntimeParameterSettingScheme runtime_parameter_setting_scheme =
+      RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme;
+  int initial_sample_rate_hz = 16000;
+  AecType aec_type = AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec;
+  int min_number_of_calls = 300;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() {
+    rtc::CritScope cs(&crit_);
+    render_count++;
+  }
+
+  void IncreaseCaptureCounter() {
+    rtc::CritScope cs(&crit_);
+    capture_count++;
+  }
+
+  int GetCaptureCounter() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count;
+  }
+
+  int GetRenderCounter() const {
+    rtc::CritScope cs(&crit_);
+    return render_count;
+  }
+
+  int CaptureMinusRenderCounters() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count - render_count;
+  }
+
+  int RenderMinusCaptureCounters() const {
+    return -CaptureMinusRenderCounters();
+  }
+
+  bool BothCountersExceedeThreshold(int threshold) {
+    rtc::CritScope cs(&crit_);
+    return (render_count > threshold && capture_count > threshold);
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  int render_count GUARDED_BY(crit_) = 0;
+  int capture_count GUARDED_BY(crit_) = 0;
+};
+
+// Class for handling the capture side processing.
+class CaptureProcessor {
+ public:
+  CaptureProcessor(int max_frame_size,
+                   RandomGenerator* rand_gen,
+                   rtc::Event* render_call_event,
+                   rtc::Event* capture_call_event,
+                   FrameCounters* shared_counters_state,
+                   AudioProcessingImplLockTest* test_framework,
+                   TestConfig* test_config,
+                   AudioProcessing* apm);
+  bool Process();
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const float kCaptureInputFloatLevel;
+  static const int kCaptureInputFixLevel = 1024;
+
+  void PrepareFrame();
+  void CallApmCaptureSide();
+  void ApplyRuntimeSettingScheme();
+
+  RandomGenerator* const rand_gen_ = nullptr;
+  rtc::Event* const render_call_event_ = nullptr;
+  rtc::Event* const capture_call_event_ = nullptr;
+  FrameCounters* const frame_counters_ = nullptr;
+  AudioProcessingImplLockTest* const test_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* const apm_ = nullptr;
+  AudioFrameData frame_data_;
+};
+
+// Class for handling the stats processing.
+class StatsProcessor {
+ public:
+  StatsProcessor(RandomGenerator* rand_gen,
+                 TestConfig* test_config,
+                 AudioProcessing* apm);
+  bool Process();
+
+ private:
+  RandomGenerator* rand_gen_ = nullptr;
+  TestConfig* test_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+};
+
+// Class for handling the render side processing.
+class RenderProcessor {
+ public:
+  RenderProcessor(int max_frame_size,
+                  RandomGenerator* rand_gen,
+                  rtc::Event* render_call_event,
+                  rtc::Event* capture_call_event,
+                  FrameCounters* shared_counters_state,
+                  AudioProcessingImplLockTest* test_framework,
+                  TestConfig* test_config,
+                  AudioProcessing* apm);
+  bool Process();
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const int kRenderInputFixLevel = 16384;
+  static const float kRenderInputFloatLevel;
+
+  void PrepareFrame();
+  void CallApmRenderSide();
+  void ApplyRuntimeSettingScheme();
+
+  RandomGenerator* const rand_gen_ = nullptr;
+  rtc::Event* const render_call_event_ = nullptr;
+  rtc::Event* const capture_call_event_ = nullptr;
+  FrameCounters* const frame_counters_ = nullptr;
+  AudioProcessingImplLockTest* const test_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* const apm_ = nullptr;
+  AudioFrameData frame_data_;
+  bool first_render_call_ = true;
+};
+
+class AudioProcessingImplLockTest
+    : public ::testing::TestWithParam<TestConfig> {
+ public:
+  AudioProcessingImplLockTest();
+  bool RunTest();
+  bool MaybeEndTest();
+
+ private:
+  static const int kTestTimeOutLimit = 10 * 60 * 1000;
+  static const int kMaxFrameSize = 480;
+
+  // ::testing::TestWithParam<> implementation
+  void SetUp() override;
+  void TearDown() override;
+
+  // Thread callback for the render thread
+  static bool RenderProcessorThreadFunc(void* context) {
+    return reinterpret_cast<AudioProcessingImplLockTest*>(context)
+        ->render_thread_state_.Process();
+  }
+
+  // Thread callback for the capture thread
+  static bool CaptureProcessorThreadFunc(void* context) {
+    return reinterpret_cast<AudioProcessingImplLockTest*>(context)
+        ->capture_thread_state_.Process();
+  }
+
+  // Thread callback for the stats thread
+  static bool StatsProcessorThreadFunc(void* context) {
+    return reinterpret_cast<AudioProcessingImplLockTest*>(context)
+        ->stats_thread_state_.Process();
+  }
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool TestDone() {
+    return frame_counters_.BothCountersExceedeThreshold(
+        test_config_.min_number_of_calls);
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    render_thread_.Start();
+    render_thread_.SetPriority(rtc::kRealtimePriority);
+    capture_thread_.Start();
+    capture_thread_.SetPriority(rtc::kRealtimePriority);
+    stats_thread_.Start();
+    stats_thread_.SetPriority(rtc::kNormalPriority);
+  }
+
+  // Event handlers for the test.
+  rtc::Event test_complete_;
+  rtc::Event render_call_event_;
+  rtc::Event capture_call_event_;
+
+  // Thread related variables.
+  rtc::PlatformThread render_thread_;
+  rtc::PlatformThread capture_thread_;
+  rtc::PlatformThread stats_thread_;
+  mutable RandomGenerator rand_gen_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+  TestConfig test_config_;
+  FrameCounters frame_counters_;
+  RenderProcessor render_thread_state_;
+  CaptureProcessor capture_thread_state_;
+  StatsProcessor stats_thread_state_;
+};
+
+// Sleeps a random time between 0 and max_sleep milliseconds.
+void SleepRandomMs(int max_sleep, RandomGenerator* rand_gen) {
+  int sleeptime = rand_gen->RandInt(0, max_sleep);
+  SleepMs(sleeptime);
+}
+
+// Populates a float audio frame with random data.
+void PopulateAudioFrame(float** frame,
+                        float amplitude,
+                        size_t num_channels,
+                        size_t samples_per_channel,
+                        RandomGenerator* rand_gen) {
+  for (size_t ch = 0; ch < num_channels; ch++) {
+    for (size_t k = 0; k < samples_per_channel; k++) {
+      // Store random 16 bit quantized float number between +-amplitude.
+      frame[ch][k] = amplitude * (2 * rand_gen->RandFloat() - 1);
+    }
+  }
+}
+
+// Populates an audioframe frame of AudioFrame type with random data.
+void PopulateAudioFrame(AudioFrame* frame,
+                        int16_t amplitude,
+                        RandomGenerator* rand_gen) {
+  ASSERT_GT(amplitude, 0);
+  ASSERT_LE(amplitude, 32767);
+  for (size_t ch = 0; ch < frame->num_channels_; ch++) {
+    for (size_t k = 0; k < frame->samples_per_channel_; k++) {
+      // Store random 16 bit number between -(amplitude+1) and
+      // amplitude.
+      frame->data_[k * ch] =
+          rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
+    }
+  }
+}
+
+AudioProcessingImplLockTest::AudioProcessingImplLockTest()
+    : test_complete_(false, false),
+      render_call_event_(false, false),
+      capture_call_event_(false, false),
+      render_thread_(RenderProcessorThreadFunc, this, "render"),
+      capture_thread_(CaptureProcessorThreadFunc, this, "capture"),
+      stats_thread_(StatsProcessorThreadFunc, this, "stats"),
+      apm_(AudioProcessingImpl::Create()),
+      render_thread_state_(kMaxFrameSize,
+                           &rand_gen_,
+                           &render_call_event_,
+                           &capture_call_event_,
+                           &frame_counters_,
+                           this,
+                           &test_config_,
+                           apm_.get()),
+      capture_thread_state_(kMaxFrameSize,
+                            &rand_gen_,
+                            &render_call_event_,
+                            &capture_call_event_,
+                            &frame_counters_,
+                            this,
+                            &test_config_,
+                            apm_.get()),
+      stats_thread_state_(&rand_gen_, &test_config_, apm_.get()) {}
+
+// Run the test with a timeout.
+bool AudioProcessingImplLockTest::RunTest() {
+  StartThreads();
+  return test_complete_.Wait(kTestTimeOutLimit);
+}
+
+bool AudioProcessingImplLockTest::MaybeEndTest() {
+  if (HasFatalFailure() || TestDone()) {
+    test_complete_.Set();
+    return true;
+  }
+  return false;
+}
+
+// Setup of test and APM.
+void AudioProcessingImplLockTest::SetUp() {
+  test_config_ = static_cast<TestConfig>(GetParam());
+
+  ASSERT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
+  ASSERT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+
+  ASSERT_EQ(apm_->kNoError,
+            apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+  ASSERT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
+
+  ASSERT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
+  ASSERT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
+
+  Config config;
+  if (test_config_.aec_type == AecType::AecTurnedOff) {
+    ASSERT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
+  } else if (test_config_.aec_type ==
+             AecType::BasicWebRtcAecSettingsWithAecMobile) {
+    ASSERT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
+  } else {
+    ASSERT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
+    ASSERT_EQ(apm_->kNoError, apm_->echo_cancellation()->enable_metrics(true));
+    ASSERT_EQ(apm_->kNoError,
+              apm_->echo_cancellation()->enable_delay_logging(true));
+
+    config.Set<ExtendedFilter>(
+        new ExtendedFilter(test_config_.aec_type ==
+                           AecType::BasicWebRtcAecSettingsWithExtentedFilter));
+
+    config.Set<DelayAgnostic>(
+        new DelayAgnostic(test_config_.aec_type ==
+                          AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec));
+
+    apm_->SetExtraOptions(config);
+  }
+}
+
+void AudioProcessingImplLockTest::TearDown() {
+  render_call_event_.Set();
+  capture_call_event_.Set();
+  render_thread_.Stop();
+  capture_thread_.Stop();
+  stats_thread_.Stop();
+}
+
+StatsProcessor::StatsProcessor(RandomGenerator* rand_gen,
+                               TestConfig* test_config,
+                               AudioProcessing* apm)
+    : rand_gen_(rand_gen), test_config_(test_config), apm_(apm) {}
+
+// Implements the callback functionality for the statistics
+// collection thread.
+bool StatsProcessor::Process() {
+  SleepRandomMs(100, rand_gen_);
+
+  EXPECT_EQ(apm_->echo_cancellation()->is_enabled(),
+            ((test_config_->aec_type != AecType::AecTurnedOff) &&
+             (test_config_->aec_type !=
+              AecType::BasicWebRtcAecSettingsWithAecMobile)));
+  apm_->echo_cancellation()->stream_drift_samples();
+  EXPECT_EQ(apm_->echo_control_mobile()->is_enabled(),
+            (test_config_->aec_type != AecType::AecTurnedOff) &&
+                (test_config_->aec_type ==
+                 AecType::BasicWebRtcAecSettingsWithAecMobile));
+  EXPECT_TRUE(apm_->gain_control()->is_enabled());
+  apm_->gain_control()->stream_analog_level();
+  EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
+
+  // The below return values are not testable.
+  apm_->noise_suppression()->speech_probability();
+  apm_->voice_detection()->is_enabled();
+
+  return true;
+}
+
+const float CaptureProcessor::kCaptureInputFloatLevel = 0.03125f;
+
+CaptureProcessor::CaptureProcessor(int max_frame_size,
+                                   RandomGenerator* rand_gen,
+                                   rtc::Event* render_call_event,
+                                   rtc::Event* capture_call_event,
+                                   FrameCounters* shared_counters_state,
+                                   AudioProcessingImplLockTest* test_framework,
+                                   TestConfig* test_config,
+                                   AudioProcessing* apm)
+    : rand_gen_(rand_gen),
+      render_call_event_(render_call_event),
+      capture_call_event_(capture_call_event),
+      frame_counters_(shared_counters_state),
+      test_(test_framework),
+      test_config_(test_config),
+      apm_(apm),
+      frame_data_(max_frame_size) {}
+
+// Implements the callback functionality for the capture thread.
+bool CaptureProcessor::Process() {
+  // Sleep a random time to simulate thread jitter.
+  SleepRandomMs(3, rand_gen_);
+
+  // Check whether the test is done.
+  if (test_->MaybeEndTest()) {
+    return false;
+  }
+
+  // Ensure that the number of render and capture calls do not
+  // differ too much.
+  if (frame_counters_->CaptureMinusRenderCounters() > kMaxCallDifference) {
+    render_call_event_->Wait(rtc::Event::kForever);
+  }
+
+  // Apply any specified capture side APM non-processing runtime calls.
+  ApplyRuntimeSettingScheme();
+
+  // Apply the capture side processing call.
+  CallApmCaptureSide();
+
+  // Increase the number of capture-side calls.
+  frame_counters_->IncreaseCaptureCounter();
+
+  // Flag to the render thread that another capture API call has occurred
+  // by triggering this threads call event.
+  capture_call_event_->Set();
+
+  return true;
+}
+
+// Prepares a frame with relevant audio data and metadata.
+void CaptureProcessor::PrepareFrame() {
+  // Restrict to a common fixed sample rate if the AudioFrame
+  // interface is used.
+  if (test_config_->capture_api_function ==
+      CaptureApiImpl::ProcessStreamImpl1) {
+    frame_data_.input_sample_rate_hz = test_config_->initial_sample_rate_hz;
+    frame_data_.output_sample_rate_hz = test_config_->initial_sample_rate_hz;
+  }
+
+  // Prepare the audioframe data and metadata.
+  frame_data_.input_samples_per_channel =
+      frame_data_.input_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.frame.sample_rate_hz_ = frame_data_.input_sample_rate_hz;
+  frame_data_.frame.num_channels_ = frame_data_.input_number_of_channels;
+  frame_data_.frame.samples_per_channel_ =
+      frame_data_.input_samples_per_channel;
+  PopulateAudioFrame(&frame_data_.frame, kCaptureInputFixLevel, rand_gen_);
+
+  // Prepare the float audio input data and metadata.
+  frame_data_.input_stream_config.set_sample_rate_hz(
+      frame_data_.input_sample_rate_hz);
+  frame_data_.input_stream_config.set_num_channels(
+      frame_data_.input_number_of_channels);
+  frame_data_.input_stream_config.set_has_keyboard(false);
+  PopulateAudioFrame(&frame_data_.input_frame[0], kCaptureInputFloatLevel,
+                     frame_data_.input_number_of_channels,
+                     frame_data_.input_samples_per_channel, rand_gen_);
+  frame_data_.input_channel_layout =
+      (frame_data_.input_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+
+  // Prepare the float audio output data and metadata.
+  frame_data_.output_samples_per_channel =
+      frame_data_.output_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.output_stream_config.set_sample_rate_hz(
+      frame_data_.output_sample_rate_hz);
+  frame_data_.output_stream_config.set_num_channels(
+      frame_data_.output_number_of_channels);
+  frame_data_.output_stream_config.set_has_keyboard(false);
+  frame_data_.output_channel_layout =
+      (frame_data_.output_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+}
+
+// Applies the capture side processing API call.
+void CaptureProcessor::CallApmCaptureSide() {
+  // Prepare a proper capture side processing API call input.
+  PrepareFrame();
+
+  // Set the stream delay
+  apm_->set_stream_delay_ms(30);
+
+  // Call the specified capture side API processing method.
+  int result = AudioProcessing::kNoError;
+  switch (test_config_->capture_api_function) {
+    case CaptureApiImpl::ProcessStreamImpl1:
+      result = apm_->ProcessStream(&frame_data_.frame);
+      break;
+    case CaptureApiImpl::ProcessStreamImpl2:
+      result = apm_->ProcessStream(
+          &frame_data_.input_frame[0], frame_data_.input_samples_per_channel,
+          frame_data_.input_sample_rate_hz, frame_data_.input_channel_layout,
+          frame_data_.output_sample_rate_hz, frame_data_.output_channel_layout,
+          &frame_data_.output_frame[0]);
+      break;
+    case CaptureApiImpl::ProcessStreamImpl3:
+      result = apm_->ProcessStream(
+          &frame_data_.input_frame[0], frame_data_.input_stream_config,
+          frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+      break;
+    default:
+      FAIL();
+  }
+
+  // Check the return code for error.
+  ASSERT_EQ(AudioProcessing::kNoError, result);
+}
+
+// Applies any runtime capture APM API calls and audio stream characteristics
+// specified by the scheme for the test.
+void CaptureProcessor::ApplyRuntimeSettingScheme() {
+  const int capture_count_local = frame_counters_->GetCaptureCounter();
+
+  // Update the number of channels and sample rates for the input and output.
+  // Note that the counts frequencies for when to set parameters
+  // are set using prime numbers in order to ensure that the
+  // permutation scheme in the parameter setting changes.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+      if (capture_count_local == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (capture_count_local % 11 == 0)
+        frame_data_.input_sample_rate_hz = 32000;
+      else if (capture_count_local % 73 == 0)
+        frame_data_.input_sample_rate_hz = 48000;
+      else if (capture_count_local % 89 == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (capture_count_local % 97 == 0)
+        frame_data_.input_sample_rate_hz = 8000;
+
+      if (capture_count_local == 0)
+        frame_data_.input_number_of_channels = 1;
+      else if (capture_count_local % 4 == 0)
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+
+      if (capture_count_local == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (capture_count_local % 5 == 0)
+        frame_data_.output_sample_rate_hz = 32000;
+      else if (capture_count_local % 47 == 0)
+        frame_data_.output_sample_rate_hz = 48000;
+      else if (capture_count_local % 53 == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (capture_count_local % 71 == 0)
+        frame_data_.output_sample_rate_hz = 8000;
+
+      if (capture_count_local == 0)
+        frame_data_.output_number_of_channels = 1;
+      else if (capture_count_local % 8 == 0)
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+      if (capture_count_local % 2 == 0) {
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+      } else {
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.input_sample_rate_hz == 8000)
+          frame_data_.input_sample_rate_hz = 16000;
+        else if (frame_data_.input_sample_rate_hz == 16000)
+          frame_data_.input_sample_rate_hz = 32000;
+        else if (frame_data_.input_sample_rate_hz == 32000)
+          frame_data_.input_sample_rate_hz = 48000;
+        else if (frame_data_.input_sample_rate_hz == 48000)
+          frame_data_.input_sample_rate_hz = 8000;
+
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.output_sample_rate_hz == 8000)
+          frame_data_.output_sample_rate_hz = 16000;
+        else if (frame_data_.output_sample_rate_hz == 16000)
+          frame_data_.output_sample_rate_hz = 32000;
+        else if (frame_data_.output_sample_rate_hz == 32000)
+          frame_data_.output_sample_rate_hz = 48000;
+        else if (frame_data_.output_sample_rate_hz == 48000)
+          frame_data_.output_sample_rate_hz = 8000;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      if (capture_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (capture_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 2;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 2;
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Call any specified runtime APM setter and
+  // getter calls.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (capture_count_local % 2 == 0) {
+        ASSERT_EQ(AudioProcessing::Error::kNoError,
+                  apm_->set_stream_delay_ms(30));
+        apm_->set_stream_key_pressed(true);
+        apm_->set_delay_offset_ms(15);
+        EXPECT_EQ(apm_->delay_offset_ms(), 15);
+      } else {
+        ASSERT_EQ(AudioProcessing::Error::kNoError,
+                  apm_->set_stream_delay_ms(50));
+        apm_->set_stream_key_pressed(false);
+        apm_->set_delay_offset_ms(20);
+        EXPECT_EQ(apm_->delay_offset_ms(), 20);
+        apm_->delay_offset_ms();
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Restric the number of output channels not to exceed
+  // the number of input channels.
+  frame_data_.output_number_of_channels =
+      std::min(frame_data_.output_number_of_channels,
+               frame_data_.input_number_of_channels);
+}
+
+const float RenderProcessor::kRenderInputFloatLevel = 0.5f;
+
+RenderProcessor::RenderProcessor(int max_frame_size,
+                                 RandomGenerator* rand_gen,
+                                 rtc::Event* render_call_event,
+                                 rtc::Event* capture_call_event,
+                                 FrameCounters* shared_counters_state,
+                                 AudioProcessingImplLockTest* test_framework,
+                                 TestConfig* test_config,
+                                 AudioProcessing* apm)
+    : rand_gen_(rand_gen),
+      render_call_event_(render_call_event),
+      capture_call_event_(capture_call_event),
+      frame_counters_(shared_counters_state),
+      test_(test_framework),
+      test_config_(test_config),
+      apm_(apm),
+      frame_data_(max_frame_size) {}
+
+// Implements the callback functionality for the render thread.
+bool RenderProcessor::Process() {
+  // Conditional wait to ensure that a capture call has been done
+  // before the first render call is performed (implicitly
+  // required by the APM API).
+  if (first_render_call_) {
+    capture_call_event_->Wait(rtc::Event::kForever);
+    first_render_call_ = false;
+  }
+
+  // Sleep a random time to simulate thread jitter.
+  SleepRandomMs(3, rand_gen_);
+
+  // Check whether the test is done.
+  if (test_->MaybeEndTest()) {
+    return false;
+  }
+
+  // Ensure that the number of render and capture calls do not
+  // differ too much.
+  if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+    capture_call_event_->Wait(rtc::Event::kForever);
+  }
+
+  // Apply any specified render side APM non-processing runtime calls.
+  ApplyRuntimeSettingScheme();
+
+  // Apply the render side processing call.
+  CallApmRenderSide();
+
+  // Increase the number of render-side calls.
+  frame_counters_->IncreaseRenderCounter();
+
+  // Flag to the capture thread that another render API call has occurred
+  // by triggering this threads call event.
+  render_call_event_->Set();
+  return true;
+}
+
+// Prepares the render side frame and the accompanying metadata
+// with the appropriate information.
+void RenderProcessor::PrepareFrame() {
+  // Restrict to a common fixed sample rate if the AudioFrame interface is
+  // used.
+  if ((test_config_->render_api_function ==
+       RenderApiImpl::AnalyzeReverseStreamImpl1) ||
+      (test_config_->render_api_function ==
+       RenderApiImpl::ProcessReverseStreamImpl1) ||
+      (test_config_->aec_type !=
+       AecType::BasicWebRtcAecSettingsWithAecMobile)) {
+    frame_data_.input_sample_rate_hz = test_config_->initial_sample_rate_hz;
+    frame_data_.output_sample_rate_hz = test_config_->initial_sample_rate_hz;
+  }
+
+  // Prepare the audioframe data and metadata
+  frame_data_.input_samples_per_channel =
+      frame_data_.input_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.frame.sample_rate_hz_ = frame_data_.input_sample_rate_hz;
+  frame_data_.frame.num_channels_ = frame_data_.input_number_of_channels;
+  frame_data_.frame.samples_per_channel_ =
+      frame_data_.input_samples_per_channel;
+  PopulateAudioFrame(&frame_data_.frame, kRenderInputFixLevel, rand_gen_);
+
+  // Prepare the float audio input data and metadata.
+  frame_data_.input_stream_config.set_sample_rate_hz(
+      frame_data_.input_sample_rate_hz);
+  frame_data_.input_stream_config.set_num_channels(
+      frame_data_.input_number_of_channels);
+  frame_data_.input_stream_config.set_has_keyboard(false);
+  PopulateAudioFrame(&frame_data_.input_frame[0], kRenderInputFloatLevel,
+                     frame_data_.input_number_of_channels,
+                     frame_data_.input_samples_per_channel, rand_gen_);
+  frame_data_.input_channel_layout =
+      (frame_data_.input_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+
+  // Prepare the float audio output data and metadata.
+  frame_data_.output_samples_per_channel =
+      frame_data_.output_sample_rate_hz * AudioProcessing::kChunkSizeMs / 1000;
+  frame_data_.output_stream_config.set_sample_rate_hz(
+      frame_data_.output_sample_rate_hz);
+  frame_data_.output_stream_config.set_num_channels(
+      frame_data_.output_number_of_channels);
+  frame_data_.output_stream_config.set_has_keyboard(false);
+  frame_data_.output_channel_layout =
+      (frame_data_.output_number_of_channels == 1
+           ? AudioProcessing::ChannelLayout::kMono
+           : AudioProcessing::ChannelLayout::kStereo);
+}
+
+// Makes the render side processing API call.
+void RenderProcessor::CallApmRenderSide() {
+  // Prepare a proper render side processing API call input.
+  PrepareFrame();
+
+  // Call the specified render side API processing method.
+  int result = AudioProcessing::kNoError;
+  switch (test_config_->render_api_function) {
+    case RenderApiImpl::ProcessReverseStreamImpl1:
+      result = apm_->ProcessReverseStream(&frame_data_.frame);
+      break;
+    case RenderApiImpl::ProcessReverseStreamImpl2:
+      result = apm_->ProcessReverseStream(
+          &frame_data_.input_frame[0], frame_data_.input_stream_config,
+          frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+      break;
+    case RenderApiImpl::AnalyzeReverseStreamImpl1:
+      result = apm_->AnalyzeReverseStream(&frame_data_.frame);
+      break;
+    case RenderApiImpl::AnalyzeReverseStreamImpl2:
+      result = apm_->AnalyzeReverseStream(
+          &frame_data_.input_frame[0], frame_data_.input_samples_per_channel,
+          frame_data_.input_sample_rate_hz, frame_data_.input_channel_layout);
+      break;
+    default:
+      FAIL();
+  }
+
+  // Check the return code for error.
+  ASSERT_EQ(AudioProcessing::kNoError, result);
+}
+
+// Applies any render capture side APM API calls and audio stream
+// characteristics
+// specified by the scheme for the test.
+void RenderProcessor::ApplyRuntimeSettingScheme() {
+  const int render_count_local = frame_counters_->GetRenderCounter();
+
+  // Update the number of channels and sample rates for the input and output.
+  // Note that the counts frequencies for when to set parameters
+  // are set using prime numbers in order to ensure that the
+  // permutation scheme in the parameter setting changes.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+      if (render_count_local == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (render_count_local % 47 == 0)
+        frame_data_.input_sample_rate_hz = 32000;
+      else if (render_count_local % 71 == 0)
+        frame_data_.input_sample_rate_hz = 48000;
+      else if (render_count_local % 79 == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (render_count_local % 83 == 0)
+        frame_data_.input_sample_rate_hz = 8000;
+
+      if (render_count_local == 0)
+        frame_data_.input_number_of_channels = 1;
+      else if (render_count_local % 4 == 0)
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+
+      if (render_count_local == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (render_count_local % 17 == 0)
+        frame_data_.output_sample_rate_hz = 32000;
+      else if (render_count_local % 19 == 0)
+        frame_data_.output_sample_rate_hz = 48000;
+      else if (render_count_local % 29 == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (render_count_local % 61 == 0)
+        frame_data_.output_sample_rate_hz = 8000;
+
+      if (render_count_local == 0)
+        frame_data_.output_number_of_channels = 1;
+      else if (render_count_local % 8 == 0)
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+      } else {
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.input_sample_rate_hz == 8000)
+          frame_data_.input_sample_rate_hz = 16000;
+        else if (frame_data_.input_sample_rate_hz == 16000)
+          frame_data_.input_sample_rate_hz = 32000;
+        else if (frame_data_.input_sample_rate_hz == 32000)
+          frame_data_.input_sample_rate_hz = 48000;
+        else if (frame_data_.input_sample_rate_hz == 48000)
+          frame_data_.input_sample_rate_hz = 8000;
+
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.output_sample_rate_hz == 8000)
+          frame_data_.output_sample_rate_hz = 16000;
+        else if (frame_data_.output_sample_rate_hz == 16000)
+          frame_data_.output_sample_rate_hz = 32000;
+        else if (frame_data_.output_sample_rate_hz == 32000)
+          frame_data_.output_sample_rate_hz = 48000;
+        else if (frame_data_.output_sample_rate_hz == 48000)
+          frame_data_.output_sample_rate_hz = 8000;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 2;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 2;
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Restric the number of output channels not to exceed
+  // the number of input channels.
+  frame_data_.output_number_of_channels =
+      std::min(frame_data_.output_number_of_channels,
+               frame_data_.input_number_of_channels);
+}
+
+}  // anonymous namespace
+
+TEST_P(AudioProcessingImplLockTest, LockTest) {
+  // Run test and verify that it did not time out.
+  ASSERT_TRUE(RunTest());
+}
+
+// Instantiate tests from the extreme test configuration set.
+INSTANTIATE_TEST_CASE_P(
+    DISABLED_AudioProcessingImplLockExtensive,
+    AudioProcessingImplLockTest,
+    ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
+
+INSTANTIATE_TEST_CASE_P(
+    AudioProcessingImplLockBrief,
+    AudioProcessingImplLockTest,
+    ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
index f4c36d0009..ed20daaa61 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -14,7 +14,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 using ::testing::Invoke;
 using ::testing::Return;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
new file mode 100644
index 0000000000..0c8c060ea3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -0,0 +1,724 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/random.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/testsupport/perf_test.h"
+
+namespace webrtc {
+
+namespace {
+
+static const bool kPrintAllDurations = false;
+
+class CallSimulator;
+
+// Type of the render thread APM API call to use in the test.
+enum class ProcessorType { kRender, kCapture };
+
+// Variant of APM processing settings to use in the test.
+enum class SettingsType {
+  kDefaultApmDesktop,
+  kDefaultApmMobile,
+  kDefaultApmDesktopAndBeamformer,
+  kDefaultApmDesktopAndIntelligibilityEnhancer,
+  kAllSubmodulesTurnedOff,
+  kDefaultDesktopApmWithoutDelayAgnostic,
+  kDefaultDesktopApmWithoutExtendedFilter
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(size_t max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+  }
+
+  std::vector<float> output_frame_channels;
+  std::vector<float*> output_frame;
+  std::vector<float> input_framechannels;
+  std::vector<float*> input_frame;
+  StreamConfig input_stream_config;
+  StreamConfig output_stream_config;
+};
+
+// The configuration for the test.
+struct SimulationConfig {
+  SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
+      : sample_rate_hz(sample_rate_hz),
+        simulation_settings(simulation_settings) {}
+
+  static std::vector<SimulationConfig> GenerateSimulationConfigs() {
+    std::vector<SimulationConfig> simulation_configs;
+#ifndef WEBRTC_ANDROID
+    const SettingsType desktop_settings[] = {
+        SettingsType::kDefaultApmDesktop, SettingsType::kAllSubmodulesTurnedOff,
+        SettingsType::kDefaultDesktopApmWithoutDelayAgnostic,
+        SettingsType::kDefaultDesktopApmWithoutExtendedFilter};
+
+    const int desktop_sample_rates[] = {8000, 16000, 32000, 48000};
+
+    for (auto sample_rate : desktop_sample_rates) {
+      for (auto settings : desktop_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    const SettingsType intelligibility_enhancer_settings[] = {
+        SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer};
+
+    const int intelligibility_enhancer_sample_rates[] = {8000, 16000, 32000,
+                                                         48000};
+
+    for (auto sample_rate : intelligibility_enhancer_sample_rates) {
+      for (auto settings : intelligibility_enhancer_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    const SettingsType beamformer_settings[] = {
+        SettingsType::kDefaultApmDesktopAndBeamformer};
+
+    const int beamformer_sample_rates[] = {8000, 16000, 32000, 48000};
+
+    for (auto sample_rate : beamformer_sample_rates) {
+      for (auto settings : beamformer_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+#endif
+
+    const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
+
+    const int mobile_sample_rates[] = {8000, 16000};
+
+    for (auto sample_rate : mobile_sample_rates) {
+      for (auto settings : mobile_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    return simulation_configs;
+  }
+
+  std::string SettingsDescription() const {
+    std::string description;
+    switch (simulation_settings) {
+      case SettingsType::kDefaultApmMobile:
+        description = "DefaultApmMobile";
+        break;
+      case SettingsType::kDefaultApmDesktop:
+        description = "DefaultApmDesktop";
+        break;
+      case SettingsType::kDefaultApmDesktopAndBeamformer:
+        description = "DefaultApmDesktopAndBeamformer";
+        break;
+      case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer:
+        description = "DefaultApmDesktopAndIntelligibilityEnhancer";
+        break;
+      case SettingsType::kAllSubmodulesTurnedOff:
+        description = "AllSubmodulesOff";
+        break;
+      case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic:
+        description = "DefaultDesktopApmWithoutDelayAgnostic";
+        break;
+      case SettingsType::kDefaultDesktopApmWithoutExtendedFilter:
+        description = "DefaultDesktopApmWithoutExtendedFilter";
+        break;
+    }
+    return description;
+  }
+
+  int sample_rate_hz = 16000;
+  SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() {
+    rtc::CritScope cs(&crit_);
+    render_count_++;
+  }
+
+  void IncreaseCaptureCounter() {
+    rtc::CritScope cs(&crit_);
+    capture_count_++;
+  }
+
+  int GetCaptureCounter() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count_;
+  }
+
+  int GetRenderCounter() const {
+    rtc::CritScope cs(&crit_);
+    return render_count_;
+  }
+
+  int CaptureMinusRenderCounters() const {
+    rtc::CritScope cs(&crit_);
+    return capture_count_ - render_count_;
+  }
+
+  int RenderMinusCaptureCounters() const {
+    return -CaptureMinusRenderCounters();
+  }
+
+  bool BothCountersExceedeThreshold(int threshold) const {
+    rtc::CritScope cs(&crit_);
+    return (render_count_ > threshold && capture_count_ > threshold);
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  int render_count_ GUARDED_BY(crit_) = 0;
+  int capture_count_ GUARDED_BY(crit_) = 0;
+};
+
+// Class that protects a flag using a lock.
+class LockedFlag {
+ public:
+  bool get_flag() const {
+    rtc::CritScope cs(&crit_);
+    return flag_;
+  }
+
+  void set_flag() {
+    rtc::CritScope cs(&crit_);
+    flag_ = true;
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  bool flag_ GUARDED_BY(crit_) = false;
+};
+
+// Parent class for the thread processors.
+class TimedThreadApiProcessor {
+ public:
+  TimedThreadApiProcessor(ProcessorType processor_type,
+                          Random* rand_gen,
+                          FrameCounters* shared_counters_state,
+                          LockedFlag* capture_call_checker,
+                          CallSimulator* test_framework,
+                          const SimulationConfig* simulation_config,
+                          AudioProcessing* apm,
+                          int num_durations_to_store,
+                          float input_level,
+                          int num_channels)
+      : rand_gen_(rand_gen),
+        frame_counters_(shared_counters_state),
+        capture_call_checker_(capture_call_checker),
+        test_(test_framework),
+        simulation_config_(simulation_config),
+        apm_(apm),
+        frame_data_(kMaxFrameSize),
+        clock_(webrtc::Clock::GetRealTimeClock()),
+        num_durations_to_store_(num_durations_to_store),
+        input_level_(input_level),
+        processor_type_(processor_type),
+        num_channels_(num_channels) {
+    api_call_durations_.reserve(num_durations_to_store_);
+  }
+
+  // Implements the callback functionality for the threads.
+  bool Process();
+
+  // Method for printing out the simulation statistics.
+  void print_processor_statistics(std::string processor_name) const {
+    const std::string modifier = "_api_call_duration";
+
+    // Lambda function for creating a test printout string.
+    auto create_mean_and_std_string = [](int64_t average,
+                                         int64_t standard_dev) {
+      std::string s = std::to_string(average);
+      s += ", ";
+      s += std::to_string(standard_dev);
+      return s;
+    };
+
+    const std::string sample_rate_name =
+        "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
+
+    webrtc::test::PrintResultMeanAndError(
+        "apm_timing", sample_rate_name, processor_name,
+        create_mean_and_std_string(GetDurationAverage(),
+                                   GetDurationStandardDeviation()),
+        "us", false);
+
+    if (kPrintAllDurations) {
+      std::string value_string = "";
+      for (int64_t duration : api_call_durations_) {
+        value_string += std::to_string(duration) + ",";
+      }
+      webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
+                                    processor_name, value_string, "us", false);
+    }
+  }
+
+  void AddDuration(int64_t duration) {
+    if (api_call_durations_.size() < num_durations_to_store_) {
+      api_call_durations_.push_back(duration);
+    }
+  }
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const int kMaxFrameSize = 480;
+  static const int kNumInitializationFrames = 5;
+
+  int64_t GetDurationStandardDeviation() const {
+    double variance = 0;
+    const int64_t average_duration = GetDurationAverage();
+    for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
+         k++) {
+      int64_t tmp = api_call_durations_[k] - average_duration;
+      variance += static_cast<double>(tmp * tmp);
+    }
+    const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
+                            kNumInitializationFrames;
+    return (denominator > 0
+                ? rtc::checked_cast<int64_t>(sqrt(variance / denominator))
+                : -1);
+  }
+
+  int64_t GetDurationAverage() const {
+    int64_t average_duration = 0;
+    for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
+         k++) {
+      average_duration += api_call_durations_[k];
+    }
+    const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
+                            kNumInitializationFrames;
+    return (denominator > 0 ? average_duration / denominator : -1);
+  }
+
+  int ProcessCapture() {
+    // Set the stream delay.
+    apm_->set_stream_delay_ms(30);
+
+    // Call and time the specified capture side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+
+    frame_counters_->IncreaseCaptureCounter();
+
+    AddDuration(end_time - start_time);
+
+    if (first_process_call_) {
+      // Flag that the capture side has been called at least once
+      // (needed to ensure that a capture call has been done
+      // before the first render call is performed (implicitly
+      // required by the APM API).
+      capture_call_checker_->set_flag();
+      first_process_call_ = false;
+    }
+    return result;
+  }
+
+  bool ReadyToProcessCapture() {
+    return (frame_counters_->CaptureMinusRenderCounters() <=
+            kMaxCallDifference);
+  }
+
+  int ProcessRender() {
+    // Call and time the specified render side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessReverseStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+    frame_counters_->IncreaseRenderCounter();
+
+    AddDuration(end_time - start_time);
+
+    return result;
+  }
+
+  bool ReadyToProcessRender() {
+    // Do not process until at least one capture call has been done.
+    // (implicitly required by the APM API).
+    if (first_process_call_ && !capture_call_checker_->get_flag()) {
+      return false;
+    }
+
+    // Ensure that the number of render and capture calls do not differ too
+    // much.
+    if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+      return false;
+    }
+
+    first_process_call_ = false;
+    return true;
+  }
+
+  void PrepareFrame() {
+    // Lambda function for populating a float multichannel audio frame
+    // with random data.
+    auto populate_audio_frame = [](float amplitude, size_t num_channels,
+                                   size_t samples_per_channel, Random* rand_gen,
+                                   float** frame) {
+      for (size_t ch = 0; ch < num_channels; ch++) {
+        for (size_t k = 0; k < samples_per_channel; k++) {
+          // Store random float number with a value between +-amplitude.
+          frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
+        }
+      }
+    };
+
+    // Prepare the audio input data and metadata.
+    frame_data_.input_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.input_stream_config.set_num_channels(num_channels_);
+    frame_data_.input_stream_config.set_has_keyboard(false);
+    populate_audio_frame(input_level_, num_channels_,
+                         (simulation_config_->sample_rate_hz *
+                          AudioProcessing::kChunkSizeMs / 1000),
+                         rand_gen_, &frame_data_.input_frame[0]);
+
+    // Prepare the float audio output data and metadata.
+    frame_data_.output_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.output_stream_config.set_num_channels(1);
+    frame_data_.output_stream_config.set_has_keyboard(false);
+  }
+
+  bool ReadyToProcess() {
+    switch (processor_type_) {
+      case ProcessorType::kRender:
+        return ReadyToProcessRender();
+        break;
+      case ProcessorType::kCapture:
+        return ReadyToProcessCapture();
+        break;
+    }
+
+    // Should not be reached, but the return statement is needed for the code to
+    // build successfully on Android.
+    RTC_NOTREACHED();
+    return false;
+  }
+
+  Random* rand_gen_ = nullptr;
+  FrameCounters* frame_counters_ = nullptr;
+  LockedFlag* capture_call_checker_ = nullptr;
+  CallSimulator* test_ = nullptr;
+  const SimulationConfig* const simulation_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+  AudioFrameData frame_data_;
+  webrtc::Clock* clock_;
+  const size_t num_durations_to_store_;
+  std::vector<int64_t> api_call_durations_;
+  const float input_level_;
+  bool first_process_call_ = true;
+  const ProcessorType processor_type_;
+  const int num_channels_ = 1;
+};
+
+// Class for managing the test simulation.
+class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
+ public:
+  CallSimulator()
+      : test_complete_(EventWrapper::Create()),
+        render_thread_(
+            new rtc::PlatformThread(RenderProcessorThreadFunc, this, "render")),
+        capture_thread_(new rtc::PlatformThread(CaptureProcessorThreadFunc,
+                                                this,
+                                                "capture")),
+        rand_gen_(42U),
+        simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
+
+  // Run the call simulation with a timeout.
+  EventTypeWrapper Run() {
+    StartThreads();
+
+    EventTypeWrapper result = test_complete_->Wait(kTestTimeout);
+
+    StopThreads();
+
+    render_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_render");
+    capture_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_capture");
+
+    return result;
+  }
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool MaybeEndTest() {
+    if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
+      test_complete_->Set();
+      return true;
+    }
+    return false;
+  }
+
+ private:
+  static const float kCaptureInputFloatLevel;
+  static const float kRenderInputFloatLevel;
+  static const int kMinNumFramesToProcess = 150;
+  static const int32_t kTestTimeout = 3 * 10 * kMinNumFramesToProcess;
+
+  // ::testing::TestWithParam<> implementation.
+  void TearDown() override { StopThreads(); }
+
+  // Stop all running threads.
+  void StopThreads() {
+    render_thread_->Stop();
+    capture_thread_->Stop();
+  }
+
+  // Simulator and APM setup.
+  void SetUp() override {
+    // Lambda function for setting the default APM runtime settings for desktop.
+    auto set_default_desktop_apm_runtime_settings = [](AudioProcessing* apm) {
+      ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError,
+                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(true));
+      ASSERT_EQ(apm->kNoError,
+                apm->echo_cancellation()->enable_delay_logging(true));
+    };
+
+    // Lambda function for setting the default APM runtime settings for mobile.
+    auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
+      ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError,
+                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
+    };
+
+    // Lambda function for turning off all of the APM runtime settings
+    // submodules.
+    auto turn_off_default_apm_runtime_settings = [](AudioProcessing* apm) {
+      ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
+      ASSERT_EQ(apm->kNoError,
+                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
+      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
+      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(false));
+      ASSERT_EQ(apm->kNoError,
+                apm->echo_cancellation()->enable_delay_logging(false));
+    };
+
+    // Lambda function for adding default desktop APM settings to a config.
+    auto add_default_desktop_config = [](Config* config) {
+      config->Set<ExtendedFilter>(new ExtendedFilter(true));
+      config->Set<DelayAgnostic>(new DelayAgnostic(true));
+    };
+
+    // Lambda function for adding beamformer settings to a config.
+    auto add_beamformer_config = [](Config* config) {
+      const size_t num_mics = 2;
+      const std::vector<Point> array_geometry =
+          ParseArrayGeometry("0 0 0 0.05 0 0", num_mics);
+      RTC_CHECK_EQ(array_geometry.size(), num_mics);
+
+      config->Set<Beamforming>(
+          new Beamforming(true, array_geometry,
+                          SphericalPointf(DegreesToRadians(90), 0.f, 1.f)));
+    };
+
+    int num_capture_channels = 1;
+    switch (simulation_config_.simulation_settings) {
+      case SettingsType::kDefaultApmMobile: {
+        apm_.reset(AudioProcessingImpl::Create());
+        ASSERT_TRUE(!!apm_);
+        set_default_mobile_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktop: {
+        Config config;
+        add_default_desktop_config(&config);
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopAndBeamformer: {
+        Config config;
+        add_beamformer_config(&config);
+        add_default_desktop_config(&config);
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        num_capture_channels = 2;
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer: {
+        Config config;
+        config.Set<Intelligibility>(new Intelligibility(true));
+        add_default_desktop_config(&config);
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+      case SettingsType::kAllSubmodulesTurnedOff: {
+        apm_.reset(AudioProcessingImpl::Create());
+        ASSERT_TRUE(!!apm_);
+        turn_off_default_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic: {
+        Config config;
+        config.Set<ExtendedFilter>(new ExtendedFilter(true));
+        config.Set<DelayAgnostic>(new DelayAgnostic(false));
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+      case SettingsType::kDefaultDesktopApmWithoutExtendedFilter: {
+        Config config;
+        config.Set<ExtendedFilter>(new ExtendedFilter(false));
+        config.Set<DelayAgnostic>(new DelayAgnostic(true));
+        apm_.reset(AudioProcessingImpl::Create(config));
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        apm_->SetExtraOptions(config);
+        break;
+      }
+    }
+
+    render_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kRender, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kRenderInputFloatLevel, 1));
+    capture_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kCapture, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kCaptureInputFloatLevel, num_capture_channels));
+  }
+
+  // Thread callback for the render thread.
+  static bool RenderProcessorThreadFunc(void* context) {
+    return reinterpret_cast<CallSimulator*>(context)
+        ->render_thread_state_->Process();
+  }
+
+  // Thread callback for the capture thread.
+  static bool CaptureProcessorThreadFunc(void* context) {
+    return reinterpret_cast<CallSimulator*>(context)
+        ->capture_thread_state_->Process();
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    ASSERT_NO_FATAL_FAILURE(render_thread_->Start());
+    render_thread_->SetPriority(rtc::kRealtimePriority);
+    ASSERT_NO_FATAL_FAILURE(capture_thread_->Start());
+    capture_thread_->SetPriority(rtc::kRealtimePriority);
+  }
+
+  // Event handler for the test.
+  const rtc::scoped_ptr<EventWrapper> test_complete_;
+
+  // Thread related variables.
+  rtc::scoped_ptr<rtc::PlatformThread> render_thread_;
+  rtc::scoped_ptr<rtc::PlatformThread> capture_thread_;
+  Random rand_gen_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+  const SimulationConfig simulation_config_;
+  FrameCounters frame_counters_;
+  LockedFlag capture_call_checker_;
+  rtc::scoped_ptr<TimedThreadApiProcessor> render_thread_state_;
+  rtc::scoped_ptr<TimedThreadApiProcessor> capture_thread_state_;
+};
+
+// Implements the callback functionality for the threads.
+bool TimedThreadApiProcessor::Process() {
+  PrepareFrame();
+
+  // Wait in a spinlock manner until it is ok to start processing.
+  // Note that SleepMs is not applicable since it only allows sleeping
+  // on a millisecond basis which is too long.
+  while (!ReadyToProcess()) {
+  }
+
+  int result = AudioProcessing::kNoError;
+  switch (processor_type_) {
+    case ProcessorType::kRender:
+      result = ProcessRender();
+      break;
+    case ProcessorType::kCapture:
+      result = ProcessCapture();
+      break;
+  }
+
+  EXPECT_EQ(result, AudioProcessing::kNoError);
+
+  return !test_->MaybeEndTest();
+}
+
+const float CallSimulator::kRenderInputFloatLevel = 0.5f;
+const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
+}  // anonymous namespace
+
+TEST_P(CallSimulator, ApiCallDurationTest) {
+  // Run test and verify that it did not time out.
+  EXPECT_EQ(kEventSignaled, Run());
+}
+
+INSTANTIATE_TEST_CASE_P(
+    AudioProcessingPerformanceTest,
+    CallSimulator,
+    ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_tests.gypi b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_tests.gypi
index a535144589..523602baba 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_tests.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_tests.gypi
@@ -8,6 +8,18 @@
 
 {
   'targets': [
+    {
+      'target_name': 'audioproc_test_utils',
+      'type': 'static_library',
+      'dependencies': [
+        '<(webrtc_root)/base/base.gyp:rtc_base_approved',
+        '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+      ],
+      'sources': [
+        'test/test_utils.cc',
+        'test/test_utils.h',
+      ],
+    },
     {
       'target_name': 'transient_suppression_test',
       'type': 'executable',
@@ -39,15 +51,28 @@
       'target_name': 'nonlinear_beamformer_test',
       'type': 'executable',
       'dependencies': [
+        'audioproc_test_utils',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/modules/modules.gyp:audio_processing',
       ],
       'sources': [
         'beamformer/nonlinear_beamformer_test.cc',
-        'beamformer/pcm_utils.cc',
-        'beamformer/pcm_utils.h',
       ],
     }, # nonlinear_beamformer_test
+    {
+      'target_name': 'intelligibility_proc',
+      'type': 'executable',
+      'dependencies': [
+        'audioproc_test_utils',
+        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(webrtc_root)/modules/modules.gyp:audio_processing',
+        '<(webrtc_root)/test/test.gyp:test_support',
+      ],
+      'sources': [
+        'intelligibility/test/intelligibility_proc.cc',
+      ],
+    }, # intelligibility_proc
   ],
   'conditions': [
     ['enable_protobuf==1', {
@@ -65,14 +90,28 @@
           },
           'includes': [ '../../build/protoc.gypi', ],
         },
+        {
+          'target_name': 'audioproc_protobuf_utils',
+          'type': 'static_library',
+          'dependencies': [
+            'audioproc_debug_proto',
+          ],
+          'sources': [
+            'test/protobuf_utils.cc',
+            'test/protobuf_utils.h',
+          ],
+        },
         {
           'target_name': 'audioproc',
           'type': 'executable',
           'dependencies': [
             'audio_processing',
             'audioproc_debug_proto',
+            'audioproc_test_utils',
+            'audioproc_protobuf_utils',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:test_support',
           ],
           'sources': [ 'test/process_test.cc', ],
@@ -83,15 +122,25 @@
           'dependencies': [
             'audio_processing',
             'audioproc_debug_proto',
+            'audioproc_test_utils',
+            'audioproc_protobuf_utils',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            '<(webrtc_root)/test/test.gyp:test_support',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
           ],
-          'sources': [ 'test/audioproc_float.cc', ],
+          'sources': [
+            'test/audio_file_processor.cc',
+            'test/audio_file_processor.h',
+            'test/audioproc_float.cc',
+          ],
         },
         {
           'target_name': 'unpack_aecdump',
           'type': 'executable',
           'dependencies': [
             'audioproc_debug_proto',
+            'audioproc_test_utils',
+            'audioproc_protobuf_utils',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.cc
new file mode 100644
index 0000000000..6b1c474269
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/beamformer/array_util.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace {
+
+const float kMaxDotProduct = 1e-6f;
+
+}  // namespace
+
+float GetMinimumSpacing(const std::vector<Point>& array_geometry) {
+  RTC_CHECK_GT(array_geometry.size(), 1u);
+  float mic_spacing = std::numeric_limits<float>::max();
+  for (size_t i = 0; i < (array_geometry.size() - 1); ++i) {
+    for (size_t j = i + 1; j < array_geometry.size(); ++j) {
+      mic_spacing =
+          std::min(mic_spacing, Distance(array_geometry[i], array_geometry[j]));
+    }
+  }
+  return mic_spacing;
+}
+
+Point PairDirection(const Point& a, const Point& b) {
+  return {b.x() - a.x(), b.y() - a.y(), b.z() - a.z()};
+}
+
+float DotProduct(const Point& a, const Point& b) {
+  return a.x() * b.x() + a.y() * b.y() + a.z() * b.z();
+}
+
+Point CrossProduct(const Point& a, const Point& b) {
+  return {a.y() * b.z() - a.z() * b.y(), a.z() * b.x() - a.x() * b.z(),
+          a.x() * b.y() - a.y() * b.x()};
+}
+
+bool AreParallel(const Point& a, const Point& b) {
+  Point cross_product = CrossProduct(a, b);
+  return DotProduct(cross_product, cross_product) < kMaxDotProduct;
+}
+
+bool ArePerpendicular(const Point& a, const Point& b) {
+  return std::abs(DotProduct(a, b)) < kMaxDotProduct;
+}
+
+rtc::Optional<Point> GetDirectionIfLinear(
+    const std::vector<Point>& array_geometry) {
+  RTC_DCHECK_GT(array_geometry.size(), 1u);
+  const Point first_pair_direction =
+      PairDirection(array_geometry[0], array_geometry[1]);
+  for (size_t i = 2u; i < array_geometry.size(); ++i) {
+    const Point pair_direction =
+        PairDirection(array_geometry[i - 1], array_geometry[i]);
+    if (!AreParallel(first_pair_direction, pair_direction)) {
+      return rtc::Optional<Point>();
+    }
+  }
+  return rtc::Optional<Point>(first_pair_direction);
+}
+
+rtc::Optional<Point> GetNormalIfPlanar(
+    const std::vector<Point>& array_geometry) {
+  RTC_DCHECK_GT(array_geometry.size(), 1u);
+  const Point first_pair_direction =
+      PairDirection(array_geometry[0], array_geometry[1]);
+  Point pair_direction(0.f, 0.f, 0.f);
+  size_t i = 2u;
+  bool is_linear = true;
+  for (; i < array_geometry.size() && is_linear; ++i) {
+    pair_direction = PairDirection(array_geometry[i - 1], array_geometry[i]);
+    if (!AreParallel(first_pair_direction, pair_direction)) {
+      is_linear = false;
+    }
+  }
+  if (is_linear) {
+    return rtc::Optional<Point>();
+  }
+  const Point normal_direction =
+      CrossProduct(first_pair_direction, pair_direction);
+  for (; i < array_geometry.size(); ++i) {
+    pair_direction = PairDirection(array_geometry[i - 1], array_geometry[i]);
+    if (!ArePerpendicular(normal_direction, pair_direction)) {
+      return rtc::Optional<Point>();
+    }
+  }
+  return rtc::Optional<Point>(normal_direction);
+}
+
+rtc::Optional<Point> GetArrayNormalIfExists(
+    const std::vector<Point>& array_geometry) {
+  const rtc::Optional<Point> direction = GetDirectionIfLinear(array_geometry);
+  if (direction) {
+    return rtc::Optional<Point>(Point(direction->y(), -direction->x(), 0.f));
+  }
+  const rtc::Optional<Point> normal = GetNormalIfPlanar(array_geometry);
+  if (normal && normal->z() < kMaxDotProduct) {
+    return normal;
+  }
+  return rtc::Optional<Point>();
+}
+
+Point AzimuthToPoint(float azimuth) {
+  return Point(std::cos(azimuth), std::sin(azimuth), 0.f);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.h
index 8d1cda783a..f86ad5dee6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util.h
@@ -12,30 +12,67 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_ARRAY_UTIL_H_
 
 #include <cmath>
+#include <vector>
+
+#include "webrtc/base/optional.h"
 
 namespace webrtc {
 
-// Coordinates in meters.
+// Coordinates in meters. The convention used is:
+// x: the horizontal dimension, with positive to the right from the camera's
+//    perspective.
+// y: the depth dimension, with positive forward from the camera's
+//    perspective.
+// z: the vertical dimension, with positive upwards.
 template<typename T>
 struct CartesianPoint {
+  CartesianPoint() {
+    c[0] = 0;
+    c[1] = 0;
+    c[2] = 0;
+  }
   CartesianPoint(T x, T y, T z) {
     c[0] = x;
     c[1] = y;
     c[2] = z;
   }
-  T x() const {
-    return c[0];
-  }
-  T y() const {
-    return c[1];
-  }
-  T z() const {
-    return c[2];
-  }
+  T x() const { return c[0]; }
+  T y() const { return c[1]; }
+  T z() const { return c[2]; }
   T c[3];
 };
 
-typedef CartesianPoint<float> Point;
+using Point = CartesianPoint<float>;
+
+// Calculates the direction from a to b.
+Point PairDirection(const Point& a, const Point& b);
+
+float DotProduct(const Point& a, const Point& b);
+Point CrossProduct(const Point& a, const Point& b);
+
+bool AreParallel(const Point& a, const Point& b);
+bool ArePerpendicular(const Point& a, const Point& b);
+
+// Returns the minimum distance between any two Points in the given
+// |array_geometry|.
+float GetMinimumSpacing(const std::vector<Point>& array_geometry);
+
+// If the given array geometry is linear it returns the direction without
+// normalizing.
+rtc::Optional<Point> GetDirectionIfLinear(
+    const std::vector<Point>& array_geometry);
+
+// If the given array geometry is planar it returns the normal without
+// normalizing.
+rtc::Optional<Point> GetNormalIfPlanar(
+    const std::vector<Point>& array_geometry);
+
+// Returns the normal of an array if it has one and it is in the xy-plane.
+rtc::Optional<Point> GetArrayNormalIfExists(
+    const std::vector<Point>& array_geometry);
+
+// The resulting Point will be in the xy-plane.
+Point AzimuthToPoint(float azimuth);
 
 template<typename T>
 float Distance(CartesianPoint<T> a, CartesianPoint<T> b) {
@@ -44,6 +81,37 @@ float Distance(CartesianPoint<T> a, CartesianPoint<T> b) {
                    (a.z() - b.z()) * (a.z() - b.z()));
 }
 
+// The convention used:
+// azimuth: zero is to the right from the camera's perspective, with positive
+//          angles in radians counter-clockwise.
+// elevation: zero is horizontal, with positive angles in radians upwards.
+// radius: distance from the camera in meters.
+template <typename T>
+struct SphericalPoint {
+  SphericalPoint(T azimuth, T elevation, T radius) {
+    s[0] = azimuth;
+    s[1] = elevation;
+    s[2] = radius;
+  }
+  T azimuth() const { return s[0]; }
+  T elevation() const { return s[1]; }
+  T distance() const { return s[2]; }
+  T s[3];
+};
+
+using SphericalPointf = SphericalPoint<float>;
+
+// Helper functions to transform degrees to radians and the inverse.
+template <typename T>
+T DegreesToRadians(T angle_degrees) {
+  return M_PI * angle_degrees / 180;
+}
+
+template <typename T>
+T RadiansToDegrees(T angle_radians) {
+  return 180 * angle_radians / M_PI;
+}
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_ARRAY_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util_unittest.cc
new file mode 100644
index 0000000000..e3a7bbd7aa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/array_util_unittest.cc
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#define _USE_MATH_DEFINES
+
+#include "webrtc/modules/audio_processing/beamformer/array_util.h"
+
+#include <math.h>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+bool operator==(const Point& lhs, const Point& rhs) {
+  return lhs.x() == rhs.x() && lhs.y() == rhs.y() && lhs.z() == rhs.z();
+}
+
+TEST(ArrayUtilTest, PairDirection) {
+  EXPECT_EQ(Point(1.f, 2.f, 3.f),
+            PairDirection(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
+  EXPECT_EQ(Point(-1.f, -2.f, -3.f),
+            PairDirection(Point(1.f, 2.f, 3.f), Point(0.f, 0.f, 0.f)));
+  EXPECT_EQ(Point(0.f, 0.f, 0.f),
+            PairDirection(Point(1.f, 0.f, 0.f), Point(1.f, 0.f, 0.f)));
+  EXPECT_EQ(Point(-1.f, 2.f, 0.f),
+            PairDirection(Point(1.f, 0.f, 0.f), Point(0.f, 2.f, 0.f)));
+  EXPECT_EQ(Point(-4.f, 4.f, -4.f),
+            PairDirection(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
+}
+
+TEST(ArrayUtilTest, DotProduct) {
+  EXPECT_FLOAT_EQ(0.f, DotProduct(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
+  EXPECT_FLOAT_EQ(0.f, DotProduct(Point(1.f, 0.f, 2.f), Point(0.f, 3.f, 0.f)));
+  EXPECT_FLOAT_EQ(0.f, DotProduct(Point(1.f, 1.f, 0.f), Point(1.f, -1.f, 0.f)));
+  EXPECT_FLOAT_EQ(2.f, DotProduct(Point(1.f, 0.f, 0.f), Point(2.f, 0.f, 0.f)));
+  EXPECT_FLOAT_EQ(-6.f,
+                  DotProduct(Point(-2.f, 0.f, 0.f), Point(3.f, 0.f, 0.f)));
+  EXPECT_FLOAT_EQ(-10.f,
+                  DotProduct(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
+}
+
+TEST(ArrayUtilTest, CrossProduct) {
+  EXPECT_EQ(Point(0.f, 0.f, 0.f),
+            CrossProduct(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
+  EXPECT_EQ(Point(0.f, 0.f, 1.f),
+            CrossProduct(Point(1.f, 0.f, 0.f), Point(0.f, 1.f, 0.f)));
+  EXPECT_EQ(Point(1.f, 0.f, 0.f),
+            CrossProduct(Point(0.f, 1.f, 0.f), Point(0.f, 0.f, 1.f)));
+  EXPECT_EQ(Point(0.f, -1.f, 0.f),
+            CrossProduct(Point(1.f, 0.f, 0.f), Point(0.f, 0.f, 1.f)));
+  EXPECT_EQ(Point(-4.f, -8.f, -4.f),
+            CrossProduct(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
+}
+
+TEST(ArrayUtilTest, AreParallel) {
+  EXPECT_TRUE(AreParallel(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
+  EXPECT_FALSE(AreParallel(Point(1.f, 0.f, 2.f), Point(0.f, 3.f, 0.f)));
+  EXPECT_FALSE(AreParallel(Point(1.f, 2.f, 0.f), Point(1.f, -0.5f, 0.f)));
+  EXPECT_FALSE(AreParallel(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
+  EXPECT_TRUE(AreParallel(Point(1.f, 0.f, 0.f), Point(2.f, 0.f, 0.f)));
+  EXPECT_TRUE(AreParallel(Point(1.f, 2.f, 3.f), Point(-2.f, -4.f, -6.f)));
+}
+
+TEST(ArrayUtilTest, ArePerpendicular) {
+  EXPECT_TRUE(ArePerpendicular(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
+  EXPECT_TRUE(ArePerpendicular(Point(1.f, 0.f, 2.f), Point(0.f, 3.f, 0.f)));
+  EXPECT_TRUE(ArePerpendicular(Point(1.f, 2.f, 0.f), Point(1.f, -0.5f, 0.f)));
+  EXPECT_FALSE(ArePerpendicular(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
+  EXPECT_FALSE(ArePerpendicular(Point(1.f, 0.f, 0.f), Point(2.f, 0.f, 0.f)));
+  EXPECT_FALSE(ArePerpendicular(Point(1.f, 2.f, 3.f), Point(-2.f, -4.f, -6.f)));
+}
+
+TEST(ArrayUtilTest, GetMinimumSpacing) {
+  std::vector<Point> geometry;
+  geometry.push_back(Point(0.f, 0.f, 0.f));
+  geometry.push_back(Point(0.1f, 0.f, 0.f));
+  EXPECT_FLOAT_EQ(0.1f, GetMinimumSpacing(geometry));
+  geometry.push_back(Point(0.f, 0.05f, 0.f));
+  EXPECT_FLOAT_EQ(0.05f, GetMinimumSpacing(geometry));
+  geometry.push_back(Point(0.f, 0.f, 0.02f));
+  EXPECT_FLOAT_EQ(0.02f, GetMinimumSpacing(geometry));
+  geometry.push_back(Point(-0.003f, -0.004f, 0.02f));
+  EXPECT_FLOAT_EQ(0.005f, GetMinimumSpacing(geometry));
+}
+
+TEST(ArrayUtilTest, GetDirectionIfLinear) {
+  std::vector<Point> geometry;
+  geometry.push_back(Point(0.f, 0.f, 0.f));
+  geometry.push_back(Point(0.1f, 0.f, 0.f));
+  EXPECT_TRUE(
+      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
+  geometry.push_back(Point(0.15f, 0.f, 0.f));
+  EXPECT_TRUE(
+      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
+  geometry.push_back(Point(-0.2f, 0.f, 0.f));
+  EXPECT_TRUE(
+      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
+  geometry.push_back(Point(0.05f, 0.f, 0.f));
+  EXPECT_TRUE(
+      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
+  geometry.push_back(Point(0.1f, 0.1f, 0.f));
+  EXPECT_FALSE(GetDirectionIfLinear(geometry));
+  geometry.push_back(Point(0.f, 0.f, -0.2f));
+  EXPECT_FALSE(GetDirectionIfLinear(geometry));
+}
+
+TEST(ArrayUtilTest, GetNormalIfPlanar) {
+  std::vector<Point> geometry;
+  geometry.push_back(Point(0.f, 0.f, 0.f));
+  geometry.push_back(Point(0.1f, 0.f, 0.f));
+  EXPECT_FALSE(GetNormalIfPlanar(geometry));
+  geometry.push_back(Point(0.15f, 0.f, 0.f));
+  EXPECT_FALSE(GetNormalIfPlanar(geometry));
+  geometry.push_back(Point(0.1f, 0.2f, 0.f));
+  EXPECT_TRUE(AreParallel(Point(0.f, 0.f, 1.f), *GetNormalIfPlanar(geometry)));
+  geometry.push_back(Point(0.f, -0.15f, 0.f));
+  EXPECT_TRUE(AreParallel(Point(0.f, 0.f, 1.f), *GetNormalIfPlanar(geometry)));
+  geometry.push_back(Point(0.f, 0.1f, 0.2f));
+  EXPECT_FALSE(GetNormalIfPlanar(geometry));
+  geometry.push_back(Point(0.f, 0.f, -0.15f));
+  EXPECT_FALSE(GetNormalIfPlanar(geometry));
+  geometry.push_back(Point(0.1f, 0.2f, 0.f));
+  EXPECT_FALSE(GetNormalIfPlanar(geometry));
+}
+
+TEST(ArrayUtilTest, GetArrayNormalIfExists) {
+  std::vector<Point> geometry;
+  geometry.push_back(Point(0.f, 0.f, 0.f));
+  geometry.push_back(Point(0.1f, 0.f, 0.f));
+  EXPECT_TRUE(
+      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
+  geometry.push_back(Point(0.15f, 0.f, 0.f));
+  EXPECT_TRUE(
+      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
+  geometry.push_back(Point(0.1f, 0.f, 0.2f));
+  EXPECT_TRUE(
+      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
+  geometry.push_back(Point(0.f, 0.f, -0.1f));
+  EXPECT_TRUE(
+      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
+  geometry.push_back(Point(0.1f, 0.2f, 0.3f));
+  EXPECT_FALSE(GetArrayNormalIfExists(geometry));
+  geometry.push_back(Point(0.f, -0.1f, 0.f));
+  EXPECT_FALSE(GetArrayNormalIfExists(geometry));
+  geometry.push_back(Point(1.f, 0.f, -0.2f));
+  EXPECT_FALSE(GetArrayNormalIfExists(geometry));
+}
+
+TEST(ArrayUtilTest, DegreesToRadians) {
+  EXPECT_FLOAT_EQ(0.f, DegreesToRadians(0.f));
+  EXPECT_FLOAT_EQ(static_cast<float>(M_PI) / 6.f, DegreesToRadians(30.f));
+  EXPECT_FLOAT_EQ(-static_cast<float>(M_PI) / 4.f, DegreesToRadians(-45.f));
+  EXPECT_FLOAT_EQ(static_cast<float>(M_PI) / 3.f, DegreesToRadians(60.f));
+  EXPECT_FLOAT_EQ(-static_cast<float>(M_PI) / 2.f, DegreesToRadians(-90.f));
+  EXPECT_FLOAT_EQ(2.f * static_cast<float>(M_PI) / 3.f,
+                  DegreesToRadians(120.f));
+  EXPECT_FLOAT_EQ(-3.f * static_cast<float>(M_PI) / 4.f,
+                  DegreesToRadians(-135.f));
+  EXPECT_FLOAT_EQ(5.f * static_cast<float>(M_PI) / 6.f,
+                  DegreesToRadians(150.f));
+  EXPECT_FLOAT_EQ(-static_cast<float>(M_PI), DegreesToRadians(-180.f));
+}
+
+TEST(ArrayUtilTest, RadiansToDegrees) {
+  EXPECT_FLOAT_EQ(0.f, RadiansToDegrees(0.f));
+  EXPECT_FLOAT_EQ(30.f, RadiansToDegrees(M_PI / 6.f));
+  EXPECT_FLOAT_EQ(-45.f, RadiansToDegrees(-M_PI / 4.f));
+  EXPECT_FLOAT_EQ(60.f, RadiansToDegrees(M_PI / 3.f));
+  EXPECT_FLOAT_EQ(-90.f, RadiansToDegrees(-M_PI / 2.f));
+  EXPECT_FLOAT_EQ(120.f, RadiansToDegrees(2.f * M_PI / 3.f));
+  EXPECT_FLOAT_EQ(-135.f, RadiansToDegrees(-3.f * M_PI / 4.f));
+  EXPECT_FLOAT_EQ(150.f, RadiansToDegrees(5.f * M_PI / 6.f));
+  EXPECT_FLOAT_EQ(-180.f, RadiansToDegrees(-M_PI));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/beamformer.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/beamformer.h
index 04cb659c6d..6a9ff45d12 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/beamformer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/beamformer.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_BEAMFORMER_H_
 
 #include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_processing/beamformer/array_util.h"
 
 namespace webrtc {
 
@@ -31,6 +32,12 @@ class Beamformer {
   // Needs to be called before the the Beamformer can be used.
   virtual void Initialize(int chunk_size_ms, int sample_rate_hz) = 0;
 
+  // Aim the beamformer at a point in space.
+  virtual void AimAt(const SphericalPointf& spherical_point) = 0;
+
+  // Indicates whether a given point is inside of the beam.
+  virtual bool IsInBeam(const SphericalPointf& spherical_point) { return true; }
+
   // Returns true if the current data contains the target signal.
   // Which signals are considered "targets" is implementation dependent.
   virtual bool is_target_present() = 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/complex_matrix.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/complex_matrix.h
index f5be2b2f63..707c51564b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/complex_matrix.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/complex_matrix.h
@@ -27,10 +27,10 @@ class ComplexMatrix : public Matrix<complex<T> > {
  public:
   ComplexMatrix() : Matrix<complex<T> >() {}
 
-  ComplexMatrix(int num_rows, int num_columns)
+  ComplexMatrix(size_t num_rows, size_t num_columns)
       : Matrix<complex<T> >(num_rows, num_columns) {}
 
-  ComplexMatrix(const complex<T>* data, int num_rows, int num_columns)
+  ComplexMatrix(const complex<T>* data, size_t num_rows, size_t num_columns)
       : Matrix<complex<T> >(data, num_rows, num_columns) {}
 
   // Complex Matrix operations.
@@ -51,7 +51,7 @@ class ComplexMatrix : public Matrix<complex<T> > {
 
   ComplexMatrix& ConjugateTranspose() {
     this->CopyDataToScratch();
-    int num_rows = this->num_rows();
+    size_t num_rows = this->num_rows();
     this->SetNumRows(this->num_columns());
     this->SetNumColumns(num_rows);
     this->Resize();
@@ -59,8 +59,8 @@ class ComplexMatrix : public Matrix<complex<T> > {
   }
 
   ComplexMatrix& ConjugateTranspose(const ComplexMatrix& operand) {
-    CHECK_EQ(operand.num_rows(), this->num_columns());
-    CHECK_EQ(operand.num_columns(), this->num_rows());
+    RTC_CHECK_EQ(operand.num_rows(), this->num_columns());
+    RTC_CHECK_EQ(operand.num_columns(), this->num_rows());
     return ConjugateTranspose(operand.elements());
   }
 
@@ -82,8 +82,8 @@ class ComplexMatrix : public Matrix<complex<T> > {
  private:
   ComplexMatrix& ConjugateTranspose(const complex<T>* const* src) {
     complex<T>* const* elements = this->elements();
-    for (int i = 0; i < this->num_rows(); ++i) {
-      for (int j = 0; j < this->num_columns(); ++j) {
+    for (size_t i = 0; i < this->num_rows(); ++i) {
+      for (size_t j = 0; j < this->num_columns(); ++j) {
         elements[i][j] = conj(src[j][i]);
       }
     }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
index 51a4ad088f..1752765bd4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
@@ -14,6 +14,7 @@
 
 #include "webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h"
 
+namespace webrtc {
 namespace {
 
 float BesselJ0(float x) {
@@ -24,16 +25,26 @@ float BesselJ0(float x) {
 #endif
 }
 
-}  // namespace
+// Calculates the Euclidean norm for a row vector.
+float Norm(const ComplexMatrix<float>& x) {
+  RTC_CHECK_EQ(1u, x.num_rows());
+  const size_t length = x.num_columns();
+  const complex<float>* elems = x.elements()[0];
+  float result = 0.f;
+  for (size_t i = 0u; i < length; ++i) {
+    result += std::norm(elems[i]);
+  }
+  return std::sqrt(result);
+}
 
-namespace webrtc {
+}  // namespace
 
 void CovarianceMatrixGenerator::UniformCovarianceMatrix(
     float wave_number,
     const std::vector<Point>& geometry,
     ComplexMatrix<float>* mat) {
-  CHECK_EQ(static_cast<int>(geometry.size()), mat->num_rows());
-  CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+  RTC_CHECK_EQ(geometry.size(), mat->num_rows());
+  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
 
   complex<float>* const* mat_els = mat->elements();
   for (size_t i = 0; i < geometry.size(); ++i) {
@@ -51,14 +62,14 @@ void CovarianceMatrixGenerator::UniformCovarianceMatrix(
 void CovarianceMatrixGenerator::AngledCovarianceMatrix(
     float sound_speed,
     float angle,
-    int frequency_bin,
-    int fft_size,
-    int num_freq_bins,
+    size_t frequency_bin,
+    size_t fft_size,
+    size_t num_freq_bins,
     int sample_rate,
     const std::vector<Point>& geometry,
     ComplexMatrix<float>* mat) {
-  CHECK_EQ(static_cast<int>(geometry.size()), mat->num_rows());
-  CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+  RTC_CHECK_EQ(geometry.size(), mat->num_rows());
+  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
 
   ComplexMatrix<float> interf_cov_vector(1, geometry.size());
   ComplexMatrix<float> interf_cov_vector_transposed(geometry.size(), 1);
@@ -69,21 +80,22 @@ void CovarianceMatrixGenerator::AngledCovarianceMatrix(
                       geometry,
                       angle,
                       &interf_cov_vector);
+  interf_cov_vector.Scale(1.f / Norm(interf_cov_vector));
   interf_cov_vector_transposed.Transpose(interf_cov_vector);
   interf_cov_vector.PointwiseConjugate();
   mat->Multiply(interf_cov_vector_transposed, interf_cov_vector);
 }
 
 void CovarianceMatrixGenerator::PhaseAlignmentMasks(
-    int frequency_bin,
-    int fft_size,
+    size_t frequency_bin,
+    size_t fft_size,
     int sample_rate,
     float sound_speed,
     const std::vector<Point>& geometry,
     float angle,
     ComplexMatrix<float>* mat) {
-  CHECK_EQ(1, mat->num_rows());
-  CHECK_EQ(static_cast<int>(geometry.size()), mat->num_columns());
+  RTC_CHECK_EQ(1u, mat->num_rows());
+  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
 
   float freq_in_hertz =
       (static_cast<float>(frequency_bin) / fft_size) * sample_rate;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h
index 5979462751..5375518e8a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h
@@ -30,9 +30,9 @@ class CovarianceMatrixGenerator {
   // The covariance matrix of a source at the given angle.
   static void AngledCovarianceMatrix(float sound_speed,
                                      float angle,
-                                     int frequency_bin,
-                                     int fft_size,
-                                     int num_freq_bins,
+                                     size_t frequency_bin,
+                                     size_t fft_size,
+                                     size_t num_freq_bins,
                                      int sample_rate,
                                      const std::vector<Point>& geometry,
                                      ComplexMatrix<float>* mat);
@@ -40,8 +40,8 @@ class CovarianceMatrixGenerator {
   // Calculates phase shifts that, when applied to a multichannel signal and
   // added together, cause constructive interferernce for sources located at
   // the given angle.
-  static void PhaseAlignmentMasks(int frequency_bin,
-                                  int fft_size,
+  static void PhaseAlignmentMasks(size_t frequency_bin,
+                                  size_t fft_size,
                                   int sample_rate,
                                   float sound_speed,
                                   const std::vector<Point>& geometry,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc
index 4ea341d84c..23d672fb4e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc
@@ -165,14 +165,14 @@ TEST(CovarianceMatrixGeneratorTest, TestAngledCovarianceMatrix2Mics) {
 
   complex<float>* const* actual_els = actual_covariance_matrix.elements();
 
-  EXPECT_NEAR(actual_els[0][0].real(), 1.f, kTolerance);
-  EXPECT_NEAR(actual_els[0][1].real(), 0.9952f, kTolerance);
-  EXPECT_NEAR(actual_els[1][0].real(), 0.9952f, kTolerance);
-  EXPECT_NEAR(actual_els[1][1].real(), 1.f, kTolerance);
+  EXPECT_NEAR(actual_els[0][0].real(), 0.5f, kTolerance);
+  EXPECT_NEAR(actual_els[0][1].real(), 0.4976f, kTolerance);
+  EXPECT_NEAR(actual_els[1][0].real(), 0.4976f, kTolerance);
+  EXPECT_NEAR(actual_els[1][1].real(), 0.5f, kTolerance);
 
   EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
-  EXPECT_NEAR(actual_els[0][1].imag(), 0.0978f, kTolerance);
-  EXPECT_NEAR(actual_els[1][0].imag(), -0.0978f, kTolerance);
+  EXPECT_NEAR(actual_els[0][1].imag(), 0.0489f, kTolerance);
+  EXPECT_NEAR(actual_els[1][0].imag(), -0.0489f, kTolerance);
   EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
 }
 
@@ -203,24 +203,24 @@ TEST(CovarianceMatrixGeneratorTest, TestAngledCovarianceMatrix3Mics) {
 
   complex<float>* const* actual_els = actual_covariance_matrix.elements();
 
-  EXPECT_NEAR(actual_els[0][0].real(), 1.f, kTolerance);
-  EXPECT_NEAR(actual_els[0][1].real(), 0.8859f, kTolerance);
-  EXPECT_NEAR(actual_els[0][2].real(), 0.5696f, kTolerance);
-  EXPECT_NEAR(actual_els[1][0].real(), 0.8859f, kTolerance);
-  EXPECT_NEAR(actual_els[1][1].real(), 1.f, kTolerance);
-  EXPECT_NEAR(actual_els[1][2].real(), 0.8859f, kTolerance);
-  EXPECT_NEAR(actual_els[2][0].real(), 0.5696f, kTolerance);
-  EXPECT_NEAR(actual_els[2][1].real(), 0.8859f, kTolerance);
-  EXPECT_NEAR(actual_els[2][2].real(), 1.f, kTolerance);
+  EXPECT_NEAR(actual_els[0][0].real(), 0.3333f, kTolerance);
+  EXPECT_NEAR(actual_els[0][1].real(), 0.2953f, kTolerance);
+  EXPECT_NEAR(actual_els[0][2].real(), 0.1899f, kTolerance);
+  EXPECT_NEAR(actual_els[1][0].real(), 0.2953f, kTolerance);
+  EXPECT_NEAR(actual_els[1][1].real(), 0.3333f, kTolerance);
+  EXPECT_NEAR(actual_els[1][2].real(), 0.2953f, kTolerance);
+  EXPECT_NEAR(actual_els[2][0].real(), 0.1899f, kTolerance);
+  EXPECT_NEAR(actual_els[2][1].real(), 0.2953f, kTolerance);
+  EXPECT_NEAR(actual_els[2][2].real(), 0.3333f, kTolerance);
 
   EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
-  EXPECT_NEAR(actual_els[0][1].imag(), 0.4639f, kTolerance);
-  EXPECT_NEAR(actual_els[0][2].imag(), 0.8219f, kTolerance);
-  EXPECT_NEAR(actual_els[1][0].imag(), -0.4639f, kTolerance);
+  EXPECT_NEAR(actual_els[0][1].imag(), 0.1546f, kTolerance);
+  EXPECT_NEAR(actual_els[0][2].imag(), 0.274f, kTolerance);
+  EXPECT_NEAR(actual_els[1][0].imag(), -0.1546f, kTolerance);
   EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
-  EXPECT_NEAR(actual_els[1][2].imag(), 0.4639f, kTolerance);
-  EXPECT_NEAR(actual_els[2][0].imag(), -0.8219f, kTolerance);
-  EXPECT_NEAR(actual_els[2][1].imag(), -0.4639f, kTolerance);
+  EXPECT_NEAR(actual_els[1][2].imag(), 0.1546f, kTolerance);
+  EXPECT_NEAR(actual_els[2][0].imag(), -0.274f, kTolerance);
+  EXPECT_NEAR(actual_els[2][1].imag(), -0.1546f, kTolerance);
   EXPECT_NEAR(actual_els[2][2].imag(), 0.f, kTolerance);
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix.h
index 990f6a4a1b..51c1cece97 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix.h
@@ -67,7 +67,7 @@ class Matrix {
   Matrix() : num_rows_(0), num_columns_(0) {}
 
   // Allocates space for the elements and initializes all values to zero.
-  Matrix(int num_rows, int num_columns)
+  Matrix(size_t num_rows, size_t num_columns)
       : num_rows_(num_rows), num_columns_(num_columns) {
     Resize();
     scratch_data_.resize(num_rows_ * num_columns_);
@@ -75,7 +75,7 @@ class Matrix {
   }
 
   // Copies |data| into the new Matrix.
-  Matrix(const T* data, int num_rows, int num_columns)
+  Matrix(const T* data, size_t num_rows, size_t num_columns)
       : num_rows_(0), num_columns_(0) {
     CopyFrom(data, num_rows, num_columns);
     scratch_data_.resize(num_rows_ * num_columns_);
@@ -90,21 +90,23 @@ class Matrix {
   }
 
   // Copy |data| into the Matrix. The current data is lost.
-  void CopyFrom(const T* const data, int num_rows, int num_columns) {
+  void CopyFrom(const T* const data, size_t num_rows, size_t num_columns) {
     Resize(num_rows, num_columns);
     memcpy(&data_[0], data, num_rows_ * num_columns_ * sizeof(data_[0]));
   }
 
-  Matrix& CopyFromColumn(const T* const* src, int column_index, int num_rows) {
+  Matrix& CopyFromColumn(const T* const* src,
+                         size_t column_index,
+                         size_t num_rows) {
     Resize(1, num_rows);
-    for (int i = 0; i < num_columns_; ++i) {
+    for (size_t i = 0; i < num_columns_; ++i) {
       data_[i] = src[i][column_index];
     }
 
     return *this;
   }
 
-  void Resize(int num_rows, int num_columns) {
+  void Resize(size_t num_rows, size_t num_columns) {
     if (num_rows != num_rows_ || num_columns != num_columns_) {
       num_rows_ = num_rows;
       num_columns_ = num_columns;
@@ -113,16 +115,16 @@ class Matrix {
   }
 
   // Accessors and mutators.
-  int num_rows() const { return num_rows_; }
-  int num_columns() const { return num_columns_; }
+  size_t num_rows() const { return num_rows_; }
+  size_t num_columns() const { return num_columns_; }
   T* const* elements() { return &elements_[0]; }
   const T* const* elements() const { return &elements_[0]; }
 
   T Trace() {
-    CHECK_EQ(num_rows_, num_columns_);
+    RTC_CHECK_EQ(num_rows_, num_columns_);
 
     T trace = 0;
-    for (int i = 0; i < num_rows_; ++i) {
+    for (size_t i = 0; i < num_rows_; ++i) {
       trace += elements_[i][i];
     }
     return trace;
@@ -136,8 +138,8 @@ class Matrix {
   }
 
   Matrix& Transpose(const Matrix& operand) {
-    CHECK_EQ(operand.num_rows_, num_columns_);
-    CHECK_EQ(operand.num_columns_, num_rows_);
+    RTC_CHECK_EQ(operand.num_rows_, num_columns_);
+    RTC_CHECK_EQ(operand.num_columns_, num_rows_);
 
     return Transpose(operand.elements());
   }
@@ -158,8 +160,8 @@ class Matrix {
   }
 
   Matrix& Add(const Matrix& operand) {
-    CHECK_EQ(num_rows_, operand.num_rows_);
-    CHECK_EQ(num_columns_, operand.num_columns_);
+    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
+    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
 
     for (size_t i = 0; i < data_.size(); ++i) {
       data_[i] += operand.data_[i];
@@ -174,8 +176,8 @@ class Matrix {
   }
 
   Matrix& Subtract(const Matrix& operand) {
-    CHECK_EQ(num_rows_, operand.num_rows_);
-    CHECK_EQ(num_columns_, operand.num_columns_);
+    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
+    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
 
     for (size_t i = 0; i < data_.size(); ++i) {
       data_[i] -= operand.data_[i];
@@ -190,8 +192,8 @@ class Matrix {
   }
 
   Matrix& PointwiseMultiply(const Matrix& operand) {
-    CHECK_EQ(num_rows_, operand.num_rows_);
-    CHECK_EQ(num_columns_, operand.num_columns_);
+    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
+    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
 
     for (size_t i = 0; i < data_.size(); ++i) {
       data_[i] *= operand.data_[i];
@@ -206,8 +208,8 @@ class Matrix {
   }
 
   Matrix& PointwiseDivide(const Matrix& operand) {
-    CHECK_EQ(num_rows_, operand.num_rows_);
-    CHECK_EQ(num_columns_, operand.num_columns_);
+    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
+    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
 
     for (size_t i = 0; i < data_.size(); ++i) {
       data_[i] /= operand.data_[i];
@@ -261,15 +263,15 @@ class Matrix {
   }
 
   Matrix& Multiply(const Matrix& lhs, const Matrix& rhs) {
-    CHECK_EQ(lhs.num_columns_, rhs.num_rows_);
-    CHECK_EQ(num_rows_, lhs.num_rows_);
-    CHECK_EQ(num_columns_, rhs.num_columns_);
+    RTC_CHECK_EQ(lhs.num_columns_, rhs.num_rows_);
+    RTC_CHECK_EQ(num_rows_, lhs.num_rows_);
+    RTC_CHECK_EQ(num_columns_, rhs.num_columns_);
 
     return Multiply(lhs.elements(), rhs.num_rows_, rhs.elements());
   }
 
   Matrix& Multiply(const Matrix& rhs) {
-    CHECK_EQ(num_columns_, rhs.num_rows_);
+    RTC_CHECK_EQ(num_columns_, rhs.num_rows_);
 
     CopyDataToScratch();
     Resize(num_rows_, rhs.num_columns_);
@@ -280,8 +282,8 @@ class Matrix {
     std::ostringstream ss;
     ss << std::endl << "Matrix" << std::endl;
 
-    for (int i = 0; i < num_rows_; ++i) {
-      for (int j = 0; j < num_columns_; ++j) {
+    for (size_t i = 0; i < num_rows_; ++i) {
+      for (size_t j = 0; j < num_columns_; ++j) {
         ss << elements_[i][j] << " ";
       }
       ss << std::endl;
@@ -292,8 +294,8 @@ class Matrix {
   }
 
  protected:
-  void SetNumRows(const int num_rows) { num_rows_ = num_rows; }
-  void SetNumColumns(const int num_columns) { num_columns_ = num_columns; }
+  void SetNumRows(const size_t num_rows) { num_rows_ = num_rows; }
+  void SetNumColumns(const size_t num_columns) { num_columns_ = num_columns; }
   T* data() { return &data_[0]; }
   const T* data() const { return &data_[0]; }
   const T* const* scratch_elements() const { return &scratch_elements_[0]; }
@@ -305,7 +307,7 @@ class Matrix {
     data_.resize(size);
     elements_.resize(num_rows_);
 
-    for (int i = 0; i < num_rows_; ++i) {
+    for (size_t i = 0; i < num_rows_; ++i) {
       elements_[i] = &data_[i * num_columns_];
     }
   }
@@ -315,14 +317,14 @@ class Matrix {
     scratch_data_ = data_;
     scratch_elements_.resize(num_rows_);
 
-    for (int i = 0; i < num_rows_; ++i) {
+    for (size_t i = 0; i < num_rows_; ++i) {
       scratch_elements_[i] = &scratch_data_[i * num_columns_];
     }
   }
 
  private:
-  int num_rows_;
-  int num_columns_;
+  size_t num_rows_;
+  size_t num_columns_;
   std::vector<T> data_;
   std::vector<T*> elements_;
 
@@ -334,8 +336,8 @@ class Matrix {
   // Helpers for Transpose and Multiply operations that unify in-place and
   // out-of-place solutions.
   Matrix& Transpose(const T* const* src) {
-    for (int i = 0; i < num_rows_; ++i) {
-      for (int j = 0; j < num_columns_; ++j) {
+    for (size_t i = 0; i < num_rows_; ++i) {
+      for (size_t j = 0; j < num_columns_; ++j) {
         elements_[i][j] = src[j][i];
       }
     }
@@ -343,11 +345,13 @@ class Matrix {
     return *this;
   }
 
-  Matrix& Multiply(const T* const* lhs, int num_rows_rhs, const T* const* rhs) {
-    for (int row = 0; row < num_rows_; ++row) {
-      for (int col = 0; col < num_columns_; ++col) {
+  Matrix& Multiply(const T* const* lhs,
+                   size_t num_rows_rhs,
+                   const T* const* rhs) {
+    for (size_t row = 0; row < num_rows_; ++row) {
+      for (size_t col = 0; col < num_columns_; ++col) {
         T cur_element = 0;
-        for (int i = 0; i < num_rows_rhs; ++i) {
+        for (size_t i = 0; i < num_rows_rhs; ++i) {
           cur_element += lhs[row][i] * rhs[i][col];
         }
 
@@ -358,7 +362,7 @@ class Matrix {
     return *this;
   }
 
-  DISALLOW_COPY_AND_ASSIGN(Matrix);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Matrix);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
index 7c58670068..9891a8220c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
@@ -34,8 +34,8 @@ class MatrixTestHelpers {
 
     const T* const* expected_elements = expected.elements();
     const T* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_EQ(expected_elements[i][j], actual_elements[i][j]);
       }
     }
@@ -48,8 +48,8 @@ class MatrixTestHelpers {
 
     const float* const* expected_elements = expected.elements();
     const float* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_NEAR(expected_elements[i][j], actual_elements[i][j], kTolerance);
       }
     }
@@ -63,8 +63,8 @@ class MatrixTestHelpers {
 
     const complex<float>* const* expected_elements = expected.elements();
     const complex<float>* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_NEAR(expected_elements[i][j].real(),
                     actual_elements[i][j].real(),
                     kTolerance);
@@ -84,8 +84,8 @@ class MatrixTestHelpers {
 
     const complex<float>* const* expected_elements = expected.elements();
     const complex<float>* const* actual_elements = actual.elements();
-    for (int i = 0; i < expected.num_rows(); ++i) {
-      for (int j = 0; j < expected.num_columns(); ++j) {
+    for (size_t i = 0; i < expected.num_rows(); ++i) {
+      for (size_t j = 0; j < expected.num_columns(); ++j) {
         EXPECT_NEAR(expected_elements[i][j].real(),
                     actual_elements[i][j].real(),
                     tolerance);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
index eb05ecdab3..e2b4417c13 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
@@ -20,11 +20,13 @@ namespace webrtc {
 
 class MockNonlinearBeamformer : public NonlinearBeamformer {
  public:
-  explicit MockNonlinearBeamformer(const std::vector<Point>& array_geometry);
+  explicit MockNonlinearBeamformer(const std::vector<Point>& array_geometry)
+      : NonlinearBeamformer(array_geometry) {}
 
   MOCK_METHOD2(Initialize, void(int chunk_size_ms, int sample_rate_hz));
   MOCK_METHOD2(ProcessChunk, void(const ChannelBuffer<float>& input,
                                   ChannelBuffer<float>* output));
+  MOCK_METHOD1(IsInBeam, bool(const SphericalPointf& spherical_point));
   MOCK_METHOD0(is_target_present, bool());
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
index 8fd6c687f4..1039eddfc3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
@@ -8,12 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef _USE_MATH_DEFINES
 #define _USE_MATH_DEFINES
+#endif
 
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
 
 #include <algorithm>
 #include <cmath>
+#include <numeric>
 #include <vector>
 
 #include "webrtc/base/arraysize.h"
@@ -24,61 +27,63 @@ namespace webrtc {
 namespace {
 
 // Alpha for the Kaiser Bessel Derived window.
-const float kAlpha = 1.5f;
-
-// The minimum value a post-processing mask can take.
-const float kMaskMinimum = 0.01f;
+const float kKbdAlpha = 1.5f;
 
 const float kSpeedOfSoundMeterSeconds = 343;
 
-// For both target and interference angles, PI / 2 is perpendicular to the
-// microphone array, facing forwards. The positive direction goes
-// counterclockwise.
-// The angle at which we amplify sound.
-const float kTargetAngleRadians = static_cast<float>(M_PI) / 2.f;
+// The minimum separation in radians between the target direction and an
+// interferer scenario.
+const float kMinAwayRadians = 0.2f;
 
-// The angle at which we suppress sound. Suppression is symmetric around PI / 2
-// radians, so sound is suppressed at both +|kInterfAngleRadians| and
-// PI - |kInterfAngleRadians|. Since the beamformer is robust, this should
-// suppress sound coming from close angles as well.
-const float kInterfAngleRadians = static_cast<float>(M_PI) / 4.f;
+// The separation between the target direction and the closest interferer
+// scenario is proportional to this constant.
+const float kAwaySlope = 0.008f;
 
 // When calculating the interference covariance matrix, this is the weight for
 // the weighted average between the uniform covariance matrix and the angled
 // covariance matrix.
 // Rpsi = Rpsi_angled * kBalance + Rpsi_uniform * (1 - kBalance)
-const float kBalance = 0.4f;
+const float kBalance = 0.95f;
 
-// TODO(claguna): need comment here.
-const float kBeamwidthConstant = 0.00002f;
-
-// Alpha coefficient for mask smoothing.
-const float kMaskSmoothAlpha = 0.2f;
+// Alpha coefficients for mask smoothing.
+const float kMaskTimeSmoothAlpha = 0.2f;
+const float kMaskFrequencySmoothAlpha = 0.6f;
 
 // The average mask is computed from masks in this mid-frequency range. If these
 // ranges are changed |kMaskQuantile| might need to be adjusted.
-const int kLowAverageStartHz = 200;
-const int kLowAverageEndHz = 400;
+const int kLowMeanStartHz = 200;
+const int kLowMeanEndHz = 400;
 
-const int kHighAverageStartHz = 6000;
-const int kHighAverageEndHz = 6500;
+// Range limiter for subtractive terms in the nominator and denominator of the
+// postfilter expression. It handles the scenario mismatch between the true and
+// model sources (target and interference).
+const float kCutOffConstant = 0.9999f;
 
 // Quantile of mask values which is used to estimate target presence.
-const float kMaskQuantile = 0.3f;
+const float kMaskQuantile = 0.7f;
 // Mask threshold over which the data is considered signal and not interference.
-const float kMaskTargetThreshold = 0.3f;
+// It has to be updated every time the postfilter calculation is changed
+// significantly.
+// TODO(aluebs): Write a tool to tune the target threshold automatically based
+// on files annotated with target and interference ground truth.
+const float kMaskTargetThreshold = 0.01f;
 // Time in seconds after which the data is considered interference if the mask
 // does not pass |kMaskTargetThreshold|.
 const float kHoldTargetSeconds = 0.25f;
 
+// To compensate for the attenuation this algorithm introduces to the target
+// signal. It was estimated empirically from a low-noise low-reverberation
+// recording from broadside.
+const float kCompensationGain = 2.f;
+
 // Does conjugate(|norm_mat|) * |mat| * transpose(|norm_mat|). No extra space is
 // used; to accomplish this, we compute both multiplications in the same loop.
 // The returned norm is clamped to be non-negative.
 float Norm(const ComplexMatrix<float>& mat,
            const ComplexMatrix<float>& norm_mat) {
-  CHECK_EQ(norm_mat.num_rows(), 1);
-  CHECK_EQ(norm_mat.num_columns(), mat.num_rows());
-  CHECK_EQ(norm_mat.num_columns(), mat.num_columns());
+  RTC_CHECK_EQ(1u, norm_mat.num_rows());
+  RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_rows());
+  RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_columns());
 
   complex<float> first_product = complex<float>(0.f, 0.f);
   complex<float> second_product = complex<float>(0.f, 0.f);
@@ -86,8 +91,8 @@ float Norm(const ComplexMatrix<float>& mat,
   const complex<float>* const* mat_els = mat.elements();
   const complex<float>* const* norm_mat_els = norm_mat.elements();
 
-  for (int i = 0; i < norm_mat.num_columns(); ++i) {
-    for (int j = 0; j < norm_mat.num_columns(); ++j) {
+  for (size_t i = 0; i < norm_mat.num_columns(); ++i) {
+    for (size_t j = 0; j < norm_mat.num_columns(); ++j) {
       first_product += conj(norm_mat_els[0][j]) * mat_els[j][i];
     }
     second_product += first_product * norm_mat_els[0][i];
@@ -99,15 +104,15 @@ float Norm(const ComplexMatrix<float>& mat,
 // Does conjugate(|lhs|) * |rhs| for row vectors |lhs| and |rhs|.
 complex<float> ConjugateDotProduct(const ComplexMatrix<float>& lhs,
                                    const ComplexMatrix<float>& rhs) {
-  CHECK_EQ(lhs.num_rows(), 1);
-  CHECK_EQ(rhs.num_rows(), 1);
-  CHECK_EQ(lhs.num_columns(), rhs.num_columns());
+  RTC_CHECK_EQ(1u, lhs.num_rows());
+  RTC_CHECK_EQ(1u, rhs.num_rows());
+  RTC_CHECK_EQ(lhs.num_columns(), rhs.num_columns());
 
   const complex<float>* const* lhs_elements = lhs.elements();
   const complex<float>* const* rhs_elements = rhs.elements();
 
   complex<float> result = complex<float>(0.f, 0.f);
-  for (int i = 0; i < lhs.num_columns(); ++i) {
+  for (size_t i = 0; i < lhs.num_columns(); ++i) {
     result += conj(lhs_elements[0][i]) * rhs_elements[0][i];
   }
 
@@ -115,16 +120,16 @@ complex<float> ConjugateDotProduct(const ComplexMatrix<float>& lhs,
 }
 
 // Works for positive numbers only.
-int Round(float x) {
-  return std::floor(x + 0.5f);
+size_t Round(float x) {
+  return static_cast<size_t>(std::floor(x + 0.5f));
 }
 
 // Calculates the sum of absolute values of a complex matrix.
 float SumAbs(const ComplexMatrix<float>& mat) {
   float sum_abs = 0.f;
   const complex<float>* const* mat_els = mat.elements();
-  for (int i = 0; i < mat.num_rows(); ++i) {
-    for (int j = 0; j < mat.num_columns(); ++j) {
+  for (size_t i = 0; i < mat.num_rows(); ++i) {
+    for (size_t j = 0; j < mat.num_columns(); ++j) {
       sum_abs += std::abs(mat_els[i][j]);
     }
   }
@@ -135,8 +140,8 @@ float SumAbs(const ComplexMatrix<float>& mat) {
 float SumSquares(const ComplexMatrix<float>& mat) {
   float sum_squares = 0.f;
   const complex<float>* const* mat_els = mat.elements();
-  for (int i = 0; i < mat.num_rows(); ++i) {
-    for (int j = 0; j < mat.num_columns(); ++j) {
+  for (size_t i = 0; i < mat.num_rows(); ++i) {
+    for (size_t j = 0; j < mat.num_columns(); ++j) {
       float abs_value = std::abs(mat_els[i][j]);
       sum_squares += abs_value * abs_value;
     }
@@ -147,20 +152,20 @@ float SumSquares(const ComplexMatrix<float>& mat) {
 // Does |out| = |in|.' * conj(|in|) for row vector |in|.
 void TransposedConjugatedProduct(const ComplexMatrix<float>& in,
                                  ComplexMatrix<float>* out) {
-  CHECK_EQ(in.num_rows(), 1);
-  CHECK_EQ(out->num_rows(), in.num_columns());
-  CHECK_EQ(out->num_columns(), in.num_columns());
+  RTC_CHECK_EQ(1u, in.num_rows());
+  RTC_CHECK_EQ(out->num_rows(), in.num_columns());
+  RTC_CHECK_EQ(out->num_columns(), in.num_columns());
   const complex<float>* in_elements = in.elements()[0];
   complex<float>* const* out_elements = out->elements();
-  for (int i = 0; i < out->num_rows(); ++i) {
-    for (int j = 0; j < out->num_columns(); ++j) {
+  for (size_t i = 0; i < out->num_rows(); ++i) {
+    for (size_t j = 0; j < out->num_columns(); ++j) {
       out_elements[i][j] = in_elements[i] * conj(in_elements[j]);
     }
   }
 }
 
 std::vector<Point> GetCenteredArray(std::vector<Point> array_geometry) {
-  for (int dim = 0; dim < 3; ++dim) {
+  for (size_t dim = 0; dim < 3; ++dim) {
     float center = 0.f;
     for (size_t i = 0; i < array_geometry.size(); ++i) {
       center += array_geometry[i].c[dim];
@@ -175,34 +180,36 @@ std::vector<Point> GetCenteredArray(std::vector<Point> array_geometry) {
 
 }  // namespace
 
+const float NonlinearBeamformer::kHalfBeamWidthRadians = DegreesToRadians(20.f);
+
+// static
+const size_t NonlinearBeamformer::kNumFreqBins;
+
 NonlinearBeamformer::NonlinearBeamformer(
-    const std::vector<Point>& array_geometry)
-  : num_input_channels_(array_geometry.size()),
-      array_geometry_(GetCenteredArray(array_geometry)) {
-  WindowGenerator::KaiserBesselDerived(kAlpha, kFftSize, window_);
+    const std::vector<Point>& array_geometry,
+    SphericalPointf target_direction)
+    : num_input_channels_(array_geometry.size()),
+      array_geometry_(GetCenteredArray(array_geometry)),
+      array_normal_(GetArrayNormalIfExists(array_geometry)),
+      min_mic_spacing_(GetMinimumSpacing(array_geometry)),
+      target_angle_radians_(target_direction.azimuth()),
+      away_radians_(std::min(
+          static_cast<float>(M_PI),
+          std::max(kMinAwayRadians,
+                   kAwaySlope * static_cast<float>(M_PI) / min_mic_spacing_))) {
+  WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_);
 }
 
 void NonlinearBeamformer::Initialize(int chunk_size_ms, int sample_rate_hz) {
-  chunk_length_ = sample_rate_hz / (1000.f / chunk_size_ms);
+  chunk_length_ =
+      static_cast<size_t>(sample_rate_hz / (1000.f / chunk_size_ms));
   sample_rate_hz_ = sample_rate_hz;
-  low_average_start_bin_ =
-      Round(kLowAverageStartHz * kFftSize / sample_rate_hz_);
-  low_average_end_bin_ =
-      Round(kLowAverageEndHz * kFftSize / sample_rate_hz_);
-  high_average_start_bin_ =
-      Round(kHighAverageStartHz * kFftSize / sample_rate_hz_);
-  high_average_end_bin_ =
-      Round(kHighAverageEndHz * kFftSize / sample_rate_hz_);
+
   high_pass_postfilter_mask_ = 1.f;
   is_target_present_ = false;
   hold_target_blocks_ = kHoldTargetSeconds * 2 * sample_rate_hz / kFftSize;
   interference_blocks_count_ = hold_target_blocks_;
 
-  DCHECK_LE(low_average_end_bin_, kNumFreqBins);
-  DCHECK_LT(low_average_start_bin_, low_average_end_bin_);
-  DCHECK_LE(high_average_end_bin_, kNumFreqBins);
-  DCHECK_LT(high_average_start_bin_, high_average_end_bin_);
-
   lapped_transform_.reset(new LappedTransform(num_input_channels_,
                                               1,
                                               chunk_length_,
@@ -210,38 +217,93 @@ void NonlinearBeamformer::Initialize(int chunk_size_ms, int sample_rate_hz) {
                                               kFftSize,
                                               kFftSize / 2,
                                               this));
-  for (int i = 0; i < kNumFreqBins; ++i) {
-    postfilter_mask_[i] = 1.f;
+  for (size_t i = 0; i < kNumFreqBins; ++i) {
+    time_smooth_mask_[i] = 1.f;
+    final_mask_[i] = 1.f;
     float freq_hz = (static_cast<float>(i) / kFftSize) * sample_rate_hz_;
     wave_numbers_[i] = 2 * M_PI * freq_hz / kSpeedOfSoundMeterSeconds;
-    mask_thresholds_[i] = num_input_channels_ * num_input_channels_ *
-                          kBeamwidthConstant * wave_numbers_[i] *
-                          wave_numbers_[i];
   }
 
-  // Initialize all nonadaptive values before looping through the frames.
-  InitDelaySumMasks();
-  InitTargetCovMats();
-  InitInterfCovMats();
+  InitLowFrequencyCorrectionRanges();
+  InitDiffuseCovMats();
+  AimAt(SphericalPointf(target_angle_radians_, 0.f, 1.f));
+}
 
-  for (int i = 0; i < kNumFreqBins; ++i) {
-    rxiws_[i] = Norm(target_cov_mats_[i], delay_sum_masks_[i]);
-    rpsiws_[i] = Norm(interf_cov_mats_[i], delay_sum_masks_[i]);
-    reflected_rpsiws_[i] =
-        Norm(reflected_interf_cov_mats_[i], delay_sum_masks_[i]);
+// These bin indexes determine the regions over which a mean is taken. This is
+// applied as a constant value over the adjacent end "frequency correction"
+// regions.
+//
+//             low_mean_start_bin_     high_mean_start_bin_
+//                   v                         v              constant
+// |----------------|--------|----------------|-------|----------------|
+//   constant               ^                        ^
+//             low_mean_end_bin_       high_mean_end_bin_
+//
+void NonlinearBeamformer::InitLowFrequencyCorrectionRanges() {
+  low_mean_start_bin_ = Round(kLowMeanStartHz * kFftSize / sample_rate_hz_);
+  low_mean_end_bin_ = Round(kLowMeanEndHz * kFftSize / sample_rate_hz_);
+
+  RTC_DCHECK_GT(low_mean_start_bin_, 0U);
+  RTC_DCHECK_LT(low_mean_start_bin_, low_mean_end_bin_);
+}
+
+void NonlinearBeamformer::InitHighFrequencyCorrectionRanges() {
+  const float kAliasingFreqHz =
+      kSpeedOfSoundMeterSeconds /
+      (min_mic_spacing_ * (1.f + std::abs(std::cos(target_angle_radians_))));
+  const float kHighMeanStartHz = std::min(0.5f *  kAliasingFreqHz,
+                                          sample_rate_hz_ / 2.f);
+  const float kHighMeanEndHz = std::min(0.75f *  kAliasingFreqHz,
+                                        sample_rate_hz_ / 2.f);
+  high_mean_start_bin_ = Round(kHighMeanStartHz * kFftSize / sample_rate_hz_);
+  high_mean_end_bin_ = Round(kHighMeanEndHz * kFftSize / sample_rate_hz_);
+
+  RTC_DCHECK_LT(low_mean_end_bin_, high_mean_end_bin_);
+  RTC_DCHECK_LT(high_mean_start_bin_, high_mean_end_bin_);
+  RTC_DCHECK_LT(high_mean_end_bin_, kNumFreqBins - 1);
+}
+
+void NonlinearBeamformer::InitInterfAngles() {
+  interf_angles_radians_.clear();
+  const Point target_direction = AzimuthToPoint(target_angle_radians_);
+  const Point clockwise_interf_direction =
+      AzimuthToPoint(target_angle_radians_ - away_radians_);
+  if (!array_normal_ ||
+      DotProduct(*array_normal_, target_direction) *
+              DotProduct(*array_normal_, clockwise_interf_direction) >=
+          0.f) {
+    // The target and clockwise interferer are in the same half-plane defined
+    // by the array.
+    interf_angles_radians_.push_back(target_angle_radians_ - away_radians_);
+  } else {
+    // Otherwise, the interferer will begin reflecting back at the target.
+    // Instead rotate it away 180 degrees.
+    interf_angles_radians_.push_back(target_angle_radians_ - away_radians_ +
+                                     M_PI);
+  }
+  const Point counterclock_interf_direction =
+      AzimuthToPoint(target_angle_radians_ + away_radians_);
+  if (!array_normal_ ||
+      DotProduct(*array_normal_, target_direction) *
+              DotProduct(*array_normal_, counterclock_interf_direction) >=
+          0.f) {
+    // The target and counter-clockwise interferer are in the same half-plane
+    // defined by the array.
+    interf_angles_radians_.push_back(target_angle_radians_ + away_radians_);
+  } else {
+    // Otherwise, the interferer will begin reflecting back at the target.
+    // Instead rotate it away 180 degrees.
+    interf_angles_radians_.push_back(target_angle_radians_ + away_radians_ -
+                                     M_PI);
   }
 }
 
 void NonlinearBeamformer::InitDelaySumMasks() {
-  for (int f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
+  for (size_t f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
     delay_sum_masks_[f_ix].Resize(1, num_input_channels_);
-    CovarianceMatrixGenerator::PhaseAlignmentMasks(f_ix,
-                                                   kFftSize,
-                                                   sample_rate_hz_,
-                                                   kSpeedOfSoundMeterSeconds,
-                                                   array_geometry_,
-                                                   kTargetAngleRadians,
-                                                   &delay_sum_masks_[f_ix]);
+    CovarianceMatrixGenerator::PhaseAlignmentMasks(
+        f_ix, kFftSize, sample_rate_hz_, kSpeedOfSoundMeterSeconds,
+        array_geometry_, target_angle_radians_, &delay_sum_masks_[f_ix]);
 
     complex_f norm_factor = sqrt(
         ConjugateDotProduct(delay_sum_masks_[f_ix], delay_sum_masks_[f_ix]));
@@ -253,50 +315,63 @@ void NonlinearBeamformer::InitDelaySumMasks() {
 }
 
 void NonlinearBeamformer::InitTargetCovMats() {
-  for (int i = 0; i < kNumFreqBins; ++i) {
+  for (size_t i = 0; i < kNumFreqBins; ++i) {
     target_cov_mats_[i].Resize(num_input_channels_, num_input_channels_);
     TransposedConjugatedProduct(delay_sum_masks_[i], &target_cov_mats_[i]);
-    complex_f normalization_factor = target_cov_mats_[i].Trace();
-    target_cov_mats_[i].Scale(1.f / normalization_factor);
+  }
+}
+
+void NonlinearBeamformer::InitDiffuseCovMats() {
+  for (size_t i = 0; i < kNumFreqBins; ++i) {
+    uniform_cov_mat_[i].Resize(num_input_channels_, num_input_channels_);
+    CovarianceMatrixGenerator::UniformCovarianceMatrix(
+        wave_numbers_[i], array_geometry_, &uniform_cov_mat_[i]);
+    complex_f normalization_factor = uniform_cov_mat_[i].elements()[0][0];
+    uniform_cov_mat_[i].Scale(1.f / normalization_factor);
+    uniform_cov_mat_[i].Scale(1 - kBalance);
   }
 }
 
 void NonlinearBeamformer::InitInterfCovMats() {
-  for (int i = 0; i < kNumFreqBins; ++i) {
-    interf_cov_mats_[i].Resize(num_input_channels_, num_input_channels_);
-    ComplexMatrixF uniform_cov_mat(num_input_channels_, num_input_channels_);
-    ComplexMatrixF angled_cov_mat(num_input_channels_, num_input_channels_);
+  for (size_t i = 0; i < kNumFreqBins; ++i) {
+    interf_cov_mats_[i].clear();
+    for (size_t j = 0; j < interf_angles_radians_.size(); ++j) {
+      interf_cov_mats_[i].push_back(new ComplexMatrixF(num_input_channels_,
+                                                       num_input_channels_));
+      ComplexMatrixF angled_cov_mat(num_input_channels_, num_input_channels_);
+      CovarianceMatrixGenerator::AngledCovarianceMatrix(
+          kSpeedOfSoundMeterSeconds,
+          interf_angles_radians_[j],
+          i,
+          kFftSize,
+          kNumFreqBins,
+          sample_rate_hz_,
+          array_geometry_,
+          &angled_cov_mat);
+      // Normalize matrices before averaging them.
+      complex_f normalization_factor = angled_cov_mat.elements()[0][0];
+      angled_cov_mat.Scale(1.f / normalization_factor);
+      // Weighted average of matrices.
+      angled_cov_mat.Scale(kBalance);
+      interf_cov_mats_[i][j]->Add(uniform_cov_mat_[i], angled_cov_mat);
+    }
+  }
+}
 
-    CovarianceMatrixGenerator::UniformCovarianceMatrix(wave_numbers_[i],
-                                                       array_geometry_,
-                                                       &uniform_cov_mat);
-
-    CovarianceMatrixGenerator::AngledCovarianceMatrix(kSpeedOfSoundMeterSeconds,
-                                                      kInterfAngleRadians,
-                                                      i,
-                                                      kFftSize,
-                                                      kNumFreqBins,
-                                                      sample_rate_hz_,
-                                                      array_geometry_,
-                                                      &angled_cov_mat);
-    // Normalize matrices before averaging them.
-    complex_f normalization_factor = uniform_cov_mat.Trace();
-    uniform_cov_mat.Scale(1.f / normalization_factor);
-    normalization_factor = angled_cov_mat.Trace();
-    angled_cov_mat.Scale(1.f / normalization_factor);
-
-    // Average matrices.
-    uniform_cov_mat.Scale(1 - kBalance);
-    angled_cov_mat.Scale(kBalance);
-    interf_cov_mats_[i].Add(uniform_cov_mat, angled_cov_mat);
-    reflected_interf_cov_mats_[i].PointwiseConjugate(interf_cov_mats_[i]);
+void NonlinearBeamformer::NormalizeCovMats() {
+  for (size_t i = 0; i < kNumFreqBins; ++i) {
+    rxiws_[i] = Norm(target_cov_mats_[i], delay_sum_masks_[i]);
+    rpsiws_[i].clear();
+    for (size_t j = 0; j < interf_angles_radians_.size(); ++j) {
+      rpsiws_[i].push_back(Norm(*interf_cov_mats_[i][j], delay_sum_masks_[i]));
+    }
   }
 }
 
 void NonlinearBeamformer::ProcessChunk(const ChannelBuffer<float>& input,
-                              ChannelBuffer<float>* output) {
-  DCHECK_EQ(input.num_channels(), num_input_channels_);
-  DCHECK_EQ(input.num_frames_per_band(), chunk_length_);
+                                       ChannelBuffer<float>* output) {
+  RTC_DCHECK_EQ(input.num_channels(), num_input_channels_);
+  RTC_DCHECK_EQ(input.num_frames_per_band(), chunk_length_);
 
   float old_high_pass_mask = high_pass_postfilter_mask_;
   lapped_transform_->ProcessChunk(input.channels(0), output->channels(0));
@@ -305,37 +380,48 @@ void NonlinearBeamformer::ProcessChunk(const ChannelBuffer<float>& input,
   const float ramp_increment =
       (high_pass_postfilter_mask_ - old_high_pass_mask) /
       input.num_frames_per_band();
-  // Apply delay and sum and post-filter in the time domain. WARNING: only works
-  // because delay-and-sum is not frequency dependent.
-  for (int i = 1; i < input.num_bands(); ++i) {
+  // Apply the smoothed high-pass mask to the first channel of each band.
+  // This can be done because the effect of the linear beamformer is negligible
+  // compared to the post-filter.
+  for (size_t i = 1; i < input.num_bands(); ++i) {
     float smoothed_mask = old_high_pass_mask;
-    for (int j = 0; j < input.num_frames_per_band(); ++j) {
+    for (size_t j = 0; j < input.num_frames_per_band(); ++j) {
       smoothed_mask += ramp_increment;
-
-      // Applying the delay and sum (at zero degrees, this is equivalent to
-      // averaging).
-      float sum = 0.f;
-      for (int k = 0; k < input.num_channels(); ++k) {
-        sum += input.channels(i)[k][j];
-      }
-      output->channels(i)[0][j] = sum / input.num_channels() * smoothed_mask;
+      output->channels(i)[0][j] = input.channels(i)[0][j] * smoothed_mask;
     }
   }
 }
 
+void NonlinearBeamformer::AimAt(const SphericalPointf& target_direction) {
+  target_angle_radians_ = target_direction.azimuth();
+  InitHighFrequencyCorrectionRanges();
+  InitInterfAngles();
+  InitDelaySumMasks();
+  InitTargetCovMats();
+  InitInterfCovMats();
+  NormalizeCovMats();
+}
+
+bool NonlinearBeamformer::IsInBeam(const SphericalPointf& spherical_point) {
+  // If more than half-beamwidth degrees away from the beam's center,
+  // you are out of the beam.
+  return fabs(spherical_point.azimuth() - target_angle_radians_) <
+         kHalfBeamWidthRadians;
+}
+
 void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input,
-                                   int num_input_channels,
-                                   int num_freq_bins,
-                                   int num_output_channels,
-                                   complex_f* const* output) {
-  CHECK_EQ(num_freq_bins, kNumFreqBins);
-  CHECK_EQ(num_input_channels, num_input_channels_);
-  CHECK_EQ(num_output_channels, 1);
+                                            size_t num_input_channels,
+                                            size_t num_freq_bins,
+                                            size_t num_output_channels,
+                                            complex_f* const* output) {
+  RTC_CHECK_EQ(kNumFreqBins, num_freq_bins);
+  RTC_CHECK_EQ(num_input_channels_, num_input_channels);
+  RTC_CHECK_EQ(1u, num_output_channels);
 
   // Calculating the post-filter masks. Note that we need two for each
   // frequency bin to account for the positive and negative interferer
   // angle.
-  for (int i = low_average_start_bin_; i < high_average_end_bin_; ++i) {
+  for (size_t i = low_mean_start_bin_; i <= high_mean_end_bin_; ++i) {
     eig_m_.CopyFromColumn(input, i, num_input_channels_);
     float eig_m_norm_factor = std::sqrt(SumSquares(eig_m_));
     if (eig_m_norm_factor != 0.f) {
@@ -352,106 +438,129 @@ void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input,
     rmw *= rmw;
     float rmw_r = rmw.real();
 
-    new_mask_[i] = CalculatePostfilterMask(interf_cov_mats_[i],
-                                           rpsiws_[i],
+    new_mask_[i] = CalculatePostfilterMask(*interf_cov_mats_[i][0],
+                                           rpsiws_[i][0],
                                            ratio_rxiw_rxim,
-                                           rmw_r,
-                                           mask_thresholds_[i]);
-
-    new_mask_[i] *= CalculatePostfilterMask(reflected_interf_cov_mats_[i],
-                                            reflected_rpsiws_[i],
-                                            ratio_rxiw_rxim,
-                                            rmw_r,
-                                            mask_thresholds_[i]);
+                                           rmw_r);
+    for (size_t j = 1; j < interf_angles_radians_.size(); ++j) {
+      float tmp_mask = CalculatePostfilterMask(*interf_cov_mats_[i][j],
+                                               rpsiws_[i][j],
+                                               ratio_rxiw_rxim,
+                                               rmw_r);
+      if (tmp_mask < new_mask_[i]) {
+        new_mask_[i] = tmp_mask;
+      }
+    }
   }
 
-  ApplyMaskSmoothing();
+  ApplyMaskTimeSmoothing();
+  EstimateTargetPresence();
   ApplyLowFrequencyCorrection();
   ApplyHighFrequencyCorrection();
+  ApplyMaskFrequencySmoothing();
   ApplyMasks(input, output);
-
-  EstimateTargetPresence();
 }
 
 float NonlinearBeamformer::CalculatePostfilterMask(
     const ComplexMatrixF& interf_cov_mat,
     float rpsiw,
     float ratio_rxiw_rxim,
-    float rmw_r,
-    float mask_threshold) {
+    float rmw_r) {
   float rpsim = Norm(interf_cov_mat, eig_m_);
 
-  // Find lambda.
   float ratio = 0.f;
   if (rpsim > 0.f) {
     ratio = rpsiw / rpsim;
   }
-  float numerator = rmw_r - ratio;
-  float denominator = ratio_rxiw_rxim - ratio;
 
-  float mask = 1.f;
-  if (denominator > mask_threshold) {
-    float lambda = numerator / denominator;
-    mask = std::max(lambda * ratio_rxiw_rxim / rmw_r, kMaskMinimum);
-  }
-  return mask;
+  return (1.f - std::min(kCutOffConstant, ratio / rmw_r)) /
+         (1.f - std::min(kCutOffConstant, ratio / ratio_rxiw_rxim));
 }
 
 void NonlinearBeamformer::ApplyMasks(const complex_f* const* input,
-                            complex_f* const* output) {
+                                     complex_f* const* output) {
   complex_f* output_channel = output[0];
-  for (int f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
+  for (size_t f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
     output_channel[f_ix] = complex_f(0.f, 0.f);
 
     const complex_f* delay_sum_mask_els =
         normalized_delay_sum_masks_[f_ix].elements()[0];
-    for (int c_ix = 0; c_ix < num_input_channels_; ++c_ix) {
+    for (size_t c_ix = 0; c_ix < num_input_channels_; ++c_ix) {
       output_channel[f_ix] += input[c_ix][f_ix] * delay_sum_mask_els[c_ix];
     }
 
-    output_channel[f_ix] *= postfilter_mask_[f_ix];
+    output_channel[f_ix] *= kCompensationGain * final_mask_[f_ix];
   }
 }
 
-void NonlinearBeamformer::ApplyMaskSmoothing() {
-  for (int i = 0; i < kNumFreqBins; ++i) {
-    postfilter_mask_[i] = kMaskSmoothAlpha * new_mask_[i] +
-                          (1.f - kMaskSmoothAlpha) * postfilter_mask_[i];
+// Smooth new_mask_ into time_smooth_mask_.
+void NonlinearBeamformer::ApplyMaskTimeSmoothing() {
+  for (size_t i = low_mean_start_bin_; i <= high_mean_end_bin_; ++i) {
+    time_smooth_mask_[i] = kMaskTimeSmoothAlpha * new_mask_[i] +
+                           (1 - kMaskTimeSmoothAlpha) * time_smooth_mask_[i];
   }
 }
 
+// Copy time_smooth_mask_ to final_mask_ and smooth over frequency.
+void NonlinearBeamformer::ApplyMaskFrequencySmoothing() {
+  // Smooth over frequency in both directions. The "frequency correction"
+  // regions have constant value, but we enter them to smooth over the jump
+  // that exists at the boundary. However, this does mean when smoothing "away"
+  // from the region that we only need to use the last element.
+  //
+  // Upward smoothing:
+  //   low_mean_start_bin_
+  //         v
+  // |------|------------|------|
+  //       ^------------------>^
+  //
+  // Downward smoothing:
+  //         high_mean_end_bin_
+  //                    v
+  // |------|------------|------|
+  //  ^<------------------^
+  std::copy(time_smooth_mask_, time_smooth_mask_ + kNumFreqBins, final_mask_);
+  for (size_t i = low_mean_start_bin_; i < kNumFreqBins; ++i) {
+    final_mask_[i] = kMaskFrequencySmoothAlpha * final_mask_[i] +
+                     (1 - kMaskFrequencySmoothAlpha) * final_mask_[i - 1];
+  }
+  for (size_t i = high_mean_end_bin_ + 1; i > 0; --i) {
+    final_mask_[i - 1] = kMaskFrequencySmoothAlpha * final_mask_[i - 1] +
+                         (1 - kMaskFrequencySmoothAlpha) * final_mask_[i];
+  }
+}
+
+// Apply low frequency correction to time_smooth_mask_.
 void NonlinearBeamformer::ApplyLowFrequencyCorrection() {
-  float low_frequency_mask = 0.f;
-  for (int i = low_average_start_bin_; i < low_average_end_bin_; ++i) {
-    low_frequency_mask += postfilter_mask_[i];
-  }
-
-  low_frequency_mask /= low_average_end_bin_ - low_average_start_bin_;
-
-  for (int i = 0; i < low_average_start_bin_; ++i) {
-    postfilter_mask_[i] = low_frequency_mask;
-  }
+  const float low_frequency_mask =
+      MaskRangeMean(low_mean_start_bin_, low_mean_end_bin_ + 1);
+  std::fill(time_smooth_mask_, time_smooth_mask_ + low_mean_start_bin_,
+            low_frequency_mask);
 }
 
+// Apply high frequency correction to time_smooth_mask_. Update
+// high_pass_postfilter_mask_ to use for the high frequency time-domain bands.
 void NonlinearBeamformer::ApplyHighFrequencyCorrection() {
-  high_pass_postfilter_mask_ = 0.f;
-  for (int i = high_average_start_bin_; i < high_average_end_bin_; ++i) {
-    high_pass_postfilter_mask_ += postfilter_mask_[i];
-  }
+  high_pass_postfilter_mask_ =
+      MaskRangeMean(high_mean_start_bin_, high_mean_end_bin_ + 1);
+  std::fill(time_smooth_mask_ + high_mean_end_bin_ + 1,
+            time_smooth_mask_ + kNumFreqBins, high_pass_postfilter_mask_);
+}
 
-  high_pass_postfilter_mask_ /= high_average_end_bin_ - high_average_start_bin_;
-
-  for (int i = high_average_end_bin_; i < kNumFreqBins; ++i) {
-    postfilter_mask_[i] = high_pass_postfilter_mask_;
-  }
+// Compute mean over the given range of time_smooth_mask_, [first, last).
+float NonlinearBeamformer::MaskRangeMean(size_t first, size_t last) {
+  RTC_DCHECK_GT(last, first);
+  const float sum = std::accumulate(time_smooth_mask_ + first,
+                                    time_smooth_mask_ + last, 0.f);
+  return sum / (last - first);
 }
 
 void NonlinearBeamformer::EstimateTargetPresence() {
-  const int quantile = (1.f - kMaskQuantile) * high_average_end_bin_ +
-                       kMaskQuantile * low_average_start_bin_;
-  std::nth_element(new_mask_ + low_average_start_bin_,
-                   new_mask_ + quantile,
-                   new_mask_ + high_average_end_bin_);
+  const size_t quantile = static_cast<size_t>(
+      (high_mean_end_bin_ - low_mean_start_bin_) * kMaskQuantile +
+      low_mean_start_bin_);
+  std::nth_element(new_mask_ + low_mean_start_bin_, new_mask_ + quantile,
+                   new_mask_ + high_mean_end_bin_ + 1);
   if (new_mask_[quantile] > kMaskTargetThreshold) {
     is_target_present_ = true;
     interference_blocks_count_ = 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
index bebfad8b1f..fe5b866c81 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
@@ -11,13 +11,19 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
 
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#ifndef _USE_MATH_DEFINES
+#define _USE_MATH_DEFINES
+#endif
+
+#include <math.h>
 #include <vector>
 
 #include "webrtc/common_audio/lapped_transform.h"
 #include "webrtc/common_audio/channel_buffer.h"
-#include "webrtc/modules/audio_processing/beamformer/array_util.h"
 #include "webrtc/modules/audio_processing/beamformer/beamformer.h"
 #include "webrtc/modules/audio_processing/beamformer/complex_matrix.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
 
 namespace webrtc {
 
@@ -27,15 +33,16 @@ namespace webrtc {
 //
 // The implemented nonlinear postfilter algorithm taken from "A Robust Nonlinear
 // Beamforming Postprocessor" by Bastiaan Kleijn.
-//
-// TODO: Target angle assumed to be 0. Parameterize target angle.
 class NonlinearBeamformer
   : public Beamformer<float>,
     public LappedTransform::Callback {
  public:
-  // At the moment it only accepts uniform linear microphone arrays. Using the
-  // first microphone as a reference position [0, 0, 0] is a natural choice.
-  explicit NonlinearBeamformer(const std::vector<Point>& array_geometry);
+  static const float kHalfBeamWidthRadians;
+
+  explicit NonlinearBeamformer(
+      const std::vector<Point>& array_geometry,
+      SphericalPointf target_direction =
+          SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f));
 
   // Sample rate corresponds to the lower band.
   // Needs to be called before the NonlinearBeamformer can be used.
@@ -48,6 +55,10 @@ class NonlinearBeamformer
   void ProcessChunk(const ChannelBuffer<float>& input,
                     ChannelBuffer<float>* output) override;
 
+  void AimAt(const SphericalPointf& target_direction) override;
+
+  bool IsInBeam(const SphericalPointf& spherical_point) override;
+
   // After processing each block |is_target_present_| is set to true if the
   // target signal es present and to false otherwise. This methods can be called
   // to know if the data is target signal or interference and process it
@@ -58,33 +69,39 @@ class NonlinearBeamformer
   // Process one frequency-domain block of audio. This is where the fun
   // happens. Implements LappedTransform::Callback.
   void ProcessAudioBlock(const complex<float>* const* input,
-                         int num_input_channels,
-                         int num_freq_bins,
-                         int num_output_channels,
+                         size_t num_input_channels,
+                         size_t num_freq_bins,
+                         size_t num_output_channels,
                          complex<float>* const* output) override;
 
  private:
+  FRIEND_TEST_ALL_PREFIXES(NonlinearBeamformerTest,
+                           InterfAnglesTakeAmbiguityIntoAccount);
+
   typedef Matrix<float> MatrixF;
   typedef ComplexMatrix<float> ComplexMatrixF;
   typedef complex<float> complex_f;
 
+  void InitLowFrequencyCorrectionRanges();
+  void InitHighFrequencyCorrectionRanges();
+  void InitInterfAngles();
   void InitDelaySumMasks();
-  void InitTargetCovMats();  // TODO: Make this depend on target angle.
+  void InitTargetCovMats();
+  void InitDiffuseCovMats();
   void InitInterfCovMats();
+  void NormalizeCovMats();
 
-  // An implementation of equation 18, which calculates postfilter masks that,
-  // when applied, minimize the mean-square error of our estimation of the
-  // desired signal. A sub-task is to calculate lambda, which is solved via
-  // equation 13.
+  // Calculates postfilter masks that minimize the mean squared error of our
+  // estimation of the desired signal.
   float CalculatePostfilterMask(const ComplexMatrixF& interf_cov_mat,
                                 float rpsiw,
                                 float ratio_rxiw_rxim,
-                                float rmxi_r,
-                                float mask_threshold);
+                                float rmxi_r);
 
   // Prevents the postfilter masks from degenerating too quickly (a cause of
   // musical noise).
-  void ApplyMaskSmoothing();
+  void ApplyMaskTimeSmoothing();
+  void ApplyMaskFrequencySmoothing();
 
   // The postfilter masks are unreliable at low frequencies. Calculates a better
   // mask by averaging mid-low frequency values.
@@ -97,57 +114,73 @@ class NonlinearBeamformer
   // both transforming and blocking the high-frequency signal.
   void ApplyHighFrequencyCorrection();
 
+  // Compute the means needed for the above frequency correction.
+  float MaskRangeMean(size_t start_bin, size_t end_bin);
+
   // Applies both sets of masks to |input| and store in |output|.
   void ApplyMasks(const complex_f* const* input, complex_f* const* output);
 
   void EstimateTargetPresence();
 
-  static const int kFftSize = 256;
-  static const int kNumFreqBins = kFftSize / 2 + 1;
+  static const size_t kFftSize = 256;
+  static const size_t kNumFreqBins = kFftSize / 2 + 1;
 
   // Deals with the fft transform and blocking.
-  int chunk_length_;
+  size_t chunk_length_;
   rtc::scoped_ptr<LappedTransform> lapped_transform_;
   float window_[kFftSize];
 
   // Parameters exposed to the user.
-  const int num_input_channels_;
+  const size_t num_input_channels_;
   int sample_rate_hz_;
 
   const std::vector<Point> array_geometry_;
+  // The normal direction of the array if it has one and it is in the xy-plane.
+  const rtc::Optional<Point> array_normal_;
+
+  // Minimum spacing between microphone pairs.
+  const float min_mic_spacing_;
 
   // Calculated based on user-input and constants in the .cc file.
-  int low_average_start_bin_;
-  int low_average_end_bin_;
-  int high_average_start_bin_;
-  int high_average_end_bin_;
+  size_t low_mean_start_bin_;
+  size_t low_mean_end_bin_;
+  size_t high_mean_start_bin_;
+  size_t high_mean_end_bin_;
 
-  // Old masks are saved for smoothing. Matrix of size 1 x |kNumFreqBins|.
-  float postfilter_mask_[kNumFreqBins];
+  // Quickly varying mask updated every block.
   float new_mask_[kNumFreqBins];
+  // Time smoothed mask.
+  float time_smooth_mask_[kNumFreqBins];
+  // Time and frequency smoothed mask.
+  float final_mask_[kNumFreqBins];
+
+  float target_angle_radians_;
+  // Angles of the interferer scenarios.
+  std::vector<float> interf_angles_radians_;
+  // The angle between the target and the interferer scenarios.
+  const float away_radians_;
 
   // Array of length |kNumFreqBins|, Matrix of size |1| x |num_channels_|.
   ComplexMatrixF delay_sum_masks_[kNumFreqBins];
   ComplexMatrixF normalized_delay_sum_masks_[kNumFreqBins];
 
-  // Array of length |kNumFreqBins|, Matrix of size |num_input_channels_| x
+  // Arrays of length |kNumFreqBins|, Matrix of size |num_input_channels_| x
   // |num_input_channels_|.
   ComplexMatrixF target_cov_mats_[kNumFreqBins];
-
+  ComplexMatrixF uniform_cov_mat_[kNumFreqBins];
   // Array of length |kNumFreqBins|, Matrix of size |num_input_channels_| x
-  // |num_input_channels_|.
-  ComplexMatrixF interf_cov_mats_[kNumFreqBins];
-  ComplexMatrixF reflected_interf_cov_mats_[kNumFreqBins];
+  // |num_input_channels_|. ScopedVector has a size equal to the number of
+  // interferer scenarios.
+  ScopedVector<ComplexMatrixF> interf_cov_mats_[kNumFreqBins];
 
   // Of length |kNumFreqBins|.
-  float mask_thresholds_[kNumFreqBins];
   float wave_numbers_[kNumFreqBins];
 
   // Preallocated for ProcessAudioBlock()
   // Of length |kNumFreqBins|.
   float rxiws_[kNumFreqBins];
-  float rpsiws_[kNumFreqBins];
-  float reflected_rpsiws_[kNumFreqBins];
+  // The vector has a size equal to the number of interferer scenarios.
+  std::vector<float> rpsiws_[kNumFreqBins];
 
   // The microphone normalization factor.
   ComplexMatrixF eig_m_;
@@ -159,9 +192,9 @@ class NonlinearBeamformer
   bool is_target_present_;
   // Number of blocks after which the data is considered interference if the
   // mask does not pass |kMaskSignalThreshold|.
-  int hold_target_blocks_;
+  size_t hold_target_blocks_;
   // Number of blocks since the last mask that passed |kMaskSignalThreshold|.
-  int interference_blocks_count_;
+  size_t interference_blocks_count_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
index 48d7c2b2ae..d187552692 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
@@ -8,75 +8,83 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <iostream>
 #include <vector>
 
 #include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
-#include "webrtc/modules/audio_processing/beamformer/pcm_utils.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
 
-DEFINE_int32(sample_rate,
-             48000,
-             "The sample rate of the input file. The output"
-             "file will be of the same sample rate.");
-DEFINE_int32(num_input_channels,
-             2,
-             "The number of channels in the input file.");
-DEFINE_double(mic_spacing,
-              0.05,
-              "The spacing between microphones on the chromebook which "
-              "recorded the input file.");
-DEFINE_string(input_file_path,
-              "input.wav",
-              "The absolute path to the input file.");
-DEFINE_string(output_file_path,
-              "beamformer_test_output.wav",
-              "The absolute path to the output file.");
+DEFINE_string(i, "", "The name of the input file to read from.");
+DEFINE_string(o, "out.wav", "Name of the output file to write to.");
+DEFINE_string(mic_positions, "",
+    "Space delimited cartesian coordinates of microphones in meters. "
+    "The coordinates of each point are contiguous. "
+    "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
 
-using webrtc::ChannelBuffer;
+namespace webrtc {
+namespace {
+
+const int kChunksPerSecond = 100;
+const int kChunkSizeMs = 1000 / kChunksPerSecond;
+
+const char kUsage[] =
+    "Command-line tool to run beamforming on WAV files. The signal is passed\n"
+    "in as a single band, unlike the audio processing interface which splits\n"
+    "signals into multiple bands.";
+
+}  // namespace
 
 int main(int argc, char* argv[]) {
+  google::SetUsageMessage(kUsage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
-  const float kChunkTimeMilliseconds = 10;
-  const int kChunkSize = FLAGS_sample_rate / (1000.f / kChunkTimeMilliseconds);
-  const int kInputSamplesPerChunk = kChunkSize * FLAGS_num_input_channels;
+  WavReader in_file(FLAGS_i);
+  WavWriter out_file(FLAGS_o, in_file.sample_rate(), 1);
 
-  ChannelBuffer<float> captured_audio_cb(kChunkSize, FLAGS_num_input_channels);
+  const size_t num_mics = in_file.num_channels();
+  const std::vector<Point> array_geometry =
+      ParseArrayGeometry(FLAGS_mic_positions, num_mics);
+  RTC_CHECK_EQ(array_geometry.size(), num_mics);
 
-  FILE* read_file = fopen(FLAGS_input_file_path.c_str(), "rb");
-  if (!read_file) {
-    std::cerr << "Input file '" << FLAGS_input_file_path << "' not found."
-              << std::endl;
-    return -1;
+  NonlinearBeamformer bf(array_geometry);
+  bf.Initialize(kChunkSizeMs, in_file.sample_rate());
+
+  printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
+         FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
+  printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
+         FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
+
+  ChannelBuffer<float> in_buf(
+      rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
+      in_file.num_channels());
+  ChannelBuffer<float> out_buf(
+      rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
+      out_file.num_channels());
+
+  std::vector<float> interleaved(in_buf.size());
+  while (in_file.ReadSamples(interleaved.size(),
+                             &interleaved[0]) == interleaved.size()) {
+    FloatS16ToFloat(&interleaved[0], interleaved.size(), &interleaved[0]);
+    Deinterleave(&interleaved[0], in_buf.num_frames(),
+                 in_buf.num_channels(), in_buf.channels());
+
+    bf.ProcessChunk(in_buf, &out_buf);
+
+    Interleave(out_buf.channels(), out_buf.num_frames(),
+               out_buf.num_channels(), &interleaved[0]);
+    FloatToFloatS16(&interleaved[0], interleaved.size(), &interleaved[0]);
+    out_file.WriteSamples(&interleaved[0], interleaved.size());
   }
 
-  // Skipping the .wav header. TODO: Add .wav header parsing.
-  fseek(read_file, 44, SEEK_SET);
-
-  FILE* write_file = fopen(FLAGS_output_file_path.c_str(), "wb");
-
-  std::vector<webrtc::Point> array_geometry;
-  for (int i = 0; i < FLAGS_num_input_channels; ++i) {
-    array_geometry.push_back(webrtc::Point(i * FLAGS_mic_spacing, 0.f, 0.f));
-  }
-  webrtc::NonlinearBeamformer bf(array_geometry);
-  bf.Initialize(kChunkTimeMilliseconds, FLAGS_sample_rate);
-  while (true) {
-    size_t samples_read = webrtc::PcmReadToFloat(read_file,
-                                                 kInputSamplesPerChunk,
-                                                 FLAGS_num_input_channels,
-                                                 captured_audio_cb.channels());
-
-    if (static_cast<int>(samples_read) != kInputSamplesPerChunk) {
-      break;
-    }
-
-    bf.ProcessChunk(captured_audio_cb, &captured_audio_cb);
-    webrtc::PcmWriteFromFloat(
-        write_file, kChunkSize, 1, captured_audio_cb.channels());
-  }
-  fclose(read_file);
-  fclose(write_file);
   return 0;
 }
+
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::main(argc, argv);
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_unittest.cc
new file mode 100644
index 0000000000..a38a49b1e1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_unittest.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#define _USE_MATH_DEFINES
+
+#include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
+
+#include <math.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const int kChunkSizeMs = 10;
+const int kSampleRateHz = 16000;
+
+SphericalPointf AzimuthToSphericalPoint(float azimuth_radians) {
+  return SphericalPointf(azimuth_radians, 0.f, 1.f);
+}
+
+void Verify(NonlinearBeamformer* bf, float target_azimuth_radians) {
+  EXPECT_TRUE(bf->IsInBeam(AzimuthToSphericalPoint(target_azimuth_radians)));
+  EXPECT_TRUE(bf->IsInBeam(AzimuthToSphericalPoint(
+      target_azimuth_radians - NonlinearBeamformer::kHalfBeamWidthRadians +
+      0.001f)));
+  EXPECT_TRUE(bf->IsInBeam(AzimuthToSphericalPoint(
+      target_azimuth_radians + NonlinearBeamformer::kHalfBeamWidthRadians -
+      0.001f)));
+  EXPECT_FALSE(bf->IsInBeam(AzimuthToSphericalPoint(
+      target_azimuth_radians - NonlinearBeamformer::kHalfBeamWidthRadians -
+      0.001f)));
+  EXPECT_FALSE(bf->IsInBeam(AzimuthToSphericalPoint(
+      target_azimuth_radians + NonlinearBeamformer::kHalfBeamWidthRadians +
+      0.001f)));
+}
+
+void AimAndVerify(NonlinearBeamformer* bf, float target_azimuth_radians) {
+  bf->AimAt(AzimuthToSphericalPoint(target_azimuth_radians));
+  Verify(bf, target_azimuth_radians);
+}
+
+}  // namespace
+
+TEST(NonlinearBeamformerTest, AimingModifiesBeam) {
+  std::vector<Point> array_geometry;
+  array_geometry.push_back(Point(-0.025f, 0.f, 0.f));
+  array_geometry.push_back(Point(0.025f, 0.f, 0.f));
+  NonlinearBeamformer bf(array_geometry);
+  bf.Initialize(kChunkSizeMs, kSampleRateHz);
+  // The default constructor parameter sets the target angle to PI / 2.
+  Verify(&bf, static_cast<float>(M_PI) / 2.f);
+  AimAndVerify(&bf, static_cast<float>(M_PI) / 3.f);
+  AimAndVerify(&bf, 3.f * static_cast<float>(M_PI) / 4.f);
+  AimAndVerify(&bf, static_cast<float>(M_PI) / 6.f);
+  AimAndVerify(&bf, static_cast<float>(M_PI));
+}
+
+TEST(NonlinearBeamformerTest, InterfAnglesTakeAmbiguityIntoAccount) {
+  {
+    // For linear arrays there is ambiguity.
+    std::vector<Point> array_geometry;
+    array_geometry.push_back(Point(-0.1f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.2f, 0.f, 0.f));
+    NonlinearBeamformer bf(array_geometry);
+    bf.Initialize(kChunkSizeMs, kSampleRateHz);
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
+                    bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
+                    bf.interf_angles_radians_[1]);
+    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(M_PI - bf.away_radians_ / 2.f,
+                    bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
+  }
+  {
+    // For planar arrays with normal in the xy-plane there is ambiguity.
+    std::vector<Point> array_geometry;
+    array_geometry.push_back(Point(-0.1f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.2f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.1f, 0.f, 0.2f));
+    array_geometry.push_back(Point(0.f, 0.f, -0.1f));
+    NonlinearBeamformer bf(array_geometry);
+    bf.Initialize(kChunkSizeMs, kSampleRateHz);
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
+                    bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
+                    bf.interf_angles_radians_[1]);
+    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(M_PI - bf.away_radians_ / 2.f,
+                    bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
+  }
+  {
+    // For planar arrays with normal not in the xy-plane there is no ambiguity.
+    std::vector<Point> array_geometry;
+    array_geometry.push_back(Point(0.f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.2f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.f, 0.1f, -0.2f));
+    NonlinearBeamformer bf(array_geometry);
+    bf.Initialize(kChunkSizeMs, kSampleRateHz);
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
+                    bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
+                    bf.interf_angles_radians_[1]);
+    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(-bf.away_radians_ / 2.f, bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
+  }
+  {
+    // For arrays which are not linear or planar there is no ambiguity.
+    std::vector<Point> array_geometry;
+    array_geometry.push_back(Point(0.f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.1f, 0.f, 0.f));
+    array_geometry.push_back(Point(0.f, 0.2f, 0.f));
+    array_geometry.push_back(Point(0.f, 0.f, 0.3f));
+    NonlinearBeamformer bf(array_geometry);
+    bf.Initialize(kChunkSizeMs, kSampleRateHz);
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
+                    bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
+                    bf.interf_angles_radians_[1]);
+    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
+    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
+    EXPECT_FLOAT_EQ(-bf.away_radians_ / 2.f, bf.interf_angles_radians_[0]);
+    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/pcm_utils.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/pcm_utils.cc
deleted file mode 100644
index 0999de34f8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/pcm_utils.cc
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_processing/beamformer/pcm_utils.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/common_audio/include/audio_util.h"
-#include "webrtc/common_audio/channel_buffer.h"
-
-namespace webrtc {
-
-size_t PcmRead(FILE* file,
-               size_t length,
-               int num_channels,
-               int16_t* const* buffer) {
-  CHECK_GE(num_channels, 1);
-
-  rtc::scoped_ptr<int16_t[]> interleaved_buffer(new int16_t[length]);
-  size_t elements_read = fread(interleaved_buffer.get(), sizeof(int16_t),
-                               length, file);
-  if (elements_read != length) {
-    // This is only an error if we haven't reached the end of the file.
-    CHECK_NE(0, feof(file));
-  }
-
-  Deinterleave(interleaved_buffer.get(),
-               static_cast<int>(elements_read) / num_channels,
-               num_channels,
-               buffer);
-  return elements_read;
-}
-
-size_t PcmReadToFloat(FILE* file,
-                      size_t length,
-                      int num_channels,
-                      float* const* buffer) {
-  CHECK_GE(num_channels, 1);
-
-  int num_frames = static_cast<int>(length) / num_channels;
-  rtc::scoped_ptr<ChannelBuffer<int16_t> > deinterleaved_buffer(
-      new ChannelBuffer<int16_t>(num_frames, num_channels));
-
-  size_t elements_read =
-      PcmRead(file, length, num_channels, deinterleaved_buffer->channels());
-
-  for (int i = 0; i < num_channels; ++i) {
-    S16ToFloat(deinterleaved_buffer->channels()[i], num_frames, buffer[i]);
-  }
-  return elements_read;
-}
-
-void PcmWrite(FILE* file,
-              size_t length,
-              int num_channels,
-              const int16_t* const* buffer) {
-  CHECK_GE(num_channels, 1);
-
-  rtc::scoped_ptr<int16_t[]> interleaved_buffer(new int16_t[length]);
-  Interleave(buffer,
-             static_cast<int>(length) / num_channels,
-             num_channels,
-             interleaved_buffer.get());
-  CHECK_EQ(length,
-           fwrite(interleaved_buffer.get(), sizeof(int16_t), length, file));
-}
-
-void PcmWriteFromFloat(FILE* file,
-                       size_t length,
-                       int num_channels,
-                       const float* const* buffer) {
-  CHECK_GE(num_channels, 1);
-
-  int num_frames = static_cast<int>(length) / num_channels;
-  rtc::scoped_ptr<ChannelBuffer<int16_t> > deinterleaved_buffer(
-      new ChannelBuffer<int16_t>(num_frames, num_channels));
-
-  for (int i = 0; i < num_channels; ++i) {
-    FloatToS16(buffer[i], num_frames, deinterleaved_buffer->channels()[i]);
-  }
-  PcmWrite(file, length, num_channels, deinterleaved_buffer->channels());
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/pcm_utils.h b/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/pcm_utils.h
deleted file mode 100644
index 3a6a3b9057..0000000000
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/beamformer/pcm_utils.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_PCM_UTILS_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_PCM_UTILS_H_
-
-#include <inttypes.h>
-#include <stdio.h>
-
-// Utilities for reading from and writing to multichannel pcm files.
-// Assumes a bit depth of 16 and little-endian. Note that in these functions,
-// length refers to the number of samples to read from/write to the file,
-// such that length / num_channels is the number of frames.
-namespace webrtc {
-
-// Reads audio from a pcm into a 2D array: buffer[channel_index][frame_index].
-// Returns the number of frames written. If this is less than |length|, it's
-// safe to assume the end-of-file was reached, as otherwise this will crash.
-// In PcmReadToFloat, the floats are within the range [-1, 1].
-size_t PcmRead(FILE* file,
-               size_t length,
-               int num_channels,
-               int16_t* const* buffer);
-size_t PcmReadToFloat(FILE* file,
-                      size_t length,
-                      int num_channels,
-                      float* const* buffer);
-
-// Writes to a pcm file. The resulting file contains the channels interleaved.
-// Crashes if the correct number of frames aren't written to the file. For
-// PcmWriteFromFloat, floats must be within the range [-1, 1].
-void PcmWrite(FILE* file,
-              size_t length,
-              int num_channels,
-              const int16_t* const* buffer);
-void PcmWriteFromFloat(FILE* file,
-                       size_t length,
-                       int num_channels,
-                       const float* const* buffer);
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_PCM_UTILS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/common.h b/media/webrtc/trunk/webrtc/modules/audio_processing/common.h
index ed8a0544c3..d4ddb92b50 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/common.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/common.h
@@ -17,7 +17,7 @@
 
 namespace webrtc {
 
-static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+static inline size_t ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
     case AudioProcessing::kMonoAndKeyboard:
@@ -27,7 +27,7 @@ static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
       return 2;
   }
   assert(false);
-  return -1;
+  return 0;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/debug.proto b/media/webrtc/trunk/webrtc/modules/audio_processing/debug.proto
index dce2f79209..227271298c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/debug.proto
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/debug.proto
@@ -2,6 +2,8 @@ syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 package webrtc.audioproc;
 
+// Contains the format of input/output/reverse audio. An Init message is added
+// when any of the fields are changed.
 message Init {
   optional int32 sample_rate = 1;
   optional int32 device_sample_rate = 2 [deprecated=true];
@@ -39,11 +41,41 @@ message Stream {
   repeated bytes output_channel = 8;
 }
 
+// Contains the configurations of various APM component. A Config message is
+// added when any of the fields are changed.
+message Config {
+  // Next field number 17.
+  // Acoustic echo canceler.
+  optional bool aec_enabled = 1;
+  optional bool aec_delay_agnostic_enabled = 2;
+  optional bool aec_drift_compensation_enabled = 3;
+  optional bool aec_extended_filter_enabled = 4;
+  optional int32 aec_suppression_level = 5;
+  // Mobile AEC.
+  optional bool aecm_enabled = 6;
+  optional bool aecm_comfort_noise_enabled = 7;
+  optional int32 aecm_routing_mode = 8;
+  // Automatic gain controller.
+  optional bool agc_enabled = 9;
+  optional int32 agc_mode = 10;
+  optional bool agc_limiter_enabled = 11;
+  optional bool noise_robust_agc_enabled = 12;
+  // High pass filter.
+  optional bool hpf_enabled = 13;
+  // Noise suppression.
+  optional bool ns_enabled = 14;
+  optional int32 ns_level = 15;
+  // Transient suppression.
+  optional bool transient_suppression_enabled = 16;
+}
+
 message Event {
   enum Type {
     INIT = 0;
     REVERSE_STREAM = 1;
     STREAM = 2;
+    CONFIG = 3;
+    UNKNOWN_EVENT = 4;
   }
 
   required Type type = 1;
@@ -51,4 +83,5 @@ message Event {
   optional Init init = 2;
   optional ReverseStream reverse_stream = 3;
   optional Stream stream = 4;
+  optional Config config = 5;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index c6a35f1b68..debc597c54 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -16,9 +16,8 @@
 extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -53,102 +52,155 @@ AudioProcessing::Error MapError(int err) {
       return AudioProcessing::kUnspecifiedError;
   }
 }
+
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
 }  // namespace
 
 EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
-                                           CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    drift_compensation_enabled_(false),
-    metrics_enabled_(false),
-    suppression_level_(kModerateSuppression),
-    stream_drift_samples_(0),
-    was_stream_drift_set_(false),
-    stream_has_echo_(false),
-    delay_logging_enabled_(false),
-    extended_filter_enabled_(false),
-    delay_agnostic_enabled_(false) {
+                                           rtc::CriticalSection* crit_render,
+                                           rtc::CriticalSection* crit_capture)
+    : ProcessingComponent(),
+      apm_(apm),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
+      drift_compensation_enabled_(false),
+      metrics_enabled_(false),
+      suppression_level_(kModerateSuppression),
+      stream_drift_samples_(0),
+      was_stream_drift_set_(false),
+      stream_has_echo_(false),
+      delay_logging_enabled_(false),
+      extended_filter_enabled_(false),
+      delay_agnostic_enabled_(false),
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
 }
 
 EchoCancellationImpl::~EchoCancellationImpl() {}
 
 int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+  rtc::CritScope cs_render(crit_render_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_reverse_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
+  render_queue_buffer_.clear();
+  for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAec_BufferFarend(
-          my_handle,
-          audio->split_bands_const_f(j)[kBand0To8kHz],
+      // Retrieve any error code produced by the buffering of the farend
+      // signal
+      err = WebRtcAec_GetBufferFarendError(
+          my_handle, audio->split_bands_const_f(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
+      if (err != AudioProcessing::kNoError) {
+        return MapError(err);  // TODO(ajm): warning possible?
       }
 
-      handle_index++;
+      // Buffer the samples in the render queue.
+      render_queue_buffer_.insert(render_queue_buffer_.end(),
+                                  audio->split_bands_const_f(j)[kBand0To8kHz],
+                                  (audio->split_bands_const_f(j)[kBand0To8kHz] +
+                                   audio->num_frames_per_band()));
     }
   }
 
-  return apm_->kNoError;
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AEC.
+void EchoCancellationImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs_capture(crit_capture_);
+  if (!is_component_enabled()) {
+    return;
+  }
+
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    size_t handle_index = 0;
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
+        capture_queue_buffer_.size() /
+        (apm_->num_output_channels() * apm_->num_reverse_channels());
+    for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+      for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
+        Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+        WebRtcAec_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
+                               num_frames_per_band);
+
+        buffer_index += num_frames_per_band;
+        handle_index++;
+      }
+    }
+  }
 }
 
 int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs_capture(crit_capture_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   if (drift_compensation_enabled_ && !was_stream_drift_set_) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
-  assert(audio->num_channels() == apm_->num_output_channels());
+  assert(audio->num_channels() == apm_->num_proc_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   stream_has_echo_ = false;
-  for (int i = 0; i < audio->num_channels(); i++) {
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+  for (size_t i = 0; i < audio->num_channels(); i++) {
+    for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
       Handle* my_handle = handle(handle_index);
-      err = WebRtcAec_Process(
-          my_handle,
-          audio->split_bands_const_f(i),
-          audio->num_bands(),
-          audio->split_bands_f(i),
-          audio->num_frames_per_band(),
-          apm_->stream_delay_ms(),
-          stream_drift_samples_);
+      err = WebRtcAec_Process(my_handle, audio->split_bands_const_f(i),
+                              audio->num_bands(), audio->split_bands_f(i),
+                              audio->num_frames_per_band(),
+                              apm_->stream_delay_ms(), stream_drift_samples_);
 
-      if (err != apm_->kNoError) {
-        err = GetHandleError(my_handle);
+      if (err != AudioProcessing::kNoError) {
+        err = MapError(err);
         // TODO(ajm): Figure out how to return warnings properly.
-        if (err != apm_->kBadStreamParameterWarning) {
+        if (err != AudioProcessing::kBadStreamParameterWarning) {
           return err;
         }
       }
 
       int status = 0;
       err = WebRtcAec_get_echo_status(my_handle, &status);
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);
+      if (err != AudioProcessing::kNoError) {
+        return MapError(err);
       }
 
       if (status == 1) {
@@ -160,77 +212,92 @@ int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   }
 
   was_stream_drift_set_ = false;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoCancellationImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  // Run in a single-threaded manner.
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   // Ensure AEC and AECM are not both enabled.
+  // The is_enabled call is safe from a deadlock perspective
+  // as both locks are already held in the correct order.
   if (enable && apm_->echo_control_mobile()->is_enabled()) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   return EnableComponent(enable);
 }
 
 bool EchoCancellationImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
+  {
+    if (MapSetting(level) == -1) {
+      return AudioProcessing::kBadParameterError;
+    }
+    rtc::CritScope cs(crit_capture_);
+    suppression_level_ = level;
   }
-
-  suppression_level_ = level;
   return Configure();
 }
 
 EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
     const {
+  rtc::CritScope cs(crit_capture_);
   return suppression_level_;
 }
 
 int EchoCancellationImpl::enable_drift_compensation(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  drift_compensation_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    drift_compensation_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_drift_compensation_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return drift_compensation_enabled_;
 }
 
 void EchoCancellationImpl::set_stream_drift_samples(int drift) {
+  rtc::CritScope cs(crit_capture_);
   was_stream_drift_set_ = true;
   stream_drift_samples_ = drift;
 }
 
 int EchoCancellationImpl::stream_drift_samples() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_drift_samples_;
 }
 
 int EchoCancellationImpl::enable_metrics(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  metrics_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    metrics_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::are_metrics_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return metrics_enabled_;
 }
 
 // TODO(ajm): we currently just use the metrics from the first AEC. Think more
 //            aboue the best way to extend this to multi-channel.
 int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (metrics == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
 
   if (!is_component_enabled() || !metrics_enabled_) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   AecMetrics my_metrics;
@@ -239,8 +306,8 @@ int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
 
   Handle* my_handle = static_cast<Handle*>(handle(0));
   int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
-  if (err != apm_->kNoError) {
-    return GetHandleError(my_handle);
+  if (err != AudioProcessing::kNoError) {
+    return MapError(err);
   }
 
   metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
@@ -263,62 +330,70 @@ int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
   metrics->a_nlp.maximum = my_metrics.aNlp.max;
   metrics->a_nlp.minimum = my_metrics.aNlp.min;
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 bool EchoCancellationImpl::stream_has_echo() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_has_echo_;
 }
 
 int EchoCancellationImpl::enable_delay_logging(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  delay_logging_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    delay_logging_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoCancellationImpl::is_delay_logging_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return delay_logging_enabled_;
 }
 
 bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return delay_agnostic_enabled_;
 }
 
 bool EchoCancellationImpl::is_extended_filter_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return extended_filter_enabled_;
 }
 
 // TODO(bjornv): How should we handle the multi-channel case?
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
+  rtc::CritScope cs(crit_capture_);
   float fraction_poor_delays = 0;
   return GetDelayMetrics(median, std, &fraction_poor_delays);
 }
 
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
                                           float* fraction_poor_delays) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (median == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
   if (std == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
 
   if (!is_component_enabled() || !delay_logging_enabled_) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   Handle* my_handle = static_cast<Handle*>(handle(0));
-  if (WebRtcAec_GetDelayMetrics(my_handle, median, std, fraction_poor_delays) !=
-      apm_->kNoError) {
-    return GetHandleError(my_handle);
+  const int err =
+      WebRtcAec_GetDelayMetrics(my_handle, median, std, fraction_poor_delays);
+  if (err != AudioProcessing::kNoError) {
+    return MapError(err);
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 struct AecCore* EchoCancellationImpl::aec_core() const {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (!is_component_enabled()) {
     return NULL;
   }
@@ -328,16 +403,51 @@ struct AecCore* EchoCancellationImpl::aec_core() const {
 
 int EchoCancellationImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
+  {
+    rtc::CritScope cs(crit_capture_);
+    if (err != AudioProcessing::kNoError || !is_component_enabled()) {
+      return err;
+    }
   }
 
-  return apm_->kNoError;
+  AllocateRenderQueue();
+
+  return AudioProcessing::kNoError;
+}
+
+void EchoCancellationImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size = std::max<size_t>(
+      static_cast<size_t>(1),
+      kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
+
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  // Reallocate the queue if the queue item size is too small to fit the
+  // data to put in the queue.
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+
+    std::vector<float> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<float>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
 }
 
 void EchoCancellationImpl::SetExtraOptions(const Config& config) {
-  extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
-  delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
+  {
+    rtc::CritScope cs(crit_capture_);
+    extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
+    delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
+  }
   Configure();
 }
 
@@ -351,23 +461,25 @@ void EchoCancellationImpl::DestroyHandle(void* handle) const {
 }
 
 int EchoCancellationImpl::InitializeHandle(void* handle) const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+
   assert(handle != NULL);
   // TODO(ajm): Drift compensation is disabled in practice. If restored, it
   // should be managed internally and not depend on the hardware sample rate.
   // For now, just hardcode a 48 kHz value.
   return WebRtcAec_Init(static_cast<Handle*>(handle),
-                       apm_->proc_sample_rate_hz(),
-                       48000);
+                        apm_->proc_sample_rate_hz(), 48000);
 }
 
 int EchoCancellationImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   assert(handle != NULL);
   AecConfig config;
   config.metricsMode = metrics_enabled_;
   config.nlpMode = MapSetting(suppression_level_);
   config.skewMode = drift_compensation_enabled_;
   config.delay_logging = delay_logging_enabled_;
-
   WebRtcAec_enable_extended_filter(
       WebRtcAec_aec_core(static_cast<Handle*>(handle)),
       extended_filter_enabled_ ? 1 : 0);
@@ -377,12 +489,13 @@ int EchoCancellationImpl::ConfigureHandle(void* handle) const {
   return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
 }
 
-int EchoCancellationImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
+size_t EchoCancellationImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+  return apm_->num_output_channels() * apm_->num_reverse_channels();
 }
 
 int EchoCancellationImpl::GetHandleError(void* handle) const {
+  // Not locked as it does not rely on anything in the state.
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.h
index 070dcabc5d..a40a267e32 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.h
@@ -11,19 +11,22 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
 class EchoCancellationImpl : public EchoCancellation,
                              public ProcessingComponent {
  public:
   EchoCancellationImpl(const AudioProcessing* apm,
-                       CriticalSectionWrapper* crit);
+                       rtc::CriticalSection* crit_render,
+                       rtc::CriticalSection* crit_capture);
   virtual ~EchoCancellationImpl();
 
   int ProcessRenderAudio(const AudioBuffer* audio);
@@ -38,10 +41,13 @@ class EchoCancellationImpl : public EchoCancellation,
   // ProcessingComponent implementation.
   int Initialize() override;
   void SetExtraOptions(const Config& config) override;
-
   bool is_delay_agnostic_enabled() const;
   bool is_extended_filter_enabled() const;
 
+  // Reads render side data that has been queued on the render call.
+  // Called holding the capture lock.
+  void ReadQueuedRenderData();
+
  private:
   // EchoCancellation implementation.
   int Enable(bool enable) override;
@@ -58,6 +64,7 @@ class EchoCancellationImpl : public EchoCancellation,
   int GetDelayMetrics(int* median,
                       int* std,
                       float* fraction_poor_delays) override;
+
   struct AecCore* aec_core() const override;
 
   // ProcessingComponent implementation.
@@ -65,20 +72,35 @@ class EchoCancellationImpl : public EchoCancellation,
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
+  // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  bool drift_compensation_enabled_;
-  bool metrics_enabled_;
-  SuppressionLevel suppression_level_;
-  int stream_drift_samples_;
-  bool was_stream_drift_set_;
-  bool stream_has_echo_;
-  bool delay_logging_enabled_;
-  bool extended_filter_enabled_;
-  bool delay_agnostic_enabled_;
+
+  rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection* const crit_capture_;
+
+  bool drift_compensation_enabled_ GUARDED_BY(crit_capture_);
+  bool metrics_enabled_ GUARDED_BY(crit_capture_);
+  SuppressionLevel suppression_level_ GUARDED_BY(crit_capture_);
+  int stream_drift_samples_ GUARDED_BY(crit_capture_);
+  bool was_stream_drift_set_ GUARDED_BY(crit_capture_);
+  bool stream_has_echo_ GUARDED_BY(crit_capture_);
+  bool delay_logging_enabled_ GUARDED_BY(crit_capture_);
+  bool extended_filter_enabled_ GUARDED_BY(crit_capture_);
+  bool delay_agnostic_enabled_ GUARDED_BY(crit_capture_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+  std::vector<float> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<float> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  rtc::scoped_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
+      render_signal_queue_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
index aac9a1e705..7f152bf942 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
@@ -14,11 +14,10 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
-TEST(EchoCancellationInternalTest, DelayCorrection) {
+TEST(EchoCancellationInternalTest, ExtendedFilter) {
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create());
   EXPECT_TRUE(ap->echo_cancellation()->aec_core() == NULL);
 
@@ -28,27 +27,27 @@ TEST(EchoCancellationInternalTest, DelayCorrection) {
   AecCore* aec_core = ap->echo_cancellation()->aec_core();
   ASSERT_TRUE(aec_core != NULL);
   // Disabled by default.
-  EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
+  EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(aec_core));
 
   Config config;
-  config.Set<DelayCorrection>(new DelayCorrection(true));
+  config.Set<ExtendedFilter>(new ExtendedFilter(true));
   ap->SetExtraOptions(config);
-  EXPECT_EQ(1, WebRtcAec_delay_correction_enabled(aec_core));
+  EXPECT_EQ(1, WebRtcAec_extended_filter_enabled(aec_core));
 
   // Retains setting after initialization.
   EXPECT_EQ(ap->kNoError, ap->Initialize());
-  EXPECT_EQ(1, WebRtcAec_delay_correction_enabled(aec_core));
+  EXPECT_EQ(1, WebRtcAec_extended_filter_enabled(aec_core));
 
-  config.Set<DelayCorrection>(new DelayCorrection(false));
+  config.Set<ExtendedFilter>(new ExtendedFilter(false));
   ap->SetExtraOptions(config);
-  EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
+  EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(aec_core));
 
   // Retains setting after initialization.
   EXPECT_EQ(ap->kNoError, ap->Initialize());
-  EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
+  EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(aec_core));
 }
 
-TEST(EchoCancellationInternalTest, ReportedDelay) {
+TEST(EchoCancellationInternalTest, DelayAgnostic) {
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create());
   EXPECT_TRUE(ap->echo_cancellation()->aec_core() == NULL);
 
@@ -58,24 +57,24 @@ TEST(EchoCancellationInternalTest, ReportedDelay) {
   AecCore* aec_core = ap->echo_cancellation()->aec_core();
   ASSERT_TRUE(aec_core != NULL);
   // Enabled by default.
-  EXPECT_EQ(1, WebRtcAec_reported_delay_enabled(aec_core));
+  EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(aec_core));
 
   Config config;
-  config.Set<ReportedDelay>(new ReportedDelay(false));
+  config.Set<DelayAgnostic>(new DelayAgnostic(true));
   ap->SetExtraOptions(config);
-  EXPECT_EQ(0, WebRtcAec_reported_delay_enabled(aec_core));
+  EXPECT_EQ(1, WebRtcAec_delay_agnostic_enabled(aec_core));
 
   // Retains setting after initialization.
   EXPECT_EQ(ap->kNoError, ap->Initialize());
-  EXPECT_EQ(0, WebRtcAec_reported_delay_enabled(aec_core));
+  EXPECT_EQ(1, WebRtcAec_delay_agnostic_enabled(aec_core));
 
-  config.Set<ReportedDelay>(new ReportedDelay(true));
+  config.Set<DelayAgnostic>(new DelayAgnostic(false));
   ap->SetExtraOptions(config);
-  EXPECT_EQ(1, WebRtcAec_reported_delay_enabled(aec_core));
+  EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(aec_core));
 
   // Retains setting after initialization.
   EXPECT_EQ(ap->kNoError, ap->Initialize());
-  EXPECT_EQ(1, WebRtcAec_reported_delay_enabled(aec_core));
+  EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(aec_core));
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index d0161e0b46..0264dd3852 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -13,10 +13,9 @@
 #include <assert.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -40,6 +39,12 @@ int16_t MapSetting(EchoControlMobile::RoutingMode mode) {
   return -1;
 }
 
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
 }  // namespace
 
 size_t EchoControlMobile::echo_path_size_bytes() {
@@ -47,13 +52,20 @@ size_t EchoControlMobile::echo_path_size_bytes() {
 }
 
 EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessing* apm,
-                                             CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    routing_mode_(kSpeakerphone),
-    comfort_noise_enabled_(true),
-    external_echo_path_(NULL) {}
+                                             rtc::CriticalSection* crit_render,
+                                             rtc::CriticalSection* crit_capture)
+    : ProcessingComponent(),
+      apm_(apm),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
+      routing_mode_(kSpeakerphone),
+      comfort_noise_enabled_(true),
+      external_echo_path_(NULL),
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
+}
 
 EchoControlMobileImpl::~EchoControlMobileImpl() {
     if (external_echo_path_ != NULL) {
@@ -63,53 +75,98 @@ EchoControlMobileImpl::~EchoControlMobileImpl() {
 }
 
 int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
+  rtc::CritScope cs_render(crit_render_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_reverse_channels());
 
-  int err = apm_->kNoError;
-
+  int err = AudioProcessing::kNoError;
   // The ordering convention must be followed to pass to the correct AECM.
   size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
+  render_queue_buffer_.clear();
+  for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAecm_BufferFarend(
-          my_handle,
-          audio->split_bands_const(j)[kBand0To8kHz],
+      err = WebRtcAecm_GetBufferFarendError(
+          my_handle, audio->split_bands_const(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
+      if (err != AudioProcessing::kNoError)
+        return AudioProcessing::kUnspecifiedError;
+
+      // Buffer the samples in the render queue.
+      render_queue_buffer_.insert(render_queue_buffer_.end(),
+                                  audio->split_bands_const(j)[kBand0To8kHz],
+                                  (audio->split_bands_const(j)[kBand0To8kHz] +
+                                   audio->num_frames_per_band()));
 
       handle_index++;
     }
   }
 
-  return apm_->kNoError;
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AEC.
+void EchoControlMobileImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs_capture(crit_capture_);
+
+  if (!is_component_enabled()) {
+    return;
+  }
+
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    size_t handle_index = 0;
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
+        capture_queue_buffer_.size() /
+        (apm_->num_output_channels() * apm_->num_reverse_channels());
+    for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+      for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
+        Handle* my_handle = static_cast<Handle*>(handle(handle_index));
+        WebRtcAecm_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
+                                num_frames_per_band);
+
+        buffer_index += num_frames_per_band;
+        handle_index++;
+      }
+    }
+  }
 }
 
 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs_capture(crit_capture_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_output_channels());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AECM.
   size_t handle_index = 0;
-  for (int i = 0; i < audio->num_channels(); i++) {
+  for (size_t i = 0; i < audio->num_channels(); i++) {
     // TODO(ajm): improve how this works, possibly inside AECM.
     //            This is kind of hacked up.
     const int16_t* noisy = audio->low_pass_reference(i);
@@ -118,7 +175,7 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
       noisy = clean;
       clean = NULL;
     }
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+    for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
       err = WebRtcAecm_Process(
           my_handle,
@@ -128,109 +185,158 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
           audio->num_frames_per_band(),
           apm_->stream_delay_ms());
 
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
+      if (err != AudioProcessing::kNoError)
+        return AudioProcessing::kUnspecifiedError;
 
       handle_index++;
     }
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoControlMobileImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
   // Ensure AEC and AECM are not both enabled.
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+  // The is_enabled call is safe from a deadlock perspective
+  // as both locks are allready held in the correct order.
   if (enable && apm_->echo_cancellation()->is_enabled()) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   return EnableComponent(enable);
 }
 
 bool EchoControlMobileImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
-  CriticalSectionScoped crit_scoped(crit_);
   if (MapSetting(mode) == -1) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
-  routing_mode_ = mode;
+  {
+    rtc::CritScope cs(crit_capture_);
+    routing_mode_ = mode;
+  }
   return Configure();
 }
 
 EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
     const {
+  rtc::CritScope cs(crit_capture_);
   return routing_mode_;
 }
 
 int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  comfort_noise_enabled_ = enable;
+  {
+    rtc::CritScope cs(crit_capture_);
+    comfort_noise_enabled_ = enable;
+  }
   return Configure();
 }
 
 bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return comfort_noise_enabled_;
 }
 
 int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
                                        size_t size_bytes) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (echo_path == NULL) {
-    return apm_->kNullPointerError;
-  }
-  if (size_bytes != echo_path_size_bytes()) {
-    // Size mismatch
-    return apm_->kBadParameterError;
-  }
+  {
+    rtc::CritScope cs_render(crit_render_);
+    rtc::CritScope cs_capture(crit_capture_);
+    if (echo_path == NULL) {
+      return AudioProcessing::kNullPointerError;
+    }
+    if (size_bytes != echo_path_size_bytes()) {
+      // Size mismatch
+      return AudioProcessing::kBadParameterError;
+    }
 
-  if (external_echo_path_ == NULL) {
-    external_echo_path_ = new unsigned char[size_bytes];
+    if (external_echo_path_ == NULL) {
+      external_echo_path_ = new unsigned char[size_bytes];
+    }
+    memcpy(external_echo_path_, echo_path, size_bytes);
   }
-  memcpy(external_echo_path_, echo_path, size_bytes);
 
   return Initialize();
 }
 
 int EchoControlMobileImpl::GetEchoPath(void* echo_path,
                                        size_t size_bytes) const {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (echo_path == NULL) {
-    return apm_->kNullPointerError;
+    return AudioProcessing::kNullPointerError;
   }
   if (size_bytes != echo_path_size_bytes()) {
     // Size mismatch
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
   if (!is_component_enabled()) {
-    return apm_->kNotEnabledError;
+    return AudioProcessing::kNotEnabledError;
   }
 
   // Get the echo path from the first channel
   Handle* my_handle = static_cast<Handle*>(handle(0));
-  if (WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes) != 0) {
-      return GetHandleError(my_handle);
-  }
+  int32_t err = WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes);
+  if (err != 0)
+    return AudioProcessing::kUnspecifiedError;
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoControlMobileImpl::Initialize() {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+  {
+    rtc::CritScope cs_capture(crit_capture_);
+    if (!is_component_enabled()) {
+      return AudioProcessing::kNoError;
+    }
   }
 
-  if (apm_->proc_sample_rate_hz() > apm_->kSampleRate16kHz) {
+  if (apm_->proc_sample_rate_hz() > AudioProcessing::kSampleRate16kHz) {
     LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates";
-    return apm_->kBadSampleRateError;
+    return AudioProcessing::kBadSampleRateError;
   }
 
-  return ProcessingComponent::Initialize();
+  int err = ProcessingComponent::Initialize();
+  if (err != AudioProcessing::kNoError) {
+    return err;
+  }
+
+  AllocateRenderQueue();
+
+  return AudioProcessing::kNoError;
+}
+
+void EchoControlMobileImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size = std::max<size_t>(
+      static_cast<size_t>(1),
+      kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
+
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  // Reallocate the queue if the queue item size is too small to fit the
+  // data to put in the queue.
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+
+    std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
 }
 
 void* EchoControlMobileImpl::CreateHandle() const {
@@ -238,10 +344,14 @@ void* EchoControlMobileImpl::CreateHandle() const {
 }
 
 void EchoControlMobileImpl::DestroyHandle(void* handle) const {
+  // This method is only called in a non-concurrent manner during APM
+  // destruction.
   WebRtcAecm_Free(static_cast<Handle*>(handle));
 }
 
 int EchoControlMobileImpl::InitializeHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   assert(handle != NULL);
   Handle* my_handle = static_cast<Handle*>(handle);
   if (WebRtcAecm_Init(my_handle, apm_->proc_sample_rate_hz()) != 0) {
@@ -255,10 +365,12 @@ int EchoControlMobileImpl::InitializeHandle(void* handle) const {
     }
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   AecmConfig config;
   config.cngMode = comfort_noise_enabled_;
   config.echoMode = MapSetting(routing_mode_);
@@ -266,12 +378,13 @@ int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
   return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
 }
 
-int EchoControlMobileImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
+size_t EchoControlMobileImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+  return apm_->num_output_channels() * apm_->num_reverse_channels();
 }
 
 int EchoControlMobileImpl::GetHandleError(void* handle) const {
+  // Not locked as it does not rely on anything in the state.
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.h
index f399f480b2..4d6529d3ac 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/echo_control_mobile_impl.h
@@ -11,19 +11,23 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
 class EchoControlMobileImpl : public EchoControlMobile,
                               public ProcessingComponent {
  public:
   EchoControlMobileImpl(const AudioProcessing* apm,
-                        CriticalSectionWrapper* crit);
+                        rtc::CriticalSection* crit_render,
+                        rtc::CriticalSection* crit_capture);
+
   virtual ~EchoControlMobileImpl();
 
   int ProcessRenderAudio(const AudioBuffer* audio);
@@ -31,33 +35,55 @@ class EchoControlMobileImpl : public EchoControlMobile,
 
   // EchoControlMobile implementation.
   bool is_enabled() const override;
+  RoutingMode routing_mode() const override;
+  bool is_comfort_noise_enabled() const override;
 
   // ProcessingComponent implementation.
   int Initialize() override;
 
+  // Reads render side data that has been queued on the render call.
+  void ReadQueuedRenderData();
+
  private:
   // EchoControlMobile implementation.
   int Enable(bool enable) override;
   int set_routing_mode(RoutingMode mode) override;
-  RoutingMode routing_mode() const override;
   int enable_comfort_noise(bool enable) override;
-  bool is_comfort_noise_enabled() const override;
   int SetEchoPath(const void* echo_path, size_t size_bytes) override;
   int GetEchoPath(void* echo_path, size_t size_bytes) const override;
 
   // ProcessingComponent implementation.
+  // Called holding both the render and capture locks.
   void* CreateHandle() const override;
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
+  // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  RoutingMode routing_mode_;
-  bool comfort_noise_enabled_;
-  unsigned char* external_echo_path_;
+
+  rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection* const crit_capture_;
+
+  RoutingMode routing_mode_ GUARDED_BY(crit_capture_);
+  bool comfort_noise_enabled_ GUARDED_BY(crit_capture_);
+  unsigned char* external_echo_path_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+
+  std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  rtc::scoped_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      render_signal_queue_;
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.cc
index 6211c4985c..04a6c7ba29 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -14,7 +14,6 @@
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -33,74 +32,127 @@ int16_t MapSetting(GainControl::Mode mode) {
   assert(false);
   return -1;
 }
+
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
+
 }  // namespace
 
 GainControlImpl::GainControlImpl(const AudioProcessing* apm,
-                                 CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    mode_(kAdaptiveAnalog),
-    minimum_capture_level_(0),
-    maximum_capture_level_(255),
-    limiter_enabled_(true),
-    target_level_dbfs_(3),
-    compression_gain_db_(9),
-    analog_capture_level_(0),
-    was_analog_level_set_(false),
-    stream_is_saturated_(false) {}
+                                 rtc::CriticalSection* crit_render,
+                                 rtc::CriticalSection* crit_capture)
+    : ProcessingComponent(),
+      apm_(apm),
+      crit_render_(crit_render),
+      crit_capture_(crit_capture),
+      mode_(kAdaptiveAnalog),
+      minimum_capture_level_(0),
+      maximum_capture_level_(255),
+      limiter_enabled_(true),
+      target_level_dbfs_(3),
+      compression_gain_db_(9),
+      analog_capture_level_(0),
+      was_analog_level_set_(false),
+      stream_is_saturated_(false),
+      render_queue_element_max_size_(0) {
+  RTC_DCHECK(apm);
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
+}
 
 GainControlImpl::~GainControlImpl() {}
 
 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_render_);
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
 
-  for (int i = 0; i < num_handles(); i++) {
+  render_queue_buffer_.resize(0);
+  for (size_t i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
-    int err = WebRtcAgc_AddFarend(
-        my_handle,
-        audio->mixed_low_pass_data(),
-        static_cast<int16_t>(audio->num_frames_per_band()));
+    int err =
+        WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());
 
-    if (err != apm_->kNoError) {
+    if (err != AudioProcessing::kNoError)
       return GetHandleError(my_handle);
-    }
+
+    // Buffer the samples in the render queue.
+    render_queue_buffer_.insert(
+        render_queue_buffer_.end(), audio->mixed_low_pass_data(),
+        (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
   }
 
-  return apm_->kNoError;
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AGC.
+void GainControlImpl::ReadQueuedRenderData() {
+  rtc::CritScope cs(crit_capture_);
+
+  if (!is_component_enabled()) {
+    return;
+  }
+
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
+    size_t buffer_index = 0;
+    const size_t num_frames_per_band =
+        capture_queue_buffer_.size() / num_handles();
+    for (size_t i = 0; i < num_handles(); i++) {
+      Handle* my_handle = static_cast<Handle*>(handle(i));
+      WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
+                          num_frames_per_band);
+
+      buffer_index += num_frames_per_band;
+    }
+  }
 }
 
 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_capture_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
-  int err = apm_->kNoError;
+  int err = AudioProcessing::kNoError;
 
   if (mode_ == kAdaptiveAnalog) {
     capture_levels_.assign(num_handles(), analog_capture_level_);
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       err = WebRtcAgc_AddMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
-          static_cast<int16_t>(audio->num_frames_per_band()));
+          audio->num_frames_per_band());
 
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }
     }
   } else if (mode_ == kAdaptiveDigital) {
 
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       int32_t capture_level_out = 0;
 
@@ -108,52 +160,56 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
-          static_cast<int16_t>(audio->num_frames_per_band()),
+          audio->num_frames_per_band(),
           analog_capture_level_,
           &capture_level_out);
 
       capture_levels_[i] = capture_level_out;
 
-      if (err != apm_->kNoError) {
+      if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }
 
     }
   }
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_capture_);
+
   if (!is_component_enabled()) {
-    return apm_->kNoError;
+    return AudioProcessing::kNoError;
   }
 
   if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
-    return apm_->kStreamParameterNotSetError;
+    return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
   stream_is_saturated_ = false;
-  for (int i = 0; i < num_handles(); i++) {
+  for (size_t i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int32_t capture_level_out = 0;
     uint8_t saturation_warning = 0;
 
+    // The call to stream_has_echo() is ok from a deadlock perspective
+    // as the capture lock is allready held.
     int err = WebRtcAgc_Process(
         my_handle,
         audio->split_bands_const(i),
         audio->num_bands(),
-        static_cast<int16_t>(audio->num_frames_per_band()),
+        audio->num_frames_per_band(),
         audio->split_bands(i),
         capture_levels_[i],
         &capture_level_out,
         apm_->echo_cancellation()->stream_has_echo(),
         &saturation_warning);
 
-    if (err != apm_->kNoError) {
+    if (err != AudioProcessing::kNoError) {
       return GetHandleError(my_handle);
     }
 
@@ -166,7 +222,7 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   if (mode_ == kAdaptiveAnalog) {
     // Take the analog level to be the average across the handles.
     analog_capture_level_ = 0;
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       analog_capture_level_ += capture_levels_[i];
     }
 
@@ -174,21 +230,24 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   }
 
   was_analog_level_set_ = false;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
+  rtc::CritScope cs(crit_capture_);
+
   was_analog_level_set_ = true;
   if (level < minimum_capture_level_ || level > maximum_capture_level_) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
   analog_capture_level_ = level;
 
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 int GainControlImpl::stream_analog_level() {
+  rtc::CritScope cs(crit_capture_);
   // TODO(ajm): enable this assertion?
   //assert(mode_ == kAdaptiveAnalog);
 
@@ -196,18 +255,21 @@ int GainControlImpl::stream_analog_level() {
 }
 
 int GainControlImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   return EnableComponent(enable);
 }
 
 bool GainControlImpl::is_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }
 
 int GainControlImpl::set_mode(Mode mode) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   if (MapSetting(mode) == -1) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   mode_ = mode;
@@ -215,22 +277,23 @@ int GainControlImpl::set_mode(Mode mode) {
 }
 
 GainControl::Mode GainControlImpl::mode() const {
+  rtc::CritScope cs(crit_capture_);
   return mode_;
 }
 
 int GainControlImpl::set_analog_level_limits(int minimum,
                                              int maximum) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (minimum < 0) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   if (maximum > 65535) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   if (maximum < minimum) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   minimum_capture_level_ = minimum;
@@ -240,21 +303,24 @@ int GainControlImpl::set_analog_level_limits(int minimum,
 }
 
 int GainControlImpl::analog_level_minimum() const {
+  rtc::CritScope cs(crit_capture_);
   return minimum_capture_level_;
 }
 
 int GainControlImpl::analog_level_maximum() const {
+  rtc::CritScope cs(crit_capture_);
   return maximum_capture_level_;
 }
 
 bool GainControlImpl::stream_is_saturated() const {
+  rtc::CritScope cs(crit_capture_);
   return stream_is_saturated_;
 }
 
 int GainControlImpl::set_target_level_dbfs(int level) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (level > 31 || level < 0) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   target_level_dbfs_ = level;
@@ -262,13 +328,14 @@ int GainControlImpl::set_target_level_dbfs(int level) {
 }
 
 int GainControlImpl::target_level_dbfs() const {
+  rtc::CritScope cs(crit_capture_);
   return target_level_dbfs_;
 }
 
 int GainControlImpl::set_compression_gain_db(int gain) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   if (gain < 0 || gain > 90) {
-    return apm_->kBadParameterError;
+    return AudioProcessing::kBadParameterError;
   }
 
   compression_gain_db_ = gain;
@@ -276,38 +343,63 @@ int GainControlImpl::set_compression_gain_db(int gain) {
 }
 
 int GainControlImpl::compression_gain_db() const {
+  rtc::CritScope cs(crit_capture_);
   return compression_gain_db_;
 }
 
 int GainControlImpl::enable_limiter(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
+  rtc::CritScope cs(crit_capture_);
   limiter_enabled_ = enable;
   return Configure();
 }
 
 bool GainControlImpl::is_limiter_enabled() const {
+  rtc::CritScope cs(crit_capture_);
   return limiter_enabled_;
 }
 
 int GainControlImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
+  if (err != AudioProcessing::kNoError || !is_component_enabled()) {
     return err;
   }
 
-  capture_levels_.assign(num_handles(), analog_capture_level_);
-  return apm_->kNoError;
+  AllocateRenderQueue();
+
+  rtc::CritScope cs_capture(crit_capture_);
+  const int n = num_handles();
+  RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;
+
+  capture_levels_.assign(n, analog_capture_level_);
+  return AudioProcessing::kNoError;
+}
+
+void GainControlImpl::AllocateRenderQueue() {
+  const size_t new_render_queue_element_max_size =
+      std::max<size_t>(static_cast<size_t>(1),
+                       kMaxAllowedValuesOfSamplesPerFrame * num_handles());
+
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
+    std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
+
+    render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
+  } else {
+    render_signal_queue_->Clear();
+  }
 }
 
 void* GainControlImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAgc_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
+  return WebRtcAgc_Create();
 }
 
 void GainControlImpl::DestroyHandle(void* handle) const {
@@ -315,6 +407,9 @@ void GainControlImpl::DestroyHandle(void* handle) const {
 }
 
 int GainControlImpl::InitializeHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
   return WebRtcAgc_Init(static_cast<Handle*>(handle),
                           minimum_capture_level_,
                           maximum_capture_level_,
@@ -323,6 +418,8 @@ int GainControlImpl::InitializeHandle(void* handle) const {
 }
 
 int GainControlImpl::ConfigureHandle(void* handle) const {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
   WebRtcAgcConfig config;
   // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
   //            change the interface.
@@ -336,14 +433,15 @@ int GainControlImpl::ConfigureHandle(void* handle) const {
   return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
 }
 
-int GainControlImpl::num_handles_required() const {
-  return apm_->num_output_channels();
+size_t GainControlImpl::num_handles_required() const {
+  // Not locked as it only relies on APM public API which is threadsafe.
+  return apm_->num_proc_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
-  return apm_->kUnspecifiedError;
+  return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.h
index d64894367c..72789ba5e1 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/gain_control_impl.h
@@ -13,19 +13,23 @@
 
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
 class GainControlImpl : public GainControl,
                         public ProcessingComponent {
  public:
   GainControlImpl(const AudioProcessing* apm,
-                  CriticalSectionWrapper* crit);
+                  rtc::CriticalSection* crit_render,
+                  rtc::CriticalSection* crit_capture);
   virtual ~GainControlImpl();
 
   int ProcessRenderAudio(AudioBuffer* audio);
@@ -38,19 +42,22 @@ class GainControlImpl : public GainControl,
   // GainControl implementation.
   bool is_enabled() const override;
   int stream_analog_level() override;
+  bool is_limiter_enabled() const override;
+  Mode mode() const override;
+
+  // Reads render side data that has been queued on the render call.
+  void ReadQueuedRenderData();
 
  private:
   // GainControl implementation.
   int Enable(bool enable) override;
   int set_stream_analog_level(int level) override;
   int set_mode(Mode mode) override;
-  Mode mode() const override;
   int set_target_level_dbfs(int level) override;
   int target_level_dbfs() const override;
   int set_compression_gain_db(int gain) override;
   int compression_gain_db() const override;
   int enable_limiter(bool enable) override;
-  bool is_limiter_enabled() const override;
   int set_analog_level_limits(int minimum, int maximum) override;
   int analog_level_minimum() const override;
   int analog_level_maximum() const override;
@@ -61,21 +68,37 @@ class GainControlImpl : public GainControl,
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
+  size_t num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
+  // Not guarded as its public API is thread safe.
   const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  Mode mode_;
-  int minimum_capture_level_;
-  int maximum_capture_level_;
-  bool limiter_enabled_;
-  int target_level_dbfs_;
-  int compression_gain_db_;
-  std::vector<int> capture_levels_;
-  int analog_capture_level_;
-  bool was_analog_level_set_;
-  bool stream_is_saturated_;
+
+  rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
+  rtc::CriticalSection* const crit_capture_;
+
+  Mode mode_ GUARDED_BY(crit_capture_);
+  int minimum_capture_level_ GUARDED_BY(crit_capture_);
+  int maximum_capture_level_ GUARDED_BY(crit_capture_);
+  bool limiter_enabled_ GUARDED_BY(crit_capture_);
+  int target_level_dbfs_ GUARDED_BY(crit_capture_);
+  int compression_gain_db_ GUARDED_BY(crit_capture_);
+  std::vector<int> capture_levels_ GUARDED_BY(crit_capture_);
+  int analog_capture_level_ GUARDED_BY(crit_capture_);
+  bool was_analog_level_set_ GUARDED_BY(crit_capture_);
+  bool stream_is_saturated_ GUARDED_BY(crit_capture_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_);
+  std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  rtc::scoped_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      render_signal_queue_;
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.cc
index 588ba41415..375d58febb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.cc
@@ -10,159 +10,125 @@
 
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
 
-#include <assert.h>
-
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/typedefs.h"
-
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 namespace {
-const int16_t kFilterCoefficients8kHz[5] =
-    {3798, -7596, 3798, 7807, -3733};
-
-const int16_t kFilterCoefficients[5] =
-    {4012, -8024, 4012, 8002, -3913};
-
-struct FilterState {
-  int16_t y[4];
-  int16_t x[2];
-  const int16_t* ba;
-};
-
-int InitializeFilter(FilterState* hpf, int sample_rate_hz) {
-  assert(hpf != NULL);
-
-  if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) {
-    hpf->ba = kFilterCoefficients8kHz;
-  } else {
-    hpf->ba = kFilterCoefficients;
-  }
-
-  WebRtcSpl_MemSetW16(hpf->x, 0, 2);
-  WebRtcSpl_MemSetW16(hpf->y, 0, 4);
-
-  return AudioProcessing::kNoError;
-}
-
-int Filter(FilterState* hpf, int16_t* data, int length) {
-  assert(hpf != NULL);
-
-  int32_t tmp_int32 = 0;
-  int16_t* y = hpf->y;
-  int16_t* x = hpf->x;
-  const int16_t* ba = hpf->ba;
-
-  for (int i = 0; i < length; i++) {
-    //  y[i] = b[0] * x[i] + b[1] * x[i-1] + b[2] * x[i-2]
-    //         + -a[1] * y[i-1] + -a[2] * y[i-2];
-
-    tmp_int32 = y[1] * ba[3];  // -a[1] * y[i-1] (low part)
-    tmp_int32 += y[3] * ba[4];  // -a[2] * y[i-2] (low part)
-    tmp_int32 = (tmp_int32 >> 15);
-    tmp_int32 += y[0] * ba[3];  // -a[1] * y[i-1] (high part)
-    tmp_int32 += y[2] * ba[4];  // -a[2] * y[i-2] (high part)
-    tmp_int32 = (tmp_int32 << 1);
-
-    tmp_int32 += data[i] * ba[0];  // b[0]*x[0]
-    tmp_int32 += x[0] * ba[1];  // b[1]*x[i-1]
-    tmp_int32 += x[1] * ba[2];  // b[2]*x[i-2]
-
-    // Update state (input part)
-    x[1] = x[0];
-    x[0] = data[i];
-
-    // Update state (filtered part)
-    y[2] = y[0];
-    y[3] = y[1];
-    y[0] = static_cast<int16_t>(tmp_int32 >> 13);
-    y[1] = static_cast<int16_t>(
-        (tmp_int32 - (static_cast<int32_t>(y[0]) << 13)) << 2);
-
-    // Rounding in Q12, i.e. add 2^11
-    tmp_int32 += 2048;
-
-    // Saturate (to 2^27) so that the HP filtered signal does not overflow
-    tmp_int32 = WEBRTC_SPL_SAT(static_cast<int32_t>(134217727),
-                               tmp_int32,
-                               static_cast<int32_t>(-134217728));
-
-    // Convert back to Q0 and use rounding.
-    data[i] = (int16_t)(tmp_int32 >> 12);
-  }
-
-  return AudioProcessing::kNoError;
-}
+const int16_t kFilterCoefficients8kHz[5] = {3798, -7596, 3798, 7807, -3733};
+const int16_t kFilterCoefficients[5] = {4012, -8024, 4012, 8002, -3913};
 }  // namespace
 
-typedef FilterState Handle;
-
-HighPassFilterImpl::HighPassFilterImpl(const AudioProcessing* apm,
-                                       CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit) {}
-
-HighPassFilterImpl::~HighPassFilterImpl() {}
-
-int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  int err = apm_->kNoError;
-
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+class HighPassFilterImpl::BiquadFilter {
+ public:
+  explicit BiquadFilter(int sample_rate_hz) :
+      ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ?
+          kFilterCoefficients8kHz : kFilterCoefficients)
+  {
+    Reset();
   }
 
-  assert(audio->num_frames_per_band() <= 160);
+  void Reset() {
+    std::memset(x_, 0, sizeof(x_));
+    std::memset(y_, 0, sizeof(y_));
+  }
 
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    err = Filter(my_handle,
-                 audio->split_bands(i)[kBand0To8kHz],
-                 audio->num_frames_per_band());
+  void Process(int16_t* data, size_t length) {
+    const int16_t* const ba = ba_;
+    int16_t* x = x_;
+    int16_t* y = y_;
+    int32_t tmp_int32 = 0;
 
-    if (err != apm_->kNoError) {
-      return GetHandleError(my_handle);
+    for (size_t i = 0; i < length; i++) {
+      //  y[i] = b[0] * x[i] +  b[1] * x[i-1] +  b[2] * x[i-2]
+      //                     + -a[1] * y[i-1] + -a[2] * y[i-2];
+
+      tmp_int32 = y[1] * ba[3];      // -a[1] * y[i-1] (low part)
+      tmp_int32 += y[3] * ba[4];     // -a[2] * y[i-2] (low part)
+      tmp_int32 = (tmp_int32 >> 15);
+      tmp_int32 += y[0] * ba[3];     // -a[1] * y[i-1] (high part)
+      tmp_int32 += y[2] * ba[4];     // -a[2] * y[i-2] (high part)
+      tmp_int32 = (tmp_int32 << 1);
+
+      tmp_int32 += data[i] * ba[0];  // b[0] * x[0]
+      tmp_int32 += x[0] * ba[1];     // b[1] * x[i-1]
+      tmp_int32 += x[1] * ba[2];     // b[2] * x[i-2]
+
+      // Update state (input part).
+      x[1] = x[0];
+      x[0] = data[i];
+
+      // Update state (filtered part).
+      y[2] = y[0];
+      y[3] = y[1];
+      y[0] = static_cast<int16_t>(tmp_int32 >> 13);
+      y[1] = static_cast<int16_t>(
+          (tmp_int32 - (static_cast<int32_t>(y[0]) << 13)) << 2);
+
+      // Rounding in Q12, i.e. add 2^11.
+      tmp_int32 += 2048;
+
+      // Saturate (to 2^27) so that the HP filtered signal does not overflow.
+      tmp_int32 = WEBRTC_SPL_SAT(static_cast<int32_t>(134217727),
+                                 tmp_int32,
+                                 static_cast<int32_t>(-134217728));
+
+      // Convert back to Q0 and use rounding.
+      data[i] = static_cast<int16_t>(tmp_int32 >> 12);
     }
   }
 
-  return apm_->kNoError;
+ private:
+  const int16_t* const ba_ = nullptr;
+  int16_t x_[2];
+  int16_t y_[4];
+};
+
+HighPassFilterImpl::HighPassFilterImpl(rtc::CriticalSection* crit)
+    : crit_(crit) {
+  RTC_DCHECK(crit_);
+}
+
+HighPassFilterImpl::~HighPassFilterImpl() {}
+
+void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) {
+  std::vector<rtc::scoped_ptr<BiquadFilter>> new_filters(channels);
+  for (size_t i = 0; i < channels; i++) {
+    new_filters[i].reset(new BiquadFilter(sample_rate_hz));
+  }
+  rtc::CritScope cs(crit_);
+  filters_.swap(new_filters);
+}
+
+void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
+
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
+  for (size_t i = 0; i < filters_.size(); i++) {
+    filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
+                         audio->num_frames_per_band());
+  }
 }
 
 int HighPassFilterImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (!enabled_ && enable) {
+    for (auto& filter : filters_) {
+      filter->Reset();
+    }
+  }
+  enabled_ = enable;
+  return AudioProcessing::kNoError;
 }
 
 bool HighPassFilterImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-void* HighPassFilterImpl::CreateHandle() const {
-  return new FilterState;
-}
-
-void HighPassFilterImpl::DestroyHandle(void* handle) const {
-  delete static_cast<Handle*>(handle);
-}
-
-int HighPassFilterImpl::InitializeHandle(void* handle) const {
-  return InitializeFilter(static_cast<Handle*>(handle),
-                          apm_->proc_sample_rate_hz());
-}
-
-int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const {
-  return apm_->kNoError; // Not configurable.
-}
-
-int HighPassFilterImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int HighPassFilterImpl::GetHandleError(void* handle) const {
-  // The component has no detailed errors.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.h
index 90b393e903..0e985bac7a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/high_pass_filter_impl.h
@@ -11,39 +11,34 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
-class HighPassFilterImpl : public HighPassFilter,
-                           public ProcessingComponent {
+class HighPassFilterImpl : public HighPassFilter {
  public:
-  HighPassFilterImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
-  virtual ~HighPassFilterImpl();
+  explicit HighPassFilterImpl(rtc::CriticalSection* crit);
+  ~HighPassFilterImpl() override;
 
-  int ProcessCaptureAudio(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize(size_t channels, int sample_rate_hz);
+  void ProcessCaptureAudio(AudioBuffer* audio);
 
   // HighPassFilter implementation.
+  int Enable(bool enable) override;
   bool is_enabled() const override;
 
  private:
-  // HighPassFilter implementation.
-  int Enable(bool enable) override;
-
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
+  class BiquadFilter;
+  rtc::CriticalSection* const crit_ = nullptr;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  std::vector<rtc::scoped_ptr<BiquadFilter>> filters_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/include/audio_processing.h b/media/webrtc/trunk/webrtc/modules/audio_processing/include/audio_processing.h
index 97b14b0d9c..dec358b8cc 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/include/audio_processing.h
@@ -11,10 +11,17 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#ifndef _USE_MATH_DEFINES
+#define _USE_MATH_DEFINES
+#endif
+
+#include <math.h>
 #include <stddef.h>  // size_t
 #include <stdio.h>  // FILE
 #include <vector>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/beamformer/array_util.h"
@@ -29,6 +36,9 @@ class AudioFrame;
 template<typename T>
 class Beamformer;
 
+class StreamConfig;
+class ProcessingConfig;
+
 class EchoCancellation;
 class EchoControlMobile;
 class GainControl;
@@ -37,22 +47,10 @@ class LevelEstimator;
 class NoiseSuppression;
 class VoiceDetection;
 
-struct ExtendedFilter {
-  ExtendedFilter() : enabled(false) {}
-  explicit ExtendedFilter(bool enabled) : enabled(enabled) {}
-  bool enabled;
-};
-
-struct DelayAgnostic {
-  DelayAgnostic() : enabled(false) {}
-  explicit DelayAgnostic(bool enabled) : enabled(enabled) {}
-  bool enabled;
-};
-
-// Use to enable the delay correction feature. This now engages an extended
-// filter mode in the AEC, along with robustness measures around the reported
-// system delays. It comes with a significant increase in AEC complexity, but is
-// much more robust to unreliable reported delays.
+// Use to enable the extended filter mode in the AEC, along with robustness
+// measures around the reported system delays. It comes with a significant
+// increase in AEC complexity, but is much more robust to unreliable reported
+// delays.
 //
 // Detailed changes to the algorithm:
 // - The filter length is changed from 48 to 128 ms. This comes with tuning of
@@ -66,30 +64,45 @@ struct DelayAgnostic {
 //   the delay difference more heavily, and back off from the difference more.
 //   Adjustments force a readaptation of the filter, so they should be avoided
 //   except when really necessary.
-struct DelayCorrection {
-  DelayCorrection() : enabled(false) {}
-  explicit DelayCorrection(bool enabled) : enabled(enabled) {}
+struct ExtendedFilter {
+  ExtendedFilter() : enabled(false) {}
+  explicit ExtendedFilter(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kExtendedFilter;
   bool enabled;
 };
 
-// Use to disable the reported system delays. By disabling the reported system
-// delays the echo cancellation algorithm assumes the process and reverse
-// streams to be aligned. This configuration only applies to EchoCancellation
-// and not EchoControlMobile and is set with AudioProcessing::SetExtraOptions().
-// Note that by disabling reported system delays the EchoCancellation may
-// regress in performance.
-struct ReportedDelay {
-  ReportedDelay() : enabled(true) {}
-  explicit ReportedDelay(bool enabled) : enabled(enabled) {}
+// Enables delay-agnostic echo cancellation. This feature relies on internally
+// estimated delays between the process and reverse streams, thus not relying
+// on reported system delays. This configuration only applies to
+// EchoCancellation and not EchoControlMobile. It can be set in the constructor
+// or using AudioProcessing::SetExtraOptions().
+struct DelayAgnostic {
+  DelayAgnostic() : enabled(false) {}
+  explicit DelayAgnostic(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kDelayAgnostic;
   bool enabled;
 };
 
-// Must be provided through AudioProcessing::Create(Confg&). It will have no
-// impact if used with AudioProcessing::SetExtraOptions().
+// Use to enable experimental gain control (AGC). At startup the experimental
+// AGC moves the microphone volume up to |startup_min_volume| if the current
+// microphone volume is set too low. The value is clamped to its operating range
+// [12, 255]. Here, 255 maps to 100%.
+//
+// Must be provided through AudioProcessing::Create(Confg&).
+#if defined(WEBRTC_CHROMIUM_BUILD)
+static const int kAgcStartupMinVolume = 85;
+#else
+static const int kAgcStartupMinVolume = 0;
+#endif  // defined(WEBRTC_CHROMIUM_BUILD)
 struct ExperimentalAgc {
-  ExperimentalAgc() : enabled(true) {}
-  explicit ExperimentalAgc(bool enabled) : enabled(enabled) {}
+  ExperimentalAgc() : enabled(true), startup_min_volume(kAgcStartupMinVolume) {}
+  explicit ExperimentalAgc(bool enabled)
+      : enabled(enabled), startup_min_volume(kAgcStartupMinVolume) {}
+  ExperimentalAgc(bool enabled, int startup_min_volume)
+      : enabled(enabled), startup_min_volume(startup_min_volume) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kExperimentalAgc;
   bool enabled;
+  int startup_min_volume;
 };
 
 // Use to enable experimental noise suppression. It can be set in the
@@ -97,31 +110,48 @@ struct ExperimentalAgc {
 struct ExperimentalNs {
   ExperimentalNs() : enabled(false) {}
   explicit ExperimentalNs(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kExperimentalNs;
   bool enabled;
 };
 
 // Use to enable beamforming. Must be provided through the constructor. It will
 // have no impact if used with AudioProcessing::SetExtraOptions().
 struct Beamforming {
-  Beamforming() : enabled(false) {}
+  Beamforming()
+      : enabled(false),
+        array_geometry(),
+        target_direction(
+            SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f)) {}
   Beamforming(bool enabled, const std::vector<Point>& array_geometry)
+      : Beamforming(enabled,
+                    array_geometry,
+                    SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f)) {
+  }
+  Beamforming(bool enabled,
+              const std::vector<Point>& array_geometry,
+              SphericalPointf target_direction)
       : enabled(enabled),
-        array_geometry(array_geometry) {}
+        array_geometry(array_geometry),
+        target_direction(target_direction) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kBeamforming;
   const bool enabled;
   const std::vector<Point> array_geometry;
+  const SphericalPointf target_direction;
 };
 
-// Use to enable 48kHz support in audio processing. Must be provided through the
-// constructor. It will have no impact if used with
+// Use to enable intelligibility enhancer in audio processing. Must be provided
+// though the constructor. It will have no impact if used with
 // AudioProcessing::SetExtraOptions().
-struct AudioProcessing48kHzSupport {
-  AudioProcessing48kHzSupport() : enabled(false) {}
-  explicit AudioProcessing48kHzSupport(bool enabled) : enabled(enabled) {}
+//
+// Note: If enabled and the reverse stream has more than one output channel,
+// the reverse stream will become an upmixed mono signal.
+struct Intelligibility {
+  Intelligibility() : enabled(false) {}
+  explicit Intelligibility(bool enabled) : enabled(enabled) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kIntelligibility;
   bool enabled;
 };
 
-static const int kAudioProcMaxNativeSampleRateHz = 32000;
-
 // The Audio Processing Module (APM) provides a collection of voice processing
 // components designed for real-time communications software.
 //
@@ -197,6 +227,7 @@ static const int kAudioProcMaxNativeSampleRateHz = 32000;
 //
 class AudioProcessing {
  public:
+  // TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
   enum ChannelLayout {
     kMono,
     // Left, right.
@@ -234,10 +265,17 @@ class AudioProcessing {
   // The int16 interfaces require:
   //   - only |NativeRate|s be used
   //   - that the input, output and reverse rates must match
-  //   - that |output_layout| matches |input_layout|
+  //   - that |processing_config.output_stream()| matches
+  //     |processing_config.input_stream()|.
   //
-  // The float interfaces accept arbitrary rates and support differing input
-  // and output layouts, but the output may only remove channels, not add.
+  // The float interfaces accept arbitrary rates and support differing input and
+  // output layouts, but the output must have either one channel or the same
+  // number of channels as the input.
+  virtual int Initialize(const ProcessingConfig& processing_config) = 0;
+
+  // Initialize with unpacked parameters. See Initialize() above for details.
+  //
+  // TODO(mgraczyk): Remove once clients are updated to use the new interface.
   virtual int Initialize(int input_sample_rate_hz,
                          int output_sample_rate_hz,
                          int reverse_sample_rate_hz,
@@ -249,29 +287,24 @@ class AudioProcessing {
   // ensures the options are applied immediately.
   virtual void SetExtraOptions(const Config& config) = 0;
 
-  // DEPRECATED.
-  // TODO(ajm): Remove after Chromium has upgraded to using Initialize().
-  virtual int set_sample_rate_hz(int rate) = 0;
-  // TODO(ajm): Remove after voice engine no longer requires it to resample
+  // TODO(peah): Remove after voice engine no longer requires it to resample
   // the reverse stream to the forward rate.
   virtual int input_sample_rate_hz() const = 0;
-  // TODO(ajm): Remove after Chromium no longer depends on it.
-  virtual int sample_rate_hz() const = 0;
 
   // TODO(ajm): Only intended for internal use. Make private and friend the
   // necessary classes?
   virtual int proc_sample_rate_hz() const = 0;
   virtual int proc_split_sample_rate_hz() const = 0;
-  virtual int num_input_channels() const = 0;
-  virtual int num_output_channels() const = 0;
-  virtual int num_reverse_channels() const = 0;
+  virtual size_t num_input_channels() const = 0;
+  virtual size_t num_proc_channels() const = 0;
+  virtual size_t num_output_channels() const = 0;
+  virtual size_t num_reverse_channels() const = 0;
 
   // Set to true when the output of AudioProcessing will be muted or in some
   // other way not used. Ideally, the captured audio would still be processed,
   // but some components may change behavior based on this information.
   // Default false.
   virtual void set_output_will_be_muted(bool muted) = 0;
-  virtual bool output_will_be_muted() const = 0;
 
   // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
   // this is the near-end (or captured) audio.
@@ -290,16 +323,30 @@ class AudioProcessing {
   // |input_layout|. At output, the channels will be arranged according to
   // |output_layout| at |output_sample_rate_hz| in |dest|.
   //
-  // The output layout may only remove channels, not add. |src| and |dest|
-  // may use the same memory, if desired.
+  // The output layout must have one channel or as many channels as the input.
+  // |src| and |dest| may use the same memory, if desired.
+  //
+  // TODO(mgraczyk): Remove once clients are updated to use the new interface.
   virtual int ProcessStream(const float* const* src,
-                            int samples_per_channel,
+                            size_t samples_per_channel,
                             int input_sample_rate_hz,
                             ChannelLayout input_layout,
                             int output_sample_rate_hz,
                             ChannelLayout output_layout,
                             float* const* dest) = 0;
 
+  // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
+  // |src| points to a channel buffer, arranged according to |input_stream|. At
+  // output, the channels will be arranged according to |output_stream| in
+  // |dest|.
+  //
+  // The output must have one channel or as many channels as the input. |src|
+  // and |dest| may use the same memory, if desired.
+  virtual int ProcessStream(const float* const* src,
+                            const StreamConfig& input_config,
+                            const StreamConfig& output_config,
+                            float* const* dest) = 0;
+
   // Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame
   // will not be modified. On the client-side, this is the far-end (or to be
   // rendered) audio.
@@ -315,15 +362,29 @@ class AudioProcessing {
   // |input_sample_rate_hz()|
   //
   // TODO(ajm): add const to input; requires an implementation fix.
+  // DEPRECATED: Use |ProcessReverseStream| instead.
+  // TODO(ekm): Remove once all users have updated to |ProcessReverseStream|.
   virtual int AnalyzeReverseStream(AudioFrame* frame) = 0;
 
+  // Same as |AnalyzeReverseStream|, but may modify |frame| if intelligibility
+  // is enabled.
+  virtual int ProcessReverseStream(AudioFrame* frame) = 0;
+
   // Accepts deinterleaved float audio with the range [-1, 1]. Each element
   // of |data| points to a channel buffer, arranged according to |layout|.
+  // TODO(mgraczyk): Remove once clients are updated to use the new interface.
   virtual int AnalyzeReverseStream(const float* const* data,
-                                   int samples_per_channel,
-                                   int sample_rate_hz,
+                                   size_t samples_per_channel,
+                                   int rev_sample_rate_hz,
                                    ChannelLayout layout) = 0;
 
+  // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
+  // |data| points to a channel buffer, arranged according to |reverse_config|.
+  virtual int ProcessReverseStream(const float* const* src,
+                                   const StreamConfig& reverse_input_config,
+                                   const StreamConfig& reverse_output_config,
+                                   float* const* dest) = 0;
+
   // This must be called if and only if echo processing is enabled.
   //
   // Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end
@@ -344,7 +405,6 @@ class AudioProcessing {
   // Call to signal that a key press occurred (true) or did not occur (false)
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
-  virtual bool stream_key_pressed() const = 0;
 
   // Sets a delay |offset| in ms to add to the values passed in through
   // set_stream_delay_ms(). May be positive or negative.
@@ -376,6 +436,10 @@ class AudioProcessing {
   // cannot be resumed in the same file (without overwriting it).
   virtual int StopDebugRecording() = 0;
 
+  // Use to send UMA histograms at end of a call. Note that all histogram
+  // specific member variables are reset.
+  virtual void UpdateHistogramsOnCallEnd() = 0;
+
   // These provide access to the component interfaces and should never return
   // NULL. The pointers will be valid for the lifetime of the APM instance.
   // The memory for these objects is entirely managed internally.
@@ -423,9 +487,121 @@ class AudioProcessing {
     kSampleRate48kHz = 48000
   };
 
+  static const int kNativeSampleRatesHz[];
+  static const size_t kNumNativeSampleRates;
+  static const int kMaxNativeSampleRateHz;
+  static const int kMaxAECMSampleRateHz;
+
   static const int kChunkSizeMs = 10;
 };
 
+class StreamConfig {
+ public:
+  // sample_rate_hz: The sampling rate of the stream.
+  //
+  // num_channels: The number of audio channels in the stream, excluding the
+  //               keyboard channel if it is present. When passing a
+  //               StreamConfig with an array of arrays T*[N],
+  //
+  //                N == {num_channels + 1  if  has_keyboard
+  //                     {num_channels      if  !has_keyboard
+  //
+  // has_keyboard: True if the stream has a keyboard channel. When has_keyboard
+  //               is true, the last channel in any corresponding list of
+  //               channels is the keyboard channel.
+  StreamConfig(int sample_rate_hz = 0,
+               size_t num_channels = 0,
+               bool has_keyboard = false)
+      : sample_rate_hz_(sample_rate_hz),
+        num_channels_(num_channels),
+        has_keyboard_(has_keyboard),
+        num_frames_(calculate_frames(sample_rate_hz)) {}
+
+  void set_sample_rate_hz(int value) {
+    sample_rate_hz_ = value;
+    num_frames_ = calculate_frames(value);
+  }
+  void set_num_channels(size_t value) { num_channels_ = value; }
+  void set_has_keyboard(bool value) { has_keyboard_ = value; }
+
+  int sample_rate_hz() const { return sample_rate_hz_; }
+
+  // The number of channels in the stream, not including the keyboard channel if
+  // present.
+  size_t num_channels() const { return num_channels_; }
+
+  bool has_keyboard() const { return has_keyboard_; }
+  size_t num_frames() const { return num_frames_; }
+  size_t num_samples() const { return num_channels_ * num_frames_; }
+
+  bool operator==(const StreamConfig& other) const {
+    return sample_rate_hz_ == other.sample_rate_hz_ &&
+           num_channels_ == other.num_channels_ &&
+           has_keyboard_ == other.has_keyboard_;
+  }
+
+  bool operator!=(const StreamConfig& other) const { return !(*this == other); }
+
+ private:
+  static size_t calculate_frames(int sample_rate_hz) {
+    return static_cast<size_t>(
+        AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000);
+  }
+
+  int sample_rate_hz_;
+  size_t num_channels_;
+  bool has_keyboard_;
+  size_t num_frames_;
+};
+
+class ProcessingConfig {
+ public:
+  enum StreamName {
+    kInputStream,
+    kOutputStream,
+    kReverseInputStream,
+    kReverseOutputStream,
+    kNumStreamNames,
+  };
+
+  const StreamConfig& input_stream() const {
+    return streams[StreamName::kInputStream];
+  }
+  const StreamConfig& output_stream() const {
+    return streams[StreamName::kOutputStream];
+  }
+  const StreamConfig& reverse_input_stream() const {
+    return streams[StreamName::kReverseInputStream];
+  }
+  const StreamConfig& reverse_output_stream() const {
+    return streams[StreamName::kReverseOutputStream];
+  }
+
+  StreamConfig& input_stream() { return streams[StreamName::kInputStream]; }
+  StreamConfig& output_stream() { return streams[StreamName::kOutputStream]; }
+  StreamConfig& reverse_input_stream() {
+    return streams[StreamName::kReverseInputStream];
+  }
+  StreamConfig& reverse_output_stream() {
+    return streams[StreamName::kReverseOutputStream];
+  }
+
+  bool operator==(const ProcessingConfig& other) const {
+    for (int i = 0; i < StreamName::kNumStreamNames; ++i) {
+      if (this->streams[i] != other.streams[i]) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  bool operator!=(const ProcessingConfig& other) const {
+    return !(*this == other);
+  }
+
+  StreamConfig streams[StreamName::kNumStreamNames];
+};
+
 // The acoustic echo cancellation (AEC) component provides better performance
 // than AECM but also requires more processing power and is dependent on delay
 // stability and reporting accuracy. As such it is well-suited and recommended
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/include/mock_audio_processing.h b/media/webrtc/trunk/webrtc/modules/audio_processing/include/mock_audio_processing.h
index 63161c891c..9e1f2d5861 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -186,6 +186,8 @@ class MockAudioProcessing : public AudioProcessing {
           ChannelLayout input_layout,
           ChannelLayout output_layout,
           ChannelLayout reverse_layout));
+  MOCK_METHOD1(Initialize,
+      int(const ProcessingConfig& processing_config));
   MOCK_METHOD1(SetExtraOptions,
       void(const Config& config));
   MOCK_METHOD1(set_sample_rate_hz,
@@ -199,11 +201,11 @@ class MockAudioProcessing : public AudioProcessing {
   MOCK_CONST_METHOD0(proc_split_sample_rate_hz,
       int());
   MOCK_CONST_METHOD0(num_input_channels,
-      int());
+      size_t());
   MOCK_CONST_METHOD0(num_output_channels,
-      int());
+      size_t());
   MOCK_CONST_METHOD0(num_reverse_channels,
-      int());
+      size_t());
   MOCK_METHOD1(set_output_will_be_muted,
       void(bool muted));
   MOCK_CONST_METHOD0(output_will_be_muted,
@@ -212,17 +214,28 @@ class MockAudioProcessing : public AudioProcessing {
       int(AudioFrame* frame));
   MOCK_METHOD7(ProcessStream,
       int(const float* const* src,
-          int samples_per_channel,
+          size_t samples_per_channel,
           int input_sample_rate_hz,
           ChannelLayout input_layout,
           int output_sample_rate_hz,
           ChannelLayout output_layout,
           float* const* dest));
+  MOCK_METHOD4(ProcessStream,
+               int(const float* const* src,
+                   const StreamConfig& input_config,
+                   const StreamConfig& output_config,
+                   float* const* dest));
   MOCK_METHOD1(AnalyzeReverseStream,
       int(AudioFrame* frame));
+  MOCK_METHOD1(ProcessReverseStream, int(AudioFrame* frame));
   MOCK_METHOD4(AnalyzeReverseStream,
-      int(const float* const* data, int frames, int sample_rate_hz,
+      int(const float* const* data, size_t frames, int sample_rate_hz,
           ChannelLayout input_layout));
+  MOCK_METHOD4(ProcessReverseStream,
+               int(const float* const* src,
+                   const StreamConfig& input_config,
+                   const StreamConfig& output_config,
+                   float* const* dest));
   MOCK_METHOD1(set_stream_delay_ms,
       int(int delay));
   MOCK_CONST_METHOD0(stream_delay_ms,
@@ -243,6 +256,7 @@ class MockAudioProcessing : public AudioProcessing {
       int(FILE* handle));
   MOCK_METHOD0(StopDebugRecording,
       int());
+  MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
   virtual MockEchoCancellation* echo_cancellation() const {
     return echo_cancellation_.get();
   }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
new file mode 100644
index 0000000000..fe964aba8c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
@@ -0,0 +1,381 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Implements core class for intelligibility enhancer.
+//
+//  Details of the model and algorithm can be found in the original paper:
+//  http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788
+//
+
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <algorithm>
+#include <numeric>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/window_generator.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kErbResolution = 2;
+const int kWindowSizeMs = 2;
+const int kChunkSizeMs = 10;  // Size provided by APM.
+const float kClipFreq = 200.0f;
+const float kConfigRho = 0.02f;  // Default production and interpretation SNR.
+const float kKbdAlpha = 1.5f;
+const float kLambdaBot = -1.0f;      // Extreme values in bisection
+const float kLambdaTop = -10e-18f;  // search for lamda.
+
+}  // namespace
+
+using std::complex;
+using std::max;
+using std::min;
+using VarianceType = intelligibility::VarianceArray::StepType;
+
+IntelligibilityEnhancer::TransformCallback::TransformCallback(
+    IntelligibilityEnhancer* parent,
+    IntelligibilityEnhancer::AudioSource source)
+    : parent_(parent), source_(source) {
+}
+
+void IntelligibilityEnhancer::TransformCallback::ProcessAudioBlock(
+    const complex<float>* const* in_block,
+    size_t in_channels,
+    size_t frames,
+    size_t /* out_channels */,
+    complex<float>* const* out_block) {
+  RTC_DCHECK_EQ(parent_->freqs_, frames);
+  for (size_t i = 0; i < in_channels; ++i) {
+    parent_->DispatchAudio(source_, in_block[i], out_block[i]);
+  }
+}
+
+IntelligibilityEnhancer::IntelligibilityEnhancer()
+    : IntelligibilityEnhancer(IntelligibilityEnhancer::Config()) {
+}
+
+IntelligibilityEnhancer::IntelligibilityEnhancer(const Config& config)
+    : freqs_(RealFourier::ComplexLength(
+          RealFourier::FftOrder(config.sample_rate_hz * kWindowSizeMs / 1000))),
+      window_size_(static_cast<size_t>(1 << RealFourier::FftOrder(freqs_))),
+      chunk_length_(
+          static_cast<size_t>(config.sample_rate_hz * kChunkSizeMs / 1000)),
+      bank_size_(GetBankSize(config.sample_rate_hz, kErbResolution)),
+      sample_rate_hz_(config.sample_rate_hz),
+      erb_resolution_(kErbResolution),
+      num_capture_channels_(config.num_capture_channels),
+      num_render_channels_(config.num_render_channels),
+      analysis_rate_(config.analysis_rate),
+      active_(true),
+      clear_variance_(freqs_,
+                      config.var_type,
+                      config.var_window_size,
+                      config.var_decay_rate),
+      noise_variance_(freqs_,
+                      config.var_type,
+                      config.var_window_size,
+                      config.var_decay_rate),
+      filtered_clear_var_(new float[bank_size_]),
+      filtered_noise_var_(new float[bank_size_]),
+      filter_bank_(bank_size_),
+      center_freqs_(new float[bank_size_]),
+      rho_(new float[bank_size_]),
+      gains_eq_(new float[bank_size_]),
+      gain_applier_(freqs_, config.gain_change_limit),
+      temp_render_out_buffer_(chunk_length_, num_render_channels_),
+      temp_capture_out_buffer_(chunk_length_, num_capture_channels_),
+      kbd_window_(new float[window_size_]),
+      render_callback_(this, AudioSource::kRenderStream),
+      capture_callback_(this, AudioSource::kCaptureStream),
+      block_count_(0),
+      analysis_step_(0) {
+  RTC_DCHECK_LE(config.rho, 1.0f);
+
+  CreateErbBank();
+
+  // Assumes all rho equal.
+  for (size_t i = 0; i < bank_size_; ++i) {
+    rho_[i] = config.rho * config.rho;
+  }
+
+  float freqs_khz = kClipFreq / 1000.0f;
+  size_t erb_index = static_cast<size_t>(ceilf(
+      11.17f * logf((freqs_khz + 0.312f) / (freqs_khz + 14.6575f)) + 43.0f));
+  start_freq_ = std::max(static_cast<size_t>(1), erb_index * erb_resolution_);
+
+  WindowGenerator::KaiserBesselDerived(kKbdAlpha, window_size_,
+                                       kbd_window_.get());
+  render_mangler_.reset(new LappedTransform(
+      num_render_channels_, num_render_channels_, chunk_length_,
+      kbd_window_.get(), window_size_, window_size_ / 2, &render_callback_));
+  capture_mangler_.reset(new LappedTransform(
+      num_capture_channels_, num_capture_channels_, chunk_length_,
+      kbd_window_.get(), window_size_, window_size_ / 2, &capture_callback_));
+}
+
+void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio,
+                                                 int sample_rate_hz,
+                                                 size_t num_channels) {
+  RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
+  RTC_CHECK_EQ(num_render_channels_, num_channels);
+
+  if (active_) {
+    render_mangler_->ProcessChunk(audio, temp_render_out_buffer_.channels());
+  }
+
+  if (active_) {
+    for (size_t i = 0; i < num_render_channels_; ++i) {
+      memcpy(audio[i], temp_render_out_buffer_.channels()[i],
+             chunk_length_ * sizeof(**audio));
+    }
+  }
+}
+
+void IntelligibilityEnhancer::AnalyzeCaptureAudio(float* const* audio,
+                                                  int sample_rate_hz,
+                                                  size_t num_channels) {
+  RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
+  RTC_CHECK_EQ(num_capture_channels_, num_channels);
+
+  capture_mangler_->ProcessChunk(audio, temp_capture_out_buffer_.channels());
+}
+
+void IntelligibilityEnhancer::DispatchAudio(
+    IntelligibilityEnhancer::AudioSource source,
+    const complex<float>* in_block,
+    complex<float>* out_block) {
+  switch (source) {
+    case kRenderStream:
+      ProcessClearBlock(in_block, out_block);
+      break;
+    case kCaptureStream:
+      ProcessNoiseBlock(in_block, out_block);
+      break;
+  }
+}
+
+void IntelligibilityEnhancer::ProcessClearBlock(const complex<float>* in_block,
+                                                complex<float>* out_block) {
+  if (block_count_ < 2) {
+    memset(out_block, 0, freqs_ * sizeof(*out_block));
+    ++block_count_;
+    return;
+  }
+
+  // TODO(ekm): Use VAD to |Step| and |AnalyzeClearBlock| only if necessary.
+  if (true) {
+    clear_variance_.Step(in_block, false);
+    if (block_count_ % analysis_rate_ == analysis_rate_ - 1) {
+      const float power_target = std::accumulate(
+          clear_variance_.variance(), clear_variance_.variance() + freqs_, 0.f);
+      AnalyzeClearBlock(power_target);
+      ++analysis_step_;
+    }
+    ++block_count_;
+  }
+
+  if (active_) {
+    gain_applier_.Apply(in_block, out_block);
+  }
+}
+
+void IntelligibilityEnhancer::AnalyzeClearBlock(float power_target) {
+  FilterVariance(clear_variance_.variance(), filtered_clear_var_.get());
+  FilterVariance(noise_variance_.variance(), filtered_noise_var_.get());
+
+  SolveForGainsGivenLambda(kLambdaTop, start_freq_, gains_eq_.get());
+  const float power_top =
+      DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
+  SolveForGainsGivenLambda(kLambdaBot, start_freq_, gains_eq_.get());
+  const float power_bot =
+      DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
+  if (power_target >= power_bot && power_target <= power_top) {
+    SolveForLambda(power_target, power_bot, power_top);
+    UpdateErbGains();
+  }  // Else experiencing variance underflow, so do nothing.
+}
+
+void IntelligibilityEnhancer::SolveForLambda(float power_target,
+                                             float power_bot,
+                                             float power_top) {
+  const float kConvergeThresh = 0.001f;  // TODO(ekmeyerson): Find best values
+  const int kMaxIters = 100;             // for these, based on experiments.
+
+  const float reciprocal_power_target = 1.f / power_target;
+  float lambda_bot = kLambdaBot;
+  float lambda_top = kLambdaTop;
+  float power_ratio = 2.0f;  // Ratio of achieved power to target power.
+  int iters = 0;
+  while (std::fabs(power_ratio - 1.0f) > kConvergeThresh &&
+         iters <= kMaxIters) {
+    const float lambda = lambda_bot + (lambda_top - lambda_bot) / 2.0f;
+    SolveForGainsGivenLambda(lambda, start_freq_, gains_eq_.get());
+    const float power =
+        DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
+    if (power < power_target) {
+      lambda_bot = lambda;
+    } else {
+      lambda_top = lambda;
+    }
+    power_ratio = std::fabs(power * reciprocal_power_target);
+    ++iters;
+  }
+}
+
+void IntelligibilityEnhancer::UpdateErbGains() {
+  // (ERB gain) = filterbank' * (freq gain)
+  float* gains = gain_applier_.target();
+  for (size_t i = 0; i < freqs_; ++i) {
+    gains[i] = 0.0f;
+    for (size_t j = 0; j < bank_size_; ++j) {
+      gains[i] = fmaf(filter_bank_[j][i], gains_eq_[j], gains[i]);
+    }
+  }
+}
+
+void IntelligibilityEnhancer::ProcessNoiseBlock(const complex<float>* in_block,
+                                                complex<float>* /*out_block*/) {
+  noise_variance_.Step(in_block);
+}
+
+size_t IntelligibilityEnhancer::GetBankSize(int sample_rate,
+                                            size_t erb_resolution) {
+  float freq_limit = sample_rate / 2000.0f;
+  size_t erb_scale = static_cast<size_t>(ceilf(
+      11.17f * logf((freq_limit + 0.312f) / (freq_limit + 14.6575f)) + 43.0f));
+  return erb_scale * erb_resolution;
+}
+
+void IntelligibilityEnhancer::CreateErbBank() {
+  size_t lf = 1, rf = 4;
+
+  for (size_t i = 0; i < bank_size_; ++i) {
+    float abs_temp = fabsf((i + 1.0f) / static_cast<float>(erb_resolution_));
+    center_freqs_[i] = 676170.4f / (47.06538f - expf(0.08950404f * abs_temp));
+    center_freqs_[i] -= 14678.49f;
+  }
+  float last_center_freq = center_freqs_[bank_size_ - 1];
+  for (size_t i = 0; i < bank_size_; ++i) {
+    center_freqs_[i] *= 0.5f * sample_rate_hz_ / last_center_freq;
+  }
+
+  for (size_t i = 0; i < bank_size_; ++i) {
+    filter_bank_[i].resize(freqs_);
+  }
+
+  for (size_t i = 1; i <= bank_size_; ++i) {
+    size_t lll, ll, rr, rrr;
+    static const size_t kOne = 1;  // Avoids repeated static_cast<>s below.
+    lll = static_cast<size_t>(round(
+        center_freqs_[max(kOne, i - lf) - 1] * freqs_ /
+            (0.5f * sample_rate_hz_)));
+    ll = static_cast<size_t>(round(
+        center_freqs_[max(kOne, i) - 1] * freqs_ / (0.5f * sample_rate_hz_)));
+    lll = min(freqs_, max(lll, kOne)) - 1;
+    ll = min(freqs_, max(ll, kOne)) - 1;
+
+    rrr = static_cast<size_t>(round(
+        center_freqs_[min(bank_size_, i + rf) - 1] * freqs_ /
+            (0.5f * sample_rate_hz_)));
+    rr = static_cast<size_t>(round(
+        center_freqs_[min(bank_size_, i + 1) - 1] * freqs_ /
+            (0.5f * sample_rate_hz_)));
+    rrr = min(freqs_, max(rrr, kOne)) - 1;
+    rr = min(freqs_, max(rr, kOne)) - 1;
+
+    float step, element;
+
+    step = 1.0f / (ll - lll);
+    element = 0.0f;
+    for (size_t j = lll; j <= ll; ++j) {
+      filter_bank_[i - 1][j] = element;
+      element += step;
+    }
+    step = 1.0f / (rrr - rr);
+    element = 1.0f;
+    for (size_t j = rr; j <= rrr; ++j) {
+      filter_bank_[i - 1][j] = element;
+      element -= step;
+    }
+    for (size_t j = ll; j <= rr; ++j) {
+      filter_bank_[i - 1][j] = 1.0f;
+    }
+  }
+
+  float sum;
+  for (size_t i = 0; i < freqs_; ++i) {
+    sum = 0.0f;
+    for (size_t j = 0; j < bank_size_; ++j) {
+      sum += filter_bank_[j][i];
+    }
+    for (size_t j = 0; j < bank_size_; ++j) {
+      filter_bank_[j][i] /= sum;
+    }
+  }
+}
+
+void IntelligibilityEnhancer::SolveForGainsGivenLambda(float lambda,
+                                                       size_t start_freq,
+                                                       float* sols) {
+  bool quadratic = (kConfigRho < 1.0f);
+  const float* var_x0 = filtered_clear_var_.get();
+  const float* var_n0 = filtered_noise_var_.get();
+
+  for (size_t n = 0; n < start_freq; ++n) {
+    sols[n] = 1.0f;
+  }
+
+  // Analytic solution for optimal gains. See paper for derivation.
+  for (size_t n = start_freq - 1; n < bank_size_; ++n) {
+    float alpha0, beta0, gamma0;
+    gamma0 = 0.5f * rho_[n] * var_x0[n] * var_n0[n] +
+             lambda * var_x0[n] * var_n0[n] * var_n0[n];
+    beta0 = lambda * var_x0[n] * (2 - rho_[n]) * var_x0[n] * var_n0[n];
+    if (quadratic) {
+      alpha0 = lambda * var_x0[n] * (1 - rho_[n]) * var_x0[n] * var_x0[n];
+      sols[n] =
+          (-beta0 - sqrtf(beta0 * beta0 - 4 * alpha0 * gamma0)) / (2 * alpha0);
+    } else {
+      sols[n] = -gamma0 / beta0;
+    }
+    sols[n] = fmax(0, sols[n]);
+  }
+}
+
+void IntelligibilityEnhancer::FilterVariance(const float* var, float* result) {
+  RTC_DCHECK_GT(freqs_, 0u);
+  for (size_t i = 0; i < bank_size_; ++i) {
+    result[i] = DotProduct(&filter_bank_[i][0], var, freqs_);
+  }
+}
+
+float IntelligibilityEnhancer::DotProduct(const float* a,
+                                          const float* b,
+                                          size_t length) {
+  float ret = 0.0f;
+
+  for (size_t i = 0; i < length; ++i) {
+    ret = fmaf(a[i], b[i], ret);
+  }
+  return ret;
+}
+
+bool IntelligibilityEnhancer::active() const {
+  return active_;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
new file mode 100644
index 0000000000..1eb22342ad
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Specifies core class for intelligbility enhancement.
+//
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
+
+#include <complex>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/lapped_transform.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
+
+namespace webrtc {
+
+// Speech intelligibility enhancement module. Reads render and capture
+// audio streams and modifies the render stream with a set of gains per
+// frequency bin to enhance speech against the noise background.
+// Note: assumes speech and noise streams are already separated.
+class IntelligibilityEnhancer {
+ public:
+  struct Config {
+    // |var_*| are parameters for the VarianceArray constructor for the
+    // clear speech stream.
+    // TODO(bercic): the |var_*|, |*_rate| and |gain_limit| parameters should
+    // probably go away once fine tuning is done.
+    Config()
+        : sample_rate_hz(16000),
+          num_capture_channels(1),
+          num_render_channels(1),
+          var_type(intelligibility::VarianceArray::kStepDecaying),
+          var_decay_rate(0.9f),
+          var_window_size(10),
+          analysis_rate(800),
+          gain_change_limit(0.1f),
+          rho(0.02f) {}
+    int sample_rate_hz;
+    size_t num_capture_channels;
+    size_t num_render_channels;
+    intelligibility::VarianceArray::StepType var_type;
+    float var_decay_rate;
+    size_t var_window_size;
+    int analysis_rate;
+    float gain_change_limit;
+    float rho;
+  };
+
+  explicit IntelligibilityEnhancer(const Config& config);
+  IntelligibilityEnhancer();  // Initialize with default config.
+
+  // Reads and processes chunk of noise stream in time domain.
+  void AnalyzeCaptureAudio(float* const* audio,
+                           int sample_rate_hz,
+                           size_t num_channels);
+
+  // Reads chunk of speech in time domain and updates with modified signal.
+  void ProcessRenderAudio(float* const* audio,
+                          int sample_rate_hz,
+                          size_t num_channels);
+  bool active() const;
+
+ private:
+  enum AudioSource {
+    kRenderStream = 0,  // Clear speech stream.
+    kCaptureStream,  // Noise stream.
+  };
+
+  // Provides access point to the frequency domain.
+  class TransformCallback : public LappedTransform::Callback {
+   public:
+    TransformCallback(IntelligibilityEnhancer* parent, AudioSource source);
+
+    // All in frequency domain, receives input |in_block|, applies
+    // intelligibility enhancement, and writes result to |out_block|.
+    void ProcessAudioBlock(const std::complex<float>* const* in_block,
+                           size_t in_channels,
+                           size_t frames,
+                           size_t out_channels,
+                           std::complex<float>* const* out_block) override;
+
+   private:
+    IntelligibilityEnhancer* parent_;
+    AudioSource source_;
+  };
+  friend class TransformCallback;
+  FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation);
+  FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains);
+
+  // Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source.
+  void DispatchAudio(AudioSource source,
+                     const std::complex<float>* in_block,
+                     std::complex<float>* out_block);
+
+  // Updates variance computation and analysis with |in_block_|,
+  // and writes modified speech to |out_block|.
+  void ProcessClearBlock(const std::complex<float>* in_block,
+                         std::complex<float>* out_block);
+
+  // Computes and sets modified gains.
+  void AnalyzeClearBlock(float power_target);
+
+  // Bisection search for optimal |lambda|.
+  void SolveForLambda(float power_target, float power_bot, float power_top);
+
+  // Transforms freq gains to ERB gains.
+  void UpdateErbGains();
+
+  // Updates variance calculation for noise input with |in_block|.
+  void ProcessNoiseBlock(const std::complex<float>* in_block,
+                         std::complex<float>* out_block);
+
+  // Returns number of ERB filters.
+  static size_t GetBankSize(int sample_rate, size_t erb_resolution);
+
+  // Initializes ERB filterbank.
+  void CreateErbBank();
+
+  // Analytically solves quadratic for optimal gains given |lambda|.
+  // Negative gains are set to 0. Stores the results in |sols|.
+  void SolveForGainsGivenLambda(float lambda, size_t start_freq, float* sols);
+
+  // Computes variance across ERB filters from freq variance |var|.
+  // Stores in |result|.
+  void FilterVariance(const float* var, float* result);
+
+  // Returns dot product of vectors specified by size |length| arrays |a|,|b|.
+  static float DotProduct(const float* a, const float* b, size_t length);
+
+  const size_t freqs_;         // Num frequencies in frequency domain.
+  const size_t window_size_;   // Window size in samples; also the block size.
+  const size_t chunk_length_;  // Chunk size in samples.
+  const size_t bank_size_;     // Num ERB filters.
+  const int sample_rate_hz_;
+  const int erb_resolution_;
+  const size_t num_capture_channels_;
+  const size_t num_render_channels_;
+  const int analysis_rate_;    // Num blocks before gains recalculated.
+
+  const bool active_;          // Whether render gains are being updated.
+                               // TODO(ekm): Add logic for updating |active_|.
+
+  intelligibility::VarianceArray clear_variance_;
+  intelligibility::VarianceArray noise_variance_;
+  rtc::scoped_ptr<float[]> filtered_clear_var_;
+  rtc::scoped_ptr<float[]> filtered_noise_var_;
+  std::vector<std::vector<float>> filter_bank_;
+  rtc::scoped_ptr<float[]> center_freqs_;
+  size_t start_freq_;
+  rtc::scoped_ptr<float[]> rho_;  // Production and interpretation SNR.
+                                  // for each ERB band.
+  rtc::scoped_ptr<float[]> gains_eq_;  // Pre-filter modified gains.
+  intelligibility::GainApplier gain_applier_;
+
+  // Destination buffers used to reassemble blocked chunks before overwriting
+  // the original input array with modifications.
+  ChannelBuffer<float> temp_render_out_buffer_;
+  ChannelBuffer<float> temp_capture_out_buffer_;
+
+  rtc::scoped_ptr<float[]> kbd_window_;
+  TransformCallback render_callback_;
+  TransformCallback capture_callback_;
+  rtc::scoped_ptr<LappedTransform> render_mangler_;
+  rtc::scoped_ptr<LappedTransform> capture_mangler_;
+  int block_count_;
+  int analysis_step_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc
new file mode 100644
index 0000000000..ce146deaf5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Unit tests for intelligibility enhancer.
+//
+
+#include <math.h>
+#include <stdlib.h>
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
+
+namespace webrtc {
+
+namespace {
+
+// Target output for ERB create test. Generated with matlab.
+const float kTestCenterFreqs[] = {
+    13.169f, 26.965f, 41.423f, 56.577f, 72.461f, 89.113f, 106.57f, 124.88f,
+    144.08f, 164.21f, 185.34f, 207.5f,  230.75f, 255.16f, 280.77f, 307.66f,
+    335.9f,  365.56f, 396.71f, 429.44f, 463.84f, 500.f};
+const float kTestFilterBank[][2] = {{0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.f},
+                                    {0.055556f, 0.2f},
+                                    {0, 0.2f},
+                                    {0, 0.2f},
+                                    {0, 0.2f},
+                                    {0, 0.2f}};
+static_assert(arraysize(kTestCenterFreqs) == arraysize(kTestFilterBank),
+              "Test filterbank badly initialized.");
+
+// Target output for gain solving test. Generated with matlab.
+const size_t kTestStartFreq = 12;  // Lowest integral frequency for ERBs.
+const float kTestZeroVar[] = {1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f,
+                              1.f, 1.f, 1.f, 0.f, 0.f, 0.f, 0.f, 0.f,
+                              0.f, 0.f, 0.f, 0.f, 0.f, 0.f};
+static_assert(arraysize(kTestCenterFreqs) == arraysize(kTestZeroVar),
+              "Variance test data badly initialized.");
+const float kTestNonZeroVarLambdaTop[] = {
+    1.f,     1.f,     1.f,     1.f,     1.f,     1.f,     1.f,     1.f,
+    1.f,     1.f,     1.f,     0.f,     0.f,     0.0351f, 0.0636f, 0.0863f,
+    0.1037f, 0.1162f, 0.1236f, 0.1251f, 0.1189f, 0.0993f};
+static_assert(arraysize(kTestCenterFreqs) ==
+                  arraysize(kTestNonZeroVarLambdaTop),
+              "Variance test data badly initialized.");
+const float kMaxTestError = 0.005f;
+
+// Enhancer initialization parameters.
+const int kSamples = 2000;
+const int kSampleRate = 1000;
+const int kNumChannels = 1;
+const int kFragmentSize = kSampleRate / 100;
+
+}  // namespace
+
+using std::vector;
+using intelligibility::VarianceArray;
+
+class IntelligibilityEnhancerTest : public ::testing::Test {
+ protected:
+  IntelligibilityEnhancerTest()
+      : clear_data_(kSamples), noise_data_(kSamples), orig_data_(kSamples) {
+    config_.sample_rate_hz = kSampleRate;
+    enh_.reset(new IntelligibilityEnhancer(config_));
+  }
+
+  bool CheckUpdate(VarianceArray::StepType step_type) {
+    config_.sample_rate_hz = kSampleRate;
+    config_.var_type = step_type;
+    enh_.reset(new IntelligibilityEnhancer(config_));
+    float* clear_cursor = &clear_data_[0];
+    float* noise_cursor = &noise_data_[0];
+    for (int i = 0; i < kSamples; i += kFragmentSize) {
+      enh_->AnalyzeCaptureAudio(&noise_cursor, kSampleRate, kNumChannels);
+      enh_->ProcessRenderAudio(&clear_cursor, kSampleRate, kNumChannels);
+      clear_cursor += kFragmentSize;
+      noise_cursor += kFragmentSize;
+    }
+    for (int i = 0; i < kSamples; i++) {
+      if (std::fabs(clear_data_[i] - orig_data_[i]) > kMaxTestError) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  IntelligibilityEnhancer::Config config_;
+  rtc::scoped_ptr<IntelligibilityEnhancer> enh_;
+  vector<float> clear_data_;
+  vector<float> noise_data_;
+  vector<float> orig_data_;
+};
+
+// For each class of generated data, tests that render stream is
+// updated when it should be for each variance update method.
+TEST_F(IntelligibilityEnhancerTest, TestRenderUpdate) {
+  vector<VarianceArray::StepType> step_types;
+  step_types.push_back(VarianceArray::kStepInfinite);
+  step_types.push_back(VarianceArray::kStepDecaying);
+  step_types.push_back(VarianceArray::kStepWindowed);
+  step_types.push_back(VarianceArray::kStepBlocked);
+  step_types.push_back(VarianceArray::kStepBlockBasedMovingAverage);
+  std::fill(noise_data_.begin(), noise_data_.end(), 0.0f);
+  std::fill(orig_data_.begin(), orig_data_.end(), 0.0f);
+  for (auto step_type : step_types) {
+    std::fill(clear_data_.begin(), clear_data_.end(), 0.0f);
+    EXPECT_FALSE(CheckUpdate(step_type));
+  }
+  std::srand(1);
+  auto float_rand = []() { return std::rand() * 2.f / RAND_MAX - 1; };
+  std::generate(noise_data_.begin(), noise_data_.end(), float_rand);
+  for (auto step_type : step_types) {
+    EXPECT_FALSE(CheckUpdate(step_type));
+  }
+  for (auto step_type : step_types) {
+    std::generate(clear_data_.begin(), clear_data_.end(), float_rand);
+    orig_data_ = clear_data_;
+    EXPECT_TRUE(CheckUpdate(step_type));
+  }
+}
+
+// Tests ERB bank creation, comparing against matlab output.
+TEST_F(IntelligibilityEnhancerTest, TestErbCreation) {
+  ASSERT_EQ(arraysize(kTestCenterFreqs), enh_->bank_size_);
+  for (size_t i = 0; i < enh_->bank_size_; ++i) {
+    EXPECT_NEAR(kTestCenterFreqs[i], enh_->center_freqs_[i], kMaxTestError);
+    ASSERT_EQ(arraysize(kTestFilterBank[0]), enh_->freqs_);
+    for (size_t j = 0; j < enh_->freqs_; ++j) {
+      EXPECT_NEAR(kTestFilterBank[i][j], enh_->filter_bank_[i][j],
+                  kMaxTestError);
+    }
+  }
+}
+
+// Tests analytic solution for optimal gains, comparing
+// against matlab output.
+TEST_F(IntelligibilityEnhancerTest, TestSolveForGains) {
+  ASSERT_EQ(kTestStartFreq, enh_->start_freq_);
+  vector<float> sols(enh_->bank_size_);
+  float lambda = -0.001f;
+  for (size_t i = 0; i < enh_->bank_size_; i++) {
+    enh_->filtered_clear_var_[i] = 0.0f;
+    enh_->filtered_noise_var_[i] = 0.0f;
+    enh_->rho_[i] = 0.02f;
+  }
+  enh_->SolveForGainsGivenLambda(lambda, enh_->start_freq_, &sols[0]);
+  for (size_t i = 0; i < enh_->bank_size_; i++) {
+    EXPECT_NEAR(kTestZeroVar[i], sols[i], kMaxTestError);
+  }
+  for (size_t i = 0; i < enh_->bank_size_; i++) {
+    enh_->filtered_clear_var_[i] = static_cast<float>(i + 1);
+    enh_->filtered_noise_var_[i] = static_cast<float>(enh_->bank_size_ - i);
+  }
+  enh_->SolveForGainsGivenLambda(lambda, enh_->start_freq_, &sols[0]);
+  for (size_t i = 0; i < enh_->bank_size_; i++) {
+    EXPECT_NEAR(kTestNonZeroVarLambdaTop[i], sols[i], kMaxTestError);
+  }
+  lambda = -1.0;
+  enh_->SolveForGainsGivenLambda(lambda, enh_->start_freq_, &sols[0]);
+  for (size_t i = 0; i < enh_->bank_size_; i++) {
+    EXPECT_NEAR(kTestZeroVar[i], sols[i], kMaxTestError);
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
new file mode 100644
index 0000000000..7da9b957a4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
@@ -0,0 +1,314 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Implements helper functions and classes for intelligibility enhancement.
+//
+
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <algorithm>
+
+using std::complex;
+using std::min;
+
+namespace webrtc {
+
+namespace intelligibility {
+
+float UpdateFactor(float target, float current, float limit) {
+  float delta = fabsf(target - current);
+  float sign = copysign(1.0f, target - current);
+  return current + sign * fminf(delta, limit);
+}
+
+float AddDitherIfZero(float value) {
+  return value == 0.f ? std::rand() * 0.01f / RAND_MAX : value;
+}
+
+complex<float> zerofudge(complex<float> c) {
+  return complex<float>(AddDitherIfZero(c.real()), AddDitherIfZero(c.imag()));
+}
+
+complex<float> NewMean(complex<float> mean, complex<float> data, size_t count) {
+  return mean + (data - mean) / static_cast<float>(count);
+}
+
+void AddToMean(complex<float> data, size_t count, complex<float>* mean) {
+  (*mean) = NewMean(*mean, data, count);
+}
+
+
+static const size_t kWindowBlockSize = 10;
+
+VarianceArray::VarianceArray(size_t num_freqs,
+                             StepType type,
+                             size_t window_size,
+                             float decay)
+    : running_mean_(new complex<float>[num_freqs]()),
+      running_mean_sq_(new complex<float>[num_freqs]()),
+      sub_running_mean_(new complex<float>[num_freqs]()),
+      sub_running_mean_sq_(new complex<float>[num_freqs]()),
+      variance_(new float[num_freqs]()),
+      conj_sum_(new float[num_freqs]()),
+      num_freqs_(num_freqs),
+      window_size_(window_size),
+      decay_(decay),
+      history_cursor_(0),
+      count_(0),
+      array_mean_(0.0f),
+      buffer_full_(false) {
+  history_.reset(new rtc::scoped_ptr<complex<float>[]>[num_freqs_]());
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    history_[i].reset(new complex<float>[window_size_]());
+  }
+  subhistory_.reset(new rtc::scoped_ptr<complex<float>[]>[num_freqs_]());
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    subhistory_[i].reset(new complex<float>[window_size_]());
+  }
+  subhistory_sq_.reset(new rtc::scoped_ptr<complex<float>[]>[num_freqs_]());
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    subhistory_sq_[i].reset(new complex<float>[window_size_]());
+  }
+  switch (type) {
+    case kStepInfinite:
+      step_func_ = &VarianceArray::InfiniteStep;
+      break;
+    case kStepDecaying:
+      step_func_ = &VarianceArray::DecayStep;
+      break;
+    case kStepWindowed:
+      step_func_ = &VarianceArray::WindowedStep;
+      break;
+    case kStepBlocked:
+      step_func_ = &VarianceArray::BlockedStep;
+      break;
+    case kStepBlockBasedMovingAverage:
+      step_func_ = &VarianceArray::BlockBasedMovingAverage;
+      break;
+  }
+}
+
+// Compute the variance with Welford's algorithm, adding some fudge to
+// the input in case of all-zeroes.
+void VarianceArray::InfiniteStep(const complex<float>* data, bool skip_fudge) {
+  array_mean_ = 0.0f;
+  ++count_;
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    complex<float> sample = data[i];
+    if (!skip_fudge) {
+      sample = zerofudge(sample);
+    }
+    if (count_ == 1) {
+      running_mean_[i] = sample;
+      variance_[i] = 0.0f;
+    } else {
+      float old_sum = conj_sum_[i];
+      complex<float> old_mean = running_mean_[i];
+      running_mean_[i] =
+          old_mean + (sample - old_mean) / static_cast<float>(count_);
+      conj_sum_[i] =
+          (old_sum + std::conj(sample - old_mean) * (sample - running_mean_[i]))
+              .real();
+      variance_[i] =
+          conj_sum_[i] / (count_ - 1);
+    }
+    array_mean_ += (variance_[i] - array_mean_) / (i + 1);
+  }
+}
+
+// Compute the variance from the beginning, with exponential decaying of the
+// series data.
+void VarianceArray::DecayStep(const complex<float>* data, bool /*dummy*/) {
+  array_mean_ = 0.0f;
+  ++count_;
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    complex<float> sample = data[i];
+    sample = zerofudge(sample);
+
+    if (count_ == 1) {
+      running_mean_[i] = sample;
+      running_mean_sq_[i] = sample * std::conj(sample);
+      variance_[i] = 0.0f;
+    } else {
+      complex<float> prev = running_mean_[i];
+      complex<float> prev2 = running_mean_sq_[i];
+      running_mean_[i] = decay_ * prev + (1.0f - decay_) * sample;
+      running_mean_sq_[i] =
+          decay_ * prev2 + (1.0f - decay_) * sample * std::conj(sample);
+      variance_[i] = (running_mean_sq_[i] -
+                      running_mean_[i] * std::conj(running_mean_[i])).real();
+    }
+
+    array_mean_ += (variance_[i] - array_mean_) / (i + 1);
+  }
+}
+
+// Windowed variance computation. On each step, the variances for the
+// window are recomputed from scratch, using Welford's algorithm.
+void VarianceArray::WindowedStep(const complex<float>* data, bool /*dummy*/) {
+  size_t num = min(count_ + 1, window_size_);
+  array_mean_ = 0.0f;
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    complex<float> mean;
+    float conj_sum = 0.0f;
+
+    history_[i][history_cursor_] = data[i];
+
+    mean = history_[i][history_cursor_];
+    variance_[i] = 0.0f;
+    for (size_t j = 1; j < num; ++j) {
+      complex<float> sample =
+          zerofudge(history_[i][(history_cursor_ + j) % window_size_]);
+      sample = history_[i][(history_cursor_ + j) % window_size_];
+      float old_sum = conj_sum;
+      complex<float> old_mean = mean;
+
+      mean = old_mean + (sample - old_mean) / static_cast<float>(j + 1);
+      conj_sum =
+          (old_sum + std::conj(sample - old_mean) * (sample - mean)).real();
+      variance_[i] = conj_sum / (j);
+    }
+    array_mean_ += (variance_[i] - array_mean_) / (i + 1);
+  }
+  history_cursor_ = (history_cursor_ + 1) % window_size_;
+  ++count_;
+}
+
+// Variance with a window of blocks. Within each block, the variances are
+// recomputed from scratch at every stp, using |Var(X) = E(X^2) - E^2(X)|.
+// Once a block is filled with kWindowBlockSize samples, it is added to the
+// history window and a new block is started. The variances for the window
+// are recomputed from scratch at each of these transitions.
+void VarianceArray::BlockedStep(const complex<float>* data, bool /*dummy*/) {
+  size_t blocks = min(window_size_, history_cursor_ + 1);
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    AddToMean(data[i], count_ + 1, &sub_running_mean_[i]);
+    AddToMean(data[i] * std::conj(data[i]), count_ + 1,
+              &sub_running_mean_sq_[i]);
+    subhistory_[i][history_cursor_ % window_size_] = sub_running_mean_[i];
+    subhistory_sq_[i][history_cursor_ % window_size_] = sub_running_mean_sq_[i];
+
+    variance_[i] =
+        (NewMean(running_mean_sq_[i], sub_running_mean_sq_[i], blocks) -
+         NewMean(running_mean_[i], sub_running_mean_[i], blocks) *
+             std::conj(NewMean(running_mean_[i], sub_running_mean_[i], blocks)))
+            .real();
+    if (count_ == kWindowBlockSize - 1) {
+      sub_running_mean_[i] = complex<float>(0.0f, 0.0f);
+      sub_running_mean_sq_[i] = complex<float>(0.0f, 0.0f);
+      running_mean_[i] = complex<float>(0.0f, 0.0f);
+      running_mean_sq_[i] = complex<float>(0.0f, 0.0f);
+      for (size_t j = 0; j < min(window_size_, history_cursor_); ++j) {
+        AddToMean(subhistory_[i][j], j + 1, &running_mean_[i]);
+        AddToMean(subhistory_sq_[i][j], j + 1, &running_mean_sq_[i]);
+      }
+      ++history_cursor_;
+    }
+  }
+  ++count_;
+  if (count_ == kWindowBlockSize) {
+    count_ = 0;
+  }
+}
+
+// Recomputes variances for each window from scratch based on previous window.
+void VarianceArray::BlockBasedMovingAverage(const std::complex<float>* data,
+                                            bool /*dummy*/) {
+  // TODO(ekmeyerson) To mitigate potential divergence, add counter so that
+  // after every so often sums are computed scratch by summing over all
+  // elements instead of subtracting oldest and adding newest.
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    sub_running_mean_[i] += data[i];
+    sub_running_mean_sq_[i] += data[i] * std::conj(data[i]);
+  }
+  ++count_;
+
+  // TODO(ekmeyerson) Make kWindowBlockSize nonconstant to allow
+  // experimentation with different block size,window size pairs.
+  if (count_ >= kWindowBlockSize) {
+    count_ = 0;
+
+    for (size_t i = 0; i < num_freqs_; ++i) {
+      running_mean_[i] -= subhistory_[i][history_cursor_];
+      running_mean_sq_[i] -= subhistory_sq_[i][history_cursor_];
+
+      float scale = 1.f / kWindowBlockSize;
+      subhistory_[i][history_cursor_] = sub_running_mean_[i] * scale;
+      subhistory_sq_[i][history_cursor_] = sub_running_mean_sq_[i] * scale;
+
+      sub_running_mean_[i] = std::complex<float>(0.0f, 0.0f);
+      sub_running_mean_sq_[i] = std::complex<float>(0.0f, 0.0f);
+
+      running_mean_[i] += subhistory_[i][history_cursor_];
+      running_mean_sq_[i] += subhistory_sq_[i][history_cursor_];
+
+      scale = 1.f / (buffer_full_ ? window_size_ : history_cursor_ + 1);
+      variance_[i] = std::real(running_mean_sq_[i] * scale -
+                               running_mean_[i] * scale *
+                                   std::conj(running_mean_[i]) * scale);
+    }
+
+    ++history_cursor_;
+    if (history_cursor_ >= window_size_) {
+      buffer_full_ = true;
+      history_cursor_ = 0;
+    }
+  }
+}
+
+void VarianceArray::Clear() {
+  memset(running_mean_.get(), 0, sizeof(*running_mean_.get()) * num_freqs_);
+  memset(running_mean_sq_.get(), 0,
+         sizeof(*running_mean_sq_.get()) * num_freqs_);
+  memset(variance_.get(), 0, sizeof(*variance_.get()) * num_freqs_);
+  memset(conj_sum_.get(), 0, sizeof(*conj_sum_.get()) * num_freqs_);
+  history_cursor_ = 0;
+  count_ = 0;
+  array_mean_ = 0.0f;
+}
+
+void VarianceArray::ApplyScale(float scale) {
+  array_mean_ = 0.0f;
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    variance_[i] *= scale * scale;
+    array_mean_ += (variance_[i] - array_mean_) / (i + 1);
+  }
+}
+
+GainApplier::GainApplier(size_t freqs, float change_limit)
+    : num_freqs_(freqs),
+      change_limit_(change_limit),
+      target_(new float[freqs]()),
+      current_(new float[freqs]()) {
+  for (size_t i = 0; i < freqs; ++i) {
+    target_[i] = 1.0f;
+    current_[i] = 1.0f;
+  }
+}
+
+void GainApplier::Apply(const complex<float>* in_block,
+                        complex<float>* out_block) {
+  for (size_t i = 0; i < num_freqs_; ++i) {
+    float factor = sqrtf(fabsf(current_[i]));
+    if (!std::isnormal(factor)) {
+      factor = 1.0f;
+    }
+    out_block[i] = factor * in_block[i];
+    current_[i] = UpdateFactor(target_[i], current_[i], change_limit_);
+  }
+}
+
+}  // namespace intelligibility
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h
new file mode 100644
index 0000000000..4ac1167147
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Specifies helper classes for intelligibility enhancement.
+//
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
+
+#include <complex>
+
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+namespace intelligibility {
+
+// Return |current| changed towards |target|, with the change being at most
+// |limit|.
+float UpdateFactor(float target, float current, float limit);
+
+// Apply a small fudge to degenerate complex values. The numbers in the array
+// were chosen randomly, so that even a series of all zeroes has some small
+// variability.
+std::complex<float> zerofudge(std::complex<float> c);
+
+// Incremental mean computation. Return the mean of the series with the
+// mean |mean| with added |data|.
+std::complex<float> NewMean(std::complex<float> mean,
+                            std::complex<float> data,
+                            size_t count);
+
+// Updates |mean| with added |data|;
+void AddToMean(std::complex<float> data,
+               size_t count,
+               std::complex<float>* mean);
+
+// Internal helper for computing the variances of a stream of arrays.
+// The result is an array of variances per position: the i-th variance
+// is the variance of the stream of data on the i-th positions in the
+// input arrays.
+// There are four methods of computation:
+//  * kStepInfinite computes variances from the beginning onwards
+//  * kStepDecaying uses a recursive exponential decay formula with a
+//    settable forgetting factor
+//  * kStepWindowed computes variances within a moving window
+//  * kStepBlocked is similar to kStepWindowed, but history is kept
+//    as a rolling window of blocks: multiple input elements are used for
+//    one block and the history then consists of the variances of these blocks
+//    with the same effect as kStepWindowed, but less storage, so the window
+//    can be longer
+class VarianceArray {
+ public:
+  enum StepType {
+    kStepInfinite = 0,
+    kStepDecaying,
+    kStepWindowed,
+    kStepBlocked,
+    kStepBlockBasedMovingAverage
+  };
+
+  // Construct an instance for the given input array length (|freqs|) and
+  // computation algorithm (|type|), with the appropriate parameters.
+  // |window_size| is the number of samples for kStepWindowed and
+  // the number of blocks for kStepBlocked. |decay| is the forgetting factor
+  // for kStepDecaying.
+  VarianceArray(size_t freqs, StepType type, size_t window_size, float decay);
+
+  // Add a new data point to the series and compute the new variances.
+  // TODO(bercic) |skip_fudge| is a flag for kStepWindowed and kStepDecaying,
+  // whether they should skip adding some small dummy values to the input
+  // to prevent problems with all-zero inputs. Can probably be removed.
+  void Step(const std::complex<float>* data, bool skip_fudge = false) {
+    (this->*step_func_)(data, skip_fudge);
+  }
+  // Reset variances to zero and forget all history.
+  void Clear();
+  // Scale the input data by |scale|. Effectively multiply variances
+  // by |scale^2|.
+  void ApplyScale(float scale);
+
+  // The current set of variances.
+  const float* variance() const { return variance_.get(); }
+
+  // The mean value of the current set of variances.
+  float array_mean() const { return array_mean_; }
+
+ private:
+  void InfiniteStep(const std::complex<float>* data, bool dummy);
+  void DecayStep(const std::complex<float>* data, bool dummy);
+  void WindowedStep(const std::complex<float>* data, bool dummy);
+  void BlockedStep(const std::complex<float>* data, bool dummy);
+  void BlockBasedMovingAverage(const std::complex<float>* data, bool dummy);
+
+  // TODO(ekmeyerson): Switch the following running means
+  // and histories from rtc::scoped_ptr to std::vector.
+
+  // The current average X and X^2.
+  rtc::scoped_ptr<std::complex<float>[]> running_mean_;
+  rtc::scoped_ptr<std::complex<float>[]> running_mean_sq_;
+
+  // Average X and X^2 for the current block in kStepBlocked.
+  rtc::scoped_ptr<std::complex<float>[]> sub_running_mean_;
+  rtc::scoped_ptr<std::complex<float>[]> sub_running_mean_sq_;
+
+  // Sample history for the rolling window in kStepWindowed and block-wise
+  // histories for kStepBlocked.
+  rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> history_;
+  rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> subhistory_;
+  rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> subhistory_sq_;
+
+  // The current set of variances and sums for Welford's algorithm.
+  rtc::scoped_ptr<float[]> variance_;
+  rtc::scoped_ptr<float[]> conj_sum_;
+
+  const size_t num_freqs_;
+  const size_t window_size_;
+  const float decay_;
+  size_t history_cursor_;
+  size_t count_;
+  float array_mean_;
+  bool buffer_full_;
+  void (VarianceArray::*step_func_)(const std::complex<float>*, bool);
+};
+
+// Helper class for smoothing gain changes. On each applicatiion step, the
+// currently used gains are changed towards a set of settable target gains,
+// constrained by a limit on the magnitude of the changes.
+class GainApplier {
+ public:
+  GainApplier(size_t freqs, float change_limit);
+
+  // Copy |in_block| to |out_block|, multiplied by the current set of gains,
+  // and step the current set of gains towards the target set.
+  void Apply(const std::complex<float>* in_block,
+             std::complex<float>* out_block);
+
+  // Return the current target gain set. Modify this array to set the targets.
+  float* target() const { return target_.get(); }
+
+ private:
+  const size_t num_freqs_;
+  const float change_limit_;
+  rtc::scoped_ptr<float[]> target_;
+  rtc::scoped_ptr<float[]> current_;
+};
+
+}  // namespace intelligibility
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils_unittest.cc
new file mode 100644
index 0000000000..9caa2eb0a1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/intelligibility_utils_unittest.cc
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Unit tests for intelligibility utils.
+//
+
+#include <math.h>
+#include <complex>
+#include <iostream>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
+
+using std::complex;
+using std::vector;
+
+namespace webrtc {
+
+namespace intelligibility {
+
+vector<vector<complex<float>>> GenerateTestData(int freqs, int samples) {
+  vector<vector<complex<float>>> data(samples);
+  for (int i = 0; i < samples; i++) {
+    for (int j = 0; j < freqs; j++) {
+      const float val = 0.99f / ((i + 1) * (j + 1));
+      data[i].push_back(complex<float>(val, val));
+    }
+  }
+  return data;
+}
+
+// Tests UpdateFactor.
+TEST(IntelligibilityUtilsTest, TestUpdateFactor) {
+  EXPECT_EQ(0, intelligibility::UpdateFactor(0, 0, 0));
+  EXPECT_EQ(4, intelligibility::UpdateFactor(4, 2, 3));
+  EXPECT_EQ(3, intelligibility::UpdateFactor(4, 2, 1));
+  EXPECT_EQ(2, intelligibility::UpdateFactor(2, 4, 3));
+  EXPECT_EQ(3, intelligibility::UpdateFactor(2, 4, 1));
+}
+
+// Tests zerofudge.
+TEST(IntelligibilityUtilsTest, TestCplx) {
+  complex<float> t0(1.f, 0.f);
+  t0 = intelligibility::zerofudge(t0);
+  EXPECT_NE(t0.imag(), 0.f);
+  EXPECT_NE(t0.real(), 0.f);
+}
+
+// Tests NewMean and AddToMean.
+TEST(IntelligibilityUtilsTest, TestMeanUpdate) {
+  const complex<float> data[] = {{3, 8}, {7, 6}, {2, 1}, {8, 9}, {0, 6}};
+  const complex<float> means[] = {{3, 8}, {5, 7}, {4, 5}, {5, 6}, {4, 6}};
+  complex<float> mean(3, 8);
+  for (size_t i = 0; i < arraysize(data); i++) {
+    EXPECT_EQ(means[i], NewMean(mean, data[i], i + 1));
+    AddToMean(data[i], i + 1, &mean);
+    EXPECT_EQ(means[i], mean);
+  }
+}
+
+// Tests VarianceArray, for all variance step types.
+TEST(IntelligibilityUtilsTest, TestVarianceArray) {
+  const int kFreqs = 10;
+  const int kSamples = 100;
+  const int kWindowSize = 10;  // Should pass for all kWindowSize > 1.
+  const float kDecay = 0.5f;
+  vector<VarianceArray::StepType> step_types;
+  step_types.push_back(VarianceArray::kStepInfinite);
+  step_types.push_back(VarianceArray::kStepDecaying);
+  step_types.push_back(VarianceArray::kStepWindowed);
+  step_types.push_back(VarianceArray::kStepBlocked);
+  step_types.push_back(VarianceArray::kStepBlockBasedMovingAverage);
+  const vector<vector<complex<float>>> test_data(
+      GenerateTestData(kFreqs, kSamples));
+  for (auto step_type : step_types) {
+    VarianceArray variance_array(kFreqs, step_type, kWindowSize, kDecay);
+    EXPECT_EQ(0, variance_array.variance()[0]);
+    EXPECT_EQ(0, variance_array.array_mean());
+    variance_array.ApplyScale(2.0f);
+    EXPECT_EQ(0, variance_array.variance()[0]);
+    EXPECT_EQ(0, variance_array.array_mean());
+
+    // Makes sure Step is doing something.
+    variance_array.Step(&test_data[0][0]);
+    for (int i = 1; i < kSamples; i++) {
+      variance_array.Step(&test_data[i][0]);
+      EXPECT_GE(variance_array.array_mean(), 0.0f);
+      EXPECT_LE(variance_array.array_mean(), 1.0f);
+      for (int j = 0; j < kFreqs; j++) {
+        EXPECT_GE(variance_array.variance()[j], 0.0f);
+        EXPECT_LE(variance_array.variance()[j], 1.0f);
+      }
+    }
+    variance_array.Clear();
+    EXPECT_EQ(0, variance_array.variance()[0]);
+    EXPECT_EQ(0, variance_array.array_mean());
+  }
+}
+
+// Tests exact computation on synthetic data.
+TEST(IntelligibilityUtilsTest, TestMovingBlockAverage) {
+  // Exact, not unbiased estimates.
+  const float kTestVarianceBufferNotFull = 16.5f;
+  const float kTestVarianceBufferFull1 = 66.5f;
+  const float kTestVarianceBufferFull2 = 333.375f;
+  const int kFreqs = 2;
+  const int kSamples = 50;
+  const int kWindowSize = 2;
+  const float kDecay = 0.5f;
+  const float kMaxError = 0.0001f;
+
+  VarianceArray variance_array(
+      kFreqs, VarianceArray::kStepBlockBasedMovingAverage, kWindowSize, kDecay);
+
+  vector<vector<complex<float>>> test_data(kSamples);
+  for (int i = 0; i < kSamples; i++) {
+    for (int j = 0; j < kFreqs; j++) {
+      if (i < 30) {
+        test_data[i].push_back(complex<float>(static_cast<float>(kSamples - i),
+                                              static_cast<float>(i + 1)));
+      } else {
+        test_data[i].push_back(complex<float>(0.f, 0.f));
+      }
+    }
+  }
+
+  for (int i = 0; i < kSamples; i++) {
+    variance_array.Step(&test_data[i][0]);
+    for (int j = 0; j < kFreqs; j++) {
+      if (i < 9) {  // In utils, kWindowBlockSize = 10.
+        EXPECT_EQ(0, variance_array.variance()[j]);
+      } else if (i < 19) {
+        EXPECT_NEAR(kTestVarianceBufferNotFull, variance_array.variance()[j],
+                    kMaxError);
+      } else if (i < 39) {
+        EXPECT_NEAR(kTestVarianceBufferFull1, variance_array.variance()[j],
+                    kMaxError);
+      } else if (i < 49) {
+        EXPECT_NEAR(kTestVarianceBufferFull2, variance_array.variance()[j],
+                    kMaxError);
+      } else {
+        EXPECT_EQ(0, variance_array.variance()[j]);
+      }
+    }
+  }
+}
+
+// Tests gain applier.
+TEST(IntelligibilityUtilsTest, TestGainApplier) {
+  const int kFreqs = 10;
+  const int kSamples = 100;
+  const float kChangeLimit = 0.1f;
+  GainApplier gain_applier(kFreqs, kChangeLimit);
+  const vector<vector<complex<float>>> in_data(
+      GenerateTestData(kFreqs, kSamples));
+  vector<vector<complex<float>>> out_data(GenerateTestData(kFreqs, kSamples));
+  for (int i = 0; i < kSamples; i++) {
+    gain_applier.Apply(&in_data[i][0], &out_data[i][0]);
+    for (int j = 0; j < kFreqs; j++) {
+      EXPECT_GT(out_data[i][j].real(), 0.0f);
+      EXPECT_LT(out_data[i][j].real(), 1.0f);
+      EXPECT_GT(out_data[i][j].imag(), 0.0f);
+      EXPECT_LT(out_data[i][j].imag(), 1.0f);
+    }
+  }
+}
+
+}  // namespace intelligibility
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
new file mode 100644
index 0000000000..4d2f5f4c5d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+//  Command line tool for speech intelligibility enhancement. Provides for
+//  running and testing intelligibility_enhancer as an independent process.
+//  Use --help for options.
+//
+
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <string>
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/real_fourier.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using std::complex;
+using webrtc::intelligibility::VarianceArray;
+
+namespace webrtc {
+namespace {
+
+bool ValidateClearWindow(const char* flagname, int32_t value) {
+  return value > 0;
+}
+
+DEFINE_int32(clear_type,
+             webrtc::intelligibility::VarianceArray::kStepDecaying,
+             "Variance algorithm for clear data.");
+DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
+DEFINE_int32(clear_window,
+             475,
+             "Window size for windowed variance for clear data.");
+const bool clear_window_dummy =
+    google::RegisterFlagValidator(&FLAGS_clear_window, &ValidateClearWindow);
+DEFINE_int32(sample_rate,
+             16000,
+             "Audio sample rate used in the input and output files.");
+DEFINE_int32(ana_rate,
+             800,
+             "Analysis rate; gains recalculated every N blocks.");
+DEFINE_int32(
+    var_rate,
+    2,
+    "Variance clear rate; history is forgotten every N gain recalculations.");
+DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
+
+DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
+DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
+DEFINE_string(out_file,
+              "proc_enhanced.wav",
+              "Enhanced output. Use '-' to "
+              "play through aplay immediately.");
+
+const size_t kNumChannels = 1;
+
+// void function for gtest
+void void_main(int argc, char* argv[]) {
+  google::SetUsageMessage(
+      "\n\nVariance algorithm types are:\n"
+      "  0 - infinite/normal,\n"
+      "  1 - exponentially decaying,\n"
+      "  2 - rolling window.\n"
+      "\nInput files must be little-endian 16-bit signed raw PCM.\n");
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  size_t samples;        // Number of samples in input PCM file
+  size_t fragment_size;  // Number of samples to process at a time
+                         // to simulate APM stream processing
+
+  // Load settings and wav input.
+
+  fragment_size = FLAGS_sample_rate / 100;  // Mirror real time APM chunk size.
+                                            // Duplicates chunk_length_ in
+                                            // IntelligibilityEnhancer.
+
+  struct stat in_stat, noise_stat;
+  ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
+      << "Empty speech file.";
+  ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
+      << "Empty noise file.";
+
+  samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
+
+  WavReader in_file(FLAGS_clear_file);
+  std::vector<float> in_fpcm(samples);
+  in_file.ReadSamples(samples, &in_fpcm[0]);
+
+  WavReader noise_file(FLAGS_noise_file);
+  std::vector<float> noise_fpcm(samples);
+  noise_file.ReadSamples(samples, &noise_fpcm[0]);
+
+  // Run intelligibility enhancement.
+  IntelligibilityEnhancer::Config config;
+  config.sample_rate_hz = FLAGS_sample_rate;
+  config.var_type = static_cast<VarianceArray::StepType>(FLAGS_clear_type);
+  config.var_decay_rate = static_cast<float>(FLAGS_clear_alpha);
+  config.var_window_size = static_cast<size_t>(FLAGS_clear_window);
+  config.analysis_rate = FLAGS_ana_rate;
+  config.gain_change_limit = FLAGS_gain_limit;
+  IntelligibilityEnhancer enh(config);
+
+  // Slice the input into smaller chunks, as the APM would do, and feed them
+  // through the enhancer.
+  float* clear_cursor = &in_fpcm[0];
+  float* noise_cursor = &noise_fpcm[0];
+
+  for (size_t i = 0; i < samples; i += fragment_size) {
+    enh.AnalyzeCaptureAudio(&noise_cursor, FLAGS_sample_rate, kNumChannels);
+    enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels);
+    clear_cursor += fragment_size;
+    noise_cursor += fragment_size;
+  }
+
+  if (FLAGS_out_file.compare("-") == 0) {
+    const std::string temp_out_filename =
+        test::TempFilename(test::WorkingDir(), "temp_wav_file");
+    {
+      WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
+      out_file.WriteSamples(&in_fpcm[0], samples);
+    }
+    system(("aplay " + temp_out_filename).c_str());
+    system(("rm " + temp_out_filename).c_str());
+  } else {
+    WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
+    out_file.WriteSamples(&in_fpcm[0], samples);
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  webrtc::void_main(argc, argv);
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.cc
index 26a61dcdb5..187873e33e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.cc
@@ -11,76 +11,55 @@
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
-LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessing* apm,
-                                       CriticalSectionWrapper* crit)
-    : ProcessingComponent(),
-      crit_(crit) {}
+LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
+    : crit_(crit), rms_(new RMSLevel()) {
+  RTC_DCHECK(crit);
+}
 
 LevelEstimatorImpl::~LevelEstimatorImpl() {}
 
-int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return AudioProcessing::kNoError;
+void LevelEstimatorImpl::Initialize() {
+  rtc::CritScope cs(crit_);
+  rms_->Reset();
+}
+
+void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
   }
 
-  RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
-  for (int i = 0; i < audio->num_channels(); ++i) {
-    rms_level->Process(audio->channels_const()[i],
-                       audio->num_frames());
+  for (size_t i = 0; i < audio->num_channels(); i++) {
+    rms_->Process(audio->channels_const()[i], audio->num_frames());
   }
-
-  return AudioProcessing::kNoError;
 }
 
 int LevelEstimatorImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (enable && !enabled_) {
+    rms_->Reset();
+  }
+  enabled_ = enable;
+  return AudioProcessing::kNoError;
 }
 
 bool LevelEstimatorImpl::is_enabled() const {
-  return is_component_enabled();
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 
 int LevelEstimatorImpl::RMS() {
-  if (!is_component_enabled()) {
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
     return AudioProcessing::kNotEnabledError;
   }
 
-  RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
-  return rms_level->RMS();
+  return rms_->RMS();
 }
-
-// The ProcessingComponent implementation is pretty weird in this class since
-// we have only a single instance of the trivial underlying component.
-void* LevelEstimatorImpl::CreateHandle() const {
-  return new RMSLevel;
-}
-
-void LevelEstimatorImpl::DestroyHandle(void* handle) const {
-  delete static_cast<RMSLevel*>(handle);
-}
-
-int LevelEstimatorImpl::InitializeHandle(void* handle) const {
-  static_cast<RMSLevel*>(handle)->Reset();
-  return AudioProcessing::kNoError;
-}
-
-int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
-  return AudioProcessing::kNoError;
-}
-
-int LevelEstimatorImpl::num_handles_required() const {
-  return 1;
-}
-
-int LevelEstimatorImpl::GetHandleError(void* /*handle*/) const {
-  return AudioProcessing::kUnspecifiedError;
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.h
index 0d0050c7e7..4401da37e4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/level_estimator_impl.h
@@ -11,43 +11,36 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
-#include "webrtc/modules/audio_processing/rms_level.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
+class RMSLevel;
 
-class LevelEstimatorImpl : public LevelEstimator,
-                           public ProcessingComponent {
+class LevelEstimatorImpl : public LevelEstimator {
  public:
-  LevelEstimatorImpl(const AudioProcessing* apm,
-                     CriticalSectionWrapper* crit);
-  virtual ~LevelEstimatorImpl();
+  explicit LevelEstimatorImpl(rtc::CriticalSection* crit);
+  ~LevelEstimatorImpl() override;
 
-  int ProcessStream(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize();
+  void ProcessStream(AudioBuffer* audio);
 
-  // LevelEstimator implementation.
-  bool is_enabled() const override;
-
- private:
   // LevelEstimator implementation.
   int Enable(bool enable) override;
+  bool is_enabled() const override;
   int RMS() override;
 
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  CriticalSectionWrapper* crit_;
+ private:
+  rtc::CriticalSection* const crit_ = nullptr;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  rtc::scoped_ptr<RMSLevel> rms_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LevelEstimatorImpl);
 };
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging.h b/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging.h
new file mode 100644
index 0000000000..464098b683
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
+
+#include <stdio.h>
+
+#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
+
+// To enable AEC logging, invoke GYP with -Daec_debug_dump=1.
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+// Dumps a wav data to file.
+#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
+  do {                                                   \
+    rtc_WavWriteSamples(file, data, num_samples);        \
+  } while (0)
+
+// (Re)opens a wav file for writing using the specified sample rate.
+#define RTC_AEC_DEBUG_WAV_REOPEN(name, instance_index, count,     \
+                                 sample_rate, wav_file)                  \
+  do {                                                                   \
+    WebRtcAec_ReopenWav(name, instance_index, process_rate, sample_rate, \
+                        wav_file);                                       \
+  } while (0)
+
+// Closes a wav file.
+#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
+  do {                                    \
+    rtc_WavClose(wav_file);               \
+  } while (0)
+
+// Dumps a raw data to file.
+#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
+  do {                                                 \
+    (void) fwrite(data, data_size, 1, file);           \
+  } while (0)
+
+// Dumps a raw scalar int32 to file.
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_INT32(file, data)             \
+  do {                                                               \
+    int32_t value_to_store = data;                                   \
+    (void) fwrite(&value_to_store, sizeof(value_to_store), 1, file); \
+  } while (0)
+
+// Dumps a raw scalar double to file.
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_DOUBLE(file, data)            \
+  do {                                                               \
+    double value_to_store = data;                                    \
+    (void) fwrite(&value_to_store, sizeof(value_to_store), 1, file); \
+  } while (0)
+
+// Opens a raw data file for writing using the specified sample rate.
+#define RTC_AEC_DEBUG_RAW_OPEN(name, instance_index, counter, file)     \
+  do {                                                       \
+    WebRtcAec_RawFileOpen(name, instance_index, counter, file); \
+  } while (0)
+
+// Closes a raw data file.
+#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
+  do {                                \
+    if (file) {                       \
+      fclose(file);                   \
+    }                                 \
+  } while (0)
+
+#else  // RTC_AEC_DEBUG_DUMP
+#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
+  do {                                                   \
+  } while (0)
+
+#define RTC_AEC_DEBUG_WAV_REOPEN(wav_file, name, instance_index, process_rate, \
+                                 sample_rate)                                  \
+  do {                                                                         \
+  } while (0)
+
+#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
+  do {                                    \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
+  do {                                                 \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_INT32(file, data) \
+  do {                                                   \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_DOUBLE(file, data) \
+  do {                                                    \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_OPEN(file, name, instance_counter) \
+  do {                                                       \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
+  do {                                \
+  } while (0)
+
+#endif  // WEBRTC_AEC_DEBUG_DUMP
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
new file mode 100644
index 0000000000..13cdc74d97
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
+
+#include <stdint.h>
+#include <stdio.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/stringutils.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/typedefs.h"
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+void WebRtcAec_ReopenWav(const char* name,
+                         int instance_index,
+                         int count,
+                         int sample_rate,
+                         rtc_WavWriter** wav_file) {
+  if (*wav_file) {
+    if (rtc_WavSampleRate(*wav_file) == sample_rate)
+      return;
+    rtc_WavClose(*wav_file);
+  }
+  char filename[64];
+  int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.wav", name,
+                              instance_index, count);
+
+  // Ensure there was no buffer output error.
+  RTC_DCHECK_GE(written, 0);
+  // Ensure that the buffer size was sufficient.
+  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
+
+  *wav_file = rtc_WavOpen(filename, sample_rate, 1);
+}
+
+void WebRtcAec_RawFileOpen(const char* name, int instance_index, int counter, FILE** file) {
+  char filename[64];
+  int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.dat", name,
+                              instance_index, counter);
+
+  // Ensure there was no buffer output error.
+  RTC_DCHECK_GE(written, 0);
+  // Ensure that the buffer size was sufficient.
+  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
+
+  *file = fopen(filename, "wb");
+}
+
+#endif  // WEBRTC_AEC_DEBUG_DUMP
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h b/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
new file mode 100644
index 0000000000..c1ce0a0d43
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
+
+#include <stdio.h>
+
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/typedefs.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+// Opens a new Wav file for writing. If it was already open with a different
+// sample frequency, it closes it first.
+void WebRtcAec_ReopenWav(const char* name,
+                         int instance_index,
+                         int count,
+                         int sample_rate,
+                         rtc_WavWriter** wav_file);
+
+// Opens dumpfile with instance-specific filename.
+void WebRtcAec_RawFileOpen(const char* name, int instance_index, int counter, FILE** file);
+
+#endif  // WEBRTC_AEC_DEBUG_DUMP
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.cc
index aa37e67fa8..de7e856676 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -10,180 +10,166 @@
 
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 
-#include <assert.h>
-
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #if defined(WEBRTC_NS_FLOAT)
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
+#define NS_CREATE WebRtcNs_Create
+#define NS_FREE WebRtcNs_Free
+#define NS_INIT WebRtcNs_Init
+#define NS_SET_POLICY WebRtcNs_set_policy
+typedef NsHandle NsState;
 #elif defined(WEBRTC_NS_FIXED)
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
+#define NS_CREATE WebRtcNsx_Create
+#define NS_FREE WebRtcNsx_Free
+#define NS_INIT WebRtcNsx_Init
+#define NS_SET_POLICY WebRtcNsx_set_policy
+typedef NsxHandle NsState;
 #endif
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-
 
 namespace webrtc {
-
-#if defined(WEBRTC_NS_FLOAT)
-typedef NsHandle Handle;
-#elif defined(WEBRTC_NS_FIXED)
-typedef NsxHandle Handle;
-#endif
-
-namespace {
-int MapSetting(NoiseSuppression::Level level) {
-  switch (level) {
-    case NoiseSuppression::kLow:
-      return 0;
-    case NoiseSuppression::kModerate:
-      return 1;
-    case NoiseSuppression::kHigh:
-      return 2;
-    case NoiseSuppression::kVeryHigh:
-      return 3;
+class NoiseSuppressionImpl::Suppressor {
+ public:
+  explicit Suppressor(int sample_rate_hz) {
+    state_ = NS_CREATE();
+    RTC_CHECK(state_);
+    int error = NS_INIT(state_, sample_rate_hz);
+    RTC_DCHECK_EQ(0, error);
   }
-  assert(false);
-  return -1;
-}
-}  // namespace
+  ~Suppressor() {
+    NS_FREE(state_);
+  }
+  NsState* state() { return state_; }
+ private:
+  NsState* state_ = nullptr;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
+};
 
-NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessing* apm,
-                                           CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    level_(kModerate) {}
+NoiseSuppressionImpl::NoiseSuppressionImpl(rtc::CriticalSection* crit)
+    : crit_(crit) {
+  RTC_DCHECK(crit);
+}
 
 NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
 
-int NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
-#if defined(WEBRTC_NS_FLOAT)
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
+  rtc::CritScope cs(crit_);
+  channels_ = channels;
+  sample_rate_hz_ = sample_rate_hz;
+  std::vector<rtc::scoped_ptr<Suppressor>> new_suppressors;
+  if (enabled_) {
+    new_suppressors.resize(channels);
+    for (size_t i = 0; i < channels; i++) {
+      new_suppressors[i].reset(new Suppressor(sample_rate_hz));
+    }
   }
-  assert(audio->num_frames_per_band() <= 160);
-  assert(audio->num_channels() == num_handles());
-
-  for (int i = 0; i < num_handles(); ++i) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-
-    WebRtcNs_Analyze(my_handle, audio->split_bands_const_f(i)[kBand0To8kHz]);
-  }
-#endif
-  return apm_->kNoError;
+  suppressors_.swap(new_suppressors);
+  set_level(level_);
 }
 
-int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-  assert(audio->num_frames_per_band() <= 160);
-  assert(audio->num_channels() == num_handles());
-
-  for (int i = 0; i < num_handles(); ++i) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
+void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
 #if defined(WEBRTC_NS_FLOAT)
-    WebRtcNs_Process(my_handle,
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
+
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
+  for (size_t i = 0; i < suppressors_.size(); i++) {
+    WebRtcNs_Analyze(suppressors_[i]->state(),
+                     audio->split_bands_const_f(i)[kBand0To8kHz]);
+  }
+#endif
+}
+
+void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  RTC_DCHECK(audio);
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
+
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
+  for (size_t i = 0; i < suppressors_.size(); i++) {
+#if defined(WEBRTC_NS_FLOAT)
+    WebRtcNs_Process(suppressors_[i]->state(),
                      audio->split_bands_const_f(i),
                      audio->num_bands(),
                      audio->split_bands_f(i));
 #elif defined(WEBRTC_NS_FIXED)
-    WebRtcNsx_Process(my_handle,
+    WebRtcNsx_Process(suppressors_[i]->state(),
                       audio->split_bands_const(i),
                       audio->num_bands(),
                       audio->split_bands(i));
 #endif
   }
-  return apm_->kNoError;
 }
 
 int NoiseSuppressionImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (enabled_ != enable) {
+    enabled_ = enable;
+    Initialize(channels_, sample_rate_hz_);
+  }
+  return AudioProcessing::kNoError;
 }
 
 bool NoiseSuppressionImpl::is_enabled() const {
-  return is_component_enabled();
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 
 int NoiseSuppressionImpl::set_level(Level level) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
+  int policy = 1;
+  switch (level) {
+    case NoiseSuppression::kLow:
+      policy = 0;
+      break;
+    case NoiseSuppression::kModerate:
+      policy = 1;
+      break;
+    case NoiseSuppression::kHigh:
+      policy = 2;
+      break;
+    case NoiseSuppression::kVeryHigh:
+      policy = 3;
+      break;
+    default:
+      RTC_NOTREACHED();
   }
-
+  rtc::CritScope cs(crit_);
   level_ = level;
-  return Configure();
+  for (auto& suppressor : suppressors_) {
+    int error = NS_SET_POLICY(suppressor->state(), policy);
+    RTC_DCHECK_EQ(0, error);
+  }
+  return AudioProcessing::kNoError;
 }
 
 NoiseSuppression::Level NoiseSuppressionImpl::level() const {
+  rtc::CritScope cs(crit_);
   return level_;
 }
 
 float NoiseSuppressionImpl::speech_probability() const {
+  rtc::CritScope cs(crit_);
 #if defined(WEBRTC_NS_FLOAT)
   float probability_average = 0.0f;
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    probability_average += WebRtcNs_prior_speech_probability(my_handle);
+  for (auto& suppressor : suppressors_) {
+    probability_average +=
+        WebRtcNs_prior_speech_probability(suppressor->state());
   }
-  return probability_average / num_handles();
+  if (!suppressors_.empty()) {
+    probability_average /= suppressors_.size();
+  }
+  return probability_average;
 #elif defined(WEBRTC_NS_FIXED)
+  // TODO(peah): Returning error code as a float! Remove this.
   // Currently not available for the fixed point implementation.
-  return apm_->kUnsupportedFunctionError;
+  return AudioProcessing::kUnsupportedFunctionError;
 #endif
 }
-
-void* NoiseSuppressionImpl::CreateHandle() const {
-  Handle* handle = NULL;
-#if defined(WEBRTC_NS_FLOAT)
-  if (WebRtcNs_Create(&handle) != apm_->kNoError)
-#elif defined(WEBRTC_NS_FIXED)
-  if (WebRtcNsx_Create(&handle) != apm_->kNoError)
-#endif
-  {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-void NoiseSuppressionImpl::DestroyHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  WebRtcNs_Free(static_cast<Handle*>(handle));
-#elif defined(WEBRTC_NS_FIXED)
-  WebRtcNsx_Free(static_cast<Handle*>(handle));
-#endif
-}
-
-int NoiseSuppressionImpl::InitializeHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_Init(static_cast<Handle*>(handle),
-                       apm_->proc_sample_rate_hz());
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_Init(static_cast<Handle*>(handle),
-                        apm_->proc_sample_rate_hz());
-#endif
-}
-
-int NoiseSuppressionImpl::ConfigureHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_set_policy(static_cast<Handle*>(handle),
-                             MapSetting(level_));
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_set_policy(static_cast<Handle*>(handle),
-                              MapSetting(level_));
-#endif
-}
-
-int NoiseSuppressionImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int NoiseSuppressionImpl::GetHandleError(void* handle) const {
-  // The NS has no get_error() function.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.h
index 33a0e060a6..debbc61bc9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/noise_suppression_impl.h
@@ -11,47 +11,42 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
-class NoiseSuppressionImpl : public NoiseSuppression,
-                             public ProcessingComponent {
+class NoiseSuppressionImpl : public NoiseSuppression {
  public:
-  NoiseSuppressionImpl(const AudioProcessing* apm,
-                       CriticalSectionWrapper* crit);
-  virtual ~NoiseSuppressionImpl();
+  explicit NoiseSuppressionImpl(rtc::CriticalSection* crit);
+  ~NoiseSuppressionImpl() override;
 
-  int AnalyzeCaptureAudio(AudioBuffer* audio);
-  int ProcessCaptureAudio(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize(size_t channels, int sample_rate_hz);
+  void AnalyzeCaptureAudio(AudioBuffer* audio);
+  void ProcessCaptureAudio(AudioBuffer* audio);
 
   // NoiseSuppression implementation.
+  int Enable(bool enable) override;
   bool is_enabled() const override;
+  int set_level(Level level) override;
+  Level level() const override;
   float speech_probability() const override;
 
  private:
-  // NoiseSuppression implementation.
-  int Enable(bool enable) override;
-  int set_level(Level level) override;
-  Level level() const override;
-
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  Level level_;
+  class Suppressor;
+  rtc::CriticalSection* const crit_;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  Level level_ GUARDED_BY(crit_) = kModerate;
+  size_t channels_ GUARDED_BY(crit_) = 0;
+  int sample_rate_hz_ GUARDED_BY(crit_) = 0;
+  std::vector<rtc::scoped_ptr<Suppressor>> suppressors_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSuppressionImpl);
 };
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.c
index bae0f2e1e7..dd05e0ab3d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
 
 #include <stdlib.h>
 #include <string.h>
@@ -17,23 +17,16 @@
 #include "webrtc/modules/audio_processing/ns/defines.h"
 #include "webrtc/modules/audio_processing/ns/ns_core.h"
 
-int WebRtcNs_Create(NsHandle** NS_inst) {
-  *NS_inst = (NsHandle*)malloc(sizeof(NoiseSuppressionC));
-  if (*NS_inst != NULL) {
-    (*(NoiseSuppressionC**)NS_inst)->initFlag = 0;
-    return 0;
-  } else {
-    return -1;
-  }
-
+NsHandle* WebRtcNs_Create() {
+  NoiseSuppressionC* self = malloc(sizeof(NoiseSuppressionC));
+  self->initFlag = 0;
+  return (NsHandle*)self;
 }
 
-int WebRtcNs_Free(NsHandle* NS_inst) {
+void WebRtcNs_Free(NsHandle* NS_inst) {
   free(NS_inst);
-  return 0;
 }
 
-
 int WebRtcNs_Init(NsHandle* NS_inst, uint32_t fs) {
   return WebRtcNs_InitCore((NoiseSuppressionC*)NS_inst, fs);
 }
@@ -48,7 +41,7 @@ void WebRtcNs_Analyze(NsHandle* NS_inst, const float* spframe) {
 
 void WebRtcNs_Process(NsHandle* NS_inst,
                       const float* const* spframe,
-                      int num_bands,
+                      size_t num_bands,
                       float* const* outframe) {
   WebRtcNs_ProcessCore((NoiseSuppressionC*)NS_inst, spframe, num_bands,
                        outframe);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression.h b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.h
similarity index 79%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.h
index d912f7112c..8018118b60 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
+
+#include <stddef.h>
 
 #include "webrtc/typedefs.h"
 
@@ -20,20 +22,9 @@ extern "C" {
 #endif
 
 /*
- * This function creates an instance to the noise suppression structure
- *
- * Input:
- *      - NS_inst       : Pointer to noise suppression instance that should be
- *                        created
- *
- * Output:
- *      - NS_inst       : Pointer to created noise suppression instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
+ * This function creates an instance of the floating point Noise Suppression.
  */
-int WebRtcNs_Create(NsHandle** NS_inst);
-
+NsHandle* WebRtcNs_Create();
 
 /*
  * This function frees the dynamic memory of a specified noise suppression
@@ -41,12 +32,8 @@ int WebRtcNs_Create(NsHandle** NS_inst);
  *
  * Input:
  *      - NS_inst       : Pointer to NS instance that should be freed
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
  */
-int WebRtcNs_Free(NsHandle* NS_inst);
-
+void WebRtcNs_Free(NsHandle* NS_inst);
 
 /*
  * This function initializes a NS instance and has to be called before any other
@@ -107,7 +94,7 @@ void WebRtcNs_Analyze(NsHandle* NS_inst, const float* spframe);
  */
 void WebRtcNs_Process(NsHandle* NS_inst,
                      const float* const* spframe,
-                     int num_bands,
+                     size_t num_bands,
                      float* const* outframe);
 
 /* Returns the internally used prior speech probability of the current frame.
@@ -126,4 +113,4 @@ float WebRtcNs_prior_speech_probability(NsHandle* handle);
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.c
index 920b50100d..0a5ba13300 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 
 #include <stdlib.h>
 
@@ -16,25 +16,17 @@
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/modules/audio_processing/ns/nsx_defines.h"
 
-int WebRtcNsx_Create(NsxHandle** nsxInst) {
+NsxHandle* WebRtcNsx_Create() {
   NoiseSuppressionFixedC* self = malloc(sizeof(NoiseSuppressionFixedC));
-  *nsxInst = (NsxHandle*)self;
-
-  if (self != NULL) {
-    WebRtcSpl_Init();
-    self->real_fft = NULL;
-    self->initFlag = 0;
-    return 0;
-  } else {
-    return -1;
-  }
-
+  WebRtcSpl_Init();
+  self->real_fft = NULL;
+  self->initFlag = 0;
+  return (NsxHandle*)self;
 }
 
-int WebRtcNsx_Free(NsxHandle* nsxInst) {
+void WebRtcNsx_Free(NsxHandle* nsxInst) {
   WebRtcSpl_FreeRealFFT(((NoiseSuppressionFixedC*)nsxInst)->real_fft);
   free(nsxInst);
-  return 0;
 }
 
 int WebRtcNsx_Init(NsxHandle* nsxInst, uint32_t fs) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.h
index e1671a60a2..02b44cc091 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/noise_suppression_x.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
 
 #include "webrtc/typedefs.h"
 
@@ -20,20 +20,9 @@ extern "C" {
 #endif
 
 /*
- * This function creates an instance to the noise reduction structure
- *
- * Input:
- *      - nsxInst       : Pointer to noise reduction instance that should be
- *                       created
- *
- * Output:
- *      - nsxInst       : Pointer to created noise reduction instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
+ * This function creates an instance of the fixed point Noise Suppression.
  */
-int WebRtcNsx_Create(NsxHandle** nsxInst);
-
+NsxHandle* WebRtcNsx_Create();
 
 /*
  * This function frees the dynamic memory of a specified Noise Suppression
@@ -41,12 +30,8 @@ int WebRtcNsx_Create(NsxHandle** nsxInst);
  *
  * Input:
  *      - nsxInst       : Pointer to NS instance that should be freed
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
  */
-int WebRtcNsx_Free(NsxHandle* nsxInst);
-
+void WebRtcNsx_Free(NsxHandle* nsxInst);
 
 /*
  * This function initializes a NS instance
@@ -100,4 +85,4 @@ void WebRtcNsx_Process(NsxHandle* nsxInst,
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.c
index 9e230dd140..5ce64cee29 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.c
@@ -15,7 +15,7 @@
 
 #include "webrtc/common_audio/fft4g.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression.h"
 #include "webrtc/modules/audio_processing/ns/ns_core.h"
 #include "webrtc/modules/audio_processing/ns/windows_private.h"
 
@@ -217,7 +217,7 @@ int WebRtcNs_InitCore(NoiseSuppressionC* self, uint32_t fs) {
 static void NoiseEstimation(NoiseSuppressionC* self,
                             float* magn,
                             float* noise) {
-  int i, s, offset;
+  size_t i, s, offset;
   float lmagn[HALF_ANAL_BLOCKL], delta;
 
   if (self->updates < END_STARTUP_LONG) {
@@ -522,8 +522,8 @@ static void FeatureParameterExtraction(NoiseSuppressionC* self, int flag) {
 // Spectral flatness is returned in self->featureData[0].
 static void ComputeSpectralFlatness(NoiseSuppressionC* self,
                                     const float* magnIn) {
-  int i;
-  int shiftLP = 1;  // Option to remove first bin(s) from spectral measures.
+  size_t i;
+  size_t shiftLP = 1;  // Option to remove first bin(s) from spectral measures.
   float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
 
   // Compute spectral measures.
@@ -568,7 +568,7 @@ static void ComputeSnr(const NoiseSuppressionC* self,
                        const float* noise,
                        float* snrLocPrior,
                        float* snrLocPost) {
-  int i;
+  size_t i;
 
   for (i = 0; i < self->magnLen; i++) {
     // Previous post SNR.
@@ -596,7 +596,7 @@ static void ComputeSpectralDifference(NoiseSuppressionC* self,
                                       const float* magnIn) {
   // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 /
   // var(magnAvgPause)
-  int i;
+  size_t i;
   float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
 
   avgPause = 0.0;
@@ -606,8 +606,8 @@ static void ComputeSpectralDifference(NoiseSuppressionC* self,
     // Conservative smooth noise spectrum from pause frames.
     avgPause += self->magnAvgPause[i];
   }
-  avgPause = avgPause / ((float)self->magnLen);
-  avgMagn = avgMagn / ((float)self->magnLen);
+  avgPause /= self->magnLen;
+  avgMagn /= self->magnLen;
 
   covMagnPause = 0.0;
   varPause = 0.0;
@@ -619,9 +619,9 @@ static void ComputeSpectralDifference(NoiseSuppressionC* self,
         (self->magnAvgPause[i] - avgPause) * (self->magnAvgPause[i] - avgPause);
     varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
   }
-  covMagnPause = covMagnPause / ((float)self->magnLen);
-  varPause = varPause / ((float)self->magnLen);
-  varMagn = varMagn / ((float)self->magnLen);
+  covMagnPause /= self->magnLen;
+  varPause /= self->magnLen;
+  varMagn /= self->magnLen;
   // Update of average magnitude spectrum.
   self->featureData[6] += self->signalEnergy;
 
@@ -643,7 +643,8 @@ static void SpeechNoiseProb(NoiseSuppressionC* self,
                             float* probSpeechFinal,
                             const float* snrLocPrior,
                             const float* snrLocPost) {
-  int i, sgnMap;
+  size_t i;
+  int sgnMap;
   float invLrt, gainPrior, indPrior;
   float logLrtTimeAvgKsum, besselTmp;
   float indicator0, indicator1, indicator2;
@@ -802,7 +803,7 @@ static void UpdateNoiseEstimate(NoiseSuppressionC* self,
                                 const float* snrLocPrior,
                                 const float* snrLocPost,
                                 float* noise) {
-  int i;
+  size_t i;
   float probSpeech, probNonSpeech;
   // Time-avg parameter for noise update.
   float gammaNoiseTmp = NOISE_UPDATE;
@@ -853,8 +854,8 @@ static void UpdateNoiseEstimate(NoiseSuppressionC* self,
 // Output:
 //   * |buffer| is the updated buffer.
 static void UpdateBuffer(const float* frame,
-                         int frame_length,
-                         int buffer_length,
+                         size_t frame_length,
+                         size_t buffer_length,
                          float* buffer) {
   assert(buffer_length < 2 * frame_length);
 
@@ -885,12 +886,12 @@ static void UpdateBuffer(const float* frame,
 //   * |magn| is the calculated signal magnitude in the frequency domain.
 static void FFT(NoiseSuppressionC* self,
                 float* time_data,
-                int time_data_length,
-                int magnitude_length,
+                size_t time_data_length,
+                size_t magnitude_length,
                 float* real,
                 float* imag,
                 float* magn) {
-  int i;
+  size_t i;
 
   assert(magnitude_length == time_data_length / 2 + 1);
 
@@ -898,10 +899,10 @@ static void FFT(NoiseSuppressionC* self,
 
   imag[0] = 0;
   real[0] = time_data[0];
-  magn[0] = fabs(real[0]) + 1.f;
+  magn[0] = fabsf(real[0]) + 1.f;
   imag[magnitude_length - 1] = 0;
   real[magnitude_length - 1] = time_data[1];
-  magn[magnitude_length - 1] = fabs(real[magnitude_length - 1]) + 1.f;
+  magn[magnitude_length - 1] = fabsf(real[magnitude_length - 1]) + 1.f;
   for (i = 1; i < magnitude_length - 1; ++i) {
     real[i] = time_data[2 * i];
     imag[i] = time_data[2 * i + 1];
@@ -923,10 +924,10 @@ static void FFT(NoiseSuppressionC* self,
 static void IFFT(NoiseSuppressionC* self,
                  const float* real,
                  const float* imag,
-                 int magnitude_length,
-                 int time_data_length,
+                 size_t magnitude_length,
+                 size_t time_data_length,
                  float* time_data) {
-  int i;
+  size_t i;
 
   assert(time_data_length == 2 * (magnitude_length - 1));
 
@@ -948,8 +949,8 @@ static void IFFT(NoiseSuppressionC* self,
 //   * |buffer| is the buffer over which the energy is calculated.
 //   * |length| is the length of the buffer.
 // Returns the calculated energy.
-static float Energy(const float* buffer, int length) {
-  int i;
+static float Energy(const float* buffer, size_t length) {
+  size_t i;
   float energy = 0.f;
 
   for (i = 0; i < length; ++i) {
@@ -968,9 +969,9 @@ static float Energy(const float* buffer, int length) {
 //   * |data_windowed| is the windowed data.
 static void Windowing(const float* window,
                       const float* data,
-                      int length,
+                      size_t length,
                       float* data_windowed) {
-  int i;
+  size_t i;
 
   for (i = 0; i < length; ++i) {
     data_windowed[i] = window[i] * data[i];
@@ -985,7 +986,7 @@ static void Windowing(const float* window,
 static void ComputeDdBasedWienerFilter(const NoiseSuppressionC* self,
                                        const float* magn,
                                        float* theFilter) {
-  int i;
+  size_t i;
   float snrPrior, previousEstimateStsa, currentEstimateStsa;
 
   for (i = 0; i < self->magnLen; i++) {
@@ -1041,8 +1042,8 @@ int WebRtcNs_set_policy_core(NoiseSuppressionC* self, int mode) {
 }
 
 void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
-  int i;
-  const int kStartBand = 5;  // Skip first frequency bins during estimation.
+  size_t i;
+  const size_t kStartBand = 5;  // Skip first frequency bins during estimation.
   int updateParsFlag;
   float energy;
   float signalEnergy = 0.f;
@@ -1090,16 +1091,16 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
     sumMagn += magn[i];
     if (self->blockInd < END_STARTUP_SHORT) {
       if (i >= kStartBand) {
-        tmpFloat2 = log((float)i);
+        tmpFloat2 = logf((float)i);
         sum_log_i += tmpFloat2;
         sum_log_i_square += tmpFloat2 * tmpFloat2;
-        tmpFloat1 = log(magn[i]);
+        tmpFloat1 = logf(magn[i]);
         sum_log_magn += tmpFloat1;
         sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
       }
     }
   }
-  signalEnergy = signalEnergy / ((float)self->magnLen);
+  signalEnergy /= self->magnLen;
   self->signalEnergy = signalEnergy;
   self->sumMagn = sumMagn;
 
@@ -1108,9 +1109,9 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
   // Compute simplified noise model during startup.
   if (self->blockInd < END_STARTUP_SHORT) {
     // Estimate White noise.
-    self->whiteNoiseLevel += sumMagn / ((float)self->magnLen) * self->overdrive;
+    self->whiteNoiseLevel += sumMagn / self->magnLen * self->overdrive;
     // Estimate Pink noise parameters.
-    tmpFloat1 = sum_log_i_square * ((float)(self->magnLen - kStartBand));
+    tmpFloat1 = sum_log_i_square * (self->magnLen - kStartBand);
     tmpFloat1 -= (sum_log_i * sum_log_i);
     tmpFloat2 =
         (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
@@ -1121,7 +1122,7 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
     }
     self->pinkNoiseNumerator += tmpFloat3;
     tmpFloat2 = (sum_log_i * sum_log_magn);
-    tmpFloat2 -= ((float)(self->magnLen - kStartBand)) * sum_log_i_log_magn;
+    tmpFloat2 -= (self->magnLen - kStartBand) * sum_log_i_log_magn;
     tmpFloat3 = tmpFloat2 / tmpFloat1;
     // Constrain the pink noise power to be in the interval [0, 1].
     if (tmpFloat3 < 0.f) {
@@ -1136,7 +1137,7 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
     if (self->pinkNoiseExp > 0.f) {
       // Use pink noise estimate.
       parametric_num =
-          exp(self->pinkNoiseNumerator / (float)(self->blockInd + 1));
+          expf(self->pinkNoiseNumerator / (float)(self->blockInd + 1));
       parametric_num *= (float)(self->blockInd + 1);
       parametric_exp = self->pinkNoiseExp / (float)(self->blockInd + 1);
     }
@@ -1150,7 +1151,7 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
         // Use pink noise estimate.
         float use_band = (float)(i < kStartBand ? kStartBand : i);
         self->parametricNoise[i] =
-            parametric_num / pow(use_band, parametric_exp);
+            parametric_num / powf(use_band, parametric_exp);
       }
       // Weight quantile noise with modeled noise.
       noise[i] *= (self->blockInd);
@@ -1182,11 +1183,11 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
 
 void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
                           const float* const* speechFrame,
-                          int num_bands,
+                          size_t num_bands,
                           float* const* outFrame) {
   // Main routine for noise reduction.
   int flagHB = 0;
-  int i, j;
+  size_t i, j;
 
   float energy1, energy2, gain, factor, factor1, factor2;
   float fout[BLOCKL_MAX];
@@ -1210,7 +1211,7 @@ void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
 
   const float* const* speechFrameHB = NULL;
   float* const* outFrameHB = NULL;
-  int num_high_bands = 0;
+  size_t num_high_bands = 0;
   if (num_bands > 1) {
     speechFrameHB = &speechFrame[1];
     outFrameHB = &outFrame[1];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.h b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.h
index 8a7992ec5a..aba1c468ed 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/ns_core.h
@@ -51,10 +51,10 @@ typedef struct NSParaExtract_ {
 
 typedef struct NoiseSuppressionC_ {
   uint32_t fs;
-  int blockLen;
-  int windShift;
-  int anaLen;
-  int magnLen;
+  size_t blockLen;
+  size_t windShift;
+  size_t anaLen;
+  size_t magnLen;
   int aggrMode;
   const float* window;
   float analyzeBuf[ANAL_BLOCKL_MAX];
@@ -74,7 +74,7 @@ typedef struct NoiseSuppressionC_ {
   float denoiseBound;
   int gainmap;
   // FFT work arrays.
-  int ip[IP_LENGTH];
+  size_t ip[IP_LENGTH];
   float wfft[W_LENGTH];
 
   // Parameters for new method: some not needed, will reduce/cleanup later.
@@ -181,7 +181,7 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame);
  */
 void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
                           const float* const* inFrame,
-                          int num_bands,
+                          size_t num_bands,
                           float* const* outFrame);
 
 #ifdef __cplusplus
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.c
index 6faaebac42..25f16d26ab 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 
 #include <assert.h>
 #include <math.h>
@@ -17,9 +17,9 @@
 
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
-#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON)
+#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
 /* Tables are defined in ARM assembly files. */
 extern const int16_t WebRtcNsx_kLogTable[9];
 extern const int16_t WebRtcNsx_kCounterDiv[201];
@@ -65,10 +65,10 @@ static const int16_t WebRtcNsx_kLogTableFrac[256] = {
   237, 238, 238, 239, 240, 241, 241, 242, 243, 244, 244, 245, 246, 247, 247,
   248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
 };
-#endif  // WEBRTC_DETECT_ARM_NEON || WEBRTC_ARCH_ARM_NEON
+#endif  // WEBRTC_DETECT_NEON || WEBRTC_HAS_NEON
 
 // Skip first frequency bins during estimation. (0 <= value < 64)
-static const int kStartBand = 5;
+static const size_t kStartBand = 5;
 
 // hybrib Hanning & flat window
 static const int16_t kBlocks80w128x[128] = {
@@ -306,7 +306,7 @@ static void UpdateNoiseEstimate(NoiseSuppressionFixedC* inst, int offset) {
   int16_t tmp16 = 0;
   const int16_t kExp2Const = 11819; // Q13
 
-  int i = 0;
+  size_t i = 0;
 
   tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
                                    inst->magnLen);
@@ -341,7 +341,7 @@ static void NoiseEstimationC(NoiseSuppressionFixedC* inst,
   const int16_t log2_const = 22713; // Q15
   const int16_t width_factor = 21845;
 
-  int i, s, offset;
+  size_t i, s, offset;
 
   tabind = inst->stages - inst->normData;
   assert(tabind < 9);
@@ -454,7 +454,7 @@ static void NoiseEstimationC(NoiseSuppressionFixedC* inst,
 
 // Filter the data in the frequency domain, and create spectrum.
 static void PrepareSpectrumC(NoiseSuppressionFixedC* inst, int16_t* freq_buf) {
-  int i = 0, j = 0;
+  size_t i = 0, j = 0;
 
   for (i = 0; i < inst->magnLen; i++) {
     inst->real[i] = (int16_t)((inst->real[i] *
@@ -477,7 +477,7 @@ static void PrepareSpectrumC(NoiseSuppressionFixedC* inst, int16_t* freq_buf) {
 static void DenormalizeC(NoiseSuppressionFixedC* inst,
                          int16_t* in,
                          int factor) {
-  int i = 0;
+  size_t i = 0;
   int32_t tmp32 = 0;
   for (i = 0; i < inst->anaLen; i += 1) {
     tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[i],
@@ -491,7 +491,7 @@ static void DenormalizeC(NoiseSuppressionFixedC* inst,
 static void SynthesisUpdateC(NoiseSuppressionFixedC* inst,
                              int16_t* out_frame,
                              int16_t gain_factor) {
-  int i = 0;
+  size_t i = 0;
   int16_t tmp16a = 0;
   int16_t tmp16b = 0;
   int32_t tmp32 = 0;
@@ -513,9 +513,8 @@ static void SynthesisUpdateC(NoiseSuppressionFixedC* inst,
   }
 
   // update synthesis buffer
-  WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
-                        inst->synthesisBuffer + inst->blockLen10ms,
-                        inst->anaLen - inst->blockLen10ms);
+  memcpy(inst->synthesisBuffer, inst->synthesisBuffer + inst->blockLen10ms,
+      (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->synthesisBuffer));
   WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
       + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
 }
@@ -524,14 +523,13 @@ static void SynthesisUpdateC(NoiseSuppressionFixedC* inst,
 static void AnalysisUpdateC(NoiseSuppressionFixedC* inst,
                             int16_t* out,
                             int16_t* new_speech) {
-  int i = 0;
+  size_t i = 0;
 
   // For lower band update analysis buffer.
-  WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
-                        inst->analysisBuffer + inst->blockLen10ms,
-                        inst->anaLen - inst->blockLen10ms);
-  WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
-      + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+  memcpy(inst->analysisBuffer, inst->analysisBuffer + inst->blockLen10ms,
+      (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->analysisBuffer));
+  memcpy(inst->analysisBuffer + inst->anaLen - inst->blockLen10ms, new_speech,
+      inst->blockLen10ms * sizeof(*inst->analysisBuffer));
 
   // Window data before FFT.
   for (i = 0; i < inst->anaLen; i++) {
@@ -544,7 +542,7 @@ static void AnalysisUpdateC(NoiseSuppressionFixedC* inst,
 static void NormalizeRealBufferC(NoiseSuppressionFixedC* inst,
                                  const int16_t* in,
                                  int16_t* out) {
-  int i = 0;
+  size_t i = 0;
   assert(inst->normData >= 0);
   for (i = 0; i < inst->anaLen; ++i) {
     out[i] = in[i] << inst->normData;  // Q(normData)
@@ -559,8 +557,7 @@ AnalysisUpdate WebRtcNsx_AnalysisUpdate;
 Denormalize WebRtcNsx_Denormalize;
 NormalizeRealBuffer WebRtcNsx_NormalizeRealBuffer;
 
-#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON || \
-     defined WEBRTC_ARCH_ARM64_NEON)
+#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
 // Initialize function pointers for ARM Neon platform.
 static void WebRtcNsx_InitNeon(void) {
   WebRtcNsx_NoiseEstimation = WebRtcNsx_NoiseEstimationNeon;
@@ -765,12 +762,12 @@ int32_t WebRtcNsx_InitCore(NoiseSuppressionFixedC* inst, uint32_t fs) {
   WebRtcNsx_Denormalize = DenormalizeC;
   WebRtcNsx_NormalizeRealBuffer = NormalizeRealBufferC;
 
-#ifdef WEBRTC_DETECT_ARM_NEON
+#ifdef WEBRTC_DETECT_NEON
   uint64_t features = WebRtc_GetCPUFeaturesARM();
   if ((features & kCPUFeatureNEON) != 0) {
       WebRtcNsx_InitNeon();
   }
-#elif defined(WEBRTC_ARCH_ARM_NEON) || defined(WEBRTC_ARCH_ARM64_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   WebRtcNsx_InitNeon();
 #endif
 
@@ -1029,7 +1026,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NoiseSuppressionFixedC* inst,
 
   int16_t zeros, frac, intPart;
 
-  int i;
+  size_t i;
 
   // for flatness
   avgSpectralFlatnessNum = 0;
@@ -1102,7 +1099,8 @@ void WebRtcNsx_ComputeSpectralDifference(NoiseSuppressionFixedC* inst,
 
   int16_t tmp16no1;
 
-  int i, norm32, nShifts;
+  size_t i;
+  int norm32, nShifts;
 
   avgPauseFX = 0;
   maxPause = 0;
@@ -1201,7 +1199,7 @@ void WebRtcNsx_DataAnalysis(NoiseSuppressionFixedC* inst,
   int16_t   matrix_determinant = 0;
   int16_t   maxWinData;
 
-  int i, j;
+  size_t i, j;
   int zeros;
   int net_norm = 0;
   int right_shifts_in_magnU16 = 0;
@@ -1218,7 +1216,8 @@ void WebRtcNsx_DataAnalysis(NoiseSuppressionFixedC* inst,
   WebRtcNsx_AnalysisUpdate(inst, winData, speechFrame);
 
   // Get input energy
-  inst->energyIn = WebRtcSpl_Energy(winData, (int)inst->anaLen, &(inst->scaleEnergyIn));
+  inst->energyIn =
+      WebRtcSpl_Energy(winData, inst->anaLen, &inst->scaleEnergyIn);
 
   // Reset zero input flag
   inst->zeroInputSignal = 0;
@@ -1432,7 +1431,7 @@ void WebRtcNsx_DataSynthesis(NoiseSuppressionFixedC* inst, short* outFrame) {
   int16_t energyRatio;
   int16_t gainFactor, gainFactor1, gainFactor2;
 
-  int i;
+  size_t i;
   int outCIFFT;
   int scaleEnergyOut = 0;
 
@@ -1443,9 +1442,8 @@ void WebRtcNsx_DataSynthesis(NoiseSuppressionFixedC* inst, short* outFrame) {
       outFrame[i] = inst->synthesisBuffer[i]; // Q0
     }
     // update synthesis buffer
-    WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
-                          inst->synthesisBuffer + inst->blockLen10ms,
-                          inst->anaLen - inst->blockLen10ms);
+    memcpy(inst->synthesisBuffer, inst->synthesisBuffer + inst->blockLen10ms,
+        (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->synthesisBuffer));
     WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
                             inst->blockLen10ms);
     return;
@@ -1464,7 +1462,8 @@ void WebRtcNsx_DataSynthesis(NoiseSuppressionFixedC* inst, short* outFrame) {
   if (inst->gainMap == 1 &&
       inst->blockIndex > END_STARTUP_LONG &&
       inst->energyIn > 0) {
-    energyOut = WebRtcSpl_Energy(inst->real, (int)inst->anaLen, &scaleEnergyOut); // Q(-scaleEnergyOut)
+    // Q(-scaleEnergyOut)
+    energyOut = WebRtcSpl_Energy(inst->real, inst->anaLen, &scaleEnergyOut);
     if (scaleEnergyOut == 0 && !(energyOut & 0x7f800000)) {
       energyOut = WEBRTC_SPL_SHIFT_W32(energyOut, 8 + scaleEnergyOut
                                        - inst->scaleEnergyIn);
@@ -1533,7 +1532,7 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
   int16_t avgProbSpeechHB, gainModHB, avgFilterGainHB, gainTimeDomainHB;
   int16_t pink_noise_exp_avg = 0;
 
-  int i, j;
+  size_t i, j;
   int nShifts, postShifts;
   int norm32no1, norm32no2;
   int flag, sign;
@@ -1561,11 +1560,11 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
 
   const short* const* speechFrameHB = NULL;
   short* const* outFrameHB = NULL;
-  int num_high_bands = 0;
+  size_t num_high_bands = 0;
   if (num_bands > 1) {
     speechFrameHB = &speechFrame[1];
     outFrameHB = &outFrame[1];
-    num_high_bands = num_bands - 1;
+    num_high_bands = (size_t)(num_bands - 1);
   }
 
   // Store speechFrame and transform to frequency domain
@@ -1578,13 +1577,11 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
       // update analysis buffer for H band
       // append new data to buffer FX
       for (i = 0; i < num_high_bands; ++i) {
-        WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX[i],
-                              inst->dataBufHBFX[i] + inst->blockLen10ms,
-                              inst->anaLen - inst->blockLen10ms);
-        WEBRTC_SPL_MEMCPY_W16(
-            inst->dataBufHBFX[i] + inst->anaLen - inst->blockLen10ms,
-            speechFrameHB[i],
-            inst->blockLen10ms);
+        int block_shift = inst->anaLen - inst->blockLen10ms;
+        memcpy(inst->dataBufHBFX[i], inst->dataBufHBFX[i] + inst->blockLen10ms,
+            block_shift * sizeof(*inst->dataBufHBFX[i]));
+        memcpy(inst->dataBufHBFX[i] + block_shift, speechFrameHB[i],
+            inst->blockLen10ms * sizeof(*inst->dataBufHBFX[i]));
         for (j = 0; j < inst->blockLen10ms; j++) {
           outFrameHB[i][j] = inst->dataBufHBFX[i][j]; // Q0
         }
@@ -2043,13 +2040,10 @@ void WebRtcNsx_ProcessCore(NoiseSuppressionFixedC* inst,
     // update analysis buffer for H band
     // append new data to buffer FX
     for (i = 0; i < num_high_bands; ++i) {
-      WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX[i],
-                            inst->dataBufHBFX[i] + inst->blockLen10ms,
-                            inst->anaLen - inst->blockLen10ms);
-      WEBRTC_SPL_MEMCPY_W16(
-          inst->dataBufHBFX[i] + inst->anaLen - inst->blockLen10ms,
-          speechFrameHB[i],
-          inst->blockLen10ms);
+      memcpy(inst->dataBufHBFX[i], inst->dataBufHBFX[i] + inst->blockLen10ms,
+          (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->dataBufHBFX[i]));
+      memcpy(inst->dataBufHBFX[i] + inst->anaLen - inst->blockLen10ms,
+          speechFrameHB[i], inst->blockLen10ms * sizeof(*inst->dataBufHBFX[i]));
     }
     // range for averaging low band quantities for H band gain
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.h b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.h
index 8f0db6d016..f463dbbe1a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core.h
@@ -34,9 +34,9 @@ typedef struct NoiseSuppressionFixedC_ {
   int16_t                 noiseEstCounter[SIMULT];
   int16_t                 noiseEstQuantile[HALF_ANAL_BLOCKL];
 
-  int                     anaLen;
-  int                     anaLen2;
-  int                     magnLen;
+  size_t                  anaLen;
+  size_t                  anaLen2;
+  size_t                  magnLen;
   int                     aggrMode;
   int                     stages;
   int                     initFlag;
@@ -98,7 +98,7 @@ typedef struct NoiseSuppressionFixedC_ {
   int                     qNoise;
   int                     prevQNoise;
   int                     prevQMagn;
-  int                     blockLen10ms;
+  size_t                  blockLen10ms;
 
   int16_t                 real[ANAL_BLOCKL_MAX];
   int16_t                 imag[ANAL_BLOCKL_MAX];
@@ -215,8 +215,7 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
                                uint32_t* priorLocSnr,
                                uint32_t* postLocSnr);
 
-#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON || \
-     defined WEBRTC_ARCH_ARM64_NEON)
+#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
 // For the above function pointers, functions for generic platforms are declared
 // and defined as static in file nsx_core.c, while those for ARM Neon platforms
 // are declared below and defined in file nsx_core_neon.c.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_c.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_c.c
index 9c929d1865..da7aa3d5db 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -10,7 +10,7 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/modules/audio_processing/ns/nsx_defines.h"
 
@@ -33,7 +33,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
   int32_t logLrtTimeAvgKsumFX;
   int16_t indPriorFX16;
   int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
-  int i, normTmp, normTmp2, nShifts;
+  size_t i;
+  int normTmp, normTmp2, nShifts;
 
   // compute feature based on average LR factor
   // this is the average over all frequencies of the smooth log LRT
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_mips.c
index 6c29a04c32..7688d82d78 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@@ -9,8 +9,9 @@
  */
 
 #include <assert.h>
+#include <string.h>
 
-#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/noise_suppression_x.h"
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 
 static const int16_t kIndicatorTable[17] = {
@@ -31,7 +32,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
   int32_t logLrtTimeAvgKsumFX;
   int16_t indPriorFX16;
   int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac;
-  int i, normTmp, nShifts;
+  size_t i;
+  int normTmp, nShifts;
 
   int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9;
   int32_t const_max = 0x7fffffff;
@@ -330,7 +332,7 @@ void WebRtcNsx_AnalysisUpdate_mips(NoiseSuppressionFixedC* inst,
                                    int16_t* out,
                                    int16_t* new_speech) {
   int iters, after;
-  int anaLen = inst->anaLen;
+  int anaLen = (int)inst->anaLen;
   int *window = (int*)inst->window;
   int *anaBuf = (int*)inst->analysisBuffer;
   int *outBuf = (int*)out;
@@ -340,11 +342,10 @@ void WebRtcNsx_AnalysisUpdate_mips(NoiseSuppressionFixedC* inst,
 #endif
 
   // For lower band update analysis buffer.
-  WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
-                        inst->analysisBuffer + inst->blockLen10ms,
-                        inst->anaLen - inst->blockLen10ms);
-  WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
-      + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+  memcpy(inst->analysisBuffer, inst->analysisBuffer + inst->blockLen10ms,
+      (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->analysisBuffer));
+  memcpy(inst->analysisBuffer + inst->anaLen - inst->blockLen10ms, new_speech,
+      inst->blockLen10ms * sizeof(*inst->analysisBuffer));
 
   // Window data before FFT.
 #if defined(MIPS_DSP_R1_LE)
@@ -504,7 +505,7 @@ void WebRtcNsx_AnalysisUpdate_mips(NoiseSuppressionFixedC* inst,
 void WebRtcNsx_SynthesisUpdate_mips(NoiseSuppressionFixedC* inst,
                                     int16_t* out_frame,
                                     int16_t gain_factor) {
-  int iters = inst->blockLen10ms >> 2;
+  int iters = (int)inst->blockLen10ms >> 2;
   int after = inst->blockLen10ms & 3;
   int r0, r1, r2, r3, r4, r5, r6, r7;
   int16_t *window = (int16_t*)inst->window;
@@ -744,9 +745,8 @@ void WebRtcNsx_SynthesisUpdate_mips(NoiseSuppressionFixedC* inst,
   );
 
   // update synthesis buffer
-  WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
-                        inst->synthesisBuffer + inst->blockLen10ms,
-                        inst->anaLen - inst->blockLen10ms);
+  memcpy(inst->synthesisBuffer, inst->synthesisBuffer + inst->blockLen10ms,
+      (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->synthesisBuffer));
   WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
       + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
 }
@@ -759,7 +759,7 @@ void WebRtcNsx_PrepareSpectrum_mips(NoiseSuppressionFixedC* inst,
   int16_t *imag = inst->imag;
   int32_t loop_count = 2;
   int16_t tmp_1, tmp_2, tmp_3, tmp_4, tmp_5, tmp_6;
-  int16_t tmp16 = (inst->anaLen << 1) - 4;
+  int16_t tmp16 = (int16_t)(inst->anaLen << 1) - 4;
   int16_t* freq_buf_f = freq_buf;
   int16_t* freq_buf_s = &freq_buf[tmp16];
 
@@ -862,7 +862,7 @@ void WebRtcNsx_Denormalize_mips(NoiseSuppressionFixedC* inst,
                                 int16_t* in,
                                 int factor) {
   int32_t r0, r1, r2, r3, t0;
-  int len = inst->anaLen;
+  int len = (int)inst->anaLen;
   int16_t *out = &inst->real[0];
   int shift = factor - inst->normData;
 
@@ -952,7 +952,7 @@ void WebRtcNsx_NormalizeRealBuffer_mips(NoiseSuppressionFixedC* inst,
                                         const int16_t* in,
                                         int16_t* out) {
   int32_t r0, r1, r2, r3, t0;
-  int len = inst->anaLen;
+  int len = (int)inst->anaLen;
   int shift = inst->normData;
 
   __asm __volatile (
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_neon.c b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_neon.c
index ed735e8b8b..65788ae230 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_neon.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/ns/nsx_core_neon.c
@@ -141,7 +141,7 @@ void WebRtcNsx_NoiseEstimationNeon(NoiseSuppressionFixedC* inst,
   const int16_t log2_const = 22713;
   const int16_t width_factor = 21845;
 
-  int i, s, offset;
+  size_t i, s, offset;
 
   tabind = inst->stages - inst->normData;
   assert(tabind < 9);
@@ -208,7 +208,7 @@ void WebRtcNsx_NoiseEstimationNeon(NoiseSuppressionFixedC* inst,
     uint16x8_t tmp16x8_4;
     int32x4_t tmp32x4;
 
-    for (i = 0; i < inst->magnLen - 7; i += 8) {
+    for (i = 0; i + 7 < inst->magnLen; i += 8) {
       // Compute delta.
       // Smaller step size during startup. This prevents from using
       // unrealistic values causing overflow.
@@ -541,9 +541,8 @@ void WebRtcNsx_AnalysisUpdateNeon(NoiseSuppressionFixedC* inst,
   assert(inst->anaLen % 16 == 0);
 
   // For lower band update analysis buffer.
-  // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
-  //                      inst->analysisBuffer + inst->blockLen10ms,
-  //                      inst->anaLen - inst->blockLen10ms);
+  // memcpy(inst->analysisBuffer, inst->analysisBuffer + inst->blockLen10ms,
+  //     (inst->anaLen - inst->blockLen10ms) * sizeof(*inst->analysisBuffer));
   int16_t* p_start_src = inst->analysisBuffer + inst->blockLen10ms;
   int16_t* p_end_src = inst->analysisBuffer + inst->anaLen;
   int16_t* p_start_dst = inst->analysisBuffer;
@@ -555,8 +554,8 @@ void WebRtcNsx_AnalysisUpdateNeon(NoiseSuppressionFixedC* inst,
     p_start_dst += 8;
   }
 
-  // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
-  //    + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
+  // memcpy(inst->analysisBuffer + inst->anaLen - inst->blockLen10ms,
+  //     new_speech, inst->blockLen10ms * sizeof(*inst->analysisBuffer));
   p_start_src = new_speech;
   p_end_src = new_speech + inst->blockLen10ms;
   p_start_dst = inst->analysisBuffer + inst->anaLen - inst->blockLen10ms;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.cc
index 9e16d7c4ee..7abd8e2100 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.cc
@@ -55,12 +55,12 @@ bool ProcessingComponent::is_component_enabled() const {
   return enabled_;
 }
 
-void* ProcessingComponent::handle(int index) const {
+void* ProcessingComponent::handle(size_t index) const {
   assert(index < num_handles_);
   return handles_[index];
 }
 
-int ProcessingComponent::num_handles() const {
+size_t ProcessingComponent::num_handles() const {
   return num_handles_;
 }
 
@@ -70,12 +70,12 @@ int ProcessingComponent::Initialize() {
   }
 
   num_handles_ = num_handles_required();
-  if (num_handles_ > static_cast<int>(handles_.size())) {
+  if (num_handles_ > handles_.size()) {
     handles_.resize(num_handles_, NULL);
   }
 
-  assert(static_cast<int>(handles_.size()) >= num_handles_);
-  for (int i = 0; i < num_handles_; i++) {
+  assert(handles_.size() >= num_handles_);
+  for (size_t i = 0; i < num_handles_; i++) {
     if (handles_[i] == NULL) {
       handles_[i] = CreateHandle();
       if (handles_[i] == NULL) {
@@ -98,8 +98,8 @@ int ProcessingComponent::Configure() {
     return AudioProcessing::kNoError;
   }
 
-  assert(static_cast<int>(handles_.size()) >= num_handles_);
-  for (int i = 0; i < num_handles_; i++) {
+  assert(handles_.size() >= num_handles_);
+  for (size_t i = 0; i < num_handles_; i++) {
     int err = ConfigureHandle(handles_[i]);
     if (err != AudioProcessing::kNoError) {
       return GetHandleError(handles_[i]);
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.h b/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.h
index 8ee3ac6c7d..577f1570ad 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/processing_component.h
@@ -17,6 +17,22 @@
 
 namespace webrtc {
 
+// Functor to use when supplying a verifier function for the queue item
+// verifcation.
+template <typename T>
+class RenderQueueItemVerifier {
+ public:
+  explicit RenderQueueItemVerifier(size_t minimum_capacity)
+      : minimum_capacity_(minimum_capacity) {}
+
+  bool operator()(const std::vector<T>& v) const {
+    return v.capacity() >= minimum_capacity_;
+  }
+
+ private:
+  size_t minimum_capacity_;
+};
+
 class ProcessingComponent {
  public:
   ProcessingComponent();
@@ -31,21 +47,21 @@ class ProcessingComponent {
  protected:
   virtual int Configure();
   int EnableComponent(bool enable);
-  void* handle(int index) const;
-  int num_handles() const;
+  void* handle(size_t index) const;
+  size_t num_handles() const;
 
  private:
   virtual void* CreateHandle() const = 0;
   virtual int InitializeHandle(void* handle) const = 0;
   virtual int ConfigureHandle(void* handle) const = 0;
   virtual void DestroyHandle(void* handle) const = 0;
-  virtual int num_handles_required() const = 0;
+  virtual size_t num_handles_required() const = 0;
   virtual int GetHandleError(void* handle) const = 0;
 
   std::vector<void*> handles_;
   bool initialized_;
   bool enabled_;
-  int num_handles_;
+  size_t num_handles_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.cc
index 14136bf304..70c4422d34 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.cc
@@ -28,14 +28,14 @@ void RMSLevel::Reset() {
   sample_count_ = 0;
 }
 
-void RMSLevel::Process(const int16_t* data, int length) {
-  for (int i = 0; i < length; ++i) {
+void RMSLevel::Process(const int16_t* data, size_t length) {
+  for (size_t i = 0; i < length; ++i) {
     sum_square_ += data[i] * data[i];
   }
   sample_count_ += length;
 }
 
-void RMSLevel::ProcessMuted(int length) {
+void RMSLevel::ProcessMuted(size_t length) {
   sample_count_ += length;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.h b/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.h
index 055d271bb1..12fa2125f0 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
 
+#include <cstddef>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -35,11 +37,11 @@ class RMSLevel {
   void Reset();
 
   // Pass each chunk of audio to Process() to accumulate the level.
-  void Process(const int16_t* data, int length);
+  void Process(const int16_t* data, size_t length);
 
   // If all samples with the given |length| have a magnitude of zero, this is
   // a shortcut to avoid some computation.
-  void ProcessMuted(int length);
+  void ProcessMuted(size_t length);
 
   // Computes the RMS level over all data passed to Process() since the last
   // call to RMS(). The returned value is positive but should be interpreted as
@@ -48,7 +50,7 @@ class RMSLevel {
 
  private:
   float sum_square_;
-  int sample_count_;
+  size_t sample_count_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.cc
index 623bb05891..46cc9352c2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.cc
@@ -11,32 +11,31 @@
 #include "webrtc/modules/audio_processing/splitting_filter.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/channel_buffer.h"
 
 namespace webrtc {
 
-SplittingFilter::SplittingFilter(int channels)
-    : channels_(channels),
-      two_bands_states_(new TwoBandsStates[channels]),
-      band1_states_(new TwoBandsStates[channels]),
-      band2_states_(new TwoBandsStates[channels]) {
-  for (int i = 0; i < channels; ++i) {
-    analysis_resamplers_.push_back(new PushSincResampler(
-        kSamplesPer48kHzChannel, kSamplesPer64kHzChannel));
-    synthesis_resamplers_.push_back(new PushSincResampler(
-        kSamplesPer64kHzChannel, kSamplesPer48kHzChannel));
+SplittingFilter::SplittingFilter(size_t num_channels,
+                                 size_t num_bands,
+                                 size_t num_frames)
+    : num_bands_(num_bands) {
+  RTC_CHECK(num_bands_ == 2 || num_bands_ == 3);
+  if (num_bands_ == 2) {
+    two_bands_states_.resize(num_channels);
+  } else if (num_bands_ == 3) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      three_band_filter_banks_.push_back(new ThreeBandFilterBank(num_frames));
+    }
   }
 }
 
 void SplittingFilter::Analysis(const IFChannelBuffer* data,
                                IFChannelBuffer* bands) {
-  DCHECK(bands->num_bands() == 2 || bands->num_bands() == 3);
-  DCHECK_EQ(channels_, data->num_channels());
-  DCHECK_EQ(channels_, bands->num_channels());
-  DCHECK_EQ(data->num_frames(),
-            bands->num_frames_per_band() * bands->num_bands());
+  RTC_DCHECK_EQ(num_bands_, bands->num_bands());
+  RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
+  RTC_DCHECK_EQ(data->num_frames(),
+                bands->num_frames_per_band() * bands->num_bands());
   if (bands->num_bands() == 2) {
     TwoBandsAnalysis(data, bands);
   } else if (bands->num_bands() == 3) {
@@ -46,11 +45,10 @@ void SplittingFilter::Analysis(const IFChannelBuffer* data,
 
 void SplittingFilter::Synthesis(const IFChannelBuffer* bands,
                                 IFChannelBuffer* data) {
-  DCHECK(bands->num_bands() == 2 || bands->num_bands() == 3);
-  DCHECK_EQ(channels_, data->num_channels());
-  DCHECK_EQ(channels_, bands->num_channels());
-  DCHECK_EQ(data->num_frames(),
-            bands->num_frames_per_band() * bands->num_bands());
+  RTC_DCHECK_EQ(num_bands_, bands->num_bands());
+  RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
+  RTC_DCHECK_EQ(data->num_frames(),
+                bands->num_frames_per_band() * bands->num_bands());
   if (bands->num_bands() == 2) {
     TwoBandsSynthesis(bands, data);
   } else if (bands->num_bands() == 3) {
@@ -60,7 +58,8 @@ void SplittingFilter::Synthesis(const IFChannelBuffer* bands,
 
 void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data,
                                        IFChannelBuffer* bands) {
-  for (int i = 0; i < channels_; ++i) {
+  RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
+  for (size_t i = 0; i < two_bands_states_.size(); ++i) {
     WebRtcSpl_AnalysisQMF(data->ibuf_const()->channels()[i],
                           data->num_frames(),
                           bands->ibuf()->channels(0)[i],
@@ -72,7 +71,8 @@ void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data,
 
 void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands,
                                         IFChannelBuffer* data) {
-  for (int i = 0; i < channels_; ++i) {
+  RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
+  for (size_t i = 0; i < two_bands_states_.size(); ++i) {
     WebRtcSpl_SynthesisQMF(bands->ibuf_const()->channels(0)[i],
                            bands->ibuf_const()->channels(1)[i],
                            bands->num_frames_per_band(),
@@ -82,82 +82,23 @@ void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands,
   }
 }
 
-// This is a simple implementation using the existing code and will be replaced
-// by a proper 3 band filter bank.
-// It up-samples from 48kHz to 64kHz, splits twice into 2 bands and discards the
-// uppermost band, because it is empty anyway.
 void SplittingFilter::ThreeBandsAnalysis(const IFChannelBuffer* data,
                                          IFChannelBuffer* bands) {
-  DCHECK_EQ(kSamplesPer48kHzChannel,
-            data->num_frames());
-  InitBuffers();
-  for (int i = 0; i < channels_; ++i) {
-    analysis_resamplers_[i]->Resample(data->ibuf_const()->channels()[i],
-                                      kSamplesPer48kHzChannel,
-                                      int_buffer_.get(),
-                                      kSamplesPer64kHzChannel);
-    WebRtcSpl_AnalysisQMF(int_buffer_.get(),
-                          kSamplesPer64kHzChannel,
-                          int_buffer_.get(),
-                          int_buffer_.get() + kSamplesPer32kHzChannel,
-                          two_bands_states_[i].analysis_state1,
-                          two_bands_states_[i].analysis_state2);
-    WebRtcSpl_AnalysisQMF(int_buffer_.get(),
-                          kSamplesPer32kHzChannel,
-                          bands->ibuf()->channels(0)[i],
-                          bands->ibuf()->channels(1)[i],
-                          band1_states_[i].analysis_state1,
-                          band1_states_[i].analysis_state2);
-    WebRtcSpl_AnalysisQMF(int_buffer_.get() + kSamplesPer32kHzChannel,
-                          kSamplesPer32kHzChannel,
-                          int_buffer_.get(),
-                          bands->ibuf()->channels(2)[i],
-                          band2_states_[i].analysis_state1,
-                          band2_states_[i].analysis_state2);
+  RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
+  for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
+    three_band_filter_banks_[i]->Analysis(data->fbuf_const()->channels()[i],
+                                          data->num_frames(),
+                                          bands->fbuf()->bands(i));
   }
 }
 
-// This is a simple implementation using the existing code and will be replaced
-// by a proper 3 band filter bank.
-// Using an empty uppermost band, it merges the 4 bands in 2 steps and
-// down-samples from 64kHz to 48kHz.
 void SplittingFilter::ThreeBandsSynthesis(const IFChannelBuffer* bands,
                                           IFChannelBuffer* data) {
-  DCHECK_EQ(kSamplesPer48kHzChannel,
-            data->num_frames());
-  InitBuffers();
-  for (int i = 0; i < channels_; ++i) {
-    memset(int_buffer_.get(),
-           0,
-           kSamplesPer64kHzChannel * sizeof(int_buffer_[0]));
-    WebRtcSpl_SynthesisQMF(bands->ibuf_const()->channels(0)[i],
-                           bands->ibuf_const()->channels(1)[i],
-                           kSamplesPer16kHzChannel,
-                           int_buffer_.get(),
-                           band1_states_[i].synthesis_state1,
-                           band1_states_[i].synthesis_state2);
-    WebRtcSpl_SynthesisQMF(int_buffer_.get() + kSamplesPer32kHzChannel,
-                           bands->ibuf_const()->channels(2)[i],
-                           kSamplesPer16kHzChannel,
-                           int_buffer_.get() + kSamplesPer32kHzChannel,
-                           band2_states_[i].synthesis_state1,
-                           band2_states_[i].synthesis_state2);
-    WebRtcSpl_SynthesisQMF(int_buffer_.get(),
-                           int_buffer_.get() + kSamplesPer32kHzChannel,
-                           kSamplesPer32kHzChannel,
-                           int_buffer_.get(),
-                           two_bands_states_[i].synthesis_state1,
-                           two_bands_states_[i].synthesis_state2);
-    synthesis_resamplers_[i]->Resample(int_buffer_.get(),
-                                       kSamplesPer64kHzChannel,
-                                       data->ibuf()->channels()[i],
-                                       kSamplesPer48kHzChannel);
-  }
-}
-
-void SplittingFilter::InitBuffers() {
-  if (!int_buffer_) {
-    int_buffer_.reset(new int16_t[kSamplesPer64kHzChannel]);
+  RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
+  for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
+    three_band_filter_banks_[i]->Synthesis(bands->fbuf_const()->bands(i),
+                                           bands->num_frames_per_band(),
+                                           data->fbuf()->channels()[i]);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.h b/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.h
index 8df5310f05..6b81c2fb05 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter.h
@@ -11,25 +11,16 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_SPLITTING_FILTER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_SPLITTING_FILTER_H_
 
-#include <string.h>
+#include <cstring>
+#include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
-#include "webrtc/typedefs.h"
+#include "webrtc/modules/audio_processing/three_band_filter_bank.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
 
 namespace webrtc {
 
 class IFChannelBuffer;
 
-enum {
-  kSamplesPer8kHzChannel = 80,
-  kSamplesPer16kHzChannel = 160,
-  kSamplesPer32kHzChannel = 320,
-  kSamplesPer48kHzChannel = 480,
-  kSamplesPer64kHzChannel = 640
-};
-
 struct TwoBandsStates {
   TwoBandsStates() {
     memset(analysis_state1, 0, sizeof(analysis_state1));
@@ -54,27 +45,22 @@ struct TwoBandsStates {
 // used.
 class SplittingFilter {
  public:
-  SplittingFilter(int channels);
+  SplittingFilter(size_t num_channels, size_t num_bands, size_t num_frames);
 
   void Analysis(const IFChannelBuffer* data, IFChannelBuffer* bands);
   void Synthesis(const IFChannelBuffer* bands, IFChannelBuffer* data);
 
  private:
-  // These work for 640 samples or less.
+  // Two-band analysis and synthesis work for 640 samples or less.
   void TwoBandsAnalysis(const IFChannelBuffer* data, IFChannelBuffer* bands);
   void TwoBandsSynthesis(const IFChannelBuffer* bands, IFChannelBuffer* data);
-  // These only work for 480 samples at the moment.
   void ThreeBandsAnalysis(const IFChannelBuffer* data, IFChannelBuffer* bands);
   void ThreeBandsSynthesis(const IFChannelBuffer* bands, IFChannelBuffer* data);
   void InitBuffers();
 
-  int channels_;
-  rtc::scoped_ptr<TwoBandsStates[]> two_bands_states_;
-  rtc::scoped_ptr<TwoBandsStates[]> band1_states_;
-  rtc::scoped_ptr<TwoBandsStates[]> band2_states_;
-  ScopedVector<PushSincResampler> analysis_resamplers_;
-  ScopedVector<PushSincResampler> synthesis_resamplers_;
-  rtc::scoped_ptr<int16_t[]> int_buffer_;
+  const size_t num_bands_;
+  std::vector<TwoBandsStates> two_bands_states_;
+  ScopedVector<ThreeBandFilterBank> three_band_filter_banks_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter_unittest.cc
index 598057f8c2..e7af65115c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/splitting_filter_unittest.cc
@@ -11,14 +11,19 @@
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
 
-#include <math.h>
+#include <cmath>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
-#include "webrtc/common_audio/include/audio_util.h"
 
 namespace webrtc {
+namespace {
+
+const size_t kSamplesPer16kHzChannel = 160;
+const size_t kSamplesPer48kHzChannel = 480;
+
+}  // namespace
 
 // Generates a signal from presence or absence of sine waves of different
 // frequencies.
@@ -30,37 +35,40 @@ namespace webrtc {
 TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
   static const int kChannels = 1;
   static const int kSampleRateHz = 48000;
-  static const int kNumBands = 3;
+  static const size_t kNumBands = 3;
   static const int kFrequenciesHz[kNumBands] = {1000, 12000, 18000};
-  static const float kAmplitude = 8192;
-  static const int kChunks = 8;
-  SplittingFilter splitting_filter(kChannels);
+  static const float kAmplitude = 8192.f;
+  static const size_t kChunks = 8;
+  SplittingFilter splitting_filter(kChannels,
+                                   kNumBands,
+                                   kSamplesPer48kHzChannel);
   IFChannelBuffer in_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
+  IFChannelBuffer bands(kSamplesPer48kHzChannel, kChannels, kNumBands);
   IFChannelBuffer out_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
-  for (int i = 0; i < kChunks; ++i) {
+  for (size_t i = 0; i < kChunks; ++i) {
     // Input signal generation.
     bool is_present[kNumBands];
     memset(in_data.fbuf()->channels()[0],
            0,
            kSamplesPer48kHzChannel * sizeof(in_data.fbuf()->channels()[0][0]));
-    for (int j = 0; j < kNumBands; ++j) {
-      is_present[j] = i & (1 << j);
-      float amplitude = is_present[j] ? kAmplitude : 0;
-      for (int k = 0; k < kSamplesPer48kHzChannel; ++k) {
+    for (size_t j = 0; j < kNumBands; ++j) {
+      is_present[j] = i & (static_cast<size_t>(1) << j);
+      float amplitude = is_present[j] ? kAmplitude : 0.f;
+      for (size_t k = 0; k < kSamplesPer48kHzChannel; ++k) {
         in_data.fbuf()->channels()[0][k] +=
-            amplitude * sin(2 * M_PI * kFrequenciesHz[j] *
+            amplitude * sin(2.f * M_PI * kFrequenciesHz[j] *
                 (i * kSamplesPer48kHzChannel + k) / kSampleRateHz);
       }
     }
     // Three band splitting filter.
-    splitting_filter.Analysis(&in_data, &out_data);
+    splitting_filter.Analysis(&in_data, &bands);
     // Energy calculation.
     float energy[kNumBands];
-    for (int j = 0; j < kNumBands; ++j) {
-      energy[j] = 0;
-      for (int k = 0; k < kSamplesPer16kHzChannel; ++k) {
-        energy[j] += out_data.fbuf_const()->channels(j)[0][k] *
-                     out_data.fbuf_const()->channels(j)[0][k];
+    for (size_t j = 0; j < kNumBands; ++j) {
+      energy[j] = 0.f;
+      for (size_t k = 0; k < kSamplesPer16kHzChannel; ++k) {
+        energy[j] += bands.fbuf_const()->channels(j)[0][k] *
+                     bands.fbuf_const()->channels(j)[0][k];
       }
       energy[j] /= kSamplesPer16kHzChannel;
       if (is_present[j]) {
@@ -70,14 +78,14 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
       }
     }
     // Three band merge.
-    splitting_filter.Synthesis(&out_data, &out_data);
+    splitting_filter.Synthesis(&bands, &out_data);
     // Delay and cross correlation estimation.
-    float xcorr = 0;
-    for (int delay = 0; delay < kSamplesPer48kHzChannel; ++delay) {
-      float tmpcorr = 0;
-      for (int j = delay; j < kSamplesPer48kHzChannel; ++j) {
-        tmpcorr += in_data.fbuf_const()->channels()[0][j] *
-                   out_data.fbuf_const()->channels()[0][j - delay];
+    float xcorr = 0.f;
+    for (size_t delay = 0; delay < kSamplesPer48kHzChannel; ++delay) {
+      float tmpcorr = 0.f;
+      for (size_t j = delay; j < kSamplesPer48kHzChannel; ++j) {
+        tmpcorr += in_data.fbuf_const()->channels()[0][j - delay] *
+                   out_data.fbuf_const()->channels()[0][j];
       }
       tmpcorr /= kSamplesPer48kHzChannel;
       if (tmpcorr > xcorr) {
@@ -86,7 +94,7 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
     }
     // High cross correlation check.
     bool any_present = false;
-    for (int j = 0; j < kNumBands; ++j) {
+    for (size_t j = 0; j < kNumBands; ++j) {
       any_present |= is_present[j];
     }
     if (any_present) {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_file_processor.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_file_processor.cc
new file mode 100644
index 0000000000..56e9b4b96f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_file_processor.cc
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+
+using rtc::scoped_ptr;
+using rtc::CheckedDivExact;
+using std::vector;
+using webrtc::audioproc::Event;
+using webrtc::audioproc::Init;
+using webrtc::audioproc::ReverseStream;
+using webrtc::audioproc::Stream;
+
+namespace webrtc {
+namespace {
+
+// Returns a StreamConfig corresponding to file.
+StreamConfig GetStreamConfig(const WavFile& file) {
+  return StreamConfig(file.sample_rate(), file.num_channels());
+}
+
+// Returns a ChannelBuffer corresponding to file.
+ChannelBuffer<float> GetChannelBuffer(const WavFile& file) {
+  return ChannelBuffer<float>(
+      CheckedDivExact(file.sample_rate(), AudioFileProcessor::kChunksPerSecond),
+      file.num_channels());
+}
+
+}  // namespace
+
+WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
+                                   scoped_ptr<WavReader> in_file,
+                                   scoped_ptr<WavWriter> out_file)
+    : ap_(std::move(ap)),
+      in_buf_(GetChannelBuffer(*in_file)),
+      out_buf_(GetChannelBuffer(*out_file)),
+      input_config_(GetStreamConfig(*in_file)),
+      output_config_(GetStreamConfig(*out_file)),
+      buffer_reader_(std::move(in_file)),
+      buffer_writer_(std::move(out_file)) {}
+
+bool WavFileProcessor::ProcessChunk() {
+  if (!buffer_reader_.Read(&in_buf_)) {
+    return false;
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(kNoErr,
+                 ap_->ProcessStream(in_buf_.channels(), input_config_,
+                                    output_config_, out_buf_.channels()));
+  }
+  buffer_writer_.Write(out_buf_);
+  return true;
+}
+
+AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,
+                                           FILE* dump_file,
+                                           scoped_ptr<WavWriter> out_file)
+    : ap_(std::move(ap)),
+      dump_file_(dump_file),
+      out_buf_(GetChannelBuffer(*out_file)),
+      output_config_(GetStreamConfig(*out_file)),
+      buffer_writer_(std::move(out_file)) {
+  RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
+}
+
+AecDumpFileProcessor::~AecDumpFileProcessor() {
+  fclose(dump_file_);
+}
+
+bool AecDumpFileProcessor::ProcessChunk() {
+  Event event_msg;
+
+  // Continue until we process our first Stream message.
+  do {
+    if (!ReadMessageFromFile(dump_file_, &event_msg)) {
+      return false;
+    }
+
+    if (event_msg.type() == Event::INIT) {
+      RTC_CHECK(event_msg.has_init());
+      HandleMessage(event_msg.init());
+
+    } else if (event_msg.type() == Event::STREAM) {
+      RTC_CHECK(event_msg.has_stream());
+      HandleMessage(event_msg.stream());
+
+    } else if (event_msg.type() == Event::REVERSE_STREAM) {
+      RTC_CHECK(event_msg.has_reverse_stream());
+      HandleMessage(event_msg.reverse_stream());
+    }
+  } while (event_msg.type() != Event::STREAM);
+
+  return true;
+}
+
+void AecDumpFileProcessor::HandleMessage(const Init& msg) {
+  RTC_CHECK(msg.has_sample_rate());
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_num_reverse_channels());
+
+  in_buf_.reset(new ChannelBuffer<float>(
+      CheckedDivExact(msg.sample_rate(), kChunksPerSecond),
+      msg.num_input_channels()));
+  const int reverse_sample_rate = msg.has_reverse_sample_rate()
+                                      ? msg.reverse_sample_rate()
+                                      : msg.sample_rate();
+  reverse_buf_.reset(new ChannelBuffer<float>(
+      CheckedDivExact(reverse_sample_rate, kChunksPerSecond),
+      msg.num_reverse_channels()));
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  reverse_config_ =
+      StreamConfig(reverse_sample_rate, msg.num_reverse_channels());
+
+  const ProcessingConfig config = {
+      {input_config_, output_config_, reverse_config_, reverse_config_}};
+  RTC_CHECK_EQ(kNoErr, ap_->Initialize(config));
+}
+
+void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
+  RTC_CHECK(!msg.has_input_data());
+  RTC_CHECK_EQ(in_buf_->num_channels(),
+               static_cast<size_t>(msg.input_channel_size()));
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+    RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                 msg.input_channel(i).size());
+    std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
+                msg.input_channel(i).size());
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    RTC_CHECK_EQ(kNoErr, ap_->set_stream_delay_ms(msg.delay()));
+    ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+    if (msg.has_keypress()) {
+      ap_->set_stream_key_pressed(msg.keypress());
+    }
+    RTC_CHECK_EQ(kNoErr,
+                 ap_->ProcessStream(in_buf_->channels(), input_config_,
+                                    output_config_, out_buf_.channels()));
+  }
+
+  buffer_writer_.Write(out_buf_);
+}
+
+void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
+  RTC_CHECK(!msg.has_data());
+  RTC_CHECK_EQ(reverse_buf_->num_channels(),
+               static_cast<size_t>(msg.channel_size()));
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+    RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                 msg.channel(i).size());
+    std::memcpy(reverse_buf_->channels()[i], msg.channel(i).data(),
+                msg.channel(i).size());
+  }
+  {
+    const auto st = ScopedTimer(mutable_proc_time());
+    // TODO(ajm): This currently discards the processed output, which is needed
+    // for e.g. intelligibility enhancement.
+    RTC_CHECK_EQ(kNoErr, ap_->ProcessReverseStream(
+                             reverse_buf_->channels(), reverse_config_,
+                             reverse_config_, reverse_buf_->channels()));
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_file_processor.h b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_file_processor.h
new file mode 100644
index 0000000000..a3153b2244
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_file_processor.h
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
+
+#include <algorithm>
+#include <limits>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/audio_processing/debug.pb.h"
+#endif
+
+namespace webrtc {
+
+// Holds a few statistics about a series of TickIntervals.
+struct TickIntervalStats {
+  TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
+  TickInterval sum;
+  TickInterval max;
+  TickInterval min;
+};
+
+// Interface for processing an input file with an AudioProcessing instance and
+// dumping the results to an output file.
+class AudioFileProcessor {
+ public:
+  static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
+
+  virtual ~AudioFileProcessor() {}
+
+  // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
+  // writes to the output file.
+  virtual bool ProcessChunk() = 0;
+
+  // Returns the execution time of all AudioProcessing calls.
+  const TickIntervalStats& proc_time() const { return proc_time_; }
+
+ protected:
+  // RAII class for execution time measurement. Updates the provided
+  // TickIntervalStats based on the time between ScopedTimer creation and
+  // leaving the enclosing scope.
+  class ScopedTimer {
+   public:
+    explicit ScopedTimer(TickIntervalStats* proc_time)
+        : proc_time_(proc_time), start_time_(TickTime::Now()) {}
+
+    ~ScopedTimer() {
+      TickInterval interval = TickTime::Now() - start_time_;
+      proc_time_->sum += interval;
+      proc_time_->max = std::max(proc_time_->max, interval);
+      proc_time_->min = std::min(proc_time_->min, interval);
+    }
+
+   private:
+    TickIntervalStats* const proc_time_;
+    TickTime start_time_;
+  };
+
+  TickIntervalStats* mutable_proc_time() { return &proc_time_; }
+
+ private:
+  TickIntervalStats proc_time_;
+};
+
+// Used to read from and write to WavFile objects.
+class WavFileProcessor final : public AudioFileProcessor {
+ public:
+  // Takes ownership of all parameters.
+  WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
+                   rtc::scoped_ptr<WavReader> in_file,
+                   rtc::scoped_ptr<WavWriter> out_file);
+  virtual ~WavFileProcessor() {}
+
+  // Processes one chunk from the WAV input and writes to the WAV output.
+  bool ProcessChunk() override;
+
+ private:
+  rtc::scoped_ptr<AudioProcessing> ap_;
+
+  ChannelBuffer<float> in_buf_;
+  ChannelBuffer<float> out_buf_;
+  const StreamConfig input_config_;
+  const StreamConfig output_config_;
+  ChannelBufferWavReader buffer_reader_;
+  ChannelBufferWavWriter buffer_writer_;
+};
+
+// Used to read from an aecdump file and write to a WavWriter.
+class AecDumpFileProcessor final : public AudioFileProcessor {
+ public:
+  // Takes ownership of all parameters.
+  AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
+                       FILE* dump_file,
+                       rtc::scoped_ptr<WavWriter> out_file);
+
+  virtual ~AecDumpFileProcessor();
+
+  // Processes messages from the aecdump file until the first Stream message is
+  // completed. Passes other data from the aecdump messages as appropriate.
+  bool ProcessChunk() override;
+
+ private:
+  void HandleMessage(const webrtc::audioproc::Init& msg);
+  void HandleMessage(const webrtc::audioproc::Stream& msg);
+  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
+
+  rtc::scoped_ptr<AudioProcessing> ap_;
+  FILE* dump_file_;
+
+  rtc::scoped_ptr<ChannelBuffer<float>> in_buf_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_;
+  ChannelBuffer<float> out_buf_;
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  const StreamConfig output_config_;
+  ChannelBufferWavWriter buffer_writer_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 6546cfae0f..94aea17277 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -14,6 +14,7 @@
 #include <limits>
 #include <queue>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
@@ -22,12 +23,12 @@
 #include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "gtest/gtest.h"
 #include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
@@ -39,6 +40,9 @@
 namespace webrtc {
 namespace {
 
+// TODO(ekmeyerson): Switch to using StreamConfig and ProcessingConfig where
+// applicable.
+
 // TODO(bjornv): This is not feasible until the functionality has been
 // re-implemented; see comment at the bottom of this file. For now, the user has
 // to hard code the |write_ref_data| value.
@@ -46,20 +50,17 @@ namespace {
 // file. This is the typical case. When the file should be updated, it can
 // be set to true with the command-line switch --write_ref_data.
 bool write_ref_data = false;
-const int kChannels[] = {1, 2};
-const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
-
-const int kSampleRates[] = {8000, 16000, 32000};
-const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
+const google::protobuf::int32 kChannels[] = {1, 2};
+const int kSampleRates[] = {8000, 16000, 32000, 48000};
 
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
 // AECM doesn't support super-wb.
 const int kProcessSampleRates[] = {8000, 16000};
 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
-const int kProcessSampleRates[] = {8000, 16000, 32000};
+const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
 #endif
-const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
-    sizeof(*kProcessSampleRates);
+
+enum StreamDirection { kForward = 0, kReverse };
 
 void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
   ChannelBuffer<int16_t> cb_int(cb->num_frames(),
@@ -68,7 +69,7 @@ void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
                cb->num_frames(),
                cb->num_channels(),
                cb_int.channels());
-  for (int i = 0; i < cb->num_channels(); ++i) {
+  for (size_t i = 0; i < cb->num_channels(); ++i) {
     S16ToFloat(cb_int.channels()[i],
                cb->num_frames(),
                cb->channels()[i]);
@@ -80,7 +81,7 @@ void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) {
 }
 
 // Number of channels including the keyboard channel.
-int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+size_t TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
       return 1;
@@ -91,7 +92,7 @@ int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
       return 3;
   }
   assert(false);
-  return -1;
+  return 0;
 }
 
 int TruncateToMultipleOf10(int value) {
@@ -99,45 +100,47 @@ int TruncateToMultipleOf10(int value) {
 }
 
 void MixStereoToMono(const float* stereo, float* mono,
-                     int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; ++i)
+                     size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; ++i)
     mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2;
 }
 
 void MixStereoToMono(const int16_t* stereo, int16_t* mono,
-                     int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; ++i)
+                     size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; ++i)
     mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
 }
 
-void CopyLeftToRightChannel(int16_t* stereo, int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; i++) {
+void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     stereo[i * 2 + 1] = stereo[i * 2];
   }
 }
 
-void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) {
-  for (int i = 0; i < samples_per_channel; i++) {
+void VerifyChannelsAreEqual(int16_t* stereo, size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
   }
 }
 
 void SetFrameTo(AudioFrame* frame, int16_t value) {
-  for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) {
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       ++i) {
     frame->data_[i] = value;
   }
 }
 
 void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
-  ASSERT_EQ(2, frame->num_channels_);
-  for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+  ASSERT_EQ(2u, frame->num_channels_);
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     frame->data_[i] = left;
     frame->data_[i + 1] = right;
   }
 }
 
 void ScaleFrame(AudioFrame* frame, float scale) {
-  for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) {
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       ++i) {
     frame->data_[i] = FloatS16ToS16(frame->data_[i] * scale);
   }
 }
@@ -188,9 +191,9 @@ T AbsValue(T a) {
 }
 
 int16_t MaxAudioFrame(const AudioFrame& frame) {
-  const int length = frame.samples_per_channel_ * frame.num_channels_;
+  const size_t length = frame.samples_per_channel_ * frame.num_channels_;
   int16_t max_data = AbsValue(frame.data_[0]);
-  for (int i = 1; i < length; i++) {
+  for (size_t i = 1; i < length; i++) {
     max_data = std::max(max_data, AbsValue(frame.data_[i]));
   }
 
@@ -251,13 +254,16 @@ std::map<std::string, std::string> temp_filenames;
 std::string OutputFilePath(std::string name,
                            int input_rate,
                            int output_rate,
-                           int reverse_rate,
-                           int num_input_channels,
-                           int num_output_channels,
-                           int num_reverse_channels) {
+                           int reverse_input_rate,
+                           int reverse_output_rate,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
+                           size_t num_reverse_input_channels,
+                           size_t num_reverse_output_channels,
+                           StreamDirection file_direction) {
   std::ostringstream ss;
-  ss << name << "_i" << num_input_channels << "_" << input_rate / 1000
-     << "_r" << num_reverse_channels << "_" << reverse_rate  / 1000 << "_";
+  ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
+     << num_reverse_input_channels << "_" << reverse_input_rate / 1000 << "_";
   if (num_output_channels == 1) {
     ss << "mono";
   } else if (num_output_channels == 2) {
@@ -265,10 +271,19 @@ std::string OutputFilePath(std::string name,
   } else {
     assert(false);
   }
-  ss << output_rate / 1000 << "_pcm";
+  ss << output_rate / 1000;
+  if (num_reverse_output_channels == 1) {
+    ss << "_rmono";
+  } else if (num_reverse_output_channels == 2) {
+    ss << "_rstereo";
+  } else {
+    assert(false);
+  }
+  ss << reverse_output_rate / 1000;
+  ss << "_d" << file_direction << "_pcm";
 
   std::string filename = ss.str();
-  if (temp_filenames[filename] == "")
+  if (temp_filenames[filename].empty())
     temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename);
   return temp_filenames[filename];
 }
@@ -323,9 +338,6 @@ class ApmTest : public ::testing::Test {
 
   static void SetUpTestCase() {
     Trace::CreateTrace();
-    std::string trace_filename =
-        test::TempFilename(test::OutputPath(), "audioproc_trace");
-    ASSERT_EQ(0, Trace::SetTraceFile(trace_filename.c_str()));
   }
 
   static void TearDownTestCase() {
@@ -342,9 +354,9 @@ class ApmTest : public ::testing::Test {
   void Init(int sample_rate_hz,
             int output_sample_rate_hz,
             int reverse_sample_rate_hz,
-            int num_reverse_channels,
-            int num_input_channels,
-            int num_output_channels,
+            size_t num_input_channels,
+            size_t num_output_channels,
+            size_t num_reverse_channels,
             bool open_output_file);
   void Init(AudioProcessing* ap);
   void EnableAllComponents();
@@ -356,8 +368,14 @@ class ApmTest : public ::testing::Test {
   void ProcessWithDefaultStreamParameters(AudioFrame* frame);
   void ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
                                     int delay_min, int delay_max);
-  void TestChangingChannels(int num_channels,
-                            AudioProcessing::Error expected_return);
+  void TestChangingChannelsInt16Interface(
+      size_t num_channels,
+      AudioProcessing::Error expected_return);
+  void TestChangingForwardChannels(size_t num_in_channels,
+                                   size_t num_out_channels,
+                                   AudioProcessing::Error expected_return);
+  void TestChangingReverseChannels(size_t num_rev_channels,
+                                   AudioProcessing::Error expected_return);
   void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate);
   void RunManualVolumeChangeIsPossibleTest(int sample_rate);
   void StreamParametersTest(Format format);
@@ -377,7 +395,7 @@ class ApmTest : public ::testing::Test {
   rtc::scoped_ptr<ChannelBuffer<float> > float_cb_;
   rtc::scoped_ptr<ChannelBuffer<float> > revfloat_cb_;
   int output_sample_rate_hz_;
-  int num_output_channels_;
+  size_t num_output_channels_;
   FILE* far_file_;
   FILE* near_file_;
   FILE* out_file_;
@@ -451,20 +469,19 @@ void ApmTest::TearDown() {
 
 void ApmTest::Init(AudioProcessing* ap) {
   ASSERT_EQ(kNoErr,
-            ap->Initialize(frame_->sample_rate_hz_,
-                           output_sample_rate_hz_,
-                           revframe_->sample_rate_hz_,
-                           LayoutFromChannels(frame_->num_channels_),
-                           LayoutFromChannels(num_output_channels_),
-                           LayoutFromChannels(revframe_->num_channels_)));
+            ap->Initialize(
+                {{{frame_->sample_rate_hz_, frame_->num_channels_},
+                  {output_sample_rate_hz_, num_output_channels_},
+                  {revframe_->sample_rate_hz_, revframe_->num_channels_},
+                  {revframe_->sample_rate_hz_, revframe_->num_channels_}}}));
 }
 
 void ApmTest::Init(int sample_rate_hz,
                    int output_sample_rate_hz,
                    int reverse_sample_rate_hz,
-                   int num_input_channels,
-                   int num_output_channels,
-                   int num_reverse_channels,
+                   size_t num_input_channels,
+                   size_t num_output_channels,
+                   size_t num_reverse_channels,
                    bool open_output_file) {
   SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_);
   output_sample_rate_hz_ = output_sample_rate_hz;
@@ -494,13 +511,10 @@ void ApmTest::Init(int sample_rate_hz,
     if (out_file_) {
       ASSERT_EQ(0, fclose(out_file_));
     }
-    filename = OutputFilePath("out",
-                              sample_rate_hz,
-                              output_sample_rate_hz,
-                              reverse_sample_rate_hz,
-                              num_input_channels,
-                              num_output_channels,
-                              num_reverse_channels);
+    filename = OutputFilePath(
+        "out", sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
+        reverse_sample_rate_hz, num_input_channels, num_output_channels,
+        num_reverse_channels, num_reverse_channels, kForward);
     out_file_ = fopen(filename.c_str(), "wb");
     ASSERT_TRUE(out_file_ != NULL) << "Could not open file " <<
           filename << "\n";
@@ -659,13 +673,18 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
   }
   // Calculate expected delay estimate and acceptable regions. Further,
   // limit them w.r.t. AEC delay estimation support.
-  const int samples_per_ms = std::min(16, frame_->samples_per_channel_ / 10);
+  const size_t samples_per_ms =
+      std::min(static_cast<size_t>(16), frame_->samples_per_channel_ / 10);
   int expected_median = std::min(std::max(delay_ms - system_delay_ms,
                                           delay_min), delay_max);
-  int expected_median_high = std::min(std::max(
-      expected_median + 96 / samples_per_ms, delay_min), delay_max);
-  int expected_median_low = std::min(std::max(
-      expected_median - 96 / samples_per_ms, delay_min), delay_max);
+  int expected_median_high = std::min(
+      std::max(expected_median + static_cast<int>(96 / samples_per_ms),
+               delay_min),
+      delay_max);
+  int expected_median_low = std::min(
+      std::max(expected_median - static_cast<int>(96 / samples_per_ms),
+               delay_min),
+      delay_max);
   // Verify delay metrics.
   int median;
   int std;
@@ -793,23 +812,79 @@ TEST_F(ApmTest, DelayOffsetWithLimitsIsSetProperly) {
   EXPECT_EQ(50, apm_->stream_delay_ms());
 }
 
-void ApmTest::TestChangingChannels(int num_channels,
-                                   AudioProcessing::Error expected_return) {
+void ApmTest::TestChangingChannelsInt16Interface(
+    size_t num_channels,
+    AudioProcessing::Error expected_return) {
   frame_->num_channels_ = num_channels;
   EXPECT_EQ(expected_return, apm_->ProcessStream(frame_));
   EXPECT_EQ(expected_return, apm_->AnalyzeReverseStream(frame_));
 }
 
-TEST_F(ApmTest, Channels) {
-  // Testing number of invalid channels.
-  TestChangingChannels(0, apm_->kBadNumberChannelsError);
-  TestChangingChannels(3, apm_->kBadNumberChannelsError);
-  // Testing number of valid channels.
-  for (int i = 1; i < 3; i++) {
-    TestChangingChannels(i, kNoErr);
+void ApmTest::TestChangingForwardChannels(
+    size_t num_in_channels,
+    size_t num_out_channels,
+    AudioProcessing::Error expected_return) {
+  const StreamConfig input_stream = {frame_->sample_rate_hz_, num_in_channels};
+  const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels};
+
+  EXPECT_EQ(expected_return,
+            apm_->ProcessStream(float_cb_->channels(), input_stream,
+                                output_stream, float_cb_->channels()));
+}
+
+void ApmTest::TestChangingReverseChannels(
+    size_t num_rev_channels,
+    AudioProcessing::Error expected_return) {
+  const ProcessingConfig processing_config = {
+      {{frame_->sample_rate_hz_, apm_->num_input_channels()},
+       {output_sample_rate_hz_, apm_->num_output_channels()},
+       {frame_->sample_rate_hz_, num_rev_channels},
+       {frame_->sample_rate_hz_, num_rev_channels}}};
+
+  EXPECT_EQ(
+      expected_return,
+      apm_->ProcessReverseStream(
+          float_cb_->channels(), processing_config.reverse_input_stream(),
+          processing_config.reverse_output_stream(), float_cb_->channels()));
+}
+
+TEST_F(ApmTest, ChannelsInt16Interface) {
+  // Testing number of invalid and valid channels.
+  Init(16000, 16000, 16000, 4, 4, 4, false);
+
+  TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError);
+
+  for (size_t i = 1; i < 4; i++) {
+    TestChangingChannelsInt16Interface(i, kNoErr);
     EXPECT_EQ(i, apm_->num_input_channels());
     // We always force the number of reverse channels used for processing to 1.
-    EXPECT_EQ(1, apm_->num_reverse_channels());
+    EXPECT_EQ(1u, apm_->num_reverse_channels());
+  }
+}
+
+TEST_F(ApmTest, Channels) {
+  // Testing number of invalid and valid channels.
+  Init(16000, 16000, 16000, 4, 4, 4, false);
+
+  TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError);
+  TestChangingReverseChannels(0, apm_->kBadNumberChannelsError);
+
+  for (size_t i = 1; i < 4; ++i) {
+    for (size_t j = 0; j < 1; ++j) {
+      // Output channels much be one or match input channels.
+      if (j == 1 || i == j) {
+        TestChangingForwardChannels(i, j, kNoErr);
+        TestChangingReverseChannels(i, kNoErr);
+
+        EXPECT_EQ(i, apm_->num_input_channels());
+        EXPECT_EQ(j, apm_->num_output_channels());
+        // The number of reverse channels used for processing to is always 1.
+        EXPECT_EQ(1u, apm_->num_reverse_channels());
+      } else {
+        TestChangingForwardChannels(i, j,
+                                    AudioProcessing::kBadNumberChannelsError);
+      }
+    }
   }
 }
 
@@ -818,11 +893,10 @@ TEST_F(ApmTest, SampleRatesInt) {
   SetContainerFormat(10000, 2, frame_, &float_cb_);
   EXPECT_EQ(apm_->kBadSampleRateError, ProcessStreamChooser(kIntFormat));
   // Testing valid sample rates
-  int fs[] = {8000, 16000, 32000};
-  for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
+  int fs[] = {8000, 16000, 32000, 48000};
+  for (size_t i = 0; i < arraysize(fs); i++) {
     SetContainerFormat(fs[i], 2, frame_, &float_cb_);
     EXPECT_NOERR(ProcessStreamChooser(kIntFormat));
-    EXPECT_EQ(fs[i], apm_->input_sample_rate_hz());
   }
 }
 
@@ -839,7 +913,7 @@ TEST_F(ApmTest, EchoCancellation) {
     EchoCancellation::kModerateSuppression,
     EchoCancellation::kHighSuppression,
   };
-  for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+  for (size_t i = 0; i < arraysize(level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->echo_cancellation()->set_suppression_level(level[i]));
     EXPECT_EQ(level[i],
@@ -895,7 +969,7 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
             apm_->echo_cancellation()->enable_delay_logging(true));
   EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
   Config config;
-  config.Set<ReportedDelay>(new ReportedDelay(true));
+  config.Set<DelayAgnostic>(new DelayAgnostic(false));
   apm_->SetExtraOptions(config);
 
   // Internally in the AEC the amount of lookahead the delay estimation can
@@ -916,7 +990,7 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
   // Test a couple of corner cases and verify that the estimated delay is
   // within a valid region (set to +-1.5 blocks). Note that these cases are
   // sampling frequency dependent.
-  for (size_t i = 0; i < kProcessSampleRatesSize; i++) {
+  for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
     Init(kProcessSampleRates[i],
          kProcessSampleRates[i],
          kProcessSampleRates[i],
@@ -925,8 +999,8 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
          2,
          false);
     // Sampling frequency dependent variables.
-    const int num_ms_per_block = std::max(4,
-                                          640 / frame_->samples_per_channel_);
+    const int num_ms_per_block =
+        std::max(4, static_cast<int>(640 / frame_->samples_per_channel_));
     const int delay_min_ms = -kLookaheadBlocks * num_ms_per_block;
     const int delay_max_ms = (kMaxDelayBlocks - 1) * num_ms_per_block;
 
@@ -988,7 +1062,7 @@ TEST_F(ApmTest, EchoControlMobile) {
       EchoControlMobile::kSpeakerphone,
       EchoControlMobile::kLoudSpeakerphone,
   };
-  for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+  for (size_t i = 0; i < arraysize(mode); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->echo_control_mobile()->set_routing_mode(mode[i]));
     EXPECT_EQ(mode[i],
@@ -1053,7 +1127,7 @@ TEST_F(ApmTest, GainControl) {
     GainControl::kAdaptiveDigital,
     GainControl::kFixedDigital
   };
-  for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
+  for (size_t i = 0; i < arraysize(mode); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_mode(mode[i]));
     EXPECT_EQ(mode[i], apm_->gain_control()->mode());
@@ -1069,7 +1143,7 @@ TEST_F(ApmTest, GainControl) {
       apm_->gain_control()->target_level_dbfs()));
 
   int level_dbfs[] = {0, 6, 31};
-  for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
+  for (size_t i = 0; i < arraysize(level_dbfs); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
     EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
@@ -1087,7 +1161,7 @@ TEST_F(ApmTest, GainControl) {
       apm_->gain_control()->compression_gain_db()));
 
   int gain_db[] = {0, 10, 90};
-  for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
+  for (size_t i = 0; i < arraysize(gain_db); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_compression_gain_db(gain_db[i]));
     EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
@@ -1118,14 +1192,14 @@ TEST_F(ApmTest, GainControl) {
       apm_->gain_control()->analog_level_maximum()));
 
   int min_level[] = {0, 255, 1024};
-  for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+  for (size_t i = 0; i < arraysize(min_level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
     EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
   }
 
   int max_level[] = {0, 1024, 65535};
-  for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
+  for (size_t i = 0; i < arraysize(min_level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
     EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
@@ -1164,7 +1238,7 @@ void ApmTest::RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate) {
 // Verifies that despite volume slider quantization, the AGC can continue to
 // increase its volume.
 TEST_F(ApmTest, QuantizedVolumeDoesNotGetStuck) {
-  for (size_t i = 0; i < kSampleRatesSize; ++i) {
+  for (size_t i = 0; i < arraysize(kSampleRates); ++i) {
     RunQuantizedVolumeDoesNotGetStuckTest(kSampleRates[i]);
   }
 }
@@ -1209,7 +1283,7 @@ void ApmTest::RunManualVolumeChangeIsPossibleTest(int sample_rate) {
 }
 
 TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
-  for (size_t i = 0; i < kSampleRatesSize; ++i) {
+  for (size_t i = 0; i < arraysize(kSampleRates); ++i) {
     RunManualVolumeChangeIsPossibleTest(kSampleRates[i]);
   }
 }
@@ -1217,11 +1291,11 @@ TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
 TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
   const int kSampleRateHz = 16000;
-  const int kSamplesPerChannel =
-      AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000;
-  const int kNumInputChannels = 2;
-  const int kNumOutputChannels = 1;
-  const int kNumChunks = 700;
+  const size_t kSamplesPerChannel =
+      static_cast<size_t>(AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000);
+  const size_t kNumInputChannels = 2;
+  const size_t kNumOutputChannels = 1;
+  const size_t kNumChunks = 700;
   const float kScaleFactor = 0.25f;
   Config config;
   std::vector<webrtc::Point> geometry;
@@ -1235,8 +1309,8 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
   EXPECT_EQ(kNoErr, apm->gain_control()->Enable(true));
   ChannelBuffer<float> src_buf(kSamplesPerChannel, kNumInputChannels);
   ChannelBuffer<float> dest_buf(kSamplesPerChannel, kNumOutputChannels);
-  const int max_length = kSamplesPerChannel * std::max(kNumInputChannels,
-                                                       kNumOutputChannels);
+  const size_t max_length = kSamplesPerChannel * std::max(kNumInputChannels,
+                                                          kNumOutputChannels);
   rtc::scoped_ptr<int16_t[]> int_data(new int16_t[max_length]);
   rtc::scoped_ptr<float[]> float_data(new float[max_length]);
   std::string filename = ResourceFilePath("far", kSampleRateHz);
@@ -1248,13 +1322,13 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
   bool is_target = false;
   EXPECT_CALL(*beamformer, is_target_present())
       .WillRepeatedly(testing::ReturnPointee(&is_target));
-  for (int i = 0; i < kNumChunks; ++i) {
+  for (size_t i = 0; i < kNumChunks; ++i) {
     ASSERT_TRUE(ReadChunk(far_file,
                           int_data.get(),
                           float_data.get(),
                           &src_buf));
-    for (int j = 0; j < kNumInputChannels; ++j) {
-      for (int k = 0; k < kSamplesPerChannel; ++k) {
+    for (size_t j = 0; j < kNumInputChannels; ++j) {
+      for (size_t k = 0; k < kSamplesPerChannel; ++k) {
         src_buf.channels()[j][k] *= kScaleFactor;
       }
     }
@@ -1273,13 +1347,13 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
             apm->gain_control()->compression_gain_db());
   rewind(far_file);
   is_target = true;
-  for (int i = 0; i < kNumChunks; ++i) {
+  for (size_t i = 0; i < kNumChunks; ++i) {
     ASSERT_TRUE(ReadChunk(far_file,
                           int_data.get(),
                           float_data.get(),
                           &src_buf));
-    for (int j = 0; j < kNumInputChannels; ++j) {
-      for (int k = 0; k < kSamplesPerChannel; ++k) {
+    for (size_t j = 0; j < kNumInputChannels; ++j) {
+      for (size_t k = 0; k < kSamplesPerChannel; ++k) {
         src_buf.channels()[j][k] *= kScaleFactor;
       }
     }
@@ -1308,7 +1382,7 @@ TEST_F(ApmTest, NoiseSuppression) {
     NoiseSuppression::kHigh,
     NoiseSuppression::kVeryHigh
   };
-  for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
+  for (size_t i = 0; i < arraysize(level); i++) {
     EXPECT_EQ(apm_->kNoError,
         apm_->noise_suppression()->set_level(level[i]));
     EXPECT_EQ(level[i], apm_->noise_suppression()->level());
@@ -1410,7 +1484,7 @@ TEST_F(ApmTest, VoiceDetection) {
       VoiceDetection::kModerateLikelihood,
       VoiceDetection::kHighLikelihood
   };
-  for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
+  for (size_t i = 0; i < arraysize(likelihood); i++) {
     EXPECT_EQ(apm_->kNoError,
               apm_->voice_detection()->set_likelihood(likelihood[i]));
     EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
@@ -1442,7 +1516,7 @@ TEST_F(ApmTest, VoiceDetection) {
       AudioFrame::kVadPassive,
       AudioFrame::kVadUnknown
   };
-  for (size_t i = 0; i < sizeof(activity)/sizeof(*activity); i++) {
+  for (size_t i = 0; i < arraysize(activity); i++) {
     frame_->vad_activity_ = activity[i];
     EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
     EXPECT_EQ(activity[i], frame_->vad_activity_);
@@ -1468,7 +1542,7 @@ TEST_F(ApmTest, AllProcessingDisabledByDefault) {
 }
 
 TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
-  for (size_t i = 0; i < kSampleRatesSize; i++) {
+  for (size_t i = 0; i < arraysize(kSampleRates); i++) {
     Init(kSampleRates[i], kSampleRates[i], kSampleRates[i], 2, 2, 2, false);
     SetFrameTo(frame_, 1000, 2000);
     AudioFrame frame_copy;
@@ -1476,6 +1550,8 @@ TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
     for (int j = 0; j < 1000; j++) {
       EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
       EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
+      EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(frame_));
+      EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
     }
   }
 }
@@ -1500,12 +1576,25 @@ TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) {
   for (size_t i = 0; i < kSamples; ++i) {
     EXPECT_EQ(src[i], dest[i]);
   }
+
+  // Same for ProcessReverseStream.
+  float rev_dest[kSamples] = {};
+  auto rev_dest_channels = &rev_dest[0];
+
+  StreamConfig input_stream = {sample_rate, 1};
+  StreamConfig output_stream = {sample_rate, 1};
+  EXPECT_NOERR(apm_->ProcessReverseStream(&src_channels, input_stream,
+                                          output_stream, &rev_dest_channels));
+
+  for (size_t i = 0; i < kSamples; ++i) {
+    EXPECT_EQ(src[i], rev_dest[i]);
+  }
 }
 
 TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
   EnableAllComponents();
 
-  for (size_t i = 0; i < kProcessSampleRatesSize; i++) {
+  for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
     Init(kProcessSampleRates[i],
          kProcessSampleRates[i],
          kProcessSampleRates[i],
@@ -1589,7 +1678,7 @@ TEST_F(ApmTest, SplittingFilter) {
   // Make sure we have extended filter enabled. This makes sure nothing is
   // touched until we have a farend frame.
   Config config;
-  config.Set<DelayCorrection>(new DelayCorrection(true));
+  config.Set<ExtendedFilter>(new ExtendedFilter(true));
   apm_->SetExtraOptions(config);
   SetFrameTo(frame_, 1000);
   frame_copy.CopyFrom(*frame_);
@@ -1653,7 +1742,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
       const audioproc::ReverseStream msg = event_msg.reverse_stream();
 
       if (msg.channel_size() > 0) {
-        ASSERT_EQ(revframe_->num_channels_, msg.channel_size());
+        ASSERT_EQ(revframe_->num_channels_,
+                  static_cast<size_t>(msg.channel_size()));
         for (int i = 0; i < msg.channel_size(); ++i) {
            memcpy(revfloat_cb_->channels()[i],
                   msg.channel(i).data(),
@@ -1683,7 +1773,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
       }
 
       if (msg.input_channel_size() > 0) {
-        ASSERT_EQ(frame_->num_channels_, msg.input_channel_size());
+        ASSERT_EQ(frame_->num_channels_,
+                  static_cast<size_t>(msg.input_channel_size()));
         for (int i = 0; i < msg.input_channel_size(); ++i) {
            memcpy(float_cb_->channels()[i],
                   msg.input_channel(i).data(),
@@ -1744,6 +1835,8 @@ void ApmTest::VerifyDebugDumpTest(Format format) {
   EXPECT_NE(0, feof(out_file));
   ASSERT_EQ(0, fclose(ref_file));
   ASSERT_EQ(0, fclose(out_file));
+  remove(ref_filename.c_str());
+  remove(out_filename.c_str());
 }
 
 TEST_F(ApmTest, VerifyDebugDumpInt) {
@@ -1839,11 +1932,14 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
     if (test->num_input_channels() != test->num_output_channels())
       continue;
 
-    const int num_render_channels = test->num_reverse_channels();
-    const int num_input_channels = test->num_input_channels();
-    const int num_output_channels = test->num_output_channels();
-    const int samples_per_channel = test->sample_rate() *
-        AudioProcessing::kChunkSizeMs / 1000;
+    const size_t num_render_channels =
+        static_cast<size_t>(test->num_reverse_channels());
+    const size_t num_input_channels =
+        static_cast<size_t>(test->num_input_channels());
+    const size_t num_output_channels =
+        static_cast<size_t>(test->num_output_channels());
+    const size_t samples_per_channel = static_cast<size_t>(
+        test->sample_rate() * AudioProcessing::kChunkSizeMs / 1000);
 
     Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
          num_input_channels, num_output_channels, num_render_channels, true);
@@ -1884,7 +1980,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
           test->sample_rate(),
           LayoutFromChannels(num_output_channels),
           float_cb_->channels()));
-      for (int j = 0; j < num_output_channels; ++j) {
+      for (size_t j = 0; j < num_output_channels; ++j) {
         FloatToS16(float_cb_->channels()[j],
                    samples_per_channel,
                    output_cb.channels()[j]);
@@ -1914,10 +2010,10 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
                 fapm->echo_cancellation()->stream_has_echo());
       EXPECT_NEAR(apm_->noise_suppression()->speech_probability(),
                   fapm->noise_suppression()->speech_probability(),
-                  0.0005);
+                  0.01);
 
       // Reset in case of downmixing.
-      frame_->num_channels_ = test->num_input_channels();
+      frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
     }
     rewind(far_file_);
     rewind(near_file_);
@@ -1935,9 +2031,9 @@ TEST_F(ApmTest, Process) {
     OpenFileAndReadMessage(ref_filename_, &ref_data);
   } else {
     // Write the desired tests to the protobuf reference file.
-    for (size_t i = 0; i < kChannelsSize; i++) {
-      for (size_t j = 0; j < kChannelsSize; j++) {
-        for (size_t l = 0; l < kProcessSampleRatesSize; l++) {
+    for (size_t i = 0; i < arraysize(kChannels); i++) {
+      for (size_t j = 0; j < arraysize(kChannels); j++) {
+        for (size_t l = 0; l < arraysize(kProcessSampleRates); l++) {
           audioproc::Test* test = ref_data.add_test();
           test->set_num_reverse_channels(kChannels[i]);
           test->set_num_input_channels(kChannels[j]);
@@ -1969,8 +2065,8 @@ TEST_F(ApmTest, Process) {
 
     Config config;
     config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
-    config.Set<DelayCorrection>(
-        new DelayCorrection(test->use_aec_extended_filter()));
+    config.Set<ExtendedFilter>(
+        new ExtendedFilter(test->use_aec_extended_filter()));
     apm_.reset(AudioProcessing::Create(config));
 
     EnableAllComponents();
@@ -1978,9 +2074,9 @@ TEST_F(ApmTest, Process) {
     Init(test->sample_rate(),
          test->sample_rate(),
          test->sample_rate(),
-         test->num_input_channels(),
-         test->num_output_channels(),
-         test->num_reverse_channels(),
+         static_cast<size_t>(test->num_input_channels()),
+         static_cast<size_t>(test->num_output_channels()),
+         static_cast<size_t>(test->num_reverse_channels()),
          true);
 
     int frame_count = 0;
@@ -2005,7 +2101,8 @@ TEST_F(ApmTest, Process) {
       EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
 
       // Ensure the frame was downmixed properly.
-      EXPECT_EQ(test->num_output_channels(), frame_->num_channels_);
+      EXPECT_EQ(static_cast<size_t>(test->num_output_channels()),
+                frame_->num_channels_);
 
       max_output_average += MaxAudioFrame(*frame_);
 
@@ -2035,7 +2132,7 @@ TEST_F(ApmTest, Process) {
       ASSERT_EQ(frame_size, write_count);
 
       // Reset in case of downmixing.
-      frame_->num_channels_ = test->num_input_channels();
+      frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
       frame_count++;
     }
     max_output_average /= frame_count;
@@ -2164,12 +2261,11 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
     {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono},
     {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo},
   };
-  size_t channel_format_size = sizeof(cf) / sizeof(*cf);
 
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create());
   // Enable one component just to ensure some processing takes place.
   ap->noise_suppression()->Enable(true);
-  for (size_t i = 0; i < channel_format_size; ++i) {
+  for (size_t i = 0; i < arraysize(cf); ++i) {
     const int in_rate = 44100;
     const int out_rate = 48000;
     ChannelBuffer<float> in_cb(SamplesFromRate(in_rate),
@@ -2196,7 +2292,7 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
 // error results to the supplied accumulators.
 void UpdateBestSNR(const float* ref,
                    const float* test,
-                   int length,
+                   size_t length,
                    int expected_delay,
                    double* variance_acc,
                    double* sq_error_acc) {
@@ -2208,7 +2304,7 @@ void UpdateBestSNR(const float* ref,
        ++delay) {
     double sq_error = 0;
     double variance = 0;
-    for (int i = 0; i < length - delay; ++i) {
+    for (size_t i = 0; i < length - delay; ++i) {
       double error = test[i + delay] - ref[i];
       sq_error += error * error;
       variance += ref[i] * ref[i];
@@ -2242,37 +2338,32 @@ void UpdateBestSNR(const float* ref,
 // Due to the resampling distortion, we don't expect identical results, but
 // enforce SNR thresholds which vary depending on the format. 0 is a special
 // case SNR which corresponds to inf, or zero error.
-typedef std::tr1::tuple<int, int, int, double> AudioProcessingTestData;
+typedef std::tr1::tuple<int, int, int, int, double, double>
+    AudioProcessingTestData;
 class AudioProcessingTest
     : public testing::TestWithParam<AudioProcessingTestData> {
  public:
   AudioProcessingTest()
       : input_rate_(std::tr1::get<0>(GetParam())),
         output_rate_(std::tr1::get<1>(GetParam())),
-        reverse_rate_(std::tr1::get<2>(GetParam())),
-        expected_snr_(std::tr1::get<3>(GetParam())) {}
+        reverse_input_rate_(std::tr1::get<2>(GetParam())),
+        reverse_output_rate_(std::tr1::get<3>(GetParam())),
+        expected_snr_(std::tr1::get<4>(GetParam())),
+        expected_reverse_snr_(std::tr1::get<5>(GetParam())) {}
 
   virtual ~AudioProcessingTest() {}
 
   static void SetUpTestCase() {
     // Create all needed output reference files.
-    const int kNativeRates[] = {8000, 16000, 32000};
-    const size_t kNativeRatesSize =
-        sizeof(kNativeRates) / sizeof(*kNativeRates);
-    const int kNumChannels[] = {1, 2};
-    const size_t kNumChannelsSize =
-        sizeof(kNumChannels) / sizeof(*kNumChannels);
-    for (size_t i = 0; i < kNativeRatesSize; ++i) {
-      for (size_t j = 0; j < kNumChannelsSize; ++j) {
-        for (size_t k = 0; k < kNumChannelsSize; ++k) {
+    const int kNativeRates[] = {8000, 16000, 32000, 48000};
+    const size_t kNumChannels[] = {1, 2};
+    for (size_t i = 0; i < arraysize(kNativeRates); ++i) {
+      for (size_t j = 0; j < arraysize(kNumChannels); ++j) {
+        for (size_t k = 0; k < arraysize(kNumChannels); ++k) {
           // The reference files always have matching input and output channels.
-          ProcessFormat(kNativeRates[i],
-                        kNativeRates[i],
-                        kNativeRates[i],
-                        kNumChannels[j],
-                        kNumChannels[j],
-                        kNumChannels[k],
-                        "ref");
+          ProcessFormat(kNativeRates[i], kNativeRates[i], kNativeRates[i],
+                        kNativeRates[i], kNumChannels[j], kNumChannels[j],
+                        kNumChannels[k], kNumChannels[k], "ref");
         }
       }
     }
@@ -2281,62 +2372,75 @@ class AudioProcessingTest
   static void TearDownTestCase() {
     ClearTempFiles();
   }
+
   // Runs a process pass on files with the given parameters and dumps the output
-  // to a file specified with |output_file_prefix|.
+  // to a file specified with |output_file_prefix|. Both forward and reverse
+  // output streams are dumped.
   static void ProcessFormat(int input_rate,
                             int output_rate,
-                            int reverse_rate,
-                            int num_input_channels,
-                            int num_output_channels,
-                            int num_reverse_channels,
+                            int reverse_input_rate,
+                            int reverse_output_rate,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            size_t num_reverse_input_channels,
+                            size_t num_reverse_output_channels,
                             std::string output_file_prefix) {
     Config config;
     config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
     rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
     EnableAllAPComponents(ap.get());
-    ap->Initialize(input_rate,
-                   output_rate,
-                   reverse_rate,
-                   LayoutFromChannels(num_input_channels),
-                   LayoutFromChannels(num_output_channels),
-                   LayoutFromChannels(num_reverse_channels));
 
-    FILE* far_file = fopen(ResourceFilePath("far", reverse_rate).c_str(), "rb");
+    ProcessingConfig processing_config = {
+        {{input_rate, num_input_channels},
+         {output_rate, num_output_channels},
+         {reverse_input_rate, num_reverse_input_channels},
+         {reverse_output_rate, num_reverse_output_channels}}};
+    ap->Initialize(processing_config);
+
+    FILE* far_file =
+        fopen(ResourceFilePath("far", reverse_input_rate).c_str(), "rb");
     FILE* near_file = fopen(ResourceFilePath("near", input_rate).c_str(), "rb");
-    FILE* out_file = fopen(OutputFilePath(output_file_prefix,
-                                          input_rate,
-                                          output_rate,
-                                          reverse_rate,
-                                          num_input_channels,
-                                          num_output_channels,
-                                          num_reverse_channels).c_str(), "wb");
+    FILE* out_file =
+        fopen(OutputFilePath(output_file_prefix, input_rate, output_rate,
+                             reverse_input_rate, reverse_output_rate,
+                             num_input_channels, num_output_channels,
+                             num_reverse_input_channels,
+                             num_reverse_output_channels, kForward).c_str(),
+              "wb");
+    FILE* rev_out_file =
+        fopen(OutputFilePath(output_file_prefix, input_rate, output_rate,
+                             reverse_input_rate, reverse_output_rate,
+                             num_input_channels, num_output_channels,
+                             num_reverse_input_channels,
+                             num_reverse_output_channels, kReverse).c_str(),
+              "wb");
     ASSERT_TRUE(far_file != NULL);
     ASSERT_TRUE(near_file != NULL);
     ASSERT_TRUE(out_file != NULL);
+    ASSERT_TRUE(rev_out_file != NULL);
 
     ChannelBuffer<float> fwd_cb(SamplesFromRate(input_rate),
                                 num_input_channels);
-    ChannelBuffer<float> rev_cb(SamplesFromRate(reverse_rate),
-                                num_reverse_channels);
+    ChannelBuffer<float> rev_cb(SamplesFromRate(reverse_input_rate),
+                                num_reverse_input_channels);
     ChannelBuffer<float> out_cb(SamplesFromRate(output_rate),
                                 num_output_channels);
+    ChannelBuffer<float> rev_out_cb(SamplesFromRate(reverse_output_rate),
+                                    num_reverse_output_channels);
 
     // Temporary buffers.
     const int max_length =
-        2 * std::max(out_cb.num_frames(),
-                     std::max(fwd_cb.num_frames(),
-                              rev_cb.num_frames()));
+        2 * std::max(std::max(out_cb.num_frames(), rev_out_cb.num_frames()),
+                     std::max(fwd_cb.num_frames(), rev_cb.num_frames()));
     rtc::scoped_ptr<float[]> float_data(new float[max_length]);
     rtc::scoped_ptr<int16_t[]> int_data(new int16_t[max_length]);
 
     int analog_level = 127;
     while (ReadChunk(far_file, int_data.get(), float_data.get(), &rev_cb) &&
            ReadChunk(near_file, int_data.get(), float_data.get(), &fwd_cb)) {
-      EXPECT_NOERR(ap->AnalyzeReverseStream(
-          rev_cb.channels(),
-          rev_cb.num_frames(),
-          reverse_rate,
-          LayoutFromChannels(num_reverse_channels)));
+      EXPECT_NOERR(ap->ProcessReverseStream(
+          rev_cb.channels(), processing_config.reverse_input_stream(),
+          processing_config.reverse_output_stream(), rev_out_cb.channels()));
 
       EXPECT_NOERR(ap->set_stream_delay_ms(0));
       ap->echo_cancellation()->set_stream_drift_samples(0);
@@ -2351,276 +2455,295 @@ class AudioProcessingTest
           LayoutFromChannels(num_output_channels),
           out_cb.channels()));
 
-      Interleave(out_cb.channels(),
-                 out_cb.num_frames(),
-                 out_cb.num_channels(),
+      // Dump forward output to file.
+      Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(),
                  float_data.get());
-      // Dump output to file.
-      int out_length = out_cb.num_channels() * out_cb.num_frames();
-      ASSERT_EQ(static_cast<size_t>(out_length),
+      size_t out_length = out_cb.num_channels() * out_cb.num_frames();
+
+      ASSERT_EQ(out_length,
                 fwrite(float_data.get(), sizeof(float_data[0]),
                        out_length, out_file));
 
+      // Dump reverse output to file.
+      Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(),
+                 rev_out_cb.num_channels(), float_data.get());
+      size_t rev_out_length =
+          rev_out_cb.num_channels() * rev_out_cb.num_frames();
+
+      ASSERT_EQ(rev_out_length,
+                fwrite(float_data.get(), sizeof(float_data[0]), rev_out_length,
+                       rev_out_file));
+
       analog_level = ap->gain_control()->stream_analog_level();
     }
     fclose(far_file);
     fclose(near_file);
     fclose(out_file);
+    fclose(rev_out_file);
   }
 
  protected:
   int input_rate_;
   int output_rate_;
-  int reverse_rate_;
+  int reverse_input_rate_;
+  int reverse_output_rate_;
   double expected_snr_;
+  double expected_reverse_snr_;
 };
 
 TEST_P(AudioProcessingTest, Formats) {
   struct ChannelFormat {
     int num_input;
     int num_output;
-    int num_reverse;
+    int num_reverse_input;
+    int num_reverse_output;
   };
   ChannelFormat cf[] = {
-    {1, 1, 1},
-    {1, 1, 2},
-    {2, 1, 1},
-    {2, 1, 2},
-    {2, 2, 1},
-    {2, 2, 2},
+      {1, 1, 1, 1},
+      {1, 1, 2, 1},
+      {2, 1, 1, 1},
+      {2, 1, 2, 1},
+      {2, 2, 1, 1},
+      {2, 2, 2, 2},
   };
-  size_t channel_format_size = sizeof(cf) / sizeof(*cf);
 
-  for (size_t i = 0; i < channel_format_size; ++i) {
-    ProcessFormat(input_rate_,
-                  output_rate_,
-                  reverse_rate_,
-                  cf[i].num_input,
-                  cf[i].num_output,
-                  cf[i].num_reverse,
-                  "out");
-    int min_ref_rate = std::min(input_rate_, output_rate_);
-    int ref_rate;
-    if (min_ref_rate > 16000) {
-      ref_rate = 32000;
-    } else if (min_ref_rate > 8000) {
-      ref_rate = 16000;
-    } else {
-      ref_rate = 8000;
-    }
+  for (size_t i = 0; i < arraysize(cf); ++i) {
+    ProcessFormat(input_rate_, output_rate_, reverse_input_rate_,
+                  reverse_output_rate_, cf[i].num_input, cf[i].num_output,
+                  cf[i].num_reverse_input, cf[i].num_reverse_output, "out");
+
+    // Verify output for both directions.
+    std::vector<StreamDirection> stream_directions;
+    stream_directions.push_back(kForward);
+    stream_directions.push_back(kReverse);
+    for (StreamDirection file_direction : stream_directions) {
+      const int in_rate = file_direction ? reverse_input_rate_ : input_rate_;
+      const int out_rate = file_direction ? reverse_output_rate_ : output_rate_;
+      const int out_num =
+          file_direction ? cf[i].num_reverse_output : cf[i].num_output;
+      const double expected_snr =
+          file_direction ? expected_reverse_snr_ : expected_snr_;
+
+      const int min_ref_rate = std::min(in_rate, out_rate);
+      int ref_rate;
+
+      if (min_ref_rate > 32000) {
+        ref_rate = 48000;
+      } else if (min_ref_rate > 16000) {
+        ref_rate = 32000;
+      } else if (min_ref_rate > 8000) {
+        ref_rate = 16000;
+      } else {
+        ref_rate = 8000;
+      }
 #ifdef WEBRTC_AUDIOPROC_FIXED_PROFILE
-    ref_rate = std::min(ref_rate, 16000);
+      if (file_direction == kForward) {
+        ref_rate = std::min(ref_rate, 16000);
+      }
 #endif
+      FILE* out_file = fopen(
+          OutputFilePath("out", input_rate_, output_rate_, reverse_input_rate_,
+                         reverse_output_rate_, cf[i].num_input,
+                         cf[i].num_output, cf[i].num_reverse_input,
+                         cf[i].num_reverse_output, file_direction).c_str(),
+          "rb");
+      // The reference files always have matching input and output channels.
+      FILE* ref_file = fopen(
+          OutputFilePath("ref", ref_rate, ref_rate, ref_rate, ref_rate,
+                         cf[i].num_output, cf[i].num_output,
+                         cf[i].num_reverse_output, cf[i].num_reverse_output,
+                         file_direction).c_str(),
+          "rb");
+      ASSERT_TRUE(out_file != NULL);
+      ASSERT_TRUE(ref_file != NULL);
 
-    FILE* out_file = fopen(OutputFilePath("out",
-                                          input_rate_,
-                                          output_rate_,
-                                          reverse_rate_,
-                                          cf[i].num_input,
-                                          cf[i].num_output,
-                                          cf[i].num_reverse).c_str(), "rb");
-    // The reference files always have matching input and output channels.
-    FILE* ref_file = fopen(OutputFilePath("ref",
-                                          ref_rate,
-                                          ref_rate,
-                                          ref_rate,
-                                          cf[i].num_output,
-                                          cf[i].num_output,
-                                          cf[i].num_reverse).c_str(), "rb");
-    ASSERT_TRUE(out_file != NULL);
-    ASSERT_TRUE(ref_file != NULL);
+      const size_t ref_length = SamplesFromRate(ref_rate) * out_num;
+      const size_t out_length = SamplesFromRate(out_rate) * out_num;
+      // Data from the reference file.
+      rtc::scoped_ptr<float[]> ref_data(new float[ref_length]);
+      // Data from the output file.
+      rtc::scoped_ptr<float[]> out_data(new float[out_length]);
+      // Data from the resampled output, in case the reference and output rates
+      // don't match.
+      rtc::scoped_ptr<float[]> cmp_data(new float[ref_length]);
 
-    const int ref_length = SamplesFromRate(ref_rate) * cf[i].num_output;
-    const int out_length = SamplesFromRate(output_rate_) * cf[i].num_output;
-    // Data from the reference file.
-    rtc::scoped_ptr<float[]> ref_data(new float[ref_length]);
-    // Data from the output file.
-    rtc::scoped_ptr<float[]> out_data(new float[out_length]);
-    // Data from the resampled output, in case the reference and output rates
-    // don't match.
-    rtc::scoped_ptr<float[]> cmp_data(new float[ref_length]);
+      PushResampler<float> resampler;
+      resampler.InitializeIfNeeded(out_rate, ref_rate, out_num);
 
-    PushResampler<float> resampler;
-    resampler.InitializeIfNeeded(output_rate_, ref_rate, cf[i].num_output);
+      // Compute the resampling delay of the output relative to the reference,
+      // to find the region over which we should search for the best SNR.
+      float expected_delay_sec = 0;
+      if (in_rate != ref_rate) {
+        // Input resampling delay.
+        expected_delay_sec +=
+            PushSincResampler::AlgorithmicDelaySeconds(in_rate);
+      }
+      if (out_rate != ref_rate) {
+        // Output resampling delay.
+        expected_delay_sec +=
+            PushSincResampler::AlgorithmicDelaySeconds(ref_rate);
+        // Delay of converting the output back to its processing rate for
+        // testing.
+        expected_delay_sec +=
+            PushSincResampler::AlgorithmicDelaySeconds(out_rate);
+      }
+      int expected_delay =
+          floor(expected_delay_sec * ref_rate + 0.5f) * out_num;
 
-    // Compute the resampling delay of the output relative to the reference,
-    // to find the region over which we should search for the best SNR.
-    float expected_delay_sec = 0;
-    if (input_rate_ != ref_rate) {
-      // Input resampling delay.
-      expected_delay_sec +=
-          PushSincResampler::AlgorithmicDelaySeconds(input_rate_);
-    }
-    if (output_rate_ != ref_rate) {
-      // Output resampling delay.
-      expected_delay_sec +=
-          PushSincResampler::AlgorithmicDelaySeconds(ref_rate);
-      // Delay of converting the output back to its processing rate for testing.
-      expected_delay_sec +=
-          PushSincResampler::AlgorithmicDelaySeconds(output_rate_);
-    }
-    int expected_delay = floor(expected_delay_sec * ref_rate + 0.5f) *
-                         cf[i].num_output;
+      double variance = 0;
+      double sq_error = 0;
+      while (fread(out_data.get(), sizeof(out_data[0]), out_length, out_file) &&
+             fread(ref_data.get(), sizeof(ref_data[0]), ref_length, ref_file)) {
+        float* out_ptr = out_data.get();
+        if (out_rate != ref_rate) {
+          // Resample the output back to its internal processing rate if
+          // necssary.
+          ASSERT_EQ(ref_length,
+                    static_cast<size_t>(resampler.Resample(
+                        out_ptr, out_length, cmp_data.get(), ref_length)));
+          out_ptr = cmp_data.get();
+        }
 
-    double variance = 0;
-    double sq_error = 0;
-    while (fread(out_data.get(), sizeof(out_data[0]), out_length, out_file) &&
-           fread(ref_data.get(), sizeof(ref_data[0]), ref_length, ref_file)) {
-      float* out_ptr = out_data.get();
-      if (output_rate_ != ref_rate) {
-        // Resample the output back to its internal processing rate if necssary.
-        ASSERT_EQ(ref_length, resampler.Resample(out_ptr,
-                                                 out_length,
-                                                 cmp_data.get(),
-                                                 ref_length));
-        out_ptr = cmp_data.get();
+        // Update the |sq_error| and |variance| accumulators with the highest
+        // SNR of reference vs output.
+        UpdateBestSNR(ref_data.get(), out_ptr, ref_length, expected_delay,
+                      &variance, &sq_error);
       }
 
-      // Update the |sq_error| and |variance| accumulators with the highest SNR
-      // of reference vs output.
-      UpdateBestSNR(ref_data.get(),
-                    out_ptr,
-                    ref_length,
-                    expected_delay,
-                    &variance,
-                    &sq_error);
-    }
+      std::cout << "(" << input_rate_ << ", " << output_rate_ << ", "
+                << reverse_input_rate_ << ", " << reverse_output_rate_ << ", "
+                << cf[i].num_input << ", " << cf[i].num_output << ", "
+                << cf[i].num_reverse_input << ", " << cf[i].num_reverse_output
+                << ", " << file_direction << "): ";
+      if (sq_error > 0) {
+        double snr = 10 * log10(variance / sq_error);
+        EXPECT_GE(snr, expected_snr);
+        EXPECT_NE(0, expected_snr);
+        std::cout << "SNR=" << snr << " dB" << std::endl;
+      } else {
+        EXPECT_EQ(expected_snr, 0);
+        std::cout << "SNR="
+                  << "inf dB" << std::endl;
+      }
 
-    std::cout << "(" << input_rate_ << ", "
-                     << output_rate_ << ", "
-                     << reverse_rate_ << ", "
-                     << cf[i].num_input << ", "
-                     << cf[i].num_output << ", "
-                     << cf[i].num_reverse << "): ";
-    if (sq_error > 0) {
-      double snr = 10 * log10(variance / sq_error);
-      EXPECT_GE(snr, expected_snr_);
-      EXPECT_NE(0, expected_snr_);
-      std::cout << "SNR=" << snr << " dB" << std::endl;
-    } else {
-      EXPECT_EQ(expected_snr_, 0);
-      std::cout << "SNR=" << "inf dB" << std::endl;
+      fclose(out_file);
+      fclose(ref_file);
     }
-
-    fclose(out_file);
-    fclose(ref_file);
   }
 }
 
 #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 INSTANTIATE_TEST_CASE_P(
-    CommonFormats, AudioProcessingTest, testing::Values(
-        std::tr1::make_tuple(48000, 48000, 48000, 20),
-        std::tr1::make_tuple(48000, 48000, 32000, 20),
-        std::tr1::make_tuple(48000, 48000, 16000, 20),
-        std::tr1::make_tuple(48000, 44100, 48000, 15),
-        std::tr1::make_tuple(48000, 44100, 32000, 15),
-        std::tr1::make_tuple(48000, 44100, 16000, 15),
-        std::tr1::make_tuple(48000, 32000, 48000, 20),
-        std::tr1::make_tuple(48000, 32000, 32000, 20),
-        std::tr1::make_tuple(48000, 32000, 16000, 20),
-        std::tr1::make_tuple(48000, 16000, 48000, 20),
-        std::tr1::make_tuple(48000, 16000, 32000, 20),
-        std::tr1::make_tuple(48000, 16000, 16000, 20),
+    CommonFormats,
+    AudioProcessingTest,
+    testing::Values(std::tr1::make_tuple(48000, 48000, 48000, 48000, 0, 0),
+                    std::tr1::make_tuple(48000, 48000, 32000, 48000, 40, 30),
+                    std::tr1::make_tuple(48000, 48000, 16000, 48000, 40, 20),
+                    std::tr1::make_tuple(48000, 44100, 48000, 44100, 20, 20),
+                    std::tr1::make_tuple(48000, 44100, 32000, 44100, 20, 15),
+                    std::tr1::make_tuple(48000, 44100, 16000, 44100, 20, 15),
+                    std::tr1::make_tuple(48000, 32000, 48000, 32000, 30, 35),
+                    std::tr1::make_tuple(48000, 32000, 32000, 32000, 30, 0),
+                    std::tr1::make_tuple(48000, 32000, 16000, 32000, 30, 20),
+                    std::tr1::make_tuple(48000, 16000, 48000, 16000, 25, 20),
+                    std::tr1::make_tuple(48000, 16000, 32000, 16000, 25, 20),
+                    std::tr1::make_tuple(48000, 16000, 16000, 16000, 25, 0),
 
-        std::tr1::make_tuple(44100, 48000, 48000, 20),
-        std::tr1::make_tuple(44100, 48000, 32000, 20),
-        std::tr1::make_tuple(44100, 48000, 16000, 20),
-        std::tr1::make_tuple(44100, 44100, 48000, 15),
-        std::tr1::make_tuple(44100, 44100, 32000, 15),
-        std::tr1::make_tuple(44100, 44100, 16000, 15),
-        std::tr1::make_tuple(44100, 32000, 48000, 20),
-        std::tr1::make_tuple(44100, 32000, 32000, 20),
-        std::tr1::make_tuple(44100, 32000, 16000, 20),
-        std::tr1::make_tuple(44100, 16000, 48000, 20),
-        std::tr1::make_tuple(44100, 16000, 32000, 20),
-        std::tr1::make_tuple(44100, 16000, 16000, 20),
+                    std::tr1::make_tuple(44100, 48000, 48000, 48000, 30, 0),
+                    std::tr1::make_tuple(44100, 48000, 32000, 48000, 30, 30),
+                    std::tr1::make_tuple(44100, 48000, 16000, 48000, 30, 20),
+                    std::tr1::make_tuple(44100, 44100, 48000, 44100, 20, 20),
+                    std::tr1::make_tuple(44100, 44100, 32000, 44100, 20, 15),
+                    std::tr1::make_tuple(44100, 44100, 16000, 44100, 20, 15),
+                    std::tr1::make_tuple(44100, 32000, 48000, 32000, 30, 35),
+                    std::tr1::make_tuple(44100, 32000, 32000, 32000, 30, 0),
+                    std::tr1::make_tuple(44100, 32000, 16000, 32000, 30, 20),
+                    std::tr1::make_tuple(44100, 16000, 48000, 16000, 25, 20),
+                    std::tr1::make_tuple(44100, 16000, 32000, 16000, 25, 20),
+                    std::tr1::make_tuple(44100, 16000, 16000, 16000, 25, 0),
 
-        std::tr1::make_tuple(32000, 48000, 48000, 25),
-        std::tr1::make_tuple(32000, 48000, 32000, 25),
-        std::tr1::make_tuple(32000, 48000, 16000, 25),
-        std::tr1::make_tuple(32000, 44100, 48000, 20),
-        std::tr1::make_tuple(32000, 44100, 32000, 20),
-        std::tr1::make_tuple(32000, 44100, 16000, 20),
-        std::tr1::make_tuple(32000, 32000, 48000, 30),
-        std::tr1::make_tuple(32000, 32000, 32000, 0),
-        std::tr1::make_tuple(32000, 32000, 16000, 30),
-        std::tr1::make_tuple(32000, 16000, 48000, 20),
-        std::tr1::make_tuple(32000, 16000, 32000, 20),
-        std::tr1::make_tuple(32000, 16000, 16000, 20),
+                    std::tr1::make_tuple(32000, 48000, 48000, 48000, 30, 0),
+                    std::tr1::make_tuple(32000, 48000, 32000, 48000, 35, 30),
+                    std::tr1::make_tuple(32000, 48000, 16000, 48000, 30, 20),
+                    std::tr1::make_tuple(32000, 44100, 48000, 44100, 20, 20),
+                    std::tr1::make_tuple(32000, 44100, 32000, 44100, 20, 15),
+                    std::tr1::make_tuple(32000, 44100, 16000, 44100, 20, 15),
+                    std::tr1::make_tuple(32000, 32000, 48000, 32000, 40, 35),
+                    std::tr1::make_tuple(32000, 32000, 32000, 32000, 0, 0),
+                    std::tr1::make_tuple(32000, 32000, 16000, 32000, 40, 20),
+                    std::tr1::make_tuple(32000, 16000, 48000, 16000, 25, 20),
+                    std::tr1::make_tuple(32000, 16000, 32000, 16000, 25, 20),
+                    std::tr1::make_tuple(32000, 16000, 16000, 16000, 25, 0),
 
-        std::tr1::make_tuple(16000, 48000, 48000, 25),
-        std::tr1::make_tuple(16000, 48000, 32000, 25),
-        std::tr1::make_tuple(16000, 48000, 16000, 25),
-        std::tr1::make_tuple(16000, 44100, 48000, 15),
-        std::tr1::make_tuple(16000, 44100, 32000, 15),
-        std::tr1::make_tuple(16000, 44100, 16000, 15),
-        std::tr1::make_tuple(16000, 32000, 48000, 25),
-        std::tr1::make_tuple(16000, 32000, 32000, 25),
-        std::tr1::make_tuple(16000, 32000, 16000, 25),
-        std::tr1::make_tuple(16000, 16000, 48000, 30),
-        std::tr1::make_tuple(16000, 16000, 32000, 30),
-        std::tr1::make_tuple(16000, 16000, 16000, 0)));
+                    std::tr1::make_tuple(16000, 48000, 48000, 48000, 25, 0),
+                    std::tr1::make_tuple(16000, 48000, 32000, 48000, 25, 30),
+                    std::tr1::make_tuple(16000, 48000, 16000, 48000, 25, 20),
+                    std::tr1::make_tuple(16000, 44100, 48000, 44100, 15, 20),
+                    std::tr1::make_tuple(16000, 44100, 32000, 44100, 15, 15),
+                    std::tr1::make_tuple(16000, 44100, 16000, 44100, 15, 15),
+                    std::tr1::make_tuple(16000, 32000, 48000, 32000, 25, 35),
+                    std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
+                    std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
+                    std::tr1::make_tuple(16000, 16000, 48000, 16000, 40, 20),
+                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 50, 20),
+                    std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 
 #elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
 INSTANTIATE_TEST_CASE_P(
-    CommonFormats, AudioProcessingTest, testing::Values(
-        std::tr1::make_tuple(48000, 48000, 48000, 20),
-        std::tr1::make_tuple(48000, 48000, 32000, 20),
-        std::tr1::make_tuple(48000, 48000, 16000, 20),
-        std::tr1::make_tuple(48000, 44100, 48000, 15),
-        std::tr1::make_tuple(48000, 44100, 32000, 15),
-        std::tr1::make_tuple(48000, 44100, 16000, 15),
-        std::tr1::make_tuple(48000, 32000, 48000, 20),
-        std::tr1::make_tuple(48000, 32000, 32000, 20),
-        std::tr1::make_tuple(48000, 32000, 16000, 20),
-        std::tr1::make_tuple(48000, 16000, 48000, 20),
-        std::tr1::make_tuple(48000, 16000, 32000, 20),
-        std::tr1::make_tuple(48000, 16000, 16000, 20),
+    CommonFormats,
+    AudioProcessingTest,
+    testing::Values(std::tr1::make_tuple(48000, 48000, 48000, 48000, 20, 0),
+                    std::tr1::make_tuple(48000, 48000, 32000, 48000, 20, 30),
+                    std::tr1::make_tuple(48000, 48000, 16000, 48000, 20, 20),
+                    std::tr1::make_tuple(48000, 44100, 48000, 44100, 15, 20),
+                    std::tr1::make_tuple(48000, 44100, 32000, 44100, 15, 15),
+                    std::tr1::make_tuple(48000, 44100, 16000, 44100, 15, 15),
+                    std::tr1::make_tuple(48000, 32000, 48000, 32000, 20, 35),
+                    std::tr1::make_tuple(48000, 32000, 32000, 32000, 20, 0),
+                    std::tr1::make_tuple(48000, 32000, 16000, 32000, 20, 20),
+                    std::tr1::make_tuple(48000, 16000, 48000, 16000, 20, 20),
+                    std::tr1::make_tuple(48000, 16000, 32000, 16000, 20, 20),
+                    std::tr1::make_tuple(48000, 16000, 16000, 16000, 20, 0),
 
-        std::tr1::make_tuple(44100, 48000, 48000, 19),
-        std::tr1::make_tuple(44100, 48000, 32000, 19),
-        std::tr1::make_tuple(44100, 48000, 16000, 19),
-        std::tr1::make_tuple(44100, 44100, 48000, 15),
-        std::tr1::make_tuple(44100, 44100, 32000, 15),
-        std::tr1::make_tuple(44100, 44100, 16000, 15),
-        std::tr1::make_tuple(44100, 32000, 48000, 19),
-        std::tr1::make_tuple(44100, 32000, 32000, 19),
-        std::tr1::make_tuple(44100, 32000, 16000, 19),
-        std::tr1::make_tuple(44100, 16000, 48000, 19),
-        std::tr1::make_tuple(44100, 16000, 32000, 19),
-        std::tr1::make_tuple(44100, 16000, 16000, 19),
+                    std::tr1::make_tuple(44100, 48000, 48000, 48000, 20, 0),
+                    std::tr1::make_tuple(44100, 48000, 32000, 48000, 20, 30),
+                    std::tr1::make_tuple(44100, 48000, 16000, 48000, 20, 20),
+                    std::tr1::make_tuple(44100, 44100, 48000, 44100, 15, 20),
+                    std::tr1::make_tuple(44100, 44100, 32000, 44100, 15, 15),
+                    std::tr1::make_tuple(44100, 44100, 16000, 44100, 15, 15),
+                    std::tr1::make_tuple(44100, 32000, 48000, 32000, 20, 35),
+                    std::tr1::make_tuple(44100, 32000, 32000, 32000, 20, 0),
+                    std::tr1::make_tuple(44100, 32000, 16000, 32000, 20, 20),
+                    std::tr1::make_tuple(44100, 16000, 48000, 16000, 20, 20),
+                    std::tr1::make_tuple(44100, 16000, 32000, 16000, 20, 20),
+                    std::tr1::make_tuple(44100, 16000, 16000, 16000, 20, 0),
 
-        std::tr1::make_tuple(32000, 48000, 48000, 19),
-        std::tr1::make_tuple(32000, 48000, 32000, 19),
-        std::tr1::make_tuple(32000, 48000, 16000, 19),
-        std::tr1::make_tuple(32000, 44100, 48000, 15),
-        std::tr1::make_tuple(32000, 44100, 32000, 15),
-        std::tr1::make_tuple(32000, 44100, 16000, 15),
-        std::tr1::make_tuple(32000, 32000, 48000, 19),
-        std::tr1::make_tuple(32000, 32000, 32000, 19),
-        std::tr1::make_tuple(32000, 32000, 16000, 19),
-        std::tr1::make_tuple(32000, 16000, 48000, 19),
-        std::tr1::make_tuple(32000, 16000, 32000, 19),
-        std::tr1::make_tuple(32000, 16000, 16000, 19),
+                    std::tr1::make_tuple(32000, 48000, 48000, 48000, 20, 0),
+                    std::tr1::make_tuple(32000, 48000, 32000, 48000, 20, 30),
+                    std::tr1::make_tuple(32000, 48000, 16000, 48000, 20, 20),
+                    std::tr1::make_tuple(32000, 44100, 48000, 44100, 15, 20),
+                    std::tr1::make_tuple(32000, 44100, 32000, 44100, 15, 15),
+                    std::tr1::make_tuple(32000, 44100, 16000, 44100, 15, 15),
+                    std::tr1::make_tuple(32000, 32000, 48000, 32000, 20, 35),
+                    std::tr1::make_tuple(32000, 32000, 32000, 32000, 20, 0),
+                    std::tr1::make_tuple(32000, 32000, 16000, 32000, 20, 20),
+                    std::tr1::make_tuple(32000, 16000, 48000, 16000, 20, 20),
+                    std::tr1::make_tuple(32000, 16000, 32000, 16000, 20, 20),
+                    std::tr1::make_tuple(32000, 16000, 16000, 16000, 20, 0),
 
-        std::tr1::make_tuple(16000, 48000, 48000, 25),
-        std::tr1::make_tuple(16000, 48000, 32000, 25),
-        std::tr1::make_tuple(16000, 48000, 16000, 25),
-        std::tr1::make_tuple(16000, 44100, 48000, 15),
-        std::tr1::make_tuple(16000, 44100, 32000, 15),
-        std::tr1::make_tuple(16000, 44100, 16000, 15),
-        std::tr1::make_tuple(16000, 32000, 48000, 25),
-        std::tr1::make_tuple(16000, 32000, 32000, 25),
-        std::tr1::make_tuple(16000, 32000, 16000, 25),
-        std::tr1::make_tuple(16000, 16000, 48000, 30),
-        std::tr1::make_tuple(16000, 16000, 32000, 30),
-        std::tr1::make_tuple(16000, 16000, 16000, 0)));
+                    std::tr1::make_tuple(16000, 48000, 48000, 48000, 25, 0),
+                    std::tr1::make_tuple(16000, 48000, 32000, 48000, 25, 30),
+                    std::tr1::make_tuple(16000, 48000, 16000, 48000, 25, 20),
+                    std::tr1::make_tuple(16000, 44100, 48000, 44100, 15, 20),
+                    std::tr1::make_tuple(16000, 44100, 32000, 44100, 15, 15),
+                    std::tr1::make_tuple(16000, 44100, 16000, 44100, 15, 15),
+                    std::tr1::make_tuple(16000, 32000, 48000, 32000, 25, 35),
+                    std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
+                    std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
+                    std::tr1::make_tuple(16000, 16000, 48000, 16000, 35, 20),
+                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
+                    std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 #endif
 
-// TODO(henrike): re-implement functionality lost when removing the old main
-//                function. See
-//                https://code.google.com/p/webrtc/issues/detail?id=1981
-
 }  // namespace
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/audioproc_float.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audioproc_float.cc
index a451d0a4cd..a489d255c8 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -9,29 +9,50 @@
  */
 
 #include <stdio.h>
+#include <iostream>
 #include <sstream>
 #include <string>
+#include <utility>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
 
-DEFINE_string(dump, "", "The name of the debug dump file to read from.");
-DEFINE_string(c, "", "The name of the capture input file to read from.");
-DEFINE_string(o, "out.wav", "Name of the capture output file to write to.");
-DEFINE_int32(o_channels, 0, "Number of output channels. Defaults to input.");
-DEFINE_int32(o_sample_rate, 0, "Output sample rate in Hz. Defaults to input.");
-DEFINE_double(mic_spacing, 0.0,
-    "Alternate way to specify mic_positions. "
-    "Assumes uniform linear array with specified spacings.");
+namespace {
+
+bool ValidateOutChannels(const char* flagname, int32_t value) {
+  return value >= 0;
+}
+
+}  // namespace
+
+DEFINE_string(dump, "", "Name of the aecdump debug file to read from.");
+DEFINE_string(i, "", "Name of the capture input stream file to read from.");
+DEFINE_string(
+    o,
+    "out.wav",
+    "Name of the output file to write the processed capture stream to.");
+DEFINE_int32(out_channels, 1, "Number of output channels.");
+const bool out_channels_dummy =
+    google::RegisterFlagValidator(&FLAGS_out_channels, &ValidateOutChannels);
+DEFINE_int32(out_sample_rate, 48000, "Output sample rate in Hz.");
 DEFINE_string(mic_positions, "",
     "Space delimited cartesian coordinates of microphones in meters. "
     "The coordinates of each point are contiguous. "
     "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
+DEFINE_double(
+    target_angle_degrees,
+    90,
+    "The azimuth of the target in degrees. Only applies to beamforming.");
 
 DEFINE_bool(aec, false, "Enable echo cancellation.");
 DEFINE_bool(agc, false, "Enable automatic gain control.");
@@ -39,12 +60,18 @@ DEFINE_bool(hpf, false, "Enable high-pass filtering.");
 DEFINE_bool(ns, false, "Enable noise suppression.");
 DEFINE_bool(ts, false, "Enable transient suppression.");
 DEFINE_bool(bf, false, "Enable beamforming.");
+DEFINE_bool(ie, false, "Enable intelligibility enhancer.");
 DEFINE_bool(all, false, "Enable all components.");
 
 DEFINE_int32(ns_level, -1, "Noise suppression level [0 - 3].");
 
-static const int kChunksPerSecond = 100;
-static const char kUsage[] =
+DEFINE_bool(perf, false, "Enable performance tests.");
+
+namespace webrtc {
+namespace {
+
+const int kChunksPerSecond = 100;
+const char kUsage[] =
     "Command-line tool to run audio processing on WAV files. Accepts either\n"
     "an input capture WAV file or protobuf debug dump and writes to an output\n"
     "WAV file.\n"
@@ -52,155 +79,84 @@ static const char kUsage[] =
     "All components are disabled by default. If any bi-directional components\n"
     "are enabled, only debug dump files are permitted.";
 
-namespace webrtc {
-
-namespace {
-
-// Returns a vector<T> parsed from whitespace delimited values in to_parse,
-// or an empty vector if the string could not be parsed.
-template<typename T>
-std::vector<T> parse_list(std::string to_parse) {
-  std::vector<T> values;
-
-  std::istringstream str(to_parse);
-  std::copy(
-      std::istream_iterator<T>(str),
-      std::istream_iterator<T>(),
-      std::back_inserter(values));
-
-  return values;
-}
-
-// Parses the array geometry from the command line.
-//
-// If a vector with size != num_mics is returned, an error has occurred and an
-// appropriate error message has been printed to stdout.
-std::vector<Point> get_array_geometry(size_t num_mics) {
-  std::vector<Point> result;
-  result.reserve(num_mics);
-
-  if (FLAGS_mic_positions.length()) {
-    CHECK(FLAGS_mic_spacing == 0.0 &&
-        "mic_positions and mic_spacing should not both be specified");
-
-    const std::vector<float> values = parse_list<float>(FLAGS_mic_positions);
-    if (values.size() != 3 * num_mics) {
-      fprintf(stderr,
-          "Could not parse mic_positions or incorrect number of points.\n");
-    } else {
-      for (size_t i = 0; i < values.size(); i += 3) {
-        double x = values[i + 0];
-        double y = values[i + 1];
-        double z = values[i + 2];
-        result.push_back(Point(x, y, z));
-      }
-    }
-  } else {
-    if (FLAGS_mic_spacing <= 0) {
-      fprintf(stderr,
-          "mic_spacing must a positive value when beamforming is enabled.\n");
-    } else {
-      for (size_t i = 0; i < num_mics; ++i) {
-        result.push_back(Point(i * FLAGS_mic_spacing, 0.f, 0.f));
-      }
-    }
-  }
-
-  return result;
-}
-
 }  // namespace
 
 int main(int argc, char* argv[]) {
-  {
-    const std::string program_name = argv[0];
-    const std::string usage = kUsage;
-    google::SetUsageMessage(usage);
-  }
+  google::SetUsageMessage(kUsage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
-  if (!((FLAGS_c == "") ^ (FLAGS_dump == ""))) {
+  if (!((FLAGS_i.empty()) ^ (FLAGS_dump.empty()))) {
     fprintf(stderr,
-            "An input file must be specified with either -c or -dump.\n");
+            "An input file must be specified with either -i or -dump.\n");
     return 1;
   }
-  if (FLAGS_dump != "") {
-    fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
+  if (FLAGS_dump.empty() && (FLAGS_aec || FLAGS_ie)) {
+    fprintf(stderr, "-aec and -ie require a -dump file.\n");
+    return 1;
+  }
+  if (FLAGS_ie) {
+    fprintf(stderr,
+            "FIXME(ajm): The intelligibility enhancer output is not dumped.\n");
     return 1;
   }
 
-  WavReader c_file(FLAGS_c);
-  // If the output format is uninitialized, use the input format.
-  int o_channels = FLAGS_o_channels;
-  if (!o_channels)
-    o_channels = c_file.num_channels();
-  int o_sample_rate = FLAGS_o_sample_rate;
-  if (!o_sample_rate)
-    o_sample_rate = c_file.sample_rate();
-  WavWriter o_file(FLAGS_o, o_sample_rate, o_channels);
-
+  test::TraceToStderr trace_to_stderr(true);
   Config config;
-  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
-
   if (FLAGS_bf || FLAGS_all) {
-    const size_t num_mics = c_file.num_channels();
-    const std::vector<Point> array_geometry = get_array_geometry(num_mics);
-    if (array_geometry.size() != num_mics) {
+    if (FLAGS_mic_positions.empty()) {
+      fprintf(stderr, "-mic_positions must be specified when -bf is used.\n");
       return 1;
     }
-
-    config.Set<Beamforming>(new Beamforming(true, array_geometry));
+    config.Set<Beamforming>(new Beamforming(
+        true, ParseArrayGeometry(FLAGS_mic_positions),
+        SphericalPointf(DegreesToRadians(FLAGS_target_angle_degrees), 0.f,
+                        1.f)));
   }
+  config.Set<ExperimentalNs>(new ExperimentalNs(FLAGS_ts || FLAGS_all));
+  config.Set<Intelligibility>(new Intelligibility(FLAGS_ie || FLAGS_all));
 
   rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
-  if (FLAGS_dump != "") {
-    CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
-  } else if (FLAGS_aec) {
-    fprintf(stderr, "-aec requires a -dump file.\n");
-    return -1;
+  RTC_CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
+  RTC_CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
+  RTC_CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
+  RTC_CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
+  if (FLAGS_ns_level != -1) {
+    RTC_CHECK_EQ(kNoErr,
+                 ap->noise_suppression()->set_level(
+                     static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
   }
-  CHECK_EQ(kNoErr, ap->gain_control()->Enable(FLAGS_agc || FLAGS_all));
-  CHECK_EQ(kNoErr, ap->gain_control()->set_mode(GainControl::kFixedDigital));
-  CHECK_EQ(kNoErr, ap->high_pass_filter()->Enable(FLAGS_hpf || FLAGS_all));
-  CHECK_EQ(kNoErr, ap->noise_suppression()->Enable(FLAGS_ns || FLAGS_all));
-  if (FLAGS_ns_level != -1)
-    CHECK_EQ(kNoErr, ap->noise_suppression()->set_level(
-        static_cast<NoiseSuppression::Level>(FLAGS_ns_level)));
+  ap->set_stream_key_pressed(FLAGS_ts);
 
-  printf("Input file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-         FLAGS_c.c_str(), c_file.num_channels(), c_file.sample_rate());
-  printf("Output file: %s\nChannels: %d, Sample rate: %d Hz\n\n",
-         FLAGS_o.c_str(), o_file.num_channels(), o_file.sample_rate());
+  rtc::scoped_ptr<AudioFileProcessor> processor;
+  auto out_file = rtc_make_scoped_ptr(new WavWriter(
+      FLAGS_o, FLAGS_out_sample_rate, static_cast<size_t>(FLAGS_out_channels)));
+  std::cout << FLAGS_o << ": " << out_file->FormatAsString() << std::endl;
+  if (FLAGS_dump.empty()) {
+    auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
+    std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl;
+    processor.reset(new WavFileProcessor(std::move(ap), std::move(in_file),
+                                         std::move(out_file)));
 
-  ChannelBuffer<float> c_buf(c_file.sample_rate() / kChunksPerSecond,
-                             c_file.num_channels());
-  ChannelBuffer<float> o_buf(o_file.sample_rate() / kChunksPerSecond,
-                             o_file.num_channels());
+  } else {
+    processor.reset(new AecDumpFileProcessor(
+        std::move(ap), fopen(FLAGS_dump.c_str(), "rb"), std::move(out_file)));
+  }
 
-  const size_t c_length =
-      static_cast<size_t>(c_buf.num_channels() * c_buf.num_frames());
-  const size_t o_length =
-      static_cast<size_t>(o_buf.num_channels() * o_buf.num_frames());
-  rtc::scoped_ptr<float[]> c_interleaved(new float[c_length]);
-  rtc::scoped_ptr<float[]> o_interleaved(new float[o_length]);
-  while (c_file.ReadSamples(c_length, c_interleaved.get()) == c_length) {
-    FloatS16ToFloat(c_interleaved.get(), c_length, c_interleaved.get());
-    Deinterleave(c_interleaved.get(), c_buf.num_frames(),
-                 c_buf.num_channels(), c_buf.channels());
+  int num_chunks = 0;
+  while (processor->ProcessChunk()) {
+    trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond);
+    ++num_chunks;
+  }
 
-    CHECK_EQ(kNoErr,
-        ap->ProcessStream(c_buf.channels(),
-                          c_buf.num_frames(),
-                          c_file.sample_rate(),
-                          LayoutFromChannels(c_buf.num_channels()),
-                          o_file.sample_rate(),
-                          LayoutFromChannels(o_buf.num_channels()),
-                          o_buf.channels()));
-
-    Interleave(o_buf.channels(), o_buf.num_frames(),
-               o_buf.num_channels(), o_interleaved.get());
-    FloatToFloatS16(o_interleaved.get(), o_length, o_interleaved.get());
-    o_file.WriteSamples(o_interleaved.get(), o_length);
+  if (FLAGS_perf) {
+    const auto& proc_time = processor->proc_time();
+    int64_t exec_time_us = proc_time.sum.Microseconds();
+    printf(
+        "\nExecution time: %.3f s, File time: %.2f s\n"
+        "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n",
+        exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond,
+        exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(),
+        1.f * proc_time.min.Microseconds());
   }
 
   return 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/debug_dump_test.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/debug_dump_test.cc
new file mode 100644
index 0000000000..005faa0f44
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -0,0 +1,612 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>  // size_t
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+void MaybeResetBuffer(rtc::scoped_ptr<ChannelBuffer<float>>* buffer,
+                      const StreamConfig& config) {
+  auto& buffer_ref = *buffer;
+  if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
+      buffer_ref->num_channels() != config.num_channels()) {
+    buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
+                                             config.num_channels()));
+  }
+}
+
+class DebugDumpGenerator {
+ public:
+  DebugDumpGenerator(const std::string& input_file_name,
+                     int input_file_rate_hz,
+                     int input_channels,
+                     const std::string& reverse_file_name,
+                     int reverse_file_rate_hz,
+                     int reverse_channels,
+                     const Config& config,
+                     const std::string& dump_file_name);
+
+  // Constructor that uses default input files.
+  explicit DebugDumpGenerator(const Config& config);
+
+  ~DebugDumpGenerator();
+
+  // Changes the sample rate of the input audio to the APM.
+  void SetInputRate(int rate_hz);
+
+  // Sets if converts stereo input signal to mono by discarding other channels.
+  void ForceInputMono(bool mono);
+
+  // Changes the sample rate of the reverse audio to the APM.
+  void SetReverseRate(int rate_hz);
+
+  // Sets if converts stereo reverse signal to mono by discarding other
+  // channels.
+  void ForceReverseMono(bool mono);
+
+  // Sets the required sample rate of the APM output.
+  void SetOutputRate(int rate_hz);
+
+  // Sets the required channels of the APM output.
+  void SetOutputChannels(int channels);
+
+  std::string dump_file_name() const { return dump_file_name_; }
+
+  void StartRecording();
+  void Process(size_t num_blocks);
+  void StopRecording();
+  AudioProcessing* apm() const { return apm_.get(); }
+
+ private:
+  static void ReadAndDeinterleave(ResampleInputAudioFile* audio, int channels,
+                                  const StreamConfig& config,
+                                  float* const* buffer);
+
+  // APM input/output settings.
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  StreamConfig output_config_;
+
+  // Input file format.
+  const std::string input_file_name_;
+  ResampleInputAudioFile input_audio_;
+  const int input_file_channels_;
+
+  // Reverse file format.
+  const std::string reverse_file_name_;
+  ResampleInputAudioFile reverse_audio_;
+  const int reverse_file_channels_;
+
+  // Buffer for APM input/output.
+  rtc::scoped_ptr<ChannelBuffer<float>> input_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
+  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+
+  const std::string dump_file_name_;
+};
+
+DebugDumpGenerator::DebugDumpGenerator(const std::string& input_file_name,
+                                       int input_rate_hz,
+                                       int input_channels,
+                                       const std::string& reverse_file_name,
+                                       int reverse_rate_hz,
+                                       int reverse_channels,
+                                       const Config& config,
+                                       const std::string& dump_file_name)
+    : input_config_(input_rate_hz, input_channels),
+      reverse_config_(reverse_rate_hz, reverse_channels),
+      output_config_(input_rate_hz, input_channels),
+      input_audio_(input_file_name, input_rate_hz, input_rate_hz),
+      input_file_channels_(input_channels),
+      reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz),
+      reverse_file_channels_(reverse_channels),
+      input_(new ChannelBuffer<float>(input_config_.num_frames(),
+                                      input_config_.num_channels())),
+      reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(),
+                                        reverse_config_.num_channels())),
+      output_(new ChannelBuffer<float>(output_config_.num_frames(),
+                                       output_config_.num_channels())),
+      apm_(AudioProcessing::Create(config)),
+      dump_file_name_(dump_file_name) {
+}
+
+DebugDumpGenerator::DebugDumpGenerator(const Config& config)
+  : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"), 32000, 2,
+                       ResourcePath("far32_stereo", "pcm"), 32000, 2,
+                       config,
+                       TempFilename(OutputPath(), "debug_aec")) {
+}
+
+DebugDumpGenerator::~DebugDumpGenerator() {
+  remove(dump_file_name_.c_str());
+}
+
+void DebugDumpGenerator::SetInputRate(int rate_hz) {
+  input_audio_.set_output_rate_hz(rate_hz);
+  input_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&input_, input_config_);
+}
+
+void DebugDumpGenerator::ForceInputMono(bool mono) {
+  const int channels = mono ? 1 : input_file_channels_;
+  input_config_.set_num_channels(channels);
+  MaybeResetBuffer(&input_, input_config_);
+}
+
+void DebugDumpGenerator::SetReverseRate(int rate_hz) {
+  reverse_audio_.set_output_rate_hz(rate_hz);
+  reverse_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+void DebugDumpGenerator::ForceReverseMono(bool mono) {
+  const int channels = mono ? 1 : reverse_file_channels_;
+  reverse_config_.set_num_channels(channels);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+void DebugDumpGenerator::SetOutputRate(int rate_hz) {
+  output_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&output_, output_config_);
+}
+
+void DebugDumpGenerator::SetOutputChannels(int channels) {
+  output_config_.set_num_channels(channels);
+  MaybeResetBuffer(&output_, output_config_);
+}
+
+void DebugDumpGenerator::StartRecording() {
+  apm_->StartDebugRecording(dump_file_name_.c_str());
+}
+
+void DebugDumpGenerator::Process(size_t num_blocks) {
+  for (size_t i = 0; i < num_blocks; ++i) {
+    ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_,
+                        reverse_config_, reverse_->channels());
+    ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
+                        input_->channels());
+    RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
+    apm_->set_stream_key_pressed(i % 10 == 9);
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->ProcessStream(input_->channels(), input_config_,
+                                     output_config_, output_->channels()));
+
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->ProcessReverseStream(reverse_->channels(),
+                                            reverse_config_,
+                                            reverse_config_,
+                                            reverse_->channels()));
+  }
+}
+
+void DebugDumpGenerator::StopRecording() {
+  apm_->StopDebugRecording();
+}
+
+void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,
+                                             int channels,
+                                             const StreamConfig& config,
+                                             float* const* buffer) {
+  const size_t num_frames = config.num_frames();
+  const int out_channels = config.num_channels();
+
+  std::vector<int16_t> signal(channels * num_frames);
+
+  audio->Read(num_frames * channels, &signal[0]);
+
+  // We only allow reducing number of channels by discarding some channels.
+  RTC_CHECK_LE(out_channels, channels);
+  for (int channel = 0; channel < out_channels; ++channel) {
+    for (size_t i = 0; i < num_frames; ++i) {
+      buffer[channel][i] = S16ToFloat(signal[i * channels + channel]);
+    }
+  }
+}
+
+}  // namespace
+
+class DebugDumpTest : public ::testing::Test {
+ public:
+  DebugDumpTest();
+
+  // VerifyDebugDump replays a debug dump using APM and verifies that the result
+  // is bit-exact-identical to the output channel in the dump. This is only
+  // guaranteed if the debug dump is started on the first frame.
+  void VerifyDebugDump(const std::string& dump_file_name);
+
+ private:
+  // Following functions are facilities for replaying debug dumps.
+  void OnInitEvent(const audioproc::Init& msg);
+  void OnStreamEvent(const audioproc::Stream& msg);
+  void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
+  void OnConfigEvent(const audioproc::Config& msg);
+
+  void MaybeRecreateApm(const audioproc::Config& msg);
+  void ConfigureApm(const audioproc::Config& msg);
+
+  // Buffer for APM input/output.
+  rtc::scoped_ptr<ChannelBuffer<float>> input_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
+  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  StreamConfig output_config_;
+};
+
+DebugDumpTest::DebugDumpTest()
+    : input_(nullptr),  // will be created upon usage.
+      reverse_(nullptr),
+      output_(nullptr),
+      apm_(nullptr) {
+}
+
+void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
+  FILE* in_file = fopen(in_filename.c_str(), "rb");
+  ASSERT_TRUE(in_file);
+  audioproc::Event event_msg;
+
+  while (ReadMessageFromFile(in_file, &event_msg)) {
+    switch (event_msg.type()) {
+      case audioproc::Event::INIT:
+        OnInitEvent(event_msg.init());
+        break;
+      case audioproc::Event::STREAM:
+        OnStreamEvent(event_msg.stream());
+        break;
+      case audioproc::Event::REVERSE_STREAM:
+        OnReverseStreamEvent(event_msg.reverse_stream());
+        break;
+      case audioproc::Event::CONFIG:
+        OnConfigEvent(event_msg.config());
+        break;
+      case audioproc::Event::UNKNOWN_EVENT:
+        // We do not expect receive UNKNOWN event currently.
+        FAIL();
+    }
+  }
+  fclose(in_file);
+}
+
+// OnInitEvent reset the input/output/reserve channel format.
+void DebugDumpTest::OnInitEvent(const audioproc::Init& msg) {
+  ASSERT_TRUE(msg.has_num_input_channels());
+  ASSERT_TRUE(msg.has_output_sample_rate());
+  ASSERT_TRUE(msg.has_num_output_channels());
+  ASSERT_TRUE(msg.has_reverse_sample_rate());
+  ASSERT_TRUE(msg.has_num_reverse_channels());
+
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  output_config_ =
+      StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
+  reverse_config_ =
+      StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
+
+  MaybeResetBuffer(&input_, input_config_);
+  MaybeResetBuffer(&output_, output_config_);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+// OnStreamEvent replays an input signal and verifies the output.
+void DebugDumpTest::OnStreamEvent(const audioproc::Stream& msg) {
+  // APM should have been created.
+  ASSERT_TRUE(apm_.get());
+
+  EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(msg.level()));
+  EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
+  apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+  if (msg.has_keypress())
+    apm_->set_stream_key_pressed(msg.keypress());
+  else
+    apm_->set_stream_key_pressed(true);
+
+  ASSERT_EQ(input_config_.num_channels(),
+            static_cast<size_t>(msg.input_channel_size()));
+  ASSERT_EQ(input_config_.num_frames() * sizeof(float),
+            msg.input_channel(0).size());
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+     memcpy(input_->channels()[i], msg.input_channel(i).data(),
+            msg.input_channel(i).size());
+  }
+
+  ASSERT_EQ(AudioProcessing::kNoError,
+            apm_->ProcessStream(input_->channels(), input_config_,
+                                output_config_, output_->channels()));
+
+  // Check that output of APM is bit-exact to the output in the dump.
+  ASSERT_EQ(output_config_.num_channels(),
+            static_cast<size_t>(msg.output_channel_size()));
+  ASSERT_EQ(output_config_.num_frames() * sizeof(float),
+            msg.output_channel(0).size());
+  for (int i = 0; i < msg.output_channel_size(); ++i) {
+    ASSERT_EQ(0, memcmp(output_->channels()[i], msg.output_channel(i).data(),
+                        msg.output_channel(i).size()));
+  }
+}
+
+void DebugDumpTest::OnReverseStreamEvent(const audioproc::ReverseStream& msg) {
+  // APM should have been created.
+  ASSERT_TRUE(apm_.get());
+
+  ASSERT_GT(msg.channel_size(), 0);
+  ASSERT_EQ(reverse_config_.num_channels(),
+            static_cast<size_t>(msg.channel_size()));
+  ASSERT_EQ(reverse_config_.num_frames() * sizeof(float),
+            msg.channel(0).size());
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+     memcpy(reverse_->channels()[i], msg.channel(i).data(),
+            msg.channel(i).size());
+  }
+
+  ASSERT_EQ(AudioProcessing::kNoError,
+            apm_->ProcessReverseStream(reverse_->channels(),
+                                       reverse_config_,
+                                       reverse_config_,
+                                       reverse_->channels()));
+}
+
+void DebugDumpTest::OnConfigEvent(const audioproc::Config& msg) {
+  MaybeRecreateApm(msg);
+  ConfigureApm(msg);
+}
+
+void DebugDumpTest::MaybeRecreateApm(const audioproc::Config& msg) {
+  // These configurations cannot be changed on the fly.
+  Config config;
+  ASSERT_TRUE(msg.has_aec_delay_agnostic_enabled());
+  config.Set<DelayAgnostic>(
+      new DelayAgnostic(msg.aec_delay_agnostic_enabled()));
+
+  ASSERT_TRUE(msg.has_noise_robust_agc_enabled());
+  config.Set<ExperimentalAgc>(
+       new ExperimentalAgc(msg.noise_robust_agc_enabled()));
+
+  ASSERT_TRUE(msg.has_transient_suppression_enabled());
+  config.Set<ExperimentalNs>(
+      new ExperimentalNs(msg.transient_suppression_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_extended_filter_enabled());
+  config.Set<ExtendedFilter>(new ExtendedFilter(
+      msg.aec_extended_filter_enabled()));
+
+  // We only create APM once, since changes on these fields should not
+  // happen in current implementation.
+  if (!apm_.get()) {
+    apm_.reset(AudioProcessing::Create(config));
+  }
+}
+
+void DebugDumpTest::ConfigureApm(const audioproc::Config& msg) {
+  // AEC configs.
+  ASSERT_TRUE(msg.has_aec_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->Enable(msg.aec_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_drift_compensation_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->enable_drift_compensation(
+                msg.aec_drift_compensation_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_suppression_level());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->set_suppression_level(
+                static_cast<EchoCancellation::SuppressionLevel>(
+                    msg.aec_suppression_level())));
+
+  // AECM configs.
+  ASSERT_TRUE(msg.has_aecm_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->Enable(msg.aecm_enabled()));
+
+  ASSERT_TRUE(msg.has_aecm_comfort_noise_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->enable_comfort_noise(
+                msg.aecm_comfort_noise_enabled()));
+
+  ASSERT_TRUE(msg.has_aecm_routing_mode());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->set_routing_mode(
+                static_cast<EchoControlMobile::RoutingMode>(
+                    msg.aecm_routing_mode())));
+
+  // AGC configs.
+  ASSERT_TRUE(msg.has_agc_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->Enable(msg.agc_enabled()));
+
+  ASSERT_TRUE(msg.has_agc_mode());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->set_mode(
+                static_cast<GainControl::Mode>(msg.agc_mode())));
+
+  ASSERT_TRUE(msg.has_agc_limiter_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->enable_limiter(msg.agc_limiter_enabled()));
+
+  // HPF configs.
+  ASSERT_TRUE(msg.has_hpf_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->high_pass_filter()->Enable(msg.hpf_enabled()));
+
+  // NS configs.
+  ASSERT_TRUE(msg.has_ns_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->noise_suppression()->Enable(msg.ns_enabled()));
+
+  ASSERT_TRUE(msg.has_ns_level());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->noise_suppression()->set_level(
+                static_cast<NoiseSuppression::Level>(msg.ns_level())));
+}
+
+TEST_F(DebugDumpTest, SimpleCase) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeInputFormat) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetInputRate(48000);
+
+  generator.ForceInputMono(true);
+  // Number of output channel should not be larger than that of input. APM will
+  // fail otherwise.
+  generator.SetOutputChannels(1);
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeReverseFormat) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetReverseRate(48000);
+  generator.ForceReverseMono(true);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeOutputFormat) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetOutputRate(48000);
+  generator.SetOutputChannels(1);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleAec) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleDelayAgnosticAec) {
+  Config config;
+  config.Set<DelayAgnostic>(new DelayAgnostic(true));
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleAecLevel) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(true));
+  EXPECT_EQ(AudioProcessing::kNoError,
+            aec->set_suppression_level(EchoCancellation::kLowSuppression));
+  generator.StartRecording();
+  generator.Process(100);
+
+  EXPECT_EQ(AudioProcessing::kNoError,
+            aec->set_suppression_level(EchoCancellation::kHighSuppression));
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+#if defined(WEBRTC_ANDROID)
+// AGC may not be supported on Android.
+#define MAYBE_ToggleAgc DISABLED_ToggleAgc
+#else
+#define MAYBE_ToggleAgc ToggleAgc
+#endif
+TEST_F(DebugDumpTest, MAYBE_ToggleAgc) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  GainControl* agc = generator.apm()->gain_control();
+  EXPECT_EQ(AudioProcessing::kNoError, agc->Enable(!agc->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleNs) {
+  Config config;
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  NoiseSuppression* ns = generator.apm()->noise_suppression();
+  EXPECT_EQ(AudioProcessing::kNoError, ns->Enable(!ns->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, TransientSuppressionOn) {
+  Config config;
+  config.Set<ExperimentalNs>(new ExperimentalNs(true));
+  DebugDumpGenerator generator(config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/process_test.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/process_test.cc
index d32dcb89a7..6e20a787e7 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/test/process_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/process_test.cc
@@ -17,13 +17,15 @@
 
 #include <algorithm>
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@@ -158,9 +160,9 @@ void void_main(int argc, char* argv[]) {
 
   int32_t sample_rate_hz = 16000;
 
-  int num_capture_input_channels = 1;
-  int num_capture_output_channels = 1;
-  int num_render_channels = 1;
+  size_t num_capture_input_channels = 1;
+  size_t num_capture_output_channels = 1;
+  size_t num_render_channels = 1;
 
   int samples_per_channel = sample_rate_hz / 100;
 
@@ -171,6 +173,7 @@ void void_main(int argc, char* argv[]) {
   bool raw_output = false;
   int extra_delay_ms = 0;
   int override_delay_ms = 0;
+  Config config;
 
   ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
   for (int i = 1; i < argc; i++) {
@@ -205,14 +208,14 @@ void void_main(int argc, char* argv[]) {
     } else if (strcmp(argv[i], "-ch") == 0) {
       i++;
       ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
+      ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_capture_input_channels));
       i++;
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
+      ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_capture_output_channels));
 
     } else if (strcmp(argv[i], "-rch") == 0) {
       i++;
       ASSERT_LT(i, argc) << "Specify number of channels after -rch";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
+      ASSERT_EQ(1, sscanf(argv[i], "%" PRIuS, &num_render_channels));
 
     } else if (strcmp(argv[i], "-aec") == 0) {
       ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
@@ -256,14 +259,13 @@ void void_main(int argc, char* argv[]) {
                         suppression_level)));
 
     } else if (strcmp(argv[i], "--extended_filter") == 0) {
-      Config config;
-      config.Set<DelayCorrection>(new DelayCorrection(true));
-      apm->SetExtraOptions(config);
+      config.Set<ExtendedFilter>(new ExtendedFilter(true));
 
     } else if (strcmp(argv[i], "--no_reported_delay") == 0) {
-      Config config;
-      config.Set<ReportedDelay>(new ReportedDelay(false));
-      apm->SetExtraOptions(config);
+      config.Set<DelayAgnostic>(new DelayAgnostic(true));
+
+    } else if (strcmp(argv[i], "--delay_agnostic") == 0) {
+      config.Set<DelayAgnostic>(new DelayAgnostic(true));
 
     } else if (strcmp(argv[i], "-aecm") == 0) {
       ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
@@ -402,9 +404,7 @@ void void_main(int argc, char* argv[]) {
       vad_out_filename = argv[i];
 
     } else if (strcmp(argv[i], "-expns") == 0) {
-      Config config;
       config.Set<ExperimentalNs>(new ExperimentalNs(true));
-      apm->SetExtraOptions(config);
 
     } else if (strcmp(argv[i], "--noasm") == 0) {
       WebRtc_GetCPUInfo = WebRtc_GetCPUInfoNoASM;
@@ -440,16 +440,18 @@ void void_main(int argc, char* argv[]) {
       FAIL() << "Unrecognized argument " << argv[i];
     }
   }
+  apm->SetExtraOptions(config);
+
   // If we're reading a protobuf file, ensure a simulation hasn't also
   // been requested (which makes no sense...)
   ASSERT_FALSE(pb_filename && simulating);
 
   if (verbose) {
     printf("Sample rate: %d Hz\n", sample_rate_hz);
-    printf("Primary channels: %d (in), %d (out)\n",
+    printf("Primary channels: %" PRIuS " (in), %" PRIuS " (out)\n",
            num_capture_input_channels,
            num_capture_output_channels);
-    printf("Reverse channels: %d \n", num_render_channels);
+    printf("Reverse channels: %" PRIuS "\n", num_render_channels);
   }
 
   const std::string out_path = webrtc::test::OutputPath();
@@ -600,14 +602,18 @@ void void_main(int argc, char* argv[]) {
         if (msg.has_output_sample_rate()) {
           output_sample_rate = msg.output_sample_rate();
         }
-        output_layout = LayoutFromChannels(msg.num_output_channels());
-        ASSERT_EQ(kNoErr, apm->Initialize(
-                              msg.sample_rate(),
-                              output_sample_rate,
-                              reverse_sample_rate,
-                              LayoutFromChannels(msg.num_input_channels()),
-                              output_layout,
-                              LayoutFromChannels(msg.num_reverse_channels())));
+        output_layout =
+            LayoutFromChannels(static_cast<size_t>(msg.num_output_channels()));
+        ASSERT_EQ(kNoErr,
+                  apm->Initialize(
+                      msg.sample_rate(),
+                      output_sample_rate,
+                      reverse_sample_rate,
+                      LayoutFromChannels(
+                          static_cast<size_t>(msg.num_input_channels())),
+                      output_layout,
+                      LayoutFromChannels(
+                          static_cast<size_t>(msg.num_reverse_channels()))));
 
         samples_per_channel = msg.sample_rate() / 100;
         far_frame.sample_rate_hz_ = reverse_sample_rate;
@@ -635,11 +641,11 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (!raw_output) {
-          // The WAV file needs to be reset every time, because it cant change
-          // it's sample rate or number of channels.
-          output_wav_file.reset(new WavWriter(out_filename + ".wav",
-                                              output_sample_rate,
-                                              msg.num_output_channels()));
+          // The WAV file needs to be reset every time, because it can't change
+          // its sample rate or number of channels.
+          output_wav_file.reset(new WavWriter(
+              out_filename + ".wav", output_sample_rate,
+              static_cast<size_t>(msg.num_output_channels())));
         }
 
       } else if (event_msg.type() == Event::REVERSE_STREAM) {
@@ -908,7 +914,7 @@ void void_main(int argc, char* argv[]) {
             // not reaching end-of-file.
             EXPECT_EQ(0, fseek(near_file, read_count * sizeof(int16_t),
                       SEEK_CUR));
-            break; // This is expected.
+            break;  // This is expected.
           }
         } else {
           ASSERT_EQ(size, read_count);
@@ -951,7 +957,7 @@ void void_main(int argc, char* argv[]) {
         }
         if (simulating) {
           if (read_count != size) {
-            break; // This is expected.
+            break;  // This is expected.
           }
 
           delay_ms = 0;
@@ -1043,13 +1049,14 @@ void void_main(int argc, char* argv[]) {
                      size,
                      output_wav_file.get(),
                      output_raw_file.get());
-      }
-      else {
+      } else {
         FAIL() << "Event " << event << " is unrecognized";
       }
     }
   }
-  printf("100%% complete\r");
+  if (progress) {
+    printf("100%% complete\r");
+  }
 
   if (aecm_echo_path_out_file != NULL) {
     const size_t path_size =
@@ -1139,8 +1146,7 @@ void void_main(int argc, char* argv[]) {
 }  // namespace
 }  // namespace webrtc
 
-int main(int argc, char* argv[])
-{
+int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
 
   // Optional, but removes memory leak noise from Valgrind.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/protobuf_utils.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/protobuf_utils.cc
new file mode 100644
index 0000000000..37042cdc14
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/protobuf_utils.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+
+namespace webrtc {
+
+size_t ReadMessageBytesFromFile(FILE* file, rtc::scoped_ptr<uint8_t[]>* bytes) {
+  // The "wire format" for the size is little-endian. Assume we're running on
+  // a little-endian machine.
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Need to convert messsage from little-endian."
+#endif
+  int32_t size = 0;
+  if (fread(&size, sizeof(size), 1, file) != 1)
+    return 0;
+  if (size <= 0)
+    return 0;
+
+  bytes->reset(new uint8_t[size]);
+  return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
+}
+
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file, ::google::protobuf::MessageLite* msg) {
+  rtc::scoped_ptr<uint8_t[]> bytes;
+  size_t size = ReadMessageBytesFromFile(file, &bytes);
+  if (!size)
+    return false;
+
+  msg->Clear();
+  return msg->ParseFromArray(bytes.get(), size);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/protobuf_utils.h b/media/webrtc/trunk/webrtc/modules/audio_processing/test/protobuf_utils.h
new file mode 100644
index 0000000000..230fcaad76
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/protobuf_utils.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
+
+#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+// Allocates new memory in the scoped_ptr to fit the raw message and returns the
+// number of bytes read.
+size_t ReadMessageBytesFromFile(FILE* file, rtc::scoped_ptr<uint8_t[]>* bytes);
+
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file, ::google::protobuf::MessageLite* msg);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.cc
new file mode 100644
index 0000000000..0bd70126ae
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.cc
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+
+namespace webrtc {
+
+RawFile::RawFile(const std::string& filename)
+    : file_handle_(fopen(filename.c_str(), "wb")) {}
+
+RawFile::~RawFile() {
+  fclose(file_handle_);
+}
+
+void RawFile::WriteSamples(const int16_t* samples, size_t num_samples) {
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Need to convert samples to little-endian when writing to PCM file"
+#endif
+  fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+}
+
+void RawFile::WriteSamples(const float* samples, size_t num_samples) {
+  fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+}
+
+ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
+    : file_(std::move(file)) {}
+
+bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
+  interleaved_.resize(buffer->size());
+  if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
+      interleaved_.size()) {
+    return false;
+  }
+
+  FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
+               buffer->channels());
+  return true;
+}
+
+ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+    : file_(std::move(file)) {}
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
+  interleaved_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             &interleaved_[0]);
+  FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  file_->WriteSamples(&interleaved_[0], interleaved_.size());
+}
+
+void WriteIntData(const int16_t* data,
+                  size_t length,
+                  WavWriter* wav_file,
+                  RawFile* raw_file) {
+  if (wav_file) {
+    wav_file->WriteSamples(data, length);
+  }
+  if (raw_file) {
+    raw_file->WriteSamples(data, length);
+  }
+}
+
+void WriteFloatData(const float* const* data,
+                    size_t samples_per_channel,
+                    size_t num_channels,
+                    WavWriter* wav_file,
+                    RawFile* raw_file) {
+  size_t length = num_channels * samples_per_channel;
+  rtc::scoped_ptr<float[]> buffer(new float[length]);
+  Interleave(data, samples_per_channel, num_channels, buffer.get());
+  if (raw_file) {
+    raw_file->WriteSamples(buffer.get(), length);
+  }
+  // TODO(aluebs): Use ScaleToInt16Range() from audio_util
+  for (size_t i = 0; i < length; ++i) {
+    buffer[i] = buffer[i] > 0 ?
+                buffer[i] * std::numeric_limits<int16_t>::max() :
+                -buffer[i] * std::numeric_limits<int16_t>::min();
+  }
+  if (wav_file) {
+    wav_file->WriteSamples(buffer.get(), length);
+  }
+}
+
+FILE* OpenFile(const std::string& filename, const char* mode) {
+  FILE* file = fopen(filename.c_str(), mode);
+  if (!file) {
+    printf("Unable to open file %s\n", filename.c_str());
+    exit(1);
+  }
+  return file;
+}
+
+size_t SamplesFromRate(int rate) {
+  return static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate / 1000);
+}
+
+void SetFrameSampleRate(AudioFrame* frame,
+                        int sample_rate_hz) {
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
+      sample_rate_hz / 1000;
+}
+
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
+  switch (num_channels) {
+    case 1:
+      return AudioProcessing::kMono;
+    case 2:
+      return AudioProcessing::kStereo;
+    default:
+      RTC_CHECK(false);
+      return AudioProcessing::kMono;
+  }
+}
+
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
+  const std::vector<float> values = ParseList<float>(mic_positions);
+  const size_t num_mics =
+      rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
+  RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";
+
+  std::vector<Point> result;
+  result.reserve(num_mics);
+  for (size_t i = 0; i < values.size(); i += 3) {
+    result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
+  }
+
+  return result;
+}
+
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
+                                      size_t num_mics) {
+  std::vector<Point> result = ParseArrayGeometry(mic_positions);
+  RTC_CHECK_EQ(result.size(), num_mics)
+      << "Could not parse mic_positions or incorrect number of points.";
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.h b/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.h
index 52274d7853..e23beb66f4 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/test_utils.h
@@ -8,104 +8,92 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <math.h>
-#include <limits>
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
 
-#include "webrtc/audio_processing/debug.pb.h"
+#include <math.h>
+#include <iterator>
+#include <limits>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
-#include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
 #define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
 
-class RawFile {
+class RawFile final {
  public:
-  RawFile(const std::string& filename)
-      : file_handle_(fopen(filename.c_str(), "wb")) {}
+  explicit RawFile(const std::string& filename);
+  ~RawFile();
 
-  ~RawFile() {
-    fclose(file_handle_);
-  }
-
-  void WriteSamples(const int16_t* samples, size_t num_samples) {
-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
-#error "Need to convert samples to little-endian when writing to PCM file"
-#endif
-    fwrite(samples, sizeof(*samples), num_samples, file_handle_);
-  }
-
-  void WriteSamples(const float* samples, size_t num_samples) {
-    fwrite(samples, sizeof(*samples), num_samples, file_handle_);
-  }
+  void WriteSamples(const int16_t* samples, size_t num_samples);
+  void WriteSamples(const float* samples, size_t num_samples);
 
  private:
   FILE* file_handle_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
 };
 
-static inline void WriteIntData(const int16_t* data,
-                                size_t length,
-                                WavWriter* wav_file,
-                                RawFile* raw_file) {
-  if (wav_file) {
-    wav_file->WriteSamples(data, length);
-  }
-  if (raw_file) {
-    raw_file->WriteSamples(data, length);
-  }
-}
+// Reads ChannelBuffers from a provided WavReader.
+class ChannelBufferWavReader final {
+ public:
+  explicit ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file);
 
-static inline void WriteFloatData(const float* const* data,
-                                  size_t samples_per_channel,
-                                  int num_channels,
-                                  WavWriter* wav_file,
-                                  RawFile* raw_file) {
-  size_t length = num_channels * samples_per_channel;
-  rtc::scoped_ptr<float[]> buffer(new float[length]);
-  Interleave(data, samples_per_channel, num_channels, buffer.get());
-  if (raw_file) {
-    raw_file->WriteSamples(buffer.get(), length);
-  }
-  // TODO(aluebs): Use ScaleToInt16Range() from audio_util
-  for (size_t i = 0; i < length; ++i) {
-    buffer[i] = buffer[i] > 0 ?
-                buffer[i] * std::numeric_limits<int16_t>::max() :
-                -buffer[i] * std::numeric_limits<int16_t>::min();
-  }
-  if (wav_file) {
-    wav_file->WriteSamples(buffer.get(), length);
-  }
-}
+  // Reads data from the file according to the |buffer| format. Returns false if
+  // a full buffer can't be read from the file.
+  bool Read(ChannelBuffer<float>* buffer);
+
+ private:
+  rtc::scoped_ptr<WavReader> file_;
+  std::vector<float> interleaved_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavReader);
+};
+
+// Writes ChannelBuffers to a provided WavWriter.
+class ChannelBufferWavWriter final {
+ public:
+  explicit ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file);
+  void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+  rtc::scoped_ptr<WavWriter> file_;
+  std::vector<float> interleaved_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelBufferWavWriter);
+};
+
+void WriteIntData(const int16_t* data,
+                  size_t length,
+                  WavWriter* wav_file,
+                  RawFile* raw_file);
+
+void WriteFloatData(const float* const* data,
+                    size_t samples_per_channel,
+                    size_t num_channels,
+                    WavWriter* wav_file,
+                    RawFile* raw_file);
 
 // Exits on failure; do not use in unit tests.
-static inline FILE* OpenFile(const std::string& filename, const char* mode) {
-  FILE* file = fopen(filename.c_str(), mode);
-  if (!file) {
-    printf("Unable to open file %s\n", filename.c_str());
-    exit(1);
-  }
-  return file;
-}
+FILE* OpenFile(const std::string& filename, const char* mode);
 
-static inline int SamplesFromRate(int rate) {
-  return AudioProcessing::kChunkSizeMs * rate / 1000;
-}
+size_t SamplesFromRate(int rate);
 
-static inline void SetFrameSampleRate(AudioFrame* frame,
-                                      int sample_rate_hz) {
-  frame->sample_rate_hz_ = sample_rate_hz;
-  frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
-      sample_rate_hz / 1000;
-}
+void SetFrameSampleRate(AudioFrame* frame,
+                        int sample_rate_hz);
 
 template <typename T>
 void SetContainerFormat(int sample_rate_hz,
-                        int num_channels,
+                        size_t num_channels,
                         AudioFrame* frame,
                         rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
   SetFrameSampleRate(frame, sample_rate_hz);
@@ -113,54 +101,14 @@ void SetContainerFormat(int sample_rate_hz,
   cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
 }
 
-static inline AudioProcessing::ChannelLayout LayoutFromChannels(
-    int num_channels) {
-  switch (num_channels) {
-    case 1:
-      return AudioProcessing::kMono;
-    case 2:
-      return AudioProcessing::kStereo;
-    default:
-      assert(false);
-      return AudioProcessing::kMono;
-  }
-}
-
-// Allocates new memory in the scoped_ptr to fit the raw message and returns the
-// number of bytes read.
-static inline size_t ReadMessageBytesFromFile(
-    FILE* file,
-    rtc::scoped_ptr<uint8_t[]>* bytes) {
-  // The "wire format" for the size is little-endian. Assume we're running on
-  // a little-endian machine.
-  int32_t size = 0;
-  if (fread(&size, sizeof(size), 1, file) != 1)
-    return 0;
-  if (size <= 0)
-    return 0;
-
-  bytes->reset(new uint8_t[size]);
-  return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
-}
-
-// Returns true on success, false on error or end-of-file.
-static inline bool ReadMessageFromFile(FILE* file,
-                                       ::google::protobuf::MessageLite* msg) {
-  rtc::scoped_ptr<uint8_t[]> bytes;
-  size_t size = ReadMessageBytesFromFile(file, &bytes);
-  if (!size)
-    return false;
-
-  msg->Clear();
-  return msg->ParseFromArray(bytes.get(), size);
-}
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels);
 
 template <typename T>
-float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
+float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
   float mse = 0;
   float mean = 0;
   *variance = 0;
-  for (int i = 0; i < length; ++i) {
+  for (size_t i = 0; i < length; ++i) {
     T error = ref[i] - test[i];
     mse += error * error;
     *variance += ref[i] * ref[i];
@@ -177,4 +125,31 @@ float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
   return snr;
 }
 
+// Returns a vector<T> parsed from whitespace delimited values in to_parse,
+// or an empty vector if the string could not be parsed.
+template<typename T>
+std::vector<T> ParseList(const std::string& to_parse) {
+  std::vector<T> values;
+
+  std::istringstream str(to_parse);
+  std::copy(
+      std::istream_iterator<T>(str),
+      std::istream_iterator<T>(),
+      std::back_inserter(values));
+
+  return values;
+}
+
+// Parses the array geometry from the command line.
+//
+// If a vector with size != num_mics is returned, an error has occurred and an
+// appropriate error message has been printed to stdout.
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
+                                      size_t num_mics);
+
+// Same as above, but without the num_mics check for when it isn't available.
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
+
 }  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/test/unpack.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/test/unpack.cc
index af0f5cae89..8b2b082f97 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/test/unpack.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/test/unpack.cc
@@ -17,7 +17,9 @@
 
 #include "gflags/gflags.h"
 #include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/typedefs.h"
 
@@ -35,6 +37,14 @@ DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
 DEFINE_bool(full, false,
             "Unpack the full set of files (normally not needed).");
 DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
+DEFINE_bool(text,
+            false,
+            "Write non-audio files as text files instead of binary files.");
+
+#define PRINT_CONFIG(field_name) \
+  if (msg.has_##field_name()) { \
+    fprintf(settings_file, "  " #field_name ": %d\n", msg.field_name()); \
+  }
 
 namespace webrtc {
 
@@ -67,18 +77,21 @@ int do_main(int argc, char* argv[]) {
 
   Event event_msg;
   int frame_count = 0;
-  int reverse_samples_per_channel = 0;
-  int input_samples_per_channel = 0;
-  int output_samples_per_channel = 0;
-  int num_reverse_channels = 0;
-  int num_input_channels = 0;
-  int num_output_channels = 0;
+  size_t reverse_samples_per_channel = 0;
+  size_t input_samples_per_channel = 0;
+  size_t output_samples_per_channel = 0;
+  size_t num_reverse_channels = 0;
+  size_t num_input_channels = 0;
+  size_t num_output_channels = 0;
   rtc::scoped_ptr<WavWriter> reverse_wav_file;
   rtc::scoped_ptr<WavWriter> input_wav_file;
   rtc::scoped_ptr<WavWriter> output_wav_file;
   rtc::scoped_ptr<RawFile> reverse_raw_file;
   rtc::scoped_ptr<RawFile> input_raw_file;
   rtc::scoped_ptr<RawFile> output_raw_file;
+
+  FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
+
   while (ReadMessageFromFile(debug_file, &event_msg)) {
     if (event_msg.type() == Event::REVERSE_STREAM) {
       if (!event_msg.has_reverse_stream()) {
@@ -105,7 +118,7 @@ int do_main(int argc, char* argv[]) {
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_reverse_channels]);
-        for (int i = 0; i < num_reverse_channels; ++i) {
+        for (size_t i = 0; i < num_reverse_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
         }
         WriteFloatData(data.get(),
@@ -136,7 +149,7 @@ int do_main(int argc, char* argv[]) {
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_input_channels]);
-        for (int i = 0; i < num_input_channels; ++i) {
+        for (size_t i = 0; i < num_input_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
         }
         WriteFloatData(data.get(),
@@ -160,7 +173,7 @@ int do_main(int argc, char* argv[]) {
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_output_channels]);
-        for (int i = 0; i < num_output_channels; ++i) {
+        for (size_t i = 0; i < num_output_channels; ++i) {
           data[i] =
               reinterpret_cast<const float*>(msg.output_channel(i).data());
         }
@@ -175,35 +188,75 @@ int do_main(int argc, char* argv[]) {
         if (msg.has_delay()) {
           static FILE* delay_file = OpenFile(FLAGS_delay_file, "wb");
           int32_t delay = msg.delay();
-          WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
+          if (FLAGS_text) {
+            fprintf(delay_file, "%d\n", delay);
+          } else {
+            WriteData(&delay, sizeof(delay), delay_file, FLAGS_delay_file);
+          }
         }
 
         if (msg.has_drift()) {
           static FILE* drift_file = OpenFile(FLAGS_drift_file, "wb");
           int32_t drift = msg.drift();
-          WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
+          if (FLAGS_text) {
+            fprintf(drift_file, "%d\n", drift);
+          } else {
+            WriteData(&drift, sizeof(drift), drift_file, FLAGS_drift_file);
+          }
         }
 
         if (msg.has_level()) {
           static FILE* level_file = OpenFile(FLAGS_level_file, "wb");
           int32_t level = msg.level();
-          WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
+          if (FLAGS_text) {
+            fprintf(level_file, "%d\n", level);
+          } else {
+            WriteData(&level, sizeof(level), level_file, FLAGS_level_file);
+          }
         }
 
         if (msg.has_keypress()) {
           static FILE* keypress_file = OpenFile(FLAGS_keypress_file, "wb");
           bool keypress = msg.keypress();
-          WriteData(&keypress, sizeof(keypress), keypress_file,
-                    FLAGS_keypress_file);
+          if (FLAGS_text) {
+            fprintf(keypress_file, "%d\n", keypress);
+          } else {
+            WriteData(&keypress, sizeof(keypress), keypress_file,
+                      FLAGS_keypress_file);
+          }
         }
       }
+    } else if (event_msg.type() == Event::CONFIG) {
+      if (!event_msg.has_config()) {
+        printf("Corrupt input file: Config missing.\n");
+        return 1;
+      }
+      const audioproc::Config msg = event_msg.config();
+
+      fprintf(settings_file, "APM re-config at frame: %d\n", frame_count);
+
+      PRINT_CONFIG(aec_enabled);
+      PRINT_CONFIG(aec_delay_agnostic_enabled);
+      PRINT_CONFIG(aec_drift_compensation_enabled);
+      PRINT_CONFIG(aec_extended_filter_enabled);
+      PRINT_CONFIG(aec_suppression_level);
+      PRINT_CONFIG(aecm_enabled);
+      PRINT_CONFIG(aecm_comfort_noise_enabled);
+      PRINT_CONFIG(aecm_routing_mode);
+      PRINT_CONFIG(agc_enabled);
+      PRINT_CONFIG(agc_mode);
+      PRINT_CONFIG(agc_limiter_enabled);
+      PRINT_CONFIG(noise_robust_agc_enabled);
+      PRINT_CONFIG(hpf_enabled);
+      PRINT_CONFIG(ns_enabled);
+      PRINT_CONFIG(ns_level);
+      PRINT_CONFIG(transient_suppression_enabled);
     } else if (event_msg.type() == Event::INIT) {
       if (!event_msg.has_init()) {
         printf("Corrupt input file: Init missing.\n");
         return 1;
       }
 
-      static FILE* settings_file = OpenFile(FLAGS_settings_file, "wb");
       const Init msg = event_msg.init();
       // These should print out zeros if they're missing.
       fprintf(settings_file, "Init at frame: %d\n", frame_count);
@@ -216,11 +269,14 @@ int do_main(int argc, char* argv[]) {
               "  Reverse sample rate: %d\n",
               reverse_sample_rate);
       num_input_channels = msg.num_input_channels();
-      fprintf(settings_file, "  Input channels: %d\n", num_input_channels);
+      fprintf(settings_file, "  Input channels: %" PRIuS "\n",
+              num_input_channels);
       num_output_channels = msg.num_output_channels();
-      fprintf(settings_file, "  Output channels: %d\n", num_output_channels);
+      fprintf(settings_file, "  Output channels: %" PRIuS "\n",
+              num_output_channels);
       num_reverse_channels = msg.num_reverse_channels();
-      fprintf(settings_file, "  Reverse channels: %d\n", num_reverse_channels);
+      fprintf(settings_file, "  Reverse channels: %" PRIuS "\n",
+              num_reverse_channels);
 
       fprintf(settings_file, "\n");
 
@@ -231,9 +287,12 @@ int do_main(int argc, char* argv[]) {
         output_sample_rate = input_sample_rate;
       }
 
-      reverse_samples_per_channel = reverse_sample_rate / 100;
-      input_samples_per_channel = input_sample_rate / 100;
-      output_samples_per_channel = output_sample_rate / 100;
+      reverse_samples_per_channel =
+          static_cast<size_t>(reverse_sample_rate / 100);
+      input_samples_per_channel =
+          static_cast<size_t>(input_sample_rate / 100);
+      output_samples_per_channel =
+          static_cast<size_t>(output_sample_rate / 100);
 
       if (!FLAGS_raw) {
         // The WAV files need to be reset every time, because they cant change
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/three_band_filter_bank.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/three_band_filter_bank.cc
new file mode 100644
index 0000000000..388f6e8cf4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/three_band_filter_bank.cc
@@ -0,0 +1,214 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// An implementation of a 3-band FIR filter-bank with DCT modulation, similar to
+// the proposed in "Multirate Signal Processing for Communication Systems" by
+// Fredric J Harris.
+//
+// The idea is to take a heterodyne system and change the order of the
+// components to get something which is efficient to implement digitally.
+//
+// It is possible to separate the filter using the noble identity as follows:
+//
+// H(z) = H0(z^3) + z^-1 * H1(z^3) + z^-2 * H2(z^3)
+//
+// This is used in the analysis stage to first downsample serial to parallel
+// and then filter each branch with one of these polyphase decompositions of the
+// lowpass prototype. Because each filter is only a modulation of the prototype,
+// it is enough to multiply each coefficient by the respective cosine value to
+// shift it to the desired band. But because the cosine period is 12 samples,
+// it requires separating the prototype even further using the noble identity.
+// After filtering and modulating for each band, the output of all filters is
+// accumulated to get the downsampled bands.
+//
+// A similar logic can be applied to the synthesis stage.
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#ifndef _USE_MATH_DEFINES
+#define _USE_MATH_DEFINES
+#endif
+
+#include "webrtc/modules/audio_processing/three_band_filter_bank.h"
+
+#include <cmath>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace {
+
+const size_t kNumBands = 3;
+const size_t kSparsity = 4;
+
+// Factors to take into account when choosing |kNumCoeffs|:
+//   1. Higher |kNumCoeffs|, means faster transition, which ensures less
+//      aliasing. This is especially important when there is non-linear
+//      processing between the splitting and merging.
+//   2. The delay that this filter bank introduces is
+//      |kNumBands| * |kSparsity| * |kNumCoeffs| / 2, so it increases linearly
+//      with |kNumCoeffs|.
+//   3. The computation complexity also increases linearly with |kNumCoeffs|.
+const size_t kNumCoeffs = 4;
+
+// The Matlab code to generate these |kLowpassCoeffs| is:
+//
+// N = kNumBands * kSparsity * kNumCoeffs - 1;
+// h = fir1(N, 1 / (2 * kNumBands), kaiser(N + 1, 3.5));
+// reshape(h, kNumBands * kSparsity, kNumCoeffs);
+//
+// Because the total bandwidth of the lower and higher band is double the middle
+// one (because of the spectrum parity), the low-pass prototype is half the
+// bandwidth of 1 / (2 * |kNumBands|) and is then shifted with cosine modulation
+// to the right places.
+// A Kaiser window is used because of its flexibility and the alpha is set to
+// 3.5, since that sets a stop band attenuation of 40dB ensuring a fast
+// transition.
+const float kLowpassCoeffs[kNumBands * kSparsity][kNumCoeffs] =
+    {{-0.00047749f, -0.00496888f, +0.16547118f, +0.00425496f},
+     {-0.00173287f, -0.01585778f, +0.14989004f, +0.00994113f},
+     {-0.00304815f, -0.02536082f, +0.12154542f, +0.01157993f},
+     {-0.00383509f, -0.02982767f, +0.08543175f, +0.00983212f},
+     {-0.00346946f, -0.02587886f, +0.04760441f, +0.00607594f},
+     {-0.00154717f, -0.01136076f, +0.01387458f, +0.00186353f},
+     {+0.00186353f, +0.01387458f, -0.01136076f, -0.00154717f},
+     {+0.00607594f, +0.04760441f, -0.02587886f, -0.00346946f},
+     {+0.00983212f, +0.08543175f, -0.02982767f, -0.00383509f},
+     {+0.01157993f, +0.12154542f, -0.02536082f, -0.00304815f},
+     {+0.00994113f, +0.14989004f, -0.01585778f, -0.00173287f},
+     {+0.00425496f, +0.16547118f, -0.00496888f, -0.00047749f}};
+
+// Downsamples |in| into |out|, taking one every |kNumbands| starting from
+// |offset|. |split_length| is the |out| length. |in| has to be at least
+// |kNumBands| * |split_length| long.
+void Downsample(const float* in,
+                size_t split_length,
+                size_t offset,
+                float* out) {
+  for (size_t i = 0; i < split_length; ++i) {
+    out[i] = in[kNumBands * i + offset];
+  }
+}
+
+// Upsamples |in| into |out|, scaling by |kNumBands| and accumulating it every
+// |kNumBands| starting from |offset|. |split_length| is the |in| length. |out|
+// has to be at least |kNumBands| * |split_length| long.
+void Upsample(const float* in, size_t split_length, size_t offset, float* out) {
+  for (size_t i = 0; i < split_length; ++i) {
+    out[kNumBands * i + offset] += kNumBands * in[i];
+  }
+}
+
+}  // namespace
+
+// Because the low-pass filter prototype has half bandwidth it is possible to
+// use a DCT to shift it in both directions at the same time, to the center
+// frequencies [1 / 12, 3 / 12, 5 / 12].
+ThreeBandFilterBank::ThreeBandFilterBank(size_t length)
+    : in_buffer_(rtc::CheckedDivExact(length, kNumBands)),
+      out_buffer_(in_buffer_.size()) {
+  for (size_t i = 0; i < kSparsity; ++i) {
+    for (size_t j = 0; j < kNumBands; ++j) {
+      analysis_filters_.push_back(new SparseFIRFilter(
+          kLowpassCoeffs[i * kNumBands + j], kNumCoeffs, kSparsity, i));
+      synthesis_filters_.push_back(new SparseFIRFilter(
+          kLowpassCoeffs[i * kNumBands + j], kNumCoeffs, kSparsity, i));
+    }
+  }
+  dct_modulation_.resize(kNumBands * kSparsity);
+  for (size_t i = 0; i < dct_modulation_.size(); ++i) {
+    dct_modulation_[i].resize(kNumBands);
+    for (size_t j = 0; j < kNumBands; ++j) {
+      dct_modulation_[i][j] =
+          2.f * cos(2.f * M_PI * i * (2.f * j + 1.f) / dct_modulation_.size());
+    }
+  }
+}
+
+// The analysis can be separated in these steps:
+//   1. Serial to parallel downsampling by a factor of |kNumBands|.
+//   2. Filtering of |kSparsity| different delayed signals with polyphase
+//      decomposition of the low-pass prototype filter and upsampled by a factor
+//      of |kSparsity|.
+//   3. Modulating with cosines and accumulating to get the desired band.
+void ThreeBandFilterBank::Analysis(const float* in,
+                                   size_t length,
+                                   float* const* out) {
+  RTC_CHECK_EQ(in_buffer_.size(), rtc::CheckedDivExact(length, kNumBands));
+  for (size_t i = 0; i < kNumBands; ++i) {
+    memset(out[i], 0, in_buffer_.size() * sizeof(*out[i]));
+  }
+  for (size_t i = 0; i < kNumBands; ++i) {
+    Downsample(in, in_buffer_.size(), kNumBands - i - 1, &in_buffer_[0]);
+    for (size_t j = 0; j < kSparsity; ++j) {
+      const size_t offset = i + j * kNumBands;
+      analysis_filters_[offset]->Filter(&in_buffer_[0],
+                                        in_buffer_.size(),
+                                        &out_buffer_[0]);
+      DownModulate(&out_buffer_[0], out_buffer_.size(), offset, out);
+    }
+  }
+}
+
+// The synthesis can be separated in these steps:
+//   1. Modulating with cosines.
+//   2. Filtering each one with a polyphase decomposition of the low-pass
+//      prototype filter upsampled by a factor of |kSparsity| and accumulating
+//      |kSparsity| signals with different delays.
+//   3. Parallel to serial upsampling by a factor of |kNumBands|.
+void ThreeBandFilterBank::Synthesis(const float* const* in,
+                                    size_t split_length,
+                                    float* out) {
+  RTC_CHECK_EQ(in_buffer_.size(), split_length);
+  memset(out, 0, kNumBands * in_buffer_.size() * sizeof(*out));
+  for (size_t i = 0; i < kNumBands; ++i) {
+    for (size_t j = 0; j < kSparsity; ++j) {
+      const size_t offset = i + j * kNumBands;
+      UpModulate(in, in_buffer_.size(), offset, &in_buffer_[0]);
+      synthesis_filters_[offset]->Filter(&in_buffer_[0],
+                                         in_buffer_.size(),
+                                         &out_buffer_[0]);
+      Upsample(&out_buffer_[0], out_buffer_.size(), i, out);
+    }
+  }
+}
+
+
+// Modulates |in| by |dct_modulation_| and accumulates it in each of the
+// |kNumBands| bands of |out|. |offset| is the index in the period of the
+// cosines used for modulation. |split_length| is the length of |in| and each
+// band of |out|.
+void ThreeBandFilterBank::DownModulate(const float* in,
+                                       size_t split_length,
+                                       size_t offset,
+                                       float* const* out) {
+  for (size_t i = 0; i < kNumBands; ++i) {
+    for (size_t j = 0; j < split_length; ++j) {
+      out[i][j] += dct_modulation_[offset][i] * in[j];
+    }
+  }
+}
+
+// Modulates each of the |kNumBands| bands of |in| by |dct_modulation_| and
+// accumulates them in |out|. |out| is cleared before starting to accumulate.
+// |offset| is the index in the period of the cosines used for modulation.
+// |split_length| is the length of each band of |in| and |out|.
+void ThreeBandFilterBank::UpModulate(const float* const* in,
+                                     size_t split_length,
+                                     size_t offset,
+                                     float* out) {
+  memset(out, 0, split_length * sizeof(*out));
+  for (size_t i = 0; i < kNumBands; ++i) {
+    for (size_t j = 0; j < split_length; ++j) {
+      out[j] += dct_modulation_[offset][i] * in[i][j];
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/three_band_filter_bank.h b/media/webrtc/trunk/webrtc/modules/audio_processing/three_band_filter_bank.h
new file mode 100644
index 0000000000..cb9cfbe7b1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/three_band_filter_bank.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_THREE_BAND_FILTER_BANK_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_THREE_BAND_FILTER_BANK_H_
+
+#include <cstring>
+#include <vector>
+
+#include "webrtc/common_audio/sparse_fir_filter.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
+
+namespace webrtc {
+
+// An implementation of a 3-band FIR filter-bank with DCT modulation, similar to
+// the proposed in "Multirate Signal Processing for Communication Systems" by
+// Fredric J Harris.
+// The low-pass filter prototype has these characteristics:
+// * Pass-band ripple = 0.3dB
+// * Pass-band frequency = 0.147 (7kHz at 48kHz)
+// * Stop-band attenuation = 40dB
+// * Stop-band frequency = 0.192 (9.2kHz at 48kHz)
+// * Delay = 24 samples (500us at 48kHz)
+// * Linear phase
+// This filter bank does not satisfy perfect reconstruction. The SNR after
+// analysis and synthesis (with no processing in between) is approximately 9.5dB
+// depending on the input signal after compensating for the delay.
+class ThreeBandFilterBank final {
+ public:
+  explicit ThreeBandFilterBank(size_t length);
+
+  // Splits |in| into 3 downsampled frequency bands in |out|.
+  // |length| is the |in| length. Each of the 3 bands of |out| has to have a
+  // length of |length| / 3.
+  void Analysis(const float* in, size_t length, float* const* out);
+
+  // Merges the 3 downsampled frequency bands in |in| into |out|.
+  // |split_length| is the length of each band of |in|. |out| has to have at
+  // least a length of 3 * |split_length|.
+  void Synthesis(const float* const* in, size_t split_length, float* out);
+
+ private:
+  void DownModulate(const float* in,
+                    size_t split_length,
+                    size_t offset,
+                    float* const* out);
+  void UpModulate(const float* const* in,
+                  size_t split_length,
+                  size_t offset,
+                  float* out);
+
+  std::vector<float> in_buffer_;
+  std::vector<float> out_buffer_;
+  ScopedVector<SparseFIRFilter> analysis_filters_;
+  ScopedVector<SparseFIRFilter> synthesis_filters_;
+  std::vector<std::vector<float>> dct_modulation_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_THREE_BAND_FILTER_BANK_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/click_annotate.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/click_annotate.cc
index f913cfd716..38f7a8eede 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/click_annotate.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/click_annotate.cc
@@ -16,7 +16,7 @@
 #include "webrtc/modules/audio_processing/transient/transient_detector.h"
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 
 using rtc::scoped_ptr;
 using webrtc::FileWrapper;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.cc
index 2325bd6cb2..e043286f80 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.h b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.h
index dbc3b5f788..cc76953215 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils.h
@@ -13,7 +13,7 @@
 
 #include <string.h>
 
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
index 8507d90065..7fb7d2d6a9 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
@@ -15,7 +15,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -58,7 +58,12 @@ class TransientFileUtilsTest: public ::testing::Test {
   const std::string kTestFileNamef;
 };
 
-TEST_F(TransientFileUtilsTest, ConvertByteArrayToFloat) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToFloat DISABLED_ConvertByteArrayToFloat
+#else
+#define MAYBE_ConvertByteArrayToFloat ConvertByteArrayToFloat
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToFloat) {
   float value = 0.0;
 
   EXPECT_EQ(0, ConvertByteArrayToFloat(kPiBytesf, &value));
@@ -71,7 +76,12 @@ TEST_F(TransientFileUtilsTest, ConvertByteArrayToFloat) {
   EXPECT_FLOAT_EQ(kAvogadro, value);
 }
 
-TEST_F(TransientFileUtilsTest, ConvertByteArrayToDouble) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToDouble DISABLED_ConvertByteArrayToDouble
+#else
+#define MAYBE_ConvertByteArrayToDouble ConvertByteArrayToDouble
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToDouble) {
   double value = 0.0;
 
   EXPECT_EQ(0, ConvertByteArrayToDouble(kPiBytes, &value));
@@ -84,7 +94,12 @@ TEST_F(TransientFileUtilsTest, ConvertByteArrayToDouble) {
   EXPECT_DOUBLE_EQ(kAvogadro, value);
 }
 
-TEST_F(TransientFileUtilsTest, ConvertFloatToByteArray) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertFloatToByteArray DISABLED_ConvertFloatToByteArray
+#else
+#define MAYBE_ConvertFloatToByteArray ConvertFloatToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertFloatToByteArray) {
   rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[4]);
 
   EXPECT_EQ(0, ConvertFloatToByteArray(kPi, bytes.get()));
@@ -97,7 +112,12 @@ TEST_F(TransientFileUtilsTest, ConvertFloatToByteArray) {
   EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytesf, 4));
 }
 
-TEST_F(TransientFileUtilsTest, ConvertDoubleToByteArray) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertDoubleToByteArray DISABLED_ConvertDoubleToByteArray
+#else
+#define MAYBE_ConvertDoubleToByteArray ConvertDoubleToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertDoubleToByteArray) {
   rtc::scoped_ptr<uint8_t[]> bytes(new uint8_t[8]);
 
   EXPECT_EQ(0, ConvertDoubleToByteArray(kPi, bytes.get()));
@@ -110,7 +130,12 @@ TEST_F(TransientFileUtilsTest, ConvertDoubleToByteArray) {
   EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytes, 8));
 }
 
-TEST_F(TransientFileUtilsTest, ReadInt16BufferFromFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16BufferFromFile DISABLED_ReadInt16BufferFromFile
+#else
+#define MAYBE_ReadInt16BufferFromFile ReadInt16BufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -148,7 +173,13 @@ TEST_F(TransientFileUtilsTest, ReadInt16BufferFromFile) {
   EXPECT_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest, ReadInt16FromFileToFloatBuffer) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToFloatBuffer \
+  DISABLED_ReadInt16FromFileToFloatBuffer
+#else
+#define MAYBE_ReadInt16FromFileToFloatBuffer ReadInt16FromFileToFloatBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -189,7 +220,13 @@ TEST_F(TransientFileUtilsTest, ReadInt16FromFileToFloatBuffer) {
   EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest, ReadInt16FromFileToDoubleBuffer) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToDoubleBuffer \
+  DISABLED_ReadInt16FromFileToDoubleBuffer
+#else
+#define MAYBE_ReadInt16FromFileToDoubleBuffer ReadInt16FromFileToDoubleBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -229,7 +266,12 @@ TEST_F(TransientFileUtilsTest, ReadInt16FromFileToDoubleBuffer) {
   EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
 }
 
-TEST_F(TransientFileUtilsTest, ReadFloatBufferFromFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadFloatBufferFromFile DISABLED_ReadFloatBufferFromFile
+#else
+#define MAYBE_ReadFloatBufferFromFile ReadFloatBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
   std::string test_filename = kTestFileNamef;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -266,7 +308,12 @@ TEST_F(TransientFileUtilsTest, ReadFloatBufferFromFile) {
   EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
 }
 
-TEST_F(TransientFileUtilsTest, ReadDoubleBufferFromFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadDoubleBufferFromFile DISABLED_ReadDoubleBufferFromFile
+#else
+#define MAYBE_ReadDoubleBufferFromFile ReadDoubleBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
   std::string test_filename = kTestFileName;
 
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
@@ -303,7 +350,12 @@ TEST_F(TransientFileUtilsTest, ReadDoubleBufferFromFile) {
   EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
 }
 
-TEST_F(TransientFileUtilsTest, WriteInt16BufferToFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteInt16BufferToFile DISABLED_WriteInt16BufferToFile
+#else
+#define MAYBE_WriteInt16BufferToFile WriteInt16BufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -345,7 +397,12 @@ TEST_F(TransientFileUtilsTest, WriteInt16BufferToFile) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, WriteFloatBufferToFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteFloatBufferToFile DISABLED_WriteFloatBufferToFile
+#else
+#define MAYBE_WriteFloatBufferToFile WriteFloatBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -387,7 +444,12 @@ TEST_F(TransientFileUtilsTest, WriteFloatBufferToFile) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, WriteDoubleBufferToFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteDoubleBufferToFile DISABLED_WriteDoubleBufferToFile
+#else
+#define MAYBE_WriteDoubleBufferToFile WriteDoubleBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
   rtc::scoped_ptr<FileWrapper> file(FileWrapper::Create());
 
   std::string kOutFileName = test::TempFilename(test::OutputPath(),
@@ -429,7 +491,12 @@ TEST_F(TransientFileUtilsTest, WriteDoubleBufferToFile) {
                       kBufferLength * sizeof(written_buffer[0])));
 }
 
-TEST_F(TransientFileUtilsTest, ExpectedErrorReturnValues) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_ExpectedErrorReturnValues DISABLED_ExpectedErrorReturnValues
+#else
+#define MAYBE_ExpectedErrorReturnValues ExpectedErrorReturnValues
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
   std::string test_filename = kTestFileName;
 
   double value;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
index 16a79158bd..b60077510b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
@@ -17,7 +17,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/transient/common.h"
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -36,7 +36,11 @@ static const size_t kNumberOfSampleRates =
 // The files contain all the results in double precision (Little endian).
 // The audio files used with different sample rates are stored in the same
 // directory.
+#if defined(WEBRTC_IOS)
+TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
+#else
 TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
+#endif
   for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
     int sample_rate_hz = kSampleRatesHz[i];
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
index fdc5686763..b7b7595abf 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -19,7 +19,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -150,13 +150,13 @@ void void_main() {
 
   // Prepare the detection file.
   FILE* detection_file = NULL;
-  if (FLAGS_detection_file_name != "") {
+  if (!FLAGS_detection_file_name.empty()) {
     detection_file = fopen(FLAGS_detection_file_name.c_str(), "rb");
   }
 
   // Prepare the reference file.
   FILE* reference_file = NULL;
-  if (FLAGS_reference_file_name != "") {
+  if (!FLAGS_reference_file_name.empty()) {
     reference_file = fopen(FLAGS_reference_file_name.c_str(), "rb");
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.cc
index 2f79a20ac7..c8d9e65858 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.cc
@@ -24,7 +24,7 @@
 #include "webrtc/modules/audio_processing/transient/common.h"
 #include "webrtc/modules/audio_processing/transient/transient_detector.h"
 #include "webrtc/modules/audio_processing/ns/windows_private.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -124,7 +124,7 @@ int TransientSuppressor::Initialize(int sample_rate_hz,
          analysis_length_ * num_channels_ * sizeof(out_buffer_[0]));
   // ip[0] must be zero to trigger initialization using rdft().
   size_t ip_length = 2 + sqrtf(analysis_length_);
-  ip_.reset(new int[ip_length]());
+  ip_.reset(new size_t[ip_length]());
   memset(ip_.get(), 0, ip_length * sizeof(ip_[0]));
   wfft_.reset(new float[complex_analysis_length_ - 1]);
   memset(wfft_.get(), 0, (complex_analysis_length_ - 1) * sizeof(wfft_[0]));
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.h b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.h
index 12e4b5ed1f..5a6f117629 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/transient_suppressor.h
@@ -86,7 +86,7 @@ class TransientSuppressor {
   rtc::scoped_ptr<float[]> out_buffer_;
 
   // Arrays for fft.
-  rtc::scoped_ptr<int[]> ip_;
+  rtc::scoped_ptr<size_t[]> ip_;
   rtc::scoped_ptr<float[]> wfft_;
 
   rtc::scoped_ptr<float[]> spectral_mean_;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
index dd5b5c810d..e4e9048f88 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
@@ -17,7 +17,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h"
 #include "webrtc/modules/audio_processing/transient/file_utils.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -68,7 +68,11 @@ TEST(WPDTreeTest, Construction) {
 // It also writes the results in its own set of files in the out directory.
 // Matlab and output files contain all the results in double precision (Little
 // endian) appended.
+#if defined(WEBRTC_IOS)
+TEST(WPDTreeTest, DISABLED_CorrectnessBasedOnMatlabFiles) {
+#else
 TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
+#endif
   // 10 ms at 16000 Hz.
   const size_t kTestBufferSize = 160;
   const int kLevels = 3;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/typing_detection.h b/media/webrtc/trunk/webrtc/modules/audio_processing/typing_detection.h
index 5fa6456e9e..40608f885d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/typing_detection.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/typing_detection.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.c b/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.c
index 9c6f19f4d6..02df75a101 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.c
@@ -632,8 +632,7 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
   }
 
   if (self->robust_validation_enabled) {
-    int is_histogram_valid = 0;
-    is_histogram_valid = HistogramBasedValidation(self, candidate_delay);
+    int is_histogram_valid = HistogramBasedValidation(self, candidate_delay);
     valid_candidate = RobustValidation(self, candidate_delay, valid_candidate,
                                        is_histogram_valid);
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c b/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c
index 270588f38c..b5448bc5bd 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c
@@ -16,7 +16,7 @@
 
 #include "webrtc/modules/audio_processing/utility/delay_estimator.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_internal.h"
-#include "webrtc/system_wrappers/interface/compile_assert_c.h"
+#include "webrtc/system_wrappers/include/compile_assert_c.h"
 
 // Only bit |kBandFirst| through bit |kBandLast| are processed and
 // |kBandFirst| - |kBandLast| must be < 32.
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/common.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/common.h
similarity index 69%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/common.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/common.h
index e9ed1edadd..be99c1c59d 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/common.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/common.h
@@ -8,20 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_COMMON_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_COMMON_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
 
 static const int kSampleRateHz = 16000;
-static const int kLength10Ms = kSampleRateHz / 100;
-static const int kMaxNumFrames = 4;
+static const size_t kLength10Ms = kSampleRateHz / 100;
+static const size_t kMaxNumFrames = 4;
 
 struct AudioFeatures {
   double log_pitch_gain[kMaxNumFrames];
   double pitch_lag_hz[kMaxNumFrames];
   double spectral_peak[kMaxNumFrames];
   double rms[kMaxNumFrames];
-  int num_frames;
+  size_t num_frames;
   bool silence;
 };
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_COMMON_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.cc
similarity index 81%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.cc
index 9ad8ef95ae..9651975913 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/gmm.h"
+#include "webrtc/modules/audio_processing/vad/gmm.h"
 
 #include <math.h>
 #include <stdlib.h>
@@ -19,13 +19,16 @@ namespace webrtc {
 
 static const int kMaxDimension = 10;
 
-static void RemoveMean(const double* in, const double* mean_vec,
-                       int dimension, double* out) {
+static void RemoveMean(const double* in,
+                       const double* mean_vec,
+                       int dimension,
+                       double* out) {
   for (int n = 0; n < dimension; ++n)
     out[n] = in[n] - mean_vec[n];
 }
 
-static double ComputeExponent(const double* in, const double* covar_inv,
+static double ComputeExponent(const double* in,
+                              const double* covar_inv,
                               int dimension) {
   double q = 0;
   for (int i = 0; i < dimension; ++i) {
@@ -50,7 +53,7 @@ double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters) {
   for (int n = 0; n < gmm_parameters.num_mixtures; n++) {
     RemoveMean(x, mean_vec, gmm_parameters.dimension, v);
     double q = ComputeExponent(v, covar_inv, gmm_parameters.dimension) +
-        gmm_parameters.weight[n];
+               gmm_parameters.weight[n];
     f += exp(q);
     mean_vec += gmm_parameters.dimension;
     covar_inv += gmm_parameters.dimension * gmm_parameters.dimension;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.h
index 90ce95d4dd..9f3e578fef 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_GMM_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_GMM_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_GMM_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_GMM_H_
 
 namespace webrtc {
 
@@ -42,4 +42,4 @@ struct GmmParameters {
 double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters);
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_GMM_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_GMM_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm_unittest.cc
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm_unittest.cc
index 4ca658d732..f8e1bde776 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/gmm_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/gmm_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/gmm.h"
+#include "webrtc/modules/audio_processing/vad/gmm.h"
 
 #include <math.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/noise_gmm_tables.h"
-#include "webrtc/modules/audio_processing/agc/voice_gmm_tables.h"
+#include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h"
+#include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/vad/noise_gmm_tables.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/noise_gmm_tables.h
new file mode 100644
index 0000000000..293af57a2a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/noise_gmm_tables.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// GMM tables for inactive segments. Generated by MakeGmmTables.m.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
+
+static const int kNoiseGmmNumMixtures = 12;
+static const int kNoiseGmmDim = 3;
+
+static const double
+    kNoiseGmmCovarInverse[kNoiseGmmNumMixtures][kNoiseGmmDim][kNoiseGmmDim] = {
+        {{7.36219567592941e+00, 4.83060785179861e-03, 1.23335151497610e-02},
+         {4.83060785179861e-03, 1.65289507047817e-04, -2.41490588169997e-04},
+         {1.23335151497610e-02, -2.41490588169997e-04, 6.59472060689382e-03}},
+        {{8.70265239309140e+00, -5.30636201431086e-04, 5.44014966585347e-03},
+         {-5.30636201431086e-04, 3.11095453521008e-04, -1.86287206836035e-04},
+         {5.44014966585347e-03, -1.86287206836035e-04, 6.29493388790744e-04}},
+        {{4.53467851955055e+00, -3.92977536695197e-03, -2.46521420693317e-03},
+         {-3.92977536695197e-03, 4.94650752632750e-05, -1.08587438501826e-05},
+         {-2.46521420693317e-03, -1.08587438501826e-05, 9.28793975422261e-05}},
+        {{9.26817997114275e-01, -4.03976069276753e-04, -3.56441427392165e-03},
+         {-4.03976069276753e-04, 2.51976251631430e-06, 1.46914206734572e-07},
+         {-3.56441427392165e-03, 1.46914206734572e-07, 8.19914567685373e-05}},
+        {{7.61715986787441e+00, -1.54889041216888e-04, 2.41756280071656e-02},
+         {-1.54889041216888e-04, 3.50282550461672e-07, -6.27251196972490e-06},
+         {2.41756280071656e-02, -6.27251196972490e-06, 1.45061847649872e-02}},
+        {{8.31193642663158e+00, -3.84070508164323e-04, -3.09750630821876e-02},
+         {-3.84070508164323e-04, 3.80433432277336e-07, -1.14321142836636e-06},
+         {-3.09750630821876e-02, -1.14321142836636e-06, 8.35091486289997e-04}},
+        {{9.67283151270894e-01, 5.82465812445039e-05, -3.18350798617053e-03},
+         {5.82465812445039e-05, 2.23762672000318e-07, -7.74196587408623e-07},
+         {-3.18350798617053e-03, -7.74196587408623e-07, 3.85120938338325e-04}},
+        {{8.28066236985388e+00, 5.87634508319763e-05, 6.99303090891743e-03},
+         {5.87634508319763e-05, 2.93746018618058e-07, 3.40843332882272e-07},
+         {6.99303090891743e-03, 3.40843332882272e-07, 1.99379171190344e-04}},
+        {{6.07488998675646e+00, -1.11494526618473e-02, 5.10013111123381e-03},
+         {-1.11494526618473e-02, 6.99238879921751e-04, 5.36718550370870e-05},
+         {5.10013111123381e-03, 5.36718550370870e-05, 5.26909853276753e-04}},
+        {{6.90492021419175e+00, 4.20639355257863e-04, -2.38612752336481e-03},
+         {4.20639355257863e-04, 3.31246767338153e-06, -2.42052288150859e-08},
+         {-2.38612752336481e-03, -2.42052288150859e-08, 4.46608368363412e-04}},
+        {{1.31069150869715e+01, -1.73718583865670e-04, -1.97591814508578e-02},
+         {-1.73718583865670e-04, 2.80451716300124e-07, 9.96570755379865e-07},
+         {-1.97591814508578e-02, 9.96570755379865e-07, 2.41361900868847e-03}},
+        {{4.69566344239814e+00, -2.61077567563690e-04, 5.26359000761433e-03},
+         {-2.61077567563690e-04, 1.82420859823767e-06, -7.83645887541601e-07},
+         {5.26359000761433e-03, -7.83645887541601e-07, 1.33586288288802e-02}}};
+
+static const double kNoiseGmmMean[kNoiseGmmNumMixtures][kNoiseGmmDim] = {
+    {-2.01386094766163e+00, 1.69702162045397e+02, 7.41715804872181e+01},
+    {-1.94684591777290e+00, 1.42398396732668e+02, 1.64186321157831e+02},
+    {-2.29319297562437e+00, 3.86415425589868e+02, 2.13452215267125e+02},
+    {-3.25487177070268e+00, 1.08668712553616e+03, 2.33119949467419e+02},
+    {-2.13159632447467e+00, 4.83821702557717e+03, 6.86786166673740e+01},
+    {-2.26171410780526e+00, 4.79420193982422e+03, 1.53222513286450e+02},
+    {-3.32166740703185e+00, 4.35161135834358e+03, 1.33206448431316e+02},
+    {-2.19290322814343e+00, 3.98325506609408e+03, 2.13249167359934e+02},
+    {-2.02898459255404e+00, 7.37039893155007e+03, 1.12518527491926e+02},
+    {-2.26150236399500e+00, 1.54896745196145e+03, 1.49717357868579e+02},
+    {-2.00417668301790e+00, 3.82434760310304e+03, 1.07438913004312e+02},
+    {-2.30193040814533e+00, 1.43953696546439e+03, 7.04085275122649e+01}};
+
+static const double kNoiseGmmWeights[kNoiseGmmNumMixtures] = {
+    -1.09422832086193e+01,
+    -1.10847897513425e+01,
+    -1.36767587732187e+01,
+    -1.79789356118641e+01,
+    -1.42830169160894e+01,
+    -1.56500228061379e+01,
+    -1.83124990950113e+01,
+    -1.69979436177477e+01,
+    -1.12329424387828e+01,
+    -1.41311785780639e+01,
+    -1.47171861448585e+01,
+    -1.35963362781839e+01};
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
index 0cfa52a010..fce144de6b 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/pitch_based_vad.h"
+#include "webrtc/modules/audio_processing/vad/pitch_based_vad.h"
 
 #include <assert.h>
 #include <math.h>
 #include <string.h>
 
-#include "webrtc/modules/audio_processing/agc/circular_buffer.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
-#include "webrtc/modules/audio_processing/agc/noise_gmm_tables.h"
-#include "webrtc/modules/audio_processing/agc/voice_gmm_tables.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_processing/vad/vad_circular_buffer.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
+#include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h"
+#include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -44,7 +44,7 @@ static double LimitProbability(double p) {
 
 PitchBasedVad::PitchBasedVad()
     : p_prior_(kInitialPriorProbability),
-      circular_buffer_(AgcCircularBuffer::Create(kPosteriorHistorySize)) {
+      circular_buffer_(VadCircularBuffer::Create(kPosteriorHistorySize)) {
   // Setup noise GMM.
   noise_gmm_.dimension = kNoiseGmmDim;
   noise_gmm_.num_mixtures = kNoiseGmmNumMixtures;
@@ -60,7 +60,8 @@ PitchBasedVad::PitchBasedVad()
   voice_gmm_.covar_inverse = &kVoiceGmmCovarInverse[0][0][0];
 }
 
-PitchBasedVad::~PitchBasedVad() {}
+PitchBasedVad::~PitchBasedVad() {
+}
 
 int PitchBasedVad::VoicingProbability(const AudioFeatures& features,
                                       double* p_combined) {
@@ -74,7 +75,7 @@ int PitchBasedVad::VoicingProbability(const AudioFeatures& features,
   const double kLimLowSpectralPeak = 200;
   const double kLimHighSpectralPeak = 2000;
   const double kEps = 1e-12;
-  for (int n = 0; n < features.num_frames; n++) {
+  for (size_t n = 0; n < features.num_frames; n++) {
     gmm_features[0] = features.log_pitch_gain[n];
     gmm_features[1] = features.spectral_peak[n];
     gmm_features[2] = features.pitch_lag_hz[n];
@@ -90,8 +91,9 @@ int PitchBasedVad::VoicingProbability(const AudioFeatures& features,
       pdf_features_given_noise = kEps * pdf_features_given_voice;
     }
 
-    p = p_prior_ * pdf_features_given_voice / (pdf_features_given_voice *
-        p_prior_ + pdf_features_given_noise * (1 - p_prior_));
+    p = p_prior_ * pdf_features_given_voice /
+        (pdf_features_given_voice * p_prior_ +
+         pdf_features_given_noise * (1 - p_prior_));
 
     p = LimitProbability(p);
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.h
similarity index 80%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.h
index 2295505cc3..c502184aea 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad.h
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_PITCH_BASED_VAD_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_PITCH_BASED_VAD_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
-#include "webrtc/modules/audio_processing/agc/gmm.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
+#include "webrtc/modules/audio_processing/vad/gmm.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioFrame;
-class AgcCircularBuffer;
+class VadCircularBuffer;
 
 // Computes the probability of the input audio frame to be active given
 // the corresponding pitch-gain and lag of the frame.
@@ -37,6 +37,7 @@ class PitchBasedVad {
   //               then, computes the voicing probabilities and combine them
   //               with the given values. The result are returned in |p|.
   int VoicingProbability(const AudioFeatures& features, double* p_combined);
+
  private:
   int UpdatePrior(double p);
 
@@ -49,8 +50,8 @@ class PitchBasedVad {
 
   double p_prior_;
 
-  rtc::scoped_ptr<AgcCircularBuffer> circular_buffer_;
+  rtc::scoped_ptr<VadCircularBuffer> circular_buffer_;
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_PITCH_BASED_VAD_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc
similarity index 73%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc
index 3ec0baac95..04ddcab5cb 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_based_vad_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc
@@ -8,20 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/pitch_based_vad.h"
+#include "webrtc/modules/audio_processing/vad/pitch_based_vad.h"
 
 #include <math.h>
 #include <stdio.h>
-#include <string.h>
 
-#include "gtest/gtest.h"
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 TEST(PitchBasedVadTest, VoicingProbabilityTest) {
-  std::string spectral_peak_file_name = test::ResourcePath(
-      "audio_processing/agc/agc_spectral_peak", "dat");
+  std::string spectral_peak_file_name =
+      test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat");
   FILE* spectral_peak_file = fopen(spectral_peak_file_name.c_str(), "rb");
   ASSERT_TRUE(spectral_peak_file != NULL);
 
@@ -51,12 +52,15 @@ TEST(PitchBasedVadTest, VoicingProbabilityTest) {
                sizeof(audio_features.spectral_peak[0]), 1,
                spectral_peak_file) == 1u) {
     double p;
-    ASSERT_EQ(1u, fread(audio_features.log_pitch_gain, sizeof(
-        audio_features.log_pitch_gain[0]), 1, pitch_gain_file));
-    ASSERT_EQ(1u, fread(audio_features.pitch_lag_hz, sizeof(
-        audio_features.pitch_lag_hz[0]), 1, pitch_lag_file));
-    ASSERT_EQ(1u, fread(&reference_activity_probability, sizeof(
-        reference_activity_probability), 1, voicing_prob_file));
+    ASSERT_EQ(1u, fread(audio_features.log_pitch_gain,
+                        sizeof(audio_features.log_pitch_gain[0]), 1,
+                        pitch_gain_file));
+    ASSERT_EQ(1u,
+              fread(audio_features.pitch_lag_hz,
+                    sizeof(audio_features.pitch_lag_hz[0]), 1, pitch_lag_file));
+    ASSERT_EQ(1u, fread(&reference_activity_probability,
+                        sizeof(reference_activity_probability), 1,
+                        voicing_prob_file));
 
     p = 0.5;  // Initialize to the neutral value for combining probabilities.
     EXPECT_EQ(0, vad_.VoicingProbability(audio_features, &p));
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.cc
similarity index 96%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.cc
index b394074bd3..309b45acf5 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/pitch_internal.h"
+#include "webrtc/modules/audio_processing/vad/pitch_internal.h"
 
 #include <cmath>
 
@@ -25,7 +25,6 @@ static void PitchInterpolation(double old_val, const double* in, double* out) {
   out[2] = 0.5 * in[2] + 0.5 * in[3];
 }
 
-
 void GetSubframesPitchParameters(int sampling_rate_hz,
                                  double* gains,
                                  double* lags,
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.h
index ed73760e3a..b25b1a82a2 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_PITCH_INTERNAL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_PITCH_INTERNAL_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
 
 // TODO(turajs): Write a description of this function. Also be consistent with
 // usage of |sampling_rate_hz| vs |kSamplingFreqHz|.
@@ -23,4 +23,4 @@ void GetSubframesPitchParameters(int sampling_rate_hz,
                                  double* log_pitch_gain,
                                  double* pitch_lag_hz);
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_PITCH_INTERNAL_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
similarity index 82%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
index 8998f9014b..8b5959d03e 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pitch_internal_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/pitch_internal.h"
+#include "webrtc/modules/audio_processing/vad/pitch_internal.h"
 
 #include <math.h>
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
 
 TEST(PitchInternalTest, test) {
   const int kSamplingRateHz = 8000;
@@ -26,12 +26,12 @@ TEST(PitchInternalTest, test) {
   double lags[] = {90, 111, 122, 50};
 
   // Expected outputs
-  double expected_log_pitch_gain[] = {-0.541212549898316, -1.45672279045507,
-      -0.80471895621705};
+  double expected_log_pitch_gain[] = {
+      -0.541212549898316, -1.45672279045507, -0.80471895621705};
   double expected_log_old_gain = log(gains[kNumInputParameters - 1]);
 
-  double expected_pitch_lag_hz[] = {92.3076923076923, 70.9010339734121,
-      93.0232558139535};
+  double expected_pitch_lag_hz[] = {
+      92.3076923076923, 70.9010339734121, 93.0232558139535};
   double expected_old_lag = lags[kNumInputParameters - 1];
 
   double log_pitch_gain[kNumOutputParameters];
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
similarity index 68%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
index 3c41e33dd6..9769515c57 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/pole_zero_filter.h"
+#include "webrtc/modules/audio_processing/vad/pole_zero_filter.h"
 
 #include <stdlib.h>
 #include <string.h>
@@ -17,25 +17,21 @@
 namespace webrtc {
 
 PoleZeroFilter* PoleZeroFilter::Create(const float* numerator_coefficients,
-                                       int order_numerator,
+                                       size_t order_numerator,
                                        const float* denominator_coefficients,
-                                       int order_denominator) {
-  if (order_numerator < 0 ||
-      order_denominator < 0 ||
-      order_numerator > kMaxFilterOrder ||
-      order_denominator > kMaxFilterOrder ||
-      denominator_coefficients[0] == 0 ||
-      numerator_coefficients == NULL ||
-      denominator_coefficients == NULL)
+                                       size_t order_denominator) {
+  if (order_numerator > kMaxFilterOrder ||
+      order_denominator > kMaxFilterOrder || denominator_coefficients[0] == 0 ||
+      numerator_coefficients == NULL || denominator_coefficients == NULL)
     return NULL;
   return new PoleZeroFilter(numerator_coefficients, order_numerator,
                             denominator_coefficients, order_denominator);
 }
 
 PoleZeroFilter::PoleZeroFilter(const float* numerator_coefficients,
-                               int order_numerator,
+                               size_t order_numerator,
                                const float* denominator_coefficients,
-                               int order_denominator)
+                               size_t order_denominator)
     : past_input_(),
       past_output_(),
       numerator_coefficients_(),
@@ -49,31 +45,31 @@ PoleZeroFilter::PoleZeroFilter(const float* numerator_coefficients,
          sizeof(denominator_coefficients_[0]) * (order_denominator_ + 1));
 
   if (denominator_coefficients_[0] != 1) {
-    for (int n = 0; n <= order_numerator_; n++)
+    for (size_t n = 0; n <= order_numerator_; n++)
       numerator_coefficients_[n] /= denominator_coefficients_[0];
-    for (int n = 0; n <= order_denominator_; n++)
+    for (size_t n = 0; n <= order_denominator_; n++)
       denominator_coefficients_[n] /= denominator_coefficients_[0];
   }
 }
 
 template <typename T>
-static float FilterArPast(const T* past, int order,
+static float FilterArPast(const T* past, size_t order,
                           const float* coefficients) {
   float sum = 0.0f;
-  int past_index = order - 1;
-  for (int k = 1; k <= order; k++, past_index--)
+  size_t past_index = order - 1;
+  for (size_t k = 1; k <= order; k++, past_index--)
     sum += coefficients[k] * past[past_index];
   return sum;
 }
 
 int PoleZeroFilter::Filter(const int16_t* in,
-                           int num_input_samples,
+                           size_t num_input_samples,
                            float* output) {
-  if (in == NULL || num_input_samples < 0 || output == NULL)
+  if (in == NULL || output == NULL)
     return -1;
   // This is the typical case, just a memcpy.
-  const int k = std::min(num_input_samples, highest_order_);
-  int n;
+  const size_t k = std::min(num_input_samples, highest_order_);
+  size_t n;
   for (n = 0; n < k; n++) {
     output[n] = in[n] * numerator_coefficients_[0];
     output[n] += FilterArPast(&past_input_[n], order_numerator_,
@@ -85,10 +81,10 @@ int PoleZeroFilter::Filter(const int16_t* in,
     past_output_[n + order_denominator_] = output[n];
   }
   if (highest_order_ < num_input_samples) {
-    for (int m = 0; n < num_input_samples; n++, m++) {
+    for (size_t m = 0; n < num_input_samples; n++, m++) {
       output[n] = in[n] * numerator_coefficients_[0];
-      output[n] += FilterArPast(&in[m], order_numerator_,
-                                numerator_coefficients_);
+      output[n] +=
+          FilterArPast(&in[m], order_numerator_, numerator_coefficients_);
       output[n] -= FilterArPast(&output[m], order_denominator_,
                                 denominator_coefficients_);
     }
@@ -99,13 +95,12 @@ int PoleZeroFilter::Filter(const int16_t* in,
            sizeof(output[0]) * order_denominator_);
   } else {
     // Odd case that the length of the input is shorter that filter order.
-    memmove(past_input_, &past_input_[num_input_samples], order_numerator_ *
-            sizeof(past_input_[0]));
-    memmove(past_output_, &past_output_[num_input_samples], order_denominator_ *
-            sizeof(past_output_[0]));
+    memmove(past_input_, &past_input_[num_input_samples],
+            order_numerator_ * sizeof(past_input_[0]));
+    memmove(past_output_, &past_output_[num_input_samples],
+            order_denominator_ * sizeof(past_output_[0]));
   }
   return 0;
 }
 
 }  // namespace webrtc
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter.h
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter.h
index c9d96fdd42..bd13050a5c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_POLE_ZERO_FILTER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_POLE_ZERO_FILTER_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
+
+#include <cstddef>
 
 #include "webrtc/typedefs.h"
 
@@ -20,17 +22,17 @@ class PoleZeroFilter {
   ~PoleZeroFilter() {}
 
   static PoleZeroFilter* Create(const float* numerator_coefficients,
-                                int order_numerator,
+                                size_t order_numerator,
                                 const float* denominator_coefficients,
-                                int order_denominator);
+                                size_t order_denominator);
 
-  int Filter(const int16_t* in, int num_input_samples, float* output);
+  int Filter(const int16_t* in, size_t num_input_samples, float* output);
 
  private:
   PoleZeroFilter(const float* numerator_coefficients,
-                 int order_numerator,
+                 size_t order_numerator,
                  const float* denominator_coefficients,
-                 int order_denominator);
+                 size_t order_denominator);
 
   static const int kMaxFilterOrder = 24;
 
@@ -40,11 +42,11 @@ class PoleZeroFilter {
   float numerator_coefficients_[kMaxFilterOrder + 1];
   float denominator_coefficients_[kMaxFilterOrder + 1];
 
-  int order_numerator_;
-  int order_denominator_;
-  int highest_order_;
+  size_t order_numerator_;
+  size_t order_denominator_;
+  size_t highest_order_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_POLE_ZERO_FILTER_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
similarity index 50%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
index b198b0eed1..492c3f0c94 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/pole_zero_filter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
@@ -8,44 +8,49 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/pole_zero_filter.h"
+#include "webrtc/modules/audio_processing/vad/pole_zero_filter.h"
 
 #include <math.h>
 #include <stdio.h>
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc_internal.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 static const int kInputSamples = 50;
 
-static const int16_t kInput[kInputSamples] = {-2136, -7116, 10715, 2464, 3164,
-    8139, 11393, 24013, -32117, -5544, -27740, 10181, 14190, -24055, -15912,
-    17393, 6359, -9950, -13894, 32432, -23944, 3437, -8381, 19768, 3087, -19795,
-    -5920, 13310, 1407, 3876, 4059, 3524, -23130, 19121, -27900, -24840, 4089,
-    21422, -3625, 3015, -11236, 28856, 13424, 6571, -19761, -6361, 15821, -9469,
-    29727, 32229};
+static const int16_t kInput[kInputSamples] = {
+    -2136,  -7116, 10715,  2464,   3164,   8139,   11393, 24013, -32117, -5544,
+    -27740, 10181, 14190,  -24055, -15912, 17393,  6359,  -9950, -13894, 32432,
+    -23944, 3437,  -8381,  19768,  3087,   -19795, -5920, 13310, 1407,   3876,
+    4059,   3524,  -23130, 19121,  -27900, -24840, 4089,  21422, -3625,  3015,
+    -11236, 28856, 13424,  6571,   -19761, -6361,  15821, -9469, 29727,  32229};
 
-static const float kReferenceOutput[kInputSamples] = {-2082.230472f,
-    -6878.572941f, 10697.090871f, 2358.373952f, 2973.936512f, 7738.580650f,
-    10690.803213f, 22687.091576f, -32676.684717f, -5879.621684f, -27359.297432f,
-    10368.735888f, 13994.584604f, -23676.126249f, -15078.250390f, 17818.253338f,
-    6577.743123f,  -9498.369315f, -13073.651079f, 32460.026588f, -23391.849347f,
-    3953.805667f, -7667.761363f, 19995.153447f, 3185.575477f, -19207.365160f,
-    -5143.103201f, 13756.317237f, 1779.654794f, 4142.269755f, 4209.475034f,
-    3572.991789f, -22509.089546f, 19307.878964f, -27060.439759f, -23319.042810f,
-    5547.685267f, 22312.718676f, -2707.309027f, 3852.358490f, -10135.510093f,
-    29241.509970f, 13394.397233f, 6340.721417f, -19510.207905f, -5908.442086f,
-    15882.301634f, -9211.335255f, 29253.056735f, 30874.443046f};
+static const float kReferenceOutput[kInputSamples] = {
+    -2082.230472f,  -6878.572941f,  10697.090871f,  2358.373952f,
+    2973.936512f,   7738.580650f,   10690.803213f,  22687.091576f,
+    -32676.684717f, -5879.621684f,  -27359.297432f, 10368.735888f,
+    13994.584604f,  -23676.126249f, -15078.250390f, 17818.253338f,
+    6577.743123f,   -9498.369315f,  -13073.651079f, 32460.026588f,
+    -23391.849347f, 3953.805667f,   -7667.761363f,  19995.153447f,
+    3185.575477f,   -19207.365160f, -5143.103201f,  13756.317237f,
+    1779.654794f,   4142.269755f,   4209.475034f,   3572.991789f,
+    -22509.089546f, 19307.878964f,  -27060.439759f, -23319.042810f,
+    5547.685267f,   22312.718676f,  -2707.309027f,  3852.358490f,
+    -10135.510093f, 29241.509970f,  13394.397233f,  6340.721417f,
+    -19510.207905f, -5908.442086f,  15882.301634f,  -9211.335255f,
+    29253.056735f,  30874.443046f};
 
 class PoleZeroFilterTest : public ::testing::Test {
  protected:
   PoleZeroFilterTest()
-     : my_filter_(PoleZeroFilter::Create(
-         kCoeffNumerator, kFilterOrder, kCoeffDenominator, kFilterOrder)) {}
+      : my_filter_(PoleZeroFilter::Create(kCoeffNumerator,
+                                          kFilterOrder,
+                                          kCoeffDenominator,
+                                          kFilterOrder)) {}
 
   ~PoleZeroFilterTest() {}
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.cc
similarity index 77%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.cc
index afd9d7b6dd..1209526a92 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/standalone_vad.h"
+#include "webrtc/modules/audio_processing/vad/standalone_vad.h"
 
 #include <assert.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -21,30 +21,28 @@ namespace webrtc {
 static const int kDefaultStandaloneVadMode = 3;
 
 StandaloneVad::StandaloneVad(VadInst* vad)
-    : vad_(vad),
-      buffer_(),
-      index_(0),
-      mode_(kDefaultStandaloneVadMode) {}
+    : vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {
+}
 
 StandaloneVad::~StandaloneVad() {
   WebRtcVad_Free(vad_);
 }
 
 StandaloneVad* StandaloneVad::Create() {
-  VadInst* vad = NULL;
-  if (WebRtcVad_Create(&vad) < 0)
-    return NULL;
+  VadInst* vad = WebRtcVad_Create();
+  if (!vad)
+    return nullptr;
 
   int err = WebRtcVad_Init(vad);
   err |= WebRtcVad_set_mode(vad, kDefaultStandaloneVadMode);
   if (err != 0) {
     WebRtcVad_Free(vad);
-    return NULL;
+    return nullptr;
   }
   return new StandaloneVad(vad);
 }
 
-int StandaloneVad::AddAudio(const int16_t* data, int length) {
+int StandaloneVad::AddAudio(const int16_t* data, size_t length) {
   if (length != kLength10Ms)
     return -1;
 
@@ -59,11 +57,11 @@ int StandaloneVad::AddAudio(const int16_t* data, int length) {
   return 0;
 }
 
-int StandaloneVad::GetActivity(double* p, int length_p) {
+int StandaloneVad::GetActivity(double* p, size_t length_p) {
   if (index_ == 0)
     return -1;
 
-  const int num_frames = index_ / kLength10Ms;
+  const size_t num_frames = index_ / kLength10Ms;
   if (num_frames > length_p)
     return -1;
   assert(WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_) == 0);
@@ -75,7 +73,7 @@ int StandaloneVad::GetActivity(double* p, int length_p) {
     p[0] = 0.01;  // Arbitrary but small and non-zero.
   else
     p[0] = 0.5;  // 0.5 is neutral values when combinned by other probabilities.
-  for (int n = 1; n < num_frames; n++)
+  for (size_t n = 1; n < num_frames; n++)
     p[n] = p[0];
   // Reset the buffer to start from the beginning.
   index_ = 0;
@@ -93,4 +91,3 @@ int StandaloneVad::set_mode(int mode) {
 }
 
 }  // namespace webrtc
-
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.h
similarity index 90%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.h
index 3cace01286..6a25424dab 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -41,10 +41,10 @@ class StandaloneVad {
   // classified as passive. In this way, when probabilities are combined, the
   // effect of the stand-alone VAD is neutral if the input is classified as
   // active.
-  int GetActivity(double* p, int length_p);
+  int GetActivity(double* p, size_t length_p);
 
   // Expecting 10 ms of 16 kHz audio to be pushed in.
-  int AddAudio(const int16_t* data, int length);
+  int AddAudio(const int16_t* data, size_t length);
 
   // Set aggressiveness of VAD, 0 is the least aggressive and 3 is the most
   // aggressive mode. Returns -1 if the input is less than 0 or larger than 3,
@@ -56,12 +56,12 @@ class StandaloneVad {
  private:
   explicit StandaloneVad(VadInst* vad);
 
-  static const int kMaxNum10msFrames = 3;
+  static const size_t kMaxNum10msFrames = 3;
 
   // TODO(turajs): Is there a way to use scoped-pointer here?
   VadInst* vad_;
   int16_t buffer_[kMaxNum10msFrames * kLength10Ms];
-  int index_;
+  size_t index_;
   int mode_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
index 3887828ed8..1d1dcc7066 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/standalone_vad_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -8,27 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/standalone_vad.h"
+#include "webrtc/modules/audio_processing/vad/standalone_vad.h"
 
 #include <string.h>
 
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 TEST(StandaloneVadTest, Api) {
   rtc::scoped_ptr<StandaloneVad> vad(StandaloneVad::Create());
-  int16_t data[kLength10Ms] = { 0 };
+  int16_t data[kLength10Ms] = {0};
 
   // Valid frame length (for 32 kHz rate), but not what the VAD is expecting.
   EXPECT_EQ(-1, vad->AddAudio(data, 320));
 
-  const int kMaxNumFrames = 3;
+  const size_t kMaxNumFrames = 3;
   double p[kMaxNumFrames];
-  for (int n = 0; n < kMaxNumFrames; n++)
+  for (size_t n = 0; n < kMaxNumFrames; n++)
     EXPECT_EQ(0, vad->AddAudio(data, kLength10Ms));
 
   // Pretend |p| is shorter that it should be.
@@ -40,7 +40,7 @@ TEST(StandaloneVadTest, Api) {
   EXPECT_EQ(-1, vad->GetActivity(p, kMaxNumFrames));
 
   // Should reset and result in one buffer.
-  for (int n = 0; n < kMaxNumFrames + 1; n++)
+  for (size_t n = 0; n < kMaxNumFrames + 1; n++)
     EXPECT_EQ(0, vad->AddAudio(data, kLength10Ms));
   EXPECT_EQ(0, vad->GetActivity(p, 1));
 
@@ -54,10 +54,14 @@ TEST(StandaloneVadTest, Api) {
   EXPECT_EQ(kMode, vad->mode());
 }
 
+#if defined(WEBRTC_IOS)
+TEST(StandaloneVadTest, DISABLED_ActivityDetection) {
+#else
 TEST(StandaloneVadTest, ActivityDetection) {
+#endif
   rtc::scoped_ptr<StandaloneVad> vad(StandaloneVad::Create());
   const size_t kDataLength = kLength10Ms;
-  int16_t data[kDataLength] = { 0 };
+  int16_t data[kDataLength] = {0};
 
   FILE* pcm_file =
       fopen(test::ResourcePath("audio_processing/agc/agc_audio", "pcm").c_str(),
@@ -100,4 +104,4 @@ TEST(StandaloneVadTest, ActivityDetection) {
   fclose(reference_file);
   fclose(pcm_file);
 }
-}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
similarity index 70%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
index dc4a5a711c..1a595597b6 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
@@ -8,44 +8,46 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h"
 
 #include <math.h>
 #include <stdio.h>
 
 #include "webrtc/common_audio/fft4g.h"
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc_internal.h"
-#include "webrtc/modules/audio_processing/agc/pitch_internal.h"
-#include "webrtc/modules/audio_processing/agc/pole_zero_filter.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h"
+#include "webrtc/modules/audio_processing/vad/pitch_internal.h"
+#include "webrtc/modules/audio_processing/vad/pole_zero_filter.h"
 extern "C" {
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/codec.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h"
 }
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 // The following structures are declared anonymous in iSAC's structs.h. To
 // forward declare them, we use this derived class trick.
-struct AgcAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
-struct AgcAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
+struct VadAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
+struct VadAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
 
-static const float kFrequencyResolution = kSampleRateHz /
-    static_cast<float>(AgcAudioProc::kDftSize);
+static const float kFrequencyResolution =
+    kSampleRateHz / static_cast<float>(VadAudioProc::kDftSize);
 static const int kSilenceRms = 5;
 
-// TODO(turajs): Make a Create or Init for AgcAudioProc.
-AgcAudioProc::AgcAudioProc()
+// TODO(turajs): Make a Create or Init for VadAudioProc.
+VadAudioProc::VadAudioProc()
     : audio_buffer_(),
       num_buffer_samples_(kNumPastSignalSamples),
       log_old_gain_(-2),
       old_lag_(50),  // Arbitrary but valid as pitch-lag (in samples).
       pitch_analysis_handle_(new PitchAnalysisStruct),
       pre_filter_handle_(new PreFiltBankstr),
-      high_pass_filter_(PoleZeroFilter::Create(
-          kCoeffNumerator, kFilterOrder, kCoeffDenominator, kFilterOrder)) {
+      high_pass_filter_(PoleZeroFilter::Create(kCoeffNumerator,
+                                               kFilterOrder,
+                                               kCoeffDenominator,
+                                               kFilterOrder)) {
   static_assert(kNumPastSignalSamples + kNumSubframeSamples ==
                     sizeof(kLpcAnalWin) / sizeof(kLpcAnalWin[0]),
                 "lpc analysis window incorrect size");
@@ -64,16 +66,17 @@ AgcAudioProc::AgcAudioProc()
   WebRtcIsac_InitPitchAnalysis(pitch_analysis_handle_.get());
 }
 
-AgcAudioProc::~AgcAudioProc() {}
+VadAudioProc::~VadAudioProc() {
+}
 
-void AgcAudioProc::ResetBuffer() {
+void VadAudioProc::ResetBuffer() {
   memcpy(audio_buffer_, &audio_buffer_[kNumSamplesToProcess],
          sizeof(audio_buffer_[0]) * kNumPastSignalSamples);
   num_buffer_samples_ = kNumPastSignalSamples;
 }
 
-int AgcAudioProc::ExtractFeatures(const int16_t* frame,
-                                  int length,
+int VadAudioProc::ExtractFeatures(const int16_t* frame,
+                                  size_t length,
                                   AudioFeatures* features) {
   features->num_frames = 0;
   if (length != kNumSubframeSamples) {
@@ -85,7 +88,7 @@ int AgcAudioProc::ExtractFeatures(const int16_t* frame,
   // classification.
   if (high_pass_filter_->Filter(frame, kNumSubframeSamples,
                                 &audio_buffer_[num_buffer_samples_]) != 0) {
-      return -1;
+    return -1;
   }
 
   num_buffer_samples_ += kNumSubframeSamples;
@@ -97,7 +100,7 @@ int AgcAudioProc::ExtractFeatures(const int16_t* frame,
   features->silence = false;
 
   Rms(features->rms, kMaxNumFrames);
-  for (int i = 0; i < kNum10msSubframes; ++i) {
+  for (size_t i = 0; i < kNum10msSubframes; ++i) {
     if (features->rms[i] < kSilenceRms) {
       // PitchAnalysis can cause NaNs in the pitch gain if it's fed silence.
       // Bail out here instead.
@@ -115,33 +118,34 @@ int AgcAudioProc::ExtractFeatures(const int16_t* frame,
 }
 
 // Computes |kLpcOrder + 1| correlation coefficients.
-void AgcAudioProc::SubframeCorrelation(double* corr, int length_corr,
-                                       int subframe_index) {
+void VadAudioProc::SubframeCorrelation(double* corr,
+                                       size_t length_corr,
+                                       size_t subframe_index) {
   assert(length_corr >= kLpcOrder + 1);
   double windowed_audio[kNumSubframeSamples + kNumPastSignalSamples];
-  int buffer_index = subframe_index * kNumSubframeSamples;
+  size_t buffer_index = subframe_index * kNumSubframeSamples;
 
-  for (int n = 0; n < kNumSubframeSamples + kNumPastSignalSamples; n++)
+  for (size_t n = 0; n < kNumSubframeSamples + kNumPastSignalSamples; n++)
     windowed_audio[n] = audio_buffer_[buffer_index++] * kLpcAnalWin[n];
 
-  WebRtcIsac_AutoCorr(corr, windowed_audio, kNumSubframeSamples +
-                      kNumPastSignalSamples, kLpcOrder);
+  WebRtcIsac_AutoCorr(corr, windowed_audio,
+                      kNumSubframeSamples + kNumPastSignalSamples, kLpcOrder);
 }
 
 // Compute |kNum10msSubframes| sets of LPC coefficients, one per 10 ms input.
 // The analysis window is 15 ms long and it is centered on the first half of
 // each 10ms sub-frame. This is equivalent to computing LPC coefficients for the
 // first half of each 10 ms subframe.
-void AgcAudioProc::GetLpcPolynomials(double* lpc, int length_lpc) {
+void VadAudioProc::GetLpcPolynomials(double* lpc, size_t length_lpc) {
   assert(length_lpc >= kNum10msSubframes * (kLpcOrder + 1));
   double corr[kLpcOrder + 1];
   double reflec_coeff[kLpcOrder];
-  for (int i = 0, offset_lpc = 0; i < kNum10msSubframes;
-      i++, offset_lpc += kLpcOrder + 1) {
+  for (size_t i = 0, offset_lpc = 0; i < kNum10msSubframes;
+       i++, offset_lpc += kLpcOrder + 1) {
     SubframeCorrelation(corr, kLpcOrder + 1, i);
     corr[0] *= 1.0001;
     // This makes Lev-Durb a bit more stable.
-    for (int k = 0; k < kLpcOrder + 1; k++) {
+    for (size_t k = 0; k < kLpcOrder + 1; k++) {
       corr[k] *= kCorrWeight[k];
     }
     WebRtcIsac_LevDurb(&lpc[offset_lpc], reflec_coeff, corr, kLpcOrder);
@@ -150,7 +154,8 @@ void AgcAudioProc::GetLpcPolynomials(double* lpc, int length_lpc) {
 
 // Fit a second order curve to these 3 points and find the location of the
 // extremum. The points are inverted before curve fitting.
-static float QuadraticInterpolation(float prev_val, float curr_val,
+static float QuadraticInterpolation(float prev_val,
+                                    float curr_val,
                                     float next_val) {
   // Doing the interpolation in |1 / A(z)|^2.
   float fractional_index = 0;
@@ -158,8 +163,8 @@ static float QuadraticInterpolation(float prev_val, float curr_val,
   prev_val = 1.0f / prev_val;
   curr_val = 1.0f / curr_val;
 
-  fractional_index = -(next_val - prev_val) * 0.5f / (next_val + prev_val -
-      2.f * curr_val);
+  fractional_index =
+      -(next_val - prev_val) * 0.5f / (next_val + prev_val - 2.f * curr_val);
   assert(fabs(fractional_index) < 1);
   return fractional_index;
 }
@@ -169,32 +174,33 @@ static float QuadraticInterpolation(float prev_val, float curr_val,
 // with the local minimum of A(z). It saves complexity, as we save one
 // inversion. Furthermore, we find the first local maximum of magnitude squared,
 // to save on one square root.
-void AgcAudioProc::FindFirstSpectralPeaks(double* f_peak, int length_f_peak) {
+void VadAudioProc::FindFirstSpectralPeaks(double* f_peak,
+                                          size_t length_f_peak) {
   assert(length_f_peak >= kNum10msSubframes);
   double lpc[kNum10msSubframes * (kLpcOrder + 1)];
   // For all sub-frames.
   GetLpcPolynomials(lpc, kNum10msSubframes * (kLpcOrder + 1));
 
-  const int kNumDftCoefficients = kDftSize / 2 + 1;
+  const size_t kNumDftCoefficients = kDftSize / 2 + 1;
   float data[kDftSize];
 
-  for (int i = 0; i < kNum10msSubframes; i++) {
+  for (size_t i = 0; i < kNum10msSubframes; i++) {
     // Convert to float with zero pad.
     memset(data, 0, sizeof(data));
-    for (int n = 0; n < kLpcOrder + 1; n++) {
+    for (size_t n = 0; n < kLpcOrder + 1; n++) {
       data[n] = static_cast<float>(lpc[i * (kLpcOrder + 1) + n]);
     }
     // Transform to frequency domain.
     WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
 
-    int index_peak = 0;
+    size_t index_peak = 0;
     float prev_magn_sqr = data[0] * data[0];
     float curr_magn_sqr = data[2] * data[2] + data[3] * data[3];
     float next_magn_sqr;
     bool found_peak = false;
-    for (int n = 2; n < kNumDftCoefficients - 1; n++) {
-      next_magn_sqr = data[2 * n] * data[2 * n] +
-          data[2 * n + 1] * data[2 * n + 1];
+    for (size_t n = 2; n < kNumDftCoefficients - 1; n++) {
+      next_magn_sqr =
+          data[2 * n] * data[2 * n] + data[2 * n + 1] * data[2 * n + 1];
       if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
         found_peak = true;
         index_peak = n - 1;
@@ -213,16 +219,17 @@ void AgcAudioProc::FindFirstSpectralPeaks(double* f_peak, int length_f_peak) {
     } else {
       // A peak is found, do a simple quadratic interpolation to get a more
       // accurate estimate of the peak location.
-      fractional_index = QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr,
-                                                next_magn_sqr);
+      fractional_index =
+          QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr, next_magn_sqr);
     }
     f_peak[i] = (index_peak + fractional_index) * kFrequencyResolution;
   }
 }
 
 // Using iSAC functions to estimate pitch gains & lags.
-void AgcAudioProc::PitchAnalysis(double* log_pitch_gains, double* pitch_lags_hz,
-                                 int length) {
+void VadAudioProc::PitchAnalysis(double* log_pitch_gains,
+                                 double* pitch_lags_hz,
+                                 size_t length) {
   // TODO(turajs): This can be "imported" from iSAC & and the next two
   // constants.
   assert(length >= kNum10msSubframes);
@@ -241,28 +248,27 @@ void AgcAudioProc::PitchAnalysis(double* log_pitch_gains, double* pitch_lags_hz,
                                     kNumLookaheadSamples];
 
   // Split signal to lower and upper bands
-  WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples],
-                                 lower, upper, lower_lookahead, upper_lookahead,
+  WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples], lower,
+                                 upper, lower_lookahead, upper_lookahead,
                                  pre_filter_handle_.get());
   WebRtcIsac_PitchAnalysis(lower_lookahead, lower_lookahead_pre_filter,
                            pitch_analysis_handle_.get(), lags, gains);
 
   // Lags are computed on lower-band signal with sampling rate half of the
   // input signal.
-  GetSubframesPitchParameters(kSampleRateHz / 2, gains, lags,
-                              kNumPitchSubframes, kNum10msSubframes,
-                              &log_old_gain_, &old_lag_,
-                              log_pitch_gains, pitch_lags_hz);
+  GetSubframesPitchParameters(
+      kSampleRateHz / 2, gains, lags, kNumPitchSubframes, kNum10msSubframes,
+      &log_old_gain_, &old_lag_, log_pitch_gains, pitch_lags_hz);
 }
 
-void AgcAudioProc::Rms(double* rms, int length_rms) {
+void VadAudioProc::Rms(double* rms, size_t length_rms) {
   assert(length_rms >= kNum10msSubframes);
-  int offset = kNumPastSignalSamples;
-  for (int i = 0; i < kNum10msSubframes; i++) {
+  size_t offset = kNumPastSignalSamples;
+  for (size_t i = 0; i < kNum10msSubframes; i++) {
     rms[i] = 0;
-    for (int n = 0; n < kNumSubframeSamples; n++, offset++)
+    for (size_t n = 0; n < kNumSubframeSamples; n++, offset++)
       rms[i] += audio_buffer_[offset] * audio_buffer_[offset];
-    rms[i] =  sqrt(rms[i] / kNumSubframeSamples);
+    rms[i] = sqrt(rms[i] / kNumSubframeSamples);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc.h
similarity index 57%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc.h
index 8c8fc31552..85500aed84 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -20,55 +20,61 @@ namespace webrtc {
 class AudioFrame;
 class PoleZeroFilter;
 
-class AgcAudioProc {
+class VadAudioProc {
  public:
   // Forward declare iSAC structs.
   struct PitchAnalysisStruct;
   struct PreFiltBankstr;
 
-  AgcAudioProc();
-  ~AgcAudioProc();
+  VadAudioProc();
+  ~VadAudioProc();
 
   int ExtractFeatures(const int16_t* audio_frame,
-                      int length,
+                      size_t length,
                       AudioFeatures* audio_features);
 
-  static const int kDftSize = 512;
+  static const size_t kDftSize = 512;
 
  private:
-  void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, int length);
-  void SubframeCorrelation(double* corr, int lenght_corr, int subframe_index);
-  void GetLpcPolynomials(double* lpc, int length_lpc);
-  void FindFirstSpectralPeaks(double* f_peak, int length_f_peak);
-  void Rms(double* rms, int length_rms);
+  void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, size_t length);
+  void SubframeCorrelation(double* corr,
+                           size_t length_corr,
+                           size_t subframe_index);
+  void GetLpcPolynomials(double* lpc, size_t length_lpc);
+  void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
+  void Rms(double* rms, size_t length_rms);
   void ResetBuffer();
 
   // To compute spectral peak we perform LPC analysis to get spectral envelope.
   // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
   // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
   // we need 5 ms of past signal to create the input of LPC analysis.
-  static const int kNumPastSignalSamples = kSampleRateHz / 200;
+  static const size_t kNumPastSignalSamples =
+      static_cast<size_t>(kSampleRateHz / 200);
 
   // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
   // all the code recognize it as "no-error."
   static const int kNoError = 0;
 
-  static const int kNum10msSubframes = 3;
-  static const int kNumSubframeSamples = kSampleRateHz / 100;
-  static const int kNumSamplesToProcess = kNum10msSubframes *
+  static const size_t kNum10msSubframes = 3;
+  static const size_t kNumSubframeSamples =
+      static_cast<size_t>(kSampleRateHz / 100);
+  static const size_t kNumSamplesToProcess =
+      kNum10msSubframes *
       kNumSubframeSamples;  // Samples in 30 ms @ given sampling rate.
-  static const int kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess;
-  static const int kIpLength = kDftSize >> 1;
-  static const int kWLength = kDftSize >> 1;
+  static const size_t kBufferLength =
+      kNumPastSignalSamples + kNumSamplesToProcess;
+  static const size_t kIpLength = kDftSize >> 1;
+  static const size_t kWLength = kDftSize >> 1;
 
-  static const int kLpcOrder = 16;
+  static const size_t kLpcOrder = 16;
 
-  int ip_[kIpLength];
+  size_t ip_[kIpLength];
   float w_fft_[kWLength];
 
   // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
   float audio_buffer_[kBufferLength];
-  int num_buffer_samples_;
+  size_t num_buffer_samples_;
 
   double log_old_gain_;
   double old_lag_;
@@ -80,4 +86,4 @@ class AgcAudioProc {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
new file mode 100644
index 0000000000..45586b9be6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
+
+namespace webrtc {
+
+// These values should match MATLAB counterparts for unit-tests to pass.
+static const double kCorrWeight[] = {1.000000,
+                                     0.985000,
+                                     0.970225,
+                                     0.955672,
+                                     0.941337,
+                                     0.927217,
+                                     0.913308,
+                                     0.899609,
+                                     0.886115,
+                                     0.872823,
+                                     0.859730,
+                                     0.846834,
+                                     0.834132,
+                                     0.821620,
+                                     0.809296,
+                                     0.797156,
+                                     0.785199};
+
+static const double kLpcAnalWin[] = {
+    0.00000000, 0.01314436, 0.02628645, 0.03942400, 0.05255473, 0.06567639,
+    0.07878670, 0.09188339, 0.10496421, 0.11802689, 0.13106918, 0.14408883,
+    0.15708358, 0.17005118, 0.18298941, 0.19589602, 0.20876878, 0.22160547,
+    0.23440387, 0.24716177, 0.25987696, 0.27254725, 0.28517045, 0.29774438,
+    0.31026687, 0.32273574, 0.33514885, 0.34750406, 0.35979922, 0.37203222,
+    0.38420093, 0.39630327, 0.40833713, 0.42030043, 0.43219112, 0.44400713,
+    0.45574642, 0.46740697, 0.47898676, 0.49048379, 0.50189608, 0.51322164,
+    0.52445853, 0.53560481, 0.54665854, 0.55761782, 0.56848075, 0.57924546,
+    0.58991008, 0.60047278, 0.61093173, 0.62128512, 0.63153117, 0.64166810,
+    0.65169416, 0.66160761, 0.67140676, 0.68108990, 0.69065536, 0.70010148,
+    0.70942664, 0.71862923, 0.72770765, 0.73666033, 0.74548573, 0.75418233,
+    0.76274862, 0.77118312, 0.77948437, 0.78765094, 0.79568142, 0.80357442,
+    0.81132858, 0.81894256, 0.82641504, 0.83374472, 0.84093036, 0.84797069,
+    0.85486451, 0.86161063, 0.86820787, 0.87465511, 0.88095122, 0.88709512,
+    0.89308574, 0.89892206, 0.90460306, 0.91012776, 0.91549520, 0.92070447,
+    0.92575465, 0.93064488, 0.93537432, 0.93994213, 0.94434755, 0.94858979,
+    0.95266814, 0.95658189, 0.96033035, 0.96391289, 0.96732888, 0.97057773,
+    0.97365889, 0.97657181, 0.97931600, 0.98189099, 0.98429632, 0.98653158,
+    0.98859639, 0.99049038, 0.99221324, 0.99376466, 0.99514438, 0.99635215,
+    0.99738778, 0.99825107, 0.99894188, 0.99946010, 0.99980562, 0.99997840,
+    0.99997840, 0.99980562, 0.99946010, 0.99894188, 0.99825107, 0.99738778,
+    0.99635215, 0.99514438, 0.99376466, 0.99221324, 0.99049038, 0.98859639,
+    0.98653158, 0.98429632, 0.98189099, 0.97931600, 0.97657181, 0.97365889,
+    0.97057773, 0.96732888, 0.96391289, 0.96033035, 0.95658189, 0.95266814,
+    0.94858979, 0.94434755, 0.93994213, 0.93537432, 0.93064488, 0.92575465,
+    0.92070447, 0.91549520, 0.91012776, 0.90460306, 0.89892206, 0.89308574,
+    0.88709512, 0.88095122, 0.87465511, 0.86820787, 0.86161063, 0.85486451,
+    0.84797069, 0.84093036, 0.83374472, 0.82641504, 0.81894256, 0.81132858,
+    0.80357442, 0.79568142, 0.78765094, 0.77948437, 0.77118312, 0.76274862,
+    0.75418233, 0.74548573, 0.73666033, 0.72770765, 0.71862923, 0.70942664,
+    0.70010148, 0.69065536, 0.68108990, 0.67140676, 0.66160761, 0.65169416,
+    0.64166810, 0.63153117, 0.62128512, 0.61093173, 0.60047278, 0.58991008,
+    0.57924546, 0.56848075, 0.55761782, 0.54665854, 0.53560481, 0.52445853,
+    0.51322164, 0.50189608, 0.49048379, 0.47898676, 0.46740697, 0.45574642,
+    0.44400713, 0.43219112, 0.42030043, 0.40833713, 0.39630327, 0.38420093,
+    0.37203222, 0.35979922, 0.34750406, 0.33514885, 0.32273574, 0.31026687,
+    0.29774438, 0.28517045, 0.27254725, 0.25987696, 0.24716177, 0.23440387,
+    0.22160547, 0.20876878, 0.19589602, 0.18298941, 0.17005118, 0.15708358,
+    0.14408883, 0.13106918, 0.11802689, 0.10496421, 0.09188339, 0.07878670,
+    0.06567639, 0.05255473, 0.03942400, 0.02628645, 0.01314436, 0.00000000};
+
+static const size_t kFilterOrder = 2;
+static const float kCoeffNumerator[kFilterOrder + 1] = {0.974827f,
+                                                        -1.949650f,
+                                                        0.974827f};
+static const float kCoeffDenominator[kFilterOrder + 1] = {1.0f,
+                                                          -1.971999f,
+                                                          0.972457f};
+
+static_assert(kFilterOrder + 1 ==
+                  sizeof(kCoeffNumerator) / sizeof(kCoeffNumerator[0]),
+              "numerator coefficients incorrect size");
+static_assert(kFilterOrder + 1 ==
+                  sizeof(kCoeffDenominator) / sizeof(kCoeffDenominator[0]),
+              "denominator coefficients incorrect size");
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROCESSING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
similarity index 83%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
index 9534aec2ec..a8a4ead2e3 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc_audio_proc_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -12,20 +12,22 @@
 // routines. However, interpolation of pitch-gain and lags is in a separate
 // class and has its own unit-test.
 
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h"
 
 #include <math.h>
 #include <stdio.h>
 
-#include "gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
-  AgcAudioProc audioproc;
+  VadAudioProc audioproc;
 
   std::string peak_file_name =
       test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat");
@@ -39,7 +41,7 @@ TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
 
   // Read 10 ms audio in each iteration.
   const size_t kDataLength = kLength10Ms;
-  int16_t data[kDataLength] = { 0 };
+  int16_t data[kDataLength] = {0};
   AudioFeatures features;
   double sp[kMaxNumFrames];
   while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) {
@@ -49,7 +51,7 @@ TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
       // Read reference values.
       const size_t num_frames = features.num_frames;
       ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file));
-      for (int n = 0; n < features.num_frames; n++)
+      for (size_t n = 0; n < features.num_frames; n++)
         EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
     }
   }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
index 8ecb76008f..d337893c45 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
@@ -8,42 +8,44 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/circular_buffer.h"
+#include "webrtc/modules/audio_processing/vad/vad_circular_buffer.h"
 
 #include <assert.h>
 #include <stdlib.h>
 
 namespace webrtc {
 
-AgcCircularBuffer::AgcCircularBuffer(int buffer_size)
+VadCircularBuffer::VadCircularBuffer(int buffer_size)
     : buffer_(new double[buffer_size]),
       is_full_(false),
       index_(0),
       buffer_size_(buffer_size),
-      sum_(0) {}
+      sum_(0) {
+}
 
-AgcCircularBuffer::~AgcCircularBuffer() {}
+VadCircularBuffer::~VadCircularBuffer() {
+}
 
-void AgcCircularBuffer::Reset() {
+void VadCircularBuffer::Reset() {
   is_full_ = false;
   index_ = 0;
   sum_ = 0;
 }
 
-AgcCircularBuffer* AgcCircularBuffer::Create(int buffer_size) {
+VadCircularBuffer* VadCircularBuffer::Create(int buffer_size) {
   if (buffer_size <= 0)
     return NULL;
-  return new AgcCircularBuffer(buffer_size);
+  return new VadCircularBuffer(buffer_size);
 }
 
-double AgcCircularBuffer::Oldest() const {
+double VadCircularBuffer::Oldest() const {
   if (!is_full_)
     return buffer_[0];
   else
     return buffer_[index_];
 }
 
-double AgcCircularBuffer::Mean() {
+double VadCircularBuffer::Mean() {
   double m;
   if (is_full_) {
     m = sum_ / buffer_size_;
@@ -56,7 +58,7 @@ double AgcCircularBuffer::Mean() {
   return m;
 }
 
-void AgcCircularBuffer::Insert(double value) {
+void VadCircularBuffer::Insert(double value) {
   if (is_full_) {
     sum_ -= buffer_[index_];
   }
@@ -68,13 +70,13 @@ void AgcCircularBuffer::Insert(double value) {
     index_ = 0;
   }
 }
-int AgcCircularBuffer::BufferLevel() {
+int VadCircularBuffer::BufferLevel() {
   if (is_full_)
     return buffer_size_;
   return index_;
 }
 
-int AgcCircularBuffer::Get(int index, double* value) const {
+int VadCircularBuffer::Get(int index, double* value) const {
   int err = ConvertToLinearIndex(&index);
   if (err < 0)
     return -1;
@@ -82,7 +84,7 @@ int AgcCircularBuffer::Get(int index, double* value) const {
   return 0;
 }
 
-int AgcCircularBuffer::Set(int index, double value) {
+int VadCircularBuffer::Set(int index, double value) {
   int err = ConvertToLinearIndex(&index);
   if (err < 0)
     return -1;
@@ -93,7 +95,7 @@ int AgcCircularBuffer::Set(int index, double value) {
   return 0;
 }
 
-int AgcCircularBuffer::ConvertToLinearIndex(int* index) const {
+int VadCircularBuffer::ConvertToLinearIndex(int* index) const {
   if (*index < 0 || *index >= buffer_size_)
     return -1;
 
@@ -106,7 +108,7 @@ int AgcCircularBuffer::ConvertToLinearIndex(int* index) const {
   return 0;
 }
 
-int AgcCircularBuffer::RemoveTransient(int width_threshold,
+int VadCircularBuffer::RemoveTransient(int width_threshold,
                                        double val_threshold) {
   if (!is_full_ && index_ < width_threshold + 2)
     return 0;
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.h
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.h
index eee60977d1..5238f77257 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_CIRCULAR_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_CIRCULAR_BUFFER_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
 
 #include "webrtc/base/scoped_ptr.h"
 
@@ -21,10 +21,10 @@ namespace webrtc {
 // It is used in class "PitchBasedActivity" to keep track of posterior
 // probabilities in the past few seconds. The posterior probabilities are used
 // to recursively update prior probabilities.
-class AgcCircularBuffer {
+class VadCircularBuffer {
  public:
-  static AgcCircularBuffer* Create(int buffer_size);
-  ~AgcCircularBuffer();
+  static VadCircularBuffer* Create(int buffer_size);
+  ~VadCircularBuffer();
 
   // If buffer is wrapped around.
   bool is_full() const { return is_full_; }
@@ -44,7 +44,7 @@ class AgcCircularBuffer {
   int RemoveTransient(int width_threshold, double val_threshold);
 
  private:
-  explicit AgcCircularBuffer(int buffer_size);
+  explicit VadCircularBuffer(int buffer_size);
   // Get previous values. |index = 0| corresponds to the most recent
   // insertion. |index = 1| is the one before the most recent insertion, and
   // so on.
@@ -66,4 +66,4 @@ class AgcCircularBuffer {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_CIRCULAR_BUFFER_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc
similarity index 76%
rename from media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc
index e80a5d0fa1..11945e042c 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/agc/circular_buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/agc/circular_buffer.h"
+#include "webrtc/modules/audio_processing/vad/vad_circular_buffer.h"
 
 #include <stdio.h>
 
@@ -22,7 +22,7 @@ static const double kValThreshold = 1.0;
 static const int kLongBuffSize = 100;
 static const int kShortBuffSize = 10;
 
-static void InsertSequentially(int k, AgcCircularBuffer* circular_buffer) {
+static void InsertSequentially(int k, VadCircularBuffer* circular_buffer) {
   double mean_val;
   for (int n = 1; n <= k; n++) {
     EXPECT_TRUE(!circular_buffer->is_full());
@@ -32,19 +32,20 @@ static void InsertSequentially(int k, AgcCircularBuffer* circular_buffer) {
   }
 }
 
-static void Insert(double value, int num_insertion,
-                   AgcCircularBuffer* circular_buffer) {
+static void Insert(double value,
+                   int num_insertion,
+                   VadCircularBuffer* circular_buffer) {
   for (int n = 0; n < num_insertion; n++)
     circular_buffer->Insert(value);
 }
 
-static void InsertZeros(int num_zeros, AgcCircularBuffer* circular_buffer) {
+static void InsertZeros(int num_zeros, VadCircularBuffer* circular_buffer) {
   Insert(0.0, num_zeros, circular_buffer);
 }
 
-TEST(AgcCircularBufferTest, GeneralTest) {
-  rtc::scoped_ptr<AgcCircularBuffer> circular_buffer(
-      AgcCircularBuffer::Create(kShortBuffSize));
+TEST(VadCircularBufferTest, GeneralTest) {
+  rtc::scoped_ptr<VadCircularBuffer> circular_buffer(
+      VadCircularBuffer::Create(kShortBuffSize));
   double mean_val;
 
   // Mean should return zero if nothing is inserted.
@@ -70,9 +71,9 @@ TEST(AgcCircularBufferTest, GeneralTest) {
   EXPECT_TRUE(circular_buffer->is_full());
 }
 
-TEST(AgcCircularBufferTest, TransientsRemoval) {
-  rtc::scoped_ptr<AgcCircularBuffer> circular_buffer(
-      AgcCircularBuffer::Create(kLongBuffSize));
+TEST(VadCircularBufferTest, TransientsRemoval) {
+  rtc::scoped_ptr<VadCircularBuffer> circular_buffer(
+      VadCircularBuffer::Create(kLongBuffSize));
   // Let the first transient be in wrap-around.
   InsertZeros(kLongBuffSize - kWidthThreshold / 2, circular_buffer.get());
 
@@ -89,9 +90,9 @@ TEST(AgcCircularBufferTest, TransientsRemoval) {
   }
 }
 
-TEST(AgcCircularBufferTest, TransientDetection) {
-  rtc::scoped_ptr<AgcCircularBuffer> circular_buffer(
-      AgcCircularBuffer::Create(kLongBuffSize));
+TEST(VadCircularBufferTest, TransientDetection) {
+  rtc::scoped_ptr<VadCircularBuffer> circular_buffer(
+      VadCircularBuffer::Create(kLongBuffSize));
   // Let the first transient be in wrap-around.
   int num_insertion = kLongBuffSize - kWidthThreshold / 2;
   InsertZeros(num_insertion, circular_buffer.get());
@@ -104,8 +105,8 @@ TEST(AgcCircularBufferTest, TransientDetection) {
   double mean_val = circular_buffer->Mean();
   EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
   circular_buffer->Insert(0);
-  EXPECT_EQ(0, circular_buffer->RemoveTransient(kWidthThreshold,
-                                                kValThreshold));
+  EXPECT_EQ(0,
+            circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold));
   mean_val = circular_buffer->Mean();
   EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
 
@@ -114,8 +115,8 @@ TEST(AgcCircularBufferTest, TransientDetection) {
   num_insertion = 3;
   Insert(push_val, num_insertion, circular_buffer.get());
   circular_buffer->Insert(0);
-  EXPECT_EQ(0, circular_buffer->RemoveTransient(kWidthThreshold,
-                                                kValThreshold));
+  EXPECT_EQ(0,
+            circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold));
   mean_val = circular_buffer->Mean();
   EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
 
@@ -123,8 +124,8 @@ TEST(AgcCircularBufferTest, TransientDetection) {
   // it shouldn't be considered transient.
   Insert(push_val, num_insertion, circular_buffer.get());
   num_non_zero_elements += num_insertion;
-  EXPECT_EQ(0, circular_buffer->RemoveTransient(kWidthThreshold,
-                                                kValThreshold));
+  EXPECT_EQ(0,
+            circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold));
   mean_val = circular_buffer->Mean();
   EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
new file mode 100644
index 0000000000..fc9d103918
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace {
+
+const size_t kMaxLength = 320;
+const size_t kNumChannels = 1;
+
+const double kDefaultVoiceValue = 1.0;
+const double kNeutralProbability = 0.5;
+const double kLowProbability = 0.01;
+
+}  // namespace
+
+VoiceActivityDetector::VoiceActivityDetector()
+    : last_voice_probability_(kDefaultVoiceValue),
+      standalone_vad_(StandaloneVad::Create()) {
+}
+
+// Because ISAC has a different chunk length, it updates
+// |chunkwise_voice_probabilities_| and |chunkwise_rms_| when there is new data.
+// Otherwise it clears them.
+void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
+                                         size_t length,
+                                         int sample_rate_hz) {
+  RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100);
+  RTC_DCHECK_LE(length, kMaxLength);
+  // Resample to the required rate.
+  const int16_t* resampled_ptr = audio;
+  if (sample_rate_hz != kSampleRateHz) {
+    RTC_CHECK_EQ(
+        resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
+        0);
+    resampler_.Push(audio, length, resampled_, kLength10Ms, length);
+    resampled_ptr = resampled_;
+  }
+  RTC_DCHECK_EQ(length, kLength10Ms);
+
+  // Each chunk needs to be passed into |standalone_vad_|, because internally it
+  // buffers the audio and processes it all at once when GetActivity() is
+  // called.
+  RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
+
+  audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
+
+  chunkwise_voice_probabilities_.resize(features_.num_frames);
+  chunkwise_rms_.resize(features_.num_frames);
+  std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
+            chunkwise_rms_.begin());
+  if (features_.num_frames > 0) {
+    if (features_.silence) {
+      // The other features are invalid, so set the voice probabilities to an
+      // arbitrary low value.
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kLowProbability);
+    } else {
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kNeutralProbability);
+      RTC_CHECK_GE(
+          standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
+                                       chunkwise_voice_probabilities_.size()),
+          0);
+      RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
+                       features_, &chunkwise_voice_probabilities_[0]),
+                   0);
+    }
+    last_voice_probability_ = chunkwise_voice_probabilities_.back();
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector.h
new file mode 100644
index 0000000000..e2dcf022a9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
+
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
+#include "webrtc/modules/audio_processing/vad/pitch_based_vad.h"
+#include "webrtc/modules/audio_processing/vad/standalone_vad.h"
+
+namespace webrtc {
+
+// A Voice Activity Detector (VAD) that combines the voice probability from the
+// StandaloneVad and PitchBasedVad to get a more robust estimation.
+class VoiceActivityDetector {
+ public:
+  VoiceActivityDetector();
+
+  // Processes each audio chunk and estimates the voice probability. The maximum
+  // supported sample rate is 32kHz.
+  // TODO(aluebs): Change |length| to size_t.
+  void ProcessChunk(const int16_t* audio, size_t length, int sample_rate_hz);
+
+  // Returns a vector of voice probabilities for each chunk. It can be empty for
+  // some chunks, but it catches up afterwards returning multiple values at
+  // once.
+  const std::vector<double>& chunkwise_voice_probabilities() const {
+    return chunkwise_voice_probabilities_;
+  }
+
+  // Returns a vector of RMS values for each chunk. It has the same length as
+  // chunkwise_voice_probabilities().
+  const std::vector<double>& chunkwise_rms() const { return chunkwise_rms_; }
+
+  // Returns the last voice probability, regardless of the internal
+  // implementation, although it has a few chunks of delay.
+  float last_voice_probability() const { return last_voice_probability_; }
+
+ private:
+  // TODO(aluebs): Change these to float.
+  std::vector<double> chunkwise_voice_probabilities_;
+  std::vector<double> chunkwise_rms_;
+
+  float last_voice_probability_;
+
+  Resampler resampler_;
+  VadAudioProc audio_processing_;
+
+  rtc::scoped_ptr<StandaloneVad> standalone_vad_;
+  PitchBasedVad pitch_based_vad_;
+
+  int16_t resampled_[kLength10Ms];
+  AudioFeatures features_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector_unittest.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector_unittest.cc
new file mode 100644
index 0000000000..f4ee17760e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_activity_detector_unittest.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace {
+
+const int kStartTimeSec = 16;
+const float kMeanSpeechProbability = 0.3f;
+const float kMaxNoiseProbability = 0.1f;
+const size_t kNumChunks = 300u;
+const size_t kNumChunksPerIsacBlock = 3;
+
+void GenerateNoise(std::vector<int16_t>* data) {
+  for (size_t i = 0; i < data->size(); ++i) {
+    // std::rand returns between 0 and RAND_MAX, but this will work because it
+    // wraps into some random place.
+    (*data)[i] = std::rand();
+  }
+}
+
+}  // namespace
+
+TEST(VoiceActivityDetectorTest, ConstructorSetsDefaultValues) {
+  const float kDefaultVoiceValue = 1.f;
+
+  VoiceActivityDetector vad;
+
+  std::vector<double> p = vad.chunkwise_voice_probabilities();
+  std::vector<double> rms = vad.chunkwise_rms();
+
+  EXPECT_EQ(p.size(), 0u);
+  EXPECT_EQ(rms.size(), 0u);
+
+  EXPECT_FLOAT_EQ(vad.last_voice_probability(), kDefaultVoiceValue);
+}
+
+TEST(VoiceActivityDetectorTest, Speech16kHzHasHighVoiceProbabilities) {
+  const int kSampleRateHz = 16000;
+  const int kLength10Ms = kSampleRateHz / 100;
+
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(kLength10Ms);
+  float mean_probability = 0.f;
+
+  FILE* pcm_file =
+      fopen(test::ResourcePath("audio_processing/transient/audio16kHz", "pcm")
+                .c_str(),
+            "rb");
+  ASSERT_TRUE(pcm_file != nullptr);
+  // The silences in the file are skipped to get a more robust voice probability
+  // for speech.
+  ASSERT_EQ(fseek(pcm_file, kStartTimeSec * kSampleRateHz * sizeof(data[0]),
+                  SEEK_SET),
+            0);
+
+  size_t num_chunks = 0;
+  while (fread(&data[0], sizeof(data[0]), data.size(), pcm_file) ==
+         data.size()) {
+    vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
+
+    mean_probability += vad.last_voice_probability();
+
+    ++num_chunks;
+  }
+
+  mean_probability /= num_chunks;
+
+  EXPECT_GT(mean_probability, kMeanSpeechProbability);
+}
+
+TEST(VoiceActivityDetectorTest, Speech32kHzHasHighVoiceProbabilities) {
+  const int kSampleRateHz = 32000;
+  const int kLength10Ms = kSampleRateHz / 100;
+
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(kLength10Ms);
+  float mean_probability = 0.f;
+
+  FILE* pcm_file =
+      fopen(test::ResourcePath("audio_processing/transient/audio32kHz", "pcm")
+                .c_str(),
+            "rb");
+  ASSERT_TRUE(pcm_file != nullptr);
+  // The silences in the file are skipped to get a more robust voice probability
+  // for speech.
+  ASSERT_EQ(fseek(pcm_file, kStartTimeSec * kSampleRateHz * sizeof(data[0]),
+                  SEEK_SET),
+            0);
+
+  size_t num_chunks = 0;
+  while (fread(&data[0], sizeof(data[0]), data.size(), pcm_file) ==
+         data.size()) {
+    vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
+
+    mean_probability += vad.last_voice_probability();
+
+    ++num_chunks;
+  }
+
+  mean_probability /= num_chunks;
+
+  EXPECT_GT(mean_probability, kMeanSpeechProbability);
+}
+
+TEST(VoiceActivityDetectorTest, Noise16kHzHasLowVoiceProbabilities) {
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(kLength10Ms);
+  float max_probability = 0.f;
+
+  std::srand(42);
+
+  for (size_t i = 0; i < kNumChunks; ++i) {
+    GenerateNoise(&data);
+
+    vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
+
+    // Before the |vad has enough data to process an ISAC block it will return
+    // the default value, 1.f, which would ruin the |max_probability| value.
+    if (i > kNumChunksPerIsacBlock) {
+      max_probability = std::max(max_probability, vad.last_voice_probability());
+    }
+  }
+
+  EXPECT_LT(max_probability, kMaxNoiseProbability);
+}
+
+TEST(VoiceActivityDetectorTest, Noise32kHzHasLowVoiceProbabilities) {
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(2 * kLength10Ms);
+  float max_probability = 0.f;
+
+  std::srand(42);
+
+  for (size_t i = 0; i < kNumChunks; ++i) {
+    GenerateNoise(&data);
+
+    vad.ProcessChunk(&data[0], data.size(), 2 * kSampleRateHz);
+
+    // Before the |vad has enough data to process an ISAC block it will return
+    // the default value, 1.f, which would ruin the |max_probability| value.
+    if (i > kNumChunksPerIsacBlock) {
+      max_probability = std::max(max_probability, vad.last_voice_probability());
+    }
+  }
+
+  EXPECT_LT(max_probability, kMaxNoiseProbability);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_gmm_tables.h b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_gmm_tables.h
new file mode 100644
index 0000000000..2f247c3798
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/vad/voice_gmm_tables.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// GMM tables for active segments. Generated by MakeGmmTables.m.
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
+
+static const int kVoiceGmmNumMixtures = 12;
+static const int kVoiceGmmDim = 3;
+
+static const double
+    kVoiceGmmCovarInverse[kVoiceGmmNumMixtures][kVoiceGmmDim][kVoiceGmmDim] = {
+        {{1.83673825579513e+00, -8.09791637570095e-04, 4.60106414365986e-03},
+         {-8.09791637570095e-04, 8.89351738394608e-04, -9.80188953277734e-04},
+         {4.60106414365986e-03, -9.80188953277734e-04, 1.38706060206582e-03}},
+        {{6.76228912850703e+01, -1.98893120119660e-02, -3.53548357253551e-03},
+         {-1.98893120119660e-02, 3.96216858500530e-05, -4.08492938394097e-05},
+         {-3.53548357253551e-03, -4.08492938394097e-05, 9.31864352856416e-04}},
+        {{9.98612435944558e+00, -5.27880954316893e-03, -6.30342541619017e-03},
+         {-5.27880954316893e-03, 4.54359480225226e-05, 6.30804591626044e-05},
+         {-6.30342541619017e-03, 6.30804591626044e-05, 5.36466441382942e-04}},
+        {{3.39917474216349e+01, -1.56213579433191e-03, -4.01459014990225e-02},
+         {-1.56213579433191e-03, 6.40415424897724e-05, 6.20076342427833e-05},
+         {-4.01459014990225e-02, 6.20076342427833e-05, 3.51199070103063e-03}},
+        {{1.34545062271428e+01, -7.94513610147144e-03, -5.34401019341728e-02},
+         {-7.94513610147144e-03, 1.16511820098649e-04, 4.66063702069293e-05},
+         {-5.34401019341728e-02, 4.66063702069293e-05, 2.72354323774163e-03}},
+        {{1.08557844314806e+02, -1.54885805673668e-02, -1.88029692674851e-02},
+         {-1.54885805673668e-02, 1.16404042786406e-04, 6.45579292702802e-06},
+         {-1.88029692674851e-02, 6.45579292702802e-06, 4.32330478391416e-04}},
+        {{8.22940066541450e+01, -1.15903110231303e-02, -4.92166764865343e-02},
+         {-1.15903110231303e-02, 7.42510742165261e-05, 3.73007314191290e-06},
+         {-4.92166764865343e-02, 3.73007314191290e-06, 3.64005221593244e-03}},
+        {{2.31133605685660e+00, -7.83261568950254e-04, 7.45744012346313e-04},
+         {-7.83261568950254e-04, 1.29460648214142e-05, -2.22774455093730e-06},
+         {7.45744012346313e-04, -2.22774455093730e-06, 1.05117294093010e-04}},
+        {{3.78767849189611e+02, 1.57759761011568e-03, -2.08551217988774e-02},
+         {1.57759761011568e-03, 4.76066236886865e-05, -2.33977412299324e-05},
+         {-2.08551217988774e-02, -2.33977412299324e-05, 5.24261005371196e-04}},
+        {{6.98580096506135e-01, -5.13850255217378e-04, -4.01124551717056e-04},
+         {-5.13850255217378e-04, 1.40501021984840e-06, -2.09496928716569e-06},
+         {-4.01124551717056e-04, -2.09496928716569e-06, 2.82879357740037e-04}},
+        {{2.62770945162399e+00, -2.31825753241430e-03, -5.30447217466318e-03},
+         {-2.31825753241430e-03, 4.59108572227649e-05, 7.67631886355405e-05},
+         {-5.30447217466318e-03, 7.67631886355405e-05, 2.28521601674098e-03}},
+        {{1.89940391362152e+02, -4.23280856852379e-03, -2.70608873541399e-02},
+         {-4.23280856852379e-03, 6.77547582742563e-05, 2.69154203800467e-05},
+         {-2.70608873541399e-02, 2.69154203800467e-05, 3.88574543373470e-03}}};
+
+static const double kVoiceGmmMean[kVoiceGmmNumMixtures][kVoiceGmmDim] = {
+    {-2.15020241646536e+00, 4.97079062999877e+02, 4.77078119504505e+02},
+    {-8.92097680029190e-01, 5.92064964199921e+02, 1.81045145941059e+02},
+    {-1.29435784144398e+00, 4.98450293410611e+02, 1.71991263804064e+02},
+    {-1.03925228397884e+00, 4.99511274321571e+02, 1.05838336539105e+02},
+    {-1.29229047206129e+00, 4.15026762566707e+02, 1.12861119017125e+02},
+    {-7.88748114599810e-01, 4.48739336688113e+02, 1.89784216956337e+02},
+    {-8.77777402332642e-01, 4.86620285054533e+02, 1.13477708016491e+02},
+    {-2.06465957063057e+00, 6.33385049870607e+02, 2.32758546796149e+02},
+    {-6.98893789231685e-01, 5.93622051503385e+02, 1.92536982473203e+02},
+    {-2.55901217508894e+00, 1.55914919756205e+03, 1.39769980835570e+02},
+    {-1.92070024165837e+00, 4.87983940444185e+02, 1.02745468128289e+02},
+    {-7.29187507662854e-01, 5.22717685022855e+02, 1.16377942283991e+02}};
+
+static const double kVoiceGmmWeights[kVoiceGmmNumMixtures] = {
+    -1.39789694361035e+01,
+    -1.19527720202104e+01,
+    -1.32396317929055e+01,
+    -1.09436815209238e+01,
+    -1.13440027478149e+01,
+    -1.12200721834504e+01,
+    -1.02537324043693e+01,
+    -1.60789861938302e+01,
+    -1.03394494048344e+01,
+    -1.83207938586818e+01,
+    -1.31186044948288e+01,
+    -9.52479998673554e+00};
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.cc b/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.cc
index da51a10ab1..22d218c371 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.cc
@@ -10,61 +10,61 @@
 
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 
-#include <assert.h>
-
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
-
-typedef VadInst Handle;
-
-namespace {
-int MapSetting(VoiceDetection::Likelihood likelihood) {
-  switch (likelihood) {
-    case VoiceDetection::kVeryLowLikelihood:
-      return 3;
-    case VoiceDetection::kLowLikelihood:
-      return 2;
-    case VoiceDetection::kModerateLikelihood:
-      return 1;
-    case VoiceDetection::kHighLikelihood:
-      return 0;
+class VoiceDetectionImpl::Vad {
+ public:
+  Vad() {
+    state_ = WebRtcVad_Create();
+    RTC_CHECK(state_);
+    int error = WebRtcVad_Init(state_);
+    RTC_DCHECK_EQ(0, error);
   }
-  assert(false);
-  return -1;
-}
-}  // namespace
+  ~Vad() {
+    WebRtcVad_Free(state_);
+  }
+  VadInst* state() { return state_; }
+ private:
+  VadInst* state_ = nullptr;
+  RTC_DISALLOW_COPY_AND_ASSIGN(Vad);
+};
 
-VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessing* apm,
-                                       CriticalSectionWrapper* crit)
-  : ProcessingComponent(),
-    apm_(apm),
-    crit_(crit),
-    stream_has_voice_(false),
-    using_external_vad_(false),
-    likelihood_(kLowLikelihood),
-    frame_size_ms_(10),
-    frame_size_samples_(0) {}
+VoiceDetectionImpl::VoiceDetectionImpl(rtc::CriticalSection* crit)
+    : crit_(crit) {
+  RTC_DCHECK(crit);
+}
 
 VoiceDetectionImpl::~VoiceDetectionImpl() {}
 
-int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
+void VoiceDetectionImpl::Initialize(int sample_rate_hz) {
+  rtc::CritScope cs(crit_);
+  sample_rate_hz_ = sample_rate_hz;
+  rtc::scoped_ptr<Vad> new_vad;
+  if (enabled_) {
+    new_vad.reset(new Vad());
   }
+  vad_.swap(new_vad);
+  using_external_vad_ = false;
+  frame_size_samples_ =
+      static_cast<size_t>(frame_size_ms_ * sample_rate_hz_) / 1000;
+  set_likelihood(likelihood_);
+}
 
+void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs(crit_);
+  if (!enabled_) {
+    return;
+  }
   if (using_external_vad_) {
     using_external_vad_ = false;
-    return apm_->kNoError;
+    return;
   }
-  assert(audio->num_frames_per_band() <= 160);
 
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
   // TODO(ajm): concatenate data in frame buffer here.
-
-  int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
-                                  apm_->proc_split_sample_rate_hz(),
+  int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
                                   audio->mixed_low_pass_data(),
                                   frame_size_samples_);
   if (vad_ret == 0) {
@@ -74,110 +74,81 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
     stream_has_voice_ = true;
     audio->set_activity(AudioFrame::kVadActive);
   } else {
-    return apm_->kUnspecifiedError;
+    RTC_NOTREACHED();
   }
-
-  return apm_->kNoError;
 }
 
 int VoiceDetectionImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(crit_);
-  return EnableComponent(enable);
+  rtc::CritScope cs(crit_);
+  if (enabled_ != enable) {
+    enabled_ = enable;
+    Initialize(sample_rate_hz_);
+  }
+  return AudioProcessing::kNoError;
 }
 
 bool VoiceDetectionImpl::is_enabled() const {
-  return is_component_enabled();
+  rtc::CritScope cs(crit_);
+  return enabled_;
 }
 
 int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
+  rtc::CritScope cs(crit_);
   using_external_vad_ = true;
   stream_has_voice_ = has_voice;
-  return apm_->kNoError;
+  return AudioProcessing::kNoError;
 }
 
 bool VoiceDetectionImpl::stream_has_voice() const {
+  rtc::CritScope cs(crit_);
   // TODO(ajm): enable this assertion?
   //assert(using_external_vad_ || is_component_enabled());
   return stream_has_voice_;
 }
 
 int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
-  CriticalSectionScoped crit_scoped(crit_);
-  if (MapSetting(likelihood) == -1) {
-    return apm_->kBadParameterError;
-  }
-
+  rtc::CritScope cs(crit_);
   likelihood_ = likelihood;
-  return Configure();
+  if (enabled_) {
+    int mode = 2;
+    switch (likelihood) {
+      case VoiceDetection::kVeryLowLikelihood:
+        mode = 3;
+        break;
+      case VoiceDetection::kLowLikelihood:
+        mode = 2;
+        break;
+      case VoiceDetection::kModerateLikelihood:
+        mode = 1;
+        break;
+      case VoiceDetection::kHighLikelihood:
+        mode = 0;
+        break;
+      default:
+        RTC_NOTREACHED();
+        break;
+    }
+    int error = WebRtcVad_set_mode(vad_->state(), mode);
+    RTC_DCHECK_EQ(0, error);
+  }
+  return AudioProcessing::kNoError;
 }
 
 VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
+  rtc::CritScope cs(crit_);
   return likelihood_;
 }
 
 int VoiceDetectionImpl::set_frame_size_ms(int size) {
-  CriticalSectionScoped crit_scoped(crit_);
-  assert(size == 10); // TODO(ajm): remove when supported.
-  if (size != 10 &&
-      size != 20 &&
-      size != 30) {
-    return apm_->kBadParameterError;
-  }
-
+  rtc::CritScope cs(crit_);
+  RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported.
   frame_size_ms_ = size;
-
-  return Initialize();
+  Initialize(sample_rate_hz_);
+  return AudioProcessing::kNoError;
 }
 
 int VoiceDetectionImpl::frame_size_ms() const {
+  rtc::CritScope cs(crit_);
   return frame_size_ms_;
 }
-
-int VoiceDetectionImpl::Initialize() {
-  int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
-  }
-
-  using_external_vad_ = false;
-  frame_size_samples_ = frame_size_ms_ *
-      apm_->proc_split_sample_rate_hz() / 1000;
-  // TODO(ajm): intialize frame buffer here.
-
-  return apm_->kNoError;
-}
-
-void* VoiceDetectionImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcVad_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-void VoiceDetectionImpl::DestroyHandle(void* handle) const {
-  WebRtcVad_Free(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::InitializeHandle(void* handle) const {
-  return WebRtcVad_Init(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::ConfigureHandle(void* handle) const {
-  return WebRtcVad_set_mode(static_cast<Handle*>(handle),
-                            MapSetting(likelihood_));
-}
-
-int VoiceDetectionImpl::num_handles_required() const {
-  return 1;
-}
-
-int VoiceDetectionImpl::GetHandleError(void* handle) const {
-  // The VAD has no get_error() function.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.h b/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.h
index 32f031edf2..0d6d8cf14a 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_processing/voice_detection_impl.h
@@ -11,31 +11,27 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
-class CriticalSectionWrapper;
 
-class VoiceDetectionImpl : public VoiceDetection,
-                           public ProcessingComponent {
+class VoiceDetectionImpl : public VoiceDetection {
  public:
-  VoiceDetectionImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit);
-  virtual ~VoiceDetectionImpl();
+  explicit VoiceDetectionImpl(rtc::CriticalSection* crit);
+  ~VoiceDetectionImpl() override;
 
-  int ProcessCaptureAudio(AudioBuffer* audio);
+  // TODO(peah): Fold into ctor, once public API is removed.
+  void Initialize(int sample_rate_hz);
+  void ProcessCaptureAudio(AudioBuffer* audio);
 
-  // VoiceDetection implementation.
-  bool is_enabled() const override;
-
-  // ProcessingComponent implementation.
-  int Initialize() override;
-
- private:
   // VoiceDetection implementation.
   int Enable(bool enable) override;
+  bool is_enabled() const override;
   int set_stream_has_voice(bool has_voice) override;
   bool stream_has_voice() const override;
   int set_likelihood(Likelihood likelihood) override;
@@ -43,21 +39,18 @@ class VoiceDetectionImpl : public VoiceDetection,
   int set_frame_size_ms(int size) override;
   int frame_size_ms() const override;
 
-  // ProcessingComponent implementation.
-  void* CreateHandle() const override;
-  int InitializeHandle(void* handle) const override;
-  int ConfigureHandle(void* handle) const override;
-  void DestroyHandle(void* handle) const override;
-  int num_handles_required() const override;
-  int GetHandleError(void* handle) const override;
-
-  const AudioProcessing* apm_;
-  CriticalSectionWrapper* crit_;
-  bool stream_has_voice_;
-  bool using_external_vad_;
-  Likelihood likelihood_;
-  int frame_size_ms_;
-  int frame_size_samples_;
+ private:
+  class Vad;
+  rtc::CriticalSection* const crit_;
+  bool enabled_ GUARDED_BY(crit_) = false;
+  bool stream_has_voice_ GUARDED_BY(crit_) = false;
+  bool using_external_vad_ GUARDED_BY(crit_) = false;
+  Likelihood likelihood_ GUARDED_BY(crit_) = kLowLikelihood;
+  int frame_size_ms_ GUARDED_BY(crit_) = 10;
+  size_t frame_size_samples_ GUARDED_BY(crit_) = 0;
+  int sample_rate_hz_ GUARDED_BY(crit_) = 0;
+  rtc::scoped_ptr<Vad> vad_ GUARDED_BY(crit_);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VoiceDetectionImpl);
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/BUILD.gn b/media/webrtc/trunk/webrtc/modules/bitrate_controller/BUILD.gn
index f7f67b899c..5e3741ba93 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/BUILD.gn
@@ -10,21 +10,18 @@ import("../../build/webrtc.gni")
 
 source_set("bitrate_controller") {
   sources = [
-    "bitrate_allocator.cc",
     "bitrate_controller_impl.cc",
     "bitrate_controller_impl.h",
     "include/bitrate_allocator.h",
     "include/bitrate_controller.h",
     "send_side_bandwidth_estimation.cc",
     "send_side_bandwidth_estimation.h",
-    "send_time_history.cc",
-    "send_time_history.h",
   ]
 
   if (is_win) {
     cflags = [
       # TODO(jschuh): Bug 1348: fix this warning.
-      "/wd4267"  # size_t to int truncations
+      "/wd4267",  # size_t to int truncations
     ]
   }
 
@@ -37,5 +34,7 @@ source_set("bitrate_controller") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  deps = [ "../../system_wrappers" ]
+  deps = [
+    "../../system_wrappers",
+  ]
 }
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller.gypi b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller.gypi
index a0c2fc92f6..3d86f2e32a 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller.gypi
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller.gypi
@@ -15,15 +15,11 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'bitrate_allocator.cc',
         'bitrate_controller_impl.cc',
         'bitrate_controller_impl.h',
         'include/bitrate_controller.h',
-        'include/bitrate_allocator.h',
         'send_side_bandwidth_estimation.cc',
         'send_side_bandwidth_estimation.h',
-        'send_time_history.cc',
-        'send_time_history.h',
       ],
       # TODO(jschuh): Bug 1348: fix size_t to int truncations.
       'msvs_disabled_warnings': [ 4267, ],
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
index d54da99bef..f8fd2bb987 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
@@ -14,7 +14,7 @@
 #include <algorithm>
 #include <utility>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -87,7 +87,6 @@ BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
     : clock_(clock),
       observer_(observer),
       last_bitrate_update_ms_(clock_->TimeInMilliseconds()),
-      critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       bandwidth_estimation_(),
       reserved_bitrate_bps_(0),
       last_bitrate_bps_(0),
@@ -107,7 +106,7 @@ RtcpBandwidthObserver* BitrateControllerImpl::CreateRtcpBandwidthObserver() {
 
 void BitrateControllerImpl::SetStartBitrate(int start_bitrate_bps) {
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     bandwidth_estimation_.SetSendBitrate(start_bitrate_bps);
   }
   MaybeTriggerOnNetworkChanged();
@@ -116,7 +115,7 @@ void BitrateControllerImpl::SetStartBitrate(int start_bitrate_bps) {
 void BitrateControllerImpl::SetMinMaxBitrate(int min_bitrate_bps,
                                              int max_bitrate_bps) {
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     bandwidth_estimation_.SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
   }
   MaybeTriggerOnNetworkChanged();
@@ -124,23 +123,29 @@ void BitrateControllerImpl::SetMinMaxBitrate(int min_bitrate_bps,
 
 void BitrateControllerImpl::SetReservedBitrate(uint32_t reserved_bitrate_bps) {
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     reserved_bitrate_bps_ = reserved_bitrate_bps;
   }
   MaybeTriggerOnNetworkChanged();
 }
 
+void BitrateControllerImpl::SetEventLog(RtcEventLog* event_log) {
+  rtc::CritScope cs(&critsect_);
+  bandwidth_estimation_.SetEventLog(event_log);
+}
+
 void BitrateControllerImpl::OnReceivedEstimatedBitrate(uint32_t bitrate) {
   {
-    CriticalSectionScoped cs(critsect_.get());
-    bandwidth_estimation_.UpdateReceiverEstimate(bitrate);
+    rtc::CritScope cs(&critsect_);
+    bandwidth_estimation_.UpdateReceiverEstimate(clock_->TimeInMilliseconds(),
+                                                 bitrate);
   }
   MaybeTriggerOnNetworkChanged();
 }
 
 int64_t BitrateControllerImpl::TimeUntilNextProcess() {
   const int64_t kBitrateControllerUpdateIntervalMs = 25;
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   int64_t time_since_update_ms =
       clock_->TimeInMilliseconds() - last_bitrate_update_ms_;
   return std::max<int64_t>(
@@ -151,7 +156,7 @@ int32_t BitrateControllerImpl::Process() {
   if (TimeUntilNextProcess() > 0)
     return 0;
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     bandwidth_estimation_.UpdateEstimate(clock_->TimeInMilliseconds());
   }
   MaybeTriggerOnNetworkChanged();
@@ -165,7 +170,7 @@ void BitrateControllerImpl::OnReceivedRtcpReceiverReport(
     int number_of_packets,
     int64_t now_ms) {
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     bandwidth_estimation_.UpdateReceiverBlock(fraction_loss, rtt,
                                               number_of_packets, now_ms);
   }
@@ -183,7 +188,7 @@ void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() {
 bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,
                                                  uint8_t* fraction_loss,
                                                  int64_t* rtt) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   int current_bitrate;
   bandwidth_estimation_.CurrentEstimate(&current_bitrate, fraction_loss, rtt);
   *bitrate = current_bitrate;
@@ -205,7 +210,7 @@ bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,
 }
 
 bool BitrateControllerImpl::AvailableBandwidth(uint32_t* bandwidth) const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   int bitrate;
   uint8_t fraction_loss;
   int64_t rtt;
@@ -218,5 +223,4 @@ bool BitrateControllerImpl::AvailableBandwidth(uint32_t* bandwidth) const {
   }
   return false;
 }
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
index 3d38a54f53..b601899631 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
@@ -20,9 +20,9 @@
 #include <list>
 #include <utility>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -41,6 +41,8 @@ class BitrateControllerImpl : public BitrateController {
 
   void SetReservedBitrate(uint32_t reserved_bitrate_bps) override;
 
+  void SetEventLog(RtcEventLog* event_log) override;
+
   int64_t TimeUntilNextProcess() override;
   int32_t Process() override;
 
@@ -64,24 +66,23 @@ class BitrateControllerImpl : public BitrateController {
 
   void OnNetworkChanged(uint32_t bitrate,
                         uint8_t fraction_loss,  // 0 - 255.
-                        int64_t rtt)
-      EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+                        int64_t rtt) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
   // Used by process thread.
   Clock* clock_;
   BitrateObserver* observer_;
   int64_t last_bitrate_update_ms_;
 
-  const rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
-  SendSideBandwidthEstimation bandwidth_estimation_ GUARDED_BY(*critsect_);
-  uint32_t reserved_bitrate_bps_ GUARDED_BY(*critsect_);
+  mutable rtc::CriticalSection critsect_;
+  SendSideBandwidthEstimation bandwidth_estimation_ GUARDED_BY(critsect_);
+  uint32_t reserved_bitrate_bps_ GUARDED_BY(critsect_);
 
-  uint32_t last_bitrate_bps_ GUARDED_BY(*critsect_);
-  uint8_t last_fraction_loss_ GUARDED_BY(*critsect_);
-  int64_t last_rtt_ms_ GUARDED_BY(*critsect_);
-  uint32_t last_reserved_bitrate_bps_ GUARDED_BY(*critsect_);
+  uint32_t last_bitrate_bps_ GUARDED_BY(critsect_);
+  uint8_t last_fraction_loss_ GUARDED_BY(critsect_);
+  int64_t last_rtt_ms_ GUARDED_BY(critsect_);
+  uint32_t last_reserved_bitrate_bps_ GUARDED_BY(critsect_);
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(BitrateControllerImpl);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BitrateControllerImpl);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index 72831c78d6..2b9e589fbd 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 using webrtc::RtcpBandwidthObserver;
 using webrtc::BitrateObserver;
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index 7303d069a4..d1eca8e0fe 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -17,12 +17,14 @@
 
 #include <map>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
+class RtcEventLog;
+struct PacketInfo;
 
 class BitrateObserver {
   // Observer class for bitrate changes announced due to change in bandwidth
@@ -55,6 +57,8 @@ class BitrateController : public Module {
   virtual void SetStartBitrate(int start_bitrate_bps) = 0;
   virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
 
+  virtual void SetEventLog(RtcEventLog* event_log) = 0;
+
   // Gets the available payload bandwidth in bits per second. Note that
   // this bandwidth excludes packet headers.
   virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h b/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
new file mode 100644
index 0000000000..7a7d2e406b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_MOCK_MOCK_BITRATE_CONTROLLER_H_
+#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_MOCK_MOCK_BITRATE_CONTROLLER_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+
+namespace webrtc {
+namespace test {
+
+class MockBitrateObserver : public BitrateObserver {
+ public:
+  MOCK_METHOD3(OnNetworkChanged,
+               void(uint32_t bitrate_bps,
+                    uint8_t fraction_loss,
+                    int64_t rtt_ms));
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_MOCK_MOCK_BITRATE_CONTROLLER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 247361df47..258c4d94de 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -13,9 +13,10 @@
 #include <cmath>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/field_trial.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/call/rtc_event_log.h"
 
 namespace webrtc {
 namespace {
@@ -24,9 +25,9 @@ const int64_t kBweDecreaseIntervalMs = 300;
 const int64_t kStartPhaseMs = 2000;
 const int64_t kBweConverganceTimeMs = 20000;
 const int kLimitNumPackets = 20;
-const int kAvgPacketSizeBytes = 1000;
 const int kDefaultMinBitrateBps = 10000;
 const int kDefaultMaxBitrateBps = 1000000000;
+const int64_t kLowBitrateLogPeriodMs = 10000;
 
 struct UmaRampUpMetric {
   const char* metric_name;
@@ -40,38 +41,17 @@ const UmaRampUpMetric kUmaRampupMetrics[] = {
 const size_t kNumUmaRampupMetrics =
     sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);
 
-// Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
-// The formula in RFC 3448, Section 3.1, is used.
-uint32_t CalcTfrcBps(int64_t rtt, uint8_t loss) {
-  if (rtt == 0 || loss == 0) {
-    // Input variables out of range.
-    return 0;
-  }
-  double R = static_cast<double>(rtt) / 1000;  // RTT in seconds.
-  int b = 1;  // Number of packets acknowledged by a single TCP acknowledgement:
-              // recommended = 1.
-  double t_RTO = 4.0 * R;  // TCP retransmission timeout value in seconds
-                           // recommended = 4*R.
-  double p = static_cast<double>(loss) / 255;  // Packet loss rate in [0, 1).
-  double s = static_cast<double>(kAvgPacketSizeBytes);
-
-  // Calculate send rate in bytes/second.
-  double X =
-      s / (R * std::sqrt(2 * b * p / 3) +
-           (t_RTO * (3 * std::sqrt(3 * b * p / 8) * p * (1 + 32 * p * p))));
-
-  // Convert to bits/second.
-  return (static_cast<uint32_t>(X * 8));
-}
 }
 
 SendSideBandwidthEstimation::SendSideBandwidthEstimation()
-    : accumulate_lost_packets_Q8_(0),
-      accumulate_expected_packets_(0),
+    : lost_packets_since_last_loss_update_Q8_(0),
+      expected_packets_since_last_loss_update_(0),
       bitrate_(0),
       min_bitrate_configured_(kDefaultMinBitrateBps),
       max_bitrate_configured_(kDefaultMaxBitrateBps),
-      time_last_receiver_block_ms_(0),
+      last_low_bitrate_log_ms_(-1),
+      has_decreased_since_last_fraction_loss_(false),
+      time_last_receiver_block_ms_(-1),
       last_fraction_loss_(0),
       last_round_trip_time_ms_(0),
       bwe_incoming_(0),
@@ -80,13 +60,13 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
       initially_lost_packets_(0),
       bitrate_at_2_seconds_kbps_(0),
       uma_update_state_(kNoUpdate),
-      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false) {
-}
+      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
+      event_log_(nullptr) {}
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
 void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
-  DCHECK_GT(bitrate, 0);
+  RTC_DCHECK_GT(bitrate, 0);
   bitrate_ = bitrate;
 
   // Clear last sent bitrate history so the new value can be used directly
@@ -96,7 +76,7 @@ void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
 
 void SendSideBandwidthEstimation::SetMinMaxBitrate(int min_bitrate,
                                                    int max_bitrate) {
-  DCHECK_GE(min_bitrate, 0);
+  RTC_DCHECK_GE(min_bitrate, 0);
   min_bitrate_configured_ = std::max(min_bitrate, kDefaultMinBitrateBps);
   if (max_bitrate > 0) {
     max_bitrate_configured_ =
@@ -118,9 +98,10 @@ void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate,
   *rtt = last_round_trip_time_ms_;
 }
 
-void SendSideBandwidthEstimation::UpdateReceiverEstimate(uint32_t bandwidth) {
+void SendSideBandwidthEstimation::UpdateReceiverEstimate(
+    int64_t now_ms, uint32_t bandwidth) {
   bwe_incoming_ = bandwidth;
-  bitrate_ = CapBitrateToThresholds(bitrate_);
+  bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
 }
 
 void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
@@ -138,21 +119,20 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
     // Calculate number of lost packets.
     const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
     // Accumulate reports.
-    accumulate_lost_packets_Q8_ += num_lost_packets_Q8;
-    accumulate_expected_packets_ += number_of_packets;
+    lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8;
+    expected_packets_since_last_loss_update_ += number_of_packets;
 
-    // Report loss if the total report is based on sufficiently many packets.
-    if (accumulate_expected_packets_ >= kLimitNumPackets) {
-      last_fraction_loss_ =
-          accumulate_lost_packets_Q8_ / accumulate_expected_packets_;
-
-      // Reset accumulators.
-      accumulate_lost_packets_Q8_ = 0;
-      accumulate_expected_packets_ = 0;
-    } else {
-      // Early return without updating estimate.
+    // Don't generate a loss rate until it can be based on enough packets.
+    if (expected_packets_since_last_loss_update_ < kLimitNumPackets)
       return;
-    }
+
+    has_decreased_since_last_fraction_loss_ = false;
+    last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ /
+                          expected_packets_since_last_loss_update_;
+
+    // Reset accumulators.
+    lost_packets_since_last_loss_update_Q8_ = 0;
+    expected_packets_since_last_loss_update_ = 0;
   }
   time_last_receiver_block_ms_ = now_ms;
   UpdateEstimate(now_ms);
@@ -166,8 +146,8 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
   for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
     if (!rampup_uma_stats_updated_[i] &&
         bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
-      RTC_HISTOGRAM_COUNTS_100000(kUmaRampupMetrics[i].metric_name,
-                                  now_ms - first_report_time_ms_);
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(kUmaRampupMetrics[i].metric_name,
+                                         now_ms - first_report_time_ms_);
       rampup_uma_stats_updated_[i] = true;
     }
   }
@@ -176,22 +156,19 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
   } else if (uma_update_state_ == kNoUpdate) {
     uma_update_state_ = kFirstDone;
     bitrate_at_2_seconds_kbps_ = bitrate_kbps;
-    RTC_HISTOGRAM_COUNTS(
-        "WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
-    RTC_HISTOGRAM_COUNTS(
-        "WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0, 2000, 50);
-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
-                         bitrate_at_2_seconds_kbps_,
-                         0,
-                         2000,
-                         50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitiallyLostPackets",
+                                initially_lost_packets_, 0, 100, 50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialRtt", static_cast<int>(rtt),
+                                0, 2000, 50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialBandwidthEstimate",
+                                bitrate_at_2_seconds_kbps_, 0, 2000, 50);
   } else if (uma_update_state_ == kFirstDone &&
              now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
     uma_update_state_ = kDone;
     int bitrate_diff_kbps =
         std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
-    RTC_HISTOGRAM_COUNTS(
-        "WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
+    RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialVsConvergedDiff",
+                                bitrate_diff_kbps, 0, 2000, 50);
   }
 }
 
@@ -200,14 +177,16 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
   // packet loss reported, to allow startup bitrate probing.
   if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms) &&
       bwe_incoming_ > bitrate_) {
-    bitrate_ = CapBitrateToThresholds(bwe_incoming_);
+    bitrate_ = CapBitrateToThresholds(now_ms, bwe_incoming_);
     min_bitrate_history_.clear();
     min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
     return;
   }
   UpdateMinHistory(now_ms);
   // Only start updating bitrate when receiving receiver blocks.
-  if (time_last_receiver_block_ms_ != 0) {
+  // TODO(pbos): Handle the case when no receiver report is received for a very
+  // long time.
+  if (time_last_receiver_block_ms_ != -1) {
     if (last_fraction_loss_ <= 5) {
       // Loss < 2%: Increase rate by 8% of the min bitrate in the last
       // kBweIncreaseIntervalMs.
@@ -226,14 +205,19 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
       // rates).
       bitrate_ += 1000;
 
+      if (event_log_) {
+        event_log_->LogBwePacketLossEvent(
+            bitrate_, last_fraction_loss_,
+            expected_packets_since_last_loss_update_);
+      }
     } else if (last_fraction_loss_ <= 26) {
       // Loss between 2% - 10%: Do nothing.
-
     } else {
       // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs +
       // rtt.
-      if ((now_ms - time_last_decrease_ms_) >=
-          (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {
+      if (!has_decreased_since_last_fraction_loss_ &&
+          (now_ms - time_last_decrease_ms_) >=
+              (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {
         time_last_decrease_ms_ = now_ms;
 
         // Reduce rate:
@@ -242,16 +226,16 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
         bitrate_ = static_cast<uint32_t>(
             (bitrate_ * static_cast<double>(512 - last_fraction_loss_)) /
             512.0);
-
-        // Calculate what rate TFRC would apply in this situation and to not
-        // reduce further than it.
-        bitrate_ = std::max(
-            bitrate_,
-            CalcTfrcBps(last_round_trip_time_ms_, last_fraction_loss_));
+        has_decreased_since_last_fraction_loss_ = true;
+      }
+      if (event_log_) {
+        event_log_->LogBwePacketLossEvent(
+            bitrate_, last_fraction_loss_,
+            expected_packets_since_last_loss_update_);
       }
     }
   }
-  bitrate_ = CapBitrateToThresholds(bitrate_);
+  bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
 }
 
 bool SendSideBandwidthEstimation::IsInStartPhase(int64_t now_ms) const {
@@ -279,7 +263,8 @@ void SendSideBandwidthEstimation::UpdateMinHistory(int64_t now_ms) {
   min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
 }
 
-uint32_t SendSideBandwidthEstimation::CapBitrateToThresholds(uint32_t bitrate) {
+uint32_t SendSideBandwidthEstimation::CapBitrateToThresholds(
+    int64_t now_ms, uint32_t bitrate) {
   if (bwe_incoming_ > 0 && bitrate > bwe_incoming_) {
     bitrate = bwe_incoming_;
   }
@@ -287,11 +272,20 @@ uint32_t SendSideBandwidthEstimation::CapBitrateToThresholds(uint32_t bitrate) {
     bitrate = max_bitrate_configured_;
   }
   if (bitrate < min_bitrate_configured_) {
-    LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate / 1000
-                    << " kbps is below configured min bitrate "
-                    << min_bitrate_configured_ / 1000 << " kbps.";
+    if (last_low_bitrate_log_ms_ == -1 ||
+        now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) {
+      LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate / 1000
+                      << " kbps is below configured min bitrate "
+                      << min_bitrate_configured_ / 1000 << " kbps.";
+      last_low_bitrate_log_ms_ = now_ms;
+    }
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
+
+void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
+  event_log_ = event_log;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index fb8962ad50..7ffb42cb54 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -15,10 +15,13 @@
 
 #include <deque>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
+
+class RtcEventLog;
+
 class SendSideBandwidthEstimation {
  public:
   SendSideBandwidthEstimation();
@@ -30,7 +33,7 @@ class SendSideBandwidthEstimation {
   void UpdateEstimate(int64_t now_ms);
 
   // Call when we receive a RTCP message with TMMBR or REMB.
-  void UpdateReceiverEstimate(uint32_t bandwidth);
+  void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
 
   // Call when we receive a RTCP message with a ReceiveBlock.
   void UpdateReceiverBlock(uint8_t fraction_loss,
@@ -42,6 +45,8 @@ class SendSideBandwidthEstimation {
   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
   int GetMinBitrate() const;
 
+  void SetEventLog(RtcEventLog* event_log);
+
  private:
   enum UmaState { kNoUpdate, kFirstDone, kDone };
 
@@ -51,7 +56,7 @@ class SendSideBandwidthEstimation {
 
   // Returns the input bitrate capped to the thresholds defined by the max,
   // min and incoming bandwidth.
-  uint32_t CapBitrateToThresholds(uint32_t bitrate);
+  uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate);
 
   // Updates history of min bitrates.
   // After this method returns min_bitrate_history_.front().second contains the
@@ -61,13 +66,15 @@ class SendSideBandwidthEstimation {
   std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
 
   // incoming filters
-  int accumulate_lost_packets_Q8_;
-  int accumulate_expected_packets_;
+  int lost_packets_since_last_loss_update_Q8_;
+  int expected_packets_since_last_loss_update_;
 
   uint32_t bitrate_;
   uint32_t min_bitrate_configured_;
   uint32_t max_bitrate_configured_;
+  int64_t last_low_bitrate_log_ms_;
 
+  bool has_decreased_since_last_fraction_loss_;
   int64_t time_last_receiver_block_ms_;
   uint8_t last_fraction_loss_;
   int64_t last_round_trip_time_ms_;
@@ -79,6 +86,7 @@ class SendSideBandwidthEstimation {
   int bitrate_at_2_seconds_kbps_;
   UmaState uma_update_state_;
   std::vector<bool> rampup_uma_stats_updated_;
+  RtcEventLog* event_log_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
index ab052b5244..0424d22bd6 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
@@ -28,7 +28,7 @@ TEST(SendSideBweTest, InitialRembWithProbing) {
   bwe.UpdateReceiverBlock(0, 50, 1, now_ms);
 
   // Initial REMB applies immediately.
-  bwe.UpdateReceiverEstimate(kRembBps);
+  bwe.UpdateReceiverEstimate(now_ms, kRembBps);
   bwe.UpdateEstimate(now_ms);
   int bitrate;
   uint8_t fraction_loss;
@@ -38,10 +38,61 @@ TEST(SendSideBweTest, InitialRembWithProbing) {
 
   // Second REMB doesn't apply immediately.
   now_ms += 2001;
-  bwe.UpdateReceiverEstimate(kSecondRembBps);
+  bwe.UpdateReceiverEstimate(now_ms, kSecondRembBps);
   bwe.UpdateEstimate(now_ms);
   bitrate = 0;
   bwe.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
   EXPECT_EQ(kRembBps, bitrate);
 }
+
+TEST(SendSideBweTest, DoesntReapplyBitrateDecreaseWithoutFollowingRemb) {
+  SendSideBandwidthEstimation bwe;
+  static const int kMinBitrateBps = 100000;
+  static const int kInitialBitrateBps = 1000000;
+  bwe.SetMinMaxBitrate(kMinBitrateBps, 1500000);
+  bwe.SetSendBitrate(kInitialBitrateBps);
+
+  static const uint8_t kFractionLoss = 128;
+  static const int64_t kRttMs = 50;
+
+  int64_t now_ms = 0;
+  int bitrate_bps;
+  uint8_t fraction_loss;
+  int64_t rtt_ms;
+  bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms);
+  EXPECT_EQ(kInitialBitrateBps, bitrate_bps);
+  EXPECT_EQ(0, fraction_loss);
+  EXPECT_EQ(0, rtt_ms);
+
+  // Signal heavy loss to go down in bitrate.
+  bwe.UpdateReceiverBlock(kFractionLoss, kRttMs, 100, now_ms);
+  // Trigger an update 2 seconds later to not be rate limited.
+  now_ms += 2000;
+  bwe.UpdateEstimate(now_ms);
+
+  bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms);
+  EXPECT_LT(bitrate_bps, kInitialBitrateBps);
+  // Verify that the obtained bitrate isn't hitting the min bitrate, or this
+  // test doesn't make sense. If this ever happens, update the thresholds or
+  // loss rates so that it doesn't hit min bitrate after one bitrate update.
+  EXPECT_GT(bitrate_bps, kMinBitrateBps);
+  EXPECT_EQ(kFractionLoss, fraction_loss);
+  EXPECT_EQ(kRttMs, rtt_ms);
+
+  // Triggering an update shouldn't apply further downgrade nor upgrade since
+  // there's no intermediate receiver block received indicating whether this is
+  // currently good or not.
+  int last_bitrate_bps = bitrate_bps;
+  // Trigger an update 2 seconds later to not be rate limited (but it still
+  // shouldn't update).
+  now_ms += 2000;
+  bwe.UpdateEstimate(now_ms);
+  bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms);
+
+  EXPECT_EQ(last_bitrate_bps, bitrate_bps);
+  // The old loss rate should still be applied though.
+  EXPECT_EQ(kFractionLoss, fraction_loss);
+  EXPECT_EQ(kRttMs, rtt_ms);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history.h b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history.h
deleted file mode 100644
index 8835856353..0000000000
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
-#define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
-
-#include <map>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/basictypes.h"
-
-namespace webrtc {
-
-class SendTimeHistory {
- public:
-  explicit SendTimeHistory(int64_t packet_age_limit);
-  virtual ~SendTimeHistory();
-
-  void AddAndRemoveOldSendTimes(uint16_t sequence_number, int64_t timestamp);
-  bool GetSendTime(uint16_t sequence_number, int64_t* timestamp, bool remove);
-  void Clear();
-
- private:
-  void EraseOld(int64_t limit);
-  void UpdateOldestSequenceNumber();
-
-  const int64_t packet_age_limit_;
-  uint16_t oldest_sequence_number_;  // Oldest may not be lowest.
-  std::map<uint16_t, int64_t> history_;
-
-  DISALLOW_COPY_AND_ASSIGN(SendTimeHistory);
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history_unittest.cc b/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history_unittest.cc
deleted file mode 100644
index fc7099dbdd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history_unittest.cc
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <limits>
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/bitrate_controller/send_time_history.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-
-namespace webrtc {
-
-static const int kDefaultHistoryLengthMs = 1000;
-
-class SendTimeHistoryTest : public ::testing::Test {
- protected:
-  SendTimeHistoryTest() : history_(kDefaultHistoryLengthMs), clock_(0) {}
-  ~SendTimeHistoryTest() {}
-
-  virtual void SetUp() {}
-
-  virtual void TearDown() {}
-
-  SendTimeHistory history_;
-  webrtc::SimulatedClock clock_;
-};
-
-TEST_F(SendTimeHistoryTest, AddRemoveOne) {
-  const uint16_t kSeqNo = 1;
-  const int64_t kTimestamp = 2;
-  history_.AddAndRemoveOldSendTimes(kSeqNo, kTimestamp);
-
-  int64_t time = 0;
-  EXPECT_TRUE(history_.GetSendTime(kSeqNo, &time, false));
-  EXPECT_EQ(kTimestamp, time);
-
-  time = 0;
-  EXPECT_TRUE(history_.GetSendTime(kSeqNo, &time, true));
-  EXPECT_EQ(kTimestamp, time);
-
-  time = 0;
-  EXPECT_FALSE(history_.GetSendTime(kSeqNo, &time, true));
-}
-
-TEST_F(SendTimeHistoryTest, AddThenRemoveOutOfOrder) {
-  struct Timestamp {
-    Timestamp(uint16_t sequence_number, int64_t timestamp)
-        : sequence_number(sequence_number), timestamp(timestamp) {}
-    uint16_t sequence_number;
-    int64_t timestamp;
-  };
-  std::vector<Timestamp> timestamps;
-  const size_t num_items = 100;
-  for (size_t i = 0; i < num_items; ++i) {
-    timestamps.push_back(
-        Timestamp(static_cast<uint16_t>(i), static_cast<int64_t>(i)));
-  }
-  std::vector<Timestamp> randomized_timestamps = timestamps;
-  std::random_shuffle(randomized_timestamps.begin(),
-                      randomized_timestamps.end());
-  for (size_t i = 0; i < num_items; ++i) {
-    history_.AddAndRemoveOldSendTimes(timestamps[i].sequence_number,
-                                      timestamps[i].timestamp);
-  }
-  for (size_t i = 0; i < num_items; ++i) {
-    int64_t timestamp;
-    EXPECT_TRUE(history_.GetSendTime(randomized_timestamps[i].sequence_number,
-                                     &timestamp, false));
-    EXPECT_EQ(randomized_timestamps[i].timestamp, timestamp);
-    EXPECT_TRUE(history_.GetSendTime(randomized_timestamps[i].sequence_number,
-                                     &timestamp, true));
-  }
-  for (size_t i = 0; i < num_items; ++i) {
-    int64_t timestamp;
-    EXPECT_FALSE(
-        history_.GetSendTime(timestamps[i].sequence_number, &timestamp, false));
-  }
-}
-
-TEST_F(SendTimeHistoryTest, HistorySize) {
-  const int kItems = kDefaultHistoryLengthMs / 100;
-  for (int i = 0; i < kItems; ++i) {
-    history_.AddAndRemoveOldSendTimes(i, i * 100);
-  }
-  int64_t timestamp;
-  for (int i = 0; i < kItems; ++i) {
-    EXPECT_TRUE(history_.GetSendTime(i, &timestamp, false));
-    EXPECT_EQ(i * 100, timestamp);
-  }
-  history_.AddAndRemoveOldSendTimes(kItems, kItems * 100);
-  EXPECT_FALSE(history_.GetSendTime(0, &timestamp, false));
-  for (int i = 1; i < (kItems + 1); ++i) {
-    EXPECT_TRUE(history_.GetSendTime(i, &timestamp, false));
-    EXPECT_EQ(i * 100, timestamp);
-  }
-}
-
-TEST_F(SendTimeHistoryTest, HistorySizeWithWraparound) {
-  const int kMaxSeqNo = std::numeric_limits<uint16_t>::max();
-  history_.AddAndRemoveOldSendTimes(kMaxSeqNo - 2, 0);
-  history_.AddAndRemoveOldSendTimes(kMaxSeqNo - 1, 100);
-  history_.AddAndRemoveOldSendTimes(kMaxSeqNo, 200);
-  history_.AddAndRemoveOldSendTimes(0, 1000);
-  int64_t timestamp;
-  EXPECT_FALSE(history_.GetSendTime(kMaxSeqNo - 2, &timestamp, false));
-  EXPECT_TRUE(history_.GetSendTime(kMaxSeqNo - 1, &timestamp, false));
-  EXPECT_TRUE(history_.GetSendTime(kMaxSeqNo, &timestamp, false));
-  EXPECT_TRUE(history_.GetSendTime(0, &timestamp, false));
-
-  // Create a gap (kMaxSeqNo - 1) -> 0.
-  EXPECT_TRUE(history_.GetSendTime(kMaxSeqNo, &timestamp, true));
-
-  history_.AddAndRemoveOldSendTimes(1, 1100);
-
-  EXPECT_FALSE(history_.GetSendTime(kMaxSeqNo - 2, &timestamp, false));
-  EXPECT_FALSE(history_.GetSendTime(kMaxSeqNo - 1, &timestamp, false));
-  EXPECT_FALSE(history_.GetSendTime(kMaxSeqNo, &timestamp, false));
-  EXPECT_TRUE(history_.GetSendTime(0, &timestamp, false));
-  EXPECT_TRUE(history_.GetSendTime(1, &timestamp, false));
-}
-
-TEST_F(SendTimeHistoryTest, InterlievedGetAndRemove) {
-  const uint16_t kSeqNo = 1;
-  const int64_t kTimestamp = 2;
-
-  history_.AddAndRemoveOldSendTimes(kSeqNo, kTimestamp);
-  history_.AddAndRemoveOldSendTimes(kSeqNo + 1, kTimestamp + 1);
-
-  int64_t time = 0;
-  EXPECT_TRUE(history_.GetSendTime(kSeqNo, &time, true));
-  EXPECT_EQ(kTimestamp, time);
-
-  history_.AddAndRemoveOldSendTimes(kSeqNo + 2, kTimestamp + 2);
-
-  EXPECT_TRUE(history_.GetSendTime(kSeqNo + 1, &time, true));
-  EXPECT_EQ(kTimestamp + 1, time);
-  EXPECT_TRUE(history_.GetSendTime(kSeqNo + 2, &time, true));
-  EXPECT_EQ(kTimestamp + 2, time);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/BUILD.gn b/media/webrtc/trunk/webrtc/modules/desktop_capture/BUILD.gn
index c23aa0312d..aa33993192 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/BUILD.gn
@@ -10,7 +10,21 @@ import("//build/config/ui.gni")
 import("../../build/webrtc.gni")
 
 use_desktop_capture_differ_sse2 =
-    (!is_ios && (current_cpu == "x86" || current_cpu == "x64"))
+    !is_ios && (current_cpu == "x86" || current_cpu == "x64")
+
+source_set("primitives") {
+  sources = [
+    "desktop_capture_types.h",
+    "desktop_frame.cc",
+    "desktop_frame.h",
+    "desktop_geometry.cc",
+    "desktop_geometry.h",
+    "desktop_region.cc",
+    "desktop_region.h",
+  ]
+
+  public_configs = [ "../..:common_inherited_config" ]
+}
 
 source_set("desktop_capture") {
   sources = [
@@ -21,27 +35,20 @@ source_set("desktop_capture") {
     "cropping_window_capturer_win.cc",
     "desktop_and_cursor_composer.cc",
     "desktop_and_cursor_composer.h",
-    "desktop_capture_types.h",
+    "desktop_capture_options.cc",
+    "desktop_capture_options.h",
+    "desktop_capturer.h",
     "desktop_capturer.h",
-    "desktop_frame.cc",
-    "desktop_frame.h",
     "desktop_frame_win.cc",
     "desktop_frame_win.h",
-    "desktop_geometry.cc",
-    "desktop_geometry.h",
-    "desktop_capture_options.h",
-    "desktop_capture_options.cc",
-    "desktop_capturer.h",
-    "desktop_region.cc",
-    "desktop_region.h",
     "differ.cc",
     "differ.h",
     "differ_block.cc",
     "differ_block.h",
     "mac/desktop_configuration.h",
     "mac/desktop_configuration.mm",
-    "mac/desktop_configuration_monitor.h",
     "mac/desktop_configuration_monitor.cc",
+    "mac/desktop_configuration_monitor.h",
     "mac/full_screen_chrome_window_detector.cc",
     "mac/full_screen_chrome_window_detector.h",
     "mac/scoped_pixel_buffer_object.cc",
@@ -72,12 +79,12 @@ source_set("desktop_capture") {
     "win/scoped_gdi_object.h",
     "win/scoped_thread_desktop.cc",
     "win/scoped_thread_desktop.h",
+    "win/screen_capture_utils.cc",
+    "win/screen_capture_utils.h",
     "win/screen_capturer_win_gdi.cc",
     "win/screen_capturer_win_gdi.h",
     "win/screen_capturer_win_magnifier.cc",
     "win/screen_capturer_win_magnifier.h",
-    "win/screen_capture_utils.cc",
-    "win/screen_capture_utils.h",
     "win/window_capture_utils.cc",
     "win/window_capture_utils.h",
     "window_capturer.cc",
@@ -91,14 +98,14 @@ source_set("desktop_capture") {
       "mouse_cursor_monitor_x11.cc",
       "screen_capturer_x11.cc",
       "window_capturer_x11.cc",
-      "x11/shared_x_display.h",
       "x11/shared_x_display.cc",
+      "x11/shared_x_display.h",
       "x11/x_error_trap.cc",
       "x11/x_error_trap.h",
       "x11/x_server_pixel_buffer.cc",
       "x11/x_server_pixel_buffer.h",
     ]
-    configs += ["//build/config/linux:x11"]
+    configs += [ "//build/config/linux:x11" ]
   }
 
   if (!is_win && !is_mac && !use_x11) {
@@ -118,21 +125,22 @@ source_set("desktop_capture") {
   }
 
   configs += [ "../..:common_config" ]
-  public_configs = [ "../..:common_inherited_config"]
+  public_configs = [ "../..:common_inherited_config" ]
 
-  if (is_clang) {
+  if (is_clang && !is_nacl) {
     # Suppress warnings from Chrome's Clang plugins.
     # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
+    ":primitives",
     "../../base:rtc_base_approved",
     "../../system_wrappers",
   ]
 
   if (use_desktop_capture_differ_sse2) {
-    deps += [":desktop_capture_differ_sse2"]
+    deps += [ ":desktop_capture_differ_sse2" ]
   }
 }
 
@@ -150,7 +158,7 @@ if (use_desktop_capture_differ_sse2) {
     public_configs = [ "../..:common_inherited_config" ]
 
     if (is_posix && !is_mac) {
-      cflags = ["-msse2"]
+      cflags = [ "-msse2" ]
     }
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_mac.mm b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_mac.mm
index 65016cb445..6383e3a2af 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_mac.mm
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_mac.mm
@@ -19,7 +19,8 @@
 #include "webrtc/modules/desktop_capture/app_capturer.h"
 
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/base/constructormagic.h"
 
 namespace webrtc {
 
@@ -44,7 +45,7 @@ class AppCapturerMac : public AppCapturer {
   Callback* callback_;
   ProcessId process_id_;
 
-  DISALLOW_COPY_AND_ASSIGN(AppCapturerMac);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AppCapturerMac);
 };
 
 AppCapturerMac::AppCapturerMac()
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_null.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_null.cc
index 428b6043f1..5a0b1a28f0 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_null.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_null.cc
@@ -36,7 +36,7 @@ public:
 private:
   Callback* callback_;
 
-  DISALLOW_COPY_AND_ASSIGN(AppCapturerNull);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AppCapturerNull);
 };
 
 AppCapturerNull::AppCapturerNull()
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_unittest.cc
index eb8c238bd1..f11133eb19 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_unittest.cc
@@ -13,7 +13,7 @@
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_win.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_win.cc
index 4de4bdc3e8..f3344172aa 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_win.cc
@@ -19,7 +19,7 @@
 #include <cassert>
 
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
@@ -118,7 +118,7 @@ private:
   // WebRTC Window mode
   WindowsCapturerProxy window_capturer_proxy_;
 
-  DISALLOW_COPY_AND_ASSIGN(AppCapturerWin);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AppCapturerWin);
 };
 
 AppCapturerWin::AppCapturerWin(const DesktopCaptureOptions& options)
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_x11.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_x11.cc
index fde8353622..66a85aa757 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_x11.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/app_capturer_x11.cc
@@ -28,9 +28,8 @@
 #include "webrtc/modules/desktop_capture/x11/shared_x_display.h"
 #include "webrtc/modules/desktop_capture/x11/x_error_trap.h"
 #include "webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
 
 namespace webrtc {
 
@@ -111,8 +110,8 @@ private:
   // WebRtc Window mode
   WindowsCapturerProxy window_capturer_proxy_;
 
-  scoped_refptr<SharedXDisplay> x_display_;
-  DISALLOW_COPY_AND_ASSIGN(AppCapturerLinux);
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AppCapturerLinux);
 };
 
 AppCapturerLinux::AppCapturerLinux(const DesktopCaptureOptions& options)
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/cropped_desktop_frame.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/cropped_desktop_frame.cc
index 9ab6fe9c7c..2c709733e1 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/cropped_desktop_frame.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/cropped_desktop_frame.cc
@@ -20,7 +20,7 @@ class CroppedDesktopFrame : public DesktopFrame {
  private:
   rtc::scoped_ptr<DesktopFrame> frame_;
 
-  DISALLOW_COPY_AND_ASSIGN(CroppedDesktopFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(CroppedDesktopFrame);
 };
 
 DesktopFrame*
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer.cc
index c12437ca33..ab620ad2fb 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/desktop_capture/cropping_window_capturer.h"
 
 #include "webrtc/modules/desktop_capture/cropped_desktop_frame.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc
index deffe665ee..fe696eba67 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc
@@ -14,7 +14,7 @@
 #include "webrtc/modules/desktop_capture/win/scoped_gdi_object.h"
 #include "webrtc/modules/desktop_capture/win/screen_capture_utils.h"
 #include "webrtc/modules/desktop_capture/win/window_capture_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -119,10 +119,12 @@ class CroppingWindowCapturerWin : public CroppingWindowCapturer {
   // The region from GetWindowRgn in the desktop coordinate if the region is
   // rectangular, or the rect from GetWindowRect if the region is not set.
   DesktopRect window_region_rect_;
+
+  AeroChecker aero_checker_;
 };
 
 bool CroppingWindowCapturerWin::ShouldUseScreenCapturer() {
-  if (!rtc::IsWindows8OrLater())
+  if (!rtc::IsWindows8OrLater() && aero_checker_.IsAeroEnabled())
     return false;
 
   // Check if the window is a translucent layered window.
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
index 92670f85b1..2e6b380969 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
@@ -67,7 +67,7 @@ class DesktopFrameWithCursor : public DesktopFrame {
   DesktopVector restore_position_;
   rtc::scoped_ptr<DesktopFrame> restore_frame_;
 
-  DISALLOW_COPY_AND_ASSIGN(DesktopFrameWithCursor);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameWithCursor);
 };
 
 DesktopFrameWithCursor::DesktopFrameWithCursor(DesktopFrame* frame,
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
index cb26cc160c..a78756271b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
@@ -56,7 +56,7 @@ class DesktopAndCursorComposer : public DesktopCapturer,
   MouseCursorMonitor::CursorState cursor_state_;
   DesktopVector cursor_position_;
 
-  DISALLOW_COPY_AND_ASSIGN(DesktopAndCursorComposer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopAndCursorComposer);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
index 7fdfa1c4b9..bee9e087f9 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
 #include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
 #include "webrtc/modules/desktop_capture/window_capturer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture.gypi b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture.gypi
index de1732d09e..1b5bccca0a 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture.gypi
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture.gypi
@@ -202,9 +202,12 @@
             "differ_block_sse2.h",
           ],
           'conditions': [
-            [ 'os_posix == 1 and OS != "mac"', {
+            ['os_posix==1', {
               'cflags': [ '-msse2', ],
               'cflags_mozilla': [ '-msse2', ],
+              'xcode_settings': {
+                'OTHER_CFLAGS': [ '-msse2', ],
+              },
             }],
           ],
         },
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture_options.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture_options.h
index 030cb2b777..68bb588445 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture_options.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture_options.h
@@ -11,7 +11,7 @@
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_OPTIONS_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 
 #if defined(USE_X11)
 #include "webrtc/modules/desktop_capture/x11/shared_x_display.h"
@@ -39,7 +39,7 @@ class DesktopCaptureOptions {
 
 #if defined(USE_X11)
   SharedXDisplay* x_display() const { return x_display_; }
-  void set_x_display(scoped_refptr<SharedXDisplay> x_display) {
+  void set_x_display(rtc::scoped_refptr<SharedXDisplay> x_display) {
     x_display_ = x_display;
   }
 #endif
@@ -48,7 +48,8 @@ class DesktopCaptureOptions {
   DesktopConfigurationMonitor* configuration_monitor() const {
     return configuration_monitor_;
   }
-  void set_configuration_monitor(scoped_refptr<DesktopConfigurationMonitor> m) {
+  void set_configuration_monitor(
+      rtc::scoped_refptr<DesktopConfigurationMonitor> m) {
     configuration_monitor_ = m;
   }
 
@@ -56,7 +57,7 @@ class DesktopCaptureOptions {
     return full_screen_window_detector_;
   }
   void set_full_screen_chrome_window_detector(
-      scoped_refptr<FullScreenChromeWindowDetector> detector) {
+      rtc::scoped_refptr<FullScreenChromeWindowDetector> detector) {
     full_screen_window_detector_ = detector;
   }
 #endif
@@ -86,12 +87,13 @@ class DesktopCaptureOptions {
 
  private:
 #if defined(USE_X11)
-  scoped_refptr<SharedXDisplay> x_display_;
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
 #endif
 
 #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-  scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
-  scoped_refptr<FullScreenChromeWindowDetector> full_screen_window_detector_;
+  rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
+  rtc::scoped_refptr<FullScreenChromeWindowDetector>
+      full_screen_window_detector_;
 #endif
 
 #if defined(WEBRTC_WIN)
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame.h
index 29d5076959..49b964630c 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame.h
@@ -91,20 +91,20 @@ class DesktopFrame {
   rtc::scoped_ptr<DesktopRegion> shape_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(DesktopFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrame);
 };
 
 // A DesktopFrame that stores data in the heap.
 class BasicDesktopFrame : public DesktopFrame {
  public:
   explicit BasicDesktopFrame(DesktopSize size);
-  virtual ~BasicDesktopFrame();
+  ~BasicDesktopFrame() override;
 
   // Creates a BasicDesktopFrame that contains copy of |frame|.
   static DesktopFrame* CopyOf(const DesktopFrame& frame);
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(BasicDesktopFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BasicDesktopFrame);
 };
 
 // A DesktopFrame that stores data in shared memory.
@@ -114,10 +114,10 @@ class SharedMemoryDesktopFrame : public DesktopFrame {
   SharedMemoryDesktopFrame(DesktopSize size,
                            int stride,
                            SharedMemory* shared_memory);
-  virtual ~SharedMemoryDesktopFrame();
+  ~SharedMemoryDesktopFrame() override;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(SharedMemoryDesktopFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemoryDesktopFrame);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.cc
index 98bc902c4b..6b97b132d8 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -35,7 +35,7 @@ DesktopFrameWin* DesktopFrameWin::Create(DesktopSize size,
   int bytes_per_row = size.width() * kBytesPerPixel;
 
   // Describe a device independent bitmap (DIB) that is the size of the desktop.
-  BITMAPINFO bmi = {0};
+  BITMAPINFO bmi = {};
   bmi.bmiHeader.biHeight = -size.height();
   bmi.bmiHeader.biWidth = size.width();
   bmi.bmiHeader.biPlanes = 1;
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.h
index 9530fdc89b..15b5883c36 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_frame_win.h
@@ -40,7 +40,7 @@ class DesktopFrameWin : public DesktopFrame {
   HBITMAP bitmap_;
   rtc::scoped_ptr<SharedMemory> owned_shared_memory_;
 
-  DISALLOW_COPY_AND_ASSIGN(DesktopFrameWin);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameWin);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.cc
index 90428199a4..bc9972660a 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.cc
@@ -511,6 +511,8 @@ DesktopRegion::Iterator::Iterator(const DesktopRegion& region)
   }
 }
 
+DesktopRegion::Iterator::~Iterator() {}
+
 bool DesktopRegion::Iterator::IsAtEnd() const {
   return row_ == region_.rows_.end();
 }
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.h
index c4528ae349..c86da56e17 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_region.h
@@ -67,6 +67,7 @@ class DesktopRegion {
   class Iterator {
    public:
     explicit Iterator(const DesktopRegion& target);
+    ~Iterator();
 
     bool IsAtEnd() const;
     void Advance();
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.cc
index 5a347a7dd6..8140e612a1 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.cc
@@ -13,7 +13,7 @@
 #include "string.h"
 
 #include "webrtc/modules/desktop_capture/differ_block.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -28,13 +28,14 @@ Differ::Differ(int width, int height, int bpp, int stride) {
   // One additional row/column is added as a boundary on the right & bottom.
   diff_info_width_ = ((width_ + kBlockSize - 1) / kBlockSize) + 1;
   diff_info_height_ = ((height_ + kBlockSize - 1) / kBlockSize) + 1;
-  diff_info_size_ = diff_info_width_ * diff_info_height_ * sizeof(DiffInfo);
-  diff_info_.reset(new DiffInfo[diff_info_size_]);
+  diff_info_size_ = diff_info_width_ * diff_info_height_ * sizeof(bool);
+  diff_info_.reset(new bool[diff_info_size_]);
 }
 
 Differ::~Differ() {}
 
-void Differ::CalcDirtyRegion(const void* prev_buffer, const void* curr_buffer,
+void Differ::CalcDirtyRegion(const uint8_t* prev_buffer,
+                             const uint8_t* curr_buffer,
                              DesktopRegion* region) {
   // Identify all the blocks that contain changed pixels.
   MarkDirtyBlocks(prev_buffer, curr_buffer);
@@ -44,7 +45,8 @@ void Differ::CalcDirtyRegion(const void* prev_buffer, const void* curr_buffer,
   MergeBlocks(region);
 }
 
-void Differ::MarkDirtyBlocks(const void* prev_buffer, const void* curr_buffer) {
+void Differ::MarkDirtyBlocks(const uint8_t* prev_buffer,
+                             const uint8_t* curr_buffer) {
   memset(diff_info_.get(), 0, diff_info_size_);
 
   // Calc number of full blocks.
@@ -60,18 +62,16 @@ void Differ::MarkDirtyBlocks(const void* prev_buffer, const void* curr_buffer) {
   // Offset from the start of one block-row to the next.
   int block_y_stride = (width_ * bytes_per_pixel_) * kBlockSize;
   // Offset from the start of one diff_info row to the next.
-  int diff_info_stride = diff_info_width_ * sizeof(DiffInfo);
+  int diff_info_stride = diff_info_width_ * sizeof(bool);
 
-  const uint8_t* prev_block_row_start =
-      static_cast<const uint8_t*>(prev_buffer);
-  const uint8_t* curr_block_row_start =
-      static_cast<const uint8_t*>(curr_buffer);
-  DiffInfo* diff_info_row_start = static_cast<DiffInfo*>(diff_info_.get());
+  const uint8_t* prev_block_row_start = prev_buffer;
+  const uint8_t* curr_block_row_start = curr_buffer;
+  bool* diff_info_row_start = diff_info_.get();
 
   for (int y = 0; y < y_full_blocks; y++) {
     const uint8_t* prev_block = prev_block_row_start;
     const uint8_t* curr_block = curr_block_row_start;
-    DiffInfo* diff_info = diff_info_row_start;
+    bool* diff_info = diff_info_row_start;
 
     for (int x = 0; x < x_full_blocks; x++) {
       // Mark this block as being modified so that it gets incorporated into
@@ -79,15 +79,15 @@ void Differ::MarkDirtyBlocks(const void* prev_buffer, const void* curr_buffer) {
       *diff_info = BlockDifference(prev_block, curr_block, bytes_per_row_);
       prev_block += block_x_offset;
       curr_block += block_x_offset;
-      diff_info += sizeof(DiffInfo);
+      diff_info += sizeof(bool);
     }
 
     // If there is a partial column at the end, handle it.
     // This condition should rarely, if ever, occur.
     if (partial_column_width != 0) {
-      *diff_info = DiffPartialBlock(prev_block, curr_block, bytes_per_row_,
-                                    partial_column_width, kBlockSize);
-      diff_info += sizeof(DiffInfo);
+      *diff_info = !PartialBlocksEqual(prev_block, curr_block, bytes_per_row_,
+                                       partial_column_width, kBlockSize);
+      diff_info += sizeof(bool);
     }
 
     // Update pointers for next row.
@@ -102,74 +102,75 @@ void Differ::MarkDirtyBlocks(const void* prev_buffer, const void* curr_buffer) {
   if (partial_row_height != 0) {
     const uint8_t* prev_block = prev_block_row_start;
     const uint8_t* curr_block = curr_block_row_start;
-    DiffInfo* diff_info = diff_info_row_start;
+    bool* diff_info = diff_info_row_start;
     for (int x = 0; x < x_full_blocks; x++) {
-      *diff_info = DiffPartialBlock(prev_block, curr_block,
-                                    bytes_per_row_,
-                                    kBlockSize, partial_row_height);
+      *diff_info = !PartialBlocksEqual(prev_block, curr_block,
+                                       bytes_per_row_,
+                                       kBlockSize, partial_row_height);
       prev_block += block_x_offset;
       curr_block += block_x_offset;
-      diff_info += sizeof(DiffInfo);
+      diff_info += sizeof(bool);
     }
     if (partial_column_width != 0) {
-      *diff_info = DiffPartialBlock(prev_block, curr_block, bytes_per_row_,
-                                    partial_column_width, partial_row_height);
-      diff_info += sizeof(DiffInfo);
+      *diff_info = !PartialBlocksEqual(prev_block, curr_block, bytes_per_row_,
+                                       partial_column_width,
+                                       partial_row_height);
+      diff_info += sizeof(bool);
     }
   }
 }
 
-DiffInfo Differ::DiffPartialBlock(const uint8_t* prev_buffer,
-                                  const uint8_t* curr_buffer,
-                                  int stride, int width, int height) {
+bool Differ::PartialBlocksEqual(const uint8_t* prev_buffer,
+                                const uint8_t* curr_buffer,
+                                int stride, int width, int height) {
   int width_bytes = width * bytes_per_pixel_;
   for (int y = 0; y < height; y++) {
     if (memcmp(prev_buffer, curr_buffer, width_bytes) != 0)
-      return 1;
+      return false;
     prev_buffer += bytes_per_row_;
     curr_buffer += bytes_per_row_;
   }
-  return 0;
+  return true;
 }
 
 void Differ::MergeBlocks(DesktopRegion* region) {
   region->Clear();
 
-  uint8_t* diff_info_row_start = static_cast<uint8_t*>(diff_info_.get());
-  int diff_info_stride = diff_info_width_ * sizeof(DiffInfo);
+  bool* diff_info_row_start = diff_info_.get();
+  int diff_info_stride = diff_info_width_ * sizeof(bool);
 
   for (int y = 0; y < diff_info_height_; y++) {
-    uint8_t* diff_info = diff_info_row_start;
+    bool* diff_info = diff_info_row_start;
     for (int x = 0; x < diff_info_width_; x++) {
-      if (*diff_info != 0) {
+      if (*diff_info) {
         // We've found a modified block. Look at blocks to the right and below
         // to group this block with as many others as we can.
         int left = x * kBlockSize;
         int top = y * kBlockSize;
         int width = 1;
         int height = 1;
-        *diff_info = 0;
+        *diff_info = false;
 
         // Group with blocks to the right.
         // We can keep looking until we find an unchanged block because we
         // have a boundary block which is never marked as having diffs.
-        uint8_t* right = diff_info + 1;
+        bool* right = diff_info + 1;
         while (*right) {
-          *right++ = 0;
+          *right++ = false;
           width++;
         }
 
         // Group with blocks below.
         // The entire width of blocks that we matched above much match for
         // each row that we add.
-        uint8_t* bottom = diff_info;
+        bool* bottom = diff_info;
         bool found_new_row;
         do {
           found_new_row = true;
           bottom += diff_info_stride;
           right = bottom;
           for (int x2 = 0; x2 < width; x2++) {
-            if (*right++ == 0) {
+            if (!*right++) {
               found_new_row = false;
             }
           }
@@ -181,7 +182,7 @@ void Differ::MergeBlocks(DesktopRegion* region) {
             // try to add these blocks a second time.
             right = bottom;
             for (int x2 = 0; x2 < width; x2++) {
-              *right++ = 0;
+              *right++ = false;
             }
           }
         } while (found_new_row);
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.h
index d5e21db953..b3b0e7c244 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ.h
@@ -18,8 +18,6 @@
 
 namespace webrtc {
 
-typedef uint8_t DiffInfo;
-
 // TODO(sergeyu): Simplify differ now that we are working with DesktopRegion.
 // diff_info_ should no longer be needed, as we can put our data directly into
 // the region that we are calculating.
@@ -40,7 +38,7 @@ class Differ {
 
   // Given the previous and current screen buffer, calculate the dirty region
   // that encloses all of the changed pixels in the new screen.
-  void CalcDirtyRegion(const void* prev_buffer, const void* curr_buffer,
+  void CalcDirtyRegion(const uint8_t* prev_buffer, const uint8_t* curr_buffer,
                        DesktopRegion* region);
 
  private:
@@ -48,21 +46,21 @@ class Differ {
   friend class DifferTest;
 
   // Identify all of the blocks that contain changed pixels.
-  void MarkDirtyBlocks(const void* prev_buffer, const void* curr_buffer);
+  void MarkDirtyBlocks(const uint8_t* prev_buffer, const uint8_t* curr_buffer);
 
   // After the dirty blocks have been identified, this routine merges adjacent
   // blocks into a region.
   // The goal is to minimize the region that covers the dirty blocks.
   void MergeBlocks(DesktopRegion* region);
 
-  // Check for diffs in upper-left portion of the block. The size of the portion
-  // to check is specified by the |width| and |height| values.
+  // Checks whether the upper-left portions of the buffers are equal. The size
+  // of the portion to check is specified by the |width| and |height| values.
   // Note that if we force the capturer to always return images whose width and
   // height are multiples of kBlockSize, then this will never be called.
-  DiffInfo DiffPartialBlock(const uint8_t* prev_buffer,
-                            const uint8_t* curr_buffer,
-                            int stride,
-                            int width, int height);
+  bool PartialBlocksEqual(const uint8_t* prev_buffer,
+                          const uint8_t* curr_buffer,
+                          int stride,
+                          int width, int height);
 
   // Dimensions of screen.
   int width_;
@@ -76,14 +74,14 @@ class Differ {
   int bytes_per_row_;
 
   // Diff information for each block in the image.
-  rtc::scoped_ptr<DiffInfo[]> diff_info_;
+  rtc::scoped_ptr<bool[]> diff_info_;
 
   // Dimensions and total size of diff info array.
   int diff_info_width_;
   int diff_info_height_;
   int diff_info_size_;
 
-  DISALLOW_COPY_AND_ASSIGN(Differ);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Differ);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.cc
index a1cc93b386..f13d225600 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.cc
@@ -12,28 +12,30 @@
 
 #include <string.h>
 
-#include "build/build_config.h"
+#include "webrtc/typedefs.h"
 #include "webrtc/modules/desktop_capture/differ_block_sse2.h"
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
 namespace webrtc {
 
-int BlockDifference_C(const uint8_t* image1,
-                      const uint8_t* image2,
-                      int stride) {
+bool BlockDifference_C(const uint8_t* image1,
+                       const uint8_t* image2,
+                       int stride) {
   int width_bytes = kBlockSize * kBytesPerPixel;
 
   for (int y = 0; y < kBlockSize; y++) {
     if (memcmp(image1, image2, width_bytes) != 0)
-      return 1;
+      return true;
     image1 += stride;
     image2 += stride;
   }
-  return 0;
+  return false;
 }
 
-int BlockDifference(const uint8_t* image1, const uint8_t* image2, int stride) {
-  static int (*diff_proc)(const uint8_t*, const uint8_t*, int) = NULL;
+bool BlockDifference(const uint8_t* image1,
+                     const uint8_t* image2,
+                     int stride) {
+  static bool (*diff_proc)(const uint8_t*, const uint8_t*, int) = NULL;
 
   if (!diff_proc) {
 #if !defined(WEBRTC_ARCH_X86_FAMILY)
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.h
index 2b43f4ed06..e1d487d68b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_DIFFER_BLOCK_H_
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_DIFFER_BLOCK_H_
 
-#include "webrtc/typedefs.h"
+#include <stdint.h>
 
 namespace webrtc {
 
@@ -22,9 +22,11 @@ const int kBlockSize = 32;
 // Format: BGRA 32 bit.
 const int kBytesPerPixel = 4;
 
-// Low level functions to compare 2 blocks of pixels. Zero means the blocks
-// are identical. One - the blocks are different.
-int BlockDifference(const uint8_t* image1, const uint8_t* image2, int stride);
+// Low level function to compare 2 blocks of pixels of size
+// (kBlockSize, kBlockSize).  Returns whether the blocks differ.
+bool BlockDifference(const uint8_t* image1,
+                     const uint8_t* image2,
+                     int stride);
 
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.cc
index 7f31bd3a84..8d35df2226 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.cc
@@ -21,9 +21,9 @@
 
 namespace webrtc {
 
-extern int BlockDifference_SSE2_W16(const uint8_t* image1,
-                                    const uint8_t* image2,
-                                    int stride) {
+extern bool BlockDifference_SSE2_W16(const uint8_t* image1,
+                                     const uint8_t* image2,
+                                     int stride) {
   __m128i acc = _mm_setzero_si128();
   __m128i v0;
   __m128i v1;
@@ -54,16 +54,16 @@ extern int BlockDifference_SSE2_W16(const uint8_t* image1,
     sad = _mm_adds_epu16(sad, acc);
     int diff = _mm_cvtsi128_si32(sad);
     if (diff)
-      return 1;
+      return true;
     image1 += stride;
     image2 += stride;
   }
-  return 0;
+  return false;
 }
 
-extern int BlockDifference_SSE2_W32(const uint8_t* image1,
-                                    const uint8_t* image2,
-                                    int stride) {
+extern bool BlockDifference_SSE2_W32(const uint8_t* image1,
+                                     const uint8_t* image2,
+                                     int stride) {
   __m128i acc = _mm_setzero_si128();
   __m128i v0;
   __m128i v1;
@@ -110,11 +110,11 @@ extern int BlockDifference_SSE2_W32(const uint8_t* image1,
     sad = _mm_adds_epu16(sad, acc);
     int diff = _mm_cvtsi128_si32(sad);
     if (diff)
-      return 1;
+      return true;
     image1 += stride;
     image2 += stride;
   }
-  return 0;
+  return false;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.h
index 081e6fa235..90426dafab 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_sse2.h
@@ -19,14 +19,14 @@
 namespace webrtc {
 
 // Find block difference of dimension 16x16.
-extern int BlockDifference_SSE2_W16(const uint8_t* image1,
-                                    const uint8_t* image2,
-                                    int stride);
+extern bool BlockDifference_SSE2_W16(const uint8_t* image1,
+                                     const uint8_t* image2,
+                                     int stride);
 
 // Find block difference of dimension 32x32.
-extern int BlockDifference_SSE2_W32(const uint8_t* image1,
-                                    const uint8_t* image2,
-                                    int stride);
+extern bool BlockDifference_SSE2_W32(const uint8_t* image1,
+                                     const uint8_t* image2,
+                                     int stride);
 
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_unittest.cc
index 4f79eed983..df9f4d517a 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block_unittest.cc
@@ -10,7 +10,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/modules/desktop_capture/differ_block.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_unittest.cc
index 019e952e9c..642cb37448 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_unittest.cc
@@ -51,7 +51,7 @@ class DifferTest : public testing::Test {
   }
 
   // Here in DifferTest so that tests can access private methods of Differ.
-  void MarkDirtyBlocks(const void* prev_buffer, const void* curr_buffer) {
+  void MarkDirtyBlocks(const uint8_t* prev_buffer, const uint8_t* curr_buffer) {
     differ_->MarkDirtyBlocks(prev_buffer, curr_buffer);
   }
 
@@ -71,7 +71,7 @@ class DifferTest : public testing::Test {
 
   // Convenience wrapper for Differ's DiffBlock that calculates the appropriate
   // offset to the start of the desired block.
-  DiffInfo DiffBlock(int block_x, int block_y) {
+  bool DiffBlock(int block_x, int block_y) {
     // Offset from upper-left of buffer to upper-left of requested block.
     int block_offset = ((block_y * stride_) + (block_x * bytes_per_pixel_))
                         * kBlockSize;
@@ -114,8 +114,8 @@ class DifferTest : public testing::Test {
   }
 
   // Get the value in the |diff_info_| array at (x,y).
-  DiffInfo GetDiffInfo(int x, int y) {
-    DiffInfo* diff_info = differ_->diff_info_.get();
+  bool GetDiffInfo(int x, int y) {
+    bool* diff_info = differ_->diff_info_.get();
     return diff_info[(y * GetDiffInfoWidth()) + x];
   }
 
@@ -134,8 +134,8 @@ class DifferTest : public testing::Test {
     return differ_->diff_info_size_;
   }
 
-  void SetDiffInfo(int x, int y, const DiffInfo& value) {
-    DiffInfo* diff_info = differ_->diff_info_.get();
+  void SetDiffInfo(int x, int y, bool value) {
+    bool* diff_info = differ_->diff_info_.get();
     diff_info[(y * GetDiffInfoWidth()) + x] = value;
   }
 
@@ -143,7 +143,7 @@ class DifferTest : public testing::Test {
   void MarkBlocks(int x_origin, int y_origin, int width, int height) {
     for (int y = 0; y < height; y++) {
       for (int x = 0; x < width; x++) {
-        SetDiffInfo(x_origin + x, y_origin + y, 1);
+        SetDiffInfo(x_origin + x, y_origin + y, true);
       }
     }
   }
@@ -204,7 +204,7 @@ class DifferTest : public testing::Test {
   rtc::scoped_ptr<uint8_t[]> curr_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(DifferTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DifferTest);
 };
 
 TEST_F(DifferTest, Setup) {
@@ -240,7 +240,7 @@ TEST_F(DifferTest, MarkDirtyBlocks_All) {
   // Make sure each block is marked as dirty.
   for (int y = 0; y < GetDiffInfoHeight() - 1; y++) {
     for (int x = 0; x < GetDiffInfoWidth() - 1; x++) {
-      EXPECT_EQ(1, GetDiffInfo(x, y))
+      EXPECT_TRUE(GetDiffInfo(x, y))
           << "when x = " << x << ", and y = " << y;
     }
   }
@@ -258,23 +258,23 @@ TEST_F(DifferTest, MarkDirtyBlocks_Sampling) {
   MarkDirtyBlocks(prev_.get(), curr_.get());
 
   // Make sure corresponding blocks are updated.
-  EXPECT_EQ(0, GetDiffInfo(0, 0));
-  EXPECT_EQ(0, GetDiffInfo(0, 1));
-  EXPECT_EQ(1, GetDiffInfo(0, 2));
-  EXPECT_EQ(1, GetDiffInfo(1, 0));
-  EXPECT_EQ(0, GetDiffInfo(1, 1));
-  EXPECT_EQ(0, GetDiffInfo(1, 2));
-  EXPECT_EQ(0, GetDiffInfo(2, 0));
-  EXPECT_EQ(1, GetDiffInfo(2, 1));
-  EXPECT_EQ(0, GetDiffInfo(2, 2));
+  EXPECT_FALSE(GetDiffInfo(0, 0));
+  EXPECT_FALSE(GetDiffInfo(0, 1));
+  EXPECT_TRUE(GetDiffInfo(0, 2));
+  EXPECT_TRUE(GetDiffInfo(1, 0));
+  EXPECT_FALSE(GetDiffInfo(1, 1));
+  EXPECT_FALSE(GetDiffInfo(1, 2));
+  EXPECT_FALSE(GetDiffInfo(2, 0));
+  EXPECT_TRUE(GetDiffInfo(2, 1));
+  EXPECT_FALSE(GetDiffInfo(2, 2));
 }
 
 TEST_F(DifferTest, DiffBlock) {
   InitDiffer(kScreenWidth, kScreenHeight);
 
   // Verify no differences at start.
-  EXPECT_EQ(0, DiffBlock(0, 0));
-  EXPECT_EQ(0, DiffBlock(1, 1));
+  EXPECT_FALSE(DiffBlock(0, 0));
+  EXPECT_FALSE(DiffBlock(1, 1));
 
   // Write new data into the 4 corners of the middle block and verify that
   // neighboring blocks are not affected.
@@ -283,15 +283,15 @@ TEST_F(DifferTest, DiffBlock) {
   WriteBlockPixel(curr_.get(), 1, 1, 0, max, 0xffffff);
   WriteBlockPixel(curr_.get(), 1, 1, max, 0, 0xffffff);
   WriteBlockPixel(curr_.get(), 1, 1, max, max, 0xffffff);
-  EXPECT_EQ(0, DiffBlock(0, 0));
-  EXPECT_EQ(0, DiffBlock(0, 1));
-  EXPECT_EQ(0, DiffBlock(0, 2));
-  EXPECT_EQ(0, DiffBlock(1, 0));
-  EXPECT_EQ(1, DiffBlock(1, 1));  // Only this block should change.
-  EXPECT_EQ(0, DiffBlock(1, 2));
-  EXPECT_EQ(0, DiffBlock(2, 0));
-  EXPECT_EQ(0, DiffBlock(2, 1));
-  EXPECT_EQ(0, DiffBlock(2, 2));
+  EXPECT_FALSE(DiffBlock(0, 0));
+  EXPECT_FALSE(DiffBlock(0, 1));
+  EXPECT_FALSE(DiffBlock(0, 2));
+  EXPECT_FALSE(DiffBlock(1, 0));
+  EXPECT_TRUE(DiffBlock(1, 1));  // Only this block should change.
+  EXPECT_FALSE(DiffBlock(1, 2));
+  EXPECT_FALSE(DiffBlock(2, 0));
+  EXPECT_FALSE(DiffBlock(2, 1));
+  EXPECT_FALSE(DiffBlock(2, 2));
 }
 
 TEST_F(DifferTest, Partial_Setup) {
@@ -328,7 +328,7 @@ TEST_F(DifferTest, Partial_FirstPixel) {
   // Make sure each block is marked as dirty.
   for (int y = 0; y < GetDiffInfoHeight() - 1; y++) {
     for (int x = 0; x < GetDiffInfoWidth() - 1; x++) {
-      EXPECT_EQ(1, GetDiffInfo(x, y))
+      EXPECT_TRUE(GetDiffInfo(x, y))
           << "when x = " << x << ", and y = " << y;
     }
   }
@@ -351,18 +351,18 @@ TEST_F(DifferTest, Partial_BorderPixel) {
   // Make sure last (partial) block in each row/column is marked as dirty.
   int x_last = GetDiffInfoWidth() - 2;
   for (int y = 0; y < GetDiffInfoHeight() - 1; y++) {
-    EXPECT_EQ(1, GetDiffInfo(x_last, y))
+    EXPECT_TRUE(GetDiffInfo(x_last, y))
         << "when x = " << x_last << ", and y = " << y;
   }
   int y_last = GetDiffInfoHeight() - 2;
   for (int x = 0; x < GetDiffInfoWidth() - 1; x++) {
-    EXPECT_EQ(1, GetDiffInfo(x, y_last))
+    EXPECT_TRUE(GetDiffInfo(x, y_last))
         << "when x = " << x << ", and y = " << y_last;
   }
   // All other blocks are clean.
   for (int y = 0; y < GetDiffInfoHeight() - 2; y++) {
     for (int x = 0; x < GetDiffInfoWidth() - 2; x++) {
-      EXPECT_EQ(0, GetDiffInfo(x, y)) << "when x = " << x << ", and y = " << y;
+      EXPECT_FALSE(GetDiffInfo(x, y)) << "when x = " << x << ", and y = " << y;
     }
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration.mm b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration.mm
index 35fa65be2d..9e483e5b81 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration.mm
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration.mm
@@ -14,17 +14,17 @@
 #include <algorithm>
 #include <Cocoa/Cocoa.h>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 #if !defined(MAC_OS_X_VERSION_10_7) || \
-    MAC_OS_X_VERSION_MAX_ALLOWED < MAC_OS_X_VERSION_10_7
+    MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_X_VERSION_10_7
 
 @interface NSScreen (LionAPI)
 - (CGFloat)backingScaleFactor;
 - (NSRect)convertRectToBacking:(NSRect)aRect;
 @end
 
-#endif  // 10.7
+#endif  // MAC_OS_X_VERSION_10_7
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
index f0d5c34be6..eeccecb6cc 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
@@ -11,8 +11,8 @@
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h"
 
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
index bd502f0815..b2fa81a416 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
@@ -17,7 +17,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
@@ -58,7 +58,7 @@ class DesktopConfigurationMonitor {
   MacDesktopConfiguration desktop_configuration_;
   rtc::scoped_ptr<EventWrapper> display_configuration_capture_event_;
 
-  DISALLOW_COPY_AND_ASSIGN(DesktopConfigurationMonitor);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopConfigurationMonitor);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
index 23c432f60f..84579c4149 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
@@ -17,7 +17,7 @@
 #include "webrtc/base/macutils.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
 #include "webrtc/modules/desktop_capture/mac/window_list_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h
index b24fc997e4..4e6008966e 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h
@@ -14,8 +14,8 @@
 #include <ApplicationServices/ApplicationServices.h>
 
 #include "webrtc/modules/desktop_capture/window_capturer.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -61,7 +61,7 @@ class FullScreenChromeWindowDetector {
   WindowCapturer::WindowList previous_window_list_;
   TickTime last_udpate_time_;
 
-  DISALLOW_COPY_AND_ASSIGN(FullScreenChromeWindowDetector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(FullScreenChromeWindowDetector);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h
index 4d1dd1ffd6..a32d470954 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h
@@ -33,7 +33,7 @@ class ScopedPixelBufferObject {
   CGLContextObj cgl_context_;
   GLuint pixel_buffer_object_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScopedPixelBufferObject);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedPixelBufferObject);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor.h
index 1da98a4905..dd5dc0eb44 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor.h
@@ -40,7 +40,7 @@ class MouseCursor {
   rtc::scoped_ptr<DesktopFrame> image_;
   DesktopVector hotspot_;
 
-  DISALLOW_COPY_AND_ASSIGN(MouseCursor);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MouseCursor);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
index a839e7735a..18fb1c9128 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
@@ -17,14 +17,14 @@
 
 #include "webrtc/base/macutils.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h"
 #include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -48,13 +48,13 @@ class MouseCursorMonitorMac : public MouseCursorMonitor {
 
   void CaptureImage();
 
-  scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
+  rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
   CGWindowID window_id_;
   ScreenId screen_id_;
   Callback* callback_;
   Mode mode_;
   rtc::scoped_ptr<MouseCursor> last_cursor_;
-  scoped_refptr<FullScreenChromeWindowDetector>
+  rtc::scoped_refptr<FullScreenChromeWindowDetector>
       full_screen_chrome_window_detector_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
index c37f0ddbb8..1aa5fb871f 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
@@ -16,7 +16,7 @@
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
 #include "webrtc/modules/desktop_capture/window_capturer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
index 9e72de5d38..c6f7668381 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
@@ -16,7 +16,7 @@
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
 #include "webrtc/modules/desktop_capture/win/cursor.h"
 #include "webrtc/modules/desktop_capture/win/window_capture_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
index 857331f139..bb36660dab 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
@@ -19,7 +19,7 @@
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
 #include "webrtc/modules/desktop_capture/x11/x_error_trap.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace {
 
@@ -77,7 +77,7 @@ class MouseCursorMonitorX11 : public MouseCursorMonitor,
   // Captures current cursor shape and stores it in |cursor_shape_|.
   void CaptureCursor();
 
-  scoped_refptr<SharedXDisplay> x_display_;
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
   Callback* callback_;
   Mode mode_;
   Window window_;
@@ -90,7 +90,6 @@ class MouseCursorMonitorX11 : public MouseCursorMonitor,
   rtc::scoped_ptr<MouseCursor> cursor_shape_;
 };
 
-// For screens, we pass the same windowid for window and inner_window
 MouseCursorMonitorX11::MouseCursorMonitorX11(
     const DesktopCaptureOptions& options,
     Window window, Window inner_window)
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc
index 45a3507b92..94d8a27b13 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc
@@ -15,7 +15,7 @@
 
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
index f3b11cfbff..6cd9e3bfc8 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
@@ -66,7 +66,7 @@ class ScreenCaptureFrameQueue {
   static const int kQueueLength = 2;
   rtc::scoped_ptr<SharedDesktopFrame> frames_[kQueueLength];
 
-  DISALLOW_COPY_AND_ASSIGN(ScreenCaptureFrameQueue);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCaptureFrameQueue);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.cc
index 86761c170f..fa7096d24d 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.cc
@@ -13,7 +13,7 @@
 #include <assert.h>
 #include <algorithm>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.h
index a8be989791..da1f2bfeeb 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_helper.h
@@ -14,7 +14,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 namespace webrtc {
 
@@ -80,7 +80,7 @@ class ScreenCapturerHelper {
   // If the value is <= 0, then the invalid region is not expanded to a grid.
   int log_grid_size_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScreenCapturerHelper);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerHelper);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mac.mm b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mac.mm
index ecab62e751..2c857b4f1e 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mac.mm
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mac.mm
@@ -32,8 +32,8 @@
 #include "webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -191,7 +191,7 @@ CGImageRef CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
 class ScreenCapturerMac : public ScreenCapturer {
  public:
   explicit ScreenCapturerMac(
-      scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor);
+      rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor);
   virtual ~ScreenCapturerMac();
 
   bool Init();
@@ -273,7 +273,7 @@ class ScreenCapturerMac : public ScreenCapturer {
   DesktopRegion last_invalid_region_;
 
   // Monitoring display reconfiguration.
-  scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
+  rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
 
   // Power management assertion to prevent the screen from sleeping.
   IOPMAssertionID power_assertion_id_display_;
@@ -291,7 +291,7 @@ class ScreenCapturerMac : public ScreenCapturer {
 
   CGWindowID excluded_window_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac);
 };
 
 // DesktopFrame wrapper that flips wrapped frame upside down by inverting
@@ -314,11 +314,11 @@ class InvertedDesktopFrame : public DesktopFrame {
  private:
   rtc::scoped_ptr<DesktopFrame> original_frame_;
 
-  DISALLOW_COPY_AND_ASSIGN(InvertedDesktopFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(InvertedDesktopFrame);
 };
 
 ScreenCapturerMac::ScreenCapturerMac(
-    scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor)
+    rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor)
     : screen_callback_data_(new ScreenCallbackData(this)),
       callback_(NULL),
       cgl_context_(NULL),
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h
index 373e66f7bd..8b83f41252 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h
@@ -27,7 +27,7 @@ class MockScreenCapturer : public ScreenCapturer {
   MOCK_METHOD1(SelectScreen, bool(ScreenId id));
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(MockScreenCapturer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MockScreenCapturer);
 };
 
 class MockScreenCapturerCallback : public ScreenCapturer::Callback {
@@ -39,7 +39,7 @@ class MockScreenCapturerCallback : public ScreenCapturer::Callback {
   MOCK_METHOD1(OnCaptureCompleted, void(DesktopFrame*));
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(MockScreenCapturerCallback);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MockScreenCapturerCallback);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
index 606c06153d..a3cf6d93cc 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
@@ -51,7 +51,7 @@ class FakeSharedMemory : public SharedMemory {
   }
  private:
   char* buffer_;
-  DISALLOW_COPY_AND_ASSIGN(FakeSharedMemory);
+  RTC_DISALLOW_COPY_AND_ASSIGN(FakeSharedMemory);
 };
 
 SharedMemory* ScreenCapturerTest::CreateSharedMemory(size_t size) {
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_win.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_win.cc
index 1f33155656..18be4eb30b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_win.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 
+#include <utility>
+
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h"
 #include "webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h"
@@ -22,7 +24,7 @@ ScreenCapturer* ScreenCapturer::Create(const DesktopCaptureOptions& options) {
       new ScreenCapturerWinGdi(options));
 
   if (options.allow_use_magnification_api())
-    return new ScreenCapturerWinMagnifier(gdi_capturer.Pass());
+    return new ScreenCapturerWinMagnifier(std::move(gdi_capturer));
 
   return gdi_capturer.release();
 }
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_x11.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_x11.cc
index 233be2ff40..37f4a34c9f 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_x11.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/screen_capturer_x11.cc
@@ -18,6 +18,7 @@
 #include <X11/Xlib.h>
 #include <X11/Xutil.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
@@ -25,18 +26,10 @@
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
 #include "webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-
-// TODO(sergeyu): Move this to a header where it can be shared.
-#if defined(NDEBUG)
-#define DCHECK(condition) (void)(condition)
-#else  // NDEBUG
-#define DCHECK(condition) if (!(condition)) {abort();}
-#endif
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
-
 namespace {
 
 // A class to perform video frame capturing for Linux.
@@ -122,7 +115,7 @@ class ScreenCapturerLinux : public ScreenCapturer,
   // |Differ| for use when polling for changes.
   rtc::scoped_ptr<Differ> differ_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScreenCapturerLinux);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerLinux);
 };
 
 ScreenCapturerLinux::ScreenCapturerLinux()
@@ -234,8 +227,8 @@ void ScreenCapturerLinux::InitXDamage() {
 }
 
 void ScreenCapturerLinux::Start(Callback* callback) {
-  DCHECK(!callback_);
-  DCHECK(callback);
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
 
   callback_ = callback;
 }
@@ -290,7 +283,7 @@ void ScreenCapturerLinux::Capture(const DesktopRegion& region) {
 }
 
 bool ScreenCapturerLinux::GetScreenList(ScreenList* screens) {
-  DCHECK(screens->size() == 0);
+  RTC_DCHECK(screens->size() == 0);
   // TODO(jiayl): implement screen enumeration.
   Screen default_screen;
   default_screen.id = 0;
@@ -309,7 +302,7 @@ bool ScreenCapturerLinux::HandleXEvent(const XEvent& event) {
         reinterpret_cast<const XDamageNotifyEvent*>(&event);
     if (damage_event->damage != damage_handle_)
       return false;
-    DCHECK(damage_event->level == XDamageReportNonEmpty);
+    RTC_DCHECK(damage_event->level == XDamageReportNonEmpty);
     return true;
   } else if (event.type == ConfigureNotify) {
     ScreenConfigurationChanged();
@@ -372,8 +365,8 @@ DesktopFrame* ScreenCapturerLinux::CaptureScreen() {
     if (queue_.previous_frame()) {
       // Full-screen polling, so calculate the invalid rects here, based on the
       // changed pixels between current and previous buffers.
-      DCHECK(differ_.get() != NULL);
-      DCHECK(queue_.previous_frame()->data());
+      RTC_DCHECK(differ_.get() != NULL);
+      RTC_DCHECK(queue_.previous_frame()->data());
       differ_->CalcDirtyRegion(queue_.previous_frame()->data(),
                                frame->data(), updated_region);
     } else {
@@ -408,11 +401,11 @@ void ScreenCapturerLinux::SynchronizeFrame() {
   // TODO(hclam): We can reduce the amount of copying here by subtracting
   // |capturer_helper_|s region from |last_invalid_region_|.
   // http://crbug.com/92354
-  DCHECK(queue_.previous_frame());
+  RTC_DCHECK(queue_.previous_frame());
 
   DesktopFrame* current = queue_.current_frame();
   DesktopFrame* last = queue_.previous_frame();
-  DCHECK(current != last);
+  RTC_DCHECK(current != last);
   for (DesktopRegion::Iterator it(last_invalid_region_);
        !it.IsAtEnd(); it.Advance()) {
     current->CopyPixelsFrom(*last, it.rect().top_left(), it.rect());
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.cc
index 591b6225c7..1f1aefa13b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
@@ -41,7 +41,7 @@ class SharedDesktopFrame::Core {
   Atomic32 ref_count_;
   rtc::scoped_ptr<DesktopFrame> frame_;
 
-  DISALLOW_COPY_AND_ASSIGN(Core);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Core);
 };
 
 SharedDesktopFrame::~SharedDesktopFrame() {}
@@ -49,7 +49,7 @@ SharedDesktopFrame::~SharedDesktopFrame() {}
 // static
 SharedDesktopFrame* SharedDesktopFrame::Wrap(
     DesktopFrame* desktop_frame) {
-  scoped_refptr<Core> core(new Core(desktop_frame));
+  rtc::scoped_refptr<Core> core(new Core(desktop_frame));
   return new SharedDesktopFrame(core);
 }
 
@@ -69,9 +69,11 @@ bool SharedDesktopFrame::IsShared() {
   return !core_->HasOneRef();
 }
 
-SharedDesktopFrame::SharedDesktopFrame(scoped_refptr<Core> core)
-    : DesktopFrame(core->frame()->size(), core->frame()->stride(),
-                   core->frame()->data(), core->frame()->shared_memory()),
+SharedDesktopFrame::SharedDesktopFrame(rtc::scoped_refptr<Core> core)
+    : DesktopFrame(core->frame()->size(),
+                   core->frame()->stride(),
+                   core->frame()->data(),
+                   core->frame()->shared_memory()),
       core_(core) {
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.h
index d77cb15b4d..7d18db153c 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_desktop_frame.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
 
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
 
 namespace webrtc {
 
@@ -37,11 +37,11 @@ class SharedDesktopFrame : public DesktopFrame {
  private:
   class Core;
 
-  SharedDesktopFrame(scoped_refptr<Core> core);
+  SharedDesktopFrame(rtc::scoped_refptr<Core> core);
 
-  scoped_refptr<Core> core_;
+  rtc::scoped_refptr<Core> core_;
 
-  DISALLOW_COPY_AND_ASSIGN(SharedDesktopFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedDesktopFrame);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_memory.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_memory.h
index 7870d833f1..631f119b5f 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_memory.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/shared_memory.h
@@ -59,7 +59,7 @@ class SharedMemory {
   const int id_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(SharedMemory);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemory);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/cursor.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/cursor.cc
index 35c5190b72..a3acaf822b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/cursor.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/cursor.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -50,7 +50,6 @@ const uint32_t kPixelRgbaWhite = RGBA(0xff, 0xff, 0xff, 0xff);
 const uint32_t kPixelRgbaTransparent = RGBA(0, 0, 0, 0);
 
 const uint32_t kPixelRgbWhite = RGB(0xff, 0xff, 0xff);
-const uint32_t kPixelRgbBlack = RGB(0, 0, 0);
 
 // Expands the cursor shape to add a white outline for visibility against
 // dark backgrounds.
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.cc
index e665751636..97bbfb717b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.cc
@@ -12,7 +12,7 @@
 
 #include <vector>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.h
index 0f3e64d05d..dc3b8c61b9 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/desktop.h
@@ -56,7 +56,7 @@ class Desktop {
   // True if |desktop_| must be closed on teardown.
   bool own_;
 
-  DISALLOW_COPY_AND_ASSIGN(Desktop);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Desktop);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_gdi_object.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_gdi_object.h
index 366df6d4ff..1cac63e43d 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_gdi_object.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_gdi_object.h
@@ -56,7 +56,7 @@ class ScopedGDIObject {
  private:
   T handle_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScopedGDIObject);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedGDIObject);
 };
 
 // The traits class that uses DeleteObject() to close a handle.
@@ -70,7 +70,7 @@ class DeleteObjectTraits {
   }
 
  private:
-  DISALLOW_IMPLICIT_CONSTRUCTORS(DeleteObjectTraits);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DeleteObjectTraits);
 };
 
 // The traits class that uses DestroyCursor() to close a handle.
@@ -83,7 +83,7 @@ class DestroyCursorTraits {
   }
 
  private:
-  DISALLOW_IMPLICIT_CONSTRUCTORS(DestroyCursorTraits);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DestroyCursorTraits);
 };
 
 typedef ScopedGDIObject<HBITMAP, DeleteObjectTraits<HBITMAP> > ScopedBitmap;
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
index 5666fc8977..12f9e89e96 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h"
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 #include "webrtc/modules/desktop_capture/win/desktop.h"
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h
index 7566e6a0e1..df8652ac9d 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/scoped_thread_desktop.h
@@ -45,7 +45,7 @@ class ScopedThreadDesktop {
   // The desktop handle assigned to the calling thread at creation.
   rtc::scoped_ptr<Desktop> initial_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScopedThreadDesktop);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedThreadDesktop);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
index 756bdcb3cd..4ae234b925 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
@@ -21,8 +21,8 @@
 #include "webrtc/modules/desktop_capture/win/cursor.h"
 #include "webrtc/modules/desktop_capture/win/desktop.h"
 #include "webrtc/modules/desktop_capture/win/screen_capture_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -263,9 +263,10 @@ bool ScreenCapturerWinGdi::CaptureImage() {
         DesktopFrame::kBytesPerPixel;
     SharedMemory* shared_memory = callback_->CreateSharedMemory(buffer_size);
 
-    rtc::scoped_ptr<DesktopFrame> buffer;
-    buffer.reset(
+    rtc::scoped_ptr<DesktopFrame> buffer(
         DesktopFrameWin::Create(size, shared_memory, desktop_dc_));
+    if (!buffer.get())
+      return false;
     queue_.ReplaceCurrentFrame(buffer.release());
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
index d58bd48b46..09dd52ec7b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
@@ -86,7 +86,7 @@ class ScreenCapturerWinGdi : public ScreenCapturer {
   // Used to suppress duplicate logging of SetThreadExecutionState errors.
   bool set_thread_execution_state_failed_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWinGdi);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWinGdi);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
index 16199ca58b..a4ea706339 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -12,6 +12,8 @@
 
 #include <assert.h>
 
+#include <utility>
+
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
@@ -21,8 +23,8 @@
 #include "webrtc/modules/desktop_capture/win/cursor.h"
 #include "webrtc/modules/desktop_capture/win/desktop.h"
 #include "webrtc/modules/desktop_capture/win/screen_capture_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -37,7 +39,7 @@ Atomic32 ScreenCapturerWinMagnifier::tls_index_(TLS_OUT_OF_INDEXES);
 
 ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier(
     rtc::scoped_ptr<ScreenCapturer> fallback_capturer)
-    : fallback_capturer_(fallback_capturer.Pass()),
+    : fallback_capturer_(std::move(fallback_capturer)),
       fallback_capturer_started_(false),
       callback_(NULL),
       current_screen_id_(kFullDesktopScreenId),
@@ -53,8 +55,7 @@ ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier(
       host_window_(NULL),
       magnifier_window_(NULL),
       magnifier_initialized_(false),
-      magnifier_capture_succeeded_(true) {
-}
+      magnifier_capture_succeeded_(true) {}
 
 ScreenCapturerWinMagnifier::~ScreenCapturerWinMagnifier() {
   Stop();
@@ -123,9 +124,9 @@ void ScreenCapturerWinMagnifier::Capture(const DesktopRegion& region) {
 
   bool succeeded = false;
 
-  // Do not try to use the magnfiier if it's capturing non-primary screen, or it
-  // failed before.
-  if (magnifier_initialized_ && IsCapturingPrimaryScreenOnly() &&
+  // Do not try to use the magnifier if it failed before and in multi-screen
+  // setup (where the API crashes sometimes).
+  if (magnifier_initialized_ && (GetSystemMetrics(SM_CMONITORS) == 1) &&
       magnifier_capture_succeeded_) {
     DesktopRect rect = GetScreenRect(current_screen_id_, current_device_key_);
     CreateCurrentFrameIfNecessary(rect.size());
@@ -250,7 +251,7 @@ BOOL ScreenCapturerWinMagnifier::OnMagImageScalingCallback(
     RECT unclipped,
     RECT clipped,
     HRGN dirty) {
-  assert(tls_index_.Value() != TLS_OUT_OF_INDEXES);
+  assert(tls_index_.Value() != static_cast<int32_t>(TLS_OUT_OF_INDEXES));
 
   ScreenCapturerWinMagnifier* owner =
       reinterpret_cast<ScreenCapturerWinMagnifier*>(
@@ -383,7 +384,7 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
     }
   }
 
-  if (tls_index_.Value() == TLS_OUT_OF_INDEXES) {
+  if (tls_index_.Value() == static_cast<int32_t>(TLS_OUT_OF_INDEXES)) {
     // More than one threads may get here at the same time, but only one will
     // write to tls_index_ using CompareExchange.
     DWORD new_tls_index = TlsAlloc();
@@ -391,7 +392,7 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
       TlsFree(new_tls_index);
   }
 
-  assert(tls_index_.Value() != TLS_OUT_OF_INDEXES);
+  assert(tls_index_.Value() != static_cast<int32_t>(TLS_OUT_OF_INDEXES));
   TlsSetValue(tls_index_.Value(), this);
 
   magnifier_initialized_ = true;
@@ -450,13 +451,6 @@ void ScreenCapturerWinMagnifier::CreateCurrentFrameIfNecessary(
   }
 }
 
-bool ScreenCapturerWinMagnifier::IsCapturingPrimaryScreenOnly() const {
-  if (current_screen_id_ != kFullDesktopScreenId)
-    return current_screen_id_ == 0;  // the primary screen is always '0'.
-
-  return GetSystemMetrics(SM_CMONITORS) == 1;
-}
-
 void ScreenCapturerWinMagnifier::StartFallbackCapturer() {
   assert(fallback_capturer_);
   if (!fallback_capturer_started_) {
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
index c1375a62db..0d9d316d1b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
@@ -21,7 +21,7 @@
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
 #include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
@@ -97,9 +97,6 @@ class ScreenCapturerWinMagnifier : public ScreenCapturer {
   // Makes sure the current frame exists and matches |size|.
   void CreateCurrentFrameIfNecessary(const DesktopSize& size);
 
-  // Returns true if we are capturing the primary screen only.
-  bool IsCapturingPrimaryScreenOnly() const;
-
   // Start the fallback capturer and select the screen.
   void StartFallbackCapturer();
 
@@ -149,7 +146,7 @@ class ScreenCapturerWinMagnifier : public ScreenCapturer {
   // successfully. Reset at the beginning of each CaptureImage call.
   bool magnifier_capture_succeeded_;
 
-  DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWinMagnifier);
+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerWinMagnifier);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.cc
index 03e021954b..83922ea7f8 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.cc
@@ -29,7 +29,7 @@ GetCroppedWindowRect(HWND window,
   *original_rect = DesktopRect::MakeLTRB(
       rect.left, rect.top, rect.right, rect.bottom);
 
-  if (window_placement.showCmd & SW_SHOWMAXIMIZED) {
+  if (window_placement.showCmd == SW_SHOWMAXIMIZED) {
     DesktopSize border = DesktopSize(GetSystemMetrics(SM_CXSIZEFRAME),
                                      GetSystemMetrics(SM_CYSIZEFRAME));
     *cropped_rect = DesktopRect::MakeLTRB(
@@ -43,4 +43,27 @@ GetCroppedWindowRect(HWND window,
   return true;
 }
 
+AeroChecker::AeroChecker() : dwmapi_library_(nullptr), func_(nullptr) {
+  // Try to load dwmapi.dll dynamically since it is not available on XP.
+  dwmapi_library_ = LoadLibrary(L"dwmapi.dll");
+  if (dwmapi_library_) {
+    func_ = reinterpret_cast<DwmIsCompositionEnabledFunc>(
+        GetProcAddress(dwmapi_library_, "DwmIsCompositionEnabled"));
+  }
+}
+
+AeroChecker::~AeroChecker() {
+  if (dwmapi_library_) {
+    FreeLibrary(dwmapi_library_);
+  }
+}
+
+bool AeroChecker::IsAeroEnabled() {
+  BOOL result = FALSE;
+  if (func_) {
+    func_(&result);
+  }
+  return result != FALSE;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.h
index 2a3a470c59..225f08b97b 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/win/window_capture_utils.h
@@ -7,8 +7,11 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURE_UTILS_H_
+#define WEBRTC_MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURE_UTILS_H_
 
 #include <windows.h>
+#include <dwmapi.h>
 
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 
@@ -22,4 +25,22 @@ bool GetCroppedWindowRect(HWND window,
                           DesktopRect* cropped_rect,
                           DesktopRect* original_rect);
 
+ typedef HRESULT (WINAPI *DwmIsCompositionEnabledFunc)(BOOL*);
+
+class AeroChecker {
+ public:
+  AeroChecker();
+  ~AeroChecker();
+
+  bool IsAeroEnabled();
+
+ private:
+  HMODULE dwmapi_library_;
+  DwmIsCompositionEnabledFunc func_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AeroChecker);
+};
+
 }  // namespace webrtc
+
+#endif
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_mac.mm b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_mac.mm
index db37cb8675..2cc8c0a761 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_mac.mm
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_mac.mm
@@ -16,14 +16,14 @@
 #include <CoreFoundation/CoreFoundation.h>
 
 #include "webrtc/base/macutils.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
 #include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h"
 #include "webrtc/modules/desktop_capture/mac/window_list_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -44,9 +44,8 @@ bool IsWindowValid(CGWindowID id) {
 
 class WindowCapturerMac : public WindowCapturer {
  public:
-  explicit WindowCapturerMac(
-      scoped_refptr<FullScreenChromeWindowDetector>
-          full_screen_chrome_window_detector);
+  explicit WindowCapturerMac(rtc::scoped_refptr<FullScreenChromeWindowDetector>
+                                 full_screen_chrome_window_detector);
   virtual ~WindowCapturerMac();
 
   // WindowCapturer interface.
@@ -65,15 +64,14 @@ class WindowCapturerMac : public WindowCapturer {
   // The window being captured.
   CGWindowID window_id_;
 
-  scoped_refptr<FullScreenChromeWindowDetector>
+  rtc::scoped_refptr<FullScreenChromeWindowDetector>
       full_screen_chrome_window_detector_;
 
-  DISALLOW_COPY_AND_ASSIGN(WindowCapturerMac);
+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerMac);
 };
 
-WindowCapturerMac::WindowCapturerMac(
-    scoped_refptr<FullScreenChromeWindowDetector>
-        full_screen_chrome_window_detector)
+WindowCapturerMac::WindowCapturerMac(rtc::scoped_refptr<
+    FullScreenChromeWindowDetector> full_screen_chrome_window_detector)
     : callback_(NULL),
       window_id_(0),
       full_screen_chrome_window_detector_(full_screen_chrome_window_detector) {
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_null.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_null.cc
index 584b474e9d..bee22319bb 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_null.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_null.cc
@@ -36,7 +36,7 @@ class WindowCapturerNull : public WindowCapturer {
  private:
   Callback* callback_;
 
-  DISALLOW_COPY_AND_ASSIGN(WindowCapturerNull);
+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerNull);
 };
 
 WindowCapturerNull::WindowCapturerNull()
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_unittest.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_unittest.cc
index bb4ddf07c3..445a4e9848 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_unittest.cc
@@ -15,7 +15,7 @@
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -75,8 +75,8 @@ TEST_F(WindowCapturerTest, Capture) {
     if (!frame_.get()) {
       WindowCapturer::WindowList new_list;
       EXPECT_TRUE(capturer_->GetWindowList(&new_list));
-      for (WindowCapturer::WindowList::iterator new_list_it = windows.begin();
-           new_list_it != windows.end(); ++new_list_it) {
+      for (WindowCapturer::WindowList::iterator new_list_it = new_list.begin();
+           new_list_it != new_list.end(); ++new_list_it) {
         EXPECT_FALSE(it->id == new_list_it->id);
       }
       continue;
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_win.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_win.cc
index 313ed4a4c2..569ef846e0 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_win.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_win.cc
@@ -13,18 +13,17 @@
 #include <assert.h>
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/win32.h"
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
 #include "webrtc/modules/desktop_capture/win/window_capture_utils.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include <VersionHelpers.h>
 
 namespace webrtc {
 
 namespace {
 
-typedef HRESULT (WINAPI *DwmIsCompositionEnabledFunc)(BOOL* enabled);
-
 BOOL CALLBACK WindowsEnumerationHandler(HWND hwnd, LPARAM param) {
   assert(IsGUIThread(false));
   WindowCapturer::WindowList* list =
@@ -43,13 +42,26 @@ BOOL CALLBACK WindowsEnumerationHandler(HWND hwnd, LPARAM param) {
   // Skip the Program Manager window and the Start button.
   const size_t kClassLength = 256;
   WCHAR class_name[kClassLength];
-  GetClassName(hwnd, class_name, kClassLength);
+  const int class_name_length = GetClassName(hwnd, class_name, kClassLength);
+  RTC_DCHECK(class_name_length)
+      << "Error retrieving the application's class name";
+
   // Skip Program Manager window and the Start button. This is the same logic
   // that's used in Win32WindowPicker in libjingle. Consider filtering other
   // windows as well (e.g. toolbars).
   if (wcscmp(class_name, L"Progman") == 0 || wcscmp(class_name, L"Button") == 0)
     return TRUE;
 
+  // Windows 8 introduced a "Modern App" identified by their class name being
+  // either ApplicationFrameWindow or windows.UI.Core.coreWindow. The
+  // associated windows cannot be captured, so we skip them.
+  // http://crbug.com/526883.
+  if (rtc::IsWindows8OrLater() &&
+      (wcscmp(class_name, L"ApplicationFrameWindow") == 0 ||
+       wcscmp(class_name, L"Windows.UI.Core.CoreWindow") == 0)) {
+    return TRUE;
+  }
+
   // Win8 introduced "Modern Apps" whose associated window is
   // non-shareable. We want to filter them out.
   if (IsWindows8OrGreater() &&
@@ -102,48 +114,25 @@ class WindowCapturerWin : public WindowCapturer {
   void Capture(const DesktopRegion& region) override;
 
  private:
-  bool IsAeroEnabled();
-
   Callback* callback_;
 
   // HWND and HDC for the currently selected window or NULL if window is not
   // selected.
   HWND window_;
 
-  // dwmapi.dll is used to determine if desktop compositing is enabled.
-  HMODULE dwmapi_library_;
-  DwmIsCompositionEnabledFunc is_composition_enabled_func_;
-
   DesktopSize previous_size_;
 
-  DISALLOW_COPY_AND_ASSIGN(WindowCapturerWin);
+  AeroChecker aero_checker_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerWin);
 };
 
 WindowCapturerWin::WindowCapturerWin()
     : callback_(NULL),
       window_(NULL) {
-  // Try to load dwmapi.dll dynamically since it is not available on XP.
-  dwmapi_library_ = LoadLibrary(L"dwmapi.dll");
-  if (dwmapi_library_) {
-    is_composition_enabled_func_ =
-        reinterpret_cast<DwmIsCompositionEnabledFunc>(
-            GetProcAddress(dwmapi_library_, "DwmIsCompositionEnabled"));
-    assert(is_composition_enabled_func_);
-  } else {
-    is_composition_enabled_func_ = NULL;
-  }
 }
 
 WindowCapturerWin::~WindowCapturerWin() {
-  if (dwmapi_library_)
-    FreeLibrary(dwmapi_library_);
-}
-
-bool WindowCapturerWin::IsAeroEnabled() {
-  BOOL result = FALSE;
-  if (is_composition_enabled_func_)
-    is_composition_enabled_func_(&result);
-  return result != FALSE;
 }
 
 bool WindowCapturerWin::GetWindowList(WindowList* windows) {
@@ -196,15 +185,16 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) {
     return;
   }
 
-  // Stop capturing if the window has been closed or hidden.
-  if (!IsWindow(window_) || !IsWindowVisible(window_)) {
+  // Stop capturing if the window has been closed.
+  if (!IsWindow(window_)) {
     callback_->OnCaptureCompleted(NULL);
     return;
   }
 
-  // Return a 1x1 black frame if the window is minimized, to match the behavior
-  // on Mac.
-  if (IsIconic(window_)) {
+  // Return a 1x1 black frame if the window is minimized or invisible, to match
+  // behavior on mace. Window can be temporarily invisible during the
+  // transition of full screen mode on/off.
+  if (IsIconic(window_) || !IsWindowVisible(window_)) {
     BasicDesktopFrame* frame = new BasicDesktopFrame(DesktopSize(1, 1));
     memset(frame->data(), 0, frame->stride() * frame->size().height());
 
@@ -257,7 +247,7 @@ void WindowCapturerWin::Capture(const DesktopRegion& region) {
   // capturing - it somehow affects what we get from BitBlt() on the subsequent
   // captures.
 
-  if (!IsAeroEnabled() || !previous_size_.equals(frame->size())) {
+  if (!aero_checker_.IsAeroEnabled() || !previous_size_.equals(frame->size())) {
     result = PrintWindow(window_, mem_dc, 0);
   }
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc
index 63f3d7d3f8..50d25f033d 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc
@@ -20,13 +20,13 @@
 #include <algorithm>
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/x11/shared_x_display.h"
 #include "webrtc/modules/desktop_capture/x11/x_error_trap.h"
 #include "webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/modules/desktop_capture/x11/shared_x_util.h"
 
 namespace webrtc {
@@ -71,7 +71,7 @@ class WindowCapturerLinux : public WindowCapturer,
 
   Callback* callback_;
 
-  scoped_refptr<SharedXDisplay> x_display_;
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
 
   Atom wm_state_atom_;
   Atom window_type_atom_;
@@ -81,7 +81,7 @@ class WindowCapturerLinux : public WindowCapturer,
   ::Window selected_window_;
   XServerPixelBuffer x_server_pixel_buffer_;
 
-  DISALLOW_COPY_AND_ASSIGN(WindowCapturerLinux);
+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerLinux);
 };
 
 WindowCapturerLinux::WindowCapturerLinux(const DesktopCaptureOptions& options)
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/desktop_device_info_x11.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/desktop_device_info_x11.cc
index 631681c3b0..ee1a58cbf5 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/desktop_device_info_x11.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/desktop_device_info_x11.cc
@@ -3,9 +3,8 @@
  * You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "webrtc/modules/desktop_capture/x11/desktop_device_info_x11.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
 #include <inttypes.h>
 #include <unistd.h>
 #include <stdio.h>
@@ -51,7 +50,7 @@ void DesktopDeviceInfoX11::InitializeScreenList() {
 
 void DesktopDeviceInfoX11::InitializeApplicationList() {
   //List all running applications exclude background process.
-  scoped_refptr<SharedXDisplay> SharedDisplay = SharedXDisplay::CreateDefault();
+  rtc::scoped_refptr<SharedXDisplay> SharedDisplay = SharedXDisplay::CreateDefault();
   XErrorTrap error_trap(SharedDisplay->display());
 
   WindowUtilX11 window_util_x11(SharedDisplay);
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.cc
index 05b7f572e0..3eb5eb10a9 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.cc
@@ -12,7 +12,7 @@
 
 #include <algorithm>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
@@ -27,7 +27,7 @@ SharedXDisplay::~SharedXDisplay() {
 }
 
 // static
-scoped_refptr<SharedXDisplay> SharedXDisplay::Create(
+rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::Create(
     const std::string& display_name) {
   Display* display =
       XOpenDisplay(display_name.empty() ? NULL : display_name.c_str());
@@ -39,7 +39,7 @@ scoped_refptr<SharedXDisplay> SharedXDisplay::Create(
 }
 
 // static
-scoped_refptr<SharedXDisplay> SharedXDisplay::CreateDefault() {
+rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::CreateDefault() {
   return Create(std::string());
 }
 
@@ -64,7 +64,7 @@ void SharedXDisplay::RemoveEventHandler(int type, XEventHandler* handler) {
 void SharedXDisplay::ProcessPendingXEvents() {
   // Hold reference to |this| to prevent it from being destroyed while
   // processing events.
-  scoped_refptr<SharedXDisplay> self(this);
+  rtc::scoped_refptr<SharedXDisplay> self(this);
 
   // Find the number of events that are outstanding "now."  We don't just loop
   // on XPending because we want to guarantee this terminates.
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.h
index 81b5ef6606..d905b9e51c 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_display.h
@@ -19,8 +19,8 @@
 
 #include <string>
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
@@ -41,11 +41,12 @@ class SharedXDisplay {
   // Creates a new X11 Display for the |display_name|. NULL is returned if X11
   // connection failed. Equivalent to CreateDefault() when |display_name| is
   // empty.
-  static scoped_refptr<SharedXDisplay> Create(const std::string& display_name);
+  static rtc::scoped_refptr<SharedXDisplay> Create(
+      const std::string& display_name);
 
   // Creates X11 Display connection for the default display (e.g. specified in
   // DISPLAY). NULL is returned if X11 connection failed.
-  static scoped_refptr<SharedXDisplay> CreateDefault();
+  static rtc::scoped_refptr<SharedXDisplay> CreateDefault();
 
   void AddRef() { ++ref_count_; }
   void Release() {
@@ -75,7 +76,7 @@ class SharedXDisplay {
 
   EventHandlersMap event_handlers_;
 
-  DISALLOW_COPY_AND_ASSIGN(SharedXDisplay);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedXDisplay);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.cc
index bf064ac276..c903aa66f7 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.cc
@@ -12,7 +12,7 @@
 
 namespace webrtc {
 
-WindowUtilX11::WindowUtilX11(scoped_refptr<SharedXDisplay> x_display) {
+WindowUtilX11::WindowUtilX11(rtc::scoped_refptr<SharedXDisplay> x_display) {
   x_display_ = x_display;
   wm_state_atom_ = XInternAtom(display(), "WM_STATE", True);
   window_type_atom_ = XInternAtom(display(), "_NET_WM_WINDOW_TYPE", True);
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.h
index 2382409210..325b034139 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/shared_x_util.h
@@ -11,10 +11,9 @@
 #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_X11_SHARED_X_UTIL_H_
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_X11_SHARED_X_UTIL_H_
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 #include "webrtc/modules/desktop_capture/x11/shared_x_display.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 #include <unistd.h>
 #include <string.h>
 
@@ -88,12 +87,12 @@ private:
   unsigned long size_;  // NOLINT: type required by XGetWindowProperty
   unsigned char* data_;
 
-  DISALLOW_COPY_AND_ASSIGN(XWindowProperty);
+  RTC_DISALLOW_COPY_AND_ASSIGN(XWindowProperty);
 };
 
 class WindowUtilX11 {
 public:
-  WindowUtilX11(scoped_refptr<SharedXDisplay> x_display);
+  WindowUtilX11(rtc::scoped_refptr<SharedXDisplay> x_display);
   ~WindowUtilX11();
   // Iterates through |window| hierarchy to find first visible window, i.e. one
   // that has WM_STATE property set to NormalState.
@@ -112,7 +111,7 @@ public:
 protected:
   Display* display() { return x_display_->display(); }
 
-  scoped_refptr<SharedXDisplay> x_display_;
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
   Atom wm_state_atom_;
   Atom window_type_atom_;
   Atom normal_window_type_atom_;
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_error_trap.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_error_trap.h
index 98d5680d64..670c81343a 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_error_trap.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_error_trap.h
@@ -44,7 +44,7 @@ class XErrorTrap {
   int last_xserver_error_code_;
   bool enabled_;
 
-  DISALLOW_COPY_AND_ASSIGN(XErrorTrap);
+  RTC_DISALLOW_COPY_AND_ASSIGN(XErrorTrap);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
index be00fa7697..bcfcb7e027 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
@@ -16,7 +16,7 @@
 
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/x11/x_error_trap.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace {
 
diff --git a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
index 98f263f3a8..d1e6632f08 100644
--- a/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
@@ -77,7 +77,7 @@ class XServerPixelBuffer {
   Pixmap shm_pixmap_;
   GC shm_gc_;
 
-  DISALLOW_COPY_AND_ASSIGN(XServerPixelBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(XServerPixelBuffer);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/interface/module.h b/media/webrtc/trunk/webrtc/modules/include/module.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/interface/module.h
rename to media/webrtc/trunk/webrtc/modules/include/module.h
index a83f148cf8..d02aa95dc8 100644
--- a/media/webrtc/trunk/webrtc/modules/interface/module.h
+++ b/media/webrtc/trunk/webrtc/modules/include/module.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_INTERFACE_MODULE_H_
-#define MODULES_INTERFACE_MODULE_H_
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_H_
 
 #include "webrtc/typedefs.h"
 
@@ -64,18 +64,18 @@ class RefCountedModule : public Module {
  public:
   // Increase the reference count by one.
   // Returns the incremented reference count.
-  virtual int32_t AddRef() = 0;
+  virtual int32_t AddRef() const = 0;
 
   // Decrease the reference count by one.
   // Returns the decreased reference count.
   // Returns 0 if the last reference was just released.
   // When the reference count reaches 0 the object will self-destruct.
-  virtual int32_t Release() = 0;
+  virtual int32_t Release() const = 0;
 
  protected:
-  virtual ~RefCountedModule() {}
+  ~RefCountedModule() override = default;
 };
 
 }  // namespace webrtc
 
-#endif  // MODULES_INTERFACE_MODULE_H_
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/interface/module_common_types.h b/media/webrtc/trunk/webrtc/modules/include/module_common_types.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/interface/module_common_types.h
rename to media/webrtc/trunk/webrtc/modules/include/module_common_types.h
index 248ffc0e31..d220c5688f 100644
--- a/media/webrtc/trunk/webrtc/modules/interface/module_common_types.h
+++ b/media/webrtc/trunk/webrtc/modules/include/module_common_types.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULE_COMMON_TYPES_H
-#define MODULE_COMMON_TYPES_H
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
 
 #include <assert.h>
 #include <string.h>  // memcpy
@@ -28,7 +28,7 @@ struct RTPAudioHeader {
   uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
   uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
   bool isCNG;                         // is this CNG
-  uint8_t channel;                    // number of channels 2 = stereo
+  size_t channel;                     // number of channels 2 = stereo
 };
 
 const int16_t kNoPictureId = -1;
@@ -173,10 +173,10 @@ struct RTPVideoHeaderVP9 {
   bool ss_data_available;  // True if SS data is available in this payload
                            // descriptor.
   int16_t picture_id;      // PictureID index, 15 bits;
-                              // kNoPictureId if PictureID does not exist.
+                           // kNoPictureId if PictureID does not exist.
   int16_t max_picture_id;  // Maximum picture ID index; either 0x7F or 0x7FFF;
   int16_t tl0_pic_idx;     // TL0PIC_IDX, 8 bits;
-                              // kNoTl0PicIdx means no value provided.
+                           // kNoTl0PicIdx means no value provided.
   uint8_t temporal_idx;    // Temporal layer index, or kNoTemporalIdx.
   uint8_t spatial_idx;     // Spatial layer index, or kNoSpatialIdx.
   bool temporal_up_switch;  // True if upswitch to higher frame rate is possible
@@ -343,15 +343,17 @@ class RTPFragmentationHeader {
     }
   }
 
-  void VerifyAndAllocateFragmentationHeader(const uint16_t size) {
-    if (fragmentationVectorSize < size) {
+  void VerifyAndAllocateFragmentationHeader(const size_t size) {
+    assert(size <= std::numeric_limits<uint16_t>::max());
+    const uint16_t size16 = static_cast<uint16_t>(size);
+    if (fragmentationVectorSize < size16) {
       uint16_t oldVectorSize = fragmentationVectorSize;
       {
         // offset
         size_t* oldOffsets = fragmentationOffset;
-        fragmentationOffset = new size_t[size];
+        fragmentationOffset = new size_t[size16];
         memset(fragmentationOffset + oldVectorSize, 0,
-               sizeof(size_t) * (size - oldVectorSize));
+               sizeof(size_t) * (size16 - oldVectorSize));
         // copy old values
         memcpy(fragmentationOffset, oldOffsets,
                sizeof(size_t) * oldVectorSize);
@@ -360,9 +362,9 @@ class RTPFragmentationHeader {
       // length
       {
         size_t* oldLengths = fragmentationLength;
-        fragmentationLength = new size_t[size];
+        fragmentationLength = new size_t[size16];
         memset(fragmentationLength + oldVectorSize, 0,
-               sizeof(size_t) * (size - oldVectorSize));
+               sizeof(size_t) * (size16 - oldVectorSize));
         memcpy(fragmentationLength, oldLengths,
                sizeof(size_t) * oldVectorSize);
         delete[] oldLengths;
@@ -370,9 +372,9 @@ class RTPFragmentationHeader {
       // time diff
       {
         uint16_t* oldTimeDiffs = fragmentationTimeDiff;
-        fragmentationTimeDiff = new uint16_t[size];
+        fragmentationTimeDiff = new uint16_t[size16];
         memset(fragmentationTimeDiff + oldVectorSize, 0,
-               sizeof(uint16_t) * (size - oldVectorSize));
+               sizeof(uint16_t) * (size16 - oldVectorSize));
         memcpy(fragmentationTimeDiff, oldTimeDiffs,
                sizeof(uint16_t) * oldVectorSize);
         delete[] oldTimeDiffs;
@@ -380,14 +382,14 @@ class RTPFragmentationHeader {
       // payload type
       {
         uint8_t* oldTimePlTypes = fragmentationPlType;
-        fragmentationPlType = new uint8_t[size];
+        fragmentationPlType = new uint8_t[size16];
         memset(fragmentationPlType + oldVectorSize, 0,
-               sizeof(uint8_t) * (size - oldVectorSize));
+               sizeof(uint8_t) * (size16 - oldVectorSize));
         memcpy(fragmentationPlType, oldTimePlTypes,
                sizeof(uint8_t) * oldVectorSize);
         delete[] oldTimePlTypes;
       }
-      fragmentationVectorSize = size;
+      fragmentationVectorSize = size16;
     }
   }
 
@@ -400,7 +402,7 @@ class RTPFragmentationHeader {
   uint8_t* fragmentationPlType;      // Payload type of each fragmentation
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader);
 };
 
 struct RTCPVoIPMetric {
@@ -449,103 +451,11 @@ struct FecProtectionParams {
 // CallStats object using RegisterStatsObserver.
 class CallStatsObserver {
  public:
-  virtual void OnRttUpdate(int64_t rtt_ms) = 0;
+  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
 
   virtual ~CallStatsObserver() {}
 };
 
-// class describing a complete, or parts of an encoded frame.
-class EncodedVideoData {
- public:
-  EncodedVideoData()
-      : payloadType(0),
-        timeStamp(0),
-        renderTimeMs(0),
-        encodedWidth(0),
-        encodedHeight(0),
-        completeFrame(false),
-        missingFrame(false),
-        payloadData(NULL),
-        payloadSize(0),
-        bufferSize(0),
-        fragmentationHeader(),
-        frameType(kVideoFrameDelta),
-        codec(kVideoCodecUnknown) {};
-
-  EncodedVideoData(const EncodedVideoData& data) {
-    payloadType = data.payloadType;
-    timeStamp = data.timeStamp;
-    renderTimeMs = data.renderTimeMs;
-    encodedWidth = data.encodedWidth;
-    encodedHeight = data.encodedHeight;
-    completeFrame = data.completeFrame;
-    missingFrame = data.missingFrame;
-    payloadSize = data.payloadSize;
-    fragmentationHeader.CopyFrom(data.fragmentationHeader);
-    frameType = data.frameType;
-    codec = data.codec;
-    if (data.payloadSize > 0) {
-      payloadData = new uint8_t[data.payloadSize];
-      memcpy(payloadData, data.payloadData, data.payloadSize);
-      bufferSize = data.payloadSize;
-    } else {
-      payloadData = NULL;
-    }
-  }
-
-  ~EncodedVideoData() {
-    delete[] payloadData;
-  };
-
-  EncodedVideoData& operator=(const EncodedVideoData& data) {
-    if (this == &data) {
-      return *this;
-    }
-    payloadType = data.payloadType;
-    timeStamp = data.timeStamp;
-    renderTimeMs = data.renderTimeMs;
-    encodedWidth = data.encodedWidth;
-    encodedHeight = data.encodedHeight;
-    completeFrame = data.completeFrame;
-    missingFrame = data.missingFrame;
-    payloadSize = data.payloadSize;
-    fragmentationHeader.CopyFrom(data.fragmentationHeader);
-    frameType = data.frameType;
-    codec = data.codec;
-    if (data.payloadSize > 0) {
-      delete[] payloadData;
-      payloadData = new uint8_t[data.payloadSize];
-      memcpy(payloadData, data.payloadData, data.payloadSize);
-      bufferSize = data.payloadSize;
-    }
-    return *this;
-  };
-  void VerifyAndAllocate(const size_t size) {
-    if (bufferSize < size) {
-      uint8_t* oldPayload = payloadData;
-      payloadData = new uint8_t[size];
-      memcpy(payloadData, oldPayload, sizeof(uint8_t) * payloadSize);
-
-      bufferSize = size;
-      delete[] oldPayload;
-    }
-  }
-
-  uint8_t payloadType;
-  uint32_t timeStamp;
-  int64_t renderTimeMs;
-  uint32_t encodedWidth;
-  uint32_t encodedHeight;
-  bool completeFrame;
-  bool missingFrame;
-  uint8_t* payloadData;
-  size_t payloadSize;
-  size_t bufferSize;
-  RTPFragmentationHeader fragmentationHeader;
-  FrameType frameType;
-  VideoCodecType codec;
-};
-
 struct VideoContentMetrics {
   VideoContentMetrics()
       : motion_magnitude(0.0f),
@@ -581,7 +491,7 @@ struct VideoContentMetrics {
 class AudioFrame {
  public:
   // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
-  static const int kMaxDataSizeSamples = 3840;
+  static const size_t kMaxDataSizeSamples = 3840;
 
   enum VADActivity {
     kVadActive = 0,
@@ -605,9 +515,9 @@ class AudioFrame {
 
   // |interleaved_| is not changed by this method.
   void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
-                   int samples_per_channel, int sample_rate_hz,
+                   size_t samples_per_channel, int sample_rate_hz,
                    SpeechType speech_type, VADActivity vad_activity,
-                   int num_channels = 1, uint32_t energy = -1);
+                   size_t num_channels = 1, uint32_t energy = -1);
 
   AudioFrame& Append(const AudioFrame& rhs);
 
@@ -629,9 +539,9 @@ class AudioFrame {
   // -1 represents an uninitialized value.
   int64_t ntp_time_ms_;
   int16_t data_[kMaxDataSizeSamples];
-  int samples_per_channel_;
+  size_t samples_per_channel_;
   int sample_rate_hz_;
-  int num_channels_;
+  size_t num_channels_;
   SpeechType speech_type_;
   VADActivity vad_activity_;
   // Note that there is no guarantee that |energy_| is correct. Any user of this
@@ -642,7 +552,7 @@ class AudioFrame {
   bool interleaved_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(AudioFrame);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
 };
 
 inline AudioFrame::AudioFrame()
@@ -666,11 +576,14 @@ inline void AudioFrame::Reset() {
   interleaved_ = true;
 }
 
-inline void AudioFrame::UpdateFrame(int id, uint32_t timestamp,
+inline void AudioFrame::UpdateFrame(int id,
+                                    uint32_t timestamp,
                                     const int16_t* data,
-                                    int samples_per_channel, int sample_rate_hz,
+                                    size_t samples_per_channel,
+                                    int sample_rate_hz,
                                     SpeechType speech_type,
-                                    VADActivity vad_activity, int num_channels,
+                                    VADActivity vad_activity,
+                                    size_t num_channels,
                                     uint32_t energy) {
   id_ = id;
   timestamp_ = timestamp;
@@ -681,8 +594,8 @@ inline void AudioFrame::UpdateFrame(int id, uint32_t timestamp,
   num_channels_ = num_channels;
   energy_ = energy;
 
-  const int length = samples_per_channel * num_channels;
-  assert(length <= kMaxDataSizeSamples && length >= 0);
+  const size_t length = samples_per_channel * num_channels;
+  assert(length <= kMaxDataSizeSamples);
   if (data != NULL) {
     memcpy(data_, data, sizeof(int16_t) * length);
   } else {
@@ -705,8 +618,8 @@ inline void AudioFrame::CopyFrom(const AudioFrame& src) {
   energy_ = src.energy_;
   interleaved_ = src.interleaved_;
 
-  const int length = samples_per_channel_ * num_channels_;
-  assert(length <= kMaxDataSizeSamples && length >= 0);
+  const size_t length = samples_per_channel_ * num_channels_;
+  assert(length <= kMaxDataSizeSamples);
   memcpy(data_, src.data_, sizeof(int16_t) * length);
 }
 
@@ -718,7 +631,7 @@ inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
   assert((num_channels_ > 0) && (num_channels_ < 3));
   if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
 
-  for (int i = 0; i < samples_per_channel_ * num_channels_; i++) {
+  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
     data_[i] = static_cast<int16_t>(data_[i] >> rhs);
   }
   return *this;
@@ -740,8 +653,8 @@ inline AudioFrame& AudioFrame::Append(const AudioFrame& rhs) {
     speech_type_ = kUndefined;
   }
 
-  int offset = samples_per_channel_ * num_channels_;
-  for (int i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) {
+  size_t offset = samples_per_channel_ * num_channels_;
+  for (size_t i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) {
     data_[offset + i] = rhs.data_[i];
   }
   samples_per_channel_ += rhs.samples_per_channel_;
@@ -791,7 +704,7 @@ inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
            sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
   } else {
     // IMPROVEMENT this can be done very fast in assembly
-    for (int i = 0; i < samples_per_channel_ * num_channels_; i++) {
+    for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
       int32_t wrap_guard =
           static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
       data_[i] = ClampToInt16(wrap_guard);
@@ -816,7 +729,7 @@ inline AudioFrame& AudioFrame::operator-=(const AudioFrame& rhs) {
   }
   speech_type_ = kUndefined;
 
-  for (int i = 0; i < samples_per_channel_ * num_channels_; i++) {
+  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
     int32_t wrap_guard =
         static_cast<int32_t>(data_[i]) - static_cast<int32_t>(rhs.data_[i]);
     data_[i] = ClampToInt16(wrap_guard);
@@ -848,7 +761,7 @@ inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
   return timestamp != prev_timestamp &&
          static_cast<uint32_t>(timestamp - prev_timestamp) < 0x80000000;
 }
-
+ 
 inline bool IsNewerOrSameTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
   return timestamp == prev_timestamp ||
       static_cast<uint32_t>(timestamp - prev_timestamp) < 0x80000000;
@@ -907,4 +820,4 @@ class SequenceNumberUnwrapper {
 
 }  // namespace webrtc
 
-#endif  // MODULE_COMMON_TYPES_H
+#endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/BUILD.gn b/media/webrtc/trunk/webrtc/modules/media_file/BUILD.gn
index 05cfb4e555..2a4be728f3 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/media_file/BUILD.gn
@@ -14,12 +14,12 @@ config("media_file_config") {
 
 source_set("media_file") {
   sources = [
-    "interface/media_file.h",
-    "interface/media_file_defines.h",
-    "source/media_file_impl.cc",
-    "source/media_file_impl.h",
-    "source/media_file_utility.cc",
-    "source/media_file_utility.h",
+    "media_file.h",
+    "media_file_defines.h",
+    "media_file_impl.cc",
+    "media_file_impl.h",
+    "media_file_utility.cc",
+    "media_file_utility.h",
   ]
 
   if (is_win) {
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/OWNERS b/media/webrtc/trunk/webrtc/modules/media_file/OWNERS
index a5d3cf6ce2..f6467a4161 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/media_file/OWNERS
@@ -2,4 +2,9 @@ mflodman@webrtc.org
 perkj@webrtc.org
 niklas.enbom@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/media_file.gypi b/media/webrtc/trunk/webrtc/modules/media_file/media_file.gypi
index 4ec80c3c52..94a99a22f1 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/media_file.gypi
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file.gypi
@@ -17,12 +17,12 @@
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       ],
       'sources': [
-        'interface/media_file.h',
-        'interface/media_file_defines.h',
-        'source/media_file_impl.cc',
-        'source/media_file_impl.h',
-        'source/media_file_utility.cc',
-        'source/media_file_utility.h',
+        'media_file.h',
+        'media_file_defines.h',
+        'media_file_impl.cc',
+        'media_file_impl.h',
+        'media_file_utility.cc',
+        'media_file_utility.h',
       ], # source
       # TODO(jschuh): Bug 1348: fix size_t to int truncations.
       'msvs_disabled_warnings': [ 4267, ],
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/interface/media_file.h b/media/webrtc/trunk/webrtc/modules/media_file/media_file.h
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/media_file/interface/media_file.h
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file.h
index 5b09ad4383..f6924d6bb0 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/interface/media_file.h
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -177,4 +177,4 @@ protected:
     virtual ~MediaFile() {}
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/interface/media_file_defines.h b/media/webrtc/trunk/webrtc/modules/media_file/media_file_defines.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/media_file/interface/media_file_defines.h
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file_defines.h
index ded71a8ca7..a021a148a5 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/interface/media_file_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file_defines.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -48,4 +48,4 @@ protected:
     FileCallback() {}
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.cc b/media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.cc
similarity index 99%
rename from media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.cc
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.cc
index 83bb9b5355..abc7b9d9e0 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.cc
@@ -11,11 +11,11 @@
 #include <assert.h>
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/modules/media_file/source/media_file_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/media_file/media_file_impl.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 MediaFile* MediaFile::CreateMediaFile(const int32_t id)
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.h b/media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.h
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.h
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.h
index c5038bbdf5..c23f514c75 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file_impl.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/media_file/source/media_file_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 class MediaFileImpl : public MediaFile
@@ -145,4 +145,4 @@ private:
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_unittest.cc b/media/webrtc/trunk/webrtc/modules/media_file/media_file_unittest.cc
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/media_file/source/media_file_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file_unittest.cc
index ea6b953d93..6541a8fb7c 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file_unittest.cc
@@ -9,10 +9,9 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 class MediaFileTest : public testing::Test {
  protected:
@@ -28,7 +27,14 @@ class MediaFileTest : public testing::Test {
   webrtc::MediaFile* media_file_;
 };
 
-TEST_F(MediaFileTest, DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError)) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  DISABLED_StartPlayingAudioFileWithoutError
+#else
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  StartPlayingAudioFileWithoutError
+#endif
+TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
   // TODO(leozwang): Use hard coded filename here, we want to
   // loop through all audio files in future
   const std::string audio_file = webrtc::test::ProjectRootPath() +
@@ -46,7 +52,12 @@ TEST_F(MediaFileTest, DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError)) {
   ASSERT_EQ(0, media_file_->StopPlaying());
 }
 
-TEST_F(MediaFileTest, WriteWavFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteWavFile DISABLED_WriteWavFile
+#else
+#define MAYBE_WriteWavFile WriteWavFile
+#endif
+TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
   // Write file.
   static const size_t kHeaderSize = 44;
   static const size_t kPayloadSize = 320;
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.cc b/media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.cc
similarity index 83%
rename from media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.cc
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.cc
index aadf8e19ba..6c435c9590 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.cc
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/media_file/source/media_file_utility.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
 
 #include <assert.h>
 #include <sys/stat.h>
@@ -19,9 +19,9 @@
 #include "webrtc/common_audio/wav_header.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace {
 
@@ -38,8 +38,8 @@ struct WAVE_RIFF_header
 // the chunk size (16, 18 or 40 byte)
 struct WAVE_CHUNK_header
 {
-   int8_t  fmt_ckID[4];
-   int32_t fmt_ckSize;
+   int8_t   fmt_ckID[4];
+   uint32_t fmt_ckSize;
 };
 }  // unnamed namespace
 
@@ -79,15 +79,15 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
     // TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
     char tmpStr[6] = "FOUR";
     unsigned char tmpStr2[4];
-    int32_t i, len;
+    size_t i;
     bool dataFound = false;
     bool fmtFound = false;
     int8_t dummyRead;
 
 
     _dataSize = 0;
-    len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
-    if(len != sizeof(WAVE_RIFF_header))
+    int len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
+    if (len != static_cast<int>(sizeof(WAVE_RIFF_header)))
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
                      "Not a wave file (too short)");
@@ -123,14 +123,13 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
     //                 in a subroutine.
     memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
     CHUNKheaderObj.fmt_ckSize =
-        (int32_t) ((uint32_t) tmpStr2[0] +
-                         (((uint32_t)tmpStr2[1])<<8) +
-                         (((uint32_t)tmpStr2[2])<<16) +
-                         (((uint32_t)tmpStr2[3])<<24));
+        (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
+        (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
 
     memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
 
-    while ((len == sizeof(WAVE_CHUNK_header)) && (!fmtFound || !dataFound))
+    while ((len == static_cast<int>(sizeof(WAVE_CHUNK_header))) &&
+           (!fmtFound || !dataFound))
     {
         if(strcmp(tmpStr, "fmt ") == 0)
         {
@@ -164,9 +163,14 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
                 (int16_t) ((uint32_t)tmpStr2[0] +
                                  (((uint32_t)tmpStr2[1])<<8));
 
+            if (CHUNKheaderObj.fmt_ckSize < sizeof(WAVE_FMTINFO_header))
+            {
+                WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+                             "Chunk size is too small");
+                return -1;
+            }
             for (i = 0;
-                 i < (CHUNKheaderObj.fmt_ckSize -
-                      (int32_t)sizeof(WAVE_FMTINFO_header));
+                 i < CHUNKheaderObj.fmt_ckSize - sizeof(WAVE_FMTINFO_header);
                  i++)
             {
                 len = wav.Read(&dummyRead, 1);
@@ -187,7 +191,7 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
         }
         else
         {
-            for (i = 0; i < (CHUNKheaderObj.fmt_ckSize); i++)
+            for (i = 0; i < CHUNKheaderObj.fmt_ckSize; i++)
             {
                 len = wav.Read(&dummyRead, 1);
                 if(len != 1)
@@ -203,10 +207,8 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
 
         memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
         CHUNKheaderObj.fmt_ckSize =
-            (int32_t) ((uint32_t)tmpStr2[0] +
-                             (((uint32_t)tmpStr2[1])<<8) +
-                             (((uint32_t)tmpStr2[2])<<16) +
-                             (((uint32_t)tmpStr2[3])<<24));
+            (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
+            (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
 
         memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
     }
@@ -241,35 +243,17 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
     }
 
     // Calculate the number of bytes that 10 ms of audio data correspond to.
-    if(_wavFormatObj.formatTag == kWavFormatPcm)
-    {
-        // TODO (hellner): integer division for 22050 and 11025 would yield
-        //                 the same result as the else statement. Remove those
-        //                 special cases?
-        if(_wavFormatObj.nSamplesPerSec == 44100)
-        {
-            _readSizeBytes = 441 * _wavFormatObj.nChannels *
-                (_wavFormatObj.nBitsPerSample / 8);
-        } else if(_wavFormatObj.nSamplesPerSec == 22050) {
-            _readSizeBytes = 220 * _wavFormatObj.nChannels * // XXX inexact!
-                (_wavFormatObj.nBitsPerSample / 8);
-        } else if(_wavFormatObj.nSamplesPerSec == 11025) {
-            _readSizeBytes = 110 * _wavFormatObj.nChannels * // XXX inexact!
-                (_wavFormatObj.nBitsPerSample / 8);
-        } else {
-            _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
-              _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
-        }
-
-    } else {
-        _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
-            _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
-    }
+    size_t samples_per_10ms =
+        ((_wavFormatObj.formatTag == kWavFormatPcm) &&
+         (_wavFormatObj.nSamplesPerSec == 44100)) ?
+        440 : static_cast<size_t>(_wavFormatObj.nSamplesPerSec / 100);
+    _readSizeBytes = samples_per_10ms * _wavFormatObj.nChannels *
+        (_wavFormatObj.nBitsPerSample / 8);
     return 0;
 }
 
 int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec,
-                                        uint32_t channels,
+                                        size_t channels,
                                         uint32_t bitsPerSample,
                                         uint32_t formatTag)
 {
@@ -376,15 +360,15 @@ int32_t ModuleFileUtility::InitWavReading(InStream& wav,
     if(start > 0)
     {
         uint8_t dummy[WAV_MAX_BUFFER_SIZE];
-        int32_t readLength;
+        int readLength;
         if(_readSizeBytes <= WAV_MAX_BUFFER_SIZE)
         {
             while (_playoutPositionMs < start)
             {
                 readLength = wav.Read(dummy, _readSizeBytes);
-                if(readLength == _readSizeBytes)
+                if(readLength == static_cast<int>(_readSizeBytes))
                 {
-                    _readPos += readLength;
+                    _readPos += _readSizeBytes;
                     _playoutPositionMs += 10;
                 }
                 else // Must have reached EOF before start position!
@@ -406,7 +390,7 @@ int32_t ModuleFileUtility::InitWavReading(InStream& wav,
     {
         return -1;
     }
-    _bytesPerSample = _wavFormatObj.nBitsPerSample / 8;
+    _bytesPerSample = static_cast<size_t>(_wavFormatObj.nBitsPerSample / 8);
 
 
     _startPointInMs = start;
@@ -431,9 +415,9 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
         bufferSize);
 
     // The number of bytes that should be read from file.
-    const uint32_t totalBytesNeeded = _readSizeBytes;
+    const size_t totalBytesNeeded = _readSizeBytes;
     // The number of bytes that will be written to outData.
-    const uint32_t bytesRequested = (codec_info_.channels == 2) ?
+    const size_t bytesRequested = (codec_info_.channels == 2) ?
         totalBytesNeeded >> 1 : totalBytesNeeded;
     if(bufferSize < bytesRequested)
     {
@@ -472,7 +456,7 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
     // Output data is should be mono.
     if(codec_info_.channels == 2)
     {
-        for (uint32_t i = 0; i < bytesRequested / _bytesPerSample; i++)
+        for (size_t i = 0; i < bytesRequested / _bytesPerSample; i++)
         {
             // Sample value is the average of left and right buffer rounded to
             // closest integer value. Note samples can be either 1 or 2 byte.
@@ -490,7 +474,7 @@ int32_t ModuleFileUtility::ReadWavDataAsMono(
         }
         memcpy(outData, _tempData, bytesRequested);
     }
-    return bytesRequested;
+    return static_cast<int32_t>(bytesRequested);
 }
 
 int32_t ModuleFileUtility::ReadWavDataAsStereo(
@@ -534,10 +518,10 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
     }
 
     // The number of bytes that should be read from file.
-    const uint32_t totalBytesNeeded = _readSizeBytes;
+    const size_t totalBytesNeeded = _readSizeBytes;
     // The number of bytes that will be written to the left and the right
     // buffers.
-    const uint32_t bytesRequested = totalBytesNeeded >> 1;
+    const size_t bytesRequested = totalBytesNeeded >> 1;
     if(bufferSize < bytesRequested)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
@@ -558,7 +542,7 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
     // either 1 or 2 bytes
     if(_bytesPerSample == 1)
     {
-        for (uint32_t i = 0; i < bytesRequested; i++)
+        for (size_t i = 0; i < bytesRequested; i++)
         {
             outDataLeft[i]  = _tempData[2 * i];
             outDataRight[i] = _tempData[(2 * i) + 1];
@@ -572,35 +556,29 @@ int32_t ModuleFileUtility::ReadWavDataAsStereo(
             outDataRight);
 
         // Bytes requested to samples requested.
-        uint32_t sampleCount = bytesRequested >> 1;
-        for (uint32_t i = 0; i < sampleCount; i++)
+        size_t sampleCount = bytesRequested >> 1;
+        for (size_t i = 0; i < sampleCount; i++)
         {
             outLeft[i] = sampleData[2 * i];
             outRight[i] = sampleData[(2 * i) + 1];
         }
     } else {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
-                   "ReadWavStereoData: unsupported sample size %d!",
+                   "ReadWavStereoData: unsupported sample size %" PRIuS "!",
                    _bytesPerSample);
         assert(false);
         return -1;
     }
-    return bytesRequested;
+    return static_cast<int32_t>(bytesRequested);
 }
 
-int32_t ModuleFileUtility::ReadWavData(
-    InStream& wav,
-    uint8_t* buffer,
-    const uint32_t dataLengthInBytes)
+int32_t ModuleFileUtility::ReadWavData(InStream& wav,
+                                       uint8_t* buffer,
+                                       size_t dataLengthInBytes)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, dataLen= %ld)",
-        &wav,
-        buffer,
-        dataLengthInBytes);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, "
+                 "dataLen= %" PRIuS ")", &wav, buffer, dataLengthInBytes);
 
 
     if(buffer == NULL)
@@ -613,7 +591,7 @@ int32_t ModuleFileUtility::ReadWavData(
     // Make sure that a read won't return too few samples.
     // TODO (hellner): why not read the remaining bytes needed from the start
     //                 of the file?
-    if((_dataSize - _readPos) < (int32_t)dataLengthInBytes)
+    if(_dataSize < (_readPos + dataLengthInBytes))
     {
         // Rewind() being -1 may be due to the file not supposed to be looped.
         if(wav.Rewind() == -1)
@@ -685,8 +663,7 @@ int32_t ModuleFileUtility::InitWavWriting(OutStream& wav,
         return -1;
     }
     _writing = false;
-    uint32_t channels = (codecInst.channels == 0) ?
-        1 : codecInst.channels;
+    size_t channels = (codecInst.channels == 0) ? 1 : codecInst.channels;
 
     if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
     {
@@ -696,7 +673,8 @@ int32_t ModuleFileUtility::InitWavWriting(OutStream& wav,
         {
             return -1;
         }
-    }else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
+    }
+    else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
     {
         _bytesPerSample = 1;
         if(WriteWavHeader(wav, 8000, _bytesPerSample, channels, kWavFormatALaw,
@@ -729,15 +707,9 @@ int32_t ModuleFileUtility::WriteWavData(OutStream& out,
                                         const int8_t*  buffer,
                                         const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, dataLen= %" PRIuS
-        ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, "
+                 "dataLen= %" PRIuS ")", &out, buffer, dataLength);
 
     if(buffer == NULL)
     {
@@ -757,17 +729,17 @@ int32_t ModuleFileUtility::WriteWavData(OutStream& out,
 
 int32_t ModuleFileUtility::WriteWavHeader(
     OutStream& wav,
-    const uint32_t freq,
-    const uint32_t bytesPerSample,
-    const uint32_t channels,
-    const uint32_t format,
-    const uint32_t lengthInBytes)
+    uint32_t freq,
+    size_t bytesPerSample,
+    size_t channels,
+    uint32_t format,
+    size_t lengthInBytes)
 {
     // Frame size in bytes for 10 ms of audio.
-    const int32_t frameSize = (freq / 100) * channels;
+    const size_t frameSize = (freq / 100) * channels;
 
     // Calculate the number of full frames that the wave file contain.
-    const int32_t dataLengthInBytes = frameSize * (lengthInBytes / frameSize);
+    const size_t dataLengthInBytes = frameSize * (lengthInBytes / frameSize);
 
     uint8_t buf[kWavHeaderSize];
     webrtc::WriteWavHeader(buf, channels, freq, static_cast<WavFormat>(format),
@@ -783,8 +755,7 @@ int32_t ModuleFileUtility::UpdateWavHeader(OutStream& wav)
     {
         return -1;
     }
-    uint32_t channels = (codec_info_.channels == 0) ?
-        1 : codec_info_.channels;
+    size_t channels = (codec_info_.channels == 0) ? 1 : codec_info_.channels;
 
     if(STR_CASE_CMP(codec_info_.plname, "L16") == 0)
     {
@@ -837,22 +808,17 @@ int32_t ModuleFileUtility::ReadPreEncodedData(
     int8_t* outData,
     const size_t bufferSize)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, outData= 0x%x, "
-        "bufferSize= %" PRIuS ")",
-        &in,
-        outData,
-        bufferSize);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, "
+                 "outData= 0x%x, bufferSize= %" PRIuS ")", &in, outData,
+                 bufferSize);
 
     if(outData == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id, "output buffer NULL");
     }
 
-    uint32_t frameLen;
+    size_t frameLen;
     uint8_t buf[64];
     // Each frame has a two byte header containing the frame length.
     int32_t res = in.Read(buf, 2);
@@ -872,12 +838,9 @@ int32_t ModuleFileUtility::ReadPreEncodedData(
     frameLen = buf[0] + buf[1] * 256;
     if(bufferSize < frameLen)
     {
-        WEBRTC_TRACE(
-            kTraceError,
-            kTraceFile,
-            _id,
-            "buffer not large enough to read %d bytes of pre-encoded data!",
-            frameLen);
+        WEBRTC_TRACE(kTraceError, kTraceFile, _id,
+                     "buffer not large enough to read %" PRIuS " bytes of "
+                     "pre-encoded data!", frameLen);
         return -1;
     }
     return in.Read(outData, frameLen);
@@ -895,24 +858,19 @@ int32_t ModuleFileUtility::InitPreEncodedWriting(
     }
     _writing = true;
     _bytesWritten = 1;
-     out.Write(&_codecId, 1);
-     return 0;
+    out.Write(&_codecId, 1);
+    return 0;
 }
 
 int32_t ModuleFileUtility::WritePreEncodedData(
     OutStream& out,
-    const int8_t*  buffer,
+    const int8_t* buffer,
     const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WritePreEncodedData(out= 0x%x, inData= 0x%x, "
-        "dataLen= %" PRIuS ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WritePreEncodedData(out= 0x%x, "
+                 "inData= 0x%x, dataLen= %" PRIuS ")", &out, buffer,
+                 dataLength);
 
     if(buffer == NULL)
     {
@@ -943,15 +901,9 @@ int32_t ModuleFileUtility::InitCompressedReading(
     const uint32_t start,
     const uint32_t stop)
 {
-    WEBRTC_TRACE(
-        kTraceDebug,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::InitCompressedReading(in= 0x%x, start= %d,\
- stop= %d)",
-        &in,
-        start,
-        stop);
+    WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
+                 "ModuleFileUtility::InitCompressedReading(in= 0x%x, "
+                 "start= %d, stop= %d)", &in, start, stop);
 
 #if defined(WEBRTC_CODEC_ILBC)
     int16_t read_len = 0;
@@ -974,9 +926,8 @@ int32_t ModuleFileUtility::InitCompressedReading(
     if(cnt==64)
     {
         return -1;
-    } else {
-        buf[cnt]=0;
     }
+    buf[cnt]=0;
 
 #ifdef WEBRTC_CODEC_ILBC
     if(!strcmp("#!iLBC20\n", buf))
@@ -994,14 +945,11 @@ int32_t ModuleFileUtility::InitCompressedReading(
             while (_playoutPositionMs <= _startPointInMs)
             {
                 read_len = in.Read(buf, 38);
-                if(read_len == 38)
-                {
-                    _playoutPositionMs += 20;
-                }
-                else
+                if(read_len != 38)
                 {
                     return -1;
                 }
+                _playoutPositionMs += 20;
             }
         }
     }
@@ -1021,14 +969,11 @@ int32_t ModuleFileUtility::InitCompressedReading(
             while (_playoutPositionMs <= _startPointInMs)
             {
                 read_len = in.Read(buf, 50);
-                if(read_len == 50)
-                {
-                    _playoutPositionMs += 20;
-                }
-                else
+                if(read_len != 50)
                 {
                     return -1;
                 }
+                _playoutPositionMs += 20;
             }
         }
     }
@@ -1045,17 +990,11 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
                                               int8_t* outData,
                                               size_t bufferSize)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x, bytes=%"
-        PRIuS ")",
-        &in,
-        outData,
-        bufferSize);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x, "
+                 "bytes=%" PRIuS ")", &in, outData, bufferSize);
 
-    uint32_t bytesRead = 0;
+    int bytesRead = 0;
 
     if(! _reading)
     {
@@ -1067,8 +1006,8 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
     if((_codecId == kCodecIlbc20Ms) ||
         (_codecId == kCodecIlbc30Ms))
     {
-        uint32_t byteSize = 0;
-         if(_codecId == kCodecIlbc30Ms)
+        size_t byteSize = 0;
+        if(_codecId == kCodecIlbc30Ms)
         {
             byteSize = 50;
         }
@@ -1079,20 +1018,20 @@ int32_t ModuleFileUtility::ReadCompressedData(InStream& in,
         if(bufferSize < byteSize)
         {
             WEBRTC_TRACE(kTraceError, kTraceFile, _id,
-                           "output buffer is too short to read ILBC compressed\
- data.");
+                         "output buffer is too short to read ILBC compressed "
+                         "data.");
             assert(false);
             return -1;
         }
 
         bytesRead = in.Read(outData, byteSize);
-        if(bytesRead != byteSize)
+        if(bytesRead != static_cast<int>(byteSize))
         {
             if(!in.Rewind())
             {
                 InitCompressedReading(in, _startPointInMs, _stopPointInMs);
                 bytesRead = in.Read(outData, byteSize);
-                if(bytesRead != byteSize)
+                if(bytesRead != static_cast<int>(byteSize))
                 {
                     _reading = false;
                     return -1;
@@ -1134,9 +1073,8 @@ int32_t ModuleFileUtility::InitCompressedWriting(
     const CodecInst& codecInst)
 {
     WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
-               "ModuleFileUtility::InitCompressedWriting(out= 0x%x,\
- codecName= %s)",
-               &out, codecInst.plname);
+                 "ModuleFileUtility::InitCompressedWriting(out= 0x%x, "
+                 "codecName= %s)", &out, codecInst.plname);
 
     _writing = false;
 
@@ -1175,15 +1113,9 @@ int32_t ModuleFileUtility::WriteCompressedData(
     const int8_t* buffer,
     const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x, "
-        "dataLen= %" PRIuS ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x, "
+                 "dataLen= %" PRIuS ")", &out, buffer, dataLength);
 
     if(buffer == NULL)
     {
@@ -1202,19 +1134,12 @@ int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
                                           const uint32_t stop,
                                           uint32_t freq)
 {
-    WEBRTC_TRACE(
-        kTraceInfo,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, stop=%d,\
- freq=%d)",
-        &pcm,
-        start,
-        stop,
-        freq);
+    WEBRTC_TRACE(kTraceInfo, kTraceFile, _id,
+                 "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, "
+                 "stop=%d, freq=%d)", &pcm, start, stop, freq);
 
     int8_t dummy[320];
-    int32_t read_len;
+    int read_len;
 
     _playoutPositionMs = 0;
     _startPointInMs = start;
@@ -1259,14 +1184,11 @@ int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
         while (_playoutPositionMs < _startPointInMs)
         {
             read_len = pcm.Read(dummy, _readSizeBytes);
-            if(read_len == _readSizeBytes)
+            if(read_len != static_cast<int>(_readSizeBytes))
             {
-                _playoutPositionMs += 10;
-            }
-            else // Must have reached EOF before start position!
-            {
-                return -1;
+                return -1;  // Must have reached EOF before start position!
             }
+            _playoutPositionMs += 10;
         }
     }
     _reading = true;
@@ -1277,23 +1199,17 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
                                        int8_t* outData,
                                        size_t bufferSize)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, bufSize= %"
-        PRIuS ")",
-        &pcm,
-        outData,
-        bufferSize);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, "
+                 "bufSize= %" PRIuS ")", &pcm, outData, bufferSize);
 
     if(outData == NULL)
     {
-        WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
+        WEBRTC_TRACE(kTraceError, kTraceFile, _id, "buffer NULL");
     }
 
     // Readsize for 10ms of audio data (2 bytes per sample).
-    uint32_t bytesRequested = 2 * codec_info_.plfreq / 100;
+    size_t bytesRequested = static_cast<size_t>(2 * codec_info_.plfreq / 100);
     if(bufferSize <  bytesRequested)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
@@ -1302,8 +1218,8 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
         return -1;
     }
 
-    uint32_t bytesRead = pcm.Read(outData, bytesRequested);
-    if(bytesRead < bytesRequested)
+    int bytesRead = pcm.Read(outData, bytesRequested);
+    if(bytesRead < static_cast<int>(bytesRequested))
     {
         if(pcm.Rewind() == -1)
         {
@@ -1318,9 +1234,9 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
             }
             else
             {
-                int32_t rest = bytesRequested - bytesRead;
-                int32_t len = pcm.Read(&(outData[bytesRead]), rest);
-                if(len == rest)
+                size_t rest = bytesRequested - bytesRead;
+                int len = pcm.Read(&(outData[bytesRead]), rest);
+                if(len == static_cast<int>(rest))
                 {
                     bytesRead += len;
                 }
@@ -1332,7 +1248,7 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
             if(bytesRead <= 0)
             {
                 WEBRTC_TRACE(kTraceError, kTraceFile, _id,
-                        "ReadPCMData: Failed to rewind audio file.");
+                             "ReadPCMData: Failed to rewind audio file.");
                 return -1;
             }
         }
@@ -1341,7 +1257,7 @@ int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
     if(bytesRead <= 0)
     {
         WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
-                   "ReadPCMData: end of file");
+                     "ReadPCMData: end of file");
         return -1;
     }
     _playoutPositionMs += 10;
@@ -1412,15 +1328,9 @@ int32_t ModuleFileUtility::WritePCMData(OutStream& out,
                                         const int8_t*  buffer,
                                         const size_t dataLength)
 {
-    WEBRTC_TRACE(
-        kTraceStream,
-        kTraceFile,
-        _id,
-        "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, dataLen= %" PRIuS
-        ")",
-        &out,
-        buffer,
-        dataLength);
+    WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
+                 "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, "
+                 "dataLen= %" PRIuS ")", &out, buffer, dataLength);
 
     if(buffer == NULL)
     {
@@ -1583,7 +1493,7 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName,
         case kFileFormatCompressedFile:
         {
             int32_t cnt = 0;
-            int32_t read_len = 0;
+            int read_len = 0;
             char buf[64];
             do
             {
@@ -1640,15 +1550,8 @@ int32_t ModuleFileUtility::FileDurationMs(const char* fileName,
 uint32_t ModuleFileUtility::PlayoutPositionMs()
 {
     WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
-               "ModuleFileUtility::PlayoutPosition()");
+                 "ModuleFileUtility::PlayoutPosition()");
 
-    if(_reading)
-    {
-        return _playoutPositionMs;
-    }
-    else
-    {
-        return 0;
-    }
+    return _reading ? _playoutPositionMs : 0;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.h b/media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.h
similarity index 93%
rename from media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.h
rename to media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.h
index 2823ceca8a..bc2fa5a2f0 100644
--- a/media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.h
+++ b/media/webrtc/trunk/webrtc/modules/media_file/media_file_utility.h
@@ -9,13 +9,13 @@
  */
 
 // Note: the class cannot be used for reading and writing at the same time.
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
 
 #include <stdio.h>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
 
 namespace webrtc {
 class InStream;
@@ -176,11 +176,11 @@ public:
 
 private:
     // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
-    enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
+    static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
 
 
     int32_t InitWavCodec(uint32_t samplesPerSec,
-                         uint32_t channels,
+                         size_t channels,
                          uint32_t bitsPerSample,
                          uint32_t formatTag);
 
@@ -194,16 +194,16 @@ private:
     // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
     // lengthInBytes is the number of bytes the audio samples are using up.
     int32_t WriteWavHeader(OutStream& stream,
-                           const uint32_t freqInHz,
-                           const uint32_t bytesPerSample,
-                           const uint32_t channels,
-                           const uint32_t format,
-                           const uint32_t lengthInBytes);
+                           uint32_t freqInHz,
+                           size_t bytesPerSample,
+                           size_t channels,
+                           uint32_t format,
+                           size_t lengthInBytes);
 
     // Put dataLengthInBytes of audio data from stream into the audioBuffer.
     // The return value is the number of bytes written to audioBuffer.
     int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
-                        const uint32_t dataLengthInBytes);
+                        size_t dataLengthInBytes);
 
     // Update the current audio codec being used for reading or writing
     // according to codecInst.
@@ -254,10 +254,10 @@ private:
     // TODO (hellner): why store multiple formats. Just store either codec_info_
     //                 or _wavFormatObj and supply conversion functions.
     WAVE_FMTINFO_header _wavFormatObj;
-    int32_t _dataSize;      // Chunk size if reading a WAV file
+    size_t _dataSize;      // Chunk size if reading a WAV file
     // Number of bytes to read. I.e. frame size in bytes. May be multiple
     // chunks if reading WAV.
-    int32_t _readSizeBytes;
+    size_t _readSizeBytes;
 
     int32_t _id;
 
@@ -270,8 +270,8 @@ private:
     MediaFileUtility_CodecType _codecId;
 
     // The amount of bytes, on average, used for one audio sample.
-    int32_t  _bytesPerSample;
-    int32_t  _readPos;
+    size_t _bytesPerSample;
+    size_t _readPos;
 
     // Only reading or writing can be enabled, not both.
     bool _reading;
@@ -281,4 +281,4 @@ private:
     uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/module_common_types_unittest.cc b/media/webrtc/trunk/webrtc/modules/module_common_types_unittest.cc
index ee23bf020e..acd58476a1 100644
--- a/media/webrtc/trunk/webrtc/modules/module_common_types_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/module_common_types_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -38,6 +38,11 @@ TEST(IsNewerSequenceNumber, BackwardWrap) {
   EXPECT_FALSE(IsNewerSequenceNumber(0xFF00, 0x00FF));
 }
 
+TEST(IsNewerSequenceNumber, HalfWayApart) {
+  EXPECT_TRUE(IsNewerSequenceNumber(0x8000, 0x0000));
+  EXPECT_FALSE(IsNewerSequenceNumber(0x0000, 0x8000));
+}
+
 TEST(IsNewerTimestamp, Equal) {
   EXPECT_FALSE(IsNewerTimestamp(0x00000001, 0x000000001));
 }
@@ -62,6 +67,11 @@ TEST(IsNewerTimestamp, BackwardWrap) {
   EXPECT_FALSE(IsNewerTimestamp(0xFFFF0000, 0x0000FFFF));
 }
 
+TEST(IsNewerTimestamp, HalfWayApart) {
+  EXPECT_TRUE(IsNewerTimestamp(0x80000000, 0x00000000));
+  EXPECT_FALSE(IsNewerTimestamp(0x00000000, 0x80000000));
+}
+
 TEST(LatestSequenceNumber, NoWrap) {
   EXPECT_EQ(0xFFFFu, LatestSequenceNumber(0xFFFF, 0xFFFE));
   EXPECT_EQ(0x0001u, LatestSequenceNumber(0x0001, 0x0000));
@@ -101,4 +111,74 @@ TEST(LatestTimestamp, Wrap) {
   EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFFFFFF, 0x0000FFFF));
   EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFF0000, 0x0000FFFF));
 }
+
+TEST(ClampToInt16, TestCases) {
+  EXPECT_EQ(0x0000, ClampToInt16(0x00000000));
+  EXPECT_EQ(0x0001, ClampToInt16(0x00000001));
+  EXPECT_EQ(0x7FFF, ClampToInt16(0x00007FFF));
+  EXPECT_EQ(0x7FFF, ClampToInt16(0x7FFFFFFF));
+  EXPECT_EQ(-0x0001, ClampToInt16(-0x00000001));
+  EXPECT_EQ(-0x8000, ClampToInt16(-0x8000));
+  EXPECT_EQ(-0x8000, ClampToInt16(-0x7FFFFFFF));
+}
+
+TEST(SequenceNumberUnwrapper, Limits) {
+  SequenceNumberUnwrapper unwrapper;
+
+  EXPECT_EQ(0, unwrapper.Unwrap(0));
+  EXPECT_EQ(0x8000, unwrapper.Unwrap(0x8000));
+  // Delta is exactly 0x8000 but current is lower than input, wrap backwards.
+  EXPECT_EQ(0x0, unwrapper.Unwrap(0x0000));
+
+  EXPECT_EQ(0x8000, unwrapper.Unwrap(0x8000));
+  EXPECT_EQ(0xFFFF, unwrapper.Unwrap(0xFFFF));
+  EXPECT_EQ(0x10000, unwrapper.Unwrap(0x0000));
+  EXPECT_EQ(0xFFFF, unwrapper.Unwrap(0xFFFF));
+  EXPECT_EQ(0x8000, unwrapper.Unwrap(0x8000));
+  EXPECT_EQ(0, unwrapper.Unwrap(0));
+
+  // Don't allow negative values.
+  EXPECT_EQ(0xFFFF, unwrapper.Unwrap(0xFFFF));
+}
+
+TEST(SequenceNumberUnwrapper, ForwardWraps) {
+  int64_t seq = 0;
+  SequenceNumberUnwrapper unwrapper;
+
+  const int kMaxIncrease = 0x8000 - 1;
+  const int kNumWraps = 4;
+  for (int i = 0; i < kNumWraps * 2; ++i) {
+    int64_t unwrapped = unwrapper.Unwrap(static_cast<uint16_t>(seq & 0xFFFF));
+    EXPECT_EQ(seq, unwrapped);
+    seq += kMaxIncrease;
+  }
+
+  unwrapper.UpdateLast(0);
+  for (int seq = 0; seq < kNumWraps * 0xFFFF; ++seq) {
+    int64_t unwrapped = unwrapper.Unwrap(static_cast<uint16_t>(seq & 0xFFFF));
+    EXPECT_EQ(seq, unwrapped);
+  }
+}
+
+TEST(SequenceNumberUnwrapper, BackwardWraps) {
+  SequenceNumberUnwrapper unwrapper;
+
+  const int kMaxDecrease = 0x8000 - 1;
+  const int kNumWraps = 4;
+  int64_t seq = kNumWraps * 2 * kMaxDecrease;
+  unwrapper.UpdateLast(seq);
+  for (int i = kNumWraps * 2; i >= 0; --i) {
+    int64_t unwrapped = unwrapper.Unwrap(static_cast<uint16_t>(seq & 0xFFFF));
+    EXPECT_EQ(seq, unwrapped);
+    seq -= kMaxDecrease;
+  }
+
+  seq = kNumWraps * 0xFFFF;
+  unwrapper.UpdateLast(seq);
+  for (; seq >= 0; --seq) {
+    int64_t unwrapped = unwrapper.Unwrap(static_cast<uint16_t>(seq & 0xFFFF));
+    EXPECT_EQ(seq, unwrapped);
+  }
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/modules.gyp b/media/webrtc/trunk/webrtc/modules/modules.gyp
index c004d5a3f5..a7dc0ffe53 100644
--- a/media/webrtc/trunk/webrtc/modules/modules.gyp
+++ b/media/webrtc/trunk/webrtc/modules/modules.gyp
@@ -20,7 +20,8 @@
     'remote_bitrate_estimator/remote_bitrate_estimator.gypi',
     'rtp_rtcp/rtp_rtcp.gypi',
     'utility/utility.gypi',
-    'video_coding/codecs/i420/main/source/i420.gypi',
+    'video_coding/codecs/h264/h264.gypi',
+    'video_coding/codecs/i420/i420.gypi',
     'video_coding/video_coding.gypi',
     'video_capture/video_capture.gypi',
     'video_processing/video_processing.gypi',
@@ -47,330 +48,6 @@
         ],
       },
       'targets': [
-        {
-          'target_name': 'modules_unittests',
-          'type': '<(gtest_target_type)',
-          'defines': [
-            '<@(audio_coding_defines)',
-          ],
-          'dependencies': [
-            'acm_receive_test',
-            'acm_send_test',
-            'audio_coding_module',
-            'audio_device'  ,
-            'audio_processing',
-            'bitrate_controller',
-            'CNG',
-            'desktop_capture',
-            'iSACFix',
-            'media_file',
-            'neteq',
-            'neteq_test_support',
-            'neteq_unittest_tools',
-            'paced_sender',
-            'PCM16B',  # Needed by NetEq tests.
-            'red',
-            'remote_bitrate_estimator',
-            'rtp_rtcp',
-            'video_codecs_test_framework',
-            'video_processing',
-            'webrtc_utility',
-            'webrtc_video_coding',
-            '<@(neteq_dependencies)',
-            '<(DEPTH)/testing/gmock.gyp:gmock',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-            '<(webrtc_root)/modules/modules.gyp:video_capture',
-            '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
-            '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/test/test.gyp:frame_generator',
-            '<(webrtc_root)/test/test.gyp:rtp_test_utils',
-            '<(webrtc_root)/test/test.gyp:test_support_main',
-            '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
-          ],
-          'sources': [
-            'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
-            'audio_coding/main/acm2/acm_generic_codec_test.cc',
-            'audio_coding/main/acm2/acm_generic_codec_opus_test.cc',
-            'audio_coding/main/acm2/acm_receiver_unittest.cc',
-            'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc',
-            'audio_coding/main/acm2/audio_coding_module_unittest.cc',
-            'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc',
-            'audio_coding/main/acm2/call_statistics_unittest.cc',
-            'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
-            'audio_coding/main/acm2/nack_unittest.cc',
-            'audio_coding/codecs/cng/cng_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
-            'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
-            'audio_coding/codecs/isac/main/source/isac_unittest.cc',
-            'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
-            'audio_coding/codecs/opus/opus_unittest.cc',
-            'audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc',
-            'audio_coding/neteq/audio_classifier_unittest.cc',
-            'audio_coding/neteq/audio_multi_vector_unittest.cc',
-            'audio_coding/neteq/audio_vector_unittest.cc',
-            'audio_coding/neteq/background_noise_unittest.cc',
-            'audio_coding/neteq/buffer_level_filter_unittest.cc',
-            'audio_coding/neteq/comfort_noise_unittest.cc',
-            'audio_coding/neteq/decision_logic_unittest.cc',
-            'audio_coding/neteq/decoder_database_unittest.cc',
-            'audio_coding/neteq/delay_manager_unittest.cc',
-            'audio_coding/neteq/delay_peak_detector_unittest.cc',
-            'audio_coding/neteq/dsp_helper_unittest.cc',
-            'audio_coding/neteq/dtmf_buffer_unittest.cc',
-            'audio_coding/neteq/dtmf_tone_generator_unittest.cc',
-            'audio_coding/neteq/expand_unittest.cc',
-            'audio_coding/neteq/merge_unittest.cc',
-            'audio_coding/neteq/neteq_external_decoder_unittest.cc',
-            'audio_coding/neteq/neteq_impl_unittest.cc',
-            'audio_coding/neteq/neteq_network_stats_unittest.cc',
-            'audio_coding/neteq/neteq_stereo_unittest.cc',
-            'audio_coding/neteq/neteq_unittest.cc',
-            'audio_coding/neteq/normal_unittest.cc',
-            'audio_coding/neteq/packet_buffer_unittest.cc',
-            'audio_coding/neteq/payload_splitter_unittest.cc',
-            'audio_coding/neteq/post_decode_vad_unittest.cc',
-            'audio_coding/neteq/random_vector_unittest.cc',
-            'audio_coding/neteq/sync_buffer_unittest.cc',
-            'audio_coding/neteq/timestamp_scaler_unittest.cc',
-            'audio_coding/neteq/time_stretch_unittest.cc',
-            'audio_coding/neteq/mock/mock_audio_decoder.h',
-            'audio_coding/neteq/mock/mock_audio_vector.h',
-            'audio_coding/neteq/mock/mock_buffer_level_filter.h',
-            'audio_coding/neteq/mock/mock_decoder_database.h',
-            'audio_coding/neteq/mock/mock_delay_manager.h',
-            'audio_coding/neteq/mock/mock_delay_peak_detector.h',
-            'audio_coding/neteq/mock/mock_dtmf_buffer.h',
-            'audio_coding/neteq/mock/mock_dtmf_tone_generator.h',
-            'audio_coding/neteq/mock/mock_expand.h',
-            'audio_coding/neteq/mock/mock_external_decoder_pcm16b.h',
-            'audio_coding/neteq/mock/mock_packet_buffer.h',
-            'audio_coding/neteq/mock/mock_payload_splitter.h',
-            'audio_coding/neteq/tools/input_audio_file_unittest.cc',
-            'audio_coding/neteq/tools/packet_unittest.cc',
-            'audio_processing/aec/echo_cancellation_unittest.cc',
-            'audio_processing/aec/system_delay_unittest.cc',
-            # TODO(ajm): Fix to match new interface.
-            # 'audio_processing/agc/agc_unittest.cc',
-            'audio_processing/agc/agc_audio_proc_unittest.cc',
-            'audio_processing/agc/circular_buffer_unittest.cc',
-            'audio_processing/agc/gmm_unittest.cc',
-            'audio_processing/agc/histogram_unittest.cc',
-            'audio_processing/agc/mock_agc.h',
-            'audio_processing/agc/pitch_based_vad_unittest.cc',
-            'audio_processing/agc/pitch_internal_unittest.cc',
-            'audio_processing/agc/pole_zero_filter_unittest.cc',
-            'audio_processing/agc/standalone_vad_unittest.cc',
-            'audio_processing/beamformer/complex_matrix_unittest.cc',
-            'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
-            'audio_processing/beamformer/matrix_unittest.cc',
-            'audio_processing/beamformer/mock_nonlinear_beamformer.cc',
-            'audio_processing/beamformer/mock_nonlinear_beamformer.h',
-            'audio_processing/beamformer/pcm_utils.cc',
-            'audio_processing/beamformer/pcm_utils.h',
-            'audio_processing/echo_cancellation_impl_unittest.cc',
-            'audio_processing/splitting_filter_unittest.cc',
-            'audio_processing/transient/dyadic_decimator_unittest.cc',
-            'audio_processing/transient/file_utils.cc',
-            'audio_processing/transient/file_utils.h',
-            'audio_processing/transient/file_utils_unittest.cc',
-            'audio_processing/transient/moving_moments_unittest.cc',
-            'audio_processing/transient/transient_detector_unittest.cc',
-            'audio_processing/transient/transient_suppressor_unittest.cc',
-            'audio_processing/transient/wpd_node_unittest.cc',
-            'audio_processing/transient/wpd_tree_unittest.cc',
-            'audio_processing/utility/delay_estimator_unittest.cc',
-            'bitrate_controller/bitrate_allocator_unittest.cc',
-            'bitrate_controller/bitrate_controller_unittest.cc',
-            'bitrate_controller/send_side_bandwidth_estimation_unittest.cc',
-            'bitrate_controller/send_time_history_unittest.cc',
-            'desktop_capture/desktop_and_cursor_composer_unittest.cc',
-            'desktop_capture/desktop_region_unittest.cc',
-            'desktop_capture/differ_block_unittest.cc',
-            'desktop_capture/differ_unittest.cc',
-            'desktop_capture/mouse_cursor_monitor_unittest.cc',
-            'desktop_capture/screen_capturer_helper_unittest.cc',
-            'desktop_capture/screen_capturer_mac_unittest.cc',
-            'desktop_capture/screen_capturer_mock_objects.h',
-            'desktop_capture/screen_capturer_unittest.cc',
-            'desktop_capture/window_capturer_unittest.cc',
-            'desktop_capture/win/cursor_unittest.cc',
-            'desktop_capture/win/cursor_unittest_resources.h',
-            'desktop_capture/win/cursor_unittest_resources.rc',
-            'media_file/source/media_file_unittest.cc',
-            'module_common_types_unittest.cc',
-            'pacing/bitrate_prober_unittest.cc',
-            'pacing/paced_sender_unittest.cc',
-            'pacing/packet_router_unittest.cc',
-            'remote_bitrate_estimator/bwe_simulations.cc',
-            'remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h',
-            'remote_bitrate_estimator/inter_arrival_unittest.cc',
-            'remote_bitrate_estimator/overuse_detector_unittest.cc',
-            'remote_bitrate_estimator/rate_statistics_unittest.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
-            'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
-            'remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
-            'remote_bitrate_estimator/test/bwe.cc',
-            'remote_bitrate_estimator/test/bwe.h',
-            'remote_bitrate_estimator/test/bwe_test.cc',
-            'remote_bitrate_estimator/test/bwe_test.h',
-            'remote_bitrate_estimator/test/bwe_test_baselinefile.cc',
-            'remote_bitrate_estimator/test/bwe_test_baselinefile.h',
-            'remote_bitrate_estimator/test/bwe_test_fileutils.cc',
-            'remote_bitrate_estimator/test/bwe_test_fileutils.h',
-            'remote_bitrate_estimator/test/bwe_test_framework.cc',
-            'remote_bitrate_estimator/test/bwe_test_framework.h',
-            'remote_bitrate_estimator/test/bwe_test_framework_unittest.cc',
-            'remote_bitrate_estimator/test/bwe_test_logging.cc',
-            'remote_bitrate_estimator/test/bwe_test_logging.h',
-            'remote_bitrate_estimator/test/packet_receiver.cc',
-            'remote_bitrate_estimator/test/packet_receiver.h',
-            'remote_bitrate_estimator/test/packet_sender.cc',
-            'remote_bitrate_estimator/test/packet_sender.h',
-            'remote_bitrate_estimator/test/packet.h',
-            'remote_bitrate_estimator/test/estimators/nada.cc',
-            'remote_bitrate_estimator/test/estimators/nada.h',
-            'remote_bitrate_estimator/test/estimators/remb.cc',
-            'remote_bitrate_estimator/test/estimators/remb.h',
-            'remote_bitrate_estimator/test/estimators/send_side.cc',
-            'remote_bitrate_estimator/test/estimators/send_side.h',
-            'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
-            'rtp_rtcp/source/byte_io_unittest.cc',
-            'rtp_rtcp/source/fec_receiver_unittest.cc',
-            'rtp_rtcp/source/fec_test_helper.cc',
-            'rtp_rtcp/source/fec_test_helper.h',
-            'rtp_rtcp/source/nack_rtx_unittest.cc',
-            'rtp_rtcp/source/producer_fec_unittest.cc',
-            'rtp_rtcp/source/receive_statistics_unittest.cc',
-            'rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc',
-            'rtp_rtcp/source/rtcp_format_remb_unittest.cc',
-            'rtp_rtcp/source/rtcp_packet_unittest.cc',
-            'rtp_rtcp/source/rtcp_receiver_unittest.cc',
-            'rtp_rtcp/source/rtcp_sender_unittest.cc',
-            'rtp_rtcp/source/rtcp_utility_unittest.cc',
-            'rtp_rtcp/source/rtp_fec_unittest.cc',
-            'rtp_rtcp/source/rtp_format_h264_unittest.cc',
-            'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
-            'rtp_rtcp/source/rtp_format_vp8_test_helper.cc',
-            'rtp_rtcp/source/rtp_format_vp8_test_helper.h',
-            'rtp_rtcp/source/rtp_packet_history_unittest.cc',
-            'rtp_rtcp/source/rtp_payload_registry_unittest.cc',
-            'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
-            'rtp_rtcp/source/rtp_header_extension_unittest.cc',
-            'rtp_rtcp/source/rtp_sender_unittest.cc',
-            'rtp_rtcp/source/vp8_partition_aggregator_unittest.cc',
-            'rtp_rtcp/test/testAPI/test_api.cc',
-            'rtp_rtcp/test/testAPI/test_api.h',
-            'rtp_rtcp/test/testAPI/test_api_audio.cc',
-            'rtp_rtcp/test/testAPI/test_api_rtcp.cc',
-            'rtp_rtcp/test/testAPI/test_api_video.cc',
-            'utility/source/audio_frame_operations_unittest.cc',
-            'utility/source/file_player_unittests.cc',
-            'utility/source/process_thread_impl_unittest.cc',
-            'video_coding/codecs/test/packet_manipulator_unittest.cc',
-            'video_coding/codecs/test/stats_unittest.cc',
-            'video_coding/codecs/test/videoprocessor_unittest.cc',
-            'video_coding/codecs/vp8/default_temporal_layers_unittest.cc',
-            'video_coding/codecs/vp8/reference_picture_selection_unittest.cc',
-            'video_coding/codecs/vp8/screenshare_layers_unittest.cc',
-            'video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc',
-            'video_coding/codecs/vp8/simulcast_unittest.cc',
-            'video_coding/codecs/vp8/simulcast_unittest.h',
-            'video_coding/main/interface/mock/mock_vcm_callbacks.h',
-            'video_coding/main/source/decoding_state_unittest.cc',
-            'video_coding/main/source/jitter_buffer_unittest.cc',
-            'video_coding/main/source/jitter_estimator_tests.cc',
-            'video_coding/main/source/media_optimization_unittest.cc',
-            'video_coding/main/source/receiver_unittest.cc',
-            'video_coding/main/source/session_info_unittest.cc',
-            'video_coding/main/source/timing_unittest.cc',
-            'video_coding/main/source/video_coding_robustness_unittest.cc',
-            'video_coding/main/source/video_receiver_unittest.cc',
-            'video_coding/main/source/video_sender_unittest.cc',
-            'video_coding/main/source/qm_select_unittest.cc',
-            'video_coding/main/source/test/stream_generator.cc',
-            'video_coding/main/source/test/stream_generator.h',
-            'video_coding/utility/quality_scaler_unittest.cc',
-            'video_processing/main/test/unit_test/brightness_detection_test.cc',
-            'video_processing/main/test/unit_test/color_enhancement_test.cc',
-            'video_processing/main/test/unit_test/content_metrics_test.cc',
-            'video_processing/main/test/unit_test/deflickering_test.cc',
-            'video_processing/main/test/unit_test/video_processing_unittest.cc',
-            'video_processing/main/test/unit_test/video_processing_unittest.h',
-          ],
-          'conditions': [
-            ['enable_bwe_test_logging==1', {
-              'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
-            }, {
-              'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
-              'sources!': [
-                'remote_bitrate_estimator/test/bwe_test_logging.cc'
-              ],
-            }],
-            # Run screen/window capturer tests only on platforms where they are
-            # supported.
-            ['desktop_capture_supported==0', {
-              'sources!': [
-                'desktop_capture/desktop_and_cursor_composer_unittest.cc',
-                'desktop_capture/mouse_cursor_monitor_unittest.cc',
-                'desktop_capture/screen_capturer_helper_unittest.cc',
-                'desktop_capture/screen_capturer_mac_unittest.cc',
-                'desktop_capture/screen_capturer_mock_objects.h',
-                'desktop_capture/screen_capturer_unittest.cc',
-                'desktop_capture/window_capturer_unittest.cc',
-              ],
-            }],
-            ['prefer_fixed_point==1', {
-              'defines': [ 'WEBRTC_AUDIOPROC_FIXED_PROFILE' ],
-            }, {
-              'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
-            }],
-            ['enable_protobuf==1', {
-              'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
-              'dependencies': [
-                'audioproc_unittest_proto',
-              ],
-              'sources': [
-                'audio_processing/audio_processing_impl_unittest.cc',
-                'audio_processing/test/audio_processing_unittest.cc',
-                'audio_processing/test/test_utils.h',
-              ],
-            }],
-            ['build_libvpx==1', {
-              'dependencies': [
-                '<(libvpx_dir)/libvpx.gyp:libvpx',
-              ],
-            }],
-            ['OS=="android"', {
-              'dependencies': [
-                '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
-              ],
-              # Need to disable error due to the line in
-              # base/android/jni_android.h triggering it:
-              # const BASE_EXPORT jobject GetApplicationContext()
-              # error: type qualifiers ignored on function return type
-              'cflags': [
-                '-Wno-ignored-qualifiers',
-              ],
-              'sources': [
-                'audio_device/android/audio_device_unittest.cc',
-                'audio_device/android/ensure_initialized.cc',
-                'audio_device/android/ensure_initialized.h',
-              ],
-            }],
-          ],
-          # Disable warnings to enable Win64 build, issue 1323.
-          'msvs_disabled_warnings': [
-            4267,  # size_t to int truncation.
-          ],
-        },
         {
           'target_name': 'modules_tests',
           'type': '<(gtest_target_type)',
@@ -394,25 +71,24 @@
             '<@(audio_coding_defines)',
           ],
           'sources': [
-            'audio_coding/main/test/APITest.cc',
-            'audio_coding/main/test/Channel.cc',
-            'audio_coding/main/test/EncodeDecodeTest.cc',
-            'audio_coding/main/test/PCMFile.cc',
-            'audio_coding/main/test/PacketLossTest.cc',
-            'audio_coding/main/test/RTPFile.cc',
-            'audio_coding/main/test/SpatialAudio.cc',
-            'audio_coding/main/test/TestAllCodecs.cc',
-            'audio_coding/main/test/TestRedFec.cc',
-            'audio_coding/main/test/TestStereo.cc',
-            'audio_coding/main/test/TestVADDTX.cc',
-            'audio_coding/main/test/Tester.cc',
-            'audio_coding/main/test/TimedTrace.cc',
-            'audio_coding/main/test/TwoWayCommunication.cc',
-            'audio_coding/main/test/iSACTest.cc',
-            'audio_coding/main/test/initial_delay_unittest.cc',
-            'audio_coding/main/test/opus_test.cc',
-            'audio_coding/main/test/target_delay_unittest.cc',
-            'audio_coding/main/test/utility.cc',
+            'audio_coding/test/APITest.cc',
+            'audio_coding/test/Channel.cc',
+            'audio_coding/test/EncodeDecodeTest.cc',
+            'audio_coding/test/PCMFile.cc',
+            'audio_coding/test/PacketLossTest.cc',
+            'audio_coding/test/RTPFile.cc',
+            'audio_coding/test/SpatialAudio.cc',
+            'audio_coding/test/TestAllCodecs.cc',
+            'audio_coding/test/TestRedFec.cc',
+            'audio_coding/test/TestStereo.cc',
+            'audio_coding/test/TestVADDTX.cc',
+            'audio_coding/test/Tester.cc',
+            'audio_coding/test/TimedTrace.cc',
+            'audio_coding/test/TwoWayCommunication.cc',
+            'audio_coding/test/iSACTest.cc',
+            'audio_coding/test/opus_test.cc',
+            'audio_coding/test/target_delay_unittest.cc',
+            'audio_coding/test/utility.cc',
             'rtp_rtcp/test/testFec/test_fec.cc',
             'video_coding/codecs/test/videoprocessor_integrationtest.cc',
             'video_coding/codecs/vp8/test/vp8_impl_unittest.cc',
@@ -427,6 +103,379 @@
         },
       ],
       'conditions': [
+        # Does not compile on iOS for ia32 or x64: webrtc:4755.
+        ['OS!="ios" or target_arch=="arm" or target_arch=="arm64"', {
+          'targets': [
+            {
+              'target_name': 'modules_unittests',
+              'type': '<(gtest_target_type)',
+              'defines': [
+                '<@(audio_coding_defines)',
+              ],
+              'dependencies': [
+                'acm_receive_test',
+                'acm_send_test',
+                'audio_coding_module',
+                'audio_conference_mixer',
+                'audio_device'  ,
+                'audio_processing',
+                'audioproc_test_utils',
+                'bitrate_controller',
+                'bwe_simulator',
+                'cng',
+                'desktop_capture',
+                'isac_fix',
+                'media_file',
+                'neteq',
+                'neteq_test_support',
+                'neteq_unittest_tools',
+                'paced_sender',
+                'pcm16b',  # Needed by NetEq tests.
+                'red',
+                'remote_bitrate_estimator',
+                'rtp_rtcp',
+                'video_codecs_test_framework',
+                'video_processing',
+                'webrtc_utility',
+                'webrtc_video_coding',
+                '<@(neteq_dependencies)',
+                '<(DEPTH)/testing/gmock.gyp:gmock',
+                '<(DEPTH)/testing/gtest.gyp:gtest',
+                '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+                '<(webrtc_root)/common.gyp:webrtc_common',
+                '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+                '<(webrtc_root)/common_video/common_video.gyp:common_video',
+                '<(webrtc_root)/modules/modules.gyp:video_capture',
+                '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
+                '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9',
+                '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+                '<(webrtc_root)/test/test.gyp:fake_video_frames',
+                '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+                '<(webrtc_root)/test/test.gyp:test_support_main',
+                '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
+                '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
+              ],
+              'sources': [
+                'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
+                'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',
+                'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',
+                'audio_coding/acm2/call_statistics_unittest.cc',
+                'audio_coding/acm2/codec_manager_unittest.cc',
+                'audio_coding/acm2/initial_delay_manager_unittest.cc',
+                'audio_coding/acm2/rent_a_codec_unittest.cc',
+                'audio_coding/codecs/cng/cng_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
+                'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
+                'audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc',
+                'audio_coding/codecs/isac/main/source/isac_unittest.cc',
+                'audio_coding/codecs/isac/unittest.cc',
+                'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
+                'audio_coding/codecs/opus/opus_unittest.cc',
+                'audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc',
+                'audio_coding/neteq/audio_classifier_unittest.cc',
+                'audio_coding/neteq/audio_multi_vector_unittest.cc',
+                'audio_coding/neteq/audio_vector_unittest.cc',
+                'audio_coding/neteq/background_noise_unittest.cc',
+                'audio_coding/neteq/buffer_level_filter_unittest.cc',
+                'audio_coding/neteq/comfort_noise_unittest.cc',
+                'audio_coding/neteq/decision_logic_unittest.cc',
+                'audio_coding/neteq/decoder_database_unittest.cc',
+                'audio_coding/neteq/delay_manager_unittest.cc',
+                'audio_coding/neteq/delay_peak_detector_unittest.cc',
+                'audio_coding/neteq/dsp_helper_unittest.cc',
+                'audio_coding/neteq/dtmf_buffer_unittest.cc',
+                'audio_coding/neteq/dtmf_tone_generator_unittest.cc',
+                'audio_coding/neteq/expand_unittest.cc',
+                'audio_coding/neteq/merge_unittest.cc',
+                'audio_coding/neteq/nack_unittest.cc',
+                'audio_coding/neteq/neteq_external_decoder_unittest.cc',
+                'audio_coding/neteq/neteq_impl_unittest.cc',
+                'audio_coding/neteq/neteq_network_stats_unittest.cc',
+                'audio_coding/neteq/neteq_stereo_unittest.cc',
+                'audio_coding/neteq/neteq_unittest.cc',
+                'audio_coding/neteq/normal_unittest.cc',
+                'audio_coding/neteq/packet_buffer_unittest.cc',
+                'audio_coding/neteq/payload_splitter_unittest.cc',
+                'audio_coding/neteq/post_decode_vad_unittest.cc',
+                'audio_coding/neteq/random_vector_unittest.cc',
+                'audio_coding/neteq/sync_buffer_unittest.cc',
+                'audio_coding/neteq/timestamp_scaler_unittest.cc',
+                'audio_coding/neteq/time_stretch_unittest.cc',
+                'audio_coding/neteq/mock/mock_audio_decoder.h',
+                'audio_coding/neteq/mock/mock_audio_vector.h',
+                'audio_coding/neteq/mock/mock_buffer_level_filter.h',
+                'audio_coding/neteq/mock/mock_decoder_database.h',
+                'audio_coding/neteq/mock/mock_delay_manager.h',
+                'audio_coding/neteq/mock/mock_delay_peak_detector.h',
+                'audio_coding/neteq/mock/mock_dtmf_buffer.h',
+                'audio_coding/neteq/mock/mock_dtmf_tone_generator.h',
+                'audio_coding/neteq/mock/mock_expand.h',
+                'audio_coding/neteq/mock/mock_external_decoder_pcm16b.h',
+                'audio_coding/neteq/mock/mock_packet_buffer.h',
+                'audio_coding/neteq/mock/mock_payload_splitter.h',
+                'audio_coding/neteq/tools/input_audio_file_unittest.cc',
+                'audio_coding/neteq/tools/packet_unittest.cc',
+                'audio_conference_mixer/test/audio_conference_mixer_unittest.cc',
+                'audio_device/fine_audio_buffer_unittest.cc',
+                'audio_processing/aec/echo_cancellation_unittest.cc',
+                'audio_processing/aec/system_delay_unittest.cc',
+                'audio_processing/agc/agc_manager_direct_unittest.cc',
+                # TODO(ajm): Fix to match new interface.
+                # 'audio_processing/agc/agc_unittest.cc',
+                'audio_processing/agc/histogram_unittest.cc',
+                'audio_processing/agc/mock_agc.h',
+                'audio_processing/beamformer/array_util_unittest.cc',
+                'audio_processing/beamformer/complex_matrix_unittest.cc',
+                'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
+                'audio_processing/beamformer/matrix_unittest.cc',
+                'audio_processing/beamformer/mock_nonlinear_beamformer.h',
+                'audio_processing/beamformer/nonlinear_beamformer_unittest.cc',
+                'audio_processing/echo_cancellation_impl_unittest.cc',
+                'audio_processing/intelligibility/intelligibility_enhancer_unittest.cc',
+                'audio_processing/intelligibility/intelligibility_utils_unittest.cc',
+                'audio_processing/splitting_filter_unittest.cc',
+                'audio_processing/transient/dyadic_decimator_unittest.cc',
+                'audio_processing/transient/file_utils.cc',
+                'audio_processing/transient/file_utils.h',
+                'audio_processing/transient/file_utils_unittest.cc',
+                'audio_processing/transient/moving_moments_unittest.cc',
+                'audio_processing/transient/transient_detector_unittest.cc',
+                'audio_processing/transient/transient_suppressor_unittest.cc',
+                'audio_processing/transient/wpd_node_unittest.cc',
+                'audio_processing/transient/wpd_tree_unittest.cc',
+                'audio_processing/utility/delay_estimator_unittest.cc',
+                'audio_processing/vad/gmm_unittest.cc',
+                'audio_processing/vad/pitch_based_vad_unittest.cc',
+                'audio_processing/vad/pitch_internal_unittest.cc',
+                'audio_processing/vad/pole_zero_filter_unittest.cc',
+                'audio_processing/vad/standalone_vad_unittest.cc',
+                'audio_processing/vad/vad_audio_proc_unittest.cc',
+                'audio_processing/vad/vad_circular_buffer_unittest.cc',
+                'audio_processing/vad/voice_activity_detector_unittest.cc',
+                'bitrate_controller/bitrate_controller_unittest.cc',
+                'bitrate_controller/send_side_bandwidth_estimation_unittest.cc',
+                'desktop_capture/desktop_and_cursor_composer_unittest.cc',
+                'desktop_capture/desktop_region_unittest.cc',
+                'desktop_capture/differ_block_unittest.cc',
+                'desktop_capture/differ_unittest.cc',
+                'desktop_capture/mouse_cursor_monitor_unittest.cc',
+                'desktop_capture/screen_capturer_helper_unittest.cc',
+                'desktop_capture/screen_capturer_mac_unittest.cc',
+                'desktop_capture/screen_capturer_mock_objects.h',
+                'desktop_capture/screen_capturer_unittest.cc',
+                'desktop_capture/window_capturer_unittest.cc',
+                'desktop_capture/win/cursor_unittest.cc',
+                'desktop_capture/win/cursor_unittest_resources.h',
+                'desktop_capture/win/cursor_unittest_resources.rc',
+                'media_file/media_file_unittest.cc',
+                'module_common_types_unittest.cc',
+                'pacing/bitrate_prober_unittest.cc',
+                'pacing/paced_sender_unittest.cc',
+                'pacing/packet_router_unittest.cc',
+                'remote_bitrate_estimator/bwe_simulations.cc',
+                'remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h',
+                'remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h',
+                'remote_bitrate_estimator/inter_arrival_unittest.cc',
+                'remote_bitrate_estimator/overuse_detector_unittest.cc',
+                'remote_bitrate_estimator/rate_statistics_unittest.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
+                'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
+                'remote_bitrate_estimator/remote_estimator_proxy_unittest.cc',
+                'remote_bitrate_estimator/send_time_history_unittest.cc',
+                'remote_bitrate_estimator/test/bwe_test_framework_unittest.cc',
+                'remote_bitrate_estimator/test/bwe_unittest.cc',
+                'remote_bitrate_estimator/test/metric_recorder_unittest.cc',
+                'remote_bitrate_estimator/test/estimators/nada_unittest.cc',
+                'remote_bitrate_estimator/transport_feedback_adapter_unittest.cc',
+                'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
+                'rtp_rtcp/source/byte_io_unittest.cc',
+                'rtp_rtcp/source/fec_receiver_unittest.cc',
+                'rtp_rtcp/source/fec_test_helper.cc',
+                'rtp_rtcp/source/fec_test_helper.h',
+                'rtp_rtcp/source/h264_sps_parser_unittest.cc',
+                'rtp_rtcp/source/h264_bitstream_parser_unittest.cc',
+                'rtp_rtcp/source/nack_rtx_unittest.cc',
+                'rtp_rtcp/source/packet_loss_stats_unittest.cc',
+                'rtp_rtcp/source/producer_fec_unittest.cc',
+                'rtp_rtcp/source/receive_statistics_unittest.cc',
+                'rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc',
+                'rtp_rtcp/source/rtcp_format_remb_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/app_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/bye_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/nack_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/pli_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/sli_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc',
+                'rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc',
+                'rtp_rtcp/source/rtcp_receiver_unittest.cc',
+                'rtp_rtcp/source/rtcp_sender_unittest.cc',
+                'rtp_rtcp/source/rtcp_utility_unittest.cc',
+                'rtp_rtcp/source/rtp_fec_unittest.cc',
+                'rtp_rtcp/source/rtp_format_h264_unittest.cc',
+                'rtp_rtcp/source/rtp_format_vp8_test_helper.cc',
+                'rtp_rtcp/source/rtp_format_vp8_test_helper.h',
+                'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
+                'rtp_rtcp/source/rtp_format_vp9_unittest.cc',
+                'rtp_rtcp/source/rtp_packet_history_unittest.cc',
+                'rtp_rtcp/source/rtp_payload_registry_unittest.cc',
+                'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
+                'rtp_rtcp/source/rtp_header_extension_unittest.cc',
+                'rtp_rtcp/source/rtp_sender_unittest.cc',
+                'rtp_rtcp/source/time_util_unittest.cc',
+                'rtp_rtcp/source/vp8_partition_aggregator_unittest.cc',
+                'rtp_rtcp/test/testAPI/test_api.cc',
+                'rtp_rtcp/test/testAPI/test_api.h',
+                'rtp_rtcp/test/testAPI/test_api_audio.cc',
+                'rtp_rtcp/test/testAPI/test_api_rtcp.cc',
+                'rtp_rtcp/test/testAPI/test_api_video.cc',
+                'utility/source/audio_frame_operations_unittest.cc',
+                'utility/source/file_player_unittests.cc',
+                'utility/source/process_thread_impl_unittest.cc',
+                'video_coding/codecs/test/packet_manipulator_unittest.cc',
+                'video_coding/codecs/test/stats_unittest.cc',
+                'video_coding/codecs/test/videoprocessor_unittest.cc',
+                'video_coding/codecs/vp8/default_temporal_layers_unittest.cc',
+                'video_coding/codecs/vp8/reference_picture_selection_unittest.cc',
+                'video_coding/codecs/vp8/screenshare_layers_unittest.cc',
+                'video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc',
+                'video_coding/codecs/vp8/simulcast_unittest.cc',
+                'video_coding/codecs/vp8/simulcast_unittest.h',
+                'video_coding/codecs/vp9/screenshare_layers_unittest.cc',
+                'video_coding/include/mock/mock_vcm_callbacks.h',
+                'video_coding/decoding_state_unittest.cc',
+                'video_coding/jitter_buffer_unittest.cc',
+                'video_coding/jitter_estimator_tests.cc',
+                'video_coding/media_optimization_unittest.cc',
+                'video_coding/receiver_unittest.cc',
+                'video_coding/session_info_unittest.cc',
+                'video_coding/timing_unittest.cc',
+                'video_coding/video_coding_robustness_unittest.cc',
+                'video_coding/video_receiver_unittest.cc',
+                'video_coding/video_sender_unittest.cc',
+                'video_coding/qm_select_unittest.cc',
+                'video_coding/test/stream_generator.cc',
+                'video_coding/test/stream_generator.h',
+                'video_coding/utility/quality_scaler_unittest.cc',
+                'video_processing/test/brightness_detection_test.cc',
+                'video_processing/test/content_metrics_test.cc',
+                'video_processing/test/deflickering_test.cc',
+                'video_processing/test/denoiser_test.cc',
+                'video_processing/test/video_processing_unittest.cc',
+                'video_processing/test/video_processing_unittest.h',
+              ],
+              'conditions': [
+                ['enable_bwe_test_logging==1', {
+                  'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
+                }, {
+                  'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
+                  'sources!': [
+                    'remote_bitrate_estimator/test/bwe_test_logging.cc'
+                  ],
+                }],
+                # Run screen/window capturer tests only on platforms where they are
+                # supported.
+                ['desktop_capture_supported==0', {
+                  'sources!': [
+                    'desktop_capture/desktop_and_cursor_composer_unittest.cc',
+                    'desktop_capture/mouse_cursor_monitor_unittest.cc',
+                    'desktop_capture/screen_capturer_helper_unittest.cc',
+                    'desktop_capture/screen_capturer_mac_unittest.cc',
+                    'desktop_capture/screen_capturer_mock_objects.h',
+                    'desktop_capture/screen_capturer_unittest.cc',
+                    'desktop_capture/window_capturer_unittest.cc',
+                  ],
+                }],
+                ['prefer_fixed_point==1', {
+                  'defines': [ 'WEBRTC_AUDIOPROC_FIXED_PROFILE' ],
+                }, {
+                  'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
+                }],
+                ['enable_protobuf==1', {
+                  'defines': [
+                    'WEBRTC_AUDIOPROC_DEBUG_DUMP',
+                    'WEBRTC_NETEQ_UNITTEST_BITEXACT',
+                  ],
+                  'dependencies': [
+                    'audioproc_protobuf_utils',
+                    'audioproc_unittest_proto',
+                    'neteq_unittest_proto',
+                  ],
+                  'sources': [
+                    'audio_processing/audio_processing_impl_locking_unittest.cc',
+                    'audio_processing/audio_processing_impl_unittest.cc',
+                    'audio_processing/test/audio_processing_unittest.cc',
+                    'audio_processing/test/debug_dump_test.cc',
+                    'audio_processing/test/test_utils.h',
+                  ],
+                }],
+                ['build_libvpx==1', {
+                  'dependencies': [
+                    '<(libvpx_dir)/libvpx.gyp:libvpx_new',
+                  ],
+                }],
+                ['OS=="android"', {
+                  'dependencies': [
+                    '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+                  ],
+                  # Need to disable error due to the line in
+                  # base/android/jni_android.h triggering it:
+                  # const BASE_EXPORT jobject GetApplicationContext()
+                  # error: type qualifiers ignored on function return type
+                  'cflags': [
+                    '-Wno-ignored-qualifiers',
+                  ],
+                  'sources': [
+                    'audio_device/android/audio_device_unittest.cc',
+                    'audio_device/android/audio_manager_unittest.cc',
+                    'audio_device/android/ensure_initialized.cc',
+                    'audio_device/android/ensure_initialized.h',
+                  ],
+                }],
+                ['OS=="ios"', {
+                  'sources': [
+                    'video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc',
+                    'audio_device/ios/audio_device_unittest_ios.cc',
+                  ],
+                  'mac_bundle_resources': [
+                    '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
+                    '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
+                    '<(DEPTH)/resources/audio_coding/teststereo32kHz.pcm',
+                    '<(DEPTH)/resources/audio_device/audio_short16.pcm',
+                    '<(DEPTH)/resources/audio_device/audio_short44.pcm',
+                    '<(DEPTH)/resources/audio_device/audio_short48.pcm',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_no_circular_buffer.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_pitch_gain.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_pitch_lag.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_spectral_peak.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_voicing_prob.dat',
+                    '<(DEPTH)/resources/audio_processing/agc/agc_with_circular_buffer.dat',
+                    '<(DEPTH)/resources/short_mixed_mono_48.dat',
+                    '<(DEPTH)/resources/short_mixed_mono_48.pcm',
+                    '<(DEPTH)/resources/short_mixed_stereo_48.dat',
+                    '<(DEPTH)/resources/short_mixed_stereo_48.pcm',
+                  ],
+                }],
+              ],
+              # Disable warnings to enable Win64 build, issue 1323.
+              'msvs_disabled_warnings': [
+                4267,  # size_t to int truncation.
+              ],
+            },
+          ],
+        }],
         ['OS=="android"', {
           'targets': [
             {
@@ -447,6 +496,45 @@
         }],
         ['test_isolation_mode != "noop"', {
           'targets': [
+            {
+              'target_name': 'audio_codec_speed_tests_run',
+              'type': 'none',
+              'dependencies': [
+                'audio_codec_speed_tests',
+              ],
+              'includes': [
+                '../build/isolate.gypi',
+              ],
+              'sources': [
+                'audio_codec_speed_tests.isolate',
+              ],
+            },
+            {
+              'target_name': 'audio_decoder_unittests_run',
+              'type': 'none',
+              'dependencies': [
+                'audio_decoder_unittests',
+              ],
+              'includes': [
+                '../build/isolate.gypi',
+              ],
+              'sources': [
+                'audio_decoder_unittests.isolate',
+              ],
+            },
+            {
+              'target_name': 'audio_device_tests_run',
+              'type': 'none',
+              'dependencies': [
+                'audio_device_tests',
+              ],
+              'includes': [
+                '../build/isolate.gypi',
+              ],
+              'sources': [
+                'audio_device_tests.isolate',
+              ],
+            },
             {
               'target_name': 'modules_tests_run',
               'type': 'none',
@@ -473,6 +561,19 @@
                 'modules_unittests.isolate',
               ],
             },
+            {
+              'target_name': 'video_render_tests_run',
+              'type': 'none',
+              'dependencies': [
+                'video_render_tests',
+              ],
+              'includes': [
+                '../build/isolate.gypi',
+              ],
+              'sources': [
+                'video_render_tests.isolate',
+              ],
+            },
           ],
         }],
       ],
diff --git a/media/webrtc/trunk/webrtc/modules/modules_java.gyp b/media/webrtc/trunk/webrtc/modules/modules_java.gyp
index e59d2bd41c..060de2a067 100644
--- a/media/webrtc/trunk/webrtc/modules/modules_java.gyp
+++ b/media/webrtc/trunk/webrtc/modules/modules_java.gyp
@@ -13,25 +13,17 @@
       'type': 'none',
       'variables': {
         'java_in_dir': 'audio_device/android/java',
+        'additional_src_dirs': [ '../base/java/src', ],
       },
+
       'includes': [ '../../build/java.gypi' ],
     }, # audio_device_module_java
-    {
-      'target_name': 'video_capture_module_java',
-      'type': 'none',
-      'dependencies': [
-        'video_render_module_java',
-      ],
-      'variables': {
-        'java_in_dir': 'video_capture/android/java',
-      },
-      'includes': [ '../../build/java.gypi' ],
-    }, # video_capture_module_java
     {
       'target_name': 'video_render_module_java',
       'type': 'none',
       'variables': {
         'java_in_dir': 'video_render/android/java',
+        'additional_src_dirs': [ '../base/java/src', ],
       },
       'includes': [ '../../build/java.gypi' ],
     }, # video_render_module_java
diff --git a/media/webrtc/trunk/webrtc/modules/modules_java_chromium.gyp b/media/webrtc/trunk/webrtc/modules/modules_java_chromium.gyp
index 247a81d929..32d2d8d24e 100644
--- a/media/webrtc/trunk/webrtc/modules/modules_java_chromium.gyp
+++ b/media/webrtc/trunk/webrtc/modules/modules_java_chromium.gyp
@@ -16,17 +16,6 @@
       },
       'includes': [ '../../../build/java.gypi' ],
     }, # audio_device_module_java
-    {
-      'target_name': 'video_capture_module_java',
-      'type': 'none',
-      'dependencies': [
-        'video_render_module_java',
-      ],
-      'variables': {
-        'java_in_dir': 'video_capture/android/java',
-      },
-      'includes': [ '../../../build/java.gypi' ],
-    }, # video_capture_module_java
     {
       'target_name': 'video_render_module_java',
       'type': 'none',
diff --git a/media/webrtc/trunk/webrtc/modules/modules_unittests.isolate b/media/webrtc/trunk/webrtc/modules/modules_unittests.isolate
index bc6437f37f..d988821af0 100644
--- a/media/webrtc/trunk/webrtc/modules/modules_unittests.isolate
+++ b/media/webrtc/trunk/webrtc/modules/modules_unittests.isolate
@@ -18,6 +18,7 @@
       'variables': {
         'files': [
           '<(DEPTH)/data/audio_processing/output_data_float.pb',
+          '<(DEPTH)/data/audio_processing/output_data_mac.pb',
           '<(DEPTH)/data/voice_engine/audio_tiny48.wav',
           '<(DEPTH)/resources/att-downlink.rx',
           '<(DEPTH)/resources/att-uplink.rx',
@@ -26,10 +27,7 @@
           '<(DEPTH)/resources/audio_coding/neteq4_universal_ref.pcm',
           '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_32.pcm',
           '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_64.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq_network_stats.dat',
-          '<(DEPTH)/resources/audio_coding/neteq_rtcp_stats.dat',
           '<(DEPTH)/resources/audio_coding/neteq_universal_new.rtp',
-          '<(DEPTH)/resources/audio_coding/neteq_universal_ref.pcm',
           '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
           '<(DEPTH)/resources/audio_coding/speech_mono_32_48kHz.pcm',
           '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
@@ -129,7 +127,7 @@
         ],
       },
     }],
-    ['OS=="linux" or OS=="mac" or OS=="win"', {
+    ['(OS=="linux" or OS=="mac" or OS=="win") and use_x11==0', {
       'variables': {
         'command': [
           '<(DEPTH)/testing/test_env.py',
@@ -142,5 +140,22 @@
         ],
       },
     }],
+    ['(OS=="linux" or OS=="mac" or OS=="win") and use_x11==1', {
+      'variables': {
+        'command': [
+          '<(DEPTH)/testing/xvfb.py',
+          '<(PRODUCT_DIR)',
+          '<(DEPTH)/testing/test_env.py',
+          '<(PRODUCT_DIR)/modules_unittests<(EXECUTABLE_SUFFIX)',
+        ],
+        'files': [
+          '<(DEPTH)/DEPS',
+          '<(DEPTH)/testing/test_env.py',
+          '<(DEPTH)/testing/xvfb.py',
+          '<(PRODUCT_DIR)/modules_unittests<(EXECUTABLE_SUFFIX)',
+          '<(PRODUCT_DIR)/xdisplaycheck<(EXECUTABLE_SUFFIX)',
+        ],
+      },
+    }],
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/BUILD.gn b/media/webrtc/trunk/webrtc/modules/pacing/BUILD.gn
index ffced4d882..0354c64fcb 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/pacing/BUILD.gn
@@ -8,12 +8,12 @@
 
 source_set("pacing") {
   sources = [
-    "include/paced_sender.h",
-    "include/packet_router.h",
     "bitrate_prober.cc",
     "bitrate_prober.h",
     "paced_sender.cc",
+    "paced_sender.h",
     "packet_router.cc",
+    "packet_router.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -25,5 +25,9 @@ source_set("pacing") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  deps = [ "../../system_wrappers" ]
+  deps = [
+    "../../system_wrappers",
+    "../bitrate_controller",
+    "../rtp_rtcp",
+  ]
 }
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.cc b/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.cc
index 5475ef3f1d..41ad5fa11a 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.cc
+++ b/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.cc
@@ -11,10 +11,12 @@
 #include "webrtc/modules/pacing/bitrate_prober.h"
 
 #include <assert.h>
+#include <algorithm>
 #include <limits>
 #include <sstream>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 
 namespace webrtc {
 
@@ -87,7 +89,8 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
   // We will send the first probe packet immediately if no packet has been
   // sent before.
   int time_until_probe_ms = 0;
-  if (packet_size_last_send_ > 0 && probing_state_ == kProbing) {
+  if (packet_size_last_send_ > PacedSender::kMinProbePacketSize &&
+      probing_state_ == kProbing) {
     int next_delta_ms = ComputeDeltaFromBitrate(packet_size_last_send_,
                                                 probe_bitrates_.front());
     time_until_probe_ms = next_delta_ms - elapsed_time_ms;
@@ -107,7 +110,11 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
       time_until_probe_ms = 0;
     }
   }
-  return time_until_probe_ms;
+  return std::max(time_until_probe_ms, 0);
+}
+
+size_t BitrateProber::RecommendedPacketSize() const {
+  return packet_size_last_send_;
 }
 
 void BitrateProber::PacketSent(int64_t now_ms, size_t packet_size) {
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.h b/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.h
index 04a858058f..b3f52afeb6 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.h
+++ b/media/webrtc/trunk/webrtc/modules/pacing/bitrate_prober.h
@@ -38,6 +38,10 @@ class BitrateProber {
   // get accurate probing.
   int TimeUntilNextProbe(int64_t now_ms);
 
+  // Returns the number of bytes that the prober recommends for the next probe
+  // packet.
+  size_t RecommendedPacketSize() const;
+
   // Called to report to the prober that a packet has been sent, which helps the
   // prober know when to move to the next packet in a probe.
   void PacketSent(int64_t now_ms, size_t packet_size);
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/include/mock/mock_paced_sender.h b/media/webrtc/trunk/webrtc/modules/pacing/mock/mock_paced_sender.h
similarity index 77%
rename from media/webrtc/trunk/webrtc/modules/pacing/include/mock/mock_paced_sender.h
rename to media/webrtc/trunk/webrtc/modules/pacing/mock/mock_paced_sender.h
index 632ec9bd05..01d5f6a6e9 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/include/mock/mock_paced_sender.h
+++ b/media/webrtc/trunk/webrtc/modules/pacing/mock/mock_paced_sender.h
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_PACING_INCLUDE_MOCK_MOCK_PACED_SENDER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_MOCK_MOCK_PACED_SENDER_H_
+#ifndef WEBRTC_MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
+#define WEBRTC_MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
 
 #include <vector>
 
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -35,4 +35,4 @@ class MockPacedSender : public PacedSender {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_MOCK_MOCK_PACED_SENDER_H_
+#endif  // WEBRTC_MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/paced_sender.cc b/media/webrtc/trunk/webrtc/modules/pacing/paced_sender.cc
index 6186f96c3f..121f860c7d 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/paced_sender.cc
+++ b/media/webrtc/trunk/webrtc/modules/pacing/paced_sender.cc
@@ -8,20 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/pacing/include/paced_sender.h"
-
-#include <assert.h>
+#include "webrtc/modules/pacing/paced_sender.h"
 
 #include <map>
 #include <queue>
 #include <set>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/bitrate_prober.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/field_trial.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 // Time limit in milliseconds between packet bursts.
@@ -33,10 +32,13 @@ const int64_t kMaxIntervalTimeMs = 30;
 
 }  // namespace
 
+// TODO(sprang): Move at least PacketQueue and MediaBudget out to separate
+// files, so that we can more easily test them.
+
 namespace webrtc {
 namespace paced_sender {
 struct Packet {
-  Packet(PacedSender::Priority priority,
+  Packet(RtpPacketSender::Priority priority,
          uint32_t ssrc,
          uint16_t seq_number,
          int64_t capture_time_ms,
@@ -53,7 +55,7 @@ struct Packet {
         retransmission(retransmission),
         enqueue_order(enqueue_order) {}
 
-  PacedSender::Priority priority;
+  RtpPacketSender::Priority priority;
   uint32_t ssrc;
   uint16_t sequence_number;
   int64_t capture_time_ms;
@@ -86,13 +88,19 @@ struct Comparator {
 // Class encapsulating a priority queue with some extensions.
 class PacketQueue {
  public:
-  PacketQueue() : bytes_(0) {}
+  explicit PacketQueue(Clock* clock)
+      : bytes_(0),
+        clock_(clock),
+        queue_time_sum_(0),
+        time_last_updated_(clock_->TimeInMilliseconds()) {}
   virtual ~PacketQueue() {}
 
   void Push(const Packet& packet) {
-    if (!AddToDupeSet(packet)) {
+    if (!AddToDupeSet(packet))
       return;
-    }
+
+    UpdateQueueTime(packet.enqueue_time_ms);
+
     // Store packet in list, use pointers in priority queue for cheaper moves.
     // Packets have a handle to its own iterator in the list, for easy removal
     // when popping from queue.
@@ -114,7 +122,11 @@ class PacketQueue {
   void FinalizePop(const Packet& packet) {
     RemoveFromDupeSet(packet);
     bytes_ -= packet.bytes;
+    queue_time_sum_ -= (time_last_updated_ - packet.enqueue_time_ms);
     packet_list_.erase(packet.this_it);
+    RTC_DCHECK_EQ(packet_list_.size(), prio_queue_.size());
+    if (packet_list_.empty())
+      RTC_DCHECK_EQ(0u, queue_time_sum_);
   }
 
   bool Empty() const { return prio_queue_.empty(); }
@@ -123,13 +135,29 @@ class PacketQueue {
 
   uint64_t SizeInBytes() const { return bytes_; }
 
-  int64_t OldestEnqueueTime() const {
-    std::list<Packet>::const_reverse_iterator it = packet_list_.rbegin();
+  int64_t OldestEnqueueTimeMs() const {
+    auto it = packet_list_.rbegin();
     if (it == packet_list_.rend())
       return 0;
     return it->enqueue_time_ms;
   }
 
+  void UpdateQueueTime(int64_t timestamp_ms) {
+    RTC_DCHECK_GE(timestamp_ms, time_last_updated_);
+    int64_t delta = timestamp_ms - time_last_updated_;
+    // Use packet packet_list_.size() not prio_queue_.size() here, as there
+    // might be an outstanding element popped from prio_queue_ currently in the
+    // SendPacket() call, while packet_list_ will always be correct.
+    queue_time_sum_ += delta * packet_list_.size();
+    time_last_updated_ = timestamp_ms;
+  }
+
+  int64_t AverageQueueTimeMs() const {
+    if (prio_queue_.empty())
+      return 0;
+    return queue_time_sum_ / packet_list_.size();
+  }
+
  private:
   // Try to add a packet to the set of ssrc/seqno identifiers currently in the
   // queue. Return true if inserted, false if this is a duplicate.
@@ -147,7 +175,7 @@ class PacketQueue {
 
   void RemoveFromDupeSet(const Packet& packet) {
     SsrcSeqNoMap::iterator it = dupe_map_.find(packet.ssrc);
-    assert(it != dupe_map_.end());
+    RTC_DCHECK(it != dupe_map_.end());
     it->second.erase(packet.sequence_number);
     if (it->second.empty()) {
       dupe_map_.erase(it);
@@ -165,6 +193,9 @@ class PacketQueue {
   // Map<ssrc, set<seq_no> >, for checking duplicates.
   typedef std::map<uint32_t, std::set<uint16_t> > SsrcSeqNoMap;
   SsrcSeqNoMap dupe_map_;
+  Clock* const clock_;
+  int64_t queue_time_sum_;
+  int64_t time_last_updated_;
 };
 
 class IntervalBudget {
@@ -175,6 +206,8 @@ class IntervalBudget {
 
   void set_target_rate_kbps(int target_rate_kbps) {
     target_rate_kbps_ = target_rate_kbps;
+    bytes_remaining_ =
+        std::max(-kWindowMs * target_rate_kbps_ / 8, bytes_remaining_);
   }
 
   void IncreaseBudget(int64_t delta_time_ms) {
@@ -190,19 +223,24 @@ class IntervalBudget {
 
   void UseBudget(size_t bytes) {
     bytes_remaining_ = std::max(bytes_remaining_ - static_cast<int>(bytes),
-                                -500 * target_rate_kbps_ / 8);
+                                -kWindowMs * target_rate_kbps_ / 8);
   }
 
-  int bytes_remaining() const { return bytes_remaining_; }
+  size_t bytes_remaining() const {
+    return static_cast<size_t>(std::max(0, bytes_remaining_));
+  }
 
   int target_rate_kbps() const { return target_rate_kbps_; }
 
  private:
+  static const int kWindowMs = 500;
+
   int target_rate_kbps_;
   int bytes_remaining_;
 };
 }  // namespace paced_sender
 
+const int64_t PacedSender::kMaxQueueLengthMs = 2000;
 const float PacedSender::kDefaultPaceMultiplier = 2.5f;
 
 PacedSender::PacedSender(Clock* clock,
@@ -213,15 +251,15 @@ PacedSender::PacedSender(Clock* clock,
     : clock_(clock),
       callback_(callback),
       critsect_(CriticalSectionWrapper::CreateCriticalSection()),
-      enabled_(true),
       paused_(false),
       probing_enabled_(true),
       media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)),
       padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)),
       prober_(new BitrateProber()),
       bitrate_bps_(1000 * bitrate_kbps),
+      max_bitrate_kbps_(max_bitrate_kbps),
       time_last_update_us_(clock->TimeInMicroseconds()),
-      packets_(new paced_sender::PacketQueue()),
+      packets_(new paced_sender::PacketQueue(clock)),
       packet_counter_(0) {
   UpdateBytesPerInterval(kMinPacketLimitMs);
 }
@@ -239,57 +277,47 @@ void PacedSender::Resume() {
 }
 
 void PacedSender::SetProbingEnabled(bool enabled) {
-  assert(packet_counter_ == 0);
+  RTC_CHECK_EQ(0u, packet_counter_);
   probing_enabled_ = enabled;
 }
 
-void PacedSender::SetStatus(bool enable) {
-  CriticalSectionScoped cs(critsect_.get());
-  enabled_ = enable;
-}
-
-bool PacedSender::Enabled() const {
-  CriticalSectionScoped cs(critsect_.get());
-  return enabled_;
-}
-
 void PacedSender::UpdateBitrate(int bitrate_kbps,
                                 int max_bitrate_kbps,
                                 int min_bitrate_kbps) {
   CriticalSectionScoped cs(critsect_.get());
-  media_budget_->set_target_rate_kbps(max_bitrate_kbps);
+  // Don't set media bitrate here as it may be boosted in order to meet max
+  // queue time constraint. Just update max_bitrate_kbps_ and let media_budget_
+  // be updated in Process().
   padding_budget_->set_target_rate_kbps(min_bitrate_kbps);
   bitrate_bps_ = 1000 * bitrate_kbps;
+  max_bitrate_kbps_ = max_bitrate_kbps;
 }
 
-bool PacedSender::SendPacket(Priority priority, uint32_t ssrc,
-    uint16_t sequence_number, int64_t capture_time_ms, size_t bytes,
-    bool retransmission) {
+void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
+                               uint32_t ssrc,
+                               uint16_t sequence_number,
+                               int64_t capture_time_ms,
+                               size_t bytes,
+                               bool retransmission) {
   CriticalSectionScoped cs(critsect_.get());
 
-  if (!enabled_) {
-    return true;  // We can send now.
-  }
-  if (probing_enabled_ && !prober_->IsProbing()) {
+  if (probing_enabled_ && !prober_->IsProbing())
     prober_->SetEnabled(true);
-  }
   prober_->MaybeInitializeProbe(bitrate_bps_);
 
-  if (capture_time_ms < 0) {
-    capture_time_ms = clock_->TimeInMilliseconds();
-  }
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (capture_time_ms < 0)
+    capture_time_ms = now_ms;
 
-  packets_->Push(paced_sender::Packet(
-      priority, ssrc, sequence_number, capture_time_ms,
-      clock_->TimeInMilliseconds(), bytes, retransmission, packet_counter_++));
-  return false;
+  packets_->Push(paced_sender::Packet(priority, ssrc, sequence_number,
+                                      capture_time_ms, now_ms, bytes,
+                                      retransmission, packet_counter_++));
 }
 
 int64_t PacedSender::ExpectedQueueTimeMs() const {
   CriticalSectionScoped cs(critsect_.get());
-  int target_rate = media_budget_->target_rate_kbps();
-  assert(target_rate > 0);
-  return static_cast<int64_t>(packets_->SizeInBytes() * 8 / target_rate);
+  RTC_DCHECK_GT(max_bitrate_kbps_, 0);
+  return static_cast<int64_t>(packets_->SizeInBytes() * 8 / max_bitrate_kbps_);
 }
 
 size_t PacedSender::QueueSizePackets() const {
@@ -300,20 +328,25 @@ size_t PacedSender::QueueSizePackets() const {
 int64_t PacedSender::QueueInMs() const {
   CriticalSectionScoped cs(critsect_.get());
 
-  int64_t oldest_packet = packets_->OldestEnqueueTime();
+  int64_t oldest_packet = packets_->OldestEnqueueTimeMs();
   if (oldest_packet == 0)
     return 0;
 
   return clock_->TimeInMilliseconds() - oldest_packet;
 }
 
+int64_t PacedSender::AverageQueueTimeMs() {
+  CriticalSectionScoped cs(critsect_.get());
+  packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
+  return packets_->AverageQueueTimeMs();
+}
+
 int64_t PacedSender::TimeUntilNextProcess() {
   CriticalSectionScoped cs(critsect_.get());
   if (prober_->IsProbing()) {
     int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
-    if (ret >= 0) {
+    if (ret >= 0)
       return ret;
-    }
   }
   int64_t elapsed_time_us = clock_->TimeInMicroseconds() - time_last_update_us_;
   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
@@ -325,45 +358,71 @@ int32_t PacedSender::Process() {
   CriticalSectionScoped cs(critsect_.get());
   int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
   time_last_update_us_ = now_us;
-  if (!enabled_) {
+  int target_bitrate_kbps = max_bitrate_kbps_;
+  // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
+  if (!paused_ && elapsed_time_ms > 0) {
+    size_t queue_size_bytes = packets_->SizeInBytes();
+    if (queue_size_bytes > 0) {
+      // Assuming equal size packets and input/output rate, the average packet
+      // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
+      // time constraint shall be met. Determine bitrate needed for that.
+      packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
+      int64_t avg_time_left_ms = std::max<int64_t>(
+          1, kMaxQueueLengthMs - packets_->AverageQueueTimeMs());
+      int min_bitrate_needed_kbps =
+          static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
+      if (min_bitrate_needed_kbps > target_bitrate_kbps)
+        target_bitrate_kbps = min_bitrate_needed_kbps;
+    }
+
+    media_budget_->set_target_rate_kbps(target_bitrate_kbps);
+
+    int64_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms);
+    UpdateBytesPerInterval(delta_time_ms);
+  }
+  while (!packets_->Empty()) {
+    if (media_budget_->bytes_remaining() == 0 && !prober_->IsProbing())
+      return 0;
+
+    // Since we need to release the lock in order to send, we first pop the
+    // element from the priority queue but keep it in storage, so that we can
+    // reinsert it if send fails.
+    const paced_sender::Packet& packet = packets_->BeginPop();
+
+    if (SendPacket(packet)) {
+      // Send succeeded, remove it from the queue.
+      packets_->FinalizePop(packet);
+      if (prober_->IsProbing())
+        return 0;
+    } else {
+      // Send failed, put it back into the queue.
+      packets_->CancelPop(packet);
+      return 0;
+    }
+  }
+
+  // TODO(holmer): Remove the paused_ check when issue 5307 has been fixed.
+  if (paused_ || !packets_->Empty())
     return 0;
-  }
-  if (!paused_) {
-    if (elapsed_time_ms > 0) {
-      int64_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms);
-      UpdateBytesPerInterval(delta_time_ms);
-    }
-    while (!packets_->Empty()) {
-      if (media_budget_->bytes_remaining() <= 0 && !prober_->IsProbing()) {
-        return 0;
-      }
 
-      // Since we need to release the lock in order to send, we first pop the
-      // element from the priority queue but keep it in storage, so that we can
-      // reinsert it if send fails.
-      const paced_sender::Packet& packet = packets_->BeginPop();
-      if (SendPacket(packet)) {
-        // Send succeeded, remove it from the queue.
-        packets_->FinalizePop(packet);
-        if (prober_->IsProbing()) {
-          return 0;
-        }
-      } else {
-        // Send failed, put it back into the queue.
-        packets_->CancelPop(packet);
-        return 0;
-      }
-    }
-
-    int padding_needed = padding_budget_->bytes_remaining();
-    if (padding_needed > 0) {
-      SendPadding(static_cast<size_t>(padding_needed));
-    }
+  size_t padding_needed;
+  if (prober_->IsProbing()) {
+    padding_needed = prober_->RecommendedPacketSize();
+  } else {
+    padding_needed = padding_budget_->bytes_remaining();
   }
+
+  if (padding_needed > 0)
+    SendPadding(static_cast<size_t>(padding_needed));
   return 0;
 }
 
 bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
+  // TODO(holmer): Because of this bug issue 5307 we have to send audio
+  // packets even when the pacer is paused. Here we assume audio packets are
+  // always high priority and that they are the only high priority packets.
+  if (paused_ && packet.priority != kHighPriority)
+    return false;
   critsect_->Leave();
   const bool success = callback_->TimeToSendPacket(packet.ssrc,
                                                    packet.sequence_number,
@@ -371,7 +430,9 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
                                                    packet.retransmission);
   critsect_->Enter();
 
-  if (success) {
+  // TODO(holmer): High priority packets should only be accounted for if we are
+  // allocating bandwidth for audio.
+  if (success && packet.priority != kHighPriority) {
     // Update media bytes sent.
     prober_->PacketSent(clock_->TimeInMilliseconds(), packet.bytes);
     media_budget_->UseBudget(packet.bytes);
@@ -386,9 +447,11 @@ void PacedSender::SendPadding(size_t padding_needed) {
   size_t bytes_sent = callback_->TimeToSendPadding(padding_needed);
   critsect_->Enter();
 
-  // Update padding bytes sent.
-  media_budget_->UseBudget(bytes_sent);
-  padding_budget_->UseBudget(bytes_sent);
+  if (bytes_sent > 0) {
+    prober_->PacketSent(clock_->TimeInMilliseconds(), bytes_sent);
+    media_budget_->UseBudget(bytes_sent);
+    padding_budget_->UseBudget(bytes_sent);
+  }
 }
 
 void PacedSender::UpdateBytesPerInterval(int64_t delta_time_ms) {
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/include/paced_sender.h b/media/webrtc/trunk/webrtc/modules/pacing/paced_sender.h
similarity index 79%
rename from media/webrtc/trunk/webrtc/modules/pacing/include/paced_sender.h
rename to media/webrtc/trunk/webrtc/modules/pacing/paced_sender.h
index 645999d507..62e794fdbc 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/include/paced_sender.h
+++ b/media/webrtc/trunk/webrtc/modules/pacing/paced_sender.h
@@ -8,15 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
+#ifndef WEBRTC_MODULES_PACING_PACED_SENDER_H_
+#define WEBRTC_MODULES_PACING_PACED_SENDER_H_
 
 #include <list>
 #include <set>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -30,16 +31,8 @@ struct Packet;
 class PacketQueue;
 }  // namespace paced_sender
 
-class PacedSender : public Module {
+class PacedSender : public Module, public RtpPacketSender {
  public:
-  enum Priority {
-    kHighPriority = 0,  // Pass through; will be sent immediately.
-    kNormalPriority = 2,  // Put in back of the line.
-    kLowPriority = 3,  // Put in back of the low priority line.
-  };
-  // Low priority packets are mixed with the normal priority packets
-  // while we are paused.
-
   class Callback {
    public:
     // Note: packets sent as a result of a callback should not pass by this
@@ -58,7 +51,11 @@ class PacedSender : public Module {
     virtual ~Callback() {}
   };
 
-  static const int64_t kDefaultMaxQueueLengthMs = 2000;
+  // Expected max pacer delay in ms. If ExpectedQueueTimeMs() is higher than
+  // this value, the packet producers should wait (eg drop frames rather than
+  // encoding them). Bitrate sent may temporarily exceed target set by
+  // UpdateBitrate() so that this limit will be upheld.
+  static const int64_t kMaxQueueLengthMs;
   // Pace in kbits/s until we receive first estimate.
   static const int kDefaultInitialPaceKbps = 2000;
   // Pacing-rate relative to our target send rate.
@@ -68,6 +65,8 @@ class PacedSender : public Module {
   // overshoots from the encoder.
   static const float kDefaultPaceMultiplier;
 
+  static const size_t kMinProbePacketSize = 200;
+
   PacedSender(Clock* clock,
               Callback* callback,
               int bitrate_kbps,
@@ -76,11 +75,6 @@ class PacedSender : public Module {
 
   virtual ~PacedSender();
 
-  // Enable/disable pacing.
-  void SetStatus(bool enable);
-
-  bool Enabled() const;
-
   // Temporarily pause all sending.
   void Pause();
 
@@ -103,12 +97,12 @@ class PacedSender : public Module {
 
   // Returns true if we send the packet now, else it will add the packet
   // information to the queue and call TimeToSendPacket when it's time to send.
-  virtual bool SendPacket(Priority priority,
-                          uint32_t ssrc,
-                          uint16_t sequence_number,
-                          int64_t capture_time_ms,
-                          size_t bytes,
-                          bool retransmission);
+  void InsertPacket(RtpPacketSender::Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission) override;
 
   // Returns the time since the oldest queued packet was enqueued.
   virtual int64_t QueueInMs() const;
@@ -119,6 +113,10 @@ class PacedSender : public Module {
   // packets in the queue, given the current size and bitrate, ignoring prio.
   virtual int64_t ExpectedQueueTimeMs() const;
 
+  // Returns the average time since being enqueued, in milliseconds, for all
+  // packets currently in the pacer queue, or 0 if queue is empty.
+  virtual int64_t AverageQueueTimeMs();
+
   // Returns the number of milliseconds until the module want a worker thread
   // to call Process.
   int64_t TimeUntilNextProcess() override;
@@ -139,7 +137,6 @@ class PacedSender : public Module {
   Callback* const callback_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
-  bool enabled_ GUARDED_BY(critsect_);
   bool paused_ GUARDED_BY(critsect_);
   bool probing_enabled_;
   // This is the media budget, keeping track of how many bits of media
@@ -153,7 +150,10 @@ class PacedSender : public Module {
       GUARDED_BY(critsect_);
 
   rtc::scoped_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
+  // Actual configured bitrates (media_budget_ may temporarily be higher in
+  // order to meet pace time constraint).
   int bitrate_bps_ GUARDED_BY(critsect_);
+  int max_bitrate_kbps_ GUARDED_BY(critsect_);
 
   int64_t time_last_update_us_ GUARDED_BY(critsect_);
 
@@ -161,4 +161,4 @@ class PacedSender : public Module {
   uint64_t packet_counter_;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
+#endif  // WEBRTC_MODULES_PACING_PACED_SENDER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/paced_sender_unittest.cc b/media/webrtc/trunk/webrtc/modules/pacing/paced_sender_unittest.cc
index e82a49e7cc..588bf3b669 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/paced_sender_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/pacing/paced_sender_unittest.cc
@@ -12,8 +12,8 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 using testing::_;
 using testing::Return;
@@ -71,22 +71,26 @@ class PacedSenderProbing : public PacedSender::Callback {
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
                         bool retransmission) {
+    ExpectAndCountPacket();
+    return true;
+  }
+
+  size_t TimeToSendPadding(size_t bytes) {
+    ExpectAndCountPacket();
+    return bytes;
+  }
+
+  void ExpectAndCountPacket() {
     ++packets_sent_;
     EXPECT_FALSE(expected_deltas_.empty());
     if (expected_deltas_.empty())
-      return false;
+      return;
     int64_t now_ms = clock_->TimeInMilliseconds();
     if (prev_packet_time_ms_ >= 0) {
       EXPECT_EQ(expected_deltas_.front(), now_ms - prev_packet_time_ms_);
       expected_deltas_.pop_front();
     }
     prev_packet_time_ms_ = now_ms;
-    return true;
-  }
-
-  size_t TimeToSendPadding(size_t bytes) {
-    EXPECT_TRUE(false);
-    return bytes;
   }
 
   int packets_sent() const { return packets_sent_; }
@@ -120,8 +124,8 @@ class PacedSenderTest : public ::testing::Test {
                            int64_t capture_time_ms,
                            size_t size,
                            bool retransmission) {
-    EXPECT_FALSE(send_bucket_->SendPacket(priority, ssrc,
-        sequence_number, capture_time_ms, size, retransmission));
+    send_bucket_->InsertPacket(priority, ssrc, sequence_number, capture_time_ms,
+                               size, retransmission);
     EXPECT_CALL(callback_,
                 TimeToSendPacket(ssrc, sequence_number, capture_time_ms, false))
         .Times(1)
@@ -156,8 +160,9 @@ TEST_F(PacedSenderTest, QueuePacket) {
                       250,
                       false);
   int64_t queued_packet_timestamp = clock_.TimeInMilliseconds();
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-      sequence_number, queued_packet_timestamp, 250, false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number, queued_packet_timestamp, 250,
+                             false);
   send_bucket_->Process();
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
@@ -184,8 +189,9 @@ TEST_F(PacedSenderTest, QueuePacket) {
                       clock_.TimeInMilliseconds(),
                       250,
                       false);
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-      sequence_number++, clock_.TimeInMilliseconds(), 250, false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, clock_.TimeInMilliseconds(),
+                             250, false);
   send_bucket_->Process();
 }
 
@@ -203,8 +209,9 @@ TEST_F(PacedSenderTest, PaceQueuedPackets) {
                         false);
   }
   for (int j = 0; j < 30; ++j) {
-    EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-        sequence_number++, clock_.TimeInMilliseconds(), 250, false));
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number++, clock_.TimeInMilliseconds(),
+                               250, false);
   }
   send_bucket_->Process();
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
@@ -239,8 +246,9 @@ TEST_F(PacedSenderTest, PaceQueuedPackets) {
                       clock_.TimeInMilliseconds(),
                       250,
                       false);
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-      sequence_number, clock_.TimeInMilliseconds(), 250, false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number, clock_.TimeInMilliseconds(), 250,
+                             false);
   send_bucket_->Process();
 }
 
@@ -262,10 +270,12 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
 
   for (int j = 0; j < 30; ++j) {
     // Send in duplicate packets.
-    EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-        sequence_number, clock_.TimeInMilliseconds(), 250, false));
-    EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-        sequence_number++, clock_.TimeInMilliseconds(), 250, false));
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number, clock_.TimeInMilliseconds(),
+                               250, false);
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number++, clock_.TimeInMilliseconds(),
+                               250, false);
   }
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
   send_bucket_->Process();
@@ -304,8 +314,9 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
                       clock_.TimeInMilliseconds(),
                       250,
                       false);
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-      sequence_number++, clock_.TimeInMilliseconds(), 250, false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, clock_.TimeInMilliseconds(),
+                             250, false);
   send_bucket_->Process();
 }
 
@@ -373,23 +384,6 @@ TEST_F(PacedSenderTest, Padding) {
   EXPECT_EQ(0, send_bucket_->Process());
 }
 
-TEST_F(PacedSenderTest, NoPaddingWhenDisabled) {
-  send_bucket_->SetStatus(false);
-  send_bucket_->UpdateBitrate(
-      kTargetBitrate, kPaceMultiplier * kTargetBitrate, kTargetBitrate);
-  // No padding is expected since the pacer is disabled.
-  EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
-  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
-  clock_.AdvanceTimeMilliseconds(5);
-  EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
-  EXPECT_EQ(0, send_bucket_->Process());
-  EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
-  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
-  clock_.AdvanceTimeMilliseconds(5);
-  EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
-  EXPECT_EQ(0, send_bucket_->Process());
-}
-
 TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) {
   uint32_t ssrc = 12345;
   uint16_t sequence_number = 1234;
@@ -429,9 +423,9 @@ TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
   size_t media_bytes = 0;
   while (clock_.TimeInMilliseconds() - start_time < kBitrateWindow) {
     size_t media_payload = rand() % 100 + 200;  // [200, 300] bytes.
-    EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority, ssrc,
-                                          sequence_number++, capture_time_ms,
-                                          media_payload, false));
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number++, capture_time_ms,
+                               media_payload, false);
     media_bytes += media_payload;
     clock_.AdvanceTimeMilliseconds(kTimeStep);
     send_bucket_->Process();
@@ -470,20 +464,22 @@ TEST_F(PacedSenderTest, Priority) {
   send_bucket_->Process();
 
   // Expect normal and low priority to be queued and high to pass through.
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kLowPriority,
-      ssrc_low_priority, sequence_number++, capture_time_ms_low_priority, 250,
-      false));
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-      ssrc, sequence_number++, capture_time_ms, 250, false));
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-      ssrc, sequence_number++, capture_time_ms, 250, false));
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kHighPriority,
-      ssrc, sequence_number++, capture_time_ms, 250, false));
+  send_bucket_->InsertPacket(PacedSender::kLowPriority, ssrc_low_priority,
+                             sequence_number++, capture_time_ms_low_priority,
+                             250, false);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
+  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
 
   // Expect all high and normal priority to be sent out first.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
   EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, capture_time_ms, false))
-      .Times(3)
+      .Times(4)
       .WillRepeatedly(Return(true));
 
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
@@ -503,6 +499,37 @@ TEST_F(PacedSenderTest, Priority) {
   EXPECT_EQ(0, send_bucket_->Process());
 }
 
+TEST_F(PacedSenderTest, HighPrioDoesntAffectBudget) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 56789;
+
+  // As high prio packets doesn't affect the budget, we should be able to send
+  // a high number of them at once.
+  for (int i = 0; i < 25; ++i) {
+    SendAndExpectPacket(PacedSender::kHighPriority, ssrc, sequence_number++,
+                        capture_time_ms, 250, false);
+  }
+  send_bucket_->Process();
+  // Low prio packets does affect the budget, so we should only be able to send
+  // 3 at once, the 4th should be queued.
+  for (int i = 0; i < 3; ++i) {
+    SendAndExpectPacket(PacedSender::kLowPriority, ssrc, sequence_number++,
+                        capture_time_ms, 250, false);
+  }
+  send_bucket_->InsertPacket(PacedSender::kLowPriority, ssrc, sequence_number,
+                             capture_time_ms, 250, false);
+  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
+  clock_.AdvanceTimeMilliseconds(5);
+  send_bucket_->Process();
+  EXPECT_CALL(callback_,
+              TimeToSendPacket(ssrc, sequence_number++, capture_time_ms, false))
+      .Times(1);
+  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
+  clock_.AdvanceTimeMilliseconds(5);
+  send_bucket_->Process();
+}
+
 TEST_F(PacedSenderTest, Pause) {
   uint32_t ssrc_low_priority = 12345;
   uint32_t ssrc = 12346;
@@ -534,20 +561,20 @@ TEST_F(PacedSenderTest, Pause) {
 
   send_bucket_->Pause();
 
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-      ssrc, sequence_number++, capture_time_ms, 250, false));
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-      ssrc, sequence_number++, capture_time_ms, 250, false));
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kHighPriority,
-      ssrc, sequence_number++, capture_time_ms, 250, false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
+  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
+                             sequence_number++, capture_time_ms, 250, false);
 
   clock_.AdvanceTimeMilliseconds(10000);
   int64_t second_capture_time_ms = clock_.TimeInMilliseconds();
 
   // Expect everything to be queued.
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kLowPriority,
-      ssrc_low_priority, sequence_number++, second_capture_time_ms, 250,
-      false));
+  send_bucket_->InsertPacket(PacedSender::kLowPriority, ssrc_low_priority,
+                             sequence_number++, second_capture_time_ms, 250,
+                             false);
 
   EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms,
             send_bucket_->QueueInMs());
@@ -566,6 +593,9 @@ TEST_F(PacedSenderTest, Pause) {
   EXPECT_CALL(callback_, TimeToSendPacket(_, _, capture_time_ms, false))
       .Times(3)
       .WillRepeatedly(Return(true));
+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, second_capture_time_ms, false))
+      .Times(1)
+      .WillRepeatedly(Return(true));
   send_bucket_->Resume();
 
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
@@ -573,13 +603,6 @@ TEST_F(PacedSenderTest, Pause) {
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
   EXPECT_EQ(0, send_bucket_->Process());
 
-  EXPECT_CALL(callback_, TimeToSendPacket(_, _, second_capture_time_ms, false))
-      .Times(1)
-      .WillRepeatedly(Return(true));
-  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
-  clock_.AdvanceTimeMilliseconds(5);
-  EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
-  EXPECT_EQ(0, send_bucket_->Process());
   EXPECT_EQ(0, send_bucket_->QueueInMs());
 }
 
@@ -589,19 +612,12 @@ TEST_F(PacedSenderTest, ResendPacket) {
   int64_t capture_time_ms = clock_.TimeInMilliseconds();
   EXPECT_EQ(0, send_bucket_->QueueInMs());
 
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-                                        ssrc,
-                                        sequence_number,
-                                        capture_time_ms,
-                                        250,
-                                        false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number, capture_time_ms, 250, false);
   clock_.AdvanceTimeMilliseconds(1);
-  EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-                                        ssrc,
-                                        sequence_number + 1,
-                                        capture_time_ms + 1,
-                                        250,
-                                        false));
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number + 1, capture_time_ms + 1, 250,
+                             false);
   clock_.AdvanceTimeMilliseconds(9999);
   EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms,
             send_bucket_->QueueInMs());
@@ -677,10 +693,9 @@ TEST_F(PacedSenderTest, ExpectedQueueTimeMs) {
 
   EXPECT_EQ(0, send_bucket_->ExpectedQueueTimeMs());
 
-  // Allow for aliasing, duration should be in [expected(n - 1), expected(n)].
-  EXPECT_LE(duration, queue_in_ms);
-  EXPECT_GE(duration,
-            queue_in_ms - static_cast<int64_t>(kPacketSize * 8 / kMaxBitrate));
+  // Allow for aliasing, duration should be within one pack of max time limit.
+  EXPECT_NEAR(duration, PacedSender::kMaxQueueLengthMs,
+              static_cast<int64_t>(kPacketSize * 8 / kMaxBitrate));
 }
 
 TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) {
@@ -714,8 +729,6 @@ TEST_F(PacedSenderTest, ProbingWithInitialFrame) {
   std::list<int> expected_deltas_list(expected_deltas,
                                       expected_deltas + kNumPackets - 1);
   PacedSenderProbing callback(expected_deltas_list, &clock_);
-  // Probing implicitly enabled by creating a new PacedSender which defaults to
-  // probing on.
   send_bucket_.reset(
       new PacedSender(&clock_,
                       &callback,
@@ -724,12 +737,9 @@ TEST_F(PacedSenderTest, ProbingWithInitialFrame) {
                       0));
 
   for (int i = 0; i < kNumPackets; ++i) {
-    EXPECT_FALSE(send_bucket_->SendPacket(PacedSender::kNormalPriority,
-                                          ssrc,
-                                          sequence_number++,
-                                          clock_.TimeInMilliseconds(),
-                                          kPacketSize,
-                                          false));
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number++, clock_.TimeInMilliseconds(),
+                               kPacketSize, false);
   }
   while (callback.packets_sent() < kNumPackets) {
     int time_until_process = send_bucket_->TimeUntilNextProcess();
@@ -741,26 +751,60 @@ TEST_F(PacedSenderTest, ProbingWithInitialFrame) {
   }
 }
 
+TEST_F(PacedSenderTest, ProbingWithTooSmallInitialFrame) {
+  const int kNumPackets = 11;
+  const int kNumDeltas = kNumPackets - 1;
+  const size_t kPacketSize = 1200;
+  const int kInitialBitrateKbps = 300;
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  const int expected_deltas[kNumDeltas] = {10, 10, 10, 10, 10, 5, 5, 5, 5, 5};
+  std::list<int> expected_deltas_list(expected_deltas,
+                                      expected_deltas + kNumPackets - 1);
+  PacedSenderProbing callback(expected_deltas_list, &clock_);
+  send_bucket_.reset(new PacedSender(&clock_, &callback, kInitialBitrateKbps,
+                                     kPaceMultiplier * kInitialBitrateKbps, 0));
+
+  for (int i = 0; i < kNumPackets - 5; ++i) {
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number++, clock_.TimeInMilliseconds(),
+                               kPacketSize, false);
+  }
+  while (callback.packets_sent() < kNumPackets) {
+    int time_until_process = send_bucket_->TimeUntilNextProcess();
+    if (time_until_process <= 0) {
+      send_bucket_->Process();
+    } else {
+      clock_.AdvanceTimeMilliseconds(time_until_process);
+    }
+  }
+
+  // Process one more time and make sure we don't send any more probes.
+  int time_until_process = send_bucket_->TimeUntilNextProcess();
+  clock_.AdvanceTimeMilliseconds(time_until_process);
+  send_bucket_->Process();
+  EXPECT_EQ(kNumPackets, callback.packets_sent());
+}
+
 TEST_F(PacedSenderTest, PriorityInversion) {
   uint32_t ssrc = 12346;
   uint16_t sequence_number = 1234;
   const size_t kPacketSize = 1200;
 
-  EXPECT_FALSE(send_bucket_->SendPacket(
+  send_bucket_->InsertPacket(
       PacedSender::kHighPriority, ssrc, sequence_number + 3,
-      clock_.TimeInMilliseconds() + 33, kPacketSize, true));
+      clock_.TimeInMilliseconds() + 33, kPacketSize, true);
 
-  EXPECT_FALSE(send_bucket_->SendPacket(
+  send_bucket_->InsertPacket(
       PacedSender::kHighPriority, ssrc, sequence_number + 2,
-      clock_.TimeInMilliseconds() + 33, kPacketSize, true));
+      clock_.TimeInMilliseconds() + 33, kPacketSize, true);
 
-  EXPECT_FALSE(send_bucket_->SendPacket(
-      PacedSender::kHighPriority, ssrc, sequence_number,
-      clock_.TimeInMilliseconds(), kPacketSize, true));
+  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc, sequence_number,
+                             clock_.TimeInMilliseconds(), kPacketSize, true);
 
-  EXPECT_FALSE(send_bucket_->SendPacket(
-      PacedSender::kHighPriority, ssrc, sequence_number + 1,
-      clock_.TimeInMilliseconds(), kPacketSize, true));
+  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
+                             sequence_number + 1, clock_.TimeInMilliseconds(),
+                             kPacketSize, true);
 
   // Packets from earlier frames should be sent first.
   {
@@ -805,14 +849,60 @@ TEST_F(PacedSenderTest, PaddingOveruse) {
   clock_.AdvanceTimeMilliseconds(5);
   send_bucket_->UpdateBitrate(60, 90, 30);
 
-  EXPECT_FALSE(send_bucket_->SendPacket(
-      PacedSender::kHighPriority, ssrc, sequence_number++,
-      clock_.TimeInMilliseconds(), kPacketSize, false));
+  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
+                             sequence_number++, clock_.TimeInMilliseconds(),
+                             kPacketSize, false);
 
   // Don't send padding if queue is non-empty, even if padding budget > 0.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
   send_bucket_->Process();
 }
 
+TEST_F(PacedSenderTest, AverageQueueTime) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 1200;
+  const int kBitrateBps = 10 * kPacketSize * 8;  // 10 packets per second.
+  const int kBitrateKbps = (kBitrateBps + 500) / 1000;
+
+  send_bucket_->UpdateBitrate(kBitrateKbps, kBitrateKbps, kBitrateKbps);
+
+  EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
+
+  int64_t first_capture_time = clock_.TimeInMilliseconds();
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number, first_capture_time, kPacketSize,
+                             false);
+  clock_.AdvanceTimeMilliseconds(10);
+  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                             sequence_number + 1, clock_.TimeInMilliseconds(),
+                             kPacketSize, false);
+  clock_.AdvanceTimeMilliseconds(10);
+
+  EXPECT_EQ((20 + 10) / 2, send_bucket_->AverageQueueTimeMs());
+
+  // Only first packet (queued for 20ms) should be removed, leave the second
+  // packet (queued for 10ms) alone in the queue.
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
+                                          first_capture_time, false))
+      .Times(1)
+      .WillRepeatedly(Return(true));
+  send_bucket_->Process();
+
+  EXPECT_EQ(10, send_bucket_->AverageQueueTimeMs());
+
+  clock_.AdvanceTimeMilliseconds(10);
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 1,
+                                          first_capture_time + 10, false))
+      .Times(1)
+      .WillRepeatedly(Return(true));
+  for (int i = 0; i < 3; ++i) {
+    clock_.AdvanceTimeMilliseconds(30);  // Max delta.
+    send_bucket_->Process();
+  }
+
+  EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/pacing.gypi b/media/webrtc/trunk/webrtc/modules/pacing/pacing.gypi
index 09be38f414..90f663c1b0 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/pacing.gypi
+++ b/media/webrtc/trunk/webrtc/modules/pacing/pacing.gypi
@@ -13,14 +13,16 @@
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
+        '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       ],
       'sources': [
-        'include/paced_sender.h',
-        'include/packet_router.h',
         'bitrate_prober.cc',
         'bitrate_prober.h',
         'paced_sender.cc',
+        'paced_sender.h',
         'packet_router.cc',
+        'packet_router.h',
       ],
     },
   ], # targets
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/packet_router.cc b/media/webrtc/trunk/webrtc/modules/pacing/packet_router.cc
index 9e15a71317..5fd350834a 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/packet_router.cc
+++ b/media/webrtc/trunk/webrtc/modules/pacing/packet_router.cc
@@ -8,39 +8,42 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/pacing/include/packet_router.h"
+#include "webrtc/modules/pacing/packet_router.h"
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
 
-PacketRouter::PacketRouter()
-    : crit_(CriticalSectionWrapper::CreateCriticalSection()) {
+PacketRouter::PacketRouter() : transport_seq_(0) {
 }
 
 PacketRouter::~PacketRouter() {
+  RTC_DCHECK(rtp_modules_.empty());
 }
 
 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
-  CriticalSectionScoped cs(crit_.get());
-  DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
-         rtp_modules_.end());
+  rtc::CritScope cs(&modules_lock_);
+  RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
+             rtp_modules_.end());
   rtp_modules_.push_back(rtp_module);
 }
 
 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
-  CriticalSectionScoped cs(crit_.get());
-  rtp_modules_.remove(rtp_module);
+  rtc::CritScope cs(&modules_lock_);
+  auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module);
+  RTC_DCHECK(it != rtp_modules_.end());
+  rtp_modules_.erase(it);
 }
 
 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
                                     uint16_t sequence_number,
                                     int64_t capture_timestamp,
                                     bool retransmission) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&modules_lock_);
   for (auto* rtp_module : rtp_modules_) {
     if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
       return rtp_module->TimeToSendPacket(ssrc, sequence_number,
@@ -50,12 +53,51 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
   return true;
 }
 
-size_t PacketRouter::TimeToSendPadding(size_t bytes) {
-  CriticalSectionScoped cs(crit_.get());
-  for (auto* rtp_module : rtp_modules_) {
-    if (rtp_module->SendingMedia())
-      return rtp_module->TimeToSendPadding(bytes);
+size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
+  size_t total_bytes_sent = 0;
+  rtc::CritScope cs(&modules_lock_);
+  for (RtpRtcp* module : rtp_modules_) {
+    if (module->SendingMedia()) {
+      size_t bytes_sent =
+          module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
+      total_bytes_sent += bytes_sent;
+      if (total_bytes_sent >= bytes_to_send)
+        break;
+    }
   }
-  return 0;
+  return total_bytes_sent;
 }
+
+void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
+  rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
+}
+
+uint16_t PacketRouter::AllocateSequenceNumber() {
+  int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
+  int desired_prev_seq;
+  int new_seq;
+  do {
+    desired_prev_seq = prev_seq;
+    new_seq = (desired_prev_seq + 1) & 0xFFFF;
+    // Note: CompareAndSwap returns the actual value of transport_seq at the
+    // time the CAS operation was executed. Thus, if prev_seq is returned, the
+    // operation was successful - otherwise we need to retry. Saving the
+    // return value saves us a load on retry.
+    prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
+                                              new_seq);
+  } while (prev_seq != desired_prev_seq);
+
+  return new_seq;
+}
+
+bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
+  rtc::CritScope cs(&modules_lock_);
+  for (auto* rtp_module : rtp_modules_) {
+    packet->WithPacketSenderSsrc(rtp_module->SSRC());
+    if (rtp_module->SendFeedbackPacket(*packet))
+      return true;
+  }
+  return false;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/include/packet_router.h b/media/webrtc/trunk/webrtc/modules/pacing/packet_router.h
similarity index 58%
rename from media/webrtc/trunk/webrtc/modules/pacing/include/packet_router.h
rename to media/webrtc/trunk/webrtc/modules/pacing/packet_router.h
index c1b332a6bf..edef1aa9b3 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/include/packet_router.h
+++ b/media/webrtc/trunk/webrtc/modules/pacing/packet_router.h
@@ -8,27 +8,30 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
+#ifndef WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
+#define WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
 
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-class RTPFragmentationHeader;
 class RtpRtcp;
-struct RTPVideoHeader;
+namespace rtcp {
+class TransportFeedback;
+}  // namespace rtcp
 
 // PacketRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
-class PacketRouter : public PacedSender::Callback {
+class PacketRouter : public PacedSender::Callback,
+                     public TransportSequenceNumberAllocator {
  public:
   PacketRouter();
   virtual ~PacketRouter();
@@ -44,16 +47,20 @@ class PacketRouter : public PacedSender::Callback {
 
   size_t TimeToSendPadding(size_t bytes) override;
 
+  void SetTransportWideSequenceNumber(uint16_t sequence_number);
+  uint16_t AllocateSequenceNumber() override;
+
+  // Send transport feedback packet to send-side.
+  virtual bool SendFeedback(rtcp::TransportFeedback* packet);
+
  private:
-  // TODO(holmer): When the new video API has launched, remove crit_ and
-  // assume rtp_modules_ will never change during a call. We should then also
-  // switch rtp_modules_ to a map from ssrc to rtp module.
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-
+  rtc::CriticalSection modules_lock_;
   // Map from ssrc to sending rtp module.
-  std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
+  std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_lock_);
 
-  DISALLOW_COPY_AND_ASSIGN(PacketRouter);
+  volatile int transport_seq_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
+#endif  // WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/pacing/packet_router_unittest.cc b/media/webrtc/trunk/webrtc/modules/pacing/packet_router_unittest.cc
index f7fdf7bbca..31acf44b9b 100644
--- a/media/webrtc/trunk/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/pacing/packet_router_unittest.cc
@@ -13,8 +13,8 @@
 #include "webrtc/base/checks.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/base/scoped_ptr.h"
 
@@ -102,20 +102,30 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
 }
 
 TEST_F(PacketRouterTest, TimeToSendPadding) {
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+
   MockRtpRtcp rtp_1;
+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
   MockRtpRtcp rtp_2;
+  EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
   packet_router_->AddRtpModule(&rtp_1);
   packet_router_->AddRtpModule(&rtp_2);
 
-  // Default configuration, sending padding on the first sending module.
+  // Default configuration, sending padding on all modules sending media,
+  // ordered by SSRC.
   const size_t requested_padding_bytes = 1000;
   const size_t sent_padding_bytes = 890;
   EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
   EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
       .Times(1)
       .WillOnce(Return(sent_padding_bytes));
-  EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
-  EXPECT_EQ(sent_padding_bytes,
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2,
+              TimeToSendPadding(requested_padding_bytes - sent_padding_bytes))
+      .Times(1)
+      .WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
+  EXPECT_EQ(requested_padding_bytes,
             packet_router_->TimeToSendPadding(requested_padding_bytes));
 
   // Let only the second module be sending and verify the padding request is
@@ -134,8 +144,7 @@ TEST_F(PacketRouterTest, TimeToSendPadding) {
   EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
   EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
   EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
-  EXPECT_EQ(static_cast<size_t>(0),
-            packet_router_->TimeToSendPadding(requested_padding_bytes));
+  EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
 
   packet_router_->RemoveRtpModule(&rtp_1);
 
@@ -143,9 +152,21 @@ TEST_F(PacketRouterTest, TimeToSendPadding) {
   // to send by not expecting any calls. Instead verify rtp_2 is called.
   EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
   EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1);
-  EXPECT_EQ(static_cast<size_t>(0),
-            packet_router_->TimeToSendPadding(requested_padding_bytes));
+  EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
 
   packet_router_->RemoveRtpModule(&rtp_2);
 }
+
+TEST_F(PacketRouterTest, AllocateSequenceNumbers) {
+  const uint16_t kStartSeq = 0xFFF0;
+  const size_t kNumPackets = 32;
+
+  packet_router_->SetTransportWideSequenceNumber(kStartSeq - 1);
+
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    uint16_t seq = packet_router_->AllocateSequenceNumber();
+    uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
+    EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq);
+  }
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/BUILD.gn
index fe06c6d777..99c297dda6 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -14,7 +14,7 @@ source_set("remote_bitrate_estimator") {
     "rate_statistics.h",
   ]
 
-  configs += [ "../../:common_inherited_config"]
+  configs += [ "../../:common_inherited_config" ]
 
   deps = [
     ":rbe_components",
@@ -27,23 +27,27 @@ source_set("rbe_components") {
   sources = [
     "aimd_rate_control.cc",
     "aimd_rate_control.h",
+    "include/send_time_history.h",
     "inter_arrival.cc",
     "inter_arrival.h",
-    "mimd_rate_control.cc",
-    "mimd_rate_control.h",
     "overuse_detector.cc",
     "overuse_detector.h",
     "overuse_estimator.cc",
     "overuse_estimator.h",
     "remote_bitrate_estimator_abs_send_time.cc",
     "remote_bitrate_estimator_single_stream.cc",
-    "remote_rate_control.cc",
-    "remote_rate_control.h",
+    "remote_estimator_proxy.cc",
+    "remote_estimator_proxy.h",
+    "send_time_history.cc",
+    "transport_feedback_adapter.cc",
+    "transport_feedback_adapter.h",
   ]
 
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
-  deps = [ "../..:webrtc_common" ]
+  deps = [
+    "../..:webrtc_common",
+  ]
 
   if (is_clang) {
     # Suppress warnings from Chrome's Clang plugins.
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
index bed5d99cab..4820e6295f 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -14,6 +14,10 @@
 #include <cassert>
 #include <cmath>
 
+#include "webrtc/base/checks.h"
+
+#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 
 namespace webrtc {
@@ -21,32 +25,30 @@ namespace webrtc {
 static const int64_t kDefaultRttMs = 200;
 static const int64_t kLogIntervalMs = 1000;
 static const double kWithinIncomingBitrateHysteresis = 1.05;
+static const int64_t kMaxFeedbackIntervalMs = 1000;
 
-AimdRateControl::AimdRateControl(uint32_t min_bitrate_bps)
-    : min_configured_bitrate_bps_(min_bitrate_bps),
+AimdRateControl::AimdRateControl()
+    : min_configured_bitrate_bps_(
+          RemoteBitrateEstimator::kDefaultMinBitrateBps),
       max_configured_bitrate_bps_(30000000),
       current_bitrate_bps_(max_configured_bitrate_bps_),
-      max_hold_rate_bps_(0),
       avg_max_bitrate_kbps_(-1.0f),
       var_max_bitrate_kbps_(0.4f),
       rate_control_state_(kRcHold),
-      came_from_state_(kRcDecrease),
       rate_control_region_(kRcMaxUnknown),
       time_last_bitrate_change_(-1),
       current_input_(kBwNormal, 0, 1.0),
       updated_(false),
       time_first_incoming_estimate_(-1),
       bitrate_is_initialized_(false),
-      beta_(0.9f),
+      beta_(0.85f),
       rtt_(kDefaultRttMs),
-      time_of_last_log_(-1) {}
+      time_of_last_log_(-1),
+      in_experiment_(AdaptiveThresholdExperimentIsEnabled()) {}
 
-RateControlType AimdRateControl::GetControlType() const {
-  return kAimdControl;
-}
-
-uint32_t AimdRateControl::GetMinBitrate() const {
-  return min_configured_bitrate_bps_;
+void AimdRateControl::SetMinBitrate(int min_bitrate_bps) {
+  min_configured_bitrate_bps_ = min_bitrate_bps;
+  current_bitrate_bps_ = std::max<int>(min_bitrate_bps, current_bitrate_bps_);
 }
 
 bool AimdRateControl::ValidEstimate() const {
@@ -86,8 +88,7 @@ uint32_t AimdRateControl::LatestEstimate() const {
 
 uint32_t AimdRateControl::UpdateBandwidthEstimate(int64_t now_ms) {
   current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_,
-                                       current_input_._incomingBitRate,
-                                       now_ms);
+                                       current_input_.incoming_bitrate, now_ms);
   if (now_ms - time_of_last_log_ > kLogIntervalMs) {
     time_of_last_log_ = now_ms;
   }
@@ -98,34 +99,34 @@ void AimdRateControl::SetRtt(int64_t rtt) {
   rtt_ = rtt;
 }
 
-RateControlRegion AimdRateControl::Update(const RateControlInput* input,
-                                          int64_t now_ms) {
+void AimdRateControl::Update(const RateControlInput* input, int64_t now_ms) {
   assert(input);
 
   // Set the initial bit rate value to what we're receiving the first half
   // second.
   if (!bitrate_is_initialized_) {
+    const int64_t kInitializationTimeMs = 5000;
+    RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
     if (time_first_incoming_estimate_ < 0) {
-      if (input->_incomingBitRate > 0) {
+      if (input->incoming_bitrate > 0) {
         time_first_incoming_estimate_ = now_ms;
       }
-    } else if (now_ms - time_first_incoming_estimate_ > 500 &&
-               input->_incomingBitRate > 0) {
-      current_bitrate_bps_ = input->_incomingBitRate;
+    } else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs &&
+               input->incoming_bitrate > 0) {
+      current_bitrate_bps_ = input->incoming_bitrate;
       bitrate_is_initialized_ = true;
     }
   }
 
-  if (updated_ && current_input_._bwState == kBwOverusing) {
+  if (updated_ && current_input_.bw_state == kBwOverusing) {
     // Only update delay factor and incoming bit rate. We always want to react
     // on an over-use.
-    current_input_._noiseVar = input->_noiseVar;
-    current_input_._incomingBitRate = input->_incomingBitRate;
+    current_input_.noise_var = input->noise_var;
+    current_input_.incoming_bitrate = input->incoming_bitrate;
   } else {
     updated_ = true;
     current_input_ = *input;
   }
-  return rate_control_region_;
 }
 
 void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
@@ -137,11 +138,14 @@ void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
 uint32_t AimdRateControl::ChangeBitrate(uint32_t current_bitrate_bps,
                                         uint32_t incoming_bitrate_bps,
                                         int64_t now_ms) {
-  BWE_TEST_LOGGING_PLOT("estimated_incoming#1", -1,
-                        incoming_bitrate_bps / 1000);
   if (!updated_) {
     return current_bitrate_bps_;
   }
+  // An over-use should always trigger us to reduce the bitrate, even though
+  // we have not yet established our first estimate. By acting on the over-use,
+  // we will end up with a valid estimate.
+  if (!bitrate_is_initialized_ && current_input_.bw_state != kBwOverusing)
+    return current_bitrate_bps_;
   updated_ = false;
   ChangeState(current_input_, now_ms);
   // Calculated here because it's used in multiple places.
@@ -150,52 +154,35 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t current_bitrate_bps,
   // variance and the current incoming bit rate.
   const float std_max_bit_rate = sqrt(var_max_bitrate_kbps_ *
                                       avg_max_bitrate_kbps_);
-  bool fast_recovery_after_hold = false;
   switch (rate_control_state_) {
-    case kRcHold: {
-      max_hold_rate_bps_ = std::max(max_hold_rate_bps_, incoming_bitrate_bps);
+    case kRcHold:
       break;
-    }
-    case kRcIncrease: {
-      if (avg_max_bitrate_kbps_ >= 0) {
-        if (incoming_bitrate_kbps > avg_max_bitrate_kbps_ +
-            3 * std_max_bit_rate) {
-          ChangeRegion(kRcMaxUnknown);
-          avg_max_bitrate_kbps_ = -1.0;
-        } else if (incoming_bitrate_kbps > avg_max_bitrate_kbps_ +
-            2.5 * std_max_bit_rate) {
-          ChangeRegion(kRcAboveMax);
-        }
+
+    case kRcIncrease:
+      if (avg_max_bitrate_kbps_ >= 0 &&
+          incoming_bitrate_kbps >
+              avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
+        ChangeRegion(kRcMaxUnknown);
+        avg_max_bitrate_kbps_ = -1.0;
       }
       if (rate_control_region_ == kRcNearMax) {
         // Approximate the over-use estimator delay to 100 ms.
         const int64_t response_time = rtt_ + 100;
         uint32_t additive_increase_bps = AdditiveRateIncrease(
             now_ms, time_last_bitrate_change_, response_time);
-        BWE_TEST_LOGGING_PLOT("add_increase#1", -1,
-                              additive_increase_bps / 1000);
         current_bitrate_bps += additive_increase_bps;
 
       } else {
         uint32_t multiplicative_increase_bps = MultiplicativeRateIncrease(
             now_ms, time_last_bitrate_change_, current_bitrate_bps);
-        BWE_TEST_LOGGING_PLOT("mult_increase#1", -1,
-                              multiplicative_increase_bps / 1000);
         current_bitrate_bps += multiplicative_increase_bps;
       }
 
-      if (max_hold_rate_bps_ > 0 &&
-          beta_ * max_hold_rate_bps_ > current_bitrate_bps) {
-        current_bitrate_bps = static_cast<uint32_t>(beta_ * max_hold_rate_bps_);
-        avg_max_bitrate_kbps_ = beta_ * max_hold_rate_bps_ / 1000.0f;
-        ChangeRegion(kRcNearMax);
-        fast_recovery_after_hold = true;
-      }
-      max_hold_rate_bps_ = 0;
       time_last_bitrate_change_ = now_ms;
       break;
-    }
-    case kRcDecrease: {
+
+    case kRcDecrease:
+      bitrate_is_initialized_ = true;
       if (incoming_bitrate_bps < min_configured_bitrate_bps_) {
         current_bitrate_bps = min_configured_bitrate_bps_;
       } else {
@@ -225,12 +212,11 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t current_bitrate_bps,
       ChangeState(kRcHold);
       time_last_bitrate_change_ = now_ms;
       break;
-    }
+
     default:
       assert(false);
   }
-  if (!fast_recovery_after_hold && (incoming_bitrate_bps > 100000 ||
-      current_bitrate_bps > 150000) &&
+  if ((incoming_bitrate_bps > 100000 || current_bitrate_bps > 150000) &&
       current_bitrate_bps > 1.5 * incoming_bitrate_bps) {
     // Allow changing the bit rate if we are operating at very low rates
     // Don't change the bit rate if the send side is too far off
@@ -258,8 +244,10 @@ uint32_t AimdRateControl::AdditiveRateIncrease(
   assert(response_time_ms > 0);
   double beta = 0.0;
   if (last_ms > 0) {
-    beta = std::min((now_ms - last_ms) /
-                    static_cast<double>(response_time_ms), 1.0);
+    beta = std::min((now_ms - last_ms) / static_cast<double>(response_time_ms),
+                    1.0);
+    if (in_experiment_)
+      beta /= 2.0;
   }
   double bits_per_frame = static_cast<double>(current_bitrate_bps_) / 30.0;
   double packets_per_frame = std::ceil(bits_per_frame / (8.0 * 1200.0));
@@ -295,7 +283,7 @@ void AimdRateControl::UpdateMaxBitRateEstimate(float incoming_bitrate_kbps) {
 
 void AimdRateControl::ChangeState(const RateControlInput& input,
                                   int64_t now_ms) {
-  switch (current_input_._bwState) {
+  switch (current_input_.bw_state) {
     case kBwNormal:
       if (rate_control_state_ == kRcHold) {
         time_last_bitrate_change_ = now_ms;
@@ -317,21 +305,9 @@ void AimdRateControl::ChangeState(const RateControlInput& input,
 
 void AimdRateControl::ChangeRegion(RateControlRegion region) {
   rate_control_region_ = region;
-  switch (rate_control_region_) {
-    case kRcAboveMax:
-    case kRcMaxUnknown:
-      beta_ = 0.9f;
-      break;
-    case kRcNearMax:
-      beta_ = 0.95f;
-      break;
-    default:
-      assert(false);
-  }
 }
 
 void AimdRateControl::ChangeState(RateControlState new_state) {
-  came_from_state_ = rate_control_state_;
   rate_control_state_ = new_state;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
index 56f1c1a866..93ae2190d6 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
@@ -11,38 +11,37 @@
 #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h"
 
 namespace webrtc {
 
-// A RemoteRateControl implementation based on additive increases of
+// A rate control implementation based on additive increases of
 // bitrate when no over-use is detected and multiplicative decreases when
 // over-uses are detected. When we think the available bandwidth has changes or
 // is unknown, we will switch to a "slow-start mode" where we increase
 // multiplicatively.
-class AimdRateControl : public RemoteRateControl {
+class AimdRateControl {
  public:
-  explicit AimdRateControl(uint32_t min_bitrate_bps);
+  AimdRateControl();
   virtual ~AimdRateControl() {}
 
-  // Implements RemoteRateControl.
-  bool ValidEstimate() const override;
-  RateControlType GetControlType() const override;
-  uint32_t GetMinBitrate() const override;
-  int64_t GetFeedbackInterval() const override;
+  // Returns true if there is a valid estimate of the incoming bitrate, false
+  // otherwise.
+  bool ValidEstimate() const;
+  void SetMinBitrate(int min_bitrate_bps);
+  int64_t GetFeedbackInterval() const;
   // Returns true if the bitrate estimate hasn't been changed for more than
   // an RTT, or if the incoming_bitrate is more than 5% above the current
   // estimate. Should be used to decide if we should reduce the rate further
   // when over-using.
   bool TimeToReduceFurther(int64_t time_now,
-                           uint32_t incoming_bitrate_bps) const override;
-  uint32_t LatestEstimate() const override;
-  uint32_t UpdateBandwidthEstimate(int64_t now_ms) override;
-  void SetRtt(int64_t rtt) override;
-  RateControlRegion Update(const RateControlInput* input,
-                           int64_t now_ms) override;
-  void SetEstimate(int bitrate_bps, int64_t now_ms) override;
+                           uint32_t incoming_bitrate_bps) const;
+  uint32_t LatestEstimate() const;
+  uint32_t UpdateBandwidthEstimate(int64_t now_ms);
+  void SetRtt(int64_t rtt);
+  void Update(const RateControlInput* input, int64_t now_ms);
+  void SetEstimate(int bitrate_bps, int64_t now_ms);
 
  private:
   // Update the target bitrate according based on, among other things,
@@ -72,7 +71,6 @@ class AimdRateControl : public RemoteRateControl {
   float avg_max_bitrate_kbps_;
   float var_max_bitrate_kbps_;
   RateControlState rate_control_state_;
-  RateControlState came_from_state_;
   RateControlRegion rate_control_region_;
   int64_t time_last_bitrate_change_;
   RateControlInput current_input_;
@@ -82,9 +80,8 @@ class AimdRateControl : public RemoteRateControl {
   float beta_;
   int64_t rtt_;
   int64_t time_of_last_log_;
-
-  DISALLOW_IMPLICIT_CONSTRUCTORS(AimdRateControl);
+  bool in_experiment_;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
index df82764a6c..11fd64f84e 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -16,7 +16,6 @@
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
-using std::string;
 
 namespace webrtc {
 namespace testing {
@@ -27,14 +26,20 @@ namespace bwe {
 class BweSimulation : public BweTest,
                       public ::testing::TestWithParam<BandwidthEstimatorType> {
  public:
-  BweSimulation() : BweTest() {}
+  BweSimulation()
+      : BweTest(), random_(Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
   virtual ~BweSimulation() {}
 
  protected:
-  void SetUp() override { BweTest::SetUp(); }
+  void SetUp() override {
+    BweTest::SetUp();
+    VerboseLogging(true);
+  }
+
+  Random random_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(BweSimulation);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweSimulation);
 };
 
 INSTANTIATE_TEST_CASE_P(VideoSendersTest,
@@ -44,189 +49,224 @@ INSTANTIATE_TEST_CASE_P(VideoSendersTest,
                                           kNadaEstimator));
 
 TEST_P(BweSimulation, SprintUplinkTest) {
-  VerboseLogging(true);
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
-  RateCounterFilter counter1(&uplink_, 0, "sender_output");
+  VideoSender sender(&uplink_, &source, GetParam());
+  RateCounterFilter counter1(&uplink_, 0, "sender_output",
+                             bwe_names[GetParam()]);
   TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");
-  RateCounterFilter counter2(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter2(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
   ASSERT_TRUE(filter.Init(test::ResourcePath("sprint-uplink", "rx")));
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, Verizon4gDownlinkTest) {
-  VerboseLogging(true);
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&downlink_, &source, GetParam());
-  RateCounterFilter counter1(&downlink_, 0, "sender_output");
+  VideoSender sender(&downlink_, &source, GetParam());
+  RateCounterFilter counter1(&downlink_, 0, "sender_output",
+                             bwe_names[GetParam()] + "_up");
   TraceBasedDeliveryFilter filter(&downlink_, 0, "link_capacity");
-  RateCounterFilter counter2(&downlink_, 0, "receiver_input");
+  RateCounterFilter counter2(&downlink_, 0, "Receiver",
+                             bwe_names[GetParam()] + "_down");
   PacketReceiver receiver(&downlink_, 0, GetParam(), true, true);
   ASSERT_TRUE(filter.Init(test::ResourcePath("verizon4g-downlink", "rx")));
   RunFor(22 * 60 * 1000);
 }
 
 TEST_P(BweSimulation, Choke1000kbps500kbps1000kbpsBiDirectional) {
-  VerboseLogging(true);
-
   const int kFlowIds[] = {0, 1};
   const size_t kNumFlows = sizeof(kFlowIds) / sizeof(kFlowIds[0]);
 
   AdaptiveVideoSource source(kFlowIds[0], 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, kFlowIds[0]);
-  RateCounterFilter counter(&uplink_, kFlowIds[0], "receiver_input_0");
+  RateCounterFilter counter(&uplink_, kFlowIds[0], "Receiver_0",
+                            bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, kFlowIds[0], GetParam(), true, false);
 
   AdaptiveVideoSource source2(kFlowIds[1], 30, 300, 0, 0);
-  PacketSender sender2(&downlink_, &source2, GetParam());
+  VideoSender sender2(&downlink_, &source2, GetParam());
   ChokeFilter choke2(&downlink_, kFlowIds[1]);
   DelayFilter delay(&downlink_, CreateFlowIds(kFlowIds, kNumFlows));
-  RateCounterFilter counter2(&downlink_, kFlowIds[1], "receiver_input_1");
+  RateCounterFilter counter2(&downlink_, kFlowIds[1], "Receiver_1",
+                             bwe_names[GetParam()]);
   PacketReceiver receiver2(&downlink_, kFlowIds[1], GetParam(), true, false);
 
-  choke2.SetCapacity(500);
-  delay.SetDelay(0);
+  choke2.set_capacity_kbps(500);
+  delay.SetOneWayDelayMs(0);
 
-  choke.SetCapacity(1000);
-  choke.SetMaxDelay(500);
+  choke.set_capacity_kbps(1000);
+  choke.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  choke.SetCapacity(500);
+  choke.set_capacity_kbps(500);
   RunFor(60 * 1000);
-  choke.SetCapacity(1000);
+  choke.set_capacity_kbps(1000);
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) {
-  VerboseLogging(true);
-
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, false);
 
-  choke.SetCapacity(1000);
-  choke.SetMaxDelay(500);
+  choke.set_capacity_kbps(1000);
+  choke.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  choke.SetCapacity(500);
+  choke.set_capacity_kbps(500);
   RunFor(60 * 1000);
-  choke.SetCapacity(1000);
+  choke.set_capacity_kbps(1000);
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, PacerChoke1000kbps500kbps1000kbps) {
-  VerboseLogging(true);
-  PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000);
+  AdaptiveVideoSource source(0, 30, 300, 0, 0);
   PacedVideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter filter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
-  filter.SetCapacity(1000);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(1000);
+  filter.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  filter.SetCapacity(500);
+  filter.set_capacity_kbps(500);
   RunFor(60 * 1000);
-  filter.SetCapacity(1000);
+  filter.set_capacity_kbps(1000);
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, PacerChoke10000kbps) {
-  VerboseLogging(true);
   PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000);
   PacedVideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter filter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
-  filter.SetCapacity(10000);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(10000);
+  filter.set_max_delay_ms(500);
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, PacerChoke200kbps30kbps200kbps) {
-  VerboseLogging(true);
-  PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000);
+  AdaptiveVideoSource source(0, 30, 300, 0, 0);
   PacedVideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter filter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
-  filter.SetCapacity(200);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(200);
+  filter.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  filter.SetCapacity(30);
+  filter.set_capacity_kbps(30);
   RunFor(60 * 1000);
-  filter.SetCapacity(200);
+  filter.set_capacity_kbps(200);
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, Choke200kbps30kbps200kbps) {
-  VerboseLogging(true);
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter filter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
-  filter.SetCapacity(200);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(200);
+  filter.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  filter.SetCapacity(30);
+  filter.set_capacity_kbps(30);
   RunFor(60 * 1000);
-  filter.SetCapacity(200);
+  filter.set_capacity_kbps(200);
   RunFor(60 * 1000);
 }
 
 TEST_P(BweSimulation, GoogleWifiTrace3Mbps) {
-  VerboseLogging(true);
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, kRembEstimator);
-  RateCounterFilter counter1(&uplink_, 0, "sender_output");
+  VideoSender sender(&uplink_, &source, GetParam());
+  RateCounterFilter counter1(&uplink_, 0, "sender_output",
+                             bwe_names[GetParam()]);
   TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");
-  filter.SetMaxDelay(500);
-  RateCounterFilter counter2(&uplink_, 0, "receiver_input");
+  filter.set_max_delay_ms(500);
+  RateCounterFilter counter2(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
   ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx")));
   RunFor(300 * 1000);
 }
 
+TEST_P(BweSimulation, LinearIncreasingCapacity) {
+  PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000000);
+  PacedVideoSender sender(&uplink_, &source, GetParam());
+  ChokeFilter filter(&uplink_, 0);
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
+  PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
+  filter.set_max_delay_ms(500);
+  const int kStartingCapacityKbps = 150;
+  const int kEndingCapacityKbps = 1500;
+  const int kStepKbps = 5;
+  const int kStepTimeMs = 1000;
+
+  for (int i = kStartingCapacityKbps; i <= kEndingCapacityKbps;
+       i += kStepKbps) {
+    filter.set_capacity_kbps(i);
+    RunFor(kStepTimeMs);
+  }
+}
+
+TEST_P(BweSimulation, LinearDecreasingCapacity) {
+  PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000000);
+  PacedVideoSender sender(&uplink_, &source, GetParam());
+  ChokeFilter filter(&uplink_, 0);
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
+  PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
+  filter.set_max_delay_ms(500);
+  const int kStartingCapacityKbps = 1500;
+  const int kEndingCapacityKbps = 150;
+  const int kStepKbps = -5;
+  const int kStepTimeMs = 1000;
+
+  for (int i = kStartingCapacityKbps; i >= kEndingCapacityKbps;
+       i += kStepKbps) {
+    filter.set_capacity_kbps(i);
+    RunFor(kStepTimeMs);
+  }
+}
+
 TEST_P(BweSimulation, PacerGoogleWifiTrace3Mbps) {
-  VerboseLogging(true);
   PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000);
-  PacedVideoSender sender(&uplink_, &source, kRembEstimator);
-  RateCounterFilter counter1(&uplink_, 0, "sender_output");
+  PacedVideoSender sender(&uplink_, &source, GetParam());
+  RateCounterFilter counter1(&uplink_, 0, "sender_output",
+                             bwe_names[GetParam()]);
   TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");
-  filter.SetMaxDelay(500);
-  RateCounterFilter counter2(&uplink_, 0, "receiver_input");
+  filter.set_max_delay_ms(500);
+  RateCounterFilter counter2(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
   ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx")));
   RunFor(300 * 1000);
 }
 
 TEST_P(BweSimulation, SelfFairnessTest) {
-  VerboseLogging(true);
-  const int kAllFlowIds[] = {0, 1, 2};
+  Random prng(Clock::GetRealTimeClock()->TimeInMicroseconds());
+  const int kAllFlowIds[] = {0, 1, 2, 3};
   const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]);
-  rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumFlows];
-  rtc::scoped_ptr<PacketSender> senders[kNumFlows];
+  rtc::scoped_ptr<VideoSource> sources[kNumFlows];
+  rtc::scoped_ptr<VideoSender> senders[kNumFlows];
   for (size_t i = 0; i < kNumFlows; ++i) {
     // Streams started 20 seconds apart to give them different advantage when
     // competing for the bandwidth.
-    sources[i].reset(
-        new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0, i * 20000));
-    senders[i].reset(new PacketSender(&uplink_, sources[i].get(), GetParam()));
+    sources[i].reset(new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0,
+                                             i * prng.Rand(39999)));
+    senders[i].reset(new VideoSender(&uplink_, sources[i].get(), GetParam()));
   }
 
   ChokeFilter choke(&uplink_, CreateFlowIds(kAllFlowIds, kNumFlows));
-  choke.SetCapacity(1000);
+  choke.set_capacity_kbps(1000);
 
   rtc::scoped_ptr<RateCounterFilter> rate_counters[kNumFlows];
   for (size_t i = 0; i < kNumFlows; ++i) {
-    rate_counters[i].reset(new RateCounterFilter(
-        &uplink_, CreateFlowIds(&kAllFlowIds[i], 1), "receiver_input"));
+    rate_counters[i].reset(
+        new RateCounterFilter(&uplink_, CreateFlowIds(&kAllFlowIds[i], 1),
+                              "Receiver", bwe_names[GetParam()]));
   }
 
   RateCounterFilter total_utilization(
-      &uplink_, CreateFlowIds(kAllFlowIds, kNumFlows), "total_utilization");
+      &uplink_, CreateFlowIds(kAllFlowIds, kNumFlows), "total_utilization",
+      "Total_link_utilization");
 
   rtc::scoped_ptr<PacketReceiver> receivers[kNumFlows];
   for (size_t i = 0; i < kNumFlows; ++i) {
@@ -237,43 +277,174 @@ TEST_P(BweSimulation, SelfFairnessTest) {
   RunFor(30 * 60 * 1000);
 }
 
-TEST_P(BweSimulation, PacedSelfFairnessTest) {
-  VerboseLogging(true);
-  const int kAllFlowIds[] = {0, 1, 2};
-  const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]);
-  rtc::scoped_ptr<PeriodicKeyFrameSource> sources[kNumFlows];
-  rtc::scoped_ptr<PacedVideoSender> senders[kNumFlows];
-
-  for (size_t i = 0; i < kNumFlows; ++i) {
-    // Streams started 20 seconds apart to give them different advantage when
-    // competing for the bandwidth.
-    sources[i].reset(new PeriodicKeyFrameSource(kAllFlowIds[i], 30, 300, 0,
-                                                i * 20000, 1000));
-    senders[i].reset(
-        new PacedVideoSender(&uplink_, sources[i].get(), GetParam()));
+TEST_P(BweSimulation, PacedSelfFairness50msTest) {
+  const int64_t kAverageOffsetMs = 20 * 1000;
+  const int kNumRmcatFlows = 4;
+  int64_t offsets_ms[kNumRmcatFlows];
+  offsets_ms[0] = random_.Rand(2 * kAverageOffsetMs);
+  for (int i = 1; i < kNumRmcatFlows; ++i) {
+    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(2 * kAverageOffsetMs);
   }
-
-  ChokeFilter choke(&uplink_, CreateFlowIds(kAllFlowIds, kNumFlows));
-  choke.SetCapacity(1000);
-
-  rtc::scoped_ptr<RateCounterFilter> rate_counters[kNumFlows];
-  for (size_t i = 0; i < kNumFlows; ++i) {
-    rate_counters[i].reset(new RateCounterFilter(
-        &uplink_, CreateFlowIds(&kAllFlowIds[i], 1), "receiver_input"));
-  }
-
-  RateCounterFilter total_utilization(
-      &uplink_, CreateFlowIds(kAllFlowIds, kNumFlows), "total_utilization");
-
-  rtc::scoped_ptr<PacketReceiver> receivers[kNumFlows];
-  for (size_t i = 0; i < kNumFlows; ++i) {
-    receivers[i].reset(new PacketReceiver(&uplink_, kAllFlowIds[i], GetParam(),
-                                          i == 0, false));
-  }
-
-  RunFor(30 * 60 * 1000);
+  RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 1000, 3000, 50, 50, 0,
+                  offsets_ms);
 }
+
+TEST_P(BweSimulation, PacedSelfFairness500msTest) {
+  const int64_t kAverageOffsetMs = 20 * 1000;
+  const int kNumRmcatFlows = 4;
+  int64_t offsets_ms[kNumRmcatFlows];
+  offsets_ms[0] = random_.Rand(2 * kAverageOffsetMs);
+  for (int i = 1; i < kNumRmcatFlows; ++i) {
+    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(2 * kAverageOffsetMs);
+  }
+  RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 1000, 3000, 500, 50, 0,
+                  offsets_ms);
+}
+
+TEST_P(BweSimulation, PacedSelfFairness1000msTest) {
+  const int64_t kAverageOffsetMs = 20 * 1000;
+  const int kNumRmcatFlows = 4;
+  int64_t offsets_ms[kNumRmcatFlows];
+  offsets_ms[0] = random_.Rand(2 * kAverageOffsetMs);
+  for (int i = 1; i < kNumRmcatFlows; ++i) {
+    offsets_ms[i] = offsets_ms[i - 1] + random_.Rand(2 * kAverageOffsetMs);
+  }
+  RunFairnessTest(GetParam(), 4, 0, 1000, 3000, 1000, 50, 0, offsets_ms);
+}
+
+TEST_P(BweSimulation, TcpFairness50msTest) {
+  const int64_t kAverageOffsetMs = 20 * 1000;
+  int64_t offset_ms[] = {random_.Rand(2 * kAverageOffsetMs), 0};
+  RunFairnessTest(GetParam(), 1, 1, 1000, 2000, 50, 50, 0, offset_ms);
+}
+
+TEST_P(BweSimulation, TcpFairness500msTest) {
+  const int64_t kAverageOffsetMs = 20 * 1000;
+  int64_t offset_ms[] = {random_.Rand(2 * kAverageOffsetMs), 0};
+  RunFairnessTest(GetParam(), 1, 1, 1000, 2000, 500, 50, 0, offset_ms);
+}
+
+TEST_P(BweSimulation, TcpFairness1000msTest) {
+  const int kAverageOffsetMs = 20 * 1000;
+  int64_t offset_ms[] = {random_.Rand(2 * kAverageOffsetMs), 0};
+  RunFairnessTest(GetParam(), 1, 1, 1000, 2000, 1000, 50, 0, offset_ms);
+}
+
+// The following test cases begin with "Evaluation" as a referrence to the
+// Internet draft https://tools.ietf.org/html/draft-ietf-rmcat-eval-test-01.
+
+TEST_P(BweSimulation, Evaluation1) {
+  RunVariableCapacity1SingleFlow(GetParam());
+}
+
+TEST_P(BweSimulation, Evaluation2) {
+  const size_t kNumFlows = 2;
+  RunVariableCapacity2MultipleFlows(GetParam(), kNumFlows);
+}
+
+TEST_P(BweSimulation, Evaluation3) {
+  RunBidirectionalFlow(GetParam());
+}
+
+TEST_P(BweSimulation, Evaluation4) {
+  RunSelfFairness(GetParam());
+}
+
+TEST_P(BweSimulation, Evaluation5) {
+  RunRoundTripTimeFairness(GetParam());
+}
+
+TEST_P(BweSimulation, Evaluation6) {
+  RunLongTcpFairness(GetParam());
+}
+
+// Different calls to the Evaluation7 will create the same FileSizes
+// and StartingTimes as long as the seeds remain unchanged. This is essential
+// when calling it with multiple estimators for comparison purposes.
+TEST_P(BweSimulation, Evaluation7) {
+  const int kNumTcpFiles = 10;
+  RunMultipleShortTcpFairness(GetParam(),
+                              BweTest::GetFileSizesBytes(kNumTcpFiles),
+                              BweTest::GetStartingTimesMs(kNumTcpFiles));
+}
+
+TEST_P(BweSimulation, Evaluation8) {
+  RunPauseResumeFlows(GetParam());
+}
+
+// Following test cases begin with "GccComparison" run the
+// evaluation test cases for both GCC and other calling RMCAT.
+
+TEST_P(BweSimulation, GccComparison1) {
+  RunVariableCapacity1SingleFlow(GetParam());
+  BweTest gcc_test(false);
+  gcc_test.RunVariableCapacity1SingleFlow(kFullSendSideEstimator);
+}
+
+TEST_P(BweSimulation, GccComparison2) {
+  const size_t kNumFlows = 2;
+  RunVariableCapacity2MultipleFlows(GetParam(), kNumFlows);
+  BweTest gcc_test(false);
+  gcc_test.RunVariableCapacity2MultipleFlows(kFullSendSideEstimator, kNumFlows);
+}
+
+TEST_P(BweSimulation, GccComparison3) {
+  RunBidirectionalFlow(GetParam());
+  BweTest gcc_test(false);
+  gcc_test.RunBidirectionalFlow(kFullSendSideEstimator);
+}
+
+TEST_P(BweSimulation, GccComparison4) {
+  RunSelfFairness(GetParam());
+  BweTest gcc_test(false);
+  gcc_test.RunSelfFairness(GetParam());
+}
+
+TEST_P(BweSimulation, GccComparison5) {
+  RunRoundTripTimeFairness(GetParam());
+  BweTest gcc_test(false);
+  gcc_test.RunRoundTripTimeFairness(kFullSendSideEstimator);
+}
+
+TEST_P(BweSimulation, GccComparison6) {
+  RunLongTcpFairness(GetParam());
+  BweTest gcc_test(false);
+  gcc_test.RunLongTcpFairness(kFullSendSideEstimator);
+}
+
+TEST_P(BweSimulation, GccComparison7) {
+  const int kNumTcpFiles = 10;
+
+  std::vector<int> tcp_file_sizes_bytes =
+      BweTest::GetFileSizesBytes(kNumTcpFiles);
+  std::vector<int64_t> tcp_starting_times_ms =
+      BweTest::GetStartingTimesMs(kNumTcpFiles);
+
+  RunMultipleShortTcpFairness(GetParam(), tcp_file_sizes_bytes,
+                              tcp_starting_times_ms);
+
+  BweTest gcc_test(false);
+  gcc_test.RunMultipleShortTcpFairness(
+      kFullSendSideEstimator, tcp_file_sizes_bytes, tcp_starting_times_ms);
+}
+
+TEST_P(BweSimulation, GccComparison8) {
+  RunPauseResumeFlows(GetParam());
+  BweTest gcc_test(false);
+  gcc_test.RunPauseResumeFlows(kFullSendSideEstimator);
+}
+
+TEST_P(BweSimulation, GccComparisonChoke) {
+  int array[] = {1000, 500, 1000};
+  std::vector<int> capacities_kbps(array, array + 3);
+  RunChoke(GetParam(), capacities_kbps);
+
+  BweTest gcc_test(false);
+  gcc_test.RunChoke(kFullSendSideEstimator, capacities_kbps);
+}
+
 #endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
+
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
index f915a0fa52..3fb7e29e5b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
@@ -8,50 +8,40 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 
 #include "webrtc/typedefs.h"
 
-#define BWE_MAX(a,b) ((a)>(b)?(a):(b))
-#define BWE_MIN(a,b) ((a)<(b)?(a):(b))
+#define BWE_MAX(a, b) ((a) > (b) ? (a) : (b))
+#define BWE_MIN(a, b) ((a) < (b) ? (a) : (b))
 
 namespace webrtc {
-enum BandwidthUsage
-{
-    kBwNormal = 0,
-    kBwUnderusing = 1,
-    kBwOverusing = 2,
+
+static const int64_t kBitrateWindowMs = 1000;
+
+enum BandwidthUsage {
+  kBwNormal = 0,
+  kBwUnderusing = 1,
+  kBwOverusing = 2,
 };
 
-enum RateControlState
-{
-    kRcHold,
-    kRcIncrease,
-    kRcDecrease
-};
+enum RateControlState { kRcHold, kRcIncrease, kRcDecrease };
 
-enum RateControlRegion
-{
-    kRcNearMax,
-    kRcAboveMax,
-    kRcMaxUnknown
-};
+enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown };
 
-class RateControlInput
-{
-public:
-    RateControlInput(BandwidthUsage bwState,
-                     uint32_t incomingBitRate,
-                     double noiseVar)
-        : _bwState(bwState),
-          _incomingBitRate(incomingBitRate),
-          _noiseVar(noiseVar) {}
+struct RateControlInput {
+  RateControlInput(BandwidthUsage bw_state,
+                   uint32_t incoming_bitrate,
+                   double noise_var)
+      : bw_state(bw_state),
+        incoming_bitrate(incoming_bitrate),
+        noise_var(noise_var) {}
 
-    BandwidthUsage  _bwState;
-    uint32_t      _incomingBitRate;
-    double              _noiseVar;
+  BandwidthUsage bw_state;
+  uint32_t incoming_bitrate;
+  double noise_var;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h
new file mode 100644
index 0000000000..91a8ac8707
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+
+namespace webrtc {
+
+class MockRemoteBitrateEstimator : public RemoteBitrateEstimator {
+ public:
+  MOCK_METHOD1(IncomingPacketFeedbackVector,
+               void(const std::vector<PacketInfo>&));
+  MOCK_METHOD4(IncomingPacket, void(int64_t, size_t, const RTPHeader&, bool));
+  MOCK_METHOD1(RemoveStream, void(unsigned int));
+  MOCK_CONST_METHOD2(LatestEstimate,
+                     bool(std::vector<unsigned int>*, unsigned int*));
+  MOCK_CONST_METHOD1(GetStats, bool(ReceiveBandwidthEstimatorStats*));
+
+  // From CallStatsObserver;
+  MOCK_METHOD2(OnRttUpdate, void(int64_t, int64_t));
+
+  // From Module.
+  MOCK_METHOD0(TimeUntilNextProcess, int64_t());
+  MOCK_METHOD0(Process, int32_t());
+  MOCK_METHOD1(SetMinBitrate, void(int));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
index edfac977a2..ae05912b5f 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
 
 #include <vector>
 
@@ -26,4 +26,4 @@ class MockRemoteBitrateObserver : public RemoteBitrateObserver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
index 4986467e8d..0734cbf255 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -17,19 +17,15 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Clock;
 
-enum RateControlType {
-  kMimdControl,
-  kAimdControl
-};
-
 // RemoteBitrateObserver is used to signal changes in bitrate estimates for
 // the incoming streams.
 class RemoteBitrateObserver {
@@ -63,30 +59,9 @@ struct ReceiveBandwidthEstimatorStats {
   std::vector<int64_t> recent_arrival_time_ms;
 };
 
-struct PacketInfo {
-  PacketInfo(int64_t arrival_time_ms,
-             int64_t send_time_ms,
-             uint16_t sequence_number,
-             size_t payload_size)
-      : arrival_time_ms(arrival_time_ms),
-        send_time_ms(send_time_ms),
-        sequence_number(sequence_number),
-        payload_size(payload_size) {}
-  // Time corresponding to when the packet was received. Timestamped with the
-  // receiver's clock.
-  int64_t arrival_time_ms;
-  // Time corresponding to when the packet was sent, timestamped with the
-  // sender's clock.
-  int64_t send_time_ms;
-  // Packet identifier, incremented with 1 for every packet generated by the
-  // sender.
-  uint16_t sequence_number;
-  // Size of the packet excluding RTP headers.
-  size_t payload_size;
-};
-
 class RemoteBitrateEstimator : public CallStatsObserver, public Module {
  public:
+  static const int kDefaultMinBitrateBps = 30000;
   virtual ~RemoteBitrateEstimator() {}
 
   virtual void IncomingPacketFeedbackVector(
@@ -101,7 +76,8 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module {
   // Note that |arrival_time_ms| can be of an arbitrary time base.
   virtual void IncomingPacket(int64_t arrival_time_ms,
                               size_t payload_size,
-                              const RTPHeader& header) = 0;
+                              const RTPHeader& header,
+                              bool was_paced) = 0;
 
   // Removes all data for |ssrc|.
   virtual void RemoveStream(unsigned int ssrc) = 0;
@@ -115,33 +91,13 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module {
   // Returns true if the statistics are available.
   virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const = 0;
 
+  virtual void SetMinBitrate(int min_bitrate_bps) = 0;
+
  protected:
-  static const int64_t kProcessIntervalMs = 1000;
+  static const int64_t kProcessIntervalMs = 500;
   static const int64_t kStreamTimeOutMs = 2000;
 };
 
-struct RemoteBitrateEstimatorFactory {
-  RemoteBitrateEstimatorFactory() {}
-  virtual ~RemoteBitrateEstimatorFactory() {}
-
-  virtual RemoteBitrateEstimator* Create(
-      RemoteBitrateObserver* observer,
-      Clock* clock,
-      RateControlType control_type,
-      uint32_t min_bitrate_bps) const;
-};
-
-struct AbsoluteSendTimeRemoteBitrateEstimatorFactory
-    : public RemoteBitrateEstimatorFactory {
-  AbsoluteSendTimeRemoteBitrateEstimatorFactory() {}
-  virtual ~AbsoluteSendTimeRemoteBitrateEstimatorFactory() {}
-
-  virtual RemoteBitrateEstimator* Create(
-      RemoteBitrateObserver* observer,
-      Clock* clock,
-      RateControlType control_type,
-      uint32_t min_bitrate_bps) const;
-};
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
new file mode 100644
index 0000000000..a643c1f103
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
+
+#include <map>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+
+namespace webrtc {
+
+class SendTimeHistory {
+ public:
+  SendTimeHistory(Clock* clock, int64_t packet_age_limit);
+  virtual ~SendTimeHistory();
+
+  void AddAndRemoveOld(uint16_t sequence_number, size_t length, bool was_paced);
+  bool OnSentPacket(uint16_t sequence_number, int64_t timestamp);
+  // Look up PacketInfo for a sent packet, based on the sequence number, and
+  // populate all fields except for receive_time. The packet parameter must
+  // thus be non-null and have the sequence_number field set.
+  bool GetInfo(PacketInfo* packet, bool remove);
+  void Clear();
+
+ private:
+  void EraseOld();
+  void UpdateOldestSequenceNumber();
+
+  Clock* const clock_;
+  const int64_t packet_age_limit_;
+  uint16_t oldest_sequence_number_;  // Oldest may not be lowest.
+  std::map<uint16_t, PacketInfo> history_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(SendTimeHistory);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
index a9a7ae7d07..f75bc2b03e 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
@@ -13,7 +13,8 @@
 #include <algorithm>
 #include <cassert>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -52,6 +53,14 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
                          prev_timestamp_group_.timestamp;
       *arrival_time_delta_ms = current_timestamp_group_.complete_time_ms -
                                prev_timestamp_group_.complete_time_ms;
+      if (*arrival_time_delta_ms < 0) {
+        // The group of packets has been reordered since receiving its local
+        // arrival timestamp.
+        LOG(LS_WARNING) << "Packets are being reordered on the path from the "
+                           "socket to the bandwidth estimator. Ignoring this "
+                           "packet for bandwidth estimation.";
+        return false;
+      }
       assert(*arrival_time_delta_ms >= 0);
       *packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
           static_cast<int>(prev_timestamp_group_.size);
@@ -62,8 +71,7 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
     current_timestamp_group_.first_timestamp = timestamp;
     current_timestamp_group_.timestamp = timestamp;
     current_timestamp_group_.size = 0;
-  }
-  else {
+  } else {
     current_timestamp_group_.timestamp = LatestTimestamp(
         current_timestamp_group_.timestamp, timestamp);
   }
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.h
index ace855118e..427bafcf96 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/inter_arrival.h
@@ -78,7 +78,7 @@ class InterArrival {
   double timestamp_to_ms_coeff_;
   bool burst_grouping_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(InterArrival);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(InterArrival);
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.cc
deleted file mode 100644
index ab8f4db826..0000000000
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.cc
+++ /dev/null
@@ -1,328 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/remote_bitrate_estimator/mimd_rate_control.h"
-
-#include <algorithm>
-#include <cassert>
-#include <cmath>
-#include <cstring>
-
-namespace webrtc {
-
-const int64_t kDefaultRttMs = 200;
-const int64_t kLogIntervalMs = 1000;
-
-MimdRateControl::MimdRateControl(uint32_t min_bitrate_bps)
-    : min_configured_bit_rate_(min_bitrate_bps),
-    max_configured_bit_rate_(30000000),
-    current_bit_rate_(max_configured_bit_rate_),
-    max_hold_rate_(0),
-    avg_max_bit_rate_(-1.0f),
-    var_max_bit_rate_(0.4f),
-    rate_control_state_(kRcHold),
-    came_from_state_(kRcDecrease),
-    rate_control_region_(kRcMaxUnknown),
-    last_bit_rate_change_(-1),
-    current_input_(kBwNormal, 0, 1.0),
-    updated_(false),
-    time_first_incoming_estimate_(-1),
-    initialized_bit_rate_(false),
-    avg_change_period_(1000.0f),
-    last_change_ms_(-1),
-    beta_(0.9f),
-    rtt_(kDefaultRttMs),
-    time_of_last_log_(-1)
-{
-}
-
-RateControlType MimdRateControl::GetControlType() const {
-  return kMimdControl;
-}
-
-uint32_t MimdRateControl::GetMinBitrate() const {
-  return min_configured_bit_rate_;
-}
-
-bool MimdRateControl::ValidEstimate() const {
-  return initialized_bit_rate_;
-}
-
-int64_t MimdRateControl::GetFeedbackInterval() const {
-  return kMaxFeedbackIntervalMs;
-}
-
-bool MimdRateControl::TimeToReduceFurther(int64_t time_now,
-                                          uint32_t incoming_bitrate_bps) const {
-  const int64_t bitrate_reduction_interval =
-      std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
-  if (time_now - last_bit_rate_change_ >= bitrate_reduction_interval) {
-    return true;
-  }
-  if (ValidEstimate()) {
-    const int threshold = static_cast<int>(1.05 * incoming_bitrate_bps);
-    const int bitrate_difference = LatestEstimate() - incoming_bitrate_bps;
-    return bitrate_difference > threshold;
-  }
-  return false;
-}
-
-uint32_t MimdRateControl::LatestEstimate() const {
-  return current_bit_rate_;
-}
-
-uint32_t MimdRateControl::UpdateBandwidthEstimate(int64_t now_ms) {
-  current_bit_rate_ = ChangeBitRate(current_bit_rate_,
-                                    current_input_._incomingBitRate,
-                                    current_input_._noiseVar,
-                                    now_ms);
-  if (now_ms - time_of_last_log_ > kLogIntervalMs) {
-    time_of_last_log_ = now_ms;
-  }
-  return current_bit_rate_;
-}
-
-void MimdRateControl::SetRtt(int64_t rtt) {
-  rtt_ = rtt;
-}
-
-RateControlRegion MimdRateControl::Update(const RateControlInput* input,
-                                          int64_t now_ms) {
-  assert(input);
-
-  // Set the initial bit rate value to what we're receiving the first half
-  // second.
-  if (!initialized_bit_rate_) {
-    if (time_first_incoming_estimate_ < 0) {
-      if (input->_incomingBitRate > 0) {
-        time_first_incoming_estimate_ = now_ms;
-      }
-    } else if (now_ms - time_first_incoming_estimate_ > 500 &&
-               input->_incomingBitRate > 0) {
-      current_bit_rate_ = input->_incomingBitRate;
-      initialized_bit_rate_ = true;
-    }
-  }
-
-  if (updated_ && current_input_._bwState == kBwOverusing) {
-    // Only update delay factor and incoming bit rate. We always want to react
-    // on an over-use.
-    current_input_._noiseVar = input->_noiseVar;
-    current_input_._incomingBitRate = input->_incomingBitRate;
-    return rate_control_region_;
-  }
-  updated_ = true;
-  current_input_ = *input;
-  return rate_control_region_;
-}
-
-void MimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
-}
-
-uint32_t MimdRateControl::ChangeBitRate(uint32_t current_bit_rate,
-                                        uint32_t incoming_bit_rate,
-                                        double noise_var,
-                                        int64_t now_ms) {
-  if (!updated_) {
-    return current_bit_rate_;
-  }
-  updated_ = false;
-  UpdateChangePeriod(now_ms);
-  ChangeState(current_input_, now_ms);
-  // calculated here because it's used in multiple places
-  const float incoming_bit_rate_kbps = incoming_bit_rate / 1000.0f;
-  // Calculate the max bit rate std dev given the normalized
-  // variance and the current incoming bit rate.
-  const float std_max_bit_rate = sqrt(var_max_bit_rate_ * avg_max_bit_rate_);
-  bool recovery = false;
-  switch (rate_control_state_) {
-    case kRcHold: {
-      max_hold_rate_ = std::max(max_hold_rate_, incoming_bit_rate);
-      break;
-    }
-    case kRcIncrease: {
-      if (avg_max_bit_rate_ >= 0) {
-        if (incoming_bit_rate_kbps > avg_max_bit_rate_ + 3 * std_max_bit_rate) {
-          ChangeRegion(kRcMaxUnknown);
-          avg_max_bit_rate_ = -1.0;
-        } else if (incoming_bit_rate_kbps > avg_max_bit_rate_ + 2.5 *
-                   std_max_bit_rate) {
-          ChangeRegion(kRcAboveMax);
-        }
-      }
-      const int64_t response_time =
-          static_cast<int64_t>(avg_change_period_ + 0.5f) + rtt_ + 300;
-      double alpha = RateIncreaseFactor(now_ms, last_bit_rate_change_,
-                                        response_time, noise_var);
-
-      current_bit_rate = static_cast<uint32_t>(current_bit_rate * alpha) + 1000;
-      if (max_hold_rate_ > 0 && beta_ * max_hold_rate_ > current_bit_rate) {
-        current_bit_rate = static_cast<uint32_t>(beta_ * max_hold_rate_);
-        avg_max_bit_rate_ = beta_ * max_hold_rate_ / 1000.0f;
-        ChangeRegion(kRcNearMax);
-        recovery = true;
-      }
-      max_hold_rate_ = 0;
-      last_bit_rate_change_ = now_ms;
-      break;
-    }
-    case kRcDecrease: {
-      if (incoming_bit_rate < min_configured_bit_rate_) {
-        current_bit_rate = min_configured_bit_rate_;
-      } else {
-        // Set bit rate to something slightly lower than max
-        // to get rid of any self-induced delay.
-        current_bit_rate = static_cast<uint32_t>(beta_ * incoming_bit_rate +
-            0.5);
-        if (current_bit_rate > current_bit_rate_) {
-          // Avoid increasing the rate when over-using.
-          if (rate_control_region_ != kRcMaxUnknown) {
-            current_bit_rate = static_cast<uint32_t>(beta_ * avg_max_bit_rate_ *
-                1000 + 0.5f);
-          }
-          current_bit_rate = std::min(current_bit_rate, current_bit_rate_);
-        }
-        ChangeRegion(kRcNearMax);
-
-        if (incoming_bit_rate_kbps < avg_max_bit_rate_ - 3 * std_max_bit_rate) {
-          avg_max_bit_rate_ = -1.0f;
-        }
-
-        UpdateMaxBitRateEstimate(incoming_bit_rate_kbps);
-      }
-      // Stay on hold until the pipes are cleared.
-      ChangeState(kRcHold);
-      last_bit_rate_change_ = now_ms;
-      break;
-    }
-    default:
-      assert(false);
-  }
-  if (!recovery && (incoming_bit_rate > 100000 || current_bit_rate > 150000) &&
-      current_bit_rate > 1.5 * incoming_bit_rate) {
-    // Allow changing the bit rate if we are operating at very low rates
-    // Don't change the bit rate if the send side is too far off
-    current_bit_rate = current_bit_rate_;
-    last_bit_rate_change_ = now_ms;
-  }
-  return current_bit_rate;
-}
-
-double MimdRateControl::RateIncreaseFactor(int64_t now_ms,
-                                           int64_t last_ms,
-                                           int64_t reaction_time_ms,
-                                           double noise_var) const {
-  // alpha = 1.02 + B ./ (1 + exp(b*(tr - (c1*s2 + c2))))
-  // Parameters
-  const double B = 0.0407;
-  const double b = 0.0025;
-  const double c1 = -6700.0 / (33 * 33);
-  const double c2 = 800.0;
-  const double d = 0.85;
-
-  double alpha = 1.005 + B / (1 + exp( b * (d * reaction_time_ms -
-      (c1 * noise_var + c2))));
-
-  if (alpha < 1.005) {
-    alpha = 1.005;
-  } else if (alpha > 1.3) {
-    alpha = 1.3;
-  }
-
-  if (last_ms > -1) {
-    alpha = pow(alpha, (now_ms - last_ms) / 1000.0);
-  }
-
-  if (rate_control_region_ == kRcNearMax) {
-    // We're close to our previous maximum. Try to stabilize the
-    // bit rate in this region, by increasing in smaller steps.
-    alpha = alpha - (alpha - 1.0) / 2.0;
-  } else if (rate_control_region_ == kRcMaxUnknown) {
-    alpha = alpha + (alpha - 1.0) * 2.0;
-  }
-
-  return alpha;
-}
-
-void MimdRateControl::UpdateChangePeriod(int64_t now_ms) {
-  int64_t change_period = 0;
-  if (last_change_ms_ > -1) {
-    change_period = now_ms - last_change_ms_;
-  }
-  last_change_ms_ = now_ms;
-  avg_change_period_ = 0.9f * avg_change_period_ + 0.1f * change_period;
-}
-
-void MimdRateControl::UpdateMaxBitRateEstimate(float incoming_bit_rate_kbps) {
-  const float alpha = 0.05f;
-  if (avg_max_bit_rate_ == -1.0f) {
-    avg_max_bit_rate_ = incoming_bit_rate_kbps;
-  } else {
-    avg_max_bit_rate_ = (1 - alpha) * avg_max_bit_rate_ +
-        alpha * incoming_bit_rate_kbps;
-  }
-  // Estimate the max bit rate variance and normalize the variance
-  // with the average max bit rate.
-  const float norm = std::max(avg_max_bit_rate_, 1.0f);
-  var_max_bit_rate_ = (1 - alpha) * var_max_bit_rate_ +
-      alpha * (avg_max_bit_rate_ - incoming_bit_rate_kbps) *
-          (avg_max_bit_rate_ - incoming_bit_rate_kbps) / norm;
-  // 0.4 ~= 14 kbit/s at 500 kbit/s
-  if (var_max_bit_rate_ < 0.4f) {
-    var_max_bit_rate_ = 0.4f;
-  }
-  // 2.5f ~= 35 kbit/s at 500 kbit/s
-  if (var_max_bit_rate_ > 2.5f) {
-    var_max_bit_rate_ = 2.5f;
-  }
-}
-
-void MimdRateControl::ChangeState(const RateControlInput& input,
-                                    int64_t now_ms) {
-  switch (current_input_._bwState) {
-    case kBwNormal:
-      if (rate_control_state_ == kRcHold) {
-        last_bit_rate_change_ = now_ms;
-        ChangeState(kRcIncrease);
-      }
-      break;
-    case kBwOverusing:
-      if (rate_control_state_ != kRcDecrease) {
-        ChangeState(kRcDecrease);
-      }
-      break;
-    case kBwUnderusing:
-      ChangeState(kRcHold);
-      break;
-    default:
-      assert(false);
-  }
-}
-
-void MimdRateControl::ChangeRegion(RateControlRegion region) {
-  rate_control_region_ = region;
-  switch (rate_control_region_) {
-    case kRcAboveMax:
-    case kRcMaxUnknown:
-      beta_ = 0.9f;
-      break;
-    case kRcNearMax:
-      beta_ = 0.95f;
-      break;
-    default:
-      assert(false);
-  }
-}
-
-void MimdRateControl::ChangeState(RateControlState new_state) {
-  came_from_state_ = rate_control_state_;
-  rate_control_state_ = new_state;
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.h
deleted file mode 100644
index d15799344e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_MIMD_RATE_CONTROL_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_MIMD_RATE_CONTROL_H_
-
-#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h"
-
-namespace webrtc {
-
-// A RemoteRateControl implementation based on multiplicative increases of
-// bitrate when no over-use is detected and multiplicative decreases when
-// over-uses are detected.
-class MimdRateControl : public RemoteRateControl {
- public:
-  explicit MimdRateControl(uint32_t min_bitrate_bps);
-  virtual ~MimdRateControl() {}
-
-  // Implements RemoteRateControl.
-  RateControlType GetControlType() const override;
-  uint32_t GetMinBitrate() const override;
-  bool ValidEstimate() const override;
-  int64_t GetFeedbackInterval() const override;
-  bool TimeToReduceFurther(int64_t time_now,
-                           uint32_t incoming_bitrate_bps) const override;
-  uint32_t LatestEstimate() const override;
-  uint32_t UpdateBandwidthEstimate(int64_t now_ms) override;
-  void SetRtt(int64_t rtt) override;
-  RateControlRegion Update(const RateControlInput* input,
-                           int64_t now_ms) override;
-  void SetEstimate(int bitrate_bps, int64_t now_ms) override;
-
- private:
-  uint32_t ChangeBitRate(uint32_t current_bit_rate,
-                         uint32_t incoming_bit_rate,
-                         double delay_factor,
-                         int64_t now_ms);
-  double RateIncreaseFactor(int64_t now_ms,
-                            int64_t last_ms,
-                            int64_t reaction_time_ms,
-                            double noise_var) const;
-  void UpdateChangePeriod(int64_t now_ms);
-  void UpdateMaxBitRateEstimate(float incoming_bit_rate_kbps);
-  void ChangeState(const RateControlInput& input, int64_t now_ms);
-  void ChangeState(RateControlState new_state);
-  void ChangeRegion(RateControlRegion region);
-
-  uint32_t min_configured_bit_rate_;
-  uint32_t max_configured_bit_rate_;
-  uint32_t current_bit_rate_;
-  uint32_t max_hold_rate_;
-  float avg_max_bit_rate_;
-  float var_max_bit_rate_;
-  RateControlState rate_control_state_;
-  RateControlState came_from_state_;
-  RateControlRegion rate_control_region_;
-  int64_t last_bit_rate_change_;
-  RateControlInput current_input_;
-  bool updated_;
-  int64_t time_first_incoming_estimate_;
-  bool initialized_bit_rate_;
-  float avg_change_period_;
-  int64_t last_change_ms_;
-  float beta_;
-  int64_t rtt_;
-  int64_t time_of_last_log_;
-
-  DISALLOW_IMPLICIT_CONSTRUCTORS(MimdRateControl);
-};
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_MIMD_RATE_CONTROL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
index fa5b8f71d7..d337aac73d 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
@@ -10,25 +10,71 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
 
-#include <algorithm>
 #include <math.h>
 #include <stdlib.h>
 
+#include <algorithm>
+#include <sstream>
+#include <string>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
-enum { kOverUsingTimeThreshold = 100 };
+const char kAdaptiveThresholdExperiment[] = "WebRTC-AdaptiveBweThreshold";
+const char kEnabledPrefix[] = "Enabled";
+const size_t kEnabledPrefixLength = sizeof(kEnabledPrefix) - 1;
+const size_t kMinExperimentLength = kEnabledPrefixLength + 3;
+
+const double kMaxAdaptOffsetMs = 15.0;
+const double kOverUsingTimeThreshold = 10;
+
+bool AdaptiveThresholdExperimentIsEnabled() {
+#ifdef CONVERT_TO_MOZILLA_ABOUT_CONFIG
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kAdaptiveThresholdExperiment);
+  if (experiment_string.length() < kMinExperimentLength)
+    return false;
+  return experiment_string.substr(0, kEnabledPrefixLength) == kEnabledPrefix;
+#else
+  return false;
+#endif
+}
+
+// Gets thresholds from the experiment name following the format
+// "WebRTC-AdaptiveBweThreshold/Enabled-0.5,0.002/".
+bool ReadExperimentConstants(double* k_up, double* k_down) {
+#ifdef CONVERT_TO_MOZILLA_ABOUT_CONFIG
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kAdaptiveThresholdExperiment);
+  return sscanf(experiment_string.substr(kEnabledPrefixLength + 1).c_str(),
+                "%lf,%lf", k_up, k_down) == 2;
+#else
+  return false;
+#endif
+}
 
 OveruseDetector::OveruseDetector(const OverUseDetectorOptions& options)
-    : options_(options),
-      threshold_(options_.initial_threshold),
+    : in_experiment_(AdaptiveThresholdExperimentIsEnabled()),
+      k_up_(0.01),
+      k_down_(0.00018),
+      overusing_time_threshold_(100),
+      options_(options),
+      threshold_(12.5),
+      last_update_ms_(-1),
       prev_offset_(0.0),
       time_over_using_(-1),
       overuse_counter_(0),
-      hypothesis_(kBwNormal) {}
+      hypothesis_(kBwNormal) {
+  if (in_experiment_)
+    InitializeExperiment();
+}
 
 OveruseDetector::~OveruseDetector() {}
 
@@ -36,29 +82,18 @@ BandwidthUsage OveruseDetector::State() const {
   return hypothesis_;
 }
 
-
-void OveruseDetector::SetRateControlRegion(RateControlRegion region) {
-  switch (region) {
-    case kRcMaxUnknown: {
-      threshold_ = options_.initial_threshold;
-      break;
-    }
-    case kRcAboveMax:
-    case kRcNearMax: {
-      threshold_ = options_.initial_threshold / 2;
-      break;
-    }
-  }
-}
-
-BandwidthUsage OveruseDetector::Detect(double offset, double ts_delta,
-                                       int num_of_deltas) {
+BandwidthUsage OveruseDetector::Detect(double offset,
+                                       double ts_delta,
+                                       int num_of_deltas,
+                                       int64_t now_ms) {
   if (num_of_deltas < 2) {
     return kBwNormal;
   }
   const double prev_offset = prev_offset_;
   prev_offset_ = offset;
   const double T = std::min(num_of_deltas, 60) * offset;
+  BWE_TEST_LOGGING_PLOT(1, "offset", now_ms, T);
+  BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
   if (T > threshold_) {
     if (time_over_using_ == -1) {
       // Initialize the timer. Assume that we've been
@@ -70,8 +105,7 @@ BandwidthUsage OveruseDetector::Detect(double offset, double ts_delta,
       time_over_using_ += ts_delta;
     }
     overuse_counter_++;
-    if (time_over_using_ > kOverUsingTimeThreshold
-        && overuse_counter_ > 1) {
+    if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
       if (offset >= prev_offset) {
         time_over_using_ = 0;
         overuse_counter_ = 0;
@@ -87,6 +121,45 @@ BandwidthUsage OveruseDetector::Detect(double offset, double ts_delta,
     overuse_counter_ = 0;
     hypothesis_ = kBwNormal;
   }
+
+  UpdateThreshold(T, now_ms);
+
   return hypothesis_;
 }
+
+void OveruseDetector::UpdateThreshold(double modified_offset, int64_t now_ms) {
+  if (!in_experiment_)
+    return;
+
+  if (last_update_ms_ == -1)
+    last_update_ms_ = now_ms;
+
+  if (fabs(modified_offset) > threshold_ + kMaxAdaptOffsetMs) {
+    // Avoid adapting the threshold to big latency spikes, caused e.g.,
+    // by a sudden capacity drop.
+    last_update_ms_ = now_ms;
+    return;
+  }
+
+  const double k = fabs(modified_offset) < threshold_ ? k_down_ : k_up_;
+  threshold_ +=
+      k * (fabs(modified_offset) - threshold_) * (now_ms - last_update_ms_);
+
+  const double kMinThreshold = 6;
+  const double kMaxThreshold = 600;
+  threshold_ = std::min(std::max(threshold_, kMinThreshold), kMaxThreshold);
+
+  last_update_ms_ = now_ms;
+}
+
+void OveruseDetector::InitializeExperiment() {
+  RTC_DCHECK(in_experiment_);
+  double k_up = 0.0;
+  double k_down = 0.0;
+  overusing_time_threshold_ = kOverUsingTimeThreshold;
+  if (ReadExperimentConstants(&k_up, &k_down)) {
+    k_up_ = k_up;
+    k_down_ = k_down;
+  }
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
index 2a6cdd764f..56e9c14206 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
@@ -13,17 +13,19 @@
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 enum RateControlRegion;
 
+bool AdaptiveThresholdExperimentIsEnabled();
+
 class OveruseDetector {
  public:
   explicit OveruseDetector(const OverUseDetectorOptions& options);
-  ~OveruseDetector();
+  virtual ~OveruseDetector();
 
   // Update the detection state based on the estimated inter-arrival time delta
   // offset. |timestamp_delta| is the delta between the last timestamp which the
@@ -31,27 +33,33 @@ class OveruseDetector {
   // offset was based on, representing the time between detector updates.
   // |num_of_deltas| is the number of deltas the offset estimate is based on.
   // Returns the state after the detection update.
-  BandwidthUsage Detect(double offset, double timestamp_delta,
-                        int num_of_deltas);
+  BandwidthUsage Detect(double offset,
+                        double timestamp_delta,
+                        int num_of_deltas,
+                        int64_t now_ms);
 
   // Returns the current detector state.
   BandwidthUsage State() const;
 
-  // Sets the current rate-control region as decided by RemoteRateControl. This
-  // affects the sensitivity of the detector.
-  void SetRateControlRegion(webrtc::RateControlRegion region);
-
  private:
+  void UpdateThreshold(double modified_offset, int64_t now_ms);
+  void InitializeExperiment();
+
+  const bool in_experiment_;
+  double k_up_;
+  double k_down_;
+  double overusing_time_threshold_;
   // Must be first member variable. Cannot be const because we need to be
   // copyable.
   webrtc::OverUseDetectorOptions options_;
   double threshold_;
+  int64_t last_update_ms_;
   double prev_offset_;
   double time_over_using_;
   int overuse_counter_;
   BandwidthUsage hypothesis_;
 
-  DISALLOW_COPY_AND_ASSIGN(OveruseDetector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OveruseDetector);
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
index 59a3056a3b..50909ebd01 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
@@ -9,17 +9,21 @@
  */
 
 #include <math.h>
+
+#include <algorithm>
 #include <cmath>
 #include <cstdlib>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
 #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
+#include "webrtc/test/field_trial.h"
 
 namespace webrtc {
 namespace testing {
@@ -27,25 +31,19 @@ namespace testing {
 const double kRtpTimestampToMs = 1.0 / 90.0;
 
 class OveruseDetectorTest : public ::testing::Test {
+ public:
+  OveruseDetectorTest()
+      : now_ms_(0),
+        receive_time_ms_(0),
+        rtp_timestamp_(10 * 90),
+        overuse_detector_(),
+        overuse_estimator_(new OveruseEstimator(options_)),
+        inter_arrival_(new InterArrival(5 * 90, kRtpTimestampToMs, true)),
+        random_(123456789) {}
+
  protected:
-  void SetUp() {
-    srand(1234);
-    now_ms_ = 0;
-    receive_time_ms_ = 0;
-    rtp_timestamp_ = 10 * 90;
+  void SetUp() override {
     overuse_detector_.reset(new OveruseDetector(options_));
-    overuse_estimator_.reset(new OveruseEstimator(options_));
-    inter_arrival_.reset(new InterArrival(5 * 90, kRtpTimestampToMs, true));
-  }
-  // Normal Distribution.
-  #define PI  3.14159265
-  int GaussianRandom(int mean_ms, int standard_deviation_ms) {
-    // Creating a Normal distribution variable from two independent uniform
-    // variables based on the Box-Muller transform.
-    double uniform1 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
-    double uniform2 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
-    return static_cast<int>(mean_ms + standard_deviation_ms *
-        sqrt(-2 * log(uniform1)) * cos(2 * PI * uniform2));
   }
 
   int Run100000Samples(int packets_per_frame, size_t packet_size, int mean_ms,
@@ -58,8 +56,10 @@ class OveruseDetectorTest : public ::testing::Test {
       }
       rtp_timestamp_ += mean_ms * 90;
       now_ms_ += mean_ms;
-      receive_time_ms_ = std::max(receive_time_ms_,
-          now_ms_ + GaussianRandom(0, standard_deviation_ms));
+      receive_time_ms_ = std::max<int64_t>(
+          receive_time_ms_,
+          now_ms_ + static_cast<int64_t>(
+                        random_.Gaussian(0, standard_deviation_ms) + 0.5));
       if (kBwOverusing == overuse_detector_->State()) {
         if (last_overuse + 1 != i) {
           unique_overuse++;
@@ -79,8 +79,10 @@ class OveruseDetectorTest : public ::testing::Test {
       }
       rtp_timestamp_ += mean_ms * 90;
       now_ms_ += mean_ms + drift_per_frame_ms;
-      receive_time_ms_ = std::max(receive_time_ms_,
-          now_ms_ + GaussianRandom(0, standard_deviation_ms));
+      receive_time_ms_ = std::max<int64_t>(
+          receive_time_ms_,
+          now_ms_ + static_cast<int64_t>(
+                        random_.Gaussian(0, standard_deviation_ms) + 0.5));
       if (kBwOverusing == overuse_detector_->State()) {
         return i + 1;
       }
@@ -102,9 +104,9 @@ class OveruseDetectorTest : public ::testing::Test {
       double timestamp_delta_ms = timestamp_delta / 90.0;
       overuse_estimator_->Update(time_delta, timestamp_delta_ms, size_delta,
                                  overuse_detector_->State());
-      overuse_detector_->Detect(overuse_estimator_->offset(),
-                                timestamp_delta_ms,
-                                overuse_estimator_->num_of_deltas());
+      overuse_detector_->Detect(
+          overuse_estimator_->offset(), timestamp_delta_ms,
+          overuse_estimator_->num_of_deltas(), receive_time_ms);
     }
   }
 
@@ -115,13 +117,14 @@ class OveruseDetectorTest : public ::testing::Test {
   rtc::scoped_ptr<OveruseDetector> overuse_detector_;
   rtc::scoped_ptr<OveruseEstimator> overuse_estimator_;
   rtc::scoped_ptr<InterArrival> inter_arrival_;
+  Random random_;
 };
 
 TEST_F(OveruseDetectorTest, GaussianRandom) {
   int buckets[100];
   memset(buckets, 0, sizeof(buckets));
   for (int i = 0; i < 100000; ++i) {
-    int index = GaussianRandom(49, 10);
+    int index = random_.Gaussian(49, 10);
     if (index >= 0 && index < 100)
       buckets[index]++;
   }
@@ -192,7 +195,7 @@ TEST_F(OveruseDetectorTest, SimpleOveruse2000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(6, frames_until_overuse);
+  EXPECT_EQ(8, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, SimpleOveruse100kbit10fps) {
@@ -207,7 +210,7 @@ TEST_F(OveruseDetectorTest, SimpleOveruse100kbit10fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_EQ(4, frames_until_overuse);
+  EXPECT_EQ(6, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, DISABLED_OveruseWithHighVariance100Kbit10fps) {
@@ -222,7 +225,7 @@ TEST_F(OveruseDetectorTest, DISABLED_OveruseWithHighVariance100Kbit10fps) {
     UpdateDetector(rtp_timestamp, now_ms_, packet_size);
     rtp_timestamp += frame_duration_ms * 90;
     if (i % 2) {
-      offset = rand() % 50;
+      offset = random_.Rand(0, 49);
       now_ms_ += frame_duration_ms - offset;
     } else {
       now_ms_ += frame_duration_ms + offset;
@@ -254,7 +257,7 @@ TEST_F(OveruseDetectorTest, DISABLED_OveruseWithLowVariance100Kbit10fps) {
     UpdateDetector(rtp_timestamp, now_ms_, packet_size);
     rtp_timestamp += frame_duration_ms * 90;
     if (i % 2) {
-      offset = rand() % 2;
+      offset = random_.Rand(0, 1);
       now_ms_ += frame_duration_ms - offset;
     } else {
       now_ms_ += frame_duration_ms + offset;
@@ -290,7 +293,7 @@ TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
     UpdateDetector(rtp_timestamp, now_ms_, packet_size);
     rtp_timestamp += frame_duration_ms * 90;
     if (i % 2) {
-      offset = rand() % 2;
+      offset = random_.Rand(0, 1);
       now_ms_ += frame_duration_ms - offset;
     } else {
       now_ms_ += frame_duration_ms + offset;
@@ -314,8 +317,13 @@ TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
   EXPECT_EQ(kBwOverusing, overuse_detector_->State());
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance30Kbit3fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance30Kbit3fps \
+  DISABLED_LowGaussianVariance30Kbit3fps
+#else
+#define MAYBE_LowGaussianVariance30Kbit3fps LowGaussianVariance30Kbit3fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance30Kbit3fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 333;
@@ -323,10 +331,10 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(0, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(29, frames_until_overuse, 5);
+  EXPECT_EQ(13, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift30Kbit3fps) {
@@ -337,10 +345,10 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift30Kbit3fps) {
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(0, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(4, frames_until_overuse, 1);
+  EXPECT_EQ(4, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVariance30Kbit3fps) {
@@ -351,10 +359,10 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance30Kbit3fps) {
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(0, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(79, frames_until_overuse, 30);
+  EXPECT_EQ(32, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
@@ -365,14 +373,19 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(0, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(4, frames_until_overuse, 1);
+  EXPECT_EQ(4, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance100Kbit5fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance100Kbit5fps \
+  DISABLED_LowGaussianVariance100Kbit5fps
+#else
+#define MAYBE_LowGaussianVariance100Kbit5fps LowGaussianVariance100Kbit5fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit5fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 2;
   int frame_duration_ms = 200;
@@ -383,11 +396,16 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(29, frames_until_overuse, 5);
+  EXPECT_EQ(13, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(HighGaussianVariance100Kbit5fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_HighGaussianVariance100Kbit5fps \
+  DISABLED_HighGaussianVariance100Kbit5fps
+#else
+#define MAYBE_HighGaussianVariance100Kbit5fps HighGaussianVariance100Kbit5fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit5fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 2;
   int frame_duration_ms = 200;
@@ -395,14 +413,19 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 10;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(0, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(79, frames_until_overuse, 15);
+  EXPECT_EQ(32, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance100Kbit10fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance100Kbit10fps \
+  DISABLED_LowGaussianVariance100Kbit10fps
+#else
+#define MAYBE_LowGaussianVariance100Kbit10fps LowGaussianVariance100Kbit10fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit10fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 100;
@@ -410,14 +433,19 @@ TEST_F(OveruseDetectorTest,
   int sigma_ms = 3;
   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
                                         frame_duration_ms, sigma_ms);
-  EXPECT_EQ(0, unique_overuse);
+  EXPECT_EQ(1, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(29, frames_until_overuse, 5);
+  EXPECT_EQ(13, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(HighGaussianVariance100Kbit10fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_HighGaussianVariance100Kbit10fps \
+  DISABLED_HighGaussianVariance100Kbit10fps
+#else
+#define MAYBE_HighGaussianVariance100Kbit10fps HighGaussianVariance100Kbit10fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit10fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 100;
@@ -428,11 +456,16 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(79, frames_until_overuse, 15);
+  EXPECT_EQ(32, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance300Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance300Kbit30fps \
+  DISABLED_LowGaussianVariance300Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance300Kbit30fps LowGaussianVariance300Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance300Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 1;
   int frame_duration_ms = 33;
@@ -443,7 +476,7 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(30, frames_until_overuse, 5);
+  EXPECT_EQ(15, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift300Kbit30fps) {
@@ -457,7 +490,7 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift300Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(7, frames_until_overuse, 1);
+  EXPECT_EQ(6, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVariance300Kbit30fps) {
@@ -471,7 +504,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance300Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(98, frames_until_overuse, 22);
+  EXPECT_EQ(41, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
@@ -485,11 +518,16 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(12, frames_until_overuse, 2);
+  EXPECT_EQ(10, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance1000Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance1000Kbit30fps \
+  DISABLED_LowGaussianVariance1000Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance1000Kbit30fps LowGaussianVariance1000Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance1000Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 3;
   int frame_duration_ms = 33;
@@ -500,7 +538,7 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(30, frames_until_overuse, 5);
+  EXPECT_EQ(15, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift1000Kbit30fps) {
@@ -514,7 +552,7 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift1000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(7, frames_until_overuse, 1);
+  EXPECT_EQ(6, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVariance1000Kbit30fps) {
@@ -528,7 +566,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance1000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(98, frames_until_overuse, 22);
+  EXPECT_EQ(41, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
@@ -542,11 +580,16 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(12, frames_until_overuse, 2);
+  EXPECT_EQ(10, frames_until_overuse);
 }
 
-TEST_F(OveruseDetectorTest,
-       DISABLED_ON_ANDROID(LowGaussianVariance2000Kbit30fps)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LowGaussianVariance2000Kbit30fps \
+  DISABLED_LowGaussianVariance2000Kbit30fps
+#else
+#define MAYBE_LowGaussianVariance2000Kbit30fps LowGaussianVariance2000Kbit30fps
+#endif
+TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance2000Kbit30fps) {
   size_t packet_size = 1200;
   int packets_per_frame = 6;
   int frame_duration_ms = 33;
@@ -557,7 +600,7 @@ TEST_F(OveruseDetectorTest,
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(30, frames_until_overuse, 5);
+  EXPECT_EQ(15, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift2000Kbit30fps) {
@@ -571,7 +614,7 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift2000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(7, frames_until_overuse, 1);
+  EXPECT_EQ(6, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVariance2000Kbit30fps) {
@@ -585,7 +628,7 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance2000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(98, frames_until_overuse, 22);
+  EXPECT_EQ(41, frames_until_overuse);
 }
 
 TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift2000Kbit30fps) {
@@ -599,7 +642,142 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift2000Kbit30fps) {
   EXPECT_EQ(0, unique_overuse);
   int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
       frame_duration_ms, sigma_ms, drift_per_frame_ms);
-  EXPECT_NEAR(12, frames_until_overuse, 2);
+  EXPECT_EQ(10, frames_until_overuse);
+}
+
+class OveruseDetectorExperimentTest : public OveruseDetectorTest {
+ public:
+  OveruseDetectorExperimentTest()
+      : override_field_trials_(
+            "WebRTC-AdaptiveBweThreshold/Enabled-0.01,0.00018/") {}
+
+ protected:
+  void SetUp() override {
+    overuse_detector_.reset(new OveruseDetector(options_));
+  }
+
+  test::ScopedFieldTrials override_field_trials_;
+};
+
+TEST_F(OveruseDetectorExperimentTest, ThresholdAdapts) {
+  const double kOffset = 0.21;
+  double kTsDelta = 3000.0;
+  int64_t now_ms = 0;
+  int num_deltas = 60;
+  const int kBatchLength = 10;
+
+  // Pass in a positive offset and verify it triggers overuse.
+  bool overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+
+  // Force the threshold to increase by passing in a higher offset.
+  overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(1.1 * kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+
+  // Verify that the same offset as before no longer triggers overuse.
+  overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_FALSE(overuse_detected);
+
+  // Pass in a low offset to make the threshold adapt down.
+  for (int i = 0; i < 15 * kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(0.7 * kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_FALSE(overuse_detected);
+
+  // Make sure the original offset now again triggers overuse.
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+}
+
+TEST_F(OveruseDetectorExperimentTest, DoesntAdaptToSpikes) {
+  const double kOffset = 1.0;
+  const double kLargeOffset = 20.0;
+  double kTsDelta = 3000.0;
+  int64_t now_ms = 0;
+  int num_deltas = 60;
+  const int kBatchLength = 10;
+  const int kShortBatchLength = 3;
+
+  // Pass in a positive offset and verify it triggers overuse.
+  bool overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+
+  // Pass in a large offset. This shouldn't have a too big impact on the
+  // threshold, but still trigger an overuse.
+  now_ms += 100;
+  overuse_detected = false;
+  for (int i = 0; i < kShortBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kLargeOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+
+  // Pass in a positive normal offset and verify it still triggers.
+  overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
 }
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
index 2f6e3305ac..83917912e8 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
@@ -10,14 +10,15 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
 
-#include <algorithm>
 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include <algorithm>
+
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
@@ -146,8 +147,8 @@ void OveruseEstimator::UpdateNoiseEstimate(double residual,
               + (1 - beta) * residual;
   var_noise_ = beta * var_noise_
               + (1 - beta) * (avg_noise_ - residual) * (avg_noise_ - residual);
-  if (var_noise_ < 1e-7) {
-    var_noise_ = 1e-7;
+  if (var_noise_ < 1) {
+    var_noise_ = 1;
   }
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
index 6499d8d043..d671f39166 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
@@ -63,7 +63,7 @@ class OveruseEstimator {
   double var_noise_;
   std::list<double> ts_delta_hist_;
 
-  DISALLOW_COPY_AND_ASSIGN(OveruseEstimator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OveruseEstimator);
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 84e8324674..d2af81e2f0 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -21,12 +21,11 @@
       'sources': [
         'include/bwe_defines.h',
         'include/remote_bitrate_estimator.h',
+        'include/send_time_history.h',
         'aimd_rate_control.cc',
         'aimd_rate_control.h',
         'inter_arrival.cc',
         'inter_arrival.h',
-        'mimd_rate_control.cc',
-        'mimd_rate_control.h',
         'overuse_detector.cc',
         'overuse_detector.h',
         'overuse_estimator.cc',
@@ -34,21 +33,83 @@
         'rate_statistics.cc',
         'rate_statistics.h',
         'remote_bitrate_estimator_abs_send_time.cc',
+        'remote_bitrate_estimator_abs_send_time.h',
         'remote_bitrate_estimator_single_stream.cc',
-        'remote_rate_control.cc',
-        'remote_rate_control.h',
+        'remote_bitrate_estimator_single_stream.h',
+        'remote_estimator_proxy.cc',
+        'remote_estimator_proxy.h',
+        'send_time_history.cc',
+        'transport_feedback_adapter.cc',
+        'transport_feedback_adapter.h',
         'test/bwe_test_logging.cc',
         'test/bwe_test_logging.h',
       ], # source
+      'conditions': [
+        ['enable_bwe_test_logging==1', {
+          'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
+        }, {
+          'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
+          'sources!': [
+            'remote_bitrate_estimator/test/bwe_test_logging.cc'
+          ],
+        }],
+      ],
     },
   ], # targets
   'conditions': [
     ['include_tests==1', {
       'targets': [
+        {
+          'target_name': 'bwe_simulator',
+          'type': 'static_library',
+          'dependencies': [
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'test/bwe.cc',
+            'test/bwe.h',
+            'test/bwe_test.cc',
+            'test/bwe_test.h',
+            'test/bwe_test_baselinefile.cc',
+            'test/bwe_test_baselinefile.h',
+            'test/bwe_test_fileutils.cc',
+            'test/bwe_test_fileutils.h',
+            'test/bwe_test_framework.cc',
+            'test/bwe_test_framework.h',
+            'test/bwe_test_logging.cc',
+            'test/bwe_test_logging.h',
+            'test/metric_recorder.cc',
+            'test/metric_recorder.h',
+            'test/packet_receiver.cc',
+            'test/packet_receiver.h',
+            'test/packet_sender.cc',
+            'test/packet_sender.h',
+            'test/packet.h',
+            'test/estimators/nada.cc',
+            'test/estimators/nada.h',
+            'test/estimators/remb.cc',
+            'test/estimators/remb.h',
+            'test/estimators/send_side.cc',
+            'test/estimators/send_side.h',
+            'test/estimators/tcp.cc',
+            'test/estimators/tcp.h',
+          ],
+          'conditions': [
+            ['enable_bwe_test_logging==1', {
+              'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
+            }, {
+              'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
+              'sources!': [
+                'remote_bitrate_estimator/test/bwe_test_logging.cc'
+              ],
+            }],
+          ],
+        },
         {
           'target_name': 'bwe_tools_util',
           'type': 'static_library',
           'dependencies': [
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             'rtp_rtcp',
           ],
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 65e5401494..97e5cd32e5 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -8,21 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+
 #include <math.h>
-#include <map>
+
+#include <algorithm>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
-#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
-#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
-#include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h"
-#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -67,73 +66,18 @@ std::vector<K> Keys(const std::map<K, V>& map) {
   return keys;
 }
 
-struct Probe {
-  Probe(int64_t send_time_ms, int64_t recv_time_ms, size_t payload_size)
-      : send_time_ms(send_time_ms),
-        recv_time_ms(recv_time_ms),
-        payload_size(payload_size) {}
-  int64_t send_time_ms;
-  int64_t recv_time_ms;
-  size_t payload_size;
-};
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(
+          ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+          1000) &
+      0x00FFFFFF;
+  return time_24_bits;
+}
 
-struct Cluster {
-  Cluster()
-      : send_mean_ms(0.0f),
-        recv_mean_ms(0.0f),
-        mean_size(0),
-        count(0),
-        num_above_min_delta(0) {}
-
-  int GetSendBitrateBps() const {
-    assert(send_mean_ms > 0);
-    return mean_size * 8 * 1000 / send_mean_ms;
-  }
-
-  int GetRecvBitrateBps() const {
-    assert(recv_mean_ms > 0);
-    return mean_size * 8 * 1000 / recv_mean_ms;
-  }
-
-  float send_mean_ms;
-  float recv_mean_ms;
-  // TODO(holmer): Add some variance metric as well?
-  size_t mean_size;
-  int count;
-  int num_above_min_delta;
-};
-
-class RemoteBitrateEstimatorAbsSendTimeImpl : public RemoteBitrateEstimator {
- public:
-  RemoteBitrateEstimatorAbsSendTimeImpl(RemoteBitrateObserver* observer,
-                                        Clock* clock,
-                                        RateControlType control_type,
-                                        uint32_t min_bitrate_bps);
-  virtual ~RemoteBitrateEstimatorAbsSendTimeImpl() {}
-
-  void IncomingPacketFeedbackVector(
-      const std::vector<PacketInfo>& packet_feedback_vector) override;
-
-  void IncomingPacket(int64_t arrival_time_ms,
-                      size_t payload_size,
-                      const RTPHeader& header) override;
-  // This class relies on Process() being called periodically (at least once
-  // every other second) for streams to be timed out properly. Therefore it
-  // shouldn't be detached from the ProcessThread except if it's about to be
-  // deleted.
-  int32_t Process() override;
-  int64_t TimeUntilNextProcess() override;
-  void OnRttUpdate(int64_t rtt) override;
-  void RemoveStream(unsigned int ssrc) override;
-  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
-                      unsigned int* bitrate_bps) const override;
-  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override;
-
- private:
-  typedef std::map<unsigned int, int64_t> Ssrcs;
-
-  static bool IsWithinClusterBounds(int send_delta_ms,
-                                    const Cluster& cluster_aggregate) {
+bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds(
+    int send_delta_ms,
+    const Cluster& cluster_aggregate) {
     if (cluster_aggregate.count == 0)
       return true;
     float cluster_mean = cluster_aggregate.send_mean_ms /
@@ -141,90 +85,41 @@ class RemoteBitrateEstimatorAbsSendTimeImpl : public RemoteBitrateEstimator {
     return fabs(static_cast<float>(send_delta_ms) - cluster_mean) < 2.5f;
   }
 
-  static void AddCluster(std::list<Cluster>* clusters, Cluster* cluster) {
+  void RemoteBitrateEstimatorAbsSendTime::AddCluster(
+      std::list<Cluster>* clusters,
+      Cluster* cluster) {
     cluster->send_mean_ms /= static_cast<float>(cluster->count);
     cluster->recv_mean_ms /= static_cast<float>(cluster->count);
     cluster->mean_size /= cluster->count;
     clusters->push_back(*cluster);
   }
 
-  int Id() const {
+  int RemoteBitrateEstimatorAbsSendTime::Id() const {
     return static_cast<int>(reinterpret_cast<uint64_t>(this));
   }
 
-  void IncomingPacketInfo(int64_t arrival_time_ms,
-                          uint32_t send_time_24bits,
-                          size_t payload_size,
-                          uint32_t ssrc);
-
-  bool IsProbe(int64_t send_time_ms, int payload_size) const
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  // Triggers a new estimate calculation.
-  void UpdateEstimate(int64_t now_ms)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  void UpdateStats(int propagation_delta_ms, int64_t now_ms)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  void ComputeClusters(std::list<Cluster>* clusters) const;
-
-  std::list<Cluster>::const_iterator FindBestProbe(
-      const std::list<Cluster>& clusters) const
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  void ProcessClusters(int64_t now_ms)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  bool IsBitrateImproving(int probe_bitrate_bps) const
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  RemoteBitrateObserver* observer_ GUARDED_BY(crit_sect_.get());
-  Clock* clock_;
-  Ssrcs ssrcs_ GUARDED_BY(crit_sect_.get());
-  rtc::scoped_ptr<InterArrival> inter_arrival_ GUARDED_BY(crit_sect_.get());
-  OveruseEstimator estimator_ GUARDED_BY(crit_sect_.get());
-  OveruseDetector detector_ GUARDED_BY(crit_sect_.get());
-  RateStatistics incoming_bitrate_ GUARDED_BY(crit_sect_.get());
-  rtc::scoped_ptr<RemoteRateControl> remote_rate_ GUARDED_BY(crit_sect_.get());
-  int64_t last_process_time_;
-  std::vector<int> recent_propagation_delta_ms_ GUARDED_BY(crit_sect_.get());
-  std::vector<int64_t> recent_update_time_ms_ GUARDED_BY(crit_sect_.get());
-  int64_t process_interval_ms_ GUARDED_BY(crit_sect_.get());
-  int total_propagation_delta_ms_ GUARDED_BY(crit_sect_.get());
-
-  std::list<Probe> probes_;
-  size_t total_probes_received_;
-  int64_t first_packet_time_ms_;
-
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorAbsSendTimeImpl);
-};
-
-RemoteBitrateEstimatorAbsSendTimeImpl::RemoteBitrateEstimatorAbsSendTimeImpl(
-    RemoteBitrateObserver* observer,
-    Clock* clock,
-    RateControlType control_type,
-    uint32_t min_bitrate_bps)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      observer_(observer),
-      clock_(clock),
-      ssrcs_(),
-      inter_arrival_(),
-      estimator_(OverUseDetectorOptions()),
-      detector_(OverUseDetectorOptions()),
-      incoming_bitrate_(1000, 8000),
-      remote_rate_(RemoteRateControl::Create(control_type, min_bitrate_bps)),
-      last_process_time_(-1),
-      process_interval_ms_(kProcessIntervalMs),
-      total_propagation_delta_ms_(0),
-      total_probes_received_(0),
-      first_packet_time_ms_(-1) {
+  RemoteBitrateEstimatorAbsSendTime::RemoteBitrateEstimatorAbsSendTime(
+      RemoteBitrateObserver* observer,
+      Clock* clock)
+      : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+        observer_(observer),
+        clock_(clock),
+        ssrcs_(),
+        inter_arrival_(),
+        estimator_(OverUseDetectorOptions()),
+        detector_(OverUseDetectorOptions()),
+        incoming_bitrate_(kBitrateWindowMs, 8000),
+        last_process_time_(-1),
+        process_interval_ms_(kProcessIntervalMs),
+        total_propagation_delta_ms_(0),
+        total_probes_received_(0),
+        first_packet_time_ms_(-1) {
   assert(observer_);
   assert(clock_);
+  LOG(LS_INFO) << "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::ComputeClusters(
+void RemoteBitrateEstimatorAbsSendTime::ComputeClusters(
     std::list<Cluster>* clusters) const {
   Cluster current;
   int64_t prev_send_time = -1;
@@ -256,7 +151,7 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::ComputeClusters(
 }
 
 std::list<Cluster>::const_iterator
-RemoteBitrateEstimatorAbsSendTimeImpl::FindBestProbe(
+RemoteBitrateEstimatorAbsSendTime::FindBestProbe(
     const std::list<Cluster>& clusters) const {
   int highest_probe_bitrate_bps = 0;
   std::list<Cluster>::const_iterator best_it = clusters.end();
@@ -288,7 +183,7 @@ RemoteBitrateEstimatorAbsSendTimeImpl::FindBestProbe(
   return best_it;
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::ProcessClusters(int64_t now_ms) {
+void RemoteBitrateEstimatorAbsSendTime::ProcessClusters(int64_t now_ms) {
   std::list<Cluster> clusters;
   ComputeClusters(&clusters);
   if (clusters.empty()) {
@@ -303,14 +198,17 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::ProcessClusters(int64_t now_ms) {
   if (best_it != clusters.end()) {
     int probe_bitrate_bps =
         std::min(best_it->GetSendBitrateBps(), best_it->GetRecvBitrateBps());
-    if (IsBitrateImproving(probe_bitrate_bps)) {
+    // Make sure that a probe sent on a lower bitrate than our estimate can't
+    // reduce the estimate.
+    if (IsBitrateImproving(probe_bitrate_bps) &&
+        probe_bitrate_bps > static_cast<int>(incoming_bitrate_.Rate(now_ms))) {
       LOG(LS_INFO) << "Probe successful, sent at "
                    << best_it->GetSendBitrateBps() << " bps, received at "
                    << best_it->GetRecvBitrateBps()
                    << " bps. Mean send delta: " << best_it->send_mean_ms
                    << " ms, mean recv delta: " << best_it->recv_mean_ms
                    << " ms, num probes: " << best_it->count;
-      remote_rate_->SetEstimate(probe_bitrate_bps, now_ms);
+      remote_rate_.SetEstimate(probe_bitrate_bps, now_ms);
     }
   }
 
@@ -320,45 +218,43 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::ProcessClusters(int64_t now_ms) {
     probes_.clear();
 }
 
-bool RemoteBitrateEstimatorAbsSendTimeImpl::IsBitrateImproving(
+bool RemoteBitrateEstimatorAbsSendTime::IsBitrateImproving(
     int new_bitrate_bps) const {
-  bool initial_probe = !remote_rate_->ValidEstimate() && new_bitrate_bps > 0;
+  bool initial_probe = !remote_rate_.ValidEstimate() && new_bitrate_bps > 0;
   bool bitrate_above_estimate =
-      remote_rate_->ValidEstimate() &&
-      new_bitrate_bps > static_cast<int>(remote_rate_->LatestEstimate());
+      remote_rate_.ValidEstimate() &&
+      new_bitrate_bps > static_cast<int>(remote_rate_.LatestEstimate());
   return initial_probe || bitrate_above_estimate;
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::IncomingPacketFeedbackVector(
+void RemoteBitrateEstimatorAbsSendTime::IncomingPacketFeedbackVector(
     const std::vector<PacketInfo>& packet_feedback_vector) {
   for (const auto& packet_info : packet_feedback_vector) {
-    // TODO(holmer): We should get rid of this conversion if possible as we may
-    // lose precision.
-    uint32_t send_time_32bits = (packet_info.send_time_ms) / kTimestampToMs;
-    uint32_t send_time_24bits =
-        send_time_32bits >> kAbsSendTimeInterArrivalUpshift;
-    IncomingPacketInfo(packet_info.arrival_time_ms, send_time_24bits,
-                       packet_info.payload_size, 0);
+    IncomingPacketInfo(packet_info.arrival_time_ms,
+                       ConvertMsTo24Bits(packet_info.send_time_ms),
+                       packet_info.payload_size, 0, packet_info.was_paced);
   }
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::IncomingPacket(
-    int64_t arrival_time_ms,
-    size_t payload_size,
-    const RTPHeader& header) {
+void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(int64_t arrival_time_ms,
+                                                       size_t payload_size,
+                                                       const RTPHeader& header,
+                                                       bool was_paced) {
   if (!header.extension.hasAbsoluteSendTime) {
     LOG(LS_WARNING) << "RemoteBitrateEstimatorAbsSendTimeImpl: Incoming packet "
                        "is missing absolute send time extension!";
+    return;
   }
   IncomingPacketInfo(arrival_time_ms, header.extension.absoluteSendTime,
-                     payload_size, header.ssrc);
+                     payload_size, header.ssrc, was_paced);
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::IncomingPacketInfo(
+void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
     int64_t arrival_time_ms,
     uint32_t send_time_24bits,
     size_t payload_size,
-    uint32_t ssrc) {
+    uint32_t ssrc,
+    bool was_paced) {
   assert(send_time_24bits < (1ul << 24));
   // Shift up send time to use the full 32 bits that inter_arrival works with,
   // so wrapping works properly.
@@ -379,9 +275,13 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::IncomingPacketInfo(
   uint32_t ts_delta = 0;
   int64_t t_delta = 0;
   int size_delta = 0;
-  // For now only try to detect probes while we don't have a valid estimate.
-  if (!remote_rate_->ValidEstimate() ||
-      now_ms - first_packet_time_ms_ < kInitialProbingIntervalMs) {
+  // For now only try to detect probes while we don't have a valid estimate, and
+  // make sure the packet was paced. We currently assume that only packets
+  // larger than 200 bytes are paced by the sender.
+  was_paced = was_paced && payload_size > PacedSender::kMinProbePacketSize;
+  if (was_paced &&
+      (!remote_rate_.ValidEstimate() ||
+       now_ms - first_packet_time_ms_ < kInitialProbingIntervalMs)) {
     // TODO(holmer): Use a map instead to get correct order?
     if (total_probes_received_ < kMaxProbePackets) {
       int send_delta_ms = -1;
@@ -400,22 +300,22 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::IncomingPacketInfo(
     ProcessClusters(now_ms);
   }
   if (!inter_arrival_.get()) {
-    inter_arrival_.reset(new InterArrival(
-        (kTimestampGroupLengthMs << kInterArrivalShift) / 1000, kTimestampToMs,
-        remote_rate_->GetControlType() == kAimdControl));
+    inter_arrival_.reset(
+        new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
+                         kTimestampToMs, true));
   }
   if (inter_arrival_->ComputeDeltas(timestamp, arrival_time_ms, payload_size,
                                     &ts_delta, &t_delta, &size_delta)) {
     double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
     estimator_.Update(t_delta, ts_delta_ms, size_delta, detector_.State());
     detector_.Detect(estimator_.offset(), ts_delta_ms,
-                     estimator_.num_of_deltas());
+                     estimator_.num_of_deltas(), arrival_time_ms);
     UpdateStats(static_cast<int>(t_delta - ts_delta_ms), now_ms);
   }
   if (detector_.State() == kBwOverusing) {
-    unsigned int incoming_bitrate = incoming_bitrate_.Rate(now_ms);
+    uint32_t incoming_bitrate_bps = incoming_bitrate_.Rate(now_ms);
     if (prior_state != kBwOverusing ||
-        remote_rate_->TimeToReduceFurther(now_ms, incoming_bitrate)) {
+        remote_rate_.TimeToReduceFurther(now_ms, incoming_bitrate_bps)) {
       // The first overuse should immediately trigger a new estimate.
       // We also have to update the estimate immediately if we are overusing
       // and the target bitrate is too high compared to what we are receiving.
@@ -424,7 +324,7 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::IncomingPacketInfo(
   }
 }
 
-int32_t RemoteBitrateEstimatorAbsSendTimeImpl::Process() {
+int32_t RemoteBitrateEstimatorAbsSendTime::Process() {
   if (TimeUntilNextProcess() > 0) {
     return 0;
   }
@@ -436,7 +336,7 @@ int32_t RemoteBitrateEstimatorAbsSendTimeImpl::Process() {
   return 0;
 }
 
-int64_t RemoteBitrateEstimatorAbsSendTimeImpl::TimeUntilNextProcess() {
+int64_t RemoteBitrateEstimatorAbsSendTime::TimeUntilNextProcess() {
   if (last_process_time_ < 0) {
     return 0;
   }
@@ -447,7 +347,7 @@ int64_t RemoteBitrateEstimatorAbsSendTimeImpl::TimeUntilNextProcess() {
   }
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::UpdateEstimate(int64_t now_ms) {
+void RemoteBitrateEstimatorAbsSendTime::UpdateEstimate(int64_t now_ms) {
   if (!inter_arrival_.get()) {
     // No packets have been received on the active streams.
     return;
@@ -470,44 +370,44 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::UpdateEstimate(int64_t now_ms) {
   const RateControlInput input(detector_.State(),
                                incoming_bitrate_.Rate(now_ms),
                                estimator_.var_noise());
-  const RateControlRegion region = remote_rate_->Update(&input, now_ms);
-  unsigned int target_bitrate = remote_rate_->UpdateBandwidthEstimate(now_ms);
-  if (remote_rate_->ValidEstimate()) {
-    process_interval_ms_ = remote_rate_->GetFeedbackInterval();
+  remote_rate_.Update(&input, now_ms);
+  unsigned int target_bitrate = remote_rate_.UpdateBandwidthEstimate(now_ms);
+  if (remote_rate_.ValidEstimate()) {
+    process_interval_ms_ = remote_rate_.GetFeedbackInterval();
     observer_->OnReceiveBitrateChanged(Keys(ssrcs_), target_bitrate);
   }
-  detector_.SetRateControlRegion(region);
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::OnRttUpdate(int64_t rtt) {
+void RemoteBitrateEstimatorAbsSendTime::OnRttUpdate(int64_t avg_rtt_ms,
+                                                    int64_t max_rtt_ms) {
   CriticalSectionScoped cs(crit_sect_.get());
-  remote_rate_->SetRtt(rtt);
+  remote_rate_.SetRtt(avg_rtt_ms);
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::RemoveStream(unsigned int ssrc) {
+void RemoteBitrateEstimatorAbsSendTime::RemoveStream(unsigned int ssrc) {
   CriticalSectionScoped cs(crit_sect_.get());
   ssrcs_.erase(ssrc);
 }
 
-bool RemoteBitrateEstimatorAbsSendTimeImpl::LatestEstimate(
+bool RemoteBitrateEstimatorAbsSendTime::LatestEstimate(
     std::vector<unsigned int>* ssrcs,
     unsigned int* bitrate_bps) const {
   CriticalSectionScoped cs(crit_sect_.get());
   assert(ssrcs);
   assert(bitrate_bps);
-  if (!remote_rate_->ValidEstimate()) {
+  if (!remote_rate_.ValidEstimate()) {
     return false;
   }
   *ssrcs = Keys(ssrcs_);
   if (ssrcs_.empty()) {
     *bitrate_bps = 0;
   } else {
-    *bitrate_bps = remote_rate_->LatestEstimate();
+    *bitrate_bps = remote_rate_.LatestEstimate();
   }
   return true;
 }
 
-bool RemoteBitrateEstimatorAbsSendTimeImpl::GetStats(
+bool RemoteBitrateEstimatorAbsSendTime::GetStats(
     ReceiveBandwidthEstimatorStats* output) const {
   {
     CriticalSectionScoped cs(crit_sect_.get());
@@ -522,8 +422,8 @@ bool RemoteBitrateEstimatorAbsSendTimeImpl::GetStats(
   return true;
 }
 
-void RemoteBitrateEstimatorAbsSendTimeImpl::UpdateStats(
-    int propagation_delta_ms, int64_t now_ms) {
+void RemoteBitrateEstimatorAbsSendTime::UpdateStats(int propagation_delta_ms,
+                                                    int64_t now_ms) {
   // The caller must enter crit_sect_ before the call.
 
   // Remove the oldest entry if the size limit is reached.
@@ -544,16 +444,8 @@ void RemoteBitrateEstimatorAbsSendTimeImpl::UpdateStats(
       std::max(total_propagation_delta_ms_ + propagation_delta_ms, 0);
 }
 
-RemoteBitrateEstimator* AbsoluteSendTimeRemoteBitrateEstimatorFactory::Create(
-    RemoteBitrateObserver* observer,
-    Clock* clock,
-    RateControlType control_type,
-    uint32_t min_bitrate_bps) const {
-  LOG(LS_INFO) << "AbsoluteSendTimeRemoteBitrateEstimatorFactory: "
-                  "Instantiating.";
-  return new RemoteBitrateEstimatorAbsSendTimeImpl(observer,
-                                                   clock,
-                                                   control_type,
-                                                   min_bitrate_bps);
+void RemoteBitrateEstimatorAbsSendTime::SetMinBitrate(int min_bitrate_bps) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  remote_rate_.SetMinBitrate(min_bitrate_bps);
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
new file mode 100644
index 0000000000..549c437faf
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_ABS_SEND_TIME_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_ABS_SEND_TIME_H_
+
+#include <list>
+#include <map>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
+#include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
+#include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+struct Probe {
+  Probe(int64_t send_time_ms, int64_t recv_time_ms, size_t payload_size)
+      : send_time_ms(send_time_ms),
+        recv_time_ms(recv_time_ms),
+        payload_size(payload_size) {}
+  int64_t send_time_ms;
+  int64_t recv_time_ms;
+  size_t payload_size;
+};
+
+struct Cluster {
+  Cluster()
+      : send_mean_ms(0.0f),
+        recv_mean_ms(0.0f),
+        mean_size(0),
+        count(0),
+        num_above_min_delta(0) {}
+
+  int GetSendBitrateBps() const {
+    RTC_CHECK_GT(send_mean_ms, 0.0f);
+    return mean_size * 8 * 1000 / send_mean_ms;
+  }
+
+  int GetRecvBitrateBps() const {
+    RTC_CHECK_GT(recv_mean_ms, 0.0f);
+    return mean_size * 8 * 1000 / recv_mean_ms;
+  }
+
+  float send_mean_ms;
+  float recv_mean_ms;
+  // TODO(holmer): Add some variance metric as well?
+  size_t mean_size;
+  int count;
+  int num_above_min_delta;
+};
+
+class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
+ public:
+  RemoteBitrateEstimatorAbsSendTime(RemoteBitrateObserver* observer,
+                                    Clock* clock);
+  virtual ~RemoteBitrateEstimatorAbsSendTime() {}
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketInfo>& packet_feedback_vector) override;
+
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header,
+                      bool was_paced) override;
+  // This class relies on Process() being called periodically (at least once
+  // every other second) for streams to be timed out properly. Therefore it
+  // shouldn't be detached from the ProcessThread except if it's about to be
+  // deleted.
+  int32_t Process() override;
+  int64_t TimeUntilNextProcess() override;
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+  void RemoveStream(unsigned int ssrc) override;
+  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
+                      unsigned int* bitrate_bps) const override;
+  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override;
+  void SetMinBitrate(int min_bitrate_bps) override;
+
+ private:
+  typedef std::map<unsigned int, int64_t> Ssrcs;
+
+  static bool IsWithinClusterBounds(int send_delta_ms,
+                                    const Cluster& cluster_aggregate);
+
+  static void AddCluster(std::list<Cluster>* clusters, Cluster* cluster);
+
+  int Id() const;
+
+  void IncomingPacketInfo(int64_t arrival_time_ms,
+                          uint32_t send_time_24bits,
+                          size_t payload_size,
+                          uint32_t ssrc,
+                          bool was_paced);
+
+  bool IsProbe(int64_t send_time_ms, int payload_size) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  // Triggers a new estimate calculation.
+  void UpdateEstimate(int64_t now_ms)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  void UpdateStats(int propagation_delta_ms, int64_t now_ms)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  void ComputeClusters(std::list<Cluster>* clusters) const;
+
+  std::list<Cluster>::const_iterator FindBestProbe(
+      const std::list<Cluster>& clusters) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  void ProcessClusters(int64_t now_ms)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  bool IsBitrateImproving(int probe_bitrate_bps) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  RemoteBitrateObserver* observer_ GUARDED_BY(crit_sect_.get());
+  Clock* clock_;
+  Ssrcs ssrcs_ GUARDED_BY(crit_sect_.get());
+  rtc::scoped_ptr<InterArrival> inter_arrival_ GUARDED_BY(crit_sect_.get());
+  OveruseEstimator estimator_ GUARDED_BY(crit_sect_.get());
+  OveruseDetector detector_ GUARDED_BY(crit_sect_.get());
+  RateStatistics incoming_bitrate_ GUARDED_BY(crit_sect_.get());
+  AimdRateControl remote_rate_ GUARDED_BY(crit_sect_.get());
+  int64_t last_process_time_;
+  std::vector<int> recent_propagation_delta_ms_ GUARDED_BY(crit_sect_.get());
+  std::vector<int64_t> recent_update_time_ms_ GUARDED_BY(crit_sect_.get());
+  int64_t process_interval_ms_ GUARDED_BY(crit_sect_.get());
+  int total_propagation_delta_ms_ GUARDED_BY(crit_sect_.get());
+
+  std::list<Probe> probes_;
+  size_t total_probes_received_;
+  int64_t first_packet_time_ms_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorAbsSendTime);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_ABS_SEND_TIME_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
index e1268dad22..908daf6c31 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
 
 namespace webrtc {
@@ -16,19 +17,13 @@ namespace webrtc {
 class RemoteBitrateEstimatorAbsSendTimeTest :
     public RemoteBitrateEstimatorTest {
  public:
-  static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
-
   RemoteBitrateEstimatorAbsSendTimeTest() {}
   virtual void SetUp() {
-    bitrate_estimator_.reset(
-        AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
-            bitrate_observer_.get(),
-            &clock_,
-            kAimdControl,
-            kRemoteBitrateEstimatorMinBitrateBps));
+    bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime(
+        bitrate_observer_.get(), &clock_));
   }
  protected:
-  DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorAbsSendTimeTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorAbsSendTimeTest);
 };
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, InitialBehavior) {
@@ -40,19 +35,19 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseReordering) {
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseRtpTimestamps) {
-  RateIncreaseRtpTimestampsTestHelper(1090);
+  RateIncreaseRtpTimestampsTestHelper(1240);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStream) {
-  CapacityDropTestHelper(1, false, 700);
+  CapacityDropTestHelper(1, false, 600);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStreamWrap) {
-  CapacityDropTestHelper(1, true, 700);
+  CapacityDropTestHelper(1, true, 600);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropTwoStreamsWrap) {
-  CapacityDropTestHelper(2, true, 700);
+  CapacityDropTestHelper(2, true, 533);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThreeStreamsWrap) {
@@ -60,15 +55,15 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThreeStreamsWrap) {
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThirteenStreamsWrap) {
-  CapacityDropTestHelper(13, true, 666);
+  CapacityDropTestHelper(13, true, 700);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropNineteenStreamsWrap) {
-  CapacityDropTestHelper(19, true, 666);
+  CapacityDropTestHelper(19, true, 700);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThirtyStreamsWrap) {
-  CapacityDropTestHelper(30, true, 666);
+  CapacityDropTestHelper(30, true, 700);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestTimestampGrouping) {
@@ -99,7 +94,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProcessAfterTimeout) {
   // RemoteBitrateEstimator.
   const int64_t kStreamTimeOutMs = 2000;
   const int64_t kProcessIntervalMs = 1000;
-  IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0);
+  IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0, true);
   clock_.AdvanceTimeMilliseconds(kStreamTimeOutMs + 1);
   // Trigger timeout.
   EXPECT_EQ(0, bitrate_estimator_->Process());
@@ -115,14 +110,16 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetection) {
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(10);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
+                   true);
   }
 
   // Second burst sent at 8 * 1000 / 5 = 1600 kbps.
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(5);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
+                   true);
   }
 
   EXPECT_EQ(0, bitrate_estimator_->Process());
@@ -130,6 +127,28 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetection) {
   EXPECT_GT(bitrate_observer_->latest_bitrate(), 1500000u);
 }
 
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
+       TestProbeDetectionNonPacedPackets) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps, but with every other packet
+  // not being paced which could mess things up.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
+                   true);
+    // Non-paced packet, arriving 5 ms after.
+    clock_.AdvanceTimeMilliseconds(5);
+    IncomingPacket(0, 100, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
+                   false);
+  }
+
+  EXPECT_EQ(0, bitrate_estimator_->Process());
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_GT(bitrate_observer_->latest_bitrate(), 800000u);
+}
+
 // Packets will require 5 ms to be transmitted to the receiver, causing packets
 // of the second probe to be dispersed.
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
@@ -143,7 +162,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     now_ms = clock_.TimeInMilliseconds();
     send_time_ms += 10;
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000));
+                   AbsSendTime(send_time_ms, 1000), true);
   }
 
   // Second burst sent at 8 * 1000 / 5 = 1600 kbps, arriving at 8 * 1000 / 8 =
@@ -153,7 +172,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     now_ms = clock_.TimeInMilliseconds();
     send_time_ms += 5;
     IncomingPacket(0, 1000, now_ms, send_time_ms,
-                   AbsSendTime(send_time_ms, 1000));
+                   AbsSendTime(send_time_ms, 1000), true);
   }
 
   EXPECT_EQ(0, bitrate_estimator_->Process());
@@ -173,7 +192,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     send_time_ms += 10;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000));
+                   AbsSendTime(send_time_ms, 1000), true);
   }
 
   EXPECT_EQ(0, bitrate_estimator_->Process());
@@ -192,7 +211,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetectionFasterArrival) {
     send_time_ms += 10;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000));
+                   AbsSendTime(send_time_ms, 1000), true);
   }
 
   EXPECT_EQ(0, bitrate_estimator_->Process());
@@ -210,7 +229,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetectionSlowerArrival) {
     send_time_ms += 5;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000));
+                   AbsSendTime(send_time_ms, 1000), true);
   }
 
   EXPECT_EQ(0, bitrate_estimator_->Process());
@@ -230,11 +249,43 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     send_time_ms += 1;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000));
+                   AbsSendTime(send_time_ms, 1000), true);
   }
 
   EXPECT_EQ(0, bitrate_estimator_->Process());
   EXPECT_TRUE(bitrate_observer_->updated());
   EXPECT_NEAR(bitrate_observer_->latest_bitrate(), 4000000u, 10000);
 }
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, ProbingIgnoresSmallPackets) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // Probing with 200 bytes every 10 ms, should be ignored by the probe
+  // detection.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 200, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
+                   true);
+  }
+
+  EXPECT_EQ(0, bitrate_estimator_->Process());
+  EXPECT_FALSE(bitrate_observer_->updated());
+
+  // Followed by a probe with 1000 bytes packets, should be detected as a
+  // probe.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
+                   true);
+  }
+
+  // Wait long enough so that we can call Process again.
+  clock_.AdvanceTimeMilliseconds(1000);
+
+  EXPECT_EQ(0, bitrate_estimator_->Process());
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(bitrate_observer_->latest_bitrate(), 800000u, 10000);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index 72c476d202..4b7732c80f 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -7,20 +7,21 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include <map>
+
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+
+#include <utility>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
 #include "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
 #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
-#include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -28,78 +29,37 @@ namespace webrtc {
 enum { kTimestampGroupLengthMs = 5 };
 static const double kTimestampToMs = 1.0 / 90.0;
 
-class RemoteBitrateEstimatorImpl : public RemoteBitrateEstimator {
- public:
-  RemoteBitrateEstimatorImpl(RemoteBitrateObserver* observer,
-                             Clock* clock,
-                             RateControlType control_type,
-                             uint32_t min_bitrate_bps);
-  virtual ~RemoteBitrateEstimatorImpl();
-
-  void IncomingPacket(int64_t arrival_time_ms,
-                      size_t payload_size,
-                      const RTPHeader& header) override;
-  int32_t Process() override;
-  int64_t TimeUntilNextProcess() override;
-  void OnRttUpdate(int64_t rtt) override;
-  void RemoveStream(unsigned int ssrc) override;
-  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
-                      unsigned int* bitrate_bps) const override;
-  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override;
-
- private:
-  struct Detector {
-    explicit Detector(int64_t last_packet_time_ms,
-                      const OverUseDetectorOptions& options,
-                      bool enable_burst_grouping)
-        : last_packet_time_ms(last_packet_time_ms),
-          inter_arrival(90 * kTimestampGroupLengthMs, kTimestampToMs,
-                        enable_burst_grouping),
-          estimator(options),
-          detector(options) {}
-    int64_t last_packet_time_ms;
-    InterArrival inter_arrival;
-    OveruseEstimator estimator;
-    OveruseDetector detector;
-  };
-
-  typedef std::map<unsigned int, Detector*> SsrcOveruseEstimatorMap;
-
-  // Triggers a new estimate calculation.
-  void UpdateEstimate(int64_t time_now)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
-
-  void GetSsrcs(std::vector<unsigned int>* ssrcs) const
-      SHARED_LOCKS_REQUIRED(crit_sect_.get());
-
-  Clock* clock_;
-  SsrcOveruseEstimatorMap overuse_detectors_ GUARDED_BY(crit_sect_.get());
-  RateStatistics incoming_bitrate_ GUARDED_BY(crit_sect_.get());
-  rtc::scoped_ptr<RemoteRateControl> remote_rate_ GUARDED_BY(crit_sect_.get());
-  RemoteBitrateObserver* observer_ GUARDED_BY(crit_sect_.get());
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  int64_t last_process_time_;
-  int64_t process_interval_ms_ GUARDED_BY(crit_sect_.get());
-
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorImpl);
+struct RemoteBitrateEstimatorSingleStream::Detector {
+  explicit Detector(int64_t last_packet_time_ms,
+                    const OverUseDetectorOptions& options,
+                    bool enable_burst_grouping)
+      : last_packet_time_ms(last_packet_time_ms),
+        inter_arrival(90 * kTimestampGroupLengthMs,
+                      kTimestampToMs,
+                      enable_burst_grouping),
+        estimator(options),
+        detector(options) {}
+  int64_t last_packet_time_ms;
+  InterArrival inter_arrival;
+  OveruseEstimator estimator;
+  OveruseDetector detector;
 };
 
-RemoteBitrateEstimatorImpl::RemoteBitrateEstimatorImpl(
-    RemoteBitrateObserver* observer,
-    Clock* clock,
-    RateControlType control_type,
-    uint32_t min_bitrate_bps)
-    : clock_(clock),
-      incoming_bitrate_(1000, 8000),
-      remote_rate_(RemoteRateControl::Create(control_type, min_bitrate_bps)),
-      observer_(observer),
-      crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      last_process_time_(-1),
-      process_interval_ms_(kProcessIntervalMs) {
+  RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream(
+      RemoteBitrateObserver* observer,
+      Clock* clock)
+      : clock_(clock),
+        incoming_bitrate_(kBitrateWindowMs, 8000),
+        remote_rate_(new AimdRateControl()),
+        observer_(observer),
+        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+        last_process_time_(-1),
+        process_interval_ms_(kProcessIntervalMs) {
   assert(observer_);
+  LOG(LS_INFO) << "RemoteBitrateEstimatorSingleStream: Instantiating.";
 }
 
-RemoteBitrateEstimatorImpl::~RemoteBitrateEstimatorImpl() {
+RemoteBitrateEstimatorSingleStream::~RemoteBitrateEstimatorSingleStream() {
   while (!overuse_detectors_.empty()) {
     SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.begin();
     delete it->second;
@@ -107,10 +67,10 @@ RemoteBitrateEstimatorImpl::~RemoteBitrateEstimatorImpl() {
   }
 }
 
-void RemoteBitrateEstimatorImpl::IncomingPacket(
-    int64_t arrival_time_ms,
-    size_t payload_size,
-    const RTPHeader& header) {
+void RemoteBitrateEstimatorSingleStream::IncomingPacket(int64_t arrival_time_ms,
+                                                        size_t payload_size,
+                                                        const RTPHeader& header,
+                                                        bool was_paced) {
   uint32_t ssrc = header.ssrc;
   uint32_t rtp_timestamp = header.timestamp +
       header.extension.transmissionTimeOffset;
@@ -125,10 +85,8 @@ void RemoteBitrateEstimatorImpl::IncomingPacket(
     // automatically cleaned up when we have one RemoteBitrateEstimator per REMB
     // group.
     std::pair<SsrcOveruseEstimatorMap::iterator, bool> insert_result =
-        overuse_detectors_.insert(std::make_pair(ssrc, new Detector(
-            now_ms,
-            OverUseDetectorOptions(),
-            remote_rate_->GetControlType() == kAimdControl)));
+        overuse_detectors_.insert(std::make_pair(
+            ssrc, new Detector(now_ms, OverUseDetectorOptions(), true)));
     it = insert_result.first;
   }
   Detector* estimator = it->second;
@@ -146,12 +104,12 @@ void RemoteBitrateEstimatorImpl::IncomingPacket(
                                 estimator->detector.State());
     estimator->detector.Detect(estimator->estimator.offset(),
                                timestamp_delta_ms,
-                               estimator->estimator.num_of_deltas());
+                               estimator->estimator.num_of_deltas(), now_ms);
   }
   if (estimator->detector.State() == kBwOverusing) {
-    uint32_t incoming_bitrate = incoming_bitrate_.Rate(now_ms);
+    uint32_t incoming_bitrate_bps = incoming_bitrate_.Rate(now_ms);
     if (prior_state != kBwOverusing ||
-        remote_rate_->TimeToReduceFurther(now_ms, incoming_bitrate)) {
+        remote_rate_->TimeToReduceFurther(now_ms, incoming_bitrate_bps)) {
       // The first overuse should immediately trigger a new estimate.
       // We also have to update the estimate immediately if we are overusing
       // and the target bitrate is too high compared to what we are receiving.
@@ -160,7 +118,7 @@ void RemoteBitrateEstimatorImpl::IncomingPacket(
   }
 }
 
-int32_t RemoteBitrateEstimatorImpl::Process() {
+int32_t RemoteBitrateEstimatorSingleStream::Process() {
   if (TimeUntilNextProcess() > 0) {
     return 0;
   }
@@ -172,7 +130,7 @@ int32_t RemoteBitrateEstimatorImpl::Process() {
   return 0;
 }
 
-int64_t RemoteBitrateEstimatorImpl::TimeUntilNextProcess() {
+int64_t RemoteBitrateEstimatorSingleStream::TimeUntilNextProcess() {
   if (last_process_time_ < 0) {
     return 0;
   }
@@ -183,7 +141,7 @@ int64_t RemoteBitrateEstimatorImpl::TimeUntilNextProcess() {
   }
 }
 
-void RemoteBitrateEstimatorImpl::UpdateEstimate(int64_t now_ms) {
+void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
   BandwidthUsage bw_state = kBwNormal;
   double sum_var_noise = 0.0;
   SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.begin();
@@ -208,8 +166,7 @@ void RemoteBitrateEstimatorImpl::UpdateEstimate(int64_t now_ms) {
   }
   // We can't update the estimate if we don't have any active streams.
   if (overuse_detectors_.empty()) {
-    remote_rate_.reset(RemoteRateControl::Create(
-        remote_rate_->GetControlType(), remote_rate_->GetMinBitrate()));
+    remote_rate_.reset(new AimdRateControl());
     return;
   }
   double mean_noise_var = sum_var_noise /
@@ -217,7 +174,7 @@ void RemoteBitrateEstimatorImpl::UpdateEstimate(int64_t now_ms) {
   const RateControlInput input(bw_state,
                                incoming_bitrate_.Rate(now_ms),
                                mean_noise_var);
-  const RateControlRegion region = remote_rate_->Update(&input, now_ms);
+  remote_rate_->Update(&input, now_ms);
   unsigned int target_bitrate = remote_rate_->UpdateBandwidthEstimate(now_ms);
   if (remote_rate_->ValidEstimate()) {
     process_interval_ms_ = remote_rate_->GetFeedbackInterval();
@@ -225,17 +182,15 @@ void RemoteBitrateEstimatorImpl::UpdateEstimate(int64_t now_ms) {
     GetSsrcs(&ssrcs);
     observer_->OnReceiveBitrateChanged(ssrcs, target_bitrate);
   }
-  for (it = overuse_detectors_.begin(); it != overuse_detectors_.end(); ++it) {
-    it->second->detector.SetRateControlRegion(region);
-  }
 }
 
-void RemoteBitrateEstimatorImpl::OnRttUpdate(int64_t rtt) {
+void RemoteBitrateEstimatorSingleStream::OnRttUpdate(int64_t avg_rtt_ms,
+                                                     int64_t max_rtt_ms) {
   CriticalSectionScoped cs(crit_sect_.get());
-  remote_rate_->SetRtt(rtt);
+  remote_rate_->SetRtt(avg_rtt_ms);
 }
 
-void RemoteBitrateEstimatorImpl::RemoveStream(unsigned int ssrc) {
+void RemoteBitrateEstimatorSingleStream::RemoveStream(unsigned int ssrc) {
   CriticalSectionScoped cs(crit_sect_.get());
   SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc);
   if (it != overuse_detectors_.end()) {
@@ -244,7 +199,7 @@ void RemoteBitrateEstimatorImpl::RemoveStream(unsigned int ssrc) {
   }
 }
 
-bool RemoteBitrateEstimatorImpl::LatestEstimate(
+bool RemoteBitrateEstimatorSingleStream::LatestEstimate(
     std::vector<unsigned int>* ssrcs,
     unsigned int* bitrate_bps) const {
   CriticalSectionScoped cs(crit_sect_.get());
@@ -260,13 +215,13 @@ bool RemoteBitrateEstimatorImpl::LatestEstimate(
   return true;
 }
 
-bool RemoteBitrateEstimatorImpl::GetStats(
+bool RemoteBitrateEstimatorSingleStream::GetStats(
     ReceiveBandwidthEstimatorStats* output) const {
   // Not implemented.
   return false;
 }
 
-void RemoteBitrateEstimatorImpl::GetSsrcs(
+void RemoteBitrateEstimatorSingleStream::GetSsrcs(
     std::vector<unsigned int>* ssrcs) const {
   assert(ssrcs);
   ssrcs->resize(overuse_detectors_.size());
@@ -277,13 +232,9 @@ void RemoteBitrateEstimatorImpl::GetSsrcs(
   }
 }
 
-RemoteBitrateEstimator* RemoteBitrateEstimatorFactory::Create(
-    webrtc::RemoteBitrateObserver* observer,
-    webrtc::Clock* clock,
-    RateControlType control_type,
-    uint32_t min_bitrate_bps) const {
-  LOG(LS_INFO) << "RemoteBitrateEstimatorFactory: Instantiating.";
-  return new RemoteBitrateEstimatorImpl(observer, clock, control_type,
-                                        min_bitrate_bps);
+void RemoteBitrateEstimatorSingleStream::SetMinBitrate(int min_bitrate_bps) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  remote_rate_->SetMinBitrate(min_bitrate_bps);
 }
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
new file mode 100644
index 0000000000..35fe7216a5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
+
+#include <map>
+#include <vector>
+
+#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
+ public:
+  RemoteBitrateEstimatorSingleStream(RemoteBitrateObserver* observer,
+                                     Clock* clock);
+  virtual ~RemoteBitrateEstimatorSingleStream();
+
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header,
+                      bool was_paced) override;
+  int32_t Process() override;
+  int64_t TimeUntilNextProcess() override;
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+  void RemoveStream(unsigned int ssrc) override;
+  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
+                      unsigned int* bitrate_bps) const override;
+  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override;
+  void SetMinBitrate(int min_bitrate_bps) override;
+
+ private:
+  struct Detector;
+
+  typedef std::map<unsigned int, Detector*> SsrcOveruseEstimatorMap;
+
+  // Triggers a new estimate calculation.
+  void UpdateEstimate(int64_t time_now)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get());
+
+  void GetSsrcs(std::vector<unsigned int>* ssrcs) const
+      SHARED_LOCKS_REQUIRED(crit_sect_.get());
+
+  Clock* clock_;
+  SsrcOveruseEstimatorMap overuse_detectors_ GUARDED_BY(crit_sect_.get());
+  RateStatistics incoming_bitrate_ GUARDED_BY(crit_sect_.get());
+  rtc::scoped_ptr<AimdRateControl> remote_rate_ GUARDED_BY(crit_sect_.get());
+  RemoteBitrateObserver* observer_ GUARDED_BY(crit_sect_.get());
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  int64_t last_process_time_;
+  int64_t process_interval_ms_ GUARDED_BY(crit_sect_.get());
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorSingleStream);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
index 2ce7a8b75d..7a26a7e63b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
 
 namespace webrtc {
@@ -16,18 +17,13 @@ namespace webrtc {
 class RemoteBitrateEstimatorSingleTest :
     public RemoteBitrateEstimatorTest {
  public:
-  static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
-
   RemoteBitrateEstimatorSingleTest() {}
   virtual void SetUp() {
-    bitrate_estimator_.reset(RemoteBitrateEstimatorFactory().Create(
-        bitrate_observer_.get(),
-        &clock_,
-        kMimdControl,
-        kRemoteBitrateEstimatorMinBitrateBps));
+    bitrate_estimator_.reset(new RemoteBitrateEstimatorSingleStream(
+        bitrate_observer_.get(), &clock_));
   }
  protected:
-  DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorSingleTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorSingleTest);
 };
 
 TEST_F(RemoteBitrateEstimatorSingleTest, InitialBehavior) {
@@ -39,15 +35,15 @@ TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseReordering) {
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseRtpTimestamps) {
-  RateIncreaseRtpTimestampsTestHelper(1621);
+  RateIncreaseRtpTimestampsTestHelper(1240);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStream) {
-  CapacityDropTestHelper(1, false, 733);
+  CapacityDropTestHelper(1, false, 600);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStreamWrap) {
-  CapacityDropTestHelper(1, true, 733);
+  CapacityDropTestHelper(1, true, 600);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropTwoStreamsWrap) {
@@ -55,19 +51,19 @@ TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropTwoStreamsWrap) {
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThreeStreamsWrap) {
-  CapacityDropTestHelper(3, true, 733);
+  CapacityDropTestHelper(3, true, 734);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThirteenStreamsWrap) {
-  CapacityDropTestHelper(13, true, 733);
+  CapacityDropTestHelper(13, true, 700);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropNineteenStreamsWrap) {
-  CapacityDropTestHelper(19, true, 733);
+  CapacityDropTestHelper(19, true, 700);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThirtyStreamsWrap) {
-  CapacityDropTestHelper(30, true, 733);
+  CapacityDropTestHelper(30, true, 700);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, TestTimestampGrouping) {
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
index 4aa6b52d00..8b9c0b9a1d 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
@@ -10,6 +10,7 @@
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
 
 #include <algorithm>
+#include <limits>
 #include <utility>
 
 namespace webrtc {
@@ -221,7 +222,8 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
                                                 size_t payload_size,
                                                 int64_t arrival_time,
                                                 uint32_t rtp_timestamp,
-                                                uint32_t absolute_send_time) {
+                                                uint32_t absolute_send_time,
+                                                bool was_paced) {
   RTPHeader header;
   memset(&header, 0, sizeof(header));
   header.ssrc = ssrc;
@@ -229,7 +231,7 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
   header.extension.hasAbsoluteSendTime = true;
   header.extension.absoluteSendTime = absolute_send_time;
   bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs,
-      payload_size, header);
+                                     payload_size, header, was_paced);
 }
 
 // Generates a frame of packets belonging to a stream at a given bitrate and
@@ -252,11 +254,9 @@ bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(unsigned int ssrc,
     // since both are used in IncomingPacket().
     clock_.AdvanceTimeMicroseconds(packet->arrival_time -
                                    clock_.TimeInMicroseconds());
-    IncomingPacket(packet->ssrc,
-                   packet->size,
-                   (packet->arrival_time + 500) / 1000,
-                   packet->rtp_timestamp,
-                   AbsSendTime(packet->send_time, 1000000));
+    IncomingPacket(packet->ssrc, packet->size,
+                   (packet->arrival_time + 500) / 1000, packet->rtp_timestamp,
+                   AbsSendTime(packet->send_time, 1000000), true);
     if (bitrate_observer_->updated()) {
       // Verify that new estimates only are triggered by an overuse and a
       // rate decrease.
@@ -321,18 +321,18 @@ void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper(
   EXPECT_FALSE(bitrate_observer_->updated());
   bitrate_observer_->Reset();
   clock_.AdvanceTimeMilliseconds(1000);
-  // Inserting a packet. Still no valid estimate. We need to wait 1 second.
+  // Inserting a packet. Still no valid estimate. We need to wait 5 seconds.
   IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
-                 absolute_send_time);
+                 absolute_send_time, true);
   bitrate_estimator_->Process();
   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
   EXPECT_EQ(0u, ssrcs.size());
   EXPECT_FALSE(bitrate_observer_->updated());
   bitrate_observer_->Reset();
-  // Inserting packets for one second to get a valid estimate.
-  for (int i = 0; i < kFramerate; ++i) {
+  // Inserting packets for 5 seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1; ++i) {
     IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
     timestamp += 90 * kFrameIntervalMs;
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -360,13 +360,13 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
   uint32_t timestamp = 0;
   uint32_t absolute_send_time = 0;
   IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                 absolute_send_time);
+                 absolute_send_time, true);
   bitrate_estimator_->Process();
   EXPECT_FALSE(bitrate_observer_->updated());  // No valid estimate.
   // Inserting packets for one second to get a valid estimate.
-  for (int i = 0; i < kFramerate; ++i) {
+  for (int i = 0; i < 5 * kFramerate + 1; ++i) {
     IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     timestamp += 90 * kFrameIntervalMs;
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -383,11 +383,12 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
     absolute_send_time = AddAbsSendTime(absolute_send_time,
                                         2 * kFrameIntervalAbsSendTime);
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
                    timestamp - 90 * kFrameIntervalMs,
                    AddAbsSendTime(absolute_send_time,
-                                  -int(kFrameIntervalAbsSendTime)));
+                                  -static_cast<int>(kFrameIntervalAbsSendTime)),
+                   true);
   }
   bitrate_estimator_->Process();
   EXPECT_TRUE(bitrate_observer_->updated());
@@ -439,7 +440,7 @@ void RemoteBitrateEstimatorTest::CapacityDropTestHelper(
     steady_state_time = 10;
     AddDefaultStream();
   } else {
-    steady_state_time = 8 * number_of_streams;
+    steady_state_time = 10 * number_of_streams;
     int bitrate_sum = 0;
     int kBitrateDenom = number_of_streams * (number_of_streams - 1);
     for (int i = 0; i < number_of_streams; i++) {
@@ -473,7 +474,7 @@ void RemoteBitrateEstimatorTest::CapacityDropTestHelper(
                                             kMinExpectedBitrate,
                                             kMaxExpectedBitrate,
                                             kInitialCapacityBps);
-  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 100000u);
+  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 110000u);
   bitrate_observer_->Reset();
 
   // Reduce the capacity and verify the decrease time.
@@ -493,8 +494,8 @@ void RemoteBitrateEstimatorTest::CapacityDropTestHelper(
     }
   }
 
-  EXPECT_EQ(expected_bitrate_drop_delta,
-            bitrate_drop_time - overuse_start_time);
+  EXPECT_NEAR(expected_bitrate_drop_delta,
+              bitrate_drop_time - overuse_start_time, 33);
 
   // Remove stream one by one.
   unsigned int latest_bps = 0;
@@ -520,12 +521,13 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
   uint32_t timestamp = 0;
   // Initialize absolute_send_time (24 bits) so that it will definitely wrap
   // during the test.
-  uint32_t absolute_send_time =
-      AddAbsSendTime((1 << 24), -int(50 * kFrameIntervalAbsSendTime));
-  // Initial set of frames to increase the bitrate.
-  for (int i = 0; i <= 100; ++i) {
+  uint32_t absolute_send_time = AddAbsSendTime(
+      (1 << 24), -static_cast<int>(50 * kFrameIntervalAbsSendTime));
+  // Initial set of frames to increase the bitrate. 6 seconds to have enough
+  // time for the first estimate to be generated and for Process() to be called.
+  for (int i = 0; i <= 6 * kFramerate; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     bitrate_estimator_->Process();
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     timestamp += 90 * kFrameIntervalMs;
@@ -533,7 +535,7 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
                                         kFrameIntervalAbsSendTime);
   }
   EXPECT_TRUE(bitrate_observer_->updated());
-  EXPECT_NEAR(450000u, bitrate_observer_->latest_bitrate(), 20000u);
+  EXPECT_GE(bitrate_observer_->latest_bitrate(), 400000u);
 
   // Insert batches of frames which were sent very close in time. Also simulate
   // capacity over-use to see that we back off correctly.
@@ -546,7 +548,7 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
       // Insert |kTimestampGroupLength| frames with just 1 timestamp ticks in
       // between. Should be treated as part of the same group by the estimator.
       IncomingPacket(kDefaultSsrc, 100, clock_.TimeInMilliseconds(), timestamp,
-                     absolute_send_time);
+                     absolute_send_time, true);
       clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
       timestamp += 1;
       absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -555,13 +557,15 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
     // Increase time until next batch to simulate over-use.
     clock_.AdvanceTimeMilliseconds(10);
     timestamp += 90 * kFrameIntervalMs - kTimestampGroupLength;
-    absolute_send_time = AddAbsSendTime(absolute_send_time, AddAbsSendTime(
-        kFrameIntervalAbsSendTime, -int(kTimestampGroupLengthAbsSendTime)));
+    absolute_send_time = AddAbsSendTime(
+        absolute_send_time,
+        AddAbsSendTime(kFrameIntervalAbsSendTime,
+                       -static_cast<int>(kTimestampGroupLengthAbsSendTime)));
     bitrate_estimator_->Process();
   }
   EXPECT_TRUE(bitrate_observer_->updated());
   // Should have reduced the estimate.
-  EXPECT_EQ(378720u, bitrate_observer_->latest_bitrate());
+  EXPECT_LT(bitrate_observer_->latest_bitrate(), 400000u);
 }
 
 void RemoteBitrateEstimatorTest::TestGetStatsHelper() {
@@ -579,7 +583,7 @@ void RemoteBitrateEstimatorTest::TestGetStatsHelper() {
   // Inject propagation_time_delta of kFrameIntervalMs.
   for (size_t i = 0; i < 3; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     timestamp += kFrameIntervalMs;
     // Insert a kFrameIntervalMs propagation_time_delta.
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs * 2);
@@ -597,7 +601,7 @@ void RemoteBitrateEstimatorTest::TestGetStatsHelper() {
   // should be adjusted to 0.
   for (size_t i = 0; i < 3; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     timestamp += 10 * kFrameIntervalMs;
     clock_.AdvanceTimeMilliseconds(kBurstThresholdMs + 1);
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -610,7 +614,7 @@ void RemoteBitrateEstimatorTest::TestGetStatsHelper() {
   // limits.
   for (size_t i = 0; i < 1001; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     timestamp += kFrameIntervalMs;
     absolute_send_time = AddAbsSendTime(absolute_send_time,
                                         kFrameIntervalAbsSendTime);
@@ -635,7 +639,7 @@ void RemoteBitrateEstimatorTest::TestWrappingHelper(
 
   for (size_t i = 0; i < 3000; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     timestamp += kFrameIntervalMs;
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -652,7 +656,7 @@ void RemoteBitrateEstimatorTest::TestWrappingHelper(
   bitrate_estimator_->Process();
   for (size_t i = 0; i < 100; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time);
+                   absolute_send_time, true);
     timestamp += kFrameIntervalMs;
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     absolute_send_time = AddAbsSendTime(absolute_send_time,
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
index 8da880f6c2..8343d7d57b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
@@ -14,12 +14,13 @@
 #include <list>
 #include <map>
 #include <utility>
+#include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace testing {
@@ -100,7 +101,7 @@ class RtpStream {
   uint32_t rtp_timestamp_offset_;
   const double kNtpFracPerMs;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpStream);
 };
 
 class StreamGenerator {
@@ -138,7 +139,7 @@ class StreamGenerator {
   // All streams being transmitted on this simulated channel.
   StreamMap streams_;
 
-  DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
 };
 }  // namespace testing
 
@@ -168,7 +169,8 @@ class RemoteBitrateEstimatorTest : public ::testing::Test {
                       size_t payload_size,
                       int64_t arrival_time,
                       uint32_t rtp_timestamp,
-                      uint32_t absolute_send_time);
+                      uint32_t absolute_send_time,
+                      bool was_paced);
 
   // Generates a frame of packets belonging to a stream at a given bitrate and
   // with a given ssrc. The stream is pushed through a very simple simulated
@@ -189,7 +191,6 @@ class RemoteBitrateEstimatorTest : public ::testing::Test {
                               unsigned int max_bitrate,
                               unsigned int target_bitrate);
 
-
   void TestTimestampGroupingTestHelper();
 
   void TestGetStatsHelper();
@@ -211,7 +212,7 @@ class RemoteBitrateEstimatorTest : public ::testing::Test {
   rtc::scoped_ptr<RemoteBitrateEstimator> bitrate_estimator_;
   rtc::scoped_ptr<testing::StreamGenerator> stream_generator_;
 
-  DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorTest);
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
index d1b9ea79f7..2ce144129b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
@@ -8,14 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef WEBRTC_WIN
+#include <sys/types.h>
+#include <unistd.h>
+#endif
+
+#include <algorithm>
 #include <sstream>
 
+#include "webrtc/base/random.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/perf_test.h"
 
 using std::string;
 
@@ -36,14 +42,14 @@ INSTANTIATE_TEST_CASE_P(VideoSendersTest,
 
 TEST_P(DefaultBweTest, UnlimitedSpeed) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   RunFor(10 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, SteadyLoss) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   LossFilter loss(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   loss.SetLoss(20.0);
@@ -52,7 +58,7 @@ TEST_P(DefaultBweTest, SteadyLoss) {
 
 TEST_P(DefaultBweTest, IncreasingLoss1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   LossFilter loss(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   for (int i = 0; i < 76; ++i) {
@@ -63,21 +69,21 @@ TEST_P(DefaultBweTest, IncreasingLoss1) {
 
 TEST_P(DefaultBweTest, SteadyDelay) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   DelayFilter delay(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
-  delay.SetDelay(1000);
+  delay.SetOneWayDelayMs(1000);
   RunFor(10 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, IncreasingDelay1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   DelayFilter delay(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   RunFor(10 * 60 * 1000);
   for (int i = 0; i < 30 * 2; ++i) {
-    delay.SetDelay(i);
+    delay.SetOneWayDelayMs(i);
     RunFor(10 * 1000);
   }
   RunFor(10 * 60 * 1000);
@@ -85,52 +91,52 @@ TEST_P(DefaultBweTest, IncreasingDelay1) {
 
 TEST_P(DefaultBweTest, IncreasingDelay2) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   DelayFilter delay(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "");
+  RateCounterFilter counter(&uplink_, 0, "", "");
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   RunFor(1 * 60 * 1000);
   for (int i = 1; i < 51; ++i) {
-    delay.SetDelay(10.0f * i);
+    delay.SetOneWayDelayMs(10.0f * i);
     RunFor(10 * 1000);
   }
-  delay.SetDelay(0.0f);
+  delay.SetOneWayDelayMs(0.0f);
   RunFor(10 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, JumpyDelay1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   DelayFilter delay(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   RunFor(10 * 60 * 1000);
   for (int i = 1; i < 200; ++i) {
-    delay.SetDelay((10 * i) % 500);
+    delay.SetOneWayDelayMs((10 * i) % 500);
     RunFor(1000);
-    delay.SetDelay(1.0f);
+    delay.SetOneWayDelayMs(1.0f);
     RunFor(1000);
   }
-  delay.SetDelay(0.0f);
+  delay.SetOneWayDelayMs(0.0f);
   RunFor(10 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, SteadyJitter) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   JitterFilter jitter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "");
+  RateCounterFilter counter(&uplink_, 0, "", "");
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
-  jitter.SetJitter(20);
+  jitter.SetMaxJitter(20);
   RunFor(2 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, IncreasingJitter1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   JitterFilter jitter(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   for (int i = 0; i < 2 * 60 * 2; ++i) {
-    jitter.SetJitter(i);
+    jitter.SetMaxJitter(i);
     RunFor(10 * 1000);
   }
   RunFor(10 * 60 * 1000);
@@ -138,21 +144,21 @@ TEST_P(DefaultBweTest, IncreasingJitter1) {
 
 TEST_P(DefaultBweTest, IncreasingJitter2) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   JitterFilter jitter(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   RunFor(30 * 1000);
   for (int i = 1; i < 51; ++i) {
-    jitter.SetJitter(10.0f * i);
+    jitter.SetMaxJitter(10.0f * i);
     RunFor(10 * 1000);
   }
-  jitter.SetJitter(0.0f);
+  jitter.SetMaxJitter(0.0f);
   RunFor(10 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, SteadyReorder) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ReorderFilter reorder(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   reorder.SetReorder(20.0);
@@ -161,7 +167,7 @@ TEST_P(DefaultBweTest, SteadyReorder) {
 
 TEST_P(DefaultBweTest, IncreasingReorder1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ReorderFilter reorder(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   for (int i = 0; i < 76; ++i) {
@@ -172,63 +178,63 @@ TEST_P(DefaultBweTest, IncreasingReorder1) {
 
 TEST_P(DefaultBweTest, SteadyChoke) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
-  choke.SetCapacity(140);
+  choke.set_capacity_kbps(140);
   RunFor(10 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, IncreasingChoke1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   for (int i = 1200; i >= 100; i -= 100) {
-    choke.SetCapacity(i);
+    choke.set_capacity_kbps(i);
     RunFor(5000);
   }
 }
 
 TEST_P(DefaultBweTest, IncreasingChoke2) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   RunFor(60 * 1000);
   for (int i = 1200; i >= 100; i -= 20) {
-    choke.SetCapacity(i);
+    choke.set_capacity_kbps(i);
     RunFor(1000);
   }
 }
 
 TEST_P(DefaultBweTest, Multi1) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   DelayFilter delay(&uplink_, 0);
   ChokeFilter choke(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "");
+  RateCounterFilter counter(&uplink_, 0, "", "");
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
-  choke.SetCapacity(1000);
+  choke.set_capacity_kbps(1000);
   RunFor(1 * 60 * 1000);
   for (int i = 1; i < 51; ++i) {
-    delay.SetDelay(100.0f * i);
+    delay.SetOneWayDelayMs(100.0f * i);
     RunFor(10 * 1000);
   }
   RunFor(500 * 1000);
-  delay.SetDelay(0.0f);
+  delay.SetOneWayDelayMs(0.0f);
   RunFor(5 * 60 * 1000);
 }
 
 TEST_P(DefaultBweTest, Multi2) {
   VideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
   JitterFilter jitter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "");
+  RateCounterFilter counter(&uplink_, 0, "", "");
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
-  choke.SetCapacity(2000);
-  jitter.SetJitter(120);
+  choke.set_capacity_kbps(2000);
+  jitter.SetMaxJitter(120);
   RunFor(5 * 60 * 1000);
 }
 
@@ -238,39 +244,23 @@ class BweFeedbackTest
     : public BweTest,
       public ::testing::TestWithParam<BandwidthEstimatorType> {
  public:
-  BweFeedbackTest() : BweTest() {}
+#ifdef WEBRTC_WIN
+  BweFeedbackTest()
+      : BweTest(), random_(Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
+#else
+  BweFeedbackTest()
+      : BweTest(),
+        // Multiply the time by a random-ish odd number derived from the PID.
+        random_((getpid() | 1) *
+                Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
+#endif
   virtual ~BweFeedbackTest() {}
 
-  void PrintResults(double max_throughput_kbps, Stats<double> throughput_kbps,
-                    Stats<double> delay_ms) {
-    double utilization = throughput_kbps.GetMean() / max_throughput_kbps;
-    webrtc::test::PrintResult("BwePerformance",
-                              GetTestName(),
-                              "Utilization",
-                              utilization * 100.0,
-                              "%",
-                              false);
-    std::stringstream ss;
-    ss << throughput_kbps.GetStdDev() / throughput_kbps.GetMean();
-    webrtc::test::PrintResult("BwePerformance",
-                              GetTestName(),
-                              "Utilization var coeff",
-                              ss.str(),
-                              "",
-                              false);
-    webrtc::test::PrintResult("BwePerformance",
-                              GetTestName(),
-                              "Average delay",
-                              delay_ms.AsString(),
-                              "ms",
-                              false);
-  }
-
  protected:
-  void SetUp() override { BweTest::SetUp(); }
+  Random random_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(BweFeedbackTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweFeedbackTest);
 };
 
 INSTANTIATE_TEST_CASE_P(VideoSendersTest,
@@ -278,71 +268,170 @@ INSTANTIATE_TEST_CASE_P(VideoSendersTest,
                         ::testing::Values(kRembEstimator,
                                           kFullSendSideEstimator));
 
+TEST_P(BweFeedbackTest, ConstantCapacity) {
+  AdaptiveVideoSource source(0, 30, 300, 0, 0);
+  PacedVideoSender sender(&uplink_, &source, GetParam());
+  ChokeFilter filter(&uplink_, 0);
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
+  PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
+  const int kCapacityKbps = 1000;
+  filter.set_capacity_kbps(kCapacityKbps);
+  filter.set_max_delay_ms(500);
+  RunFor(180 * 1000);
+  PrintResults(kCapacityKbps, counter.GetBitrateStats(), 0,
+               receiver.GetDelayStats(), counter.GetBitrateStats());
+}
+
 TEST_P(BweFeedbackTest, Choke1000kbps500kbps1000kbps) {
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  PacedVideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter filter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   const int kHighCapacityKbps = 1000;
   const int kLowCapacityKbps = 500;
-  filter.SetCapacity(kHighCapacityKbps);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(kHighCapacityKbps);
+  filter.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  filter.SetCapacity(kLowCapacityKbps);
+  filter.set_capacity_kbps(kLowCapacityKbps);
   RunFor(60 * 1000);
-  filter.SetCapacity(kHighCapacityKbps);
+  filter.set_capacity_kbps(kHighCapacityKbps);
   RunFor(60 * 1000);
   PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0,
-               counter.GetBitrateStats(), filter.GetDelayStats());
+               counter.GetBitrateStats(), 0, receiver.GetDelayStats(),
+               counter.GetBitrateStats());
 }
 
 TEST_P(BweFeedbackTest, Choke200kbps30kbps200kbps) {
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
+  PacedVideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter filter(&uplink_, 0);
-  RateCounterFilter counter(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   const int kHighCapacityKbps = 200;
   const int kLowCapacityKbps = 30;
-  filter.SetCapacity(kHighCapacityKbps);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(kHighCapacityKbps);
+  filter.set_max_delay_ms(500);
   RunFor(60 * 1000);
-  filter.SetCapacity(kLowCapacityKbps);
+  filter.set_capacity_kbps(kLowCapacityKbps);
   RunFor(60 * 1000);
-  filter.SetCapacity(kHighCapacityKbps);
+  filter.set_capacity_kbps(kHighCapacityKbps);
   RunFor(60 * 1000);
 
   PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0,
-               counter.GetBitrateStats(), filter.GetDelayStats());
+               counter.GetBitrateStats(), 0, receiver.GetDelayStats(),
+               counter.GetBitrateStats());
 }
 
 TEST_P(BweFeedbackTest, Verizon4gDownlinkTest) {
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
-  RateCounterFilter counter1(&uplink_, 0, "sender_output");
+  VideoSender sender(&uplink_, &source, GetParam());
+  RateCounterFilter counter1(&uplink_, 0, "sender_output",
+                             bwe_names[GetParam()]);
   TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");
-  RateCounterFilter counter2(&uplink_, 0, "receiver_input");
+  RateCounterFilter counter2(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   ASSERT_TRUE(filter.Init(test::ResourcePath("verizon4g-downlink", "rx")));
   RunFor(22 * 60 * 1000);
   PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(),
-               filter.GetDelayStats());
+               0, receiver.GetDelayStats(), counter2.GetBitrateStats());
 }
 
 // webrtc:3277
-TEST_P(BweFeedbackTest, DISABLED_GoogleWifiTrace3Mbps) {
+TEST_P(BweFeedbackTest, GoogleWifiTrace3Mbps) {
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
-  PacketSender sender(&uplink_, &source, GetParam());
-  RateCounterFilter counter1(&uplink_, 0, "sender_output");
+  VideoSender sender(&uplink_, &source, GetParam());
+  RateCounterFilter counter1(&uplink_, 0, "sender_output",
+                             bwe_names[GetParam()]);
   TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");
-  filter.SetMaxDelay(500);
-  RateCounterFilter counter2(&uplink_, 0, "receiver_input");
+  filter.set_max_delay_ms(500);
+  RateCounterFilter counter2(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), false, false);
   ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx")));
   RunFor(300 * 1000);
   PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(),
-               filter.GetDelayStats());
+               0, receiver.GetDelayStats(), counter2.GetBitrateStats());
+}
+
+TEST_P(BweFeedbackTest, PacedSelfFairness50msTest) {
+  int64_t kRttMs = 100;
+  int64_t kMaxJitterMs = 15;
+
+  const int kNumRmcatFlows = 4;
+  int64_t offset_ms[kNumRmcatFlows];
+  for (int i = 0; i < kNumRmcatFlows; ++i) {
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
+  }
+
+  RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 50, kRttMs,
+                  kMaxJitterMs, offset_ms);
+}
+
+TEST_P(BweFeedbackTest, PacedSelfFairness500msTest) {
+  int64_t kRttMs = 100;
+  int64_t kMaxJitterMs = 15;
+
+  const int kNumRmcatFlows = 4;
+  int64_t offset_ms[kNumRmcatFlows];
+  for (int i = 0; i < kNumRmcatFlows; ++i) {
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
+  }
+
+  RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 500, kRttMs,
+                  kMaxJitterMs, offset_ms);
+}
+
+TEST_P(BweFeedbackTest, PacedSelfFairness1000msTest) {
+  int64_t kRttMs = 100;
+  int64_t kMaxJitterMs = 15;
+
+  const int kNumRmcatFlows = 4;
+  int64_t offset_ms[kNumRmcatFlows];
+  for (int i = 0; i < kNumRmcatFlows; ++i) {
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
+  }
+
+  RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 1000, kRttMs,
+                  kMaxJitterMs, offset_ms);
+}
+
+TEST_P(BweFeedbackTest, TcpFairness50msTest) {
+  int64_t kRttMs = 100;
+  int64_t kMaxJitterMs = 15;
+
+  int64_t offset_ms[2];  // One TCP, one RMCAT flow.
+  for (int i = 0; i < 2; ++i) {
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
+  }
+
+  RunFairnessTest(GetParam(), 1, 1, 300, 2000, 50, kRttMs, kMaxJitterMs,
+                  offset_ms);
+}
+
+TEST_P(BweFeedbackTest, TcpFairness500msTest) {
+  int64_t kRttMs = 100;
+  int64_t kMaxJitterMs = 15;
+
+  int64_t offset_ms[2];  // One TCP, one RMCAT flow.
+  for (int i = 0; i < 2; ++i) {
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
+  }
+
+  RunFairnessTest(GetParam(), 1, 1, 300, 2000, 500, kRttMs, kMaxJitterMs,
+                  offset_ms);
+}
+
+TEST_P(BweFeedbackTest, TcpFairness1000msTest) {
+  int64_t kRttMs = 100;
+  int64_t kMaxJitterMs = 15;
+
+  int64_t offset_ms[2];  // One TCP, one RMCAT flow.
+  for (int i = 0; i < 2; ++i) {
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
+  }
+
+  RunFairnessTest(GetParam(), 1, 1, 300, 2000, 1000, kRttMs, kMaxJitterMs,
+                  offset_ms);
 }
 }  // namespace bwe
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
new file mode 100644
index 0000000000..15ca42dda9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -0,0 +1,164 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+
+namespace webrtc {
+
+// TODO(sprang): Tune these!
+const int RemoteEstimatorProxy::kDefaultProcessIntervalMs = 50;
+const int RemoteEstimatorProxy::kBackWindowMs = 500;
+
+RemoteEstimatorProxy::RemoteEstimatorProxy(Clock* clock,
+                                           PacketRouter* packet_router)
+    : clock_(clock),
+      packet_router_(packet_router),
+      last_process_time_ms_(-1),
+      media_ssrc_(0),
+      feedback_sequence_(0),
+      window_start_seq_(-1) {}
+
+RemoteEstimatorProxy::~RemoteEstimatorProxy() {}
+
+void RemoteEstimatorProxy::IncomingPacketFeedbackVector(
+    const std::vector<PacketInfo>& packet_feedback_vector) {
+  rtc::CritScope cs(&lock_);
+  for (PacketInfo info : packet_feedback_vector)
+    OnPacketArrival(info.sequence_number, info.arrival_time_ms);
+}
+
+void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms,
+                                          size_t payload_size,
+                                          const RTPHeader& header,
+                                          bool was_paced) {
+  if (!header.extension.hasTransportSequenceNumber) {
+    LOG(LS_WARNING) << "RemoteEstimatorProxy: Incoming packet "
+                       "is missing the transport sequence number extension!";
+    return;
+  }
+  rtc::CritScope cs(&lock_);
+  media_ssrc_ = header.ssrc;
+  OnPacketArrival(header.extension.transportSequenceNumber, arrival_time_ms);
+}
+
+void RemoteEstimatorProxy::RemoveStream(unsigned int ssrc) {}
+
+bool RemoteEstimatorProxy::LatestEstimate(std::vector<unsigned int>* ssrcs,
+                                          unsigned int* bitrate_bps) const {
+  return false;
+}
+
+bool RemoteEstimatorProxy::GetStats(
+    ReceiveBandwidthEstimatorStats* output) const {
+  return false;
+}
+
+
+int64_t RemoteEstimatorProxy::TimeUntilNextProcess() {
+  int64_t now = clock_->TimeInMilliseconds();
+  int64_t time_until_next = 0;
+  if (last_process_time_ms_ != -1 &&
+      now - last_process_time_ms_ < kDefaultProcessIntervalMs) {
+    time_until_next = (last_process_time_ms_ + kDefaultProcessIntervalMs - now);
+  }
+  return time_until_next;
+}
+
+int32_t RemoteEstimatorProxy::Process() {
+  // TODO(sprang): Perhaps we need a dedicated thread here instead?
+
+  if (TimeUntilNextProcess() > 0)
+    return 0;
+  last_process_time_ms_ = clock_->TimeInMilliseconds();
+
+  bool more_to_build = true;
+  while (more_to_build) {
+    rtcp::TransportFeedback feedback_packet;
+    if (BuildFeedbackPacket(&feedback_packet)) {
+      RTC_DCHECK(packet_router_ != nullptr);
+      packet_router_->SendFeedback(&feedback_packet);
+    } else {
+      more_to_build = false;
+    }
+  }
+
+  return 0;
+}
+
+void RemoteEstimatorProxy::OnPacketArrival(uint16_t sequence_number,
+                                           int64_t arrival_time) {
+  int64_t seq = unwrapper_.Unwrap(sequence_number);
+
+  if (window_start_seq_ == -1) {
+    window_start_seq_ = seq;
+    // Start new feedback packet, cull old packets.
+    for (auto it = packet_arrival_times_.begin();
+         it != packet_arrival_times_.end() && it->first < seq &&
+         arrival_time - it->second >= kBackWindowMs;) {
+      auto delete_it = it;
+      ++it;
+      packet_arrival_times_.erase(delete_it);
+    }
+  } else if (seq < window_start_seq_) {
+    window_start_seq_ = seq;
+  }
+
+  RTC_DCHECK(packet_arrival_times_.end() == packet_arrival_times_.find(seq));
+  packet_arrival_times_[seq] = arrival_time;
+}
+
+bool RemoteEstimatorProxy::BuildFeedbackPacket(
+    rtcp::TransportFeedback* feedback_packet) {
+  rtc::CritScope cs(&lock_);
+  if (window_start_seq_ == -1)
+    return false;
+
+  // window_start_seq_ is the first sequence number to include in the current
+  // feedback packet. Some older may still be in the map, in case a reordering
+  // happens and we need to retransmit them.
+  auto it = packet_arrival_times_.find(window_start_seq_);
+  RTC_DCHECK(it != packet_arrival_times_.end());
+
+  // TODO(sprang): Measure receive times in microseconds and remove the
+  // conversions below.
+  feedback_packet->WithMediaSourceSsrc(media_ssrc_);
+  feedback_packet->WithBase(static_cast<uint16_t>(it->first & 0xFFFF),
+                            it->second * 1000);
+  feedback_packet->WithFeedbackSequenceNumber(feedback_sequence_++);
+  for (; it != packet_arrival_times_.end(); ++it) {
+    if (!feedback_packet->WithReceivedPacket(
+            static_cast<uint16_t>(it->first & 0xFFFF), it->second * 1000)) {
+      // If we can't even add the first seq to the feedback packet, we won't be
+      // able to build it at all.
+      RTC_CHECK_NE(window_start_seq_, it->first);
+
+      // Could not add timestamp, feedback packet might be full. Return and
+      // try again with a fresh packet.
+      window_start_seq_ = it->first;
+      break;
+    }
+    // Note: Don't erase items from packet_arrival_times_ after sending, in case
+    // they need to be re-sent after a reordering. Removal will be handled
+    // by OnPacketArrival once packets are too old.
+  }
+  if (it == packet_arrival_times_.end())
+    window_start_seq_ = -1;
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
new file mode 100644
index 0000000000..98a68b3dcf
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
+
+#include <map>
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+
+namespace webrtc {
+
+class Clock;
+class PacketRouter;
+namespace rtcp {
+class TransportFeedback;
+}
+
+// Class used when send-side BWE is enabled: This proxy is instantiated on the
+// receive side. It buffers a number of receive timestamps and then sends
+// transport feedback messages back too the send side.
+
+class RemoteEstimatorProxy : public RemoteBitrateEstimator {
+ public:
+  RemoteEstimatorProxy(Clock* clock, PacketRouter* packet_router);
+  virtual ~RemoteEstimatorProxy();
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketInfo>& packet_feedback_vector) override;
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header,
+                      bool was_paced) override;
+  void RemoveStream(unsigned int ssrc) override;
+  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
+                      unsigned int* bitrate_bps) const override;
+  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override;
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override {}
+  void SetMinBitrate(int min_bitrate_bps) override {}
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
+
+  static const int kDefaultProcessIntervalMs;
+  static const int kBackWindowMs;
+
+ private:
+  void OnPacketArrival(uint16_t sequence_number, int64_t arrival_time)
+      EXCLUSIVE_LOCKS_REQUIRED(&lock_);
+  bool BuildFeedbackPacket(rtcp::TransportFeedback* feedback_packetket);
+
+  Clock* const clock_;
+  PacketRouter* const packet_router_;
+  int64_t last_process_time_ms_;
+
+  rtc::CriticalSection lock_;
+
+  uint32_t media_ssrc_ GUARDED_BY(&lock_);
+  uint8_t feedback_sequence_ GUARDED_BY(&lock_);
+  SequenceNumberUnwrapper unwrapper_ GUARDED_BY(&lock_);
+  int64_t window_start_seq_ GUARDED_BY(&lock_);
+  // Map unwrapped seq -> time.
+  std::map<int64_t, int64_t> packet_arrival_times_ GUARDED_BY(&lock_);
+};
+
+}  // namespace webrtc
+
+#endif  //  WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
new file mode 100644
index 0000000000..7ddd31467b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+using ::testing::_;
+using ::testing::InSequence;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+class MockPacketRouter : public PacketRouter {
+ public:
+  MOCK_METHOD1(SendFeedback, bool(rtcp::TransportFeedback* packet));
+};
+
+class RemoteEstimatorProxyTest : public ::testing::Test {
+ public:
+  RemoteEstimatorProxyTest() : clock_(0), proxy_(&clock_, &router_) {}
+
+ protected:
+  void IncomingPacket(uint16_t seq, int64_t time_ms) {
+    RTPHeader header;
+    header.extension.hasTransportSequenceNumber = true;
+    header.extension.transportSequenceNumber = seq;
+    header.ssrc = kMediaSsrc;
+    proxy_.IncomingPacket(time_ms, kDefaultPacketSize, header, true);
+  }
+
+  void Process() {
+    clock_.AdvanceTimeMilliseconds(
+        RemoteEstimatorProxy::kDefaultProcessIntervalMs);
+    proxy_.Process();
+  }
+
+  SimulatedClock clock_;
+  MockPacketRouter router_;
+  RemoteEstimatorProxy proxy_;
+
+  const size_t kDefaultPacketSize = 100;
+  const uint32_t kMediaSsrc = 456;
+  const uint16_t kBaseSeq = 10;
+  const int64_t kBaseTimeMs = 123;
+  const int64_t kMaxSmallDeltaMs =
+      (rtcp::TransportFeedback::kDeltaScaleFactor * 0xFF) / 1000;
+};
+
+TEST_F(RemoteEstimatorProxyTest, SendsSinglePacketFeedback) {
+  IncomingPacket(kBaseSeq, kBaseTimeMs);
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq, packet->GetBaseSequence());
+        EXPECT_EQ(kMediaSsrc, packet->GetMediaSourceSsrc());
+
+        std::vector<rtcp::TransportFeedback::StatusSymbol> status_vec =
+            packet->GetStatusVector();
+        EXPECT_EQ(1u, status_vec.size());
+        EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta,
+                  status_vec[0]);
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(1u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs, (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        return true;
+      }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, SendsFeedbackWithVaryingDeltas) {
+  IncomingPacket(kBaseSeq, kBaseTimeMs);
+  IncomingPacket(kBaseSeq + 1, kBaseTimeMs + kMaxSmallDeltaMs);
+  IncomingPacket(kBaseSeq + 2, kBaseTimeMs + (2 * kMaxSmallDeltaMs) + 1);
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq, packet->GetBaseSequence());
+        EXPECT_EQ(kMediaSsrc, packet->GetMediaSourceSsrc());
+
+        std::vector<rtcp::TransportFeedback::StatusSymbol> status_vec =
+            packet->GetStatusVector();
+        EXPECT_EQ(3u, status_vec.size());
+        EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta,
+                  status_vec[0]);
+        EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta,
+                  status_vec[1]);
+        EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedLargeDelta,
+                  status_vec[2]);
+
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(3u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs, (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        EXPECT_EQ(kMaxSmallDeltaMs, delta_vec[1] / 1000);
+        EXPECT_EQ(kMaxSmallDeltaMs + 1, delta_vec[2] / 1000);
+        return true;
+      }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, SendsFragmentedFeedback) {
+  const int64_t kTooLargeDelta =
+      rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 16);
+
+  IncomingPacket(kBaseSeq, kBaseTimeMs);
+  IncomingPacket(kBaseSeq + 1, kBaseTimeMs + kTooLargeDelta);
+
+  InSequence s;
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([kTooLargeDelta, this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq, packet->GetBaseSequence());
+        EXPECT_EQ(kMediaSsrc, packet->GetMediaSourceSsrc());
+
+        std::vector<rtcp::TransportFeedback::StatusSymbol> status_vec =
+            packet->GetStatusVector();
+        EXPECT_EQ(1u, status_vec.size());
+        EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta,
+                  status_vec[0]);
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(1u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs, (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        return true;
+      }))
+      .RetiresOnSaturation();
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([kTooLargeDelta, this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq + 1, packet->GetBaseSequence());
+        EXPECT_EQ(kMediaSsrc, packet->GetMediaSourceSsrc());
+
+        std::vector<rtcp::TransportFeedback::StatusSymbol> status_vec =
+            packet->GetStatusVector();
+        EXPECT_EQ(1u, status_vec.size());
+        EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta,
+                  status_vec[0]);
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(1u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs + kTooLargeDelta,
+                  (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        return true;
+      }))
+      .RetiresOnSaturation();
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, ResendsTimestampsOnReordering) {
+  IncomingPacket(kBaseSeq, kBaseTimeMs);
+  IncomingPacket(kBaseSeq + 2, kBaseTimeMs + 2);
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq, packet->GetBaseSequence());
+        EXPECT_EQ(kMediaSsrc, packet->GetMediaSourceSsrc());
+
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(2u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs, (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        EXPECT_EQ(2, delta_vec[1] / 1000);
+        return true;
+      }));
+
+  Process();
+
+  IncomingPacket(kBaseSeq + 1, kBaseTimeMs + 1);
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq + 1, packet->GetBaseSequence());
+        EXPECT_EQ(kMediaSsrc, packet->GetMediaSourceSsrc());
+
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(2u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs + 1,
+                  (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        EXPECT_EQ(1, delta_vec[1] / 1000);
+        return true;
+      }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, RemovesTimestampsOutOfScope) {
+  const int64_t kTimeoutTimeMs =
+      kBaseTimeMs + RemoteEstimatorProxy::kBackWindowMs;
+
+  IncomingPacket(kBaseSeq + 2, kBaseTimeMs);
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([kTimeoutTimeMs, this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq + 2, packet->GetBaseSequence());
+
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(1u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs, (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        return true;
+      }));
+
+  Process();
+
+  IncomingPacket(kBaseSeq + 3, kTimeoutTimeMs);  // kBaseSeq + 2 times out here.
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([kTimeoutTimeMs, this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq + 3, packet->GetBaseSequence());
+
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(1u, delta_vec.size());
+        EXPECT_EQ(kTimeoutTimeMs,
+                  (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        return true;
+      }));
+
+  Process();
+
+  // New group, with sequence starting below the first so that they may be
+  // retransmitted.
+  IncomingPacket(kBaseSeq, kBaseTimeMs - 1);
+  IncomingPacket(kBaseSeq + 1, kTimeoutTimeMs - 1);
+
+  EXPECT_CALL(router_, SendFeedback(_))
+      .Times(1)
+      .WillOnce(Invoke([kTimeoutTimeMs, this](rtcp::TransportFeedback* packet) {
+        packet->Build();
+        EXPECT_EQ(kBaseSeq, packet->GetBaseSequence());
+
+        // Four status entries (kBaseSeq + 3 missing).
+        EXPECT_EQ(4u, packet->GetStatusVector().size());
+
+        // Only three actual timestamps.
+        std::vector<int64_t> delta_vec = packet->GetReceiveDeltasUs();
+        EXPECT_EQ(3u, delta_vec.size());
+        EXPECT_EQ(kBaseTimeMs - 1,
+                  (packet->GetBaseTimeUs() + delta_vec[0]) / 1000);
+        EXPECT_EQ(kTimeoutTimeMs - kBaseTimeMs, delta_vec[1] / 1000);
+        EXPECT_EQ(1, delta_vec[2] / 1000);
+        return true;
+      }));
+
+  Process();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc
deleted file mode 100644
index 763a5751d7..0000000000
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/remote_bitrate_estimator/remote_rate_control.h"
-
-#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
-#include "webrtc/modules/remote_bitrate_estimator/mimd_rate_control.h"
-
-namespace webrtc {
-
-// static
-const int64_t RemoteRateControl::kMaxFeedbackIntervalMs = 1000;
-
-RemoteRateControl* RemoteRateControl::Create(RateControlType control_type,
-                                             uint32_t min_bitrate_bps) {
-  if (control_type == kAimdControl) {
-    return new AimdRateControl(min_bitrate_bps);
-  } else {
-    return new MimdRateControl(min_bitrate_bps);
-  }
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h
deleted file mode 100644
index 4398c57fa6..0000000000
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_rate_control.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_RATE_CONTROL_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_RATE_CONTROL_H_
-
-#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-
-namespace webrtc {
-
-class RemoteRateControl {
- public:
-  static RemoteRateControl* Create(RateControlType control_type,
-                                   uint32_t min_bitrate_bps);
-
-  virtual ~RemoteRateControl() {}
-
-  // Returns true if there is a valid estimate of the incoming bitrate, false
-  // otherwise.
-  virtual bool ValidEstimate() const = 0;
-  virtual RateControlType GetControlType() const = 0;
-  virtual uint32_t GetMinBitrate() const = 0;
-  virtual int64_t GetFeedbackInterval() const = 0;
-
-  // Returns true if the bitrate estimate hasn't been changed for more than
-  // an RTT, or if the incoming_bitrate is more than 5% above the current
-  // estimate. Should be used to decide if we should reduce the rate further
-  // when over-using.
-  virtual bool TimeToReduceFurther(int64_t time_now,
-                                   uint32_t incoming_bitrate_bps) const = 0;
-  virtual uint32_t LatestEstimate() const = 0;
-  virtual uint32_t UpdateBandwidthEstimate(int64_t now_ms) = 0;
-  virtual void SetRtt(int64_t rtt) = 0;
-  virtual RateControlRegion Update(const RateControlInput* input,
-                                   int64_t now_ms) = 0;
-  virtual void SetEstimate(int bitrate_bps, int64_t time_now_ms) = 0;
-
- protected:
-  static const int64_t kMaxFeedbackIntervalMs;
-};
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_RATE_CONTROL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
similarity index 59%
rename from media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history.cc
rename to media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
index 7e0c89e73d..a58d12a160 100644
--- a/media/webrtc/trunk/webrtc/modules/bitrate_controller/send_time_history.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
@@ -10,13 +10,14 @@
 
 #include <assert.h>
 
-#include "webrtc/modules/bitrate_controller/send_time_history.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
 
 namespace webrtc {
 
-SendTimeHistory::SendTimeHistory(int64_t packet_age_limit)
-    : packet_age_limit_(packet_age_limit), oldest_sequence_number_(0) {
-}
+SendTimeHistory::SendTimeHistory(Clock* clock, int64_t packet_age_limit)
+    : clock_(clock),
+      packet_age_limit_(packet_age_limit),
+      oldest_sequence_number_(0) {}
 
 SendTimeHistory::~SendTimeHistory() {
 }
@@ -25,23 +26,37 @@ void SendTimeHistory::Clear() {
   history_.clear();
 }
 
-void SendTimeHistory::AddAndRemoveOldSendTimes(uint16_t sequence_number,
-                                               int64_t timestamp) {
-  EraseOld(timestamp - packet_age_limit_);
+void SendTimeHistory::AddAndRemoveOld(uint16_t sequence_number,
+                                      size_t length,
+                                      bool was_paced) {
+  EraseOld();
 
   if (history_.empty())
     oldest_sequence_number_ = sequence_number;
 
-  history_[sequence_number] = timestamp;
+  history_.insert(std::pair<uint16_t, PacketInfo>(
+      sequence_number, PacketInfo(clock_->TimeInMilliseconds(), 0, -1,
+                                  sequence_number, length, was_paced)));
 }
 
-void SendTimeHistory::EraseOld(int64_t limit) {
+bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
+                                   int64_t send_time_ms) {
+  auto it = history_.find(sequence_number);
+  if (it == history_.end())
+    return false;
+  it->second.send_time_ms = send_time_ms;
+  return true;
+}
+
+void SendTimeHistory::EraseOld() {
   while (!history_.empty()) {
     auto it = history_.find(oldest_sequence_number_);
     assert(it != history_.end());
 
-    if (it->second > limit)
+    if (clock_->TimeInMilliseconds() - it->second.creation_time_ms <=
+        packet_age_limit_) {
       return;  // Oldest packet within age limit, return.
+    }
 
     // TODO(sprang): Warn if erasing (too many) old items?
     history_.erase(it);
@@ -68,16 +83,16 @@ void SendTimeHistory::UpdateOldestSequenceNumber() {
   oldest_sequence_number_ = it->first;
 }
 
-bool SendTimeHistory::GetSendTime(uint16_t sequence_number,
-                                  int64_t* timestamp,
-                                  bool remove) {
-  auto it = history_.find(sequence_number);
+bool SendTimeHistory::GetInfo(PacketInfo* packet, bool remove) {
+  auto it = history_.find(packet->sequence_number);
   if (it == history_.end())
     return false;
-  *timestamp = it->second;
+  int64_t receive_time = packet->arrival_time_ms;
+  *packet = it->second;
+  packet->arrival_time_ms = receive_time;
   if (remove) {
     history_.erase(it);
-    if (sequence_number == oldest_sequence_number_)
+    if (packet->sequence_number == oldest_sequence_number_)
       UpdateOldestSequenceNumber();
   }
   return true;
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
new file mode 100644
index 0000000000..b525813cdc
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
@@ -0,0 +1,234 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <limits>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+static const int kDefaultHistoryLengthMs = 1000;
+
+class SendTimeHistoryTest : public ::testing::Test {
+ protected:
+  SendTimeHistoryTest()
+      : clock_(0), history_(&clock_, kDefaultHistoryLengthMs) {}
+  ~SendTimeHistoryTest() {}
+
+  virtual void SetUp() {}
+
+  virtual void TearDown() {}
+
+  void AddPacketWithSendTime(uint16_t sequence_number,
+                             size_t length,
+                             bool was_paced,
+                             int64_t send_time_ms) {
+    history_.AddAndRemoveOld(sequence_number, length, was_paced);
+    history_.OnSentPacket(sequence_number, send_time_ms);
+  }
+
+  webrtc::SimulatedClock clock_;
+  SendTimeHistory history_;
+};
+
+// Help class extended so we can do EXPECT_EQ and collections.
+class PacketInfo : public webrtc::PacketInfo {
+ public:
+  PacketInfo() : webrtc::PacketInfo(-1, 0, 0, 0, 0, false) {}
+  PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
+      : PacketInfo(arrival_time_ms, 0, sequence_number, 0, false) {}
+  PacketInfo(int64_t arrival_time_ms,
+             int64_t send_time_ms,
+             uint16_t sequence_number,
+             size_t payload_size,
+             bool was_paced)
+      : webrtc::PacketInfo(-1,
+                           arrival_time_ms,
+                           send_time_ms,
+                           sequence_number,
+                           payload_size,
+                           was_paced) {}
+  bool operator==(const PacketInfo& other) const {
+    return arrival_time_ms == other.arrival_time_ms &&
+           send_time_ms == other.send_time_ms &&
+           sequence_number == other.sequence_number &&
+           payload_size == other.payload_size && was_paced == other.was_paced;
+  }
+};
+
+TEST_F(SendTimeHistoryTest, AddRemoveOne) {
+  const uint16_t kSeqNo = 10;
+  const PacketInfo kSentPacket(0, 1, kSeqNo, 1, true);
+  AddPacketWithSendTime(kSeqNo, 1, true, 1);
+
+  PacketInfo received_packet(0, 0, kSeqNo, 0, false);
+  EXPECT_TRUE(history_.GetInfo(&received_packet, false));
+  EXPECT_EQ(kSentPacket, received_packet);
+
+  PacketInfo received_packet2(0, 0, kSeqNo, 0, false);
+  EXPECT_TRUE(history_.GetInfo(&received_packet2, true));
+  EXPECT_EQ(kSentPacket, received_packet2);
+
+  PacketInfo received_packet3(0, 0, kSeqNo, 0, false);
+  EXPECT_FALSE(history_.GetInfo(&received_packet3, true));
+}
+
+TEST_F(SendTimeHistoryTest, PopulatesExpectedFields) {
+  const uint16_t kSeqNo = 10;
+  const int64_t kSendTime = 1000;
+  const int64_t kReceiveTime = 2000;
+  const size_t kPayloadSize = 42;
+  const bool kPaced = true;
+
+  AddPacketWithSendTime(kSeqNo, kPayloadSize, kPaced, kSendTime);
+
+  PacketInfo info(kReceiveTime, kSeqNo);
+  EXPECT_TRUE(history_.GetInfo(&info, true));
+  EXPECT_EQ(kReceiveTime, info.arrival_time_ms);
+  EXPECT_EQ(kSendTime, info.send_time_ms);
+  EXPECT_EQ(kSeqNo, info.sequence_number);
+  EXPECT_EQ(kPayloadSize, info.payload_size);
+  EXPECT_EQ(kPaced, info.was_paced);
+}
+
+TEST_F(SendTimeHistoryTest, AddThenRemoveOutOfOrder) {
+  std::vector<PacketInfo> sent_packets;
+  std::vector<PacketInfo> received_packets;
+  const size_t num_items = 100;
+  const size_t kPacketSize = 400;
+  const size_t kTransmissionTime = 1234;
+  const bool kPaced = true;
+  for (size_t i = 0; i < num_items; ++i) {
+    sent_packets.push_back(PacketInfo(0, static_cast<int64_t>(i),
+                                      static_cast<uint16_t>(i), kPacketSize,
+                                      kPaced));
+    received_packets.push_back(
+        PacketInfo(static_cast<int64_t>(i) + kTransmissionTime, 0,
+                   static_cast<uint16_t>(i), kPacketSize, false));
+  }
+  for (size_t i = 0; i < num_items; ++i) {
+    history_.AddAndRemoveOld(sent_packets[i].sequence_number,
+                             sent_packets[i].payload_size,
+                             sent_packets[i].was_paced);
+  }
+  for (size_t i = 0; i < num_items; ++i)
+    history_.OnSentPacket(sent_packets[i].sequence_number,
+                          sent_packets[i].send_time_ms);
+  std::random_shuffle(received_packets.begin(), received_packets.end());
+  for (size_t i = 0; i < num_items; ++i) {
+    PacketInfo packet = received_packets[i];
+    EXPECT_TRUE(history_.GetInfo(&packet, false));
+    PacketInfo sent_packet = sent_packets[packet.sequence_number];
+    sent_packet.arrival_time_ms = packet.arrival_time_ms;
+    EXPECT_EQ(sent_packet, packet);
+    EXPECT_TRUE(history_.GetInfo(&packet, true));
+  }
+  for (PacketInfo packet : sent_packets)
+    EXPECT_FALSE(history_.GetInfo(&packet, false));
+}
+
+TEST_F(SendTimeHistoryTest, HistorySize) {
+  const int kItems = kDefaultHistoryLengthMs / 100;
+  for (int i = 0; i < kItems; ++i) {
+    clock_.AdvanceTimeMilliseconds(100);
+    AddPacketWithSendTime(i, 0, false, i * 100);
+  }
+  for (int i = 0; i < kItems; ++i) {
+    PacketInfo info(0, 0, static_cast<uint16_t>(i), 0, false);
+    EXPECT_TRUE(history_.GetInfo(&info, false));
+    EXPECT_EQ(i * 100, info.send_time_ms);
+  }
+  clock_.AdvanceTimeMilliseconds(101);
+  AddPacketWithSendTime(kItems, 0, false, kItems * 101);
+  PacketInfo info(0, 0, 0, 0, false);
+  EXPECT_FALSE(history_.GetInfo(&info, false));
+  for (int i = 1; i < (kItems + 1); ++i) {
+    PacketInfo info2(0, 0, static_cast<uint16_t>(i), 0, false);
+    EXPECT_TRUE(history_.GetInfo(&info2, false));
+    int64_t expected_time_ms = (i == kItems) ? i * 101 : i * 100;
+    EXPECT_EQ(expected_time_ms, info2.send_time_ms);
+  }
+}
+
+TEST_F(SendTimeHistoryTest, HistorySizeWithWraparound) {
+  const uint16_t kMaxSeqNo = std::numeric_limits<uint16_t>::max();
+  AddPacketWithSendTime(kMaxSeqNo - 2, 0, false, 0);
+
+  clock_.AdvanceTimeMilliseconds(100);
+  AddPacketWithSendTime(kMaxSeqNo - 1, 1, false, 100);
+
+  clock_.AdvanceTimeMilliseconds(100);
+  AddPacketWithSendTime(kMaxSeqNo, 0, false, 200);
+
+  clock_.AdvanceTimeMilliseconds(kDefaultHistoryLengthMs - 200 + 1);
+  AddPacketWithSendTime(0, 0, false, kDefaultHistoryLengthMs);
+
+  PacketInfo info(0, static_cast<uint16_t>(kMaxSeqNo - 2));
+  EXPECT_FALSE(history_.GetInfo(&info, false));
+  PacketInfo info2(0, static_cast<uint16_t>(kMaxSeqNo - 1));
+  EXPECT_TRUE(history_.GetInfo(&info2, false));
+  PacketInfo info3(0, static_cast<uint16_t>(kMaxSeqNo));
+  EXPECT_TRUE(history_.GetInfo(&info3, false));
+  PacketInfo info4(0, 0);
+  EXPECT_TRUE(history_.GetInfo(&info4, false));
+
+  // Create a gap (kMaxSeqNo - 1) -> 0.
+  PacketInfo info5(0, kMaxSeqNo);
+  EXPECT_TRUE(history_.GetInfo(&info5, true));
+
+  clock_.AdvanceTimeMilliseconds(100);
+  AddPacketWithSendTime(1, 0, false, 1100);
+
+  PacketInfo info6(0, static_cast<uint16_t>(kMaxSeqNo - 2));
+  EXPECT_FALSE(history_.GetInfo(&info6, false));
+  PacketInfo info7(0, static_cast<uint16_t>(kMaxSeqNo - 1));
+  EXPECT_FALSE(history_.GetInfo(&info7, false));
+  PacketInfo info8(0, kMaxSeqNo);
+  EXPECT_FALSE(history_.GetInfo(&info8, false));
+  PacketInfo info9(0, 0);
+  EXPECT_TRUE(history_.GetInfo(&info9, false));
+  PacketInfo info10(0, 1);
+  EXPECT_TRUE(history_.GetInfo(&info10, false));
+}
+
+TEST_F(SendTimeHistoryTest, InterlievedGetAndRemove) {
+  const uint16_t kSeqNo = 1;
+  const int64_t kTimestamp = 2;
+  PacketInfo packets[3] = {{0, kTimestamp, kSeqNo, 0, false},
+                           {0, kTimestamp + 1, kSeqNo + 1, 0, false},
+                           {0, kTimestamp + 2, kSeqNo + 2, 0, false}};
+
+  AddPacketWithSendTime(packets[0].sequence_number, packets[0].payload_size,
+                        packets[0].was_paced, packets[0].send_time_ms);
+  AddPacketWithSendTime(packets[1].sequence_number, packets[1].payload_size,
+                        packets[1].was_paced, packets[1].send_time_ms);
+  PacketInfo info(0, 0, packets[0].sequence_number, 0, false);
+  EXPECT_TRUE(history_.GetInfo(&info, true));
+  EXPECT_EQ(packets[0], info);
+
+  AddPacketWithSendTime(packets[2].sequence_number, packets[2].payload_size,
+                        packets[2].was_paced, packets[2].send_time_ms);
+
+  PacketInfo info2(0, 0, packets[1].sequence_number, 0, false);
+  EXPECT_TRUE(history_.GetInfo(&info2, true));
+  EXPECT_EQ(packets[1], info2);
+
+  PacketInfo info3(0, 0, packets[2].sequence_number, 0, false);
+  EXPECT_TRUE(history_.GetInfo(&info3, true));
+  EXPECT_EQ(packets[2], info3);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
index 3b233709c8..c667b6864e 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
@@ -16,11 +16,46 @@
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
+// With the assumption that packet loss is lower than 97%, the max gap
+// between elements in the set is lower than 0x8000, hence we have a
+// total order in the set. For (x,y,z) subset of the LinkedSet,
+// (x<=y and y<=z) ==> x<=z so the set can be sorted.
+const int kSetCapacity = 1000;
+
+BweReceiver::BweReceiver(int flow_id)
+    : flow_id_(flow_id),
+      received_packets_(kSetCapacity),
+      rate_counter_(),
+      loss_account_() {
+}
+
+BweReceiver::BweReceiver(int flow_id, int64_t window_size_ms)
+    : flow_id_(flow_id),
+      received_packets_(kSetCapacity),
+      rate_counter_(window_size_ms),
+      loss_account_() {
+}
+
+void BweReceiver::ReceivePacket(int64_t arrival_time_ms,
+                                const MediaPacket& media_packet) {
+  if (received_packets_.size() == kSetCapacity) {
+    RelieveSetAndUpdateLoss();
+  }
+
+  received_packets_.Insert(media_packet.sequence_number(),
+                           media_packet.send_time_ms(), arrival_time_ms,
+                           media_packet.payload_size());
+
+  rate_counter_.UpdateRates(media_packet.send_time_ms() * 1000,
+                            static_cast<uint32_t>(media_packet.payload_size()));
+}
+
 class NullBweSender : public BweSender {
  public:
   NullBweSender() {}
@@ -35,7 +70,7 @@ class NullBweSender : public BweSender {
   int Process() override { return 0; }
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(NullBweSender);
+  RTC_DISALLOW_COPY_AND_ASSIGN(NullBweSender);
 };
 
 int64_t GetAbsSendTimeInMs(uint32_t abs_send_time) {
@@ -58,6 +93,8 @@ BweSender* CreateBweSender(BandwidthEstimatorType estimator,
       return new FullBweSender(kbps, observer, clock);
     case kNadaEstimator:
       return new NadaBweSender(kbps, observer, clock);
+    case kTcpEstimator:
+      FALLTHROUGH();
     case kNullEstimator:
       return new NullBweSender();
   }
@@ -75,12 +112,169 @@ BweReceiver* CreateBweReceiver(BandwidthEstimatorType type,
       return new SendSideBweReceiver(flow_id);
     case kNadaEstimator:
       return new NadaBweReceiver(flow_id);
+    case kTcpEstimator:
+      return new TcpBweReceiver(flow_id);
     case kNullEstimator:
       return new BweReceiver(flow_id);
   }
   assert(false);
   return NULL;
 }
+
+// Take into account all LinkedSet content.
+void BweReceiver::UpdateLoss() {
+  loss_account_.Add(LinkedSetPacketLossRatio());
+}
+
+// Preserve 10% latest packets and update packet loss based on the oldest
+// 90%, that will be removed.
+void BweReceiver::RelieveSetAndUpdateLoss() {
+  // Compute Loss for the whole LinkedSet and updates loss_account_.
+  UpdateLoss();
+
+  size_t num_preserved_elements = received_packets_.size() / 10;
+  PacketNodeIt it = received_packets_.begin();
+  std::advance(it, num_preserved_elements);
+
+  while (it != received_packets_.end()) {
+    received_packets_.Erase(it++);
+  }
+
+  // Compute Loss for the preserved elements
+  loss_account_.Subtract(LinkedSetPacketLossRatio());
+}
+
+float BweReceiver::GlobalReceiverPacketLossRatio() {
+  UpdateLoss();
+  return loss_account_.LossRatio();
+}
+
+// This function considers at most kSetCapacity = 1000 packets.
+LossAccount BweReceiver::LinkedSetPacketLossRatio() {
+  if (received_packets_.empty()) {
+    return LossAccount();
+  }
+
+  uint16_t oldest_seq_num = received_packets_.OldestSeqNumber();
+  uint16_t newest_seq_num = received_packets_.NewestSeqNumber();
+
+  size_t set_total_packets =
+      static_cast<uint16_t>(newest_seq_num - oldest_seq_num + 1);
+
+  size_t set_received_packets = received_packets_.size();
+  size_t set_lost_packets = set_total_packets - set_received_packets;
+
+  return LossAccount(set_total_packets, set_lost_packets);
+}
+
+uint32_t BweReceiver::RecentKbps() const {
+  return (rate_counter_.bits_per_second() + 500) / 1000;
+}
+
+// Go through a fixed time window of most recent packets received and
+// counts packets missing to obtain the packet loss ratio. If an unordered
+// packet falls out of the timewindow it will be counted as missing.
+// E.g.: for a timewindow covering 5 packets of the following arrival sequence
+// {10 7 9 5 6} 8 3 2 4 1, the output will be 1/6 (#8 is considered as missing).
+float BweReceiver::RecentPacketLossRatio() {
+  if (received_packets_.empty()) {
+    return 0.0f;
+  }
+  int number_packets_received = 0;
+
+  PacketNodeIt node_it = received_packets_.begin();  // Latest.
+
+  // Lowest timestamp limit, oldest one that should be checked.
+  int64_t time_limit_ms = (*node_it)->arrival_time_ms - kPacketLossTimeWindowMs;
+  // Oldest and newest values found within the given time window.
+  uint16_t oldest_seq_num = (*node_it)->sequence_number;
+  uint16_t newest_seq_num = oldest_seq_num;
+
+  while (node_it != received_packets_.end()) {
+    if ((*node_it)->arrival_time_ms < time_limit_ms) {
+      break;
+    }
+    uint16_t seq_num = (*node_it)->sequence_number;
+    if (IsNewerSequenceNumber(seq_num, newest_seq_num)) {
+      newest_seq_num = seq_num;
+    }
+    if (IsNewerSequenceNumber(oldest_seq_num, seq_num)) {
+      oldest_seq_num = seq_num;
+    }
+    ++node_it;
+    ++number_packets_received;
+  }
+  // Interval width between oldest and newest sequence number.
+  // There was an overflow if newest_seq_num < oldest_seq_num.
+  int gap = static_cast<uint16_t>(newest_seq_num - oldest_seq_num + 1);
+
+  return static_cast<float>(gap - number_packets_received) / gap;
+}
+
+LinkedSet::~LinkedSet() {
+  while (!empty())
+    RemoveTail();
+}
+
+void LinkedSet::Insert(uint16_t sequence_number,
+                       int64_t send_time_ms,
+                       int64_t arrival_time_ms,
+                       size_t payload_size) {
+  auto it = map_.find(sequence_number);
+  if (it != map_.end()) {
+    PacketNodeIt node_it = it->second;
+    PacketIdentifierNode* node = *node_it;
+    node->arrival_time_ms = arrival_time_ms;
+    if (node_it != list_.begin()) {
+      list_.erase(node_it);
+      list_.push_front(node);
+      map_[sequence_number] = list_.begin();
+    }
+  } else {
+    if (size() == capacity_) {
+      RemoveTail();
+    }
+    UpdateHead(new PacketIdentifierNode(sequence_number, send_time_ms,
+                                        arrival_time_ms, payload_size));
+  }
+}
+
+void LinkedSet::Insert(PacketIdentifierNode packet_identifier) {
+  Insert(packet_identifier.sequence_number, packet_identifier.send_time_ms,
+         packet_identifier.arrival_time_ms, packet_identifier.payload_size);
+}
+
+void LinkedSet::RemoveTail() {
+  map_.erase(list_.back()->sequence_number);
+  delete list_.back();
+  list_.pop_back();
+}
+void LinkedSet::UpdateHead(PacketIdentifierNode* new_head) {
+  list_.push_front(new_head);
+  map_[new_head->sequence_number] = list_.begin();
+}
+
+void LinkedSet::Erase(PacketNodeIt node_it) {
+  map_.erase((*node_it)->sequence_number);
+  delete (*node_it);
+  list_.erase(node_it);
+}
+
+void LossAccount::Add(LossAccount rhs) {
+  num_total += rhs.num_total;
+  num_lost += rhs.num_lost;
+}
+void LossAccount::Subtract(LossAccount rhs) {
+  num_total -= rhs.num_total;
+  num_lost -= rhs.num_lost;
+}
+
+float LossAccount::LossRatio() {
+  if (num_total == 0)
+    return 0.0f;
+  return static_cast<float>(num_lost) / num_total;
+}
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.h
index 0bab5a98a1..8d29de2619 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe.h
@@ -11,51 +11,176 @@
 #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_H_
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_H_
 
+#include <list>
+#include <map>
 #include <sstream>
+#include <string>
 
+#include "webrtc/test/testsupport/gtest_prod_util.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
-const int kMinBitrateKbps = 150;
-const int kMaxBitrateKbps = 2000;
+// Overload map comparator.
+class SequenceNumberOlderThan {
+ public:
+  bool operator()(uint16_t seq_num_1, uint16_t seq_num_2) const {
+    return IsNewerSequenceNumber(seq_num_2, seq_num_1);
+  }
+};
+
+// Holds information for computing global packet loss.
+struct LossAccount {
+  LossAccount() : num_total(0), num_lost(0) {}
+  LossAccount(size_t num_total, size_t num_lost)
+      : num_total(num_total), num_lost(num_lost) {}
+  void Add(LossAccount rhs);
+  void Subtract(LossAccount rhs);
+  float LossRatio();
+  size_t num_total;
+  size_t num_lost;
+};
+
+// Holds only essential information about packets to be saved for
+// further use, e.g. for calculating packet loss and receiving rate.
+struct PacketIdentifierNode {
+  PacketIdentifierNode(uint16_t sequence_number,
+                       int64_t send_time_ms,
+                       int64_t arrival_time_ms,
+                       size_t payload_size)
+      : sequence_number(sequence_number),
+        send_time_ms(send_time_ms),
+        arrival_time_ms(arrival_time_ms),
+        payload_size(payload_size) {}
+
+  uint16_t sequence_number;
+  int64_t send_time_ms;
+  int64_t arrival_time_ms;
+  size_t payload_size;
+};
+
+typedef std::list<PacketIdentifierNode*>::iterator PacketNodeIt;
+
+// FIFO implementation for a limited capacity set.
+// Used for keeping the latest arrived packets while avoiding duplicates.
+// Allows efficient insertion, deletion and search.
+class LinkedSet {
+ public:
+  explicit LinkedSet(int capacity) : capacity_(capacity) {}
+  ~LinkedSet();
+
+  // If the arriving packet (identified by its sequence number) is already
+  // in the LinkedSet, move its Node to the head of the list. Else, create
+  // a PacketIdentifierNode n_ and then UpdateHead(n_), calling RemoveTail()
+  // if the LinkedSet reached its maximum capacity.
+  void Insert(uint16_t sequence_number,
+              int64_t send_time_ms,
+              int64_t arrival_time_ms,
+              size_t payload_size);
+
+  void Insert(PacketIdentifierNode packet_identifier);
+
+  PacketNodeIt begin() { return list_.begin(); }
+  PacketNodeIt end() { return list_.end(); }
+
+  bool empty() const { return list_.empty(); }
+  size_t size() const { return list_.size(); }
+  size_t capacity() const { return capacity_; }
+
+  uint16_t OldestSeqNumber() const { return empty() ? 0 : map_.begin()->first; }
+  uint16_t NewestSeqNumber() const {
+    return empty() ? 0 : map_.rbegin()->first;
+  }
+
+  void Erase(PacketNodeIt node_it);
+
+ private:
+  // Pop oldest element from the back of the list and remove it from the map.
+  void RemoveTail();
+  // Add new element to the front of the list and insert it in the map.
+  void UpdateHead(PacketIdentifierNode* new_head);
+  size_t capacity_;
+  std::map<uint16_t, PacketNodeIt, SequenceNumberOlderThan> map_;
+  std::list<PacketIdentifierNode*> list_;
+};
+
+const int kMinBitrateKbps = 50;
+const int kMaxBitrateKbps = 2500;
 
 class BweSender : public Module {
  public:
   BweSender() {}
+  explicit BweSender(int bitrate_kbps) : bitrate_kbps_(bitrate_kbps) {}
   virtual ~BweSender() {}
 
   virtual int GetFeedbackIntervalMs() const = 0;
   virtual void GiveFeedback(const FeedbackPacket& feedback) = 0;
   virtual void OnPacketsSent(const Packets& packets) = 0;
 
+ protected:
+  int bitrate_kbps_;
+
  private:
-  DISALLOW_COPY_AND_ASSIGN(BweSender);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweSender);
 };
 
 class BweReceiver {
  public:
-  explicit BweReceiver(int flow_id) : flow_id_(flow_id) {}
+  explicit BweReceiver(int flow_id);
+  BweReceiver(int flow_id, int64_t window_size_ms);
+
   virtual ~BweReceiver() {}
 
   virtual void ReceivePacket(int64_t arrival_time_ms,
-                             const MediaPacket& media_packet) {}
+                             const MediaPacket& media_packet);
   virtual FeedbackPacket* GetFeedback(int64_t now_ms) { return NULL; }
 
+  size_t GetSetCapacity() { return received_packets_.capacity(); }
+  double BitrateWindowS() const { return rate_counter_.BitrateWindowS(); }
+  uint32_t RecentKbps() const;  // Receiving Rate.
+
+  // Computes packet loss during an entire simulation, up to 4 billion packets.
+  float GlobalReceiverPacketLossRatio();  // Plot histogram.
+  float RecentPacketLossRatio();          // Plot dynamics.
+
+  static const int64_t kPacketLossTimeWindowMs = 500;
+  static const int64_t kReceivingRateTimeWindowMs = 1000;
+
  protected:
   int flow_id_;
+  // Deals with packets sent more than once.
+  LinkedSet received_packets_;
+  // Used for calculating recent receiving rate.
+  RateCounter rate_counter_;
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(BweReceiverTest, RecentKbps);
+  FRIEND_TEST_ALL_PREFIXES(BweReceiverTest, Loss);
+
+  void UpdateLoss();
+  void RelieveSetAndUpdateLoss();
+  // Packet loss for packets stored in the LinkedSet, up to 1000 packets.
+  // Used to update global loss account whenever the set is filled and cleared.
+  LossAccount LinkedSetPacketLossRatio();
+
+  // Used for calculating global packet loss ratio.
+  LossAccount loss_account_;
 };
 
 enum BandwidthEstimatorType {
   kNullEstimator,
   kNadaEstimator,
   kRembEstimator,
-  kFullSendSideEstimator
+  kFullSendSideEstimator,
+  kTcpEstimator
 };
 
+const std::string bwe_names[] = {"Null", "NADA", "REMB", "GCC", "TCP"};
+
 int64_t GetAbsSendTimeInMs(uint32_t abs_send_time);
 
 BweSender* CreateBweSender(BandwidthEstimatorType estimator,
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_plot.sh b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_plot.sh
deleted file mode 100644
index 4695af45cd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_plot.sh
+++ /dev/null
@@ -1,63 +0,0 @@
-#!/bin/bash
-
-# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# To set up in e.g. Eclipse, run a separate shell and pipe the output from the
-# test into this script.
-#
-# In Eclipse, that amounts to creating a Run Configuration which starts
-# "/bin/bash" with the arguments "-c [trunk_path]/out/Debug/modules_unittests
-# --gtest_filter=*BweTest* | [trunk_path]/webrtc/modules/
-# remote_bitrate_estimator/bwe_plot.
-
-# bwe_plot.sh has a single y axis and a dual y axis mode. If any line specifies
-# a an axis by ending with "#<axis number (1 or 2)>" two y axis will be used,
-# the first will be assumed to represent bitrate (in kbps) and the second will
-# be assumed to represent time deltas (in ms).
-
-log=$(</dev/stdin)
-
-function gen_gnuplot_input {
-  colors=(a7001f 0a60c2 b2582b 21a66c d6604d 4393c3 f4a582 92c5de edcbb7 b1c5d0)
-  data_sets=$(echo "$log" | grep "^PLOT" | cut -f 2 | sort | uniq)
-  linetypes=($(echo "$data_sets" | cut -d '#' -f 2 | cut -d ' ' -f 1))
-  echo -n "reset; "
-  echo -n "set terminal wxt size 1440,900 font \"Arial,9\"; "
-  echo -n "set xlabel \"Seconds\"; "
-  if [ -n $linetypes ]; then
-    echo -n "set ylabel 'bitrate (kbps)';"
-    echo -n "set ytics nomirror;"
-    echo -n "set y2label 'time delta (ms)';"
-    echo -n "set y2tics nomirror;"
-  fi
-  echo -n "plot "
-  i=0
-  for set in $data_sets ; do
-    (( i++ )) && echo -n ","
-    echo -n "'-' with "
-    echo -n "linespoints "
-    echo -n "ps 0.5 "
-    echo -n "lc rgbcolor \"#${colors[$(($i % 10))]}\" "
-    if [ -n ${linetypes[$i - 1]} ]; then
-      echo -n "axes x1y${linetypes[$i - 1]} "
-    elif [ -n $linestypes ]; then
-      # If no line type is specified, but line types are used, we will default
-      # to the bitrate axis.
-      echo -n "axes x1y1 "
-    fi
-    echo -n "title \"$set\" "
-  done
-  echo
-  for set in $data_sets ; do
-    echo "$log" | grep "^PLOT.$set" | cut -f 3,4
-    echo "e"
-  done
-}
-
-gen_gnuplot_input "$log" | gnuplot -persist
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
index 64406c7e48..9da21c1aaa 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
@@ -10,13 +10,18 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
 
+#include <sstream>
+
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/testsupport/perf_test.h"
 
 using std::string;
 using std::vector;
@@ -47,7 +52,7 @@ void PacketProcessorRunner::RunFor(int64_t time_ms,
   processor_->RunFor(time_ms, &to_process);
   QueuePackets(&to_process, time_now_ms * 1000);
   if (!to_process.empty()) {
-    processor_->Plot((to_process.back()->send_time_us() + 500) / 1000);
+    processor_->Plot(to_process.back()->send_time_ms());
   }
   in_out->merge(to_process, DereferencingComparator<Packet>);
 }
@@ -88,8 +93,15 @@ void PacketProcessorRunner::QueuePackets(Packets* batch,
   batch->merge(to_transfer, DereferencingComparator<Packet>);
 }
 
-BweTest::BweTest()
-    : run_time_ms_(0), time_now_ms_(-1), simulation_interval_ms_(-1) {
+// Plot link capacity by default.
+BweTest::BweTest() : BweTest(true) {
+}
+
+BweTest::BweTest(bool plot_capacity)
+    : run_time_ms_(0),
+      time_now_ms_(-1),
+      simulation_interval_ms_(-1),
+      plot_total_available_capacity_(plot_capacity) {
   links_.push_back(&uplink_);
   links_.push_back(&downlink_);
 }
@@ -175,6 +187,798 @@ string BweTest::GetTestName() const {
       ::testing::UnitTest::GetInstance()->current_test_info();
   return string(test_info->name());
 }
+
+void BweTest::PrintResults(double max_throughput_kbps,
+                           Stats<double> throughput_kbps,
+                           int flow_id,
+                           Stats<double> flow_delay_ms,
+                           Stats<double> flow_throughput_kbps) {
+  std::map<int, Stats<double>> flow_delays_ms;
+  flow_delays_ms[flow_id] = flow_delay_ms;
+  std::map<int, Stats<double>> flow_throughputs_kbps;
+  flow_throughputs_kbps[flow_id] = flow_throughput_kbps;
+  PrintResults(max_throughput_kbps, throughput_kbps, flow_delays_ms,
+               flow_throughputs_kbps);
+}
+
+void BweTest::PrintResults(double max_throughput_kbps,
+                           Stats<double> throughput_kbps,
+                           std::map<int, Stats<double>> flow_delay_ms,
+                           std::map<int, Stats<double>> flow_throughput_kbps) {
+  double utilization = throughput_kbps.GetMean() / max_throughput_kbps;
+  webrtc::test::PrintResult("BwePerformance", GetTestName(), "Utilization",
+                            utilization * 100.0, "%", false);
+  std::stringstream ss;
+  ss << throughput_kbps.GetStdDev() / throughput_kbps.GetMean();
+  webrtc::test::PrintResult("BwePerformance", GetTestName(),
+                            "Utilization var coeff", ss.str(), "", false);
+  for (auto& kv : flow_throughput_kbps) {
+    ss.str("");
+    ss << "Throughput flow " << kv.first;
+    webrtc::test::PrintResultMeanAndError("BwePerformance", GetTestName(),
+                                          ss.str(), kv.second.AsString(),
+                                          "kbps", false);
+  }
+  for (auto& kv : flow_delay_ms) {
+    ss.str("");
+    ss << "Delay flow " << kv.first;
+    webrtc::test::PrintResultMeanAndError("BwePerformance", GetTestName(),
+                                          ss.str(), kv.second.AsString(), "ms",
+                                          false);
+  }
+  double fairness_index = 1.0;
+  if (!flow_throughput_kbps.empty()) {
+    double squared_bitrate_sum = 0.0;
+    fairness_index = 0.0;
+    for (auto kv : flow_throughput_kbps) {
+      squared_bitrate_sum += kv.second.GetMean() * kv.second.GetMean();
+      fairness_index += kv.second.GetMean();
+    }
+    fairness_index *= fairness_index;
+    fairness_index /= flow_throughput_kbps.size() * squared_bitrate_sum;
+  }
+  webrtc::test::PrintResult("BwePerformance", GetTestName(), "Fairness",
+                            fairness_index * 100, "%", false);
+}
+
+void BweTest::RunFairnessTest(BandwidthEstimatorType bwe_type,
+                              size_t num_media_flows,
+                              size_t num_tcp_flows,
+                              int64_t run_time_seconds,
+                              uint32_t capacity_kbps,
+                              int64_t max_delay_ms,
+                              int64_t rtt_ms,
+                              int64_t max_jitter_ms,
+                              const int64_t* offsets_ms) {
+  RunFairnessTest(bwe_type, num_media_flows, num_tcp_flows, run_time_seconds,
+                  capacity_kbps, max_delay_ms, rtt_ms, max_jitter_ms,
+                  offsets_ms, "Fairness_test", bwe_names[bwe_type]);
+}
+
+void BweTest::RunFairnessTest(BandwidthEstimatorType bwe_type,
+                              size_t num_media_flows,
+                              size_t num_tcp_flows,
+                              int64_t run_time_seconds,
+                              uint32_t capacity_kbps,
+                              int64_t max_delay_ms,
+                              int64_t rtt_ms,
+                              int64_t max_jitter_ms,
+                              const int64_t* offsets_ms,
+                              const std::string& title,
+                              const std::string& flow_name) {
+  std::set<int> all_flow_ids;
+  std::set<int> media_flow_ids;
+  std::set<int> tcp_flow_ids;
+  int next_flow_id = 0;
+  for (size_t i = 0; i < num_media_flows; ++i) {
+    media_flow_ids.insert(next_flow_id);
+    all_flow_ids.insert(next_flow_id);
+    ++next_flow_id;
+  }
+  for (size_t i = 0; i < num_tcp_flows; ++i) {
+    tcp_flow_ids.insert(next_flow_id);
+    all_flow_ids.insert(next_flow_id);
+    ++next_flow_id;
+  }
+
+  std::vector<VideoSource*> sources;
+  std::vector<PacketSender*> senders;
+  std::vector<MetricRecorder*> metric_recorders;
+
+  int64_t max_offset_ms = 0;
+
+  for (int media_flow : media_flow_ids) {
+    sources.push_back(new AdaptiveVideoSource(media_flow, 30, 300, 0,
+                                              offsets_ms[media_flow]));
+    senders.push_back(new PacedVideoSender(&uplink_, sources.back(), bwe_type));
+    max_offset_ms = std::max(max_offset_ms, offsets_ms[media_flow]);
+  }
+
+  for (int tcp_flow : tcp_flow_ids) {
+    senders.push_back(new TcpSender(&uplink_, tcp_flow, offsets_ms[tcp_flow]));
+    max_offset_ms = std::max(max_offset_ms, offsets_ms[tcp_flow]);
+  }
+
+  ChokeFilter choke(&uplink_, all_flow_ids);
+  choke.set_capacity_kbps(capacity_kbps);
+  choke.set_max_delay_ms(max_delay_ms);
+  LinkShare link_share(&choke);
+
+  int64_t one_way_delay_ms = rtt_ms / 2;
+  DelayFilter delay_uplink(&uplink_, all_flow_ids);
+  delay_uplink.SetOneWayDelayMs(one_way_delay_ms);
+
+  JitterFilter jitter(&uplink_, all_flow_ids);
+  jitter.SetMaxJitter(max_jitter_ms);
+
+  std::vector<RateCounterFilter*> rate_counters;
+  for (int flow : media_flow_ids) {
+    rate_counters.push_back(
+        new RateCounterFilter(&uplink_, flow, "Receiver", bwe_names[bwe_type]));
+  }
+  for (int flow : tcp_flow_ids) {
+    rate_counters.push_back(new RateCounterFilter(&uplink_, flow, "Receiver",
+                                                  bwe_names[kTcpEstimator]));
+  }
+
+  RateCounterFilter total_utilization(
+      &uplink_, all_flow_ids, "total_utilization", "Total_link_utilization");
+
+  std::vector<PacketReceiver*> receivers;
+  // Delays is being plotted only for the first flow.
+  // To plot all of them, replace "i == 0" with "true" on new PacketReceiver().
+  for (int media_flow : media_flow_ids) {
+    metric_recorders.push_back(
+        new MetricRecorder(bwe_names[bwe_type], static_cast<int>(media_flow),
+                           senders[media_flow], &link_share));
+    receivers.push_back(new PacketReceiver(&uplink_, media_flow, bwe_type,
+                                           media_flow == 0, false,
+                                           metric_recorders[media_flow]));
+    metric_recorders[media_flow]->set_plot_available_capacity(
+        media_flow == 0 && plot_total_available_capacity_);
+    metric_recorders[media_flow]->set_start_computing_metrics_ms(max_offset_ms);
+  }
+  // Delays is not being plotted only for TCP flows. To plot all of them,
+  // replace first "false" occurence with "true" on new PacketReceiver().
+  for (int tcp_flow : tcp_flow_ids) {
+    metric_recorders.push_back(
+        new MetricRecorder(bwe_names[kTcpEstimator], static_cast<int>(tcp_flow),
+                           senders[tcp_flow], &link_share));
+    receivers.push_back(new PacketReceiver(&uplink_, tcp_flow, kTcpEstimator,
+                                           false, false,
+                                           metric_recorders[tcp_flow]));
+    metric_recorders[tcp_flow]->set_plot_available_capacity(
+        tcp_flow == 0 && plot_total_available_capacity_);
+  }
+
+  DelayFilter delay_downlink(&downlink_, all_flow_ids);
+  delay_downlink.SetOneWayDelayMs(one_way_delay_ms);
+
+  RunFor(run_time_seconds * 1000);
+
+  std::map<int, Stats<double>> flow_throughput_kbps;
+  for (RateCounterFilter* rate_counter : rate_counters) {
+    int flow_id = *rate_counter->flow_ids().begin();
+    flow_throughput_kbps[flow_id] = rate_counter->GetBitrateStats();
+  }
+
+  std::map<int, Stats<double>> flow_delay_ms;
+  for (PacketReceiver* receiver : receivers) {
+    int flow_id = *receiver->flow_ids().begin();
+    flow_delay_ms[flow_id] = receiver->GetDelayStats();
+  }
+
+  PrintResults(capacity_kbps, total_utilization.GetBitrateStats(),
+               flow_delay_ms, flow_throughput_kbps);
+
+  for (int i : all_flow_ids) {
+    metric_recorders[i]->PlotThroughputHistogram(
+        title, flow_name, static_cast<int>(num_media_flows), 0);
+
+    metric_recorders[i]->PlotLossHistogram(title, flow_name,
+                                           static_cast<int>(num_media_flows),
+                                           receivers[i]->GlobalPacketLoss());
+  }
+
+  // Pointless to show delay histogram for TCP flow.
+  for (int i : media_flow_ids) {
+    metric_recorders[i]->PlotDelayHistogram(title, bwe_names[bwe_type],
+                                            static_cast<int>(num_media_flows),
+                                            one_way_delay_ms);
+    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], one_way_delay_ms, i);
+  }
+
+  for (VideoSource* source : sources)
+    delete source;
+  for (PacketSender* sender : senders)
+    delete sender;
+  for (RateCounterFilter* rate_counter : rate_counters)
+    delete rate_counter;
+  for (PacketReceiver* receiver : receivers)
+    delete receiver;
+  for (MetricRecorder* recorder : metric_recorders)
+    delete recorder;
+}
+
+void BweTest::RunChoke(BandwidthEstimatorType bwe_type,
+                       std::vector<int> capacities_kbps) {
+  int flow_id = bwe_type;
+  AdaptiveVideoSource source(flow_id, 30, 300, 0, 0);
+  VideoSender sender(&uplink_, &source, bwe_type);
+  ChokeFilter choke(&uplink_, flow_id);
+  LinkShare link_share(&choke);
+  MetricRecorder metric_recorder(bwe_names[bwe_type], flow_id, &sender,
+                                 &link_share);
+  PacketReceiver receiver(&uplink_, flow_id, bwe_type, true, false,
+                          &metric_recorder);
+  metric_recorder.set_plot_available_capacity(plot_total_available_capacity_);
+
+  choke.set_max_delay_ms(500);
+  const int64_t kRunTimeMs = 60 * 1000;
+
+  std::stringstream title("Choke");
+  char delimiter = '_';
+
+  for (auto it = capacities_kbps.begin(); it != capacities_kbps.end(); ++it) {
+    choke.set_capacity_kbps(*it);
+    RunFor(kRunTimeMs);
+    title << delimiter << (*it);
+    delimiter = '-';
+  }
+
+  title << "_kbps,_" << (kRunTimeMs / 1000) << "s_each";
+  metric_recorder.PlotThroughputHistogram(title.str(), bwe_names[bwe_type], 1,
+                                          0);
+  metric_recorder.PlotDelayHistogram(title.str(), bwe_names[bwe_type], 1, 0);
+  // receiver.PlotLossHistogram(title, bwe_names[bwe_type], 1);
+  // receiver.PlotObjectiveHistogram(title, bwe_names[bwe_type], 1);
+}
+
+// 5.1. Single Video and Audio media traffic, forward direction.
+void BweTest::RunVariableCapacity1SingleFlow(BandwidthEstimatorType bwe_type) {
+  const int kFlowId = 0;  // Arbitrary value.
+  AdaptiveVideoSource source(kFlowId, 30, 300, 0, 0);
+  PacedVideoSender sender(&uplink_, &source, bwe_type);
+
+  DefaultEvaluationFilter up_filter(&uplink_, kFlowId);
+  LinkShare link_share(&(up_filter.choke));
+  MetricRecorder metric_recorder(bwe_names[bwe_type], kFlowId, &sender,
+                                 &link_share);
+
+  PacketReceiver receiver(&uplink_, kFlowId, bwe_type, true, true,
+                          &metric_recorder);
+
+  metric_recorder.set_plot_available_capacity(plot_total_available_capacity_);
+
+  DelayFilter down_filter(&downlink_, kFlowId);
+  down_filter.SetOneWayDelayMs(kOneWayDelayMs);
+
+  // Test also with one way propagation delay = 100ms.
+  // up_filter.delay.SetOneWayDelayMs(100);
+  // down_filter.SetOneWayDelayMs(100);
+
+  up_filter.choke.set_capacity_kbps(1000);
+  RunFor(40 * 1000);  // 0-40s.
+  up_filter.choke.set_capacity_kbps(2500);
+  RunFor(20 * 1000);  // 40-60s.
+  up_filter.choke.set_capacity_kbps(600);
+  RunFor(20 * 1000);  // 60-80s.
+  up_filter.choke.set_capacity_kbps(1000);
+  RunFor(20 * 1000);  // 80-100s.
+
+  std::string title("5.1_Variable_capacity_single_flow");
+  metric_recorder.PlotThroughputHistogram(title, bwe_names[bwe_type], 1, 0);
+  metric_recorder.PlotDelayHistogram(title, bwe_names[bwe_type], 1,
+                                     kOneWayDelayMs);
+  metric_recorder.PlotLossHistogram(title, bwe_names[bwe_type], 1,
+                                    receiver.GlobalPacketLoss());
+  BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], kOneWayDelayMs, kFlowId);
+}
+
+// 5.2. Two forward direction competing flows, variable capacity.
+void BweTest::RunVariableCapacity2MultipleFlows(BandwidthEstimatorType bwe_type,
+                                                size_t num_flows) {
+  std::vector<VideoSource*> sources;
+  std::vector<PacketSender*> senders;
+  std::vector<MetricRecorder*> metric_recorders;
+  std::vector<PacketReceiver*> receivers;
+
+  const int64_t kStartingApartMs = 0;  // Flows initialized simultaneously.
+
+  for (size_t i = 0; i < num_flows; ++i) {
+    sources.push_back(new AdaptiveVideoSource(static_cast<int>(i), 30, 300, 0,
+                                              i * kStartingApartMs));
+    senders.push_back(new VideoSender(&uplink_, sources[i], bwe_type));
+  }
+
+  FlowIds flow_ids = CreateFlowIdRange(0, static_cast<int>(num_flows - 1));
+
+  DefaultEvaluationFilter up_filter(&uplink_, flow_ids);
+  LinkShare link_share(&(up_filter.choke));
+
+  RateCounterFilter total_utilization(&uplink_, flow_ids, "Total_utilization",
+                                      "Total_link_utilization");
+
+  // Delays is being plotted only for the first flow.
+  // To plot all of them, replace "i == 0" with "true" on new PacketReceiver().
+  for (size_t i = 0; i < num_flows; ++i) {
+    metric_recorders.push_back(new MetricRecorder(
+        bwe_names[bwe_type], static_cast<int>(i), senders[i], &link_share));
+
+    receivers.push_back(new PacketReceiver(&uplink_, static_cast<int>(i),
+                                           bwe_type, i == 0, false,
+                                           metric_recorders[i]));
+    metric_recorders[i]->set_plot_available_capacity(
+        i == 0 && plot_total_available_capacity_);
+  }
+
+  DelayFilter down_filter(&downlink_, flow_ids);
+  down_filter.SetOneWayDelayMs(kOneWayDelayMs);
+  // Test also with one way propagation delay = 100ms.
+  // up_filter.delay.SetOneWayDelayMs(100);
+  // down_filter.SetOneWayDelayMs(100);
+
+  up_filter.choke.set_capacity_kbps(4000);
+  RunFor(25 * 1000);  // 0-25s.
+  up_filter.choke.set_capacity_kbps(2000);
+  RunFor(25 * 1000);  // 25-50s.
+  up_filter.choke.set_capacity_kbps(3500);
+  RunFor(25 * 1000);  // 50-75s.
+  up_filter.choke.set_capacity_kbps(1000);
+  RunFor(25 * 1000);  // 75-100s.
+  up_filter.choke.set_capacity_kbps(2000);
+  RunFor(25 * 1000);  // 100-125s.
+
+  std::string title("5.2_Variable_capacity_two_flows");
+  for (size_t i = 0; i < num_flows; ++i) {
+    metric_recorders[i]->PlotThroughputHistogram(title, bwe_names[bwe_type],
+                                                 num_flows, 0);
+    metric_recorders[i]->PlotDelayHistogram(title, bwe_names[bwe_type],
+                                            num_flows, kOneWayDelayMs);
+    metric_recorders[i]->PlotLossHistogram(title, bwe_names[bwe_type],
+                                           num_flows,
+                                           receivers[i]->GlobalPacketLoss());
+    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], kOneWayDelayMs, i);
+  }
+
+  for (VideoSource* source : sources)
+    delete source;
+  for (PacketSender* sender : senders)
+    delete sender;
+  for (MetricRecorder* recorder : metric_recorders)
+    delete recorder;
+  for (PacketReceiver* receiver : receivers)
+    delete receiver;
+}
+
+// 5.3. Bi-directional RMCAT flows.
+void BweTest::RunBidirectionalFlow(BandwidthEstimatorType bwe_type) {
+  enum direction { kForward = 0, kBackward };
+  const size_t kNumFlows = 2;
+  rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumFlows];
+  rtc::scoped_ptr<VideoSender> senders[kNumFlows];
+  rtc::scoped_ptr<MetricRecorder> metric_recorders[kNumFlows];
+  rtc::scoped_ptr<PacketReceiver> receivers[kNumFlows];
+
+  sources[kForward].reset(new AdaptiveVideoSource(kForward, 30, 300, 0, 0));
+  senders[kForward].reset(
+      new VideoSender(&uplink_, sources[kForward].get(), bwe_type));
+
+  sources[kBackward].reset(new AdaptiveVideoSource(kBackward, 30, 300, 0, 0));
+  senders[kBackward].reset(
+      new VideoSender(&downlink_, sources[kBackward].get(), bwe_type));
+
+  DefaultEvaluationFilter up_filter(&uplink_, kForward);
+  LinkShare up_link_share(&(up_filter.choke));
+
+  metric_recorders[kForward].reset(new MetricRecorder(
+      bwe_names[bwe_type], kForward, senders[kForward].get(), &up_link_share));
+  receivers[kForward].reset(
+      new PacketReceiver(&uplink_, kForward, bwe_type, true, false,
+                         metric_recorders[kForward].get()));
+
+  metric_recorders[kForward].get()->set_plot_available_capacity(
+      plot_total_available_capacity_);
+
+  DefaultEvaluationFilter down_filter(&downlink_, kBackward);
+  LinkShare down_link_share(&(down_filter.choke));
+
+  metric_recorders[kBackward].reset(
+      new MetricRecorder(bwe_names[bwe_type], kBackward,
+                         senders[kBackward].get(), &down_link_share));
+  receivers[kBackward].reset(
+      new PacketReceiver(&downlink_, kBackward, bwe_type, true, false,
+                         metric_recorders[kBackward].get()));
+
+  metric_recorders[kBackward].get()->set_plot_available_capacity(
+      plot_total_available_capacity_);
+
+  // Test also with one way propagation delay = 100ms.
+  // up_filter.delay.SetOneWayDelayMs(100);
+  // down_filter.delay.SetOneWayDelayMs(100);
+
+  up_filter.choke.set_capacity_kbps(2000);
+  down_filter.choke.set_capacity_kbps(2000);
+  RunFor(20 * 1000);  // 0-20s.
+
+  up_filter.choke.set_capacity_kbps(1000);
+  RunFor(15 * 1000);  // 20-35s.
+
+  down_filter.choke.set_capacity_kbps(800);
+  RunFor(5 * 1000);  // 35-40s.
+
+  up_filter.choke.set_capacity_kbps(500);
+  RunFor(20 * 1000);  // 40-60s.
+
+  up_filter.choke.set_capacity_kbps(2000);
+  RunFor(10 * 1000);  // 60-70s.
+
+  down_filter.choke.set_capacity_kbps(2000);
+  RunFor(30 * 1000);  // 70-100s.
+
+  std::string title("5.3_Bidirectional_flows");
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    metric_recorders[i].get()->PlotThroughputHistogram(
+        title, bwe_names[bwe_type], kNumFlows, 0);
+    metric_recorders[i].get()->PlotDelayHistogram(title, bwe_names[bwe_type],
+                                                  kNumFlows, kOneWayDelayMs);
+    metric_recorders[i].get()->PlotLossHistogram(
+        title, bwe_names[bwe_type], kNumFlows,
+        receivers[i].get()->GlobalPacketLoss());
+    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], kOneWayDelayMs, i);
+  }
+}
+
+// 5.4. Three forward direction competing flows, constant capacity.
+void BweTest::RunSelfFairness(BandwidthEstimatorType bwe_type) {
+  const int kNumRmcatFlows = 3;
+  const int kNumTcpFlows = 0;
+  const int64_t kRunTimeS = 120;
+  const int kLinkCapacity = 3500;
+
+  int64_t max_delay_ms = kMaxQueueingDelayMs;
+  int64_t rtt_ms = 2 * kOneWayDelayMs;
+
+  const int64_t kStartingApartMs = 20 * 1000;
+  int64_t offsets_ms[kNumRmcatFlows];
+  for (int i = 0; i < kNumRmcatFlows; ++i) {
+    offsets_ms[i] = kStartingApartMs * i;
+  }
+
+  // Test also with one way propagation delay = 100ms.
+  // rtt_ms = 2 * 100;
+  // Test also with bottleneck queue size = 20ms and 1000ms.
+  // max_delay_ms = 20;
+  // max_delay_ms = 1000;
+
+  std::string title("5.4_Self_fairness_test");
+
+  // Test also with one way propagation delay = 100ms.
+  RunFairnessTest(bwe_type, kNumRmcatFlows, kNumTcpFlows, kRunTimeS,
+                  kLinkCapacity, max_delay_ms, rtt_ms, kMaxJitterMs, offsets_ms,
+                  title, bwe_names[bwe_type]);
+}
+
+// 5.5. Five competing RMCAT flows under different RTTs.
+void BweTest::RunRoundTripTimeFairness(BandwidthEstimatorType bwe_type) {
+  const int kAllFlowIds[] = {0, 1, 2, 3, 4};  // Five RMCAT flows.
+  const int64_t kAllOneWayDelayMs[] = {10, 25, 50, 100, 150};
+  const size_t kNumFlows = arraysize(kAllFlowIds);
+  rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumFlows];
+  rtc::scoped_ptr<VideoSender> senders[kNumFlows];
+  rtc::scoped_ptr<MetricRecorder> metric_recorders[kNumFlows];
+
+  // Flows initialized 10 seconds apart.
+  const int64_t kStartingApartMs = 10 * 1000;
+
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    sources[i].reset(new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0,
+                                             i * kStartingApartMs));
+    senders[i].reset(new VideoSender(&uplink_, sources[i].get(), bwe_type));
+  }
+
+  ChokeFilter choke_filter(&uplink_, CreateFlowIds(kAllFlowIds, kNumFlows));
+  LinkShare link_share(&choke_filter);
+
+  JitterFilter jitter_filter(&uplink_, CreateFlowIds(kAllFlowIds, kNumFlows));
+
+  rtc::scoped_ptr<DelayFilter> up_delay_filters[kNumFlows];
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    up_delay_filters[i].reset(new DelayFilter(&uplink_, kAllFlowIds[i]));
+  }
+
+  RateCounterFilter total_utilization(
+      &uplink_, CreateFlowIds(kAllFlowIds, kNumFlows), "Total_utilization",
+      "Total_link_utilization");
+
+  // Delays is being plotted only for the first flow.
+  // To plot all of them, replace "i == 0" with "true" on new PacketReceiver().
+  rtc::scoped_ptr<PacketReceiver> receivers[kNumFlows];
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    metric_recorders[i].reset(
+        new MetricRecorder(bwe_names[bwe_type], static_cast<int>(i),
+                           senders[i].get(), &link_share));
+
+    receivers[i].reset(new PacketReceiver(&uplink_, kAllFlowIds[i], bwe_type,
+                                          i == 0, false,
+                                          metric_recorders[i].get()));
+    metric_recorders[i].get()->set_start_computing_metrics_ms(kStartingApartMs *
+                                                              (kNumFlows - 1));
+    metric_recorders[i].get()->set_plot_available_capacity(
+        i == 0 && plot_total_available_capacity_);
+  }
+
+  rtc::scoped_ptr<DelayFilter> down_delay_filters[kNumFlows];
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    down_delay_filters[i].reset(new DelayFilter(&downlink_, kAllFlowIds[i]));
+  }
+
+  jitter_filter.SetMaxJitter(kMaxJitterMs);
+  choke_filter.set_max_delay_ms(kMaxQueueingDelayMs);
+
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    up_delay_filters[i]->SetOneWayDelayMs(kAllOneWayDelayMs[i]);
+    down_delay_filters[i]->SetOneWayDelayMs(kAllOneWayDelayMs[i]);
+  }
+
+  choke_filter.set_capacity_kbps(3500);
+
+  RunFor(300 * 1000);  // 0-300s.
+
+  std::string title("5.5_Round_Trip_Time_Fairness");
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    metric_recorders[i].get()->PlotThroughputHistogram(
+        title, bwe_names[bwe_type], kNumFlows, 0);
+    metric_recorders[i].get()->PlotDelayHistogram(title, bwe_names[bwe_type],
+                                                  kNumFlows, kOneWayDelayMs);
+    metric_recorders[i].get()->PlotLossHistogram(
+        title, bwe_names[bwe_type], kNumFlows,
+        receivers[i].get()->GlobalPacketLoss());
+    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], kAllOneWayDelayMs[i],
+                                 i);
+  }
+}
+
+// 5.6. RMCAT Flow competing with a long TCP Flow.
+void BweTest::RunLongTcpFairness(BandwidthEstimatorType bwe_type) {
+  const size_t kNumRmcatFlows = 1;
+  const size_t kNumTcpFlows = 1;
+  const int64_t kRunTimeS = 120;
+  const int kCapacityKbps = 2000;
+  // Tcp starts at t = 0, media flow at t = 5s.
+  const int64_t kOffSetsMs[] = {5000, 0};
+
+  int64_t max_delay_ms = kMaxQueueingDelayMs;
+  int64_t rtt_ms = 2 * kOneWayDelayMs;
+
+  // Test also with one way propagation delay = 100ms.
+  // rtt_ms = 2 * 100;
+  // Test also with bottleneck queue size = 20ms and 1000ms.
+  // max_delay_ms = 20;
+  // max_delay_ms = 1000;
+
+  std::string title("5.6_Long_TCP_Fairness");
+  std::string flow_name(bwe_names[bwe_type] + 'x' + bwe_names[kTcpEstimator]);
+
+  RunFairnessTest(bwe_type, kNumRmcatFlows, kNumTcpFlows, kRunTimeS,
+                  kCapacityKbps, max_delay_ms, rtt_ms, kMaxJitterMs, kOffSetsMs,
+                  title, flow_name);
+}
+
+// 5.7. RMCAT Flows competing with multiple short TCP Flows.
+void BweTest::RunMultipleShortTcpFairness(
+    BandwidthEstimatorType bwe_type,
+    std::vector<int> tcp_file_sizes_bytes,
+    std::vector<int64_t> tcp_starting_times_ms) {
+  // Two RMCAT flows and ten TCP flows.
+  const int kAllRmcatFlowIds[] = {0, 1};
+  const int kAllTcpFlowIds[] = {2, 3, 4, 5, 6, 7, 8, 9, 10, 11};
+
+  assert(tcp_starting_times_ms.size() == tcp_file_sizes_bytes.size() &&
+         tcp_starting_times_ms.size() == arraysize(kAllTcpFlowIds));
+
+  const size_t kNumRmcatFlows = arraysize(kAllRmcatFlowIds);
+  const size_t kNumTotalFlows = kNumRmcatFlows + arraysize(kAllTcpFlowIds);
+
+  rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumRmcatFlows];
+  rtc::scoped_ptr<PacketSender> senders[kNumTotalFlows];
+  rtc::scoped_ptr<MetricRecorder> metric_recorders[kNumTotalFlows];
+  rtc::scoped_ptr<PacketReceiver> receivers[kNumTotalFlows];
+
+  // RMCAT Flows are initialized simultaneosly at t=5 seconds.
+  const int64_t kRmcatStartingTimeMs = 5 * 1000;
+  for (size_t id : kAllRmcatFlowIds) {
+    sources[id].reset(new AdaptiveVideoSource(static_cast<int>(id), 30, 300, 0,
+                                              kRmcatStartingTimeMs));
+    senders[id].reset(new VideoSender(&uplink_, sources[id].get(), bwe_type));
+  }
+
+  for (size_t id : kAllTcpFlowIds) {
+    senders[id].reset(new TcpSender(&uplink_, static_cast<int>(id),
+                                    tcp_starting_times_ms[id - kNumRmcatFlows],
+                                    tcp_file_sizes_bytes[id - kNumRmcatFlows]));
+  }
+
+  FlowIds flow_ids = CreateFlowIdRange(0, static_cast<int>(kNumTotalFlows - 1));
+  DefaultEvaluationFilter up_filter(&uplink_, flow_ids);
+
+  LinkShare link_share(&(up_filter.choke));
+
+  RateCounterFilter total_utilization(&uplink_, flow_ids, "Total_utilization",
+                                      "Total_link_utilization");
+
+  // Delays is being plotted only for the first flow.
+  // To plot all of them, replace "i == 0" with "true" on new PacketReceiver().
+  for (size_t id : kAllRmcatFlowIds) {
+    metric_recorders[id].reset(
+        new MetricRecorder(bwe_names[bwe_type], static_cast<int>(id),
+                           senders[id].get(), &link_share));
+    receivers[id].reset(new PacketReceiver(&uplink_, static_cast<int>(id),
+                                           bwe_type, id == 0, false,
+                                           metric_recorders[id].get()));
+    metric_recorders[id].get()->set_start_computing_metrics_ms(
+        kRmcatStartingTimeMs);
+    metric_recorders[id].get()->set_plot_available_capacity(
+        id == 0 && plot_total_available_capacity_);
+  }
+
+  // Delays is not being plotted only for TCP flows. To plot all of them,
+  // replace first "false" occurence with "true" on new PacketReceiver().
+  for (size_t id : kAllTcpFlowIds) {
+    metric_recorders[id].reset(
+        new MetricRecorder(bwe_names[kTcpEstimator], static_cast<int>(id),
+                           senders[id].get(), &link_share));
+    receivers[id].reset(new PacketReceiver(&uplink_, static_cast<int>(id),
+                                           kTcpEstimator, false, false,
+                                           metric_recorders[id].get()));
+    metric_recorders[id].get()->set_plot_available_capacity(
+        id == 0 && plot_total_available_capacity_);
+  }
+
+  DelayFilter down_filter(&downlink_, flow_ids);
+  down_filter.SetOneWayDelayMs(kOneWayDelayMs);
+
+  // Test also with one way propagation delay = 100ms.
+  // up_filter.delay.SetOneWayDelayMs(100);
+  // down_filter.SetOneWayDelayms(100);
+
+  // Test also with bottleneck queue size = 20ms and 1000ms.
+  // up_filter.choke.set_max_delay_ms(20);
+  // up_filter.choke.set_max_delay_ms(1000);
+
+  // Test also with no Jitter:
+  // up_filter.jitter.SetMaxJitter(0);
+
+  up_filter.choke.set_capacity_kbps(2000);
+
+  RunFor(300 * 1000);  // 0-300s.
+
+  std::string title("5.7_Multiple_short_TCP_flows");
+  for (size_t id : kAllRmcatFlowIds) {
+    metric_recorders[id].get()->PlotThroughputHistogram(
+        title, bwe_names[bwe_type], kNumRmcatFlows, 0);
+    metric_recorders[id].get()->PlotDelayHistogram(
+        title, bwe_names[bwe_type], kNumRmcatFlows, kOneWayDelayMs);
+    metric_recorders[id].get()->PlotLossHistogram(
+        title, bwe_names[bwe_type], kNumRmcatFlows,
+        receivers[id].get()->GlobalPacketLoss());
+    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], kOneWayDelayMs, id);
+  }
+}
+
+// 5.8. Three forward direction competing flows, constant capacity.
+// During the test, one of the flows is paused and later resumed.
+void BweTest::RunPauseResumeFlows(BandwidthEstimatorType bwe_type) {
+  const int kAllFlowIds[] = {0, 1, 2};  // Three RMCAT flows.
+  const size_t kNumFlows = arraysize(kAllFlowIds);
+
+  rtc::scoped_ptr<AdaptiveVideoSource> sources[kNumFlows];
+  rtc::scoped_ptr<VideoSender> senders[kNumFlows];
+  rtc::scoped_ptr<MetricRecorder> metric_recorders[kNumFlows];
+  rtc::scoped_ptr<PacketReceiver> receivers[kNumFlows];
+
+  // Flows initialized simultaneously.
+  const int64_t kStartingApartMs = 0;
+
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    sources[i].reset(new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0,
+                                             i * kStartingApartMs));
+    senders[i].reset(new VideoSender(&uplink_, sources[i].get(), bwe_type));
+  }
+
+  DefaultEvaluationFilter filter(&uplink_,
+                                 CreateFlowIds(kAllFlowIds, kNumFlows));
+
+  LinkShare link_share(&(filter.choke));
+
+  RateCounterFilter total_utilization(
+      &uplink_, CreateFlowIds(kAllFlowIds, kNumFlows), "Total_utilization",
+      "Total_link_utilization");
+
+  // Delays is being plotted only for the first flow.
+  // To plot all of them, replace "i == 0" with "true" on new PacketReceiver().
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    metric_recorders[i].reset(
+        new MetricRecorder(bwe_names[bwe_type], static_cast<int>(i),
+                           senders[i].get(), &link_share));
+    receivers[i].reset(new PacketReceiver(&uplink_, kAllFlowIds[i], bwe_type,
+                                          i == 0, false,
+                                          metric_recorders[i].get()));
+    metric_recorders[i].get()->set_start_computing_metrics_ms(kStartingApartMs *
+                                                              (kNumFlows - 1));
+    metric_recorders[i].get()->set_plot_available_capacity(
+        i == 0 && plot_total_available_capacity_);
+  }
+
+  // Test also with one way propagation delay = 100ms.
+  // filter.delay.SetOneWayDelayMs(100);
+  filter.choke.set_capacity_kbps(3500);
+
+  RunFor(40 * 1000);  // 0-40s.
+  senders[0].get()->Pause();
+  RunFor(20 * 1000);  // 40-60s.
+  senders[0].get()->Resume(20 * 1000);
+  RunFor(60 * 1000);  // 60-120s.
+
+  int64_t paused[] = {20 * 1000, 0, 0};
+
+  // First flow is being paused, hence having a different optimum.
+  const std::string optima_lines[] = {"1", "2", "2"};
+
+  std::string title("5.8_Pause_and_resume_media_flow");
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    metric_recorders[i].get()->PlotThroughputHistogram(
+        title, bwe_names[bwe_type], kNumFlows, paused[i], optima_lines[i]);
+    metric_recorders[i].get()->PlotDelayHistogram(title, bwe_names[bwe_type],
+                                                  kNumFlows, kOneWayDelayMs);
+    metric_recorders[i].get()->PlotLossHistogram(
+        title, bwe_names[bwe_type], kNumFlows,
+        receivers[i].get()->GlobalPacketLoss());
+    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], kOneWayDelayMs, i);
+  }
+}
+
+// Following functions are used for randomizing TCP file size and
+// starting time, used on 5.7 RunMultipleShortTcpFairness.
+// They are pseudo-random generators, creating always the same
+// value sequence for a given Random seed.
+
+std::vector<int> BweTest::GetFileSizesBytes(int num_files) {
+  // File size chosen from uniform distribution between [100,1000] kB.
+  const int kMinKbytes = 100;
+  const int kMaxKbytes = 1000;
+
+  Random random(0x12345678);
+  std::vector<int> tcp_file_sizes_bytes;
+
+  while (num_files-- > 0) {
+    tcp_file_sizes_bytes.push_back(random.Rand(kMinKbytes, kMaxKbytes) * 1000);
+  }
+
+  return tcp_file_sizes_bytes;
+}
+
+std::vector<int64_t> BweTest::GetStartingTimesMs(int num_files) {
+  // OFF state behaves as an exp. distribution with mean = 10 seconds.
+  const float kMeanMs = 10000.0f;
+  Random random(0x12345678);
+
+  std::vector<int64_t> tcp_starting_times_ms;
+
+  // Two TCP Flows are initialized simultaneosly at t=0 seconds.
+  for (int i = 0; i < 2; ++i, --num_files) {
+    tcp_starting_times_ms.push_back(0);
+  }
+
+  // Other TCP Flows are initialized in an OFF state.
+  while (num_files-- > 0) {
+    tcp_starting_times_ms.push_back(
+        static_cast<int64_t>(random.Exponential(1.0f / kMeanMs)));
+  }
+
+  return tcp_starting_times_ms;
+}
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h
index 9dcb6bcb8f..5fb3252195 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h
@@ -17,6 +17,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 
 namespace webrtc {
@@ -66,8 +67,49 @@ class Link : public PacketProcessorListener {
 class BweTest {
  public:
   BweTest();
+  explicit BweTest(bool plot_capacity);
   ~BweTest();
 
+  void RunChoke(BandwidthEstimatorType bwe_type,
+                std::vector<int> capacities_kbps);
+
+  void RunVariableCapacity1SingleFlow(BandwidthEstimatorType bwe_type);
+  void RunVariableCapacity2MultipleFlows(BandwidthEstimatorType bwe_type,
+                                         size_t num_flows);
+  void RunBidirectionalFlow(BandwidthEstimatorType bwe_type);
+  void RunSelfFairness(BandwidthEstimatorType bwe_type);
+  void RunRoundTripTimeFairness(BandwidthEstimatorType bwe_type);
+  void RunLongTcpFairness(BandwidthEstimatorType bwe_type);
+  void RunMultipleShortTcpFairness(BandwidthEstimatorType bwe_type,
+                                   std::vector<int> tcp_file_sizes_bytes,
+                                   std::vector<int64_t> tcp_starting_times_ms);
+  void RunPauseResumeFlows(BandwidthEstimatorType bwe_type);
+
+  void RunFairnessTest(BandwidthEstimatorType bwe_type,
+                       size_t num_media_flows,
+                       size_t num_tcp_flows,
+                       int64_t run_time_seconds,
+                       uint32_t capacity_kbps,
+                       int64_t max_delay_ms,
+                       int64_t rtt_ms,
+                       int64_t max_jitter_ms,
+                       const int64_t* offsets_ms);
+
+  void RunFairnessTest(BandwidthEstimatorType bwe_type,
+                       size_t num_media_flows,
+                       size_t num_tcp_flows,
+                       int64_t run_time_seconds,
+                       uint32_t capacity_kbps,
+                       int64_t max_delay_ms,
+                       int64_t rtt_ms,
+                       int64_t max_jitter_ms,
+                       const int64_t* offsets_ms,
+                       const std::string& title,
+                       const std::string& flow_name);
+
+  static std::vector<int> GetFileSizesBytes(int num_files);
+  static std::vector<int64_t> GetStartingTimesMs(int num_files);
+
  protected:
   void SetUp();
 
@@ -75,6 +117,17 @@ class BweTest {
   void RunFor(int64_t time_ms);
   std::string GetTestName() const;
 
+  void PrintResults(double max_throughput_kbps,
+                    Stats<double> throughput_kbps,
+                    int flow_id,
+                    Stats<double> flow_delay_ms,
+                    Stats<double> flow_throughput_kbps);
+
+  void PrintResults(double max_throughput_kbps,
+                    Stats<double> throughput_kbps,
+                    std::map<int, Stats<double>> flow_delay_ms,
+                    std::map<int, Stats<double>> flow_throughput_kbps);
+
   Link downlink_;
   Link uplink_;
 
@@ -88,9 +141,53 @@ class BweTest {
   int64_t simulation_interval_ms_;
   std::vector<Link*> links_;
   Packets packets_;
+  bool plot_total_available_capacity_;
 
-  DISALLOW_COPY_AND_ASSIGN(BweTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweTest);
 };
+
+// Default Evaluation parameters:
+// Link capacity: 4000ms;
+// Queueing delay capacity: 300ms.
+// One-Way propagation delay: 50ms.
+// Jitter model: Truncated gaussian.
+// Maximum end-to-end jitter: 30ms = 2*standard_deviation.
+// Bottleneck queue type: Drop tail.
+// Path loss ratio: 0%.
+
+const int kOneWayDelayMs = 50;
+const int kMaxQueueingDelayMs = 300;
+const int kMaxCapacityKbps = 4000;
+const int kMaxJitterMs = 15;
+
+struct DefaultEvaluationFilter {
+  DefaultEvaluationFilter(PacketProcessorListener* listener, int flow_id)
+      : choke(listener, flow_id),
+        delay(listener, flow_id),
+        jitter(listener, flow_id) {
+    SetDefaultParameters();
+  }
+
+  DefaultEvaluationFilter(PacketProcessorListener* listener,
+                          const FlowIds& flow_ids)
+      : choke(listener, flow_ids),
+        delay(listener, flow_ids),
+        jitter(listener, flow_ids) {
+    SetDefaultParameters();
+  }
+
+  void SetDefaultParameters() {
+    delay.SetOneWayDelayMs(kOneWayDelayMs);
+    choke.set_max_delay_ms(kMaxQueueingDelayMs);
+    choke.set_capacity_kbps(kMaxCapacityKbps);
+    jitter.SetMaxJitter(kMaxJitterMs);
+  }
+
+  ChokeFilter choke;
+  DelayFilter delay;
+  JitterFilter jitter;
+};
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc
index dbb5ade0b2..d7abede707 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc
@@ -94,7 +94,7 @@ class BaseLineFileVerify : public BaseLineFileInterface {
   rtc::scoped_ptr<ResourceFileReader> reader_;
   bool fail_to_read_response_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(BaseLineFileVerify);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BaseLineFileVerify);
 };
 
 class BaseLineFileUpdate : public BaseLineFileInterface {
@@ -146,7 +146,7 @@ class BaseLineFileUpdate : public BaseLineFileInterface {
   std::vector<uint32_t> output_content_;
   std::string filepath_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(BaseLineFileUpdate);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BaseLineFileUpdate);
 };
 
 BaseLineFileInterface* BaseLineFileInterface::Create(
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
index 64dfa85535..b3df7124e3 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_BASELINEFILE_H_
 
 #include <string>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace testing {
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
index e73a545e53..d470324ac3 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
@@ -16,7 +16,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace testing {
@@ -35,7 +35,7 @@ class ResourceFileReader {
  private:
   explicit ResourceFileReader(FILE* file) : file_(file) {}
   FILE* file_;
-  DISALLOW_IMPLICIT_CONSTRUCTORS(ResourceFileReader);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ResourceFileReader);
 };
 
 class OutputFileWriter {
@@ -50,7 +50,7 @@ class OutputFileWriter {
  private:
   explicit OutputFileWriter(FILE* file) : file_(file) {}
   FILE* file_;
-  DISALLOW_IMPLICIT_CONSTRUCTORS(OutputFileWriter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OutputFileWriter);
 };
 }  // namespace bwe
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
index d5a95066c5..41bf836c9e 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
@@ -20,9 +20,10 @@ namespace bwe {
 
 class DelayCapHelper {
  public:
+  // Max delay = 0 stands for +infinite.
   DelayCapHelper() : max_delay_us_(0), delay_stats_() {}
 
-  void SetMaxDelay(int max_delay_ms) {
+  void set_max_delay_ms(int64_t max_delay_ms) {
     BWE_TEST_LOGGING_ENABLE(false);
     BWE_TEST_LOGGING_LOG1("Max Delay", "%d ms", static_cast<int>(max_delay_ms));
     assert(max_delay_ms >= 0);
@@ -31,7 +32,7 @@ class DelayCapHelper {
 
   bool ShouldSendPacket(int64_t send_time_us, int64_t arrival_time_us) {
     int64_t packet_delay_us = send_time_us - arrival_time_us;
-    delay_stats_.Push(std::min(packet_delay_us, max_delay_us_) / 1000);
+    delay_stats_.Push((std::min(packet_delay_us, max_delay_us_) + 500) / 1000);
     return (max_delay_us_ == 0 || max_delay_us_ >= packet_delay_us);
   }
 
@@ -43,7 +44,7 @@ class DelayCapHelper {
   int64_t max_delay_us_;
   Stats<double> delay_stats_;
 
-  DISALLOW_COPY_AND_ASSIGN(DelayCapHelper);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DelayCapHelper);
 };
 
 const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids) {
@@ -51,85 +52,62 @@ const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids) {
   return flow_ids;
 }
 
-class RateCounter {
- public:
-  RateCounter()
-      : kWindowSizeUs(1000000),
-        packets_per_second_(0),
-        bytes_per_second_(0),
-        last_accumulated_us_(0),
-        window_() {}
+const FlowIds CreateFlowIdRange(int initial_value, int last_value) {
+  int size = last_value - initial_value + 1;
+  assert(size > 0);
+  int* flow_ids_array = new int[size];
+  for (int i = initial_value; i <= last_value; ++i) {
+    flow_ids_array[i - initial_value] = i;
+  }
+  return CreateFlowIds(flow_ids_array, size);
+}
 
-  void UpdateRates(int64_t send_time_us, uint32_t payload_size) {
-    packets_per_second_++;
-    bytes_per_second_ += payload_size;
-    last_accumulated_us_ = send_time_us;
-    window_.push_back(std::make_pair(send_time_us, payload_size));
-    while (!window_.empty()) {
-      const TimeSizePair& packet = window_.front();
-      if (packet.first > (last_accumulated_us_ - kWindowSizeUs)) {
-        break;
-      }
-      assert(packets_per_second_ >= 1);
-      assert(bytes_per_second_ >= packet.second);
-      packets_per_second_--;
-      bytes_per_second_ -= packet.second;
-      window_.pop_front();
+void RateCounter::UpdateRates(int64_t send_time_us, uint32_t payload_size) {
+  ++recently_received_packets_;
+  recently_received_bytes_ += payload_size;
+  last_accumulated_us_ = send_time_us;
+  window_.push_back(std::make_pair(send_time_us, payload_size));
+  while (!window_.empty()) {
+    const TimeSizePair& packet = window_.front();
+    if (packet.first > (last_accumulated_us_ - window_size_us_)) {
+      break;
     }
+    assert(recently_received_packets_ >= 1);
+    assert(recently_received_bytes_ >= packet.second);
+    --recently_received_packets_;
+    recently_received_bytes_ -= packet.second;
+    window_.pop_front();
   }
-
-  uint32_t bits_per_second() const {
-    return bytes_per_second_ * 8;
-  }
-
-  uint32_t packets_per_second() const { return packets_per_second_; }
-
- private:
-  typedef std::pair<int64_t, uint32_t> TimeSizePair;
-
-  const int64_t kWindowSizeUs;
-  uint32_t packets_per_second_;
-  uint32_t bytes_per_second_;
-  int64_t last_accumulated_us_;
-  std::list<TimeSizePair> window_;
-};
-
-Random::Random(uint32_t seed)
-    : a_(0x531FDB97 ^ seed),
-      b_(0x6420ECA8 + seed) {
 }
 
-float Random::Rand() {
-  const float kScale = 1.0f / 0xffffffff;
-  float result = kScale * b_;
-  a_ ^= b_;
-  b_ += a_;
-  return result;
+uint32_t RateCounter::bits_per_second() const {
+  return (8 * recently_received_bytes_) / BitrateWindowS();
 }
 
-int Random::Gaussian(int mean, int standard_deviation) {
-  // Creating a Normal distribution variable from two independent uniform
-  // variables based on the Box-Muller transform, which is defined on the
-  // interval (0, 1], hence the mask+add below.
-  const double kPi = 3.14159265358979323846;
-  const double kScale = 1.0 / 0x80000000ul;
-  double u1 = kScale * ((a_ & 0x7ffffffful) + 1);
-  double u2 = kScale * ((b_ & 0x7ffffffful) + 1);
-  a_ ^= b_;
-  b_ += a_;
-  return static_cast<int>(mean + standard_deviation *
-      sqrt(-2 * log(u1)) * cos(2 * kPi * u2));
+uint32_t RateCounter::packets_per_second() const {
+  return recently_received_packets_ / BitrateWindowS();
+}
+
+double RateCounter::BitrateWindowS() const {
+  return static_cast<double>(window_size_us_) / (1000 * 1000);
 }
 
 Packet::Packet()
-    : flow_id_(0), creation_time_us_(-1), send_time_us_(-1), payload_size_(0) {
+    : flow_id_(0),
+      creation_time_us_(-1),
+      send_time_us_(-1),
+      sender_timestamp_us_(-1),
+      payload_size_(0),
+      paced_(false) {
 }
 
 Packet::Packet(int flow_id, int64_t send_time_us, size_t payload_size)
     : flow_id_(flow_id),
       creation_time_us_(send_time_us),
       send_time_us_(send_time_us),
-      payload_size_(payload_size) {
+      sender_timestamp_us_(send_time_us),
+      payload_size_(payload_size),
+      paced_(false) {
 }
 
 Packet::~Packet() {
@@ -148,6 +126,15 @@ MediaPacket::MediaPacket() {
   memset(&header_, 0, sizeof(header_));
 }
 
+MediaPacket::MediaPacket(int flow_id,
+                         int64_t send_time_us,
+                         size_t payload_size,
+                         uint16_t sequence_number)
+    : Packet(flow_id, send_time_us, payload_size) {
+  header_ = RTPHeader();
+  header_.sequenceNumber = sequence_number;
+}
+
 MediaPacket::MediaPacket(int flow_id,
                          int64_t send_time_us,
                          size_t payload_size,
@@ -155,9 +142,9 @@ MediaPacket::MediaPacket(int flow_id,
     : Packet(flow_id, send_time_us, payload_size), header_(header) {
 }
 
-MediaPacket::MediaPacket(int64_t send_time_us, uint32_t sequence_number)
+MediaPacket::MediaPacket(int64_t send_time_us, uint16_t sequence_number)
     : Packet(0, send_time_us, 0) {
-  memset(&header_, 0, sizeof(header_));
+  header_ = RTPHeader();
   header_.sequenceNumber = sequence_number;
 }
 
@@ -169,9 +156,10 @@ void MediaPacket::SetAbsSendTimeMs(int64_t abs_send_time_ms) {
 
 RembFeedback::RembFeedback(int flow_id,
                            int64_t send_time_us,
+                           int64_t last_send_time_ms,
                            uint32_t estimated_bps,
                            RTCPReportBlock report_block)
-    : FeedbackPacket(flow_id, send_time_us),
+    : FeedbackPacket(flow_id, send_time_us, last_send_time_ms),
       estimated_bps_(estimated_bps),
       report_block_(report_block) {
 }
@@ -179,8 +167,9 @@ RembFeedback::RembFeedback(int flow_id,
 SendSideBweFeedback::SendSideBweFeedback(
     int flow_id,
     int64_t send_time_us,
+    int64_t last_send_time_ms,
     const std::vector<PacketInfo>& packet_feedback_vector)
-    : FeedbackPacket(flow_id, send_time_us),
+    : FeedbackPacket(flow_id, send_time_us, last_send_time_ms),
       packet_feedback_vector_(packet_feedback_vector) {
 }
 
@@ -198,8 +187,7 @@ bool IsTimeSorted(const Packets& packets) {
 PacketProcessor::PacketProcessor(PacketProcessorListener* listener,
                                  int flow_id,
                                  ProcessorType type)
-    : listener_(listener) {
-  flow_ids_.insert(flow_id);
+    : listener_(listener), flow_ids_(&flow_id, &flow_id + 1) {
   if (listener_) {
     listener_->AddPacketProcessor(this, type);
   }
@@ -220,43 +208,60 @@ PacketProcessor::~PacketProcessor() {
   }
 }
 
+uint32_t PacketProcessor::packets_per_second() const {
+  return rate_counter_.packets_per_second();
+}
+
+uint32_t PacketProcessor::bits_per_second() const {
+  return rate_counter_.bits_per_second();
+}
+
 RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
                                      int flow_id,
-                                     const char* name)
+                                     const char* name,
+                                     const std::string& plot_name)
     : PacketProcessor(listener, flow_id, kRegular),
-      rate_counter_(new RateCounter()),
       packets_per_second_stats_(),
       kbps_stats_(),
-      name_(name) {
+      start_plotting_time_ms_(0),
+      plot_name_(plot_name) {
+  std::stringstream ss;
+  ss << name << "_" << flow_id;
+  name_ = ss.str();
 }
 
 RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
                                      const FlowIds& flow_ids,
-                                     const char* name)
+                                     const char* name,
+                                     const std::string& plot_name)
     : PacketProcessor(listener, flow_ids, kRegular),
-      rate_counter_(new RateCounter()),
       packets_per_second_stats_(),
       kbps_stats_(),
-      name_(name) {
+      start_plotting_time_ms_(0),
+      plot_name_(plot_name) {
   std::stringstream ss;
-  ss << name_ << "_";
+  ss << name;
+  char delimiter = '_';
   for (int flow_id : flow_ids) {
-    ss << flow_id << ",";
+    ss << delimiter << flow_id;
+    delimiter = ',';
   }
   name_ = ss.str();
 }
 
+RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
+                                     const FlowIds& flow_ids,
+                                     const char* name,
+                                     int64_t start_plotting_time_ms,
+                                     const std::string& plot_name)
+    : RateCounterFilter(listener, flow_ids, name, plot_name) {
+  start_plotting_time_ms_ = start_plotting_time_ms;
+}
+
 RateCounterFilter::~RateCounterFilter() {
   LogStats();
 }
 
-uint32_t RateCounterFilter::packets_per_second() const {
-  return rate_counter_->packets_per_second();
-}
-
-uint32_t RateCounterFilter::bits_per_second() const {
-  return rate_counter_->bits_per_second();
-}
 
 void RateCounterFilter::LogStats() {
   BWE_TEST_LOGGING_CONTEXT("RateCounterFilter");
@@ -269,18 +274,29 @@ Stats<double> RateCounterFilter::GetBitrateStats() const {
 }
 
 void RateCounterFilter::Plot(int64_t timestamp_ms) {
+  uint32_t plot_kbps = 0;
+  if (timestamp_ms >= start_plotting_time_ms_) {
+    plot_kbps = rate_counter_.bits_per_second() / 1000.0;
+  }
   BWE_TEST_LOGGING_CONTEXT(name_.c_str());
-  BWE_TEST_LOGGING_PLOT("Throughput_#1", timestamp_ms,
-                        rate_counter_->bits_per_second() / 1000.0);
+  if (plot_name_.empty()) {
+    BWE_TEST_LOGGING_PLOT(0, "Throughput_kbps#1", timestamp_ms, plot_kbps);
+  } else {
+    BWE_TEST_LOGGING_PLOT_WITH_NAME(0, "Throughput_kbps#1", timestamp_ms,
+                                    plot_kbps, plot_name_);
+  }
+
+  RTC_UNUSED(plot_kbps);
 }
 
 void RateCounterFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (const Packet* packet : *in_out) {
-    rate_counter_->UpdateRates(packet->send_time_us(), packet->payload_size());
+    rate_counter_.UpdateRates(packet->send_time_us(),
+                              static_cast<int>(packet->payload_size()));
   }
-  packets_per_second_stats_.Push(rate_counter_->packets_per_second());
-  kbps_stats_.Push(rate_counter_->bits_per_second() / 1000.0);
+  packets_per_second_stats_.Push(rate_counter_.packets_per_second());
+  kbps_stats_.Push(rate_counter_.bits_per_second() / 1000.0);
 }
 
 LossFilter::LossFilter(PacketProcessorListener* listener, int flow_id)
@@ -307,7 +323,7 @@ void LossFilter::SetLoss(float loss_percent) {
 void LossFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
-    if (random_.Rand() < loss_fraction_) {
+    if (random_.Rand<float>() < loss_fraction_) {
       delete *it;
       it = in_out->erase(it);
     } else {
@@ -316,30 +332,32 @@ void LossFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   }
 }
 
+const int64_t kDefaultOneWayDelayUs = 0;
+
 DelayFilter::DelayFilter(PacketProcessorListener* listener, int flow_id)
     : PacketProcessor(listener, flow_id, kRegular),
-      delay_us_(0),
+      one_way_delay_us_(kDefaultOneWayDelayUs),
       last_send_time_us_(0) {
 }
 
 DelayFilter::DelayFilter(PacketProcessorListener* listener,
                          const FlowIds& flow_ids)
     : PacketProcessor(listener, flow_ids, kRegular),
-      delay_us_(0),
+      one_way_delay_us_(kDefaultOneWayDelayUs),
       last_send_time_us_(0) {
 }
 
-void DelayFilter::SetDelay(int64_t delay_ms) {
+void DelayFilter::SetOneWayDelayMs(int64_t one_way_delay_ms) {
   BWE_TEST_LOGGING_ENABLE(false);
-  BWE_TEST_LOGGING_LOG1("Delay", "%d ms", static_cast<int>(delay_ms));
-  assert(delay_ms >= 0);
-  delay_us_ = delay_ms * 1000;
+  BWE_TEST_LOGGING_LOG1("Delay", "%d ms", static_cast<int>(one_way_delay_ms));
+  assert(one_way_delay_ms >= 0);
+  one_way_delay_us_ = one_way_delay_ms * 1000;
 }
 
 void DelayFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (Packet* packet : *in_out) {
-    int64_t new_send_time_us = packet->send_time_us() + delay_us_;
+    int64_t new_send_time_us = packet->send_time_us() + one_way_delay_us_;
     last_send_time_us_ = std::max(last_send_time_us_, new_send_time_us);
     packet->set_send_time_us(last_send_time_us_);
   }
@@ -349,7 +367,8 @@ JitterFilter::JitterFilter(PacketProcessorListener* listener, int flow_id)
     : PacketProcessor(listener, flow_id, kRegular),
       random_(0x89674523),
       stddev_jitter_us_(0),
-      last_send_time_us_(0) {
+      last_send_time_us_(0),
+      reordering_(false) {
 }
 
 JitterFilter::JitterFilter(PacketProcessorListener* listener,
@@ -357,27 +376,62 @@ JitterFilter::JitterFilter(PacketProcessorListener* listener,
     : PacketProcessor(listener, flow_ids, kRegular),
       random_(0x89674523),
       stddev_jitter_us_(0),
-      last_send_time_us_(0) {
+      last_send_time_us_(0),
+      reordering_(false) {
 }
 
-void JitterFilter::SetJitter(int64_t stddev_jitter_ms) {
+const int kN = 3;  // Truncated N sigma gaussian.
+
+void JitterFilter::SetMaxJitter(int64_t max_jitter_ms) {
   BWE_TEST_LOGGING_ENABLE(false);
-  BWE_TEST_LOGGING_LOG1("Jitter", "%d ms",
-                        static_cast<int>(stddev_jitter_ms));
-  assert(stddev_jitter_ms >= 0);
-  stddev_jitter_us_ = stddev_jitter_ms * 1000;
+  BWE_TEST_LOGGING_LOG1("Max Jitter", "%d ms", static_cast<int>(max_jitter_ms));
+  assert(max_jitter_ms >= 0);
+  // Truncated gaussian, Max jitter = kN*sigma.
+  stddev_jitter_us_ = (max_jitter_ms * 1000 + kN / 2) / kN;
+}
+
+namespace {
+inline int64_t TruncatedNSigmaGaussian(Random* const random,
+                                       int64_t mean,
+                                       int64_t std_dev) {
+  int64_t gaussian_random = random->Gaussian(mean, std_dev);
+  return std::max(std::min(gaussian_random, kN * std_dev), -kN * std_dev);
+}
 }
 
 void JitterFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (Packet* packet : *in_out) {
-    int64_t new_send_time_us = packet->send_time_us();
-    new_send_time_us += random_.Gaussian(0, stddev_jitter_us_);
-    last_send_time_us_ = std::max(last_send_time_us_, new_send_time_us);
-    packet->set_send_time_us(last_send_time_us_);
+    int64_t jitter_us =
+        std::abs(TruncatedNSigmaGaussian(&random_, 0, stddev_jitter_us_));
+    int64_t new_send_time_us = packet->send_time_us() + jitter_us;
+
+    if (!reordering_) {
+      new_send_time_us = std::max(last_send_time_us_, new_send_time_us);
+    }
+
+    // Receiver timestamp cannot be lower than sender timestamp.
+    assert(new_send_time_us >= packet->sender_timestamp_us());
+
+    packet->set_send_time_us(new_send_time_us);
+    last_send_time_us_ = new_send_time_us;
   }
 }
 
+// Computes the expected value for a right sided (abs) truncated gaussian.
+// Does not take into account  possible reoerdering updates.
+int64_t JitterFilter::MeanUs() {
+  const double kPi = 3.1415926535897932;
+  double max_jitter_us = static_cast<double>(kN * stddev_jitter_us_);
+  double right_sided_mean_us =
+      static_cast<double>(stddev_jitter_us_) / sqrt(kPi / 2.0);
+  double truncated_mean_us =
+      right_sided_mean_us *
+          (1.0 - exp(-pow(static_cast<double>(kN), 2.0) / 2.0)) +
+      max_jitter_us * erfc(static_cast<double>(kN));
+  return static_cast<int64_t>(truncated_mean_us + 0.5);
+}
+
 ReorderFilter::ReorderFilter(PacketProcessorListener* listener, int flow_id)
     : PacketProcessor(listener, flow_id, kRegular),
       random_(0x27452389),
@@ -405,7 +459,7 @@ void ReorderFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
     PacketsIt last_it = in_out->begin();
     PacketsIt it = last_it;
     while (++it != in_out->end()) {
-      if (random_.Rand() < reorder_fraction_) {
+      if (random_.Rand<float>() < reorder_fraction_) {
         int64_t t1 = (*last_it)->send_time_us();
         int64_t t2 = (*it)->send_time_us();
         std::swap(*last_it, *it);
@@ -417,9 +471,11 @@ void ReorderFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   }
 }
 
+const uint32_t kDefaultKbps = 1200;
+
 ChokeFilter::ChokeFilter(PacketProcessorListener* listener, int flow_id)
     : PacketProcessor(listener, flow_id, kRegular),
-      kbps_(1200),
+      capacity_kbps_(kDefaultKbps),
       last_send_time_us_(0),
       delay_cap_helper_(new DelayCapHelper()) {
 }
@@ -427,27 +483,34 @@ ChokeFilter::ChokeFilter(PacketProcessorListener* listener, int flow_id)
 ChokeFilter::ChokeFilter(PacketProcessorListener* listener,
                          const FlowIds& flow_ids)
     : PacketProcessor(listener, flow_ids, kRegular),
-      kbps_(1200),
+      capacity_kbps_(kDefaultKbps),
       last_send_time_us_(0),
       delay_cap_helper_(new DelayCapHelper()) {
 }
 
 ChokeFilter::~ChokeFilter() {}
 
-void ChokeFilter::SetCapacity(uint32_t kbps) {
+void ChokeFilter::set_capacity_kbps(uint32_t kbps) {
   BWE_TEST_LOGGING_ENABLE(false);
   BWE_TEST_LOGGING_LOG1("BitrateChoke", "%d kbps", kbps);
-  kbps_ = kbps;
+  capacity_kbps_ = kbps;
+}
+
+uint32_t ChokeFilter::capacity_kbps() {
+  return capacity_kbps_;
 }
 
 void ChokeFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
     int64_t earliest_send_time_us =
-        last_send_time_us_ +
-        ((*it)->payload_size() * 8 * 1000 + kbps_ / 2) / kbps_;
+        std::max(last_send_time_us_, (*it)->send_time_us());
+
     int64_t new_send_time_us =
-        std::max((*it)->send_time_us(), earliest_send_time_us);
+        earliest_send_time_us +
+        ((*it)->payload_size() * 8 * 1000 + capacity_kbps_ / 2) /
+            capacity_kbps_;
+
     if (delay_cap_helper_->ShouldSendPacket(new_send_time_us,
                                             (*it)->send_time_us())) {
       (*it)->set_send_time_us(new_send_time_us);
@@ -460,8 +523,8 @@ void ChokeFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   }
 }
 
-void ChokeFilter::SetMaxDelay(int max_delay_ms) {
-  delay_cap_helper_->SetMaxDelay(max_delay_ms);
+void ChokeFilter::set_max_delay_ms(int64_t max_delay_ms) {
+  delay_cap_helper_->set_max_delay_ms(max_delay_ms);
 }
 
 Stats<double> ChokeFilter::GetDelayStats() const {
@@ -523,7 +586,7 @@ bool TraceBasedDeliveryFilter::Init(const std::string& filename) {
     return false;
   }
   int64_t first_timestamp = -1;
-  while(!feof(trace_file)) {
+  while (!feof(trace_file)) {
     const size_t kMaxLineLength = 100;
     char line[kMaxLineLength];
     if (fgets(line, kMaxLineLength, trace_file)) {
@@ -549,7 +612,7 @@ void TraceBasedDeliveryFilter::Plot(int64_t timestamp_ms) {
   BWE_TEST_LOGGING_CONTEXT(name_.c_str());
   // This plots the max possible throughput of the trace-based delivery filter,
   // which will be reached if a packet sent on every packet slot of the trace.
-  BWE_TEST_LOGGING_PLOT("MaxThroughput_#1", timestamp_ms,
+  BWE_TEST_LOGGING_PLOT(0, "MaxThroughput_#1", timestamp_ms,
                         rate_counter_->bits_per_second() / 1000.0);
 }
 
@@ -578,8 +641,8 @@ void TraceBasedDeliveryFilter::RunFor(int64_t time_ms, Packets* in_out) {
   kbps_stats_.Push(rate_counter_->bits_per_second() / 1000.0);
 }
 
-void TraceBasedDeliveryFilter::SetMaxDelay(int max_delay_ms) {
-  delay_cap_helper_->SetMaxDelay(max_delay_ms);
+void TraceBasedDeliveryFilter::set_max_delay_ms(int64_t max_delay_ms) {
+  delay_cap_helper_->set_max_delay_ms(max_delay_ms);
 }
 
 Stats<double> TraceBasedDeliveryFilter::GetDelayStats() const {
@@ -617,8 +680,10 @@ VideoSource::VideoSource(int flow_id,
       frame_period_ms_(1000.0 / fps),
       bits_per_second_(1000 * kbps),
       frame_size_bytes_(bits_per_second_ / 8 / fps),
+      random_(0x12345678),
       flow_id_(flow_id),
       next_frame_ms_(first_frame_offset_ms),
+      next_frame_rand_ms_(0),
       now_ms_(0),
       prototype_header_() {
   memset(&prototype_header_, 0, sizeof(prototype_header_));
@@ -630,6 +695,10 @@ uint32_t VideoSource::NextFrameSize() {
   return frame_size_bytes_;
 }
 
+int64_t VideoSource::GetTimeUntilNextFrameMs() const {
+  return next_frame_ms_ + next_frame_rand_ms_ - now_ms_;
+}
+
 uint32_t VideoSource::NextPacketSize(uint32_t frame_size,
                                      uint32_t remaining_payload) {
   return std::min(kMaxPayloadSizeBytes, remaining_payload);
@@ -637,18 +706,31 @@ uint32_t VideoSource::NextPacketSize(uint32_t frame_size,
 
 void VideoSource::RunFor(int64_t time_ms, Packets* in_out) {
   assert(in_out);
+
   now_ms_ += time_ms;
   Packets new_packets;
+
   while (now_ms_ >= next_frame_ms_) {
-    prototype_header_.timestamp = kTimestampBase +
-        static_cast<uint32_t>(next_frame_ms_ * 90.0);
+    const int64_t kRandAmplitude = 2;
+    // A variance picked uniformly from {-1, 0, 1} ms is added to the frame
+    // timestamp.
+    next_frame_rand_ms_ = kRandAmplitude * (random_.Rand<float>() - 0.5);
+
+    // Ensure frame will not have a negative timestamp.
+    int64_t next_frame_ms =
+        std::max<int64_t>(next_frame_ms_ + next_frame_rand_ms_, 0);
+
+    prototype_header_.timestamp =
+        kTimestampBase + static_cast<uint32_t>(next_frame_ms * 90.0);
     prototype_header_.extension.transmissionTimeOffset = 0;
 
     // Generate new packets for this frame, all with the same timestamp,
     // but the payload size is capped, so if the whole frame doesn't fit in
     // one packet, we will see a number of equally sized packets followed by
     // one smaller at the tail.
-    int64_t send_time_us = next_frame_ms_ * 1000.0;
+
+    int64_t send_time_us = next_frame_ms * 1000.0;
+
     uint32_t frame_size = NextFrameSize();
     uint32_t payload_size = frame_size;
 
@@ -658,12 +740,14 @@ void VideoSource::RunFor(int64_t time_ms, Packets* in_out) {
       MediaPacket* new_packet =
           new MediaPacket(flow_id_, send_time_us, size, prototype_header_);
       new_packets.push_back(new_packet);
-      new_packet->SetAbsSendTimeMs(next_frame_ms_);
+      new_packet->SetAbsSendTimeMs(next_frame_ms);
+      new_packet->set_sender_timestamp_us(send_time_us);
       payload_size -= size;
     }
 
     next_frame_ms_ += frame_period_ms_;
   }
+
   in_out->merge(new_packets, DereferencingComparator<Packet>);
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
index 9e893de511..3bb9b95f4b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
@@ -17,19 +17,23 @@
 #include <algorithm>
 #include <list>
 #include <numeric>
+#include <set>
 #include <sstream>
 #include <string>
+#include <utility>
 #include <vector>
 
+#include "webrtc/base/common.h"
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -39,10 +43,39 @@ namespace testing {
 namespace bwe {
 
 class DelayCapHelper;
-class RateCounter;
+
+class RateCounter {
+ public:
+  explicit RateCounter(int64_t window_size_ms)
+      : window_size_us_(1000 * window_size_ms),
+        recently_received_packets_(0),
+        recently_received_bytes_(0),
+        last_accumulated_us_(0),
+        window_() {}
+
+  RateCounter() : RateCounter(1000) {}
+
+  void UpdateRates(int64_t send_time_us, uint32_t payload_size);
+
+  int64_t window_size_ms() const { return (window_size_us_ + 500) / 1000; }
+  uint32_t packets_per_second() const;
+  uint32_t bits_per_second() const;
+
+  double BitrateWindowS() const;
+
+ private:
+  typedef std::pair<int64_t, uint32_t> TimeSizePair;
+
+  int64_t window_size_us_;
+  uint32_t recently_received_packets_;
+  uint32_t recently_received_bytes_;
+  int64_t last_accumulated_us_;
+  std::list<TimeSizePair> window_;
+};
 
 typedef std::set<int> FlowIds;
 const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids);
+const FlowIds CreateFlowIdRange(int initial_value, int last_value);
 
 template <typename T>
 bool DereferencingComparator(const T* const& a, const T* const& b) {
@@ -142,26 +175,6 @@ template<typename T> class Stats {
   T max_;
 };
 
-class Random {
- public:
-  explicit Random(uint32_t seed);
-
-  // Return pseudo random number in the interval [0.0, 1.0].
-  float Rand();
-
-  // Normal Distribution.
-  int Gaussian(int mean, int standard_deviation);
-
-  // TODO(solenberg): Random from histogram.
-  // template<typename T> int Distribution(const std::vector<T> histogram) {
-
- private:
-  uint32_t a_;
-  uint32_t b_;
-
-  DISALLOW_IMPLICIT_CONSTRUCTORS(Random);
-};
-
 bool IsTimeSorted(const Packets& packets);
 
 class PacketProcessor;
@@ -191,45 +204,57 @@ class PacketProcessor {
   // internal data.
   virtual void Plot(int64_t timestamp_ms) {}
 
-  // Run simulation for |time_ms| micro seconds, consuming packets from, and
+  // Run simulation for |time_ms| milliseconds, consuming packets from, and
   // producing packets into in_out. The outgoing packet list must be sorted on
   // |send_time_us_|. The simulation time |time_ms| is optional to use.
   virtual void RunFor(int64_t time_ms, Packets* in_out) = 0;
 
   const FlowIds& flow_ids() const { return flow_ids_; }
 
+  uint32_t packets_per_second() const;
+  uint32_t bits_per_second() const;
+
+ protected:
+  RateCounter rate_counter_;
+
  private:
   PacketProcessorListener* listener_;
-  FlowIds flow_ids_;
+  const FlowIds flow_ids_;
 
-  DISALLOW_COPY_AND_ASSIGN(PacketProcessor);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PacketProcessor);
 };
 
 class RateCounterFilter : public PacketProcessor {
  public:
   RateCounterFilter(PacketProcessorListener* listener,
                     int flow_id,
-                    const char* name);
+                    const char* name,
+                    const std::string& plot_name);
   RateCounterFilter(PacketProcessorListener* listener,
                     const FlowIds& flow_ids,
-                    const char* name);
+                    const char* name,
+                    const std::string& plot_name);
+  RateCounterFilter(PacketProcessorListener* listener,
+                    const FlowIds& flow_ids,
+                    const char* name,
+                    int64_t start_plotting_time_ms,
+                    const std::string& plot_name);
   virtual ~RateCounterFilter();
 
-  uint32_t packets_per_second() const;
-  uint32_t bits_per_second() const;
-
   void LogStats();
   Stats<double> GetBitrateStats() const;
   virtual void Plot(int64_t timestamp_ms);
   virtual void RunFor(int64_t time_ms, Packets* in_out);
 
  private:
-  rtc::scoped_ptr<RateCounter> rate_counter_;
   Stats<double> packets_per_second_stats_;
   Stats<double> kbps_stats_;
   std::string name_;
+  int64_t start_plotting_time_ms_;
+  // Algorithm name if single flow, Total link utilization if all flows.
+  std::string plot_name_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RateCounterFilter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RateCounterFilter);
 };
 
 class LossFilter : public PacketProcessor {
@@ -245,7 +270,7 @@ class LossFilter : public PacketProcessor {
   Random random_;
   float loss_fraction_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(LossFilter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LossFilter);
 };
 
 class DelayFilter : public PacketProcessor {
@@ -254,14 +279,14 @@ class DelayFilter : public PacketProcessor {
   DelayFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
   virtual ~DelayFilter() {}
 
-  void SetDelay(int64_t delay_ms);
+  void SetOneWayDelayMs(int64_t one_way_delay_ms);
   virtual void RunFor(int64_t time_ms, Packets* in_out);
 
  private:
-  int64_t delay_us_;
+  int64_t one_way_delay_us_;
   int64_t last_send_time_us_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(DelayFilter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DelayFilter);
 };
 
 class JitterFilter : public PacketProcessor {
@@ -270,15 +295,18 @@ class JitterFilter : public PacketProcessor {
   JitterFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
   virtual ~JitterFilter() {}
 
-  void SetJitter(int64_t stddev_jitter_ms);
+  void SetMaxJitter(int64_t stddev_jitter_ms);
   virtual void RunFor(int64_t time_ms, Packets* in_out);
+  void set_reorderdering(bool reordering) { reordering_ = reordering; }
+  int64_t MeanUs();
 
  private:
   Random random_;
   int64_t stddev_jitter_us_;
   int64_t last_send_time_us_;
+  bool reordering_;  // False by default.
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(JitterFilter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(JitterFilter);
 };
 
 // Reorders two consecutive packets with a probability of reorder_percent.
@@ -295,7 +323,7 @@ class ReorderFilter : public PacketProcessor {
   Random random_;
   float reorder_fraction_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(ReorderFilter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ReorderFilter);
 };
 
 // Apply a bitrate choke with an infinite queue on the packet stream.
@@ -305,18 +333,21 @@ class ChokeFilter : public PacketProcessor {
   ChokeFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
   virtual ~ChokeFilter();
 
-  void SetCapacity(uint32_t kbps);
-  void SetMaxDelay(int max_delay_ms);
+  void set_capacity_kbps(uint32_t kbps);
+  void set_max_delay_ms(int64_t max_queueing_delay_ms);
+
+  uint32_t capacity_kbps();
+
   virtual void RunFor(int64_t time_ms, Packets* in_out);
 
   Stats<double> GetDelayStats() const;
 
  private:
-  uint32_t kbps_;
+  uint32_t capacity_kbps_;
   int64_t last_send_time_us_;
   rtc::scoped_ptr<DelayCapHelper> delay_cap_helper_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(ChokeFilter);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ChokeFilter);
 };
 
 class TraceBasedDeliveryFilter : public PacketProcessor {
@@ -336,7 +367,7 @@ class TraceBasedDeliveryFilter : public PacketProcessor {
   virtual void Plot(int64_t timestamp_ms);
   virtual void RunFor(int64_t time_ms, Packets* in_out);
 
-  void SetMaxDelay(int max_delay_ms);
+  void set_max_delay_ms(int64_t max_delay_ms);
   Stats<double> GetDelayStats() const;
   Stats<double> GetBitrateStats() const;
 
@@ -354,7 +385,7 @@ class TraceBasedDeliveryFilter : public PacketProcessor {
   Stats<double> packets_per_second_stats_;
   Stats<double> kbps_stats_;
 
-  DISALLOW_COPY_AND_ASSIGN(TraceBasedDeliveryFilter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TraceBasedDeliveryFilter);
 };
 
 class VideoSource {
@@ -372,7 +403,7 @@ class VideoSource {
   virtual void SetBitrateBps(int bitrate_bps) {}
   uint32_t bits_per_second() const { return bits_per_second_; }
   uint32_t max_payload_size_bytes() const { return kMaxPayloadSizeBytes; }
-  int64_t GetTimeUntilNextFrameMs() const { return next_frame_ms_ - now_ms_; }
+  int64_t GetTimeUntilNextFrameMs() const;
 
  protected:
   virtual uint32_t NextFrameSize();
@@ -386,12 +417,14 @@ class VideoSource {
   uint32_t frame_size_bytes_;
 
  private:
+  Random random_;
   const int flow_id_;
   int64_t next_frame_ms_;
+  int64_t next_frame_rand_ms_;
   int64_t now_ms_;
   RTPHeader prototype_header_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(VideoSource);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VideoSource);
 };
 
 class AdaptiveVideoSource : public VideoSource {
@@ -406,7 +439,7 @@ class AdaptiveVideoSource : public VideoSource {
   void SetBitrateBps(int bitrate_bps) override;
 
  private:
-  DISALLOW_IMPLICIT_CONSTRUCTORS(AdaptiveVideoSource);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AdaptiveVideoSource);
 };
 
 class PeriodicKeyFrameSource : public AdaptiveVideoSource {
@@ -429,7 +462,7 @@ class PeriodicKeyFrameSource : public AdaptiveVideoSource {
   uint32_t frame_counter_;
   int compensation_bytes_;
   int compensation_per_frame_;
-  DISALLOW_IMPLICIT_CONSTRUCTORS(PeriodicKeyFrameSource);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PeriodicKeyFrameSource);
 };
 }  // namespace bwe
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
index c611b073b6..6bdfa847df 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
@@ -14,48 +14,14 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
-using std::vector;
-
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
-TEST(BweTestFramework_RandomTest, Gaussian) {
-  enum {
-    kN = 100000,
-    kBuckets = 100,
-    kMean = 49,
-    kStddev = 10
-  };
-
-  Random random(0x12345678);
-
-  int buckets[kBuckets] = {0};
-  for (int i = 0; i < kN; ++i) {
-    int index = random.Gaussian(kMean, kStddev);
-    if (index >= 0 && index < kBuckets) {
-      buckets[index]++;
-    }
-  }
-
-  const double kPi = 3.14159265358979323846;
-  const double kScale = kN / (kStddev * sqrt(2.0 * kPi));
-  const double kDiv = -2.0 * kStddev * kStddev;
-  double self_corr = 0.0;
-  double bucket_corr = 0.0;
-  for (int n = 0; n < kBuckets; ++n) {
-    double normal_dist = kScale * exp((n - kMean) * (n - kMean) / kDiv);
-    self_corr += normal_dist * normal_dist;
-    bucket_corr += normal_dist * buckets[n];
-  }
-  printf("Correlation: %f (random sample), %f (self), %f (quotient)\n",
-         bucket_corr, self_corr, bucket_corr / self_corr);
-  EXPECT_NEAR(1.0, bucket_corr / self_corr, 0.0004);
-}
-
 static bool IsSequenceNumberSorted(const Packets& packets) {
   PacketsConstIt last_it = packets.begin();
   for (PacketsConstIt it = last_it; it != packets.end(); ++it) {
@@ -182,7 +148,8 @@ TEST(BweTestFramework_StatsTest, MinMax) {
 
 class BweTestFramework_RateCounterFilterTest : public ::testing::Test {
  public:
-  BweTestFramework_RateCounterFilterTest() : filter_(NULL, 0, ""), now_ms_(0) {}
+  BweTestFramework_RateCounterFilterTest()
+      : filter_(NULL, 0, "", ""), now_ms_(0) {}
   virtual ~BweTestFramework_RateCounterFilterTest() {}
 
  protected:
@@ -208,7 +175,7 @@ class BweTestFramework_RateCounterFilterTest : public ::testing::Test {
   RateCounterFilter filter_;
   int64_t now_ms_;
 
-  DISALLOW_COPY_AND_ASSIGN(BweTestFramework_RateCounterFilterTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweTestFramework_RateCounterFilterTest);
 };
 
 TEST_F(BweTestFramework_RateCounterFilterTest, Short) {
@@ -333,7 +300,7 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
   }
 
   void TestDelayFilter(int64_t delay_ms) {
-    filter_.SetDelay(delay_ms);
+    filter_.SetOneWayDelayMs(delay_ms);
     TestDelayFilter(1, 0, 0);    // No input should yield no output
 
     // Single packet
@@ -341,7 +308,7 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
     TestDelayFilter(delay_ms, 0, 0);
 
     for (int i = 0; i < delay_ms; ++i) {
-      filter_.SetDelay(i);
+      filter_.SetOneWayDelayMs(i);
       TestDelayFilter(1, 10, 10);
     }
     TestDelayFilter(0, 0, 0);
@@ -351,11 +318,11 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
     TestDelayFilter(delay_ms, 0, 0);
 
     for (int i = 1; i < delay_ms + 1; ++i) {
-      filter_.SetDelay(i);
+      filter_.SetOneWayDelayMs(i);
       TestDelayFilter(1, 5, 5);
     }
     TestDelayFilter(0, 0, 0);
-    filter_.SetDelay(2 * delay_ms);
+    filter_.SetOneWayDelayMs(2 * delay_ms);
     TestDelayFilter(1, 0, 0);
     TestDelayFilter(delay_ms, 13, 13);
     TestDelayFilter(delay_ms, 0, 0);
@@ -364,11 +331,11 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
     TestDelayFilter(delay_ms, 0, 0);
 
     for (int i = 0; i < 2 * delay_ms; ++i) {
-      filter_.SetDelay(2 * delay_ms - i - 1);
+      filter_.SetOneWayDelayMs(2 * delay_ms - i - 1);
       TestDelayFilter(1, 5, 5);
     }
     TestDelayFilter(0, 0, 0);
-    filter_.SetDelay(0);
+    filter_.SetOneWayDelayMs(0);
     TestDelayFilter(0, 7, 7);
 
     ASSERT_TRUE(IsTimeSorted(accumulated_packets_));
@@ -380,16 +347,16 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
 
  private:
   int64_t now_ms_;
-  uint32_t sequence_number_;
+  uint16_t sequence_number_;
 
-  DISALLOW_COPY_AND_ASSIGN(BweTestFramework_DelayFilterTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweTestFramework_DelayFilterTest);
 };
 
 TEST_F(BweTestFramework_DelayFilterTest, Delay0) {
   TestDelayFilter(1, 0, 0);    // No input should yield no output
   TestDelayFilter(1, 10, 10);  // Expect no delay (delay time is zero)
   TestDelayFilter(1, 0, 0);    // Check no packets are still in buffer
-  filter_.SetDelay(0);
+  filter_.SetOneWayDelayMs(0);
   TestDelayFilter(1, 5, 5);    // Expect no delay (delay time is zero)
   TestDelayFilter(1, 0, 0);    // Check no packets are still in buffer
 }
@@ -416,7 +383,7 @@ TEST_F(BweTestFramework_DelayFilterTest, JumpToZeroDelay) {
   Packets packets;
 
   // Delay a bunch of packets, accumulate them to the 'acc' list.
-  delay.SetDelay(100.0f);
+  delay.SetOneWayDelayMs(100.0f);
   for (uint32_t i = 0; i < 10; ++i) {
     packets.push_back(new MediaPacket(i * 100, i));
   }
@@ -427,7 +394,7 @@ TEST_F(BweTestFramework_DelayFilterTest, JumpToZeroDelay) {
 
   // Drop delay to zero, send a few more packets through the delay, append them
   // to the 'acc' list and verify that it is all sorted.
-  delay.SetDelay(0.0f);
+  delay.SetOneWayDelayMs(0.0f);
   for (uint32_t i = 10; i < 50; ++i) {
     packets.push_back(new MediaPacket(i * 100, i));
   }
@@ -446,24 +413,24 @@ TEST_F(BweTestFramework_DelayFilterTest, IncreasingDelay) {
     TestDelayFilter(i);
   }
   // Reach a steady state.
-  filter_.SetDelay(100);
+  filter_.SetOneWayDelayMs(100);
   TestDelayFilter(1, 20, 20);
   TestDelayFilter(2, 0, 0);
   TestDelayFilter(99, 20, 20);
   // Drop delay back down to zero.
-  filter_.SetDelay(0);
+  filter_.SetOneWayDelayMs(0);
   TestDelayFilter(1, 100, 100);
   TestDelayFilter(23010, 0, 0);
   ASSERT_TRUE(IsTimeSorted(accumulated_packets_));
   ASSERT_TRUE(IsSequenceNumberSorted(accumulated_packets_));
 }
 
-static void TestJitterFilter(int64_t stddev_jitter_ms) {
+static void TestJitterFilter(int64_t max_jitter_ms) {
   JitterFilter filter(NULL, 0);
-  filter.SetJitter(stddev_jitter_ms);
+  filter.SetMaxJitter(max_jitter_ms);
 
   int64_t now_ms = 0;
-  uint32_t sequence_number = 0;
+  uint16_t sequence_number = 0;
 
   // Generate packets, add jitter to them, accumulate the altered packets.
   Packets original;
@@ -474,9 +441,9 @@ static void TestJitterFilter(int64_t stddev_jitter_ms) {
       packets.push_back(new MediaPacket(now_ms * 1000, sequence_number));
       original.push_back(new MediaPacket(now_ms * 1000, sequence_number));
       ++sequence_number;
-      now_ms += 5 * stddev_jitter_ms;
+      now_ms += 5 * max_jitter_ms;
     }
-    filter.RunFor(stddev_jitter_ms, &packets);
+    filter.RunFor(max_jitter_ms, &packets);
     jittered.splice(jittered.end(), packets);
   }
 
@@ -491,17 +458,22 @@ static void TestJitterFilter(int64_t stddev_jitter_ms) {
   // difference (jitter) in stats, then check that mean jitter is close to zero
   // and standard deviation of jitter is what we set it to.
   Stats<double> jitter_us;
+  int64_t max_jitter_obtained_us = 0;
   for (PacketsIt it1 = original.begin(), it2 = jittered.begin();
        it1 != original.end() && it2 != jittered.end(); ++it1, ++it2) {
     const MediaPacket* packet1 = static_cast<const MediaPacket*>(*it1);
     const MediaPacket* packet2 = static_cast<const MediaPacket*>(*it2);
     EXPECT_EQ(packet1->header().sequenceNumber,
               packet2->header().sequenceNumber);
-    jitter_us.Push(packet1->send_time_us() - packet2->send_time_us());
+    max_jitter_obtained_us =
+        std::max(max_jitter_obtained_us,
+                 packet2->send_time_us() - packet1->send_time_us());
+    jitter_us.Push(packet2->send_time_us() - packet1->send_time_us());
   }
-  EXPECT_NEAR(0.0, jitter_us.GetMean(), stddev_jitter_ms * 1000.0 * 0.008);
-  EXPECT_NEAR(stddev_jitter_ms * 1000.0, jitter_us.GetStdDev(),
-              stddev_jitter_ms * 1000.0 * 0.02);
+  EXPECT_NEAR(filter.MeanUs(), jitter_us.GetMean(),
+              max_jitter_ms * 1000.0 * 0.01);
+  EXPECT_NEAR(max_jitter_ms * 1000.0, max_jitter_obtained_us,
+              max_jitter_ms * 1000.0 * 0.01);
   for (auto* packet : original)
     delete packet;
   for (auto* packet : jittered)
@@ -528,14 +500,14 @@ TEST(BweTestFramework_JitterFilterTest, Jitter1031) {
   TestJitterFilter(1031);
 }
 
-static void TestReorderFilter(uint32_t reorder_percent, uint32_t near_value) {
-  const uint32_t kPacketCount = 10000;
+static void TestReorderFilter(uint16_t reorder_percent) {
+  const uint16_t kPacketCount = 10000;
 
   // Generate packets with 10 ms interval.
   Packets packets;
   int64_t now_ms = 0;
-  uint32_t sequence_number = 1;
-  for (uint32_t i = 0; i < kPacketCount; ++i, now_ms += 10) {
+  uint16_t sequence_number = 1;
+  for (uint16_t i = 0; i < kPacketCount; ++i, now_ms += 10) {
     packets.push_back(new MediaPacket(now_ms * 1000, sequence_number++));
   }
   ASSERT_TRUE(IsTimeSorted(packets));
@@ -549,21 +521,28 @@ static void TestReorderFilter(uint32_t reorder_percent, uint32_t near_value) {
 
   // We measure the amount of reordering by summing the distance by which out-
   // of-order packets have been moved in the stream.
-  uint32_t distance = 0;
-  uint32_t last_sequence_number = 0;
+  uint16_t distance = 0;
+  uint16_t last_sequence_number = 0;
   for (auto* packet : packets) {
     const MediaPacket* media_packet = static_cast<const MediaPacket*>(packet);
-    uint32_t sequence_number = media_packet->header().sequenceNumber;
+    uint16_t sequence_number = media_packet->header().sequenceNumber;
+    // The expected position for sequence number s is in position s-1.
     if (sequence_number < last_sequence_number) {
       distance += last_sequence_number - sequence_number;
     }
     last_sequence_number = sequence_number;
   }
 
-  // Because reordering is random, we allow a threshold when comparing. The
-  // maximum distance a packet can be moved is PacketCount - 1.
-  EXPECT_NEAR(
-    ((kPacketCount - 1) * reorder_percent) / 100, distance, near_value);
+  // The probability that two elements are swapped is p = reorder_percent / 100.
+  double p = static_cast<double>(reorder_percent) / 100;
+  // The expected number of swaps we perform is p * (PacketCount - 1),
+  // and each swap increases the distance by one.
+  double mean = p * (kPacketCount - 1);
+  // If pair i is chosen to be swapped with probability p, the variance for that
+  // pair is p * (1 - p). Since there are (kPacketCount - 1) independent pairs,
+  // the variance for the number of swaps is (kPacketCount - 1) * p * (1 - p).
+  double std_deviation = sqrt((kPacketCount - 1) * p * (1 - p));
+  EXPECT_NEAR(mean, distance, 3 * std_deviation);
 
   for (auto* packet : packets)
     delete packet;
@@ -571,23 +550,23 @@ static void TestReorderFilter(uint32_t reorder_percent, uint32_t near_value) {
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder0) {
   // For 0% reordering, no packets should have been moved, so result is exact.
-  TestReorderFilter(0, 0);
+  TestReorderFilter(0);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder10) {
-  TestReorderFilter(10, 30);
+  TestReorderFilter(10);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder20) {
-  TestReorderFilter(20, 20);
+  TestReorderFilter(20);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder50) {
-  TestReorderFilter(50, 20);
+  TestReorderFilter(50);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder70) {
-  TestReorderFilter(70, 20);
+  TestReorderFilter(70);
 }
 
 TEST(BweTestFramework_ReorderFilterTest, Reorder100) {
@@ -595,7 +574,7 @@ TEST(BweTestFramework_ReorderFilterTest, Reorder100) {
   // adjacent packets, when the likelihood of a swap is 1.0, a swap will always
   // occur, so the stream will be in order except for the first packet, which
   // has been moved to the end. Therefore we expect the result to be exact here.
-  TestReorderFilter(100.0, 0);
+  TestReorderFilter(100.0);
 }
 
 class BweTestFramework_ChokeFilterTest : public ::testing::Test {
@@ -615,7 +594,7 @@ class BweTestFramework_ChokeFilterTest : public ::testing::Test {
   void TestChoke(PacketProcessor* filter,
                  int64_t run_for_ms,
                  uint32_t packets_to_generate,
-                 uint32_t expected_kbit_transmitted) {
+                 size_t expected_kbit_transmitted) {
     // Generate a bunch of packets, apply choke, verify output is ordered.
     Packets packets;
     RTPHeader header;
@@ -634,7 +613,7 @@ class BweTestFramework_ChokeFilterTest : public ::testing::Test {
     ASSERT_TRUE(IsSequenceNumberSorted(output_packets_));
 
     // Sum up the transmitted bytes up until the current time.
-    uint32_t bytes_transmitted = 0;
+    size_t bytes_transmitted = 0;
     while (!output_packets_.empty()) {
       const Packet* packet = output_packets_.front();
       if (packet->send_time_us() > now_ms_ * 1000) {
@@ -644,7 +623,7 @@ class BweTestFramework_ChokeFilterTest : public ::testing::Test {
       delete output_packets_.front();
       output_packets_.pop_front();
     }
-    EXPECT_EQ(expected_kbit_transmitted, (bytes_transmitted * 8) / 1000);
+    EXPECT_EQ(expected_kbit_transmitted, (bytes_transmitted * 8 + 500) / 1000);
   }
 
   void CheckMaxDelay(int64_t max_delay_ms) {
@@ -658,26 +637,56 @@ class BweTestFramework_ChokeFilterTest : public ::testing::Test {
 
  private:
   int64_t now_ms_;
-  uint32_t sequence_number_;
+  uint16_t sequence_number_;
   Packets output_packets_;
   std::vector<int64_t> send_times_us_;
 
-  DISALLOW_COPY_AND_ASSIGN(BweTestFramework_ChokeFilterTest);
+  RTC_DISALLOW_COPY_AND_ASSIGN(BweTestFramework_ChokeFilterTest);
 };
 
+TEST_F(BweTestFramework_ChokeFilterTest, NoQueue) {
+  const int kCapacityKbps = 10;
+  const size_t kPacketSizeBytes = 125;
+  const int64_t kExpectedSendTimeUs =
+      (kPacketSizeBytes * 8 * 1000 + kCapacityKbps / 2) / kCapacityKbps;
+  uint16_t sequence_number = 0;
+  int64_t send_time_us = 0;
+  ChokeFilter filter(NULL, 0);
+  filter.set_capacity_kbps(10);
+  Packets packets;
+  RTPHeader header;
+  for (int i = 0; i < 2; ++i) {
+    header.sequenceNumber = sequence_number++;
+    // Payload is 1000 bits.
+    packets.push_back(
+        new MediaPacket(0, send_time_us, kPacketSizeBytes, header));
+    // Packets are sent far enough a part plus an extra millisecond so that they
+    // will never be in the choke queue at the same time.
+    send_time_us += kExpectedSendTimeUs + 1000;
+  }
+  ASSERT_TRUE(IsTimeSorted(packets));
+  filter.RunFor(2 * kExpectedSendTimeUs + 1000, &packets);
+  EXPECT_EQ(kExpectedSendTimeUs, packets.front()->send_time_us());
+  delete packets.front();
+  packets.pop_front();
+  EXPECT_EQ(2 * kExpectedSendTimeUs + 1000, packets.front()->send_time_us());
+  delete packets.front();
+  packets.pop_front();
+}
+
 TEST_F(BweTestFramework_ChokeFilterTest, Short) {
   // 100ms, 100 packets, 10 kbps choke -> 1 kbit of data should have propagated.
   // That is actually just a single packet, since each packet has 1000 bits of
   // payload.
   ChokeFilter filter(NULL, 0);
-  filter.SetCapacity(10);
+  filter.set_capacity_kbps(10);
   TestChoke(&filter, 100, 100, 1);
 }
 
 TEST_F(BweTestFramework_ChokeFilterTest, Medium) {
   // 100ms, 10 packets, 10 kbps choke -> 1 packet through, or 1 kbit.
   ChokeFilter filter(NULL, 0);
-  filter.SetCapacity(10);
+  filter.set_capacity_kbps(10);
   TestChoke(&filter, 100, 10, 1);
   // 200ms, no new packets -> another packet through.
   TestChoke(&filter, 100, 0, 1);
@@ -690,7 +699,7 @@ TEST_F(BweTestFramework_ChokeFilterTest, Medium) {
 TEST_F(BweTestFramework_ChokeFilterTest, Long) {
   // 100ms, 100 packets in queue, 10 kbps choke -> 1 packet through, or 1 kbit.
   ChokeFilter filter(NULL, 0);
-  filter.SetCapacity(10);
+  filter.set_capacity_kbps(10);
   TestChoke(&filter, 100, 100, 1);
   // 200ms, no input, another packet through.
   TestChoke(&filter, 100, 0, 1);
@@ -698,22 +707,22 @@ TEST_F(BweTestFramework_ChokeFilterTest, Long) {
   TestChoke(&filter, 800, 0, 8);
   // 10000ms, no input, raise choke to 100 kbps. Remaining 90 packets in queue
   // should be propagated, for a total of 90 kbps.
-  filter.SetCapacity(100);
+  filter.set_capacity_kbps(100);
   TestChoke(&filter, 9000, 0, 90);
   // 10100ms, 20 more packets -> 10 packets or 10 kbit through.
   TestChoke(&filter, 100, 20, 10);
   // 10300ms, 10 more packets -> 20 packets out.
   TestChoke(&filter, 200, 10, 20);
   // 11300ms, no input, queue should be empty.
-  filter.SetCapacity(10);
+  filter.set_capacity_kbps(10);
   TestChoke(&filter, 1000, 0, 0);
 }
 
 TEST_F(BweTestFramework_ChokeFilterTest, MaxDelay) {
   // 10 kbps choke, 500 ms delay cap
   ChokeFilter filter(NULL, 0);
-  filter.SetCapacity(10);
-  filter.SetMaxDelay(500);
+  filter.set_capacity_kbps(10);
+  filter.set_max_delay_ms(500);
   // 100ms, 100 packets in queue, 10 kbps choke -> 1 packet through, or 1 kbit.
   TestChoke(&filter, 100, 100, 1);
   CheckMaxDelay(500);
@@ -723,18 +732,18 @@ TEST_F(BweTestFramework_ChokeFilterTest, MaxDelay) {
   TestChoke(&filter, 9500, 0, 0);
 
   // 100 ms delay cap
-  filter.SetMaxDelay(100);
-  // 10100ms, 50 more packets -> 2 packets or 2 kbit through.
-  TestChoke(&filter, 100, 50, 2);
+  filter.set_max_delay_ms(100);
+  // 10100ms, 50 more packets -> 1 packets or 1 kbit through.
+  TestChoke(&filter, 100, 50, 1);
   CheckMaxDelay(100);
   // 20000ms, no input, remaining packets in queue should have been dropped.
   TestChoke(&filter, 9900, 0, 0);
 
   // Reset delay cap (0 is no cap) and verify no packets are dropped.
-  filter.SetCapacity(10);
-  filter.SetMaxDelay(0);
-  TestChoke(&filter, 100, 100, 2);
-  TestChoke(&filter, 9900, 0, 98);
+  filter.set_capacity_kbps(10);
+  filter.set_max_delay_ms(0);
+  TestChoke(&filter, 100, 100, 1);
+  TestChoke(&filter, 9900, 0, 99);
 }
 
 TEST_F(BweTestFramework_ChokeFilterTest, ShortTrace) {
@@ -755,7 +764,7 @@ TEST_F(BweTestFramework_ChokeFilterTest, ShortTraceTwoWraps) {
 
 TEST_F(BweTestFramework_ChokeFilterTest, ShortTraceMaxDelay) {
   TraceBasedDeliveryFilter filter(NULL, 0);
-  filter.SetMaxDelay(25);
+  filter.set_max_delay_ms(25);
   ASSERT_TRUE(filter.Init(test::ResourcePath("synthetic-trace", "rx")));
   // Uses all slots up to 110 ms. Several packets are being dropped.
   TestChoke(&filter, 110, 20, 9);
@@ -765,23 +774,25 @@ TEST_F(BweTestFramework_ChokeFilterTest, ShortTraceMaxDelay) {
   TestChoke(&filter, 25, 1, 1);
 }
 
-void TestVideoSender(PacketSender* sender,
+void TestVideoSender(VideoSender* sender,
                      int64_t run_for_ms,
                      uint32_t expected_packets,
                      uint32_t expected_payload_size,
-                     uint32_t expected_total_payload_size) {
+                     size_t expected_total_payload_size) {
   assert(sender);
   Packets packets;
   sender->RunFor(run_for_ms, &packets);
   ASSERT_TRUE(IsTimeSorted(packets));
   ASSERT_TRUE(IsSequenceNumberSorted(packets));
   EXPECT_EQ(expected_packets, packets.size());
+
   int64_t send_time_us = -1;
-  uint32_t total_payload_size = 0;
+  size_t total_payload_size = 0;
   uint32_t absolute_send_time = 0;
   uint32_t absolute_send_time_wraps = 0;
   uint32_t rtp_timestamp = 0;
   uint32_t rtp_timestamp_wraps = 0;
+
   for (const auto* packet : packets) {
     const MediaPacket* media_packet = static_cast<const MediaPacket*>(packet);
     EXPECT_LE(send_time_us, media_packet->send_time_us());
@@ -801,6 +812,7 @@ void TestVideoSender(PacketSender* sender,
     }
     rtp_timestamp = media_packet->header().timestamp;
   }
+
   EXPECT_EQ(expected_total_payload_size, total_payload_size);
   EXPECT_GE(1u, absolute_send_time_wraps);
   EXPECT_GE(1u, rtp_timestamp_wraps);
@@ -809,105 +821,109 @@ void TestVideoSender(PacketSender* sender,
     delete packet;
 }
 
+// Random {-1, 0, +1} ms was added to frame timestamps.
+
 TEST(BweTestFramework_VideoSenderTest, Fps1Kbps80_1s) {
   // 1 fps, 80 kbps
   VideoSource source(0, 1.0f, 80, 0x1234, 0);
-  PacketSender sender(NULL, &source, kNullEstimator);
+  VideoSender sender(NULL, &source, kNullEstimator);
   EXPECT_EQ(80000u, source.bits_per_second());
   // We're at 1 fps, so all packets should be generated on first call, giving 10
   // packets of each 1000 bytes, total 10000 bytes.
   TestVideoSender(&sender, 1, 9, 400, 10000);
-  // 999ms, should see no output here.
-  TestVideoSender(&sender, 998, 0, 0, 0);
-  // 1999ms, should get data for one more frame.
-  TestVideoSender(&sender, 1000, 9, 400, 10000);
-  // 2000ms, one more frame.
-  TestVideoSender(&sender, 1, 9, 400, 10000);
-  // 2999ms, should see nothing.
-  TestVideoSender(&sender, 999, 0, 0, 0);
+  // 998ms, should see no output here.
+  TestVideoSender(&sender, 997, 0, 0, 0);
+  // 1001ms, should get data for one more frame.
+  TestVideoSender(&sender, 3, 9, 400, 10000);
+  // 1998ms, should see no output here.
+  TestVideoSender(&sender, 997, 0, 0, 0);
+  // 2001ms, one more frame.
+  TestVideoSender(&sender, 3, 9, 400, 10000);
+  // 2998ms, should see nothing.
+  TestVideoSender(&sender, 997, 0, 0, 0);
 }
 
 TEST(BweTestFramework_VideoSenderTest, Fps1Kbps80_1s_Offset) {
   // 1 fps, 80 kbps, offset 0.5 of a frame period, ==0.5s in this case.
   VideoSource source(0, 1.0f, 80, 0x1234, 500);
-  PacketSender sender(NULL, &source, kNullEstimator);
+  VideoSender sender(NULL, &source, kNullEstimator);
   EXPECT_EQ(80000u, source.bits_per_second());
-  // 499ms, no output.
-  TestVideoSender(&sender, 499, 0, 0, 0);
-  // 500ms, first frame (this is the offset we set), 10 packets of 1000 bytes.
-  TestVideoSender(&sender, 1, 9, 400, 10000);
-  // 1499ms, nothing.
-  TestVideoSender(&sender, 999, 0, 0, 0);
-  // 1999ms, second frame.
-  TestVideoSender(&sender, 500, 9, 400, 10000);
-  // 2499ms, nothing.
-  TestVideoSender(&sender, 500, 0, 0, 0);
-  // 2500ms, third frame.
-  TestVideoSender(&sender, 1, 9, 400, 10000);
-  // 3499ms, nothing.
-  TestVideoSender(&sender, 999, 0, 0, 0);
+  // 498ms, no output.
+  TestVideoSender(&sender, 498, 0, 0, 0);
+  // 501ms, first frame (this is the offset we set), 10 packets of 1000 bytes.
+  TestVideoSender(&sender, 3, 9, 400, 10000);
+  // 1498ms, nothing.
+  TestVideoSender(&sender, 997, 0, 0, 0);
+  // 1501ms, second frame.
+  TestVideoSender(&sender, 3, 9, 400, 10000);
+  // 2498ms, nothing.
+  TestVideoSender(&sender, 997, 0, 0, 0);
+  // 2501ms, third frame.
+  TestVideoSender(&sender, 3, 9, 400, 10000);
+  // 3498ms, nothing.
+  TestVideoSender(&sender, 997, 0, 0, 0);
 }
 
 TEST(BweTestFramework_VideoSenderTest, Fps50Kpbs80_11s) {
   // 50 fps, 80 kbps.
   VideoSource source(0, 50.0f, 80, 0x1234, 0);
-  PacketSender sender(NULL, &source, kNullEstimator);
+  VideoSender sender(NULL, &source, kNullEstimator);
   EXPECT_EQ(80000u, source.bits_per_second());
-  // 9998ms, should see 500 frames, 200 byte payloads, total 100000 bytes.
-  TestVideoSender(&sender, 9998, 500, 200, 100000);
-  // 9999ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
-  // 10000ms, 501st frame as a single packet.
-  TestVideoSender(&sender, 1, 1, 200, 200);
-  // 10998ms, 49 more frames.
-  TestVideoSender(&sender, 998, 49, 200, 9800);
-  // 10999ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
+  // 9981, should see 500 frames, 200 byte payloads, total 100000 bytes.
+  TestVideoSender(&sender, 9981, 500, 200, 100000);
+  // 9998ms, nothing.
+  TestVideoSender(&sender, 17, 0, 0, 0);
+  // 10001ms, 501st frame as a single packet.
+  TestVideoSender(&sender, 3, 1, 200, 200);
+  // 10981ms, 49 more frames.
+  TestVideoSender(&sender, 981, 49, 200, 9800);
+  // 10998ms, nothing.
+  TestVideoSender(&sender, 17, 0, 0, 0);
 }
 
-TEST(BweTestFramework_VideoSenderTest, Fps10Kpbs120_1s) {
+TEST(BweTestFramework_VideoSenderTest, Fps20Kpbs120_1s) {
   // 20 fps, 120 kbps.
   VideoSource source(0, 20.0f, 120, 0x1234, 0);
-  PacketSender sender(NULL, &source, kNullEstimator);
+  VideoSender sender(NULL, &source, kNullEstimator);
   EXPECT_EQ(120000u, source.bits_per_second());
-  // 498ms, 10 frames with 750 byte payloads, total 7500 bytes.
-  TestVideoSender(&sender, 498, 10, 750, 7500);
-  // 499ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
-  // 500ms, one more frame.
-  TestVideoSender(&sender, 1, 1, 750, 750);
-  // 998ms, 9 more frames.
-  TestVideoSender(&sender, 498, 9, 750, 6750);
-  // 999ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
+  // 451ms, 10 frames with 750 byte payloads, total 7500 bytes.
+  TestVideoSender(&sender, 451, 10, 750, 7500);
+  // 498ms, nothing.
+  TestVideoSender(&sender, 47, 0, 0, 0);
+  // 501ms, one more frame.
+  TestVideoSender(&sender, 3, 1, 750, 750);
+  // 951ms, 9 more frames.
+  TestVideoSender(&sender, 450, 9, 750, 6750);
+  // 998ms, nothing.
+  TestVideoSender(&sender, 47, 0, 0, 0);
 }
 
-TEST(BweTestFramework_VideoSenderTest, Fps30Kbps800_20s) {
-  // 20 fps, 820 kbps.
+TEST(BweTestFramework_VideoSenderTest, Fps25Kbps820_20s) {
+  // 25 fps, 820 kbps.
   VideoSource source(0, 25.0f, 820, 0x1234, 0);
-  PacketSender sender(NULL, &source, kNullEstimator);
+  VideoSender sender(NULL, &source, kNullEstimator);
   EXPECT_EQ(820000u, source.bits_per_second());
-  // 9998ms, 250 frames. 820 kbps = 102500 bytes/s, so total should be 1025000.
+  // 9961ms, 250 frames. 820 kbps = 102500 bytes/s, so total should be 1025000.
   // Each frame is 102500/25=4100 bytes, or 5 packets (4 @1000 bytes, 1 @100),
   // so packet count should be 5*250=1250 and last packet of each frame has
   // 100 bytes of payload.
-  TestVideoSender(&sender, 9998, 1000, 500, 1025000);
-  // 9999ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
-  // 19998ms, 250 more frames.
-  TestVideoSender(&sender, 9999, 1000, 500, 1025000);
-  // 19999ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
-  // 20038ms, one more frame, as described above (25fps == 40ms/frame).
-  TestVideoSender(&sender, 39, 4, 500, 4100);
-  // 20039ms, nothing.
-  TestVideoSender(&sender, 1, 0, 0, 0);
+  TestVideoSender(&sender, 9961, 1000, 500, 1025000);
+  // 9998ms, nothing.
+  TestVideoSender(&sender, 37, 0, 0, 0);
+  // 19961ms, 250 more frames.
+  TestVideoSender(&sender, 9963, 1000, 500, 1025000);
+  // 19998ms, nothing.
+  TestVideoSender(&sender, 37, 0, 0, 0);
+  // 20001ms, one more frame, as described above (25fps == 40ms/frame).
+  TestVideoSender(&sender, 3, 4, 500, 4100);
+  // 20038ms, nothing.
+  TestVideoSender(&sender, 37, 0, 0, 0);
 }
 
 TEST(BweTestFramework_VideoSenderTest, TestAppendInOrder) {
   // 1 fps, 80 kbps, 250ms offset.
   VideoSource source1(0, 1.0f, 80, 0x1234, 250);
-  PacketSender sender1(NULL, &source1, kNullEstimator);
+  VideoSender sender1(NULL, &source1, kNullEstimator);
   EXPECT_EQ(80000u, source1.bits_per_second());
   Packets packets;
   // Generate some packets, verify they are sorted.
@@ -923,7 +939,7 @@ TEST(BweTestFramework_VideoSenderTest, TestAppendInOrder) {
 
   // Another sender, 2 fps, 160 kbps, 150ms offset
   VideoSource source2(0, 2.0f, 160, 0x2234, 150);
-  PacketSender sender2(NULL, &source2, kNullEstimator);
+  VideoSender sender2(NULL, &source2, kNullEstimator);
   EXPECT_EQ(160000u, source2.bits_per_second());
   // Generate some packets, verify that they are merged with the packets already
   // on the list.
@@ -941,37 +957,37 @@ TEST(BweTestFramework_VideoSenderTest, TestAppendInOrder) {
 
 TEST(BweTestFramework_VideoSenderTest, FeedbackIneffective) {
   VideoSource source(0, 25.0f, 820, 0x1234, 0);
-  PacketSender sender(NULL, &source, kNullEstimator);
+  VideoSender sender(NULL, &source, kNullEstimator);
 
   EXPECT_EQ(820000u, source.bits_per_second());
-  TestVideoSender(&sender, 9998, 1000, 500, 1025000);
+  TestVideoSender(&sender, 9961, 1000, 500, 1025000);
 
   // Make sure feedback has no effect on a regular video sender.
-  RembFeedback* feedback = new RembFeedback(0, 0, 512000, RTCPReportBlock());
+  RembFeedback* feedback = new RembFeedback(0, 0, 0, 512000, RTCPReportBlock());
   Packets packets;
   packets.push_back(feedback);
   sender.RunFor(0, &packets);
   EXPECT_EQ(820000u, source.bits_per_second());
-  TestVideoSender(&sender, 9998, 1000, 500, 1025000);
+  TestVideoSender(&sender, 10000, 1000, 500, 1025000);
 }
 
 TEST(BweTestFramework_AdaptiveVideoSenderTest, FeedbackChangesBitrate) {
   AdaptiveVideoSource source(0, 25.0f, 820, 0x1234, 0);
-  PacketSender sender(NULL, &source, kRembEstimator);
+  VideoSender sender(NULL, &source, kRembEstimator);
   EXPECT_EQ(820000u, source.bits_per_second());
-  TestVideoSender(&sender, 9998, 1000, 500, 1025000);
+  TestVideoSender(&sender, 9961, 1000, 500, 1025000);
 
   // Make sure we can reduce the bitrate.
-  RembFeedback* feedback = new RembFeedback(0, 0, 512000, RTCPReportBlock());
+  RembFeedback* feedback = new RembFeedback(0, 0, 0, 512000, RTCPReportBlock());
   Packets packets;
   packets.push_back(feedback);
   sender.RunFor(0, &packets);
   EXPECT_EQ(512000u, source.bits_per_second());
-  TestVideoSender(&sender, 9998, 750, 160, 640000);
+  TestVideoSender(&sender, 10000, 750, 160, 640000);
 
   // Increase the bitrate to the initial bitrate and verify that the output is
   // the same.
-  feedback = new RembFeedback(0, 0, 820000, RTCPReportBlock());
+  feedback = new RembFeedback(0, 0, 0, 820000, RTCPReportBlock());
   packets.push_back(feedback);
   sender.RunFor(10000, &packets);
   EXPECT_EQ(820000u, source.bits_per_second());
@@ -987,16 +1003,16 @@ TEST(BweTestFramework_AdaptiveVideoSenderTest, Paced_FeedbackChangesBitrate) {
   TestVideoSender(&sender, 9998, 1000, 500, 1025000);
 
   // Make sure we can reduce the bitrate.
-  RembFeedback* feedback = new RembFeedback(0, 1, 512000, RTCPReportBlock());
+  RembFeedback* feedback = new RembFeedback(0, 1, 0, 512000, RTCPReportBlock());
   Packets packets;
   packets.push_back(feedback);
   sender.RunFor(10000, &packets);
   ASSERT_EQ(512000u, source.bits_per_second());
-  TestVideoSender(&sender, 9998, 750, 160, 640000);
+  TestVideoSender(&sender, 10000, 750, 160, 640000);
 
   // Increase the bitrate to the initial bitrate and verify that the output is
   // the same.
-  feedback = new RembFeedback(0, 0, 820000, RTCPReportBlock());
+  feedback = new RembFeedback(0, 0, 0, 820000, RTCPReportBlock());
   packets.push_back(feedback);
   sender.RunFor(10000, &packets);
   EXPECT_EQ(820000u, source.bits_per_second());
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
index 65369fde36..3a84e81a0b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
@@ -16,9 +16,10 @@
 #include <stdio.h>
 
 #include <algorithm>
+#include <sstream>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 namespace testing {
@@ -27,14 +28,9 @@ namespace bwe {
 Logging Logging::g_Logging;
 
 static std::string ToString(uint32_t v) {
-  const size_t kBufferSize = 16;
-  char string_buffer[kBufferSize] = {0};
-#if defined(_MSC_VER) && defined(_WIN32)
-  _snprintf(string_buffer, kBufferSize - 1, "%08x", v);
-#else
-  snprintf(string_buffer, kBufferSize, "%08x", v);
-#endif
-  return string_buffer;
+  std::stringstream ss;
+  ss << v;
+  return ss.str();
 }
 
 Logging::Context::Context(uint32_t name, int64_t timestamp_ms, bool enabled) {
@@ -61,27 +57,27 @@ Logging* Logging::GetInstance() {
 
 void Logging::SetGlobalContext(uint32_t name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.tag = ToString(name);
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = ToString(name);
 }
 
 void Logging::SetGlobalContext(const std::string& name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.tag = name;
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = name;
 }
 
 void Logging::SetGlobalContext(const char* name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.tag = name;
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = name;
 }
 
 void Logging::SetGlobalEnable(bool enabled) {
   CriticalSectionScoped cs(crit_sect_.get());
-  thread_map_[ThreadWrapper::GetThreadId()].global_state.enabled = enabled;
+  thread_map_[rtc::CurrentThreadId()].global_state.enabled = enabled;
 }
 
 void Logging::Log(const char format[], ...) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
@@ -94,14 +90,104 @@ void Logging::Log(const char format[], ...) {
   }
 }
 
-void Logging::Plot(double value) {
+void Logging::Plot(int figure, double value) {
+  Plot(figure, value, "-");
+}
+
+void Logging::Plot(int figure, double value, const std::string& alg_name) {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  assert(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  std::string label = state.tag + '@' + alg_name;
+  std::string prefix("Available");
+  if (alg_name.compare(0, prefix.length(), prefix) == 0) {
+    std::string receiver("Receiver");
+    size_t start_pos = label.find(receiver);
+    if (start_pos != std::string::npos) {
+      label.replace(start_pos, receiver.length(), "Sender");
+    }
+  }
+  if (state.enabled) {
+    printf("PLOT\t%d\t%s\t%f\t%f\n", figure, label.c_str(),
+           state.timestamp_ms * 0.001, value);
+  }
+}
+
+void Logging::PlotBar(int figure,
+                      const std::string& name,
+                      double value,
+                      int flow_id) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   const State& state = it->second.stack.top();
   if (state.enabled) {
-    printf("PLOT\t%s\t%f\t%f\n", state.tag.c_str(), state.timestamp_ms * 0.001,
-           value);
+    printf("BAR\t%d\t%s_%d\t%f\n", figure, name.c_str(), flow_id, value);
+  }
+}
+
+void Logging::PlotBaselineBar(int figure,
+                              const std::string& name,
+                              double value,
+                              int flow_id) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  assert(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("BASELINE\t%d\t%s_%d\t%f\n", figure, name.c_str(), flow_id, value);
+  }
+}
+
+void Logging::PlotErrorBar(int figure,
+                           const std::string& name,
+                           double value,
+                           double ylow,
+                           double yhigh,
+                           const std::string& error_title,
+                           int flow_id) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  assert(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("ERRORBAR\t%d\t%s_%d\t%f\t%f\t%f\t%s\n", figure, name.c_str(),
+           flow_id, value, ylow, yhigh, error_title.c_str());
+  }
+}
+
+void Logging::PlotLimitErrorBar(int figure,
+                                const std::string& name,
+                                double value,
+                                double ylow,
+                                double yhigh,
+                                const std::string& error_title,
+                                double ymax,
+                                const std::string& limit_title,
+                                int flow_id) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  assert(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("LIMITERRORBAR\t%d\t%s_%d\t%f\t%f\t%f\t%s\t%f\t%s\n", figure,
+           name.c_str(), flow_id, value, ylow, yhigh, error_title.c_str(), ymax,
+           limit_title.c_str());
+  }
+}
+
+void Logging::PlotLabel(int figure,
+                        const std::string& title,
+                        const std::string& y_label,
+                        int num_flows) {
+  CriticalSectionScoped cs(crit_sect_.get());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  assert(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("LABEL\t%d\t%s\t%s\t%d\n", figure, title.c_str(), y_label.c_str(),
+           num_flows);
   }
 }
 
@@ -120,9 +206,9 @@ Logging::State::State(const std::string& tag, int64_t timestamp_ms,
 }
 
 void Logging::State::MergePrevious(const State& previous) {
-  if (tag == "") {
+  if (tag.empty()) {
     tag = previous.tag;
-  } else if (previous.tag != "") {
+  } else if (!previous.tag.empty()) {
     tag = previous.tag + "_" + tag;
   }
   timestamp_ms = std::max(previous.timestamp_ms, timestamp_ms);
@@ -133,7 +219,7 @@ void Logging::PushState(const std::string& append_to_tag, int64_t timestamp_ms,
                         bool enabled) {
   CriticalSectionScoped cs(crit_sect_.get());
   State new_state(append_to_tag, timestamp_ms, enabled);
-  ThreadState* thread_state = &thread_map_[ThreadWrapper::GetThreadId()];
+  ThreadState* thread_state = &thread_map_[rtc::CurrentThreadId()];
   std::stack<State>* stack = &thread_state->stack;
   if (stack->empty()) {
     new_state.MergePrevious(thread_state->global_state);
@@ -145,7 +231,7 @@ void Logging::PushState(const std::string& append_to_tag, int64_t timestamp_ms,
 
 void Logging::PopState() {
   CriticalSectionScoped cs(crit_sect_.get());
-  ThreadMap::iterator it = thread_map_.find(ThreadWrapper::GetThreadId());
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
   assert(it != thread_map_.end());
   std::stack<State>* stack = &it->second.stack;
   int64_t newest_timestamp_ms = stack->top().timestamp_ms;
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
index 728e98e629..cc7807ba8a 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
@@ -28,7 +28,7 @@
 //      BWE_TEST_LOGGING_CONTEXT(i);
 //      BWE_TEST_LOGGING_LOG1("weight", "%f tonnes", weights_[i]);
 //      for (float j=0.0f; j<1.0; j+=0.4f) {
-//        BWE_TEST_LOGGING_PLOT("bps", -1, j);
+//        BWE_TEST_LOGGING_PLOT(0, "bps", -1, j);
 //      }
 //    }
 //  }
@@ -86,11 +86,36 @@
 #define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5)
 
 // Print to stdout in tab-separated format suitable for plotting, e.g.:
-//   PLOT  Context1_Context2_Name  time  value
+//   PLOT figure Context1_Context2_Name  time  value
+// |figure| is a figure id. Different figures are plotted in different windows.
 // |name| is a char*, std::string or uint32_t to name the plotted value.
 // |time| is an int64_t time in ms, or -1 to inherit time from previous context.
 // |value| is a double precision float to be plotted.
-#define BWE_TEST_LOGGING_PLOT(name, time, value)
+// |alg_name| is an optional argument, a string
+#define BWE_TEST_LOGGING_PLOT(figure, name, time, value)
+#define BWE_TEST_LOGGING_PLOT_WITH_NAME(figure, name, time, value, alg_name)
+
+// Print to stdout in tab-separated format suitable for plotting, e.g.:
+//   BAR figure Context1_Context2_Name  x_left  width  value
+// |figure| is a figure id. Different figures are plotted in different windows.
+// |name| is a char*, std::string or uint32_t to name the plotted value.
+// |value| is a double precision float to be plotted.
+// |ylow| and |yhigh| are double precision float for the error line.
+// |title| is a string and refers to the error label.
+// |ymax| is a double precision float for the limit horizontal line.
+// |limit_title| is a string and refers to the limit label.
+#define BWE_TEST_LOGGING_BAR(figure, name, value, flow_id)
+#define BWE_TEST_LOGGING_ERRORBAR(figure, name, value, ylow, yhigh, \
+                                  error_title, flow_id)
+#define BWE_TEST_LOGGING_LIMITERRORBAR( \
+    figure, name, value, ylow, yhigh, error_title, ymax, limit_title, flow_id)
+
+#define BWE_TEST_LOGGING_BASELINEBAR(figure, name, value, flow_id)
+
+// |num_flows| is an integer refering to the number of RMCAT flows in the
+// scenario.
+// Define |x_label| and |y_label| for plots.
+#define BWE_TEST_LOGGING_LABEL(figure, x_label, y_label, num_flows)
 
 #else  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
 
@@ -105,12 +130,12 @@
 #define BWE_TEST_LOGGING_GLOBAL_CONTEXT(name) \
     do { \
       webrtc::testing::bwe::Logging::GetInstance()->SetGlobalContext(name); \
-    } while (0);
+    } while (0)
 
 #define BWE_TEST_LOGGING_GLOBAL_ENABLE(enabled) \
     do { \
       webrtc::testing::bwe::Logging::GetInstance()->SetGlobalEnable(enabled); \
-    } while (0);
+    } while (0)
 
 #define __BWE_TEST_LOGGING_CONTEXT_NAME(ctx, line) ctx ## line
 #define __BWE_TEST_LOGGING_CONTEXT_DECLARE(ctx, line, name, time, enabled) \
@@ -130,36 +155,82 @@
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG2(name, format, _1, _2) \
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG3(name, format, _1, _2, _3) \
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG4(name, format, _1, _2, _3, _4) \
     do { \
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, \
                                                         _4); \
-    } while (0);
+    } while (0)
 #define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5) \
     do {\
       BWE_TEST_LOGGING_CONTEXT(name); \
       webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, \
                                                         _4, _5); \
-    } while (0);
+    } while (0)
 
-#define BWE_TEST_LOGGING_PLOT(name, time, value)\
-    do { \
-      __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, name, \
-                                         static_cast<int64_t>(time), true); \
-      webrtc::testing::bwe::Logging::GetInstance()->Plot(value); \
-    } while (0);
+#define BWE_TEST_LOGGING_PLOT(figure, name, time, value)                  \
+  do {                                                                    \
+    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,        \
+                                       static_cast<int64_t>(time), true); \
+    webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, value);    \
+  } while (0)
+
+#define BWE_TEST_LOGGING_PLOT_WITH_NAME(figure, name, time, value, alg_name) \
+  do {                                                                       \
+    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,           \
+                                       static_cast<int64_t>(time), true);    \
+    webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, value,        \
+                                                       alg_name);            \
+  } while (0)
+
+#define BWE_TEST_LOGGING_BAR(figure, name, value, flow_id)                     \
+  do {                                                                         \
+    BWE_TEST_LOGGING_CONTEXT(name);                                            \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotBar(figure, name, value, \
+                                                          flow_id);            \
+  } while (0)
+
+#define BWE_TEST_LOGGING_BASELINEBAR(figure, name, value, flow_id) \
+  do {                                                             \
+    BWE_TEST_LOGGING_CONTEXT(name);                                \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotBaselineBar( \
+        figure, name, value, flow_id);                             \
+  } while (0)
+
+#define BWE_TEST_LOGGING_ERRORBAR(figure, name, value, ylow, yhigh, title, \
+                                  flow_id)                                 \
+  do {                                                                     \
+    BWE_TEST_LOGGING_CONTEXT(name);                                        \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotErrorBar(            \
+        figure, name, value, ylow, yhigh, title, flow_id);                 \
+  } while (0)
+
+#define BWE_TEST_LOGGING_LIMITERRORBAR(                                        \
+    figure, name, value, ylow, yhigh, error_title, ymax, limit_title, flow_id) \
+  do {                                                                         \
+    BWE_TEST_LOGGING_CONTEXT(name);                                            \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotLimitErrorBar(           \
+        figure, name, value, ylow, yhigh, error_title, ymax, limit_title,      \
+        flow_id);                                                              \
+  } while (0)
+
+#define BWE_TEST_LOGGING_LABEL(figure, title, y_label, num_flows) \
+  do {                                                            \
+    BWE_TEST_LOGGING_CONTEXT(title);                              \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotLabel(      \
+        figure, title, y_label, num_flows);                       \
+  } while (0)
 
 namespace webrtc {
 
@@ -177,7 +248,7 @@ class Logging {
     Context(const char* name, int64_t timestamp_ms, bool enabled);
     ~Context();
    private:
-    DISALLOW_IMPLICIT_CONSTRUCTORS(Context);
+    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Context);
   };
 
   static Logging* GetInstance();
@@ -188,7 +259,34 @@ class Logging {
   void SetGlobalEnable(bool enabled);
 
   void Log(const char format[], ...);
-  void Plot(double value);
+  void Plot(int figure, double value);
+  void Plot(int figure, double value, const std::string& alg_name);
+  void PlotBar(int figure, const std::string& name, double value, int flow_id);
+  void PlotBaselineBar(int figure,
+                       const std::string& name,
+                       double value,
+                       int flow_id);
+  void PlotErrorBar(int figure,
+                    const std::string& name,
+                    double value,
+                    double ylow,
+                    double yhigh,
+                    const std::string& error_title,
+                    int flow_id);
+
+  void PlotLimitErrorBar(int figure,
+                         const std::string& name,
+                         double value,
+                         double ylow,
+                         double yhigh,
+                         const std::string& error_title,
+                         double ymax,
+                         const std::string& limit_title,
+                         int flow_id);
+  void PlotLabel(int figure,
+                 const std::string& title,
+                 const std::string& y_label,
+                 int num_flows);
 
  private:
   struct State {
@@ -215,7 +313,7 @@ class Logging {
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   ThreadMap thread_map_;
 
-  DISALLOW_COPY_AND_ASSIGN(Logging);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Logging);
 };
 }  // namespace bwe
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
new file mode 100644
index 0000000000..6245ccd25d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
@@ -0,0 +1,394 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
+
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+const int kSetCapacity = 1000;
+
+class LinkedSetTest : public ::testing::Test {
+ public:
+  LinkedSetTest() : linked_set_(kSetCapacity) {}
+
+  ~LinkedSetTest() {}
+
+ protected:
+  LinkedSet linked_set_;
+};
+
+TEST_F(LinkedSetTest, EmptySet) {
+  EXPECT_EQ(linked_set_.OldestSeqNumber(), 0);
+  EXPECT_EQ(linked_set_.NewestSeqNumber(), 0);
+}
+
+TEST_F(LinkedSetTest, SinglePacket) {
+  const uint16_t kSeqNumber = 1;  // Arbitrary.
+  // Other parameters don't matter here.
+  linked_set_.Insert(kSeqNumber, 0, 0, 0);
+
+  EXPECT_EQ(linked_set_.OldestSeqNumber(), kSeqNumber);
+  EXPECT_EQ(linked_set_.NewestSeqNumber(), kSeqNumber);
+}
+
+TEST_F(LinkedSetTest, MultiplePackets) {
+  const uint16_t kNumberPackets = 100;
+
+  std::vector<uint16_t> sequence_numbers;
+  for (size_t i = 0; i < kNumberPackets; ++i) {
+    sequence_numbers.push_back(static_cast<uint16_t>(i + 1));
+  }
+  random_shuffle(sequence_numbers.begin(), sequence_numbers.end());
+
+  for (size_t i = 0; i < kNumberPackets; ++i) {
+    // Other parameters don't matter here.
+    linked_set_.Insert(static_cast<uint16_t>(i), 0, 0, 0);
+  }
+
+  // Packets arriving out of order should not affect the following values:
+  EXPECT_EQ(linked_set_.OldestSeqNumber(), 0);
+  EXPECT_EQ(linked_set_.NewestSeqNumber(), kNumberPackets - 1);
+}
+
+TEST_F(LinkedSetTest, Overflow) {
+  const int kFirstSeqNumber = -100;
+  const int kLastSeqNumber = 100;
+
+  for (int i = kFirstSeqNumber; i <= kLastSeqNumber; ++i) {
+    // Other parameters don't matter here.
+    linked_set_.Insert(static_cast<uint16_t>(i), 0, 0, 0);
+  }
+
+  // Packets arriving out of order should not affect the following values:
+  EXPECT_EQ(linked_set_.OldestSeqNumber(),
+            static_cast<uint16_t>(kFirstSeqNumber));
+  EXPECT_EQ(linked_set_.NewestSeqNumber(),
+            static_cast<uint16_t>(kLastSeqNumber));
+}
+
+class SequenceNumberOlderThanTest : public ::testing::Test {
+ public:
+  SequenceNumberOlderThanTest() {}
+  ~SequenceNumberOlderThanTest() {}
+
+ protected:
+  SequenceNumberOlderThan comparator_;
+};
+
+TEST_F(SequenceNumberOlderThanTest, Operator) {
+  // Operator()(x, y) returns true <==> y is newer than x.
+  EXPECT_TRUE(comparator_.operator()(0x0000, 0x0001));
+  EXPECT_TRUE(comparator_.operator()(0x0001, 0x1000));
+  EXPECT_FALSE(comparator_.operator()(0x0001, 0x0000));
+  EXPECT_FALSE(comparator_.operator()(0x0002, 0x0002));
+  EXPECT_TRUE(comparator_.operator()(0xFFF6, 0x000A));
+  EXPECT_FALSE(comparator_.operator()(0x000A, 0xFFF6));
+  EXPECT_TRUE(comparator_.operator()(0x0000, 0x8000));
+  EXPECT_FALSE(comparator_.operator()(0x8000, 0x0000));
+}
+
+class LossAccountTest : public ::testing::Test {
+ public:
+  LossAccountTest() {}
+  ~LossAccountTest() {}
+
+ protected:
+  LossAccount loss_account_;
+};
+
+TEST_F(LossAccountTest, Operations) {
+  const size_t kTotal = 100;  // Arbitrary values.
+  const size_t kLost = 10;
+
+  LossAccount rhs(kTotal, kLost);
+
+  loss_account_.Add(rhs);
+  EXPECT_EQ(loss_account_.num_total, kTotal);
+  EXPECT_EQ(loss_account_.num_lost, kLost);
+  EXPECT_NEAR(loss_account_.LossRatio(), static_cast<float>(kLost) / kTotal,
+              0.001f);
+
+  loss_account_.Subtract(rhs);
+  EXPECT_EQ(loss_account_.num_total, 0UL);
+  EXPECT_EQ(loss_account_.num_lost, 0UL);
+  EXPECT_NEAR(loss_account_.LossRatio(), 0.0f, 0.001f);
+}
+
+class BweReceiverTest : public ::testing::Test {
+ public:
+  BweReceiverTest() : bwe_receiver_(kFlowId) {}
+  ~BweReceiverTest() {}
+
+ protected:
+  const int kFlowId = 1;  // Arbitrary.
+  BweReceiver bwe_receiver_;
+};
+
+TEST_F(BweReceiverTest, ReceivingRateNoPackets) {
+  EXPECT_EQ(bwe_receiver_.RecentKbps(), static_cast<size_t>(0));
+}
+
+TEST_F(BweReceiverTest, ReceivingRateSinglePacket) {
+  const size_t kPayloadSizeBytes = 500 * 1000;
+  const int64_t kSendTimeUs = 300 * 1000;
+  const int64_t kArrivalTimeMs = kSendTimeUs / 1000 + 100;
+  const uint16_t kSequenceNumber = 1;
+  const int64_t kTimeWindowMs = BweReceiver::kReceivingRateTimeWindowMs;
+
+  const MediaPacket media_packet(kFlowId, kSendTimeUs, kPayloadSizeBytes,
+                                 kSequenceNumber);
+  bwe_receiver_.ReceivePacket(kArrivalTimeMs, media_packet);
+
+  const size_t kReceivingRateKbps = 8 * kPayloadSizeBytes / kTimeWindowMs;
+
+  EXPECT_NEAR(bwe_receiver_.RecentKbps(), kReceivingRateKbps,
+              static_cast<float>(kReceivingRateKbps) / 100.0f);
+}
+
+TEST_F(BweReceiverTest, ReceivingRateSmallPackets) {
+  const size_t kPayloadSizeBytes = 100 * 1000;
+  const int64_t kTimeGapMs = 50;  // Between each packet.
+  const int64_t kOneWayDelayMs = 50;
+
+  for (int i = 1; i < 50; ++i) {
+    int64_t send_time_us = i * kTimeGapMs * 1000;
+    int64_t arrival_time_ms = send_time_us / 1000 + kOneWayDelayMs;
+    uint16_t sequence_number = i;
+    const MediaPacket media_packet(kFlowId, send_time_us, kPayloadSizeBytes,
+                                   sequence_number);
+    bwe_receiver_.ReceivePacket(arrival_time_ms, media_packet);
+  }
+
+  const size_t kReceivingRateKbps = 8 * kPayloadSizeBytes / kTimeGapMs;
+  EXPECT_NEAR(bwe_receiver_.RecentKbps(), kReceivingRateKbps,
+              static_cast<float>(kReceivingRateKbps) / 100.0f);
+}
+
+TEST_F(BweReceiverTest, PacketLossNoPackets) {
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+}
+
+TEST_F(BweReceiverTest, PacketLossSinglePacket) {
+  const MediaPacket media_packet(kFlowId, 0, 0, 0);
+  bwe_receiver_.ReceivePacket(0, media_packet);
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+}
+
+TEST_F(BweReceiverTest, PacketLossContiguousPackets) {
+  const int64_t kTimeWindowMs = BweReceiver::kPacketLossTimeWindowMs;
+  size_t set_capacity = bwe_receiver_.GetSetCapacity();
+
+  for (int i = 0; i < 10; ++i) {
+    uint16_t sequence_number = static_cast<uint16_t>(i);
+    // Sequence_number and flow_id are the only members that matter here.
+    const MediaPacket media_packet(kFlowId, 0, 0, sequence_number);
+    // Arrival time = 0, all packets will be considered.
+    bwe_receiver_.ReceivePacket(0, media_packet);
+  }
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+
+  for (int i = 30; i > 20; i--) {
+    uint16_t sequence_number = static_cast<uint16_t>(i);
+    // Sequence_number and flow_id are the only members that matter here.
+    const MediaPacket media_packet(kFlowId, 0, 0, sequence_number);
+    // Only the packets sent in this for loop will be considered.
+    bwe_receiver_.ReceivePacket(2 * kTimeWindowMs, media_packet);
+  }
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+
+  // Should handle uint16_t overflow.
+  for (int i = 0xFFFF - 10; i < 0xFFFF + 10; ++i) {
+    uint16_t sequence_number = static_cast<uint16_t>(i);
+    const MediaPacket media_packet(kFlowId, 0, 0, sequence_number);
+    // Only the packets sent in this for loop will be considered.
+    bwe_receiver_.ReceivePacket(4 * kTimeWindowMs, media_packet);
+  }
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+
+  // Should handle set overflow.
+  for (int i = 0; i < set_capacity * 1.5; ++i) {
+    uint16_t sequence_number = static_cast<uint16_t>(i);
+    const MediaPacket media_packet(kFlowId, 0, 0, sequence_number);
+    // Only the packets sent in this for loop will be considered.
+    bwe_receiver_.ReceivePacket(6 * kTimeWindowMs, media_packet);
+  }
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+}
+
+// Should handle duplicates.
+TEST_F(BweReceiverTest, PacketLossDuplicatedPackets) {
+  const int64_t kTimeWindowMs = BweReceiver::kPacketLossTimeWindowMs;
+
+  for (int i = 0; i < 10; ++i) {
+    const MediaPacket media_packet(kFlowId, 0, 0, 0);
+    // Arrival time = 0, all packets will be considered.
+    bwe_receiver_.ReceivePacket(0, media_packet);
+  }
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+
+  // Missing the element 5.
+  const uint16_t kSequenceNumbers[] = {1, 2, 3, 4, 6, 7, 8};
+  const int kNumPackets = arraysize(kSequenceNumbers);
+
+  // Insert each sequence number twice.
+  for (int i = 0; i < 2; ++i) {
+    for (int j = 0; j < kNumPackets; j++) {
+      const MediaPacket media_packet(kFlowId, 0, 0, kSequenceNumbers[j]);
+      // Only the packets sent in this for loop will be considered.
+      bwe_receiver_.ReceivePacket(2 * kTimeWindowMs, media_packet);
+    }
+  }
+
+  EXPECT_NEAR(bwe_receiver_.RecentPacketLossRatio(), 1.0f / (kNumPackets + 1),
+              0.1f / (kNumPackets + 1));
+}
+
+TEST_F(BweReceiverTest, PacketLossLakingPackets) {
+  size_t set_capacity = bwe_receiver_.GetSetCapacity();
+  EXPECT_LT(set_capacity, static_cast<size_t>(0xFFFF));
+
+  // Missing every other packet.
+  for (size_t i = 0; i < set_capacity; ++i) {
+    if ((i & 1) == 0) {  // Only even sequence numbers.
+      uint16_t sequence_number = static_cast<uint16_t>(i);
+      const MediaPacket media_packet(kFlowId, 0, 0, sequence_number);
+      // Arrival time = 0, all packets will be considered.
+      bwe_receiver_.ReceivePacket(0, media_packet);
+    }
+  }
+  EXPECT_NEAR(bwe_receiver_.RecentPacketLossRatio(), 0.5f, 0.01f);
+}
+
+TEST_F(BweReceiverTest, PacketLossLakingFewPackets) {
+  size_t set_capacity = bwe_receiver_.GetSetCapacity();
+  EXPECT_LT(set_capacity, static_cast<size_t>(0xFFFF));
+
+  const int kPeriod = 100;
+  // Missing one for each kPeriod packets.
+  for (size_t i = 0; i < set_capacity; ++i) {
+    if ((i % kPeriod) != 0) {
+      uint16_t sequence_number = static_cast<uint16_t>(i);
+      const MediaPacket media_packet(kFlowId, 0, 0, sequence_number);
+      // Arrival time = 0, all packets will be considered.
+      bwe_receiver_.ReceivePacket(0, media_packet);
+    }
+  }
+  EXPECT_NEAR(bwe_receiver_.RecentPacketLossRatio(), 1.0f / kPeriod,
+              0.1f / kPeriod);
+}
+
+// Packet's sequence numbers greatly apart, expect high loss.
+TEST_F(BweReceiverTest, PacketLossWideGap) {
+  const int64_t kTimeWindowMs = BweReceiver::kPacketLossTimeWindowMs;
+
+  const MediaPacket media_packet1(0, 0, 0, 1);
+  const MediaPacket media_packet2(0, 0, 0, 1000);
+  // Only these two packets will be considered.
+  bwe_receiver_.ReceivePacket(0, media_packet1);
+  bwe_receiver_.ReceivePacket(0, media_packet2);
+  EXPECT_NEAR(bwe_receiver_.RecentPacketLossRatio(), 0.998f, 0.0001f);
+
+  const MediaPacket media_packet3(0, 0, 0, 0);
+  const MediaPacket media_packet4(0, 0, 0, 0x8000);
+  // Only these two packets will be considered.
+  bwe_receiver_.ReceivePacket(2 * kTimeWindowMs, media_packet3);
+  bwe_receiver_.ReceivePacket(2 * kTimeWindowMs, media_packet4);
+  EXPECT_NEAR(bwe_receiver_.RecentPacketLossRatio(), 0.99994f, 0.00001f);
+}
+
+// Packets arriving unordered should not be counted as losted.
+TEST_F(BweReceiverTest, PacketLossUnorderedPackets) {
+  size_t num_packets = bwe_receiver_.GetSetCapacity() / 2;
+  std::vector<uint16_t> sequence_numbers;
+
+  for (size_t i = 0; i < num_packets; ++i) {
+    sequence_numbers.push_back(static_cast<uint16_t>(i + 1));
+  }
+
+  random_shuffle(sequence_numbers.begin(), sequence_numbers.end());
+
+  for (size_t i = 0; i < num_packets; ++i) {
+    const MediaPacket media_packet(kFlowId, 0, 0, sequence_numbers[i]);
+    // Arrival time = 0, all packets will be considered.
+    bwe_receiver_.ReceivePacket(0, media_packet);
+  }
+
+  EXPECT_EQ(bwe_receiver_.RecentPacketLossRatio(), 0.0f);
+}
+
+TEST_F(BweReceiverTest, RecentKbps) {
+  EXPECT_EQ(bwe_receiver_.RecentKbps(), 0U);
+
+  const size_t kPacketSizeBytes = 1200;
+  const int kNumPackets = 100;
+
+  double window_size_s = bwe_receiver_.BitrateWindowS();
+
+  // Receive packets at the same time.
+  for (int i = 0; i < kNumPackets; ++i) {
+    MediaPacket packet(kFlowId, 0L, kPacketSizeBytes, static_cast<uint16_t>(i));
+    bwe_receiver_.ReceivePacket(0, packet);
+  }
+
+  EXPECT_NEAR(bwe_receiver_.RecentKbps(),
+              (8 * kNumPackets * kPacketSizeBytes) / (1000 * window_size_s),
+              10);
+
+  int64_t time_gap_ms =
+      2 * 1000 * window_size_s;  // Larger than rate_counter time window.
+
+  MediaPacket packet(kFlowId, time_gap_ms * 1000, kPacketSizeBytes,
+                     static_cast<uint16_t>(kNumPackets));
+  bwe_receiver_.ReceivePacket(time_gap_ms, packet);
+
+  EXPECT_NEAR(bwe_receiver_.RecentKbps(),
+              (8 * kPacketSizeBytes) / (1000 * window_size_s), 10);
+}
+
+TEST_F(BweReceiverTest, Loss) {
+  EXPECT_NEAR(bwe_receiver_.GlobalReceiverPacketLossRatio(), 0.0f, 0.001f);
+
+  LossAccount loss_account = bwe_receiver_.LinkedSetPacketLossRatio();
+  EXPECT_NEAR(loss_account.LossRatio(), 0.0f, 0.001f);
+
+  // Insert packets 1-50 and 151-200;
+  for (int i = 1; i <= 200; ++i) {
+    // Packet size and timestamp do not matter here.
+    MediaPacket packet(kFlowId, 0L, 0UL, static_cast<uint16_t>(i));
+    bwe_receiver_.ReceivePacket(0, packet);
+    if (i == 50) {
+      i += 100;
+    }
+  }
+
+  loss_account = bwe_receiver_.LinkedSetPacketLossRatio();
+  EXPECT_NEAR(loss_account.LossRatio(), 0.5f, 0.001f);
+
+  bwe_receiver_.RelieveSetAndUpdateLoss();
+  EXPECT_EQ(bwe_receiver_.received_packets_.size(), 100U / 10);
+
+  // No packet loss within the preserved packets.
+  loss_account = bwe_receiver_.LinkedSetPacketLossRatio();
+  EXPECT_NEAR(loss_account.LossRatio(), 0.0f, 0.001f);
+
+  // RelieveSetAndUpdateLoss automatically updates loss account.
+  EXPECT_NEAR(bwe_receiver_.GlobalReceiverPacketLossRatio(), 0.5f, 0.001f);
+}
+
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
index 32f772c02f..6166ff8c2d 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
@@ -6,26 +6,41 @@
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
+ *
  */
 
+//  Implementation of Network-Assisted Dynamic Adaptation's (NADA's) proposal.
+//  Version according to Draft Document (mentioned in references)
+//  http://tools.ietf.org/html/draft-zhu-rmcat-nada-06
+//  From March 26, 2015.
+
+#include <math.h>
 #include <algorithm>
+#include <vector>
 
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
-
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
+const int64_t NadaBweReceiver::kReceivingRateTimeWindowMs = 500;
+
 NadaBweReceiver::NadaBweReceiver(int flow_id)
-    : BweReceiver(flow_id),
+    : BweReceiver(flow_id, kReceivingRateTimeWindowMs),
       clock_(0),
       last_feedback_ms_(0),
       recv_stats_(ReceiveStatistics::Create(&clock_)),
-      baseline_delay_ms_(0),
+      baseline_delay_ms_(10000),  // Initialized as an upper bound.
       delay_signal_ms_(0),
-      last_congestion_signal_ms_(0) {
+      last_congestion_signal_ms_(0),
+      last_delays_index_(0),
+      exp_smoothed_delay_ms_(-1),
+      est_queuing_delay_signal_ms_(0) {
 }
 
 NadaBweReceiver::~NadaBweReceiver() {
@@ -33,34 +48,59 @@ NadaBweReceiver::~NadaBweReceiver() {
 
 void NadaBweReceiver::ReceivePacket(int64_t arrival_time_ms,
                                     const MediaPacket& media_packet) {
+  const float kAlpha = 0.1f;                 // Used for exponential smoothing.
+  const int64_t kDelayLowThresholdMs = 50;   // Referred as d_th.
+  const int64_t kDelayMaxThresholdMs = 400;  // Referred as d_max.
+
   clock_.AdvanceTimeMilliseconds(arrival_time_ms - clock_.TimeInMilliseconds());
   recv_stats_->IncomingPacket(media_packet.header(),
                               media_packet.payload_size(), false);
-  int64_t delay_ms = arrival_time_ms - media_packet.creation_time_us() / 1000;
-  // TODO(holmer): The min should time out after 10 minutes.
-  if (delay_ms < baseline_delay_ms_) {
-    baseline_delay_ms_ = delay_ms;
+  // Refered as x_n.
+  int64_t delay_ms = arrival_time_ms - media_packet.sender_timestamp_ms();
+
+  // The min should be updated within the first 10 minutes.
+  if (clock_.TimeInMilliseconds() < 10 * 60 * 1000) {
+    baseline_delay_ms_ = std::min(baseline_delay_ms_, delay_ms);
   }
-  delay_signal_ms_ = delay_ms - baseline_delay_ms_;
+
+  delay_signal_ms_ = delay_ms - baseline_delay_ms_;  // Refered as d_n.
+  const int kMedian = arraysize(last_delays_ms_);
+  last_delays_ms_[(last_delays_index_++) % kMedian] = delay_signal_ms_;
+  int size = std::min(last_delays_index_, kMedian);
+
+  int64_t median_filtered_delay_ms_ = MedianFilter(last_delays_ms_, size);
+  exp_smoothed_delay_ms_ = ExponentialSmoothingFilter(
+      median_filtered_delay_ms_, exp_smoothed_delay_ms_, kAlpha);
+
+  if (exp_smoothed_delay_ms_ < kDelayLowThresholdMs) {
+    est_queuing_delay_signal_ms_ = exp_smoothed_delay_ms_;
+  } else if (exp_smoothed_delay_ms_ < kDelayMaxThresholdMs) {
+    est_queuing_delay_signal_ms_ = static_cast<int64_t>(
+        pow((static_cast<double>(kDelayMaxThresholdMs -
+                                 exp_smoothed_delay_ms_)) /
+                (kDelayMaxThresholdMs - kDelayLowThresholdMs),
+            4.0) *
+        kDelayLowThresholdMs);
+  } else {
+    est_queuing_delay_signal_ms_ = 0;
+  }
+
+  // Log received packet information.
+  BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
 }
 
 FeedbackPacket* NadaBweReceiver::GetFeedback(int64_t now_ms) {
-  if (now_ms - last_feedback_ms_ < 100)
-    return NULL;
+  const int64_t kPacketLossPenaltyMs = 1000;  // Referred as d_L.
 
-  StatisticianMap statisticians = recv_stats_->GetActiveStatisticians();
-  int64_t loss_signal_ms = 0.0f;
-  if (!statisticians.empty()) {
-    RtcpStatistics stats;
-    if (!statisticians.begin()->second->GetStatistics(&stats, true)) {
-      const float kLossSignalWeight = 1000.0f;
-      loss_signal_ms =
-          (kLossSignalWeight * static_cast<float>(stats.fraction_lost) + 127) /
-          255;
-    }
+  if (now_ms - last_feedback_ms_ < 100) {
+    return NULL;
   }
 
-  int64_t congestion_signal_ms = delay_signal_ms_ + loss_signal_ms;
+  float loss_fraction = RecentPacketLossRatio();
+
+  int64_t loss_signal_ms =
+      static_cast<int64_t>(loss_fraction * kPacketLossPenaltyMs + 0.5f);
+  int64_t congestion_signal_ms = est_queuing_delay_signal_ms_ + loss_signal_ms;
 
   float derivative = 0.0f;
   if (last_feedback_ms_ > 0) {
@@ -69,14 +109,61 @@ FeedbackPacket* NadaBweReceiver::GetFeedback(int64_t now_ms) {
   }
   last_feedback_ms_ = now_ms;
   last_congestion_signal_ms_ = congestion_signal_ms;
-  return new NadaFeedback(flow_id_, now_ms, congestion_signal_ms, derivative);
+
+  int64_t corrected_send_time_ms = 0L;
+
+  if (!received_packets_.empty()) {
+    PacketIdentifierNode* latest = *(received_packets_.begin());
+    corrected_send_time_ms =
+        latest->send_time_ms + now_ms - latest->arrival_time_ms;
+  }
+
+  // Sends a tuple containing latest values of <d_hat_n, d_tilde_n, x_n, x'_n,
+  // R_r> and additional information.
+  return new NadaFeedback(flow_id_, now_ms * 1000, exp_smoothed_delay_ms_,
+                          est_queuing_delay_signal_ms_, congestion_signal_ms,
+                          derivative, RecentKbps(), corrected_send_time_ms);
 }
 
+// If size is even, the median is the average of the two middlemost numbers.
+int64_t NadaBweReceiver::MedianFilter(int64_t* last_delays_ms, int size) {
+  std::vector<int64_t> array_copy(last_delays_ms, last_delays_ms + size);
+  std::nth_element(array_copy.begin(), array_copy.begin() + size / 2,
+                   array_copy.end());
+  if (size % 2 == 1) {
+    // Typically, size = 5. For odd size values, right and left are equal.
+    return array_copy.at(size / 2);
+  }
+  int64_t right = array_copy.at(size / 2);
+  std::nth_element(array_copy.begin(), array_copy.begin() + (size - 1) / 2,
+                   array_copy.end());
+  int64_t left = array_copy.at((size - 1) / 2);
+  return (left + right + 1) / 2;
+}
+
+int64_t NadaBweReceiver::ExponentialSmoothingFilter(int64_t new_value,
+                                                    int64_t last_smoothed_value,
+                                                    float alpha) {
+  if (last_smoothed_value < 0) {
+    return new_value;  // Handling initial case.
+  }
+  return static_cast<int64_t>(alpha * new_value +
+                              (1.0f - alpha) * last_smoothed_value + 0.5f);
+}
+
+// Implementation according to Cisco's proposal by default.
 NadaBweSender::NadaBweSender(int kbps, BitrateObserver* observer, Clock* clock)
-    : clock_(clock),
+    : BweSender(kbps),  // Referred as "Reference Rate" = R_n.,
+      clock_(clock),
       observer_(observer),
-      bitrate_kbps_(kbps),
-      last_feedback_ms_(0) {
+      original_operating_mode_(true) {
+}
+
+NadaBweSender::NadaBweSender(BitrateObserver* observer, Clock* clock)
+    : BweSender(kMinBitrateKbps),  // Referred as "Reference Rate" = R_n.
+      clock_(clock),
+      observer_(observer),
+      original_operating_mode_(true) {
 }
 
 NadaBweSender::~NadaBweSender() {
@@ -89,30 +176,61 @@ int NadaBweSender::GetFeedbackIntervalMs() const {
 void NadaBweSender::GiveFeedback(const FeedbackPacket& feedback) {
   const NadaFeedback& fb = static_cast<const NadaFeedback&>(feedback);
 
-  // TODO(holmer): Implement special start-up behavior.
-
-  const float kEta = 2.0f;
-  const float kTaoO = 500.0f;
-  float x_hat = fb.congestion_signal() + kEta * kTaoO * fb.derivative();
+  // Following parameters might be optimized.
+  const int64_t kQueuingDelayUpperBoundMs = 10;
+  const float kDerivativeUpperBound = 10.0f / min_feedback_delay_ms_;
+  // In the modified version, a higher kMinUpperBound allows a higher d_hat
+  // upper bound for calling AcceleratedRampUp.
+  const float kProportionalityDelayBits = 20.0f;
 
   int64_t now_ms = clock_->TimeInMilliseconds();
   float delta_s = now_ms - last_feedback_ms_;
   last_feedback_ms_ = now_ms;
+  // Update delta_0.
+  min_feedback_delay_ms_ =
+      std::min(min_feedback_delay_ms_, static_cast<int64_t>(delta_s));
 
-  const float kPriorityWeight = 1.0f;
-  const float kReferenceDelayS = 10.0f;
-  float kTheta =
-      kPriorityWeight * (kMaxBitrateKbps - kMinBitrateKbps) * kReferenceDelayS;
+  // Update RTT_0.
+  int64_t rtt_ms = now_ms - fb.latest_send_time_ms();
+  min_round_trip_time_ms_ = std::min(min_round_trip_time_ms_, rtt_ms);
+
+  // Independent limits for AcceleratedRampUp conditions variables:
+  // x_n, d_tilde and x'_n in the original implementation, plus
+  // d_hat and receiving_rate in the modified one.
+  // There should be no packet losses/marking, hence x_n == d_tilde.
+  if (original_operating_mode_) {
+    // Original if conditions and rate update.
+    if (fb.congestion_signal() == fb.est_queuing_delay_signal_ms() &&
+        fb.est_queuing_delay_signal_ms() < kQueuingDelayUpperBoundMs &&
+        fb.derivative() < kDerivativeUpperBound) {
+      AcceleratedRampUp(fb);
+    } else {
+      GradualRateUpdate(fb, delta_s, 1.0);
+    }
+  } else {
+    // Modified if conditions and rate update; new ramp down mode.
+    if (fb.congestion_signal() == fb.est_queuing_delay_signal_ms() &&
+        fb.est_queuing_delay_signal_ms() < kQueuingDelayUpperBoundMs &&
+        fb.exp_smoothed_delay_ms() <
+            kMinBitrateKbps / kProportionalityDelayBits &&
+        fb.derivative() < kDerivativeUpperBound &&
+        fb.receiving_rate() > kMinBitrateKbps) {
+      AcceleratedRampUp(fb);
+    } else if (fb.congestion_signal() > kMaxCongestionSignalMs ||
+               fb.exp_smoothed_delay_ms() > kMaxCongestionSignalMs) {
+      AcceleratedRampDown(fb);
+    } else {
+      double bitrate_reference =
+          (2.0 * bitrate_kbps_) / (kMaxBitrateKbps + kMinBitrateKbps);
+      double smoothing_factor = pow(bitrate_reference, 0.75);
+      GradualRateUpdate(fb, delta_s, smoothing_factor);
+    }
+  }
 
-  const float kKappa = 1.0f;
-  bitrate_kbps_ = bitrate_kbps_ +
-                  kKappa * delta_s / (kTaoO * kTaoO) *
-                      (kTheta - (bitrate_kbps_ - kMinBitrateKbps) * x_hat) +
-                  0.5f;
   bitrate_kbps_ = std::min(bitrate_kbps_, kMaxBitrateKbps);
   bitrate_kbps_ = std::max(bitrate_kbps_, kMinBitrateKbps);
 
-  observer_->OnNetworkChanged(1000 * bitrate_kbps_, 0, 0);
+  observer_->OnNetworkChanged(1000 * bitrate_kbps_, 0, rtt_ms);
 }
 
 int64_t NadaBweSender::TimeUntilNextProcess() {
@@ -123,6 +241,48 @@ int NadaBweSender::Process() {
   return 0;
 }
 
+void NadaBweSender::AcceleratedRampUp(const NadaFeedback& fb) {
+  const int kMaxRampUpQueuingDelayMs = 50;  // Referred as T_th.
+  const float kGamma0 = 0.5f;               // Referred as gamma_0.
+
+  float gamma =
+      std::min(kGamma0, static_cast<float>(kMaxRampUpQueuingDelayMs) /
+                            (min_round_trip_time_ms_ + min_feedback_delay_ms_));
+
+  bitrate_kbps_ = static_cast<int>((1.0f + gamma) * fb.receiving_rate() + 0.5f);
+}
+
+void NadaBweSender::AcceleratedRampDown(const NadaFeedback& fb) {
+  const float kGamma0 = 0.9f;
+  float gamma = 3.0f * kMaxCongestionSignalMs /
+                (fb.congestion_signal() + fb.exp_smoothed_delay_ms());
+  gamma = std::min(gamma, kGamma0);
+  bitrate_kbps_ = gamma * fb.receiving_rate() + 0.5f;
+}
+
+void NadaBweSender::GradualRateUpdate(const NadaFeedback& fb,
+                                      float delta_s,
+                                      double smoothing_factor) {
+  const float kTauOMs = 500.0f;           // Referred as tau_o.
+  const float kEta = 2.0f;                // Referred as eta.
+  const float kKappa = 1.0f;              // Referred as kappa.
+  const float kReferenceDelayMs = 10.0f;  // Referred as x_ref.
+  const float kPriorityWeight = 1.0f;     // Referred as w.
+
+  float x_hat = fb.congestion_signal() + kEta * kTauOMs * fb.derivative();
+
+  float kTheta =
+      kPriorityWeight * (kMaxBitrateKbps - kMinBitrateKbps) * kReferenceDelayMs;
+
+  int original_increase =
+      static_cast<int>((kKappa * delta_s *
+                        (kTheta - (bitrate_kbps_ - kMinBitrateKbps) * x_hat)) /
+                           (kTauOMs * kTauOMs) +
+                       0.5f);
+
+  bitrate_kbps_ = bitrate_kbps_ + smoothing_factor * original_increase;
+}
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
index 14cc8a2733..bf23d09884 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
@@ -6,12 +6,23 @@
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
- */
+ *
+*/
+
+//  Implementation of Network-Assisted Dynamic Adaptation's (NADA's) proposal
+//  Version according to Draft Document (mentioned in references)
+//  http://tools.ietf.org/html/draft-zhu-rmcat-nada-06
+//  From March 26, 2015.
 
 #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_ESTIMATORS_NADA_H_
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_ESTIMATORS_NADA_H_
 
+#include <list>
+#include <map>
+
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
+#include "webrtc/voice_engine/channel.h"
 
 namespace webrtc {
 
@@ -29,33 +40,66 @@ class NadaBweReceiver : public BweReceiver {
                      const MediaPacket& media_packet) override;
   FeedbackPacket* GetFeedback(int64_t now_ms) override;
 
+  static int64_t MedianFilter(int64_t* v, int size);
+  static int64_t ExponentialSmoothingFilter(int64_t new_value,
+                                            int64_t last_smoothed_value,
+                                            float alpha);
+
+  static const int64_t kReceivingRateTimeWindowMs;
+
  private:
   SimulatedClock clock_;
   int64_t last_feedback_ms_;
   rtc::scoped_ptr<ReceiveStatistics> recv_stats_;
-  int64_t baseline_delay_ms_;
-  int64_t delay_signal_ms_;
+  int64_t baseline_delay_ms_;  // Referred as d_f.
+  int64_t delay_signal_ms_;    // Referred as d_n.
   int64_t last_congestion_signal_ms_;
+  int last_delays_index_;
+  int64_t exp_smoothed_delay_ms_;        // Referred as d_hat_n.
+  int64_t est_queuing_delay_signal_ms_;  // Referred as d_tilde_n.
+  int64_t last_delays_ms_[5];            // Used for Median Filter.
 };
 
 class NadaBweSender : public BweSender {
  public:
   NadaBweSender(int kbps, BitrateObserver* observer, Clock* clock);
+  NadaBweSender(BitrateObserver* observer, Clock* clock);
   virtual ~NadaBweSender();
 
   int GetFeedbackIntervalMs() const override;
+  // Updates the min_feedback_delay_ms_ and the min_round_trip_time_ms_.
   void GiveFeedback(const FeedbackPacket& feedback) override;
   void OnPacketsSent(const Packets& packets) override {}
   int64_t TimeUntilNextProcess() override;
   int Process() override;
+  void AcceleratedRampUp(const NadaFeedback& fb);
+  void AcceleratedRampDown(const NadaFeedback& fb);
+  void GradualRateUpdate(const NadaFeedback& fb,
+                         float delta_s,
+                         double smoothing_factor);
+
+  int bitrate_kbps() const { return bitrate_kbps_; }
+  void set_bitrate_kbps(int bitrate_kbps) { bitrate_kbps_ = bitrate_kbps; }
+  bool original_operating_mode() const { return original_operating_mode_; }
+  void set_original_operating_mode(bool original_operating_mode) {
+    original_operating_mode_ = original_operating_mode;
+  }
+  int64_t NowMs() const { return clock_->TimeInMilliseconds(); }
 
  private:
   Clock* const clock_;
   BitrateObserver* const observer_;
-  int bitrate_kbps_;
-  int64_t last_feedback_ms_;
+  // Used as an upper bound for calling AcceleratedRampDown.
+  const float kMaxCongestionSignalMs = 40.0f + kMinBitrateKbps / 15;
+  // Referred as R_min, default initialization for bitrate R_n.
+  int64_t last_feedback_ms_ = 0;
+  // Referred as delta_0, initialized as an upper bound.
+  int64_t min_feedback_delay_ms_ = 200;
+  // Referred as RTT_0, initialized as an upper bound.
+  int64_t min_round_trip_time_ms_ = 100;
+  bool original_operating_mode_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(NadaBweSender);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NadaBweSender);
 };
 
 }  // namespace bwe
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
new file mode 100644
index 0000000000..51afae1df4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
@@ -0,0 +1,496 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
+
+#include <algorithm>
+#include <numeric>
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/common.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+class FilterTest : public ::testing::Test {
+ public:
+  void MedianFilterConstantArray() {
+    std::fill_n(raw_signal_, kNumElements, kSignalValue);
+    for (int i = 0; i < kNumElements; ++i) {
+      int size = std::min(5, i + 1);
+      median_filtered_[i] =
+          NadaBweReceiver::MedianFilter(&raw_signal_[i + 1 - size], size);
+    }
+  }
+
+  void MedianFilterIntermittentNoise() {
+    const int kValue = 500;
+    const int kNoise = 100;
+
+    for (int i = 0; i < kNumElements; ++i) {
+      raw_signal_[i] = kValue + kNoise * (i % 10 == 9 ? 1 : 0);
+    }
+    for (int i = 0; i < kNumElements; ++i) {
+      int size = std::min(5, i + 1);
+      median_filtered_[i] =
+          NadaBweReceiver::MedianFilter(&raw_signal_[i + 1 - size], size);
+      EXPECT_EQ(median_filtered_[i], kValue);
+    }
+  }
+
+  void ExponentialSmoothingFilter(const int64_t raw_signal_[],
+                                  int num_elements,
+                                  int64_t exp_smoothed[]) {
+    exp_smoothed[0] =
+        NadaBweReceiver::ExponentialSmoothingFilter(raw_signal_[0], -1, kAlpha);
+    for (int i = 1; i < num_elements; ++i) {
+      exp_smoothed[i] = NadaBweReceiver::ExponentialSmoothingFilter(
+          raw_signal_[i], exp_smoothed[i - 1], kAlpha);
+    }
+  }
+
+  void ExponentialSmoothingConstantArray(int64_t exp_smoothed[]) {
+    std::fill_n(raw_signal_, kNumElements, kSignalValue);
+    ExponentialSmoothingFilter(raw_signal_, kNumElements, exp_smoothed);
+  }
+
+ protected:
+  static const int kNumElements = 1000;
+  static const int64_t kSignalValue;
+  static const float kAlpha;
+  int64_t raw_signal_[kNumElements];
+  int64_t median_filtered_[kNumElements];
+};
+
+const int64_t FilterTest::kSignalValue = 200;
+const float FilterTest::kAlpha = 0.1f;
+
+class TestBitrateObserver : public BitrateObserver {
+ public:
+  TestBitrateObserver()
+      : last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
+
+  virtual void OnNetworkChanged(uint32_t bitrate,
+                                uint8_t fraction_loss,
+                                int64_t rtt) {
+    last_bitrate_ = bitrate;
+    last_fraction_loss_ = fraction_loss;
+    last_rtt_ = rtt;
+  }
+  uint32_t last_bitrate_;
+  uint8_t last_fraction_loss_;
+  int64_t last_rtt_;
+};
+
+class NadaSenderSideTest : public ::testing::Test {
+ public:
+  NadaSenderSideTest()
+      : observer_(),
+        simulated_clock_(0),
+        nada_sender_(&observer_, &simulated_clock_) {}
+  ~NadaSenderSideTest() {}
+
+ private:
+  TestBitrateObserver observer_;
+  SimulatedClock simulated_clock_;
+
+ protected:
+  NadaBweSender nada_sender_;
+};
+
+class NadaReceiverSideTest : public ::testing::Test {
+ public:
+  NadaReceiverSideTest() : nada_receiver_(kFlowId) {}
+  ~NadaReceiverSideTest() {}
+
+ protected:
+  const int kFlowId = 1;  // Arbitrary.
+  NadaBweReceiver nada_receiver_;
+};
+
+class NadaFbGenerator {
+ public:
+  NadaFbGenerator();
+
+  static NadaFeedback NotCongestedFb(size_t receiving_rate,
+                                     int64_t ref_signal_ms,
+                                     int64_t send_time_ms) {
+    int64_t exp_smoothed_delay_ms = ref_signal_ms;
+    int64_t est_queuing_delay_signal_ms = ref_signal_ms;
+    int64_t congestion_signal_ms = ref_signal_ms;
+    float derivative = 0.0f;
+    return NadaFeedback(kFlowId, kNowMs, exp_smoothed_delay_ms,
+                        est_queuing_delay_signal_ms, congestion_signal_ms,
+                        derivative, receiving_rate, send_time_ms);
+  }
+
+  static NadaFeedback CongestedFb(size_t receiving_rate, int64_t send_time_ms) {
+    int64_t exp_smoothed_delay_ms = 1000;
+    int64_t est_queuing_delay_signal_ms = 800;
+    int64_t congestion_signal_ms = 1000;
+    float derivative = 1.0f;
+    return NadaFeedback(kFlowId, kNowMs, exp_smoothed_delay_ms,
+                        est_queuing_delay_signal_ms, congestion_signal_ms,
+                        derivative, receiving_rate, send_time_ms);
+  }
+
+  static NadaFeedback ExtremelyCongestedFb(size_t receiving_rate,
+                                           int64_t send_time_ms) {
+    int64_t exp_smoothed_delay_ms = 100000;
+    int64_t est_queuing_delay_signal_ms = 0;
+    int64_t congestion_signal_ms = 100000;
+    float derivative = 10000.0f;
+    return NadaFeedback(kFlowId, kNowMs, exp_smoothed_delay_ms,
+                        est_queuing_delay_signal_ms, congestion_signal_ms,
+                        derivative, receiving_rate, send_time_ms);
+  }
+
+ private:
+  // Arbitrary values, won't change these test results.
+  static const int kFlowId = 2;
+  static const int64_t kNowMs = 1000;
+};
+
+// Verify if AcceleratedRampUp is called and that bitrate increases.
+TEST_F(NadaSenderSideTest, AcceleratedRampUp) {
+  const int64_t kRefSignalMs = 1;
+  const int64_t kOneWayDelayMs = 50;
+  int original_bitrate = 2 * kMinBitrateKbps;
+  size_t receiving_rate = static_cast<size_t>(original_bitrate);
+  int64_t send_time_ms = nada_sender_.NowMs() - kOneWayDelayMs;
+
+  NadaFeedback not_congested_fb = NadaFbGenerator::NotCongestedFb(
+      receiving_rate, kRefSignalMs, send_time_ms);
+
+  nada_sender_.set_original_operating_mode(true);
+  nada_sender_.set_bitrate_kbps(original_bitrate);
+
+  // Trigger AcceleratedRampUp mode.
+  nada_sender_.GiveFeedback(not_congested_fb);
+  int bitrate_1_kbps = nada_sender_.bitrate_kbps();
+  EXPECT_GT(bitrate_1_kbps, original_bitrate);
+  // Updates the bitrate according to the receiving rate and other constant
+  // parameters.
+  nada_sender_.AcceleratedRampUp(not_congested_fb);
+  EXPECT_EQ(nada_sender_.bitrate_kbps(), bitrate_1_kbps);
+
+  nada_sender_.set_original_operating_mode(false);
+  nada_sender_.set_bitrate_kbps(original_bitrate);
+  // Trigger AcceleratedRampUp mode.
+  nada_sender_.GiveFeedback(not_congested_fb);
+  bitrate_1_kbps = nada_sender_.bitrate_kbps();
+  EXPECT_GT(bitrate_1_kbps, original_bitrate);
+  nada_sender_.AcceleratedRampUp(not_congested_fb);
+  EXPECT_EQ(nada_sender_.bitrate_kbps(), bitrate_1_kbps);
+}
+
+// Verify if AcceleratedRampDown is called and if bitrate decreases.
+TEST_F(NadaSenderSideTest, AcceleratedRampDown) {
+  const int64_t kOneWayDelayMs = 50;
+  int original_bitrate = 3 * kMinBitrateKbps;
+  size_t receiving_rate = static_cast<size_t>(original_bitrate);
+  int64_t send_time_ms = nada_sender_.NowMs() - kOneWayDelayMs;
+
+  NadaFeedback congested_fb =
+      NadaFbGenerator::CongestedFb(receiving_rate, send_time_ms);
+
+  nada_sender_.set_original_operating_mode(false);
+  nada_sender_.set_bitrate_kbps(original_bitrate);
+  nada_sender_.GiveFeedback(congested_fb);  // Trigger AcceleratedRampDown mode.
+  int bitrate_1_kbps = nada_sender_.bitrate_kbps();
+  EXPECT_LE(bitrate_1_kbps, original_bitrate * 0.9f + 0.5f);
+  EXPECT_LT(bitrate_1_kbps, original_bitrate);
+
+  // Updates the bitrate according to the receiving rate and other constant
+  // parameters.
+  nada_sender_.AcceleratedRampDown(congested_fb);
+  int bitrate_2_kbps = std::max(nada_sender_.bitrate_kbps(), kMinBitrateKbps);
+  EXPECT_EQ(bitrate_2_kbps, bitrate_1_kbps);
+}
+
+TEST_F(NadaSenderSideTest, GradualRateUpdate) {
+  const int64_t kDeltaSMs = 20;
+  const int64_t kRefSignalMs = 20;
+  const int64_t kOneWayDelayMs = 50;
+  int original_bitrate = 2 * kMinBitrateKbps;
+  size_t receiving_rate = static_cast<size_t>(original_bitrate);
+  int64_t send_time_ms = nada_sender_.NowMs() - kOneWayDelayMs;
+
+  NadaFeedback congested_fb =
+      NadaFbGenerator::CongestedFb(receiving_rate, send_time_ms);
+  NadaFeedback not_congested_fb = NadaFbGenerator::NotCongestedFb(
+      original_bitrate, kRefSignalMs, send_time_ms);
+
+  nada_sender_.set_bitrate_kbps(original_bitrate);
+  double smoothing_factor = 0.0;
+  nada_sender_.GradualRateUpdate(congested_fb, kDeltaSMs, smoothing_factor);
+  EXPECT_EQ(nada_sender_.bitrate_kbps(), original_bitrate);
+
+  smoothing_factor = 1.0;
+  nada_sender_.GradualRateUpdate(congested_fb, kDeltaSMs, smoothing_factor);
+  EXPECT_LT(nada_sender_.bitrate_kbps(), original_bitrate);
+
+  nada_sender_.set_bitrate_kbps(original_bitrate);
+  nada_sender_.GradualRateUpdate(not_congested_fb, kDeltaSMs, smoothing_factor);
+  EXPECT_GT(nada_sender_.bitrate_kbps(), original_bitrate);
+}
+
+// Sending bitrate should decrease and reach its Min bound.
+TEST_F(NadaSenderSideTest, VeryLowBandwith) {
+  const int64_t kOneWayDelayMs = 50;
+
+  size_t receiving_rate = static_cast<size_t>(kMinBitrateKbps);
+  int64_t send_time_ms = nada_sender_.NowMs() - kOneWayDelayMs;
+
+  NadaFeedback extremely_congested_fb =
+      NadaFbGenerator::ExtremelyCongestedFb(receiving_rate, send_time_ms);
+  NadaFeedback congested_fb =
+      NadaFbGenerator::CongestedFb(receiving_rate, send_time_ms);
+
+  nada_sender_.set_bitrate_kbps(5 * kMinBitrateKbps);
+  nada_sender_.set_original_operating_mode(true);
+  for (int i = 0; i < 100; ++i) {
+    // Trigger GradualRateUpdate mode.
+    nada_sender_.GiveFeedback(extremely_congested_fb);
+  }
+  // The original implementation doesn't allow the bitrate to stay at kMin,
+  // even if the congestion signal is very high.
+  EXPECT_GE(nada_sender_.bitrate_kbps(), kMinBitrateKbps);
+
+  nada_sender_.set_original_operating_mode(false);
+  nada_sender_.set_bitrate_kbps(5 * kMinBitrateKbps);
+
+  for (int i = 0; i < 1000; ++i) {
+    int previous_bitrate = nada_sender_.bitrate_kbps();
+    // Trigger AcceleratedRampDown mode.
+    nada_sender_.GiveFeedback(congested_fb);
+    EXPECT_LE(nada_sender_.bitrate_kbps(), previous_bitrate);
+  }
+  EXPECT_EQ(nada_sender_.bitrate_kbps(), kMinBitrateKbps);
+}
+
+// Sending bitrate should increase and reach its Max bound.
+TEST_F(NadaSenderSideTest, VeryHighBandwith) {
+  const int64_t kOneWayDelayMs = 50;
+  const size_t kRecentReceivingRate = static_cast<size_t>(kMaxBitrateKbps);
+  const int64_t kRefSignalMs = 1;
+  int64_t send_time_ms = nada_sender_.NowMs() - kOneWayDelayMs;
+
+  NadaFeedback not_congested_fb = NadaFbGenerator::NotCongestedFb(
+      kRecentReceivingRate, kRefSignalMs, send_time_ms);
+
+  nada_sender_.set_original_operating_mode(true);
+  for (int i = 0; i < 100; ++i) {
+    int previous_bitrate = nada_sender_.bitrate_kbps();
+    nada_sender_.GiveFeedback(not_congested_fb);
+    EXPECT_GE(nada_sender_.bitrate_kbps(), previous_bitrate);
+  }
+  EXPECT_EQ(nada_sender_.bitrate_kbps(), kMaxBitrateKbps);
+
+  nada_sender_.set_original_operating_mode(false);
+  nada_sender_.set_bitrate_kbps(kMinBitrateKbps);
+
+  for (int i = 0; i < 100; ++i) {
+    int previous_bitrate = nada_sender_.bitrate_kbps();
+    nada_sender_.GiveFeedback(not_congested_fb);
+    EXPECT_GE(nada_sender_.bitrate_kbps(), previous_bitrate);
+  }
+  EXPECT_EQ(nada_sender_.bitrate_kbps(), kMaxBitrateKbps);
+}
+
+TEST_F(NadaReceiverSideTest, FeedbackInitialCases) {
+  rtc::scoped_ptr<NadaFeedback> nada_feedback(
+      static_cast<NadaFeedback*>(nada_receiver_.GetFeedback(0)));
+  EXPECT_EQ(nada_feedback, nullptr);
+
+  nada_feedback.reset(
+      static_cast<NadaFeedback*>(nada_receiver_.GetFeedback(100)));
+  EXPECT_EQ(nada_feedback->exp_smoothed_delay_ms(), -1);
+  EXPECT_EQ(nada_feedback->est_queuing_delay_signal_ms(), 0L);
+  EXPECT_EQ(nada_feedback->congestion_signal(), 0L);
+  EXPECT_EQ(nada_feedback->derivative(), 0.0f);
+  EXPECT_EQ(nada_feedback->receiving_rate(), 0.0f);
+}
+
+TEST_F(NadaReceiverSideTest, FeedbackEmptyQueues) {
+  const int64_t kTimeGapMs = 50;  // Between each packet.
+  const int64_t kOneWayDelayMs = 50;
+
+  // No added latency, delay = kOneWayDelayMs.
+  for (int i = 1; i < 10; ++i) {
+    int64_t send_time_us = i * kTimeGapMs * 1000;
+    int64_t arrival_time_ms = send_time_us / 1000 + kOneWayDelayMs;
+    uint16_t sequence_number = static_cast<uint16_t>(i);
+    // Payload sizes are not important here.
+    const MediaPacket media_packet(kFlowId, send_time_us, 0, sequence_number);
+    nada_receiver_.ReceivePacket(arrival_time_ms, media_packet);
+  }
+
+  // Baseline delay will be equal kOneWayDelayMs.
+  rtc::scoped_ptr<NadaFeedback> nada_feedback(
+      static_cast<NadaFeedback*>(nada_receiver_.GetFeedback(500)));
+  EXPECT_EQ(nada_feedback->exp_smoothed_delay_ms(), 0L);
+  EXPECT_EQ(nada_feedback->est_queuing_delay_signal_ms(), 0L);
+  EXPECT_EQ(nada_feedback->congestion_signal(), 0L);
+  EXPECT_EQ(nada_feedback->derivative(), 0.0f);
+}
+
+TEST_F(NadaReceiverSideTest, FeedbackIncreasingDelay) {
+  // Since packets are 100ms apart, each one corresponds to a feedback.
+  const int64_t kTimeGapMs = 100;  // Between each packet.
+
+  // Raw delays are = [10 20 30 40 50 60 70 80] ms.
+  // Baseline delay will be 50 ms.
+  // Delay signals should be: [0 10 20 30 40 50 60 70] ms.
+  const int64_t kMedianFilteredDelaysMs[] = {0, 5, 10, 15, 20, 30, 40, 50};
+  const int kNumPackets = arraysize(kMedianFilteredDelaysMs);
+  const float kAlpha = 0.1f;  // Used for exponential smoothing.
+
+  int64_t exp_smoothed_delays_ms[kNumPackets];
+  exp_smoothed_delays_ms[0] = kMedianFilteredDelaysMs[0];
+
+  for (int i = 1; i < kNumPackets; ++i) {
+    exp_smoothed_delays_ms[i] = static_cast<int64_t>(
+        kAlpha * kMedianFilteredDelaysMs[i] +
+        (1.0f - kAlpha) * exp_smoothed_delays_ms[i - 1] + 0.5f);
+  }
+
+  for (int i = 0; i < kNumPackets; ++i) {
+    int64_t send_time_us = (i + 1) * kTimeGapMs * 1000;
+    int64_t arrival_time_ms = send_time_us / 1000 + 10 * (i + 1);
+    uint16_t sequence_number = static_cast<uint16_t>(i + 1);
+    // Payload sizes are not important here.
+    const MediaPacket media_packet(kFlowId, send_time_us, 0, sequence_number);
+    nada_receiver_.ReceivePacket(arrival_time_ms, media_packet);
+
+    rtc::scoped_ptr<NadaFeedback> nada_feedback(static_cast<NadaFeedback*>(
+        nada_receiver_.GetFeedback(arrival_time_ms)));
+    EXPECT_EQ(nada_feedback->exp_smoothed_delay_ms(),
+              exp_smoothed_delays_ms[i]);
+    // Since delay signals are lower than 50ms, they will not be non-linearly
+    // warped.
+    EXPECT_EQ(nada_feedback->est_queuing_delay_signal_ms(),
+              exp_smoothed_delays_ms[i]);
+    // Zero loss, congestion signal = queuing_delay
+    EXPECT_EQ(nada_feedback->congestion_signal(), exp_smoothed_delays_ms[i]);
+    if (i == 0) {
+      EXPECT_NEAR(nada_feedback->derivative(),
+                  static_cast<float>(exp_smoothed_delays_ms[i]) / kTimeGapMs,
+                  0.005f);
+    } else {
+      EXPECT_NEAR(nada_feedback->derivative(),
+                  static_cast<float>(exp_smoothed_delays_ms[i] -
+                                     exp_smoothed_delays_ms[i - 1]) /
+                      kTimeGapMs,
+                  0.005f);
+    }
+  }
+}
+
+int64_t Warp(int64_t input) {
+  const int64_t kMinThreshold = 50;   // Referred as d_th.
+  const int64_t kMaxThreshold = 400;  // Referred as d_max.
+  if (input < kMinThreshold) {
+    return input;
+  } else if (input < kMaxThreshold) {
+    return static_cast<int64_t>(
+        pow((static_cast<double>(kMaxThreshold - input)) /
+                (kMaxThreshold - kMinThreshold),
+            4.0) *
+        kMinThreshold);
+  } else {
+    return 0L;
+  }
+}
+
+TEST_F(NadaReceiverSideTest, FeedbackWarpedDelay) {
+  // Since packets are 100ms apart, each one corresponds to a feedback.
+  const int64_t kTimeGapMs = 100;  // Between each packet.
+
+  // Raw delays are = [50 250 450 650 850 1050 1250 1450] ms.
+  // Baseline delay will be 50 ms.
+  // Delay signals should be: [0 200 400 600 800 1000 1200 1400] ms.
+  const int64_t kMedianFilteredDelaysMs[] = {
+      0, 100, 200, 300, 400, 600, 800, 1000};
+  const int kNumPackets = arraysize(kMedianFilteredDelaysMs);
+  const float kAlpha = 0.1f;  // Used for exponential smoothing.
+
+  int64_t exp_smoothed_delays_ms[kNumPackets];
+  exp_smoothed_delays_ms[0] = kMedianFilteredDelaysMs[0];
+
+  for (int i = 1; i < kNumPackets; ++i) {
+    exp_smoothed_delays_ms[i] = static_cast<int64_t>(
+        kAlpha * kMedianFilteredDelaysMs[i] +
+        (1.0f - kAlpha) * exp_smoothed_delays_ms[i - 1] + 0.5f);
+  }
+
+  for (int i = 0; i < kNumPackets; ++i) {
+    int64_t send_time_us = (i + 1) * kTimeGapMs * 1000;
+    int64_t arrival_time_ms = send_time_us / 1000 + 50 + 200 * i;
+    uint16_t sequence_number = static_cast<uint16_t>(i + 1);
+    // Payload sizes are not important here.
+    const MediaPacket media_packet(kFlowId, send_time_us, 0, sequence_number);
+    nada_receiver_.ReceivePacket(arrival_time_ms, media_packet);
+
+    rtc::scoped_ptr<NadaFeedback> nada_feedback(static_cast<NadaFeedback*>(
+        nada_receiver_.GetFeedback(arrival_time_ms)));
+    EXPECT_EQ(nada_feedback->exp_smoothed_delay_ms(),
+              exp_smoothed_delays_ms[i]);
+    // Delays can be non-linearly warped.
+    EXPECT_EQ(nada_feedback->est_queuing_delay_signal_ms(),
+              Warp(exp_smoothed_delays_ms[i]));
+    // Zero loss, congestion signal = queuing_delay
+    EXPECT_EQ(nada_feedback->congestion_signal(),
+              Warp(exp_smoothed_delays_ms[i]));
+  }
+}
+
+TEST_F(FilterTest, MedianConstantArray) {
+  MedianFilterConstantArray();
+  for (int i = 0; i < kNumElements; ++i) {
+    EXPECT_EQ(median_filtered_[i], raw_signal_[i]);
+  }
+}
+
+TEST_F(FilterTest, MedianIntermittentNoise) {
+  MedianFilterIntermittentNoise();
+}
+
+TEST_F(FilterTest, ExponentialSmoothingConstantArray) {
+  int64_t exp_smoothed[kNumElements];
+  ExponentialSmoothingConstantArray(exp_smoothed);
+  for (int i = 0; i < kNumElements; ++i) {
+    EXPECT_EQ(exp_smoothed[i], kSignalValue);
+  }
+}
+
+TEST_F(FilterTest, ExponentialSmoothingInitialPertubation) {
+  const int64_t kSignal[] = {90000, 0, 0, 0, 0, 0};
+  const int kNumElements = arraysize(kSignal);
+  int64_t exp_smoothed[kNumElements];
+  ExponentialSmoothingFilter(kSignal, kNumElements, exp_smoothed);
+  for (int i = 1; i < kNumElements; ++i) {
+    EXPECT_EQ(
+        exp_smoothed[i],
+        static_cast<int64_t>(exp_smoothed[i - 1] * (1.0f - kAlpha) + 0.5f));
+  }
+}
+
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
index 18c63e3bb8..9599b01933 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
@@ -15,8 +15,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/common.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
@@ -67,16 +68,14 @@ RembReceiver::RembReceiver(int flow_id, bool plot)
       clock_(0),
       recv_stats_(ReceiveStatistics::Create(&clock_)),
       latest_estimate_bps_(-1),
-      estimator_(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
-          this,
-          &clock_,
-          kAimdControl,
-          kRemoteBitrateEstimatorMinBitrateBps)) {
+      last_feedback_ms_(-1),
+      estimator_(new RemoteBitrateEstimatorAbsSendTime(this, &clock_)) {
   std::stringstream ss;
   ss << "Estimate_" << flow_id_ << "#1";
   estimate_log_prefix_ = ss.str();
   // Default RTT in RemoteRateControl is 200 ms ; 50 ms is more realistic.
-  estimator_->OnRttUpdate(50);
+  estimator_->OnRttUpdate(50, 50);
+  estimator_->SetMinBitrate(kRemoteBitrateEstimatorMinBitrateBps);
 }
 
 RembReceiver::~RembReceiver() {
@@ -96,9 +95,12 @@ void RembReceiver::ReceivePacket(int64_t arrival_time_ms,
     step_ms = std::max<int64_t>(estimator_->TimeUntilNextProcess(), 0);
   }
   estimator_->IncomingPacket(arrival_time_ms, media_packet.payload_size(),
-                             media_packet.header());
+                             media_packet.header(), true);
   clock_.AdvanceTimeMilliseconds(arrival_time_ms - clock_.TimeInMilliseconds());
   ASSERT_TRUE(arrival_time_ms == clock_.TimeInMilliseconds());
+
+  // Log received packet information.
+  BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
 }
 
 FeedbackPacket* RembReceiver::GetFeedback(int64_t now_ms) {
@@ -111,14 +113,16 @@ FeedbackPacket* RembReceiver::GetFeedback(int64_t now_ms) {
     if (!statisticians.empty()) {
       report_block = BuildReportBlock(statisticians.begin()->second);
     }
-    feedback =
-        new RembFeedback(flow_id_, now_ms * 1000, estimated_bps, report_block);
+
+    feedback = new RembFeedback(flow_id_, now_ms * 1000, last_feedback_ms_,
+                                estimated_bps, report_block);
+    last_feedback_ms_ = now_ms;
 
     double estimated_kbps = static_cast<double>(estimated_bps) / 1000.0;
     RTC_UNUSED(estimated_kbps);
     if (plot_estimate_) {
-      BWE_TEST_LOGGING_PLOT(estimate_log_prefix_, clock_.TimeInMilliseconds(),
-                            estimated_kbps);
+      BWE_TEST_LOGGING_PLOT(0, estimate_log_prefix_,
+                            clock_.TimeInMilliseconds(), estimated_kbps);
     }
   }
   return feedback;
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
index 753f152598..7dfd7a8459 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
@@ -44,7 +44,7 @@ class RembBweSender : public BweSender {
  private:
   Clock* clock_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RembBweSender);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RembBweSender);
 };
 
 class RembReceiver : public BweReceiver, public RemoteBitrateObserver {
@@ -70,9 +70,10 @@ class RembReceiver : public BweReceiver, public RemoteBitrateObserver {
   SimulatedClock clock_;
   rtc::scoped_ptr<ReceiveStatistics> recv_stats_;
   int64_t latest_estimate_bps_;
+  int64_t last_feedback_ms_;
   rtc::scoped_ptr<RemoteBitrateEstimator> estimator_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RembReceiver);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RembReceiver);
 };
 
 }  // namespace bwe
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
index c890b6aa0f..8a7352874b 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
@@ -11,31 +11,37 @@
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h"
 
 #include "webrtc/base/logging.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
+const int kFeedbackIntervalMs = 50;
+
 FullBweSender::FullBweSender(int kbps, BitrateObserver* observer, Clock* clock)
     : bitrate_controller_(
           BitrateController::CreateBitrateController(clock, observer)),
-      rbe_(AbsoluteSendTimeRemoteBitrateEstimatorFactory()
-               .Create(this, clock, kAimdControl, 1000 * kMinBitrateKbps)),
+      rbe_(new RemoteBitrateEstimatorAbsSendTime(this, clock)),
       feedback_observer_(bitrate_controller_->CreateRtcpBandwidthObserver()),
       clock_(clock),
-      send_time_history_(10000) {
+      send_time_history_(clock_, 10000),
+      has_received_ack_(false),
+      last_acked_seq_num_(0) {
   assert(kbps >= kMinBitrateKbps);
   assert(kbps <= kMaxBitrateKbps);
   bitrate_controller_->SetStartBitrate(1000 * kbps);
   bitrate_controller_->SetMinMaxBitrate(1000 * kMinBitrateKbps,
                                         1000 * kMaxBitrateKbps);
+  rbe_->SetMinBitrate(1000 * kMinBitrateKbps);
 }
 
 FullBweSender::~FullBweSender() {
 }
 
 int FullBweSender::GetFeedbackIntervalMs() const {
-  return 100;
+  return kFeedbackIntervalMs;
 }
 
 void FullBweSender::GiveFeedback(const FeedbackPacket& feedback) {
@@ -43,39 +49,54 @@ void FullBweSender::GiveFeedback(const FeedbackPacket& feedback) {
       static_cast<const SendSideBweFeedback&>(feedback);
   if (fb.packet_feedback_vector().empty())
     return;
-  // TODO(sprang): Unconstify PacketInfo so we don't need temp copy?
   std::vector<PacketInfo> packet_feedback_vector(fb.packet_feedback_vector());
-  for (PacketInfo& packet : packet_feedback_vector) {
-    if (!send_time_history_.GetSendTime(packet.sequence_number,
-                                        &packet.send_time_ms, true)) {
+  for (PacketInfo& packet_info : packet_feedback_vector) {
+    if (!send_time_history_.GetInfo(&packet_info, true)) {
       LOG(LS_WARNING) << "Ack arrived too late.";
     }
   }
+
+  int64_t rtt_ms =
+      clock_->TimeInMilliseconds() - feedback.latest_send_time_ms();
+  rbe_->OnRttUpdate(rtt_ms, rtt_ms);
+  BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms);
+
   rbe_->IncomingPacketFeedbackVector(packet_feedback_vector);
-  // TODO(holmer): Handle losses in between feedback packets.
-  int expected_packets = fb.packet_feedback_vector().back().sequence_number -
-                         fb.packet_feedback_vector().front().sequence_number +
-                         1;
-  // Assuming no reordering for now.
-  if (expected_packets <= 0)
-    return;
-  int lost_packets = expected_packets - fb.packet_feedback_vector().size();
-  report_block_.fractionLost = (lost_packets << 8) / expected_packets;
-  report_block_.cumulativeLost += lost_packets;
-  ReportBlockList report_blocks;
-  report_blocks.push_back(report_block_);
-  feedback_observer_->OnReceivedRtcpReceiverReport(
-      report_blocks, 0, clock_->TimeInMilliseconds());
-  bitrate_controller_->Process();
+  if (has_received_ack_) {
+    int expected_packets = fb.packet_feedback_vector().back().sequence_number -
+                           last_acked_seq_num_;
+    // Assuming no reordering for now.
+    if (expected_packets > 0) {
+      int lost_packets = expected_packets -
+                         static_cast<int>(fb.packet_feedback_vector().size());
+      report_block_.fractionLost = (lost_packets << 8) / expected_packets;
+      report_block_.cumulativeLost += lost_packets;
+      report_block_.extendedHighSeqNum =
+          packet_feedback_vector.back().sequence_number;
+      ReportBlockList report_blocks;
+      report_blocks.push_back(report_block_);
+      feedback_observer_->OnReceivedRtcpReceiverReport(
+          report_blocks, rtt_ms, clock_->TimeInMilliseconds());
+    }
+    bitrate_controller_->Process();
+
+    last_acked_seq_num_ = LatestSequenceNumber(
+        packet_feedback_vector.back().sequence_number, last_acked_seq_num_);
+  } else {
+    last_acked_seq_num_ = packet_feedback_vector.back().sequence_number;
+    has_received_ack_ = true;
+  }
 }
 
 void FullBweSender::OnPacketsSent(const Packets& packets) {
   for (Packet* packet : packets) {
     if (packet->GetPacketType() == Packet::kMedia) {
       MediaPacket* media_packet = static_cast<MediaPacket*>(packet);
-      send_time_history_.AddAndRemoveOldSendTimes(
-          media_packet->header().sequenceNumber,
-          media_packet->GetAbsSendTimeInMs());
+      send_time_history_.AddAndRemoveOld(media_packet->header().sequenceNumber,
+                                         media_packet->payload_size(),
+                                         packet->paced());
+      send_time_history_.OnSentPacket(media_packet->header().sequenceNumber,
+                                      media_packet->sender_timestamp_ms());
     }
   }
 }
@@ -105,17 +126,22 @@ SendSideBweReceiver::~SendSideBweReceiver() {
 void SendSideBweReceiver::ReceivePacket(int64_t arrival_time_ms,
                                         const MediaPacket& media_packet) {
   packet_feedback_vector_.push_back(PacketInfo(
-      arrival_time_ms,
-      GetAbsSendTimeInMs(media_packet.header().extension.absoluteSendTime),
-      media_packet.header().sequenceNumber, media_packet.payload_size()));
+      -1, arrival_time_ms, media_packet.sender_timestamp_ms(),
+      media_packet.header().sequenceNumber, media_packet.payload_size(), true));
+
+  // Log received packet information.
+  BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
 }
 
 FeedbackPacket* SendSideBweReceiver::GetFeedback(int64_t now_ms) {
-  if (now_ms - last_feedback_ms_ < 100)
+  if (now_ms - last_feedback_ms_ < kFeedbackIntervalMs)
     return NULL;
   last_feedback_ms_ = now_ms;
-  FeedbackPacket* fb =
-      new SendSideBweFeedback(flow_id_, now_ms * 1000, packet_feedback_vector_);
+  int64_t corrected_send_time_ms =
+      packet_feedback_vector_.back().send_time_ms + now_ms -
+      packet_feedback_vector_.back().arrival_time_ms;
+  FeedbackPacket* fb = new SendSideBweFeedback(
+      flow_id_, now_ms * 1000, corrected_send_time_ms, packet_feedback_vector_);
   packet_feedback_vector_.clear();
   return fb;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
index 007ea4e0cc..ab9abc5cbc 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/modules/bitrate_controller/send_time_history.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 
 namespace webrtc {
@@ -42,8 +42,10 @@ class FullBweSender : public BweSender, public RemoteBitrateObserver {
   Clock* const clock_;
   RTCPReportBlock report_block_;
   SendTimeHistory send_time_history_;
+  bool has_received_ack_;
+  uint16_t last_acked_seq_num_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(FullBweSender);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FullBweSender);
 };
 
 class SendSideBweReceiver : public BweReceiver {
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
new file mode 100644
index 0000000000..b7e4f971fa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/common.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+TcpBweReceiver::TcpBweReceiver(int flow_id)
+    : BweReceiver(flow_id),
+      last_feedback_ms_(0),
+      latest_owd_ms_(0) {
+}
+
+TcpBweReceiver::~TcpBweReceiver() {
+}
+
+void TcpBweReceiver::ReceivePacket(int64_t arrival_time_ms,
+                                   const MediaPacket& media_packet) {
+  latest_owd_ms_ = arrival_time_ms - media_packet.sender_timestamp_ms() / 1000;
+  acks_.push_back(media_packet.header().sequenceNumber);
+
+  // Log received packet information.
+  BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
+}
+
+FeedbackPacket* TcpBweReceiver::GetFeedback(int64_t now_ms) {
+  int64_t corrected_send_time_ms = now_ms - latest_owd_ms_;
+  FeedbackPacket* fb =
+      new TcpFeedback(flow_id_, now_ms * 1000, corrected_send_time_ms, acks_);
+  last_feedback_ms_ = now_ms;
+  acks_.clear();
+  return fb;
+}
+
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h
new file mode 100644
index 0000000000..b33c93eef7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_ESTIMATORS_TCP_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_ESTIMATORS_TCP_H_
+
+#include <vector>
+
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+class TcpBweReceiver : public BweReceiver {
+ public:
+  explicit TcpBweReceiver(int flow_id);
+  virtual ~TcpBweReceiver();
+
+  void ReceivePacket(int64_t arrival_time_ms,
+                     const MediaPacket& media_packet) override;
+  FeedbackPacket* GetFeedback(int64_t now_ms) override;
+
+ private:
+  int64_t last_feedback_ms_;
+  int64_t latest_owd_ms_;
+  std::vector<uint16_t> acks_;
+};
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_ESTIMATORS_TCP_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
new file mode 100644
index 0000000000..559757c0eb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
@@ -0,0 +1,445 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
+
+#include <algorithm>
+
+#include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+namespace {
+// Holder mean, Manhattan distance for p=1, EuclidianNorm/sqrt(n) for p=2.
+template <typename T>
+double NormLp(T sum, size_t size, double p) {
+  return pow(sum / size, 1.0 / p);
+}
+}
+
+const double kP = 1.0;  // Used for Norm Lp.
+
+LinkShare::LinkShare(ChokeFilter* choke_filter)
+    : choke_filter_(choke_filter), running_flows_(choke_filter->flow_ids()) {
+}
+
+void LinkShare::PauseFlow(int flow_id) {
+  running_flows_.erase(flow_id);
+}
+
+void LinkShare::ResumeFlow(int flow_id) {
+  running_flows_.insert(flow_id);
+}
+
+uint32_t LinkShare::TotalAvailableKbps() {
+  return choke_filter_->capacity_kbps();
+}
+
+uint32_t LinkShare::AvailablePerFlowKbps(int flow_id) {
+  uint32_t available_capacity_per_flow_kbps = 0;
+  if (running_flows_.find(flow_id) != running_flows_.end()) {
+    available_capacity_per_flow_kbps =
+        TotalAvailableKbps() / static_cast<uint32_t>(running_flows_.size());
+  }
+  return available_capacity_per_flow_kbps;
+}
+
+MetricRecorder::MetricRecorder(const std::string algorithm_name,
+                               int flow_id,
+                               PacketSender* packet_sender,
+                               LinkShare* link_share)
+    : algorithm_name_(algorithm_name),
+      flow_id_(flow_id),
+      link_share_(link_share),
+      now_ms_(0),
+      sum_delays_ms_(0),
+      delay_histogram_ms_(),
+      sum_delays_square_ms2_(0),
+      sum_throughput_bytes_(0),
+      last_unweighted_estimate_error_(0),
+      optimal_throughput_bits_(0),
+      last_available_bitrate_per_flow_kbps_(0),
+      start_computing_metrics_ms_(0),
+      started_computing_metrics_(false),
+      num_packets_received_(0) {
+  std::fill_n(sum_lp_weighted_estimate_error_, 2, 0);
+  if (packet_sender != nullptr)
+    packet_sender->set_metric_recorder(this);
+}
+
+void MetricRecorder::SetPlotInformation(
+    const std::vector<std::string>& prefixes,
+    bool plot_delay,
+    bool plot_loss) {
+  assert(prefixes.size() == kNumMetrics);
+  for (size_t i = 0; i < kNumMetrics; ++i) {
+    plot_information_[i].prefix = prefixes[i];
+  }
+  plot_information_[kThroughput].plot_interval_ms = 100;
+  plot_information_[kSendingEstimate].plot_interval_ms = 100;
+  plot_information_[kDelay].plot_interval_ms = 100;
+  plot_information_[kLoss].plot_interval_ms = 500;
+  plot_information_[kObjective].plot_interval_ms = 1000;
+  plot_information_[kTotalAvailable].plot_interval_ms = 1000;
+  plot_information_[kAvailablePerFlow].plot_interval_ms = 1000;
+
+  for (int i = kThroughput; i < kNumMetrics; ++i) {
+    plot_information_[i].last_plot_ms = 0;
+    switch (i) {
+      case kSendingEstimate:
+      case kObjective:
+      case kAvailablePerFlow:
+        plot_information_[i].plot = false;
+        break;
+      case kLoss:
+        plot_information_[i].plot = plot_loss;
+        break;
+      case kDelay:
+        plot_information_[i].plot = plot_delay;
+        break;
+      default:
+        plot_information_[i].plot = true;
+    }
+  }
+}
+
+void MetricRecorder::PlotAllDynamics() {
+  for (int i = kThroughput; i < kNumMetrics; ++i) {
+    if (plot_information_[i].plot &&
+        now_ms_ - plot_information_[i].last_plot_ms >=
+            plot_information_[i].plot_interval_ms) {
+      PlotDynamics(i);
+    }
+  }
+}
+
+void MetricRecorder::PlotDynamics(int metric) {
+  if (metric == kTotalAvailable) {
+    BWE_TEST_LOGGING_PLOT_WITH_NAME(
+        0, plot_information_[kTotalAvailable].prefix, now_ms_,
+        GetTotalAvailableKbps(), "Available");
+  } else if (metric == kAvailablePerFlow) {
+    BWE_TEST_LOGGING_PLOT_WITH_NAME(
+        0, plot_information_[kAvailablePerFlow].prefix, now_ms_,
+        GetAvailablePerFlowKbps(), "Available_per_flow");
+  } else {
+    PlotLine(metric, plot_information_[metric].prefix,
+             plot_information_[metric].time_ms,
+             plot_information_[metric].value);
+  }
+  plot_information_[metric].last_plot_ms = now_ms_;
+}
+
+template <typename T>
+void MetricRecorder::PlotLine(int windows_id,
+                              const std::string& prefix,
+                              int64_t time_ms,
+                              T y) {
+  BWE_TEST_LOGGING_PLOT_WITH_NAME(windows_id, prefix, time_ms,
+                                  static_cast<double>(y), algorithm_name_);
+}
+
+void MetricRecorder::UpdateTimeMs(int64_t time_ms) {
+  now_ms_ = std::max(now_ms_, time_ms);
+}
+
+void MetricRecorder::UpdateThroughput(int64_t bitrate_kbps,
+                                      size_t payload_size) {
+  // Total throughput should be computed before updating the time.
+  PushThroughputBytes(payload_size, now_ms_);
+  plot_information_[kThroughput].Update(now_ms_, bitrate_kbps);
+}
+
+void MetricRecorder::UpdateSendingEstimateKbps(int64_t bitrate_kbps) {
+  plot_information_[kSendingEstimate].Update(now_ms_, bitrate_kbps);
+}
+
+void MetricRecorder::UpdateDelayMs(int64_t delay_ms) {
+  PushDelayMs(delay_ms, now_ms_);
+  plot_information_[kDelay].Update(now_ms_, delay_ms);
+}
+
+void MetricRecorder::UpdateLoss(float loss_ratio) {
+  plot_information_[kLoss].Update(now_ms_, loss_ratio);
+}
+
+void MetricRecorder::UpdateObjective() {
+  plot_information_[kObjective].Update(now_ms_, ObjectiveFunction());
+}
+
+uint32_t MetricRecorder::GetTotalAvailableKbps() {
+  if (link_share_ == nullptr)
+    return 0;
+  return link_share_->TotalAvailableKbps();
+}
+
+uint32_t MetricRecorder::GetAvailablePerFlowKbps() {
+  if (link_share_ == nullptr)
+    return 0;
+  return link_share_->AvailablePerFlowKbps(flow_id_);
+}
+
+uint32_t MetricRecorder::GetSendingEstimateKbps() {
+  return static_cast<uint32_t>(plot_information_[kSendingEstimate].value);
+}
+
+void MetricRecorder::PushDelayMs(int64_t delay_ms, int64_t arrival_time_ms) {
+  if (ShouldRecord(arrival_time_ms)) {
+    sum_delays_ms_ += delay_ms;
+    sum_delays_square_ms2_ += delay_ms * delay_ms;
+    if (delay_histogram_ms_.find(delay_ms) == delay_histogram_ms_.end()) {
+      delay_histogram_ms_[delay_ms] = 0;
+    }
+    ++delay_histogram_ms_[delay_ms];
+  }
+}
+
+void MetricRecorder::UpdateEstimateError(int64_t new_value) {
+  int64_t lp_value = pow(static_cast<double>(std::abs(new_value)), kP);
+  if (new_value < 0) {
+    sum_lp_weighted_estimate_error_[0] += lp_value;
+  } else {
+    sum_lp_weighted_estimate_error_[1] += lp_value;
+  }
+}
+
+void MetricRecorder::PushThroughputBytes(size_t payload_size,
+                                         int64_t arrival_time_ms) {
+  if (ShouldRecord(arrival_time_ms)) {
+    ++num_packets_received_;
+    sum_throughput_bytes_ += payload_size;
+
+    int64_t current_available_per_flow_kbps =
+        static_cast<int64_t>(GetAvailablePerFlowKbps());
+
+    int64_t current_bitrate_diff_kbps =
+        static_cast<int64_t>(GetSendingEstimateKbps()) -
+        current_available_per_flow_kbps;
+
+    int64_t weighted_estimate_error =
+        (((current_bitrate_diff_kbps + last_unweighted_estimate_error_) *
+          (arrival_time_ms - plot_information_[kThroughput].time_ms)) /
+         2);
+
+    UpdateEstimateError(weighted_estimate_error);
+
+    optimal_throughput_bits_ +=
+        ((current_available_per_flow_kbps +
+          last_available_bitrate_per_flow_kbps_) *
+         (arrival_time_ms - plot_information_[kThroughput].time_ms)) /
+        2;
+
+    last_available_bitrate_per_flow_kbps_ = current_available_per_flow_kbps;
+  }
+}
+
+bool MetricRecorder::ShouldRecord(int64_t arrival_time_ms) {
+  if (arrival_time_ms >= start_computing_metrics_ms_) {
+    if (!started_computing_metrics_) {
+      start_computing_metrics_ms_ = arrival_time_ms;
+      now_ms_ = arrival_time_ms;
+      started_computing_metrics_ = true;
+    }
+    return true;
+  } else {
+    return false;
+  }
+}
+
+void MetricRecorder::PlotThroughputHistogram(
+    const std::string& title,
+    const std::string& bwe_name,
+    size_t num_flows,
+    int64_t extra_offset_ms,
+    const std::string optimum_id) const {
+  double optimal_bitrate_per_flow_kbps = static_cast<double>(
+      optimal_throughput_bits_ / RunDurationMs(extra_offset_ms));
+
+  double neg_error = Renormalize(
+      NormLp(sum_lp_weighted_estimate_error_[0], num_packets_received_, kP));
+  double pos_error = Renormalize(
+      NormLp(sum_lp_weighted_estimate_error_[1], num_packets_received_, kP));
+
+  double average_bitrate_kbps = AverageBitrateKbps(extra_offset_ms);
+
+  // Prevent the error to be too close to zero (plotting issue).
+  double extra_error = average_bitrate_kbps / 500;
+
+  std::string optimum_title =
+      optimum_id.empty() ? "optimal_bitrate" : "optimal_bitrates#" + optimum_id;
+
+  BWE_TEST_LOGGING_LABEL(4, title, "average_bitrate_(kbps)", num_flows);
+  BWE_TEST_LOGGING_LIMITERRORBAR(
+      4, bwe_name, average_bitrate_kbps,
+      average_bitrate_kbps - neg_error - extra_error,
+      average_bitrate_kbps + pos_error + extra_error, "estimate_error",
+      optimal_bitrate_per_flow_kbps, optimum_title, flow_id_);
+
+  BWE_TEST_LOGGING_LOG1("RESULTS >>> " + bwe_name + " Channel utilization : ",
+                        "%lf %%",
+                        100.0 * static_cast<double>(average_bitrate_kbps) /
+                            optimal_bitrate_per_flow_kbps);
+
+  RTC_UNUSED(pos_error);
+  RTC_UNUSED(neg_error);
+  RTC_UNUSED(extra_error);
+  RTC_UNUSED(optimal_bitrate_per_flow_kbps);
+}
+
+void MetricRecorder::PlotThroughputHistogram(const std::string& title,
+                                             const std::string& bwe_name,
+                                             size_t num_flows,
+                                             int64_t extra_offset_ms) const {
+  PlotThroughputHistogram(title, bwe_name, num_flows, extra_offset_ms, "");
+}
+
+void MetricRecorder::PlotDelayHistogram(const std::string& title,
+                                        const std::string& bwe_name,
+                                        size_t num_flows,
+                                        int64_t one_way_path_delay_ms) const {
+  double average_delay_ms =
+      static_cast<double>(sum_delays_ms_) / num_packets_received_;
+
+  // Prevent the error to be too close to zero (plotting issue).
+  double extra_error = average_delay_ms / 500;
+  double tenth_sigma_ms = DelayStdDev() / 10.0 + extra_error;
+  int64_t percentile_5_ms = NthDelayPercentile(5);
+  int64_t percentile_95_ms = NthDelayPercentile(95);
+
+  BWE_TEST_LOGGING_LABEL(5, title, "average_delay_(ms)", num_flows)
+  BWE_TEST_LOGGING_ERRORBAR(5, bwe_name, average_delay_ms, percentile_5_ms,
+                            percentile_95_ms, "5th and 95th percentiles",
+                            flow_id_);
+
+  // Log added latency, disregard baseline path delay.
+  BWE_TEST_LOGGING_LOG1("RESULTS >>> " + bwe_name + " Delay average : ",
+                        "%lf ms", average_delay_ms - one_way_path_delay_ms);
+  BWE_TEST_LOGGING_LOG1("RESULTS >>> " + bwe_name + " Delay 5th percentile : ",
+                        "%ld ms", percentile_5_ms - one_way_path_delay_ms);
+  BWE_TEST_LOGGING_LOG1("RESULTS >>> " + bwe_name + " Delay 95th percentile : ",
+                        "%ld ms", percentile_95_ms - one_way_path_delay_ms);
+
+  RTC_UNUSED(tenth_sigma_ms);
+  RTC_UNUSED(percentile_5_ms);
+  RTC_UNUSED(percentile_95_ms);
+}
+
+void MetricRecorder::PlotLossHistogram(const std::string& title,
+                                       const std::string& bwe_name,
+                                       size_t num_flows,
+                                       float global_loss_ratio) const {
+  BWE_TEST_LOGGING_LABEL(6, title, "packet_loss_ratio_(%)", num_flows)
+  BWE_TEST_LOGGING_BAR(6, bwe_name, 100.0f * global_loss_ratio, flow_id_);
+
+  BWE_TEST_LOGGING_LOG1("RESULTS >>> " + bwe_name + " Loss Ratio : ", "%f %%",
+                        100.0f * global_loss_ratio);
+}
+
+void MetricRecorder::PlotObjectiveHistogram(const std::string& title,
+                                            const std::string& bwe_name,
+                                            size_t num_flows) const {
+  BWE_TEST_LOGGING_LABEL(7, title, "objective_function", num_flows)
+  BWE_TEST_LOGGING_BAR(7, bwe_name, ObjectiveFunction(), flow_id_);
+}
+
+void MetricRecorder::PlotZero() {
+  for (int i = kThroughput; i <= kLoss; ++i) {
+    if (plot_information_[i].plot) {
+      std::stringstream prefix;
+      prefix << "Receiver_" << flow_id_ << "_" + plot_information_[i].prefix;
+      PlotLine(i, prefix.str(), now_ms_, 0);
+      plot_information_[i].last_plot_ms = now_ms_;
+    }
+  }
+}
+
+void MetricRecorder::PauseFlow() {
+  PlotZero();
+  link_share_->PauseFlow(flow_id_);
+}
+
+void MetricRecorder::ResumeFlow(int64_t paused_time_ms) {
+  UpdateTimeMs(now_ms_ + paused_time_ms);
+  PlotZero();
+  link_share_->ResumeFlow(flow_id_);
+}
+
+double MetricRecorder::AverageBitrateKbps(int64_t extra_offset_ms) const {
+  int64_t duration_ms = RunDurationMs(extra_offset_ms);
+  if (duration_ms == 0)
+    return 0.0;
+  return static_cast<double>(8 * sum_throughput_bytes_ / duration_ms);
+}
+
+int64_t MetricRecorder::RunDurationMs(int64_t extra_offset_ms) const {
+  return now_ms_ - start_computing_metrics_ms_ - extra_offset_ms;
+}
+
+double MetricRecorder::DelayStdDev() const {
+  if (num_packets_received_ == 0) {
+    return 0.0;
+  }
+  double mean = static_cast<double>(sum_delays_ms_) / num_packets_received_;
+  double mean2 =
+      static_cast<double>(sum_delays_square_ms2_) / num_packets_received_;
+  return sqrt(mean2 - pow(mean, 2.0));
+}
+
+// Since delay values are bounded in a subset of [0, 5000] ms,
+// this function's execution time is O(1), independend of num_packets_received_.
+int64_t MetricRecorder::NthDelayPercentile(int n) const {
+  if (num_packets_received_ == 0) {
+    return 0;
+  }
+  size_t num_packets_remaining = (n * num_packets_received_) / 100;
+  for (auto hist : delay_histogram_ms_) {
+    if (num_packets_remaining <= hist.second)
+      return static_cast<int64_t>(hist.first);
+    num_packets_remaining -= hist.second;
+  }
+
+  assert(false);
+  return -1;
+}
+
+// The weighted_estimate_error_ was weighted based on time windows.
+// This function scales back the result before plotting.
+double MetricRecorder::Renormalize(double x) const {
+  return (x * num_packets_received_) / now_ms_;
+}
+
+inline double U(int64_t x, double alpha) {
+  if (alpha == 1.0) {
+    return log(static_cast<double>(x));
+  }
+  return pow(static_cast<double>(x), 1.0 - alpha) / (1.0 - alpha);
+}
+
+inline double U(size_t x, double alpha) {
+  return U(static_cast<int64_t>(x), alpha);
+}
+
+// TODO(magalhaesc): Update ObjectiveFunction.
+double MetricRecorder::ObjectiveFunction() const {
+  const double kDelta = 0.15;  // Delay penalty factor.
+  const double kAlpha = 1.0;
+  const double kBeta = 1.0;
+
+  double throughput_metric = U(sum_throughput_bytes_, kAlpha);
+  double delay_penalty = kDelta * U(sum_delays_ms_, kBeta);
+
+  return throughput_metric - delay_penalty;
+}
+
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h
new file mode 100644
index 0000000000..2be13e0b0b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_METRIC_RECORDER_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_METRIC_RECORDER_H_
+
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/common.h"
+#include "webrtc/test/testsupport/gtest_prod_util.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+class ChokeFilter;
+class PacketSender;
+
+class LinkShare {
+ public:
+  explicit LinkShare(ChokeFilter* choke_filter);
+
+  void PauseFlow(int flow_id);   // Increases available capacity per flow.
+  void ResumeFlow(int flow_id);  // Decreases available capacity per flow.
+
+  uint32_t TotalAvailableKbps();
+  // If the given flow is paused, its output is zero.
+  uint32_t AvailablePerFlowKbps(int flow_id);
+
+ private:
+  ChokeFilter* choke_filter_;
+  std::set<int> running_flows_;
+};
+
+struct PlotInformation {
+  PlotInformation()
+      : prefix(),
+        last_plot_ms(0),
+        time_ms(0),
+        value(0.0),
+        plot_interval_ms(0) {}
+  template <typename T>
+  void Update(int64_t now_ms, T new_value) {
+    time_ms = now_ms;
+    value = static_cast<double>(new_value);
+  }
+  std::string prefix;
+  bool plot;
+  int64_t last_plot_ms;
+  int64_t time_ms;
+  double value;
+  int64_t plot_interval_ms;
+};
+
+class MetricRecorder {
+ public:
+  MetricRecorder(const std::string algorithm_name,
+                 int flow_id,
+                 PacketSender* packet_sender,
+                 LinkShare* link_share);
+
+  void SetPlotInformation(const std::vector<std::string>& prefixes,
+                          bool plot_delay,
+                          bool plot_loss);
+
+  template <typename T>
+  void PlotLine(int windows_id,
+                const std::string& prefix,
+                int64_t time_ms,
+                T y);
+
+  void PlotDynamics(int metric);
+  void PlotAllDynamics();
+
+  void UpdateTimeMs(int64_t time_ms);
+  void UpdateThroughput(int64_t bitrate_kbps, size_t payload_size);
+  void UpdateSendingEstimateKbps(int64_t bitrate_kbps);
+  void UpdateDelayMs(int64_t delay_ms);
+  void UpdateLoss(float loss_ratio);
+  void UpdateObjective();
+
+  void PlotThroughputHistogram(const std::string& title,
+                               const std::string& bwe_name,
+                               size_t num_flows,
+                               int64_t extra_offset_ms,
+                               const std::string optimum_id) const;
+
+  void PlotThroughputHistogram(const std::string& title,
+                               const std::string& bwe_name,
+                               size_t num_flows,
+                               int64_t extra_offset_ms) const;
+
+  void PlotDelayHistogram(const std::string& title,
+                          const std::string& bwe_name,
+                          size_t num_flows,
+                          int64_t one_way_path_delay_ms) const;
+
+  void PlotLossHistogram(const std::string& title,
+                         const std::string& bwe_name,
+                         size_t num_flows,
+                         float global_loss_ratio) const;
+
+  void PlotObjectiveHistogram(const std::string& title,
+                              const std::string& bwe_name,
+                              size_t num_flows) const;
+
+  void set_start_computing_metrics_ms(int64_t start_computing_metrics_ms) {
+    start_computing_metrics_ms_ = start_computing_metrics_ms;
+  }
+
+  void set_plot_available_capacity(bool plot) {
+    plot_information_[kTotalAvailable].plot = plot;
+  }
+
+  void PauseFlow();                         // Plot zero.
+  void ResumeFlow(int64_t paused_time_ms);  // Plot zero.
+  void PlotZero();
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(MetricRecorderTest, NoPackets);
+  FRIEND_TEST_ALL_PREFIXES(MetricRecorderTest, RegularPackets);
+  FRIEND_TEST_ALL_PREFIXES(MetricRecorderTest, VariableDelayPackets);
+
+  uint32_t GetTotalAvailableKbps();
+  uint32_t GetAvailablePerFlowKbps();
+  uint32_t GetSendingEstimateKbps();
+  double ObjectiveFunction() const;
+
+  double Renormalize(double x) const;
+  bool ShouldRecord(int64_t arrival_time_ms);
+
+  void PushDelayMs(int64_t delay_ms, int64_t arrival_time_ms);
+  void PushThroughputBytes(size_t throughput_bytes, int64_t arrival_time_ms);
+
+  void UpdateEstimateError(int64_t new_value);
+  double DelayStdDev() const;
+  int64_t NthDelayPercentile(int n) const;
+  double AverageBitrateKbps(int64_t extra_offset_ms) const;
+  int64_t RunDurationMs(int64_t extra_offset_ms) const;
+
+  enum Metrics {
+    kThroughput = 0,
+    kSendingEstimate,
+    kDelay,
+    kLoss,
+    kObjective,
+    kTotalAvailable,
+    kAvailablePerFlow,
+    kNumMetrics
+  };
+
+  std::string algorithm_name_;
+  int flow_id_;
+  LinkShare* link_share_;
+
+  int64_t now_ms_;
+
+  PlotInformation plot_information_[kNumMetrics];
+
+  int64_t sum_delays_ms_;
+  // delay_histogram_ms_[i] counts how many packets have delay = i ms.
+  std::map<int64_t, size_t> delay_histogram_ms_;
+  int64_t sum_delays_square_ms2_;  // Used to compute standard deviation.
+  size_t sum_throughput_bytes_;
+  // ((Receiving rate - available bitrate per flow) * time window)^p.
+  // 0 for negative values, 1 for positive values.
+  int64_t sum_lp_weighted_estimate_error_[2];
+  int64_t last_unweighted_estimate_error_;
+  int64_t optimal_throughput_bits_;
+  int64_t last_available_bitrate_per_flow_kbps_;
+  int64_t start_computing_metrics_ms_;
+  bool started_computing_metrics_;
+  size_t num_packets_received_;
+};
+
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_METRIC_RECORDER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder_unittest.cc
new file mode 100644
index 0000000000..7d4ed5fd5f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/metric_recorder_unittest.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
+
+#include <math.h>
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+class MetricRecorderTest : public ::testing::Test {
+ public:
+  MetricRecorderTest() : metric_recorder_("Test", 0, nullptr, nullptr) {}
+
+  ~MetricRecorderTest() {}
+
+ protected:
+  MetricRecorder metric_recorder_;
+};
+
+TEST_F(MetricRecorderTest, NoPackets) {
+  EXPECT_EQ(metric_recorder_.AverageBitrateKbps(0), 0);
+  EXPECT_EQ(metric_recorder_.DelayStdDev(), 0.0);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(0), 0);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(5), 0);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(95), 0);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(100), 0);
+}
+
+TEST_F(MetricRecorderTest, RegularPackets) {
+  const size_t kPayloadSizeBytes = 1200;
+  const int64_t kDelayMs = 20;
+  const int64_t kInterpacketGapMs = 5;
+  const int kNumPackets = 1000;
+
+  for (int i = 0; i < kNumPackets; ++i) {
+    int64_t arrival_time_ms = kInterpacketGapMs * i + kDelayMs;
+    metric_recorder_.UpdateTimeMs(arrival_time_ms);
+    metric_recorder_.PushDelayMs(kDelayMs, arrival_time_ms);
+    metric_recorder_.PushThroughputBytes(kPayloadSizeBytes, arrival_time_ms);
+  }
+
+  EXPECT_NEAR(
+      metric_recorder_.AverageBitrateKbps(0),
+      static_cast<uint32_t>(kPayloadSizeBytes * 8) / (kInterpacketGapMs), 10);
+
+  EXPECT_EQ(metric_recorder_.DelayStdDev(), 0.0);
+
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(0), kDelayMs);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(5), kDelayMs);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(95), kDelayMs);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(100), kDelayMs);
+}
+
+TEST_F(MetricRecorderTest, VariableDelayPackets) {
+  const size_t kPayloadSizeBytes = 1200;
+  const int64_t kInterpacketGapMs = 2000;
+  const int kNumPackets = 1000;
+
+  std::vector<int64_t> delays_ms;
+  for (int i = 0; i < kNumPackets; ++i) {
+    delays_ms.push_back(static_cast<int64_t>(i + 1));
+  }
+  // Order of packets should not matter here.
+  std::random_shuffle(delays_ms.begin(), delays_ms.end());
+
+  int first_received_ms = delays_ms[0];
+  int64_t last_received_ms = 0;
+  for (int i = 0; i < kNumPackets; ++i) {
+    int64_t arrival_time_ms = kInterpacketGapMs * i + delays_ms[i];
+    last_received_ms = std::max(last_received_ms, arrival_time_ms);
+    metric_recorder_.UpdateTimeMs(arrival_time_ms);
+    metric_recorder_.PushDelayMs(delays_ms[i], arrival_time_ms);
+    metric_recorder_.PushThroughputBytes(kPayloadSizeBytes, arrival_time_ms);
+  }
+
+  size_t received_bits = kPayloadSizeBytes * 8 * kNumPackets;
+  EXPECT_NEAR(metric_recorder_.AverageBitrateKbps(0),
+              static_cast<uint32_t>(received_bits) /
+                  ((last_received_ms - first_received_ms)),
+              10);
+
+  double expected_x = (kNumPackets + 1) / 2.0;
+  double expected_x2 = ((kNumPackets + 1) * (2 * kNumPackets + 1)) / 6.0;
+  double var = expected_x2 - pow(expected_x, 2.0);
+  EXPECT_NEAR(metric_recorder_.DelayStdDev(), sqrt(var), kNumPackets / 1000.0);
+
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(0), 1);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(5), (5 * kNumPackets) / 100);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(95), (95 * kNumPackets) / 100);
+  EXPECT_EQ(metric_recorder_.NthDelayPercentile(100), kNumPackets);
+}
+
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet.h
index 12d4a3ee93..4a361c4dc2 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet.h
@@ -16,7 +16,7 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 
 namespace webrtc {
@@ -34,27 +34,46 @@ class Packet {
   virtual bool operator<(const Packet& rhs) const;
 
   virtual int flow_id() const { return flow_id_; }
-  virtual int64_t creation_time_us() const { return creation_time_us_; }
   virtual void set_send_time_us(int64_t send_time_us);
   virtual int64_t send_time_us() const { return send_time_us_; }
+  virtual int64_t sender_timestamp_us() const { return sender_timestamp_us_; }
   virtual size_t payload_size() const { return payload_size_; }
   virtual Packet::Type GetPacketType() const = 0;
+  virtual void set_sender_timestamp_us(int64_t sender_timestamp_us) {
+    sender_timestamp_us_ = sender_timestamp_us;
+  }
+  virtual void set_paced(bool paced) { paced_ = paced; }
+  virtual bool paced() const { return paced_; }
+  virtual int64_t creation_time_ms() const {
+    return (creation_time_us_ + 500) / 1000;
+  }
+  virtual int64_t sender_timestamp_ms() const {
+    return (sender_timestamp_us_ + 500) / 1000;
+  }
+  virtual int64_t send_time_ms() const { return (send_time_us_ + 500) / 1000; }
 
- private:
+ protected:
   int flow_id_;
   int64_t creation_time_us_;  // Time when the packet was created.
   int64_t send_time_us_;  // Time the packet left last processor touching it.
-  size_t payload_size_;   // Size of the (non-existent, simulated) payload.
+  int64_t sender_timestamp_us_;  // Time the packet left the Sender.
+  size_t payload_size_;  // Size of the (non-existent, simulated) payload.
+  bool paced_;           // True if sent through paced sender.
 };
 
 class MediaPacket : public Packet {
  public:
   MediaPacket();
+  MediaPacket(int flow_id,
+              int64_t send_time_us,
+              size_t payload_size,
+              uint16_t sequence_number);
   MediaPacket(int flow_id,
               int64_t send_time_us,
               size_t payload_size,
               const RTPHeader& header);
-  MediaPacket(int64_t send_time_us, uint32_t sequence_number);
+  MediaPacket(int64_t send_time_us, uint16_t sequence_number);
+
   virtual ~MediaPacket() {}
 
   int64_t GetAbsSendTimeInMs() const {
@@ -65,6 +84,7 @@ class MediaPacket : public Packet {
   void SetAbsSendTimeMs(int64_t abs_send_time_ms);
   const RTPHeader& header() const { return header_; }
   virtual Packet::Type GetPacketType() const { return kMedia; }
+  uint16_t sequence_number() const { return header_.sequenceNumber; }
 
  private:
   static const int kAbsSendTimeFraction = 18;
@@ -77,17 +97,25 @@ class MediaPacket : public Packet {
 
 class FeedbackPacket : public Packet {
  public:
-  FeedbackPacket(int flow_id, int64_t send_time_us)
-      : Packet(flow_id, send_time_us, 0) {}
+  FeedbackPacket(int flow_id,
+                 int64_t this_send_time_us,
+                 int64_t latest_send_time_ms)
+      : Packet(flow_id, this_send_time_us, 0),
+        latest_send_time_ms_(latest_send_time_ms) {}
   virtual ~FeedbackPacket() {}
 
   virtual Packet::Type GetPacketType() const { return kFeedback; }
+  int64_t latest_send_time_ms() const { return latest_send_time_ms_; }
+
+ private:
+  int64_t latest_send_time_ms_;  // Time stamp for the latest sent FbPacket.
 };
 
 class RembFeedback : public FeedbackPacket {
  public:
   RembFeedback(int flow_id,
                int64_t send_time_us,
+               int64_t latest_send_time_ms,
                uint32_t estimated_bps,
                RTCPReportBlock report_block);
   virtual ~RembFeedback() {}
@@ -105,6 +133,7 @@ class SendSideBweFeedback : public FeedbackPacket {
   typedef std::map<uint16_t, int64_t> ArrivalTimesMap;
   SendSideBweFeedback(int flow_id,
                       int64_t send_time_us,
+                      int64_t latest_send_time_ms,
                       const std::vector<PacketInfo>& packet_feedback_vector);
   virtual ~SendSideBweFeedback() {}
 
@@ -119,25 +148,57 @@ class SendSideBweFeedback : public FeedbackPacket {
 class NadaFeedback : public FeedbackPacket {
  public:
   NadaFeedback(int flow_id,
-               int64_t send_time_us,
+               int64_t this_send_time_us,
+               int64_t exp_smoothed_delay_ms,
+               int64_t est_queuing_delay_signal_ms,
                int64_t congestion_signal,
-               float derivative)
-      : FeedbackPacket(flow_id, send_time_us),
+               float derivative,
+               float receiving_rate,
+               int64_t latest_send_time_ms)
+      : FeedbackPacket(flow_id, this_send_time_us, latest_send_time_ms),
+        exp_smoothed_delay_ms_(exp_smoothed_delay_ms),
+        est_queuing_delay_signal_ms_(est_queuing_delay_signal_ms),
         congestion_signal_(congestion_signal),
-        derivative_(derivative) {}
+        derivative_(derivative),
+        receiving_rate_(receiving_rate) {}
   virtual ~NadaFeedback() {}
 
+  int64_t exp_smoothed_delay_ms() const { return exp_smoothed_delay_ms_; }
+  int64_t est_queuing_delay_signal_ms() const {
+    return est_queuing_delay_signal_ms_;
+  }
   int64_t congestion_signal() const { return congestion_signal_; }
   float derivative() const { return derivative_; }
+  float receiving_rate() const { return receiving_rate_; }
 
  private:
-  int64_t congestion_signal_;
-  float derivative_;
+  int64_t exp_smoothed_delay_ms_;        // Referred as d_hat_n.
+  int64_t est_queuing_delay_signal_ms_;  // Referred as d_tilde_n.
+  int64_t congestion_signal_;            // Referred as x_n.
+  float derivative_;                     // Referred as x'_n.
+  float receiving_rate_;                 // Referred as R_r.
+};
+
+class TcpFeedback : public FeedbackPacket {
+ public:
+  TcpFeedback(int flow_id,
+              int64_t send_time_us,
+              int64_t latest_send_time_ms,
+              const std::vector<uint16_t>& acked_packets)
+      : FeedbackPacket(flow_id, send_time_us, latest_send_time_ms),
+        acked_packets_(acked_packets) {}
+  virtual ~TcpFeedback() {}
+
+  const std::vector<uint16_t>& acked_packets() const { return acked_packets_; }
+
+ private:
+  const std::vector<uint16_t> acked_packets_;
 };
 
 typedef std::list<Packet*> Packets;
 typedef std::list<Packet*>::iterator PacketsIt;
 typedef std::list<Packet*>::const_iterator PacketsConstIt;
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
index 0f85d7f22c..793e06421f 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
@@ -14,11 +14,11 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace testing {
@@ -28,16 +28,48 @@ PacketReceiver::PacketReceiver(PacketProcessorListener* listener,
                                int flow_id,
                                BandwidthEstimatorType bwe_type,
                                bool plot_delay,
-                               bool plot_bwe)
+                               bool plot_bwe,
+                               MetricRecorder* metric_recorder)
     : PacketProcessor(listener, flow_id, kReceiver),
-      delay_log_prefix_(),
-      last_delay_plot_ms_(0),
+      bwe_receiver_(CreateBweReceiver(bwe_type, flow_id, plot_bwe)),
+      metric_recorder_(metric_recorder),
       plot_delay_(plot_delay),
-      bwe_receiver_(CreateBweReceiver(bwe_type, flow_id, plot_bwe)) {
-  // Setup the prefix ststd::rings used when logging.
-  std::stringstream ss;
-  ss << "Delay_" << flow_id << "#2";
-  delay_log_prefix_ = ss.str();
+      last_delay_plot_ms_(0),
+      // #2 aligns the plot with the right axis.
+      delay_prefix_("Delay_ms#2"),
+      bwe_type_(bwe_type) {
+  if (metric_recorder_ != nullptr) {
+    // Setup the prefix std::strings used when logging.
+    std::vector<std::string> prefixes;
+
+    // Metric recorder plots them in separated figures,
+    // alignment will take place with the #1 left axis.
+    prefixes.push_back("Throughput_kbps#1");
+    prefixes.push_back("Sending_Estimate_kbps#1");
+    prefixes.push_back("Delay_ms_#1");
+    prefixes.push_back("Packet_Loss_#1");
+    prefixes.push_back("Objective_function_#1");
+
+    // Plot Total/PerFlow Available capacity together with throughputs.
+    prefixes.push_back("Throughput_kbps#1");  // Total Available.
+    prefixes.push_back("Throughput_kbps#1");  // Available per flow.
+
+    bool plot_loss = plot_delay;  // Plot loss if delay is plotted.
+    metric_recorder_->SetPlotInformation(prefixes, plot_delay, plot_loss);
+  }
+}
+
+PacketReceiver::PacketReceiver(PacketProcessorListener* listener,
+                               int flow_id,
+                               BandwidthEstimatorType bwe_type,
+                               bool plot_delay,
+                               bool plot_bwe)
+    : PacketReceiver(listener,
+                     flow_id,
+                     bwe_type,
+                     plot_delay,
+                     plot_bwe,
+                     nullptr) {
 }
 
 PacketReceiver::~PacketReceiver() {
@@ -50,16 +82,25 @@ void PacketReceiver::RunFor(int64_t time_ms, Packets* in_out) {
     // should only process a single flow id.
     // TODO(holmer): Break this out into a Demuxer which implements both
     // PacketProcessorListener and PacketProcessor.
+    BWE_TEST_LOGGING_CONTEXT("Receiver");
     if ((*it)->GetPacketType() == Packet::kMedia &&
         (*it)->flow_id() == *flow_ids().begin()) {
-      BWE_TEST_LOGGING_CONTEXT("Receiver");
+      BWE_TEST_LOGGING_CONTEXT(*flow_ids().begin());
       const MediaPacket* media_packet = static_cast<const MediaPacket*>(*it);
       // We're treating the send time (from previous filter) as the arrival
       // time once packet reaches the estimator.
-      int64_t arrival_time_ms = (media_packet->send_time_us() + 500) / 1000;
-      BWE_TEST_LOGGING_TIME(arrival_time_ms);
-      PlotDelay(arrival_time_ms,
-                (media_packet->creation_time_us() + 500) / 1000);
+      int64_t arrival_time_ms = media_packet->send_time_ms();
+      int64_t send_time_ms = media_packet->creation_time_ms();
+      delay_stats_.Push(arrival_time_ms - send_time_ms);
+
+      if (metric_recorder_ != nullptr) {
+        metric_recorder_->UpdateTimeMs(arrival_time_ms);
+        UpdateMetrics(arrival_time_ms, send_time_ms,
+                      media_packet->payload_size());
+        metric_recorder_->PlotAllDynamics();
+      } else if (plot_delay_) {
+        PlotDelay(arrival_time_ms, send_time_ms);
+      }
 
       bwe_receiver_->ReceivePacket(arrival_time_ms, *media_packet);
       FeedbackPacket* fb = bwe_receiver_->GetFeedback(arrival_time_ms);
@@ -75,16 +116,32 @@ void PacketReceiver::RunFor(int64_t time_ms, Packets* in_out) {
   in_out->merge(feedback, DereferencingComparator<Packet>);
 }
 
+void PacketReceiver::UpdateMetrics(int64_t arrival_time_ms,
+                                   int64_t send_time_ms,
+                                   size_t payload_size) {
+  metric_recorder_->UpdateThroughput(bwe_receiver_->RecentKbps(), payload_size);
+  metric_recorder_->UpdateDelayMs(arrival_time_ms - send_time_ms);
+  metric_recorder_->UpdateLoss(bwe_receiver_->RecentPacketLossRatio());
+  metric_recorder_->UpdateObjective();
+}
+
 void PacketReceiver::PlotDelay(int64_t arrival_time_ms, int64_t send_time_ms) {
-  static const int kDelayPlotIntervalMs = 100;
-  if (!plot_delay_)
-    return;
-  if (arrival_time_ms - last_delay_plot_ms_ > kDelayPlotIntervalMs) {
-    BWE_TEST_LOGGING_PLOT(delay_log_prefix_, arrival_time_ms,
-                          arrival_time_ms - send_time_ms);
+  const int64_t kDelayPlotIntervalMs = 100;
+  if (arrival_time_ms >= last_delay_plot_ms_ + kDelayPlotIntervalMs) {
+    BWE_TEST_LOGGING_PLOT_WITH_NAME(0, delay_prefix_, arrival_time_ms,
+                                    arrival_time_ms - send_time_ms,
+                                    bwe_names[bwe_type_]);
     last_delay_plot_ms_ = arrival_time_ms;
   }
 }
+
+float PacketReceiver::GlobalPacketLoss() {
+  return bwe_receiver_->GlobalReceiverPacketLossRatio();
+}
+
+Stats<double> PacketReceiver::GetDelayStats() const {
+  return delay_stats_;
+}
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h
index 6d94ea7a05..fb9e9fd7ab 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h
@@ -17,6 +17,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 
 namespace webrtc {
 namespace testing {
@@ -29,6 +30,12 @@ class PacketReceiver : public PacketProcessor {
                  BandwidthEstimatorType bwe_type,
                  bool plot_delay,
                  bool plot_bwe);
+  PacketReceiver(PacketProcessorListener* listener,
+                 int flow_id,
+                 BandwidthEstimatorType bwe_type,
+                 bool plot_delay,
+                 bool plot_bwe,
+                 MetricRecorder* metric_recorder);
   ~PacketReceiver();
 
   // Implements PacketProcessor.
@@ -36,17 +43,27 @@ class PacketReceiver : public PacketProcessor {
 
   void LogStats();
 
- protected:
-  void PlotDelay(int64_t arrival_time_ms, int64_t send_time_ms);
+  Stats<double> GetDelayStats() const;
 
-  int64_t now_ms_;
-  std::string delay_log_prefix_;
-  int64_t last_delay_plot_ms_;
-  bool plot_delay_;
+  float GlobalPacketLoss();
+
+ protected:
+  void UpdateMetrics(int64_t arrival_time_ms,
+                     int64_t send_time_ms,
+                     size_t payload_size);
+
+  Stats<double> delay_stats_;
   rtc::scoped_ptr<BweReceiver> bwe_receiver_;
 
  private:
-  DISALLOW_IMPLICIT_CONSTRUCTORS(PacketReceiver);
+  void PlotDelay(int64_t arrival_time_ms, int64_t send_time_ms);
+  MetricRecorder* metric_recorder_;
+  bool plot_delay_;  // Used in case there isn't a metric recorder.
+  int64_t last_delay_plot_ms_;
+  std::string delay_prefix_;
+  BandwidthEstimatorType bwe_type_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacketReceiver);
 };
 }  // namespace bwe
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
index 2a34d22c63..3bcbc0a071 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
@@ -14,81 +14,51 @@
 #include <list>
 #include <sstream>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
-PacketSender::PacketSender(PacketProcessorListener* listener,
-                           VideoSource* source,
-                           BandwidthEstimatorType estimator_type)
-    : PacketProcessor(listener, source->flow_id(), kSender),
-      // For Packet::send_time_us() to be comparable with timestamps from
-      // clock_, the clock of the PacketSender and the Source must be aligned.
-      // We assume that both start at time 0.
-      clock_(0),
-      source_(source),
-      bwe_(CreateBweSender(estimator_type,
-                           source_->bits_per_second() / 1000,
-                           this,
-                           &clock_)) {
-  modules_.push_back(bwe_.get());
+void PacketSender::Pause() {
+  running_ = false;
+  if (metric_recorder_ != nullptr) {
+    metric_recorder_->PauseFlow();
+  }
 }
 
-PacketSender::~PacketSender() {
+void PacketSender::Resume(int64_t paused_time_ms) {
+  running_ = true;
+  if (metric_recorder_ != nullptr) {
+    metric_recorder_->ResumeFlow(paused_time_ms);
+  }
 }
 
-void PacketSender::RunFor(int64_t time_ms, Packets* in_out) {
-  int64_t now_ms = clock_.TimeInMilliseconds();
-  std::list<FeedbackPacket*> feedbacks =
-      GetFeedbackPackets(in_out, now_ms + time_ms);
-  ProcessFeedbackAndGeneratePackets(time_ms, &feedbacks, in_out);
+void PacketSender::set_metric_recorder(MetricRecorder* metric_recorder) {
+  metric_recorder_ = metric_recorder;
 }
 
-void PacketSender::ProcessFeedbackAndGeneratePackets(
-    int64_t time_ms,
-    std::list<FeedbackPacket*>* feedbacks,
-    Packets* packets) {
-  do {
-    // Make sure to at least run Process() below every 100 ms.
-    int64_t time_to_run_ms = std::min<int64_t>(time_ms, 100);
-    if (!feedbacks->empty()) {
-      int64_t time_until_feedback_ms =
-          feedbacks->front()->send_time_us() / 1000 -
-          clock_.TimeInMilliseconds();
-      time_to_run_ms =
-          std::max<int64_t>(std::min(time_ms, time_until_feedback_ms), 0);
-    }
-    Packets generated;
-    source_->RunFor(time_to_run_ms, &generated);
-    bwe_->OnPacketsSent(generated);
-    packets->merge(generated, DereferencingComparator<Packet>);
-    clock_.AdvanceTimeMilliseconds(time_to_run_ms);
-    if (!feedbacks->empty()) {
-      bwe_->GiveFeedback(*feedbacks->front());
-      delete feedbacks->front();
-      feedbacks->pop_front();
-    }
-    bwe_->Process();
-    time_ms -= time_to_run_ms;
-  } while (time_ms > 0);
-  assert(feedbacks->empty());
+void PacketSender::RecordBitrate() {
+  if (metric_recorder_ != nullptr) {
+    BWE_TEST_LOGGING_CONTEXT("Sender");
+    BWE_TEST_LOGGING_CONTEXT(*flow_ids().begin());
+    metric_recorder_->UpdateTimeMs(clock_.TimeInMilliseconds());
+    metric_recorder_->UpdateSendingEstimateKbps(TargetBitrateKbps());
+  }
 }
 
-int PacketSender::GetFeedbackIntervalMs() const {
-  return bwe_->GetFeedbackIntervalMs();
-}
-
-std::list<FeedbackPacket*> PacketSender::GetFeedbackPackets(
-    Packets* in_out,
-    int64_t end_time_ms) {
+std::list<FeedbackPacket*> GetFeedbackPackets(Packets* in_out,
+                                              int64_t end_time_ms,
+                                              int flow_id) {
   std::list<FeedbackPacket*> fb_packets;
   for (auto it = in_out->begin(); it != in_out->end();) {
     if ((*it)->send_time_us() > 1000 * end_time_ms)
       break;
     if ((*it)->GetPacketType() == Packet::kFeedback &&
-        source()->flow_id() == (*it)->flow_id()) {
+        flow_id == (*it)->flow_id()) {
       fb_packets.push_back(static_cast<FeedbackPacket*>(*it));
       it = in_out->erase(it);
     } else {
@@ -98,20 +68,95 @@ std::list<FeedbackPacket*> PacketSender::GetFeedbackPackets(
   return fb_packets;
 }
 
-void PacketSender::OnNetworkChanged(uint32_t target_bitrate_bps,
-                                    uint8_t fraction_lost,
-                                    int64_t rtt) {
+VideoSender::VideoSender(PacketProcessorListener* listener,
+                         VideoSource* source,
+                         BandwidthEstimatorType estimator_type)
+    : PacketSender(listener, source->flow_id()),
+      source_(source),
+      bwe_(CreateBweSender(estimator_type,
+                           source_->bits_per_second() / 1000,
+                           this,
+                           &clock_)),
+      previous_sending_bitrate_(0) {
+  modules_.push_back(bwe_.get());
+}
+
+VideoSender::~VideoSender() {
+}
+
+void VideoSender::Pause() {
+  previous_sending_bitrate_ = TargetBitrateKbps();
+  PacketSender::Pause();
+}
+
+void VideoSender::Resume(int64_t paused_time_ms) {
+  source_->SetBitrateBps(previous_sending_bitrate_);
+  PacketSender::Resume(paused_time_ms);
+}
+
+void VideoSender::RunFor(int64_t time_ms, Packets* in_out) {
+  std::list<FeedbackPacket*> feedbacks = GetFeedbackPackets(
+      in_out, clock_.TimeInMilliseconds() + time_ms, source_->flow_id());
+  ProcessFeedbackAndGeneratePackets(time_ms, &feedbacks, in_out);
+}
+
+void VideoSender::ProcessFeedbackAndGeneratePackets(
+    int64_t time_ms,
+    std::list<FeedbackPacket*>* feedbacks,
+    Packets* packets) {
+  do {
+    // Make sure to at least run Process() below every 100 ms.
+    int64_t time_to_run_ms = std::min<int64_t>(time_ms, 100);
+    if (!feedbacks->empty()) {
+      int64_t time_until_feedback_ms =
+          feedbacks->front()->send_time_ms() - clock_.TimeInMilliseconds();
+      time_to_run_ms =
+          std::max<int64_t>(std::min(time_ms, time_until_feedback_ms), 0);
+    }
+
+    if (!running_) {
+      source_->SetBitrateBps(0);
+    }
+
+    Packets generated;
+    source_->RunFor(time_to_run_ms, &generated);
+    bwe_->OnPacketsSent(generated);
+    packets->merge(generated, DereferencingComparator<Packet>);
+
+    clock_.AdvanceTimeMilliseconds(time_to_run_ms);
+
+    if (!feedbacks->empty()) {
+      bwe_->GiveFeedback(*feedbacks->front());
+      delete feedbacks->front();
+      feedbacks->pop_front();
+    }
+
+    bwe_->Process();
+
+    time_ms -= time_to_run_ms;
+  } while (time_ms > 0);
+  assert(feedbacks->empty());
+}
+
+int VideoSender::GetFeedbackIntervalMs() const {
+  return bwe_->GetFeedbackIntervalMs();
+}
+
+void VideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
+                                   uint8_t fraction_lost,
+                                   int64_t rtt) {
   source_->SetBitrateBps(target_bitrate_bps);
-  std::stringstream ss;
-  ss << "SendEstimate_" << source_->flow_id() << "#1";
-  BWE_TEST_LOGGING_PLOT(ss.str(), clock_.TimeInMilliseconds(),
-                        target_bitrate_bps / 1000);
+  RecordBitrate();
+}
+
+uint32_t VideoSender::TargetBitrateKbps() {
+  return (source_->bits_per_second() + 500) / 1000;
 }
 
 PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
                                    VideoSource* source,
                                    BandwidthEstimatorType estimator)
-    : PacketSender(listener, source, estimator),
+    : VideoSender(listener, source, estimator),
       pacer_(&clock_,
              this,
              source->bits_per_second() / 1000,
@@ -132,14 +177,16 @@ void PacedVideoSender::RunFor(int64_t time_ms, Packets* in_out) {
   int64_t end_time_ms = clock_.TimeInMilliseconds() + time_ms;
   // Run process periodically to allow the packets to be paced out.
   std::list<FeedbackPacket*> feedbacks =
-      GetFeedbackPackets(in_out, end_time_ms);
+      GetFeedbackPackets(in_out, end_time_ms, source_->flow_id());
   int64_t last_run_time_ms = -1;
+  BWE_TEST_LOGGING_CONTEXT("Sender");
+  BWE_TEST_LOGGING_CONTEXT(source_->flow_id());
   do {
     int64_t time_until_process_ms = TimeUntilNextProcess(modules_);
     int64_t time_until_feedback_ms = time_ms;
     if (!feedbacks.empty())
-      time_until_feedback_ms = feedbacks.front()->send_time_us() / 1000 -
-                               clock_.TimeInMilliseconds();
+      time_until_feedback_ms = std::max<int64_t>(
+          feedbacks.front()->send_time_ms() - clock_.TimeInMilliseconds(), 0);
 
     int64_t time_until_next_event_ms =
         std::min(time_until_feedback_ms, time_until_process_ms);
@@ -162,11 +209,10 @@ void PacedVideoSender::RunFor(int64_t time_ms, Packets* in_out) {
     if (!generated_packets.empty()) {
       for (Packet* packet : generated_packets) {
         MediaPacket* media_packet = static_cast<MediaPacket*>(packet);
-        pacer_.SendPacket(PacedSender::kNormalPriority,
-                          media_packet->header().ssrc,
-                          media_packet->header().sequenceNumber,
-                          (media_packet->send_time_us() + 500) / 1000,
-                          media_packet->payload_size(), false);
+        pacer_.InsertPacket(
+            PacedSender::kNormalPriority, media_packet->header().ssrc,
+            media_packet->header().sequenceNumber, media_packet->send_time_ms(),
+            media_packet->payload_size(), false);
         pacer_queue_.push_back(packet);
         assert(pacer_queue_.size() < 10000);
       }
@@ -225,6 +271,8 @@ void PacedVideoSender::QueuePackets(Packets* batch,
   }
   Packets to_transfer;
   to_transfer.splice(to_transfer.begin(), queue_, queue_.begin(), it);
+  for (Packet* packet : to_transfer)
+    packet->set_paced(true);
   bwe_->OnPacketsSent(to_transfer);
   batch->merge(to_transfer, DereferencingComparator<Packet>);
 }
@@ -238,11 +286,14 @@ bool PacedVideoSender::TimeToSendPacket(uint32_t ssrc,
     MediaPacket* media_packet = static_cast<MediaPacket*>(*it);
     if (media_packet->header().sequenceNumber == sequence_number) {
       int64_t pace_out_time_ms = clock_.TimeInMilliseconds();
+
       // Make sure a packet is never paced out earlier than when it was put into
       // the pacer.
-      assert(pace_out_time_ms >= (media_packet->send_time_us() + 500) / 1000);
+      assert(pace_out_time_ms >= media_packet->send_time_ms());
+
       media_packet->SetAbsSendTimeMs(pace_out_time_ms);
       media_packet->set_send_time_us(1000 * pace_out_time_ms);
+      media_packet->set_sender_timestamp_us(1000 * pace_out_time_ms);
       queue_.push_back(media_packet);
       pacer_queue_.erase(it);
       return true;
@@ -258,11 +309,186 @@ size_t PacedVideoSender::TimeToSendPadding(size_t bytes) {
 void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
                                         uint8_t fraction_lost,
                                         int64_t rtt) {
-  PacketSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
+  VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
   pacer_.UpdateBitrate(
       target_bitrate_bps / 1000,
       PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000, 0);
 }
+
+const int kNoLimit = std::numeric_limits<int>::max();
+const int kPacketSizeBytes = 1200;
+
+TcpSender::TcpSender(PacketProcessorListener* listener,
+                     int flow_id,
+                     int64_t offset_ms)
+    : TcpSender(listener, flow_id, offset_ms, kNoLimit) {
+}
+
+TcpSender::TcpSender(PacketProcessorListener* listener,
+                     int flow_id,
+                     int64_t offset_ms,
+                     int send_limit_bytes)
+    : PacketSender(listener, flow_id),
+      cwnd_(10),
+      ssthresh_(kNoLimit),
+      ack_received_(false),
+      last_acked_seq_num_(0),
+      next_sequence_number_(0),
+      offset_ms_(offset_ms),
+      last_reduction_time_ms_(-1),
+      last_rtt_ms_(0),
+      total_sent_bytes_(0),
+      send_limit_bytes_(send_limit_bytes),
+      last_generated_packets_ms_(0),
+      num_recent_sent_packets_(0),
+      bitrate_kbps_(0) {
+}
+
+void TcpSender::RunFor(int64_t time_ms, Packets* in_out) {
+  if (clock_.TimeInMilliseconds() + time_ms < offset_ms_) {
+    clock_.AdvanceTimeMilliseconds(time_ms);
+    if (running_) {
+      Pause();
+    }
+    return;
+  }
+
+  if (!running_ && total_sent_bytes_ == 0) {
+    Resume(offset_ms_);
+  }
+
+  int64_t start_time_ms = clock_.TimeInMilliseconds();
+
+  std::list<FeedbackPacket*> feedbacks = GetFeedbackPackets(
+      in_out, clock_.TimeInMilliseconds() + time_ms, *flow_ids().begin());
+  // The number of packets which are sent in during time_ms depends on the
+  // number of packets in_flight_ and the max number of packets in flight
+  // (cwnd_). Therefore SendPackets() isn't directly dependent on time_ms.
+  for (FeedbackPacket* fb : feedbacks) {
+    clock_.AdvanceTimeMilliseconds(fb->send_time_ms() -
+                                   clock_.TimeInMilliseconds());
+    last_rtt_ms_ = fb->send_time_ms() - fb->latest_send_time_ms();
+    UpdateCongestionControl(fb);
+    SendPackets(in_out);
+  }
+
+  for (auto it = in_flight_.begin(); it != in_flight_.end();) {
+    if (it->time_ms < clock_.TimeInMilliseconds() - 1000)
+      in_flight_.erase(it++);
+    else
+      ++it;
+  }
+
+  clock_.AdvanceTimeMilliseconds(time_ms -
+                                 (clock_.TimeInMilliseconds() - start_time_ms));
+  SendPackets(in_out);
+}
+
+void TcpSender::SendPackets(Packets* in_out) {
+  int cwnd = ceil(cwnd_);
+  int packets_to_send = std::max(cwnd - static_cast<int>(in_flight_.size()), 0);
+  int timed_out = TriggerTimeouts();
+  if (timed_out > 0) {
+    HandleLoss();
+  }
+  if (packets_to_send > 0) {
+    Packets generated = GeneratePackets(packets_to_send);
+    for (Packet* packet : generated)
+      in_flight_.insert(InFlight(*static_cast<MediaPacket*>(packet)));
+
+    in_out->merge(generated, DereferencingComparator<Packet>);
+  }
+}
+
+void TcpSender::UpdateCongestionControl(const FeedbackPacket* fb) {
+  const TcpFeedback* tcp_fb = static_cast<const TcpFeedback*>(fb);
+  RTC_DCHECK(!tcp_fb->acked_packets().empty());
+  ack_received_ = true;
+
+  uint16_t expected = tcp_fb->acked_packets().back() - last_acked_seq_num_;
+  uint16_t missing =
+      expected - static_cast<uint16_t>(tcp_fb->acked_packets().size());
+
+  for (uint16_t ack_seq_num : tcp_fb->acked_packets())
+    in_flight_.erase(InFlight(ack_seq_num, clock_.TimeInMilliseconds()));
+
+  if (missing > 0) {
+    HandleLoss();
+  } else if (cwnd_ <= ssthresh_) {
+    cwnd_ += tcp_fb->acked_packets().size();
+  } else {
+    cwnd_ += 1.0f / cwnd_;
+  }
+
+  last_acked_seq_num_ =
+      LatestSequenceNumber(tcp_fb->acked_packets().back(), last_acked_seq_num_);
+}
+
+int TcpSender::TriggerTimeouts() {
+  int timed_out = 0;
+  for (auto it = in_flight_.begin(); it != in_flight_.end();) {
+    if (it->time_ms < clock_.TimeInMilliseconds() - 1000) {
+      in_flight_.erase(it++);
+      ++timed_out;
+    } else {
+      ++it;
+    }
+  }
+  return timed_out;
+}
+
+void TcpSender::HandleLoss() {
+  if (clock_.TimeInMilliseconds() - last_reduction_time_ms_ < last_rtt_ms_)
+    return;
+  last_reduction_time_ms_ = clock_.TimeInMilliseconds();
+  ssthresh_ = std::max(static_cast<int>(in_flight_.size() / 2), 2);
+  cwnd_ = ssthresh_;
+}
+
+Packets TcpSender::GeneratePackets(size_t num_packets) {
+  Packets generated;
+
+  UpdateSendBitrateEstimate(num_packets);
+
+  for (size_t i = 0; i < num_packets; ++i) {
+    if ((total_sent_bytes_ + kPacketSizeBytes) > send_limit_bytes_) {
+      if (running_) {
+        Pause();
+      }
+      break;
+    }
+    generated.push_back(
+        new MediaPacket(*flow_ids().begin(), 1000 * clock_.TimeInMilliseconds(),
+                        kPacketSizeBytes, next_sequence_number_++));
+    generated.back()->set_sender_timestamp_us(
+        1000 * clock_.TimeInMilliseconds());
+
+    total_sent_bytes_ += kPacketSizeBytes;
+  }
+
+  return generated;
+}
+
+void TcpSender::UpdateSendBitrateEstimate(size_t num_packets) {
+  const int kTimeWindowMs = 500;
+  num_recent_sent_packets_ += num_packets;
+
+  int64_t delta_ms = clock_.TimeInMilliseconds() - last_generated_packets_ms_;
+  if (delta_ms >= kTimeWindowMs) {
+    bitrate_kbps_ =
+        static_cast<uint32_t>(8 * num_recent_sent_packets_ * kPacketSizeBytes) /
+        delta_ms;
+    last_generated_packets_ms_ = clock_.TimeInMilliseconds();
+    num_recent_sent_packets_ = 0;
+  }
+
+  RecordBitrate();
+}
+
+uint32_t TcpSender::TargetBitrateKbps() {
+  return bitrate_kbps_;
+}
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
index 8c13cb16ef..f48ed62f57 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -12,11 +12,13 @@
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_
 
 #include <list>
+#include <limits>
+#include <set>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 
@@ -24,46 +26,81 @@ namespace webrtc {
 namespace testing {
 namespace bwe {
 
-class PacketSender : public PacketProcessor, public BitrateObserver {
- public:
-  PacketSender(PacketProcessorListener* listener,
-               VideoSource* source,
-               BandwidthEstimatorType estimator);
-  virtual ~PacketSender();
+class MetricRecorder;
 
+class PacketSender : public PacketProcessor {
+ public:
+  PacketSender(PacketProcessorListener* listener, int flow_id)
+      : PacketProcessor(listener, flow_id, kSender),
+        running_(true),
+        // For Packet::send_time_us() to be comparable with timestamps from
+        // clock_, the clock of the PacketSender and the Source must be aligned.
+        // We assume that both start at time 0.
+        clock_(0),
+        metric_recorder_(nullptr) {}
+  virtual ~PacketSender() {}
   // Call GiveFeedback() with the returned interval in milliseconds, provided
   // there is a new estimate available.
   // Note that changing the feedback interval affects the timing of when the
   // output of the estimators is sampled and therefore the baseline files may
   // have to be regenerated.
-  virtual int GetFeedbackIntervalMs() const;
+  virtual int GetFeedbackIntervalMs() const = 0;
+  void SetSenderTimestamps(Packets* in_out);
+
+  virtual uint32_t TargetBitrateKbps() { return 0; }
+
+  virtual void Pause();
+  virtual void Resume(int64_t paused_time_ms);
+
+  void set_metric_recorder(MetricRecorder* metric_recorder);
+  virtual void RecordBitrate();
+
+ protected:
+  bool running_;  // Initialized by default as true.
+  SimulatedClock clock_;
+
+ private:
+  MetricRecorder* metric_recorder_;
+};
+
+class VideoSender : public PacketSender, public BitrateObserver {
+ public:
+  VideoSender(PacketProcessorListener* listener,
+              VideoSource* source,
+              BandwidthEstimatorType estimator);
+  virtual ~VideoSender();
+
+  int GetFeedbackIntervalMs() const override;
   void RunFor(int64_t time_ms, Packets* in_out) override;
 
   virtual VideoSource* source() const { return source_; }
 
+  uint32_t TargetBitrateKbps() override;
+
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t target_bitrate_bps,
                         uint8_t fraction_lost,
                         int64_t rtt) override;
 
+  void Pause() override;
+  void Resume(int64_t paused_time_ms) override;
+
  protected:
   void ProcessFeedbackAndGeneratePackets(int64_t time_ms,
                                          std::list<FeedbackPacket*>* feedbacks,
                                          Packets* generated);
-  std::list<FeedbackPacket*> GetFeedbackPackets(Packets* in_out,
-                                                int64_t end_time_ms);
 
-  SimulatedClock clock_;
   VideoSource* source_;
   rtc::scoped_ptr<BweSender> bwe_;
   int64_t start_of_run_ms_;
   std::list<Module*> modules_;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(PacketSender);
+  uint32_t previous_sending_bitrate_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(VideoSender);
 };
 
-class PacedVideoSender : public PacketSender, public PacedSender::Callback {
+class PacedVideoSender : public VideoSender, public PacedSender::Callback {
  public:
   PacedVideoSender(PacketProcessorListener* listener,
                    VideoSource* source,
@@ -93,7 +130,64 @@ class PacedVideoSender : public PacketSender, public PacedSender::Callback {
   Packets queue_;
   Packets pacer_queue_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(PacedVideoSender);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacedVideoSender);
+};
+
+class TcpSender : public PacketSender {
+ public:
+  TcpSender(PacketProcessorListener* listener, int flow_id, int64_t offset_ms);
+  TcpSender(PacketProcessorListener* listener,
+            int flow_id,
+            int64_t offset_ms,
+            int send_limit_bytes);
+  virtual ~TcpSender() {}
+
+  void RunFor(int64_t time_ms, Packets* in_out) override;
+  int GetFeedbackIntervalMs() const override { return 10; }
+
+  uint32_t TargetBitrateKbps() override;
+
+ private:
+  struct InFlight {
+   public:
+    explicit InFlight(const MediaPacket& packet)
+        : sequence_number(packet.header().sequenceNumber),
+          time_ms(packet.send_time_ms()) {}
+
+    InFlight(uint16_t seq_num, int64_t now_ms)
+        : sequence_number(seq_num), time_ms(now_ms) {}
+
+    bool operator<(const InFlight& rhs) const {
+      return sequence_number < rhs.sequence_number;
+    }
+
+    uint16_t sequence_number;  // Sequence number of a packet in flight, or a
+                               // packet which has just been acked.
+    int64_t time_ms;  // Time of when the packet left the sender, or when the
+                      // ack was received.
+  };
+
+  void SendPackets(Packets* in_out);
+  void UpdateCongestionControl(const FeedbackPacket* fb);
+  int TriggerTimeouts();
+  void HandleLoss();
+  Packets GeneratePackets(size_t num_packets);
+  void UpdateSendBitrateEstimate(size_t num_packets);
+
+  float cwnd_;
+  int ssthresh_;
+  std::set<InFlight> in_flight_;
+  bool ack_received_;
+  uint16_t last_acked_seq_num_;
+  uint16_t next_sequence_number_;
+  int64_t offset_ms_;
+  int64_t last_reduction_time_ms_;
+  int64_t last_rtt_ms_;
+  int total_sent_bytes_;
+  int send_limit_bytes_;  // Initialized by default as kNoLimit.
+  int64_t last_generated_packets_ms_;
+  size_t num_recent_sent_packets_;
+  uint32_t bitrate_kbps_;
 };
 }  // namespace bwe
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_bars.sh b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_bars.sh
new file mode 100644
index 0000000000..9f7fb16203
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_bars.sh
@@ -0,0 +1,286 @@
+#!/bin/bash
+
+# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# To set up in e.g. Eclipse, run a separate shell and pipe the output from the
+# test into this script.
+#
+# In Eclipse, that amounts to creating a Run Configuration which starts
+# "/bin/bash" with the arguments "-c [trunk_path]/out/Debug/modules_unittests
+# --gtest_filter=*BweTest* | [trunk_path]/webrtc/modules/
+# remote_bitrate_estimator/test/plot_bars.sh
+
+# This script supports multiple figures (windows), the figure is specified as an
+# identifier at the first argument after the PLOT command. Each figure has a
+# single y axis and a dual y axis mode. If any line specifies an axis by ending
+# with "#<axis number (1 or 2)>" two y axis will be used, the first will be
+# assumed to represent bitrate (in kbps) and the second will be assumed to
+# represent time deltas (in ms).
+
+log=$(</dev/stdin)
+
+# Plot histograms.
+function gen_gnuplot_bar_input {
+  x_start=1
+  x_end=3.75
+  bars=$(echo "$log" | grep "BAR")
+
+  labels=$(echo "$log" | grep "^LABEL")
+  figures=($(echo "$bars" | cut -f 2 | sort | uniq))
+
+  echo "reset"  # Clears previous settings.
+
+  echo "set title font 'Verdana,22'"
+  echo "set xtics font 'Verdana,24'"
+  echo "set ytics font 'Verdana,14'"
+  echo "set ylabel font 'Verdana,16'"
+
+  echo "set xrange[$x_start:$x_end]"
+  echo "set style fill solid 0.5"
+  echo "set style fill solid border -1"
+
+  declare -a ydist=(11.5 10.5 10.5)  # Used to correctly offset the y label.
+  i=0
+  for figure in "${figures[@]}" ; do
+
+    echo "set terminal wxt $figure size 440,440 dashed"
+    echo "set ylabel offset ${ydist[$i]}, -3"
+    (( i++ ))
+
+    title=$(echo "$labels" | grep "^LABEL.$figure" | cut -f 3 | \
+                                         head -n 1 | sed 's/_/ /g')
+    y_label=$(echo "$labels" | grep "^LABEL.$figure" | cut -f 4 | \
+                                         head -n 1 | sed 's/_/ /g')
+
+    # RMCAT flows.
+    num_flows=$(echo "$labels" | grep "^LABEL.$figure" | cut -f 5 | \
+                                         head -n 1)
+
+    # RMCAT algorithm 1.
+    x_label_1=$(echo "$log" | grep "BAR.$figure" | cut -f 3 | sed 's/_/\t/g' \
+                                 | cut -f 1  | sort | uniq | head -n 1 )
+
+    # RMCAT algorithm 2.
+    x_label_2=$(echo "$log" | grep "BAR.$figure" | cut -f 3 | sed 's/_/\t/g' \
+                                 | cut -f 1  | sort | uniq |  sed -n 2p)
+
+    x_labels="('$x_label_1' 2, '$x_label_2' 3)"
+    tcp_flow=false
+
+    tcp_space=0.2  # Extra horizontal space between bars.
+
+    # Parse labels if there are other flows in addition to RMCAT ones.
+    IFS='x' read -ra split_label_1 <<< "$x_label_1"
+
+    if (( ${#split_label_1[@]} > "1" )); then
+      tcp_flow=true
+      box_width=$(echo "(1.0-$tcp_space/2)/$num_flows" | bc -l)
+      echo "set xtics font 'Verdana,16'"
+      x_labels="("
+      delimiter=""
+      abscissa=$(echo $x_start + 0.5 + 0.5*$box_width | bc)
+      for label in "${split_label_1[@]}" ; do
+        x_labels+="$delimiter'$label' $abscissa"
+        abscissa=$(echo $abscissa + $box_width | bc)
+        delimiter=", "
+      done
+      abscissa=$(echo $abscissa + $tcp_space | bc)
+      IFS='x' read -ra split_label_2 <<< "$x_label_2"
+      for label in "${split_label_2[@]}" ; do
+        x_labels+="$delimiter'$label' $abscissa"
+        abscissa=$(echo $abscissa + $box_width | bc)
+      done
+      x_labels="$x_labels)"
+    else
+      box_width=$(echo 1.0/$num_flows | bc -l)
+    fi
+
+    echo "set boxwidth $box_width"
+
+    # Plots can be directly exported to image files.
+    file_name=$(echo "$labels" | grep "^LABEL.$figure" | cut -f 5 | head -n 1)
+
+    y_max=0  # Used to scale the plot properly.
+
+    # Scale all latency plots with the same vertical scale.
+    delay_figure=5
+    if (( $figure==$delay_figure )) ; then
+      y_max=400
+    else  # Take y_max = 1.1 * highest plot value.
+
+      # Since only the optimal bitrate for the first flow is being ploted,
+      # consider only this one for scalling purposes.
+      data_sets=$(echo "$bars" | grep "LIMITERRORBAR.$figure" | cut -f 3 | \
+                                     sed 's/_/\t/g' | cut -f 1 | sort | uniq)
+
+      if (( ${#data_sets[@]} > "0" )); then
+        for set in $data_sets ; do
+          y=$(echo "$bars" | grep "LIMITERRORBAR.$figure.$set" | cut -f 8 | \
+                                                                  head -n 1)
+          if (( $(bc <<< "$y > $y_max") == 1 )); then
+            y_max=$y
+          fi
+        done
+      fi
+
+      data_sets=$(echo "$bars" | grep "ERRORBAR.$figure" | cut -f 3 | \
+                                                               sort | uniq)
+      if (( ${#data_sets[@]} > "0" )); then
+        for set in $data_sets ; do
+          y=$(echo "$bars" | grep "ERRORBAR.$figure.$set" | cut -f 6 | \
+                                                                 head -n 1)
+          if (( $(bc <<< "$y > $y_max") == 1 )) ; then
+            y_max=$y
+          fi
+        done
+      fi
+
+      data_sets=$(echo "$bars" | grep "BAR.$figure" | cut -f 3 | sort | uniq)
+
+      for set in $data_sets ; do
+        y=$(echo "$bars" | grep "BAR.$figure.$set" | cut -f 4 | head -n 1)
+        if (( $(bc <<< "$y > $y_max") == 1 )) ; then
+          y_max=$y
+        fi
+      done
+
+      y_max=$(echo $y_max*1.1 | bc)
+    fi
+
+
+    echo "set ylabel \"$y_label\""
+    echo "set yrange[0:$y_max]"
+
+    echo "set multiplot"
+
+    # Plot bars.
+    data_sets=$(echo "$bars" | grep "BAR.$figure" | cut -f 3 | sort | uniq)
+
+    echo "set xtics $x_labels"
+    echo "plot '-' using 1:4:2 with boxes lc variable notitle"
+
+    echo
+
+    color=11  # Green.
+    x_bar=$(echo $x_start + 0.5 + 0.5*$box_width | bc)
+    for set in $data_sets ; do
+      echo -n "$x_bar  $color  "
+      echo "$bars" | grep "BAR.$figure.$set" | cut -f 3,4
+
+      # Add extra space if TCP flows are being plotted.
+      if $tcp_flow && \
+          (( $(bc <<< "$x_bar < $x_start + 1.5 - 0.5*$tcp_space") == 1 )) && \
+          (( $(bc <<< "$x_bar + $box_width > $x_start + 1.5 + 0.5*$tcp_space") \
+           == 1 )); then
+          x_bar=$(echo $x_bar + $tcp_space | bc)
+      fi
+
+      x_bar=$(echo $x_bar + $box_width | bc)
+
+      if (( $(bc <<< "$x_bar > 2.5") == 1 )) ; then
+        color=12  # Blue.
+      fi
+      # Different bar color for TCP flows:
+      if $tcp_flow && \
+         (( $(bc <<< "(100*$x_bar)%100 < 50") == 1 ))
+      then
+        color=18  # Gray.
+      fi
+    done
+    echo "e"
+
+    # Plot Baseline bars, e.g. one-way path delay on latency plots.
+    data_sets=$(echo "$log" | grep "BASELINE.$figure" | cut -f 3 | sort | uniq)
+
+    if (( ${#data_sets} > "0" )); then
+      echo "set xtics $x_labels"
+      echo "plot '-' using 1:4:2 with boxes lc variable notitle"
+
+      echo
+
+      color=18  # Gray.
+      x_bar=$(echo $x_start + 0.5 + 0.5*$box_width | bc)
+      for set in $data_sets ; do
+        echo -n "$x_bar  $color  "
+        echo "$log" | grep "BASELINE.$figure.$set" | cut -f 3,4
+
+        # Add extra space if TCP flows are being plotted.
+        if $tcp_flow && \
+            (( $(bc <<< "$x_bar < $x_start + 1.5 - 0.5*$tcp_space") == 1 )) && \
+            (( $(bc <<< "$x_bar + $box_width > $x_start + 1.5 \
+            + 0.5*$tcp_space") == 1 )); then
+            x_bar=$(echo $x_bar + $tcp_space | bc)
+        fi
+
+        x_bar=$(echo $x_bar + $box_width | bc)
+
+      done
+      echo "e"
+    fi
+
+    # Plot vertical error lines, e.g. y +- sigma.
+    data_sets=$(echo "$bars" | grep "ERRORBAR.$figure" | cut -f 3 | sort | uniq)
+
+    if (( ${#data_sets} > "0" )); then
+
+      echo "set key left"
+      error_title=$(echo "$bars" | grep "ERRORBAR.$figure" | cut -f 7 | \
+                                                 head -n 1 | sed 's/_/ /g')
+
+      echo "set xtics $x_labels"
+      echo "plot '-' using 1:3:4:5 title '$error_title' with yerr"
+
+      x_error_line=$(echo $x_start + 0.5 + 0.5*$box_width | bc)
+      for set in $data_sets ; do
+        echo -n "$x_error_line  "
+        echo "$bars" | grep "ERRORBAR.$figure.$set" | cut -f 3,4,5,6
+
+        # Add extra space if TCP flows are being plotted.
+        if $tcp_flow && \
+          (( $(bc <<< "$x_error_line < $x_start + 1.5 - 0.5*$tcp_space") == 1 \
+          )) && (( $(bc <<< "$x_error_line + $box_width > $x_start + 1.5 \
+          + 0.5*$tcp_space") == 1 )); then
+          x_error_line=$(echo $x_error_line + $tcp_space | bc)
+        fi
+
+        x_error_line=$(echo $x_error_line + $box_width | bc)
+      done
+      echo "e"
+    fi
+
+    # Plot horizontal dashed lines, e.g. y = optimal bitrate.
+    data_sets=$(echo "$bars" | grep "LIMITERRORBAR.$figure" | cut -f 3 \
+                                                     | sort | uniq)
+    if (( ${#data_sets} > "0" )); then
+
+      echo "set style line 1 lt 1 lw 3 pt 3 ps 0 linecolor rgb 'black'"
+
+      limit_titles=$(echo "$bars" | grep "LIMITERRORBAR.$figure" | cut -f 9 \
+                                                         | sort | uniq)
+
+      for title in $limit_titles ; do
+        y_max=$(echo "$bars" | grep "LIMITERRORBAR.$figure" | grep "$title" \
+                                               | cut -f 8 | head -n 1)
+
+        retouched_title=$(echo "$title" | sed 's/#/\t/g' | cut -f 1 \
+                                                         | sed 's/_/ /g')
+
+        echo "set key right top"
+        echo "set xtics $x_labels"
+        echo "plot $y_max lt 7 lw 1 linecolor rgb 'black' \
+                                  title '$retouched_title'"
+      done
+
+    fi
+
+    echo "unset multiplot"
+  done
+}
+
+gen_gnuplot_bar_input | gnuplot -persist
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
new file mode 100644
index 0000000000..1bae1e81f0
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
@@ -0,0 +1,166 @@
+#!/usr/bin/env python
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# This script is used to plot simulation dynamics.
+# Able to plot each flow separately. Other plot boxes can be added,
+# currently one for Throughput, one for Latency and one for Packet Loss.
+
+import matplotlib
+import matplotlib.pyplot as plt
+import numpy
+import re
+import sys
+
+# Change this to True to save the figure to a file. Look below for details.
+save_figure = False
+
+class Variable(object):
+  def __init__(self, variable):
+    self._ID = variable[0]
+    self._xlabel = variable[1]
+    self._ylabel = variable[2]
+    self._subplot = variable[3]
+    self._y_max = variable[4]
+    self.samples = dict()
+
+  def getID(self):
+    return self._ID
+
+  def getXLabel(self):
+    return self._xlabel
+
+  def getYLabel(self):
+    return self._ylabel
+
+  def getSubplot(self):
+    return self._subplot
+
+  def getYMax(self):
+    return self._y_max
+
+  def getNumberOfFlows(self):
+    return len(self.samples)
+
+
+  def addSample(self, line):
+    groups = re.search(r'_(((\d)+((,(\d)+)*))_(\D+))#\d@(\S+)', line)
+
+    # Each variable will be plotted in a separated box.
+    var_name = groups.group(1)
+    alg_name = groups.group(8)
+
+    alg_name = alg_name.replace('_', ' ')
+
+    if alg_name not in self.samples.keys():
+      self.samples[alg_name] = {}
+
+    if var_name not in self.samples[alg_name].keys():
+      self.samples[alg_name][var_name] = []
+
+    sample = re.search(r'(\d+\.\d+)\t([-]?\d+\.\d+)', line)
+
+    s = (sample.group(1), sample.group(2))
+    self.samples[alg_name][var_name].append(s)
+
+def plotVar(v, ax, show_legend, show_x_label):
+  if show_x_label:
+    ax.set_xlabel(v.getXLabel(), fontsize='large')
+  ax.set_ylabel(v.getYLabel(), fontsize='large')
+
+  for alg in v.samples.keys():
+
+    for series in v.samples[alg].keys():
+
+      x = [sample[0] for sample in v.samples[alg][series]]
+      y = [sample[1] for sample in v.samples[alg][series]]
+      x = numpy.array(x)
+      y = numpy.array(y)
+
+      line = plt.plot(x, y, label=alg, linewidth=4.0)
+
+      colormap = {'Available0':'#AAAAAA',
+                  'Available1':'#AAAAAA',
+                  'GCC0':'#80D000',
+                  'GCC1':'#008000',
+                  'GCC2':'#00F000',
+                  'GCC3':'#00B000',
+                  'GCC4':'#70B020',
+                  'NADA0':'#0000AA',
+                  'NADA1':'#A0A0FF',
+                  'NADA2':'#0000FF',
+                  'NADA3':'#C0A0FF',
+                  'NADA4':'#9060B0',}
+
+      flow_id = re.search(r'(\d+(,\d+)*)', series)  # One or multiple ids.
+      key = alg + flow_id.group(1)
+
+      if key in colormap:
+        plt.setp(line, color=colormap[key])
+      elif alg == 'TCP':
+        plt.setp(line, color='#AAAAAA')
+      else:
+        plt.setp(line, color='#654321')
+
+      if alg.startswith('Available'):
+        plt.setp(line, linestyle='--')
+      plt.grid(True)
+
+      # x1, x2, y1, y2
+      _, x2, _, y2 = plt.axis()
+      if v.getYMax() >= 0:
+        y2 = v.getYMax()
+      plt.axis((0, x2, 0, y2))
+
+    if show_legend:
+      plt.legend(loc='upper center', bbox_to_anchor=(0.5, 1.40),
+                 shadow=True, fontsize='large', ncol=len(v.samples))
+
+def main():
+  variables = [
+          ('Throughput_kbps', "Time (s)", "Throughput (kbps)", 1, 4000),
+          ('Delay_ms', "Time (s)", "One-way Delay (ms)", 2, 500),
+          ('Packet_Loss', "Time (s)", "Packet Loss Ratio", 3, 1.0),
+          # ('Sending_Estimate_kbps', "Time (s)", "Sending Estimate (kbps)",
+          #                                                        4, 4000),
+          ]
+
+  var = []
+
+  # Create objects.
+  for variable in variables:
+    var.append(Variable(variable))
+
+  # Add samples to the objects.
+  for line in sys.stdin:
+    if line.startswith("[ RUN      ]"):
+      test_name = re.search(r'\.(\w+)', line).group(1)
+    if line.startswith("PLOT"):
+      for v in var:
+        if v.getID() in line:
+          v.addSample(line)
+
+  matplotlib.rcParams.update({'font.size': 48/len(variables)})
+
+  # Plot variables.
+  fig = plt.figure()
+
+  # Offest and threshold on the same plot.
+  n = var[-1].getSubplot()
+  i = 0
+  for v in var:
+    ax = fig.add_subplot(n, 1, v.getSubplot())
+    plotVar(v, ax, i == 0, i == n - 1)
+    i += 1
+
+  if save_figure:
+    fig.savefig(test_name + ".png")
+  plt.show()
+
+if __name__ == '__main__':
+  main()
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.sh b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.sh
new file mode 100644
index 0000000000..fd104a1704
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.sh
@@ -0,0 +1,87 @@
+#!/bin/bash
+
+# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# To set up in e.g. Eclipse, run a separate shell and pipe the output from the
+# test into this script.
+#
+# In Eclipse, that amounts to creating a Run Configuration which starts
+# "/bin/bash" with the arguments "-c [trunk_path]/out/Debug/modules_unittests
+# --gtest_filter=*BweTest* | [trunk_path]/webrtc/modules/
+# remote_bitrate_estimator/test/plot_dynamics.sh
+
+# This script supports multiple figures (windows), the figure is specified as an
+# identifier at the first argument after the PLOT command. Each figure has a
+# single y axis and a dual y axis mode. If any line specifies an axis by ending
+# with "#<axis number (1 or 2)>" two y axis will be used, the first will be
+# assumed to represent bitrate (in kbps) and the second will be assumed to
+# represent time deltas (in ms).
+
+log=$(</dev/stdin)
+
+# Plot dynamics.
+function gen_gnuplot_input {
+  colors=(a7001f 0a60c2 b2582b 21a66c d6604d 4393c3 f4a582 92c5de edcbb7 b1c5d0)
+  plots=$(echo "$log" | grep "^PLOT")
+  # Each figure corresponds to a separate plot window.
+  figures=($(echo "$plots" | cut -f 2 | sort | uniq))
+
+  for figure in "${figures[@]}" ; do
+    # Each data set corresponds to a plot line.
+    data_sets=$(echo "$plots" | grep "^PLOT.$figure" | cut -f 3 | sort | uniq)
+    # Lines can be scaled on the left (1) or right (2) axis.
+    linetypes=($(echo "$data_sets" | grep "#" | cut -d '#' -f 2 | \
+                 cut -d '@' -f 1 | uniq))
+
+    # Set plot configurations.
+    echo "reset; "
+    echo "set terminal wxt $figure size 1440,900 font \"Arial,9\"; "
+    echo "set xlabel \"Seconds\"; "
+    if (( "${#linetypes[@]}" > "1" )); then
+      echo "set ylabel 'Bitrate (kbps)';"  # Left side.
+      echo "set ytics nomirror;"
+      echo "set y2label 'Time delta (ms)';"  # Right side.
+      echo "set y2tics nomirror;"
+    else
+      # Single axis (left side), set its label according to data.
+      y_label=$(echo "$data_sets" | grep "#" | cut -d '#' -f 1  | \
+          cut -d ' ' -f 1 | cut -d '/' -f 3 | sed 's/[0-9]/#/g' | \
+          cut -d '#' -f 3 | head -n 1 | sed 's/_/ /g')
+      echo "set ylabel \"$y_label\";"
+    fi
+
+    i=0
+    echo -n "plot "
+    for set in $data_sets ; do
+      (( i++ )) && echo -n ","
+      echo -n "'-' with "
+      echo -n "linespoints "
+      echo -n "ps 0.5 "
+      echo -n "lc rgbcolor \"#${colors[$(($i % 10))]}\" "
+      if (( "${#linetypes[@]}" > "1" )); then
+        # Multiple sets can have a same line plot.
+        linetype=$(echo "$set" | grep "#" | cut -d '#' -f 2 | cut -d '@' -f 1)
+        if (( "${#linetype}" > "0")); then
+          echo -n "axes x1y$linetype "
+        else
+          # If no line type is specified, but line types are used, we will
+          # default to scale on the left axis.
+          echo -n "axes x1y1 "
+        fi
+      fi
+      echo -n "title \"$set\" "
+    done
+    echo
+    for set in $data_sets ; do
+      echo "$log" | grep "^PLOT.$figure.$set" | cut -f 4,5
+      echo "e"
+    done
+  done
+}
+gen_gnuplot_input | gnuplot -persist
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index e71c75ce39..f138035de5 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -11,14 +11,60 @@
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
 
 #include <stdio.h>
+
+#include <set>
+#include <sstream>
 #include <string>
 
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "gflags/gflags.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
-const int kMinBitrateBps = 30000;
+namespace flags {
+
+DEFINE_string(extension_type,
+              "abs",
+              "Extension type, either abs for absolute send time or tsoffset "
+              "for timestamp offset.");
+std::string ExtensionType() {
+  return static_cast<std::string>(FLAGS_extension_type);
+}
+
+DEFINE_int32(extension_id, 3, "Extension id.");
+int ExtensionId() {
+  return static_cast<int>(FLAGS_extension_id);
+}
+
+DEFINE_string(input_file, "", "Input file.");
+std::string InputFile() {
+  return static_cast<std::string>(FLAGS_input_file);
+}
+
+DEFINE_string(ssrc_filter,
+              "",
+              "Comma-separated list of SSRCs in hexadecimal which are to be "
+              "used as input to the BWE (only applicable to pcap files).");
+std::set<uint32_t> SsrcFilter() {
+  std::string ssrc_filter_string = static_cast<std::string>(FLAGS_ssrc_filter);
+  if (ssrc_filter_string.empty())
+    return std::set<uint32_t>();
+  std::stringstream ss;
+  std::string ssrc_filter = ssrc_filter_string;
+  std::set<uint32_t> ssrcs;
+
+  // Parse the ssrcs in hexadecimal format.
+  ss << std::hex << ssrc_filter;
+  uint32_t ssrc;
+  while (ss >> ssrc) {
+    ssrcs.insert(ssrc);
+    ss.ignore(1, ',');
+  }
+  return ssrcs;
+}
+}  // namespace flags
 
 bool ParseArgsAndSetupEstimator(int argc,
                                 char** argv,
@@ -28,39 +74,56 @@ bool ParseArgsAndSetupEstimator(int argc,
                                 webrtc::RtpHeaderParser** parser,
                                 webrtc::RemoteBitrateEstimator** estimator,
                                 std::string* estimator_used) {
-  *rtp_reader = webrtc::test::RtpFileReader::Create(
-      webrtc::test::RtpFileReader::kRtpDump, argv[3]);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  std::string filename = flags::InputFile();
+
+  std::set<uint32_t> ssrc_filter = flags::SsrcFilter();
+  fprintf(stderr, "Filter on SSRC: ");
+  for (auto& s : ssrc_filter) {
+    fprintf(stderr, "0x%08x, ", s);
+  }
+  fprintf(stderr, "\n");
+  if (filename.substr(filename.find_last_of(".")) == ".pcap") {
+    fprintf(stderr, "Opening as pcap\n");
+    *rtp_reader = webrtc::test::RtpFileReader::Create(
+        webrtc::test::RtpFileReader::kPcap, filename.c_str(),
+        flags::SsrcFilter());
+  } else {
+    fprintf(stderr, "Opening as rtp\n");
+    *rtp_reader = webrtc::test::RtpFileReader::Create(
+        webrtc::test::RtpFileReader::kRtpDump, filename.c_str());
+  }
   if (!*rtp_reader) {
-    fprintf(stderr, "Cannot open input file %s\n", argv[3]);
+    fprintf(stderr, "Cannot open input file %s\n", filename.c_str());
     return false;
   }
-  fprintf(stderr, "Input file: %s\n\n", argv[3]);
-  webrtc::RTPExtensionType extension = webrtc::kRtpExtensionAbsoluteSendTime;
+  fprintf(stderr, "Input file: %s\n\n", filename.c_str());
 
-  if (strncmp("tsoffset", argv[1], 8) == 0) {
+  webrtc::RTPExtensionType extension = webrtc::kRtpExtensionAbsoluteSendTime;
+  if (flags::ExtensionType() == "tsoffset") {
     extension = webrtc::kRtpExtensionTransmissionTimeOffset;
     fprintf(stderr, "Extension: toffset\n");
-  } else {
+  } else if (flags::ExtensionType() == "abs") {
     fprintf(stderr, "Extension: abs\n");
+  } else {
+    fprintf(stderr, "Unknown extension type\n");
+    return false;
   }
-  int id = atoi(argv[2]);
 
   // Setup the RTP header parser and the bitrate estimator.
   *parser = webrtc::RtpHeaderParser::Create();
-  (*parser)->RegisterRtpHeaderExtension(extension, id);
+  (*parser)->RegisterRtpHeaderExtension(extension, flags::ExtensionId());
   if (estimator) {
     switch (extension) {
       case webrtc::kRtpExtensionAbsoluteSendTime: {
-          webrtc::AbsoluteSendTimeRemoteBitrateEstimatorFactory factory;
-          *estimator = factory.Create(observer, clock, webrtc::kAimdControl,
-                                      kMinBitrateBps);
+        *estimator =
+            new webrtc::RemoteBitrateEstimatorAbsSendTime(observer, clock);
           *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator";
           break;
         }
       case webrtc::kRtpExtensionTransmissionTimeOffset: {
-          webrtc::RemoteBitrateEstimatorFactory factory;
-          *estimator = factory.Create(observer, clock, webrtc::kAimdControl,
-                                      kMinBitrateBps);
+        *estimator =
+            new webrtc::RemoteBitrateEstimatorSingleStream(observer, clock);
           *estimator_used = "RemoteBitrateEstimator";
           break;
         }
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
index 1c505ad835..4574faf8b7 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
@@ -14,8 +14,8 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 class Observer : public webrtc::RemoteBitrateObserver {
@@ -38,15 +38,6 @@ class Observer : public webrtc::RemoteBitrateObserver {
 };
 
 int main(int argc, char** argv) {
-  if (argc < 4) {
-    printf("Usage: bwe_rtp_play <extension type> <extension id> "
-           "<input_file.rtp>\n");
-    printf("<extension type> can either be:\n"
-           "  abs for absolute send time or\n"
-           "  tsoffset for timestamp offset.\n"
-           "<extension id> is the id associated with the extension.\n");
-    return -1;
-  }
   webrtc::test::RtpFileReader* reader;
   webrtc::RemoteBitrateEstimator* estimator;
   webrtc::RtpHeaderParser* parser;
@@ -76,23 +67,24 @@ int main(int argc, char** argv) {
   packet.time_ms = packet.time_ms - first_rtp_time_ms;
   while (true) {
     if (next_rtp_time_ms <= clock.TimeInMilliseconds()) {
-      webrtc::RTPHeader header;
-      parser->Parse(packet.data, packet.length, &header);
-      if (header.extension.hasAbsoluteSendTime)
-        ++abs_send_time_count;
-      if (header.extension.hasTransmissionTimeOffset)
-        ++ts_offset_count;
-      size_t packet_length = packet.length;
-      // Some RTP dumps only include the header, in which case packet.length
-      // is equal to the header length. In those cases packet.original_length
-      // usually contains the original packet length.
-      if (packet.original_length > 0) {
-        packet_length = packet.original_length;
+      if (!parser->IsRtcp(packet.data, packet.length)) {
+        webrtc::RTPHeader header;
+        parser->Parse(packet.data, packet.length, &header);
+        if (header.extension.hasAbsoluteSendTime)
+          ++abs_send_time_count;
+        if (header.extension.hasTransmissionTimeOffset)
+          ++ts_offset_count;
+        size_t packet_length = packet.length;
+        // Some RTP dumps only include the header, in which case packet.length
+        // is equal to the header length. In those cases packet.original_length
+        // usually contains the original packet length.
+        if (packet.original_length > 0) {
+          packet_length = packet.original_length;
+        }
+        rbe->IncomingPacket(clock.TimeInMilliseconds(),
+                            packet_length - header.headerLength, header, true);
+        ++packet_counter;
       }
-      rbe->IncomingPacket(clock.TimeInMilliseconds(),
-                          packet_length - header.headerLength,
-                          header);
-      ++packet_counter;
       if (!rtp_reader->NextPacket(&packet)) {
         break;
       }
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index f2ff7dfb85..bf698728e8 100644
--- a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -14,22 +14,11 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 int main(int argc, char** argv) {
-  if (argc < 4) {
-    fprintf(stderr, "Usage: rtp_to_text <extension type> <extension id>"
-           " <input_file.rtp> [-t]\n");
-    fprintf(stderr, "<extension type> can either be:\n"
-           "  abs for absolute send time or\n"
-           "  tsoffset for timestamp offset.\n"
-           "<extension id> is the id associated with the extension.\n"
-           "  -t is an optional flag, if set only packet arrival time will be"
-           " output.\n");
-    return -1;
-  }
   webrtc::test::RtpFileReader* reader;
   webrtc::RtpHeaderParser* parser;
   if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser,
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
new file mode 100644
index 0000000000..5904594ac8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
+
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+
+namespace webrtc {
+
+const int64_t kNoTimestamp = -1;
+const int64_t kSendTimeHistoryWindowMs = 10000;
+const int64_t kBaseTimestampScaleFactor =
+    rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
+const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
+
+TransportFeedbackAdapter::TransportFeedbackAdapter(
+    RtcpBandwidthObserver* bandwidth_observer,
+    Clock* clock,
+    ProcessThread* process_thread)
+    : send_time_history_(clock, kSendTimeHistoryWindowMs),
+      rtcp_bandwidth_observer_(bandwidth_observer),
+      process_thread_(process_thread),
+      clock_(clock),
+      current_offset_ms_(kNoTimestamp),
+      last_timestamp_us_(kNoTimestamp) {}
+
+TransportFeedbackAdapter::~TransportFeedbackAdapter() {
+  if (bitrate_estimator_.get())
+    process_thread_->DeRegisterModule(bitrate_estimator_.get());
+}
+
+void TransportFeedbackAdapter::SetBitrateEstimator(
+    RemoteBitrateEstimator* rbe) {
+  if (bitrate_estimator_.get() != rbe) {
+    bitrate_estimator_.reset(rbe);
+    process_thread_->RegisterModule(rbe);
+  }
+}
+
+void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
+                                         size_t length,
+                                         bool was_paced) {
+  rtc::CritScope cs(&lock_);
+  send_time_history_.AddAndRemoveOld(sequence_number, length, was_paced);
+}
+
+void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
+                                            int64_t send_time_ms) {
+  rtc::CritScope cs(&lock_);
+  send_time_history_.OnSentPacket(sequence_number, send_time_ms);
+}
+
+void TransportFeedbackAdapter::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  int64_t timestamp_us = feedback.GetBaseTimeUs();
+  // Add timestamp deltas to a local time base selected on first packet arrival.
+  // This won't be the true time base, but makes it easier to manually inspect
+  // time stamps.
+  if (last_timestamp_us_ == kNoTimestamp) {
+    current_offset_ms_ = clock_->TimeInMilliseconds();
+  } else {
+    int64_t delta = timestamp_us - last_timestamp_us_;
+
+    // Detect and compensate for wrap-arounds in base time.
+    if (std::abs(delta - kBaseTimestampRangeSizeUs) < std::abs(delta)) {
+      delta -= kBaseTimestampRangeSizeUs;  // Wrap backwards.
+    } else if (std::abs(delta + kBaseTimestampRangeSizeUs) < std::abs(delta)) {
+      delta += kBaseTimestampRangeSizeUs;  // Wrap forwards.
+    }
+
+    current_offset_ms_ += delta / 1000;
+  }
+  last_timestamp_us_ = timestamp_us;
+
+  uint16_t sequence_number = feedback.GetBaseSequence();
+  std::vector<int64_t> delta_vec = feedback.GetReceiveDeltasUs();
+  auto delta_it = delta_vec.begin();
+  std::vector<PacketInfo> packet_feedback_vector;
+  packet_feedback_vector.reserve(delta_vec.size());
+
+  {
+    rtc::CritScope cs(&lock_);
+    size_t failed_lookups = 0;
+    int64_t offset_us = 0;
+    for (auto symbol : feedback.GetStatusVector()) {
+      if (symbol != rtcp::TransportFeedback::StatusSymbol::kNotReceived) {
+        RTC_DCHECK(delta_it != delta_vec.end());
+        offset_us += *(delta_it++);
+        int64_t timestamp_ms = current_offset_ms_ + (offset_us / 1000);
+        PacketInfo info(timestamp_ms, sequence_number);
+        if (send_time_history_.GetInfo(&info, true) && info.send_time_ms >= 0) {
+          packet_feedback_vector.push_back(info);
+        } else {
+          ++failed_lookups;
+        }
+      }
+      ++sequence_number;
+    }
+    RTC_DCHECK(delta_it == delta_vec.end());
+    if (failed_lookups > 0) {
+      LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
+                      << " packet" << (failed_lookups > 1 ? "s" : "")
+                      << ". Send time history too small?";
+    }
+  }
+
+  RTC_DCHECK(bitrate_estimator_.get() != nullptr);
+  bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector);
+}
+
+void TransportFeedbackAdapter::OnReceiveBitrateChanged(
+    const std::vector<unsigned int>& ssrcs,
+    unsigned int bitrate) {
+  rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
+}
+
+void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
+                                           int64_t max_rtt_ms) {
+  RTC_DCHECK(bitrate_estimator_.get() != nullptr);
+  bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
new file mode 100644
index 0000000000..93f30e6cee
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TRANSPORT_FEEDBACK_ADAPTER_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TRANSPORT_FEEDBACK_ADAPTER_H_
+
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
+
+namespace webrtc {
+
+class ProcessThread;
+
+class TransportFeedbackAdapter : public TransportFeedbackObserver,
+                                 public CallStatsObserver,
+                                 public RemoteBitrateObserver {
+ public:
+  TransportFeedbackAdapter(RtcpBandwidthObserver* bandwidth_observer,
+                           Clock* clock,
+                           ProcessThread* process_thread);
+  virtual ~TransportFeedbackAdapter();
+
+  void AddPacket(uint16_t sequence_number,
+                 size_t length,
+                 bool was_paced) override;
+
+  void OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
+
+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
+
+  void SetBitrateEstimator(RemoteBitrateEstimator* rbe);
+
+  RemoteBitrateEstimator* GetBitrateEstimator() const {
+    return bitrate_estimator_.get();
+  }
+
+ private:
+  void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
+                               unsigned int bitrate) override;
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+
+  rtc::CriticalSection lock_;
+  SendTimeHistory send_time_history_ GUARDED_BY(&lock_);
+  rtc::scoped_ptr<RtcpBandwidthObserver> rtcp_bandwidth_observer_;
+  rtc::scoped_ptr<RemoteBitrateEstimator> bitrate_estimator_;
+  ProcessThread* const process_thread_;
+  Clock* const clock_;
+  int64_t current_offset_ms_;
+  int64_t last_timestamp_us_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TRANSPORT_FEEDBACK_ADAPTER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
new file mode 100644
index 0000000000..64d0e55397
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
@@ -0,0 +1,325 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <limits>
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+namespace test {
+
+class TransportFeedbackAdapterTest : public ::testing::Test {
+ public:
+  TransportFeedbackAdapterTest()
+      : clock_(0),
+        bitrate_estimator_(nullptr),
+        receiver_estimated_bitrate_(0) {}
+
+  virtual ~TransportFeedbackAdapterTest() {}
+
+  virtual void SetUp() {
+    adapter_.reset(new TransportFeedbackAdapter(
+        new RtcpBandwidthObserverAdapter(this), &clock_, &process_thread_));
+
+    bitrate_estimator_ = new MockRemoteBitrateEstimator();
+    EXPECT_CALL(process_thread_, RegisterModule(bitrate_estimator_)).Times(1);
+    adapter_->SetBitrateEstimator(bitrate_estimator_);
+  }
+
+  virtual void TearDown() {
+    EXPECT_CALL(process_thread_, DeRegisterModule(bitrate_estimator_)).Times(1);
+    adapter_.reset();
+  }
+
+ protected:
+  // Proxy class used since TransportFeedbackAdapter will own the instance
+  // passed at construction.
+  class RtcpBandwidthObserverAdapter : public RtcpBandwidthObserver {
+   public:
+    explicit RtcpBandwidthObserverAdapter(TransportFeedbackAdapterTest* owner)
+        : owner_(owner) {}
+
+    void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
+      owner_->receiver_estimated_bitrate_ = bitrate;
+    }
+
+    void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                      int64_t rtt,
+                                      int64_t now_ms) override {
+      RTC_NOTREACHED();
+    }
+
+    TransportFeedbackAdapterTest* const owner_;
+  };
+
+  void OnReceivedEstimatedBitrate(uint32_t bitrate) {}
+
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                    int64_t rtt,
+                                    int64_t now_ms) {}
+
+  void ComparePacketVectors(const std::vector<PacketInfo>& truth,
+                            const std::vector<PacketInfo>& input) {
+    ASSERT_EQ(truth.size(), input.size());
+    size_t len = truth.size();
+    // truth contains the input data for the test, and input is what will be
+    // sent to the bandwidth estimator. truth.arrival_tims_ms is used to
+    // populate the transport feedback messages. As these times may be changed
+    // (because of resolution limits in the packets, and because of the time
+    // base adjustment performed by the TransportFeedbackAdapter at the first
+    // packet, the truth[x].arrival_time and input[x].arrival_time may not be
+    // equal. However, the difference must be the same for all x.
+    int64_t arrival_time_delta =
+        truth[0].arrival_time_ms - input[0].arrival_time_ms;
+    for (size_t i = 0; i < len; ++i) {
+      EXPECT_EQ(truth[i].arrival_time_ms,
+                input[i].arrival_time_ms + arrival_time_delta);
+      EXPECT_EQ(truth[i].send_time_ms, input[i].send_time_ms);
+      EXPECT_EQ(truth[i].sequence_number, input[i].sequence_number);
+      EXPECT_EQ(truth[i].payload_size, input[i].payload_size);
+      EXPECT_EQ(truth[i].was_paced, input[i].was_paced);
+    }
+  }
+
+  // Utility method, to reset arrival_time_ms before adding send time.
+  void OnSentPacket(PacketInfo info) {
+    info.arrival_time_ms = 0;
+    adapter_->AddPacket(info.sequence_number, info.payload_size,
+                        info.was_paced);
+    adapter_->OnSentPacket(info.sequence_number, info.send_time_ms);
+  }
+
+  SimulatedClock clock_;
+  MockProcessThread process_thread_;
+  MockRemoteBitrateEstimator* bitrate_estimator_;
+  rtc::scoped_ptr<TransportFeedbackAdapter> adapter_;
+
+  uint32_t receiver_estimated_bitrate_;
+};
+
+TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(100, 200, 0, 1500, true));
+  packets.push_back(PacketInfo(110, 210, 1, 1500, true));
+  packets.push_back(PacketInfo(120, 220, 2, 1500, true));
+  packets.push_back(PacketInfo(130, 230, 3, 1500, true));
+  packets.push_back(PacketInfo(140, 240, 4, 1500, true));
+
+  for (const PacketInfo& packet : packets)
+    OnSentPacket(packet);
+
+  rtcp::TransportFeedback feedback;
+  feedback.WithBase(packets[0].sequence_number,
+                    packets[0].arrival_time_ms * 1000);
+
+  for (const PacketInfo& packet : packets) {
+    EXPECT_TRUE(feedback.WithReceivedPacket(packet.sequence_number,
+                                            packet.arrival_time_ms * 1000));
+  }
+
+  feedback.Build();
+
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke(
+          [packets, this](const std::vector<PacketInfo>& feedback_vector) {
+            ComparePacketVectors(packets, feedback_vector);
+          }));
+  adapter_->OnTransportFeedback(feedback);
+}
+
+TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(100, 200, 0, 1500, true));
+  packets.push_back(PacketInfo(110, 210, 1, 1500, true));
+  packets.push_back(PacketInfo(120, 220, 2, 1500, true));
+  packets.push_back(PacketInfo(130, 230, 3, 1500, true));
+  packets.push_back(PacketInfo(140, 240, 4, 1500, true));
+
+  const uint16_t kSendSideDropBefore = 1;
+  const uint16_t kReceiveSideDropAfter = 3;
+
+  for (const PacketInfo& packet : packets) {
+    if (packet.sequence_number >= kSendSideDropBefore)
+      OnSentPacket(packet);
+  }
+
+  rtcp::TransportFeedback feedback;
+  feedback.WithBase(packets[0].sequence_number,
+                    packets[0].arrival_time_ms * 1000);
+
+  for (const PacketInfo& packet : packets) {
+    if (packet.sequence_number <= kReceiveSideDropAfter) {
+      EXPECT_TRUE(feedback.WithReceivedPacket(packet.sequence_number,
+                                              packet.arrival_time_ms * 1000));
+    }
+  }
+
+  feedback.Build();
+
+  std::vector<PacketInfo> expected_packets(
+      packets.begin() + kSendSideDropBefore,
+      packets.begin() + kReceiveSideDropAfter + 1);
+
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke([expected_packets,
+                        this](const std::vector<PacketInfo>& feedback_vector) {
+        ComparePacketVectors(expected_packets, feedback_vector);
+      }));
+  adapter_->OnTransportFeedback(feedback);
+}
+
+TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
+  int64_t kHighArrivalTimeMs = rtcp::TransportFeedback::kDeltaScaleFactor *
+                               static_cast<int64_t>(1 << 8) *
+                               static_cast<int64_t>((1 << 23) - 1) / 1000;
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(kHighArrivalTimeMs - 64, 200, 0, 1500, true));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs + 64, 210, 1, 1500, true));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs, 220, 2, 1500, true));
+
+  for (const PacketInfo& packet : packets)
+    OnSentPacket(packet);
+
+  for (size_t i = 0; i < packets.size(); ++i) {
+    rtc::scoped_ptr<rtcp::TransportFeedback> feedback(
+        new rtcp::TransportFeedback());
+    feedback->WithBase(packets[i].sequence_number,
+                       packets[i].arrival_time_ms * 1000);
+
+    EXPECT_TRUE(feedback->WithReceivedPacket(
+        packets[i].sequence_number, packets[i].arrival_time_ms * 1000));
+
+    rtc::scoped_ptr<rtcp::RawPacket> raw_packet = feedback->Build();
+    feedback = rtcp::TransportFeedback::ParseFrom(raw_packet->Buffer(),
+                                                  raw_packet->Length());
+
+    std::vector<PacketInfo> expected_packets;
+    expected_packets.push_back(packets[i]);
+
+    EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+        .Times(1)
+        .WillOnce(Invoke([expected_packets, this](
+            const std::vector<PacketInfo>& feedback_vector) {
+          ComparePacketVectors(expected_packets, feedback_vector);
+        }));
+    adapter_->OnTransportFeedback(*feedback.get());
+  }
+}
+
+TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
+  std::vector<PacketInfo> sent_packets;
+  const int64_t kSmallDeltaUs =
+      rtcp::TransportFeedback::kDeltaScaleFactor * ((1 << 8) - 1);
+  const int64_t kLargePositiveDeltaUs =
+      rtcp::TransportFeedback::kDeltaScaleFactor *
+      std::numeric_limits<int16_t>::max();
+  const int64_t kLargeNegativeDeltaUs =
+      rtcp::TransportFeedback::kDeltaScaleFactor *
+      std::numeric_limits<int16_t>::min();
+
+  PacketInfo info(100, 200, 0, 1500, true);
+  sent_packets.push_back(info);
+
+  info.send_time_ms += kSmallDeltaUs / 1000;
+  info.arrival_time_ms += kSmallDeltaUs / 1000;
+  ++info.sequence_number;
+  sent_packets.push_back(info);
+
+  info.send_time_ms += kLargePositiveDeltaUs / 1000;
+  info.arrival_time_ms += kLargePositiveDeltaUs / 1000;
+  ++info.sequence_number;
+  sent_packets.push_back(info);
+
+  info.send_time_ms += kLargeNegativeDeltaUs / 1000;
+  info.arrival_time_ms += kLargeNegativeDeltaUs / 1000;
+  ++info.sequence_number;
+  sent_packets.push_back(info);
+
+  // Too large, delta - will need two feedback messages.
+  info.send_time_ms += (kLargePositiveDeltaUs + 1000) / 1000;
+  info.arrival_time_ms += (kLargePositiveDeltaUs + 1000) / 1000;
+  ++info.sequence_number;
+
+  // Packets will be added to send history.
+  for (const PacketInfo& packet : sent_packets)
+    OnSentPacket(packet);
+  OnSentPacket(info);
+
+  // Create expected feedback and send into adapter.
+  rtc::scoped_ptr<rtcp::TransportFeedback> feedback(
+      new rtcp::TransportFeedback());
+  feedback->WithBase(sent_packets[0].sequence_number,
+                     sent_packets[0].arrival_time_ms * 1000);
+
+  for (const PacketInfo& packet : sent_packets) {
+    EXPECT_TRUE(feedback->WithReceivedPacket(packet.sequence_number,
+                                             packet.arrival_time_ms * 1000));
+  }
+  EXPECT_FALSE(feedback->WithReceivedPacket(info.sequence_number,
+                                            info.arrival_time_ms * 1000));
+
+  rtc::scoped_ptr<rtcp::RawPacket> raw_packet = feedback->Build();
+  feedback = rtcp::TransportFeedback::ParseFrom(raw_packet->Buffer(),
+                                                raw_packet->Length());
+
+  std::vector<PacketInfo> received_feedback;
+
+  EXPECT_TRUE(feedback.get() != nullptr);
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke([sent_packets, &received_feedback](
+          const std::vector<PacketInfo>& feedback_vector) {
+        EXPECT_EQ(sent_packets.size(), feedback_vector.size());
+        received_feedback = feedback_vector;
+      }));
+  adapter_->OnTransportFeedback(*feedback.get());
+
+  // Create a new feedback message and add the trailing item.
+  feedback.reset(new rtcp::TransportFeedback());
+  feedback->WithBase(info.sequence_number, info.arrival_time_ms * 1000);
+  EXPECT_TRUE(feedback->WithReceivedPacket(info.sequence_number,
+                                           info.arrival_time_ms * 1000));
+  raw_packet = feedback->Build();
+  feedback = rtcp::TransportFeedback::ParseFrom(raw_packet->Buffer(),
+                                                raw_packet->Length());
+
+  EXPECT_TRUE(feedback.get() != nullptr);
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke(
+          [&received_feedback](const std::vector<PacketInfo>& feedback_vector) {
+            EXPECT_EQ(1u, feedback_vector.size());
+            received_feedback.push_back(feedback_vector[0]);
+          }));
+  adapter_->OnTransportFeedback(*feedback.get());
+
+  sent_packets.push_back(info);
+
+  ComparePacketVectors(sent_packets, received_feedback);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/BUILD.gn b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/BUILD.gn
index 528637fe59..a3d3403172 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -10,29 +10,76 @@ import("../../build/webrtc.gni")
 
 source_set("rtp_rtcp") {
   sources = [
-    # Common
-    "interface/fec_receiver.h",
-    "interface/receive_statistics.h",
-    "interface/remote_ntp_time_estimator.h",
-    "interface/rtp_header_parser.h",
-    "interface/rtp_payload_registry.h",
-    "interface/rtp_receiver.h",
-    "interface/rtp_rtcp.h",
-    "interface/rtp_rtcp_defines.h",
+    "include/fec_receiver.h",
+    "include/receive_statistics.h",
+    "include/remote_ntp_time_estimator.h",
+    "include/rtp_header_parser.h",
+    "include/rtp_payload_registry.h",
+    "include/rtp_receiver.h",
+    "include/rtp_rtcp.h",
+    "include/rtp_rtcp_defines.h",
+    "mocks/mock_rtp_rtcp.h",
     "source/bitrate.cc",
     "source/bitrate.h",
     "source/byte_io.h",
+    "source/dtmf_queue.cc",
+    "source/dtmf_queue.h",
+    "source/fec_private_tables_bursty.h",
+    "source/fec_private_tables_random.h",
     "source/fec_receiver_impl.cc",
     "source/fec_receiver_impl.h",
+    "source/forward_error_correction.cc",
+    "source/forward_error_correction.h",
+    "source/forward_error_correction_internal.cc",
+    "source/forward_error_correction_internal.h",
+    "source/h264_bitstream_parser.cc",
+    "source/h264_bitstream_parser.h",
+    "source/h264_sps_parser.cc",
+    "source/h264_sps_parser.h",
+    "source/mock/mock_rtp_payload_strategy.h",
+    "source/packet_loss_stats.cc",
+    "source/packet_loss_stats.h",
+    "source/producer_fec.cc",
+    "source/producer_fec.h",
     "source/receive_statistics_impl.cc",
     "source/receive_statistics_impl.h",
     "source/remote_ntp_time_estimator.cc",
-    "source/rtp_header_parser.cc",
-    "source/rtp_rtcp_config.h",
-    "source/rtp_rtcp_impl.cc",
-    "source/rtp_rtcp_impl.h",
     "source/rtcp_packet.cc",
     "source/rtcp_packet.h",
+    "source/rtcp_packet/app.cc",
+    "source/rtcp_packet/app.h",
+    "source/rtcp_packet/bye.cc",
+    "source/rtcp_packet/bye.h",
+    "source/rtcp_packet/compound_packet.cc",
+    "source/rtcp_packet/compound_packet.h",
+    "source/rtcp_packet/dlrr.cc",
+    "source/rtcp_packet/dlrr.h",
+    "source/rtcp_packet/extended_jitter_report.cc",
+    "source/rtcp_packet/extended_jitter_report.h",
+    "source/rtcp_packet/nack.cc",
+    "source/rtcp_packet/nack.h",
+    "source/rtcp_packet/pli.cc",
+    "source/rtcp_packet/pli.h",
+    "source/rtcp_packet/psfb.cc",
+    "source/rtcp_packet/psfb.h",
+    "source/rtcp_packet/receiver_report.cc",
+    "source/rtcp_packet/receiver_report.h",
+    "source/rtcp_packet/report_block.cc",
+    "source/rtcp_packet/report_block.h",
+    "source/rtcp_packet/rrtr.cc",
+    "source/rtcp_packet/rrtr.h",
+    "source/rtcp_packet/rtpfb.cc",
+    "source/rtcp_packet/rtpfb.h",
+    "source/rtcp_packet/sli.cc",
+    "source/rtcp_packet/sli.h",
+    "source/rtcp_packet/tmmbn.cc",
+    "source/rtcp_packet/tmmbn.h",
+    "source/rtcp_packet/tmmbr.cc",
+    "source/rtcp_packet/tmmbr.h",
+    "source/rtcp_packet/transport_feedback.cc",
+    "source/rtcp_packet/transport_feedback.h",
+    "source/rtcp_packet/voip_metric.cc",
+    "source/rtcp_packet/voip_metric.h",
     "source/rtcp_receiver.cc",
     "source/rtcp_receiver.h",
     "source/rtcp_receiver_help.cc",
@@ -41,57 +88,48 @@ source_set("rtp_rtcp") {
     "source/rtcp_sender.h",
     "source/rtcp_utility.cc",
     "source/rtcp_utility.h",
+    "source/rtp_format.cc",
+    "source/rtp_format.h",
+    "source/rtp_format_h264.cc",
+    "source/rtp_format_h264.h",
+    "source/rtp_format_video_generic.cc",
+    "source/rtp_format_video_generic.h",
+    "source/rtp_format_vp8.cc",
+    "source/rtp_format_vp8.h",
+    "source/rtp_format_vp9.cc",
+    "source/rtp_format_vp9.h",
     "source/rtp_header_extension.cc",
     "source/rtp_header_extension.h",
+    "source/rtp_header_parser.cc",
+    "source/rtp_packet_history.cc",
+    "source/rtp_packet_history.h",
+    "source/rtp_payload_registry.cc",
+    "source/rtp_receiver_audio.cc",
+    "source/rtp_receiver_audio.h",
     "source/rtp_receiver_impl.cc",
     "source/rtp_receiver_impl.h",
+    "source/rtp_receiver_strategy.cc",
+    "source/rtp_receiver_strategy.h",
+    "source/rtp_receiver_video.cc",
+    "source/rtp_receiver_video.h",
+    "source/rtp_rtcp_config.h",
+    "source/rtp_rtcp_impl.cc",
+    "source/rtp_rtcp_impl.h",
     "source/rtp_sender.cc",
     "source/rtp_sender.h",
+    "source/rtp_sender_audio.cc",
+    "source/rtp_sender_audio.h",
+    "source/rtp_sender_video.cc",
+    "source/rtp_sender_video.h",
     "source/rtp_utility.cc",
     "source/rtp_utility.h",
     "source/ssrc_database.cc",
     "source/ssrc_database.h",
     "source/tmmbr_help.cc",
     "source/tmmbr_help.h",
-    # Audio Files
-    "source/dtmf_queue.cc",
-    "source/dtmf_queue.h",
-    "source/rtp_receiver_audio.cc",
-    "source/rtp_receiver_audio.h",
-    "source/rtp_sender_audio.cc",
-    "source/rtp_sender_audio.h",
-    # Video Files
-    "source/fec_private_tables_random.h",
-    "source/fec_private_tables_bursty.h",
-    "source/forward_error_correction.cc",
-    "source/forward_error_correction.h",
-    "source/forward_error_correction_internal.cc",
-    "source/forward_error_correction_internal.h",
-    "source/producer_fec.cc",
-    "source/producer_fec.h",
-    "source/rtp_packet_history.cc",
-    "source/rtp_packet_history.h",
-    "source/rtp_payload_registry.cc",
-    "source/rtp_receiver_strategy.cc",
-    "source/rtp_receiver_strategy.h",
-    "source/rtp_receiver_video.cc",
-    "source/rtp_receiver_video.h",
-    "source/rtp_sender_video.cc",
-    "source/rtp_sender_video.h",
     "source/video_codec_information.h",
-    "source/rtp_format.cc",
-    "source/rtp_format.h",
-    "source/rtp_format_h264.cc",
-    "source/rtp_format_h264.h",
-    "source/rtp_format_vp8.cc",
-    "source/rtp_format_vp8.h",
-    "source/rtp_format_video_generic.cc",
-    "source/rtp_format_video_generic.h",
     "source/vp8_partition_aggregator.cc",
     "source/vp8_partition_aggregator.h",
-    # Mocks
-    "mocks/mock_rtp_rtcp.h",
-    "source/mock/mock_rtp_payload_strategy.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -106,7 +144,6 @@ source_set("rtp_rtcp") {
   deps = [
     "../..:webrtc_common",
     "../../system_wrappers",
-    "../pacing",
     "../remote_bitrate_estimator",
   ]
 
@@ -114,6 +151,7 @@ source_set("rtp_rtcp") {
     cflags = [
       # TODO(jschuh): Bug 1348: fix this warning.
       "/wd4267",  # size_t to int truncations
+
       # TODO(kjellander): Bug 261: fix this warning.
       "/wd4373",  # virtual function override.
     ]
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/OWNERS b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/OWNERS
index 4b553124af..fd12dcea0c 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/OWNERS
@@ -3,4 +3,9 @@ henrik.lundin@webrtc.org
 mflodman@webrtc.org
 asapersson@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/fec_receiver.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/fec_receiver.h
index 3608165dab..65e85ad7a5 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/fec_receiver.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -43,4 +43,4 @@ class FecReceiver {
   virtual FecPacketCounter GetPacketCounter() const = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/receive_statistics.h
similarity index 90%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/receive_statistics.h
index f7de91fc4b..b4a7cd0de2 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/receive_statistics.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
 
 #include <map>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -35,9 +35,6 @@ class StreamStatistician {
 
   virtual uint32_t BitrateReceived() const = 0;
 
-  // Resets all statistics.
-  virtual void ResetStatistics() = 0;
-
   // Returns true if the packet with RTP header |header| is likely to be a
   // retransmitted packet, false otherwise.
   virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
@@ -102,4 +99,4 @@ class NullReceiveStatistics : public ReceiveStatistics {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
similarity index 82%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
index 63949f7619..56c6e48691 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 
 namespace webrtc {
 
@@ -43,9 +43,9 @@ class RemoteNtpTimeEstimator {
   rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
   RtcpList rtcp_list_;
   int64_t last_timing_log_ms_;
-  DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
index c7a0268ef0..2e30d898ec 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
 
 #include "webrtc/common_video/rotation.h"
 
@@ -51,4 +51,4 @@ inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) {
 }
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
index 2809996b25..329de32611 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
@@ -7,10 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -41,4 +41,4 @@ class RtpHeaderParser {
   virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
similarity index 70%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index bc1ba2be41..fae864107f 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -8,8 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#include <map>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -27,7 +29,7 @@ class RTPPayloadStrategy {
 
   virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                                    const uint32_t frequency,
-                                   const uint8_t channels,
+                                   const size_t channels,
                                    const uint32_t rate) const = 0;
 
   virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
@@ -37,7 +39,7 @@ class RTPPayloadStrategy {
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payloadType,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) const = 0;
 
   virtual int GetPayloadTypeFrequency(
@@ -52,14 +54,14 @@ class RTPPayloadStrategy {
 class RTPPayloadRegistry {
  public:
   // The registry takes ownership of the strategy.
-  RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
+  explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
   ~RTPPayloadRegistry();
 
   int32_t RegisterReceivePayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate,
       bool* created_new_payload_type);
 
@@ -69,7 +71,7 @@ class RTPPayloadRegistry {
   int32_t ReceivePayloadType(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate,
       int8_t* payload_type) const;
 
@@ -79,16 +81,25 @@ class RTPPayloadRegistry {
 
   bool GetRtxSsrc(uint32_t* ssrc) const;
 
-  void SetRtxPayloadType(int payload_type);
+  void SetRtxPayloadType(int payload_type, int associated_payload_type);
 
   bool IsRtx(const RTPHeader& header) const;
 
+  // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t*
+  // restored_packet, instead of a uint8_t**.
+  // TODO(noahric): Remove this when all callers have been updated.
   bool RestoreOriginalPacket(uint8_t** restored_packet,
                              const uint8_t* packet,
                              size_t* packet_length,
                              uint32_t original_ssrc,
                              const RTPHeader& header) const;
 
+  bool RestoreOriginalPacket(uint8_t* restored_packet,
+                             const uint8_t* packet,
+                             size_t* packet_length,
+                             uint32_t original_ssrc,
+                             const RTPHeader& header) const;
+
   bool IsRed(const RTPHeader& header) const;
 
   // Returns true if the media of this RTP packet is encapsulated within an
@@ -99,8 +110,16 @@ class RTPPayloadRegistry {
 
   int GetPayloadTypeFrequency(uint8_t payload_type) const;
 
+  // DEPRECATED. Use PayloadTypeToPayload below that returns const Payload*
+  // instead of taking output parameter.
+  // TODO(danilchap): Remove this when all callers have been updated.
   bool PayloadTypeToPayload(const uint8_t payload_type,
-                            RtpUtility::Payload*& payload) const;
+                            RtpUtility::Payload*& payload) const {  // NOLINT
+    payload =
+        const_cast<RtpUtility::Payload*>(PayloadTypeToPayload(payload_type));
+    return payload != nullptr;
+  }
+  const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
 
   void ResetLastReceivedPayloadTypes() {
     CriticalSectionScoped cs(crit_sect_.get());
@@ -136,7 +155,17 @@ class RTPPayloadRegistry {
   int8_t last_received_media_payload_type() const {
     CriticalSectionScoped cs(crit_sect_.get());
     return last_received_media_payload_type_;
-  };
+  }
+
+  bool use_rtx_payload_mapping_on_restore() const {
+    CriticalSectionScoped cs(crit_sect_.get());
+    return use_rtx_payload_mapping_on_restore_;
+  }
+
+  void set_use_rtx_payload_mapping_on_restore(bool val) {
+    CriticalSectionScoped cs(crit_sect_.get());
+    use_rtx_payload_mapping_on_restore_ = val;
+  }
 
  private:
   // Prunes the payload type map of the specific payload type, if it exists.
@@ -144,7 +173,7 @@ class RTPPayloadRegistry {
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const size_t payload_name_length,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate);
 
   bool IsRtxInternal(const RTPHeader& header) const;
@@ -158,10 +187,17 @@ class RTPPayloadRegistry {
   int8_t  last_received_payload_type_;
   int8_t  last_received_media_payload_type_;
   bool rtx_;
-  int8_t payload_type_rtx_;
+  // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that
+  // only understand one RTX PT is no longer needed.
+  int rtx_payload_type_;
+  // Mapping rtx_payload_type_map_[rtx] = associated.
+  std::map<int, int> rtx_payload_type_map_;
+  // When true, use rtx_payload_type_map_ when restoring RTX packets to get the
+  // correct payload type.
+  bool use_rtx_payload_mapping_on_restore_;
   uint32_t ssrc_rtx_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
index e383923c7f..241825bd5f 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -37,14 +37,14 @@ class RtpReceiver {
  public:
   // Creates a video-enabled RTP receiver.
   static RtpReceiver* CreateVideoReceiver(
-      int id, Clock* clock,
+      Clock* clock,
       RtpData* incoming_payload_callback,
       RtpFeedback* incoming_messages_callback,
       RTPPayloadRegistry* rtp_payload_registry);
 
   // Creates an audio-enabled RTP receiver.
   static RtpReceiver* CreateAudioReceiver(
-      int id, Clock* clock,
+      Clock* clock,
       RtpAudioFeedback* incoming_audio_feedback,
       RtpData* incoming_payload_callback,
       RtpFeedback* incoming_messages_callback,
@@ -61,7 +61,7 @@ class RtpReceiver {
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) = 0;
 
   // De-registers |payload_type| from the payload registry.
@@ -94,7 +94,7 @@ class RtpReceiver {
 
   // Returns the current remote CSRCs.
   virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
-
+ 
   virtual void GetRID(char rid[256]) const = 0;
 
   // Returns the current energy of the RTP stream received.
@@ -102,4 +102,4 @@ class RtpReceiver {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
rename to media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 01a515b0fd..df38f3b5db 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -8,21 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
 
+#include <set>
+#include <utility>
 #include <vector>
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 // Forward declarations.
-class PacedSender;
 class ReceiveStatistics;
 class RemoteBitrateEstimator;
 class RtpReceiver;
 class Transport;
+namespace rtcp {
+class TransportFeedback;
+}
 
 class RtpRtcp : public Module {
  public:
@@ -52,18 +56,20 @@ class RtpRtcp : public Module {
     *  paced_sender             - Spread any bursts of packets into smaller
     *                             bursts to minimize packet loss.
     */
-    int32_t id;
     bool audio;
+    bool receiver_only;
     Clock* clock;
     ReceiveStatistics* receive_statistics;
     Transport* outgoing_transport;
     RtcpIntraFrameObserver* intra_frame_callback;
     RtcpBandwidthObserver* bandwidth_callback;
+    TransportFeedbackObserver* transport_feedback_callback;
     RtcpRttStats* rtt_stats;
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
     RtpAudioFeedback* audio_messages;
     RemoteBitrateEstimator* remote_bitrate_estimator;
-    PacedSender* paced_sender;
+    RtpPacketSender* paced_sender;
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator;
     BitrateStatisticsObserver* send_bitrate_observer;
     FrameCountObserver* send_frame_count_observer;
     SendSideDelayObserver* send_side_delay_observer;
@@ -209,6 +215,11 @@ class RtpRtcp : public Module {
     *   configure SSRC, default is a random number
     */
     virtual void SetSSRC(uint32_t ssrc) = 0;
+ 
+    /*
+    *   Set RID value for the RID header extension or RTCP SDES
+    */
+    virtual int32_t SetRID(const char *rid) = 0;
 
     /*
     *   Set CSRC
@@ -217,11 +228,6 @@ class RtpRtcp : public Module {
     */
     virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
 
-    /*
-    *   Set RID value for the RID header extension or RTCP SDES
-    */
-    virtual int32_t SetRID(const char *rid) = 0;
-
     /*
     * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination
     * of values of the enumerator RtxMode.
@@ -240,7 +246,12 @@ class RtpRtcp : public Module {
 
     // Sets the payload type to use when sending RTX packets. Note that this
     // doesn't enable RTX, only the payload type is set.
-    virtual void SetRtxSendPayloadType(int payload_type) = 0;
+    virtual void SetRtxSendPayloadType(int payload_type,
+                                       int associated_payload_type) = 0;
+
+    // Gets the payload type pair of (RTX, associated) to use when sending RTX
+    // packets.
+    virtual std::pair<int, int> RtxSendPayloadType() const = 0;
 
     /*
     *   sends kRtcpByeCode when going from true to false
@@ -307,9 +318,6 @@ class RtpRtcp : public Module {
 
     virtual size_t TimeToSendPadding(size_t bytes) = 0;
 
-    virtual bool GetSendSideDelay(int* avg_send_delay_ms,
-                                  int* max_send_delay_ms) const = 0;
-
     // Called on generation of new statistics after an RTP send.
     virtual void RegisterSendChannelRtpStatisticsCallback(
         StreamDataCountersCallback* callback) = 0;
@@ -325,21 +333,21 @@ class RtpRtcp : public Module {
     /*
     *    Get RTCP status
     */
-    virtual RTCPMethod RTCP() const = 0;
+    virtual RtcpMode RTCP() const = 0;
 
     /*
     *   configure RTCP status i.e on(compound or non- compound)/off
     *
     *   method  - RTCP method to use
     */
-    virtual void SetRTCPStatus(RTCPMethod method) = 0;
+    virtual void SetRTCPStatus(RtcpMode method) = 0;
 
     /*
     *   Set RTCP CName (i.e unique identifier)
     *
     *   return -1 on failure else 0
     */
-    virtual int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0;
+    virtual int32_t SetCNAME(const char* c_name) = 0;
 
     /*
     *   Get remote CName
@@ -366,8 +374,7 @@ class RtpRtcp : public Module {
     *
     *   return -1 on failure else 0
     */
-    virtual int32_t AddMixedCNAME(uint32_t SSRC,
-                                  const char cName[RTCP_CNAME_SIZE]) = 0;
+    virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
 
     /*
     *   RemoveMixedCNAME
@@ -387,7 +394,7 @@ class RtpRtcp : public Module {
                         int64_t* minRTT,
                         int64_t* maxRTT) const = 0;
 
-    /*
+     /*
      *   Get time of last rr, as well as packets received remotely
      *   (derived from rr report + cached sender-side info).
      *
@@ -400,12 +407,20 @@ class RtpRtcp : public Module {
                                        uint64_t* octets_received) const = 0;
     /*
     *   Force a send of a RTCP packet
-    *   normal SR and RR are triggered via the process function
+    *   periodic SR and RR are triggered via the process function
     *
     *   return -1 on failure else 0
     */
-    virtual int32_t SendRTCP(
-        uint32_t rtcpPacketType = kRtcpReport) = 0;
+    virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
+
+    /*
+    *   Force a send of a RTCP packet with more than one packet type.
+    *   periodic SR and RR are triggered via the process function
+    *
+    *   return -1 on failure else 0
+    */
+    virtual int32_t SendCompoundRTCP(
+        const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
 
     /*
     *    Good state of RTP receiver inform sender
@@ -419,13 +434,6 @@ class RtpRtcp : public Module {
     */
     virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
 
-    /*
-    *   Reset RTP data counters for the sending side
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t ResetSendDataCountersRTP() = 0;
-
     /*
     *   Statistics of the amount of data sent
     *
@@ -442,6 +450,14 @@ class RtpRtcp : public Module {
         StreamDataCounters* rtp_counters,
         StreamDataCounters* rtx_counters) const = 0;
 
+    /*
+     *  Get packet loss statistics for the RTP stream.
+     */
+    virtual void GetRtpPacketLossStats(
+        bool outgoing,
+        uint32_t ssrc,
+        struct RtpPacketLossStats* loss_stats) const = 0;
+
     /*
     *   Get received RTCP sender info
     *
@@ -457,21 +473,6 @@ class RtpRtcp : public Module {
     virtual int32_t RemoteRTCPStat(
         std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
 
-    /*
-    *   Set received RTCP report block
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t AddRTCPReportBlock(uint32_t SSRC,
-                                       const RTCPReportBlock* receiveBlock) = 0;
-
-    /*
-    *   RemoveRTCPReportBlock
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoveRTCPReportBlock(uint32_t SSRC) = 0;
-
     /*
     *   (APP) Application specific data
     *
@@ -506,13 +507,6 @@ class RtpRtcp : public Module {
     virtual void SetREMBData(uint32_t bitrate,
                              const std::vector<uint32_t>& ssrcs) = 0;
 
-    /*
-    *   (IJ) Extended jitter report.
-    */
-    virtual bool IJ() const = 0;
-
-    virtual void SetIJStatus(bool enable) = 0;
-
     /*
     *   (TMMBR) Temporary Max Media Bit Rate
     */
@@ -565,6 +559,8 @@ class RtpRtcp : public Module {
         RtcpStatisticsCallback* callback) = 0;
     virtual RtcpStatisticsCallback*
         GetRtcpStatisticsCallback() = 0;
+    // BWE feedback packets.
+    virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
 
     /**************************************************************************
     *
@@ -601,9 +597,13 @@ class RtpRtcp : public Module {
     *
     *   return -1 on failure else 0
     */
-     virtual int32_t SendREDPayloadType(
-         int8_t& payloadType) const = 0;
-
+    // DEPRECATED. Use SendREDPayloadType below that takes output parameter
+    // by pointer instead of by reference.
+    // TODO(danilchap): Remove this when all callers have been updated.
+    int32_t SendREDPayloadType(int8_t& payloadType) const {  // NOLINT
+      return SendREDPayloadType(&payloadType);
+    }
+    virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 0;
      /*
      * Store the audio level in dBov for header-extension-for-audio-level-
      * indication.
@@ -627,22 +627,25 @@ class RtpRtcp : public Module {
 
     /*
     *   Turn on/off generic FEC
-    *
-    *   return -1 on failure else 0
     */
-    virtual int32_t SetGenericFECStatus(bool enable,
-                                        uint8_t payloadTypeRED,
-                                        uint8_t payloadTypeFEC) = 0;
+    virtual void SetGenericFECStatus(bool enable,
+                                     uint8_t payload_type_red,
+                                     uint8_t payload_type_fec) = 0;
 
     /*
     *   Get generic FEC setting
-    *
-    *   return -1 on failure else 0
     */
-    virtual int32_t GenericFECStatus(bool& enable,
-                                     uint8_t& payloadTypeRED,
-                                     uint8_t& payloadTypeFEC) = 0;
-
+    // DEPRECATED. Use GenericFECStatus below that takes output parameters
+    // by pointers instead of by references.
+    // TODO(danilchap): Remove this when all callers have been updated.
+    void GenericFECStatus(bool& enable,               // NOLINT
+                          uint8_t& payloadTypeRED,    // NOLINT
+                          uint8_t& payloadTypeFEC) {  // NOLINT
+      GenericFECStatus(&enable, &payloadTypeRED, &payloadTypeFEC);
+    }
+    virtual void GenericFECStatus(bool* enable,
+                                  uint8_t* payload_type_red,
+                                  uint8_t* payload_type_fec) = 0;
 
     virtual int32_t SetFecParameters(
         const FecProtectionParams* delta_params,
@@ -663,4 +666,4 @@ class RtpRtcp : public Module {
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
new file mode 100644
index 0000000000..cad979d6a2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -0,0 +1,417 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#include <stddef.h>
+#include <list>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/typedefs.h"
+
+#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500    // we assume ethernet
+#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
+#define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
+
+namespace webrtc {
+namespace rtcp {
+class TransportFeedback;
+}
+
+const int kVideoPayloadTypeFrequency = 90000;
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+struct AudioPayload {
+    uint32_t    frequency;
+    size_t      channels;
+    uint32_t    rate;
+};
+
+struct VideoPayload {
+    RtpVideoCodecTypes   videoCodecType;
+    uint32_t       maxRate;
+};
+
+union PayloadUnion {
+    AudioPayload Audio;
+    VideoPayload Video;
+};
+
+enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
+
+enum ProtectionType {
+  kUnprotectedPacket,
+  kProtectedPacket
+};
+
+enum StorageType {
+  kDontRetransmit,
+  kAllowRetransmission
+};
+
+enum RTPExtensionType {
+  kRtpExtensionNone,
+  kRtpExtensionTransmissionTimeOffset,
+  kRtpExtensionAudioLevel,
+  kRtpExtensionAbsoluteSendTime,
+  kRtpExtensionVideoRotation,
+  kRtpExtensionTransportSequenceNumber,
+  kRtpExtensionRtpStreamId,
+};
+
+enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
+
+// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
+enum RTCPPacketType : uint32_t {
+  kRtcpReport = 0x0001,
+  kRtcpSr = 0x0002,
+  kRtcpRr = 0x0004,
+  kRtcpSdes = 0x0008,
+  kRtcpBye = 0x0010,
+  kRtcpPli = 0x0020,
+  kRtcpNack = 0x0040,
+  kRtcpFir = 0x0080,
+  kRtcpTmmbr = 0x0100,
+  kRtcpTmmbn = 0x0200,
+  kRtcpSrReq = 0x0400,
+  kRtcpXrVoipMetric = 0x0800,
+  kRtcpApp = 0x1000,
+  kRtcpSli = 0x4000,
+  kRtcpRpsi = 0x8000,
+  kRtcpRemb = 0x10000,
+  kRtcpTransmissionTimeOffset = 0x20000,
+  kRtcpXrReceiverReferenceTime = 0x40000,
+  kRtcpXrDlrrReportBlock = 0x80000,
+  kRtcpTransportFeedback = 0x100000,
+};
+
+enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
+
+enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };
+
+enum NACKMethod { kNackOff = 0, kNackRtcp = 2 };
+
+enum RetransmissionMode : uint8_t {
+  kRetransmitOff = 0x0,
+  kRetransmitFECPackets = 0x1,
+  kRetransmitBaseLayer = 0x2,
+  kRetransmitHigherLayers = 0x4,
+  kRetransmitAllPackets = 0xFF
+};
+
+enum RtxMode {
+  kRtxOff                 = 0x0,
+  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
+  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
+                                  // instead of padding.
+};
+
+const size_t kRtxHeaderSize = 2;
+
+struct RTCPSenderInfo {
+    uint32_t NTPseconds;
+    uint32_t NTPfraction;
+    uint32_t RTPtimeStamp;
+    uint32_t sendPacketCount;
+    uint32_t sendOctetCount;
+};
+
+struct RTCPReportBlock {
+  RTCPReportBlock()
+      : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
+        extendedHighSeqNum(0), jitter(0), lastSR(0),
+        delaySinceLastSR(0) {}
+
+  RTCPReportBlock(uint32_t remote_ssrc,
+                  uint32_t source_ssrc,
+                  uint8_t fraction_lost,
+                  uint32_t cumulative_lost,
+                  uint32_t extended_high_sequence_number,
+                  uint32_t jitter,
+                  uint32_t last_sender_report,
+                  uint32_t delay_since_last_sender_report)
+      : remoteSSRC(remote_ssrc),
+        sourceSSRC(source_ssrc),
+        fractionLost(fraction_lost),
+        cumulativeLost(cumulative_lost),
+        extendedHighSeqNum(extended_high_sequence_number),
+        jitter(jitter),
+        lastSR(last_sender_report),
+        delaySinceLastSR(delay_since_last_sender_report) {}
+
+  // Fields as described by RFC 3550 6.4.2.
+  uint32_t remoteSSRC;  // SSRC of sender of this report.
+  uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
+  uint8_t fractionLost;
+  uint32_t cumulativeLost;  // 24 bits valid.
+  uint32_t extendedHighSeqNum;
+  uint32_t jitter;
+  uint32_t lastSR;
+  uint32_t delaySinceLastSR;
+};
+
+struct RtcpReceiveTimeInfo {
+  // Fields as described by RFC 3611 4.5.
+  uint32_t sourceSSRC;
+  uint32_t lastRR;
+  uint32_t delaySinceLastRR;
+};
+
+typedef std::list<RTCPReportBlock> ReportBlockList;
+
+struct RtpState {
+  RtpState()
+      : sequence_number(0),
+        start_timestamp(0),
+        timestamp(0),
+        capture_time_ms(-1),
+        last_timestamp_time_ms(-1),
+        media_has_been_sent(false) {}
+  uint16_t sequence_number;
+  uint32_t start_timestamp;
+  uint32_t timestamp;
+  int64_t capture_time_ms;
+  int64_t last_timestamp_time_ms;
+  bool media_has_been_sent;
+};
+
+class RtpData {
+ public:
+  virtual ~RtpData() {}
+
+  virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+                                        const size_t payloadSize,
+                                        const WebRtcRTPHeader* rtpHeader) = 0;
+
+  virtual bool OnRecoveredPacket(const uint8_t* packet,
+                                 size_t packet_length) = 0;
+};
+
+class RtpFeedback {
+ public:
+  virtual ~RtpFeedback() {}
+
+  // Receiving payload change or SSRC change. (return success!)
+  /*
+  *   channels    - number of channels in codec (1 = mono, 2 = stereo)
+  */
+  virtual int32_t OnInitializeDecoder(
+      const int8_t payloadType,
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int frequency,
+      const size_t channels,
+      const uint32_t rate) = 0;
+
+  virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
+
+  virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0;
+};
+
+class RtpAudioFeedback {
+ public:
+  virtual void OnPlayTelephoneEvent(const uint8_t event,
+                                    const uint16_t lengthMs,
+                                    const uint8_t volume) = 0;
+
+ protected:
+  virtual ~RtpAudioFeedback() {}
+};
+
+class RtcpIntraFrameObserver {
+ public:
+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
+
+  virtual void OnReceivedSLI(uint32_t ssrc,
+                             uint8_t picture_id) = 0;
+
+  virtual void OnReceivedRPSI(uint32_t ssrc,
+                              uint64_t picture_id) = 0;
+
+  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0;
+
+  virtual ~RtcpIntraFrameObserver() {}
+};
+
+class RtcpBandwidthObserver {
+ public:
+  // REMB or TMMBR
+  virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
+
+  virtual void OnReceivedRtcpReceiverReport(
+      const ReportBlockList& report_blocks,
+      int64_t rtt,
+      int64_t now_ms) = 0;
+
+  virtual ~RtcpBandwidthObserver() {}
+};
+
+struct PacketInfo {
+  PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
+      : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {}
+
+  PacketInfo(int64_t arrival_time_ms,
+             int64_t send_time_ms,
+             uint16_t sequence_number,
+             size_t payload_size,
+             bool was_paced)
+      : PacketInfo(-1,
+                   arrival_time_ms,
+                   send_time_ms,
+                   sequence_number,
+                   payload_size,
+                   was_paced) {}
+
+  PacketInfo(int64_t creation_time_ms,
+             int64_t arrival_time_ms,
+             int64_t send_time_ms,
+             uint16_t sequence_number,
+             size_t payload_size,
+             bool was_paced)
+      : creation_time_ms(creation_time_ms),
+        arrival_time_ms(arrival_time_ms),
+        send_time_ms(send_time_ms),
+        sequence_number(sequence_number),
+        payload_size(payload_size),
+        was_paced(was_paced) {}
+
+  // Time corresponding to when this object was created.
+  int64_t creation_time_ms;
+  // Time corresponding to when the packet was received. Timestamped with the
+  // receiver's clock.
+  int64_t arrival_time_ms;
+  // Time corresponding to when the packet was sent, timestamped with the
+  // sender's clock.
+  int64_t send_time_ms;
+  // Packet identifier, incremented with 1 for every packet generated by the
+  // sender.
+  uint16_t sequence_number;
+  // Size of the packet excluding RTP headers.
+  size_t payload_size;
+  // True if the packet was paced out by the pacer.
+  bool was_paced;
+};
+
+class TransportFeedbackObserver {
+ public:
+  TransportFeedbackObserver() {}
+  virtual ~TransportFeedbackObserver() {}
+
+  // Note: Transport-wide sequence number as sequence number. Arrival time
+  // must be set to 0.
+  virtual void AddPacket(uint16_t sequence_number,
+                         size_t length,
+                         bool was_paced) = 0;
+
+  virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
+};
+
+class RtcpRttStats {
+ public:
+  virtual void OnRttUpdate(int64_t rtt) = 0;
+
+  virtual int64_t LastProcessedRtt() const = 0;
+
+  virtual ~RtcpRttStats() {}
+};
+
+// Null object version of RtpFeedback.
+class NullRtpFeedback : public RtpFeedback {
+ public:
+  virtual ~NullRtpFeedback() {}
+
+  int32_t OnInitializeDecoder(const int8_t payloadType,
+                              const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                              const int frequency,
+                              const size_t channels,
+                              const uint32_t rate) override {
+    return 0;
+  }
+
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
+  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
+};
+
+// Null object version of RtpData.
+class NullRtpData : public RtpData {
+ public:
+  virtual ~NullRtpData() {}
+
+  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+                                const size_t payloadSize,
+                                const WebRtcRTPHeader* rtpHeader) override {
+    return 0;
+  }
+
+  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
+    return true;
+  }
+};
+
+// Null object version of RtpAudioFeedback.
+class NullRtpAudioFeedback : public RtpAudioFeedback {
+ public:
+  virtual ~NullRtpAudioFeedback() {}
+
+  void OnPlayTelephoneEvent(const uint8_t event,
+                            const uint16_t lengthMs,
+                            const uint8_t volume) override {}
+};
+
+// Statistics about packet loss for a single directional connection. All values
+// are totals since the connection initiated.
+struct RtpPacketLossStats {
+  // The number of packets lost in events where no adjacent packets were also
+  // lost.
+  uint64_t single_packet_loss_count;
+  // The number of events in which more than one adjacent packet was lost.
+  uint64_t multiple_packet_loss_event_count;
+  // The number of packets lost in events where more than one adjacent packet
+  // was lost.
+  uint64_t multiple_packet_loss_packet_count;
+};
+
+class RtpPacketSender {
+ public:
+  RtpPacketSender() {}
+  virtual ~RtpPacketSender() {}
+
+  enum Priority {
+    kHighPriority = 0,    // Pass through; will be sent immediately.
+    kNormalPriority = 2,  // Put in back of the line.
+    kLowPriority = 3,     // Put in back of the low priority line.
+  };
+  // Low priority packets are mixed with the normal priority packets
+  // while we are paused.
+
+  // Returns true if we send the packet now, else it will add the packet
+  // information to the queue and call TimeToSendPacket when it's time to send.
+  virtual void InsertPacket(Priority priority,
+                            uint32_t ssrc,
+                            uint16_t sequence_number,
+                            int64_t capture_time_ms,
+                            size_t bytes,
+                            bool retransmission) = 0;
+};
+
+class TransportSequenceNumberAllocator {
+ public:
+  TransportSequenceNumberAllocator() {}
+  virtual ~TransportSequenceNumberAllocator() {}
+
+  virtual uint16_t AllocateSequenceNumber() = 0;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
deleted file mode 100644
index daa9aa73b8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ /dev/null
@@ -1,358 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-
-#include <stddef.h>
-#include <list>
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/typedefs.h"
-
-#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
-#define IP_PACKET_SIZE 1500    // we assume ethernet
-#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
-#define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
-
-namespace webrtc {
-
-const int kVideoPayloadTypeFrequency = 90000;
-
-// Minimum RTP header size in bytes.
-const uint8_t kRtpHeaderSize = 12;
-
-struct AudioPayload
-{
-    uint32_t    frequency;
-    uint8_t     channels;
-    uint32_t    rate;
-};
-
-struct VideoPayload
-{
-    RtpVideoCodecTypes   videoCodecType;
-    uint32_t       maxRate;
-};
-
-union PayloadUnion
-{
-    AudioPayload Audio;
-    VideoPayload Video;
-};
-
-enum RTCPMethod
-{
-    kRtcpOff          = 0,
-    kRtcpCompound     = 1,
-    kRtcpNonCompound = 2
-};
-
-enum RTPAliveType
-{
-    kRtpDead   = 0,
-    kRtpNoRtp = 1,
-    kRtpAlive  = 2
-};
-
-enum ProtectionType {
-  kUnprotectedPacket,
-  kProtectedPacket
-};
-
-enum StorageType {
-  kDontStore,
-  kDontRetransmit,
-  kAllowRetransmission
-};
-
-enum RTPExtensionType {
-  kRtpExtensionNone,
-  kRtpExtensionTransmissionTimeOffset,
-  kRtpExtensionAudioLevel,
-  kRtpExtensionAbsoluteSendTime,
-  kRtpExtensionVideoRotation,
-  kRtpExtensionTransportSequenceNumber,
-  kRtpExtensionRtpStreamId,
-};
-
-enum RTCPAppSubTypes
-{
-    kAppSubtypeBwe     = 0x00
-};
-
-enum RTCPPacketType
-{
-    kRtcpReport         = 0x0001,
-    kRtcpSr             = 0x0002,
-    kRtcpRr             = 0x0004,
-    kRtcpBye            = 0x0008,
-    kRtcpPli            = 0x0010,
-    kRtcpNack           = 0x0020,
-    kRtcpFir            = 0x0040,
-    kRtcpTmmbr          = 0x0080,
-    kRtcpTmmbn          = 0x0100,
-    kRtcpSrReq          = 0x0200,
-    kRtcpXrVoipMetric   = 0x0400,
-    kRtcpApp            = 0x0800,
-    kRtcpSli            = 0x4000,
-    kRtcpRpsi           = 0x8000,
-    kRtcpRemb           = 0x10000,
-    kRtcpTransmissionTimeOffset = 0x20000,
-    kRtcpXrReceiverReferenceTime = 0x40000,
-    kRtcpXrDlrrReportBlock = 0x80000
-};
-
-enum KeyFrameRequestMethod
-{
-    kKeyFrameReqFirRtp    = 1,
-    kKeyFrameReqPliRtcp   = 2,
-    kKeyFrameReqFirRtcp   = 3
-};
-
-enum RtpRtcpPacketType
-{
-    kPacketRtp        = 0,
-    kPacketKeepAlive = 1
-};
-
-enum NACKMethod
-{
-    kNackOff      = 0,
-    kNackRtcp     = 2
-};
-
-enum RetransmissionMode {
-  kRetransmitOff          = 0x0,
-  kRetransmitFECPackets   = 0x1,
-  kRetransmitBaseLayer    = 0x2,
-  kRetransmitHigherLayers = 0x4,
-  kRetransmitAllPackets   = 0xFF
-};
-
-enum RtxMode {
-  kRtxOff                 = 0x0,
-  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
-  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
-                                  // instead of padding.
-};
-
-const size_t kRtxHeaderSize = 2;
-
-struct RTCPSenderInfo
-{
-    uint32_t NTPseconds;
-    uint32_t NTPfraction;
-    uint32_t RTPtimeStamp;
-    uint32_t sendPacketCount;
-    uint32_t sendOctetCount;
-};
-
-struct RTCPReportBlock {
-  RTCPReportBlock()
-      : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
-        extendedHighSeqNum(0), jitter(0), lastSR(0),
-        delaySinceLastSR(0) {}
-
-  RTCPReportBlock(uint32_t remote_ssrc,
-                  uint32_t source_ssrc,
-                  uint8_t fraction_lost,
-                  uint32_t cumulative_lost,
-                  uint32_t extended_high_sequence_number,
-                  uint32_t jitter,
-                  uint32_t last_sender_report,
-                  uint32_t delay_since_last_sender_report)
-      : remoteSSRC(remote_ssrc),
-        sourceSSRC(source_ssrc),
-        fractionLost(fraction_lost),
-        cumulativeLost(cumulative_lost),
-        extendedHighSeqNum(extended_high_sequence_number),
-        jitter(jitter),
-        lastSR(last_sender_report),
-        delaySinceLastSR(delay_since_last_sender_report) {}
-
-  // Fields as described by RFC 3550 6.4.2.
-  uint32_t remoteSSRC;  // SSRC of sender of this report.
-  uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
-  uint8_t fractionLost;
-  uint32_t cumulativeLost;  // 24 bits valid.
-  uint32_t extendedHighSeqNum;
-  uint32_t jitter;
-  uint32_t lastSR;
-  uint32_t delaySinceLastSR;
-};
-
-struct RtcpReceiveTimeInfo {
-  // Fields as described by RFC 3611 4.5.
-  uint32_t sourceSSRC;
-  uint32_t lastRR;
-  uint32_t delaySinceLastRR;
-};
-
-typedef std::list<RTCPReportBlock> ReportBlockList;
-
-struct RtpState {
-  RtpState()
-      : sequence_number(0),
-        start_timestamp(0),
-        timestamp(0),
-        capture_time_ms(-1),
-        last_timestamp_time_ms(-1),
-        media_has_been_sent(false) {}
-  uint16_t sequence_number;
-  uint32_t start_timestamp;
-  uint32_t timestamp;
-  int64_t capture_time_ms;
-  int64_t last_timestamp_time_ms;
-  bool media_has_been_sent;
-};
-
-class RtpData
-{
-public:
-    virtual ~RtpData() {}
-
-    virtual int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const WebRtcRTPHeader* rtpHeader) = 0;
-
-    virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                   size_t packet_length) = 0;
-};
-
-class RtpFeedback
-{
-public:
-    virtual ~RtpFeedback() {}
-
-    // Receiving payload change or SSRC change. (return success!)
-    /*
-    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
-    */
-    virtual int32_t OnInitializeDecoder(
-        const int32_t id,
-        const int8_t payloadType,
-        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-        const int frequency,
-        const uint8_t channels,
-        const uint32_t rate) = 0;
-
-    virtual void OnIncomingSSRCChanged( const int32_t id,
-                                        const uint32_t ssrc) = 0;
-
-    virtual void OnIncomingCSRCChanged( const int32_t id,
-                                        const uint32_t CSRC,
-                                        const bool added) = 0;
-
-    virtual void ResetStatistics(uint32_t ssrc) = 0;
-};
-
-class RtpAudioFeedback {
- public:
-
-  virtual void OnPlayTelephoneEvent(const int32_t id,
-                                    const uint8_t event,
-                                    const uint16_t lengthMs,
-                                    const uint8_t volume) = 0;
- protected:
-  virtual ~RtpAudioFeedback() {}
-};
-
-class RtcpIntraFrameObserver {
- public:
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
-
-  virtual void OnReceivedSLI(uint32_t ssrc,
-                             uint8_t picture_id) = 0;
-
-  virtual void OnReceivedRPSI(uint32_t ssrc,
-                              uint64_t picture_id) = 0;
-
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0;
-
-  virtual ~RtcpIntraFrameObserver() {}
-};
-
-class RtcpBandwidthObserver {
- public:
-  // REMB or TMMBR
-  virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
-
-  virtual void OnReceivedRtcpReceiverReport(
-      const ReportBlockList& report_blocks,
-      int64_t rtt,
-      int64_t now_ms) = 0;
-
-  virtual ~RtcpBandwidthObserver() {}
-};
-
-class RtcpRttStats {
- public:
-  virtual void OnRttUpdate(int64_t rtt) = 0;
-
-  virtual int64_t LastProcessedRtt() const = 0;
-
-  virtual ~RtcpRttStats() {};
-};
-
-// Null object version of RtpFeedback.
-class NullRtpFeedback : public RtpFeedback {
- public:
-  virtual ~NullRtpFeedback() {}
-
-  int32_t OnInitializeDecoder(const int32_t id,
-                              const int8_t payloadType,
-                              const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                              const int frequency,
-                              const uint8_t channels,
-                              const uint32_t rate) override {
-    return 0;
-  }
-
-  void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {}
-
-  void OnIncomingCSRCChanged(const int32_t id,
-                             const uint32_t CSRC,
-                             const bool added) override {}
-
-  void ResetStatistics(uint32_t ssrc) override {}
-};
-
-// Null object version of RtpData.
-class NullRtpData : public RtpData {
- public:
-  virtual ~NullRtpData() {}
-
-  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                const size_t payloadSize,
-                                const WebRtcRTPHeader* rtpHeader) override {
-    return 0;
-  }
-
-  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
-    return true;
-  }
-};
-
-// Null object version of RtpAudioFeedback.
-class NullRtpAudioFeedback : public RtpAudioFeedback {
- public:
-  virtual ~NullRtpAudioFeedback() {}
-
-  void OnPlayTelephoneEvent(const int32_t id,
-                            const uint8_t event,
-                            const uint16_t lengthMs,
-                            const uint8_t volume) override {}
-};
-
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 0dce70ad2a..30478aa071 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -11,11 +11,16 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
 
+#include <set>
+#include <utility>
+#include <vector>
+
 #include "testing/gmock/include/gmock/gmock.h"
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
 
@@ -97,8 +102,8 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(RtxSendStatus, int());
   MOCK_METHOD1(SetRtxSsrc,
       void(uint32_t));
-  MOCK_METHOD1(SetRtxSendPayloadType,
-      void(int));
+  MOCK_METHOD2(SetRtxSendPayloadType, void(int, int));
+  MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair<int, int>());
   MOCK_METHOD1(SetSendingStatus,
       int32_t(const bool sending));
   MOCK_CONST_METHOD0(Sending,
@@ -107,7 +112,10 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(SendingMedia,
       bool());
   MOCK_CONST_METHOD4(BitrateSent,
-      void(uint32_t* totalRate, uint32_t* videoRate, uint32_t* fecRate, uint32_t* nackRate));
+                     void(uint32_t* totalRate,
+                          uint32_t* videoRate,
+                          uint32_t* fecRate,
+                          uint32_t* nackRate));
   MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver*));
   MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver*(void));
   MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
@@ -126,14 +134,11 @@ class MockRtpRtcp : public RtpRtcp {
            bool retransmission));
   MOCK_METHOD1(TimeToSendPadding,
       size_t(size_t bytes));
-  MOCK_CONST_METHOD2(GetSendSideDelay,
-      bool(int* avg_send_delay_ms, int* max_send_delay_ms));
   MOCK_METHOD2(RegisterRtcpObservers,
       void(RtcpIntraFrameObserver* intraFrameCallback,
            RtcpBandwidthObserver* bandwidthCallback));
-  MOCK_CONST_METHOD0(RTCP,
-      RTCPMethod());
-  MOCK_METHOD1(SetRTCPStatus, void(const RTCPMethod method));
+  MOCK_CONST_METHOD0(RTCP, RtcpMode());
+  MOCK_METHOD1(SetRTCPStatus, void(const RtcpMode method));
   MOCK_METHOD1(SetCNAME,
       int32_t(const char cName[RTCP_CNAME_SIZE]));
   MOCK_CONST_METHOD2(RemoteCNAME,
@@ -156,28 +161,28 @@ class MockRtpRtcp : public RtpRtcp {
               int64_t* avgRTT,
               int64_t* minRTT,
               int64_t* maxRTT));
-  MOCK_METHOD1(SendRTCP,
-      int32_t(uint32_t rtcpPacketType));
+  MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packetType));
+  MOCK_METHOD1(SendCompoundRTCP,
+               int32_t(const std::set<RTCPPacketType>& packetTypes));
   MOCK_METHOD1(SendRTCPReferencePictureSelection,
       int32_t(const uint64_t pictureID));
   MOCK_METHOD1(SendRTCPSliceLossIndication,
       int32_t(const uint8_t pictureID));
-  MOCK_METHOD0(ResetSendDataCountersRTP,
-      int32_t());
   MOCK_CONST_METHOD2(DataCountersRTP,
       int32_t(size_t *bytesSent, uint32_t *packetsSent));
   MOCK_CONST_METHOD2(GetSendStreamDataCounters,
       void(StreamDataCounters*, StreamDataCounters*));
+  MOCK_CONST_METHOD3(GetRtpPacketLossStats,
+      void(bool, uint32_t, struct RtpPacketLossStats*));
   MOCK_METHOD1(RemoteRTCPStat,
       int32_t(RTCPSenderInfo* senderInfo));
   MOCK_CONST_METHOD1(RemoteRTCPStat,
       int32_t(std::vector<RTCPReportBlock>* receiveBlocks));
-  MOCK_METHOD2(AddRTCPReportBlock,
-      int32_t(const uint32_t SSRC, const RTCPReportBlock* receiveBlock));
-  MOCK_METHOD1(RemoveRTCPReportBlock,
-      int32_t(const uint32_t SSRC));
   MOCK_METHOD4(SetRTCPApplicationSpecificData,
-      int32_t(const uint8_t subType, const uint32_t name, const uint8_t* data, const uint16_t length));
+               int32_t(const uint8_t subType,
+                       const uint32_t name,
+                       const uint8_t* data,
+                       const uint16_t length));
   MOCK_METHOD1(SetRTCPVoIPMetrics,
       int32_t(const RTCPVoIPMetric* VoIPMetric));
   MOCK_METHOD1(SetRtcpXrRrtrStatus,
@@ -190,9 +195,6 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD2(SetREMBData,
                void(const uint32_t bitrate,
                     const std::vector<uint32_t>& ssrcs));
-  MOCK_CONST_METHOD0(IJ,
-      bool());
-  MOCK_METHOD1(SetIJStatus, void(const bool));
   MOCK_CONST_METHOD0(TMMBR,
       bool());
   MOCK_METHOD1(SetTMMBRStatus, void(const bool enable));
@@ -213,6 +215,7 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(StorePackets, bool());
   MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
   MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
+  MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
   MOCK_METHOD1(RegisterAudioCallback,
       int32_t(RtpAudioFeedback* messagesCallback));
   MOCK_METHOD1(SetAudioPacketSize,
@@ -221,22 +224,21 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level));
   MOCK_METHOD1(SetSendREDPayloadType,
       int32_t(const int8_t payloadType));
-  MOCK_CONST_METHOD1(SendREDPayloadType,
-      int32_t(int8_t& payloadType));
+  MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payloadType));
   MOCK_METHOD2(SetAudioLevelIndicationStatus,
       int32_t(const bool enable, const uint8_t ID));
-  MOCK_CONST_METHOD2(GetAudioLevelIndicationStatus,
-      int32_t(bool& enable, uint8_t& ID));
   MOCK_METHOD1(SetAudioLevel,
       int32_t(const uint8_t level_dBov));
   MOCK_METHOD1(SetTargetSendBitrate,
       void(uint32_t bitrate_bps));
   MOCK_METHOD3(SetGenericFECStatus,
-      int32_t(const bool enable,
-              const uint8_t payloadTypeRED,
-              const uint8_t payloadTypeFEC));
+               void(const bool enable,
+                    const uint8_t payload_type_red,
+                    const uint8_t payload_type_fec));
   MOCK_METHOD3(GenericFECStatus,
-      int32_t(bool& enable, uint8_t& payloadTypeRED, uint8_t& payloadTypeFEC));
+               void(bool* enable,
+                    uint8_t* payloadTypeRED,
+                    uint8_t* payloadTypeFEC));
   MOCK_METHOD2(SetFecParameters,
       int32_t(const FecProtectionParams* delta_params,
               const FecProtectionParams* key_params));
@@ -244,8 +246,6 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const KeyFrameRequestMethod method));
   MOCK_METHOD0(RequestKeyFrame,
       int32_t());
-  MOCK_CONST_METHOD3(Version,
-      int32_t(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position));
   MOCK_METHOD0(TimeUntilNextProcess,
         int64_t());
   MOCK_METHOD0(Process,
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index c4b9b3b43d..d340f746be 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -13,24 +13,25 @@
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/modules/modules.gyp:paced_sender',
         '<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
       ],
       'sources': [
         # Common
-        'interface/fec_receiver.h',
-        'interface/receive_statistics.h',
-        'interface/remote_ntp_time_estimator.h',
-        'interface/rtp_header_parser.h',
-        'interface/rtp_payload_registry.h',
-        'interface/rtp_receiver.h',
-        'interface/rtp_rtcp.h',
-        'interface/rtp_rtcp_defines.h',
+        'include/fec_receiver.h',
+        'include/receive_statistics.h',
+        'include/remote_ntp_time_estimator.h',
+        'include/rtp_header_parser.h',
+        'include/rtp_payload_registry.h',
+        'include/rtp_receiver.h',
+        'include/rtp_rtcp.h',
+        'include/rtp_rtcp_defines.h',
         'source/bitrate.cc',
         'source/bitrate.h',
         'source/byte_io.h',
         'source/fec_receiver_impl.cc',
         'source/fec_receiver_impl.h',
+        'source/packet_loss_stats.cc',
+        'source/packet_loss_stats.h',
         'source/receive_statistics_impl.cc',
         'source/receive_statistics_impl.h',
         'source/remote_ntp_time_estimator.cc',
@@ -40,6 +41,40 @@
         'source/rtp_rtcp_impl.h',
         'source/rtcp_packet.cc',
         'source/rtcp_packet.h',
+        'source/rtcp_packet/app.cc',
+        'source/rtcp_packet/app.h',
+        'source/rtcp_packet/bye.cc',
+        'source/rtcp_packet/bye.h',
+        'source/rtcp_packet/compound_packet.cc',
+        'source/rtcp_packet/compound_packet.h',
+        'source/rtcp_packet/dlrr.cc',
+        'source/rtcp_packet/dlrr.h',
+        'source/rtcp_packet/extended_jitter_report.cc',
+        'source/rtcp_packet/extended_jitter_report.h',
+        'source/rtcp_packet/nack.cc',
+        'source/rtcp_packet/nack.h',
+        'source/rtcp_packet/pli.cc',
+        'source/rtcp_packet/pli.h',
+        'source/rtcp_packet/psfb.cc',
+        'source/rtcp_packet/psfb.h',
+        'source/rtcp_packet/receiver_report.cc',
+        'source/rtcp_packet/receiver_report.h',
+        'source/rtcp_packet/report_block.cc',
+        'source/rtcp_packet/report_block.h',
+        'source/rtcp_packet/rrtr.cc',
+        'source/rtcp_packet/rrtr.h',
+        'source/rtcp_packet/rtpfb.cc',
+        'source/rtcp_packet/rtpfb.h',
+        'source/rtcp_packet/sli.cc',
+        'source/rtcp_packet/sli.h',
+        'source/rtcp_packet/tmmbn.cc',
+        'source/rtcp_packet/tmmbn.h',
+        'source/rtcp_packet/tmmbr.cc',
+        'source/rtcp_packet/tmmbr.h',
+        'source/rtcp_packet/transport_feedback.cc',
+        'source/rtcp_packet/transport_feedback.h',
+        'source/rtcp_packet/voip_metric.cc',
+        'source/rtcp_packet/voip_metric.h',
         'source/rtcp_receiver.cc',
         'source/rtcp_receiver.h',
         'source/rtcp_receiver_help.cc',
@@ -74,6 +109,10 @@
         'source/forward_error_correction.h',
         'source/forward_error_correction_internal.cc',
         'source/forward_error_correction_internal.h',
+        'source/h264_bitstream_parser.cc',
+        'source/h264_bitstream_parser.h',
+        'source/h264_sps_parser.cc',
+        'source/h264_sps_parser.h',
         'source/producer_fec.cc',
         'source/producer_fec.h',
         'source/rtp_packet_history.cc',
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
new file mode 100644
index 0000000000..c318452482
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/CPPLINT.cfg
@@ -0,0 +1,6 @@
+#tmmbr_help is refactored in CL#1474693002
+exclude_files=tmmbr_help.*
+#rtcp_utility planned to be removed when webrtc:5260 will be finished.
+exclude_files=rtcp_utility.*
+#rtcp_receiver/rtcp_receiver_help will be refactored more deeply as part of webrtc:5260
+exclude_files=rtcp_receiver.*
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/bitrate.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/bitrate.cc
index 0d502213aa..4e9fc72c1f 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/bitrate.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/bitrate.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/byte_io.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/byte_io.h
index 2617806dd9..c69c178078 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/byte_io.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/byte_io.h
@@ -42,40 +42,107 @@
 
 namespace webrtc {
 
+// According to ISO C standard ISO/IEC 9899, section 6.2.6.2 (2), the three
+// representations of signed integers allowed are two's complement, one's
+// complement and sign/magnitude. We can detect which is used by looking at
+// the two last bits of -1, which will be 11 in two's complement, 10 in one's
+// complement and 01 in sign/magnitude.
+// TODO(sprang): In the unlikely event that we actually need to support a
+// platform that doesn't use two's complement, implement conversion to/from
+// wire format.
+
+// Assume the if any one signed integer type is two's complement, then all
+// other will be too.
+static_assert(
+    (-1 & 0x03) == 0x03,
+    "Only two's complement representation of signed integers supported.");
+
+// Plain const char* won't work for static_assert, use #define instead.
+#define kSizeErrorMsg "Byte size must be less than or equal to data type size."
+
+// Utility class for getting the unsigned equivalent of a signed type.
+template <typename T>
+struct UnsignedOf;
+
 // Class for reading integers from a sequence of bytes.
-// T = type of integer, B = bytes to read, is_signed = true if signed integer
-// If is_signed is true and B < sizeof(T), sign extension might be needed
-template<typename T, unsigned int B = sizeof(T),
-    bool is_signed = std::numeric_limits<T>::is_signed>
-class ByteReader {
+// T = type of integer, B = bytes to read, is_signed = true if signed integer.
+// If is_signed is true and B < sizeof(T), sign extension might be needed.
+template <typename T,
+          unsigned int B = sizeof(T),
+          bool is_signed = std::numeric_limits<T>::is_signed>
+class ByteReader;
+
+// Specialization of ByteReader for unsigned types.
+template <typename T, unsigned int B>
+class ByteReader<T, B, false> {
  public:
   static T ReadBigEndian(const uint8_t* data) {
-    if (is_signed && B < sizeof(T)) {
-      return SignExtend(InternalReadBigEndian(data));
-    }
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
     return InternalReadBigEndian(data);
   }
 
   static T ReadLittleEndian(const uint8_t* data) {
-    if (is_signed && B < sizeof(T)) {
-      return SignExtend(InternalReadLittleEndian(data));
-    }
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
     return InternalReadLittleEndian(data);
   }
 
  private:
   static T InternalReadBigEndian(const uint8_t* data) {
     T val(0);
-    for (unsigned int i = 0; i < B; ++i) {
+    for (unsigned int i = 0; i < B; ++i)
       val |= static_cast<T>(data[i]) << ((B - 1 - i) * 8);
-    }
     return val;
   }
 
   static T InternalReadLittleEndian(const uint8_t* data) {
     T val(0);
-    for (unsigned int i = 0; i < B; ++i) {
+    for (unsigned int i = 0; i < B; ++i)
       val |= static_cast<T>(data[i]) << (i * 8);
+    return val;
+  }
+};
+
+// Specialization of ByteReader for signed types.
+template <typename T, unsigned int B>
+class ByteReader<T, B, true> {
+ public:
+  typedef typename UnsignedOf<T>::Type U;
+
+  static T ReadBigEndian(const uint8_t* data) {
+    U unsigned_val = ByteReader<T, B, false>::ReadBigEndian(data);
+    if (B < sizeof(T))
+      unsigned_val = SignExtend(unsigned_val);
+    return ReinterpretAsSigned(unsigned_val);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    U unsigned_val = ByteReader<T, B, false>::ReadLittleEndian(data);
+    if (B < sizeof(T))
+      unsigned_val = SignExtend(unsigned_val);
+    return ReinterpretAsSigned(unsigned_val);
+  }
+
+ private:
+  // As a hack to avoid implementation-specific or undefined behavior when
+  // bit-shifting or casting signed integers, read as a signed equivalent
+  // instead and convert to signed. This is safe since we have asserted that
+  // two's complement for is used.
+  static T ReinterpretAsSigned(U unsigned_val) {
+    // An unsigned value with only the highest order bit set (ex 0x80).
+    const U kUnsignedHighestBitMask =
+        static_cast<U>(1) << ((sizeof(U) * 8) - 1);
+    // A signed value with only the highest bit set. Since this is two's
+    // complement form, we can use the min value from std::numeric_limits.
+    const T kSignedHighestBitMask = std::numeric_limits<T>::min();
+
+    T val;
+    if ((unsigned_val & kUnsignedHighestBitMask) != 0) {
+      // Casting is only safe when unsigned value can be represented in the
+      // signed target type, so mask out highest bit and mask it back manually.
+      val = static_cast<T>(unsigned_val & ~kUnsignedHighestBitMask);
+      val |= kSignedHighestBitMask;
+    } else {
+      val = static_cast<T>(unsigned_val);
     }
     return val;
   }
@@ -85,16 +152,16 @@ class ByteReader {
   // extend the remaining byte(s) with ones so that the correct negative
   // number is retained.
   // Ex: 0x810A0B -> 0xFF810A0B, but 0x710A0B -> 0x00710A0B
-  static T SignExtend(const T val) {
-    uint8_t msb = static_cast<uint8_t>(val >> ((B - 1) * 8));
-    if (msb & 0x80) {
-      // Sign extension is -1 (all ones) shifted left B bytes.
-      // The "B % sizeof(T)"-part is there to avoid compiler warning for
-      // shifting the whole size of the data type.
-      T sign_extend = (sizeof(T) == B ? 0 :
-          (static_cast<T>(-1L) << ((B % sizeof(T)) * 8)));
-
-      return val | sign_extend;
+  static U SignExtend(const U val) {
+    const uint8_t kMsb = static_cast<uint8_t>(val >> ((B - 1) * 8));
+    if ((kMsb & 0x80) != 0) {
+      // Create a mask where all bits used by the B bytes are set to one,
+      // for instance 0x00FFFFFF for B = 3. Bit-wise invert that mask (to
+      // (0xFF000000 in the example above) and add it to the input value.
+      // The "B % sizeof(T)" is a workaround to undefined values warnings for
+      // B == sizeof(T), in which case this code won't be called anyway.
+      const U kUsedBitsMask = (1 << ((B % sizeof(T)) * 8)) - 1;
+      return ~kUsedBitsMask | val;
     }
     return val;
   }
@@ -102,71 +169,162 @@ class ByteReader {
 
 // Class for writing integers to a sequence of bytes
 // T = type of integer, B = bytes to write
-template<typename T, unsigned int B = sizeof(T)>
-class ByteWriter {
+template <typename T,
+          unsigned int B = sizeof(T),
+          bool is_signed = std::numeric_limits<T>::is_signed>
+class ByteWriter;
+
+// Specialization of ByteWriter for unsigned types.
+template <typename T, unsigned int B>
+class ByteWriter<T, B, false> {
  public:
   static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
     for (unsigned int i = 0; i < B; ++i) {
       data[i] = val >> ((B - 1 - i) * 8);
     }
   }
 
   static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
     for (unsigned int i = 0; i < B; ++i) {
       data[i] = val >> (i * 8);
     }
   }
 };
 
+// Specialization of ByteWriter for signed types.
+template <typename T, unsigned int B>
+class ByteWriter<T, B, true> {
+ public:
+  typedef typename UnsignedOf<T>::Type U;
 
-// -------- Below follows specializations for B in { 2, 4, 8 } --------
+  static void WriteBigEndian(uint8_t* data, T val) {
+    ByteWriter<U, B, false>::WriteBigEndian(data, ReinterpretAsUnsigned(val));
+  }
 
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    ByteWriter<U, B, false>::WriteLittleEndian(data,
+                                               ReinterpretAsUnsigned(val));
+  }
 
-// Specializations for two byte words
-template<typename T, bool is_signed>
-class ByteReader<T, 2, is_signed> {
+ private:
+  static U ReinterpretAsUnsigned(T val) {
+    // According to ISO C standard ISO/IEC 9899, section 6.3.1.3 (1, 2) a
+    // conversion from signed to unsigned keeps the value if the new type can
+    // represent it, and otherwise adds one more than the max value of T until
+    // the value is in range. For two's complement, this fortunately means
+    // that the bit-wise value will be intact. Thus, since we have asserted that
+    // two's complement form is actually used, a simple cast is sufficient.
+    return static_cast<U>(val);
+  }
+};
+
+// ----- Below follows specializations of UnsignedOf utility class -----
+
+template <>
+struct UnsignedOf<int8_t> {
+  typedef uint8_t Type;
+};
+template <>
+struct UnsignedOf<int16_t> {
+  typedef uint16_t Type;
+};
+template <>
+struct UnsignedOf<int32_t> {
+  typedef uint32_t Type;
+};
+template <>
+struct UnsignedOf<int64_t> {
+  typedef uint64_t Type;
+};
+
+// ----- Below follows specializations for unsigned, B in { 1, 2, 4, 8 } -----
+
+// TODO(sprang): Check if these actually help or if generic cases will be
+// unrolled to and optimized to similar performance.
+
+// Specializations for single bytes
+template <typename T>
+class ByteReader<T, 1, false> {
  public:
   static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    return data[0];
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    return data[0];
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 1, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    data[0] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    data[0] = val;
+  }
+};
+
+// Specializations for two byte words
+template <typename T>
+class ByteReader<T, 2, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
     return (data[0] << 8) | data[1];
   }
 
   static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
     return data[0] | (data[1] << 8);
   }
 };
 
-template<typename T>
-class ByteWriter<T, 2> {
+template <typename T>
+class ByteWriter<T, 2, false> {
  public:
   static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
     data[0] = val >> 8;
     data[1] = val;
   }
 
   static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
     data[0] = val;
     data[1] = val >> 8;
   }
 };
 
 // Specializations for four byte words.
-template<typename T, bool is_signed>
-class ByteReader<T, 4, is_signed> {
+template <typename T>
+class ByteReader<T, 4, false> {
  public:
   static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
     return (data[0] << 24) | (data[1] << 16) | (data[2] << 8) | data[3];
   }
 
   static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
     return data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
   }
 };
 
 // Specializations for four byte words.
-template<typename T>
-class ByteWriter<T, 4> {
+template <typename T>
+class ByteWriter<T, 4, false> {
  public:
   static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
     data[0] = val >> 24;
     data[1] = val >> 16;
     data[2] = val >> 8;
@@ -174,6 +332,7 @@ class ByteWriter<T, 4> {
   }
 
   static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
     data[0] = val;
     data[1] = val >> 8;
     data[2] = val >> 16;
@@ -182,10 +341,11 @@ class ByteWriter<T, 4> {
 };
 
 // Specializations for eight byte words.
-template<typename T, bool is_signed>
-class ByteReader<T, 8, is_signed> {
+template <typename T>
+class ByteReader<T, 8, false> {
  public:
   static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
     return
         (Get(data, 0) << 56) | (Get(data, 1) << 48) |
         (Get(data, 2) << 40) | (Get(data, 3) << 32) |
@@ -194,6 +354,7 @@ class ByteReader<T, 8, is_signed> {
   }
 
   static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
     return
          Get(data, 0)        | (Get(data, 1) << 8)  |
         (Get(data, 2) << 16) | (Get(data, 3) << 24) |
@@ -207,10 +368,11 @@ class ByteReader<T, 8, is_signed> {
   }
 };
 
-template<typename T>
-class ByteWriter<T, 8> {
+template <typename T>
+class ByteWriter<T, 8, false> {
  public:
   static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
     data[0] = val >> 56;
     data[1] = val >> 48;
     data[2] = val >> 40;
@@ -222,6 +384,7 @@ class ByteWriter<T, 8> {
   }
 
   static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
     data[0] = val;
     data[1] = val >> 8;
     data[2] = val >> 16;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
index becea912ab..ab21b8704a 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 
-#include <string.h>  //memset
+#include <string.h>
 
 namespace webrtc {
 DTMFqueue::DTMFqueue()
@@ -21,7 +21,9 @@ DTMFqueue::DTMFqueue()
   memset(dtmf_level_, 0, sizeof(dtmf_level_));
 }
 
-DTMFqueue::~DTMFqueue() { delete dtmf_critsect_; }
+DTMFqueue::~DTMFqueue() {
+  delete dtmf_critsect_;
+}
 
 int32_t DTMFqueue::AddDTMF(uint8_t key, uint16_t len, uint8_t level) {
   CriticalSectionScoped lock(dtmf_critsect_);
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
index 320f8f57a9..d1b3f5667c 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
index 6105ae1d24..0b39908bb1 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
@@ -27,7 +27,8 @@
 
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace fec_private_tables {
 
 const uint8_t kMaskBursty1_1[2] = {
   0x80, 0x00
@@ -756,5 +757,6 @@ const uint8_t** kPacketMaskBurstyTbl[12] = {
   kPacketMaskBursty12
 };
 
-}  // namespace
+}  // namespace fec_private_tables
+}  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
index ff6de43b76..295d749873 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
 
 // This file contains a set of packets masks for the FEC code. The masks in
 // this table are specifically designed to favor recovery to random loss.
@@ -17,7 +17,8 @@
 
 #include "webrtc/typedefs.h"
 
-namespace {
+namespace webrtc {
+namespace fec_private_tables {
 
 const uint8_t kMaskRandom10_1[2] = {
   0xff, 0xc0
@@ -24518,5 +24519,6 @@ const uint8_t** kPacketMaskRandomTbl[48] = {
   kPacketMaskRandom48
 };
 
-}  // namespace
-#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
+}  // namespace fec_private_tables
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
index 9d9550b0b1..2109574e39 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
@@ -12,11 +12,11 @@
 
 #include <assert.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // RFC 5109
 namespace webrtc {
@@ -81,11 +81,16 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
   uint8_t REDHeaderLength = 1;
   size_t payload_data_length = packet_length - header.headerLength;
 
+  if (payload_data_length == 0) {
+    LOG(LS_WARNING) << "Corrupt/truncated FEC packet.";
+    return -1;
+  }
+
   // Add to list without RED header, aka a virtual RTP packet
   // we remove the RED header
 
-  ForwardErrorCorrection::ReceivedPacket* received_packet =
-      new ForwardErrorCorrection::ReceivedPacket;
+  rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
+      new ForwardErrorCorrection::ReceivedPacket);
   received_packet->pkt = new ForwardErrorCorrection::Packet;
 
   // get payload type from RED header
@@ -99,16 +104,18 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
   if (incoming_rtp_packet[header.headerLength] & 0x80) {
     // f bit set in RED header
     REDHeaderLength = 4;
+    if (payload_data_length < REDHeaderLength + 1u) {
+      LOG(LS_WARNING) << "Corrupt/truncated FEC packet.";
+      return -1;
+    }
+
     uint16_t timestamp_offset =
         (incoming_rtp_packet[header.headerLength + 1]) << 8;
     timestamp_offset +=
         incoming_rtp_packet[header.headerLength + 2];
     timestamp_offset = timestamp_offset >> 2;
     if (timestamp_offset != 0) {
-      // |timestampOffset| should be 0. However, it's possible this is the first
-      // location a corrupt payload can be caught, so don't assert.
       LOG(LS_WARNING) << "Corrupt payload found.";
-      delete received_packet;
       return -1;
     }
 
@@ -118,21 +125,20 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
 
     // check next RED header
     if (incoming_rtp_packet[header.headerLength + 4] & 0x80) {
-      // more than 2 blocks in packet not supported
-      delete received_packet;
-      assert(false);
+      LOG(LS_WARNING) << "More than 2 blocks in packet not supported.";
       return -1;
     }
-    if (blockLength > payload_data_length - REDHeaderLength) {
-      // block length longer than packet
-      delete received_packet;
-      assert(false);
+    // Check that the packet is long enough to contain data in the following
+    // block.
+    if (blockLength > payload_data_length - (REDHeaderLength + 1)) {
+      LOG(LS_WARNING) << "Block length longer than packet.";
       return -1;
     }
   }
   ++packet_counter_.num_packets;
 
-  ForwardErrorCorrection::ReceivedPacket* second_received_packet = NULL;
+  rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket>
+      second_received_packet;
   if (blockLength > 0) {
     // handle block length, split into 2 packets
     REDHeaderLength = 5;
@@ -154,7 +160,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
 
     received_packet->pkt->length = blockLength;
 
-    second_received_packet = new ForwardErrorCorrection::ReceivedPacket;
+    second_received_packet.reset(new ForwardErrorCorrection::ReceivedPacket);
     second_received_packet->pkt = new ForwardErrorCorrection::Packet;
 
     second_received_packet->is_fec = true;
@@ -202,14 +208,12 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
   }
 
   if (received_packet->pkt->length == 0) {
-    delete second_received_packet;
-    delete received_packet;
     return 0;
   }
 
-  received_packet_list_.push_back(received_packet);
+  received_packet_list_.push_back(received_packet.release());
   if (second_received_packet) {
-    received_packet_list_.push_back(second_received_packet);
+    received_packet_list_.push_back(second_received_packet.release());
   }
   return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 24db39b902..6a63813f40 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -14,8 +14,8 @@
 // This header is included to get the nested declaration of Packet structure.
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/typedefs.h"
 
@@ -25,7 +25,7 @@ class CriticalSectionWrapper;
 
 class FecReceiverImpl : public FecReceiver {
  public:
-  FecReceiverImpl(RtpData* callback);
+  explicit FecReceiverImpl(RtpData* callback);
   virtual ~FecReceiverImpl();
 
   int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index 31baf4e767..bb22e1d580 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -15,8 +15,10 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
@@ -81,6 +83,11 @@ class ReceiverFecTest : public ::testing::Test {
     delete red_packet;
   }
 
+  void InjectGarbagePacketLength(size_t fec_garbage_offset);
+  static void SurvivesMaliciousPacket(const uint8_t* data,
+                                      size_t length,
+                                      uint8_t ulpfec_payload_type);
+
   MockRtpData rtp_data_callback_;
   rtc::scoped_ptr<ForwardErrorCorrection> fec_;
   rtc::scoped_ptr<FecReceiver> receiver_fec_;
@@ -104,8 +111,7 @@ TEST_F(ReceiverFecTest, TwoMediaOneFec) {
 
   // Recovery
   std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
-  std::list<RtpPacket*>::iterator media_it = media_rtp_packets.begin();
-  BuildAndAddRedMediaPacket(*media_it);
+  BuildAndAddRedMediaPacket(*it);
   VerifyReconstructedMediaPacket(*it, 1);
   EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
   // Drop one media packet.
@@ -123,6 +129,44 @@ TEST_F(ReceiverFecTest, TwoMediaOneFec) {
   DeletePackets(&media_packets);
 }
 
+void ReceiverFecTest::InjectGarbagePacketLength(size_t fec_garbage_offset) {
+  EXPECT_CALL(rtp_data_callback_, OnRecoveredPacket(_, _))
+      .WillRepeatedly(Return(true));
+
+  const unsigned int kNumFecPackets = 1u;
+  std::list<RtpPacket*> media_rtp_packets;
+  std::list<Packet*> media_packets;
+  GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
+  std::list<Packet*> fec_packets;
+  GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &fec_packets.front()->data[fec_garbage_offset], 0x4711);
+
+  // Inject first media packet, then first FEC packet, skipping the second media
+  // packet to cause a recovery from the FEC packet.
+  BuildAndAddRedMediaPacket(media_rtp_packets.front());
+  BuildAndAddRedFecPacket(fec_packets.front());
+  EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
+
+  FecPacketCounter counter = receiver_fec_->GetPacketCounter();
+  EXPECT_EQ(2u, counter.num_packets);
+  EXPECT_EQ(1u, counter.num_fec_packets);
+  EXPECT_EQ(0u, counter.num_recovered_packets);
+
+  DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, InjectGarbageFecHeaderLengthRecovery) {
+  // Byte offset 8 is the 'length recovery' field of the FEC header.
+  InjectGarbagePacketLength(8);
+}
+
+TEST_F(ReceiverFecTest, InjectGarbageFecLevelHeaderProtectionLength) {
+  // Byte offset 10 is the 'protection length' field in the first FEC level
+  // header.
+  InjectGarbagePacketLength(10);
+}
+
 TEST_F(ReceiverFecTest, TwoMediaTwoFec) {
   const unsigned int kNumFecPackets = 2u;
   std::list<RtpPacket*> media_rtp_packets;
@@ -362,4 +406,132 @@ TEST_F(ReceiverFecTest, OldFecPacketDropped) {
   DeletePackets(&media_packets);
 }
 
+void ReceiverFecTest::SurvivesMaliciousPacket(const uint8_t* data,
+                                              size_t length,
+                                              uint8_t ulpfec_payload_type) {
+  webrtc::RTPHeader header;
+  rtc::scoped_ptr<webrtc::RtpHeaderParser> parser(
+      webrtc::RtpHeaderParser::Create());
+  ASSERT_TRUE(parser->Parse(data, length, &header));
+
+  webrtc::NullRtpData null_callback;
+  rtc::scoped_ptr<webrtc::FecReceiver> receiver_fec(
+      webrtc::FecReceiver::Create(&null_callback));
+
+  receiver_fec->AddReceivedRedPacket(header, data, length, ulpfec_payload_type);
+}
+
+TEST_F(ReceiverFecTest, TruncatedPacketWithFBitSet) {
+  const uint8_t kTruncatedPacket[] = {0x80,
+                                      0x2a,
+                                      0x68,
+                                      0x71,
+                                      0x29,
+                                      0xa1,
+                                      0x27,
+                                      0x3a,
+                                      0x29,
+                                      0x12,
+                                      0x2a,
+                                      0x98,
+                                      0xe0,
+                                      0x29};
+
+  SurvivesMaliciousPacket(kTruncatedPacket, sizeof(kTruncatedPacket), 100);
+}
+
+TEST_F(ReceiverFecTest, TruncatedPacketWithFBitSetEndingAfterFirstRedHeader) {
+  const uint8_t kPacket[] = {0x89,
+                             0x27,
+                             0x3a,
+                             0x83,
+                             0x27,
+                             0x3a,
+                             0x3a,
+                             0xf3,
+                             0x67,
+                             0xbe,
+                             0x2a,
+                             0xa9,
+                             0x27,
+                             0x54,
+                             0x3a,
+                             0x3a,
+                             0x2a,
+                             0x67,
+                             0x3a,
+                             0xf3,
+                             0x67,
+                             0xbe,
+                             0x2a,
+                             0x27,
+                             0xe6,
+                             0xf6,
+                             0x03,
+                             0x3e,
+                             0x29,
+                             0x27,
+                             0x21,
+                             0x27,
+                             0x2a,
+                             0x29,
+                             0x21,
+                             0x4b,
+                             0x29,
+                             0x3a,
+                             0x28,
+                             0x29,
+                             0xbf,
+                             0x29,
+                             0x2a,
+                             0x26,
+                             0x29,
+                             0xae,
+                             0x27,
+                             0xa6,
+                             0xf6,
+                             0x00,
+                             0x03,
+                             0x3e};
+  SurvivesMaliciousPacket(kPacket, sizeof(kPacket), 100);
+}
+
+TEST_F(ReceiverFecTest, TruncatedPacketWithoutDataPastFirstBlock) {
+  const uint8_t kPacket[] = {0x82,
+                             0x38,
+                             0x92,
+                             0x38,
+                             0x92,
+                             0x38,
+                             0xde,
+                             0x2a,
+                             0x11,
+                             0xc8,
+                             0xa3,
+                             0xc4,
+                             0x82,
+                             0x38,
+                             0x2a,
+                             0x21,
+                             0x2a,
+                             0x28,
+                             0x92,
+                             0x38,
+                             0x92,
+                             0x00,
+                             0x00,
+                             0x0a,
+                             0x3a,
+                             0xc8,
+                             0xa3,
+                             0x3a,
+                             0x27,
+                             0xc4,
+                             0x2a,
+                             0x21,
+                             0x2a,
+                             0x28};
+  SurvivesMaliciousPacket(kPacket, sizeof(kPacket), 100);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
index e1791adba3..aacc2d1ecc 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
 namespace webrtc {
@@ -54,6 +54,6 @@ class FrameGenerator {
   uint16_t seq_num_;
   uint32_t timestamp_;
 };
-}
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index abef1dda30..b85d813790 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -10,17 +10,17 @@
 
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
-#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
 #include <algorithm>
 #include <iterator>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
@@ -36,11 +36,11 @@ const uint8_t kUlpHeaderSizeLBitClear = (2 + kMaskSizeLBitClear);
 // Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
 const uint8_t kTransportOverhead = 28;
 
-enum {
-  kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets
-};
+enum { kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets };
 
-int32_t ForwardErrorCorrection::Packet::AddRef() { return ++ref_count_; }
+int32_t ForwardErrorCorrection::Packet::AddRef() {
+  return ++ref_count_;
+}
 
 int32_t ForwardErrorCorrection::Packet::Release() {
   int32_t ref_count;
@@ -55,7 +55,7 @@ int32_t ForwardErrorCorrection::Packet::Release() {
 // TODO(holmer): Refactor into a proper class.
 class ProtectedPacket : public ForwardErrorCorrection::SortablePacket {
  public:
-  scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+  rtc::scoped_refptr<ForwardErrorCorrection::Packet> pkt;
 };
 
 typedef std::list<ProtectedPacket*> ProtectedPacketList;
@@ -68,11 +68,12 @@ class FecPacket : public ForwardErrorCorrection::SortablePacket {
  public:
   ProtectedPacketList protected_pkt_list;
   uint32_t ssrc;  // SSRC of the current frame.
-  scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+  rtc::scoped_refptr<ForwardErrorCorrection::Packet> pkt;
 };
 
 bool ForwardErrorCorrection::SortablePacket::LessThan(
-    const SortablePacket* first, const SortablePacket* second) {
+    const SortablePacket* first,
+    const SortablePacket* second) {
   return IsNewerSequenceNumber(second->seq_num, first->seq_num);
 }
 
@@ -83,8 +84,7 @@ ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() {}
 ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() {}
 
 ForwardErrorCorrection::ForwardErrorCorrection()
-    : generated_fec_packets_(kMaxMediaPackets),
-      fec_packet_received_(false) {}
+    : generated_fec_packets_(kMaxMediaPackets), fec_packet_received_(false) {}
 
 ForwardErrorCorrection::~ForwardErrorCorrection() {}
 
@@ -112,7 +112,6 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
                                             FecMaskType fec_mask_type,
                                             PacketList* fec_packet_list) {
   const uint16_t num_media_packets = media_packet_list.size();
-
   // Sanity check arguments.
   assert(num_media_packets > 0);
   assert(num_important_packets >= 0 &&
@@ -126,12 +125,10 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
   }
 
   bool l_bit = (num_media_packets > 8 * kMaskSizeLBitClear);
-  int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+  int num_mask_bytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
   // Do some error checking on the media packets.
-  PacketList::const_iterator media_list_it = media_packet_list.begin();
-  while (media_list_it != media_packet_list.end()) {
-    Packet* media_packet = *media_list_it;
+  for (Packet* media_packet : media_packet_list) {
     assert(media_packet);
 
     if (media_packet->length < kRtpHeaderSize) {
@@ -146,7 +143,6 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
       LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
                       << "with overhead is larger than " << IP_PACKET_SIZE;
     }
-    media_list_it++;
   }
 
   int num_fec_packets =
@@ -167,29 +163,29 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
 
   // -- Generate packet masks --
   // Always allocate space for a large mask.
-  uint8_t* packet_mask = new uint8_t[num_fec_packets * kMaskSizeLBitSet];
-  memset(packet_mask, 0, num_fec_packets * num_maskBytes);
+  rtc::scoped_ptr<uint8_t[]> packet_mask(
+      new uint8_t[num_fec_packets * kMaskSizeLBitSet]);
+  memset(packet_mask.get(), 0, num_fec_packets * num_mask_bytes);
   internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
                                 num_important_packets, use_unequal_protection,
-                                mask_table, packet_mask);
+                                mask_table, packet_mask.get());
 
-  int num_maskBits = InsertZerosInBitMasks(media_packet_list, packet_mask,
-                                           num_maskBytes, num_fec_packets);
+  int num_mask_bits = InsertZerosInBitMasks(
+      media_packet_list, packet_mask.get(), num_mask_bytes, num_fec_packets);
 
-  l_bit = (num_maskBits > 8 * kMaskSizeLBitClear);
-
-  if (num_maskBits < 0) {
-    delete[] packet_mask;
+  if (num_mask_bits < 0) {
     return -1;
   }
+  l_bit = (num_mask_bits > 8 * kMaskSizeLBitClear);
   if (l_bit) {
-    num_maskBytes = kMaskSizeLBitSet;
+    num_mask_bytes = kMaskSizeLBitSet;
   }
 
-  GenerateFecBitStrings(media_packet_list, packet_mask, num_fec_packets, l_bit);
-  GenerateFecUlpHeaders(media_packet_list, packet_mask, l_bit, num_fec_packets);
+  GenerateFecBitStrings(media_packet_list, packet_mask.get(), num_fec_packets,
+                        l_bit);
+  GenerateFecUlpHeaders(media_packet_list, packet_mask.get(), l_bit,
+                        num_fec_packets);
 
-  delete[] packet_mask;
   return 0;
 }
 
@@ -206,26 +202,30 @@ int ForwardErrorCorrection::GetNumberOfFecPackets(int num_media_packets,
 }
 
 void ForwardErrorCorrection::GenerateFecBitStrings(
-    const PacketList& media_packet_list, uint8_t* packet_mask,
-    int num_fec_packets, bool l_bit) {
+    const PacketList& media_packet_list,
+    uint8_t* packet_mask,
+    int num_fec_packets,
+    bool l_bit) {
   if (media_packet_list.empty()) {
     return;
   }
   uint8_t media_payload_length[2];
-  const int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+  const int num_mask_bytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
   const uint16_t ulp_header_size =
       l_bit ? kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
   const uint16_t fec_rtp_offset =
       kFecHeaderSize + ulp_header_size - kRtpHeaderSize;
 
   for (int i = 0; i < num_fec_packets; ++i) {
+    Packet* const fec_packet = &generated_fec_packets_[i];
     PacketList::const_iterator media_list_it = media_packet_list.begin();
-    uint32_t pkt_mask_idx = i * num_maskBytes;
+    uint32_t pkt_mask_idx = i * num_mask_bytes;
     uint32_t media_pkt_idx = 0;
     uint16_t fec_packet_length = 0;
     uint16_t prev_seq_num = ParseSequenceNumber((*media_list_it)->data);
     while (media_list_it != media_packet_list.end()) {
-      // Each FEC packet has a multiple byte mask.
+      // Each FEC packet has a multiple byte mask. Determine if this media
+      // packet should be included in FEC packet i.
       if (packet_mask[pkt_mask_idx] & (1 << (7 - media_pkt_idx))) {
         Packet* media_packet = *media_list_it;
 
@@ -235,42 +235,40 @@ void ForwardErrorCorrection::GenerateFecBitStrings(
 
         fec_packet_length = media_packet->length + fec_rtp_offset;
         // On the first protected packet, we don't need to XOR.
-        if (generated_fec_packets_[i].length == 0) {
+        if (fec_packet->length == 0) {
           // Copy the first 2 bytes of the RTP header.
-          memcpy(generated_fec_packets_[i].data, media_packet->data, 2);
+          memcpy(fec_packet->data, media_packet->data, 2);
           // Copy the 5th to 8th bytes of the RTP header.
-          memcpy(&generated_fec_packets_[i].data[4], &media_packet->data[4], 4);
+          memcpy(&fec_packet->data[4], &media_packet->data[4], 4);
           // Copy network-ordered payload size.
-          memcpy(&generated_fec_packets_[i].data[8], media_payload_length, 2);
+          memcpy(&fec_packet->data[8], media_payload_length, 2);
 
           // Copy RTP payload, leaving room for the ULP header.
-          memcpy(
-              &generated_fec_packets_[i].data[kFecHeaderSize + ulp_header_size],
-              &media_packet->data[kRtpHeaderSize],
-              media_packet->length - kRtpHeaderSize);
+          memcpy(&fec_packet->data[kFecHeaderSize + ulp_header_size],
+                 &media_packet->data[kRtpHeaderSize],
+                 media_packet->length - kRtpHeaderSize);
         } else {
           // XOR with the first 2 bytes of the RTP header.
-          generated_fec_packets_[i].data[0] ^= media_packet->data[0];
-          generated_fec_packets_[i].data[1] ^= media_packet->data[1];
+          fec_packet->data[0] ^= media_packet->data[0];
+          fec_packet->data[1] ^= media_packet->data[1];
 
           // XOR with the 5th to 8th bytes of the RTP header.
           for (uint32_t j = 4; j < 8; ++j) {
-            generated_fec_packets_[i].data[j] ^= media_packet->data[j];
+            fec_packet->data[j] ^= media_packet->data[j];
           }
 
           // XOR with the network-ordered payload size.
-          generated_fec_packets_[i].data[8] ^= media_payload_length[0];
-          generated_fec_packets_[i].data[9] ^= media_payload_length[1];
+          fec_packet->data[8] ^= media_payload_length[0];
+          fec_packet->data[9] ^= media_payload_length[1];
 
           // XOR with RTP payload, leaving room for the ULP header.
           for (int32_t j = kFecHeaderSize + ulp_header_size;
                j < fec_packet_length; j++) {
-            generated_fec_packets_[i].data[j] ^=
-                media_packet->data[j - fec_rtp_offset];
+            fec_packet->data[j] ^= media_packet->data[j - fec_rtp_offset];
           }
         }
-        if (fec_packet_length > generated_fec_packets_[i].length) {
-          generated_fec_packets_[i].length = fec_packet_length;
+        if (fec_packet_length > fec_packet->length) {
+          fec_packet->length = fec_packet_length;
         }
       }
       media_list_it++;
@@ -279,19 +277,18 @@ void ForwardErrorCorrection::GenerateFecBitStrings(
         media_pkt_idx += static_cast<uint16_t>(seq_num - prev_seq_num);
         prev_seq_num = seq_num;
       }
-      if (media_pkt_idx == 8) {
-        // Switch to the next mask byte.
-        media_pkt_idx = 0;
-        pkt_mask_idx++;
-      }
+      pkt_mask_idx += media_pkt_idx / 8;
+      media_pkt_idx %= 8;
     }
-    assert(generated_fec_packets_[i].length);
-    //Note: This shouldn't happen: means packet mask is wrong or poorly designed
+    RTC_DCHECK_GT(fec_packet->length, 0u)
+        << "Packet mask is wrong or poorly designed.";
   }
 }
 
 int ForwardErrorCorrection::InsertZerosInBitMasks(
-    const PacketList& media_packets, uint8_t* packet_mask, int num_mask_bytes,
+    const PacketList& media_packets,
+    uint8_t* packet_mask,
+    int num_mask_bytes,
     int num_fec_packets) {
   uint8_t* new_mask = NULL;
   if (media_packets.size() <= 1) {
@@ -307,6 +304,9 @@ int ForwardErrorCorrection::InsertZerosInBitMasks(
     // required.
     return media_packets.size();
   }
+  // We can only protect 8 * kMaskSizeLBitSet packets.
+  if (total_missing_seq_nums + media_packets.size() > 8 * kMaskSizeLBitSet)
+    return -1;
   // Allocate the new mask.
   int new_mask_bytes = kMaskSizeLBitClear;
   if (media_packets.size() + total_missing_seq_nums > 8 * kMaskSizeLBitClear) {
@@ -357,7 +357,8 @@ int ForwardErrorCorrection::InsertZerosInBitMasks(
   return new_bit_index;
 }
 
-void ForwardErrorCorrection::InsertZeroColumns(int num_zeros, uint8_t* new_mask,
+void ForwardErrorCorrection::InsertZeroColumns(int num_zeros,
+                                               uint8_t* new_mask,
                                                int new_mask_bytes,
                                                int num_fec_packets,
                                                int new_bit_index) {
@@ -368,9 +369,12 @@ void ForwardErrorCorrection::InsertZeroColumns(int num_zeros, uint8_t* new_mask,
   }
 }
 
-void ForwardErrorCorrection::CopyColumn(uint8_t* new_mask, int new_mask_bytes,
-                                        uint8_t* old_mask, int old_mask_bytes,
-                                        int num_fec_packets, int new_bit_index,
+void ForwardErrorCorrection::CopyColumn(uint8_t* new_mask,
+                                        int new_mask_bytes,
+                                        uint8_t* old_mask,
+                                        int old_mask_bytes,
+                                        int num_fec_packets,
+                                        int new_bit_index,
                                         int old_bit_index) {
   // Copy column from the old mask to the beginning of the new mask and shift it
   // out from the old mask.
@@ -386,7 +390,9 @@ void ForwardErrorCorrection::CopyColumn(uint8_t* new_mask, int new_mask_bytes,
 }
 
 void ForwardErrorCorrection::GenerateFecUlpHeaders(
-    const PacketList& media_packet_list, uint8_t* packet_mask, bool l_bit,
+    const PacketList& media_packet_list,
+    uint8_t* packet_mask,
+    bool l_bit,
     int num_fec_packets) {
   // -- Generate FEC and ULP headers --
   //
@@ -412,33 +418,34 @@ void ForwardErrorCorrection::GenerateFecUlpHeaders(
   PacketList::const_iterator media_list_it = media_packet_list.begin();
   Packet* media_packet = *media_list_it;
   assert(media_packet != NULL);
-  int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
+  int num_mask_bytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
   const uint16_t ulp_header_size =
       l_bit ? kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
 
   for (int i = 0; i < num_fec_packets; ++i) {
+    Packet* const fec_packet = &generated_fec_packets_[i];
     // -- FEC header --
-    generated_fec_packets_[i].data[0] &= 0x7f;  // Set E to zero.
+    fec_packet->data[0] &= 0x7f;  // Set E to zero.
     if (l_bit == 0) {
-      generated_fec_packets_[i].data[0] &= 0xbf;  // Clear the L bit.
+      fec_packet->data[0] &= 0xbf;  // Clear the L bit.
     } else {
-      generated_fec_packets_[i].data[0] |= 0x40;  // Set the L bit.
+      fec_packet->data[0] |= 0x40;  // Set the L bit.
     }
     // Two byte sequence number from first RTP packet to SN base.
     // We use the same sequence number base for every FEC packet,
     // but that's not required in general.
-    memcpy(&generated_fec_packets_[i].data[2], &media_packet->data[2], 2);
+    memcpy(&fec_packet->data[2], &media_packet->data[2], 2);
 
     // -- ULP header --
     // Copy the payload size to the protection length field.
     // (We protect the entire packet.)
     ByteWriter<uint16_t>::WriteBigEndian(
-        &generated_fec_packets_[i].data[10],
-        generated_fec_packets_[i].length - kFecHeaderSize - ulp_header_size);
+        &fec_packet->data[10],
+        fec_packet->length - kFecHeaderSize - ulp_header_size);
 
     // Copy the packet mask.
-    memcpy(&generated_fec_packets_[i].data[12], &packet_mask[i * num_maskBytes],
-           num_maskBytes);
+    memcpy(&fec_packet->data[12], &packet_mask[i * num_mask_bytes],
+           num_mask_bytes);
   }
 }
 
@@ -460,7 +467,7 @@ void ForwardErrorCorrection::ResetState(
     ProtectedPacketList::iterator protected_packet_list_it;
     protected_packet_list_it = fec_packet->protected_pkt_list.begin();
     while (protected_packet_list_it != fec_packet->protected_pkt_list.end()) {
-      delete* protected_packet_list_it;
+      delete *protected_packet_list_it;
       protected_packet_list_it =
           fec_packet->protected_pkt_list.erase(protected_packet_list_it);
     }
@@ -472,7 +479,8 @@ void ForwardErrorCorrection::ResetState(
 }
 
 void ForwardErrorCorrection::InsertMediaPacket(
-    ReceivedPacket* rx_packet, RecoveredPacketList* recovered_packet_list) {
+    ReceivedPacket* rx_packet,
+    RecoveredPacketList* recovered_packet_list) {
   RecoveredPacketList::iterator recovered_packet_list_it =
       recovered_packet_list->begin();
 
@@ -538,9 +546,9 @@ void ForwardErrorCorrection::InsertFECPacket(
 
   const uint16_t seq_num_base =
       ByteReader<uint16_t>::ReadBigEndian(&fec_packet->pkt->data[2]);
-  const uint16_t maskSizeBytes =
-      (fec_packet->pkt->data[0] & 0x40) ? kMaskSizeLBitSet
-                                        : kMaskSizeLBitClear;  // L bit set?
+  const uint16_t maskSizeBytes = (fec_packet->pkt->data[0] & 0x40)
+                                     ? kMaskSizeLBitSet
+                                     : kMaskSizeLBitClear;  // L bit set?
 
   for (uint16_t byte_idx = 0; byte_idx < maskSizeBytes; ++byte_idx) {
     uint8_t packet_mask = fec_packet->pkt->data[12 + byte_idx];
@@ -574,7 +582,8 @@ void ForwardErrorCorrection::InsertFECPacket(
 }
 
 void ForwardErrorCorrection::AssignRecoveredPackets(
-    FecPacket* fec_packet, const RecoveredPacketList* recovered_packets) {
+    FecPacket* fec_packet,
+    const RecoveredPacketList* recovered_packets) {
   // Search for missing packets which have arrived or have been recovered by
   // another FEC packet.
   ProtectedPacketList* not_recovered = &fec_packet->protected_pkt_list;
@@ -599,7 +608,6 @@ void ForwardErrorCorrection::AssignRecoveredPackets(
 void ForwardErrorCorrection::InsertPackets(
     ReceivedPacketList* received_packet_list,
     RecoveredPacketList* recovered_packet_list) {
-
   while (!received_packet_list->empty()) {
     ReceivedPacket* rx_packet = received_packet_list->front();
 
@@ -611,9 +619,9 @@ void ForwardErrorCorrection::InsertPackets(
     // old FEC packets based on timestamp information or better sequence number
     // thresholding (e.g., to distinguish between wrap-around and reordering).
     if (!fec_packet_list_.empty()) {
-      uint16_t seq_num_diff = abs(
-          static_cast<int>(rx_packet->seq_num) -
-          static_cast<int>(fec_packet_list_.front()->seq_num));
+      uint16_t seq_num_diff =
+          abs(static_cast<int>(rx_packet->seq_num) -
+              static_cast<int>(fec_packet_list_.front()->seq_num));
       if (seq_num_diff > 0x3fff) {
         DiscardFECPacket(fec_packet_list_.front());
         fec_packet_list_.pop_front();
@@ -634,23 +642,35 @@ void ForwardErrorCorrection::InsertPackets(
   DiscardOldPackets(recovered_packet_list);
 }
 
-void ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet,
+bool ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet,
                                           RecoveredPacket* recovered) {
   // This is the first packet which we try to recover with.
-  const uint16_t ulp_header_size =
-      fec_packet->pkt->data[0] & 0x40 ? kUlpHeaderSizeLBitSet
-                                      : kUlpHeaderSizeLBitClear;  // L bit set?
+  const uint16_t ulp_header_size = fec_packet->pkt->data[0] & 0x40
+                                       ? kUlpHeaderSizeLBitSet
+                                       : kUlpHeaderSizeLBitClear;  // L bit set?
+  if (fec_packet->pkt->length <
+      static_cast<size_t>(kFecHeaderSize + ulp_header_size)) {
+    LOG(LS_WARNING)
+        << "Truncated FEC packet doesn't contain room for ULP header.";
+    return false;
+  }
   recovered->pkt = new Packet;
   memset(recovered->pkt->data, 0, IP_PACKET_SIZE);
   recovered->returned = false;
   recovered->was_recovered = true;
-  uint8_t protection_length[2];
-  // Copy the protection length from the ULP header.
-  memcpy(protection_length, &fec_packet->pkt->data[10], 2);
+  uint16_t protection_length =
+      ByteReader<uint16_t>::ReadBigEndian(&fec_packet->pkt->data[10]);
+  if (protection_length >
+      std::min(
+          sizeof(recovered->pkt->data) - kRtpHeaderSize,
+          sizeof(fec_packet->pkt->data) - kFecHeaderSize - ulp_header_size)) {
+    LOG(LS_WARNING) << "Incorrect FEC protection length, dropping.";
+    return false;
+  }
   // Copy FEC payload, skipping the ULP header.
   memcpy(&recovered->pkt->data[kRtpHeaderSize],
          &fec_packet->pkt->data[kFecHeaderSize + ulp_header_size],
-         ByteReader<uint16_t>::ReadBigEndian(protection_length));
+         protection_length);
   // Copy the length recovery field.
   memcpy(recovered->length_recovery, &fec_packet->pkt->data[8], 2);
   // Copy the first 2 bytes of the FEC header.
@@ -660,9 +680,10 @@ void ForwardErrorCorrection::InitRecovery(const FecPacket* fec_packet,
   // Set the SSRC field.
   ByteWriter<uint32_t>::WriteBigEndian(&recovered->pkt->data[8],
                                        fec_packet->ssrc);
+  return true;
 }
 
-void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) {
+bool ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) {
   // Set the RTP version to 2.
   recovered->pkt->data[0] |= 0x80;  // Set the 1st bit.
   recovered->pkt->data[0] &= 0xbf;  // Clear the 2nd bit.
@@ -674,6 +695,10 @@ void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) {
   recovered->pkt->length =
       ByteReader<uint16_t>::ReadBigEndian(recovered->length_recovery) +
       kRtpHeaderSize;
+  if (recovered->pkt->length > sizeof(recovered->pkt->data) - kRtpHeaderSize)
+    return false;
+
+  return true;
 }
 
 void ForwardErrorCorrection::XorPackets(const Packet* src_packet,
@@ -700,9 +725,11 @@ void ForwardErrorCorrection::XorPackets(const Packet* src_packet,
   }
 }
 
-void ForwardErrorCorrection::RecoverPacket(
-    const FecPacket* fec_packet, RecoveredPacket* rec_packet_to_insert) {
-  InitRecovery(fec_packet, rec_packet_to_insert);
+bool ForwardErrorCorrection::RecoverPacket(
+    const FecPacket* fec_packet,
+    RecoveredPacket* rec_packet_to_insert) {
+  if (!InitRecovery(fec_packet, rec_packet_to_insert))
+    return false;
   ProtectedPacketList::const_iterator protected_it =
       fec_packet->protected_pkt_list.begin();
   while (protected_it != fec_packet->protected_pkt_list.end()) {
@@ -714,7 +741,9 @@ void ForwardErrorCorrection::RecoverPacket(
     }
     ++protected_it;
   }
-  FinishRecovery(rec_packet_to_insert);
+  if (!FinishRecovery(rec_packet_to_insert))
+    return false;
+  return true;
 }
 
 void ForwardErrorCorrection::AttemptRecover(
@@ -729,7 +758,13 @@ void ForwardErrorCorrection::AttemptRecover(
       // Recovery possible.
       RecoveredPacket* packet_to_insert = new RecoveredPacket;
       packet_to_insert->pkt = NULL;
-      RecoverPacket(*fec_packet_list_it, packet_to_insert);
+      if (!RecoverPacket(*fec_packet_list_it, packet_to_insert)) {
+        // Can't recover using this packet, drop it.
+        DiscardFECPacket(*fec_packet_list_it);
+        fec_packet_list_it = fec_packet_list_.erase(fec_packet_list_it);
+        delete packet_to_insert;
+        continue;
+      }
 
       // Add recovered packet to the list of recovered packets and update any
       // FEC packets covering this packet with a pointer to the data.
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
index a3b3fa0e49..9ba6ce0438 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
@@ -14,9 +14,9 @@
 #include <list>
 #include <vector>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -92,7 +92,7 @@ class ForwardErrorCorrection {
                     // packets, but not required for media packets.
     bool is_fec;    // Set to true if this is an FEC packet and false
                     // otherwise.
-    scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
+    rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
   };
 
   // The recovered list parameter of #DecodeFEC() will reference structs of
@@ -110,7 +110,7 @@ class ForwardErrorCorrection {
                     // caller through the callback.
     uint8_t length_recovery[2];  // Two bytes used for recovering the packet
                                  // length with XOR operations.
-    scoped_refptr<Packet> pkt;   // Pointer to the packet storage.
+    rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
   };
 
   typedef std::list<Packet*> PacketList;
@@ -279,7 +279,7 @@ class ForwardErrorCorrection {
   void AttemptRecover(RecoveredPacketList* recovered_packet_list);
 
   // Initializes the packet recovery using the FEC packet.
-  static void InitRecovery(const FecPacket* fec_packet,
+  static bool InitRecovery(const FecPacket* fec_packet,
                            RecoveredPacket* recovered);
 
   // Performs XOR between |src_packet| and |dst_packet| and stores the result
@@ -287,10 +287,10 @@ class ForwardErrorCorrection {
   static void XorPackets(const Packet* src_packet, RecoveredPacket* dst_packet);
 
   // Finish up the recovery of a packet.
-  static void FinishRecovery(RecoveredPacket* recovered);
+  static bool FinishRecovery(RecoveredPacket* recovered);
 
   // Recover a missing packet.
-  void RecoverPacket(const FecPacket* fec_packet,
+  bool RecoverPacket(const FecPacket* fec_packet,
                      RecoveredPacket* rec_packet_to_insert);
 
   // Get the number of missing media packets which are covered by this
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
index 6d9be90de1..fae59078b1 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -17,6 +17,8 @@
 #include "webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h"
 
 namespace {
+using webrtc::fec_private_tables::kPacketMaskBurstyTbl;
+using webrtc::fec_private_tables::kPacketMaskRandomTbl;
 
 // Allow for different modes of protection for packets in UEP case.
 enum ProtectionMode {
@@ -37,8 +39,11 @@ enum ProtectionMode {
 //                                [0, num_rows * num_sub_mask_bytes]
 // \param[out] packet_mask        A pointer to hold the output mask, of size
 //                                [0, x * num_mask_bytes], where x >= num_rows.
-void FitSubMask(int num_mask_bytes, int num_sub_mask_bytes, int num_rows,
-                const uint8_t* sub_mask, uint8_t* packet_mask) {
+void FitSubMask(int num_mask_bytes,
+                int num_sub_mask_bytes,
+                int num_rows,
+                const uint8_t* sub_mask,
+                uint8_t* packet_mask) {
   if (num_mask_bytes == num_sub_mask_bytes) {
     memcpy(packet_mask, sub_mask, num_rows * num_sub_mask_bytes);
   } else {
@@ -70,13 +75,15 @@ void FitSubMask(int num_mask_bytes, int num_sub_mask_bytes, int num_rows,
 // \param[out] packet_mask        A pointer to hold the output mask, of size
 //                                [0, x * num_mask_bytes],
 //                                where x >= end_row_fec.
-// TODO (marpan): This function is doing three things at the same time:
+// TODO(marpan): This function is doing three things at the same time:
 // shift within a byte, byte shift and resizing.
 // Split up into subroutines.
-void ShiftFitSubMask(int num_mask_bytes, int res_mask_bytes,
-                     int num_column_shift, int end_row, const uint8_t* sub_mask,
+void ShiftFitSubMask(int num_mask_bytes,
+                     int res_mask_bytes,
+                     int num_column_shift,
+                     int end_row,
+                     const uint8_t* sub_mask,
                      uint8_t* packet_mask) {
-
   // Number of bit shifts within a byte
   const int num_bit_shifts = (num_column_shift % 8);
   const int num_byte_shifts = num_column_shift >> 3;
@@ -128,7 +135,6 @@ void ShiftFitSubMask(int num_mask_bytes, int res_mask_bytes,
     // For the first byte in the row (j=0 case).
     shift_right_curr_byte = sub_mask[pkt_mask_idx2] >> num_bit_shifts;
     packet_mask[pkt_mask_idx] = shift_right_curr_byte;
-
   }
 }
 }  // namespace
@@ -151,7 +157,9 @@ FecMaskType PacketMaskTable::InitMaskType(FecMaskType fec_mask_type,
   assert(num_media_packets <= static_cast<int>(sizeof(kPacketMaskRandomTbl) /
                                                sizeof(*kPacketMaskRandomTbl)));
   switch (fec_mask_type) {
-    case kFecMaskRandom: { return kFecMaskRandom; }
+    case kFecMaskRandom: {
+      return kFecMaskRandom;
+    }
     case kFecMaskBursty: {
       int max_media_packets = static_cast<int>(sizeof(kPacketMaskBurstyTbl) /
                                                sizeof(*kPacketMaskBurstyTbl));
@@ -170,17 +178,24 @@ FecMaskType PacketMaskTable::InitMaskType(FecMaskType fec_mask_type,
 // |fec_mask_type|.
 const uint8_t*** PacketMaskTable::InitMaskTable(FecMaskType fec_mask_type) {
   switch (fec_mask_type) {
-    case kFecMaskRandom: { return kPacketMaskRandomTbl; }
-    case kFecMaskBursty: { return kPacketMaskBurstyTbl; }
+    case kFecMaskRandom: {
+      return kPacketMaskRandomTbl;
+    }
+    case kFecMaskBursty: {
+      return kPacketMaskBurstyTbl;
+    }
   }
   assert(false);
   return kPacketMaskRandomTbl;
 }
 
 // Remaining protection after important (first partition) packet protection
-void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
-                               int num_fec_for_imp_packets, int num_mask_bytes,
-                               ProtectionMode mode, uint8_t* packet_mask,
+void RemainingPacketProtection(int num_media_packets,
+                               int num_fec_remaining,
+                               int num_fec_for_imp_packets,
+                               int num_mask_bytes,
+                               ProtectionMode mode,
+                               uint8_t* packet_mask,
                                const PacketMaskTable& mask_table) {
   if (mode == kModeNoOverlap) {
     // sub_mask21
@@ -191,8 +206,10 @@ void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
     const int res_mask_bytes =
         (l_bit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
-    const uint8_t* packet_mask_sub_21 = mask_table.fec_packet_mask_table()[
-        num_media_packets - num_fec_for_imp_packets - 1][num_fec_remaining - 1];
+    const uint8_t* packet_mask_sub_21 =
+        mask_table.fec_packet_mask_table()[num_media_packets -
+                                           num_fec_for_imp_packets -
+                                           1][num_fec_remaining - 1];
 
     ShiftFitSubMask(num_mask_bytes, res_mask_bytes, num_fec_for_imp_packets,
                     (num_fec_for_imp_packets + num_fec_remaining),
@@ -201,8 +218,9 @@ void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
   } else if (mode == kModeOverlap || mode == kModeBiasFirstPacket) {
     // sub_mask22
 
-    const uint8_t* packet_mask_sub_22 = mask_table
-        .fec_packet_mask_table()[num_media_packets - 1][num_fec_remaining - 1];
+    const uint8_t* packet_mask_sub_22 =
+        mask_table.fec_packet_mask_table()[num_media_packets -
+                                           1][num_fec_remaining - 1];
 
     FitSubMask(num_mask_bytes, num_mask_bytes, num_fec_remaining,
                packet_mask_sub_22,
@@ -217,41 +235,42 @@ void RemainingPacketProtection(int num_media_packets, int num_fec_remaining,
   } else {
     assert(false);
   }
-
 }
 
 // Protection for important (first partition) packets
-void ImportantPacketProtection(int num_fec_for_imp_packets, int num_imp_packets,
-                               int num_mask_bytes, uint8_t* packet_mask,
+void ImportantPacketProtection(int num_fec_for_imp_packets,
+                               int num_imp_packets,
+                               int num_mask_bytes,
+                               uint8_t* packet_mask,
                                const PacketMaskTable& mask_table) {
   const int l_bit = num_imp_packets > 16 ? 1 : 0;
   const int num_imp_mask_bytes =
       (l_bit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
   // Get sub_mask1 from table
-  const uint8_t* packet_mask_sub_1 = mask_table.fec_packet_mask_table()[
-      num_imp_packets - 1][num_fec_for_imp_packets - 1];
+  const uint8_t* packet_mask_sub_1 =
+      mask_table.fec_packet_mask_table()[num_imp_packets -
+                                         1][num_fec_for_imp_packets - 1];
 
   FitSubMask(num_mask_bytes, num_imp_mask_bytes, num_fec_for_imp_packets,
              packet_mask_sub_1, packet_mask);
-
 }
 
 // This function sets the protection allocation: i.e., how many FEC packets
 // to use for num_imp (1st partition) packets, given the: number of media
 // packets, number of FEC packets, and number of 1st partition packets.
-int SetProtectionAllocation(int num_media_packets, int num_fec_packets,
+int SetProtectionAllocation(int num_media_packets,
+                            int num_fec_packets,
                             int num_imp_packets) {
-
-  // TODO (marpan): test different cases for protection allocation:
+  // TODO(marpan): test different cases for protection allocation:
 
   // Use at most (alloc_par * num_fec_packets) for important packets.
   float alloc_par = 0.5;
   int max_num_fec_for_imp = alloc_par * num_fec_packets;
 
-  int num_fec_for_imp_packets =
-      (num_imp_packets < max_num_fec_for_imp) ? num_imp_packets
-                                              : max_num_fec_for_imp;
+  int num_fec_for_imp_packets = (num_imp_packets < max_num_fec_for_imp)
+                                    ? num_imp_packets
+                                    : max_num_fec_for_imp;
 
   // Fall back to equal protection in this case
   if (num_fec_packets == 1 && (num_media_packets > 2 * num_imp_packets)) {
@@ -268,7 +287,7 @@ int SetProtectionAllocation(int num_media_packets, int num_fec_packets,
 // Current version has 3 modes (options) to build UEP mask from existing ones.
 // Various other combinations may be added in future versions.
 // Longer-term, we may add another set of tables specifically for UEP cases.
-// TODO (marpan): also consider modification of masks for bursty loss cases.
+// TODO(marpan): also consider modification of masks for bursty loss cases.
 
 // Mask is characterized as (#packets_to_protect, #fec_for_protection).
 // Protection factor defined as: (#fec_for_protection / #packets_to_protect).
@@ -306,13 +325,14 @@ int SetProtectionAllocation(int num_media_packets, int num_fec_packets,
 // Protection Mode 2 may be extended for a sort of sliding protection
 // (i.e., vary the number/density of "1s" across columns) across packets.
 
-void UnequalProtectionMask(int num_media_packets, int num_fec_packets,
-                           int num_imp_packets, int num_mask_bytes,
+void UnequalProtectionMask(int num_media_packets,
+                           int num_fec_packets,
+                           int num_imp_packets,
+                           int num_mask_bytes,
                            uint8_t* packet_mask,
                            const PacketMaskTable& mask_table) {
-
   // Set Protection type and allocation
-  // TODO (marpan): test/update for best mode and some combinations thereof.
+  // TODO(marpan): test/update for best mode and some combinations thereof.
 
   ProtectionMode mode = kModeOverlap;
   int num_fec_for_imp_packets = 0;
@@ -341,11 +361,12 @@ void UnequalProtectionMask(int num_media_packets, int num_fec_packets,
                               num_fec_for_imp_packets, num_mask_bytes, mode,
                               packet_mask, mask_table);
   }
-
 }
 
-void GeneratePacketMasks(int num_media_packets, int num_fec_packets,
-                         int num_imp_packets, bool use_unequal_protection,
+void GeneratePacketMasks(int num_media_packets,
+                         int num_fec_packets,
+                         int num_imp_packets,
+                         bool use_unequal_protection,
                          const PacketMaskTable& mask_table,
                          uint8_t* packet_mask) {
   assert(num_media_packets > 0);
@@ -361,16 +382,15 @@ void GeneratePacketMasks(int num_media_packets, int num_fec_packets,
     // Retrieve corresponding mask table directly:for equal-protection case.
     // Mask = (k,n-k), with protection factor = (n-k)/k,
     // where k = num_media_packets, n=total#packets, (n-k)=num_fec_packets.
-    memcpy(packet_mask, mask_table.fec_packet_mask_table()[
-                            num_media_packets - 1][num_fec_packets - 1],
+    memcpy(packet_mask,
+           mask_table.fec_packet_mask_table()[num_media_packets -
+                                              1][num_fec_packets - 1],
            num_fec_packets * num_mask_bytes);
-  } else  //UEP case
-      {
+  } else {  // UEP case
     UnequalProtectionMask(num_media_packets, num_fec_packets, num_imp_packets,
                           num_mask_bytes, packet_mask, mask_table);
-
   }  // End of UEP modification
-}  //End of GetPacketMasks
+}  // End of GetPacketMasks
 
 }  // namespace internal
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc
new file mode 100644
index 0000000000..b78b96dc86
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc
@@ -0,0 +1,565 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
+
+#include <vector>
+
+#include "webrtc/base/bitbuffer.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+namespace {
+// The size of a NALU header {0 0 0 1}.
+static const size_t kNaluHeaderSize = 4;
+
+// The size of a NALU header plus the type byte.
+static const size_t kNaluHeaderAndTypeSize = kNaluHeaderSize + 1;
+
+// The NALU type.
+static const uint8_t kNaluSps = 0x7;
+static const uint8_t kNaluPps = 0x8;
+static const uint8_t kNaluIdr = 0x5;
+static const uint8_t kNaluTypeMask = 0x1F;
+
+static const uint8_t kSliceTypeP = 0x0;
+static const uint8_t kSliceTypeB = 0x1;
+static const uint8_t kSliceTypeSp = 0x3;
+
+// Returns a vector of the NALU start sequences (0 0 0 1) in the given buffer.
+std::vector<size_t> FindNaluStartSequences(const uint8_t* buffer,
+                                           size_t buffer_size) {
+  std::vector<size_t> sequences;
+  // This is sorta like Boyer-Moore, but with only the first optimization step:
+  // given a 4-byte sequence we're looking at, if the 4th byte isn't 1 or 0,
+  // skip ahead to the next 4-byte sequence. 0s and 1s are relatively rare, so
+  // this will skip the majority of reads/checks.
+  const uint8_t* end = buffer + buffer_size - 4;
+  for (const uint8_t* head = buffer; head < end;) {
+    if (head[3] > 1) {
+      head += 4;
+    } else if (head[3] == 1 && head[2] == 0 && head[1] == 0 && head[0] == 0) {
+      sequences.push_back(static_cast<size_t>(head - buffer));
+      head += 4;
+    } else {
+      head++;
+    }
+  }
+
+  return sequences;
+}
+}  // namespace
+
+// Parses RBSP from source bytes. Removes emulation bytes, but leaves the
+// rbsp_trailing_bits() in the stream, since none of the parsing reads all the
+// way to the end of a parsed RBSP sequence. When writing, that means the
+// rbsp_trailing_bits() should be preserved and don't need to be restored (i.e.
+// the rbsp_stop_one_bit, which is just a 1, then zero padded), and alignment
+// should "just work".
+// TODO(pbos): Make parsing RBSP something that can be integrated into BitBuffer
+// so we don't have to copy the entire frames when only interested in the
+// headers.
+rtc::ByteBuffer* ParseRbsp(const uint8_t* bytes, size_t length) {
+  // Copied from webrtc::H264SpsParser::Parse.
+  rtc::ByteBuffer* rbsp_buffer = new rtc::ByteBuffer;
+  for (size_t i = 0; i < length;) {
+    if (length - i >= 3 && bytes[i] == 0 && bytes[i + 1] == 0 &&
+        bytes[i + 2] == 3) {
+      rbsp_buffer->WriteBytes(reinterpret_cast<const char*>(bytes) + i, 2);
+      i += 3;
+    } else {
+      rbsp_buffer->WriteBytes(reinterpret_cast<const char*>(bytes) + i, 1);
+      i++;
+    }
+  }
+  return rbsp_buffer;
+}
+
+#define RETURN_FALSE_ON_FAIL(x)       \
+  if (!(x)) {                         \
+    LOG_F(LS_ERROR) << "FAILED: " #x; \
+    return false;                     \
+  }
+
+H264BitstreamParser::PpsState::PpsState() {}
+
+H264BitstreamParser::SpsState::SpsState() {}
+
+// These functions are similar to webrtc::H264SpsParser::Parse, and based on the
+// same version of the H.264 standard. You can find it here:
+// http://www.itu.int/rec/T-REC-H.264
+bool H264BitstreamParser::ParseSpsNalu(const uint8_t* sps, size_t length) {
+  // Reset SPS state.
+  sps_ = SpsState();
+  sps_parsed_ = false;
+  // Parse out the SPS RBSP. It should be small, so it's ok that we create a
+  // copy. We'll eventually write this back.
+  rtc::scoped_ptr<rtc::ByteBuffer> sps_rbsp(
+      ParseRbsp(sps + kNaluHeaderAndTypeSize, length - kNaluHeaderAndTypeSize));
+  rtc::BitBuffer sps_parser(reinterpret_cast<const uint8_t*>(sps_rbsp->Data()),
+                            sps_rbsp->Length());
+
+  uint8_t byte_tmp;
+  uint32_t golomb_tmp;
+  uint32_t bits_tmp;
+
+  // profile_idc: u(8).
+  uint8_t profile_idc;
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadUInt8(&profile_idc));
+  // constraint_set0_flag through constraint_set5_flag + reserved_zero_2bits
+  // 1 bit each for the flags + 2 bits = 8 bits = 1 byte.
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadUInt8(&byte_tmp));
+  // level_idc: u(8)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadUInt8(&byte_tmp));
+  // seq_parameter_set_id: ue(v)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+  sps_.separate_colour_plane_flag = 0;
+  // See if profile_idc has chroma format information.
+  if (profile_idc == 100 || profile_idc == 110 || profile_idc == 122 ||
+      profile_idc == 244 || profile_idc == 44 || profile_idc == 83 ||
+      profile_idc == 86 || profile_idc == 118 || profile_idc == 128 ||
+      profile_idc == 138 || profile_idc == 139 || profile_idc == 134) {
+    // chroma_format_idc: ue(v)
+    uint32_t chroma_format_idc;
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&chroma_format_idc));
+    if (chroma_format_idc == 3) {
+      // separate_colour_plane_flag: u(1)
+      RETURN_FALSE_ON_FAIL(
+          sps_parser.ReadBits(&sps_.separate_colour_plane_flag, 1));
+    }
+    // bit_depth_luma_minus8: ue(v)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+    // bit_depth_chroma_minus8: ue(v)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+    // qpprime_y_zero_transform_bypass_flag: u(1)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadBits(&bits_tmp, 1));
+    // seq_scaling_matrix_present_flag: u(1)
+    uint32_t seq_scaling_matrix_present_flag;
+    RETURN_FALSE_ON_FAIL(
+        sps_parser.ReadBits(&seq_scaling_matrix_present_flag, 1));
+    if (seq_scaling_matrix_present_flag) {
+      // seq_scaling_list_present_flags. Either 8 or 12, depending on
+      // chroma_format_idc.
+      uint32_t seq_scaling_list_present_flags;
+      if (chroma_format_idc != 3) {
+        RETURN_FALSE_ON_FAIL(
+            sps_parser.ReadBits(&seq_scaling_list_present_flags, 8));
+      } else {
+        RETURN_FALSE_ON_FAIL(
+            sps_parser.ReadBits(&seq_scaling_list_present_flags, 12));
+      }
+      // TODO(pbos): Support parsing scaling lists if they're seen in practice.
+      RTC_CHECK(seq_scaling_list_present_flags == 0)
+          << "SPS contains scaling lists, which are unsupported.";
+    }
+  }
+  // log2_max_frame_num_minus4: ue(v)
+  RETURN_FALSE_ON_FAIL(
+      sps_parser.ReadExponentialGolomb(&sps_.log2_max_frame_num_minus4));
+  // pic_order_cnt_type: ue(v)
+  RETURN_FALSE_ON_FAIL(
+      sps_parser.ReadExponentialGolomb(&sps_.pic_order_cnt_type));
+
+  if (sps_.pic_order_cnt_type == 0) {
+    // log2_max_pic_order_cnt_lsb_minus4: ue(v)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(
+        &sps_.log2_max_pic_order_cnt_lsb_minus4));
+  } else if (sps_.pic_order_cnt_type == 1) {
+    // delta_pic_order_always_zero_flag: u(1)
+    RETURN_FALSE_ON_FAIL(
+        sps_parser.ReadBits(&sps_.delta_pic_order_always_zero_flag, 1));
+    // offset_for_non_ref_pic: se(v)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+    // offset_for_top_to_bottom_field: se(v)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+    uint32_t num_ref_frames_in_pic_order_cnt_cycle;
+    // num_ref_frames_in_pic_order_cnt_cycle: ue(v)
+    RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(
+        &num_ref_frames_in_pic_order_cnt_cycle));
+    for (uint32_t i = 0; i < num_ref_frames_in_pic_order_cnt_cycle; i++) {
+      // offset_for_ref_frame[i]: se(v)
+      RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+    }
+  }
+  // max_num_ref_frames: ue(v)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+  // gaps_in_frame_num_value_allowed_flag: u(1)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadBits(&bits_tmp, 1));
+  // pic_width_in_mbs_minus1: ue(v)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+  // pic_height_in_map_units_minus1: ue(v)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadExponentialGolomb(&golomb_tmp));
+  // frame_mbs_only_flag: u(1)
+  RETURN_FALSE_ON_FAIL(sps_parser.ReadBits(&sps_.frame_mbs_only_flag, 1));
+  sps_parsed_ = true;
+  return true;
+}
+
+bool H264BitstreamParser::ParsePpsNalu(const uint8_t* pps, size_t length) {
+  RTC_CHECK(sps_parsed_);
+  // We're starting a new stream, so reset picture type rewriting values.
+  pps_ = PpsState();
+  pps_parsed_ = false;
+  rtc::scoped_ptr<rtc::ByteBuffer> buffer(
+      ParseRbsp(pps + kNaluHeaderAndTypeSize, length - kNaluHeaderAndTypeSize));
+  rtc::BitBuffer parser(reinterpret_cast<const uint8_t*>(buffer->Data()),
+                        buffer->Length());
+
+  uint32_t bits_tmp;
+  uint32_t golomb_ignored;
+  // pic_parameter_set_id: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // seq_parameter_set_id: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // entropy_coding_mode_flag: u(1)
+  uint32_t entropy_coding_mode_flag;
+  RETURN_FALSE_ON_FAIL(parser.ReadBits(&entropy_coding_mode_flag, 1));
+  // TODO(pbos): Implement CABAC support if spotted in the wild.
+  RTC_CHECK(entropy_coding_mode_flag == 0)
+      << "Don't know how to parse CABAC streams.";
+  // bottom_field_pic_order_in_frame_present_flag: u(1)
+  uint32_t bottom_field_pic_order_in_frame_present_flag;
+  RETURN_FALSE_ON_FAIL(
+      parser.ReadBits(&bottom_field_pic_order_in_frame_present_flag, 1));
+  pps_.bottom_field_pic_order_in_frame_present_flag =
+      bottom_field_pic_order_in_frame_present_flag != 0;
+
+  // num_slice_groups_minus1: ue(v)
+  uint32_t num_slice_groups_minus1;
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&num_slice_groups_minus1));
+  if (num_slice_groups_minus1 > 0) {
+    uint32_t slice_group_map_type;
+    // slice_group_map_type: ue(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&slice_group_map_type));
+    if (slice_group_map_type == 0) {
+      for (uint32_t i_group = 0; i_group <= num_slice_groups_minus1;
+           ++i_group) {
+        // run_length_minus1[iGroup]: ue(v)
+        RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+      }
+    } else if (slice_group_map_type == 2) {
+      for (uint32_t i_group = 0; i_group <= num_slice_groups_minus1;
+           ++i_group) {
+        // top_left[iGroup]: ue(v)
+        RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+        // bottom_right[iGroup]: ue(v)
+        RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+      }
+    } else if (slice_group_map_type == 3 || slice_group_map_type == 4 ||
+               slice_group_map_type == 5) {
+      // slice_group_change_direction_flag: u(1)
+      RETURN_FALSE_ON_FAIL(parser.ReadBits(&bits_tmp, 1));
+      // slice_group_change_rate_minus1: ue(v)
+      RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+    } else if (slice_group_map_type == 6) {
+      // pic_size_in_map_units_minus1: ue(v)
+      uint32_t pic_size_in_map_units_minus1;
+      RETURN_FALSE_ON_FAIL(
+          parser.ReadExponentialGolomb(&pic_size_in_map_units_minus1));
+      uint32_t slice_group_id_bits = 0;
+      uint32_t num_slice_groups = num_slice_groups_minus1 + 1;
+      // If num_slice_groups is not a power of two an additional bit is required
+      // to account for the ceil() of log2() below.
+      if ((num_slice_groups & (num_slice_groups - 1)) != 0)
+        ++slice_group_id_bits;
+      while (num_slice_groups > 0) {
+        num_slice_groups >>= 1;
+        ++slice_group_id_bits;
+      }
+      for (uint32_t i = 0; i <= pic_size_in_map_units_minus1; i++) {
+        // slice_group_id[i]: u(v)
+        // Represented by ceil(log2(num_slice_groups_minus1 + 1)) bits.
+        RETURN_FALSE_ON_FAIL(parser.ReadBits(&bits_tmp, slice_group_id_bits));
+      }
+    }
+  }
+  // num_ref_idx_l0_default_active_minus1: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // num_ref_idx_l1_default_active_minus1: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // weighted_pred_flag: u(1)
+  uint32_t weighted_pred_flag;
+  RETURN_FALSE_ON_FAIL(parser.ReadBits(&weighted_pred_flag, 1));
+  pps_.weighted_pred_flag = weighted_pred_flag != 0;
+  // weighted_bipred_idc: u(2)
+  RETURN_FALSE_ON_FAIL(parser.ReadBits(&pps_.weighted_bipred_idc, 2));
+
+  // pic_init_qp_minus26: se(v)
+  RETURN_FALSE_ON_FAIL(
+      parser.ReadSignedExponentialGolomb(&pps_.pic_init_qp_minus26));
+  // pic_init_qs_minus26: se(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // chroma_qp_index_offset: se(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // deblocking_filter_control_present_flag: u(1)
+  // constrained_intra_pred_flag: u(1)
+  RETURN_FALSE_ON_FAIL(parser.ReadBits(&bits_tmp, 2));
+  // redundant_pic_cnt_present_flag: u(1)
+  RETURN_FALSE_ON_FAIL(
+      parser.ReadBits(&pps_.redundant_pic_cnt_present_flag, 1));
+
+  pps_parsed_ = true;
+  return true;
+}
+
+bool H264BitstreamParser::ParseNonParameterSetNalu(const uint8_t* source,
+                                                   size_t source_length,
+                                                   uint8_t nalu_type) {
+  RTC_CHECK(sps_parsed_);
+  RTC_CHECK(pps_parsed_);
+  last_slice_qp_delta_parsed_ = false;
+  rtc::scoped_ptr<rtc::ByteBuffer> slice_rbsp(ParseRbsp(
+      source + kNaluHeaderAndTypeSize, source_length - kNaluHeaderAndTypeSize));
+  rtc::BitBuffer slice_reader(
+      reinterpret_cast<const uint8_t*>(slice_rbsp->Data()),
+      slice_rbsp->Length());
+  // Check to see if this is an IDR slice, which has an extra field to parse
+  // out.
+  bool is_idr = (source[kNaluHeaderSize] & 0x0F) == kNaluIdr;
+  uint8_t nal_ref_idc = (source[kNaluHeaderSize] & 0x60) >> 5;
+  uint32_t golomb_tmp;
+  uint32_t bits_tmp;
+
+  // first_mb_in_slice: ue(v)
+  RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+  // slice_type: ue(v)
+  uint32_t slice_type;
+  RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&slice_type));
+  // slice_type's 5..9 range is used to indicate that all slices of a picture
+  // have the same value of slice_type % 5, we don't care about that, so we map
+  // to the corresponding 0..4 range.
+  slice_type %= 5;
+  // pic_parameter_set_id: ue(v)
+  RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+  if (sps_.separate_colour_plane_flag == 1) {
+    // colour_plane_id
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadBits(&bits_tmp, 2));
+  }
+  // frame_num: u(v)
+  // Represented by log2_max_frame_num_minus4 + 4 bits.
+  RETURN_FALSE_ON_FAIL(
+      slice_reader.ReadBits(&bits_tmp, sps_.log2_max_frame_num_minus4 + 4));
+  uint32_t field_pic_flag = 0;
+  if (sps_.frame_mbs_only_flag == 0) {
+    // field_pic_flag: u(1)
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadBits(&field_pic_flag, 1));
+    if (field_pic_flag != 0) {
+      // bottom_field_flag: u(1)
+      RETURN_FALSE_ON_FAIL(slice_reader.ReadBits(&bits_tmp, 1));
+    }
+  }
+  if (is_idr) {
+    // idr_pic_id: ue(v)
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+  }
+  // pic_order_cnt_lsb: u(v)
+  // Represented by sps_.log2_max_pic_order_cnt_lsb_minus4 + 4 bits.
+  if (sps_.pic_order_cnt_type == 0) {
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadBits(
+        &bits_tmp, sps_.log2_max_pic_order_cnt_lsb_minus4 + 4));
+    if (pps_.bottom_field_pic_order_in_frame_present_flag &&
+        field_pic_flag == 0) {
+      // delta_pic_order_cnt_bottom: se(v)
+      RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+    }
+  }
+  if (sps_.pic_order_cnt_type == 1 && !sps_.delta_pic_order_always_zero_flag) {
+    // delta_pic_order_cnt[0]: se(v)
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+    if (pps_.bottom_field_pic_order_in_frame_present_flag && !field_pic_flag) {
+      // delta_pic_order_cnt[1]: se(v)
+      RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+    }
+  }
+  if (pps_.redundant_pic_cnt_present_flag) {
+    // redundant_pic_cnt: ue(v)
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+  }
+  if (slice_type == kSliceTypeB) {
+    // direct_spatial_mv_pred_flag: u(1)
+    RETURN_FALSE_ON_FAIL(slice_reader.ReadBits(&bits_tmp, 1));
+  }
+  if (slice_type == kSliceTypeP || slice_type == kSliceTypeSp ||
+      slice_type == kSliceTypeB) {
+    uint32_t num_ref_idx_active_override_flag;
+    // num_ref_idx_active_override_flag: u(1)
+    RETURN_FALSE_ON_FAIL(
+        slice_reader.ReadBits(&num_ref_idx_active_override_flag, 1));
+    if (num_ref_idx_active_override_flag != 0) {
+      // num_ref_idx_l0_active_minus1: ue(v)
+      RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+      if (slice_type == kSliceTypeB) {
+        // num_ref_idx_l1_active_minus1: ue(v)
+        RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
+      }
+    }
+  }
+  // assume nal_unit_type != 20 && nal_unit_type != 21:
+  RTC_CHECK_NE(nalu_type, 20);
+  RTC_CHECK_NE(nalu_type, 21);
+  // if (nal_unit_type == 20 || nal_unit_type == 21)
+  //   ref_pic_list_mvc_modification()
+  // else
+  {
+    // ref_pic_list_modification():
+    // |slice_type| checks here don't use named constants as they aren't named
+    // in the spec for this segment. Keeping them consistent makes it easier to
+    // verify that they are both the same.
+    if (slice_type % 5 != 2 && slice_type % 5 != 4) {
+      // ref_pic_list_modification_flag_l0: u(1)
+      uint32_t ref_pic_list_modification_flag_l0;
+      RETURN_FALSE_ON_FAIL(
+          slice_reader.ReadBits(&ref_pic_list_modification_flag_l0, 1));
+      if (ref_pic_list_modification_flag_l0) {
+        uint32_t modification_of_pic_nums_idc;
+        do {
+          // modification_of_pic_nums_idc: ue(v)
+          RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(
+              &modification_of_pic_nums_idc));
+          if (modification_of_pic_nums_idc == 0 ||
+              modification_of_pic_nums_idc == 1) {
+            // abs_diff_pic_num_minus1: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          } else if (modification_of_pic_nums_idc == 2) {
+            // long_term_pic_num: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          }
+        } while (modification_of_pic_nums_idc != 3);
+      }
+    }
+    if (slice_type % 5 == 1) {
+      // ref_pic_list_modification_flag_l1: u(1)
+      uint32_t ref_pic_list_modification_flag_l1;
+      RETURN_FALSE_ON_FAIL(
+          slice_reader.ReadBits(&ref_pic_list_modification_flag_l1, 1));
+      if (ref_pic_list_modification_flag_l1) {
+        uint32_t modification_of_pic_nums_idc;
+        do {
+          // modification_of_pic_nums_idc: ue(v)
+          RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(
+              &modification_of_pic_nums_idc));
+          if (modification_of_pic_nums_idc == 0 ||
+              modification_of_pic_nums_idc == 1) {
+            // abs_diff_pic_num_minus1: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          } else if (modification_of_pic_nums_idc == 2) {
+            // long_term_pic_num: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          }
+        } while (modification_of_pic_nums_idc != 3);
+      }
+    }
+  }
+  // TODO(pbos): Do we need support for pred_weight_table()?
+  RTC_CHECK(!((pps_.weighted_pred_flag &&
+               (slice_type == kSliceTypeP || slice_type == kSliceTypeSp)) ||
+              (pps_.weighted_bipred_idc != 0 && slice_type == kSliceTypeB)))
+      << "Missing support for pred_weight_table().";
+  // if ((weighted_pred_flag && (slice_type == P || slice_type == SP)) ||
+  //    (weighted_bipred_idc == 1 && slice_type == B)) {
+  //  pred_weight_table()
+  // }
+  if (nal_ref_idc != 0) {
+    // dec_ref_pic_marking():
+    if (is_idr) {
+      // no_output_of_prior_pics_flag: u(1)
+      // long_term_reference_flag: u(1)
+      RETURN_FALSE_ON_FAIL(slice_reader.ReadBits(&bits_tmp, 2));
+    } else {
+      // adaptive_ref_pic_marking_mode_flag: u(1)
+      uint32_t adaptive_ref_pic_marking_mode_flag;
+      RETURN_FALSE_ON_FAIL(
+          slice_reader.ReadBits(&adaptive_ref_pic_marking_mode_flag, 1));
+      if (adaptive_ref_pic_marking_mode_flag) {
+        uint32_t memory_management_control_operation;
+        do {
+          // memory_management_control_operation: ue(v)
+          RETURN_FALSE_ON_FAIL(slice_reader.ReadExponentialGolomb(
+              &memory_management_control_operation));
+          if (memory_management_control_operation == 1 ||
+              memory_management_control_operation == 3) {
+            // difference_of_pic_nums_minus1: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          }
+          if (memory_management_control_operation == 2) {
+            // long_term_pic_num: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          }
+          if (memory_management_control_operation == 3 ||
+              memory_management_control_operation == 6) {
+            // long_term_frame_idx: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          }
+          if (memory_management_control_operation == 4) {
+            // max_long_term_frame_idx_plus1: ue(v)
+            RETURN_FALSE_ON_FAIL(
+                slice_reader.ReadExponentialGolomb(&golomb_tmp));
+          }
+        } while (memory_management_control_operation != 0);
+      }
+    }
+  }
+  // cabac not supported: entropy_coding_mode_flag == 0 asserted above.
+  // if (entropy_coding_mode_flag && slice_type != I && slice_type != SI)
+  //   cabac_init_idc
+  RETURN_FALSE_ON_FAIL(
+      slice_reader.ReadSignedExponentialGolomb(&last_slice_qp_delta_));
+  last_slice_qp_delta_parsed_ = true;
+  return true;
+}
+
+void H264BitstreamParser::ParseSlice(const uint8_t* slice, size_t length) {
+  uint8_t nalu_type = slice[4] & kNaluTypeMask;
+  switch (nalu_type) {
+    case kNaluSps:
+      RTC_CHECK(ParseSpsNalu(slice, length))
+          << "Failed to parse bitstream SPS.";
+      break;
+    case kNaluPps:
+      RTC_CHECK(ParsePpsNalu(slice, length))
+          << "Failed to parse bitstream PPS.";
+      break;
+    default:
+      RTC_CHECK(ParseNonParameterSetNalu(slice, length, nalu_type))
+          << "Failed to parse picture slice.";
+      break;
+  }
+}
+
+void H264BitstreamParser::ParseBitstream(const uint8_t* bitstream,
+                                         size_t length) {
+  RTC_CHECK_GE(length, 4u);
+  std::vector<size_t> slice_markers = FindNaluStartSequences(bitstream, length);
+  RTC_CHECK(!slice_markers.empty());
+  for (size_t i = 0; i < slice_markers.size() - 1; ++i) {
+    ParseSlice(bitstream + slice_markers[i],
+               slice_markers[i + 1] - slice_markers[i]);
+  }
+  // Parse the last slice.
+  ParseSlice(bitstream + slice_markers.back(), length - slice_markers.back());
+}
+
+bool H264BitstreamParser::GetLastSliceQp(int* qp) const {
+  if (!last_slice_qp_delta_parsed_)
+    return false;
+  *qp = 26 + pps_.pic_init_qp_minus26 + last_slice_qp_delta_;
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
new file mode 100644
index 0000000000..28276afb72
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace rtc {
+class BitBuffer;
+}
+
+namespace webrtc {
+
+// Stateful H264 bitstream parser (due to SPS/PPS). Used to parse out QP values
+// from the bitstream.
+// TODO(pbos): Unify with RTP SPS parsing and only use one H264 parser.
+// TODO(pbos): If/when this gets used on the receiver side CHECKs must be
+// removed and gracefully abort as we have no control over receive-side
+// bitstreams.
+class H264BitstreamParser {
+ public:
+  // Parse an additional chunk of H264 bitstream.
+  void ParseBitstream(const uint8_t* bitstream, size_t length);
+
+  // Get the last extracted QP value from the parsed bitstream.
+  bool GetLastSliceQp(int* qp) const;
+
+ private:
+  // Captured in SPS and used when parsing slice NALUs.
+  struct SpsState {
+    SpsState();
+
+    uint32_t delta_pic_order_always_zero_flag = 0;
+    uint32_t separate_colour_plane_flag = 0;
+    uint32_t frame_mbs_only_flag = 0;
+    uint32_t log2_max_frame_num_minus4 = 0;
+    uint32_t log2_max_pic_order_cnt_lsb_minus4 = 0;
+    uint32_t pic_order_cnt_type = 0;
+  };
+
+  struct PpsState {
+    PpsState();
+
+    bool bottom_field_pic_order_in_frame_present_flag = false;
+    bool weighted_pred_flag = false;
+    uint32_t weighted_bipred_idc = false;
+    uint32_t redundant_pic_cnt_present_flag = 0;
+    int pic_init_qp_minus26 = 0;
+  };
+
+  void ParseSlice(const uint8_t* slice, size_t length);
+  bool ParseSpsNalu(const uint8_t* sps_nalu, size_t length);
+  bool ParsePpsNalu(const uint8_t* pps_nalu, size_t length);
+  bool ParseNonParameterSetNalu(const uint8_t* source,
+                                size_t source_length,
+                                uint8_t nalu_type);
+
+  // SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices.
+  bool sps_parsed_ = false;
+  SpsState sps_;
+  bool pps_parsed_ = false;
+  PpsState pps_;
+
+  // Last parsed slice QP.
+  bool last_slice_qp_delta_parsed_ = false;
+  int32_t last_slice_qp_delta_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser_unittest.cc
new file mode 100644
index 0000000000..6c726c3120
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser_unittest.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+// SPS/PPS part of below chunk.
+uint8_t kH264SpsPps[] = {0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0x80, 0x20, 0xda,
+                         0x01, 0x40, 0x16, 0xe8, 0x06, 0xd0, 0xa1, 0x35, 0x00,
+                         0x00, 0x00, 0x01, 0x68, 0xce, 0x06, 0xe2};
+
+// Contains enough of the image slice to contain slice QP.
+uint8_t kH264BitstreamChunk[] = {
+    0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0x80, 0x20, 0xda, 0x01, 0x40, 0x16,
+    0xe8, 0x06, 0xd0, 0xa1, 0x35, 0x00, 0x00, 0x00, 0x01, 0x68, 0xce, 0x06,
+    0xe2, 0x00, 0x00, 0x00, 0x01, 0x65, 0xb8, 0x40, 0xf0, 0x8c, 0x03, 0xf2,
+    0x75, 0x67, 0xad, 0x41, 0x64, 0x24, 0x0e, 0xa0, 0xb2, 0x12, 0x1e, 0xf8,
+};
+
+// Contains enough of the image slice to contain slice QP.
+uint8_t kH264BitstreamNextImageSliceChunk[] = {
+    0x00, 0x00, 0x00, 0x01, 0x41, 0xe2, 0x01, 0x16, 0x0e, 0x3e, 0x2b, 0x86,
+};
+
+TEST(H264BitstreamParserTest, ReportsNoQpWithoutParsedSlices) {
+  H264BitstreamParser h264_parser;
+  int qp;
+  EXPECT_FALSE(h264_parser.GetLastSliceQp(&qp));
+}
+
+TEST(H264BitstreamParserTest, ReportsNoQpWithOnlyParsedPpsAndSpsSlices) {
+  H264BitstreamParser h264_parser;
+  h264_parser.ParseBitstream(kH264SpsPps, sizeof(kH264SpsPps));
+  int qp;
+  EXPECT_FALSE(h264_parser.GetLastSliceQp(&qp));
+}
+
+TEST(H264BitstreamParserTest, ReportsLastSliceQpForImageSlices) {
+  H264BitstreamParser h264_parser;
+  h264_parser.ParseBitstream(kH264BitstreamChunk, sizeof(kH264BitstreamChunk));
+  int qp;
+  ASSERT_TRUE(h264_parser.GetLastSliceQp(&qp));
+  EXPECT_EQ(35, qp);
+
+  // Parse an additional image slice.
+  h264_parser.ParseBitstream(kH264BitstreamNextImageSliceChunk,
+                             sizeof(kH264BitstreamNextImageSliceChunk));
+  ASSERT_TRUE(h264_parser.GetLastSliceQp(&qp));
+  EXPECT_EQ(37, qp);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
new file mode 100644
index 0000000000..2fb723e3f9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
@@ -0,0 +1,232 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
+
+#include "webrtc/base/bitbuffer.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/logging.h"
+
+#define RETURN_FALSE_ON_FAIL(x) \
+  if (!(x)) {                   \
+    return false;               \
+  }
+
+namespace webrtc {
+
+H264SpsParser::H264SpsParser(const uint8_t* sps, size_t byte_length)
+    : sps_(sps), byte_length_(byte_length), width_(), height_() {
+}
+
+bool H264SpsParser::Parse() {
+  // General note: this is based off the 02/2014 version of the H.264 standard.
+  // You can find it on this page:
+  // http://www.itu.int/rec/T-REC-H.264
+
+  const char* sps_bytes = reinterpret_cast<const char*>(sps_);
+  // First, parse out rbsp, which is basically the source buffer minus emulation
+  // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
+  // section 7.3.1 of the H.264 standard.
+  rtc::ByteBuffer rbsp_buffer;
+  for (size_t i = 0; i < byte_length_;) {
+    // Be careful about over/underflow here. byte_length_ - 3 can underflow, and
+    // i + 3 can overflow, but byte_length_ - i can't, because i < byte_length_
+    // above, and that expression will produce the number of bytes left in
+    // the stream including the byte at i.
+    if (byte_length_ - i >= 3 && sps_[i] == 0 && sps_[i + 1] == 0 &&
+        sps_[i + 2] == 3) {
+      // Two rbsp bytes + the emulation byte.
+      rbsp_buffer.WriteBytes(sps_bytes + i, 2);
+      i += 3;
+    } else {
+      // Single rbsp byte.
+      rbsp_buffer.WriteBytes(sps_bytes + i, 1);
+      i++;
+    }
+  }
+
+  // Now, we need to use a bit buffer to parse through the actual AVC SPS
+  // format. See Section 7.3.2.1.1 ("Sequence parameter set data syntax") of the
+  // H.264 standard for a complete description.
+  // Since we only care about resolution, we ignore the majority of fields, but
+  // we still have to actively parse through a lot of the data, since many of
+  // the fields have variable size.
+  // We're particularly interested in:
+  // chroma_format_idc -> affects crop units
+  // pic_{width,height}_* -> resolution of the frame in macroblocks (16x16).
+  // frame_crop_*_offset -> crop information
+  rtc::BitBuffer parser(reinterpret_cast<const uint8_t*>(rbsp_buffer.Data()),
+                        rbsp_buffer.Length());
+
+  // The golomb values we have to read, not just consume.
+  uint32_t golomb_ignored;
+
+  // separate_colour_plane_flag is optional (assumed 0), but has implications
+  // about the ChromaArrayType, which modifies how we treat crop coordinates.
+  uint32_t separate_colour_plane_flag = 0;
+  // chroma_format_idc will be ChromaArrayType if separate_colour_plane_flag is
+  // 0. It defaults to 1, when not specified.
+  uint32_t chroma_format_idc = 1;
+
+  // profile_idc: u(8). We need it to determine if we need to read/skip chroma
+  // formats.
+  uint8_t profile_idc;
+  RETURN_FALSE_ON_FAIL(parser.ReadUInt8(&profile_idc));
+  // constraint_set0_flag through constraint_set5_flag + reserved_zero_2bits
+  // 1 bit each for the flags + 2 bits = 8 bits = 1 byte.
+  RETURN_FALSE_ON_FAIL(parser.ConsumeBytes(1));
+  // level_idc: u(8)
+  RETURN_FALSE_ON_FAIL(parser.ConsumeBytes(1));
+  // seq_parameter_set_id: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // See if profile_idc has chroma format information.
+  if (profile_idc == 100 || profile_idc == 110 || profile_idc == 122 ||
+      profile_idc == 244 || profile_idc == 44 || profile_idc == 83 ||
+      profile_idc == 86 || profile_idc == 118 || profile_idc == 128 ||
+      profile_idc == 138 || profile_idc == 139 || profile_idc == 134) {
+    // chroma_format_idc: ue(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&chroma_format_idc));
+    if (chroma_format_idc == 3) {
+      // separate_colour_plane_flag: u(1)
+      RETURN_FALSE_ON_FAIL(parser.ReadBits(&separate_colour_plane_flag, 1));
+    }
+    // bit_depth_luma_minus8: ue(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+    // bit_depth_chroma_minus8: ue(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+    // qpprime_y_zero_transform_bypass_flag: u(1)
+    RETURN_FALSE_ON_FAIL(parser.ConsumeBits(1));
+    // seq_scaling_matrix_present_flag: u(1)
+    uint32_t seq_scaling_matrix_present_flag;
+    RETURN_FALSE_ON_FAIL(parser.ReadBits(&seq_scaling_matrix_present_flag, 1));
+    if (seq_scaling_matrix_present_flag) {
+      // seq_scaling_list_present_flags. Either 8 or 12, depending on
+      // chroma_format_idc.
+      uint32_t seq_scaling_list_present_flags;
+      if (chroma_format_idc != 3) {
+        RETURN_FALSE_ON_FAIL(
+            parser.ReadBits(&seq_scaling_list_present_flags, 8));
+      } else {
+        RETURN_FALSE_ON_FAIL(
+            parser.ReadBits(&seq_scaling_list_present_flags, 12));
+      }
+      // We don't support reading the sequence scaling list, and we don't really
+      // see/use them in practice, so we'll just reject the full sps if we see
+      // any provided.
+      if (seq_scaling_list_present_flags > 0) {
+        LOG(LS_WARNING) << "SPS contains scaling lists, which are unsupported.";
+        return false;
+      }
+    }
+  }
+  // log2_max_frame_num_minus4: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // pic_order_cnt_type: ue(v)
+  uint32_t pic_order_cnt_type;
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&pic_order_cnt_type));
+  if (pic_order_cnt_type == 0) {
+    // log2_max_pic_order_cnt_lsb_minus4: ue(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  } else if (pic_order_cnt_type == 1) {
+    // delta_pic_order_always_zero_flag: u(1)
+    RETURN_FALSE_ON_FAIL(parser.ConsumeBits(1));
+    // offset_for_non_ref_pic: se(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+    // offset_for_top_to_bottom_field: se(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+    // num_ref_frames_in_pic_order_cnt_cycle: ue(v)
+    uint32_t num_ref_frames_in_pic_order_cnt_cycle;
+    RETURN_FALSE_ON_FAIL(
+        parser.ReadExponentialGolomb(&num_ref_frames_in_pic_order_cnt_cycle));
+    for (size_t i = 0; i < num_ref_frames_in_pic_order_cnt_cycle; ++i) {
+      // offset_for_ref_frame[i]: se(v)
+      RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+    }
+  }
+  // max_num_ref_frames: ue(v)
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&golomb_ignored));
+  // gaps_in_frame_num_value_allowed_flag: u(1)
+  RETURN_FALSE_ON_FAIL(parser.ConsumeBits(1));
+  //
+  // IMPORTANT ONES! Now we're getting to resolution. First we read the pic
+  // width/height in macroblocks (16x16), which gives us the base resolution,
+  // and then we continue on until we hit the frame crop offsets, which are used
+  // to signify resolutions that aren't multiples of 16.
+  //
+  // pic_width_in_mbs_minus1: ue(v)
+  uint32_t pic_width_in_mbs_minus1;
+  RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&pic_width_in_mbs_minus1));
+  // pic_height_in_map_units_minus1: ue(v)
+  uint32_t pic_height_in_map_units_minus1;
+  RETURN_FALSE_ON_FAIL(
+      parser.ReadExponentialGolomb(&pic_height_in_map_units_minus1));
+  // frame_mbs_only_flag: u(1)
+  uint32_t frame_mbs_only_flag;
+  RETURN_FALSE_ON_FAIL(parser.ReadBits(&frame_mbs_only_flag, 1));
+  if (!frame_mbs_only_flag) {
+    // mb_adaptive_frame_field_flag: u(1)
+    RETURN_FALSE_ON_FAIL(parser.ConsumeBits(1));
+  }
+  // direct_8x8_inference_flag: u(1)
+  RETURN_FALSE_ON_FAIL(parser.ConsumeBits(1));
+  //
+  // MORE IMPORTANT ONES! Now we're at the frame crop information.
+  //
+  // frame_cropping_flag: u(1)
+  uint32_t frame_cropping_flag;
+  uint32_t frame_crop_left_offset = 0;
+  uint32_t frame_crop_right_offset = 0;
+  uint32_t frame_crop_top_offset = 0;
+  uint32_t frame_crop_bottom_offset = 0;
+  RETURN_FALSE_ON_FAIL(parser.ReadBits(&frame_cropping_flag, 1));
+  if (frame_cropping_flag) {
+    // frame_crop_{left, right, top, bottom}_offset: ue(v)
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&frame_crop_left_offset));
+    RETURN_FALSE_ON_FAIL(
+        parser.ReadExponentialGolomb(&frame_crop_right_offset));
+    RETURN_FALSE_ON_FAIL(parser.ReadExponentialGolomb(&frame_crop_top_offset));
+    RETURN_FALSE_ON_FAIL(
+        parser.ReadExponentialGolomb(&frame_crop_bottom_offset));
+  }
+
+  // Far enough! We don't use the rest of the SPS.
+
+  // Start with the resolution determined by the pic_width/pic_height fields.
+  int width = 16 * (pic_width_in_mbs_minus1 + 1);
+  int height =
+      16 * (2 - frame_mbs_only_flag) * (pic_height_in_map_units_minus1 + 1);
+
+  // Figure out the crop units in pixels. That's based on the chroma format's
+  // sampling, which is indicated by chroma_format_idc.
+  if (separate_colour_plane_flag || chroma_format_idc == 0) {
+    frame_crop_bottom_offset *= (2 - frame_mbs_only_flag);
+    frame_crop_top_offset *= (2 - frame_mbs_only_flag);
+  } else if (!separate_colour_plane_flag && chroma_format_idc > 0) {
+    // Width multipliers for formats 1 (4:2:0) and 2 (4:2:2).
+    if (chroma_format_idc == 1 || chroma_format_idc == 2) {
+      frame_crop_left_offset *= 2;
+      frame_crop_right_offset *= 2;
+    }
+    // Height multipliers for format 1 (4:2:0).
+    if (chroma_format_idc == 1) {
+      frame_crop_top_offset *= 2;
+      frame_crop_bottom_offset *= 2;
+    }
+  }
+  // Subtract the crop for each dimension.
+  width -= (frame_crop_left_offset + frame_crop_right_offset);
+  height -= (frame_crop_top_offset + frame_crop_bottom_offset);
+
+  width_ = width;
+  height_ = height;
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser.h
new file mode 100644
index 0000000000..c05ee67923
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_SPS_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_SPS_PARSER_H_
+
+#include "webrtc/base/common.h"
+
+namespace webrtc {
+
+// A class for parsing out sequence parameter set (SPS) data from an H264 NALU.
+// Currently, only resolution is read without being ignored.
+class H264SpsParser {
+ public:
+  H264SpsParser(const uint8_t* sps, size_t byte_length);
+  // Parses the SPS to completion. Returns true if the SPS was parsed correctly.
+  bool Parse();
+  uint16_t width() { return width_; }
+  uint16_t height() { return height_; }
+
+ private:
+  const uint8_t* const sps_;
+  const size_t byte_length_;
+
+  uint16_t width_;
+  uint16_t height_;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_SPS_PARSER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc
new file mode 100644
index 0000000000..7a7e3ed293
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/bitbuffer.h"
+
+namespace webrtc {
+
+// Example SPS can be generated with ffmpeg. Here's an example set of commands,
+// runnable on OS X:
+// 1) Generate a video, from the camera:
+// ffmpeg -f avfoundation -i "0" -video_size 640x360 camera.mov
+//
+// 2) Scale the video to the desired size:
+// ffmpeg -i camera.mov -vf scale=640x360 scaled.mov
+//
+// 3) Get just the H.264 bitstream in AnnexB:
+// ffmpeg -i scaled.mov -vcodec copy -vbsf h264_mp4toannexb -an out.h264
+//
+// 4) Open out.h264 and find the SPS, generally everything between the first
+// two start codes (0 0 0 1 or 0 0 1). The first byte should be 0x67,
+// which should be stripped out before being passed to the parser.
+
+static const size_t kSpsBufferMaxSize = 256;
+
+// Generates a fake SPS with basically everything empty but the width/height.
+// Pass in a buffer of at least kSpsBufferMaxSize.
+// The fake SPS that this generates also always has at least one emulation byte
+// at offset 2, since the first two bytes are always 0, and has a 0x3 as the
+// level_idc, to make sure the parser doesn't eat all 0x3 bytes.
+void GenerateFakeSps(uint16_t width, uint16_t height, uint8_t buffer[]) {
+  uint8_t rbsp[kSpsBufferMaxSize] = {0};
+  rtc::BitBufferWriter writer(rbsp, kSpsBufferMaxSize);
+  // Profile byte.
+  writer.WriteUInt8(0);
+  // Constraint sets and reserved zero bits.
+  writer.WriteUInt8(0);
+  // level_idc.
+  writer.WriteUInt8(0x3u);
+  // seq_paramter_set_id.
+  writer.WriteExponentialGolomb(0);
+  // Profile is not special, so we skip all the chroma format settings.
+
+  // Now some bit magic.
+  // log2_max_frame_num_minus4: ue(v). 0 is fine.
+  writer.WriteExponentialGolomb(0);
+  // pic_order_cnt_type: ue(v). 0 is the type we want.
+  writer.WriteExponentialGolomb(0);
+  // log2_max_pic_order_cnt_lsb_minus4: ue(v). 0 is fine.
+  writer.WriteExponentialGolomb(0);
+  // max_num_ref_frames: ue(v). 0 is fine.
+  writer.WriteExponentialGolomb(0);
+  // gaps_in_frame_num_value_allowed_flag: u(1).
+  writer.WriteBits(0, 1);
+  // Next are width/height. First, calculate the mbs/map_units versions.
+  uint16_t width_in_mbs_minus1 = (width + 15) / 16 - 1;
+
+  // For the height, we're going to define frame_mbs_only_flag, so we need to
+  // divide by 2. See the parser for the full calculation.
+  uint16_t height_in_map_units_minus1 = ((height + 15) / 16 - 1) / 2;
+  // Write each as ue(v).
+  writer.WriteExponentialGolomb(width_in_mbs_minus1);
+  writer.WriteExponentialGolomb(height_in_map_units_minus1);
+  // frame_mbs_only_flag: u(1). Needs to be false.
+  writer.WriteBits(0, 1);
+  // mb_adaptive_frame_field_flag: u(1).
+  writer.WriteBits(0, 1);
+  // direct_8x8_inferene_flag: u(1).
+  writer.WriteBits(0, 1);
+  // frame_cropping_flag: u(1). 1, so we can supply crop.
+  writer.WriteBits(1, 1);
+  // Now we write the left/right/top/bottom crop. For simplicity, we'll put all
+  // the crop at the left/top.
+  // We picked a 4:2:0 format, so the crops are 1/2 the pixel crop values.
+  // Left/right.
+  writer.WriteExponentialGolomb(((16 - (width % 16)) % 16) / 2);
+  writer.WriteExponentialGolomb(0);
+  // Top/bottom.
+  writer.WriteExponentialGolomb(((16 - (height % 16)) % 16) / 2);
+  writer.WriteExponentialGolomb(0);
+
+  // Get the number of bytes written (including the last partial byte).
+  size_t byte_count, bit_offset;
+  writer.GetCurrentOffset(&byte_count, &bit_offset);
+  if (bit_offset > 0) {
+    byte_count++;
+  }
+
+  // Now, we need to write the rbsp into bytes. To do that, we'll need to add
+  // emulation 0x03 bytes if there's ever a sequence of 00 00 01 or 00 00 00 01.
+  // To be simple, just add a 0x03 after every 0x00. Extra emulation doesn't
+  // hurt.
+  for (size_t i = 0; i < byte_count;) {
+    // The -3 is intentional; we never need to write an emulation byte if the 00
+    // is at the end.
+    if (i < byte_count - 3 && rbsp[i] == 0 && rbsp[i + 1] == 0) {
+      *buffer++ = rbsp[i];
+      *buffer++ = rbsp[i + 1];
+      *buffer++ = 0x3u;
+      i += 2;
+    } else {
+      *buffer++ = rbsp[i];
+      ++i;
+    }
+  }
+}
+
+TEST(H264SpsParserTest, TestSampleSPSHdLandscape) {
+  // SPS for a 1280x720 camera capture from ffmpeg on osx. Contains
+  // emulation bytes but no cropping.
+  const uint8_t buffer[] = {0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50, 0x05,
+                            0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0, 0x00,
+                            0x00, 0x2A, 0xE0, 0xF1, 0x83, 0x19, 0x60};
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
+  EXPECT_TRUE(parser.Parse());
+  EXPECT_EQ(1280u, parser.width());
+  EXPECT_EQ(720u, parser.height());
+}
+
+TEST(H264SpsParserTest, TestSampleSPSVgaLandscape) {
+  // SPS for a 640x360 camera capture from ffmpeg on osx. Contains emulation
+  // bytes and cropping (360 isn't divisible by 16).
+  const uint8_t buffer[] = {0x7A, 0x00, 0x1E, 0xBC, 0xD9, 0x40, 0xA0, 0x2F,
+                            0xF8, 0x98, 0x40, 0x00, 0x00, 0x03, 0x01, 0x80,
+                            0x00, 0x00, 0x56, 0x83, 0xC5, 0x8B, 0x65, 0x80};
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
+  EXPECT_TRUE(parser.Parse());
+  EXPECT_EQ(640u, parser.width());
+  EXPECT_EQ(360u, parser.height());
+}
+
+TEST(H264SpsParserTest, TestSampleSPSWeirdResolution) {
+  // SPS for a 200x400 camera capture from ffmpeg on osx. Horizontal and
+  // veritcal crop (neither dimension is divisible by 16).
+  const uint8_t buffer[] = {0x7A, 0x00, 0x0D, 0xBC, 0xD9, 0x43, 0x43, 0x3E,
+                            0x5E, 0x10, 0x00, 0x00, 0x03, 0x00, 0x60, 0x00,
+                            0x00, 0x15, 0xA0, 0xF1, 0x42, 0x99, 0x60};
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
+  EXPECT_TRUE(parser.Parse());
+  EXPECT_EQ(200u, parser.width());
+  EXPECT_EQ(400u, parser.height());
+}
+
+TEST(H264SpsParserTest, TestSyntheticSPSQvgaLandscape) {
+  uint8_t buffer[kSpsBufferMaxSize] = {0};
+  GenerateFakeSps(320u, 180u, buffer);
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
+  EXPECT_TRUE(parser.Parse());
+  EXPECT_EQ(320u, parser.width());
+  EXPECT_EQ(180u, parser.height());
+}
+
+TEST(H264SpsParserTest, TestSyntheticSPSWeirdResolution) {
+  uint8_t buffer[kSpsBufferMaxSize] = {0};
+  GenerateFakeSps(156u, 122u, buffer);
+  H264SpsParser parser = H264SpsParser(buffer, arraysize(buffer));
+  EXPECT_TRUE(parser.Parse());
+  EXPECT_EQ(156u, parser.width());
+  EXPECT_EQ(122u, parser.height());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
index f577cbaad1..011829cc6c 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 namespace webrtc {
 
@@ -23,7 +23,7 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy {
   MOCK_CONST_METHOD4(PayloadIsCompatible,
                      bool(const RtpUtility::Payload& payload,
                           const uint32_t frequency,
-                          const uint8_t channels,
+                          const size_t channels,
                           const uint32_t rate));
   MOCK_CONST_METHOD2(UpdatePayloadRate,
                      void(RtpUtility::Payload* payload, const uint32_t rate));
@@ -34,10 +34,10 @@ class MockRTPPayloadStrategy : public RTPPayloadStrategy {
       RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                            const int8_t payloadType,
                            const uint32_t frequency,
-                           const uint8_t channels,
+                           const size_t channels,
                            const uint32_t rate));
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_STRATEGY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index e5c4faff34..e19c31bfec 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -16,25 +16,26 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/transport.h"
 
-using namespace webrtc;
+namespace webrtc {
 
 const int kVideoNackListSize = 30;
-const int kTestId = 123;
 const uint32_t kTestSsrc = 3456;
 const uint16_t kTestSequenceNumber = 2345;
 const uint32_t kTestNumberOfPackets = 1350;
 const int kTestNumberOfRtxPackets = 149;
 const int kNumFrames = 30;
+const int kPayloadType = 123;
+const int kRtxPayloadType = 98;
 
-class VerifyingRtxReceiver : public NullRtpData
-{
+class VerifyingRtxReceiver : public NullRtpData {
  public:
   VerifyingRtxReceiver() {}
 
@@ -52,10 +53,10 @@ class VerifyingRtxReceiver : public NullRtpData
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
-  TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
+  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
-  void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override {
     rtp_rtcp_->SetRemoteSSRC(ssrc);
   }
 
@@ -84,9 +85,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
     rtp_receiver_ = receiver;
   }
 
-  void DropEveryNthPacket(int n) {
-    packet_loss_ = n;
-  }
+  void DropEveryNthPacket(int n) { packet_loss_ = n; }
 
   void DropConsecutivePackets(int start, int total) {
     consecutive_drop_start_ = start;
@@ -94,62 +93,64 @@ class RtxLoopBackTransport : public webrtc::Transport {
     packet_loss_ = 0;
   }
 
-  int SendPacket(int channel, const void* data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
     count_++;
-    const unsigned char* ptr = static_cast<const unsigned  char*>(data);
+    const unsigned char* ptr = static_cast<const unsigned char*>(data);
     uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
-    if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
+    if (ssrc == rtx_ssrc_)
+      count_rtx_ssrc_++;
     uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
-    expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
-        sequence_number);
-    if (packet_loss_ > 0) {
-      if ((count_ % packet_loss_) == 0) {
-        return static_cast<int>(len);
-      }
-    } else if (count_ >= consecutive_drop_start_ &&
-        count_ < consecutive_drop_end_) {
-      return static_cast<int>(len);
-    }
     size_t packet_length = len;
-    // TODO(pbos): Figure out why this needs to be initialized. Likely this
-    // is hiding a bug either in test setup or other code.
-    // https://code.google.com/p/webrtc/issues/detail?id=3183
-    uint8_t restored_packet[1500] = {0};
-    uint8_t* restored_packet_ptr = restored_packet;
+    uint8_t restored_packet[1500];
     RTPHeader header;
     rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     if (!parser->Parse(ptr, len, &header)) {
-      return -1;
+      return false;
+    }
+
+    if (!rtp_payload_registry_->IsRtx(header)) {
+      // Don't store retransmitted packets since we compare it to the list
+      // created by the receiver.
+      expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
+                                        sequence_number);
+    }
+    if (packet_loss_ > 0) {
+      if ((count_ % packet_loss_) == 0) {
+        return true;
+      }
+    } else if (count_ >= consecutive_drop_start_ &&
+               count_ < consecutive_drop_end_) {
+      return true;
     }
     if (rtp_payload_registry_->IsRtx(header)) {
       // Remove the RTX header and parse the original RTP header.
       EXPECT_TRUE(rtp_payload_registry_->RestoreOriginalPacket(
-          &restored_packet_ptr, ptr, &packet_length, rtp_receiver_->SSRC(),
-          header));
-      if (!parser->Parse(restored_packet_ptr, packet_length, &header)) {
-        return -1;
+          restored_packet, ptr, &packet_length, rtp_receiver_->SSRC(), header));
+      if (!parser->Parse(restored_packet, packet_length, &header)) {
+        return false;
       }
+      ptr = restored_packet;
+    } else {
+      rtp_payload_registry_->SetIncomingPayloadType(header);
     }
-    restored_packet_ptr += header.headerLength;
-    packet_length -= header.headerLength;
+
     PayloadUnion payload_specific;
     if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                     &payload_specific)) {
-      return -1;
+      return false;
     }
-    if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_ptr,
-                                          packet_length, payload_specific,
-                                          true)) {
-      return -1;
+    if (!rtp_receiver_->IncomingRtpPacket(header, ptr + header.headerLength,
+                                          packet_length - header.headerLength,
+                                          payload_specific, true)) {
+      return false;
     }
-    return static_cast<int>(len);
+    return true;
   }
 
-  int SendRTCPPacket(int channel, const void* data, size_t len) override {
-    if (module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0) {
-      return static_cast<int>(len);
-    }
-    return -1;
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    return module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0;
   }
   int count_;
   int packet_loss_;
@@ -176,7 +177,6 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
 
   void SetUp() override {
     RtpRtcp::Configuration configuration;
-    configuration.id = kTestId;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
@@ -187,11 +187,10 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
 
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
-        kTestId, &fake_clock, &receiver_, rtp_feedback_.get(),
-        &rtp_payload_registry_));
+        &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_));
 
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
-    rtp_rtcp_module_->SetRTCPStatus(kRtcpCompound);
+    rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
     rtp_receiver_->SetNACKStatus(kNackRtcp);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
@@ -203,15 +202,15 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
 
     VideoCodec video_codec;
     memset(&video_codec, 0, sizeof(video_codec));
-    video_codec.plType = 123;
+    video_codec.plType = kPayloadType;
     memcpy(video_codec.plName, "I420", 5);
 
     EXPECT_EQ(0, rtp_rtcp_module_->RegisterSendPayload(video_codec));
-    EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
-                                                       video_codec.plType,
-                                                       90000,
-                                                       0,
-                                                       video_codec.maxBitrate));
+    rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
+    EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(
+                     video_codec.plName, video_codec.plType, 90000, 0,
+                     video_codec.maxBitrate));
+    rtp_payload_registry_.SetRtxPayloadType(kRtxPayloadType, kPayloadType);
 
     for (size_t n = 0; n < payload_data_length; n++) {
       payload_data[n] = n % 10;
@@ -221,8 +220,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
   int BuildNackList(uint16_t* nack_list) {
     receiver_.sequence_numbers_.sort();
     std::list<uint16_t> missing_sequence_numbers;
-    std::list<uint16_t>::iterator it =
-        receiver_.sequence_numbers_.begin();
+    std::list<uint16_t>::iterator it = receiver_.sequence_numbers_.begin();
 
     while (it != receiver_.sequence_numbers_.end()) {
       uint16_t sequence_number_1 = *it;
@@ -230,15 +228,14 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
       if (it != receiver_.sequence_numbers_.end()) {
         uint16_t sequence_number_2 = *it;
         // Add all missing sequence numbers to list
-        for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2;
-            ++i) {
+        for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
           missing_sequence_numbers.push_back(i);
         }
       }
     }
     int n = 0;
     for (it = missing_sequence_numbers.begin();
-        it != missing_sequence_numbers.end(); ++it) {
+         it != missing_sequence_numbers.end(); ++it) {
       nack_list[n++] = (*it);
     }
     return n;
@@ -250,7 +247,9 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
               receiver_.sequence_numbers_.end(),
               std::back_inserter(received_sorted));
     received_sorted.sort();
-    return std::equal(received_sorted.begin(), received_sorted.end(),
+    return received_sorted.size() ==
+               transport_.expected_sequence_numbers_.size() &&
+           std::equal(received_sorted.begin(), received_sorted.end(),
                       transport_.expected_sequence_numbers_.begin());
   }
 
@@ -262,12 +261,9 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     uint32_t timestamp = 3000;
     uint16_t nack_list[kVideoNackListSize];
     for (int frame = 0; frame < kNumFrames; ++frame) {
-      EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
-                                                      123,
-                                                      timestamp,
-                                                      timestamp / 90,
-                                                      payload_data,
-                                                      payload_data_length));
+      EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(
+                       webrtc::kVideoFrameDelta, kPayloadType, timestamp,
+                       timestamp / 90, payload_data, payload_data_length));
       // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
       fake_clock.AdvanceTimeMilliseconds(5);
       int length = BuildNackList(nack_list);
@@ -290,7 +286,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
   rtc::scoped_ptr<TestRtpFeedback> rtp_feedback_;
   RtxLoopBackTransport transport_;
   VerifyingRtxReceiver receiver_;
-  uint8_t  payload_data[65000];
+  uint8_t payload_data[65000];
   size_t payload_data_length;
   SimulatedClock fake_clock;
 };
@@ -310,12 +306,9 @@ TEST_F(RtpRtcpRtxNackTest, LongNackList) {
   // Send 30 frames which at the default size is roughly what we need to get
   // enough packets.
   for (int frame = 0; frame < kNumFrames; ++frame) {
-    EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(webrtc::kVideoFrameDelta,
-                                                    123,
-                                                    timestamp,
-                                                    timestamp / 90,
-                                                    payload_data,
-                                                    payload_data_length));
+    EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(
+                     webrtc::kVideoFrameDelta, kPayloadType, timestamp,
+                     timestamp / 90, payload_data, payload_data_length));
     // Prepare next frame.
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
@@ -340,8 +333,10 @@ TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
-      *(receiver_.sequence_numbers_.rbegin()));
+            *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
new file mode 100644
index 0000000000..1def671f20
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+
+// After this many packets are added, adding additional packets will cause the
+// oldest packets to be pruned from the buffer.
+static const int kBufferSize = 100;
+
+namespace webrtc {
+
+PacketLossStats::PacketLossStats()
+    : single_loss_historic_count_(0),
+      multiple_loss_historic_event_count_(0),
+      multiple_loss_historic_packet_count_(0) {
+}
+
+void PacketLossStats::AddLostPacket(uint16_t sequence_number) {
+  // Detect sequence number wrap around.
+  if (!lost_packets_buffer_.empty() &&
+      static_cast<int>(*(lost_packets_buffer_.rbegin())) - sequence_number
+      > 0x8000) {
+    // The buffer contains large numbers and this is a small number.
+    lost_packets_wrapped_buffer_.insert(sequence_number);
+  } else {
+    lost_packets_buffer_.insert(sequence_number);
+  }
+  if (lost_packets_wrapped_buffer_.size() + lost_packets_buffer_.size()
+      > kBufferSize || (!lost_packets_wrapped_buffer_.empty() &&
+                        *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000)) {
+    PruneBuffer();
+  }
+}
+
+int PacketLossStats::GetSingleLossCount() const {
+  int single_loss_count, unused1, unused2;
+  ComputeLossCounts(&single_loss_count, &unused1, &unused2);
+  return single_loss_count;
+}
+
+int PacketLossStats::GetMultipleLossEventCount() const {
+  int event_count, unused1, unused2;
+  ComputeLossCounts(&unused1, &event_count, &unused2);
+  return event_count;
+}
+
+int PacketLossStats::GetMultipleLossPacketCount() const {
+  int packet_count, unused1, unused2;
+  ComputeLossCounts(&unused1, &unused2, &packet_count);
+  return packet_count;
+}
+
+void PacketLossStats::ComputeLossCounts(
+    int* out_single_loss_count,
+    int* out_multiple_loss_event_count,
+    int* out_multiple_loss_packet_count) const {
+  *out_single_loss_count = single_loss_historic_count_;
+  *out_multiple_loss_event_count = multiple_loss_historic_event_count_;
+  *out_multiple_loss_packet_count = multiple_loss_historic_packet_count_;
+  if (lost_packets_buffer_.empty()) {
+    RTC_DCHECK(lost_packets_wrapped_buffer_.empty());
+    return;
+  }
+  uint16_t last_num = 0;
+  int sequential_count = 0;
+  std::vector<const std::set<uint16_t>*> buffers;
+  buffers.push_back(&lost_packets_buffer_);
+  buffers.push_back(&lost_packets_wrapped_buffer_);
+  for (auto buffer : buffers) {
+    for (auto it = buffer->begin(); it != buffer->end(); ++it) {
+      uint16_t current_num = *it;
+      if (sequential_count > 0 && current_num != ((last_num + 1) & 0xFFFF)) {
+        if (sequential_count == 1) {
+          (*out_single_loss_count)++;
+        } else {
+          (*out_multiple_loss_event_count)++;
+          *out_multiple_loss_packet_count += sequential_count;
+        }
+        sequential_count = 0;
+      }
+      sequential_count++;
+      last_num = current_num;
+    }
+  }
+  if (sequential_count == 1) {
+    (*out_single_loss_count)++;
+  } else if (sequential_count > 1) {
+    (*out_multiple_loss_event_count)++;
+    *out_multiple_loss_packet_count += sequential_count;
+  }
+}
+
+void PacketLossStats::PruneBuffer() {
+  // Remove the oldest lost packet and any contiguous packets and move them
+  // into the historic counts.
+  auto it = lost_packets_buffer_.begin();
+  uint16_t last_removed = 0;
+  int remove_count = 0;
+  // Count adjacent packets and continue counting if it is wrap around by
+  // swapping in the wrapped buffer and letting our value wrap as well.
+  while (remove_count == 0 || (!lost_packets_buffer_.empty() &&
+                               *it == ((last_removed + 1) & 0xFFFF))) {
+    last_removed = *it;
+    remove_count++;
+    auto to_erase = it++;
+    lost_packets_buffer_.erase(to_erase);
+    if (lost_packets_buffer_.empty()) {
+      lost_packets_buffer_.swap(lost_packets_wrapped_buffer_);
+      it = lost_packets_buffer_.begin();
+    }
+  }
+  if (remove_count > 1) {
+    multiple_loss_historic_event_count_++;
+    multiple_loss_historic_packet_count_ += remove_count;
+  } else {
+    single_loss_historic_count_++;
+  }
+  // Continue pruning if the wrapped buffer is beyond a threshold and there are
+  // things left in the pre-wrapped buffer.
+  if (!lost_packets_wrapped_buffer_.empty() &&
+      *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000) {
+    PruneBuffer();
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
new file mode 100644
index 0000000000..2eab043c0d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
+
+#include <stdint.h>
+#include <set>
+
+namespace webrtc {
+
+// Keeps track of statistics of packet loss including whether losses are a
+// single packet or multiple packets in a row.
+class PacketLossStats {
+ public:
+  PacketLossStats();
+  ~PacketLossStats() {}
+
+  // Adds a lost packet to the stats by sequence number.
+  void AddLostPacket(uint16_t sequence_number);
+
+  // Queries the number of packets that were lost by themselves, no neighboring
+  // packets were lost.
+  int GetSingleLossCount() const;
+
+  // Queries the number of times that multiple packets with sequential numbers
+  // were lost. This is the number of events with more than one packet lost,
+  // regardless of the size of the event;
+  int GetMultipleLossEventCount() const;
+
+  // Queries the number of packets lost in multiple packet loss events. Combined
+  // with the event count, this can be used to determine the average event size.
+  int GetMultipleLossPacketCount() const;
+
+ private:
+  std::set<uint16_t> lost_packets_buffer_;
+  std::set<uint16_t> lost_packets_wrapped_buffer_;
+  int single_loss_historic_count_;
+  int multiple_loss_historic_event_count_;
+  int multiple_loss_historic_packet_count_;
+
+  void ComputeLossCounts(int* out_single_loss_count,
+                         int* out_multiple_loss_event_count,
+                         int* out_multiple_loss_packet_count) const;
+  void PruneBuffer();
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats_unittest.cc
new file mode 100644
index 0000000000..660628242d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/packet_loss_stats_unittest.cc
@@ -0,0 +1,197 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
+
+namespace webrtc {
+
+class PacketLossStatsTest : public ::testing::Test {
+ protected:
+  PacketLossStats stats_;
+};
+
+// Add a lost packet as every other packet, they should all count as single
+// losses.
+TEST_F(PacketLossStatsTest, EveryOtherPacket) {
+  for (int i = 0; i < 1000; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as every other packet, but such that the sequence numbers
+// will wrap around while they are being added.
+TEST_F(PacketLossStatsTest, EveryOtherPacketWrapped) {
+  for (int i = 65500; i < 66500; i += 2) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as every other packet, but such that the sequence numbers
+// will wrap around close to the very end, such that the buffer contains packets
+// on either side of the wrapping.
+TEST_F(PacketLossStatsTest, EveryOtherPacketWrappedAtEnd) {
+  for (int i = 64600; i < 65600; i += 2) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as the first three of every eight packets. Each set of
+// three should count as a multiple loss event and three multiple loss packets.
+TEST_F(PacketLossStatsTest, FirstThreeOfEight) {
+  for (int i = 0; i < 1000; ++i) {
+    if ((i & 7) < 3) {
+      stats_.AddLostPacket(i);
+    }
+  }
+  EXPECT_EQ(0, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as the first three of every eight packets such that the
+// sequence numbers wrap in the middle of adding them.
+TEST_F(PacketLossStatsTest, FirstThreeOfEightWrapped) {
+  for (int i = 65500; i < 66500; ++i) {
+    if ((i & 7) < 3) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(0, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as the first three of every eight packets such that the
+// sequence numbers wrap near the end of adding them and there are still numbers
+// in the buffer from before the wrapping.
+TEST_F(PacketLossStatsTest, FirstThreeOfEightWrappedAtEnd) {
+  for (int i = 64600; i < 65600; ++i) {
+    if ((i & 7) < 3) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(0, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets as the first three and the fifth of every eight packets. The
+// set of three should be multiple loss and the fifth should be single loss.
+TEST_F(PacketLossStatsTest, FirstThreeAndFifthOfEight) {
+  for (int i = 0; i < 1000; ++i) {
+    if ((i & 7) < 3 || (i & 7) == 4) {
+      stats_.AddLostPacket(i);
+    }
+  }
+  EXPECT_EQ(125, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets as the first three and the fifth of every eight packets such
+// that the sequence numbers wrap in the middle of adding them.
+TEST_F(PacketLossStatsTest, FirstThreeAndFifthOfEightWrapped) {
+  for (int i = 65500; i < 66500; ++i) {
+    if ((i & 7) < 3 || (i & 7) == 4) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(125, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets as the first three and the fifth of every eight packets such
+// that the sequence numbers wrap near the end of adding them and there are
+// packets from before the wrapping still in the buffer.
+TEST_F(PacketLossStatsTest, FirstThreeAndFifthOfEightWrappedAtEnd) {
+  for (int i = 64600; i < 65600; ++i) {
+    if ((i & 7) < 3 || (i & 7) == 4) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(125, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets such that there is a multiple loss event that continues
+// around the wrapping of sequence numbers.
+TEST_F(PacketLossStatsTest, MultipleLossEventWrapped) {
+  for (int i = 60000; i < 60500; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  for (int i = 65530; i < 65540; ++i) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  EXPECT_EQ(250, stats_.GetSingleLossCount());
+  EXPECT_EQ(1, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(10, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets such that there is a multiple loss event that continues
+// around the wrapping of sequence numbers and then is pushed out of the buffer.
+TEST_F(PacketLossStatsTest, MultipleLossEventWrappedPushedOut) {
+  for (int i = 60000; i < 60500; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  for (int i = 65530; i < 65540; ++i) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  for (int i = 1000; i < 1500; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(1, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(10, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets out of order and ensure that they still get counted
+// correctly as single or multiple loss events.
+TEST_F(PacketLossStatsTest, OutOfOrder) {
+  for (int i = 0; i < 1000; i += 10) {
+    stats_.AddLostPacket(i + 5);
+    stats_.AddLostPacket(i + 7);
+    stats_.AddLostPacket(i + 4);
+    stats_.AddLostPacket(i + 1);
+    stats_.AddLostPacket(i + 2);
+  }
+  EXPECT_EQ(100, stats_.GetSingleLossCount());
+  EXPECT_EQ(200, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(400, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets out of order and ensure that they still get counted
+// correctly as single or multiple loss events, and wrap in the middle of
+// adding.
+TEST_F(PacketLossStatsTest, OutOfOrderWrapped) {
+  for (int i = 65000; i < 66000; i += 10) {
+    stats_.AddLostPacket((i + 5) & 0xFFFF);
+    stats_.AddLostPacket((i + 7) & 0xFFFF);
+    stats_.AddLostPacket((i + 4) & 0xFFFF);
+    stats_.AddLostPacket((i + 1) & 0xFFFF);
+    stats_.AddLostPacket((i + 2) & 0xFFFF);
+  }
+  EXPECT_EQ(100, stats_.GetSingleLossCount());
+  EXPECT_EQ(200, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(400, stats_.GetMultipleLossPacketCount());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.cc
index a271a75263..6ec213ee43 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.cc
@@ -88,7 +88,6 @@ ProducerFec::ProducerFec(ForwardErrorCorrection* fec)
       media_packets_fec_(),
       fec_packets_(),
       num_frames_(0),
-      incomplete_frame_(false),
       num_first_partition_(0),
       minimum_media_packets_fec_(1),
       params_(),
@@ -125,9 +124,8 @@ RedPacket* ProducerFec::BuildRedPacket(const uint8_t* data_buffer,
                                        size_t payload_length,
                                        size_t rtp_header_length,
                                        int red_pl_type) {
-  RedPacket* red_packet = new RedPacket(payload_length +
-                                        kREDForFECHeaderLength +
-                                        rtp_header_length);
+  RedPacket* red_packet = new RedPacket(
+      payload_length + kREDForFECHeaderLength + rtp_header_length);
   int pl_type = data_buffer[1] & 0x7f;
   red_packet->CreateHeader(data_buffer, rtp_header_length,
                            red_pl_type, pl_type);
@@ -142,7 +140,7 @@ int ProducerFec::AddRtpPacketAndGenerateFec(const uint8_t* data_buffer,
   if (media_packets_fec_.empty()) {
     params_ = new_params_;
   }
-  incomplete_frame_ = true;
+  bool complete_frame = false;
   const bool marker_bit = (data_buffer[1] & kRtpMarkerBitMask) ? true : false;
   if (media_packets_fec_.size() < ForwardErrorCorrection::kMaxMediaPackets) {
     // Generic FEC can only protect up to kMaxMediaPackets packets.
@@ -153,13 +151,13 @@ int ProducerFec::AddRtpPacketAndGenerateFec(const uint8_t* data_buffer,
   }
   if (marker_bit) {
     ++num_frames_;
-    incomplete_frame_ = false;
+    complete_frame = true;
   }
   // Produce FEC over at most |params_.max_fec_frames| frames, or as soon as:
   // (1) the excess overhead (actual overhead - requested/target overhead) is
   // less than |kMaxExcessOverhead|, and
   // (2) at least |minimum_media_packets_fec_| media packets is reached.
-  if (!incomplete_frame_ &&
+  if (complete_frame &&
       (num_frames_ == params_.max_fec_frames ||
           (ExcessOverheadBelowMax() && MinimumMediaPacketsReached()))) {
     assert(num_first_partition_ <=
@@ -206,37 +204,43 @@ bool ProducerFec::MinimumMediaPacketsReached() {
 }
 
 bool ProducerFec::FecAvailable() const {
-  return (fec_packets_.size() > 0);
+  return !fec_packets_.empty();
 }
 
-RedPacket* ProducerFec::GetFecPacket(int red_pl_type,
-                                     int fec_pl_type,
-                                     uint16_t seq_num,
-                                     size_t rtp_header_length) {
-  if (fec_packets_.empty())
-    return NULL;
-  // Build FEC packet. The FEC packets in |fec_packets_| doesn't
-  // have RTP headers, so we're reusing the header from the last
-  // media packet.
-  ForwardErrorCorrection::Packet* packet_to_send = fec_packets_.front();
-  ForwardErrorCorrection::Packet* last_media_packet = media_packets_fec_.back();
-  RedPacket* return_packet = new RedPacket(packet_to_send->length +
-                                           kREDForFECHeaderLength +
-                                           rtp_header_length);
-  return_packet->CreateHeader(last_media_packet->data,
-                              rtp_header_length,
-                              red_pl_type,
-                              fec_pl_type);
-  return_packet->SetSeqNum(seq_num);
-  return_packet->ClearMarkerBit();
-  return_packet->AssignPayload(packet_to_send->data, packet_to_send->length);
-  fec_packets_.pop_front();
-  if (fec_packets_.empty()) {
-    // Done with all the FEC packets. Reset for next run.
-    DeletePackets();
-    num_frames_ = 0;
+size_t ProducerFec::NumAvailableFecPackets() const {
+  return fec_packets_.size();
+}
+
+std::vector<RedPacket*> ProducerFec::GetFecPackets(int red_pl_type,
+                                                   int fec_pl_type,
+                                                   uint16_t first_seq_num,
+                                                   size_t rtp_header_length) {
+  std::vector<RedPacket*> fec_packets;
+  fec_packets.reserve(fec_packets_.size());
+  uint16_t sequence_number = first_seq_num;
+  while (!fec_packets_.empty()) {
+    // Build FEC packet. The FEC packets in |fec_packets_| doesn't
+    // have RTP headers, so we're reusing the header from the last
+    // media packet.
+    ForwardErrorCorrection::Packet* packet_to_send = fec_packets_.front();
+    ForwardErrorCorrection::Packet* last_media_packet =
+        media_packets_fec_.back();
+
+    RedPacket* red_packet = new RedPacket(
+        packet_to_send->length + kREDForFECHeaderLength + rtp_header_length);
+    red_packet->CreateHeader(last_media_packet->data, rtp_header_length,
+                             red_pl_type, fec_pl_type);
+    red_packet->SetSeqNum(sequence_number++);
+    red_packet->ClearMarkerBit();
+    red_packet->AssignPayload(packet_to_send->data, packet_to_send->length);
+
+    fec_packets.push_back(red_packet);
+
+    fec_packets_.pop_front();
   }
-  return return_packet;
+  DeletePackets();
+  num_frames_ = 0;
+  return fec_packets;
 }
 
 int ProducerFec::Overhead() const {
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.h
index ec58bcf629..b2fdfeccac 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_PRODUCER_FEC_H_
 
 #include <list>
+#include <vector>
 
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
@@ -45,6 +46,7 @@ class ProducerFec {
   void SetFecParameters(const FecProtectionParams* params,
                         int max_fec_frames);
 
+  // The caller is expected to delete the memory when done.
   RedPacket* BuildRedPacket(const uint8_t* data_buffer,
                             size_t payload_length,
                             size_t rtp_header_length,
@@ -59,11 +61,14 @@ class ProducerFec {
   bool MinimumMediaPacketsReached();
 
   bool FecAvailable() const;
+  size_t NumAvailableFecPackets() const;
 
-  RedPacket* GetFecPacket(int red_pl_type,
-                          int fec_pl_type,
-                          uint16_t seq_num,
-                          size_t rtp_header_length);
+  // GetFecPackets allocates memory and creates FEC packets, but the caller is
+  // assumed to delete the memory when done with the packets.
+  std::vector<RedPacket*> GetFecPackets(int red_pl_type,
+                                        int fec_pl_type,
+                                        uint16_t first_seq_num,
+                                        size_t rtp_header_length);
 
  private:
   void DeletePackets();
@@ -72,7 +77,6 @@ class ProducerFec {
   std::list<ForwardErrorCorrection::Packet*> media_packets_fec_;
   std::list<ForwardErrorCorrection::Packet*> fec_packets_;
   int num_frames_;
-  bool incomplete_frame_;
   int num_first_partition_;
   int minimum_media_packets_fec_;
   FecProtectionParams params_;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
index f6d36d93d3..be4b453454 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
@@ -9,8 +9,10 @@
  */
 
 #include <list>
+#include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
@@ -54,6 +56,53 @@ class ProducerFecTest : public ::testing::Test {
   FrameGenerator* generator_;
 };
 
+// Verifies bug found via fuzzing, where a gap in the packet sequence caused us
+// to move past the end of the current FEC packet mask byte without moving to
+// the next byte. That likely caused us to repeatedly read from the same byte,
+// and if that byte didn't protect packets we would generate empty FEC.
+TEST_F(ProducerFecTest, NoEmptyFecWithSeqNumGaps) {
+  struct Packet {
+    size_t header_size;
+    size_t payload_size;
+    uint16_t seq_num;
+    bool marker_bit;
+  };
+  std::vector<Packet> protected_packets;
+  protected_packets.push_back({15, 3, 41, 0});
+  protected_packets.push_back({14, 1, 43, 0});
+  protected_packets.push_back({19, 0, 48, 0});
+  protected_packets.push_back({19, 0, 50, 0});
+  protected_packets.push_back({14, 3, 51, 0});
+  protected_packets.push_back({13, 8, 52, 0});
+  protected_packets.push_back({19, 2, 53, 0});
+  protected_packets.push_back({12, 3, 54, 0});
+  protected_packets.push_back({21, 0, 55, 0});
+  protected_packets.push_back({13, 3, 57, 1});
+  FecProtectionParams params = {117, 0, 3, kFecMaskBursty};
+  producer_->SetFecParameters(&params, 0);
+  uint8_t packet[28] = {0};
+  for (Packet p : protected_packets) {
+    if (p.marker_bit) {
+      packet[1] |= 0x80;
+    } else {
+      packet[1] &= ~0x80;
+    }
+    ByteWriter<uint16_t>::WriteBigEndian(&packet[2], p.seq_num);
+    producer_->AddRtpPacketAndGenerateFec(packet, p.payload_size,
+                                          p.header_size);
+    uint16_t num_fec_packets = producer_->NumAvailableFecPackets();
+    std::vector<RedPacket*> fec_packets;
+    if (num_fec_packets > 0) {
+      fec_packets =
+          producer_->GetFecPackets(kRedPayloadType, 99, 100, p.header_size);
+      EXPECT_EQ(num_fec_packets, fec_packets.size());
+    }
+    for (RedPacket* fec_packet : fec_packets) {
+      delete fec_packet;
+    }
+  }
+}
+
 TEST_F(ProducerFecTest, OneFrameFec) {
   // The number of media packets (|kNumPackets|), number of frames (one for
   // this test), and the protection factor (|params->fec_rate|) are set to make
@@ -77,19 +126,19 @@ TEST_F(ProducerFecTest, OneFrameFec) {
   }
   EXPECT_TRUE(producer_->FecAvailable());
   uint16_t seq_num = generator_->NextSeqNum();
-  RedPacket* packet = producer_->GetFecPacket(kRedPayloadType,
-                                              kFecPayloadType,
-                                              seq_num,
-                                              kRtpHeaderSize);
+  std::vector<RedPacket*> packets = producer_->GetFecPackets(kRedPayloadType,
+                                                             kFecPayloadType,
+                                                             seq_num,
+                                                             kRtpHeaderSize);
   EXPECT_FALSE(producer_->FecAvailable());
-  ASSERT_TRUE(packet != NULL);
+  ASSERT_EQ(1u, packets.size());
   VerifyHeader(seq_num, last_timestamp,
-               kRedPayloadType, kFecPayloadType, packet, false);
+               kRedPayloadType, kFecPayloadType, packets.front(), false);
   while (!rtp_packets.empty()) {
     delete rtp_packets.front();
     rtp_packets.pop_front();
   }
-  delete packet;
+  delete packets.front();
 }
 
 TEST_F(ProducerFecTest, TwoFrameFec) {
@@ -120,39 +169,36 @@ TEST_F(ProducerFecTest, TwoFrameFec) {
   }
   EXPECT_TRUE(producer_->FecAvailable());
   uint16_t seq_num = generator_->NextSeqNum();
-  RedPacket* packet = producer_->GetFecPacket(kRedPayloadType,
-                                              kFecPayloadType,
-                                              seq_num,
-                                              kRtpHeaderSize);
+  std::vector<RedPacket*> packets = producer_->GetFecPackets(kRedPayloadType,
+                                                             kFecPayloadType,
+                                                             seq_num,
+                                                             kRtpHeaderSize);
   EXPECT_FALSE(producer_->FecAvailable());
-  EXPECT_TRUE(packet != NULL);
-  VerifyHeader(seq_num, last_timestamp,
-               kRedPayloadType, kFecPayloadType, packet, false);
+  ASSERT_EQ(1u, packets.size());
+  VerifyHeader(seq_num, last_timestamp, kRedPayloadType, kFecPayloadType,
+               packets.front(), false);
   while (!rtp_packets.empty()) {
     delete rtp_packets.front();
     rtp_packets.pop_front();
   }
-  delete packet;
+  delete packets.front();
 }
 
 TEST_F(ProducerFecTest, BuildRedPacket) {
   generator_->NewFrame(1);
   RtpPacket* packet = generator_->NextPacket(0, 10);
-  RedPacket* red_packet = producer_->BuildRedPacket(packet->data,
-                                                    packet->length -
-                                                    kRtpHeaderSize,
-                                                    kRtpHeaderSize,
-                                                    kRedPayloadType);
+  rtc::scoped_ptr<RedPacket> red_packet(producer_->BuildRedPacket(
+      packet->data, packet->length - kRtpHeaderSize, kRtpHeaderSize,
+      kRedPayloadType));
   EXPECT_EQ(packet->length + 1, red_packet->length());
   VerifyHeader(packet->header.header.sequenceNumber,
                packet->header.header.timestamp,
                kRedPayloadType,
                packet->header.header.payloadType,
-               red_packet,
+               red_packet.get(),
                true);  // Marker bit set.
   for (int i = 0; i < 10; ++i)
     EXPECT_EQ(i, red_packet->data()[kRtpHeaderSize + 1 + i]);
-  delete red_packet;
   delete packet;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 3846558975..24f1e2c96e 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -14,8 +14,8 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -37,8 +37,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
       cumulative_loss_(0),
       jitter_q4_transmission_time_offset_(0),
       last_receive_time_ms_(0),
-      last_receive_time_secs_(0),
-      last_receive_time_frac_(0),
       last_received_timestamp_(0),
       last_received_transmission_time_offset_(0),
       received_seq_first_(0),
@@ -51,22 +49,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
       rtcp_callback_(rtcp_callback),
       rtp_callback_(rtp_callback) {}
 
-void StreamStatisticianImpl::ResetStatistics() {
-  CriticalSectionScoped cs(stream_lock_.get());
-  last_report_inorder_packets_ = 0;
-  last_report_old_packets_ = 0;
-  last_report_seq_max_ = 0;
-  last_reported_statistics_ = RtcpStatistics();
-  jitter_q4_ = 0;
-  cumulative_loss_ = 0;
-  jitter_q4_transmission_time_offset_ = 0;
-  received_seq_wraps_ = 0;
-  received_seq_max_ = 0;
-  received_seq_first_ = 0;
-  stored_sum_receive_counters_.Add(receive_counters_);
-  receive_counters_ = StreamDataCounters();
-}
-
 void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
                                             size_t packet_length,
                                             bool retransmitted) {
@@ -95,9 +77,7 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
   // are received, 4 will be ignored.
   if (in_order) {
     // Current time in samples.
-    uint32_t receive_time_secs;
-    uint32_t receive_time_frac;
-    clock_->CurrentNtp(receive_time_secs, receive_time_frac);
+    NtpTime receive_time(*clock_);
 
     // Wrong if we use RetransmitOfOldPacket.
     if (receive_counters_.transmitted.packets > 1 &&
@@ -113,11 +93,10 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
     if (header.timestamp != last_received_timestamp_ &&
         (receive_counters_.transmitted.packets -
          receive_counters_.retransmitted.packets) > 1) {
-      UpdateJitter(header, receive_time_secs, receive_time_frac);
+      UpdateJitter(header, receive_time);
     }
     last_received_timestamp_ = header.timestamp;
-    last_receive_time_secs_ = receive_time_secs;
-    last_receive_time_frac_ = receive_time_frac;
+    last_receive_time_ntp_ = receive_time;
     last_receive_time_ms_ = clock_->TimeInMilliseconds();
   }
 
@@ -129,14 +108,11 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
 }
 
 void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
-                                          uint32_t receive_time_secs,
-                                          uint32_t receive_time_frac) {
-  uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP(
-      receive_time_secs, receive_time_frac, header.payload_type_frequency);
+                                          NtpTime receive_time) {
+  uint32_t receive_time_rtp =
+      NtpToRtp(receive_time, header.payload_type_frequency);
   uint32_t last_receive_time_rtp =
-      RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_,
-                                      last_receive_time_frac_,
-                                      header.payload_type_frequency);
+      NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
       (header.timestamp - last_received_timestamp_);
 
@@ -283,6 +259,7 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
   stats.fraction_lost = local_fraction_lost;
 
   // We need a counter for cumulative loss too.
+  // TODO(danilchap): Ensure cumulative loss is below maximum value of 2^24.
   cumulative_loss_ += missing;
   stats.cumulative_lost = cumulative_loss_;
   stats.extended_max_sequence_number =
@@ -320,7 +297,6 @@ void StreamStatisticianImpl::GetReceiveStreamDataCounters(
     StreamDataCounters* data_counters) const {
   CriticalSectionScoped cs(stream_lock_.get());
   *data_counters = receive_counters_;
-  data_counters->Add(stored_sum_receive_counters_);
 }
 
 uint32_t StreamStatisticianImpl::BitrateReceived() const {
@@ -336,8 +312,8 @@ void StreamStatisticianImpl::ProcessBitrate() {
 void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
                                                 uint32_t* frac) const {
   CriticalSectionScoped cs(stream_lock_.get());
-  *secs = last_receive_time_secs_;
-  *frac = last_receive_time_frac_;
+  *secs = last_receive_time_ntp_.seconds();
+  *frac = last_receive_time_ntp_.fractions();
 }
 
 bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index cd41744d8d..025dcd42c7 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -11,13 +11,15 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 #include <algorithm>
+#include <map>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
 
 namespace webrtc {
 
@@ -36,7 +38,6 @@ class StreamStatisticianImpl : public StreamStatistician {
   void GetReceiveStreamDataCounters(
       StreamDataCounters* data_counters) const override;
   uint32_t BitrateReceived() const override;
-  void ResetStatistics() override;
   bool IsRetransmitOfOldPacket(const RTPHeader& header,
                                int64_t min_rtt) const override;
   bool IsPacketInOrder(uint16_t sequence_number) const override;
@@ -52,9 +53,7 @@ class StreamStatisticianImpl : public StreamStatistician {
  private:
   bool InOrderPacketInternal(uint16_t sequence_number) const;
   RtcpStatistics CalculateRtcpStatistics();
-  void UpdateJitter(const RTPHeader& header,
-                    uint32_t receive_time_secs,
-                    uint32_t receive_time_frac);
+  void UpdateJitter(const RTPHeader& header, NtpTime receive_time);
   void UpdateCounters(const RTPHeader& rtp_header,
                       size_t packet_length,
                       bool retransmitted);
@@ -73,8 +72,7 @@ class StreamStatisticianImpl : public StreamStatistician {
   uint32_t jitter_q4_transmission_time_offset_;
 
   int64_t last_receive_time_ms_;
-  uint32_t last_receive_time_secs_;
-  uint32_t last_receive_time_frac_;
+  NtpTime last_receive_time_ntp_;
   uint32_t last_received_timestamp_;
   int32_t last_received_transmission_time_offset_;
   uint16_t received_seq_first_;
@@ -85,9 +83,6 @@ class StreamStatisticianImpl : public StreamStatistician {
   size_t received_packet_overhead_;
   StreamDataCounters receive_counters_;
 
-  // Stored counter values. Includes sum of reset counter values for the stream.
-  StreamDataCounters stored_sum_receive_counters_;
-
   // Counter values when we sent the last report.
   uint32_t last_report_inorder_packets_;
   uint32_t last_report_old_packets_;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 8b25bcf615..c265c17c04 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -11,8 +11,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -143,12 +143,6 @@ TEST_F(ReceiveStatisticsTest, GetReceiveStreamDataCounters) {
   EXPECT_GT(counters.first_packet_time_ms, -1);
   EXPECT_EQ(1u, counters.transmitted.packets);
 
-  statistician->ResetStatistics();
-  // GetReceiveStreamDataCounters includes reset counter values.
-  statistician->GetReceiveStreamDataCounters(&counters);
-  EXPECT_GT(counters.first_packet_time_ms, -1);
-  EXPECT_EQ(1u, counters.transmitted.packets);
-
   receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
   statistician->GetReceiveStreamDataCounters(&counters);
   EXPECT_GT(counters.first_packet_time_ms, -1);
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
index 0c968bdb61..ccc15ec417 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
index 45817b01e6..797c7883a9 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
@@ -11,8 +11,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
 using ::testing::DoAll;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 3c1a9a3fc1..87c0259b3e 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -13,34 +13,30 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/null_transport.h"
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
 namespace {
 
-using namespace webrtc;
-
-
 class TestTransport : public Transport {
  public:
-  TestTransport(RTCPReceiver* rtcp_receiver) :
-    rtcp_receiver_(rtcp_receiver) {
-  }
+  explicit TestTransport(RTCPReceiver* rtcp_receiver)
+      : rtcp_receiver_(rtcp_receiver) {}
 
-  int SendPacket(int /*channel*/,
-                 const void* /*data*/,
-                 size_t /*len*/) override {
-    return -1;
+  bool SendRtp(const uint8_t* /*data*/,
+               size_t /*len*/,
+               const PacketOptions& options) override {
+    return false;
   }
-  int SendRTCPPacket(int /*channel*/,
-                     const void* packet,
-                     size_t packetLength) override {
-    RTCPUtility::RTCPParserV2 rtcpParser((uint8_t*)packet,
-                                         packetLength,
-                                         true); // Allow non-compound RTCP
+  bool SendRtcp(const uint8_t* packet, size_t packetLength) override {
+    RTCPUtility::RTCPParserV2 rtcpParser(packet, packetLength,
+                                         true);  // Allow non-compound RTCP
 
     EXPECT_TRUE(rtcpParser.IsValid());
     RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
@@ -51,28 +47,27 @@ class TestTransport : public Transport {
               rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb);
     EXPECT_EQ((uint32_t)1234,
               rtcpPacketInformation.receiverEstimatedMaxBitrate);
-    return static_cast<int>(packetLength);
+    return true;
   }
+
  private:
   RTCPReceiver* rtcp_receiver_;
 };
 
-
 class RtcpFormatRembTest : public ::testing::Test {
  protected:
-  static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
-
   RtcpFormatRembTest()
       : over_use_detector_options_(),
         system_clock_(Clock::GetRealTimeClock()),
+        dummy_rtp_rtcp_impl_(nullptr),
         receive_statistics_(ReceiveStatistics::Create(system_clock_)),
+        rtcp_sender_(nullptr),
+        rtcp_receiver_(nullptr),
+        test_transport_(nullptr),
         remote_bitrate_observer_(),
         remote_bitrate_estimator_(
-            RemoteBitrateEstimatorFactory().Create(
-                &remote_bitrate_observer_,
-                system_clock_,
-                kMimdControl,
-                kRemoteBitrateEstimatorMinBitrateBps)) {}
+            new RemoteBitrateEstimatorSingleStream(&remote_bitrate_observer_,
+                                                   system_clock_)) {}
   void SetUp() override;
   void TearDown() override;
 
@@ -83,24 +78,23 @@ class RtcpFormatRembTest : public ::testing::Test {
   RTCPSender* rtcp_sender_;
   RTCPReceiver* rtcp_receiver_;
   TestTransport* test_transport_;
+  test::NullTransport null_transport_;
   MockRemoteBitrateObserver remote_bitrate_observer_;
   rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
 };
 
 void RtcpFormatRembTest::SetUp() {
   RtpRtcp::Configuration configuration;
-  configuration.id = 0;
   configuration.audio = false;
   configuration.clock = system_clock_;
   configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
+  configuration.outgoing_transport = &null_transport_;
   dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
-  rtcp_sender_ = new RTCPSender(0, false, system_clock_,
-                                receive_statistics_.get(), NULL);
-  rtcp_receiver_ = new RTCPReceiver(0, system_clock_, NULL, NULL, NULL,
-                                    dummy_rtp_rtcp_impl_);
+  rtcp_receiver_ = new RTCPReceiver(system_clock_, false, nullptr, nullptr,
+                                    nullptr, nullptr, dummy_rtp_rtcp_impl_);
   test_transport_ = new TestTransport(rtcp_receiver_);
-
-  EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
+  rtcp_sender_ = new RTCPSender(false, system_clock_, receive_statistics_.get(),
+                                nullptr, test_transport_);
 }
 
 void RtcpFormatRembTest::TearDown() {
@@ -120,7 +114,7 @@ TEST_F(RtcpFormatRembTest, TestRembStatus) {
 
 TEST_F(RtcpFormatRembTest, TestNonCompund) {
   uint32_t SSRC = 456789;
-  rtcp_sender_->SetRTCPStatus(kRtcpNonCompound);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
   rtcp_sender_->SetREMBData(1234, std::vector<uint32_t>(1, SSRC));
   RTCPSender::FeedbackState feedback_state =
       dummy_rtp_rtcp_impl_->GetFeedbackState();
@@ -129,10 +123,11 @@ TEST_F(RtcpFormatRembTest, TestNonCompund) {
 
 TEST_F(RtcpFormatRembTest, TestCompund) {
   uint32_t SSRCs[2] = {456789, 98765};
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   rtcp_sender_->SetREMBData(1234, std::vector<uint32_t>(SSRCs, SSRCs + 2));
   RTCPSender::FeedbackState feedback_state =
       dummy_rtp_rtcp_impl_->GetFeedbackState();
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRemb));
 }
 }  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
index f44021ece7..eef2978371 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
@@ -10,46 +10,36 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 using webrtc::RTCPUtility::kBtDlrr;
 using webrtc::RTCPUtility::kBtReceiverReferenceTime;
 using webrtc::RTCPUtility::kBtVoipMetric;
 
 using webrtc::RTCPUtility::PT_APP;
-using webrtc::RTCPUtility::PT_BYE;
 using webrtc::RTCPUtility::PT_IJ;
 using webrtc::RTCPUtility::PT_PSFB;
-using webrtc::RTCPUtility::PT_RR;
 using webrtc::RTCPUtility::PT_RTPFB;
 using webrtc::RTCPUtility::PT_SDES;
 using webrtc::RTCPUtility::PT_SR;
 using webrtc::RTCPUtility::PT_XR;
 
 using webrtc::RTCPUtility::RTCPPacketAPP;
-using webrtc::RTCPUtility::RTCPPacketBYE;
 using webrtc::RTCPUtility::RTCPPacketPSFBAPP;
 using webrtc::RTCPUtility::RTCPPacketPSFBFIR;
 using webrtc::RTCPUtility::RTCPPacketPSFBFIRItem;
-using webrtc::RTCPUtility::RTCPPacketPSFBPLI;
 using webrtc::RTCPUtility::RTCPPacketPSFBREMBItem;
 using webrtc::RTCPUtility::RTCPPacketPSFBRPSI;
-using webrtc::RTCPUtility::RTCPPacketPSFBSLI;
-using webrtc::RTCPUtility::RTCPPacketPSFBSLIItem;
 using webrtc::RTCPUtility::RTCPPacketReportBlockItem;
-using webrtc::RTCPUtility::RTCPPacketRR;
 using webrtc::RTCPUtility::RTCPPacketRTPFBNACK;
 using webrtc::RTCPUtility::RTCPPacketRTPFBNACKItem;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBN;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBNItem;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBR;
-using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem;
 using webrtc::RTCPUtility::RTCPPacketSR;
 using webrtc::RTCPUtility::RTCPPacketXRDLRRReportBlockItem;
-using webrtc::RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem;
 using webrtc::RTCPUtility::RTCPPacketXR;
-using webrtc::RTCPUtility::RTCPPacketXRVOIPMetricItem;
 
 namespace webrtc {
 namespace rtcp {
@@ -91,34 +81,6 @@ void ComputeMantissaAnd6bitBase2Exponent(uint32_t input_base10,
   *mantissa = (input_base10 >> exponent);
 }
 
-size_t BlockToHeaderLength(size_t length_in_bytes) {
-  // Length in 32-bit words minus 1.
-  assert(length_in_bytes > 0);
-  assert(length_in_bytes % 4 == 0);
-  return (length_in_bytes / 4) - 1;
-}
-
-// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
-//
-// RTP header format.
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P| RC/FMT  |      PT       |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateHeader(uint8_t count_or_format,  // Depends on packet type.
-                  uint8_t packet_type,
-                  size_t length,
-                  uint8_t* buffer,
-                  size_t* pos) {
-  assert(length <= 0xffff);
-  const uint8_t kVersion = 2;
-  AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format);
-  AssignUWord8(buffer, pos, packet_type);
-  AssignUWord16(buffer, pos, length);
-}
-
 //  Sender report (SR) (RFC 3550).
 //   0                   1                   2                   3
 //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ -139,10 +101,8 @@ void CreateHeader(uint8_t count_or_format,  // Depends on packet type.
 //  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 
 void CreateSenderReport(const RTCPPacketSR& sr,
-                        size_t length,
                         uint8_t* buffer,
                         size_t* pos) {
-  CreateHeader(sr.NumberOfReportBlocks, PT_SR, length, buffer, pos);
   AssignUWord32(buffer, pos, sr.SenderSSRC);
   AssignUWord32(buffer, pos, sr.NTPMostSignificant);
   AssignUWord32(buffer, pos, sr.NTPLeastSignificant);
@@ -151,23 +111,6 @@ void CreateSenderReport(const RTCPPacketSR& sr,
   AssignUWord32(buffer, pos, sr.SenderOctetCount);
 }
 
-//  Receiver report (RR), header (RFC 3550).
-//
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P|    RC   |   PT=RR=201   |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                     SSRC of packet sender                     |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-
-void CreateReceiverReport(const RTCPPacketRR& rr,
-                          size_t length,
-                          uint8_t* buffer,
-                          size_t* pos) {
-  CreateHeader(rr.NumberOfReportBlocks, PT_RR, length, buffer, pos);
-  AssignUWord32(buffer, pos, rr.SenderSSRC);
-}
-
 //  Report block (RFC 3550).
 //
 //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ -185,44 +128,12 @@ void CreateReceiverReport(const RTCPPacketRR& rr,
 //  |                   delay since last SR (DLSR)                  |
 //  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 
-void CreateReportBlocks(const std::vector<RTCPPacketReportBlockItem>& blocks,
+void CreateReportBlocks(const std::vector<ReportBlock>& blocks,
                         uint8_t* buffer,
                         size_t* pos) {
-  for (std::vector<RTCPPacketReportBlockItem>::const_iterator
-       it = blocks.begin(); it != blocks.end(); ++it) {
-    AssignUWord32(buffer, pos, (*it).SSRC);
-    AssignUWord8(buffer, pos, (*it).FractionLost);
-    AssignUWord24(buffer, pos, (*it).CumulativeNumOfPacketsLost);
-    AssignUWord32(buffer, pos, (*it).ExtendedHighestSequenceNumber);
-    AssignUWord32(buffer, pos, (*it).Jitter);
-    AssignUWord32(buffer, pos, (*it).LastSR);
-    AssignUWord32(buffer, pos, (*it).DelayLastSR);
-  }
-}
-
-// Transmission Time Offsets in RTP Streams (RFC 5450).
-//
-//      0                   1                   2                   3
-//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     .                                                               .
-//     .                                                               .
-//     .                                                               .
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateIj(const std::vector<uint32_t>& ij_items,
-              uint8_t* buffer,
-              size_t* pos) {
-  size_t length = ij_items.size();
-  CreateHeader(length, PT_IJ, length, buffer, pos);
-  for (std::vector<uint32_t>::const_iterator it = ij_items.begin();
-       it != ij_items.end(); ++it) {
-    AssignUWord32(buffer, pos, *it);
+  for (const ReportBlock& block : blocks) {
+    block.Create(buffer + *pos);
+    *pos += ReportBlock::kLength;
   }
 }
 
@@ -253,10 +164,8 @@ void CreateIj(const std::vector<uint32_t>& ij_items,
 //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 void CreateSdes(const std::vector<Sdes::Chunk>& chunks,
-                size_t length,
                 uint8_t* buffer,
                 size_t* pos) {
-  CreateHeader(chunks.size(), PT_SDES, length, buffer, pos);
   const uint8_t kSdesItemType = 1;
   for (std::vector<Sdes::Chunk>::const_iterator it = chunks.begin();
        it != chunks.end(); ++it) {
@@ -270,142 +179,6 @@ void CreateSdes(const std::vector<Sdes::Chunk>& chunks,
   }
 }
 
-// Bye packet (BYE) (RFC 3550).
-//
-//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |                           SSRC/CSRC                           |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       :                              ...                              :
-//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// (opt) |     length    |               reason for leaving            ...
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateBye(const RTCPPacketBYE& bye,
-               const std::vector<uint32_t>& csrcs,
-               size_t length,
-               uint8_t* buffer,
-               size_t* pos) {
-  CreateHeader(length, PT_BYE, length, buffer, pos);
-  AssignUWord32(buffer, pos, bye.SenderSSRC);
-  for (std::vector<uint32_t>::const_iterator it = csrcs.begin();
-       it != csrcs.end(); ++it) {
-    AssignUWord32(buffer, pos, *it);
-  }
-}
-
-// Application-Defined packet (APP) (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P| subtype |   PT=APP=204  |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                           SSRC/CSRC                           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                          name (ASCII)                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   application-dependent data                ...
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateApp(const RTCPPacketAPP& app,
-               uint32_t ssrc,
-               size_t length,
-               uint8_t* buffer,
-               size_t* pos) {
-  CreateHeader(app.SubType, PT_APP, length, buffer, pos);
-  AssignUWord32(buffer, pos, ssrc);
-  AssignUWord32(buffer, pos, app.Name);
-  memcpy(buffer + *pos, app.Data, app.Size);
-  *pos += app.Size;
-}
-
-// RFC 4585: Feedback format.
-//
-// Common packet format:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |V=2|P|   FMT   |       PT      |          length               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of packet sender                        |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of media source                         |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   :            Feedback Control Information (FCI)                 :
-//   :
-//
-
-// Picture loss indication (PLI) (RFC 4585).
-//
-// FCI: no feedback control information.
-
-void CreatePli(const RTCPPacketPSFBPLI& pli,
-               size_t length,
-               uint8_t* buffer,
-               size_t* pos) {
-  const uint8_t kFmt = 1;
-  CreateHeader(kFmt, PT_PSFB, length, buffer, pos);
-  AssignUWord32(buffer, pos, pli.SenderSSRC);
-  AssignUWord32(buffer, pos, pli.MediaSSRC);
-}
-
-// Slice loss indication (SLI) (RFC 4585).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            First        |        Number           | PictureID |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateSli(const RTCPPacketPSFBSLI& sli,
-               const RTCPPacketPSFBSLIItem& sli_item,
-               size_t length,
-               uint8_t* buffer,
-               size_t* pos) {
-  const uint8_t kFmt = 2;
-  CreateHeader(kFmt, PT_PSFB, length, buffer, pos);
-  AssignUWord32(buffer, pos, sli.SenderSSRC);
-  AssignUWord32(buffer, pos, sli.MediaSSRC);
-
-  AssignUWord8(buffer, pos, sli_item.FirstMB >> 5);
-  AssignUWord8(buffer, pos, (sli_item.FirstMB << 3) +
-                            ((sli_item.NumberOfMB >> 10) & 0x07));
-  AssignUWord8(buffer, pos, sli_item.NumberOfMB >> 2);
-  AssignUWord8(buffer, pos, (sli_item.NumberOfMB << 6) + sli_item.PictureId);
-}
-
-// Generic NACK (RFC 4585).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            PID                |             BLP               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateNack(const RTCPPacketRTPFBNACK& nack,
-                const std::vector<RTCPPacketRTPFBNACKItem>& nack_fields,
-                size_t length,
-                uint8_t* buffer,
-                size_t* pos) {
-  const uint8_t kFmt = 1;
-  CreateHeader(kFmt, PT_RTPFB, length, buffer, pos);
-  AssignUWord32(buffer, pos, nack.SenderSSRC);
-  AssignUWord32(buffer, pos, nack.MediaSSRC);
-  for (std::vector<RTCPPacketRTPFBNACKItem>::const_iterator
-      it = nack_fields.begin(); it != nack_fields.end(); ++it) {
-    AssignUWord16(buffer, pos, (*it).PacketID);
-    AssignUWord16(buffer, pos, (*it).BitMask);
-  }
-}
-
 // Reference picture selection indication (RPSI) (RFC 4585).
 //
 // FCI:
@@ -420,13 +193,10 @@ void CreateNack(const RTCPPacketRTPFBNACK& nack,
 
 void CreateRpsi(const RTCPPacketPSFBRPSI& rpsi,
                 uint8_t padding_bytes,
-                size_t length,
                 uint8_t* buffer,
                 size_t* pos) {
   // Native bit string should be a multiple of 8 bits.
   assert(rpsi.NumberOfValidBits % 8 == 0);
-  const uint8_t kFmt = 3;
-  CreateHeader(kFmt, PT_PSFB, length, buffer, pos);
   AssignUWord32(buffer, pos, rpsi.SenderSSRC);
   AssignUWord32(buffer, pos, rpsi.MediaSSRC);
   AssignUWord8(buffer, pos, padding_bytes * 8);
@@ -451,11 +221,8 @@ void CreateRpsi(const RTCPPacketPSFBRPSI& rpsi,
 
 void CreateFir(const RTCPPacketPSFBFIR& fir,
                const RTCPPacketPSFBFIRItem& fir_item,
-               size_t length,
                uint8_t* buffer,
                size_t* pos) {
-  const uint8_t kFmt = 4;
-  CreateHeader(kFmt, PT_PSFB, length, buffer, pos);
   AssignUWord32(buffer, pos, fir.SenderSSRC);
   AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
   AssignUWord32(buffer, pos, fir_item.SSRC);
@@ -463,72 +230,6 @@ void CreateFir(const RTCPPacketPSFBFIR& fir,
   AssignUWord24(buffer, pos, 0);
 }
 
-void CreateTmmbrItem(const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
-                     uint8_t* buffer,
-                     size_t* pos) {
-  uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000;
-  uint32_t mantissa = 0;
-  uint8_t exp = 0;
-  ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
-
-  AssignUWord32(buffer, pos, tmmbr_item.SSRC);
-  AssignUWord8(buffer, pos, (exp << 2) + ((mantissa >> 15) & 0x03));
-  AssignUWord8(buffer, pos, mantissa >> 7);
-  AssignUWord8(buffer, pos, (mantissa << 1) +
-                            ((tmmbr_item.MeasuredOverhead >> 8) & 0x01));
-  AssignUWord8(buffer, pos, tmmbr_item.MeasuredOverhead);
-}
-
-// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateTmmbr(const RTCPPacketRTPFBTMMBR& tmmbr,
-                 const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
-                 size_t length,
-                 uint8_t* buffer,
-                 size_t* pos) {
-  const uint8_t kFmt = 3;
-  CreateHeader(kFmt, PT_RTPFB, length, buffer, pos);
-  AssignUWord32(buffer, pos, tmmbr.SenderSSRC);
-  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
-  CreateTmmbrItem(tmmbr_item, buffer, pos);
-}
-
-// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateTmmbn(const RTCPPacketRTPFBTMMBN& tmmbn,
-                 const std::vector<RTCPPacketRTPFBTMMBRItem>& tmmbn_items,
-                 size_t length,
-                 uint8_t* buffer,
-                 size_t* pos) {
-  const uint8_t kFmt = 4;
-  CreateHeader(kFmt, PT_RTPFB, length, buffer, pos);
-  AssignUWord32(buffer, pos, tmmbn.SenderSSRC);
-  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
-  for (uint8_t i = 0; i < tmmbn_items.size(); ++i) {
-    CreateTmmbrItem(tmmbn_items[i], buffer, pos);
-  }
-}
-
 // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
 //
 //    0                   1                   2                   3
@@ -550,15 +251,12 @@ void CreateTmmbn(const RTCPPacketRTPFBTMMBN& tmmbn,
 
 void CreateRemb(const RTCPPacketPSFBAPP& remb,
                 const RTCPPacketPSFBREMBItem& remb_item,
-                size_t length,
                 uint8_t* buffer,
                 size_t* pos) {
   uint32_t mantissa = 0;
   uint8_t exp = 0;
   ComputeMantissaAnd6bitBase2Exponent(remb_item.BitRate, 18, &mantissa, &exp);
 
-  const uint8_t kFmt = 15;
-  CreateHeader(kFmt, PT_PSFB, length, buffer, pos);
   AssignUWord32(buffer, pos, remb.SenderSSRC);
   AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
   AssignUWord8(buffer, pos, 'R');
@@ -589,137 +287,11 @@ void CreateRemb(const RTCPPacketPSFBAPP& remb,
 //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 void CreateXrHeader(const RTCPPacketXR& header,
-                    size_t length,
                     uint8_t* buffer,
                     size_t* pos) {
-  CreateHeader(0U, PT_XR, length, buffer, pos);
   AssignUWord32(buffer, pos, header.OriginatorSSRC);
 }
 
-void CreateXrBlockHeader(uint8_t block_type,
-                         uint16_t block_length,
-                         uint8_t* buffer,
-                         size_t* pos) {
-  AssignUWord8(buffer, pos, block_type);
-  AssignUWord8(buffer, pos, 0);
-  AssignUWord16(buffer, pos, block_length);
-}
-
-// Receiver Reference Time Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=4      |   reserved    |       block length = 2        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |              NTP timestamp, most significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |             NTP timestamp, least significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateRrtr(const std::vector<RTCPPacketXRReceiverReferenceTimeItem>& rrtrs,
-                uint8_t* buffer,
-                size_t* pos) {
-  const uint16_t kBlockLength = 2;
-  for (std::vector<RTCPPacketXRReceiverReferenceTimeItem>::const_iterator it =
-       rrtrs.begin(); it != rrtrs.end(); ++it) {
-    CreateXrBlockHeader(kBtReceiverReferenceTime, kBlockLength, buffer, pos);
-    AssignUWord32(buffer, pos, (*it).NTPMostSignificant);
-    AssignUWord32(buffer, pos, (*it).NTPLeastSignificant);
-  }
-}
-
-// DLRR Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=5      |   reserved    |         block length          |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_1 (SSRC of first receiver)               | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  |                         last RR (LRR)                         |   1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   delay since last RR (DLRR)                  |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_2 (SSRC of second receiver)              | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  :                               ...                             :   2
-
-void CreateDlrr(const std::vector<Xr::DlrrBlock>& dlrrs,
-                uint8_t* buffer,
-                size_t* pos) {
-  for (std::vector<Xr::DlrrBlock>::const_iterator it = dlrrs.begin();
-       it != dlrrs.end(); ++it) {
-    if ((*it).empty()) {
-      continue;
-    }
-    uint16_t block_length = 3 * (*it).size();
-    CreateXrBlockHeader(kBtDlrr, block_length, buffer, pos);
-    for (Xr::DlrrBlock::const_iterator it_block = (*it).begin();
-         it_block != (*it).end(); ++it_block) {
-      AssignUWord32(buffer, pos, (*it_block).SSRC);
-      AssignUWord32(buffer, pos, (*it_block).LastRR);
-      AssignUWord32(buffer, pos, (*it_block).DelayLastRR);
-    }
-  }
-}
-
-// VoIP Metrics Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=7      |   reserved    |       block length = 8        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                        SSRC of source                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   loss rate   | discard rate  | burst density |  gap density  |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |       burst duration          |         gap duration          |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     round trip delay          |       end system delay        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  | signal level  |  noise level  |     RERL      |     Gmin      |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   RX config   |   reserved    |          JB nominal           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |          JB maximum           |          JB abs max           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-void CreateVoipMetric(const std::vector<RTCPPacketXRVOIPMetricItem>& metrics,
-                      uint8_t* buffer,
-                      size_t* pos) {
-  const uint16_t kBlockLength = 8;
-  for (std::vector<RTCPPacketXRVOIPMetricItem>::const_iterator it =
-       metrics.begin(); it != metrics.end(); ++it) {
-    CreateXrBlockHeader(kBtVoipMetric, kBlockLength, buffer, pos);
-    AssignUWord32(buffer, pos, (*it).SSRC);
-    AssignUWord8(buffer, pos, (*it).lossRate);
-    AssignUWord8(buffer, pos, (*it).discardRate);
-    AssignUWord8(buffer, pos, (*it).burstDensity);
-    AssignUWord8(buffer, pos, (*it).gapDensity);
-    AssignUWord16(buffer, pos, (*it).burstDuration);
-    AssignUWord16(buffer, pos, (*it).gapDuration);
-    AssignUWord16(buffer, pos, (*it).roundTripDelay);
-    AssignUWord16(buffer, pos, (*it).endSystemDelay);
-    AssignUWord8(buffer, pos, (*it).signalLevel);
-    AssignUWord8(buffer, pos, (*it).noiseLevel);
-    AssignUWord8(buffer, pos, (*it).RERL);
-    AssignUWord8(buffer, pos, (*it).Gmin);
-    AssignUWord8(buffer, pos, (*it).Rfactor);
-    AssignUWord8(buffer, pos, (*it).extRfactor);
-    AssignUWord8(buffer, pos, (*it).MOSLQ);
-    AssignUWord8(buffer, pos, (*it).MOSCQ);
-    AssignUWord8(buffer, pos, (*it).RXconfig);
-    AssignUWord8(buffer, pos, 0);
-    AssignUWord16(buffer, pos, (*it).JBnominal);
-    AssignUWord16(buffer, pos, (*it).JBmax);
-    AssignUWord16(buffer, pos, (*it).JBabsMax);
-  }
-}
 }  // namespace
 
 void RtcpPacket::Append(RtcpPacket* packet) {
@@ -727,105 +299,140 @@ void RtcpPacket::Append(RtcpPacket* packet) {
   appended_packets_.push_back(packet);
 }
 
-RawPacket RtcpPacket::Build() const {
+rtc::scoped_ptr<RawPacket> RtcpPacket::Build() const {
   size_t length = 0;
-  uint8_t packet[IP_PACKET_SIZE];
-  CreateAndAddAppended(packet, &length, IP_PACKET_SIZE);
-  return RawPacket(packet, length);
+  rtc::scoped_ptr<RawPacket> packet(new RawPacket(IP_PACKET_SIZE));
+
+  class PacketVerifier : public PacketReadyCallback {
+   public:
+    explicit PacketVerifier(RawPacket* packet)
+        : called_(false), packet_(packet) {}
+    virtual ~PacketVerifier() {}
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      RTC_CHECK(!called_) << "Fragmentation not supported.";
+      called_ = true;
+      packet_->SetLength(length);
+    }
+
+   private:
+    bool called_;
+    RawPacket* const packet_;
+  } verifier(packet.get());
+  CreateAndAddAppended(packet->MutableBuffer(), &length, packet->BufferLength(),
+                       &verifier);
+  OnBufferFull(packet->MutableBuffer(), &length, &verifier);
+  return packet;
 }
 
-void RtcpPacket::Build(uint8_t* packet,
-                       size_t* length,
-                       size_t max_length) const {
-  *length = 0;
-  CreateAndAddAppended(packet, length, max_length);
+bool RtcpPacket::Build(PacketReadyCallback* callback) const {
+  uint8_t buffer[IP_PACKET_SIZE];
+  return BuildExternalBuffer(buffer, IP_PACKET_SIZE, callback);
 }
 
-void RtcpPacket::CreateAndAddAppended(uint8_t* packet,
-                                      size_t* length,
-                                      size_t max_length) const {
-  Create(packet, length, max_length);
-  for (std::vector<RtcpPacket*>::const_iterator it = appended_packets_.begin();
-      it != appended_packets_.end(); ++it) {
-    (*it)->CreateAndAddAppended(packet, length, max_length);
+bool RtcpPacket::BuildExternalBuffer(uint8_t* buffer,
+                                     size_t max_length,
+                                     PacketReadyCallback* callback) const {
+  size_t index = 0;
+  if (!CreateAndAddAppended(buffer, &index, max_length, callback))
+    return false;
+  return OnBufferFull(buffer, &index, callback);
+}
+
+bool RtcpPacket::CreateAndAddAppended(uint8_t* packet,
+                                      size_t* index,
+                                      size_t max_length,
+                                      PacketReadyCallback* callback) const {
+  if (!Create(packet, index, max_length, callback))
+    return false;
+  for (RtcpPacket* appended : appended_packets_) {
+    if (!appended->CreateAndAddAppended(packet, index, max_length, callback))
+      return false;
   }
+  return true;
 }
 
-void Empty::Create(uint8_t* packet, size_t* length, size_t max_length) const {
+bool RtcpPacket::OnBufferFull(uint8_t* packet,
+                              size_t* index,
+                              RtcpPacket::PacketReadyCallback* callback) const {
+  if (*index == 0)
+    return false;
+  callback->OnPacketReady(packet, *index);
+  *index = 0;
+  return true;
 }
 
-void SenderReport::Create(uint8_t* packet,
-                          size_t* length,
-                          size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
+size_t RtcpPacket::HeaderLength() const {
+  size_t length_in_bytes = BlockLength();
+  // Length in 32-bit words minus 1.
+  assert(length_in_bytes > 0);
+  return ((length_in_bytes + 3) / 4) - 1;
+}
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTP header format.
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P| RC/FMT  |      PT       |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void RtcpPacket::CreateHeader(
+    uint8_t count_or_format,  // Depends on packet type.
+    uint8_t packet_type,
+    size_t length,
+    uint8_t* buffer,
+    size_t* pos) {
+  assert(length <= 0xffff);
+  const uint8_t kVersion = 2;
+  AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format);
+  AssignUWord8(buffer, pos, packet_type);
+  AssignUWord16(buffer, pos, length);
+}
+
+bool SenderReport::Create(uint8_t* packet,
+                          size_t* index,
+                          size_t max_length,
+                          RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
   }
-  CreateSenderReport(sr_, BlockToHeaderLength(BlockLength()), packet, length);
-  CreateReportBlocks(report_blocks_, packet, length);
+  CreateHeader(sr_.NumberOfReportBlocks, PT_SR, HeaderLength(), packet, index);
+  CreateSenderReport(sr_, packet, index);
+  CreateReportBlocks(report_blocks_, packet, index);
+  return true;
 }
 
-void SenderReport::WithReportBlock(ReportBlock* block) {
-  assert(block);
+bool SenderReport::WithReportBlock(const ReportBlock& block) {
   if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
     LOG(LS_WARNING) << "Max report blocks reached.";
-    return;
+    return false;
   }
-  report_blocks_.push_back(block->report_block_);
+  report_blocks_.push_back(block);
   sr_.NumberOfReportBlocks = report_blocks_.size();
+  return true;
 }
 
-void ReceiverReport::Create(uint8_t* packet,
-                            size_t* length,
-                            size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateReceiverReport(rr_, BlockToHeaderLength(BlockLength()), packet, length);
-  CreateReportBlocks(report_blocks_, packet, length);
-}
-
-void ReceiverReport::WithReportBlock(ReportBlock* block) {
-  assert(block);
-  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
-    LOG(LS_WARNING) << "Max report blocks reached.";
-    return;
-  }
-  report_blocks_.push_back(block->report_block_);
-  rr_.NumberOfReportBlocks = report_blocks_.size();
-}
-
-void Ij::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateIj(ij_items_, packet, length);
-}
-
-void Ij::WithJitterItem(uint32_t jitter) {
-  if (ij_items_.size() >= kMaxNumberOfIjItems) {
-    LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
-    return;
-  }
-  ij_items_.push_back(jitter);
-}
-
-void Sdes::Create(uint8_t* packet, size_t* length, size_t max_length) const {
+bool Sdes::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  RtcpPacket::PacketReadyCallback* callback) const {
   assert(!chunks_.empty());
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
   }
-  CreateSdes(chunks_, BlockToHeaderLength(BlockLength()), packet, length);
+  CreateHeader(chunks_.size(), PT_SDES, HeaderLength(), packet, index);
+  CreateSdes(chunks_, packet, index);
+  return true;
 }
 
-void Sdes::WithCName(uint32_t ssrc, std::string cname) {
+bool Sdes::WithCName(uint32_t ssrc, const std::string& cname) {
   assert(cname.length() <= 0xff);
   if (chunks_.size() >= kMaxNumberOfChunks) {
     LOG(LS_WARNING) << "Max SDES chunks reached.";
-    return;
+    return false;
   }
   // In each chunk, the list of items must be terminated by one or more null
   // octets. The next chunk must start on a 32-bit boundary.
@@ -836,6 +443,7 @@ void Sdes::WithCName(uint32_t ssrc, std::string cname) {
   chunk.name = cname;
   chunk.null_octets = null_octets;
   chunks_.push_back(chunk);
+  return true;
 }
 
 size_t Sdes::BlockLength() const {
@@ -843,97 +451,25 @@ size_t Sdes::BlockLength() const {
   // Chunk:
   // SSRC/CSRC (4 bytes) | CNAME (1 byte) | length (1 byte) | name | padding.
   size_t length = kHeaderLength;
-  for (std::vector<Chunk>::const_iterator it = chunks_.begin();
-       it != chunks_.end(); ++it) {
-    length += 6 + (*it).name.length() + (*it).null_octets;
-  }
+  for (const Chunk& chunk : chunks_)
+    length += 6 + chunk.name.length() + chunk.null_octets;
   assert(length % 4 == 0);
   return length;
 }
 
-void Bye::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateBye(bye_, csrcs_, BlockToHeaderLength(BlockLength()), packet, length);
-}
-
-void Bye::WithCsrc(uint32_t csrc) {
-  if (csrcs_.size() >= kMaxNumberOfCsrcs) {
-    LOG(LS_WARNING) << "Max CSRC size reached.";
-    return;
-  }
-  csrcs_.push_back(csrc);
-}
-
-void App::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateApp(app_, ssrc_, BlockToHeaderLength(BlockLength()), packet, length);
-}
-
-void Pli::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreatePli(pli_, BlockToHeaderLength(BlockLength()), packet, length);
-}
-
-void Sli::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateSli(sli_, sli_item_, BlockToHeaderLength(BlockLength()), packet,
-            length);
-}
-
-void Nack::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  assert(!nack_fields_.empty());
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateNack(nack_, nack_fields_, BlockToHeaderLength(BlockLength()), packet,
-             length);
-}
-
-void Nack::WithList(const uint16_t* nack_list, int length) {
-  assert(nack_list);
-  assert(nack_fields_.empty());
-  int i = 0;
-  while (i < length) {
-    uint16_t pid = nack_list[i++];
-    // Bitmask specifies losses in any of the 16 packets following the pid.
-    uint16_t bitmask = 0;
-    while (i < length) {
-      int shift = static_cast<uint16_t>(nack_list[i] - pid) - 1;
-      if (shift >= 0 && shift <= 15) {
-        bitmask |= (1 << shift);
-        ++i;
-      } else {
-        break;
-      }
-    }
-    RTCPUtility::RTCPPacketRTPFBNACKItem item;
-    item.PacketID = pid;
-    item.BitMask = bitmask;
-    nack_fields_.push_back(item);
-  }
-}
-
-void Rpsi::Create(uint8_t* packet, size_t* length, size_t max_length) const {
+bool Rpsi::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  RtcpPacket::PacketReadyCallback* callback) const {
   assert(rpsi_.NumberOfValidBits > 0);
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
   }
-  CreateRpsi(rpsi_, padding_bytes_, BlockToHeaderLength(BlockLength()), packet,
-             length);
+  const uint8_t kFmt = 3;
+  CreateHeader(kFmt, PT_PSFB, HeaderLength(), packet, index);
+  CreateRpsi(rpsi_, padding_bytes_, packet, index);
+  return true;
 }
 
 void Rpsi::WithPictureId(uint64_t picture_id) {
@@ -963,22 +499,32 @@ void Rpsi::WithPictureId(uint64_t picture_id) {
   }
 }
 
-void Fir::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
+bool Fir::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
   }
-  CreateFir(fir_, fir_item_, BlockToHeaderLength(BlockLength()), packet,
-            length);
+  const uint8_t kFmt = 4;
+  CreateHeader(kFmt, PT_PSFB, HeaderLength(), packet, index);
+  CreateFir(fir_, fir_item_, packet, index);
+  return true;
 }
 
-void Remb::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
+bool Remb::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
   }
-  CreateRemb(remb_, remb_item_, BlockToHeaderLength(BlockLength()), packet,
-             length);
+  const uint8_t kFmt = 15;
+  CreateHeader(kFmt, PT_PSFB, HeaderLength(), packet, index);
+  CreateRemb(remb_, remb_item_, packet, index);
+  return true;
 }
 
 void Remb::AppliesTo(uint32_t ssrc) {
@@ -989,101 +535,99 @@ void Remb::AppliesTo(uint32_t ssrc) {
   remb_item_.SSRCs[remb_item_.NumberOfSSRCs++] = ssrc;
 }
 
-void Tmmbr::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
+bool Xr::Create(uint8_t* packet,
+                size_t* index,
+                size_t max_length,
+                RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
   }
-  CreateTmmbr(tmmbr_, tmmbr_item_, BlockToHeaderLength(BlockLength()), packet,
-              length);
+  CreateHeader(0U, PT_XR, HeaderLength(), packet, index);
+  CreateXrHeader(xr_header_, packet, index);
+  for (const Rrtr& block : rrtr_blocks_) {
+    block.Create(packet + *index);
+    *index += Rrtr::kLength;
+  }
+  for (const Dlrr& block : dlrr_blocks_) {
+    block.Create(packet + *index);
+    *index += block.BlockLength();
+  }
+  for (const VoipMetric& block : voip_metric_blocks_) {
+    block.Create(packet + *index);
+    *index += VoipMetric::kLength;
+  }
+  return true;
 }
 
-void Tmmbn::WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead) {
-  assert(overhead <= 0x1ff);
-  if (tmmbn_items_.size() >= kMaxNumberOfTmmbrs) {
-    LOG(LS_WARNING) << "Max TMMBN size reached.";
-    return;
-  }
-  RTCPPacketRTPFBTMMBRItem tmmbn_item;
-  tmmbn_item.SSRC = ssrc;
-  tmmbn_item.MaxTotalMediaBitRate = bitrate_kbps;
-  tmmbn_item.MeasuredOverhead = overhead;
-  tmmbn_items_.push_back(tmmbn_item);
-}
-
-void Tmmbn::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateTmmbn(tmmbn_, tmmbn_items_, BlockToHeaderLength(BlockLength()), packet,
-              length);
-}
-
-void Xr::Create(uint8_t* packet, size_t* length, size_t max_length) const {
-  if (*length + BlockLength() > max_length) {
-    LOG(LS_WARNING) << "Max packet size reached.";
-    return;
-  }
-  CreateXrHeader(xr_header_, BlockToHeaderLength(BlockLength()), packet,
-                 length);
-  CreateRrtr(rrtr_blocks_, packet, length);
-  CreateDlrr(dlrr_blocks_, packet, length);
-  CreateVoipMetric(voip_metric_blocks_, packet, length);
-}
-
-void Xr::WithRrtr(Rrtr* rrtr) {
-  assert(rrtr);
+bool Xr::WithRrtr(Rrtr* rrtr) {
+  RTC_DCHECK(rrtr);
   if (rrtr_blocks_.size() >= kMaxNumberOfRrtrBlocks) {
     LOG(LS_WARNING) << "Max RRTR blocks reached.";
-    return;
+    return false;
   }
-  rrtr_blocks_.push_back(rrtr->rrtr_block_);
+  rrtr_blocks_.push_back(*rrtr);
+  return true;
 }
 
-void Xr::WithDlrr(Dlrr* dlrr) {
-  assert(dlrr);
+bool Xr::WithDlrr(Dlrr* dlrr) {
+  RTC_DCHECK(dlrr);
   if (dlrr_blocks_.size() >= kMaxNumberOfDlrrBlocks) {
     LOG(LS_WARNING) << "Max DLRR blocks reached.";
-    return;
+    return false;
   }
-  dlrr_blocks_.push_back(dlrr->dlrr_block_);
+  dlrr_blocks_.push_back(*dlrr);
+  return true;
 }
 
-void Xr::WithVoipMetric(VoipMetric* voip_metric) {
+bool Xr::WithVoipMetric(VoipMetric* voip_metric) {
   assert(voip_metric);
   if (voip_metric_blocks_.size() >= kMaxNumberOfVoipMetricBlocks) {
     LOG(LS_WARNING) << "Max Voip Metric blocks reached.";
-    return;
+    return false;
   }
-  voip_metric_blocks_.push_back(voip_metric->metric_);
+  voip_metric_blocks_.push_back(*voip_metric);
+  return true;
 }
 
 size_t Xr::DlrrLength() const {
-  const size_t kBlockHeaderLen = 4;
-  const size_t kSubBlockLen = 12;
   size_t length = 0;
-  for (std::vector<DlrrBlock>::const_iterator it = dlrr_blocks_.begin();
-       it != dlrr_blocks_.end(); ++it) {
-    if (!(*it).empty()) {
-      length += kBlockHeaderLen + kSubBlockLen * (*it).size();
-    }
+  for (const Dlrr& block : dlrr_blocks_) {
+    length += block.BlockLength();
   }
   return length;
 }
 
-void Dlrr::WithDlrrItem(uint32_t ssrc,
-                        uint32_t last_rr,
-                        uint32_t delay_last_rr) {
-  if (dlrr_block_.size() >= kMaxNumberOfDlrrItems) {
-    LOG(LS_WARNING) << "Max DLRR items reached.";
-    return;
-  }
-  RTCPPacketXRDLRRReportBlockItem dlrr;
-  dlrr.SSRC = ssrc;
-  dlrr.LastRR = last_rr;
-  dlrr.DelayLastRR = delay_last_rr;
-  dlrr_block_.push_back(dlrr);
+RawPacket::RawPacket(size_t buffer_length)
+    : buffer_length_(buffer_length), length_(0) {
+  buffer_.reset(new uint8_t[buffer_length]);
+}
+
+RawPacket::RawPacket(const uint8_t* packet, size_t packet_length)
+    : buffer_length_(packet_length), length_(packet_length) {
+  buffer_.reset(new uint8_t[packet_length]);
+  memcpy(buffer_.get(), packet, packet_length);
+}
+
+const uint8_t* RawPacket::Buffer() const {
+  return buffer_.get();
+}
+
+uint8_t* RawPacket::MutableBuffer() {
+  return buffer_.get();
+}
+
+size_t RawPacket::BufferLength() const {
+  return buffer_length_;
+}
+
+size_t RawPacket::Length() const {
+  return length_;
+}
+
+void RawPacket::SetLength(size_t length) {
+  assert(length <= buffer_length_);
+  length_ = length;
 }
 
 }  // namespace rtcp
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
index 150b5b4097..07de49a4a0 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -9,27 +9,29 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace rtcp {
 
-enum { kCommonFbFmtLength = 12 };
-enum { kReportBlockLength = 24 };
+static const int kCommonFbFmtLength = 12;
+static const int kReportBlockLength = 24;
 
-class Dlrr;
 class RawPacket;
-class Rrtr;
-class VoipMetric;
 
 // Class for building RTCP packets.
 //
@@ -64,96 +66,66 @@ class RtcpPacket {
 
   void Append(RtcpPacket* packet);
 
-  RawPacket Build() const;
+  // Callback used to signal that an RTCP packet is ready. Note that this may
+  // not contain all data in this RtcpPacket; if a packet cannot fit in
+  // max_length bytes, it will be fragmented and multiple calls to this
+  // callback will be made.
+  class PacketReadyCallback {
+   public:
+    PacketReadyCallback() {}
+    virtual ~PacketReadyCallback() {}
 
-  void Build(uint8_t* packet, size_t* length, size_t max_length) const;
+    virtual void OnPacketReady(uint8_t* data, size_t length) = 0;
+  };
+
+  // Convenience method mostly used for test. Max length of IP_PACKET_SIZE is
+  // used, will cause assertion error if fragmentation occurs.
+  rtc::scoped_ptr<RawPacket> Build() const;
+
+  // Returns true if all calls to Create succeeded. A buffer of size
+  // IP_PACKET_SIZE will be allocated and reused between calls to callback.
+  bool Build(PacketReadyCallback* callback) const;
+
+  // Returns true if all calls to Create succeeded. Provided buffer reference
+  // will be used for all calls to callback.
+  bool BuildExternalBuffer(uint8_t* buffer,
+                           size_t max_length,
+                           PacketReadyCallback* callback) const;
+
+  // Size of this packet in bytes (including headers, excluding nested packets).
+  virtual size_t BlockLength() const = 0;
 
  protected:
-  RtcpPacket() : kHeaderLength(4) {}
+  RtcpPacket() {}
 
-  virtual void Create(
-      uint8_t* packet, size_t* length, size_t max_length) const = 0;
+  virtual bool Create(uint8_t* packet,
+                      size_t* index,
+                      size_t max_length,
+                      PacketReadyCallback* callback) const = 0;
 
-  const size_t kHeaderLength;
+  static void CreateHeader(uint8_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Size in 32bit words - 1.
+                           uint8_t* buffer,
+                           size_t* pos);
 
- private:
-  void CreateAndAddAppended(
-      uint8_t* packet, size_t* length, size_t max_length) const;
+  bool OnBufferFull(uint8_t* packet,
+                    size_t* index,
+                    RtcpPacket::PacketReadyCallback* callback) const;
 
+  size_t HeaderLength() const;
+
+  static const size_t kHeaderLength = 4;
   std::vector<RtcpPacket*> appended_packets_;
-};
-
-class Empty : public RtcpPacket {
- public:
-  Empty() : RtcpPacket() {}
-
-  virtual ~Empty() {}
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(Empty);
+  bool CreateAndAddAppended(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            PacketReadyCallback* callback) const;
 };
 
-// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
-//
-// RTCP report block (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_1 (SSRC of first source)                 |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  | fraction lost |       cumulative number of packets lost       |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |           extended highest sequence number received           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                      interarrival jitter                      |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                         last SR (LSR)                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   delay since last SR (DLSR)                  |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-
-class ReportBlock {
- public:
-  ReportBlock() {
-    // TODO(asapersson): Consider adding a constructor to struct.
-    memset(&report_block_, 0, sizeof(report_block_));
-  }
-
-  ~ReportBlock() {}
-
-  void To(uint32_t ssrc) {
-    report_block_.SSRC = ssrc;
-  }
-  void WithFractionLost(uint8_t fraction_lost) {
-    report_block_.FractionLost = fraction_lost;
-  }
-  void WithCumulativeLost(uint32_t cumulative_lost) {
-    report_block_.CumulativeNumOfPacketsLost = cumulative_lost;
-  }
-  void WithExtHighestSeqNum(uint32_t ext_highest_seq_num) {
-    report_block_.ExtendedHighestSequenceNumber = ext_highest_seq_num;
-  }
-  void WithJitter(uint32_t jitter) {
-    report_block_.Jitter = jitter;
-  }
-  void WithLastSr(uint32_t last_sr) {
-    report_block_.LastSR = last_sr;
-  }
-  void WithDelayLastSr(uint32_t delay_last_sr) {
-    report_block_.DelayLastSR = delay_last_sr;
-  }
-
- private:
-  friend class SenderReport;
-  friend class ReceiverReport;
-  RTCPUtility::RTCPPacketReportBlockItem report_block_;
-};
+// TODO(sprang): Move RtcpPacket subclasses out to separate files.
 
 // RTCP sender report (RFC 3550).
 //
@@ -202,17 +174,18 @@ class SenderReport : public RtcpPacket {
   void WithOctetCount(uint32_t octet_count) {
     sr_.SenderOctetCount = octet_count;
   }
-  void WithReportBlock(ReportBlock* block);
+  bool WithReportBlock(const ReportBlock& block);
 
  protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  enum { kMaxNumberOfReportBlocks = 0x1f };
+  static const int kMaxNumberOfReportBlocks = 0x1f;
 
-  size_t BlockLength() const {
+  size_t BlockLength() const override {
     const size_t kSrHeaderLength = 8;
     const size_t kSenderInfoLength = 20;
     return kSrHeaderLength + kSenderInfoLength +
@@ -220,98 +193,9 @@ class SenderReport : public RtcpPacket {
   }
 
   RTCPUtility::RTCPPacketSR sr_;
-  std::vector<RTCPUtility::RTCPPacketReportBlockItem> report_blocks_;
+  std::vector<ReportBlock> report_blocks_;
 
-  DISALLOW_COPY_AND_ASSIGN(SenderReport);
-};
-
-//
-// RTCP receiver report (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P|    RC   |   PT=RR=201   |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                     SSRC of packet sender                     |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                         report block(s)                       |
-//  |                            ....                               |
-
-class ReceiverReport : public RtcpPacket {
- public:
-  ReceiverReport() : RtcpPacket() {
-    memset(&rr_, 0, sizeof(rr_));
-  }
-
-  virtual ~ReceiverReport() {}
-
-  void From(uint32_t ssrc) {
-    rr_.SenderSSRC = ssrc;
-  }
-  void WithReportBlock(ReportBlock* block);
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  enum { kMaxNumberOfReportBlocks = 0x1f };
-
-  size_t BlockLength() const {
-    const size_t kRrHeaderLength = 8;
-    return kRrHeaderLength + report_blocks_.size() * kReportBlockLength;
-  }
-
-  RTCPUtility::RTCPPacketRR rr_;
-  std::vector<RTCPUtility::RTCPPacketReportBlockItem> report_blocks_;
-
-  DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
-};
-
-// Transmission Time Offsets in RTP Streams (RFC 5450).
-//
-//      0                   1                   2                   3
-//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//     .                                                               .
-//     .                                                               .
-//     .                                                               .
-//     |                      inter-arrival jitter                     |
-//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//
-//  If present, this RTCP packet must be placed after a receiver report
-//  (inside a compound RTCP packet), and MUST have the same value for RC
-//  (reception report count) as the receiver report.
-
-class Ij : public RtcpPacket {
- public:
-  Ij() : RtcpPacket() {}
-
-  virtual ~Ij() {}
-
-  void WithJitterItem(uint32_t jitter);
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  enum { kMaxNumberOfIjItems = 0x1f };
-
-  size_t BlockLength() const {
-    return kHeaderLength + 4 * ij_items_.size();
-  }
-
-  std::vector<uint32_t> ij_items_;
-
-  DISALLOW_COPY_AND_ASSIGN(Ij);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SenderReport);
 };
 
 // Source Description (SDES) (RFC 3550).
@@ -346,7 +230,7 @@ class Sdes : public RtcpPacket {
 
   virtual ~Sdes() {}
 
-  void WithCName(uint32_t ssrc, std::string cname);
+  bool WithCName(uint32_t ssrc, const std::string& cname);
 
   struct Chunk {
     uint32_t ssrc;
@@ -355,268 +239,19 @@ class Sdes : public RtcpPacket {
   };
 
  protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  enum { kMaxNumberOfChunks = 0x1f };
+  static const int kMaxNumberOfChunks = 0x1f;
 
-  size_t BlockLength() const;
+  size_t BlockLength() const override;
 
   std::vector<Chunk> chunks_;
 
-  DISALLOW_COPY_AND_ASSIGN(Sdes);
-};
-
-//
-// Bye packet (BYE) (RFC 3550).
-//
-//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |                           SSRC/CSRC                           |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       :                              ...                              :
-//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// (opt) |     length    |               reason for leaving            ...
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Bye : public RtcpPacket {
- public:
-  Bye() : RtcpPacket() {
-    memset(&bye_, 0, sizeof(bye_));
-  }
-
-  virtual ~Bye() {}
-
-  void From(uint32_t ssrc) {
-    bye_.SenderSSRC = ssrc;
-  }
-  void WithCsrc(uint32_t csrc);
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  enum { kMaxNumberOfCsrcs = 0x1f - 1 };
-
-  size_t BlockLength() const {
-    size_t source_count = 1 + csrcs_.size();
-    return kHeaderLength + 4 * source_count;
-  }
-
-  RTCPUtility::RTCPPacketBYE bye_;
-  std::vector<uint32_t> csrcs_;
-
-  DISALLOW_COPY_AND_ASSIGN(Bye);
-};
-
-// Application-Defined packet (APP) (RFC 3550).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |V=2|P| subtype |   PT=APP=204  |             length            |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                           SSRC/CSRC                           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                          name (ASCII)                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   application-dependent data                ...
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class App : public RtcpPacket {
- public:
-  App()
-      : RtcpPacket(),
-        ssrc_(0) {
-    memset(&app_, 0, sizeof(app_));
-  }
-
-  virtual ~App() {}
-
-  void From(uint32_t ssrc) {
-    ssrc_ = ssrc;
-  }
-  void WithSubType(uint8_t subtype) {
-    assert(subtype <= 0x1f);
-    app_.SubType = subtype;
-  }
-  void WithName(uint32_t name) {
-    app_.Name = name;
-  }
-  void WithData(const uint8_t* data, uint16_t data_length) {
-    assert(data);
-    assert(data_length <= kRtcpAppCode_DATA_SIZE);
-    assert(data_length % 4 == 0);
-    memcpy(app_.Data, data, data_length);
-    app_.Size = data_length;
-  }
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  size_t BlockLength() const {
-    return 12 + app_.Size;
-  }
-
-  uint32_t ssrc_;
-  RTCPUtility::RTCPPacketAPP app_;
-
-  DISALLOW_COPY_AND_ASSIGN(App);
-};
-
-// RFC 4585: Feedback format.
-//
-// Common packet format:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |V=2|P|   FMT   |       PT      |          length               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of packet sender                        |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                  SSRC of media source                         |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   :            Feedback Control Information (FCI)                 :
-//   :
-
-// Picture loss indication (PLI) (RFC 4585).
-//
-// FCI: no feedback control information.
-
-class Pli : public RtcpPacket {
- public:
-  Pli() : RtcpPacket() {
-    memset(&pli_, 0, sizeof(pli_));
-  }
-
-  virtual ~Pli() {}
-
-  void From(uint32_t ssrc) {
-    pli_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    pli_.MediaSSRC = ssrc;
-  }
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  size_t BlockLength() const {
-    return kCommonFbFmtLength;
-  }
-
-  RTCPUtility::RTCPPacketPSFBPLI pli_;
-
-  DISALLOW_COPY_AND_ASSIGN(Pli);
-};
-
-// Slice loss indication (SLI) (RFC 4585).
-//
-// FCI:
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            First        |        Number           | PictureID |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Sli : public RtcpPacket {
- public:
-  Sli() : RtcpPacket() {
-    memset(&sli_, 0, sizeof(sli_));
-    memset(&sli_item_, 0, sizeof(sli_item_));
-  }
-
-  virtual ~Sli() {}
-
-  void From(uint32_t ssrc) {
-    sli_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    sli_.MediaSSRC = ssrc;
-  }
-  void WithFirstMb(uint16_t first_mb) {
-    assert(first_mb <= 0x1fff);
-    sli_item_.FirstMB = first_mb;
-  }
-  void WithNumberOfMb(uint16_t number_mb) {
-    assert(number_mb <= 0x1fff);
-    sli_item_.NumberOfMB = number_mb;
-  }
-  void WithPictureId(uint8_t picture_id) {
-    assert(picture_id <= 0x3f);
-    sli_item_.PictureId = picture_id;
-  }
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  size_t BlockLength() const {
-    const size_t kFciLength = 4;
-    return kCommonFbFmtLength + kFciLength;
-  }
-
-  RTCPUtility::RTCPPacketPSFBSLI sli_;
-  RTCPUtility::RTCPPacketPSFBSLIItem sli_item_;
-
-  DISALLOW_COPY_AND_ASSIGN(Sli);
-};
-
-// Generic NACK (RFC 4585).
-//
-// FCI:
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |            PID                |             BLP               |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Nack : public RtcpPacket {
- public:
-  Nack() : RtcpPacket() {
-    memset(&nack_, 0, sizeof(nack_));
-  }
-
-  virtual ~Nack() {}
-
-  void From(uint32_t ssrc) {
-    nack_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    nack_.MediaSSRC = ssrc;
-  }
-  void WithList(const uint16_t* nack_list, int length);
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  size_t BlockLength() const {
-    size_t fci_length = 4 * nack_fields_.size();
-    return kCommonFbFmtLength + fci_length;
-  }
-
-  RTCPUtility::RTCPPacketRTPFBNACK nack_;
-  std::vector<RTCPUtility::RTCPPacketRTPFBNACKItem> nack_fields_;
-
-  DISALLOW_COPY_AND_ASSIGN(Nack);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Sdes);
 };
 
 // Reference picture selection indication (RPSI) (RFC 4585).
@@ -654,12 +289,13 @@ class Rpsi : public RtcpPacket {
   void WithPictureId(uint64_t picture_id);
 
  protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  size_t BlockLength() const {
+  size_t BlockLength() const override {
     size_t fci_length = 2 + (rpsi_.NumberOfValidBits / 8) + padding_bytes_;
     return kCommonFbFmtLength + fci_length;
   }
@@ -667,7 +303,7 @@ class Rpsi : public RtcpPacket {
   uint8_t padding_bytes_;
   RTCPUtility::RTCPPacketPSFBRPSI rpsi_;
 
-  DISALLOW_COPY_AND_ASSIGN(Rpsi);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Rpsi);
 };
 
 // Full intra request (FIR) (RFC 5104).
@@ -702,12 +338,13 @@ class Fir : public RtcpPacket {
   }
 
  protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  size_t BlockLength() const {
+  size_t BlockLength() const override {
     const size_t kFciLength = 8;
     return kCommonFbFmtLength + kFciLength;
   }
@@ -716,102 +353,6 @@ class Fir : public RtcpPacket {
   RTCPUtility::RTCPPacketPSFBFIRItem fir_item_;
 };
 
-// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Tmmbr : public RtcpPacket {
- public:
-  Tmmbr() : RtcpPacket() {
-    memset(&tmmbr_, 0, sizeof(tmmbr_));
-    memset(&tmmbr_item_, 0, sizeof(tmmbr_item_));
-  }
-
-  virtual ~Tmmbr() {}
-
-  void From(uint32_t ssrc) {
-    tmmbr_.SenderSSRC = ssrc;
-  }
-  void To(uint32_t ssrc) {
-    tmmbr_item_.SSRC = ssrc;
-  }
-  void WithBitrateKbps(uint32_t bitrate_kbps) {
-    tmmbr_item_.MaxTotalMediaBitRate = bitrate_kbps;
-  }
-  void WithOverhead(uint16_t overhead) {
-    assert(overhead <= 0x1ff);
-    tmmbr_item_.MeasuredOverhead = overhead;
-  }
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  size_t BlockLength() const {
-    const size_t kFciLen = 8;
-    return kCommonFbFmtLength + kFciLen;
-  }
-
-  RTCPUtility::RTCPPacketRTPFBTMMBR tmmbr_;
-  RTCPUtility::RTCPPacketRTPFBTMMBRItem tmmbr_item_;
-
-  DISALLOW_COPY_AND_ASSIGN(Tmmbr);
-};
-
-// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
-//
-// FCI:
-//
-//    0                   1                   2                   3
-//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   |                              SSRC                             |
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
-//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Tmmbn : public RtcpPacket {
- public:
-  Tmmbn() : RtcpPacket() {
-    memset(&tmmbn_, 0, sizeof(tmmbn_));
-  }
-
-  virtual ~Tmmbn() {}
-
-  void From(uint32_t ssrc) {
-    tmmbn_.SenderSSRC = ssrc;
-  }
-  void WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead);
-
- protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
-
- private:
-  enum { kMaxNumberOfTmmbrs = 50 };
-
-  size_t BlockLength() const {
-    const size_t kFciLen = 8;
-    return kCommonFbFmtLength + kFciLen * tmmbn_items_.size();
-  }
-
-  RTCPUtility::RTCPPacketRTPFBTMMBN tmmbn_;
-  std::vector<RTCPUtility::RTCPPacketRTPFBTMMBRItem> tmmbn_items_;
-
-  DISALLOW_COPY_AND_ASSIGN(Tmmbn);
-};
-
 // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
 //
 //    0                   1                   2                   3
@@ -850,21 +391,22 @@ class Remb : public RtcpPacket {
   }
 
  protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  enum { kMaxNumberOfSsrcs = 0xff };
+  static const int kMaxNumberOfSsrcs = 0xff;
 
-  size_t BlockLength() const {
+  size_t BlockLength() const override {
     return (remb_item_.NumberOfSSRCs + 5) * 4;
   }
 
   RTCPUtility::RTCPPacketPSFBAPP remb_;
   RTCPUtility::RTCPPacketPSFBREMBItem remb_item_;
 
-  DISALLOW_COPY_AND_ASSIGN(Remb);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Remb);
 };
 
 // From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
@@ -893,179 +435,42 @@ class Xr : public RtcpPacket {
   void From(uint32_t ssrc) {
     xr_header_.OriginatorSSRC = ssrc;
   }
-  void WithRrtr(Rrtr* rrtr);
-  void WithDlrr(Dlrr* dlrr);
-  void WithVoipMetric(VoipMetric* voip_metric);
+
+  // Max 50 items of each of {Rrtr, Dlrr, VoipMetric} allowed per Xr.
+  bool WithRrtr(Rrtr* rrtr);
+  bool WithDlrr(Dlrr* dlrr);
+  bool WithVoipMetric(VoipMetric* voip_metric);
 
  protected:
-  void Create(uint8_t* packet,
-              size_t* length,
-              size_t max_length) const override;
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
 
  private:
-  enum { kMaxNumberOfRrtrBlocks = 50 };
-  enum { kMaxNumberOfDlrrBlocks = 50 };
-  enum { kMaxNumberOfVoipMetricBlocks = 50 };
+  static const int kMaxNumberOfRrtrBlocks = 50;
+  static const int kMaxNumberOfDlrrBlocks = 50;
+  static const int kMaxNumberOfVoipMetricBlocks = 50;
 
-  size_t BlockLength() const {
+  size_t BlockLength() const override {
     const size_t kXrHeaderLength = 8;
     return kXrHeaderLength + RrtrLength() + DlrrLength() + VoipMetricLength();
   }
 
-  size_t RrtrLength() const {
-    const size_t kRrtrBlockLength = 12;
-    return kRrtrBlockLength * rrtr_blocks_.size();
-  }
+  size_t RrtrLength() const { return Rrtr::kLength * rrtr_blocks_.size(); }
 
   size_t DlrrLength() const;
 
   size_t VoipMetricLength() const {
-    const size_t kVoipMetricBlockLength = 36;
-    return kVoipMetricBlockLength * voip_metric_blocks_.size();
+    return VoipMetric::kLength * voip_metric_blocks_.size();
   }
 
   RTCPUtility::RTCPPacketXR xr_header_;
-  std::vector<RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem> rrtr_blocks_;
-  std::vector<DlrrBlock> dlrr_blocks_;
-  std::vector<RTCPUtility::RTCPPacketXRVOIPMetricItem> voip_metric_blocks_;
+  std::vector<Rrtr> rrtr_blocks_;
+  std::vector<Dlrr> dlrr_blocks_;
+  std::vector<VoipMetric> voip_metric_blocks_;
 
-  DISALLOW_COPY_AND_ASSIGN(Xr);
-};
-
-// Receiver Reference Time Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=4      |   reserved    |       block length = 2        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |              NTP timestamp, most significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |             NTP timestamp, least significant word             |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class Rrtr {
- public:
-  Rrtr() {
-    memset(&rrtr_block_, 0, sizeof(rrtr_block_));
-  }
-  ~Rrtr() {}
-
-  void WithNtpSec(uint32_t sec) {
-    rrtr_block_.NTPMostSignificant = sec;
-  }
-  void WithNtpFrac(uint32_t frac) {
-    rrtr_block_.NTPLeastSignificant = frac;
-  }
-
- private:
-  friend class Xr;
-  RTCPUtility::RTCPPacketXRReceiverReferenceTimeItem rrtr_block_;
-
-  DISALLOW_COPY_AND_ASSIGN(Rrtr);
-};
-
-// DLRR Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=5      |   reserved    |         block length          |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_1 (SSRC of first receiver)               | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  |                         last RR (LRR)                         |   1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                   delay since last RR (DLRR)                  |
-//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-//  |                 SSRC_2 (SSRC of second receiver)              | sub-
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-//  :                               ...                             :   2
-
-class Dlrr {
- public:
-  Dlrr() {}
-  ~Dlrr() {}
-
-  void WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
-
- private:
-  friend class Xr;
-  enum { kMaxNumberOfDlrrItems = 100 };
-
-  std::vector<RTCPUtility::RTCPPacketXRDLRRReportBlockItem> dlrr_block_;
-
-  DISALLOW_COPY_AND_ASSIGN(Dlrr);
-};
-
-// VoIP Metrics Report Block (RFC 3611).
-//
-//   0                   1                   2                   3
-//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     BT=7      |   reserved    |       block length = 8        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |                        SSRC of source                         |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   loss rate   | discard rate  | burst density |  gap density  |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |       burst duration          |         gap duration          |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |     round trip delay          |       end system delay        |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  | signal level  |  noise level  |     RERL      |     Gmin      |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |   RX config   |   reserved    |          JB nominal           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//  |          JB maximum           |          JB abs max           |
-//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-class VoipMetric {
- public:
-  VoipMetric() {
-    memset(&metric_, 0, sizeof(metric_));
-  }
-  ~VoipMetric() {}
-
-  void To(uint32_t ssrc) { metric_.SSRC = ssrc; }
-  void LossRate(uint8_t loss_rate) { metric_.lossRate = loss_rate; }
-  void DiscardRate(uint8_t discard_rate) { metric_.discardRate = discard_rate; }
-  void BurstDensity(uint8_t burst_density) {
-    metric_.burstDensity = burst_density;
-  }
-  void GapDensity(uint8_t gap_density) { metric_.gapDensity = gap_density; }
-  void BurstDuration(uint16_t burst_duration) {
-    metric_.burstDuration = burst_duration;
-  }
-  void GapDuration(uint16_t gap_duration) {
-    metric_.gapDuration = gap_duration;
-  }
-  void RoundTripDelay(uint16_t round_trip_delay) {
-    metric_.roundTripDelay = round_trip_delay;
-  }
-  void EndSystemDelay(uint16_t end_system_delay) {
-    metric_.endSystemDelay = end_system_delay;
-  }
-  void SignalLevel(uint8_t signal_level) { metric_.signalLevel = signal_level; }
-  void NoiseLevel(uint8_t noise_level) { metric_.noiseLevel = noise_level; }
-  void Rerl(uint8_t rerl) { metric_.RERL = rerl; }
-  void Gmin(uint8_t gmin) { metric_.Gmin = gmin; }
-  void Rfactor(uint8_t rfactor) { metric_.Rfactor = rfactor; }
-  void ExtRfactor(uint8_t extrfactor) { metric_.extRfactor = extrfactor; }
-  void MosLq(uint8_t moslq) { metric_.MOSLQ = moslq; }
-  void MosCq(uint8_t moscq) { metric_.MOSCQ = moscq; }
-  void RxConfig(uint8_t rxconfig) { metric_.RXconfig = rxconfig; }
-  void JbNominal(uint16_t jbnominal) { metric_.JBnominal = jbnominal; }
-  void JbMax(uint16_t jbmax) { metric_.JBmax = jbmax; }
-  void JbAbsMax(uint16_t jbabsmax) { metric_.JBabsMax = jbabsmax; }
-
- private:
-  friend class Xr;
-  RTCPUtility::RTCPPacketXRVOIPMetricItem metric_;
-
-  DISALLOW_COPY_AND_ASSIGN(VoipMetric);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Xr);
 };
 
 // Class holding a RTCP packet.
@@ -1074,29 +479,26 @@ class VoipMetric {
 //  RawPacket raw_packet(buffer, length);
 //
 // To access the raw packet:
-//  raw_packet.buffer();         - pointer to the raw packet
-//  raw_packet.buffer_length();  - the length of the raw packet
+//  raw_packet.Buffer();         - pointer to the raw packet
+//  raw_packet.BufferLength();   - the length of the raw packet
 
 class RawPacket {
  public:
-  RawPacket(const uint8_t* packet, size_t length) {
-    assert(length <= IP_PACKET_SIZE);
-    memcpy(buffer_, packet, length);
-    buffer_length_ = length;
-  }
+  explicit RawPacket(size_t buffer_length);
+  RawPacket(const uint8_t* packet, size_t packet_length);
 
-  const uint8_t* buffer() {
-    return buffer_;
-  }
-  size_t buffer_length() const {
-    return buffer_length_;
-  }
+  const uint8_t* Buffer() const;
+  uint8_t* MutableBuffer();
+  size_t BufferLength() const;
+  size_t Length() const;
+  void SetLength(size_t length);
 
  private:
-  size_t buffer_length_;
-  uint8_t buffer_[IP_PACKET_SIZE];
+  const size_t buffer_length_;
+  size_t length_;
+  rtc::scoped_ptr<uint8_t[]> buffer_;
 };
 
 }  // namespace rtcp
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_RTCP_PACKET_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
new file mode 100644
index 0000000000..a1ad8d6427
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Application-Defined packet (APP) (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| subtype |   PT=APP=204  |             length            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                           SSRC/CSRC                           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          name (ASCII)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |                   application-dependent data                ...
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool App::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  sub_type_ = header.count_or_format;
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  name_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+  data_.SetData(&payload[8], header.payload_size_bytes - 8);
+  return true;
+}
+
+void App::WithSubType(uint8_t subtype) {
+  RTC_DCHECK_LE(subtype, 0x1f);
+  sub_type_ = subtype;
+}
+
+void App::WithData(const uint8_t* data, size_t data_length) {
+  RTC_DCHECK(data);
+  RTC_DCHECK_EQ(0u, data_length % 4) << "Data must be 32 bits aligned.";
+  RTC_DCHECK(data_length <= kMaxDataSize) << "App data size << " << data_length
+                                          << "exceed maximum of "
+                                          << kMaxDataSize << " bytes.";
+  data_.SetData(data, data_length);
+}
+
+bool App::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
+
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
+  memcpy(&packet[*index + 8], data_.data(), data_.size());
+  *index += (8 + data_.size());
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
new file mode 100644
index 0000000000..16bd3fc2a2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class App : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 204;
+  // 28 bytes for UDP header
+  // 12 bytes for RTCP app header
+  static const size_t kMaxDataSize = IP_PACKET_SIZE - 12 - 28;
+  App() : sub_type_(0), ssrc_(0), name_(0) {}
+
+  virtual ~App() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void From(uint32_t ssrc) { ssrc_ = ssrc; }
+  void WithSubType(uint8_t subtype);
+  void WithName(uint32_t name) { name_ = name; }
+  void WithData(const uint8_t* data, size_t data_length);
+
+  uint8_t sub_type() const { return sub_type_; }
+  uint32_t ssrc() const { return ssrc_; }
+  uint32_t name() const { return name_; }
+  size_t data_size() const { return data_.size(); }
+  const uint8_t* data() const { return data_.data(); }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override { return 12 + data_.size(); }
+
+  uint8_t sub_type_;
+  uint32_t ssrc_;
+  uint32_t name_;
+  rtc::Buffer data_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(App);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
new file mode 100644
index 0000000000..4451fe8fb5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::App;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kName = ((uint32_t)'n' << 24) | ((uint32_t)'a' << 16) |
+                       ((uint32_t)'m' << 8) | (uint32_t)'e';
+const uint32_t kSenderSsrc = 0x12345678;
+
+class RtcpPacketAppTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = app.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    // Check there is exactly one RTCP packet in the buffer.
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  App app;
+  rtc::scoped_ptr<RawPacket> packet;
+  const App& parsed() { return parsed_; }
+
+ private:
+  App parsed_;
+};
+
+TEST_F(RtcpPacketAppTest, WithNoData) {
+  app.WithSubType(30);
+  app.WithName(kName);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(30U, parsed().sub_type());
+  EXPECT_EQ(kName, parsed().name());
+  EXPECT_EQ(0u, parsed().data_size());
+}
+
+TEST_F(RtcpPacketAppTest, WithData) {
+  app.From(kSenderSsrc);
+  app.WithSubType(30);
+  app.WithName(kName);
+  const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'};
+  const size_t kDataLength = sizeof(kData) / sizeof(kData[0]);
+  app.WithData(kData, kDataLength);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(30U, parsed().sub_type());
+  EXPECT_EQ(kName, parsed().name());
+  EXPECT_EQ(kDataLength, parsed().data_size());
+  EXPECT_EQ(0, memcmp(kData, parsed().data(), kDataLength));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
new file mode 100644
index 0000000000..4cfc921ce5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Bye packet (BYE) (RFC 3550).
+//
+//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |                           SSRC/CSRC                           |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       :                              ...                              :
+//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// (opt) |     length    |               reason for leaving            ...
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Bye::Bye() : sender_ssrc_(0) {}
+
+bool Bye::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t src_count = header.count_or_format;
+  // Validate packet.
+  if (header.payload_size_bytes < 4u * src_count) {
+    LOG(LS_WARNING)
+        << "Packet is too small to contain CSRCs it promise to have.";
+    return false;
+  }
+  bool has_reason = (header.payload_size_bytes > 4u * src_count);
+  uint8_t reason_length = 0;
+  if (has_reason) {
+    reason_length = payload[4u * src_count];
+    if (header.payload_size_bytes - 4u * src_count < 1u + reason_length) {
+      LOG(LS_WARNING) << "Invalid reason length: " << reason_length;
+      return false;
+    }
+  }
+  // Once sure packet is valid, copy values.
+  if (src_count == 0) {  // A count value of zero is valid, but useless.
+    sender_ssrc_ = 0;
+    csrcs_.clear();
+  } else {
+    sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload);
+    csrcs_.resize(src_count - 1);
+    for (size_t i = 1; i < src_count; ++i)
+      csrcs_[i - 1] = ByteReader<uint32_t>::ReadBigEndian(&payload[4 * i]);
+  }
+
+  if (has_reason) {
+    reason_.assign(reinterpret_cast<const char*>(&payload[4u * src_count + 1]),
+                   reason_length);
+  } else {
+    reason_.clear();
+  }
+
+  return true;
+}
+
+bool Bye::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(1 + csrcs_.size(), kPacketType, HeaderLength(), packet, index);
+  // Store srcs of the leaving clients.
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], sender_ssrc_);
+  *index += sizeof(uint32_t);
+  for (uint32_t csrc : csrcs_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], csrc);
+    *index += sizeof(uint32_t);
+  }
+  // Store the reason to leave.
+  if (!reason_.empty()) {
+    uint8_t reason_length = reason_.size();
+    packet[(*index)++] = reason_length;
+    memcpy(&packet[*index], reason_.data(), reason_length);
+    *index += reason_length;
+    // Add padding bytes if needed.
+    size_t bytes_to_pad = index_end - *index;
+    RTC_DCHECK_LE(bytes_to_pad, 3u);
+    if (bytes_to_pad > 0) {
+      memset(&packet[*index], 0, bytes_to_pad);
+      *index += bytes_to_pad;
+    }
+  }
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+bool Bye::WithCsrc(uint32_t csrc) {
+  if (csrcs_.size() >= kMaxNumberOfCsrcs) {
+    LOG(LS_WARNING) << "Max CSRC size reached.";
+    return false;
+  }
+  csrcs_.push_back(csrc);
+  return true;
+}
+
+void Bye::WithReason(const std::string& reason) {
+  RTC_DCHECK_LE(reason.size(), 0xffu);
+  reason_ = reason;
+}
+
+size_t Bye::BlockLength() const {
+  size_t src_count = (1 + csrcs_.size());
+  size_t reason_size_in_32bits = reason_.empty() ? 0 : (reason_.size() / 4 + 1);
+  return kHeaderLength + 4 * (src_count + reason_size_in_32bits);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
new file mode 100644
index 0000000000..6b4a181330
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Bye : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 203;
+
+  Bye();
+  virtual ~Bye() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  bool WithCsrc(uint32_t csrc);
+  void WithReason(const std::string& reason);
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  const std::string& reason() const { return reason_; }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfCsrcs = 0x1f - 1;  // First item is sender SSRC.
+
+  size_t BlockLength() const override;
+
+  uint32_t sender_ssrc_;
+  std::vector<uint32_t> csrcs_;
+  std::string reason_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Bye);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
new file mode 100644
index 0000000000..d2ae8ed782
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using ::testing::ElementsAre;
+
+using webrtc::rtcp::Bye;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kCsrc1 = 0x22232425;
+const uint32_t kCsrc2 = 0x33343536;
+
+class RtcpPacketByeTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = bye.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    // Check that there is exactly one RTCP packet in the buffer.
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_bye.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  Bye bye;
+  rtc::scoped_ptr<RawPacket> packet;
+  Bye parsed_bye;
+};
+
+TEST_F(RtcpPacketByeTest, Bye) {
+  bye.From(kSenderSsrc);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST_F(RtcpPacketByeTest, WithCsrcs) {
+  bye.From(kSenderSsrc);
+  EXPECT_TRUE(bye.WithCsrc(kCsrc1));
+  EXPECT_TRUE(bye.WithCsrc(kCsrc2));
+  EXPECT_TRUE(bye.reason().empty());
+
+  BuildPacket();
+  EXPECT_EQ(16u, packet->Length());  // Header: 4, 3xSRCs: 12, Reason: 0.
+
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2));
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST_F(RtcpPacketByeTest, WithCsrcsAndReason) {
+  const std::string kReason = "Some Reason";
+
+  bye.From(kSenderSsrc);
+  EXPECT_TRUE(bye.WithCsrc(kCsrc1));
+  EXPECT_TRUE(bye.WithCsrc(kCsrc2));
+  bye.WithReason(kReason);
+
+  BuildPacket();
+  EXPECT_EQ(28u, packet->Length());  // Header: 4, 3xSRCs: 12, Reason: 12.
+
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2));
+  EXPECT_EQ(kReason, parsed_bye.reason());
+}
+
+TEST_F(RtcpPacketByeTest, WithTooManyCsrcs) {
+  bye.From(kSenderSsrc);
+  const int kMaxCsrcs = (1 << 5) - 2;  // 5 bit len, first item is sender SSRC.
+  for (int i = 0; i < kMaxCsrcs; ++i) {
+    EXPECT_TRUE(bye.WithCsrc(i));
+  }
+  EXPECT_FALSE(bye.WithCsrc(kMaxCsrcs));
+}
+
+TEST_F(RtcpPacketByeTest, WithAReason) {
+  const std::string kReason = "Some Random Reason";
+
+  bye.From(kSenderSsrc);
+  bye.WithReason(kReason);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_EQ(kReason, parsed_bye.reason());
+}
+
+TEST_F(RtcpPacketByeTest, WithReasons) {
+  // Test that packet creation/parsing behave with reasons of different length
+  // both when it require padding and when it does not.
+  for (size_t reminder = 0; reminder < 4; ++reminder) {
+    const std::string kReason(4 + reminder, 'a' + reminder);
+    bye.From(kSenderSsrc);
+    bye.WithReason(kReason);
+
+    BuildPacket();
+    ParsePacket();
+
+    EXPECT_EQ(kReason, parsed_bye.reason());
+  }
+}
+
+TEST_F(RtcpPacketByeTest, ParseEmptyPacket) {
+  RtcpCommonHeader header;
+  header.packet_type = Bye::kPacketType;
+  header.count_or_format = 0;
+  header.payload_size_bytes = 0;
+  uint8_t empty_payload[1];
+
+  EXPECT_TRUE(parsed_bye.Parse(header, empty_payload + 1));
+  EXPECT_EQ(0u, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST_F(RtcpPacketByeTest, ParseFailOnInvalidSrcCount) {
+  bye.From(kSenderSsrc);
+
+  BuildPacket();
+
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
+  header.count_or_format = 2;  // Lie there are 2 ssrcs, not one.
+
+  EXPECT_FALSE(parsed_bye.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+TEST_F(RtcpPacketByeTest, ParseFailOnInvalidReasonLength) {
+  bye.From(kSenderSsrc);
+  bye.WithReason("18 characters long");
+
+  BuildPacket();
+
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
+  header.payload_size_bytes -= 4;  // Payload is usually 32bit aligned.
+
+  EXPECT_FALSE(parsed_bye.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
new file mode 100644
index 0000000000..8f5afd5dd1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+bool CompoundPacket::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            RtcpPacket::PacketReadyCallback* callback) const {
+  return true;
+}
+
+size_t CompoundPacket::BlockLength() const {
+  return 0;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
new file mode 100644
index 0000000000..f2f49a8ffb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class CompoundPacket : public RtcpPacket {
+ public:
+  CompoundPacket() : RtcpPacket() {}
+
+  virtual ~CompoundPacket() {}
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+  size_t BlockLength() const override;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(CompoundPacket);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
new file mode 100644
index 0000000000..83dc5f6ed3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::Bye;
+using webrtc::rtcp::CompoundPacket;
+using webrtc::rtcp::Fir;
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+using webrtc::rtcp::SenderReport;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+
+const uint32_t kSenderSsrc = 0x12345678;
+
+TEST(RtcpCompoundPacketTest, AppendPacket) {
+  Fir fir;
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  rtc::scoped_ptr<RawPacket> packet(rr.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
+  EXPECT_EQ(1, parser.report_block()->num_packets());
+  EXPECT_EQ(1, parser.fir()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, AppendPacketOnEmpty) {
+  CompoundPacket empty;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  empty.Append(&rr);
+
+  rtc::scoped_ptr<RawPacket> packet(empty.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(0, parser.report_block()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, AppendPacketWithOwnAppendedPacket) {
+  Fir fir;
+  Bye bye;
+  ReportBlock rb;
+
+  ReceiverReport rr;
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  SenderReport sr;
+  sr.Append(&bye);
+  sr.Append(&rr);
+
+  rtc::scoped_ptr<RawPacket> packet(sr.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.sender_report()->num_packets());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(1, parser.report_block()->num_packets());
+  EXPECT_EQ(1, parser.bye()->num_packets());
+  EXPECT_EQ(1, parser.fir()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, BuildWithInputBuffer) {
+  Fir fir;
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+  const size_t kFirLength = 20;
+
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      RtcpPacketParser parser;
+      parser.Parse(data, length);
+      EXPECT_EQ(1, parser.receiver_report()->num_packets());
+      EXPECT_EQ(1, parser.report_block()->num_packets());
+      EXPECT_EQ(1, parser.fir()->num_packets());
+      ++packets_created_;
+    }
+
+    int packets_created_ = 0;
+  } verifier;
+  const size_t kBufferSize = kRrLength + kReportBlockLength + kFirLength;
+  uint8_t buffer[kBufferSize];
+  EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
+  EXPECT_EQ(1, verifier.packets_created_);
+}
+
+TEST(RtcpCompoundPacketTest, BuildWithTooSmallBuffer_FragmentedSend) {
+  Fir fir;
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
+  rr.Append(&fir);
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      RtcpPacketParser parser;
+      parser.Parse(data, length);
+      switch (packets_created_++) {
+        case 0:
+          EXPECT_EQ(1, parser.receiver_report()->num_packets());
+          EXPECT_EQ(1, parser.report_block()->num_packets());
+          EXPECT_EQ(0, parser.fir()->num_packets());
+          break;
+        case 1:
+          EXPECT_EQ(0, parser.receiver_report()->num_packets());
+          EXPECT_EQ(0, parser.report_block()->num_packets());
+          EXPECT_EQ(1, parser.fir()->num_packets());
+          break;
+        default:
+          ADD_FAILURE() << "OnPacketReady not expected to be called "
+                        << packets_created_ << " times.";
+      }
+    }
+
+    int packets_created_ = 0;
+  } verifier;
+  const size_t kBufferSize = kRrLength + kReportBlockLength;
+  uint8_t buffer[kBufferSize];
+  EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
+  EXPECT_EQ(2, verifier.packets_created_);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
new file mode 100644
index 0000000000..6d6c48fada
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+// DLRR Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=5      |   reserved    |         block length          |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_1 (SSRC of first receiver)               | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  |                         last RR (LRR)                         |   1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                   delay since last RR (DLRR)                  |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_2 (SSRC of second receiver)              | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  :                               ...                             :   2
+bool Dlrr::Parse(const uint8_t* buffer, uint16_t block_length_32bits) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // kReserved = buffer[1];
+  RTC_DCHECK_EQ(block_length_32bits,
+                ByteReader<uint16_t>::ReadBigEndian(&buffer[2]));
+  if (block_length_32bits % 3 != 0) {
+    LOG(LS_WARNING) << "Invalid size for dlrr block.";
+    return false;
+  }
+
+  size_t blocks_count = block_length_32bits / 3;
+  const uint8_t* read_at = buffer + kBlockHeaderLength;
+  sub_blocks_.resize(blocks_count);
+  for (SubBlock& sub_block : sub_blocks_) {
+    sub_block.ssrc = ByteReader<uint32_t>::ReadBigEndian(&read_at[0]);
+    sub_block.last_rr = ByteReader<uint32_t>::ReadBigEndian(&read_at[4]);
+    sub_block.delay_since_last_rr =
+        ByteReader<uint32_t>::ReadBigEndian(&read_at[8]);
+    read_at += kSubBlockLength;
+  }
+  return true;
+}
+
+size_t Dlrr::BlockLength() const {
+  if (sub_blocks_.empty())
+    return 0;
+  return kBlockHeaderLength + kSubBlockLength * sub_blocks_.size();
+}
+
+void Dlrr::Create(uint8_t* buffer) const {
+  if (sub_blocks_.empty())  // No subblocks, no need to write header either.
+    return;
+  // Create block header.
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], 3 * sub_blocks_.size());
+  // Create sub blocks.
+  uint8_t* write_at = buffer + kBlockHeaderLength;
+  for (const SubBlock& sub_block : sub_blocks_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[0], sub_block.ssrc);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[4], sub_block.last_rr);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[8],
+                                         sub_block.delay_since_last_rr);
+    write_at += kSubBlockLength;
+  }
+  RTC_DCHECK_EQ(buffer + BlockLength(), write_at);
+}
+
+bool Dlrr::WithDlrrItem(uint32_t ssrc,
+                        uint32_t last_rr,
+                        uint32_t delay_last_rr) {
+  if (sub_blocks_.size() >= kMaxNumberOfDlrrItems) {
+    LOG(LS_WARNING) << "Max DLRR items reached.";
+    return false;
+  }
+  SubBlock block;
+  block.ssrc = ssrc;
+  block.last_rr = last_rr;
+  block.delay_since_last_rr = delay_last_rr;
+  sub_blocks_.push_back(block);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
new file mode 100644
index 0000000000..9af2dedf3f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// DLRR Report Block: Delay since the Last Receiver Report (RFC 3611).
+class Dlrr {
+ public:
+  struct SubBlock {
+    // RFC 3611 4.5
+    uint32_t ssrc;
+    uint32_t last_rr;
+    uint32_t delay_since_last_rr;
+  };
+
+  static const uint8_t kBlockType = 5;
+  static const size_t kMaxNumberOfDlrrItems = 100;
+
+  Dlrr() {}
+  Dlrr(const Dlrr& other) = default;
+  ~Dlrr() {}
+
+  Dlrr& operator=(const Dlrr& other) = default;
+
+  // Second parameter is value read from block header,
+  // i.e. size of block in 32bits excluding block header itself.
+  bool Parse(const uint8_t* buffer, uint16_t block_length_32bits);
+
+  size_t BlockLength() const;
+  // Fills buffer with the Dlrr.
+  // Consumes BlockLength() bytes.
+  void Create(uint8_t* buffer) const;
+
+  // Max 100 DLRR Items can be added per DLRR report block.
+  bool WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
+
+  const std::vector<SubBlock>& sub_blocks() const { return sub_blocks_; }
+
+ private:
+  static const size_t kBlockHeaderLength = 4;
+  static const size_t kSubBlockLength = 12;
+
+  std::vector<SubBlock> sub_blocks_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc
new file mode 100644
index 0000000000..c7c139c560
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::rtcp::Dlrr;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSsrc = 0x12345678;
+const uint32_t kLastRR = 0x23344556;
+const uint32_t kDelay = 0x33343536;
+const uint8_t kBlock[] = {0x05, 0x00, 0x00, 0x03, 0x12, 0x34, 0x56, 0x78,
+                          0x23, 0x34, 0x45, 0x56, 0x33, 0x34, 0x35, 0x36};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+
+TEST(RtcpPacketDlrrTest, Empty) {
+  Dlrr dlrr;
+
+  EXPECT_EQ(0u, dlrr.BlockLength());
+}
+
+TEST(RtcpPacketDlrrTest, Create) {
+  Dlrr dlrr;
+  EXPECT_TRUE(dlrr.WithDlrrItem(kSsrc, kLastRR, kDelay));
+
+  ASSERT_EQ(kBlockSizeBytes, dlrr.BlockLength());
+  uint8_t buffer[kBlockSizeBytes];
+
+  dlrr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketDlrrTest, Parse) {
+  Dlrr dlrr;
+  uint16_t block_length = ByteReader<uint16_t>::ReadBigEndian(&kBlock[2]);
+  EXPECT_TRUE(dlrr.Parse(kBlock, block_length));
+
+  EXPECT_EQ(1u, dlrr.sub_blocks().size());
+  const Dlrr::SubBlock& block = dlrr.sub_blocks().front();
+  EXPECT_EQ(kSsrc, block.ssrc);
+  EXPECT_EQ(kLastRR, block.last_rr);
+  EXPECT_EQ(kDelay, block.delay_since_last_rr);
+}
+
+TEST(RtcpPacketDlrrTest, ParseFailsOnBadSize) {
+  const size_t kBigBufferSize = 0x100;  // More than enough.
+  uint8_t buffer[kBigBufferSize];
+  buffer[0] = Dlrr::kBlockType;
+  buffer[1] = 0;  // Reserved.
+  buffer[2] = 0;  // Most significant size byte.
+  for (uint8_t size = 3; size < 6; ++size) {
+    buffer[3] = size;
+    Dlrr dlrr;
+    // Parse should be successful only when size is multiple of 3.
+    EXPECT_EQ(size % 3 == 0, dlrr.Parse(buffer, static_cast<uint16_t>(size)));
+  }
+}
+
+TEST(RtcpPacketDlrrTest, FailsOnTooManySubBlocks) {
+  Dlrr dlrr;
+  for (size_t i = 1; i <= Dlrr::kMaxNumberOfDlrrItems; ++i) {
+    EXPECT_TRUE(dlrr.WithDlrrItem(kSsrc + i, kLastRR + i, kDelay + i));
+  }
+  EXPECT_FALSE(dlrr.WithDlrrItem(kSsrc, kLastRR, kDelay));
+}
+
+TEST(RtcpPacketDlrrTest, CreateAndParseMaxSubBlocks) {
+  const size_t kBufferSize = 0x1000;  // More than enough.
+  uint8_t buffer[kBufferSize];
+
+  // Create.
+  Dlrr dlrr;
+  for (size_t i = 1; i <= Dlrr::kMaxNumberOfDlrrItems; ++i) {
+    EXPECT_TRUE(dlrr.WithDlrrItem(kSsrc + i, kLastRR + i, kDelay + i));
+  }
+  size_t used_buffer_size = dlrr.BlockLength();
+  ASSERT_LE(used_buffer_size, kBufferSize);
+  dlrr.Create(buffer);
+
+  // Parse.
+  Dlrr parsed;
+  uint16_t block_length = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]);
+  EXPECT_EQ(used_buffer_size, (block_length + 1) * 4u);
+  EXPECT_TRUE(parsed.Parse(buffer, block_length));
+  EXPECT_TRUE(parsed.sub_blocks().size() == Dlrr::kMaxNumberOfDlrrItems);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
new file mode 100644
index 0000000000..030f9f81fa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Transmission Time Offsets in RTP Streams (RFC 5450).
+//
+//      0                   1                   2                   3
+//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//     |                      inter-arrival jitter                     |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//     .                                                               .
+//     .                                                               .
+//     .                                                               .
+//     |                      inter-arrival jitter                     |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  If present, this RTCP packet must be placed after a receiver report
+//  (inside a compound RTCP packet), and MUST have the same value for RC
+//  (reception report count) as the receiver report.
+
+bool ExtendedJitterReport::Parse(const RtcpCommonHeader& header,
+                                 const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t jitters_count = header.count_or_format;
+  const size_t kJitterSizeBytes = 4u;
+
+  if (header.payload_size_bytes < jitters_count * kJitterSizeBytes) {
+    LOG(LS_WARNING) << "Packet is too small to contain all the jitter.";
+    return false;
+  }
+
+  inter_arrival_jitters_.resize(jitters_count);
+  for (size_t index = 0; index < jitters_count; ++index) {
+    inter_arrival_jitters_[index] =
+        ByteReader<uint32_t>::ReadBigEndian(&payload[index * kJitterSizeBytes]);
+  }
+
+  return true;
+}
+
+bool ExtendedJitterReport::WithJitter(uint32_t jitter) {
+  if (inter_arrival_jitters_.size() >= kMaxNumberOfJitters) {
+    LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
+    return false;
+  }
+  inter_arrival_jitters_.push_back(jitter);
+  return true;
+}
+
+bool ExtendedJitterReport::Create(
+    uint8_t* packet,
+    size_t* index,
+    size_t max_length,
+    RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  size_t length = inter_arrival_jitters_.size();
+  CreateHeader(length, kPacketType, length, packet, index);
+
+  for (uint32_t jitter : inter_arrival_jitters_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, jitter);
+    *index += sizeof(uint32_t);
+  }
+  // Sanity check.
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
new file mode 100644
index 0000000000..49de7be1a8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ExtendedJitterReport : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 195;
+
+  ExtendedJitterReport() : RtcpPacket() {}
+
+  virtual ~ExtendedJitterReport() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  bool WithJitter(uint32_t jitter);
+
+  size_t jitters_count() const { return inter_arrival_jitters_.size(); }
+  uint32_t jitter(size_t index) const {
+    RTC_DCHECK_LT(index, jitters_count());
+    return inter_arrival_jitters_[index];
+  }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfJitters = 0x1f;
+
+  size_t BlockLength() const override {
+    return kHeaderLength + 4 * inter_arrival_jitters_.size();
+  }
+
+  std::vector<uint32_t> inter_arrival_jitters_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ExtendedJitterReport);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
new file mode 100644
index 0000000000..09d7b6305f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ExtendedJitterReport;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+class RtcpPacketExtendedJitterReportTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = ij.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  ExtendedJitterReport ij;
+  rtc::scoped_ptr<RawPacket> packet;
+  const ExtendedJitterReport& parsed() { return parsed_; }
+
+ private:
+  ExtendedJitterReport parsed_;
+};
+
+TEST_F(RtcpPacketExtendedJitterReportTest, NoItem) {
+  // No initialization because packet is empty.
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(0u, parsed().jitters_count());
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, OneItem) {
+  EXPECT_TRUE(ij.WithJitter(0x11121314));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(1u, parsed().jitters_count());
+  EXPECT_EQ(0x11121314U, parsed().jitter(0));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, TwoItems) {
+  EXPECT_TRUE(ij.WithJitter(0x11121418));
+  EXPECT_TRUE(ij.WithJitter(0x22242628));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(2u, parsed().jitters_count());
+  EXPECT_EQ(0x11121418U, parsed().jitter(0));
+  EXPECT_EQ(0x22242628U, parsed().jitter(1));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, TooManyItems) {
+  const int kMaxIjItems = (1 << 5) - 1;
+  for (int i = 0; i < kMaxIjItems; ++i) {
+    EXPECT_TRUE(ij.WithJitter(i));
+  }
+  EXPECT_FALSE(ij.WithJitter(kMaxIjItems));
+}
+
+TEST_F(RtcpPacketExtendedJitterReportTest, ParseFailWithTooManyItems) {
+  ij.WithJitter(0x11121418);
+  BuildPacket();
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
+  header.count_or_format++;  // Damage package.
+
+  ExtendedJitterReport parsed;
+
+  EXPECT_FALSE(parsed.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
new file mode 100644
index 0000000000..8b9b354a06
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+//
+// Generic NACK (RFC 4585).
+//
+// FCI:
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |            PID                |             BLP               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool Nack::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes < kCommonFeedbackLength + kNackItemLength) {
+    LOG(LS_WARNING) << "Payload length " << header.payload_size_bytes
+                    << " is too small for a Nack.";
+    return false;
+  }
+  size_t nack_items =
+      (header.payload_size_bytes - kCommonFeedbackLength) / kNackItemLength;
+
+  ParseCommonFeedback(payload);
+  const uint8_t* next_nack = payload + kCommonFeedbackLength;
+
+  packet_ids_.clear();
+  packed_.resize(nack_items);
+  for (size_t index = 0; index < nack_items; ++index) {
+    packed_[index].first_pid = ByteReader<uint16_t>::ReadBigEndian(next_nack);
+    packed_[index].bitmask = ByteReader<uint16_t>::ReadBigEndian(next_nack + 2);
+    next_nack += kNackItemLength;
+  }
+  Unpack();
+
+  return true;
+}
+
+bool Nack::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  RtcpPacket::PacketReadyCallback* callback) const {
+  RTC_DCHECK(!packed_.empty());
+  // If nack list can't fit in packet, try to fragment.
+  size_t nack_index = 0;
+  const size_t kCommonFbFmtLength = kHeaderLength + kCommonFeedbackLength;
+  do {
+    size_t bytes_left_in_buffer = max_length - *index;
+    if (bytes_left_in_buffer < kCommonFbFmtLength + kNackItemLength) {
+      if (!OnBufferFull(packet, index, callback))
+        return false;
+      continue;
+    }
+    size_t num_nack_fields =
+        std::min((bytes_left_in_buffer - kCommonFbFmtLength) / kNackItemLength,
+                 packed_.size() - nack_index);
+
+    size_t size_bytes =
+        (num_nack_fields * kNackItemLength) + kCommonFbFmtLength;
+    size_t header_length = ((size_bytes + 3) / 4) - 1;  // As 32bit words - 1
+    CreateHeader(kFeedbackMessageType, kPacketType, header_length, packet,
+                 index);
+    CreateCommonFeedback(packet + *index);
+    *index += kCommonFeedbackLength;
+    size_t end_index = nack_index + num_nack_fields;
+    for (; nack_index < end_index; ++nack_index) {
+      const auto& item = packed_[nack_index];
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 0, item.first_pid);
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 2, item.bitmask);
+      *index += kNackItemLength;
+    }
+    RTC_DCHECK_LE(*index, max_length);
+  } while (nack_index < packed_.size());
+
+  return true;
+}
+
+size_t Nack::BlockLength() const {
+  return (packed_.size() * kNackItemLength) + kCommonFeedbackLength +
+         kHeaderLength;
+}
+
+void Nack::WithList(const uint16_t* nack_list, size_t length) {
+  RTC_DCHECK(nack_list);
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  packet_ids_.assign(nack_list, nack_list + length);
+  Pack();
+}
+
+void Nack::Pack() {
+  RTC_DCHECK(!packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  auto it = packet_ids_.begin();
+  const auto end = packet_ids_.end();
+  while (it != end) {
+    PackedNack item;
+    item.first_pid = *it++;
+    // Bitmask specifies losses in any of the 16 packets following the pid.
+    item.bitmask = 0;
+    while (it != end) {
+      uint16_t shift = static_cast<uint16_t>(*it - item.first_pid - 1);
+      if (shift <= 15) {
+        item.bitmask |= (1 << shift);
+        ++it;
+      } else {
+        break;
+      }
+    }
+    packed_.push_back(item);
+  }
+}
+
+void Nack::Unpack() {
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(!packed_.empty());
+  for (const PackedNack& item : packed_) {
+    packet_ids_.push_back(item.first_pid);
+    uint16_t pid = item.first_pid + 1;
+    for (uint16_t bitmask = item.bitmask; bitmask != 0; bitmask >>= 1, ++pid)
+      if (bitmask & 1)
+        packet_ids_.push_back(pid);
+  }
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
new file mode 100644
index 0000000000..fb2be113a2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Nack : public Rtpfb {
+ public:
+  const uint8_t kFeedbackMessageType = 1;
+  Nack() {}
+
+  virtual ~Nack() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void WithList(const uint16_t* nack_list, size_t length);
+  const std::vector<uint16_t>& packet_ids() const { return packet_ids_; }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+  size_t BlockLength() const override;
+
+ private:
+  const size_t kNackItemLength = 4;
+  struct PackedNack {
+    uint16_t first_pid;
+    uint16_t bitmask;
+  };
+
+  void Pack();    // Fills packed_ using packed_ids_. (used in WithList).
+  void Unpack();  // Fills packet_ids_ using packed_. (used in Parse).
+
+  std::vector<PackedNack> packed_;
+  std::vector<uint16_t> packet_ids_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Nack);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
new file mode 100644
index 0000000000..01e30f5644
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using ::testing::_;
+using ::testing::ElementsAreArray;
+using ::testing::Invoke;
+using ::testing::UnorderedElementsAreArray;
+
+using webrtc::rtcp::Nack;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+const uint16_t kList[] = {0, 1, 3, 8, 16};
+const size_t kListLength = sizeof(kList) / sizeof(kList[0]);
+const uint8_t kPacket[] = {0x81, 205,  0x00, 0x03, 0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89, 0x00, 0x00, 0x80, 0x85};
+const size_t kPacketLength = sizeof(kPacket);
+
+const uint16_t kWrapList[] = {0xffdc, 0xffec, 0xfffe, 0xffff, 0x0000,
+                              0x0001, 0x0003, 0x0014, 0x0064};
+const size_t kWrapListLength = sizeof(kWrapList) / sizeof(kWrapList[0]);
+const uint8_t kWrapPacket[] = {0x81, 205,  0x00, 0x06, 0x12, 0x34, 0x56, 0x78,
+                               0x23, 0x45, 0x67, 0x89, 0xff, 0xdc, 0x80, 0x00,
+                               0xff, 0xfe, 0x00, 0x17, 0x00, 0x14, 0x00, 0x00,
+                               0x00, 0x64, 0x00, 0x00};
+const size_t kWrapPacketLength = sizeof(kWrapPacket);
+
+TEST(RtcpPacketNackTest, Create) {
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kList, kListLength);
+
+  rtc::scoped_ptr<RawPacket> packet = nack.Build();
+
+  EXPECT_EQ(kPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST(RtcpPacketNackTest, Parse) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  EXPECT_EQ(kPacketLength, header.BlockSize());
+  Nack parsed;
+
+  EXPECT_TRUE(
+      parsed.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const Nack& const_parsed = parsed;
+
+  EXPECT_EQ(kSenderSsrc, const_parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, const_parsed.media_ssrc());
+  EXPECT_THAT(const_parsed.packet_ids(), ElementsAreArray(kList));
+}
+
+TEST(RtcpPacketNackTest, CreateWrap) {
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kWrapList, kWrapListLength);
+
+  rtc::scoped_ptr<RawPacket> packet = nack.Build();
+
+  EXPECT_EQ(kWrapPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kWrapPacket, packet->Buffer(), kWrapPacketLength));
+}
+
+TEST(RtcpPacketNackTest, ParseWrap) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kWrapPacket, kWrapPacketLength, &header));
+  EXPECT_EQ(kWrapPacketLength, header.BlockSize());
+
+  Nack parsed;
+  EXPECT_TRUE(
+      parsed.Parse(header, kWrapPacket + RtcpCommonHeader::kHeaderSizeBytes));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_THAT(parsed.packet_ids(), ElementsAreArray(kWrapList));
+}
+
+TEST(RtcpPacketNackTest, BadOrder) {
+  // Does not guarantee optimal packing, but should guarantee correctness.
+  const uint16_t kUnorderedList[] = {1, 25, 13, 12, 9, 27, 29};
+  const size_t kUnorderedListLength =
+      sizeof(kUnorderedList) / sizeof(kUnorderedList[0]);
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kUnorderedList, kUnorderedListLength);
+
+  rtc::scoped_ptr<RawPacket> packet = nack.Build();
+
+  Nack parsed;
+  RtcpCommonHeader header;
+  EXPECT_TRUE(
+      RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+  EXPECT_TRUE(parsed.Parse(
+      header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_THAT(parsed.packet_ids(), UnorderedElementsAreArray(kUnorderedList));
+}
+
+TEST(RtcpPacketNackTest, CreateFragmented) {
+  Nack nack;
+  const uint16_t kList[] = {1, 100, 200, 300, 400};
+  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kList, kListLength);
+
+  class MockPacketReadyCallback : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    MOCK_METHOD2(OnPacketReady, void(uint8_t*, size_t));
+  } verifier;
+
+  class NackVerifier {
+   public:
+    explicit NackVerifier(std::vector<uint16_t> ids) : ids_(ids) {}
+    void operator()(uint8_t* data, size_t length) {
+      RtcpCommonHeader header;
+      EXPECT_TRUE(RtcpParseCommonHeader(data, length, &header));
+      EXPECT_EQ(length, header.BlockSize());
+      Nack nack;
+      EXPECT_TRUE(
+          nack.Parse(header, data + RtcpCommonHeader::kHeaderSizeBytes));
+      EXPECT_EQ(kSenderSsrc, nack.sender_ssrc());
+      EXPECT_EQ(kRemoteSsrc, nack.media_ssrc());
+      EXPECT_THAT(nack.packet_ids(), ElementsAreArray(ids_));
+    }
+    std::vector<uint16_t> ids_;
+  } packet1({1, 100, 200}), packet2({300, 400});
+
+  EXPECT_CALL(verifier, OnPacketReady(_, _))
+      .WillOnce(Invoke(packet1))
+      .WillOnce(Invoke(packet2));
+  const size_t kBufferSize = 12 + (3 * 4);  // Fits common header + 3 nack items
+  uint8_t buffer[kBufferSize];
+  EXPECT_TRUE(nack.BuildExternalBuffer(buffer, kBufferSize, &verifier));
+}
+
+TEST(RtcpPacketNackTest, CreateFailsWithTooSmallBuffer) {
+  const uint16_t kList[] = {1};
+  const size_t kMinNackBlockSize = 16;
+  Nack nack;
+  nack.From(kSenderSsrc);
+  nack.To(kRemoteSsrc);
+  nack.WithList(kList, 1);
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      ADD_FAILURE() << "Buffer should be too small.";
+    }
+  } verifier;
+  uint8_t buffer[kMinNackBlockSize - 1];
+  EXPECT_FALSE(
+      nack.BuildExternalBuffer(buffer, kMinNackBlockSize - 1, &verifier));
+}
+
+TEST(RtcpPacketNackTest, ParseFailsWithTooSmallBuffer) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  header.payload_size_bytes--;  // Damage the packet
+  Nack parsed;
+  EXPECT_FALSE(
+      parsed.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc
new file mode 100644
index 0000000000..3673491058
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+
+//
+// Picture loss indication (PLI) (RFC 4585).
+// FCI: no feedback control information.
+bool Pli::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes < kCommonFeedbackLength) {
+    LOG(LS_WARNING) << "Packet is too small to be a valid PLI packet";
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+  return true;
+}
+
+bool Pli::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
new file mode 100644
index 0000000000..5567825830
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Picture loss indication (PLI) (RFC 4585).
+class Pli : public Psfb {
+ public:
+  static const uint8_t kFeedbackMessageType = 1;
+
+  Pli() {}
+  virtual ~Pli() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override {
+    return kHeaderLength + kCommonFeedbackLength;
+  }
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Pli);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
new file mode 100644
index 0000000000..1c47c3ffb1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+using webrtc::rtcp::Pli;
+using webrtc::rtcp::RawPacket;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+// Manually created Pli packet matching constants above.
+const uint8_t kPacket[] = {0x81, 206,  0x00, 0x02,
+                           0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89};
+const size_t kPacketLength = sizeof(kPacket);
+
+TEST(RtcpPacketPliTest, Parse) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  Pli mutable_parsed;
+  EXPECT_TRUE(mutable_parsed.Parse(
+      header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const Pli& parsed = mutable_parsed;  // Read values from constant object.
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+}
+
+TEST(RtcpPacketPliTest, Create) {
+  Pli pli;
+  pli.From(kSenderSsrc);
+  pli.To(kRemoteSsrc);
+
+  rtc::scoped_ptr<RawPacket> packet(pli.Build());
+
+  ASSERT_EQ(kPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST(RtcpPacketPliTest, ParseFailsOnTooSmallPacket) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(kPacket, kPacketLength, &header));
+  header.payload_size_bytes--;
+
+  Pli parsed;
+  EXPECT_FALSE(
+      parsed.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc
new file mode 100644
index 0000000000..d1ee401dab
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Psfb::ParseCommonFeedback(const uint8_t* payload) {
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  media_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+}
+
+void Psfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc_);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
new file mode 100644
index 0000000000..dddcdecba6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// PSFB: Payload-specific feedback message.
+// RFC 4585, Section 6.3.
+class Psfb : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 206;
+
+  Psfb() : sender_ssrc_(0), media_ssrc_(0) {}
+  virtual ~Psfb() {}
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static const size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t sender_ssrc_;
+  uint32_t media_ssrc_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
new file mode 100644
index 0000000000..ef64b4f51b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+//
+// RTCP receiver report (RFC 3550).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|    RC   |   PT=RR=201   |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                     SSRC of packet sender                     |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                         report block(s)                       |
+//  |                            ....                               |
+bool ReceiverReport::Parse(const RTCPUtility::RtcpCommonHeader& header,
+                           const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t report_blocks_count = header.count_or_format;
+
+  if (header.payload_size_bytes <
+      kRrBaseLength + report_blocks_count * ReportBlock::kLength) {
+    LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+    return false;
+  }
+
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(payload);
+
+  const uint8_t* next_report_block = payload + kRrBaseLength;
+
+  report_blocks_.resize(report_blocks_count);
+  for (ReportBlock& block : report_blocks_) {
+    block.Parse(next_report_block, ReportBlock::kLength);
+    next_report_block += ReportBlock::kLength;
+  }
+
+  RTC_DCHECK_LE(next_report_block, payload + header.payload_size_bytes);
+  return true;
+}
+
+bool ReceiverReport::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+               index);
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc_);
+  *index += kRrBaseLength;
+  for (const ReportBlock& block : report_blocks_) {
+    block.Create(packet + *index);
+    *index += ReportBlock::kLength;
+  }
+  return true;
+}
+
+bool ReceiverReport::WithReportBlock(const ReportBlock& block) {
+  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+    LOG(LS_WARNING) << "Max report blocks reached.";
+    return false;
+  }
+  report_blocks_.push_back(block);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
new file mode 100644
index 0000000000..396be535e9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ReceiverReport : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 201;
+  ReceiverReport() : sender_ssrc_(0) {}
+
+  virtual ~ReceiverReport() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  bool WithReportBlock(const ReportBlock& block);
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  const std::vector<ReportBlock>& report_blocks() const {
+    return report_blocks_;
+  }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const size_t kRrBaseLength = 4;
+  static const size_t kMaxNumberOfReportBlocks = 0x1F;
+
+  size_t BlockLength() const override{
+    return kHeaderLength + kRrBaseLength +
+           report_blocks_.size() * ReportBlock::kLength;
+  }
+
+  uint32_t sender_ssrc_;
+  std::vector<ReportBlock> report_blocks_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
new file mode 100644
index 0000000000..ff3da600a5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+const uint32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+// Manually created ReceiverReport with one ReportBlock matching constants
+// above.
+// Having this block allows to test Create and Parse separately.
+const uint8_t kPacket[] = {0x81, 201,  0x00, 0x07, 0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89, 55,   0x11, 0x12, 0x13,
+                           0x22, 0x23, 0x24, 0x25, 0x33, 0x34, 0x35, 0x36,
+                           0x44, 0x45, 0x46, 0x47, 0x55, 0x56, 0x57, 0x58};
+const size_t kPacketLength = sizeof(kPacket);
+
+class RtcpPacketReceiverReportTest : public ::testing::Test {
+ protected:
+  void BuildPacket() { packet = rr.Build(); }
+  void ParsePacket() {
+    RtcpCommonHeader header;
+    EXPECT_TRUE(
+        RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
+    EXPECT_EQ(header.BlockSize(), packet->Length());
+    EXPECT_TRUE(parsed_.Parse(
+        header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
+  }
+
+  ReceiverReport rr;
+  rtc::scoped_ptr<RawPacket> packet;
+  const ReceiverReport& parsed() { return parsed_; }
+
+ private:
+  ReceiverReport parsed_;
+};
+
+TEST_F(RtcpPacketReceiverReportTest, Parse) {
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(kPacket, kPacketLength, &header);
+  EXPECT_TRUE(rr.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+  const ReceiverReport& parsed = rr;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(1u, parsed.report_blocks().size());
+  const ReportBlock& rb = parsed.report_blocks().front();
+  EXPECT_EQ(kRemoteSsrc, rb.source_ssrc());
+  EXPECT_EQ(kFractionLost, rb.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, rb.cumulative_lost());
+  EXPECT_EQ(kExtHighestSeqNum, rb.extended_high_seq_num());
+  EXPECT_EQ(kJitter, rb.jitter());
+  EXPECT_EQ(kLastSr, rb.last_sr());
+  EXPECT_EQ(kDelayLastSr, rb.delay_since_last_sr());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, ParseFailsOnIncorrectSize) {
+  RtcpCommonHeader header;
+  RtcpParseCommonHeader(kPacket, kPacketLength, &header);
+  header.count_or_format++;  // Damage the packet.
+  EXPECT_FALSE(rr.Parse(header, kPacket + RtcpCommonHeader::kHeaderSizeBytes));
+}
+
+TEST_F(RtcpPacketReceiverReportTest, Create) {
+  rr.From(kSenderSsrc);
+  ReportBlock rb;
+  rb.To(kRemoteSsrc);
+  rb.WithFractionLost(kFractionLost);
+  rb.WithCumulativeLost(kCumulativeLost);
+  rb.WithExtHighestSeqNum(kExtHighestSeqNum);
+  rb.WithJitter(kJitter);
+  rb.WithLastSr(kLastSr);
+  rb.WithDelayLastSr(kDelayLastSr);
+  rr.WithReportBlock(rb);
+
+  BuildPacket();
+
+  ASSERT_EQ(kPacketLength, packet->Length());
+  EXPECT_EQ(0, memcmp(kPacket, packet->Buffer(), kPacketLength));
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithoutReportBlocks) {
+  rr.From(kSenderSsrc);
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed().sender_ssrc());
+  EXPECT_EQ(0u, parsed().report_blocks().size());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithTwoReportBlocks) {
+  ReportBlock rb1;
+  rb1.To(kRemoteSsrc);
+  ReportBlock rb2;
+  rb2.To(kRemoteSsrc + 1);
+
+  rr.From(kSenderSsrc);
+  EXPECT_TRUE(rr.WithReportBlock(rb1));
+  EXPECT_TRUE(rr.WithReportBlock(rb2));
+
+  BuildPacket();
+  ParsePacket();
+
+  EXPECT_EQ(kSenderSsrc, parsed().sender_ssrc());
+  EXPECT_EQ(2u, parsed().report_blocks().size());
+  EXPECT_EQ(kRemoteSsrc, parsed().report_blocks()[0].source_ssrc());
+  EXPECT_EQ(kRemoteSsrc + 1, parsed().report_blocks()[1].source_ssrc());
+}
+
+TEST_F(RtcpPacketReceiverReportTest, WithTooManyReportBlocks) {
+  rr.From(kSenderSsrc);
+  const size_t kMaxReportBlocks = (1 << 5) - 1;
+  ReportBlock rb;
+  for (size_t i = 0; i < kMaxReportBlocks; ++i) {
+    rb.To(kRemoteSsrc + i);
+    EXPECT_TRUE(rr.WithReportBlock(rb));
+  }
+  rb.To(kRemoteSsrc + kMaxReportBlocks);
+  EXPECT_FALSE(rr.WithReportBlock(rb));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
new file mode 100644
index 0000000000..4911dbf5b7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTCP report block (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                 SSRC_1 (SSRC of first source)                 |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 | fraction lost |       cumulative number of packets lost       |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |           extended highest sequence number received           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                      interarrival jitter                      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                         last SR (LSR)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |                   delay since last SR (DLSR)                  |
+// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+ReportBlock::ReportBlock()
+    : source_ssrc_(0),
+      fraction_lost_(0),
+      cumulative_lost_(0),
+      extended_high_seq_num_(0),
+      jitter_(0),
+      last_sr_(0),
+      delay_since_last_sr_(0) {}
+
+bool ReportBlock::Parse(const uint8_t* buffer, size_t length) {
+  RTC_DCHECK(buffer != nullptr);
+  if (length < ReportBlock::kLength) {
+    LOG(LS_ERROR) << "Report Block should be 24 bytes long";
+    return false;
+  }
+
+  source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
+  fraction_lost_ = buffer[4];
+  cumulative_lost_ = ByteReader<uint32_t, 3>::ReadBigEndian(&buffer[5]);
+  extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
+  last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
+  delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);
+
+  return true;
+}
+
+void ReportBlock::Create(uint8_t* buffer) const {
+  // Runtime check should be done while setting cumulative_lost.
+  RTC_DCHECK_LT(cumulative_lost(), (1u << 24));  // Have only 3 bytes for it.
+
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc());
+  ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost());
+  ByteWriter<uint32_t, 3>::WriteBigEndian(&buffer[5], cumulative_lost());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], extended_high_seq_num());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[16], last_sr());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[20], delay_since_last_sr());
+}
+
+bool ReportBlock::WithCumulativeLost(uint32_t cumulative_lost) {
+  if (cumulative_lost >= (1u << 24)) {  // Have only 3 bytes to store it.
+    LOG(LS_WARNING) << "Cumulative lost is too big to fit into Report Block";
+    return false;
+  }
+  cumulative_lost_ = cumulative_lost;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
new file mode 100644
index 0000000000..ef99e17297
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+
+#include "webrtc/base/basictypes.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class ReportBlock {
+ public:
+  static const size_t kLength = 24;
+
+  ReportBlock();
+  ~ReportBlock() {}
+
+  bool Parse(const uint8_t* buffer, size_t length);
+
+  // Fills buffer with the ReportBlock.
+  // Consumes ReportBlock::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void To(uint32_t ssrc) { source_ssrc_ = ssrc; }
+  void WithFractionLost(uint8_t fraction_lost) {
+    fraction_lost_ = fraction_lost;
+  }
+  bool WithCumulativeLost(uint32_t cumulative_lost);
+  void WithExtHighestSeqNum(uint32_t ext_highest_seq_num) {
+    extended_high_seq_num_ = ext_highest_seq_num;
+  }
+  void WithJitter(uint32_t jitter) { jitter_ = jitter; }
+  void WithLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
+  void WithDelayLastSr(uint32_t delay_last_sr) {
+    delay_since_last_sr_ = delay_last_sr;
+  }
+
+  uint32_t source_ssrc() const { return source_ssrc_; }
+  uint8_t fraction_lost() const { return fraction_lost_; }
+  uint32_t cumulative_lost() const { return cumulative_lost_; }
+  uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
+  uint32_t jitter() const { return jitter_; }
+  uint32_t last_sr() const { return last_sr_; }
+  uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
+
+ private:
+  uint32_t source_ssrc_;
+  uint8_t fraction_lost_;
+  uint32_t cumulative_lost_;
+  uint32_t extended_high_seq_num_;
+  uint32_t jitter_;
+  uint32_t last_sr_;
+  uint32_t delay_since_last_sr_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
new file mode 100644
index 0000000000..85bbb404a4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+
+using webrtc::rtcp::ReportBlock;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+// Use values that are streamed differently LE and BE.
+const uint32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+const size_t kBufferLength = ReportBlock::kLength;
+
+TEST(RtcpPacketReportBlockTest, ParseChecksLength) {
+  uint8_t buffer[kBufferLength];
+  memset(buffer, 0, sizeof(buffer));
+
+  ReportBlock rb;
+  EXPECT_FALSE(rb.Parse(buffer, kBufferLength - 1));
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseAnyData) {
+  uint8_t buffer[kBufferLength];
+  // Fill buffer with semi-random data.
+  Random generator(0x256F8A285EC829ull);
+  for (size_t i = 0; i < kBufferLength; ++i)
+    buffer[i] = static_cast<uint8_t>(generator.Rand(0, 0xff));
+
+  ReportBlock rb;
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseMatchCreate) {
+  ReportBlock rb;
+  rb.To(kRemoteSsrc);
+  rb.WithFractionLost(kFractionLost);
+  rb.WithCumulativeLost(kCumulativeLost);
+  rb.WithExtHighestSeqNum(kExtHighestSeqNum);
+  rb.WithJitter(kJitter);
+  rb.WithLastSr(kLastSr);
+  rb.WithDelayLastSr(kDelayLastSr);
+
+  uint8_t buffer[kBufferLength];
+  rb.Create(buffer);
+
+  ReportBlock parsed;
+  EXPECT_TRUE(parsed.Parse(buffer, kBufferLength));
+
+  EXPECT_EQ(kRemoteSsrc, parsed.source_ssrc());
+  EXPECT_EQ(kFractionLost, parsed.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, parsed.cumulative_lost());
+  EXPECT_EQ(kExtHighestSeqNum, parsed.extended_high_seq_num());
+  EXPECT_EQ(kJitter, parsed.jitter());
+  EXPECT_EQ(kLastSr, parsed.last_sr());
+  EXPECT_EQ(kDelayLastSr, parsed.delay_since_last_sr());
+}
+
+TEST(RtcpPacketReportBlockTest, ValidateCumulativeLost) {
+  const uint32_t kMaxCumulativeLost = 0xffffff;
+  ReportBlock rb;
+  EXPECT_FALSE(rb.WithCumulativeLost(kMaxCumulativeLost + 1));
+  EXPECT_TRUE(rb.WithCumulativeLost(kMaxCumulativeLost));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
new file mode 100644
index 0000000000..db4ae67326
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+// Receiver Reference Time Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=4      |   reserved    |       block length = 2        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |              NTP timestamp, most significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             NTP timestamp, least significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void Rrtr::Parse(const uint8_t* buffer) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // reserved = buffer[1];
+  RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
+  uint32_t seconds = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  uint32_t fraction = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  ntp_.Set(seconds, fraction);
+}
+
+void Rrtr::Create(uint8_t* buffer) const {
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ntp_.seconds());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], ntp_.fractions());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
new file mode 100644
index 0000000000..3354f61df6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Rrtr {
+ public:
+  static const uint8_t kBlockType = 4;
+  static const uint16_t kBlockLength = 2;
+  static const size_t kLength = 4 * (kBlockLength + 1);  // 12
+
+  Rrtr() {}
+  Rrtr(const Rrtr&) = default;
+  ~Rrtr() {}
+
+  Rrtr& operator=(const Rrtr&) = default;
+
+  void Parse(const uint8_t* buffer);
+
+  // Fills buffer with the Rrtr.
+  // Consumes Rrtr::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void WithNtp(const NtpTime& ntp) { ntp_ = ntp; }
+
+  NtpTime ntp() const { return ntp_; }
+
+ private:
+  NtpTime ntp_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
new file mode 100644
index 0000000000..6536e06186
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+using webrtc::rtcp::Rrtr;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kNtpSec = 0x12345678;
+const uint32_t kNtpFrac = 0x23456789;
+const uint8_t kBlock[] = {0x04, 0x00, 0x00, 0x02,
+                          0x12, 0x34, 0x56, 0x78,
+                          0x23, 0x45, 0x67, 0x89};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+static_assert(
+    kBlockSizeBytes == Rrtr::kLength,
+    "Size of manually created Rrtr block should match class constant");
+
+TEST(RtcpPacketRrtrTest, Create) {
+  uint8_t buffer[Rrtr::kLength];
+  Rrtr rrtr;
+  rrtr.WithNtp(NtpTime(kNtpSec, kNtpFrac));
+
+  rrtr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketRrtrTest, Parse) {
+  Rrtr read_rrtr;
+  read_rrtr.Parse(kBlock);
+
+  // Run checks on const object to ensure all accessors have const modifier.
+  const Rrtr& parsed = read_rrtr;
+
+  EXPECT_EQ(kNtpSec, parsed.ntp().seconds());
+  EXPECT_EQ(kNtpFrac, parsed.ntp().fractions());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc
new file mode 100644
index 0000000000..b5571d45a3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RFC 4585, Section 6.1: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Rtpfb::ParseCommonFeedback(const uint8_t* payload) {
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  media_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+}
+
+void Rtpfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc_);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
new file mode 100644
index 0000000000..801aa085c4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RTPFB: Transport layer feedback message.
+// RFC4585, Section 6.2
+class Rtpfb : public RtcpPacket {
+ public:
+  static const uint8_t kPacketType = 205;
+
+  Rtpfb() : sender_ssrc_(0), media_ssrc_(0) {}
+  virtual ~Rtpfb() {}
+
+  void From(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  void To(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static const size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t sender_ssrc_;
+  uint32_t media_ssrc_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.cc
new file mode 100644
index 0000000000..829f3a9db9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+//
+// Slice loss indication (SLI) (RFC 4585).
+// FCI:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |            First        |        Number           | PictureID |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Sli::Macroblocks::Macroblocks(uint8_t picture_id,
+                              uint16_t first,
+                              uint16_t number) {
+  RTC_DCHECK_LE(first, 0x1fff);
+  RTC_DCHECK_LE(number, 0x1fff);
+  RTC_DCHECK_LE(picture_id, 0x3f);
+  item_ = (first << 19) | (number << 6) | picture_id;
+}
+
+void Sli::Macroblocks::Parse(const uint8_t* buffer) {
+  item_ = ByteReader<uint32_t>::ReadBigEndian(buffer);
+}
+
+void Sli::Macroblocks::Create(uint8_t* buffer) const {
+  ByteWriter<uint32_t>::WriteBigEndian(buffer, item_);
+}
+
+bool Sli::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+
+  if (header.payload_size_bytes <
+      kCommonFeedbackLength + Macroblocks::kLength) {
+    LOG(LS_WARNING) << "Packet is too small to be a valid SLI packet";
+    return false;
+  }
+
+  size_t number_of_items =
+      (header.payload_size_bytes - kCommonFeedbackLength) /
+      Macroblocks::kLength;
+
+  ParseCommonFeedback(payload);
+  items_.resize(number_of_items);
+
+  const uint8_t* next_item = payload + kCommonFeedbackLength;
+  for (Macroblocks& item : items_) {
+    item.Parse(next_item);
+    next_item += Macroblocks::kLength;
+  }
+
+  return true;
+}
+
+bool Sli::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 RtcpPacket::PacketReadyCallback* callback) const {
+  RTC_DCHECK(!items_.empty());
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  for (const Macroblocks& item : items_) {
+    item.Create(packet + *index);
+    *index += Macroblocks::kLength;
+  }
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
new file mode 100644
index 0000000000..5d9e6c93e9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SLI_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SLI_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Slice loss indication (SLI) (RFC 4585).
+class Sli : public Psfb {
+ public:
+  static const uint8_t kFeedbackMessageType = 2;
+  class Macroblocks {
+   public:
+    static const size_t kLength = 4;
+    Macroblocks() : item_(0) {}
+    Macroblocks(uint8_t picture_id, uint16_t first, uint16_t number);
+    ~Macroblocks() {}
+
+    void Parse(const uint8_t* buffer);
+    void Create(uint8_t* buffer) const;
+
+    uint16_t first() const { return item_ >> 19; }
+    uint16_t number() const { return (item_ >> 6) & 0x1fff; }
+    uint8_t picture_id() const { return (item_ & 0x3f); }
+
+   private:
+    uint32_t item_;
+  };
+
+  Sli() {}
+  virtual ~Sli() {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
+             const uint8_t* payload);  // Size of the payload is in the header.
+
+  void WithPictureId(uint8_t picture_id,
+                     uint16_t first_macroblock = 0,
+                     uint16_t number_macroblocks = 0x1fff) {
+    items_.push_back(
+        Macroblocks(picture_id, first_macroblock, number_macroblocks));
+  }
+
+  const std::vector<Macroblocks>& macroblocks() const { return items_; }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override {
+    return RtcpPacket::kHeaderLength + Psfb::kCommonFeedbackLength +
+           items_.size() * Macroblocks::kLength;
+  }
+
+  std::vector<Macroblocks> items_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Sli);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SLI_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli_unittest.cc
new file mode 100644
index 0000000000..c2be16846b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli_unittest.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using testing::ElementsAreArray;
+using testing::make_tuple;
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::Sli;
+using webrtc::RTCPUtility::RtcpCommonHeader;
+using webrtc::RTCPUtility::RtcpParseCommonHeader;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+const uint8_t kPictureId = 0x3f;
+const uint16_t kFirstMb = 0x1e61;
+const uint16_t kNumberOfMb = 0x1a0a;
+const uint32_t kSliItem = (static_cast<uint32_t>(kFirstMb) << 19) |
+                          (static_cast<uint32_t>(kNumberOfMb) << 6) |
+                           static_cast<uint32_t>(kPictureId);
+
+// Manually created Sli packet matching constants above.
+const uint8_t kPacket[] = {0x82,  206, 0x00, 0x03,
+                           0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89,
+                           (kSliItem >> 24) & 0xff,
+                           (kSliItem >> 16) & 0xff,
+                           (kSliItem >> 8) & 0xff,
+                           kSliItem & 0xff};
+const size_t kPacketLength = sizeof(kPacket);
+
+bool ParseSli(const uint8_t* buffer, size_t length, Sli* sli) {
+  RtcpCommonHeader header;
+  EXPECT_TRUE(RtcpParseCommonHeader(buffer, length, &header));
+  EXPECT_EQ(length, header.BlockSize());
+  return sli->Parse(header, buffer + RtcpCommonHeader::kHeaderSizeBytes);
+}
+
+TEST(RtcpPacketSliTest, Create) {
+  Sli sli;
+  sli.From(kSenderSsrc);
+  sli.To(kRemoteSsrc);
+  sli.WithPictureId(kPictureId, kFirstMb, kNumberOfMb);
+
+  rtc::scoped_ptr<RawPacket> packet(sli.Build());
+
+  EXPECT_THAT(make_tuple(packet->Buffer(), packet->Length()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketSliTest, Parse) {
+  Sli mutable_parsed;
+  EXPECT_TRUE(ParseSli(kPacket, kPacketLength, &mutable_parsed));
+  const Sli& parsed = mutable_parsed;  // Read values from constant object.
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_EQ(1u, parsed.macroblocks().size());
+  EXPECT_EQ(kFirstMb, parsed.macroblocks()[0].first());
+  EXPECT_EQ(kNumberOfMb, parsed.macroblocks()[0].number());
+  EXPECT_EQ(kPictureId, parsed.macroblocks()[0].picture_id());
+}
+
+TEST(RtcpPacketSliTest, ParseFailsOnTooSmallPacket) {
+  Sli sli;
+  sli.From(kSenderSsrc);
+  sli.To(kRemoteSsrc);
+  sli.WithPictureId(kPictureId, kFirstMb, kNumberOfMb);
+
+  rtc::scoped_ptr<RawPacket> packet(sli.Build());
+  packet->MutableBuffer()[3]--;  // Decrease size by 1 word (4 bytes).
+
+  EXPECT_FALSE(ParseSli(packet->Buffer(), packet->Length() - 4, &sli));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
new file mode 100644
index 0000000000..fd0219cf82
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::PT_RTPFB;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBN;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem;
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+const uint32_t kUnusedMediaSourceSsrc0 = 0;
+void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) {
+  buffer[(*offset)++] = value;
+}
+void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) {
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + *offset, value);
+  *offset += 4;
+}
+
+void ComputeMantissaAnd6bitBase2Exponent(uint32_t input_base10,
+                                         uint8_t bits_mantissa,
+                                         uint32_t* mantissa,
+                                         uint8_t* exp) {
+  // input_base10 = mantissa * 2^exp
+  assert(bits_mantissa <= 32);
+  uint32_t mantissa_max = (1 << bits_mantissa) - 1;
+  uint8_t exponent = 0;
+  for (uint32_t i = 0; i < 64; ++i) {
+    if (input_base10 <= (mantissa_max << i)) {
+      exponent = i;
+      break;
+    }
+  }
+  *exp = exponent;
+  *mantissa = (input_base10 >> exponent);
+}
+
+void CreateTmmbrItem(const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
+                     uint8_t* buffer,
+                     size_t* pos) {
+  uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000;
+  uint32_t mantissa = 0;
+  uint8_t exp = 0;
+  ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
+
+  AssignUWord32(buffer, pos, tmmbr_item.SSRC);
+  AssignUWord8(buffer, pos, (exp << 2) + ((mantissa >> 15) & 0x03));
+  AssignUWord8(buffer, pos, mantissa >> 7);
+  AssignUWord8(buffer, pos, (mantissa << 1) +
+                            ((tmmbr_item.MeasuredOverhead >> 8) & 0x01));
+  AssignUWord8(buffer, pos, tmmbr_item.MeasuredOverhead);
+}
+
+// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
+//
+// FCI:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                              SSRC                             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void CreateTmmbn(const RTCPPacketRTPFBTMMBN& tmmbn,
+                 const std::vector<RTCPPacketRTPFBTMMBRItem>& tmmbn_items,
+                 uint8_t* buffer,
+                 size_t* pos) {
+  AssignUWord32(buffer, pos, tmmbn.SenderSSRC);
+  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
+  for (uint8_t i = 0; i < tmmbn_items.size(); ++i) {
+    CreateTmmbrItem(tmmbn_items[i], buffer, pos);
+  }
+}
+}  // namespace
+
+bool Tmmbn::WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead) {
+  assert(overhead <= 0x1ff);
+  if (tmmbn_items_.size() >= kMaxNumberOfTmmbrs) {
+    LOG(LS_WARNING) << "Max TMMBN size reached.";
+    return false;
+  }
+  RTCPPacketRTPFBTMMBRItem tmmbn_item;
+  tmmbn_item.SSRC = ssrc;
+  tmmbn_item.MaxTotalMediaBitRate = bitrate_kbps;
+  tmmbn_item.MeasuredOverhead = overhead;
+  tmmbn_items_.push_back(tmmbn_item);
+  return true;
+}
+
+bool Tmmbn::Create(uint8_t* packet,
+                   size_t* index,
+                   size_t max_length,
+                   RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const uint8_t kFmt = 4;
+  CreateHeader(kFmt, PT_RTPFB, HeaderLength(), packet, index);
+  CreateTmmbn(tmmbn_, tmmbn_items_, packet, index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
new file mode 100644
index 0000000000..82bf9dd9e9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
+
+#include <vector>
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
+class Tmmbn : public RtcpPacket {
+ public:
+  Tmmbn() : RtcpPacket() {
+    memset(&tmmbn_, 0, sizeof(tmmbn_));
+  }
+
+  virtual ~Tmmbn() {}
+
+  void From(uint32_t ssrc) {
+    tmmbn_.SenderSSRC = ssrc;
+  }
+  // Max 50 TMMBR can be added per TMMBN.
+  bool WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead);
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfTmmbrs = 50;
+
+  size_t BlockLength() const {
+    const size_t kFciLen = 8;
+    return kCommonFbFmtLength + kFciLen * tmmbn_items_.size();
+  }
+
+  RTCPUtility::RTCPPacketRTPFBTMMBN tmmbn_;
+  std::vector<RTCPUtility::RTCPPacketRTPFBTMMBRItem> tmmbn_items_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Tmmbn);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
new file mode 100644
index 0000000000..32d64a97b4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::Tmmbn;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+TEST(RtcpPacketTest, TmmbnWithNoItem) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
+  EXPECT_EQ(0, parser.tmmbn_items()->num_packets());
+}
+
+TEST(RtcpPacketTest, TmmbnWithOneItem) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc, 312, 60));
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
+  EXPECT_EQ(1, parser.tmmbn_items()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
+  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
+  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
+}
+
+TEST(RtcpPacketTest, TmmbnWithTwoItems) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc, 312, 60));
+  EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc + 1, 1288, 40));
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbn.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
+  EXPECT_EQ(2, parser.tmmbn_items()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
+  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
+  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
+  EXPECT_EQ(kRemoteSsrc + 1, parser.tmmbn_items()->Ssrc(1));
+  EXPECT_EQ(1288U, parser.tmmbn_items()->BitrateKbps(1));
+  EXPECT_EQ(40U, parser.tmmbn_items()->Overhead(1));
+}
+
+TEST(RtcpPacketTest, TmmbnWithTooManyItems) {
+  Tmmbn tmmbn;
+  tmmbn.From(kSenderSsrc);
+  const int kMaxTmmbrItems = 50;
+  for (int i = 0; i < kMaxTmmbrItems; ++i)
+    EXPECT_TRUE(tmmbn.WithTmmbr(kRemoteSsrc + i, 312, 60));
+
+  EXPECT_FALSE(tmmbn.WithTmmbr(kRemoteSsrc + kMaxTmmbrItems, 312, 60));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
new file mode 100644
index 0000000000..4df167de79
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::PT_RTPFB;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBR;
+using webrtc::RTCPUtility::RTCPPacketRTPFBTMMBRItem;
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+const uint32_t kUnusedMediaSourceSsrc0 = 0;
+
+void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) {
+  buffer[(*offset)++] = value;
+}
+
+void AssignUWord32(uint8_t* buffer, size_t* offset, uint32_t value) {
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + *offset, value);
+  *offset += 4;
+}
+
+void ComputeMantissaAnd6bitBase2Exponent(uint32_t input_base10,
+                                         uint8_t bits_mantissa,
+                                         uint32_t* mantissa,
+                                         uint8_t* exp) {
+  // input_base10 = mantissa * 2^exp
+  assert(bits_mantissa <= 32);
+  uint32_t mantissa_max = (1 << bits_mantissa) - 1;
+  uint8_t exponent = 0;
+  for (uint32_t i = 0; i < 64; ++i) {
+    if (input_base10 <= (mantissa_max << i)) {
+      exponent = i;
+      break;
+    }
+  }
+  *exp = exponent;
+  *mantissa = (input_base10 >> exponent);
+}
+
+void CreateTmmbrItem(const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
+                     uint8_t* buffer,
+                     size_t* pos) {
+  uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000;
+  uint32_t mantissa = 0;
+  uint8_t exp = 0;
+  ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
+
+  AssignUWord32(buffer, pos, tmmbr_item.SSRC);
+  AssignUWord8(buffer, pos, (exp << 2) + ((mantissa >> 15) & 0x03));
+  AssignUWord8(buffer, pos, mantissa >> 7);
+  AssignUWord8(buffer, pos, (mantissa << 1) +
+                            ((tmmbr_item.MeasuredOverhead >> 8) & 0x01));
+  AssignUWord8(buffer, pos, tmmbr_item.MeasuredOverhead);
+}
+
+// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
+//
+// FCI:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                              SSRC                             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void CreateTmmbr(const RTCPPacketRTPFBTMMBR& tmmbr,
+                 const RTCPPacketRTPFBTMMBRItem& tmmbr_item,
+                 uint8_t* buffer,
+                 size_t* pos) {
+  AssignUWord32(buffer, pos, tmmbr.SenderSSRC);
+  AssignUWord32(buffer, pos, kUnusedMediaSourceSsrc0);
+  CreateTmmbrItem(tmmbr_item, buffer, pos);
+}
+}  // namespace
+
+bool Tmmbr::Create(uint8_t* packet,
+                   size_t* index,
+                   size_t max_length,
+                   RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const uint8_t kFmt = 3;
+  CreateHeader(kFmt, PT_RTPFB, HeaderLength(), packet, index);
+  CreateTmmbr(tmmbr_, tmmbr_item_, packet, index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
new file mode 100644
index 0000000000..cb97ea3237
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
+class Tmmbr : public RtcpPacket {
+ public:
+  Tmmbr() : RtcpPacket() {
+    memset(&tmmbr_, 0, sizeof(tmmbr_));
+    memset(&tmmbr_item_, 0, sizeof(tmmbr_item_));
+  }
+
+  virtual ~Tmmbr() {}
+
+  void From(uint32_t ssrc) {
+    tmmbr_.SenderSSRC = ssrc;
+  }
+  void To(uint32_t ssrc) {
+    tmmbr_item_.SSRC = ssrc;
+  }
+  void WithBitrateKbps(uint32_t bitrate_kbps) {
+    tmmbr_item_.MaxTotalMediaBitRate = bitrate_kbps;
+  }
+  void WithOverhead(uint16_t overhead) {
+    assert(overhead <= 0x1ff);
+    tmmbr_item_.MeasuredOverhead = overhead;
+  }
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  size_t BlockLength() const override {
+    const size_t kFciLen = 8;
+    return kCommonFbFmtLength + kFciLen;
+  }
+
+  RTCPUtility::RTCPPacketRTPFBTMMBR tmmbr_;
+  RTCPUtility::RTCPPacketRTPFBTMMBRItem tmmbr_item_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Tmmbr);
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
new file mode 100644
index 0000000000..6d71caa251
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::RawPacket;
+using webrtc::rtcp::Tmmbr;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+
+TEST(RtcpPacketTest, Tmmbr) {
+  Tmmbr tmmbr;
+  tmmbr.From(kSenderSsrc);
+  tmmbr.To(kRemoteSsrc);
+  tmmbr.WithBitrateKbps(312);
+  tmmbr.WithOverhead(60);
+
+  rtc::scoped_ptr<RawPacket> packet(tmmbr.Build());
+  RtcpPacketParser parser;
+  parser.Parse(packet->Buffer(), packet->Length());
+  EXPECT_EQ(1, parser.tmmbr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.tmmbr()->Ssrc());
+  EXPECT_EQ(1, parser.tmmbr_item()->num_packets());
+  EXPECT_EQ(312U, parser.tmmbr_item()->BitrateKbps());
+  EXPECT_EQ(60U, parser.tmmbr_item()->Overhead());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
new file mode 100644
index 0000000000..4ad49561b8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
@@ -0,0 +1,776 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Header size:
+// * 12 bytes Common Packet Format for RTCP Feedback Messages
+// * 8 bytes FeedbackPacket header
+static const uint32_t kHeaderSizeBytes = 12 + 8;
+static const uint32_t kChunkSizeBytes = 2;
+static const uint32_t kOneBitVectorCapacity = 14;
+static const uint32_t kTwoBitVectorCapacity = 7;
+static const uint32_t kRunLengthCapacity = 0x1FFF;
+// TODO(sprang): Add support for dynamic max size for easier fragmentation,
+// eg. set it to what's left in the buffer or IP_PACKET_SIZE.
+// Size constraint imposed by RTCP common header: 16bit size field interpreted
+// as number of four byte words minus the first header word.
+static const uint32_t kMaxSizeBytes = (1 << 16) * 4;
+static const uint32_t kMinSizeBytes = kHeaderSizeBytes + kChunkSizeBytes;
+static const uint32_t kBaseScaleFactor =
+    TransportFeedback::kDeltaScaleFactor * (1 << 8);
+
+class PacketStatusChunk {
+ public:
+  virtual ~PacketStatusChunk() {}
+  virtual uint16_t NumSymbols() const = 0;
+  virtual void AppendSymbolsTo(
+      std::vector<TransportFeedback::StatusSymbol>* vec) const = 0;
+  virtual void WriteTo(uint8_t* buffer) const = 0;
+};
+
+uint8_t EncodeSymbol(TransportFeedback::StatusSymbol symbol) {
+  switch (symbol) {
+    case TransportFeedback::StatusSymbol::kNotReceived:
+      return 0;
+    case TransportFeedback::StatusSymbol::kReceivedSmallDelta:
+      return 1;
+    case TransportFeedback::StatusSymbol::kReceivedLargeDelta:
+      return 2;
+    default:
+      RTC_NOTREACHED();
+      return 0;
+  }
+}
+
+TransportFeedback::StatusSymbol DecodeSymbol(uint8_t value) {
+  switch (value) {
+    case 0:
+      return TransportFeedback::StatusSymbol::kNotReceived;
+    case 1:
+      return TransportFeedback::StatusSymbol::kReceivedSmallDelta;
+    case 2:
+      return TransportFeedback::StatusSymbol::kReceivedLargeDelta;
+    default:
+      RTC_NOTREACHED();
+      return TransportFeedback::StatusSymbol::kNotReceived;
+  }
+}
+
+TransportFeedback::TransportFeedback()
+    : packet_sender_ssrc_(0),
+      media_source_ssrc_(0),
+      base_seq_(-1),
+      base_time_(-1),
+      feedback_seq_(0),
+      last_seq_(-1),
+      last_timestamp_(-1),
+      first_symbol_cardinality_(0),
+      vec_needs_two_bit_symbols_(false),
+      size_bytes_(kHeaderSizeBytes) {
+}
+
+TransportFeedback::~TransportFeedback() {
+  for (PacketStatusChunk* chunk : status_chunks_)
+    delete chunk;
+}
+
+//  One Bit Status Vector Chunk
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |T|S|       symbol list         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  T = 1
+//  S = 0
+//  symbol list = 14 entries where 0 = not received, 1 = received
+
+class OneBitVectorChunk : public PacketStatusChunk {
+ public:
+  static const int kCapacity = 14;
+
+  explicit OneBitVectorChunk(
+      std::deque<TransportFeedback::StatusSymbol>* symbols) {
+    size_t input_size = symbols->size();
+    for (size_t i = 0; i < kCapacity; ++i) {
+      if (i < input_size) {
+        symbols_[i] = symbols->front();
+        symbols->pop_front();
+      } else {
+        symbols_[i] = TransportFeedback::StatusSymbol::kNotReceived;
+      }
+    }
+  }
+
+  virtual ~OneBitVectorChunk() {}
+
+  uint16_t NumSymbols() const override { return kCapacity; }
+
+  void AppendSymbolsTo(
+      std::vector<TransportFeedback::StatusSymbol>* vec) const override {
+    vec->insert(vec->end(), &symbols_[0], &symbols_[kCapacity]);
+  }
+
+  void WriteTo(uint8_t* buffer) const override {
+    const int kSymbolsInFirstByte = 6;
+    const int kSymbolsInSecondByte = 8;
+    buffer[0] = 0x80u;
+    for (int i = 0; i < kSymbolsInFirstByte; ++i) {
+      uint8_t encoded_symbol = EncodeSymbol(symbols_[i]);
+      RTC_DCHECK_LE(encoded_symbol, 1u);
+      buffer[0] |= encoded_symbol << (kSymbolsInFirstByte - (i + 1));
+    }
+    buffer[1] = 0x00u;
+    for (int i = 0; i < kSymbolsInSecondByte; ++i) {
+      uint8_t encoded_symbol = EncodeSymbol(symbols_[i + kSymbolsInFirstByte]);
+      RTC_DCHECK_LE(encoded_symbol, 1u);
+      buffer[1] |= encoded_symbol << (kSymbolsInSecondByte - (i + 1));
+    }
+  }
+
+  static OneBitVectorChunk* ParseFrom(const uint8_t* data) {
+    OneBitVectorChunk* chunk = new OneBitVectorChunk();
+
+    size_t index = 0;
+    for (int i = 5; i >= 0; --i)  // Last 5 bits from first byte.
+      chunk->symbols_[index++] = DecodeSymbol((data[0] >> i) & 0x01);
+    for (int i = 7; i >= 0; --i)  // 8 bits from the last byte.
+      chunk->symbols_[index++] = DecodeSymbol((data[1] >> i) & 0x01);
+
+    return chunk;
+  }
+
+ private:
+  OneBitVectorChunk() {}
+
+  TransportFeedback::StatusSymbol symbols_[kCapacity];
+};
+
+//  Two Bit Status Vector Chunk
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |T|S|       symbol list         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  T = 1
+//  S = 1
+//  symbol list = 7 entries of two bits each, see (Encode|Decode)Symbol
+
+class TwoBitVectorChunk : public PacketStatusChunk {
+ public:
+  static const int kCapacity = 7;
+
+  explicit TwoBitVectorChunk(
+      std::deque<TransportFeedback::StatusSymbol>* symbols) {
+    size_t input_size = symbols->size();
+    for (size_t i = 0; i < kCapacity; ++i) {
+      if (i < input_size) {
+        symbols_[i] = symbols->front();
+        symbols->pop_front();
+      } else {
+        symbols_[i] = TransportFeedback::StatusSymbol::kNotReceived;
+      }
+    }
+  }
+
+  virtual ~TwoBitVectorChunk() {}
+
+  uint16_t NumSymbols() const override { return kCapacity; }
+
+  void AppendSymbolsTo(
+      std::vector<TransportFeedback::StatusSymbol>* vec) const override {
+    vec->insert(vec->end(), &symbols_[0], &symbols_[kCapacity]);
+  }
+
+  void WriteTo(uint8_t* buffer) const override {
+    buffer[0] = 0xC0;
+    buffer[0] |= EncodeSymbol(symbols_[0]) << 4;
+    buffer[0] |= EncodeSymbol(symbols_[1]) << 2;
+    buffer[0] |= EncodeSymbol(symbols_[2]);
+    buffer[1] = EncodeSymbol(symbols_[3]) << 6;
+    buffer[1] |= EncodeSymbol(symbols_[4]) << 4;
+    buffer[1] |= EncodeSymbol(symbols_[5]) << 2;
+    buffer[1] |= EncodeSymbol(symbols_[6]);
+  }
+
+  static TwoBitVectorChunk* ParseFrom(const uint8_t* buffer) {
+    TwoBitVectorChunk* chunk = new TwoBitVectorChunk();
+
+    chunk->symbols_[0] = DecodeSymbol((buffer[0] >> 4) & 0x03);
+    chunk->symbols_[1] = DecodeSymbol((buffer[0] >> 2) & 0x03);
+    chunk->symbols_[2] = DecodeSymbol(buffer[0] & 0x03);
+    chunk->symbols_[3] = DecodeSymbol((buffer[1] >> 6) & 0x03);
+    chunk->symbols_[4] = DecodeSymbol((buffer[1] >> 4) & 0x03);
+    chunk->symbols_[5] = DecodeSymbol((buffer[1] >> 2) & 0x03);
+    chunk->symbols_[6] = DecodeSymbol(buffer[1] & 0x03);
+
+    return chunk;
+  }
+
+ private:
+  TwoBitVectorChunk() {}
+
+  TransportFeedback::StatusSymbol symbols_[kCapacity];
+};
+
+//  Two Bit Status Vector Chunk
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |T| S |       Run Length        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  T = 0
+//  S = symbol, see (Encode|Decode)Symbol
+//  Run Length = Unsigned integer denoting the run length of the symbol
+
+class RunLengthChunk : public PacketStatusChunk {
+ public:
+  RunLengthChunk(TransportFeedback::StatusSymbol symbol, size_t size)
+      : symbol_(symbol), size_(size) {
+    RTC_DCHECK_LE(size, 0x1FFFu);
+  }
+
+  virtual ~RunLengthChunk() {}
+
+  uint16_t NumSymbols() const override { return size_; }
+
+  void AppendSymbolsTo(
+      std::vector<TransportFeedback::StatusSymbol>* vec) const override {
+    vec->insert(vec->end(), size_, symbol_);
+  }
+
+  void WriteTo(uint8_t* buffer) const override {
+    buffer[0] = EncodeSymbol(symbol_) << 5;  // Write S (T = 0 implicitly)
+    buffer[0] |= (size_ >> 8) & 0x1F;  // 5 most significant bits of run length.
+    buffer[1] = size_ & 0xFF;  // 8 least significant bits of run length.
+  }
+
+  static RunLengthChunk* ParseFrom(const uint8_t* buffer) {
+    RTC_DCHECK_EQ(0, buffer[0] & 0x80);
+    TransportFeedback::StatusSymbol symbol =
+        DecodeSymbol((buffer[0] >> 5) & 0x03);
+    uint16_t count = (static_cast<uint16_t>(buffer[0] & 0x1F) << 8) | buffer[1];
+
+    return new RunLengthChunk(symbol, count);
+  }
+
+ private:
+  const TransportFeedback::StatusSymbol symbol_;
+  const size_t size_;
+};
+
+// Unwrap to a larger type, for easier handling of wraps.
+int64_t TransportFeedback::Unwrap(uint16_t sequence_number) {
+  if (last_seq_ == -1)
+    return sequence_number;
+
+  int64_t delta = sequence_number - last_seq_;
+  if (IsNewerSequenceNumber(sequence_number,
+                            static_cast<uint16_t>(last_seq_))) {
+    if (delta < 0)
+      delta += (1 << 16);
+  } else if (delta > 0) {
+    delta -= (1 << 16);
+  }
+
+  return last_seq_ + delta;
+}
+
+void TransportFeedback::WithPacketSenderSsrc(uint32_t ssrc) {
+  packet_sender_ssrc_ = ssrc;
+}
+
+void TransportFeedback::WithMediaSourceSsrc(uint32_t ssrc) {
+  media_source_ssrc_ = ssrc;
+}
+
+uint32_t TransportFeedback::GetPacketSenderSsrc() const {
+  return packet_sender_ssrc_;
+}
+
+uint32_t TransportFeedback::GetMediaSourceSsrc() const {
+  return media_source_ssrc_;
+}
+void TransportFeedback::WithBase(uint16_t base_sequence,
+                                 int64_t ref_timestamp_us) {
+  RTC_DCHECK_EQ(-1, base_seq_);
+  RTC_DCHECK_NE(-1, ref_timestamp_us);
+  base_seq_ = base_sequence;
+  last_seq_ = base_sequence;
+  base_time_ = ref_timestamp_us / kBaseScaleFactor;
+  last_timestamp_ = base_time_ * kBaseScaleFactor;
+}
+
+void TransportFeedback::WithFeedbackSequenceNumber(uint8_t feedback_sequence) {
+  feedback_seq_ = feedback_sequence;
+}
+
+bool TransportFeedback::WithReceivedPacket(uint16_t sequence_number,
+                                           int64_t timestamp) {
+  RTC_DCHECK_NE(-1, base_seq_);
+  int64_t seq = Unwrap(sequence_number);
+  if (seq != base_seq_ && seq <= last_seq_)
+    return false;
+
+  // Convert to ticks and round.
+  int64_t delta_full = timestamp - last_timestamp_;
+  delta_full +=
+      delta_full < 0 ? -(kDeltaScaleFactor / 2) : kDeltaScaleFactor / 2;
+  delta_full /= kDeltaScaleFactor;
+
+  int16_t delta = static_cast<int16_t>(delta_full);
+  // If larger than 16bit signed, we can't represent it - need new fb packet.
+  if (delta != delta_full) {
+    LOG(LS_WARNING) << "Delta value too large ( >= 2^16 ticks )";
+    return false;
+  }
+
+  StatusSymbol symbol;
+  if (delta >= 0 && delta <= 0xFF) {
+    symbol = StatusSymbol::kReceivedSmallDelta;
+  } else {
+    symbol = StatusSymbol::kReceivedLargeDelta;
+  }
+
+  if (!AddSymbol(symbol, seq))
+    return false;
+
+  receive_deltas_.push_back(delta);
+  last_timestamp_ += delta * kDeltaScaleFactor;
+  return true;
+}
+
+// Add a symbol for a received packet, with the given sequence number. This
+// method will add any "packet not received" symbols needed before this one.
+bool TransportFeedback::AddSymbol(StatusSymbol symbol, int64_t seq) {
+  while (last_seq_ < seq - 1) {
+    if (!Encode(StatusSymbol::kNotReceived))
+      return false;
+    ++last_seq_;
+  }
+
+  if (!Encode(symbol))
+    return false;
+
+  last_seq_ = seq;
+  return true;
+}
+
+// Append a symbol to the internal symbol vector. If the new state cannot be
+// represented using a single status chunk, a chunk will first be emitted and
+// the associated symbols removed from the internal symbol vector.
+bool TransportFeedback::Encode(StatusSymbol symbol) {
+  if (last_seq_ - base_seq_ + 1 > 0xFFFF) {
+    LOG(LS_WARNING) << "Packet status count too large ( >= 2^16 )";
+    return false;
+  }
+
+  bool is_two_bit;
+  int delta_size;
+  switch (symbol) {
+    case StatusSymbol::kReceivedSmallDelta:
+      delta_size = 1;
+      is_two_bit = false;
+      break;
+    case StatusSymbol::kReceivedLargeDelta:
+      delta_size = 2;
+      is_two_bit = true;
+      break;
+    case StatusSymbol::kNotReceived:
+      is_two_bit = false;
+      delta_size = 0;
+      break;
+    default:
+      RTC_NOTREACHED();
+      return false;
+  }
+
+  if (symbol_vec_.empty()) {
+    if (size_bytes_ + delta_size + kChunkSizeBytes > kMaxSizeBytes)
+      return false;
+
+    symbol_vec_.push_back(symbol);
+    vec_needs_two_bit_symbols_ = is_two_bit;
+    first_symbol_cardinality_ = 1;
+    size_bytes_ += delta_size + kChunkSizeBytes;
+    return true;
+  }
+  if (size_bytes_ + delta_size > kMaxSizeBytes)
+    return false;
+
+  // Capacity, in number of symbols, that a vector chunk could hold.
+  size_t capacity = vec_needs_two_bit_symbols_ ? kTwoBitVectorCapacity
+                                               : kOneBitVectorCapacity;
+
+  // first_symbol_cardinality_ is the number of times the first symbol in
+  // symbol_vec is repeated. So if that is equal to the size of symbol_vec,
+  // there is only one kind of symbol - we can potentially RLE encode it.
+  // If we have less than (capacity) symbols in symbol_vec, we can't know
+  // for certain this will be RLE-encoded; if a different symbol is added
+  // these symbols will be needed to emit a vector chunk instead. However,
+  // if first_symbol_cardinality_ > capacity, then we cannot encode the
+  // current state as a vector chunk - we must first emit symbol_vec as an
+  // RLE-chunk and then add the new symbol.
+  bool rle_candidate = symbol_vec_.size() == first_symbol_cardinality_ ||
+                       first_symbol_cardinality_ > capacity;
+  if (rle_candidate) {
+    if (symbol_vec_.back() == symbol) {
+      ++first_symbol_cardinality_;
+      if (first_symbol_cardinality_ <= capacity) {
+        symbol_vec_.push_back(symbol);
+      } else if (first_symbol_cardinality_ == kRunLengthCapacity) {
+        // Max length for an RLE-chunk reached.
+        EmitRunLengthChunk();
+      }
+      size_bytes_ += delta_size;
+      return true;
+    } else {
+      // New symbol does not match what's already in symbol_vec.
+      if (first_symbol_cardinality_ >= capacity) {
+        // Symbols in symbol_vec can only be RLE-encoded. Emit the RLE-chunk
+        // and re-add input. symbol_vec is then guaranteed to have room for the
+        // symbol, so recursion cannot continue.
+        EmitRunLengthChunk();
+        return Encode(symbol);
+      }
+      // Fall through and treat state as non RLE-candidate.
+    }
+  }
+
+  // If this code point is reached, symbols in symbol_vec cannot be RLE-encoded.
+
+  if (is_two_bit && !vec_needs_two_bit_symbols_) {
+    // If the symbols in symbol_vec can be encoded using a one-bit chunk but
+    // the input symbol cannot, first check if we can simply change target type.
+    vec_needs_two_bit_symbols_ = true;
+    if (symbol_vec_.size() >= kTwoBitVectorCapacity) {
+      // symbol_vec contains more symbols than we can encode in a single
+      // two-bit chunk. Emit a new vector append to the remains, if any.
+      if (size_bytes_ + delta_size + kChunkSizeBytes > kMaxSizeBytes)
+        return false;
+      EmitVectorChunk();
+      // If symbol_vec isn't empty after emitting a vector chunk, we need to
+      // account for chunk size (otherwise handled by Encode method).
+      if (!symbol_vec_.empty())
+        size_bytes_ += kChunkSizeBytes;
+      return Encode(symbol);
+    }
+    // symbol_vec symbols fit within a single two-bit vector chunk.
+    capacity = kTwoBitVectorCapacity;
+  }
+
+  symbol_vec_.push_back(symbol);
+  if (symbol_vec_.size() == capacity)
+    EmitVectorChunk();
+
+  size_bytes_ += delta_size;
+  return true;
+}
+
+// Upon packet completion, emit any remaining symbols in symbol_vec that have
+// not yet been emitted in a status chunk.
+void TransportFeedback::EmitRemaining() {
+  if (symbol_vec_.empty())
+    return;
+
+  size_t capacity = vec_needs_two_bit_symbols_ ? kTwoBitVectorCapacity
+                                               : kOneBitVectorCapacity;
+  if (first_symbol_cardinality_ > capacity) {
+    EmitRunLengthChunk();
+  } else {
+    EmitVectorChunk();
+  }
+}
+
+void TransportFeedback::EmitVectorChunk() {
+  if (vec_needs_two_bit_symbols_) {
+    status_chunks_.push_back(new TwoBitVectorChunk(&symbol_vec_));
+  } else {
+    status_chunks_.push_back(new OneBitVectorChunk(&symbol_vec_));
+  }
+  // Update first symbol cardinality to match what is potentially left in in
+  // symbol_vec.
+  first_symbol_cardinality_ = 1;
+  for (size_t i = 1; i < symbol_vec_.size(); ++i) {
+    if (symbol_vec_[i] != symbol_vec_[0])
+      break;
+    ++first_symbol_cardinality_;
+  }
+}
+
+void TransportFeedback::EmitRunLengthChunk() {
+  RTC_DCHECK_GE(first_symbol_cardinality_, symbol_vec_.size());
+  status_chunks_.push_back(
+      new RunLengthChunk(symbol_vec_.front(), first_symbol_cardinality_));
+  symbol_vec_.clear();
+}
+
+size_t TransportFeedback::BlockLength() const {
+  return size_bytes_;
+}
+
+uint16_t TransportFeedback::GetBaseSequence() const {
+  return base_seq_;
+}
+
+int64_t TransportFeedback::GetBaseTimeUs() const {
+  return base_time_ * kBaseScaleFactor;
+}
+
+std::vector<TransportFeedback::StatusSymbol>
+TransportFeedback::GetStatusVector() const {
+  std::vector<TransportFeedback::StatusSymbol> symbols;
+  for (PacketStatusChunk* chunk : status_chunks_)
+    chunk->AppendSymbolsTo(&symbols);
+  int64_t status_count = last_seq_ - base_seq_ + 1;
+  // If packet ends with a vector chunk, it may contain extraneous "packet not
+  // received"-symbols at the end. Crop any such symbols.
+  symbols.erase(symbols.begin() + status_count, symbols.end());
+  return symbols;
+}
+
+std::vector<int16_t> TransportFeedback::GetReceiveDeltas() const {
+  return receive_deltas_;
+}
+
+std::vector<int64_t> TransportFeedback::GetReceiveDeltasUs() const {
+  if (receive_deltas_.empty())
+    return std::vector<int64_t>();
+
+  std::vector<int64_t> us_deltas;
+  for (int16_t delta : receive_deltas_)
+    us_deltas.push_back(static_cast<int64_t>(delta) * kDeltaScaleFactor);
+
+  return us_deltas;
+}
+
+// Serialize packet.
+bool TransportFeedback::Create(uint8_t* packet,
+                               size_t* position,
+                               size_t max_length,
+                               PacketReadyCallback* callback) const {
+  if (base_seq_ == -1)
+    return false;
+
+  while (*position + size_bytes_ > max_length) {
+    if (!OnBufferFull(packet, position, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPayloadType, HeaderLength(), packet,
+               position);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*position], packet_sender_ssrc_);
+  *position += 4;
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*position], media_source_ssrc_);
+  *position += 4;
+
+  RTC_DCHECK_LE(base_seq_, 0xFFFF);
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], base_seq_);
+  *position += 2;
+
+  int64_t status_count = last_seq_ - base_seq_ + 1;
+  RTC_DCHECK_LE(status_count, 0xFFFF);
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], status_count);
+  *position += 2;
+
+  ByteWriter<int32_t, 3>::WriteBigEndian(&packet[*position],
+                                         static_cast<int32_t>(base_time_));
+  *position += 3;
+
+  packet[(*position)++] = feedback_seq_;
+
+  // TODO(sprang): Get rid of this cast.
+  const_cast<TransportFeedback*>(this)->EmitRemaining();
+  for (PacketStatusChunk* chunk : status_chunks_) {
+    chunk->WriteTo(&packet[*position]);
+    *position += 2;
+  }
+
+  for (int16_t delta : receive_deltas_) {
+    if (delta >= 0 && delta <= 0xFF) {
+      packet[(*position)++] = delta;
+    } else {
+      ByteWriter<int16_t>::WriteBigEndian(&packet[*position], delta);
+      *position += 2;
+    }
+  }
+
+  while ((*position % 4) != 0)
+    packet[(*position)++] = 0;
+
+  return true;
+}
+
+//    Message format
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P|  FMT=15 |    PT=205     |           length              |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |                     SSRC of packet sender                     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |                      SSRC of media source                     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |      base sequence number     |      packet status count      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |                 reference time                | fb pkt. count |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |          packet chunk         |         packet chunk          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    .                                                               .
+//    .                                                               .
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |         packet chunk          |  recv delta   |  recv delta   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    .                                                               .
+//    .                                                               .
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |           recv delta          |  recv delta   | zero padding  |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+// De-serialize packet.
+rtc::scoped_ptr<TransportFeedback> TransportFeedback::ParseFrom(
+    const uint8_t* buffer,
+    size_t length) {
+  rtc::scoped_ptr<TransportFeedback> packet(new TransportFeedback());
+
+  if (length < kMinSizeBytes) {
+    LOG(LS_WARNING) << "Buffer too small (" << length
+                    << " bytes) to fit a "
+                       "FeedbackPacket. Minimum size = " << kMinSizeBytes;
+    return nullptr;
+  }
+
+  RTCPUtility::RtcpCommonHeader header;
+  if (!RtcpParseCommonHeader(buffer, length, &header))
+    return nullptr;
+
+  if (header.count_or_format != kFeedbackMessageType) {
+    LOG(LS_WARNING) << "Invalid RTCP header: FMT must be "
+                    << kFeedbackMessageType << " but was "
+                    << header.count_or_format;
+    return nullptr;
+  }
+
+  if (header.packet_type != kPayloadType) {
+    LOG(LS_WARNING) << "Invalid RTCP header: PT must be " << kPayloadType
+                    << " but was " << header.packet_type;
+    return nullptr;
+  }
+
+  packet->packet_sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  packet->media_source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  packet->base_seq_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[12]);
+  uint16_t num_packets = ByteReader<uint16_t>::ReadBigEndian(&buffer[14]);
+  packet->base_time_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[16]);
+  packet->feedback_seq_ = buffer[19];
+  size_t index = 20;
+  const size_t end_index = kHeaderLength + header.payload_size_bytes;
+
+  if (num_packets == 0) {
+    LOG(LS_WARNING) << "Empty feedback messages not allowed.";
+    return nullptr;
+  }
+  packet->last_seq_ = packet->base_seq_ + num_packets - 1;
+
+  size_t packets_read = 0;
+  while (packets_read < num_packets) {
+    if (index + 2 > end_index) {
+      LOG(LS_WARNING) << "Buffer overflow while parsing packet.";
+      return nullptr;
+    }
+
+    PacketStatusChunk* chunk =
+        ParseChunk(&buffer[index], num_packets - packets_read);
+    if (chunk == nullptr)
+      return nullptr;
+
+    index += 2;
+    packet->status_chunks_.push_back(chunk);
+    packets_read += chunk->NumSymbols();
+  }
+
+  std::vector<StatusSymbol> symbols = packet->GetStatusVector();
+
+  RTC_DCHECK_EQ(num_packets, symbols.size());
+
+  for (StatusSymbol symbol : symbols) {
+    switch (symbol) {
+      case StatusSymbol::kReceivedSmallDelta:
+        if (index + 1 > end_index) {
+          LOG(LS_WARNING) << "Buffer overflow while parsing packet.";
+          return nullptr;
+        }
+        packet->receive_deltas_.push_back(buffer[index]);
+        ++index;
+        break;
+      case StatusSymbol::kReceivedLargeDelta:
+        if (index + 2 > end_index) {
+          LOG(LS_WARNING) << "Buffer overflow while parsing packet.";
+          return nullptr;
+        }
+        packet->receive_deltas_.push_back(
+            ByteReader<int16_t>::ReadBigEndian(&buffer[index]));
+        index += 2;
+        break;
+      default:
+        continue;
+    }
+  }
+
+  RTC_DCHECK_GE(index, end_index - 3);
+  RTC_DCHECK_LE(index, end_index);
+
+  return packet;
+}
+
+PacketStatusChunk* TransportFeedback::ParseChunk(const uint8_t* buffer,
+                                                 size_t max_size) {
+  if (buffer[0] & 0x80) {
+    // First bit set => vector chunk.
+    std::deque<StatusSymbol> symbols;
+    if (buffer[0] & 0x40) {
+      // Second bit set => two bits per symbol vector.
+      return TwoBitVectorChunk::ParseFrom(buffer);
+    }
+
+    // Second bit not set => one bit per symbol vector.
+    return OneBitVectorChunk::ParseFrom(buffer);
+  }
+
+  // First bit not set => RLE chunk.
+  RunLengthChunk* rle_chunk = RunLengthChunk::ParseFrom(buffer);
+  if (rle_chunk->NumSymbols() > max_size) {
+    LOG(LS_WARNING) << "Header/body mismatch. "
+                       "RLE block of size " << rle_chunk->NumSymbols()
+                    << " but only " << max_size << " left to read.";
+    delete rle_chunk;
+    return nullptr;
+  }
+  return rle_chunk;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
new file mode 100644
index 0000000000..ad6fd166f2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
+
+#include <deque>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class PacketStatusChunk;
+
+class TransportFeedback : public RtcpPacket {
+ public:
+  TransportFeedback();
+  virtual ~TransportFeedback();
+
+  void WithPacketSenderSsrc(uint32_t ssrc);
+  void WithMediaSourceSsrc(uint32_t ssrc);
+  void WithBase(uint16_t base_sequence,     // Seq# of first packet in this msg.
+                int64_t ref_timestamp_us);  // Reference timestamp for this msg.
+  void WithFeedbackSequenceNumber(uint8_t feedback_sequence);
+  // NOTE: This method requires increasing sequence numbers (excepting wraps).
+  bool WithReceivedPacket(uint16_t sequence_number, int64_t timestamp_us);
+
+  enum class StatusSymbol {
+    kNotReceived,
+    kReceivedSmallDelta,
+    kReceivedLargeDelta,
+  };
+
+  uint16_t GetBaseSequence() const;
+  std::vector<TransportFeedback::StatusSymbol> GetStatusVector() const;
+  std::vector<int16_t> GetReceiveDeltas() const;
+
+  // Get the reference time in microseconds, including any precision loss.
+  int64_t GetBaseTimeUs() const;
+  // Convenience method for getting all deltas as microseconds. The first delta
+  // is relative the base time.
+  std::vector<int64_t> GetReceiveDeltasUs() const;
+
+  uint32_t GetPacketSenderSsrc() const;
+  uint32_t GetMediaSourceSsrc() const;
+  static const int kDeltaScaleFactor = 250;  // Convert to multiples of 0.25ms.
+  static const uint8_t kFeedbackMessageType = 15;  // TODO(sprang): IANA reg?
+  static const uint8_t kPayloadType = 205;         // RTPFB, see RFC4585.
+
+  static rtc::scoped_ptr<TransportFeedback> ParseFrom(const uint8_t* buffer,
+                                                      size_t length);
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* position,
+              size_t max_length,
+              PacketReadyCallback* callback) const override;
+
+  size_t BlockLength() const override;
+
+ private:
+  static PacketStatusChunk* ParseChunk(const uint8_t* buffer, size_t max_size);
+
+  int64_t Unwrap(uint16_t sequence_number);
+  bool AddSymbol(StatusSymbol symbol, int64_t seq);
+  bool Encode(StatusSymbol symbol);
+  bool HandleRleCandidate(StatusSymbol symbol,
+                          int current_capacity,
+                          int delta_size);
+  void EmitRemaining();
+  void EmitVectorChunk();
+  void EmitRunLengthChunk();
+
+  uint32_t packet_sender_ssrc_;
+  uint32_t media_source_ssrc_;
+  int32_t base_seq_;
+  int64_t base_time_;
+  uint8_t feedback_seq_;
+  std::vector<PacketStatusChunk*> status_chunks_;
+  std::vector<int16_t> receive_deltas_;
+
+  int64_t last_seq_;
+  int64_t last_timestamp_;
+  std::deque<StatusSymbol> symbol_vec_;
+  uint16_t first_symbol_cardinality_;
+  bool vec_needs_two_bit_symbols_;
+  uint32_t size_bytes_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(TransportFeedback);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
new file mode 100644
index 0000000000..ceb911d308
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
@@ -0,0 +1,482 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+#include <limits>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::rtcp::TransportFeedback;
+
+namespace webrtc {
+namespace {
+
+static const int kHeaderSize = 20;
+static const int kStatusChunkSize = 2;
+static const int kSmallDeltaSize = 1;
+static const int kLargeDeltaSize = 2;
+
+static const int64_t kDeltaLimit = 0xFF * TransportFeedback::kDeltaScaleFactor;
+
+class FeedbackTester {
+ public:
+  FeedbackTester()
+      : expected_size_(kAnySize),
+        default_delta_(TransportFeedback::kDeltaScaleFactor * 4) {}
+
+  void WithExpectedSize(size_t expected_size) {
+    expected_size_ = expected_size;
+  }
+
+  void WithDefaultDelta(int64_t delta) { default_delta_ = delta; }
+
+  void WithInput(const uint16_t received_seq[],
+                 const int64_t received_ts[],
+                 uint16_t length) {
+    rtc::scoped_ptr<int64_t[]> temp_deltas;
+    if (received_ts == nullptr) {
+      temp_deltas.reset(new int64_t[length]);
+      GenerateDeltas(received_seq, length, temp_deltas.get());
+      received_ts = temp_deltas.get();
+    }
+
+    expected_seq_.clear();
+    expected_deltas_.clear();
+    feedback_.reset(new TransportFeedback());
+
+    feedback_->WithBase(received_seq[0], received_ts[0]);
+    int64_t last_time = feedback_->GetBaseTimeUs();
+    for (int i = 0; i < length; ++i) {
+      int64_t time = received_ts[i];
+      EXPECT_TRUE(feedback_->WithReceivedPacket(received_seq[i], time));
+
+      if (last_time != -1) {
+        int64_t delta = time - last_time;
+        expected_deltas_.push_back(delta);
+      }
+      last_time = time;
+    }
+    expected_seq_.insert(expected_seq_.begin(), &received_seq[0],
+                         &received_seq[length]);
+  }
+
+  void VerifyPacket() {
+    serialized_ = feedback_->Build();
+    VerifyInternal();
+    feedback_ = TransportFeedback::ParseFrom(serialized_->Buffer(),
+                                             serialized_->Length());
+    ASSERT_NE(nullptr, feedback_.get());
+    VerifyInternal();
+  }
+
+  static const size_t kAnySize = static_cast<size_t>(0) - 1;
+
+ private:
+  void VerifyInternal() {
+    if (expected_size_ != kAnySize) {
+      // Round up to whole 32-bit words.
+      size_t expected_size_words = (expected_size_ + 3) / 4;
+      size_t expected_size_bytes = expected_size_words * 4;
+      EXPECT_EQ(expected_size_bytes, serialized_->Length());
+    }
+
+    std::vector<TransportFeedback::StatusSymbol> symbols =
+        feedback_->GetStatusVector();
+    uint16_t seq = feedback_->GetBaseSequence();
+    auto seq_it = expected_seq_.begin();
+    for (TransportFeedback::StatusSymbol symbol : symbols) {
+      bool received =
+          (symbol == TransportFeedback::StatusSymbol::kReceivedSmallDelta ||
+           symbol == TransportFeedback::StatusSymbol::kReceivedLargeDelta);
+      if (seq_it != expected_seq_.end()) {
+        if (seq == *seq_it) {
+          ASSERT_NE(expected_seq_.end(), seq_it);
+          ASSERT_TRUE(received) << "Expected received packet @ " << seq;
+          ++seq_it;
+        } else {
+          ASSERT_FALSE(received) << "Did not expect received packet @ " << seq;
+        }
+      }
+      ++seq;
+    }
+    ASSERT_EQ(expected_seq_.end(), seq_it);
+
+    std::vector<int64_t> deltas = feedback_->GetReceiveDeltasUs();
+    ASSERT_EQ(expected_deltas_.size(), deltas.size());
+    for (size_t i = 0; i < expected_deltas_.size(); ++i)
+      EXPECT_EQ(expected_deltas_[i], deltas[i]) << "Delta mismatch @ " << i;
+  }
+
+  void GenerateDeltas(const uint16_t seq[],
+                      const size_t length,
+                      int64_t* deltas) {
+    uint16_t last_seq = seq[0];
+    int64_t offset = 0;
+
+    for (size_t i = 0; i < length; ++i) {
+      if (seq[i] < last_seq)
+        offset += 0x10000 * default_delta_;
+      last_seq = seq[i];
+
+      deltas[i] = offset + (last_seq * default_delta_);
+    }
+  }
+
+  std::vector<uint16_t> expected_seq_;
+  std::vector<int64_t> expected_deltas_;
+  size_t expected_size_;
+  int64_t default_delta_;
+  rtc::scoped_ptr<TransportFeedback> feedback_;
+  rtc::scoped_ptr<rtcp::RawPacket> serialized_;
+};
+
+TEST(RtcpPacketTest, TransportFeedback_OneBitVector) {
+  const uint16_t kReceived[] = {1, 2, 7, 8, 9, 10, 13};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, nullptr, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_FullOneBitVector) {
+  const uint16_t kReceived[] = {1, 2, 7, 8, 9, 10, 13, 14};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, nullptr, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_OneBitVector_WrapReceived) {
+  const uint16_t kMax = 0xFFFF;
+  const uint16_t kReceived[] = {kMax - 2, kMax - 1, kMax, 0, 1, 2};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, nullptr, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_OneBitVector_WrapMissing) {
+  const uint16_t kMax = 0xFFFF;
+  const uint16_t kReceived[] = {kMax - 2, kMax - 1, 1, 2};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, nullptr, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_TwoBitVector) {
+  const uint16_t kReceived[] = {1, 2, 6, 7};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kLargeDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithDefaultDelta(kDeltaLimit + TransportFeedback::kDeltaScaleFactor);
+  test.WithInput(kReceived, nullptr, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_TwoBitVectorFull) {
+  const uint16_t kReceived[] = {1, 2, 6, 7, 8};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (2 * kStatusChunkSize) + (kLength * kLargeDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithDefaultDelta(kDeltaLimit + TransportFeedback::kDeltaScaleFactor);
+  test.WithInput(kReceived, nullptr, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_LargeAndNegativeDeltas) {
+  const uint16_t kReceived[] = {1, 2, 6, 7, 8};
+  const int64_t kReceiveTimes[] = {
+      2000,
+      1000,
+      4000,
+      3000,
+      3000 + TransportFeedback::kDeltaScaleFactor * (1 << 8)};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (3 * kLargeDeltaSize) + kSmallDeltaSize;
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_MaxRle) {
+  // Expected chunks created:
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+  // * RLE chunk of max length for dropped symbol
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+
+  const size_t kPacketCount = (1 << 13) - 1 + 14;
+  const uint16_t kReceived[] = {0, kPacketCount};
+  const int64_t kReceiveTimes[] = {1000, 2000};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (3 * kStatusChunkSize) + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_MinRle) {
+  // Expected chunks created:
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+  // * RLE chunk of length 15 for dropped symbol
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+
+  const uint16_t kReceived[] = {0, (14 * 2) + 1};
+  const int64_t kReceiveTimes[] = {1000, 2000};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (3 * kStatusChunkSize) + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_OneToTwoBitVector) {
+  const size_t kTwoBitVectorCapacity = 7;
+  const uint16_t kReceived[] = {0, kTwoBitVectorCapacity - 1};
+  const int64_t kReceiveTimes[] = {
+      0, kDeltaLimit + TransportFeedback::kDeltaScaleFactor};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + kSmallDeltaSize + kLargeDeltaSize;
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_OneToTwoBitVectorSimpleSplit) {
+  const size_t kTwoBitVectorCapacity = 7;
+  const uint16_t kReceived[] = {0, kTwoBitVectorCapacity};
+  const int64_t kReceiveTimes[] = {
+      0, kDeltaLimit + TransportFeedback::kDeltaScaleFactor};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (kStatusChunkSize * 2) + kSmallDeltaSize + kLargeDeltaSize;
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes, kLength);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_OneToTwoBitVectorSplit) {
+  // With received small delta = S, received large delta = L, use input
+  // SSSSSSSSLSSSSSSSSSSSS. This will cause a 1:2 split at the L.
+  // After split there will be two symbols in symbol_vec: SL.
+
+  const int64_t kLargeDelta = TransportFeedback::kDeltaScaleFactor * (1 << 8);
+  const size_t kNumPackets = (3 * 7) + 1;
+  const size_t kExpectedSizeBytes = kHeaderSize + (kStatusChunkSize * 3) +
+                                    (kSmallDeltaSize * (kNumPackets - 1)) +
+                                    (kLargeDeltaSize * 1);
+
+  uint16_t kReceived[kNumPackets];
+  for (size_t i = 0; i < kNumPackets; ++i)
+    kReceived[i] = i;
+
+  int64_t kReceiveTimes[kNumPackets];
+  kReceiveTimes[0] = 1000;
+  for (size_t i = 1; i < kNumPackets; ++i) {
+    int delta = (i == 8) ? kLargeDelta : 1000;
+    kReceiveTimes[i] = kReceiveTimes[i - 1] + delta;
+  }
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes, kNumPackets);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedback_Aliasing) {
+  TransportFeedback feedback;
+  feedback.WithBase(0, 0);
+
+  const int kSamples = 100;
+  const int64_t kTooSmallDelta = TransportFeedback::kDeltaScaleFactor / 3;
+
+  for (int i = 0; i < kSamples; ++i)
+    feedback.WithReceivedPacket(i, i * kTooSmallDelta);
+
+  feedback.Build();
+  std::vector<int64_t> deltas = feedback.GetReceiveDeltasUs();
+
+  int64_t accumulated_delta = 0;
+  int num_samples = 0;
+  for (int64_t delta : deltas) {
+    accumulated_delta += delta;
+    int64_t expected_time = num_samples * kTooSmallDelta;
+    ++num_samples;
+
+    EXPECT_NEAR(expected_time, accumulated_delta,
+                TransportFeedback::kDeltaScaleFactor / 2);
+  }
+}
+
+TEST(RtcpPacketTest, TransportFeedback_Limits) {
+  // Sequence number wrap above 0x8000.
+  rtc::scoped_ptr<TransportFeedback> packet(new TransportFeedback());
+  packet->WithBase(0, 0);
+  EXPECT_TRUE(packet->WithReceivedPacket(0x8000, 1000));
+
+  packet.reset(new TransportFeedback());
+  packet->WithBase(0, 0);
+  EXPECT_FALSE(packet->WithReceivedPacket(0x8000 + 1, 1000));
+
+  // Packet status count max 0xFFFF.
+  packet.reset(new TransportFeedback());
+  packet->WithBase(0, 0);
+  EXPECT_TRUE(packet->WithReceivedPacket(0x8000, 1000));
+  EXPECT_TRUE(packet->WithReceivedPacket(0xFFFF, 2000));
+  EXPECT_FALSE(packet->WithReceivedPacket(0, 3000));
+
+  // Too large delta.
+  packet.reset(new TransportFeedback());
+  packet->WithBase(0, 0);
+  int64_t kMaxPositiveTimeDelta = std::numeric_limits<int16_t>::max() *
+                                  TransportFeedback::kDeltaScaleFactor;
+  EXPECT_FALSE(packet->WithReceivedPacket(
+      1, kMaxPositiveTimeDelta + TransportFeedback::kDeltaScaleFactor));
+  EXPECT_TRUE(packet->WithReceivedPacket(1, kMaxPositiveTimeDelta));
+
+  // Too large negative delta.
+  packet.reset(new TransportFeedback());
+  packet->WithBase(0, 0);
+  int64_t kMaxNegativeTimeDelta = std::numeric_limits<int16_t>::min() *
+                                  TransportFeedback::kDeltaScaleFactor;
+  EXPECT_FALSE(packet->WithReceivedPacket(
+      1, kMaxNegativeTimeDelta - TransportFeedback::kDeltaScaleFactor));
+  EXPECT_TRUE(packet->WithReceivedPacket(1, kMaxNegativeTimeDelta));
+
+  // Base time at maximum value.
+  int64_t kMaxBaseTime =
+      static_cast<int64_t>(TransportFeedback::kDeltaScaleFactor) * (1L << 8) *
+      ((1L << 23) - 1);
+  packet.reset(new TransportFeedback());
+  packet->WithBase(0, kMaxBaseTime);
+  packet->WithReceivedPacket(0, kMaxBaseTime);
+  // Serialize and de-serialize (verify 24bit parsing).
+  rtc::scoped_ptr<rtcp::RawPacket> raw_packet = packet->Build();
+  packet =
+      TransportFeedback::ParseFrom(raw_packet->Buffer(), raw_packet->Length());
+  EXPECT_EQ(kMaxBaseTime, packet->GetBaseTimeUs());
+
+  // Base time above maximum value.
+  int64_t kTooLargeBaseTime =
+      kMaxBaseTime + (TransportFeedback::kDeltaScaleFactor * (1L << 8));
+  packet.reset(new TransportFeedback());
+  packet->WithBase(0, kTooLargeBaseTime);
+  packet->WithReceivedPacket(0, kTooLargeBaseTime);
+  raw_packet = packet->Build();
+  packet =
+      TransportFeedback::ParseFrom(raw_packet->Buffer(), raw_packet->Length());
+  EXPECT_NE(kTooLargeBaseTime, packet->GetBaseTimeUs());
+
+  // TODO(sprang): Once we support max length lower than RTCP length limit,
+  // add back test for max size in bytes.
+}
+
+TEST(RtcpPacketTest, TransportFeedback_Padding) {
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + kSmallDeltaSize;
+  const size_t kExpectedSizeWords = (kExpectedSizeBytes + 3) / 4;
+
+  TransportFeedback feedback;
+  feedback.WithBase(0, 0);
+  EXPECT_TRUE(feedback.WithReceivedPacket(0, 0));
+
+  rtc::scoped_ptr<rtcp::RawPacket> packet(feedback.Build());
+  EXPECT_EQ(kExpectedSizeWords * 4, packet->Length());
+  ASSERT_GT(kExpectedSizeWords * 4, kExpectedSizeBytes);
+  for (size_t i = kExpectedSizeBytes; i < kExpectedSizeWords * 4; ++i)
+    EXPECT_EQ(0u, packet->Buffer()[i]);
+
+  // Modify packet by adding 4 bytes of padding at the end. Not currently used
+  // when we're sending, but need to be able to handle it when receiving.
+
+  const int kPaddingBytes = 4;
+  const size_t kExpectedSizeWithPadding =
+      (kExpectedSizeWords * 4) + kPaddingBytes;
+  uint8_t mod_buffer[kExpectedSizeWithPadding];
+  memcpy(mod_buffer, packet->Buffer(), kExpectedSizeWords * 4);
+  memset(&mod_buffer[kExpectedSizeWords * 4], 0, kPaddingBytes - 1);
+  mod_buffer[kExpectedSizeWithPadding - 1] = kPaddingBytes;
+  const uint8_t padding_flag = 1 << 5;
+  mod_buffer[0] |= padding_flag;
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &mod_buffer[2], ByteReader<uint16_t>::ReadBigEndian(&mod_buffer[2]) +
+                          ((kPaddingBytes + 3) / 4));
+
+  rtc::scoped_ptr<TransportFeedback> parsed_packet(
+      TransportFeedback::ParseFrom(mod_buffer, kExpectedSizeWithPadding));
+  ASSERT_TRUE(parsed_packet.get() != nullptr);
+  EXPECT_EQ(kExpectedSizeWords * 4, packet->Length());  // Padding not included.
+}
+
+TEST(RtcpPacketTest, TransportFeedback_CorrectlySplitsVectorChunks) {
+  const int kOneBitVectorCapacity = 14;
+  const int64_t kLargeTimeDelta =
+      TransportFeedback::kDeltaScaleFactor * (1 << 8);
+
+  // Test that a number of small deltas followed by a large delta results in a
+  // correct split into multiple chunks, as needed.
+
+  for (int deltas = 0; deltas <= kOneBitVectorCapacity + 1; ++deltas) {
+    TransportFeedback feedback;
+    feedback.WithBase(0, 0);
+    for (int i = 0; i < deltas; ++i)
+      feedback.WithReceivedPacket(i, i * 1000);
+    feedback.WithReceivedPacket(deltas, deltas * 1000 + kLargeTimeDelta);
+
+    rtc::scoped_ptr<rtcp::RawPacket> serialized_packet = feedback.Build();
+    EXPECT_TRUE(serialized_packet.get() != nullptr);
+    rtc::scoped_ptr<TransportFeedback> deserialized_packet =
+        TransportFeedback::ParseFrom(serialized_packet->Buffer(),
+                                     serialized_packet->Length());
+    EXPECT_TRUE(deserialized_packet.get() != nullptr);
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.cc
new file mode 100644
index 0000000000..a79d48e1ca
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+// VoIP Metrics Report Block (RFC 3611).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |     BT=7      |   reserved    |       block length = 8        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                        SSRC of source                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |   loss rate   | discard rate  | burst density |  gap density  |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |       burst duration          |         gap duration          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |     round trip delay          |       end system delay        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 | signal level  |  noise level  |     RERL      |     Gmin      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 24 |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 28 |   RX config   |   reserved    |          JB nominal           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 32 |          JB maximum           |          JB abs max           |
+// 36 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+VoipMetric::VoipMetric() : ssrc_(0) {
+  memset(&voip_metric_, 0, sizeof(voip_metric_));
+}
+
+void VoipMetric::Parse(const uint8_t* buffer) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // reserved = buffer[1];
+  RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  voip_metric_.lossRate = buffer[8];
+  voip_metric_.discardRate = buffer[9];
+  voip_metric_.burstDensity = buffer[10];
+  voip_metric_.gapDensity = buffer[11];
+  voip_metric_.burstDuration = ByteReader<uint16_t>::ReadBigEndian(&buffer[12]);
+  voip_metric_.gapDuration = ByteReader<uint16_t>::ReadBigEndian(&buffer[14]);
+  voip_metric_.roundTripDelay =
+      ByteReader<uint16_t>::ReadBigEndian(&buffer[16]);
+  voip_metric_.endSystemDelay =
+      ByteReader<uint16_t>::ReadBigEndian(&buffer[18]);
+  voip_metric_.signalLevel = buffer[20];
+  voip_metric_.noiseLevel = buffer[21];
+  voip_metric_.RERL = buffer[22];
+  voip_metric_.Gmin = buffer[23];
+  voip_metric_.Rfactor = buffer[24];
+  voip_metric_.extRfactor = buffer[25];
+  voip_metric_.MOSLQ = buffer[26];
+  voip_metric_.MOSCQ = buffer[27];
+  voip_metric_.RXconfig = buffer[28];
+  // reserved = buffer[29];
+  voip_metric_.JBnominal = ByteReader<uint16_t>::ReadBigEndian(&buffer[30]);
+  voip_metric_.JBmax = ByteReader<uint16_t>::ReadBigEndian(&buffer[32]);
+  voip_metric_.JBabsMax = ByteReader<uint16_t>::ReadBigEndian(&buffer[34]);
+}
+
+void VoipMetric::Create(uint8_t* buffer) const {
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ssrc_);
+  buffer[8] = voip_metric_.lossRate;
+  buffer[9] = voip_metric_.discardRate;
+  buffer[10] = voip_metric_.burstDensity;
+  buffer[11] = voip_metric_.gapDensity;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[12], voip_metric_.burstDuration);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[14], voip_metric_.gapDuration);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[16],
+                                       voip_metric_.roundTripDelay);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[18],
+                                       voip_metric_.endSystemDelay);
+  buffer[20] = voip_metric_.signalLevel;
+  buffer[21] = voip_metric_.noiseLevel;
+  buffer[22] = voip_metric_.RERL;
+  buffer[23] = voip_metric_.Gmin;
+  buffer[24] = voip_metric_.Rfactor;
+  buffer[25] = voip_metric_.extRfactor;
+  buffer[26] = voip_metric_.MOSLQ;
+  buffer[27] = voip_metric_.MOSCQ;
+  buffer[28] = voip_metric_.RXconfig;
+  buffer[29] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[30], voip_metric_.JBnominal);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[32], voip_metric_.JBmax);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[34], voip_metric_.JBabsMax);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
new file mode 100644
index 0000000000..9e3e41995a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class VoipMetric {
+ public:
+  static const uint8_t kBlockType = 7;
+  static const uint16_t kBlockLength = 8;
+  static const size_t kLength = 4 * (kBlockLength + 1);  // 36
+  VoipMetric();
+  VoipMetric(const VoipMetric&) = default;
+  ~VoipMetric() {}
+
+  VoipMetric& operator=(const VoipMetric&) = default;
+
+  void Parse(const uint8_t* buffer);
+
+  // Fills buffer with the VoipMetric.
+  // Consumes VoipMetric::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void To(uint32_t ssrc) { ssrc_ = ssrc; }
+  void WithVoipMetric(const RTCPVoIPMetric& voip_metric) {
+    voip_metric_ = voip_metric;
+  }
+
+  uint32_t ssrc() const { return ssrc_; }
+  const RTCPVoIPMetric& voip_metric() const { return voip_metric_; }
+
+ private:
+  uint32_t ssrc_;
+  RTCPVoIPMetric voip_metric_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
new file mode 100644
index 0000000000..44c82d67a9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kBlock[] = {0x07, 0x00, 0x00, 0x08, 0x23, 0x45, 0x67, 0x89,
+                          0x01, 0x02, 0x03, 0x04, 0x11, 0x12, 0x22, 0x23,
+                          0x33, 0x34, 0x44, 0x45, 0x05, 0x06, 0x07, 0x08,
+                          0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x00, 0x55, 0x56,
+                          0x66, 0x67, 0x77, 0x78};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+static_assert(
+    kBlockSizeBytes == VoipMetric::kLength,
+    "Size of manually created Voip Metric block should match class constant");
+
+TEST(RtcpPacketVoipMetricTest, Create) {
+  uint8_t buffer[VoipMetric::kLength];
+  RTCPVoIPMetric metric;
+  metric.lossRate = 1;
+  metric.discardRate = 2;
+  metric.burstDensity = 3;
+  metric.gapDensity = 4;
+  metric.burstDuration = 0x1112;
+  metric.gapDuration = 0x2223;
+  metric.roundTripDelay = 0x3334;
+  metric.endSystemDelay = 0x4445;
+  metric.signalLevel = 5;
+  metric.noiseLevel = 6;
+  metric.RERL = 7;
+  metric.Gmin = 8;
+  metric.Rfactor = 9;
+  metric.extRfactor = 10;
+  metric.MOSLQ = 11;
+  metric.MOSCQ = 12;
+  metric.RXconfig = 13;
+  metric.JBnominal = 0x5556;
+  metric.JBmax = 0x6667;
+  metric.JBabsMax = 0x7778;
+  VoipMetric metric_block;
+  metric_block.To(kRemoteSsrc);
+  metric_block.WithVoipMetric(metric);
+
+  metric_block.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketVoipMetricTest, Parse) {
+  VoipMetric read_metric;
+  read_metric.Parse(kBlock);
+
+  // Run checks on const object to ensure all accessors have const modifier.
+  const VoipMetric& parsed = read_metric;
+
+  EXPECT_EQ(kRemoteSsrc, parsed.ssrc());
+  EXPECT_EQ(1, parsed.voip_metric().lossRate);
+  EXPECT_EQ(2, parsed.voip_metric().discardRate);
+  EXPECT_EQ(3, parsed.voip_metric().burstDensity);
+  EXPECT_EQ(4, parsed.voip_metric().gapDensity);
+  EXPECT_EQ(0x1112, parsed.voip_metric().burstDuration);
+  EXPECT_EQ(0x2223, parsed.voip_metric().gapDuration);
+  EXPECT_EQ(0x3334, parsed.voip_metric().roundTripDelay);
+  EXPECT_EQ(0x4445, parsed.voip_metric().endSystemDelay);
+  EXPECT_EQ(5, parsed.voip_metric().signalLevel);
+  EXPECT_EQ(6, parsed.voip_metric().noiseLevel);
+  EXPECT_EQ(7, parsed.voip_metric().RERL);
+  EXPECT_EQ(8, parsed.voip_metric().Gmin);
+  EXPECT_EQ(9, parsed.voip_metric().Rfactor);
+  EXPECT_EQ(10, parsed.voip_metric().extRfactor);
+  EXPECT_EQ(11, parsed.voip_metric().MOSLQ);
+  EXPECT_EQ(12, parsed.voip_metric().MOSCQ);
+  EXPECT_EQ(13, parsed.voip_metric().RXconfig);
+  EXPECT_EQ(0x5556, parsed.voip_metric().JBnominal);
+  EXPECT_EQ(0x6667, parsed.voip_metric().JBmax);
+  EXPECT_EQ(0x7778, parsed.voip_metric().JBabsMax);
+}
+
+}  // namespace
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
index c0ba3075d1..22f61f5cab 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
@@ -10,31 +10,29 @@
  * This file includes unit tests for the RtcpPacket.
  */
 
+#include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 
+using ::testing::ElementsAre;
+
 using webrtc::rtcp::App;
 using webrtc::rtcp::Bye;
 using webrtc::rtcp::Dlrr;
-using webrtc::rtcp::Empty;
 using webrtc::rtcp::Fir;
-using webrtc::rtcp::Ij;
-using webrtc::rtcp::Nack;
-using webrtc::rtcp::Pli;
-using webrtc::rtcp::Sdes;
-using webrtc::rtcp::SenderReport;
-using webrtc::rtcp::Sli;
 using webrtc::rtcp::RawPacket;
 using webrtc::rtcp::ReceiverReport;
 using webrtc::rtcp::Remb;
 using webrtc::rtcp::ReportBlock;
 using webrtc::rtcp::Rpsi;
 using webrtc::rtcp::Rrtr;
+using webrtc::rtcp::Sdes;
 using webrtc::rtcp::SenderReport;
-using webrtc::rtcp::Tmmbn;
-using webrtc::rtcp::Tmmbr;
 using webrtc::rtcp::VoipMetric;
 using webrtc::rtcp::Xr;
 using webrtc::test::RtcpPacketParser;
@@ -44,68 +42,6 @@ namespace webrtc {
 const uint32_t kSenderSsrc = 0x12345678;
 const uint32_t kRemoteSsrc = 0x23456789;
 
-TEST(RtcpPacketTest, Rr) {
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-
-  RawPacket packet = rr.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(0, parser.report_block()->num_packets());
-}
-
-TEST(RtcpPacketTest, RrWithOneReportBlock) {
-  ReportBlock rb;
-  rb.To(kRemoteSsrc);
-  rb.WithFractionLost(55);
-  rb.WithCumulativeLost(0x111111);
-  rb.WithExtHighestSeqNum(0x22222222);
-  rb.WithJitter(0x33333333);
-  rb.WithLastSr(0x44444444);
-  rb.WithDelayLastSr(0x55555555);
-
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-
-  RawPacket packet = rr.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(kRemoteSsrc, parser.report_block()->Ssrc());
-  EXPECT_EQ(55U, parser.report_block()->FractionLost());
-  EXPECT_EQ(0x111111U, parser.report_block()->CumPacketLost());
-  EXPECT_EQ(0x22222222U, parser.report_block()->ExtHighestSeqNum());
-  EXPECT_EQ(0x33333333U, parser.report_block()->Jitter());
-  EXPECT_EQ(0x44444444U, parser.report_block()->LastSr());
-  EXPECT_EQ(0x55555555U, parser.report_block()->DelayLastSr());
-}
-
-TEST(RtcpPacketTest, RrWithTwoReportBlocks) {
-  ReportBlock rb1;
-  rb1.To(kRemoteSsrc);
-  ReportBlock rb2;
-  rb2.To(kRemoteSsrc + 1);
-
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb1);
-  rr.WithReportBlock(&rb2);
-
-  RawPacket packet = rr.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(2, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc));
-  EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc + 1));
-}
-
 TEST(RtcpPacketTest, Sr) {
   SenderReport sr;
   sr.From(kSenderSsrc);
@@ -115,9 +51,9 @@ TEST(RtcpPacketTest, Sr) {
   sr.WithPacketCount(0x44444444);
   sr.WithOctetCount(0x55555555);
 
-  RawPacket packet = sr.Build();
+  rtc::scoped_ptr<RawPacket> packet(sr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
 
   EXPECT_EQ(1, parser.sender_report()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.sender_report()->Ssrc());
@@ -135,11 +71,11 @@ TEST(RtcpPacketTest, SrWithOneReportBlock) {
 
   SenderReport sr;
   sr.From(kSenderSsrc);
-  sr.WithReportBlock(&rb);
+  EXPECT_TRUE(sr.WithReportBlock(rb));
 
-  RawPacket packet = sr.Build();
+  rtc::scoped_ptr<RawPacket> packet(sr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.sender_report()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.sender_report()->Ssrc());
   EXPECT_EQ(1, parser.report_block()->num_packets());
@@ -154,12 +90,12 @@ TEST(RtcpPacketTest, SrWithTwoReportBlocks) {
 
   SenderReport sr;
   sr.From(kSenderSsrc);
-  sr.WithReportBlock(&rb1);
-  sr.WithReportBlock(&rb2);
+  EXPECT_TRUE(sr.WithReportBlock(rb1));
+  EXPECT_TRUE(sr.WithReportBlock(rb2));
 
-  RawPacket packet = sr.Build();
+  rtc::scoped_ptr<RawPacket> packet(sr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.sender_report()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.sender_report()->Ssrc());
   EXPECT_EQ(2, parser.report_block()->num_packets());
@@ -167,39 +103,17 @@ TEST(RtcpPacketTest, SrWithTwoReportBlocks) {
   EXPECT_EQ(1, parser.report_blocks_per_ssrc(kRemoteSsrc + 1));
 }
 
-TEST(RtcpPacketTest, IjNoItem) {
-  Ij ij;
-
-  RawPacket packet = ij.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.ij()->num_packets());
-  EXPECT_EQ(0, parser.ij_item()->num_packets());
-}
-
-TEST(RtcpPacketTest, IjOneItem) {
-  Ij ij;
-  ij.WithJitterItem(0x11111111);
-
-  RawPacket packet = ij.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.ij()->num_packets());
-  EXPECT_EQ(1, parser.ij_item()->num_packets());
-  EXPECT_EQ(0x11111111U, parser.ij_item()->Jitter());
-}
-
-TEST(RtcpPacketTest, IjTwoItems) {
-  Ij ij;
-  ij.WithJitterItem(0x11111111);
-  ij.WithJitterItem(0x22222222);
-
-  RawPacket packet = ij.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.ij()->num_packets());
-  EXPECT_EQ(2, parser.ij_item()->num_packets());
-  EXPECT_EQ(0x22222222U, parser.ij_item()->Jitter());
+TEST(RtcpPacketTest, SrWithTooManyReportBlocks) {
+  SenderReport sr;
+  sr.From(kSenderSsrc);
+  const int kMaxReportBlocks = (1 << 5) - 1;
+  ReportBlock rb;
+  for (int i = 0; i < kMaxReportBlocks; ++i) {
+    rb.To(kRemoteSsrc + i);
+    EXPECT_TRUE(sr.WithReportBlock(rb));
+  }
+  rb.To(kRemoteSsrc + kMaxReportBlocks);
+  EXPECT_FALSE(sr.WithReportBlock(rb));
 }
 
 TEST(RtcpPacketTest, AppWithNoData) {
@@ -211,9 +125,9 @@ TEST(RtcpPacketTest, AppWithNoData) {
   name += 'e';
   app.WithName(name);
 
-  RawPacket packet = app.Build();
+  rtc::scoped_ptr<RawPacket> packet(app.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.app()->num_packets());
   EXPECT_EQ(30U, parser.app()->SubType());
   EXPECT_EQ(name, parser.app()->Name());
@@ -233,9 +147,9 @@ TEST(RtcpPacketTest, App) {
   const size_t kDataLength = sizeof(kData) / sizeof(kData[0]);
   app.WithData((const uint8_t*)kData, kDataLength);
 
-  RawPacket packet = app.Build();
+  rtc::scoped_ptr<RawPacket> packet(app.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.app()->num_packets());
   EXPECT_EQ(30U, parser.app()->SubType());
   EXPECT_EQ(name, parser.app()->Name());
@@ -247,11 +161,11 @@ TEST(RtcpPacketTest, App) {
 
 TEST(RtcpPacketTest, SdesWithOneChunk) {
   Sdes sdes;
-  sdes.WithCName(kSenderSsrc, "alice@host");
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc, "alice@host"));
 
-  RawPacket packet = sdes.Build();
+  rtc::scoped_ptr<RawPacket> packet(sdes.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.sdes()->num_packets());
   EXPECT_EQ(1, parser.sdes_chunk()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.sdes_chunk()->Ssrc());
@@ -260,113 +174,47 @@ TEST(RtcpPacketTest, SdesWithOneChunk) {
 
 TEST(RtcpPacketTest, SdesWithMultipleChunks) {
   Sdes sdes;
-  sdes.WithCName(kSenderSsrc, "a");
-  sdes.WithCName(kSenderSsrc + 1, "ab");
-  sdes.WithCName(kSenderSsrc + 2, "abc");
-  sdes.WithCName(kSenderSsrc + 3, "abcd");
-  sdes.WithCName(kSenderSsrc + 4, "abcde");
-  sdes.WithCName(kSenderSsrc + 5, "abcdef");
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc, "a"));
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc + 1, "ab"));
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc + 2, "abc"));
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc + 3, "abcd"));
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc + 4, "abcde"));
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc + 5, "abcdef"));
 
-  RawPacket packet = sdes.Build();
+  rtc::scoped_ptr<RawPacket> packet(sdes.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.sdes()->num_packets());
   EXPECT_EQ(6, parser.sdes_chunk()->num_packets());
   EXPECT_EQ(kSenderSsrc + 5, parser.sdes_chunk()->Ssrc());
   EXPECT_EQ("abcdef", parser.sdes_chunk()->Cname());
 }
 
+TEST(RtcpPacketTest, SdesWithTooManyChunks) {
+  Sdes sdes;
+  const int kMaxChunks = (1 << 5) - 1;
+  for (int i = 0; i < kMaxChunks; ++i) {
+    uint32_t ssrc = kSenderSsrc + i;
+    std::ostringstream oss;
+    oss << "cname" << i;
+    EXPECT_TRUE(sdes.WithCName(ssrc, oss.str()));
+  }
+  EXPECT_FALSE(sdes.WithCName(kSenderSsrc + kMaxChunks, "foo"));
+}
+
 TEST(RtcpPacketTest, CnameItemWithEmptyString) {
   Sdes sdes;
-  sdes.WithCName(kSenderSsrc, "");
+  EXPECT_TRUE(sdes.WithCName(kSenderSsrc, ""));
 
-  RawPacket packet = sdes.Build();
+  rtc::scoped_ptr<RawPacket> packet(sdes.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.sdes()->num_packets());
   EXPECT_EQ(1, parser.sdes_chunk()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.sdes_chunk()->Ssrc());
   EXPECT_EQ("", parser.sdes_chunk()->Cname());
 }
 
-TEST(RtcpPacketTest, Pli) {
-  Pli pli;
-  pli.From(kSenderSsrc);
-  pli.To(kRemoteSsrc);
-
-  RawPacket packet = pli.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.pli()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.pli()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.pli()->MediaSsrc());
-}
-
-TEST(RtcpPacketTest, Sli) {
-  const uint16_t kFirstMb = 7777;
-  const uint16_t kNumberOfMb = 6666;
-  const uint8_t kPictureId = 60;
-  Sli sli;
-  sli.From(kSenderSsrc);
-  sli.To(kRemoteSsrc);
-  sli.WithFirstMb(kFirstMb);
-  sli.WithNumberOfMb(kNumberOfMb);
-  sli.WithPictureId(kPictureId);
-
-  RawPacket packet = sli.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.sli()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.sli()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.sli()->MediaSsrc());
-  EXPECT_EQ(1, parser.sli_item()->num_packets());
-  EXPECT_EQ(kFirstMb, parser.sli_item()->FirstMb());
-  EXPECT_EQ(kNumberOfMb, parser.sli_item()->NumberOfMb());
-  EXPECT_EQ(kPictureId, parser.sli_item()->PictureId());
-}
-
-TEST(RtcpPacketTest, Nack) {
-  Nack nack;
-  const uint16_t kList[] = {0, 1, 3, 8, 16};
-  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
-  nack.From(kSenderSsrc);
-  nack.To(kRemoteSsrc);
-  nack.WithList(kList, kListLength);
-  RawPacket packet = nack.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.nack()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc());
-  EXPECT_EQ(1, parser.nack_item()->num_packets());
-  std::vector<uint16_t> seqs = parser.nack_item()->last_nack_list();
-  EXPECT_EQ(kListLength, seqs.size());
-  for (size_t i = 0; i < kListLength; ++i) {
-    EXPECT_EQ(kList[i], seqs[i]);
-  }
-}
-
-TEST(RtcpPacketTest, NackWithWrap) {
-  Nack nack;
-  const uint16_t kList[] = {65500, 65516, 65534, 65535, 0, 1, 3, 20, 100};
-  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
-  nack.From(kSenderSsrc);
-  nack.To(kRemoteSsrc);
-  nack.WithList(kList, kListLength);
-  RawPacket packet = nack.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.nack()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.nack()->Ssrc());
-  EXPECT_EQ(kRemoteSsrc, parser.nack()->MediaSsrc());
-  EXPECT_EQ(4, parser.nack_item()->num_packets());
-  std::vector<uint16_t> seqs = parser.nack_item()->last_nack_list();
-  EXPECT_EQ(kListLength, seqs.size());
-  for (size_t i = 0; i < kListLength; ++i) {
-    EXPECT_EQ(kList[i], seqs[i]);
-  }
-}
-
 TEST(RtcpPacketTest, Rpsi) {
   Rpsi rpsi;
   // 1000001 (7 bits = 1 byte in native string).
@@ -375,9 +223,9 @@ TEST(RtcpPacketTest, Rpsi) {
   rpsi.WithPayloadType(100);
   rpsi.WithPictureId(kPictureId);
 
-  RawPacket packet = rpsi.Build();
+  rtc::scoped_ptr<RawPacket> packet(rpsi.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(100, parser.rpsi()->PayloadType());
   EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits());
   EXPECT_EQ(kPictureId, parser.rpsi()->PictureId());
@@ -390,9 +238,9 @@ TEST(RtcpPacketTest, RpsiWithTwoByteNativeString) {
   const uint16_t kNumberOfValidBytes = 2;
   rpsi.WithPictureId(kPictureId);
 
-  RawPacket packet = rpsi.Build();
+  rtc::scoped_ptr<RawPacket> packet(rpsi.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits());
   EXPECT_EQ(kPictureId, parser.rpsi()->PictureId());
 }
@@ -404,9 +252,9 @@ TEST(RtcpPacketTest, RpsiWithThreeByteNativeString) {
   const uint16_t kNumberOfValidBytes = 3;
   rpsi.WithPictureId(kPictureId);
 
-  RawPacket packet = rpsi.Build();
+  rtc::scoped_ptr<RawPacket> packet(rpsi.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits());
   EXPECT_EQ(kPictureId, parser.rpsi()->PictureId());
 }
@@ -418,9 +266,9 @@ TEST(RtcpPacketTest, RpsiWithFourByteNativeString) {
   const uint16_t kNumberOfValidBytes = 4;
   rpsi.WithPictureId(kPictureId);
 
-  RawPacket packet = rpsi.Build();
+  rtc::scoped_ptr<RawPacket> packet(rpsi.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits());
   EXPECT_EQ(kPictureId, parser.rpsi()->PictureId());
 }
@@ -433,9 +281,9 @@ TEST(RtcpPacketTest, RpsiWithMaxPictureId) {
   const uint16_t kNumberOfValidBytes = 10;
   rpsi.WithPictureId(kPictureId);
 
-  RawPacket packet = rpsi.Build();
+  rtc::scoped_ptr<RawPacket> packet(rpsi.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits());
   EXPECT_EQ(kPictureId, parser.rpsi()->PictureId());
 }
@@ -446,9 +294,9 @@ TEST(RtcpPacketTest, Fir) {
   fir.To(kRemoteSsrc);
   fir.WithCommandSeqNum(123);
 
-  RawPacket packet = fir.Build();
+  rtc::scoped_ptr<RawPacket> packet(fir.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.fir()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.fir()->Ssrc());
   EXPECT_EQ(1, parser.fir_item()->num_packets());
@@ -456,144 +304,24 @@ TEST(RtcpPacketTest, Fir) {
   EXPECT_EQ(123U, parser.fir_item()->SeqNum());
 }
 
-TEST(RtcpPacketTest, AppendPacket) {
-  Fir fir;
-  ReportBlock rb;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-  rr.Append(&fir);
-
-  RawPacket packet = rr.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
-TEST(RtcpPacketTest, AppendPacketOnEmpty) {
-  Empty empty;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  empty.Append(&rr);
-
-  RawPacket packet = empty.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(0, parser.report_block()->num_packets());
-}
-
-TEST(RtcpPacketTest, AppendPacketWithOwnAppendedPacket) {
-  Fir fir;
-  Bye bye;
-  ReportBlock rb;
-
-  ReceiverReport rr;
-  rr.WithReportBlock(&rb);
-  rr.Append(&fir);
-
-  SenderReport sr;
-  sr.Append(&bye);
-  sr.Append(&rr);
-
-  RawPacket packet = sr.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.sender_report()->num_packets());
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.bye()->num_packets());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
-TEST(RtcpPacketTest, Bye) {
-  Bye bye;
-  bye.From(kSenderSsrc);
-
-  RawPacket packet = bye.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.bye()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc());
-}
-
-TEST(RtcpPacketTest, ByeWithCsrcs) {
-  Fir fir;
-  Bye bye;
-  bye.From(kSenderSsrc);
-  bye.WithCsrc(0x22222222);
-  bye.WithCsrc(0x33333333);
-  bye.Append(&fir);
-
-  RawPacket packet = bye.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.bye()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.bye()->Ssrc());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
-TEST(RtcpPacketTest, BuildWithInputBuffer) {
-  Fir fir;
-  ReportBlock rb;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-  rr.Append(&fir);
-
-  const size_t kRrLength = 8;
-  const size_t kReportBlockLength = 24;
-  const size_t kFirLength = 20;
-
-  size_t len = 0;
-  uint8_t packet[kRrLength + kReportBlockLength + kFirLength];
-  rr.Build(packet, &len, kRrLength + kReportBlockLength + kFirLength);
-
-  RtcpPacketParser parser;
-  parser.Parse(packet, len);
-  EXPECT_EQ(1, parser.receiver_report()->num_packets());
-  EXPECT_EQ(1, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.fir()->num_packets());
-}
-
 TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
   ReportBlock rb;
   ReceiverReport rr;
   rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
+  EXPECT_TRUE(rr.WithReportBlock(rb));
 
   const size_t kRrLength = 8;
   const size_t kReportBlockLength = 24;
 
   // No packet.
-  size_t len = 0;
-  uint8_t packet[kRrLength + kReportBlockLength - 1];
-  rr.Build(packet, &len, kRrLength + kReportBlockLength - 1);
-  EXPECT_EQ(0U, len);
-}
-
-TEST(RtcpPacketTest, BuildWithTooSmallBuffer_LastBlockFits) {
-  Fir fir;
-  ReportBlock rb;
-  ReceiverReport rr;
-  rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-  rr.Append(&fir);
-
-  const size_t kRrLength = 8;
-  const size_t kReportBlockLength = 24;
-
-  size_t len = 0;
-  uint8_t packet[kRrLength + kReportBlockLength - 1];
-  rr.Build(packet, &len, kRrLength + kReportBlockLength - 1);
-  RtcpPacketParser parser;
-  parser.Parse(packet, len);
-  EXPECT_EQ(0, parser.receiver_report()->num_packets());
-  EXPECT_EQ(0, parser.report_block()->num_packets());
-  EXPECT_EQ(1, parser.fir()->num_packets());
+  class Verifier : public rtcp::RtcpPacket::PacketReadyCallback {
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      ADD_FAILURE() << "Packet should not fit within max size.";
+    }
+  } verifier;
+  const size_t kBufferSize = kRrLength + kReportBlockLength - 1;
+  uint8_t buffer[kBufferSize];
+  EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
 }
 
 TEST(RtcpPacketTest, Remb) {
@@ -604,9 +332,9 @@ TEST(RtcpPacketTest, Remb) {
   remb.AppliesTo(kRemoteSsrc + 2);
   remb.WithBitrateBps(261011);
 
-  RawPacket packet = remb.Build();
+  rtc::scoped_ptr<RawPacket> packet(remb.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.psfb_app()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.psfb_app()->Ssrc());
   EXPECT_EQ(1, parser.remb_item()->num_packets());
@@ -617,93 +345,27 @@ TEST(RtcpPacketTest, Remb) {
   EXPECT_EQ(kRemoteSsrc + 2, ssrcs[2]);
 }
 
-TEST(RtcpPacketTest, Tmmbr) {
-  Tmmbr tmmbr;
-  tmmbr.From(kSenderSsrc);
-  tmmbr.To(kRemoteSsrc);
-  tmmbr.WithBitrateKbps(312);
-  tmmbr.WithOverhead(60);
-
-  RawPacket packet = tmmbr.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.tmmbr()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbr()->Ssrc());
-  EXPECT_EQ(1, parser.tmmbr_item()->num_packets());
-  EXPECT_EQ(312U, parser.tmmbr_item()->BitrateKbps());
-  EXPECT_EQ(60U, parser.tmmbr_item()->Overhead());
-}
-
-TEST(RtcpPacketTest, TmmbnWithNoItem) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-
-  RawPacket packet = tmmbn.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.tmmbn()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
-  EXPECT_EQ(0, parser.tmmbn_items()->num_packets());
-}
-
-TEST(RtcpPacketTest, TmmbnWithOneItem) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-  tmmbn.WithTmmbr(kRemoteSsrc, 312, 60);
-
-  RawPacket packet = tmmbn.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.tmmbn()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
-  EXPECT_EQ(1, parser.tmmbn_items()->num_packets());
-  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
-  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
-  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
-}
-
-TEST(RtcpPacketTest, TmmbnWithTwoItems) {
-  Tmmbn tmmbn;
-  tmmbn.From(kSenderSsrc);
-  tmmbn.WithTmmbr(kRemoteSsrc, 312, 60);
-  tmmbn.WithTmmbr(kRemoteSsrc + 1, 1288, 40);
-
-  RawPacket packet = tmmbn.Build();
-  RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
-  EXPECT_EQ(1, parser.tmmbn()->num_packets());
-  EXPECT_EQ(kSenderSsrc, parser.tmmbn()->Ssrc());
-  EXPECT_EQ(2, parser.tmmbn_items()->num_packets());
-  EXPECT_EQ(kRemoteSsrc, parser.tmmbn_items()->Ssrc(0));
-  EXPECT_EQ(312U, parser.tmmbn_items()->BitrateKbps(0));
-  EXPECT_EQ(60U, parser.tmmbn_items()->Overhead(0));
-  EXPECT_EQ(kRemoteSsrc + 1, parser.tmmbn_items()->Ssrc(1));
-  EXPECT_EQ(1288U, parser.tmmbn_items()->BitrateKbps(1));
-  EXPECT_EQ(40U, parser.tmmbn_items()->Overhead(1));
-}
-
 TEST(RtcpPacketTest, XrWithNoReportBlocks) {
   Xr xr;
   xr.From(kSenderSsrc);
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
 }
 
 TEST(RtcpPacketTest, XrWithRrtr) {
   Rrtr rrtr;
-  rrtr.WithNtpSec(0x11111111);
-  rrtr.WithNtpFrac(0x22222222);
+  rrtr.WithNtp(NtpTime(0x11111111, 0x22222222));
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithRrtr(&rrtr);
+  EXPECT_TRUE(xr.WithRrtr(&rrtr));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(1, parser.rrtr()->num_packets());
@@ -713,19 +375,17 @@ TEST(RtcpPacketTest, XrWithRrtr) {
 
 TEST(RtcpPacketTest, XrWithTwoRrtrBlocks) {
   Rrtr rrtr1;
-  rrtr1.WithNtpSec(0x11111111);
-  rrtr1.WithNtpFrac(0x22222222);
+  rrtr1.WithNtp(NtpTime(0x11111111, 0x22222222));
   Rrtr rrtr2;
-  rrtr2.WithNtpSec(0x33333333);
-  rrtr2.WithNtpFrac(0x44444444);
+  rrtr2.WithNtp(NtpTime(0x33333333, 0x44444444));
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithRrtr(&rrtr1);
-  xr.WithRrtr(&rrtr2);
+  EXPECT_TRUE(xr.WithRrtr(&rrtr1));
+  EXPECT_TRUE(xr.WithRrtr(&rrtr2));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(2, parser.rrtr()->num_packets());
@@ -735,14 +395,14 @@ TEST(RtcpPacketTest, XrWithTwoRrtrBlocks) {
 
 TEST(RtcpPacketTest, XrWithDlrrWithOneSubBlock) {
   Dlrr dlrr;
-  dlrr.WithDlrrItem(0x11111111, 0x22222222, 0x33333333);
+  EXPECT_TRUE(dlrr.WithDlrrItem(0x11111111, 0x22222222, 0x33333333));
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithDlrr(&dlrr);
+  EXPECT_TRUE(xr.WithDlrr(&dlrr));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(1, parser.dlrr()->num_packets());
@@ -754,15 +414,15 @@ TEST(RtcpPacketTest, XrWithDlrrWithOneSubBlock) {
 
 TEST(RtcpPacketTest, XrWithDlrrWithTwoSubBlocks) {
   Dlrr dlrr;
-  dlrr.WithDlrrItem(0x11111111, 0x22222222, 0x33333333);
-  dlrr.WithDlrrItem(0x44444444, 0x55555555, 0x66666666);
+  EXPECT_TRUE(dlrr.WithDlrrItem(0x11111111, 0x22222222, 0x33333333));
+  EXPECT_TRUE(dlrr.WithDlrrItem(0x44444444, 0x55555555, 0x66666666));
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithDlrr(&dlrr);
+  EXPECT_TRUE(xr.WithDlrr(&dlrr));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(1, parser.dlrr()->num_packets());
@@ -775,19 +435,27 @@ TEST(RtcpPacketTest, XrWithDlrrWithTwoSubBlocks) {
   EXPECT_EQ(0x66666666U, parser.dlrr_items()->DelayLastRr(1));
 }
 
+TEST(RtcpPacketTest, DlrrWithTooManySubBlocks) {
+  const int kMaxItems = 100;
+  Dlrr dlrr;
+  for (int i = 0; i < kMaxItems; ++i)
+    EXPECT_TRUE(dlrr.WithDlrrItem(i, i, i));
+  EXPECT_FALSE(dlrr.WithDlrrItem(kMaxItems, kMaxItems, kMaxItems));
+}
+
 TEST(RtcpPacketTest, XrWithTwoDlrrBlocks) {
   Dlrr dlrr1;
-  dlrr1.WithDlrrItem(0x11111111, 0x22222222, 0x33333333);
+  EXPECT_TRUE(dlrr1.WithDlrrItem(0x11111111, 0x22222222, 0x33333333));
   Dlrr dlrr2;
-  dlrr2.WithDlrrItem(0x44444444, 0x55555555, 0x66666666);
+  EXPECT_TRUE(dlrr2.WithDlrrItem(0x44444444, 0x55555555, 0x66666666));
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithDlrr(&dlrr1);
-  xr.WithDlrr(&dlrr2);
+  EXPECT_TRUE(xr.WithDlrr(&dlrr1));
+  EXPECT_TRUE(xr.WithDlrr(&dlrr2));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(2, parser.dlrr()->num_packets());
@@ -801,36 +469,37 @@ TEST(RtcpPacketTest, XrWithTwoDlrrBlocks) {
 }
 
 TEST(RtcpPacketTest, XrWithVoipMetric) {
-  VoipMetric metric;
-  metric.To(kRemoteSsrc);
-  metric.LossRate(1);
-  metric.DiscardRate(2);
-  metric.BurstDensity(3);
-  metric.GapDensity(4);
-  metric.BurstDuration(0x1111);
-  metric.GapDuration(0x2222);
-  metric.RoundTripDelay(0x3333);
-  metric.EndSystemDelay(0x4444);
-  metric.SignalLevel(5);
-  metric.NoiseLevel(6);
-  metric.Rerl(7);
-  metric.Gmin(8);
-  metric.Rfactor(9);
-  metric.ExtRfactor(10);
-  metric.MosLq(11);
-  metric.MosCq(12);
-  metric.RxConfig(13);
-  metric.JbNominal(0x5555);
-  metric.JbMax(0x6666);
-  metric.JbAbsMax(0x7777);
-
+  RTCPVoIPMetric metric;
+  metric.lossRate = 1;
+  metric.discardRate = 2;
+  metric.burstDensity = 3;
+  metric.gapDensity = 4;
+  metric.burstDuration = 0x1111;
+  metric.gapDuration = 0x2222;
+  metric.roundTripDelay = 0x3333;
+  metric.endSystemDelay = 0x4444;
+  metric.signalLevel = 5;
+  metric.noiseLevel = 6;
+  metric.RERL = 7;
+  metric.Gmin = 8;
+  metric.Rfactor = 9;
+  metric.extRfactor = 10;
+  metric.MOSLQ = 11;
+  metric.MOSCQ = 12;
+  metric.RXconfig = 13;
+  metric.JBnominal = 0x5555;
+  metric.JBmax = 0x6666;
+  metric.JBabsMax = 0x7777;
+  VoipMetric metric_block;
+  metric_block.To(kRemoteSsrc);
+  metric_block.WithVoipMetric(metric);
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithVoipMetric(&metric);
+  EXPECT_TRUE(xr.WithVoipMetric(&metric_block));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(1, parser.voip_metric()->num_packets());
@@ -860,17 +529,17 @@ TEST(RtcpPacketTest, XrWithVoipMetric) {
 TEST(RtcpPacketTest, XrWithMultipleReportBlocks) {
   Rrtr rrtr;
   Dlrr dlrr;
-  dlrr.WithDlrrItem(1, 2, 3);
+  EXPECT_TRUE(dlrr.WithDlrrItem(1, 2, 3));
   VoipMetric metric;
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithRrtr(&rrtr);
-  xr.WithDlrr(&dlrr);
-  xr.WithVoipMetric(&metric);
+  EXPECT_TRUE(xr.WithRrtr(&rrtr));
+  EXPECT_TRUE(xr.WithDlrr(&dlrr));
+  EXPECT_TRUE(xr.WithVoipMetric(&metric));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(1, parser.rrtr()->num_packets());
@@ -885,17 +554,38 @@ TEST(RtcpPacketTest, DlrrWithoutItemNotIncludedInPacket) {
   VoipMetric metric;
   Xr xr;
   xr.From(kSenderSsrc);
-  xr.WithRrtr(&rrtr);
-  xr.WithDlrr(&dlrr);
-  xr.WithVoipMetric(&metric);
+  EXPECT_TRUE(xr.WithRrtr(&rrtr));
+  EXPECT_TRUE(xr.WithDlrr(&dlrr));
+  EXPECT_TRUE(xr.WithVoipMetric(&metric));
 
-  RawPacket packet = xr.Build();
+  rtc::scoped_ptr<RawPacket> packet(xr.Build());
   RtcpPacketParser parser;
-  parser.Parse(packet.buffer(), packet.buffer_length());
+  parser.Parse(packet->Buffer(), packet->Length());
   EXPECT_EQ(1, parser.xr_header()->num_packets());
   EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
   EXPECT_EQ(1, parser.rrtr()->num_packets());
   EXPECT_EQ(0, parser.dlrr()->num_packets());
   EXPECT_EQ(1, parser.voip_metric()->num_packets());
 }
+
+TEST(RtcpPacketTest, XrWithTooManyBlocks) {
+  const int kMaxBlocks = 50;
+  Xr xr;
+
+  Rrtr rrtr;
+  for (int i = 0; i < kMaxBlocks; ++i)
+    EXPECT_TRUE(xr.WithRrtr(&rrtr));
+  EXPECT_FALSE(xr.WithRrtr(&rrtr));
+
+  Dlrr dlrr;
+  for (int i = 0; i < kMaxBlocks; ++i)
+    EXPECT_TRUE(xr.WithDlrr(&dlrr));
+  EXPECT_FALSE(xr.WithDlrr(&dlrr));
+
+  VoipMetric voip_metric;
+  for (int i = 0; i < kMaxBlocks; ++i)
+    EXPECT_TRUE(xr.WithVoipMetric(&voip_metric));
+  EXPECT_FALSE(xr.WithVoipMetric(&voip_metric));
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index beb8538ef8..ba8630c718 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -10,40 +10,51 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 
-#include <assert.h> //assert
-#include <string.h> //memset
+#include <assert.h>
+#include <string.h>
 
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
-using namespace RTCPUtility;
-using namespace RTCPHelp;
+using RTCPHelp::RTCPPacketInformation;
+using RTCPHelp::RTCPReceiveInformation;
+using RTCPHelp::RTCPReportBlockInformation;
+using RTCPUtility::kBtVoipMetric;
+using RTCPUtility::RTCPCnameInformation;
+using RTCPUtility::RTCPPacketReportBlockItem;
+using RTCPUtility::RTCPPacketTypes;
 
 // The number of RTCP time intervals needed to trigger a timeout.
 const int kRrTimeoutIntervals = 3;
 
+const int64_t kMaxWarningLogIntervalMs = 10000;
+
 RTCPReceiver::RTCPReceiver(
-    int32_t id,
     Clock* clock,
+    bool receiver_only,
     RtcpPacketTypeCounterObserver* packet_type_counter_observer,
     RtcpBandwidthObserver* rtcp_bandwidth_observer,
     RtcpIntraFrameObserver* rtcp_intra_frame_observer,
+    TransportFeedbackObserver* transport_feedback_observer,
     ModuleRtpRtcpImpl* owner)
     : TMMBRHelp(),
       _clock(clock),
-      _method(kRtcpOff),
+      receiver_only_(receiver_only),
+      _method(RtcpMode::kOff),
       _lastReceived(0),
       _rtpRtcp(*owner),
       _criticalSectionFeedbacks(
           CriticalSectionWrapper::CreateCriticalSection()),
       _cbRtcpBandwidthObserver(rtcp_bandwidth_observer),
       _cbRtcpIntraFrameObserver(rtcp_intra_frame_observer),
+      _cbTransportFeedbackObserver(transport_feedback_observer),
       _criticalSectionRTCPReceiver(
           CriticalSectionWrapper::CreateCriticalSection()),
       main_ssrc_(0),
@@ -59,7 +70,9 @@ RTCPReceiver::RTCPReceiver(
       _lastReceivedRrMs(0),
       _lastIncreasedSequenceNumberMs(0),
       stats_callback_(NULL),
-      packet_type_counter_observer_(packet_type_counter_observer) {
+      packet_type_counter_observer_(packet_type_counter_observer),
+      num_skipped_packets_(0),
+      last_skipped_packets_warning_(clock->TimeInMilliseconds()) {
   memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
 }
 
@@ -90,12 +103,12 @@ RTCPReceiver::~RTCPReceiver() {
   }
 }
 
-RTCPMethod RTCPReceiver::Status() const {
+RtcpMode RTCPReceiver::Status() const {
   CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
   return _method;
 }
 
-void RTCPReceiver::SetRTCPStatus(RTCPMethod method) {
+void RTCPReceiver::SetRTCPStatus(RtcpMode method) {
   CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
   _method = method;
 }
@@ -304,72 +317,74 @@ RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
     }
 
     RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
-    while (pktType != RTCPUtility::kRtcpNotValidCode)
-    {
+    while (pktType != RTCPPacketTypes::kInvalid) {
         // Each "case" is responsible for iterate the parser to the
         // next top level packet.
         switch (pktType)
         {
-        case RTCPUtility::kRtcpSrCode:
-        case RTCPUtility::kRtcpRrCode:
+          case RTCPPacketTypes::kSr:
+          case RTCPPacketTypes::kRr:
             HandleSenderReceiverReport(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpSdesCode:
-            HandleSDES(*rtcpParser);
+          case RTCPPacketTypes::kSdes:
+            HandleSDES(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpXrHeaderCode:
+          case RTCPPacketTypes::kXrHeader:
             HandleXrHeader(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpXrReceiverReferenceTimeCode:
+          case RTCPPacketTypes::kXrReceiverReferenceTime:
             HandleXrReceiveReferenceTime(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpXrDlrrReportBlockCode:
+          case RTCPPacketTypes::kXrDlrrReportBlock:
             HandleXrDlrrReportBlock(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpXrVoipMetricCode:
+          case RTCPPacketTypes::kXrVoipMetric:
             HandleXRVOIPMetric(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpByeCode:
+          case RTCPPacketTypes::kBye:
             HandleBYE(*rtcpParser);
             break;
-        case RTCPUtility::kRtcpRtpfbNackCode:
+          case RTCPPacketTypes::kRtpfbNack:
             HandleNACK(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpRtpfbTmmbrCode:
+          case RTCPPacketTypes::kRtpfbTmmbr:
             HandleTMMBR(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpRtpfbTmmbnCode:
+          case RTCPPacketTypes::kRtpfbTmmbn:
             HandleTMMBN(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpRtpfbSrReqCode:
+          case RTCPPacketTypes::kRtpfbSrReq:
             HandleSR_REQ(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpPsfbPliCode:
+          case RTCPPacketTypes::kPsfbPli:
             HandlePLI(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpPsfbSliCode:
+          case RTCPPacketTypes::kPsfbSli:
             HandleSLI(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpPsfbRpsiCode:
+          case RTCPPacketTypes::kPsfbRpsi:
             HandleRPSI(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpExtendedIjCode:
+          case RTCPPacketTypes::kExtendedIj:
             HandleIJ(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpPsfbFirCode:
+          case RTCPPacketTypes::kPsfbFir:
             HandleFIR(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpPsfbAppCode:
+          case RTCPPacketTypes::kPsfbApp:
             HandlePsfbApp(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpAppCode:
+          case RTCPPacketTypes::kApp:
             // generic application messages
             HandleAPP(*rtcpParser, rtcpPacketInformation);
             break;
-        case RTCPUtility::kRtcpAppItemCode:
+          case RTCPPacketTypes::kAppItem:
             // generic application messages
             HandleAPPItem(*rtcpParser, rtcpPacketInformation);
             break;
+          case RTCPPacketTypes::kTransportFeedback:
+            HandleTransportFeedback(rtcpParser, &rtcpPacketInformation);
+            break;
         default:
             rtcpParser->Iterate();
             break;
@@ -382,10 +397,22 @@ RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
           main_ssrc_, packet_type_counter_);
     }
 
+    num_skipped_packets_ += rtcpParser->NumSkippedBlocks();
+
+    int64_t now = _clock->TimeInMilliseconds();
+    if (now - last_skipped_packets_warning_ >= kMaxWarningLogIntervalMs &&
+        num_skipped_packets_ > 0) {
+      last_skipped_packets_warning_ = now;
+      LOG(LS_WARNING)
+          << num_skipped_packets_
+          << " RTCP blocks were skipped due to being malformed or of "
+             "unrecognized/unsupported type, during the past "
+          << (kMaxWarningLogIntervalMs / 1000) << " second period.";
+    }
+
     return 0;
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void
 RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
                                          RTCPPacketInformation& rtcpPacketInformation)
@@ -393,7 +420,8 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
     RTCPUtility::RTCPPacketTypes rtcpPacketType = rtcpParser.PacketType();
     const RTCPUtility::RTCPPacket& rtcpPacket   = rtcpParser.Packet();
 
-    assert((rtcpPacketType == RTCPUtility::kRtcpRrCode) || (rtcpPacketType == RTCPUtility::kRtcpSrCode));
+    assert((rtcpPacketType == RTCPPacketTypes::kRr) ||
+           (rtcpPacketType == RTCPPacketTypes::kSr));
 
     // SR.SenderSSRC
     // The synchronization source identifier for the originator of this SR packet
@@ -401,7 +429,9 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
     // rtcpPacket.RR.SenderSSRC
     // The source of the packet sender, same as of SR? or is this a CE?
 
-    const uint32_t remoteSSRC = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.SenderSSRC:rtcpPacket.SR.SenderSSRC;
+    const uint32_t remoteSSRC = (rtcpPacketType == RTCPPacketTypes::kRr)
+                                    ? rtcpPacket.RR.SenderSSRC
+                                    : rtcpPacket.SR.SenderSSRC;
 
     rtcpPacketInformation.remoteSSRC = remoteSSRC;
 
@@ -412,8 +442,7 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
         return;
     }
 
-    if (rtcpPacketType == RTCPUtility::kRtcpSrCode)
-    {
+    if (rtcpPacketType == RTCPPacketTypes::kSr) {
       TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "SR",
                            "remote_ssrc", remoteSSRC, "ssrc", main_ssrc_);
 
@@ -453,14 +482,12 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
 
     rtcpPacketType = rtcpParser.Iterate();
 
-    while (rtcpPacketType == RTCPUtility::kRtcpReportBlockItemCode)
-    {
+    while (rtcpPacketType == RTCPPacketTypes::kReportBlockItem) {
         HandleReportBlock(rtcpPacket, rtcpPacketInformation, remoteSSRC);
         rtcpPacketType = rtcpParser.Iterate();
     }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleReportBlock(
     const RTCPUtility::RTCPPacket& rtcpPacket,
     RTCPPacketInformation& rtcpPacketInformation,
@@ -541,7 +568,7 @@ void RTCPReceiver::HandleReportBlock(
                      reportBlock->lastReceivedRRNTPfrac);
 
   // time when we received this in MS
-  uint64_t receiveTimeMS = Clock::NtpToMs(reportBlock->lastReceivedRRNTPsecs,
+  uint32_t receiveTimeMS = Clock::NtpToMs(reportBlock->lastReceivedRRNTPsecs,
                                           reportBlock->lastReceivedRRNTPfrac);
 
   // Estimate RTT
@@ -751,7 +778,7 @@ bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
   return updateBoundingSet;
 }
 
-int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
+int32_t RTCPReceiver::BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec) {
   CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::iterator receiveInfoIt =
@@ -771,7 +798,7 @@ int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
         i++) {
       if(receiveInfo->TmmbnBoundingSet.Ssrc(i) == main_ssrc_) {
         // owner of bounding set
-        tmmbrOwner = true;
+        *tmmbrOwner = true;
       }
       boundingSetRec->SetEntry(i,
                                receiveInfo->TmmbnBoundingSet.Tmmbr(i),
@@ -782,16 +809,16 @@ int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
   return receiveInfo->TmmbnBoundingSet.lengthOfSet();
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
-void RTCPReceiver::HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser) {
+void RTCPReceiver::HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser,
+                              RTCPPacketInformation& rtcpPacketInformation) {
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpSdesChunkCode) {
+  while (pktType == RTCPPacketTypes::kSdesChunk) {
     HandleSDESChunk(rtcpParser);
     pktType = rtcpParser.Iterate();
   }
+  rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSdes;
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
   RTCPCnameInformation* cnameInfo =
@@ -809,11 +836,10 @@ void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
                               RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-  if (main_ssrc_ != rtcpPacket.NACK.MediaSSRC) {
+  if (receiver_only_ || main_ssrc_ != rtcpPacket.NACK.MediaSSRC) {
     // Not to us.
     rtcpParser.Iterate();
     return;
@@ -821,7 +847,7 @@ void RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
   rtcpPacketInformation.ResetNACKPacketIdArray();
 
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpRtpfbNackItemCode) {
+  while (pktType == RTCPPacketTypes::kRtpfbNackItem) {
     HandleNACKItem(rtcpPacket, rtcpPacketInformation);
     pktType = rtcpParser.Iterate();
   }
@@ -833,7 +859,6 @@ void RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void
 RTCPReceiver::HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
                              RTCPPacketInformation& rtcpPacketInformation) {
@@ -853,12 +878,10 @@ RTCPReceiver::HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
   rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpNack;
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
 
   // clear our lists
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
   ReportBlockMap::iterator it = _receivedReportBlockMap.begin();
   for (; it != _receivedReportBlockMap.end(); ++it) {
     ReportBlockInfoMap* info_map = &(it->second);
@@ -925,7 +948,7 @@ void RTCPReceiver::HandleXrDlrrReportBlock(
   // Iterate through sub-block(s), if any.
   RTCPUtility::RTCPPacketTypes packet_type = parser.Iterate();
 
-  while (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) {
+  while (packet_type == RTCPPacketTypes::kXrDlrrReportBlockItem) {
     HandleXrDlrrReportBlockItem(packet, rtcpPacketInformation);
     packet_type = parser.Iterate();
   }
@@ -969,15 +992,12 @@ void RTCPReceiver::HandleXrDlrrReportBlockItem(
   rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpXrDlrrReportBlock;
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void
 RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
                                  RTCPPacketInformation& rtcpPacketInformation)
 {
     const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
 
-    CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
     if(rtcpPacket.XRVOIPMetricItem.SSRC == main_ssrc_)
     {
         // Store VoIP metrics block if it's about me
@@ -1013,7 +1033,6 @@ RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
     rtcpParser.Iterate();
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
                              RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
@@ -1027,7 +1046,6 @@ void RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
   rtcpParser.Iterate();
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
                                RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
@@ -1058,13 +1076,12 @@ void RTCPReceiver::HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
   ptrReceiveInfo->VerifyAndAllocateTMMBRSet((uint32_t)maxNumOfTMMBRBlocks);
 
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpRtpfbTmmbrItemCode) {
+  while (pktType == RTCPPacketTypes::kRtpfbTmmbrItem) {
     HandleTMMBRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation, senderSSRC);
     pktType = rtcpParser.Iterate();
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo,
                                    const RTCPUtility::RTCPPacket& rtcpPacket,
                                    RTCPPacketInformation& rtcpPacketInformation,
@@ -1077,7 +1094,6 @@ void RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo,
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser,
                                RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
@@ -1103,20 +1119,18 @@ void RTCPReceiver::HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser,
   ptrReceiveInfo->VerifyAndAllocateBoundingSet((uint32_t)maxNumOfTMMBNBlocks);
 
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode) {
+  while (pktType == RTCPPacketTypes::kRtpfbTmmbnItem) {
     HandleTMMBNItem(*ptrReceiveInfo, rtcpPacket);
     pktType = rtcpParser.Iterate();
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
                                 RTCPPacketInformation& rtcpPacketInformation) {
   rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSrReq;
   rtcpParser.Iterate();
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleTMMBNItem(RTCPReceiveInformation& receiveInfo,
                                    const RTCPUtility::RTCPPacket& rtcpPacket) {
   receiveInfo.TmmbnBoundingSet.AddEntry(
@@ -1125,18 +1139,16 @@ void RTCPReceiver::HandleTMMBNItem(RTCPReceiveInformation& receiveInfo,
       rtcpPacket.TMMBNItem.SSRC);
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
                              RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpPsfbSliItemCode) {
+  while (pktType == RTCPPacketTypes::kPsfbSliItem) {
     HandleSLIItem(rtcpPacket, rtcpPacketInformation);
     pktType = rtcpParser.Iterate();
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
                                  RTCPPacketInformation& rtcpPacketInformation) {
   // in theory there could be multiple slices lost
@@ -1150,8 +1162,7 @@ RTCPReceiver::HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
 {
     const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
     RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-    if(pktType == RTCPUtility::kRtcpPsfbRpsiCode)
-    {
+    if (pktType == RTCPPacketTypes::kPsfbRpsi) {
         rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRpsi; // received signal that we have a confirmed reference picture
         if(rtcpPacket.RPSI.NumberOfValidBits%8 != 0)
         {
@@ -1173,26 +1184,24 @@ RTCPReceiver::HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
     }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
                                  RTCPPacketInformation& rtcpPacketInformation) {
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  if (pktType == RTCPUtility::kRtcpPsfbRembCode) {
+  if (pktType == RTCPPacketTypes::kPsfbRemb) {
     pktType = rtcpParser.Iterate();
-    if (pktType == RTCPUtility::kRtcpPsfbRembItemCode) {
+    if (pktType == RTCPPacketTypes::kPsfbRembItem) {
       HandleREMBItem(rtcpParser, rtcpPacketInformation);
       rtcpParser.Iterate();
     }
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
                             RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
 
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpExtendedIjItemCode) {
+  while (pktType == RTCPPacketTypes::kExtendedIjItem) {
     HandleIJItem(rtcpPacket, rtcpPacketInformation);
     pktType = rtcpParser.Iterate();
   }
@@ -1214,7 +1223,6 @@ void RTCPReceiver::HandleREMBItem(
       rtcpPacket.REMBItem.BitRate;
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
                              RTCPPacketInformation& rtcpPacketInformation) {
   const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
@@ -1222,13 +1230,12 @@ void RTCPReceiver::HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
       GetReceiveInformation(rtcpPacket.FIR.SenderSSRC);
 
   RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
-  while (pktType == RTCPUtility::kRtcpPsfbFirItemCode) {
+  while (pktType == RTCPPacketTypes::kPsfbFirItem) {
     HandleFIRItem(ptrReceiveInfo, rtcpPacket, rtcpPacketInformation);
     pktType = rtcpParser.Iterate();
   }
 }
 
-// no need for critsect we have _criticalSectionRTCPReceiver
 void RTCPReceiver::HandleFIRItem(RTCPReceiveInformation* receiveInfo,
                                  const RTCPUtility::RTCPPacket& rtcpPacket,
                                  RTCPPacketInformation& rtcpPacketInformation) {
@@ -1281,6 +1288,17 @@ void RTCPReceiver::HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
   rtcpParser.Iterate();
 }
 
+void RTCPReceiver::HandleTransportFeedback(
+    RTCPUtility::RTCPParserV2* rtcp_parser,
+    RTCPHelp::RTCPPacketInformation* rtcp_packet_information) {
+  rtcp::RtcpPacket* packet = rtcp_parser->ReleaseRtcpPacket();
+  RTC_DCHECK(packet != nullptr);
+  rtcp_packet_information->rtcpPacketTypeFlags |= kRtcpTransportFeedback;
+  rtcp_packet_information->transport_feedback_.reset(
+      static_cast<rtcp::TransportFeedback*>(packet));
+
+  rtcp_parser->Iterate();
+}
 int32_t RTCPReceiver::UpdateTMMBR() {
   int32_t numBoundingSet = 0;
   uint32_t bitrate = 0;
@@ -1343,16 +1361,20 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
     // Might trigger a OnReceivedBandwidthEstimateUpdate.
     UpdateTMMBR();
   }
-  unsigned int local_ssrc = 0;
+  uint32_t local_ssrc;
+  std::set<uint32_t> registered_ssrcs;
   {
     // We don't want to hold this critsect when triggering the callbacks below.
     CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
     local_ssrc = main_ssrc_;
+    registered_ssrcs = registered_ssrcs_;
   }
-  if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq) {
+  if (!receiver_only_ &&
+      (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq)) {
     _rtpRtcp.OnRequestSendReport();
   }
-  if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) {
+  if (!receiver_only_ &&
+      (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack)) {
     if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
       LOG(LS_VERBOSE) << "Incoming NACK length: "
                    << rtcpPacketInformation.nackSequenceNumbers.size();
@@ -1365,6 +1387,7 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
     // report can generate several RTCP packets, based on number relayed/mixed
     // a send report block should go out to all receivers.
     if (_cbRtcpIntraFrameObserver) {
+      RTC_DCHECK(!receiver_only_);
       if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
           (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) {
         if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) {
@@ -1386,14 +1409,15 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
       }
     }
     if (_cbRtcpBandwidthObserver) {
+      RTC_DCHECK(!receiver_only_);
       if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) {
         LOG(LS_VERBOSE) << "Incoming REMB: "
                         << rtcpPacketInformation.receiverEstimatedMaxBitrate;
         _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(
             rtcpPacketInformation.receiverEstimatedMaxBitrate);
       }
-      if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr ||
-          rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr) {
+      if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) ||
+          (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr)) {
         int64_t now = _clock->TimeInMilliseconds();
         _cbRtcpBandwidthObserver->OnReceivedRtcpReceiverReport(
             rtcpPacketInformation.report_blocks,
@@ -1401,9 +1425,19 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
             now);
       }
     }
+    if (_cbTransportFeedbackObserver &&
+        (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTransportFeedback)) {
+      uint32_t media_source_ssrc =
+          rtcpPacketInformation.transport_feedback_->GetMediaSourceSsrc();
+      if (media_source_ssrc == local_ssrc ||
+          registered_ssrcs.find(media_source_ssrc) != registered_ssrcs.end()) {
+        _cbTransportFeedbackObserver->OnTransportFeedback(
+            *rtcpPacketInformation.transport_feedback_.get());
+      }
+    }
   }
 
-  {
+  if (!receiver_only_) {
     CriticalSectionScoped cs(_criticalSectionFeedbacks);
     if (stats_callback_) {
       for (ReportBlockList::const_iterator it =
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 62df34f0dd..54f59eface 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -12,11 +12,11 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
 
 #include <map>
-#include <vector>
 #include <set>
+#include <vector>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -29,16 +29,17 @@ class ModuleRtpRtcpImpl;
 class RTCPReceiver : public TMMBRHelp
 {
 public:
- RTCPReceiver(int32_t id,
-              Clock* clock,
+ RTCPReceiver(Clock* clock,
+              bool receiver_only,
               RtcpPacketTypeCounterObserver* packet_type_counter_observer,
               RtcpBandwidthObserver* rtcp_bandwidth_observer,
               RtcpIntraFrameObserver* rtcp_intra_frame_observer,
+              TransportFeedbackObserver* transport_feedback_observer,
               ModuleRtpRtcpImpl* owner);
     virtual ~RTCPReceiver();
 
-    RTCPMethod Status() const;
-    void SetRTCPStatus(RTCPMethod method);
+    RtcpMode Status() const;
+    void SetRTCPStatus(RtcpMode method);
 
     int64_t LastReceived();
     int64_t LastReceivedReceiverReport() const;
@@ -108,7 +109,7 @@ public:
 
     bool UpdateRTCPReceiveInformationTimers();
 
-    int32_t BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec);
+    int32_t BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec);
 
     int32_t UpdateTMMBR();
 
@@ -129,97 +130,132 @@ protected:
 
     void HandleSenderReceiverReport(
         RTCPUtility::RTCPParserV2& rtcpParser,
-        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleReportBlock(
         const RTCPUtility::RTCPPacket& rtcpPacket,
         RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
-        uint32_t remoteSSRC);
+        uint32_t remoteSSRC)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
-    void HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser);
+    void HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser,
+                    RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
-    void HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser);
+    void HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleXrHeader(RTCPUtility::RTCPParserV2& parser,
-                        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleXrReceiveReferenceTime(
         RTCPUtility::RTCPParserV2& parser,
-        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleXrDlrrReportBlock(
         RTCPUtility::RTCPParserV2& parser,
-        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleXrDlrrReportBlockItem(
         const RTCPUtility::RTCPPacket& packet,
-        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleXRVOIPMetric(
         RTCPUtility::RTCPParserV2& rtcpParser,
-        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
-                    RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                    RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
-                        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
-    void HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser);
+    void HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
-                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
-                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
-                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
-                    RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                    RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
-                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser,
-                        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                        RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
-                  RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                  RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket,
-                      RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                      RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
-                     RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                     RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleTMMBRItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
                          const RTCPUtility::RTCPPacket& rtcpPacket,
                          RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
-                         uint32_t senderSSRC);
+                         uint32_t senderSSRC)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser,
-                     RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                     RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
-                      RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                      RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleTMMBNItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
-                         const RTCPUtility::RTCPPacket& rtcpPacket);
+                         const RTCPUtility::RTCPPacket& rtcpPacket)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
-                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleFIRItem(RTCPHelp::RTCPReceiveInformation* receiveInfo,
                        const RTCPUtility::RTCPPacket& rtcpPacket,
-                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser,
-                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                   RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
     void HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
-                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
+                       RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
+
+    void HandleTransportFeedback(
+        RTCPUtility::RTCPParserV2* rtcp_parser,
+        RTCPHelp::RTCPPacketInformation* rtcp_packet_information)
+        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
  private:
   typedef std::map<uint32_t, RTCPHelp::RTCPReceiveInformation*>
@@ -237,19 +273,21 @@ protected:
       uint32_t remote_ssrc, uint32_t source_ssrc) const
           EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPReceiver);
 
-  Clock* _clock;
-  RTCPMethod _method;
+  Clock* const _clock;
+  const bool receiver_only_;
+  RtcpMode _method;
   int64_t _lastReceived;
   ModuleRtpRtcpImpl& _rtpRtcp;
 
   CriticalSectionWrapper* _criticalSectionFeedbacks;
   RtcpBandwidthObserver* const _cbRtcpBandwidthObserver;
   RtcpIntraFrameObserver* const _cbRtcpIntraFrameObserver;
+  TransportFeedbackObserver* const _cbTransportFeedbackObserver;
 
   CriticalSectionWrapper* _criticalSectionRTCPReceiver;
-  uint32_t main_ssrc_;
-  uint32_t _remoteSSRC;
-  std::set<uint32_t> registered_ssrcs_;
+  uint32_t main_ssrc_ GUARDED_BY(_criticalSectionRTCPReceiver);
+  uint32_t _remoteSSRC GUARDED_BY(_criticalSectionRTCPReceiver);
+  std::set<uint32_t> registered_ssrcs_ GUARDED_BY(_criticalSectionRTCPReceiver);
 
   // Received send report
   RTCPSenderInfo _remoteSenderInfo;
@@ -286,6 +324,9 @@ protected:
   RtcpPacketTypeCounter packet_type_counter_;
 
   RTCPUtility::NackStats nack_stats_;
+
+  size_t num_skipped_packets_;
+  int64_t last_skipped_packets_warning_;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
index 7b9c70b227..4e3955d594 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
@@ -13,10 +13,11 @@
 #include <assert.h>  // assert
 #include <string.h>  // memset
 
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
-using namespace RTCPHelp;
+namespace RTCPHelp {
 
 RTCPPacketInformation::RTCPPacketInformation()
     : rtcpPacketTypeFlags(0),
@@ -36,8 +37,7 @@ RTCPPacketInformation::RTCPPacketInformation()
       rtp_timestamp(0),
       xr_originator_ssrc(0),
       xr_dlrr_item(false),
-      VoIPMetric(NULL) {
-}
+      VoIPMetric(nullptr) {}
 
 RTCPPacketInformation::~RTCPPacketInformation()
 {
@@ -197,4 +197,5 @@ void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
     const uint32_t minimumSize) {
   TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
 }
+}  // namespace RTCPHelp
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index de3b8c517a..1dc8412487 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -11,15 +11,20 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 
+#include <list>
+#include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"  // RTCPReportBlock
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPReportBlock
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace rtcp {
+class TransportFeedback;
+}
 namespace RTCPHelp
 {
 
@@ -88,8 +93,10 @@ public:
     bool xr_dlrr_item;
     RTCPVoIPMetric*  VoIPMetric;
 
+    rtc::scoped_ptr<rtcp::TransportFeedback> transport_feedback_;
+
 private:
-    DISALLOW_COPY_AND_ASSIGN(RTCPPacketInformation);
+    RTC_DISALLOW_COPY_AND_ASSIGN(RTCPPacketInformation);
 };
 
 class RTCPReceiveInformation
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index f2ea04be5c..5d2fda347e 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -15,11 +15,19 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-// Note: This file has no directory. Lint warning must be ignored.
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
@@ -33,23 +41,20 @@ namespace {  // Anonymous namespace; hide utility functions and classes.
 class TestTransport : public Transport,
                       public NullRtpData {
  public:
-  explicit TestTransport()
-      : rtcp_receiver_(NULL) {
-  }
+  explicit TestTransport() : rtcp_receiver_(nullptr) {}
   void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
     rtcp_receiver_ = rtcp_receiver;
   }
-  int SendPacket(int /*ch*/, const void* /*data*/, size_t /*len*/) override {
+  bool SendRtp(const uint8_t* /*data*/,
+               size_t /*len*/,
+               const PacketOptions& options) override {
     ADD_FAILURE();  // FAIL() gives a compile error.
-    return -1;
+    return false;
   }
 
-  // Injects an RTCP packet into the receiver.
-  int SendRTCPPacket(int /* ch */,
-                     const void* packet,
-                     size_t packet_len) override {
+  bool SendRtcp(const uint8_t* packet, size_t packet_len) override {
     ADD_FAILURE();
-    return 0;
+    return true;
   }
 
   int OnReceivedPayloadData(const uint8_t* payloadData,
@@ -63,29 +68,23 @@ class TestTransport : public Transport,
 
 class RtcpReceiverTest : public ::testing::Test {
  protected:
-  static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
-
   RtcpReceiverTest()
       : over_use_detector_options_(),
         system_clock_(1335900000),
         remote_bitrate_observer_(),
         remote_bitrate_estimator_(
-            RemoteBitrateEstimatorFactory().Create(
-                &remote_bitrate_observer_,
-                &system_clock_,
-                kMimdControl,
-                kRemoteBitrateEstimatorMinBitrateBps)) {
+            new RemoteBitrateEstimatorSingleStream(&remote_bitrate_observer_,
+                                                   &system_clock_)) {
     test_transport_ = new TestTransport();
 
     RtpRtcp::Configuration configuration;
-    configuration.id = 0;
     configuration.audio = false;
     configuration.clock = &system_clock_;
     configuration.outgoing_transport = test_transport_;
     configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
     rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
-    rtcp_receiver_ = new RTCPReceiver(0, &system_clock_, NULL, NULL, NULL,
-                                      rtp_rtcp_impl_);
+    rtcp_receiver_ = new RTCPReceiver(&system_clock_, false, nullptr, nullptr,
+                                      nullptr, nullptr, rtp_rtcp_impl_);
     test_transport_->SetRTCPReceiver(rtcp_receiver_);
   }
   ~RtcpReceiverTest() {
@@ -128,9 +127,10 @@ class RtcpReceiverTest : public ::testing::Test {
     rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac;
     rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
     rtcp_packet_info_.xr_dlrr_item = rtcpPacketInformation.xr_dlrr_item;
-    if (rtcpPacketInformation.VoIPMetric) {
+    if (rtcpPacketInformation.VoIPMetric)
       rtcp_packet_info_.AddVoIPMetric(rtcpPacketInformation.VoIPMetric);
-    }
+    rtcp_packet_info_.transport_feedback_.reset(
+        rtcpPacketInformation.transport_feedback_.release());
     return 0;
   }
 
@@ -155,8 +155,8 @@ TEST_F(RtcpReceiverTest, InjectSrPacket) {
   const uint32_t kSenderSsrc = 0x10203;
   rtcp::SenderReport sr;
   sr.From(kSenderSsrc);
-  rtcp::RawPacket p = sr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   // The parser will note the remote SSRC on a SR from other than his
   // expected peer, but will not flag that he's gotten a packet.
   EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
@@ -169,8 +169,8 @@ TEST_F(RtcpReceiverTest, InjectSrPacketFromExpectedPeer) {
   rtcp_receiver_->SetRemoteSSRC(kSenderSsrc);
   rtcp::SenderReport sr;
   sr.From(kSenderSsrc);
-  rtcp::RawPacket p = sr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
   EXPECT_EQ(kRtcpSr, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
@@ -179,8 +179,8 @@ TEST_F(RtcpReceiverTest, InjectRrPacket) {
   const uint32_t kSenderSsrc = 0x10203;
   rtcp::ReceiverReport rr;
   rr.From(kSenderSsrc);
-  rtcp::RawPacket p = rr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(rr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
   EXPECT_EQ(kRtcpRr, rtcp_packet_info_.rtcpPacketTypeFlags);
   ASSERT_EQ(0u, rtcp_packet_info_.report_blocks.size());
@@ -197,9 +197,9 @@ TEST_F(RtcpReceiverTest, InjectRrPacketWithReportBlockNotToUsIgnored) {
   rb.To(kSourceSsrc + 1);
   rtcp::ReceiverReport rr;
   rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-  rtcp::RawPacket p = rr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rr.WithReportBlock(rb);
+  rtc::scoped_ptr<rtcp::RawPacket> packet(rr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
   EXPECT_EQ(kRtcpRr, rtcp_packet_info_.rtcpPacketTypeFlags);
   ASSERT_EQ(0u, rtcp_packet_info_.report_blocks.size());
@@ -220,9 +220,9 @@ TEST_F(RtcpReceiverTest, InjectRrPacketWithOneReportBlock) {
   rb.To(kSourceSsrc);
   rtcp::ReceiverReport rr;
   rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-  rtcp::RawPacket p = rr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rr.WithReportBlock(rb);
+  rtc::scoped_ptr<rtcp::RawPacket> packet(rr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
   EXPECT_EQ(kRtcpRr, rtcp_packet_info_.rtcpPacketTypeFlags);
   ASSERT_EQ(1u, rtcp_packet_info_.report_blocks.size());
@@ -255,11 +255,11 @@ TEST_F(RtcpReceiverTest, InjectRrPacketWithTwoReportBlocks) {
 
   rtcp::ReceiverReport rr1;
   rr1.From(kSenderSsrc);
-  rr1.WithReportBlock(&rb1);
-  rr1.WithReportBlock(&rb2);
+  rr1.WithReportBlock(rb1);
+  rr1.WithReportBlock(rb2);
 
-  rtcp::RawPacket p1 = rr1.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> p1(rr1.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   ASSERT_EQ(2u, rtcp_packet_info_.report_blocks.size());
   EXPECT_EQ(10, rtcp_packet_info_.report_blocks.front().fractionLost);
   EXPECT_EQ(0, rtcp_packet_info_.report_blocks.back().fractionLost);
@@ -278,11 +278,11 @@ TEST_F(RtcpReceiverTest, InjectRrPacketWithTwoReportBlocks) {
 
   rtcp::ReceiverReport rr2;
   rr2.From(kSenderSsrc);
-  rr2.WithReportBlock(&rb3);
-  rr2.WithReportBlock(&rb4);
+  rr2.WithReportBlock(rb3);
+  rr2.WithReportBlock(rb4);
 
-  rtcp::RawPacket p2 = rr2.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> p2(rr2.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   ASSERT_EQ(2u, rtcp_packet_info_.report_blocks.size());
   EXPECT_EQ(kFracLost[0], rtcp_packet_info_.report_blocks.front().fractionLost);
   EXPECT_EQ(kFracLost[1], rtcp_packet_info_.report_blocks.back().fractionLost);
@@ -318,10 +318,10 @@ TEST_F(RtcpReceiverTest, InjectRrPacketsFromTwoRemoteSsrcs) {
   rb1.WithCumulativeLost(kCumLost[0]);
   rtcp::ReceiverReport rr1;
   rr1.From(kSenderSsrc1);
-  rr1.WithReportBlock(&rb1);
+  rr1.WithReportBlock(rb1);
 
-  rtcp::RawPacket p1 = rr1.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> p1(rr1.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   ASSERT_EQ(1u, rtcp_packet_info_.report_blocks.size());
   EXPECT_EQ(kFracLost[0], rtcp_packet_info_.report_blocks.front().fractionLost);
 
@@ -341,9 +341,9 @@ TEST_F(RtcpReceiverTest, InjectRrPacketsFromTwoRemoteSsrcs) {
   rb2.WithCumulativeLost(kCumLost[1]);
   rtcp::ReceiverReport rr2;
   rr2.From(kSenderSsrc2);
-  rr2.WithReportBlock(&rb2);
-  rtcp::RawPacket p2 = rr2.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  rr2.WithReportBlock(rb2);
+  rtc::scoped_ptr<rtcp::RawPacket> p2(rr2.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   ASSERT_EQ(1u, rtcp_packet_info_.report_blocks.size());
   EXPECT_EQ(kFracLost[1], rtcp_packet_info_.report_blocks.front().fractionLost);
 
@@ -368,39 +368,42 @@ TEST_F(RtcpReceiverTest, GetRtt) {
   rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs);
 
   // No report block received.
-  EXPECT_EQ(-1, rtcp_receiver_->RTT(kSenderSsrc, NULL, NULL, NULL, NULL));
+  EXPECT_EQ(
+      -1, rtcp_receiver_->RTT(kSenderSsrc, nullptr, nullptr, nullptr, nullptr));
 
   rtcp::ReportBlock rb;
   rb.To(kSourceSsrc);
   rtcp::ReceiverReport rr;
   rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb);
-  rtcp::RawPacket p = rr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rr.WithReportBlock(rb);
+  rtc::scoped_ptr<rtcp::RawPacket> packet(rr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
   EXPECT_EQ(kRtcpRr, rtcp_packet_info_.rtcpPacketTypeFlags);
   EXPECT_EQ(1u, rtcp_packet_info_.report_blocks.size());
-  EXPECT_EQ(0, rtcp_receiver_->RTT(kSenderSsrc, NULL, NULL, NULL, NULL));
+  EXPECT_EQ(
+      0, rtcp_receiver_->RTT(kSenderSsrc, nullptr, nullptr, nullptr, nullptr));
 
   // Report block not received.
-  EXPECT_EQ(-1, rtcp_receiver_->RTT(kSenderSsrc + 1, NULL, NULL, NULL, NULL));
+  EXPECT_EQ(-1, rtcp_receiver_->RTT(kSenderSsrc + 1, nullptr, nullptr, nullptr,
+                                    nullptr));
 }
 
 TEST_F(RtcpReceiverTest, InjectIjWithNoItem) {
-  rtcp::Ij ij;
-  rtcp::RawPacket p = ij.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtcp::ExtendedJitterReport ij;
+  rtc::scoped_ptr<rtcp::RawPacket> packet(ij.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
 TEST_F(RtcpReceiverTest, InjectIjWithOneItem) {
-  rtcp::Ij ij;
-  ij.WithJitterItem(0x11111111);
+  rtcp::ExtendedJitterReport ij;
+  ij.WithJitter(0x11213141);
 
-  rtcp::RawPacket p = ij.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(ij.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpTransmissionTimeOffset, rtcp_packet_info_.rtcpPacketTypeFlags);
-  EXPECT_EQ(0x11111111U, rtcp_packet_info_.interArrivalJitter);
+  EXPECT_EQ(0x11213141U, rtcp_packet_info_.interArrivalJitter);
 }
 
 TEST_F(RtcpReceiverTest, InjectAppWithNoData) {
@@ -412,8 +415,8 @@ TEST_F(RtcpReceiverTest, InjectAppWithNoData) {
   name += 'e';
   app.WithName(name);
 
-  rtcp::RawPacket p = app.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(app.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpApp, rtcp_packet_info_.rtcpPacketTypeFlags);
   EXPECT_EQ(30, rtcp_packet_info_.applicationSubType);
   EXPECT_EQ(name, rtcp_packet_info_.applicationName);
@@ -432,8 +435,8 @@ TEST_F(RtcpReceiverTest, InjectAppWithData) {
   const size_t kDataLength = sizeof(kData) / sizeof(kData[0]);
   app.WithData((const uint8_t*)kData, kDataLength);
 
-  rtcp::RawPacket p = app.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(app.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpApp, rtcp_packet_info_.rtcpPacketTypeFlags);
   EXPECT_EQ(30, rtcp_packet_info_.applicationSubType);
   EXPECT_EQ(name, rtcp_packet_info_.applicationName);
@@ -445,8 +448,8 @@ TEST_F(RtcpReceiverTest, InjectSdesWithOneChunk) {
   rtcp::Sdes sdes;
   sdes.WithCName(kSenderSsrc, "alice@host");
 
-  rtcp::RawPacket p = sdes.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sdes.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   char cName[RTCP_CNAME_SIZE];
   EXPECT_EQ(0, rtcp_receiver_->CNAME(kSenderSsrc, cName));
   EXPECT_EQ(0, strncmp(cName, "alice@host", RTCP_CNAME_SIZE));
@@ -457,16 +460,16 @@ TEST_F(RtcpReceiverTest, InjectByePacket_RemovesCname) {
   rtcp::Sdes sdes;
   sdes.WithCName(kSenderSsrc, "alice@host");
 
-  rtcp::RawPacket p = sdes.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sdes.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   char cName[RTCP_CNAME_SIZE];
   EXPECT_EQ(0, rtcp_receiver_->CNAME(kSenderSsrc, cName));
 
   // Verify that BYE removes the CNAME.
   rtcp::Bye bye;
   bye.From(kSenderSsrc);
-  rtcp::RawPacket p2 = bye.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> p2(bye.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   EXPECT_EQ(-1, rtcp_receiver_->CNAME(kSenderSsrc, cName));
 }
 
@@ -484,11 +487,11 @@ TEST_F(RtcpReceiverTest, InjectByePacket_RemovesReportBlocks) {
   rb2.To(kSourceSsrcs[1]);
   rtcp::ReceiverReport rr;
   rr.From(kSenderSsrc);
-  rr.WithReportBlock(&rb1);
-  rr.WithReportBlock(&rb2);
+  rr.WithReportBlock(rb1);
+  rr.WithReportBlock(rb2);
 
-  rtcp::RawPacket p1 = rr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> p1(rr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   ASSERT_EQ(2u, rtcp_packet_info_.report_blocks.size());
   std::vector<RTCPReportBlock> received_blocks;
   rtcp_receiver_->StatisticsReceived(&received_blocks);
@@ -497,14 +500,14 @@ TEST_F(RtcpReceiverTest, InjectByePacket_RemovesReportBlocks) {
   // Verify that BYE removes the report blocks.
   rtcp::Bye bye;
   bye.From(kSenderSsrc);
-  rtcp::RawPacket p2 = bye.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> p2(bye.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   received_blocks.clear();
   rtcp_receiver_->StatisticsReceived(&received_blocks);
   EXPECT_TRUE(received_blocks.empty());
 
   // Inject packet.
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   ASSERT_EQ(2u, rtcp_packet_info_.report_blocks.size());
   received_blocks.clear();
   rtcp_receiver_->StatisticsReceived(&received_blocks);
@@ -519,8 +522,8 @@ TEST_F(RtcpReceiverTest, InjectPliPacket) {
 
   rtcp::Pli pli;
   pli.To(kSourceSsrc);
-  rtcp::RawPacket p = pli.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(pli.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpPli, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
@@ -532,8 +535,8 @@ TEST_F(RtcpReceiverTest, PliPacketNotToUsIgnored) {
 
   rtcp::Pli pli;
   pli.To(kSourceSsrc + 1);
-  rtcp::RawPacket p = pli.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(pli.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
@@ -545,8 +548,8 @@ TEST_F(RtcpReceiverTest, InjectFirPacket) {
 
   rtcp::Fir fir;
   fir.To(kSourceSsrc);
-  rtcp::RawPacket p = fir.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(fir.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpFir, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
@@ -558,16 +561,16 @@ TEST_F(RtcpReceiverTest, FirPacketNotToUsIgnored) {
 
   rtcp::Fir fir;
   fir.To(kSourceSsrc + 1);
-  rtcp::RawPacket p = fir.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(fir.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
 TEST_F(RtcpReceiverTest, InjectSliPacket) {
   rtcp::Sli sli;
   sli.WithPictureId(40);
-  rtcp::RawPacket p = sli.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sli.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpSli, rtcp_packet_info_.rtcpPacketTypeFlags);
   EXPECT_EQ(40, rtcp_packet_info_.sliPictureId);
 }
@@ -575,8 +578,8 @@ TEST_F(RtcpReceiverTest, InjectSliPacket) {
 TEST_F(RtcpReceiverTest, XrPacketWithZeroReportBlocksIgnored) {
   rtcp::Xr xr;
   xr.From(0x2345);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
@@ -589,13 +592,15 @@ TEST_F(RtcpReceiverTest, InjectXrVoipPacket) {
   const uint8_t kLossRate = 123;
   rtcp::VoipMetric voip_metric;
   voip_metric.To(kSourceSsrc);
-  voip_metric.LossRate(kLossRate);
+  RTCPVoIPMetric metric;
+  metric.lossRate = kLossRate;
+  voip_metric.WithVoipMetric(metric);
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithVoipMetric(&voip_metric);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
-  ASSERT_TRUE(rtcp_packet_info_.VoIPMetric != NULL);
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
+  ASSERT_TRUE(rtcp_packet_info_.VoIPMetric != nullptr);
   EXPECT_EQ(kLossRate, rtcp_packet_info_.VoIPMetric->lossRate);
   EXPECT_EQ(kRtcpXrVoipMetric, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
@@ -611,21 +616,20 @@ TEST_F(RtcpReceiverTest, XrVoipPacketNotToUsIgnored) {
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithVoipMetric(&voip_metric);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
 TEST_F(RtcpReceiverTest, InjectXrReceiverReferenceTimePacket) {
   rtcp::Rrtr rrtr;
-  rrtr.WithNtpSec(0x10203);
-  rrtr.WithNtpFrac(0x40506);
+  rrtr.WithNtp(NtpTime(0x10203, 0x40506));
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithRrtr(&rrtr);
 
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpXrReceiverReferenceTime,
             rtcp_packet_info_.rtcpPacketTypeFlags);
 }
@@ -641,8 +645,8 @@ TEST_F(RtcpReceiverTest, XrDlrrPacketNotToUsIgnored) {
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithDlrr(&dlrr);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
   EXPECT_FALSE(rtcp_packet_info_.xr_dlrr_item);
 }
@@ -658,8 +662,8 @@ TEST_F(RtcpReceiverTest, InjectXrDlrrPacketWithSubBlock) {
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithDlrr(&dlrr);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   // The parser should note the DLRR report block item, but not flag the packet
   // since the RTT is not estimated.
   EXPECT_TRUE(rtcp_packet_info_.xr_dlrr_item);
@@ -678,8 +682,8 @@ TEST_F(RtcpReceiverTest, InjectXrDlrrPacketWithMultipleSubBlocks) {
   rtcp::Xr xr;
   xr.From(0x2345);
   xr.WithDlrr(&dlrr);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   // The parser should note the DLRR report block item, but not flag the packet
   // since the RTT is not estimated.
   EXPECT_TRUE(rtcp_packet_info_.xr_dlrr_item);
@@ -701,8 +705,8 @@ TEST_F(RtcpReceiverTest, InjectXrPacketWithMultipleReportBlocks) {
   xr.WithRrtr(&rrtr);
   xr.WithDlrr(&dlrr);
   xr.WithVoipMetric(&metric);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(static_cast<unsigned int>(kRtcpXrReceiverReferenceTime +
                                       kRtcpXrVoipMetric),
             rtcp_packet_info_.rtcpPacketTypeFlags);
@@ -729,13 +733,13 @@ TEST_F(RtcpReceiverTest, InjectXrPacketWithUnknownReportBlock) {
   xr.WithRrtr(&rrtr);
   xr.WithDlrr(&dlrr);
   xr.WithVoipMetric(&metric);
-  rtcp::RawPacket p = xr.Build();
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
   // Modify the DLRR block to have an unsupported block type, from 5 to 6.
-  uint8_t* buffer = const_cast<uint8_t*>(p.buffer());
+  uint8_t* buffer = const_cast<uint8_t*>(packet->Buffer());
   EXPECT_EQ(5, buffer[20]);
   buffer[20] = 6;
 
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(static_cast<unsigned int>(kRtcpXrReceiverReferenceTime +
                                       kRtcpXrVoipMetric),
             rtcp_packet_info_.rtcpPacketTypeFlags);
@@ -754,18 +758,17 @@ TEST_F(RtcpReceiverTest, LastReceivedXrReferenceTimeInfoInitiallyFalse) {
 
 TEST_F(RtcpReceiverTest, GetLastReceivedXrReferenceTimeInfo) {
   const uint32_t kSenderSsrc = 0x123456;
-  const uint32_t kNtpSec = 0x10203;
-  const uint32_t kNtpFrac = 0x40506;
-  const uint32_t kNtpMid = RTCPUtility::MidNtp(kNtpSec, kNtpFrac);
+  const NtpTime kNtp(0x10203, 0x40506);
+  const uint32_t kNtpMid =
+      RTCPUtility::MidNtp(kNtp.seconds(), kNtp.fractions());
 
   rtcp::Rrtr rrtr;
-  rrtr.WithNtpSec(kNtpSec);
-  rrtr.WithNtpFrac(kNtpFrac);
+  rrtr.WithNtp(kNtp);
   rtcp::Xr xr;
   xr.From(kSenderSsrc);
   xr.WithRrtr(&rrtr);
-  rtcp::RawPacket p = xr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpXrReceiverReferenceTime,
       rtcp_packet_info_.rtcpPacketTypeFlags);
 
@@ -802,16 +805,16 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
   rb1.WithExtHighestSeqNum(kSequenceNumber);
   rtcp::ReceiverReport rr1;
   rr1.From(kSenderSsrc);
-  rr1.WithReportBlock(&rb1);
-  rtcp::RawPacket p1 = rr1.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  rr1.WithReportBlock(rb1);
+  rtc::scoped_ptr<rtcp::RawPacket> p1(rr1.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs - 1);
   EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
   EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
 
   // Add a RR with the same extended max as the previous RR to trigger a
   // sequence number timeout, but not a RR timeout.
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   system_clock_.AdvanceTimeMilliseconds(2);
   EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
   EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
@@ -831,15 +834,15 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
   rb2.WithExtHighestSeqNum(kSequenceNumber + 1);
   rtcp::ReceiverReport rr2;
   rr2.From(kSenderSsrc);
-  rr2.WithReportBlock(&rb2);
-  rtcp::RawPacket p2 = rr2.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  rr2.WithReportBlock(rb2);
+  rtc::scoped_ptr<rtcp::RawPacket> p2(rr2.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
   EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
 
   // Verify we can get a timeout again once we've received new RR.
   system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   system_clock_.AdvanceTimeMilliseconds(kRtcpIntervalMs + 1);
   EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
   EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
@@ -849,7 +852,7 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
 
 TEST_F(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
   // This call is expected to fail because no data has arrived.
-  EXPECT_EQ(-1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  EXPECT_EQ(-1, rtcp_receiver_->TMMBRReceived(0, 0, nullptr));
 }
 
 TEST_F(RtcpReceiverTest, TmmbrPacketAccepted) {
@@ -867,10 +870,10 @@ TEST_F(RtcpReceiverTest, TmmbrPacketAccepted) {
   rtcp::SenderReport sr;
   sr.From(kSenderSsrc);
   sr.Append(&tmmbr);
-  rtcp::RawPacket p = sr.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
 
-  EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(0, 0, nullptr));
   TMMBRSet candidate_set;
   candidate_set.VerifyAndAllocateSet(1);
   EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(1, 0, &candidate_set));
@@ -890,13 +893,13 @@ TEST_F(RtcpReceiverTest, TmmbrPacketNotForUsIgnored) {
   rtcp::SenderReport sr;
   sr.From(kSenderSsrc);
   sr.Append(&tmmbr);
-  rtcp::RawPacket p = sr.Build();
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build());
 
   std::set<uint32_t> ssrcs;
   ssrcs.insert(kMediaFlowSsrc);
   rtcp_receiver_->SetSsrcs(kMediaFlowSsrc, ssrcs);
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
-  EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
+  EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, nullptr));
 }
 
 TEST_F(RtcpReceiverTest, TmmbrPacketZeroRateIgnored) {
@@ -914,10 +917,10 @@ TEST_F(RtcpReceiverTest, TmmbrPacketZeroRateIgnored) {
   rtcp::SenderReport sr;
   sr.From(kSenderSsrc);
   sr.Append(&tmmbr);
-  rtcp::RawPacket p = sr.Build();
+  rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build());
 
-  EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
-  EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
+  EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, nullptr));
 }
 
 TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
@@ -938,13 +941,13 @@ TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
     rtcp::SenderReport sr;
     sr.From(ssrc);
     sr.Append(&tmmbr);
-    rtcp::RawPacket p = sr.Build();
-    EXPECT_EQ(0, InjectRtcpPacket(p.buffer(), p.buffer_length()));
+    rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build());
+    EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
     // 5 seconds between each packet.
     system_clock_.AdvanceTimeMilliseconds(5000);
   }
   // It is now starttime + 15.
-  EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, nullptr));
   TMMBRSet candidate_set;
   candidate_set.VerifyAndAllocateSet(3);
   EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(3, 0, &candidate_set));
@@ -953,7 +956,7 @@ TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
   // seconds, timing out the first packet.
   system_clock_.AdvanceTimeMilliseconds(12000);
   // Odd behaviour: Just counting them does not trigger the timeout.
-  EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, nullptr));
   EXPECT_EQ(2, rtcp_receiver_->TMMBRReceived(3, 0, &candidate_set));
   EXPECT_EQ(kSenderSsrc + 1, candidate_set.Ssrc(0));
 }
@@ -1008,13 +1011,13 @@ TEST_F(RtcpReceiverTest, Callbacks) {
 
   rtcp::ReceiverReport rr1;
   rr1.From(kSenderSsrc);
-  rr1.WithReportBlock(&rb1);
-  rtcp::RawPacket p1 = rr1.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p1.buffer(), p1.buffer_length()));
+  rr1.WithReportBlock(rb1);
+  rtc::scoped_ptr<rtcp::RawPacket> p1(rr1.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p1->Buffer(), p1->Length()));
   EXPECT_TRUE(callback.Matches(kSourceSsrc, kSequenceNumber, kFractionLoss,
                                kCumulativeLoss, kJitter));
 
-  rtcp_receiver_->RegisterRtcpStatisticsCallback(NULL);
+  rtcp_receiver_->RegisterRtcpStatisticsCallback(nullptr);
 
   // Add arbitrary numbers, callback should not be called (retain old values).
   rtcp::ReportBlock rb2;
@@ -1026,13 +1029,75 @@ TEST_F(RtcpReceiverTest, Callbacks) {
 
   rtcp::ReceiverReport rr2;
   rr2.From(kSenderSsrc);
-  rr2.WithReportBlock(&rb2);
-  rtcp::RawPacket p2 = rr2.Build();
-  EXPECT_EQ(0, InjectRtcpPacket(p2.buffer(), p2.buffer_length()));
+  rr2.WithReportBlock(rb2);
+  rtc::scoped_ptr<rtcp::RawPacket> p2(rr2.Build());
+  EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
   EXPECT_TRUE(callback.Matches(kSourceSsrc, kSequenceNumber, kFractionLoss,
                                kCumulativeLoss, kJitter));
 }
 
+TEST_F(RtcpReceiverTest, ReceivesTransportFeedback) {
+  const uint32_t kSenderSsrc = 0x10203;
+  const uint32_t kSourceSsrc = 0x123456;
+
+  std::set<uint32_t> ssrcs;
+  ssrcs.insert(kSourceSsrc);
+  rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs);
+
+  rtcp::TransportFeedback packet;
+  packet.WithMediaSourceSsrc(kSourceSsrc);
+  packet.WithPacketSenderSsrc(kSenderSsrc);
+  packet.WithBase(1, 1000);
+  packet.WithReceivedPacket(1, 1000);
+
+  rtc::scoped_ptr<rtcp::RawPacket> built_packet = packet.Build();
+  ASSERT_TRUE(built_packet.get() != nullptr);
+
+  EXPECT_EQ(0,
+            InjectRtcpPacket(built_packet->Buffer(), built_packet->Length()));
+
+  EXPECT_NE(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpTransportFeedback);
+  EXPECT_TRUE(rtcp_packet_info_.transport_feedback_.get() != nullptr);
+}
+
+TEST_F(RtcpReceiverTest, HandlesInvalidTransportFeedback) {
+  const uint32_t kSenderSsrc = 0x10203;
+  const uint32_t kSourceSsrc = 0x123456;
+
+  std::set<uint32_t> ssrcs;
+  ssrcs.insert(kSourceSsrc);
+  rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs);
+
+  // Send a compound packet with a TransportFeedback followed by something else.
+  rtcp::TransportFeedback packet;
+  packet.WithMediaSourceSsrc(kSourceSsrc);
+  packet.WithPacketSenderSsrc(kSenderSsrc);
+  packet.WithBase(1, 1000);
+  packet.WithReceivedPacket(1, 1000);
+
+  static uint32_t kBitrateBps = 50000;
+  rtcp::Remb remb;
+  remb.From(kSourceSsrc);
+  remb.WithBitrateBps(kBitrateBps);
+  packet.Append(&remb);
+
+  rtc::scoped_ptr<rtcp::RawPacket> built_packet = packet.Build();
+  ASSERT_TRUE(built_packet.get() != nullptr);
+
+  // Modify the TransportFeedback packet so that it is invalid.
+  const size_t kStatusCountOffset = 14;
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &built_packet->MutableBuffer()[kStatusCountOffset], 42);
+
+  EXPECT_EQ(0,
+            InjectRtcpPacket(built_packet->Buffer(), built_packet->Length()));
+
+  // Transport feedback should be ignored, but next packet should work.
+  EXPECT_EQ(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpTransportFeedback);
+  EXPECT_NE(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpRemb);
+  EXPECT_EQ(kBitrateBps, rtcp_packet_info_.receiverEstimatedMaxBitrate);
+}
+
 }  // Anonymous namespace
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index b87a6a71dc..8684a429af 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -11,59 +11,61 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 
 #include <assert.h>  // assert
-#include <stdlib.h>  // rand
 #include <string.h>  // memcpy
 
 #include <algorithm>  // min
+#include <limits>     // max
+#include <utility>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 using RTCPUtility::RTCPCnameInformation;
 
-NACKStringBuilder::NACKStringBuilder() :
-    _stream(""), _count(0), _consecutive(false)
-{
-    // Empty.
-}
+NACKStringBuilder::NACKStringBuilder()
+    : stream_(""), count_(0), prevNack_(0), consecutive_(false) {}
 
 NACKStringBuilder::~NACKStringBuilder() {}
 
-void NACKStringBuilder::PushNACK(uint16_t nack)
-{
-    if (_count == 0)
-    {
-        _stream << nack;
-    } else if (nack == _prevNack + 1)
-    {
-        _consecutive = true;
-    } else
-    {
-        if (_consecutive)
-        {
-            _stream << "-" << _prevNack;
-            _consecutive = false;
-        }
-        _stream << "," << nack;
+void NACKStringBuilder::PushNACK(uint16_t nack) {
+  if (count_ == 0) {
+    stream_ << nack;
+  } else if (nack == prevNack_ + 1) {
+    consecutive_ = true;
+  } else {
+    if (consecutive_) {
+      stream_ << "-" << prevNack_;
+      consecutive_ = false;
     }
-    _count++;
-    _prevNack = nack;
+    stream_ << "," << nack;
+  }
+  count_++;
+  prevNack_ = nack;
 }
 
-std::string NACKStringBuilder::GetResult()
-{
-    if (_consecutive)
-    {
-        _stream << "-" << _prevNack;
-        _consecutive = false;
-    }
-    return _stream.str();
+std::string NACKStringBuilder::GetResult() {
+  if (consecutive_) {
+    stream_ << "-" << prevNack_;
+    consecutive_ = false;
+  }
+  return stream_.str();
 }
 
 RTCPSender::FeedbackState::FeedbackState()
@@ -75,367 +77,355 @@ RTCPSender::FeedbackState::FeedbackState()
       last_rr_ntp_secs(0),
       last_rr_ntp_frac(0),
       remote_sr(0),
-      has_last_xr_rr(false) {}
+      has_last_xr_rr(false),
+      module(nullptr) {}
+
+class PacketContainer : public rtcp::CompoundPacket,
+                        public rtcp::RtcpPacket::PacketReadyCallback {
+ public:
+  explicit PacketContainer(Transport* transport)
+      : transport_(transport), bytes_sent_(0) {}
+  virtual ~PacketContainer() {
+    for (RtcpPacket* packet : appended_packets_)
+      delete packet;
+  }
+
+  void OnPacketReady(uint8_t* data, size_t length) override {
+    if (transport_->SendRtcp(data, length))
+      bytes_sent_ += length;
+  }
+
+  size_t SendPackets() {
+    rtcp::CompoundPacket::Build(this);
+    return bytes_sent_;
+  }
+
+ private:
+  Transport* transport_;
+  size_t bytes_sent_;
+};
+
+class RTCPSender::RtcpContext {
+ public:
+  RtcpContext(const FeedbackState& feedback_state,
+              int32_t nack_size,
+              const uint16_t* nack_list,
+              bool repeat,
+              uint64_t picture_id,
+              uint32_t ntp_sec,
+              uint32_t ntp_frac,
+              PacketContainer* container)
+      : feedback_state_(feedback_state),
+        nack_size_(nack_size),
+        nack_list_(nack_list),
+        repeat_(repeat),
+        picture_id_(picture_id),
+        ntp_sec_(ntp_sec),
+        ntp_frac_(ntp_frac),
+        container_(container) {}
+
+  virtual ~RtcpContext() {}
+
+  const FeedbackState& feedback_state_;
+  const int32_t nack_size_;
+  const uint16_t* nack_list_;
+  const bool repeat_;
+  const uint64_t picture_id_;
+  const uint32_t ntp_sec_;
+  const uint32_t ntp_frac_;
+
+  PacketContainer* const container_;
+};
 
 RTCPSender::RTCPSender(
-    int32_t id,
     bool audio,
     Clock* clock,
     ReceiveStatistics* receive_statistics,
-    RtcpPacketTypeCounterObserver* packet_type_counter_observer)
-    : _id(id),
-      _audio(audio),
-      _clock(clock),
-      _method(kRtcpOff),
-      _criticalSectionTransport(
-          CriticalSectionWrapper::CreateCriticalSection()),
-      _cbTransport(NULL),
+    RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+    Transport* outgoing_transport)
+    : audio_(audio),
+      clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
+      method_(RtcpMode::kOff),
+      transport_(outgoing_transport),
 
-      _criticalSectionRTCPSender(
+      critical_section_rtcp_sender_(
           CriticalSectionWrapper::CreateCriticalSection()),
-      _usingNack(false),
-      _sending(false),
-      _sendTMMBN(false),
-      _REMB(false),
-      _sendREMB(false),
-      _TMMBR(false),
-      _IJ(false),
-      _nextTimeToSendRTCP(clock->TimeInMilliseconds()),
+      using_nack_(false),
+      sending_(false),
+      remb_enabled_(false),
+      next_time_to_send_rtcp_(clock->TimeInMilliseconds()),
       start_timestamp_(0),
       last_rtp_timestamp_(0),
       last_frame_capture_time_ms_(-1),
-      _SSRC(0),
-      _remoteSSRC(0),
-      _CNAME(),
+      ssrc_(0),
+      remote_ssrc_(0),
       receive_statistics_(receive_statistics),
-      internal_report_blocks_(),
-      external_report_blocks_(),
-      _csrcCNAMEs(),
-      _lastSendReport(),
-      _lastRTCPTime(),
-      _lastSRPacketCount(),
-      _lastSROctetCount(),
 
-      last_xr_rr_(),
+      sequence_number_fir_(0),
 
-      _sequenceNumberFIR(0),
+      remb_bitrate_(0),
 
-      _rembBitrate(0),
+      tmmbr_help_(),
+      tmmbr_send_(0),
+      packet_oh_send_(0),
 
-      _tmmbrHelp(),
-      _tmmbr_Send(0),
-      _packetOH_Send(0),
+      app_sub_type_(0),
+      app_name_(0),
+      app_data_(nullptr),
+      app_length_(0),
 
-      _appSend(false),
-      _appSubType(0),
-      _appName(),
-      _appData(NULL),
-      _appLength(0),
-
-      xrSendReceiverReferenceTimeEnabled_(false),
-      _xrSendVoIPMetric(false),
-      _xrVoIPMetric(),
+      xr_send_receiver_reference_time_enabled_(false),
       packet_type_counter_observer_(packet_type_counter_observer) {
-    memset(_CNAME, 0, sizeof(_CNAME));
-    memset(_lastSendReport, 0, sizeof(_lastSendReport));
-    memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
-    memset(_lastSRPacketCount, 0, sizeof(_lastSRPacketCount));
-    memset(_lastSROctetCount, 0, sizeof(_lastSROctetCount));
+  memset(last_send_report_, 0, sizeof(last_send_report_));
+  memset(last_rtcp_time_, 0, sizeof(last_rtcp_time_));
+  memset(lastSRPacketCount_, 0, sizeof(lastSRPacketCount_));
+  memset(lastSROctetCount_, 0, sizeof(lastSROctetCount_));
+  RTC_DCHECK(transport_ != nullptr);
+
+  builders_[kRtcpSr] = &RTCPSender::BuildSR;
+  builders_[kRtcpRr] = &RTCPSender::BuildRR;
+  builders_[kRtcpSdes] = &RTCPSender::BuildSDES;
+  builders_[kRtcpPli] = &RTCPSender::BuildPLI;
+  builders_[kRtcpFir] = &RTCPSender::BuildFIR;
+  builders_[kRtcpSli] = &RTCPSender::BuildSLI;
+  builders_[kRtcpRpsi] = &RTCPSender::BuildRPSI;
+  builders_[kRtcpRemb] = &RTCPSender::BuildREMB;
+  builders_[kRtcpBye] = &RTCPSender::BuildBYE;
+  builders_[kRtcpApp] = &RTCPSender::BuildAPP;
+  builders_[kRtcpTmmbr] = &RTCPSender::BuildTMMBR;
+  builders_[kRtcpTmmbn] = &RTCPSender::BuildTMMBN;
+  builders_[kRtcpNack] = &RTCPSender::BuildNACK;
+  builders_[kRtcpXrVoipMetric] = &RTCPSender::BuildVoIPMetric;
+  builders_[kRtcpXrReceiverReferenceTime] =
+      &RTCPSender::BuildReceiverReferenceTime;
+  builders_[kRtcpXrDlrrReportBlock] = &RTCPSender::BuildDlrr;
 }
 
-RTCPSender::~RTCPSender() {
-  delete [] _appData;
+RTCPSender::~RTCPSender() {}
 
-  while (!internal_report_blocks_.empty()) {
-    delete internal_report_blocks_.begin()->second;
-    internal_report_blocks_.erase(internal_report_blocks_.begin());
-  }
-  while (!external_report_blocks_.empty()) {
-    std::map<uint32_t, RTCPReportBlock*>::iterator it =
-        external_report_blocks_.begin();
-    delete it->second;
-    external_report_blocks_.erase(it);
-  }
-  while (!_csrcCNAMEs.empty()) {
-    std::map<uint32_t, RTCPCnameInformation*>::iterator it =
-        _csrcCNAMEs.begin();
-    delete it->second;
-    _csrcCNAMEs.erase(it);
-  }
-  delete _criticalSectionTransport;
-  delete _criticalSectionRTCPSender;
+RtcpMode RTCPSender::Status() const {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  return method_;
 }
 
-int32_t
-RTCPSender::RegisterSendTransport(Transport* outgoingTransport)
-{
-    CriticalSectionScoped lock(_criticalSectionTransport);
-    _cbTransport = outgoingTransport;
-    return 0;
-}
+void RTCPSender::SetRTCPStatus(RtcpMode method) {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  method_ = method;
 
-RTCPMethod
-RTCPSender::Status() const
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    return _method;
-}
-
-void RTCPSender::SetRTCPStatus(RTCPMethod method) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  _method = method;
-
-  if (method == kRtcpOff)
+  if (method == RtcpMode::kOff)
     return;
-  _nextTimeToSendRTCP =
-    _clock->TimeInMilliseconds() + RTCP_INTERVAL_RAPID_SYNC_MS / 2;
+  next_time_to_send_rtcp_ =
+    clock_->TimeInMilliseconds() + RTCP_INTERVAL_RAPID_SYNC_MS / 2;
 }
 
-bool
-RTCPSender::Sending() const
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    return _sending;
+bool RTCPSender::Sending() const {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  return sending_;
 }
 
-int32_t
-RTCPSender::SetSendingStatus(const FeedbackState& feedback_state, bool sending)
-{
-    bool sendRTCPBye = false;
-    {
-        CriticalSectionScoped lock(_criticalSectionRTCPSender);
+int32_t RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
+                                     bool sending) {
+  bool sendRTCPBye = false;
+  {
+    CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-        if(_method != kRtcpOff)
-        {
-            if(sending == false && _sending == true)
-            {
-                // Trigger RTCP bye
-                sendRTCPBye = true;
-            }
-        }
-        _sending = sending;
+    if (method_ != RtcpMode::kOff) {
+      if (sending == false && sending_ == true) {
+        // Trigger RTCP bye
+        sendRTCPBye = true;
+      }
     }
-    if(sendRTCPBye)
-    {
-        return SendRTCP(feedback_state, kRtcpBye);
-    }
-    return 0;
+    sending_ = sending;
+  }
+  if (sendRTCPBye)
+    return SendRTCP(feedback_state, kRtcpBye);
+  return 0;
 }
 
-bool
-RTCPSender::REMB() const
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    return _REMB;
+bool RTCPSender::REMB() const {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  return remb_enabled_;
 }
 
 void RTCPSender::SetREMBStatus(bool enable) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  _REMB = enable;
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  remb_enabled_ = enable;
 }
 
 void RTCPSender::SetREMBData(uint32_t bitrate,
                              const std::vector<uint32_t>& ssrcs) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  _rembBitrate = bitrate;
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  remb_bitrate_ = bitrate;
   remb_ssrcs_ = ssrcs;
 
-  _sendREMB = true;
+  if (remb_enabled_)
+    SetFlag(kRtcpRemb, false);
   // Send a REMB immediately if we have a new REMB. The frequency of REMBs is
   // throttled by the caller.
-  _nextTimeToSendRTCP = _clock->TimeInMilliseconds();
+  next_time_to_send_rtcp_ = clock_->TimeInMilliseconds();
 }
 
-bool
-RTCPSender::TMMBR() const
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    return _TMMBR;
+bool RTCPSender::TMMBR() const {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  return IsFlagPresent(RTCPPacketType::kRtcpTmmbr);
 }
 
 void RTCPSender::SetTMMBRStatus(bool enable) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  _TMMBR = enable;
-}
-
-bool
-RTCPSender::IJ() const
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    return _IJ;
-}
-
-void RTCPSender::SetIJStatus(bool enable) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  _IJ = enable;
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  if (enable) {
+    SetFlag(RTCPPacketType::kRtcpTmmbr, false);
+  } else {
+    ConsumeFlag(RTCPPacketType::kRtcpTmmbr, true);
+  }
 }
 
 void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   start_timestamp_ = start_timestamp;
 }
 
 void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
                                 int64_t capture_time_ms) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   last_rtp_timestamp_ = rtp_timestamp;
   if (capture_time_ms < 0) {
     // We don't currently get a capture time from VoiceEngine.
-    last_frame_capture_time_ms_ = _clock->TimeInMilliseconds();
+    last_frame_capture_time_ms_ = clock_->TimeInMilliseconds();
   } else {
     last_frame_capture_time_ms_ = capture_time_ms;
   }
 }
 
 void RTCPSender::SetSSRC(uint32_t ssrc) {
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-    if(_SSRC != 0)
-    {
-        // not first SetSSRC, probably due to a collision
-        // schedule a new RTCP report
-        // make sure that we send a RTP packet
-        _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 100;
-    }
-    _SSRC = ssrc;
+  if (ssrc_ != 0) {
+    // not first SetSSRC, probably due to a collision
+    // schedule a new RTCP report
+    // make sure that we send a RTP packet
+    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + 100;
+  }
+  ssrc_ = ssrc;
 }
 
-void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    _remoteSSRC = ssrc;
+void RTCPSender::SetRemoteSSRC(uint32_t ssrc) {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  remote_ssrc_ = ssrc;
 }
 
-int32_t RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
-  if (!cName)
+int32_t RTCPSender::SetCNAME(const char* c_name) {
+  if (!c_name)
     return -1;
 
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  _CNAME[RTCP_CNAME_SIZE - 1] = 0;
-  strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1);
+  RTC_DCHECK_LT(strlen(c_name), static_cast<size_t>(RTCP_CNAME_SIZE));
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  cname_ = c_name;
   return 0;
 }
 
-int32_t RTCPSender::AddMixedCNAME(uint32_t SSRC,
-                                  const char cName[RTCP_CNAME_SIZE]) {
-  assert(cName);
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  if (_csrcCNAMEs.size() >= kRtpCsrcSize) {
+int32_t RTCPSender::AddMixedCNAME(uint32_t SSRC, const char* c_name) {
+  assert(c_name);
+  RTC_DCHECK_LT(strlen(c_name), static_cast<size_t>(RTCP_CNAME_SIZE));
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  if (csrc_cnames_.size() >= kRtpCsrcSize)
     return -1;
-  }
-  RTCPCnameInformation* ptr = new RTCPCnameInformation();
-  ptr->name[RTCP_CNAME_SIZE - 1] = 0;
-  strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1);
-  _csrcCNAMEs[SSRC] = ptr;
+
+  csrc_cnames_[SSRC] = c_name;
   return 0;
 }
 
 int32_t RTCPSender::RemoveMixedCNAME(uint32_t SSRC) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  std::map<uint32_t, RTCPCnameInformation*>::iterator it =
-      _csrcCNAMEs.find(SSRC);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  auto it = csrc_cnames_.find(SSRC);
 
-  if (it == _csrcCNAMEs.end()) {
+  if (it == csrc_cnames_.end())
     return -1;
-  }
-  delete it->second;
-  _csrcCNAMEs.erase(it);
+
+  csrc_cnames_.erase(it);
   return 0;
 }
 
 bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
-/*
-    For audio we use a fix 5 sec interval
+  /*
+      For audio we use a fix 5 sec interval
 
-    For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
-        technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
-        that should be extremely rare
+      For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
+          technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
+          that should be extremely rare
 
 
-From RFC 3550
+  From RFC 3550
 
-    MAX RTCP BW is 5% if the session BW
-        A send report is approximately 65 bytes inc CNAME
-        A receiver report is approximately 28 bytes
+      MAX RTCP BW is 5% if the session BW
+          A send report is approximately 65 bytes inc CNAME
+          A receiver report is approximately 28 bytes
 
-    The RECOMMENDED value for the reduced minimum in seconds is 360
-      divided by the session bandwidth in kilobits/second.  This minimum
-      is smaller than 5 seconds for bandwidths greater than 72 kb/s.
+      The RECOMMENDED value for the reduced minimum in seconds is 360
+        divided by the session bandwidth in kilobits/second.  This minimum
+        is smaller than 5 seconds for bandwidths greater than 72 kb/s.
 
-    If the participant has not yet sent an RTCP packet (the variable
-      initial is true), the constant Tmin is set to 2.5 seconds, else it
-      is set to 5 seconds.
+      If the participant has not yet sent an RTCP packet (the variable
+        initial is true), the constant Tmin is set to 2.5 seconds, else it
+        is set to 5 seconds.
 
-    The interval between RTCP packets is varied randomly over the
-      range [0.5,1.5] times the calculated interval to avoid unintended
-      synchronization of all participants
+      The interval between RTCP packets is varied randomly over the
+        range [0.5,1.5] times the calculated interval to avoid unintended
+        synchronization of all participants
 
-    if we send
-    If the participant is a sender (we_sent true), the constant C is
-      set to the average RTCP packet size (avg_rtcp_size) divided by 25%
-      of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
-      number of senders.
+      if we send
+      If the participant is a sender (we_sent true), the constant C is
+        set to the average RTCP packet size (avg_rtcp_size) divided by 25%
+        of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
+        number of senders.
 
-    if we receive only
-      If we_sent is not true, the constant C is set
-      to the average RTCP packet size divided by 75% of the RTCP
-      bandwidth.  The constant n is set to the number of receivers
-      (members - senders).  If the number of senders is greater than
-      25%, senders and receivers are treated together.
+      if we receive only
+        If we_sent is not true, the constant C is set
+        to the average RTCP packet size divided by 75% of the RTCP
+        bandwidth.  The constant n is set to the number of receivers
+        (members - senders).  If the number of senders is greater than
+        25%, senders and receivers are treated together.
 
-    reconsideration NOT required for peer-to-peer
-      "timer reconsideration" is
-      employed.  This algorithm implements a simple back-off mechanism
-      which causes users to hold back RTCP packet transmission if the
-      group sizes are increasing.
+      reconsideration NOT required for peer-to-peer
+        "timer reconsideration" is
+        employed.  This algorithm implements a simple back-off mechanism
+        which causes users to hold back RTCP packet transmission if the
+        group sizes are increasing.
 
-      n = number of members
-      C = avg_size/(rtcpBW/4)
+        n = number of members
+        C = avg_size/(rtcpBW/4)
 
-   3. The deterministic calculated interval Td is set to max(Tmin, n*C).
+     3. The deterministic calculated interval Td is set to max(Tmin, n*C).
 
-   4. The calculated interval T is set to a number uniformly distributed
-      between 0.5 and 1.5 times the deterministic calculated interval.
+     4. The calculated interval T is set to a number uniformly distributed
+        between 0.5 and 1.5 times the deterministic calculated interval.
 
-   5. The resulting value of T is divided by e-3/2=1.21828 to compensate
-      for the fact that the timer reconsideration algorithm converges to
-      a value of the RTCP bandwidth below the intended average
-*/
+     5. The resulting value of T is divided by e-3/2=1.21828 to compensate
+        for the fact that the timer reconsideration algorithm converges to
+        a value of the RTCP bandwidth below the intended average
+  */
 
-    int64_t now = _clock->TimeInMilliseconds();
+  int64_t now = clock_->TimeInMilliseconds();
 
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-    if(_method == kRtcpOff)
-    {
-        return false;
-    }
-
-    if(!_audio && sendKeyframeBeforeRTP)
-    {
-        // for video key-frames we want to send the RTCP before the large key-frame
-        // if we have a 100 ms margin
-        now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
-    }
-
-    if(now >= _nextTimeToSendRTCP)
-    {
-        return true;
-
-    } else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin
-    {
-        // wrap
-        return true;
-    }
+  if (method_ == RtcpMode::kOff)
     return false;
-}
 
-uint32_t RTCPSender::LastSendReport(int64_t& lastRTCPTime)
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  if (!audio_ && sendKeyframeBeforeRTP) {
+    // for video key-frames we want to send the RTCP before the large key-frame
+    // if we have a 100 ms margin
+    now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
+  }
 
-    lastRTCPTime = _lastRTCPTime[0];
-    return _lastSendReport[0];
+  if (now >= next_time_to_send_rtcp_) {
+    return true;
+  } else if (now < 0x0000ffff &&
+             next_time_to_send_rtcp_ > 0xffff0000) {  // 65 sec margin
+    // wrap
+    return true;
+  }
+  return false;
 }
 
 bool
@@ -444,31 +434,27 @@ RTCPSender::GetSendReportMetadata(const uint32_t sendReport,
                                   uint32_t *packetCount,
                                   uint64_t *octetCount)
 {
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-    // This is only saved when we are the sender
-    if((_lastSendReport[0] == 0) || (sendReport == 0))
-    {
-        return false;
-    } else
-    {
-        for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i)
-        {
-            if( _lastSendReport[i] == sendReport)
-            {
-                *timeOfSend = _lastRTCPTime[i];
-                *packetCount = _lastSRPacketCount[i];
-                *octetCount = _lastSROctetCount[i];
-                return true;
-            }
-        }
-    }
+  // This is only saved when we are the sender
+  if ((last_send_report_[0] == 0) || (sendReport == 0)) {
     return false;
+  } else {
+    for (int i = 0; i < RTCP_NUMBER_OF_SR; ++i) {
+      if (last_send_report_[i] == sendReport) {
+        *timeOfSend = last_rtcp_time_[i];
+        *packetCount = lastSRPacketCount_[i];
+        *octetCount = lastSROctetCount_[i];
+        return true;
+      }
+    }
+  }
+  return false;
 }
 
 bool RTCPSender::SendTimeOfXrRrReport(uint32_t mid_ntp,
                                       int64_t* time_ms) const {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
   if (last_xr_rr_.empty()) {
     return false;
@@ -481,385 +467,99 @@ bool RTCPSender::SendTimeOfXrRrReport(uint32_t mid_ntp,
   return true;
 }
 
-int32_t RTCPSender::AddExternalReportBlock(
-    uint32_t SSRC,
-    const RTCPReportBlock* reportBlock) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  return AddReportBlock(SSRC, &external_report_blocks_, reportBlock);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
+  for (int i = (RTCP_NUMBER_OF_SR - 2); i >= 0; i--) {
+    // shift old
+    last_send_report_[i + 1] = last_send_report_[i];
+    last_rtcp_time_[i + 1] = last_rtcp_time_[i];
+    lastSRPacketCount_[i+1] = lastSRPacketCount_[i];
+    lastSROctetCount_[i+1] = lastSROctetCount_[i];
+  }
+
+  last_rtcp_time_[0] = Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_);
+  last_send_report_[0] = (ctx.ntp_sec_ << 16) + (ctx.ntp_frac_ >> 16);
+  lastSRPacketCount_[0] = ctx.feedback_state_.packets_sent;
+  lastSROctetCount_[0] = ctx.feedback_state_.media_bytes_sent;
+
+  // The timestamp of this RTCP packet should be estimated as the timestamp of
+  // the frame being captured at this moment. We are calculating that
+  // timestamp as the last frame's timestamp + the time since the last frame
+  // was captured.
+  uint32_t rtp_timestamp =
+      start_timestamp_ + last_rtp_timestamp_ +
+      (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) *
+          (ctx.feedback_state_.frequency_hz / 1000);
+
+  rtcp::SenderReport* report = new rtcp::SenderReport();
+  report->From(ssrc_);
+  report->WithNtpSec(ctx.ntp_sec_);
+  report->WithNtpFrac(ctx.ntp_frac_);
+  report->WithRtpTimestamp(rtp_timestamp);
+  report->WithPacketCount(ctx.feedback_state_.packets_sent);
+  report->WithOctetCount(ctx.feedback_state_.media_bytes_sent);
+
+  for (auto it : report_blocks_)
+    report->WithReportBlock(it.second);
+
+  report_blocks_.clear();
+
+  return rtc::scoped_ptr<rtcp::SenderReport>(report);
 }
 
-int32_t RTCPSender::AddReportBlock(
-    uint32_t SSRC,
-    std::map<uint32_t, RTCPReportBlock*>* report_blocks,
-    const RTCPReportBlock* reportBlock) {
-  assert(reportBlock);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES(
+    const RtcpContext& ctx) {
+  size_t length_cname = cname_.length();
+  RTC_CHECK_LT(length_cname, static_cast<size_t>(RTCP_CNAME_SIZE));
 
-  if (report_blocks->size() >= RTCP_MAX_REPORT_BLOCKS) {
-    LOG(LS_WARNING) << "Too many report blocks.";
-    return -1;
-  }
-  std::map<uint32_t, RTCPReportBlock*>::iterator it =
-      report_blocks->find(SSRC);
-  if (it != report_blocks->end()) {
-    delete it->second;
-    report_blocks->erase(it);
-  }
-  RTCPReportBlock* copyReportBlock = new RTCPReportBlock();
-  memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock));
-  (*report_blocks)[SSRC] = copyReportBlock;
-  return 0;
+  rtcp::Sdes* sdes = new rtcp::Sdes();
+  sdes->WithCName(ssrc_, cname_);
+
+  for (const auto it : csrc_cnames_)
+    sdes->WithCName(it.first, it.second);
+
+  return rtc::scoped_ptr<rtcp::Sdes>(sdes);
 }
 
-int32_t RTCPSender::RemoveExternalReportBlock(uint32_t SSRC) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildRR(const RtcpContext& ctx) {
+  rtcp::ReceiverReport* report = new rtcp::ReceiverReport();
+  report->From(ssrc_);
+  for (auto it : report_blocks_)
+    report->WithReportBlock(it.second);
 
-  std::map<uint32_t, RTCPReportBlock*>::iterator it =
-      external_report_blocks_.find(SSRC);
-
-  if (it == external_report_blocks_.end()) {
-    return -1;
-  }
-  delete it->second;
-  external_report_blocks_.erase(it);
-  return 0;
+  report_blocks_.clear();
+  return rtc::scoped_ptr<rtcp::ReceiverReport>(report);
 }
 
-int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state,
-                            uint8_t* rtcpbuffer,
-                            int& pos,
-                            uint32_t NTPsec,
-                            uint32_t NTPfrac)
-{
-    // sanity
-    if(pos + 52 >= IP_PACKET_SIZE)
-    {
-        LOG(LS_WARNING) << "Failed to build Sender Report.";
-        return -2;
-    }
-    uint32_t RTPtime;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildPLI(const RtcpContext& ctx) {
+  rtcp::Pli* pli = new rtcp::Pli();
+  pli->From(ssrc_);
+  pli->To(remote_ssrc_);
 
-    uint32_t posNumberOfReportBlocks = pos;
-    rtcpbuffer[pos++]=(uint8_t)0x80;
+  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                       "RTCPSender::PLI");
+  ++packet_type_counter_.pli_packets;
+  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_PLICount",
+                    ssrc_, packet_type_counter_.pli_packets);
 
-    // Sender report
-    rtcpbuffer[pos++]=(uint8_t)200;
-
-    for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--)
-    {
-        // shift old
-        _lastSendReport[i+1] = _lastSendReport[i];
-        _lastRTCPTime[i+1] =_lastRTCPTime[i];
-        _lastSRPacketCount[i+1] = _lastSRPacketCount[i];
-        _lastSROctetCount[i+1] = _lastSROctetCount[i];
-    }
-
-    _lastRTCPTime[0] = Clock::NtpToMs(NTPsec, NTPfrac);
-    _lastSendReport[0] = (NTPsec << 16) + (NTPfrac >> 16);
-    _lastSRPacketCount[0] = feedback_state.packets_sent;
-    _lastSROctetCount[0] = feedback_state.media_bytes_sent;
-
-    // The timestamp of this RTCP packet should be estimated as the timestamp of
-    // the frame being captured at this moment. We are calculating that
-    // timestamp as the last frame's timestamp + the time since the last frame
-    // was captured.
-    RTPtime = start_timestamp_ + last_rtp_timestamp_ +
-              (_clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
-                  (feedback_state.frequency_hz / 1000);
-
-    // Add sender data
-    // Save  for our length field
-    pos++;
-    pos++;
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-    // NTP
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, NTPsec);
-    pos += 4;
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, NTPfrac);
-    pos += 4;
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, RTPtime);
-    pos += 4;
-
-    //sender's packet count
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos,
-                                         feedback_state.packets_sent);
-    pos += 4;
-
-    //sender's octet count
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos,
-                                         feedback_state.media_bytes_sent);
-    pos += 4;
-
-    uint8_t numberOfReportBlocks = 0;
-    int32_t retVal = WriteAllReportBlocksToBuffer(rtcpbuffer, pos,
-                                                  numberOfReportBlocks,
-                                                  NTPsec, NTPfrac);
-    if(retVal < 0)
-    {
-        //
-        return retVal ;
-    }
-    pos = retVal;
-    rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
-
-    uint16_t len = uint16_t((pos/4) -1);
-    ByteWriter<uint16_t>::WriteBigEndian(rtcpbuffer + 2, len);
-    return 0;
+  return rtc::scoped_ptr<rtcp::Pli>(pli);
 }
 
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildFIR(const RtcpContext& ctx) {
+  if (!ctx.repeat_)
+    ++sequence_number_fir_;  // Do not increase if repetition.
 
-int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) {
-  size_t lengthCname = strlen(_CNAME);
-  assert(lengthCname < RTCP_CNAME_SIZE);
+  rtcp::Fir* fir = new rtcp::Fir();
+  fir->From(ssrc_);
+  fir->To(remote_ssrc_);
+  fir->WithCommandSeqNum(sequence_number_fir_);
 
-  // sanity
-  if(pos + 12 + lengthCname  >= IP_PACKET_SIZE) {
-    LOG(LS_WARNING) << "Failed to build SDEC.";
-    return -2;
-  }
-  // SDEC Source Description
+  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                       "RTCPSender::FIR");
+  ++packet_type_counter_.fir_packets;
+  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_FIRCount",
+                    ssrc_, packet_type_counter_.fir_packets);
 
-  // We always need to add SDES CNAME
-  rtcpbuffer[pos++] = static_cast<uint8_t>(0x80 + 1 + _csrcCNAMEs.size());
-  rtcpbuffer[pos++] = static_cast<uint8_t>(202);
-
-  // handle SDES length later on
-  uint32_t SDESLengthPos = pos;
-  pos++;
-  pos++;
-
-  // Add our own SSRC
-  ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-  pos += 4;
-
-  // CNAME = 1
-  rtcpbuffer[pos++] = static_cast<uint8_t>(1);
-
-  //
-  rtcpbuffer[pos++] = static_cast<uint8_t>(lengthCname);
-
-  uint16_t SDESLength = 10;
-
-  memcpy(&rtcpbuffer[pos], _CNAME, lengthCname);
-  pos += lengthCname;
-  SDESLength += (uint16_t)lengthCname;
-
-  uint16_t padding = 0;
-  // We must have a zero field even if we have an even multiple of 4 bytes
-  if ((pos % 4) == 0) {
-    padding++;
-    rtcpbuffer[pos++]=0;
-  }
-  while ((pos % 4) != 0) {
-    padding++;
-    rtcpbuffer[pos++]=0;
-  }
-  SDESLength += padding;
-
-  std::map<uint32_t, RTCPUtility::RTCPCnameInformation*>::iterator it =
-      _csrcCNAMEs.begin();
-
-  for(; it != _csrcCNAMEs.end(); it++) {
-    RTCPCnameInformation* cname = it->second;
-    uint32_t SSRC = it->first;
-
-    // Add SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, SSRC);
-    pos += 4;
-
-    // CNAME = 1
-    rtcpbuffer[pos++] = static_cast<uint8_t>(1);
-
-    size_t length = strlen(cname->name);
-    assert(length < RTCP_CNAME_SIZE);
-
-    rtcpbuffer[pos++]= static_cast<uint8_t>(length);
-    SDESLength += 6;
-
-    memcpy(&rtcpbuffer[pos],cname->name, length);
-
-    pos += length;
-    SDESLength += length;
-    uint16_t padding = 0;
-
-    // We must have a zero field even if we have an even multiple of 4 bytes
-    if((pos % 4) == 0){
-      padding++;
-      rtcpbuffer[pos++]=0;
-    }
-    while((pos % 4) != 0){
-      padding++;
-      rtcpbuffer[pos++] = 0;
-    }
-    SDESLength += padding;
-  }
-  // in 32-bit words minus one and we don't count the header
-  uint16_t buffer_length = (SDESLength / 4) - 1;
-  ByteWriter<uint16_t>::WriteBigEndian(rtcpbuffer + SDESLengthPos,
-                                       buffer_length);
-  return 0;
-}
-
-int32_t RTCPSender::BuildRR(uint8_t* rtcpbuffer,
-                            int& pos,
-                            uint32_t NTPsec,
-                            uint32_t NTPfrac) {
-    // sanity one block
-    if(pos + 32 >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
-    uint32_t posNumberOfReportBlocks = pos;
-
-    rtcpbuffer[pos++]=(uint8_t)0x80;
-    rtcpbuffer[pos++]=(uint8_t)201;
-
-    // Save  for our length field
-    pos++;
-    pos++;
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    uint8_t numberOfReportBlocks = 0;
-    int retVal = WriteAllReportBlocksToBuffer(rtcpbuffer, pos,
-                                              numberOfReportBlocks,
-                                              NTPsec, NTPfrac);
-    if(retVal < 0)
-    {
-        return pos;
-    }
-    pos = retVal;
-    rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
-
-    uint16_t len = uint16_t((pos)/4 -1);
-    ByteWriter<uint16_t>::WriteBigEndian(rtcpbuffer + 2, len);
-    return 0;
-}
-
-// From RFC 5450: Transmission Time Offsets in RTP Streams.
-//        0                   1                   2                   3
-//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//   hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       |                      inter-arrival jitter                     |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//       .                                                               .
-//       .                                                               .
-//       .                                                               .
-//       |                      inter-arrival jitter                     |
-//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//
-//  If present, this RTCP packet must be placed after a receiver report
-//  (inside a compound RTCP packet), and MUST have the same value for RC
-//  (reception report count) as the receiver report.
-
-int32_t
-RTCPSender::BuildExtendedJitterReport(
-    uint8_t* rtcpbuffer,
-    int& pos,
-    const uint32_t jitterTransmissionTimeOffset)
-{
-    if (external_report_blocks_.size() > 0)
-    {
-        // TODO(andresp): Remove external report blocks since they are not
-        // supported.
-        LOG(LS_ERROR) << "Handling of external report blocks not implemented.";
-        return 0;
-    }
-
-    // sanity
-    if(pos + 8 >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
-    // add picture loss indicator
-    uint8_t RC = 1;
-    rtcpbuffer[pos++]=(uint8_t)0x80 + RC;
-    rtcpbuffer[pos++]=(uint8_t)195;
-
-    // Used fixed length of 2
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=(uint8_t)(1);
-
-    // Add inter-arrival jitter
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos,
-                                         jitterTransmissionTimeOffset);
-    pos += 4;
-    return 0;
-}
-
-int32_t
-RTCPSender::BuildPLI(uint8_t* rtcpbuffer, int& pos)
-{
-    // sanity
-    if(pos + 12 >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
-    // add picture loss indicator
-    uint8_t FMT = 1;
-    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-    rtcpbuffer[pos++]=(uint8_t)206;
-
-    //Used fixed length of 2
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=(uint8_t)(2);
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Add the remote SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-    pos += 4;
-    return 0;
-}
-
-int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer,
-                             int& pos,
-                             bool repeat) {
-  // sanity
-  if(pos + 20 >= IP_PACKET_SIZE)  {
-    return -2;
-  }
-  if (!repeat) {
-    _sequenceNumberFIR++;   // do not increase if repetition
-  }
-
-  // add full intra request indicator
-  uint8_t FMT = 4;
-  rtcpbuffer[pos++] = (uint8_t)0x80 + FMT;
-  rtcpbuffer[pos++] = (uint8_t)206;
-
-  //Length of 4
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)(4);
-
-  // Add our own SSRC
-  ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-  pos += 4;
-
-  // RFC 5104     4.3.1.2.  Semantics
-  // SSRC of media source
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)0;
-
-  // Additional Feedback Control Information (FCI)
-  ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-  pos += 4;
-
-  rtcpbuffer[pos++] = (uint8_t)(_sequenceNumberFIR);
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)0;
-  return 0;
+  return rtc::scoped_ptr<rtcp::Fir>(fir);
 }
 
 /*
@@ -869,36 +569,14 @@ int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer,
    |            First        |        Number           | PictureID |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 */
-int32_t RTCPSender::BuildSLI(uint8_t* rtcpbuffer, int& pos, uint8_t pictureID) {
-    // sanity
-    if(pos + 16 >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
-    // add slice loss indicator
-    uint8_t FMT = 2;
-    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-    rtcpbuffer[pos++]=(uint8_t)206;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSLI(const RtcpContext& ctx) {
+  rtcp::Sli* sli = new rtcp::Sli();
+  sli->From(ssrc_);
+  sli->To(remote_ssrc_);
+  // Crop picture id to 6 least significant bits.
+  sli->WithPictureId(ctx.picture_id_ & 0x3F);
 
-    //Used fixed length of 3
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=(uint8_t)(3);
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Add the remote SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-    pos += 4;
-
-    // Add first, number & picture ID 6 bits
-    // first  = 0, 13 - bits
-    // number = 0x1fff, 13 - bits only ones for now
-    uint32_t sliField = (0x1fff << 6)+ (0x3f & pictureID);
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, sliField);
-    pos += 4;
-    return 0;
+  return rtc::scoped_ptr<rtcp::Sli>(sli);
 }
 
 /*
@@ -913,1064 +591,385 @@ int32_t RTCPSender::BuildSLI(uint8_t* rtcpbuffer, int& pos, uint8_t pictureID) {
 /*
 *    Note: not generic made for VP8
 */
-int32_t RTCPSender::BuildRPSI(uint8_t* rtcpbuffer,
-                              int& pos,
-                              uint64_t pictureID,
-                              uint8_t payloadType) {
-    // sanity
-    if(pos + 24 >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
-    // add Reference Picture Selection Indication
-    uint8_t FMT = 3;
-    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-    rtcpbuffer[pos++]=(uint8_t)206;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildRPSI(
+    const RtcpContext& ctx) {
+  if (ctx.feedback_state_.send_payload_type == 0xFF)
+    return nullptr;
 
-    // calc length
-    uint32_t bitsRequired = 7;
-    uint8_t bytesRequired = 1;
-    while((pictureID>>bitsRequired) > 0)
-    {
-        bitsRequired += 7;
-        bytesRequired++;
-    }
+  rtcp::Rpsi* rpsi = new rtcp::Rpsi();
+  rpsi->From(ssrc_);
+  rpsi->To(remote_ssrc_);
+  rpsi->WithPayloadType(ctx.feedback_state_.send_payload_type);
+  rpsi->WithPictureId(ctx.picture_id_);
 
-    uint8_t size = 3;
-    if(bytesRequired > 6)
-    {
-        size = 5;
-    } else if(bytesRequired > 2)
-    {
-        size = 4;
-    }
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=size;
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Add the remote SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-    pos += 4;
-
-    // calc padding length
-    uint8_t paddingBytes = 4-((2+bytesRequired)%4);
-    if(paddingBytes == 4)
-    {
-        paddingBytes = 0;
-    }
-    // add padding length in bits
-    rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24
-    pos++;
-
-    // add payload type
-    rtcpbuffer[pos] = payloadType;
-    pos++;
-
-    // add picture ID
-    for(int i = bytesRequired-1; i > 0; i--)
-    {
-        rtcpbuffer[pos] = 0x80 | uint8_t(pictureID >> (i*7));
-        pos++;
-    }
-    // add last byte of picture ID
-    rtcpbuffer[pos] = uint8_t(pictureID & 0x7f);
-    pos++;
-
-    // add padding
-    for(int j = 0; j <paddingBytes; j++)
-    {
-        rtcpbuffer[pos] = 0;
-        pos++;
-    }
-    return 0;
+  return rtc::scoped_ptr<rtcp::Rpsi>(rpsi);
 }
 
-int32_t
-RTCPSender::BuildREMB(uint8_t* rtcpbuffer, int& pos)
-{
-    // sanity
-    if(pos + 20 + 4 * remb_ssrcs_.size() >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
-    // add application layer feedback
-    uint8_t FMT = 15;
-    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-    rtcpbuffer[pos++]=(uint8_t)206;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildREMB(
+    const RtcpContext& ctx) {
+  rtcp::Remb* remb = new rtcp::Remb();
+  remb->From(ssrc_);
+  for (uint32_t ssrc : remb_ssrcs_)
+    remb->AppliesTo(ssrc);
+  remb->WithBitrateBps(remb_bitrate_);
 
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=remb_ssrcs_.size() + 4;
+  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                       "RTCPSender::REMB");
 
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Remote SSRC must be 0
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, 0);
-    pos += 4;
-
-    rtcpbuffer[pos++]='R';
-    rtcpbuffer[pos++]='E';
-    rtcpbuffer[pos++]='M';
-    rtcpbuffer[pos++]='B';
-
-    rtcpbuffer[pos++] = remb_ssrcs_.size();
-    // 6 bit Exp
-    // 18 bit mantissa
-    uint8_t brExp = 0;
-    for(uint32_t i=0; i<64; i++)
-    {
-        if(_rembBitrate <= ((uint32_t)262143 << i))
-        {
-            brExp = i;
-            break;
-        }
-    }
-    const uint32_t brMantissa = (_rembBitrate >> brExp);
-    rtcpbuffer[pos++]=(uint8_t)((brExp << 2) + ((brMantissa >> 16) & 0x03));
-    rtcpbuffer[pos++]=(uint8_t)(brMantissa >> 8);
-    rtcpbuffer[pos++]=(uint8_t)(brMantissa);
-
-    for (size_t i = 0; i < remb_ssrcs_.size(); i++)
-    {
-      ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, remb_ssrcs_[i]);
-      pos += 4;
-    }
-    return 0;
+  return rtc::scoped_ptr<rtcp::Remb>(remb);
 }
 
-void
-RTCPSender::SetTargetBitrate(unsigned int target_bitrate)
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    _tmmbr_Send = target_bitrate / 1000;
+void RTCPSender::SetTargetBitrate(unsigned int target_bitrate) {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  tmmbr_send_ = target_bitrate / 1000;
 }
 
-int32_t RTCPSender::BuildTMMBR(ModuleRtpRtcpImpl* rtp_rtcp_module,
-                               uint8_t* rtcpbuffer,
-                               int& pos) {
-    if (rtp_rtcp_module == NULL)
-      return -1;
-    // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate
-    // If the sender is an owner of the TMMBN -> send TMMBR
-    // If not an owner but the TMMBR would enter the TMMBN -> send TMMBR
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBR(
+    const RtcpContext& ctx) {
+  if (ctx.feedback_state_.module == nullptr)
+    return nullptr;
+  // Before sending the TMMBR check the received TMMBN, only an owner is
+  // allowed to raise the bitrate:
+  // * If the sender is an owner of the TMMBN -> send TMMBR
+  // * If not an owner but the TMMBR would enter the TMMBN -> send TMMBR
 
-    // get current bounding set from RTCP receiver
-    bool tmmbrOwner = false;
-    // store in candidateSet, allocates one extra slot
-    TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet();
+  // get current bounding set from RTCP receiver
+  bool tmmbrOwner = false;
+  // store in candidateSet, allocates one extra slot
+  TMMBRSet* candidateSet = tmmbr_help_.CandidateSet();
 
-    // holding _criticalSectionRTCPSender while calling RTCPreceiver which
-    // will accuire _criticalSectionRTCPReceiver is a potental deadlock but
-    // since RTCPreceiver is not doing the reverse we should be fine
-    int32_t lengthOfBoundingSet =
-        rtp_rtcp_module->BoundingSet(tmmbrOwner, candidateSet);
+  // holding critical_section_rtcp_sender_ while calling RTCPreceiver which
+  // will accuire criticalSectionRTCPReceiver_ is a potental deadlock but
+  // since RTCPreceiver is not doing the reverse we should be fine
+  int32_t lengthOfBoundingSet =
+      ctx.feedback_state_.module->BoundingSet(&tmmbrOwner, candidateSet);
 
-    if(lengthOfBoundingSet > 0)
-    {
-        for (int32_t i = 0; i < lengthOfBoundingSet; i++)
-        {
-            if( candidateSet->Tmmbr(i) == _tmmbr_Send &&
-                candidateSet->PacketOH(i) == _packetOH_Send)
-            {
-                // do not send the same tuple
-                return 0;
-            }
-        }
-        if(!tmmbrOwner)
-        {
-            // use received bounding set as candidate set
-            // add current tuple
-            candidateSet->SetEntry(lengthOfBoundingSet,
-                                   _tmmbr_Send,
-                                   _packetOH_Send,
-                                   _SSRC);
-            int numCandidates = lengthOfBoundingSet+ 1;
-
-            // find bounding set
-            TMMBRSet* boundingSet = NULL;
-            int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet);
-            if(numBoundingSet > 0 || numBoundingSet <= numCandidates)
-            {
-                tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet);
-            }
-            if(!tmmbrOwner)
-            {
-                // did not enter bounding set, no meaning to send this request
-                return 0;
-            }
-        }
-    }
-
-    if(_tmmbr_Send)
-    {
-        // sanity
-        if(pos + 20 >= IP_PACKET_SIZE)
-        {
-            return -2;
-        }
-        // add TMMBR indicator
-        uint8_t FMT = 3;
-        rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-        rtcpbuffer[pos++]=(uint8_t)205;
-
-        //Length of 4
-        rtcpbuffer[pos++]=(uint8_t)0;
-        rtcpbuffer[pos++]=(uint8_t)(4);
-
-        // Add our own SSRC
-        ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-        pos += 4;
-
-        // RFC 5104     4.2.1.2.  Semantics
-
-        // SSRC of media source
-        rtcpbuffer[pos++]=(uint8_t)0;
-        rtcpbuffer[pos++]=(uint8_t)0;
-        rtcpbuffer[pos++]=(uint8_t)0;
-        rtcpbuffer[pos++]=(uint8_t)0;
-
-        // Additional Feedback Control Information (FCI)
-        ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-        pos += 4;
-
-        uint32_t bitRate = _tmmbr_Send*1000;
-        uint32_t mmbrExp = 0;
-        for(uint32_t i=0;i<64;i++)
-        {
-            if(bitRate <= ((uint32_t)131071 << i))
-            {
-                mmbrExp = i;
-                break;
-            }
-        }
-        uint32_t mmbrMantissa = (bitRate >> mmbrExp);
-
-        rtcpbuffer[pos++]=(uint8_t)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
-        rtcpbuffer[pos++]=(uint8_t)(mmbrMantissa >> 7);
-        rtcpbuffer[pos++]=(uint8_t)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01));
-        rtcpbuffer[pos++]=(uint8_t)(_packetOH_Send);
-    }
-    return 0;
-}
-
-int32_t
-RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos)
-{
-    TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend();
-    if(boundingSet == NULL)
-    {
-        return -1;
-    }
-    // sanity
-    if(pos + 12 + boundingSet->lengthOfSet()*8 >= IP_PACKET_SIZE)
-    {
-        LOG(LS_WARNING) << "Failed to build TMMBN.";
-        return -2;
-    }
-    uint8_t FMT = 4;
-    // add TMMBN indicator
-    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-    rtcpbuffer[pos++]=(uint8_t)205;
-
-    //Add length later
-    int posLength = pos;
-    pos++;
-    pos++;
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // RFC 5104     4.2.2.2.  Semantics
-
-    // SSRC of media source
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=(uint8_t)0;
-    rtcpbuffer[pos++]=(uint8_t)0;
-
-    // Additional Feedback Control Information (FCI)
-    int numBoundingSet = 0;
-    for(uint32_t n=0; n< boundingSet->lengthOfSet(); n++)
-    {
-        if (boundingSet->Tmmbr(n) > 0)
-        {
-            uint32_t tmmbrSSRC = boundingSet->Ssrc(n);
-            ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, tmmbrSSRC);
-            pos += 4;
-
-            uint32_t bitRate = boundingSet->Tmmbr(n) * 1000;
-            uint32_t mmbrExp = 0;
-            for(int i=0; i<64; i++)
-            {
-                if(bitRate <=  ((uint32_t)131071 << i))
-                {
-                    mmbrExp = i;
-                    break;
-                }
-            }
-            uint32_t mmbrMantissa = (bitRate >> mmbrExp);
-            uint32_t measuredOH = boundingSet->PacketOH(n);
-
-            rtcpbuffer[pos++]=(uint8_t)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
-            rtcpbuffer[pos++]=(uint8_t)(mmbrMantissa >> 7);
-            rtcpbuffer[pos++]=(uint8_t)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01));
-            rtcpbuffer[pos++]=(uint8_t)(measuredOH);
-            numBoundingSet++;
-        }
-    }
-    uint16_t length= (uint16_t)(2+2*numBoundingSet);
-    rtcpbuffer[posLength++]=(uint8_t)(length>>8);
-    rtcpbuffer[posLength]=(uint8_t)(length);
-    return 0;
-}
-
-int32_t
-RTCPSender::BuildAPP(uint8_t* rtcpbuffer, int& pos)
-{
-    // sanity
-    if(_appData == NULL)
-    {
-        LOG(LS_WARNING) << "Failed to build app specific.";
-        return -1;
-    }
-    if(pos + 12 + _appLength >= IP_PACKET_SIZE)
-    {
-        LOG(LS_WARNING) << "Failed to build app specific.";
-        return -2;
-    }
-    rtcpbuffer[pos++]=(uint8_t)0x80 + _appSubType;
-
-    // Add APP ID
-    rtcpbuffer[pos++]=(uint8_t)204;
-
-    uint16_t length = (_appLength>>2) + 2; // include SSRC and name
-    rtcpbuffer[pos++]=(uint8_t)(length>>8);
-    rtcpbuffer[pos++]=(uint8_t)(length);
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Add our application name
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _appName);
-    pos += 4;
-
-    // Add the data
-    memcpy(rtcpbuffer +pos, _appData,_appLength);
-    pos += _appLength;
-    return 0;
-}
-
-int32_t RTCPSender::BuildNACK(uint8_t* rtcpbuffer,
-                              int& pos,
-                              int32_t nackSize,
-                              const uint16_t* nackList,
-                              std::string* nackString) {
-    // sanity
-    if(pos + 16 >= IP_PACKET_SIZE)
-    {
-        LOG(LS_WARNING) << "Failed to build NACK.";
-        return -2;
-    }
-
-    // int size, uint16_t* nackList
-    // add nack list
-    uint8_t FMT = 1;
-    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
-    rtcpbuffer[pos++]=(uint8_t)205;
-
-    rtcpbuffer[pos++]=(uint8_t) 0;
-    int nackSizePos = pos;
-    rtcpbuffer[pos++]=(uint8_t)(3); //setting it to one kNACK signal as default
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Add the remote SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-    pos += 4;
-
-    // Build NACK bitmasks and write them to the RTCP message.
-    // The nack list should be sorted and not contain duplicates if one
-    // wants to build the smallest rtcp nack packet.
-    int numOfNackFields = 0;
-    int maxNackFields = std::min<int>(kRtcpMaxNackFields,
-                                      (IP_PACKET_SIZE - pos) / 4);
-    int i = 0;
-    while (i < nackSize && numOfNackFields < maxNackFields) {
-      uint16_t nack = nackList[i++];
-      uint16_t bitmask = 0;
-      while (i < nackSize) {
-        int shift = static_cast<uint16_t>(nackList[i] - nack) - 1;
-        if (shift >= 0 && shift <= 15) {
-          bitmask |= (1 << shift);
-          ++i;
-        } else {
-          break;
-        }
+  if (lengthOfBoundingSet > 0) {
+    for (int32_t i = 0; i < lengthOfBoundingSet; i++) {
+      if (candidateSet->Tmmbr(i) == tmmbr_send_ &&
+          candidateSet->PacketOH(i) == packet_oh_send_) {
+        // Do not send the same tuple.
+        return nullptr;
       }
-      // Write the sequence number and the bitmask to the packet.
-      assert(pos + 4 < IP_PACKET_SIZE);
-      ByteWriter<uint16_t>::WriteBigEndian(rtcpbuffer + pos, nack);
-      pos += 2;
-      ByteWriter<uint16_t>::WriteBigEndian(rtcpbuffer + pos, bitmask);
-      pos += 2;
-      numOfNackFields++;
     }
-    rtcpbuffer[nackSizePos] = static_cast<uint8_t>(2 + numOfNackFields);
+    if (!tmmbrOwner) {
+      // use received bounding set as candidate set
+      // add current tuple
+      candidateSet->SetEntry(lengthOfBoundingSet, tmmbr_send_, packet_oh_send_,
+                             ssrc_);
+      int numCandidates = lengthOfBoundingSet + 1;
 
-    if (i != nackSize) {
-      LOG(LS_WARNING) << "Nack list too large for one packet.";
+      // find bounding set
+      TMMBRSet* boundingSet = nullptr;
+      int numBoundingSet = tmmbr_help_.FindTMMBRBoundingSet(boundingSet);
+      if (numBoundingSet > 0 || numBoundingSet <= numCandidates)
+        tmmbrOwner = tmmbr_help_.IsOwner(ssrc_, numBoundingSet);
+      if (!tmmbrOwner) {
+        // Did not enter bounding set, no meaning to send this request.
+        return nullptr;
+      }
     }
+  }
 
-    // Report stats.
-    NACKStringBuilder stringBuilder;
-    for (int idx = 0; idx < i; ++idx) {
-      stringBuilder.PushNACK(nackList[idx]);
-      nack_stats_.ReportRequest(nackList[idx]);
-    }
-    *nackString = stringBuilder.GetResult();
-    packet_type_counter_.nack_requests = nack_stats_.requests();
-    packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
-    return 0;
+  if (!tmmbr_send_)
+    return nullptr;
+
+  rtcp::Tmmbr* tmmbr = new rtcp::Tmmbr();
+  tmmbr->From(ssrc_);
+  tmmbr->To(remote_ssrc_);
+  tmmbr->WithBitrateKbps(tmmbr_send_);
+  tmmbr->WithOverhead(packet_oh_send_);
+
+  return rtc::scoped_ptr<rtcp::Tmmbr>(tmmbr);
 }
 
-int32_t RTCPSender::BuildBYE(uint8_t* rtcpbuffer, int& pos) {
-  // sanity
-  if (pos + 8 >= IP_PACKET_SIZE) {
-    return -2;
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBN(
+    const RtcpContext& ctx) {
+  TMMBRSet* boundingSet = tmmbr_help_.BoundingSetToSend();
+  if (boundingSet == nullptr)
+    return nullptr;
+
+  rtcp::Tmmbn* tmmbn = new rtcp::Tmmbn();
+  tmmbn->From(ssrc_);
+  for (uint32_t i = 0; i < boundingSet->lengthOfSet(); i++) {
+    if (boundingSet->Tmmbr(i) > 0) {
+      tmmbn->WithTmmbr(boundingSet->Ssrc(i), boundingSet->Tmmbr(i),
+                       boundingSet->PacketOH(i));
+    }
   }
 
-  // Add a bye packet
-  // Number of SSRC + CSRCs.
-  rtcpbuffer[pos++] = (uint8_t)0x80 + 1 + csrcs_.size();
-  rtcpbuffer[pos++] = (uint8_t)203;
-
-  // length
-  rtcpbuffer[pos++] = (uint8_t)0;
-  rtcpbuffer[pos++] = (uint8_t)(1 + csrcs_.size());
-
-  // Add our own SSRC
-  ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-  pos += 4;
-
-  // add CSRCs
-  for (size_t i = 0; i < csrcs_.size(); i++) {
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, csrcs_[i]);
-    pos += 4;
-  }
-
-  return 0;
+  return rtc::scoped_ptr<rtcp::Tmmbn>(tmmbn);
 }
 
-int32_t RTCPSender::BuildReceiverReferenceTime(uint8_t* buffer,
-                                               int& pos,
-                                               uint32_t ntp_sec,
-                                               uint32_t ntp_frac) {
-  const int kRrTimeBlockLength = 20;
-  if (pos + kRrTimeBlockLength >= IP_PACKET_SIZE) {
-    return -2;
-  }
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildAPP(const RtcpContext& ctx) {
+  rtcp::App* app = new rtcp::App();
+  app->From(ssrc_);
+  app->WithSubType(app_sub_type_);
+  app->WithName(app_name_);
+  app->WithData(app_data_.get(), app_length_);
 
-  if (last_xr_rr_.size() >= RTCP_NUMBER_OF_SR) {
+  return rtc::scoped_ptr<rtcp::App>(app);
+}
+
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildNACK(
+    const RtcpContext& ctx) {
+  rtcp::Nack* nack = new rtcp::Nack();
+  nack->From(ssrc_);
+  nack->To(remote_ssrc_);
+  nack->WithList(ctx.nack_list_, ctx.nack_size_);
+
+  // Report stats.
+  NACKStringBuilder stringBuilder;
+  for (int idx = 0; idx < ctx.nack_size_; ++idx) {
+    stringBuilder.PushNACK(ctx.nack_list_[idx]);
+    nack_stats_.ReportRequest(ctx.nack_list_[idx]);
+  }
+  packet_type_counter_.nack_requests = nack_stats_.requests();
+  packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
+
+  TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                       "RTCPSender::NACK", "nacks",
+                       TRACE_STR_COPY(stringBuilder.GetResult().c_str()));
+  ++packet_type_counter_.nack_packets;
+  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_NACKCount",
+                    ssrc_, packet_type_counter_.nack_packets);
+
+  return rtc::scoped_ptr<rtcp::Nack>(nack);
+}
+
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildBYE(const RtcpContext& ctx) {
+  rtcp::Bye* bye = new rtcp::Bye();
+  bye->From(ssrc_);
+  for (uint32_t csrc : csrcs_)
+    bye->WithCsrc(csrc);
+
+  return rtc::scoped_ptr<rtcp::Bye>(bye);
+}
+
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildReceiverReferenceTime(
+    const RtcpContext& ctx) {
+  if (last_xr_rr_.size() >= RTCP_NUMBER_OF_SR)
     last_xr_rr_.erase(last_xr_rr_.begin());
-  }
   last_xr_rr_.insert(std::pair<uint32_t, int64_t>(
-      RTCPUtility::MidNtp(ntp_sec, ntp_frac),
-      Clock::NtpToMs(ntp_sec, ntp_frac)));
+      RTCPUtility::MidNtp(ctx.ntp_sec_, ctx.ntp_frac_),
+      Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_)));
 
-  // Add XR header.
-  buffer[pos++] = 0x80;
-  buffer[pos++] = 207;
-  buffer[pos++] = 0;  // XR packet length.
-  buffer[pos++] = 4;  // XR packet length.
+  rtcp::Xr* xr = new rtcp::Xr();
+  xr->From(ssrc_);
 
-  // Add our own SSRC.
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, _SSRC);
-  pos += 4;
+  rtcp::Rrtr rrtr;
+  rrtr.WithNtp(NtpTime(ctx.ntp_sec_, ctx.ntp_frac_));
 
-  //    0                   1                   2                   3
-  //    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-  //   |     BT=4      |   reserved    |       block length = 2        |
-  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-  //   |              NTP timestamp, most significant word             |
-  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-  //   |             NTP timestamp, least significant word             |
-  //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  xr->WithRrtr(&rrtr);
 
-  // Add Receiver Reference Time Report block.
-  buffer[pos++] = 4;  // BT.
-  buffer[pos++] = 0;  // Reserved.
-  buffer[pos++] = 0;  // Block length.
-  buffer[pos++] = 2;  // Block length.
+  // TODO(sprang): Merge XR report sending to contain all of RRTR, DLRR, VOIP?
 
-  // NTP timestamp.
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, ntp_sec);
-  pos += 4;
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, ntp_frac);
-  pos += 4;
-
-  return 0;
+  return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
-int32_t RTCPSender::BuildDlrr(uint8_t* buffer,
-                              int& pos,
-                              const RtcpReceiveTimeInfo& info) {
-  const int kDlrrBlockLength = 24;
-  if (pos + kDlrrBlockLength >= IP_PACKET_SIZE) {
-    return -2;
-  }
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildDlrr(
+    const RtcpContext& ctx) {
+  rtcp::Xr* xr = new rtcp::Xr();
+  xr->From(ssrc_);
 
-  // Add XR header.
-  buffer[pos++] = 0x80;
-  buffer[pos++] = 207;
-  buffer[pos++] = 0;  // XR packet length.
-  buffer[pos++] = 5;  // XR packet length.
+  rtcp::Dlrr dlrr;
+  const RtcpReceiveTimeInfo& info = ctx.feedback_state_.last_xr_rr;
+  dlrr.WithDlrrItem(info.sourceSSRC, info.lastRR, info.delaySinceLastRR);
 
-  // Add our own SSRC.
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, _SSRC);
-  pos += 4;
+  xr->WithDlrr(&dlrr);
 
-  //   0                   1                   2                   3
-  //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-  //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-  //  |     BT=5      |   reserved    |         block length          |
-  //  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-  //  |                 SSRC_1 (SSRC of first receiver)               | sub-
-  //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-  //  |                         last RR (LRR)                         |   1
-  //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-  //  |                   delay since last RR (DLRR)                  |
-  //  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-  //  |                 SSRC_2 (SSRC of second receiver)              | sub-
-  //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
-  //  :                               ...                             :   2
-
-  // Add DLRR sub block.
-  buffer[pos++] = 5;  // BT.
-  buffer[pos++] = 0;  // Reserved.
-  buffer[pos++] = 0;  // Block length.
-  buffer[pos++] = 3;  // Block length.
-
-  // NTP timestamp.
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, info.sourceSSRC);
-  pos += 4;
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, info.lastRR);
-  pos += 4;
-  ByteWriter<uint32_t>::WriteBigEndian(buffer + pos, info.delaySinceLastRR);
-  pos += 4;
-
-  return 0;
+  return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
-int32_t
-RTCPSender::BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos)
-{
-    // sanity
-    if(pos + 44 >= IP_PACKET_SIZE)
-    {
-        return -2;
-    }
+// TODO(sprang): Add a unit test for this, or remove if the code isn't used.
+rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildVoIPMetric(
+    const RtcpContext& context) {
+  rtcp::Xr* xr = new rtcp::Xr();
+  xr->From(ssrc_);
 
-    // Add XR header
-    rtcpbuffer[pos++]=(uint8_t)0x80;
-    rtcpbuffer[pos++]=(uint8_t)207;
+  rtcp::VoipMetric voip;
+  voip.To(remote_ssrc_);
+  voip.WithVoipMetric(xr_voip_metric_);
 
-    uint32_t XRLengthPos = pos;
+  xr->WithVoipMetric(&voip);
 
-    // handle length later on
-    pos++;
-    pos++;
-
-    // Add our own SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _SSRC);
-    pos += 4;
-
-    // Add a VoIP metrics block
-    rtcpbuffer[pos++]=7;
-    rtcpbuffer[pos++]=0;
-    rtcpbuffer[pos++]=0;
-    rtcpbuffer[pos++]=8;
-
-    // Add the remote SSRC
-    ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + pos, _remoteSSRC);
-    pos += 4;
-
-    rtcpbuffer[pos++] = _xrVoIPMetric.lossRate;
-    rtcpbuffer[pos++] = _xrVoIPMetric.discardRate;
-    rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity;
-    rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity;
-
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.burstDuration >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.burstDuration);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.gapDuration >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.gapDuration);
-
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.roundTripDelay >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.roundTripDelay);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.endSystemDelay >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.endSystemDelay);
-
-    rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel;
-    rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel;
-    rtcpbuffer[pos++] = _xrVoIPMetric.RERL;
-    rtcpbuffer[pos++] = _xrVoIPMetric.Gmin;
-
-    rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor;
-    rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor;
-    rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ;
-    rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ;
-
-    rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig;
-    rtcpbuffer[pos++] = 0; // reserved
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBnominal >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBnominal);
-
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBmax >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBmax);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBabsMax >> 8);
-    rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBabsMax);
-
-    rtcpbuffer[XRLengthPos]=(uint8_t)(0);
-    rtcpbuffer[XRLengthPos+1]=(uint8_t)(10);
-    return 0;
+  return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
 int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
-                             uint32_t packetTypeFlags,
-                             int32_t nackSize,
-                             const uint16_t* nackList,
+                             RTCPPacketType packetType,
+                             int32_t nack_size,
+                             const uint16_t* nack_list,
                              bool repeat,
                              uint64_t pictureID) {
-  {
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    if(_method == kRtcpOff)
-    {
-        LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
-        return -1;
-    }
-  }
-  uint8_t rtcp_buffer[IP_PACKET_SIZE];
-  int rtcp_length = PrepareRTCP(feedback_state,
-                                packetTypeFlags,
-                                nackSize,
-                                nackList,
-                                repeat,
-                                pictureID,
-                                rtcp_buffer,
-                                IP_PACKET_SIZE);
-  if (rtcp_length < 0) {
-    return -1;
-  }
-  // Sanity don't send empty packets.
-  if (rtcp_length == 0)
-  {
-      return -1;
-  }
-  return SendToNetwork(rtcp_buffer, static_cast<size_t>(rtcp_length));
+  return SendCompoundRTCP(
+      feedback_state, std::set<RTCPPacketType>(&packetType, &packetType + 1),
+      nack_size, nack_list, repeat, pictureID);
 }
 
-int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
-                            uint32_t packetTypeFlags,
-                            int32_t nackSize,
-                            const uint16_t* nackList,
-                            bool repeat,
-                            uint64_t pictureID,
-                            uint8_t* rtcp_buffer,
-                            int buffer_size) {
-  uint32_t rtcpPacketTypeFlags = packetTypeFlags;
-  // Collect the received information.
-  uint32_t NTPsec = 0;
-  uint32_t NTPfrac = 0;
-  uint32_t jitterTransmissionOffset = 0;
-  int position = 0;
+int32_t RTCPSender::SendCompoundRTCP(
+    const FeedbackState& feedback_state,
+    const std::set<RTCPPacketType>& packet_types,
+    int32_t nack_size,
+    const uint16_t* nack_list,
+    bool repeat,
+    uint64_t pictureID) {
+  PacketContainer container(transport_);
+  {
+    CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+    if (method_ == RtcpMode::kOff) {
+      LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
+      return -1;
+    }
 
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+    // We need to send our NTP even if we haven't received any reports.
+    uint32_t ntp_sec;
+    uint32_t ntp_frac;
+    clock_->CurrentNtp(ntp_sec, ntp_frac);
+    RtcpContext context(feedback_state, nack_size, nack_list, repeat, pictureID,
+                        ntp_sec, ntp_frac, &container);
 
-  if (packet_type_counter_.first_packet_time_ms == -1) {
-    packet_type_counter_.first_packet_time_ms = _clock->TimeInMilliseconds();
-  }
+    PrepareReport(packet_types, feedback_state);
 
-  if(_TMMBR )  // Attach TMMBR to send and receive reports.
-  {
-      rtcpPacketTypeFlags |= kRtcpTmmbr;
-  }
-  if(_appSend)
-  {
-      rtcpPacketTypeFlags |= kRtcpApp;
-      _appSend = false;
-  }
-  if(_REMB && _sendREMB)
-  {
-      // Always attach REMB to SR if that is configured. Note that REMB is
-      // only sent on one of the RTP modules in the REMB group.
-      rtcpPacketTypeFlags |= kRtcpRemb;
-  }
-  if(_xrSendVoIPMetric)
-  {
-      rtcpPacketTypeFlags |= kRtcpXrVoipMetric;
-      _xrSendVoIPMetric = false;
-  }
-  if(_sendTMMBN)  // Set when having received a TMMBR.
-  {
-      rtcpPacketTypeFlags |= kRtcpTmmbn;
-      _sendTMMBN = false;
-  }
-  if (rtcpPacketTypeFlags & kRtcpReport)
-  {
-      if (xrSendReceiverReferenceTimeEnabled_ && !_sending)
-      {
-          rtcpPacketTypeFlags |= kRtcpXrReceiverReferenceTime;
+    auto it = report_flags_.begin();
+    while (it != report_flags_.end()) {
+      auto builder_it = builders_.find(it->type);
+      RTC_DCHECK(builder_it != builders_.end());
+      if (it->is_volatile) {
+        report_flags_.erase(it++);
+      } else {
+        ++it;
       }
-      if (feedback_state.has_last_xr_rr)
-      {
-          rtcpPacketTypeFlags |= kRtcpXrDlrrReportBlock;
-      }
-  }
-  if(_method == kRtcpCompound)
-  {
-      if(_sending)
-      {
-          rtcpPacketTypeFlags |= kRtcpSr;
-      } else
-      {
-          rtcpPacketTypeFlags |= kRtcpRr;
-      }
-  } else if(_method == kRtcpNonCompound)
-  {
-      if(rtcpPacketTypeFlags & kRtcpReport)
-      {
-          if(_sending)
-          {
-              rtcpPacketTypeFlags |= kRtcpSr;
-          } else
-          {
-              rtcpPacketTypeFlags |= kRtcpRr;
-          }
-      }
-  }
-  if( rtcpPacketTypeFlags & kRtcpRr ||
-      rtcpPacketTypeFlags & kRtcpSr)
-  {
-      // generate next time to send a RTCP report
-      // seeded from RTP constructor
-      int32_t random = rand() % 1000;
-      int32_t timeToNext = RTCP_INTERVAL_AUDIO_MS;
 
-      if(_audio)
-      {
-          timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) +
-              (RTCP_INTERVAL_AUDIO_MS*random/1000);
-      }else
-      {
-          uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
-          if(_sending)
-          {
-            // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
-            uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
-            if (send_bitrate_kbit != 0)
-              minIntervalMs = 360000 / send_bitrate_kbit;
-          }
-          if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
-          {
-              minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
-          }
-          timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000);
-      }
-      _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + timeToNext;
+      BuilderFunc func = builder_it->second;
+      rtc::scoped_ptr<rtcp::RtcpPacket> packet = (this->*func)(context);
+      if (packet.get() == nullptr)
+        return -1;
+      container.Append(packet.release());
+    }
+
+    if (packet_type_counter_observer_ != nullptr) {
+      packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
+          remote_ssrc_, packet_type_counter_);
+    }
+
+    RTC_DCHECK(AllVolatileFlagsConsumed());
   }
 
-  // If the data does not fit in the packet we fill it as much as possible.
-  int32_t buildVal = 0;
+  size_t bytes_sent = container.SendPackets();
+  return bytes_sent == 0 ? -1 : 0;
+}
+
+void RTCPSender::PrepareReport(const std::set<RTCPPacketType>& packetTypes,
+                               const FeedbackState& feedback_state) {
+  // Add all flags as volatile. Non volatile entries will not be overwritten
+  // and all new volatile flags added will be consumed by the end of this call.
+  SetFlags(packetTypes, true);
+
+  if (packet_type_counter_.first_packet_time_ms == -1)
+    packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();
+
+  bool generate_report;
+  if (IsFlagPresent(kRtcpSr) || IsFlagPresent(kRtcpRr)) {
+    // Report type already explicitly set, don't automatically populate.
+    generate_report = true;
+    RTC_DCHECK(ConsumeFlag(kRtcpReport) == false);
+  } else {
+    generate_report =
+        (ConsumeFlag(kRtcpReport) && method_ == RtcpMode::kReducedSize) ||
+        method_ == RtcpMode::kCompound;
+    if (generate_report)
+      SetFlag(sending_ ? kRtcpSr : kRtcpRr, true);
+  }
+
+  if (IsFlagPresent(kRtcpSr) || (IsFlagPresent(kRtcpRr) && !cname_.empty()))
+    SetFlag(kRtcpSdes, true);
+
+  if (generate_report) {
+    if (!sending_ && xr_send_receiver_reference_time_enabled_)
+      SetFlag(kRtcpXrReceiverReferenceTime, true);
+    if (feedback_state.has_last_xr_rr)
+      SetFlag(kRtcpXrDlrrReportBlock, true);
+
+    // generate next time to send an RTCP report
+    uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
+
+    if (!audio_) {
+      if (sending_) {
+        // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
+        uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
+        if (send_bitrate_kbit != 0)
+          minIntervalMs = 360000 / send_bitrate_kbit;
+      }
+      if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
+        minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
+    }
+    // The interval between RTCP packets is varied randomly over the
+    // range [1/2,3/2] times the calculated interval.
+    uint32_t timeToNext =
+        random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
+    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
 
-  // We need to send our NTP even if we haven't received any reports.
-  _clock->CurrentNtp(NTPsec, NTPfrac);
-  if (ShouldSendReportBlocks(rtcpPacketTypeFlags)) {
     StatisticianMap statisticians =
         receive_statistics_->GetActiveStatisticians();
-    if (!statisticians.empty()) {
-      StatisticianMap::const_iterator it;
-      int i;
-      for (it = statisticians.begin(), i = 0; it != statisticians.end();
-           ++it, ++i) {
-        RTCPReportBlock report_block;
-        if (PrepareReport(
-                feedback_state, it->second, &report_block, &NTPsec, &NTPfrac))
-          AddReportBlock(it->first, &internal_report_blocks_, &report_block);
-      }
-      if (_IJ && !statisticians.empty()) {
-        rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
-      }
+    RTC_DCHECK(report_blocks_.empty());
+    for (auto& it : statisticians) {
+      AddReportBlock(feedback_state, it.first, it.second);
     }
   }
-
-  if(rtcpPacketTypeFlags & kRtcpSr)
-  {
-    buildVal = BuildSR(feedback_state, rtcp_buffer, position, NTPsec, NTPfrac);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-      buildVal = BuildSDEC(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }else if(rtcpPacketTypeFlags & kRtcpRr)
-  {
-      buildVal = BuildRR(rtcp_buffer, position, NTPsec, NTPfrac);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-      // only of set
-      if(_CNAME[0] != 0)
-      {
-          buildVal = BuildSDEC(rtcp_buffer, position);
-          if (buildVal == -1) {
-            return -1;
-          }
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpTransmissionTimeOffset)
-  {
-      // If present, this RTCP packet must be placed after a
-      // receiver report.
-      buildVal = BuildExtendedJitterReport(rtcp_buffer,
-                                           position,
-                                           jitterTransmissionOffset);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpPli)
-  {
-      buildVal = BuildPLI(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-      TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                           "RTCPSender::PLI");
-      ++packet_type_counter_.pli_packets;
-      TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                        "RTCP_PLICount", _SSRC,
-                        packet_type_counter_.pli_packets);
-  }
-  if(rtcpPacketTypeFlags & kRtcpFir)
-  {
-      buildVal = BuildFIR(rtcp_buffer, position, repeat);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-      TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                           "RTCPSender::FIR");
-      ++packet_type_counter_.fir_packets;
-      TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                        "RTCP_FIRCount", _SSRC,
-                        packet_type_counter_.fir_packets);
-  }
-  if(rtcpPacketTypeFlags & kRtcpSli)
-  {
-      buildVal = BuildSLI(rtcp_buffer, position, (uint8_t)pictureID);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpRpsi)
-  {
-      const int8_t payloadType = feedback_state.send_payload_type;
-      if (payloadType == -1) {
-        return -1;
-      }
-      buildVal = BuildRPSI(rtcp_buffer, position, pictureID,
-                           (uint8_t)payloadType);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpRemb)
-  {
-      buildVal = BuildREMB(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-      TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                           "RTCPSender::REMB");
-  }
-  if(rtcpPacketTypeFlags & kRtcpBye)
-  {
-      buildVal = BuildBYE(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpApp)
-  {
-      buildVal = BuildAPP(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpTmmbr)
-  {
-      buildVal = BuildTMMBR(feedback_state.module, rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpTmmbn)
-  {
-      buildVal = BuildTMMBN(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if(rtcpPacketTypeFlags & kRtcpNack)
-  {
-      std::string nackString;
-      buildVal = BuildNACK(rtcp_buffer, position, nackSize, nackList,
-                           &nackString);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-      TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                           "RTCPSender::NACK", "nacks",
-                           TRACE_STR_COPY(nackString.c_str()));
-      ++packet_type_counter_.nack_packets;
-      TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                        "RTCP_NACKCount", _SSRC,
-                        packet_type_counter_.nack_packets);
-  }
-  if(rtcpPacketTypeFlags & kRtcpXrVoipMetric)
-  {
-      buildVal = BuildVoIPMetric(rtcp_buffer, position);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if (rtcpPacketTypeFlags & kRtcpXrReceiverReferenceTime)
-  {
-      buildVal = BuildReceiverReferenceTime(rtcp_buffer,
-                                            position,
-                                            NTPsec,
-                                            NTPfrac);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-  if (rtcpPacketTypeFlags & kRtcpXrDlrrReportBlock)
-  {
-      buildVal = BuildDlrr(rtcp_buffer, position, feedback_state.last_xr_rr);
-      if (buildVal == -1) {
-        return -1;
-      } else if (buildVal == -2) {
-        return position;
-      }
-  }
-
-  if (packet_type_counter_observer_ != NULL) {
-    packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
-        _remoteSSRC, packet_type_counter_);
-  }
-
-  return position;
 }
 
-bool RTCPSender::ShouldSendReportBlocks(uint32_t rtcp_packet_type) const {
-  return Status() == kRtcpCompound ||
-      (rtcp_packet_type & kRtcpReport) ||
-      (rtcp_packet_type & kRtcpSr) ||
-      (rtcp_packet_type & kRtcpRr);
-}
-
-bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
-                               StreamStatistician* statistician,
-                               RTCPReportBlock* report_block,
-                               uint32_t* ntp_secs, uint32_t* ntp_frac) {
+bool RTCPSender::AddReportBlock(const FeedbackState& feedback_state,
+                                uint32_t ssrc,
+                                StreamStatistician* statistician) {
   // Do we have receive statistics to send?
   RtcpStatistics stats;
   if (!statistician->GetStatistics(&stats, true))
     return false;
-  report_block->fractionLost = stats.fraction_lost;
-  report_block->cumulativeLost = stats.cumulative_lost;
-  report_block->extendedHighSeqNum =
-      stats.extended_max_sequence_number;
-  report_block->jitter = stats.jitter;
 
-  // get our NTP as late as possible to avoid a race
-  _clock->CurrentNtp(*ntp_secs, *ntp_frac);
+  if (report_blocks_.size() >= RTCP_MAX_REPORT_BLOCKS) {
+    LOG(LS_WARNING) << "Too many report blocks.";
+    return false;
+  }
+  RTC_DCHECK(report_blocks_.find(ssrc) == report_blocks_.end());
+  rtcp::ReportBlock* block = &report_blocks_[ssrc];
+  block->To(ssrc);
+  block->WithFractionLost(stats.fraction_lost);
+  if (!block->WithCumulativeLost(stats.cumulative_lost)) {
+    report_blocks_.erase(ssrc);
+    LOG(LS_WARNING) << "Cumulative lost is oversized.";
+    return false;
+  }
+  block->WithExtHighestSeqNum(stats.extended_max_sequence_number);
+  block->WithJitter(stats.jitter);
+  block->WithLastSr(feedback_state.remote_sr);
 
-  // Delay since last received report
-  uint32_t delaySinceLastReceivedSR = 0;
+  // TODO(sprang): Do we really need separate time stamps for each report?
+  // Get our NTP as late as possible to avoid a race.
+  uint32_t ntp_secs;
+  uint32_t ntp_frac;
+  clock_->CurrentNtp(ntp_secs, ntp_frac);
+
+  // Delay since last received report.
   if ((feedback_state.last_rr_ntp_secs != 0) ||
       (feedback_state.last_rr_ntp_frac != 0)) {
-    // get the 16 lowest bits of seconds and the 16 higest bits of fractions
-    uint32_t now=*ntp_secs&0x0000FFFF;
-    now <<=16;
-    now += (*ntp_frac&0xffff0000)>>16;
+    // Get the 16 lowest bits of seconds and the 16 highest bits of fractions.
+    uint32_t now = ntp_secs & 0x0000FFFF;
+    now <<= 16;
+    now += (ntp_frac & 0xffff0000) >> 16;
 
-    uint32_t receiveTime = feedback_state.last_rr_ntp_secs&0x0000FFFF;
-    receiveTime <<=16;
-    receiveTime += (feedback_state.last_rr_ntp_frac&0xffff0000)>>16;
+    uint32_t receiveTime = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
+    receiveTime <<= 16;
+    receiveTime += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;
 
-    delaySinceLastReceivedSR = now-receiveTime;
+    block->WithDelayLastSr(now - receiveTime);
   }
-  report_block->delaySinceLastSR = delaySinceLastReceivedSR;
-  report_block->lastSR = feedback_state.remote_sr;
   return true;
 }
 
-int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
-    CriticalSectionScoped lock(_criticalSectionTransport);
-    if(_cbTransport)
-    {
-        if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0)
-        {
-            return 0;
-        }
-    }
-    return -1;
-}
-
 void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   assert(csrcs.size() <= kRtpCsrcSize);
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   csrcs_ = csrcs;
 }
 
@@ -1978,122 +977,100 @@ int32_t RTCPSender::SetApplicationSpecificData(uint8_t subType,
                                                uint32_t name,
                                                const uint8_t* data,
                                                uint16_t length) {
-    if(length %4 != 0)
-    {
-        LOG(LS_ERROR) << "Failed to SetApplicationSpecificData.";
-        return -1;
-    }
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  if (length % 4 != 0) {
+    LOG(LS_ERROR) << "Failed to SetApplicationSpecificData.";
+    return -1;
+  }
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-    if(_appData)
-    {
-        delete [] _appData;
-    }
-
-    _appSend = true;
-    _appSubType = subType;
-    _appName = name;
-    _appData = new uint8_t[length];
-    _appLength = length;
-    memcpy(_appData, data, length);
-    return 0;
+  SetFlag(kRtcpApp, true);
+  app_sub_type_ = subType;
+  app_name_ = name;
+  app_data_.reset(new uint8_t[length]);
+  app_length_ = length;
+  memcpy(app_data_.get(), data, length);
+  return 0;
 }
 
-int32_t
-RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric)
-{
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric));
+int32_t RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) {
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  memcpy(&xr_voip_metric_, VoIPMetric, sizeof(RTCPVoIPMetric));
 
-    _xrSendVoIPMetric = true;
-    return 0;
+  SetFlag(kRtcpXrVoipMetric, true);
+  return 0;
 }
 
 void RTCPSender::SendRtcpXrReceiverReferenceTime(bool enable) {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  xrSendReceiverReferenceTimeEnabled_ = enable;
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  xr_send_receiver_reference_time_enabled_ = enable;
 }
 
 bool RTCPSender::RtcpXrReceiverReferenceTime() const {
-  CriticalSectionScoped lock(_criticalSectionRTCPSender);
-  return xrSendReceiverReferenceTimeEnabled_;
-}
-
-// called under critsect _criticalSectionRTCPSender
-int32_t RTCPSender::WriteAllReportBlocksToBuffer(uint8_t* rtcpbuffer,
-                                                 int pos,
-                                                 uint8_t& numberOfReportBlocks,
-                                                 uint32_t NTPsec,
-                                                 uint32_t NTPfrac) {
-  numberOfReportBlocks = external_report_blocks_.size();
-  numberOfReportBlocks += internal_report_blocks_.size();
-  if ((pos + numberOfReportBlocks * 24) >= IP_PACKET_SIZE) {
-    LOG(LS_WARNING) << "Can't fit all report blocks.";
-    return -1;
-  }
-  pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, internal_report_blocks_);
-  while (!internal_report_blocks_.empty()) {
-    delete internal_report_blocks_.begin()->second;
-    internal_report_blocks_.erase(internal_report_blocks_.begin());
-  }
-  pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, external_report_blocks_);
-  return pos;
-}
-
-int32_t RTCPSender::WriteReportBlocksToBuffer(
-    uint8_t* rtcpbuffer,
-    int32_t position,
-    const std::map<uint32_t, RTCPReportBlock*>& report_blocks) {
-  std::map<uint32_t, RTCPReportBlock*>::const_iterator it =
-      report_blocks.begin();
-  for (; it != report_blocks.end(); it++) {
-    uint32_t remoteSSRC = it->first;
-    RTCPReportBlock* reportBlock = it->second;
-    if (reportBlock) {
-      // Remote SSRC
-      ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + position, remoteSSRC);
-      position += 4;
-
-      // fraction lost
-      rtcpbuffer[position++] = reportBlock->fractionLost;
-
-      // cumulative loss
-      ByteWriter<uint32_t, 3>::WriteBigEndian(rtcpbuffer + position,
-                                              reportBlock->cumulativeLost);
-      position += 3;
-
-      // extended highest seq_no, contain the highest sequence number received
-      ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + position,
-                                           reportBlock->extendedHighSeqNum);
-      position += 4;
-
-      // Jitter
-      ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + position,
-                                           reportBlock->jitter);
-      position += 4;
-
-      ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + position,
-                                           reportBlock->lastSR);
-      position += 4;
-
-      ByteWriter<uint32_t>::WriteBigEndian(rtcpbuffer + position,
-                                           reportBlock->delaySinceLastSR);
-      position += 4;
-    }
-  }
-  return position;
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
+  return xr_send_receiver_reference_time_enabled_;
 }
 
 // no callbacks allowed inside this function
 int32_t RTCPSender::SetTMMBN(const TMMBRSet* boundingSet,
                              uint32_t maxBitrateKbit) {
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
+  CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-    if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit))
-    {
-        _sendTMMBN = true;
-        return 0;
-    }
-    return -1;
+  if (0 == tmmbr_help_.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit)) {
+    SetFlag(kRtcpTmmbn, true);
+    return 0;
+  }
+  return -1;
 }
+
+void RTCPSender::SetFlag(RTCPPacketType type, bool is_volatile) {
+  report_flags_.insert(ReportFlag(type, is_volatile));
+}
+
+void RTCPSender::SetFlags(const std::set<RTCPPacketType>& types,
+                          bool is_volatile) {
+  for (RTCPPacketType type : types)
+    SetFlag(type, is_volatile);
+}
+
+bool RTCPSender::IsFlagPresent(RTCPPacketType type) const {
+  return report_flags_.find(ReportFlag(type, false)) != report_flags_.end();
+}
+
+bool RTCPSender::ConsumeFlag(RTCPPacketType type, bool forced) {
+  auto it = report_flags_.find(ReportFlag(type, false));
+  if (it == report_flags_.end())
+    return false;
+  if (it->is_volatile || forced)
+    report_flags_.erase((it));
+  return true;
+}
+
+bool RTCPSender::AllVolatileFlagsConsumed() const {
+  for (const ReportFlag& flag : report_flags_) {
+    if (flag.is_volatile)
+      return false;
+  }
+  return true;
+}
+
+bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
+  class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
+   public:
+    explicit Sender(Transport* transport)
+        : transport_(transport), send_failure_(false) {}
+
+    void OnPacketReady(uint8_t* data, size_t length) override {
+      if (!transport_->SendRtcp(data, length))
+        send_failure_ = true;
+    }
+
+    Transport* const transport_;
+    bool send_failure_;
+  } sender(transport_);
+
+  uint8_t buffer[IP_PACKET_SIZE];
+  return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
+         !sender.send_failure_;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index b6099deb8f..8cf4e15429 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -12,18 +12,23 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
 
 #include <map>
+#include <set>
 #include <sstream>
 #include <string>
+#include <vector>
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
+#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -40,319 +45,265 @@ class NACKStringBuilder {
   std::string GetResult();
 
  private:
-  std::ostringstream _stream;
-  int _count;
-  uint16_t _prevNack;
-  bool _consecutive;
+  std::ostringstream stream_;
+  int count_;
+  uint16_t prevNack_;
+  bool consecutive_;
 };
 
 class RTCPSender {
-public:
- struct FeedbackState {
-   FeedbackState();
+ public:
+  struct FeedbackState {
+    FeedbackState();
 
-   uint8_t send_payload_type;
-   uint32_t frequency_hz;
-   uint32_t packets_sent;
-   size_t media_bytes_sent;
-   uint32_t send_bitrate;
+    uint8_t send_payload_type;
+    uint32_t frequency_hz;
+    uint32_t packets_sent;
+    size_t media_bytes_sent;
+    uint32_t send_bitrate;
 
-   uint32_t last_rr_ntp_secs;
-   uint32_t last_rr_ntp_frac;
-   uint32_t remote_sr;
+    uint32_t last_rr_ntp_secs;
+    uint32_t last_rr_ntp_frac;
+    uint32_t remote_sr;
 
-   bool has_last_xr_rr;
-   RtcpReceiveTimeInfo last_xr_rr;
+    bool has_last_xr_rr;
+    RtcpReceiveTimeInfo last_xr_rr;
 
-   // Used when generating TMMBR.
-   ModuleRtpRtcpImpl* module;
- };
- RTCPSender(int32_t id,
-            bool audio,
-            Clock* clock,
-            ReceiveStatistics* receive_statistics,
-            RtcpPacketTypeCounterObserver* packet_type_counter_observer);
-    virtual ~RTCPSender();
+    // Used when generating TMMBR.
+    ModuleRtpRtcpImpl* module;
+  };
 
-    int32_t RegisterSendTransport(Transport* outgoingTransport);
+  RTCPSender(bool audio,
+             Clock* clock,
+             ReceiveStatistics* receive_statistics,
+             RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+             Transport* outgoing_transport);
+  virtual ~RTCPSender();
 
-    RTCPMethod Status() const;
-    void SetRTCPStatus(RTCPMethod method);
+  RtcpMode Status() const;
+  void SetRTCPStatus(RtcpMode method);
 
-    bool Sending() const;
-    int32_t SetSendingStatus(const FeedbackState& feedback_state,
-                             bool enabled);  // combine the functions
+  bool Sending() const;
+  int32_t SetSendingStatus(const FeedbackState& feedback_state,
+                           bool enabled);  // combine the functions
 
-    int32_t SetNackStatus(bool enable);
+  int32_t SetNackStatus(bool enable);
 
-    void SetStartTimestamp(uint32_t start_timestamp);
+  void SetStartTimestamp(uint32_t start_timestamp);
 
-    void SetLastRtpTime(uint32_t rtp_timestamp,
-                        int64_t capture_time_ms);
+  void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
 
-    void SetSSRC(uint32_t ssrc);
+  void SetSSRC(uint32_t ssrc);
 
-    void SetRemoteSSRC(uint32_t ssrc);
-
-    int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]);
-
-    int32_t AddMixedCNAME(uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]);
-
-    int32_t RemoveMixedCNAME(uint32_t SSRC);
-
-    bool GetSendReportMetadata(const uint32_t sendReport,
-                               uint64_t *timeOfSend,
-                               uint32_t *packetCount,
-                               uint64_t *octetCount);
-
-    bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
-
-    bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
-
-    uint32_t LastSendReport(int64_t& lastRTCPTime);
-
-    int32_t SendRTCP(
-        const FeedbackState& feedback_state,
-        uint32_t rtcpPacketTypeFlags,
-        int32_t nackSize = 0,
-        const uint16_t* nackList = 0,
-        bool repeat = false,
-        uint64_t pictureID = 0);
-
-    int32_t AddExternalReportBlock(
-        uint32_t SSRC,
-        const RTCPReportBlock* receiveBlock);
-
-    int32_t RemoveExternalReportBlock(uint32_t SSRC);
-
-    /*
-    *  REMB
-    */
-    bool REMB() const;
-
-    void SetREMBStatus(bool enable);
-
-    void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
-
-    /*
-    *   TMMBR
-    */
-    bool TMMBR() const;
-
-    void SetTMMBRStatus(bool enable);
-
-    int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
-
-    /*
-    *   Extended jitter report
-    */
-    bool IJ() const;
-
-    void SetIJStatus(bool enable);
-
-    /*
-    *
-    */
-
-    int32_t SetApplicationSpecificData(uint8_t subType,
-                                       uint32_t name,
-                                       const uint8_t* data,
-                                       uint16_t length);
-
-    int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
-
-    void SendRtcpXrReceiverReferenceTime(bool enable);
-
-    bool RtcpXrReceiverReferenceTime() const;
-
-    void SetCsrcs(const std::vector<uint32_t>& csrcs);
-
-    void SetTargetBitrate(unsigned int target_bitrate);
-
-private:
- int32_t SendToNetwork(const uint8_t* dataBuffer, size_t length);
-
- int32_t WriteAllReportBlocksToBuffer(uint8_t* rtcpbuffer,
-                                      int pos,
-                                      uint8_t& numberOfReportBlocks,
-                                      uint32_t NTPsec,
-                                      uint32_t NTPfrac)
-     EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-
-    int32_t WriteReportBlocksToBuffer(
-        uint8_t* rtcpbuffer,
-        int32_t position,
-        const std::map<uint32_t, RTCPReportBlock*>& report_blocks);
-
-    int32_t AddReportBlock(
-        uint32_t SSRC,
-        std::map<uint32_t, RTCPReportBlock*>* report_blocks,
-        const RTCPReportBlock* receiveBlock);
-
-    bool PrepareReport(const FeedbackState& feedback_state,
-                       StreamStatistician* statistician,
-                       RTCPReportBlock* report_block,
-                       uint32_t* ntp_secs, uint32_t* ntp_frac);
-
-    int32_t BuildSR(const FeedbackState& feedback_state,
-                    uint8_t* rtcpbuffer,
-                    int& pos,
-                    uint32_t NTPsec,
-                    uint32_t NTPfrac)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-
-    int32_t BuildRR(uint8_t* rtcpbuffer,
-                    int& pos,
-                    uint32_t NTPsec,
-                    uint32_t NTPfrac)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-
-    int PrepareRTCP(
-        const FeedbackState& feedback_state,
-        uint32_t packetTypeFlags,
-        int32_t nackSize,
-        const uint16_t* nackList,
-        bool repeat,
-        uint64_t pictureID,
-        uint8_t* rtcp_buffer,
-        int buffer_size);
-
-    bool ShouldSendReportBlocks(uint32_t rtcp_packet_type) const;
-
-    int32_t BuildExtendedJitterReport(uint8_t* rtcpbuffer,
-                                      int& pos,
-                                      uint32_t jitterTransmissionTimeOffset)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-
-    int32_t BuildSDEC(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildPLI(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildREMB(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildTMMBR(ModuleRtpRtcpImpl* module, uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildTMMBN(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildAPP(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildBYE(uint8_t* rtcpbuffer, int& pos)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildFIR(uint8_t* rtcpbuffer, int& pos, bool repeat)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildSLI(uint8_t* rtcpbuffer, int& pos, uint8_t pictureID)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildRPSI(uint8_t* rtcpbuffer,
-                      int& pos,
-                      uint64_t pictureID,
-                      uint8_t payloadType)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-
-    int32_t BuildNACK(uint8_t* rtcpbuffer,
-                      int& pos,
-                      int32_t nackSize,
-                      const uint16_t* nackList,
-                      std::string* nackString)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildReceiverReferenceTime(uint8_t* buffer,
-                                       int& pos,
-                                       uint32_t ntp_sec,
-                                       uint32_t ntp_frac)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-    int32_t BuildDlrr(uint8_t* buffer,
-                      int& pos,
-                      const RtcpReceiveTimeInfo& info)
-        EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
-
-private:
-    const int32_t _id;
-    const bool _audio;
-    Clock* const _clock;
-    RTCPMethod _method GUARDED_BY(_criticalSectionRTCPSender);
-
-    CriticalSectionWrapper* _criticalSectionTransport;
-    Transport* _cbTransport GUARDED_BY(_criticalSectionTransport);
-
-    CriticalSectionWrapper* _criticalSectionRTCPSender;
-    bool _usingNack GUARDED_BY(_criticalSectionRTCPSender);
-    bool _sending GUARDED_BY(_criticalSectionRTCPSender);
-    bool _sendTMMBN GUARDED_BY(_criticalSectionRTCPSender);
-    bool _REMB GUARDED_BY(_criticalSectionRTCPSender);
-    bool _sendREMB GUARDED_BY(_criticalSectionRTCPSender);
-    bool _TMMBR GUARDED_BY(_criticalSectionRTCPSender);
-    bool _IJ GUARDED_BY(_criticalSectionRTCPSender);
-
-    int64_t _nextTimeToSendRTCP GUARDED_BY(_criticalSectionRTCPSender);
-
-    uint32_t start_timestamp_ GUARDED_BY(_criticalSectionRTCPSender);
-    uint32_t last_rtp_timestamp_ GUARDED_BY(_criticalSectionRTCPSender);
-    int64_t last_frame_capture_time_ms_ GUARDED_BY(_criticalSectionRTCPSender);
-    uint32_t _SSRC GUARDED_BY(_criticalSectionRTCPSender);
-    // SSRC that we receive on our RTP channel
-    uint32_t _remoteSSRC GUARDED_BY(_criticalSectionRTCPSender);
-    char _CNAME[RTCP_CNAME_SIZE] GUARDED_BY(_criticalSectionRTCPSender);
-
-    ReceiveStatistics* receive_statistics_
-        GUARDED_BY(_criticalSectionRTCPSender);
-    std::map<uint32_t, RTCPReportBlock*> internal_report_blocks_
-        GUARDED_BY(_criticalSectionRTCPSender);
-    std::map<uint32_t, RTCPReportBlock*> external_report_blocks_
-        GUARDED_BY(_criticalSectionRTCPSender);
-    std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs
-        GUARDED_BY(_criticalSectionRTCPSender);
-
-    // Sent
-    uint32_t _lastSendReport[RTCP_NUMBER_OF_SR] GUARDED_BY(
-        _criticalSectionRTCPSender);  // allow packet loss and RTT above 1 sec
-    int64_t _lastRTCPTime[RTCP_NUMBER_OF_SR] GUARDED_BY(
-        _criticalSectionRTCPSender);
-    uint32_t        _lastSRPacketCount[RTCP_NUMBER_OF_SR] GUARDED_BY(
-        _criticalSectionRTCPSender); 
-    uint64_t        _lastSROctetCount[RTCP_NUMBER_OF_SR] GUARDED_BY(
-        _criticalSectionRTCPSender);
-
-    // Sent XR receiver reference time report.
-    // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
-    std::map<uint32_t, int64_t> last_xr_rr_
-        GUARDED_BY(_criticalSectionRTCPSender);
-
-    // send CSRCs
-    std::vector<uint32_t> csrcs_ GUARDED_BY(_criticalSectionRTCPSender);
-
-    // Full intra request
-    uint8_t _sequenceNumberFIR GUARDED_BY(_criticalSectionRTCPSender);
-
-    // REMB
-    uint32_t _rembBitrate GUARDED_BY(_criticalSectionRTCPSender);
-    std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(_criticalSectionRTCPSender);
-
-    TMMBRHelp _tmmbrHelp GUARDED_BY(_criticalSectionRTCPSender);
-    uint32_t _tmmbr_Send GUARDED_BY(_criticalSectionRTCPSender);
-    uint32_t _packetOH_Send GUARDED_BY(_criticalSectionRTCPSender);
-
-    // APP
-    bool _appSend GUARDED_BY(_criticalSectionRTCPSender);
-    uint8_t _appSubType GUARDED_BY(_criticalSectionRTCPSender);
-    uint32_t _appName GUARDED_BY(_criticalSectionRTCPSender);
-    uint8_t* _appData GUARDED_BY(_criticalSectionRTCPSender);
-    uint16_t _appLength GUARDED_BY(_criticalSectionRTCPSender);
-
-    // True if sending of XR Receiver reference time report is enabled.
-    bool xrSendReceiverReferenceTimeEnabled_
-        GUARDED_BY(_criticalSectionRTCPSender);
-
-    // XR VoIP metric
-    bool _xrSendVoIPMetric GUARDED_BY(_criticalSectionRTCPSender);
-    RTCPVoIPMetric _xrVoIPMetric GUARDED_BY(_criticalSectionRTCPSender);
-
-    RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
-    RtcpPacketTypeCounter packet_type_counter_
-        GUARDED_BY(_criticalSectionRTCPSender);
-
-    RTCPUtility::NackStats nack_stats_ GUARDED_BY(_criticalSectionRTCPSender);
+  void SetRemoteSSRC(uint32_t ssrc);
+
+  int32_t SetCNAME(const char* cName);
+
+  int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
+
+  int32_t RemoveMixedCNAME(uint32_t SSRC);
+
+  bool GetSendReportMetadata(const uint32_t sendReport,
+                             uint64_t *timeOfSend,
+                             uint32_t *packetCount,
+                             uint64_t *octetCount);
+
+  bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
+
+  bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
+
+  int32_t SendRTCP(const FeedbackState& feedback_state,
+                   RTCPPacketType packetType,
+                   int32_t nackSize = 0,
+                   const uint16_t* nackList = 0,
+                   bool repeat = false,
+                   uint64_t pictureID = 0);
+
+  int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
+                           const std::set<RTCPPacketType>& packetTypes,
+                           int32_t nackSize = 0,
+                           const uint16_t* nackList = 0,
+                           bool repeat = false,
+                           uint64_t pictureID = 0);
+
+  bool REMB() const;
+
+  void SetREMBStatus(bool enable);
+
+  void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
+
+  bool TMMBR() const;
+
+  void SetTMMBRStatus(bool enable);
+
+  int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
+
+  int32_t SetApplicationSpecificData(uint8_t subType,
+                                     uint32_t name,
+                                     const uint8_t* data,
+                                     uint16_t length);
+  int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
+
+  void SendRtcpXrReceiverReferenceTime(bool enable);
+
+  bool RtcpXrReceiverReferenceTime() const;
+
+  void SetCsrcs(const std::vector<uint32_t>& csrcs);
+
+  void SetTargetBitrate(unsigned int target_bitrate);
+  bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
+
+ private:
+  class RtcpContext;
+
+  // Determine which RTCP messages should be sent and setup flags.
+  void PrepareReport(const std::set<RTCPPacketType>& packetTypes,
+                     const FeedbackState& feedback_state)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+
+  bool AddReportBlock(const FeedbackState& feedback_state,
+                      uint32_t ssrc,
+                      StreamStatistician* statistician)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBN(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildVoIPMetric(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildBYE(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildFIR(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildSLI(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildRPSI(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildNACK(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildReceiverReferenceTime(
+      const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+
+ private:
+  const bool audio_;
+  Clock* const clock_;
+  Random random_ GUARDED_BY(critical_section_rtcp_sender_);
+  RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  Transport* const transport_;
+
+  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
+  bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
+  bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
+  bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
+  int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
+  // SSRC that we receive on our RTP channel
+  uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
+  std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  ReceiveStatistics* receive_statistics_
+      GUARDED_BY(critical_section_rtcp_sender_);
+  std::map<uint32_t, rtcp::ReportBlock> report_blocks_
+      GUARDED_BY(critical_section_rtcp_sender_);
+  std::map<uint32_t, std::string> csrc_cnames_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  // Sent
+  uint32_t last_send_report_[RTCP_NUMBER_OF_SR] GUARDED_BY(
+      critical_section_rtcp_sender_);  // allow packet loss and RTT above 1 sec
+  int64_t last_rtcp_time_[RTCP_NUMBER_OF_SR] GUARDED_BY(
+      critical_section_rtcp_sender_);
+  uint32_t lastSRPacketCount_[RTCP_NUMBER_OF_SR] GUARDED_BY(
+      critical_section_rtcp_sender_); 
+  uint64_t lastSROctetCount_[RTCP_NUMBER_OF_SR] GUARDED_BY(
+      critical_section_rtcp_sender_);
+
+  // Sent XR receiver reference time report.
+  // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
+  std::map<uint32_t, int64_t> last_xr_rr_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  // send CSRCs
+  std::vector<uint32_t> csrcs_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // Full intra request
+  uint8_t sequence_number_fir_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // REMB
+  uint32_t remb_bitrate_ GUARDED_BY(critical_section_rtcp_sender_);
+  std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  TMMBRHelp tmmbr_help_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t tmmbr_send_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t packet_oh_send_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // APP
+  uint8_t app_sub_type_ GUARDED_BY(critical_section_rtcp_sender_);
+  uint32_t app_name_ GUARDED_BY(critical_section_rtcp_sender_);
+  rtc::scoped_ptr<uint8_t[]> app_data_
+      GUARDED_BY(critical_section_rtcp_sender_);
+  uint16_t app_length_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  // True if sending of XR Receiver reference time report is enabled.
+  bool xr_send_receiver_reference_time_enabled_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  // XR VoIP metric
+  RTCPVoIPMetric xr_voip_metric_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
+  RtcpPacketTypeCounter packet_type_counter_
+      GUARDED_BY(critical_section_rtcp_sender_);
+
+  RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  void SetFlag(RTCPPacketType type, bool is_volatile)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  bool IsFlagPresent(RTCPPacketType type) const
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  bool ConsumeFlag(RTCPPacketType type, bool forced = false)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  bool AllVolatileFlagsConsumed() const
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  struct ReportFlag {
+    ReportFlag(RTCPPacketType type, bool is_volatile)
+        : type(type), is_volatile(is_volatile) {}
+    bool operator<(const ReportFlag& flag) const { return type < flag.type; }
+    bool operator==(const ReportFlag& flag) const { return type == flag.type; }
+    const RTCPPacketType type;
+    const bool is_volatile;
+  };
+
+  std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
+
+  typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
+      const RtcpContext&);
+  std::map<RTCPPacketType, BuilderFunc> builders_;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index a35b7c382f..ba42c8dd50 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -17,16 +17,11 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAre;
 
 namespace webrtc {
 
@@ -185,259 +180,466 @@ TEST(NACKStringBuilderTest, TestCase13) {
   EXPECT_EQ(std::string("5-6,9"), builder.GetResult());
 }
 
-void CreateRtpPacket(const bool marker_bit, const uint8_t payload_type,
-    const uint16_t seq_num, const uint32_t timestamp,
-    const uint32_t ssrc, uint8_t* array,
-    size_t* cur_pos) {
-  ASSERT_LE(payload_type, 127);
-  array[(*cur_pos)++] = 0x80;
-  array[(*cur_pos)++] = payload_type | (marker_bit ? 0x80 : 0);
-  array[(*cur_pos)++] = seq_num >> 8;
-  array[(*cur_pos)++] = seq_num & 0xFF;
-  array[(*cur_pos)++] = timestamp >> 24;
-  array[(*cur_pos)++] = (timestamp >> 16) & 0xFF;
-  array[(*cur_pos)++] = (timestamp >> 8) & 0xFF;
-  array[(*cur_pos)++] = timestamp & 0xFF;
-  array[(*cur_pos)++] = ssrc >> 24;
-  array[(*cur_pos)++] = (ssrc >> 16) & 0xFF;
-  array[(*cur_pos)++] = (ssrc >> 8) & 0xFF;
-  array[(*cur_pos)++] = ssrc & 0xFF;
-  // VP8 payload header
-  array[(*cur_pos)++] = 0x90;  // X bit = 1
-  array[(*cur_pos)++] = 0x20;  // T bit = 1
-  array[(*cur_pos)++] = 0x00;  // TID = 0
-  array[(*cur_pos)++] = 0x00;  // Key frame
-  array[(*cur_pos)++] = 0x00;
-  array[(*cur_pos)++] = 0x00;
-  array[(*cur_pos)++] = 0x9d;
-  array[(*cur_pos)++] = 0x01;
-  array[(*cur_pos)++] = 0x2a;
-  array[(*cur_pos)++] = 128;
-  array[(*cur_pos)++] = 0;
-  array[(*cur_pos)++] = 96;
-  array[(*cur_pos)++] = 0;
-}
+class RtcpPacketTypeCounterObserverImpl : public RtcpPacketTypeCounterObserver {
+ public:
+  RtcpPacketTypeCounterObserverImpl() : ssrc_(0) {}
+  virtual ~RtcpPacketTypeCounterObserverImpl() {}
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override {
+    ssrc_ = ssrc;
+    counter_ = packet_counter;
+  }
+  uint32_t ssrc_;
+  RtcpPacketTypeCounter counter_;
+};
 
 class TestTransport : public Transport,
                       public NullRtpData {
  public:
-  TestTransport()
-      : rtcp_receiver_(NULL) {
-  }
-  void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
-    rtcp_receiver_ = rtcp_receiver;
-  }
-  int SendPacket(int /*ch*/, const void* /*data*/, size_t /*len*/) override {
-    return -1;
-  }
+  TestTransport() {}
 
-  int SendRTCPPacket(int /*ch*/,
-                     const void* packet,
-                     size_t packet_len) override {
-    RTCPUtility::RTCPParserV2 rtcpParser((uint8_t*)packet,
-                                         packet_len,
-                                         true); // Allow non-compound RTCP
-
-    EXPECT_TRUE(rtcpParser.IsValid());
-    RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
-    EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
-                                                    &rtcpParser));
-    rtcp_packet_info_.rtcpPacketTypeFlags =
-        rtcpPacketInformation.rtcpPacketTypeFlags;
-    rtcp_packet_info_.remoteSSRC = rtcpPacketInformation.remoteSSRC;
-    rtcp_packet_info_.applicationSubType =
-        rtcpPacketInformation.applicationSubType;
-    rtcp_packet_info_.applicationName = rtcpPacketInformation.applicationName;
-    rtcp_packet_info_.report_blocks = rtcpPacketInformation.report_blocks;
-    rtcp_packet_info_.rtt = rtcpPacketInformation.rtt;
-    rtcp_packet_info_.interArrivalJitter =
-        rtcpPacketInformation.interArrivalJitter;
-    rtcp_packet_info_.sliPictureId = rtcpPacketInformation.sliPictureId;
-    rtcp_packet_info_.rpsiPictureId = rtcpPacketInformation.rpsiPictureId;
-    rtcp_packet_info_.receiverEstimatedMaxBitrate =
-        rtcpPacketInformation.receiverEstimatedMaxBitrate;
-    rtcp_packet_info_.ntp_secs = rtcpPacketInformation.ntp_secs;
-    rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac;
-    rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
-
-    return static_cast<int>(packet_len);
+  bool SendRtp(const uint8_t* /*data*/,
+               size_t /*len*/,
+               const PacketOptions& options) override {
+    return false;
   }
-
-  int OnReceivedPayloadData(const uint8_t* payloadData,
-                            const size_t payloadSize,
-                            const WebRtcRTPHeader* rtpHeader) override {
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    parser_.Parse(static_cast<const uint8_t*>(data), len);
+    return true;
+  }
+  int OnReceivedPayloadData(const uint8_t* payload_data,
+                            const size_t payload_size,
+                            const WebRtcRTPHeader* rtp_header) override {
     return 0;
   }
-  RTCPReceiver* rtcp_receiver_;
-  RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
+  test::RtcpPacketParser parser_;
 };
 
+namespace {
+static const uint32_t kSenderSsrc = 0x11111111;
+static const uint32_t kRemoteSsrc = 0x22222222;
+}
+
 class RtcpSenderTest : public ::testing::Test {
  protected:
-  static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
-
   RtcpSenderTest()
-      : over_use_detector_options_(),
-        clock_(1335900000),
-        rtp_payload_registry_(new RTPPayloadRegistry(
-            RTPPayloadStrategy::CreateStrategy(false))),
-        remote_bitrate_observer_(),
-        remote_bitrate_estimator_(
-            RemoteBitrateEstimatorFactory().Create(
-                &remote_bitrate_observer_,
-                &clock_,
-                kMimdControl,
-                kRemoteBitrateEstimatorMinBitrateBps)),
+      : clock_(1335900000),
         receive_statistics_(ReceiveStatistics::Create(&clock_)) {
-    test_transport_ = new TestTransport();
-
     RtpRtcp::Configuration configuration;
-    configuration.id = 0;
     configuration.audio = false;
     configuration.clock = &clock_;
-    configuration.outgoing_transport = test_transport_;
-    configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
+    configuration.outgoing_transport = &test_transport_;
 
-    rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
-    rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
-        0, &clock_, test_transport_, NULL, rtp_payload_registry_.get()));
-    rtcp_sender_ =
-        new RTCPSender(0, false, &clock_, receive_statistics_.get(), NULL);
-    rtcp_receiver_ = new RTCPReceiver(0, &clock_, NULL, NULL, NULL,
-                                      rtp_rtcp_impl_);
-    test_transport_->SetRTCPReceiver(rtcp_receiver_);
-    // Initialize
-    EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
-  }
-  ~RtcpSenderTest() {
-    delete rtcp_sender_;
-    delete rtcp_receiver_;
-    delete rtp_rtcp_impl_;
-    delete test_transport_;
+    rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
+    rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+                                      nullptr, &test_transport_));
+    rtcp_sender_->SetSSRC(kSenderSsrc);
+    rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
   }
 
-  // Helper function: Incoming RTCP has a specific packet type.
-  bool gotPacketType(RTCPPacketType packet_type) {
-    return ((test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags) &
-            packet_type) != 0U;
+  void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
+    RTPHeader header;
+    header.ssrc = remote_ssrc;
+    header.sequenceNumber = seq_num;
+    header.timestamp = 12345;
+    header.headerLength = 12;
+    size_t kPacketLength = 100;
+    receive_statistics_->IncomingPacket(header, kPacketLength, false);
+  }
+
+  test::RtcpPacketParser* parser() { return &test_transport_.parser_; }
+
+  RTCPSender::FeedbackState feedback_state() {
+    return rtp_rtcp_impl_->GetFeedbackState();
   }
 
-  OverUseDetectorOptions over_use_detector_options_;
   SimulatedClock clock_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
-  ModuleRtpRtcpImpl* rtp_rtcp_impl_;
-  RTCPSender* rtcp_sender_;
-  RTCPReceiver* rtcp_receiver_;
-  TestTransport* test_transport_;
-  MockRemoteBitrateObserver remote_bitrate_observer_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  TestTransport test_transport_;
   rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
-
-  enum {kMaxPacketLength = 1500};
-  uint8_t packet_[kMaxPacketLength];
+  rtc::scoped_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
+  rtc::scoped_ptr<RTCPSender> rtcp_sender_;
 };
 
-TEST_F(RtcpSenderTest, RtcpOff) {
-  rtcp_sender_->SetRTCPStatus(kRtcpOff);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(-1, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
+TEST_F(RtcpSenderTest, SetRtcpStatus) {
+  EXPECT_EQ(RtcpMode::kOff, rtcp_sender_->Status());
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(RtcpMode::kReducedSize, rtcp_sender_->Status());
 }
 
-TEST_F(RtcpSenderTest, IJStatus) {
-  ASSERT_FALSE(rtcp_sender_->IJ());
-  rtcp_sender_->SetIJStatus(true);
-  EXPECT_TRUE(rtcp_sender_->IJ());
+TEST_F(RtcpSenderTest, SetSendingStatus) {
+  EXPECT_FALSE(rtcp_sender_->Sending());
+  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true));
+  EXPECT_TRUE(rtcp_sender_->Sending());
 }
 
-TEST_F(RtcpSenderTest, TestCompound) {
-  const bool marker_bit = false;
-  const uint8_t payload_type = 100;
-  const uint16_t seq_num = 11111;
-  const uint32_t timestamp = 1234567;
-  const uint32_t ssrc = 0x11111111;
-  size_t packet_length = 0;
-  CreateRtpPacket(marker_bit, payload_type, seq_num, timestamp, ssrc, packet_,
-      &packet_length);
-  EXPECT_EQ(25u, packet_length);
-
-  VideoCodec codec_inst;
-  strncpy(codec_inst.plName, "VP8", webrtc::kPayloadNameSize - 1);
-  codec_inst.codecType = webrtc::kVideoCodecVP8;
-  codec_inst.plType = payload_type;
-  EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(codec_inst.plName,
-                                                     codec_inst.plType,
-                                                     90000,
-                                                     0,
-                                                     codec_inst.maxBitrate));
-
-  // Make sure RTP packet has been received.
-  rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-  RTPHeader header;
-  EXPECT_TRUE(parser->Parse(packet_, packet_length, &header));
-  PayloadUnion payload_specific;
-  EXPECT_TRUE(rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
-                                                        &payload_specific));
-  receive_statistics_->IncomingPacket(header, packet_length, false);
-  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, packet_, packet_length,
-                                               payload_specific, true));
-
-  rtcp_sender_->SetIJStatus(true);
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRr));
-
-  // Transmission time offset packet should be received.
-  ASSERT_TRUE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
-      kRtcpTransmissionTimeOffset);
+TEST_F(RtcpSenderTest, NoPacketSentIfOff) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kOff);
+  EXPECT_EQ(-1, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
 }
 
-TEST_F(RtcpSenderTest, TestCompound_NoRtpReceived) {
-  rtcp_sender_->SetIJStatus(true);
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
+TEST_F(RtcpSenderTest, SendSr) {
+  const uint32_t kPacketCount = 0x12345;
+  const uint32_t kOctetCount = 0x23456;
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
   RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRr));
-
-  // Transmission time offset packet should not be received.
-  ASSERT_FALSE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
-      kRtcpTransmissionTimeOffset);
+  feedback_state.packets_sent = kPacketCount;
+  feedback_state.media_bytes_sent = kOctetCount;
+  uint32_t ntp_secs;
+  uint32_t ntp_frac;
+  clock_.CurrentNtp(ntp_secs, ntp_frac);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->sender_report()->Ssrc());
+  EXPECT_EQ(ntp_secs, parser()->sender_report()->NtpSec());
+  EXPECT_EQ(ntp_frac, parser()->sender_report()->NtpFrac());
+  EXPECT_EQ(kPacketCount, parser()->sender_report()->PacketCount());
+  EXPECT_EQ(kOctetCount, parser()->sender_report()->OctetCount());
+  EXPECT_EQ(0, parser()->report_block()->num_packets());
 }
 
-TEST_F(RtcpSenderTest, TestXrReceiverReferenceTime) {
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
+TEST_F(RtcpSenderTest, SendRr) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->Ssrc());
+  EXPECT_EQ(0, parser()->report_block()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SendRrWithOneReportBlock) {
+  const uint16_t kSeqNum = 11111;
+  InsertIncomingPacket(kRemoteSsrc, kSeqNum);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->Ssrc());
+  EXPECT_EQ(1, parser()->report_block()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser()->report_block()->Ssrc());
+  EXPECT_EQ(0U, parser()->report_block()->FractionLost());
+  EXPECT_EQ(0U, parser()->report_block()->CumPacketLost());
+  EXPECT_EQ(kSeqNum, parser()->report_block()->ExtHighestSeqNum());
+}
+
+TEST_F(RtcpSenderTest, SendRrWithTwoReportBlocks) {
+  const uint16_t kSeqNum = 11111;
+  InsertIncomingPacket(kRemoteSsrc, kSeqNum);
+  InsertIncomingPacket(kRemoteSsrc + 1, kSeqNum + 1);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRr));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->Ssrc());
+  EXPECT_EQ(2, parser()->report_block()->num_packets());
+  EXPECT_EQ(1, parser()->report_blocks_per_ssrc(kRemoteSsrc));
+  EXPECT_EQ(1, parser()->report_blocks_per_ssrc(kRemoteSsrc + 1));
+}
+
+TEST_F(RtcpSenderTest, SendSdes) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SetCNAME("alice@host"));
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSdes));
+  EXPECT_EQ(1, parser()->sdes()->num_packets());
+  EXPECT_EQ(1, parser()->sdes_chunk()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->sdes_chunk()->Ssrc());
+  EXPECT_EQ("alice@host", parser()->sdes_chunk()->Cname());
+}
+
+TEST_F(RtcpSenderTest, SdesIncludedInCompoundPacket) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender_->SetCNAME("alice@host"));
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(1, parser()->sdes()->num_packets());
+  EXPECT_EQ(1, parser()->sdes_chunk()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SendBye) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
+  EXPECT_EQ(1, parser()->bye()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->bye()->Ssrc());
+}
+
+TEST_F(RtcpSenderTest, StopSendingTriggersBye) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true));
+  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
+  EXPECT_EQ(1, parser()->bye()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->bye()->Ssrc());
+}
+
+TEST_F(RtcpSenderTest, SendApp) {
+  const uint8_t kSubType = 30;
+  uint32_t name = 'n' << 24;
+  name += 'a' << 16;
+  name += 'm' << 8;
+  name += 'e';
+  const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'};
+  const uint16_t kDataLength = sizeof(kData) / sizeof(kData[0]);
+  EXPECT_EQ(0, rtcp_sender_->SetApplicationSpecificData(kSubType, name, kData,
+                                                        kDataLength));
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpApp));
+  EXPECT_EQ(1, parser()->app()->num_packets());
+  EXPECT_EQ(kSubType, parser()->app()->SubType());
+  EXPECT_EQ(name, parser()->app()->Name());
+  EXPECT_EQ(1, parser()->app_item()->num_packets());
+  EXPECT_EQ(kDataLength, parser()->app_item()->DataLength());
+  EXPECT_EQ(0, strncmp(reinterpret_cast<const char*>(kData),
+      reinterpret_cast<const char*>(parser()->app_item()->Data()),
+      parser()->app_item()->DataLength()));
+}
+
+TEST_F(RtcpSenderTest, SendEmptyApp) {
+  const uint8_t kSubType = 30;
+  const uint32_t kName = 0x6E616D65;
+
+  EXPECT_EQ(
+      0, rtcp_sender_->SetApplicationSpecificData(kSubType, kName, nullptr, 0));
+
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpApp));
+  EXPECT_EQ(1, parser()->app()->num_packets());
+  EXPECT_EQ(kSubType, parser()->app()->SubType());
+  EXPECT_EQ(kName, parser()->app()->Name());
+  EXPECT_EQ(0, parser()->app_item()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SetInvalidApplicationSpecificData) {
+  const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't'};
+  const uint16_t kInvalidDataLength = sizeof(kData) / sizeof(kData[0]);
+  EXPECT_EQ(-1, rtcp_sender_->SetApplicationSpecificData(
+      0, 0, kData, kInvalidDataLength));  // Should by multiple of 4.
+}
+
+TEST_F(RtcpSenderTest, SendFirNonRepeat) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpFir));
+  EXPECT_EQ(1, parser()->fir()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->fir()->Ssrc());
+  EXPECT_EQ(1, parser()->fir_item()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser()->fir_item()->Ssrc());
+  uint8_t seq = parser()->fir_item()->SeqNum();
+  // Sends non-repeat FIR as default.
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpFir));
+  EXPECT_EQ(2, parser()->fir()->num_packets());
+  EXPECT_EQ(2, parser()->fir_item()->num_packets());
+  EXPECT_EQ(seq + 1, parser()->fir_item()->SeqNum());
+}
+
+TEST_F(RtcpSenderTest, SendFirRepeat) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpFir));
+  EXPECT_EQ(1, parser()->fir()->num_packets());
+  EXPECT_EQ(1, parser()->fir_item()->num_packets());
+  uint8_t seq = parser()->fir_item()->SeqNum();
+  const bool kRepeat = true;
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpFir, 0, nullptr,
+                                      kRepeat));
+  EXPECT_EQ(2, parser()->fir()->num_packets());
+  EXPECT_EQ(2, parser()->fir_item()->num_packets());
+  EXPECT_EQ(seq, parser()->fir_item()->SeqNum());
+}
+
+TEST_F(RtcpSenderTest, SendPli) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
+  EXPECT_EQ(1, parser()->pli()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->pli()->Ssrc());
+  EXPECT_EQ(kRemoteSsrc, parser()->pli()->MediaSsrc());
+}
+
+TEST_F(RtcpSenderTest, SendRpsi) {
+  const uint64_t kPictureId = 0x41;
+  const int8_t kPayloadType = 100;
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
   RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false));
-  rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
+  feedback_state.send_payload_type = kPayloadType;
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpRpsi, 0, nullptr,
+                                      false, kPictureId));
+  EXPECT_EQ(kPayloadType, parser()->rpsi()->PayloadType());
+  EXPECT_EQ(kPictureId, parser()->rpsi()->PictureId());
+}
+
+TEST_F(RtcpSenderTest, SendSli) {
+  const uint16_t kFirstMb = 0;
+  const uint16_t kNumberOfMb = 0x1FFF;
+  const uint8_t kPictureId = 60;
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSli, 0, nullptr,
+                                      false, kPictureId));
+  EXPECT_EQ(1, parser()->sli()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->sli()->Ssrc());
+  EXPECT_EQ(kRemoteSsrc, parser()->sli()->MediaSsrc());
+  EXPECT_EQ(1, parser()->sli_item()->num_packets());
+  EXPECT_EQ(kFirstMb, parser()->sli_item()->FirstMb());
+  EXPECT_EQ(kNumberOfMb, parser()->sli_item()->NumberOfMb());
+  EXPECT_EQ(kPictureId, parser()->sli_item()->PictureId());
+}
+
+TEST_F(RtcpSenderTest, SendNack) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  const uint16_t kList[] = {0, 1, 16};
+  const int32_t kListLength = sizeof(kList) / sizeof(kList[0]);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpNack, kListLength,
+                                      kList));
+  EXPECT_EQ(1, parser()->nack()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->nack()->Ssrc());
+  EXPECT_EQ(kRemoteSsrc, parser()->nack()->MediaSsrc());
+  EXPECT_EQ(1, parser()->nack_item()->num_packets());
+  EXPECT_THAT(parser()->nack_item()->last_nack_list(), ElementsAre(0, 1, 16));
+}
+
+TEST_F(RtcpSenderTest, SendRemb) {
+  const int kBitrate = 261011;
+  std::vector<uint32_t> ssrcs;
+  ssrcs.push_back(kRemoteSsrc);
+  ssrcs.push_back(kRemoteSsrc + 1);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender_->SetREMBData(kBitrate, ssrcs);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpRemb));
+  EXPECT_EQ(1, parser()->psfb_app()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->psfb_app()->Ssrc());
+  EXPECT_EQ(1, parser()->remb_item()->num_packets());
+  EXPECT_EQ(kBitrate, parser()->remb_item()->last_bitrate_bps());
+  EXPECT_THAT(parser()->remb_item()->last_ssrc_list(),
+              ElementsAre(kRemoteSsrc, kRemoteSsrc + 1));
+}
+
+TEST_F(RtcpSenderTest, RembIncludedInCompoundPacketIfEnabled) {
+  const int kBitrate = 261011;
+  std::vector<uint32_t> ssrcs;
+  ssrcs.push_back(kRemoteSsrc);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender_->SetREMBStatus(true);
+  EXPECT_TRUE(rtcp_sender_->REMB());
+  rtcp_sender_->SetREMBData(kBitrate, ssrcs);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->psfb_app()->num_packets());
+  EXPECT_EQ(1, parser()->remb_item()->num_packets());
+  // REMB should be included in each compound packet.
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(2, parser()->psfb_app()->num_packets());
+  EXPECT_EQ(2, parser()->remb_item()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, RembNotIncludedInCompoundPacketIfNotEnabled) {
+  const int kBitrate = 261011;
+  std::vector<uint32_t> ssrcs;
+  ssrcs.push_back(kRemoteSsrc);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender_->SetREMBData(kBitrate, ssrcs);
+  EXPECT_FALSE(rtcp_sender_->REMB());
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(0, parser()->psfb_app()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SendXrWithVoipMetric) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  RTCPVoIPMetric metric;
+  metric.lossRate = 1;
+  metric.discardRate = 2;
+  metric.burstDensity = 3;
+  metric.gapDensity = 4;
+  metric.burstDuration = 0x1111;
+  metric.gapDuration = 0x2222;
+  metric.roundTripDelay = 0x3333;
+  metric.endSystemDelay = 0x4444;
+  metric.signalLevel = 5;
+  metric.noiseLevel = 6;
+  metric.RERL = 7;
+  metric.Gmin = 8;
+  metric.Rfactor = 9;
+  metric.extRfactor = 10;
+  metric.MOSLQ = 11;
+  metric.MOSCQ = 12;
+  metric.RXconfig = 13;
+  metric.JBnominal = 0x5555;
+  metric.JBmax = 0x6666;
+  metric.JBabsMax = 0x7777;
+  EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric));
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpXrVoipMetric));
+  EXPECT_EQ(1, parser()->xr_header()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr_header()->Ssrc());
+  EXPECT_EQ(1, parser()->voip_metric()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, parser()->voip_metric()->Ssrc());
+  EXPECT_EQ(metric.lossRate, parser()->voip_metric()->LossRate());
+  EXPECT_EQ(metric.discardRate, parser()->voip_metric()->DiscardRate());
+  EXPECT_EQ(metric.burstDensity, parser()->voip_metric()->BurstDensity());
+  EXPECT_EQ(metric.gapDensity, parser()->voip_metric()->GapDensity());
+  EXPECT_EQ(metric.burstDuration, parser()->voip_metric()->BurstDuration());
+  EXPECT_EQ(metric.gapDuration, parser()->voip_metric()->GapDuration());
+  EXPECT_EQ(metric.roundTripDelay, parser()->voip_metric()->RoundTripDelay());
+  EXPECT_EQ(metric.endSystemDelay, parser()->voip_metric()->EndSystemDelay());
+  EXPECT_EQ(metric.signalLevel, parser()->voip_metric()->SignalLevel());
+  EXPECT_EQ(metric.noiseLevel, parser()->voip_metric()->NoiseLevel());
+  EXPECT_EQ(metric.RERL, parser()->voip_metric()->Rerl());
+  EXPECT_EQ(metric.Gmin, parser()->voip_metric()->Gmin());
+  EXPECT_EQ(metric.Rfactor, parser()->voip_metric()->Rfactor());
+  EXPECT_EQ(metric.extRfactor, parser()->voip_metric()->ExtRfactor());
+  EXPECT_EQ(metric.MOSLQ, parser()->voip_metric()->MosLq());
+  EXPECT_EQ(metric.MOSCQ, parser()->voip_metric()->MosCq());
+  EXPECT_EQ(metric.RXconfig, parser()->voip_metric()->RxConfig());
+  EXPECT_EQ(metric.JBnominal, parser()->voip_metric()->JbNominal());
+  EXPECT_EQ(metric.JBmax, parser()->voip_metric()->JbMax());
+  EXPECT_EQ(metric.JBabsMax, parser()->voip_metric()->JbAbsMax());
+}
+
+TEST_F(RtcpSenderTest, SendXrWithDlrr) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
+  feedback_state.has_last_xr_rr = true;
+  RtcpReceiveTimeInfo last_xr_rr;
+  last_xr_rr.sourceSSRC = 0x11111111;
+  last_xr_rr.lastRR = 0x22222222;
+  last_xr_rr.delaySinceLastRR = 0x33333333;
+  feedback_state.last_xr_rr = last_xr_rr;
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
-
-  EXPECT_TRUE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
-      kRtcpXrReceiverReferenceTime);
+  EXPECT_EQ(1, parser()->xr_header()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr_header()->Ssrc());
+  EXPECT_EQ(1, parser()->dlrr()->num_packets());
+  EXPECT_EQ(1, parser()->dlrr_items()->num_packets());
+  EXPECT_EQ(last_xr_rr.sourceSSRC, parser()->dlrr_items()->Ssrc(0));
+  EXPECT_EQ(last_xr_rr.lastRR, parser()->dlrr_items()->LastRr(0));
+  EXPECT_EQ(last_xr_rr.delaySinceLastRR,
+            parser()->dlrr_items()->DelayLastRr(0));
 }
 
-TEST_F(RtcpSenderTest, TestNoXrReceiverReferenceTimeIfSending) {
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, true));
+TEST_F(RtcpSenderTest, SendXrWithRrtr) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
   rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
-  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
-
-  EXPECT_FALSE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
-      kRtcpXrReceiverReferenceTime);
+  uint32_t ntp_secs;
+  uint32_t ntp_frac;
+  clock_.CurrentNtp(ntp_secs, ntp_frac);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->xr_header()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr_header()->Ssrc());
+  EXPECT_EQ(0, parser()->dlrr()->num_packets());
+  EXPECT_EQ(1, parser()->rrtr()->num_packets());
+  EXPECT_EQ(ntp_secs, parser()->rrtr()->NtpSec());
+  EXPECT_EQ(ntp_frac, parser()->rrtr()->NtpFrac());
 }
 
-TEST_F(RtcpSenderTest, TestNoXrReceiverReferenceTimeIfNotEnabled) {
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false));
+TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfSending) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true));
+  rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(0, parser()->xr_header()->num_packets());
+  EXPECT_EQ(0, parser()->rrtr()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
   rtcp_sender_->SendRtcpXrReceiverReferenceTime(false);
-  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
-
-  EXPECT_FALSE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
-      kRtcpXrReceiverReferenceTime);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(0, parser()->xr_header()->num_packets());
+  EXPECT_EQ(0, parser()->rrtr()->num_packets());
 }
 
-TEST_F(RtcpSenderTest, TestSendTimeOfXrRrReport) {
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
+TEST_F(RtcpSenderTest, TestSendTimeOfXrRrtr) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
   EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state, false));
   rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
@@ -453,20 +655,81 @@ TEST_F(RtcpSenderTest, TestSendTimeOfXrRrReport) {
   // Send XR RR packets.
   for (int i = 0; i <= RTCP_NUMBER_OF_SR; ++i) {
     EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
-    EXPECT_TRUE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
-        kRtcpXrReceiverReferenceTime);
-
+    EXPECT_EQ(i + 1, test_transport_.parser_.rrtr()->num_packets());
     clock_.CurrentNtp(ntp_sec, ntp_frac);
     uint32_t mid_ntp = RTCPUtility::MidNtp(ntp_sec, ntp_frac);
     EXPECT_TRUE(rtcp_sender_->SendTimeOfXrRrReport(mid_ntp, &time_ms));
     EXPECT_EQ(clock_.CurrentNtpInMilliseconds(), time_ms);
     clock_.AdvanceTimeMilliseconds(1000);
   }
-
   // The first report should no longer be stored.
   EXPECT_FALSE(rtcp_sender_->SendTimeOfXrRrReport(initial_mid_ntp, &time_ms));
 }
 
+TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
+  RtcpPacketTypeCounterObserverImpl observer;
+  rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+                                    &observer, &test_transport_));
+  rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
+  EXPECT_EQ(1, parser()->pli()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, observer.ssrc_);
+  EXPECT_EQ(1U, observer.counter_.pli_packets);
+  EXPECT_EQ(clock_.TimeInMilliseconds(),
+            observer.counter_.first_packet_time_ms);
+}
+
+TEST_F(RtcpSenderTest, SendTmmbr) {
+  const unsigned int kBitrateBps = 312000;
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender_->SetTargetBitrate(kBitrateBps);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpTmmbr));
+  EXPECT_EQ(1, parser()->tmmbr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->tmmbr()->Ssrc());
+  EXPECT_EQ(1, parser()->tmmbr_item()->num_packets());
+  EXPECT_EQ(kBitrateBps / 1000, parser()->tmmbr_item()->BitrateKbps());
+  // TODO(asapersson): tmmbr_item()->Overhead() looks broken, always zero.
+}
+
+TEST_F(RtcpSenderTest, TmmbrIncludedInCompoundPacketIfEnabled) {
+  const unsigned int kBitrateBps = 312000;
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_FALSE(rtcp_sender_->TMMBR());
+  rtcp_sender_->SetTMMBRStatus(true);
+  EXPECT_TRUE(rtcp_sender_->TMMBR());
+  rtcp_sender_->SetTargetBitrate(kBitrateBps);
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->tmmbr()->num_packets());
+  EXPECT_EQ(1, parser()->tmmbr_item()->num_packets());
+  // TMMBR should be included in each compound packet.
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(2, parser()->tmmbr()->num_packets());
+  EXPECT_EQ(2, parser()->tmmbr_item()->num_packets());
+
+  rtcp_sender_->SetTMMBRStatus(false);
+  EXPECT_FALSE(rtcp_sender_->TMMBR());
+}
+
+TEST_F(RtcpSenderTest, SendTmmbn) {
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  TMMBRSet bounding_set;
+  bounding_set.VerifyAndAllocateSet(1);
+  const uint32_t kBitrateKbps = 32768;
+  const uint32_t kPacketOh = 40;
+  const uint32_t kSourceSsrc = 12345;
+  bounding_set.AddEntry(kBitrateKbps, kPacketOh, kSourceSsrc);
+  EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set, 0));
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  EXPECT_EQ(1, parser()->tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->tmmbn()->Ssrc());
+  EXPECT_EQ(1, parser()->tmmbn_items()->num_packets());
+  EXPECT_EQ(kBitrateKbps, parser()->tmmbn_items()->BitrateKbps(0));
+  EXPECT_EQ(kPacketOh, parser()->tmmbn_items()->Overhead(0));
+  EXPECT_EQ(kSourceSsrc, parser()->tmmbn_items()->Ssrc(0));
+}
+
 // This test is written to verify actual behaviour. It does not seem
 // to make much sense to send an empty TMMBN, since there is no place
 // to put an actual limit here. It's just information that no limit
@@ -474,44 +737,28 @@ TEST_F(RtcpSenderTest, TestSendTimeOfXrRrReport) {
 // See http://code.google.com/p/webrtc/issues/detail?id=468 for one
 // situation where this caused confusion.
 TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) {
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
   TMMBRSet bounding_set;
   EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set, 3));
-  ASSERT_EQ(0U, test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state,kRtcpSr));
-  // We now expect the packet to show up in the rtcp_packet_info_ of
-  // test_transport_.
-  ASSERT_NE(0U, test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags);
-  EXPECT_TRUE(gotPacketType(kRtcpTmmbn));
-  TMMBRSet* incoming_set = NULL;
-  bool owner = false;
-  // The BoundingSet function returns the number of members of the
-  // bounding set, and touches the incoming set only if there's > 1.
-  EXPECT_EQ(0, test_transport_->rtcp_receiver_->BoundingSet(owner,
-      incoming_set));
+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  EXPECT_EQ(1, parser()->tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->tmmbn()->Ssrc());
+  EXPECT_EQ(0, parser()->tmmbn_items()->num_packets());
 }
 
-TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndValid) {
-  rtcp_sender_->SetRTCPStatus(kRtcpCompound);
-  TMMBRSet bounding_set;
-  bounding_set.VerifyAndAllocateSet(1);
-  const uint32_t kSourceSsrc = 12345;
-  bounding_set.AddEntry(32768, 0, kSourceSsrc);
-
-  EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set, 3));
-  ASSERT_EQ(0U, test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags);
-  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
-  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
-  // We now expect the packet to show up in the rtcp_packet_info_ of
-  // test_transport_.
-  ASSERT_NE(0U, test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags);
-  EXPECT_TRUE(gotPacketType(kRtcpTmmbn));
-  TMMBRSet incoming_set;
-  bool owner = false;
-  // We expect 1 member of the incoming set.
-  EXPECT_EQ(1, test_transport_->rtcp_receiver_->BoundingSet(owner,
-      &incoming_set));
-  EXPECT_EQ(kSourceSsrc, incoming_set.Ssrc(0));
+TEST_F(RtcpSenderTest, SendCompoundPliRemb) {
+  const int kBitrate = 261011;
+  std::vector<uint32_t> ssrcs;
+  ssrcs.push_back(kRemoteSsrc);
+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender_->SetREMBData(kBitrate, ssrcs);
+  std::set<RTCPPacketType> packet_types;
+  packet_types.insert(kRtcpRemb);
+  packet_types.insert(kRtcpPli);
+  EXPECT_EQ(0, rtcp_sender_->SendCompoundRTCP(feedback_state(), packet_types));
+  EXPECT_EQ(1, parser()->remb_item()->num_packets());
+  EXPECT_EQ(1, parser()->pli()->num_packets());
 }
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
index 0423389632..e19499612d 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
@@ -14,6 +14,11 @@
 #include <math.h>   // ceil
 #include <string.h> // memcpy
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
 namespace webrtc {
 
 namespace RTCPUtility {
@@ -36,8 +41,8 @@ void NackStats::ReportRequest(uint16_t sequence_number) {
 
 uint32_t MidNtp(uint32_t ntp_sec, uint32_t ntp_frac) {
   return (ntp_sec << 16) + (ntp_frac >> 16);
-}  // end RTCPUtility
 }
+}  // namespace RTCPUtility
 
 // RTCPParserV2 : currently read only
 RTCPUtility::RTCPParserV2::RTCPParserV2(const uint8_t* rtcpData,
@@ -49,9 +54,10 @@ RTCPUtility::RTCPParserV2::RTCPParserV2(const uint8_t* rtcpData,
       _validPacket(false),
       _ptrRTCPData(rtcpData),
       _ptrRTCPBlockEnd(NULL),
-      _state(State_TopLevel),
+      _state(ParseState::State_TopLevel),
       _numberOfBlocks(0),
-      _packetType(kRtcpNotValidCode) {
+      num_skipped_blocks_(0),
+      _packetType(RTCPPacketTypes::kInvalid) {
   Validate();
 }
 
@@ -76,6 +82,9 @@ RTCPUtility::RTCPParserV2::Packet() const
     return _packet;
 }
 
+rtcp::RtcpPacket* RTCPUtility::RTCPParserV2::ReleaseRtcpPacket() {
+  return rtcp_packet_.release();
+}
 RTCPUtility::RTCPPacketTypes
 RTCPUtility::RTCPParserV2::Begin()
 {
@@ -88,62 +97,62 @@ RTCPUtility::RTCPPacketTypes
 RTCPUtility::RTCPParserV2::Iterate()
 {
     // Reset packet type
-    _packetType = kRtcpNotValidCode;
+  _packetType = RTCPPacketTypes::kInvalid;
 
     if (IsValid())
     {
         switch (_state)
         {
-        case State_TopLevel:
+          case ParseState::State_TopLevel:
             IterateTopLevel();
             break;
-        case State_ReportBlockItem:
+          case ParseState::State_ReportBlockItem:
             IterateReportBlockItem();
             break;
-        case State_SDESChunk:
+          case ParseState::State_SDESChunk:
             IterateSDESChunk();
             break;
-        case State_BYEItem:
+          case ParseState::State_BYEItem:
             IterateBYEItem();
             break;
-        case State_ExtendedJitterItem:
+          case ParseState::State_ExtendedJitterItem:
             IterateExtendedJitterItem();
             break;
-        case State_RTPFB_NACKItem:
+          case ParseState::State_RTPFB_NACKItem:
             IterateNACKItem();
             break;
-        case State_RTPFB_TMMBRItem:
+          case ParseState::State_RTPFB_TMMBRItem:
             IterateTMMBRItem();
             break;
-        case State_RTPFB_TMMBNItem:
+          case ParseState::State_RTPFB_TMMBNItem:
             IterateTMMBNItem();
             break;
-        case State_PSFB_SLIItem:
+          case ParseState::State_PSFB_SLIItem:
             IterateSLIItem();
             break;
-        case State_PSFB_RPSIItem:
+          case ParseState::State_PSFB_RPSIItem:
             IterateRPSIItem();
             break;
-        case State_PSFB_FIRItem:
+          case ParseState::State_PSFB_FIRItem:
             IterateFIRItem();
             break;
-        case State_PSFB_AppItem:
+          case ParseState::State_PSFB_AppItem:
             IteratePsfbAppItem();
             break;
-        case State_PSFB_REMBItem:
+          case ParseState::State_PSFB_REMBItem:
             IteratePsfbREMBItem();
             break;
-        case State_XRItem:
+          case ParseState::State_XRItem:
             IterateXrItem();
             break;
-        case State_XR_DLLRItem:
+          case ParseState::State_XR_DLLRItem:
             IterateXrDlrrItem();
             break;
-        case State_AppItem:
+          case ParseState::State_AppItem:
             IterateAppItem();
             break;
         default:
-            assert(false); // Invalid state!
+          RTC_NOTREACHED() << "Invalid state!";
             break;
         }
     }
@@ -155,43 +164,40 @@ RTCPUtility::RTCPParserV2::IterateTopLevel()
 {
     for (;;)
     {
-        RTCPCommonHeader header;
+      RtcpCommonHeader header;
+      if (_ptrRTCPDataEnd <= _ptrRTCPData)
+        return;
 
-        const bool success = RTCPParseCommonHeader(_ptrRTCPData,
-                                                    _ptrRTCPDataEnd,
-                                                    header);
-
-        if (!success)
-        {
+      if (!RtcpParseCommonHeader(_ptrRTCPData, _ptrRTCPDataEnd - _ptrRTCPData,
+                                 &header)) {
             return;
         }
-        _ptrRTCPBlockEnd = _ptrRTCPData + header.LengthInOctets;
+        _ptrRTCPBlockEnd = _ptrRTCPData + header.BlockSize();
         if (_ptrRTCPBlockEnd > _ptrRTCPDataEnd)
         {
-            // Bad block!
+          ++num_skipped_blocks_;
             return;
         }
 
-        switch (header.PT)
-        {
+        switch (header.packet_type) {
         case PT_SR:
         {
             // number of Report blocks
-            _numberOfBlocks = header.IC;
+            _numberOfBlocks = header.count_or_format;
             ParseSR();
             return;
         }
         case PT_RR:
         {
             // number of Report blocks
-            _numberOfBlocks = header.IC;
+            _numberOfBlocks = header.count_or_format;
             ParseRR();
             return;
         }
         case PT_SDES:
         {
             // number of SDES blocks
-            _numberOfBlocks = header.IC;
+            _numberOfBlocks = header.count_or_format;
             const bool ok = ParseSDES();
             if (!ok)
             {
@@ -202,7 +208,7 @@ RTCPUtility::RTCPParserV2::IterateTopLevel()
         }
         case PT_BYE:
         {
-            _numberOfBlocks = header.IC;
+          _numberOfBlocks = header.count_or_format;
             const bool ok = ParseBYE();
             if (!ok)
             {
@@ -214,20 +220,19 @@ RTCPUtility::RTCPParserV2::IterateTopLevel()
         case PT_IJ:
         {
             // number of Report blocks
-            _numberOfBlocks = header.IC;
+            _numberOfBlocks = header.count_or_format;
             ParseIJ();
             return;
         }
-        case PT_RTPFB: // Fall through!
+        case PT_RTPFB:
+          FALLTHROUGH();
         case PT_PSFB:
         {
-            const bool ok = ParseFBCommon(header);
-            if (!ok)
-            {
-                // Nothing supported found, continue to next block!
-                break;
-            }
-            return;
+          if (!ParseFBCommon(header)) {
+            // Nothing supported found, continue to next block!
+            break;
+          }
+          return;
         }
         case PT_APP:
         {
@@ -251,6 +256,7 @@ RTCPUtility::RTCPParserV2::IterateTopLevel()
         }
         default:
             // Not supported! Skip!
+            ++num_skipped_blocks_;
             EndCurrentBlock();
             break;
         }
@@ -410,20 +416,16 @@ RTCPUtility::RTCPParserV2::IterateAppItem()
 void
 RTCPUtility::RTCPParserV2::Validate()
 {
-    if (_ptrRTCPData == NULL)
-    {
-        return; // NOT VALID
-    }
+  if (_ptrRTCPData == nullptr)
+    return;  // NOT VALID
 
-    RTCPCommonHeader header;
-    const bool success = RTCPParseCommonHeader(_ptrRTCPDataBegin,
-                                               _ptrRTCPDataEnd,
-                                               header);
+  RtcpCommonHeader header;
+  if (_ptrRTCPDataEnd <= _ptrRTCPDataBegin)
+    return;  // NOT VALID
 
-    if (!success)
-    {
-        return; // NOT VALID!
-    }
+  if (!RtcpParseCommonHeader(_ptrRTCPDataBegin,
+                             _ptrRTCPDataEnd - _ptrRTCPDataBegin, &header))
+    return;  // NOT VALID!
 
     // * if (!reducedSize) : first packet must be RR or SR.
     //
@@ -437,8 +439,7 @@ RTCPUtility::RTCPParserV2::Validate()
 
     if (!_RTCPReducedSizeEnable)
     {
-        if ((header.PT != PT_SR) && (header.PT != PT_RR))
-        {
+      if ((header.packet_type != PT_SR) && (header.packet_type != PT_RR)) {
             return; // NOT VALID
         }
     }
@@ -458,48 +459,74 @@ RTCPUtility::RTCPParserV2::EndCurrentBlock()
     _ptrRTCPData = _ptrRTCPBlockEnd;
 }
 
-bool
-RTCPUtility::RTCPParseCommonHeader( const uint8_t* ptrDataBegin,
-                                    const uint8_t* ptrDataEnd,
-                                    RTCPCommonHeader& parsedHeader)
-{
-    if (!ptrDataBegin || !ptrDataEnd)
-    {
-        return false;
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |V=2|P|    IC   |      PT       |             length            |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Common header for all RTCP packets, 4 octets.
+
+bool RTCPUtility::RtcpParseCommonHeader(const uint8_t* packet,
+                                        size_t size_bytes,
+                                        RtcpCommonHeader* parsed_header) {
+  RTC_DCHECK(parsed_header != nullptr);
+  if (size_bytes < RtcpCommonHeader::kHeaderSizeBytes) {
+    LOG(LS_WARNING) << "Too little data (" << size_bytes << " byte"
+                    << (size_bytes != 1 ? "s" : "")
+                    << ") remaining in buffer to parse RTCP header (4 bytes).";
+    return false;
+  }
+
+  const uint8_t kRtcpVersion = 2;
+  uint8_t version = packet[0] >> 6;
+  if (version != kRtcpVersion) {
+    LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
+                    << static_cast<int>(kRtcpVersion) << " but was "
+                    << static_cast<int>(version);
+    return false;
+  }
+
+  bool has_padding = (packet[0] & 0x20) != 0;
+  uint8_t format = packet[0] & 0x1F;
+  uint8_t packet_type = packet[1];
+  size_t packet_size_words =
+      ByteReader<uint16_t>::ReadBigEndian(&packet[2]) + 1;
+
+  if (size_bytes < packet_size_words * 4) {
+    LOG(LS_WARNING) << "Buffer too small (" << size_bytes
+                    << " bytes) to fit an RtcpPacket of " << packet_size_words
+                    << " 32bit words.";
+    return false;
+  }
+
+  size_t payload_size = packet_size_words * 4;
+  size_t padding_bytes = 0;
+  if (has_padding) {
+    if (payload_size <= RtcpCommonHeader::kHeaderSizeBytes) {
+      LOG(LS_WARNING) << "Invalid RTCP header: Padding bit set but 0 payload "
+                         "size specified.";
+      return false;
     }
 
-    //  0                   1                   2                   3
-    //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-    // |V=2|P|    IC   |      PT       |             length            |
-    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-    //
-    // Common header for all RTCP packets, 4 octets.
-
-    if ((ptrDataEnd - ptrDataBegin) < 4)
-    {
-        return false;
+    padding_bytes = packet[payload_size - 1];
+    if (RtcpCommonHeader::kHeaderSizeBytes + padding_bytes > payload_size) {
+      LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
+                      << padding_bytes << ") for a packet size of "
+                      << payload_size << "bytes.";
+      return false;
     }
+    payload_size -= padding_bytes;
+  }
+  payload_size -= RtcpCommonHeader::kHeaderSizeBytes;
 
-    parsedHeader.V              = ptrDataBegin[0] >> 6;
-    parsedHeader.P              = ((ptrDataBegin[0] & 0x20) == 0) ? false : true;
-    parsedHeader.IC             = ptrDataBegin[0] & 0x1f;
-    parsedHeader.PT             = ptrDataBegin[1];
+  parsed_header->version = kRtcpVersion;
+  parsed_header->count_or_format = format;
+  parsed_header->packet_type = packet_type;
+  parsed_header->payload_size_bytes = payload_size;
+  parsed_header->padding_bytes = padding_bytes;
 
-    parsedHeader.LengthInOctets = (ptrDataBegin[2] << 8) + ptrDataBegin[3] + 1;
-    parsedHeader.LengthInOctets *= 4;
-
-    if(parsedHeader.LengthInOctets == 0)
-    {
-        return false;
-    }
-    // Check if RTP version field == 2
-    if (parsedHeader.V != 2)
-    {
-        return false;
-    }
-
-    return true;
+  return true;
 }
 
 bool
@@ -515,7 +542,7 @@ RTCPUtility::RTCPParserV2::ParseRR()
 
     _ptrRTCPData += 4; // Skip header
 
-    _packetType = kRtcpRrCode;
+    _packetType = RTCPPacketTypes::kRr;
 
     _packet.RR.SenderSSRC = *_ptrRTCPData++ << 24;
     _packet.RR.SenderSSRC += *_ptrRTCPData++ << 16;
@@ -525,7 +552,7 @@ RTCPUtility::RTCPParserV2::ParseRR()
     _packet.RR.NumberOfReportBlocks = _numberOfBlocks;
 
     // State transition
-    _state = State_ReportBlockItem;
+    _state = ParseState::State_ReportBlockItem;
 
     return true;
 }
@@ -543,7 +570,7 @@ RTCPUtility::RTCPParserV2::ParseSR()
 
     _ptrRTCPData += 4; // Skip header
 
-    _packetType = kRtcpSrCode;
+    _packetType = RTCPPacketTypes::kSr;
 
     _packet.SR.SenderSSRC = *_ptrRTCPData++ << 24;
     _packet.SR.SenderSSRC += *_ptrRTCPData++ << 16;
@@ -580,11 +607,11 @@ RTCPUtility::RTCPParserV2::ParseSR()
     // State transition
     if(_numberOfBlocks != 0)
     {
-        _state = State_ReportBlockItem;
+      _state = ParseState::State_ReportBlockItem;
     }else
     {
         // don't go to state report block item if 0 report blocks
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
         EndCurrentBlock();
     }
     return true;
@@ -597,7 +624,7 @@ RTCPUtility::RTCPParserV2::ParseReportBlockItem()
 
     if (length < 24 || _numberOfBlocks <= 0)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
@@ -634,7 +661,7 @@ RTCPUtility::RTCPParserV2::ParseReportBlockItem()
     _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++;
 
     _numberOfBlocks--;
-    _packetType = kRtcpReportBlockItemCode;
+    _packetType = RTCPPacketTypes::kReportBlockItem;
     return true;
 }
 
@@ -665,10 +692,10 @@ RTCPUtility::RTCPParserV2::ParseIJ()
 
     _ptrRTCPData += 4; // Skip header
 
-    _packetType = kRtcpExtendedIjCode;
+    _packetType = RTCPPacketTypes::kExtendedIj;
 
     // State transition
-    _state = State_ExtendedJitterItem;
+    _state = ParseState::State_ExtendedJitterItem;
     return true;
 }
 
@@ -679,7 +706,7 @@ RTCPUtility::RTCPParserV2::ParseIJItem()
 
     if (length < 4 || _numberOfBlocks <= 0)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
         EndCurrentBlock();
         return false;
     }
@@ -690,7 +717,7 @@ RTCPUtility::RTCPParserV2::ParseIJItem()
     _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++;
 
     _numberOfBlocks--;
-    _packetType = kRtcpExtendedIjItemCode;
+    _packetType = RTCPPacketTypes::kExtendedIjItem;
     return true;
 }
 
@@ -701,15 +728,15 @@ RTCPUtility::RTCPParserV2::ParseSDES()
 
     if (length < 8)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
     _ptrRTCPData += 4; // Skip header
 
-    _state = State_SDESChunk;
-    _packetType = kRtcpSdesCode;
+    _state = ParseState::State_SDESChunk;
+    _packetType = RTCPPacketTypes::kSdes;
     return true;
 }
 
@@ -718,7 +745,7 @@ RTCPUtility::RTCPParserV2::ParseSDESChunk()
 {
     if(_numberOfBlocks <= 0)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
@@ -731,7 +758,7 @@ RTCPUtility::RTCPParserV2::ParseSDESChunk()
         const ptrdiff_t dataLen = _ptrRTCPBlockEnd - _ptrRTCPData;
         if (dataLen < 4)
         {
-            _state = State_TopLevel;
+          _state = ParseState::State_TopLevel;
 
             EndCurrentBlock();
             return false;
@@ -749,7 +776,7 @@ RTCPUtility::RTCPParserV2::ParseSDESChunk()
             return true;
         }
     }
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
 
     EndCurrentBlock();
     return false;
@@ -790,7 +817,7 @@ RTCPUtility::RTCPParserV2::ParseSDESItem()
                 // Sanity
                 if ((_ptrRTCPData + len) >= _ptrRTCPBlockEnd)
                 {
-                    _state = State_TopLevel;
+                  _state = ParseState::State_TopLevel;
 
                     EndCurrentBlock();
                     return false;
@@ -802,7 +829,7 @@ RTCPUtility::RTCPParserV2::ParseSDESItem()
                     if ((c < ' ') || (c > '{') || (c == '%') || (c == '\\'))
                     {
                         // Illegal char
-                        _state = State_TopLevel;
+                      _state = ParseState::State_TopLevel;
 
                         EndCurrentBlock();
                         return false;
@@ -811,7 +838,7 @@ RTCPUtility::RTCPParserV2::ParseSDESItem()
                 }
                 // Make sure we are null terminated.
                 _packet.CName.CName[i] = 0;
-                _packetType = kRtcpSdesChunkCode;
+                _packetType = RTCPPacketTypes::kSdesChunk;
 
                 foundCName = true;
             }
@@ -821,7 +848,7 @@ RTCPUtility::RTCPParserV2::ParseSDESItem()
     }
 
     // No end tag found!
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
 
     EndCurrentBlock();
     return false;
@@ -832,7 +859,7 @@ RTCPUtility::RTCPParserV2::ParseBYE()
 {
     _ptrRTCPData += 4; // Skip header
 
-    _state = State_BYEItem;
+    _state = ParseState::State_BYEItem;
 
     return ParseBYEItem();
 }
@@ -843,13 +870,13 @@ RTCPUtility::RTCPParserV2::ParseBYEItem()
     const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
     if (length < 4 || _numberOfBlocks == 0)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpByeCode;
+    _packetType = RTCPPacketTypes::kBye;
 
     _packet.BYE.SenderSSRC = *_ptrRTCPData++ << 24;
     _packet.BYE.SenderSSRC += *_ptrRTCPData++ << 16;
@@ -894,8 +921,8 @@ bool RTCPUtility::RTCPParserV2::ParseXr()
     _packet.XR.OriginatorSSRC += *_ptrRTCPData++ << 8;
     _packet.XR.OriginatorSSRC += *_ptrRTCPData++;
 
-    _packetType = kRtcpXrHeaderCode;
-    _state = State_XRItem;
+    _packetType = RTCPPacketTypes::kXrHeader;
+    _state = ParseState::State_XRItem;
     return true;
 }
 
@@ -915,7 +942,7 @@ bool RTCPUtility::RTCPParserV2::ParseXrItem() {
   const int kBlockHeaderLengthInBytes = 4;
   const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
   if (length < kBlockHeaderLengthInBytes) {
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
     EndCurrentBlock();
     return false;
   }
@@ -956,7 +983,7 @@ bool RTCPUtility::RTCPParserV2::ParseXrReceiverReferenceTimeItem(
   const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
   if (block_length_4bytes != kBlockLengthIn4Bytes ||
       length < kBlockLengthInBytes) {
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
     EndCurrentBlock();
     return false;
   }
@@ -971,8 +998,8 @@ bool RTCPUtility::RTCPParserV2::ParseXrReceiverReferenceTimeItem(
   _packet.XRReceiverReferenceTimeItem.NTPLeastSignificant+= *_ptrRTCPData++<<8;
   _packet.XRReceiverReferenceTimeItem.NTPLeastSignificant+= *_ptrRTCPData++;
 
-  _packetType = kRtcpXrReceiverReferenceTimeCode;
-  _state = State_XRItem;
+  _packetType = RTCPPacketTypes::kXrReceiverReferenceTime;
+  _state = ParseState::State_XRItem;
   return true;
 }
 
@@ -997,25 +1024,25 @@ bool RTCPUtility::RTCPParserV2::ParseXrDlrr(int block_length_4bytes) {
   const int kSubBlockLengthIn4Bytes = 3;
   if (block_length_4bytes < 0 ||
       (block_length_4bytes % kSubBlockLengthIn4Bytes) != 0) {
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
     EndCurrentBlock();
     return false;
   }
-  _packetType = kRtcpXrDlrrReportBlockCode;
-  _state = State_XR_DLLRItem;
+  _packetType = RTCPPacketTypes::kXrDlrrReportBlock;
+  _state = ParseState::State_XR_DLLRItem;
   _numberOfBlocks = block_length_4bytes / kSubBlockLengthIn4Bytes;
   return true;
 }
 
 bool RTCPUtility::RTCPParserV2::ParseXrDlrrItem() {
   if (_numberOfBlocks == 0) {
-    _state = State_XRItem;
+    _state = ParseState::State_XRItem;
     return false;
   }
   const int kSubBlockLengthInBytes = 12;
   const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
   if (length < kSubBlockLengthInBytes) {
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
     EndCurrentBlock();
     return false;
   }
@@ -1035,9 +1062,9 @@ bool RTCPUtility::RTCPParserV2::ParseXrDlrrItem() {
   _packet.XRDLRRReportBlockItem.DelayLastRR += *_ptrRTCPData++ << 8;
   _packet.XRDLRRReportBlockItem.DelayLastRR += *_ptrRTCPData++;
 
-  _packetType = kRtcpXrDlrrReportBlockItemCode;
+  _packetType = RTCPPacketTypes::kXrDlrrReportBlockItem;
   --_numberOfBlocks;
-  _state = State_XR_DLLRItem;
+  _state = ParseState::State_XR_DLLRItem;
   return true;
 }
 /*  VoIP Metrics Report Block.
@@ -1070,7 +1097,7 @@ bool RTCPUtility::RTCPParserV2::ParseXrVoipMetricItem(int block_length_4bytes) {
   const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
   if (block_length_4bytes != kBlockLengthIn4Bytes ||
       length < kBlockLengthInBytes) {
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
     EndCurrentBlock();
     return false;
   }
@@ -1117,8 +1144,8 @@ bool RTCPUtility::RTCPParserV2::ParseXrVoipMetricItem(int block_length_4bytes) {
   _packet.XRVOIPMetricItem.JBabsMax = *_ptrRTCPData++ << 8;
   _packet.XRVOIPMetricItem.JBabsMax += *_ptrRTCPData++;
 
-  _packetType = kRtcpXrVoipMetricCode;
-  _state = State_XRItem;
+  _packetType = RTCPPacketTypes::kXrVoipMetric;
+  _state = ParseState::State_XRItem;
   return true;
 }
 
@@ -1127,83 +1154,72 @@ bool RTCPUtility::RTCPParserV2::ParseXrUnsupportedBlockType(
   const int32_t kBlockLengthInBytes = block_length_4bytes * 4;
   const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
   if (length < kBlockLengthInBytes) {
-    _state = State_TopLevel;
+    _state = ParseState::State_TopLevel;
     EndCurrentBlock();
     return false;
   }
   // Skip block.
   _ptrRTCPData += kBlockLengthInBytes;
-  _state = State_XRItem;
+  _state = ParseState::State_XRItem;
   return false;
 }
 
-bool
-RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header)
-{
-    assert((header.PT == PT_RTPFB) || (header.PT == PT_PSFB)); // Parser logic check
+bool RTCPUtility::RTCPParserV2::ParseFBCommon(const RtcpCommonHeader& header) {
+  RTC_CHECK((header.packet_type == PT_RTPFB) ||
+            (header.packet_type == PT_PSFB));  // Parser logic check
 
     const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
 
-    if (length < 12) // 4 * 3, RFC4585 section 6.1
-    {
-        EndCurrentBlock();
+    // 4 * 3, RFC4585 section 6.1
+    if (length < 12) {
+      LOG(LS_WARNING)
+          << "Invalid RTCP packet: Too little data (" << length
+          << " bytes) left in buffer to parse a 12 byte RTPFB/PSFB message.";
         return false;
     }
 
     _ptrRTCPData += 4; // Skip RTCP header
 
-    uint32_t senderSSRC = *_ptrRTCPData++ << 24;
-    senderSSRC += *_ptrRTCPData++ << 16;
-    senderSSRC += *_ptrRTCPData++ << 8;
-    senderSSRC += *_ptrRTCPData++;
+    uint32_t senderSSRC = ByteReader<uint32_t>::ReadBigEndian(_ptrRTCPData);
+    _ptrRTCPData += 4;
 
-    uint32_t mediaSSRC = *_ptrRTCPData++ << 24;
-    mediaSSRC += *_ptrRTCPData++ << 16;
-    mediaSSRC += *_ptrRTCPData++ << 8;
-    mediaSSRC += *_ptrRTCPData++;
+    uint32_t mediaSSRC = ByteReader<uint32_t>::ReadBigEndian(_ptrRTCPData);
+    _ptrRTCPData += 4;
 
-    if (header.PT == PT_RTPFB)
-    {
+    if (header.packet_type == PT_RTPFB) {
         // Transport layer feedback
 
-        switch (header.IC)
-        {
+        switch (header.count_or_format) {
         case 1:
         {
             // NACK
-            _packetType             = kRtcpRtpfbNackCode;
+          _packetType = RTCPPacketTypes::kRtpfbNack;
             _packet.NACK.SenderSSRC = senderSSRC;
             _packet.NACK.MediaSSRC  = mediaSSRC;
 
-            _state = State_RTPFB_NACKItem;
+            _state = ParseState::State_RTPFB_NACKItem;
 
             return true;
         }
-        case 2:
-        {
-            // used to be ACK is this code point, which is removed
-            // conficts with http://tools.ietf.org/html/draft-levin-avt-rtcp-burst-00
-            break;
-        }
         case 3:
         {
             // TMMBR
-            _packetType              = kRtcpRtpfbTmmbrCode;
+          _packetType = RTCPPacketTypes::kRtpfbTmmbr;
             _packet.TMMBR.SenderSSRC = senderSSRC;
             _packet.TMMBR.MediaSSRC  = mediaSSRC;
 
-            _state = State_RTPFB_TMMBRItem;
+            _state = ParseState::State_RTPFB_TMMBRItem;
 
             return true;
         }
         case 4:
         {
             // TMMBN
-            _packetType              = kRtcpRtpfbTmmbnCode;
+          _packetType = RTCPPacketTypes::kRtpfbTmmbn;
             _packet.TMMBN.SenderSSRC = senderSSRC;
             _packet.TMMBN.MediaSSRC  = mediaSSRC;
 
-            _state = State_RTPFB_TMMBNItem;
+            _state = ParseState::State_RTPFB_TMMBNItem;
 
             return true;
         }
@@ -1212,25 +1228,35 @@ RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header)
             // RTCP-SR-REQ Rapid Synchronisation of RTP Flows
             // draft-perkins-avt-rapid-rtp-sync-03.txt
             // trigger a new RTCP SR
-            _packetType = kRtcpRtpfbSrReqCode;
+          _packetType = RTCPPacketTypes::kRtpfbSrReq;
 
             // Note: No state transition, SR REQ is empty!
             return true;
         }
+        case 15: {
+          rtcp_packet_ =
+              rtcp::TransportFeedback::ParseFrom(_ptrRTCPData - 12, length);
+          // Since we parse the whole packet here, keep the TopLevel state and
+          // just end the current block.
+          EndCurrentBlock();
+          if (rtcp_packet_.get()) {
+            _packetType = RTCPPacketTypes::kTransportFeedback;
+            return true;
+          }
+          break;
+        }
         default:
             break;
         }
-        EndCurrentBlock();
+        // Unsupported RTPFB message. Skip and move to next block.
+        ++num_skipped_blocks_;
         return false;
-    }
-    else if (header.PT == PT_PSFB)
-    {
+    } else if (header.packet_type == PT_PSFB) {
         // Payload specific feedback
-        switch (header.IC)
-        {
+        switch (header.count_or_format) {
         case 1:
             // PLI
-            _packetType            = kRtcpPsfbPliCode;
+          _packetType = RTCPPacketTypes::kPsfbPli;
             _packet.PLI.SenderSSRC = senderSSRC;
             _packet.PLI.MediaSSRC  = mediaSSRC;
 
@@ -1238,47 +1264,44 @@ RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header)
             return true;
         case 2:
             // SLI
-            _packetType            = kRtcpPsfbSliCode;
+          _packetType = RTCPPacketTypes::kPsfbSli;
             _packet.SLI.SenderSSRC = senderSSRC;
             _packet.SLI.MediaSSRC  = mediaSSRC;
 
-            _state = State_PSFB_SLIItem;
+            _state = ParseState::State_PSFB_SLIItem;
 
             return true;
         case 3:
-            _packetType             = kRtcpPsfbRpsiCode;
+          _packetType = RTCPPacketTypes::kPsfbRpsi;
             _packet.RPSI.SenderSSRC = senderSSRC;
             _packet.RPSI.MediaSSRC  = mediaSSRC;
 
-            _state = State_PSFB_RPSIItem;
+            _state = ParseState::State_PSFB_RPSIItem;
             return true;
         case 4:
             // FIR
-            _packetType            = kRtcpPsfbFirCode;
+          _packetType = RTCPPacketTypes::kPsfbFir;
             _packet.FIR.SenderSSRC = senderSSRC;
             _packet.FIR.MediaSSRC  = mediaSSRC;
 
-            _state = State_PSFB_FIRItem;
+            _state = ParseState::State_PSFB_FIRItem;
             return true;
         case 15:
-            _packetType                = kRtcpPsfbAppCode;
+          _packetType = RTCPPacketTypes::kPsfbApp;
             _packet.PSFBAPP.SenderSSRC = senderSSRC;
             _packet.PSFBAPP.MediaSSRC  = mediaSSRC;
 
-            _state = State_PSFB_AppItem;
+            _state = ParseState::State_PSFB_AppItem;
             return true;
         default:
             break;
         }
 
-        EndCurrentBlock();
         return false;
     }
     else
     {
-        assert(false);
-
-        EndCurrentBlock();
+      RTC_NOTREACHED();
         return false;
     }
 }
@@ -1298,19 +1321,19 @@ bool RTCPUtility::RTCPParserV2::ParseRPSIItem() {
     const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
 
     if (length < 4) {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
     if (length > 2 + RTCP_RPSI_DATA_SIZE) {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpPsfbRpsiCode;
+    _packetType = RTCPPacketTypes::kPsfbRpsi;
 
     uint8_t padding_bits = *_ptrRTCPData++;
     _packet.RPSI.PayloadType = *_ptrRTCPData++;
@@ -1332,13 +1355,13 @@ RTCPUtility::RTCPParserV2::ParseNACKItem()
 
     if (length < 4)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpRtpfbNackItemCode;
+    _packetType = RTCPPacketTypes::kRtpfbNackItem;
 
     _packet.NACKItem.PacketID = *_ptrRTCPData++ << 8;
     _packet.NACKItem.PacketID += *_ptrRTCPData++;
@@ -1356,41 +1379,41 @@ RTCPUtility::RTCPParserV2::ParsePsfbAppItem()
 
     if (length < 4)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
     if(*_ptrRTCPData++ != 'R')
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
     if(*_ptrRTCPData++ != 'E')
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
     if(*_ptrRTCPData++ != 'M')
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
     if(*_ptrRTCPData++ != 'B')
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
-    _packetType = kRtcpPsfbRembCode;
-    _state = State_PSFB_REMBItem;
+    _packetType = RTCPPacketTypes::kPsfbRemb;
+    _state = ParseState::State_PSFB_REMBItem;
     return true;
 }
 
@@ -1401,7 +1424,7 @@ RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
 
     if (length < 4)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
@@ -1420,13 +1443,13 @@ RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
     const ptrdiff_t length_ssrcs = _ptrRTCPBlockEnd - _ptrRTCPData;
     if (length_ssrcs < 4 * _packet.REMBItem.NumberOfSSRCs)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpPsfbRembItemCode;
+    _packetType = RTCPPacketTypes::kPsfbRembItem;
 
     for (int i = 0; i < _packet.REMBItem.NumberOfSSRCs; i++)
     {
@@ -1447,13 +1470,13 @@ RTCPUtility::RTCPParserV2::ParseTMMBRItem()
 
     if (length < 8)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpRtpfbTmmbrItemCode;
+    _packetType = RTCPPacketTypes::kRtpfbTmmbrItem;
 
     _packet.TMMBRItem.SSRC = *_ptrRTCPData++ << 24;
     _packet.TMMBRItem.SSRC += *_ptrRTCPData++ << 16;
@@ -1486,13 +1509,13 @@ RTCPUtility::RTCPParserV2::ParseTMMBNItem()
 
     if (length < 8)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpRtpfbTmmbnItemCode;
+    _packetType = RTCPPacketTypes::kRtpfbTmmbnItem;
 
     _packet.TMMBNItem.SSRC = *_ptrRTCPData++ << 24;
     _packet.TMMBNItem.SSRC += *_ptrRTCPData++ << 16;
@@ -1532,12 +1555,12 @@ RTCPUtility::RTCPParserV2::ParseSLIItem()
 
     if (length < 4)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
-    _packetType = kRtcpPsfbSliItemCode;
+    _packetType = RTCPPacketTypes::kPsfbSliItem;
 
     uint32_t buffer;
     buffer = *_ptrRTCPData++ << 24;
@@ -1561,13 +1584,13 @@ RTCPUtility::RTCPParserV2::ParseFIRItem()
 
     if (length < 8)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
 
-    _packetType = kRtcpPsfbFirItemCode;
+    _packetType = RTCPPacketTypes::kPsfbFirItem;
 
     _packet.FIRItem.SSRC = *_ptrRTCPData++ << 24;
     _packet.FIRItem.SSRC += *_ptrRTCPData++ << 16;
@@ -1579,9 +1602,7 @@ RTCPUtility::RTCPParserV2::ParseFIRItem()
     return true;
 }
 
-bool
-RTCPUtility::RTCPParserV2::ParseAPP( const RTCPCommonHeader& header)
-{
+bool RTCPUtility::RTCPParserV2::ParseAPP(const RtcpCommonHeader& header) {
     ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
 
     if (length < 12) // 4 * 3, RFC 3550 6.7 APP: Application-Defined RTCP Packet
@@ -1604,12 +1625,12 @@ RTCPUtility::RTCPParserV2::ParseAPP( const RTCPCommonHeader& header)
 
     length  = _ptrRTCPBlockEnd - _ptrRTCPData;
 
-    _packetType = kRtcpAppCode;
+    _packetType = RTCPPacketTypes::kApp;
 
-    _packet.APP.SubType = header.IC;
+    _packet.APP.SubType = header.count_or_format;
     _packet.APP.Name = name;
 
-    _state = State_AppItem;
+    _state = ParseState::State_AppItem;
     return true;
 }
 
@@ -1619,12 +1640,12 @@ RTCPUtility::RTCPParserV2::ParseAPPItem()
     const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
     if (length < 4)
     {
-        _state = State_TopLevel;
+      _state = ParseState::State_TopLevel;
 
         EndCurrentBlock();
         return false;
     }
-    _packetType = kRtcpAppItemCode;
+    _packetType = RTCPPacketTypes::kAppItem;
 
     if(length > kRtcpAppCode_DATA_SIZE)
     {
@@ -1640,6 +1661,10 @@ RTCPUtility::RTCPParserV2::ParseAPPItem()
     return true;
 }
 
+size_t RTCPUtility::RTCPParserV2::NumSkippedBlocks() const {
+  return num_skipped_blocks_;
+}
+
 RTCPUtility::RTCPPacketIterator::RTCPPacketIterator(uint8_t* rtcpData,
                                                     size_t rtcpDataLength)
     : _ptrBegin(rtcpData),
@@ -1651,37 +1676,31 @@ RTCPUtility::RTCPPacketIterator::RTCPPacketIterator(uint8_t* rtcpData,
 RTCPUtility::RTCPPacketIterator::~RTCPPacketIterator() {
 }
 
-const RTCPUtility::RTCPCommonHeader*
-RTCPUtility::RTCPPacketIterator::Begin()
-{
+const RTCPUtility::RtcpCommonHeader* RTCPUtility::RTCPPacketIterator::Begin() {
     _ptrBlock = _ptrBegin;
 
     return Iterate();
 }
 
-const RTCPUtility::RTCPCommonHeader*
-RTCPUtility::RTCPPacketIterator::Iterate()
-{
-    const bool success = RTCPParseCommonHeader(_ptrBlock, _ptrEnd, _header);
-    if (!success)
-    {
-        _ptrBlock = NULL;
-        return NULL;
-    }
-    _ptrBlock += _header.LengthInOctets;
+const RTCPUtility::RtcpCommonHeader*
+RTCPUtility::RTCPPacketIterator::Iterate() {
+  if ((_ptrEnd <= _ptrBlock) ||
+      !RtcpParseCommonHeader(_ptrBlock, _ptrEnd - _ptrBlock, &_header)) {
+    _ptrBlock = nullptr;
+    return nullptr;
+  }
+  _ptrBlock += _header.BlockSize();
 
-    if (_ptrBlock > _ptrEnd)
-    {
-        _ptrBlock = NULL;
-        return  NULL;
-    }
+  if (_ptrBlock > _ptrEnd) {
+    _ptrBlock = nullptr;
+    return nullptr;
+  }
 
-    return &_header;
+  return &_header;
 }
 
-const RTCPUtility::RTCPCommonHeader*
-RTCPUtility::RTCPPacketIterator::Current()
-{
+const RTCPUtility::RtcpCommonHeader*
+RTCPUtility::RTCPPacketIterator::Current() {
     if (!_ptrBlock)
     {
         return NULL;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
index 804e8b9478..0b03ceb56e 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
@@ -1,23 +1,27 @@
 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
+*  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+*
+*  Use of this source code is governed by a BSD-style license
+*  that can be found in the LICENSE file in the root of the source
+*  tree. An additional intellectual property rights grant can be found
+*  in the file PATENTS.  All contributing project authors may
+*  be found in the AUTHORS file in the root of the source tree.
+*/
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
 
 #include <stddef.h> // size_t, ptrdiff_t
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace rtcp {
+class RtcpPacket;
+}
 namespace RTCPUtility {
 
 class NackStats {
@@ -42,463 +46,447 @@ class NackStats {
   uint32_t unique_requests_;
 };
 
-    uint32_t MidNtp(uint32_t ntp_sec, uint32_t ntp_frac);
+uint32_t MidNtp(uint32_t ntp_sec, uint32_t ntp_frac);
 
-    // CNAME
-    struct RTCPCnameInformation
-    {
-        char name[RTCP_CNAME_SIZE];
-    };
-    struct RTCPPacketRR
-    {
-        uint32_t SenderSSRC;
-        uint8_t  NumberOfReportBlocks;
-    };
-    struct RTCPPacketSR
-    {
-        uint32_t SenderSSRC;
-        uint8_t  NumberOfReportBlocks;
+// CNAME
+struct RTCPCnameInformation {
+  char name[RTCP_CNAME_SIZE];
+};
+struct RTCPPacketRR {
+  uint32_t SenderSSRC;
+  uint8_t NumberOfReportBlocks;
+};
+struct RTCPPacketSR {
+  uint32_t SenderSSRC;
+  uint8_t NumberOfReportBlocks;
 
-        // sender info
-        uint32_t NTPMostSignificant;
-        uint32_t NTPLeastSignificant;
-        uint32_t RTPTimestamp;
-        uint32_t SenderPacketCount;
-        uint32_t SenderOctetCount;
-    };
-    struct RTCPPacketReportBlockItem
-    {
-        // report block
-        uint32_t SSRC;
-        uint8_t  FractionLost;
-        uint32_t CumulativeNumOfPacketsLost;
-        uint32_t ExtendedHighestSequenceNumber;
-        uint32_t Jitter;
-        uint32_t LastSR;
-        uint32_t DelayLastSR;
-    };
-    struct RTCPPacketSDESCName
-    {
-        // RFC3550
-        uint32_t SenderSSRC;
-        char CName[RTCP_CNAME_SIZE];
-    };
+  // sender info
+  uint32_t NTPMostSignificant;
+  uint32_t NTPLeastSignificant;
+  uint32_t RTPTimestamp;
+  uint32_t SenderPacketCount;
+  uint32_t SenderOctetCount;
+};
+struct RTCPPacketReportBlockItem {
+  // report block
+  uint32_t SSRC;
+  uint8_t FractionLost;
+  uint32_t CumulativeNumOfPacketsLost;
+  uint32_t ExtendedHighestSequenceNumber;
+  uint32_t Jitter;
+  uint32_t LastSR;
+  uint32_t DelayLastSR;
+};
+struct RTCPPacketSDESCName {
+  // RFC3550
+  uint32_t SenderSSRC;
+  char CName[RTCP_CNAME_SIZE];
+};
 
-    struct RTCPPacketExtendedJitterReportItem
-    {
-        // RFC 5450
-        uint32_t Jitter;
-    };
+struct RTCPPacketExtendedJitterReportItem {
+  // RFC 5450
+  uint32_t Jitter;
+};
 
-    struct RTCPPacketBYE
-    {
-        uint32_t SenderSSRC;
-    };
-    struct RTCPPacketXR
-    {
-        // RFC 3611
-        uint32_t OriginatorSSRC;
-    };
-    struct RTCPPacketXRReceiverReferenceTimeItem
-    {
-        // RFC 3611 4.4
-        uint32_t NTPMostSignificant;
-        uint32_t NTPLeastSignificant;
-    };
-    struct RTCPPacketXRDLRRReportBlockItem
-    {
-        // RFC 3611 4.5
-        uint32_t SSRC;
-        uint32_t LastRR;
-        uint32_t DelayLastRR;
-    };
-    struct RTCPPacketXRVOIPMetricItem
-    {
-        // RFC 3611 4.7
-        uint32_t    SSRC;
-        uint8_t     lossRate;
-        uint8_t     discardRate;
-        uint8_t     burstDensity;
-        uint8_t     gapDensity;
-        uint16_t    burstDuration;
-        uint16_t    gapDuration;
-        uint16_t    roundTripDelay;
-        uint16_t    endSystemDelay;
-        uint8_t     signalLevel;
-        uint8_t     noiseLevel;
-        uint8_t     RERL;
-        uint8_t     Gmin;
-        uint8_t     Rfactor;
-        uint8_t     extRfactor;
-        uint8_t     MOSLQ;
-        uint8_t     MOSCQ;
-        uint8_t     RXconfig;
-        uint16_t    JBnominal;
-        uint16_t    JBmax;
-        uint16_t    JBabsMax;
-    };
+struct RTCPPacketBYE {
+  uint32_t SenderSSRC;
+};
+struct RTCPPacketXR {
+  // RFC 3611
+  uint32_t OriginatorSSRC;
+};
+struct RTCPPacketXRReceiverReferenceTimeItem {
+  // RFC 3611 4.4
+  uint32_t NTPMostSignificant;
+  uint32_t NTPLeastSignificant;
+};
+struct RTCPPacketXRDLRRReportBlockItem {
+  // RFC 3611 4.5
+  uint32_t SSRC;
+  uint32_t LastRR;
+  uint32_t DelayLastRR;
+};
+struct RTCPPacketXRVOIPMetricItem {
+  // RFC 3611 4.7
+  uint32_t SSRC;
+  uint8_t lossRate;
+  uint8_t discardRate;
+  uint8_t burstDensity;
+  uint8_t gapDensity;
+  uint16_t burstDuration;
+  uint16_t gapDuration;
+  uint16_t roundTripDelay;
+  uint16_t endSystemDelay;
+  uint8_t signalLevel;
+  uint8_t noiseLevel;
+  uint8_t RERL;
+  uint8_t Gmin;
+  uint8_t Rfactor;
+  uint8_t extRfactor;
+  uint8_t MOSLQ;
+  uint8_t MOSCQ;
+  uint8_t RXconfig;
+  uint16_t JBnominal;
+  uint16_t JBmax;
+  uint16_t JBabsMax;
+};
 
-    struct RTCPPacketRTPFBNACK
-    {
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC;
-    };
-    struct RTCPPacketRTPFBNACKItem
-    {
-        // RFC4585
-        uint16_t PacketID;
-        uint16_t BitMask;
-    };
+struct RTCPPacketRTPFBNACK {
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;
+};
+struct RTCPPacketRTPFBNACKItem {
+  // RFC4585
+  uint16_t PacketID;
+  uint16_t BitMask;
+};
 
-    struct RTCPPacketRTPFBTMMBR
-    {
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC; // zero!
-    };
-    struct RTCPPacketRTPFBTMMBRItem
-    {
-        // RFC5104
-        uint32_t SSRC;
-        uint32_t MaxTotalMediaBitRate; // In Kbit/s
-        uint32_t MeasuredOverhead;
-    };
+struct RTCPPacketRTPFBTMMBR {
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;  // zero!
+};
+struct RTCPPacketRTPFBTMMBRItem {
+  // RFC5104
+  uint32_t SSRC;
+  uint32_t MaxTotalMediaBitRate;  // In Kbit/s
+  uint32_t MeasuredOverhead;
+};
 
-    struct RTCPPacketRTPFBTMMBN
-    {
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC; // zero!
-    };
-    struct RTCPPacketRTPFBTMMBNItem
-    {
-        // RFC5104
-        uint32_t SSRC; // "Owner"
-        uint32_t MaxTotalMediaBitRate;
-        uint32_t MeasuredOverhead;
-    };
+struct RTCPPacketRTPFBTMMBN {
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;  // zero!
+};
+struct RTCPPacketRTPFBTMMBNItem {
+  // RFC5104
+  uint32_t SSRC;  // "Owner"
+  uint32_t MaxTotalMediaBitRate;
+  uint32_t MeasuredOverhead;
+};
 
-    struct RTCPPacketPSFBFIR
-    {
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC; // zero!
-    };
-    struct RTCPPacketPSFBFIRItem
-    {
-        // RFC5104
-        uint32_t SSRC;
-        uint8_t  CommandSequenceNumber;
-    };
+struct RTCPPacketPSFBFIR {
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;  // zero!
+};
+struct RTCPPacketPSFBFIRItem {
+  // RFC5104
+  uint32_t SSRC;
+  uint8_t CommandSequenceNumber;
+};
 
-    struct RTCPPacketPSFBPLI
-    {
-        // RFC4585
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC;
-    };
+struct RTCPPacketPSFBPLI {
+  // RFC4585
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;
+};
 
-    struct RTCPPacketPSFBSLI
-    {
-        // RFC4585
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC;
-    };
-    struct RTCPPacketPSFBSLIItem
-    {
-        // RFC4585
-        uint16_t FirstMB;
-        uint16_t NumberOfMB;
-        uint8_t PictureId;
-    };
-    struct RTCPPacketPSFBRPSI
-    {
-        // RFC4585
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC;
-        uint8_t  PayloadType;
-        uint16_t NumberOfValidBits;
-        uint8_t  NativeBitString[RTCP_RPSI_DATA_SIZE];
-    };
-    struct RTCPPacketPSFBAPP
-    {
-        uint32_t SenderSSRC;
-        uint32_t MediaSSRC;
-    };
-    struct RTCPPacketPSFBREMBItem
-    {
-        uint32_t BitRate;
-        uint8_t NumberOfSSRCs;
-        uint32_t SSRCs[MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS];
-    };
-    // generic name APP
-    struct RTCPPacketAPP
-    {
-        uint8_t     SubType;
-        uint32_t    Name;
-        uint8_t     Data[kRtcpAppCode_DATA_SIZE];
-        uint16_t    Size;
-    };
+struct RTCPPacketPSFBSLI {
+  // RFC4585
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;
+};
+struct RTCPPacketPSFBSLIItem {
+  // RFC4585
+  uint16_t FirstMB;
+  uint16_t NumberOfMB;
+  uint8_t PictureId;
+};
+struct RTCPPacketPSFBRPSI {
+  // RFC4585
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;
+  uint8_t PayloadType;
+  uint16_t NumberOfValidBits;
+  uint8_t NativeBitString[RTCP_RPSI_DATA_SIZE];
+};
+struct RTCPPacketPSFBAPP {
+  uint32_t SenderSSRC;
+  uint32_t MediaSSRC;
+};
+struct RTCPPacketPSFBREMBItem {
+  uint32_t BitRate;
+  uint8_t NumberOfSSRCs;
+  uint32_t SSRCs[MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS];
+};
+// generic name APP
+struct RTCPPacketAPP {
+  uint8_t SubType;
+  uint32_t Name;
+  uint8_t Data[kRtcpAppCode_DATA_SIZE];
+  uint16_t Size;
+};
 
-    union RTCPPacket
-    {
-        RTCPPacketRR              RR;
-        RTCPPacketSR              SR;
-        RTCPPacketReportBlockItem ReportBlockItem;
+union RTCPPacket {
+  RTCPPacketRR RR;
+  RTCPPacketSR SR;
+  RTCPPacketReportBlockItem ReportBlockItem;
 
-        RTCPPacketSDESCName       CName;
-        RTCPPacketBYE             BYE;
+  RTCPPacketSDESCName CName;
+  RTCPPacketBYE BYE;
 
-        RTCPPacketExtendedJitterReportItem ExtendedJitterReportItem;
+  RTCPPacketExtendedJitterReportItem ExtendedJitterReportItem;
 
-        RTCPPacketRTPFBNACK       NACK;
-        RTCPPacketRTPFBNACKItem   NACKItem;
+  RTCPPacketRTPFBNACK NACK;
+  RTCPPacketRTPFBNACKItem NACKItem;
 
-        RTCPPacketPSFBPLI         PLI;
-        RTCPPacketPSFBSLI         SLI;
-        RTCPPacketPSFBSLIItem     SLIItem;
-        RTCPPacketPSFBRPSI        RPSI;
-        RTCPPacketPSFBAPP         PSFBAPP;
-        RTCPPacketPSFBREMBItem    REMBItem;
+  RTCPPacketPSFBPLI PLI;
+  RTCPPacketPSFBSLI SLI;
+  RTCPPacketPSFBSLIItem SLIItem;
+  RTCPPacketPSFBRPSI RPSI;
+  RTCPPacketPSFBAPP PSFBAPP;
+  RTCPPacketPSFBREMBItem REMBItem;
 
-        RTCPPacketRTPFBTMMBR      TMMBR;
-        RTCPPacketRTPFBTMMBRItem  TMMBRItem;
-        RTCPPacketRTPFBTMMBN      TMMBN;
-        RTCPPacketRTPFBTMMBNItem  TMMBNItem;
-        RTCPPacketPSFBFIR         FIR;
-        RTCPPacketPSFBFIRItem     FIRItem;
+  RTCPPacketRTPFBTMMBR TMMBR;
+  RTCPPacketRTPFBTMMBRItem TMMBRItem;
+  RTCPPacketRTPFBTMMBN TMMBN;
+  RTCPPacketRTPFBTMMBNItem TMMBNItem;
+  RTCPPacketPSFBFIR FIR;
+  RTCPPacketPSFBFIRItem FIRItem;
 
-        RTCPPacketXR               XR;
-        RTCPPacketXRReceiverReferenceTimeItem XRReceiverReferenceTimeItem;
-        RTCPPacketXRDLRRReportBlockItem XRDLRRReportBlockItem;
-        RTCPPacketXRVOIPMetricItem XRVOIPMetricItem;
+  RTCPPacketXR XR;
+  RTCPPacketXRReceiverReferenceTimeItem XRReceiverReferenceTimeItem;
+  RTCPPacketXRDLRRReportBlockItem XRDLRRReportBlockItem;
+  RTCPPacketXRVOIPMetricItem XRVOIPMetricItem;
 
-        RTCPPacketAPP             APP;
-    };
+  RTCPPacketAPP APP;
+};
 
-    enum RTCPPacketTypes
-    {
-        kRtcpNotValidCode,
+enum class RTCPPacketTypes {
+  kInvalid,
 
-        // RFC3550
-        kRtcpRrCode,
-        kRtcpSrCode,
-        kRtcpReportBlockItemCode,
+  // RFC3550
+  kRr,
+  kSr,
+  kReportBlockItem,
 
-        kRtcpSdesCode,
-        kRtcpSdesChunkCode,
-        kRtcpByeCode,
+  kSdes,
+  kSdesChunk,
+  kBye,
 
-        // RFC5450
-        kRtcpExtendedIjCode,
-        kRtcpExtendedIjItemCode,
+  // RFC5450
+  kExtendedIj,
+  kExtendedIjItem,
 
-        // RFC4585
-        kRtcpRtpfbNackCode,
-        kRtcpRtpfbNackItemCode,
+  // RFC4585
+  kRtpfbNack,
+  kRtpfbNackItem,
 
-        kRtcpPsfbPliCode,
-        kRtcpPsfbRpsiCode,
-        kRtcpPsfbSliCode,
-        kRtcpPsfbSliItemCode,
-        kRtcpPsfbAppCode,
-        kRtcpPsfbRembCode,
-        kRtcpPsfbRembItemCode,
+  kPsfbPli,
+  kPsfbRpsi,
+  kPsfbSli,
+  kPsfbSliItem,
+  kPsfbApp,
+  kPsfbRemb,
+  kPsfbRembItem,
 
-        // RFC5104
-        kRtcpRtpfbTmmbrCode,
-        kRtcpRtpfbTmmbrItemCode,
-        kRtcpRtpfbTmmbnCode,
-        kRtcpRtpfbTmmbnItemCode,
-        kRtcpPsfbFirCode,
-        kRtcpPsfbFirItemCode,
+  // RFC5104
+  kRtpfbTmmbr,
+  kRtpfbTmmbrItem,
+  kRtpfbTmmbn,
+  kRtpfbTmmbnItem,
+  kPsfbFir,
+  kPsfbFirItem,
 
-        // draft-perkins-avt-rapid-rtp-sync
-        kRtcpRtpfbSrReqCode,
+  // draft-perkins-avt-rapid-rtp-sync
+  kRtpfbSrReq,
 
-        // RFC 3611
-        kRtcpXrHeaderCode,
-        kRtcpXrReceiverReferenceTimeCode,
-        kRtcpXrDlrrReportBlockCode,
-        kRtcpXrDlrrReportBlockItemCode,
-        kRtcpXrVoipMetricCode,
+  // RFC 3611
+  kXrHeader,
+  kXrReceiverReferenceTime,
+  kXrDlrrReportBlock,
+  kXrDlrrReportBlockItem,
+  kXrVoipMetric,
 
-        kRtcpAppCode,
-        kRtcpAppItemCode,
-    };
+  kApp,
+  kAppItem,
 
-    struct RTCPRawPacket
-    {
-        const uint8_t* _ptrPacketBegin;
-        const uint8_t* _ptrPacketEnd;
-    };
+  // draft-holmer-rmcat-transport-wide-cc-extensions
+  kTransportFeedback,
+};
 
-    struct RTCPModRawPacket
-    {
-        uint8_t* _ptrPacketBegin;
-        uint8_t* _ptrPacketEnd;
-    };
+struct RTCPRawPacket {
+  const uint8_t* _ptrPacketBegin;
+  const uint8_t* _ptrPacketEnd;
+};
 
-    struct RTCPCommonHeader
-    {
-        uint8_t  V;  // Version
-        bool           P;  // Padding
-        uint8_t  IC; // Item count/subtype
-        uint8_t  PT; // Packet Type
-        uint16_t LengthInOctets;
-    };
+struct RTCPModRawPacket {
+  uint8_t* _ptrPacketBegin;
+  uint8_t* _ptrPacketEnd;
+};
 
-    enum RTCPPT
-    {
-        PT_IJ    = 195,
-        PT_SR    = 200,
-        PT_RR    = 201,
-        PT_SDES  = 202,
-        PT_BYE   = 203,
-        PT_APP   = 204,
-        PT_RTPFB = 205,
-        PT_PSFB  = 206,
-        PT_XR    = 207
-    };
+struct RtcpCommonHeader {
+  static const uint8_t kHeaderSizeBytes = 4;
+  RtcpCommonHeader()
+      : version(2),
+        count_or_format(0),
+        packet_type(0),
+        payload_size_bytes(0),
+        padding_bytes(0) {}
 
-    // Extended report blocks, RFC 3611.
-    enum RtcpXrBlockType {
-      kBtReceiverReferenceTime = 4,
-      kBtDlrr = 5,
-      kBtVoipMetric = 7
-    };
+  uint32_t BlockSize() const {
+    return kHeaderSizeBytes + payload_size_bytes + padding_bytes;
+  }
 
-    bool RTCPParseCommonHeader( const uint8_t* ptrDataBegin,
-                                const uint8_t* ptrDataEnd,
-                                RTCPCommonHeader& parsedHeader);
+  uint8_t version;
+  uint8_t count_or_format;
+  uint8_t packet_type;
+  uint32_t payload_size_bytes;
+  uint8_t padding_bytes;
+};
 
-    class RTCPParserV2
-    {
-    public:
-        RTCPParserV2(const uint8_t* rtcpData,
-                     size_t rtcpDataLength,
-                     bool rtcpReducedSizeEnable); // Set to true, to allow non-compound RTCP!
-        ~RTCPParserV2();
+enum RTCPPT : uint8_t {
+  PT_IJ = 195,
+  PT_SR = 200,
+  PT_RR = 201,
+  PT_SDES = 202,
+  PT_BYE = 203,
+  PT_APP = 204,
+  PT_RTPFB = 205,
+  PT_PSFB = 206,
+  PT_XR = 207
+};
 
-        RTCPPacketTypes PacketType() const;
-        const RTCPPacket& Packet() const;
-        const RTCPRawPacket& RawPacket() const;
-        ptrdiff_t LengthLeft() const;
+// Extended report blocks, RFC 3611.
+enum RtcpXrBlockType : uint8_t {
+  kBtReceiverReferenceTime = 4,
+  kBtDlrr = 5,
+  kBtVoipMetric = 7
+};
 
-        bool IsValid() const;
+bool RtcpParseCommonHeader(const uint8_t* buffer,
+                           size_t size_bytes,
+                           RtcpCommonHeader* parsed_header);
 
-        RTCPPacketTypes Begin();
-        RTCPPacketTypes Iterate();
+class RTCPParserV2 {
+ public:
+  RTCPParserV2(
+      const uint8_t* rtcpData,
+      size_t rtcpDataLength,
+      bool rtcpReducedSizeEnable);  // Set to true, to allow non-compound RTCP!
+  ~RTCPParserV2();
 
-    private:
-        enum ParseState
-        {
-            State_TopLevel,        // Top level packet
-            State_ReportBlockItem, // SR/RR report block
-            State_SDESChunk,       // SDES chunk
-            State_BYEItem,         // BYE item
-            State_ExtendedJitterItem, // Extended jitter report item
-            State_RTPFB_NACKItem,  // NACK FCI item
-            State_RTPFB_TMMBRItem, // TMMBR FCI item
-            State_RTPFB_TMMBNItem, // TMMBN FCI item
-            State_PSFB_SLIItem,    // SLI FCI item
-            State_PSFB_RPSIItem,   // RPSI FCI item
-            State_PSFB_FIRItem,    // FIR FCI item
-            State_PSFB_AppItem,    // Application specific FCI item
-            State_PSFB_REMBItem,   // Application specific REMB item
-            State_XRItem,
-            State_XR_DLLRItem,
-            State_AppItem
-        };
+  RTCPPacketTypes PacketType() const;
+  const RTCPPacket& Packet() const;
+  rtcp::RtcpPacket* ReleaseRtcpPacket();
+  const RTCPRawPacket& RawPacket() const;
+  ptrdiff_t LengthLeft() const;
 
-    private:
-        void IterateTopLevel();
-        void IterateReportBlockItem();
-        void IterateSDESChunk();
-        void IterateBYEItem();
-        void IterateExtendedJitterItem();
-        void IterateNACKItem();
-        void IterateTMMBRItem();
-        void IterateTMMBNItem();
-        void IterateSLIItem();
-        void IterateRPSIItem();
-        void IterateFIRItem();
-        void IteratePsfbAppItem();
-        void IteratePsfbREMBItem();
-        void IterateAppItem();
-        void IterateXrItem();
-        void IterateXrDlrrItem();
+  bool IsValid() const;
+  size_t NumSkippedBlocks() const;
 
-        void Validate();
-        void EndCurrentBlock();
+  RTCPPacketTypes Begin();
+  RTCPPacketTypes Iterate();
 
-        bool ParseRR();
-        bool ParseSR();
-        bool ParseReportBlockItem();
+ private:
+  enum class ParseState {
+    State_TopLevel,            // Top level packet
+    State_ReportBlockItem,     // SR/RR report block
+    State_SDESChunk,           // SDES chunk
+    State_BYEItem,             // BYE item
+    State_ExtendedJitterItem,  // Extended jitter report item
+    State_RTPFB_NACKItem,      // NACK FCI item
+    State_RTPFB_TMMBRItem,     // TMMBR FCI item
+    State_RTPFB_TMMBNItem,     // TMMBN FCI item
+    State_PSFB_SLIItem,        // SLI FCI item
+    State_PSFB_RPSIItem,       // RPSI FCI item
+    State_PSFB_FIRItem,        // FIR FCI item
+    State_PSFB_AppItem,        // Application specific FCI item
+    State_PSFB_REMBItem,       // Application specific REMB item
+    State_XRItem,
+    State_XR_DLLRItem,
+    State_AppItem
+  };
 
-        bool ParseSDES();
-        bool ParseSDESChunk();
-        bool ParseSDESItem();
+ private:
+  void IterateTopLevel();
+  void IterateReportBlockItem();
+  void IterateSDESChunk();
+  void IterateBYEItem();
+  void IterateExtendedJitterItem();
+  void IterateNACKItem();
+  void IterateTMMBRItem();
+  void IterateTMMBNItem();
+  void IterateSLIItem();
+  void IterateRPSIItem();
+  void IterateFIRItem();
+  void IteratePsfbAppItem();
+  void IteratePsfbREMBItem();
+  void IterateAppItem();
+  void IterateXrItem();
+  void IterateXrDlrrItem();
 
-        bool ParseBYE();
-        bool ParseBYEItem();
+  void Validate();
+  void EndCurrentBlock();
 
-        bool ParseIJ();
-        bool ParseIJItem();
+  bool ParseRR();
+  bool ParseSR();
+  bool ParseReportBlockItem();
 
-        bool ParseXr();
-        bool ParseXrItem();
-        bool ParseXrReceiverReferenceTimeItem(int block_length_4bytes);
-        bool ParseXrDlrr(int block_length_4bytes);
-        bool ParseXrDlrrItem();
-        bool ParseXrVoipMetricItem(int block_length_4bytes);
-        bool ParseXrUnsupportedBlockType(int block_length_4bytes);
+  bool ParseSDES();
+  bool ParseSDESChunk();
+  bool ParseSDESItem();
 
-        bool ParseFBCommon(const RTCPCommonHeader& header);
-        bool ParseNACKItem();
-        bool ParseTMMBRItem();
-        bool ParseTMMBNItem();
-        bool ParseSLIItem();
-        bool ParseRPSIItem();
-        bool ParseFIRItem();
-        bool ParsePsfbAppItem();
-        bool ParsePsfbREMBItem();
+  bool ParseBYE();
+  bool ParseBYEItem();
 
-        bool ParseAPP(const RTCPCommonHeader& header);
-        bool ParseAPPItem();
+  bool ParseIJ();
+  bool ParseIJItem();
 
-    private:
-        const uint8_t* const _ptrRTCPDataBegin;
-        const bool                 _RTCPReducedSizeEnable;
-        const uint8_t* const _ptrRTCPDataEnd;
+  bool ParseXr();
+  bool ParseXrItem();
+  bool ParseXrReceiverReferenceTimeItem(int block_length_4bytes);
+  bool ParseXrDlrr(int block_length_4bytes);
+  bool ParseXrDlrrItem();
+  bool ParseXrVoipMetricItem(int block_length_4bytes);
+  bool ParseXrUnsupportedBlockType(int block_length_4bytes);
 
-        bool                     _validPacket;
-        const uint8_t*     _ptrRTCPData;
-        const uint8_t*     _ptrRTCPBlockEnd;
+  bool ParseFBCommon(const RtcpCommonHeader& header);
+  bool ParseNACKItem();
+  bool ParseTMMBRItem();
+  bool ParseTMMBNItem();
+  bool ParseSLIItem();
+  bool ParseRPSIItem();
+  bool ParseFIRItem();
+  bool ParsePsfbAppItem();
+  bool ParsePsfbREMBItem();
 
-        ParseState               _state;
-        uint8_t            _numberOfBlocks;
+  bool ParseAPP(const RtcpCommonHeader& header);
+  bool ParseAPPItem();
 
-        RTCPPacketTypes          _packetType;
-        RTCPPacket               _packet;
-    };
+ private:
+  const uint8_t* const _ptrRTCPDataBegin;
+  const bool _RTCPReducedSizeEnable;
+  const uint8_t* const _ptrRTCPDataEnd;
 
-    class RTCPPacketIterator
-    {
-    public:
-        RTCPPacketIterator(uint8_t* rtcpData,
-                            size_t rtcpDataLength);
-        ~RTCPPacketIterator();
+  bool _validPacket;
+  const uint8_t* _ptrRTCPData;
+  const uint8_t* _ptrRTCPBlockEnd;
 
-        const RTCPCommonHeader* Begin();
-        const RTCPCommonHeader* Iterate();
-        const RTCPCommonHeader* Current();
+  ParseState _state;
+  uint8_t _numberOfBlocks;
+  size_t num_skipped_blocks_;
 
-    private:
-        uint8_t* const     _ptrBegin;
-        uint8_t* const     _ptrEnd;
+  RTCPPacketTypes _packetType;
+  RTCPPacket _packet;
+  rtc::scoped_ptr<webrtc::rtcp::RtcpPacket> rtcp_packet_;
+};
 
-        uint8_t*           _ptrBlock;
+class RTCPPacketIterator {
+ public:
+  RTCPPacketIterator(uint8_t* rtcpData, size_t rtcpDataLength);
+  ~RTCPPacketIterator();
 
-        RTCPCommonHeader         _header;
-    };
-}  // RTCPUtility
+  const RtcpCommonHeader* Begin();
+  const RtcpCommonHeader* Iterate();
+  const RtcpCommonHeader* Current();
+
+ private:
+  uint8_t* const _ptrBegin;
+  uint8_t* const _ptrEnd;
+
+  uint8_t* _ptrBlock;
+
+  RtcpCommonHeader _header;
+};
+}  // namespace RTCPUtility
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility_unittest.cc
index 275b007bef..1a13812f02 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_utility_unittest.cc
@@ -10,10 +10,16 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
 namespace webrtc {
 
+using RTCPUtility::RtcpCommonHeader;
+
+namespace rtcp {
+
 TEST(RtcpUtilityTest, MidNtp) {
   const uint32_t kNtpSec = 0x12345678;
   const uint32_t kNtpFrac = 0x23456789;
@@ -68,5 +74,88 @@ TEST(RtcpUtilityTest, NackRequestsWithWrap) {
   EXPECT_EQ(8U, stats.requests());
 }
 
+class RtcpParseCommonHeaderTest : public ::testing::Test {
+ public:
+  RtcpParseCommonHeaderTest() { memset(buffer, 0, kBufferCapacityBytes); }
+  virtual ~RtcpParseCommonHeaderTest() {}
+
+ protected:
+  static const size_t kBufferCapacityBytes = 40;
+  uint8_t buffer[kBufferCapacityBytes];
+  RtcpCommonHeader header;
+};
+
+TEST_F(RtcpParseCommonHeaderTest, TooSmallBuffer) {
+  // Buffer needs to be able to hold the header.
+  for (size_t i = 0; i < RtcpCommonHeader::kHeaderSizeBytes; ++i)
+    EXPECT_FALSE(RtcpParseCommonHeader(buffer, i, &header));
+}
+
+TEST_F(RtcpParseCommonHeaderTest, Version) {
+  // Version 2 is the only allowed for now.
+  for (int v = 0; v < 4; ++v) {
+    buffer[0] = v << 6;
+    EXPECT_EQ(v == 2, RtcpParseCommonHeader(
+                          buffer, RtcpCommonHeader::kHeaderSizeBytes, &header));
+  }
+}
+
+TEST_F(RtcpParseCommonHeaderTest, PacketSize) {
+  // Set v = 2, leave p, fmt, pt as 0.
+  buffer[0] = 2 << 6;
+
+  const size_t kBlockSize = 3;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize);
+  const size_t kSizeInBytes = (kBlockSize + 1) * 4;
+
+  EXPECT_FALSE(RtcpParseCommonHeader(buffer, kSizeInBytes - 1, &header));
+  EXPECT_TRUE(RtcpParseCommonHeader(buffer, kSizeInBytes, &header));
+}
+
+TEST_F(RtcpParseCommonHeaderTest, PayloadSize) {
+  // Set v = 2, p = 1, but leave fmt, pt as 0.
+  buffer[0] = (2 << 6) | (1 << 5);
+
+  // Padding bit set, but no byte for padding (can't specify padding length).
+  EXPECT_FALSE(RtcpParseCommonHeader(buffer, 4, &header));
+
+  const size_t kBlockSize = 3;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize);
+  const size_t kSizeInBytes = (kBlockSize + 1) * 4;
+  const size_t kPayloadSizeBytes =
+      kSizeInBytes - RtcpCommonHeader::kHeaderSizeBytes;
+
+  // Padding one byte larger than possible.
+  buffer[kSizeInBytes - 1] = kPayloadSizeBytes + 1;
+  EXPECT_FALSE(RtcpParseCommonHeader(buffer, kSizeInBytes, &header));
+
+  // Pure padding packet?
+  buffer[kSizeInBytes - 1] = kPayloadSizeBytes;
+  EXPECT_TRUE(RtcpParseCommonHeader(buffer, kSizeInBytes, &header));
+  EXPECT_EQ(kPayloadSizeBytes, header.padding_bytes);
+  EXPECT_EQ(0u, header.payload_size_bytes);
+
+  // Single byte of actual data.
+  buffer[kSizeInBytes - 1] = kPayloadSizeBytes - 1;
+  EXPECT_TRUE(RtcpParseCommonHeader(buffer, kSizeInBytes, &header));
+  EXPECT_EQ(kPayloadSizeBytes - 1, header.padding_bytes);
+  EXPECT_EQ(1u, header.payload_size_bytes);
+}
+
+TEST_F(RtcpParseCommonHeaderTest, FormatAndPayloadType) {
+  // Format/count and packet type both set to max values.
+  const uint8_t kCountOrFormat = 0x1F;
+  const uint8_t kPacketType = 0xFF;
+  buffer[0] = 2 << 6;  // V = 2.
+  buffer[0] |= kCountOrFormat;
+  buffer[1] = kPacketType;
+
+  EXPECT_TRUE(RtcpParseCommonHeader(buffer, RtcpCommonHeader::kHeaderSizeBytes,
+                                    &header));
+  EXPECT_EQ(kCountOrFormat, header.count_or_format);
+  EXPECT_EQ(kPacketType, header.packet_type);
+}
+
+}  // namespace rtcp
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
index 541f522f8d..80f961bd1e 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -11,6 +11,7 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
@@ -41,8 +42,12 @@ template <typename T> void ClearList(std::list<T*>* my_list) {
 class RtpFecTest : public ::testing::Test {
  protected:
   RtpFecTest()
-      : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {}
+      : random_(0xfec133700742),
+        fec_(new ForwardErrorCorrection()),
+        ssrc_(random_.Rand<uint32_t>()),
+        fec_seq_num_(0) {}
 
+  webrtc::Random random_;
   ForwardErrorCorrection* fec_;
   int ssrc_;
   uint16_t fec_seq_num_;
@@ -891,22 +896,20 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
   assert(num_media_packets > 0);
   ForwardErrorCorrection::Packet* media_packet = NULL;
   int sequence_number = start_seq_num;
-  int time_stamp = rand();
+  int time_stamp = random_.Rand<int>();
 
   for (int i = 0; i < num_media_packets; ++i) {
     media_packet = new ForwardErrorCorrection::Packet;
     media_packet_list_.push_back(media_packet);
-    media_packet->length = static_cast<size_t>(
-        (static_cast<float>(rand()) / RAND_MAX) *
-        (IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead -
-         ForwardErrorCorrection::PacketOverhead()));
+    const uint32_t kMinPacketSize = kRtpHeaderSize;
+    const uint32_t kMaxPacketSize = IP_PACKET_SIZE - kRtpHeaderSize -
+                                    kTransportOverhead -
+                                    ForwardErrorCorrection::PacketOverhead();
+    media_packet->length = random_.Rand(kMinPacketSize, kMaxPacketSize);
 
-    if (media_packet->length < kRtpHeaderSize) {
-      media_packet->length = kRtpHeaderSize;
-    }
     // Generate random values for the first 2 bytes
-    media_packet->data[0] = static_cast<uint8_t>(rand() % 256);
-    media_packet->data[1] = static_cast<uint8_t>(rand() % 256);
+    media_packet->data[0] = random_.Rand<uint8_t>();
+    media_packet->data[1] = random_.Rand<uint8_t>();
 
     // The first two bits are assumed to be 10 by the FEC encoder.
     // In fact the FEC decoder will set the two first bits to 10 regardless of
@@ -929,7 +932,7 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
 
     // Generate random values for payload.
     for (size_t j = 12; j < media_packet->length; ++j) {
-      media_packet->data[j] = static_cast<uint8_t>(rand() % 256);
+      media_packet->data[j] = random_.Rand<uint8_t>();
     }
     sequence_number++;
   }
@@ -940,5 +943,5 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
 }
 
 int RtpFecTest::ConstructMediaPackets(int num_media_packets) {
-  return ConstructMediaPacketsSeqNum(num_media_packets, rand());
+  return ConstructMediaPacketsSeqNum(num_media_packets, random_.Rand<int>());
 }
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 18225f9bb4..3519499248 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -14,8 +14,8 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index cd17853b15..a6dde789d1 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -10,10 +10,12 @@
 
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace {
@@ -33,6 +35,7 @@ enum Nalu { // 0-23 from H.264, 24-31 from RFC 6184
 static const size_t kNalHeaderSize = 1;
 static const size_t kFuAHeaderSize = 2;
 static const size_t kLengthFieldSize = 2;
+static const size_t kStapAHeaderSize = kNalHeaderSize + kLengthFieldSize;
 
 // Bit masks for FU (A and B) indicators.
 enum NalDefs { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F };
@@ -40,7 +43,24 @@ enum NalDefs { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F };
 // Bit masks for FU (A and B) headers.
 enum FuDefs { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 };
 
-void ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
+// TODO(pbos): Avoid parsing this here as well as inside the jitter buffer.
+bool VerifyStapANaluLengths(const uint8_t* nalu_ptr, size_t length_remaining) {
+  while (length_remaining > 0) {
+    // Buffer doesn't contain room for additional nalu length.
+    if (length_remaining < sizeof(uint16_t))
+      return false;
+    uint16_t nalu_size = nalu_ptr[0] << 8 | nalu_ptr[1];
+    nalu_ptr += sizeof(uint16_t);
+    length_remaining -= sizeof(uint16_t);
+    if (nalu_size > length_remaining)
+      return false;
+    nalu_ptr += nalu_size;
+    length_remaining -= nalu_size;
+  }
+  return true;
+}
+
+bool ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                      const uint8_t* payload_data,
                      size_t payload_data_length) {
   parsed_payload->type.Video.width = 0;
@@ -57,7 +77,7 @@ void ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
   if (nal_type == kStapA) {
     offset = 3;
     if (offset >= payload_data_length) {
-      return; // XXX malformed
+      return false; // XXX malformed
     }
     nal_type = payload_data[offset] & kTypeMask;
     h264_header->stap_a = true;
@@ -68,10 +88,11 @@ void ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
   // send large iframes, and instead use forms of incremental/continuous refresh.
   switch (nal_type) {
     case kSei: // check if it is a Recovery Point SEI (aka GDR)
-      if (offset+1 >= payload_data_length) {
-        return; // XXX malformed
+      if (offset + 1 >= payload_data_length) {
+        LOG(LS_ERROR) << "KSei packet with incorrect pachet length.";
+        return false; // XXX malformed
       }
-      if (payload_data[offset+1] != kSeiRecPt) {
+      if (payload_data[offset + 1] != kSeiRecPt) {
         parsed_payload->frame_type = kVideoFrameDelta;
         break; // some other form of SEI - not a keyframe
       }
@@ -92,12 +113,17 @@ void ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
       parsed_payload->frame_type = kVideoFrameDelta;
       break;
   }
+  return true;
 }
 
-void ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
+bool ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                   const uint8_t* payload_data,
                   size_t payload_data_length,
                   size_t* offset) {
+  if (payload_data_length < kFuAHeaderSize) {
+    LOG(LS_ERROR) << "FU-A NAL units truncated.";
+    return false;
+  }
   uint8_t fnri = payload_data[0] & (kFBit | kNriMask);
   uint8_t original_nal_type = payload_data[1] & kTypeMask;
   bool first_fragment = (payload_data[1] & kSBit) > 0;
@@ -124,6 +150,7 @@ void ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
       &parsed_payload->type.Video.codecHeader.H264;
   h264_header->single_nalu = false;
   h264_header->stap_a = false;
+  return true;
 }
 }  // namespace
 
@@ -133,7 +160,6 @@ RtpPacketizerH264::RtpPacketizerH264(FrameType frame_type,
     : payload_data_(NULL),
       payload_size_(0),
       max_payload_len_(max_payload_len),
-      frame_type_(frame_type),
       packetization_mode_(packetization_mode) {
 }
 
@@ -327,8 +353,7 @@ void RtpPacketizerH264::NextFragmentPacket(uint8_t* buffer,
 }
 
 ProtectionType RtpPacketizerH264::GetProtectionType() {
-  return (frame_type_ == kVideoFrameKey) ? kProtectedPacket
-                                         : kUnprotectedPacket;
+  return kProtectedPacket;
 }
 
 StorageType RtpPacketizerH264::GetStorageType(
@@ -344,15 +369,24 @@ bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
                                 const uint8_t* payload_data,
                                 size_t payload_data_length) {
   assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
+
   uint8_t nal_type = payload_data[0] & kTypeMask;
   size_t offset = 0;
   if (nal_type == kFuA) {
     // Fragmented NAL units (FU-A).
-    ParseFuaNalu(parsed_payload, payload_data, payload_data_length, &offset);
+    if (!ParseFuaNalu(
+            parsed_payload, payload_data, payload_data_length, &offset)) {
+      return false;
+    }
   } else {
     // We handle STAP-A and single NALU's the same way here. The jitter buffer
     // will depacketize the STAP-A into NAL units later.
-    ParseSingleNalu(parsed_payload, payload_data, payload_data_length);
+    if (!ParseSingleNalu(parsed_payload, payload_data, payload_data_length))
+      return false;
   }
 
   parsed_payload->payload = payload_data + offset;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
index 00510badda..13898fa1a8 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -87,10 +87,9 @@ class RtpPacketizerH264 : public RtpPacketizer {
   const size_t max_payload_len_;
   RTPFragmentationHeader fragmentation_;
   PacketQueue packets_;
-  FrameType frame_type_;
   uint8_t packetization_mode_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
 };
 
 // Depacketizer for H264.
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index caae400550..d29e3d4f21 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -13,7 +13,7 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
@@ -83,7 +83,7 @@ void TestFua(size_t frame_size,
   fragmentation.fragmentationOffset[0] = 0;
   fragmentation.fragmentationLength[0] = frame_size;
   rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
-      kRtpVideoH264, max_payload_size, NULL, kFrameEmpty));
+      kRtpVideoH264, max_payload_size, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame.get(), frame_size, &fragmentation);
 
   rtc::scoped_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]);
@@ -157,7 +157,7 @@ TEST(RtpPacketizerH264Test, TestSingleNalu) {
   fragmentation.fragmentationOffset[0] = 0;
   fragmentation.fragmentationLength[0] = sizeof(frame);
   rtc::scoped_ptr<RtpPacketizer> packetizer(
-      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, sizeof(frame), &fragmentation);
   uint8_t packet[kMaxPayloadSize] = {0};
   size_t length = 0;
@@ -186,7 +186,7 @@ TEST(RtpPacketizerH264Test, TestSingleNaluTwoPackets) {
   frame[fragmentation.fragmentationOffset[1]] = 0x01;
 
   rtc::scoped_ptr<RtpPacketizer> packetizer(
-      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
   uint8_t packet[kMaxPayloadSize] = {0};
@@ -207,9 +207,9 @@ TEST(RtpPacketizerH264Test, TestSingleNaluTwoPackets) {
 TEST(RtpPacketizerH264Test, TestStapA) {
   const size_t kFrameSize =
       kMaxPayloadSize - 3 * kLengthFieldLength - kNalHeaderSize;
-  uint8_t frame[kFrameSize] = {0x07, 0xFF,  // F=0, NRI=0, Type=7.
-                               0x08, 0xFF,  // F=0, NRI=0, Type=8.
-                               0x05};       // F=0, NRI=0, Type=5.
+  uint8_t frame[kFrameSize] = {0x07, 0xFF,  // F=0, NRI=0, Type=7 (SPS).
+                               0x08, 0xFF,  // F=0, NRI=0, Type=8 (PPS).
+                               0x05};       // F=0, NRI=0, Type=5 (IDR).
   const size_t kPayloadOffset = 5;
   for (size_t i = 0; i < kFrameSize - kPayloadOffset; ++i)
     frame[i + kPayloadOffset] = i;
@@ -223,7 +223,7 @@ TEST(RtpPacketizerH264Test, TestStapA) {
   fragmentation.fragmentationLength[2] =
       kNalHeaderSize + kFrameSize - kPayloadOffset;
   rtc::scoped_ptr<RtpPacketizer> packetizer(
-      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
   uint8_t packet[kMaxPayloadSize] = {0};
@@ -258,7 +258,7 @@ TEST(RtpPacketizerH264Test, TestTooSmallForStapAHeaders) {
   fragmentation.fragmentationLength[2] =
       kNalHeaderSize + kFrameSize - kPayloadOffset;
   rtc::scoped_ptr<RtpPacketizer> packetizer(
-      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
   uint8_t packet[kMaxPayloadSize] = {0};
@@ -306,7 +306,7 @@ TEST(RtpPacketizerH264Test, TestMixedStapA_FUA) {
     }
   }
   rtc::scoped_ptr<RtpPacketizer> packetizer(
-      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kFrameEmpty));
+      RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
   // First expecting two FU-A packets.
@@ -398,7 +398,7 @@ class RtpDepacketizerH264Test : public ::testing::Test {
 };
 
 TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
-  uint8_t packet[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5.
+  uint8_t packet[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5 (IDR).
   RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
@@ -406,15 +406,34 @@ TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
-  EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kH264SingleNalu,
+            payload.type.Video.codecHeader.H264.packetization_type);
+  EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
+}
+
+TEST_F(RtpDepacketizerH264Test, TestSingleNaluSpsWithResolution) {
+  uint8_t packet[] = {kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50,
+                      0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
+                      0x00, 0x00, 0x2A, 0xE0, 0xF1, 0x83, 0x19, 0x60};
+  RtpDepacketizer::ParsedPayload payload;
+
+  ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
+  ExpectPacket(&payload, packet, sizeof(packet));
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_TRUE(payload.type.Video.isFirstPacket);
+  EXPECT_EQ(kH264SingleNalu,
+            payload.type.Video.codecHeader.H264.packetization_type);
+  EXPECT_EQ(1280u, payload.type.Video.width);
+  EXPECT_EQ(720u, payload.type.Video.height);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
   uint8_t packet[16] = {kStapA,  // F=0, NRI=0, Type=24.
                         // Length, nal header, payload.
-                        0,      0x02, kIdr, 0xFF, 0,    0x03, kIdr, 0xFF,
-                        0x00,   0,    0x04, kIdr, 0xFF, 0x00, 0x11};
+                        0, 0x02, kSps, 0xFF,
+                        0, 0x03, kPps, 0xFF, 0x00,
+                        0, 0x04, kIdr, 0xFF, 0x00, 0x11};
   RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
@@ -422,8 +441,29 @@ TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
-  EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
-  EXPECT_TRUE(payload.type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
+  // NALU type for aggregated packets is the type of the first packet only.
+  EXPECT_EQ(kSps, payload.type.Video.codecHeader.H264.nalu_type);
+}
+
+TEST_F(RtpDepacketizerH264Test, TestStapANaluSpsWithResolution) {
+  uint8_t packet[] = {kStapA,  // F=0, NRI=0, Type=24.
+                      // Length (2 bytes), nal header, payload.
+                      0,      24,   kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9,
+                      0x40,   0x50, 0x05, 0xBA, 0x10, 0x00, 0x00, 0x03,
+                      0x00,   0xC0, 0x00, 0x00, 0x2A, 0xE0, 0xF1, 0x83,
+                      0x19,   0x60, 0,    0x03, kIdr, 0xFF, 0x00, 0,
+                      0x04,   kIdr, 0xFF, 0x00, 0x11};
+  RtpDepacketizer::ParsedPayload payload;
+
+  ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
+  ExpectPacket(&payload, packet, sizeof(packet));
+  EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+  EXPECT_TRUE(payload.type.Video.isFirstPacket);
+  EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
+  EXPECT_EQ(1280u, payload.type.Video.width);
+  EXPECT_EQ(720u, payload.type.Video.height);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
@@ -438,8 +478,9 @@ TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
-  EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
-  EXPECT_TRUE(payload.type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
+  // NALU type for aggregated packets is the type of the first packet only.
+  EXPECT_EQ(kSlice, payload.type.Video.codecHeader.H264.nalu_type);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestFuA) {
@@ -473,8 +514,8 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kH264FuA, payload.type.Video.codecHeader.H264.packetization_type);
+  EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
 
   // Following packets will be 2 bytes shorter since they will only be appended
   // onto the first packet.
@@ -484,8 +525,8 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_FALSE(payload.type.Video.isFirstPacket);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kH264FuA, payload.type.Video.codecHeader.H264.packetization_type);
+  EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
 
   payload = RtpDepacketizer::ParsedPayload();
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet3, sizeof(packet3)));
@@ -493,7 +534,39 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_FALSE(payload.type.Video.isFirstPacket);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
-  EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
+  EXPECT_EQ(kH264FuA, payload.type.Video.codecHeader.H264.packetization_type);
+  EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
 }
+
+TEST_F(RtpDepacketizerH264Test, TestEmptyPayload) {
+  // Using a wild pointer to crash on accesses from inside the depacketizer.
+  uint8_t* garbage_ptr = reinterpret_cast<uint8_t*>(0x4711);
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, garbage_ptr, 0));
+}
+
+TEST_F(RtpDepacketizerH264Test, TestTruncatedFuaNalu) {
+  const uint8_t kPayload[] = {0x9c};
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
+}
+
+TEST_F(RtpDepacketizerH264Test, TestTruncatedSingleStapANalu) {
+  const uint8_t kPayload[] = {0xd8, 0x27};
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
+}
+
+TEST_F(RtpDepacketizerH264Test, TestTruncationJustAfterSingleStapANalu) {
+  const uint8_t kPayload[] = {0x38, 0x27, 0x27};
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
+}
+
+TEST_F(RtpDepacketizerH264Test, TestShortSpsPacket) {
+  const uint8_t kPayload[] = {0x27, 0x80, 0x00};
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 1fa288acad..b47e9b9359 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -10,7 +10,8 @@
 
 #include <string>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 
 namespace webrtc {
@@ -90,6 +91,10 @@ bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
                                    const uint8_t* payload_data,
                                    size_t payload_data_length) {
   assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
 
   uint8_t generic_header = *payload_data++;
   --payload_data_length;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
index 2e7bca5c48..3bf72e9dd3 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
@@ -59,7 +59,7 @@ class RtpPacketizerGeneric : public RtpPacketizer {
   size_t payload_length_;
   uint8_t generic_header_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
 };
 
 // Depacketizer for generic codec.
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
index 5202754caf..7beeb181e2 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -15,8 +15,8 @@
 
 #include <vector>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 namespace {
@@ -234,16 +234,16 @@ ProtectionType RtpPacketizerVp8::GetProtectionType() {
 }
 
 StorageType RtpPacketizerVp8::GetStorageType(uint32_t retransmission_settings) {
-  StorageType storage = kAllowRetransmission;
   if (hdr_info_.temporalIdx == 0 &&
       !(retransmission_settings & kRetransmitBaseLayer)) {
-    storage = kDontRetransmit;
-  } else if (hdr_info_.temporalIdx != kNoTemporalIdx &&
+    return kDontRetransmit;
+  }
+  if (hdr_info_.temporalIdx != kNoTemporalIdx &&
              hdr_info_.temporalIdx > 0 &&
              !(retransmission_settings & kRetransmitHigherLayers)) {
-    storage = kDontRetransmit;
+    return kDontRetransmit;
   }
-  return storage;
+  return kAllowRetransmission;
 }
 
 std::string RtpPacketizerVp8::ToString() {
@@ -668,6 +668,10 @@ bool RtpDepacketizerVp8::Parse(ParsedPayload* parsed_payload,
                                const uint8_t* payload_data,
                                size_t payload_data_length) {
   assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
 
   // Parse mandatory first byte of payload descriptor.
   bool extension = (*payload_data & 0x80) ? true : false;               // X bit
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
index d73dfc1b50..d62ecba85f 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -30,7 +30,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
@@ -212,7 +212,7 @@ class RtpPacketizerVp8 : public RtpPacketizer {
   InfoQueue packets_;
   bool packets_calculated_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp8);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp8);
 };
 
 // Depacketizer for VP8.
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
index 2454fb70cc..668476833d 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -19,7 +19,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/typedefs.h"
 
@@ -65,7 +65,7 @@ class RtpFormatVp8TestHelper {
   bool sloppy_partitioning_;
   bool inited_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8TestHelper);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8TestHelper);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
index 804dc09038..4283a778d0 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -596,4 +596,11 @@ TEST_F(RtpDepacketizerVp8Test, TestWithPacketizer) {
   EXPECT_EQ(payload.type.Video.codecHeader.VP8.layerSync,
             input_header.layerSync);
 }
+
+TEST_F(RtpDepacketizerVp8Test, TestEmptyPayload) {
+  // Using a wild pointer to crash on accesses from inside the depacketizer.
+  uint8_t* garbage_ptr = reinterpret_cast<uint8_t*>(0x4711);
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, garbage_ptr, 0));
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
index 883fbce5c8..3feca4392a 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -25,7 +25,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
@@ -91,7 +91,7 @@ class RtpPacketizerVp9 : public RtpPacketizer {
   size_t payload_size_;              // The size in bytes of the payload data.
   PacketInfoQueue packets_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
 };
 
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 07c651ea82..a41c9a9f62 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
 
 #include <map>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -26,11 +26,11 @@ const size_t kAudioLevelLength = 2;
 const size_t kAbsoluteSendTimeLength = 4;
 const size_t kVideoRotationLength = 2;
 const size_t kTransportSequenceNumberLength = 3;
-// kRIDLength is variable
+// kRtpStreamIdLength is variable
 const size_t kRtpStreamIdLength = 4; // max 1-byte header extension length
 
 struct HeaderExtension {
-  HeaderExtension(RTPExtensionType extension_type)
+  explicit HeaderExtension(RTPExtensionType extension_type)
       : type(extension_type), length(0), active(true) {
     Init();
   }
@@ -117,6 +117,7 @@ class RtpHeaderExtensionMap {
   int32_t Register(const RTPExtensionType type, const uint8_t id, bool active);
   std::map<uint8_t, HeaderExtension*> extensionMap_;
 };
-}
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
 
-#endif // WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
index 520cf7a962..ca37750621 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
@@ -15,7 +15,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/typedefs.h"
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index 0b8280de1e..d4cbe544cc 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -7,12 +7,12 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -58,7 +58,7 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
     rtp_header_extension_map_.GetCopy(&map);
   }
 
-  const bool valid_rtpheader = rtp_parser.Parse(*header, &map);
+  const bool valid_rtpheader = rtp_parser.Parse(header, &map);
   if (!valid_rtpheader) {
     return false;
   }
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index 8fb183543a..49f9d8530a 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -13,24 +13,25 @@
 #include <assert.h>
 #include <stdlib.h>
 #include <string.h>   // memset
+
+#include <algorithm>
 #include <limits>
 #include <set>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 static const int kMinPacketRequestBytes = 50;
 
 RTPPacketHistory::RTPPacketHistory(Clock* clock)
-  : clock_(clock),
-    critsect_(CriticalSectionWrapper::CreateCriticalSection()),
-    store_(false),
-    prev_index_(0),
-    max_packet_length_(0) {
-}
+    : clock_(clock),
+      critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+      store_(false),
+      prev_index_(0) {}
 
 RTPPacketHistory::~RTPPacketHistory() {
 }
@@ -55,11 +56,6 @@ void RTPPacketHistory::Allocate(size_t number_to_store) {
   assert(number_to_store <= kMaxHistoryCapacity);
   store_ = true;
   stored_packets_.resize(number_to_store);
-  stored_seq_nums_.resize(number_to_store);
-  stored_lengths_.resize(number_to_store);
-  stored_times_.resize(number_to_store);
-  stored_send_times_.resize(number_to_store);
-  stored_types_.resize(number_to_store);
 }
 
 void RTPPacketHistory::Free() {
@@ -67,21 +63,10 @@ void RTPPacketHistory::Free() {
     return;
   }
 
-  std::vector<std::vector<uint8_t> >::iterator it;
-  for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) {
-    it->clear();
-  }
-
   stored_packets_.clear();
-  stored_seq_nums_.clear();
-  stored_lengths_.clear();
-  stored_times_.clear();
-  stored_send_times_.clear();
-  stored_types_.clear();
 
   store_ = false;
   prev_index_ = 0;
-  max_packet_length_ = 0;
 }
 
 bool RTPPacketHistory::StorePackets() const {
@@ -89,37 +74,10 @@ bool RTPPacketHistory::StorePackets() const {
   return store_;
 }
 
-void RTPPacketHistory::VerifyAndAllocatePacketLength(size_t packet_length,
-                                                     uint32_t start_index) {
-  assert(packet_length > 0);
-  if (!store_) {
-    return;
-  }
-
-  // If start_index > 0 this is a resize and we must check any new (empty)
-  // packets created during the resize.
-  if (start_index == 0 && packet_length <= max_packet_length_) {
-    return;
-  }
-
-  max_packet_length_ = std::max(packet_length, max_packet_length_);
-
-  std::vector<std::vector<uint8_t> >::iterator it;
-  for (it = stored_packets_.begin() + start_index; it != stored_packets_.end();
-       ++it) {
-    it->resize(max_packet_length_);
-  }
-}
-
 int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
                                        size_t packet_length,
-                                       size_t max_packet_length,
                                        int64_t capture_time_ms,
                                        StorageType type) {
-  if (type == kDontStore) {
-    return 0;
-  }
-
   CriticalSectionScoped cs(critsect_.get());
   if (!store_) {
     return 0;
@@ -128,9 +86,7 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
   assert(packet);
   assert(packet_length > 3);
 
-  VerifyAndAllocatePacketLength(max_packet_length, 0);
-
-  if (packet_length > max_packet_length_) {
+  if (packet_length > IP_PACKET_SIZE) {
     LOG(LS_WARNING) << "Failed to store RTP packet with length: "
                     << packet_length;
     return -1;
@@ -141,14 +97,13 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
   // If index we're about to overwrite contains a packet that has not
   // yet been sent (probably pending in paced sender), we need to expand
   // the buffer.
-  if (stored_lengths_[prev_index_] > 0 &&
-      stored_send_times_[prev_index_] == 0) {
+  if (stored_packets_[prev_index_].length > 0 &&
+      stored_packets_[prev_index_].send_time == 0) {
     size_t current_size = static_cast<uint16_t>(stored_packets_.size());
     if (current_size < kMaxHistoryCapacity) {
       size_t expanded_size = std::max(current_size * 3 / 2, current_size + 1);
       expanded_size = std::min(expanded_size, kMaxHistoryCapacity);
       Allocate(expanded_size);
-      VerifyAndAllocatePacketLength(max_packet_length, current_size);
       // Causes discontinuity, but that's OK-ish. FindSeqNum() will still work,
       // but may be slower - at least until buffer has wrapped around once.
       prev_index_ = current_size;
@@ -156,21 +111,20 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
   }
 
   // Store packet
-  std::vector<std::vector<uint8_t> >::iterator it =
-      stored_packets_.begin() + prev_index_;
   // TODO(sprang): Overhaul this class and get rid of this copy step.
   //               (Finally introduce the RtpPacket class?)
-  std::copy(packet, packet + packet_length, it->begin());
+  memcpy(stored_packets_[prev_index_].data, packet, packet_length);
+  stored_packets_[prev_index_].length = packet_length;
 
-  stored_seq_nums_[prev_index_] = seq_num;
-  stored_lengths_[prev_index_] = packet_length;
-  stored_times_[prev_index_] = (capture_time_ms > 0) ? capture_time_ms :
-      clock_->TimeInMilliseconds();
-  stored_send_times_[prev_index_] = 0;  // Packet not sent.
-  stored_types_[prev_index_] = type;
+  stored_packets_[prev_index_].sequence_number = seq_num;
+  stored_packets_[prev_index_].time_ms =
+      (capture_time_ms > 0) ? capture_time_ms : clock_->TimeInMilliseconds();
+  stored_packets_[prev_index_].send_time = 0;  // Packet not sent.
+  stored_packets_[prev_index_].storage_type = type;
+  stored_packets_[prev_index_].has_been_retransmitted = false;
 
   ++prev_index_;
-  if (prev_index_ >= stored_seq_nums_.size()) {
+  if (prev_index_ >= stored_packets_.size()) {
     prev_index_ = 0;
   }
   return 0;
@@ -188,8 +142,7 @@ bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const {
     return false;
   }
 
-  size_t length = stored_lengths_.at(index);
-  if (length == 0 || length > max_packet_length_) {
+  if (stored_packets_[index].length == 0) {
     // Invalid length.
     return false;
   }
@@ -209,11 +162,11 @@ bool RTPPacketHistory::SetSent(uint16_t sequence_number) {
   }
 
   // Send time already set.
-  if (stored_send_times_[index] != 0) {
+  if (stored_packets_[index].send_time != 0) {
     return false;
   }
 
-  stored_send_times_[index] = clock_->TimeInMilliseconds();
+  stored_packets_[index].send_time = clock_->TimeInMilliseconds();
   return true;
 }
 
@@ -224,10 +177,9 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
                                                size_t* packet_length,
                                                int64_t* stored_time_ms) {
   CriticalSectionScoped cs(critsect_.get());
-  assert(*packet_length >= max_packet_length_);
-  if (!store_) {
+  RTC_CHECK_GE(*packet_length, static_cast<size_t>(IP_PACKET_SIZE));
+  if (!store_)
     return false;
-  }
 
   int32_t index = 0;
   bool found = FindSeqNum(sequence_number, &index);
@@ -236,27 +188,32 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
     return false;
   }
 
-  size_t length = stored_lengths_.at(index);
-  assert(length <= max_packet_length_);
+  size_t length = stored_packets_[index].length;
+  assert(length <= IP_PACKET_SIZE);
   if (length == 0) {
     LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number
                     << ", len " << length;
     return false;
   }
 
-  // Verify elapsed time since last retrieve.
+  // Verify elapsed time since last retrieve, but only for retransmissions and
+  // always send packet upon first retransmission request.
   int64_t now = clock_->TimeInMilliseconds();
-  if (min_elapsed_time_ms > 0 &&
-      ((now - stored_send_times_.at(index)) < min_elapsed_time_ms)) {
+  if (min_elapsed_time_ms > 0 && retransmit &&
+      stored_packets_[index].has_been_retransmitted &&
+      ((now - stored_packets_[index].send_time) < min_elapsed_time_ms)) {
     return false;
   }
 
-  if (retransmit && stored_types_.at(index) == kDontRetransmit) {
-    // No bytes copied since this packet shouldn't be retransmitted or is
-    // of zero size.
-    return false;
+  if (retransmit) {
+    if (stored_packets_[index].storage_type == kDontRetransmit) {
+      // No bytes copied since this packet shouldn't be retransmitted or is
+      // of zero size.
+      return false;
+    }
+    stored_packets_[index].has_been_retransmitted = true;
   }
-  stored_send_times_[index] = clock_->TimeInMilliseconds();
+  stored_packets_[index].send_time = clock_->TimeInMilliseconds();
   GetPacket(index, packet, packet_length, stored_time_ms);
   return true;
 }
@@ -266,13 +223,10 @@ void RTPPacketHistory::GetPacket(int index,
                                  size_t* packet_length,
                                  int64_t* stored_time_ms) const {
   // Get packet.
-  size_t length = stored_lengths_.at(index);
-  std::vector<std::vector<uint8_t> >::const_iterator it_found_packet =
-      stored_packets_.begin() + index;
-  std::copy(it_found_packet->begin(), it_found_packet->begin() + length,
-            packet);
+  size_t length = stored_packets_[index].length;
+  memcpy(packet, stored_packets_[index].data, length);
   *packet_length = length;
-  *stored_time_ms = stored_times_.at(index);
+  *stored_time_ms = stored_packets_[index].time_ms;
 }
 
 bool RTPPacketHistory::GetBestFittingPacket(uint8_t* packet,
@@ -294,24 +248,24 @@ bool RTPPacketHistory::FindSeqNum(uint16_t sequence_number,
   uint16_t temp_sequence_number = 0;
   if (prev_index_ > 0) {
     *index = prev_index_ - 1;
-    temp_sequence_number = stored_seq_nums_[*index];
+    temp_sequence_number = stored_packets_[*index].sequence_number;
   } else {
-    *index = stored_seq_nums_.size() - 1;
-    temp_sequence_number = stored_seq_nums_[*index];  // wrap
+    *index = stored_packets_.size() - 1;
+    temp_sequence_number = stored_packets_[*index].sequence_number;  // wrap
   }
 
   int32_t idx = (prev_index_ - 1) - (temp_sequence_number - sequence_number);
-  if (idx >= 0 && idx < static_cast<int>(stored_seq_nums_.size())) {
+  if (idx >= 0 && idx < static_cast<int>(stored_packets_.size())) {
     *index = idx;
-    temp_sequence_number = stored_seq_nums_[*index];
+    temp_sequence_number = stored_packets_[*index].sequence_number;
   }
 
   if (temp_sequence_number != sequence_number) {
     // We did not found a match, search all.
-    for (uint16_t m = 0; m < stored_seq_nums_.size(); m++) {
-      if (stored_seq_nums_[m] == sequence_number) {
+    for (uint16_t m = 0; m < stored_packets_.size(); m++) {
+      if (stored_packets_[m].sequence_number == sequence_number) {
         *index = m;
-        temp_sequence_number = stored_seq_nums_[*index];
+        temp_sequence_number = stored_packets_[*index].sequence_number;
         break;
       }
     }
@@ -324,15 +278,16 @@ bool RTPPacketHistory::FindSeqNum(uint16_t sequence_number,
 }
 
 int RTPPacketHistory::FindBestFittingPacket(size_t size) const {
-  if (size < kMinPacketRequestBytes || stored_lengths_.empty())
+  if (size < kMinPacketRequestBytes || stored_packets_.empty())
     return -1;
   size_t min_diff = std::numeric_limits<size_t>::max();
   int best_index = -1;  // Returned unchanged if we don't find anything.
-  for (size_t i = 0; i < stored_lengths_.size(); ++i) {
-    if (stored_lengths_[i] == 0)
+  for (size_t i = 0; i < stored_packets_.size(); ++i) {
+    if (stored_packets_[i].length == 0)
       continue;
-    size_t diff = (stored_lengths_[i] > size) ?
-        (stored_lengths_[i] - size) : (size - stored_lengths_[i]);
+    size_t diff = (stored_packets_[i].length > size)
+                      ? (stored_packets_[i].length - size)
+                      : (size - stored_packets_[i].length);
     if (diff < min_diff) {
       min_diff = diff;
       best_index = static_cast<int>(i);
@@ -340,4 +295,7 @@ int RTPPacketHistory::FindBestFittingPacket(size_t size) const {
   }
   return best_index;
 }
+
+RTPPacketHistory::StoredPacket::StoredPacket() {}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 212aa21266..8e1a732b19 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -10,14 +10,14 @@
  *  Class for storing RTP packets.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
 
 #include <vector>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,7 +29,7 @@ static const size_t kMaxHistoryCapacity = 9600;
 
 class RTPPacketHistory {
  public:
-  RTPPacketHistory(Clock* clock);
+  explicit RTPPacketHistory(Clock* clock);
   ~RTPPacketHistory();
 
   void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
@@ -39,7 +39,6 @@ class RTPPacketHistory {
   // Stores RTP packet.
   int32_t PutRTPPacket(const uint8_t* packet,
                        size_t packet_length,
-                       size_t max_packet_length,
                        int64_t capture_time_ms,
                        StorageType type);
 
@@ -47,13 +46,12 @@ class RTPPacketHistory {
   // The packet is copied to the buffer pointed to by ptr_rtp_packet.
   // The rtp_packet_length should show the available buffer size.
   // Returns true if packet is found.
-  // rtp_packet_length: returns the copied packet length on success.
+  // packet_length: returns the copied packet length on success.
   // min_elapsed_time_ms: the minimum time that must have elapsed since the last
   // time the packet was resent (parameter is ignored if set to zero).
-  // If the packet is found but the minimum time has not elaped, no bytes are
+  // If the packet is found but the minimum time has not elapsed, no bytes are
   // copied.
   // stored_time_ms: returns the time when the packet was stored.
-  // type: returns the storage type set in PutRTPPacket.
   bool GetPacketAndSetSendTime(uint16_t sequence_number,
                                int64_t min_elapsed_time_ms,
                                bool retransmit,
@@ -88,14 +86,19 @@ class RTPPacketHistory {
   rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
   bool store_ GUARDED_BY(critsect_);
   uint32_t prev_index_ GUARDED_BY(critsect_);
-  size_t max_packet_length_ GUARDED_BY(critsect_);
 
-  std::vector<std::vector<uint8_t> > stored_packets_ GUARDED_BY(critsect_);
-  std::vector<uint16_t> stored_seq_nums_ GUARDED_BY(critsect_);
-  std::vector<size_t> stored_lengths_ GUARDED_BY(critsect_);
-  std::vector<int64_t> stored_times_ GUARDED_BY(critsect_);
-  std::vector<int64_t> stored_send_times_ GUARDED_BY(critsect_);
-  std::vector<StorageType> stored_types_ GUARDED_BY(critsect_);
+  struct StoredPacket {
+    StoredPacket();
+    uint16_t sequence_number = 0;
+    int64_t time_ms = 0;
+    int64_t send_time = 0;
+    StorageType storage_type = kDontRetransmit;
+    bool has_been_retransmitted = false;
+
+    uint8_t data[IP_PACKET_SIZE];
+    size_t length = 0;
+  };
+  std::vector<StoredPacket> stored_packets_ GUARDED_BY(critsect_);
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index fe33b01e06..a406d8bc9b 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -12,10 +12,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/video_engine/vie_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,7 +28,7 @@ class RtpPacketHistoryTest : public ::testing::Test {
   ~RtpPacketHistoryTest() {
     delete hist_;
   }
-  
+
   SimulatedClock fake_clock_;
   RTPPacketHistory* hist_;
   enum {kPayload = 127};
@@ -54,7 +53,7 @@ class RtpPacketHistoryTest : public ::testing::Test {
     array[(*cur_pos)++] = ssrc >> 16;
     array[(*cur_pos)++] = ssrc >> 8;
     array[(*cur_pos)++] = ssrc;
-  } 
+  }
 };
 
 TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
@@ -70,23 +69,8 @@ TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
   size_t len = 0;
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kAllowRetransmission));
-  // Packet should not be stored.
-  len = kMaxPacketLength;
-  int64_t time;
-  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_, &len,
-                                              &time));
-}
-
-TEST_F(RtpPacketHistoryTest, DontStore) {
-  hist_->SetStorePacketsStatus(true, 10);
-  size_t len = 0;
-  int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
-  CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kDontStore));
-
+  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                   kAllowRetransmission));
   // Packet should not be stored.
   len = kMaxPacketLength;
   int64_t time;
@@ -97,11 +81,8 @@ TEST_F(RtpPacketHistoryTest, DontStore) {
 TEST_F(RtpPacketHistoryTest, PutRtpPacket_TooLargePacketLength) {
   hist_->SetStorePacketsStatus(true, 10);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
-  EXPECT_EQ(-1, hist_->PutRTPPacket(packet_,
-                                    kMaxPacketLength + 1,
-                                    kMaxPacketLength,
-                                    capture_time_ms,
-                                    kAllowRetransmission));
+  EXPECT_EQ(-1, hist_->PutRTPPacket(packet_, kMaxPacketLength + 1,
+                                    capture_time_ms, kAllowRetransmission));
 }
 
 TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
@@ -119,8 +100,8 @@ TEST_F(RtpPacketHistoryTest, PutRtpPacket) {
 
   EXPECT_FALSE(hist_->HasRTPPacket(kSeqNum));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kAllowRetransmission));
+  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                   kAllowRetransmission));
   EXPECT_TRUE(hist_->HasRTPPacket(kSeqNum));
 }
 
@@ -129,8 +110,8 @@ TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
   size_t len = 0;
   int64_t capture_time_ms = 1;
   CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kAllowRetransmission));
+  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                   kAllowRetransmission));
 
   size_t len_out = kMaxPacketLength;
   int64_t time;
@@ -149,8 +130,7 @@ TEST_F(RtpPacketHistoryTest, NoCaptureTime) {
   fake_clock_.AdvanceTimeMilliseconds(1);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   -1, kAllowRetransmission));
+  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, -1, kAllowRetransmission));
 
   size_t len_out = kMaxPacketLength;
   int64_t time;
@@ -168,8 +148,8 @@ TEST_F(RtpPacketHistoryTest, DontRetransmit) {
   size_t len = 0;
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kDontRetransmit));
+  EXPECT_EQ(
+      0, hist_->PutRTPPacket(packet_, len, capture_time_ms, kDontRetransmit));
 
   size_t len_out = kMaxPacketLength;
   int64_t time;
@@ -180,29 +160,66 @@ TEST_F(RtpPacketHistoryTest, DontRetransmit) {
 }
 
 TEST_F(RtpPacketHistoryTest, MinResendTime) {
+  static const int64_t kMinRetransmitIntervalMs = 100;
+
   hist_->SetStorePacketsStatus(true, 10);
   size_t len = 0;
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kAllowRetransmission));
+  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                   kAllowRetransmission));
 
+  // First transmission: TimeToSendPacket() call from pacer.
   int64_t time;
   len = kMaxPacketLength;
-  EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 100, false, packet_, &len,
-                                             &time));
-  fake_clock_.AdvanceTimeMilliseconds(100);
+  EXPECT_TRUE(
+      hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_, &len, &time));
+
+  fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs);
   // Time has elapsed.
   len = kMaxPacketLength;
-  EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 100, false, packet_, &len,
-                                             &time));
+  EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs,
+                                             true, packet_, &len, &time));
   EXPECT_GT(len, 0u);
   EXPECT_EQ(capture_time_ms, time);
 
+  fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
   // Time has not elapsed. Packet should be found, but no bytes copied.
   len = kMaxPacketLength;
-  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 101, false, packet_,
-                                              &len, &time));
+  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs,
+                                              true, packet_, &len, &time));
+}
+
+TEST_F(RtpPacketHistoryTest, EarlyFirstResend) {
+  static const int64_t kMinRetransmitIntervalMs = 100;
+
+  hist_->SetStorePacketsStatus(true, 10);
+  size_t len = 0;
+  int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
+  CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                   kAllowRetransmission));
+
+  // First transmission: TimeToSendPacket() call from pacer.
+  int64_t time;
+  len = kMaxPacketLength;
+  EXPECT_TRUE(
+      hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_, &len, &time));
+
+  fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
+  // Time has not elapsed, but this is the first retransmission request so
+  // allow anyway.
+  len = kMaxPacketLength;
+  EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs,
+                                             true, packet_, &len, &time));
+  EXPECT_GT(len, 0u);
+  EXPECT_EQ(capture_time_ms, time);
+
+  fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
+  // Time has not elapsed. Packet should be found, but no bytes copied.
+  len = kMaxPacketLength;
+  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs,
+                                              true, packet_, &len, &time));
 }
 
 TEST_F(RtpPacketHistoryTest, DynamicExpansion) {
@@ -215,8 +232,8 @@ TEST_F(RtpPacketHistoryTest, DynamicExpansion) {
   for (int i = 0; i < 4; ++i) {
     len = 0;
     CreateRtpPacket(kSeqNum + i, kSsrc, kPayload, kTimestamp, packet_, &len);
-    EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                     capture_time_ms, kAllowRetransmission));
+    EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                     kAllowRetransmission));
   }
   for (int i = 0; i < 4; ++i) {
     len = kMaxPacketLength;
@@ -230,8 +247,8 @@ TEST_F(RtpPacketHistoryTest, DynamicExpansion) {
   for (int i = 4; i < 20; ++i) {
     len = 0;
     CreateRtpPacket(kSeqNum + i, kSsrc, kPayload, kTimestamp, packet_, &len);
-    EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                     capture_time_ms, kAllowRetransmission));
+    EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                     kAllowRetransmission));
   }
   for (int i = 4; i < 20; ++i) {
     len = kMaxPacketLength;
@@ -250,6 +267,7 @@ TEST_F(RtpPacketHistoryTest, DynamicExpansion) {
 }
 
 TEST_F(RtpPacketHistoryTest, FullExpansion) {
+  static const int kSendSidePacketHistorySize = 600;
   hist_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
   size_t len;
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
@@ -257,8 +275,8 @@ TEST_F(RtpPacketHistoryTest, FullExpansion) {
   for (size_t i = 0; i < kMaxHistoryCapacity + 1; ++i) {
     len = 0;
     CreateRtpPacket(kSeqNum + i, kSsrc, kPayload, kTimestamp, packet_, &len);
-    EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                     capture_time_ms, kAllowRetransmission));
+    EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
+                                     kAllowRetransmission));
   }
 
   fake_clock_.AdvanceTimeMilliseconds(100);
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 9005598a86..ce0bcd7fed 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -8,15 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
-RTPPayloadRegistry::RTPPayloadRegistry(
-    RTPPayloadStrategy* rtp_payload_strategy)
+RTPPayloadRegistry::RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_payload_strategy_(rtp_payload_strategy),
       red_payload_type_(-1),
@@ -25,7 +24,8 @@ RTPPayloadRegistry::RTPPayloadRegistry(
       last_received_payload_type_(-1),
       last_received_media_payload_type_(-1),
       rtx_(false),
-      payload_type_rtx_(-1),
+      rtx_payload_type_(-1),
+      use_rtx_payload_mapping_on_restore_(false),
       ssrc_rtx_(0) {}
 
 RTPPayloadRegistry::~RTPPayloadRegistry() {
@@ -40,7 +40,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate,
     bool* created_new_payload) {
   assert(payload_type >= 0);
@@ -139,7 +139,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const size_t payload_name_length,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate) {
   RtpUtility::PayloadTypeMap::iterator iterator = payload_type_map_.begin();
   for (; iterator != payload_type_map_.end(); ++iterator) {
@@ -171,7 +171,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
 int32_t RTPPayloadRegistry::ReceivePayloadType(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate,
     int8_t* payload_type) const {
   assert(payload_type);
@@ -237,37 +237,58 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet,
                                                size_t* packet_length,
                                                uint32_t original_ssrc,
                                                const RTPHeader& header) const {
-  if (kRtxHeaderSize + header.headerLength > *packet_length) {
+  return RestoreOriginalPacket(*restored_packet, packet, packet_length,
+                               original_ssrc, header);
+}
+
+bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
+                                               const uint8_t* packet,
+                                               size_t* packet_length,
+                                               uint32_t original_ssrc,
+                                               const RTPHeader& header) const {
+  if (kRtxHeaderSize + header.headerLength + header.paddingLength >
+      *packet_length) {
     return false;
   }
   const uint8_t* rtx_header = packet + header.headerLength;
   uint16_t original_sequence_number = (rtx_header[0] << 8) + rtx_header[1];
 
   // Copy the packet into the restored packet, except for the RTX header.
-  memcpy(*restored_packet, packet, header.headerLength);
-  memcpy(*restored_packet + header.headerLength,
+  memcpy(restored_packet, packet, header.headerLength);
+  memcpy(restored_packet + header.headerLength,
          packet + header.headerLength + kRtxHeaderSize,
          *packet_length - header.headerLength - kRtxHeaderSize);
   *packet_length -= kRtxHeaderSize;
 
   // Replace the SSRC and the sequence number with the originals.
-  ByteWriter<uint16_t>::WriteBigEndian(*restored_packet + 2,
+  ByteWriter<uint16_t>::WriteBigEndian(restored_packet + 2,
                                        original_sequence_number);
-  ByteWriter<uint32_t>::WriteBigEndian(*restored_packet + 8, original_ssrc);
+  ByteWriter<uint32_t>::WriteBigEndian(restored_packet + 8, original_ssrc);
 
   CriticalSectionScoped cs(crit_sect_.get());
+  if (!rtx_)
+    return true;
 
-  if (payload_type_rtx_ != -1) {
-    if (header.payloadType == payload_type_rtx_ &&
-        incoming_payload_type_ != -1) {
-      (*restored_packet)[1] = static_cast<uint8_t>(incoming_payload_type_);
-      if (header.markerBit) {
-        (*restored_packet)[1] |= kRtpMarkerBitMask;  // Marker bit is set.
-      }
-    } else {
+  int associated_payload_type;
+  auto apt_mapping = rtx_payload_type_map_.find(header.payloadType);
+  if (use_rtx_payload_mapping_on_restore_ &&
+      apt_mapping != rtx_payload_type_map_.end()) {
+    associated_payload_type = apt_mapping->second;
+  } else {
+    // In the future, this will be a bug. For now, just assume this RTX packet
+    // matches the last non-RTX payload type we received. There are cases where
+    // this could break, especially where RTX is sent outside of NACKing (e.g.
+    // padding with redundant payloads).
+    if (rtx_payload_type_ == -1 || incoming_payload_type_ == -1) {
       LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet.";
       return false;
     }
+    associated_payload_type = incoming_payload_type_;
+  }
+
+  restored_packet[1] = static_cast<uint8_t>(associated_payload_type);
+  if (header.markerBit) {
+    restored_packet[1] |= kRtpMarkerBitMask;  // Marker bit is set.
   }
   return true;
 }
@@ -284,11 +305,17 @@ bool RTPPayloadRegistry::GetRtxSsrc(uint32_t* ssrc) const {
   return rtx_;
 }
 
-void RTPPayloadRegistry::SetRtxPayloadType(int payload_type) {
+void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
+                                           int associated_payload_type) {
   CriticalSectionScoped cs(crit_sect_.get());
-  assert(payload_type >= 0);
-  payload_type_rtx_ = payload_type;
+  if (payload_type < 0) {
+    LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
+    return;
+  }
+
+  rtx_payload_type_map_[payload_type] = associated_payload_type;
   rtx_ = true;
+  rtx_payload_type_ = payload_type;
 }
 
 bool RTPPayloadRegistry::IsRed(const RTPHeader& header) const {
@@ -316,17 +343,16 @@ bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
 
 int RTPPayloadRegistry::GetPayloadTypeFrequency(
     uint8_t payload_type) const {
-  RtpUtility::Payload* payload;
-  if (!PayloadTypeToPayload(payload_type, payload)) {
+  const RtpUtility::Payload* payload = PayloadTypeToPayload(payload_type);
+  if (!payload) {
     return -1;
   }
   CriticalSectionScoped cs(crit_sect_.get());
   return rtp_payload_strategy_->GetPayloadTypeFrequency(*payload);
 }
 
-bool RTPPayloadRegistry::PayloadTypeToPayload(
-    const uint8_t payload_type,
-    RtpUtility::Payload*& payload) const {
+const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
+    uint8_t payload_type) const {
   CriticalSectionScoped cs(crit_sect_.get());
 
   RtpUtility::PayloadTypeMap::const_iterator it =
@@ -334,11 +360,10 @@ bool RTPPayloadRegistry::PayloadTypeToPayload(
 
   // Check that this is a registered payload type.
   if (it == payload_type_map_.end()) {
-    return false;
+    return nullptr;
   }
 
-  payload = it->second;
-  return true;
+  return it->second;
 }
 
 void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
@@ -363,7 +388,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy {
 
   bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                            const uint32_t frequency,
-                           const uint8_t channels,
+                           const size_t channels,
                            const uint32_t rate) const override {
     return
         payload.audio &&
@@ -382,7 +407,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy {
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payloadType,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) const override {
     RtpUtility::Payload* payload = new RtpUtility::Payload;
     payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
@@ -406,7 +431,7 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy {
 
   bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                            const uint32_t frequency,
-                           const uint8_t channels,
+                           const size_t channels,
                            const uint32_t rate) const override {
     return !payload.audio;
   }
@@ -420,7 +445,7 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy {
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payloadType,
       const uint32_t frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) const override {
     RtpVideoCodecTypes videoType = kRtpVideoGeneric;
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 5026986858..b73666d1af 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -8,11 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
@@ -23,7 +25,7 @@ using ::testing::Return;
 using ::testing::_;
 
 static const char* kTypicalPayloadName = "name";
-static const uint8_t kTypicalChannels = 1;
+static const size_t kTypicalChannels = 1;
 static const int kTypicalFrequency = 44000;
 static const int kTypicalRate = 32 * 1024;
 
@@ -50,10 +52,9 @@ class RtpPayloadRegistryTest : public ::testing::Test {
     RtpUtility::Payload* returned_payload_on_heap =
         new RtpUtility::Payload(returned_payload);
     EXPECT_CALL(*mock_payload_strategy_,
-        CreatePayloadType(kTypicalPayloadName, payload_type,
-            kTypicalFrequency,
-            kTypicalChannels,
-            rate)).WillOnce(Return(returned_payload_on_heap));
+                CreatePayloadType(kTypicalPayloadName, payload_type,
+                                  kTypicalFrequency, kTypicalChannels, rate))
+        .WillOnce(Return(returned_payload_on_heap));
     return returned_payload_on_heap;
   }
 
@@ -68,14 +69,14 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) {
 
   bool new_payload_created = false;
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &new_payload_created));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &new_payload_created));
 
   EXPECT_TRUE(new_payload_created) << "A new payload WAS created.";
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                          retrieved_payload));
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+  EXPECT_TRUE(retrieved_payload);
 
   // We should get back the exact pointer to the payload returned by the
   // payload strategy.
@@ -83,32 +84,30 @@ TEST_F(RtpPayloadRegistryTest, RegistersAndRemembersPayloadsUntilDeregistered) {
 
   // Now forget about it and verify it's gone.
   EXPECT_EQ(0, rtp_payload_registry_->DeRegisterReceivePayload(payload_type));
-  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type, retrieved_payload));
+  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type));
 }
 
 TEST_F(RtpPayloadRegistryTest, AudioRedWorkProperly) {
   const uint8_t kRedPayloadType = 127;
   const int kRedSampleRate = 8000;
-  const int kRedChannels = 1;
+  const size_t kRedChannels = 1;
   const int kRedBitRate = 0;
 
   // This creates an audio RTP payload strategy.
-  rtp_payload_registry_.reset(new RTPPayloadRegistry(
-      RTPPayloadStrategy::CreateStrategy(true)));
+  rtp_payload_registry_.reset(
+      new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true)));
 
   bool new_payload_created = false;
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      "red", kRedPayloadType, kRedSampleRate, kRedChannels, kRedBitRate,
-      &new_payload_created));
+                   "red", kRedPayloadType, kRedSampleRate, kRedChannels,
+                   kRedBitRate, &new_payload_created));
   EXPECT_TRUE(new_payload_created);
 
   EXPECT_EQ(kRedPayloadType, rtp_payload_registry_->red_payload_type());
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(kRedPayloadType,
-                                                          retrieved_payload));
-  ASSERT_TRUE(retrieved_payload);
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(kRedPayloadType);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_TRUE(retrieved_payload->audio);
   EXPECT_STRCASEEQ("red", retrieved_payload->name);
 
@@ -125,27 +124,29 @@ TEST_F(RtpPayloadRegistryTest,
   RtpUtility::Payload* first_payload_on_heap =
       ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 
   EXPECT_EQ(-1, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored)) << "Adding same codec twice = bad.";
+                    kTypicalPayloadName, payload_type, kTypicalFrequency,
+                    kTypicalChannels, kTypicalRate, &ignored))
+      << "Adding same codec twice = bad.";
 
   RtpUtility::Payload* second_payload_on_heap =
       ExpectReturnOfTypicalAudioPayload(payload_type - 1, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
-      kTypicalChannels, kTypicalRate, &ignored)) <<
-          "With a different payload type is fine though.";
+                   kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored))
+      << "With a different payload type is fine though.";
 
   // Ensure both payloads are preserved.
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type,
-                                                          retrieved_payload));
+  const RtpUtility::Payload* retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_EQ(first_payload_on_heap, retrieved_payload);
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1,
-                                                          retrieved_payload));
+  retrieved_payload =
+      rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1);
+  EXPECT_TRUE(retrieved_payload);
   EXPECT_EQ(second_payload_on_heap, retrieved_payload);
 
   // Ok, update the rate for one of the codecs. If either the incoming rate or
@@ -156,8 +157,8 @@ TEST_F(RtpPayloadRegistryTest,
   EXPECT_CALL(*mock_payload_strategy_,
               UpdatePayloadRate(first_payload_on_heap, kTypicalRate));
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 }
 
 TEST_F(RtpPayloadRegistryTest,
@@ -172,35 +173,31 @@ TEST_F(RtpPayloadRegistryTest,
   bool ignored = false;
   ExpectReturnOfTypicalAudioPayload(payload_type, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
   ExpectReturnOfTypicalAudioPayload(payload_type - 1, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
-      kTypicalChannels, kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type - 1, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 
-  RtpUtility::Payload* retrieved_payload = NULL;
-  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type, retrieved_payload)) << "The first payload should be "
-          "deregistered because the only thing that differs is payload type.";
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type - 1, retrieved_payload)) <<
-          "The second payload should still be registered though.";
+  EXPECT_FALSE(rtp_payload_registry_->PayloadTypeToPayload(payload_type))
+      << "The first payload should be "
+         "deregistered because the only thing that differs is payload type.";
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1))
+      << "The second payload should still be registered though.";
 
   // Now ensure non-compatible codecs aren't removed.
   ON_CALL(*mock_payload_strategy_, PayloadIsCompatible(_, _, _, _))
       .WillByDefault(Return(false));
   ExpectReturnOfTypicalAudioPayload(payload_type + 1, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, payload_type + 1, kTypicalFrequency,
-      kTypicalChannels, kTypicalRate, &ignored));
+                   kTypicalPayloadName, payload_type + 1, kTypicalFrequency,
+                   kTypicalChannels, kTypicalRate, &ignored));
 
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type - 1, retrieved_payload)) <<
-          "Not compatible; both payloads should be kept.";
-  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(
-      payload_type + 1, retrieved_payload)) <<
-          "Not compatible; both payloads should be kept.";
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type - 1))
+      << "Not compatible; both payloads should be kept.";
+  EXPECT_TRUE(rtp_payload_registry_->PayloadTypeToPayload(payload_type + 1))
+      << "Not compatible; both payloads should be kept.";
 }
 
 TEST_F(RtpPayloadRegistryTest,
@@ -216,18 +213,17 @@ TEST_F(RtpPayloadRegistryTest,
   bool ignored;
   ExpectReturnOfTypicalAudioPayload(34, kTypicalRate);
   EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
-      kTypicalPayloadName, 34, kTypicalFrequency, kTypicalChannels,
-      kTypicalRate, &ignored));
+                   kTypicalPayloadName, 34, kTypicalFrequency, kTypicalChannels,
+                   kTypicalRate, &ignored));
 
   EXPECT_EQ(-1, rtp_payload_registry_->last_received_payload_type());
   media_type_unchanged = rtp_payload_registry_->ReportMediaPayloadType(18);
   EXPECT_FALSE(media_type_unchanged);
 }
 
-class ParameterizedRtpPayloadRegistryTest :
-    public RtpPayloadRegistryTest,
-    public ::testing::WithParamInterface<int> {
-};
+class ParameterizedRtpPayloadRegistryTest
+    : public RtpPayloadRegistryTest,
+      public ::testing::WithParamInterface<int> {};
 
 TEST_P(ParameterizedRtpPayloadRegistryTest,
        FailsToRegisterKnownPayloadsWeAreNotInterestedIn) {
@@ -235,29 +231,166 @@ TEST_P(ParameterizedRtpPayloadRegistryTest,
 
   bool ignored;
   EXPECT_EQ(-1, rtp_payload_registry_->RegisterReceivePayload(
-      "whatever", static_cast<uint8_t>(payload_type), 19, 1, 17, &ignored));
+                    "whatever", static_cast<uint8_t>(payload_type), 19, 1, 17,
+                    &ignored));
 }
 
 INSTANTIATE_TEST_CASE_P(TestKnownBadPayloadTypes,
                         ParameterizedRtpPayloadRegistryTest,
                         testing::Values(64, 72, 73, 74, 75, 76, 77, 78, 79));
 
-class RtpPayloadRegistryGenericTest :
-    public RtpPayloadRegistryTest,
-    public ::testing::WithParamInterface<int> {
-};
+class RtpPayloadRegistryGenericTest
+    : public RtpPayloadRegistryTest,
+      public ::testing::WithParamInterface<int> {};
 
 TEST_P(RtpPayloadRegistryGenericTest, RegisterGenericReceivePayloadType) {
   int payload_type = GetParam();
 
   bool ignored;
 
-  EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload("generic-codec",
-    static_cast<int8_t>(payload_type),
-    19, 1, 17, &ignored)); // dummy values, except for payload_type
+  EXPECT_EQ(0, rtp_payload_registry_->RegisterReceivePayload(
+                   "generic-codec", static_cast<int8_t>(payload_type), 19, 1,
+                   17, &ignored));  // dummy values, except for payload_type
 }
 
-INSTANTIATE_TEST_CASE_P(TestDynamicRange, RtpPayloadRegistryGenericTest,
-                        testing::Range(96, 127+1));
+// Generates an RTX packet for the given length and original sequence number.
+// The RTX sequence number and ssrc will use the default value of 9999. The
+// caller takes ownership of the returned buffer.
+const uint8_t* GenerateRtxPacket(size_t header_length,
+                                 size_t payload_length,
+                                 uint16_t original_sequence_number) {
+  uint8_t* packet =
+      new uint8_t[kRtxHeaderSize + header_length + payload_length]();
+  // Write the RTP version to the first byte, so the resulting header can be
+  // parsed.
+  static const int kRtpExpectedVersion = 2;
+  packet[0] = static_cast<uint8_t>(kRtpExpectedVersion << 6);
+  // Write a junk sequence number. It should be thrown away when the packet is
+  // restored.
+  ByteWriter<uint16_t>::WriteBigEndian(packet + 2, 9999);
+  // Write a junk ssrc. It should also be thrown away when the packet is
+  // restored.
+  ByteWriter<uint32_t>::WriteBigEndian(packet + 8, 9999);
+
+  // Now write the RTX header. It occurs at the start of the payload block, and
+  // contains just the sequence number.
+  ByteWriter<uint16_t>::WriteBigEndian(packet + header_length,
+                                       original_sequence_number);
+  return packet;
+}
+
+void TestRtxPacket(RTPPayloadRegistry* rtp_payload_registry,
+                   int rtx_payload_type,
+                   int expected_payload_type,
+                   bool should_succeed) {
+  size_t header_length = 100;
+  size_t payload_length = 200;
+  size_t original_length = header_length + payload_length + kRtxHeaderSize;
+
+  RTPHeader header;
+  header.ssrc = 1000;
+  header.sequenceNumber = 100;
+  header.payloadType = rtx_payload_type;
+  header.headerLength = header_length;
+
+  uint16_t original_sequence_number = 1234;
+  uint32_t original_ssrc = 500;
+
+  rtc::scoped_ptr<const uint8_t[]> packet(GenerateRtxPacket(
+      header_length, payload_length, original_sequence_number));
+  rtc::scoped_ptr<uint8_t[]> restored_packet(
+      new uint8_t[header_length + payload_length]);
+  size_t length = original_length;
+  bool success = rtp_payload_registry->RestoreOriginalPacket(
+      restored_packet.get(), packet.get(), &length, original_ssrc, header);
+  ASSERT_EQ(should_succeed, success)
+      << "Test success should match should_succeed.";
+  if (!success) {
+    return;
+  }
+
+  EXPECT_EQ(original_length - kRtxHeaderSize, length)
+      << "The restored packet should be exactly kRtxHeaderSize smaller.";
+
+  rtc::scoped_ptr<RtpHeaderParser> header_parser(RtpHeaderParser::Create());
+  RTPHeader restored_header;
+  ASSERT_TRUE(
+      header_parser->Parse(restored_packet.get(), length, &restored_header));
+  EXPECT_EQ(original_sequence_number, restored_header.sequenceNumber)
+      << "The restored packet should have the original sequence number "
+      << "in the correct location in the RTP header.";
+  EXPECT_EQ(expected_payload_type, restored_header.payloadType)
+      << "The restored packet should have the correct payload type.";
+  EXPECT_EQ(original_ssrc, restored_header.ssrc)
+      << "The restored packet should have the correct ssrc.";
+}
+
+TEST_F(RtpPayloadRegistryTest, MultipleRtxPayloadTypes) {
+  // Set the incoming payload type to 90.
+  RTPHeader header;
+  header.payloadType = 90;
+  header.ssrc = 1;
+  rtp_payload_registry_->SetIncomingPayloadType(header);
+  rtp_payload_registry_->SetRtxSsrc(100);
+  // Map two RTX payload types.
+  rtp_payload_registry_->SetRtxPayloadType(105, 95);
+  rtp_payload_registry_->SetRtxPayloadType(106, 96);
+  rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(true);
+
+  TestRtxPacket(rtp_payload_registry_.get(), 105, 95, true);
+  TestRtxPacket(rtp_payload_registry_.get(), 106, 96, true);
+
+  // If the option is off, the map will be ignored.
+  rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(false);
+  TestRtxPacket(rtp_payload_registry_.get(), 105, 90, true);
+  TestRtxPacket(rtp_payload_registry_.get(), 106, 90, true);
+}
+
+// TODO(holmer): Ignored by default for compatibility with misconfigured RTX
+// streams in Chrome. When that is fixed, remove this.
+TEST_F(RtpPayloadRegistryTest, IgnoresRtxPayloadTypeMappingByDefault) {
+  // Set the incoming payload type to 90.
+  RTPHeader header;
+  header.payloadType = 90;
+  header.ssrc = 1;
+  rtp_payload_registry_->SetIncomingPayloadType(header);
+  rtp_payload_registry_->SetRtxSsrc(100);
+  // Map two RTX payload types.
+  rtp_payload_registry_->SetRtxPayloadType(105, 95);
+  rtp_payload_registry_->SetRtxPayloadType(106, 96);
+
+  TestRtxPacket(rtp_payload_registry_.get(), 105, 90, true);
+  TestRtxPacket(rtp_payload_registry_.get(), 106, 90, true);
+}
+
+TEST_F(RtpPayloadRegistryTest, InferLastReceivedPacketIfPayloadTypeUnknown) {
+  rtp_payload_registry_->SetRtxSsrc(100);
+  // Set the incoming payload type to 90.
+  RTPHeader header;
+  header.payloadType = 90;
+  header.ssrc = 1;
+  rtp_payload_registry_->SetIncomingPayloadType(header);
+  rtp_payload_registry_->SetRtxPayloadType(105, 95);
+  rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(true);
+  // Mapping respected for known type.
+  TestRtxPacket(rtp_payload_registry_.get(), 105, 95, true);
+  // Mapping ignored for unknown type, even though the option is on.
+  TestRtxPacket(rtp_payload_registry_.get(), 106, 90, true);
+}
+
+TEST_F(RtpPayloadRegistryTest, InvalidRtxConfiguration) {
+  rtp_payload_registry_->SetRtxSsrc(100);
+  // Fails because no mappings exist and the incoming payload type isn't known.
+  TestRtxPacket(rtp_payload_registry_.get(), 105, 0, false);
+  // Succeeds when the mapping is used, but fails for the implicit fallback.
+  rtp_payload_registry_->SetRtxPayloadType(105, 95);
+  rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(true);
+  TestRtxPacket(rtp_payload_registry_.get(), 105, 95, true);
+  TestRtxPacket(rtp_payload_registry_.get(), 106, 0, false);
+}
+
+INSTANTIATE_TEST_CASE_P(TestDynamicRange,
+                        RtpPayloadRegistryGenericTest,
+                        testing::Range(96, 127 + 1));
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index e19378aa2e..c4c7dbb4cd 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -14,23 +14,21 @@
 #include <math.h>   // pow()
 #include <string.h>  // memcpy()
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
-    int32_t id, RtpData* data_callback,
+    RtpData* data_callback,
     RtpAudioFeedback* incoming_messages_callback) {
-  return new RTPReceiverAudio(id, data_callback, incoming_messages_callback);
+  return new RTPReceiverAudio(data_callback, incoming_messages_callback);
 }
 
-RTPReceiverAudio::RTPReceiverAudio(const int32_t id,
-                                   RtpData* data_callback,
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
                                    RtpAudioFeedback* incoming_messages_callback)
     : RTPReceiverStrategy(data_callback),
       TelephoneEventHandler(),
-      id_(id),
       last_received_frequency_(8000),
       telephone_event_forward_to_decoder_(false),
       telephone_event_payload_type_(-1),
@@ -228,10 +226,8 @@ RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
 
 void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
                                            PayloadUnion* specific_payload,
-                                           bool* should_reset_statistics,
                                            bool* should_discard_changes) {
   *should_discard_changes = false;
-  *should_reset_statistics = false;
 
   if (TelephoneEventPayloadType(payload_type)) {
     // Don't do callbacks for DTMF packets.
@@ -244,8 +240,6 @@ void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
                                             &specific_payload->Audio.frequency,
                                             &cng_payload_type_has_changed);
 
-  *should_reset_statistics = cng_payload_type_has_changed;
-
   if (is_cng_payload_type) {
     // Don't do callbacks for DTMF packets.
     *should_discard_changes = true;
@@ -267,16 +261,13 @@ int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
 
 int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
     RtpFeedback* callback,
-    int32_t id,
     int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const PayloadUnion& specific_payload) const {
-  if (-1 == callback->OnInitializeDecoder(id,
-                                          payload_type,
-                                          payload_name,
-                                          specific_payload.Audio.frequency,
-                                          specific_payload.Audio.channels,
-                                          specific_payload.Audio.rate)) {
+  if (-1 ==
+      callback->OnInitializeDecoder(
+          payload_type, payload_name, specific_payload.Audio.frequency,
+          specific_payload.Audio.channels, specific_payload.Audio.rate)) {
     LOG(LS_ERROR) << "Failed to create decoder for payload type: "
                   << payload_name << "/" << static_cast<int>(payload_type);
     return -1;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index cd146b9332..1dd07d1cc9 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -14,8 +14,8 @@
 #include <set>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
@@ -28,8 +28,7 @@ class CriticalSectionWrapper;
 class RTPReceiverAudio : public RTPReceiverStrategy,
                          public TelephoneEventHandler {
  public:
-  RTPReceiverAudio(const int32_t id,
-                   RtpData* data_callback,
+  RTPReceiverAudio(RtpData* data_callback,
                    RtpAudioFeedback* incoming_messages_callback);
   virtual ~RTPReceiverAudio() {}
 
@@ -43,9 +42,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
   // Is TelephoneEvent configured with payload type payload_type
   bool TelephoneEventPayloadType(const int8_t payload_type) const;
 
-  TelephoneEventHandler* GetTelephoneEventHandler() {
-    return this;
-  }
+  TelephoneEventHandler* GetTelephoneEventHandler() { return this; }
 
   // Returns true if CNG is configured with payload type payload_type. If so,
   // the frequency and cng_payload_type_has_changed are filled in.
@@ -74,7 +71,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
 
   int32_t InvokeOnInitializeDecoder(
       RtpFeedback* callback,
-      int32_t id,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const override;
@@ -93,21 +89,16 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
   // statistics. In addition we sometimes need to tweak the frequency.
   void CheckPayloadChanged(int8_t payload_type,
                            PayloadUnion* specific_payload,
-                           bool* should_reset_statistics,
                            bool* should_discard_changes) override;
 
   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
  private:
-
-  int32_t ParseAudioCodecSpecific(
-      WebRtcRTPHeader* rtp_header,
-      const uint8_t* payload_data,
-      size_t payload_length,
-      const AudioPayload& audio_specific,
-      bool is_red);
-
-  int32_t id_;
+  int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
+                                  const uint8_t* payload_data,
+                                  size_t payload_length,
+                                  const AudioPayload& audio_specific,
+                                  bool is_red);
 
   uint32_t last_received_frequency_;
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 1c14ec365c..c7af3611b0 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -15,19 +15,18 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
-using RtpUtility::GetCurrentRTP;
 using RtpUtility::Payload;
 using RtpUtility::StringCompare;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
-    int id, Clock* clock,
+    Clock* clock,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
@@ -36,13 +35,13 @@ RtpReceiver* RtpReceiver::CreateVideoReceiver(
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
-      id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
+      clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
       rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
-    int id, Clock* clock,
+    Clock* clock,
     RtpAudioFeedback* incoming_audio_feedback,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
@@ -54,25 +53,24 @@ RtpReceiver* RtpReceiver::CreateAudioReceiver(
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
-      id, clock, incoming_audio_feedback, incoming_messages_callback,
+      clock, incoming_audio_feedback, incoming_messages_callback,
       rtp_payload_registry,
-      RTPReceiverStrategy::CreateAudioStrategy(id, incoming_payload_callback,
+      RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
                                                incoming_audio_feedback));
 }
 
-RtpReceiverImpl::RtpReceiverImpl(int32_t id,
-                         Clock* clock,
-                         RtpAudioFeedback* incoming_audio_messages_callback,
-                         RtpFeedback* incoming_messages_callback,
-                         RTPPayloadRegistry* rtp_payload_registry,
-                         RTPReceiverStrategy* rtp_media_receiver)
+RtpReceiverImpl::RtpReceiverImpl(
+    Clock* clock,
+    RtpAudioFeedback* incoming_audio_messages_callback,
+    RtpFeedback* incoming_messages_callback,
+    RTPPayloadRegistry* rtp_payload_registry,
+    RTPReceiverStrategy* rtp_media_receiver)
     : clock_(clock),
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
-      id_(id),
       cb_rtp_feedback_(incoming_messages_callback),
       critical_section_rtp_receiver_(
-        CriticalSectionWrapper::CreateCriticalSection()),
+          CriticalSectionWrapper::CreateCriticalSection()),
       last_receive_time_(0),
       last_received_payload_length_(0),
       ssrc_(0),
@@ -91,8 +89,7 @@ RtpReceiverImpl::RtpReceiverImpl(int32_t id,
 
 RtpReceiverImpl::~RtpReceiverImpl() {
   for (int i = 0; i < num_csrcs_; ++i) {
-    cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
-                                            false);
+    cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
   }
 }
 
@@ -100,7 +97,7 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
@@ -181,13 +178,9 @@ bool RtpReceiverImpl::IncomingRtpPacket(
 
   int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
   bool is_red = false;
-  bool should_reset_statistics = false;
 
-  if (CheckPayloadChanged(rtp_header,
-                          first_payload_byte,
-                          is_red,
-                          &payload_specific,
-                          &should_reset_statistics) == -1) {
+  if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
+                          &payload_specific) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
       return true;
@@ -196,10 +189,6 @@ bool RtpReceiverImpl::IncomingRtpPacket(
     return false;
   }
 
-  if (should_reset_statistics) {
-    cb_rtp_feedback_->ResetStatistics(ssrc_);
-  }
-
   WebRtcRTPHeader webrtc_rtp_header;
   memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
   webrtc_rtp_header.header = rtp_header;
@@ -279,7 +268,7 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   bool new_ssrc = false;
   bool re_initialize_decoder = false;
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
-  uint8_t channels = 1;
+  size_t channels = 1;
   uint32_t rate = 0;
 
   {
@@ -292,8 +281,6 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
       // We need the payload_type_ to make the call if the remote SSRC is 0.
       new_ssrc = true;
 
-      cb_rtp_feedback_->ResetStatistics(ssrc_);
-
       last_received_timestamp_ = 0;
       last_received_sequence_number_ = 0;
       last_received_frame_time_ms_ = -1;
@@ -304,12 +291,11 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
         if (rtp_header.payloadType == last_received_payload_type) {
           re_initialize_decoder = true;
 
-          Payload* payload;
-          if (!rtp_payload_registry_->PayloadTypeToPayload(
-              rtp_header.payloadType, payload)) {
+          const Payload* payload = rtp_payload_registry_->PayloadTypeToPayload(
+              rtp_header.payloadType);
+          if (!payload) {
             return;
           }
-          assert(payload);
           payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
           strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
           if (payload->audio) {
@@ -325,13 +311,14 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   if (new_ssrc) {
     // We need to get this to our RTCP sender and receiver.
     // We need to do this outside critical section.
-    cb_rtp_feedback_->OnIncomingSSRCChanged(id_, rtp_header.ssrc);
+    cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc);
   }
 
   if (re_initialize_decoder) {
-    if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
-        id_, rtp_header.payloadType, payload_name,
-        rtp_header.payload_type_frequency, channels, rate)) {
+    if (-1 ==
+        cb_rtp_feedback_->OnInitializeDecoder(
+            rtp_header.payloadType, payload_name,
+            rtp_header.payload_type_frequency, channels, rate)) {
       // New stream, same codec.
       LOG(LS_ERROR) << "Failed to create decoder for payload type: "
                     << static_cast<int>(rtp_header.payloadType);
@@ -346,12 +333,10 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
 // this code path moves we can get rid of some of the rtp_receiver ->
 // media_specific interface (such as CheckPayloadChange, possibly get/set
 // last known payload).
-int32_t RtpReceiverImpl::CheckPayloadChanged(
-  const RTPHeader& rtp_header,
-  const int8_t first_payload_byte,
-  bool& is_red,
-  PayloadUnion* specific_payload,
-  bool* should_reset_statistics) {
+int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
+                                             const int8_t first_payload_byte,
+                                             bool* is_red,
+                                             PayloadUnion* specific_payload) {
   bool re_initialize_decoder = false;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
@@ -368,7 +353,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
       if (rtp_payload_registry_->red_payload_type() == payload_type) {
         // Get the real codec payload type.
         payload_type = first_payload_byte & 0x7f;
-        is_red = true;
+        *is_red = true;
 
         if (rtp_payload_registry_->red_payload_type() == payload_type) {
           // Invalid payload type, traced by caller. If we proceeded here,
@@ -383,24 +368,23 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
           return 0;
         }
       }
-      *should_reset_statistics = false;
       bool should_discard_changes = false;
 
       rtp_media_receiver_->CheckPayloadChanged(
-        payload_type, specific_payload, should_reset_statistics,
+        payload_type, specific_payload,
         &should_discard_changes);
 
       if (should_discard_changes) {
-        is_red = false;
+        *is_red = false;
         return 0;
       }
 
-      Payload* payload;
-      if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
+      const Payload* payload =
+          rtp_payload_registry_->PayloadTypeToPayload(payload_type);
+      if (!payload) {
         // Not a registered payload type.
         return -1;
       }
-      assert(payload);
       payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
       strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
 
@@ -419,19 +403,16 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
           re_initialize_decoder = false;
         }
       }
-      if (re_initialize_decoder) {
-        *should_reset_statistics = true;
-      }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
-      is_red = false;
+      *is_red = false;
     }
   }  // End critsect.
 
   if (re_initialize_decoder) {
-    if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
-        cb_rtp_feedback_, id_, payload_type, payload_name,
-        *specific_payload)) {
+    if (-1 ==
+        rtp_media_receiver_->InvokeOnInitializeDecoder(
+            cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) {
       return -1;  // Wrong payload type.
     }
   }
@@ -488,7 +469,7 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
     if (!found_match && csrc) {
       // Didn't find it, report it as new.
       have_called_callback = true;
-      cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, true);
+      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true);
     }
   }
   // Search for old CSRC in new array.
@@ -505,7 +486,7 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
     if (!found_match && csrc) {
       // Did not find it, report as removed.
       have_called_callback = true;
-      cb_rtp_feedback_->OnIncomingCSRCChanged(id_, csrc, false);
+      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false);
     }
   }
   if (!have_called_callback) {
@@ -513,9 +494,9 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
     // Using CSRC 0 to signal this event, not interop safe, other
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
-      cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, true);
+      cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
-      cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
+      cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 33c7fdad47..e80a31e251 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -12,10 +12,10 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -25,8 +25,7 @@ class RtpReceiverImpl : public RtpReceiver {
   // Callbacks passed in here may not be NULL (use Null Object callbacks if you
   // want callbacks to do nothing). This class takes ownership of the media
   // receiver but nothing else.
-  RtpReceiverImpl(int32_t id,
-                  Clock* clock,
+  RtpReceiverImpl(Clock* clock,
                   RtpAudioFeedback* incoming_audio_messages_callback,
                   RtpFeedback* incoming_messages_callback,
                   RTPPayloadRegistry* rtp_payload_registry,
@@ -37,7 +36,7 @@ class RtpReceiverImpl : public RtpReceiver {
   int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                                  const int8_t payload_type,
                                  const uint32_t frequency,
-                                 const uint8_t channels,
+                                 const size_t channels,
                                  const uint32_t rate) override;
 
   int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
@@ -74,16 +73,13 @@ class RtpReceiverImpl : public RtpReceiver {
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
-                              bool& is_red,
-                              PayloadUnion* payload,
-                              bool* should_reset_statistics);
+                              bool* is_red,
+                              PayloadUnion* payload);
 
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
   rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
 
-  int32_t id_;
-
   RtpFeedback* cb_rtp_feedback_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
index 56dd081fc8..3797b1bcc2 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
@@ -12,7 +12,7 @@
 
 #include <stdlib.h>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -36,11 +36,9 @@ void RTPReceiverStrategy::SetLastMediaSpecificPayload(
 
 void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type,
                                               PayloadUnion* specific_payload,
-                                              bool* should_reset_statistics,
                                               bool* should_discard_changes) {
-  // Default: Keep changes and don't reset statistics.
+  // Default: Keep changes.
   *should_discard_changes = false;
-  *should_reset_statistics = false;
 }
 
 int RTPReceiverStrategy::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index b34ad38cdb..0f7ad30e87 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -12,10 +12,10 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -28,7 +28,7 @@ class RTPReceiverStrategy {
  public:
   static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
   static RTPReceiverStrategy* CreateAudioStrategy(
-      int32_t id, RtpData* data_callback,
+      RtpData* data_callback,
       RtpAudioFeedback* incoming_messages_callback);
 
   virtual ~RTPReceiverStrategy() {}
@@ -70,7 +70,6 @@ class RTPReceiverStrategy {
   // Invokes the OnInitializeDecoder callback in a media-specific way.
   virtual int32_t InvokeOnInitializeDecoder(
       RtpFeedback* callback,
-      int32_t id,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const = 0;
@@ -79,7 +78,6 @@ class RTPReceiverStrategy {
   // reset statistics and/or discard this packet.
   virtual void CheckPayloadChanged(int8_t payload_type,
                                    PayloadUnion* specific_payload,
-                                   bool* should_reset_statistics,
                                    bool* should_discard_changes);
 
   virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
@@ -97,7 +95,7 @@ class RTPReceiverStrategy {
   // Note: Implementations may call the callback for other reasons than calls
   // to ParseRtpPacket, for instance if the implementation somehow recovers a
   // packet.
-  RTPReceiverStrategy(RtpData* data_callback);
+  explicit RTPReceiverStrategy(RtpData* data_callback);
 
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   PayloadUnion last_payload_;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index fb690c3f7d..dadca4cf2f 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -19,15 +19,16 @@
 #include <arpa/inet.h>
 #endif
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -67,6 +68,7 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                rtp_header->header.timestamp);
   rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
 
+  RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
   const size_t payload_data_length =
       payload_length - rtp_header->header.paddingLength;
 
@@ -116,14 +118,13 @@ RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
 
 int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
     RtpFeedback* callback,
-    int32_t id,
     int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const PayloadUnion& specific_payload) const {
   // For video we just go with default values.
   if (-1 ==
-      callback->OnInitializeDecoder(
-          id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
+      callback->OnInitializeDecoder(payload_type, payload_name,
+                                    kVideoPayloadTypeFrequency, 1, 0)) {
     LOG(LS_ERROR) << "Failed to created decoder for payload type: "
                   << static_cast<int>(payload_type);
     return -1;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 8528a7d6b7..56f761a2e1 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -12,7 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -49,7 +49,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
 
   int32_t InvokeOnInitializeDecoder(
       RtpFeedback* callback,
-      int32_t id,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const override;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
index 2acc77a4ec..68e512bdd8 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -13,37 +13,38 @@
 
 // Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
 namespace webrtc {
-enum { NACK_BYTECOUNT_SIZE      = 60};   // size of our NACK history
+enum { NACK_BYTECOUNT_SIZE = 60 };  // size of our NACK history
 // A sanity for the NACK list parsing at the send-side.
 enum { kSendSideNackListSizeSanity = 20000 };
 enum { kDefaultMaxReorderingThreshold = 50 };  // In sequence numbers.
 enum { kRtcpMaxNackFields = 253 };
 
-enum { RTCP_INTERVAL_VIDEO_MS       = 1000 };
-enum { RTCP_INTERVAL_AUDIO_MS       = 5000 };
+enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
+enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
 enum { RTCP_INTERVAL_RAPID_SYNC_MS = 100 }; // RFX 6051
-enum { RTCP_SEND_BEFORE_KEY_FRAME_MS= 100 };
-enum { RTCP_MAX_REPORT_BLOCKS       = 31};      // RFC 3550 page 37
-enum { RTCP_MIN_FRAME_LENGTH_MS     = 17};
-enum { kRtcpAppCode_DATA_SIZE           = 32*4};    // multiple of 4, this is not a limitation of the size
-enum { RTCP_RPSI_DATA_SIZE          = 30};
-enum { RTCP_NUMBER_OF_SR            = 60 };
+enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
+enum { RTCP_MAX_REPORT_BLOCKS = 31 };  // RFC 3550 page 37
+enum { RTCP_MIN_FRAME_LENGTH_MS = 17 };
+enum {
+  kRtcpAppCode_DATA_SIZE = 32 * 4
+};  // multiple of 4, this is not a limitation of the size
+enum { RTCP_RPSI_DATA_SIZE = 30 };
+enum { RTCP_NUMBER_OF_SR = 60 };
 
-enum { MAX_NUMBER_OF_TEMPORAL_ID    = 8 };          // RFC
-enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID  = 128 };// RFC
+enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 };              // RFC
+enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };  // RFC
 enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
 
-enum { BW_HISTORY_SIZE          = 35};
+enum { BW_HISTORY_SIZE = 35 };
 
-#define MIN_AUDIO_BW_MANAGEMENT_BITRATE   6
-#define MIN_VIDEO_BW_MANAGEMENT_BITRATE   30
+#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
+#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
 
-enum { DTMF_OUTBAND_MAX         = 20};
+enum { DTMF_OUTBAND_MAX = 20 };
 
 enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
 enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
-enum { RTP_MAX_PACKETS_PER_FRAME= 512 }; // must be multiple of 32
+enum { RTP_MAX_PACKETS_PER_FRAME = 512 };  // must be multiple of 32
 }  // namespace webrtc
 
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index cb661f1d8b..d8e708d72d 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -15,9 +15,9 @@
 #include <set>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #ifdef _WIN32
 // Disable warning C4355: 'this' : used in base member initializer list.
@@ -27,22 +27,23 @@
 namespace webrtc {
 
 RtpRtcp::Configuration::Configuration()
-    : id(-1),
-      audio(false),
-      clock(NULL),
+    : audio(false),
+      receiver_only(false),
+      clock(nullptr),
       receive_statistics(NullObjectReceiveStatistics()),
-      outgoing_transport(NULL),
-      intra_frame_callback(NULL),
-      bandwidth_callback(NULL),
-      rtt_stats(NULL),
-      rtcp_packet_type_counter_observer(NULL),
+      outgoing_transport(nullptr),
+      intra_frame_callback(nullptr),
+      bandwidth_callback(nullptr),
+      transport_feedback_callback(nullptr),
+      rtt_stats(nullptr),
+      rtcp_packet_type_counter_observer(nullptr),
       audio_messages(NullObjectRtpAudioFeedback()),
-      remote_bitrate_estimator(NULL),
-      paced_sender(NULL),
-      send_bitrate_observer(NULL),
-      send_frame_count_observer(NULL),
-      send_side_delay_observer(NULL) {
-}
+      remote_bitrate_estimator(nullptr),
+      paced_sender(nullptr),
+      transport_sequence_number_allocator(nullptr),
+      send_bitrate_observer(nullptr),
+      send_frame_count_observer(nullptr),
+      send_side_delay_observer(nullptr) {}
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
@@ -58,49 +59,47 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
 }
 
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
-    : rtp_sender_(configuration.id,
-                  configuration.audio,
+    : rtp_sender_(configuration.audio,
                   configuration.clock,
                   configuration.outgoing_transport,
                   configuration.audio_messages,
                   configuration.paced_sender,
+                  configuration.transport_sequence_number_allocator,
+                  configuration.transport_feedback_callback,
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer),
-      rtcp_sender_(configuration.id,
-                   configuration.audio,
+      rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
-                   configuration.rtcp_packet_type_counter_observer),
-      rtcp_receiver_(configuration.id,
-                     configuration.clock,
+                   configuration.rtcp_packet_type_counter_observer,
+                   configuration.outgoing_transport),
+      rtcp_receiver_(configuration.clock,
+                     configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
                      configuration.intra_frame_callback,
+                     configuration.transport_feedback_callback,
                      this),
       clock_(configuration.clock),
-      id_(configuration.id),
       audio_(configuration.audio),
       collision_detected_(false),
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
-      packet_overhead_(28),  // IPV4 UDP.
+      packet_overhead_(28),                     // IPV4 UDP.
       padding_index_(static_cast<size_t>(-1)),  // Start padding at first child.
       nack_method_(kNackOff),
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
-      key_frame_req_method_(kKeyFrameReqFirRtp),
+      key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
       rtt_ms_(0) {
   send_video_codec_.codecType = kVideoCodecUnknown;
 
-  // TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
-  rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
-
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
@@ -184,8 +183,13 @@ int32_t ModuleRtpRtcpImpl::Process() {
       set_rtt_ms(rtt_stats_->LastProcessedRtt());
   }
 
-  if (rtcp_sender_.TimeToSendRTCPReport())
-    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+  // For sending streams, make sure to not send a SR before media has been sent.
+  if (rtcp_sender_.TimeToSendRTCPReport()) {
+    RTCPSender::FeedbackState state = GetFeedbackState();
+    // Prevent sending streams to send SR before any media has been sent.
+    if (!rtcp_sender_.Sending() || state.packets_sent > 0)
+      rtcp_sender_.SendRTCP(state, kRtcpReport);
+  }
 
   if (UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out
@@ -206,8 +210,13 @@ void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
   rtp_sender_.SetRtxSsrc(ssrc);
 }
 
-void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) {
-  rtp_sender_.SetRtxPayloadType(payload_type);
+void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
+                                              int associated_payload_type) {
+  rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type);
+}
+
+std::pair<int, int> ModuleRtpRtcpImpl::RtxSendPayloadType() const {
+  return rtp_sender_.RtxPayloadType();
 }
 
 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
@@ -403,12 +412,13 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_hdr) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
+  // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
   return rtp_sender_.SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_size, fragmentation, NULL, rtp_video_hdr);
+      payload_size, fragmentation, rtp_video_hdr);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
@@ -427,13 +437,6 @@ size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes) {
   return rtp_sender_.TimeToSendPadding(bytes);
 }
 
-bool ModuleRtpRtcpImpl::GetSendSideDelay(int* avg_send_delay_ms,
-                                         int* max_send_delay_ms) const {
-  DCHECK(avg_send_delay_ms);
-  DCHECK(max_send_delay_ms);
-  return rtp_sender_.GetSendSideDelay(avg_send_delay_ms, max_send_delay_ms);
-}
-
 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const {
   return rtp_sender_.MaxPayloadLength();
 }
@@ -477,39 +480,29 @@ int32_t ModuleRtpRtcpImpl::SetTransportOverhead(
 }
 
 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
-  if (mtu > IP_PACKET_SIZE) {
-    LOG(LS_ERROR) << "Invalid mtu: " << mtu;
-    return -1;
-  }
+  RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu;
   return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
                                          packet_overhead_);
 }
 
-RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
-  if (rtcp_sender_.Status() != kRtcpOff) {
+RtcpMode ModuleRtpRtcpImpl::RTCP() const {
+  if (rtcp_sender_.Status() != RtcpMode::kOff) {
     return rtcp_receiver_.Status();
   }
-  return kRtcpOff;
+  return RtcpMode::kOff;
 }
 
 // Configure RTCP status i.e on/off.
-void ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) {
+void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
   rtcp_sender_.SetRTCPStatus(method);
   rtcp_receiver_.SetRTCPStatus(method);
 }
 
-// Only for internal test.
-uint32_t ModuleRtpRtcpImpl::LastSendReport(
-    int64_t& last_rtcptime) {
-  return rtcp_sender_.LastSendReport(last_rtcptime);
-}
-
-int32_t ModuleRtpRtcpImpl::SetCNAME(const char c_name[RTCP_CNAME_SIZE]) {
+int32_t ModuleRtpRtcpImpl::SetCNAME(const char* c_name) {
   return rtcp_sender_.SetCNAME(c_name);
 }
 
-int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc,
-                                         const char c_name[RTCP_CNAME_SIZE]) {
+int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) {
   return rtcp_sender_.AddMixedCNAME(ssrc, c_name);
 }
 
@@ -552,29 +545,17 @@ int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
   return ret;
 }
 
-int32_t
-ModuleRtpRtcpImpl::GetReportBlockInfo(const uint32_t remote_ssrc,
-                                      uint32_t* ntp_high,
-                                      uint32_t* ntp_low,
-                                      uint32_t* packets_received,
-                                      uint64_t* octets_received) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemotePacketsReceived()");
-
-  return rtcp_receiver_.GetReportBlockInfo(remote_ssrc,
-                                           ntp_high, ntp_low,
-                                           packets_received, octets_received);
-}
-
-// Reset RTP data counters for the sending side.
-int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
-  rtp_sender_.ResetDataCounters();
-  return 0;  // TODO(pwestin): change to void.
+// Force a send of an RTCP packet.
+// Normal SR and RR are triggered via the process function.
+int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
 }
 
 // Force a send of an RTCP packet.
 // Normal SR and RR are triggered via the process function.
-int32_t ModuleRtpRtcpImpl::SendRTCP(uint32_t rtcp_packet_type) {
-  return rtcp_sender_.SendRTCP(GetFeedbackState(), rtcp_packet_type);
+int32_t ModuleRtpRtcpImpl::SendCompoundRTCP(
+    const std::set<RTCPPacketType>& packet_types) {
+  return rtcp_sender_.SendCompoundRTCP(GetFeedbackState(), packet_types);
 }
 
 int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
@@ -628,6 +609,42 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
   rtp_sender_.GetDataCounters(rtp_counters, rtx_counters);
 }
 
+void ModuleRtpRtcpImpl::GetRtpPacketLossStats(
+    bool outgoing,
+    uint32_t ssrc,
+    struct RtpPacketLossStats* loss_stats) const {
+  if (!loss_stats) return;
+  const PacketLossStats* stats_source = NULL;
+  if (outgoing) {
+    if (SSRC() == ssrc) {
+      stats_source = &send_loss_stats_;
+    }
+  } else {
+    if (rtcp_receiver_.RemoteSSRC() == ssrc) {
+      stats_source = &receive_loss_stats_;
+    }
+  }
+  if (stats_source) {
+    loss_stats->single_packet_loss_count =
+        stats_source->GetSingleLossCount();
+    loss_stats->multiple_packet_loss_event_count =
+        stats_source->GetMultipleLossEventCount();
+    loss_stats->multiple_packet_loss_packet_count =
+        stats_source->GetMultipleLossPacketCount();
+  }
+}
+ 
+int32_t
+ModuleRtpRtcpImpl::GetReportBlockInfo(const uint32_t remote_ssrc,
+                                      uint32_t* ntp_high,
+                                      uint32_t* ntp_low,
+                                      uint32_t* packets_received,
+                                      uint64_t* octets_received) const {
+  return rtcp_receiver_.GetReportBlockInfo(remote_ssrc,
+                                           ntp_high, ntp_low,
+                                           packets_received, octets_received);
+}
+
 int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) {
   return rtcp_receiver_.SenderInfoReceived(sender_info);
 }
@@ -638,17 +655,6 @@ int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
   return rtcp_receiver_.StatisticsReceived(receive_blocks);
 }
 
-int32_t ModuleRtpRtcpImpl::AddRTCPReportBlock(
-    const uint32_t ssrc,
-    const RTCPReportBlock* report_block) {
-  return rtcp_sender_.AddExternalReportBlock(ssrc, report_block);
-}
-
-int32_t ModuleRtpRtcpImpl::RemoveRTCPReportBlock(
-  const uint32_t ssrc) {
-  return rtcp_sender_.RemoveExternalReportBlock(ssrc);
-}
-
 // (REMB) Receiver Estimated Max Bitrate.
 bool ModuleRtpRtcpImpl::REMB() const {
   return rtcp_sender_.REMB();
@@ -663,15 +669,6 @@ void ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate,
   rtcp_sender_.SetREMBData(bitrate, ssrcs);
 }
 
-// (IJ) Extended jitter report.
-bool ModuleRtpRtcpImpl::IJ() const {
-  return rtcp_sender_.IJ();
-}
-
-void ModuleRtpRtcpImpl::SetIJStatus(const bool enable) {
-  rtcp_sender_.SetIJStatus(enable);
-}
-
 int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
     const RTPExtensionType type,
     const uint8_t id) {
@@ -712,6 +709,9 @@ int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
 // Send a Negative acknowledgment packet.
 int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
                                     const uint16_t size) {
+  for (int i = 0; i < size; ++i) {
+    receive_loss_stats_.AddLostPacket(nack_list[i]);
+  }
   uint16_t nack_length = size;
   uint16_t start_id = 0;
   int64_t now = clock_->TimeInMilliseconds();
@@ -784,6 +784,11 @@ RtcpStatisticsCallback* ModuleRtpRtcpImpl::GetRtcpStatisticsCallback() {
   return rtcp_receiver_.GetRtcpStatisticsCallback();
 }
 
+bool ModuleRtpRtcpImpl::SendFeedbackPacket(
+    const rtcp::TransportFeedback& packet) {
+  return rtcp_sender_.SendFeedbackPacket(packet);
+}
+
 // Send a TelephoneEvent tone using RFC 2833 (4733).
 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
     const uint8_t key,
@@ -811,9 +816,8 @@ int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
 }
 
 // Get payload type for Redundant Audio Data RFC 2198.
-int32_t ModuleRtpRtcpImpl::SendREDPayloadType(
-    int8_t& payload_type) const {
-  return rtp_sender_.RED(&payload_type);
+int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const {
+  return rtp_sender_.RED(payload_type);
 }
 
 void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) {
@@ -828,8 +832,6 @@ int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
 
 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
   switch (key_frame_req_method_) {
-    case kKeyFrameReqFirRtp:
-      return rtp_sender_.SendRTPIntraRequest();
     case kKeyFrameReqPliRtcp:
       return SendRTCP(kRtcpPli);
     case kKeyFrameReqFirRtcp:
@@ -844,21 +846,17 @@ int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(
       GetFeedbackState(), kRtcpSli, 0, 0, false, picture_id);
 }
 
-int32_t ModuleRtpRtcpImpl::SetGenericFECStatus(
+void ModuleRtpRtcpImpl::SetGenericFECStatus(
     const bool enable,
     const uint8_t payload_type_red,
     const uint8_t payload_type_fec) {
-  return rtp_sender_.SetGenericFECStatus(enable,
-                                         payload_type_red,
-                                         payload_type_fec);
+  rtp_sender_.SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
-int32_t ModuleRtpRtcpImpl::GenericFECStatus(
-    bool& enable,
-    uint8_t& payload_type_red,
-    uint8_t& payload_type_fec) {
-  return rtp_sender_.GenericFECStatus(&enable, &payload_type_red,
-                                      &payload_type_fec);
+void ModuleRtpRtcpImpl::GenericFECStatus(bool* enable,
+                                         uint8_t* payload_type_red,
+                                         uint8_t* payload_type_fec) {
+  rtp_sender_.GenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 int32_t ModuleRtpRtcpImpl::SetFecParameters(
@@ -881,7 +879,7 @@ void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
       // Configured via API ignore.
       return;
     }
-    if (kRtcpOff != rtcp_sender_.Status()) {
+    if (RtcpMode::kOff != rtcp_sender_.Status()) {
       // Send RTCP bye on the current SSRC.
       SendRTCP(kRtcpBye);
     }
@@ -932,6 +930,9 @@ bool ModuleRtpRtcpImpl::SendTimeOfXrRrReport(
 
 void ModuleRtpRtcpImpl::OnReceivedNACK(
     const std::list<uint16_t>& nack_sequence_numbers) {
+  for (uint16_t nack_sequence_number : nack_sequence_numbers) {
+    send_loss_stats_.AddLostPacket(nack_sequence_number);
+  }
   if (!rtp_sender_.StorePackets() ||
       nack_sequence_numbers.size() == 0) {
     return;
@@ -976,8 +977,8 @@ bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
 }
 
 // Called from RTCPsender.
-int32_t ModuleRtpRtcpImpl::BoundingSet(bool& tmmbr_owner,
-                                       TMMBRSet*& bounding_set) {
+int32_t ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner,
+                                       TMMBRSet* bounding_set) {
   return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set);
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index e946860d9a..7f4faa66d1 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -12,10 +12,13 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
 #include <list>
+#include <set>
+#include <utility>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
@@ -90,7 +93,9 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   void SetRtxSsrc(uint32_t ssrc) override;
 
-  void SetRtxSendPayloadType(int payload_type) override;
+  void SetRtxSendPayloadType(int payload_type,
+                             int associated_payload_type) override;
+  std::pair<int, int> RtxSendPayloadType() const override;
 
   // Sends kRtcpByeCode when going from true to false.
   int32_t SetSendingStatus(bool sending) override;
@@ -122,19 +127,16 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // less than |bytes|.
   size_t TimeToSendPadding(size_t bytes) override;
 
-  bool GetSendSideDelay(int* avg_send_delay_ms,
-                        int* max_send_delay_ms) const override;
-
   // RTCP part.
 
   // Get RTCP status.
-  RTCPMethod RTCP() const override;
+  RtcpMode RTCP() const override;
 
   // Configure RTCP status i.e on/off.
-  void SetRTCPStatus(RTCPMethod method) override;
+  void SetRTCPStatus(RtcpMode method) override;
 
   // Set RTCP CName.
-  int32_t SetCNAME(const char c_name[RTCP_CNAME_SIZE]) override;
+  int32_t SetCNAME(const char* c_name) override;
 
   // Get remote CName.
   int32_t RemoteCNAME(uint32_t remote_ssrc,
@@ -147,8 +149,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
                     uint32_t* rtcp_arrival_time_frac,
                     uint32_t* rtcp_timestamp) const override;
 
-  int32_t AddMixedCNAME(uint32_t ssrc,
-                        const char c_name[RTCP_CNAME_SIZE]) override;
+  int32_t AddMixedCNAME(uint32_t ssrc, const char* c_name) override;
 
   int32_t RemoveMixedCNAME(uint32_t ssrc) override;
 
@@ -159,17 +160,12 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
               int64_t* min_rtt,
               int64_t* max_rtt) const override;
 
-  virtual int32_t GetReportBlockInfo(const uint32_t remote_ssrc,
-                                     uint32_t* ntp_high,
-                                     uint32_t* ntp_low,
-                                     uint32_t* packets_received,
-                                     uint64_t* octets_received) const override;
-
   // Force a send of an RTCP packet.
   // Normal SR and RR are triggered via the process function.
-  int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) override;
+  int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
 
-  int32_t ResetSendDataCountersRTP() override;
+  int32_t SendCompoundRTCP(
+      const std::set<RTCPPacketType>& rtcpPacketTypes) override;
 
   // Statistics of the amount of data sent and received.
   int32_t DataCountersRTP(size_t* bytes_sent,
@@ -179,6 +175,17 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
       StreamDataCounters* rtp_counters,
       StreamDataCounters* rtx_counters) const override;
 
+  void GetRtpPacketLossStats(
+      bool outgoing,
+      uint32_t ssrc,
+      struct RtpPacketLossStats* loss_stats) const override;
+ 
+  int32_t GetReportBlockInfo(const uint32_t remote_ssrc,
+                             uint32_t* ntp_high,
+                             uint32_t* ntp_low,
+                             uint32_t* packets_received,
+                             uint64_t* octets_received) const override;
+
   // Get received RTCP report, sender info.
   int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) override;
 
@@ -186,12 +193,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   int32_t RemoteRTCPStat(
       std::vector<RTCPReportBlock>* receive_blocks) const override;
 
-  // Set received RTCP report block.
-  int32_t AddRTCPReportBlock(uint32_t ssrc,
-                             const RTCPReportBlock* receive_block) override;
-
-  int32_t RemoveRTCPReportBlock(uint32_t ssrc) override;
-
   // (REMB) Receiver Estimated Max Bitrate.
   bool REMB() const override;
 
@@ -200,11 +201,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   void SetREMBData(uint32_t bitrate,
                    const std::vector<uint32_t>& ssrcs) override;
 
-  // (IJ) Extended jitter report.
-  bool IJ() const override;
-
-  void SetIJStatus(bool enable) override;
-
   // (TMMBR) Temporary Max Media Bit Rate.
   bool TMMBR() const override;
 
@@ -242,6 +238,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
       RtcpStatisticsCallback* callback) override;
   RtcpStatisticsCallback* GetRtcpStatisticsCallback() override;
 
+  bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override;
   // (APP) Application specific data.
   int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
                                          uint32_t name,
@@ -271,7 +268,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   int32_t SetSendREDPayloadType(int8_t payload_type) override;
 
   // Get payload type for Redundant Audio Data RFC 2198.
-  int32_t SendREDPayloadType(int8_t& payload_type) const override;
+  int32_t SendREDPayloadType(int8_t* payload_type) const override;
 
   // Store the audio level in d_bov for header-extension-for-audio-level-
   // indication.
@@ -289,13 +286,13 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   void SetTargetSendBitrate(uint32_t bitrate_bps) override;
 
-  int32_t SetGenericFECStatus(bool enable,
-                              uint8_t payload_type_red,
-                              uint8_t payload_type_fec) override;
+  void SetGenericFECStatus(bool enable,
+                           uint8_t payload_type_red,
+                           uint8_t payload_type_fec) override;
 
-  int32_t GenericFECStatus(bool& enable,
-                           uint8_t& payload_type_red,
-                           uint8_t& payload_type_fec) override;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payload_type_red,
+                        uint8_t* payload_type_fec) override;
 
   int32_t SetFecParameters(const FecProtectionParams* delta_params,
                            const FecProtectionParams* key_params) override;
@@ -306,7 +303,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
 
-  virtual int32_t BoundingSet(bool& tmmbr_owner, TMMBRSet*& bounding_set_rec);
+  int32_t BoundingSet(bool* tmmbr_owner, TMMBRSet* bounding_set_rec);
 
   void BitrateSent(uint32_t* total_rate,
                    uint32_t* video_rate,
@@ -351,9 +348,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Get remote SequenceNumber.
   uint16_t RemoteSequenceNumber() const;
 
-  // Only for internal testing.
-  uint32_t LastSendReport(int64_t& last_rtcptime);
-
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;
@@ -372,7 +366,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   bool TimeToSendFullNackList(int64_t now) const;
 
-  int32_t id_;
   const bool audio_;
   bool collision_detected_;
   int64_t last_process_time_;
@@ -395,6 +388,9 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   RtcpRttStats* rtt_stats_;
 
+  PacketLossStats send_loss_stats_;
+  PacketLossStats receive_loss_stats_;
+
   // The processed RTT from RtcpRttStats.
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
   int64_t rtt_ms_;
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 3eb05a9b84..8329f603f9 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -8,16 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <map>
+#include <set>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/pacing/include/mock/mock_paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 
 using ::testing::_;
@@ -62,15 +65,17 @@ class SendTransport : public Transport,
     clock_ = clock;
     delay_ms_ = delay_ms;
   }
-  int SendPacket(int /*ch*/, const void* data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
     RTPHeader header;
     rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header));
     ++rtp_packets_sent_;
     last_rtp_header_ = header;
-    return static_cast<int>(len);
+    return true;
   }
-  int SendRTCPPacket(int /*ch*/, const void* data, size_t len) override {
+  bool SendRtcp(const uint8_t* data, size_t len) override {
     test::RtcpPacketParser parser;
     parser.Parse(static_cast<const uint8_t*>(data), len);
     last_nack_list_ = parser.nack_item()->last_nack_list();
@@ -81,7 +86,7 @@ class SendTransport : public Transport,
     EXPECT_TRUE(receiver_ != NULL);
     EXPECT_EQ(0, receiver_->IncomingRtcpPacket(
         static_cast<const uint8_t*>(data), len));
-    return static_cast<int>(len);
+    return true;
   }
   ModuleRtpRtcpImpl* receiver_;
   SimulatedClock* clock_;
@@ -93,7 +98,7 @@ class SendTransport : public Transport,
 
 class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
  public:
-  RtpRtcpModule(SimulatedClock* clock)
+  explicit RtpRtcpModule(SimulatedClock* clock)
       : receive_statistics_(ReceiveStatistics::Create(clock)) {
     RtpRtcp::Configuration config;
     config.audio = false;
@@ -104,7 +109,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
     config.rtt_stats = &rtt_stats_;
 
     impl_.reset(new ModuleRtpRtcpImpl(config));
-    impl_->SetRTCPStatus(kRtcpCompound);
+    impl_->SetRTCPStatus(RtcpMode::kCompound);
 
     transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, clock);
   }
@@ -218,9 +223,9 @@ class RtpRtcpImplTest : public ::testing::Test {
     nack.From(sender ? kReceiverSsrc : kSenderSsrc);
     nack.To(sender ? kSenderSsrc : kReceiverSsrc);
     nack.WithList(list, kListLength);
-    rtcp::RawPacket packet = nack.Build();
-    EXPECT_EQ(0, module->impl_->IncomingRtcpPacket(packet.buffer(),
-                                                   packet.buffer_length()));
+    rtc::scoped_ptr<rtcp::RawPacket> packet(nack.Build());
+    EXPECT_EQ(0, module->impl_->IncomingRtcpPacket(packet->Buffer(),
+                                                   packet->Length()));
   }
 };
 
@@ -345,6 +350,27 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
   EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
 }
 
+TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {
+  // Ignore fake transport delays in this test.
+  sender_.transport_.SimulateNetworkDelay(0, &clock_);
+  receiver_.transport_.SimulateNetworkDelay(0, &clock_);
+
+  sender_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+
+  // Verify no SR is sent before media has been sent, RR should still be sent
+  // from the receiving module though.
+  clock_.AdvanceTimeMilliseconds(2000);
+  int64_t current_time = clock_.TimeInMilliseconds();
+  sender_.impl_->Process();
+  receiver_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+  EXPECT_EQ(receiver_.RtcpSent().first_packet_time_ms, current_time);
+
+  SendFrame(&sender_, kBaseLayerTid);
+  EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, current_time);
+}
+
 TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) {
   EXPECT_EQ(-1, receiver_.RtcpSent().first_packet_time_ms);
   EXPECT_EQ(-1, sender_.RtcpReceived().first_packet_time_ms);
@@ -373,7 +399,10 @@ TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_FirAndPli) {
   EXPECT_EQ(1U, sender_.RtcpReceived().fir_packets);
 
   // Receive module sends a FIR and PLI.
-  EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpFir | kRtcpPli));
+  std::set<RTCPPacketType> packet_types;
+  packet_types.insert(kRtcpFir);
+  packet_types.insert(kRtcpPli);
+  EXPECT_EQ(0, receiver_.impl_->SendCompoundRTCP(packet_types));
   EXPECT_EQ(2U, receiver_.RtcpSent().fir_packets);
   EXPECT_EQ(1U, receiver_.RtcpSent().pli_packets);
   // Send module receives the FIR and PLI.
@@ -518,5 +547,4 @@ TEST_F(RtpRtcpImplTest, UniqueNackRequests) {
   EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
   EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
 }
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index e85012ae16..3072a512ab 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -11,29 +11,36 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <stdlib.h>  // srand
+#include <algorithm>
+#include <utility>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
-const size_t kMaxPaddingLength = 224;
-const int kSendSideDelayWindowMs = 1000;
+static const size_t kMaxPaddingLength = 224;
+static const int kSendSideDelayWindowMs = 1000;
+static const uint32_t kAbsSendTimeFraction = 18;
 
 namespace {
 
 const size_t kRtpHeaderLength = 12;
+const uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
-    case kFrameEmpty: return "empty";
+    case kEmptyFrame:
+      return "empty";
     case kAudioFrameSpeech: return "audio_speech";
     case kAudioFrameCN: return "audio_cn";
     case kVideoFrameKey: return "video_key";
@@ -42,6 +49,16 @@ const char* FrameTypeToString(FrameType frame_type) {
   return "";
 }
 
+// TODO(holmer): Merge this with the implementation in
+// remote_bitrate_estimator_abs_send_time.cc.
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(
+          ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+          1000) &
+      0x00FFFFFF;
+  return time_24_bits;
+}
 }  // namespace
 
 class BitrateAggregator {
@@ -95,28 +112,31 @@ class BitrateAggregator {
   uint32_t ssrc_;
 };
 
-RTPSender::RTPSender(int32_t id,
-                     bool audio,
-                     Clock* clock,
-                     Transport* transport,
-                     RtpAudioFeedback* audio_feedback,
-                     PacedSender* paced_sender,
-                     BitrateStatisticsObserver* bitrate_callback,
-                     FrameCountObserver* frame_count_observer,
-                     SendSideDelayObserver* send_side_delay_observer)
+RTPSender::RTPSender(
+    bool audio,
+    Clock* clock,
+    Transport* transport,
+    RtpAudioFeedback* audio_feedback,
+    RtpPacketSender* paced_sender,
+    TransportSequenceNumberAllocator* sequence_number_allocator,
+    TransportFeedbackObserver* transport_feedback_observer,
+    BitrateStatisticsObserver* bitrate_callback,
+    FrameCountObserver* frame_count_observer,
+    SendSideDelayObserver* send_side_delay_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
+      random_(clock_->TimeInMicroseconds()),
       bitrates_(new BitrateAggregator(bitrate_callback)),
       total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
-      id_(id),
       audio_configured_(audio),
-      audio_(audio ? new RTPSenderAudio(id, clock, this, audio_feedback)
-                   : nullptr),
+      audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
       paced_sender_(paced_sender),
+      transport_sequence_number_allocator_(sequence_number_allocator),
+      transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       transport_(transport),
@@ -156,7 +176,7 @@ RTPSender::RTPSender(int32_t id,
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
-      payload_type_rtx_(-1),
+      rtx_payload_type_(-1),
       target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       target_bitrate_(0) {
   memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
@@ -167,8 +187,8 @@ RTPSender::RTPSender(int32_t id,
   ssrc_rtx_ = ssrc_db_.CreateSSRC();  // Can't be 0.
   bitrates_->set_ssrc(ssrc_);
   // Random start, 16 bits. Can't be 0.
-  sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
-  sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
+  sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+  sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
 }
 
 RTPSender::~RTPSender() {
@@ -218,23 +238,6 @@ uint32_t RTPSender::NackOverheadRate() const {
   return nack_bitrate_.BitrateLast();
 }
 
-bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms,
-                                 int* max_send_delay_ms) const {
-  CriticalSectionScoped lock(statistics_crit_.get());
-  SendDelayMap::const_iterator it = send_delays_.upper_bound(
-      clock_->TimeInMilliseconds() - kSendSideDelayWindowMs);
-  if (it == send_delays_.end())
-    return false;
-  int num_delays = 0;
-  for (; it != send_delays_.end(); ++it) {
-    *max_send_delay_ms = std::max(*max_send_delay_ms, it->second);
-    *avg_send_delay_ms += it->second;
-    ++num_delays;
-  }
-  *avg_send_delay_ms = (*avg_send_delay_ms + num_delays / 2) / num_delays;
-  return true;
-}
-
 int32_t RTPSender::SetTransmissionTimeOffset(int32_t transmission_time_offset) {
   if (transmission_time_offset > (0x800000 - 1) ||
       transmission_time_offset < -(0x800000 - 1)) {  // Word24.
@@ -306,7 +309,7 @@ int32_t RTPSender::RegisterPayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     int8_t payload_number,
     uint32_t frequency,
-    uint8_t channels,
+    size_t channels,
     uint32_t rate) {
   assert(payload_name);
   CriticalSectionScoped cs(send_critsect_.get());
@@ -336,14 +339,14 @@ int32_t RTPSender::RegisterPayload(
     }
     return -1;
   }
-  int32_t ret_val = -1;
-  RtpUtility::Payload* payload = NULL;
+  int32_t ret_val = 0;
+  RtpUtility::Payload* payload = nullptr;
   if (audio_configured_) {
+    // TODO(mflodman): Change to CreateAudioPayload and make static.
     ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
-                                           frequency, channels, rate, payload);
+                                           frequency, channels, rate, &payload);
   } else {
-    ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate,
-                                           payload);
+    payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
   }
   if (payload) {
     payload_type_map_[payload_number] = payload;
@@ -383,10 +386,8 @@ int RTPSender::SendPayloadFrequency() const {
 int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
                                        uint16_t packet_over_head) {
   // Sanity check.
-  if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) {
-    LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length;
-    return -1;
-  }
+  RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
+      << "Invalid max payload length: " << max_payload_length;
   CriticalSectionScoped cs(send_critsect_.get());
   max_payload_length_ = max_payload_length;
   packet_over_head_ = packet_over_head;
@@ -434,9 +435,28 @@ uint32_t RTPSender::RtxSsrc() const {
   return ssrc_rtx_;
 }
 
-void RTPSender::SetRtxPayloadType(int payload_type) {
+void RTPSender::SetRtxPayloadType(int payload_type,
+                                  int associated_payload_type) {
   CriticalSectionScoped cs(send_critsect_.get());
-  payload_type_rtx_ = payload_type;
+  RTC_DCHECK_LE(payload_type, 127);
+  RTC_DCHECK_LE(associated_payload_type, 127);
+  if (payload_type < 0) {
+    LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
+    return;
+  }
+
+  rtx_payload_type_map_[associated_payload_type] = payload_type;
+  rtx_payload_type_ = payload_type;
+}
+
+std::pair<int, int> RTPSender::RtxPayloadType() const {
+  CriticalSectionScoped cs(send_critsect_.get());
+  for (const auto& kv : rtx_payload_type_map_) {
+    if (kv.second == rtx_payload_type_) {
+      return std::make_pair(rtx_payload_type_, kv.first);
+    }
+  }
+  return std::make_pair(-1, -1);
 }
 
 int32_t RTPSender::CheckPayloadType(int8_t payload_type,
@@ -449,7 +469,7 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   }
   if (audio_configured_) {
     int8_t red_pl_type = -1;
-    if (audio_->RED(red_pl_type) == 0) {
+    if (audio_->RED(&red_pl_type) == 0) {
       // We have configured RED.
       if (red_pl_type == payload_type) {
         // And it's a match...
@@ -466,7 +486,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
-    LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+    LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+                    << " not registered.";
     return -1;
   }
   SetSendPayloadType(payload_type);
@@ -497,7 +518,6 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
                                     const uint8_t* payload_data,
                                     size_t payload_size,
                                     const RTPFragmentationHeader* fragmentation,
-                                    VideoCodecInformation* codec_info,
                                     const RTPVideoHeader* rtp_hdr) {
   uint32_t ssrc;
   {
@@ -510,16 +530,17 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
-    LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+    LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+                  << static_cast<int>(payload_type) << ".";
     return -1;
   }
 
-  uint32_t ret_val;
+  int32_t ret_val;
   if (audio_configured_) {
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
-           frame_type == kFrameEmpty);
+           frame_type == kEmptyFrame);
 
     ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
                                 payload_data, payload_size, fragmentation);
@@ -528,13 +549,13 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
 
-    if (frame_type == kFrameEmpty)
+    if (frame_type == kEmptyFrame)
       return 0;
 
     ret_val =
         video_->SendVideo(video_type, frame_type, payload_type,
                           capture_timestamp, capture_time_ms, payload_data,
-                          payload_size, fragmentation, codec_info, rtp_hdr);
+                          payload_size, fragmentation, rtp_hdr);
   }
 
   CriticalSectionScoped cs(statistics_crit_.get());
@@ -571,54 +592,44 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
       break;
     RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
     bytes_left -= static_cast<int>(length - rtp_header.headerLength);
   }
   return bytes_to_send - bytes_left;
 }
 
-size_t RTPSender::BuildPaddingPacket(uint8_t* packet, size_t header_length) {
-  size_t padding_bytes_in_packet = kMaxPaddingLength;
+void RTPSender::BuildPaddingPacket(uint8_t* packet,
+                                   size_t header_length,
+                                   size_t padding_length) {
   packet[0] |= 0x20;  // Set padding bit.
-  int32_t *data =
-      reinterpret_cast<int32_t *>(&(packet[header_length]));
+  int32_t* data = reinterpret_cast<int32_t*>(&(packet[header_length]));
 
   // Fill data buffer with random data.
-  for (size_t j = 0; j < (padding_bytes_in_packet >> 2); ++j) {
+  for (size_t j = 0; j < (padding_length >> 2); ++j) {
     data[j] = rand();  // NOLINT
   }
   // Set number of padding bytes in the last byte of the packet.
-  packet[header_length + padding_bytes_in_packet - 1] =
-      static_cast<uint8_t>(padding_bytes_in_packet);
-  return padding_bytes_in_packet;
+  packet[header_length + padding_length - 1] =
+      static_cast<uint8_t>(padding_length);
 }
 
-size_t RTPSender::TrySendPadData(size_t bytes) {
-  int64_t capture_time_ms;
-  uint32_t timestamp;
-  {
-    CriticalSectionScoped cs(send_critsect_.get());
-    timestamp = timestamp_;
-    capture_time_ms = capture_time_ms_;
-    if (last_timestamp_time_ms_ > 0) {
-      timestamp +=
-          (clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
-      capture_time_ms +=
-          (clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
-    }
-  }
-  return SendPadData(timestamp, capture_time_ms, bytes);
-}
-
-size_t RTPSender::SendPadData(uint32_t timestamp,
-                              int64_t capture_time_ms,
-                              size_t bytes) {
-  size_t padding_bytes_in_packet = 0;
+size_t RTPSender::SendPadData(size_t bytes,
+                              bool timestamp_provided,
+                              uint32_t timestamp,
+                              int64_t capture_time_ms) {
+  // Always send full padding packets. This is accounted for by the
+  // RtpPacketSender,
+  // which will make sure we don't send too much padding even if a single packet
+  // is larger than requested.
+  size_t padding_bytes_in_packet =
+      std::min(MaxDataPayloadLength(), kMaxPaddingLength);
   size_t bytes_sent = 0;
+  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
+                                 kRtpExtensionTransportSequenceNumber) &&
+                             transport_sequence_number_allocator_;
   for (; bytes > 0; bytes -= padding_bytes_in_packet) {
-    // Always send full padding packets.
-    if (bytes < kMaxPaddingLength)
-      bytes = kMaxPaddingLength;
+    if (bytes < padding_bytes_in_packet)
+      bytes = padding_bytes_in_packet;
 
     uint32_t ssrc;
     uint16_t sequence_number;
@@ -626,8 +637,10 @@ size_t RTPSender::SendPadData(uint32_t timestamp,
     bool over_rtx;
     {
       CriticalSectionScoped cs(send_critsect_.get());
-      // Only send padding packets following the last packet of a frame,
-      // indicated by the marker bit.
+      if (!timestamp_provided) {
+        timestamp = timestamp_;
+        capture_time_ms = capture_time_ms_;
+      }
       if (rtx_ == kRtxOff) {
         // Without RTX we can't send padding in the middle of frames.
         if (!last_packet_marker_bit_)
@@ -643,11 +656,19 @@ size_t RTPSender::SendPadData(uint32_t timestamp,
         if (!media_has_been_sent_ && !rtp_header_extension_map_.IsRegistered(
             kRtpExtensionAbsoluteSendTime))
           return 0;
+        // Only change change the timestamp of padding packets sent over RTX.
+        // Padding only packets over RTP has to be sent as part of a media
+        // frame (and therefore the same timestamp).
+        if (last_timestamp_time_ms_ > 0) {
+          timestamp +=
+              (clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
+          capture_time_ms +=
+              (clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
+        }
         ssrc = ssrc_rtx_;
         sequence_number = sequence_number_rtx_;
         ++sequence_number_rtx_;
-        payload_type = ((rtx_ & kRtxRedundantPayloads) > 0) ? payload_type_rtx_
-                                                            : payload_type_;
+        payload_type = rtx_payload_type_;
         over_rtx = true;
       }
     }
@@ -656,15 +677,13 @@ size_t RTPSender::SendPadData(uint32_t timestamp,
     size_t header_length =
         CreateRtpHeader(padding_packet, payload_type, ssrc, false, timestamp,
                         sequence_number, std::vector<uint32_t>());
-    assert(header_length != static_cast<size_t>(-1));
-    padding_bytes_in_packet = BuildPaddingPacket(padding_packet, header_length);
-    assert(padding_bytes_in_packet <= bytes);
+    BuildPaddingPacket(padding_packet, header_length, padding_bytes_in_packet);
     size_t length = padding_bytes_in_packet + header_length;
     int64_t now_ms = clock_->TimeInMilliseconds();
 
     RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
 
     if (capture_time_ms > 0) {
       UpdateTransmissionTimeOffset(
@@ -672,8 +691,20 @@ size_t RTPSender::SendPadData(uint32_t timestamp,
     }
 
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
-    if (!SendPacketToNetwork(padding_packet, length))
+
+    PacketOptions options;
+    if (using_transport_seq) {
+      options.packet_id =
+          UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
+    }
+
+    if (using_transport_seq && transport_feedback_observer_) {
+      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+    }
+
+    if (!SendPacketToNetwork(padding_packet, length, options))
       break;
+
     bytes_sent += padding_bytes_in_packet;
     UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
   }
@@ -693,6 +724,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
   size_t length = IP_PACKET_SIZE;
   uint8_t data_buffer[IP_PACKET_SIZE];
   int64_t capture_time_ms;
+
   if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
                                                data_buffer, &length,
                                                &capture_time_ms)) {
@@ -703,35 +735,39 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
-    if (!rtp_parser.Parse(header)) {
+    if (!rtp_parser.Parse(&header)) {
       assert(false);
       return -1;
     }
     // Convert from TickTime to Clock since capture_time_ms is based on
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
-    if (!paced_sender_->SendPacket(
-            PacedSender::kHighPriority, header.ssrc, header.sequenceNumber,
-            corrected_capture_tims_ms, length - header.headerLength, true)) {
-      // We can't send the packet right now.
-      // We will be called when it is time.
-      return length;
-    }
+    paced_sender_->InsertPacket(
+        RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
+        corrected_capture_tims_ms, length - header.headerLength, true);
+
+    return length;
   }
   int rtx = kRtxOff;
   {
     CriticalSectionScoped lock(send_critsect_.get());
     rtx = rtx_;
   }
-  return PrepareAndSendPacket(data_buffer, length, capture_time_ms,
-                              (rtx & kRtxRetransmitted) > 0, true) ?
-      static_cast<int32_t>(length) : -1;
+  if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
+                            (rtx & kRtxRetransmitted) > 0, true)) {
+    return -1;
+  }
+  return static_cast<int32_t>(length);
 }
 
-bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
+bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
+                                    size_t size,
+                                    const PacketOptions& options) {
   int bytes_sent = -1;
   if (transport_) {
-    bytes_sent = transport_->SendPacket(id_, packet, size);
+    bytes_sent = transport_->SendRtp(packet, size, options)
+                     ? static_cast<int>(size)
+                     : -1;
   }
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTPSender::SendPacketToNetwork", "size", size, "sent",
@@ -753,7 +789,8 @@ int RTPSender::SelectiveRetransmissions() const {
 int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
   if (!video_)
     return -1;
-  return video_->SetSelectiveRetransmissions(settings);
+  video_->SetSelectiveRetransmissions(settings);
+  return 0;
 }
 
 void RTPSender::OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
@@ -884,11 +921,11 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                      int64_t capture_time_ms,
                                      bool send_over_rtx,
                                      bool is_retransmit) {
-  uint8_t *buffer_to_send_ptr = buffer;
+  uint8_t* buffer_to_send_ptr = buffer;
 
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   if (!is_retransmit && rtp_header.markerBit) {
     TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
                            capture_time_ms);
@@ -909,7 +946,23 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
   UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
                                diff_ms);
   UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
-  bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
+
+  // TODO(sprang): Potentially too much overhead in IsRegistered()?
+  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
+                                 kRtpExtensionTransportSequenceNumber) &&
+                             transport_sequence_number_allocator_;
+
+  PacketOptions options;
+  if (using_transport_seq) {
+    options.packet_id =
+        UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
+  }
+
+  if (using_transport_seq && transport_feedback_observer_) {
+    transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+  }
+
+  bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
     CriticalSectionScoped lock(send_critsect_.get());
     media_has_been_sent_ = true;
@@ -961,34 +1014,37 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
   bool fec_enabled;
   uint8_t pt_red;
   uint8_t pt_fec;
-  video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
+  video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
+  if (audio_configured_ || bytes == 0)
+    return 0;
   {
     CriticalSectionScoped cs(send_critsect_.get());
-    if (!sending_media_) return 0;
+    if (!sending_media_)
+      return 0;
   }
-  if (bytes == 0)
-    return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes);
   if (bytes_sent < bytes)
-    bytes_sent += TrySendPadData(bytes - bytes_sent);
+    bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);
   return bytes_sent;
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
-int32_t RTPSender::SendToNetwork(
-    uint8_t *buffer, size_t payload_length, size_t rtp_header_length,
-    int64_t capture_time_ms, StorageType storage,
-    PacedSender::Priority priority) {
+int32_t RTPSender::SendToNetwork(uint8_t* buffer,
+                                 size_t payload_length,
+                                 size_t rtp_header_length,
+                                 int64_t capture_time_ms,
+                                 StorageType storage,
+                                 RtpPacketSender::Priority priority) {
   RtpUtility::RtpHeaderParser rtp_parser(buffer,
                                          payload_length + rtp_header_length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   int64_t now_ms = clock_->TimeInMilliseconds();
 
@@ -1005,43 +1061,38 @@ int32_t RTPSender::SendToNetwork(
 
   // Used for NACK and to spread out the transmission of packets.
   if (packet_history_.PutRTPPacket(buffer, rtp_header_length + payload_length,
-                                   max_payload_length_, capture_time_ms,
-                                   storage) != 0) {
+                                   capture_time_ms, storage) != 0) {
     return -1;
   }
 
-  if (paced_sender_ && storage != kDontStore) {
+  if (paced_sender_) {
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
-    if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
-                                   rtp_header.sequenceNumber, corrected_time_ms,
-                                   payload_length, false)) {
-      if (last_capture_time_ms_sent_ == 0 ||
-          corrected_time_ms > last_capture_time_ms_sent_) {
-        last_capture_time_ms_sent_ = corrected_time_ms;
-        TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                                 "PacedSend", corrected_time_ms,
-                                 "capture_time_ms", corrected_time_ms);
-      }
-      // We can't send the packet right now.
-      // We will be called when it is time.
-      return 0;
+    paced_sender_->InsertPacket(priority, rtp_header.ssrc,
+                                rtp_header.sequenceNumber, corrected_time_ms,
+                                payload_length, false);
+    if (last_capture_time_ms_sent_ == 0 ||
+        corrected_time_ms > last_capture_time_ms_sent_) {
+      last_capture_time_ms_sent_ = corrected_time_ms;
+      TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                               "PacedSend", corrected_time_ms,
+                               "capture_time_ms", corrected_time_ms);
     }
+    return 0;
   }
   if (capture_time_ms > 0) {
     UpdateDelayStatistics(capture_time_ms, now_ms);
   }
 
   size_t length = payload_length + rtp_header_length;
-  bool sent = SendPacketToNetwork(buffer, length);
+  bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
+
+  // Mark the packet as sent in the history even if send failed. Dropping a
+  // packet here should be treated as any other packet drop so we should be
+  // ready for a retransmission.
+  packet_history_.SetSent(rtp_header.sequenceNumber);
 
-  if (storage != kDontStore) {
-    // Mark the packet as sent in the history even if send failed. Dropping a
-    // packet here should be treated as any other packet drop so we should be
-    // ready for a retransmission.
-    packet_history_.SetSent(rtp_header.sequenceNumber);
-  }
   if (!sent)
     return -1;
 
@@ -1054,6 +1105,9 @@ int32_t RTPSender::SendToNetwork(
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
+  if (!send_side_delay_observer_)
+    return;
+
   uint32_t ssrc;
   int avg_delay_ms = 0;
   int max_delay_ms = 0;
@@ -1068,12 +1122,19 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
     send_delays_.erase(send_delays_.begin(),
                        send_delays_.lower_bound(now_ms -
                        kSendSideDelayWindowMs));
+    int num_delays = 0;
+    for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
+         it != send_delays_.end(); ++it) {
+      max_delay_ms = std::max(max_delay_ms, it->second);
+      avg_delay_ms += it->second;
+      ++num_delays;
+    }
+    if (num_delays == 0)
+      return;
+    avg_delay_ms = (avg_delay_ms + num_delays / 2) / num_delays;
   }
-  if (send_side_delay_observer_ &&
-      GetSendSideDelay(&avg_delay_ms, &max_delay_ms)) {
-    send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms,
-        max_delay_ms, ssrc);
-  }
+  send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
+                                                  ssrc);
 }
 
 void RTPSender::ProcessBitrate() {
@@ -1094,29 +1155,11 @@ size_t RTPSender::RTPHeaderLength() const {
   return rtp_header_length;
 }
 
-uint16_t RTPSender::IncrementSequenceNumber() {
+uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
   CriticalSectionScoped cs(send_critsect_.get());
-  return sequence_number_++;
-}
-
-void RTPSender::ResetDataCounters() {
-  uint32_t ssrc;
-  uint32_t ssrc_rtx;
-  bool report_rtx;
-  {
-    CriticalSectionScoped ssrc_lock(send_critsect_.get());
-    ssrc = ssrc_;
-    ssrc_rtx = ssrc_rtx_;
-    report_rtx = rtx_ != kRtxOff;
-  }
-  CriticalSectionScoped lock(statistics_crit_.get());
-  rtp_stats_ = StreamDataCounters();
-  rtx_rtp_stats_ = StreamDataCounters();
-  if (rtp_stats_callback_) {
-    rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc);
-    if (report_rtx)
-      rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx);
-  }
+  uint16_t first_allocated_sequence_number = sequence_number_;
+  sequence_number_ += packets_to_send;
+  return first_allocated_sequence_number;
 }
 
 void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
@@ -1144,7 +1187,7 @@ size_t RTPSender::CreateRtpHeader(uint8_t* header,
   int32_t rtp_header_length = kRtpHeaderLength;
 
   if (csrcs.size() > 0) {
-    uint8_t *ptr = &header[rtp_header_length];
+    uint8_t* ptr = &header[rtp_header_length];
     for (size_t i = 0; i < csrcs.size(); ++i) {
       ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
       ptr += 4;
@@ -1232,7 +1275,8 @@ uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer,
         block_length = BuildVideoRotationExtension(extension_data);
         break;
       case kRtpExtensionTransportSequenceNumber:
-        block_length = BuildTransportSequenceNumberExtension(extension_data);
+        block_length = BuildTransportSequenceNumberExtension(
+            extension_data, transport_sequence_number_);
         break;
       case kRtpExtensionRtpStreamId:
         block_length = BuildRIDExtension(extension_data);
@@ -1388,7 +1432,8 @@ uint8_t RTPSender::BuildVideoRotationExtension(uint8_t* data_buffer) const {
 }
 
 uint8_t RTPSender::BuildTransportSequenceNumberExtension(
-    uint8_t* data_buffer) const {
+    uint8_t* data_buffer,
+    uint16_t sequence_number) const {
   //   0                   1                   2
   //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
   //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
@@ -1405,8 +1450,7 @@ uint8_t RTPSender::BuildTransportSequenceNumberExtension(
   size_t pos = 0;
   const uint8_t len = 1;
   data_buffer[pos++] = (id << 4) + len;
-  ByteWriter<uint16_t>::WriteBigEndian(data_buffer + pos,
-                                       transport_sequence_number_);
+  ByteWriter<uint16_t>::WriteBigEndian(data_buffer + pos, sequence_number);
   pos += 2;
   assert(pos == kTransportSequenceNumberLength);
   return kTransportSequenceNumberLength;
@@ -1423,15 +1467,20 @@ uint8_t RTPSender::BuildRIDExtension(
   // Get id defined by user.
   uint8_t id;
   if (!rid_ ||
-    rtp_header_extension_map_.GetId(kRtpExtensionRtpStreamId,
+      rtp_header_extension_map_.GetId(kRtpExtensionRtpStreamId,
                                       &id) != 0) {
-    // No RtpStreamId or not registered
+    // Not registered or not set
     return 0;
   }
   size_t pos = 0;
   // RID value is not null-terminated in header, so no +1
   const uint8_t len = strlen(rid_);
-  data_buffer[pos++] = (id << 4) + len;
+  if (len > 16 || len == 0) {
+    LOG(LS_ERROR) << "Failed to add RID header because of unsupported RID"
+                    " length: " << len;
+    return 0;
+  }
+  data_buffer[pos++] = (id << 4) + (len - 1);
   memcpy(data_buffer + pos, rid_, len);
   pos += len;
   return pos;
@@ -1474,35 +1523,62 @@ bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
   return true;
 }
 
+RTPSender::ExtensionStatus RTPSender::VerifyExtension(
+    RTPExtensionType extension_type,
+    uint8_t* rtp_packet,
+    size_t rtp_packet_length,
+    const RTPHeader& rtp_header,
+    size_t extension_length_bytes,
+    size_t* extension_offset) const {
+  // Get id.
+  uint8_t id = 0;
+  if (rtp_header_extension_map_.GetId(extension_type, &id) != 0)
+    return ExtensionStatus::kNotRegistered;
+
+  size_t block_pos = 0;
+  if (!FindHeaderExtensionPosition(extension_type, rtp_packet,
+                                   rtp_packet_length, rtp_header, &block_pos))
+    return ExtensionStatus::kError;
+
+  // Verify that header contains extension.
+  if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
+        (rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
+    LOG(LS_WARNING)
+        << "Failed to update absolute send time, hdr extension not found.";
+    return ExtensionStatus::kError;
+  }
+
+  // Verify first byte in block.
+  const uint8_t first_block_byte = (id << 4) + (extension_length_bytes - 2);
+  if (rtp_packet[block_pos] != first_block_byte)
+    return ExtensionStatus::kError;
+
+  *extension_offset = block_pos;
+  return ExtensionStatus::kOk;
+}
+
 void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
                                              size_t rtp_packet_length,
                                              const RTPHeader& rtp_header,
                                              int64_t time_diff_ms) const {
+  size_t offset;
   CriticalSectionScoped cs(send_critsect_.get());
-  // Get id.
-  uint8_t id = 0;
-  if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
-                                      &id) != 0) {
-    // Not registered.
-    return;
+  switch (VerifyExtension(kRtpExtensionTransmissionTimeOffset, rtp_packet,
+                          rtp_packet_length, rtp_header,
+                          kTransmissionTimeOffsetLength, &offset)) {
+    case ExtensionStatus::kNotRegistered:
+      return;
+    case ExtensionStatus::kError:
+      LOG(LS_WARNING) << "Failed to update transmission time offset.";
+      return;
+    case ExtensionStatus::kOk:
+      break;
+    default:
+      RTC_NOTREACHED();
   }
 
-  size_t block_pos = 0;
-  if (!FindHeaderExtensionPosition(kRtpExtensionTransmissionTimeOffset,
-                                   rtp_packet, rtp_packet_length, rtp_header,
-                                   &block_pos)) {
-    LOG(LS_WARNING) << "Failed to update transmission time offset.";
-    return;
-  }
-
-  // Verify first byte in block.
-  const uint8_t first_block_byte = (id << 4) + 2;
-  if (rtp_packet[block_pos] != first_block_byte) {
-    LOG(LS_WARNING) << "Failed to update transmission time offset.";
-    return;
-  }
   // Update transmission offset field (converting to a 90 kHz timestamp).
-  ByteWriter<int32_t, 3>::WriteBigEndian(rtp_packet + block_pos + 1,
+  ByteWriter<int32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
                                          time_diff_ms * 90);  // RTP timestamp.
 }
 
@@ -1511,29 +1587,24 @@ bool RTPSender::UpdateAudioLevel(uint8_t* rtp_packet,
                                  const RTPHeader& rtp_header,
                                  bool is_voiced,
                                  uint8_t dBov) const {
+  size_t offset;
   CriticalSectionScoped cs(send_critsect_.get());
 
-  // Get id.
-  uint8_t id = 0;
-  if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
-    // Not registered.
-    return false;
+  switch (VerifyExtension(kRtpExtensionAudioLevel, rtp_packet,
+                          rtp_packet_length, rtp_header, kAudioLevelLength,
+                          &offset)) {
+    case ExtensionStatus::kNotRegistered:
+      return false;
+    case ExtensionStatus::kError:
+      LOG(LS_WARNING) << "Failed to update audio level.";
+      return false;
+    case ExtensionStatus::kOk:
+      break;
+    default:
+      RTC_NOTREACHED();
   }
 
-  size_t block_pos = 0;
-  if (!FindHeaderExtensionPosition(kRtpExtensionAudioLevel, rtp_packet,
-                                   rtp_packet_length, rtp_header, &block_pos)) {
-    LOG(LS_WARNING) << "Failed to update audio level.";
-    return false;
-  }
-
-  // Verify first byte in block.
-  const uint8_t first_block_byte = (id << 4) + 0;
-  if (rtp_packet[block_pos] != first_block_byte) {
-    LOG(LS_WARNING) << "Failed to update audio level.";
-    return false;
-  }
-  rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
+  rtp_packet[offset + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
   return true;
 }
 
@@ -1541,37 +1612,24 @@ bool RTPSender::UpdateVideoRotation(uint8_t* rtp_packet,
                                     size_t rtp_packet_length,
                                     const RTPHeader& rtp_header,
                                     VideoRotation rotation) const {
+  size_t offset;
   CriticalSectionScoped cs(send_critsect_.get());
 
-  // Get id.
-  uint8_t id = 0;
-  if (rtp_header_extension_map_.GetId(kRtpExtensionVideoRotation, &id) != 0) {
-    // Not registered.
-    return false;
+  switch (VerifyExtension(kRtpExtensionVideoRotation, rtp_packet,
+                          rtp_packet_length, rtp_header, kVideoRotationLength,
+                          &offset)) {
+    case ExtensionStatus::kNotRegistered:
+      return false;
+    case ExtensionStatus::kError:
+      LOG(LS_WARNING) << "Failed to update CVO.";
+      return false;
+    case ExtensionStatus::kOk:
+      break;
+    default:
+      RTC_NOTREACHED();
   }
 
-  size_t block_pos = 0;
-  if (!FindHeaderExtensionPosition(kRtpExtensionVideoRotation, rtp_packet,
-                                   rtp_packet_length, rtp_header, &block_pos)) {
-    LOG(LS_WARNING) << "Failed to update video rotation (CVO).";
-    return false;
-  }
-  // Get length until start of header extension block.
-  int extension_block_pos =
-      rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
-          kRtpExtensionVideoRotation);
-  if (extension_block_pos < 0) {
-    // The feature is not enabled.
-    return false;
-  }
-
-  // Verify first byte in block.
-  const uint8_t first_block_byte = (id << 4) + 0;
-  if (rtp_packet[block_pos] != first_block_byte) {
-    LOG(LS_WARNING) << "Failed to update CVO.";
-    return false;
-  }
-  rtp_packet[block_pos + 1] = ConvertVideoRotationToCVOByte(rotation);
+  rtp_packet[offset + 1] = ConvertVideoRotationToCVOByte(rotation);
   return true;
 }
 
@@ -1579,53 +1637,59 @@ void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                                        size_t rtp_packet_length,
                                        const RTPHeader& rtp_header,
                                        int64_t now_ms) const {
+  size_t offset;
   CriticalSectionScoped cs(send_critsect_.get());
 
-  // Get id.
-  uint8_t id = 0;
-  if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
-                                      &id) != 0) {
-    // Not registered.
-    return;
-  }
-  // Get length until start of header extension block.
-  int extension_block_pos =
-      rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
-          kRtpExtensionAbsoluteSendTime);
-  if (extension_block_pos < 0) {
-    // The feature is not enabled.
-    return;
-  }
-  size_t block_pos =
-      kRtpHeaderLength + rtp_header.numCSRCs + extension_block_pos;
-  if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength ||
-      rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) {
-    LOG(LS_WARNING) << "Failed to update absolute send time, invalid length.";
-    return;
-  }
-  // Verify that header contains extension.
-  if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
-        (rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
-    LOG(LS_WARNING)
-        << "Failed to update absolute send time, hdr extension not found.";
-    return;
-  }
-  // Verify first byte in block.
-  const uint8_t first_block_byte = (id << 4) + 2;
-  if (rtp_packet[block_pos] != first_block_byte) {
-    LOG(LS_WARNING) << "Failed to update absolute send time.";
-    return;
+  switch (VerifyExtension(kRtpExtensionAbsoluteSendTime, rtp_packet,
+                          rtp_packet_length, rtp_header,
+                          kAbsoluteSendTimeLength, &offset)) {
+    case ExtensionStatus::kNotRegistered:
+      return;
+    case ExtensionStatus::kError:
+      LOG(LS_WARNING) << "Failed to update absolute send time";
+      return;
+    case ExtensionStatus::kOk:
+      break;
+    default:
+      RTC_NOTREACHED();
   }
+
   // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
   // fractional part).
-  ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + block_pos + 1,
-                                          ((now_ms << 18) / 1000) & 0x00ffffff);
+  ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
+                                          ConvertMsTo24Bits(now_ms));
+}
+
+uint16_t RTPSender::UpdateTransportSequenceNumber(
+    uint8_t* rtp_packet,
+    size_t rtp_packet_length,
+    const RTPHeader& rtp_header) const {
+  size_t offset;
+  CriticalSectionScoped cs(send_critsect_.get());
+
+  switch (VerifyExtension(kRtpExtensionTransportSequenceNumber, rtp_packet,
+                          rtp_packet_length, rtp_header,
+                          kTransportSequenceNumberLength, &offset)) {
+    case ExtensionStatus::kNotRegistered:
+      return 0;
+    case ExtensionStatus::kError:
+      LOG(LS_WARNING) << "Failed to update transport sequence number";
+      return 0;
+    case ExtensionStatus::kOk:
+      break;
+    default:
+      RTC_NOTREACHED();
+  }
+
+  uint16_t seq = transport_sequence_number_allocator_->AllocateSequenceNumber();
+  BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
+  return seq;
 }
 
 void RTPSender::SetSendingStatus(bool enabled) {
   if (enabled) {
     uint32_t frequency_hz = SendPayloadFrequency();
-    uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
+    uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
 
     // Will be ignored if it's already configured via API.
     SetStartTimestamp(RTPtime, false);
@@ -1640,8 +1704,7 @@ void RTPSender::SetSendingStatus(bool enabled) {
     // Don't initialize seq number if SSRC passed externally.
     if (!sequence_number_forced_ && !ssrc_forced_) {
       // Generate a new sequence number.
-      sequence_number_ =
-          rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);  // NOLINT
+      sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
     }
   }
 }
@@ -1703,8 +1766,7 @@ void RTPSender::SetSSRC(uint32_t ssrc) {
   ssrc_ = ssrc;
   bitrates_->set_ssrc(ssrc_);
   if (!sequence_number_forced_) {
-    sequence_number_ =
-        rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);  // NOLINT
+    sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
 }
 
@@ -1762,15 +1824,7 @@ int32_t RTPSender::RED(int8_t *payload_type) const {
   if (!audio_configured_) {
     return -1;
   }
-  return audio_->RED(*payload_type);
-}
-
-// Video
-VideoCodecInformation *RTPSender::CodecInformationVideo() {
-  if (audio_configured_) {
-    return NULL;
-  }
-  return video_->CodecInformationVideo();
+  return audio_->RED(payload_type);
 }
 
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
@@ -1785,31 +1839,18 @@ uint32_t RTPSender::MaxConfiguredBitrateVideo() const {
   return video_->MaxConfiguredBitrateVideo();
 }
 
-int32_t RTPSender::SendRTPIntraRequest() {
-  if (audio_configured_) {
-    return -1;
-  }
-  return video_->SendRTPIntraRequest();
+void RTPSender::SetGenericFECStatus(bool enable,
+                                    uint8_t payload_type_red,
+                                    uint8_t payload_type_fec) {
+  RTC_DCHECK(!audio_configured_);
+  video_->SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
-int32_t RTPSender::SetGenericFECStatus(bool enable,
-                                       uint8_t payload_type_red,
-                                       uint8_t payload_type_fec) {
-  if (audio_configured_) {
-    return -1;
-  }
-  return video_->SetGenericFECStatus(enable, payload_type_red,
-                                     payload_type_fec);
-}
-
-int32_t RTPSender::GenericFECStatus(bool* enable,
+void RTPSender::GenericFECStatus(bool* enable,
                                     uint8_t* payload_type_red,
                                     uint8_t* payload_type_fec) const {
-  if (audio_configured_) {
-    return -1;
-  }
-  return video_->GenericFECStatus(
-      *enable, *payload_type_red, *payload_type_fec);
+  RTC_DCHECK(!audio_configured_);
+  video_->GenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 int32_t RTPSender::SetFecParameters(
@@ -1818,7 +1859,8 @@ int32_t RTPSender::SetFecParameters(
   if (audio_configured_) {
     return -1;
   }
-  return video_->SetFecParameters(delta_params, key_params);
+  video_->SetFecParameters(delta_params, key_params);
+  return 0;
 }
 
 void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
@@ -1830,20 +1872,20 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
       reinterpret_cast<const uint8_t*>(buffer), *length);
 
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   // Add original RTP header.
   memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
 
   // Replace payload type, if a specific type is set for RTX.
-  if (payload_type_rtx_ != -1) {
-    data_buffer_rtx[1] = static_cast<uint8_t>(payload_type_rtx_);
+  if (rtx_payload_type_ != -1) {
+    data_buffer_rtx[1] = static_cast<uint8_t>(rtx_payload_type_);
     if (rtp_header.markerBit)
       data_buffer_rtx[1] |= kRtpMarkerBitMask;
   }
 
   // Replace sequence number.
-  uint8_t *ptr = data_buffer_rtx + 2;
+  uint8_t* ptr = data_buffer_rtx + 2;
   ByteWriter<uint16_t>::WriteBigEndian(ptr, sequence_number_rtx_++);
 
   // Replace SSRC.
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 3d766d7781..b1898216d4 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -11,23 +11,22 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 
-#include <assert.h>
-#include <math.h>
-
+#include <list>
 #include <map>
+#include <utility>
+#include <vector>
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
-#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
-
-#define MAX_INIT_RTP_SEQ_NUMBER 32767  // 2^15 -1.
+#include "webrtc/transport.h"
 
 namespace webrtc {
 
@@ -61,17 +60,22 @@ class RTPSenderInterface {
                                  bool inc_sequence_number = true) = 0;
 
   virtual size_t RTPHeaderLength() const = 0;
-  virtual uint16_t IncrementSequenceNumber() = 0;
+  // Returns the next sequence number to use for a packet and allocates
+  // 'packets_to_send' number of sequence numbers. It's important all allocated
+  // sequence numbers are used in sequence to avoid perceived packet loss.
+  virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0;
   virtual uint16_t SequenceNumber() const = 0;
   virtual size_t MaxPayloadLength() const = 0;
   virtual size_t MaxDataPayloadLength() const = 0;
   virtual uint16_t PacketOverHead() const = 0;
   virtual uint16_t ActualSendBitrateKbit() const = 0;
 
-  virtual int32_t SendToNetwork(
-      uint8_t *data_buffer, size_t payload_length, size_t rtp_header_length,
-      int64_t capture_time_ms, StorageType storage,
-      PacedSender::Priority priority) = 0;
+  virtual int32_t SendToNetwork(uint8_t* data_buffer,
+                                size_t payload_length,
+                                size_t rtp_header_length,
+                                int64_t capture_time_ms,
+                                StorageType storage,
+                                RtpPacketSender::Priority priority) = 0;
 
   virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
@@ -83,12 +87,13 @@ class RTPSenderInterface {
 
 class RTPSender : public RTPSenderInterface {
  public:
-  RTPSender(int32_t id,
-            bool audio,
+  RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
             RtpAudioFeedback* audio_feedback,
-            PacedSender* paced_sender,
+            RtpPacketSender* paced_sender,
+            TransportSequenceNumberAllocator* sequence_number_allocator,
+            TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer);
@@ -102,10 +107,6 @@ class RTPSender : public RTPSenderInterface {
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
-  // Returns true if the statistics have been calculated, and false if no frame
-  // was sent within the statistics window.
-  bool GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const;
-
   void SetTargetBitrate(uint32_t bitrate);
   uint32_t GetTargetBitrate();
 
@@ -115,7 +116,7 @@ class RTPSender : public RTPSenderInterface {
   int32_t RegisterPayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type, const uint32_t frequency,
-      const uint8_t channels, const uint32_t rate);
+      const size_t channels, const uint32_t rate);
 
   int32_t DeRegisterSendPayload(const int8_t payload_type);
 
@@ -133,8 +134,6 @@ class RTPSender : public RTPSenderInterface {
   void GetDataCounters(StreamDataCounters* rtp_stats,
                        StreamDataCounters* rtx_stats) const;
 
-  void ResetDataCounters();
-
   uint32_t StartTimestamp() const;
   void SetStartTimestamp(uint32_t timestamp, bool force);
 
@@ -155,7 +154,6 @@ class RTPSender : public RTPSenderInterface {
                            const uint8_t* payload_data,
                            size_t payload_size,
                            const RTPFragmentationHeader* fragmentation,
-                           VideoCodecInformation* codec_info = NULL,
                            const RTPVideoHeader* rtp_hdr = NULL);
 
   // RTP header extension
@@ -166,7 +164,7 @@ class RTPSender : public RTPSenderInterface {
   int32_t SetRID(const char* rid);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
-  virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
+  bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   size_t RtpHeaderExtensionTotalLength() const;
@@ -177,19 +175,39 @@ class RTPSender : public RTPSenderInterface {
   uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
   uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
   uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const;
-  uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer) const;
+  uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer,
+                                                uint16_t sequence_number) const;
   uint8_t BuildRIDExtension(uint8_t* data_buffer) const;
 
+  // Verifies that the specified extension is registered, and that it is
+  // present in rtp packet. If extension is not registered kNotRegistered is
+  // returned. If extension cannot be found in the rtp header, or if it is
+  // malformed, kError is returned. Otherwise *extension_offset is set to the
+  // offset of the extension from the beginning of the rtp packet and kOk is
+  // returned.
+  enum class ExtensionStatus {
+    kNotRegistered,
+    kOk,
+    kError,
+  };
+  ExtensionStatus VerifyExtension(RTPExtensionType extension_type,
+                                  uint8_t* rtp_packet,
+                                  size_t rtp_packet_length,
+                                  const RTPHeader& rtp_header,
+                                  size_t extension_length_bytes,
+                                  size_t* extension_offset) const
+      EXCLUSIVE_LOCKS_REQUIRED(send_critsect_.get());
+
   bool UpdateAudioLevel(uint8_t* rtp_packet,
                         size_t rtp_packet_length,
                         const RTPHeader& rtp_header,
                         bool is_voiced,
                         uint8_t dBov) const;
 
-  virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
-                                   size_t rtp_packet_length,
-                                   const RTPHeader& rtp_header,
-                                   VideoRotation rotation) const override;
+  bool UpdateVideoRotation(uint8_t* rtp_packet,
+                           size_t rtp_packet_length,
+                           const RTPHeader& rtp_header,
+                           VideoRotation rotation) const override;
 
   bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
                         bool retransmission);
@@ -216,7 +234,8 @@ class RTPSender : public RTPSenderInterface {
   uint32_t RtxSsrc() const;
   void SetRtxSsrc(uint32_t ssrc);
 
-  void SetRtxPayloadType(int payloadType);
+  void SetRtxPayloadType(int payload_type, int associated_payload_type);
+  std::pair<int, int> RtxPayloadType() const;
 
   // Functions wrapping RTPSenderInterface.
   int32_t BuildRTPheader(uint8_t* data_buffer,
@@ -228,7 +247,7 @@ class RTPSender : public RTPSenderInterface {
                          const bool inc_sequence_number = true) override;
 
   size_t RTPHeaderLength() const override;
-  uint16_t IncrementSequenceNumber() override;
+  uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
   uint16_t PacketOverHead() const override;
 
@@ -241,7 +260,7 @@ class RTPSender : public RTPSenderInterface {
                         size_t rtp_header_length,
                         int64_t capture_time_ms,
                         StorageType storage,
-                        PacedSender::Priority priority) override;
+                        RtpPacketSender::Priority priority) override;
 
   // Audio.
 
@@ -262,29 +281,26 @@ class RTPSender : public RTPSenderInterface {
   // Get payload type for Redundant Audio Data RFC 2198.
   int32_t RED(int8_t *payload_type) const;
 
-  // Video.
-  VideoCodecInformation *CodecInformationVideo();
-
   RtpVideoCodecTypes VideoCodecType() const;
 
   uint32_t MaxConfiguredBitrateVideo() const;
 
-  int32_t SendRTPIntraRequest();
-
   // FEC.
-  int32_t SetGenericFECStatus(bool enable,
-                              uint8_t payload_type_red,
-                              uint8_t payload_type_fec);
+  void SetGenericFECStatus(bool enable,
+                           uint8_t payload_type_red,
+                           uint8_t payload_type_fec);
 
-  int32_t GenericFECStatus(bool *enable, uint8_t *payload_type_red,
-                           uint8_t *payload_type_fec) const;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payload_type_red,
+                        uint8_t* payload_type_fec) const;
 
   int32_t SetFecParameters(const FecProtectionParams *delta_params,
                            const FecProtectionParams *key_params);
 
-  size_t SendPadData(uint32_t timestamp,
-                     int64_t capture_time_ms,
-                     size_t bytes);
+  size_t SendPadData(size_t bytes,
+                     bool timestamp_provided,
+                     uint32_t timestamp,
+                     int64_t capture_time_ms);
 
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
@@ -326,14 +342,17 @@ class RTPSender : public RTPSenderInterface {
   // Return the number of bytes sent.  Note that both of these functions may
   // return a larger value that their argument.
   size_t TrySendRedundantPayloads(size_t bytes);
-  size_t TrySendPadData(size_t bytes);
 
-  size_t BuildPaddingPacket(uint8_t* packet, size_t header_length);
+  void BuildPaddingPacket(uint8_t* packet,
+                          size_t header_length,
+                          size_t padding_length);
 
   void BuildRtxPacket(uint8_t* buffer, size_t* length,
                       uint8_t* buffer_rtx);
 
-  bool SendPacketToNetwork(const uint8_t *packet, size_t size);
+  bool SendPacketToNetwork(const uint8_t* packet,
+                           size_t size,
+                           const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
 
@@ -353,6 +372,12 @@ class RTPSender : public RTPSenderInterface {
                               size_t rtp_packet_length,
                               const RTPHeader& rtp_header,
                               int64_t now_ms) const;
+  // Update the transport sequence number of the packet using a new sequence
+  // number allocated by SequenceNumberAllocator. Returns the assigned sequence
+  // number, or 0 if extension could not be updated.
+  uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet,
+                                         size_t rtp_packet_length,
+                                         const RTPHeader& rtp_header) const;
 
   void UpdateRtpStats(const uint8_t* buffer,
                       size_t packet_length,
@@ -363,17 +388,18 @@ class RTPSender : public RTPSenderInterface {
 
   Clock* clock_;
   int64_t clock_delta_ms_;
+  Random random_ GUARDED_BY(send_critsect_);
 
   rtc::scoped_ptr<BitrateAggregator> bitrates_;
   Bitrate total_bitrate_sent_;
 
-  int32_t id_;
-
   const bool audio_configured_;
   rtc::scoped_ptr<RTPSenderAudio> audio_;
   rtc::scoped_ptr<RTPSenderVideo> video_;
 
-  PacedSender *paced_sender_;
+  RtpPacketSender* const paced_sender_;
+  TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
+  TransportFeedbackObserver* const transport_feedback_observer_;
   int64_t last_capture_time_ms_sent_;
   rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_;
 
@@ -429,7 +455,11 @@ class RTPSender : public RTPSenderInterface {
   std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_);
   int rtx_ GUARDED_BY(send_critsect_);
   uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
-  int payload_type_rtx_ GUARDED_BY(send_critsect_);
+  // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that
+  // only understand one RTX PT is no longer needed.
+  int rtx_payload_type_ GUARDED_BY(send_critsect_);
+  // Mapping rtx_payload_type_map_[associated] = rtx.
+  std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   // Note: Don't access this variable directly, always go through
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index de728f0860..2aa4961cdc 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -10,47 +10,44 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
-#include <assert.h> //assert
-#include <string.h> //memcpy
+#include <string.h>
 
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
-RTPSenderAudio::RTPSenderAudio(const int32_t id,
-                               Clock* clock,
+RTPSenderAudio::RTPSenderAudio(Clock* clock,
                                RTPSender* rtpSender,
-                               RtpAudioFeedback* audio_feedback) :
-    _id(id),
-    _clock(clock),
-    _rtpSender(rtpSender),
-    _audioFeedback(audio_feedback),
-    _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
-    _packetSizeSamples(160),
-    _dtmfEventIsOn(false),
-    _dtmfEventFirstPacketSent(false),
-    _dtmfPayloadType(-1),
-    _dtmfTimestamp(0),
-    _dtmfKey(0),
-    _dtmfLengthSamples(0),
-    _dtmfLevel(0),
-    _dtmfTimeLastSent(0),
-    _dtmfTimestampLastSent(0),
-    _REDPayloadType(-1),
-    _inbandVADactive(false),
-    _cngNBPayloadType(-1),
-    _cngWBPayloadType(-1),
-    _cngSWBPayloadType(-1),
-    _cngFBPayloadType(-1),
-    _lastPayloadType(-1),
-    _audioLevel_dBov(0) {
-}
+                               RtpAudioFeedback* audio_feedback)
+    : _clock(clock),
+      _rtpSender(rtpSender),
+      _audioFeedback(audio_feedback),
+      _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
+      _packetSizeSamples(160),
+      _dtmfEventIsOn(false),
+      _dtmfEventFirstPacketSent(false),
+      _dtmfPayloadType(-1),
+      _dtmfTimestamp(0),
+      _dtmfKey(0),
+      _dtmfLengthSamples(0),
+      _dtmfLevel(0),
+      _dtmfTimeLastSent(0),
+      _dtmfTimestampLastSent(0),
+      _REDPayloadType(-1),
+      _inbandVADactive(false),
+      _cngNBPayloadType(-1),
+      _cngWBPayloadType(-1),
+      _cngSWBPayloadType(-1),
+      _cngFBPayloadType(-1),
+      _lastPayloadType(-1),
+      _audioLevel_dBov(0) {}
 
-RTPSenderAudio::~RTPSenderAudio() {
-}
+RTPSenderAudio::~RTPSenderAudio() {}
 
 int RTPSenderAudio::AudioFrequency() const {
   return kDtmfFrequencyHz;
@@ -58,22 +55,20 @@ int RTPSenderAudio::AudioFrequency() const {
 
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
-int32_t
-RTPSenderAudio::SetAudioPacketSize(const uint16_t packetSizeSamples)
-{
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) {
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
 
-    _packetSizeSamples = packetSizeSamples;
-    return 0;
+  _packetSizeSamples = packetSizeSamples;
+  return 0;
 }
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payloadType,
     const uint32_t frequency,
-    const uint8_t channels,
+    const size_t channels,
     const uint32_t rate,
-    RtpUtility::Payload*& payload) {
+    RtpUtility::Payload** payload) {
   if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     //  we can have multiple CNG payload types
@@ -101,72 +96,65 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     return 0;
     // The default timestamp rate is 8000 Hz, but other rates may be defined.
   }
-  payload = new RtpUtility::Payload;
-  payload->typeSpecific.Audio.frequency = frequency;
-  payload->typeSpecific.Audio.channels = channels;
-  payload->typeSpecific.Audio.rate = rate;
-  payload->audio = true;
-  payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
-  strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+  *payload = new RtpUtility::Payload;
+  (*payload)->typeSpecific.Audio.frequency = frequency;
+  (*payload)->typeSpecific.Audio.channels = channels;
+  (*payload)->typeSpecific.Audio.rate = rate;
+  (*payload)->audio = true;
+  (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
+  strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
   return 0;
 }
 
-bool
-RTPSenderAudio::MarkerBit(const FrameType frameType,
-                          const int8_t payload_type)
-{
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
-    // for audio true for first packet in a speech burst
-    bool markerBit = false;
-    if (_lastPayloadType != payload_type) {
-      if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
-                                 _cngWBPayloadType == payload_type ||
-                                 _cngSWBPayloadType == payload_type ||
-                                 _cngFBPayloadType == payload_type)) {
-        // Only set a marker bit when we change payload type to a non CNG
+bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  // for audio true for first packet in a speech burst
+  bool markerBit = false;
+  if (_lastPayloadType != payload_type) {
+    if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
+                               _cngWBPayloadType == payload_type ||
+                               _cngSWBPayloadType == payload_type ||
+                               _cngFBPayloadType == payload_type)) {
+      // Only set a marker bit when we change payload type to a non CNG
+      return false;
+    }
+
+    // payload_type differ
+    if (_lastPayloadType == -1) {
+      if (frameType != kAudioFrameCN) {
+        // first packet and NOT CNG
+        return true;
+      } else {
+        // first packet and CNG
+        _inbandVADactive = true;
         return false;
       }
-
-      // payload_type differ
-      if (_lastPayloadType == -1) {
-        if (frameType != kAudioFrameCN) {
-          // first packet and NOT CNG
-          return true;
-        } else {
-          // first packet and CNG
-          _inbandVADactive = true;
-          return false;
-        }
-      }
-
-      // not first packet AND
-      // not CNG AND
-      // payload_type changed
-
-      // set a marker bit when we change payload type
-      markerBit = true;
     }
 
-    // For G.723 G.729, AMR etc we can have inband VAD
-    if(frameType == kAudioFrameCN)
-    {
-        _inbandVADactive = true;
+    // not first packet AND
+    // not CNG AND
+    // payload_type changed
 
-    } else if(_inbandVADactive)
-    {
-        _inbandVADactive = false;
-        markerBit = true;
-    }
-    return markerBit;
+    // set a marker bit when we change payload type
+    markerBit = true;
+  }
+
+  // For G.723 G.729, AMR etc we can have inband VAD
+  if (frameType == kAudioFrameCN) {
+    _inbandVADactive = true;
+  } else if (_inbandVADactive) {
+    _inbandVADactive = false;
+    markerBit = true;
+  }
+  return markerBit;
 }
 
-int32_t RTPSenderAudio::SendAudio(
-    const FrameType frameType,
-    const int8_t payloadType,
-    const uint32_t captureTimeStamp,
-    const uint8_t* payloadData,
-    const size_t dataSize,
-    const RTPFragmentationHeader* fragmentation) {
+int32_t RTPSenderAudio::SendAudio(FrameType frameType,
+                                  int8_t payloadType,
+                                  uint32_t captureTimeStamp,
+                                  const uint8_t* payloadData,
+                                  size_t dataSize,
+                                  const RTPFragmentationHeader* fragmentation) {
   // TODO(pwestin) Breakup function in smaller functions.
   size_t payloadSize = dataSize;
   size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
@@ -187,8 +175,8 @@ int32_t RTPSenderAudio::SendAudio(
 
   // Check if we have pending DTMFs to send
   if (!_dtmfEventIsOn && PendingDTMF()) {
-    int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() -
-        _dtmfTimeLastSent;
+    int64_t delaySinceLastDTMF =
+        _clock->TimeInMilliseconds() - _dtmfTimeLastSent;
 
     if (delaySinceLastDTMF > 100) {
       // New tone to play
@@ -204,14 +192,14 @@ int32_t RTPSenderAudio::SendAudio(
   }
   if (dtmfToneStarted) {
     if (_audioFeedback)
-      _audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
+      _audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel);
   }
 
   // A source MAY send events and coded audio packets for the same time
   // but we don't support it
   if (_dtmfEventIsOn) {
-    if (frameType == kFrameEmpty) {
-      // kFrameEmpty is used to drive the DTMF when in CN mode
+    if (frameType == kEmptyFrame) {
+      // kEmptyFrame is used to drive the DTMF when in CN mode
       // it can be triggered more frequently than we want to send the
       // DTMF packets.
       if (packet_size_samples > (captureTimeStamp - _dtmfTimestampLastSent)) {
@@ -261,7 +249,7 @@ int32_t RTPSenderAudio::SendAudio(
     return 0;
   }
   if (payloadSize == 0 || payloadData == NULL) {
-    if (frameType == kFrameEmpty) {
+    if (frameType == kEmptyFrame) {
       // we don't send empty audio RTP packets
       // no error since we use it to drive DTMF when we use VAD
       return 0;
@@ -296,128 +284,120 @@ int32_t RTPSenderAudio::SendAudio(
     // Too large payload buffer.
     return -1;
   }
-    if (red_payload_type >= 0 &&  // Have we configured RED?
-        fragmentation && fragmentation->fragmentationVectorSize > 1 &&
-        !markerBit) {
-      if (timestampOffset <= 0x3fff) {
-        if (fragmentation->fragmentationVectorSize != 2) {
-          // we only support 2 codecs when using RED
-          return -1;
-        }
-        // only 0x80 if we have multiple blocks
-        dataBuffer[rtpHeaderLength++] =
-            0x80 + fragmentation->fragmentationPlType[1];
-        size_t blockLength = fragmentation->fragmentationLength[1];
-
-        // sanity blockLength
-        if (blockLength > 0x3ff) {  // block length 10 bits 1023 bytes
-          return -1;
-        }
-        uint32_t REDheader = (timestampOffset << 10) + blockLength;
-        ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + rtpHeaderLength,
-                                             REDheader);
-        rtpHeaderLength += 3;
-
-        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-        // copy the RED data
-        memcpy(dataBuffer + rtpHeaderLength,
-               payloadData + fragmentation->fragmentationOffset[1],
-               fragmentation->fragmentationLength[1]);
-
-        // copy the normal data
-        memcpy(dataBuffer + rtpHeaderLength +
-                   fragmentation->fragmentationLength[1],
-               payloadData + fragmentation->fragmentationOffset[0],
-               fragmentation->fragmentationLength[0]);
-
-        payloadSize = fragmentation->fragmentationLength[0] +
-                      fragmentation->fragmentationLength[1];
-      } else {
-        // silence for too long send only new data
-        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-        memcpy(dataBuffer + rtpHeaderLength,
-               payloadData + fragmentation->fragmentationOffset[0],
-               fragmentation->fragmentationLength[0]);
-
-        payloadSize = fragmentation->fragmentationLength[0];
+  if (red_payload_type >= 0 &&  // Have we configured RED?
+      fragmentation && fragmentation->fragmentationVectorSize > 1 &&
+      !markerBit) {
+    if (timestampOffset <= 0x3fff) {
+      if (fragmentation->fragmentationVectorSize != 2) {
+        // we only support 2 codecs when using RED
+        return -1;
       }
+      // only 0x80 if we have multiple blocks
+      dataBuffer[rtpHeaderLength++] =
+          0x80 + fragmentation->fragmentationPlType[1];
+      size_t blockLength = fragmentation->fragmentationLength[1];
+
+      // sanity blockLength
+      if (blockLength > 0x3ff) {  // block length 10 bits 1023 bytes
+        return -1;
+      }
+      uint32_t REDheader = (timestampOffset << 10) + blockLength;
+      ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + rtpHeaderLength,
+                                           REDheader);
+      rtpHeaderLength += 3;
+
+      dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+      // copy the RED data
+      memcpy(dataBuffer + rtpHeaderLength,
+             payloadData + fragmentation->fragmentationOffset[1],
+             fragmentation->fragmentationLength[1]);
+
+      // copy the normal data
+      memcpy(
+          dataBuffer + rtpHeaderLength + fragmentation->fragmentationLength[1],
+          payloadData + fragmentation->fragmentationOffset[0],
+          fragmentation->fragmentationLength[0]);
+
+      payloadSize = fragmentation->fragmentationLength[0] +
+                    fragmentation->fragmentationLength[1];
     } else {
-      if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
-        // use the fragment info if we have one
-        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-        memcpy(dataBuffer + rtpHeaderLength,
-               payloadData + fragmentation->fragmentationOffset[0],
-               fragmentation->fragmentationLength[0]);
+      // silence for too long send only new data
+      dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+      memcpy(dataBuffer + rtpHeaderLength,
+             payloadData + fragmentation->fragmentationOffset[0],
+             fragmentation->fragmentationLength[0]);
 
-        payloadSize = fragmentation->fragmentationLength[0];
-      } else {
-        memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
-      }
+      payloadSize = fragmentation->fragmentationLength[0];
     }
-    {
-      CriticalSectionScoped cs(_sendAudioCritsect.get());
-      _lastPayloadType = payloadType;
+  } else {
+    if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
+      // use the fragment info if we have one
+      dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
+      memcpy(dataBuffer + rtpHeaderLength,
+             payloadData + fragmentation->fragmentationOffset[0],
+             fragmentation->fragmentationLength[0]);
+
+      payloadSize = fragmentation->fragmentationLength[0];
+    } else {
+      memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
     }
-    // Update audio level extension, if included.
-    size_t packetSize = payloadSize + rtpHeaderLength;
-    RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
-    RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
-    _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
-                                 (frameType == kAudioFrameSpeech),
-                                 audio_level_dbov);
-    TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
-                           _rtpSender->Timestamp(), "seqnum",
-                           _rtpSender->SequenceNumber());
-    return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                     -1, kAllowRetransmission,
-                                     PacedSender::kHighPriority);
   }
-
-    // Audio level magnitude and voice activity flag are set for each RTP packet
-int32_t
-RTPSenderAudio::SetAudioLevel(const uint8_t level_dBov)
-{
-    if (level_dBov > 127)
-    {
-        return -1;
-    }
+  {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
-    _audioLevel_dBov = level_dBov;
-    return 0;
+    _lastPayloadType = payloadType;
+  }
+  // Update audio level extension, if included.
+  size_t packetSize = payloadSize + rtpHeaderLength;
+  RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
+  RTPHeader rtp_header;
+  rtp_parser.Parse(&rtp_header);
+  _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
+                               (frameType == kAudioFrameSpeech),
+                               audio_level_dbov);
+  TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
+                         _rtpSender->Timestamp(), "seqnum",
+                         _rtpSender->SequenceNumber());
+  return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
+                                   TickTime::MillisecondTimestamp(),
+                                   kAllowRetransmission,
+                                   RtpPacketSender::kHighPriority);
 }
 
-    // Set payload type for Redundant Audio Data RFC 2198
-int32_t
-RTPSenderAudio::SetRED(const int8_t payloadType)
-{
-    if(payloadType < -1 )
-    {
-        return -1;
-    }
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
-    _REDPayloadType = payloadType;
-    return 0;
+// Audio level magnitude and voice activity flag are set for each RTP packet
+int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
+  if (level_dBov > 127) {
+    return -1;
+  }
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  _audioLevel_dBov = level_dBov;
+  return 0;
 }
 
-    // Get payload type for Redundant Audio Data RFC 2198
-int32_t
-RTPSenderAudio::RED(int8_t& payloadType) const
-{
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
-    if(_REDPayloadType == -1)
-    {
-        // not configured
-        return -1;
-    }
-    payloadType = _REDPayloadType;
-    return 0;
+// Set payload type for Redundant Audio Data RFC 2198
+int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
+  if (payloadType < -1) {
+    return -1;
+  }
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  _REDPayloadType = payloadType;
+  return 0;
+}
+
+// Get payload type for Redundant Audio Data RFC 2198
+int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
+  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  if (_REDPayloadType == -1) {
+    // not configured
+    return -1;
+  }
+  *payloadType = _REDPayloadType;
+  return 0;
 }
 
 // Send a TelephoneEvent tone using RFC 2833 (4733)
-int32_t RTPSenderAudio::SendTelephoneEvent(const uint8_t key,
-                                           const uint16_t time_ms,
-                                           const uint8_t level) {
+int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
+                                           uint16_t time_ms,
+                                           uint8_t level) {
   {
     CriticalSectionScoped lock(_sendAudioCritsect.get());
     if (_dtmfPayloadType < 0) {
@@ -428,63 +408,57 @@ int32_t RTPSenderAudio::SendTelephoneEvent(const uint8_t key,
   return AddDTMF(key, time_ms, level);
 }
 
-int32_t
-RTPSenderAudio::SendTelephoneEventPacket(bool ended,
-                                         int8_t dtmf_payload_type,
-                                         uint32_t dtmfTimeStamp,
-                                         uint16_t duration,
-                                         bool markerBit)
-{
-    uint8_t dtmfbuffer[IP_PACKET_SIZE];
-    uint8_t sendCount = 1;
-    int32_t retVal = 0;
+int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
+                                                 int8_t dtmf_payload_type,
+                                                 uint32_t dtmfTimeStamp,
+                                                 uint16_t duration,
+                                                 bool markerBit) {
+  uint8_t dtmfbuffer[IP_PACKET_SIZE];
+  uint8_t sendCount = 1;
+  int32_t retVal = 0;
 
-    if(ended)
-    {
-        // resend last packet in an event 3 times
-        sendCount = 3;
-    }
-    do
-    {
-        //Send DTMF data
-        _rtpSender->BuildRTPheader(dtmfbuffer, dtmf_payload_type, markerBit,
-                                   dtmfTimeStamp, _clock->TimeInMilliseconds());
+  if (ended) {
+    // resend last packet in an event 3 times
+    sendCount = 3;
+  }
+  do {
+    // Send DTMF data
+    _rtpSender->BuildRTPheader(dtmfbuffer, dtmf_payload_type, markerBit,
+                               dtmfTimeStamp, _clock->TimeInMilliseconds());
 
-        // reset CSRC and X bit
-        dtmfbuffer[0] &= 0xe0;
+    // reset CSRC and X bit
+    dtmfbuffer[0] &= 0xe0;
 
-        //Create DTMF data
-        /*    From RFC 2833:
+    // Create DTMF data
+    /*    From RFC 2833:
 
-         0                   1                   2                   3
-         0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-        |     event     |E|R| volume    |          duration             |
-        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-        */
-        // R bit always cleared
-        uint8_t R = 0x00;
-        uint8_t volume = _dtmfLevel;
+     0                   1                   2                   3
+     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |     event     |E|R| volume    |          duration             |
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    */
+    // R bit always cleared
+    uint8_t R = 0x00;
+    uint8_t volume = _dtmfLevel;
 
-        // First packet un-ended
-        uint8_t E = ended ? 0x80 : 0x00;
+    // First packet un-ended
+    uint8_t E = ended ? 0x80 : 0x00;
 
-        // First byte is Event number, equals key number
-        dtmfbuffer[12] = _dtmfKey;
-        dtmfbuffer[13] = E|R|volume;
-        ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
+    // First byte is Event number, equals key number
+    dtmfbuffer[12] = _dtmfKey;
+    dtmfbuffer[13] = E | R | volume;
+    ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
 
-        TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                             "Audio::SendTelephoneEvent", "timestamp",
-                             dtmfTimeStamp, "seqnum",
-                             _rtpSender->SequenceNumber());
-        retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
-                                           kAllowRetransmission,
-                                           PacedSender::kHighPriority);
-        sendCount--;
+    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                         "Audio::SendTelephoneEvent", "timestamp",
+                         dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
+    retVal = _rtpSender->SendToNetwork(
+        dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+        kAllowRetransmission, RtpPacketSender::kHighPriority);
+    sendCount--;
+  } while (sendCount > 0 && retVal == 0);
 
-    }while (sendCount > 0 && retVal == 0);
-
-    return retVal;
+  return retVal;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 762668a4e4..1e96d17a67 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -19,94 +19,91 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-class RTPSenderAudio: public DTMFqueue
-{
-public:
-    RTPSenderAudio(const int32_t id,
-                   Clock* clock,
-                   RTPSender* rtpSender,
-                   RtpAudioFeedback* audio_feedback);
-    virtual ~RTPSenderAudio();
+class RTPSenderAudio : public DTMFqueue {
+ public:
+  RTPSenderAudio(Clock* clock,
+                 RTPSender* rtpSender,
+                 RtpAudioFeedback* audio_feedback);
+  virtual ~RTPSenderAudio();
 
-    int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                 const int8_t payloadType,
-                                 const uint32_t frequency,
-                                 const uint8_t channels,
-                                 const uint32_t rate,
-                                 RtpUtility::Payload*& payload);
+  int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                               int8_t payloadType,
+                               uint32_t frequency,
+                               size_t channels,
+                               uint32_t rate,
+                               RtpUtility::Payload** payload);
 
-    int32_t SendAudio(const FrameType frameType,
-                      const int8_t payloadType,
-                      const uint32_t captureTimeStamp,
-                      const uint8_t* payloadData,
-                      const size_t payloadSize,
-                      const RTPFragmentationHeader* fragmentation);
+  int32_t SendAudio(FrameType frameType,
+                    int8_t payloadType,
+                    uint32_t captureTimeStamp,
+                    const uint8_t* payloadData,
+                    size_t payloadSize,
+                    const RTPFragmentationHeader* fragmentation);
 
-    // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
-    int32_t SetAudioPacketSize(const uint16_t packetSizeSamples);
+  // set audio packet size, used to determine when it's time to send a DTMF
+  // packet in silence (CNG)
+  int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
 
-    // Store the audio level in dBov for header-extension-for-audio-level-indication.
-    // Valid range is [0,100]. Actual value is negative.
-    int32_t SetAudioLevel(const uint8_t level_dBov);
+  // Store the audio level in dBov for
+  // header-extension-for-audio-level-indication.
+  // Valid range is [0,100]. Actual value is negative.
+  int32_t SetAudioLevel(uint8_t level_dBov);
 
-    // Send a DTMF tone using RFC 2833 (4733)
-    int32_t SendTelephoneEvent(const uint8_t key,
-                               const uint16_t time_ms,
-                               const uint8_t level);
+  // Send a DTMF tone using RFC 2833 (4733)
+  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
-    int AudioFrequency() const;
+  int AudioFrequency() const;
 
-    // Set payload type for Redundant Audio Data RFC 2198
-    int32_t SetRED(const int8_t payloadType);
+  // Set payload type for Redundant Audio Data RFC 2198
+  int32_t SetRED(int8_t payloadType);
 
-    // Get payload type for Redundant Audio Data RFC 2198
-    int32_t RED(int8_t& payloadType) const;
+  // Get payload type for Redundant Audio Data RFC 2198
+  int32_t RED(int8_t* payloadType) const;
 
-protected:
-    int32_t SendTelephoneEventPacket(bool ended,
-                                     int8_t dtmf_payload_type,
-                                     uint32_t dtmfTimeStamp,
-                                     uint16_t duration,
-                                     bool markerBit); // set on first packet in talk burst
+ protected:
+  int32_t SendTelephoneEventPacket(
+      bool ended,
+      int8_t dtmf_payload_type,
+      uint32_t dtmfTimeStamp,
+      uint16_t duration,
+      bool markerBit);  // set on first packet in talk burst
 
-    bool MarkerBit(const FrameType frameType,
-                   const int8_t payloadType);
+  bool MarkerBit(const FrameType frameType, const int8_t payloadType);
 
-private:
- const int32_t _id;
- Clock* const _clock;
- RTPSender* const _rtpSender;
- RtpAudioFeedback* const _audioFeedback;
+ private:
+  Clock* const _clock;
+  RTPSender* const _rtpSender;
+  RtpAudioFeedback* const _audioFeedback;
 
- rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
 
- uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
+  uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
 
- // DTMF
- bool _dtmfEventIsOn;
- bool _dtmfEventFirstPacketSent;
- int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
- uint32_t _dtmfTimestamp;
- uint8_t _dtmfKey;
- uint32_t _dtmfLengthSamples;
- uint8_t _dtmfLevel;
- int64_t _dtmfTimeLastSent;
- uint32_t _dtmfTimestampLastSent;
+  // DTMF
+  bool _dtmfEventIsOn;
+  bool _dtmfEventFirstPacketSent;
+  int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
+  uint32_t _dtmfTimestamp;
+  uint8_t _dtmfKey;
+  uint32_t _dtmfLengthSamples;
+  uint8_t _dtmfLevel;
+  int64_t _dtmfTimeLastSent;
+  uint32_t _dtmfTimestampLastSent;
 
- int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
 
- // VAD detection, used for markerbit
- bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
- int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
+  // VAD detection, used for markerbit
+  bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
 
- // Audio level indication
- // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
- uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+  // Audio level indication
+  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
+  uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index f6ea40b7b7..6bc122201a 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -12,20 +12,22 @@
  * This file includes unit tests for the RTPSender.
  */
 
-#include "testing/gtest/include/gtest/gtest.h"
+#include <list>
+#include <vector>
 
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/pacing/include/mock/mock_paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
-#include "webrtc/system_wrappers/interface/stl_util.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/include/stl_util.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/typedefs.h"
 
@@ -36,6 +38,7 @@ const int kTransmissionTimeOffsetExtensionId = 1;
 const int kAbsoluteSendTimeExtensionId = 14;
 const int kTransportSequenceNumberExtensionId = 13;
 const int kPayload = 100;
+const int kRtxPayload = 98;
 const uint32_t kTimestamp = 10;
 const uint16_t kSeqNum = 33;
 const int kTimeOffset = 22222;
@@ -63,7 +66,7 @@ size_t GetPayloadDataLength(const RTPHeader& rtp_header,
 }
 
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
-  return 0x00fffffful & ((time_ms << 18) / 1000);
+  return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
 }
 
 class LoopbackTransportTest : public webrtc::Transport {
@@ -72,23 +75,24 @@ class LoopbackTransportTest : public webrtc::Transport {
       : packets_sent_(0),
         last_sent_packet_len_(0),
         total_bytes_sent_(0),
-        last_sent_packet_(NULL) {}
+        last_sent_packet_(nullptr) {}
 
   ~LoopbackTransportTest() {
     STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
   }
-  int SendPacket(int channel, const void *data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
     packets_sent_++;
-    rtc::Buffer* buffer = new rtc::Buffer(data, len);
-    last_sent_packet_ = reinterpret_cast<uint8_t*>(buffer->data());
+    rtc::Buffer* buffer =
+        new rtc::Buffer(reinterpret_cast<const uint8_t*>(data), len);
+    last_sent_packet_ = buffer->data();
     last_sent_packet_len_ = len;
     total_bytes_sent_ += len;
     sent_packets_.push_back(buffer);
-    return static_cast<int>(len);
-  }
-  int SendRTCPPacket(int channel, const void* data, size_t len) override {
-    return -1;
+    return true;
   }
+  bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
   int packets_sent_;
   size_t last_sent_packet_len_;
   size_t total_bytes_sent_;
@@ -98,6 +102,20 @@ class LoopbackTransportTest : public webrtc::Transport {
 
 }  // namespace
 
+class MockRtpPacketSender : public RtpPacketSender {
+ public:
+  MockRtpPacketSender() {}
+  virtual ~MockRtpPacketSender() {}
+
+  MOCK_METHOD6(InsertPacket,
+               void(Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission));
+};
+
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
@@ -107,18 +125,22 @@ class RtpSenderTest : public ::testing::Test {
         payload_(kPayload),
         transport_(),
         kMarkerBit(true) {
-    EXPECT_CALL(mock_paced_sender_,
-        SendPacket(_, _, _, _, _, _)).WillRepeatedly(testing::Return(true));
+    EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
+        .WillRepeatedly(testing::Return());
   }
 
-  void SetUp() override {
-    rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
-                                    &mock_paced_sender_, NULL, NULL, NULL));
+  void SetUp() override { SetUpRtpSender(true); }
+
+  void SetUpRtpSender(bool pacer) {
+    rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+                                    pacer ? &mock_paced_sender_ : nullptr,
+                                    nullptr, nullptr, nullptr, nullptr,
+                                    nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
-  MockPacedSender mock_paced_sender_;
+  MockRtpPacketSender mock_paced_sender_;
   rtc::scoped_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
@@ -141,27 +163,29 @@ class RtpSenderTest : public ::testing::Test {
 
   void SendPacket(int64_t capture_time_ms, int payload_length) {
     uint32_t timestamp = capture_time_ms * 90;
-    int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
-                                                     kPayload,
-                                                     kMarkerBit,
-                                                     timestamp,
-                                                     capture_time_ms);
+    int32_t rtp_length = rtp_sender_->BuildRTPheader(
+        packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
     ASSERT_GE(rtp_length, 0);
 
     // Packet should be stored in a send bucket.
-    EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
-                                            payload_length,
-                                            rtp_length,
-                                            capture_time_ms,
-                                            kAllowRetransmission,
-                                            PacedSender::kNormalPriority));
+    EXPECT_EQ(0, rtp_sender_->SendToNetwork(
+                     packet_, payload_length, rtp_length, capture_time_ms,
+                     kAllowRetransmission, RtpPacketSender::kNormalPriority));
   }
 };
 
+// TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
+// default code path.
+class RtpSenderTestWithoutPacer : public RtpSenderTest {
+ public:
+  void SetUp() override { SetUpRtpSender(false); }
+};
+
 class RtpSenderVideoTest : public RtpSenderTest {
  protected:
-  virtual void SetUp() override {
-    RtpSenderTest::SetUp();
+  void SetUp() override {
+    // TODO(pbos): Set up to use pacer.
+    SetUpRtpSender(false);
     rtp_sender_video_.reset(
         new RTPSenderVideo(&fake_clock_, rtp_sender_.get()));
   }
@@ -184,7 +208,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
     } else {
       ASSERT_EQ(kRtpHeaderSize, length);
     }
-    ASSERT_TRUE(rtp_parser.Parse(rtp_header, map));
+    ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map));
     ASSERT_FALSE(rtp_parser.RTCP());
     EXPECT_EQ(payload_, rtp_header.payloadType);
     EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -197,56 +221,61 @@ class RtpSenderVideoTest : public RtpSenderTest {
   }
 };
 
-TEST_F(RtpSenderTest, RegisterRtpTransmissionTimeOffsetHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer,
+       RegisterRtpTransmissionTimeOffsetHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength,
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset));
+                   kRtpExtensionTransmissionTimeOffset));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpAbsoluteSendTimeHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAbsoluteSendTimeHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAbsoluteSendTimeLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime));
+                   kRtpExtensionAbsoluteSendTime));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpAudioLevelHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAudioLevelHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel));
+  EXPECT_EQ(0,
+            rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kTransmissionTimeOffsetLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kTransmissionTimeOffsetLength +
                                     kAbsoluteSendTimeLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(
                 kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength +
                 kAbsoluteSendTimeLength + kAudioLevelLength),
@@ -262,18 +291,18 @@ TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) {
 
   // Deregister starts.
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset));
+                   kRtpExtensionTransmissionTimeOffset));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAbsoluteSendTimeLength +
                                     kAudioLevelLength + kVideoRotationLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime));
+                   kRtpExtensionAbsoluteSendTime));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAudioLevelLength + kVideoRotationLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
-  EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel));
+  EXPECT_EQ(0,
+            rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel));
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
@@ -282,7 +311,7 @@ TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpVideoRotationHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpVideoRotationHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
@@ -297,7 +326,7 @@ TEST_F(RtpSenderTest, RegisterRtpVideoRotationHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacket) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize, length);
@@ -306,7 +335,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, NULL);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -317,14 +346,17 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
   EXPECT_FALSE(rtp_header.extension.hasAudioLevel);
   EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime);
+  EXPECT_FALSE(rtp_header.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header.extension.videoRotation);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
+TEST_F(RtpSenderTestWithoutPacer,
+       BuildRTPPacketWithTransmissionOffsetExtension) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -338,7 +370,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -349,7 +381,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -358,11 +390,13 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
   EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
+TEST_F(RtpSenderTestWithoutPacer,
+       BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
   const int kNegTimeOffset = -500;
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -376,7 +410,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -386,10 +420,11 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
   EXPECT_EQ(kNegTimeOffset, rtp_header.extension.transmissionTimeOffset);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -402,7 +437,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -413,7 +448,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -423,7 +458,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
 }
 
 // Test CVO header extension is only set when marker bit is true.
-TEST_F(RtpSenderTest, BuildRTPPacketWithVideoRotation_MarkerBit) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
@@ -441,7 +476,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithVideoRotation_MarkerBit) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
@@ -451,7 +486,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithVideoRotation_MarkerBit) {
 }
 
 // Test CVO header extension is not set when marker bit is false.
-TEST_F(RtpSenderTest, DISABLED_BuildRTPPacketWithVideoRotation_NoMarkerBit) {
+TEST_F(RtpSenderTestWithoutPacer,
+       DISABLED_BuildRTPPacketWithVideoRotation_NoMarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
@@ -468,16 +504,16 @@ TEST_F(RtpSenderTest, DISABLED_BuildRTPPacketWithVideoRotation_NoMarkerBit) {
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header, &map));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header, false);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_FALSE(rtp_header.extension.hasVideoRotation);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
@@ -489,44 +525,46 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
-  // Expect kAudioLevel + 0x80 because we set "voiced" to true in the call to
-  // UpdateAudioLevel(), above.
-  EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
+  EXPECT_TRUE(rtp_header.extension.voiceActivity);
+  EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
   EXPECT_EQ(length, rtp_header2.headerLength);
   EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
+  EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
   EXPECT_EQ(0,
             rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
@@ -541,7 +579,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
 
   RtpHeaderExtensionMap map;
@@ -551,7 +589,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
   map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
   map.Register(kRtpExtensionTransportSequenceNumber,
                kTransportSequenceNumberExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
@@ -563,13 +601,14 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
   EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
   EXPECT_EQ(kAbsoluteSendTime, rtp_header.extension.absoluteSendTime);
-  EXPECT_EQ(kAudioLevel + 0x80u, rtp_header.extension.audioLevel);
+  EXPECT_TRUE(rtp_header.extension.voiceActivity);
+  EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
   EXPECT_EQ(kTransportSequenceNumber,
             rtp_header.extension.transportSequenceNumber);
 
   // Parse without map extension
   webrtc::RTPHeader rtp_header2;
-  const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+  const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
@@ -581,20 +620,23 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
 
   EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
+  EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber);
 }
 
 TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
-  EXPECT_CALL(mock_paced_sender_,
-              SendPacket(PacedSender::kNormalPriority, _, kSeqNum, _, _, _)).
-                  WillOnce(testing::Return(false));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
+                                               _, kSeqNum, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
@@ -603,12 +645,9 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
-  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
-                                          0,
-                                          rtp_length,
-                                          capture_time_ms,
-                                          kAllowRetransmission,
-                                          PacedSender::kNormalPriority));
+  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+                                          capture_time_ms, kAllowRetransmission,
+                                          RtpPacketSender::kNormalPriority));
 
   EXPECT_EQ(0, transport_.packets_sent_);
 
@@ -628,7 +667,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
@@ -639,15 +678,17 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
 }
 
 TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
-  EXPECT_CALL(mock_paced_sender_,
-              SendPacket(PacedSender::kNormalPriority, _, kSeqNum, _, _, _)).
-                  WillOnce(testing::Return(false));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
+                                               _, kSeqNum, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
@@ -656,18 +697,15 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
-  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
-                                          0,
-                                          rtp_length,
-                                          capture_time_ms,
-                                          kAllowRetransmission,
-                                          PacedSender::kNormalPriority));
+  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+                                          capture_time_ms, kAllowRetransmission,
+                                          RtpPacketSender::kNormalPriority));
 
   EXPECT_EQ(0, transport_.packets_sent_);
 
   EXPECT_CALL(mock_paced_sender_,
-              SendPacket(PacedSender::kHighPriority, _, kSeqNum, _, _, _)).
-                  WillOnce(testing::Return(false));
+              InsertPacket(RtpPacketSender::kHighPriority, _, kSeqNum, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
@@ -689,7 +727,7 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
-  const bool valid_rtp_header = rtp_parser.Parse(rtp_header, &map);
+  const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
@@ -704,25 +742,27 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
 TEST_F(RtpSenderTest, SendPadding) {
   // Make all (non-padding) packets go to send queue.
   EXPECT_CALL(mock_paced_sender_,
-              SendPacket(PacedSender::kNormalPriority, _, _, _, _, _)).
-                  WillRepeatedly(testing::Return(false));
+              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   uint16_t seq_num = kSeqNum;
   uint32_t timestamp = kTimestamp;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   size_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
+                   kRtpExtensionTransmissionTimeOffset,
+                   kTransmissionTimeOffsetExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
   // Create and set up parser.
   rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
-  ASSERT_TRUE(rtp_parser.get() != NULL);
+  ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                          kTransmissionTimeOffsetExtensionId);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
@@ -733,16 +773,14 @@ TEST_F(RtpSenderTest, SendPadding) {
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
+  const uint32_t media_packet_timestamp = timestamp;
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
-  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
-                                          0,
-                                          rtp_length,
-                                          capture_time_ms,
-                                          kAllowRetransmission,
-                                          PacedSender::kNormalPriority));
+  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+                                          capture_time_ms, kAllowRetransmission,
+                                          RtpPacketSender::kNormalPriority));
 
   int total_packets_sent = 0;
   EXPECT_EQ(total_packets_sent, transport_.packets_sent_);
@@ -768,14 +806,18 @@ TEST_F(RtpSenderTest, SendPadding) {
     EXPECT_EQ(kMaxPaddingLength + rtp_header_len,
               transport_.last_sent_packet_len_);
     // Parse sent packet.
-    ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, kPaddingBytes,
+    ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
+                                  transport_.last_sent_packet_len_,
                                   &rtp_header));
+    EXPECT_EQ(kMaxPaddingLength, rtp_header.paddingLength);
 
-    // Verify sequence number and timestamp.
+    // Verify sequence number and timestamp. The timestamp should be the same
+    // as the last media packet.
     EXPECT_EQ(seq_num++, rtp_header.sequenceNumber);
-    EXPECT_EQ(timestamp, rtp_header.timestamp);
+    EXPECT_EQ(media_packet_timestamp, rtp_header.timestamp);
     // Verify transmission time offset.
-    EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
+    int offset = timestamp - media_packet_timestamp;
+    EXPECT_EQ(offset, rtp_header.extension.transmissionTimeOffset);
     uint64_t expected_send_time =
         ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
     EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
@@ -785,26 +827,23 @@ TEST_F(RtpSenderTest, SendPadding) {
 
   // Send a regular video packet again.
   capture_time_ms = fake_clock_.TimeInMilliseconds();
-  rtp_length_int = rtp_sender_->BuildRTPheader(
-      packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
+  rtp_length_int = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit,
+                                               timestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
-  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
-                                          0,
-                                          rtp_length,
-                                          capture_time_ms,
-                                          kAllowRetransmission,
-                                          PacedSender::kNormalPriority));
+  EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
+                                          capture_time_ms, kAllowRetransmission,
+                                          RtpPacketSender::kNormalPriority));
 
   rtp_sender_->TimeToSendPacket(seq_num, capture_time_ms, false);
   // Process send bucket.
   EXPECT_EQ(++total_packets_sent, transport_.packets_sent_);
   EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
   // Parse sent packet.
-  ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_, rtp_length,
-                                &rtp_header));
+  ASSERT_TRUE(
+      rtp_parser->Parse(transport_.last_sent_packet_, rtp_length, &rtp_header));
 
   // Verify sequence number and timestamp.
   EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
@@ -818,19 +857,22 @@ TEST_F(RtpSenderTest, SendPadding) {
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
-  rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
-                                  &mock_paced_sender_, NULL, NULL, NULL));
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
+                                  &mock_paced_sender_, nullptr, nullptr,
+                                  nullptr, nullptr, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
+  rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
   // Make all packets go through the pacer.
   EXPECT_CALL(mock_paced_sender_,
-              SendPacket(PacedSender::kNormalPriority, _, _, _, _, _)).
-                  WillRepeatedly(testing::Return(false));
+              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
@@ -840,45 +882,44 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
   // Create and set up parser.
   rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
-  ASSERT_TRUE(rtp_parser.get() != NULL);
+  ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                          kTransmissionTimeOffsetExtensionId);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                          kAbsoluteSendTimeExtensionId);
   rtp_sender_->SetTargetBitrate(300000);
   const size_t kNumPayloadSizes = 10;
-  const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, 750,
-      800, 850, 900, 950};
+  const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
+                                                  750, 800, 850, 900, 950};
   // Send 10 packets of increasing size.
   for (size_t i = 0; i < kNumPayloadSizes; ++i) {
     int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
-    EXPECT_CALL(transport, SendPacket(_, _, _))
-        .WillOnce(testing::ReturnArg<2>());
+    EXPECT_CALL(transport, SendRtp(_, _, _)).WillOnce(testing::Return(true));
     SendPacket(capture_time_ms, kPayloadSizes[i]);
     rtp_sender_->TimeToSendPacket(seq_num++, capture_time_ms, false);
     fake_clock_.AdvanceTimeMilliseconds(33);
   }
   // The amount of padding to send it too small to send a payload packet.
-  EXPECT_CALL(transport,
-              SendPacket(_, _, kMaxPaddingSize + rtp_header_len))
-      .WillOnce(testing::ReturnArg<2>());
+  EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
+      .WillOnce(testing::Return(true));
   EXPECT_EQ(kMaxPaddingSize, rtp_sender_->TimeToSendPadding(49));
 
-  EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[0] +
-                                    rtp_header_len + kRtxHeaderSize))
-      .WillOnce(testing::ReturnArg<2>());
+  EXPECT_CALL(transport,
+              SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
+      .WillOnce(testing::Return(true));
   EXPECT_EQ(kPayloadSizes[0], rtp_sender_->TimeToSendPadding(500));
 
-  EXPECT_CALL(transport, SendPacket(_, _, kPayloadSizes[kNumPayloadSizes - 1] +
-                                    rtp_header_len + kRtxHeaderSize))
-      .WillOnce(testing::ReturnArg<2>());
-  EXPECT_CALL(transport, SendPacket(_, _, kMaxPaddingSize + rtp_header_len))
-      .WillOnce(testing::ReturnArg<2>());
+  EXPECT_CALL(transport, SendRtp(_, kPayloadSizes[kNumPayloadSizes - 1] +
+                                        rtp_header_len + kRtxHeaderSize,
+                                 _))
+      .WillOnce(testing::Return(true));
+  EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
+      .WillOnce(testing::Return(true));
   EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
             rtp_sender_->TimeToSendPadding(999));
 }
 
-TEST_F(RtpSenderTest, SendGenericVideo) {
+TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
@@ -886,17 +927,17 @@ TEST_F(RtpSenderTest, SendGenericVideo) {
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   // Send keyframe
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                             4321, payload, sizeof(payload),
-                                             NULL));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
+                                       payload, sizeof(payload), nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
-  const uint8_t* payload_data = GetPayloadData(rtp_header,
-      transport_.last_sent_packet_);
+  const uint8_t* payload_data =
+      GetPayloadData(rtp_header, transport_.last_sent_packet_);
   uint8_t generic_header = *payload_data++;
 
   ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
@@ -914,11 +955,11 @@ TEST_F(RtpSenderTest, SendGenericVideo) {
 
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
                                              1234, 4321, payload,
-                                             sizeof(payload), NULL));
+                                             sizeof(payload), nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
                                           transport_.last_sent_packet_len_);
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_);
   generic_header = *payload_data++;
@@ -950,8 +991,9 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
     FrameCounts frame_counts_;
   } callback;
 
-  rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
-                                  &mock_paced_sender_, NULL, &callback, NULL));
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+                                  &mock_paced_sender_, nullptr, nullptr,
+                                  nullptr, &callback, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
@@ -961,18 +1003,18 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                             4321, payload, sizeof(payload),
-                                             NULL));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
+                                       payload, sizeof(payload), nullptr));
 
   EXPECT_EQ(1U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(0, callback.frame_counts_.delta_frames);
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta,
-                                             payload_type, 1234, 4321, payload,
-                                             sizeof(payload), NULL));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
+                                             1234, 4321, payload,
+                                             sizeof(payload), nullptr));
 
   EXPECT_EQ(2U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
@@ -1002,8 +1044,9 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
     BitrateStatistics total_stats_;
     BitrateStatistics retransmit_stats_;
   } callback;
-  rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
-                                  &mock_paced_sender_, &callback, NULL, NULL));
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+                                  nullptr, nullptr, nullptr, &callback, nullptr,
+                                  nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
   const uint32_t kNumPackets = 15;
@@ -1013,9 +1056,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
-  ASSERT_EQ(
-      0,
-      rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 1500));
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
+                                            0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
@@ -1027,13 +1069,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
   // Send a few frames.
   for (uint32_t i = 0; i < kNumPackets; ++i) {
     ASSERT_EQ(0,
-              rtp_sender_->SendOutgoingData(kVideoFrameKey,
-                                            payload_type,
-                                            1234,
-                                            4321,
-                                            payload,
-                                            sizeof(payload),
-                                            0));
+              rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                            4321, payload, sizeof(payload), 0));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
@@ -1060,17 +1097,17 @@ class RtpSenderAudioTest : public RtpSenderTest {
 
   void SetUp() override {
     payload_ = kAudioPayload;
-    rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL,
-                                    &mock_paced_sender_, NULL, NULL, NULL));
+    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
+                                    nullptr, nullptr, nullptr, nullptr, nullptr,
+                                    nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
-TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
+TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
-    TestCallback()
-      : StreamDataCountersCallback(), ssrc_(0), counters_() {}
+    TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
     void DataCountersUpdated(const StreamDataCounters& counters,
@@ -1096,7 +1133,6 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
       MatchPacketCounter(counters.retransmitted, counters_.retransmitted);
       EXPECT_EQ(counters.fec.packets, counters_.fec.packets);
     }
-
   } callback;
 
   const uint8_t kRedPayloadType = 96;
@@ -1112,9 +1148,9 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
   rtp_sender_->RegisterRtpStatisticsCallback(&callback);
 
   // Send a frame.
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                             4321, payload, sizeof(payload),
-                                             NULL));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
+                                       payload, sizeof(payload), nullptr));
   StreamDataCounters expected;
   expected.transmitted.payload_bytes = 6;
   expected.transmitted.header_bytes = 12;
@@ -1157,14 +1193,14 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
   rtp_sender_->SetFecParameters(&fec_params, &fec_params);
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
                                              1234, 4321, payload,
-                                             sizeof(payload), NULL));
+                                             sizeof(payload), nullptr));
   expected.transmitted.payload_bytes = 40;
   expected.transmitted.header_bytes = 60;
   expected.transmitted.packets = 5;
   expected.fec.packets = 1;
   callback.Matches(ssrc, expected);
 
-  rtp_sender_->RegisterRtpStatisticsCallback(NULL);
+  rtp_sender_->RegisterRtpStatisticsCallback(nullptr);
 }
 
 TEST_F(RtpSenderAudioTest, SendAudio) {
@@ -1174,17 +1210,17 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
-                                             4321, payload, sizeof(payload),
-                                             NULL));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
+                                       payload, sizeof(payload), nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
-  const uint8_t* payload_data = GetPayloadData(rtp_header,
-      transport_.last_sent_packet_);
+  const uint8_t* payload_data =
+      GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
@@ -1194,8 +1230,8 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
 
 TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
-  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
-      kRtpExtensionAudioLevel, kAudioLevelExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+                                                       kAudioLevelExtensionId));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
@@ -1203,28 +1239,29 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
-                                             4321, payload, sizeof(payload),
-                                             NULL));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
+                                       payload, sizeof(payload), nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
-  ASSERT_TRUE(rtp_parser.Parse(rtp_header));
+  ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
-  const uint8_t* payload_data = GetPayloadData(rtp_header,
-                                               transport_.last_sent_packet_);
+  const uint8_t* payload_data =
+      GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
 
   EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
 
-  uint8_t extension[] = { 0xbe, 0xde, 0x00, 0x01,
-                          (kAudioLevelExtensionId << 4) + 0, // ID + length.
-                          kAudioLevel,                       // Data.
-                          0x00, 0x00                         // Padding.
-                        };
+  uint8_t extension[] = {
+      0xbe, 0xde, 0x00, 0x01,
+      (kAudioLevelExtensionId << 4) + 0,  // ID + length.
+      kAudioLevel,                        // Data.
+      0x00, 0x00                          // Padding.
+  };
 
   EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension),
                       sizeof(extension)));
@@ -1235,77 +1272,66 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
 // audio channel.
 // This test checks the marker bit for the first packet and the consequent
 // packets of the same telephone event. Since it is specifically for DTMF
-// events, ignoring audio packets and sending kFrameEmpty instead of those.
+// events, ignoring audio packets and sending kEmptyFrame instead of those.
 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
   uint8_t payload_type = 126;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 0,
-                                            0, 0));
+  ASSERT_EQ(0,
+            rtp_sender_->RegisterPayload(payload_name, payload_type, 0, 0, 0));
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
-  strcpy(payload_name, "payload_name");
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 8000,
-                                            1, 0));
+  const char kPayloadName[] = "payload_name";
+  ASSERT_EQ(
+      0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
   // During start, it takes the starting timestamp as last sent timestamp.
   // The duration is calculated as the difference of current and last sent
   // timestamp. So for first call it will skip since the duration is zero.
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
-                                             capture_time_ms,
-                                             0, NULL, 0,
-                                             NULL));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
+                                             capture_time_ms, 0, nullptr, 0,
+                                             nullptr));
   // DTMF Sample Length is (Frequency/1000) * Duration.
   // So in this case, it is (8000/1000) * 500 = 4000.
   // Sending it as two packets.
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
-                                             capture_time_ms+2000,
-                                             0, NULL, 0,
-                                             NULL));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
+                                             capture_time_ms + 2000, 0, nullptr,
+                                             0, nullptr));
   rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
-  ASSERT_TRUE(rtp_parser.get() != NULL);
+  ASSERT_TRUE(rtp_parser.get() != nullptr);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
-                                transport_.last_sent_packet_len_,
-                                &rtp_header));
+                                transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(rtp_header.markerBit);
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
-                                             capture_time_ms+4000,
-                                             0, NULL, 0,
-                                             NULL));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
+                                             capture_time_ms + 4000, 0, nullptr,
+                                             0, nullptr));
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
-                                transport_.last_sent_packet_len_,
-                                &rtp_header));
+                                transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(rtp_header.markerBit);
 }
 
-TEST_F(RtpSenderTest, BytesReportedCorrectly) {
+TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   const char* kPayloadName = "GENERIC";
   const uint8_t kPayloadType = 127;
   rtp_sender_->SetSSRC(1234);
   rtp_sender_->SetRtxSsrc(4321);
-  rtp_sender_->SetRtxPayloadType(kPayloadType - 1);
+  rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
 
-  ASSERT_EQ(
-      0,
-      rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000, 0, 1500));
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
+                                            0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(0,
-            rtp_sender_->SendOutgoingData(kVideoFrameKey,
-                                          kPayloadType,
-                                          1234,
-                                          4321,
-                                          payload,
-                                          sizeof(payload),
-                                          0));
+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
+                                       payload, sizeof(payload), 0));
 
   // Will send 2 full-size padding packets.
   rtp_sender_->TimeToSendPadding(1);
@@ -1325,17 +1351,46 @@ TEST_F(RtpSenderTest, BytesReportedCorrectly) {
   EXPECT_EQ(rtx_stats.transmitted.padding_bytes, 2 * kMaxPaddingSize);
 
   EXPECT_EQ(rtp_stats.transmitted.TotalBytes(),
-      rtp_stats.transmitted.payload_bytes +
-      rtp_stats.transmitted.header_bytes +
-      rtp_stats.transmitted.padding_bytes);
+            rtp_stats.transmitted.payload_bytes +
+                rtp_stats.transmitted.header_bytes +
+                rtp_stats.transmitted.padding_bytes);
   EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
-      rtx_stats.transmitted.payload_bytes +
-      rtx_stats.transmitted.header_bytes +
-      rtx_stats.transmitted.padding_bytes);
+            rtx_stats.transmitted.payload_bytes +
+                rtx_stats.transmitted.header_bytes +
+                rtx_stats.transmitted.padding_bytes);
 
-  EXPECT_EQ(transport_.total_bytes_sent_,
-            rtp_stats.transmitted.TotalBytes() +
-            rtx_stats.transmitted.TotalBytes());
+  EXPECT_EQ(
+      transport_.total_bytes_sent_,
+      rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes());
+}
+
+TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
+  const int32_t kPacketSize = 1400;
+  const int32_t kNumPackets = 30;
+
+  rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
+  // Set bitrate (in kbps) to fit kNumPackets á kPacketSize bytes in one second.
+  rtp_sender_->SetTargetBitrate(kNumPackets * kPacketSize * 8);
+  const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();
+  std::list<uint16_t> sequence_numbers;
+  for (int32_t i = 0; i < kNumPackets; ++i) {
+    sequence_numbers.push_back(kStartSequenceNumber + i);
+    fake_clock_.AdvanceTimeMilliseconds(1);
+    SendPacket(fake_clock_.TimeInMilliseconds(), kPacketSize);
+  }
+  EXPECT_EQ(kNumPackets, transport_.packets_sent_);
+
+  fake_clock_.AdvanceTimeMilliseconds(1000 - kNumPackets);
+
+  // Resending should work - brings the bandwidth up to the limit.
+  // NACK bitrate is capped to the same bitrate as the encoder, since the max
+  // protection overhead is 50% (see MediaOptimization::SetTargetRates).
+  rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
+  EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
+
+  // Resending should not work, bandwidth exceeded.
+  rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
+  EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
 }
 
 // Verify that all packets of a frame have CVO byte set.
@@ -1352,8 +1407,8 @@ TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
       rtp_sender_->RtpHeaderExtensionTotalLength());
 
   rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
-                               kTimestamp, 0, packet_, sizeof(packet_), NULL,
-                               NULL, &hdr);
+                               kTimestamp, 0, packet_, sizeof(packet_), nullptr,
+                               &hdr);
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
@@ -1361,7 +1416,7 @@ TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
-      transport_.sent_packets_[0]->length(), true, &map, kSeqNum, hdr.rotation);
+      transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index c5af226ca6..60e786f0e8 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -10,23 +10,22 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 
-#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include <vector>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 enum { REDForFECHeaderLength = 1 };
@@ -38,20 +37,19 @@ struct RtpPacket {
 
 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
     : _rtpSender(*rtpSender),
+      crit_(CriticalSectionWrapper::CreateCriticalSection()),
       _videoType(kRtpVideoGeneric),
-      _videoCodecInformation(NULL),
       _maxBitrate(0),
       _retransmissionSettings(kRetransmitBaseLayer),
 
       // Generic FEC
-      _fec(),
-      _fecEnabled(false),
-      _payloadTypeRED(-1),
-      _payloadTypeFEC(-1),
-      _numberFirstPartition(0),
+      fec_(),
+      fec_enabled_(false),
+      red_payload_type_(-1),
+      fec_payload_type_(-1),
       delta_fec_params_(),
       key_fec_params_(),
-      producer_fec_(&_fec),
+      producer_fec_(&fec_),
       _fecOverheadRate(clock, NULL),
       _videoBitrate(clock, NULL) {
   memset(&delta_fec_params_, 0, sizeof(delta_fec_params_));
@@ -62,9 +60,6 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
 }
 
 RTPSenderVideo::~RTPSenderVideo() {
-  if (_videoCodecInformation) {
-    delete _videoCodecInformation;
-  }
 }
 
 void RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes videoType) {
@@ -75,11 +70,11 @@ RtpVideoCodecTypes RTPSenderVideo::VideoCodecType() const {
   return _videoType;
 }
 
-int32_t RTPSenderVideo::RegisterVideoPayload(
+// Static.
+RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payloadType,
-    const uint32_t maxBitRate,
-    RtpUtility::Payload*& payload) {
+    const uint32_t maxBitRate) {
   RtpVideoCodecTypes videoType = kRtpVideoGeneric;
   if (RtpUtility::StringCompare(payloadName, "VP8", 3)) {
     videoType = kRtpVideoVp8;
@@ -92,150 +87,122 @@ int32_t RTPSenderVideo::RegisterVideoPayload(
   } else {
     videoType = kRtpVideoGeneric;
   }
-  payload = new RtpUtility::Payload;
+  RtpUtility::Payload* payload = new RtpUtility::Payload();
   payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
   strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
   payload->typeSpecific.Video.videoCodecType = videoType;
   payload->typeSpecific.Video.maxRate = maxBitRate;
   payload->audio = false;
-  return 0;
+  return payload;
 }
 
-int32_t RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
-                                        const size_t payload_length,
-                                        const size_t rtp_header_length,
-                                        const uint32_t capture_timestamp,
-                                        int64_t capture_time_ms,
-                                        StorageType storage,
-                                        bool protect) {
-  if (_fecEnabled) {
-    int ret = 0;
-    size_t fec_overhead_sent = 0;
-    size_t video_sent = 0;
+void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
+                                     const size_t payload_length,
+                                     const size_t rtp_header_length,
+                                     uint16_t seq_num,
+                                     const uint32_t capture_timestamp,
+                                     int64_t capture_time_ms,
+                                     StorageType storage) {
+  if (_rtpSender.SendToNetwork(data_buffer, payload_length, rtp_header_length,
+                               capture_time_ms, storage,
+                               RtpPacketSender::kLowPriority) == 0) {
+    _videoBitrate.Update(payload_length + rtp_header_length);
+    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                         "Video::PacketNormal", "timestamp", capture_timestamp,
+                         "seqnum", seq_num);
+  } else {
+    LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
+  }
+}
 
-    RedPacket* red_packet = producer_fec_.BuildRedPacket(
-        data_buffer, payload_length, rtp_header_length, _payloadTypeRED);
+void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
+                                          const size_t payload_length,
+                                          const size_t rtp_header_length,
+                                          uint16_t media_seq_num,
+                                          const uint32_t capture_timestamp,
+                                          int64_t capture_time_ms,
+                                          StorageType media_packet_storage,
+                                          bool protect) {
+  rtc::scoped_ptr<RedPacket> red_packet;
+  std::vector<RedPacket*> fec_packets;
+  StorageType fec_storage = kDontRetransmit;
+  uint16_t next_fec_sequence_number = 0;
+  {
+    // Only protect while creating RED and FEC packets, not when sending.
+    CriticalSectionScoped cs(crit_.get());
+    red_packet.reset(producer_fec_.BuildRedPacket(
+        data_buffer, payload_length, rtp_header_length, red_payload_type_));
+    if (protect) {
+      producer_fec_.AddRtpPacketAndGenerateFec(data_buffer, payload_length,
+                                               rtp_header_length);
+    }
+    uint16_t num_fec_packets = producer_fec_.NumAvailableFecPackets();
+    if (num_fec_packets > 0) {
+      next_fec_sequence_number =
+          _rtpSender.AllocateSequenceNumber(num_fec_packets);
+      fec_packets = producer_fec_.GetFecPackets(
+          red_payload_type_, fec_payload_type_, next_fec_sequence_number,
+          rtp_header_length);
+      RTC_DCHECK_EQ(num_fec_packets, fec_packets.size());
+      if (_retransmissionSettings & kRetransmitFECPackets)
+        fec_storage = kAllowRetransmission;
+    }
+  }
+  if (_rtpSender.SendToNetwork(
+          red_packet->data(), red_packet->length() - rtp_header_length,
+          rtp_header_length, capture_time_ms, media_packet_storage,
+          RtpPacketSender::kLowPriority) == 0) {
+    _videoBitrate.Update(red_packet->length());
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Video::PacketRed", "timestamp", capture_timestamp,
-                         "seqnum", _rtpSender.SequenceNumber());
-    // Sending the media packet with RED header.
-    int packet_success =
-        _rtpSender.SendToNetwork(red_packet->data(),
-                                 red_packet->length() - rtp_header_length,
-                                 rtp_header_length,
-                                 capture_time_ms,
-                                 storage,
-                                 PacedSender::kNormalPriority);
-
-    ret |= packet_success;
-
-    if (packet_success == 0) {
-      video_sent += red_packet->length();
-    }
-    delete red_packet;
-    red_packet = NULL;
-
-    if (protect) {
-      ret = producer_fec_.AddRtpPacketAndGenerateFec(
-          data_buffer, payload_length, rtp_header_length);
-      if (ret != 0)
-        return ret;
-    }
-
-    while (producer_fec_.FecAvailable()) {
-      red_packet =
-          producer_fec_.GetFecPacket(_payloadTypeRED,
-                                     _payloadTypeFEC,
-                                     _rtpSender.IncrementSequenceNumber(),
-                                     rtp_header_length);
-      StorageType storage = kDontRetransmit;
-      if (_retransmissionSettings & kRetransmitFECPackets) {
-        storage = kAllowRetransmission;
-      }
+                         "seqnum", media_seq_num);
+  } else {
+    LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num;
+  }
+  for (RedPacket* fec_packet : fec_packets) {
+    if (_rtpSender.SendToNetwork(
+            fec_packet->data(), fec_packet->length() - rtp_header_length,
+            rtp_header_length, capture_time_ms, fec_storage,
+            RtpPacketSender::kLowPriority) == 0) {
+      _fecOverheadRate.Update(fec_packet->length());
       TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                            "Video::PacketFec", "timestamp", capture_timestamp,
-                           "seqnum", _rtpSender.SequenceNumber());
-      // Sending FEC packet with RED header.
-      int packet_success =
-          _rtpSender.SendToNetwork(red_packet->data(),
-                                   red_packet->length() - rtp_header_length,
-                                   rtp_header_length,
-                                   capture_time_ms,
-                                   storage,
-                                   PacedSender::kNormalPriority);
-
-      ret |= packet_success;
-
-      if (packet_success == 0) {
-        fec_overhead_sent += red_packet->length();
-      }
-      delete red_packet;
-      red_packet = NULL;
+                           "seqnum", next_fec_sequence_number);
+    } else {
+      LOG(LS_WARNING) << "Failed to send FEC packet "
+                      << next_fec_sequence_number;
     }
-    _videoBitrate.Update(video_sent);
-    _fecOverheadRate.Update(fec_overhead_sent);
-    return ret;
+    delete fec_packet;
+    ++next_fec_sequence_number;
   }
-  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                       "Video::PacketNormal", "timestamp", capture_timestamp,
-                       "seqnum", _rtpSender.SequenceNumber());
-  int ret = _rtpSender.SendToNetwork(data_buffer,
-                                     payload_length,
-                                     rtp_header_length,
-                                     capture_time_ms,
-                                     storage,
-                                     PacedSender::kNormalPriority);
-  if (ret == 0) {
-    _videoBitrate.Update(payload_length + rtp_header_length);
-  }
-  return ret;
 }
 
-int32_t RTPSenderVideo::SendRTPIntraRequest() {
-  // RFC 2032
-  // 5.2.1.  Full intra-frame Request (FIR) packet
-
-  size_t length = 8;
-  uint8_t data[8];
-  data[0] = 0x80;
-  data[1] = 192;
-  data[2] = 0;
-  data[3] = 1;  // length
-
-  ByteWriter<uint32_t>::WriteBigEndian(data + 4, _rtpSender.SSRC());
-
-  TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                       "Video::IntraRequest", "seqnum",
-                       _rtpSender.SequenceNumber());
-  return _rtpSender.SendToNetwork(
-      data, 0, length, -1, kDontStore, PacedSender::kNormalPriority);
-}
-
-int32_t RTPSenderVideo::SetGenericFECStatus(const bool enable,
-                                            const uint8_t payloadTypeRED,
-                                            const uint8_t payloadTypeFEC) {
-  _fecEnabled = enable;
-  _payloadTypeRED = payloadTypeRED;
-  _payloadTypeFEC = payloadTypeFEC;
+void RTPSenderVideo::SetGenericFECStatus(const bool enable,
+                                         const uint8_t payloadTypeRED,
+                                         const uint8_t payloadTypeFEC) {
+  CriticalSectionScoped cs(crit_.get());
+  fec_enabled_ = enable;
+  red_payload_type_ = payloadTypeRED;
+  fec_payload_type_ = payloadTypeFEC;
   memset(&delta_fec_params_, 0, sizeof(delta_fec_params_));
   memset(&key_fec_params_, 0, sizeof(key_fec_params_));
   delta_fec_params_.max_fec_frames = key_fec_params_.max_fec_frames = 1;
   delta_fec_params_.fec_mask_type = key_fec_params_.fec_mask_type =
       kFecMaskRandom;
-  return 0;
 }
 
-int32_t RTPSenderVideo::GenericFECStatus(bool& enable,
-                                         uint8_t& payloadTypeRED,
-                                         uint8_t& payloadTypeFEC) const {
-  enable = _fecEnabled;
-  payloadTypeRED = _payloadTypeRED;
-  payloadTypeFEC = _payloadTypeFEC;
-  return 0;
+void RTPSenderVideo::GenericFECStatus(bool* enable,
+                                      uint8_t* payloadTypeRED,
+                                      uint8_t* payloadTypeFEC) const {
+  CriticalSectionScoped cs(crit_.get());
+  *enable = fec_enabled_;
+  *payloadTypeRED = red_payload_type_;
+  *payloadTypeFEC = fec_payload_type_;
 }
 
 size_t RTPSenderVideo::FECPacketOverhead() const {
-  if (_fecEnabled) {
+  CriticalSectionScoped cs(crit_.get());
+  if (fec_enabled_) {
     // Overhead is FEC headers plus RED for FEC header plus anything in RTP
     // header beyond the 12 bytes base header (CSRC list, extensions...)
     // This reason for the header extensions to be included here is that
@@ -247,14 +214,13 @@ size_t RTPSenderVideo::FECPacketOverhead() const {
   return 0;
 }
 
-int32_t RTPSenderVideo::SetFecParameters(
-    const FecProtectionParams* delta_params,
-    const FecProtectionParams* key_params) {
-  assert(delta_params);
-  assert(key_params);
+void RTPSenderVideo::SetFecParameters(const FecProtectionParams* delta_params,
+                                      const FecProtectionParams* key_params) {
+  CriticalSectionScoped cs(crit_.get());
+  RTC_DCHECK(delta_params);
+  RTC_DCHECK(key_params);
   delta_fec_params_ = *delta_params;
   key_fec_params_ = *key_params;
-  return 0;
 }
 
 int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
@@ -265,49 +231,26 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
                                   const uint8_t* payloadData,
                                   const size_t payloadSize,
                                   const RTPFragmentationHeader* fragmentation,
-                                  VideoCodecInformation* codecInfo,
                                   const RTPVideoHeader* rtpHdr) {
   if (payloadSize == 0) {
     return -1;
   }
 
-  if (frameType == kVideoFrameKey) {
-    producer_fec_.SetFecParameters(&key_fec_params_, _numberFirstPartition);
-  } else {
-    producer_fec_.SetFecParameters(&delta_fec_params_, _numberFirstPartition);
+  rtc::scoped_ptr<RtpPacketizer> packetizer(
+      RtpPacketizer::Create(videoType, _rtpSender.MaxDataPayloadLength(),
+                            &(rtpHdr->codecHeader), frameType));
+
+  StorageType storage;
+  bool fec_enabled;
+  {
+    CriticalSectionScoped cs(crit_.get());
+    FecProtectionParams* fec_params =
+        frameType == kVideoFrameKey ? &key_fec_params_ : &delta_fec_params_;
+    producer_fec_.SetFecParameters(fec_params, 0);
+    storage = packetizer->GetStorageType(_retransmissionSettings);
+    fec_enabled = fec_enabled_;
   }
 
-  // Default setting for number of first partition packets:
-  // Will be extracted in SendVP8 for VP8 codec; other codecs use 0
-  _numberFirstPartition = 0;
-
-  return Send(videoType, frameType, payloadType, captureTimeStamp,
-              capture_time_ms, payloadData, payloadSize, fragmentation, rtpHdr)
-             ? 0
-             : -1;
-}
-
-VideoCodecInformation* RTPSenderVideo::CodecInformationVideo() {
-  return _videoCodecInformation;
-}
-
-void RTPSenderVideo::SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate) {
-  _maxBitrate = maxBitrate;
-}
-
-uint32_t RTPSenderVideo::MaxConfiguredBitrateVideo() const {
-  return _maxBitrate;
-}
-
-bool RTPSenderVideo::Send(const RtpVideoCodecTypes videoType,
-                          const FrameType frameType,
-                          const int8_t payloadType,
-                          const uint32_t captureTimeStamp,
-                          int64_t capture_time_ms,
-                          const uint8_t* payloadData,
-                          const size_t payloadSize,
-                          const RTPFragmentationHeader* fragmentation,
-                          const RTPVideoHeader* rtpHdr) {
   // Register CVO rtp header extension at the first time when we receive a frame
   // with pending rotation.
   RTPSenderInterface::CVOMode cvo_mode = RTPSenderInterface::kCVONone;
@@ -318,10 +261,6 @@ bool RTPSenderVideo::Send(const RtpVideoCodecTypes videoType,
   uint16_t rtp_header_length = _rtpSender.RTPHeaderLength();
   size_t payload_bytes_to_send = payloadSize;
   const uint8_t* data = payloadData;
-  size_t max_payload_length = _rtpSender.MaxDataPayloadLength();
-
-  rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
-      videoType, max_payload_length, &(rtpHdr->codecHeader), frameType));
 
   // TODO(changbin): we currently don't support to configure the codec to
   // output multiple partitions for VP8. Should remove below check after the
@@ -335,16 +274,14 @@ bool RTPSenderVideo::Send(const RtpVideoCodecTypes videoType,
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
-    if (!packetizer->NextPacket(
-            &dataBuffer[rtp_header_length], &payload_bytes_in_packet, &last)) {
-      return false;
+    if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
+                                &payload_bytes_in_packet, &last)) {
+      return -1;
     }
-
     // Write RTP header.
     // Set marker bit true if this is the last packet in frame.
     _rtpSender.BuildRTPheader(
         dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms);
-
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5:
@@ -357,7 +294,7 @@ bool RTPSenderVideo::Send(const RtpVideoCodecTypes videoType,
     // value sent.
     // Here we are adding it to every packet of every frame at this point.
     if (!rtpHdr) {
-      assert(!_rtpSender.IsRtpHeaderExtensionRegistered(
+      RTC_DCHECK(!_rtpSender.IsRtpHeaderExtensionRegistered(
           kRtpExtensionVideoRotation));
     } else if (cvo_mode == RTPSenderInterface::kCVOActivated) {
       // Checking whether CVO header extension is registered will require taking
@@ -368,26 +305,33 @@ bool RTPSenderVideo::Send(const RtpVideoCodecTypes videoType,
       size_t packetSize = payloadSize + rtp_header_length;
       RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
       RTPHeader rtp_header;
-      rtp_parser.Parse(rtp_header);
+      rtp_parser.Parse(&rtp_header);
       _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
                                      rtpHdr->rotation);
     }
-    if (SendVideoPacket(dataBuffer,
-                        payload_bytes_in_packet,
-                        rtp_header_length,
-                        captureTimeStamp,
-                        capture_time_ms,
-                        packetizer->GetStorageType(_retransmissionSettings),
-                        packetizer->GetProtectionType() == kProtectedPacket)) {
-      LOG(LS_WARNING) << packetizer->ToString()
-                      << " failed to send packet number "
-                      << _rtpSender.SequenceNumber();
+    if (fec_enabled) {
+      SendVideoPacketAsRed(dataBuffer, payload_bytes_in_packet,
+                           rtp_header_length, _rtpSender.SequenceNumber(),
+                           captureTimeStamp, capture_time_ms, storage,
+                           packetizer->GetProtectionType() == kProtectedPacket);
+    } else {
+      SendVideoPacket(dataBuffer, payload_bytes_in_packet, rtp_header_length,
+                      _rtpSender.SequenceNumber(), captureTimeStamp,
+                      capture_time_ms, storage);
     }
   }
 
   TRACE_EVENT_ASYNC_END1(
       "webrtc", "Video", capture_time_ms, "timestamp", _rtpSender.Timestamp());
-  return true;
+  return 0;
+}
+
+void RTPSenderVideo::SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate) {
+  _maxBitrate = maxBitrate;
+}
+
+uint32_t RTPSenderVideo::MaxConfiguredBitrateVideo() const {
+  return _maxBitrate;
 }
 
 void RTPSenderVideo::ProcessBitrate() {
@@ -404,12 +348,13 @@ uint32_t RTPSenderVideo::FecOverheadRate() const {
 }
 
 int RTPSenderVideo::SelectiveRetransmissions() const {
+  CriticalSectionScoped cs(crit_.get());
   return _retransmissionSettings;
 }
 
-int RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
+void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
+  CriticalSectionScoped cs(crit_.get());
   _retransmissionSettings = settings;
-  return 0;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 92c312f5e9..e59321ab93 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -13,8 +13,10 @@
 
 #include <list>
 
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
@@ -37,10 +39,10 @@ class RTPSenderVideo {
 
   size_t FECPacketOverhead() const;
 
-  int32_t RegisterVideoPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                               const int8_t payloadType,
-                               const uint32_t maxBitRate,
-                               RtpUtility::Payload*& payload);
+  static RtpUtility::Payload* CreateVideoPayload(
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payloadType,
+      const uint32_t maxBitRate);
 
   int32_t SendVideo(const RtpVideoCodecTypes videoType,
                     const FrameType frameType,
@@ -50,30 +52,27 @@ class RTPSenderVideo {
                     const uint8_t* payloadData,
                     const size_t payloadSize,
                     const RTPFragmentationHeader* fragmentation,
-                    VideoCodecInformation* codecInfo,
                     const RTPVideoHeader* rtpHdr);
 
   int32_t SendRTPIntraRequest();
 
   void SetVideoCodecType(RtpVideoCodecTypes type);
 
-  VideoCodecInformation* CodecInformationVideo();
-
   void SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate);
 
   uint32_t MaxConfiguredBitrateVideo() const;
 
   // FEC
-  int32_t SetGenericFECStatus(const bool enable,
-                              const uint8_t payloadTypeRED,
-                              const uint8_t payloadTypeFEC);
+  void SetGenericFECStatus(const bool enable,
+                           const uint8_t payloadTypeRED,
+                           const uint8_t payloadTypeFEC);
 
-  int32_t GenericFECStatus(bool& enable,
-                           uint8_t& payloadTypeRED,
-                           uint8_t& payloadTypeFEC) const;
+  void GenericFECStatus(bool* enable,
+                        uint8_t* payloadTypeRED,
+                        uint8_t* payloadTypeFEC) const;
 
-  int32_t SetFecParameters(const FecProtectionParams* delta_params,
-                           const FecProtectionParams* key_params);
+  void SetFecParameters(const FecProtectionParams* delta_params,
+                        const FecProtectionParams* key_params);
 
   void ProcessBitrate();
 
@@ -81,45 +80,43 @@ class RTPSenderVideo {
   uint32_t FecOverheadRate() const;
 
   int SelectiveRetransmissions() const;
-  int SetSelectiveRetransmissions(uint8_t settings);
-
- protected:
-  virtual int32_t SendVideoPacket(uint8_t* dataBuffer,
-                                  const size_t payloadLength,
-                                  const size_t rtpHeaderLength,
-                                  const uint32_t capture_timestamp,
-                                  int64_t capture_time_ms,
-                                  StorageType storage,
-                                  bool protect);
+  void SetSelectiveRetransmissions(uint8_t settings);
 
  private:
-  bool Send(const RtpVideoCodecTypes videoType,
-            const FrameType frameType,
-            const int8_t payloadType,
-            const uint32_t captureTimeStamp,
-            int64_t capture_time_ms,
-            const uint8_t* payloadData,
-            const size_t payloadSize,
-            const RTPFragmentationHeader* fragmentation,
-            const RTPVideoHeader* rtpHdr);
+  void SendVideoPacket(uint8_t* dataBuffer,
+                       const size_t payloadLength,
+                       const size_t rtpHeaderLength,
+                       uint16_t seq_num,
+                       const uint32_t capture_timestamp,
+                       int64_t capture_time_ms,
+                       StorageType storage);
+
+  void SendVideoPacketAsRed(uint8_t* dataBuffer,
+                            const size_t payloadLength,
+                            const size_t rtpHeaderLength,
+                            uint16_t video_seq_num,
+                            const uint32_t capture_timestamp,
+                            int64_t capture_time_ms,
+                            StorageType media_packet_storage,
+                            bool protect);
 
- private:
   RTPSenderInterface& _rtpSender;
 
+  // Should never be held when calling out of this class.
+  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+
   RtpVideoCodecTypes _videoType;
-  VideoCodecInformation* _videoCodecInformation;
   uint32_t _maxBitrate;
-  int32_t _retransmissionSettings;
+  int32_t _retransmissionSettings GUARDED_BY(crit_);
 
   // FEC
-  ForwardErrorCorrection _fec;
-  bool _fecEnabled;
-  int8_t _payloadTypeRED;
-  int8_t _payloadTypeFEC;
-  unsigned int _numberFirstPartition;
-  FecProtectionParams delta_fec_params_;
-  FecProtectionParams key_fec_params_;
-  ProducerFec producer_fec_;
+  ForwardErrorCorrection fec_;
+  bool fec_enabled_ GUARDED_BY(crit_);
+  int8_t red_payload_type_ GUARDED_BY(crit_);
+  int8_t fec_payload_type_ GUARDED_BY(crit_);
+  FecProtectionParams delta_fec_params_ GUARDED_BY(crit_);
+  FecProtectionParams key_fec_params_ GUARDED_BY(crit_);
+  ProducerFec producer_fec_ GUARDED_BY(crit_);
 
   // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
   // and any padding overhead.
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index 1c3b060293..c08382e52c 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -10,40 +10,10 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
-#include <assert.h>
-#include <math.h>  // ceil
-#include <string.h>  // memcpy
-
-#if defined(_WIN32)
-// Order for these headers are important
-#include <Windows.h>  // FILETIME
-
-#include <WinSock.h>  // timeval
-
-#include <MMSystem.h>  // timeGetTime
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_BSD) || (defined WEBRTC_MAC))
-#include <sys/time.h>  // gettimeofday
-#include <time.h>
-#endif
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#include <stdio.h>
-#endif
+#include <string.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#define DEBUG_PRINT(...)           \
-  {                                \
-    char msg[256];                 \
-    sprintf(msg, __VA_ARGS__);     \
-    OutputDebugString(msg);        \
-  }
-#else
-// special fix for visual 2003
-#define DEBUG_PRINT(exp)        ((void)0)
-#endif  // defined(_DEBUG) && defined(_WIN32)
 
 namespace webrtc {
 
@@ -78,38 +48,6 @@ enum {
   kRtpMinParseLength = 12
 };
 
-/*
- * Time routines.
- */
-
-uint32_t GetCurrentRTP(Clock* clock, uint32_t freq) {
-  const bool use_global_clock = (clock == NULL);
-  Clock* local_clock = clock;
-  if (use_global_clock) {
-    local_clock = Clock::GetRealTimeClock();
-  }
-  uint32_t secs = 0, frac = 0;
-  local_clock->CurrentNtp(secs, frac);
-  if (use_global_clock) {
-    delete local_clock;
-  }
-  return ConvertNTPTimeToRTP(secs, frac, freq);
-}
-
-uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec, uint32_t NTPfrac, uint32_t freq) {
-  float ftemp = (float)NTPfrac / (float)NTP_FRAC;
-  uint32_t tmp = (uint32_t)(ftemp * freq);
- return NTPsec * freq + tmp;
-}
-
-uint32_t ConvertNTPTimeToMS(uint32_t NTPsec, uint32_t NTPfrac) {
-  int freq = 1000;
-  float ftemp = (float)NTPfrac / (float)NTP_FRAC;
-  uint32_t tmp = (uint32_t)(ftemp * freq);
-  uint32_t MStime = NTPsec * freq + tmp;
-  return MStime;
-}
-
 /*
  * Misc utility routines
  */
@@ -117,12 +55,12 @@ uint32_t ConvertNTPTimeToMS(uint32_t NTPsec, uint32_t NTPfrac) {
 #if defined(_WIN32)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
-  return (_strnicmp(str1, str2, length) == 0) ? true : false;
+  return _strnicmp(str1, str2, length) == 0;
 }
 #elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
-  return (strncasecmp(str1, str2, length) == 0) ? true : false;
+  return strncasecmp(str1, str2, length) == 0;
 }
 #endif
 
@@ -133,10 +71,6 @@ size_t Word32Align(size_t size) {
   return size;
 }
 
-uint32_t pow2(uint8_t exp) {
-  return 1 << exp;
-}
-
 RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData,
                                  const size_t rtpDataLength)
     : _ptrRTPDataBegin(rtpData),
@@ -249,7 +183,7 @@ bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const {
   return true;
 }
 
-bool RtpHeaderParser::Parse(RTPHeader& header,
+bool RtpHeaderParser::Parse(RTPHeader* header,
                             RtpHeaderExtensionMap* ptrExtensionMap) const {
   const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
   if (length < kRtpMinParseLength) {
@@ -282,49 +216,50 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
     return false;
   }
 
-  header.markerBit      = M;
-  header.payloadType    = PT;
-  header.sequenceNumber = sequenceNumber;
-  header.timestamp      = RTPTimestamp;
-  header.ssrc           = SSRC;
-  header.numCSRCs       = CC;
-  header.paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
+  header->markerBit      = M;
+  header->payloadType    = PT;
+  header->sequenceNumber = sequenceNumber;
+  header->timestamp      = RTPTimestamp;
+  header->ssrc           = SSRC;
+  header->numCSRCs       = CC;
+  header->paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
 
   // 12 == sizeof(RFC rtp header) == kRtpMinParseLength, each CSRC=4 bytes
-  header.headerLength   = 12 + (CC * 4);
+  header->headerLength   = 12 + (CC * 4);
   // not a full validation, just safety against underflow.  Padding must
   // start after the header.  We can have 0 payload bytes left, note.
-  if (header.paddingLength + header.headerLength > (size_t) length) {
+  if (header->paddingLength + header->headerLength > (size_t) length) {
     return false;
   }
 
   for (uint8_t i = 0; i < CC; ++i) {
     uint32_t CSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
     ptr += 4;
-    header.arrOfCSRCs[i] = CSRC;
+    header->arrOfCSRCs[i] = CSRC;
   }
-  assert((ptr - _ptrRTPDataBegin) == (ptrdiff_t) header.headerLength);
+  assert((ptr - _ptrRTPDataBegin) == (ptrdiff_t) header->headerLength);
 
   // If in effect, MAY be omitted for those packets for which the offset
   // is zero.
-  header.extension.hasTransmissionTimeOffset = false;
-  header.extension.transmissionTimeOffset = 0;
+  header->extension.hasTransmissionTimeOffset = false;
+  header->extension.transmissionTimeOffset = 0;
 
   // May not be present in packet.
-  header.extension.hasAbsoluteSendTime = false;
-  header.extension.absoluteSendTime = 0;
+  header->extension.hasAbsoluteSendTime = false;
+  header->extension.absoluteSendTime = 0;
 
   // May not be present in packet.
-  header.extension.hasAudioLevel = false;
-  header.extension.audioLevel = 0;
+  header->extension.hasAudioLevel = false;
+  header->extension.voiceActivity = false;
+  header->extension.audioLevel = 0;
 
   // May not be present in packet.
-  header.extension.hasVideoRotation = false;
-  header.extension.videoRotation = 0;
+  header->extension.hasVideoRotation = false;
+  header->extension.videoRotation = 0;
 
   // May not be present in packet.
-  header.extension.hasRID = false;
-  header.extension.rid = NULL;
+  header->extension.hasRID = false;
+  header->extension.rid = NULL;
 
   if (X) {
     /* RTP header extension, RFC 3550.
@@ -337,12 +272,12 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
     |                             ....                              |
     */
     // earlier test ensures we have at least paddingLength bytes left
-    const ptrdiff_t remain = (_ptrRTPDataEnd - ptr) - header.paddingLength;
+    const ptrdiff_t remain = (_ptrRTPDataEnd - ptr) - header->paddingLength;
     if (remain < 4) { // minimum header extension length = 32 bits
       return false;
     }
 
-    header.headerLength += 4;
+    header->headerLength += 4;
 
     uint16_t definedByProfile = ByteReader<uint16_t>::ReadBigEndian(ptr);
     ptr += 2;
@@ -362,13 +297,16 @@ bool RtpHeaderParser::Parse(RTPHeader& header,
                                   ptrRTPDataExtensionEnd,
                                   ptr);
     }
-    header.headerLength += XLen;
+    header->headerLength += XLen;
   }
+  if (header->headerLength + header->paddingLength >
+      static_cast<size_t>(length))
+    return false;
   return true;
 }
 
 void RtpHeaderParser::ParseOneByteExtensionHeader(
-    RTPHeader& header,
+    RTPHeader* header,
     const RtpHeaderExtensionMap* ptrExtensionMap,
     const uint8_t* ptrRTPDataExtensionEnd,
     const uint8_t* ptr) const {
@@ -385,8 +323,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
 
     // Note that 'len' is the header extension element length, which is the
     // number of bytes - 1.
-    const uint8_t id = (*ptr & 0xf0) >> 4;
-    const uint8_t len = (*ptr & 0x0f);
+    const int id = (*ptr & 0xf0) >> 4;
+    const int len = (*ptr & 0x0f);
     if (ptr + len + 1 > ptrRTPDataExtensionEnd) {
       LOG(LS_WARNING)
           << "RTP extension header length out of bounds. Terminate parsing.";
@@ -403,8 +341,7 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
     RTPExtensionType type;
     if (ptrExtensionMap->GetType(id, &type) != 0) {
       // If we encounter an unknown extension, just skip over it.
-      LOG(LS_WARNING) << "Failed to find extension id: "
-                      << static_cast<int>(id);
+      LOG(LS_INFO) << "Failed to find extension id: " << id;
     } else {
       switch (type) {
         case kRtpExtensionTransmissionTimeOffset: {
@@ -419,9 +356,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=2 |              transmission offset              |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-          header.extension.transmissionTimeOffset =
+          header->extension.transmissionTimeOffset =
               ByteReader<int32_t, 3>::ReadBigEndian(ptr);
-          header.extension.hasTransmissionTimeOffset = true;
+          header->extension.hasTransmissionTimeOffset = true;
           break;
         }
         case kRtpExtensionAudioLevel: {
@@ -435,15 +372,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=0 |V|   level     |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           //
-
-          // Parse out the fields but only use it for debugging for now.
-          // const uint8_t V = (*ptr & 0x80) >> 7;
-          // const uint8_t level = (*ptr & 0x7f);
-          // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
-          // level=%u", ID, len, V, level);
-
-          header.extension.audioLevel = ptr[0];
-          header.extension.hasAudioLevel = true;
+          header->extension.audioLevel = ptr[0] & 0x7f;
+          header->extension.voiceActivity = (ptr[0] & 0x80) != 0;
+          header->extension.hasAudioLevel = true;
           break;
         }
         case kRtpExtensionAbsoluteSendTime: {
@@ -457,9 +388,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // |  ID   | len=2 |              absolute send time               |
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
-          header.extension.absoluteSendTime =
+          header->extension.absoluteSendTime =
               ByteReader<uint32_t, 3>::ReadBigEndian(ptr);
-          header.extension.hasAbsoluteSendTime = true;
+          header->extension.hasAbsoluteSendTime = true;
           break;
         }
         case kRtpExtensionVideoRotation: {
@@ -473,14 +404,14 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           // |  ID   | len=0 |0 0 0 0 C F R R|
           // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-          header.extension.hasVideoRotation = true;
-          header.extension.videoRotation = ptr[0];
+          header->extension.hasVideoRotation = true;
+          header->extension.videoRotation = ptr[0];
           break;
         }
         case kRtpExtensionTransportSequenceNumber: {
           if (len != 1) {
-            LOG(LS_WARNING)
-                << "Incorrect peer connection sequence number len: " << len;
+            LOG(LS_WARNING) << "Incorrect transport sequence number len: "
+                            << len;
             return;
           }
           //   0                   1                   2
@@ -491,8 +422,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
 
           uint16_t sequence_number = ptr[0] << 8;
           sequence_number += ptr[1];
-          header.extension.transportSequenceNumber = sequence_number;
-          header.extension.hasTransportSequenceNumber = true;
+          header->extension.transportSequenceNumber = sequence_number;
+          header->extension.hasTransportSequenceNumber = true;
           break;
         }
         case kRtpExtensionRtpStreamId: {
@@ -506,8 +437,8 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
           char* ptrRID = new char[len+1];
           memcpy(ptrRID, ptr, len);
           ptrRID[len] = '\0';
-          header.extension.rid = ptrRID;
-          header.extension.hasRID = true;
+          header->extension.rid = ptrRID;
+          header->extension.hasRID = true;
           break;
         }
         default: {
@@ -536,5 +467,4 @@ uint8_t RtpHeaderParser::ParsePaddingBytes(
   return num_zero_bytes;
 }
 }  // namespace RtpUtility
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index af20f97e82..23c175356a 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -11,10 +11,11 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
 
-#include <stddef.h> // size_t, ptrdiff_t
+#include <map>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/base/deprecation.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
@@ -29,71 +30,48 @@ RtpAudioFeedback* NullObjectRtpAudioFeedback();
 ReceiveStatistics* NullObjectReceiveStatistics();
 
 namespace RtpUtility {
-    // January 1970, in NTP seconds.
-    const uint32_t NTP_JAN_1970 = 2208988800UL;
 
-    // Magic NTP fractional unit.
-    const double NTP_FRAC = 4.294967296E+9;
+struct Payload {
+  char name[RTP_PAYLOAD_NAME_SIZE];
+  bool audio;
+  PayloadUnion typeSpecific;
+};
 
-    struct Payload
-    {
-        char name[RTP_PAYLOAD_NAME_SIZE];
-        bool audio;
-        PayloadUnion typeSpecific;
-    };
+typedef std::map<int8_t, Payload*> PayloadTypeMap;
 
-    typedef std::map<int8_t, Payload*> PayloadTypeMap;
+bool StringCompare(const char* str1, const char* str2, const uint32_t length);
 
-    // Return the current RTP timestamp from the NTP timestamp
-    // returned by the specified clock.
-    uint32_t GetCurrentRTP(Clock* clock, uint32_t freq);
+// Round up to the nearest size that is a multiple of 4.
+size_t Word32Align(size_t size);
 
-    // Return the current RTP absolute timestamp.
-    uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec,
-                                 uint32_t NTPfrac,
-                                 uint32_t freq);
+class RtpHeaderParser {
+ public:
+  RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
+  ~RtpHeaderParser();
 
-    uint32_t pow2(uint8_t exp);
+  bool RTCP() const;
+  bool ParseRtcp(RTPHeader* header) const;
+  bool Parse(RTPHeader* parsedPacket,
+             RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
+  RTC_DEPRECATED bool Parse(
+      RTPHeader& parsedPacket,  // NOLINT(runtime/references)
+      RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const {
+    return Parse(&parsedPacket, ptrExtensionMap);
+  }
 
-    // Returns true if |newTimestamp| is older than |existingTimestamp|.
-    // |wrapped| will be set to true if there has been a wraparound between the
-    // two timestamps.
-    bool OldTimestamp(uint32_t newTimestamp,
-                      uint32_t existingTimestamp,
-                      bool* wrapped);
+ private:
+  void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
+                                   const RtpHeaderExtensionMap* ptrExtensionMap,
+                                   const uint8_t* ptrRTPDataExtensionEnd,
+                                   const uint8_t* ptr) const;
 
-    bool StringCompare(const char* str1,
-                       const char* str2,
-                       const uint32_t length);
+  uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
+                            const uint8_t* ptr) const;
 
-    // Round up to the nearest size that is a multiple of 4.
-    size_t Word32Align(size_t size);
-
-    class RtpHeaderParser {
-    public:
-     RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
-     ~RtpHeaderParser();
-
-        bool RTCP() const;
-        bool ParseRtcp(RTPHeader* header) const;
-        bool Parse(RTPHeader& parsedPacket,
-                   RtpHeaderExtensionMap* ptrExtensionMap = NULL) const;
-
-    private:
-        void ParseOneByteExtensionHeader(
-            RTPHeader& parsedPacket,
-            const RtpHeaderExtensionMap* ptrExtensionMap,
-            const uint8_t* ptrRTPDataExtensionEnd,
-            const uint8_t* ptr) const;
-
-        uint8_t ParsePaddingBytes(
-            const uint8_t* ptrRTPDataExtensionEnd,
-            const uint8_t* ptr) const;
-
-        const uint8_t* const _ptrRTPDataBegin;
-        const uint8_t* const _ptrRTPDataEnd;
-    };
+  const uint8_t* const _ptrRTPDataBegin;
+  const uint8_t* const _ptrRTPDataEnd;
+};
 }  // namespace RtpUtility
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
index 4e23083385..fb02b7ef12 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
@@ -10,110 +10,51 @@
 
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 
-#include <assert.h>
-#include <stdlib.h>
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-
-#ifdef _WIN32
-    #include <windows.h>
-    #include <MMSystem.h> //timeGetTime
-
-// TODO(hellner): investigate if it is necessary to disable these warnings.
-    #pragma warning(disable:4311)
-    #pragma warning(disable:4312)
-#else
-    #include <stdio.h>
-    #include <string.h>
-    #include <time.h>
-    #include <sys/time.h>
-#endif
+#include "webrtc/base/checks.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
-SSRCDatabase*
-SSRCDatabase::StaticInstance(CountOperation count_operation)
-{
-  SSRCDatabase* impl =
-      GetStaticInstance<SSRCDatabase>(count_operation);
-  return impl;
+namespace {
+uint64_t Seed() {
+  return Clock::GetRealTimeClock()->TimeInMicroseconds();
+}
+}  // namespace
+
+SSRCDatabase* SSRCDatabase::GetSSRCDatabase() {
+  return GetStaticInstance<SSRCDatabase>(kAddRef);
 }
 
-SSRCDatabase*
-SSRCDatabase::GetSSRCDatabase()
-{
-    return StaticInstance(kAddRef);
+void SSRCDatabase::ReturnSSRCDatabase() {
+  GetStaticInstance<SSRCDatabase>(kRelease);
 }
 
-void
-SSRCDatabase::ReturnSSRCDatabase()
-{
-    StaticInstance(kRelease);
-}
+uint32_t SSRCDatabase::CreateSSRC() {
+  CriticalSectionScoped lock(crit_.get());
 
-uint32_t
-SSRCDatabase::CreateSSRC()
-{
-    CriticalSectionScoped lock(_critSect);
-
-    uint32_t ssrc = GenerateRandom();
-
-    while(_ssrcMap.find(ssrc) != _ssrcMap.end())
-    {
-        ssrc = GenerateRandom();
+  while (true) {  // Try until get a new ssrc.
+    // 0 and 0xffffffff are invalid values for SSRC.
+    uint32_t ssrc = random_.Rand(1u, 0xfffffffe);
+    if (ssrcs_.insert(ssrc).second) {
+      return ssrc;
     }
-    _ssrcMap[ssrc] = 0;
-
-    return ssrc;
+  }
 }
 
-int32_t
-SSRCDatabase::RegisterSSRC(const uint32_t ssrc)
-{
-    CriticalSectionScoped lock(_critSect);
-    _ssrcMap[ssrc] = 0;
-    return 0;
+void SSRCDatabase::RegisterSSRC(uint32_t ssrc) {
+  CriticalSectionScoped lock(crit_.get());
+  ssrcs_.insert(ssrc);
 }
 
-int32_t
-SSRCDatabase::ReturnSSRC(const uint32_t ssrc)
-{
-    CriticalSectionScoped lock(_critSect);
-    _ssrcMap.erase(ssrc);
-    return 0;
+void SSRCDatabase::ReturnSSRC(uint32_t ssrc) {
+  CriticalSectionScoped lock(crit_.get());
+  ssrcs_.erase(ssrc);
 }
 
 SSRCDatabase::SSRCDatabase()
-{
-    // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :)
-#ifdef _WIN32
-    srand(timeGetTime());
-#else
-    struct timeval tv;
-    struct timezone tz;
-    gettimeofday(&tv, &tz);
-    srand(tv.tv_usec);
-#endif
+    : crit_(CriticalSectionWrapper::CreateCriticalSection()), random_(Seed()) {}
 
-    _critSect = CriticalSectionWrapper::CreateCriticalSection();
+SSRCDatabase::~SSRCDatabase() {
 }
 
-SSRCDatabase::~SSRCDatabase()
-{
-    _ssrcMap.clear();
-    delete _critSect;
-}
-
-uint32_t SSRCDatabase::GenerateRandom()
-{
-    uint32_t ssrc = 0;
-    do
-    {
-        ssrc = rand();
-        ssrc = ssrc <<16;
-        ssrc += rand();
-
-    } while (ssrc == 0 || ssrc == 0xffffffff);
-
-    return ssrc;
-}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.h
index e95b8324d6..4fa68c1d4a 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ssrc_database.h
@@ -11,43 +11,41 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
 
-#include <map>
+#include <set>
 
-#include "webrtc/system_wrappers/interface/static_instance.h"
+#include "webrtc/base/random.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/static_instance.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
 
-class SSRCDatabase
-{
-public:
-    static SSRCDatabase* GetSSRCDatabase();
-    static void ReturnSSRCDatabase();
+class SSRCDatabase {
+ public:
+  static SSRCDatabase* GetSSRCDatabase();
+  static void ReturnSSRCDatabase();
 
-    uint32_t CreateSSRC();
-    int32_t RegisterSSRC(const uint32_t ssrc);
-    int32_t ReturnSSRC(const uint32_t ssrc);
+  uint32_t CreateSSRC();
+  void RegisterSSRC(uint32_t ssrc);
+  void ReturnSSRC(uint32_t ssrc);
 
-    SSRCDatabase();
-    virtual ~SSRCDatabase();
+  SSRCDatabase();
+  virtual ~SSRCDatabase();
 
-protected:
-    static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
+ protected:
+  static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
 
-private:
-    // Friend function to allow the SSRC destructor to be accessed from the
-    // template class.
-    friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
-        CountOperation count_operation);
-    static SSRCDatabase* StaticInstance(CountOperation count_operation);
+ private:
+  // Friend function to allow the SSRC destructor to be accessed from the
+  // template class.
+  friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
+      CountOperation count_operation);
 
-    uint32_t GenerateRandom();
-
-    std::map<uint32_t, uint32_t>    _ssrcMap;
-
-    CriticalSectionWrapper* _critSect;
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  Random random_ GUARDED_BY(crit_);
+  std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/time_util.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/time_util.h
new file mode 100644
index 0000000000..5b544ddf9a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/time_util.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+// Converts NTP timestamp to RTP timestamp.
+inline uint32_t NtpToRtp(NtpTime ntp, uint32_t freq) {
+  uint32_t tmp = (static_cast<uint64_t>(ntp.fractions()) * freq) >> 32;
+  return ntp.seconds() * freq + tmp;
+}
+// Return the current RTP timestamp from the NTP timestamp
+// returned by the specified clock.
+inline uint32_t CurrentRtp(const Clock& clock, uint32_t freq) {
+  return NtpToRtp(NtpTime(clock), freq);
+}
+
+// Helper function for compact ntp representation:
+// RFC 3550, Section 4. Time Format.
+// Wallclock time is represented using the timestamp format of
+// the Network Time Protocol (NTP).
+// ...
+// In some fields where a more compact representation is
+// appropriate, only the middle 32 bits are used; that is, the low 16
+// bits of the integer part and the high 16 bits of the fractional part.
+inline uint32_t CompactNtp(NtpTime ntp) {
+  return (ntp.seconds() << 16) | (ntp.fractions() >> 16);
+}
+// Converts interval between compact ntp timestamps to milliseconds.
+// This interval can be upto ~18.2 hours (2^16 seconds).
+inline uint32_t CompactNtpIntervalToMs(uint32_t compact_ntp_interval) {
+  return static_cast<uint64_t>(compact_ntp_interval) * 1000 / (1 << 16);
+}
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
new file mode 100644
index 0000000000..7efb83ccad
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+TEST(TimeUtilTest, CompactNtp) {
+  const uint32_t kNtpSec = 0x12345678;
+  const uint32_t kNtpFrac = 0x23456789;
+  const NtpTime kNtp(kNtpSec, kNtpFrac);
+  const uint32_t kNtpMid = 0x56782345;
+  EXPECT_EQ(kNtpMid, CompactNtp(kNtp));
+}
+
+TEST(TimeUtilTest, CompactNtpToMs) {
+  const NtpTime ntp1(0x12345, 0x23456);
+  const NtpTime ntp2(0x12654, 0x64335);
+  uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+
+  uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+  EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpToMsWithWrap) {
+  const NtpTime ntp1(0x1ffff, 0x23456);
+  const NtpTime ntp2(0x20000, 0x64335);
+  uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+
+  // While ntp2 > ntp1, there compact ntp presentation happen to be opposite.
+  // That shouldn't be a problem as long as unsigned arithmetic is used.
+  ASSERT_GT(ntp2.ToMs(), ntp1.ToMs());
+  ASSERT_LT(CompactNtp(ntp2), CompactNtp(ntp1));
+
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+  EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpToMsLarge) {
+  const NtpTime ntp1(0x10000, 0x23456);
+  const NtpTime ntp2(0x1ffff, 0x64335);
+  uint32_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  // Ntp difference close to maximum of ~18 hours should convert correctly too.
+  ASSERT_GT(ms_diff, 18u * 3600 * 1000);
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  uint32_t ntp_to_ms_diff = CompactNtpIntervalToMs(ntp_diff);
+
+  EXPECT_NEAR(ms_diff, ntp_to_ms_diff, 1);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
index fb1ed625ed..f994ff7049 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -11,8 +11,10 @@
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
 
 #include <assert.h>
-#include <limits>
 #include <string.h>
+
+#include <limits>
+
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
index 5d44384b2f..b046e28a73 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/video_codec_information.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/video_codec_information.h
index 456b3bb934..7b819d060f 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/video_codec_information.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/video_codec_information.h
@@ -15,14 +15,13 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
-class VideoCodecInformation
-{
-public:
-    virtual void Reset() = 0;
+class VideoCodecInformation {
+ public:
+  virtual void Reset() = 0;
 
-    virtual RtpVideoCodecTypes Type() = 0;
-    virtual ~VideoCodecInformation(){};
+  virtual RtpVideoCodecTypes Type() = 0;
+  virtual ~VideoCodecInformation() {}
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
index feed784839..9721a7e9ac 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
@@ -37,9 +37,8 @@ PartitionTreeNode::PartitionTreeNode(PartitionTreeNode* parent,
 
 PartitionTreeNode* PartitionTreeNode::CreateRootNode(const size_t* size_vector,
                                                      size_t num_partitions) {
-  PartitionTreeNode* root_node =
-      new PartitionTreeNode(NULL, &size_vector[1], num_partitions - 1,
-                            size_vector[0]);
+  PartitionTreeNode* root_node = new PartitionTreeNode(
+      NULL, &size_vector[1], num_partitions - 1, size_vector[0]);
   root_node->set_packet_start(true);
   return root_node;
 }
@@ -54,7 +53,7 @@ int PartitionTreeNode::Cost(size_t penalty) {
   if (num_partitions_ == 0) {
     // This is a solution node.
     cost = std::max(max_parent_size_, this_size_int()) -
-        std::min(min_parent_size_, this_size_int());
+           std::min(min_parent_size_, this_size_int());
   } else {
     cost = std::max(max_parent_size_, this_size_int()) - min_parent_size_;
   }
@@ -68,9 +67,7 @@ bool PartitionTreeNode::CreateChildren(size_t max_size) {
     if (this_size_ + size_vector_[0] <= max_size) {
       assert(!children_[kLeftChild]);
       children_[kLeftChild] =
-          new PartitionTreeNode(this,
-                                &size_vector_[1],
-                                num_partitions_ - 1,
+          new PartitionTreeNode(this, &size_vector_[1], num_partitions_ - 1,
                                 this_size_ + size_vector_[0]);
       children_[kLeftChild]->set_max_parent_size(max_parent_size_);
       children_[kLeftChild]->set_min_parent_size(min_parent_size_);
@@ -80,10 +77,8 @@ bool PartitionTreeNode::CreateChildren(size_t max_size) {
     }
     if (this_size_ > 0) {
       assert(!children_[kRightChild]);
-      children_[kRightChild] = new PartitionTreeNode(this,
-                                                     &size_vector_[1],
-                                                     num_partitions_ - 1,
-                                                     size_vector_[0]);
+      children_[kRightChild] = new PartitionTreeNode(
+          this, &size_vector_[1], num_partitions_ - 1, size_vector_[0]);
       children_[kRightChild]->set_max_parent_size(
           std::max(max_parent_size_, this_size_int()));
       children_[kRightChild]->set_min_parent_size(
@@ -148,7 +143,8 @@ PartitionTreeNode* PartitionTreeNode::GetOptimalNode(size_t max_size,
 
 Vp8PartitionAggregator::Vp8PartitionAggregator(
     const RTPFragmentationHeader& fragmentation,
-    size_t first_partition_idx, size_t last_partition_idx)
+    size_t first_partition_idx,
+    size_t last_partition_idx)
     : root_(NULL),
       num_partitions_(last_partition_idx - first_partition_idx + 1),
       size_vector_(new size_t[num_partitions_]),
@@ -158,14 +154,14 @@ Vp8PartitionAggregator::Vp8PartitionAggregator(
   for (size_t i = 0; i < num_partitions_; ++i) {
     size_vector_[i] =
         fragmentation.fragmentationLength[i + first_partition_idx];
-    largest_partition_size_ = std::max(largest_partition_size_,
-                                       size_vector_[i]);
+    largest_partition_size_ =
+        std::max(largest_partition_size_, size_vector_[i]);
   }
   root_ = PartitionTreeNode::CreateRootNode(size_vector_, num_partitions_);
 }
 
 Vp8PartitionAggregator::~Vp8PartitionAggregator() {
-  delete [] size_vector_;
+  delete[] size_vector_;
   delete root_;
 }
 
@@ -190,14 +186,16 @@ Vp8PartitionAggregator::FindOptimalConfiguration(size_t max_size,
     assert(packet_index > 0);
     assert(temp_node != NULL);
     config_vector[i - 1] = packet_index - 1;
-    if (temp_node->packet_start()) --packet_index;
+    if (temp_node->packet_start())
+      --packet_index;
     temp_node = temp_node->parent();
   }
   return config_vector;
 }
 
 void Vp8PartitionAggregator::CalcMinMax(const ConfigVec& config,
-                                        int* min_size, int* max_size) const {
+                                        int* min_size,
+                                        int* max_size) const {
   if (*min_size < 0) {
     *min_size = std::numeric_limits<int>::max();
   }
@@ -263,8 +261,8 @@ size_t Vp8PartitionAggregator::CalcNumberOfFragments(
   }
   assert(num_fragments > 0);
   // TODO(mflodman) Assert disabled since it's falsely triggered, see issue 293.
-  //assert(large_partition_size / num_fragments + 1 <= max_payload_size);
+  // assert(large_partition_size / num_fragments + 1 <= max_payload_size);
   return num_fragments;
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
index 67babcb330..ccd22e5be2 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
@@ -14,7 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -78,7 +78,7 @@ class PartitionTreeNode {
   int min_parent_size_;
   bool packet_start_;
 
-  DISALLOW_COPY_AND_ASSIGN(PartitionTreeNode);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PartitionTreeNode);
 };
 
 // Class that calculates the optimal aggregation of VP8 partitions smaller than
@@ -130,8 +130,8 @@ class Vp8PartitionAggregator {
   size_t* size_vector_;
   size_t largest_partition_size_;
 
-  DISALLOW_COPY_AND_ASSIGN(Vp8PartitionAggregator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Vp8PartitionAggregator);
 };
-}  // namespace
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
index 4650c94047..726d83ec50 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
@@ -209,4 +209,4 @@ TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
                 1600, kMTU, 1, 900, 1000));
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
deleted file mode 100644
index ea8ac9876a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// BWEStandAlone.cpp : Defines the entry point for the console application.
-//
-
-#include <stdio.h>
-#include <string>
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
-
-//#include "vld.h"
-
-class myTransportCB: public UdpTransportData
-{
-public:
-    myTransportCB (RtpRtcp *rtpMod) : _rtpMod(rtpMod) {};
-protected:
-    // Inherited from UdpTransportData
- void IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                        const size_t rtpPacketLength,
-                        const int8_t* fromIP,
-                        const uint16_t fromPort) override;
-
- void IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                         const size_t rtcpPacketLength,
-                         const int8_t* fromIP,
-                         const uint16_t fromPort) override;
-
-private:
-    RtpRtcp *_rtpMod;
-};
-
-void myTransportCB::IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                                      const size_t rtpPacketLength,
-                                      const int8_t* fromIP,
-                                      const uint16_t fromPort)
-{
-    printf("Receiving RTP from IP %s, port %u\n", fromIP, fromPort);
-    _rtpMod->IncomingPacket((uint8_t *) incomingRtpPacket, rtpPacketLength);
-}
-
-void myTransportCB::IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                                       const size_t rtcpPacketLength,
-                                       const int8_t* fromIP,
-                                       const uint16_t fromPort)
-{
-    printf("Receiving RTCP from IP %s, port %u\n", fromIP, fromPort);
-    _rtpMod->IncomingPacket((uint8_t *) incomingRtcpPacket, rtcpPacketLength);
-}
-
-
-int main(int argc, char* argv[])
-{
-    bool isSender = false;
-    bool isReceiver = false;
-    uint16_t port;
-    std::string ip;
-    TestSenderReceiver *sendrec = new TestSenderReceiver();
-    TestLoadGenerator *gen;
-
-    if (argc == 2)
-    {
-        // receiver only
-        isReceiver = true;
-
-        // read port
-        port = atoi(argv[1]);
-    }
-    else if (argc == 3)
-    {
-        // sender and receiver
-        isSender = true;
-        isReceiver = true;
-
-        // read IP
-        ip = argv[1];
-
-        // read port
-        port = atoi(argv[2]);
-    }
-
-    Trace::CreateTrace();
-    Trace::SetTraceFile("BWEStandAloneTrace.txt");
-    Trace::set_level_filter(webrtc::kTraceAll);
-
-    sendrec->InitReceiver(port);
-
-    sendrec->Start();
-
-    if (isSender)
-    {
-        const uint32_t startRateKbps = 1000;
-        //gen = new CBRGenerator(sendrec, 1000, 500);
-        gen = new CBRFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2);
-        //gen = new PeriodicKeyFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2, 7, 300);
-        //const uint16_t numFrameRates = 5;
-        //const uint8_t frameRates[numFrameRates] = {30, 15, 20, 23, 25};
-        //gen = new CBRVarFRGenerator(sendrec, 1000, frameRates, numFrameRates, 90000, 4.0, 0.1, 0.2);
-        //gen = new CBRFrameDropGenerator(sendrec, startRateKbps, 90000, 0.2);
-        sendrec->SetLoadGenerator(gen);
-        sendrec->InitSender(startRateKbps, ip.c_str(), port);
-        gen->Start();
-    }
-
-    while (1)
-    {
-    }
-
-    if (isSender)
-    {
-        gen->Stop();
-        delete gen;
-    }
-
-    delete sendrec;
-
-    //uint8_t numberOfSocketThreads = 1;
-    //UdpTransport* transport = UdpTransport::Create(0, numberOfSocketThreads);
-
-    //RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(1, false);
-    //if (rtp->InitSender() != 0)
-    //{
-    //    exit(1);
-    //}
-    //if (rtp->RegisterSendTransport(transport) != 0)
-    //{
-    //    exit(1);
-    //}
-
-//    transport->InitializeSendSockets("192.168.200.39", 8000);
-    //transport->InitializeSendSockets("127.0.0.1", 10000);
-    //transport->InitializeSourcePorts(8000);
-
-
-    return(0);
- //   myTransportCB *tp = new myTransportCB(rtp);
- //   transport->InitializeReceiveSockets(tp, 10000, "0.0.0.0");
- //   transport->StartReceiving(500);
-
- //   int8_t data[100];
- //   for (int i = 0; i < 100; data[i] = i++);
-
- //   for (int i = 0; i < 100; i++)
- //   {
- //       transport->SendRaw(data, 100, false);
- //   }
-
-
-
- //   int32_t totTime = 0;
- //   while (totTime < 10000)
- //   {
- //       transport->Process();
- //       int32_t wTime = transport->TimeUntilNextProcess();
- //       totTime += wTime;
- //       Sleep(wTime);
- //   }
-
-
-    //if (transport)
-    //{
-    //    // Destroy the Socket Transport module
-    //    transport->StopReceiving();
-    //    transport->InitializeReceiveSockets(NULL,0);// deregister callback
- //       UdpTransport::Destroy(transport);
-    //    transport = NULL;
- //   }
-
- //   if (tp)
- //   {
- //       delete tp;
- //       tp = NULL;
- //   }
-
- //   if (rtp)
- //   {
- //       RtpRtcp::DestroyRtpRtcp(rtp);
- //       rtp = NULL;
- //   }
-
-
-    //return 0;
-}
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
deleted file mode 100644
index 9e79a8cda8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
+++ /dev/null
@@ -1,1055 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h"
-
-#include <math.h>
-#include <stdio.h>
-
-#include <algorithm>
-#include <sstream>
-
-#ifdef MATLAB
-#include "engine.h"
-#endif
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-
-using namespace webrtc;
-
-#ifdef MATLAB
-MatlabEngine eng;
-
-MatlabLine::MatlabLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
-:
-_xArray(NULL),
-_yArray(NULL),
-_maxLen(maxLen),
-_plotAttribute(),
-_name()
-{
-    if (_maxLen > 0)
-    {
-        _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-        _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-    }
-
-    if (plotAttrib)
-    {
-        _plotAttribute = plotAttrib;
-    }
-
-    if (name)
-    {
-        _name = name;
-    }
-}
-
-MatlabLine::~MatlabLine()
-{
-    if (_xArray != NULL)
-    {
-        mxDestroyArray(_xArray);
-    }
-    if (_yArray != NULL)
-    {
-        mxDestroyArray(_yArray);
-    }
-}
-
-void MatlabLine::Append(double x, double y)
-{
-    if (_maxLen > 0 && _xData.size() > static_cast<uint32_t>(_maxLen))
-    {
-        _xData.resize(_maxLen);
-        _yData.resize(_maxLen);
-    }
-
-    _xData.push_front(x);
-    _yData.push_front(y);
-}
-
-
-// append y-data with running integer index as x-data
-void MatlabLine::Append(double y)
-{
-    if (_xData.empty())
-    {
-        // first element is index 0
-        Append(0, y);
-    }
-    else
-    {
-        // take last x-value and increment
-        double temp = _xData.back(); // last x-value
-        Append(temp + 1, y);
-    }
-}
-
-
-void MatlabLine::SetMaxLen(int maxLen)
-{
-    if (maxLen <= 0)
-    {
-        // means no maxLen
-        _maxLen = -1;
-    }
-    else
-    {
-        _maxLen = maxLen;
-
-        if (_xArray != NULL)
-        {
-            mxDestroyArray(_xArray);
-            mxDestroyArray(_yArray);
-        }
-        _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-        _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-
-        maxLen = ((unsigned int)maxLen <= _xData.size()) ? maxLen : (int)_xData.size();
-        _xData.resize(maxLen);
-        _yData.resize(maxLen);
-
-        //// reserve the right amount of memory
-        //_xData.reserve(_maxLen);
-        //_yData.reserve(_maxLen);
-    }
-}
-
-void MatlabLine::SetAttribute(char *plotAttrib)
-{
-    _plotAttribute = plotAttrib;
-}
-
-void MatlabLine::SetName(char *name)
-{
-    _name = name;
-}
-
-void MatlabLine::GetPlotData(mxArray** xData, mxArray** yData)
-{
-    // Make sure we have enough Matlab allocated memory.
-    // Assuming both arrays (x and y) are of the same size.
-    if (_xData.empty())
-    {
-        return; // No data
-    }
-    unsigned int size = 0;
-    if (_xArray != NULL)
-    {
-        size = (unsigned int)mxGetNumberOfElements(_xArray);
-    }
-    if (size < _xData.size())
-    {
-        if (_xArray != NULL)
-        {
-            mxDestroyArray(_xArray);
-            mxDestroyArray(_yArray);
-        }
-        _xArray = mxCreateDoubleMatrix(1, _xData.size(), mxREAL);
-        _yArray = mxCreateDoubleMatrix(1, _yData.size(), mxREAL);
-    }
-
-    if (!_xData.empty())
-    {
-        double* x = mxGetPr(_xArray);
-
-        std::list<double>::iterator it = _xData.begin();
-
-        for (int i = 0; it != _xData.end(); it++, i++)
-        {
-            x[i] = *it;
-        }
-    }
-
-    if (!_yData.empty())
-    {
-        double* y = mxGetPr(_yArray);
-
-        std::list<double>::iterator it = _yData.begin();
-
-        for (int i = 0; it != _yData.end(); it++, i++)
-        {
-            y[i] = *it;
-        }
-    }
-    *xData = _xArray;
-    *yData = _yArray;
-}
-
-std::string MatlabLine::GetXName()
-{
-    std::ostringstream xString;
-    xString << "x_" << _name;
-    return xString.str();
-}
-
-std::string MatlabLine::GetYName()
-{
-    std::ostringstream yString;
-    yString << "y_" << _name;
-    return yString.str();
-}
-
-std::string MatlabLine::GetPlotString()
-{
-
-    std::ostringstream s;
-
-    if (_xData.size() == 0)
-    {
-        s << "[0 1], [0 1]"; // To get an empty plot
-    }
-    else
-    {
-        s << GetXName() << "(1:" << _xData.size() << "),";
-        s << GetYName() << "(1:" << _yData.size() << ")";
-    }
-
-    s << ", '";
-    s << _plotAttribute;
-    s << "'";
-
-    return s.str();
-}
-
-std::string MatlabLine::GetRefreshString()
-{
-    std::ostringstream s;
-
-    if (_xData.size() > 0)
-    {
-        s << "set(h,'xdata',"<< GetXName() <<"(1:" << _xData.size() << "),'ydata',"<< GetYName() << "(1:" << _yData.size() << "));";
-    }
-    else
-    {
-        s << "set(h,'xdata',[NaN],'ydata',[NaN]);";
-    }
-    return s.str();
-}
-
-std::string MatlabLine::GetLegendString()
-{
-    return ("'" + _name + "'");
-}
-
-bool MatlabLine::hasLegend()
-{
-    return (!_name.empty());
-}
-
-
-// remove data points, but keep attributes
-void MatlabLine::Reset()
-{
-    _xData.clear();
-    _yData.clear();
-}
-
-
-void MatlabLine::UpdateTrendLine(MatlabLine * sourceData, double slope, double offset)
-{
-    Reset(); // reset data, not attributes and name
-
-    double thexMin = sourceData->xMin();
-    double thexMax = sourceData->xMax();
-    Append(thexMin, thexMin * slope + offset);
-    Append(thexMax, thexMax * slope + offset);
-}
-
-double MatlabLine::xMin()
-{
-    if (!_xData.empty())
-    {
-        std::list<double>::iterator theStart = _xData.begin();
-        std::list<double>::iterator theEnd = _xData.end();
-        return(*min_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-double MatlabLine::xMax()
-{
-    if (!_xData.empty())
-    {
-        std::list<double>::iterator theStart = _xData.begin();
-        std::list<double>::iterator theEnd = _xData.end();
-        return(*max_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-double MatlabLine::yMin()
-{
-    if (!_yData.empty())
-    {
-        std::list<double>::iterator theStart = _yData.begin();
-        std::list<double>::iterator theEnd = _yData.end();
-        return(*min_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-double MatlabLine::yMax()
-{
-    if (!_yData.empty())
-    {
-        std::list<double>::iterator theStart = _yData.begin();
-        std::list<double>::iterator theEnd = _yData.end();
-        return(*max_element(theStart, theEnd));
-    }
-    return (0.0);
-}
-
-
-
-MatlabTimeLine::MatlabTimeLine(int horizonSeconds /*= -1*/, const char *plotAttrib /*= NULL*/,
-                               const char *name /*= NULL*/,
-                               int64_t refTimeMs /* = -1*/)
-                               :
-_timeHorizon(horizonSeconds),
-MatlabLine(-1, plotAttrib, name) // infinite number of elements
-{
-    if (refTimeMs < 0)
-        _refTimeMs = TickTime::MillisecondTimestamp();
-    else
-        _refTimeMs = refTimeMs;
-}
-
-void MatlabTimeLine::Append(double y)
-{
-    MatlabLine::Append(static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0, y);
-
-    PurgeOldData();
-}
-
-
-void MatlabTimeLine::PurgeOldData()
-{
-    if (_timeHorizon > 0)
-    {
-        // remove old data
-        double historyLimit = static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0
-            - _timeHorizon; // remove data points older than this
-
-        std::list<double>::reverse_iterator ritx = _xData.rbegin();
-        uint32_t removeCount = 0;
-        while (ritx != _xData.rend())
-        {
-            if (*ritx >= historyLimit)
-            {
-                break;
-            }
-            ritx++;
-            removeCount++;
-        }
-        if (removeCount == 0)
-        {
-            return;
-        }
-
-        // remove the range [begin, it).
-        //if (removeCount > 10)
-        //{
-        //    printf("Removing %lu elements\n", removeCount);
-        //}
-        _xData.resize(_xData.size() - removeCount);
-        _yData.resize(_yData.size() - removeCount);
-    }
-}
-
-
-int64_t MatlabTimeLine::GetRefTime()
-{
-    return(_refTimeMs);
-}
-
-
-
-
-MatlabPlot::MatlabPlot()
-:
-_figHandle(-1),
-_smartAxis(false),
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_timeToPlot(false),
-_plotting(false),
-_enabled(true),
-_firstPlot(true),
-_legendEnabled(true),
-_donePlottingEvent(EventWrapper::Create())
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _xlim[0] = 0;
-    _xlim[1] = 0;
-    _ylim[0] = 0;
-    _ylim[1] = 0;
-
-#ifdef PLOT_TESTING
-    _plotStartTime = -1;
-    _plotDelay = 0;
-#endif
-
-}
-
-
-MatlabPlot::~MatlabPlot()
-{
-    _critSect->Enter();
-
-    // delete all line objects
-    while (!_line.empty())
-    {
-        delete *(_line.end() - 1);
-        _line.pop_back();
-    }
-
-    delete _critSect;
-    delete _donePlottingEvent;
-}
-
-
-int MatlabPlot::AddLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
-{
-    CriticalSectionScoped cs(_critSect);
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    MatlabLine *newLine = new MatlabLine(maxLen, plotAttrib, name);
-    _line.push_back(newLine);
-
-    return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
-}
-
-
-int MatlabPlot::AddTimeLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/,
-                            int64_t refTimeMs /*= -1*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    MatlabTimeLine *newLine = new MatlabTimeLine(maxLen, plotAttrib, name, refTimeMs);
-    _line.push_back(newLine);
-
-    return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
-}
-
-
-int MatlabPlot::GetLineIx(const char *name)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    // search the list for a matching line name
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    bool matchFound = false;
-    int lineIx = 0;
-
-    for (; it != _line.end(); it++, lineIx++)
-    {
-        if ((*it)->_name == name)
-        {
-            matchFound = true;
-            break;
-        }
-    }
-
-    if (matchFound)
-    {
-        return (lineIx);
-    }
-    else
-    {
-        return (-1);
-    }
-}
-
-
-void MatlabPlot::Append(int lineIndex, double x, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return;
-    }
-
-    // sanity for index
-    if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
-    {
-        throw "Line index out of range";
-        exit(1);
-    }
-
-    return (_line[lineIndex]->Append(x, y));
-}
-
-
-void MatlabPlot::Append(int lineIndex, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return;
-    }
-
-    // sanity for index
-    if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
-    {
-        throw "Line index out of range";
-        exit(1);
-    }
-
-    return (_line[lineIndex]->Append(y));
-}
-
-
-int MatlabPlot::Append(const char *name, double x, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    // search the list for a matching line name
-    int lineIx = GetLineIx(name);
-
-    if (lineIx < 0) //(!matchFound)
-    {
-        // no match; append new line
-        lineIx = AddLine(-1, NULL, name);
-    }
-
-    // append data to line
-    Append(lineIx, x, y);
-    return (lineIx);
-}
-
-int MatlabPlot::Append(const char *name, double y)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    // search the list for a matching line name
-    int lineIx = GetLineIx(name);
-
-    if (lineIx < 0) //(!matchFound)
-    {
-        // no match; append new line
-        lineIx = AddLine(-1, NULL, name);
-    }
-
-    // append data to line
-    Append(lineIx, y);
-    return (lineIx);
-}
-
-int MatlabPlot::Length(char *name)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return -1;
-    }
-
-    int ix = GetLineIx(name);
-    if (ix >= 0)
-    {
-        return (static_cast<int>(_line[ix]->_xData.size()));
-    }
-    else
-    {
-        return (-1);
-    }
-}
-
-
-void MatlabPlot::SetPlotAttribute(char *name, char *plotAttrib)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (!_enabled)
-    {
-        return;
-    }
-
-    int lineIx = GetLineIx(name);
-
-    if (lineIx >= 0)
-    {
-        _line[lineIx]->SetAttribute(plotAttrib);
-    }
-}
-
-// Must be called under critical section _critSect
-void MatlabPlot::UpdateData(Engine* ep)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-
-    for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-    {
-        mxArray* xData = NULL;
-        mxArray* yData = NULL;
-        (*it)->GetPlotData(&xData, &yData);
-        if (xData != NULL)
-        {
-            std::string xName = (*it)->GetXName();
-            std::string yName = (*it)->GetYName();
-            _critSect->Leave();
-#ifdef MATLAB6
-            mxSetName(xData, xName.c_str());
-            mxSetName(yData, yName.c_str());
-            engPutArray(ep, xData);
-            engPutArray(ep, yData);
-#else
-            int ret = engPutVariable(ep, xName.c_str(), xData);
-            assert(ret == 0);
-            ret = engPutVariable(ep, yName.c_str(), yData);
-            assert(ret == 0);
-#endif
-            _critSect->Enter();
-        }
-    }
-}
-
-bool MatlabPlot::GetPlotCmd(std::ostringstream & cmd, Engine* ep)
-{
-    _critSect->Enter();
-
-    if (!DataAvailable())
-    {
-        return false;
-    }
-
-    if (_firstPlot)
-    {
-        GetPlotCmd(cmd);
-        _firstPlot = false;
-    }
-    else
-    {
-        GetRefreshCmd(cmd);
-    }
-
-    UpdateData(ep);
-
-    _critSect->Leave();
-
-    return true;
-}
-
-// Call inside critsect
-void MatlabPlot::GetPlotCmd(std::ostringstream & cmd)
-{
-    // we have something to plot
-    // empty the stream
-    cmd.str(""); // (this seems to be the only way)
-
-    cmd << "figure; h" << _figHandle << "= plot(";
-
-    // first line
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    cmd << (*it)->GetPlotString();
-
-    it++;
-
-    // remaining lines
-    for (; it != _line.end(); it++)
-    {
-        cmd << ", ";
-        cmd << (*it)->GetPlotString();
-    }
-
-    cmd << "); ";
-
-    if (_legendEnabled)
-    {
-        GetLegendCmd(cmd);
-    }
-
-    if (_smartAxis)
-    {
-        double xMin = _xlim[0];
-        double xMax = _xlim[1];
-        double yMax = _ylim[1];
-        for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-        {
-            xMax = std::max(xMax, (*it)->xMax());
-            xMin = std::min(xMin, (*it)->xMin());
-
-            yMax = std::max(yMax, (*it)->yMax());
-            yMax = std::max(yMax, fabs((*it)->yMin()));
-        }
-        _xlim[0] = xMin;
-        _xlim[1] = xMax;
-        _ylim[0] = -yMax;
-        _ylim[1] = yMax;
-
-        cmd << "axis([" << _xlim[0] << ", " << _xlim[1] << ", " << _ylim[0] << ", " << _ylim[1] << "]);";
-    }
-
-    int i=1;
-    for (it = _line.begin(); it != _line.end(); i++, it++)
-    {
-        cmd << "set(h" << _figHandle << "(" << i << "), 'Tag', " << (*it)->GetLegendString() << ");";
-    }
-}
-
-// Call inside critsect
-void MatlabPlot::GetRefreshCmd(std::ostringstream & cmd)
-{
-    cmd.str(""); // (this seems to be the only way)
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    for (it = _line.begin(); it != _line.end(); it++)
-    {
-        cmd << "h = findobj(0, 'Tag', " << (*it)->GetLegendString() << ");";
-        cmd << (*it)->GetRefreshString();
-    }
-    //if (_legendEnabled)
-    //{
-    //    GetLegendCmd(cmd);
-    //}
-}
-
-void MatlabPlot::GetLegendCmd(std::ostringstream & cmd)
-{
-    std::vector<MatlabLine*>::iterator it = _line.begin();
-    bool anyLegend = false;
-    for (; it != _line.end(); it++)
-    {
-        anyLegend = anyLegend || (*it)->hasLegend();
-    }
-    if (anyLegend)
-    {
-        // create the legend
-
-        cmd << "legend(h" << _figHandle << ",{";
-
-
-        // iterate lines
-        int i = 0;
-        for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-        {
-            if (i > 0)
-            {
-                cmd << ", ";
-            }
-            cmd << (*it)->GetLegendString();
-            i++;
-        }
-
-        cmd << "}, 2); "; // place legend in upper-left corner
-    }
-}
-
-// Call inside critsect
-bool MatlabPlot::DataAvailable()
-{
-    if (!_enabled)
-    {
-        return false;
-    }
-
-    for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-    {
-        (*it)->PurgeOldData();
-    }
-
-    return true;
-}
-
-void MatlabPlot::Plot()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _timeToPlot = true;
-
-#ifdef PLOT_TESTING
-    _plotStartTime = TickTime::MillisecondTimestamp();
-#endif
-}
-
-
-void MatlabPlot::Reset()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _enabled = true;
-
-    for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
-    {
-        (*it)->Reset();
-    }
-
-}
-
-void MatlabPlot::SetFigHandle(int handle)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (handle > 0)
-        _figHandle = handle;
-}
-
-bool
-MatlabPlot::TimeToPlot()
-{
-    CriticalSectionScoped cs(_critSect);
-    return _enabled && _timeToPlot;
-}
-
-void
-MatlabPlot::Plotting()
-{
-    CriticalSectionScoped cs(_critSect);
-    _plotting = true;
-}
-
-void
-MatlabPlot::DonePlotting()
-{
-    CriticalSectionScoped cs(_critSect);
-    _timeToPlot = false;
-    _plotting = false;
-    _donePlottingEvent->Set();
-}
-
-void
-MatlabPlot::DisablePlot()
-{
-    _critSect->Enter();
-    while (_plotting)
-    {
-        _critSect->Leave();
-        _donePlottingEvent->Wait(WEBRTC_EVENT_INFINITE);
-        _critSect->Enter();
-    }
-    _enabled = false;
-}
-
-int MatlabPlot::MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    int sourceIx;
-    int trendIx;
-
-    sourceIx = GetLineIx(sourceName);
-    if (sourceIx < 0)
-    {
-        // could not find source
-        return (-1);
-    }
-
-    trendIx = GetLineIx(trendName);
-    if (trendIx < 0)
-    {
-        // no trend found; add new line
-        trendIx = AddLine(2 /*maxLen*/, plotAttrib, trendName);
-    }
-
-    _line[trendIx]->UpdateTrendLine(_line[sourceIx], slope, offset);
-
-    return (trendIx);
-
-}
-
-
-MatlabEngine::MatlabEngine()
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_running(false),
-_numPlots(0)
-{
-    _eventPtr = EventWrapper::Create();
-
-    _plotThread = ThreadWrapper::CreateThread(MatlabEngine::PlotThread, this,
-                                              kLowPriority, "MatlabPlot");
-    _running = true;
-    _plotThread->Start();
-}
-
-MatlabEngine::~MatlabEngine()
-{
-    _critSect->Enter();
-
-    if (_plotThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _plotThread->Stop();
-    }
-
-    _plots.clear();
-
-    delete _eventPtr;
-    _eventPtr = NULL;
-
-    _critSect->Leave();
-    delete _critSect;
-
-}
-
-MatlabPlot * MatlabEngine::NewPlot(MatlabPlot *newPlot)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    //MatlabPlot *newPlot = new MatlabPlot();
-
-    if (newPlot)
-    {
-        newPlot->SetFigHandle(++_numPlots); // first plot is number 1
-        _plots.push_back(newPlot);
-    }
-
-    return (newPlot);
-
-}
-
-
-void MatlabEngine::DeletePlot(MatlabPlot *plot)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (plot == NULL)
-    {
-        return;
-    }
-
-    std::vector<MatlabPlot *>::iterator it;
-    for (it = _plots.begin(); it < _plots.end(); it++)
-    {
-        if (plot == *it)
-        {
-            break;
-        }
-    }
-
-    assert (plot == *it);
-
-    (*it)->DisablePlot();
-
-    _plots.erase(it);
-    --_numPlots;
-
-    delete plot;
-}
-
-
-bool MatlabEngine::PlotThread(void *obj)
-{
-    if (!obj)
-    {
-        return (false);
-    }
-
-    MatlabEngine *eng = (MatlabEngine *) obj;
-
-    Engine *ep = engOpen(NULL);
-    if (!ep)
-    {
-        throw "Cannot open Matlab engine";
-        return (false);
-    }
-
-    engSetVisible(ep, true);
-    engEvalString(ep, "close all;");
-
-    while (eng->_running)
-    {
-        eng->_critSect->Enter();
-
-        // iterate through all plots
-        for (unsigned int ix = 0; ix < eng->_plots.size(); ix++)
-        {
-            MatlabPlot *plot = eng->_plots[ix];
-            if (plot->TimeToPlot())
-            {
-                plot->Plotting();
-                eng->_critSect->Leave();
-                std::ostringstream cmd;
-
-                if (engEvalString(ep, cmd.str().c_str()))
-                {
-                    // engine dead
-                    return (false);
-                }
-
-                // empty the stream
-                cmd.str(""); // (this seems to be the only way)
-                if (plot->GetPlotCmd(cmd, ep))
-                {
-                    // things to plot, we have already accessed what we need in the plot
-                    plot->DonePlotting();
-
-                    int64_t start = TickTime::MillisecondTimestamp();
-                    // plot it
-                    int ret = engEvalString(ep, cmd.str().c_str());
-                    printf("time=%I64i\n", TickTime::MillisecondTimestamp() - start);
-                    if (ret)
-                    {
-                        // engine dead
-                        return (false);
-                    }
-
-#ifdef PLOT_TESTING
-                    if(plot->_plotStartTime >= 0)
-                    {
-                        plot->_plotDelay = TickTime::MillisecondTimestamp() - plot->_plotStartTime;
-                        plot->_plotStartTime = -1;
-                    }
-#endif
-                }
-                eng->_critSect->Enter();
-            }
-        }
-
-        eng->_critSect->Leave();
-        // wait a while
-        eng->_eventPtr->Wait(66); // 33 ms
-    }
-
-    if (ep)
-    {
-        engClose(ep);
-        ep = NULL;
-    }
-
-    return (true);
-
-}
-
-#endif // MATLAB
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
deleted file mode 100644
index 7623daab94..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
+++ /dev/null
@@ -1,170 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
-
-#include <list>
-#include <string>
-#include <vector>
-
-#include "webrtc/typedefs.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-}
-
-//#define PLOT_TESTING
-
-#ifdef MATLAB
-
-typedef struct engine Engine;
-typedef struct mxArray_tag mxArray;
-
-class MatlabLine
-{
-    friend class MatlabPlot;
-
-public:
-    MatlabLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
-    ~MatlabLine();
-    virtual void Append(double x, double y);
-    virtual void Append(double y);
-    void SetMaxLen(int maxLen);
-    void SetAttribute(char *plotAttrib);
-    void SetName(char *name);
-    void Reset();
-    virtual void PurgeOldData() {};
-
-    void UpdateTrendLine(MatlabLine * sourceData, double slope, double offset);
-
-    double xMin();
-    double xMax();
-    double yMin();
-    double yMax();
-
-protected:
-    void GetPlotData(mxArray** xData, mxArray** yData);
-    std::string GetXName();
-    std::string GetYName();
-    std::string GetPlotString();
-    std::string GetRefreshString();
-    std::string GetLegendString();
-    bool hasLegend();
-    std::list<double> _xData;
-    std::list<double> _yData;
-    mxArray* _xArray;
-    mxArray* _yArray;
-    int _maxLen;
-    std::string _plotAttribute;
-    std::string _name;
-};
-
-
-class MatlabTimeLine : public MatlabLine
-{
-public:
-    MatlabTimeLine(int horizonSeconds = -1, const char *plotAttrib = NULL, const char *name = NULL,
-        int64_t refTimeMs = -1);
-    ~MatlabTimeLine() {};
-    void Append(double y);
-    void PurgeOldData();
-    int64_t GetRefTime();
-
-private:
-    int64_t _refTimeMs;
-    int _timeHorizon;
-};
-
-
-class MatlabPlot
-{
-    friend class MatlabEngine;
-
-public:
-    MatlabPlot();
-    ~MatlabPlot();
-
-    int AddLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
-    int AddTimeLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL,
-        int64_t refTimeMs = -1);
-    int GetLineIx(const char *name);
-    void Append(int lineIndex, double x, double y);
-    void Append(int lineIndex, double y);
-    int Append(const char *name, double x, double y);
-    int Append(const char *name, double y);
-    int Length(char *name);
-    void SetPlotAttribute(char *name, char *plotAttrib);
-    void Plot();
-    void Reset();
-    void SmartAxis(bool status = true) { _smartAxis = status; };
-    void SetFigHandle(int handle);
-    void EnableLegend(bool enable) { _legendEnabled = enable; };
-
-    bool TimeToPlot();
-    void Plotting();
-    void DonePlotting();
-    void DisablePlot();
-
-    int MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib = NULL);
-
-#ifdef PLOT_TESTING
-    int64_t _plotStartTime;
-    int64_t _plotDelay;
-#endif
-
-private:
-    void UpdateData(Engine* ep);
-    bool GetPlotCmd(std::ostringstream & cmd, Engine* ep);
-    void GetPlotCmd(std::ostringstream & cmd); // call inside crit sect
-    void GetRefreshCmd(std::ostringstream & cmd); // call inside crit sect
-    void GetLegendCmd(std::ostringstream & cmd);
-    bool DataAvailable();
-
-    std::vector<MatlabLine *> _line;
-    int _figHandle;
-    bool _smartAxis;
-    double _xlim[2];
-    double _ylim[2];
-    webrtc::CriticalSectionWrapper *_critSect;
-    bool _timeToPlot;
-    bool _plotting;
-    bool _enabled;
-    bool _firstPlot;
-    bool _legendEnabled;
-    webrtc::EventWrapper* _donePlottingEvent;
-};
-
-
-class MatlabEngine
-{
-public:
-    MatlabEngine();
-    ~MatlabEngine();
-
-    MatlabPlot * NewPlot(MatlabPlot *newPlot);
-    void DeletePlot(MatlabPlot *plot);
-
-private:
-    static bool PlotThread(void *obj);
-
-    std::vector<MatlabPlot *> _plots;
-    webrtc::CriticalSectionWrapper *_critSect;
-    webrtc::EventWrapper *_eventPtr;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _plotThread;
-    bool _running;
-    int _numPlots;
-};
-
-#endif //MATLAB
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
deleted file mode 100644
index 0ed35728e4..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
+++ /dev/null
@@ -1,432 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-
-#include <stdio.h>
-
-#include <algorithm>
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-
-
-bool SenderThreadFunction(void *obj)
-{
-    if (obj == NULL)
-    {
-        return false;
-    }
-    TestLoadGenerator *_genObj = static_cast<TestLoadGenerator *>(obj);
-
-    return _genObj->GeneratorLoop();
-}
-
-
-TestLoadGenerator::TestLoadGenerator(TestSenderReceiver *sender, int32_t rtpSampleRate)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_bitrateKbps(0),
-_sender(sender),
-_running(false),
-_rtpSampleRate(rtpSampleRate)
-{
-}
-
-TestLoadGenerator::~TestLoadGenerator ()
-{
-    if (_running)
-    {
-        Stop();
-    }
-
-    delete _critSect;
-}
-
-int32_t TestLoadGenerator::SetBitrate (int32_t newBitrateKbps)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    if (newBitrateKbps < 0)
-    {
-        return -1;
-    }
-
-    _bitrateKbps = newBitrateKbps;
-
-    printf("New bitrate = %i kbps\n", _bitrateKbps);
-
-    return _bitrateKbps;
-}
-
-
-int32_t TestLoadGenerator::Start (const char *threadName)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _eventPtr = EventWrapper::Create();
-
-    _genThread = ThreadWrapper::CreateThread(SenderThreadFunction, this,
-                                             threadName);
-    _running = true;
-
-    _genThread->Start();
-    _genThread->SetPriority(kRealtimePriority);
-
-    return 0;
-}
-
-
-int32_t TestLoadGenerator::Stop ()
-{
-    _critSect.Enter();
-
-    if (_genThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _genThread->Stop();
-        _genThread.reset();
-
-        delete _eventPtr;
-        _eventPtr = NULL;
-    }
-
-    _critSect.Leave();
-    return (0);
-}
-
-
-int TestLoadGenerator::generatePayload ()
-{
-    return(generatePayload( static_cast<uint32_t>( TickTime::MillisecondTimestamp() * _rtpSampleRate / 1000 )));
-}
-
-
-int TestLoadGenerator::sendPayload (const uint32_t timeStamp,
-                                    const uint8_t* payloadData,
-                                    const size_t payloadSize,
-                                    const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
-
-    return (_sender->SendOutgoingData(timeStamp, payloadData, payloadSize, frameType));
-}
-
-
-CBRGenerator::CBRGenerator (TestSenderReceiver *sender,
-                            size_t payloadSizeBytes,
-                            int32_t bitrateKbps,
-                            int32_t rtpSampleRate)
-:
-//_eventPtr(NULL),
-_payloadSizeBytes(payloadSizeBytes),
-_payload(new uint8_t[payloadSizeBytes]),
-TestLoadGenerator(sender, rtpSampleRate)
-{
-    SetBitrate (bitrateKbps);
-}
-
-CBRGenerator::~CBRGenerator ()
-{
-    if (_running)
-    {
-        Stop();
-    }
-
-    if (_payload)
-    {
-        delete [] _payload;
-    }
-
-}
-
-bool CBRGenerator::GeneratorLoop ()
-{
-    double periodMs;
-    int64_t nextSendTime = TickTime::MillisecondTimestamp();
-
-
-    // no critSect
-    while (_running)
-    {
-        // send data (critSect inside)
-        generatePayload( static_cast<uint32_t>(nextSendTime * _rtpSampleRate / 1000) );
-
-        // calculate wait time
-        periodMs = 8.0 * _payloadSizeBytes / ( _bitrateKbps );
-
-        nextSendTime = static_cast<int64_t>(nextSendTime + periodMs);
-
-        int32_t waitTime = static_cast<int32_t>(nextSendTime - TickTime::MillisecondTimestamp());
-        if (waitTime < 0)
-        {
-            waitTime = 0;
-        }
-        // wait
-        _eventPtr->Wait(static_cast<int32_t>(waitTime));
-    }
-
-    return true;
-}
-
-int CBRGenerator::generatePayload ( uint32_t timestamp )
-{
-    CriticalSectionScoped cs(_critSect);
-
-    //uint8_t *payload = new uint8_t[_payloadSizeBytes];
-
-    int ret = sendPayload(timestamp, _payload, _payloadSizeBytes);
-
-    //delete [] payload;
-    return ret;
-}
-
-
-
-
-/////////////////////
-
-CBRFixFRGenerator::CBRFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps,
-                                      int32_t rtpSampleRate, int32_t frameRateFps /*= 30*/,
-                                      double spread /*= 0.0*/)
-:
-//_eventPtr(NULL),
-_payloadSizeBytes(0),
-_payload(NULL),
-_payloadAllocLen(0),
-_frameRateFps(frameRateFps),
-_spreadFactor(spread),
-TestLoadGenerator(sender, rtpSampleRate)
-{
-    SetBitrate (bitrateKbps);
-}
-
-CBRFixFRGenerator::~CBRFixFRGenerator ()
-{
-    if (_running)
-    {
-        Stop();
-    }
-
-    if (_payload)
-    {
-        delete [] _payload;
-        _payloadAllocLen = 0;
-    }
-
-}
-
-bool CBRFixFRGenerator::GeneratorLoop ()
-{
-    double periodMs;
-    int64_t nextSendTime = TickTime::MillisecondTimestamp();
-
-    _critSect.Enter();
-
-    if (_frameRateFps <= 0)
-    {
-        return false;
-    }
-
-    _critSect.Leave();
-
-    // no critSect
-    while (_running)
-    {
-        _critSect.Enter();
-
-        // calculate payload size
-        _payloadSizeBytes = nextPayloadSize();
-
-        if (_payloadSizeBytes > 0)
-        {
-
-            if (_payloadAllocLen < _payloadSizeBytes * (1 + _spreadFactor))
-            {
-                // re-allocate _payload
-                if (_payload)
-                {
-                    delete [] _payload;
-                    _payload = NULL;
-                }
-
-                _payloadAllocLen = static_cast<int32_t>((_payloadSizeBytes * (1 + _spreadFactor) * 3) / 2 + .5); // 50% extra to avoid frequent re-alloc
-                _payload = new uint8_t[_payloadAllocLen];
-            }
-
-
-            // send data (critSect inside)
-            generatePayload( static_cast<uint32_t>(nextSendTime * _rtpSampleRate / 1000) );
-        }
-
-        _critSect.Leave();
-
-        // calculate wait time
-        periodMs = 1000.0 / _frameRateFps;
-        nextSendTime = static_cast<int64_t>(nextSendTime + periodMs + 0.5);
-
-        int32_t waitTime = static_cast<int32_t>(nextSendTime - TickTime::MillisecondTimestamp());
-        if (waitTime < 0)
-        {
-            waitTime = 0;
-        }
-        // wait
-        _eventPtr->Wait(waitTime);
-    }
-
-    return true;
-}
-
-size_t CBRFixFRGenerator::nextPayloadSize()
-{
-    const double periodMs = 1000.0 / _frameRateFps;
-    return static_cast<size_t>(_bitrateKbps * periodMs / 8 + 0.5);
-}
-
-int CBRFixFRGenerator::generatePayload ( uint32_t timestamp )
-{
-    CriticalSectionScoped cs(_critSect);
-
-    double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
-    factor = 1 + 2 * _spreadFactor * factor; // [1 - _spreadFactor ; 1 + _spreadFactor]
-
-    size_t thisPayloadBytes = static_cast<size_t>(_payloadSizeBytes * factor);
-    // sanity
-    if (thisPayloadBytes > _payloadAllocLen)
-    {
-        thisPayloadBytes = _payloadAllocLen;
-    }
-
-    int ret = sendPayload(timestamp, _payload, thisPayloadBytes);
-    return ret;
-}
-
-
-/////////////////////
-
-PeriodicKeyFixFRGenerator::PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps,
-                                                      int32_t rtpSampleRate, int32_t frameRateFps /*= 30*/,
-                                                      double spread /*= 0.0*/, double keyFactor /*= 4.0*/, uint32_t keyPeriod /*= 300*/)
-:
-_keyFactor(keyFactor),
-_keyPeriod(keyPeriod),
-_frameCount(0),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRateFps, spread)
-{
-}
-
-size_t PeriodicKeyFixFRGenerator::nextPayloadSize()
-{
-    // calculate payload size for a delta frame
-    size_t payloadSizeBytes = static_cast<size_t>(1000 * _bitrateKbps /
-        (8.0 * _frameRateFps * (1.0 + (_keyFactor - 1.0) / _keyPeriod)) + 0.5);
-
-    if (_frameCount % _keyPeriod == 0)
-    {
-        // this is a key frame, scale the payload size
-        payloadSizeBytes =
-            static_cast<size_t>(_keyFactor * _payloadSizeBytes + 0.5);
-    }
-    _frameCount++;
-
-    return payloadSizeBytes;
-}
-
-////////////////////
-
-CBRVarFRGenerator::CBRVarFRGenerator(TestSenderReceiver *sender, int32_t bitrateKbps, const uint8_t* frameRates,
-                                     uint16_t numFrameRates, int32_t rtpSampleRate, double avgFrPeriodMs,
-                                     double frSpreadFactor, double spreadFactor)
-:
-_avgFrPeriodMs(avgFrPeriodMs),
-_frSpreadFactor(frSpreadFactor),
-_frameRates(NULL),
-_numFrameRates(numFrameRates),
-_frChangeTimeMs(TickTime::MillisecondTimestamp() + _avgFrPeriodMs),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRates[0], spreadFactor)
-{
-    _frameRates = new uint8_t[_numFrameRates];
-    memcpy(_frameRates, frameRates, _numFrameRates);
-}
-
-CBRVarFRGenerator::~CBRVarFRGenerator()
-{
-    delete [] _frameRates;
-}
-
-void CBRVarFRGenerator::ChangeFrameRate()
-{
-    const int64_t nowMs = TickTime::MillisecondTimestamp();
-    if (nowMs < _frChangeTimeMs)
-    {
-        return;
-    }
-    // Time to change frame rate
-    uint16_t frIndex = static_cast<uint16_t>(static_cast<double>(rand()) / RAND_MAX
-                                            * (_numFrameRates - 1) + 0.5) ;
-    assert(frIndex < _numFrameRates);
-    _frameRateFps = _frameRates[frIndex];
-    // Update the next frame rate change time
-    double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
-    factor = 1 + 2 * _frSpreadFactor * factor; // [1 - _frSpreadFactor ; 1 + _frSpreadFactor]
-    _frChangeTimeMs = nowMs + static_cast<int64_t>(1000.0 * factor *
-                                                   _avgFrPeriodMs + 0.5);
-
-    printf("New frame rate: %d\n", _frameRateFps);
-}
-
-size_t CBRVarFRGenerator::nextPayloadSize()
-{
-    ChangeFrameRate();
-    return CBRFixFRGenerator::nextPayloadSize();
-}
-
-////////////////////
-
-CBRFrameDropGenerator::CBRFrameDropGenerator(TestSenderReceiver *sender, int32_t bitrateKbps,
-                                         int32_t rtpSampleRate, double spreadFactor)
-:
-_accBits(0),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, 30, spreadFactor)
-{
-}
-
-CBRFrameDropGenerator::~CBRFrameDropGenerator()
-{
-}
-
-size_t CBRFrameDropGenerator::nextPayloadSize()
-{
-    _accBits -= 1000 * _bitrateKbps / _frameRateFps;
-    if (_accBits < 0)
-    {
-        _accBits = 0;
-    }
-    if (_accBits > 0.3 * _bitrateKbps * 1000)
-    {
-        //printf("drop\n");
-        return 0;
-    }
-    else
-    {
-        //printf("keep\n");
-        const double periodMs = 1000.0 / _frameRateFps;
-        size_t frameSize =
-            static_cast<size_t>(_bitrateKbps * periodMs / 8 + 0.5);
-        frameSize =
-            std::max(frameSize, static_cast<size_t>(300 * periodMs / 8 + 0.5));
-        _accBits += frameSize * 8;
-        return frameSize;
-    }
-}
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
deleted file mode 100644
index fbd79177ea..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
-
-#include <stdlib.h>
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/typedefs.h"
-
-class TestSenderReceiver;
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-}
-
-class TestLoadGenerator
-{
-public:
-    TestLoadGenerator (TestSenderReceiver *sender, int32_t rtpSampleRate = 90000);
-    virtual ~TestLoadGenerator ();
-
-    int32_t SetBitrate (int32_t newBitrateKbps);
-    virtual int32_t Start (const char *threadName = NULL);
-    virtual int32_t Stop ();
-    virtual bool GeneratorLoop () = 0;
-
-protected:
-    virtual int generatePayload ( uint32_t timestamp ) = 0;
-    int generatePayload ();
-    int sendPayload (const uint32_t timeStamp,
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
-
-    webrtc::CriticalSectionWrapper* _critSect;
-    webrtc::EventWrapper *_eventPtr;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _genThread;
-    int32_t _bitrateKbps;
-    TestSenderReceiver *_sender;
-    bool _running;
-    int32_t _rtpSampleRate;
-};
-
-
-class CBRGenerator : public TestLoadGenerator
-{
-public:
-    CBRGenerator (TestSenderReceiver *sender,
-                  size_t payloadSizeBytes,
-                  int32_t bitrateKbps,
-                  int32_t rtpSampleRate = 90000);
-    virtual ~CBRGenerator ();
-
-    virtual int32_t Start () {return (TestLoadGenerator::Start("CBRGenerator"));};
-
-    virtual bool GeneratorLoop ();
-
-protected:
-    virtual int generatePayload ( uint32_t timestamp );
-
-    size_t _payloadSizeBytes;
-    uint8_t *_payload;
-};
-
-
-class CBRFixFRGenerator : public TestLoadGenerator // constant bitrate and fixed frame rate
-{
-public:
-    CBRFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps, int32_t rtpSampleRate = 90000,
-        int32_t frameRateFps = 30, double spread = 0.0);
-    virtual ~CBRFixFRGenerator ();
-
-    virtual int32_t Start () {return (TestLoadGenerator::Start("CBRFixFRGenerator"));};
-
-    virtual bool GeneratorLoop ();
-
-protected:
-    virtual size_t nextPayloadSize ();
-    virtual int generatePayload ( uint32_t timestamp );
-
-    size_t _payloadSizeBytes;
-    uint8_t *_payload;
-    size_t _payloadAllocLen;
-    int32_t _frameRateFps;
-    double      _spreadFactor;
-};
-
-class PeriodicKeyFixFRGenerator : public CBRFixFRGenerator // constant bitrate and fixed frame rate with periodically large frames
-{
-public:
-    PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, int32_t bitrateKbps, int32_t rtpSampleRate = 90000,
-        int32_t frameRateFps = 30, double spread = 0.0, double keyFactor = 4.0, uint32_t keyPeriod = 300);
-    virtual ~PeriodicKeyFixFRGenerator () {}
-
-protected:
-    virtual size_t nextPayloadSize ();
-
-    double          _keyFactor;
-    uint32_t    _keyPeriod;
-    uint32_t    _frameCount;
-};
-
-// Probably better to inherit CBRFixFRGenerator from CBRVarFRGenerator, but since
-// the fix FR version already existed this was easier.
-class CBRVarFRGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
-{
-public:
-    CBRVarFRGenerator(TestSenderReceiver *sender, int32_t bitrateKbps, const uint8_t* frameRates,
-        uint16_t numFrameRates, int32_t rtpSampleRate = 90000, double avgFrPeriodMs = 5.0,
-        double frSpreadFactor = 0.05, double spreadFactor = 0.0);
-
-    ~CBRVarFRGenerator();
-
-protected:
-    virtual void ChangeFrameRate();
-    virtual size_t nextPayloadSize ();
-
-    double       _avgFrPeriodMs;
-    double       _frSpreadFactor;
-    uint8_t* _frameRates;
-    uint16_t _numFrameRates;
-    int64_t  _frChangeTimeMs;
-};
-
-class CBRFrameDropGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
-{
-public:
-    CBRFrameDropGenerator(TestSenderReceiver *sender, int32_t bitrateKbps,
-                    int32_t rtpSampleRate = 90000, double spreadFactor = 0.0);
-
-    ~CBRFrameDropGenerator();
-
-protected:
-    virtual size_t nextPayloadSize();
-
-    double       _accBits;
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
deleted file mode 100644
index e55d363467..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ /dev/null
@@ -1,413 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-
-#define NR_OF_SOCKET_BUFFERS 500
-
-
-bool ProcThreadFunction(void *obj)
-{
-    if (obj == NULL)
-    {
-        return false;
-    }
-    TestSenderReceiver *theObj = static_cast<TestSenderReceiver *>(obj);
-
-    return theObj->ProcLoop();
-}
-
-
-TestSenderReceiver::TestSenderReceiver (void)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_running(false),
-_payloadType(0),
-_loadGenerator(NULL),
-_isSender(false),
-_isReceiver(false),
-_sendRecCB(NULL),
-_lastBytesReceived(0),
-_lastTime(-1)
-{
-    // RTP/RTCP module
-    _rtp = RtpRtcp::CreateRtpRtcp(0, false);
-    if (!_rtp)
-    {
-        throw "Could not create RTP/RTCP module";
-        exit(1);
-    }
-
-    if (_rtp->InitReceiver() != 0)
-    {
-        throw "_rtp->InitReceiver()";
-        exit(1);
-    }
-
-    if (_rtp->InitSender() != 0)
-    {
-        throw "_rtp->InitSender()";
-        exit(1);
-    }
-
-    // SocketTransport module
-    uint8_t numberOfThreads = 1;
-    _transport = UdpTransport::Create(0, numberOfThreads);
-    if (!_transport)
-    {
-        throw "Could not create transport module";
-        exit(1);
-    }
-}
-
-TestSenderReceiver::~TestSenderReceiver (void)
-{
-
-    Stop(); // N.B. without critSect
-
-    _critSect->Enter();
-
-    if (_rtp)
-    {
-        RtpRtcp::DestroyRtpRtcp(_rtp);
-        _rtp = NULL;
-    }
-
-    if (_transport)
-    {
-        UdpTransport::Destroy(_transport);
-        _transport = NULL;
-    }
-
-    delete _critSect;
-
-}
-
-
-int32_t TestSenderReceiver::InitReceiver (const uint16_t rtpPort,
-                                          const uint16_t rtcpPort,
-                                          const int8_t payloadType /*= 127*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    // init transport
-    if (_transport->InitializeReceiveSockets(this, rtpPort/*, 0, NULL, 0, true*/) != 0)
-    {
-        throw "_transport->InitializeReceiveSockets";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingRTPCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingRTPCallback";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingDataCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingRTPCallback";
-        exit(1);
-    }
-
-    if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0)
-    {
-        throw "_rtp->SetRTCPStatus";
-        exit(1);
-    }
-
-    if (_rtp->SetTMMBRStatus(true) != 0)
-    {
-        throw "_rtp->SetTMMBRStatus";
-        exit(1);
-    }
-
-    if (_rtp->RegisterReceivePayload("I420", payloadType, 90000) != 0)
-    {
-        throw "_rtp->RegisterReceivePayload";
-        exit(1);
-    }
-
-    _isReceiver = true;
-
-    return (0);
-}
-
-
-int32_t TestSenderReceiver::Start()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _eventPtr = EventWrapper::Create();
-
-    if (_rtp->SetSendingStatus(true) != 0)
-    {
-        throw "_rtp->SetSendingStatus";
-        exit(1);
-    }
-
-    _procThread = ThreadWrapper::CreateThread(ProcThreadFunction, this,
-                                              "TestSenderReceiver");
-
-    _running = true;
-
-    if (_isReceiver)
-    {
-        if (_transport->StartReceiving(NR_OF_SOCKET_BUFFERS) != 0)
-        {
-            throw "_transport->StartReceiving";
-            exit(1);
-        }
-    }
-
-    _procThread->Start();
-    _procThread->SetPriority(kRealtimePriority);
-
-    return 0;
-
-}
-
-
-int32_t TestSenderReceiver::Stop ()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _transport->StopReceiving();
-
-    if (_procThread)
-    {
-        _running = false;
-        _eventPtr->Set();
-
-        _procThread->Stop();
-        _procThread.reset();
-
-        delete _eventPtr;
-    }
-
-    return (0);
-}
-
-
-bool TestSenderReceiver::ProcLoop(void)
-{
-
-    // process RTP/RTCP module
-    _rtp->Process();
-
-    // process SocketTransport module
-    _transport->Process();
-
-    // no critSect
-    while (_running)
-    {
-        // ask RTP/RTCP module for wait time
-        int32_t rtpWait = _rtp->TimeUntilNextProcess();
-
-        // ask SocketTransport module for wait time
-        int32_t tpWait = _transport->TimeUntilNextProcess();
-
-        int32_t minWait = (rtpWait < tpWait) ? rtpWait: tpWait;
-        minWait = (minWait > 0) ? minWait : 0;
-        // wait
-        _eventPtr->Wait(minWait);
-
-        // process RTP/RTCP module
-        _rtp->Process();
-
-        // process SocketTransport module
-        _transport->Process();
-
-    }
-
-    return true;
-}
-
-
-int32_t TestSenderReceiver::ReceiveBitrateKbps ()
-{
-    size_t bytesSent;
-    uint32_t packetsSent;
-    size_t bytesReceived;
-    uint32_t packetsReceived;
-
-    if (_rtp->DataCountersRTP(&bytesSent, &packetsSent, &bytesReceived, &packetsReceived) == 0)
-    {
-        int64_t now = TickTime::MillisecondTimestamp();
-        int32_t kbps = 0;
-        if (now > _lastTime)
-        {
-            if (_lastTime > 0)
-            {
-                // 8 * bytes / ms = kbps
-                kbps = static_cast<int32_t>(
-                    (8 * (bytesReceived - _lastBytesReceived)) / (now - _lastTime));
-            }
-            _lastTime = now;
-            _lastBytesReceived = bytesReceived;
-        }
-        return (kbps);
-    }
-
-    return (-1);
-}
-
-
-int32_t TestSenderReceiver::SetPacketTimeout(const uint32_t timeoutMS)
-{
-    return (_rtp->SetPacketTimeout(timeoutMS, 0 /* RTCP timeout */));
-}
-
-
-int32_t TestSenderReceiver::OnReceivedPayloadData(const uint8_t* payloadData,
-                                                  const size_t payloadSize,
-                                                  const webrtc::WebRtcRTPHeader* rtpHeader)
-{
-    //printf("OnReceivedPayloadData\n");
-    return (0);
-}
-
-
-void TestSenderReceiver::IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                                           const size_t rtpPacketLength,
-                                           const int8_t* fromIP,
-                                           const uint16_t fromPort)
-{
-    _rtp->IncomingPacket((uint8_t *) incomingRtpPacket, rtpPacketLength);
-}
-
-
-
-void TestSenderReceiver::IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                                            const size_t rtcpPacketLength,
-                                            const int8_t* fromIP,
-                                            const uint16_t fromPort)
-{
-    _rtp->IncomingPacket((uint8_t *) incomingRtcpPacket, rtcpPacketLength);
-}
-
-
-
-
-
-///////////////////
-
-
-int32_t TestSenderReceiver::InitSender (const uint32_t startBitrateKbps,
-                                        const int8_t* ipAddr,
-                                        const uint16_t rtpPort,
-                                        const uint16_t rtcpPort /*= 0*/,
-                                        const int8_t payloadType /*= 127*/)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _payloadType = payloadType;
-
-    // check load generator valid
-    if (_loadGenerator)
-    {
-        _loadGenerator->SetBitrate(startBitrateKbps);
-    }
-
-    if (_rtp->RegisterSendTransport(_transport) != 0)
-    {
-        throw "_rtp->RegisterSendTransport";
-        exit(1);
-    }
-    if (_rtp->RegisterSendPayload("I420", _payloadType, 90000) != 0)
-    {
-        throw "_rtp->RegisterSendPayload";
-        exit(1);
-    }
-
-    if (_rtp->RegisterIncomingVideoCallback(this) != 0)
-    {
-        throw "_rtp->RegisterIncomingVideoCallback";
-        exit(1);
-    }
-
-    if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0)
-    {
-        throw "_rtp->SetRTCPStatus";
-        exit(1);
-    }
-
-    if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0)
-    {
-        throw "_rtp->SetSendBitrate";
-        exit(1);
-    }
-
-
-    // SocketTransport
-    if (_transport->InitializeSendSockets(ipAddr, rtpPort, rtcpPort))
-    {
-        throw "_transport->InitializeSendSockets";
-        exit(1);
-    }
-
-    _isSender = true;
-
-    return (0);
-}
-
-
-
-int32_t
-TestSenderReceiver::SendOutgoingData(const uint32_t timeStamp,
-                                     const uint8_t* payloadData,
-                                     const size_t payloadSize,
-                                     const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
-    return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize));
-}
-
-
-int32_t TestSenderReceiver::SetLoadGenerator(TestLoadGenerator *generator)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _loadGenerator = generator;
-    return(0);
-
-}
-
-void TestSenderReceiver::OnNetworkChanged(const int32_t id,
-                                  const uint32_t minBitrateBps,
-                                  const uint32_t maxBitrateBps,
-                                  const uint8_t fractionLost,
-                                  const uint16_t roundTripTimeMs,
-                                  const uint16_t bwEstimateKbitMin,
-                                  const uint16_t bwEstimateKbitMax)
-{
-    if (_loadGenerator)
-    {
-        _loadGenerator->SetBitrate(maxBitrateBps/1000);
-    }
-
-    if (_sendRecCB)
-    {
-        _sendRecCB->OnOnNetworkChanged(maxBitrateBps,
-            fractionLost,
-            roundTripTimeMs,
-            bwEstimateKbitMin,
-            bwEstimateKbitMax);
-    }
-}
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
deleted file mode 100644
index 49cab586a3..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
+++ /dev/null
@@ -1,153 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/test/channel_transport/udp_transport.h"
-#include "webrtc/typedefs.h"
-
-class TestLoadGenerator;
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-}
-
-using namespace webrtc;
-
-#define MAX_BITRATE_KBPS 50000
-
-
-class SendRecCB
-{
-public:
-    virtual void OnOnNetworkChanged(const uint32_t bitrateTarget,
-        const uint8_t fractionLost,
-        const uint16_t roundTripTimeMs,
-        const uint16_t bwEstimateKbitMin,
-        const uint16_t bwEstimateKbitMax) = 0;
-
-    virtual ~SendRecCB() {};
-};
-
-
-class TestSenderReceiver : public RtpFeedback, public RtpData, public UdpTransportData, public RtpVideoFeedback
-{
-
-public:
-    TestSenderReceiver (void);
-
-    ~TestSenderReceiver (void);
-
-    void SetCallback (SendRecCB *cb) { _sendRecCB = cb; };
-
-    int32_t Start();
-
-    int32_t Stop();
-
-    bool ProcLoop();
-
-    /////////////////////////////////////////////
-    // Receiver methods
-
-    int32_t InitReceiver (const uint16_t rtpPort,
-                          const uint16_t rtcpPort = 0,
-                          const int8_t payloadType = 127);
-
-    int32_t ReceiveBitrateKbps ();
-
-    int32_t SetPacketTimeout(const uint32_t timeoutMS);
-
-    // Inherited from RtpFeedback
-    int32_t OnInitializeDecoder(const int32_t id,
-                                const int8_t payloadType,
-                                const int8_t payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                const uint32_t frequency,
-                                const uint8_t channels,
-                                const uint32_t rate) override {
-      return 0;
-    }
-
-    void OnIncomingSSRCChanged(const int32_t id, const uint32_t SSRC) override {
-    }
-
-    void OnIncomingCSRCChanged(const int32_t id,
-                               const uint32_t CSRC,
-                               const bool added) override {}
-
-    // Inherited from RtpData
-    int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const webrtc::WebRtcRTPHeader* rtpHeader) override;
-
-    // Inherited from UdpTransportData
-    void IncomingRTPPacket(const int8_t* incomingRtpPacket,
-                           const size_t rtpPacketLength,
-                           const int8_t* fromIP,
-                           const uint16_t fromPort) override;
-
-    void IncomingRTCPPacket(const int8_t* incomingRtcpPacket,
-                            const size_t rtcpPacketLength,
-                            const int8_t* fromIP,
-                            const uint16_t fromPort) override;
-
-    /////////////////////////////////
-    // Sender methods
-
-    int32_t InitSender (const uint32_t startBitrateKbps,
-                        const int8_t* ipAddr,
-                        const uint16_t rtpPort,
-                        const uint16_t rtcpPort = 0,
-                        const int8_t payloadType = 127);
-
-    int32_t SendOutgoingData(const uint32_t timeStamp,
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
-
-    int32_t SetLoadGenerator(TestLoadGenerator *generator);
-
-    uint32_t BitrateSent() { return (_rtp->BitrateSent()); };
-
-
-    // Inherited from RtpVideoFeedback
-    virtual void OnReceivedIntraFrameRequest(const int32_t id,
-        const uint8_t message = 0) {};
-
-    virtual void OnNetworkChanged(const int32_t id,
-                                  const uint32_t minBitrateBps,
-                                  const uint32_t maxBitrateBps,
-                                  const uint8_t fractionLost,
-                                  const uint16_t roundTripTimeMs,
-                                  const uint16_t bwEstimateKbitMin,
-                                  const uint16_t bwEstimateKbitMax);
-
-private:
-    RtpRtcp* _rtp;
-    UdpTransport* _transport;
-    webrtc::CriticalSectionWrapper* _critSect;
-    webrtc::EventWrapper *_eventPtr;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _procThread;
-    bool _running;
-    int8_t _payloadType;
-    TestLoadGenerator* _loadGenerator;
-    bool _isSender;
-    bool _isReceiver;
-    SendRecCB * _sendRecCB;
-    size_t _lastBytesReceived;
-    int64_t _lastTime;
-
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/bwe_standalone.gypi b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/bwe_standalone.gypi
deleted file mode 100644
index e45daec77d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/bwe_standalone.gypi
+++ /dev/null
@@ -1,85 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'bwe_standalone',
-      'type': 'executable',
-      'dependencies': [
-        'matlab_plotting',
-        'rtp_rtcp',
-        'udp_transport',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
-      'sources': [
-        'BWEStandAlone/BWEStandAlone.cc',
-        'BWEStandAlone/TestLoadGenerator.cc',
-        'BWEStandAlone/TestLoadGenerator.h',
-        'BWEStandAlone/TestSenderReceiver.cc',
-        'BWEStandAlone/TestSenderReceiver.h',
-      ], # source
-      'conditions': [
-          ['OS=="linux"', {
-              'cflags': [
-                  '-fexceptions', # enable exceptions
-                  ],
-              },
-           ],
-          ],
-    },
-
-    {
-      'target_name': 'matlab_plotting',
-      'type': 'static_library',
-      'dependencies': [
-        'matlab_plotting_include',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
-      'include_dirs': [
-          '/opt/matlab2010a/extern/include',
-          ],
-      'export_dependent_settings': [
-          'matlab_plotting_include',
-          ],
-      'sources': [
-          'BWEStandAlone/MatlabPlot.cc',
-          'BWEStandAlone/MatlabPlot.h',
-          ],
-      'link_settings': {
-          'ldflags' : [
-              '-L/opt/matlab2010a/bin/glnxa64',
-              '-leng',
-              '-lmx',
-              '-Wl,-rpath,/opt/matlab2010a/bin/glnxa64',
-              ],
-          },
-      'defines': [
-          'MATLAB',
-          ],
-      'conditions': [
-          ['OS=="linux"', {
-              'cflags': [
-                  '-fexceptions', # enable exceptions
-                  ],
-              },
-           ],
-          ],
-      },
-
-    {
-      'target_name': 'matlab_plotting_include',
-      'type': 'none',
-      'direct_dependent_settings': {
-          'include_dirs': [
-              'BWEStandAlone',
-              ],
-          },
-      },
-  ],
-}
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 6b4e55df66..1d4d6d04a5 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -13,9 +13,10 @@
 #include <algorithm>
 #include <vector>
 
-using namespace webrtc;
+#include "webrtc/test/null_transport.h"
 
 namespace webrtc {
+
 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
                                       RTPPayloadRegistry* payload_registry,
                                       RtpReceiver* receiver,
@@ -30,39 +31,39 @@ void LoopBackTransport::DropEveryNthPacket(int n) {
   packet_loss_ = n;
 }
 
-int LoopBackTransport::SendPacket(int channel, const void* data, size_t len) {
+bool LoopBackTransport::SendRtp(const uint8_t* data,
+                                size_t len,
+                                const PacketOptions& options) {
   count_++;
   if (packet_loss_ > 0) {
     if ((count_ % packet_loss_) == 0) {
-      return len;
+      return true;
     }
   }
   RTPHeader header;
   rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
   if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
-    return -1;
+    return false;
   }
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
-    return -1;
+    return false;
   }
   receive_statistics_->IncomingPacket(header, len, false);
   if (!rtp_receiver_->IncomingRtpPacket(header,
                                         static_cast<const uint8_t*>(data), len,
                                         payload_specific, true)) {
-    return -1;
+    return false;
   }
-  return len;
+  return true;
 }
 
-int LoopBackTransport::SendRTCPPacket(int channel,
-                                      const void* data,
-                                      size_t len) {
+bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
   if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
-    return -1;
+    return false;
   }
-  return static_cast<int>(len);
+  return true;
 }
 
 int32_t TestRtpReceiver::OnReceivedPayloadData(
@@ -75,14 +76,12 @@ int32_t TestRtpReceiver::OnReceivedPayloadData(
   payload_size_ = payload_size;
   return 0;
 }
-}  // namespace webrtc
 
 class RtpRtcpAPITest : public ::testing::Test {
  protected:
   RtpRtcpAPITest() : fake_clock_(123456) {
     test_csrcs_.push_back(1234);
     test_csrcs_.push_back(2345);
-    test_id = 123;
     test_ssrc_ = 3456;
     test_timestamp_ = 4567;
     test_sequence_number_ = 2345;
@@ -91,17 +90,16 @@ class RtpRtcpAPITest : public ::testing::Test {
 
   void SetUp() override {
     RtpRtcp::Configuration configuration;
-    configuration.id = test_id;
     configuration.audio = true;
     configuration.clock = &fake_clock_;
+    configuration.outgoing_transport = &null_transport_;
     module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
     rtp_payload_registry_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
     rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
-        test_id, &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
+        &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
   }
 
-  int test_id;
   rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
   rtc::scoped_ptr<RtpRtcp> module_;
@@ -110,6 +108,7 @@ class RtpRtcpAPITest : public ::testing::Test {
   uint16_t test_sequence_number_;
   std::vector<uint32_t> test_csrcs_;
   SimulatedClock fake_clock_;
+  test::NullTransport null_transport_;
 };
 
 TEST_F(RtpRtcpAPITest, Basic) {
@@ -125,8 +124,6 @@ TEST_F(RtpRtcpAPITest, Basic) {
 }
 
 TEST_F(RtpRtcpAPITest, MTU) {
-  EXPECT_EQ(-1, module_->SetMaxTransferUnit(10));
-  EXPECT_EQ(-1, module_->SetMaxTransferUnit(IP_PACKET_SIZE + 1));
   EXPECT_EQ(0, module_->SetMaxTransferUnit(1234));
   EXPECT_EQ(1234 - 20 - 8, module_->MaxPayloadLength());
 
@@ -143,9 +140,9 @@ TEST_F(RtpRtcpAPITest, SSRC) {
 }
 
 TEST_F(RtpRtcpAPITest, RTCP) {
-  EXPECT_EQ(kRtcpOff, module_->RTCP());
-  module_->SetRTCPStatus(kRtcpCompound);
-  EXPECT_EQ(kRtcpCompound, module_->RTCP());
+  EXPECT_EQ(RtcpMode::kOff, module_->RTCP());
+  module_->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(RtcpMode::kCompound, module_->RTCP());
 
   EXPECT_EQ(0, module_->SetCNAME("john.doe@test.test"));
 
@@ -174,9 +171,10 @@ TEST_F(RtpRtcpAPITest, RtxSender) {
 TEST_F(RtpRtcpAPITest, RtxReceiver) {
   const uint32_t kRtxSsrc = 1;
   const int kRtxPayloadType = 119;
+  const int kPayloadType = 100;
   EXPECT_FALSE(rtp_payload_registry_->RtxEnabled());
   rtp_payload_registry_->SetRtxSsrc(kRtxSsrc);
-  rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType);
+  rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType, kPayloadType);
   EXPECT_TRUE(rtp_payload_registry_->RtxEnabled());
   RTPHeader rtx_header;
   rtx_header.ssrc = kRtxSsrc;
@@ -188,3 +186,5 @@ TEST_F(RtpRtcpAPITest, RtxReceiver) {
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index 069cdc77df..d8040f7902 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -7,22 +7,25 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/transport.h"
 
 namespace webrtc {
 
 // This class sends all its packet straight to the provided RtpRtcp module.
 // with optional packet loss.
-class LoopBackTransport : public webrtc::Transport {
+class LoopBackTransport : public Transport {
  public:
   LoopBackTransport()
       : count_(0),
@@ -35,8 +38,10 @@ class LoopBackTransport : public webrtc::Transport {
                      RtpReceiver* receiver,
                      ReceiveStatistics* receive_statistics);
   void DropEveryNthPacket(int n);
-  int SendPacket(int channel, const void* data, size_t len) override;
-  int SendRTCPPacket(int channel, const void* data, size_t len) override;
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* data, size_t len) override;
 
  private:
   int count_;
@@ -65,3 +70,4 @@ class TestRtpReceiver : public NullRtpData {
 };
 
 }  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 61923aa447..634969b311 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -15,12 +15,12 @@
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
-using namespace webrtc;
-
+namespace webrtc {
+namespace {
 #define test_rate 64000u
 
 class VerifyingAudioReceiver : public NullRtpData {
@@ -61,11 +61,10 @@ class VerifyingAudioReceiver : public NullRtpData {
 
 class RTPCallback : public NullRtpFeedback {
  public:
-  int32_t OnInitializeDecoder(const int32_t id,
-                              const int8_t payloadType,
+  int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
-                              const uint8_t channels,
+                              const size_t channels,
                               const uint32_t rate) override {
     if (payloadType == 96) {
       EXPECT_EQ(test_rate, rate) <<
@@ -80,7 +79,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
   RtpRtcpAudioTest() : fake_clock(123456) {
     test_CSRC[0] = 1234;
     test_CSRC[2] = 2345;
-    test_id = 123;
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
@@ -104,7 +102,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
         RTPPayloadStrategy::CreateStrategy(true)));
 
     RtpRtcp::Configuration configuration;
-    configuration.id = test_id;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
@@ -113,18 +110,17 @@ class RtpRtcpAudioTest : public ::testing::Test {
 
     module1 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
-        test_id, &fake_clock, audioFeedback, data_receiver1, NULL,
+        &fake_clock, audioFeedback, data_receiver1, NULL,
         rtp_payload_registry1_.get()));
 
-    configuration.id = test_id + 1;
     configuration.receive_statistics = receive_statistics2_.get();
     configuration.outgoing_transport = transport2;
     configuration.audio_messages = audioFeedback;
 
     module2 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
-            test_id + 1, &fake_clock, audioFeedback, data_receiver2, NULL,
-            rtp_payload_registry2_.get()));
+        &fake_clock, audioFeedback, data_receiver2, NULL,
+        rtp_payload_registry2_.get()));
 
     transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
                               rtp_receiver2_.get(), receive_statistics2_.get());
@@ -143,7 +139,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
     delete rtp_callback;
   }
 
-  int test_id;
   RtpRtcp* module1;
   RtpRtcp* module2;
   rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
@@ -170,7 +165,7 @@ TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetStartTimestamp(test_timestamp);
 
   // Test detection at the end of a DTMF tone.
-  //EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
+  // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
 
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
@@ -246,7 +241,7 @@ TEST_F(RtpRtcpAudioTest, RED) {
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(voice_codec.pltype));
   int8_t red = 0;
-  EXPECT_EQ(0, module1->SendREDPayloadType(red));
+  EXPECT_EQ(0, module1->SendREDPayloadType(&red));
   EXPECT_EQ(voice_codec.pltype, red);
   EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
       voice_codec.plname,
@@ -283,7 +278,7 @@ TEST_F(RtpRtcpAudioTest, RED) {
                                          &fragmentation));
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
-  EXPECT_EQ(-1, module1->SendREDPayloadType(red));
+  EXPECT_EQ(-1, module1->SendREDPayloadType(&red));
 }
 
 TEST_F(RtpRtcpAudioTest, DTMF) {
@@ -339,7 +334,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
 
   // Send RTP packets for 16 tones a 160 ms  100ms
   // pause between = 2560ms + 1600ms = 4160ms
-  for (;timeStamp <= 250 * 160; timeStamp += 160) {
+  for (; timeStamp <= 250 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
@@ -347,10 +342,13 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
 
-  for (;timeStamp <= 740 * 160; timeStamp += 160) {
+  for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index 10b561deb5..6c60bf1f6d 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -14,48 +14,48 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
-using namespace webrtc;
+namespace webrtc {
+namespace {
 
 const uint64_t kTestPictureId = 12345678;
+const uint8_t kSliPictureId = 156;
 
 class RtcpCallback : public RtcpIntraFrameObserver {
  public:
   void SetModule(RtpRtcp* module) {
     _rtpRtcpModule = module;
-  };
+  }
   virtual void OnRTCPPacketTimeout(const int32_t id) {
   }
   virtual void OnLipSyncUpdate(const int32_t id,
-                               const int32_t audioVideoOffset) {
-  };
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
-  };
+                               const int32_t audioVideoOffset) {}
+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
   virtual void OnReceivedSLI(uint32_t ssrc,
                              uint8_t pictureId) {
-    EXPECT_EQ(28, pictureId);
-  };
+    EXPECT_EQ(kSliPictureId & 0x3f, pictureId);
+  }
   virtual void OnReceivedRPSI(uint32_t ssrc,
                               uint64_t pictureId) {
     EXPECT_EQ(kTestPictureId, pictureId);
-  };
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {};
+  }
+  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {}
+
  private:
   RtpRtcp* _rtpRtcpModule;
 };
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
-  TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
+  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
-  virtual void OnIncomingSSRCChanged(const int32_t id,
-                                     const uint32_t ssrc) {
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override {
     rtp_rtcp_->SetRemoteSSRC(ssrc);
   }
 
@@ -68,7 +68,6 @@ class RtpRtcpRtcpTest : public ::testing::Test {
   RtpRtcpRtcpTest() : fake_clock(123456) {
     test_csrcs.push_back(1234);
     test_csrcs.push_back(2345);
-    test_id = 123;
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
@@ -86,7 +85,6 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
 
     RtpRtcp::Configuration configuration;
-    configuration.id = test_id;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
@@ -103,11 +101,10 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     rtp_feedback1_.reset(new TestRtpFeedback(module1));
 
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
-        test_id, &fake_clock, NULL, receiver, rtp_feedback1_.get(),
+        &fake_clock, NULL, receiver, rtp_feedback1_.get(),
         rtp_payload_registry1_.get()));
 
     configuration.receive_statistics = receive_statistics2_.get();
-    configuration.id = test_id + 1;
     configuration.outgoing_transport = transport2;
     configuration.intra_frame_callback = myRTCPFeedback2;
 
@@ -116,7 +113,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     rtp_feedback2_.reset(new TestRtpFeedback(module2));
 
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
-        test_id + 1, &fake_clock, NULL, receiver, rtp_feedback2_.get(),
+        &fake_clock, NULL, receiver, rtp_feedback2_.get(),
         rtp_payload_registry2_.get()));
 
     transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
@@ -126,8 +123,8 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     myRTCPFeedback1->SetModule(module1);
     myRTCPFeedback2->SetModule(module2);
 
-    module1->SetRTCPStatus(kRtcpCompound);
-    module2->SetRTCPStatus(kRtcpCompound);
+    module1->SetRTCPStatus(RtcpMode::kCompound);
+    module2->SetRTCPStatus(RtcpMode::kCompound);
 
     module2->SetSSRC(test_ssrc + 1);
     module1->SetSSRC(test_ssrc);
@@ -178,7 +175,6 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     delete receiver;
   }
 
-  int test_id;
   rtc::scoped_ptr<TestRtpFeedback> rtp_feedback1_;
   rtc::scoped_ptr<TestRtpFeedback> rtp_feedback2_;
   rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
@@ -204,7 +200,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
 
 TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {
   EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId));
-  EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(156));
+  EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(kSliPictureId));
 }
 
 TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
@@ -246,103 +242,6 @@ TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
   EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
 }
 
-TEST_F(RtpRtcpRtcpTest, RTCP) {
-  RTCPReportBlock reportBlock;
-  reportBlock.remoteSSRC = 1;
-  reportBlock.sourceSSRC = 2;
-  reportBlock.cumulativeLost = 1;
-  reportBlock.delaySinceLastSR = 2;
-  reportBlock.extendedHighSeqNum = 3;
-  reportBlock.fractionLost= 4;
-  reportBlock.jitter = 5;
-  reportBlock.lastSR = 6;
-
-  // Set report blocks.
-  EXPECT_EQ(0, module1->AddRTCPReportBlock(test_csrcs[0], &reportBlock));
-
-  reportBlock.lastSR= 7;
-  EXPECT_EQ(0, module1->AddRTCPReportBlock(test_csrcs[1], &reportBlock));
-
-  uint32_t name = 't' << 24;
-  name += 'e' << 16;
-  name += 's' << 8;
-  name += 't';
-  EXPECT_EQ(0, module1->SetRTCPApplicationSpecificData(
-      3,
-      name,
-      (const uint8_t *)"test test test test test test test test test"\
-          " test test test test test test test test test test test test test"\
-          " test test test test test test test test test test test test test"\
-          " test test test test test test test test test test test test test"\
-          " test test test test test test test test test test test test ",
-          300));
-
-  // send RTCP packet, triggered by timer
-  fake_clock.AdvanceTimeMilliseconds(7500);
-  module1->Process();
-  fake_clock.AdvanceTimeMilliseconds(100);
-  module2->Process();
-
-  uint32_t receivedNTPsecs = 0;
-  uint32_t receivedNTPfrac = 0;
-  uint32_t RTCPArrivalTimeSecs = 0;
-  uint32_t RTCPArrivalTimeFrac = 0;
-  EXPECT_EQ(0, module2->RemoteNTP(&receivedNTPsecs,
-                                  &receivedNTPfrac,
-                                  &RTCPArrivalTimeSecs,
-                                  &RTCPArrivalTimeFrac,
-                                  NULL));
-
-
-  // get all report blocks
-  std::vector<RTCPReportBlock> report_blocks;
-  EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
-  ASSERT_EQ(1u, report_blocks.size());
-  const RTCPReportBlock& reportBlockReceived = report_blocks[0];
-
-  float secSinceLastReport =
-      static_cast<float>(reportBlockReceived.delaySinceLastSR) / 65536.0f;
-  EXPECT_GE(0.101f, secSinceLastReport);
-  EXPECT_LE(0.100f, secSinceLastReport);
-  EXPECT_EQ(test_sequence_number, reportBlockReceived.extendedHighSeqNum);
-  EXPECT_EQ(0, reportBlockReceived.fractionLost);
-
-  EXPECT_EQ(static_cast<uint32_t>(0),
-            reportBlockReceived.cumulativeLost);
-
-  StreamStatistician *statistician =
-      receive_statistics2_->GetStatistician(reportBlockReceived.sourceSSRC);
-  RtcpStatistics stats;
-  EXPECT_TRUE(statistician->GetStatistics(&stats, true));
-  EXPECT_EQ(0, stats.fraction_lost);
-  EXPECT_EQ((uint32_t)0, stats.cumulative_lost);
-  EXPECT_EQ(test_sequence_number, stats.extended_max_sequence_number);
-  EXPECT_EQ(reportBlockReceived.jitter, stats.jitter);
-
-  int64_t RTT;
-  int64_t avgRTT;
-  int64_t minRTT;
-  int64_t maxRTT;
-
-  // Get RoundTripTime.
-  EXPECT_EQ(0, module1->RTT(test_ssrc + 1, &RTT, &avgRTT, &minRTT, &maxRTT));
-  EXPECT_GE(10, RTT);
-  EXPECT_GE(10, avgRTT);
-  EXPECT_GE(10, minRTT);
-  EXPECT_GE(10, maxRTT);
-
-  // Set report blocks.
-  EXPECT_EQ(0, module1->AddRTCPReportBlock(test_csrcs[0], &reportBlock));
-
-  // Test receive report.
-  EXPECT_EQ(0, module1->SetSendingStatus(false));
-
-  // Send RTCP packet, triggered by timer.
-  fake_clock.AdvanceTimeMilliseconds(5000);
-  module1->Process();
-  module2->Process();
-}
-
 TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
   std::vector<RTCPReportBlock> report_blocks;
 
@@ -367,3 +266,6 @@ TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
   EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
   EXPECT_EQ(0u, report_blocks[0].fractionLost);
 }
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index e28d5ceaf5..16ea540bd5 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -15,9 +15,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
@@ -33,13 +33,11 @@ namespace webrtc {
 class RtpRtcpVideoTest : public ::testing::Test {
  protected:
   RtpRtcpVideoTest()
-      : test_id_(123),
-        rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
+      : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
         test_ssrc_(3456),
         test_timestamp_(4567),
         test_sequence_number_(2345),
-        fake_clock(123456) {
-  }
+        fake_clock(123456) {}
   ~RtpRtcpVideoTest() {}
 
   virtual void SetUp() {
@@ -47,16 +45,15 @@ class RtpRtcpVideoTest : public ::testing::Test {
     receiver_ = new TestRtpReceiver();
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
     RtpRtcp::Configuration configuration;
-    configuration.id = test_id_;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     configuration.outgoing_transport = transport_;
 
     video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
-        test_id_, &fake_clock, receiver_, NULL, &rtp_payload_registry_));
+        &fake_clock, receiver_, NULL, &rtp_payload_registry_));
 
-    video_module_->SetRTCPStatus(kRtcpCompound);
+    video_module_->SetRTCPStatus(RtcpMode::kCompound);
     video_module_->SetSSRC(test_ssrc_);
     rtp_receiver_->SetNACKStatus(kNackRtcp);
     video_module_->SetStorePacketsStatus(true, 600);
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
index 2e8d676e47..6c233bba17 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
@@ -7,8 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
 
-namespace {
+namespace webrtc {
 
 // Maximum number of media packets allowed in this test. The burst mask types
 // are currently defined up to (kMaxMediaPacketsTest, kMaxMediaPacketsTest).
@@ -185,4 +187,5 @@ const float kMaxResidualLossBurstyMask[kNumberCodes] = {
   0.009657f
 };
 
-}  // namespace
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
index a8eafdd27e..b164b7e04c 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -22,43 +22,49 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+#include "webrtc/base/random.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
-
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
-//#define VERBOSE_OUTPUT
+// #define VERBOSE_OUTPUT
 
 namespace webrtc {
+namespace fec_private_tables {
+extern const uint8_t** kPacketMaskBurstyTbl[12];
+}
 namespace test {
+using fec_private_tables::kPacketMaskBurstyTbl;
 
 void ReceivePackets(
     ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
     ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
-    uint32_t numPacketsToDecode, float reorderRate, float duplicateRate) {
+    size_t numPacketsToDecode,
+    float reorderRate,
+    float duplicateRate,
+    Random* random) {
   assert(toDecodeList->empty());
   assert(numPacketsToDecode <= receivedPacketList->size());
 
   ForwardErrorCorrection::ReceivedPacketList::iterator it;
-  for (uint32_t i = 0; i < numPacketsToDecode; i++) {
+  for (size_t i = 0; i < numPacketsToDecode; i++) {
     it = receivedPacketList->begin();
     // Reorder packets.
-    float randomVariable = static_cast<float>(rand()) / RAND_MAX;
+    float randomVariable = random->Rand<float>();
     while (randomVariable < reorderRate) {
       ++it;
       if (it == receivedPacketList->end()) {
         --it;
         break;
       }
-      randomVariable = static_cast<float>(rand()) / RAND_MAX;
+      randomVariable = random->Rand<float>();
     }
     ForwardErrorCorrection::ReceivedPacket* receivedPacket = *it;
     toDecodeList->push_back(receivedPacket);
 
     // Duplicate packets.
-    randomVariable = static_cast<float>(rand()) / RAND_MAX;
+    randomVariable = random->Rand<float>();
     while (randomVariable < duplicateRate) {
       ForwardErrorCorrection::ReceivedPacket* duplicatePacket =
           new ForwardErrorCorrection::ReceivedPacket;
@@ -69,7 +75,7 @@ void ReceivePackets(
       duplicatePacket->pkt->length = receivedPacket->pkt->length;
 
       toDecodeList->push_back(duplicatePacket);
-      randomVariable = static_cast<float>(rand()) / RAND_MAX;
+      randomVariable = random->Rand<float>();
     }
     receivedPacketList->erase(it);
   }
@@ -77,12 +83,8 @@ void ReceivePackets(
 
 TEST(FecTest, FecTest) {
   // TODO(marpan): Split this function into subroutines/helper functions.
-  enum {
-    kMaxNumberMediaPackets = 48
-  };
-  enum {
-    kMaxNumberFecPackets = 48
-  };
+  enum { kMaxNumberMediaPackets = 48 };
+  enum { kMaxNumberFecPackets = 48 };
 
   const uint32_t kNumMaskBytesL0 = 2;
   const uint32_t kNumMaskBytesL1 = 6;
@@ -91,15 +93,12 @@ TEST(FecTest, FecTest) {
   const bool kUseUnequalProtection = true;
 
   // FEC mask types.
-  const FecMaskType kMaskTypes[] = { kFecMaskRandom, kFecMaskBursty };
+  const FecMaskType kMaskTypes[] = {kFecMaskRandom, kFecMaskBursty};
   const int kNumFecMaskTypes = sizeof(kMaskTypes) / sizeof(*kMaskTypes);
 
-  // TODO(pbos): Fix this. Hack to prevent a warning
-  // ('-Wunneeded-internal-declaration') from clang.
-  (void) kPacketMaskBurstyTbl;
-
   // Maximum number of media packets allowed for the mask type.
-  const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets,
+  const uint16_t kMaxMediaPackets[] = {
+      kMaxNumberMediaPackets,
       sizeof(kPacketMaskBurstyTbl) / sizeof(*kPacketMaskBurstyTbl)};
 
   ASSERT_EQ(12, kMaxMediaPackets[1]) << "Max media packets for bursty mode not "
@@ -115,7 +114,7 @@ TEST(FecTest, FecTest) {
 
   ForwardErrorCorrection::Packet* mediaPacket = NULL;
   // Running over only one loss rate to limit execution time.
-  const float lossRate[] = { 0.5f };
+  const float lossRate[] = {0.5f};
   const uint32_t lossRateSize = sizeof(lossRate) / sizeof(*lossRate);
   const float reorderRate = 0.1f;
   const float duplicateRate = 0.1f;
@@ -127,7 +126,7 @@ TEST(FecTest, FecTest) {
   // Seed the random number generator, storing the seed to file in order to
   // reproduce past results.
   const unsigned int randomSeed = static_cast<unsigned int>(time(NULL));
-  srand(randomSeed);
+  Random random(randomSeed);
   std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
   FILE* randomSeedFile = fopen(filename.c_str(), "a");
   fprintf(randomSeedFile, "%u\n", randomSeed);
@@ -135,15 +134,13 @@ TEST(FecTest, FecTest) {
   randomSeedFile = NULL;
 
   uint16_t seqNum = 0;
-  uint32_t timeStamp = static_cast<uint32_t>(rand());
-  const uint32_t ssrc = static_cast<uint32_t>(rand());
+  uint32_t timeStamp = random.Rand<uint32_t>();
+  const uint32_t ssrc = random.Rand(1u, 0xfffffffe);
 
   // Loop over the mask types: random and bursty.
   for (int mask_type_idx = 0; mask_type_idx < kNumFecMaskTypes;
        ++mask_type_idx) {
-
     for (uint32_t lossRateIdx = 0; lossRateIdx < lossRateSize; ++lossRateIdx) {
-
       printf("Loss rate: %.2f, Mask type %d \n", lossRate[lossRateIdx],
              mask_type_idx);
 
@@ -159,14 +156,12 @@ TEST(FecTest, FecTest) {
         for (uint32_t numFecPackets = 1;
              numFecPackets <= numMediaPackets && numFecPackets <= packetMaskMax;
              numFecPackets++) {
-
           // Loop over numImpPackets: usually <= (0.3*numMediaPackets).
           // For this test we check up to ~ (numMediaPackets / 4).
           uint32_t maxNumImpPackets = numMediaPackets / 4 + 1;
           for (uint32_t numImpPackets = 0; numImpPackets <= maxNumImpPackets &&
-                                               numImpPackets <= packetMaskMax;
+                                           numImpPackets <= packetMaskMax;
                numImpPackets++) {
-
             uint8_t protectionFactor =
                 static_cast<uint8_t>(numFecPackets * 255 / numMediaPackets);
 
@@ -181,10 +176,11 @@ TEST(FecTest, FecTest) {
                                           mask_table, packetMask);
 
 #ifdef VERBOSE_OUTPUT
-            printf("%u media packets, %u FEC packets, %u numImpPackets, "
-                   "loss rate = %.2f \n",
-                   numMediaPackets, numFecPackets, numImpPackets,
-                   lossRate[lossRateIdx]);
+            printf(
+                "%u media packets, %u FEC packets, %u numImpPackets, "
+                "loss rate = %.2f \n",
+                numMediaPackets, numFecPackets, numImpPackets,
+                lossRate[lossRateIdx]);
             printf("Packet mask matrix \n");
 #endif
 
@@ -232,16 +228,15 @@ TEST(FecTest, FecTest) {
             for (uint32_t i = 0; i < numMediaPackets; ++i) {
               mediaPacket = new ForwardErrorCorrection::Packet;
               mediaPacketList.push_back(mediaPacket);
-              mediaPacket->length = static_cast<size_t>(
-                  (static_cast<float>(rand()) / RAND_MAX) *
-                  (IP_PACKET_SIZE - 12 - 28 -
-                   ForwardErrorCorrection::PacketOverhead()));
-              if (mediaPacket->length < 12) {
-                mediaPacket->length = 12;
-              }
+              const uint32_t kMinPacketSize = 12;
+              const uint32_t kMaxPacketSize = static_cast<uint32_t>(
+                  IP_PACKET_SIZE - 12 - 28 -
+                  ForwardErrorCorrection::PacketOverhead());
+              mediaPacket->length = random.Rand(kMinPacketSize, kMaxPacketSize);
+
               // Generate random values for the first 2 bytes.
-              mediaPacket->data[0] = static_cast<uint8_t>(rand() % 256);
-              mediaPacket->data[1] = static_cast<uint8_t>(rand() % 256);
+              mediaPacket->data[0] = random.Rand<uint8_t>();
+              mediaPacket->data[1] = random.Rand<uint8_t>();
 
               // The first two bits are assumed to be 10 by the
               // FEC encoder. In fact the FEC decoder will set the
@@ -266,7 +261,7 @@ TEST(FecTest, FecTest) {
               ByteWriter<uint32_t>::WriteBigEndian(&mediaPacket->data[8], ssrc);
               // Generate random values for payload
               for (size_t j = 12; j < mediaPacket->length; ++j) {
-                mediaPacket->data[j] = static_cast<uint8_t>(rand() % 256);
+                mediaPacket->data[j] = random.Rand<uint8_t>();
               }
               seqNum++;
             }
@@ -289,8 +284,7 @@ TEST(FecTest, FecTest) {
             while (mediaPacketListItem != mediaPacketList.end()) {
               mediaPacket = *mediaPacketListItem;
               // We want a value between 0 and 1.
-              const float lossRandomVariable =
-                  (static_cast<float>(rand()) / (RAND_MAX));
+              const float lossRandomVariable = random.Rand<float>();
 
               if (lossRandomVariable >= lossRate[lossRateIdx]) {
                 mediaLossMask[mediaPacketIdx] = 1;
@@ -315,8 +309,7 @@ TEST(FecTest, FecTest) {
             uint32_t fecPacketIdx = 0;
             while (fecPacketListItem != fecPacketList.end()) {
               fecPacket = *fecPacketListItem;
-              const float lossRandomVariable =
-                  (static_cast<float>(rand()) / (RAND_MAX));
+              const float lossRandomVariable = random.Rand<float>();
               if (lossRandomVariable >= lossRate[lossRateIdx]) {
                 fecLossMask[fecPacketIdx] = 1;
                 receivedPacket = new ForwardErrorCorrection::ReceivedPacket;
@@ -387,18 +380,15 @@ TEST(FecTest, FecTest) {
             // For error-checking frame completion.
             bool fecPacketReceived = false;
             while (!receivedPacketList.empty()) {
-              uint32_t numPacketsToDecode = static_cast<uint32_t>(
-                  (static_cast<float>(rand()) / RAND_MAX) *
-                      receivedPacketList.size() + 0.5);
-              if (numPacketsToDecode < 1) {
-                numPacketsToDecode = 1;
-              }
+              size_t numPacketsToDecode = random.Rand(
+                  1u, static_cast<uint32_t>(receivedPacketList.size()));
               ReceivePackets(&toDecodeList, &receivedPacketList,
-                             numPacketsToDecode, reorderRate, duplicateRate);
+                             numPacketsToDecode, reorderRate, duplicateRate,
+                             &random);
 
               if (fecPacketReceived == false) {
                 ForwardErrorCorrection::ReceivedPacketList::iterator
-                toDecodeIt = toDecodeList.begin();
+                    toDecodeIt = toDecodeList.begin();
                 while (toDecodeIt != toDecodeList.end()) {
                   receivedPacket = *toDecodeIt;
                   if (receivedPacket->is_fec) {
@@ -418,11 +408,11 @@ TEST(FecTest, FecTest) {
               if (mediaLossMask[mediaPacketIdx] == 1) {
                 // Should have recovered this packet.
                 ForwardErrorCorrection::RecoveredPacketList::iterator
-                recoveredPacketListItem = recoveredPacketList.begin();
+                    recoveredPacketListItem = recoveredPacketList.begin();
 
-                ASSERT_FALSE(
-                    recoveredPacketListItem == recoveredPacketList.end())
-                  << "Insufficient number of recovered packets.";
+                ASSERT_FALSE(recoveredPacketListItem ==
+                             recoveredPacketList.end())
+                    << "Insufficient number of recovered packets.";
                 mediaPacket = *mediaPacketListItem;
                 ForwardErrorCorrection::RecoveredPacket* recoveredPacket =
                     *recoveredPacketListItem;
@@ -462,7 +452,7 @@ TEST(FecTest, FecTest) {
             // Delete received packets we didn't pass to DecodeFEC(), due to
             // early frame completion.
             ForwardErrorCorrection::ReceivedPacketList::iterator
-            receivedPacketIt = receivedPacketList.begin();
+                receivedPacketIt = receivedPacketList.begin();
             while (receivedPacketIt != receivedPacketList.end()) {
               receivedPacket = *receivedPacketIt;
               delete receivedPacket;
@@ -476,11 +466,11 @@ TEST(FecTest, FecTest) {
             }
             timeStamp += 90000 / 30;
           }  // loop over numImpPackets
-        }  // loop over FecPackets
-      }  // loop over numMediaPackets
+        }    // loop over FecPackets
+      }      // loop over numMediaPackets
       delete[] packetMask;
     }  // loop over loss rates
-  }  // loop over mask types
+  }    // loop over mask types
 
   // Have DecodeFEC free allocated memory.
   fec.ResetState(&recoveredPacketList);
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
index 843a7f77f5..466214c740 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -59,13 +59,6 @@ enum { kMaxNumberMediaPackets = 48 };
 // Maximum number of media packets allowed for each mask type.
 const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets, 12};
 
-// Maximum number of media packets allowed in this test. The burst mask types
-// are currently defined up to (k=12,m=12).
-const int kMaxMediaPacketsTest = 12;
-
-// Maximum number of FEC codes considered in this test.
-const int kNumberCodes = kMaxMediaPacketsTest * (kMaxMediaPacketsTest + 1) / 2;
-
 // Maximum gap size for characterizing the consecutiveness of the loss.
 const int kMaxGapSize = 2 * kMaxMediaPacketsTest;
 
@@ -407,7 +400,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
     // Loop over all loss configurations for the symbol sequence of length
     // |tot_num_packets|. In this version we process up to (k=12, m=12) codes,
     // and get exact expressions for the residual loss.
-    // TODO (marpan): For larger codes, loop over some random sample of loss
+    // TODO(marpan): For larger codes, loop over some random sample of loss
     // configurations, sampling driven by the underlying statistical loss model
     // (importance sampling).
 
@@ -427,7 +420,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
 
       // Map configuration number to a loss state.
       for (int j = 0; j < tot_num_packets; j++) {
-        state[j]=0;  // Received state.
+        state[j] = 0;  // Received state.
         int bit_value = i >> (tot_num_packets - j - 1) & 1;
         if (bit_value == 1) {
           state[j] = 1;  // Lost state.
@@ -860,9 +853,9 @@ TEST_F(FecPacketMaskMetricsTest, FecXorVsRS) {
         EXPECT_GE(kMetricsXorBursty[code_index].average_residual_loss[k],
                   kMetricsReedSolomon[code_index].average_residual_loss[k]);
        }
-      // TODO (marpan): There are some cases (for high loss rates and/or
-      // burst loss models) where XOR is better than RS. Is there some pattern
-      // we can identify and enforce as a constraint?
+       // TODO(marpan): There are some cases (for high loss rates and/or
+       // burst loss models) where XOR is better than RS. Is there some pattern
+       // we can identify and enforce as a constraint?
     }
   }
 }
@@ -874,7 +867,7 @@ TEST_F(FecPacketMaskMetricsTest, FecXorVsRS) {
 TEST_F(FecPacketMaskMetricsTest, FecTrendXorVsRsLossRate) {
   SetLossModels();
   SetCodeParams();
-  // TODO (marpan): Examine this further to see if the condition can be strictly
+  // TODO(marpan): Examine this further to see if the condition can be strictly
   // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
   double scale = 0.90;
   int num_loss_rates = sizeof(kAverageLossRate) /
@@ -898,7 +891,7 @@ TEST_F(FecPacketMaskMetricsTest, FecTrendXorVsRsLossRate) {
                kMetricsXorRandom[code_index].average_residual_loss[k+1];
           EXPECT_GE(diff_rs_xor_random_loss1, scale * diff_rs_xor_random_loss2);
         }
-        // TODO (marpan): Investigate the cases for the bursty mask where
+        // TODO(marpan): Investigate the cases for the bursty mask where
         // this trend is not strictly satisfied.
       }
     }
@@ -937,7 +930,7 @@ TEST_F(FecPacketMaskMetricsTest, FecBehaviorViaProtectionLevelAndLength) {
             EXPECT_LT(
                 kMetricsReedSolomon[code_index2].average_residual_loss[k],
                 kMetricsReedSolomon[code_index1].average_residual_loss[k]);
-            // TODO (marpan): There are some corner cases where this is not
+            // TODO(marpan): There are some corner cases where this is not
             // satisfied with the current packet masks. Look into updating
             // these cases to see if this behavior should/can be satisfied,
             // with overall lower residual loss for those XOR codes.
@@ -963,7 +956,7 @@ TEST_F(FecPacketMaskMetricsTest, FecVarianceBehaviorXorVsRs) {
   SetCodeParams();
   // The condition is not strictly satisfied with the current masks,
   // i.e., for some codes, the variance of XOR may be slightly higher than RS.
-  // TODO (marpan): Examine this further to see if the condition can be strictly
+  // TODO(marpan): Examine this further to see if the condition can be strictly
   // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
   double scale = 0.95;
   for (int code_index = 0; code_index < max_num_codes_; code_index++) {
@@ -998,7 +991,7 @@ TEST_F(FecPacketMaskMetricsTest, FecXorBurstyPerfectRecoveryConsecutiveLoss) {
 // bursty mask type, for random loss models at low loss rates.
 // The XOR codes with bursty mask types are generally better than the one with
 // random mask type, for bursty loss models and/or high loss rates.
-// TODO (marpan): Enable this test when some of the packet masks are updated.
+// TODO(marpan): Enable this test when some of the packet masks are updated.
 // Some isolated cases of the codes don't pass this currently.
 /*
 TEST_F(FecPacketMaskMetricsTest, FecXorRandomVsBursty) {
diff --git a/media/webrtc/trunk/webrtc/modules/utility/BUILD.gn b/media/webrtc/trunk/webrtc/modules/utility/BUILD.gn
index 4503be698b..6704cd6d9a 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/utility/BUILD.gn
@@ -10,12 +10,12 @@ import("../../build/webrtc.gni")
 
 source_set("utility") {
   sources = [
-    "interface/audio_frame_operations.h",
-    "interface/file_player.h",
-    "interface/file_recorder.h",
-    "interface/helpers_android.h",
-    "interface/process_thread.h",
-    "interface/rtp_dump.h",
+    "include/audio_frame_operations.h",
+    "include/file_player.h",
+    "include/file_recorder.h",
+    "include/helpers_android.h",
+    "include/jvm_android.h",
+    "include/process_thread.h",
     "source/audio_frame_operations.cc",
     "source/coder.cc",
     "source/coder.h",
@@ -24,10 +24,9 @@ source_set("utility") {
     "source/file_recorder_impl.cc",
     "source/file_recorder_impl.h",
     "source/helpers_android.cc",
+    "source/jvm_android.cc",
     "source/process_thread_impl.cc",
     "source/process_thread_impl.h",
-    "source/rtp_dump_impl.cc",
-    "source/rtp_dump_impl.h",
   ]
 
   configs += [ "../..:common_config" ]
diff --git a/media/webrtc/trunk/webrtc/modules/utility/OWNERS b/media/webrtc/trunk/webrtc/modules/utility/OWNERS
index 347d278614..65cb70c9b9 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/utility/OWNERS
@@ -1,4 +1,9 @@
 asapersson@webrtc.org
 perkj@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/audio_frame_operations.h b/media/webrtc/trunk/webrtc/modules/utility/include/audio_frame_operations.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/utility/interface/audio_frame_operations.h
rename to media/webrtc/trunk/webrtc/modules/utility/include/audio_frame_operations.h
index f439dacbcf..1551d86894 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/audio_frame_operations.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/audio_frame_operations.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
-#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
 
 #include "webrtc/typedefs.h"
 
@@ -26,7 +26,7 @@ class AudioFrameOperations {
   // operation, meaning src_audio and dst_audio must point to different
   // buffers. It is the caller's responsibility to ensure that |dst_audio| is
   // sufficiently large.
-  static void MonoToStereo(const int16_t* src_audio, int samples_per_channel,
+  static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
                            int16_t* dst_audio);
   // |frame.num_channels_| will be updated. This version checks for sufficient
   // buffer size and that |num_channels_| is mono.
@@ -35,7 +35,7 @@ class AudioFrameOperations {
   // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
   // operation, meaning |src_audio| and |dst_audio| may point to the same
   // buffer.
-  static void StereoToMono(const int16_t* src_audio, int samples_per_channel,
+  static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
                            int16_t* dst_audio);
   // |frame.num_channels_| will be updated. This version checks that
   // |num_channels_| is stereo.
@@ -55,4 +55,4 @@ class AudioFrameOperations {
 
 }  // namespace webrtc
 
-#endif  // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/file_player.h b/media/webrtc/trunk/webrtc/modules/utility/include/file_player.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/utility/interface/file_player.h
rename to media/webrtc/trunk/webrtc/modules/utility/include/file_player.h
index d812deb09e..4ca134a669 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/file_player.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/file_player.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 class FileCallback;
@@ -38,7 +38,7 @@ public:
     // channel).
     virtual int Get10msAudioFromFile(
         int16_t* outBuffer,
-        int& lengthInSamples,
+        size_t& lengthInSamples,
         int frequencyInHz) = 0;
 
     // Register callback for receiving file playing notifications.
@@ -93,18 +93,19 @@ public:
     virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const
     {return -1;}
 
-    virtual int32_t GetVideoFromFile(I420VideoFrame& /*videoFrame*/)
-    { return -1;}
+    virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; }
 
     // Same as GetVideoFromFile(). videoFrame will have the resolution specified
     // by the width outWidth and height outHeight in pixels.
-    virtual int32_t GetVideoFromFile(I420VideoFrame& /*videoFrame*/,
+    virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/,
                                      const uint32_t /*outWidth*/,
-                                     const uint32_t /*outHeight*/)
-    {return -1;}
+                                     const uint32_t /*outHeight*/) {
+      return -1;
+    }
+
 protected:
     virtual ~FilePlayer() {}
 
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/file_recorder.h b/media/webrtc/trunk/webrtc/modules/utility/include/file_recorder.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/utility/interface/file_recorder.h
rename to media/webrtc/trunk/webrtc/modules/utility/include/file_recorder.h
index f0ceccb4c0..09ed8ae350 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/file_recorder.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/file_recorder.h
@@ -8,17 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
 
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -40,14 +39,12 @@ public:
     virtual int32_t StartRecordingAudioFile(
         const char* fileName,
         const CodecInst& codecInst,
-        uint32_t notification,
-        ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
+        uint32_t notification) = 0;
 
     virtual int32_t StartRecordingAudioFile(
         OutStream& destStream,
         const CodecInst& codecInst,
-        uint32_t notification,
-        ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
+        uint32_t notification) = 0;
 
     // Stop recording.
     // Note: this API is for both audio and video.
@@ -74,16 +71,14 @@ public:
         const char* fileName,
         const CodecInst& audioCodecInst,
         const VideoCodec& videoCodecInst,
-        ACMAMRPackingFormat amrFormat = AMRFileStorage,
         bool videoOnly = false) = 0;
 
     // Record the video frame in videoFrame to AVI file.
-    virtual int32_t RecordVideoToFile(
-        const I420VideoFrame& videoFrame) = 0;
+    virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) = 0;
 
 protected:
     virtual ~FileRecorder() {}
 
 };
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/helpers_android.h b/media/webrtc/trunk/webrtc/modules/utility/include/helpers_android.h
similarity index 66%
rename from media/webrtc/trunk/webrtc/modules/utility/interface/helpers_android.h
rename to media/webrtc/trunk/webrtc/modules/utility/include/helpers_android.h
index 3424e28ef9..2840ca965e 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/helpers_android.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/helpers_android.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
 
 #include <jni.h>
 #include <string>
 
 // Abort the process if |jni| has a Java exception pending.
 // TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h.
-#define CHECK_EXCEPTION(jni)    \
-  CHECK(!jni->ExceptionCheck()) \
+#define CHECK_EXCEPTION(jni)        \
+  RTC_CHECK(!jni->ExceptionCheck()) \
       << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
 
 namespace webrtc {
@@ -25,13 +25,21 @@ namespace webrtc {
 // Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
 JNIEnv* GetEnv(JavaVM* jvm);
 
-// JNIEnv-helper methods that wraps the API which uses the JNI interface
-// pointer (JNIEnv*). It allows us to CHECK success and that no Java exception
-// is thrown while calling the method.
-jmethodID GetMethodID (
-    JNIEnv* jni, jclass c, const std::string& name, const char* signature);
+// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
+// because the alternative (of silently passing a 32-bit pointer to a vararg
+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
+jlong PointerTojlong(void* ptr);
 
-jclass FindClass(JNIEnv* jni, const std::string& name);
+// JNIEnv-helper methods that wraps the API which uses the JNI interface
+// pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java
+// exception is thrown while calling the method.
+jmethodID GetMethodID(
+    JNIEnv* jni, jclass c, const char* name, const char* signature);
+
+jmethodID GetStaticMethodID(
+    JNIEnv* jni, jclass c, const char* name, const char* signature);
+
+jclass FindClass(JNIEnv* jni, const char* name);
 
 jobject NewGlobalRef(JNIEnv* jni, jobject o);
 
@@ -76,4 +84,4 @@ class ScopedGlobalRef {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/include/helpers_ios.h b/media/webrtc/trunk/webrtc/modules/utility/include/helpers_ios.h
new file mode 100644
index 0000000000..a5a07ace17
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/helpers_ios.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+
+#if defined(WEBRTC_IOS)
+
+#include <string>
+
+namespace webrtc {
+namespace ios {
+
+bool CheckAndLogError(BOOL success, NSError* error);
+
+std::string StdStringFromNSString(NSString* nsString);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Returns [NSThread currentThread] description as string.
+// Example: <NSThread: 0x170066d80>{number = 1, name = main}
+std::string GetCurrentThreadDescription();
+
+std::string GetAudioSessionCategory();
+
+// Returns the current name of the operating system.
+std::string GetSystemName();
+
+// Returns the current version of the operating system.
+std::string GetSystemVersion();
+
+// Returns the version of the operating system as a floating point value.
+float GetSystemVersionAsFloat();
+
+// Returns the device type.
+// Examples: ”iPhone” and ”iPod touch”.
+std::string GetDeviceType();
+
+// Returns a more detailed device name.
+// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus".
+std::string GetDeviceName();
+
+}  // namespace ios
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_IOS)
+
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/include/jvm_android.h b/media/webrtc/trunk/webrtc/modules/utility/include/jvm_android.h
new file mode 100644
index 0000000000..f527dff632
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/jvm_android.h
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+
+#include <jni.h>
+#include <string>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+// The JNI interface pointer (JNIEnv) is valid only in the current thread.
+// Should another thread need to access the Java VM, it must first call
+// AttachCurrentThread() to attach itself to the VM and obtain a JNI interface
+// pointer. The native thread remains attached to the VM until it calls
+// DetachCurrentThread() to detach.
+class AttachCurrentThreadIfNeeded {
+ public:
+  AttachCurrentThreadIfNeeded();
+  ~AttachCurrentThreadIfNeeded();
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  bool attached_;
+};
+
+// This class is created by the NativeRegistration class and is used to wrap
+// the actual Java object handle (jobject) on which we can call methods from
+// C++ in to Java. See example in JVM for more details.
+// TODO(henrika): extend support for type of function calls.
+class GlobalRef {
+ public:
+  GlobalRef(JNIEnv* jni, jobject object);
+  ~GlobalRef();
+
+  jboolean CallBooleanMethod(jmethodID methodID, ...);
+  jint CallIntMethod(jmethodID methodID, ...);
+  void CallVoidMethod(jmethodID methodID, ...);
+
+ private:
+  JNIEnv* const jni_;
+  const jobject j_object_;
+};
+
+// Wraps the jclass object on which we can call GetMethodId() functions to
+// query method IDs.
+class JavaClass {
+ public:
+  JavaClass(JNIEnv* jni, jclass clazz) : jni_(jni), j_class_(clazz) {}
+  ~JavaClass() {}
+
+  jmethodID GetMethodId(const char* name, const char* signature);
+  jmethodID GetStaticMethodId(const char* name, const char* signature);
+  jobject CallStaticObjectMethod(jmethodID methodID, ...);
+
+ protected:
+  JNIEnv* const jni_;
+  jclass const j_class_;
+};
+
+// Adds support of the NewObject factory method to the JavaClass class.
+// See example in JVM for more details on how to use it.
+class NativeRegistration : public JavaClass {
+ public:
+  NativeRegistration(JNIEnv* jni, jclass clazz);
+  ~NativeRegistration();
+
+  rtc::scoped_ptr<GlobalRef> NewObject(
+      const char* name, const char* signature, ...);
+
+ private:
+  JNIEnv* const jni_;
+};
+
+// This class is created by the JVM class and is used to expose methods that
+// needs the JNI interface pointer but its main purpose is to create a
+// NativeRegistration object given name of a Java class and a list of native
+// methods. See example in JVM for more details.
+class JNIEnvironment {
+ public:
+  explicit JNIEnvironment(JNIEnv* jni);
+  ~JNIEnvironment();
+
+  // Registers native methods with the Java class specified by |name|.
+  // Note that the class name must be one of the names in the static
+  // |loaded_classes| array defined in jvm_android.cc.
+  // This method must be called on the construction thread.
+  rtc::scoped_ptr<NativeRegistration> RegisterNatives(
+      const char* name, const JNINativeMethod *methods, int num_methods);
+
+  // Converts from Java string to std::string.
+  // This method must be called on the construction thread.
+  std::string JavaToStdString(const jstring& j_string);
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  JNIEnv* const jni_;
+};
+
+// Main class for working with Java from C++ using JNI in WebRTC.
+//
+// Example usage:
+//
+//   // At initialization (e.g. in JNI_OnLoad), call JVM::Initialize.
+//   JNIEnv* jni = ::base::android::AttachCurrentThread();
+//   JavaVM* jvm = NULL;
+//   jni->GetJavaVM(&jvm);
+//   jobject context = ::base::android::GetApplicationContext();
+//   webrtc::JVM::Initialize(jvm, context);
+//
+//   // Header (.h) file of example class called User.
+//   rtc::scoped_ptr<JNIEnvironment> env;
+//   rtc::scoped_ptr<NativeRegistration> reg;
+//   rtc::scoped_ptr<GlobalRef> obj;
+//
+//   // Construction (in .cc file) of User class.
+//   User::User() {
+//     // Calling thread must be attached to the JVM.
+//     env = JVM::GetInstance()->environment();
+//     reg = env->RegisterNatives("org/webrtc/WebRtcTest", ,);
+//     obj = reg->NewObject("<init>", ,);
+//   }
+//
+//   // Each User method can now use |reg| and |obj| and call Java functions
+//   // in WebRtcTest.java, e.g. boolean init() {}.
+//   bool User::Foo() {
+//     jmethodID id = reg->GetMethodId("init", "()Z");
+//     return obj->CallBooleanMethod(id);
+//   }
+//
+//   // And finally, e.g. in JNI_OnUnLoad, call JVM::Uninitialize.
+//   JVM::Uninitialize();
+class JVM {
+ public:
+  // Stores global handles to the Java VM interface and the application context.
+  // Should be called once on a thread that is attached to the JVM.
+  static void Initialize(JavaVM* jvm, jobject context);
+  // Clears handles stored in Initialize(). Must be called on same thread as
+  // Initialize().
+  static void Uninitialize();
+  // Gives access to the global Java VM interface pointer, which then can be
+  // used to create a valid JNIEnvironment object or to get a JavaClass object.
+  static JVM* GetInstance();
+
+  // Creates a JNIEnvironment object.
+  // This method returns a NULL pointer if AttachCurrentThread() has not been
+  // called successfully. Use the AttachCurrentThreadIfNeeded class if needed.
+  rtc::scoped_ptr<JNIEnvironment> environment();
+
+  // Returns a JavaClass object given class |name|.
+  // Note that the class name must be one of the names in the static
+  // |loaded_classes| array defined in jvm_android.cc.
+  // This method must be called on the construction thread.
+  JavaClass GetClass(const char* name);
+
+  // TODO(henrika): can we make these private?
+  JavaVM* jvm() const { return jvm_; }
+  jobject context() const { return context_; }
+
+ protected:
+  JVM(JavaVM* jvm, jobject context);
+  ~JVM();
+
+ private:
+  JNIEnv* jni() const { return GetEnv(jvm_); }
+
+  rtc::ThreadChecker thread_checker_;
+  JavaVM* const jvm_;
+  jobject context_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/mock/mock_process_thread.h b/media/webrtc/trunk/webrtc/modules/utility/include/mock/mock_process_thread.h
similarity index 81%
rename from media/webrtc/trunk/webrtc/modules/utility/interface/mock/mock_process_thread.h
rename to media/webrtc/trunk/webrtc/modules/utility/include/mock/mock_process_thread.h
index fd108a8354..56d92f4527 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/mock/mock_process_thread.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
 
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 
@@ -35,4 +35,4 @@ class MockProcessThread : public ProcessThread {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/process_thread.h b/media/webrtc/trunk/webrtc/modules/utility/include/process_thread.h
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/utility/interface/process_thread.h
rename to media/webrtc/trunk/webrtc/modules/utility/include/process_thread.h
index 0e84506f1a..285a5ea587 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/process_thread.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/include/process_thread.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
 
 #include "webrtc/typedefs.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -29,7 +29,7 @@ class ProcessThread {
  public:
   virtual ~ProcessThread();
 
-  static rtc::scoped_ptr<ProcessThread> Create();
+  static rtc::scoped_ptr<ProcessThread> Create(const char* thread_name);
 
   // Starts the worker thread.  Must be called from the construction thread.
   virtual void Start() = 0;
@@ -63,4 +63,4 @@ class ProcessThread {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/interface/rtp_dump.h b/media/webrtc/trunk/webrtc/modules/utility/interface/rtp_dump.h
deleted file mode 100644
index df45ae209d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/utility/interface/rtp_dump.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file implements a class that writes a stream of RTP and RTCP packets
-// to a file according to the format specified by rtpplay. See
-// http://www.cs.columbia.edu/irt/software/rtptools/.
-// Notes: supported platforms are Windows, Linux and Mac OSX
-
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
-
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class RtpDump
-{
-public:
-    // Factory method.
-    static RtpDump* CreateRtpDump();
-
-    // Delete function. Destructor disabled.
-    static void DestroyRtpDump(RtpDump* object);
-
-    // Open the file fileNameUTF8 for writing RTP/RTCP packets.
-    // Note: this API also adds the rtpplay header.
-    virtual int32_t Start(const char* fileNameUTF8) = 0;
-
-    // Close the existing file. No more packets will be recorded.
-    virtual int32_t Stop() = 0;
-
-    // Return true if a file is open for recording RTP/RTCP packets.
-    virtual bool IsActive() const = 0;
-
-    // Writes the RTP/RTCP packet in packet with length packetLength in bytes.
-    // Note: packet should contain the RTP/RTCP part of the packet. I.e. the
-    // first bytes of packet should be the RTP/RTCP header.
-    virtual int32_t DumpPacket(const uint8_t* packet,
-                               size_t packetLength) = 0;
-
-protected:
-    virtual ~RtpDump();
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations.cc b/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations.cc
index e3b0010476..fe09d7972f 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations.cc
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 
 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
-                                        int samples_per_channel,
+                                        size_t samples_per_channel,
                                         int16_t* dst_audio) {
-  for (int i = 0; i < samples_per_channel; i++) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     dst_audio[2 * i] = src_audio[i];
     dst_audio[2 * i + 1] = src_audio[i];
   }
@@ -41,9 +41,9 @@ int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
 }
 
 void AudioFrameOperations::StereoToMono(const int16_t* src_audio,
-                                        int samples_per_channel,
+                                        size_t samples_per_channel,
                                         int16_t* dst_audio) {
-  for (int i = 0; i < samples_per_channel; i++) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
     dst_audio[i] = (src_audio[2 * i] + src_audio[2 * i + 1]) >> 1;
   }
 }
@@ -62,7 +62,7 @@ int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
   if (frame->num_channels_ != 2) return;
 
-  for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     int16_t temp_data = frame->data_[i];
     frame->data_[i] = frame->data_[i + 1];
     frame->data_[i + 1] = temp_data;
@@ -79,7 +79,7 @@ int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
     return -1;
   }
 
-  for (int i = 0; i < frame.samples_per_channel_; i++) {
+  for (size_t i = 0; i < frame.samples_per_channel_; i++) {
     frame.data_[2 * i] =
         static_cast<int16_t>(left * frame.data_[2 * i]);
     frame.data_[2 * i + 1] =
@@ -92,7 +92,7 @@ int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
   int32_t temp_data = 0;
 
   // Ensure that the output result is saturated [-32768, +32767].
-  for (int i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
+  for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
        i++) {
     temp_data = static_cast<int32_t>(scale * frame.data_[i]);
     if (temp_data < -32768) {
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
index f4d881cf87..fff8f4407b 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
@@ -10,8 +10,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 namespace {
@@ -28,14 +28,14 @@ class AudioFrameOperationsTest : public ::testing::Test {
 };
 
 void SetFrameData(AudioFrame* frame, int16_t left, int16_t right) {
-  for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     frame->data_[i] = left;
     frame->data_[i + 1] = right;
   }
 }
 
 void SetFrameData(AudioFrame* frame, int16_t data) {
-  for (int i = 0; i < frame->samples_per_channel_; i++) {
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
     frame->data_[i] = data;
   }
 }
@@ -45,7 +45,7 @@ void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
   EXPECT_EQ(frame1.samples_per_channel_,
             frame2.samples_per_channel_);
 
-  for (int i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_;
+  for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_;
       i++) {
     EXPECT_EQ(frame1.data_[i], frame2.data_[i]);
   }
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/coder.cc b/media/webrtc/trunk/webrtc/modules/utility/source/coder.cc
index dc0799a245..18b690dc67 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/coder.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/coder.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/source/coder.h"
 
 namespace webrtc {
@@ -29,8 +29,7 @@ AudioCoder::~AudioCoder()
 {
 }
 
-int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
-                                   ACMAMRPackingFormat amrFormat)
+int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst)
 {
     if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
     {
@@ -39,8 +38,7 @@ int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
     return 0;
 }
 
-int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
-                                   ACMAMRPackingFormat amrFormat)
+int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst)
 {
     if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
     {
@@ -85,7 +83,7 @@ int32_t AudioCoder::Encode(const AudioFrame& audio,
     AudioFrame audioFrame;
     audioFrame.CopyFrom(audio);
     audioFrame.timestamp_ = _encodeTimestamp;
-    _encodeTimestamp += audioFrame.samples_per_channel_;
+    _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_);
 
     // For any codec with a frame size that is longer than 10 ms the encoded
     // length in bytes should be zero until a a full frame has been encoded.
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/coder.h b/media/webrtc/trunk/webrtc/modules/utility/source/coder.h
index 57eada18cc..abfa87efe1 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/coder.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/coder.h
@@ -13,7 +13,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -25,13 +25,9 @@ public:
     AudioCoder(uint32_t instanceID);
     ~AudioCoder();
 
-    int32_t SetEncodeCodec(
-        const CodecInst& codecInst,
-        ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
+    int32_t SetEncodeCodec(const CodecInst& codecInst);
 
-    int32_t SetDecodeCodec(
-        const CodecInst& codecInst,
-        ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
+    int32_t SetDecodeCodec(const CodecInst& codecInst);
 
     int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
                    const int8_t* incomingPayload, size_t payloadLength);
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.cc b/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.cc
index 01a6e1636e..cb8fb49921 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/utility/source/file_player_impl.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 FilePlayer* FilePlayer::CreateFilePlayer(uint32_t instanceID,
@@ -95,7 +95,7 @@ int32_t FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const
 
 int32_t FilePlayerImpl::Get10msAudioFromFile(
     int16_t* outBuffer,
-    int& lengthInSamples,
+    size_t& lengthInSamples,
     int frequencyInHz)
 {
     if(_codec.plfreq == 0)
@@ -127,8 +127,7 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(
             return 0;
         }
         // One sample is two bytes.
-        unresampledAudioFrame.samples_per_channel_ =
-            (uint16_t)lengthInBytes >> 1;
+        unresampledAudioFrame.samples_per_channel_ = lengthInBytes >> 1;
 
     } else {
         // Decode will generate 10 ms of audio data. PlayoutAudioData(..)
@@ -156,14 +155,14 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(
         }
     }
 
-    int outLen = 0;
+    size_t outLen = 0;
     if(_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_,
                                 frequencyInHz, 1))
     {
         LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";
 
         // New sampling frequency. Update state.
-        outLen = frequencyInHz / 100;
+        outLen = static_cast<size_t>(frequencyInHz / 100);
         memset(outBuffer, 0, outLen * sizeof(int16_t));
         return 0;
     }
@@ -177,7 +176,7 @@ int32_t FilePlayerImpl::Get10msAudioFromFile(
 
     if(_scaling != 1.0)
     {
-        for (int i = 0;i < outLen; i++)
+        for (size_t i = 0;i < outLen; i++)
         {
             outBuffer[i] = (int16_t)(outBuffer[i] * _scaling);
         }
@@ -390,7 +389,7 @@ int32_t FilePlayerImpl::SetUpAudioDecoder()
         return -1;
     }
     if( STR_CASE_CMP(_codec.plname, "L16") != 0 &&
-        _audioDecoder.SetDecodeCodec(_codec,AMRFileStorage) == -1)
+        _audioDecoder.SetDecodeCodec(_codec) == -1)
     {
         LOG(LS_WARNING) << "SetUpAudioDecoder() codec " << _codec.plname
                         << " not supported.";
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.h b/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.h
index f81e7101ea..beb6379ff0 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.h
@@ -14,12 +14,12 @@
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/source/coder.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -31,7 +31,7 @@ public:
 
     virtual int Get10msAudioFromFile(
         int16_t* outBuffer,
-        int& lengthInSamples,
+        size_t& lengthInSamples,
         int frequencyInHz);
     virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
     virtual int32_t StartPlayingFile(
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/file_player_unittests.cc b/media/webrtc/trunk/webrtc/modules/utility/source/file_player_unittests.cc
index 7ce9d47296..58471e5e8d 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/file_player_unittests.cc
@@ -10,7 +10,7 @@
 
 // Unit tests for FilePlayer.
 
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/utility/include/file_player.h"
 
 #include <stdio.h>
 #include <string>
@@ -61,12 +61,12 @@ class FilePlayerTest : public ::testing::Test {
     rtc::Md5Digest checksum;
     for (int i = 0; i < output_length_ms / 10; ++i) {
       int16_t out[10 * kSampleRateHz / 1000] = {0};
-      int num_samples;
+      size_t num_samples;
       EXPECT_EQ(0,
                 player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz));
       checksum.Update(out, num_samples * sizeof(out[0]));
       if (FLAGS_file_player_output) {
-        ASSERT_EQ(static_cast<size_t>(num_samples),
+        ASSERT_EQ(num_samples,
                   fwrite(out, sizeof(out[0]), num_samples, output_file_));
       }
     }
@@ -81,7 +81,12 @@ class FilePlayerTest : public ::testing::Test {
   FILE* output_file_;
 };
 
-TEST_F(FilePlayerTest, PlayWavPcmuFile) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_PlayWavPcmuFile DISABLED_PlayWavPcmuFile
+#else
+#define MAYBE_PlayWavPcmuFile PlayWavPcmuFile
+#endif
+TEST_F(FilePlayerTest, MAYBE_PlayWavPcmuFile) {
   const std::string kFileName =
       test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
@@ -92,7 +97,12 @@ TEST_F(FilePlayerTest, PlayWavPcmuFile) {
   PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
 }
 
-TEST_F(FilePlayerTest, PlayWavPcm16File) {
+#if defined(WEBRTC_IOS)
+#define MAYBE_PlayWavPcm16File DISABLED_PlayWavPcm16File
+#else
+#define MAYBE_PlayWavPcm16File PlayWavPcm16File
+#endif
+TEST_F(FilePlayerTest, MAYBE_PlayWavPcm16File) {
   const std::string kFileName =
       test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
   // The file is longer than this, but keeping the output shorter limits the
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.cc b/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.cc
index 0a2c9a0886..88b20eeac2 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -10,9 +10,9 @@
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/utility/source/file_recorder_impl.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 FileRecorder* FileRecorder::CreateFileRecorder(uint32_t instanceID,
@@ -32,7 +32,6 @@ FileRecorderImpl::FileRecorderImpl(uint32_t instanceID,
       _fileFormat(fileFormat),
       _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
       codec_info_(),
-      _amrFormat(AMRFileStorage),
       _audioBuffer(),
       _audioEncoder(instanceID),
       _audioResampler()
@@ -62,16 +61,13 @@ int32_t FileRecorderImpl::RegisterModuleFileCallback(
 int32_t FileRecorderImpl::StartRecordingAudioFile(
     const char* fileName,
     const CodecInst& codecInst,
-    uint32_t notificationTimeMs,
-    ACMAMRPackingFormat amrFormat)
+    uint32_t notificationTimeMs)
 {
     if(_moduleFile == NULL)
     {
         return -1;
     }
     codec_info_ = codecInst;
-    _amrFormat = amrFormat;
-
     int32_t retVal = 0;
     retVal =_moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
                                                  codecInst,
@@ -97,12 +93,9 @@ int32_t FileRecorderImpl::StartRecordingAudioFile(
 int32_t FileRecorderImpl::StartRecordingAudioFile(
     OutStream& destStream,
     const CodecInst& codecInst,
-    uint32_t notificationTimeMs,
-    ACMAMRPackingFormat amrFormat)
+    uint32_t notificationTimeMs)
 {
     codec_info_ = codecInst;
-    _amrFormat = amrFormat;
-
     int32_t retVal = _moduleFile->StartRecordingAudioStream(
         destStream,
         _fileFormat,
@@ -156,7 +149,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
         tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
         tempAudioFrame.samples_per_channel_ =
           incomingAudioFrame.samples_per_channel_;
-        for (uint16_t i = 0;
+        for (size_t i = 0;
              i < (incomingAudioFrame.samples_per_channel_); i++)
         {
             // Sample value is the average of left and right buffer rounded to
@@ -174,7 +167,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
         tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
         tempAudioFrame.samples_per_channel_ =
           incomingAudioFrame.samples_per_channel_;
-        for (uint16_t i = 0;
+        for (size_t i = 0;
              i < (incomingAudioFrame.samples_per_channel_); i++)
         {
             // Duplicate sample to both channels
@@ -210,7 +203,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
             return -1;
         }
     } else {
-        int outLen = 0;
+        size_t outLen = 0;
         _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
                                       codec_info_.plfreq,
                                       ptrAudioFrame->num_channels_);
@@ -227,11 +220,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
     // will be available. Wait until then.
     if (encodedLenInBytes)
     {
-        uint16_t msOfData =
-            ptrAudioFrame->samples_per_channel_ /
-            uint16_t(ptrAudioFrame->sample_rate_hz_ / 1000);
-        if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes, msOfData,
-                                  playoutTS) == -1)
+        if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1)
         {
             return -1;
         }
@@ -244,7 +233,7 @@ int32_t FileRecorderImpl::SetUpAudioEncoder()
     if (_fileFormat == kFileFormatPreencodedFile ||
         STR_CASE_CMP(codec_info_.plname, "L16") != 0)
     {
-        if(_audioEncoder.SetEncodeCodec(codec_info_,_amrFormat) == -1)
+        if(_audioEncoder.SetEncodeCodec(codec_info_) == -1)
         {
             LOG(LS_ERROR) << "SetUpAudioEncoder() codec "
                           << codec_info_.plname << " not supported.";
@@ -264,11 +253,8 @@ int32_t FileRecorderImpl::codec_info(CodecInst& codecInst) const
     return 0;
 }
 
-int32_t FileRecorderImpl::WriteEncodedAudioData(
-    const int8_t* audioBuffer,
-    size_t bufferLength,
-    uint16_t /*millisecondsOfData*/,
-    const TickTime* /*playoutTS*/)
+int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
+                                                size_t bufferLength)
 {
     return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.h b/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.h
index 776654b8e7..697d759375 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/file_recorder_impl.h
@@ -17,17 +17,17 @@
 
 #include <list>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/modules/utility/source/coder.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -50,13 +50,11 @@ public:
     virtual int32_t StartRecordingAudioFile(
         const char* fileName,
         const CodecInst& codecInst,
-        uint32_t notificationTimeMs,
-        ACMAMRPackingFormat amrFormat = AMRFileStorage);
+        uint32_t notificationTimeMs) override;
     virtual int32_t StartRecordingAudioFile(
         OutStream& destStream,
         const CodecInst& codecInst,
-        uint32_t notificationTimeMs,
-        ACMAMRPackingFormat amrFormat = AMRFileStorage);
+        uint32_t notificationTimeMs) override;
     virtual int32_t StopRecording();
     virtual bool IsRecording() const;
     virtual int32_t codec_info(CodecInst& codecInst) const;
@@ -67,22 +65,17 @@ public:
         const char* fileName,
         const CodecInst& audioCodecInst,
         const VideoCodec& videoCodecInst,
-        ACMAMRPackingFormat amrFormat = AMRFileStorage,
-        bool videoOnly = false)
+        bool videoOnly = false) override
     {
         return -1;
     }
-    virtual int32_t RecordVideoToFile(const I420VideoFrame& videoFrame)
-    {
+    virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) {
         return -1;
     }
 
 protected:
-    virtual int32_t WriteEncodedAudioData(
-        const int8_t* audioBuffer,
-        size_t bufferLength,
-        uint16_t millisecondsOfData,
-        const TickTime* playoutTS);
+    int32_t WriteEncodedAudioData(const int8_t* audioBuffer,
+                                  size_t bufferLength);
 
     int32_t SetUpAudioEncoder();
 
@@ -92,8 +85,6 @@ protected:
 
 private:
     CodecInst codec_info_;
-    ACMAMRPackingFormat _amrFormat;
-
     int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
     AudioCoder _audioEncoder;
     Resampler _audioResampler;
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/helpers_android.cc b/media/webrtc/trunk/webrtc/modules/utility/source/helpers_android.cc
index f429db123b..aea35f8d5a 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/helpers_android.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/helpers_android.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 
 #include <android/log.h>
 #include <assert.h>
@@ -25,32 +25,55 @@ namespace webrtc {
 JNIEnv* GetEnv(JavaVM* jvm) {
   void* env = NULL;
   jint status = jvm->GetEnv(&env, JNI_VERSION_1_6);
-  CHECK(((env != NULL) && (status == JNI_OK)) ||
-        ((env == NULL) && (status == JNI_EDETACHED)))
+  RTC_CHECK(((env != NULL) && (status == JNI_OK)) ||
+            ((env == NULL) && (status == JNI_EDETACHED)))
       << "Unexpected GetEnv return: " << status << ":" << env;
   return reinterpret_cast<JNIEnv*>(env);
 }
 
+// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
+// because the alternative (of silently passing a 32-bit pointer to a vararg
+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
+jlong PointerTojlong(void* ptr) {
+  static_assert(sizeof(intptr_t) <= sizeof(jlong),
+                "Time to rethink the use of jlongs");
+  // Going through intptr_t to be obvious about the definedness of the
+  // conversion from pointer to integral type.  intptr_t to jlong is a standard
+  // widening by the static_assert above.
+  jlong ret = reinterpret_cast<intptr_t>(ptr);
+  RTC_DCHECK(reinterpret_cast<void*>(ret) == ptr);
+  return ret;
+}
+
 jmethodID GetMethodID (
-    JNIEnv* jni, jclass c, const std::string& name, const char* signature) {
-  jmethodID m = jni->GetMethodID(c, name.c_str(), signature);
+    JNIEnv* jni, jclass c, const char* name, const char* signature) {
+  jmethodID m = jni->GetMethodID(c, name, signature);
   CHECK_EXCEPTION(jni) << "Error during GetMethodID: " << name << ", "
                        << signature;
-  CHECK(m) << name << ", " << signature;
+  RTC_CHECK(m) << name << ", " << signature;
   return m;
 }
 
-jclass FindClass(JNIEnv* jni, const std::string& name) {
-  jclass c = jni->FindClass(name.c_str());
+jmethodID GetStaticMethodID (
+    JNIEnv* jni, jclass c, const char* name, const char* signature) {
+  jmethodID m = jni->GetStaticMethodID(c, name, signature);
+  CHECK_EXCEPTION(jni) << "Error during GetStaticMethodID: " << name << ", "
+                       << signature;
+  RTC_CHECK(m) << name << ", " << signature;
+  return m;
+}
+
+jclass FindClass(JNIEnv* jni, const char* name) {
+  jclass c = jni->FindClass(name);
   CHECK_EXCEPTION(jni) << "Error during FindClass: " << name;
-  CHECK(c) << name;
+  RTC_CHECK(c) << name;
   return c;
 }
 
 jobject NewGlobalRef(JNIEnv* jni, jobject o) {
   jobject ret = jni->NewGlobalRef(o);
   CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
-  CHECK(ret);
+  RTC_CHECK(ret);
   return ret;
 }
 
@@ -62,8 +85,9 @@ void DeleteGlobalRef(JNIEnv* jni, jobject o) {
 std::string GetThreadId() {
   char buf[21];  // Big enough to hold a kuint64max plus terminating NULL.
   int thread_id = gettid();
-  CHECK_LT(snprintf(buf, sizeof(buf), "%i", thread_id),
-      static_cast<int>(sizeof(buf))) << "Thread id is bigger than uint64??";
+  RTC_CHECK_LT(snprintf(buf, sizeof(buf), "%i", thread_id),
+               static_cast<int>(sizeof(buf)))
+      << "Thread id is bigger than uint64??";
   return std::string(buf);
 }
 
@@ -81,7 +105,7 @@ AttachThreadScoped::AttachThreadScoped(JavaVM* jvm)
     ALOGD("Attaching thread to JVM%s", GetThreadInfo().c_str());
     jint res = jvm->AttachCurrentThread(&env_, NULL);
     attached_ = (res == JNI_OK);
-    CHECK(attached_) << "AttachCurrentThread failed: " << res;
+    RTC_CHECK(attached_) << "AttachCurrentThread failed: " << res;
   }
 }
 
@@ -89,8 +113,8 @@ AttachThreadScoped::~AttachThreadScoped() {
   if (attached_) {
     ALOGD("Detaching thread from JVM%s", GetThreadInfo().c_str());
     jint res = jvm_->DetachCurrentThread();
-    CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
-    CHECK(!GetEnv(jvm_));
+    RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
+    RTC_CHECK(!GetEnv(jvm_));
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/helpers_ios.mm b/media/webrtc/trunk/webrtc/modules/utility/source/helpers_ios.mm
new file mode 100644
index 0000000000..2d0ac098c1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/helpers_ios.mm
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if defined(WEBRTC_IOS)
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+#import <sys/sysctl.h>
+#import <UIKit/UIKit.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
+
+namespace webrtc {
+namespace ios {
+
+// TODO(henrika): move to shared location.
+// See https://code.google.com/p/webrtc/issues/detail?id=4773 for details.
+NSString* NSStringFromStdString(const std::string& stdString) {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+std::string StdStringFromNSString(NSString* nsString) {
+  NSData* charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char*>([charData bytes]),
+                     [charData length]);
+}
+
+bool CheckAndLogError(BOOL success, NSError* error) {
+  if (!success) {
+    NSString* msg =
+        [NSString stringWithFormat:@"Error: %ld, %@, %@", (long)error.code,
+                                   error.localizedDescription,
+                                   error.localizedFailureReason];
+    LOG(LS_ERROR) << StdStringFromNSString(msg);
+    return false;
+  }
+  return true;
+}
+
+// TODO(henrika): see if it is possible to move to GetThreadName in
+// platform_thread.h and base it on pthread methods instead.
+std::string GetCurrentThreadDescription() {
+  NSString* name = [NSString stringWithFormat:@"%@", [NSThread currentThread]];
+  return StdStringFromNSString(name);
+}
+
+std::string GetAudioSessionCategory() {
+  NSString* category = [[AVAudioSession sharedInstance] category];
+  return StdStringFromNSString(category);
+}
+
+std::string GetSystemName() {
+  NSString* osName = [[UIDevice currentDevice] systemName];
+  return StdStringFromNSString(osName);
+}
+
+std::string GetSystemVersion() {
+  NSString* osVersion = [[UIDevice currentDevice] systemVersion];
+  return StdStringFromNSString(osVersion);
+}
+
+float GetSystemVersionAsFloat() {
+  NSString* osVersion = [[UIDevice currentDevice] systemVersion];
+  return osVersion.floatValue;
+}
+
+std::string GetDeviceType() {
+  NSString* deviceModel = [[UIDevice currentDevice] model];
+  return StdStringFromNSString(deviceModel);
+}
+
+std::string GetDeviceName() {
+  size_t size;
+  sysctlbyname("hw.machine", NULL, &size, NULL, 0);
+  rtc::scoped_ptr<char[]> machine;
+  machine.reset(new char[size]);
+  sysctlbyname("hw.machine", machine.get(), &size, NULL, 0);
+  std::string raw_name(machine.get());
+  if (!raw_name.compare("iPhone1,1"))
+    return std::string("iPhone 1G");
+  if (!raw_name.compare("iPhone1,2"))
+    return std::string("iPhone 3G");
+  if (!raw_name.compare("iPhone2,1"))
+    return std::string("iPhone 3GS");
+  if (!raw_name.compare("iPhone3,1"))
+    return std::string("iPhone 4");
+  if (!raw_name.compare("iPhone3,3"))
+    return std::string("Verizon iPhone 4");
+  if (!raw_name.compare("iPhone4,1"))
+    return std::string("iPhone 4S");
+  if (!raw_name.compare("iPhone5,1"))
+    return std::string("iPhone 5 (GSM)");
+  if (!raw_name.compare("iPhone5,2"))
+    return std::string("iPhone 5 (GSM+CDMA)");
+  if (!raw_name.compare("iPhone5,3"))
+    return std::string("iPhone 5c (GSM)");
+  if (!raw_name.compare("iPhone5,4"))
+    return std::string("iPhone 5c (GSM+CDMA)");
+  if (!raw_name.compare("iPhone6,1"))
+    return std::string("iPhone 5s (GSM)");
+  if (!raw_name.compare("iPhone6,2"))
+    return std::string("iPhone 5s (GSM+CDMA)");
+  if (!raw_name.compare("iPhone7,1"))
+    return std::string("iPhone 6 Plus");
+  if (!raw_name.compare("iPhone7,2"))
+    return std::string("iPhone 6");
+  if (!raw_name.compare("iPhone8,1"))
+    return std::string("iPhone 6s");
+  if (!raw_name.compare("iPhone8,2"))
+    return std::string("iPhone 6s Plus");
+  if (!raw_name.compare("iPod1,1"))
+    return std::string("iPod Touch 1G");
+  if (!raw_name.compare("iPod2,1"))
+    return std::string("iPod Touch 2G");
+  if (!raw_name.compare("iPod3,1"))
+    return std::string("iPod Touch 3G");
+  if (!raw_name.compare("iPod4,1"))
+    return std::string("iPod Touch 4G");
+  if (!raw_name.compare("iPod5,1"))
+    return std::string("iPod Touch 5G");
+  if (!raw_name.compare("iPad1,1"))
+    return std::string("iPad");
+  if (!raw_name.compare("iPad2,1"))
+    return std::string("iPad 2 (WiFi)");
+  if (!raw_name.compare("iPad2,2"))
+    return std::string("iPad 2 (GSM)");
+  if (!raw_name.compare("iPad2,3"))
+    return std::string("iPad 2 (CDMA)");
+  if (!raw_name.compare("iPad2,4"))
+    return std::string("iPad 2 (WiFi)");
+  if (!raw_name.compare("iPad2,5"))
+    return std::string("iPad Mini (WiFi)");
+  if (!raw_name.compare("iPad2,6"))
+    return std::string("iPad Mini (GSM)");
+  if (!raw_name.compare("iPad2,7"))
+    return std::string("iPad Mini (GSM+CDMA)");
+  if (!raw_name.compare("iPad3,1"))
+    return std::string("iPad 3 (WiFi)");
+  if (!raw_name.compare("iPad3,2"))
+    return std::string("iPad 3 (GSM+CDMA)");
+  if (!raw_name.compare("iPad3,3"))
+    return std::string("iPad 3 (GSM)");
+  if (!raw_name.compare("iPad3,4"))
+    return std::string("iPad 4 (WiFi)");
+  if (!raw_name.compare("iPad3,5"))
+    return std::string("iPad 4 (GSM)");
+  if (!raw_name.compare("iPad3,6"))
+    return std::string("iPad 4 (GSM+CDMA)");
+  if (!raw_name.compare("iPad4,1"))
+    return std::string("iPad Air (WiFi)");
+  if (!raw_name.compare("iPad4,2"))
+    return std::string("iPad Air (Cellular)");
+  if (!raw_name.compare("iPad4,4"))
+    return std::string("iPad mini 2G (WiFi)");
+  if (!raw_name.compare("iPad4,5"))
+    return std::string("iPad mini 2G (Cellular)");
+  if (!raw_name.compare("i386"))
+    return std::string("Simulator");
+  if (!raw_name.compare("x86_64"))
+    return std::string("Simulator");
+  LOG(LS_WARNING) << "Failed to find device name (" << raw_name << ")";
+  return raw_name;
+}
+
+}  // namespace ios
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_IOS)
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/jvm_android.cc b/media/webrtc/trunk/webrtc/modules/utility/source/jvm_android.cc
new file mode 100644
index 0000000000..2a9b2b469c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/jvm_android.cc
@@ -0,0 +1,263 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <android/log.h>
+
+#include "webrtc/modules/utility/include/jvm_android.h"
+
+#include "webrtc/base/checks.h"
+#include "AndroidJNIWrapper.h"
+
+#define TAG "JVM"
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+
+namespace webrtc {
+
+JVM* g_jvm;
+
+// TODO(henrika): add more clases here if needed.
+struct {
+  const char* name;
+  jclass clazz;
+} loaded_classes[] = {
+  {"org/webrtc/voiceengine/BuildInfo", nullptr},
+  {"org/webrtc/voiceengine/WebRtcAudioManager", nullptr},
+  {"org/webrtc/voiceengine/WebRtcAudioRecord", nullptr},
+  {"org/webrtc/voiceengine/WebRtcAudioTrack", nullptr},
+};
+
+// Android's FindClass() is trickier than usual because the app-specific
+// ClassLoader is not consulted when there is no app-specific frame on the
+// stack.  Consequently, we only look up all classes once in native WebRTC.
+// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
+void LoadClasses(JNIEnv* jni) {
+  for (auto& c : loaded_classes) {
+    jclass globalRef = jsjni_GetGlobalClassRef(c.name);
+    RTC_CHECK(globalRef) << c.name;
+    c.clazz = globalRef;
+  }
+}
+
+void FreeClassReferences(JNIEnv* jni) {
+  for (auto& c : loaded_classes) {
+    jni->DeleteGlobalRef(c.clazz);
+    c.clazz = nullptr;
+  }
+}
+
+jclass LookUpClass(const char* name) {
+  for (auto& c : loaded_classes) {
+    if (strcmp(c.name, name) == 0)
+      return c.clazz;
+  }
+  RTC_CHECK(false) << "Unable to find class in lookup table";
+  return 0;
+}
+
+// AttachCurrentThreadIfNeeded implementation.
+AttachCurrentThreadIfNeeded::AttachCurrentThreadIfNeeded()
+    : attached_(false) {
+  ALOGD("AttachCurrentThreadIfNeeded::ctor%s", GetThreadInfo().c_str());
+  JavaVM* jvm = JVM::GetInstance()->jvm();
+  RTC_CHECK(jvm);
+  JNIEnv* jni = GetEnv(jvm);
+  if (!jni) {
+    ALOGD("Attaching thread to JVM");
+    JNIEnv* env = nullptr;
+    jint ret = jvm->AttachCurrentThread(&env, nullptr);
+    attached_ = (ret == JNI_OK);
+  }
+}
+
+AttachCurrentThreadIfNeeded::~AttachCurrentThreadIfNeeded() {
+  ALOGD("AttachCurrentThreadIfNeeded::dtor%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (attached_) {
+    ALOGD("Detaching thread from JVM");
+    jint res = JVM::GetInstance()->jvm()->DetachCurrentThread();
+    RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
+  }
+}
+
+// GlobalRef implementation.
+GlobalRef::GlobalRef(JNIEnv* jni, jobject object)
+    : jni_(jni), j_object_(NewGlobalRef(jni, object)) {
+  ALOGD("GlobalRef::ctor%s", GetThreadInfo().c_str());
+}
+
+GlobalRef::~GlobalRef() {
+  ALOGD("GlobalRef::dtor%s", GetThreadInfo().c_str());
+  DeleteGlobalRef(jni_, j_object_);
+}
+
+jboolean GlobalRef::CallBooleanMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jboolean res = jni_->CallBooleanMethodV(j_object_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallBooleanMethod";
+  va_end(args);
+  return res;
+}
+
+jint GlobalRef::CallIntMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jint res = jni_->CallIntMethodV(j_object_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallIntMethod";
+  va_end(args);
+  return res;
+}
+
+void GlobalRef::CallVoidMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jni_->CallVoidMethodV(j_object_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallVoidMethod";
+  va_end(args);
+}
+
+// NativeRegistration implementation.
+NativeRegistration::NativeRegistration(JNIEnv* jni, jclass clazz)
+    : JavaClass(jni, clazz), jni_(jni) {
+  ALOGD("NativeRegistration::ctor%s", GetThreadInfo().c_str());
+}
+
+NativeRegistration::~NativeRegistration() {
+  ALOGD("NativeRegistration::dtor%s", GetThreadInfo().c_str());
+  jni_->UnregisterNatives(j_class_);
+  CHECK_EXCEPTION(jni_) << "Error during UnregisterNatives";
+}
+
+rtc::scoped_ptr<GlobalRef> NativeRegistration::NewObject(
+    const char* name, const char* signature, ...) {
+  ALOGD("NativeRegistration::NewObject%s", GetThreadInfo().c_str());
+  va_list args;
+  va_start(args, signature);
+  jobject obj = jni_->NewObjectV(j_class_,
+                                 GetMethodID(jni_, j_class_, name, signature),
+                                 args);
+  CHECK_EXCEPTION(jni_) << "Error during NewObjectV";
+  va_end(args);
+  return rtc::scoped_ptr<GlobalRef>(new GlobalRef(jni_, obj));
+}
+
+// JavaClass implementation.
+jmethodID JavaClass::GetMethodId(
+    const char* name, const char* signature) {
+  return GetMethodID(jni_, j_class_, name, signature);
+}
+
+jmethodID JavaClass::GetStaticMethodId(
+    const char* name, const char* signature) {
+  return GetStaticMethodID(jni_, j_class_, name, signature);
+}
+
+jobject JavaClass::CallStaticObjectMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jobject res = jni_->CallStaticObjectMethod(j_class_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallStaticObjectMethod";
+  return res;
+}
+
+// JNIEnvironment implementation.
+JNIEnvironment::JNIEnvironment(JNIEnv* jni) : jni_(jni) {
+  ALOGD("JNIEnvironment::ctor%s", GetThreadInfo().c_str());
+}
+
+JNIEnvironment::~JNIEnvironment() {
+  ALOGD("JNIEnvironment::dtor%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+rtc::scoped_ptr<NativeRegistration> JNIEnvironment::RegisterNatives(
+    const char* name, const JNINativeMethod *methods, int num_methods) {
+  ALOGD("JNIEnvironment::RegisterNatives(%s)", name);
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  jclass clazz = LookUpClass(name);
+  jni_->RegisterNatives(clazz, methods, num_methods);
+  CHECK_EXCEPTION(jni_) << "Error during RegisterNatives";
+  return rtc::scoped_ptr<NativeRegistration>(
+      new NativeRegistration(jni_, clazz));
+}
+
+std::string JNIEnvironment::JavaToStdString(const jstring& j_string) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  const char* jchars = jni_->GetStringUTFChars(j_string, nullptr);
+  CHECK_EXCEPTION(jni_);
+  const int size = jni_->GetStringUTFLength(j_string);
+  CHECK_EXCEPTION(jni_);
+  std::string ret(jchars, size);
+  jni_->ReleaseStringUTFChars(j_string, jchars);
+  CHECK_EXCEPTION(jni_);
+  return ret;
+}
+
+// static
+void JVM::Initialize(JavaVM* jvm, jobject context) {
+  ALOGD("JVM::Initialize%s", GetThreadInfo().c_str());
+  if (g_jvm) {
+    return;
+  }
+  g_jvm = new JVM(jvm, context);
+}
+
+// static
+void JVM::Uninitialize() {
+  ALOGD("JVM::Uninitialize%s", GetThreadInfo().c_str());
+  RTC_DCHECK(g_jvm);
+  delete g_jvm;
+  g_jvm = nullptr;
+}
+
+// static
+JVM* JVM::GetInstance() {
+  RTC_DCHECK(g_jvm);
+  return g_jvm;
+}
+
+JVM::JVM(JavaVM* jvm, jobject context)
+    : jvm_(jvm) {
+  ALOGD("JVM::JVM%s", GetThreadInfo().c_str());
+  RTC_CHECK(jni()) << "AttachCurrentThread() must be called on this thread.";
+  context_ = NewGlobalRef(jni(), context);
+  LoadClasses(jni());
+}
+
+JVM::~JVM() {
+  ALOGD("JVM::~JVM%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  FreeClassReferences(jni());
+  DeleteGlobalRef(jni(), context_);
+}
+
+rtc::scoped_ptr<JNIEnvironment> JVM::environment() {
+  ALOGD("JVM::environment%s", GetThreadInfo().c_str());
+  // The JNIEnv is used for thread-local storage. For this reason, we cannot
+  // share a JNIEnv between threads. If a piece of code has no other way to get
+  // its JNIEnv, we should share the JavaVM, and use GetEnv to discover the
+  // thread's JNIEnv. (Assuming it has one, if not, use AttachCurrentThread).
+  // See // http://developer.android.com/training/articles/perf-jni.html.
+  JNIEnv* jni = GetEnv(jvm_);
+  if (!jni) {
+    ALOGE("AttachCurrentThread() has not been called on this thread.");
+    return rtc::scoped_ptr<JNIEnvironment>();
+  }
+  return rtc::scoped_ptr<JNIEnvironment>(new JNIEnvironment(jni));
+}
+
+JavaClass JVM::GetClass(const char* name) {
+  ALOGD("JVM::GetClass(%s)%s", name, GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return JavaClass(jni(), LookUpClass(name));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.cc b/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.cc
index e80d32dd7a..c5f6bf9277 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.cc
@@ -11,9 +11,9 @@
 #include "webrtc/modules/utility/source/process_thread_impl.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
@@ -25,12 +25,9 @@ const int64_t kCallProcessImmediately = -1;
 
 int64_t GetNextCallbackTime(Module* module, int64_t time_now) {
   int64_t interval = module->TimeUntilNextProcess();
-  // Currently some implementations erroneously return error codes from
-  // TimeUntilNextProcess(). So, as is, we correct that and log an error.
   if (interval < 0) {
-    LOG(LS_ERROR) << "TimeUntilNextProcess returned an invalid value "
-                  << interval;
-    interval = 0;
+    // Falling behind, we should call the callback now.
+    return time_now;
   }
   return time_now + interval;
 }
@@ -39,18 +36,20 @@ int64_t GetNextCallbackTime(Module* module, int64_t time_now) {
 ProcessThread::~ProcessThread() {}
 
 // static
-rtc::scoped_ptr<ProcessThread> ProcessThread::Create() {
-  return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl()).Pass();
+rtc::scoped_ptr<ProcessThread> ProcessThread::Create(
+    const char* thread_name) {
+  return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
 }
 
-ProcessThreadImpl::ProcessThreadImpl()
-    : wake_up_(EventWrapper::Create()), stop_(false) {
-}
+ProcessThreadImpl::ProcessThreadImpl(const char* thread_name)
+    : wake_up_(EventWrapper::Create()),
+      stop_(false),
+      thread_name_(thread_name) {}
 
 ProcessThreadImpl::~ProcessThreadImpl() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!thread_.get());
-  DCHECK(!stop_);
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!thread_.get());
+  RTC_DCHECK(!stop_);
 
   while (!queue_.empty()) {
     delete queue_.front();
@@ -59,12 +58,12 @@ ProcessThreadImpl::~ProcessThreadImpl() {
 }
 
 void ProcessThreadImpl::Start() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!thread_.get());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!thread_.get());
   if (thread_.get())
     return;
 
-  DCHECK(!stop_);
+  RTC_DCHECK(!stop_);
 
   {
     // TODO(tommi): Since DeRegisterModule is currently being called from
@@ -76,13 +75,13 @@ void ProcessThreadImpl::Start() {
       m.module->ProcessThreadAttached(this);
   }
 
-  thread_ = ThreadWrapper::CreateThread(
-      &ProcessThreadImpl::Run, this, "ProcessThread");
-  CHECK(thread_->Start());
+  thread_.reset(
+      new rtc::PlatformThread(&ProcessThreadImpl::Run, this, thread_name_));
+  thread_->Start();
 }
 
 void ProcessThreadImpl::Stop() {
-  DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if(!thread_.get())
     return;
 
@@ -93,7 +92,7 @@ void ProcessThreadImpl::Stop() {
 
   wake_up_->Set();
 
-  CHECK(thread_->Stop());
+  thread_->Stop();
   stop_ = false;
 
   // TODO(tommi): Since DeRegisterModule is currently being called from
@@ -130,15 +129,15 @@ void ProcessThreadImpl::PostTask(rtc::scoped_ptr<ProcessTask> task) {
 }
 
 void ProcessThreadImpl::RegisterModule(Module* module) {
-  //  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(module);
+  //  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(module);
 
 #if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
   {
     // Catch programmer error.
     rtc::CritScope lock(&lock_);
     for (const ModuleCallback& mc : modules_)
-      DCHECK(mc.module != module);
+      RTC_DCHECK(mc.module != module);
   }
 #endif
 
@@ -162,7 +161,7 @@ void ProcessThreadImpl::RegisterModule(Module* module) {
 void ProcessThreadImpl::DeRegisterModule(Module* module) {
   // Allowed to be called on any thread.
   // TODO(tommi): Disallow this ^^^
-  DCHECK(module);
+  RTC_DCHECK(module);
 
   {
     rtc::CritScope lock(&lock_);
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.h b/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.h
index 1fd2bf3adc..1c0a0cdfdd 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl.h
@@ -15,17 +15,17 @@
 #include <queue>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class ProcessThreadImpl : public ProcessThread {
  public:
-  ProcessThreadImpl();
+  explicit ProcessThreadImpl(const char* thread_name);
   ~ProcessThreadImpl() override;
 
   void Start() override;
@@ -70,12 +70,14 @@ class ProcessThreadImpl : public ProcessThread {
 
   rtc::ThreadChecker thread_checker_;
   const rtc::scoped_ptr<EventWrapper> wake_up_;
-  rtc::scoped_ptr<ThreadWrapper> thread_;
+  // TODO(pbos): Remove scoped_ptr and stop recreating the thread.
+  rtc::scoped_ptr<rtc::PlatformThread> thread_;
 
   ModuleList modules_;
   // TODO(tommi): Support delayed tasks.
   std::queue<ProcessTask*> queue_;
   bool stop_;
+  const char* thread_name_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl_unittest.cc
index cd1f956dd5..0b35fad7d2 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/utility/source/process_thread_impl_unittest.cc
@@ -8,11 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <utility>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/utility/source/process_thread_impl.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -52,13 +54,13 @@ ACTION_P(SetTimestamp, ptr) {
 }
 
 TEST(ProcessThreadImpl, StartStop) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   thread.Start();
   thread.Stop();
 }
 
 TEST(ProcessThreadImpl, MultipleStartStop) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   for (int i = 0; i < 5; ++i) {
     thread.Start();
     thread.Stop();
@@ -67,7 +69,7 @@ TEST(ProcessThreadImpl, MultipleStartStop) {
 
 // Verifies that we get at least call back to Process() on the worker thread.
 TEST(ProcessThreadImpl, ProcessCall) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   thread.Start();
 
   rtc::scoped_ptr<EventWrapper> event(EventWrapper::Create());
@@ -89,7 +91,7 @@ TEST(ProcessThreadImpl, ProcessCall) {
 // Same as ProcessCall except the module is registered before the
 // call to Start().
 TEST(ProcessThreadImpl, ProcessCall2) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   rtc::scoped_ptr<EventWrapper> event(EventWrapper::Create());
 
   MockModule module;
@@ -111,7 +113,7 @@ TEST(ProcessThreadImpl, ProcessCall2) {
 // Tests setting up a module for callbacks and then unregister that module.
 // After unregistration, we should not receive any further callbacks.
 TEST(ProcessThreadImpl, Deregister) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   rtc::scoped_ptr<EventWrapper> event(EventWrapper::Create());
 
   int process_count = 0;
@@ -146,7 +148,7 @@ TEST(ProcessThreadImpl, Deregister) {
 // time.  There's some variance of timing built into it to reduce chance of
 // flakiness on bots.
 void ProcessCallAfterAFewMs(int64_t milliseconds) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   thread.Start();
 
   rtc::scoped_ptr<EventWrapper> event(EventWrapper::Create());
@@ -211,7 +213,7 @@ TEST(ProcessThreadImpl, DISABLED_ProcessCallAfter200ms) {
 // build bots.
 // TODO(tommi): Fix.
 TEST(ProcessThreadImpl, DISABLED_Process50Times) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   thread.Start();
 
   rtc::scoped_ptr<EventWrapper> event(EventWrapper::Create());
@@ -244,15 +246,16 @@ TEST(ProcessThreadImpl, DISABLED_Process50Times) {
 // Tests that we can wake up the worker thread to give us a callback right
 // away when we know the thread is sleeping.
 TEST(ProcessThreadImpl, WakeUp) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   thread.Start();
 
   rtc::scoped_ptr<EventWrapper> started(EventWrapper::Create());
   rtc::scoped_ptr<EventWrapper> called(EventWrapper::Create());
 
   MockModule module;
-  int64_t start_time = 0;
-  int64_t called_time = 0;
+  int64_t start_time;
+  int64_t called_time;
+
   // Ask for a callback after 1000ms.
   // TimeUntilNextProcess will be called twice.
   // The first time we use it to get the thread into a waiting state.
@@ -281,8 +284,6 @@ TEST(ProcessThreadImpl, WakeUp) {
   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
   thread.Stop();
 
-  ASSERT_GT(start_time, 0);
-  ASSERT_GT(called_time, 0);
   EXPECT_GE(called_time, start_time);
   uint32_t diff = called_time - start_time;
   // We should have been called back much quicker than 1sec.
@@ -292,11 +293,11 @@ TEST(ProcessThreadImpl, WakeUp) {
 // Tests that we can post a task that gets run straight away on the worker
 // thread.
 TEST(ProcessThreadImpl, PostTask) {
-  ProcessThreadImpl thread;
+  ProcessThreadImpl thread("ProcessThread");
   rtc::scoped_ptr<EventWrapper> task_ran(EventWrapper::Create());
   rtc::scoped_ptr<RaiseEventTask> task(new RaiseEventTask(task_ran.get()));
   thread.Start();
-  thread.PostTask(task.Pass());
+  thread.PostTask(std::move(task));
   EXPECT_EQ(kEventSignaled, task_ran->Wait(100));
   thread.Stop();
 }
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.cc b/media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.cc
deleted file mode 100644
index 1aaed0db2f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.cc
+++ /dev/null
@@ -1,249 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/utility/source/rtp_dump_impl.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <limits>
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-#if defined(_WIN32)
-#include <Windows.h>
-#include <mmsystem.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_BSD)
-#include <string.h>
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-#if (defined(_DEBUG) && defined(_WIN32))
-#define DEBUG_PRINT(expr)   OutputDebugString(##expr)
-#define DEBUG_PRINTP(expr, p)   \
-{                               \
-    char msg[128];              \
-    sprintf(msg, ##expr, p);    \
-    OutputDebugString(msg);     \
-}
-#else
-#define DEBUG_PRINT(expr)    ((void)0)
-#define DEBUG_PRINTP(expr,p) ((void)0)
-#endif  // defined(_DEBUG) && defined(_WIN32)
-
-namespace webrtc {
-const char RTPFILE_VERSION[] = "1.0";
-const uint32_t MAX_UWORD32 = 0xffffffff;
-
-// This stucture is specified in the rtpdump documentation.
-// This struct corresponds to RD_packet_t in
-// http://www.cs.columbia.edu/irt/software/rtptools/
-typedef struct
-{
-    // Length of packet, including this header (may be smaller than plen if not
-    // whole packet recorded).
-    uint16_t length;
-    // Actual header+payload length for RTP, 0 for RTCP.
-    uint16_t plen;
-    // Milliseconds since the start of recording.
-    uint32_t offset;
-} RtpDumpPacketHeader;
-
-RtpDump* RtpDump::CreateRtpDump()
-{
-    return new RtpDumpImpl();
-}
-
-void RtpDump::DestroyRtpDump(RtpDump* object)
-{
-    delete object;
-}
-
-RtpDumpImpl::RtpDumpImpl()
-    : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
-      _file(*FileWrapper::Create()),
-      _startTime(0)
-{
-}
-
-RtpDump::~RtpDump()
-{
-}
-
-RtpDumpImpl::~RtpDumpImpl()
-{
-    _file.Flush();
-    _file.CloseFile();
-    delete &_file;
-    delete _critSect;
-}
-
-int32_t RtpDumpImpl::Start(const char* fileNameUTF8)
-{
-
-    if (fileNameUTF8 == NULL)
-    {
-        return -1;
-    }
-
-    CriticalSectionScoped lock(_critSect);
-    _file.Flush();
-    _file.CloseFile();
-    if (_file.OpenFile(fileNameUTF8, false, false, false) == -1)
-    {
-        LOG(LS_ERROR) << "Failed to open file.";
-        return -1;
-    }
-
-    // Store start of RTP dump (to be used for offset calculation later).
-    _startTime = GetTimeInMS();
-
-    // All rtp dump files start with #!rtpplay.
-    char magic[16];
-    sprintf(magic, "#!rtpplay%s \n", RTPFILE_VERSION);
-    if (_file.WriteText(magic) == -1)
-    {
-        LOG(LS_ERROR) << "Error writing to file.";
-        return -1;
-    }
-
-    // The header according to the rtpdump documentation is sizeof(RD_hdr_t)
-    // which is 8 + 4 + 2 = 14 bytes for 32-bit architecture (and 22 bytes on
-    // 64-bit architecture). However, Wireshark use 16 bytes for the header
-    // regardless of if the binary is 32-bit or 64-bit. Go by the same approach
-    // as Wireshark since it makes more sense.
-    // http://wiki.wireshark.org/rtpdump explains that an additional 2 bytes
-    // of padding should be added to the header.
-    char dummyHdr[16];
-    memset(dummyHdr, 0, 16);
-    if (!_file.Write(dummyHdr, sizeof(dummyHdr)))
-    {
-        LOG(LS_ERROR) << "Error writing to file.";
-        return -1;
-    }
-    return 0;
-}
-
-int32_t RtpDumpImpl::Stop()
-{
-    CriticalSectionScoped lock(_critSect);
-    _file.Flush();
-    _file.CloseFile();
-    return 0;
-}
-
-bool RtpDumpImpl::IsActive() const
-{
-    CriticalSectionScoped lock(_critSect);
-    return _file.Open();
-}
-
-int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, size_t packetLength)
-{
-    CriticalSectionScoped lock(_critSect);
-    if (!IsActive())
-    {
-        return 0;
-    }
-
-    if (packet == NULL)
-    {
-        return -1;
-    }
-
-    RtpDumpPacketHeader hdr;
-    size_t total_size = packetLength + sizeof hdr;
-    if (packetLength < 1 || total_size > std::numeric_limits<uint16_t>::max())
-    {
-        return -1;
-    }
-
-    // If the packet doesn't contain a valid RTCP header the packet will be
-    // considered RTP (without further verification).
-    bool isRTCP = RTCP(packet);
-
-    // Offset is relative to when recording was started.
-    uint32_t offset = GetTimeInMS();
-    if (offset < _startTime)
-    {
-        // Compensate for wraparound.
-        offset += MAX_UWORD32 - _startTime + 1;
-    } else {
-        offset -= _startTime;
-    }
-    hdr.offset = RtpDumpHtonl(offset);
-
-    hdr.length = RtpDumpHtons((uint16_t)(total_size));
-    if (isRTCP)
-    {
-        hdr.plen = 0;
-    }
-    else
-    {
-        hdr.plen = RtpDumpHtons((uint16_t)packetLength);
-    }
-
-    if (!_file.Write(&hdr, sizeof(hdr)))
-    {
-        LOG(LS_ERROR) << "Error writing to file.";
-        return -1;
-    }
-    if (!_file.Write(packet, packetLength))
-    {
-        LOG(LS_ERROR) << "Error writing to file.";
-        return -1;
-    }
-
-    return 0;
-}
-
-bool RtpDumpImpl::RTCP(const uint8_t* packet) const
-{
-    return packet[1] == 192 || packet[1] == 200 || packet[1] == 201 ||
-        packet[1] == 202 || packet[1] == 203 || packet[1] == 204 ||
-        packet[1] == 205 || packet[1] == 206 || packet[1] == 207;
-}
-
-// TODO (hellner): why is TickUtil not used here?
-inline uint32_t RtpDumpImpl::GetTimeInMS() const
-{
-#if defined(_WIN32)
-    return timeGetTime();
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-    struct timeval tv;
-    struct timezone tz;
-    unsigned long val;
-
-    gettimeofday(&tv, &tz);
-    val = tv.tv_sec * 1000 + tv.tv_usec / 1000;
-    return val;
-#endif
-}
-
-inline uint32_t RtpDumpImpl::RtpDumpHtonl(uint32_t x) const
-{
-#if defined(WEBRTC_ARCH_BIG_ENDIAN)
-    return x;
-#elif defined(WEBRTC_ARCH_LITTLE_ENDIAN)
-    return (x >> 24) + ((((x >> 16) & 0xFF) << 8) + ((((x >> 8) & 0xFF) << 16) +
-                                                     ((x & 0xFF) << 24)));
-#endif
-}
-
-inline uint16_t RtpDumpImpl::RtpDumpHtons(uint16_t x) const
-{
-#if defined(WEBRTC_ARCH_BIG_ENDIAN)
-    return x;
-#elif defined(WEBRTC_ARCH_LITTLE_ENDIAN)
-    return (x >> 8) + ((x & 0xFF) << 8);
-#endif
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.h b/media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.h
deleted file mode 100644
index b49a690427..0000000000
--- a/media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
-
-#include "webrtc/modules/utility/interface/rtp_dump.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class FileWrapper;
-class RtpDumpImpl : public RtpDump
-{
-public:
-    RtpDumpImpl();
-    virtual ~RtpDumpImpl();
-
-    int32_t Start(const char* fileNameUTF8) override;
-    int32_t Stop() override;
-    bool IsActive() const override;
-    int32_t DumpPacket(const uint8_t* packet, size_t packetLength) override;
-
-private:
-    // Return the system time in ms.
-    inline uint32_t GetTimeInMS() const;
-    // Return x in network byte order (big endian).
-    inline uint32_t RtpDumpHtonl(uint32_t x) const;
-    // Return x in network byte order (big endian).
-    inline uint16_t RtpDumpHtons(uint16_t x) const;
-
-    // Return true if the packet starts with a valid RTCP header.
-    // Note: See RtpUtility::RtpHeaderParser::RTCP() for details on how
-    //       to determine if the packet is an RTCP packet.
-    bool RTCP(const uint8_t* packet) const;
-
-private:
-    CriticalSectionWrapper* _critSect;
-    FileWrapper& _file;
-    uint32_t _startTime;
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/utility/utility.gypi b/media/webrtc/trunk/webrtc/modules/utility/utility.gypi
index 46014e81df..e5b0a4d9c0 100644
--- a/media/webrtc/trunk/webrtc/modules/utility/utility.gypi
+++ b/media/webrtc/trunk/webrtc/modules/utility/utility.gypi
@@ -18,12 +18,13 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'interface/audio_frame_operations.h',
-        'interface/file_player.h',
-        'interface/file_recorder.h',
-        'interface/helpers_android.h',
-        'interface/process_thread.h',
-        'interface/rtp_dump.h',
+        'include/audio_frame_operations.h',
+        'include/file_player.h',
+        'include/file_recorder.h',
+        'include/helpers_android.h',
+        'include/helpers_ios.h',
+        'include/jvm_android.h',
+        'include/process_thread.h',
         'source/audio_frame_operations.cc',
         'source/coder.cc',
         'source/coder.h',
@@ -32,10 +33,10 @@
         'source/file_recorder_impl.cc',
         'source/file_recorder_impl.h',
         'source/helpers_android.cc',
+        'source/helpers_ios.mm',
+        'source/jvm_android.cc',
         'source/process_thread_impl.cc',
         'source/process_thread_impl.h',
-        'source/rtp_dump_impl.cc',
-        'source/rtp_dump_impl.h',
       ],
     },
   ], # targets
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/BUILD.gn b/media/webrtc/trunk/webrtc/modules/video_capture/BUILD.gn
index 89574915e9..94d6e14751 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/BUILD.gn
@@ -16,11 +16,11 @@ source_set("video_capture_module") {
   sources = [
     "device_info_impl.cc",
     "device_info_impl.h",
-    "include/video_capture.h",
-    "include/video_capture_defines.h",
-    "include/video_capture_factory.h",
+    "video_capture.h",
     "video_capture_config.h",
+    "video_capture_defines.h",
     "video_capture_delay.h",
+    "video_capture_factory.h",
     "video_capture_factory.cc",
     "video_capture_impl.cc",
     "video_capture_impl.h",
@@ -127,20 +127,7 @@ if (!build_with_chromium) {
 
       libs = [ "Strmiids.lib" ]
 
-      deps += [ "//third_party/winsdk_samples"]
-    }
-    if (is_android) {
-      sources = [
-        "android/device_info_android.cc",
-        "android/device_info_android.h",
-        "android/video_capture_android.cc",
-        "android/video_capture_android.h",
-      ]
-
-      deps += [
-        "//third_party/icu:icuuc",
-        "//third_party/jsoncpp",
-      ]
+      deps += [ "//third_party/winsdk_samples" ]
     }
     if (is_ios) {
       sources = [
@@ -156,6 +143,7 @@ if (!build_with_chromium) {
 
       cflags = [
         "-fobjc-arc",  # CLANG_ENABLE_OBJC_ARC = YES.
+
         # To avoid warnings for deprecated videoMinFrameDuration and
         # videoMaxFrameDuration properties in iOS 7.0.
         # See webrtc:3705 for more details.
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/android/device_info_android.cc b/media/webrtc/trunk/webrtc/modules/video_capture/android/device_info_android.cc
index 624b0feb66..333862b48d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/android/device_info_android.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/android/device_info_android.cc
@@ -15,11 +15,11 @@
 #include <sstream>
 #include <vector>
 
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 #include "webrtc/modules/video_capture/android/video_capture_android.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include "AndroidJNIWrapper.h"
 
@@ -35,8 +35,9 @@ typedef std::vector<IntPair> IntPairs;
 static std::string IntPairsToString(const IntPairs& pairs, char separator) {
   std::stringstream stream;
   for (size_t i = 0; i < pairs.size(); ++i) {
-    if (i > 0)
+    if (i > 0) {
       stream << ", ";
+    }
     stream << "(" << pairs[i].first << separator << pairs[i].second << ")";
   }
   return stream.str();
@@ -64,7 +65,7 @@ struct AndroidCameraInfo {
 // Camera info; populated during DeviceInfoAndroid::Refresh()
 static std::vector<AndroidCameraInfo>* g_camera_info = NULL;
 
-static JavaVM* g_jvm = NULL;
+static JavaVM* g_jvm_dev_info = NULL;
 
 // Set |*index| to the index of |name| in g_camera_info or return false if no
 // match found.
@@ -82,8 +83,9 @@ static bool FindCameraIndexByName(const std::string& name, size_t* index) {
 // is found.
 static AndroidCameraInfo* FindCameraInfoByName(const std::string& name) {
   size_t index = 0;
-  if (FindCameraIndexByName(name, &index))
+  if (FindCameraIndexByName(name, &index)) {
     return &g_camera_info->at(index);
+  }
   return NULL;
 }
 
@@ -95,18 +97,20 @@ void DeviceInfoAndroid::Initialize(JavaVM* javaVM) {
   // prevent this.  Once that code is made to only
   // VideoEngine::SetAndroidObjects() once per process, this can turn into an
   // assert.
-  if (g_camera_info)
-    return;
-
-  g_jvm = javaVM;
-}
-
-void DeviceInfoAndroid::BuildDeviceList() {
-  if (!g_jvm) {
+  if (g_camera_info) {
     return;
   }
 
-  AttachThreadScoped ats(g_jvm);
+  g_jvm_dev_info = javaVM;
+  BuildDeviceList();
+}
+
+void DeviceInfoAndroid::BuildDeviceList() {
+  if (!g_jvm_dev_info) {
+    return;
+  }
+
+  AttachThreadScoped ats(g_jvm_dev_info);
   JNIEnv* jni = ats.env();
 
   g_camera_info = new std::vector<AndroidCameraInfo>();
@@ -233,8 +237,9 @@ int32_t DeviceInfoAndroid::GetDeviceName(
     char* /*productUniqueIdUTF8*/,
     uint32_t /*productUniqueIdUTF8Length*/,
     pid_t* /*pid*/) {
-  if (deviceNumber >= g_camera_info->size())
+  if (deviceNumber >= g_camera_info->size()) {
     return -1;
+  }
   const AndroidCameraInfo& info = g_camera_info->at(deviceNumber);
   if (info.name.length() + 1 > deviceNameLength ||
       info.name.length() + 1 > deviceUniqueIdUTF8Length) {
@@ -249,8 +254,9 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
     const char* deviceUniqueIdUTF8) {
   _captureCapabilities.clear();
   const AndroidCameraInfo* info = FindCameraInfoByName(deviceUniqueIdUTF8);
-  if (info == NULL)
+  if (info == NULL) {
     return -1;
+  }
 
   for (size_t i = 0; i < info->resolutions.size(); ++i) {
     for (size_t j = 0; j < info->mfpsRanges.size(); ++j) {
@@ -283,8 +289,9 @@ void DeviceInfoAndroid::GetMFpsRange(const char* deviceUniqueIdUTF8,
                                      int max_fps_to_match,
                                      int* min_mfps, int* max_mfps) {
   const AndroidCameraInfo* info = FindCameraInfoByName(deviceUniqueIdUTF8);
-  if (info == NULL)
+  if (info == NULL) {
     return;
+  }
   int desired_mfps = max_fps_to_match * 1000;
   int best_diff_mfps = 0;
   LOG(LS_INFO) << "Search for best target mfps " << desired_mfps;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java b/media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java
index 7546aa5ad4..7ece5cb182 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java
@@ -11,7 +11,6 @@
 package org.webrtc.videoengine;
 
 import java.io.IOException;
-import java.util.ArrayList;
 import java.util.List;
 import java.util.concurrent.Exchanger;
 
@@ -33,9 +32,6 @@ import android.view.SurfaceHolder;
 import android.view.WindowManager;
 
 import org.mozilla.gecko.annotation.WebRTCJNITarget;
-import org.mozilla.gecko.GeckoAppShell;
-import org.mozilla.gecko.GeckoAppShell.AppStateListener;
-
 
 // Wrapper for android Camera, with support for direct local preview rendering.
 // Threading notes: this class is called from ViE C++ code, and from Camera &
@@ -46,92 +42,47 @@ import org.mozilla.gecko.GeckoAppShell.AppStateListener;
 // uncontended.  Note that each of these synchronized methods must check
 // |camera| for null to account for having possibly waited for stopCapture() to
 // complete.
-public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateListener {
+public class VideoCaptureAndroid implements PreviewCallback, Callback {
   private final static String TAG = "WEBRTC-JC";
 
-  // Only non-null while capturing, accessed exclusively from synchronized methods.
-  Camera camera;
-  private Camera.CameraInfo info;
+  private static SurfaceHolder localPreview;
+  private Camera camera;  // Only non-null while capturing.
   private CameraThread cameraThread;
   private Handler cameraThreadHandler;
   private Context context;
   private final int id;
+  private final Camera.CameraInfo info;
   private volatile long native_capturer;  // |VideoCaptureAndroid*| in C++.
   private SurfaceTexture cameraSurfaceTexture;
   private int[] cameraGlTextures = null;
-
   // Arbitrary queue depth.  Higher number means more memory allocated & held,
   // lower number means more sensitivity to processing time in the client (and
   // potentially stalling the capturer if it runs out of buffers to write to).
   private final int numCaptureBuffers = 3;
-
-  // Needed to start/stop/rotate camera.
-  volatile int mCaptureRotation;
-  int mCaptureWidth;
-  int mCaptureHeight;
-  int mCaptureMinFPS;
-  int mCaptureMaxFPS;
-  // Are we being told to start/stop the camera, or just suspending/resuming
-  // due to the application being backgrounded.
-  boolean mResumeCapture;
-
   private double averageDurationMs;
   private long lastCaptureTimeMs;
   private int frameCount;
   private int frameDropRatio;
 
-  @WebRTCJNITarget
-  public VideoCaptureAndroid(int id, long native_capturer) {
+  // Requests future capturers to send their frames to |localPreview| directly.
+  public static void setLocalPreview(SurfaceHolder localPreview) {
+    // It is a gross hack that this is a class-static.  Doing it right would
+    // mean plumbing this through the C++ API and using it from
+    // webrtc/examples/android/media_demo's MediaEngine class.
+    VideoCaptureAndroid.localPreview = localPreview;
+  }
+
+ @WebRTCJNITarget
+ public VideoCaptureAndroid(int id, long native_capturer) {
     this.id = id;
     this.native_capturer = native_capturer;
     this.context = GetContext();
     this.info = new Camera.CameraInfo();
     Camera.getCameraInfo(id, info);
-    mCaptureRotation = GetRotateAmount();
-  }
-
-  @Override
-  public synchronized void onPause() {
-    if (camera != null) {
-      mResumeCapture = true;
-      stopCapture();
-      GeckoAppShell.notifyObservers("VideoCapture:Paused", null);
-    }
-  }
-
-  @Override
-  public synchronized void onResume() {
-    if (mResumeCapture) {
-      startCapture(mCaptureWidth, mCaptureHeight, mCaptureMinFPS, mCaptureMaxFPS);
-      mResumeCapture = false;
-      GeckoAppShell.notifyObservers("VideoCapture:Resumed", null);
-    }
-  }
-
-  @Override
-  public void onOrientationChanged() {
-    mCaptureRotation = GetRotateAmount();
-  }
-
-  public int GetRotateAmount() {
-    int rotation = GeckoAppShell.getGeckoInterface().getActivity().getWindowManager().getDefaultDisplay().getRotation();
-    int degrees = 0;
-    switch (rotation) {
-      case Surface.ROTATION_0: degrees = 0; break;
-      case Surface.ROTATION_90: degrees = 90; break;
-      case Surface.ROTATION_180: degrees = 180; break;
-      case Surface.ROTATION_270: degrees = 270; break;
-    }
-    int result;
-    if (info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT) {
-      result = (info.orientation + degrees) % 360;
-    } else {  // back-facing
-      result = (info.orientation - degrees + 360) % 360;
-    }
-    return result;
   }
 
   // Return the global application context.
+  @WebRTCJNITarget
   private static native Context GetContext();
 
   private class CameraThread extends Thread {
@@ -176,68 +127,75 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
     return startResult;
   }
 
+  @WebRTCJNITarget
+  private void unlinkCapturer() {
+    // stopCapture might fail. That might leave the callbacks dangling, so make
+    // sure those don't call into dead code.
+    // Note that onPreviewCameraFrame isn't synchronized, so there's no point in
+    // synchronizing us either. ProvideCameraFrame has to do the null check.
+    native_capturer = 0;
+  }
+
   private void startCaptureOnCameraThread(
       int width, int height, int min_mfps, int max_mfps,
       Exchanger<Boolean> result) {
-    if (!mResumeCapture) {
-      ViERenderer.CreateLocalRenderer();
-    }
     Throwable error = null;
     try {
       camera = Camera.open(id);
 
-      // No local renderer (we only care about onPreviewFrame() buffers, not a
-      // directly-displayed UI element).  Camera won't capture without
-      // setPreview{Texture,Display}, so we create a SurfaceTexture and hand
-      // it over to Camera, but never listen for frame-ready callbacks,
-      // and never call updateTexImage on it.
-      try {
-        cameraGlTextures = new int[1];
+      if (localPreview != null) {
+        localPreview.addCallback(this);
+        if (localPreview.getSurface() != null &&
+            localPreview.getSurface().isValid()) {
+	  try {
+	    camera.setPreviewDisplay(localPreview);
+	  } catch (IOException e) {
+	    throw new RuntimeException(e);
+	  }
+        }
+      } else {
+        // No local renderer (we only care about onPreviewFrame() buffers, not a
+        // directly-displayed UI element).  Camera won't capture without
+        // setPreview{Texture,Display}, so we create a SurfaceTexture and hand
+        // it over to Camera, but never listen for frame-ready callbacks,
+        // and never call updateTexImage on it.
+        try {
+          cameraGlTextures = new int[1];
+          // Generate one texture pointer and bind it as an external texture.
+          GLES20.glGenTextures(1, cameraGlTextures, 0);
+          GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              cameraGlTextures[0]);
+          GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_MIN_FILTER, GLES20.GL_LINEAR);
+          GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_MAG_FILTER, GLES20.GL_LINEAR);
+          GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_WRAP_S, GLES20.GL_CLAMP_TO_EDGE);
+          GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+              GLES20.GL_TEXTURE_WRAP_T, GLES20.GL_CLAMP_TO_EDGE);
 
-        // Generate one texture pointer and bind it as an external texture.
-        GLES20.glGenTextures(1, cameraGlTextures, 0);
-        GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
-                               cameraGlTextures[0]);
-        GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
-                               GLES20.GL_TEXTURE_MIN_FILTER, GLES20.GL_LINEAR);
-        GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
-                               GLES20.GL_TEXTURE_MAG_FILTER, GLES20.GL_LINEAR);
-        GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
-                               GLES20.GL_TEXTURE_WRAP_S, GLES20.GL_CLAMP_TO_EDGE);
-        GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
-                               GLES20.GL_TEXTURE_WRAP_T, GLES20.GL_CLAMP_TO_EDGE);
-
-        cameraSurfaceTexture = new SurfaceTexture(cameraGlTextures[0]);
-        cameraSurfaceTexture.setOnFrameAvailableListener(null);
-        camera.setPreviewTexture(cameraSurfaceTexture);
-      } catch (IOException e) {
-        throw new RuntimeException(e);
+          cameraSurfaceTexture = new SurfaceTexture(cameraGlTextures[0]);
+          cameraSurfaceTexture.setOnFrameAvailableListener(null);
+          camera.setPreviewTexture(cameraSurfaceTexture);
+        } catch (IOException e) {
+          throw new RuntimeException(e);
+        }
       }
 
       Log.d(TAG, "Camera orientation: " + info.orientation +
-          ". Device orientation: " + getDeviceOrientation());
+          " .Device orientation: " + getDeviceOrientation());
       Camera.Parameters parameters = camera.getParameters();
-      // This wasn't added until ICS MR1.
-      if(android.os.Build.VERSION.SDK_INT>14) {
-        Log.d(TAG, "isVideoStabilizationSupported: " +
-              parameters.isVideoStabilizationSupported());
-        if (parameters.isVideoStabilizationSupported()) {
-          parameters.setVideoStabilization(true);
-        }
-      }
-      List<String> focusModeList = parameters.getSupportedFocusModes();
-      // Not supposed to fail, but observed on Android 4.0 emulator nevertheless
-      if (focusModeList != null) {
-        if (focusModeList.contains(Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO)) {
-            parameters.setFocusMode(Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO);
-        }
+      Log.d(TAG, "isVideoStabilizationSupported: " +
+          parameters.isVideoStabilizationSupported());
+      if (parameters.isVideoStabilizationSupported()) {
+        parameters.setVideoStabilization(true);
       }
+      parameters.setPictureSize(width, height);
       parameters.setPreviewSize(width, height);
 
       // Check if requested fps range is supported by camera,
       // otherwise calculate frame drop ratio.
-      List<int[]> supportedFpsRanges =
-          VideoCaptureDeviceInfoAndroid.getFpsRangesRobust(parameters);
+      List<int[]> supportedFpsRanges = parameters.getSupportedPreviewFpsRange();
       frameDropRatio = Integer.MAX_VALUE;
       for (int i = 0; i < supportedFpsRanges.size(); i++) {
         int[] range = supportedFpsRanges.get(i);
@@ -269,14 +227,6 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
       int format = ImageFormat.NV21;
       parameters.setPreviewFormat(format);
       camera.setParameters(parameters);
-      try {
-          // See https://code.google.com/p/webrtc/issues/detail?id=4197
-          parameters.setPictureSize(width, height);
-          camera.setParameters(parameters);
-      } catch(RuntimeException e) {
-          Log.d(TAG, "Failed to apply Nexus 7 workaround");
-      }
-
       int bufSize = width * height * ImageFormat.getBitsPerPixel(format) / 8;
       for (int i = 0; i < numCaptureBuffers; i++) {
         camera.addCallbackBuffer(new byte[bufSize]);
@@ -285,15 +235,6 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
       frameCount = 0;
       averageDurationMs = 1000000.0f / (max_mfps / frameDropRatio);
       camera.startPreview();
-      // Remember parameters we were started with.
-      mCaptureWidth = width;
-      mCaptureHeight = height;
-      mCaptureMinFPS = min_mfps;
-      mCaptureMaxFPS = max_mfps;
-      // If we are resuming a paused capture, the listener is already active.
-      if (!mResumeCapture) {
-        GeckoAppShell.getGeckoInterface().addAppStateListener(this);
-      }
       exchange(result, true);
       return;
     } catch (RuntimeException e) {
@@ -313,10 +254,6 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
   @WebRTCJNITarget
   private synchronized boolean stopCapture() {
     Log.d(TAG, "stopCapture");
-    // See comment at the top of startCaptureOnCameraThread
-    if (cameraThreadHandler == null) {
-      return true;
-    }
     final Exchanger<Boolean> result = new Exchanger<Boolean>();
     cameraThreadHandler.post(new Runnable() {
         @Override public void run() {
@@ -335,30 +272,15 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
     return status;
   }
 
-  @WebRTCJNITarget
-  private void unlinkCapturer() {
-    // stopCapture might fail. That might leave the callbacks dangling, so make
-    // sure those don't call into dead code.
-    // Note that onPreviewCameraFrame isn't synchronized, so there's no point in
-    // synchronizing us either. ProvideCameraFrame has to do the null check.
-    native_capturer = 0;
-  }
-
   private void stopCaptureOnCameraThread(
       Exchanger<Boolean> result) {
     if (camera == null) {
-      if (mResumeCapture == true) {
-        // We already got onPause, but now the native code wants us to stop.
-        // Do not resume capturing when resuming the app.
-        mResumeCapture = false;
-        return;
-      }
       throw new RuntimeException("Camera is already stopped!");
     }
     Throwable error = null;
     try {
-      camera.setPreviewCallbackWithBuffer(null);
       camera.stopPreview();
+      camera.setPreviewCallbackWithBuffer(null);
       camera.setPreviewTexture(null);
       cameraSurfaceTexture = null;
       if (cameraGlTextures != null) {
@@ -367,11 +289,6 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
       }
       camera.release();
       camera = null;
-      // If we want to resume after onResume, keep the listener in place.
-      if (!mResumeCapture) {
-        GeckoAppShell.getGeckoInterface().removeAppStateListener(this);
-        ViERenderer.DestroyLocalRenderer();
-      }
       exchange(result, true);
       Looper.myLooper().quit();
       return;
@@ -452,10 +369,45 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback, AppStateL
     }
     rotation = (info.orientation + rotation) % 360;
 
-    if (data != null) {
-      ProvideCameraFrame(data, data.length, mCaptureRotation, lastCaptureTimeMs, native_capturer);
-      camera.addCallbackBuffer(data);
+    ProvideCameraFrame(data, data.length, rotation,
+        captureTimeMs, native_capturer);
+    camera.addCallbackBuffer(data);
+  }
+
+  // Sets the rotation of the preview render window.
+  // Does not affect the captured video image.
+  // Called by native code.
+  private synchronized void setPreviewRotation(final int rotation) {
+    if (camera == null || cameraThreadHandler == null) {
+      return;
     }
+    final Exchanger<IOException> result = new Exchanger<IOException>();
+    cameraThreadHandler.post(new Runnable() {
+        @Override public void run() {
+          setPreviewRotationOnCameraThread(rotation, result);
+        }
+      });
+    // Use the exchanger below to block this function until
+    // setPreviewRotationOnCameraThread() completes, holding the synchronized
+    // lock for the duration.  The exchanged value itself is ignored.
+    exchange(result, null);
+  }
+
+  private void setPreviewRotationOnCameraThread(
+      int rotation, Exchanger<IOException> result) {
+    Log.v(TAG, "setPreviewRotation:" + rotation);
+
+    int resultRotation = 0;
+    if (info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT) {
+      // This is a front facing camera.  SetDisplayOrientation will flip
+      // the image horizontally before doing the rotation.
+      resultRotation = ( 360 - rotation ) % 360; // Compensate for the mirror.
+    } else {
+      // Back-facing camera.
+      resultRotation = rotation;
+    }
+    camera.setDisplayOrientation(resultRotation);
+    exchange(result, null);
   }
 
   @WebRTCJNITarget
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/android/video_capture_android.cc b/media/webrtc/trunk/webrtc/modules/video_capture/android/video_capture_android.cc
index e03493f247..53f2e385fb 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/android/video_capture_android.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/android/video_capture_android.cc
@@ -8,19 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "AndroidJNIWrapper.h"
 #include "webrtc/modules/video_capture/android/video_capture_android.h"
 
 #include "webrtc/base/common.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
 #include "webrtc/modules/video_capture/android/device_info_android.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logcat_trace_context.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/logcat_trace_context.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
-static JavaVM* g_jvm = NULL;
+#include "AndroidJNIWrapper.h"
+
+static JavaVM* g_jvm_capture = NULL;
 static jclass g_java_capturer_class = NULL;  // VideoCaptureAndroid.class.
 static jobject g_context = NULL;  // Owned android.content.Context.
 
@@ -41,8 +42,10 @@ void JNICALL ProvideCameraFrame(
     jint rotation,
     jlong timeStamp,
     jlong context) {
-  if (!context)
+  if (!context) {
     return;
+  }
+
   webrtc::videocapturemodule::VideoCaptureAndroid* captureModule =
       reinterpret_cast<webrtc::videocapturemodule::VideoCaptureAndroid*>(
           context);
@@ -53,17 +56,18 @@ void JNICALL ProvideCameraFrame(
 }
 
 int32_t SetCaptureAndroidVM(JavaVM* javaVM) {
-  if (g_java_capturer_class)
+  if (g_java_capturer_class) {
     return 0;
+  }
 
   if (javaVM) {
-    assert(!g_jvm);
-    g_jvm = javaVM;
-    AttachThreadScoped ats(g_jvm);
+    assert(!g_jvm_capture);
+    g_jvm_capture = javaVM;
+    AttachThreadScoped ats(g_jvm_capture);
 
     g_context = jsjni_GetGlobalContextRef();
 
-    videocapturemodule::DeviceInfoAndroid::Initialize(g_jvm);
+    videocapturemodule::DeviceInfoAndroid::Initialize(g_jvm_capture);
 
     g_java_capturer_class =
       jsjni_GetGlobalClassRef("org/webrtc/videoengine/VideoCaptureAndroid");
@@ -80,14 +84,14 @@ int32_t SetCaptureAndroidVM(JavaVM* javaVM) {
                                    native_methods, 2) != 0)
       assert(false);
   } else {
-    if (g_jvm) {
-      AttachThreadScoped ats(g_jvm);
+    if (g_jvm_capture) {
+      AttachThreadScoped ats(g_jvm_capture);
       ats.env()->UnregisterNatives(g_java_capturer_class);
       ats.env()->DeleteGlobalRef(g_java_capturer_class);
       g_java_capturer_class = NULL;
       g_context = NULL;
       videocapturemodule::DeviceInfoAndroid::DeInitialize();
-      g_jvm = NULL;
+      g_jvm_capture = NULL;
     }
   }
 
@@ -152,7 +156,7 @@ int32_t VideoCaptureAndroid::Init(const int32_t id,
   _deviceUniqueId = new char[nameLength + 1];
   memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
 
-  AttachThreadScoped ats(g_jvm);
+  AttachThreadScoped ats(g_jvm_capture);
   JNIEnv* env = ats.env();
   jmethodID ctor = env->GetMethodID(g_java_capturer_class, "<init>", "(IJ)V");
   assert(ctor);
@@ -168,7 +172,7 @@ VideoCaptureAndroid::~VideoCaptureAndroid() {
   // Ensure Java camera is released even if our caller didn't explicitly Stop.
   if (_captureStarted)
     StopCapture();
-  AttachThreadScoped ats(g_jvm);
+  AttachThreadScoped ats(g_jvm_capture);
   JNIEnv* env = ats.env();
 
   // Avoid callbacks into ourself even if the above stopCapture fails.
@@ -182,7 +186,7 @@ VideoCaptureAndroid::~VideoCaptureAndroid() {
 int32_t VideoCaptureAndroid::StartCapture(
     const VideoCaptureCapability& capability) {
   CriticalSectionScoped cs(&_apiCs);
-  AttachThreadScoped ats(g_jvm);
+  AttachThreadScoped ats(g_jvm_capture);
   JNIEnv* env = ats.env();
 
   if (_deviceInfo.GetBestMatchedCapability(
@@ -215,7 +219,7 @@ int32_t VideoCaptureAndroid::StartCapture(
 
 int32_t VideoCaptureAndroid::StopCapture() {
   _apiCs.Enter();
-  AttachThreadScoped ats(g_jvm);
+  AttachThreadScoped ats(g_jvm_capture);
   JNIEnv* env = ats.env();
 
   memset(&_requestedCapability, 0, sizeof(_requestedCapability));
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.cc b/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.cc
index 0f857f586a..2d9fa0b473 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.cc
@@ -13,7 +13,7 @@
 
 #include "webrtc/modules/video_capture/device_info_impl.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 #ifndef abs
 #define abs(a) (a>=0?a:-a)
@@ -107,7 +107,7 @@ int32_t DeviceInfoImpl::GetCapability(const char* deviceUniqueIdUTF8,
     // Make sure the number is valid
     if (deviceCapabilityNumber >= (unsigned int) _captureCapabilities.size())
     {
-        LOG(LS_ERROR) << "Invalid deviceCapabilityNumber "
+        LOG(LS_ERROR) << deviceUniqueIdUTF8 << " Invalid deviceCapabilityNumber "
                       << deviceCapabilityNumber << ">= number of capabilities ("
                       << _captureCapabilities.size() << ").";
         return -1;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.h b/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.h
index 1571ba3b8a..9d37366def 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.h
@@ -13,9 +13,9 @@
 
 #include <vector>
 
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_delay.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 namespace webrtc
 {
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ensure_initialized.cc b/media/webrtc/trunk/webrtc/modules/video_capture/ensure_initialized.cc
deleted file mode 100644
index 9d43d9f1bd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ensure_initialized.cc
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Platform-specific initialization bits, if any, go here.
-
-#ifndef ANDROID
-
-namespace webrtc {
-namespace videocapturemodule {
-void EnsureInitialized() {}
-}  // namespace videocapturemodule
-}  // namespace webrtc
-
-#else
-
-#include <pthread.h>
-
-#include "base/android/jni_android.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/video_capture/video_capture_internal.h"
-
-namespace webrtc {
-namespace videocapturemodule {
-
-static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
-
-void EnsureInitializedOnce() {
-  JNIEnv* jni = ::base::android::AttachCurrentThread();
-  jobject context = ::base::android::GetApplicationContext();
-  JavaVM* jvm = NULL;
-  CHECK_EQ(0, jni->GetJavaVM(&jvm));
-  CHECK_EQ(0, webrtc::SetCaptureAndroidVM(jvm, context));
-}
-
-void EnsureInitialized() {
-  CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
-}
-
-}  // namespace videocapturemodule
-}  // namespace webrtc
-
-#endif  // !ANDROID
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/external/video_capture_external.cc b/media/webrtc/trunk/webrtc/modules/video_capture/external/video_capture_external.cc
index ff0f1a45a9..29b161263c 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/external/video_capture_external.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/external/video_capture_external.cc
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/video_capture/video_capture_impl.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/include/mock/mock_video_capture.h b/media/webrtc/trunk/webrtc/modules/video_capture/include/mock/mock_video_capture.h
deleted file mode 100644
index 0a15c445ed..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_capture/include/mock/mock_video_capture.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_MOCK_MOCK_VIDEO_CAPTURE_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_MOCK_MOCK_VIDEO_CAPTURE_H_
-
-#include "webrtc/modules/video_capture/include/video_capture.h"
-#include "testing/gmock/include/gmock/gmock.h"
-
-namespace webrtc {
-
-class MockVideoCaptureModule : public VideoCaptureModule {
- public:
-  // from Module
-  MOCK_METHOD0(TimeUntilNextProcess, int64_t());
-  MOCK_METHOD0(Process, int32_t());
-
-  // from RefCountedModule
-  MOCK_METHOD0(AddRef, int32_t());
-  MOCK_METHOD0(Release, int32_t());
-
-  // from VideoCaptureModule
-  MOCK_METHOD1(RegisterCaptureDataCallback,
-      void(VideoCaptureDataCallback& dataCallback));
-  MOCK_METHOD0(DeRegisterCaptureDataCallback, void());
-  MOCK_METHOD1(RegisterCaptureCallback, void(VideoCaptureFeedBack& callBack));
-  MOCK_METHOD0(DeRegisterCaptureCallback, void());
-  MOCK_METHOD1(StartCapture, int32_t(const VideoCaptureCapability& capability));
-  MOCK_METHOD0(StopCapture, int32_t());
-  MOCK_CONST_METHOD0(CurrentDeviceName, const char*());
-  MOCK_METHOD0(CaptureStarted, bool());
-  MOCK_METHOD1(CaptureSettings, int32_t(VideoCaptureCapability& settings));
-  MOCK_METHOD1(SetCaptureDelay, void(int32_t delayMS));
-  MOCK_METHOD0(CaptureDelay, int32_t());
-  MOCK_METHOD1(SetCaptureRotation, int32_t(VideoRotation rotation));
-  MOCK_METHOD1(SetApplyRotation, bool(bool));
-  MOCK_METHOD0(GetApplyRotation, bool());
-  MOCK_METHOD1(GetEncodeInterface,
-               VideoCaptureEncodeInterface*(const VideoCodec& codec));
-  MOCK_METHOD1(EnableFrameRateCallback, void(const bool enable));
-  MOCK_METHOD1(EnableNoPictureAlarm, void(const bool enable));
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_MOCK_MOCK_VIDEO_CAPTURE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.h b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.h
index e10db4a8a2..6af7c33899 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.h
@@ -13,6 +13,8 @@
 
 #include "webrtc/modules/video_capture/device_info_impl.h"
 
+#include <map>
+
 namespace webrtc {
 namespace videocapturemodule {
 class DeviceInfoIos : public DeviceInfoImpl {
@@ -37,10 +39,6 @@ class DeviceInfoIos : public DeviceInfoImpl {
                         const uint32_t deviceCapabilityNumber,
                         VideoCaptureCapability& capability) override;
 
-  int32_t GetBestMatchedCapability(const char* deviceUniqueIdUTF8,
-                                   const VideoCaptureCapability& requested,
-                                   VideoCaptureCapability& resulting) override;
-
   int32_t DisplayCaptureSettingsDialogBox(const char* deviceUniqueIdUTF8,
                                           const char* dialogTitleUTF8,
                                           void* parentWindow,
@@ -51,6 +49,9 @@ class DeviceInfoIos : public DeviceInfoImpl {
                          VideoRotation& orientation) override;
 
   int32_t CreateCapabilityMap(const char* device_unique_id_utf8) override;
+
+ private:
+  std::map<std::string, VideoCaptureCapabilities> _capabilitiesMap;
 };
 
 }  // namespace videocapturemodule
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.mm b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.mm
index 21206ea50c..307e5d3605 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.mm
@@ -12,14 +12,24 @@
 #error "This file requires ARC support."
 #endif
 
+#include <AVFoundation/AVFoundation.h>
+
+#include <string>
+
 #include "webrtc/modules/video_capture/ios/device_info_ios.h"
 #include "webrtc/modules/video_capture/ios/device_info_ios_objc.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 using namespace videocapturemodule;
 
+static NSArray *camera_presets = @[AVCaptureSessionPreset352x288,
+                                   AVCaptureSessionPreset640x480,
+                                   AVCaptureSessionPreset1280x720,
+                                   AVCaptureSessionPreset1920x1080];
+
+
 #define IOS_UNSUPPORTED()                                  \
   WEBRTC_TRACE(kTraceError,                                \
                kTraceVideoCapture,                         \
@@ -34,11 +44,42 @@ VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo(
 }
 
 DeviceInfoIos::DeviceInfoIos(const int32_t device_id)
-    : DeviceInfoImpl(device_id) {}
+    : DeviceInfoImpl(device_id) {
+  this->Init();
+}
 
 DeviceInfoIos::~DeviceInfoIos() {}
 
-int32_t DeviceInfoIos::Init() { return 0; }
+int32_t DeviceInfoIos::Init() {
+  // Fill in all device capabilities.
+
+  int deviceCount = [DeviceInfoIosObjC captureDeviceCount];
+
+  for (int i = 0; i < deviceCount; i++) {
+    AVCaptureDevice *avDevice = [DeviceInfoIosObjC captureDeviceForIndex:i];
+    VideoCaptureCapabilities capabilityVector;
+
+    for (NSString *preset in camera_presets) {
+      BOOL support = [avDevice supportsAVCaptureSessionPreset:preset];
+      if (support) {
+        VideoCaptureCapability capability =
+            [DeviceInfoIosObjC capabilityForPreset:preset];
+        capabilityVector.push_back(capability);
+      }
+    }
+
+    char deviceNameUTF8[256];
+    char deviceId[256];
+    this->GetDeviceName(i, deviceNameUTF8, 256, deviceId, 256);
+    std::string deviceIdCopy(deviceId);
+    std::pair<std::string, VideoCaptureCapabilities> mapPair =
+        std::pair<std::string, VideoCaptureCapabilities>
+            (deviceIdCopy, capabilityVector);
+    _capabilitiesMap.insert(mapPair);
+  }
+
+  return 0;
+}
 
 uint32_t DeviceInfoIos::NumberOfDevices() {
   return [DeviceInfoIosObjC captureDeviceCount];
@@ -72,20 +113,36 @@ int32_t DeviceInfoIos::GetDeviceName(uint32_t deviceNumber,
 }
 
 int32_t DeviceInfoIos::NumberOfCapabilities(const char* deviceUniqueIdUTF8) {
-  IOS_UNSUPPORTED();
+  int32_t numberOfCapabilities = 0;
+  std::string deviceUniqueId(deviceUniqueIdUTF8);
+  std::map<std::string, VideoCaptureCapabilities>::iterator it =
+      _capabilitiesMap.find(deviceUniqueId);
+
+  if (it != _capabilitiesMap.end()) {
+    numberOfCapabilities = it->second.size();
+  }
+  return numberOfCapabilities;
 }
 
 int32_t DeviceInfoIos::GetCapability(const char* deviceUniqueIdUTF8,
                                      const uint32_t deviceCapabilityNumber,
                                      VideoCaptureCapability& capability) {
-  IOS_UNSUPPORTED();
-}
+  std::string deviceUniqueId(deviceUniqueIdUTF8);
+  std::map<std::string, VideoCaptureCapabilities>::iterator it =
+      _capabilitiesMap.find(deviceUniqueId);
 
-int32_t DeviceInfoIos::GetBestMatchedCapability(
-    const char* deviceUniqueIdUTF8,
-    const VideoCaptureCapability& requested,
-    VideoCaptureCapability& resulting) {
-  IOS_UNSUPPORTED();
+  if (it != _capabilitiesMap.end()) {
+    VideoCaptureCapabilities deviceCapabilities = it->second;
+
+    if (deviceCapabilityNumber < deviceCapabilities.size()) {
+      VideoCaptureCapability cap;
+      cap = deviceCapabilities[deviceCapabilityNumber];
+      capability = cap;
+      return 0;
+    }
+  }
+
+  return -1;
 }
 
 int32_t DeviceInfoIos::DisplayCaptureSettingsDialogBox(
@@ -108,5 +165,14 @@ int32_t DeviceInfoIos::GetOrientation(const char* deviceUniqueIdUTF8,
 }
 
 int32_t DeviceInfoIos::CreateCapabilityMap(const char* deviceUniqueIdUTF8) {
-  IOS_UNSUPPORTED();
+  std::string deviceName(deviceUniqueIdUTF8);
+  std::map<std::string, std::vector<VideoCaptureCapability>>::iterator it =
+      _capabilitiesMap.find(deviceName);
+  VideoCaptureCapabilities deviceCapabilities;
+  if (it != _capabilitiesMap.end()) {
+    _captureCapabilities = it->second;
+    return 0;
+  }
+
+  return -1;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.h b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.h
index b4ab0cb220..d67b559972 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.h
@@ -13,6 +13,8 @@
 
 #import <AVFoundation/AVFoundation.h>
 
+#include "webrtc/modules/video_capture/video_capture_defines.h"
+
 @interface DeviceInfoIosObjC : NSObject
 + (int)captureDeviceCount;
 + (AVCaptureDevice*)captureDeviceForIndex:(int)index;
@@ -20,6 +22,8 @@
 + (NSString*)deviceNameForIndex:(int)index;
 + (NSString*)deviceUniqueIdForIndex:(int)index;
 + (NSString*)deviceNameForUniqueId:(NSString*)uniqueId;
++ (webrtc::VideoCaptureCapability)capabilityForPreset:(NSString*)preset;
+
 @end
 
 #endif  // WEBRTC_MODULES_VIDEO_CAPTURE_IOS_DEVICE_INFO_IOS_OBJC_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.mm b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.mm
index d06d3361f4..818f8624c7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.mm
@@ -15,6 +15,7 @@
 #import <AVFoundation/AVFoundation.h>
 
 #import "webrtc/modules/video_capture/ios/device_info_ios_objc.h"
+#include "webrtc/modules/video_capture/video_capture_config.h"
 
 @implementation DeviceInfoIosObjC
 
@@ -50,4 +51,50 @@
   return [[AVCaptureDevice deviceWithUniqueID:uniqueId] localizedName];
 }
 
++ (webrtc::VideoCaptureCapability)capabilityForPreset:(NSString*)preset {
+  webrtc::VideoCaptureCapability capability;
+
+  // TODO(tkchin): Maybe query AVCaptureDevice for supported formats, and
+  // then get the dimensions / frame rate from each supported format
+  if ([preset isEqualToString:AVCaptureSessionPreset352x288]) {
+    capability.width = 352;
+    capability.height = 288;
+    capability.maxFPS = 30;
+    capability.expectedCaptureDelay =
+        webrtc::videocapturemodule::kDefaultCaptureDelay;
+    capability.rawType = webrtc::kVideoNV12;
+    capability.codecType = webrtc::kVideoCodecUnknown;
+    capability.interlaced = false;
+  } else if ([preset isEqualToString:AVCaptureSessionPreset640x480]) {
+    capability.width = 640;
+    capability.height = 480;
+    capability.maxFPS = 30;
+    capability.expectedCaptureDelay =
+        webrtc::videocapturemodule::kDefaultCaptureDelay;
+    capability.rawType = webrtc::kVideoNV12;
+    capability.codecType = webrtc::kVideoCodecUnknown;
+    capability.interlaced = false;
+  } else if ([preset isEqualToString:AVCaptureSessionPreset1280x720]) {
+    capability.width = 1280;
+    capability.height = 720;
+    capability.maxFPS = 30;
+    capability.expectedCaptureDelay =
+        webrtc::videocapturemodule::kDefaultCaptureDelay;
+    capability.rawType = webrtc::kVideoNV12;
+    capability.codecType = webrtc::kVideoCodecUnknown;
+    capability.interlaced = false;
+  } else if ([preset isEqualToString:AVCaptureSessionPreset1920x1080]) {
+    capability.width = 1920;
+    capability.height = 1080;
+    capability.maxFPS = 30;
+    capability.expectedCaptureDelay =
+        webrtc::videocapturemodule::kDefaultCaptureDelay;
+    capability.rawType = webrtc::kVideoNV12;
+    capability.codecType = webrtc::kVideoCodecUnknown;
+    capability.interlaced = false;
+  }
+
+  return capability;
+}
+
 @end
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm b/media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm
index f6302f14d7..e36c83bad9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm
@@ -17,7 +17,7 @@
 #import "webrtc/modules/video_capture/ios/device_info_ios_objc.h"
 #import "webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 using namespace webrtc::videocapturemodule;
@@ -31,6 +31,7 @@ using namespace webrtc::videocapturemodule;
   webrtc::VideoCaptureCapability _capability;
   AVCaptureSession* _captureSession;
   int _captureId;
+  BOOL _orientationHasChanged;
   AVCaptureConnection* _connection;
   BOOL _captureChanging;  // Guarded by _captureChangingCondition.
   NSCondition* _captureChangingCondition;
@@ -80,14 +81,16 @@ using namespace webrtc::videocapturemodule;
                    __LINE__);
     }
 
+    [[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
+
     NSNotificationCenter* notify = [NSNotificationCenter defaultCenter];
     [notify addObserver:self
                selector:@selector(onVideoError:)
                    name:AVCaptureSessionRuntimeErrorNotification
                  object:_captureSession];
     [notify addObserver:self
-               selector:@selector(statusBarOrientationDidChange:)
-                   name:@"StatusBarOrientationDidChange"
+               selector:@selector(deviceOrientationDidChange:)
+                   name:UIDeviceOrientationDidChangeNotification
                  object:nil];
   }
 
@@ -105,7 +108,8 @@ using namespace webrtc::videocapturemodule;
   [[self currentOutput] setSampleBufferDelegate:nil queue:NULL];
 }
 
-- (void)statusBarOrientationDidChange:(NSNotification*)notification {
+- (void)deviceOrientationDidChange:(NSNotification*)notification {
+  _orientationHasChanged = YES;
   [self setRelativeVideoOrientation];
 }
 
@@ -171,6 +175,7 @@ using namespace webrtc::videocapturemodule;
 
   [self directOutputToSelf];
 
+  _orientationHasChanged = NO;
   _captureChanging = YES;
   dispatch_async(
       dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0),
@@ -238,24 +243,34 @@ using namespace webrtc::videocapturemodule;
 }
 
 - (void)setRelativeVideoOrientation {
-  if (!_connection.supportsVideoOrientation)
+  if (!_connection.supportsVideoOrientation) {
     return;
-  switch ([UIApplication sharedApplication].statusBarOrientation) {
-    case UIInterfaceOrientationPortrait:
-#if defined(__IPHONE_8_0) && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_8_0
-    case UIInterfaceOrientationUnknown:
-#endif
-      _connection.videoOrientation = AVCaptureVideoOrientationPortrait;
+  }
+
+  switch ([UIDevice currentDevice].orientation) {
+    case UIDeviceOrientationPortrait:
+      _connection.videoOrientation =
+          AVCaptureVideoOrientationPortrait;
       break;
-    case UIInterfaceOrientationPortraitUpsideDown:
+    case UIDeviceOrientationPortraitUpsideDown:
       _connection.videoOrientation =
           AVCaptureVideoOrientationPortraitUpsideDown;
       break;
-    case UIInterfaceOrientationLandscapeLeft:
-      _connection.videoOrientation = AVCaptureVideoOrientationLandscapeLeft;
+    case UIDeviceOrientationLandscapeLeft:
+      _connection.videoOrientation =
+          AVCaptureVideoOrientationLandscapeRight;
       break;
-    case UIInterfaceOrientationLandscapeRight:
-      _connection.videoOrientation = AVCaptureVideoOrientationLandscapeRight;
+    case UIDeviceOrientationLandscapeRight:
+      _connection.videoOrientation =
+          AVCaptureVideoOrientationLandscapeLeft;
+      break;
+    case UIDeviceOrientationFaceUp:
+    case UIDeviceOrientationFaceDown:
+    case UIDeviceOrientationUnknown:
+      if (!_orientationHasChanged) {
+        _connection.videoOrientation =
+            AVCaptureVideoOrientationPortrait;
+      }
       break;
   }
 }
@@ -273,6 +288,8 @@ using namespace webrtc::videocapturemodule;
 }
 
 - (BOOL)stopCapture {
+  [[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
+  _orientationHasChanged = NO;
   [self waitForCaptureChangeToFinish];
   [self directOutputToNil];
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios.mm b/media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios.mm
index e9c77631c0..ae9b7e0805 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios.mm
@@ -14,9 +14,8 @@
 
 #include "webrtc/modules/video_capture/ios/device_info_ios_objc.h"
 #include "webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 using namespace videocapturemodule;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc b/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc
index d5924399c0..7df188504f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc
@@ -26,8 +26,8 @@
 #include <linux/videodev2.h>
 #endif
 
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #ifdef WEBRTC_LINUX
 #define EVENT_SIZE  ( sizeof (struct inotify_event) )
@@ -166,17 +166,17 @@ bool DeviceInfoLinux::InotifyProcess()
 DeviceInfoLinux::DeviceInfoLinux(const int32_t id)
     : DeviceInfoImpl(id)
 #ifdef WEBRTC_LINUX
+    , _inotifyEventThread(new rtc::PlatformThread(
+                            InotifyEventThread, this, "InotifyEventThread"))
     , _isShutdown(0)
 #endif
 {
 #ifdef WEBRTC_LINUX
-    _inotifyEventThread = ThreadWrapper::CreateThread(
-        InotifyEventThread, this, "InotifyEventThread");
 
     if (_inotifyEventThread)
     {
         _inotifyEventThread->Start();
-        _inotifyEventThread->SetPriority(kHighPriority);
+        _inotifyEventThread->SetPriority(rtc::kHighPriority);
     }
 #endif
 }
@@ -228,7 +228,7 @@ int32_t DeviceInfoLinux::GetDeviceName(
                                          uint32_t deviceUniqueIdUTF8Length,
                                          char* /*productUniqueIdUTF8*/,
                                          uint32_t /*productUniqueIdUTF8Length*/,
-                                         pid_t* pid)
+                                         pid_t* /*pid*/)
 {
     WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__);
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.h b/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.h
index 2fb6965b63..300250d1ef 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.h
@@ -14,8 +14,8 @@
 #include "webrtc/modules/video_capture/device_info_impl.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
 #ifdef WEBRTC_LINUX
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 #include <sys/inotify.h>
 #endif
 
@@ -59,7 +59,7 @@ private:
     int EventCheck();
     int HandleEvents();
     int ProcessInotifyEvents();
-    rtc::scoped_ptr<ThreadWrapper> _inotifyEventThread;
+    rtc::scoped_ptr<rtc::PlatformThread> _inotifyEventThread;
     static bool InotifyEventThread(void*);
     bool InotifyProcess();
     int _fd, _wd_v4l, _wd_snd; /* accessed on InotifyEventThread thread */
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc b/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc
index 67a2bd1458..60d7acb2d9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc
@@ -16,7 +16,6 @@
 #include <sys/mman.h>
 #include <sys/stat.h>
 #include <unistd.h>
-
 //v4l includes
 #if defined(__NetBSD__) || defined(__OpenBSD__)
 #include <sys/videoio.h>
@@ -29,9 +28,9 @@
 #include <new>
 
 #include "webrtc/modules/video_capture/linux/video_capture_linux.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc
 {
@@ -294,10 +293,10 @@ int32_t VideoCaptureModuleV4L2::StartCapture(
     //start capture thread;
     if (!_captureThread)
     {
-        _captureThread = ThreadWrapper::CreateThread(
-            VideoCaptureModuleV4L2::CaptureThread, this, "CaptureThread");
+        _captureThread.reset(new rtc::PlatformThread(
+            VideoCaptureModuleV4L2::CaptureThread, this, "CaptureThread"));
         _captureThread->Start();
-        _captureThread->SetPriority(kHighPriority);
+        _captureThread->SetPriority(rtc::kHighPriority);
     }
 
     // Needed to start UVC camera - from the uvcview application
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.h b/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.h
index b2e0f813c0..8172eb8d2a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
 #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 namespace webrtc
 {
@@ -39,7 +39,8 @@ private:
     bool AllocateVideoBuffers();
     bool DeAllocateVideoBuffers();
 
-    rtc::scoped_ptr<ThreadWrapper> _captureThread;
+    // TODO(pbos): Stop using scoped_ptr and resetting the thread.
+    rtc::scoped_ptr<rtc::PlatformThread> _captureThread;
     CriticalSectionWrapper* _captureCritSect;
 
     int32_t _deviceId;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation.mm b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation.mm
index e2356a337f..9f333781eb 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation.mm
@@ -12,8 +12,8 @@
 #import "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.h"
 #import "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_objc.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 class nsAutoreleasePool {
 public:
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info.mm b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info.mm
index fdbafb0e13..f68553defd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info.mm
@@ -9,9 +9,9 @@
  */
 
 #import "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "nsDebug.h"
 
 namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.mm b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.mm
index 631d5125a7..fd1dbb2ebc 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.mm
@@ -12,7 +12,7 @@
 
 #import "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info_objc.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 using namespace videocapturemodule;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_objc.mm b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_objc.mm
index dc6b696f5c..d930b12863 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_objc.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_objc.mm
@@ -17,7 +17,7 @@
 
 #import "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_objc.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 using namespace videocapturemodule;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/mac/video_capture_mac.mm b/media/webrtc/trunk/webrtc/modules/video_capture/mac/video_capture_mac.mm
index fa1a55aacd..41f76844bd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/mac/video_capture_mac.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/mac/video_capture_mac.mm
@@ -16,8 +16,8 @@
 #include "webrtc/modules/video_capture/device_info_impl.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation.h"
 #include "webrtc/modules/video_capture/mac/avfoundation/video_capture_avfoundation_info.h"
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/test/video_capture_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_capture/test/video_capture_unittest.cc
index 2470b2d22e..45d2d2f241 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -15,17 +15,15 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_capture/ensure_initialized.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_capture/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video_frame.h"
 
 using rtc::scoped_ptr;
 using webrtc::CriticalSectionWrapper;
@@ -48,14 +46,14 @@ using webrtc::VideoCaptureModule;
       SleepMs(5); \
       res = (ex); \
     } \
-  } while (0);\
+  } while (0)
 
 #define EXPECT_TRUE_WAIT(ex, timeout) \
   do { \
     bool res; \
     WAIT_(ex, timeout, res); \
     if (!res) EXPECT_TRUE(ex); \
-  } while (0);
+  } while (0)
 
 
 static const int kTimeOut = 5000;
@@ -64,8 +62,8 @@ static const int kTestWidth = 352;
 static const int kTestFramerate = 30;
 
 // Compares the content of two video frames.
-static bool CompareFrames(const webrtc::I420VideoFrame& frame1,
-                          const webrtc::I420VideoFrame& frame2) {
+static bool CompareFrames(const webrtc::VideoFrame& frame1,
+                          const webrtc::VideoFrame& frame2) {
   bool result =
       (frame1.stride(webrtc::kYPlane) == frame2.stride(webrtc::kYPlane)) &&
       (frame1.stride(webrtc::kUPlane) == frame2.stride(webrtc::kUPlane)) &&
@@ -104,9 +102,8 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
       printf("No of timing warnings %d\n", timing_warnings_);
   }
 
-  virtual void OnIncomingCapturedFrame(
-      const int32_t id,
-      const webrtc::I420VideoFrame& videoFrame) {
+  virtual void OnIncomingCapturedFrame(const int32_t id,
+                                       const webrtc::VideoFrame& videoFrame) {
     CriticalSectionScoped cs(capture_cs_.get());
     int height = videoFrame.height();
     int width = videoFrame.width();
@@ -175,7 +172,7 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
     return capability_;
   }
 
-  bool CompareLastFrame(const webrtc::I420VideoFrame& frame) {
+  bool CompareLastFrame(const webrtc::VideoFrame& frame) {
     CriticalSectionScoped cs(capture_cs_.get());
     return CompareFrames(last_frame_, frame);
   }
@@ -192,7 +189,7 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
   int64_t last_render_time_ms_;
   int incoming_frames_;
   int timing_warnings_;
-  webrtc::I420VideoFrame last_frame_;
+  webrtc::VideoFrame last_frame_;
   webrtc::VideoRotation rotate_frame_;
 };
 
@@ -236,14 +233,13 @@ class VideoCaptureTest : public testing::Test {
   VideoCaptureTest() : number_of_devices_(0) {}
 
   void SetUp() {
-    webrtc::videocapturemodule::EnsureInitialized();
     device_info_.reset(VideoCaptureFactory::CreateDeviceInfo(0));
     assert(device_info_.get());
     number_of_devices_ = device_info_->NumberOfDevices();
     ASSERT_GT(number_of_devices_, 0u);
   }
 
-  webrtc::scoped_refptr<VideoCaptureModule> OpenVideoCaptureDevice(
+  rtc::scoped_refptr<VideoCaptureModule> OpenVideoCaptureDevice(
       unsigned int device,
       VideoCaptureDataCallback* callback) {
     char device_name[256];
@@ -252,7 +248,7 @@ class VideoCaptureTest : public testing::Test {
     EXPECT_EQ(0, device_info_->GetDeviceName(
         device, device_name, 256, unique_name, 256));
 
-    webrtc::scoped_refptr<VideoCaptureModule> module(
+    rtc::scoped_refptr<VideoCaptureModule> module(
         VideoCaptureFactory::Create(device, unique_name));
     if (module.get() == NULL)
       return NULL;
@@ -278,12 +274,19 @@ class VideoCaptureTest : public testing::Test {
   unsigned int number_of_devices_;
 };
 
-TEST_F(VideoCaptureTest, CreateDelete) {
+#ifdef WEBRTC_MAC
+// Currently fails on Mac 64-bit, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5406
+#define MAYBE_CreateDelete DISABLED_CreateDelete
+#else
+#define MAYBE_CreateDelete CreateDelete
+#endif
+TEST_F(VideoCaptureTest, MAYBE_CreateDelete) {
   for (int i = 0; i < 5; ++i) {
     int64_t start_time = TickTime::MillisecondTimestamp();
     TestVideoCaptureCallback capture_observer;
-    webrtc::scoped_refptr<VideoCaptureModule> module(OpenVideoCaptureDevice(
-        0, &capture_observer));
+    rtc::scoped_refptr<VideoCaptureModule> module(
+        OpenVideoCaptureDevice(0, &capture_observer));
     ASSERT_TRUE(module.get() != NULL);
 
     VideoCaptureCapability capability;
@@ -315,7 +318,14 @@ TEST_F(VideoCaptureTest, CreateDelete) {
   }
 }
 
-TEST_F(VideoCaptureTest, Capabilities) {
+#ifdef WEBRTC_MAC
+// Currently fails on Mac 64-bit, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5406
+#define MAYBE_Capabilities DISABLED_Capabilities
+#else
+#define MAYBE_Capabilities Capabilities
+#endif
+TEST_F(VideoCaptureTest, MAYBE_Capabilities) {
 #ifdef WEBRTC_MAC
   printf("Video capture capabilities are not supported on Mac.\n");
   return;
@@ -323,8 +333,8 @@ TEST_F(VideoCaptureTest, Capabilities) {
 
   TestVideoCaptureCallback capture_observer;
 
-  webrtc::scoped_refptr<VideoCaptureModule> module(OpenVideoCaptureDevice(
-          0, &capture_observer));
+  rtc::scoped_refptr<VideoCaptureModule> module(
+      OpenVideoCaptureDevice(0, &capture_observer));
   ASSERT_TRUE(module.get() != NULL);
 
   int number_of_capabilities = device_info_->NumberOfCapabilities(
@@ -385,8 +395,8 @@ TEST_F(VideoCaptureTest, DISABLED_TestTwoCameras) {
   }
 
   TestVideoCaptureCallback capture_observer1;
-  webrtc::scoped_refptr<VideoCaptureModule> module1(OpenVideoCaptureDevice(
-          0, &capture_observer1));
+  rtc::scoped_refptr<VideoCaptureModule> module1(
+      OpenVideoCaptureDevice(0, &capture_observer1));
   ASSERT_TRUE(module1.get() != NULL);
   VideoCaptureCapability capability1;
 #ifndef WEBRTC_MAC
@@ -400,8 +410,8 @@ TEST_F(VideoCaptureTest, DISABLED_TestTwoCameras) {
   capture_observer1.SetExpectedCapability(capability1);
 
   TestVideoCaptureCallback capture_observer2;
-  webrtc::scoped_refptr<VideoCaptureModule> module2(OpenVideoCaptureDevice(
-          1, &capture_observer2));
+  rtc::scoped_refptr<VideoCaptureModule> module2(
+      OpenVideoCaptureDevice(1, &capture_observer2));
   ASSERT_TRUE(module1.get() != NULL);
 
 
@@ -430,7 +440,7 @@ class VideoCaptureExternalTest : public testing::Test {
  public:
   void SetUp() {
     capture_module_ = VideoCaptureFactory::Create(0, capture_input_interface_);
-    process_module_ = webrtc::ProcessThread::Create();
+    process_module_ = webrtc::ProcessThread::Create("ProcessThread");
     process_module_->Start();
     process_module_->RegisterModule(capture_module_);
 
@@ -461,9 +471,9 @@ class VideoCaptureExternalTest : public testing::Test {
   }
 
   webrtc::VideoCaptureExternal* capture_input_interface_;
-  webrtc::scoped_refptr<VideoCaptureModule> capture_module_;
+  rtc::scoped_refptr<VideoCaptureModule> capture_module_;
   rtc::scoped_ptr<webrtc::ProcessThread> process_module_;
-  webrtc::I420VideoFrame test_frame_;
+  webrtc::VideoFrame test_frame_;
   TestVideoCaptureCallback capture_callback_;
   TestVideoCaptureFeedBack capture_feedback_;
 };
@@ -482,7 +492,12 @@ TEST_F(VideoCaptureExternalTest, TestExternalCapture) {
 
 // Test frame rate and no picture alarm.
 // Flaky on Win32, see webrtc:3270.
-TEST_F(VideoCaptureExternalTest, DISABLED_ON_WIN(FrameRate)) {
+#if defined(WEBRTC_WIN)
+#define MAYBE_FrameRate DISABLED_FrameRate
+#else
+#define MAYBE_FrameRate FrameRate
+#endif
+TEST_F(VideoCaptureExternalTest, MAYBE_FrameRate) {
   int64_t testTime = 3;
   TickTime startTime = TickTime::Now();
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.gypi b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.gypi
index 55d99a413c..3c48acb72a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.gypi
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.gypi
@@ -17,7 +17,6 @@
       'type': 'static_library',
       'dependencies': [
         'webrtc_utility',
-        '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/common_video/common_video.gyp:common_video',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
@@ -27,11 +26,11 @@
       'sources': [
         'device_info_impl.cc',
         'device_info_impl.h',
-        'include/video_capture.h',
-        'include/video_capture_defines.h',
-        'include/video_capture_factory.h',
+        'video_capture.h',
         'video_capture_config.h',
+        'video_capture_defines.h',
         'video_capture_delay.h',
+        'video_capture_factory.h',
         'video_capture_factory.cc',
         'video_capture_impl.cc',
         'video_capture_impl.h',
@@ -63,19 +62,21 @@
           'dependencies': [
             'video_capture_module',
             '<(webrtc_root)/common.gyp:webrtc_common',
-          ],
-	  'cflags_mozilla': [
-	    '$(NSPR_CFLAGS)',
+           ],
+          'cflags_mozilla': [
+            '$(NSPR_CFLAGS)',
           ],
           'conditions': [
-           ['include_v4l2_video_capture==1', {
+            ['OS!="android"', {
+            }],
+            ['include_v4l2_video_capture==1', {
               'sources': [
                 'linux/device_info_linux.cc',
                 'linux/device_info_linux.h',
                 'linux/video_capture_linux.cc',
                 'linux/video_capture_linux.h',
               ],
-            }],  # linux
+            }],
             ['OS=="mac"', {
               'sources': [
                 'mac/avfoundation/video_capture_avfoundation.h',
@@ -131,6 +132,23 @@
                 ],
               },
             }],  # win
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-comment',
+                    '-Wno-ignored-attributes',
+                    '-Wno-microsoft-extra-qualification',
+                    '-Wno-missing-braces',
+                    '-Wno-overloaded-virtual',
+                    '-Wno-reorder',
+                    '-Wno-writable-strings',
+                  ],
+                },
+              },
+            }],
             ['OS=="android"', {
               'sources': [
                 'android/device_info_android.cc',
@@ -171,7 +189,7 @@
         },
       ],
     }], # build_with_chromium==0
-    ['include_tests==1', {
+    ['include_tests==1 and OS!="android"', {
       'targets': [
         {
           'target_name': 'video_capture_tests',
@@ -184,8 +202,6 @@
             '<(DEPTH)/testing/gtest.gyp:gtest',
           ],
           'sources': [
-            'ensure_initialized.cc',
-            'ensure_initialized.h',
             'test/video_capture_unittest.cc',
             'test/video_capture_main_mac.mm',
           ],
@@ -209,18 +225,6 @@
                 '-lrt',
               ],
             }],
-            ['OS=="android"', {
-              'dependencies': [
-                '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
-              ],
-              # Need to disable error due to the line in
-              # base/android/jni_android.h triggering it:
-              # const BASE_EXPORT jobject GetApplicationContext()
-              # error: type qualifiers ignored on function return type
-              'cflags': [
-                '-Wno-ignored-qualifiers',
-              ],
-            }],
             ['OS=="mac"', {
               'dependencies': [
                 # Link with a special main for mac so we can use the webcam.
@@ -242,36 +246,6 @@
           ] # conditions
         },
       ], # targets
-      'conditions': [
-        ['OS=="android"', {
-          'targets': [
-            {
-              'target_name': 'video_capture_tests_apk_target',
-              'type': 'none',
-              'dependencies': [
-                '<(apk_tests_path):video_capture_tests_apk',
-              ],
-            },
-          ],
-        }],
-        ['test_isolation_mode != "noop"', {
-          'targets': [
-            {
-              'target_name': 'video_capture_tests_run',
-              'type': 'none',
-              'dependencies': [
-                'video_capture_tests',
-              ],
-              'includes': [
-                '../../build/isolate.gypi',
-              ],
-              'sources': [
-                'video_capture_tests.isolate',
-              ],
-            },
-          ],
-        }],
-      ],
     }],
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture.h b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture.h
rename to media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h
index 9acfb1abb5..108b0f8640 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h
@@ -8,12 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
 
+#include "webrtc/common.h"
 #include "webrtc/common_video/rotation.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/video_capture/include/video_capture_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
 
 #if defined(ANDROID) && !defined(WEBRTC_GONK)
 #include <jni.h>
@@ -21,6 +22,46 @@
 
 namespace webrtc {
 
+// Mozilla addition
+enum class CaptureDeviceType {
+  Camera = 0,
+  Screen = 1,
+  Application = 2,
+  Window = 3,
+  Browser = 4
+};
+// Mozilla addition
+ 
+struct CaptureDeviceInfo {
+  CaptureDeviceType type;
+
+  CaptureDeviceInfo() : type(CaptureDeviceType::Camera) {}
+  CaptureDeviceInfo(CaptureDeviceType t) : type(t) {}
+  static const ConfigOptionID identifier = ConfigOptionID::kCaptureDeviceInfo;
+  const char * TypeName() const
+  {
+    switch(type) {
+    case CaptureDeviceType::Camera: {
+      return "Camera";
+    }
+    case CaptureDeviceType::Screen: {
+      return "Screen";
+    }
+    case CaptureDeviceType::Application: {
+      return "Application";
+    }
+    case CaptureDeviceType::Window: {
+      return "Window";
+    }
+    case CaptureDeviceType::Browser: {
+      return "Browser";
+    }
+    }
+    assert(false);
+    return "UNKOWN-CaptureDeviceType!";
+  }
+};
+
 class VideoInputFeedBack
 {
 public:
@@ -29,7 +70,7 @@ protected:
     virtual ~VideoInputFeedBack(){}
 };
 
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_GONK)
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
   int32_t SetCaptureAndroidVM(JavaVM* javaVM);
 #endif
 
@@ -58,7 +99,6 @@ class VideoCaptureModule: public RefCountedModule {
     //                      Otherwise same as deviceNameUTF8.
     // productUniqueIdUTF8 - Unique product id if it exist.
     //                       Null terminated otherwise.
-    // pid                 - Owning process id (pid).
     virtual int32_t GetDeviceName(
         uint32_t deviceNumber,
         char* deviceNameUTF8,
@@ -188,4 +228,4 @@ protected:
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
+#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_defines.h b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_defines.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_defines.h
rename to media/webrtc/trunk/webrtc/modules/video_capture/video_capture_defines.h
index ca13bc31bb..3639b537ba 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_defines.h
@@ -8,12 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
+
+#ifdef XP_WIN
+typedef int pid_t;
+#endif
 
 namespace webrtc
 {
@@ -22,10 +26,6 @@ namespace webrtc
     #define NULL    0
 #endif
 
-#ifdef XP_WIN
-typedef int pid_t;
-#endif
-
 enum {kVideoCaptureUniqueNameLength =1024}; //Max unique capture device name lenght
 enum {kVideoCaptureDeviceNameLength =256}; //Max capture device name lenght
 enum {kVideoCaptureProductIdLength =128}; //Max product id length
@@ -98,8 +98,8 @@ protected:
 class VideoCaptureDataCallback
 {
 public:
-    virtual void OnIncomingCapturedFrame(const int32_t id,
-                                         const I420VideoFrame& videoFrame) = 0;
+ virtual void OnIncomingCapturedFrame(const int32_t id,
+                                      const VideoFrame& videoFrame) = 0;
     virtual void OnCaptureDelayChanged(const int32_t id,
                                        const int32_t delay) = 0;
 protected:
@@ -119,4 +119,4 @@ protected:
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
+#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.cc b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.cc
index 5b44a6c706..a4735f083a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
 
 #include "webrtc/modules/video_capture/video_capture_impl.h"
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_factory.h b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.h
similarity index 79%
rename from media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_factory.h
rename to media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.h
index ec92d31e17..4765be1fde 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture_factory.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_factory.h
@@ -11,10 +11,10 @@
 // This file contains interfaces used for creating the VideoCaptureModule
 // and DeviceInfo.
 
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_FACTORY_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_FACTORY_H_
+#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
+#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
 
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 
 namespace webrtc {
 
@@ -36,14 +36,10 @@ class VideoCaptureFactory {
   static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(
       const int32_t id);
 
-#ifdef WEBRTC_ANDROID
-  static int32_t SetAndroidObjects(void* javaVM, void* javaContext);
-#endif
-
  private:
   ~VideoCaptureFactory();
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_FACTORY_H_
+#endif  // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.cc b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.cc
index 67ab6a31c2..913930d18b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.cc
@@ -12,15 +12,15 @@
 
 #include <stdlib.h>
 
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc
 {
@@ -211,7 +211,7 @@ int32_t VideoCaptureImpl::CaptureDelay()
     return _setCaptureDelay;
 }
 
-int32_t VideoCaptureImpl::DeliverCapturedFrame(I420VideoFrame& captureFrame) {
+int32_t VideoCaptureImpl::DeliverCapturedFrame(VideoFrame& captureFrame) {
   UpdateFrameCount();  // frame count used for local frame rate callback.
 
   const bool callOnCaptureDelayChanged = _setCaptureDelay != _captureDelay;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.h b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.h
index fd4b39894f..65c659e75a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.h
@@ -15,15 +15,16 @@
  * video_capture_impl.h
  */
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/common_video/rotation.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc
 {
+
 class CriticalSectionWrapper;
 
 namespace videocapturemodule {
@@ -104,7 +105,7 @@ public:
 protected:
     VideoCaptureImpl(const int32_t id);
     virtual ~VideoCaptureImpl();
-    int32_t DeliverCapturedFrame(I420VideoFrame& captureFrame);
+    int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
 
     int32_t _id; // Module ID
     char* _deviceUniqueId; // current Device unique name;
@@ -132,7 +133,7 @@ private:
     VideoRotation _rotateFrame;  // Set if the frame should be rotated by the
                                  // capture module.
 
-    I420VideoFrame _captureFrame;
+    VideoFrame _captureFrame;
 
     // Indicate whether rotation should be applied before delivered externally.
     bool apply_rotation_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/BaseFilter.cpp b/media/webrtc/trunk/webrtc/modules/video_capture/windows/BaseFilter.cpp
index da152a1d59..2a2bdb818b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/BaseFilter.cpp
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/BaseFilter.cpp
@@ -36,13 +36,13 @@ class DECLSPEC_UUID("0e9924bd-1cb8-48ad-ba31-2fb831b162be")
 {
 public:
 
-  EnumPins(BaseFilter* aFilter)
+  explicit EnumPins(BaseFilter* aFilter)
     : mFilter(aFilter)
     , mRefCnt(0)
   {
     Reset();
   }
-  EnumPins(EnumPins* aEnumPins)
+  explicit EnumPins(EnumPins* aEnumPins)
     : mFilter(aEnumPins->mFilter)
     , mNumPins(aEnumPins->mNumPins)
     , mPinIdx(aEnumPins->mPinIdx)
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/BasePin.cpp b/media/webrtc/trunk/webrtc/modules/video_capture/windows/BasePin.cpp
index 55bea7e7b7..50bf8d72ed 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/BasePin.cpp
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/BasePin.cpp
@@ -24,14 +24,14 @@ class DECLSPEC_UUID("4de7a03c-6c3f-4314-949a-ee7e1ad05083")
 {
 public:
 
-  EnumMediaTypes(BasePin* aPin)
+  explicit EnumMediaTypes(BasePin* aPin)
     : mPin(aPin)
     , mIndex(0)
     , mRefCnt(0)
   {
   }
 
-  EnumMediaTypes(EnumMediaTypes* aEnum)
+  explicit EnumMediaTypes(EnumMediaTypes* aEnum)
     : mPin(aEnum->mPin)
     , mIndex(aEnum->mIndex)
     , mRefCnt(0)
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/DShowTools.h b/media/webrtc/trunk/webrtc/modules/video_capture/windows/DShowTools.h
index 4bb0a4a17d..5698fa3c83 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/DShowTools.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/DShowTools.h
@@ -24,7 +24,7 @@ public:
      * CriticalSection
      * @param aName A name which can reference this monitor
      */
-    CriticalSection(const char* aName)
+    explicit CriticalSection(const char* aName)
     {
       ::InitializeCriticalSection(&mCriticalSection);
     }
@@ -78,16 +78,16 @@ public:
      * Constructor
      * The constructor aquires the given lock.  The destructor
      * releases the lock.
-     * 
-     * @param aCriticalSection A valid mozilla::CriticalSection*. 
+     *
+     * @param aCriticalSection A valid mozilla::CriticalSection*.
      **/
-    CriticalSectionAutoEnter(mozilla::CriticalSection &aCriticalSection) :
+    explicit CriticalSectionAutoEnter(mozilla::CriticalSection &aCriticalSection) :
         mCriticalSection(&aCriticalSection)
     {
         assert(mCriticalSection);
         mCriticalSection->Enter();
     }
-    
+
     ~CriticalSectionAutoEnter(void)
     {
         mCriticalSection->Leave();
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/MediaType.h b/media/webrtc/trunk/webrtc/modules/video_capture/windows/MediaType.h
index 033ce52e36..6f4660500e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/MediaType.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/MediaType.h
@@ -19,10 +19,10 @@ namespace media {
 class MediaType : public AM_MEDIA_TYPE
 {
 public:
-  
+
   MediaType();
-  MediaType(const AM_MEDIA_TYPE* aMediaType);
-  MediaType(const MediaType& aMediaType);
+  explicit MediaType(const AM_MEDIA_TYPE* aMediaType);
+  explicit MediaType(const MediaType& aMediaType);
 
   ~MediaType();
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/device_info_ds.cc b/media/webrtc/trunk/webrtc/modules/video_capture/windows/device_info_ds.cc
index e70fda43d8..e4f27812cb 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/device_info_ds.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/device_info_ds.cc
@@ -13,8 +13,8 @@
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/modules/video_capture/video_capture_delay.h"
 #include "webrtc/modules/video_capture/windows/help_functions_ds.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <Dvdmedia.h>
 #include <dbt.h>
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc b/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc
index 3d079e6620..b445d97f54 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc
@@ -11,7 +11,7 @@
 #include "webrtc/modules/video_capture/windows/sink_filter_ds.h"
 
 #include "webrtc/modules/video_capture/windows/help_functions_ds.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <Dvdmedia.h> // VIDEOINFOHEADER2
 #include <initguid.h>
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.h b/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.h
index f673208110..247ae341f7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.h
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
 #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
 
-#include "webrtc/modules/video_capture/include/video_capture_defines.h"
+#include "webrtc/modules/video_capture/video_capture_defines.h"
 #include "BaseInputPin.h"
 #include "BaseFilter.h"
 #include "MediaType.h"
@@ -81,21 +81,6 @@ public:
     {
       return mozilla::media::BaseFilter::QueryInterface(aIId, aInterface);
     }
-    STDMETHODIMP_(ULONG) AddRef()
-    {
-      return ::InterlockedIncrement(&mRefCnt);
-    }
-
-    STDMETHODIMP_(ULONG) Release()
-    {
-      unsigned long newRefCnt = ::InterlockedDecrement(&mRefCnt);
-
-      if (!newRefCnt) {
-        delete this;
-      }
-
-      return newRefCnt;
-    }
 
     STDMETHODIMP SetMatchingMediaType(const VideoCaptureCapability& capability);
 
@@ -115,7 +100,6 @@ private:
     CaptureInputPin * m_pInput;
     VideoCaptureExternal& _captureObserver;
     int32_t _moduleId;
-    unsigned long mRefCnt;
 };
 }  // namespace videocapturemodule
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_ds.cc b/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_ds.cc
index 96b63f6f82..bdd51a7954 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_ds.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_ds.cc
@@ -13,8 +13,8 @@
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/modules/video_capture/windows/help_functions_ds.h"
 #include "webrtc/modules/video_capture/windows/sink_filter_ds.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <Dvdmedia.h> // VIDEOINFOHEADER2
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_factory_windows.cc b/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_factory_windows.cc
index 89c0297c8f..747d3d60cf 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_factory_windows.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_capture/windows/video_capture_factory_windows.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/modules/video_capture/windows/video_capture_ds.h"
 #include "webrtc/modules/video_capture/windows/video_capture_mf.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
 
 namespace webrtc {
 namespace videocapturemodule {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/BUILD.gn b/media/webrtc/trunk/webrtc/modules/video_coding/BUILD.gn
index 4bcbe29d73..32ac627ed2 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/BUILD.gn
@@ -10,58 +10,57 @@ import("../../build/webrtc.gni")
 
 source_set("video_coding") {
   sources = [
-    "main/interface/video_coding.h",
-    "main/interface/video_coding_defines.h",
-    "main/source/codec_database.cc",
-    "main/source/codec_database.h",
-    "main/source/codec_timer.cc",
-    "main/source/codec_timer.h",
-    "main/source/content_metrics_processing.cc",
-    "main/source/content_metrics_processing.h",
-    "main/source/decoding_state.cc",
-    "main/source/decoding_state.h",
-    "main/source/encoded_frame.cc",
-    "main/source/encoded_frame.h",
-    "main/source/er_tables_xor.h",
-    "main/source/fec_tables_xor.h",
-    "main/source/frame_buffer.cc",
-    "main/source/frame_buffer.h",
-    "main/source/generic_decoder.cc",
-    "main/source/generic_decoder.h",
-    "main/source/generic_encoder.cc",
-    "main/source/generic_encoder.h",
-    "main/source/inter_frame_delay.cc",
-    "main/source/inter_frame_delay.h",
-    "main/source/internal_defines.h",
-    "main/source/jitter_buffer.cc",
-    "main/source/jitter_buffer.h",
-    "main/source/jitter_buffer_common.h",
-    "main/source/jitter_estimator.cc",
-    "main/source/jitter_estimator.h",
-    "main/source/media_opt_util.cc",
-    "main/source/media_opt_util.h",
-    "main/source/media_optimization.cc",
-    "main/source/media_optimization.h",
-    "main/source/nack_fec_tables.h",
-    "main/source/packet.cc",
-    "main/source/packet.h",
-    "main/source/qm_select_data.h",
-    "main/source/qm_select.cc",
-    "main/source/qm_select.h",
-    "main/source/receiver.cc",
-    "main/source/receiver.h",
-    "main/source/rtt_filter.cc",
-    "main/source/rtt_filter.h",
-    "main/source/session_info.cc",
-    "main/source/session_info.h",
-    "main/source/timestamp_map.cc",
-    "main/source/timestamp_map.h",
-    "main/source/timing.cc",
-    "main/source/timing.h",
-    "main/source/video_coding_impl.cc",
-    "main/source/video_coding_impl.h",
-    "main/source/video_receiver.cc",
-    "main/source/video_sender.cc",
+    "codec_database.cc",
+    "codec_database.h",
+    "codec_timer.cc",
+    "codec_timer.h",
+    "content_metrics_processing.cc",
+    "content_metrics_processing.h",
+    "decoding_state.cc",
+    "decoding_state.h",
+    "encoded_frame.cc",
+    "encoded_frame.h",
+    "fec_tables_xor.h",
+    "frame_buffer.cc",
+    "frame_buffer.h",
+    "generic_decoder.cc",
+    "generic_decoder.h",
+    "generic_encoder.cc",
+    "generic_encoder.h",
+    "include/video_coding.h",
+    "include/video_coding_defines.h",
+    "inter_frame_delay.cc",
+    "inter_frame_delay.h",
+    "internal_defines.h",
+    "jitter_buffer.cc",
+    "jitter_buffer.h",
+    "jitter_buffer_common.h",
+    "jitter_estimator.cc",
+    "jitter_estimator.h",
+    "media_opt_util.cc",
+    "media_opt_util.h",
+    "media_optimization.cc",
+    "media_optimization.h",
+    "nack_fec_tables.h",
+    "packet.cc",
+    "packet.h",
+    "qm_select.cc",
+    "qm_select.h",
+    "qm_select_data.h",
+    "receiver.cc",
+    "receiver.h",
+    "rtt_filter.cc",
+    "rtt_filter.h",
+    "session_info.cc",
+    "session_info.h",
+    "timestamp_map.cc",
+    "timestamp_map.h",
+    "timing.cc",
+    "timing.h",
+    "video_coding_impl.cc",
+    "video_coding_impl.h",
+    "video_receiver.cc",
+    "video_sender.cc",
   ]
 
   configs += [ "../..:common_config" ]
@@ -82,6 +81,7 @@ source_set("video_coding") {
 
   deps = [
     ":video_coding_utility",
+    ":webrtc_h264",
     ":webrtc_i420",
     ":webrtc_vp8",
     ":webrtc_vp9",
@@ -94,9 +94,14 @@ source_set("video_coding") {
 source_set("video_coding_utility") {
   sources = [
     "utility/frame_dropper.cc",
-    "utility/include/frame_dropper.h",
+    "utility/frame_dropper.h",
+    "utility/moving_average.h",
+    "utility/qp_parser.cc",
+    "utility/qp_parser.h",
     "utility/quality_scaler.cc",
     "utility/quality_scaler.h",
+    "utility/vp8_header_parser.cc",
+    "utility/vp8_header_parser.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -108,13 +113,50 @@ source_set("video_coding_utility") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  deps = [ "../../system_wrappers" ]
+  deps = [
+    "../../system_wrappers",
+  ]
 }
 
+source_set("webrtc_h264") {
+  sources = [
+    "codecs/h264/h264.cc",
+    "codecs/h264/include/h264.h",
+  ]
+
+  configs += [ "../..:common_config" ]
+  public_configs = [ "../..:common_inherited_config" ]
+
+  if (is_clang) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+
+  deps = [
+    "../../system_wrappers",
+  ]
+
+  if (use_third_party_h264) {
+    # Dependency added so that variables use_openh264 and ffmpeg_branding are
+    # recognized build arguments (avoid "Build argument has no effect" error).
+    # The variables and dependencies will be used for real as soon as
+    # https://codereview.webrtc.org/1306813009/ lands. In the meantime, the
+    # build arguments are to be used by waterfall/trybots.
+    deps += [
+      "//third_party/ffmpeg:ffmpeg",
+      "//third_party/openh264:encoder",
+    ]
+  }
+}
+
+# TODO(tkchin): Source set for webrtc_h264_video_toolbox. Currently not
+# possible to add, see https://crbug.com/297668.
+
 source_set("webrtc_i420") {
   sources = [
-    "codecs/i420/main/source/i420.cc",
-    "codecs/i420/main/interface/i420.h",
+    "codecs/i420/i420.cc",
+    "codecs/i420/include/i420.h",
   ]
 
   configs += [ "../..:common_config" ]
@@ -126,7 +168,9 @@ source_set("webrtc_i420") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  deps = [ "../../system_wrappers" ]
+  deps = [
+    "../../system_wrappers",
+  ]
 }
 
 source_set("webrtc_vp8") {
@@ -172,22 +216,20 @@ source_set("webrtc_vp8") {
     "../../system_wrappers",
   ]
   if (rtc_build_libvpx) {
-    deps += [
-      rtc_libvpx_dir,
-    ]
+    deps += [ rtc_libvpx_dir ]
   }
 }
 
 source_set("webrtc_vp9") {
-  if (rtc_build_vp9) {
-    sources = [
-      "codecs/vp9/include/vp9.h",
-      "codecs/vp9/vp9_impl.cc",
-      "codecs/vp9/vp9_impl.h",
-    ]
-  } else {
-    sources = [ "codecs/vp9/vp9_dummy_impl.cc" ]
-  }
+  sources = [
+    "codecs/vp9/include/vp9.h",
+    "codecs/vp9/screenshare_layers.cc",
+    "codecs/vp9/screenshare_layers.h",
+    "codecs/vp9/vp9_frame_buffer_pool.cc",
+    "codecs/vp9/vp9_frame_buffer_pool.h",
+    "codecs/vp9/vp9_impl.cc",
+    "codecs/vp9/vp9_impl.h",
+  ]
 
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
@@ -204,8 +246,6 @@ source_set("webrtc_vp9") {
     "../../system_wrappers",
   ]
   if (rtc_build_libvpx) {
-    deps += [
-      rtc_libvpx_dir,
-    ]
+    deps += [ rtc_libvpx_dir ]
   }
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/OWNERS b/media/webrtc/trunk/webrtc/modules/video_coding/OWNERS
index f452c9ed83..389d632dfd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/OWNERS
@@ -1,4 +1,9 @@
 stefan@webrtc.org
 marpan@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codec_database.cc
similarity index 67%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/codec_database.cc
index 9a32fd27ed..43f4263498 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codec_database.cc
@@ -8,26 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/codec_database.h"
+#include "webrtc/modules/video_coding/codec_database.h"
 
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/engine_configurations.h"
-#ifdef VIDEOCODEC_I420
-#include "webrtc/modules/video_coding/codecs/i420/main/interface/i420.h"
-#endif
-#ifdef VIDEOCODEC_VP8
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+#include "webrtc/modules/video_coding/codecs/i420/include/i420.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#endif
-#ifdef VIDEOCODEC_VP9
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
-#endif
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
 
 namespace {
 const size_t kDefaultPayloadSize = 1440;
+const uint8_t kDefaultPayloadType = 100;
 }
 
 namespace webrtc {
@@ -70,9 +66,9 @@ VideoCodecH264 VideoEncoder::GetDefaultH264Settings() {
   h264_settings.profile = kProfileBase;
   h264_settings.frameDroppingOn = true;
   h264_settings.keyFrameInterval = 3000;
-  h264_settings.spsData = NULL;
+  h264_settings.spsData = nullptr;
   h264_settings.spsLen = 0;
-  h264_settings.ppsData = NULL;
+  h264_settings.ppsData = nullptr;
   h264_settings.ppsLen = 0;
 
   return h264_settings;
@@ -89,57 +85,45 @@ VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
 
 VCMExtDecoderMapItem::VCMExtDecoderMapItem(
     VideoDecoder* external_decoder_instance,
-    uint8_t payload_type,
-    bool internal_render_timing)
+    uint8_t payload_type)
     : payload_type(payload_type),
-      external_decoder_instance(external_decoder_instance),
-      internal_render_timing(internal_render_timing) {
-}
+      external_decoder_instance(external_decoder_instance) {}
 
 VCMCodecDataBase::VCMCodecDataBase(
-    VideoEncoderRateObserver* encoder_rate_observer)
+    VideoEncoderRateObserver* encoder_rate_observer,
+    VCMEncodedFrameCallback* encoded_frame_callback)
     : number_of_cores_(0),
       max_payload_size_(kDefaultPayloadSize),
       periodic_key_frames_(false),
       pending_encoder_reset_(true),
-      current_enc_is_external_(false),
       send_codec_(),
       receive_codec_(),
-      external_payload_type_(0),
-      external_encoder_(NULL),
+      encoder_payload_type_(0),
+      external_encoder_(nullptr),
       internal_source_(false),
       encoder_rate_observer_(encoder_rate_observer),
-      ptr_encoder_(NULL),
-      ptr_decoder_(NULL),
+      encoded_frame_callback_(encoded_frame_callback),
+      ptr_decoder_(nullptr),
       dec_map_(),
-      dec_external_map_() {
-}
+      dec_external_map_() {}
 
 VCMCodecDataBase::~VCMCodecDataBase() {
-  ResetSender();
-  ResetReceiver();
+  DeleteEncoder();
+  ReleaseDecoder(ptr_decoder_);
+  for (auto& kv : dec_map_)
+    delete kv.second;
+  for (auto& kv : dec_external_map_)
+    delete kv.second;
 }
 
-int VCMCodecDataBase::NumberOfCodecs() {
-  return VCM_NUM_VIDEO_CODECS_AVAILABLE;
-}
-
-bool VCMCodecDataBase::Codec(int list_id,
-                             VideoCodec* settings) {
-  if (!settings) {
-    return false;
-  }
-  if (list_id >= VCM_NUM_VIDEO_CODECS_AVAILABLE) {
-    return false;
-  }
+void VCMCodecDataBase::Codec(VideoCodecType codec_type, VideoCodec* settings) {
   memset(settings, 0, sizeof(VideoCodec));
-  switch (list_id) {
-#ifdef VIDEOCODEC_VP8
-    case VCM_VP8_IDX: {
+  switch (codec_type) {
+    case kVideoCodecVP8:
       strncpy(settings->plName, "VP8", 4);
       settings->codecType = kVideoCodecVP8;
       // 96 to 127 dynamic payload types for video codecs.
-      settings->plType = VCM_VP8_PAYLOAD_TYPE;
+      settings->plType = kDefaultPayloadType;
       settings->startBitrate = kDefaultStartBitrateKbps;
       settings->minBitrate = VCM_MIN_BITRATE;
       settings->maxBitrate = 0;
@@ -151,15 +135,12 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       settings->qpMax = 56;
       settings->codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
-      return true;
-    }
-#endif
-#ifdef VIDEOCODEC_VP9
-    case VCM_VP9_IDX: {
+      return;
+    case kVideoCodecVP9:
       strncpy(settings->plName, "VP9", 4);
       settings->codecType = kVideoCodecVP9;
       // 96 to 127 dynamic payload types for video codecs.
-      settings->plType = VCM_VP9_PAYLOAD_TYPE;
+      settings->plType = kDefaultPayloadType;
       settings->startBitrate = 100;
       settings->minBitrate = VCM_MIN_BITRATE;
       settings->maxBitrate = 0;
@@ -171,15 +152,12 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       settings->qpMax = 56;
       settings->codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
-      return true;
-    }
-#endif
-#ifdef VIDEOCODEC_H264
-    case VCM_H264_IDX: {
+      return;
+    case kVideoCodecH264:
       strncpy(settings->plName, "H264", 5);
       settings->codecType = kVideoCodecH264;
       // 96 to 127 dynamic payload types for video codecs.
-      settings->plType = VCM_H264_PAYLOAD_TYPE;
+      settings->plType = kDefaultPayloadType;
       settings->startBitrate = kDefaultStartBitrateKbps;
       settings->minBitrate = VCM_MIN_BITRATE;
       settings->maxBitrate = 0;
@@ -191,15 +169,12 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       settings->qpMax = 56;
       settings->codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
-      return true;
-    }
-#endif
-#ifdef VIDEOCODEC_I420
-    case VCM_I420_IDX: {
+      return;
+    case kVideoCodecI420:
       strncpy(settings->plName, "I420", 5);
       settings->codecType = kVideoCodecI420;
       // 96 to 127 dynamic payload types for video codecs.
-      settings->plType = VCM_I420_PAYLOAD_TYPE;
+      settings->plType = kDefaultPayloadType;
       // Bitrate needed for this size and framerate.
       settings->startBitrate = 3 * VCM_DEFAULT_CODEC_WIDTH *
                                VCM_DEFAULT_CODEC_HEIGHT * 8 *
@@ -213,49 +188,29 @@ bool VCMCodecDataBase::Codec(int list_id,
       settings->numberOfSimulcastStreams = 0;
       // consider using 2 to avoid deal with 'odd' downscales
       settings->resolution_divisor = 1; // may not actually be needed
-      return true;
-    }
-#endif
-    default: {
-      return false;
-    }
+      return;
+    case kVideoCodecRED:
+    case kVideoCodecULPFEC:
+    case kVideoCodecGeneric:
+    case kVideoCodecUnknown:
+      RTC_NOTREACHED();
+      return;
   }
 }
 
-bool VCMCodecDataBase::Codec(VideoCodecType codec_type,
-                             VideoCodec* settings) {
-  for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++) {
-    const bool ret = VCMCodecDataBase::Codec(i, settings);
-    if (!ret) {
-      return false;
-    }
-    if (codec_type == settings->codecType) {
-      return true;
-    }
-  }
-  return false;
-}
-
-void VCMCodecDataBase::ResetSender() {
-  DeleteEncoder();
-  periodic_key_frames_ = false;
-}
-
 // Assuming only one registered encoder - since only one used, no need for more.
-bool VCMCodecDataBase::SetSendCodec(
-    const VideoCodec* send_codec,
-    int number_of_cores,
-    size_t max_payload_size,
-    VCMEncodedFrameCallback* encoded_frame_callback) {
-  DCHECK(send_codec);
+bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec,
+                                    int number_of_cores,
+                                    size_t max_payload_size) {
+  RTC_DCHECK(send_codec);
   if (max_payload_size == 0) {
     max_payload_size = kDefaultPayloadSize;
   }
-  DCHECK_GE(number_of_cores, 1);
-  DCHECK_GE(send_codec->plType, 1);
+  RTC_DCHECK_GE(number_of_cores, 1);
+  RTC_DCHECK_GE(send_codec->plType, 1);
   // Make sure the start bit rate is sane...
-  DCHECK_LE(send_codec->startBitrate, 1000000u);
-  DCHECK(send_codec->codecType != kVideoCodecUnknown);
+  RTC_DCHECK_LE(send_codec->startBitrate, 1000000u);
+  RTC_DCHECK(send_codec->codecType != kVideoCodecUnknown);
   bool reset_required = pending_encoder_reset_;
   if (number_of_cores_ != number_of_cores) {
     number_of_cores_ = number_of_cores;
@@ -272,8 +227,9 @@ bool VCMCodecDataBase::SetSendCodec(
   if (new_send_codec.maxBitrate == 0) {
     // max is one bit per pixel
     new_send_codec.maxBitrate = (static_cast<int>(send_codec->height) *
-        static_cast<int>(send_codec->width) *
-        static_cast<int>(send_codec->maxFramerate)) / 1000;
+                                 static_cast<int>(send_codec->width) *
+                                 static_cast<int>(send_codec->maxFramerate)) /
+                                1000;
     if (send_codec->startBitrate > new_send_codec.maxBitrate) {
       // But if the user tries to set a higher start bit rate we will
       // increase the max accordingly.
@@ -291,37 +247,24 @@ bool VCMCodecDataBase::SetSendCodec(
   memcpy(&send_codec_, &new_send_codec, sizeof(send_codec_));
 
   if (!reset_required) {
-    encoded_frame_callback->SetPayloadType(send_codec_.plType);
-    if (ptr_encoder_->RegisterEncodeCallback(encoded_frame_callback) < 0) {
-      LOG(LS_ERROR) << "Failed to register encoded-frame callback.";
-      return false;
-    }
+    encoded_frame_callback_->SetPayloadType(send_codec_.plType);
     return true;
   }
 
   // If encoder exists, will destroy it and create new one.
   DeleteEncoder();
-  if (send_codec_.plType == external_payload_type_) {
-    // External encoder.
-    ptr_encoder_ = new VCMGenericEncoder(
-        external_encoder_, encoder_rate_observer_, internal_source_);
-    current_enc_is_external_ = true;
-  } else {
-    ptr_encoder_ = CreateEncoder(send_codec_.codecType);
-    current_enc_is_external_ = false;
-    if (!ptr_encoder_)
-      return false;
-  }
-  encoded_frame_callback->SetPayloadType(send_codec_.plType);
+  RTC_DCHECK_EQ(encoder_payload_type_, send_codec_.plType)
+      << "Encoder not registered for payload type " << send_codec_.plType;
+  ptr_encoder_.reset(
+      new VCMGenericEncoder(external_encoder_, encoder_rate_observer_,
+                            encoded_frame_callback_, internal_source_));
+  encoded_frame_callback_->SetPayloadType(send_codec_.plType);
+  encoded_frame_callback_->SetInternalSource(internal_source_);
   if (ptr_encoder_->InitEncode(&send_codec_, number_of_cores_,
                                max_payload_size_) < 0) {
     LOG(LS_ERROR) << "Failed to initialize video encoder.";
     DeleteEncoder();
     return false;
-  } else if (ptr_encoder_->RegisterEncodeCallback(encoded_frame_callback) < 0) {
-    LOG(LS_ERROR) << "Failed to register encoded-frame callback.";
-    DeleteEncoder();
-    return false;
   }
 
   // Intentionally don't check return value since the encoder registration
@@ -349,42 +292,39 @@ VideoCodecType VCMCodecDataBase::SendCodec() const {
   return send_codec_.codecType;
 }
 
-bool VCMCodecDataBase::DeregisterExternalEncoder(
-    uint8_t payload_type, bool* was_send_codec) {
+bool VCMCodecDataBase::DeregisterExternalEncoder(uint8_t payload_type,
+                                                 bool* was_send_codec) {
   assert(was_send_codec);
   *was_send_codec = false;
-  if (external_payload_type_ != payload_type) {
+  if (encoder_payload_type_ != payload_type) {
     return false;
   }
   if (send_codec_.plType == payload_type) {
     // De-register as send codec if needed.
     DeleteEncoder();
     memset(&send_codec_, 0, sizeof(VideoCodec));
-    current_enc_is_external_ = false;
     *was_send_codec = true;
   }
-  external_payload_type_ = 0;
-  external_encoder_ = NULL;
+  encoder_payload_type_ = 0;
+  external_encoder_ = nullptr;
   internal_source_ = false;
   return true;
 }
 
-void VCMCodecDataBase::RegisterExternalEncoder(
-    VideoEncoder* external_encoder,
-    uint8_t payload_type,
-    bool internal_source) {
+void VCMCodecDataBase::RegisterExternalEncoder(VideoEncoder* external_encoder,
+                                               uint8_t payload_type,
+                                               bool internal_source) {
   // Since only one encoder can be used at a given time, only one external
   // encoder can be registered/used.
   external_encoder_ = external_encoder;
-  external_payload_type_ = payload_type;
+  encoder_payload_type_ = payload_type;
   internal_source_ = internal_source;
   pending_encoder_reset_ = true;
 }
 
 bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) {
-  if (ptr_encoder_ == NULL) {
+  if (!ptr_encoder_)
     return true;
-  }
 
   // Does not check startBitrate or maxFramerate
   if (new_send_codec.codecType != send_codec_.codecType ||
@@ -442,8 +382,7 @@ bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) {
          ++i) {
       if (memcmp(&new_send_codec.simulcastStream[i],
                  &send_codec_.simulcastStream[i],
-                 sizeof(new_send_codec.simulcastStream[i])) !=
-          0) {
+                 sizeof(new_send_codec.simulcastStream[i])) != 0) {
         return true;
       }
     }
@@ -452,7 +391,7 @@ bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) {
 }
 
 VCMGenericEncoder* VCMCodecDataBase::GetEncoder() {
-  return ptr_encoder_;
+  return ptr_encoder_.get();
 }
 
 bool VCMCodecDataBase::SetPeriodicKeyFrames(bool enable) {
@@ -463,22 +402,6 @@ bool VCMCodecDataBase::SetPeriodicKeyFrames(bool enable) {
   return true;
 }
 
-void VCMCodecDataBase::ResetReceiver() {
-  ReleaseDecoder(ptr_decoder_);
-  ptr_decoder_ = NULL;
-  memset(&receive_codec_, 0, sizeof(VideoCodec));
-  while (!dec_map_.empty()) {
-    DecoderMap::iterator it = dec_map_.begin();
-    delete (*it).second;
-    dec_map_.erase(it);
-  }
-  while (!dec_external_map_.empty()) {
-    ExternalDecoderMap::iterator external_it = dec_external_map_.begin();
-    delete (*external_it).second;
-    dec_external_map_.erase(external_it);
-  }
-}
-
 bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
   ExternalDecoderMap::iterator it = dec_external_map_.find(payload_type);
   if (it == dec_external_map_.end()) {
@@ -488,43 +411,36 @@ bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
   // We can't use payload_type to check if the decoder is currently in use,
   // because payload type may be out of date (e.g. before we decode the first
   // frame after RegisterReceiveCodec)
-  if (ptr_decoder_ != NULL &&
-      &ptr_decoder_->_decoder == (*it).second->external_decoder_instance) {
+  if (ptr_decoder_ != nullptr &&
+      ptr_decoder_->_decoder == (*it).second->external_decoder_instance) {
     // Release it if it was registered and in use.
     ReleaseDecoder(ptr_decoder_);
-    ptr_decoder_ = NULL;
+    ptr_decoder_ = nullptr;
   }
   DeregisterReceiveCodec(payload_type);
-  delete (*it).second;
+  delete it->second;
   dec_external_map_.erase(it);
   return true;
 }
 
 // Add the external encoder object to the list of external decoders.
 // Won't be registered as a receive codec until RegisterReceiveCodec is called.
-bool VCMCodecDataBase::RegisterExternalDecoder(
-    VideoDecoder* external_decoder,
-    uint8_t payload_type,
-    bool internal_render_timing) {
+void VCMCodecDataBase::RegisterExternalDecoder(VideoDecoder* external_decoder,
+                                               uint8_t payload_type) {
   // Check if payload value already exists, if so  - erase old and insert new.
-  VCMExtDecoderMapItem* ext_decoder = new VCMExtDecoderMapItem(
-      external_decoder, payload_type, internal_render_timing);
-  if (!ext_decoder) {
-    return false;
-  }
+  VCMExtDecoderMapItem* ext_decoder =
+      new VCMExtDecoderMapItem(external_decoder, payload_type);
   DeregisterExternalDecoder(payload_type);
   dec_external_map_[payload_type] = ext_decoder;
-  return true;
 }
 
 bool VCMCodecDataBase::DecoderRegistered() const {
   return !dec_map_.empty();
 }
 
-bool VCMCodecDataBase::RegisterReceiveCodec(
-    const VideoCodec* receive_codec,
-    int number_of_cores,
-    bool require_key_frame) {
+bool VCMCodecDataBase::RegisterReceiveCodec(const VideoCodec* receive_codec,
+                                            int number_of_cores,
+                                            bool require_key_frame) {
   if (number_of_cores < 0) {
     return false;
   }
@@ -534,20 +450,17 @@ bool VCMCodecDataBase::RegisterReceiveCodec(
     return false;
   }
   VideoCodec* new_receive_codec = new VideoCodec(*receive_codec);
-  dec_map_[receive_codec->plType] = new VCMDecoderMapItem(new_receive_codec,
-                                                          number_of_cores,
-                                                          require_key_frame);
+  dec_map_[receive_codec->plType] = new VCMDecoderMapItem(
+      new_receive_codec, number_of_cores, require_key_frame);
   return true;
 }
 
-bool VCMCodecDataBase::DeregisterReceiveCodec(
-    uint8_t payload_type) {
+bool VCMCodecDataBase::DeregisterReceiveCodec(uint8_t payload_type) {
   DecoderMap::iterator it = dec_map_.find(payload_type);
   if (it == dec_map_.end()) {
     return false;
   }
-  VCMDecoderMapItem* dec_item = (*it).second;
-  delete dec_item;
+  delete it->second;
   dec_map_.erase(it);
   if (receive_codec_.plType == payload_type) {
     // This codec is currently in use.
@@ -573,52 +486,50 @@ VideoCodecType VCMCodecDataBase::ReceiveCodec() const {
 }
 
 VCMGenericDecoder* VCMCodecDataBase::GetDecoder(
-    uint8_t payload_type, VCMDecodedFrameCallback* decoded_frame_callback) {
+    const VCMEncodedFrame& frame,
+    VCMDecodedFrameCallback* decoded_frame_callback) {
+  uint8_t payload_type = frame.PayloadType();
   if (payload_type == receive_codec_.plType || payload_type == 0) {
     return ptr_decoder_;
   }
   // Check for exisitng decoder, if exists - delete.
   if (ptr_decoder_) {
     ReleaseDecoder(ptr_decoder_);
-    ptr_decoder_ = NULL;
+    ptr_decoder_ = nullptr;
     memset(&receive_codec_, 0, sizeof(VideoCodec));
   }
-  ptr_decoder_ = CreateAndInitDecoder(payload_type, &receive_codec_);
+  ptr_decoder_ = CreateAndInitDecoder(frame, &receive_codec_);
   if (!ptr_decoder_) {
-    return NULL;
+    return nullptr;
   }
   VCMReceiveCallback* callback = decoded_frame_callback->UserReceiveCallback();
-  if (callback) callback->IncomingCodecChanged(receive_codec_);
-  if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback)
-      < 0) {
+  if (callback)
+    callback->OnIncomingPayloadType(receive_codec_.plType);
+  if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
+      0) {
     ReleaseDecoder(ptr_decoder_);
-    ptr_decoder_ = NULL;
+    ptr_decoder_ = nullptr;
     memset(&receive_codec_, 0, sizeof(VideoCodec));
-    return NULL;
+    return nullptr;
   }
   return ptr_decoder_;
 }
 
 void VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const {
   if (decoder) {
-    assert(&decoder->_decoder);
+    assert(decoder->_decoder);
     decoder->Release();
     if (!decoder->External()) {
-      delete &decoder->_decoder;
+      delete decoder->_decoder;
     }
     delete decoder;
   }
 }
 
-bool VCMCodecDataBase::SupportsRenderScheduling() const {
-  const VCMExtDecoderMapItem* ext_item = FindExternalDecoderItem(
-      receive_codec_.plType);
-  if (ext_item == nullptr) {
-    // Assume the receive_codec_ is internal and as an internal codec
-    // by definition it supports scheduling.
+bool VCMCodecDataBase::PrefersLateDecoding() const {
+  if (!ptr_decoder_)
     return true;
-  }
-  return ext_item->internal_render_timing;
+  return ptr_decoder_->PrefersLateDecoding();
 }
 
 bool VCMCodecDataBase::MatchesCurrentResolution(int width, int height) const {
@@ -626,90 +537,73 @@ bool VCMCodecDataBase::MatchesCurrentResolution(int width, int height) const {
 }
 
 VCMGenericDecoder* VCMCodecDataBase::CreateAndInitDecoder(
-    uint8_t payload_type,
+    const VCMEncodedFrame& frame,
     VideoCodec* new_codec) const {
+  uint8_t payload_type = frame.PayloadType();
   assert(new_codec);
   const VCMDecoderMapItem* decoder_item = FindDecoderItem(payload_type);
   if (!decoder_item) {
     LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
                   << static_cast<int>(payload_type);
-    return NULL;
+    return nullptr;
   }
-  VCMGenericDecoder* ptr_decoder = NULL;
+  VCMGenericDecoder* ptr_decoder = nullptr;
   const VCMExtDecoderMapItem* external_dec_item =
       FindExternalDecoderItem(payload_type);
   if (external_dec_item) {
     // External codec.
     ptr_decoder = new VCMGenericDecoder(
-        *external_dec_item->external_decoder_instance, true);
+        external_dec_item->external_decoder_instance, true);
   } else {
     // Create decoder.
     ptr_decoder = CreateDecoder(decoder_item->settings->codecType);
   }
   if (!ptr_decoder)
-    return NULL;
+    return nullptr;
 
+  // Copy over input resolutions to prevent codec reinitialization due to
+  // the first frame being of a different resolution than the database values.
+  // This is best effort, since there's no guarantee that width/height have been
+  // parsed yet (and may be zero).
+  if (frame.EncodedImage()._encodedWidth > 0 &&
+      frame.EncodedImage()._encodedHeight > 0) {
+    decoder_item->settings->width = frame.EncodedImage()._encodedWidth;
+    decoder_item->settings->height = frame.EncodedImage()._encodedHeight;
+  }
   if (ptr_decoder->InitDecode(decoder_item->settings.get(),
                               decoder_item->number_of_cores) < 0) {
     ReleaseDecoder(ptr_decoder);
-    return NULL;
+    return nullptr;
   }
   memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
   return ptr_decoder;
 }
 
-VCMGenericEncoder* VCMCodecDataBase::CreateEncoder(
-  const VideoCodecType type) const {
-  switch (type) {
-#ifdef VIDEOCODEC_VP8
-    case kVideoCodecVP8:
-      return new VCMGenericEncoder(VP8Encoder::Create(), encoder_rate_observer_,
-                                   false);
-#endif
-#ifdef VIDEOCODEC_VP9
-    case kVideoCodecVP9:
-      return new VCMGenericEncoder(VP9Encoder::Create(), encoder_rate_observer_,
-                                   false);
-#endif
-#ifdef VIDEOCODEC_I420
-    case kVideoCodecI420:
-      return new VCMGenericEncoder(new I420Encoder(), encoder_rate_observer_,
-                                   false);
-#endif
-    default:
-      LOG(LS_WARNING) << "No internal encoder of this type exists.";
-      return NULL;
-  }
-}
-
 void VCMCodecDataBase::DeleteEncoder() {
-  if (ptr_encoder_) {
-    ptr_encoder_->Release();
-    if (!current_enc_is_external_)
-      delete ptr_encoder_->encoder_;
-    delete ptr_encoder_;
-    ptr_encoder_ = NULL;
-  }
+  if (!ptr_encoder_)
+    return;
+  ptr_encoder_->Release();
+  ptr_encoder_.reset();
 }
 
 VCMGenericDecoder* VCMCodecDataBase::CreateDecoder(VideoCodecType type) const {
   switch (type) {
-#ifdef VIDEOCODEC_VP8
     case kVideoCodecVP8:
-      return new VCMGenericDecoder(*(VP8Decoder::Create()));
-#endif
-#ifdef VIDEOCODEC_VP9
+      return new VCMGenericDecoder(VP8Decoder::Create());
     case kVideoCodecVP9:
-      return new VCMGenericDecoder(*(VP9Decoder::Create()));
-#endif
-#ifdef VIDEOCODEC_I420
+      return new VCMGenericDecoder(VP9Decoder::Create());
     case kVideoCodecI420:
-      return new VCMGenericDecoder(*(new I420Decoder));
-#endif
+      return new VCMGenericDecoder(new I420Decoder());
+    case kVideoCodecH264:
+      if (H264Decoder::IsSupported()) {
+        return new VCMGenericDecoder(H264Decoder::Create());
+      }
+      break;
     default:
-      LOG(LS_WARNING) << "No internal decoder of this type exists.";
-      return NULL;
+      break;
   }
+  LOG(LS_WARNING) << "No internal decoder of this type exists.";
+  return nullptr;
 }
 
 const VCMDecoderMapItem* VCMCodecDataBase::FindDecoderItem(
@@ -718,7 +612,7 @@ const VCMDecoderMapItem* VCMCodecDataBase::FindDecoderItem(
   if (it != dec_map_.end()) {
     return (*it).second;
   }
-  return NULL;
+  return nullptr;
 }
 
 const VCMExtDecoderMapItem* VCMCodecDataBase::FindExternalDecoderItem(
@@ -727,6 +621,6 @@ const VCMExtDecoderMapItem* VCMCodecDataBase::FindExternalDecoderItem(
   if (it != dec_external_map_.end()) {
     return (*it).second;
   }
-  return NULL;
+  return nullptr;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.h b/media/webrtc/trunk/webrtc/modules/video_coding/codec_database.h
similarity index 70%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/codec_database.h
index 4eaae5e8d4..62ec30a46e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_database.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codec_database.h
@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
 
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/generic_decoder.h"
+#include "webrtc/modules/video_coding/generic_encoder.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -36,38 +36,28 @@ struct VCMDecoderMapItem {
 struct VCMExtDecoderMapItem {
  public:
   VCMExtDecoderMapItem(VideoDecoder* external_decoder_instance,
-                       uint8_t payload_type,
-                       bool internal_render_timing);
+                       uint8_t payload_type);
 
   uint8_t payload_type;
   VideoDecoder* external_decoder_instance;
-  bool internal_render_timing;
 };
 
 class VCMCodecDataBase {
  public:
-  explicit VCMCodecDataBase(VideoEncoderRateObserver* encoder_rate_observer);
+  VCMCodecDataBase(VideoEncoderRateObserver* encoder_rate_observer,
+                   VCMEncodedFrameCallback* encoded_frame_callback);
   ~VCMCodecDataBase();
 
   // Sender Side
-  // Returns the number of supported codecs (or -1 in case of error).
-  static int NumberOfCodecs();
-
-  // Returns the default settings for the codec with id |list_id|.
-  static bool Codec(int list_id, VideoCodec* settings);
-
   // Returns the default settings for the codec with type |codec_type|.
-  static bool Codec(VideoCodecType codec_type, VideoCodec* settings);
-
-  void ResetSender();
+  static void Codec(VideoCodecType codec_type, VideoCodec* settings);
 
   // Sets the sender side codec and initiates the desired codec given the
   // VideoCodec struct.
   // Returns true if the codec was successfully registered, false otherwise.
   bool SetSendCodec(const VideoCodec* send_codec,
                     int number_of_cores,
-                    size_t max_payload_size,
-                    VCMEncodedFrameCallback* encoded_frame_callback);
+                    size_t max_payload_size);
 
   // Gets the current send codec. Relevant for internal codecs only.
   // Returns true if there is a send codec, false otherwise.
@@ -94,19 +84,12 @@ class VCMCodecDataBase {
 
   bool SetPeriodicKeyFrames(bool enable);
 
-  // Receiver Side
-  void ResetReceiver();
-
   // Deregisters an external decoder object specified by |payload_type|.
   bool DeregisterExternalDecoder(uint8_t payload_type);
 
   // Registers an external decoder object to the payload type |payload_type|.
-  // |internal_render_timing| is set to true if the |external_decoder| has
-  // built in rendering which is able to obey the render timestamps of the
-  // encoded frames.
-  bool RegisterExternalDecoder(VideoDecoder* external_decoder,
-                               uint8_t payload_type,
-                               bool internal_render_timing);
+  void RegisterExternalDecoder(VideoDecoder* external_decoder,
+                               uint8_t payload_type);
 
   bool DecoderRegistered() const;
 
@@ -128,16 +111,16 @@ class VCMCodecDataBase {
   // NULL is returned if no encoder with the specified payload type was found
   // and the function failed to create one.
   VCMGenericDecoder* GetDecoder(
-      uint8_t payload_type, VCMDecodedFrameCallback* decoded_frame_callback);
+      const VCMEncodedFrame& frame,
+      VCMDecodedFrameCallback* decoded_frame_callback);
 
   // Deletes the memory of the decoder instance |decoder|. Used to delete
   // deep copies returned by CreateDecoderCopy().
   void ReleaseDecoder(VCMGenericDecoder* decoder) const;
 
-  // Returns true if the currently active decoder supports render scheduling,
-  // that is, it is able to render frames according to the render timestamp of
-  // the encoded frames.
-  bool SupportsRenderScheduling() const;
+  // Returns true if the currently active decoder prefer to decode frames late.
+  // That means that frames must be decoded near the render times stamp.
+  bool PrefersLateDecoding() const;
 
   bool MatchesCurrentResolution(int width, int height) const;
 
@@ -145,14 +128,12 @@ class VCMCodecDataBase {
   typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
   typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
 
-  VCMGenericDecoder* CreateAndInitDecoder(uint8_t payload_type,
+  VCMGenericDecoder* CreateAndInitDecoder(const VCMEncodedFrame& frame,
                                           VideoCodec* new_codec) const;
 
   // Determines whether a new codec has to be created or not.
   // Checks every setting apart from maxFramerate and startBitrate.
   bool RequiresEncoderReset(const VideoCodec& send_codec);
-  // Create an internal encoder given a codec type.
-  VCMGenericEncoder* CreateEncoder(const VideoCodecType type) const;
 
   void DeleteEncoder();
 
@@ -168,14 +149,14 @@ class VCMCodecDataBase {
   size_t max_payload_size_;
   bool periodic_key_frames_;
   bool pending_encoder_reset_;
-  bool current_enc_is_external_;
   VideoCodec send_codec_;
   VideoCodec receive_codec_;
-  uint8_t external_payload_type_;
+  uint8_t encoder_payload_type_;
   VideoEncoder* external_encoder_;
   bool internal_source_;
   VideoEncoderRateObserver* const encoder_rate_observer_;
-  VCMGenericEncoder* ptr_encoder_;
+  VCMEncodedFrameCallback* const encoded_frame_callback_;
+  rtc::scoped_ptr<VCMGenericEncoder> ptr_encoder_;
   VCMGenericDecoder* ptr_decoder_;
   DecoderMap dec_map_;
   ExternalDecoderMap dec_external_map_;
@@ -183,4 +164,4 @@ class VCMCodecDataBase {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_CODEC_DATABASE_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codec_timer.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codec_timer.cc
new file mode 100644
index 0000000000..60add8fc4b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codec_timer.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/codec_timer.h"
+
+#include <assert.h>
+
+namespace webrtc {
+
+// The first kIgnoredSampleCount samples will be ignored.
+static const int32_t kIgnoredSampleCount = 5;
+
+VCMCodecTimer::VCMCodecTimer()
+    : _filteredMax(0), _ignoredSampleCount(0), _shortMax(0), _history() {
+  Reset();
+}
+
+void VCMCodecTimer::Reset() {
+  _filteredMax = 0;
+  _ignoredSampleCount = 0;
+  _shortMax = 0;
+  for (int i = 0; i < MAX_HISTORY_SIZE; i++) {
+    _history[i].shortMax = 0;
+    _history[i].timeMs = -1;
+  }
+}
+
+// Update the max-value filter
+void VCMCodecTimer::MaxFilter(int32_t decodeTime, int64_t nowMs) {
+  if (_ignoredSampleCount >= kIgnoredSampleCount) {
+    UpdateMaxHistory(decodeTime, nowMs);
+    ProcessHistory(nowMs);
+  } else {
+    _ignoredSampleCount++;
+  }
+}
+
+void VCMCodecTimer::UpdateMaxHistory(int32_t decodeTime, int64_t now) {
+  if (_history[0].timeMs >= 0 && now - _history[0].timeMs < SHORT_FILTER_MS) {
+    if (decodeTime > _shortMax) {
+      _shortMax = decodeTime;
+    }
+  } else {
+    // Only add a new value to the history once a second
+    if (_history[0].timeMs == -1) {
+      // First, no shift
+      _shortMax = decodeTime;
+    } else {
+      // Shift
+      for (int i = (MAX_HISTORY_SIZE - 2); i >= 0; i--) {
+        _history[i + 1].shortMax = _history[i].shortMax;
+        _history[i + 1].timeMs = _history[i].timeMs;
+      }
+    }
+    if (_shortMax == 0) {
+      _shortMax = decodeTime;
+    }
+
+    _history[0].shortMax = _shortMax;
+    _history[0].timeMs = now;
+    _shortMax = 0;
+  }
+}
+
+void VCMCodecTimer::ProcessHistory(int64_t nowMs) {
+  _filteredMax = _shortMax;
+  if (_history[0].timeMs == -1) {
+    return;
+  }
+  for (int i = 0; i < MAX_HISTORY_SIZE; i++) {
+    if (_history[i].timeMs == -1) {
+      break;
+    }
+    if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS) {
+      // This sample (and all samples after this) is too old
+      break;
+    }
+    if (_history[i].shortMax > _filteredMax) {
+      // This sample is the largest one this far into the history
+      _filteredMax = _history[i].shortMax;
+    }
+  }
+}
+
+// Get the maximum observed time within a time window
+int32_t VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const {
+  return _filteredMax;
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codec_timer.h b/media/webrtc/trunk/webrtc/modules/video_coding/codec_timer.h
new file mode 100644
index 0000000000..8ebd82ab9c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codec_timer.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
+#define MAX_HISTORY_SIZE 10
+#define SHORT_FILTER_MS 1000
+
+class VCMShortMaxSample {
+ public:
+  VCMShortMaxSample() : shortMax(0), timeMs(-1) {}
+
+  int32_t shortMax;
+  int64_t timeMs;
+};
+
+class VCMCodecTimer {
+ public:
+  VCMCodecTimer();
+
+  // Updates the max filtered decode time.
+  void MaxFilter(int32_t newDecodeTimeMs, int64_t nowMs);
+
+  // Empty the list of timers.
+  void Reset();
+
+  // Get the required decode time in ms.
+  int32_t RequiredDecodeTimeMs(FrameType frameType) const;
+
+ private:
+  void UpdateMaxHistory(int32_t decodeTime, int64_t now);
+  void ProcessHistory(int64_t nowMs);
+
+  int32_t _filteredMax;
+  // The number of samples ignored so far.
+  int32_t _ignoredSampleCount;
+  int32_t _shortMax;
+  VCMShortMaxSample _history[MAX_HISTORY_SIZE];
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/OWNERS b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/OWNERS
index 07c2987707..37a6e6e501 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/OWNERS
@@ -1,2 +1,3 @@
 stefan@webrtc.org
 marpan@webrtc.org
+tkchin@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264.cc
new file mode 100644
index 0000000000..645ed2cad7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+#if defined(WEBRTC_IOS)
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h"
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h"
+#endif
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+// We need this file to be C++ only so it will compile properly for all
+// platforms. In order to write ObjC specific implementations we use private
+// externs. This function is defined in h264.mm.
+#if defined(WEBRTC_IOS)
+extern bool IsH264CodecSupportedObjC();
+#endif
+
+bool IsH264CodecSupported() {
+#if defined(WEBRTC_IOS)
+  return IsH264CodecSupportedObjC();
+#else
+  return false;
+#endif
+}
+
+H264Encoder* H264Encoder::Create() {
+  RTC_DCHECK(H264Encoder::IsSupported());
+#if defined(WEBRTC_IOS) && defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+  return new H264VideoToolboxEncoder();
+#else
+  RTC_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+bool H264Encoder::IsSupported() {
+  return IsH264CodecSupported();
+}
+
+H264Decoder* H264Decoder::Create() {
+  RTC_DCHECK(H264Decoder::IsSupported());
+#if defined(WEBRTC_IOS) && defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+  return new H264VideoToolboxDecoder();
+#else
+  RTC_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+bool H264Decoder::IsSupported() {
+  return IsH264CodecSupported();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264.gypi b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264.gypi
new file mode 100644
index 0000000000..a20865c3aa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264.gypi
@@ -0,0 +1,63 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [
+    '../../../../build/common.gypi',
+  ],
+  'targets': [
+    {
+      'target_name': 'webrtc_h264',
+      'type': 'static_library',
+      'conditions': [
+        ['OS=="ios"', {
+          'dependencies': [
+            'webrtc_h264_video_toolbox',
+          ],
+          'sources': [
+            'h264_objc.mm',
+          ],
+        }],
+      ],
+      'sources': [
+        'h264.cc',
+        'include/h264.h',
+      ],
+    }, # webrtc_h264
+  ],
+  'conditions': [
+    ['OS=="ios"', {
+      'targets': [
+        {
+          'target_name': 'webrtc_h264_video_toolbox',
+          'type': 'static_library',
+          'dependencies': [
+            '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
+          ],
+          'link_settings': {
+            'xcode_settings': {
+              'OTHER_LDFLAGS': [
+                '-framework CoreMedia',
+                '-framework CoreVideo',
+                '-framework VideoToolbox',
+              ],
+            },
+          },
+          'sources': [
+            'h264_video_toolbox_decoder.cc',
+            'h264_video_toolbox_decoder.h',
+            'h264_video_toolbox_encoder.cc',
+            'h264_video_toolbox_encoder.h',
+            'h264_video_toolbox_nalu.cc',
+            'h264_video_toolbox_nalu.h',
+          ],
+        }, # webrtc_h264_video_toolbox
+      ], # targets
+    }], # OS=="ios"
+  ], # conditions
+}
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_objc.mm b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_objc.mm
new file mode 100644
index 0000000000..b9e0fc0090
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_objc.mm
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+#if defined(WEBRTC_IOS)
+#import <UIKit/UIKit.h>
+#endif
+
+namespace webrtc {
+
+bool IsH264CodecSupportedObjC() {
+#if defined(WEBRTC_OBJC_H264) && \
+    defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED) && \
+    defined(WEBRTC_IOS)
+  // Supported on iOS8+.
+  return [[[UIDevice currentDevice] systemVersion] doubleValue] >= 8.0;
+#else
+  // TODO(tkchin): Support OS/X once we stop mixing libstdc++ and libc++ on
+  // OSX 10.9.
+  return false;
+#endif
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
new file mode 100644
index 0000000000..6fee2e6f36
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
@@ -0,0 +1,270 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h"
+
+#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+
+#include "libyuv/convert.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
+#include "webrtc/video_frame.h"
+
+namespace internal {
+
+// Convenience function for creating a dictionary.
+inline CFDictionaryRef CreateCFDictionary(CFTypeRef* keys,
+                                          CFTypeRef* values,
+                                          size_t size) {
+  return CFDictionaryCreate(nullptr, keys, values, size,
+                            &kCFTypeDictionaryKeyCallBacks,
+                            &kCFTypeDictionaryValueCallBacks);
+}
+
+// Struct that we pass to the decoder per frame to decode. We receive it again
+// in the decoder callback.
+struct FrameDecodeParams {
+  FrameDecodeParams(webrtc::DecodedImageCallback* cb, int64_t ts)
+      : callback(cb), timestamp(ts) {}
+  webrtc::DecodedImageCallback* callback;
+  int64_t timestamp;
+};
+
+// On decode we receive a CVPixelBuffer, which we need to convert to a frame
+// buffer for use in the rest of WebRTC. Unfortunately this involves a frame
+// copy.
+// TODO(tkchin): Stuff CVPixelBuffer into a TextureBuffer and pass that along
+// instead once the pipeline supports it.
+rtc::scoped_refptr<webrtc::VideoFrameBuffer> VideoFrameBufferForPixelBuffer(
+    CVPixelBufferRef pixel_buffer) {
+  RTC_DCHECK(pixel_buffer);
+  RTC_DCHECK(CVPixelBufferGetPixelFormatType(pixel_buffer) ==
+             kCVPixelFormatType_420YpCbCr8BiPlanarFullRange);
+  size_t width = CVPixelBufferGetWidthOfPlane(pixel_buffer, 0);
+  size_t height = CVPixelBufferGetHeightOfPlane(pixel_buffer, 0);
+  // TODO(tkchin): Use a frame buffer pool.
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+      new rtc::RefCountedObject<webrtc::I420Buffer>(width, height);
+  CVPixelBufferLockBaseAddress(pixel_buffer, kCVPixelBufferLock_ReadOnly);
+  const uint8_t* src_y = reinterpret_cast<const uint8_t*>(
+      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer, 0));
+  int src_y_stride = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 0);
+  const uint8_t* src_uv = reinterpret_cast<const uint8_t*>(
+      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer, 1));
+  int src_uv_stride = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 1);
+  int ret = libyuv::NV12ToI420(
+      src_y, src_y_stride, src_uv, src_uv_stride,
+      buffer->MutableData(webrtc::kYPlane), buffer->stride(webrtc::kYPlane),
+      buffer->MutableData(webrtc::kUPlane), buffer->stride(webrtc::kUPlane),
+      buffer->MutableData(webrtc::kVPlane), buffer->stride(webrtc::kVPlane),
+      width, height);
+  CVPixelBufferUnlockBaseAddress(pixel_buffer, kCVPixelBufferLock_ReadOnly);
+  if (ret) {
+    LOG(LS_ERROR) << "Error converting NV12 to I420: " << ret;
+    return nullptr;
+  }
+  return buffer;
+}
+
+// This is the callback function that VideoToolbox calls when decode is
+// complete.
+void VTDecompressionOutputCallback(void* decoder,
+                                   void* params,
+                                   OSStatus status,
+                                   VTDecodeInfoFlags info_flags,
+                                   CVImageBufferRef image_buffer,
+                                   CMTime timestamp,
+                                   CMTime duration) {
+  rtc::scoped_ptr<FrameDecodeParams> decode_params(
+      reinterpret_cast<FrameDecodeParams*>(params));
+  if (status != noErr) {
+    LOG(LS_ERROR) << "Failed to decode frame. Status: " << status;
+    return;
+  }
+  // TODO(tkchin): Handle CVO properly.
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+      VideoFrameBufferForPixelBuffer(image_buffer);
+  webrtc::VideoFrame decoded_frame(buffer, decode_params->timestamp, 0,
+                                   webrtc::kVideoRotation_0);
+  decode_params->callback->Decoded(decoded_frame);
+}
+
+}  // namespace internal
+
+namespace webrtc {
+
+H264VideoToolboxDecoder::H264VideoToolboxDecoder()
+    : callback_(nullptr),
+      video_format_(nullptr),
+      decompression_session_(nullptr) {}
+
+H264VideoToolboxDecoder::~H264VideoToolboxDecoder() {
+  DestroyDecompressionSession();
+  SetVideoFormat(nullptr);
+}
+
+int H264VideoToolboxDecoder::InitDecode(const VideoCodec* video_codec,
+                                        int number_of_cores) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxDecoder::Decode(
+    const EncodedImage& input_image,
+    bool missing_frames,
+    const RTPFragmentationHeader* fragmentation,
+    const CodecSpecificInfo* codec_specific_info,
+    int64_t render_time_ms) {
+  RTC_DCHECK(input_image._buffer);
+
+  CMSampleBufferRef sample_buffer = nullptr;
+  if (!H264AnnexBBufferToCMSampleBuffer(input_image._buffer,
+                                        input_image._length, video_format_,
+                                        &sample_buffer)) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  RTC_DCHECK(sample_buffer);
+  // Check if the video format has changed, and reinitialize decoder if needed.
+  CMVideoFormatDescriptionRef description =
+      CMSampleBufferGetFormatDescription(sample_buffer);
+  if (!CMFormatDescriptionEqual(description, video_format_)) {
+    SetVideoFormat(description);
+    ResetDecompressionSession();
+  }
+  VTDecodeFrameFlags decode_flags =
+      kVTDecodeFrame_EnableAsynchronousDecompression;
+  rtc::scoped_ptr<internal::FrameDecodeParams> frame_decode_params;
+  frame_decode_params.reset(
+      new internal::FrameDecodeParams(callback_, input_image._timeStamp));
+  OSStatus status = VTDecompressionSessionDecodeFrame(
+      decompression_session_, sample_buffer, decode_flags,
+      frame_decode_params.release(), nullptr);
+  CFRelease(sample_buffer);
+  if (status != noErr) {
+    LOG(LS_ERROR) << "Failed to decode frame with code: " << status;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxDecoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  RTC_DCHECK(!callback_);
+  callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxDecoder::Release() {
+  callback_ = nullptr;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxDecoder::Reset() {
+  ResetDecompressionSession();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxDecoder::ResetDecompressionSession() {
+  DestroyDecompressionSession();
+
+  // Need to wait for the first SPS to initialize decoder.
+  if (!video_format_) {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  // Set keys for OpenGL and IOSurface compatibilty, which makes the encoder
+  // create pixel buffers with GPU backed memory. The intent here is to pass
+  // the pixel buffers directly so we avoid a texture upload later during
+  // rendering. This currently is moot because we are converting back to an
+  // I420 frame after decode, but eventually we will be able to plumb
+  // CVPixelBuffers directly to the renderer.
+  // TODO(tkchin): Maybe only set OpenGL/IOSurface keys if we know that that
+  // we can pass CVPixelBuffers as native handles in decoder output.
+  static size_t const attributes_size = 3;
+  CFTypeRef keys[attributes_size] = {
+#if defined(WEBRTC_IOS)
+    kCVPixelBufferOpenGLESCompatibilityKey,
+#elif defined(WEBRTC_MAC)
+    kCVPixelBufferOpenGLCompatibilityKey,
+#endif
+    kCVPixelBufferIOSurfacePropertiesKey,
+    kCVPixelBufferPixelFormatTypeKey
+  };
+  CFDictionaryRef io_surface_value =
+      internal::CreateCFDictionary(nullptr, nullptr, 0);
+  int64_t nv12type = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
+  CFNumberRef pixel_format =
+      CFNumberCreate(nullptr, kCFNumberLongType, &nv12type);
+  CFTypeRef values[attributes_size] = {kCFBooleanTrue, io_surface_value,
+                                       pixel_format};
+  CFDictionaryRef attributes =
+      internal::CreateCFDictionary(keys, values, attributes_size);
+  if (io_surface_value) {
+    CFRelease(io_surface_value);
+    io_surface_value = nullptr;
+  }
+  if (pixel_format) {
+    CFRelease(pixel_format);
+    pixel_format = nullptr;
+  }
+  VTDecompressionOutputCallbackRecord record = {
+      internal::VTDecompressionOutputCallback, this,
+  };
+  OSStatus status =
+      VTDecompressionSessionCreate(nullptr, video_format_, nullptr, attributes,
+                                   &record, &decompression_session_);
+  CFRelease(attributes);
+  if (status != noErr) {
+    DestroyDecompressionSession();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  ConfigureDecompressionSession();
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void H264VideoToolboxDecoder::ConfigureDecompressionSession() {
+  RTC_DCHECK(decompression_session_);
+#if defined(WEBRTC_IOS)
+  VTSessionSetProperty(decompression_session_,
+                       kVTDecompressionPropertyKey_RealTime, kCFBooleanTrue);
+#endif
+}
+
+void H264VideoToolboxDecoder::DestroyDecompressionSession() {
+  if (decompression_session_) {
+    VTDecompressionSessionInvalidate(decompression_session_);
+    decompression_session_ = nullptr;
+  }
+}
+
+void H264VideoToolboxDecoder::SetVideoFormat(
+    CMVideoFormatDescriptionRef video_format) {
+  if (video_format_ == video_format) {
+    return;
+  }
+  if (video_format_) {
+    CFRelease(video_format_);
+  }
+  video_format_ = video_format;
+  if (video_format_) {
+    CFRetain(video_format_);
+  }
+}
+
+const char* H264VideoToolboxDecoder::ImplementationName() const {
+  return "VideoToolbox";
+}
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h
new file mode 100644
index 0000000000..6d64307a82
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_DECODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_DECODER_H_
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+
+#include <VideoToolbox/VideoToolbox.h>
+
+// This file provides a H264 encoder implementation using the VideoToolbox
+// APIs. Since documentation is almost non-existent, this is largely based on
+// the information in the VideoToolbox header files, a talk from WWDC 2014 and
+// experimentation.
+
+namespace webrtc {
+
+class H264VideoToolboxDecoder : public H264Decoder {
+ public:
+  H264VideoToolboxDecoder();
+
+  ~H264VideoToolboxDecoder() override;
+
+  int InitDecode(const VideoCodec* video_codec, int number_of_cores) override;
+
+  int Decode(const EncodedImage& input_image,
+             bool missing_frames,
+             const RTPFragmentationHeader* fragmentation,
+             const CodecSpecificInfo* codec_specific_info,
+             int64_t render_time_ms) override;
+
+  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
+
+  int Release() override;
+
+  int Reset() override;
+
+  const char* ImplementationName() const override;
+
+ private:
+  int ResetDecompressionSession();
+  void ConfigureDecompressionSession();
+  void DestroyDecompressionSession();
+  void SetVideoFormat(CMVideoFormatDescriptionRef video_format);
+
+  DecodedImageCallback* callback_;
+  CMVideoFormatDescriptionRef video_format_;
+  VTDecompressionSessionRef decompression_session_;
+};  // H264VideoToolboxDecoder
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_DECODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
new file mode 100644
index 0000000000..7df4ec74ba
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
@@ -0,0 +1,428 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h"
+
+#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+
+#include <string>
+#include <vector>
+
+#include "libyuv/convert_from.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
+
+namespace internal {
+
+// Convenience function for creating a dictionary.
+inline CFDictionaryRef CreateCFDictionary(CFTypeRef* keys,
+                                          CFTypeRef* values,
+                                          size_t size) {
+  return CFDictionaryCreate(kCFAllocatorDefault, keys, values, size,
+                            &kCFTypeDictionaryKeyCallBacks,
+                            &kCFTypeDictionaryValueCallBacks);
+}
+
+// Copies characters from a CFStringRef into a std::string.
+std::string CFStringToString(const CFStringRef cf_string) {
+  RTC_DCHECK(cf_string);
+  std::string std_string;
+  // Get the size needed for UTF8 plus terminating character.
+  size_t buffer_size =
+      CFStringGetMaximumSizeForEncoding(CFStringGetLength(cf_string),
+                                        kCFStringEncodingUTF8) +
+      1;
+  rtc::scoped_ptr<char[]> buffer(new char[buffer_size]);
+  if (CFStringGetCString(cf_string, buffer.get(), buffer_size,
+                         kCFStringEncodingUTF8)) {
+    // Copy over the characters.
+    std_string.assign(buffer.get());
+  }
+  return std_string;
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          int32_t value) {
+  CFNumberRef cfNum =
+      CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt32Type, &value);
+  OSStatus status = VTSessionSetProperty(session, key, cfNum);
+  CFRelease(cfNum);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                  << " to " << value << ": " << status;
+  }
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
+  CFBooleanRef cf_bool = (value) ? kCFBooleanTrue : kCFBooleanFalse;
+  OSStatus status = VTSessionSetProperty(session, key, cf_bool);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                  << " to " << value << ": " << status;
+  }
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          CFStringRef value) {
+  OSStatus status = VTSessionSetProperty(session, key, value);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    std::string val_string = CFStringToString(value);
+    LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                  << " to " << val_string << ": " << status;
+  }
+}
+
+// Struct that we pass to the encoder per frame to encode. We receive it again
+// in the encoder callback.
+struct FrameEncodeParams {
+  FrameEncodeParams(webrtc::EncodedImageCallback* cb,
+                    const webrtc::CodecSpecificInfo* csi,
+                    int32_t w,
+                    int32_t h,
+                    int64_t rtms,
+                    uint32_t ts)
+      : callback(cb), width(w), height(h), render_time_ms(rtms), timestamp(ts) {
+    if (csi) {
+      codec_specific_info = *csi;
+    } else {
+      codec_specific_info.codecType = webrtc::kVideoCodecH264;
+    }
+  }
+  webrtc::EncodedImageCallback* callback;
+  webrtc::CodecSpecificInfo codec_specific_info;
+  int32_t width;
+  int32_t height;
+  int64_t render_time_ms;
+  uint32_t timestamp;
+};
+
+// We receive I420Frames as input, but we need to feed CVPixelBuffers into the
+// encoder. This performs the copy and format conversion.
+// TODO(tkchin): See if encoder will accept i420 frames and compare performance.
+bool CopyVideoFrameToPixelBuffer(const webrtc::VideoFrame& frame,
+                                 CVPixelBufferRef pixel_buffer) {
+  RTC_DCHECK(pixel_buffer);
+  RTC_DCHECK(CVPixelBufferGetPixelFormatType(pixel_buffer) ==
+             kCVPixelFormatType_420YpCbCr8BiPlanarFullRange);
+  RTC_DCHECK(CVPixelBufferGetHeightOfPlane(pixel_buffer, 0) ==
+             static_cast<size_t>(frame.height()));
+  RTC_DCHECK(CVPixelBufferGetWidthOfPlane(pixel_buffer, 0) ==
+             static_cast<size_t>(frame.width()));
+
+  CVReturn cvRet = CVPixelBufferLockBaseAddress(pixel_buffer, 0);
+  if (cvRet != kCVReturnSuccess) {
+    LOG(LS_ERROR) << "Failed to lock base address: " << cvRet;
+    return false;
+  }
+  uint8_t* dst_y = reinterpret_cast<uint8_t*>(
+      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer, 0));
+  int dst_stride_y = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 0);
+  uint8_t* dst_uv = reinterpret_cast<uint8_t*>(
+      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer, 1));
+  int dst_stride_uv = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 1);
+  // Convert I420 to NV12.
+  int ret = libyuv::I420ToNV12(
+      frame.buffer(webrtc::kYPlane), frame.stride(webrtc::kYPlane),
+      frame.buffer(webrtc::kUPlane), frame.stride(webrtc::kUPlane),
+      frame.buffer(webrtc::kVPlane), frame.stride(webrtc::kVPlane), dst_y,
+      dst_stride_y, dst_uv, dst_stride_uv, frame.width(), frame.height());
+  CVPixelBufferUnlockBaseAddress(pixel_buffer, 0);
+  if (ret) {
+    LOG(LS_ERROR) << "Error converting I420 VideoFrame to NV12 :" << ret;
+    return false;
+  }
+  return true;
+}
+
+// This is the callback function that VideoToolbox calls when encode is
+// complete.
+void VTCompressionOutputCallback(void* encoder,
+                                 void* params,
+                                 OSStatus status,
+                                 VTEncodeInfoFlags info_flags,
+                                 CMSampleBufferRef sample_buffer) {
+  rtc::scoped_ptr<FrameEncodeParams> encode_params(
+      reinterpret_cast<FrameEncodeParams*>(params));
+  if (status != noErr) {
+    LOG(LS_ERROR) << "H264 encoding failed.";
+    return;
+  }
+  if (info_flags & kVTEncodeInfo_FrameDropped) {
+    LOG(LS_INFO) << "H264 encode dropped frame.";
+  }
+
+  bool is_keyframe = false;
+  CFArrayRef attachments =
+      CMSampleBufferGetSampleAttachmentsArray(sample_buffer, 0);
+  if (attachments != nullptr && CFArrayGetCount(attachments)) {
+    CFDictionaryRef attachment =
+        static_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(attachments, 0));
+    is_keyframe =
+        !CFDictionaryContainsKey(attachment, kCMSampleAttachmentKey_NotSync);
+  }
+
+  // Convert the sample buffer into a buffer suitable for RTP packetization.
+  // TODO(tkchin): Allocate buffers through a pool.
+  rtc::scoped_ptr<rtc::Buffer> buffer(new rtc::Buffer());
+  rtc::scoped_ptr<webrtc::RTPFragmentationHeader> header;
+  if (!H264CMSampleBufferToAnnexBBuffer(sample_buffer, is_keyframe,
+                                        buffer.get(), header.accept())) {
+    return;
+  }
+  webrtc::EncodedImage frame(buffer->data(), buffer->size(), buffer->size());
+  frame._encodedWidth = encode_params->width;
+  frame._encodedHeight = encode_params->height;
+  frame._completeFrame = true;
+  frame._frameType =
+      is_keyframe ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta;
+  frame.capture_time_ms_ = encode_params->render_time_ms;
+  frame._timeStamp = encode_params->timestamp;
+
+  int result = encode_params->callback->Encoded(
+      frame, &(encode_params->codec_specific_info), header.get());
+  if (result != 0) {
+    LOG(LS_ERROR) << "Encoded callback failed: " << result;
+  }
+}
+
+}  // namespace internal
+
+namespace webrtc {
+
+H264VideoToolboxEncoder::H264VideoToolboxEncoder()
+    : callback_(nullptr), compression_session_(nullptr) {}
+
+H264VideoToolboxEncoder::~H264VideoToolboxEncoder() {
+  DestroyCompressionSession();
+}
+
+int H264VideoToolboxEncoder::InitEncode(const VideoCodec* codec_settings,
+                                        int number_of_cores,
+                                        size_t max_payload_size) {
+  RTC_DCHECK(codec_settings);
+  RTC_DCHECK_EQ(codec_settings->codecType, kVideoCodecH264);
+  // TODO(tkchin): We may need to enforce width/height dimension restrictions
+  // to match what the encoder supports.
+  width_ = codec_settings->width;
+  height_ = codec_settings->height;
+  // We can only set average bitrate on the HW encoder.
+  bitrate_ = codec_settings->startBitrate * 1000;
+
+  // TODO(tkchin): Try setting payload size via
+  // kVTCompressionPropertyKey_MaxH264SliceBytes.
+
+  return ResetCompressionSession();
+}
+
+int H264VideoToolboxEncoder::Encode(
+    const VideoFrame& input_image,
+    const CodecSpecificInfo* codec_specific_info,
+    const std::vector<FrameType>* frame_types) {
+  if (input_image.IsZeroSize()) {
+    // It's possible to get zero sizes as a signal to produce keyframes (this
+    // happens for internal sources). But this shouldn't happen in
+    // webrtcvideoengine2.
+    RTC_NOTREACHED();
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+  if (!callback_ || !compression_session_) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  // Get a pixel buffer from the pool and copy frame data over.
+  CVPixelBufferPoolRef pixel_buffer_pool =
+      VTCompressionSessionGetPixelBufferPool(compression_session_);
+  CVPixelBufferRef pixel_buffer = nullptr;
+  CVReturn ret = CVPixelBufferPoolCreatePixelBuffer(nullptr, pixel_buffer_pool,
+                                                    &pixel_buffer);
+  if (ret != kCVReturnSuccess) {
+    LOG(LS_ERROR) << "Failed to create pixel buffer: " << ret;
+    // We probably want to drop frames here, since failure probably means
+    // that the pool is empty.
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  RTC_DCHECK(pixel_buffer);
+  if (!internal::CopyVideoFrameToPixelBuffer(input_image, pixel_buffer)) {
+    LOG(LS_ERROR) << "Failed to copy frame data.";
+    CVBufferRelease(pixel_buffer);
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // Check if we need a keyframe.
+  bool is_keyframe_required = false;
+  if (frame_types) {
+    for (auto frame_type : *frame_types) {
+      if (frame_type == kVideoFrameKey) {
+        is_keyframe_required = true;
+        break;
+      }
+    }
+  }
+
+  CMTime presentation_time_stamp =
+      CMTimeMake(input_image.render_time_ms(), 1000);
+  CFDictionaryRef frame_properties = nullptr;
+  if (is_keyframe_required) {
+    CFTypeRef keys[] = {kVTEncodeFrameOptionKey_ForceKeyFrame};
+    CFTypeRef values[] = {kCFBooleanTrue};
+    frame_properties = internal::CreateCFDictionary(keys, values, 1);
+  }
+  rtc::scoped_ptr<internal::FrameEncodeParams> encode_params;
+  encode_params.reset(new internal::FrameEncodeParams(
+      callback_, codec_specific_info, width_, height_,
+      input_image.render_time_ms(), input_image.timestamp()));
+  VTCompressionSessionEncodeFrame(
+      compression_session_, pixel_buffer, presentation_time_stamp,
+      kCMTimeInvalid, frame_properties, encode_params.release(), nullptr);
+  if (frame_properties) {
+    CFRelease(frame_properties);
+  }
+  if (pixel_buffer) {
+    CVBufferRelease(pixel_buffer);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxEncoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxEncoder::SetChannelParameters(uint32_t packet_loss,
+                                                  int64_t rtt) {
+  // Encoder doesn't know anything about packet loss or rtt so just return.
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxEncoder::SetRates(uint32_t new_bitrate_kbit,
+                                      uint32_t frame_rate) {
+  bitrate_ = new_bitrate_kbit * 1000;
+  if (compression_session_) {
+    internal::SetVTSessionProperty(compression_session_,
+                                   kVTCompressionPropertyKey_AverageBitRate,
+                                   bitrate_);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int H264VideoToolboxEncoder::Release() {
+  callback_ = nullptr;
+  // Need to reset to that the session is invalidated and won't use the
+  // callback anymore.
+  return ResetCompressionSession();
+}
+
+int H264VideoToolboxEncoder::ResetCompressionSession() {
+  DestroyCompressionSession();
+
+  // Set source image buffer attributes. These attributes will be present on
+  // buffers retrieved from the encoder's pixel buffer pool.
+  const size_t attributes_size = 3;
+  CFTypeRef keys[attributes_size] = {
+#if defined(WEBRTC_IOS)
+    kCVPixelBufferOpenGLESCompatibilityKey,
+#elif defined(WEBRTC_MAC)
+    kCVPixelBufferOpenGLCompatibilityKey,
+#endif
+    kCVPixelBufferIOSurfacePropertiesKey,
+    kCVPixelBufferPixelFormatTypeKey
+  };
+  CFDictionaryRef io_surface_value =
+      internal::CreateCFDictionary(nullptr, nullptr, 0);
+  int64_t nv12type = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
+  CFNumberRef pixel_format =
+      CFNumberCreate(nullptr, kCFNumberLongType, &nv12type);
+  CFTypeRef values[attributes_size] = {kCFBooleanTrue, io_surface_value,
+                                       pixel_format};
+  CFDictionaryRef source_attributes =
+      internal::CreateCFDictionary(keys, values, attributes_size);
+  if (io_surface_value) {
+    CFRelease(io_surface_value);
+    io_surface_value = nullptr;
+  }
+  if (pixel_format) {
+    CFRelease(pixel_format);
+    pixel_format = nullptr;
+  }
+  OSStatus status = VTCompressionSessionCreate(
+      nullptr,  // use default allocator
+      width_, height_, kCMVideoCodecType_H264,
+      nullptr,  // use default encoder
+      source_attributes,
+      nullptr,  // use default compressed data allocator
+      internal::VTCompressionOutputCallback, this, &compression_session_);
+  if (source_attributes) {
+    CFRelease(source_attributes);
+    source_attributes = nullptr;
+  }
+  if (status != noErr) {
+    LOG(LS_ERROR) << "Failed to create compression session: " << status;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  ConfigureCompressionSession();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void H264VideoToolboxEncoder::ConfigureCompressionSession() {
+  RTC_DCHECK(compression_session_);
+  internal::SetVTSessionProperty(compression_session_,
+                                 kVTCompressionPropertyKey_RealTime, true);
+  internal::SetVTSessionProperty(compression_session_,
+                                 kVTCompressionPropertyKey_ProfileLevel,
+                                 kVTProfileLevel_H264_Baseline_AutoLevel);
+  internal::SetVTSessionProperty(
+      compression_session_, kVTCompressionPropertyKey_AverageBitRate, bitrate_);
+  internal::SetVTSessionProperty(compression_session_,
+                                 kVTCompressionPropertyKey_AllowFrameReordering,
+                                 false);
+  // TODO(tkchin): Look at entropy mode and colorspace matrices.
+  // TODO(tkchin): Investigate to see if there's any way to make this work.
+  // May need it to interop with Android. Currently this call just fails.
+  // On inspecting encoder output on iOS8, this value is set to 6.
+  // internal::SetVTSessionProperty(compression_session_,
+  //     kVTCompressionPropertyKey_MaxFrameDelayCount,
+  //     1);
+  // TODO(tkchin): See if enforcing keyframe frequency is beneficial in any
+  // way.
+  // internal::SetVTSessionProperty(
+  //     compression_session_,
+  //     kVTCompressionPropertyKey_MaxKeyFrameInterval, 240);
+  // internal::SetVTSessionProperty(
+  //     compression_session_,
+  //     kVTCompressionPropertyKey_MaxKeyFrameIntervalDuration, 240);
+}
+
+void H264VideoToolboxEncoder::DestroyCompressionSession() {
+  if (compression_session_) {
+    VTCompressionSessionInvalidate(compression_session_);
+    CFRelease(compression_session_);
+    compression_session_ = nullptr;
+  }
+}
+
+const char* H264VideoToolboxEncoder::ImplementationName() const {
+  return "VideoToolbox";
+}
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
new file mode 100644
index 0000000000..269e0411b2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+
+#include <VideoToolbox/VideoToolbox.h>
+#include <vector>
+
+// This file provides a H264 encoder implementation using the VideoToolbox
+// APIs. Since documentation is almost non-existent, this is largely based on
+// the information in the VideoToolbox header files, a talk from WWDC 2014 and
+// experimentation.
+
+namespace webrtc {
+
+class H264VideoToolboxEncoder : public H264Encoder {
+ public:
+  H264VideoToolboxEncoder();
+
+  ~H264VideoToolboxEncoder() override;
+
+  int InitEncode(const VideoCodec* codec_settings,
+                 int number_of_cores,
+                 size_t max_payload_size) override;
+
+  int Encode(const VideoFrame& input_image,
+             const CodecSpecificInfo* codec_specific_info,
+             const std::vector<FrameType>* frame_types) override;
+
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
+
+  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
+
+  int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate) override;
+
+  int Release() override;
+
+  const char* ImplementationName() const override;
+
+ private:
+  int ResetCompressionSession();
+  void ConfigureCompressionSession();
+  void DestroyCompressionSession();
+
+  webrtc::EncodedImageCallback* callback_;
+  VTCompressionSessionRef compression_session_;
+  int32_t bitrate_;  // Bitrate in bits per second.
+  int32_t width_;
+  int32_t height_;
+};  // H264VideoToolboxEncoder
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc
new file mode 100644
index 0000000000..322c213f7b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc
@@ -0,0 +1,355 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
+
+#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+
+#include <CoreFoundation/CoreFoundation.h>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+const char kAnnexBHeaderBytes[4] = {0, 0, 0, 1};
+const size_t kAvccHeaderByteSize = sizeof(uint32_t);
+
+bool H264CMSampleBufferToAnnexBBuffer(
+    CMSampleBufferRef avcc_sample_buffer,
+    bool is_keyframe,
+    rtc::Buffer* annexb_buffer,
+    webrtc::RTPFragmentationHeader** out_header) {
+  RTC_DCHECK(avcc_sample_buffer);
+  RTC_DCHECK(out_header);
+  *out_header = nullptr;
+
+  // Get format description from the sample buffer.
+  CMVideoFormatDescriptionRef description =
+      CMSampleBufferGetFormatDescription(avcc_sample_buffer);
+  if (description == nullptr) {
+    LOG(LS_ERROR) << "Failed to get sample buffer's description.";
+    return false;
+  }
+
+  // Get parameter set information.
+  int nalu_header_size = 0;
+  size_t param_set_count = 0;
+  OSStatus status = CMVideoFormatDescriptionGetH264ParameterSetAtIndex(
+      description, 0, nullptr, nullptr, &param_set_count, &nalu_header_size);
+  if (status != noErr) {
+    LOG(LS_ERROR) << "Failed to get parameter set.";
+    return false;
+  }
+  // TODO(tkchin): handle other potential sizes.
+  RTC_DCHECK_EQ(nalu_header_size, 4);
+  RTC_DCHECK_EQ(param_set_count, 2u);
+
+  // Truncate any previous data in the buffer without changing its capacity.
+  annexb_buffer->SetSize(0);
+
+  size_t nalu_offset = 0;
+  std::vector<size_t> frag_offsets;
+  std::vector<size_t> frag_lengths;
+
+  // Place all parameter sets at the front of buffer.
+  if (is_keyframe) {
+    size_t param_set_size = 0;
+    const uint8_t* param_set = nullptr;
+    for (size_t i = 0; i < param_set_count; ++i) {
+      status = CMVideoFormatDescriptionGetH264ParameterSetAtIndex(
+          description, i, &param_set, &param_set_size, nullptr, nullptr);
+      if (status != noErr) {
+        LOG(LS_ERROR) << "Failed to get parameter set.";
+        return false;
+      }
+      // Update buffer.
+      annexb_buffer->AppendData(kAnnexBHeaderBytes, sizeof(kAnnexBHeaderBytes));
+      annexb_buffer->AppendData(reinterpret_cast<const char*>(param_set),
+                                param_set_size);
+      // Update fragmentation.
+      frag_offsets.push_back(nalu_offset + sizeof(kAnnexBHeaderBytes));
+      frag_lengths.push_back(param_set_size);
+      nalu_offset += sizeof(kAnnexBHeaderBytes) + param_set_size;
+    }
+  }
+
+  // Get block buffer from the sample buffer.
+  CMBlockBufferRef block_buffer =
+      CMSampleBufferGetDataBuffer(avcc_sample_buffer);
+  if (block_buffer == nullptr) {
+    LOG(LS_ERROR) << "Failed to get sample buffer's block buffer.";
+    return false;
+  }
+  CMBlockBufferRef contiguous_buffer = nullptr;
+  // Make sure block buffer is contiguous.
+  if (!CMBlockBufferIsRangeContiguous(block_buffer, 0, 0)) {
+    status = CMBlockBufferCreateContiguous(
+        nullptr, block_buffer, nullptr, nullptr, 0, 0, 0, &contiguous_buffer);
+    if (status != noErr) {
+      LOG(LS_ERROR) << "Failed to flatten non-contiguous block buffer: "
+                    << status;
+      return false;
+    }
+  } else {
+    contiguous_buffer = block_buffer;
+    // Retain to make cleanup easier.
+    CFRetain(contiguous_buffer);
+    block_buffer = nullptr;
+  }
+
+  // Now copy the actual data.
+  char* data_ptr = nullptr;
+  size_t block_buffer_size = CMBlockBufferGetDataLength(contiguous_buffer);
+  status = CMBlockBufferGetDataPointer(contiguous_buffer, 0, nullptr, nullptr,
+                                       &data_ptr);
+  if (status != noErr) {
+    LOG(LS_ERROR) << "Failed to get block buffer data.";
+    CFRelease(contiguous_buffer);
+    return false;
+  }
+  size_t bytes_remaining = block_buffer_size;
+  while (bytes_remaining > 0) {
+    // The size type here must match |nalu_header_size|, we expect 4 bytes.
+    // Read the length of the next packet of data. Must convert from big endian
+    // to host endian.
+    RTC_DCHECK_GE(bytes_remaining, (size_t)nalu_header_size);
+    uint32_t* uint32_data_ptr = reinterpret_cast<uint32_t*>(data_ptr);
+    uint32_t packet_size = CFSwapInt32BigToHost(*uint32_data_ptr);
+    // Update buffer.
+    annexb_buffer->AppendData(kAnnexBHeaderBytes, sizeof(kAnnexBHeaderBytes));
+    annexb_buffer->AppendData(data_ptr + nalu_header_size, packet_size);
+    // Update fragmentation.
+    frag_offsets.push_back(nalu_offset + sizeof(kAnnexBHeaderBytes));
+    frag_lengths.push_back(packet_size);
+    nalu_offset += sizeof(kAnnexBHeaderBytes) + packet_size;
+
+    size_t bytes_written = packet_size + nalu_header_size;
+    bytes_remaining -= bytes_written;
+    data_ptr += bytes_written;
+  }
+  RTC_DCHECK_EQ(bytes_remaining, (size_t)0);
+
+  rtc::scoped_ptr<webrtc::RTPFragmentationHeader> header;
+  header.reset(new webrtc::RTPFragmentationHeader());
+  header->VerifyAndAllocateFragmentationHeader(frag_offsets.size());
+  RTC_DCHECK_EQ(frag_lengths.size(), frag_offsets.size());
+  for (size_t i = 0; i < frag_offsets.size(); ++i) {
+    header->fragmentationOffset[i] = frag_offsets[i];
+    header->fragmentationLength[i] = frag_lengths[i];
+    header->fragmentationPlType[i] = 0;
+    header->fragmentationTimeDiff[i] = 0;
+  }
+  *out_header = header.release();
+  CFRelease(contiguous_buffer);
+  return true;
+}
+
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer) {
+  RTC_DCHECK(annexb_buffer);
+  RTC_DCHECK(out_sample_buffer);
+  *out_sample_buffer = nullptr;
+
+  // The buffer we receive via RTP has 00 00 00 01 start code artifically
+  // embedded by the RTP depacketizer. Extract NALU information.
+  // TODO(tkchin): handle potential case where sps and pps are delivered
+  // separately.
+  uint8_t first_nalu_type = annexb_buffer[4] & 0x1f;
+  bool is_first_nalu_type_sps = first_nalu_type == 0x7;
+
+  AnnexBBufferReader reader(annexb_buffer, annexb_buffer_size);
+  CMVideoFormatDescriptionRef description = nullptr;
+  OSStatus status = noErr;
+  if (is_first_nalu_type_sps) {
+    // Parse the SPS and PPS into a CMVideoFormatDescription.
+    const uint8_t* param_set_ptrs[2] = {};
+    size_t param_set_sizes[2] = {};
+    if (!reader.ReadNalu(&param_set_ptrs[0], &param_set_sizes[0])) {
+      LOG(LS_ERROR) << "Failed to read SPS";
+      return false;
+    }
+    if (!reader.ReadNalu(&param_set_ptrs[1], &param_set_sizes[1])) {
+      LOG(LS_ERROR) << "Failed to read PPS";
+      return false;
+    }
+    status = CMVideoFormatDescriptionCreateFromH264ParameterSets(
+        kCFAllocatorDefault, 2, param_set_ptrs, param_set_sizes, 4,
+        &description);
+    if (status != noErr) {
+      LOG(LS_ERROR) << "Failed to create video format description.";
+      return false;
+    }
+  } else {
+    RTC_DCHECK(video_format);
+    description = video_format;
+    // We don't need to retain, but it makes logic easier since we are creating
+    // in the other block.
+    CFRetain(description);
+  }
+
+  // Allocate memory as a block buffer.
+  // TODO(tkchin): figure out how to use a pool.
+  CMBlockBufferRef block_buffer = nullptr;
+  status = CMBlockBufferCreateWithMemoryBlock(
+      nullptr, nullptr, reader.BytesRemaining(), nullptr, nullptr, 0,
+      reader.BytesRemaining(), kCMBlockBufferAssureMemoryNowFlag,
+      &block_buffer);
+  if (status != kCMBlockBufferNoErr) {
+    LOG(LS_ERROR) << "Failed to create block buffer.";
+    CFRelease(description);
+    return false;
+  }
+
+  // Make sure block buffer is contiguous.
+  CMBlockBufferRef contiguous_buffer = nullptr;
+  if (!CMBlockBufferIsRangeContiguous(block_buffer, 0, 0)) {
+    status = CMBlockBufferCreateContiguous(
+        nullptr, block_buffer, nullptr, nullptr, 0, 0, 0, &contiguous_buffer);
+    if (status != noErr) {
+      LOG(LS_ERROR) << "Failed to flatten non-contiguous block buffer: "
+                    << status;
+      CFRelease(description);
+      CFRelease(block_buffer);
+      return false;
+    }
+  } else {
+    contiguous_buffer = block_buffer;
+    block_buffer = nullptr;
+  }
+
+  // Get a raw pointer into allocated memory.
+  size_t block_buffer_size = 0;
+  char* data_ptr = nullptr;
+  status = CMBlockBufferGetDataPointer(contiguous_buffer, 0, nullptr,
+                                       &block_buffer_size, &data_ptr);
+  if (status != kCMBlockBufferNoErr) {
+    LOG(LS_ERROR) << "Failed to get block buffer data pointer.";
+    CFRelease(description);
+    CFRelease(contiguous_buffer);
+    return false;
+  }
+  RTC_DCHECK(block_buffer_size == reader.BytesRemaining());
+
+  // Write Avcc NALUs into block buffer memory.
+  AvccBufferWriter writer(reinterpret_cast<uint8_t*>(data_ptr),
+                          block_buffer_size);
+  while (reader.BytesRemaining() > 0) {
+    const uint8_t* nalu_data_ptr = nullptr;
+    size_t nalu_data_size = 0;
+    if (reader.ReadNalu(&nalu_data_ptr, &nalu_data_size)) {
+      writer.WriteNalu(nalu_data_ptr, nalu_data_size);
+    }
+  }
+
+  // Create sample buffer.
+  status = CMSampleBufferCreate(nullptr, contiguous_buffer, true, nullptr,
+                                nullptr, description, 1, 0, nullptr, 0, nullptr,
+                                out_sample_buffer);
+  if (status != noErr) {
+    LOG(LS_ERROR) << "Failed to create sample buffer.";
+    CFRelease(description);
+    CFRelease(contiguous_buffer);
+    return false;
+  }
+  CFRelease(description);
+  CFRelease(contiguous_buffer);
+  return true;
+}
+
+AnnexBBufferReader::AnnexBBufferReader(const uint8_t* annexb_buffer,
+                                       size_t length)
+    : start_(annexb_buffer), offset_(0), next_offset_(0), length_(length) {
+  RTC_DCHECK(annexb_buffer);
+  offset_ = FindNextNaluHeader(start_, length_, 0);
+  next_offset_ =
+      FindNextNaluHeader(start_, length_, offset_ + sizeof(kAnnexBHeaderBytes));
+}
+
+bool AnnexBBufferReader::ReadNalu(const uint8_t** out_nalu,
+                                  size_t* out_length) {
+  RTC_DCHECK(out_nalu);
+  RTC_DCHECK(out_length);
+  *out_nalu = nullptr;
+  *out_length = 0;
+
+  size_t data_offset = offset_ + sizeof(kAnnexBHeaderBytes);
+  if (data_offset > length_) {
+    return false;
+  }
+  *out_nalu = start_ + data_offset;
+  *out_length = next_offset_ - data_offset;
+  offset_ = next_offset_;
+  next_offset_ =
+      FindNextNaluHeader(start_, length_, offset_ + sizeof(kAnnexBHeaderBytes));
+  return true;
+}
+
+size_t AnnexBBufferReader::BytesRemaining() const {
+  return length_ - offset_;
+}
+
+size_t AnnexBBufferReader::FindNextNaluHeader(const uint8_t* start,
+                                              size_t length,
+                                              size_t offset) const {
+  RTC_DCHECK(start);
+  if (offset + sizeof(kAnnexBHeaderBytes) > length) {
+    return length;
+  }
+  // NALUs are separated by an 00 00 00 01 header. Scan the byte stream
+  // starting from the offset for the next such sequence.
+  const uint8_t* current = start + offset;
+  // The loop reads sizeof(kAnnexBHeaderBytes) at a time, so stop when there
+  // aren't enough bytes remaining.
+  const uint8_t* const end = start + length - sizeof(kAnnexBHeaderBytes);
+  while (current < end) {
+    if (current[3] > 1) {
+      current += 4;
+    } else if (current[3] == 1 && current[2] == 0 && current[1] == 0 &&
+               current[0] == 0) {
+      return current - start;
+    } else {
+      ++current;
+    }
+  }
+  return length;
+}
+
+AvccBufferWriter::AvccBufferWriter(uint8_t* const avcc_buffer, size_t length)
+    : start_(avcc_buffer), offset_(0), length_(length) {
+  RTC_DCHECK(avcc_buffer);
+}
+
+bool AvccBufferWriter::WriteNalu(const uint8_t* data, size_t data_size) {
+  // Check if we can write this length of data.
+  if (data_size + kAvccHeaderByteSize > BytesRemaining()) {
+    return false;
+  }
+  // Write length header, which needs to be big endian.
+  uint32_t big_endian_length = CFSwapInt32HostToBig(data_size);
+  memcpy(start_ + offset_, &big_endian_length, sizeof(big_endian_length));
+  offset_ += sizeof(big_endian_length);
+  // Write data.
+  memcpy(start_ + offset_, data, data_size);
+  offset_ += data_size;
+  return true;
+}
+
+size_t AvccBufferWriter::BytesRemaining() const {
+  return length_ - offset_;
+}
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
new file mode 100644
index 0000000000..31ef525816
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_
+
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+
+#if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+
+#include <CoreMedia/CoreMedia.h>
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+
+// Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
+// suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
+// needs to be in Annex B format. Data is written directly to |annexb_buffer|
+// and a new RTPFragmentationHeader is returned in |out_header|.
+bool H264CMSampleBufferToAnnexBBuffer(
+    CMSampleBufferRef avcc_sample_buffer,
+    bool is_keyframe,
+    rtc::Buffer* annexb_buffer,
+    webrtc::RTPFragmentationHeader** out_header);
+
+// Converts a buffer received from RTP into a sample buffer suitable for the
+// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
+// buffer is in avcc format.
+// If |is_keyframe| is true then |video_format| is ignored since the format will
+// be read from the buffer. Otherwise |video_format| must be provided.
+// Caller is responsible for releasing the created sample buffer.
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer);
+
+// Helper class for reading NALUs from an RTP Annex B buffer.
+class AnnexBBufferReader final {
+ public:
+  AnnexBBufferReader(const uint8_t* annexb_buffer, size_t length);
+  ~AnnexBBufferReader() {}
+  AnnexBBufferReader(const AnnexBBufferReader& other) = delete;
+  void operator=(const AnnexBBufferReader& other) = delete;
+
+  // Returns a pointer to the beginning of the next NALU slice without the
+  // header bytes and its length. Returns false if no more slices remain.
+  bool ReadNalu(const uint8_t** out_nalu, size_t* out_length);
+
+  // Returns the number of unread NALU bytes, including the size of the header.
+  // If the buffer has no remaining NALUs this will return zero.
+  size_t BytesRemaining() const;
+
+ private:
+  // Returns the the next offset that contains NALU data.
+  size_t FindNextNaluHeader(const uint8_t* start,
+                            size_t length,
+                            size_t offset) const;
+
+  const uint8_t* const start_;
+  size_t offset_;
+  size_t next_offset_;
+  const size_t length_;
+};
+
+// Helper class for writing NALUs using avcc format into a buffer.
+class AvccBufferWriter final {
+ public:
+  AvccBufferWriter(uint8_t* const avcc_buffer, size_t length);
+  ~AvccBufferWriter() {}
+  AvccBufferWriter(const AvccBufferWriter& other) = delete;
+  void operator=(const AvccBufferWriter& other) = delete;
+
+  // Writes the data slice into the buffer. Returns false if there isn't
+  // enough space left.
+  bool WriteNalu(const uint8_t* data, size_t data_size);
+
+  // Returns the unused bytes in the buffer.
+  size_t BytesRemaining() const;
+
+ private:
+  uint8_t* const start_;
+  size_t offset_;
+  const size_t length_;
+};
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc
new file mode 100644
index 0000000000..36946f1f8e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
+
+namespace webrtc {
+
+static const uint8_t NALU_TEST_DATA_0[] = {0xAA, 0xBB, 0xCC};
+static const uint8_t NALU_TEST_DATA_1[] = {0xDE, 0xAD, 0xBE, 0xEF};
+
+TEST(AnnexBBufferReaderTest, TestReadEmptyInput) {
+  const uint8_t annex_b_test_data[] = {0x00};
+  AnnexBBufferReader reader(annex_b_test_data, 0);
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  EXPECT_EQ(0u, reader.BytesRemaining());
+  EXPECT_FALSE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(nullptr, nalu);
+  EXPECT_EQ(0u, nalu_length);
+}
+
+TEST(AnnexBBufferReaderTest, TestReadSingleNalu) {
+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xAA};
+  AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  EXPECT_EQ(arraysize(annex_b_test_data), reader.BytesRemaining());
+  EXPECT_TRUE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(annex_b_test_data + 4, nalu);
+  EXPECT_EQ(1u, nalu_length);
+  EXPECT_EQ(0u, reader.BytesRemaining());
+  EXPECT_FALSE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(nullptr, nalu);
+  EXPECT_EQ(0u, nalu_length);
+}
+
+TEST(AnnexBBufferReaderTest, TestReadMissingNalu) {
+  // clang-format off
+  const uint8_t annex_b_test_data[] = {0x01,
+                                       0x00, 0x01,
+                                       0x00, 0x00, 0x01,
+                                       0x00, 0x00, 0x00, 0xFF};
+  // clang-format on
+  AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  EXPECT_EQ(0u, reader.BytesRemaining());
+  EXPECT_FALSE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(nullptr, nalu);
+  EXPECT_EQ(0u, nalu_length);
+}
+
+TEST(AnnexBBufferReaderTest, TestReadMultipleNalus) {
+  // clang-format off
+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xFF,
+                                       0x01,
+                                       0x00, 0x01,
+                                       0x00, 0x00, 0x01,
+                                       0x00, 0x00, 0x00, 0xFF,
+                                       0x00, 0x00, 0x00, 0x01, 0xAA, 0xBB};
+  // clang-format on
+  AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  EXPECT_EQ(arraysize(annex_b_test_data), reader.BytesRemaining());
+  EXPECT_TRUE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(annex_b_test_data + 4, nalu);
+  EXPECT_EQ(11u, nalu_length);
+  EXPECT_EQ(6u, reader.BytesRemaining());
+  EXPECT_TRUE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(annex_b_test_data + 19, nalu);
+  EXPECT_EQ(2u, nalu_length);
+  EXPECT_EQ(0u, reader.BytesRemaining());
+  EXPECT_FALSE(reader.ReadNalu(&nalu, &nalu_length));
+  EXPECT_EQ(nullptr, nalu);
+  EXPECT_EQ(0u, nalu_length);
+}
+
+TEST(AvccBufferWriterTest, TestEmptyOutputBuffer) {
+  const uint8_t expected_buffer[] = {0x00};
+  const size_t buffer_size = 1;
+  rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  memset(buffer.get(), 0, buffer_size);
+  AvccBufferWriter writer(buffer.get(), 0);
+  EXPECT_EQ(0u, writer.BytesRemaining());
+  EXPECT_FALSE(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  EXPECT_EQ(0,
+            memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+TEST(AvccBufferWriterTest, TestWriteSingleNalu) {
+  const uint8_t expected_buffer[] = {
+      0x00, 0x00, 0x00, 0x03, 0xAA, 0xBB, 0xCC,
+  };
+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0) + 4;
+  rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  AvccBufferWriter writer(buffer.get(), buffer_size);
+  EXPECT_EQ(buffer_size, writer.BytesRemaining());
+  EXPECT_TRUE(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  EXPECT_EQ(0u, writer.BytesRemaining());
+  EXPECT_FALSE(writer.WriteNalu(NALU_TEST_DATA_1, arraysize(NALU_TEST_DATA_1)));
+  EXPECT_EQ(0,
+            memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+TEST(AvccBufferWriterTest, TestWriteMultipleNalus) {
+  // clang-format off
+  const uint8_t expected_buffer[] = {
+      0x00, 0x00, 0x00, 0x03, 0xAA, 0xBB, 0xCC,
+      0x00, 0x00, 0x00, 0x04, 0xDE, 0xAD, 0xBE, 0xEF
+  };
+  // clang-format on
+  const size_t buffer_size =
+      arraysize(NALU_TEST_DATA_0) + arraysize(NALU_TEST_DATA_1) + 8;
+  rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  AvccBufferWriter writer(buffer.get(), buffer_size);
+  EXPECT_EQ(buffer_size, writer.BytesRemaining());
+  EXPECT_TRUE(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  EXPECT_EQ(buffer_size - (arraysize(NALU_TEST_DATA_0) + 4),
+            writer.BytesRemaining());
+  EXPECT_TRUE(writer.WriteNalu(NALU_TEST_DATA_1, arraysize(NALU_TEST_DATA_1)));
+  EXPECT_EQ(0u, writer.BytesRemaining());
+  EXPECT_EQ(0,
+            memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+TEST(AvccBufferWriterTest, TestOverflow) {
+  const uint8_t expected_buffer[] = {0x00, 0x00, 0x00};
+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0);
+  rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  memset(buffer.get(), 0, buffer_size);
+  AvccBufferWriter writer(buffer.get(), buffer_size);
+  EXPECT_EQ(buffer_size, writer.BytesRemaining());
+  EXPECT_FALSE(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  EXPECT_EQ(buffer_size, writer.BytesRemaining());
+  EXPECT_EQ(0,
+            memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/include/h264.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/include/h264.h
index 03dc8c3ffc..50ca57c1c9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/include/h264.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/include/h264.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -12,23 +12,36 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#if defined(WEBRTC_IOS) || defined(WEBRTC_MAC)
+
+#include <Availability.h>
+#if (defined(__IPHONE_8_0) &&                            \
+     __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_8_0) || \
+    (defined(__MAC_10_8) && __MAC_OS_X_VERSION_MAX_ALLOWED >= __MAC_10_8)
+#define WEBRTC_VIDEO_TOOLBOX_SUPPORTED 1
+#endif
+
+#endif  // defined(WEBRTC_IOS) || defined(WEBRTC_MAC)
+
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
 class H264Encoder : public VideoEncoder {
  public:
   static H264Encoder* Create();
+  static bool IsSupported();
 
-  virtual ~H264Encoder() {}
-};  // H264Encoder
+  ~H264Encoder() override {}
+};
 
 class H264Decoder : public VideoDecoder {
  public:
   static H264Decoder* Create();
+  static bool IsSupported();
 
-  virtual ~H264Decoder() {}
-};  // H264Decoder
+  ~H264Decoder() override {}
+};
 
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/OWNERS b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/OWNERS
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/OWNERS
rename to media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/OWNERS
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.cc
similarity index 80%
rename from media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.cc
index be2d17d874..7f06b4cf7d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/codecs/i420/main/interface/i420.h"
+#include "webrtc/modules/video_coding/codecs/i420/include/i420.h"
 
 #include <limits>
 #include <string>
@@ -21,20 +21,19 @@ const size_t kI420HeaderSize = 4;
 
 namespace webrtc {
 
-I420Encoder::I420Encoder() : _inited(false), _encodedImage(),
-    _encodedCompleteCallback(NULL) {
-}
+I420Encoder::I420Encoder()
+    : _inited(false), _encodedImage(), _encodedCompleteCallback(NULL) {}
 
 I420Encoder::~I420Encoder() {
   _inited = false;
-  delete [] _encodedImage._buffer;
+  delete[] _encodedImage._buffer;
 }
 
 int I420Encoder::Release() {
   // Should allocate an encoded frame and then release it here, for that we
   // actually need an init flag.
   if (_encodedImage._buffer != NULL) {
-    delete [] _encodedImage._buffer;
+    delete[] _encodedImage._buffer;
     _encodedImage._buffer = NULL;
   }
   _inited = false;
@@ -53,7 +52,7 @@ int I420Encoder::InitEncode(const VideoCodec* codecSettings,
 
   // Allocating encoded memory.
   if (_encodedImage._buffer != NULL) {
-    delete [] _encodedImage._buffer;
+    delete[] _encodedImage._buffer;
     _encodedImage._buffer = NULL;
     _encodedImage._size = 0;
   }
@@ -72,11 +71,9 @@ int I420Encoder::InitEncode(const VideoCodec* codecSettings,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
-
-int I420Encoder::Encode(const I420VideoFrame& inputImage,
+int I420Encoder::Encode(const VideoFrame& inputImage,
                         const CodecSpecificInfo* /*codecSpecificInfo*/,
-                        const std::vector<VideoFrameType>* /*frame_types*/) {
+                        const std::vector<FrameType>* /*frame_types*/) {
   if (!_inited) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
@@ -84,7 +81,7 @@ int I420Encoder::Encode(const I420VideoFrame& inputImage,
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
 
-  _encodedImage._frameType = kKeyFrame;
+  _encodedImage._frameType = kVideoFrameKey;
   _encodedImage._timeStamp = inputImage.timestamp();
   _encodedImage._encodedHeight = inputImage.height();
   _encodedImage._encodedWidth = inputImage.width();
@@ -103,18 +100,18 @@ int I420Encoder::Encode(const I420VideoFrame& inputImage,
       kI420HeaderSize;
   if (_encodedImage._size > req_length) {
     // Reallocate buffer.
-    delete [] _encodedImage._buffer;
+    delete[] _encodedImage._buffer;
 
     _encodedImage._buffer = new uint8_t[req_length];
     _encodedImage._size = req_length;
   }
 
-  uint8_t *buffer = _encodedImage._buffer;
+  uint8_t* buffer = _encodedImage._buffer;
 
   buffer = InsertHeader(buffer, width, height);
 
-  int ret_length = ExtractBuffer(inputImage, req_length - kI420HeaderSize,
-                                 buffer);
+  int ret_length =
+      ExtractBuffer(inputImage, req_length - kI420HeaderSize, buffer);
   if (ret_length < 0)
     return WEBRTC_VIDEO_CODEC_MEMORY;
   _encodedImage._length = ret_length + kI420HeaderSize;
@@ -123,7 +120,8 @@ int I420Encoder::Encode(const I420VideoFrame& inputImage,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-uint8_t* I420Encoder::InsertHeader(uint8_t *buffer, uint16_t width,
+uint8_t* I420Encoder::InsertHeader(uint8_t* buffer,
+                                   uint16_t width,
                                    uint16_t height) {
   *buffer++ = static_cast<uint8_t>(width >> 8);
   *buffer++ = static_cast<uint8_t>(width & 0xFF);
@@ -132,30 +130,29 @@ uint8_t* I420Encoder::InsertHeader(uint8_t *buffer, uint16_t width,
   return buffer;
 }
 
-int
-I420Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback) {
+int I420Encoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
   _encodedCompleteCallback = callback;
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
-I420Decoder::I420Decoder() : _decodedImage(), _width(0), _height(0),
-    _inited(false), _decodeCompleteCallback(NULL) {
-}
+I420Decoder::I420Decoder()
+    : _decodedImage(),
+      _width(0),
+      _height(0),
+      _inited(false),
+      _decodeCompleteCallback(NULL) {}
 
 I420Decoder::~I420Decoder() {
   Release();
 }
 
-int
-I420Decoder::Reset() {
+int I420Decoder::Reset() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
-int
-I420Decoder::InitDecode(const VideoCodec* codecSettings,
-                        int /*numberOfCores */) {
+int I420Decoder::InitDecode(const VideoCodec* codecSettings,
+                            int /*numberOfCores */) {
   if (codecSettings == NULL) {
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   } else if (codecSettings->width < 1 || codecSettings->height < 1) {
@@ -167,7 +164,8 @@ I420Decoder::InitDecode(const VideoCodec* codecSettings,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/,
+int I420Decoder::Decode(const EncodedImage& inputImage,
+                        bool /*missingFrames*/,
                         const RTPFragmentationHeader* /*fragmentation*/,
                         const CodecSpecificInfo* /*codecSpecificInfo*/,
                         int64_t /*renderTimeMs*/) {
@@ -205,8 +203,8 @@ int I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/,
   }
   // Set decoded image parameters.
   int half_width = (_width + 1) / 2;
-  _decodedImage.CreateEmptyFrame(_width, _height,
-                                 _width, half_width, half_width);
+  _decodedImage.CreateEmptyFrame(_width, _height, _width, half_width,
+                                 half_width);
   // Converting from buffer to plane representation.
   int ret = ConvertToI420(kI420, buffer, 0, 0, _width, _height, 0,
                           kVideoRotation_0, &_decodedImage);
@@ -220,7 +218,8 @@ int I420Decoder::Decode(const EncodedImage& inputImage, bool /*missingFrames*/,
 }
 
 const uint8_t* I420Decoder::ExtractHeader(const uint8_t* buffer,
-                                          uint16_t* width, uint16_t* height) {
+                                          uint16_t* width,
+                                          uint16_t* height) {
   *width = static_cast<uint16_t>(*buffer++) << 8;
   *width |= *buffer++;
   *height = static_cast<uint16_t>(*buffer++) << 8;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.gypi b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.gypi
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.gypi
rename to media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.gypi
index 3718e8924d..4fd8e318bd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/i420.gypi
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/i420.gypi
@@ -15,7 +15,7 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        '../interface/i420.h',
+        'include/i420.h',
         'i420.cc',
       ],
     },
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/include/i420.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/include/i420.h
new file mode 100644
index 0000000000..9f77845e96
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/include/i420.h
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_INCLUDE_I420_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_INCLUDE_I420_H_
+
+#include <vector>
+
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class I420Encoder : public VideoEncoder {
+ public:
+  I420Encoder();
+
+  virtual ~I420Encoder();
+
+  // Initialize the encoder with the information from the VideoCodec.
+  //
+  // Input:
+  //          - codecSettings     : Codec settings.
+  //          - numberOfCores     : Number of cores available for the encoder.
+  //          - maxPayloadSize    : The maximum size each payload is allowed
+  //                                to have. Usually MTU - overhead.
+  //
+  // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
+  //                                <0 - Error
+  int InitEncode(const VideoCodec* codecSettings,
+                 int /*numberOfCores*/,
+                 size_t /*maxPayloadSize*/) override;
+
+  // "Encode" an I420 image (as a part of a video stream). The encoded image
+  // will be returned to the user via the encode complete callback.
+  //
+  // Input:
+  //          - inputImage        : Image to be encoded.
+  //          - codecSpecificInfo : Pointer to codec specific data.
+  //          - frameType         : Frame type to be sent (Key /Delta).
+  //
+  // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
+  //                                <0 - Error
+  int Encode(const VideoFrame& inputImage,
+             const CodecSpecificInfo* /*codecSpecificInfo*/,
+             const std::vector<FrameType>* /*frame_types*/) override;
+
+  // Register an encode complete callback object.
+  //
+  // Input:
+  //          - callback         : Callback object which handles encoded images.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
+
+  // Free encoder memory.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+  int Release() override;
+
+  int SetRates(uint32_t /*newBitRate*/, uint32_t /*frameRate*/) override {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  int SetChannelParameters(uint32_t /*packetLoss*/, int64_t /*rtt*/) override {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  void OnDroppedFrame() override {}
+
+ private:
+  static uint8_t* InsertHeader(uint8_t* buffer,
+                               uint16_t width,
+                               uint16_t height);
+
+  bool _inited;
+  EncodedImage _encodedImage;
+  EncodedImageCallback* _encodedCompleteCallback;
+};  // class I420Encoder
+
+class I420Decoder : public VideoDecoder {
+ public:
+  I420Decoder();
+
+  virtual ~I420Decoder();
+
+  // Initialize the decoder.
+  // The user must notify the codec of width and height values.
+  //
+  // Return value         :  WEBRTC_VIDEO_CODEC_OK.
+  //                        <0 - Errors
+  int InitDecode(const VideoCodec* codecSettings,
+                 int /*numberOfCores*/) override;
+
+  // Decode encoded image (as a part of a video stream). The decoded image
+  // will be returned to the user through the decode complete callback.
+  //
+  // Input:
+  //          - inputImage        : Encoded image to be decoded
+  //          - missingFrames     : True if one or more frames have been lost
+  //                                since the previous decode call.
+  //          - codecSpecificInfo : pointer to specific codec data
+  //          - renderTimeMs      : Render time in Ms
+  //
+  // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
+  //                                 <0 - Error
+  int Decode(const EncodedImage& inputImage,
+             bool missingFrames,
+             const RTPFragmentationHeader* /*fragmentation*/,
+             const CodecSpecificInfo* /*codecSpecificInfo*/,
+             int64_t /*renderTimeMs*/) override;
+
+  // Register a decode complete callback object.
+  //
+  // Input:
+  //          - callback         : Callback object which handles decoded images.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
+  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
+
+  // Free decoder memory.
+  //
+  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK.
+  //                                  <0 - Error
+  int Release() override;
+
+  // Reset decoder state and prepare for a new call.
+  //
+  // Return value         :  WEBRTC_VIDEO_CODEC_OK.
+  //                          <0 - Error
+  int Reset() override;
+
+ private:
+  static const uint8_t* ExtractHeader(const uint8_t* buffer,
+                                      uint16_t* width,
+                                      uint16_t* height);
+
+  VideoFrame _decodedImage;
+  int _width;
+  int _height;
+  bool _inited;
+  DecodedImageCallback* _decodeCompleteCallback;
+};  // class I420Decoder
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_INCLUDE_I420_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h
deleted file mode 100644
index f1159073bc..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/interface/i420.h
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_MAIN_INTERFACE_I420_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_MAIN_INTERFACE_I420_H_
-
-#include <vector>
-
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class I420Encoder : public VideoEncoder {
- public:
-  I420Encoder();
-
-  virtual ~I420Encoder();
-
-// Initialize the encoder with the information from the VideoCodec.
-//
-// Input:
-//          - codecSettings     : Codec settings.
-//          - numberOfCores     : Number of cores available for the encoder.
-//          - maxPayloadSize    : The maximum size each payload is allowed
-//                                to have. Usually MTU - overhead.
-//
-// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
-//                                <0 - Error
-  int InitEncode(const VideoCodec* codecSettings,
-                 int /*numberOfCores*/,
-                 size_t /*maxPayloadSize*/) override;
-
-// "Encode" an I420 image (as a part of a video stream). The encoded image
-// will be returned to the user via the encode complete callback.
-//
-// Input:
-//          - inputImage        : Image to be encoded.
-//          - codecSpecificInfo : Pointer to codec specific data.
-//          - frameType         : Frame type to be sent (Key /Delta).
-//
-// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK.
-//                                <0 - Error
-  int Encode(const I420VideoFrame& inputImage,
-             const CodecSpecificInfo* /*codecSpecificInfo*/,
-             const std::vector<VideoFrameType>* /*frame_types*/) override;
-
-// Register an encode complete callback object.
-//
-// Input:
-//          - callback         : Callback object which handles encoded images.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
-  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
-
-// Free encoder memory.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
-  int Release() override;
-
-  int SetRates(uint32_t /*newBitRate*/, uint32_t /*frameRate*/) override {
-    return WEBRTC_VIDEO_CODEC_OK;
-  }
-
-  int SetChannelParameters(uint32_t /*packetLoss*/, int64_t /*rtt*/) override {
-    return WEBRTC_VIDEO_CODEC_OK;
-  }
-
-  int CodecConfigParameters(uint8_t* /*buffer*/, int /*size*/) override {
-    return WEBRTC_VIDEO_CODEC_OK;
-  }
-
- private:
-  static uint8_t* InsertHeader(uint8_t* buffer, uint16_t width,
-                               uint16_t height);
-
-  bool                     _inited;
-  EncodedImage             _encodedImage;
-  EncodedImageCallback*    _encodedCompleteCallback;
-};  // class I420Encoder
-
-class I420Decoder : public VideoDecoder {
- public:
-  I420Decoder();
-
-  virtual ~I420Decoder();
-
-// Initialize the decoder.
-// The user must notify the codec of width and height values.
-//
-// Return value         :  WEBRTC_VIDEO_CODEC_OK.
-//                        <0 - Errors
-  int InitDecode(const VideoCodec* codecSettings,
-                 int /*numberOfCores*/) override;
-
-  int SetCodecConfigParameters(const uint8_t* /*buffer*/,
-                               int /*size*/) override {
-    return WEBRTC_VIDEO_CODEC_OK;
-  }
-
-// Decode encoded image (as a part of a video stream). The decoded image
-// will be returned to the user through the decode complete callback.
-//
-// Input:
-//          - inputImage        : Encoded image to be decoded
-//          - missingFrames     : True if one or more frames have been lost
-//                                since the previous decode call.
-//          - codecSpecificInfo : pointer to specific codec data
-//          - renderTimeMs      : Render time in Ms
-//
-// Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
-//                                 <0 - Error
-  int Decode(const EncodedImage& inputImage,
-             bool missingFrames,
-             const RTPFragmentationHeader* /*fragmentation*/,
-             const CodecSpecificInfo* /*codecSpecificInfo*/,
-             int64_t /*renderTimeMs*/) override;
-
-// Register a decode complete callback object.
-//
-// Input:
-//          - callback         : Callback object which handles decoded images.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
-  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
-
-// Free decoder memory.
-//
-// Return value                : WEBRTC_VIDEO_CODEC_OK if OK.
-//                                  <0 - Error
-  int Release() override;
-
-// Reset decoder state and prepare for a new call.
-//
-// Return value         :  WEBRTC_VIDEO_CODEC_OK.
-//                          <0 - Error
-  int Reset() override;
-
- private:
-  static const uint8_t* ExtractHeader(const uint8_t* buffer,
-                                      uint16_t* width,
-                                      uint16_t* height);
-
-  I420VideoFrame              _decodedImage;
-  int                         _width;
-  int                         _height;
-  bool                        _inited;
-  DecodedImageCallback*       _decodeCompleteCallback;
-};  // class I420Decoder
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_MAIN_INTERFACE_I420_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/OWNERS b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/i420/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
index ad72071840..d727e896ad 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
@@ -11,30 +11,36 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
 
+#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
+    "use video_coding/include")
 #include <string>
+#include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class MockEncodedImageCallback : public EncodedImageCallback {
  public:
-  MOCK_METHOD3(Encoded, int32_t(const EncodedImage& encodedImage,
-                                const CodecSpecificInfo* codecSpecificInfo,
-                                const RTPFragmentationHeader* fragmentation));
+  MOCK_METHOD3(Encoded,
+               int32_t(const EncodedImage& encodedImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const RTPFragmentationHeader* fragmentation));
 };
 
 class MockVideoEncoder : public VideoEncoder {
  public:
-  MOCK_CONST_METHOD2(Version, int32_t(int8_t *version, int32_t length));
-  MOCK_METHOD3(InitEncode, int32_t(const VideoCodec* codecSettings,
-                                   int32_t numberOfCores,
-                                   size_t maxPayloadSize));
-  MOCK_METHOD3(Encode, int32_t(const I420VideoFrame& inputImage,
-                               const CodecSpecificInfo* codecSpecificInfo,
-                               const std::vector<VideoFrameType>* frame_types));
+  MOCK_CONST_METHOD2(Version, int32_t(int8_t* version, int32_t length));
+  MOCK_METHOD3(InitEncode,
+               int32_t(const VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize));
+  MOCK_METHOD3(Encode,
+               int32_t(const VideoFrame& inputImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const std::vector<FrameType>* frame_types));
   MOCK_METHOD1(RegisterEncodeCompleteCallback,
                int32_t(EncodedImageCallback* callback));
   MOCK_METHOD0(Release, int32_t());
@@ -42,35 +48,32 @@ class MockVideoEncoder : public VideoEncoder {
   MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
   MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
   MOCK_METHOD1(SetPeriodicKeyFrames, int32_t(bool enable));
-  MOCK_METHOD2(CodecConfigParameters,
-               int32_t(uint8_t* /*buffer*/, int32_t));
 };
 
 class MockDecodedImageCallback : public DecodedImageCallback {
  public:
-  MOCK_METHOD1(Decoded,
-               int32_t(I420VideoFrame& decodedImage));
+  MOCK_METHOD1(Decoded, int32_t(const VideoFrame& decodedImage));
+  MOCK_METHOD2(Decoded,
+               int32_t(const VideoFrame& decodedImage, int64_t decode_time_ms));
   MOCK_METHOD1(ReceivedDecodedReferenceFrame,
                int32_t(const uint64_t pictureId));
-  MOCK_METHOD1(ReceivedDecodedFrame,
-               int32_t(const uint64_t pictureId));
+  MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
 };
 
 class MockVideoDecoder : public VideoDecoder {
  public:
-  MOCK_METHOD2(InitDecode, int32_t(const VideoCodec* codecSettings,
-                                   int32_t numberOfCores));
-  MOCK_METHOD5(Decode, int32_t(const EncodedImage& inputImage,
-                               bool missingFrames,
-                               const RTPFragmentationHeader* fragmentation,
-                               const CodecSpecificInfo* codecSpecificInfo,
-                               int64_t renderTimeMs));
+  MOCK_METHOD2(InitDecode,
+               int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
+  MOCK_METHOD5(Decode,
+               int32_t(const EncodedImage& inputImage,
+                       bool missingFrames,
+                       const RTPFragmentationHeader* fragmentation,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       int64_t renderTimeMs));
   MOCK_METHOD1(RegisterDecodeCompleteCallback,
                int32_t(DecodedImageCallback* callback));
   MOCK_METHOD0(Release, int32_t());
   MOCK_METHOD0(Reset, int32_t());
-  MOCK_METHOD2(SetCodecConfigParameters,
-               int32_t(const uint8_t* /*buffer*/, int32_t));
   MOCK_METHOD0(Copy, VideoDecoder*());
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
index 038927a6a0..a282753411 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
@@ -8,23 +8,23 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
 
+#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; use video_coding/include")
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_error_codes.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_decoder.h"
 #include "webrtc/video_encoder.h"
+#include "webrtc/video_frame.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-class RTPFragmentationHeader; // forward declaration
+class RTPFragmentationHeader;  // forward declaration
 
 // Note: if any pointers are added to this struct, it must be fitted
 // with a copy-constructor. See below.
@@ -94,12 +94,11 @@ union CodecSpecificInfoUnion {
 // Note: if any pointers are added to this struct or its sub-structs, it
 // must be fitted with a copy-constructor. This is because it is copied
 // in the copy-constructor of VCMEncodedFrame.
-struct CodecSpecificInfo
-{
-    VideoCodecType   codecType;
-    CodecSpecificInfoUnion codecSpecific;
+struct CodecSpecificInfo {
+  VideoCodecType codecType;
+  CodecSpecificInfoUnion codecSpecific;
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_error_codes.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
index dfa3f53200..ea8829df80 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
@@ -8,8 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
+
+#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
+    "use video_coding/include")
 
 // NOTE: in sync with video_coding_module_defines.h
 
@@ -26,5 +29,7 @@
 #define WEBRTC_VIDEO_CODEC_TIMEOUT -6
 #define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
 #define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
+#define WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE -13
+#define WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT -14
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
index 36ba0e8272..b554b4e9ae 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
@@ -57,7 +57,7 @@ int PacketManipulatorImpl::ManipulatePackets(
       active_burst_packets_--;
       nbr_packets_dropped++;
     } else if (RandomUniform() < config_.packet_loss_probability ||
-        packet_loss_has_occurred) {
+               packet_loss_has_occurred) {
       packet_loss_has_occurred = true;
       nbr_packets_dropped++;
       if (config_.packet_loss_mode == kBurst) {
@@ -91,9 +91,9 @@ inline double PacketManipulatorImpl::RandomUniform() {
   // get the same behavior as long as we're using a fixed initial seed.
   critsect_->Enter();
   srand(random_seed_);
-  random_seed_ = rand();
+  random_seed_ = rand();  // NOLINT (rand_r instead of rand)
   critsect_->Leave();
-  return (random_seed_ + 1.0)/(RAND_MAX + 1.0);
+  return (random_seed_ + 1.0) / (RAND_MAX + 1.0);
 }
 
 const char* PacketLossModeToStr(PacketLossMode e) {
@@ -109,4 +109,4 @@ const char* PacketLossModeToStr(PacketLossMode e) {
 }
 
 }  // namespace test
-}  // namespace webrtcc
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
index 5a1654a2a7..3334be072b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
@@ -13,8 +13,8 @@
 
 #include <stdlib.h>
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/test/testsupport/packet_reader.h"
 
 namespace webrtc {
@@ -36,10 +36,11 @@ const char* PacketLossModeToStr(PacketLossMode e);
 // scenarios caused by network interference.
 struct NetworkingConfig {
   NetworkingConfig()
-  : packet_size_in_bytes(1500), max_payload_size_in_bytes(1440),
-    packet_loss_mode(kUniform), packet_loss_probability(0.0),
-    packet_loss_burst_length(1) {
-  }
+      : packet_size_in_bytes(1500),
+        max_payload_size_in_bytes(1440),
+        packet_loss_mode(kUniform),
+        packet_loss_probability(0.0),
+        packet_loss_burst_length(1) {}
 
   // Packet size in bytes. Default: 1500 bytes.
   size_t packet_size_in_bytes;
@@ -93,9 +94,11 @@ class PacketManipulatorImpl : public PacketManipulator {
   virtual ~PacketManipulatorImpl();
   int ManipulatePackets(webrtc::EncodedImage* encoded_image) override;
   virtual void InitializeRandomSeed(unsigned int seed);
+
  protected:
   // Returns a uniformly distributed random value between 0.0 and 1.0
   virtual double RandomUniform();
+
  private:
   PacketReader* packet_reader_;
   const NetworkingConfig& config_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
index ace7bc0507..8c3d30dc0d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
@@ -13,7 +13,7 @@
 #include <queue>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h"
 #include "webrtc/test/testsupport/unittest_utils.h"
 #include "webrtc/typedefs.h"
@@ -25,7 +25,7 @@ const double kNeverDropProbability = 0.0;
 const double kAlwaysDropProbability = 1.0;
 const int kBurstLength = 1;
 
-class PacketManipulatorTest: public PacketRelatedTest {
+class PacketManipulatorTest : public PacketRelatedTest {
  protected:
   PacketReader packet_reader_;
   EncodedImage image_;
@@ -50,19 +50,15 @@ class PacketManipulatorTest: public PacketRelatedTest {
 
   virtual ~PacketManipulatorTest() {}
 
-  void SetUp() {
-    PacketRelatedTest::SetUp();
-  }
+  void SetUp() { PacketRelatedTest::SetUp(); }
 
-  void TearDown() {
-    PacketRelatedTest::TearDown();
-  }
+  void TearDown() { PacketRelatedTest::TearDown(); }
 
   void VerifyPacketLoss(int expected_nbr_packets_dropped,
                         int actual_nbr_packets_dropped,
                         size_t expected_packet_data_length,
                         uint8_t* expected_packet_data,
-                        EncodedImage& actual_image) {
+                        const EncodedImage& actual_image) {
     EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
     EXPECT_EQ(expected_packet_data_length, image_._length);
     EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
@@ -75,10 +71,10 @@ TEST_F(PacketManipulatorTest, Constructor) {
 }
 
 TEST_F(PacketManipulatorTest, DropNone) {
-  PacketManipulatorImpl manipulator(&packet_reader_,  no_drop_config_, false);
+  PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
   int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-  VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength,
-                   packet_data_, image_);
+  VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength, packet_data_,
+                   image_);
 }
 
 TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
@@ -87,15 +83,14 @@ TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
   PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
   int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
 
-  VerifyPacketLoss(0, nbr_packets_dropped, data_length,
-                     packet_data_, image_);
+  VerifyPacketLoss(0, nbr_packets_dropped, data_length, packet_data_, image_);
 }
 
 TEST_F(PacketManipulatorTest, UniformDropAll) {
   PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
   int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-  VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped,
-                   0, packet_data_, image_);
+  VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped, 0,
+                   packet_data_, image_);
 }
 
 // Use our customized test class to make the second packet being lost
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
index c92cfa48a7..9eba205a88 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
@@ -19,13 +19,11 @@ namespace webrtc {
 namespace test {
 
 PredictivePacketManipulator::PredictivePacketManipulator(
-    PacketReader* packet_reader, const NetworkingConfig& config)
-    : PacketManipulatorImpl(packet_reader, config, false) {
-}
-
-PredictivePacketManipulator::~PredictivePacketManipulator() {
-}
+    PacketReader* packet_reader,
+    const NetworkingConfig& config)
+    : PacketManipulatorImpl(packet_reader, config, false) {}
 
+PredictivePacketManipulator::~PredictivePacketManipulator() {}
 
 void PredictivePacketManipulator::AddRandomResult(double result) {
   assert(result >= 0.0 && result <= 1.0);
@@ -33,8 +31,9 @@ void PredictivePacketManipulator::AddRandomResult(double result) {
 }
 
 double PredictivePacketManipulator::RandomUniform() {
-  if(random_results_.size() == 0u) {
-    fprintf(stderr, "No more stored results, please make sure AddRandomResult()"
+  if (random_results_.size() == 0u) {
+    fprintf(stderr,
+            "No more stored results, please make sure AddRandomResult()"
             "is called same amount of times you're going to invoke the "
             "RandomUniform() function, i.e. once per packet.\n");
     assert(false);
@@ -45,4 +44,4 @@ double PredictivePacketManipulator::RandomUniform() {
 }
 
 }  // namespace test
-}  // namespace webrtcc
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
index 082712d870..45c7848c67 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h
@@ -31,6 +31,7 @@ class PredictivePacketManipulator : public PacketManipulatorImpl {
   // FIFO queue so they will be returned in the same order they were added.
   // Result parameter must be 0.0 to 1.0.
   void AddRandomResult(double result);
+
  protected:
   // Returns a uniformly distributed random value between 0.0 and 1.0
   double RandomUniform() override;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.cc
index 91a2f3c5f4..478b2f4901 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.cc
@@ -32,26 +32,26 @@ FrameStatistic::FrameStatistic()
       total_packets(0),
       bit_rate_in_kbps(0),
       encoded_frame_length_in_bytes(0),
-      frame_type(kDeltaFrame) {}
+      frame_type(kVideoFrameDelta) {}
 
 Stats::Stats() {}
 
 Stats::~Stats() {}
 
 bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.encode_time_in_us < s2.encode_time_in_us;
+  return s1.encode_time_in_us < s2.encode_time_in_us;
 }
 
 bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.decode_time_in_us < s2.decode_time_in_us;
+  return s1.decode_time_in_us < s2.decode_time_in_us;
 }
 
 bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
+  return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
 }
 
 bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
-    return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
+  return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
 }
 
 FrameStatistic& Stats::NewFrame(int frame_number) {
@@ -78,12 +78,11 @@ void Stats::PrintSummary() {
   size_t nbr_keyframes = 0;
   size_t nbr_nonkeyframes = 0;
 
-  for (FrameStatisticsIterator it = stats_.begin();
-      it != stats_.end(); ++it) {
+  for (FrameStatisticsIterator it = stats_.begin(); it != stats_.end(); ++it) {
     total_encoding_time_in_us += it->encode_time_in_us;
     total_decoding_time_in_us += it->decode_time_in_us;
     total_encoded_frames_lengths += it->encoded_frame_length_in_bytes;
-    if (it->frame_type == webrtc::kKeyFrame) {
+    if (it->frame_type == webrtc::kVideoFrameKey) {
       total_encoded_key_frames_lengths += it->encoded_frame_length_in_bytes;
       nbr_keyframes++;
     } else {
@@ -96,15 +95,13 @@ void Stats::PrintSummary() {
 
   // ENCODING
   printf("Encoding time:\n");
-  frame = std::min_element(stats_.begin(),
-                      stats_.end(), LessForEncodeTime);
-  printf("  Min     : %7d us (frame %d)\n",
-         frame->encode_time_in_us, frame->frame_number);
+  frame = std::min_element(stats_.begin(), stats_.end(), LessForEncodeTime);
+  printf("  Min     : %7d us (frame %d)\n", frame->encode_time_in_us,
+         frame->frame_number);
 
-  frame = std::max_element(stats_.begin(),
-                      stats_.end(), LessForEncodeTime);
-  printf("  Max     : %7d us (frame %d)\n",
-         frame->encode_time_in_us, frame->frame_number);
+  frame = std::max_element(stats_.begin(), stats_.end(), LessForEncodeTime);
+  printf("  Max     : %7d us (frame %d)\n", frame->encode_time_in_us,
+         frame->frame_number);
 
   printf("  Average : %7d us\n",
          static_cast<int>(total_encoding_time_in_us / stats_.size()));
@@ -115,7 +112,7 @@ void Stats::PrintSummary() {
   // failures)
   std::vector<FrameStatistic> decoded_frames;
   for (std::vector<FrameStatistic>::iterator it = stats_.begin();
-      it != stats_.end(); ++it) {
+       it != stats_.end(); ++it) {
     if (it->decoding_successful) {
       decoded_frames.push_back(*it);
     }
@@ -123,15 +120,15 @@ void Stats::PrintSummary() {
   if (decoded_frames.size() == 0) {
     printf("No successfully decoded frames exist in this statistics.\n");
   } else {
-    frame = std::min_element(decoded_frames.begin(),
-                        decoded_frames.end(), LessForDecodeTime);
-    printf("  Min     : %7d us (frame %d)\n",
-           frame->decode_time_in_us, frame->frame_number);
+    frame = std::min_element(decoded_frames.begin(), decoded_frames.end(),
+                             LessForDecodeTime);
+    printf("  Min     : %7d us (frame %d)\n", frame->decode_time_in_us,
+           frame->frame_number);
 
-    frame = std::max_element(decoded_frames.begin(),
-                        decoded_frames.end(), LessForDecodeTime);
-    printf("  Max     : %7d us (frame %d)\n",
-           frame->decode_time_in_us, frame->frame_number);
+    frame = std::max_element(decoded_frames.begin(), decoded_frames.end(),
+                             LessForDecodeTime);
+    printf("  Max     : %7d us (frame %d)\n", frame->decode_time_in_us,
+           frame->frame_number);
 
     printf("  Average : %7d us\n",
            static_cast<int>(total_decoding_time_in_us / decoded_frames.size()));
@@ -141,13 +138,11 @@ void Stats::PrintSummary() {
 
   // SIZE
   printf("Frame sizes:\n");
-  frame = std::min_element(stats_.begin(),
-                      stats_.end(), LessForEncodedSize);
+  frame = std::min_element(stats_.begin(), stats_.end(), LessForEncodedSize);
   printf("  Min     : %7" PRIuS " bytes (frame %d)\n",
          frame->encoded_frame_length_in_bytes, frame->frame_number);
 
-  frame = std::max_element(stats_.begin(),
-                      stats_.end(), LessForEncodedSize);
+  frame = std::max_element(stats_.begin(), stats_.end(), LessForEncodedSize);
   printf("  Max     : %7" PRIuS " bytes (frame %d)\n",
          frame->encoded_frame_length_in_bytes, frame->frame_number);
 
@@ -167,21 +162,17 @@ void Stats::PrintSummary() {
 
   // BIT RATE
   printf("Bit rates:\n");
-  frame = std::min_element(stats_.begin(),
-                      stats_.end(), LessForBitRate);
-  printf("  Min bit rate: %7d kbps (frame %d)\n",
-         frame->bit_rate_in_kbps, frame->frame_number);
+  frame = std::min_element(stats_.begin(), stats_.end(), LessForBitRate);
+  printf("  Min bit rate: %7d kbps (frame %d)\n", frame->bit_rate_in_kbps,
+         frame->frame_number);
 
-  frame = std::max_element(stats_.begin(),
-                      stats_.end(), LessForBitRate);
-  printf("  Max bit rate: %7d kbps (frame %d)\n",
-         frame->bit_rate_in_kbps, frame->frame_number);
+  frame = std::max_element(stats_.begin(), stats_.end(), LessForBitRate);
+  printf("  Max bit rate: %7d kbps (frame %d)\n", frame->bit_rate_in_kbps,
+         frame->frame_number);
 
   printf("\n");
-  printf("Total encoding time  : %7d ms.\n",
-         total_encoding_time_in_us / 1000);
-  printf("Total decoding time  : %7d ms.\n",
-         total_decoding_time_in_us / 1000);
+  printf("Total encoding time  : %7d ms.\n", total_encoding_time_in_us / 1000);
+  printf("Total decoding time  : %7d ms.\n", total_decoding_time_in_us / 1000);
   printf("Total processing time: %7d ms.\n",
          (total_encoding_time_in_us + total_decoding_time_in_us) / 1000);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.h
index 8dc8f159fe..9092631ca1 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/common_video/interface/video_image.h"
+#include "webrtc/common_video/include/video_image.h"
 
 namespace webrtc {
 namespace test {
@@ -39,7 +39,7 @@ struct FrameStatistic {
 
   // Copied from EncodedImage
   size_t encoded_frame_length_in_bytes;
-  webrtc::VideoFrameType frame_type;
+  webrtc::FrameType frame_type;
 };
 
 // Handles statistics from a single video processing run.
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
index a2d27e71d6..0403ccfdb3 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
@@ -16,21 +16,15 @@
 namespace webrtc {
 namespace test {
 
-class StatsTest: public testing::Test {
+class StatsTest : public testing::Test {
  protected:
-  StatsTest() {
-  }
+  StatsTest() {}
 
-  virtual ~StatsTest() {
-  }
+  virtual ~StatsTest() {}
 
-  void SetUp() {
-    stats_ = new Stats();
-  }
+  void SetUp() { stats_ = new Stats(); }
 
-  void TearDown() {
-    delete stats_;
-  }
+  void TearDown() { delete stats_; }
 
   Stats* stats_;
 };
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
index 10a3ff2259..7376000bd5 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -16,7 +16,7 @@
 #include <limits>
 #include <vector>
 
-#include "webrtc/system_wrappers/interface/cpu_info.h"
+#include "webrtc/system_wrappers/include/cpu_info.h"
 
 namespace webrtc {
 namespace test {
@@ -59,6 +59,7 @@ VideoProcessorImpl::VideoProcessorImpl(webrtc::VideoEncoder* encoder,
       last_frame_missing_(false),
       initialized_(false),
       encoded_frame_size_(0),
+      encoded_frame_type_(kVideoFrameKey),
       prev_time_stamp_(0),
       num_dropped_frames_(0),
       num_spatial_resizes_(0),
@@ -92,14 +93,18 @@ bool VideoProcessorImpl::Init() {
   int32_t register_result =
       encoder_->RegisterEncodeCompleteCallback(encode_callback_);
   if (register_result != WEBRTC_VIDEO_CODEC_OK) {
-    fprintf(stderr, "Failed to register encode complete callback, return code: "
-        "%d\n", register_result);
+    fprintf(stderr,
+            "Failed to register encode complete callback, return code: "
+            "%d\n",
+            register_result);
     return false;
   }
   register_result = decoder_->RegisterDecodeCompleteCallback(decode_callback_);
   if (register_result != WEBRTC_VIDEO_CODEC_OK) {
-    fprintf(stderr, "Failed to register decode complete callback, return code: "
-            "%d\n", register_result);
+    fprintf(stderr,
+            "Failed to register decode complete callback, return code: "
+            "%d\n",
+            register_result);
     return false;
   }
   // Init the encoder and decoder
@@ -145,13 +150,14 @@ VideoProcessorImpl::~VideoProcessorImpl() {
   delete decode_callback_;
 }
 
-
 void VideoProcessorImpl::SetRates(int bit_rate, int frame_rate) {
   int set_rates_result = encoder_->SetRates(bit_rate, frame_rate);
   assert(set_rates_result >= 0);
   if (set_rates_result < 0) {
-    fprintf(stderr, "Failed to update encoder with new rate %d, "
-            "return code: %d\n", bit_rate, set_rates_result);
+    fprintf(stderr,
+            "Failed to update encoder with new rate %d, "
+            "return code: %d\n",
+            bit_rate, set_rates_result);
   }
   num_dropped_frames_ = 0;
   num_spatial_resizes_ = 0;
@@ -161,6 +167,10 @@ size_t VideoProcessorImpl::EncodedFrameSize() {
   return encoded_frame_size_;
 }
 
+FrameType VideoProcessorImpl::EncodedFrameType() {
+  return encoded_frame_type_;
+}
+
 int VideoProcessorImpl::NumberDroppedFrames() {
   return num_dropped_frames_;
 }
@@ -170,7 +180,7 @@ int VideoProcessorImpl::NumberSpatialResizes() {
 }
 
 bool VideoProcessorImpl::ProcessFrame(int frame_number) {
-  assert(frame_number >=0);
+  assert(frame_number >= 0);
   if (!initialized_) {
     fprintf(stderr, "Attempting to use uninitialized VideoProcessor!\n");
     return false;
@@ -181,10 +191,8 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
   }
   if (frame_reader_->ReadFrame(source_buffer_)) {
     // Copy the source frame to the newly read frame data.
-    source_frame_.CreateFrame(source_buffer_,
-                              config_.codec_settings->width,
-                              config_.codec_settings->height,
-                              kVideoRotation_0);
+    source_frame_.CreateFrame(source_buffer_, config_.codec_settings->width,
+                              config_.codec_settings->height, kVideoRotation_0);
 
     // Ensure we have a new statistics data object we can fill:
     FrameStatistic& stat = stats_->NewFrame(frame_number);
@@ -194,14 +202,15 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
     source_frame_.set_timestamp(frame_number);
 
     // Decide if we're going to force a keyframe:
-    std::vector<VideoFrameType> frame_types(1, kDeltaFrame);
+    std::vector<FrameType> frame_types(1, kVideoFrameDelta);
     if (config_.keyframe_interval > 0 &&
         frame_number % config_.keyframe_interval == 0) {
-      frame_types[0] = kKeyFrame;
+      frame_types[0] = kVideoFrameKey;
     }
 
     // For dropped frames, we regard them as zero size encoded frames.
     encoded_frame_size_ = 0;
+    encoded_frame_type_ = kVideoFrameDelta;
 
     int32_t encode_result = encoder_->Encode(source_frame_, NULL, &frame_types);
 
@@ -218,10 +227,10 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
 
 void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   // Timestamp is frame number, so this gives us #dropped frames.
-  int num_dropped_from_prev_encode =  encoded_image._timeStamp -
-      prev_time_stamp_ - 1;
-  num_dropped_frames_ +=  num_dropped_from_prev_encode;
-  prev_time_stamp_ =  encoded_image._timeStamp;
+  int num_dropped_from_prev_encode =
+      encoded_image._timeStamp - prev_time_stamp_ - 1;
+  num_dropped_frames_ += num_dropped_from_prev_encode;
+  prev_time_stamp_ = encoded_image._timeStamp;
   if (num_dropped_from_prev_encode > 0) {
     // For dropped frames, we write out the last decoded frame to avoid getting
     // out of sync for the computation of PSNR and SSIM.
@@ -233,23 +242,26 @@ void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   // (encoder callback is only called for non-zero length frames).
   encoded_frame_size_ = encoded_image._length;
 
+  encoded_frame_type_ = encoded_image._frameType;
+
   TickTime encode_stop = TickTime::Now();
   int frame_number = encoded_image._timeStamp;
   FrameStatistic& stat = stats_->stats_[frame_number];
-  stat.encode_time_in_us = GetElapsedTimeMicroseconds(encode_start_,
-                                                      encode_stop);
+  stat.encode_time_in_us =
+      GetElapsedTimeMicroseconds(encode_start_, encode_stop);
   stat.encoding_successful = true;
   stat.encoded_frame_length_in_bytes = encoded_image._length;
   stat.frame_number = encoded_image._timeStamp;
   stat.frame_type = encoded_image._frameType;
   stat.bit_rate_in_kbps = encoded_image._length * bit_rate_factor_;
-  stat.total_packets = encoded_image._length /
-      config_.networking_config.packet_size_in_bytes + 1;
+  stat.total_packets =
+      encoded_image._length / config_.networking_config.packet_size_in_bytes +
+      1;
 
   // Perform packet loss if criteria is fullfilled:
   bool exclude_this_frame = false;
   // Only keyframes can be excluded
-  if (encoded_image._frameType == kKeyFrame) {
+  if (encoded_image._frameType == kVideoFrameKey) {
     switch (config_.exclude_frame_types) {
       case kExcludeOnlyFirstKeyFrame:
         if (!first_key_frame_has_been_excluded_) {
@@ -272,7 +284,7 @@ void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   copied_image._buffer = copied_buffer.get();
   if (!exclude_this_frame) {
     stat.packets_dropped =
-          packet_manipulator_->ManipulatePackets(&copied_image);
+        packet_manipulator_->ManipulatePackets(&copied_image);
   }
 
   // Keep track of if frames are lost due to packet loss so we can tell
@@ -292,31 +304,30 @@ void VideoProcessorImpl::FrameEncoded(const EncodedImage& encoded_image) {
   last_frame_missing_ = copied_image._length == 0;
 }
 
-void VideoProcessorImpl::FrameDecoded(const I420VideoFrame& image) {
+void VideoProcessorImpl::FrameDecoded(const VideoFrame& image) {
   TickTime decode_stop = TickTime::Now();
   int frame_number = image.timestamp();
   // Report stats
   FrameStatistic& stat = stats_->stats_[frame_number];
-  stat.decode_time_in_us = GetElapsedTimeMicroseconds(decode_start_,
-                                                      decode_stop);
+  stat.decode_time_in_us =
+      GetElapsedTimeMicroseconds(decode_start_, decode_stop);
   stat.decoding_successful = true;
 
   // Check for resize action (either down or up):
   if (static_cast<int>(image.width()) != last_encoder_frame_width_ ||
-      static_cast<int>(image.height()) != last_encoder_frame_height_ ) {
+      static_cast<int>(image.height()) != last_encoder_frame_height_) {
     ++num_spatial_resizes_;
     last_encoder_frame_width_ = image.width();
     last_encoder_frame_height_ = image.height();
   }
   // Check if codec size is different from native/original size, and if so,
   // upsample back to original size: needed for PSNR and SSIM computations.
-  if (image.width() !=  config_.codec_settings->width ||
+  if (image.width() != config_.codec_settings->width ||
       image.height() != config_.codec_settings->height) {
-    I420VideoFrame up_image;
-    int ret_val = scaler_.Set(image.width(), image.height(),
-                              config_.codec_settings->width,
-                              config_.codec_settings->height,
-                              kI420, kI420, kScaleBilinear);
+    VideoFrame up_image;
+    int ret_val = scaler_.Set(
+        image.width(), image.height(), config_.codec_settings->width,
+        config_.codec_settings->height, kI420, kI420, kScaleBilinear);
     assert(ret_val >= 0);
     if (ret_val < 0) {
       fprintf(stderr, "Failed to set scalar for frame: %d, return code: %d\n",
@@ -358,7 +369,8 @@ void VideoProcessorImpl::FrameDecoded(const I420VideoFrame& image) {
 }
 
 int VideoProcessorImpl::GetElapsedTimeMicroseconds(
-    const webrtc::TickTime& start, const webrtc::TickTime& stop) {
+    const webrtc::TickTime& start,
+    const webrtc::TickTime& stop) {
   uint64_t encode_time = (stop - start).Microseconds();
   assert(encode_time <
          static_cast<unsigned int>(std::numeric_limits<int>::max()));
@@ -396,17 +408,15 @@ const char* VideoCodecTypeToStr(webrtc::VideoCodecType e) {
 }
 
 // Callbacks
-int32_t
-VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
+int32_t VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
     const EncodedImage& encoded_image,
     const webrtc::CodecSpecificInfo* codec_specific_info,
     const webrtc::RTPFragmentationHeader* fragmentation) {
   video_processor_->FrameEncoded(encoded_image);  // Forward to parent class.
   return 0;
 }
-int32_t
-VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
-    I420VideoFrame& image) {
+int32_t VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
+    VideoFrame& image) {
   video_processor_->FrameDecoded(image);  // forward to parent class
   return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.h
index 63d736394e..3ee08fd46a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -13,15 +13,16 @@
 
 #include <string>
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/scaler.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/stats.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 namespace test {
@@ -146,6 +147,9 @@ class VideoProcessor {
   // encoder are regarded as zero size.
   virtual size_t EncodedFrameSize() = 0;
 
+  // Return the encoded frame type (key or delta).
+  virtual FrameType EncodedFrameType() = 0;
+
   // Return the number of dropped frames.
   virtual int NumberDroppedFrames() = 0;
 
@@ -170,7 +174,7 @@ class VideoProcessorImpl : public VideoProcessor {
   // Invoked by the callback when a frame has completed encoding.
   void FrameEncoded(const webrtc::EncodedImage& encodedImage);
   // Invoked by the callback when a frame has completed decoding.
-  void FrameDecoded(const webrtc::I420VideoFrame& image);
+  void FrameDecoded(const webrtc::VideoFrame& image);
   // Used for getting a 32-bit integer representing time
   // (checks the size is within signed 32-bit bounds before casting it)
   int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
@@ -179,6 +183,8 @@ class VideoProcessorImpl : public VideoProcessor {
   void SetRates(int bit_rate, int frame_rate) override;
   // Return the size of the encoded frame in bytes.
   size_t EncodedFrameSize() override;
+  // Return the encoded frame type (key or delta).
+  FrameType EncodedFrameType() override;
   // Return the number of dropped frames.
   int NumberDroppedFrames() override;
   // Return the number of spatial resizes.
@@ -199,7 +205,7 @@ class VideoProcessorImpl : public VideoProcessor {
   // Keep track of the last successful frame, since we need to write that
   // when decoding fails:
   uint8_t* last_successful_frame_buffer_;
-  webrtc::I420VideoFrame source_frame_;
+  webrtc::VideoFrame source_frame_;
   // To keep track of if we have excluded the first key frame from packet loss:
   bool first_key_frame_has_been_excluded_;
   // To tell the decoder previous frame have been dropped due to packet loss:
@@ -207,6 +213,7 @@ class VideoProcessorImpl : public VideoProcessor {
   // If Init() has executed successfully.
   bool initialized_;
   size_t encoded_frame_size_;
+  FrameType encoded_frame_type_;
   int prev_time_stamp_;
   int num_dropped_frames_;
   int num_spatial_resizes_;
@@ -236,12 +243,16 @@ class VideoProcessorImpl : public VideoProcessor {
 
   // Callback class required to implement according to the VideoDecoder API.
   class VideoProcessorDecodeCompleteCallback
-    : public webrtc::DecodedImageCallback {
+      : public webrtc::DecodedImageCallback {
    public:
-      explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
-      : video_processor_(vp) {
+    explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
+        : video_processor_(vp) {}
+    int32_t Decoded(webrtc::VideoFrame& image) override;
+    int32_t Decoded(webrtc::VideoFrame& image,
+                    int64_t decode_time_ms) override {
+      RTC_NOTREACHED();
+      return -1;
     }
-      int32_t Decoded(webrtc::I420VideoFrame& image) override;
 
    private:
     VideoProcessorImpl* video_processor_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 6c0e1254fc..7b92616e1b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -12,17 +12,16 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/test/testsupport/metrics/video_metrics.h"
 #include "webrtc/test/testsupport/packet_reader.h"
 #include "webrtc/typedefs.h"
@@ -78,9 +77,9 @@ struct RateControlMetrics {
   int max_encoding_rate_mismatch;
   int max_time_hit_target;
   int num_spatial_resizes;
+  int num_key_frames;
 };
 
-
 // Sequence used is foreman (CIF): may be better to use VGA for resize test.
 const int kCIFWidth = 352;
 const int kCIFHeight = 288;
@@ -100,7 +99,7 @@ const float kScaleKeyFrameSize = 0.5f;
 // dropping/spatial resize, and temporal layers. The limits for the rate
 // control metrics are set to be fairly conservative, so failure should only
 // happen when some significant regression or breakdown occurs.
-class VideoProcessorIntegrationTest: public testing::Test {
+class VideoProcessorIntegrationTest : public testing::Test {
  protected:
   VideoEncoder* encoder_;
   VideoDecoder* decoder_;
@@ -147,7 +146,6 @@ class VideoProcessorIntegrationTest: public testing::Test {
   bool frame_dropper_on_;
   bool spatial_resize_on_;
 
-
   VideoProcessorIntegrationTest() {}
   virtual ~VideoProcessorIntegrationTest() {}
 
@@ -164,14 +162,13 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
     // CIF is currently used for all tests below.
     // Setup the TestConfig struct for processing of a clip in CIF resolution.
-    config_.input_filename =
-        webrtc::test::ResourcePath("foreman_cif", "yuv");
+    config_.input_filename = webrtc::test::ResourcePath("foreman_cif", "yuv");
 
     // Generate an output filename in a safe way.
     config_.output_filename = webrtc::test::TempFilename(
         webrtc::test::OutputPath(), "videoprocessor_integrationtest");
-    config_.frame_length_in_bytes = CalcBufferSize(kI420,
-                                                   kCIFWidth, kCIFHeight);
+    config_.frame_length_in_bytes =
+        CalcBufferSize(kI420, kCIFWidth, kCIFHeight);
     config_.verbose = false;
     // Only allow encoder/decoder to use single core, for predictability.
     config_.use_single_core = true;
@@ -187,50 +184,46 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
     // These features may be set depending on the test.
     switch (config_.codec_settings->codecType) {
-     case kVideoCodecVP8:
-       config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
-           error_concealment_on_;
-       config_.codec_settings->codecSpecific.VP8.denoisingOn =
-           denoising_on_;
-       config_.codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
-           num_temporal_layers_;
-       config_.codec_settings->codecSpecific.VP8.frameDroppingOn =
-           frame_dropper_on_;
-       config_.codec_settings->codecSpecific.VP8.automaticResizeOn =
-           spatial_resize_on_;
-       config_.codec_settings->codecSpecific.VP8.keyFrameInterval =
-           kBaseKeyFrameInterval;
-       break;
-     case kVideoCodecVP9:
-       config_.codec_settings->codecSpecific.VP9.denoisingOn =
-           denoising_on_;
-       config_.codec_settings->codecSpecific.VP9.numberOfTemporalLayers =
-           num_temporal_layers_;
-       config_.codec_settings->codecSpecific.VP9.frameDroppingOn =
-           frame_dropper_on_;
-       config_.codec_settings->codecSpecific.VP9.keyFrameInterval =
-           kBaseKeyFrameInterval;
-       break;
-     default:
-       assert(false);
-       break;
-     }
-    frame_reader_ =
-        new webrtc::test::FrameReaderImpl(config_.input_filename,
-                                          config_.frame_length_in_bytes);
-    frame_writer_ =
-        new webrtc::test::FrameWriterImpl(config_.output_filename,
-                                          config_.frame_length_in_bytes);
+      case kVideoCodecVP8:
+        config_.codec_settings->codecSpecific.VP8.errorConcealmentOn =
+            error_concealment_on_;
+        config_.codec_settings->codecSpecific.VP8.denoisingOn = denoising_on_;
+        config_.codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
+            num_temporal_layers_;
+        config_.codec_settings->codecSpecific.VP8.frameDroppingOn =
+            frame_dropper_on_;
+        config_.codec_settings->codecSpecific.VP8.automaticResizeOn =
+            spatial_resize_on_;
+        config_.codec_settings->codecSpecific.VP8.keyFrameInterval =
+            kBaseKeyFrameInterval;
+        break;
+      case kVideoCodecVP9:
+        config_.codec_settings->codecSpecific.VP9.denoisingOn = denoising_on_;
+        config_.codec_settings->codecSpecific.VP9.numberOfTemporalLayers =
+            num_temporal_layers_;
+        config_.codec_settings->codecSpecific.VP9.frameDroppingOn =
+            frame_dropper_on_;
+        config_.codec_settings->codecSpecific.VP9.automaticResizeOn =
+            spatial_resize_on_;
+        config_.codec_settings->codecSpecific.VP9.keyFrameInterval =
+            kBaseKeyFrameInterval;
+        break;
+      default:
+        assert(false);
+        break;
+    }
+    frame_reader_ = new webrtc::test::FrameReaderImpl(
+        config_.input_filename, config_.frame_length_in_bytes);
+    frame_writer_ = new webrtc::test::FrameWriterImpl(
+        config_.output_filename, config_.frame_length_in_bytes);
     ASSERT_TRUE(frame_reader_->Init());
     ASSERT_TRUE(frame_writer_->Init());
 
     packet_manipulator_ = new webrtc::test::PacketManipulatorImpl(
         &packet_reader_, config_.networking_config, config_.verbose);
-    processor_ = new webrtc::test::VideoProcessorImpl(encoder_, decoder_,
-                                                      frame_reader_,
-                                                      frame_writer_,
-                                                      packet_manipulator_,
-                                                      config_, &stats_);
+    processor_ = new webrtc::test::VideoProcessorImpl(
+        encoder_, decoder_, frame_reader_, frame_writer_, packet_manipulator_,
+        config_, &stats_);
     ASSERT_TRUE(processor_->Init());
   }
 
@@ -244,7 +237,7 @@ class VideoProcessorIntegrationTest: public testing::Test {
       encoding_bitrate_[i] = 0.0f;
       // Update layer per-frame-bandwidth.
       per_frame_bandwidth_[i] = static_cast<float>(bit_rate_layer_[i]) /
-             static_cast<float>(frame_rate_layer_[i]);
+                                static_cast<float>(frame_rate_layer_[i]);
     }
     // Set maximum size of key frames, following setting in the VP8 wrapper.
     float max_key_size = kScaleKeyFrameSize * kOptimalBufferSize * frame_rate_;
@@ -265,34 +258,34 @@ class VideoProcessorIntegrationTest: public testing::Test {
   }
 
   // For every encoded frame, update the rate control metrics.
-  void UpdateRateControlMetrics(int frame_num, VideoFrameType frame_type) {
+  void UpdateRateControlMetrics(int frame_num, FrameType frame_type) {
     float encoded_size_kbits = processor_->EncodedFrameSize() * 8.0f / 1000.0f;
     // Update layer data.
     // Update rate mismatch relative to per-frame bandwidth for delta frames.
-    if (frame_type == kDeltaFrame) {
+    if (frame_type == kVideoFrameDelta) {
       // TODO(marpan): Should we count dropped (zero size) frames in mismatch?
-      sum_frame_size_mismatch_[layer_] += fabs(encoded_size_kbits -
-                                               per_frame_bandwidth_[layer_]) /
-                                               per_frame_bandwidth_[layer_];
+      sum_frame_size_mismatch_[layer_] +=
+          fabs(encoded_size_kbits - per_frame_bandwidth_[layer_]) /
+          per_frame_bandwidth_[layer_];
     } else {
-      float target_size = (frame_num == 1) ? target_size_key_frame_initial_ :
-          target_size_key_frame_;
-      sum_key_frame_size_mismatch_ += fabs(encoded_size_kbits - target_size) /
-          target_size;
+      float target_size = (frame_num == 1) ? target_size_key_frame_initial_
+                                           : target_size_key_frame_;
+      sum_key_frame_size_mismatch_ +=
+          fabs(encoded_size_kbits - target_size) / target_size;
       num_key_frames_ += 1;
     }
     sum_encoded_frame_size_[layer_] += encoded_size_kbits;
     // Encoding bitrate per layer: from the start of the update/run to the
     // current frame.
     encoding_bitrate_[layer_] = sum_encoded_frame_size_[layer_] *
-        frame_rate_layer_[layer_] /
-        num_frames_per_update_[layer_];
+                                frame_rate_layer_[layer_] /
+                                num_frames_per_update_[layer_];
     // Total encoding rate: from the start of the update/run to current frame.
     sum_encoded_frame_size_total_ += encoded_size_kbits;
-    encoding_bitrate_total_ = sum_encoded_frame_size_total_ * frame_rate_ /
-        num_frames_total_;
-    perc_encoding_rate_mismatch_ =  100 * fabs(encoding_bitrate_total_ -
-                                               bit_rate_) / bit_rate_;
+    encoding_bitrate_total_ =
+        sum_encoded_frame_size_total_ * frame_rate_ / num_frames_total_;
+    perc_encoding_rate_mismatch_ =
+        100 * fabs(encoding_bitrate_total_ - bit_rate_) / bit_rate_;
     if (perc_encoding_rate_mismatch_ < kPercTargetvsActualMismatch &&
         !encoding_rate_within_target_) {
       num_frames_to_hit_target_ = num_frames_total_;
@@ -307,37 +300,42 @@ class VideoProcessorIntegrationTest: public testing::Test {
                          int max_encoding_rate_mismatch,
                          int max_time_hit_target,
                          int max_num_dropped_frames,
-                         int num_spatial_resizes) {
+                         int num_spatial_resizes,
+                         int num_key_frames) {
     int num_dropped_frames = processor_->NumberDroppedFrames();
     int num_resize_actions = processor_->NumberSpatialResizes();
-    printf("For update #: %d,\n "
+    printf(
+        "For update #: %d,\n "
         " Target Bitrate: %d,\n"
         " Encoding bitrate: %f,\n"
         " Frame rate: %d \n",
         update_index, bit_rate_, encoding_bitrate_total_, frame_rate_);
-    printf(" Number of frames to approach target rate = %d, \n"
-           " Number of dropped frames = %d, \n"
-           " Number of spatial resizes = %d, \n",
-           num_frames_to_hit_target_, num_dropped_frames, num_resize_actions);
+    printf(
+        " Number of frames to approach target rate = %d, \n"
+        " Number of dropped frames = %d, \n"
+        " Number of spatial resizes = %d, \n",
+        num_frames_to_hit_target_, num_dropped_frames, num_resize_actions);
     EXPECT_LE(perc_encoding_rate_mismatch_, max_encoding_rate_mismatch);
     if (num_key_frames_ > 0) {
-      int perc_key_frame_size_mismatch = 100 * sum_key_frame_size_mismatch_ /
-              num_key_frames_;
-      printf(" Number of Key frames: %d \n"
-             " Key frame rate mismatch: %d \n",
-             num_key_frames_, perc_key_frame_size_mismatch);
+      int perc_key_frame_size_mismatch =
+          100 * sum_key_frame_size_mismatch_ / num_key_frames_;
+      printf(
+          " Number of Key frames: %d \n"
+          " Key frame rate mismatch: %d \n",
+          num_key_frames_, perc_key_frame_size_mismatch);
       EXPECT_LE(perc_key_frame_size_mismatch, max_key_frame_size_mismatch);
     }
     printf("\n");
     printf("Rates statistics for Layer data \n");
-    for (int i = 0; i < num_temporal_layers_ ; i++) {
+    for (int i = 0; i < num_temporal_layers_; i++) {
       printf("Layer #%d \n", i);
-      int perc_frame_size_mismatch = 100 * sum_frame_size_mismatch_[i] /
-        num_frames_per_update_[i];
-      int perc_encoding_rate_mismatch = 100 * fabs(encoding_bitrate_[i] -
-                                                   bit_rate_layer_[i]) /
-                                                   bit_rate_layer_[i];
-      printf(" Target Layer Bit rate: %f \n"
+      int perc_frame_size_mismatch =
+          100 * sum_frame_size_mismatch_[i] / num_frames_per_update_[i];
+      int perc_encoding_rate_mismatch =
+          100 * fabs(encoding_bitrate_[i] - bit_rate_layer_[i]) /
+          bit_rate_layer_[i];
+      printf(
+          " Target Layer Bit rate: %f \n"
           " Layer frame rate: %f, \n"
           " Layer per frame bandwidth: %f, \n"
           " Layer Encoding bit rate: %f, \n"
@@ -354,6 +352,7 @@ class VideoProcessorIntegrationTest: public testing::Test {
     EXPECT_LE(num_frames_to_hit_target_, max_time_hit_target);
     EXPECT_LE(num_dropped_frames, max_num_dropped_frames);
     EXPECT_EQ(num_resize_actions, num_spatial_resizes);
+    EXPECT_EQ(num_key_frames_, num_key_frames);
   }
 
   // Layer index corresponding to frame number, for up to 3 layers.
@@ -361,13 +360,13 @@ class VideoProcessorIntegrationTest: public testing::Test {
     if (num_temporal_layers_ == 1) {
       layer_ = 0;
     } else if (num_temporal_layers_ == 2) {
-        // layer 0:  0     2     4 ...
-        // layer 1:     1     3
-        if (frame_number % 2 == 0) {
-          layer_ = 0;
-        } else {
-          layer_ = 1;
-        }
+      // layer 0:  0     2     4 ...
+      // layer 1:     1     3
+      if (frame_number % 2 == 0) {
+        layer_ = 0;
+      } else {
+        layer_ = 1;
+      }
     } else if (num_temporal_layers_ == 3) {
       // layer 0:  0            4            8 ...
       // layer 1:        2            6
@@ -386,35 +385,26 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
   // Set the bitrate and frame rate per layer, for up to 3 layers.
   void SetLayerRates() {
-    assert(num_temporal_layers_<= 3);
+    assert(num_temporal_layers_ <= 3);
     for (int i = 0; i < num_temporal_layers_; i++) {
       float bit_rate_ratio =
           kVp8LayerRateAlloction[num_temporal_layers_ - 1][i];
       if (i > 0) {
-        float bit_rate_delta_ratio = kVp8LayerRateAlloction
-            [num_temporal_layers_ - 1][i] -
+        float bit_rate_delta_ratio =
+            kVp8LayerRateAlloction[num_temporal_layers_ - 1][i] -
             kVp8LayerRateAlloction[num_temporal_layers_ - 1][i - 1];
         bit_rate_layer_[i] = bit_rate_ * bit_rate_delta_ratio;
       } else {
         bit_rate_layer_[i] = bit_rate_ * bit_rate_ratio;
       }
-      frame_rate_layer_[i] = frame_rate_ / static_cast<float>(
-          1 << (num_temporal_layers_ - 1));
+      frame_rate_layer_[i] =
+          frame_rate_ / static_cast<float>(1 << (num_temporal_layers_ - 1));
     }
     if (num_temporal_layers_ == 3) {
       frame_rate_layer_[2] = frame_rate_ / 2.0f;
     }
   }
 
-  VideoFrameType FrameType(int frame_number) {
-    if (frame_number == 0 || ((frame_number) % key_frame_interval_ == 0 &&
-        key_frame_interval_ > 0)) {
-      return kKeyFrame;
-    } else {
-      return kDeltaFrame;
-    }
-  }
-
   void TearDown() {
     delete processor_;
     delete packet_manipulator_;
@@ -441,12 +431,12 @@ class VideoProcessorIntegrationTest: public testing::Test {
     spatial_resize_on_ = process.spatial_resize_on;
     SetUpCodecConfig();
     // Update the layers and the codec with the initial rates.
-    bit_rate_ =  rate_profile.target_bit_rate[0];
+    bit_rate_ = rate_profile.target_bit_rate[0];
     frame_rate_ = rate_profile.input_frame_rate[0];
     SetLayerRates();
     // Set the initial target size for key frame.
-    target_size_key_frame_initial_ = 0.5 * kInitialBufferSize *
-        bit_rate_layer_[0];
+    target_size_key_frame_initial_ =
+        0.5 * kInitialBufferSize * bit_rate_layer_[0];
     processor_->SetRates(bit_rate_, frame_rate_);
     // Process each frame, up to |num_frames|.
     int num_frames = rate_profile.num_frames;
@@ -454,12 +444,13 @@ class VideoProcessorIntegrationTest: public testing::Test {
     ResetRateControlMetrics(
         rate_profile.frame_index_rate_update[update_index + 1]);
     int frame_number = 0;
-    VideoFrameType frame_type = kDeltaFrame;
+    FrameType frame_type = kVideoFrameDelta;
     while (processor_->ProcessFrame(frame_number) &&
-        frame_number < num_frames) {
+           frame_number < num_frames) {
       // Get the layer index for the frame |frame_number|.
       LayerIndexForFrame(frame_number);
-      frame_type = FrameType(frame_number);
+      // Get the frame_type.
+      frame_type = processor_->EncodedFrameType();
       // Counter for whole sequence run.
       ++frame_number;
       // Counters for each rate update.
@@ -471,31 +462,31 @@ class VideoProcessorIntegrationTest: public testing::Test {
       if (frame_number ==
           rate_profile.frame_index_rate_update[update_index + 1]) {
         VerifyRateControl(
-            update_index,
-            rc_metrics[update_index].max_key_frame_size_mismatch,
+            update_index, rc_metrics[update_index].max_key_frame_size_mismatch,
             rc_metrics[update_index].max_delta_frame_size_mismatch,
             rc_metrics[update_index].max_encoding_rate_mismatch,
             rc_metrics[update_index].max_time_hit_target,
             rc_metrics[update_index].max_num_dropped_frames,
-            rc_metrics[update_index].num_spatial_resizes);
+            rc_metrics[update_index].num_spatial_resizes,
+            rc_metrics[update_index].num_key_frames);
         // Update layer rates and the codec with new rates.
         ++update_index;
-        bit_rate_ =  rate_profile.target_bit_rate[update_index];
+        bit_rate_ = rate_profile.target_bit_rate[update_index];
         frame_rate_ = rate_profile.input_frame_rate[update_index];
         SetLayerRates();
-        ResetRateControlMetrics(rate_profile.
-                                frame_index_rate_update[update_index + 1]);
+        ResetRateControlMetrics(
+            rate_profile.frame_index_rate_update[update_index + 1]);
         processor_->SetRates(bit_rate_, frame_rate_);
       }
     }
-    VerifyRateControl(
-        update_index,
-        rc_metrics[update_index].max_key_frame_size_mismatch,
-        rc_metrics[update_index].max_delta_frame_size_mismatch,
-        rc_metrics[update_index].max_encoding_rate_mismatch,
-        rc_metrics[update_index].max_time_hit_target,
-        rc_metrics[update_index].max_num_dropped_frames,
-        rc_metrics[update_index].num_spatial_resizes);
+    VerifyRateControl(update_index,
+                      rc_metrics[update_index].max_key_frame_size_mismatch,
+                      rc_metrics[update_index].max_delta_frame_size_mismatch,
+                      rc_metrics[update_index].max_encoding_rate_mismatch,
+                      rc_metrics[update_index].max_time_hit_target,
+                      rc_metrics[update_index].max_num_dropped_frames,
+                      rc_metrics[update_index].num_spatial_resizes,
+                      rc_metrics[update_index].num_key_frames);
     EXPECT_EQ(num_frames, frame_number);
     EXPECT_EQ(num_frames + 1, static_cast<int>(stats_.stats_.size()));
 
@@ -508,16 +499,14 @@ class VideoProcessorIntegrationTest: public testing::Test {
 
     // TODO(marpan): should compute these quality metrics per SetRates update.
     webrtc::test::QualityMetricsResult psnr_result, ssim_result;
-    EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
-        config_.input_filename.c_str(),
-        config_.output_filename.c_str(),
-        config_.codec_settings->width,
-        config_.codec_settings->height,
-        &psnr_result,
-        &ssim_result));
+    EXPECT_EQ(
+        0, webrtc::test::I420MetricsFromFiles(
+               config_.input_filename.c_str(), config_.output_filename.c_str(),
+               config_.codec_settings->width, config_.codec_settings->height,
+               &psnr_result, &ssim_result));
     printf("PSNR avg: %f, min: %f    SSIM avg: %f, min: %f\n",
-           psnr_result.average, psnr_result.min,
-           ssim_result.average, ssim_result.min);
+           psnr_result.average, psnr_result.min, ssim_result.average,
+           ssim_result.min);
     stats_.PrintSummary();
     EXPECT_GT(psnr_result.average, quality_metrics.minimum_avg_psnr);
     EXPECT_GT(psnr_result.min, quality_metrics.minimum_min_psnr);
@@ -550,7 +539,7 @@ void SetCodecParameters(CodecConfigPars* process_settings,
                         bool spatial_resize_on) {
   process_settings->codec_type = codec_type;
   process_settings->packet_loss = packet_loss;
-  process_settings->key_frame_interval =  key_frame_interval;
+  process_settings->key_frame_interval = key_frame_interval;
   process_settings->num_temporal_layers = num_temporal_layers,
   process_settings->error_concealment_on = error_concealment_on;
   process_settings->denoising_on = denoising_on;
@@ -576,7 +565,8 @@ void SetRateControlMetrics(RateControlMetrics* rc_metrics,
                            int max_delta_frame_size_mismatch,
                            int max_encoding_rate_mismatch,
                            int max_time_hit_target,
-                           int num_spatial_resizes) {
+                           int num_spatial_resizes,
+                           int num_key_frames) {
   rc_metrics[update_index].max_num_dropped_frames = max_num_dropped_frames;
   rc_metrics[update_index].max_key_frame_size_mismatch =
       max_key_frame_size_mismatch;
@@ -586,6 +576,7 @@ void SetRateControlMetrics(RateControlMetrics* rc_metrics,
       max_encoding_rate_mismatch;
   rc_metrics[update_index].max_time_hit_target = max_time_hit_target;
   rc_metrics[update_index].num_spatial_resizes = num_spatial_resizes;
+  rc_metrics[update_index].num_key_frames = num_key_frames;
 }
 
 // VP9: Run with no packet loss and fixed bitrate. Quality should be very high.
@@ -606,10 +597,8 @@ TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossVP9) {
   SetQualityMetrics(&quality_metrics, 37.0, 36.0, 0.93, 0.92);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -630,14 +619,11 @@ TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLossVP9) {
   SetQualityMetrics(&quality_metrics, 17.0, 14.0, 0.45, 0.36);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
-
 // VP9: Run with no packet loss, with varying bitrate (3 rate updates):
 // low to high to medium. Check that quality and encoder response to the new
 // target rate/per-frame bandwidth (for each rate update) is within limits.
@@ -656,15 +642,13 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossChangeBitRateVP9) {
                      false, true, false);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
-  SetQualityMetrics(&quality_metrics, 35.9, 30.0, 0.90, 0.85);
+  SetQualityMetrics(&quality_metrics, 35.7, 30.0, 0.90, 0.85);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
-  SetRateControlMetrics(rc_metrics, 0, 0, 30, 20, 20, 30, 0);
-  SetRateControlMetrics(rc_metrics, 1, 2, 0, 20, 20, 60, 0);
-  SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 20, 40, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 30, 20, 20, 30, 0, 1);
+  SetRateControlMetrics(rc_metrics, 1, 2, 0, 20, 20, 60, 0, 0);
+  SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 20, 40, 0, 0);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -694,12 +678,10 @@ TEST_F(VideoProcessorIntegrationTest,
   SetQualityMetrics(&quality_metrics, 31.5, 18.0, 0.80, 0.44);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
-  SetRateControlMetrics(rc_metrics, 0, 35, 50, 70, 15, 45, 0);
-  SetRateControlMetrics(rc_metrics, 1, 10, 0, 40, 10, 30, 0);
-  SetRateControlMetrics(rc_metrics, 2, 5, 0, 30, 5, 20, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 38, 50, 75, 15, 45, 0, 1);
+  SetRateControlMetrics(rc_metrics, 1, 10, 0, 40, 10, 30, 0, 0);
+  SetRateControlMetrics(rc_metrics, 2, 5, 0, 30, 5, 20, 0, 0);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -719,10 +701,32 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossDenoiserOnVP9) {
   SetQualityMetrics(&quality_metrics, 36.8, 35.8, 0.92, 0.91);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 20, 0, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
+                         rc_metrics);
+}
+
+// Run with no packet loss, at low bitrate.
+// spatial_resize is on, for this low bitrate expect one resize in sequence.
+// Resize happens on delta frame. Expect only one key frame (first frame).
+TEST_F(VideoProcessorIntegrationTest, ProcessNoLossSpatialResizeFrameDropVP9) {
+  config_.networking_config.packet_loss_probability = 0;
+  // Bitrate and frame rate profile.
+  RateProfile rate_profile;
+  SetRateProfilePars(&rate_profile, 0, 50, 30, 0);
+  rate_profile.frame_index_rate_update[1] = kNbrFramesLong + 1;
+  rate_profile.num_frames = kNbrFramesLong;
+  // Codec/network settings.
+  CodecConfigPars process_settings;
+  SetCodecParameters(&process_settings, kVideoCodecVP9, 0.0f, -1, 1, false,
+                     false, true, true);
+  // Metrics for expected quality.
+  QualityMetrics quality_metrics;
+  SetQualityMetrics(&quality_metrics, 24.0, 13.0, 0.65, 0.37);
+  // Metrics for rate control.
+  RateControlMetrics rc_metrics[1];
+  SetRateControlMetrics(rc_metrics, 0, 228, 70, 160, 15, 80, 1, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -747,10 +751,8 @@ TEST_F(VideoProcessorIntegrationTest, ProcessZeroPacketLoss) {
   SetQualityMetrics(&quality_metrics, 34.95, 33.0, 0.90, 0.89);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -771,10 +773,8 @@ TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLoss) {
   SetQualityMetrics(&quality_metrics, 20.0, 16.0, 0.60, 0.40);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -795,10 +795,8 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
   SetQualityMetrics(&quality_metrics, 19.0, 16.0, 0.50, 0.35);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 40, 20, 10, 15, 0, 1);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -815,8 +813,13 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
 // low to high to medium. Check that quality and encoder response to the new
 // target rate/per-frame bandwidth (for each rate update) is within limits.
 // One key frame (first frame only) in sequence.
-TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossChangeBitRateVP8)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossChangeBitRateVP8 \
+  DISABLED_ProcessNoLossChangeBitRateVP8
+#else
+#define MAYBE_ProcessNoLossChangeBitRateVP8 ProcessNoLossChangeBitRateVP8
+#endif
+TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossChangeBitRateVP8) {
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
   SetRateProfilePars(&rate_profile, 0, 200, 30, 0);
@@ -833,12 +836,10 @@ TEST_F(VideoProcessorIntegrationTest,
   SetQualityMetrics(&quality_metrics, 34.0, 32.0, 0.85, 0.80);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
-  SetRateControlMetrics(rc_metrics, 0, 0, 45, 20, 10, 15, 0);
-  SetRateControlMetrics(rc_metrics, 1, 0, 0, 25, 20, 10, 0);
-  SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 15, 10, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 45, 20, 10, 15, 0, 1);
+  SetRateControlMetrics(rc_metrics, 1, 0, 0, 25, 20, 10, 0, 0);
+  SetRateControlMetrics(rc_metrics, 2, 0, 0, 25, 15, 10, 0, 0);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -849,8 +850,15 @@ TEST_F(VideoProcessorIntegrationTest,
 // for the rate control metrics can be lower. One key frame (first frame only).
 // Note: quality after update should be higher but we currently compute quality
 // metrics averaged over whole sequence run.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
+  DISABLED_ProcessNoLossChangeFrameRateFrameDropVP8
+#else
+#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
+  ProcessNoLossChangeFrameRateFrameDropVP8
+#endif
 TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossChangeFrameRateFrameDropVP8)) {
+       MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -868,19 +876,24 @@ TEST_F(VideoProcessorIntegrationTest,
   SetQualityMetrics(&quality_metrics, 31.0, 22.0, 0.80, 0.65);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[3];
-  SetRateControlMetrics(rc_metrics, 0, 40, 20, 75, 15, 60, 0);
-  SetRateControlMetrics(rc_metrics, 1, 10, 0, 25, 10, 35, 0);
-  SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 10, 15, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 40, 20, 75, 15, 60, 0, 1);
+  SetRateControlMetrics(rc_metrics, 1, 10, 0, 25, 10, 35, 0, 0);
+  SetRateControlMetrics(rc_metrics, 2, 0, 0, 20, 10, 15, 0, 0);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
 // Run with no packet loss, at low bitrate. During this time we should've
-// resized once.
+// resized once. Expect 2 key frames generated (first and one for resize).
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
+  DISABLED_ProcessNoLossSpatialResizeFrameDropVP8
+#else
+#define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
+  ProcessNoLossSpatialResizeFrameDropVP8
+#endif
 TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossSpatialResizeFrameDropVP8)) {
+       MAYBE_ProcessNoLossSpatialResizeFrameDropVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -889,17 +902,15 @@ TEST_F(VideoProcessorIntegrationTest,
   rate_profile.num_frames = kNbrFramesLong;
   // Codec/network settings.
   CodecConfigPars process_settings;
-  SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, kNbrFramesLong,
-                     1, false, true, true, true);
+  SetCodecParameters(&process_settings, kVideoCodecVP8, 0.0f, -1, 1, false,
+                     true, true, true);
   // Metrics for expected quality.
   QualityMetrics quality_metrics;
   SetQualityMetrics(&quality_metrics, 25.0, 15.0, 0.70, 0.40);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 160, 60, 120, 20, 70, 1);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 160, 60, 120, 20, 70, 1, 2);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 
@@ -908,8 +919,13 @@ TEST_F(VideoProcessorIntegrationTest,
 // encoding rate mismatch are applied to each layer.
 // No dropped frames in this test, and internal spatial resizer is off.
 // One key frame (first frame only) in sequence, so no spatial resizing.
-TEST_F(VideoProcessorIntegrationTest,
-       DISABLED_ON_ANDROID(ProcessNoLossTemporalLayersVP8)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_ProcessNoLossTemporalLayersVP8 \
+  DISABLED_ProcessNoLossTemporalLayersVP8
+#else
+#define MAYBE_ProcessNoLossTemporalLayersVP8 ProcessNoLossTemporalLayersVP8
+#endif
+TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossTemporalLayersVP8) {
   config_.networking_config.packet_loss_probability = 0;
   // Bitrate and frame rate profile.
   RateProfile rate_profile;
@@ -926,11 +942,9 @@ TEST_F(VideoProcessorIntegrationTest,
   SetQualityMetrics(&quality_metrics, 32.5, 30.0, 0.85, 0.80);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[2];
-  SetRateControlMetrics(rc_metrics, 0, 0, 20, 30, 10, 10, 0);
-  SetRateControlMetrics(rc_metrics, 1, 0, 0, 30, 15, 10, 0);
-  ProcessFramesAndVerify(quality_metrics,
-                         rate_profile,
-                         process_settings,
+  SetRateControlMetrics(rc_metrics, 0, 0, 20, 30, 10, 10, 0, 1);
+  SetRateControlMetrics(rc_metrics, 1, 0, 0, 30, 15, 10, 0, 0);
+  ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
index 88b5467f1f..148d8dc74a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
@@ -10,10 +10,10 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/test/testsupport/mock/mock_frame_reader.h"
 #include "webrtc/test/testsupport/mock/mock_frame_writer.h"
 #include "webrtc/test/testsupport/packet_reader.h"
@@ -29,7 +29,7 @@ namespace test {
 
 // Very basic testing for VideoProcessor. It's mostly tested by running the
 // video_quality_measurement program.
-class VideoProcessorTest: public testing::Test {
+class VideoProcessorTest : public testing::Test {
  protected:
   MockVideoEncoder encoder_mock_;
   MockVideoDecoder decoder_mock_;
@@ -53,44 +53,34 @@ class VideoProcessorTest: public testing::Test {
   void TearDown() {}
 
   void ExpectInit() {
-    EXPECT_CALL(encoder_mock_, InitEncode(_, _, _))
-      .Times(1);
+    EXPECT_CALL(encoder_mock_, InitEncode(_, _, _)).Times(1);
     EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_))
-      .Times(AtLeast(1));
-    EXPECT_CALL(decoder_mock_, InitDecode(_, _))
-      .Times(1);
+        .Times(AtLeast(1));
+    EXPECT_CALL(decoder_mock_, InitDecode(_, _)).Times(1);
     EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_))
-      .Times(AtLeast(1));
-    EXPECT_CALL(frame_reader_mock_, NumberOfFrames())
-      .WillOnce(Return(1));
-    EXPECT_CALL(frame_reader_mock_, FrameLength())
-      .WillOnce(Return(152064));
+        .Times(AtLeast(1));
+    EXPECT_CALL(frame_reader_mock_, NumberOfFrames()).WillOnce(Return(1));
+    EXPECT_CALL(frame_reader_mock_, FrameLength()).WillOnce(Return(152064));
   }
 };
 
 TEST_F(VideoProcessorTest, Init) {
   ExpectInit();
-  VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
-                                     &frame_reader_mock_,
-                                     &frame_writer_mock_,
-                                     &packet_manipulator_mock_, config_,
-                                     &stats_);
+  VideoProcessorImpl video_processor(
+      &encoder_mock_, &decoder_mock_, &frame_reader_mock_, &frame_writer_mock_,
+      &packet_manipulator_mock_, config_, &stats_);
   ASSERT_TRUE(video_processor.Init());
 }
 
 TEST_F(VideoProcessorTest, ProcessFrame) {
   ExpectInit();
-  EXPECT_CALL(encoder_mock_, Encode(_, _, _))
-    .Times(1);
-  EXPECT_CALL(frame_reader_mock_, ReadFrame(_))
-    .WillOnce(Return(true));
+  EXPECT_CALL(encoder_mock_, Encode(_, _, _)).Times(1);
+  EXPECT_CALL(frame_reader_mock_, ReadFrame(_)).WillOnce(Return(true));
   // Since we don't return any callback from the mock, the decoder will not
   // be more than initialized...
-  VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
-                                     &frame_reader_mock_,
-                                     &frame_writer_mock_,
-                                     &packet_manipulator_mock_, config_,
-                                     &stats_);
+  VideoProcessorImpl video_processor(
+      &encoder_mock_, &decoder_mock_, &frame_reader_mock_, &frame_writer_mock_,
+      &packet_manipulator_mock_, config_, &stats_);
   ASSERT_TRUE(video_processor.Init());
   video_processor.ProcessFrame(0);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
index ced92bce24..37fad483f7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
@@ -16,7 +16,7 @@
 #include <sys/stat.h>  // To check for directory existence.
 
 #ifndef S_ISDIR  // Not defined in stat.h on Windows.
-#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
+#define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
 #endif
 
 #include "gflags/gflags.h"
@@ -26,76 +26,110 @@
 #include "webrtc/modules/video_coding/codecs/test/stats.h"
 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
 #include "webrtc/test/testsupport/metrics/video_metrics.h"
 #include "webrtc/test/testsupport/packet_reader.h"
 
 DEFINE_string(test_name, "Quality test", "The name of the test to run. ");
-DEFINE_string(test_description, "", "A more detailed description about what "
+DEFINE_string(test_description,
+              "",
+              "A more detailed description about what "
               "the current test is about.");
-DEFINE_string(input_filename, "", "Input file. "
+DEFINE_string(input_filename,
+              "",
+              "Input file. "
               "The source video file to be encoded and decoded. Must be in "
               ".yuv format");
 DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
 DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
-DEFINE_int32(framerate, 30, "Frame rate of the input file, in FPS "
+DEFINE_int32(framerate,
+             30,
+             "Frame rate of the input file, in FPS "
              "(frames-per-second). ");
-DEFINE_string(output_dir, ".", "Output directory. "
+DEFINE_string(output_dir,
+              ".",
+              "Output directory. "
               "The directory where the output file will be put. Must already "
               "exist.");
-DEFINE_bool(use_single_core, false, "Force using a single core. If set to "
+DEFINE_bool(use_single_core,
+            false,
+            "Force using a single core. If set to "
             "true, only one core will be used for processing. Using a single "
             "core is necessary to get a deterministic behavior for the"
             "encoded frames - using multiple cores will produce different "
             "encoded frames since multiple cores are competing to consume the "
             "byte budget for each frame in parallel. If set to false, "
             "the maximum detected number of cores will be used. ");
-DEFINE_bool(disable_fixed_random_seed , false, "Set this flag to disable the"
+DEFINE_bool(disable_fixed_random_seed,
+            false,
+            "Set this flag to disable the"
             "usage of a fixed random seed for the random generator used "
             "for packet loss. Disabling this will cause consecutive runs "
             "loose packets at different locations, which is bad for "
             "reproducibility.");
-DEFINE_string(output_filename, "", "Output file. "
+DEFINE_string(output_filename,
+              "",
+              "Output file. "
               "The name of the output video file resulting of the processing "
               "of the source file. By default this is the same name as the "
               "input file with '_out' appended before the extension.");
 DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
-DEFINE_int32(keyframe_interval, 0, "Forces a keyframe every Nth frame. "
+DEFINE_int32(keyframe_interval,
+             0,
+             "Forces a keyframe every Nth frame. "
              "0 means the encoder decides when to insert keyframes.  Note that "
              "the encoder may create a keyframe in other locations in addition "
              "to the interval that is set using this parameter.");
-DEFINE_int32(temporal_layers, 0, "The number of temporal layers to use "
+DEFINE_int32(temporal_layers,
+             0,
+             "The number of temporal layers to use "
              "(VP8 specific codec setting). Must be 0-4.");
-DEFINE_int32(packet_size, 1500, "Simulated network packet size in bytes (MTU). "
+DEFINE_int32(packet_size,
+             1500,
+             "Simulated network packet size in bytes (MTU). "
              "Used for packet loss simulation.");
-DEFINE_int32(max_payload_size, 1440, "Max payload size in bytes for the "
+DEFINE_int32(max_payload_size,
+             1440,
+             "Max payload size in bytes for the "
              "encoder.");
-DEFINE_string(packet_loss_mode, "uniform", "Packet loss mode. Two different "
+DEFINE_string(packet_loss_mode,
+              "uniform",
+              "Packet loss mode. Two different "
               "packet loss models are supported: uniform or burst. This "
               "setting has no effect unless packet_loss_rate is >0. ");
-DEFINE_double(packet_loss_probability, 0.0, "Packet loss probability. A value "
+DEFINE_double(packet_loss_probability,
+              0.0,
+              "Packet loss probability. A value "
               "between 0.0 and 1.0 that defines the probability of a packet "
               "being lost. 0.1 means 10% and so on.");
-DEFINE_int32(packet_loss_burst_length, 1, "Packet loss burst length. Defines "
+DEFINE_int32(packet_loss_burst_length,
+             1,
+             "Packet loss burst length. Defines "
              "how many packets will be lost in a burst when a packet has been "
              "decided to be lost. Must be >=1.");
-DEFINE_bool(csv, false, "CSV output. Enabling this will output all frame "
+DEFINE_bool(csv,
+            false,
+            "CSV output. Enabling this will output all frame "
             "statistics at the end of execution. Recommended to run combined "
             "with --noverbose to avoid mixing output.");
-DEFINE_bool(python, false, "Python output. Enabling this will output all frame "
+DEFINE_bool(python,
+            false,
+            "Python output. Enabling this will output all frame "
             "statistics as a Python script at the end of execution. "
             "Recommended to run combine with --noverbose to avoid mixing "
             "output.");
-DEFINE_bool(verbose, true, "Verbose mode. Prints a lot of debugging info. "
+DEFINE_bool(verbose,
+            true,
+            "Verbose mode. Prints a lot of debugging info. "
             "Suitable for tracking progress but not for capturing output. "
             "Disable with --noverbose flag.");
 
 // Custom log method that only prints if the verbose flag is given.
 // Supports all the standard printf parameters and formatting (just forwarded).
-int Log(const char *format, ...) {
+int Log(const char* format, ...) {
   int result = 0;
   if (FLAGS_verbose) {
     va_list args;
@@ -111,7 +145,7 @@ int Log(const char *format, ...) {
 // Returns 0 if everything is OK, otherwise an exit code.
 int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
   // Validate the mandatory flags:
-  if (FLAGS_input_filename == "" || FLAGS_width == -1 || FLAGS_height == -1) {
+  if (FLAGS_input_filename.empty() || FLAGS_width == -1 || FLAGS_height == -1) {
     printf("%s\n", google::ProgramUsage());
     return 1;
   }
@@ -132,15 +166,15 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
   // Verify the output dir exists.
   struct stat dir_info;
   if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
-      S_ISDIR(dir_info.st_mode))) {
+        S_ISDIR(dir_info.st_mode))) {
     fprintf(stderr, "Cannot find output directory: %s\n",
-              FLAGS_output_dir.c_str());
+            FLAGS_output_dir.c_str());
     return 3;
   }
   config->output_dir = FLAGS_output_dir;
 
   // Manufacture an output filename if none was given.
-  if (FLAGS_output_filename == "") {
+  if (FLAGS_output_filename.empty()) {
     // Cut out the filename without extension from the given input file
     // (which may include a path)
     int startIndex = FLAGS_input_filename.find_last_of("/") + 1;
@@ -148,16 +182,16 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
       startIndex = 0;
     }
     FLAGS_output_filename =
-        FLAGS_input_filename.substr(startIndex,
-                                    FLAGS_input_filename.find_last_of(".")
-                                    - startIndex) + "_out.yuv";
+        FLAGS_input_filename.substr(
+            startIndex, FLAGS_input_filename.find_last_of(".") - startIndex) +
+        "_out.yuv";
   }
 
   // Verify output file can be written.
   if (FLAGS_output_dir == ".") {
     config->output_filename = FLAGS_output_filename;
   } else {
-    config->output_filename = FLAGS_output_dir + "/"+ FLAGS_output_filename;
+    config->output_filename = FLAGS_output_dir + "/" + FLAGS_output_filename;
   }
   test_file = fopen(config->output_filename.c_str(), "wb");
   if (test_file == NULL) {
@@ -232,27 +266,32 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
   // Check packet loss settings
   if (FLAGS_packet_loss_mode != "uniform" &&
       FLAGS_packet_loss_mode != "burst") {
-    fprintf(stderr, "Unsupported packet loss mode, must be 'uniform' or "
+    fprintf(stderr,
+            "Unsupported packet loss mode, must be 'uniform' or "
             "'burst'\n.");
     return 10;
   }
   config->networking_config.packet_loss_mode = webrtc::test::kUniform;
   if (FLAGS_packet_loss_mode == "burst") {
-    config->networking_config.packet_loss_mode =  webrtc::test::kBurst;
+    config->networking_config.packet_loss_mode = webrtc::test::kBurst;
   }
 
   if (FLAGS_packet_loss_probability < 0.0 ||
       FLAGS_packet_loss_probability > 1.0) {
-    fprintf(stderr, "Invalid packet loss probability. Must be 0.0 - 1.0, "
-            "was: %f\n", FLAGS_packet_loss_probability);
+    fprintf(stderr,
+            "Invalid packet loss probability. Must be 0.0 - 1.0, "
+            "was: %f\n",
+            FLAGS_packet_loss_probability);
     return 11;
   }
   config->networking_config.packet_loss_probability =
       FLAGS_packet_loss_probability;
 
   if (FLAGS_packet_loss_burst_length < 1) {
-    fprintf(stderr, "Invalid packet loss burst length, must be >=1, "
-            "was: %d\n", FLAGS_packet_loss_burst_length);
+    fprintf(stderr,
+            "Invalid packet loss burst length, must be >=1, "
+            "was: %d\n",
+            FLAGS_packet_loss_burst_length);
     return 12;
   }
   config->networking_config.packet_loss_burst_length =
@@ -264,10 +303,9 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
 void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
                                webrtc::test::QualityMetricsResult* result) {
   Log("Calculating SSIM...\n");
-  I420SSIMFromFiles(config->input_filename.c_str(),
-                    config->output_filename.c_str(),
-                    config->codec_settings->width,
-                    config->codec_settings->height, result);
+  I420SSIMFromFiles(
+      config->input_filename.c_str(), config->output_filename.c_str(),
+      config->codec_settings->width, config->codec_settings->height, result);
   Log("  Average: %3.2f\n", result->average);
   Log("  Min    : %3.2f (frame %d)\n", result->min, result->min_frame_number);
   Log("  Max    : %3.2f (frame %d)\n", result->max, result->max_frame_number);
@@ -276,10 +314,9 @@ void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
 void CalculatePsnrVideoMetrics(webrtc::test::TestConfig* config,
                                webrtc::test::QualityMetricsResult* result) {
   Log("Calculating PSNR...\n");
-  I420PSNRFromFiles(config->input_filename.c_str(),
-                    config->output_filename.c_str(),
-                    config->codec_settings->width,
-                    config->codec_settings->height, result);
+  I420PSNRFromFiles(
+      config->input_filename.c_str(), config->output_filename.c_str(),
+      config->codec_settings->width, config->codec_settings->height, result);
   Log("  Average: %3.2f\n", result->average);
   Log("  Min    : %3.2f (frame %d)\n", result->min, result->min_frame_number);
   Log("  Max    : %3.2f (frame %d)\n", result->max, result->max_frame_number);
@@ -309,9 +346,11 @@ void PrintConfigurationSummary(const webrtc::test::TestConfig& config) {
 void PrintCsvOutput(const webrtc::test::Stats& stats,
                     const webrtc::test::QualityMetricsResult& ssim_result,
                     const webrtc::test::QualityMetricsResult& psnr_result) {
-  Log("\nCSV output (recommended to run with --noverbose to skip the "
-              "above output)\n");
-  printf("frame_number encoding_successful decoding_successful "
+  Log(
+      "\nCSV output (recommended to run with --noverbose to skip the "
+      "above output)\n");
+  printf(
+      "frame_number encoding_successful decoding_successful "
       "encode_return_code decode_return_code "
       "encode_time_in_us decode_time_in_us "
       "bit_rate_in_kbps encoded_frame_length_in_bytes frame_type "
@@ -322,22 +361,13 @@ void PrintCsvOutput(const webrtc::test::Stats& stats,
     const webrtc::test::FrameStatistic& f = stats.stats_[i];
     const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
     const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
-    printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7" PRIuS ", %d, %2d, %2"
-           PRIuS ", %5.3f, %5.2f\n",
-           f.frame_number,
-           f.encoding_successful,
-           f.decoding_successful,
-           f.encode_return_code,
-           f.decode_return_code,
-           f.encode_time_in_us,
-           f.decode_time_in_us,
-           f.bit_rate_in_kbps,
-           f.encoded_frame_length_in_bytes,
-           f.frame_type,
-           f.packets_dropped,
-           f.total_packets,
-           ssim.value,
-           psnr.value);
+    printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7" PRIuS
+           ", %d, %2d, %2" PRIuS ", %5.3f, %5.2f\n",
+           f.frame_number, f.encoding_successful, f.decoding_successful,
+           f.encode_return_code, f.decode_return_code, f.encode_time_in_us,
+           f.decode_time_in_us, f.bit_rate_in_kbps,
+           f.encoded_frame_length_in_bytes, f.frame_type, f.packets_dropped,
+           f.total_packets, ssim.value, psnr.value);
   }
 }
 
@@ -345,91 +375,85 @@ void PrintPythonOutput(const webrtc::test::TestConfig& config,
                        const webrtc::test::Stats& stats,
                        const webrtc::test::QualityMetricsResult& ssim_result,
                        const webrtc::test::QualityMetricsResult& psnr_result) {
-  Log("\nPython output (recommended to run with --noverbose to skip the "
-               "above output)\n");
-  printf("test_configuration = ["
-         "{'name': 'name',                      'value': '%s'},\n"
-         "{'name': 'description',               'value': '%s'},\n"
-         "{'name': 'test_number',               'value': '%d'},\n"
-         "{'name': 'input_filename',            'value': '%s'},\n"
-         "{'name': 'output_filename',           'value': '%s'},\n"
-         "{'name': 'output_dir',                'value': '%s'},\n"
-         "{'name': 'packet_size_in_bytes',      'value': '%" PRIuS "'},\n"
-         "{'name': 'max_payload_size_in_bytes', 'value': '%" PRIuS "'},\n"
-         "{'name': 'packet_loss_mode',          'value': '%s'},\n"
-         "{'name': 'packet_loss_probability',   'value': '%f'},\n"
-         "{'name': 'packet_loss_burst_length',  'value': '%d'},\n"
-         "{'name': 'exclude_frame_types',       'value': '%s'},\n"
-         "{'name': 'frame_length_in_bytes',     'value': '%" PRIuS "'},\n"
-         "{'name': 'use_single_core',           'value': '%s'},\n"
-         "{'name': 'keyframe_interval;',        'value': '%d'},\n"
-         "{'name': 'video_codec_type',          'value': '%s'},\n"
-         "{'name': 'width',                     'value': '%d'},\n"
-         "{'name': 'height',                    'value': '%d'},\n"
-         "{'name': 'bit_rate_in_kbps',          'value': '%d'},\n"
-         "]\n",
-         config.name.c_str(),
-         config.description.c_str(),
-         config.test_number,
-         config.input_filename.c_str(),
-         config.output_filename.c_str(),
-         config.output_dir.c_str(),
-         config.networking_config.packet_size_in_bytes,
-         config.networking_config.max_payload_size_in_bytes,
-         PacketLossModeToStr(config.networking_config.packet_loss_mode),
-         config.networking_config.packet_loss_probability,
-         config.networking_config.packet_loss_burst_length,
-         ExcludeFrameTypesToStr(config.exclude_frame_types),
-         config.frame_length_in_bytes,
-         config.use_single_core ? "True " : "False",
-         config.keyframe_interval,
-         webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
-         config.codec_settings->width,
-         config.codec_settings->height,
-         config.codec_settings->startBitrate);
-  printf("frame_data_types = {"
-         "'frame_number': ('number', 'Frame number'),\n"
-         "'encoding_successful': ('boolean', 'Encoding successful?'),\n"
-         "'decoding_successful': ('boolean', 'Decoding successful?'),\n"
-         "'encode_time': ('number', 'Encode time (us)'),\n"
-         "'decode_time': ('number', 'Decode time (us)'),\n"
-         "'encode_return_code': ('number', 'Encode return code'),\n"
-         "'decode_return_code': ('number', 'Decode return code'),\n"
-         "'bit_rate': ('number', 'Bit rate (kbps)'),\n"
-         "'encoded_frame_length': "
-         "('number', 'Encoded frame length (bytes)'),\n"
-         "'frame_type': ('string', 'Frame type'),\n"
-         "'packets_dropped': ('number', 'Packets dropped'),\n"
-         "'total_packets': ('number', 'Total packets'),\n"
-         "'ssim': ('number', 'SSIM'),\n"
-         "'psnr': ('number', 'PSNR (dB)'),\n"
-         "}\n");
+  Log(
+      "\nPython output (recommended to run with --noverbose to skip the "
+      "above output)\n");
+  printf(
+      "test_configuration = ["
+      "{'name': 'name',                      'value': '%s'},\n"
+      "{'name': 'description',               'value': '%s'},\n"
+      "{'name': 'test_number',               'value': '%d'},\n"
+      "{'name': 'input_filename',            'value': '%s'},\n"
+      "{'name': 'output_filename',           'value': '%s'},\n"
+      "{'name': 'output_dir',                'value': '%s'},\n"
+      "{'name': 'packet_size_in_bytes',      'value': '%" PRIuS
+      "'},\n"
+      "{'name': 'max_payload_size_in_bytes', 'value': '%" PRIuS
+      "'},\n"
+      "{'name': 'packet_loss_mode',          'value': '%s'},\n"
+      "{'name': 'packet_loss_probability',   'value': '%f'},\n"
+      "{'name': 'packet_loss_burst_length',  'value': '%d'},\n"
+      "{'name': 'exclude_frame_types',       'value': '%s'},\n"
+      "{'name': 'frame_length_in_bytes',     'value': '%" PRIuS
+      "'},\n"
+      "{'name': 'use_single_core',           'value': '%s'},\n"
+      "{'name': 'keyframe_interval;',        'value': '%d'},\n"
+      "{'name': 'video_codec_type',          'value': '%s'},\n"
+      "{'name': 'width',                     'value': '%d'},\n"
+      "{'name': 'height',                    'value': '%d'},\n"
+      "{'name': 'bit_rate_in_kbps',          'value': '%d'},\n"
+      "]\n",
+      config.name.c_str(), config.description.c_str(), config.test_number,
+      config.input_filename.c_str(), config.output_filename.c_str(),
+      config.output_dir.c_str(), config.networking_config.packet_size_in_bytes,
+      config.networking_config.max_payload_size_in_bytes,
+      PacketLossModeToStr(config.networking_config.packet_loss_mode),
+      config.networking_config.packet_loss_probability,
+      config.networking_config.packet_loss_burst_length,
+      ExcludeFrameTypesToStr(config.exclude_frame_types),
+      config.frame_length_in_bytes, config.use_single_core ? "True " : "False",
+      config.keyframe_interval,
+      webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
+      config.codec_settings->width, config.codec_settings->height,
+      config.codec_settings->startBitrate);
+  printf(
+      "frame_data_types = {"
+      "'frame_number': ('number', 'Frame number'),\n"
+      "'encoding_successful': ('boolean', 'Encoding successful?'),\n"
+      "'decoding_successful': ('boolean', 'Decoding successful?'),\n"
+      "'encode_time': ('number', 'Encode time (us)'),\n"
+      "'decode_time': ('number', 'Decode time (us)'),\n"
+      "'encode_return_code': ('number', 'Encode return code'),\n"
+      "'decode_return_code': ('number', 'Decode return code'),\n"
+      "'bit_rate': ('number', 'Bit rate (kbps)'),\n"
+      "'encoded_frame_length': "
+      "('number', 'Encoded frame length (bytes)'),\n"
+      "'frame_type': ('string', 'Frame type'),\n"
+      "'packets_dropped': ('number', 'Packets dropped'),\n"
+      "'total_packets': ('number', 'Total packets'),\n"
+      "'ssim': ('number', 'SSIM'),\n"
+      "'psnr': ('number', 'PSNR (dB)'),\n"
+      "}\n");
   printf("frame_data = [");
   for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
     const webrtc::test::FrameStatistic& f = stats.stats_[i];
     const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
     const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
-    printf("{'frame_number': %d, "
-           "'encoding_successful': %s, 'decoding_successful': %s, "
-           "'encode_time': %d, 'decode_time': %d, "
-           "'encode_return_code': %d, 'decode_return_code': %d, "
-           "'bit_rate': %d, 'encoded_frame_length': %" PRIuS ", "
-           "'frame_type': %s, 'packets_dropped': %d, "
-           "'total_packets': %" PRIuS ", 'ssim': %f, 'psnr': %f},\n",
-           f.frame_number,
-           f.encoding_successful ? "True " : "False",
-           f.decoding_successful ? "True " : "False",
-           f.encode_time_in_us,
-           f.decode_time_in_us,
-           f.encode_return_code,
-           f.decode_return_code,
-           f.bit_rate_in_kbps,
-           f.encoded_frame_length_in_bytes,
-           f.frame_type == webrtc::kDeltaFrame ? "'Delta'" : "'Other'",
-           f.packets_dropped,
-           f.total_packets,
-           ssim.value,
-           psnr.value);
+    printf(
+        "{'frame_number': %d, "
+        "'encoding_successful': %s, 'decoding_successful': %s, "
+        "'encode_time': %d, 'decode_time': %d, "
+        "'encode_return_code': %d, 'decode_return_code': %d, "
+        "'bit_rate': %d, 'encoded_frame_length': %" PRIuS
+        ", "
+        "'frame_type': %s, 'packets_dropped': %d, "
+        "'total_packets': %" PRIuS ", 'ssim': %f, 'psnr': %f},\n",
+        f.frame_number, f.encoding_successful ? "True " : "False",
+        f.decoding_successful ? "True " : "False", f.encode_time_in_us,
+        f.decode_time_in_us, f.encode_return_code, f.decode_return_code,
+        f.bit_rate_in_kbps, f.encoded_frame_length_in_bytes,
+        f.frame_type == webrtc::kVideoFrameDelta ? "'Delta'" : "'Other'",
+        f.packets_dropped, f.total_packets, ssim.value, psnr.value);
   }
   printf("]\n");
 }
@@ -438,10 +462,14 @@ void PrintPythonOutput(const webrtc::test::TestConfig& config,
 // The input file must be in YUV format.
 int main(int argc, char* argv[]) {
   std::string program_name = argv[0];
-  std::string usage = "Quality test application for video comparisons.\n"
-    "Run " + program_name + " --helpshort for usage.\n"
-    "Example usage:\n" + program_name +
-    " --input_filename=filename.yuv --width=352 --height=288\n";
+  std::string usage =
+      "Quality test application for video comparisons.\n"
+      "Run " +
+      program_name +
+      " --helpshort for usage.\n"
+      "Example usage:\n" +
+      program_name +
+      " --input_filename=filename.yuv --width=352 --height=288\n";
   google::SetUsageMessage(usage);
 
   google::ParseCommandLineFlags(&argc, &argv, true);
@@ -478,10 +506,8 @@ int main(int argc, char* argv[]) {
     packet_manipulator.InitializeRandomSeed(time(NULL));
   }
   webrtc::test::VideoProcessor* processor =
-      new webrtc::test::VideoProcessorImpl(encoder, decoder,
-                                           &frame_reader,
-                                           &frame_writer,
-                                           &packet_manipulator,
+      new webrtc::test::VideoProcessorImpl(encoder, decoder, &frame_reader,
+                                           &frame_writer, &packet_manipulator,
                                            config, &stats);
   processor->Init();
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
index da6008ba3d..9226fa774c 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
@@ -13,8 +13,8 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 
 #include "vpx/vpx_encoder.h"
@@ -41,7 +41,7 @@ int DefaultTemporalLayers::CurrentLayerId() const {
   int index = pattern_idx_ % temporal_ids_length_;
   assert(index >= 0);
   return temporal_ids_[index];
- }
+}
 
 bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
                                               int max_bitrate_kbit,
@@ -56,8 +56,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_periodicity = temporal_ids_length_;
       cfg->ts_target_bitrate[0] = bitrateKbit;
       cfg->ts_rate_decimator[0] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 1;
       temporal_pattern_[0] = kTemporalUpdateLastRefAll;
@@ -74,8 +73,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_target_bitrate[1] = bitrateKbit;
       cfg->ts_rate_decimator[0] = 2;
       cfg->ts_rate_decimator[1] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 8;
       temporal_pattern_[0] = kTemporalUpdateLastAndGoldenRefAltRef;
@@ -103,8 +101,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_rate_decimator[0] = 4;
       cfg->ts_rate_decimator[1] = 2;
       cfg->ts_rate_decimator[2] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 8;
       temporal_pattern_[0] = kTemporalUpdateLastAndGoldenRefAltRef;
@@ -138,8 +135,7 @@ bool DefaultTemporalLayers::ConfigureBitrates(int bitrateKbit,
       cfg->ts_rate_decimator[1] = 4;
       cfg->ts_rate_decimator[2] = 2;
       cfg->ts_rate_decimator[3] = 1;
-      memcpy(cfg->ts_layer_id,
-             temporal_ids_,
+      memcpy(cfg->ts_layer_id, temporal_ids_,
              sizeof(unsigned int) * temporal_ids_length_);
       temporal_pattern_length_ = 16;
       temporal_pattern_[0] = kTemporalUpdateLast;
@@ -243,7 +239,7 @@ int DefaultTemporalLayers::EncodeFlags(uint32_t timestamp) {
 
 void DefaultTemporalLayers::PopulateCodecSpecific(
     bool base_layer_sync,
-    CodecSpecificInfoVP8 *vp8_info,
+    CodecSpecificInfoVP8* vp8_info,
     uint32_t timestamp) {
   assert(number_of_temporal_layers_ > 0);
   assert(0 < temporal_ids_length_);
@@ -254,8 +250,8 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
     vp8_info->tl0PicIdx = kNoTl0PicIdx;
   } else {
     if (base_layer_sync) {
-    vp8_info->temporalIdx = 0;
-    vp8_info->layerSync = true;
+      vp8_info->temporalIdx = 0;
+      vp8_info->layerSync = true;
     } else {
       vp8_info->temporalIdx = CurrentLayerId();
       TemporalReferences temporal_reference =
@@ -267,7 +263,7 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
               kTemporalUpdateGoldenWithoutDependencyRefAltRef ||
           temporal_reference == kTemporalUpdateNoneNoRefGoldenRefAltRef ||
           (temporal_reference == kTemporalUpdateNone &&
-              number_of_temporal_layers_ == 4)) {
+           number_of_temporal_layers_ == 4)) {
         vp8_info->layerSync = true;
       } else {
         vp8_info->layerSync = false;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
index 61f281f2b1..19846ba5ff 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
@@ -24,20 +24,22 @@ class DefaultTemporalLayers : public TemporalLayers {
 
   // Returns the recommended VP8 encode flags needed. May refresh the decoder
   // and/or update the reference buffers.
-  virtual int EncodeFlags(uint32_t timestamp);
+  int EncodeFlags(uint32_t timestamp) override;
 
-  virtual bool ConfigureBitrates(int bitrate_kbit,
-                                 int max_bitrate_kbit,
-                                 int framerate,
-                                 vpx_codec_enc_cfg_t* cfg);
+  bool ConfigureBitrates(int bitrate_kbit,
+                         int max_bitrate_kbit,
+                         int framerate,
+                         vpx_codec_enc_cfg_t* cfg) override;
 
-  virtual void PopulateCodecSpecific(bool base_layer_sync,
-                                     CodecSpecificInfoVP8* vp8_info,
-                                     uint32_t timestamp);
+  void PopulateCodecSpecific(bool base_layer_sync,
+                             CodecSpecificInfoVP8* vp8_info,
+                             uint32_t timestamp) override;
 
-  virtual void FrameEncoded(unsigned int size, uint32_t timestamp) {}
+  void FrameEncoded(unsigned int size, uint32_t timestamp, int qp) override {}
 
-  virtual int CurrentLayerId() const;
+  bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) override { return false; }
+
+  int CurrentLayerId() const override;
 
  private:
   enum TemporalReferences {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
index 34121cbcf6..461ba69a72 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
@@ -8,9 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h"
 
 #include "vpx/vpx_encoder.h"
@@ -19,47 +18,36 @@
 namespace webrtc {
 
 enum {
-  kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF |
-                        VP8_EFLAG_NO_UPD_ARF |
+  kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
                         VP8_EFLAG_NO_REF_GF |
                         VP8_EFLAG_NO_REF_ARF,
-  kTemporalUpdateGoldenWithoutDependency = VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_UPD_ARF |
-                                           VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateGolden = VP8_EFLAG_NO_REF_ARF |
-                          VP8_EFLAG_NO_UPD_ARF |
-                          VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateAltrefWithoutDependency = VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF |
-                          VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF |
-                        VP8_EFLAG_NO_UPD_ARF |
+  kTemporalUpdateGoldenWithoutDependency =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF |
+      VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateGolden =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateAltrefWithoutDependency =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
+      VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
                         VP8_EFLAG_NO_UPD_LAST |
                         VP8_EFLAG_NO_UPD_ENTROPY,
-  kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF |
-                                   VP8_EFLAG_NO_UPD_GF |
+  kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF |
                                    VP8_EFLAG_NO_UPD_ARF |
                                    VP8_EFLAG_NO_UPD_LAST |
                                    VP8_EFLAG_NO_UPD_ENTROPY,
-  kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF |
-                                   VP8_EFLAG_NO_UPD_GF |
+  kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
                                    VP8_EFLAG_NO_UPD_ARF |
                                    VP8_EFLAG_NO_UPD_LAST |
                                    VP8_EFLAG_NO_UPD_ENTROPY,
-  kTemporalUpdateGoldenWithoutDependencyRefAltRef = VP8_EFLAG_NO_REF_GF |
-                                                    VP8_EFLAG_NO_UPD_ARF |
-                                                    VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF |
-                                   VP8_EFLAG_NO_UPD_LAST,
-  kTemporalUpdateLastRefAltRef = VP8_EFLAG_NO_UPD_GF |
-                                 VP8_EFLAG_NO_UPD_ARF |
-                                 VP8_EFLAG_NO_REF_GF,
-  kTemporalUpdateLastAndGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF |
-                                          VP8_EFLAG_NO_REF_GF,
+  kTemporalUpdateGoldenWithoutDependencyRefAltRef =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateLastRefAltRef =
+      VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
+  kTemporalUpdateLastAndGoldenRefAltRef =
+      VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
 };
 
 TEST(TemporalLayersTest, 2Layers) {
@@ -68,29 +56,30 @@ TEST(TemporalLayersTest, 2Layers) {
   CodecSpecificInfoVP8 vp8_info;
   tl.ConfigureBitrates(500, 500, 30, &cfg);
 
-  int expected_flags[16] = { kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-   };
-  int expected_temporal_idx[16] =
-      { 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1 };
+  int expected_flags[16] = {
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+  };
+  int expected_temporal_idx[16] = {0, 1, 0, 1, 0, 1, 0, 1,
+                                   0, 1, 0, 1, 0, 1, 0, 1};
 
-  bool expected_layer_sync[16] =
-      { false, true, false, false, false, false, false, false,
-        false, true, false, false, false, false, false, false };
+  bool expected_layer_sync[16] = {false, true,  false, false, false, false,
+                                  false, false, false, true,  false, false,
+                                  false, false, false, false};
 
   uint32_t timestamp = 0;
   for (int i = 0; i < 16; ++i) {
@@ -108,29 +97,30 @@ TEST(TemporalLayersTest, 3Layers) {
   CodecSpecificInfoVP8 vp8_info;
   tl.ConfigureBitrates(500, 500, 30, &cfg);
 
-  int expected_flags[16] = { kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateNoneNoRefGolden,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastAndGoldenRefAltRef,
-                             kTemporalUpdateNoneNoRefGolden,
-                             kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateLastRefAltRef,
-                             kTemporalUpdateNone,
-                             kTemporalUpdateGoldenRefAltRef,
-                             kTemporalUpdateNone,
+  int expected_flags[16] = {
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateNoneNoRefGolden,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastAndGoldenRefAltRef,
+      kTemporalUpdateNoneNoRefGolden,
+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNone,
+      kTemporalUpdateGoldenRefAltRef,
+      kTemporalUpdateNone,
   };
-  int expected_temporal_idx[16] =
-      { 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2 };
+  int expected_temporal_idx[16] = {0, 2, 1, 2, 0, 2, 1, 2,
+                                   0, 2, 1, 2, 0, 2, 1, 2};
 
-  bool expected_layer_sync[16] =
-      { false, true, true, false, false, false, false, false,
-        false, true, true, false, false, false, false, false };
+  bool expected_layer_sync[16] = {false, true,  true,  false, false, false,
+                                  false, false, false, true,  true,  false,
+                                  false, false, false, false};
 
   unsigned int timestamp = 0;
   for (int i = 0; i < 16; ++i) {
@@ -165,12 +155,12 @@ TEST(TemporalLayersTest, 4Layers) {
       kTemporalUpdateAltref,
       kTemporalUpdateNone,
   };
-  int expected_temporal_idx[16] =
-      { 0, 3, 2, 3, 1, 3, 2, 3, 0, 3, 2, 3, 1, 3, 2, 3 };
+  int expected_temporal_idx[16] = {0, 3, 2, 3, 1, 3, 2, 3,
+                                   0, 3, 2, 3, 1, 3, 2, 3};
 
-  bool expected_layer_sync[16] =
-      { false, true, true, true, true, true, false, true,
-        false, true, false, true, false, true, false, true };
+  bool expected_layer_sync[16] = {false, true, true,  true, true,  true,
+                                  false, true, false, true, false, true,
+                                  false, true, false, true};
 
   uint32_t timestamp = 0;
   for (int i = 0; i < 16; ++i) {
@@ -198,8 +188,7 @@ TEST(TemporalLayersTest, KeyFrame) {
       kTemporalUpdateGoldenRefAltRef,
       kTemporalUpdateNone,
   };
-  int expected_temporal_idx[8] =
-      { 0, 0, 0, 0, 0, 0, 0, 2};
+  int expected_temporal_idx[8] = {0, 0, 0, 0, 0, 0, 0, 2};
 
   uint32_t timestamp = 0;
   for (int i = 0; i < 7; ++i) {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
index f5dae471d2..dd3514235d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
@@ -13,7 +13,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
@@ -21,16 +21,15 @@ class VP8Encoder : public VideoEncoder {
  public:
   static VP8Encoder* Create();
 
-  virtual ~VP8Encoder() {};
+  virtual ~VP8Encoder() {}
 };  // end of VP8Encoder class
 
-
 class VP8Decoder : public VideoDecoder {
  public:
   static VP8Decoder* Create();
 
-  virtual ~VP8Decoder() {};
+  virtual ~VP8Decoder() {}
 };  // end of VP8Decoder class
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
index c2cefdd94e..7a27e4429a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
 
 #include "webrtc/common_types.h"
 
@@ -19,11 +19,11 @@ namespace webrtc {
 // Values as required for the VP8 codec (accumulating).
 static const float
     kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
-      {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
-      {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
-      {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
-      {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
+        {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
+        {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
+        {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
+        {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
index f16c756813..d22601358f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
@@ -12,7 +12,7 @@
 
 #include "vpx/vpx_encoder.h"
 #include "vpx/vp8cx.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
 
@@ -23,7 +23,8 @@ namespace webrtc {
 namespace {
 enum {
   kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-                        VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF,
+                        VP8_EFLAG_NO_REF_GF |
+                        VP8_EFLAG_NO_REF_ARF,
 
   kTemporalUpdateGolden =
       VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
@@ -37,13 +38,15 @@ enum {
       kTemporalUpdateAltref | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF,
 
   kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-                        VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+                        VP8_EFLAG_NO_UPD_LAST |
+                        VP8_EFLAG_NO_UPD_ENTROPY,
 
   kTemporalUpdateNoneNoRefAltref = kTemporalUpdateNone | VP8_EFLAG_NO_REF_ARF,
 
   kTemporalUpdateNoneNoRefGoldenRefAltRef =
       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+      VP8_EFLAG_NO_UPD_LAST |
+      VP8_EFLAG_NO_UPD_ENTROPY,
 
   kTemporalUpdateGoldenWithoutDependencyRefAltRef =
       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
@@ -133,12 +136,14 @@ class RealTimeTemporalLayers : public TemporalLayers {
         layer_ids_length_ = sizeof(layer_ids) / sizeof(*layer_ids);
 
         static const int encode_flags[] = {
-          kTemporalUpdateLastAndGoldenRefAltRef,
-          kTemporalUpdateGoldenWithoutDependencyRefAltRef,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateGoldenRefAltRef,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateGoldenRefAltRef,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateNone
-        };
+            kTemporalUpdateLastAndGoldenRefAltRef,
+            kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateGoldenRefAltRef,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateGoldenRefAltRef,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateNone};
         encode_flags_length_ = sizeof(encode_flags) / sizeof(*layer_ids);
         encode_flags_ = encode_flags;
 
@@ -153,12 +158,14 @@ class RealTimeTemporalLayers : public TemporalLayers {
         layer_ids_length_ = sizeof(layer_ids) / sizeof(*layer_ids);
 
         static const int encode_flags[] = {
-          kTemporalUpdateLastAndGoldenRefAltRef,
-          kTemporalUpdateNoneNoRefGoldenRefAltRef,
-          kTemporalUpdateGoldenWithoutDependencyRefAltRef, kTemporalUpdateNone,
-          kTemporalUpdateLastRefAltRef, kTemporalUpdateNone,
-          kTemporalUpdateGoldenRefAltRef, kTemporalUpdateNone
-        };
+            kTemporalUpdateLastAndGoldenRefAltRef,
+            kTemporalUpdateNoneNoRefGoldenRefAltRef,
+            kTemporalUpdateGoldenWithoutDependencyRefAltRef,
+            kTemporalUpdateNone,
+            kTemporalUpdateLastRefAltRef,
+            kTemporalUpdateNone,
+            kTemporalUpdateGoldenRefAltRef,
+            kTemporalUpdateNone};
         encode_flags_length_ = sizeof(encode_flags) / sizeof(*layer_ids);
         encode_flags_ = encode_flags;
 
@@ -172,8 +179,8 @@ class RealTimeTemporalLayers : public TemporalLayers {
         assert(false);
         return false;
     }
-    memcpy(
-        cfg->ts_layer_id, layer_ids_, sizeof(unsigned int) * layer_ids_length_);
+    memcpy(cfg->ts_layer_id, layer_ids_,
+           sizeof(unsigned int) * layer_ids_length_);
     return true;
   }
 
@@ -239,7 +246,9 @@ class RealTimeTemporalLayers : public TemporalLayers {
     }
   }
 
-  void FrameEncoded(unsigned int size, uint32_t timestamp) {}
+  void FrameEncoded(unsigned int size, uint32_t timestamp, int qp) override {}
+
+  bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) override { return false; }
 
  private:
   int temporal_layers_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc
index a922e35712..1838e32eb7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc
@@ -25,8 +25,7 @@ ReferencePictureSelection::ReferencePictureSelection()
       last_sent_ref_update_time_(0),
       established_ref_picture_id_(0),
       last_refresh_time_(0),
-      rtt_(0) {
-}
+      rtt_(0) {}
 
 void ReferencePictureSelection::Init() {
   update_golden_next_ = true;
@@ -62,7 +61,8 @@ bool ReferencePictureSelection::ReceivedSLI(uint32_t now_ts) {
   return send_refresh;
 }
 
-int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
+int ReferencePictureSelection::EncodeFlags(int picture_id,
+                                           bool send_refresh,
                                            uint32_t now_ts) {
   int flags = 0;
   // We can't refresh the decoder until we have established the key frame.
@@ -87,12 +87,12 @@ int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
       received_ack_) {
     flags |= VP8_EFLAG_NO_REF_LAST;  // Don't reference the last frame.
     if (update_golden_next_) {
-      flags |= VP8_EFLAG_FORCE_GF;  // Update the golden reference.
+      flags |= VP8_EFLAG_FORCE_GF;    // Update the golden reference.
       flags |= VP8_EFLAG_NO_UPD_ARF;  // Don't update alt-ref.
-      flags |= VP8_EFLAG_NO_REF_GF;  // Don't reference the golden frame.
+      flags |= VP8_EFLAG_NO_REF_GF;   // Don't reference the golden frame.
     } else {
-      flags |= VP8_EFLAG_FORCE_ARF;  // Update the alt-ref reference.
-      flags |= VP8_EFLAG_NO_UPD_GF;  // Don't update the golden frame.
+      flags |= VP8_EFLAG_FORCE_ARF;   // Update the alt-ref reference.
+      flags |= VP8_EFLAG_NO_UPD_GF;   // Don't update the golden frame.
       flags |= VP8_EFLAG_NO_REF_ARF;  // Don't reference the alt-ref frame.
     }
     last_sent_ref_picture_id_ = picture_id;
@@ -103,9 +103,9 @@ int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
     if (established_golden_)
       flags |= VP8_EFLAG_NO_REF_ARF;  // Don't reference the alt-ref frame.
     else
-      flags |= VP8_EFLAG_NO_REF_GF;   // Don't reference the golden frame.
-    flags |= VP8_EFLAG_NO_UPD_GF;  // Don't update the golden frame.
-    flags |= VP8_EFLAG_NO_UPD_ARF;  // Don't update the alt-ref frame.
+      flags |= VP8_EFLAG_NO_REF_GF;  // Don't reference the golden frame.
+    flags |= VP8_EFLAG_NO_UPD_GF;    // Don't update the golden frame.
+    flags |= VP8_EFLAG_NO_UPD_ARF;   // Don't update the alt-ref frame.
   }
   return flags;
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
index c6474e5bd1..742bb96e91 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection_unittest.cc
@@ -22,25 +22,19 @@ static const uint32_t kMinUpdateInterval = 10;
 // Should match the values set in reference_picture_selection.h
 static const int kRtt = 10;
 
-static const int kNoPropagationGolden    = VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_UPD_ARF;
-static const int kNoPropagationAltRef    = VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_UPD_ARF;
-static const int kPropagateGolden        = VP8_EFLAG_FORCE_GF |
-                                           VP8_EFLAG_NO_UPD_ARF |
-                                           VP8_EFLAG_NO_REF_GF |
-                                           VP8_EFLAG_NO_REF_LAST;
-static const int kPropagateAltRef        = VP8_EFLAG_FORCE_ARF |
-                                           VP8_EFLAG_NO_UPD_GF |
-                                           VP8_EFLAG_NO_REF_ARF |
-                                           VP8_EFLAG_NO_REF_LAST;
-static const int kRefreshFromGolden      = VP8_EFLAG_NO_REF_LAST |
-                                           VP8_EFLAG_NO_REF_ARF;
-static const int kRefreshFromAltRef      = VP8_EFLAG_NO_REF_LAST |
-                                           VP8_EFLAG_NO_REF_GF;
-
+static const int kNoPropagationGolden =
+    VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF;
+static const int kNoPropagationAltRef =
+    VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF;
+static const int kPropagateGolden = VP8_EFLAG_FORCE_GF | VP8_EFLAG_NO_UPD_ARF |
+                                    VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_LAST;
+static const int kPropagateAltRef = VP8_EFLAG_FORCE_ARF | VP8_EFLAG_NO_UPD_GF |
+                                    VP8_EFLAG_NO_REF_ARF |
+                                    VP8_EFLAG_NO_REF_LAST;
+static const int kRefreshFromGolden =
+    VP8_EFLAG_NO_REF_LAST | VP8_EFLAG_NO_REF_ARF;
+static const int kRefreshFromAltRef =
+    VP8_EFLAG_NO_REF_LAST | VP8_EFLAG_NO_REF_GF;
 
 class TestRPS : public ::testing::Test {
  protected:
@@ -84,15 +78,15 @@ TEST_F(TestRPS, TestDecoderRefresh) {
   EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
   // Enough time have elapsed since the previous reference propagation, we will
   // therefore get both a refresh from golden and a propagation of alt-ref.
-  EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time), kRefreshFromGolden |
-            kPropagateAltRef);
+  EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time),
+            kRefreshFromGolden | kPropagateAltRef);
   rps_.ReceivedRPSI(5);
   time += kRtt + 1;
   // Enough time for a new refresh, but not enough time for a reference
   // propagation.
   EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
-  EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time), kRefreshFromAltRef |
-            kNoPropagationAltRef);
+  EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time),
+            kRefreshFromAltRef | kNoPropagationAltRef);
 }
 
 TEST_F(TestRPS, TestWrap) {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
index 63ef227812..536587a13e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
@@ -11,32 +11,54 @@
 
 #include <stdlib.h>
 
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
 #include "vpx/vpx_encoder.h"
 #include "vpx/vp8cx.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
-enum { kOneSecond90Khz = 90000 };
+static const int kOneSecond90Khz = 90000;
+static const int kMinTimeBetweenSyncs = kOneSecond90Khz * 5;
+static const int kMaxTimeBetweenSyncs = kOneSecond90Khz * 10;
+static const int kQpDeltaThresholdForSync = 8;
 
 const double ScreenshareLayers::kMaxTL0FpsReduction = 2.5;
 const double ScreenshareLayers::kAcceptableTargetOvershoot = 2.0;
 
+// Since this is TL0 we only allow updating and predicting from the LAST
+// reference frame.
+const int ScreenshareLayers::kTl0Flags =
+    VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF |
+    VP8_EFLAG_NO_REF_ARF;
+
+// Allow predicting from both TL0 and TL1.
+const int ScreenshareLayers::kTl1Flags =
+    VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST;
+
+// Allow predicting from only TL0 to allow participants to switch to the high
+// bitrate stream. This means predicting only from the LAST reference frame, but
+// only updating GF to not corrupt TL0.
+const int ScreenshareLayers::kTl1SyncFlags =
+    VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF |
+    VP8_EFLAG_NO_UPD_LAST;
+
 ScreenshareLayers::ScreenshareLayers(int num_temporal_layers,
-                                     uint8_t initial_tl0_pic_idx,
-                                     FrameDropper* tl0_frame_dropper,
-                                     FrameDropper* tl1_frame_dropper)
-    : tl0_frame_dropper_(tl0_frame_dropper),
-      tl1_frame_dropper_(tl1_frame_dropper),
-      number_of_temporal_layers_(num_temporal_layers),
+                                     uint8_t initial_tl0_pic_idx)
+    : number_of_temporal_layers_(num_temporal_layers),
       last_base_layer_sync_(false),
       tl0_pic_idx_(initial_tl0_pic_idx),
-      active_layer_(0),
-      framerate_(5),
-      last_sync_timestamp_(-1) {
+      active_layer_(-1),
+      last_timestamp_(-1),
+      last_sync_timestamp_(-1),
+      min_qp_(-1),
+      max_qp_(-1),
+      max_debt_bytes_(0),
+      frame_rate_(-1) {
   assert(num_temporal_layers > 0);
   assert(num_temporal_layers <= 2);
-  assert(tl0_frame_dropper && tl1_frame_dropper);
 }
 
 int ScreenshareLayers::CurrentLayerId() const {
@@ -49,84 +71,128 @@ int ScreenshareLayers::EncodeFlags(uint32_t timestamp) {
     // No flags needed for 1 layer screenshare.
     return 0;
   }
-  CalculateFramerate(timestamp);
+
+  int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
   int flags = 0;
-  // Note that ARF on purpose isn't used in this scheme since it is allocated
-  // for the last key frame to make key frame caching possible.
-  if (tl0_frame_dropper_->DropFrame()) {
-    // Must drop TL0, encode TL1 instead.
-    if (tl1_frame_dropper_->DropFrame()) {
-      // Must drop both TL0 and TL1.
-      flags = -1;
-    } else {
-      active_layer_ = 1;
-      if (TimeToSync(timestamp)) {
-        last_sync_timestamp_ = timestamp;
-        // Allow predicting from only TL0 to allow participants to switch to the
-        // high bitrate stream. This means predicting only from the LAST
-        // reference frame, but only updating GF to not corrupt TL0.
-        flags = VP8_EFLAG_NO_REF_ARF;
-        flags |= VP8_EFLAG_NO_REF_GF;
-        flags |= VP8_EFLAG_NO_UPD_ARF;
-        flags |= VP8_EFLAG_NO_UPD_LAST;
+
+  if (active_layer_ == -1 ||
+      layers_[active_layer_].state != TemporalLayer::State::kDropped) {
+    if (layers_[0].debt_bytes_ > max_debt_bytes_) {
+      // Must drop TL0, encode TL1 instead.
+      if (layers_[1].debt_bytes_ > max_debt_bytes_) {
+        // Must drop both TL0 and TL1.
+        active_layer_ = -1;
       } else {
-        // Allow predicting from both TL0 and TL1.
-        flags = VP8_EFLAG_NO_REF_ARF;
-        flags |= VP8_EFLAG_NO_UPD_ARF;
-        flags |= VP8_EFLAG_NO_UPD_LAST;
+        active_layer_ = 1;
       }
+    } else {
+      active_layer_ = 0;
     }
-  } else {
-    active_layer_ = 0;
-    // Since this is TL0 we only allow updating and predicting from the LAST
-    // reference frame.
-    flags = VP8_EFLAG_NO_UPD_GF;
-    flags |= VP8_EFLAG_NO_UPD_ARF;
-    flags |= VP8_EFLAG_NO_REF_GF;
-    flags |= VP8_EFLAG_NO_REF_ARF;
   }
+
+  switch (active_layer_) {
+    case 0:
+      flags = kTl0Flags;
+      break;
+    case 1:
+      if (TimeToSync(unwrapped_timestamp)) {
+        last_sync_timestamp_ = unwrapped_timestamp;
+        flags = kTl1SyncFlags;
+      } else {
+        flags = kTl1Flags;
+      }
+      break;
+    case -1:
+      flags = -1;
+      break;
+    default:
+      flags = -1;
+      RTC_NOTREACHED();
+  }
+
   // Make sure both frame droppers leak out bits.
-  tl0_frame_dropper_->Leak(framerate_);
-  tl1_frame_dropper_->Leak(framerate_);
+  int64_t ts_diff;
+  if (last_timestamp_ == -1) {
+    ts_diff = kOneSecond90Khz / (frame_rate_ <= 0 ? 5 : frame_rate_);
+  } else {
+    ts_diff = unwrapped_timestamp - last_timestamp_;
+  }
+
+  layers_[0].UpdateDebt(ts_diff / 90);
+  layers_[1].UpdateDebt(ts_diff / 90);
+  last_timestamp_ = timestamp;
   return flags;
 }
 
-bool ScreenshareLayers::ConfigureBitrates(int bitrate_kbit,
-                                          int max_bitrate_kbit,
+bool ScreenshareLayers::ConfigureBitrates(int bitrate_kbps,
+                                          int max_bitrate_kbps,
                                           int framerate,
                                           vpx_codec_enc_cfg_t* cfg) {
-  if (framerate > 0)
-    framerate_ = framerate;
+  layers_[0].target_rate_kbps_ = bitrate_kbps;
+  layers_[1].target_rate_kbps_ = max_bitrate_kbps;
 
-  tl0_frame_dropper_->SetRates(bitrate_kbit, framerate_);
-  tl1_frame_dropper_->SetRates(max_bitrate_kbit, framerate_);
+  int target_bitrate_kbps = bitrate_kbps;
 
   if (cfg != nullptr) {
-    // Calculate a codec target bitrate. This may be higher than TL0, gaining
-    // quality at the expense of frame rate at TL0. Constraints:
-    // - TL0 frame rate should not be less than framerate / kMaxTL0FpsReduction.
-    // - Target rate * kAcceptableTargetOvershoot should not exceed TL1 rate.
-    double target_bitrate =
-        std::min(bitrate_kbit * kMaxTL0FpsReduction,
-                 max_bitrate_kbit / kAcceptableTargetOvershoot);
-    cfg->rc_target_bitrate =
-        std::max(static_cast<unsigned int>(bitrate_kbit),
-                 static_cast<unsigned int>(target_bitrate + 0.5));
+    if (number_of_temporal_layers_ > 1) {
+      // Calculate a codec target bitrate. This may be higher than TL0, gaining
+      // quality at the expense of frame rate at TL0. Constraints:
+      // - TL0 frame rate no less than framerate / kMaxTL0FpsReduction.
+      // - Target rate * kAcceptableTargetOvershoot should not exceed TL1 rate.
+      target_bitrate_kbps =
+          std::min(bitrate_kbps * kMaxTL0FpsReduction,
+                   max_bitrate_kbps / kAcceptableTargetOvershoot);
+
+      cfg->rc_target_bitrate = std::max(bitrate_kbps, target_bitrate_kbps);
+    }
+
+    // Don't reconfigure qp limits during quality boost frames.
+    if (layers_[active_layer_].state != TemporalLayer::State::kQualityBoost) {
+      min_qp_ = cfg->rc_min_quantizer;
+      max_qp_ = cfg->rc_max_quantizer;
+      // After a dropped frame, a frame with max qp will be encoded and the
+      // quality will then ramp up from there. To boost the speed of recovery,
+      // encode the next frame with lower max qp. TL0 is the most important to
+      // improve since the errors in this layer will propagate to TL1.
+      // Currently, reduce max qp by 20% for TL0 and 15% for TL1.
+      layers_[0].enhanced_max_qp = min_qp_ + (((max_qp_ - min_qp_) * 80) / 100);
+      layers_[1].enhanced_max_qp = min_qp_ + (((max_qp_ - min_qp_) * 85) / 100);
+    }
   }
 
+  int avg_frame_size = (target_bitrate_kbps * 1000) / (8 * framerate);
+  max_debt_bytes_ = 4 * avg_frame_size;
+
   return true;
 }
 
-void ScreenshareLayers::FrameEncoded(unsigned int size, uint32_t timestamp) {
-  if (active_layer_ == 0) {
-    tl0_frame_dropper_->Fill(size, true);
+void ScreenshareLayers::FrameEncoded(unsigned int size,
+                                     uint32_t timestamp,
+                                     int qp) {
+  if (size == 0) {
+    layers_[active_layer_].state = TemporalLayer::State::kDropped;
+    return;
+  }
+
+  if (layers_[active_layer_].state == TemporalLayer::State::kDropped) {
+    layers_[active_layer_].state = TemporalLayer::State::kQualityBoost;
+  }
+
+  if (qp != -1)
+    layers_[active_layer_].last_qp = qp;
+
+  if (active_layer_ == 0) {
+    layers_[0].debt_bytes_ += size;
+    layers_[1].debt_bytes_ += size;
+  } else if (active_layer_ == 1) {
+    layers_[1].debt_bytes_ += size;
   }
-  tl1_frame_dropper_->Fill(size, true);
 }
 
 void ScreenshareLayers::PopulateCodecSpecific(bool base_layer_sync,
-                                              CodecSpecificInfoVP8 *vp8_info,
+                                              CodecSpecificInfoVP8* vp8_info,
                                               uint32_t timestamp) {
+  int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
   if (number_of_temporal_layers_ == 1) {
     vp8_info->temporalIdx = kNoTemporalIdx;
     vp8_info->layerSync = false;
@@ -135,13 +201,14 @@ void ScreenshareLayers::PopulateCodecSpecific(bool base_layer_sync,
     vp8_info->temporalIdx = active_layer_;
     if (base_layer_sync) {
       vp8_info->temporalIdx = 0;
-      last_sync_timestamp_ = timestamp;
+      last_sync_timestamp_ = unwrapped_timestamp;
     } else if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
       // Regardless of pattern the frame after a base layer sync will always
       // be a layer sync.
-      last_sync_timestamp_ = timestamp;
+      last_sync_timestamp_ = unwrapped_timestamp;
     }
-    vp8_info->layerSync = (last_sync_timestamp_ == timestamp);
+    vp8_info->layerSync = last_sync_timestamp_ != -1 &&
+                          last_sync_timestamp_ == unwrapped_timestamp;
     if (vp8_info->temporalIdx == 0) {
       tl0_pic_idx_++;
     }
@@ -150,27 +217,65 @@ void ScreenshareLayers::PopulateCodecSpecific(bool base_layer_sync,
   }
 }
 
-bool ScreenshareLayers::TimeToSync(uint32_t timestamp) const {
-  const uint32_t timestamp_diff = timestamp - last_sync_timestamp_;
-  return last_sync_timestamp_ < 0 || timestamp_diff > kOneSecond90Khz;
+bool ScreenshareLayers::TimeToSync(int64_t timestamp) const {
+  if (active_layer_ != 1) {
+    RTC_NOTREACHED();
+    return false;
+  }
+  RTC_DCHECK_NE(-1, layers_[0].last_qp);
+  if (layers_[1].last_qp == -1) {
+    // First frame in TL1 should only depend on TL0 since there are no
+    // previous frames in TL1.
+    return true;
+  }
+
+  RTC_DCHECK_NE(-1, last_sync_timestamp_);
+  int64_t timestamp_diff = timestamp - last_sync_timestamp_;
+  if (timestamp_diff > kMaxTimeBetweenSyncs) {
+    // After a certain time, force a sync frame.
+    return true;
+  } else if (timestamp_diff < kMinTimeBetweenSyncs) {
+    // If too soon from previous sync frame, don't issue a new one.
+    return false;
+  }
+  // Issue a sync frame if difference in quality between TL0 and TL1 isn't too
+  // large.
+  if (layers_[0].last_qp - layers_[1].last_qp < kQpDeltaThresholdForSync)
+    return true;
+  return false;
 }
 
-void ScreenshareLayers::CalculateFramerate(uint32_t timestamp) {
-  timestamp_list_.push_front(timestamp);
-  // Remove timestamps older than 1 second from the list.
-  uint32_t timestamp_diff = timestamp - timestamp_list_.back();
-  while (timestamp_diff > kOneSecond90Khz) {
-    timestamp_list_.pop_back();
-    timestamp_diff = timestamp - timestamp_list_.back();
+bool ScreenshareLayers::UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) {
+  if (max_qp_ == -1 || number_of_temporal_layers_ <= 1)
+    return false;
+
+  // If layer is in the quality boost state (following a dropped frame), update
+  // the configuration with the adjusted (lower) qp and set the state back to
+  // normal.
+  unsigned int adjusted_max_qp;
+  if (layers_[active_layer_].state == TemporalLayer::State::kQualityBoost &&
+      layers_[active_layer_].enhanced_max_qp != -1) {
+    adjusted_max_qp = layers_[active_layer_].enhanced_max_qp;
+    layers_[active_layer_].state = TemporalLayer::State::kNormal;
+  } else {
+    if (max_qp_ == -1)
+      return false;
+    adjusted_max_qp = max_qp_;  // Set the normal max qp.
   }
-  // If we have encoded frames within the last second, that number of frames
-  // is a reasonable first estimate of the framerate.
-  framerate_ = timestamp_list_.size();
-  if (timestamp_diff > 0) {
-    // Estimate the framerate by dividing the number of timestamp diffs with
-    // the sum of the timestamp diffs (with rounding).
-    framerate_ = (kOneSecond90Khz * (timestamp_list_.size() - 1) +
-        timestamp_diff / 2) / timestamp_diff;
+
+  if (adjusted_max_qp == cfg->rc_max_quantizer)
+    return false;
+
+  cfg->rc_max_quantizer = adjusted_max_qp;
+  return true;
+}
+
+void ScreenshareLayers::TemporalLayer::UpdateDebt(int64_t delta_ms) {
+  uint32_t debt_reduction_bytes = target_rate_kbps_ * delta_ms / 8;
+  if (debt_reduction_bytes >= debt_bytes_) {
+    debt_bytes_ = 0;
+  } else {
+    debt_bytes_ -= debt_reduction_bytes;
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
index 0bc571ee0f..7628758209 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
@@ -13,8 +13,9 @@
 
 #include "vpx/vpx_encoder.h"
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -25,43 +26,73 @@ class ScreenshareLayers : public TemporalLayers {
  public:
   static const double kMaxTL0FpsReduction;
   static const double kAcceptableTargetOvershoot;
+  static const int kTl0Flags;
+  static const int kTl1Flags;
+  static const int kTl1SyncFlags;
 
-  ScreenshareLayers(int num_temporal_layers,
-                    uint8_t initial_tl0_pic_idx,
-                    FrameDropper* tl0_frame_dropper,
-                    FrameDropper* tl1_frame_dropper);
+  ScreenshareLayers(int num_temporal_layers, uint8_t initial_tl0_pic_idx);
   virtual ~ScreenshareLayers() {}
 
   // Returns the recommended VP8 encode flags needed. May refresh the decoder
   // and/or update the reference buffers.
-  virtual int EncodeFlags(uint32_t timestamp);
+  int EncodeFlags(uint32_t timestamp) override;
 
-  virtual bool ConfigureBitrates(int bitrate_kbit,
-                                 int max_bitrate_kbit,
-                                 int framerate,
-                                 vpx_codec_enc_cfg_t* cfg);
+  bool ConfigureBitrates(int bitrate_kbps,
+                         int max_bitrate_kbps,
+                         int framerate,
+                         vpx_codec_enc_cfg_t* cfg) override;
 
-  virtual void PopulateCodecSpecific(bool base_layer_sync,
-                                     CodecSpecificInfoVP8 *vp8_info,
-                                     uint32_t timestamp);
+  void PopulateCodecSpecific(bool base_layer_sync,
+                             CodecSpecificInfoVP8* vp8_info,
+                             uint32_t timestamp) override;
 
-  virtual void FrameEncoded(unsigned int size, uint32_t timestamp);
+  void FrameEncoded(unsigned int size, uint32_t timestamp, int qp) override;
 
-  virtual int CurrentLayerId() const;
+  int CurrentLayerId() const override;
+
+  // Allows the layers adapter to update the encoder configuration prior to a
+  // frame being encoded. Return true if the configuration should be updated
+  // and false if now change is needed.
+  bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) override;
 
  private:
-  void CalculateFramerate(uint32_t timestamp);
-  bool TimeToSync(uint32_t timestamp) const;
+  bool TimeToSync(int64_t timestamp) const;
 
-  FrameDropper* tl0_frame_dropper_;
-  FrameDropper* tl1_frame_dropper_;
   int number_of_temporal_layers_;
   bool last_base_layer_sync_;
   uint8_t tl0_pic_idx_;
   int active_layer_;
-  std::list<uint32_t> timestamp_list_;
-  int framerate_;
+  int64_t last_timestamp_;
   int64_t last_sync_timestamp_;
+  rtc::TimestampWrapAroundHandler time_wrap_handler_;
+  int min_qp_;
+  int max_qp_;
+  uint32_t max_debt_bytes_;
+  int frame_rate_;
+
+  static const int kMaxNumTemporalLayers = 2;
+  struct TemporalLayer {
+    TemporalLayer()
+        : state(State::kNormal),
+          enhanced_max_qp(-1),
+          last_qp(-1),
+          debt_bytes_(0),
+          target_rate_kbps_(0) {}
+
+    enum class State {
+      kNormal,
+      kDropped,
+      kReencoded,
+      kQualityBoost,
+    } state;
+
+    int enhanced_max_qp;
+    int last_qp;
+    uint32_t debt_bytes_;
+    uint32_t target_rate_kbps_;
+
+    void UpdateDebt(int64_t delta_ms);
+  } layers_[kMaxNumTemporalLayers];
 };
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
index e12f9ce088..f31ed5e4d8 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
@@ -12,9 +12,9 @@
 #include "vpx/vpx_encoder.h"
 #include "vpx/vp8cx.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
-#include "webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h"
 
 using ::testing::_;
 using ::testing::NiceMock;
@@ -22,62 +22,19 @@ using ::testing::Return;
 
 namespace webrtc {
 
-enum { kTimestampDelta5Fps = 90000 / 5 };  // 5 frames per second at 90 kHz.
-enum { kTimestampDelta30Fps = 90000 / 30 };  // 30 frames per second at 90 kHz.
-enum { kFrameSize = 2500 };
-
-const int kFlagsTL0 = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
-    VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF;
-const int kFlagsTL1 = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF |
-    VP8_EFLAG_NO_UPD_LAST;
-const int kFlagsTL1Sync = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF |
-    VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST;
-
-class ScreenshareLayersFT : public ScreenshareLayers {
- public:
-  ScreenshareLayersFT(int num_temporal_layers,
-                      uint8_t initial_tl0_pic_idx,
-                      FrameDropper* tl0_frame_dropper,
-                      FrameDropper* tl1_frame_dropper)
-      : ScreenshareLayers(num_temporal_layers,
-                          initial_tl0_pic_idx,
-                          tl0_frame_dropper,
-                          tl1_frame_dropper) {}
-  virtual ~ScreenshareLayersFT() {}
-};
+// 5 frames per second at 90 kHz.
+const uint32_t kTimestampDelta5Fps = 90000 / 5;
+const int kDefaultQp = 54;
+const int kDefaultTl0BitrateKbps = 200;
+const int kDefaultTl1BitrateKbps = 2000;
+const int kFrameRate = 5;
+const int kSyncPeriodSeconds = 5;
+const int kMaxSyncPeriodSeconds = 10;
 
 class ScreenshareLayerTest : public ::testing::Test {
  protected:
-  void SetEncodeExpectations(bool drop_tl0, bool drop_tl1, int framerate) {
-    EXPECT_CALL(tl0_frame_dropper_, DropFrame())
-        .Times(1)
-        .WillRepeatedly(Return(drop_tl0));
-    if (drop_tl0) {
-      EXPECT_CALL(tl1_frame_dropper_, DropFrame())
-          .Times(1)
-          .WillRepeatedly(Return(drop_tl1));
-    }
-    EXPECT_CALL(tl0_frame_dropper_, Leak(framerate))
-        .Times(1);
-    EXPECT_CALL(tl1_frame_dropper_, Leak(framerate))
-        .Times(1);
-    if (drop_tl0) {
-      EXPECT_CALL(tl0_frame_dropper_, Fill(_, _))
-          .Times(0);
-      if (drop_tl1) {
-        EXPECT_CALL(tl1_frame_dropper_, Fill(_, _))
-              .Times(0);
-      } else {
-        EXPECT_CALL(tl1_frame_dropper_, Fill(kFrameSize, true))
-            .Times(1);
-      }
-    } else {
-      EXPECT_CALL(tl0_frame_dropper_, Fill(kFrameSize, true))
-          .Times(1);
-      EXPECT_CALL(tl1_frame_dropper_, Fill(kFrameSize, true))
-          .Times(1);
-    }
-  }
+  ScreenshareLayerTest() : min_qp_(2), max_qp_(kDefaultQp), frame_size_(-1) {}
+  virtual ~ScreenshareLayerTest() {}
 
   void EncodeFrame(uint32_t timestamp,
                    bool base_sync,
@@ -85,179 +42,326 @@ class ScreenshareLayerTest : public ::testing::Test {
                    int* flags) {
     *flags = layers_->EncodeFlags(timestamp);
     layers_->PopulateCodecSpecific(base_sync, vp8_info, timestamp);
-    layers_->FrameEncoded(kFrameSize, timestamp);
+    ASSERT_NE(-1, frame_size_);
+    layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
   }
 
-  NiceMock<MockFrameDropper> tl0_frame_dropper_;
-  NiceMock<MockFrameDropper> tl1_frame_dropper_;
-  rtc::scoped_ptr<ScreenshareLayersFT> layers_;
+  void ConfigureBitrates() {
+    vpx_codec_enc_cfg_t vpx_cfg;
+    memset(&vpx_cfg, 0, sizeof(vpx_codec_enc_cfg_t));
+    vpx_cfg.rc_min_quantizer = min_qp_;
+    vpx_cfg.rc_max_quantizer = max_qp_;
+    EXPECT_TRUE(layers_->ConfigureBitrates(
+        kDefaultTl0BitrateKbps, kDefaultTl1BitrateKbps, kFrameRate, &vpx_cfg));
+    frame_size_ = ((vpx_cfg.rc_target_bitrate * 1000) / 8) / kFrameRate;
+  }
+
+  void WithQpLimits(int min_qp, int max_qp) {
+    min_qp_ = min_qp;
+    max_qp_ = max_qp;
+  }
+
+  int RunGracePeriod() {
+    int flags = 0;
+    uint32_t timestamp = 0;
+    CodecSpecificInfoVP8 vp8_info;
+    bool got_tl0 = false;
+    bool got_tl1 = false;
+    for (int i = 0; i < 10; ++i) {
+      EncodeFrame(timestamp, false, &vp8_info, &flags);
+      timestamp += kTimestampDelta5Fps;
+      if (vp8_info.temporalIdx == 0) {
+        got_tl0 = true;
+      } else {
+        got_tl1 = true;
+      }
+      if (got_tl0 && got_tl1)
+        return timestamp;
+    }
+    ADD_FAILURE() << "Frames from both layers not received in time.";
+    return 0;
+  }
+
+  int SkipUntilTl(int layer, int timestamp) {
+    CodecSpecificInfoVP8 vp8_info;
+    for (int i = 0; i < 5; ++i) {
+      layers_->EncodeFlags(timestamp);
+      timestamp += kTimestampDelta5Fps;
+      layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+      if (vp8_info.temporalIdx != layer) {
+        layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+      } else {
+        return timestamp;
+      }
+    }
+    ADD_FAILURE() << "Did not get a frame of TL" << layer << " in time.";
+    return 0;
+  }
+
+  int min_qp_;
+  int max_qp_;
+  int frame_size_;
+  rtc::scoped_ptr<ScreenshareLayers> layers_;
 };
 
 TEST_F(ScreenshareLayerTest, 1Layer) {
-  layers_.reset(
-      new ScreenshareLayersFT(1, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
-  EXPECT_TRUE(layers_->ConfigureBitrates(100, 1000, 5, NULL));
+  layers_.reset(new ScreenshareLayers(1, 0));
+  ConfigureBitrates();
   int flags = 0;
   uint32_t timestamp = 0;
   CodecSpecificInfoVP8 vp8_info;
   // One layer screenshare should not use the frame dropper as all frames will
   // belong to the base layer.
-  EXPECT_CALL(tl0_frame_dropper_, DropFrame())
-      .Times(0);
-  EXPECT_CALL(tl1_frame_dropper_, DropFrame())
-      .Times(0);
+  const int kSingleLayerFlags = 0;
   flags = layers_->EncodeFlags(timestamp);
-  EXPECT_EQ(0, flags);
+  EXPECT_EQ(kSingleLayerFlags, flags);
   layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
   EXPECT_EQ(static_cast<uint8_t>(kNoTemporalIdx), vp8_info.temporalIdx);
   EXPECT_FALSE(vp8_info.layerSync);
   EXPECT_EQ(kNoTl0PicIdx, vp8_info.tl0PicIdx);
-  layers_->FrameEncoded(kFrameSize, timestamp);
-
-  EXPECT_CALL(tl0_frame_dropper_, DropFrame())
-      .Times(0);
-  EXPECT_CALL(tl1_frame_dropper_, DropFrame())
-      .Times(0);
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
   flags = layers_->EncodeFlags(timestamp);
-  EXPECT_EQ(0, flags);
+  EXPECT_EQ(kSingleLayerFlags, flags);
   timestamp += kTimestampDelta5Fps;
   layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
   EXPECT_EQ(static_cast<uint8_t>(kNoTemporalIdx), vp8_info.temporalIdx);
   EXPECT_FALSE(vp8_info.layerSync);
   EXPECT_EQ(kNoTl0PicIdx, vp8_info.tl0PicIdx);
-  layers_->FrameEncoded(kFrameSize, timestamp);
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
 }
 
 TEST_F(ScreenshareLayerTest, 2Layer) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
-  EXPECT_TRUE(layers_->ConfigureBitrates(100, 1000, 5, NULL));
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
   int flags = 0;
   uint32_t timestamp = 0;
   uint8_t expected_tl0_idx = 0;
   CodecSpecificInfoVP8 vp8_info;
-  SetEncodeExpectations(false, false, 1);
   EncodeFrame(timestamp, false, &vp8_info, &flags);
-  EXPECT_EQ(kFlagsTL0, flags);
+  EXPECT_EQ(ScreenshareLayers::kTl0Flags, flags);
   EXPECT_EQ(0, vp8_info.temporalIdx);
   EXPECT_FALSE(vp8_info.layerSync);
   ++expected_tl0_idx;
   EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
 
-  EXPECT_CALL(tl1_frame_dropper_, SetRates(1000, 1))
-      .Times(1);
-  EXPECT_TRUE(layers_->ConfigureBitrates(100, 1000, -1, NULL));
-  // Insert 5 frames at 30 fps. All should belong to TL0.
+  // Insert 5 frames, cover grace period. All should be in TL0.
   for (int i = 0; i < 5; ++i) {
-    timestamp += kTimestampDelta30Fps;
-    // First iteration has a framerate based on a single frame, thus 1.
-    SetEncodeExpectations(false, false, 30);
+    timestamp += kTimestampDelta5Fps;
     EncodeFrame(timestamp, false, &vp8_info, &flags);
     EXPECT_EQ(0, vp8_info.temporalIdx);
     EXPECT_FALSE(vp8_info.layerSync);
     ++expected_tl0_idx;
     EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
   }
-  // Drop two frames from TL0, thus being coded in TL1.
-  timestamp += kTimestampDelta30Fps;
-  SetEncodeExpectations(true, false, 30);
+
+  // First frame in TL0.
+  timestamp += kTimestampDelta5Fps;
   EncodeFrame(timestamp, false, &vp8_info, &flags);
-  EXPECT_EQ(kFlagsTL1Sync, flags);
+  EXPECT_EQ(ScreenshareLayers::kTl0Flags, flags);
+  EXPECT_EQ(0, vp8_info.temporalIdx);
+  EXPECT_FALSE(vp8_info.layerSync);
+  ++expected_tl0_idx;
+  EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
+
+  // Drop two frames from TL0, thus being coded in TL1.
+  timestamp += kTimestampDelta5Fps;
+  EncodeFrame(timestamp, false, &vp8_info, &flags);
+  // First frame is sync frame.
+  EXPECT_EQ(ScreenshareLayers::kTl1SyncFlags, flags);
   EXPECT_EQ(1, vp8_info.temporalIdx);
   EXPECT_TRUE(vp8_info.layerSync);
   EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
 
-  timestamp += kTimestampDelta30Fps;
-  SetEncodeExpectations(true, false, 30);
+  timestamp += kTimestampDelta5Fps;
   EncodeFrame(timestamp, false, &vp8_info, &flags);
-  EXPECT_EQ(kFlagsTL1, flags);
+  EXPECT_EQ(ScreenshareLayers::kTl1Flags, flags);
   EXPECT_EQ(1, vp8_info.temporalIdx);
   EXPECT_FALSE(vp8_info.layerSync);
   EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
 }
 
 TEST_F(ScreenshareLayerTest, 2LayersPeriodicSync) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
-  EXPECT_TRUE(layers_->ConfigureBitrates(100, 1000, 5, NULL));
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
   int flags = 0;
   uint32_t timestamp = 0;
   CodecSpecificInfoVP8 vp8_info;
-  const int kNumFrames = 10;
-  const bool kDrops[kNumFrames] = {false, true, true, true, true,
-      true, true, true, true, true};
-  const int kExpectedFramerates[kNumFrames] = {1, 5, 5, 5, 5, 5, 5, 5, 5, 5};
-  const bool kExpectedSyncs[kNumFrames] = {false, true, false, false, false,
-      false, false, true, false, false};
-  const int kExpectedTemporalIdx[kNumFrames] = {0, 1, 1, 1, 1, 1, 1, 1, 1, 1};
+  std::vector<int> sync_times;
+
+  const int kNumFrames = kSyncPeriodSeconds * kFrameRate * 2 - 1;
   for (int i = 0; i < kNumFrames; ++i) {
     timestamp += kTimestampDelta5Fps;
-    SetEncodeExpectations(kDrops[i], false, kExpectedFramerates[i]);
     EncodeFrame(timestamp, false, &vp8_info, &flags);
-    EXPECT_EQ(kExpectedTemporalIdx[i], vp8_info.temporalIdx);
-    EXPECT_EQ(kExpectedSyncs[i], vp8_info.layerSync) << "Iteration: " << i;
-    EXPECT_EQ(1, vp8_info.tl0PicIdx);
+    if (vp8_info.temporalIdx == 1 && vp8_info.layerSync) {
+      sync_times.push_back(timestamp);
+    }
   }
+
+  ASSERT_EQ(2u, sync_times.size());
+  EXPECT_GE(sync_times[1] - sync_times[0], 90000 * kSyncPeriodSeconds);
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersSyncAfterTimeout) {
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
+  uint32_t timestamp = 0;
+  CodecSpecificInfoVP8 vp8_info;
+  std::vector<int> sync_times;
+
+  const int kNumFrames = kMaxSyncPeriodSeconds * kFrameRate * 2 - 1;
+  for (int i = 0; i < kNumFrames; ++i) {
+    timestamp += kTimestampDelta5Fps;
+    layers_->EncodeFlags(timestamp);
+    layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+
+    // Simulate TL1 being at least 8 qp steps better.
+    if (vp8_info.temporalIdx == 0) {
+      layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+    } else {
+      layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp - 8);
+    }
+
+    if (vp8_info.temporalIdx == 1 && vp8_info.layerSync)
+      sync_times.push_back(timestamp);
+  }
+
+  ASSERT_EQ(2u, sync_times.size());
+  EXPECT_GE(sync_times[1] - sync_times[0], 90000 * kMaxSyncPeriodSeconds);
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersSyncAfterSimilarQP) {
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
+  uint32_t timestamp = 0;
+  CodecSpecificInfoVP8 vp8_info;
+  std::vector<int> sync_times;
+
+  const int kNumFrames = (kSyncPeriodSeconds +
+                          ((kMaxSyncPeriodSeconds - kSyncPeriodSeconds) / 2)) *
+                         kFrameRate;
+  for (int i = 0; i < kNumFrames; ++i) {
+    timestamp += kTimestampDelta5Fps;
+    layers_->EncodeFlags(timestamp);
+    layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+
+    // Simulate TL1 being at least 8 qp steps better.
+    if (vp8_info.temporalIdx == 0) {
+      layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+    } else {
+      layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp - 8);
+    }
+
+    if (vp8_info.temporalIdx == 1 && vp8_info.layerSync)
+      sync_times.push_back(timestamp);
+  }
+
+  ASSERT_EQ(1u, sync_times.size());
+
+  bool bumped_tl0_quality = false;
+  for (int i = 0; i < 3; ++i) {
+    timestamp += kTimestampDelta5Fps;
+    int flags = layers_->EncodeFlags(timestamp);
+    layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+
+    if (vp8_info.temporalIdx == 0) {
+      // Bump TL0 to same quality as TL1.
+      layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp - 8);
+      bumped_tl0_quality = true;
+    } else {
+      layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp - 8);
+      if (bumped_tl0_quality) {
+        EXPECT_TRUE(vp8_info.layerSync);
+        EXPECT_EQ(ScreenshareLayers::kTl1SyncFlags, flags);
+        return;
+      }
+    }
+  }
+  ADD_FAILURE() << "No TL1 frame arrived within time limit.";
 }
 
 TEST_F(ScreenshareLayerTest, 2LayersToggling) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
-  EXPECT_TRUE(layers_->ConfigureBitrates(100, 1000, 5, NULL));
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
+  int flags = 0;
+  CodecSpecificInfoVP8 vp8_info;
+  uint32_t timestamp = RunGracePeriod();
+
+  // Insert 50 frames. 2/5 should be TL0.
+  int tl0_frames = 0;
+  int tl1_frames = 0;
+  for (int i = 0; i < 50; ++i) {
+    timestamp += kTimestampDelta5Fps;
+    EncodeFrame(timestamp, false, &vp8_info, &flags);
+    switch (vp8_info.temporalIdx) {
+      case 0:
+        ++tl0_frames;
+        break;
+      case 1:
+        ++tl1_frames;
+        break;
+      default:
+        abort();
+    }
+  }
+  EXPECT_EQ(20, tl0_frames);
+  EXPECT_EQ(30, tl1_frames);
+}
+
+TEST_F(ScreenshareLayerTest, AllFitsLayer0) {
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
+  frame_size_ = ((kDefaultTl0BitrateKbps * 1000) / 8) / kFrameRate;
+
   int flags = 0;
   uint32_t timestamp = 0;
   CodecSpecificInfoVP8 vp8_info;
-  const int kNumFrames = 10;
-  const bool kDrops[kNumFrames] = {false, true, false, true, false,
-      true, false, true, false, true};
-  const int kExpectedFramerates[kNumFrames] = {1, 5, 5, 5, 5, 5, 5, 5, 5, 5};
-  const bool kExpectedSyncs[kNumFrames] = {false, true, false, false, false,
-      false, false, true, false, false};
-  const int kExpectedTemporalIdx[kNumFrames] = {0, 1, 0, 1, 0, 1, 0, 1, 0, 1};
-  const int kExpectedTl0Idx[kNumFrames] = {1, 1, 2, 2, 3, 3, 4, 4, 5, 5};
-  for (int i = 0; i < kNumFrames; ++i) {
-    timestamp += kTimestampDelta5Fps;
-    SetEncodeExpectations(kDrops[i], false, kExpectedFramerates[i]);
+  // Insert 50 frames, small enough that all fits in TL0.
+  for (int i = 0; i < 50; ++i) {
     EncodeFrame(timestamp, false, &vp8_info, &flags);
-    EXPECT_EQ(kExpectedTemporalIdx[i], vp8_info.temporalIdx);
-    EXPECT_EQ(kExpectedSyncs[i], vp8_info.layerSync) << "Iteration: " << i;
-    EXPECT_EQ(kExpectedTl0Idx[i], vp8_info.tl0PicIdx);
+    timestamp += kTimestampDelta5Fps;
+    EXPECT_EQ(ScreenshareLayers::kTl0Flags, flags);
+    EXPECT_EQ(0, vp8_info.temporalIdx);
   }
 }
 
-TEST_F(ScreenshareLayerTest, 2LayersBothDrops) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
-  EXPECT_TRUE(layers_->ConfigureBitrates(100, 1000, 5, NULL));
+TEST_F(ScreenshareLayerTest, TooHighBitrate) {
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
+  frame_size_ = 2 * ((kDefaultTl1BitrateKbps * 1000) / 8) / kFrameRate;
   int flags = 0;
-  uint32_t timestamp = 0;
-  uint8_t expected_tl0_idx = 0;
   CodecSpecificInfoVP8 vp8_info;
-  SetEncodeExpectations(false, false, 1);
-  EncodeFrame(timestamp, false, &vp8_info, &flags);
-  EXPECT_EQ(kFlagsTL0, flags);
-  EXPECT_EQ(0, vp8_info.temporalIdx);
-  EXPECT_FALSE(vp8_info.layerSync);
-  ++expected_tl0_idx;
-  EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
+  uint32_t timestamp = RunGracePeriod();
 
-  timestamp += kTimestampDelta5Fps;
-  SetEncodeExpectations(true, false, 5);
-  EncodeFrame(timestamp, false, &vp8_info, &flags);
-  EXPECT_EQ(kFlagsTL1Sync, flags);
-  EXPECT_EQ(1, vp8_info.temporalIdx);
-  EXPECT_TRUE(vp8_info.layerSync);
-  EXPECT_EQ(expected_tl0_idx, vp8_info.tl0PicIdx);
+  // Insert 100 frames. Half should be dropped.
+  int tl0_frames = 0;
+  int tl1_frames = 0;
+  int dropped_frames = 0;
+  for (int i = 0; i < 100; ++i) {
+    timestamp += kTimestampDelta5Fps;
+    EncodeFrame(timestamp, false, &vp8_info, &flags);
+    if (flags == -1) {
+      ++dropped_frames;
+    } else {
+      switch (vp8_info.temporalIdx) {
+        case 0:
+          ++tl0_frames;
+          break;
+        case 1:
+          ++tl1_frames;
+          break;
+        default:
+          abort();
+      }
+    }
+  }
 
-  timestamp += kTimestampDelta5Fps;
-  SetEncodeExpectations(true, true, 5);
-  flags = layers_->EncodeFlags(timestamp);
-  EXPECT_EQ(-1, flags);
+  EXPECT_EQ(5, tl0_frames);
+  EXPECT_EQ(45, tl1_frames);
+  EXPECT_EQ(50, dropped_frames);
 }
 
 TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL0) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
+  layers_.reset(new ScreenshareLayers(2, 0));
 
   vpx_codec_enc_cfg_t cfg;
   layers_->ConfigureBitrates(100, 1000, 5, &cfg);
@@ -268,8 +372,7 @@ TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL0) {
 }
 
 TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL1) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
+  layers_.reset(new ScreenshareLayers(2, 0));
   vpx_codec_enc_cfg_t cfg;
   layers_->ConfigureBitrates(100, 450, 5, &cfg);
 
@@ -279,12 +382,64 @@ TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL1) {
 }
 
 TEST_F(ScreenshareLayerTest, TargetBitrateBelowTL0) {
-  layers_.reset(
-      new ScreenshareLayersFT(2, 0, &tl0_frame_dropper_, &tl1_frame_dropper_));
+  layers_.reset(new ScreenshareLayers(2, 0));
   vpx_codec_enc_cfg_t cfg;
   layers_->ConfigureBitrates(100, 100, 5, &cfg);
 
   EXPECT_EQ(100U, cfg.rc_target_bitrate);
 }
 
+TEST_F(ScreenshareLayerTest, EncoderDrop) {
+  layers_.reset(new ScreenshareLayers(2, 0));
+  ConfigureBitrates();
+  CodecSpecificInfoVP8 vp8_info;
+  vpx_codec_enc_cfg_t cfg;
+  cfg.rc_max_quantizer = kDefaultQp;
+
+  uint32_t timestamp = RunGracePeriod();
+  timestamp = SkipUntilTl(0, timestamp);
+
+  // Size 0 indicates dropped frame.
+  layers_->FrameEncoded(0, timestamp, kDefaultQp);
+  timestamp += kTimestampDelta5Fps;
+  EXPECT_FALSE(layers_->UpdateConfiguration(&cfg));
+  EXPECT_EQ(ScreenshareLayers::kTl0Flags, layers_->EncodeFlags(timestamp));
+  layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+
+  timestamp = SkipUntilTl(0, timestamp);
+  EXPECT_TRUE(layers_->UpdateConfiguration(&cfg));
+  EXPECT_LT(cfg.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+
+  layers_->EncodeFlags(timestamp);
+  timestamp += kTimestampDelta5Fps;
+  EXPECT_TRUE(layers_->UpdateConfiguration(&cfg));
+  layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+  EXPECT_EQ(cfg.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+
+  // Next drop in TL1.
+
+  timestamp = SkipUntilTl(1, timestamp);
+  layers_->FrameEncoded(0, timestamp, kDefaultQp);
+  timestamp += kTimestampDelta5Fps;
+  EXPECT_FALSE(layers_->UpdateConfiguration(&cfg));
+  EXPECT_EQ(ScreenshareLayers::kTl1Flags, layers_->EncodeFlags(timestamp));
+  layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+
+  timestamp = SkipUntilTl(1, timestamp);
+  EXPECT_TRUE(layers_->UpdateConfiguration(&cfg));
+  EXPECT_LT(cfg.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+
+  layers_->EncodeFlags(timestamp);
+  timestamp += kTimestampDelta5Fps;
+  EXPECT_TRUE(layers_->UpdateConfiguration(&cfg));
+  layers_->PopulateCodecSpecific(false, &vp8_info, timestamp);
+  EXPECT_EQ(cfg.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
index 4035412465..3dfcde048b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
@@ -15,6 +15,7 @@
 // NOTE(ajm): Path provided by gyp.
 #include "libyuv/scale.h"  // NOLINT
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
 
@@ -102,22 +103,43 @@ struct ScreenshareTemporalLayersFactory : webrtc::TemporalLayers::Factory {
 
   virtual webrtc::TemporalLayers* Create(int num_temporal_layers,
                                          uint8_t initial_tl0_pic_idx) const {
-    return new webrtc::ScreenshareLayers(num_temporal_layers,
-                                         rand(),
-                                         &tl0_frame_dropper_,
-                                         &tl1_frame_dropper_);
+    return new webrtc::ScreenshareLayers(num_temporal_layers, rand());
   }
 
   mutable webrtc::FrameDropper tl0_frame_dropper_;
   mutable webrtc::FrameDropper tl1_frame_dropper_;
 };
 
+// An EncodedImageCallback implementation that forwards on calls to a
+// SimulcastEncoderAdapter, but with the stream index it's registered with as
+// the first parameter to Encoded.
+class AdapterEncodedImageCallback : public webrtc::EncodedImageCallback {
+ public:
+  AdapterEncodedImageCallback(webrtc::SimulcastEncoderAdapter* adapter,
+                              size_t stream_idx)
+      : adapter_(adapter), stream_idx_(stream_idx) {}
+
+  int32_t Encoded(
+      const webrtc::EncodedImage& encodedImage,
+      const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
+      const webrtc::RTPFragmentationHeader* fragmentation = NULL) override {
+    return adapter_->Encoded(stream_idx_, encodedImage, codecSpecificInfo,
+                             fragmentation);
+  }
+
+ private:
+  webrtc::SimulcastEncoderAdapter* const adapter_;
+  const size_t stream_idx_;
+};
+
 }  // namespace
 
 namespace webrtc {
 
 SimulcastEncoderAdapter::SimulcastEncoderAdapter(VideoEncoderFactory* factory)
-    : factory_(factory), encoded_complete_callback_(NULL) {
+    : factory_(factory),
+      encoded_complete_callback_(NULL),
+      implementation_name_("SimulcastEncoderAdapter") {
   memset(&codec_, 0, sizeof(webrtc::VideoCodec));
 }
 
@@ -133,7 +155,9 @@ int SimulcastEncoderAdapter::Release() {
   // ~SimulcastEncoderAdapter().
   while (!streaminfos_.empty()) {
     VideoEncoder* encoder = streaminfos_.back().encoder;
+    EncodedImageCallback* callback = streaminfos_.back().callback;
     factory_->Destroy(encoder);
+    delete callback;
     streaminfos_.pop_back();
   }
   return WEBRTC_VIDEO_CODEC_OK;
@@ -173,6 +197,7 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
     codec_.extra_options = screensharing_extra_options_.get();
   }
 
+  std::string implementation_name;
   // Create |number_of_streams| of encoder instances and init them.
   for (int i = 0; i < number_of_streams; ++i) {
     VideoCodec stream_codec;
@@ -182,8 +207,9 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
       stream_codec.numberOfSimulcastStreams = 1;
     } else {
       bool highest_resolution_stream = (i == (number_of_streams - 1));
-      PopulateStreamCodec(&codec_, i, highest_resolution_stream,
-                          &stream_codec, &send_stream);
+      PopulateStreamCodec(&codec_, i, number_of_streams,
+                          highest_resolution_stream, &stream_codec,
+                          &send_stream);
     }
 
     // TODO(ronghuawu): Remove once this is handled in VP8EncoderImpl.
@@ -192,26 +218,28 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
     }
 
     VideoEncoder* encoder = factory_->Create();
-    ret = encoder->InitEncode(&stream_codec,
-                              number_of_cores,
-                              max_payload_size);
+    ret = encoder->InitEncode(&stream_codec, number_of_cores, max_payload_size);
     if (ret < 0) {
       Release();
       return ret;
     }
-    encoder->RegisterEncodeCompleteCallback(this);
-    streaminfos_.push_back(StreamInfo(encoder,
-                                      stream_codec.width,
-                                      stream_codec.height,
-                                      send_stream));
+    EncodedImageCallback* callback = new AdapterEncodedImageCallback(this, i);
+    encoder->RegisterEncodeCompleteCallback(callback);
+    streaminfos_.push_back(StreamInfo(encoder, callback, stream_codec.width,
+                                      stream_codec.height, send_stream));
+    if (i != 0)
+      implementation_name += ", ";
+    implementation_name += streaminfos_[i].encoder->ImplementationName();
   }
+  implementation_name_ =
+      "SimulcastEncoderAdapter (" + implementation_name + ")";
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
 int SimulcastEncoderAdapter::Encode(
-    const I420VideoFrame& input_image,
+    const VideoFrame& input_image,
     const CodecSpecificInfo* codec_specific_info,
-    const std::vector<VideoFrameType>* frame_types) {
+    const std::vector<FrameType>* frame_types) {
   if (!Initialized()) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
@@ -224,7 +252,7 @@ int SimulcastEncoderAdapter::Encode(
   bool send_key_frame = false;
   if (frame_types) {
     for (size_t i = 0; i < frame_types->size(); ++i) {
-      if (frame_types->at(i) == kKeyFrame) {
+      if (frame_types->at(i) == kVideoFrameKey) {
         send_key_frame = true;
         break;
       }
@@ -241,12 +269,16 @@ int SimulcastEncoderAdapter::Encode(
   int src_width = input_image.width();
   int src_height = input_image.height();
   for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
-    std::vector<VideoFrameType> stream_frame_types;
+    // Don't encode frames in resolutions that we don't intend to send.
+    if (!streaminfos_[stream_idx].send_stream)
+      continue;
+
+    std::vector<FrameType> stream_frame_types;
     if (send_key_frame) {
-      stream_frame_types.push_back(kKeyFrame);
+      stream_frame_types.push_back(kVideoFrameKey);
       streaminfos_[stream_idx].key_frame_request = false;
     } else {
-      stream_frame_types.push_back(kDeltaFrame);
+      stream_frame_types.push_back(kVideoFrameDelta);
     }
 
     int dst_width = streaminfos_[stream_idx].width;
@@ -258,35 +290,25 @@ int SimulcastEncoderAdapter::Encode(
     // scale it to match what the encoder expects (below).
     if ((dst_width == src_width && dst_height == src_height) ||
         input_image.IsZeroSize()) {
-      streaminfos_[stream_idx].encoder->Encode(input_image,
-                                               codec_specific_info,
+      streaminfos_[stream_idx].encoder->Encode(input_image, codec_specific_info,
                                                &stream_frame_types);
     } else {
-      I420VideoFrame dst_frame;
+      VideoFrame dst_frame;
       // Making sure that destination frame is of sufficient size.
       // Aligning stride values based on width.
-      dst_frame.CreateEmptyFrame(dst_width, dst_height,
-                                 dst_width, (dst_width + 1) / 2,
-                                 (dst_width + 1) / 2);
-      libyuv::I420Scale(input_image.buffer(kYPlane),
-                        input_image.stride(kYPlane),
-                        input_image.buffer(kUPlane),
-                        input_image.stride(kUPlane),
-                        input_image.buffer(kVPlane),
-                        input_image.stride(kVPlane),
-                        src_width, src_height,
-                        dst_frame.buffer(kYPlane),
-                        dst_frame.stride(kYPlane),
-                        dst_frame.buffer(kUPlane),
-                        dst_frame.stride(kUPlane),
-                        dst_frame.buffer(kVPlane),
-                        dst_frame.stride(kVPlane),
-                        dst_width, dst_height,
-                        libyuv::kFilterBilinear);
+      dst_frame.CreateEmptyFrame(dst_width, dst_height, dst_width,
+                                 (dst_width + 1) / 2, (dst_width + 1) / 2);
+      libyuv::I420Scale(
+          input_image.buffer(kYPlane), input_image.stride(kYPlane),
+          input_image.buffer(kUPlane), input_image.stride(kUPlane),
+          input_image.buffer(kVPlane), input_image.stride(kVPlane), src_width,
+          src_height, dst_frame.buffer(kYPlane), dst_frame.stride(kYPlane),
+          dst_frame.buffer(kUPlane), dst_frame.stride(kUPlane),
+          dst_frame.buffer(kVPlane), dst_frame.stride(kVPlane), dst_width,
+          dst_height, libyuv::kFilterBilinear);
       dst_frame.set_timestamp(input_image.timestamp());
       dst_frame.set_render_time_ms(input_image.render_time_ms());
-      streaminfos_[stream_idx].encoder->Encode(dst_frame,
-                                               codec_specific_info,
+      streaminfos_[stream_idx].encoder->Encode(dst_frame, codec_specific_info,
                                                &stream_frame_types);
     }
   }
@@ -331,9 +353,8 @@ int SimulcastEncoderAdapter::SetRates(uint32_t new_bitrate_kbit,
   bool send_stream = true;
   uint32_t stream_bitrate = 0;
   for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
-    stream_bitrate = GetStreamBitrate(stream_idx,
-                                      new_bitrate_kbit,
-                                      &send_stream);
+    stream_bitrate = GetStreamBitrate(stream_idx, streaminfos_.size(),
+                                      new_bitrate_kbit, &send_stream);
     // Need a key frame if we have not sent this stream before.
     if (send_stream && !streaminfos_[stream_idx].send_stream) {
       streaminfos_[stream_idx].key_frame_request = true;
@@ -362,38 +383,24 @@ int SimulcastEncoderAdapter::SetRates(uint32_t new_bitrate_kbit,
 }
 
 int32_t SimulcastEncoderAdapter::Encoded(
+    size_t stream_idx,
     const EncodedImage& encodedImage,
     const CodecSpecificInfo* codecSpecificInfo,
     const RTPFragmentationHeader* fragmentation) {
-  size_t stream_idx = GetStreamIndex(encodedImage);
-
   CodecSpecificInfo stream_codec_specific = *codecSpecificInfo;
   CodecSpecificInfoVP8* vp8Info = &(stream_codec_specific.codecSpecific.VP8);
   vp8Info->simulcastIdx = stream_idx;
 
-  if (streaminfos_[stream_idx].send_stream) {
-    return encoded_complete_callback_->Encoded(encodedImage,
-                                               &stream_codec_specific,
-                                               fragmentation);
-  } else {
-    EncodedImage dummy_image;
-    // Required in case padding is applied to dropped frames.
-    dummy_image._timeStamp = encodedImage._timeStamp;
-    dummy_image.capture_time_ms_ = encodedImage.capture_time_ms_;
-    dummy_image._encodedWidth = encodedImage._encodedWidth;
-    dummy_image._encodedHeight = encodedImage._encodedHeight;
-    dummy_image._length = 0;
-    dummy_image._frameType = kSkipFrame;
-    vp8Info->keyIdx = kNoKeyIdx;
-    return encoded_complete_callback_->Encoded(dummy_image,
-                                               &stream_codec_specific, NULL);
-  }
+  return encoded_complete_callback_->Encoded(
+      encodedImage, &stream_codec_specific, fragmentation);
 }
 
-uint32_t SimulcastEncoderAdapter::GetStreamBitrate(int stream_idx,
-                                                   uint32_t new_bitrate_kbit,
-                                                   bool* send_stream) const {
-  if (streaminfos_.size() == 1) {
+uint32_t SimulcastEncoderAdapter::GetStreamBitrate(
+    int stream_idx,
+    size_t total_number_of_streams,
+    uint32_t new_bitrate_kbit,
+    bool* send_stream) const {
+  if (total_number_of_streams == 1) {
     *send_stream = true;
     return new_bitrate_kbit;
   }
@@ -415,16 +422,17 @@ uint32_t SimulcastEncoderAdapter::GetStreamBitrate(int stream_idx,
     // current stream's |targetBitrate|, otherwise it's capped by |maxBitrate|.
     if (stream_idx < codec_.numberOfSimulcastStreams - 1) {
       unsigned int max_rate = codec_.simulcastStream[stream_idx].maxBitrate;
-      if (new_bitrate_kbit >= SumStreamTargetBitrate(stream_idx + 1, codec_) +
-          codec_.simulcastStream[stream_idx + 1].minBitrate) {
+      if (new_bitrate_kbit >=
+          SumStreamTargetBitrate(stream_idx + 1, codec_) +
+              codec_.simulcastStream[stream_idx + 1].minBitrate) {
         max_rate = codec_.simulcastStream[stream_idx].targetBitrate;
       }
       return std::min(new_bitrate_kbit - sum_target_lower_streams, max_rate);
     } else {
-        // For the highest stream (highest resolution), the |targetBitRate| and
-        // |maxBitrate| are not used. Any excess bitrate (above the targets of
-        // all lower streams) is given to this (highest resolution) stream.
-        return new_bitrate_kbit - sum_target_lower_streams;
+      // For the highest stream (highest resolution), the |targetBitRate| and
+      // |maxBitrate| are not used. Any excess bitrate (above the targets of
+      // all lower streams) is given to this (highest resolution) stream.
+      return new_bitrate_kbit - sum_target_lower_streams;
     }
   } else {
     // Not enough bitrate for this stream.
@@ -438,6 +446,7 @@ uint32_t SimulcastEncoderAdapter::GetStreamBitrate(int stream_idx,
 void SimulcastEncoderAdapter::PopulateStreamCodec(
     const webrtc::VideoCodec* inst,
     int stream_index,
+    size_t total_number_of_streams,
     bool highest_resolution_stream,
     webrtc::VideoCodec* stream_codec,
     bool* send_stream) {
@@ -469,29 +478,34 @@ void SimulcastEncoderAdapter::PopulateStreamCodec(
   }
   // TODO(ronghuawu): what to do with targetBitrate.
 
-  int stream_bitrate = GetStreamBitrate(stream_index,
-                                        inst->startBitrate,
-                                        send_stream);
+  int stream_bitrate = GetStreamBitrate(stream_index, total_number_of_streams,
+                                        inst->startBitrate, send_stream);
   stream_codec->startBitrate = stream_bitrate;
 }
 
-size_t SimulcastEncoderAdapter::GetStreamIndex(
-    const EncodedImage& encodedImage) {
-  uint32_t width = encodedImage._encodedWidth;
-  uint32_t height = encodedImage._encodedHeight;
-  for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
-    if (streaminfos_[stream_idx].width == width &&
-        streaminfos_[stream_idx].height == height) {
-      return stream_idx;
-    }
-  }
-  // should not be here
-  assert(false);
-  return 0;
-}
-
 bool SimulcastEncoderAdapter::Initialized() const {
   return !streaminfos_.empty();
 }
 
+void SimulcastEncoderAdapter::OnDroppedFrame() {
+  streaminfos_[0].encoder->OnDroppedFrame();
+}
+
+int SimulcastEncoderAdapter::GetTargetFramerate() {
+  return streaminfos_[0].encoder->GetTargetFramerate();
+}
+
+bool SimulcastEncoderAdapter::SupportsNativeHandle() const {
+  // We should not be calling this method before streaminfos_ are configured.
+  RTC_DCHECK(!streaminfos_.empty());
+  // TODO(pbos): Support textures when using more than one encoder.
+  if (streaminfos_.size() != 1)
+    return false;
+  return streaminfos_[0].encoder->SupportsNativeHandle();
+}
+
+const char* SimulcastEncoderAdapter::ImplementationName() const {
+  return implementation_name_.c_str();
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
index d185d15134..2b8f80a8bc 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
@@ -12,6 +12,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_ENCODER_ADAPTER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_ENCODER_ADAPTER_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
@@ -30,8 +31,7 @@ class VideoEncoderFactory {
 // webrtc::VideoEncoder instances with the given VideoEncoderFactory.
 // All the public interfaces are expected to be called from the same thread,
 // e.g the encoder thread.
-class SimulcastEncoderAdapter : public VP8Encoder,
-                                public EncodedImageCallback {
+class SimulcastEncoderAdapter : public VP8Encoder {
  public:
   explicit SimulcastEncoderAdapter(VideoEncoderFactory* factory);
   virtual ~SimulcastEncoderAdapter();
@@ -41,57 +41,73 @@ class SimulcastEncoderAdapter : public VP8Encoder,
   int InitEncode(const VideoCodec* inst,
                  int number_of_cores,
                  size_t max_payload_size) override;
-  int Encode(const I420VideoFrame& input_image,
+  int Encode(const VideoFrame& input_image,
              const CodecSpecificInfo* codec_specific_info,
-             const std::vector<VideoFrameType>* frame_types) override;
+             const std::vector<FrameType>* frame_types) override;
   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
   int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
   int SetRates(uint32_t new_bitrate_kbit, uint32_t new_framerate) override;
 
-  // Implements EncodedImageCallback
-  int32_t Encoded(const EncodedImage& encodedImage,
+  // Eventual handler for the contained encoders' EncodedImageCallbacks, but
+  // called from an internal helper that also knows the correct stream
+  // index.
+  int32_t Encoded(size_t stream_idx,
+                  const EncodedImage& encodedImage,
                   const CodecSpecificInfo* codecSpecificInfo = NULL,
-                  const RTPFragmentationHeader* fragmentation = NULL) override;
+                  const RTPFragmentationHeader* fragmentation = NULL);
+
+  void OnDroppedFrame() override;
+
+  int GetTargetFramerate() override;
+  bool SupportsNativeHandle() const override;
+  const char* ImplementationName() const override;
 
  private:
   struct StreamInfo {
     StreamInfo()
-        : encoder(NULL), width(0), height(0),
-          key_frame_request(false), send_stream(true) {}
+        : encoder(NULL),
+          callback(NULL),
+          width(0),
+          height(0),
+          key_frame_request(false),
+          send_stream(true) {}
     StreamInfo(VideoEncoder* encoder,
-               unsigned short width,
-               unsigned short height,
+               EncodedImageCallback* callback,
+               uint16_t width,
+               uint16_t height,
                bool send_stream)
         : encoder(encoder),
+          callback(callback),
           width(width),
           height(height),
           key_frame_request(false),
           send_stream(send_stream) {}
     // Deleted by SimulcastEncoderAdapter::Release().
     VideoEncoder* encoder;
-    unsigned short width;
-    unsigned short height;
+    EncodedImageCallback* callback;
+    uint16_t width;
+    uint16_t height;
     bool key_frame_request;
     bool send_stream;
   };
 
   // Get the stream bitrate, for the stream |stream_idx|, given the bitrate
-  // |new_bitrate_kbit|. The function also returns whether there's enough
+  // |new_bitrate_kbit| and the actual configured stream count in
+  // |total_number_of_streams|. The function also returns whether there's enough
   // bandwidth to send this stream via |send_stream|.
   uint32_t GetStreamBitrate(int stream_idx,
+                            size_t total_number_of_streams,
                             uint32_t new_bitrate_kbit,
                             bool* send_stream) const;
 
   // Populate the codec settings for each stream.
   void PopulateStreamCodec(const webrtc::VideoCodec* inst,
                            int stream_index,
+                           size_t total_number_of_streams,
                            bool highest_resolution_stream,
                            webrtc::VideoCodec* stream_codec,
                            bool* send_stream);
 
-  // Get the stream index according to |encodedImage|.
-  size_t GetStreamIndex(const EncodedImage& encodedImage);
-
   bool Initialized() const;
 
   rtc::scoped_ptr<VideoEncoderFactory> factory_;
@@ -99,9 +115,9 @@ class SimulcastEncoderAdapter : public VP8Encoder,
   VideoCodec codec_;
   std::vector<StreamInfo> streaminfos_;
   EncodedImageCallback* encoded_complete_callback_;
+  std::string implementation_name_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_ENCODER_ADAPTER_H_
-
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
index 2c2a323f95..6f0cd5ba77 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
@@ -11,7 +11,7 @@
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h"
 #include "webrtc/modules/video_coding/codecs/vp8/vp8_factory.h"
@@ -27,12 +27,10 @@ static VP8Encoder* CreateTestEncoderAdapter() {
 class TestSimulcastEncoderAdapter : public TestVp8Simulcast {
  public:
   TestSimulcastEncoderAdapter()
-     : TestVp8Simulcast(CreateTestEncoderAdapter(),
-                        VP8Decoder::Create()) {}
+      : TestVp8Simulcast(CreateTestEncoderAdapter(), VP8Decoder::Create()) {}
+
  protected:
-  virtual void SetUp() {
-    TestVp8Simulcast::SetUp();
-  }
+  virtual void SetUp() { TestVp8Simulcast::SetUp(); }
   virtual void TearDown() {
     TestVp8Simulcast::TearDown();
     VP8EncoderFactoryConfig::set_use_simulcast_adapter(false);
@@ -97,8 +95,7 @@ TEST_F(TestSimulcastEncoderAdapter, TestSpatioTemporalLayers321PatternEncoder) {
 
 // TODO(ronghuawu): Enable this test when SkipEncodingUnusedStreams option is
 // implemented for SimulcastEncoderAdapter.
-TEST_F(TestSimulcastEncoderAdapter,
-    DISABLED_TestSkipEncodingUnusedStreams) {
+TEST_F(TestSimulcastEncoderAdapter, DISABLED_TestSkipEncodingUnusedStreams) {
   TestVp8Simulcast::TestSkipEncodingUnusedStreams();
 }
 
@@ -110,43 +107,67 @@ class MockVideoEncoder : public VideoEncoder {
  public:
   int32_t InitEncode(const VideoCodec* codecSettings,
                      int32_t numberOfCores,
-                     size_t maxPayloadSize) {
+                     size_t maxPayloadSize) override {
     codec_ = *codecSettings;
     return 0;
   }
 
-  int32_t Encode(const I420VideoFrame& inputImage,
+  int32_t Encode(const VideoFrame& inputImage,
                  const CodecSpecificInfo* codecSpecificInfo,
-                 const std::vector<VideoFrameType>* frame_types) { return 0; }
-
-  int32_t RegisterEncodeCompleteCallback(EncodedImageCallback* callback) {
+                 const std::vector<FrameType>* frame_types) override {
     return 0;
   }
 
-  int32_t Release() {
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* callback) override {
+    callback_ = callback;
     return 0;
   }
 
-  int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) {
+  int32_t Release() override { return 0; }
+
+  int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override {
     return 0;
   }
 
-  MOCK_METHOD2(SetChannelParameters,
-      int32_t(uint32_t packetLoss, int64_t rtt));
+  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
 
-  virtual ~MockVideoEncoder() {
-  }
+  bool SupportsNativeHandle() const override { return supports_native_handle_; }
+
+  virtual ~MockVideoEncoder() {}
 
   const VideoCodec& codec() const { return codec_; }
 
+  void SendEncodedImage(int width, int height) {
+    // Sends a fake image of the given width/height.
+    EncodedImage image;
+    image._encodedWidth = width;
+    image._encodedHeight = height;
+    CodecSpecificInfo codecSpecificInfo;
+    memset(&codecSpecificInfo, 0, sizeof(codecSpecificInfo));
+    callback_->Encoded(image, &codecSpecificInfo, NULL);
+  }
+
+  void set_supports_native_handle(bool enabled) {
+    supports_native_handle_ = enabled;
+  }
+
+  MOCK_CONST_METHOD0(ImplementationName, const char*());
+
  private:
+  bool supports_native_handle_ = false;
   VideoCodec codec_;
+  EncodedImageCallback* callback_;
 };
 
 class MockVideoEncoderFactory : public VideoEncoderFactory {
  public:
   VideoEncoder* Create() override {
     MockVideoEncoder* encoder = new MockVideoEncoder();
+    const char* encoder_name = encoder_names_.empty()
+                                   ? "codec_implementation_name"
+                                   : encoder_names_[encoders_.size()];
+    ON_CALL(*encoder, ImplementationName()).WillByDefault(Return(encoder_name));
     encoders_.push_back(encoder);
     return encoder;
   }
@@ -156,9 +177,13 @@ class MockVideoEncoderFactory : public VideoEncoderFactory {
   virtual ~MockVideoEncoderFactory() {}
 
   const std::vector<MockVideoEncoder*>& encoders() const { return encoders_; }
+  void SetEncoderNames(const std::vector<const char*>& encoder_names) {
+    encoder_names_ = encoder_names;
+  }
 
  private:
   std::vector<MockVideoEncoder*> encoders_;
+  std::vector<const char*> encoder_names_;
 };
 
 class TestSimulcastEncoderAdapterFakeHelper {
@@ -176,7 +201,8 @@ class TestSimulcastEncoderAdapterFakeHelper {
     EXPECT_TRUE(!factory_->encoders().empty());
     for (size_t i = 0; i < factory_->encoders().size(); ++i) {
       EXPECT_CALL(*factory_->encoders()[i],
-                  SetChannelParameters(packetLoss, rtt)).Times(1);
+                  SetChannelParameters(packetLoss, rtt))
+          .Times(1);
     }
   }
 
@@ -188,18 +214,46 @@ class TestSimulcastEncoderAdapterFakeHelper {
 
 static const int kTestTemporalLayerProfile[3] = {3, 2, 1};
 
-class TestSimulcastEncoderAdapterFake : public ::testing::Test {
+class TestSimulcastEncoderAdapterFake : public ::testing::Test,
+                                        public EncodedImageCallback {
  public:
   TestSimulcastEncoderAdapterFake()
-     : helper_(new TestSimulcastEncoderAdapterFakeHelper()),
-       adapter_(helper_->CreateMockEncoderAdapter()) {}
+      : helper_(new TestSimulcastEncoderAdapterFakeHelper()),
+        adapter_(helper_->CreateMockEncoderAdapter()),
+        last_encoded_image_width_(-1),
+        last_encoded_image_height_(-1),
+        last_encoded_image_simulcast_index_(-1) {}
   virtual ~TestSimulcastEncoderAdapterFake() {}
 
+  int32_t Encoded(const EncodedImage& encodedImage,
+                  const CodecSpecificInfo* codecSpecificInfo = NULL,
+                  const RTPFragmentationHeader* fragmentation = NULL) override {
+    last_encoded_image_width_ = encodedImage._encodedWidth;
+    last_encoded_image_height_ = encodedImage._encodedHeight;
+    if (codecSpecificInfo) {
+      last_encoded_image_simulcast_index_ =
+          codecSpecificInfo->codecSpecific.VP8.simulcastIdx;
+    }
+    return 0;
+  }
+
+  bool GetLastEncodedImageInfo(int* out_width,
+                               int* out_height,
+                               int* out_simulcast_index) {
+    if (last_encoded_image_width_ == -1) {
+      return false;
+    }
+    *out_width = last_encoded_image_width_;
+    *out_height = last_encoded_image_height_;
+    *out_simulcast_index = last_encoded_image_simulcast_index_;
+    return true;
+  }
+
   void SetupCodec() {
     TestVp8Simulcast::DefaultSettings(
-      &codec_,
-      static_cast<const int*>(kTestTemporalLayerProfile));
+        &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
     EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
+    adapter_->RegisterEncodeCompleteCallback(this);
   }
 
   void VerifyCodec(const VideoCodec& ref, int stream_index) {
@@ -269,11 +323,16 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test {
     // stream 1
     InitRefCodec(1, &ref_codec);
     ref_codec.codecSpecific.VP8.denoisingOn = false;
-    ref_codec.startBitrate = 300;
+    // The start bitrate (300kbit) minus what we have for the lower layers
+    // (100kbit).
+    ref_codec.startBitrate = 200;
     VerifyCodec(ref_codec, 1);
 
     // stream 2, the biggest resolution stream.
     InitRefCodec(2, &ref_codec);
+    // We don't have enough bits to send this, so the adapter should have
+    // configured it to use the min bitrate for this layer (600kbit) but turn
+    // off sending.
     ref_codec.startBitrate = 600;
     VerifyCodec(ref_codec, 2);
   }
@@ -282,6 +341,9 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test {
   rtc::scoped_ptr<TestSimulcastEncoderAdapterFakeHelper> helper_;
   rtc::scoped_ptr<VP8Encoder> adapter_;
   VideoCodec codec_;
+  int last_encoded_image_width_;
+  int last_encoded_image_height_;
+  int last_encoded_image_simulcast_index_;
 };
 
 TEST_F(TestSimulcastEncoderAdapterFake, InitEncode) {
@@ -297,5 +359,81 @@ TEST_F(TestSimulcastEncoderAdapterFake, SetChannelParameters) {
   adapter_->SetChannelParameters(packetLoss, rtt);
 }
 
+TEST_F(TestSimulcastEncoderAdapterFake, EncodedCallbackForDifferentEncoders) {
+  SetupCodec();
+
+  // Set bitrates so that we send all layers.
+  adapter_->SetRates(1200, 30);
+
+  // At this point, the simulcast encoder adapter should have 3 streams: HD,
+  // quarter HD, and quarter quarter HD. We're going to mostly ignore the exact
+  // resolutions, to test that the adapter forwards on the correct resolution
+  // and simulcast index values, going only off the encoder that generates the
+  // image.
+  EXPECT_EQ(3u, helper_->factory()->encoders().size());
+  helper_->factory()->encoders()[0]->SendEncodedImage(1152, 704);
+  int width;
+  int height;
+  int simulcast_index;
+  EXPECT_TRUE(GetLastEncodedImageInfo(&width, &height, &simulcast_index));
+  EXPECT_EQ(1152, width);
+  EXPECT_EQ(704, height);
+  EXPECT_EQ(0, simulcast_index);
+
+  helper_->factory()->encoders()[1]->SendEncodedImage(300, 620);
+  EXPECT_TRUE(GetLastEncodedImageInfo(&width, &height, &simulcast_index));
+  EXPECT_EQ(300, width);
+  EXPECT_EQ(620, height);
+  EXPECT_EQ(1, simulcast_index);
+
+  helper_->factory()->encoders()[2]->SendEncodedImage(120, 240);
+  EXPECT_TRUE(GetLastEncodedImageInfo(&width, &height, &simulcast_index));
+  EXPECT_EQ(120, width);
+  EXPECT_EQ(240, height);
+  EXPECT_EQ(2, simulcast_index);
+}
+
+TEST_F(TestSimulcastEncoderAdapterFake, SupportsNativeHandleForSingleStreams) {
+  TestVp8Simulcast::DefaultSettings(
+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
+  codec_.numberOfSimulcastStreams = 1;
+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
+  adapter_->RegisterEncodeCompleteCallback(this);
+  ASSERT_EQ(1u, helper_->factory()->encoders().size());
+  helper_->factory()->encoders()[0]->set_supports_native_handle(true);
+  EXPECT_TRUE(adapter_->SupportsNativeHandle());
+  helper_->factory()->encoders()[0]->set_supports_native_handle(false);
+  EXPECT_FALSE(adapter_->SupportsNativeHandle());
+}
+
+TEST_F(TestSimulcastEncoderAdapterFake, SupportsImplementationName) {
+  EXPECT_STREQ("SimulcastEncoderAdapter", adapter_->ImplementationName());
+  TestVp8Simulcast::DefaultSettings(
+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
+  std::vector<const char*> encoder_names;
+  encoder_names.push_back("codec1");
+  encoder_names.push_back("codec2");
+  encoder_names.push_back("codec3");
+  helper_->factory()->SetEncoderNames(encoder_names);
+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
+  EXPECT_STREQ("SimulcastEncoderAdapter (codec1, codec2, codec3)",
+               adapter_->ImplementationName());
+}
+
+TEST_F(TestSimulcastEncoderAdapterFake,
+       SupportsNativeHandleDisabledForMultipleStreams) {
+  // TODO(pbos): Implement actual test (verify that it works) when implemented
+  // for multiple streams.
+  TestVp8Simulcast::DefaultSettings(
+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
+  codec_.numberOfSimulcastStreams = 3;
+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
+  adapter_->RegisterEncodeCompleteCallback(this);
+  ASSERT_EQ(3u, helper_->factory()->encoders().size());
+  for (MockVideoEncoder* encoder : helper_->factory()->encoders())
+    encoder->set_supports_native_handle(true);
+  EXPECT_FALSE(adapter_->SupportsNativeHandle());
+}
+
 }  // namespace testing
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
index 373a55237f..9d57dc0996 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
@@ -13,18 +13,14 @@
 namespace webrtc {
 namespace testing {
 
-class TestVp8Impl
-    : public TestVp8Simulcast {
+class TestVp8Impl : public TestVp8Simulcast {
  public:
   TestVp8Impl()
-     : TestVp8Simulcast(VP8Encoder::Create(), VP8Decoder::Create()) {}
+      : TestVp8Simulcast(VP8Encoder::Create(), VP8Decoder::Create()) {}
+
  protected:
-  virtual void SetUp() {
-    TestVp8Simulcast::SetUp();
-  }
-  virtual void TearDown() {
-    TestVp8Simulcast::TearDown();
-  }
+  virtual void SetUp() { TestVp8Simulcast::SetUp(); }
+  virtual void TearDown() { TestVp8Simulcast::TearDown(); }
 };
 
 TEST_F(TestVp8Impl, TestKeyFrameRequestsOnAllStreams) {
@@ -67,6 +63,10 @@ TEST_F(TestVp8Impl, TestSwitchingToOneOddStream) {
   TestVp8Simulcast::TestSwitchingToOneOddStream();
 }
 
+TEST_F(TestVp8Impl, TestSwitchingToOneSmallStream) {
+  TestVp8Simulcast::TestSwitchingToOneSmallStream();
+}
+
 TEST_F(TestVp8Impl, TestRPSIEncoder) {
   TestVp8Simulcast::TestRPSIEncoder();
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
index 51a3ac898e..469f8c1c66 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
@@ -14,14 +14,14 @@
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
-#include "webrtc/experiments.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
+#include "webrtc/video_frame.h"
 
 #include "gtest/gtest.h"
 
@@ -44,10 +44,8 @@ const int kMinBitrates[kNumberOfSimulcastStreams] = {50, 150, 600};
 const int kTargetBitrates[kNumberOfSimulcastStreams] = {100, 450, 1000};
 const int kDefaultTemporalLayerProfile[3] = {3, 3, 3};
 
-template<typename T> void SetExpectedValues3(T value0,
-                                             T value1,
-                                             T value2,
-                                             T* expected_values) {
+template <typename T>
+void SetExpectedValues3(T value0, T value1, T value2, T* expected_values) {
   expected_values[0] = value0;
   expected_values[1] = value1;
   expected_values[2] = value2;
@@ -55,15 +53,14 @@ template<typename T> void SetExpectedValues3(T value0,
 
 class Vp8TestEncodedImageCallback : public EncodedImageCallback {
  public:
-  Vp8TestEncodedImageCallback()
-       : picture_id_(-1) {
+  Vp8TestEncodedImageCallback() : picture_id_(-1) {
     memset(temporal_layer_, -1, sizeof(temporal_layer_));
     memset(layer_sync_, false, sizeof(layer_sync_));
   }
 
   ~Vp8TestEncodedImageCallback() {
-    delete [] encoded_key_frame_._buffer;
-    delete [] encoded_frame_._buffer;
+    delete[] encoded_key_frame_._buffer;
+    delete[] encoded_frame_._buffer;
   }
 
   virtual int32_t Encoded(const EncodedImage& encoded_image,
@@ -71,23 +68,21 @@ class Vp8TestEncodedImageCallback : public EncodedImageCallback {
                           const RTPFragmentationHeader* fragmentation) {
     // Only store the base layer.
     if (codec_specific_info->codecSpecific.VP8.simulcastIdx == 0) {
-      if (encoded_image._frameType == kKeyFrame) {
-        delete [] encoded_key_frame_._buffer;
+      if (encoded_image._frameType == kVideoFrameKey) {
+        delete[] encoded_key_frame_._buffer;
         encoded_key_frame_._buffer = new uint8_t[encoded_image._size];
         encoded_key_frame_._size = encoded_image._size;
         encoded_key_frame_._length = encoded_image._length;
-        encoded_key_frame_._frameType = kKeyFrame;
+        encoded_key_frame_._frameType = kVideoFrameKey;
         encoded_key_frame_._completeFrame = encoded_image._completeFrame;
-        memcpy(encoded_key_frame_._buffer,
-               encoded_image._buffer,
+        memcpy(encoded_key_frame_._buffer, encoded_image._buffer,
                encoded_image._length);
       } else {
-        delete [] encoded_frame_._buffer;
+        delete[] encoded_frame_._buffer;
         encoded_frame_._buffer = new uint8_t[encoded_image._size];
         encoded_frame_._size = encoded_image._size;
         encoded_frame_._length = encoded_image._length;
-        memcpy(encoded_frame_._buffer,
-               encoded_image._buffer,
+        memcpy(encoded_frame_._buffer, encoded_image._buffer,
                encoded_image._length);
       }
     }
@@ -98,8 +93,10 @@ class Vp8TestEncodedImageCallback : public EncodedImageCallback {
         codec_specific_info->codecSpecific.VP8.temporalIdx;
     return 0;
   }
-  void GetLastEncodedFrameInfo(int* picture_id, int* temporal_layer,
-                               bool* layer_sync, int stream) {
+  void GetLastEncodedFrameInfo(int* picture_id,
+                               int* temporal_layer,
+                               bool* layer_sync,
+                               int stream) {
     *picture_id = picture_id_;
     *temporal_layer = temporal_layer_[stream];
     *layer_sync = layer_sync_[stream];
@@ -121,10 +118,8 @@ class Vp8TestEncodedImageCallback : public EncodedImageCallback {
 
 class Vp8TestDecodedImageCallback : public DecodedImageCallback {
  public:
-  Vp8TestDecodedImageCallback()
-      : decoded_frames_(0) {
-  }
-  virtual int32_t Decoded(I420VideoFrame& decoded_image) {
+  Vp8TestDecodedImageCallback() : decoded_frames_(0) {}
+  int32_t Decoded(VideoFrame& decoded_image) override {
     for (int i = 0; i < decoded_image.width(); ++i) {
       EXPECT_NEAR(kColorY, decoded_image.buffer(kYPlane)[i], 1);
     }
@@ -137,9 +132,11 @@ class Vp8TestDecodedImageCallback : public DecodedImageCallback {
     decoded_frames_++;
     return 0;
   }
-  int DecodedFrames() {
-    return decoded_frames_;
+  int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override {
+    RTC_NOTREACHED();
+    return -1;
   }
+  int DecodedFrames() { return decoded_frames_; }
 
  private:
   int decoded_frames_;
@@ -162,8 +159,7 @@ class SkipEncodingUnusedStreamsTest {
     std::vector<unsigned int> configured_bitrates;
     for (std::vector<TemporalLayers*>::const_iterator it =
              spy_factory->spying_layers_.begin();
-         it != spy_factory->spying_layers_.end();
-         ++it) {
+         it != spy_factory->spying_layers_.end(); ++it) {
       configured_bitrates.push_back(
           static_cast<SpyingTemporalLayers*>(*it)->configured_bitrate_);
     }
@@ -186,8 +182,8 @@ class SkipEncodingUnusedStreamsTest {
                            int framerate,
                            vpx_codec_enc_cfg_t* cfg) override {
       configured_bitrate_ = bitrate_kbit;
-      return layers_->ConfigureBitrates(
-          bitrate_kbit, max_bitrate_kbit, framerate, cfg);
+      return layers_->ConfigureBitrates(bitrate_kbit, max_bitrate_kbit,
+                                        framerate, cfg);
     }
 
     void PopulateCodecSpecific(bool base_layer_sync,
@@ -196,12 +192,16 @@ class SkipEncodingUnusedStreamsTest {
       layers_->PopulateCodecSpecific(base_layer_sync, vp8_info, timestamp);
     }
 
-    void FrameEncoded(unsigned int size, uint32_t timestamp) override {
-      layers_->FrameEncoded(size, timestamp);
+    void FrameEncoded(unsigned int size, uint32_t timestamp, int qp) override {
+      layers_->FrameEncoded(size, timestamp, qp);
     }
 
     int CurrentLayerId() const override { return layers_->CurrentLayerId(); }
 
+    bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) override {
+      return false;
+    }
+
     int configured_bitrate_;
     TemporalLayers* layers_;
   };
@@ -225,17 +225,15 @@ class SkipEncodingUnusedStreamsTest {
 class TestVp8Simulcast : public ::testing::Test {
  public:
   TestVp8Simulcast(VP8Encoder* encoder, VP8Decoder* decoder)
-     : encoder_(encoder),
-       decoder_(decoder) {}
+      : encoder_(encoder), decoder_(decoder) {}
 
-  // Creates an I420VideoFrame from |plane_colors|.
-  static void CreateImage(I420VideoFrame* frame,
-                          int plane_colors[kNumOfPlanes]) {
+  // Creates an VideoFrame from |plane_colors|.
+  static void CreateImage(VideoFrame* frame, int plane_colors[kNumOfPlanes]) {
     for (int plane_num = 0; plane_num < kNumOfPlanes; ++plane_num) {
-      int width = (plane_num != kYPlane ? (frame->width() + 1) / 2 :
-        frame->width());
-      int height = (plane_num != kYPlane ? (frame->height() + 1) / 2 :
-        frame->height());
+      int width =
+          (plane_num != kYPlane ? (frame->width() + 1) / 2 : frame->width());
+      int height =
+          (plane_num != kYPlane ? (frame->height() + 1) / 2 : frame->height());
       PlaneType plane_type = static_cast<PlaneType>(plane_num);
       uint8_t* data = frame->buffer(plane_type);
       // Setting allocated area to zero - setting only image size to
@@ -265,24 +263,15 @@ class TestVp8Simulcast : public ::testing::Test {
     settings->height = kDefaultHeight;
     settings->numberOfSimulcastStreams = kNumberOfSimulcastStreams;
     ASSERT_EQ(3, kNumberOfSimulcastStreams);
-    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4,
-                    kMaxBitrates[0],
-                    kMinBitrates[0],
-                    kTargetBitrates[0],
-                    &settings->simulcastStream[0],
-                    temporal_layer_profile[0]);
-    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2,
-                    kMaxBitrates[1],
-                    kMinBitrates[1],
-                    kTargetBitrates[1],
-                    &settings->simulcastStream[1],
-                    temporal_layer_profile[1]);
-    ConfigureStream(kDefaultWidth, kDefaultHeight,
-                    kMaxBitrates[2],
-                    kMinBitrates[2],
-                    kTargetBitrates[2],
-                    &settings->simulcastStream[2],
-                    temporal_layer_profile[2]);
+    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, kMaxBitrates[0],
+                    kMinBitrates[0], kTargetBitrates[0],
+                    &settings->simulcastStream[0], temporal_layer_profile[0]);
+    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, kMaxBitrates[1],
+                    kMinBitrates[1], kTargetBitrates[1],
+                    &settings->simulcastStream[1], temporal_layer_profile[1]);
+    ConfigureStream(kDefaultWidth, kDefaultHeight, kMaxBitrates[2],
+                    kMinBitrates[2], kTargetBitrates[2],
+                    &settings->simulcastStream[2], temporal_layer_profile[2]);
     settings->codecSpecific.VP8.resilience = kResilientStream;
     settings->codecSpecific.VP8.denoisingOn = true;
     settings->codecSpecific.VP8.errorConcealmentOn = false;
@@ -310,9 +299,7 @@ class TestVp8Simulcast : public ::testing::Test {
   }
 
  protected:
-  virtual void SetUp() {
-    SetUpCodec(kDefaultTemporalLayerProfile);
-  }
+  virtual void SetUp() { SetUpCodec(kDefaultTemporalLayerProfile); }
 
   virtual void SetUpCodec(const int* temporal_layer_profile) {
     encoder_->RegisterEncodeCompleteCallback(&encoder_callback_);
@@ -321,14 +308,14 @@ class TestVp8Simulcast : public ::testing::Test {
     EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
     EXPECT_EQ(0, decoder_->InitDecode(&settings_, 1));
     int half_width = (kDefaultWidth + 1) / 2;
-    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight,
-                                  kDefaultWidth, half_width, half_width);
+    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight, kDefaultWidth,
+                                  half_width, half_width);
     memset(input_frame_.buffer(kYPlane), 0,
-        input_frame_.allocated_size(kYPlane));
+           input_frame_.allocated_size(kYPlane));
     memset(input_frame_.buffer(kUPlane), 0,
-        input_frame_.allocated_size(kUPlane));
+           input_frame_.allocated_size(kUPlane));
     memset(input_frame_.buffer(kVPlane), 0,
-        input_frame_.allocated_size(kVPlane));
+           input_frame_.allocated_size(kVPlane));
   }
 
   virtual void TearDown() {
@@ -336,42 +323,41 @@ class TestVp8Simulcast : public ::testing::Test {
     decoder_->Release();
   }
 
-  void ExpectStreams(VideoFrameType frame_type, int expected_video_streams) {
+  void ExpectStreams(FrameType frame_type, int expected_video_streams) {
     ASSERT_GE(expected_video_streams, 0);
     ASSERT_LE(expected_video_streams, kNumberOfSimulcastStreams);
     if (expected_video_streams >= 1) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, frame_type),
-                Field(&EncodedImage::_encodedWidth, kDefaultWidth / 4),
-                Field(&EncodedImage::_encodedHeight, kDefaultHeight / 4)), _, _)
-                  )
+      EXPECT_CALL(
+          encoder_callback_,
+          Encoded(
+              AllOf(Field(&EncodedImage::_frameType, frame_type),
+                    Field(&EncodedImage::_encodedWidth, kDefaultWidth / 4),
+                    Field(&EncodedImage::_encodedHeight, kDefaultHeight / 4)),
+              _, _))
           .Times(1)
           .WillRepeatedly(Return(0));
     }
     if (expected_video_streams >= 2) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, frame_type),
-                Field(&EncodedImage::_encodedWidth, kDefaultWidth / 2),
-                Field(&EncodedImage::_encodedHeight, kDefaultHeight / 2)), _, _)
-                  )
+      EXPECT_CALL(
+          encoder_callback_,
+          Encoded(
+              AllOf(Field(&EncodedImage::_frameType, frame_type),
+                    Field(&EncodedImage::_encodedWidth, kDefaultWidth / 2),
+                    Field(&EncodedImage::_encodedHeight, kDefaultHeight / 2)),
+              _, _))
           .Times(1)
           .WillRepeatedly(Return(0));
     }
     if (expected_video_streams >= 3) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, frame_type),
-                Field(&EncodedImage::_encodedWidth, kDefaultWidth),
-                Field(&EncodedImage::_encodedHeight, kDefaultHeight)), _, _))
+      EXPECT_CALL(
+          encoder_callback_,
+          Encoded(AllOf(Field(&EncodedImage::_frameType, frame_type),
+                        Field(&EncodedImage::_encodedWidth, kDefaultWidth),
+                        Field(&EncodedImage::_encodedHeight, kDefaultHeight)),
+                  _, _))
           .Times(1)
           .WillRepeatedly(Return(0));
     }
-    if (expected_video_streams < kNumberOfSimulcastStreams) {
-      EXPECT_CALL(encoder_callback_, Encoded(
-          AllOf(Field(&EncodedImage::_frameType, kSkipFrame),
-                Field(&EncodedImage::_length, 0)), _, _))
-          .Times(kNumberOfSimulcastStreams - expected_video_streams)
-          .WillRepeatedly(Return(0));
-    }
   }
 
   void VerifyTemporalIdxAndSyncForAllSpatialLayers(
@@ -394,34 +380,34 @@ class TestVp8Simulcast : public ::testing::Test {
   // a key frame was only requested for some of them.
   void TestKeyFrameRequestsOnAllStreams() {
     encoder_->SetRates(kMaxBitrates[2], 30);  // To get all three streams.
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, kNumberOfSimulcastStreams);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, kNumberOfSimulcastStreams);
+    ExpectStreams(kVideoFrameDelta, kNumberOfSimulcastStreams);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    frame_types[0] = kKeyFrame;
-    ExpectStreams(kKeyFrame, kNumberOfSimulcastStreams);
+    frame_types[0] = kVideoFrameKey;
+    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    std::fill(frame_types.begin(), frame_types.end(), kDeltaFrame);
-    frame_types[1] = kKeyFrame;
-    ExpectStreams(kKeyFrame, kNumberOfSimulcastStreams);
+    std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
+    frame_types[1] = kVideoFrameKey;
+    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    std::fill(frame_types.begin(), frame_types.end(), kDeltaFrame);
-    frame_types[2] = kKeyFrame;
-    ExpectStreams(kKeyFrame, kNumberOfSimulcastStreams);
+    std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
+    frame_types[2] = kVideoFrameKey;
+    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    std::fill(frame_types.begin(), frame_types.end(), kDeltaFrame);
-    ExpectStreams(kDeltaFrame, kNumberOfSimulcastStreams);
+    std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
+    ExpectStreams(kVideoFrameDelta, kNumberOfSimulcastStreams);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -429,12 +415,12 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestPaddingAllStreams() {
     // We should always encode the base layer.
     encoder_->SetRates(kMinBitrates[0] - 1, 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 1);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 1);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 1);
+    ExpectStreams(kVideoFrameDelta, 1);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -442,12 +428,12 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestPaddingTwoStreams() {
     // We have just enough to get only the first stream and padding for two.
     encoder_->SetRates(kMinBitrates[0], 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 1);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 1);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 1);
+    ExpectStreams(kVideoFrameDelta, 1);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -456,12 +442,12 @@ class TestVp8Simulcast : public ::testing::Test {
     // We are just below limit of sending second stream, so we should get
     // the first stream maxed out (at |maxBitrate|), and padding for two.
     encoder_->SetRates(kTargetBitrates[0] + kMinBitrates[1] - 1, 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 1);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 1);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 1);
+    ExpectStreams(kVideoFrameDelta, 1);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -469,12 +455,12 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestPaddingOneStream() {
     // We have just enough to send two streams, so padding for one stream.
     encoder_->SetRates(kTargetBitrates[0] + kMinBitrates[1], 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 2);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 2);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 2);
+    ExpectStreams(kVideoFrameDelta, 2);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -482,78 +468,77 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestPaddingOneStreamTwoMaxedOut() {
     // We are just below limit of sending third stream, so we should get
     // first stream's rate maxed out at |targetBitrate|, second at |maxBitrate|.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2] - 1, 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 2);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] - 1, 30);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 2);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 2);
+    ExpectStreams(kVideoFrameDelta, 2);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
 
   void TestSendAllStreams() {
     // We have just enough to send all streams.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2], 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 3);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2], 30);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 3);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 3);
+    ExpectStreams(kVideoFrameDelta, 3);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
 
   void TestDisablingStreams() {
     // We should get three media streams.
-    encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1] +
-                       kMaxBitrates[2], 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    ExpectStreams(kKeyFrame, 3);
+    encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    ExpectStreams(kVideoFrameKey, 3);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
-    ExpectStreams(kDeltaFrame, 3);
+    ExpectStreams(kVideoFrameDelta, 3);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should only get two streams and padding for one.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2] / 2, 30);
-    ExpectStreams(kDeltaFrame, 2);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
+    ExpectStreams(kVideoFrameDelta, 2);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should only get the first stream and padding for two.
     encoder_->SetRates(kTargetBitrates[0] + kMinBitrates[1] / 2, 30);
-    ExpectStreams(kDeltaFrame, 1);
+    ExpectStreams(kVideoFrameDelta, 1);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We don't have enough bitrate for the thumbnail stream, but we should get
     // it anyway with current configuration.
     encoder_->SetRates(kTargetBitrates[0] - 1, 30);
-    ExpectStreams(kDeltaFrame, 1);
+    ExpectStreams(kVideoFrameDelta, 1);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should only get two streams and padding for one.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kMinBitrates[2] / 2, 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
     // We get a key frame because a new stream is being enabled.
-    ExpectStreams(kKeyFrame, 2);
+    ExpectStreams(kVideoFrameKey, 2);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
 
     // We should get all three streams.
-    encoder_->SetRates(kTargetBitrates[0] + kTargetBitrates[1] +
-                       kTargetBitrates[2], 30);
+    encoder_->SetRates(
+        kTargetBitrates[0] + kTargetBitrates[1] + kTargetBitrates[2], 30);
     // We get a key frame because a new stream is being enabled.
-    ExpectStreams(kKeyFrame, 3);
+    ExpectStreams(kVideoFrameKey, 3);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -576,30 +561,31 @@ class TestVp8Simulcast : public ::testing::Test {
     input_frame_.CreateEmptyFrame(settings_.width, settings_.height,
                                   settings_.width, half_width, half_width);
     memset(input_frame_.buffer(kYPlane), 0,
-        input_frame_.allocated_size(kYPlane));
+           input_frame_.allocated_size(kYPlane));
     memset(input_frame_.buffer(kUPlane), 0,
-        input_frame_.allocated_size(kUPlane));
+           input_frame_.allocated_size(kUPlane));
     memset(input_frame_.buffer(kVPlane), 0,
-        input_frame_.allocated_size(kVPlane));
+           input_frame_.allocated_size(kVPlane));
 
     // The for loop above did not set the bitrate of the highest layer.
-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].
-        maxBitrate = 0;
+    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1]
+        .maxBitrate = 0;
     // The highest layer has to correspond to the non-simulcast resolution.
-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].
-        width = settings_.width;
-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].
-        height = settings_.height;
+    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].width =
+        settings_.width;
+    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].height =
+        settings_.height;
     EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
 
     // Encode one frame and verify.
     encoder_->SetRates(kMaxBitrates[0] + kMaxBitrates[1], 30);
-    std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
-                                            kDeltaFrame);
-    EXPECT_CALL(encoder_callback_, Encoded(
-        AllOf(Field(&EncodedImage::_frameType, kKeyFrame),
-              Field(&EncodedImage::_encodedWidth, width),
-              Field(&EncodedImage::_encodedHeight, height)), _, _))
+    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
+                                       kVideoFrameDelta);
+    EXPECT_CALL(encoder_callback_,
+                Encoded(AllOf(Field(&EncodedImage::_frameType, kVideoFrameKey),
+                              Field(&EncodedImage::_encodedWidth, width),
+                              Field(&EncodedImage::_encodedHeight, height)),
+                        _, _))
         .Times(1)
         .WillRepeatedly(Return(0));
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
@@ -610,27 +596,25 @@ class TestVp8Simulcast : public ::testing::Test {
     settings_.startBitrate = kMinBitrates[0];
     EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
     encoder_->SetRates(settings_.startBitrate, 30);
-    ExpectStreams(kKeyFrame, 1);
+    ExpectStreams(kVideoFrameKey, 1);
     // Resize |input_frame_| to the new resolution.
     half_width = (settings_.width + 1) / 2;
     input_frame_.CreateEmptyFrame(settings_.width, settings_.height,
                                   settings_.width, half_width, half_width);
     memset(input_frame_.buffer(kYPlane), 0,
-        input_frame_.allocated_size(kYPlane));
+           input_frame_.allocated_size(kYPlane));
     memset(input_frame_.buffer(kUPlane), 0,
-        input_frame_.allocated_size(kUPlane));
+           input_frame_.allocated_size(kUPlane));
     memset(input_frame_.buffer(kVPlane), 0,
-        input_frame_.allocated_size(kVPlane));
+           input_frame_.allocated_size(kVPlane));
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
 
-  void TestSwitchingToOneStream() {
-    SwitchingToOneStream(1024, 768);
-  }
+  void TestSwitchingToOneStream() { SwitchingToOneStream(1024, 768); }
 
-  void TestSwitchingToOneOddStream() {
-    SwitchingToOneStream(1023, 769);
-  }
+  void TestSwitchingToOneOddStream() { SwitchingToOneStream(1023, 769); }
+
+  void TestSwitchingToOneSmallStream() { SwitchingToOneStream(4, 4); }
 
   void TestRPSIEncoder() {
     Vp8TestEncodedImageCallback encoder_callback;
@@ -781,67 +765,55 @@ class TestVp8Simulcast : public ::testing::Test {
     encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
     encoder_->SetRates(kMaxBitrates[2], 30);  // To get all three streams.
 
-    int expected_temporal_idx[3] = { -1, -1, -1};
+    int expected_temporal_idx[3] = {-1, -1, -1};
     bool expected_layer_sync[3] = {false, false, false};
 
     // First frame: #0.
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #1.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #2.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(1, 1, 1, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #3.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #4.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #5.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
   }
 
   // Test the layer pattern and sync flag for various spatial-temporal patterns.
@@ -853,7 +825,7 @@ class TestVp8Simulcast : public ::testing::Test {
   // 3rd stream: -1, -1, -1, -1, ....
   // Regarding the 3rd stream, note that a stream/encoder with 1 temporal layer
   // should always have temporal layer idx set to kNoTemporalIdx = -1.
-  // Since CodecSpecificInfoVP8.temporalIdx is uint8, this will wrap to 255.
+  // Since CodecSpecificInfoVP8.temporalIdx is uint8_t, this will wrap to 255.
   // TODO(marpan): Although this seems safe for now, we should fix this.
   void TestSpatioTemporalLayers321PatternEncoder() {
     int temporal_layer_profile[3] = {3, 2, 1};
@@ -862,67 +834,55 @@ class TestVp8Simulcast : public ::testing::Test {
     encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
     encoder_->SetRates(kMaxBitrates[2], 30);  // To get all three streams.
 
-    int expected_temporal_idx[3] = { -1, -1, -1};
+    int expected_temporal_idx[3] = {-1, -1, -1};
     bool expected_layer_sync[3] = {false, false, false};
 
     // First frame: #0.
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #1.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #2.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(1, 0, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(true, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #3.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #4.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
 
     // Next frame: #5.
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
     SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
-    VerifyTemporalIdxAndSyncForAllSpatialLayers(&encoder_callback,
-                                                expected_temporal_idx,
-                                                expected_layer_sync,
-                                                3);
+    VerifyTemporalIdxAndSyncForAllSpatialLayers(
+        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
   }
 
   void TestStrideEncodeDecode() {
@@ -936,8 +896,8 @@ class TestVp8Simulcast : public ::testing::Test {
     // 1. stride > width 2. stride_y != stride_uv/2
     int stride_y = kDefaultWidth + 20;
     int stride_uv = ((kDefaultWidth + 1) / 2) + 5;
-    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight,
-                                 stride_y, stride_uv, stride_uv);
+    input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight, stride_y,
+                                  stride_uv, stride_uv);
     // Set color.
     int plane_offset[kNumOfPlanes];
     plane_offset[kYPlane] = kColorY;
@@ -967,10 +927,9 @@ class TestVp8Simulcast : public ::testing::Test {
   void TestSkipEncodingUnusedStreams() {
     SkipEncodingUnusedStreamsTest test;
     std::vector<unsigned int> configured_bitrate =
-        test.RunTest(encoder_.get(),
-                     &settings_,
-                     1);    // Target bit rate 1, to force all streams but the
-                            // base one to be exceeding bandwidth constraints.
+        test.RunTest(encoder_.get(), &settings_,
+                     1);  // Target bit rate 1, to force all streams but the
+                          // base one to be exceeding bandwidth constraints.
     EXPECT_EQ(static_cast<size_t>(kNumberOfSimulcastStreams),
               configured_bitrate.size());
 
@@ -979,8 +938,7 @@ class TestVp8Simulcast : public ::testing::Test {
     int stream = 0;
     for (std::vector<unsigned int>::const_iterator it =
              configured_bitrate.begin();
-         it != configured_bitrate.end();
-         ++it) {
+         it != configured_bitrate.end(); ++it) {
       if (stream == 0) {
         EXPECT_EQ(min_bitrate, *it);
       } else {
@@ -995,7 +953,7 @@ class TestVp8Simulcast : public ::testing::Test {
   rtc::scoped_ptr<VP8Decoder> decoder_;
   MockDecodedImageCallback decoder_callback_;
   VideoCodec settings_;
-  I420VideoFrame input_frame_;
+  VideoFrame input_frame_;
 };
 
 }  // namespace testing
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
index b3b73a6508..47112c64aa 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
@@ -14,7 +14,8 @@
 
 #include "vpx/vpx_encoder.h"
 
-#include "webrtc/common_video/interface/video_image.h"
+#include "webrtc/common.h"
+#include "webrtc/common_video/include/video_image.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -30,6 +31,8 @@ class TemporalLayers {
     virtual ~Factory() {}
     virtual TemporalLayers* Create(int temporal_layers,
                                    uint8_t initial_tl0_pic_idx) const;
+    static const ConfigOptionID identifier =
+        ConfigOptionID::kTemporalLayersFactory;
   };
 
   virtual ~TemporalLayers() {}
@@ -47,9 +50,11 @@ class TemporalLayers {
                                      CodecSpecificInfoVP8* vp8_info,
                                      uint32_t timestamp) = 0;
 
-  virtual void FrameEncoded(unsigned int size, uint32_t timestamp) = 0;
+  virtual void FrameEncoded(unsigned int size, uint32_t timestamp, int qp) = 0;
 
   virtual int CurrentLayerId() const = 0;
+
+  virtual bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) = 0;
 };
 
 // Factory for a temporal layers strategy that adaptively changes the number of
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index 43fc9c8e8a..c3d77da063 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -11,12 +11,12 @@
 #include <stdio.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
 
@@ -76,13 +76,17 @@ bool Vp8UnitTestEncodeCompleteCallback::EncodeComplete() {
 
 class Vp8UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback {
  public:
-  explicit Vp8UnitTestDecodeCompleteCallback(I420VideoFrame* frame)
+  explicit Vp8UnitTestDecodeCompleteCallback(VideoFrame* frame)
       : decoded_frame_(frame), decode_complete(false) {}
-  int Decoded(webrtc::I420VideoFrame& frame);
+  int32_t Decoded(VideoFrame& frame) override;
+  int32_t Decoded(VideoFrame& frame, int64_t decode_time_ms) override {
+    RTC_NOTREACHED();
+    return -1;
+  }
   bool DecodeComplete();
 
  private:
-  I420VideoFrame* decoded_frame_;
+  VideoFrame* decoded_frame_;
   bool decode_complete;
 };
 
@@ -94,7 +98,7 @@ bool Vp8UnitTestDecodeCompleteCallback::DecodeComplete() {
   return false;
 }
 
-int Vp8UnitTestDecodeCompleteCallback::Decoded(I420VideoFrame& image) {
+int Vp8UnitTestDecodeCompleteCallback::Decoded(VideoFrame& image) {
   decoded_frame_->CopyFrame(image);
   decode_complete = true;
   return 0;
@@ -181,11 +185,11 @@ class TestVp8Impl : public ::testing::Test {
   rtc::scoped_ptr<Vp8UnitTestDecodeCompleteCallback> decode_complete_callback_;
   rtc::scoped_ptr<uint8_t[]> source_buffer_;
   FILE* source_file_;
-  I420VideoFrame input_frame_;
+  VideoFrame input_frame_;
   rtc::scoped_ptr<VideoEncoder> encoder_;
   rtc::scoped_ptr<VideoDecoder> decoder_;
   EncodedImage encoded_frame_;
-  I420VideoFrame decoded_frame_;
+  VideoFrame decoded_frame_;
   size_t length_source_frame_;
   VideoCodec codec_inst_;
 };
@@ -216,12 +220,17 @@ TEST_F(TestVp8Impl, EncoderParameterTest) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_inst_, 1));
 }
 
-TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AlignedStrideEncodeDecode DISABLED_AlignedStrideEncodeDecode
+#else
+#define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
+#endif
+TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
   SetUpEncodeDecode();
   encoder_->Encode(input_frame_, NULL, NULL);
   EXPECT_GT(WaitForEncodedFrame(), 0u);
   // First frame should be a key frame.
-  encoded_frame_._frameType = kKeyFrame;
+  encoded_frame_._frameType = kVideoFrameKey;
   encoded_frame_.ntp_time_ms_ = kTestNtpTimeMs;
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame_, false, NULL));
@@ -232,7 +241,12 @@ TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(AlignedStrideEncodeDecode)) {
   EXPECT_EQ(kTestNtpTimeMs, decoded_frame_.ntp_time_ms());
 }
 
-TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(DecodeWithACompleteKeyFrame)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_DecodeWithACompleteKeyFrame DISABLED_DecodeWithACompleteKeyFrame
+#else
+#define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
+#endif
+TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
   SetUpEncodeDecode();
   encoder_->Encode(input_frame_, NULL, NULL);
   EXPECT_GT(WaitForEncodedFrame(), 0u);
@@ -241,12 +255,12 @@ TEST_F(TestVp8Impl, DISABLED_ON_ANDROID(DecodeWithACompleteKeyFrame)) {
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
             decoder_->Decode(encoded_frame_, false, NULL));
   // Setting complete back to true. Forcing a delta frame.
-  encoded_frame_._frameType = kDeltaFrame;
+  encoded_frame_._frameType = kVideoFrameDelta;
   encoded_frame_._completeFrame = true;
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
             decoder_->Decode(encoded_frame_, false, NULL));
   // Now setting a key frame.
-  encoded_frame_._frameType = kKeyFrame;
+  encoded_frame_._frameType = kVideoFrameKey;
   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
             decoder_->Decode(encoded_frame_, false, NULL));
   EXPECT_GT(I420PSNR(&input_frame_, &decoded_frame_), 36);
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8.gyp b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8.gyp
index 2e06f21c1e..2168a4c08c 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8.gyp
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8.gyp
@@ -26,7 +26,7 @@
       'conditions': [
         ['build_libvpx==1', {
           'dependencies': [
-            '<(libvpx_dir)/libvpx.gyp:libvpx',
+            '<(libvpx_dir)/libvpx.gyp:libvpx_new',
           ],
         },{
           'link_settings': {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h
index 84745ea5a1..52f8aa30b8 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_factory.h
@@ -32,4 +32,3 @@ class VP8EncoderFactoryConfig {
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_FACTORY_H_
-
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
index b6a9fc5962..f936bb0328 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
@@ -16,21 +16,20 @@
 #include <algorithm>
 
 // NOTE(ajm): Path provided by gyp.
-#include "libyuv/scale.h"  // NOLINT
+#include "libyuv/scale.h"    // NOLINT
 #include "libyuv/convert.h"  // NOLINT
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/experiments.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
@@ -60,12 +59,44 @@ int GCD(int a, int b) {
   return b;
 }
 
-uint32_t SumStreamTargetBitrate(int streams, const VideoCodec& codec) {
-  uint32_t bitrate_sum = 0;
-  for (int i = 0; i < streams; ++i) {
-    bitrate_sum += codec.simulcastStream[i].targetBitrate;
+std::vector<int> GetStreamBitratesKbps(const VideoCodec& codec,
+                                       int bitrate_to_allocate_kbps) {
+  if (codec.numberOfSimulcastStreams <= 1) {
+    return std::vector<int>(1, bitrate_to_allocate_kbps);
   }
-  return bitrate_sum;
+
+  std::vector<int> bitrates_kbps(codec.numberOfSimulcastStreams);
+  // Allocate min -> target bitrates as long as we have bitrate to spend.
+  size_t last_active_stream = 0;
+  for (size_t i = 0; i < static_cast<size_t>(codec.numberOfSimulcastStreams) &&
+                     bitrate_to_allocate_kbps >=
+                         static_cast<int>(codec.simulcastStream[i].minBitrate);
+       ++i) {
+    last_active_stream = i;
+    int allocated_bitrate_kbps =
+        std::min(static_cast<int>(codec.simulcastStream[i].targetBitrate),
+                 bitrate_to_allocate_kbps);
+    bitrates_kbps[i] = allocated_bitrate_kbps;
+    bitrate_to_allocate_kbps -= allocated_bitrate_kbps;
+  }
+
+  // Spend additional bits on the highest-quality active layer, up to max
+  // bitrate.
+  // TODO(pbos): Consider spending additional bits on last_active_stream-1 down
+  // to 0 and not just the top layer when we have additional bitrate to spend.
+  int allocated_bitrate_kbps = std::min(
+      static_cast<int>(codec.simulcastStream[last_active_stream].maxBitrate -
+                       bitrates_kbps[last_active_stream]),
+      bitrate_to_allocate_kbps);
+  bitrates_kbps[last_active_stream] += allocated_bitrate_kbps;
+  bitrate_to_allocate_kbps -= allocated_bitrate_kbps;
+
+  // Make sure we can always send something. Suspending below min bitrate is
+  // controlled outside the codec implementation and is not overriden by this.
+  if (bitrates_kbps[0] < static_cast<int>(codec.simulcastStream[0].minBitrate))
+    bitrates_kbps[0] = static_cast<int>(codec.simulcastStream[0].minBitrate);
+
+  return bitrates_kbps;
 }
 
 uint32_t SumStreamMaxBitrate(int streams, const VideoCodec& codec) {
@@ -99,6 +130,15 @@ bool ValidSimulcastResolutions(const VideoCodec& codec, int num_streams) {
   }
   return true;
 }
+
+int NumStreamsDisabled(const std::vector<bool>& streams) {
+  int num_disabled = 0;
+  for (bool stream : streams) {
+    if (!stream)
+      ++num_disabled;
+  }
+  return num_disabled;
+}
 }  // namespace
 
 const float kTl1MaxTimeToDropFrames = 20.0f;
@@ -116,7 +156,8 @@ VP8EncoderImpl::VP8EncoderImpl()
       down_scale_bitrate_(0),
       tl0_frame_dropper_(),
       tl1_frame_dropper_(kTl1MaxTimeToDropFrames),
-      key_frame_request_(kMaxSimulcastStreams, false) {
+      key_frame_request_(kMaxSimulcastStreams, false),
+      quality_scaler_enabled_(false) {
   uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
   srand(seed);
 
@@ -141,7 +182,7 @@ int VP8EncoderImpl::Release() {
 
   while (!encoded_images_.empty()) {
     EncodedImage& image = encoded_images_.back();
-    delete [] image._buffer;
+    delete[] image._buffer;
     encoded_images_.pop_back();
   }
   while (!encoders_.empty()) {
@@ -167,7 +208,7 @@ int VP8EncoderImpl::Release() {
 }
 
 int VP8EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
-                                     uint32_t new_framerate) {
+                             uint32_t new_framerate) {
   if (!inited_) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
@@ -224,20 +265,14 @@ int VP8EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
     }
   }
 
-  bool send_stream = true;
-  int stream_bitrate = 0;
+  std::vector<int> stream_bitrates =
+      GetStreamBitratesKbps(codec_, new_bitrate_kbit);
   size_t stream_idx = encoders_.size() - 1;
   for (size_t i = 0; i < encoders_.size(); ++i, --stream_idx) {
-    if (encoders_.size() == 1) {
-      stream_bitrate = new_bitrate_kbit;
-    } else {
-      stream_bitrate = GetStreamBitrate(stream_idx,
-                                        new_bitrate_kbit,
-                                        &send_stream);
-      SetStreamState(send_stream, stream_idx);
-    }
+    if (encoders_.size() > 1)
+      SetStreamState(stream_bitrates[stream_idx] > 0, stream_idx);
 
-    unsigned int target_bitrate = stream_bitrate;
+    unsigned int target_bitrate = stream_bitrates[stream_idx];
     unsigned int max_bitrate = codec_.maxBitrate;
     int framerate = new_framerate;
     // TODO(holmer): This is a temporary hack for screensharing, where we
@@ -251,13 +286,10 @@ int VP8EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
       int tl0_bitrate = std::min(codec_.targetBitrate, target_bitrate);
       max_bitrate = std::min(codec_.maxBitrate, target_bitrate);
       target_bitrate = tl0_bitrate;
-      framerate = -1;
     }
     configurations_[i].rc_target_bitrate = target_bitrate;
-    temporal_layers_[stream_idx]->ConfigureBitrates(target_bitrate,
-                                                    max_bitrate,
-                                                    framerate,
-                                                    &configurations_[i]);
+    temporal_layers_[stream_idx]->ConfigureBitrates(
+        target_bitrate, max_bitrate, framerate, &configurations_[i]);
     if (vpx_codec_enc_config_set(&encoders_[i], &configurations_[i])) {
       return WEBRTC_VIDEO_CODEC_ERROR;
     }
@@ -266,44 +298,8 @@ int VP8EncoderImpl::SetRates(uint32_t new_bitrate_kbit,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int VP8EncoderImpl::GetStreamBitrate(int stream_idx,
-                                             uint32_t new_bitrate_kbit,
-                                             bool* send_stream) const {
-  // The bitrate needed to start sending this stream is given by the
-  // minimum bitrate allowed for encoding this stream, plus the sum target
-  // rates of all lower streams.
-  uint32_t sum_target_lower_streams = (stream_idx == 0) ? 0 :
-      SumStreamTargetBitrate(stream_idx, codec_);
-  uint32_t bitrate_to_send_this_layer =
-      codec_.simulcastStream[stream_idx].minBitrate + sum_target_lower_streams;
-  if (new_bitrate_kbit >= bitrate_to_send_this_layer) {
-    // We have enough bandwidth to send this stream.
-    *send_stream = true;
-    // Bitrate for this stream is the new bitrate (|new_bitrate_kbit|) minus the
-    // sum target rates of the lower streams, and capped to a maximum bitrate.
-    // The maximum cap depends on whether we send the next higher stream.
-    // If we will be sending the next higher stream, |max_rate| is given by
-    // current stream's |targetBitrate|, otherwise it's capped by |maxBitrate|.
-    if (stream_idx < codec_.numberOfSimulcastStreams - 1) {
-      uint32_t max_rate = codec_.simulcastStream[stream_idx].maxBitrate;
-      if (new_bitrate_kbit >= SumStreamTargetBitrate(stream_idx + 1, codec_) +
-          codec_.simulcastStream[stream_idx + 1].minBitrate) {
-        max_rate = codec_.simulcastStream[stream_idx].targetBitrate;
-      }
-      return std::min(new_bitrate_kbit - sum_target_lower_streams, max_rate);
-    } else {
-        // For the highest stream (highest resolution), the |targetBitRate| and
-        // |maxBitrate| are not used. Any excess bitrate (above the targets of
-        // all lower streams) is given to this (highest resolution) stream.
-        return new_bitrate_kbit - sum_target_lower_streams;
-    }
-  } else {
-    // Not enough bitrate for this stream.
-    // Return our max bitrate of |stream_idx| - 1, but we don't send it. We need
-    // to keep this resolution coding in order for the multi-encoder to work.
-    *send_stream = false;
-    return 0;
-  }
+const char* VP8EncoderImpl::ImplementationName() const {
+  return "libvpx";
 }
 
 void VP8EncoderImpl::SetStreamState(bool send_stream,
@@ -316,8 +312,8 @@ void VP8EncoderImpl::SetStreamState(bool send_stream,
 }
 
 void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
-                                                 int num_temporal_layers,
-                                                 const VideoCodec& codec) {
+                                         int num_temporal_layers,
+                                         const VideoCodec& codec) {
   const Config default_options;
   const TemporalLayers::Factory& tl_factory =
       (codec.extra_options ? codec.extra_options : &default_options)
@@ -325,10 +321,8 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
   if (num_streams == 1) {
     if (codec.mode == kScreensharing) {
       // Special mode when screensharing on a single stream.
-      temporal_layers_.push_back(new ScreenshareLayers(num_temporal_layers,
-                                                       rand(),
-                                                       &tl0_frame_dropper_,
-                                                       &tl1_frame_dropper_));
+      temporal_layers_.push_back(
+          new ScreenshareLayers(num_temporal_layers, rand()));
     } else {
       temporal_layers_.push_back(
           tl_factory.Create(num_temporal_layers, rand()));
@@ -337,15 +331,16 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
     for (int i = 0; i < num_streams; ++i) {
       // TODO(andresp): crash if layers is invalid.
       int layers = codec.simulcastStream[i].numberOfTemporalLayers;
-      if (layers < 1) layers = 1;
+      if (layers < 1)
+        layers = 1;
       temporal_layers_.push_back(tl_factory.Create(layers, rand()));
     }
   }
 }
 
 int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
-                                       int number_of_cores,
-                                       size_t /*maxPayloadSize */) {
+                               int number_of_cores,
+                               size_t /*maxPayloadSize */) {
   if (inst == NULL) {
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
@@ -382,12 +377,13 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
 
-  int num_temporal_layers = doing_simulcast ?
-      inst->simulcastStream[0].numberOfTemporalLayers :
-      inst->codecSpecific.VP8.numberOfTemporalLayers;
+  int num_temporal_layers =
+      doing_simulcast ? inst->simulcastStream[0].numberOfTemporalLayers
+                      : inst->codecSpecific.VP8.numberOfTemporalLayers;
 
   // TODO(andresp): crash if num temporal layers is bananas.
-  if (num_temporal_layers < 1) num_temporal_layers = 1;
+  if (num_temporal_layers < 1)
+    num_temporal_layers = 1;
   SetupTemporalLayers(number_of_streams, num_temporal_layers, *inst);
 
   feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
@@ -417,7 +413,7 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
   int idx = number_of_streams - 1;
   for (int i = 0; i < (number_of_streams - 1); ++i, --idx) {
     int gcd = GCD(inst->simulcastStream[idx].width,
-                  inst->simulcastStream[idx-1].width);
+                  inst->simulcastStream[idx - 1].width);
     downsampling_factors_[i].num = inst->simulcastStream[idx].width / gcd;
     downsampling_factors_[i].den = inst->simulcastStream[idx - 1].width / gcd;
     send_stream_[i] = false;
@@ -429,11 +425,11 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
   }
   for (int i = 0; i < number_of_streams; ++i) {
     // Random start, 16 bits is enough.
-    picture_id_[i] = static_cast<uint16_t>(rand()) & 0x7FFF;
+    picture_id_[i] = static_cast<uint16_t>(rand()) & 0x7FFF;  // NOLINT
     last_key_frame_picture_id_[i] = -1;
     // allocate memory for encoded image
     if (encoded_images_[i]._buffer != NULL) {
-      delete [] encoded_images_[i]._buffer;
+      delete[] encoded_images_[i]._buffer;
     }
     // Reserve 100 extra bytes for overhead at small resolutions.
     encoded_images_[i]._size = CalcBufferSize(kI420, codec_.width, codec_.height)
@@ -442,8 +438,8 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
     encoded_images_[i]._completeFrame = true;
   }
   // populate encoder configuration with default values
-  if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(),
-                                   &configurations_[0], 0)) {
+  if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), &configurations_[0],
+                                   0)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
   // setting the time base of the codec
@@ -467,8 +463,8 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
       break;
     case kResilientFrames:
 #ifdef INDEPENDENT_PARTITIONS
-      configurations_[0]-g_error_resilient = VPX_ERROR_RESILIENT_DEFAULT |
-      VPX_ERROR_RESILIENT_PARTITIONS;
+      configurations_[0] - g_error_resilient =
+          VPX_ERROR_RESILIENT_DEFAULT | VPX_ERROR_RESILIENT_PARTITIONS;
       break;
 #else
       return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;  // Not supported
@@ -544,36 +540,31 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
 
   // Determine number of threads based on the image size and #cores.
   // TODO(fbarchard): Consider number of Simulcast layers.
-  configurations_[0].g_threads = NumberOfThreads(configurations_[0].g_w,
-                                                 configurations_[0].g_h,
-                                                 number_of_cores);
+  configurations_[0].g_threads = NumberOfThreads(
+      configurations_[0].g_w, configurations_[0].g_h, number_of_cores);
 
   // Creating a wrapper to the image - setting image data to NULL.
   // Actual pointer will be set in encode. Setting align to 1, as it
   // is meaningless (no memory allocation is done here).
-  vpx_img_wrap(&raw_images_[0], VPX_IMG_FMT_I420, inst->width, inst->height,
-               1, NULL);
+  vpx_img_wrap(&raw_images_[0], VPX_IMG_FMT_I420, inst->width, inst->height, 1,
+               NULL);
 
   if (encoders_.size() == 1) {
     configurations_[0].rc_target_bitrate = inst->startBitrate;
-    temporal_layers_[0]->ConfigureBitrates(inst->startBitrate,
-                                           inst->maxBitrate,
+    temporal_layers_[0]->ConfigureBitrates(inst->startBitrate, inst->maxBitrate,
                                            inst->maxFramerate,
                                            &configurations_[0]);
   } else {
     // Note the order we use is different from webm, we have lowest resolution
     // at position 0 and they have highest resolution at position 0.
     int stream_idx = encoders_.size() - 1;
-    bool send_stream = true;
-    int stream_bitrate = GetStreamBitrate(stream_idx,
-                                          inst->startBitrate,
-                                          &send_stream);
-    SetStreamState(send_stream, stream_idx);
-    configurations_[0].rc_target_bitrate = stream_bitrate;
-    temporal_layers_[stream_idx]->ConfigureBitrates(stream_bitrate,
-                                                    inst->maxBitrate,
-                                                    inst->maxFramerate,
-                                                    &configurations_[0]);
+    std::vector<int> stream_bitrates =
+        GetStreamBitratesKbps(codec_, inst->startBitrate);
+    SetStreamState(stream_bitrates[stream_idx] > 0, stream_idx);
+    configurations_[0].rc_target_bitrate = stream_bitrates[stream_idx];
+    temporal_layers_[stream_idx]->ConfigureBitrates(
+        stream_bitrates[stream_idx], inst->maxBitrate, inst->maxFramerate,
+        &configurations_[0]);
     --stream_idx;
     for (size_t i = 1; i < encoders_.size(); ++i, --stream_idx) {
       memcpy(&configurations_[i], &configurations_[0],
@@ -591,27 +582,33 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
       vpx_img_alloc(&raw_images_[i], VPX_IMG_FMT_I420,
                     inst->simulcastStream[stream_idx].width,
                     inst->simulcastStream[stream_idx].height, kVp832ByteAlign);
-      int stream_bitrate = GetStreamBitrate(stream_idx,
-                                            inst->startBitrate,
-                                            &send_stream);
-      SetStreamState(send_stream, stream_idx);
-      configurations_[i].rc_target_bitrate = stream_bitrate;
-      temporal_layers_[stream_idx]->ConfigureBitrates(stream_bitrate,
-                                                      inst->maxBitrate,
-                                                      inst->maxFramerate,
-                                                      &configurations_[i]);
+      SetStreamState(stream_bitrates[stream_idx] > 0, stream_idx);
+      configurations_[i].rc_target_bitrate = stream_bitrates[stream_idx];
+      temporal_layers_[stream_idx]->ConfigureBitrates(
+          stream_bitrates[stream_idx], inst->maxBitrate, inst->maxFramerate,
+          &configurations_[i]);
     }
   }
 
   rps_.Init();
-  quality_scaler_.Init(codec_.qpMax);
+  // Disable both high-QP limits and framedropping. Both are handled by libvpx
+  // internally.
+  const int kDisabledBadQpThreshold = 64;
+  quality_scaler_.Init(codec_.qpMax / QualityScaler::kDefaultLowQpDenominator,
+                       kDisabledBadQpThreshold, false);
   quality_scaler_.ReportFramerate(codec_.maxFramerate);
 
+  // Only apply scaling to improve for single-layer streams. The scaling metrics
+  // use frame drops as a signal and is only applicable when we drop frames.
+  quality_scaler_enabled_ = encoders_.size() == 1 &&
+                            configurations_[0].rc_dropframe_thresh > 0 &&
+                            codec_.codecSpecific.VP8.automaticResizeOn;
+
   return InitAndSetControlSettings();
 }
 
 int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
-#if defined(WEBRTC_ARCH_ARM)
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
   // On mobile platform, always set to -12 to leverage between cpu usage
   // and video quality.
   return -12;
@@ -651,20 +648,15 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
   flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
 
   if (encoders_.size() > 1) {
-    int error = vpx_codec_enc_init_multi(&encoders_[0],
-                                 vpx_codec_vp8_cx(),
-                                 &configurations_[0],
-                                 encoders_.size(),
-                                 flags,
-                                 &downsampling_factors_[0]);
+    int error = vpx_codec_enc_init_multi(&encoders_[0], vpx_codec_vp8_cx(),
+                                         &configurations_[0], encoders_.size(),
+                                         flags, &downsampling_factors_[0]);
     if (error) {
       return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
     }
   } else {
-    if (vpx_codec_enc_init(&encoders_[0],
-                           vpx_codec_vp8_cx(),
-                           &configurations_[0],
-                           flags)) {
+    if (vpx_codec_enc_init(&encoders_[0], vpx_codec_vp8_cx(),
+                           &configurations_[0], flags)) {
       return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
     }
   }
@@ -676,20 +668,21 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
   // when encoding lower resolution streams. Would it work with the
   // multi-res encoding feature?
   denoiserState denoiser_state = kDenoiserOnYOnly;
-#ifdef WEBRTC_ARCH_ARM
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
   denoiser_state = kDenoiserOnYOnly;
 #else
   denoiser_state = kDenoiserOnAdaptive;
 #endif
-  vpx_codec_control(&encoders_[0], VP8E_SET_NOISE_SENSITIVITY,
-                    codec_.codecSpecific.VP8.denoisingOn ?
-                    denoiser_state : kDenoiserOff);
+  vpx_codec_control(
+      &encoders_[0], VP8E_SET_NOISE_SENSITIVITY,
+      codec_.codecSpecific.VP8.denoisingOn ? denoiser_state : kDenoiserOff);
   if (encoders_.size() > 2) {
-    vpx_codec_control(&encoders_[1], VP8E_SET_NOISE_SENSITIVITY,
-                      codec_.codecSpecific.VP8.denoisingOn ?
-                      denoiser_state : kDenoiserOff);
+    vpx_codec_control(
+        &encoders_[1], VP8E_SET_NOISE_SENSITIVITY,
+        codec_.codecSpecific.VP8.denoisingOn ? denoiser_state : kDenoiserOff);
   }
   for (size_t i = 0; i < encoders_.size(); ++i) {
+    // Allow more screen content to be detected as static.
     vpx_codec_control(&(encoders_[i]), VP8E_SET_STATIC_THRESHOLD,
                       codec_.mode == kScreensharing ? 300 : 1);
     vpx_codec_control(&(encoders_[i]), VP8E_SET_CPUUSED, cpu_speed_[i]);
@@ -697,8 +690,10 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
                       static_cast<vp8e_token_partitions>(token_partitions_));
     vpx_codec_control(&(encoders_[i]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
                       rc_max_intra_target_);
+    // VP8E_SET_SCREEN_CONTENT_MODE 2 = screen content with more aggressive
+    // rate control (drop frames on large target bitrate overshoot)
     vpx_codec_control(&(encoders_[i]), VP8E_SET_SCREEN_CONTENT_MODE,
-                      codec_.mode == kScreensharing);
+                      codec_.mode == kScreensharing ? 2 : 0);
   }
   inited_ = true;
   return WEBRTC_VIDEO_CODEC_OK;
@@ -717,32 +712,23 @@ uint32_t VP8EncoderImpl::MaxIntraTarget(uint32_t optimalBuffersize) {
 
   // Don't go below 3 times the per frame bandwidth.
   const uint32_t minIntraTh = 300;
-  return (targetPct < minIntraTh) ? minIntraTh: targetPct;
+  return (targetPct < minIntraTh) ? minIntraTh : targetPct;
 }
 
-int VP8EncoderImpl::Encode(
-    const I420VideoFrame& frame,
-    const CodecSpecificInfo* codec_specific_info,
-    const std::vector<VideoFrameType>* frame_types) {
-  TRACE_EVENT1("webrtc", "VP8::Encode", "timestamp", frame.timestamp());
-
-  if (!inited_) {
+int VP8EncoderImpl::Encode(const VideoFrame& frame,
+                           const CodecSpecificInfo* codec_specific_info,
+                           const std::vector<FrameType>* frame_types) {
+  if (!inited_)
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
-  }
-  if (frame.IsZeroSize()) {
+  if (frame.IsZeroSize())
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-  }
-  if (encoded_complete_callback_ == NULL) {
+  if (encoded_complete_callback_ == NULL)
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
-  }
 
-  // Only apply scaling to improve for single-layer streams. The scaling metrics
-  // use framedrops as a signal and is only applicable when we drop frames.
-  const bool use_quality_scaler = encoders_.size() == 1 &&
-                                  configurations_[0].rc_dropframe_thresh > 0 &&
-                                  codec_.codecSpecific.VP8.automaticResizeOn;
-  const I420VideoFrame& input_image =
-      use_quality_scaler ? quality_scaler_.GetScaledFrame(frame) : frame;
+  if (quality_scaler_enabled_)
+    quality_scaler_.OnEncodeFrame(frame);
+  const VideoFrame& input_image =
+      quality_scaler_enabled_ ? quality_scaler_.GetScaledFrame(frame) : frame;
 
   if (input_image.width() != codec_.width ||
       input_image.height() != codec_.height) {
@@ -755,17 +741,17 @@ int VP8EncoderImpl::Encode(
   // |raw_images_[0]|, the resolution of these frames must match. Note that
   // |input_image| might be scaled from |frame|. In that case, the resolution of
   // |raw_images_[0]| should have been updated in UpdateCodecFrameSize.
-  DCHECK_EQ(input_image.width(), static_cast<int>(raw_images_[0].d_w));
-  DCHECK_EQ(input_image.height(), static_cast<int>(raw_images_[0].d_h));
+  RTC_DCHECK_EQ(input_image.width(), static_cast<int>(raw_images_[0].d_w));
+  RTC_DCHECK_EQ(input_image.height(), static_cast<int>(raw_images_[0].d_h));
 
   // Image in vpx_image_t format.
   // Input image is const. VP8's raw image is not defined as const.
   raw_images_[0].planes[VPX_PLANE_Y] =
-     const_cast<uint8_t*>(input_image.buffer(kYPlane));
+      const_cast<uint8_t*>(input_image.buffer(kYPlane));
   raw_images_[0].planes[VPX_PLANE_U] =
-     const_cast<uint8_t*>(input_image.buffer(kUPlane));
+      const_cast<uint8_t*>(input_image.buffer(kUPlane));
   raw_images_[0].planes[VPX_PLANE_V] =
-     const_cast<uint8_t*>(input_image.buffer(kVPlane));
+      const_cast<uint8_t*>(input_image.buffer(kVPlane));
 
   raw_images_[0].stride[VPX_PLANE_Y] = input_image.stride(kYPlane);
   raw_images_[0].stride[VPX_PLANE_U] = input_image.stride(kUPlane);
@@ -774,17 +760,17 @@ int VP8EncoderImpl::Encode(
   for (size_t i = 1; i < encoders_.size(); ++i) {
     // Scale the image down a number of times by downsampling factor
     libyuv::I420Scale(
-        raw_images_[i-1].planes[VPX_PLANE_Y],
-        raw_images_[i-1].stride[VPX_PLANE_Y],
-        raw_images_[i-1].planes[VPX_PLANE_U],
-        raw_images_[i-1].stride[VPX_PLANE_U],
-        raw_images_[i-1].planes[VPX_PLANE_V],
-        raw_images_[i-1].stride[VPX_PLANE_V],
-        raw_images_[i-1].d_w, raw_images_[i-1].d_h,
-        raw_images_[i].planes[VPX_PLANE_Y], raw_images_[i].stride[VPX_PLANE_Y],
-        raw_images_[i].planes[VPX_PLANE_U], raw_images_[i].stride[VPX_PLANE_U],
-        raw_images_[i].planes[VPX_PLANE_V], raw_images_[i].stride[VPX_PLANE_V],
-        raw_images_[i].d_w, raw_images_[i].d_h, libyuv::kFilterBilinear);
+        raw_images_[i - 1].planes[VPX_PLANE_Y],
+        raw_images_[i - 1].stride[VPX_PLANE_Y],
+        raw_images_[i - 1].planes[VPX_PLANE_U],
+        raw_images_[i - 1].stride[VPX_PLANE_U],
+        raw_images_[i - 1].planes[VPX_PLANE_V],
+        raw_images_[i - 1].stride[VPX_PLANE_V], raw_images_[i - 1].d_w,
+        raw_images_[i - 1].d_h, raw_images_[i].planes[VPX_PLANE_Y],
+        raw_images_[i].stride[VPX_PLANE_Y], raw_images_[i].planes[VPX_PLANE_U],
+        raw_images_[i].stride[VPX_PLANE_U], raw_images_[i].planes[VPX_PLANE_V],
+        raw_images_[i].stride[VPX_PLANE_V], raw_images_[i].d_w,
+        raw_images_[i].d_h, libyuv::kFilterBilinear);
   }
   vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
   for (size_t i = 0; i < encoders_.size(); ++i) {
@@ -806,7 +792,7 @@ int VP8EncoderImpl::Encode(
   if (!send_key_frame && frame_types) {
     for (size_t i = 0; i < frame_types->size() && i < send_stream_.size();
          ++i) {
-      if ((*frame_types)[i] == kKeyFrame && send_stream_[i]) {
+      if ((*frame_types)[i] == kVideoFrameKey && send_stream_[i]) {
         send_key_frame = true;
         break;
       }
@@ -819,8 +805,8 @@ int VP8EncoderImpl::Encode(
   if (send_key_frame) {
     // Adapt the size of the key frame when in screenshare with 1 temporal
     // layer.
-    if (encoders_.size() == 1 && codec_.mode == kScreensharing
-        && codec_.codecSpecific.VP8.numberOfTemporalLayers <= 1) {
+    if (encoders_.size() == 1 && codec_.mode == kScreensharing &&
+        codec_.codecSpecific.VP8.numberOfTemporalLayers <= 1) {
       const uint32_t forceKeyFrameIntraTh = 100;
       vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
                         forceKeyFrameIntraTh);
@@ -832,13 +818,12 @@ int VP8EncoderImpl::Encode(
     }
     std::fill(key_frame_request_.begin(), key_frame_request_.end(), false);
   } else if (codec_specific_info &&
-      codec_specific_info->codecType == kVideoCodecVP8) {
+             codec_specific_info->codecType == kVideoCodecVP8) {
     if (feedback_mode_) {
       // Handle RPSI and SLI messages and set up the appropriate encode flags.
       bool sendRefresh = false;
       if (codec_specific_info->codecSpecific.VP8.hasReceivedRPSI) {
-        rps_.ReceivedRPSI(
-            codec_specific_info->codecSpecific.VP8.pictureIdRPSI);
+        rps_.ReceivedRPSI(codec_specific_info->codecSpecific.VP8.pictureIdRPSI);
       }
       if (codec_specific_info->codecSpecific.VP8.hasReceivedSLI) {
         sendRefresh = rps_.ReceivedSLI(input_image.timestamp());
@@ -879,9 +864,18 @@ int VP8EncoderImpl::Encode(
   // whereas |encoder_| is from highest to lowest resolution.
   size_t stream_idx = encoders_.size() - 1;
   for (size_t i = 0; i < encoders_.size(); ++i, --stream_idx) {
+    // Allow the layers adapter to temporarily modify the configuration. This
+    // change isn't stored in configurations_ so change will be discarded at
+    // the next update.
+    vpx_codec_enc_cfg_t temp_config;
+    memcpy(&temp_config, &configurations_[i], sizeof(vpx_codec_enc_cfg_t));
+    if (temporal_layers_[stream_idx]->UpdateConfiguration(&temp_config)) {
+      if (vpx_codec_enc_config_set(&encoders_[i], &temp_config))
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
     vpx_codec_control(&encoders_[i], VP8E_SET_FRAME_FLAGS, flags[stream_idx]);
-    vpx_codec_control(&encoders_[i],
-                      VP8E_SET_TEMPORAL_LAYER_ID,
+    vpx_codec_control(&encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
                       temporal_layers_[stream_idx]->CurrentLayerId());
   }
   // TODO(holmer): Ideally the duration should be the timestamp diff of this
@@ -899,23 +893,25 @@ int VP8EncoderImpl::Encode(
   // Reset specific intra frame thresholds, following the key frame.
   if (send_key_frame) {
     vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
-        rc_max_intra_target_);
+                      rc_max_intra_target_);
   }
-  if (error) {
+  if (error)
     return WEBRTC_VIDEO_CODEC_ERROR;
-  }
   timestamp_ += duration;
   return GetEncodedPartitions(input_image, only_predict_from_key_frame);
 }
 
 // TODO(pbos): Make sure this works for properly for >1 encoders.
-int VP8EncoderImpl::UpdateCodecFrameSize(
-    const I420VideoFrame& input_image) {
+int VP8EncoderImpl::UpdateCodecFrameSize(const VideoFrame& input_image) {
   codec_.width = input_image.width();
   codec_.height = input_image.height();
+  if (codec_.numberOfSimulcastStreams <= 1) {
+    // For now scaling is only used for single-layer streams.
+    codec_.simulcastStream[0].width = input_image.width();
+    codec_.simulcastStream[0].height = input_image.height();
+  }
   // Update the cpu_speed setting for resolution change.
-  vpx_codec_control(&(encoders_[0]),
-                    VP8E_SET_CPUUSED,
+  vpx_codec_control(&(encoders_[0]), VP8E_SET_CPUUSED,
                     SetCpuSpeed(codec_.width, codec_.height));
   raw_images_[0].w = codec_.width;
   raw_images_[0].h = codec_.height;
@@ -948,43 +944,52 @@ void VP8EncoderImpl::PopulateCodecSpecific(
   }
   vp8Info->simulcastIdx = stream_idx;
   vp8Info->keyIdx = kNoKeyIdx;  // TODO(hlundin) populate this
-  vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) ?
-      true : false;
+  vp8Info->nonReference =
+      (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) ? true : false;
   bool base_layer_sync_point = (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ||
-                                only_predicting_from_key_frame;
+                               only_predicting_from_key_frame;
   temporal_layers_[stream_idx]->PopulateCodecSpecific(base_layer_sync_point,
-                                                      vp8Info,
-                                                      timestamp);
+                                                      vp8Info, timestamp);
   // Prepare next.
   picture_id_[stream_idx] = (picture_id_[stream_idx] + 1) & 0x7FFF;
 }
 
-int VP8EncoderImpl::GetEncodedPartitions(
-    const I420VideoFrame& input_image,
-    bool only_predicting_from_key_frame) {
+int VP8EncoderImpl::GetEncodedPartitions(const VideoFrame& input_image,
+                                         bool only_predicting_from_key_frame) {
+  int bw_resolutions_disabled =
+      (encoders_.size() > 1) ? NumStreamsDisabled(send_stream_) : -1;
+
   int stream_idx = static_cast<int>(encoders_.size()) - 1;
+  int result = WEBRTC_VIDEO_CODEC_OK;
   for (size_t encoder_idx = 0; encoder_idx < encoders_.size();
-      ++encoder_idx, --stream_idx) {
+       ++encoder_idx, --stream_idx) {
     vpx_codec_iter_t iter = NULL;
     int part_idx = 0;
     encoded_images_[encoder_idx]._length = 0;
-    encoded_images_[encoder_idx]._frameType = kDeltaFrame;
+    encoded_images_[encoder_idx]._frameType = kVideoFrameDelta;
     RTPFragmentationHeader frag_info;
     // token_partitions_ is number of bits used.
-    frag_info.VerifyAndAllocateFragmentationHeader((1 << token_partitions_)
-                                                   + 1);
+    frag_info.VerifyAndAllocateFragmentationHeader((1 << token_partitions_) +
+                                                   1);
     CodecSpecificInfo codec_specific;
-    const vpx_codec_cx_pkt_t *pkt = NULL;
-    while ((pkt = vpx_codec_get_cx_data(&encoders_[encoder_idx],
-                                        &iter)) != NULL) {
+    const vpx_codec_cx_pkt_t* pkt = NULL;
+    while ((pkt = vpx_codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
+           NULL) {
       switch (pkt->kind) {
         case VPX_CODEC_CX_FRAME_PKT: {
-          uint32_t length = encoded_images_[encoder_idx]._length;
+          size_t length = encoded_images_[encoder_idx]._length;
+          if (pkt->data.frame.sz + length >
+              encoded_images_[encoder_idx]._size) {
+            uint8_t* buffer = new uint8_t[pkt->data.frame.sz + length];
+            memcpy(buffer, encoded_images_[encoder_idx]._buffer, length);
+            delete[] encoded_images_[encoder_idx]._buffer;
+            encoded_images_[encoder_idx]._buffer = buffer;
+            encoded_images_[encoder_idx]._size = pkt->data.frame.sz + length;
+          }
           memcpy(&encoded_images_[encoder_idx]._buffer[length],
-                 pkt->data.frame.buf,
-                 pkt->data.frame.sz);
+                 pkt->data.frame.buf, pkt->data.frame.sz);
           frag_info.fragmentationOffset[part_idx] = length;
-          frag_info.fragmentationLength[part_idx] =  pkt->data.frame.sz;
+          frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
           frag_info.fragmentationPlType[part_idx] = 0;  // not known here
           frag_info.fragmentationTimeDiff[part_idx] = 0;
           encoded_images_[encoder_idx]._length += pkt->data.frame.sz;
@@ -999,7 +1004,7 @@ int VP8EncoderImpl::GetEncodedPartitions(
       if ((pkt->data.frame.flags & VPX_FRAME_IS_FRAGMENT) == 0) {
         // check if encoded frame is a key frame
         if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
-          encoded_images_[encoder_idx]._frameType = kKeyFrame;
+          encoded_images_[encoder_idx]._frameType = kVideoFrameKey;
           rps_.EncodedKeyFrame(picture_id_[stream_idx]);
         }
         PopulateCodecSpecific(&codec_specific, *pkt, stream_idx,
@@ -1011,9 +1016,12 @@ int VP8EncoderImpl::GetEncodedPartitions(
     encoded_images_[encoder_idx]._timeStamp = input_image.timestamp();
     encoded_images_[encoder_idx].capture_time_ms_ =
         input_image.render_time_ms();
+
+    int qp = -1;
+    vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
     temporal_layers_[stream_idx]->FrameEncoded(
         encoded_images_[encoder_idx]._length,
-        encoded_images_[encoder_idx]._timeStamp);
+        encoded_images_[encoder_idx]._timeStamp, qp);
     if (send_stream_[stream_idx]) {
       if (encoded_images_[encoder_idx]._length > 0) {
         TRACE_COUNTER_ID1("webrtc", "EncodedFrameSize", encoder_idx,
@@ -1022,32 +1030,29 @@ int VP8EncoderImpl::GetEncodedPartitions(
             codec_.simulcastStream[stream_idx].height;
         encoded_images_[encoder_idx]._encodedWidth =
             codec_.simulcastStream[stream_idx].width;
+        encoded_images_[encoder_idx]
+            .adapt_reason_.quality_resolution_downscales =
+            quality_scaler_enabled_ ? quality_scaler_.downscale_shift() : -1;
+        // Report once per frame (lowest stream always sent).
+        encoded_images_[encoder_idx].adapt_reason_.bw_resolutions_disabled =
+            (stream_idx == 0) ? bw_resolutions_disabled : -1;
         encoded_complete_callback_->Encoded(encoded_images_[encoder_idx],
                                             &codec_specific, &frag_info);
+      } else if (codec_.mode == kScreensharing) {
+        result = WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT;
       }
-    } else {
-      // Required in case padding is applied to dropped frames.
-      encoded_images_[encoder_idx]._length = 0;
-      encoded_images_[encoder_idx]._frameType = kSkipFrame;
-      codec_specific.codecType = kVideoCodecVP8;
-      CodecSpecificInfoVP8* vp8Info = &(codec_specific.codecSpecific.VP8);
-      vp8Info->pictureId = picture_id_[stream_idx];
-      vp8Info->simulcastIdx = stream_idx;
-      vp8Info->keyIdx = kNoKeyIdx;
-      encoded_complete_callback_->Encoded(encoded_images_[encoder_idx],
-                                          &codec_specific, NULL);
     }
   }
   if (encoders_.size() == 1 && send_stream_[0]) {
     if (encoded_images_[0]._length > 0) {
       int qp;
       vpx_codec_control(&encoders_[0], VP8E_GET_LAST_QUANTIZER_64, &qp);
-      quality_scaler_.ReportEncodedFrame(qp);
+      quality_scaler_.ReportQP(qp);
     } else {
       quality_scaler_.ReportDroppedFrame();
     }
   }
-  return WEBRTC_VIDEO_CODEC_OK;
+  return result;
 }
 
 int VP8EncoderImpl::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
@@ -1061,7 +1066,6 @@ int VP8EncoderImpl::RegisterEncodeCompleteCallback(
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-
 VP8DecoderImpl::VP8DecoderImpl()
     : decode_complete_callback_(NULL),
       inited_(false),
@@ -1073,8 +1077,7 @@ VP8DecoderImpl::VP8DecoderImpl()
       propagation_cnt_(-1),
       last_frame_width_(0),
       last_frame_height_(0),
-      key_frame_required_(true) {
-}
+      key_frame_required_(true) {}
 
 VP8DecoderImpl::~VP8DecoderImpl() {
   inited_ = true;  // in order to do the actual release
@@ -1090,8 +1093,7 @@ int VP8DecoderImpl::Reset() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int VP8DecoderImpl::InitDecode(const VideoCodec* inst,
-                                       int number_of_cores) {
+int VP8DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
   int ret_val = Release();
   if (ret_val < 0) {
     return ret_val;
@@ -1102,13 +1104,13 @@ int VP8DecoderImpl::InitDecode(const VideoCodec* inst,
   if (inst && inst->codecType == kVideoCodecVP8) {
     feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
   }
-  vpx_codec_dec_cfg_t  cfg;
+  vpx_codec_dec_cfg_t cfg;
   // Setting number of threads to a constant value (1)
   cfg.threads = 1;
   cfg.h = cfg.w = 0;  // set after decode
 
-vpx_codec_flags_t flags = 0;
-#if !defined(WEBRTC_ARCH_ARM) && !defined(WEBRTC_ARCH_MIPS)
+  vpx_codec_flags_t flags = 0;
+#if !defined(WEBRTC_ARCH_ARM) && !defined(WEBRTC_ARCH_ARM64) && !defined(WEBRTC_ARCH_MIPS)
   flags = VPX_CODEC_USE_POSTPROC;
 #ifdef INDEPENDENT_PARTITIONS
   flags |= VPX_CODEC_USE_INPUT_PARTITION;
@@ -1132,10 +1134,10 @@ vpx_codec_flags_t flags = 0;
 }
 
 int VP8DecoderImpl::Decode(const EncodedImage& input_image,
-                                   bool missing_frames,
-                                   const RTPFragmentationHeader* fragmentation,
-                                   const CodecSpecificInfo* codec_specific_info,
-                                   int64_t /*render_time_ms*/) {
+                           bool missing_frames,
+                           const RTPFragmentationHeader* fragmentation,
+                           const CodecSpecificInfo* codec_specific_info,
+                           int64_t /*render_time_ms*/) {
   if (!inited_) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
@@ -1155,7 +1157,7 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
   }
 #endif
 
-#ifndef WEBRTC_ARCH_ARM
+#if !defined(WEBRTC_ARCH_ARM) && !defined(WEBRTC_ARCH_ARM64)
   vp8_postproc_cfg_t ppcfg;
   // MFQE enabled to reduce key frame popping.
   ppcfg.post_proc_flag = VP8_MFQE | VP8_DEBLOCK;
@@ -1170,7 +1172,7 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
 
   // Always start with a complete key frame.
   if (key_frame_required_) {
-    if (input_image._frameType != kKeyFrame)
+    if (input_image._frameType != kVideoFrameKey)
       return WEBRTC_VIDEO_CODEC_ERROR;
     // We have a key frame - is it complete?
     if (input_image._completeFrame) {
@@ -1183,11 +1185,12 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
   // the feedback mode is enabled (RPS).
   // Reset on a key frame refresh.
   if (!feedback_mode_) {
-    if (input_image._frameType == kKeyFrame && input_image._completeFrame) {
+    if (input_image._frameType == kVideoFrameKey &&
+        input_image._completeFrame) {
       propagation_cnt_ = -1;
-    // Start count on first loss.
+      // Start count on first loss.
     } else if ((!input_image._completeFrame || missing_frames) &&
-        propagation_cnt_ == -1) {
+               propagation_cnt_ == -1) {
       propagation_cnt_ = 0;
     }
     if (propagation_cnt_ >= 0) {
@@ -1236,18 +1239,18 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
 #endif
 
   // Store encoded frame if key frame. (Used in Copy method.)
-  if (input_image._frameType == kKeyFrame && input_image._buffer != NULL) {
+  if (input_image._frameType == kVideoFrameKey && input_image._buffer != NULL) {
     const uint32_t bytes_to_copy = input_image._length;
     if (last_keyframe_._size < bytes_to_copy) {
-      delete [] last_keyframe_._buffer;
+      delete[] last_keyframe_._buffer;
       last_keyframe_._buffer = NULL;
       last_keyframe_._size = 0;
     }
     uint8_t* temp_buffer = last_keyframe_._buffer;  // Save buffer ptr.
-    uint32_t temp_size = last_keyframe_._size;  // Save size.
-    last_keyframe_ = input_image;  // Shallow copy.
-    last_keyframe_._buffer = temp_buffer;  // Restore buffer ptr.
-    last_keyframe_._size = temp_size;  // Restore buffer size.
+    uint32_t temp_size = last_keyframe_._size;      // Save size.
+    last_keyframe_ = input_image;                   // Shallow copy.
+    last_keyframe_._buffer = temp_buffer;           // Restore buffer ptr.
+    last_keyframe_._size = temp_size;               // Restore buffer size.
     if (!last_keyframe_._buffer) {
       // Allocate memory.
       last_keyframe_._size = bytes_to_copy;
@@ -1270,7 +1273,7 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
     // Whenever we receive an incomplete key frame all reference buffers will
     // be corrupt. If that happens we must request new key frames until we
     // decode a complete key frame.
-    if (input_image._frameType == kKeyFrame && !input_image._completeFrame)
+    if (input_image._frameType == kVideoFrameKey && !input_image._completeFrame)
       return WEBRTC_VIDEO_CODEC_ERROR;
     // Check for reference updates and last reference buffer corruption and
     // signal successful reference propagation or frame corruption to the
@@ -1297,7 +1300,8 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
     }
     if (picture_id > -1) {
       if (((reference_updates & VP8_GOLD_FRAME) ||
-          (reference_updates & VP8_ALTR_FRAME)) && !corrupted) {
+           (reference_updates & VP8_ALTR_FRAME)) &&
+          !corrupted) {
         decode_complete_callback_->ReceivedDecodedReferenceFrame(picture_id);
       }
       decode_complete_callback_->ReceivedDecodedFrame(picture_id);
@@ -1320,14 +1324,10 @@ int VP8DecoderImpl::DecodePartitions(
     const EncodedImage& input_image,
     const RTPFragmentationHeader* fragmentation) {
   for (int i = 0; i < fragmentation->fragmentationVectorSize; ++i) {
-    const uint8_t* partition = input_image._buffer +
-        fragmentation->fragmentationOffset[i];
-    const uint32_t partition_length =
-        fragmentation->fragmentationLength[i];
-    if (vpx_codec_decode(decoder_,
-                         partition,
-                         partition_length,
-                         0,
+    const uint8_t* partition =
+        input_image._buffer + fragmentation->fragmentationOffset[i];
+    const uint32_t partition_length = fragmentation->fragmentationLength[i];
+    if (vpx_codec_decode(decoder_, partition, partition_length, 0,
                          VPX_DL_REALTIME)) {
       return WEBRTC_VIDEO_CODEC_ERROR;
     }
@@ -1340,8 +1340,8 @@ int VP8DecoderImpl::DecodePartitions(
 }
 
 int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
-                                        uint32_t timestamp,
-                                        int64_t ntp_time_ms) {
+                                uint32_t timestamp,
+                                int64_t ntp_time_ms) {
   if (img == NULL) {
     // Decoder OK and NULL image => No show frame
     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
@@ -1349,16 +1349,15 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
   last_frame_width_ = img->d_w;
   last_frame_height_ = img->d_h;
   // Allocate memory for decoded image.
-  I420VideoFrame decoded_image(buffer_pool_.CreateBuffer(img->d_w, img->d_h),
-                               timestamp, 0, kVideoRotation_0);
-  libyuv::I420Copy(
-      img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
-      img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
-      img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
-      decoded_image.buffer(kYPlane), decoded_image.stride(kYPlane),
-      decoded_image.buffer(kUPlane), decoded_image.stride(kUPlane),
-      decoded_image.buffer(kVPlane), decoded_image.stride(kVPlane),
-      img->d_w, img->d_h);
+  VideoFrame decoded_image(buffer_pool_.CreateBuffer(img->d_w, img->d_h),
+                           timestamp, 0, kVideoRotation_0);
+  libyuv::I420Copy(img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
+                   img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
+                   img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
+                   decoded_image.buffer(kYPlane), decoded_image.stride(kYPlane),
+                   decoded_image.buffer(kUPlane), decoded_image.stride(kUPlane),
+                   decoded_image.buffer(kVPlane), decoded_image.stride(kVPlane),
+                   img->d_w, img->d_h);
   decoded_image.set_ntp_time_ms(ntp_time_ms);
   int ret = decode_complete_callback_->Decoded(decoded_image);
   if (ret != 0)
@@ -1377,7 +1376,7 @@ int VP8DecoderImpl::RegisterDecodeCompleteCallback(
 
 int VP8DecoderImpl::Release() {
   if (last_keyframe_._buffer != NULL) {
-    delete [] last_keyframe_._buffer;
+    delete[] last_keyframe_._buffer;
     last_keyframe_._buffer = NULL;
   }
   if (decoder_ != NULL) {
@@ -1397,95 +1396,19 @@ int VP8DecoderImpl::Release() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-VideoDecoder* VP8DecoderImpl::Copy() {
-  // Sanity checks.
-  if (!inited_) {
-    // Not initialized.
-    assert(false);
-    return NULL;
-  }
-  if (last_frame_width_ == 0 || last_frame_height_ == 0) {
-    // Nothing has been decoded before; cannot clone.
-    return NULL;
-  }
-  if (last_keyframe_._buffer == NULL) {
-    // Cannot clone if we have no key frame to start with.
-    return NULL;
-  }
-  // Create a new VideoDecoder object
-  VP8DecoderImpl* copy = new VP8DecoderImpl;
-
-  // Initialize the new decoder
-  if (copy->InitDecode(&codec_, 1) != WEBRTC_VIDEO_CODEC_OK) {
-    delete copy;
-    return NULL;
-  }
-  // Inject last key frame into new decoder.
-  if (vpx_codec_decode(copy->decoder_, last_keyframe_._buffer,
-                       last_keyframe_._length, NULL, VPX_DL_REALTIME)) {
-    delete copy;
-    return NULL;
-  }
-  // Allocate memory for reference image copy
-  assert(last_frame_width_ > 0);
-  assert(last_frame_height_ > 0);
-  assert(image_format_ > VPX_IMG_FMT_NONE);
-  // Check if frame format has changed.
-  if (ref_frame_ &&
-      (last_frame_width_ != static_cast<int>(ref_frame_->img.d_w) ||
-          last_frame_height_ != static_cast<int>(ref_frame_->img.d_h) ||
-          image_format_ != ref_frame_->img.fmt)) {
-    vpx_img_free(&ref_frame_->img);
-    delete ref_frame_;
-    ref_frame_ = NULL;
-  }
-
-
-  if (!ref_frame_) {
-    ref_frame_ = new vpx_ref_frame_t;
-    // Setting alignment to 32 - as that ensures at least 16 for all
-    // planes (32 for Y, 16 for U,V) - libvpx sets the requested stride
-    // for the y plane, but only half of it to the u and v planes.
-    if (!vpx_img_alloc(&ref_frame_->img,
-                       static_cast<vpx_img_fmt_t>(image_format_),
-                       last_frame_width_, last_frame_height_,
-                       kVp832ByteAlign)) {
-      assert(false);
-      delete copy;
-      return NULL;
-    }
-  }
-  const vpx_ref_frame_type_t type_vec[] = { VP8_LAST_FRAME, VP8_GOLD_FRAME,
-      VP8_ALTR_FRAME };
-  for (uint32_t ix = 0;
-      ix < sizeof(type_vec) / sizeof(vpx_ref_frame_type_t); ++ix) {
-    ref_frame_->frame_type = type_vec[ix];
-    if (CopyReference(copy) < 0) {
-      delete copy;
-      return NULL;
-    }
-  }
-  // Copy all member variables (that are not set in initialization).
-  copy->feedback_mode_ = feedback_mode_;
-  copy->image_format_ = image_format_;
-  copy->last_keyframe_ = last_keyframe_;  // Shallow copy.
-  // Allocate memory. (Discard copied _buffer pointer.)
-  copy->last_keyframe_._buffer = new uint8_t[last_keyframe_._size];
-  memcpy(copy->last_keyframe_._buffer, last_keyframe_._buffer,
-         last_keyframe_._length);
-
-  return static_cast<VideoDecoder*>(copy);
+const char* VP8DecoderImpl::ImplementationName() const {
+  return "libvpx";
 }
 
 int VP8DecoderImpl::CopyReference(VP8DecoderImpl* copy) {
   // The type of frame to copy should be set in ref_frame_->frame_type
   // before the call to this function.
-  if (vpx_codec_control(decoder_, VP8_COPY_REFERENCE, ref_frame_)
-      != VPX_CODEC_OK) {
+  if (vpx_codec_control(decoder_, VP8_COPY_REFERENCE, ref_frame_) !=
+      VPX_CODEC_OK) {
     return -1;
   }
-  if (vpx_codec_control(copy->decoder_, VP8_SET_REFERENCE, ref_frame_)
-      != VPX_CODEC_OK) {
+  if (vpx_codec_control(copy->decoder_, VP8_SET_REFERENCE, ref_frame_) !=
+      VPX_CODEC_OK) {
     return -1;
   }
   return 0;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
index fe7cf43342..9d5fb713a4 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
@@ -22,13 +22,13 @@
 #include "vpx/vp8cx.h"
 #include "vpx/vp8dx.h"
 
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
 #include "webrtc/modules/video_coding/utility/quality_scaler.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -46,9 +46,9 @@ class VP8EncoderImpl : public VP8Encoder {
                          int number_of_cores,
                          size_t max_payload_size);
 
-  virtual int Encode(const I420VideoFrame& input_image,
+  virtual int Encode(const VideoFrame& input_image,
                      const CodecSpecificInfo* codec_specific_info,
-                     const std::vector<VideoFrameType>* frame_types);
+                     const std::vector<FrameType>* frame_types);
 
   virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
 
@@ -56,8 +56,13 @@ class VP8EncoderImpl : public VP8Encoder {
 
   virtual int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate);
 
+  void OnDroppedFrame() override {}
+
+  const char* ImplementationName() const override;
+
  private:
-  void SetupTemporalLayers(int num_streams, int num_temporal_layers,
+  void SetupTemporalLayers(int num_streams,
+                           int num_temporal_layers,
                            const VideoCodec& codec);
 
   // Set the cpu_speed setting for encoder based on resolution and/or platform.
@@ -70,7 +75,7 @@ class VP8EncoderImpl : public VP8Encoder {
   int InitAndSetControlSettings();
 
   // Update frame size for codec.
-  int UpdateCodecFrameSize(const I420VideoFrame& input_image);
+  int UpdateCodecFrameSize(const VideoFrame& input_image);
 
   void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
                              const vpx_codec_cx_pkt& pkt,
@@ -78,15 +83,9 @@ class VP8EncoderImpl : public VP8Encoder {
                              uint32_t timestamp,
                              bool only_predicting_from_key_frame);
 
-  int GetEncodedPartitions(const I420VideoFrame& input_image,
+  int GetEncodedPartitions(const VideoFrame& input_image,
                            bool only_predicting_from_key_frame);
 
-  // Get the stream bitrate, for the stream |stream_idx|, given the bitrate
-  // |new_bitrate_kbit|.
-  int GetStreamBitrate(int stream_idx,
-                       uint32_t new_bitrate_kbit,
-                       bool* send_stream) const;
-
   // Set the stream state for stream |stream_idx|.
   void SetStreamState(bool send_stream, int stream_idx);
 
@@ -118,6 +117,7 @@ class VP8EncoderImpl : public VP8Encoder {
   std::vector<vpx_codec_enc_cfg_t> configurations_;
   std::vector<vpx_rational_t> downsampling_factors_;
   QualityScaler quality_scaler_;
+  bool quality_scaler_enabled_;
 };  // end of VP8EncoderImpl class
 
 class VP8DecoderImpl : public VP8Decoder {
@@ -126,21 +126,19 @@ class VP8DecoderImpl : public VP8Decoder {
 
   virtual ~VP8DecoderImpl();
 
-  virtual int InitDecode(const VideoCodec* inst, int number_of_cores);
+  int InitDecode(const VideoCodec* inst, int number_of_cores) override;
 
-  virtual int Decode(const EncodedImage& input_image,
-                     bool missing_frames,
-                     const RTPFragmentationHeader* fragmentation,
-                     const CodecSpecificInfo* codec_specific_info,
-                     int64_t /*render_time_ms*/);
+  int Decode(const EncodedImage& input_image,
+             bool missing_frames,
+             const RTPFragmentationHeader* fragmentation,
+             const CodecSpecificInfo* codec_specific_info,
+             int64_t /*render_time_ms*/) override;
 
-  virtual int RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
+  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
+  int Release() override;
+  int Reset() override;
 
-  virtual int Release();
-
-  virtual int Reset();
-
-  virtual VideoDecoder* Copy();
+  const char* ImplementationName() const override;
 
  private:
   // Copy reference image from this _decoder to the _decoder in copyTo. Set
@@ -172,4 +170,3 @@ class VP8DecoderImpl : public VP8Decoder {
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
-
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
index a116d74bf7..9e546653db 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
@@ -1,4 +1,4 @@
- /*
+/*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
@@ -9,21 +9,21 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/common_video/interface/video_image.h"
+#include "webrtc/common_video/include/video_image.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/metrics/video_metrics.h"
 #include "webrtc/tools/simple_command_line_parser.h"
+#include "webrtc/video_frame.h"
 
 class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
  public:
   explicit Vp8SequenceCoderEncodeCallback(FILE* encoded_file)
-      : encoded_file_(encoded_file),
-        encoded_bytes_(0) {}
+      : encoded_file_(encoded_file), encoded_bytes_(0) {}
   ~Vp8SequenceCoderEncodeCallback();
   int Encoded(const webrtc::EncodedImage& encoded_image,
               const webrtc::CodecSpecificInfo* codecSpecificInfo,
@@ -31,6 +31,7 @@ class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
   // Returns the encoded image.
   webrtc::EncodedImage encoded_image() { return encoded_image_; }
   size_t encoded_bytes() { return encoded_bytes_; }
+
  private:
   webrtc::EncodedImage encoded_image_;
   FILE* encoded_file_;
@@ -38,7 +39,7 @@ class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
 };
 
 Vp8SequenceCoderEncodeCallback::~Vp8SequenceCoderEncodeCallback() {
-  delete [] encoded_image_._buffer;
+  delete[] encoded_image_._buffer;
   encoded_image_._buffer = NULL;
 }
 int Vp8SequenceCoderEncodeCallback::Encoded(
@@ -46,7 +47,7 @@ int Vp8SequenceCoderEncodeCallback::Encoded(
     const webrtc::CodecSpecificInfo* codecSpecificInfo,
     const webrtc::RTPFragmentationHeader* fragmentation) {
   if (encoded_image_._size < encoded_image._size) {
-    delete [] encoded_image_._buffer;
+    delete[] encoded_image_._buffer;
     encoded_image_._buffer = NULL;
     encoded_image_._buffer = new uint8_t[encoded_image._size];
     encoded_image_._size = encoded_image._size;
@@ -68,28 +69,32 @@ class Vp8SequenceCoderDecodeCallback : public webrtc::DecodedImageCallback {
  public:
   explicit Vp8SequenceCoderDecodeCallback(FILE* decoded_file)
       : decoded_file_(decoded_file) {}
-  int Decoded(webrtc::I420VideoFrame& frame);
+  int32_t Decoded(webrtc::VideoFrame& frame) override;
+  int32_t Decoded(webrtc::VideoFrame& frame, int64_t decode_time_ms) override {
+    RTC_NOTREACHED();
+    return -1;
+  }
   bool DecodeComplete();
 
  private:
   FILE* decoded_file_;
 };
 
-int Vp8SequenceCoderDecodeCallback::Decoded(webrtc::I420VideoFrame& image) {
-  EXPECT_EQ(0, webrtc::PrintI420VideoFrame(image, decoded_file_));
+int Vp8SequenceCoderDecodeCallback::Decoded(webrtc::VideoFrame& image) {
+  EXPECT_EQ(0, webrtc::PrintVideoFrame(image, decoded_file_));
   return 0;
 }
 
-int SequenceCoder(webrtc::test::CommandLineParser& parser) {
-  int width = strtol((parser.GetFlag("w")).c_str(), NULL, 10);
-  int height = strtol((parser.GetFlag("h")).c_str(), NULL, 10);
-  int framerate = strtol((parser.GetFlag("f")).c_str(), NULL, 10);
+int SequenceCoder(webrtc::test::CommandLineParser* parser) {
+  int width = strtol((parser->GetFlag("w")).c_str(), NULL, 10);
+  int height = strtol((parser->GetFlag("h")).c_str(), NULL, 10);
+  int framerate = strtol((parser->GetFlag("f")).c_str(), NULL, 10);
 
   if (width <= 0 || height <= 0 || framerate <= 0) {
     fprintf(stderr, "Error: Resolution cannot be <= 0!\n");
     return -1;
   }
-  int target_bitrate = strtol((parser.GetFlag("b")).c_str(), NULL, 10);
+  int target_bitrate = strtol((parser->GetFlag("b")).c_str(), NULL, 10);
   if (target_bitrate <= 0) {
     fprintf(stderr, "Error: Bit-rate cannot be <= 0!\n");
     return -1;
@@ -97,20 +102,20 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
 
   // SetUp
   // Open input file.
-  std::string encoded_file_name = parser.GetFlag("encoded_file");
+  std::string encoded_file_name = parser->GetFlag("encoded_file");
   FILE* encoded_file = fopen(encoded_file_name.c_str(), "wb");
   if (encoded_file == NULL) {
     fprintf(stderr, "Error: Cannot open encoded file\n");
     return -1;
   }
-  std::string input_file_name = parser.GetFlag("input_file");
+  std::string input_file_name = parser->GetFlag("input_file");
   FILE* input_file = fopen(input_file_name.c_str(), "rb");
   if (input_file == NULL) {
     fprintf(stderr, "Error: Cannot open input file\n");
     return -1;
   }
   // Open output file.
-  std::string output_file_name = parser.GetFlag("output_file");
+  std::string output_file_name = parser->GetFlag("output_file");
   FILE* output_file = fopen(output_file_name.c_str(), "wb");
   if (output_file == NULL) {
     fprintf(stderr, "Error: Cannot open output file\n");
@@ -118,8 +123,8 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
   }
 
   // Get range of frames: will encode num_frames following start_frame).
-  int start_frame = strtol((parser.GetFlag("start_frame")).c_str(), NULL, 10);
-  int num_frames = strtol((parser.GetFlag("num_frames")).c_str(), NULL, 10);
+  int start_frame = strtol((parser->GetFlag("start_frame")).c_str(), NULL, 10);
+  int num_frames = strtol((parser->GetFlag("num_frames")).c_str(), NULL, 10);
 
   // Codec SetUp.
   webrtc::VideoCodec inst;
@@ -140,7 +145,7 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
     return -1;
   }
   EXPECT_EQ(0, decoder->InitDecode(&inst, 1));
-  webrtc::I420VideoFrame input_frame;
+  webrtc::VideoFrame input_frame;
   size_t length = webrtc::CalcBufferSize(webrtc::kI420, width, height);
   rtc::scoped_ptr<uint8_t[]> frame_buffer(new uint8_t[length]);
 
@@ -157,8 +162,8 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
   int frames_processed = 0;
   input_frame.CreateEmptyFrame(width, height, width, half_width, half_width);
   while (!feof(input_file) &&
-      (num_frames == -1 || frames_processed < num_frames)) {
-     if (fread(frame_buffer.get(), 1, length, input_file) != length)
+         (num_frames == -1 || frames_processed < num_frames)) {
+    if (fread(frame_buffer.get(), 1, length, input_file) != length)
       continue;
     if (frame_cnt >= start_frame) {
       webrtc::ConvertToI420(webrtc::kI420, frame_buffer.get(), 0, 0, width,
@@ -179,33 +184,35 @@ int SequenceCoder(webrtc::test::CommandLineParser& parser) {
   printf("Actual bitrate: %f kbps\n", actual_bit_rate / 1000);
   webrtc::test::QualityMetricsResult psnr_result, ssim_result;
   EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
-      input_file_name.c_str(), output_file_name.c_str(),
-      inst.width, inst.height,
-      &psnr_result, &ssim_result));
+                   input_file_name.c_str(), output_file_name.c_str(),
+                   inst.width, inst.height, &psnr_result, &ssim_result));
   printf("PSNR avg: %f[dB], min: %f[dB]\nSSIM avg: %f, min: %f\n",
-          psnr_result.average, psnr_result.min,
-          ssim_result.average, ssim_result.min);
+         psnr_result.average, psnr_result.min, ssim_result.average,
+         ssim_result.min);
   return frame_cnt;
 }
 
 int main(int argc, char** argv) {
   std::string program_name = argv[0];
-  std::string usage = "Encode and decodes a video sequence, and writes"
-  "results to a file.\n"
-  "Example usage:\n" + program_name + " functionality"
-  " --w=352 --h=288 --input_file=input.yuv --output_file=output.yuv "
-  " Command line flags:\n"
-  "  - width(int): The width of the input file. Default: 352\n"
-  "  - height(int): The height of the input file. Default: 288\n"
-  "  - input_file(string): The YUV file to encode."
-  "      Default: foreman.yuv\n"
-  "  - encoded_file(string): The vp8 encoded file (encoder output)."
-  "      Default: vp8_encoded.vp8\n"
-  "  - output_file(string): The yuv decoded file (decoder output)."
-  "      Default: vp8_decoded.yuv\n."
-  "  - start_frame - frame number in which encoding will begin. Default: 0"
-  "  - num_frames - Number of frames to be processed. "
-  "      Default: -1 (entire sequence).";
+  std::string usage =
+      "Encode and decodes a video sequence, and writes"
+      "results to a file.\n"
+      "Example usage:\n" +
+      program_name +
+      " functionality"
+      " --w=352 --h=288 --input_file=input.yuv --output_file=output.yuv "
+      " Command line flags:\n"
+      "  - width(int): The width of the input file. Default: 352\n"
+      "  - height(int): The height of the input file. Default: 288\n"
+      "  - input_file(string): The YUV file to encode."
+      "      Default: foreman.yuv\n"
+      "  - encoded_file(string): The vp8 encoded file (encoder output)."
+      "      Default: vp8_encoded.vp8\n"
+      "  - output_file(string): The yuv decoded file (decoder output)."
+      "      Default: vp8_decoded.yuv\n."
+      "  - start_frame - frame number in which encoding will begin. Default: 0"
+      "  - num_frames - Number of frames to be processed. "
+      "      Default: -1 (entire sequence).";
 
   webrtc::test::CommandLineParser parser;
 
@@ -223,15 +230,16 @@ int main(int argc, char** argv) {
   parser.SetFlag("output_file", webrtc::test::OutputPath() + "vp8_decoded.yuv");
   parser.SetFlag("encoded_file",
                  webrtc::test::OutputPath() + "vp8_encoded.vp8");
-  parser.SetFlag("input_file", webrtc::test::ResourcePath("foreman_cif",
-                                                          "yuv"));
+  parser.SetFlag("input_file",
+                 webrtc::test::ResourcePath("foreman_cif", "yuv"));
   parser.SetFlag("help", "false");
 
   parser.ProcessFlags();
   if (parser.GetFlag("help") == "true") {
     parser.PrintUsageMessage();
+    exit(EXIT_SUCCESS);
   }
   parser.PrintEnteredFlags();
 
-  return SequenceCoder(parser);
+  return SequenceCoder(&parser);
 }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/include/vp9.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
index cd77f72dcb..3bcbe46b3a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
@@ -12,7 +12,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
@@ -23,7 +23,6 @@ class VP9Encoder : public VideoEncoder {
   virtual ~VP9Encoder() {}
 };
 
-
 class VP9Decoder : public VideoDecoder {
  public:
   static VP9Decoder* Create();
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
index 53e6647cfd..c7ed78a192 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
@@ -18,8 +18,8 @@ ScreenshareLayersVP9::ScreenshareLayersVP9(uint8_t num_layers)
       start_layer_(0),
       last_timestamp_(0),
       timestamp_initialized_(false) {
-  DCHECK_GT(num_layers, 0);
-  DCHECK_LE(num_layers, kMaxVp9NumberOfSpatialLayers);
+  RTC_DCHECK_GT(num_layers, 0);
+  RTC_DCHECK_LE(num_layers, kMaxVp9NumberOfSpatialLayers);
   memset(bits_used_, 0, sizeof(bits_used_));
   memset(threshold_kbps_, 0, sizeof(threshold_kbps_));
 }
@@ -34,13 +34,13 @@ void ScreenshareLayersVP9::ConfigureBitrate(int threshold_kbps,
   // to when the bitrate becomes to high, therefore setting
   // a max limit is not allowed. The top layer bitrate is
   // never used either so configuring it makes no difference.
-  DCHECK_LT(layer_id, num_layers_ - 1);
+  RTC_DCHECK_LT(layer_id, num_layers_ - 1);
   threshold_kbps_[layer_id] = threshold_kbps;
 }
 
 void ScreenshareLayersVP9::LayerFrameEncoded(unsigned int size_bytes,
                                              uint8_t layer_id) {
-  DCHECK_LT(layer_id, num_layers_);
+  RTC_DCHECK_LT(layer_id, num_layers_);
   bits_used_[layer_id] += size_bytes * 8;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9.gyp b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
index 5387b39d01..2294906c57 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
@@ -14,11 +14,6 @@
     {
       'target_name': 'webrtc_vp9',
       'type': 'static_library',
-      'dependencies': [
-        '<(webrtc_root)/common_video/common_video.gyp:common_video',
-        '<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
       'conditions': [
         ['build_libvpx==1', {
           'dependencies': [
@@ -29,21 +24,20 @@
             '$(MOZ_LIBVPX_CFLAGS)',
           ],
 	}],
-        ['build_vp9==1', {
-          'sources': [
-            'include/vp9.h',
-            'screenshare_layers.cc',
-            'screenshare_layers.h',
-            'vp9_frame_buffer_pool.cc',
-            'vp9_frame_buffer_pool.h',
-            'vp9_impl.cc',
-            'vp9_impl.h',
-          ],
-        }, {
-          'sources': [
-            'vp9_dummy_impl.cc',
-          ],
-        }],
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common_video/common_video.gyp:common_video',
+        '<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      ],
+      'sources': [
+        'include/vp9.h',
+        'screenshare_layers.cc',
+        'screenshare_layers.h',
+        'vp9_frame_buffer_pool.cc',
+        'vp9_frame_buffer_pool.h',
+        'vp9_impl.cc',
+        'vp9_impl.h',
       ],
     },
   ],
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
index fceb4bf9d3..62c05d34fa 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
@@ -16,12 +16,12 @@
 #include "vpx/vpx_frame_buffer.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
 uint8_t* Vp9FrameBufferPool::Vp9FrameBuffer::GetData() {
-  return (uint8_t*)(data_.data()); //data<uint8_t>();
+  return data_.data<uint8_t>();
 }
 
 size_t Vp9FrameBufferPool::Vp9FrameBuffer::GetDataSize() const {
@@ -34,7 +34,7 @@ void Vp9FrameBufferPool::Vp9FrameBuffer::SetSize(size_t size) {
 
 bool Vp9FrameBufferPool::InitializeVpxUsePool(
     vpx_codec_ctx* vpx_codec_context) {
-  DCHECK(vpx_codec_context);
+  RTC_DCHECK(vpx_codec_context);
   // Tell libvpx to use this pool.
   if (vpx_codec_set_frame_buffer_functions(
           // In which context to use these callback functions.
@@ -53,7 +53,7 @@ bool Vp9FrameBufferPool::InitializeVpxUsePool(
 
 rtc::scoped_refptr<Vp9FrameBufferPool::Vp9FrameBuffer>
 Vp9FrameBufferPool::GetFrameBuffer(size_t min_size) {
-  DCHECK_GT(min_size, 0u);
+  RTC_DCHECK_GT(min_size, 0u);
   rtc::scoped_refptr<Vp9FrameBuffer> available_buffer = nullptr;
   {
     rtc::CritScope cs(&buffers_lock_);
@@ -101,8 +101,8 @@ void Vp9FrameBufferPool::ClearPool() {
 int32_t Vp9FrameBufferPool::VpxGetFrameBuffer(void* user_priv,
                                               size_t min_size,
                                               vpx_codec_frame_buffer* fb) {
-  DCHECK(user_priv);
-  DCHECK(fb);
+  RTC_DCHECK(user_priv);
+  RTC_DCHECK(fb);
   Vp9FrameBufferPool* pool = static_cast<Vp9FrameBufferPool*>(user_priv);
 
   rtc::scoped_refptr<Vp9FrameBuffer> buffer = pool->GetFrameBuffer(min_size);
@@ -120,8 +120,8 @@ int32_t Vp9FrameBufferPool::VpxGetFrameBuffer(void* user_priv,
 // static
 int32_t Vp9FrameBufferPool::VpxReleaseFrameBuffer(void* user_priv,
                                                   vpx_codec_frame_buffer* fb) {
-  DCHECK(user_priv);
-  DCHECK(fb);
+  RTC_DCHECK(user_priv);
+  RTC_DCHECK(fb);
   Vp9FrameBuffer* buffer = static_cast<Vp9FrameBuffer*>(fb->priv);
   if (buffer != nullptr) {
     buffer->Release();
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
index 9a633dc6d3..6e9f0b91e7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -21,37 +21,31 @@
 #include "vpx/vp8cx.h"
 #include "vpx/vp8dx.h"
 
-#include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-
-namespace {
-
-// VP9DecoderImpl::ReturnFrame helper function used with WrappedI420Buffer.
-static void WrappedI420BufferNoLongerUsedCb(
-    webrtc::Vp9FrameBufferPool::Vp9FrameBuffer* img_buffer) {
-  img_buffer->Release();
-}
-
-}  // anonymous namespace
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 // Only positive speeds, range for real-time coding currently is: 5 - 8.
 // Lower means slower/better quality, higher means fastest/lower quality.
 int GetCpuSpeed(int width, int height) {
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  return 8;
+#else
   // For smaller resolutions, use lower speed setting (get some coding gain at
   // the cost of increased encoding complexity).
   if (width * height <= 352 * 288)
     return 5;
   else
     return 7;
+#endif
 }
 
 VP9Encoder* VP9Encoder::Create() {
@@ -60,7 +54,7 @@ VP9Encoder* VP9Encoder::Create() {
 
 void VP9EncoderImpl::EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
                                                       void* user_data) {
-  VP9EncoderImpl* enc = (VP9EncoderImpl*)(user_data);
+  VP9EncoderImpl* enc = static_cast<VP9EncoderImpl*>(user_data);
   enc->GetEncodedLayerFrame(pkt);
 }
 
@@ -95,7 +89,7 @@ VP9EncoderImpl::~VP9EncoderImpl() {
 
 int VP9EncoderImpl::Release() {
   if (encoded_image_._buffer != NULL) {
-    delete [] encoded_image_._buffer;
+    delete[] encoded_image_._buffer;
     encoded_image_._buffer = NULL;
   }
   if (encoder_ != NULL) {
@@ -286,10 +280,10 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
     num_temporal_layers_ = 1;
 
   // Random start 16 bits is enough.
-  picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;
+  picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;  // NOLINT
   // Allocate memory for encoded image
   if (encoded_image_._buffer != NULL) {
-    delete [] encoded_image_._buffer;
+    delete[] encoded_image_._buffer;
   }
   encoded_image_._size = CalcBufferSize(kI420, codec_.width, codec_.height);
   encoded_image_._buffer = new uint8_t[encoded_image_._size];
@@ -297,8 +291,8 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   // Creating a wrapper to the image - setting image data to NULL. Actual
   // pointer will be set in encode. Setting align to 1, as it is meaningless
   // (actual memory is not allocated).
-  raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height,
-                      1, NULL);
+  raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height, 1,
+                      NULL);
   // Populate encoder configuration with default values.
   if (vpx_codec_enc_config_default(vpx_codec_vp9_cx(), config_, 0)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
@@ -313,8 +307,8 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   config_->g_lag_in_frames = 0;  // 0- no frame lagging
   config_->g_threads = 1;
   // Rate control settings.
-  config_->rc_dropframe_thresh = inst->codecSpecific.VP9.frameDroppingOn ?
-      30 : 0;
+  config_->rc_dropframe_thresh =
+      inst->codecSpecific.VP9.frameDroppingOn ? 30 : 0;
   config_->rc_end_usage = VPX_CBR;
   config_->g_pass = VPX_RC_ONE_PASS;
   config_->rc_min_quantizer = 2;
@@ -326,18 +320,20 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
   config_->rc_buf_sz = 1000;
   // Set the maximum target size of any key-frame.
   rc_max_intra_target_ = MaxIntraTarget(config_->rc_buf_optimal_sz);
-  if (inst->codecSpecific.VP9.keyFrameInterval  > 0) {
+  if (inst->codecSpecific.VP9.keyFrameInterval > 0) {
     config_->kf_mode = VPX_KF_AUTO;
     config_->kf_max_dist = inst->codecSpecific.VP9.keyFrameInterval;
+    // Needs to be set (in svc mode) to get correct periodic key frame interval
+    // (will have no effect in non-svc).
+    config_->kf_min_dist = config_->kf_max_dist;
   } else {
     config_->kf_mode = VPX_KF_DISABLED;
   }
-  config_->rc_resize_allowed = inst->codecSpecific.VP9.automaticResizeOn ?
-      1 : 0;
+  config_->rc_resize_allowed =
+      inst->codecSpecific.VP9.automaticResizeOn ? 1 : 0;
   // Determine number of threads based on the image size and #cores.
-  config_->g_threads = NumberOfThreads(config_->g_w,
-                                       config_->g_h,
-                                       num_cores_);
+  config_->g_threads =
+      NumberOfThreads(config_->g_w, config_->g_h, number_of_cores);
 
   cpu_speed_ = GetCpuSpeed(config_->g_w, config_->g_h);
 
@@ -381,7 +377,7 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
   }
 
-  tl0_pic_idx_ = static_cast<uint8_t>(rand());
+  tl0_pic_idx_ = static_cast<uint8_t>(rand());  // NOLINT
 
   return InitAndSetControlSettings(inst);
 }
@@ -402,22 +398,23 @@ int VP9EncoderImpl::NumberOfThreads(int width,
 }
 
 int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
-  config_->ss_number_layers = num_spatial_layers_;
-
 #ifdef LIBVPX_SVC
+  // Set QP-min/max per spatial and temporal layer.
+  int tot_num_layers = num_spatial_layers_ * num_temporal_layers_;
+  for (int i = 0; i < tot_num_layers; ++i) {
+    svc_internal_.svc_params.max_quantizers[i] = config_->rc_max_quantizer;
+    svc_internal_.svc_params.min_quantizers[i] = config_->rc_min_quantizer;
+  }
+  config_->ss_number_layers = num_spatial_layers_;
   if (ExplicitlyConfiguredSpatialLayers()) {
     for (int i = 0; i < num_spatial_layers_; ++i) {
       const auto& layer = codec_.spatialLayers[i];
-      svc_internal_.svc_params.max_quantizers[i] = config_->rc_max_quantizer;
-      svc_internal_.svc_params.min_quantizers[i] = config_->rc_min_quantizer;
       svc_internal_.svc_params.scaling_factor_num[i] = layer.scaling_factor_num;
       svc_internal_.svc_params.scaling_factor_den[i] = layer.scaling_factor_den;
     }
   } else {
     int scaling_factor_num = 256;
     for (int i = num_spatial_layers_ - 1; i >= 0; --i) {
-      svc_internal_.svc_params.max_quantizers[i] = config_->rc_max_quantizer;
-      svc_internal_.svc_params.min_quantizers[i] = config_->rc_min_quantizer;
       // 1:2 scaling in each dimension.
       svc_internal_.svc_params.scaling_factor_num[i] = scaling_factor_num;
       svc_internal_.svc_params.scaling_factor_den[i] = 256;
@@ -452,8 +449,10 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
 
   // Register callback for getting each spatial layer.
   vpx_codec_priv_output_cx_pkt_cb_pair_t cbp = {
-      VP9EncoderImpl::EncoderOutputCodedPacketCallback, (void*)(this)};
-  vpx_codec_control(encoder_, VP9E_REGISTER_CX_CALLBACK, (void*)(&cbp));
+      VP9EncoderImpl::EncoderOutputCodedPacketCallback,
+      reinterpret_cast<void*>(this)};
+  vpx_codec_control(encoder_, VP9E_REGISTER_CX_CALLBACK,
+                    reinterpret_cast<void*>(&cbp));
 
   // Control function to set the number of column tiles in encoding a frame, in
   // log2 unit: e.g., 0 = 1 tile column, 1 = 2 tile columns, 2 = 4 tile columns.
@@ -488,12 +487,12 @@ uint32_t VP9EncoderImpl::MaxIntraTarget(uint32_t optimal_buffer_size) {
       optimal_buffer_size * scale_par * codec_.maxFramerate / 10;
   // Don't go below 3 times the per frame bandwidth.
   const uint32_t min_intra_size = 300;
-  return (target_pct < min_intra_size) ? min_intra_size: target_pct;
+  return (target_pct < min_intra_size) ? min_intra_size : target_pct;
 }
 
-int VP9EncoderImpl::Encode(const I420VideoFrame& input_image,
+int VP9EncoderImpl::Encode(const VideoFrame& input_image,
                            const CodecSpecificInfo* codec_specific_info,
-                           const std::vector<VideoFrameType>* frame_types) {
+                           const std::vector<FrameType>* frame_types) {
   if (!inited_) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
@@ -503,20 +502,13 @@ int VP9EncoderImpl::Encode(const I420VideoFrame& input_image,
   if (encoded_complete_callback_ == NULL) {
     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
   }
-  VideoFrameType frame_type = kDeltaFrame;
+  FrameType frame_type = kVideoFrameDelta;
   // We only support one stream at the moment.
   if (frame_types && frame_types->size() > 0) {
     frame_type = (*frame_types)[0];
   }
-  if (input_image.width() != codec_.width ||
-      input_image.height() != codec_.height) {
-    int ret = UpdateCodecFrameSize(input_image);
-    if (ret < 0) {
-      return ret;
-    }
-  }
-  DCHECK_EQ(input_image.width(), static_cast<int>(raw_->d_w));
-  DCHECK_EQ(input_image.height(), static_cast<int>(raw_->d_h));
+  RTC_DCHECK_EQ(input_image.width(), static_cast<int>(raw_->d_w));
+  RTC_DCHECK_EQ(input_image.height(), static_cast<int>(raw_->d_h));
 
   // Set input image for use in the callback.
   // This was necessary since you need some information from input_image.
@@ -534,7 +526,7 @@ int VP9EncoderImpl::Encode(const I420VideoFrame& input_image,
   raw_->stride[VPX_PLANE_V] = input_image.stride(kVPlane);
 
   vpx_enc_frame_flags_t flags = 0;
-  bool send_keyframe = (frame_type == kKeyFrame);
+  bool send_keyframe = (frame_type == kVideoFrameKey);
   if (send_keyframe) {
     // Key frame request from caller.
     flags = VPX_EFLAG_FORCE_KF;
@@ -575,51 +567,13 @@ int VP9EncoderImpl::Encode(const I420VideoFrame& input_image,
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int VP9EncoderImpl::UpdateCodecFrameSize(
-    const I420VideoFrame& input_image) {
-  fprintf(stderr, "Reconfiging VP( from %dx%d to %dx%d\n",
-          codec_.width, codec_.height, input_image.width(), input_image.height());
-  // Preserve latest bitrate/framerate setting
-  uint32_t old_bitrate_kbit = config_->rc_target_bitrate;
-  uint32_t old_framerate = codec_.maxFramerate;
-
-  codec_.width = input_image.width();
-  codec_.height = input_image.height();
-
-  vpx_img_free(raw_);
-  raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height,
-                      1, NULL);
-  // Update encoder context for new frame size.
-  config_->g_w = codec_.width;
-  config_->g_h = codec_.height;
-
-  // Determine number of threads based on the image size and #cores.
-  config_->g_threads = NumberOfThreads(codec_.width, codec_.height,
-                                       num_cores_);
-  // Update the cpu_speed setting for resolution change.
-  cpu_speed_ = GetCpuSpeed(codec_.width, codec_.height);
-
-  // NOTE: We would like to do this the same way vp8 does it
-  // (with vpx_codec_enc_config_set()), but that causes asserts
-  // in AQ 3 (cyclic); and in AQ 0 it works, but on a resize to smaller
-  // than 1/2 x 1/2 original it asserts in convolve().  Given these
-  // bugs in trying to do it the "right" way, we basically re-do
-  // the initialization.
-  vpx_codec_destroy(encoder_); // clean up old state
-  int result = InitAndSetControlSettings(&codec_);
-  if (result == WEBRTC_VIDEO_CODEC_OK) {
-    return SetRates(old_bitrate_kbit, old_framerate);
-  }
-  return result;
-}
-
 void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
-                                       const vpx_codec_cx_pkt& pkt,
-                                       uint32_t timestamp) {
+                                           const vpx_codec_cx_pkt& pkt,
+                                           uint32_t timestamp) {
   assert(codec_specific != NULL);
   codec_specific->codecType = kVideoCodecVP9;
-  CodecSpecificInfoVP9 *vp9_info = &(codec_specific->codecSpecific.VP9);
-  // TODO(asapersson): Set correct values.
+  CodecSpecificInfoVP9* vp9_info = &(codec_specific->codecSpecific.VP9);
+  // TODO(asapersson): Set correct value.
   vp9_info->inter_pic_predicted =
       (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? false : true;
   vp9_info->flexible_mode = codec_.codecSpecific.VP9.flexibleMode;
@@ -650,7 +604,7 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
   }
 
   // TODO(asapersson): this info has to be obtained from the encoder.
-  vp9_info->temporal_up_switch = true;
+  vp9_info->temporal_up_switch = false;
 
   bool is_first_frame = false;
   if (is_flexible_mode_) {
@@ -720,7 +674,7 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
 
 int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
   encoded_image_._length = 0;
-  encoded_image_._frameType = kDeltaFrame;
+  encoded_image_._frameType = kVideoFrameDelta;
   RTPFragmentationHeader frag_info;
   // Note: no data partitioning in VP9, so 1 partition only. We keep this
   // fragmentation data for now, until VP9 packetizer is implemented.
@@ -728,6 +682,12 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
   int part_idx = 0;
   CodecSpecificInfo codec_specific;
 
+  if (pkt->data.frame.sz > encoded_image_._size) {
+    delete[] encoded_image_._buffer;
+    encoded_image_._size = pkt->data.frame.sz;
+    encoded_image_._buffer = new uint8_t[encoded_image_._size];
+  }
+
   assert(pkt->kind == VPX_CODEC_CX_FRAME_PKT);
   memcpy(&encoded_image_._buffer[encoded_image_._length], pkt->data.frame.buf,
          pkt->data.frame.sz);
@@ -750,7 +710,7 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
   // End of frame.
   // Check if encoded frame is a key frame.
   if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
-    encoded_image_._frameType = kKeyFrame;
+    encoded_image_._frameType = kVideoFrameKey;
   }
   PopulateCodecSpecific(&codec_specific, *pkt, input_image_->timestamp());
 
@@ -797,8 +757,8 @@ vpx_svc_ref_frame_config VP9EncoderImpl::GenerateRefsAndFlags(
         if (refs[ref_idx] == -1)
           continue;
 
-        DCHECK_GE(refs[ref_idx], 0);
-        DCHECK_LE(refs[ref_idx], 7);
+        RTC_DCHECK_GE(refs[ref_idx], 0);
+        RTC_DCHECK_LE(refs[ref_idx], 7);
         // Easier to remove flags from all flags rather than having to
         // build the flags from 0.
         switch (num_ref_pics_[layer_idx]) {
@@ -820,7 +780,7 @@ vpx_svc_ref_frame_config VP9EncoderImpl::GenerateRefsAndFlags(
         }
         // Make sure we don't reference a buffer that hasn't been
         // used at all or hasn't been used since a keyframe.
-        DCHECK_NE(buffer_updated_at_frame_[refs[ref_idx]], -1);
+        RTC_DCHECK_NE(buffer_updated_at_frame_[refs[ref_idx]], -1);
 
         p_diff_[layer_idx][num_ref_pics_[layer_idx]] =
             frames_encoded_ - buffer_updated_at_frame_[refs[ref_idx]];
@@ -880,6 +840,10 @@ int VP9EncoderImpl::RegisterEncodeCompleteCallback(
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+const char* VP9EncoderImpl::ImplementationName() const {
+  return "libvpx";
+}
+
 VP9Decoder* VP9Decoder::Create() {
   return new VP9DecoderImpl();
 }
@@ -924,7 +888,7 @@ int VP9DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
   if (decoder_ == NULL) {
     decoder_ = new vpx_codec_ctx_t;
   }
-  vpx_codec_dec_cfg_t  cfg;
+  vpx_codec_dec_cfg_t cfg;
   // Setting number of threads to a constant value (1)
   cfg.threads = 1;
   cfg.h = cfg.w = 0;  // set after decode
@@ -960,7 +924,7 @@ int VP9DecoderImpl::Decode(const EncodedImage& input_image,
   }
   // Always start with a complete key frame.
   if (key_frame_required_) {
-    if (input_image._frameType != kKeyFrame)
+    if (input_image._frameType != kVideoFrameKey)
       return WEBRTC_VIDEO_CODEC_ERROR;
     // We have a key frame - is it complete?
     if (input_image._completeFrame) {
@@ -977,10 +941,8 @@ int VP9DecoderImpl::Decode(const EncodedImage& input_image,
   }
   // During decode libvpx may get and release buffers from |frame_buffer_pool_|.
   // In practice libvpx keeps a few (~3-4) buffers alive at a time.
-  if (vpx_codec_decode(decoder_,
-                       buffer,
-                       static_cast<unsigned int>(input_image._length),
-                       0,
+  if (vpx_codec_decode(decoder_, buffer,
+                       static_cast<unsigned int>(input_image._length), 0,
                        VPX_DL_REALTIME)) {
     return WEBRTC_VIDEO_CODEC_ERROR;
   }
@@ -1001,42 +963,28 @@ int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img, uint32_t timestamp) {
     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
   }
 
-#ifdef USE_WRAPPED_I420_BUFFER
   // This buffer contains all of |img|'s image data, a reference counted
-  // Vp9FrameBuffer. Performing AddRef/Release ensures it is not released and
-  // recycled during use (libvpx is done with the buffers after a few
+  // Vp9FrameBuffer. (libvpx is done with the buffers after a few
   // vpx_codec_decode calls or vpx_codec_destroy).
   Vp9FrameBufferPool::Vp9FrameBuffer* img_buffer =
       static_cast<Vp9FrameBufferPool::Vp9FrameBuffer*>(img->fb_priv);
-  img_buffer->AddRef();
   // The buffer can be used directly by the VideoFrame (without copy) by
   // using a WrappedI420Buffer.
   rtc::scoped_refptr<WrappedI420Buffer> img_wrapped_buffer(
       new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
-          img->d_w, img->d_h,
-          img->d_w, img->d_h,
-          img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
-          img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
-          img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
+          img->d_w, img->d_h, img->planes[VPX_PLANE_Y],
+          img->stride[VPX_PLANE_Y], img->planes[VPX_PLANE_U],
+          img->stride[VPX_PLANE_U], img->planes[VPX_PLANE_V],
+          img->stride[VPX_PLANE_V],
           // WrappedI420Buffer's mechanism for allowing the release of its frame
           // buffer is through a callback function. This is where we should
           // release |img_buffer|.
-          rtc::Bind(&WrappedI420BufferNoLongerUsedCb, img_buffer)));
+          rtc::KeepRefUntilDone(img_buffer)));
 
-  I420VideoFrame decoded_image_;
-  decoded_image_.set_video_frame_buffer(img_wrapped_buffer);
-#else
-  decoded_image_.CreateFrame(img->planes[VPX_PLANE_Y],
-                             img->planes[VPX_PLANE_U],
-                             img->planes[VPX_PLANE_V],
-                             img->d_w, img->d_h,
-                             img->stride[VPX_PLANE_Y],
-                             img->stride[VPX_PLANE_U],
-                             img->stride[VPX_PLANE_V]);
-#endif
-  decoded_image_.set_timestamp(timestamp);
-
-  int ret = decode_complete_callback_->Decoded(decoded_image_);
+  VideoFrame decoded_image;
+  decoded_image.set_video_frame_buffer(img_wrapped_buffer);
+  decoded_image.set_timestamp(timestamp);
+  int ret = decode_complete_callback_->Decoded(decoded_image);
   if (ret != 0)
     return ret;
   return WEBRTC_VIDEO_CODEC_OK;
@@ -1065,4 +1013,9 @@ int VP9DecoderImpl::Release() {
   inited_ = false;
   return WEBRTC_VIDEO_CODEC_OK;
 }
+
+const char* VP9DecoderImpl::ImplementationName() const {
+  return "libvpx";
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
index eeb0492416..b83d292e4f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
@@ -9,8 +9,10 @@
  *
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_IMPL_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
+
+#include <vector>
 
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
@@ -37,9 +39,9 @@ class VP9EncoderImpl : public VP9Encoder {
                  int number_of_cores,
                  size_t max_payload_size) override;
 
-  int Encode(const I420VideoFrame& input_image,
+  int Encode(const VideoFrame& input_image,
              const CodecSpecificInfo* codec_specific_info,
-             const std::vector<VideoFrameType>* frame_types) override;
+             const std::vector<FrameType>* frame_types) override;
 
   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
 
@@ -47,6 +49,10 @@ class VP9EncoderImpl : public VP9Encoder {
 
   int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate) override;
 
+  void OnDroppedFrame() override {}
+
+  const char* ImplementationName() const override;
+
   struct LayerFrameRefSettings {
     int8_t upd_buf = -1;   // -1 - no update,    0..7 - update buffer 0..7
     int8_t ref_buf1 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
@@ -69,7 +75,7 @@ class VP9EncoderImpl : public VP9Encoder {
   int InitAndSetControlSettings(const VideoCodec* inst);
 
   // Update frame size for codec.
-  int UpdateCodecFrameSize(const I420VideoFrame& input_image);
+  int UpdateCodecFrameSize(const VideoFrame& input_image);
 
   void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
                              const vpx_codec_cx_pkt& pkt,
@@ -88,7 +94,7 @@ class VP9EncoderImpl : public VP9Encoder {
   vpx_svc_ref_frame_config GenerateRefsAndFlags(
       const SuperFrameRefSettings& settings);
 #endif
-  
+
   virtual int GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt);
 
   // Callback function for outputting packets per spatial layer.
@@ -117,7 +123,7 @@ class VP9EncoderImpl : public VP9Encoder {
 #ifdef LIBVPX_SVC
   SvcInternal_t svc_internal_;
 #endif
-  const I420VideoFrame* input_image_;
+  const VideoFrame* input_image_;
   GofInfoVP9 gof_;       // Contains each frame's temporal information for
                          // non-flexible mode.
   uint8_t tl0_pic_idx_;  // Only used in non-flexible mode.
@@ -135,7 +141,6 @@ class VP9EncoderImpl : public VP9Encoder {
   rtc::scoped_ptr<ScreenshareLayersVP9> spatial_layer_;
 };
 
-
 class VP9DecoderImpl : public VP9Decoder {
  public:
   VP9DecoderImpl();
@@ -156,14 +161,11 @@ class VP9DecoderImpl : public VP9Decoder {
 
   int Reset() override;
 
+  const char* ImplementationName() const override;
+
  private:
   int ReturnFrame(const vpx_image_t* img, uint32_t timeStamp);
 
-#ifndef USE_WRAPPED_I420_BUFFER
-  // Temporarily keep VideoFrame in a separate buffer
-  // Once we debug WrappedI420VideoFrame usage, we can get rid of this
-  I420VideoFrame decoded_image_;
-#endif
   // Memory pool used to share buffers between libvpx and webrtc.
   Vp9FrameBufferPool frame_buffer_pool_;
   DecodedImageCallback* decode_complete_callback_;
@@ -174,4 +176,4 @@ class VP9DecoderImpl : public VP9Decoder {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_IMPL_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.cc b/media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.cc
similarity index 69%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.cc
index 9e142613c5..598be3a413 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/content_metrics_processing.h"
+#include "webrtc/modules/video_coding/content_metrics_processing.h"
 
 #include <math.h>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 //////////////////////////////////
@@ -38,15 +38,15 @@ int VCMContentMetricsProcessing::Reset() {
   recursive_avg_->Reset();
   uniform_avg_->Reset();
   frame_cnt_uniform_avg_ = 0;
-  avg_motion_level_  = 0.0f;
+  avg_motion_level_ = 0.0f;
   avg_spatial_level_ = 0.0f;
   return VCM_OK;
 }
 
 void VCMContentMetricsProcessing::UpdateFrameRate(float frameRate) {
   // Update factor for recursive averaging.
-  recursive_avg_factor_ = static_cast<float> (1000.0f) /
-      static_cast<float>(frameRate *  kQmMinIntervalMs);
+  recursive_avg_factor_ = static_cast<float>(1000.0f) /
+                          static_cast<float>(frameRate * kQmMinIntervalMs);
 }
 
 VideoContentMetrics* VCMContentMetricsProcessing::LongTermAvgData() {
@@ -58,10 +58,10 @@ VideoContentMetrics* VCMContentMetricsProcessing::ShortTermAvgData() {
     return NULL;
   }
   // Two metrics are used: motion and spatial level.
-  uniform_avg_->motion_magnitude = avg_motion_level_ /
-      static_cast<float>(frame_cnt_uniform_avg_);
-  uniform_avg_->spatial_pred_err = avg_spatial_level_ /
-      static_cast<float>(frame_cnt_uniform_avg_);
+  uniform_avg_->motion_magnitude =
+      avg_motion_level_ / static_cast<float>(frame_cnt_uniform_avg_);
+  uniform_avg_->spatial_pred_err =
+      avg_spatial_level_ / static_cast<float>(frame_cnt_uniform_avg_);
   return uniform_avg_;
 }
 
@@ -73,7 +73,7 @@ void VCMContentMetricsProcessing::ResetShortTermAvgData() {
 }
 
 int VCMContentMetricsProcessing::UpdateContentData(
-    const VideoContentMetrics *contentMetrics) {
+    const VideoContentMetrics* contentMetrics) {
   if (contentMetrics == NULL) {
     return VCM_OK;
   }
@@ -81,7 +81,7 @@ int VCMContentMetricsProcessing::UpdateContentData(
 }
 
 int VCMContentMetricsProcessing::ProcessContent(
-    const VideoContentMetrics *contentMetrics) {
+    const VideoContentMetrics* contentMetrics) {
   // Update the recursive averaged metrics: average is over longer window
   // of time: over QmMinIntervalMs ms.
   UpdateRecursiveAvg(contentMetrics);
@@ -92,34 +92,33 @@ int VCMContentMetricsProcessing::ProcessContent(
 }
 
 void VCMContentMetricsProcessing::UpdateUniformAvg(
-    const VideoContentMetrics *contentMetrics) {
+    const VideoContentMetrics* contentMetrics) {
   // Update frame counter.
   frame_cnt_uniform_avg_ += 1;
   // Update averaged metrics: motion and spatial level are used.
   avg_motion_level_ += contentMetrics->motion_magnitude;
-  avg_spatial_level_ +=  contentMetrics->spatial_pred_err;
+  avg_spatial_level_ += contentMetrics->spatial_pred_err;
   return;
 }
 
 void VCMContentMetricsProcessing::UpdateRecursiveAvg(
-    const VideoContentMetrics *contentMetrics) {
-
+    const VideoContentMetrics* contentMetrics) {
   // Spatial metrics: 2x2, 1x2(H), 2x1(V).
-  recursive_avg_->spatial_pred_err = (1 - recursive_avg_factor_) *
-      recursive_avg_->spatial_pred_err +
+  recursive_avg_->spatial_pred_err =
+      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err +
       recursive_avg_factor_ * contentMetrics->spatial_pred_err;
 
-  recursive_avg_->spatial_pred_err_h = (1 - recursive_avg_factor_) *
-      recursive_avg_->spatial_pred_err_h +
+  recursive_avg_->spatial_pred_err_h =
+      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err_h +
       recursive_avg_factor_ * contentMetrics->spatial_pred_err_h;
 
-  recursive_avg_->spatial_pred_err_v = (1 - recursive_avg_factor_) *
-      recursive_avg_->spatial_pred_err_v +
+  recursive_avg_->spatial_pred_err_v =
+      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err_v +
       recursive_avg_factor_ * contentMetrics->spatial_pred_err_v;
 
   // Motion metric: Derived from NFD (normalized frame difference).
-  recursive_avg_->motion_magnitude = (1 - recursive_avg_factor_) *
-      recursive_avg_->motion_magnitude +
+  recursive_avg_->motion_magnitude =
+      (1 - recursive_avg_factor_) * recursive_avg_->motion_magnitude +
       recursive_avg_factor_ * contentMetrics->motion_magnitude;
 }
-}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.h b/media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.h
similarity index 85%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.h
index 06c036ddd2..c280103e25 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/content_metrics_processing.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/content_metrics_processing.h
@@ -18,14 +18,10 @@ namespace webrtc {
 struct VideoContentMetrics;
 
 // QM interval time (in ms)
-enum {
-  kQmMinIntervalMs = 10000
-};
+enum { kQmMinIntervalMs = 10000 };
 
 // Flag for NFD metric vs motion metric
-enum {
-  kNfdMetric = 1
-};
+enum { kNfdMetric = 1 };
 
 /**********************************/
 /* Content Metrics Processing     */
@@ -36,7 +32,7 @@ class VCMContentMetricsProcessing {
   ~VCMContentMetricsProcessing();
 
   // Update class with latest metrics.
-  int UpdateContentData(const VideoContentMetrics *contentMetrics);
+  int UpdateContentData(const VideoContentMetrics* contentMetrics);
 
   // Reset the short-term averaged content data.
   void ResetShortTermAvgData();
@@ -57,13 +53,13 @@ class VCMContentMetricsProcessing {
 
  private:
   // Compute working average.
-  int ProcessContent(const VideoContentMetrics *contentMetrics);
+  int ProcessContent(const VideoContentMetrics* contentMetrics);
 
   // Update the recursive averaged metrics: longer time average (~5/10 secs).
-  void UpdateRecursiveAvg(const VideoContentMetrics *contentMetrics);
+  void UpdateRecursiveAvg(const VideoContentMetrics* contentMetrics);
 
   // Update the uniform averaged metrics: shorter time average (~RTCP report).
-  void UpdateUniformAvg(const VideoContentMetrics *contentMetrics);
+  void UpdateUniformAvg(const VideoContentMetrics* contentMetrics);
 
   VideoContentMetrics* recursive_avg_;
   VideoContentMetrics* uniform_avg_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.cc b/media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.cc
similarity index 69%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.cc
index d034466356..36c7487f94 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/decoding_state.h"
+#include "webrtc/modules/video_coding/decoding_state.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 namespace webrtc {
 
@@ -24,7 +24,9 @@ VCMDecodingState::VCMDecodingState()
       temporal_id_(kNoTemporalIdx),
       tl0_pic_id_(kNoTl0PicIdx),
       full_sync_(true),
-      in_initial_state_(true) {}
+      in_initial_state_(true) {
+  memset(frame_decoded_, 0, sizeof(frame_decoded_));
+}
 
 VCMDecodingState::~VCMDecodingState() {}
 
@@ -37,6 +39,7 @@ void VCMDecodingState::Reset() {
   tl0_pic_id_ = kNoTl0PicIdx;
   full_sync_ = true;
   in_initial_state_ = true;
+  memset(frame_decoded_, 0, sizeof(frame_decoded_));
 }
 
 uint32_t VCMDecodingState::time_stamp() const {
@@ -63,12 +66,33 @@ bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const {
 
 void VCMDecodingState::SetState(const VCMFrameBuffer* frame) {
   assert(frame != NULL && frame->GetHighSeqNum() >= 0);
-  UpdateSyncState(frame);
+  if (!UsingFlexibleMode(frame))
+    UpdateSyncState(frame);
   sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum());
   time_stamp_ = frame->TimeStamp();
   picture_id_ = frame->PictureId();
   temporal_id_ = frame->TemporalId();
   tl0_pic_id_ = frame->Tl0PicId();
+
+  if (UsingFlexibleMode(frame)) {
+    uint16_t frame_index = picture_id_ % kFrameDecodedLength;
+    if (in_initial_state_) {
+      frame_decoded_cleared_to_ = frame_index;
+    } else if (frame->FrameType() == kVideoFrameKey) {
+      memset(frame_decoded_, 0, sizeof(frame_decoded_));
+      frame_decoded_cleared_to_ = frame_index;
+    } else {
+      if (AheadOfFramesDecodedClearedTo(frame_index)) {
+        while (frame_decoded_cleared_to_ != frame_index) {
+          frame_decoded_cleared_to_ =
+              (frame_decoded_cleared_to_ + 1) % kFrameDecodedLength;
+          frame_decoded_[frame_decoded_cleared_to_] = false;
+        }
+      }
+    }
+    frame_decoded_[frame_index] = true;
+  }
+
   in_initial_state_ = false;
 }
 
@@ -80,6 +104,8 @@ void VCMDecodingState::CopyFrom(const VCMDecodingState& state) {
   tl0_pic_id_ = state.tl0_pic_id_;
   full_sync_ = state.full_sync_;
   in_initial_state_ = state.in_initial_state_;
+  frame_decoded_cleared_to_ = state.frame_decoded_cleared_to_;
+  memcpy(frame_decoded_, state.frame_decoded_, sizeof(frame_decoded_));
 }
 
 bool VCMDecodingState::UpdateEmptyFrame(const VCMFrameBuffer* frame) {
@@ -140,8 +166,8 @@ void VCMDecodingState::UpdateSyncState(const VCMFrameBuffer* frame) {
         full_sync_ = ContinuousPictureId(frame->PictureId());
       }
     } else {
-      full_sync_ = ContinuousSeqNum(static_cast<uint16_t>(
-          frame->GetLowSeqNum()));
+      full_sync_ =
+          ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
     }
   }
 }
@@ -173,7 +199,11 @@ bool VCMDecodingState::ContinuousFrame(const VCMFrameBuffer* frame) const {
   if (!full_sync_ && !frame->LayerSync())
     return false;
   if (UsingPictureId(frame)) {
-    return ContinuousPictureId(frame->PictureId());
+    if (UsingFlexibleMode(frame)) {
+      return ContinuousFrameRefs(frame);
+    } else {
+      return ContinuousPictureId(frame->PictureId());
+    }
   } else {
     return ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
   }
@@ -199,8 +229,7 @@ bool VCMDecodingState::ContinuousSeqNum(uint16_t seq_num) const {
   return seq_num == static_cast<uint16_t>(sequence_num_ + 1);
 }
 
-bool VCMDecodingState::ContinuousLayer(int temporal_id,
-                                       int tl0_pic_id) const {
+bool VCMDecodingState::ContinuousLayer(int temporal_id, int tl0_pic_id) const {
   // First, check if applicable.
   if (temporal_id == kNoTemporalIdx || tl0_pic_id == kNoTl0PicIdx)
     return false;
@@ -216,8 +245,41 @@ bool VCMDecodingState::ContinuousLayer(int temporal_id,
   return (static_cast<uint8_t>(tl0_pic_id_ + 1) == tl0_pic_id);
 }
 
+bool VCMDecodingState::ContinuousFrameRefs(const VCMFrameBuffer* frame) const {
+  uint8_t num_refs = frame->CodecSpecific()->codecSpecific.VP9.num_ref_pics;
+  for (uint8_t r = 0; r < num_refs; ++r) {
+    uint16_t frame_ref = frame->PictureId() -
+                         frame->CodecSpecific()->codecSpecific.VP9.p_diff[r];
+    uint16_t frame_index = frame_ref % kFrameDecodedLength;
+    if (AheadOfFramesDecodedClearedTo(frame_index) ||
+        !frame_decoded_[frame_index]) {
+      return false;
+    }
+  }
+  return true;
+}
+
 bool VCMDecodingState::UsingPictureId(const VCMFrameBuffer* frame) const {
   return (frame->PictureId() != kNoPictureId && picture_id_ != kNoPictureId);
 }
 
+bool VCMDecodingState::UsingFlexibleMode(const VCMFrameBuffer* frame) const {
+  return frame->CodecSpecific()->codecType == kVideoCodecVP9 &&
+         frame->CodecSpecific()->codecSpecific.VP9.flexible_mode;
+}
+
+// TODO(philipel): change how check work, this check practially
+// limits the max p_diff to 64.
+bool VCMDecodingState::AheadOfFramesDecodedClearedTo(uint16_t index) const {
+  // No way of knowing for sure if we are actually ahead of
+  // frame_decoded_cleared_to_. We just make the assumption
+  // that we are not trying to reference back to a very old
+  // index, but instead are referencing a newer index.
+  uint16_t diff =
+      index > frame_decoded_cleared_to_
+          ? kFrameDecodedLength - (index - frame_decoded_cleared_to_)
+          : frame_decoded_cleared_to_ - index;
+  return diff > kFrameDecodedLength / 2;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.h b/media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.h
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.h
index 99ee335195..f4ea8ae081 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/decoding_state.h
@@ -21,6 +21,11 @@ class VCMPacket;
 
 class VCMDecodingState {
  public:
+  // The max number of bits used to reference back
+  // to a previous frame when using flexible mode.
+  static const uint16_t kNumRefBits = 7;
+  static const uint16_t kFrameDecodedLength = 1 << kNumRefBits;
+
   VCMDecodingState();
   ~VCMDecodingState();
   // Check for old frame
@@ -52,17 +57,24 @@ class VCMDecodingState {
   bool ContinuousPictureId(int picture_id) const;
   bool ContinuousSeqNum(uint16_t seq_num) const;
   bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
+  bool ContinuousFrameRefs(const VCMFrameBuffer* frame) const;
   bool UsingPictureId(const VCMFrameBuffer* frame) const;
+  bool UsingFlexibleMode(const VCMFrameBuffer* frame) const;
+  bool AheadOfFramesDecodedClearedTo(uint16_t index) const;
 
   // Keep state of last decoded frame.
   // TODO(mikhal/stefan): create designated classes to handle these types.
-  uint16_t    sequence_num_;
-  uint32_t    time_stamp_;
-  int         picture_id_;
-  int         temporal_id_;
-  int         tl0_pic_id_;
-  bool        full_sync_;  // Sync flag when temporal layers are used.
-  bool        in_initial_state_;
+  uint16_t sequence_num_;
+  uint32_t time_stamp_;
+  int picture_id_;
+  int temporal_id_;
+  int tl0_pic_id_;
+  bool full_sync_;  // Sync flag when temporal layers are used.
+  bool in_initial_state_;
+
+  // Used to check references in flexible mode.
+  bool frame_decoded_[kFrameDecodedLength];
+  uint16_t frame_decoded_cleared_to_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/decoding_state_unittest.cc
similarity index 67%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/decoding_state_unittest.cc
index 10f1d6e4dd..5f5d0d38b1 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/decoding_state_unittest.cc
@@ -11,11 +11,11 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/decoding_state.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/decoding_state.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 namespace webrtc {
 
@@ -181,7 +181,7 @@ TEST(TestDecodingState, UpdateOldPacket) {
   // Now insert empty packet belonging to the same frame.
   packet.timestamp = 1;
   packet.seqNum = 2;
-  packet.frameType = kFrameEmpty;
+  packet.frameType = kEmptyFrame;
   packet.sizeBytes = 0;
   dec_state.UpdateOldPacket(&packet);
   EXPECT_EQ(dec_state.sequence_num(), 2);
@@ -196,7 +196,7 @@ TEST(TestDecodingState, UpdateOldPacket) {
   // sequence number.
   packet.timestamp = 0;
   packet.seqNum = 4;
-  packet.frameType = kFrameEmpty;
+  packet.frameType = kEmptyFrame;
   packet.sizeBytes = 0;
   dec_state.UpdateOldPacket(&packet);
   EXPECT_EQ(dec_state.sequence_num(), 3);
@@ -446,4 +446,254 @@ TEST(TestDecodingState, PictureIdRepeat) {
   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
 }
 
+TEST(TestDecodingState, FrameContinuityFlexibleModeKeyFrame) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.isFirstPacket = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.codecSpecificHeader.codec = kRtpVideoVp9;
+
+  RTPVideoHeaderVP9& vp9_hdr = packet.codecSpecificHeader.codecHeader.VP9;
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Key frame again
+  vp9_hdr.picture_id = 11;
+  frame.Reset();
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = 12;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeOutOfOrderFrames) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.isFirstPacket = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.codecSpecificHeader.codec = kRtpVideoVp9;
+
+  RTPVideoHeaderVP9& vp9_hdr = packet.codecSpecificHeader.codecHeader.VP9;
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 10, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = 15;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Out of order, last id 15, this id 12, ref to 10, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 12;
+  vp9_hdr.pid_diff[0] = 2;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref 10, 12, 15, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 3;
+  vp9_hdr.pid_diff[0] = 10;
+  vp9_hdr.pid_diff[1] = 8;
+  vp9_hdr.pid_diff[2] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeGeneral) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.isFirstPacket = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.codecSpecificHeader.codec = kRtpVideoVp9;
+
+  RTPVideoHeaderVP9& vp9_hdr = packet.codecSpecificHeader.codecHeader.VP9;
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+
+  // Delta frame as first frame
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Key frame then delta frame
+  frame.Reset();
+  packet.frameType = kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  dec_state.SetState(&frame);
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.picture_id = 15;
+  vp9_hdr.pid_diff[0] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 16;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Ref to 15, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 16;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11 and 15, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 2;
+  vp9_hdr.pid_diff[0] = 9;
+  vp9_hdr.pid_diff[1] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Ref to 10, 15 and 16, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 22;
+  vp9_hdr.num_ref_pics = 3;
+  vp9_hdr.pid_diff[0] = 12;
+  vp9_hdr.pid_diff[1] = 7;
+  vp9_hdr.pid_diff[2] = 6;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Key Frame, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameKey;
+  vp9_hdr.picture_id = VCMDecodingState::kFrameDecodedLength - 2;
+  vp9_hdr.num_ref_pics = 0;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame at last index, ref to KF, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = VCMDecodingState::kFrameDecodedLength - 1;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping buffer length, ref to last index, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 0;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping start frame, ref to 0, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 20;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping start frame, ref to 10, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 23;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 13;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Key frame, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameKey;
+  vp9_hdr.picture_id = 25;
+  vp9_hdr.num_ref_pics = 0;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to KF, continuous
+  frame.Reset();
+  packet.frameType = kVideoFrameDelta;
+  vp9_hdr.picture_id = 26;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to frame previous to KF, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 30;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 30;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.cc b/media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.cc
similarity index 71%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.cc
index 8b2a39244b..a55f3ee0f6 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/generic_encoder.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
 
 namespace webrtc {
 
@@ -24,7 +24,7 @@ VCMEncodedFrame::VCMEncodedFrame()
       _fragmentation(),
       _rotation(kVideoRotation_0),
       _rotation_set(false) {
-    _codecSpecificInfo.codecType = kVideoCodecUnknown;
+  _codecSpecificInfo.codecType = kVideoCodecUnknown;
 }
 
 VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
@@ -36,15 +36,14 @@ VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
       _fragmentation(),
       _rotation(kVideoRotation_0),
       _rotation_set(false) {
-    _codecSpecificInfo.codecType = kVideoCodecUnknown;
-    _buffer = NULL;
-    _size = 0;
-    _length = 0;
-    if (rhs._buffer != NULL)
-    {
-        VerifyAndAllocate(rhs._length);
-        memcpy(_buffer, rhs._buffer, rhs._length);
-    }
+  _codecSpecificInfo.codecType = kVideoCodecUnknown;
+  _buffer = NULL;
+  _size = 0;
+  _length = 0;
+  if (rhs._buffer != NULL) {
+    VerifyAndAllocate(rhs._length);
+    memcpy(_buffer, rhs._buffer, rhs._length);
+  }
 }
 
 VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
@@ -60,49 +59,43 @@ VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
   _buffer = NULL;
   _size = 0;
   _length = 0;
-  if (rhs._buffer != NULL)
-  {
-      VerifyAndAllocate(rhs._length);
-      memcpy(_buffer, rhs._buffer, rhs._length);
-      _length = rhs._length;
+  if (rhs._buffer != NULL) {
+    VerifyAndAllocate(rhs._length);
+    memcpy(_buffer, rhs._buffer, rhs._length);
+    _length = rhs._length;
   }
   _fragmentation.CopyFrom(rhs._fragmentation);
 }
 
-VCMEncodedFrame::~VCMEncodedFrame()
-{
-    Free();
+VCMEncodedFrame::~VCMEncodedFrame() {
+  Free();
 }
 
-void VCMEncodedFrame::Free()
-{
-    Reset();
-    if (_buffer != NULL)
-    {
-        delete [] _buffer;
-        _buffer = NULL;
-    }
+void VCMEncodedFrame::Free() {
+  Reset();
+  if (_buffer != NULL) {
+    delete[] _buffer;
+    _buffer = NULL;
+  }
 }
 
-void VCMEncodedFrame::Reset()
-{
-    _renderTimeMs = -1;
-    _timeStamp = 0;
-    _payloadType = 0;
-    _frameType = kDeltaFrame;
-    _encodedWidth = 0;
-    _encodedHeight = 0;
-    _completeFrame = false;
-    _missingFrame = false;
-    _length = 0;
-    _codecSpecificInfo.codecType = kVideoCodecUnknown;
-    _codec = kVideoCodecUnknown;
-    _rotation = kVideoRotation_0;
-    _rotation_set = false;
+void VCMEncodedFrame::Reset() {
+  _renderTimeMs = -1;
+  _timeStamp = 0;
+  _payloadType = 0;
+  _frameType = kVideoFrameDelta;
+  _encodedWidth = 0;
+  _encodedHeight = 0;
+  _completeFrame = false;
+  _missingFrame = false;
+  _length = 0;
+  _codecSpecificInfo.codecType = kVideoCodecUnknown;
+  _codec = kVideoCodecUnknown;
+  _rotation = kVideoRotation_0;
+  _rotation_set = false;
 }
 
-void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
-{
+void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header) {
   if (header) {
     switch (header->codec) {
       case kRtpVideoVp8: {
@@ -132,12 +125,6 @@ void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
         }
         break;
       }
-      case kRtpVideoH264: {
-        _codecSpecificInfo.codecSpecific.H264.single_nalu =
-            header->codecHeader.H264.single_nalu;
-        _codecSpecificInfo.codecType = kVideoCodecH264;
-        break;
-      }
       case kRtpVideoVp9: {
         if (_codecSpecificInfo.codecType != kVideoCodecVP9) {
           // This is the first packet for this frame.
@@ -205,6 +192,12 @@ void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
         }
         break;
       }
+      case kRtpVideoH264: {
+        _codecSpecificInfo.codecSpecific.H264.single_nalu =
+            header->codecHeader.H264.single_nalu;
+        _codecSpecificInfo.codecType = kVideoCodecH264;
+        break;
+      }
       default: {
         _codecSpecificInfo.codecType = kVideoCodecUnknown;
         break;
@@ -217,57 +210,18 @@ const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const {
   return &_fragmentation;
 }
 
-void VCMEncodedFrame::VerifyAndAllocate(const uint32_t minimumSize)
-{
-    if(minimumSize > _size)
-    {
-        // create buffer of sufficient size
-        uint8_t* newBuffer = new uint8_t[minimumSize];
-        if(_buffer)
-        {
-            // copy old data
-            memcpy(newBuffer, _buffer, _size);
-            delete [] _buffer;
-        }
-        _buffer = newBuffer;
-        _size = minimumSize;
+void VCMEncodedFrame::VerifyAndAllocate(size_t minimumSize) {
+  if (minimumSize > _size) {
+    // create buffer of sufficient size
+    uint8_t* newBuffer = new uint8_t[minimumSize];
+    if (_buffer) {
+      // copy old data
+      memcpy(newBuffer, _buffer, _size);
+      delete[] _buffer;
     }
-}
-
-webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
-{
-  switch(frameType) {
-    case kKeyFrame:
-      return  kVideoFrameKey;
-    case kDeltaFrame:
-      return kVideoFrameDelta;
-    case kSkipFrame:
-      return kFrameEmpty;
-    default:
-      return kVideoFrameDelta;
+    _buffer = newBuffer;
+    _size = minimumSize;
   }
 }
 
-VideoFrameType VCMEncodedFrame::ConvertFrameType(webrtc::FrameType frame_type) {
-  switch (frame_type) {
-    case kVideoFrameKey:
-      return kKeyFrame;
-    case kVideoFrameDelta:
-      return kDeltaFrame;
-    default:
-      assert(false);
-      return kDeltaFrame;
-  }
-}
-
-void VCMEncodedFrame::ConvertFrameTypes(
-    const std::vector<webrtc::FrameType>& frame_types,
-    std::vector<VideoFrameType>* video_frame_types) {
-  assert(video_frame_types);
-  video_frame_types->reserve(frame_types.size());
-  for (size_t i = 0; i < frame_types.size(); ++i) {
-    (*video_frame_types)[i] = ConvertFrameType(frame_types[i]);
-  }
-}
-
-}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.h b/media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.h
new file mode 100644
index 0000000000..9034200980
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/encoded_frame.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/include/video_image.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+
+namespace webrtc {
+
+class VCMEncodedFrame : protected EncodedImage {
+ public:
+  VCMEncodedFrame();
+  explicit VCMEncodedFrame(const webrtc::EncodedImage& rhs);
+  VCMEncodedFrame(const VCMEncodedFrame& rhs);
+
+  ~VCMEncodedFrame();
+  /**
+  *   Delete VideoFrame and resets members to zero
+  */
+  void Free();
+  /**
+  *   Set render time in milliseconds
+  */
+  void SetRenderTime(const int64_t renderTimeMs) {
+    _renderTimeMs = renderTimeMs;
+  }
+
+  /**
+  *   Set the encoded frame size
+  */
+  void SetEncodedSize(uint32_t width, uint32_t height) {
+    _encodedWidth = width;
+    _encodedHeight = height;
+  }
+  /**
+  *   Get the encoded image
+  */
+  const webrtc::EncodedImage& EncodedImage() const {
+    return static_cast<const webrtc::EncodedImage&>(*this);
+  }
+  /**
+  *   Get pointer to frame buffer
+  */
+  const uint8_t* Buffer() const { return _buffer; }
+  /**
+  *   Get frame length
+  */
+  size_t Length() const { return _length; }
+  /**
+  *   Get frame timestamp (90kHz)
+  */
+  uint32_t TimeStamp() const { return _timeStamp; }
+  /**
+  *   Get render time in milliseconds
+  */
+  int64_t RenderTimeMs() const { return _renderTimeMs; }
+  /**
+  *   Get frame type
+  */
+  webrtc::FrameType FrameType() const { return _frameType; }
+  /**
+  *   Get frame rotation
+  */
+  VideoRotation rotation() const { return _rotation; }
+  /**
+  *   True if this frame is complete, false otherwise
+  */
+  bool Complete() const { return _completeFrame; }
+  /**
+  *   True if there's a frame missing before this frame
+  */
+  bool MissingFrame() const { return _missingFrame; }
+  /**
+  *   Payload type of the encoded payload
+  */
+  uint8_t PayloadType() const { return _payloadType; }
+  /**
+  *   Get codec specific info.
+  *   The returned pointer is only valid as long as the VCMEncodedFrame
+  *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
+  *   the object.
+  */
+  const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
+
+  const RTPFragmentationHeader* FragmentationHeader() const;
+
+ protected:
+  /**
+  * Verifies that current allocated buffer size is larger than or equal to the
+  * input size.
+  * If the current buffer size is smaller, a new allocation is made and the old
+  * buffer data
+  * is copied to the new buffer.
+  * Buffer size is updated to minimumSize.
+  */
+  void VerifyAndAllocate(size_t minimumSize);
+
+  void Reset();
+
+  void CopyCodecSpecific(const RTPVideoHeader* header);
+
+  int64_t _renderTimeMs;
+  uint8_t _payloadType;
+  bool _missingFrame;
+  CodecSpecificInfo _codecSpecificInfo;
+  webrtc::VideoCodecType _codec;
+  RTPFragmentationHeader _fragmentation;
+  VideoRotation _rotation;
+
+  // Video rotation is only set along with the last packet for each frame
+  // (same as marker bit). This |_rotation_set| is only for debugging purpose
+  // to ensure we don't set it twice for a frame.
+  bool _rotation_set;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/fec_tables_xor.h b/media/webrtc/trunk/webrtc/modules/video_coding/fec_tables_xor.h
new file mode 100644
index 0000000000..fa5bd7bde4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/fec_tables_xor.h
@@ -0,0 +1,459 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FEC_TABLES_XOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FEC_TABLES_XOR_H_
+
+// This is a private header for media_opt_util.cc.
+// It should not be included by other files.
+
+namespace webrtc {
+
+// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
+// Input is the packet loss and an effective rate (bits/frame).
+// Output is array kCodeRateXORTable[k], where k = rate_i*129 + loss_j;
+// loss_j = 0,1,..128, and rate_i varies over some range.
+static const int kSizeCodeRateXORTable = 6450;
+static const unsigned char kCodeRateXORTable[kSizeCodeRateXORTable] = {
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   11,  11,  11,  11,  11,  11,  11,  11,  11,  11,  11,  11,
+    11,  11,  11,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  51,  51,  51,  51,  51,  51,  51,  51,  51,
+    51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,
+    51,  51,  51,  51,  51,  51,  51,  51,  51,  0,   0,   0,   0,   0,   0,
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   8,   8,   8,
+    8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   30,  30,  30,
+    30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  56,  56,  56,
+    56,  56,  56,  56,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,
+    87,  87,  87,  87,  87,  87,  87,  87,  87,  78,  78,  78,  78,  78,  78,
+    78,  78,  78,  0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   6,   6,   6,   23,  23,  23,  23,  23,  23,  23,  23,  23,
+    23,  23,  23,  23,  23,  23,  44,  44,  44,  44,  44,  44,  50,  50,  50,
+    50,  50,  50,  50,  50,  50,  68,  68,  68,  68,  68,  68,  68,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  105, 105, 105, 105, 105, 105,
+    105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105,
+    105, 105, 105, 88,  88,  88,  88,  88,  88,  88,  88,  88,  0,   0,   0,
+    0,   0,   0,   0,   0,   0,   5,   5,   5,   5,   5,   5,   19,  19,  19,
+    36,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,
+    55,  55,  55,  55,  55,  55,  69,  69,  69,  69,  69,  69,  69,  69,  69,
+    75,  75,  80,  80,  80,  80,  80,  97,  97,  97,  97,  97,  97,  97,  97,
+    97,  97,  102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116,
+    116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 0,   0,   0,   0,   0,   0,   0,   0,   4,
+    16,  16,  16,  16,  16,  16,  30,  35,  35,  47,  58,  58,  58,  58,  58,
+    58,  58,  58,  58,  58,  58,  58,  58,  58,  63,  63,  63,  63,  63,  63,
+    77,  77,  77,  77,  77,  77,  77,  82,  82,  82,  82,  94,  94,  94,  94,
+    94,  105, 105, 105, 105, 110, 110, 110, 110, 110, 110, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 115, 115, 115, 115, 115, 115, 115, 115, 115,
+    0,   0,   0,   0,   0,   0,   0,   4,   14,  27,  27,  27,  27,  27,  31,
+    41,  52,  52,  56,  69,  69,  69,  69,  69,  69,  69,  69,  69,  69,  69,
+    69,  69,  69,  69,  69,  69,  69,  69,  69,  79,  79,  79,  79,  83,  83,
+    83,  94,  94,  94,  94,  106, 106, 106, 106, 106, 115, 115, 115, 115, 125,
+    125, 125, 125, 125, 125, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,   0,   3,   3,
+    3,   17,  28,  38,  38,  38,  38,  38,  47,  51,  63,  63,  63,  72,  72,
+    72,  72,  72,  72,  72,  76,  76,  76,  76,  80,  80,  80,  80,  80,  80,
+    80,  80,  80,  84,  84,  84,  84,  93,  93,  93,  105, 105, 105, 105, 114,
+    114, 114, 114, 114, 124, 124, 124, 124, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   0,   0,   0,   12,  12,  12,  35,  43,  47,  47,  47,
+    47,  47,  58,  58,  66,  66,  66,  70,  70,  70,  70,  70,  73,  73,  82,
+    82,  82,  86,  94,  94,  94,  94,  94,  94,  94,  94,  94,  94,  94,  94,
+    94,  105, 105, 105, 114, 114, 114, 114, 117, 117, 117, 117, 117, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,
+    0,   24,  24,  24,  49,  53,  53,  53,  53,  53,  53,  61,  61,  64,  64,
+    64,  64,  70,  70,  70,  70,  78,  78,  88,  88,  88,  96,  106, 106, 106,
+    106, 106, 106, 106, 106, 106, 106, 112, 112, 112, 120, 120, 120, 124, 124,
+    124, 124, 124, 124, 124, 124, 124, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   0,   0,   5,   36,  36,  36,  55,  55,
+    55,  55,  55,  55,  55,  58,  58,  58,  58,  58,  64,  78,  78,  78,  78,
+    87,  87,  94,  94,  94,  103, 110, 110, 110, 110, 110, 110, 110, 110, 116,
+    116, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   0,   0,   18,  43,  43,  43,  53,  53,  53,  53,  53,  53,  53,  53,
+    58,  58,  58,  58,  71,  87,  87,  87,  87,  94,  94,  97,  97,  97,  109,
+    111, 111, 111, 111, 111, 111, 111, 111, 125, 125, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,   31,  46,  46,
+    46,  48,  48,  48,  48,  48,  48,  48,  48,  66,  66,  66,  66,  80,  93,
+    93,  93,  93,  95,  95,  95,  95,  100, 115, 115, 115, 115, 115, 115, 115,
+    115, 115, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   18,  30,  36,  48,  55,  55,  67,  73,  73,  73,  73,  97,  97,  110,
+    110, 110, 110, 110, 110, 110, 110, 110, 110, 110, 116, 127, 127, 127, 127,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   18,  34,  36,  52,  55,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   17,  34,
+    36,  51,  61,  62,  70,  70,  80,  80,  80,  93,  103, 103, 103, 103, 103,
+    112, 112, 112, 112, 112, 116, 118, 124, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   16,  33,  39,  50,  59,  65,  72,  72,
+    82,  82,  82,  91,  100, 100, 100, 100, 100, 109, 109, 109, 109, 109, 121,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   16,  32,  43,  48,  54,  66,  75,  75,  81,  83,  83,  92,  97,  97,
+    97,  99,  99,  105, 105, 105, 105, 105, 123, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   15,  31,  46,  47,  49,
+    69,  77,  77,  81,  85,  85,  93,  95,  95,  95,  100, 100, 102, 102, 102,
+    102, 102, 120, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   15,  30,  46,  48,  48,  70,  75,  79,  82,  87,  87,
+    92,  94,  94,  94,  103, 103, 103, 103, 103, 104, 104, 115, 120, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   15,  30,
+    45,  50,  50,  68,  70,  80,  85,  89,  89,  90,  95,  95,  95,  104, 104,
+    104, 104, 104, 109, 109, 112, 114, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   14,  29,  44,  54,  54,  64,  64,  83,
+    87,  88,  88,  88,  98,  98,  98,  103, 103, 103, 103, 103, 113, 113, 113,
+    113, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   14,  29,  43,  56,  56,  61,  61,  84,  85,  88,  88,  88,  100, 100,
+    100, 102, 102, 102, 102, 102, 113, 116, 116, 116, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   14,  28,  42,  57,  57,
+    62,  62,  80,  80,  91,  91,  91,  100, 100, 100, 100, 100, 100, 100, 100,
+    109, 119, 119, 119, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   14,  28,  42,  56,  56,  65,  66,  76,  76,  92,  92,
+    92,  97,  97,  97,  101, 101, 101, 101, 101, 106, 121, 121, 121, 126, 126,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   13,  27,
+    41,  55,  55,  67,  72,  74,  74,  90,  90,  90,  91,  91,  91,  105, 105,
+    105, 105, 105, 107, 122, 122, 122, 123, 123, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   13,  27,  40,  54,  54,  67,  76,  76,
+    76,  85,  85,  85,  85,  85,  85,  112, 112, 112, 112, 112, 112, 121, 121,
+    121, 121, 121, 126, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_FEC_TABLES_XOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.cc b/media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.cc
new file mode 100644
index 0000000000..01a0b24e33
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/frame_buffer.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/packet.h"
+
+namespace webrtc {
+
+VCMFrameBuffer::VCMFrameBuffer()
+    : _state(kStateEmpty), _nackCount(0), _latestPacketTimeMs(-1) {}
+
+VCMFrameBuffer::~VCMFrameBuffer() {}
+
+VCMFrameBuffer::VCMFrameBuffer(const VCMFrameBuffer& rhs)
+    : VCMEncodedFrame(rhs),
+      _state(rhs._state),
+      _sessionInfo(),
+      _nackCount(rhs._nackCount),
+      _latestPacketTimeMs(rhs._latestPacketTimeMs) {
+  _sessionInfo = rhs._sessionInfo;
+  _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
+}
+
+webrtc::FrameType VCMFrameBuffer::FrameType() const {
+  return _sessionInfo.FrameType();
+}
+
+int32_t VCMFrameBuffer::GetLowSeqNum() const {
+  return _sessionInfo.LowSequenceNumber();
+}
+
+int32_t VCMFrameBuffer::GetHighSeqNum() const {
+  return _sessionInfo.HighSequenceNumber();
+}
+
+int VCMFrameBuffer::PictureId() const {
+  return _sessionInfo.PictureId();
+}
+
+int VCMFrameBuffer::TemporalId() const {
+  return _sessionInfo.TemporalId();
+}
+
+bool VCMFrameBuffer::LayerSync() const {
+  return _sessionInfo.LayerSync();
+}
+
+int VCMFrameBuffer::Tl0PicId() const {
+  return _sessionInfo.Tl0PicId();
+}
+
+bool VCMFrameBuffer::NonReference() const {
+  return _sessionInfo.NonReference();
+}
+
+void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
+  _sessionInfo.SetGofInfo(gof_info, idx);
+  // TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
+  _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
+      gof_info.temporal_idx[idx];
+  _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
+      gof_info.temporal_up_switch[idx];
+}
+
+bool VCMFrameBuffer::IsSessionComplete() const {
+  return _sessionInfo.complete();
+}
+
+// Insert packet
+VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(
+    const VCMPacket& packet,
+    int64_t timeInMs,
+    VCMDecodeErrorMode decode_error_mode,
+    const FrameData& frame_data) {
+  assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
+  if (packet.dataPtr != NULL) {
+    _payloadType = packet.payloadType;
+  }
+
+  if (kStateEmpty == _state) {
+    // First packet (empty and/or media) inserted into this frame.
+    // store some info and set some initial values.
+    _timeStamp = packet.timestamp;
+    // We only take the ntp timestamp of the first packet of a frame.
+    ntp_time_ms_ = packet.ntp_time_ms_;
+    _codec = packet.codec;
+    if (packet.frameType != kEmptyFrame) {
+      // first media packet
+      SetState(kStateIncomplete);
+    }
+  }
+
+    // add safety margin because STAP-A packets can cause it to expand by
+    // ~two bytes per NAL
+  uint32_t requiredSizeBytes = Length() + packet.sizeBytes +
+                   (packet.insertStartCode ? kH264StartCodeLengthBytes : 0) +
+                                 kBufferSafetyMargin;
+  if (requiredSizeBytes >= _size) {
+    const uint8_t* prevBuffer = _buffer;
+    const uint32_t increments =
+        requiredSizeBytes / kBufferIncStepSizeBytes +
+        (requiredSizeBytes % kBufferIncStepSizeBytes > 0);
+    const uint32_t newSize = _size + increments * kBufferIncStepSizeBytes;
+    if (newSize > kMaxJBFrameSizeBytes) {
+      LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
+                       "big.";
+      return kSizeError;
+    }
+    VerifyAndAllocate(newSize);
+    _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
+  }
+
+  if (packet.width > 0 && packet.height > 0) {
+    _encodedWidth = packet.width;
+    _encodedHeight = packet.height;
+  }
+
+  // Don't copy payload specific data for empty packets (e.g padding packets).
+  if (packet.sizeBytes > 0)
+    CopyCodecSpecific(&packet.codecSpecificHeader);
+
+  int retVal =
+      _sessionInfo.InsertPacket(packet, _buffer, decode_error_mode, frame_data);
+  if (retVal == -1) {
+    return kSizeError;
+  } else if (retVal == -2) {
+    return kDuplicatePacket;
+  } else if (retVal == -3) {
+    return kOutOfBoundsPacket;
+  }
+  // update length
+  _length = Length() + static_cast<uint32_t>(retVal);
+
+  _latestPacketTimeMs = timeInMs;
+
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+  // ts_126114v120700p.pdf Section 7.4.5.
+  // The MTSI client shall add the payload bytes as defined in this clause
+  // onto the last RTP packet in each group of packets which make up a key
+  // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
+  // (HEVC)).
+  if (packet.markerBit) {
+    RTC_DCHECK(!_rotation_set);
+    _rotation = packet.codecSpecificHeader.rotation;
+    _rotation_set = true;
+  }
+
+  if (_sessionInfo.complete()) {
+    SetState(kStateComplete);
+    return kCompleteSession;
+  } else if (_sessionInfo.decodable()) {
+    SetState(kStateDecodable);
+    return kDecodableSession;
+  }
+  return kIncomplete;
+}
+
+int64_t VCMFrameBuffer::LatestPacketTimeMs() const {
+  return _latestPacketTimeMs;
+}
+
+void VCMFrameBuffer::IncrementNackCount() {
+  _nackCount++;
+}
+
+int16_t VCMFrameBuffer::GetNackCount() const {
+  return _nackCount;
+}
+
+bool VCMFrameBuffer::HaveFirstPacket() const {
+  return _sessionInfo.HaveFirstPacket();
+}
+
+bool VCMFrameBuffer::HaveLastPacket() const {
+  return _sessionInfo.HaveLastPacket();
+}
+
+int VCMFrameBuffer::NumPackets() const {
+  return _sessionInfo.NumPackets();
+}
+
+void VCMFrameBuffer::Reset() {
+  _length = 0;
+  _timeStamp = 0;
+  _sessionInfo.Reset();
+  _payloadType = 0;
+  _nackCount = 0;
+  _latestPacketTimeMs = -1;
+  _state = kStateEmpty;
+  VCMEncodedFrame::Reset();
+}
+
+// Set state of frame
+void VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
+  if (_state == state) {
+    return;
+  }
+  switch (state) {
+    case kStateIncomplete:
+      // we can go to this state from state kStateEmpty
+      assert(_state == kStateEmpty);
+
+      // Do nothing, we received a packet
+      break;
+
+    case kStateComplete:
+      assert(_state == kStateEmpty || _state == kStateIncomplete ||
+             _state == kStateDecodable);
+
+      break;
+
+    case kStateEmpty:
+      // Should only be set to empty through Reset().
+      assert(false);
+      break;
+
+    case kStateDecodable:
+      assert(_state == kStateEmpty || _state == kStateIncomplete);
+      break;
+  }
+  _state = state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum VCMFrameBuffer::GetState() const {
+  return _state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum VCMFrameBuffer::GetState(uint32_t& timeStamp) const {
+  timeStamp = TimeStamp();
+  return GetState();
+}
+
+bool VCMFrameBuffer::IsRetransmitted() const {
+  return _sessionInfo.session_nack();
+}
+
+void VCMFrameBuffer::PrepareForDecode(bool continuous) {
+#ifdef INDEPENDENT_PARTITIONS
+  if (_codec == kVideoCodecVP8) {
+    _length = _sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
+                                                       &_fragmentation);
+  } else {
+    size_t bytes_removed = _sessionInfo.MakeDecodable();
+    _length -= bytes_removed;
+  }
+#else
+  size_t bytes_removed = _sessionInfo.MakeDecodable();
+  _length -= bytes_removed;
+#endif
+  // Transfer frame information to EncodedFrame and create any codec
+  // specific information.
+  _frameType = _sessionInfo.FrameType();
+  _completeFrame = _sessionInfo.complete();
+  _missingFrame = !continuous;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.h b/media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.h
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.h
index 3af85f31a1..f5a707efe4 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/session_info.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/session_info.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -75,19 +75,18 @@ class VCMFrameBuffer : public VCMEncodedFrame {
   webrtc::FrameType FrameType() const;
   void SetPreviousFrameLoss();
 
-  // The number of packets discarded because the decoder can't make use of
-  // them.
+  // The number of packets discarded because the decoder can't make use of them.
   int NotDecodablePackets() const;
 
  private:
   void SetState(VCMFrameBufferStateEnum state);  // Set state of frame
 
-  VCMFrameBufferStateEnum    _state;         // Current state of the frame
-  VCMSessionInfo             _sessionInfo;
-  uint16_t             _nackCount;
-  int64_t              _latestPacketTimeMs;
+  VCMFrameBufferStateEnum _state;  // Current state of the frame
+  VCMSessionInfo _sessionInfo;
+  uint16_t _nackCount;
+  int64_t _latestPacketTimeMs;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/generic_decoder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/generic_decoder.cc
new file mode 100644
index 0000000000..5cbe0f5ba0
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/generic_decoder.cc
@@ -0,0 +1,192 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/generic_decoder.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming* timing,
+                                                 Clock* clock)
+    : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
+      _clock(clock),
+      _receiveCallback(NULL),
+      _timing(timing),
+      _timestampMap(kDecoderFrameMemoryLength),
+      _lastReceivedPictureID(0) {}
+
+VCMDecodedFrameCallback::~VCMDecodedFrameCallback() {
+  delete _critSect;
+}
+
+void VCMDecodedFrameCallback::SetUserReceiveCallback(
+    VCMReceiveCallback* receiveCallback) {
+  CriticalSectionScoped cs(_critSect);
+  _receiveCallback = receiveCallback;
+}
+
+VCMReceiveCallback* VCMDecodedFrameCallback::UserReceiveCallback() {
+  CriticalSectionScoped cs(_critSect);
+  return _receiveCallback;
+}
+
+int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage) {
+  return Decoded(decodedImage, -1);
+}
+
+int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
+                                         int64_t decode_time_ms) {
+  TRACE_EVENT_INSTANT1("webrtc", "VCMDecodedFrameCallback::Decoded",
+                       "timestamp", decodedImage.timestamp());
+  // TODO(holmer): We should improve this so that we can handle multiple
+  // callbacks from one call to Decode().
+  VCMFrameInformation* frameInfo;
+  VCMReceiveCallback* callback;
+  {
+    CriticalSectionScoped cs(_critSect);
+    frameInfo = _timestampMap.Pop(decodedImage.timestamp());
+    callback = _receiveCallback;
+  }
+
+  if (frameInfo == NULL) {
+    LOG(LS_WARNING) << "Too many frames backed up in the decoder, dropping "
+                       "this one.";
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  const int64_t now_ms = _clock->TimeInMilliseconds();
+  if (decode_time_ms < 0) {
+    decode_time_ms =
+        static_cast<int32_t>(now_ms - frameInfo->decodeStartTimeMs);
+  }
+  _timing->StopDecodeTimer(decodedImage.timestamp(), decode_time_ms, now_ms,
+                           frameInfo->renderTimeMs);
+
+  if (callback != NULL) {
+    decodedImage.set_render_time_ms(frameInfo->renderTimeMs);
+    decodedImage.set_rotation(frameInfo->rotation);
+    callback->FrameToRender(decodedImage);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
+    const uint64_t pictureId) {
+  CriticalSectionScoped cs(_critSect);
+  if (_receiveCallback != NULL) {
+    return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
+  }
+  return -1;
+}
+
+int32_t VCMDecodedFrameCallback::ReceivedDecodedFrame(
+    const uint64_t pictureId) {
+  _lastReceivedPictureID = pictureId;
+  return 0;
+}
+
+uint64_t VCMDecodedFrameCallback::LastReceivedPictureID() const {
+  return _lastReceivedPictureID;
+}
+
+void VCMDecodedFrameCallback::OnDecoderImplementationName(
+    const char* implementation_name) {
+  CriticalSectionScoped cs(_critSect);
+  if (_receiveCallback)
+    _receiveCallback->OnDecoderImplementationName(implementation_name);
+}
+
+void VCMDecodedFrameCallback::Map(uint32_t timestamp,
+                                  VCMFrameInformation* frameInfo) {
+  CriticalSectionScoped cs(_critSect);
+  _timestampMap.Add(timestamp, frameInfo);
+}
+
+int32_t VCMDecodedFrameCallback::Pop(uint32_t timestamp) {
+  CriticalSectionScoped cs(_critSect);
+  if (_timestampMap.Pop(timestamp) == NULL) {
+    return VCM_GENERAL_ERROR;
+  }
+  return VCM_OK;
+}
+
+VCMGenericDecoder::VCMGenericDecoder(VideoDecoder* decoder, bool isExternal)
+    : _callback(NULL),
+      _frameInfos(),
+      _nextFrameInfoIdx(0),
+      _decoder(decoder),
+      _codecType(kVideoCodecUnknown),
+      _isExternal(isExternal),
+      _keyFrameDecoded(false) {}
+
+VCMGenericDecoder::~VCMGenericDecoder() {}
+
+int32_t VCMGenericDecoder::InitDecode(const VideoCodec* settings,
+                                      int32_t numberOfCores) {
+  TRACE_EVENT0("webrtc", "VCMGenericDecoder::InitDecode");
+  _codecType = settings->codecType;
+
+  return _decoder->InitDecode(settings, numberOfCores);
+}
+
+int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame, int64_t nowMs) {
+    TRACE_EVENT1("webrtc", "VCMGenericDecoder::Decode", "timestamp",
+                 frame.EncodedImage()._timeStamp);
+    _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
+    _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
+    _frameInfos[_nextFrameInfoIdx].rotation = frame.rotation();
+    _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
+
+    _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
+    int32_t ret = _decoder->Decode(frame.EncodedImage(), frame.MissingFrame(),
+                                   frame.FragmentationHeader(),
+                                   frame.CodecSpecific(), frame.RenderTimeMs());
+
+    _callback->OnDecoderImplementationName(_decoder->ImplementationName());
+    if (ret < WEBRTC_VIDEO_CODEC_OK) {
+        LOG(LS_WARNING) << "Failed to decode frame with timestamp "
+                        << frame.TimeStamp() << ", error code: " << ret;
+        _callback->Pop(frame.TimeStamp());
+        return ret;
+    } else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
+               ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
+        // No output
+        _callback->Pop(frame.TimeStamp());
+    }
+    return ret;
+}
+
+int32_t VCMGenericDecoder::Release() {
+  return _decoder->Release();
+}
+
+int32_t VCMGenericDecoder::Reset() {
+  return _decoder->Reset();
+}
+
+int32_t VCMGenericDecoder::RegisterDecodeCompleteCallback(
+    VCMDecodedFrameCallback* callback) {
+  _callback = callback;
+  return _decoder->RegisterDecodeCompleteCallback(callback);
+}
+
+bool VCMGenericDecoder::External() const {
+  return _isExternal;
+}
+
+bool VCMGenericDecoder::PrefersLateDecoding() const {
+  return _decoder->PrefersLateDecoding();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/generic_decoder.h b/media/webrtc/trunk/webrtc/modules/video_coding/generic_decoder.h
new file mode 100644
index 0000000000..bbf24f2bbb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/generic_decoder.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/timestamp_map.h"
+#include "webrtc/modules/video_coding/timing.h"
+
+namespace webrtc {
+
+class VCMReceiveCallback;
+
+enum { kDecoderFrameMemoryLength = 30 };
+
+struct VCMFrameInformation {
+  int64_t renderTimeMs;
+  int64_t decodeStartTimeMs;
+  void* userData;
+  VideoRotation rotation;
+};
+
+class VCMDecodedFrameCallback : public DecodedImageCallback {
+ public:
+  VCMDecodedFrameCallback(VCMTiming* timing, Clock* clock);
+    virtual ~VCMDecodedFrameCallback();
+    void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
+    VCMReceiveCallback* UserReceiveCallback();
+
+    virtual int32_t Decoded(VideoFrame& decodedImage);  // NOLINT
+    virtual int32_t Decoded(VideoFrame& decodedImage,   // NOLINT
+                            int64_t decode_time_ms);
+    virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
+    virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId);
+
+    uint64_t LastReceivedPictureID() const;
+    void OnDecoderImplementationName(const char* implementation_name);
+
+    void Map(uint32_t timestamp, VCMFrameInformation* frameInfo);
+    int32_t Pop(uint32_t timestamp);
+
+ private:
+    // Protect |_receiveCallback| and |_timestampMap|.
+    CriticalSectionWrapper* _critSect;
+    Clock* _clock;
+    VCMReceiveCallback* _receiveCallback GUARDED_BY(_critSect);
+    VCMTiming* _timing;
+    VCMTimestampMap _timestampMap GUARDED_BY(_critSect);
+    uint64_t _lastReceivedPictureID;
+};
+
+class VCMGenericDecoder {
+  friend class VCMCodecDataBase;
+
+ public:
+  explicit VCMGenericDecoder(VideoDecoder* decoder, bool isExternal = false);
+  ~VCMGenericDecoder();
+
+  /**
+  * Initialize the decoder with the information from the VideoCodec
+  */
+  int32_t InitDecode(const VideoCodec* settings, int32_t numberOfCores);
+
+  /**
+  * Decode to a raw I420 frame,
+  *
+  * inputVideoBuffer reference to encoded video frame
+  */
+  int32_t Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
+
+  /**
+  * Free the decoder memory
+  */
+  int32_t Release();
+
+  /**
+  * Reset the decoder state, prepare for a new call
+  */
+  int32_t Reset();
+
+  /**
+  * Set decode callback. Deregistering while decoding is illegal.
+  */
+  int32_t RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
+
+  bool External() const;
+  bool PrefersLateDecoding() const;
+
+ private:
+  VCMDecodedFrameCallback* _callback;
+  VCMFrameInformation _frameInfos[kDecoderFrameMemoryLength];
+  uint32_t _nextFrameInfoIdx;
+  VideoDecoder* const _decoder;
+  VideoCodecType _codecType;
+  bool _isExternal;
+  bool _keyFrameDecoded;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/generic_encoder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/generic_encoder.cc
new file mode 100644
index 0000000000..a7dbc27a08
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/generic_encoder.cc
@@ -0,0 +1,325 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/generic_encoder.h"
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+namespace {
+// Map information from info into rtp. If no relevant information is found
+// in info, rtp is set to NULL.
+void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
+  RTC_DCHECK(info);
+  switch (info->codecType) {
+    case kVideoCodecVP8: {
+      rtp->codec = kRtpVideoVp8;
+      rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
+      rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
+      rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference;
+      rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
+      rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
+      rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx;
+      rtp->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx;
+      rtp->simulcastIdx = info->codecSpecific.VP8.simulcastIdx;
+      return;
+    }
+    case kVideoCodecVP9: {
+      rtp->codec = kRtpVideoVp9;
+      rtp->codecHeader.VP9.InitRTPVideoHeaderVP9();
+      rtp->codecHeader.VP9.inter_pic_predicted =
+          info->codecSpecific.VP9.inter_pic_predicted;
+      rtp->codecHeader.VP9.flexible_mode =
+          info->codecSpecific.VP9.flexible_mode;
+      rtp->codecHeader.VP9.ss_data_available =
+          info->codecSpecific.VP9.ss_data_available;
+      rtp->codecHeader.VP9.picture_id = info->codecSpecific.VP9.picture_id;
+      rtp->codecHeader.VP9.tl0_pic_idx = info->codecSpecific.VP9.tl0_pic_idx;
+      rtp->codecHeader.VP9.temporal_idx = info->codecSpecific.VP9.temporal_idx;
+      rtp->codecHeader.VP9.spatial_idx = info->codecSpecific.VP9.spatial_idx;
+      rtp->codecHeader.VP9.temporal_up_switch =
+          info->codecSpecific.VP9.temporal_up_switch;
+      rtp->codecHeader.VP9.inter_layer_predicted =
+          info->codecSpecific.VP9.inter_layer_predicted;
+      rtp->codecHeader.VP9.gof_idx = info->codecSpecific.VP9.gof_idx;
+      rtp->codecHeader.VP9.num_spatial_layers =
+          info->codecSpecific.VP9.num_spatial_layers;
+
+      if (info->codecSpecific.VP9.ss_data_available) {
+        rtp->codecHeader.VP9.spatial_layer_resolution_present =
+            info->codecSpecific.VP9.spatial_layer_resolution_present;
+        if (info->codecSpecific.VP9.spatial_layer_resolution_present) {
+          for (size_t i = 0; i < info->codecSpecific.VP9.num_spatial_layers;
+               ++i) {
+            rtp->codecHeader.VP9.width[i] = info->codecSpecific.VP9.width[i];
+            rtp->codecHeader.VP9.height[i] = info->codecSpecific.VP9.height[i];
+          }
+        }
+        rtp->codecHeader.VP9.gof.CopyGofInfoVP9(info->codecSpecific.VP9.gof);
+      }
+
+      rtp->codecHeader.VP9.num_ref_pics = info->codecSpecific.VP9.num_ref_pics;
+      for (int i = 0; i < info->codecSpecific.VP9.num_ref_pics; ++i)
+        rtp->codecHeader.VP9.pid_diff[i] = info->codecSpecific.VP9.p_diff[i];
+      return;
+    }
+    case kVideoCodecH264:
+      rtp->codec = kRtpVideoH264;
+      rtp->codecHeader.H264.packetization_mode = info->codecSpecific.H264.packetizationMode;
+      rtp->codecHeader.H264.single_nalu = info->codecSpecific.H264.single_nalu;
+      rtp->simulcastIdx = info->codecSpecific.H264.simulcastIdx;
+      return;
+    case kVideoCodecGeneric:
+      rtp->codec = kRtpVideoGeneric;
+      rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
+      return;
+    default:
+      return;
+  }
+}
+}  // namespace
+
+// #define DEBUG_ENCODER_BIT_STREAM
+
+VCMGenericEncoder::VCMGenericEncoder(
+    VideoEncoder* encoder,
+    VideoEncoderRateObserver* rate_observer,
+    VCMEncodedFrameCallback* encoded_frame_callback,
+    bool internalSource)
+    : encoder_(encoder),
+      rate_observer_(rate_observer),
+      vcm_encoded_frame_callback_(encoded_frame_callback),
+      internal_source_(internalSource),
+      encoder_params_({0, 0, 0, 0}),
+      rotation_(kVideoRotation_0),
+      is_screenshare_(false) {}
+
+VCMGenericEncoder::~VCMGenericEncoder() {}
+
+int32_t VCMGenericEncoder::Release() {
+  encoder_->RegisterEncodeCompleteCallback(nullptr);
+  return encoder_->Release();
+}
+
+int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
+                                      int32_t numberOfCores,
+                                      size_t maxPayloadSize) {
+  TRACE_EVENT0("webrtc", "VCMGenericEncoder::InitEncode");
+  {
+    rtc::CritScope lock(&params_lock_);
+    encoder_params_.target_bitrate = settings->startBitrate * 1000;
+    encoder_params_.input_frame_rate = settings->maxFramerate;
+  }
+
+  is_screenshare_ = settings->mode == VideoCodecMode::kScreensharing;
+  if (encoder_->InitEncode(settings, numberOfCores, maxPayloadSize) != 0) {
+    LOG(LS_ERROR) << "Failed to initialize the encoder associated with "
+                     "payload name: "
+                  << settings->plName;
+    return -1;
+  }
+  encoder_->RegisterEncodeCompleteCallback(vcm_encoded_frame_callback_);
+  return 0;
+}
+
+int32_t VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
+                                  const CodecSpecificInfo* codecSpecificInfo,
+                                  const std::vector<FrameType>& frameTypes) {
+  TRACE_EVENT1("webrtc", "VCMGenericEncoder::Encode", "timestamp",
+               inputFrame.timestamp());
+
+  for (FrameType frame_type : frameTypes)
+    RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta);
+
+  rotation_ = inputFrame.rotation();
+
+  // Keep track of the current frame rotation and apply to the output of the
+  // encoder. There might not be exact as the encoder could have one frame delay
+  // but it should be close enough.
+  // TODO(pbos): Map from timestamp, this is racy (even if rotation_ is locked
+  // properly, which it isn't). More than one frame may be in the pipeline.
+  vcm_encoded_frame_callback_->SetRotation(rotation_);
+
+  int32_t result = encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
+
+  if (vcm_encoded_frame_callback_) {
+    vcm_encoded_frame_callback_->SignalLastEncoderImplementationUsed(
+        encoder_->ImplementationName());
+  }
+
+  if (is_screenshare_ &&
+      result == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT) {
+    // Target bitrate exceeded, encoder state has been reset - try again.
+    return encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
+  }
+
+  return result;
+}
+
+void VCMGenericEncoder::SetEncoderParameters(const EncoderParameters& params) {
+  bool channel_parameters_have_changed;
+  bool rates_have_changed;
+  {
+    rtc::CritScope lock(&params_lock_);
+    channel_parameters_have_changed =
+        params.loss_rate != encoder_params_.loss_rate ||
+        params.rtt != encoder_params_.rtt;
+    rates_have_changed =
+        params.target_bitrate != encoder_params_.target_bitrate ||
+        params.input_frame_rate != encoder_params_.input_frame_rate;
+    encoder_params_ = params;
+  }
+  if (channel_parameters_have_changed)
+    encoder_->SetChannelParameters(params.loss_rate, params.rtt);
+  if (rates_have_changed) {
+    uint32_t target_bitrate_kbps = (params.target_bitrate + 500) / 1000;
+    encoder_->SetRates(target_bitrate_kbps, params.input_frame_rate);
+    if (rate_observer_ != nullptr) {
+      rate_observer_->OnSetRates(params.target_bitrate,
+                                 params.input_frame_rate);
+    }
+  }
+}
+
+EncoderParameters VCMGenericEncoder::GetEncoderParameters() const {
+  rtc::CritScope lock(&params_lock_);
+  return encoder_params_;
+}
+
+int32_t VCMGenericEncoder::SetPeriodicKeyFrames(bool enable) {
+  return encoder_->SetPeriodicKeyFrames(enable);
+}
+
+int32_t VCMGenericEncoder::RequestFrame(
+    const std::vector<FrameType>& frame_types) {
+  VideoFrame image;
+  return encoder_->Encode(image, NULL, &frame_types);
+}
+
+bool VCMGenericEncoder::InternalSource() const {
+  return internal_source_;
+}
+
+void VCMGenericEncoder::OnDroppedFrame() {
+  encoder_->OnDroppedFrame();
+}
+
+bool VCMGenericEncoder::SupportsNativeHandle() const {
+  return encoder_->SupportsNativeHandle();
+}
+
+int VCMGenericEncoder::GetTargetFramerate() {
+  return encoder_->GetTargetFramerate();
+}
+
+/***************************
+ * Callback Implementation
+ ***************************/
+VCMEncodedFrameCallback::VCMEncodedFrameCallback(
+    EncodedImageCallback* post_encode_callback)
+    : send_callback_(),
+      _critSect(NULL),
+      _mediaOpt(NULL),
+      _payloadType(0),
+      _internalSource(false),
+      _rotation(kVideoRotation_0),
+      post_encode_callback_(post_encode_callback)
+#ifdef DEBUG_ENCODER_BIT_STREAM
+      ,
+      _bitStreamAfterEncoder(NULL)
+#endif
+{
+#ifdef DEBUG_ENCODER_BIT_STREAM
+  _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
+#endif
+}
+
+VCMEncodedFrameCallback::~VCMEncodedFrameCallback() {
+#ifdef DEBUG_ENCODER_BIT_STREAM
+  fclose(_bitStreamAfterEncoder);
+#endif
+}
+ 
+void
+VCMEncodedFrameCallback::SetCritSect(CriticalSectionWrapper* critSect)
+{
+    _critSect = critSect;
+}
+
+int32_t VCMEncodedFrameCallback::SetTransportCallback(
+    VCMPacketizationCallback* transport) {
+  send_callback_ = transport;
+  return VCM_OK;
+}
+
+int32_t VCMEncodedFrameCallback::Encoded(
+    const EncodedImage& encoded_image,
+    const CodecSpecificInfo* codecSpecificInfo,
+    const RTPFragmentationHeader* fragmentationHeader) {
+  TRACE_EVENT_INSTANT1("webrtc", "VCMEncodedFrameCallback::Encoded",
+                       "timestamp", encoded_image._timeStamp);
+  assert(_critSect);
+  CriticalSectionScoped cs(_critSect);
+
+  post_encode_callback_->Encoded(encoded_image, NULL, NULL);
+
+  if (send_callback_ == NULL) {
+    return VCM_UNINITIALIZED;
+  }
+
+#ifdef DEBUG_ENCODER_BIT_STREAM
+  if (_bitStreamAfterEncoder != NULL) {
+    fwrite(encoded_image._buffer, 1, encoded_image._length,
+           _bitStreamAfterEncoder);
+  }
+#endif
+
+  RTPVideoHeader rtpVideoHeader;
+  memset(&rtpVideoHeader, 0, sizeof(RTPVideoHeader));
+  RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader;
+  if (codecSpecificInfo) {
+    CopyCodecSpecific(codecSpecificInfo, rtpVideoHeaderPtr);
+  }
+  rtpVideoHeader.rotation = _rotation;
+
+  int32_t callbackReturn = send_callback_->SendData(
+      _payloadType, encoded_image, *fragmentationHeader, rtpVideoHeaderPtr);
+  if (callbackReturn < 0) {
+    return callbackReturn;
+  }
+
+  if (_mediaOpt != NULL) {
+    _mediaOpt->UpdateWithEncodedData(encoded_image);
+    if (_internalSource)
+      return _mediaOpt->DropFrame();  // Signal to encoder to drop next frame.
+  }
+  return VCM_OK;
+}
+
+void VCMEncodedFrameCallback::SetMediaOpt(
+    media_optimization::MediaOptimization* mediaOpt) {
+  _mediaOpt = mediaOpt;
+}
+
+void VCMEncodedFrameCallback::SignalLastEncoderImplementationUsed(
+    const char* implementation_name) {
+  if (send_callback_)
+    send_callback_->OnEncoderImplementationName(implementation_name);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/generic_encoder.h b/media/webrtc/trunk/webrtc/modules/video_coding/generic_encoder.h
new file mode 100644
index 0000000000..e428d9635b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/generic_encoder.h
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
+
+#include <stdio.h>
+#include <vector>
+
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+namespace media_optimization {
+class MediaOptimization;
+}  // namespace media_optimization
+
+struct EncoderParameters {
+  uint32_t target_bitrate;
+  uint8_t loss_rate;
+  int64_t rtt;
+  uint32_t input_frame_rate;
+};
+
+/*************************************/
+/* VCMEncodeFrameCallback class     */
+/***********************************/
+class VCMEncodedFrameCallback : public EncodedImageCallback {
+ public:
+    explicit VCMEncodedFrameCallback(
+      EncodedImageCallback* post_encode_callback);
+    virtual ~VCMEncodedFrameCallback();
+ 
+    void SetCritSect(CriticalSectionWrapper* critSect);
+
+    /*
+    * Callback implementation - codec encode complete
+    */
+    int32_t Encoded(
+        const EncodedImage& encodedImage,
+        const CodecSpecificInfo* codecSpecificInfo = NULL,
+        const RTPFragmentationHeader* fragmentationHeader = NULL);
+    /*
+    * Callback implementation - generic encoder encode complete
+    */
+    int32_t SetTransportCallback(VCMPacketizationCallback* transport);
+    /**
+    * Set media Optimization
+    */
+    void SetMediaOpt(media_optimization::MediaOptimization* mediaOpt);
+
+    void SetPayloadType(uint8_t payloadType) {
+      _payloadType = payloadType;
+    }
+
+    void SetInternalSource(bool internalSource) {
+      _internalSource = internalSource;
+    }
+
+    void SetRotation(VideoRotation rotation) { _rotation = rotation; }
+    void SignalLastEncoderImplementationUsed(
+        const char* encoder_implementation_name);
+
+ private:
+    VCMPacketizationCallback* send_callback_;
+    CriticalSectionWrapper* _critSect;
+    media_optimization::MediaOptimization* _mediaOpt;
+    uint8_t _payloadType;
+    bool _internalSource;
+    VideoRotation _rotation;
+
+    EncodedImageCallback* post_encode_callback_;
+
+#ifdef DEBUG_ENCODER_BIT_STREAM
+    FILE* _bitStreamAfterEncoder;
+#endif
+};  // end of VCMEncodeFrameCallback class
+
+/******************************/
+/* VCMGenericEncoder class    */
+/******************************/
+class VCMGenericEncoder {
+  friend class VCMCodecDataBase;
+
+ public:
+  VCMGenericEncoder(VideoEncoder* encoder,
+                    VideoEncoderRateObserver* rate_observer,
+                    VCMEncodedFrameCallback* encoded_frame_callback,
+                    bool internalSource);
+  ~VCMGenericEncoder();
+  /**
+  * Free encoder memory
+  */
+  int32_t Release();
+  /**
+  * Initialize the encoder with the information from the VideoCodec
+  */
+  int32_t InitEncode(const VideoCodec* settings,
+                     int32_t numberOfCores,
+                     size_t maxPayloadSize);
+  /**
+  * Encode raw image
+  * inputFrame        : Frame containing raw image
+  * codecSpecificInfo : Specific codec data
+  * cameraFrameRate   : Request or information from the remote side
+  * frameType         : The requested frame type to encode
+  */
+  int32_t Encode(const VideoFrame& inputFrame,
+                 const CodecSpecificInfo* codecSpecificInfo,
+                 const std::vector<FrameType>& frameTypes);
+
+  void SetEncoderParameters(const EncoderParameters& params);
+  EncoderParameters GetEncoderParameters() const;
+
+  int32_t SetPeriodicKeyFrames(bool enable);
+
+  int32_t RequestFrame(const std::vector<FrameType>& frame_types);
+
+  bool InternalSource() const;
+
+  void OnDroppedFrame();
+
+  bool SupportsNativeHandle() const;
+
+  int GetTargetFramerate();
+
+ private:
+  VideoEncoder* const encoder_;
+  VideoEncoderRateObserver* const rate_observer_;
+  VCMEncodedFrameCallback* const vcm_encoded_frame_callback_;
+  const bool internal_source_;
+  mutable rtc::CriticalSection params_lock_;
+  EncoderParameters encoder_params_ GUARDED_BY(params_lock_);
+  VideoRotation rotation_;
+  bool is_screenshare_;
+};  // end of VCMGenericEncoder class
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h b/media/webrtc/trunk/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
similarity index 57%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
index 302d4a3a13..0185dae333 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VCM_CALLBACKS_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VCM_CALLBACKS_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -20,16 +20,15 @@ namespace webrtc {
 class MockVCMFrameTypeCallback : public VCMFrameTypeCallback {
  public:
   MOCK_METHOD0(RequestKeyFrame, int32_t());
-  MOCK_METHOD1(SliceLossIndicationRequest,
-               int32_t(const uint64_t pictureId));
+  MOCK_METHOD1(SliceLossIndicationRequest, int32_t(const uint64_t pictureId));
 };
 
 class MockPacketRequestCallback : public VCMPacketRequestCallback {
  public:
-  MOCK_METHOD2(ResendPackets, int32_t(const uint16_t* sequenceNumbers,
-                                      uint16_t length));
+  MOCK_METHOD2(ResendPackets,
+               int32_t(const uint16_t* sequenceNumbers, uint16_t length));
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VCM_CALLBACKS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h b/media/webrtc/trunk/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
new file mode 100644
index 0000000000..9cb4a83535
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
+
+#include <string>
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class MockEncodedImageCallback : public EncodedImageCallback {
+ public:
+  MOCK_METHOD3(Encoded,
+               int32_t(const EncodedImage& encodedImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const RTPFragmentationHeader* fragmentation));
+};
+
+class MockVideoEncoder : public VideoEncoder {
+ public:
+  MOCK_CONST_METHOD2(Version, int32_t(int8_t* version, int32_t length));
+  MOCK_METHOD3(InitEncode,
+               int32_t(const VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize));
+  MOCK_METHOD3(Encode,
+               int32_t(const VideoFrame& inputImage,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       const std::vector<FrameType>* frame_types));
+  MOCK_METHOD1(RegisterEncodeCompleteCallback,
+               int32_t(EncodedImageCallback* callback));
+  MOCK_METHOD0(Release, int32_t());
+  MOCK_METHOD0(Reset, int32_t());
+  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
+  MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
+  MOCK_METHOD1(SetPeriodicKeyFrames, int32_t(bool enable));
+};
+
+class MockDecodedImageCallback : public DecodedImageCallback {
+ public:
+  MOCK_METHOD1(Decoded, int32_t(VideoFrame& decodedImage));  // NOLINT
+  MOCK_METHOD2(Decoded,
+               int32_t(VideoFrame& decodedImage,  // NOLINT
+                       int64_t decode_time_ms));
+  MOCK_METHOD1(ReceivedDecodedReferenceFrame,
+               int32_t(const uint64_t pictureId));
+  MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
+};
+
+class MockVideoDecoder : public VideoDecoder {
+ public:
+  MOCK_METHOD2(InitDecode,
+               int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
+  MOCK_METHOD5(Decode,
+               int32_t(const EncodedImage& inputImage,
+                       bool missingFrames,
+                       const RTPFragmentationHeader* fragmentation,
+                       const CodecSpecificInfo* codecSpecificInfo,
+                       int64_t renderTimeMs));
+  MOCK_METHOD1(RegisterDecodeCompleteCallback,
+               int32_t(DecodedImageCallback* callback));
+  MOCK_METHOD0(Release, int32_t());
+  MOCK_METHOD0(Reset, int32_t());
+  MOCK_METHOD0(Copy, VideoDecoder*());
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/include/video_codec_interface.h b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_codec_interface.h
new file mode 100644
index 0000000000..493b1363d9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_codec_interface.h
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_decoder.h"
+#include "webrtc/video_encoder.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+
+class RTPFragmentationHeader;  // forward declaration
+
+// Note: if any pointers are added to this struct, it must be fitted
+// with a copy-constructor. See below.
+struct CodecSpecificInfoVP8 {
+  bool hasReceivedSLI;
+  uint8_t pictureIdSLI;
+  bool hasReceivedRPSI;
+  uint64_t pictureIdRPSI;
+  int16_t pictureId;  // Negative value to skip pictureId.
+  bool nonReference;
+  uint8_t simulcastIdx;
+  uint8_t temporalIdx;
+  bool layerSync;
+  int tl0PicIdx;  // Negative value to skip tl0PicIdx.
+  int8_t keyIdx;  // Negative value to skip keyIdx.
+};
+
+struct CodecSpecificInfoVP9 {
+  bool has_received_sli;
+  uint8_t picture_id_sli;
+  bool has_received_rpsi;
+  uint64_t picture_id_rpsi;
+  int16_t picture_id;  // Negative value to skip pictureId.
+
+  bool inter_pic_predicted;  // This layer frame is dependent on previously
+                             // coded frame(s).
+  bool flexible_mode;
+  bool ss_data_available;
+
+  int tl0_pic_idx;  // Negative value to skip tl0PicIdx.
+  uint8_t temporal_idx;
+  uint8_t spatial_idx;
+  bool temporal_up_switch;
+  bool inter_layer_predicted;  // Frame is dependent on directly lower spatial
+                               // layer frame.
+  uint8_t gof_idx;
+
+  // SS data.
+  size_t num_spatial_layers;  // Always populated.
+  bool spatial_layer_resolution_present;
+  uint16_t width[kMaxVp9NumberOfSpatialLayers];
+  uint16_t height[kMaxVp9NumberOfSpatialLayers];
+  GofInfoVP9 gof;
+
+  // Frame reference data.
+  uint8_t num_ref_pics;
+  uint8_t p_diff[kMaxVp9RefPics];
+};
+
+struct CodecSpecificInfoGeneric {
+  uint8_t simulcast_idx;
+};
+
+struct CodecSpecificInfoH264 {
+  bool    single_nalu;
+  uint8_t simulcastIdx;
+  uint8_t packetizationMode;
+};
+
+union CodecSpecificInfoUnion {
+  CodecSpecificInfoGeneric generic;
+  CodecSpecificInfoVP8 VP8;
+  CodecSpecificInfoVP9 VP9;
+  CodecSpecificInfoH264 H264;
+};
+
+// Note: if any pointers are added to this struct or its sub-structs, it
+// must be fitted with a copy-constructor. This is because it is copied
+// in the copy-constructor of VCMEncodedFrame.
+struct CodecSpecificInfo {
+  VideoCodecType codecType;
+  CodecSpecificInfoUnion codecSpecific;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/include/video_coding.h b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_coding.h
new file mode 100644
index 0000000000..6287b0698b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_coding.h
@@ -0,0 +1,535 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
+
+#if defined(WEBRTC_WIN)
+// This is a workaround on Windows due to the fact that some Windows
+// headers define CreateEvent as a macro to either CreateEventW or CreateEventA.
+// This can cause problems since we use that name as well and could
+// declare them as one thing here whereas in another place a windows header
+// may have been included and then implementing CreateEvent() causes compilation
+// errors.  So for consistency, we include the main windows header here.
+#include <windows.h>
+#endif
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+
+class Clock;
+class EncodedImageCallback;
+class VideoEncoder;
+class VideoDecoder;
+struct CodecSpecificInfo;
+
+class EventFactory {
+ public:
+  virtual ~EventFactory() {}
+
+  virtual EventWrapper* CreateEvent() = 0;
+};
+
+class EventFactoryImpl : public EventFactory {
+ public:
+  virtual ~EventFactoryImpl() {}
+
+  virtual EventWrapper* CreateEvent() { return EventWrapper::Create(); }
+};
+
+// Used to indicate which decode with errors mode should be used.
+enum VCMDecodeErrorMode {
+  kNoErrors,         // Never decode with errors. Video will freeze
+                     // if nack is disabled.
+  kSelectiveErrors,  // Frames that are determined decodable in
+                     // VCMSessionInfo may be decoded with missing
+                     // packets. As not all incomplete frames will be
+                     // decodable, video will freeze if nack is disabled.
+  kWithErrors        // Release frames as needed. Errors may be
+                     // introduced as some encoded frames may not be
+                     // complete.
+};
+
+class VideoCodingModule : public Module {
+ public:
+  enum SenderNackMode { kNackNone, kNackAll, kNackSelective };
+
+  enum ReceiverRobustness { kNone, kHardNack, kSoftNack, kReferenceSelection };
+
+  static VideoCodingModule* Create(
+      Clock* clock,
+      VideoEncoderRateObserver* encoder_rate_observer,
+      VCMQMSettingsCallback* qm_settings_callback);
+
+  static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
+
+  static void Destroy(VideoCodingModule* module);
+
+  // Get supported codec settings using codec type
+  //
+  // Input:
+  //      - codecType      : The codec type to get settings for
+  //      - codec          : Memory where the codec settings will be stored
+  //
+  // Return value     : VCM_OK,              on success
+  //                    VCM_PARAMETER_ERROR  if codec not supported
+  static void Codec(VideoCodecType codecType, VideoCodec* codec);
+
+  /*
+  *   Sender
+  */
+
+  // Registers a codec to be used for encoding. Calling this
+  // API multiple times overwrites any previously registered codecs.
+  //
+  // NOTE: Must be called on the thread that constructed the VCM instance.
+  //
+  // Input:
+  //      - sendCodec      : Settings for the codec to be registered.
+  //      - numberOfCores  : The number of cores the codec is allowed
+  //                         to use.
+  //      - maxPayloadSize : The maximum size each payload is allowed
+  //                                to have. Usually MTU - overhead.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterSendCodec(const VideoCodec* sendCodec,
+                                    uint32_t numberOfCores,
+                                    uint32_t maxPayloadSize) = 0;
+
+  // Register an external encoder object. This can not be used together with
+  // external decoder callbacks.
+  //
+  // Input:
+  //      - externalEncoder : Encoder object to be used for encoding frames
+  //      inserted
+  //                          with the AddVideoFrame API.
+  //      - payloadType     : The payload type bound which this encoder is bound
+  //      to.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  // TODO(pbos): Remove return type when unused elsewhere.
+  virtual int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                          uint8_t payloadType,
+                                          bool internalSource = false) = 0;
+
+  // API to get currently configured encoder target bitrate in bits/s.
+  //
+  // Return value      : 0,   on success.
+  //                     < 0, on error.
+  virtual int Bitrate(unsigned int* bitrate) const = 0;
+
+  // API to get currently configured encoder target frame rate.
+  //
+  // Return value      : 0,   on success.
+  //                     < 0, on error.
+  virtual int FrameRate(unsigned int* framerate) const = 0;
+
+  // Sets the parameters describing the send channel. These parameters are
+  // inputs to the
+  // Media Optimization inside the VCM and also specifies the target bit rate
+  // for the
+  // encoder. Bit rate used by NACK should already be compensated for by the
+  // user.
+  //
+  // Input:
+  //      - target_bitrate        : The target bitrate for VCM in bits/s.
+  //      - lossRate              : Fractions of lost packets the past second.
+  //                                (loss rate in percent = 100 * packetLoss /
+  //                                255)
+  //      - rtt                   : Current round-trip time in ms.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,         on error.
+  virtual int32_t SetChannelParameters(uint32_t target_bitrate,
+                                       uint8_t lossRate,
+                                       int64_t rtt) = 0;
+
+  // Sets the parameters describing the receive channel. These parameters are
+  // inputs to the
+  // Media Optimization inside the VCM.
+  //
+  // Input:
+  //      - rtt                   : Current round-trip time in ms.
+  //                                with the most amount available bandwidth in
+  //                                a conference
+  //                                scenario
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
+
+  // Register a transport callback which will be called to deliver the encoded
+  // data and
+  // side information.
+  //
+  // Input:
+  //      - transport  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterTransportCallback(
+      VCMPacketizationCallback* transport) = 0;
+
+  // Register video output information callback which will be called to deliver
+  // information
+  // about the video stream produced by the encoder, for instance the average
+  // frame rate and
+  // bit rate.
+  //
+  // Input:
+  //      - outputInformation  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterSendStatisticsCallback(
+      VCMSendStatisticsCallback* sendStats) = 0;
+
+  // Register a video protection callback which will be called to deliver
+  // the requested FEC rate and NACK status (on/off).
+  //
+  // Input:
+  //      - protection  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterProtectionCallback(
+      VCMProtectionCallback* protection) = 0;
+
+  // Enable or disable a video protection method.
+  //
+  // Input:
+  //      - videoProtection  : The method to enable or disable.
+  //      - enable           : True if the method should be enabled, false if
+  //                           it should be disabled.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetVideoProtection(VCMVideoProtection videoProtection,
+                                     bool enable) = 0;
+
+  // Add one raw video frame to the encoder. This function does all the
+  // necessary
+  // processing, then decides what frame type to encode, or if the frame should
+  // be
+  // dropped. If the frame should be encoded it passes the frame to the encoder
+  // before it returns.
+  //
+  // Input:
+  //      - videoFrame        : Video frame to encode.
+  //      - codecSpecificInfo : Extra codec information, e.g., pre-parsed
+  //      in-band signaling.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t AddVideoFrame(
+      const VideoFrame& videoFrame,
+      const VideoContentMetrics* contentMetrics = NULL,
+      const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
+
+  // Next frame encoded should be an intra frame (keyframe).
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t IntraFrameRequest(int stream_index) = 0;
+
+  // Frame Dropper enable. Can be used to disable the frame dropping when the
+  // encoder
+  // over-uses its bit rate. This API is designed to be used when the encoded
+  // frames
+  // are supposed to be stored to an AVI file, or when the I420 codec is used
+  // and the
+  // target bit rate shouldn't affect the frame rate.
+  //
+  // Input:
+  //      - enable            : True to enable the setting, false to disable it.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t EnableFrameDropper(bool enable) = 0;
+
+  /*
+  *   Receiver
+  */
+
+  // Register possible receive codecs, can be called multiple times for
+  // different codecs.
+  // The module will automatically switch between registered codecs depending on
+  // the
+  // payload type of incoming frames. The actual decoder will be created when
+  // needed.
+  //
+  // Input:
+  //      - receiveCodec      : Settings for the codec to be registered.
+  //      - numberOfCores     : Number of CPU cores that the decoder is allowed
+  //      to use.
+  //      - requireKeyFrame   : Set this to true if you don't want any delta
+  //      frames
+  //                            to be decoded until the first key frame has been
+  //                            decoded.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
+                                       int32_t numberOfCores,
+                                       bool requireKeyFrame = false) = 0;
+
+  // Register an externally defined decoder/renderer object. Can be a decoder
+  // only or a
+  // decoder coupled with a renderer. Note that RegisterReceiveCodec must be
+  // called to
+  // be used for decoding incoming streams.
+  //
+  // Input:
+  //      - externalDecoder        : The external decoder/renderer object.
+  //      - payloadType            : The payload type which this decoder should
+  //      be
+  //                                 registered to.
+  //
+  virtual void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                       uint8_t payloadType) = 0;
+
+  // Register a receive callback. Will be called whenever there is a new frame
+  // ready
+  // for rendering.
+  //
+  // Input:
+  //      - receiveCallback        : The callback object to be used by the
+  //      module when a
+  //                                 frame is ready for rendering.
+  //                                 De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveCallback(
+      VCMReceiveCallback* receiveCallback) = 0;
+
+  // Register a receive statistics callback which will be called to deliver
+  // information
+  // about the video stream received by the receiving side of the VCM, for
+  // instance the
+  // average frame rate and bit rate.
+  //
+  // Input:
+  //      - receiveStats  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveStatisticsCallback(
+      VCMReceiveStatisticsCallback* receiveStats) = 0;
+
+  // Register a decoder timing callback which will be called to deliver
+  // information about the timing of the decoder in the receiving side of the
+  // VCM, for instance the current and maximum frame decode latency.
+  //
+  // Input:
+  //      - decoderTiming  : The callback object to register.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterDecoderTimingCallback(
+      VCMDecoderTimingCallback* decoderTiming) = 0;
+
+  // Register a frame type request callback. This callback will be called when
+  // the
+  // module needs to request specific frame types from the send side.
+  //
+  // Input:
+  //      - frameTypeCallback      : The callback object to be used by the
+  //      module when
+  //                                 requesting a specific type of frame from
+  //                                 the send side.
+  //                                 De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterFrameTypeCallback(
+      VCMFrameTypeCallback* frameTypeCallback) = 0;
+
+  // Registers a callback which is called whenever the receive side of the VCM
+  // encounters holes in the packet sequence and needs packets to be
+  // retransmitted.
+  //
+  // Input:
+  //              - callback      : The callback to be registered in the VCM.
+  //
+  // Return value     : VCM_OK,     on success.
+  //                    <0,         on error.
+  virtual int32_t RegisterPacketRequestCallback(
+      VCMPacketRequestCallback* callback) = 0;
+ 
+  // Register a receive state change callback. This callback will be called when the
+  // module state has changed
+  //
+  // Input:
+  //      - callback      : The callback object to be used by the module when
+  //                        the receiver decode state changes.
+  //                        De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,         on error.
+  virtual int32_t RegisterReceiveStateCallback(
+      VCMReceiveStateCallback* callback) = 0;
+
+  // Waits for the next frame in the jitter buffer to become complete
+  // (waits no longer than maxWaitTimeMs), then passes it to the decoder for
+  // decoding.
+  // Should be called as often as possible to get the most out of the decoder.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t Decode(uint16_t maxWaitTimeMs = 200) = 0;
+
+  // Registers a callback which conveys the size of the render buffer.
+  virtual int RegisterRenderBufferSizeCallback(
+      VCMRenderBufferSizeCallback* callback) = 0;
+
+  // Reset the decoder state to the initial state.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t ResetDecoder() = 0;
+
+  // API to get the codec which is currently used for decoding by the module.
+  //
+  // Input:
+  //      - currentReceiveCodec      : Settings for the codec to be registered.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
+
+  // API to get the codec type currently used for decoding by the module.
+  //
+  // Return value      : codecy type,            on success.
+  //                     kVideoCodecUnknown, on error or if no receive codec is
+  //                     registered
+  virtual VideoCodecType ReceiveCodec() const = 0;
+
+  // Insert a parsed packet into the receiver side of the module. Will be placed
+  // in the
+  // jitter buffer waiting for the frame to become complete. Returns as soon as
+  // the packet
+  // has been placed in the jitter buffer.
+  //
+  // Input:
+  //      - incomingPayload      : Payload of the packet.
+  //      - payloadLength        : Length of the payload.
+  //      - rtpInfo              : The parsed header.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
+                                 size_t payloadLength,
+                                 const WebRtcRTPHeader& rtpInfo) = 0;
+
+  // Minimum playout delay (Used for lip-sync). This is the minimum delay
+  // required
+  // to sync with audio. Not included in  VideoCodingModule::Delay()
+  // Defaults to 0 ms.
+  //
+  // Input:
+  //      - minPlayoutDelayMs   : Additional delay in ms.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) = 0;
+
+  // Set the time required by the renderer to render a frame.
+  //
+  // Input:
+  //      - timeMS        : The time in ms required by the renderer to render a
+  //      frame.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t SetRenderDelay(uint32_t timeMS) = 0;
+
+  // The total delay desired by the VCM. Can be less than the minimum
+  // delay set with SetMinimumPlayoutDelay.
+  //
+  // Return value      : Total delay in ms, on success.
+  //                     < 0,               on error.
+  virtual int32_t Delay() const = 0;
+
+  // Returns the number of packets discarded by the jitter buffer due to being
+  // too late. This can include duplicated packets which arrived after the
+  // frame was sent to the decoder. Therefore packets which were prematurely
+  // NACKed will be counted.
+  virtual uint32_t DiscardedPackets() const = 0;
+
+  // Robustness APIs
+
+  // Set the receiver robustness mode. The mode decides how the receiver
+  // responds to losses in the stream. The type of counter-measure (soft or
+  // hard NACK, dual decoder, RPS, etc.) is selected through the
+  // robustnessMode parameter. The errorMode parameter decides if it is
+  // allowed to display frames corrupted by losses. Note that not all
+  // combinations of the two parameters are feasible. An error will be
+  // returned for invalid combinations.
+  // Input:
+  //      - robustnessMode : selected robustness mode.
+  //      - errorMode      : selected error mode.
+  //
+  // Return value      : VCM_OK, on success;
+  //                     < 0, on error.
+  virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
+                                        VCMDecodeErrorMode errorMode) = 0;
+
+  // Set the decode error mode. The mode decides which errors (if any) are
+  // allowed in decodable frames. Note that setting decode_error_mode to
+  // anything other than kWithErrors without enabling nack will cause
+  // long-term freezes (resulting from frequent key frame requests) if
+  // packet loss occurs.
+  virtual void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) = 0;
+
+  // Sets the maximum number of sequence numbers that we are allowed to NACK
+  // and the oldest sequence number that we will consider to NACK. If a
+  // sequence number older than |max_packet_age_to_nack| is missing
+  // a key frame will be requested. A key frame will also be requested if the
+  // time of incomplete or non-continuous frames in the jitter buffer is above
+  // |max_incomplete_time_ms|.
+  virtual void SetNackSettings(size_t max_nack_list_size,
+                               int max_packet_age_to_nack,
+                               int max_incomplete_time_ms) = 0;
+
+  // Setting a desired delay to the VCM receiver. Video rendering will be
+  // delayed by at least desired_delay_ms.
+  virtual int SetMinReceiverDelay(int desired_delay_ms) = 0;
+ 
+  // Set current load state of the CPU
+  virtual void SetCPULoadState(CPULoadState state) = 0;
+
+  // Lets the sender suspend video when the rate drops below
+  // |threshold_bps|, and turns back on when the rate goes back up above
+  // |threshold_bps| + |window_bps|.
+  virtual void SuspendBelowMinBitrate() = 0;
+
+  // Returns true if SuspendBelowMinBitrate is engaged and the video has been
+  // suspended due to bandwidth limitations; otherwise false.
+  virtual bool VideoSuspended() const = 0;
+
+  virtual void RegisterPreDecodeImageCallback(
+      EncodedImageCallback* observer) = 0;
+  virtual void RegisterPostEncodeImageCallback(
+      EncodedImageCallback* post_encode_callback) = 0;
+  // Releases pending decode calls, permitting faster thread shutdown.
+  virtual void TriggerDecoderShutdown() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/video_coding_defines.h b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_coding_defines.h
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/interface/video_coding_defines.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/include/video_coding_defines.h
index f8004f4722..8d2bf9818f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/video_coding_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_coding_defines.h
@@ -8,52 +8,41 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
-#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
 // Error codes
-#define VCM_FRAME_NOT_READY      3
-#define VCM_REQUEST_SLI          2
-#define VCM_MISSING_CALLBACK     1
-#define VCM_OK                   0
-#define VCM_GENERAL_ERROR       -1
-#define VCM_LEVEL_EXCEEDED      -2
-#define VCM_MEMORY              -3
-#define VCM_PARAMETER_ERROR     -4
-#define VCM_UNKNOWN_PAYLOAD     -5
-#define VCM_CODEC_ERROR         -6
-#define VCM_UNINITIALIZED       -7
+#define VCM_FRAME_NOT_READY 3
+#define VCM_REQUEST_SLI 2
+#define VCM_MISSING_CALLBACK 1
+#define VCM_OK 0
+#define VCM_GENERAL_ERROR -1
+#define VCM_LEVEL_EXCEEDED -2
+#define VCM_MEMORY -3
+#define VCM_PARAMETER_ERROR -4
+#define VCM_UNKNOWN_PAYLOAD -5
+#define VCM_CODEC_ERROR -6
+#define VCM_UNINITIALIZED -7
 #define VCM_NO_CODEC_REGISTERED -8
 #define VCM_JITTER_BUFFER_ERROR -9
-#define VCM_OLD_PACKET_ERROR    -10
-#define VCM_NO_FRAME_DECODED    -11
-#define VCM_ERROR_REQUEST_SLI   -12
-#define VCM_NOT_IMPLEMENTED     -20
-
-#define VCM_RED_PAYLOAD_TYPE       122
-#define VCM_ULPFEC_PAYLOAD_TYPE    123
-#define VCM_VP8_PAYLOAD_TYPE       100
-#define VCM_VP9_PAYLOAD_TYPE       101
-#define VCM_I420_PAYLOAD_TYPE      124
-#define VCM_H264_PAYLOAD_TYPE      127
+#define VCM_OLD_PACKET_ERROR -10
+#define VCM_NO_FRAME_DECODED -11
+#define VCM_ERROR_REQUEST_SLI -12
+#define VCM_NOT_IMPLEMENTED -20
 
 enum { kDefaultStartBitrateKbps = 300 };
 
 enum VCMVideoProtection {
   kProtectionNone,
-  kProtectionNack,                // Both send-side and receive-side
-  kProtectionNackSender,          // Send-side only
-  kProtectionNackReceiver,        // Receive-side only
+  kProtectionNack,
   kProtectionFEC,
   kProtectionNackFEC,
-  kProtectionKeyOnLoss,
-  kProtectionKeyOnKeyLoss,
 };
 
 enum VCMTemporalDecimation {
@@ -73,40 +62,42 @@ class VCMPacketizationCallback {
                            const RTPFragmentationHeader& fragmentationHeader,
                            const RTPVideoHeader* rtpVideoHdr) = 0;
 
+  virtual void OnEncoderImplementationName(const char* implementation_name) {}
+
  protected:
-  virtual ~VCMPacketizationCallback() {
-  }
+  virtual ~VCMPacketizationCallback() {}
 };
 
-// Callback class used for passing decoded frames which are ready to be rendered.
+// Callback class used for passing decoded frames which are ready to be
+// rendered.
 class VCMReceiveCallback {
  public:
-  virtual int32_t FrameToRender(I420VideoFrame& videoFrame) = 0;
-  virtual int32_t ReceivedDecodedReferenceFrame(
-      const uint64_t pictureId) {
+  virtual int32_t FrameToRender(VideoFrame& videoFrame) = 0;  // NOLINT
+  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) {
     return -1;
   }
   // Called when the current receive codec changes.
-  virtual void IncomingCodecChanged(const VideoCodec& codec) {}
+  virtual void OnIncomingPayloadType(int payload_type) {}
+  virtual void OnDecoderImplementationName(const char* implementation_name) {}
 
  protected:
-  virtual ~VCMReceiveCallback() {
-  }
+  virtual ~VCMReceiveCallback() {}
 };
 
-// Callback class used for informing the user of the bit rate and frame rate produced by the
+// Callback class used for informing the user of the bit rate and frame rate
+// produced by the
 // encoder.
 class VCMSendStatisticsCallback {
  public:
   virtual int32_t SendStatistics(const uint32_t bitRate,
-                                       const uint32_t frameRate) = 0;
+                                 const uint32_t frameRate) = 0;
 
  protected:
-  virtual ~VCMSendStatisticsCallback() {
-  }
+  virtual ~VCMSendStatisticsCallback() {}
 };
 
-// Callback class used for informing the user of the incoming bit rate and frame rate.
+// Callback class used for informing the user of the incoming bit rate and frame
+// rate.
 class VCMReceiveStatisticsCallback {
  public:
   virtual void OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) = 0;
@@ -114,8 +105,7 @@ class VCMReceiveStatisticsCallback {
   virtual void OnFrameCountsUpdated(const FrameCounts& frame_counts) = 0;
 
  protected:
-  virtual ~VCMReceiveStatisticsCallback() {
-  }
+  virtual ~VCMReceiveStatisticsCallback() {}
 };
 
 // Callback class used for informing the user of decode timing info.
@@ -144,8 +134,7 @@ class VCMProtectionCallback {
                                 uint32_t* sent_fec_rate_bps) = 0;
 
  protected:
-  virtual ~VCMProtectionCallback() {
-  }
+  virtual ~VCMProtectionCallback() {}
 };
 
 class VideoEncoderRateObserver {
@@ -154,33 +143,32 @@ class VideoEncoderRateObserver {
   virtual void OnSetRates(uint32_t bitrate_bps, int framerate) = 0;
 };
 
-// Callback class used for telling the user about what frame type needed to continue decoding.
+// Callback class used for telling the user about what frame type needed to
+// continue decoding.
 // Typically a key frame when the stream has been corrupted in some way.
 class VCMFrameTypeCallback {
  public:
   virtual int32_t RequestKeyFrame() = 0;
-  virtual int32_t SliceLossIndicationRequest(
-      const uint64_t pictureId) {
+  virtual int32_t SliceLossIndicationRequest(const uint64_t pictureId) {
     return -1;
   }
 
  protected:
-  virtual ~VCMFrameTypeCallback() {
-  }
+  virtual ~VCMFrameTypeCallback() {}
 };
 
-// Callback class used for telling the user about which packet sequence numbers are currently
+// Callback class used for telling the user about which packet sequence numbers
+// are currently
 // missing and need to be resent.
 class VCMPacketRequestCallback {
  public:
   virtual int32_t ResendPackets(const uint16_t* sequenceNumbers,
-                                      uint16_t length) = 0;
+                                uint16_t length) = 0;
 
  protected:
-  virtual ~VCMPacketRequestCallback() {
-  }
+  virtual ~VCMPacketRequestCallback() {}
 };
-
+ 
 // Callback class used for telling the user about the state of the decoder & jitter buffer.
 //
 class VCMReceiveStateCallback {
@@ -197,12 +185,13 @@ class VCMReceiveStateCallback {
 class VCMQMSettingsCallback {
  public:
   virtual int32_t SetVideoQMSettings(const uint32_t frameRate,
-                                           const uint32_t width,
-                                           const uint32_t height) = 0;
+                                     const uint32_t width,
+                                     const uint32_t height) = 0;
+
+  virtual void SetTargetFramerate(int frame_rate) = 0;
 
  protected:
-  virtual ~VCMQMSettingsCallback() {
-  }
+  virtual ~VCMQMSettingsCallback() {}
 };
 
 // Callback class used for telling the user about the size (in time) of the
@@ -212,10 +201,9 @@ class VCMRenderBufferSizeCallback {
   virtual void RenderBufferSizeMs(int buffer_size_ms) = 0;
 
  protected:
-  virtual ~VCMRenderBufferSizeCallback() {
-  }
+  virtual ~VCMRenderBufferSizeCallback() {}
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/include/video_error_codes.h b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_error_codes.h
new file mode 100644
index 0000000000..360aa87744
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/include/video_error_codes.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
+
+// NOTE: in sync with video_coding_module_defines.h
+
+// Define return values
+
+#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
+#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
+#define WEBRTC_VIDEO_CODEC_OK 0
+#define WEBRTC_VIDEO_CODEC_ERROR -1
+#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
+#define WEBRTC_VIDEO_CODEC_MEMORY -3
+#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
+#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
+#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
+#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
+#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
+#define WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE -13
+#define WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT -14
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/inter_frame_delay.cc b/media/webrtc/trunk/webrtc/modules/video_coding/inter_frame_delay.cc
new file mode 100644
index 0000000000..fb3b54d204
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/inter_frame_delay.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/inter_frame_delay.h"
+
+namespace webrtc {
+
+VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock) {
+  Reset(currentWallClock);
+}
+
+// Resets the delay estimate
+void VCMInterFrameDelay::Reset(int64_t currentWallClock) {
+  _zeroWallClock = currentWallClock;
+  _wrapArounds = 0;
+  _prevWallClock = 0;
+  _prevTimestamp = 0;
+  _dTS = 0;
+}
+
+// Calculates the delay of a frame with the given timestamp.
+// This method is called when the frame is complete.
+bool VCMInterFrameDelay::CalculateDelay(uint32_t timestamp,
+                                        int64_t* delay,
+                                        int64_t currentWallClock) {
+  if (_prevWallClock == 0) {
+    // First set of data, initialization, wait for next frame
+    _prevWallClock = currentWallClock;
+    _prevTimestamp = timestamp;
+    *delay = 0;
+    return true;
+  }
+
+  int32_t prevWrapArounds = _wrapArounds;
+  CheckForWrapArounds(timestamp);
+
+  // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
+  int32_t wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
+
+  // Account for reordering in jitter variance estimate in the future?
+  // Note that this also captures incomplete frames which are grabbed
+  // for decoding after a later frame has been complete, i.e. real
+  // packet losses.
+  if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) ||
+      wrapAroundsSincePrev < 0) {
+    *delay = 0;
+    return false;
+  }
+
+  // Compute the compensated timestamp difference and convert it to ms and
+  // round it to closest integer.
+  _dTS = static_cast<int64_t>(
+      (timestamp + wrapAroundsSincePrev * (static_cast<int64_t>(1) << 32) -
+       _prevTimestamp) /
+          90.0 +
+      0.5);
+
+  // frameDelay is the difference of dT and dTS -- i.e. the difference of
+  // the wall clock time difference and the timestamp difference between
+  // two following frames.
+  *delay = static_cast<int64_t>(currentWallClock - _prevWallClock - _dTS);
+
+  _prevTimestamp = timestamp;
+  _prevWallClock = currentWallClock;
+
+  return true;
+}
+
+// Returns the current difference between incoming timestamps
+uint32_t VCMInterFrameDelay::CurrentTimeStampDiffMs() const {
+  if (_dTS < 0) {
+    return 0;
+  }
+  return static_cast<uint32_t>(_dTS);
+}
+
+// Investigates if the timestamp clock has overflowed since the last timestamp
+// and
+// keeps track of the number of wrap arounds since reset.
+void VCMInterFrameDelay::CheckForWrapArounds(uint32_t timestamp) {
+  if (timestamp < _prevTimestamp) {
+    // This difference will probably be less than -2^31 if we have had a wrap
+    // around
+    // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to
+    // a Word32,
+    // it should be positive.
+    if (static_cast<int32_t>(timestamp - _prevTimestamp) > 0) {
+      // Forward wrap around
+      _wrapArounds++;
+    }
+    // This difference will probably be less than -2^31 if we have had a
+    // backward
+    // wrap around.
+    // Since it is cast to a Word32, it should be positive.
+  } else if (static_cast<int32_t>(_prevTimestamp - timestamp) > 0) {
+    // Backward wrap around
+    _wrapArounds--;
+  }
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/inter_frame_delay.h b/media/webrtc/trunk/webrtc/modules/video_coding/inter_frame_delay.h
new file mode 100644
index 0000000000..94b73908bb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/inter_frame_delay.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMInterFrameDelay {
+ public:
+  explicit VCMInterFrameDelay(int64_t currentWallClock);
+
+  // Resets the estimate. Zeros are given as parameters.
+  void Reset(int64_t currentWallClock);
+
+  // Calculates the delay of a frame with the given timestamp.
+  // This method is called when the frame is complete.
+  //
+  // Input:
+  //          - timestamp         : RTP timestamp of a received frame
+  //          - *delay            : Pointer to memory where the result should be
+  //          stored
+  //          - currentWallClock  : The current time in milliseconds.
+  //                                Should be -1 for normal operation, only used
+  //                                for testing.
+  // Return value                 : true if OK, false when reordered timestamps
+  bool CalculateDelay(uint32_t timestamp,
+                      int64_t* delay,
+                      int64_t currentWallClock);
+
+  // Returns the current difference between incoming timestamps
+  //
+  // Return value                 : Wrap-around compensated difference between
+  // incoming
+  //                                timestamps.
+  uint32_t CurrentTimeStampDiffMs() const;
+
+ private:
+  // Controls if the RTP timestamp counter has had a wrap around
+  // between the current and the previously received frame.
+  //
+  // Input:
+  //          - timestmap         : RTP timestamp of the current frame.
+  void CheckForWrapArounds(uint32_t timestamp);
+
+  int64_t _zeroWallClock;  // Local timestamp of the first video packet received
+  int32_t _wrapArounds;    // Number of wrapArounds detected
+  // The previous timestamp passed to the delay estimate
+  uint32_t _prevTimestamp;
+  // The previous wall clock timestamp used by the delay estimate
+  int64_t _prevWallClock;
+  // Wrap-around compensated difference between incoming timestamps
+  int64_t _dTS;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/internal_defines.h b/media/webrtc/trunk/webrtc/modules/video_coding/internal_defines.h
new file mode 100644
index 0000000000..e225726dea
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/internal_defines.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
+
+inline uint32_t MaskWord64ToUWord32(int64_t w64) {
+  return static_cast<uint32_t>(MASK_32_BITS(w64));
+}
+
+#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
+#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
+
+#define VCM_DEFAULT_CODEC_WIDTH 352
+#define VCM_DEFAULT_CODEC_HEIGHT 288
+#define VCM_DEFAULT_FRAME_RATE 30
+#define VCM_MIN_BITRATE 30
+#define VCM_FLUSH_INDICATOR 4
+
+#define VCM_NO_RECEIVER_ID 0
+
+inline int32_t VCMId(const int32_t vcmId, const int32_t receiverId = 0) {
+  return static_cast<int32_t>((vcmId << 16) + receiverId);
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.cc
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.cc
index 3961334e0a..2023e67182 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
 
 #include <assert.h>
 
@@ -15,20 +15,20 @@
 #include <utility>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/inter_frame_delay.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
@@ -38,6 +38,10 @@ static const uint32_t kSsCleanupIntervalSec = 60;
 // Use this rtt if no value has been reported.
 static const int64_t kDefaultRtt = 200;
 
+// Request a keyframe if no continuous frame has been received for this
+// number of milliseconds and NACKs are disabled.
+static const int64_t kMaxDiscontinuousFramesTime = 1000;
+
 typedef std::pair<uint32_t, VCMFrameBuffer*> FrameListPair;
 
 bool IsKeyFrame(FrameListPair pair) {
@@ -122,7 +126,7 @@ int FrameList::RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
 }
 
 void FrameList::CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
-                                       UnorderedFrameList* free_frames) {
+                                        UnorderedFrameList* free_frames) {
   while (!empty()) {
     VCMFrameBuffer* oldest_frame = Front();
     bool remove_frame = false;
@@ -195,7 +199,7 @@ bool Vp9SsMap::TimeForCleanup(uint32_t timestamp) const {
 }
 
 void Vp9SsMap::AdvanceFront(uint32_t timestamp) {
-  DCHECK(!ss_map_.empty());
+  RTC_DCHECK(!ss_map_.empty());
   GofInfoVP9 gof = ss_map_.begin()->second;
   ss_map_.erase(ss_map_.begin());
   ss_map_[timestamp] = gof;
@@ -243,11 +247,12 @@ void Vp9SsMap::UpdateFrames(FrameList* frames) {
   }
 }
 
-VCMJitterBuffer::VCMJitterBuffer(Clock* clock, EventFactory* event_factory)
+VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
+                                 rtc::scoped_ptr<EventWrapper> event)
     : clock_(clock),
       running_(false),
       crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      frame_event_(event_factory->CreateEvent()),
+      frame_event_(std::move(event)),
       max_number_of_frames_(kStartNumberOfFrames),
       free_frames_(),
       decodable_frames_(),
@@ -272,7 +277,6 @@ VCMJitterBuffer::VCMJitterBuffer(Clock* clock, EventFactory* event_factory)
       low_rtt_nack_threshold_ms_(-1),
       high_rtt_nack_threshold_ms_(-1),
       missing_sequence_numbers_(SequenceNumberLessThan()),
-      nack_seq_nums_(),
       max_nack_list_size_(0),
       max_packet_age_to_nack_(0),
       max_incomplete_time_ms_(0),
@@ -310,17 +314,18 @@ void VCMJitterBuffer::UpdateHistograms() {
     return;
   }
 
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
-      num_discarded_packets_ * 100 / num_packets_);
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
-      num_duplicated_packets_ * 100 / num_packets_);
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DiscardedPacketsInPercent",
+                                  num_discarded_packets_ * 100 / num_packets_);
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DuplicatedPacketsInPercent",
+                                  num_duplicated_packets_ * 100 / num_packets_);
 
   int total_frames =
       receive_statistics_.key_frames + receive_statistics_.delta_frames;
   if (total_frames > 0) {
-    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.CompleteFramesReceivedPerSecond",
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(
+        "WebRTC.Video.CompleteFramesReceivedPerSecond",
         static_cast<int>((total_frames / elapsed_sec) + 0.5f));
-    RTC_HISTOGRAM_COUNTS_1000(
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
         "WebRTC.Video.KeyFramesReceivedInPermille",
         static_cast<int>(
             (receive_statistics_.key_frames * 1000.0f / total_frames) + 0.5f));
@@ -357,6 +362,7 @@ void VCMJitterBuffer::Stop() {
   UpdateHistograms();
   running_ = false;
   last_decoded_state_.Reset();
+
   // Make sure all frames are free and reset.
   for (FrameList::iterator it = decodable_frames_.begin();
        it != decodable_frames_.end(); ++it) {
@@ -458,8 +464,8 @@ void VCMJitterBuffer::IncomingRateStatistics(unsigned int* framerate,
     if (incoming_bit_count_ == 0) {
       *bitrate = 0;
     } else {
-      *bitrate = 10 * ((100 * incoming_bit_count_) /
-                       static_cast<unsigned int>(diff));
+      *bitrate =
+          10 * ((100 * incoming_bit_count_) / static_cast<unsigned int>(diff));
     }
     incoming_bit_rate_ = *bitrate;
 
@@ -500,8 +506,8 @@ bool VCMJitterBuffer::CompleteSequenceWithNextFrame() {
 
 // Returns immediately or a |max_wait_time_ms| ms event hang waiting for a
 // complete frame, |max_wait_time_ms| decided by caller.
-bool VCMJitterBuffer::NextCompleteTimestamp(
-    uint32_t max_wait_time_ms, uint32_t* timestamp) {
+bool VCMJitterBuffer::NextCompleteTimestamp(uint32_t max_wait_time_ms,
+                                            uint32_t* timestamp) {
   crit_sect_->Enter();
   if (!running_) {
     crit_sect_->Leave();
@@ -511,13 +517,13 @@ bool VCMJitterBuffer::NextCompleteTimestamp(
 
   if (decodable_frames_.empty() ||
       decodable_frames_.Front()->GetState() != kStateComplete) {
-    const int64_t end_wait_time_ms = clock_->TimeInMilliseconds() +
-        max_wait_time_ms;
+    const int64_t end_wait_time_ms =
+        clock_->TimeInMilliseconds() + max_wait_time_ms;
     int64_t wait_time_ms = max_wait_time_ms;
     while (wait_time_ms > 0) {
       crit_sect_->Leave();
       const EventTypeWrapper ret =
-        frame_event_->Wait(static_cast<uint32_t>(wait_time_ms));
+          frame_event_->Wait(static_cast<uint32_t>(wait_time_ms));
       crit_sect_->Enter();
       if (ret == kEventSignaled) {
         // Are we shutting down the jitter buffer?
@@ -560,16 +566,25 @@ bool VCMJitterBuffer::NextMaybeIncompleteTimestamp(uint32_t* timestamp) {
 
   CleanUpOldOrEmptyFrames();
 
+  VCMFrameBuffer* oldest_frame;
   if (decodable_frames_.empty()) {
-    return false;
-  }
-  VCMFrameBuffer* oldest_frame = decodable_frames_.Front();
-  // If we have exactly one frame in the buffer, release it only if it is
-  // complete. We know decodable_frames_ is  not empty due to the previous
-  // check.
-  if (decodable_frames_.size() == 1 && incomplete_frames_.empty()
-      && oldest_frame->GetState() != kStateComplete) {
-    return false;
+    if (nack_mode_ != kNoNack || incomplete_frames_.size() <= 1) {
+      return false;
+    }
+    oldest_frame = incomplete_frames_.Front();
+    // Frame will only be removed from buffer if it is complete (or decodable).
+    if (oldest_frame->GetState() < kStateComplete) {
+      return false;
+    }
+  } else {
+    oldest_frame = decodable_frames_.Front();
+    // If we have exactly one frame in the buffer, release it only if it is
+    // complete. We know decodable_frames_ is  not empty due to the previous
+    // check.
+    if (decodable_frames_.size() == 1 && incomplete_frames_.empty() &&
+        oldest_frame->GetState() != kStateComplete) {
+      return false;
+    }
   }
 
   *timestamp = oldest_frame->TimeStamp();
@@ -610,8 +625,7 @@ VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) {
     } else {
       // Wait for this one to get complete.
       waiting_for_completion_.frame_size = frame->Length();
-      waiting_for_completion_.latest_packet_time =
-          frame->LatestPacketTimeMs();
+      waiting_for_completion_.latest_packet_time = frame->LatestPacketTimeMs();
       waiting_for_completion_.timestamp = frame->TimeStamp();
     }
   }
@@ -753,7 +767,7 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
 
   // Empty packets may bias the jitter estimate (lacking size component),
   // therefore don't let empty packet trigger the following updates:
-  if (packet.frameType != kFrameEmpty) {
+  if (packet.frameType != kEmptyFrame) {
     if (waiting_for_completion_.timestamp == packet.timestamp) {
       // This can get bad if we have a lot of duplicate packets,
       // we will then count some packet multiple times.
@@ -778,8 +792,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
       frame->InsertPacket(packet, now_ms, decode_error_mode_, frame_data);
 
   if (previous_state != kStateComplete) {
-    TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", frame->TimeStamp(),
-                             "timestamp", frame->TimeStamp());
+    TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", frame->TimeStamp(), "timestamp",
+                             frame->TimeStamp());
   }
 
   if (buffer_state > 0) {
@@ -796,8 +810,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
         buffer_state = kFlushIndicator;
       }
 
-      latest_received_sequence_number_ = LatestSequenceNumber(
-          latest_received_sequence_number_, packet.seqNum);
+      latest_received_sequence_number_ =
+          LatestSequenceNumber(latest_received_sequence_number_, packet.seqNum);
     }
   }
 
@@ -829,6 +843,12 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
         FindAndInsertContinuousFrames(*frame);
       } else {
         incomplete_frames_.InsertFrame(frame);
+        // If NACKs are enabled, keyframes are triggered by |GetNackList|.
+        if (nack_mode_ == kNoNack &&
+            NonContinuousOrIncompleteDuration() >
+                90 * kMaxDiscontinuousFramesTime) {
+          return kFlushIndicator;
+        }
       }
       break;
     }
@@ -839,6 +859,12 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
         return kNoError;
       } else {
         incomplete_frames_.InsertFrame(frame);
+        // If NACKs are enabled, keyframes are triggered by |GetNackList|.
+        if (nack_mode_ == kNoNack &&
+            NonContinuousOrIncompleteDuration() >
+                90 * kMaxDiscontinuousFramesTime) {
+          return kFlushIndicator;
+        }
       }
       break;
     }
@@ -857,15 +883,15 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
     case kFlushIndicator:
       free_frames_.push_back(frame);
       return kFlushIndicator;
-    default: assert(false);
+    default:
+      assert(false);
   }
   return buffer_state;
 }
 
-bool VCMJitterBuffer::IsContinuousInState(const VCMFrameBuffer& frame,
+bool VCMJitterBuffer::IsContinuousInState(
+    const VCMFrameBuffer& frame,
     const VCMDecodingState& decoding_state) const {
-  if (decode_error_mode_ == kWithErrors)
-    return true;
   // Is this frame (complete or decodable) and continuous?
   // kStateDecodable will never be set when decode_error_mode_ is false
   // as SessionInfo determines this state based on the error mode (and frame
@@ -882,7 +908,7 @@ bool VCMJitterBuffer::IsContinuous(const VCMFrameBuffer& frame) const {
   VCMDecodingState decoding_state;
   decoding_state.CopyFrom(last_decoded_state_);
   for (FrameList::const_iterator it = decodable_frames_.begin();
-       it != decodable_frames_.end(); ++it)  {
+       it != decodable_frames_.end(); ++it) {
     VCMFrameBuffer* decodable_frame = it->second;
     if (IsNewerTimestamp(decodable_frame->TimeStamp(), frame.TimeStamp())) {
       break;
@@ -915,7 +941,7 @@ void VCMJitterBuffer::FindAndInsertContinuousFramesWithState(
   // 1. Continuous base or sync layer.
   // 2. The end of the list was reached.
   for (FrameList::iterator it = incomplete_frames_.begin();
-       it != incomplete_frames_.end();)  {
+       it != incomplete_frames_.end();) {
     VCMFrameBuffer* frame = it->second;
     if (IsNewerTimestamp(original_decoded_state.time_stamp(),
                          frame->TimeStamp())) {
@@ -987,7 +1013,6 @@ void VCMJitterBuffer::SetNackSettings(size_t max_nack_list_size,
   max_nack_list_size_ = max_nack_list_size;
   max_packet_age_to_nack_ = max_packet_age_to_nack;
   max_incomplete_time_ms_ = max_incomplete_time_ms;
-  nack_seq_nums_.resize(max_nack_list_size_);
 }
 
 VCMNackMode VCMJitterBuffer::nack_mode() const {
@@ -1017,33 +1042,32 @@ uint16_t VCMJitterBuffer::EstimatedLowSequenceNumber(
   return frame.GetLowSeqNum() - 1;
 }
 
-uint16_t* VCMJitterBuffer::GetNackList(uint16_t* nack_list_size,
-                                       bool* request_key_frame) {
+std::vector<uint16_t> VCMJitterBuffer::GetNackList(bool* request_key_frame) {
   CriticalSectionScoped cs(crit_sect_);
   *request_key_frame = false;
   if (nack_mode_ == kNoNack) {
-    *nack_list_size = 0;
-    return NULL;
+    return std::vector<uint16_t>();
   }
   if (last_decoded_state_.in_initial_state()) {
     VCMFrameBuffer* next_frame = NextFrame();
     const bool first_frame_is_key = next_frame &&
-        next_frame->FrameType() == kVideoFrameKey &&
-        next_frame->HaveFirstPacket();
+                                    next_frame->FrameType() == kVideoFrameKey &&
+                                    next_frame->HaveFirstPacket();
     if (!first_frame_is_key) {
-      bool have_non_empty_frame = decodable_frames_.end() != find_if(
-          decodable_frames_.begin(), decodable_frames_.end(),
-          HasNonEmptyState);
+      bool have_non_empty_frame =
+          decodable_frames_.end() != find_if(decodable_frames_.begin(),
+                                             decodable_frames_.end(),
+                                             HasNonEmptyState);
       if (!have_non_empty_frame) {
-        have_non_empty_frame = incomplete_frames_.end() != find_if(
-            incomplete_frames_.begin(), incomplete_frames_.end(),
-            HasNonEmptyState);
+        have_non_empty_frame =
+            incomplete_frames_.end() != find_if(incomplete_frames_.begin(),
+                                                incomplete_frames_.end(),
+                                                HasNonEmptyState);
       }
       bool found_key_frame = RecycleFramesUntilKeyFrame();
       if (!found_key_frame) {
         *request_key_frame = have_non_empty_frame;
-        *nack_list_size = 0;
-        return NULL;
+        return std::vector<uint16_t>();
       }
     }
   }
@@ -1057,13 +1081,12 @@ uint16_t* VCMJitterBuffer::GetNackList(uint16_t* nack_list_size,
       LOG_F(LS_WARNING) << "Too long non-decodable duration: "
                         << non_continuous_incomplete_duration << " > "
                         << 90 * max_incomplete_time_ms_;
-      FrameList::reverse_iterator rit = find_if(incomplete_frames_.rbegin(),
-          incomplete_frames_.rend(), IsKeyFrame);
+      FrameList::reverse_iterator rit = find_if(
+          incomplete_frames_.rbegin(), incomplete_frames_.rend(), IsKeyFrame);
       if (rit == incomplete_frames_.rend()) {
         // Request a key frame if we don't have one already.
         *request_key_frame = true;
-        *nack_list_size = 0;
-        return NULL;
+        return std::vector<uint16_t>();
       } else {
         // Skip to the last key frame. If it's incomplete we will start
         // NACKing it.
@@ -1074,13 +1097,9 @@ uint16_t* VCMJitterBuffer::GetNackList(uint16_t* nack_list_size,
       }
     }
   }
-  unsigned int i = 0;
-  SequenceNumberSet::iterator it = missing_sequence_numbers_.begin();
-  for (; it != missing_sequence_numbers_.end(); ++it, ++i) {
-    nack_seq_nums_[i] = *it;
-  }
-  *nack_list_size = i;
-  return &nack_seq_nums_[0];
+  std::vector<uint16_t> nack_list(missing_sequence_numbers_.begin(),
+                                  missing_sequence_numbers_.end());
+  return nack_list;
 }
 
 void VCMJitterBuffer::SetDecodeErrorMode(VCMDecodeErrorMode error_mode) {
@@ -1103,8 +1122,7 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) {
   // Make sure we don't add packets which are already too old to be decoded.
   if (!last_decoded_state_.in_initial_state()) {
     latest_received_sequence_number_ = LatestSequenceNumber(
-        latest_received_sequence_number_,
-        last_decoded_state_.sequence_num());
+        latest_received_sequence_number_, last_decoded_state_.sequence_num());
   }
   if (IsNewerSequenceNumber(sequence_number,
                             latest_received_sequence_number_)) {
@@ -1154,8 +1172,8 @@ bool VCMJitterBuffer::MissingTooOldPacket(
   if (missing_sequence_numbers_.empty()) {
     return false;
   }
-  const uint16_t age_of_oldest_missing_packet = latest_sequence_number -
-      *missing_sequence_numbers_.begin();
+  const uint16_t age_of_oldest_missing_packet =
+      latest_sequence_number - *missing_sequence_numbers_.begin();
   // Recycle frames if the NACK list contains too old sequence numbers as
   // the packets may have already been dropped by the sender.
   return age_of_oldest_missing_packet > max_packet_age_to_nack_;
@@ -1163,8 +1181,8 @@ bool VCMJitterBuffer::MissingTooOldPacket(
 
 bool VCMJitterBuffer::HandleTooOldPackets(uint16_t latest_sequence_number) {
   bool key_frame_found = false;
-  const uint16_t age_of_oldest_missing_packet = latest_sequence_number -
-      *missing_sequence_numbers_.begin();
+  const uint16_t age_of_oldest_missing_packet =
+      latest_sequence_number - *missing_sequence_numbers_.begin();
   LOG_F(LS_WARNING) << "NACK list contains too old sequence numbers: "
                     << age_of_oldest_missing_packet << " > "
                     << max_packet_age_to_nack_;
@@ -1178,9 +1196,9 @@ void VCMJitterBuffer::DropPacketsFromNackList(
     uint16_t last_decoded_sequence_number) {
   // Erase all sequence numbers from the NACK list which we won't need any
   // longer.
-  missing_sequence_numbers_.erase(missing_sequence_numbers_.begin(),
-                                  missing_sequence_numbers_.upper_bound(
-                                      last_decoded_sequence_number));
+  missing_sequence_numbers_.erase(
+      missing_sequence_numbers_.begin(),
+      missing_sequence_numbers_.upper_bound(last_decoded_sequence_number));
 }
 
 int64_t VCMJitterBuffer::LastDecodedTimestamp() const {
@@ -1264,11 +1282,11 @@ void VCMJitterBuffer::CountFrame(const VCMFrameBuffer& frame) {
   incoming_frame_count_++;
 
   if (frame.FrameType() == kVideoFrameKey) {
-    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video",
-                            frame.TimeStamp(), "KeyComplete");
+    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", frame.TimeStamp(),
+                            "KeyComplete");
   } else {
-    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video",
-                            frame.TimeStamp(), "DeltaComplete");
+    TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", frame.TimeStamp(),
+                            "DeltaComplete");
   }
 
   // Update receive statistics. We count all layers, thus when you use layers
@@ -1286,13 +1304,13 @@ void VCMJitterBuffer::CountFrame(const VCMFrameBuffer& frame) {
 
 void VCMJitterBuffer::UpdateAveragePacketsPerFrame(int current_number_packets) {
   if (frame_counter_ > kFastConvergeThreshold) {
-    average_packets_per_frame_ = average_packets_per_frame_
-              * (1 - kNormalConvergeMultiplier)
-            + current_number_packets * kNormalConvergeMultiplier;
+    average_packets_per_frame_ =
+        average_packets_per_frame_ * (1 - kNormalConvergeMultiplier) +
+        current_number_packets * kNormalConvergeMultiplier;
   } else if (frame_counter_ > 0) {
-    average_packets_per_frame_ = average_packets_per_frame_
-              * (1 - kFastConvergeMultiplier)
-            + current_number_packets * kFastConvergeMultiplier;
+    average_packets_per_frame_ =
+        average_packets_per_frame_ * (1 - kFastConvergeMultiplier) +
+        current_number_packets * kFastConvergeMultiplier;
     frame_counter_++;
   } else {
     average_packets_per_frame_ = current_number_packets;
@@ -1314,7 +1332,7 @@ void VCMJitterBuffer::CleanUpOldOrEmptyFrames() {
 // Must be called from within |crit_sect_|.
 bool VCMJitterBuffer::IsPacketRetransmitted(const VCMPacket& packet) const {
   return missing_sequence_numbers_.find(packet.seqNum) !=
-      missing_sequence_numbers_.end();
+         missing_sequence_numbers_.end();
 }
 
 // Must be called under the critical section |crit_sect_|. Should never be
@@ -1346,18 +1364,16 @@ void VCMJitterBuffer::UpdateJitterEstimate(const VCMFrameBuffer& frame,
 // Must be called under the critical section |crit_sect_|. Should never be
 // called with retransmitted frames, they must be filtered out before this
 // function is called.
-void VCMJitterBuffer::UpdateJitterEstimate(
-    int64_t latest_packet_time_ms,
-    uint32_t timestamp,
-    unsigned int frame_size,
-    bool incomplete_frame) {
+void VCMJitterBuffer::UpdateJitterEstimate(int64_t latest_packet_time_ms,
+                                           uint32_t timestamp,
+                                           unsigned int frame_size,
+                                           bool incomplete_frame) {
   if (latest_packet_time_ms == -1) {
     return;
   }
   int64_t frame_delay;
-  bool not_reordered = inter_frame_delay_.CalculateDelay(timestamp,
-                                                      &frame_delay,
-                                                      latest_packet_time_ms);
+  bool not_reordered = inter_frame_delay_.CalculateDelay(
+      timestamp, &frame_delay, latest_packet_time_ms);
   // Filter out frames which have been reordered in time by the network
   if (not_reordered) {
     // Update the jitter estimate with the new samples
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.h b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.h
index 62d5b76900..88a9117a73 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
 
 #include <list>
 #include <map>
@@ -18,22 +18,19 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/decoding_state.h"
-#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/decoding_state.h"
+#include "webrtc/modules/video_coding/inter_frame_delay.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-enum VCMNackMode {
-  kNack,
-  kNoNack
-};
+enum VCMNackMode { kNack, kNoNack };
 
 // forward declarations
 class Clock;
@@ -54,8 +51,7 @@ struct VCMJitterSample {
 
 class TimestampLessThan {
  public:
-  bool operator() (const uint32_t& timestamp1,
-                   const uint32_t& timestamp2) const {
+  bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
     return IsNewerTimestamp(timestamp2, timestamp1);
   }
 };
@@ -69,7 +65,7 @@ class FrameList
   VCMFrameBuffer* Front() const;
   VCMFrameBuffer* Back() const;
   int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
-      UnorderedFrameList* free_frames);
+                                 UnorderedFrameList* free_frames);
   void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
                                UnorderedFrameList* free_frames);
   void Reset(UnorderedFrameList* free_frames);
@@ -108,9 +104,9 @@ class Vp9SsMap {
 
 class VCMJitterBuffer {
  public:
-  VCMJitterBuffer(Clock* clock,
-                  EventFactory* event_factory);
-  virtual ~VCMJitterBuffer();
+  VCMJitterBuffer(Clock* clock, rtc::scoped_ptr<EventWrapper> event);
+
+  ~VCMJitterBuffer();
 
   // Initializes and starts jitter buffer.
   void Start();
@@ -142,8 +138,7 @@ class VCMJitterBuffer {
   int num_discarded_packets() const;
 
   // Statistics, Calculate frame and bit rates.
-  void IncomingRateStatistics(unsigned int* framerate,
-                              unsigned int* bitrate);
+  void IncomingRateStatistics(unsigned int* framerate, unsigned int* bitrate);
 
   // Checks if the packet sequence will be complete if the next frame would be
   // grabbed for decoding. That is, if a frame has been lost between the
@@ -178,8 +173,7 @@ class VCMJitterBuffer {
   // Inserts a packet into a frame returned from GetFrame().
   // If the return value is <= 0, |frame| is invalidated and the pointer must
   // be dropped after this function returns.
-  VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
-                                  bool* retransmitted);
+  VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
 
   // Returns the estimated jitter in milliseconds.
   uint32_t EstimatedJitterMs();
@@ -187,13 +181,14 @@ class VCMJitterBuffer {
   // Updates the round-trip time estimate.
   void UpdateRtt(int64_t rtt_ms);
 
-  // Set the NACK mode. |highRttNackThreshold| is an RTT threshold in ms above
-  // which NACK will be disabled if the NACK mode is |kNackHybrid|, -1 meaning
-  // that NACK is always enabled in the hybrid mode.
-  // |lowRttNackThreshold| is an RTT threshold in ms below which we expect to
-  // rely on NACK only, and therefore are using larger buffers to have time to
-  // wait for retransmissions.
-  void SetNackMode(VCMNackMode mode, int64_t low_rtt_nack_threshold_ms,
+  // Set the NACK mode. |high_rtt_nack_threshold_ms| is an RTT threshold in ms
+  // above which NACK will be disabled if the NACK mode is |kNack|, -1 meaning
+  // that NACK is always enabled in the |kNack| mode.
+  // |low_rtt_nack_threshold_ms| is an RTT threshold in ms below which we expect
+  // to rely on NACK only, and therefore are using larger buffers to have time
+  // to wait for retransmissions.
+  void SetNackMode(VCMNackMode mode,
+                   int64_t low_rtt_nack_threshold_ms,
                    int64_t high_rtt_nack_threshold_ms);
 
   void SetNackSettings(size_t max_nack_list_size,
@@ -204,13 +199,13 @@ class VCMJitterBuffer {
   VCMNackMode nack_mode() const;
 
   // Returns a list of the sequence numbers currently missing.
-  uint16_t* GetNackList(uint16_t* nack_list_size, bool* request_key_frame);
+  std::vector<uint16_t> GetNackList(bool* request_key_frame);
 
   // Set decode error mode - Should not be changed in the middle of the
   // session. Changes will not influence frames already in the buffer.
   void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
   int64_t LastDecodedTimestamp() const;
-  VCMDecodeErrorMode decode_error_mode() const {return decode_error_mode_;}
+  VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
 
   // Used to compute time of complete continuous frames. Returns the timestamps
   // corresponding to the start and end of the continuous complete buffer.
@@ -221,8 +216,8 @@ class VCMJitterBuffer {
  private:
   class SequenceNumberLessThan {
    public:
-    bool operator() (const uint16_t& sequence_number1,
-                     const uint16_t& sequence_number2) const {
+    bool operator()(const uint16_t& sequence_number1,
+                    const uint16_t& sequence_number2) const {
       return IsNewerSequenceNumber(sequence_number2, sequence_number1);
     }
   };
@@ -378,7 +373,6 @@ class VCMJitterBuffer {
   // Holds the internal NACK list (the missing sequence numbers).
   SequenceNumberSet missing_sequence_numbers_;
   uint16_t latest_received_sequence_number_;
-  std::vector<uint16_t> nack_seq_nums_;
   size_t max_nack_list_size_;
   int max_packet_age_to_nack_;  // Measured in sequence numbers.
   int max_incomplete_time_ms_;
@@ -389,8 +383,8 @@ class VCMJitterBuffer {
   // average_packets_per_frame converges fast if we have fewer than this many
   // frames.
   int frame_counter_;
-  DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer_common.h b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer_common.h
new file mode 100644
index 0000000000..0c065a06af
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer_common.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Used to estimate rolling average of packets per frame.
+static const float kFastConvergeMultiplier = 0.4f;
+static const float kNormalConvergeMultiplier = 0.2f;
+
+enum { kMaxNumberOfFrames = 300 };
+enum { kStartNumberOfFrames = 6 };
+enum { kMaxVideoDelayMs = 10000 };
+enum { kPacketsPerFrameMultiplier = 5 };
+enum { kFastConvergeThreshold = 5 };
+
+enum VCMJitterBufferEnum {
+  kMaxConsecutiveOldFrames = 60,
+  kMaxConsecutiveOldPackets = 300,
+  // TODO(sprang): Reduce this limit once codecs don't sometimes wildly
+  // overshoot bitrate target.
+  kMaxPacketsInSession = 1400,      // Allows ~2MB frames.
+  kBufferIncStepSizeBytes = 30000,  // >20 packets.
+  kMaxJBFrameSizeBytes            = 4000000, // sanity don't go above 4Mbyte.
+  kBufferSafetyMargin             = 100      // enough for ~50 NALs in a STAP-A
+};
+
+enum VCMFrameBufferEnum {
+  kOutOfBoundsPacket = -7,
+  kNotInitialized = -6,
+  kOldPacket = -5,
+  kGeneralError = -4,
+  kFlushIndicator = -3,  // Indicator that a flush has occurred.
+  kTimeStampError = -2,
+  kSizeError = -1,
+  kNoError = 0,
+  kIncomplete = 1,        // Frame incomplete.
+  kCompleteSession = 3,   // at least one layer in the frame complete.
+  kDecodableSession = 4,  // Frame incomplete, but ready to be decoded
+  kDuplicatePacket = 5    // We're receiving a duplicate packet.
+};
+
+enum VCMFrameBufferStateEnum {
+  kStateEmpty,       // frame popped by the RTP receiver
+  kStateIncomplete,  // frame that have one or more packet(s) stored
+  kStateComplete,    // frame that have all packets
+  kStateDecodable    // Hybrid mode - frame can be decoded
+};
+
+enum { kH264StartCodeLengthBytes = 4 };
+
+// Used to indicate if a received packet contain a complete NALU (or equivalent)
+enum VCMNaluCompleteness {
+  kNaluUnset = 0,     // Packet has not been filled.
+  kNaluComplete = 1,  // Packet can be decoded as is.
+  kNaluStart,         // Packet contain beginning of NALU
+  kNaluIncomplete,    // Packet is not beginning or end of NALU
+  kNaluEnd,           // Packet is the end of a NALU
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer_unittest.cc
similarity index 63%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer_unittest.cc
index 42523e9aa2..8abc1b5471 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_buffer_unittest.cc
@@ -13,22 +13,184 @@
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/test/stream_generator.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/media_opt_util.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/test/stream_generator.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/test/histogram.h"
 
 namespace webrtc {
 
+namespace {
+const uint32_t kProcessIntervalSec = 60;
+}  // namespace
+
+class Vp9SsMapTest : public ::testing::Test {
+ protected:
+  Vp9SsMapTest() : packet_(data_, 1400, 1234, 1, true) {}
+
+  virtual void SetUp() {
+    packet_.isFirstPacket = true;
+    packet_.markerBit = true;
+    packet_.frameType = kVideoFrameKey;
+    packet_.codec = kVideoCodecVP9;
+    packet_.codecSpecificHeader.codec = kRtpVideoVp9;
+    packet_.codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
+    packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+    packet_.codecSpecificHeader.codecHeader.VP9.temporal_idx = kNoTemporalIdx;
+    packet_.codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
+    packet_.codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
+    packet_.codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
+        kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
+  }
+
+  Vp9SsMap map_;
+  uint8_t data_[1500];
+  VCMPacket packet_;
+};
+
+TEST_F(Vp9SsMapTest, Insert) {
+  EXPECT_TRUE(map_.Insert(packet_));
+}
+
+TEST_F(Vp9SsMapTest, Insert_NoSsData) {
+  packet_.codecSpecificHeader.codecHeader.VP9.ss_data_available = false;
+  EXPECT_FALSE(map_.Insert(packet_));
+}
+
+TEST_F(Vp9SsMapTest, Find) {
+  EXPECT_TRUE(map_.Insert(packet_));
+  Vp9SsMap::SsMap::iterator it;
+  EXPECT_TRUE(map_.Find(packet_.timestamp, &it));
+  EXPECT_EQ(packet_.timestamp, it->first);
+}
+
+TEST_F(Vp9SsMapTest, Find_WithWrap) {
+  const uint32_t kSsTimestamp1 = 0xFFFFFFFF;
+  const uint32_t kSsTimestamp2 = 100;
+  packet_.timestamp = kSsTimestamp1;
+  EXPECT_TRUE(map_.Insert(packet_));
+  packet_.timestamp = kSsTimestamp2;
+  EXPECT_TRUE(map_.Insert(packet_));
+  Vp9SsMap::SsMap::iterator it;
+  EXPECT_FALSE(map_.Find(kSsTimestamp1 - 1, &it));
+  EXPECT_TRUE(map_.Find(kSsTimestamp1, &it));
+  EXPECT_EQ(kSsTimestamp1, it->first);
+  EXPECT_TRUE(map_.Find(0, &it));
+  EXPECT_EQ(kSsTimestamp1, it->first);
+  EXPECT_TRUE(map_.Find(kSsTimestamp2 - 1, &it));
+  EXPECT_EQ(kSsTimestamp1, it->first);
+  EXPECT_TRUE(map_.Find(kSsTimestamp2, &it));
+  EXPECT_EQ(kSsTimestamp2, it->first);
+  EXPECT_TRUE(map_.Find(kSsTimestamp2 + 1, &it));
+  EXPECT_EQ(kSsTimestamp2, it->first);
+}
+
+TEST_F(Vp9SsMapTest, Reset) {
+  EXPECT_TRUE(map_.Insert(packet_));
+  Vp9SsMap::SsMap::iterator it;
+  EXPECT_TRUE(map_.Find(packet_.timestamp, &it));
+  EXPECT_EQ(packet_.timestamp, it->first);
+
+  map_.Reset();
+  EXPECT_FALSE(map_.Find(packet_.timestamp, &it));
+}
+
+TEST_F(Vp9SsMapTest, RemoveOld) {
+  Vp9SsMap::SsMap::iterator it;
+  const uint32_t kSsTimestamp1 = 10000;
+  packet_.timestamp = kSsTimestamp1;
+  EXPECT_TRUE(map_.Insert(packet_));
+
+  const uint32_t kTimestamp = kSsTimestamp1 + kProcessIntervalSec * 90000;
+  map_.RemoveOld(kTimestamp - 1);              // Interval not passed.
+  EXPECT_TRUE(map_.Find(kSsTimestamp1, &it));  // Should not been removed.
+
+  map_.RemoveOld(kTimestamp);
+  EXPECT_FALSE(map_.Find(kSsTimestamp1, &it));
+  EXPECT_TRUE(map_.Find(kTimestamp, &it));
+  EXPECT_EQ(kTimestamp, it->first);
+}
+
+TEST_F(Vp9SsMapTest, RemoveOld_WithWrap) {
+  Vp9SsMap::SsMap::iterator it;
+  const uint32_t kSsTimestamp1 = 0xFFFFFFFF - kProcessIntervalSec * 90000;
+  const uint32_t kSsTimestamp2 = 10;
+  const uint32_t kSsTimestamp3 = 1000;
+  packet_.timestamp = kSsTimestamp1;
+  EXPECT_TRUE(map_.Insert(packet_));
+  packet_.timestamp = kSsTimestamp2;
+  EXPECT_TRUE(map_.Insert(packet_));
+  packet_.timestamp = kSsTimestamp3;
+  EXPECT_TRUE(map_.Insert(packet_));
+
+  map_.RemoveOld(kSsTimestamp3);
+  EXPECT_FALSE(map_.Find(kSsTimestamp1, &it));
+  EXPECT_FALSE(map_.Find(kSsTimestamp2, &it));
+  EXPECT_TRUE(map_.Find(kSsTimestamp3, &it));
+}
+
+TEST_F(Vp9SsMapTest, UpdatePacket_NoSsData) {
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  EXPECT_FALSE(map_.UpdatePacket(&packet_));
+}
+
+TEST_F(Vp9SsMapTest, UpdatePacket_NoGofIdx) {
+  EXPECT_TRUE(map_.Insert(packet_));
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = kNoGofIdx;
+  EXPECT_FALSE(map_.UpdatePacket(&packet_));
+}
+
+TEST_F(Vp9SsMapTest, UpdatePacket_InvalidGofIdx) {
+  EXPECT_TRUE(map_.Insert(packet_));
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 4;
+  EXPECT_FALSE(map_.UpdatePacket(&packet_));
+}
+
+TEST_F(Vp9SsMapTest, UpdatePacket) {
+  EXPECT_TRUE(map_.Insert(packet_));  // kTemporalStructureMode3: 0-2-1-2..
+
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
+  EXPECT_TRUE(map_.UpdatePacket(&packet_));
+  EXPECT_EQ(0, packet_.codecSpecificHeader.codecHeader.VP9.temporal_idx);
+  EXPECT_FALSE(packet_.codecSpecificHeader.codecHeader.VP9.temporal_up_switch);
+  EXPECT_EQ(1U, packet_.codecSpecificHeader.codecHeader.VP9.num_ref_pics);
+  EXPECT_EQ(4, packet_.codecSpecificHeader.codecHeader.VP9.pid_diff[0]);
+
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 1;
+  EXPECT_TRUE(map_.UpdatePacket(&packet_));
+  EXPECT_EQ(2, packet_.codecSpecificHeader.codecHeader.VP9.temporal_idx);
+  EXPECT_TRUE(packet_.codecSpecificHeader.codecHeader.VP9.temporal_up_switch);
+  EXPECT_EQ(1U, packet_.codecSpecificHeader.codecHeader.VP9.num_ref_pics);
+  EXPECT_EQ(1, packet_.codecSpecificHeader.codecHeader.VP9.pid_diff[0]);
+
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 2;
+  EXPECT_TRUE(map_.UpdatePacket(&packet_));
+  EXPECT_EQ(1, packet_.codecSpecificHeader.codecHeader.VP9.temporal_idx);
+  EXPECT_TRUE(packet_.codecSpecificHeader.codecHeader.VP9.temporal_up_switch);
+  EXPECT_EQ(1U, packet_.codecSpecificHeader.codecHeader.VP9.num_ref_pics);
+  EXPECT_EQ(2, packet_.codecSpecificHeader.codecHeader.VP9.pid_diff[0]);
+
+  packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 3;
+  EXPECT_TRUE(map_.UpdatePacket(&packet_));
+  EXPECT_EQ(2, packet_.codecSpecificHeader.codecHeader.VP9.temporal_idx);
+  EXPECT_FALSE(packet_.codecSpecificHeader.codecHeader.VP9.temporal_up_switch);
+  EXPECT_EQ(2U, packet_.codecSpecificHeader.codecHeader.VP9.num_ref_pics);
+  EXPECT_EQ(1, packet_.codecSpecificHeader.codecHeader.VP9.pid_diff[0]);
+  EXPECT_EQ(2, packet_.codecSpecificHeader.codecHeader.VP9.pid_diff[1]);
+}
+
 class TestBasicJitterBuffer : public ::testing::Test {
  protected:
   virtual void SetUp() {
     clock_.reset(new SimulatedClock(0));
-    jitter_buffer_.reset(
-        new VCMJitterBuffer(clock_.get(), &event_factory_));
+    jitter_buffer_.reset(new VCMJitterBuffer(
+        clock_.get(),
+        rtc::scoped_ptr<EventWrapper>(event_factory_.CreateEvent())));
     jitter_buffer_->Start();
     seq_num_ = 1234;
     timestamp_ = 0;
@@ -71,8 +233,8 @@ class TestBasicJitterBuffer : public ::testing::Test {
   }
 
   void CheckOutFrame(VCMEncodedFrame* frame_out,
-                    unsigned int size,
-                    bool startCode) {
+                     unsigned int size,
+                     bool startCode) {
     ASSERT_TRUE(frame_out);
 
     const uint8_t* outData = frame_out->Buffer();
@@ -117,7 +279,6 @@ class TestBasicJitterBuffer : public ::testing::Test {
   rtc::scoped_ptr<VCMJitterBuffer> jitter_buffer_;
 };
 
-
 class TestRunningJitterBuffer : public ::testing::Test {
  protected:
   enum { kDataBufferSize = 10 };
@@ -126,11 +287,13 @@ class TestRunningJitterBuffer : public ::testing::Test {
     clock_.reset(new SimulatedClock(0));
     max_nack_list_size_ = 150;
     oldest_packet_to_nack_ = 250;
-    jitter_buffer_ = new VCMJitterBuffer(clock_.get(), &event_factory_);
-    stream_generator_ = new StreamGenerator(0, 0, clock_->TimeInMilliseconds());
+    jitter_buffer_ = new VCMJitterBuffer(
+        clock_.get(),
+        rtc::scoped_ptr<EventWrapper>(event_factory_.CreateEvent()));
+    stream_generator_ = new StreamGenerator(0, clock_->TimeInMilliseconds());
     jitter_buffer_->Start();
-    jitter_buffer_->SetNackSettings(max_nack_list_size_,
-                                    oldest_packet_to_nack_, 0);
+    jitter_buffer_->SetNackSettings(max_nack_list_size_, oldest_packet_to_nack_,
+                                    0);
     memset(data_buffer_, 0, kDataBufferSize);
   }
 
@@ -163,10 +326,9 @@ class TestRunningJitterBuffer : public ::testing::Test {
   }
 
   VCMFrameBufferEnum InsertFrame(FrameType frame_type) {
-    stream_generator_->GenerateFrame(frame_type,
-                                    (frame_type != kFrameEmpty) ? 1 : 0,
-                                    (frame_type == kFrameEmpty) ? 1 : 0,
-                                    clock_->TimeInMilliseconds());
+    stream_generator_->GenerateFrame(
+        frame_type, (frame_type != kEmptyFrame) ? 1 : 0,
+        (frame_type == kEmptyFrame) ? 1 : 0, clock_->TimeInMilliseconds());
     VCMFrameBufferEnum ret = InsertPacketAndPop(0);
     clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
     return ret;
@@ -232,9 +394,7 @@ class TestJitterBufferNack : public TestRunningJitterBuffer {
     jitter_buffer_->SetNackMode(kNack, -1, -1);
   }
 
-  virtual void TearDown() {
-    TestRunningJitterBuffer::TearDown();
-  }
+  virtual void TearDown() { TestRunningJitterBuffer::TearDown(); }
 };
 
 TEST_F(TestBasicJitterBuffer, StopRunning) {
@@ -267,22 +427,64 @@ TEST_F(TestBasicJitterBuffer, SinglePacketFrame) {
 
   // Insert the packet to the jitter buffer and get a frame.
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
   jitter_buffer_->ReleaseFrame(frame_out);
 }
 
+TEST_F(TestBasicJitterBuffer, VerifyHistogramStats) {
+  test::ClearHistograms();
+  // Always start with a complete key frame when not allowing errors.
+  jitter_buffer_->SetDecodeErrorMode(kNoErrors);
+  packet_->frameType = kVideoFrameKey;
+  packet_->isFirstPacket = true;
+  packet_->markerBit = true;
+  packet_->timestamp += 123 * 90;
+
+  // Insert single packet frame to the jitter buffer and get a frame.
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  // Verify that histograms are updated when the jitter buffer is stopped.
+  clock_->AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000);
+  jitter_buffer_->Stop();
+  EXPECT_EQ(
+      0, test::LastHistogramSample("WebRTC.Video.DiscardedPacketsInPercent"));
+  EXPECT_EQ(
+      0, test::LastHistogramSample("WebRTC.Video.DuplicatedPacketsInPercent"));
+  EXPECT_NE(-1, test::LastHistogramSample(
+                    "WebRTC.Video.CompleteFramesReceivedPerSecond"));
+  EXPECT_EQ(1000, test::LastHistogramSample(
+                      "WebRTC.Video.KeyFramesReceivedInPermille"));
+
+  // Verify that histograms are not updated if stop is called again.
+  jitter_buffer_->Stop();
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.DiscardedPacketsInPercent"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.DuplicatedPacketsInPercent"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+                   "WebRTC.Video.CompleteFramesReceivedPerSecond"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
+}
+
 TEST_F(TestBasicJitterBuffer, DualPacketFrame) {
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
   packet_->markerBit = false;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   // Should not be complete.
   EXPECT_TRUE(frame_out == NULL);
@@ -292,8 +494,8 @@ TEST_F(TestBasicJitterBuffer, DualPacketFrame) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
@@ -308,8 +510,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
   packet_->markerBit = false;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -324,8 +526,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     loop++;
   } while (loop < 98);
 
@@ -335,8 +537,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -352,8 +554,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
   packet_->markerBit = true;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_FALSE(frame_out == NULL);
   jitter_buffer_->ReleaseFrame(frame_out);
@@ -364,8 +566,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
   packet_->frameType = kVideoFrameDelta;
   packet_->timestamp += 33 * 90;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -380,8 +582,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
     packet_->seqNum = seq_num_;
 
     // Insert a packet into a frame.
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     loop++;
   } while (loop < 98);
 
@@ -391,8 +593,8 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -411,8 +613,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -426,8 +628,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     loop++;
   } while (loop < 98);
 
@@ -437,10 +639,10 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
-  frame_out = DecodeCompleteFrame();;
+  frame_out = DecodeCompleteFrame();
 
   CheckOutFrame(frame_out, 100 * size_, false);
 
@@ -454,8 +656,8 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   packet_->markerBit = false;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -466,23 +668,23 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   // check that we fail to get frame since seqnum is not continuous
   frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
 
   seq_num_ -= 3;
-  timestamp_ -= 33*90;
+  timestamp_ -= 33 * 90;
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -494,8 +696,8 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
@@ -508,6 +710,63 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
   jitter_buffer_->ReleaseFrame(frame_out);
 }
 
+TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) {
+  packet_->frameType = kVideoFrameKey;
+  packet_->isFirstPacket = true;
+  packet_->markerBit = true;
+
+  // Send in an initial good packet/frame (Frame A) to start things off.
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  // Now send in a complete delta frame (Frame C), but with a sequence number
+  // gap. No pic index either, so no temporal scalability cheating :)
+  packet_->frameType = kVideoFrameDelta;
+  // Leave a gap of 2 sequence numbers and two frames.
+  packet_->seqNum = seq_num_ + 3;
+  packet_->timestamp = timestamp_ + (66 * 90);
+  // Still isFirst = marker = true.
+  // Session should be complete (frame is complete), but there's nothing to
+  // decode yet.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Now send in a complete delta frame (Frame B) that is continuous from A, but
+  // doesn't fill the full gap to C. The rest of the gap is going to be padding.
+  packet_->seqNum = seq_num_ + 1;
+  packet_->timestamp = timestamp_ + (33 * 90);
+  // Still isFirst = marker = true.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  // But Frame C isn't continuous yet.
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Add in the padding. These are empty packets (data length is 0) with no
+  // marker bit and matching the timestamp of Frame B.
+  VCMPacket empty_packet(data_, 0, seq_num_ + 2, timestamp_ + (33 * 90), false);
+  EXPECT_EQ(kOldPacket,
+            jitter_buffer_->InsertPacket(empty_packet, &retransmitted));
+  empty_packet.seqNum += 1;
+  EXPECT_EQ(kOldPacket,
+            jitter_buffer_->InsertPacket(empty_packet, &retransmitted));
+
+  // But now Frame C should be ready!
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
 TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
@@ -518,8 +777,8 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -528,8 +787,8 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
 
   // Insert a packet into a frame.
-  EXPECT_EQ(kDuplicatePacket, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kDuplicatePacket,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_EQ(2, jitter_buffer_->num_packets());
   EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
 
@@ -538,8 +797,8 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
   packet_->markerBit = true;
   packet_->isFirstPacket = false;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   ASSERT_TRUE(frame_out != NULL);
@@ -603,6 +862,217 @@ TEST_F(TestBasicJitterBuffer, DuplicatePreviousDeltaFramePacket) {
   }
 }
 
+TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
+  // Verify that JB skips forward to next base layer frame.
+  //  -------------------------------------------------
+  // | 65485 | 65486 | 65487 | 65488 | 65489 | ...
+  // | pid:5 | pid:6 | pid:7 | pid:8 | pid:9 | ...
+  // | tid:0 | tid:2 | tid:1 | tid:2 | tid:0 | ...
+  // |  ss   |   x   |   x   |   x   |       |
+  //  -------------------------------------------------
+  // |<----------tl0idx:200--------->|<---tl0idx:201---
+
+  bool re = false;
+  packet_->codec = kVideoCodecVP9;
+  packet_->codecSpecificHeader.codec = kRtpVideoVp9;
+  packet_->isFirstPacket = true;
+  packet_->markerBit = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
+  packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.beginning_of_frame = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.end_of_frame = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
+
+  packet_->seqNum = 65485;
+  packet_->timestamp = 1000;
+  packet_->frameType = kVideoFrameKey;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
+  packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 200;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
+      kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  // Insert next temporal layer 0.
+  packet_->seqNum = 65489;
+  packet_->timestamp = 13000;
+  packet_->frameType = kVideoFrameDelta;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 9;
+  packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 201;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = false;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(1000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(13000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameDelta, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_3TlLayers) {
+  // Verify that frames are updated with SS data when SS packet is reordered.
+  //  --------------------------------
+  // | 65486 | 65487 | 65485 |...
+  // | pid:6 | pid:7 | pid:5 |...
+  // | tid:2 | tid:1 | tid:0 |...
+  // |       |       |  ss   |
+  //  --------------------------------
+  // |<--------tl0idx:200--------->|
+
+  bool re = false;
+  packet_->codec = kVideoCodecVP9;
+  packet_->codecSpecificHeader.codec = kRtpVideoVp9;
+  packet_->isFirstPacket = true;
+  packet_->markerBit = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
+  packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.beginning_of_frame = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.end_of_frame = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 200;
+
+  packet_->seqNum = 65486;
+  packet_->timestamp = 6000;
+  packet_->frameType = kVideoFrameDelta;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 6;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 2;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  packet_->seqNum = 65487;
+  packet_->timestamp = 9000;
+  packet_->frameType = kVideoFrameDelta;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 7;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  // Insert first frame with SS data.
+  packet_->seqNum = 65485;
+  packet_->timestamp = 3000;
+  packet_->frameType = kVideoFrameKey;
+  packet_->width = 352;
+  packet_->height = 288;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
+  packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
+      kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(3000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+  EXPECT_EQ(0, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_FALSE(
+      frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(6000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameDelta, frame_out->FrameType());
+  EXPECT_EQ(2, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_TRUE(frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(9000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameDelta, frame_out->FrameType());
+  EXPECT_EQ(1, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_TRUE(frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
+  // Verify that frames are updated with SS data when SS packet is reordered.
+  //  -----------------------------------------
+  // | 65486  | 65487  | 65485  | 65484  |...
+  // | pid:6  | pid:6  | pid:5  | pid:5  |...
+  // | tid:1  | tid:1  | tid:0  | tid:0  |...
+  // | sid:0  | sid:1  | sid:1  | sid:0  |...
+  // | t:6000 | t:6000 | t:3000 | t:3000 |
+  // |        |        |        |  ss    |
+  //  -----------------------------------------
+  // |<-----------tl0idx:200------------>|
+
+  bool re = false;
+  packet_->codec = kVideoCodecVP9;
+  packet_->codecSpecificHeader.codec = kRtpVideoVp9;
+  packet_->codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
+  packet_->codecSpecificHeader.codecHeader.VP9.beginning_of_frame = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.end_of_frame = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = 200;
+
+  packet_->isFirstPacket = true;
+  packet_->markerBit = false;
+  packet_->seqNum = 65486;
+  packet_->timestamp = 6000;
+  packet_->frameType = kVideoFrameDelta;
+  packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 6;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
+  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  packet_->isFirstPacket = false;
+  packet_->markerBit = true;
+  packet_->seqNum = 65487;
+  packet_->frameType = kVideoFrameDelta;
+  packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 6;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = true;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  packet_->isFirstPacket = false;
+  packet_->markerBit = true;
+  packet_->seqNum = 65485;
+  packet_->timestamp = 3000;
+  packet_->frameType = kVideoFrameKey;
+  packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 1;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
+  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  // Insert first frame with SS data.
+  packet_->isFirstPacket = true;
+  packet_->markerBit = false;
+  packet_->seqNum = 65484;
+  packet_->frameType = kVideoFrameKey;
+  packet_->width = 352;
+  packet_->height = 288;
+  packet_->codecSpecificHeader.codecHeader.VP9.spatial_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.picture_id = 5;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_idx = 0;
+  packet_->codecSpecificHeader.codecHeader.VP9.temporal_up_switch = false;
+  packet_->codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
+  packet_->codecSpecificHeader.codecHeader.VP9.gof.SetGofInfoVP9(
+      kTemporalStructureMode2);  // kTemporalStructureMode3: 0-1-0-1..
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(3000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+  EXPECT_EQ(0, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_FALSE(
+      frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(6000U, frame_out->TimeStamp());
+  EXPECT_EQ(kVideoFrameDelta, frame_out->FrameType());
+  EXPECT_EQ(1, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_TRUE(frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
 TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
@@ -612,8 +1082,8 @@ TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
   packet_->insertStartCode = true;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -625,8 +1095,8 @@ TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_ * 2 + 4 * 2, true);
@@ -646,8 +1116,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t timestamp = 0;
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
@@ -655,8 +1125,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->isFirstPacket = false;
   for (int i = 1; i < 9; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
@@ -665,8 +1135,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->markerBit = true;
   packet_->seqNum++;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 10 * size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -680,8 +1150,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->seqNum += 100;
   packet_->timestamp += 33 * 90 * 8;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -689,23 +1159,23 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
   packet_->seqNum -= 99;
   packet_->timestamp -= 33 * 90 * 7;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
   packet_->isFirstPacket = false;
   for (int i = 1; i < 8; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kDecodableSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
 
   packet_->seqNum++;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -717,8 +1187,7 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsThresholdCheck) {
 
   packet_->markerBit = true;
   packet_->seqNum++;
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 }
 
 // Make sure first packet is present before a frame can be decoded.
@@ -732,8 +1201,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -745,9 +1214,9 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->isFirstPacket = false;
   packet_->markerBit = false;
   packet_->seqNum += 100;
-  packet_->timestamp += 33*90*8;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  packet_->timestamp += 33 * 90 * 8;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t timestamp;
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
@@ -756,10 +1225,10 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->frameType = kVideoFrameKey;
   packet_->isFirstPacket = true;
   packet_->seqNum -= 99;
-  packet_->timestamp -= 33*90*7;
+  packet_->timestamp -= 33 * 90 * 7;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -768,8 +1237,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   packet_->isFirstPacket = false;
   for (int i = 1; i < 5; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
@@ -777,8 +1246,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsIncompleteKey) {
   // Complete key frame.
   packet_->markerBit = true;
   packet_->seqNum++;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 6 * size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -796,8 +1265,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, size_, false);
   EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
@@ -809,9 +1278,9 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   packet_->isFirstPacket = false;
   packet_->markerBit = false;
   packet_->seqNum += 100;
-  packet_->timestamp += 33*90*8;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  packet_->timestamp += 33 * 90 * 8;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t timestamp;
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
@@ -819,17 +1288,17 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   // Insert second frame with the first packet missing. Make sure we're waiting
   // for the key frame to be complete.
   packet_->seqNum -= 98;
-  packet_->timestamp -= 33*90*7;
+  packet_->timestamp -= 33 * 90 * 7;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
   for (int i = 0; i < 5; ++i) {
     packet_->seqNum++;
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
     EXPECT_FALSE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
   }
@@ -837,8 +1306,8 @@ TEST_F(TestBasicJitterBuffer, PacketLossWithSelectiveErrorsMissingFirstPacket) {
   // Add first packet. Frame should now be decodable, but incomplete.
   packet_->isFirstPacket = true;
   packet_->seqNum -= 6;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&timestamp));
 
@@ -857,8 +1326,8 @@ TEST_F(TestBasicJitterBuffer, DiscontinuousStreamWhenDecodingWithErrors) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   uint32_t next_timestamp;
   EXPECT_TRUE(jitter_buffer_->NextCompleteTimestamp(0, &next_timestamp));
   EXPECT_EQ(packet_->timestamp, next_timestamp);
@@ -874,8 +1343,8 @@ TEST_F(TestBasicJitterBuffer, DiscontinuousStreamWhenDecodingWithErrors) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // Insert a packet (so the previous one will be released).
   timestamp_ += 33 * 90;
   seq_num_ += 2;
@@ -884,8 +1353,8 @@ TEST_F(TestBasicJitterBuffer, DiscontinuousStreamWhenDecodingWithErrors) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_FALSE(jitter_buffer_->NextCompleteTimestamp(0, &next_timestamp));
   EXPECT_TRUE(jitter_buffer_->NextMaybeIncompleteTimestamp(&next_timestamp));
   EXPECT_EQ(packet_->timestamp - 33 * 90, next_timestamp);
@@ -910,12 +1379,12 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
   packet_->completeNALU = kNaluStart;
 
   bool retransmitted = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   for (int i = 0; i < 11; ++i) {
     webrtc::FrameType frametype = kVideoFrameDelta;
     seq_num_++;
-    timestamp_ += 33*90;
+    timestamp_ += 33 * 90;
     packet_->frameType = frametype;
     packet_->isFirstPacket = true;
     packet_->markerBit = false;
@@ -923,8 +1392,8 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
     packet_->timestamp = timestamp_;
     packet_->completeNALU = kNaluStart;
 
-    EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                              &retransmitted));
+    EXPECT_EQ(kDecodableSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -946,7 +1415,7 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
     packet_->completeNALU = kNaluEnd;
-    packet_->frameType = kFrameEmpty;
+    packet_->frameType = kEmptyFrame;
 
     EXPECT_EQ(jitter_buffer_->InsertPacket(*packet_, &retransmitted),
               kDecodableSession);
@@ -958,9 +1427,9 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
       CheckOutFrame(frame_out, size_, false);
 
       if (i == 0) {
-          EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+        EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
       } else {
-         EXPECT_EQ(frametype, frame_out->FrameType());
+        EXPECT_EQ(frametype, frame_out->FrameType());
       }
       EXPECT_FALSE(frame_out->Complete());
       EXPECT_FALSE(frame_out->MissingFrame());
@@ -974,18 +1443,15 @@ TEST_F(TestBasicJitterBuffer, PacketLoss) {
   timestamp_ -= 33 * 90;
   packet_->timestamp = timestamp_ - 1000;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   packet_->timestamp = timestamp_ - 500;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   packet_->timestamp = timestamp_ - 100;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   EXPECT_EQ(3, jitter_buffer_->num_discarded_packets());
 
@@ -1004,8 +1470,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
@@ -1018,8 +1484,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     frame_out = DecodeCompleteFrame();
 
@@ -1033,8 +1499,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
 
@@ -1053,8 +1519,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
   packet_->seqNum = seq_num_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
 
   // Should not be complete.
@@ -1068,8 +1534,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
 
-    EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                        &retransmitted));
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     frame_out = DecodeCompleteFrame();
 
@@ -1084,8 +1550,8 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 100 * size_, false);
@@ -1107,8 +1573,8 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrame) {
   packet_->seqNum = seq_num_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(3000u, frame_out->TimeStamp());
@@ -1124,8 +1590,7 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrame) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 }
 
 TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
@@ -1143,8 +1608,8 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(timestamp_, frame_out->TimeStamp());
@@ -1163,10 +1628,8 @@ TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-
   // This timestamp is old.
-  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 }
 
 TEST_F(TestBasicJitterBuffer, TimestampWrap) {
@@ -1183,8 +1646,8 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
@@ -1194,23 +1657,23 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
   jitter_buffer_->ReleaseFrame(frame_out);
 
   seq_num_++;
-  timestamp_ += 33*90;
+  timestamp_ += 33 * 90;
   packet_->frameType = kVideoFrameDelta;
   packet_->isFirstPacket = true;
   packet_->markerBit = false;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                       &retransmitted));
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
@@ -1220,8 +1683,8 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) {
   packet_->markerBit = true;
   packet_->seqNum = seq_num_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   CheckOutFrame(frame_out, 2 * size_, false);
@@ -1243,8 +1706,8 @@ TEST_F(TestBasicJitterBuffer, 2FrameWithTimestampWrap) {
 
   bool retransmitted = false;
   // Insert first frame (session will be complete).
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   // Insert next frame.
   seq_num_++;
@@ -1255,8 +1718,8 @@ TEST_F(TestBasicJitterBuffer, 2FrameWithTimestampWrap) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(0xffffff00, frame_out->TimeStamp());
@@ -1286,8 +1749,8 @@ TEST_F(TestBasicJitterBuffer, Insert2FramesReOrderedWithTimestampWrap) {
   packet_->timestamp = timestamp_;
 
   bool retransmitted = false;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   // Insert second frame
   seq_num_--;
@@ -1298,8 +1761,8 @@ TEST_F(TestBasicJitterBuffer, Insert2FramesReOrderedWithTimestampWrap) {
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_EQ(0xffffff00, frame_out->TimeStamp());
@@ -1326,12 +1789,12 @@ TEST_F(TestBasicJitterBuffer, DeltaFrameWithMoreThanMaxNumberOfPackets) {
     packet_->seqNum = seq_num_;
 
     if (firstPacket) {
-      EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+      EXPECT_EQ(kIncomplete,
+                jitter_buffer_->InsertPacket(*packet_, &retransmitted));
       firstPacket = false;
     } else {
-      EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
-                                                          &retransmitted));
+      EXPECT_EQ(kIncomplete,
+                jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     }
 
     loop++;
@@ -1345,10 +1808,8 @@ TEST_F(TestBasicJitterBuffer, DeltaFrameWithMoreThanMaxNumberOfPackets) {
   packet_->seqNum = seq_num_;
 
   // Insert the packet -> frame recycled.
-  EXPECT_EQ(kSizeError, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+  EXPECT_EQ(kSizeError, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   EXPECT_TRUE(NULL == DecodeCompleteFrame());
-
 }
 
 TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
@@ -1360,13 +1821,18 @@ TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
   //  --------------------------------------------------------------
   // |<-----------delta frames------------->|<------key frames----->|
 
+  // Make sure the jitter doesn't request a keyframe after too much non-
+  // decodable frames.
+  jitter_buffer_->SetNackMode(kNack, -1, -1);
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+
   int loop = 0;
   seq_num_ = 65485;
   uint32_t first_key_frame_timestamp = 0;
   bool retransmitted = false;
   // Insert MAX_NUMBER_OF_FRAMES frames.
   do {
-    timestamp_ += 33*90;
+    timestamp_ += 33 * 90;
     seq_num_++;
     packet_->isFirstPacket = true;
     packet_->markerBit = true;
@@ -1379,8 +1845,8 @@ TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
     }
 
     // Insert frame.
-    EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                             &retransmitted));
+    EXPECT_EQ(kCompleteSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
     loop++;
   } while (loop < kMaxNumberOfFrames);
@@ -1388,7 +1854,7 @@ TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
   // Max number of frames inserted.
 
   // Insert one more frame.
-  timestamp_ += 33*90;
+  timestamp_ += 33 * 90;
   seq_num_++;
   packet_->isFirstPacket = true;
   packet_->markerBit = true;
@@ -1420,10 +1886,9 @@ TEST_F(TestBasicJitterBuffer, EmptyLastFrame) {
     packet_->markerBit = false;
     packet_->seqNum = seq_num_;
     packet_->timestamp = timestamp_;
-    packet_->frameType = kFrameEmpty;
+    packet_->frameType = kEmptyFrame;
 
-    EXPECT_EQ(kNoError, jitter_buffer_->InsertPacket(*packet_,
-                                                     &retransmitted));
+    EXPECT_EQ(kNoError, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
     VCMEncodedFrame* testFrame = DecodeIncompleteFrame();
     // Timestamp should never be the last TS inserted.
     if (testFrame != NULL) {
@@ -1447,8 +1912,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->markerBit = false;
   bool retransmitted = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_ += 2;  // Skip one packet.
   packet_->seqNum = seq_num_;
@@ -1457,8 +1922,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluIncomplete;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_++;
   packet_->seqNum = seq_num_;
@@ -1467,15 +1932,15 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluEnd;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_++;
   packet_->seqNum = seq_num_;
   packet_->completeNALU = kNaluComplete;
   packet_->markerBit = true;  // Last packet.
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // The JB will only output (incomplete) frames if a packet belonging to a
   // subsequent frame was already inserted. Insert one packet of a subsequent
   // frame. place high timestamp so the JB would always have a next frame
@@ -1488,8 +1953,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluStart;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   VCMEncodedFrame* frame_out = DecodeIncompleteFrame();
 
@@ -1501,7 +1966,7 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
 
   // Test reordered start frame + 1 lost.
   seq_num_ += 2;  // Re-order 1 frame.
-  timestamp_ += 33*90;
+  timestamp_ += 33 * 90;
   insertedLength = 0;
 
   packet_->seqNum = seq_num_;
@@ -1510,9 +1975,9 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluEnd;
   packet_->markerBit = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
-  insertedLength += packet_->sizeBytes; // This packet should be decoded.
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  insertedLength += packet_->sizeBytes;  // This packet should be decoded.
   seq_num_--;
   packet_->seqNum = seq_num_;
   packet_->timestamp = timestamp_;
@@ -1521,8 +1986,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluStart;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   insertedLength += packet_->sizeBytes;  // This packet should be decoded.
 
   seq_num_ += 3;  // One packet drop.
@@ -1532,8 +1997,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluComplete;
   packet_->markerBit = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   insertedLength += packet_->sizeBytes;  // This packet should be decoded.
   seq_num_++;
   packet_->seqNum = seq_num_;
@@ -1542,8 +2007,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluStart;
   packet_->markerBit = false;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // This packet should be decoded since it's the beginning of a NAL.
   insertedLength += packet_->sizeBytes;
 
@@ -1554,8 +2019,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->isFirstPacket = false;
   packet_->completeNALU = kNaluEnd;
   packet_->markerBit = true;
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   // This packet should not be decoded because it is an incomplete NAL if it
   // is the last.
   frame_out = DecodeIncompleteFrame();
@@ -1573,8 +2038,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   emptypacket.isFirstPacket = true;
   emptypacket.completeNALU = kNaluComplete;
   emptypacket.markerBit = true;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(emptypacket,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(emptypacket, &retransmitted));
   // This packet should not be decoded because it is an incomplete NAL if it
   // is the last.
 
@@ -1595,8 +2060,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   packet_->completeNALU = kNaluComplete;
   packet_->markerBit = false;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   seq_num_++;
   emptypacket.seqNum = seq_num_;
@@ -1605,8 +2070,8 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
   emptypacket.isFirstPacket = true;
   emptypacket.completeNALU = kNaluComplete;
   emptypacket.markerBit = true;
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(emptypacket,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(emptypacket, &retransmitted));
 
   frame_out = DecodeCompleteFrame();
   // Only last NALU is complete
@@ -1625,8 +2090,8 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
   packet_->markerBit = true;
   bool retransmitted = false;
 
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_,
-                                                           &retransmitted));
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
   EXPECT_TRUE(frame_out != NULL);
   jitter_buffer_->ReleaseFrame(frame_out);
@@ -1637,9 +2102,8 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
   packet_->isFirstPacket = false;
   packet_->markerBit = false;
 
-
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeIncompleteFrame();
   EXPECT_TRUE(frame_out == NULL);
@@ -1648,8 +2112,8 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
   packet_->timestamp += 33 * 90;
   packet_->isFirstPacket = true;
 
-  EXPECT_EQ(kDecodableSession, jitter_buffer_->InsertPacket(*packet_,
-                                                            &retransmitted));
+  EXPECT_EQ(kDecodableSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
 
   frame_out = DecodeIncompleteFrame();
   CheckOutFrame(frame_out, packet_->sizeBytes, false);
@@ -1657,6 +2121,10 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
 }
 
 TEST_F(TestRunningJitterBuffer, Full) {
+  // Make sure the jitter doesn't request a keyframe after too much non-
+  // decodable frames.
+  jitter_buffer_->SetNackMode(kNack, -1, -1);
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
   // Insert a key frame and decode it.
   EXPECT_GE(InsertFrame(kVideoFrameKey), kNoError);
   EXPECT_TRUE(DecodeCompleteFrame());
@@ -1791,7 +2259,7 @@ TEST_F(TestRunningJitterBuffer, TwoPacketsNonContinuous) {
 TEST_F(TestJitterBufferNack, EmptyPackets) {
   // Make sure empty packets doesn't clog the jitter buffer.
   jitter_buffer_->SetNackMode(kNack, media_optimization::kLowRttNackMs, -1);
-  EXPECT_GE(InsertFrames(kMaxNumberOfFrames, kFrameEmpty), kNoError);
+  EXPECT_GE(InsertFrames(kMaxNumberOfFrames, kEmptyFrame), kNoError);
   InsertFrame(kVideoFrameKey);
   EXPECT_TRUE(DecodeCompleteFrame());
 }
@@ -1805,25 +2273,18 @@ TEST_F(TestJitterBufferNack, NackTooOldPackets) {
   // old packet.
   DropFrame(1);
   // Insert a frame which should trigger a recycle until the next key frame.
-  EXPECT_EQ(kFlushIndicator, InsertFrames(oldest_packet_to_nack_ + 1,
-                                          kVideoFrameDelta));
+  EXPECT_EQ(kFlushIndicator,
+            InsertFrames(oldest_packet_to_nack_ + 1, kVideoFrameDelta));
   EXPECT_FALSE(DecodeCompleteFrame());
 
-  uint16_t nack_list_length = max_nack_list_size_;
   bool request_key_frame = false;
-  uint16_t* nack_list = jitter_buffer_->GetNackList(&nack_list_length,
-                                                    &request_key_frame);
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
   // No key frame will be requested since the jitter buffer is empty.
   EXPECT_FALSE(request_key_frame);
-  EXPECT_TRUE(nack_list == NULL);
-  EXPECT_EQ(0, nack_list_length);
+  EXPECT_EQ(0u, nack_list.size());
 
   EXPECT_GE(InsertFrame(kVideoFrameDelta), kNoError);
-  // Verify that the jitter buffer requests a key frame since we need one to
-  // start decoding.
-  EXPECT_FALSE(request_key_frame);
-  EXPECT_TRUE(nack_list == NULL);
-  EXPECT_EQ(0, nack_list_length);
   // Waiting for a key frame.
   EXPECT_FALSE(DecodeCompleteFrame());
   EXPECT_FALSE(DecodeIncompleteFrame());
@@ -1844,13 +2305,13 @@ TEST_F(TestJitterBufferNack, NackLargeJitterBuffer) {
   // Insert a frame which should trigger a recycle until the next key frame.
   EXPECT_GE(InsertFrames(oldest_packet_to_nack_, kVideoFrameDelta), kNoError);
 
-  uint16_t nack_list_length = max_nack_list_size_;
   bool request_key_frame = false;
-  jitter_buffer_->GetNackList(&nack_list_length, &request_key_frame);
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
   // Verify that the jitter buffer does not request a key frame.
   EXPECT_FALSE(request_key_frame);
   // Verify that no packets are NACKed.
-  EXPECT_EQ(0, nack_list_length);
+  EXPECT_EQ(0u, nack_list.size());
   // Verify that we can decode the next frame.
   EXPECT_TRUE(DecodeCompleteFrame());
 }
@@ -1866,9 +2327,8 @@ TEST_F(TestJitterBufferNack, NackListFull) {
   EXPECT_EQ(kFlushIndicator, InsertFrame(kVideoFrameDelta));
   EXPECT_FALSE(DecodeCompleteFrame());
 
-  uint16_t nack_list_length = max_nack_list_size_;
   bool request_key_frame = false;
-  jitter_buffer_->GetNackList(&nack_list_length, &request_key_frame);
+  jitter_buffer_->GetNackList(&request_key_frame);
   // The jitter buffer is empty, so we won't request key frames until we get a
   // packet.
   EXPECT_FALSE(request_key_frame);
@@ -1876,7 +2336,7 @@ TEST_F(TestJitterBufferNack, NackListFull) {
   EXPECT_GE(InsertFrame(kVideoFrameDelta), kNoError);
   // Now we have a packet in the jitter buffer, a key frame will be requested
   // since it's not a key frame.
-  jitter_buffer_->GetNackList(&nack_list_length, &request_key_frame);
+  jitter_buffer_->GetNackList(&request_key_frame);
   // The jitter buffer is empty, so we won't request key frames until we get a
   // packet.
   EXPECT_TRUE(request_key_frame);
@@ -1893,33 +2353,29 @@ TEST_F(TestJitterBufferNack, NoNackListReturnedBeforeFirstDecode) {
   DropFrame(10);
   // Insert a frame and try to generate a NACK list. Shouldn't get one.
   EXPECT_GE(InsertFrame(kVideoFrameDelta), kNoError);
-  uint16_t nack_list_size = 0;
   bool request_key_frame = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size,
-                                                  &request_key_frame);
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
   // No list generated, and a key frame request is signaled.
-  EXPECT_TRUE(list == NULL);
-  EXPECT_EQ(0, nack_list_size);
+  EXPECT_EQ(0u, nack_list.size());
   EXPECT_TRUE(request_key_frame);
 }
 
 TEST_F(TestJitterBufferNack, NackListBuiltBeforeFirstDecode) {
-  stream_generator_->Init(0, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
   InsertFrame(kVideoFrameKey);
   stream_generator_->GenerateFrame(kVideoFrameDelta, 2, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   stream_generator_->NextPacket(NULL);  // Drop packet.
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
   EXPECT_TRUE(DecodeCompleteFrame());
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(1, nack_list_size);
-  EXPECT_TRUE(list != NULL);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(1u, nack_list.size());
 }
 
 TEST_F(TestJitterBufferNack, VerifyRetransmittedFlag) {
-  stream_generator_->Init(0, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
   stream_generator_->GenerateFrame(kVideoFrameKey, 3, 0,
                                    clock_->TimeInMilliseconds());
   VCMPacket packet;
@@ -1932,40 +2388,36 @@ TEST_F(TestJitterBufferNack, VerifyRetransmittedFlag) {
   EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(packet, &retransmitted));
   EXPECT_FALSE(retransmitted);
   EXPECT_FALSE(DecodeCompleteFrame());
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(1, nack_list_size);
-  ASSERT_TRUE(list != NULL);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(1u, nack_list.size());
   stream_generator_->PopPacket(&packet, 0);
-  EXPECT_EQ(packet.seqNum, list[0]);
-  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(packet,
-                                                           &retransmitted));
+  EXPECT_EQ(packet.seqNum, nack_list[0]);
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(packet, &retransmitted));
   EXPECT_TRUE(retransmitted);
   EXPECT_TRUE(DecodeCompleteFrame());
 }
 
 TEST_F(TestJitterBufferNack, UseNackToRecoverFirstKeyFrame) {
-  stream_generator_->Init(0, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
   stream_generator_->GenerateFrame(kVideoFrameKey, 3, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
   // Drop second packet.
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(1));
   EXPECT_FALSE(DecodeCompleteFrame());
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(1, nack_list_size);
-  ASSERT_TRUE(list != NULL);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(1u, nack_list.size());
   VCMPacket packet;
   stream_generator_->GetPacket(&packet, 0);
-  EXPECT_EQ(packet.seqNum, list[0]);
+  EXPECT_EQ(packet.seqNum, nack_list[0]);
 }
 
 TEST_F(TestJitterBufferNack, UseNackToRecoverFirstKeyFrameSecondInQueue) {
   VCMPacket packet;
-  stream_generator_->Init(0, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
   // First frame is delta.
   stream_generator_->GenerateFrame(kVideoFrameDelta, 3, 0,
                                    clock_->TimeInMilliseconds());
@@ -1980,13 +2432,11 @@ TEST_F(TestJitterBufferNack, UseNackToRecoverFirstKeyFrameSecondInQueue) {
   // Drop second packet in frame.
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(1));
   EXPECT_FALSE(DecodeCompleteFrame());
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(1, nack_list_size);
-  ASSERT_TRUE(list != NULL);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(1u, nack_list.size());
   stream_generator_->GetPacket(&packet, 0);
-  EXPECT_EQ(packet.seqNum, list[0]);
+  EXPECT_EQ(packet.seqNum, nack_list[0]);
 }
 
 TEST_F(TestJitterBufferNack, NormalOperation) {
@@ -2000,7 +2450,7 @@ TEST_F(TestJitterBufferNack, NormalOperation) {
   // | 1 | 2 | .. | 8 | 9 | x | 11 | 12 | .. | 19 | x | 21 | .. | 100 |
   //  ----------------------------------------------------------------
   stream_generator_->GenerateFrame(kVideoFrameKey, 100, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
   EXPECT_EQ(kDecodableSession, InsertPacketAndPop(0));
   // Verify that the frame is incomplete.
@@ -2016,15 +2466,14 @@ TEST_F(TestJitterBufferNack, NormalOperation) {
   EXPECT_EQ(0, stream_generator_->PacketsRemaining());
   EXPECT_FALSE(DecodeCompleteFrame());
   EXPECT_FALSE(DecodeIncompleteFrame());
-  uint16_t nack_list_size = 0;
   bool request_key_frame = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size,
-                                               &request_key_frame);
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
   // Verify the NACK list.
-  const int kExpectedNackSize = 9;
-  ASSERT_EQ(kExpectedNackSize, nack_list_size);
-  for (int i = 0; i < nack_list_size; ++i)
-    EXPECT_EQ((1 + i) * 10, list[i]);
+  const size_t kExpectedNackSize = 9;
+  ASSERT_EQ(kExpectedNackSize, nack_list.size());
+  for (size_t i = 0; i < nack_list.size(); ++i)
+    EXPECT_EQ((1 + i) * 10, nack_list[i]);
 }
 
 TEST_F(TestJitterBufferNack, NormalOperationWrap) {
@@ -2032,12 +2481,12 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap) {
   //  -------   ------------------------------------------------------------
   // | 65532 | | 65533 | 65534 | 65535 | x | 1 | .. | 9 | x | 11 |.....| 96 |
   //  -------   ------------------------------------------------------------
-  stream_generator_->Init(65532, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(65532, clock_->TimeInMilliseconds());
   InsertFrame(kVideoFrameKey);
   EXPECT_FALSE(request_key_frame);
   EXPECT_TRUE(DecodeCompleteFrame());
   stream_generator_->GenerateFrame(kVideoFrameDelta, 100, 0,
-                                  clock_->TimeInMilliseconds());
+                                   clock_->TimeInMilliseconds());
   EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
   while (stream_generator_->PacketsRemaining() > 1) {
     if (stream_generator_->NextSequenceNumber() % 10 != 0) {
@@ -2052,14 +2501,13 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap) {
   EXPECT_EQ(0, stream_generator_->PacketsRemaining());
   EXPECT_FALSE(DecodeCompleteFrame());
   EXPECT_FALSE(DecodeCompleteFrame());
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size, &extended);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
   // Verify the NACK list.
-  const int kExpectedNackSize = 10;
-  ASSERT_EQ(kExpectedNackSize, nack_list_size);
-  for (int i = 0; i < nack_list_size; ++i)
-    EXPECT_EQ(i * 10, list[i]);
+  const size_t kExpectedNackSize = 10;
+  ASSERT_EQ(kExpectedNackSize, nack_list.size());
+  for (size_t i = 0; i < nack_list.size(); ++i)
+    EXPECT_EQ(i * 10, nack_list[i]);
 }
 
 TEST_F(TestJitterBufferNack, NormalOperationWrap2) {
@@ -2067,7 +2515,7 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap2) {
   //  -----------------------------------
   // | 65532 | 65533 | 65534 | x | 0 | 1 |
   //  -----------------------------------
-  stream_generator_->Init(65532, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(65532, clock_->TimeInMilliseconds());
   InsertFrame(kVideoFrameKey);
   EXPECT_FALSE(request_key_frame);
   EXPECT_TRUE(DecodeCompleteFrame());
@@ -2075,7 +2523,7 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap2) {
                                    clock_->TimeInMilliseconds());
   clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
   for (int i = 0; i < 5; ++i) {
-    if (stream_generator_->NextSequenceNumber()  != 65535) {
+    if (stream_generator_->NextSequenceNumber() != 65535) {
       EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
       EXPECT_FALSE(request_key_frame);
     } else {
@@ -2087,39 +2535,37 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap2) {
   }
   EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
   EXPECT_FALSE(request_key_frame);
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  uint16_t* list = jitter_buffer_->GetNackList(&nack_list_size, &extended);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
   // Verify the NACK list.
-  ASSERT_EQ(1, nack_list_size);
-  EXPECT_EQ(65535, list[0]);
+  ASSERT_EQ(1u, nack_list.size());
+  EXPECT_EQ(65535, nack_list[0]);
 }
 
 TEST_F(TestJitterBufferNack, ResetByFutureKeyFrameDoesntError) {
-  stream_generator_->Init(0, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
   InsertFrame(kVideoFrameKey);
   EXPECT_TRUE(DecodeCompleteFrame());
-  uint16_t nack_list_size = 0;
   bool extended = false;
-  jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(0, nack_list_size);
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(0u, nack_list.size());
 
   // Far-into-the-future video frame, could be caused by resetting the encoder
   // or otherwise restarting. This should not fail when error when the packet is
   // a keyframe, even if all of the nack list needs to be flushed.
-  stream_generator_->Init(10000, 0, clock_->TimeInMilliseconds());
+  stream_generator_->Init(10000, clock_->TimeInMilliseconds());
   clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
   InsertFrame(kVideoFrameKey);
   EXPECT_TRUE(DecodeCompleteFrame());
-  jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(0, nack_list_size);
+  nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(0u, nack_list.size());
 
   // Stream should be decodable from this point.
   clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
   InsertFrame(kVideoFrameDelta);
   EXPECT_TRUE(DecodeCompleteFrame());
-  jitter_buffer_->GetNackList(&nack_list_size, &extended);
-  EXPECT_EQ(0, nack_list_size);
+  nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(0u, nack_list.size());
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator.cc b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator.cc
new file mode 100644
index 0000000000..c416cbbe73
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator.cc
@@ -0,0 +1,445 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <string>
+
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/rtt_filter.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+enum { kStartupDelaySamples = 30 };
+enum { kFsAccuStartupSamples = 5 };
+enum { kMaxFramerateEstimate = 200 };
+
+VCMJitterEstimator::VCMJitterEstimator(const Clock* clock,
+                                       int32_t vcmId,
+                                       int32_t receiverId)
+    : _vcmId(vcmId),
+      _receiverId(receiverId),
+      _phi(0.97),
+      _psi(0.9999),
+      _alphaCountMax(400),
+      _thetaLow(0.000001),
+      _nackLimit(3),
+      _numStdDevDelayOutlier(15),
+      _numStdDevFrameSizeOutlier(3),
+      _noiseStdDevs(2.33),       // ~Less than 1% chance
+                                 // (look up in normal distribution table)...
+      _noiseStdDevOffset(30.0),  // ...of getting 30 ms freezes
+      _rttFilter(),
+      fps_counter_(30),  // TODO(sprang): Use an estimator with limit based on
+                         // time, rather than number of samples.
+      low_rate_experiment_(kInit),
+      clock_(clock) {
+  Reset();
+}
+
+VCMJitterEstimator::~VCMJitterEstimator() {}
+
+VCMJitterEstimator& VCMJitterEstimator::operator=(
+    const VCMJitterEstimator& rhs) {
+  if (this != &rhs) {
+    memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
+    memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
+
+    _vcmId = rhs._vcmId;
+    _receiverId = rhs._receiverId;
+    _avgFrameSize = rhs._avgFrameSize;
+    _varFrameSize = rhs._varFrameSize;
+    _maxFrameSize = rhs._maxFrameSize;
+    _fsSum = rhs._fsSum;
+    _fsCount = rhs._fsCount;
+    _lastUpdateT = rhs._lastUpdateT;
+    _prevEstimate = rhs._prevEstimate;
+    _prevFrameSize = rhs._prevFrameSize;
+    _avgNoise = rhs._avgNoise;
+    _alphaCount = rhs._alphaCount;
+    _filterJitterEstimate = rhs._filterJitterEstimate;
+    _startupCount = rhs._startupCount;
+    _latestNackTimestamp = rhs._latestNackTimestamp;
+    _nackCount = rhs._nackCount;
+    _rttFilter = rhs._rttFilter;
+  }
+  return *this;
+}
+
+// Resets the JitterEstimate
+void VCMJitterEstimator::Reset() {
+  _theta[0] = 1 / (512e3 / 8);
+  _theta[1] = 0;
+  _varNoise = 4.0;
+
+  _thetaCov[0][0] = 1e-4;
+  _thetaCov[1][1] = 1e2;
+  _thetaCov[0][1] = _thetaCov[1][0] = 0;
+  _Qcov[0][0] = 2.5e-10;
+  _Qcov[1][1] = 1e-10;
+  _Qcov[0][1] = _Qcov[1][0] = 0;
+  _avgFrameSize = 500;
+  _maxFrameSize = 500;
+  _varFrameSize = 100;
+  _lastUpdateT = -1;
+  _prevEstimate = -1.0;
+  _prevFrameSize = 0;
+  _avgNoise = 0.0;
+  _alphaCount = 1;
+  _filterJitterEstimate = 0.0;
+  _latestNackTimestamp = 0;
+  _nackCount = 0;
+  _fsSum = 0;
+  _fsCount = 0;
+  _startupCount = 0;
+  _rttFilter.Reset();
+  fps_counter_.Reset();
+}
+
+void VCMJitterEstimator::ResetNackCount() {
+  _nackCount = 0;
+}
+
+// Updates the estimates with the new measurements
+void VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS,
+                                        uint32_t frameSizeBytes,
+                                        bool incompleteFrame /* = false */) {
+  if (frameSizeBytes == 0) {
+    return;
+  }
+  int deltaFS = frameSizeBytes - _prevFrameSize;
+  if (_fsCount < kFsAccuStartupSamples) {
+    _fsSum += frameSizeBytes;
+    _fsCount++;
+  } else if (_fsCount == kFsAccuStartupSamples) {
+    // Give the frame size filter
+    _avgFrameSize = static_cast<double>(_fsSum) / static_cast<double>(_fsCount);
+    _fsCount++;
+  }
+  if (!incompleteFrame || frameSizeBytes > _avgFrameSize) {
+    double avgFrameSize = _phi * _avgFrameSize + (1 - _phi) * frameSizeBytes;
+    if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize)) {
+      // Only update the average frame size if this sample wasn't a
+      // key frame
+      _avgFrameSize = avgFrameSize;
+    }
+    // Update the variance anyway since we want to capture cases where we only
+    // get
+    // key frames.
+    _varFrameSize = VCM_MAX(_phi * _varFrameSize +
+                                (1 - _phi) * (frameSizeBytes - avgFrameSize) *
+                                    (frameSizeBytes - avgFrameSize),
+                            1.0);
+  }
+
+  // Update max frameSize estimate
+  _maxFrameSize =
+      VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
+
+  if (_prevFrameSize == 0) {
+    _prevFrameSize = frameSizeBytes;
+    return;
+  }
+  _prevFrameSize = frameSizeBytes;
+
+  // Only update the Kalman filter if the sample is not considered
+  // an extreme outlier. Even if it is an extreme outlier from a
+  // delay point of view, if the frame size also is large the
+  // deviation is probably due to an incorrect line slope.
+  double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
+
+  if (fabs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
+      frameSizeBytes >
+          _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize)) {
+    // Update the variance of the deviation from the
+    // line given by the Kalman filter
+    EstimateRandomJitter(deviation, incompleteFrame);
+    // Prevent updating with frames which have been congested by a large
+    // frame, and therefore arrives almost at the same time as that frame.
+    // This can occur when we receive a large frame (key frame) which
+    // has been delayed. The next frame is of normal size (delta frame),
+    // and thus deltaFS will be << 0. This removes all frame samples
+    // which arrives after a key frame.
+    if ((!incompleteFrame || deviation >= 0.0) &&
+        static_cast<double>(deltaFS) > -0.25 * _maxFrameSize) {
+      // Update the Kalman filter with the new data
+      KalmanEstimateChannel(frameDelayMS, deltaFS);
+    }
+  } else {
+    int nStdDev =
+        (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
+    EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
+  }
+  // Post process the total estimated jitter
+  if (_startupCount >= kStartupDelaySamples) {
+    PostProcessEstimate();
+  } else {
+    _startupCount++;
+  }
+}
+
+// Updates the nack/packet ratio
+void VCMJitterEstimator::FrameNacked() {
+  // Wait until _nackLimit retransmissions has been received,
+  // then always add ~1 RTT delay.
+  // TODO(holmer): Should we ever remove the additional delay if the
+  // the packet losses seem to have stopped? We could for instance scale
+  // the number of RTTs to add with the amount of retransmissions in a given
+  // time interval, or similar.
+  if (_nackCount < _nackLimit) {
+    _nackCount++;
+  }
+}
+
+// Updates Kalman estimate of the channel
+// The caller is expected to sanity check the inputs.
+void VCMJitterEstimator::KalmanEstimateChannel(int64_t frameDelayMS,
+                                               int32_t deltaFSBytes) {
+  double Mh[2];
+  double hMh_sigma;
+  double kalmanGain[2];
+  double measureRes;
+  double t00, t01;
+
+  // Kalman filtering
+
+  // Prediction
+  // M = M + Q
+  _thetaCov[0][0] += _Qcov[0][0];
+  _thetaCov[0][1] += _Qcov[0][1];
+  _thetaCov[1][0] += _Qcov[1][0];
+  _thetaCov[1][1] += _Qcov[1][1];
+
+  // Kalman gain
+  // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
+  // h = [dFS 1]
+  // Mh = M*h'
+  // hMh_sigma = h*M*h' + R
+  Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
+  Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
+  // sigma weights measurements with a small deltaFS as noisy and
+  // measurements with large deltaFS as good
+  if (_maxFrameSize < 1.0) {
+    return;
+  }
+  double sigma = (300.0 * exp(-fabs(static_cast<double>(deltaFSBytes)) /
+                              (1e0 * _maxFrameSize)) +
+                  1) *
+                 sqrt(_varNoise);
+  if (sigma < 1.0) {
+    sigma = 1.0;
+  }
+  hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
+  if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) ||
+      (hMh_sigma > -1e-9 && hMh_sigma <= 0)) {
+    assert(false);
+    return;
+  }
+  kalmanGain[0] = Mh[0] / hMh_sigma;
+  kalmanGain[1] = Mh[1] / hMh_sigma;
+
+  // Correction
+  // theta = theta + K*(dT - h*theta)
+  measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
+  _theta[0] += kalmanGain[0] * measureRes;
+  _theta[1] += kalmanGain[1] * measureRes;
+
+  if (_theta[0] < _thetaLow) {
+    _theta[0] = _thetaLow;
+  }
+
+  // M = (I - K*h)*M
+  t00 = _thetaCov[0][0];
+  t01 = _thetaCov[0][1];
+  _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
+                    kalmanGain[0] * _thetaCov[1][0];
+  _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
+                    kalmanGain[0] * _thetaCov[1][1];
+  _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
+                    kalmanGain[1] * deltaFSBytes * t00;
+  _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
+                    kalmanGain[1] * deltaFSBytes * t01;
+
+  // Covariance matrix, must be positive semi-definite
+  assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
+         _thetaCov[0][0] * _thetaCov[1][1] -
+                 _thetaCov[0][1] * _thetaCov[1][0] >=
+             0 &&
+         _thetaCov[0][0] >= 0);
+}
+
+// Calculate difference in delay between a sample and the
+// expected delay estimated by the Kalman filter
+double VCMJitterEstimator::DeviationFromExpectedDelay(
+    int64_t frameDelayMS,
+    int32_t deltaFSBytes) const {
+  return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
+}
+
+// Estimates the random jitter by calculating the variance of the
+// sample distance from the line given by theta.
+void VCMJitterEstimator::EstimateRandomJitter(double d_dT,
+                                              bool incompleteFrame) {
+  uint64_t now = clock_->TimeInMicroseconds();
+  if (_lastUpdateT != -1) {
+    fps_counter_.AddSample(now - _lastUpdateT);
+  }
+  _lastUpdateT = now;
+
+  if (_alphaCount == 0) {
+    assert(false);
+    return;
+  }
+  double alpha =
+      static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
+  _alphaCount++;
+  if (_alphaCount > _alphaCountMax)
+    _alphaCount = _alphaCountMax;
+
+  if (LowRateExperimentEnabled()) {
+    // In order to avoid a low frame rate stream to react slower to changes,
+    // scale the alpha weight relative a 30 fps stream.
+    double fps = GetFrameRate();
+    if (fps > 0.0) {
+      double rate_scale = 30.0 / fps;
+      // At startup, there can be a lot of noise in the fps estimate.
+      // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
+      // at sample #kStartupDelaySamples.
+      if (_alphaCount < kStartupDelaySamples) {
+        rate_scale =
+            (_alphaCount * rate_scale + (kStartupDelaySamples - _alphaCount)) /
+            kStartupDelaySamples;
+      }
+      alpha = pow(alpha, rate_scale);
+    }
+  }
+
+  double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
+  double varNoise =
+      alpha * _varNoise + (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
+  if (!incompleteFrame || varNoise > _varNoise) {
+    _avgNoise = avgNoise;
+    _varNoise = varNoise;
+  }
+  if (_varNoise < 1.0) {
+    // The variance should never be zero, since we might get
+    // stuck and consider all samples as outliers.
+    _varNoise = 1.0;
+  }
+}
+
+double VCMJitterEstimator::NoiseThreshold() const {
+  double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
+  if (noiseThreshold < 1.0) {
+    noiseThreshold = 1.0;
+  }
+  return noiseThreshold;
+}
+
+// Calculates the current jitter estimate from the filtered estimates
+double VCMJitterEstimator::CalculateEstimate() {
+  double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
+
+  // A very low estimate (or negative) is neglected
+  if (ret < 1.0) {
+    if (_prevEstimate <= 0.01) {
+      ret = 1.0;
+    } else {
+      ret = _prevEstimate;
+    }
+  }
+  if (ret > 10000.0) {  // Sanity
+    ret = 10000.0;
+  }
+  _prevEstimate = ret;
+  return ret;
+}
+
+void VCMJitterEstimator::PostProcessEstimate() {
+  _filterJitterEstimate = CalculateEstimate();
+}
+
+void VCMJitterEstimator::UpdateRtt(int64_t rttMs) {
+  _rttFilter.Update(rttMs);
+}
+
+void VCMJitterEstimator::UpdateMaxFrameSize(uint32_t frameSizeBytes) {
+  if (_maxFrameSize < frameSizeBytes) {
+    _maxFrameSize = frameSizeBytes;
+  }
+}
+
+// Returns the current filtered estimate if available,
+// otherwise tries to calculate an estimate.
+int VCMJitterEstimator::GetJitterEstimate(double rttMultiplier) {
+  double jitterMS = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
+  if (_filterJitterEstimate > jitterMS)
+    jitterMS = _filterJitterEstimate;
+  if (_nackCount >= _nackLimit)
+    jitterMS += _rttFilter.RttMs() * rttMultiplier;
+
+  if (LowRateExperimentEnabled()) {
+    static const double kJitterScaleLowThreshold = 5.0;
+    static const double kJitterScaleHighThreshold = 10.0;
+    double fps = GetFrameRate();
+    // Ignore jitter for very low fps streams.
+    if (fps < kJitterScaleLowThreshold) {
+      if (fps == 0.0) {
+        return jitterMS;
+      }
+      return 0;
+    }
+
+    // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
+    // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
+    if (fps < kJitterScaleHighThreshold) {
+      jitterMS =
+          (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
+          (fps - kJitterScaleLowThreshold) * jitterMS;
+    }
+  }
+
+  return static_cast<uint32_t>(jitterMS + 0.5);
+}
+
+bool VCMJitterEstimator::LowRateExperimentEnabled() {
+#ifndef WEBRTC_MOZILLA_BUILD
+  if (low_rate_experiment_ == kInit) {
+    std::string group =
+        webrtc::field_trial::FindFullName("WebRTC-ReducedJitterDelay");
+    if (group == "Disabled") {
+      low_rate_experiment_ = kDisabled;
+    } else {
+      low_rate_experiment_ = kEnabled;
+    }
+  }
+#endif
+  return low_rate_experiment_ == kEnabled ? true : false;
+}
+
+double VCMJitterEstimator::GetFrameRate() const {
+  if (fps_counter_.count() == 0)
+    return 0;
+
+  double fps = 1000000.0 / fps_counter_.ComputeMean();
+  // Sanity check.
+  assert(fps >= 0.0);
+  if (fps > kMaxFramerateEstimate) {
+    fps = kMaxFramerateEstimate;
+  }
+  return fps;
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator.h b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator.h
new file mode 100644
index 0000000000..a7b4b3e3df
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator.h
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
+
+#include "webrtc/base/rollingaccumulator.h"
+#include "webrtc/modules/video_coding/rtt_filter.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+
+class VCMJitterEstimator {
+ public:
+  VCMJitterEstimator(const Clock* clock,
+                     int32_t vcmId = 0,
+                     int32_t receiverId = 0);
+  virtual ~VCMJitterEstimator();
+  VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
+
+  // Resets the estimate to the initial state
+  void Reset();
+  void ResetNackCount();
+
+  // Updates the jitter estimate with the new data.
+  //
+  // Input:
+  //          - frameDelay      : Delay-delta calculated by UTILDelayEstimate in
+  //          milliseconds
+  //          - frameSize       : Frame size of the current frame.
+  //          - incompleteFrame : Flags if the frame is used to update the
+  //          estimate before it
+  //                              was complete. Default is false.
+  void UpdateEstimate(int64_t frameDelayMS,
+                      uint32_t frameSizeBytes,
+                      bool incompleteFrame = false);
+
+  // Returns the current jitter estimate in milliseconds and adds
+  // also adds an RTT dependent term in cases of retransmission.
+  //  Input:
+  //          - rttMultiplier  : RTT param multiplier (when applicable).
+  //
+  // Return value                   : Jitter estimate in milliseconds
+  int GetJitterEstimate(double rttMultiplier);
+
+  // Updates the nack counter.
+  void FrameNacked();
+
+  // Updates the RTT filter.
+  //
+  // Input:
+  //          - rttMs               : RTT in ms
+  void UpdateRtt(int64_t rttMs);
+
+  void UpdateMaxFrameSize(uint32_t frameSizeBytes);
+
+  // A constant describing the delay from the jitter buffer
+  // to the delay on the receiving side which is not accounted
+  // for by the jitter buffer nor the decoding delay estimate.
+  static const uint32_t OPERATING_SYSTEM_JITTER = 10;
+
+ protected:
+  // These are protected for better testing possibilities
+  double _theta[2];  // Estimated line parameters (slope, offset)
+  double _varNoise;  // Variance of the time-deviation from the line
+
+  virtual bool LowRateExperimentEnabled();
+
+ private:
+  // Updates the Kalman filter for the line describing
+  // the frame size dependent jitter.
+  //
+  // Input:
+  //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in
+  //          milliseconds
+  //          - deltaFSBytes    : Frame size delta, i.e.
+  //                            : frame size at time T minus frame size at time
+  //                            T-1
+  void KalmanEstimateChannel(int64_t frameDelayMS, int32_t deltaFSBytes);
+
+  // Updates the random jitter estimate, i.e. the variance
+  // of the time deviations from the line given by the Kalman filter.
+  //
+  // Input:
+  //          - d_dT              : The deviation from the kalman estimate
+  //          - incompleteFrame   : True if the frame used to update the
+  //          estimate
+  //                                with was incomplete
+  void EstimateRandomJitter(double d_dT, bool incompleteFrame);
+
+  double NoiseThreshold() const;
+
+  // Calculates the current jitter estimate.
+  //
+  // Return value                 : The current jitter estimate in milliseconds
+  double CalculateEstimate();
+
+  // Post process the calculated estimate
+  void PostProcessEstimate();
+
+  // Calculates the difference in delay between a sample and the
+  // expected delay estimated by the Kalman filter.
+  //
+  // Input:
+  //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in
+  //          milliseconds
+  //          - deltaFS         : Frame size delta, i.e. frame size at time
+  //                              T minus frame size at time T-1
+  //
+  // Return value                 : The difference in milliseconds
+  double DeviationFromExpectedDelay(int64_t frameDelayMS,
+                                    int32_t deltaFSBytes) const;
+
+  double GetFrameRate() const;
+
+  // Constants, filter parameters
+  int32_t _vcmId;
+  int32_t _receiverId;
+  const double _phi;
+  const double _psi;
+  const uint32_t _alphaCountMax;
+  const double _thetaLow;
+  const uint32_t _nackLimit;
+  const int32_t _numStdDevDelayOutlier;
+  const int32_t _numStdDevFrameSizeOutlier;
+  const double _noiseStdDevs;
+  const double _noiseStdDevOffset;
+
+  double _thetaCov[2][2];  // Estimate covariance
+  double _Qcov[2][2];      // Process noise covariance
+  double _avgFrameSize;    // Average frame size
+  double _varFrameSize;    // Frame size variance
+  double _maxFrameSize;    // Largest frame size received (descending
+                           // with a factor _psi)
+  uint32_t _fsSum;
+  uint32_t _fsCount;
+
+  int64_t _lastUpdateT;
+  double _prevEstimate;     // The previously returned jitter estimate
+  uint32_t _prevFrameSize;  // Frame size of the previous frame
+  double _avgNoise;         // Average of the random jitter
+  uint32_t _alphaCount;
+  double _filterJitterEstimate;  // The filtered sum of jitter estimates
+
+  uint32_t _startupCount;
+
+  int64_t
+      _latestNackTimestamp;  // Timestamp in ms when the latest nack was seen
+  uint32_t _nackCount;       // Keeps track of the number of nacks received,
+                             // but never goes above _nackLimit
+  VCMRttFilter _rttFilter;
+
+  rtc::RollingAccumulator<uint64_t> fps_counter_;
+  enum ExperimentFlag { kInit, kEnabled, kDisabled };
+  ExperimentFlag low_rate_experiment_;
+  const Clock* clock_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator_tests.cc b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator_tests.cc
similarity index 97%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator_tests.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator_tests.cc
index 5f34750572..3d46ce2bcd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator_tests.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/jitter_estimator_tests.cc
@@ -7,10 +7,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/video_coding.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/video_coding.h
deleted file mode 100644
index 1fbe19a58f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/interface/video_coding.h
+++ /dev/null
@@ -1,609 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
-#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
-
-#if defined(WEBRTC_WIN)
-// This is a workaround on Windows due to the fact that some Windows
-// headers define CreateEvent as a macro to either CreateEventW or CreateEventA.
-// This can cause problems since we use that name as well and could
-// declare them as one thing here whereas in another place a windows header
-// may have been included and then implementing CreateEvent() causes compilation
-// errors.  So for consistency, we include the main windows header here.
-#include <windows.h>
-#endif
-
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-
-namespace webrtc
-{
-
-class Clock;
-class EncodedImageCallback;
-class VideoEncoder;
-class VideoDecoder;
-struct CodecSpecificInfo;
-
-class EventFactory {
- public:
-  virtual ~EventFactory() {}
-
-  virtual EventWrapper* CreateEvent() = 0;
-};
-
-class EventFactoryImpl : public EventFactory {
- public:
-  virtual ~EventFactoryImpl() {}
-
-  virtual EventWrapper* CreateEvent() {
-    return EventWrapper::Create();
-  }
-};
-
-// Used to indicate which decode with errors mode should be used.
-enum VCMDecodeErrorMode {
-  kNoErrors,                // Never decode with errors. Video will freeze
-                            // if nack is disabled.
-  kSelectiveErrors,         // Frames that are determined decodable in
-                            // VCMSessionInfo may be decoded with missing
-                            // packets. As not all incomplete frames will be
-                            // decodable, video will freeze if nack is disabled.
-  kWithErrors               // Release frames as needed. Errors may be
-                            // introduced as some encoded frames may not be
-                            // complete.
-};
-
-class VideoCodingModule : public Module
-{
-public:
-    enum SenderNackMode {
-        kNackNone,
-        kNackAll,
-        kNackSelective
-    };
-
-    enum ReceiverRobustness {
-        kNone,
-        kHardNack,
-        kSoftNack,
-        kReferenceSelection
-    };
-
-    static VideoCodingModule* Create(
-        VideoEncoderRateObserver* encoder_rate_observer);
-
-    static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
-
-    static void Destroy(VideoCodingModule* module);
-
-    // Get number of supported codecs
-    //
-    // Return value     : Number of supported codecs
-    static uint8_t NumberOfCodecs();
-
-    // Get supported codec settings with using id
-    //
-    // Input:
-    //      - listId         : Id or index of the codec to look up
-    //      - codec          : Memory where the codec settings will be stored
-    //
-    // Return value     : VCM_OK,              on success
-    //                    VCM_PARAMETER_ERROR  if codec not supported or id too high
-    static int32_t Codec(const uint8_t listId, VideoCodec* codec);
-
-    // Get supported codec settings using codec type
-    //
-    // Input:
-    //      - codecType      : The codec type to get settings for
-    //      - codec          : Memory where the codec settings will be stored
-    //
-    // Return value     : VCM_OK,              on success
-    //                    VCM_PARAMETER_ERROR  if codec not supported
-    static int32_t Codec(VideoCodecType codecType, VideoCodec* codec);
-
-    /*
-    *   Sender
-    */
-
-    // Any encoder-related state of VCM will be initialized to the
-    // same state as when the VCM was created. This will not interrupt
-    // or effect decoding functionality of VCM. VCM will lose all the
-    // encoding-related settings by calling this function.
-    // For instance, a send codec has to be registered again.
-    //
-    // NOTE: Must be called on the thread that constructed the VCM instance.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t InitializeSender() = 0;
-
-    // Registers a codec to be used for encoding. Calling this
-    // API multiple times overwrites any previously registered codecs.
-    //
-    // NOTE: Must be called on the thread that constructed the VCM instance.
-    //
-    // Input:
-    //      - sendCodec      : Settings for the codec to be registered.
-    //      - numberOfCores  : The number of cores the codec is allowed
-    //                         to use.
-    //      - maxPayloadSize : The maximum size each payload is allowed
-    //                                to have. Usually MTU - overhead.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterSendCodec(const VideoCodec* sendCodec,
-                                            uint32_t numberOfCores,
-                                            uint32_t maxPayloadSize) = 0;
-
-    // Get the current send codec in use.
-    //
-    // If a codec has not been set yet, the |id| property of the return value
-    // will be 0 and |name| empty.
-    //
-    // NOTE: This method intentionally does not hold locks and minimizes data
-    // copying.  It must be called on the thread where the VCM was constructed.
-    virtual const VideoCodec& GetSendCodec() const = 0;
-
-    // DEPRECATED: Use GetSendCodec() instead.
-    //
-    // API to get the current send codec in use.
-    //
-    // Input:
-    //      - currentSendCodec : Address where the sendCodec will be written.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    //
-    // NOTE: The returned codec information is not guaranteed to be current when
-    // the call returns.  This method acquires a lock that is aligned with
-    // video encoding, so it should be assumed to be allowed to block for
-    // several milliseconds.
-    virtual int32_t SendCodec(VideoCodec* currentSendCodec) const = 0;
-
-    // DEPRECATED: Use GetSendCodec() instead.
-    //
-    // API to get the current send codec type
-    //
-    // Return value      : Codec type, on success.
-    //                     kVideoCodecUnknown, on error or if no send codec is set
-    // NOTE: Same notes apply as for SendCodec() above.
-    virtual VideoCodecType SendCodec() const = 0;
-
-    // Register an external encoder object. This can not be used together with
-    // external decoder callbacks.
-    //
-    // Input:
-    //      - externalEncoder : Encoder object to be used for encoding frames inserted
-    //                          with the AddVideoFrame API.
-    //      - payloadType     : The payload type bound which this encoder is bound to.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                            uint8_t payloadType,
-                                            bool internalSource = false) = 0;
-
-    // API to get codec config parameters to be sent out-of-band to a receiver.
-    //
-    // Input:
-    //      - buffer          : Memory where the codec config parameters should be written.
-    //      - size            : Size of the memory available.
-    //
-    // Return value      : Number of bytes written, on success.
-    //                     < 0,                     on error.
-    virtual int32_t CodecConfigParameters(uint8_t* buffer, int32_t size) = 0;
-
-    // API to get currently configured encoder target bitrate in bits/s.
-    //
-    // Return value      : 0,   on success.
-    //                     < 0, on error.
-    virtual int Bitrate(unsigned int* bitrate) const = 0;
-
-    // API to get currently configured encoder target frame rate.
-    //
-    // Return value      : 0,   on success.
-    //                     < 0, on error.
-    virtual int FrameRate(unsigned int* framerate) const = 0;
-
-    // Sets the parameters describing the send channel. These parameters are inputs to the
-    // Media Optimization inside the VCM and also specifies the target bit rate for the
-    // encoder. Bit rate used by NACK should already be compensated for by the user.
-    //
-    // Input:
-    //      - target_bitrate        : The target bitrate for VCM in bits/s.
-    //      - lossRate              : Fractions of lost packets the past second.
-    //                                (loss rate in percent = 100 * packetLoss / 255)
-    //      - rtt                   : Current round-trip time in ms.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t SetChannelParameters(uint32_t target_bitrate,
-                                         uint8_t lossRate,
-                                         int64_t rtt) = 0;
-
-    // Sets the parameters describing the receive channel. These parameters are inputs to the
-    // Media Optimization inside the VCM.
-    //
-    // Input:
-    //      - rtt                   : Current round-trip time in ms.
-    //                                with the most amount available bandwidth in a conference
-    //                                scenario
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
-
-    // Register a transport callback which will be called to deliver the encoded data and
-    // side information.
-    //
-    // Input:
-    //      - transport  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterTransportCallback(VCMPacketizationCallback* transport) = 0;
-
-    // Register video output information callback which will be called to deliver information
-    // about the video stream produced by the encoder, for instance the average frame rate and
-    // bit rate.
-    //
-    // Input:
-    //      - outputInformation  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterSendStatisticsCallback(
-                                     VCMSendStatisticsCallback* sendStats) = 0;
-
-    // Register a video quality settings callback which will be called when
-    // frame rate/dimensions need to be updated for video quality optimization
-    //
-    // Input:
-    //      - videoQMSettings  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error
-    virtual int32_t RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings) = 0;
-
-    // Register a video protection callback which will be called to deliver
-    // the requested FEC rate and NACK status (on/off).
-    //
-    // Input:
-    //      - protection  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterProtectionCallback(VCMProtectionCallback* protection) = 0;
-
-    // Enable or disable a video protection method.
-    //
-    // Input:
-    //      - videoProtection  : The method to enable or disable.
-    //      - enable           : True if the method should be enabled, false if
-    //                           it should be disabled.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t SetVideoProtection(VCMVideoProtection videoProtection,
-                                       bool enable) = 0;
-
-    // Add one raw video frame to the encoder. This function does all the necessary
-    // processing, then decides what frame type to encode, or if the frame should be
-    // dropped. If the frame should be encoded it passes the frame to the encoder
-    // before it returns.
-    //
-    // Input:
-    //      - videoFrame        : Video frame to encode.
-    //      - codecSpecificInfo : Extra codec information, e.g., pre-parsed in-band signaling.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t AddVideoFrame(
-        const I420VideoFrame& videoFrame,
-        const VideoContentMetrics* contentMetrics = NULL,
-        const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
-
-    // Next frame encoded should be an intra frame (keyframe).
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t IntraFrameRequest(int stream_index) = 0;
-
-    // Frame Dropper enable. Can be used to disable the frame dropping when the encoder
-    // over-uses its bit rate. This API is designed to be used when the encoded frames
-    // are supposed to be stored to an AVI file, or when the I420 codec is used and the
-    // target bit rate shouldn't affect the frame rate.
-    //
-    // Input:
-    //      - enable            : True to enable the setting, false to disable it.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t EnableFrameDropper(bool enable) = 0;
-
-    // Sent frame counters
-    virtual int32_t SentFrameCount(VCMFrameCount& frameCount) const = 0;
-
-    /*
-    *   Receiver
-    */
-
-    // The receiver state of the VCM will be initialized to the
-    // same state as when the VCM was created. This will not interrupt
-    // or effect the send side functionality of VCM. VCM will lose all the
-    // decoding-related settings by calling this function. All frames
-    // inside the jitter buffer are flushed and the delay is reset.
-    // For instance, a receive codec has to be registered again.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t InitializeReceiver() = 0;
-
-    // Register possible receive codecs, can be called multiple times for different codecs.
-    // The module will automatically switch between registered codecs depending on the
-    // payload type of incoming frames. The actual decoder will be created when needed.
-    //
-    // Input:
-    //      - receiveCodec      : Settings for the codec to be registered.
-    //      - numberOfCores     : Number of CPU cores that the decoder is allowed to use.
-    //      - requireKeyFrame   : Set this to true if you don't want any delta frames
-    //                            to be decoded until the first key frame has been decoded.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
-                                         int32_t numberOfCores,
-                                         bool requireKeyFrame = false) = 0;
-
-    // Register an externally defined decoder/renderer object. Can be a decoder only or a
-    // decoder coupled with a renderer. Note that RegisterReceiveCodec must be called to
-    // be used for decoding incoming streams.
-    //
-    // Input:
-    //      - externalDecoder        : The external decoder/renderer object.
-    //      - payloadType            : The payload type which this decoder should be
-    //                                 registered to.
-    //      - internalRenderTiming   : True if the internal renderer (if any) of the decoder
-    //                                 object can make sure to render at a given time in ms.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                            uint8_t payloadType,
-                                            bool internalRenderTiming) = 0;
-
-    // Register a receive callback. Will be called whenever there is a new frame ready
-    // for rendering.
-    //
-    // Input:
-    //      - receiveCallback        : The callback object to be used by the module when a
-    //                                 frame is ready for rendering.
-    //                                 De-register with a NULL pointer.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterReceiveCallback(VCMReceiveCallback* receiveCallback) = 0;
-
-    // Register a receive statistics callback which will be called to deliver information
-    // about the video stream received by the receiving side of the VCM, for instance the
-    // average frame rate and bit rate.
-    //
-    // Input:
-    //      - receiveStats  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterReceiveStatisticsCallback(
-                               VCMReceiveStatisticsCallback* receiveStats) = 0;
-
-    // Register a decoder timing callback which will be called to deliver
-    // information about the timing of the decoder in the receiving side of the
-    // VCM, for instance the current and maximum frame decode latency.
-    //
-    // Input:
-    //      - decoderTiming  : The callback object to register.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterDecoderTimingCallback(
-        VCMDecoderTimingCallback* decoderTiming) = 0;
-
-    // Register a frame type request callback. This callback will be called when the
-    // module needs to request specific frame types from the send side.
-    //
-    // Input:
-    //      - frameTypeCallback      : The callback object to be used by the module when
-    //                                 requesting a specific type of frame from the send side.
-    //                                 De-register with a NULL pointer.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterFrameTypeCallback(
-                                  VCMFrameTypeCallback* frameTypeCallback) = 0;
-
-    // Registers a callback which is called whenever the receive side of the VCM
-    // encounters holes in the packet sequence and needs packets to be retransmitted.
-    //
-    // Input:
-    //              - callback      : The callback to be registered in the VCM.
-    //
-    // Return value     : VCM_OK,     on success.
-    //                    <0,              on error.
-    virtual int32_t RegisterPacketRequestCallback(
-                                        VCMPacketRequestCallback* callback) = 0;
-
-    // Register a receive state change callback. This callback will be called when the
-    // module state has changed
-    //
-    // Input:
-    //      - callback      : The callback object to be used by the module when
-    //                        the receiver decode state changes.
-    //                        De-register with a NULL pointer.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t RegisterReceiveStateCallback(
-                                  VCMReceiveStateCallback* callback) = 0;
-
-    // Waits for the next frame in the jitter buffer to become complete
-    // (waits no longer than maxWaitTimeMs), then passes it to the decoder for decoding.
-    // Should be called as often as possible to get the most out of the decoder.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t Decode(uint16_t maxWaitTimeMs = 200) = 0;
-
-    // Registers a callback which conveys the size of the render buffer.
-    virtual int RegisterRenderBufferSizeCallback(
-        VCMRenderBufferSizeCallback* callback) = 0;
-
-    // Reset the decoder state to the initial state.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t ResetDecoder() = 0;
-
-    // API to get the codec which is currently used for decoding by the module.
-    //
-    // Input:
-    //      - currentReceiveCodec      : Settings for the codec to be registered.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
-
-    // API to get the codec type currently used for decoding by the module.
-    //
-    // Return value      : codecy type,            on success.
-    //                     kVideoCodecUnknown, on error or if no receive codec is registered
-    virtual VideoCodecType ReceiveCodec() const = 0;
-
-    // Insert a parsed packet into the receiver side of the module. Will be placed in the
-    // jitter buffer waiting for the frame to become complete. Returns as soon as the packet
-    // has been placed in the jitter buffer.
-    //
-    // Input:
-    //      - incomingPayload      : Payload of the packet.
-    //      - payloadLength        : Length of the payload.
-    //      - rtpInfo              : The parsed header.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
-                                   size_t payloadLength,
-                                   const WebRtcRTPHeader& rtpInfo) = 0;
-
-    // Minimum playout delay (Used for lip-sync). This is the minimum delay required
-    // to sync with audio. Not included in  VideoCodingModule::Delay()
-    // Defaults to 0 ms.
-    //
-    // Input:
-    //      - minPlayoutDelayMs   : Additional delay in ms.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) = 0;
-
-    // Set the time required by the renderer to render a frame.
-    //
-    // Input:
-    //      - timeMS        : The time in ms required by the renderer to render a frame.
-    //
-    // Return value      : VCM_OK, on success.
-    //                     < 0,         on error.
-    virtual int32_t SetRenderDelay(uint32_t timeMS) = 0;
-
-    // The total delay desired by the VCM. Can be less than the minimum
-    // delay set with SetMinimumPlayoutDelay.
-    //
-    // Return value      : Total delay in ms, on success.
-    //                     < 0,               on error.
-    virtual int32_t Delay() const = 0;
-
-    // Returns the number of packets discarded by the jitter buffer due to being
-    // too late. This can include duplicated packets which arrived after the
-    // frame was sent to the decoder. Therefore packets which were prematurely
-    // NACKed will be counted.
-    virtual uint32_t DiscardedPackets() const = 0;
-
-
-    // Robustness APIs
-
-    // Set the receiver robustness mode. The mode decides how the receiver
-    // responds to losses in the stream. The type of counter-measure (soft or
-    // hard NACK, dual decoder, RPS, etc.) is selected through the
-    // robustnessMode parameter. The errorMode parameter decides if it is
-    // allowed to display frames corrupted by losses. Note that not all
-    // combinations of the two parameters are feasible. An error will be
-    // returned for invalid combinations.
-    // Input:
-    //      - robustnessMode : selected robustness mode.
-    //      - errorMode      : selected error mode.
-    //
-    // Return value      : VCM_OK, on success;
-    //                     < 0, on error.
-    virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
-                                          VCMDecodeErrorMode errorMode) = 0;
-
-    // Set the decode error mode. The mode decides which errors (if any) are
-    // allowed in decodable frames. Note that setting decode_error_mode to
-    // anything other than kWithErrors without enabling nack will cause
-    // long-term freezes (resulting from frequent key frame requests) if
-    // packet loss occurs.
-    virtual void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) = 0;
-
-    // Sets the maximum number of sequence numbers that we are allowed to NACK
-    // and the oldest sequence number that we will consider to NACK. If a
-    // sequence number older than |max_packet_age_to_nack| is missing
-    // a key frame will be requested. A key frame will also be requested if the
-    // time of incomplete or non-continuous frames in the jitter buffer is above
-    // |max_incomplete_time_ms|.
-    virtual void SetNackSettings(size_t max_nack_list_size,
-                                 int max_packet_age_to_nack,
-                                 int max_incomplete_time_ms) = 0;
-
-    // Setting a desired delay to the VCM receiver. Video rendering will be
-    // delayed by at least desired_delay_ms.
-    virtual int SetMinReceiverDelay(int desired_delay_ms) = 0;
-
-    // Set current load state of the CPU
-    virtual void SetCPULoadState(CPULoadState state) = 0;
-
-    // Enables recording of debugging information.
-    virtual int StartDebugRecording(const char* file_name_utf8) = 0;
-
-    // Disables recording of debugging information.
-    virtual int StopDebugRecording() = 0;
-
-    // Lets the sender suspend video when the rate drops below
-    // |threshold_bps|, and turns back on when the rate goes back up above
-    // |threshold_bps| + |window_bps|.
-    virtual void SuspendBelowMinBitrate() = 0;
-
-    // Returns true if SuspendBelowMinBitrate is engaged and the video has been
-    // suspended due to bandwidth limitations; otherwise false.
-    virtual bool VideoSuspended() const = 0;
-
-    virtual void RegisterPreDecodeImageCallback(
-        EncodedImageCallback* observer) = 0;
-    virtual void RegisterPostEncodeImageCallback(
-        EncodedImageCallback* post_encode_callback) = 0;
-    // Releases pending decode calls, permitting faster thread shutdown.
-    virtual void TriggerDecoderShutdown() = 0;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/OWNERS b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_timer.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_timer.cc
deleted file mode 100644
index a462258813..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_timer.cc
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/codec_timer.h"
-
-#include <assert.h>
-
-namespace webrtc
-{
-
-// The first kIgnoredSampleCount samples will be ignored.
-static const int32_t kIgnoredSampleCount = 5;
-
-VCMCodecTimer::VCMCodecTimer()
-:
-_filteredMax(0),
-_ignoredSampleCount(0),
-_shortMax(0),
-_history()
-{
-    Reset();
-}
-
-int32_t VCMCodecTimer::StopTimer(int64_t startTimeMs, int64_t nowMs)
-{
-    const int32_t timeDiff = static_cast<int32_t>(nowMs - startTimeMs);
-    MaxFilter(timeDiff, nowMs);
-    return timeDiff;
-}
-
-void VCMCodecTimer::Reset()
-{
-    _filteredMax = 0;
-    _ignoredSampleCount = 0;
-    _shortMax = 0;
-    for (int i=0; i < MAX_HISTORY_SIZE; i++)
-    {
-        _history[i].shortMax = 0;
-        _history[i].timeMs = -1;
-    }
-}
-
-// Update the max-value filter
-void VCMCodecTimer::MaxFilter(int32_t decodeTime, int64_t nowMs)
-{
-    if (_ignoredSampleCount >= kIgnoredSampleCount)
-    {
-        UpdateMaxHistory(decodeTime, nowMs);
-        ProcessHistory(nowMs);
-    }
-    else
-    {
-        _ignoredSampleCount++;
-    }
-}
-
-void
-VCMCodecTimer::UpdateMaxHistory(int32_t decodeTime, int64_t now)
-{
-    if (_history[0].timeMs >= 0 &&
-        now - _history[0].timeMs < SHORT_FILTER_MS)
-    {
-        if (decodeTime > _shortMax)
-        {
-            _shortMax = decodeTime;
-        }
-    }
-    else
-    {
-        // Only add a new value to the history once a second
-        if(_history[0].timeMs == -1)
-        {
-            // First, no shift
-            _shortMax = decodeTime;
-        }
-        else
-        {
-            // Shift
-            for(int i = (MAX_HISTORY_SIZE - 2); i >= 0 ; i--)
-            {
-                _history[i+1].shortMax = _history[i].shortMax;
-                _history[i+1].timeMs = _history[i].timeMs;
-            }
-        }
-        if (_shortMax == 0)
-        {
-            _shortMax = decodeTime;
-        }
-
-        _history[0].shortMax = _shortMax;
-        _history[0].timeMs = now;
-        _shortMax = 0;
-    }
-}
-
-void
-VCMCodecTimer::ProcessHistory(int64_t nowMs)
-{
-    _filteredMax = _shortMax;
-    if (_history[0].timeMs == -1)
-    {
-        return;
-    }
-    for (int i=0; i < MAX_HISTORY_SIZE; i++)
-    {
-        if (_history[i].timeMs == -1)
-        {
-            break;
-        }
-        if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS)
-        {
-            // This sample (and all samples after this) is too old
-            break;
-        }
-        if (_history[i].shortMax > _filteredMax)
-        {
-            // This sample is the largest one this far into the history
-            _filteredMax = _history[i].shortMax;
-        }
-    }
-}
-
-// Get the maximum observed time within a time window
-int32_t VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const
-{
-    return _filteredMax;
-}
-
-}
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_timer.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_timer.h
deleted file mode 100644
index 9268e8d817..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/codec_timer.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
-#define MAX_HISTORY_SIZE 10
-#define SHORT_FILTER_MS 1000
-
-class VCMShortMaxSample
-{
-public:
-    VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
-
-    int32_t     shortMax;
-    int64_t     timeMs;
-};
-
-class VCMCodecTimer
-{
-public:
-    VCMCodecTimer();
-
-    // Updates and returns the max filtered decode time.
-    int32_t StopTimer(int64_t startTimeMs, int64_t nowMs);
-
-    // Empty the list of timers.
-    void Reset();
-
-    // Get the required decode time in ms.
-    int32_t RequiredDecodeTimeMs(FrameType frameType) const;
-
-private:
-    void UpdateMaxHistory(int32_t decodeTime, int64_t now);
-    void MaxFilter(int32_t newTime, int64_t nowMs);
-    void ProcessHistory(int64_t nowMs);
-
-    int32_t                     _filteredMax;
-    // The number of samples ignored so far.
-    int32_t                     _ignoredSampleCount;
-    int32_t                     _shortMax;
-    VCMShortMaxSample           _history[MAX_HISTORY_SIZE];
-
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.h
deleted file mode 100644
index d8589070d4..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/encoded_frame.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
-#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
-
-#include <vector>
-
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-
-namespace webrtc
-{
-
-class VCMEncodedFrame : protected EncodedImage
-{
-public:
-    VCMEncodedFrame();
-    VCMEncodedFrame(const webrtc::EncodedImage& rhs);
-    VCMEncodedFrame(const VCMEncodedFrame& rhs);
-
-    ~VCMEncodedFrame();
-    /**
-    *   Delete VideoFrame and resets members to zero
-    */
-    void Free();
-    /**
-    *   Set render time in milliseconds
-    */
-    void SetRenderTime(const int64_t renderTimeMs) {_renderTimeMs = renderTimeMs;}
-
-    /**
-    *   Set the encoded frame size
-    */
-    void SetEncodedSize(uint32_t width, uint32_t height)
-                       { _encodedWidth  = width; _encodedHeight = height; }
-    /**
-    *   Get the encoded image
-    */
-    const webrtc::EncodedImage& EncodedImage() const
-                       { return static_cast<const webrtc::EncodedImage&>(*this); }
-    /**
-    *   Get pointer to frame buffer
-    */
-    const uint8_t* Buffer() const {return _buffer;}
-    /**
-    *   Get frame length
-    */
-    size_t Length() const {return _length;}
-    /**
-    *   Get frame timestamp (90kHz)
-    */
-    uint32_t TimeStamp() const {return _timeStamp;}
-    /**
-    *   Get render time in milliseconds
-    */
-    int64_t RenderTimeMs() const {return _renderTimeMs;}
-    /**
-    *   Get frame type
-    */
-    webrtc::FrameType FrameType() const {return ConvertFrameType(_frameType);}
-    /**
-    *   Get frame rotation
-    */
-    VideoRotation rotation() const { return _rotation; }
-    /**
-    *   True if this frame is complete, false otherwise
-    */
-    bool Complete() const { return _completeFrame; }
-    /**
-    *   True if there's a frame missing before this frame
-    */
-    bool MissingFrame() const { return _missingFrame; }
-    /**
-    *   Payload type of the encoded payload
-    */
-    uint8_t PayloadType() const { return _payloadType; }
-    /**
-    *   Get codec specific info.
-    *   The returned pointer is only valid as long as the VCMEncodedFrame
-    *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
-    *   the object.
-    */
-    const CodecSpecificInfo* CodecSpecific() const {return &_codecSpecificInfo;}
-
-    const RTPFragmentationHeader* FragmentationHeader() const;
-
-    static webrtc::FrameType ConvertFrameType(VideoFrameType frameType);
-    static VideoFrameType ConvertFrameType(webrtc::FrameType frameType);
-    static void ConvertFrameTypes(
-        const std::vector<webrtc::FrameType>& frame_types,
-        std::vector<VideoFrameType>* video_frame_types);
-
-protected:
-    /**
-    * Verifies that current allocated buffer size is larger than or equal to the input size.
-    * If the current buffer size is smaller, a new allocation is made and the old buffer data
-    * is copied to the new buffer.
-    * Buffer size is updated to minimumSize.
-    */
-    void VerifyAndAllocate(const uint32_t minimumSize);
-
-    void Reset();
-
-    void CopyCodecSpecific(const RTPVideoHeader* header);
-
-    int64_t                 _renderTimeMs;
-    uint8_t                 _payloadType;
-    bool                          _missingFrame;
-    CodecSpecificInfo             _codecSpecificInfo;
-    webrtc::VideoCodecType        _codec;
-    RTPFragmentationHeader        _fragmentation;
-    VideoRotation                 _rotation;
-
-    // Video rotation is only set along with the last packet for each frame
-    // (same as marker bit). This |_rotation_set| is only for debugging purpose
-    // to ensure we don't set it twice for a frame.
-    bool                          _rotation_set;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/fec_tables_xor.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/fec_tables_xor.h
deleted file mode 100644
index 28c67b4565..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/fec_tables_xor.h
+++ /dev/null
@@ -1,6481 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
-
-// This is a private header for media_opt_util.cc.
-// It should not be included by other files.
-
-namespace webrtc {
-
-// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
-// Input is the packet loss and an effective rate (bits/frame).
-// Output is array kCodeRateXORTable[k], where k = rate_i*129 + loss_j;
-// loss_j = 0,1,..128, and rate_i varies over some range.
-static const int kSizeCodeRateXORTable = 6450;
-static const unsigned char kCodeRateXORTable[kSizeCodeRateXORTable] = {
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-11,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-39,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-51,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
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-
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.cc
deleted file mode 100644
index 20ac04a7a2..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_buffer.cc
+++ /dev/null
@@ -1,300 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-
-#include <assert.h>
-#include <string.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-namespace webrtc {
-
-VCMFrameBuffer::VCMFrameBuffer()
-  :
-    _state(kStateEmpty),
-    _nackCount(0),
-    _latestPacketTimeMs(-1) {
-}
-
-VCMFrameBuffer::~VCMFrameBuffer() {
-}
-
-VCMFrameBuffer::VCMFrameBuffer(const VCMFrameBuffer& rhs)
-:
-VCMEncodedFrame(rhs),
-_state(rhs._state),
-_sessionInfo(),
-_nackCount(rhs._nackCount),
-_latestPacketTimeMs(rhs._latestPacketTimeMs) {
-    _sessionInfo = rhs._sessionInfo;
-    _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
-}
-
-webrtc::FrameType
-VCMFrameBuffer::FrameType() const {
-    return _sessionInfo.FrameType();
-}
-
-int32_t
-VCMFrameBuffer::GetLowSeqNum() const {
-    return _sessionInfo.LowSequenceNumber();
-}
-
-int32_t
-VCMFrameBuffer::GetHighSeqNum() const {
-    return _sessionInfo.HighSequenceNumber();
-}
-
-int VCMFrameBuffer::PictureId() const {
-  return _sessionInfo.PictureId();
-}
-
-int VCMFrameBuffer::TemporalId() const {
-  return _sessionInfo.TemporalId();
-}
-
-bool VCMFrameBuffer::LayerSync() const {
-  return _sessionInfo.LayerSync();
-}
-
-int VCMFrameBuffer::Tl0PicId() const {
-  return _sessionInfo.Tl0PicId();
-}
-
-bool VCMFrameBuffer::NonReference() const {
-  return _sessionInfo.NonReference();
-}
-
-void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
-  _sessionInfo.SetGofInfo(gof_info, idx);
-  // TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
-  _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
-      gof_info.temporal_idx[idx];
-  _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
-      gof_info.temporal_up_switch[idx];
-}
-
-bool
-VCMFrameBuffer::IsSessionComplete() const {
-    return _sessionInfo.complete();
-}
-
-// Insert packet
-VCMFrameBufferEnum
-VCMFrameBuffer::InsertPacket(const VCMPacket& packet,
-                             int64_t timeInMs,
-                             VCMDecodeErrorMode decode_error_mode,
-                             const FrameData& frame_data) {
-    assert(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
-    if (packet.dataPtr != NULL) {
-        _payloadType = packet.payloadType;
-    }
-
-    if (kStateEmpty == _state) {
-        // First packet (empty and/or media) inserted into this frame.
-        // store some info and set some initial values.
-        _timeStamp = packet.timestamp;
-        // We only take the ntp timestamp of the first packet of a frame.
-        ntp_time_ms_ = packet.ntp_time_ms_;
-        _codec = packet.codec;
-        if (packet.frameType != kFrameEmpty) {
-            // first media packet
-            SetState(kStateIncomplete);
-        }
-    }
-
-    // add safety margin because STAP-A packets can cause it to expand by
-    // ~two bytes per NAL
-    uint32_t requiredSizeBytes = Length() + packet.sizeBytes +
-                   (packet.insertStartCode ? kH264StartCodeLengthBytes : 0) +
-                                 kBufferSafetyMargin;
-    if (requiredSizeBytes >= _size) {
-        const uint8_t* prevBuffer = _buffer;
-        const uint32_t increments = requiredSizeBytes /
-                                          kBufferIncStepSizeBytes +
-                                        (requiredSizeBytes %
-                                         kBufferIncStepSizeBytes > 0);
-        const uint32_t newSize = _size +
-                                       increments * kBufferIncStepSizeBytes;
-        if (newSize > kMaxJBFrameSizeBytes) {
-            LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
-                             "big.";
-            return kSizeError;
-        }
-        VerifyAndAllocate(newSize);
-        _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
-    }
-
-    if (packet.width > 0 && packet.height > 0) {
-      _encodedWidth = packet.width;
-      _encodedHeight = packet.height;
-    }
-
-    // Don't copy payload specific data for empty packets (e.g padding packets).
-    if (packet.sizeBytes > 0)
-      CopyCodecSpecific(&packet.codecSpecificHeader);
-
-    int retVal = _sessionInfo.InsertPacket(packet, _buffer,
-                                           decode_error_mode,
-                                           frame_data);
-    if (retVal == -1) {
-        return kSizeError;
-    } else if (retVal == -2) {
-        return kDuplicatePacket;
-    } else if (retVal == -3) {
-        return kOutOfBoundsPacket;
-    }
-    // update length
-    _length = Length() + static_cast<uint32_t>(retVal);
-
-    _latestPacketTimeMs = timeInMs;
-
-    // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
-    // ts_126114v120700p.pdf Section 7.4.5.
-    // The MTSI client shall add the payload bytes as defined in this clause
-    // onto the last RTP packet in each group of packets which make up a key
-    // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
-    // (HEVC)).
-    if (packet.markerBit) {
-      DCHECK(!_rotation_set);
-      _rotation = packet.codecSpecificHeader.rotation;
-      _rotation_set = true;
-    }
-
-    if (_sessionInfo.complete()) {
-      SetState(kStateComplete);
-      return kCompleteSession;
-    } else if (_sessionInfo.decodable()) {
-      SetState(kStateDecodable);
-      return kDecodableSession;
-    }
-    return kIncomplete;
-}
-
-int64_t
-VCMFrameBuffer::LatestPacketTimeMs() const {
-    return _latestPacketTimeMs;
-}
-
-void
-VCMFrameBuffer::IncrementNackCount() {
-    _nackCount++;
-}
-
-int16_t
-VCMFrameBuffer::GetNackCount() const {
-    return _nackCount;
-}
-
-bool
-VCMFrameBuffer::HaveFirstPacket() const {
-    return _sessionInfo.HaveFirstPacket();
-}
-
-bool
-VCMFrameBuffer::HaveLastPacket() const {
-    return _sessionInfo.HaveLastPacket();
-}
-
-int
-VCMFrameBuffer::NumPackets() const {
-    return _sessionInfo.NumPackets();
-}
-
-void
-VCMFrameBuffer::Reset() {
-    _length = 0;
-    _timeStamp = 0;
-    _sessionInfo.Reset();
-    _payloadType = 0;
-    _nackCount = 0;
-    _latestPacketTimeMs = -1;
-    _state = kStateEmpty;
-    VCMEncodedFrame::Reset();
-}
-
-// Set state of frame
-void
-VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
-    if (_state == state) {
-        return;
-    }
-    switch (state) {
-    case kStateIncomplete:
-        // we can go to this state from state kStateEmpty
-        assert(_state == kStateEmpty);
-
-        // Do nothing, we received a packet
-        break;
-
-    case kStateComplete:
-        assert(_state == kStateEmpty ||
-               _state == kStateIncomplete ||
-               _state == kStateDecodable);
-
-        break;
-
-    case kStateEmpty:
-        // Should only be set to empty through Reset().
-        assert(false);
-        break;
-
-    case kStateDecodable:
-        assert(_state == kStateEmpty ||
-               _state == kStateIncomplete);
-        break;
-    }
-    _state = state;
-}
-
-// Get current state of frame
-VCMFrameBufferStateEnum
-VCMFrameBuffer::GetState() const {
-    return _state;
-}
-
-// Get current state of frame
-VCMFrameBufferStateEnum
-VCMFrameBuffer::GetState(uint32_t& timeStamp) const {
-    timeStamp = TimeStamp();
-    return GetState();
-}
-
-bool
-VCMFrameBuffer::IsRetransmitted() const {
-    return _sessionInfo.session_nack();
-}
-
-void
-VCMFrameBuffer::PrepareForDecode(bool continuous) {
-#ifdef INDEPENDENT_PARTITIONS
-    if (_codec == kVideoCodecVP8) {
-        _length =
-            _sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
-                                                     &_fragmentation);
-    } else {
-        size_t bytes_removed = _sessionInfo.MakeDecodable();
-        _length -= bytes_removed;
-    }
-#else
-    size_t bytes_removed = _sessionInfo.MakeDecodable();
-    _length -= bytes_removed;
-#endif
-    // Transfer frame information to EncodedFrame and create any codec
-    // specific information.
-    _frameType = ConvertFrameType(_sessionInfo.FrameType());
-    _completeFrame = _sessionInfo.complete();
-    _missingFrame = !continuous;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_decoder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_decoder.cc
deleted file mode 100644
index 88bc75ae91..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_decoder.cc
+++ /dev/null
@@ -1,205 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-namespace webrtc {
-
-VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
-                                                 Clock* clock)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_clock(clock),
-_receiveCallback(NULL),
-_timing(timing),
-_timestampMap(kDecoderFrameMemoryLength),
-_lastReceivedPictureID(0)
-{
-}
-
-VCMDecodedFrameCallback::~VCMDecodedFrameCallback()
-{
-    delete _critSect;
-}
-
-void VCMDecodedFrameCallback::SetUserReceiveCallback(
-    VCMReceiveCallback* receiveCallback)
-{
-    CriticalSectionScoped cs(_critSect);
-    _receiveCallback = receiveCallback;
-}
-
-VCMReceiveCallback* VCMDecodedFrameCallback::UserReceiveCallback()
-{
-    CriticalSectionScoped cs(_critSect);
-    return _receiveCallback;
-}
-
-int32_t VCMDecodedFrameCallback::Decoded(I420VideoFrame& decodedImage)
-{
-    // TODO(holmer): We should improve this so that we can handle multiple
-    // callbacks from one call to Decode().
-    VCMFrameInformation* frameInfo;
-    VCMReceiveCallback* callback;
-    {
-        CriticalSectionScoped cs(_critSect);
-        frameInfo = static_cast<VCMFrameInformation*>(
-            _timestampMap.Pop(decodedImage.timestamp()));
-        callback = _receiveCallback;
-    }
-
-    if (frameInfo == NULL) {
-      LOG(LS_WARNING) << "Too many frames backed up in the decoder, dropping "
-                         "this one.";
-      return WEBRTC_VIDEO_CODEC_OK;
-    }
-
-    _timing.StopDecodeTimer(
-        decodedImage.timestamp(),
-        frameInfo->decodeStartTimeMs,
-        _clock->TimeInMilliseconds(),
-        frameInfo->renderTimeMs);
-
-    if (callback != NULL)
-    {
-        decodedImage.set_render_time_ms(frameInfo->renderTimeMs);
-        decodedImage.set_rotation(frameInfo->rotation);
-        callback->FrameToRender(decodedImage);
-    }
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t
-VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
-    const uint64_t pictureId)
-{
-    CriticalSectionScoped cs(_critSect);
-    if (_receiveCallback != NULL)
-    {
-        return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
-    }
-    return -1;
-}
-
-int32_t
-VCMDecodedFrameCallback::ReceivedDecodedFrame(const uint64_t pictureId)
-{
-    _lastReceivedPictureID = pictureId;
-    return 0;
-}
-
-uint64_t VCMDecodedFrameCallback::LastReceivedPictureID() const
-{
-    return _lastReceivedPictureID;
-}
-
-int32_t VCMDecodedFrameCallback::Map(uint32_t timestamp, VCMFrameInformation* frameInfo)
-{
-    CriticalSectionScoped cs(_critSect);
-    return _timestampMap.Add(timestamp, frameInfo);
-}
-
-int32_t VCMDecodedFrameCallback::Pop(uint32_t timestamp)
-{
-    CriticalSectionScoped cs(_critSect);
-    if (_timestampMap.Pop(timestamp) == NULL)
-    {
-        return VCM_GENERAL_ERROR;
-    }
-    return VCM_OK;
-}
-
-VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, bool isExternal)
-:
-_callback(NULL),
-_frameInfos(),
-_nextFrameInfoIdx(0),
-_decoder(decoder),
-_codecType(kVideoCodecUnknown),
-_isExternal(isExternal),
-_keyFrameDecoded(false)
-{
-}
-
-VCMGenericDecoder::~VCMGenericDecoder()
-{
-}
-
-int32_t VCMGenericDecoder::InitDecode(const VideoCodec* settings,
-                                      int32_t numberOfCores)
-{
-    _codecType = settings->codecType;
-
-    return _decoder.InitDecode(settings, numberOfCores);
-}
-
-int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
-                                        int64_t nowMs)
-{
-    _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
-    _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
-    _frameInfos[_nextFrameInfoIdx].rotation = frame.rotation();
-    _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
-
-    _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
-    int32_t ret = _decoder.Decode(frame.EncodedImage(),
-                                        frame.MissingFrame(),
-                                        frame.FragmentationHeader(),
-                                        frame.CodecSpecific(),
-                                        frame.RenderTimeMs());
-
-    if (ret < WEBRTC_VIDEO_CODEC_OK)
-    {
-        LOG(LS_WARNING) << "Failed to decode frame with timestamp "
-                        << frame.TimeStamp() << ", error code: " << ret;
-        _callback->Pop(frame.TimeStamp());
-        return ret;
-    }
-    else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
-             ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
-    {
-        // No output
-        _callback->Pop(frame.TimeStamp());
-    }
-    return ret;
-}
-
-int32_t
-VCMGenericDecoder::Release()
-{
-    return _decoder.Release();
-}
-
-int32_t VCMGenericDecoder::Reset()
-{
-    return _decoder.Reset();
-}
-
-int32_t VCMGenericDecoder::SetCodecConfigParameters(const uint8_t* buffer, int32_t size)
-{
-    return _decoder.SetCodecConfigParameters(buffer, size);
-}
-
-int32_t VCMGenericDecoder::RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback)
-{
-    _callback = callback;
-    return _decoder.RegisterDecodeCompleteCallback(callback);
-}
-
-bool VCMGenericDecoder::External() const
-{
-    return _isExternal;
-}
-
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_decoder.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_decoder.h
deleted file mode 100644
index a19b67084c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_decoder.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/timestamp_map.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-
-namespace webrtc
-{
-
-class VCMReceiveCallback;
-
-enum { kDecoderFrameMemoryLength = 30 };
-
-struct VCMFrameInformation
-{
-    int64_t     renderTimeMs;
-    int64_t     decodeStartTimeMs;
-    void*             userData;
-    VideoRotation rotation;
-};
-
-class VCMDecodedFrameCallback : public DecodedImageCallback
-{
-public:
-    VCMDecodedFrameCallback(VCMTiming& timing, Clock* clock);
-    virtual ~VCMDecodedFrameCallback();
-    void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
-    VCMReceiveCallback* UserReceiveCallback();
-
-    virtual int32_t Decoded(I420VideoFrame& decodedImage);
-    virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
-    virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId);
-
-    uint64_t LastReceivedPictureID() const;
-
-    int32_t Map(uint32_t timestamp, VCMFrameInformation* frameInfo);
-    int32_t Pop(uint32_t timestamp);
-
-private:
-    // Protect |_receiveCallback| and |_timestampMap|.
-    CriticalSectionWrapper* _critSect;
-    Clock* _clock;
-    VCMReceiveCallback* _receiveCallback;  // Guarded by |_critSect|.
-    VCMTiming& _timing;
-    VCMTimestampMap _timestampMap;  // Guarded by |_critSect|.
-    uint64_t _lastReceivedPictureID;
-};
-
-
-class VCMGenericDecoder
-{
-    friend class VCMCodecDataBase;
-public:
-    VCMGenericDecoder(VideoDecoder& decoder, bool isExternal = false);
-    ~VCMGenericDecoder();
-
-    /**
-    *	Initialize the decoder with the information from the VideoCodec
-    */
-    int32_t InitDecode(const VideoCodec* settings,
-                             int32_t numberOfCores);
-
-    /**
-    *	Decode to a raw I420 frame,
-    *
-    *	inputVideoBuffer	reference to encoded video frame
-    */
-    int32_t Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
-
-    /**
-    *	Free the decoder memory
-    */
-    int32_t Release();
-
-    /**
-    *	Reset the decoder state, prepare for a new call
-    */
-    int32_t Reset();
-
-    /**
-    *	Codec configuration data sent out-of-band, i.e. in SIP call setup
-    *
-    *	buffer pointer to the configuration data
-    *	size the size of the configuration data in bytes
-    */
-    int32_t SetCodecConfigParameters(const uint8_t* /*buffer*/,
-                                           int32_t /*size*/);
-
-    /**
-    * Set decode callback. Deregistering while decoding is illegal.
-    */
-    int32_t RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
-
-    bool External() const;
-
-private:
-    VCMDecodedFrameCallback*    _callback;
-    VCMFrameInformation         _frameInfos[kDecoderFrameMemoryLength];
-    uint32_t                    _nextFrameInfoIdx;
-    VideoDecoder&               _decoder;
-    VideoCodecType              _codecType;
-    bool                        _isExternal;
-    bool                        _keyFrameDecoded;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_encoder.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_encoder.cc
deleted file mode 100644
index 47ff582c5f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_encoder.cc
+++ /dev/null
@@ -1,334 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/checks.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-namespace webrtc {
-namespace {
-// Map information from info into rtp. If no relevant information is found
-// in info, rtp is set to NULL.
-void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
-  DCHECK(info);
-  switch (info->codecType) {
-    case kVideoCodecVP8: {
-      rtp->codec = kRtpVideoVp8;
-      rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
-      rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
-      rtp->codecHeader.VP8.nonReference =
-          info->codecSpecific.VP8.nonReference;
-      rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
-      rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
-      rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx;
-      rtp->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx;
-      rtp->simulcastIdx = info->codecSpecific.VP8.simulcastIdx;
-      return;
-    }
-    case kVideoCodecVP9: {
-      rtp->codec = kRtpVideoVp9;
-      rtp->codecHeader.VP9.InitRTPVideoHeaderVP9();
-      rtp->codecHeader.VP9.inter_pic_predicted =
-          info->codecSpecific.VP9.inter_pic_predicted;
-      rtp->codecHeader.VP9.flexible_mode =
-          info->codecSpecific.VP9.flexible_mode;
-      rtp->codecHeader.VP9.ss_data_available =
-          info->codecSpecific.VP9.ss_data_available;
-      rtp->codecHeader.VP9.picture_id = info->codecSpecific.VP9.picture_id;
-      rtp->codecHeader.VP9.tl0_pic_idx = info->codecSpecific.VP9.tl0_pic_idx;
-      rtp->codecHeader.VP9.temporal_idx = info->codecSpecific.VP9.temporal_idx;
-      rtp->codecHeader.VP9.spatial_idx = info->codecSpecific.VP9.spatial_idx;
-      rtp->codecHeader.VP9.temporal_up_switch =
-          info->codecSpecific.VP9.temporal_up_switch;
-      rtp->codecHeader.VP9.inter_layer_predicted =
-          info->codecSpecific.VP9.inter_layer_predicted;
-      rtp->codecHeader.VP9.gof_idx = info->codecSpecific.VP9.gof_idx;
-      rtp->codecHeader.VP9.num_spatial_layers =
-          info->codecSpecific.VP9.num_spatial_layers;
-
-      if (info->codecSpecific.VP9.ss_data_available) {
-        rtp->codecHeader.VP9.spatial_layer_resolution_present =
-            info->codecSpecific.VP9.spatial_layer_resolution_present;
-        if (info->codecSpecific.VP9.spatial_layer_resolution_present) {
-          for (size_t i = 0; i < info->codecSpecific.VP9.num_spatial_layers;
-               ++i) {
-            rtp->codecHeader.VP9.width[i] = info->codecSpecific.VP9.width[i];
-            rtp->codecHeader.VP9.height[i] = info->codecSpecific.VP9.height[i];
-          }
-        }
-        rtp->codecHeader.VP9.gof.CopyGofInfoVP9(info->codecSpecific.VP9.gof);
-      }
-
-      rtp->codecHeader.VP9.num_ref_pics = info->codecSpecific.VP9.num_ref_pics;
-      for (int i = 0; i < info->codecSpecific.VP9.num_ref_pics; ++i)
-        rtp->codecHeader.VP9.pid_diff[i] = info->codecSpecific.VP9.p_diff[i];
-      return;
-    }
-    case kVideoCodecH264:
-      rtp->codec = kRtpVideoH264;
-      rtp->codecHeader.H264.packetization_mode = info->codecSpecific.H264.packetizationMode;
-      rtp->codecHeader.H264.single_nalu = info->codecSpecific.H264.single_nalu;
-      rtp->simulcastIdx = info->codecSpecific.H264.simulcastIdx;
-      return;
-    case kVideoCodecGeneric:
-      rtp->codec = kRtpVideoGeneric;
-      rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
-      return;
-    default:
-      return;
-  }
-}
-}  // namespace
-
-//#define DEBUG_ENCODER_BIT_STREAM
-
-VCMGenericEncoder::VCMGenericEncoder(VideoEncoder* encoder,
-                                     VideoEncoderRateObserver* rate_observer,
-                                     bool internalSource)
-    : encoder_(encoder),
-      rate_observer_(rate_observer),
-      vcm_encoded_frame_callback_(nullptr),
-      bit_rate_(0),
-      frame_rate_(0),
-      internal_source_(internalSource),
-      rotation_(kVideoRotation_0) {
-}
-
-VCMGenericEncoder::~VCMGenericEncoder()
-{
-}
-
-int32_t VCMGenericEncoder::Release()
-{
-    {
-      rtc::CritScope lock(&rates_lock_);
-      bit_rate_ = 0;
-      frame_rate_ = 0;
-      encoder_->RegisterEncodeCompleteCallback(nullptr);
-      vcm_encoded_frame_callback_ = nullptr;
-    }
-
-    return encoder_->Release();
-}
-
-int32_t
-VCMGenericEncoder::InitEncode(const VideoCodec* settings,
-                              int32_t numberOfCores,
-                              size_t maxPayloadSize)
-{
-    {
-      rtc::CritScope lock(&rates_lock_);
-      bit_rate_ = settings->startBitrate * 1000;
-      frame_rate_ = settings->maxFramerate;
-    }
-
-    if (encoder_->InitEncode(settings, numberOfCores, maxPayloadSize) != 0) {
-      LOG(LS_ERROR) << "Failed to initialize the encoder associated with "
-                       "payload name: " << settings->plName;
-      return -1;
-    }
-    return 0;
-}
-
-int32_t
-VCMGenericEncoder::Encode(const I420VideoFrame& inputFrame,
-                          const CodecSpecificInfo* codecSpecificInfo,
-                          const std::vector<FrameType>& frameTypes) {
-  std::vector<VideoFrameType> video_frame_types(frameTypes.size(),
-                                                kDeltaFrame);
-  VCMEncodedFrame::ConvertFrameTypes(frameTypes, &video_frame_types);
-
-  rotation_ = inputFrame.rotation();
-
-  if (vcm_encoded_frame_callback_) {
-    // Keep track of the current frame rotation and apply to the output of the
-    // encoder. There might not be exact as the encoder could have one frame
-    // delay but it should be close enough.
-    vcm_encoded_frame_callback_->SetRotation(rotation_);
-  }
-
-  return encoder_->Encode(inputFrame, codecSpecificInfo, &video_frame_types);
-}
-
-int32_t
-VCMGenericEncoder::SetChannelParameters(int32_t packetLoss, int64_t rtt)
-{
-    return encoder_->SetChannelParameters(packetLoss, rtt);
-}
-
-int32_t
-VCMGenericEncoder::SetRates(uint32_t newBitRate, uint32_t frameRate)
-{
-    uint32_t target_bitrate_kbps = (newBitRate + 500) / 1000;
-    int32_t ret = encoder_->SetRates(target_bitrate_kbps, frameRate);
-    if (ret < 0)
-    {
-        return ret;
-    }
-
-    {
-      rtc::CritScope lock(&rates_lock_);
-      bit_rate_ = newBitRate;
-      frame_rate_ = frameRate;
-    }
-
-    if (rate_observer_ != nullptr)
-      rate_observer_->OnSetRates(newBitRate, frameRate);
-    return VCM_OK;
-}
-
-int32_t
-VCMGenericEncoder::CodecConfigParameters(uint8_t* buffer, int32_t size)
-{
-    int32_t ret = encoder_->CodecConfigParameters(buffer, size);
-    if (ret < 0)
-    {
-        return ret;
-    }
-    return ret;
-}
-
-uint32_t VCMGenericEncoder::BitRate() const
-{
-    rtc::CritScope lock(&rates_lock_);
-    return bit_rate_;
-}
-
-uint32_t VCMGenericEncoder::FrameRate() const
-{
-    rtc::CritScope lock(&rates_lock_);
-    return frame_rate_;
-}
-
-int32_t
-VCMGenericEncoder::SetPeriodicKeyFrames(bool enable)
-{
-    return encoder_->SetPeriodicKeyFrames(enable);
-}
-
-int32_t VCMGenericEncoder::RequestFrame(
-    const std::vector<FrameType>& frame_types) {
-  I420VideoFrame image;
-  std::vector<VideoFrameType> video_frame_types(frame_types.size(),
-                                                kDeltaFrame);
-  VCMEncodedFrame::ConvertFrameTypes(frame_types, &video_frame_types);
-  return encoder_->Encode(image, NULL, &video_frame_types);
-}
-
-int32_t
-VCMGenericEncoder::RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback)
-{
-    VCMencodedFrameCallback->SetInternalSource(internal_source_);
-    vcm_encoded_frame_callback_ = VCMencodedFrameCallback;
-    return encoder_->RegisterEncodeCompleteCallback(VCMencodedFrameCallback);
-}
-
-bool
-VCMGenericEncoder::InternalSource() const
-{
-    return internal_source_;
-}
-
- /***************************
-  * Callback Implementation
-  ***************************/
-VCMEncodedFrameCallback::VCMEncodedFrameCallback(
-    EncodedImageCallback* post_encode_callback)
-    : _sendCallback(),
-      _critSect(NULL),
-      _mediaOpt(NULL),
-      _payloadType(0),
-      _internalSource(false),
-      _rotation(kVideoRotation_0),
-      post_encode_callback_(post_encode_callback)
-#ifdef DEBUG_ENCODER_BIT_STREAM
-      ,
-      _bitStreamAfterEncoder(NULL)
-#endif
-{
-#ifdef DEBUG_ENCODER_BIT_STREAM
-    _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
-#endif
-}
-
-VCMEncodedFrameCallback::~VCMEncodedFrameCallback()
-{
-#ifdef DEBUG_ENCODER_BIT_STREAM
-    fclose(_bitStreamAfterEncoder);
-#endif
-}
-
-void
-VCMEncodedFrameCallback::SetCritSect(CriticalSectionWrapper* critSect)
-{
-    _critSect = critSect;
-}
-
-int32_t
-VCMEncodedFrameCallback::SetTransportCallback(VCMPacketizationCallback* transport)
-{
-    _sendCallback = transport;
-    return VCM_OK;
-}
-
-int32_t VCMEncodedFrameCallback::Encoded(
-    const EncodedImage& encodedImage,
-    const CodecSpecificInfo* codecSpecificInfo,
-    const RTPFragmentationHeader* fragmentationHeader) {
-  assert(_critSect);
-  CriticalSectionScoped cs(_critSect);
-
-  post_encode_callback_->Encoded(encodedImage, NULL, NULL);
-
-  if (_sendCallback == NULL) {
-    return VCM_UNINITIALIZED;
-  }
-
-#ifdef DEBUG_ENCODER_BIT_STREAM
-  if (_bitStreamAfterEncoder != NULL) {
-    fwrite(encodedImage._buffer, 1, encodedImage._length,
-           _bitStreamAfterEncoder);
-  }
-#endif
-
-  RTPVideoHeader rtpVideoHeader;
-  memset(&rtpVideoHeader, 0, sizeof(RTPVideoHeader));
-  RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader;
-  if (codecSpecificInfo) {
-    CopyCodecSpecific(codecSpecificInfo, rtpVideoHeaderPtr);
-  }
-  rtpVideoHeader.rotation = _rotation;
-
-  int32_t callbackReturn = _sendCallback->SendData(
-      _payloadType, encodedImage, *fragmentationHeader, rtpVideoHeaderPtr);
-  if (callbackReturn < 0) {
-    return callbackReturn;
-  }
-
-  if (_mediaOpt != NULL) {
-    _mediaOpt->UpdateWithEncodedData(encodedImage);
-    if (_internalSource)
-      return _mediaOpt->DropFrame();  // Signal to encoder to drop next frame.
-  }
-  return VCM_OK;
-}
-
-void
-VCMEncodedFrameCallback::SetMediaOpt(
-    media_optimization::MediaOptimization *mediaOpt)
-{
-    _mediaOpt = mediaOpt;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_encoder.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_encoder.h
deleted file mode 100644
index db41f7c454..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/generic_encoder.h
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
-
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-
-#include <stdio.h>
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-namespace media_optimization {
-class MediaOptimization;
-}  // namespace media_optimization
-
-/*************************************/
-/* VCMEncodeFrameCallback class     */
-/***********************************/
-class VCMEncodedFrameCallback : public EncodedImageCallback
-{
-public:
-    VCMEncodedFrameCallback(EncodedImageCallback* post_encode_callback);
-    virtual ~VCMEncodedFrameCallback();
-
-    void SetCritSect(CriticalSectionWrapper* critSect);
-
-    /*
-    * Callback implementation - codec encode complete
-    */
-    int32_t Encoded(
-        const EncodedImage& encodedImage,
-        const CodecSpecificInfo* codecSpecificInfo = NULL,
-        const RTPFragmentationHeader* fragmentationHeader = NULL);
-    /*
-    * Callback implementation - generic encoder encode complete
-    */
-    int32_t SetTransportCallback(VCMPacketizationCallback* transport);
-    /**
-    * Set media Optimization
-    */
-    void SetMediaOpt (media_optimization::MediaOptimization* mediaOpt);
-
-    void SetPayloadType(uint8_t payloadType) { _payloadType = payloadType; };
-    void SetInternalSource(bool internalSource) { _internalSource = internalSource; };
-
-    void SetRotation(VideoRotation rotation) { _rotation = rotation; }
-
-private:
-    VCMPacketizationCallback* _sendCallback;
-    CriticalSectionWrapper* _critSect;
-    media_optimization::MediaOptimization* _mediaOpt;
-    uint8_t _payloadType;
-    bool _internalSource;
-    VideoRotation _rotation;
-
-    EncodedImageCallback* post_encode_callback_;
-
-#ifdef DEBUG_ENCODER_BIT_STREAM
-    FILE* _bitStreamAfterEncoder;
-#endif
-};// end of VCMEncodeFrameCallback class
-
-
-/******************************/
-/* VCMGenericEncoder class    */
-/******************************/
-class VCMGenericEncoder
-{
-    friend class VCMCodecDataBase;
-public:
-    VCMGenericEncoder(VideoEncoder* encoder,
-                      VideoEncoderRateObserver* rate_observer,
-                      bool internalSource);
-    ~VCMGenericEncoder();
-    /**
-    * Free encoder memory
-    */
-    int32_t Release();
-    /**
-    * Initialize the encoder with the information from the VideoCodec
-    */
-    int32_t InitEncode(const VideoCodec* settings,
-                       int32_t numberOfCores,
-                       size_t maxPayloadSize);
-    /**
-    * Encode raw image
-    * inputFrame        : Frame containing raw image
-    * codecSpecificInfo : Specific codec data
-    * cameraFrameRate   : Request or information from the remote side
-    * frameType         : The requested frame type to encode
-    */
-    int32_t Encode(const I420VideoFrame& inputFrame,
-                   const CodecSpecificInfo* codecSpecificInfo,
-                   const std::vector<FrameType>& frameTypes);
-    /**
-    * Set new target bitrate (bits/s) and framerate.
-    * Return Value: new bit rate if OK, otherwise <0s.
-    */
-    // TODO(tommi): We could replace BitRate and FrameRate below with a GetRates
-    // method that matches SetRates. For fetching current rates, we'd then only
-    // grab the lock once instead of twice.
-    int32_t SetRates(uint32_t target_bitrate, uint32_t frameRate);
-    /**
-    * Set a new packet loss rate and a new round-trip time in milliseconds.
-    */
-    int32_t SetChannelParameters(int32_t packetLoss, int64_t rtt);
-    int32_t CodecConfigParameters(uint8_t* buffer, int32_t size);
-    /**
-    * Register a transport callback which will be called to deliver the encoded
-    * buffers
-    */
-    int32_t RegisterEncodeCallback(
-        VCMEncodedFrameCallback* VCMencodedFrameCallback);
-    /**
-    * Get encoder bit rate
-    */
-    uint32_t BitRate() const;
-     /**
-    * Get encoder frame rate
-    */
-    uint32_t FrameRate() const;
-
-    int32_t SetPeriodicKeyFrames(bool enable);
-
-    int32_t RequestFrame(const std::vector<FrameType>& frame_types);
-
-    bool InternalSource() const;
-
-private:
-    VideoEncoder* const encoder_;
-    VideoEncoderRateObserver* const rate_observer_;
-    VCMEncodedFrameCallback*  vcm_encoded_frame_callback_;
-    uint32_t bit_rate_;
-    uint32_t frame_rate_;
-    const bool internal_source_;
-    mutable rtc::CriticalSection rates_lock_;
-    VideoRotation rotation_;
-}; // end of VCMGenericEncoder class
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/inter_frame_delay.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/inter_frame_delay.cc
deleted file mode 100644
index 4786917e16..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/inter_frame_delay.cc
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
-
-namespace webrtc {
-
-VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock)
-{
-    Reset(currentWallClock);
-}
-
-// Resets the delay estimate
-void
-VCMInterFrameDelay::Reset(int64_t currentWallClock)
-{
-    _zeroWallClock = currentWallClock;
-    _wrapArounds = 0;
-    _prevWallClock = 0;
-    _prevTimestamp = 0;
-    _dTS = 0;
-}
-
-// Calculates the delay of a frame with the given timestamp.
-// This method is called when the frame is complete.
-bool
-VCMInterFrameDelay::CalculateDelay(uint32_t timestamp,
-                                int64_t *delay,
-                                int64_t currentWallClock)
-{
-    if (_prevWallClock == 0)
-    {
-        // First set of data, initialization, wait for next frame
-        _prevWallClock = currentWallClock;
-        _prevTimestamp = timestamp;
-        *delay = 0;
-        return true;
-    }
-
-    int32_t prevWrapArounds = _wrapArounds;
-    CheckForWrapArounds(timestamp);
-
-    // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
-    int32_t wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
-
-    // Account for reordering in jitter variance estimate in the future?
-    // Note that this also captures incomplete frames which are grabbed
-    // for decoding after a later frame has been complete, i.e. real
-    // packet losses.
-    if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) || wrapAroundsSincePrev < 0)
-    {
-        *delay = 0;
-        return false;
-    }
-
-    // Compute the compensated timestamp difference and convert it to ms and
-    // round it to closest integer.
-    _dTS = static_cast<int64_t>((timestamp + wrapAroundsSincePrev *
-                (static_cast<int64_t>(1)<<32) - _prevTimestamp) / 90.0 + 0.5);
-
-    // frameDelay is the difference of dT and dTS -- i.e. the difference of
-    // the wall clock time difference and the timestamp difference between
-    // two following frames.
-    *delay = static_cast<int64_t>(currentWallClock - _prevWallClock - _dTS);
-
-    _prevTimestamp = timestamp;
-    _prevWallClock = currentWallClock;
-
-    return true;
-}
-
-// Returns the current difference between incoming timestamps
-uint32_t VCMInterFrameDelay::CurrentTimeStampDiffMs() const
-{
-    if (_dTS < 0)
-    {
-        return 0;
-    }
-    return static_cast<uint32_t>(_dTS);
-}
-
-// Investigates if the timestamp clock has overflowed since the last timestamp and
-// keeps track of the number of wrap arounds since reset.
-void
-VCMInterFrameDelay::CheckForWrapArounds(uint32_t timestamp)
-{
-    if (timestamp < _prevTimestamp)
-    {
-        // This difference will probably be less than -2^31 if we have had a wrap around
-        // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to a Word32,
-        // it should be positive.
-        if (static_cast<int32_t>(timestamp - _prevTimestamp) > 0)
-        {
-            // Forward wrap around
-            _wrapArounds++;
-        }
-    }
-    // This difference will probably be less than -2^31 if we have had a backward wrap around.
-    // Since it is cast to a Word32, it should be positive.
-    else if (static_cast<int32_t>(_prevTimestamp - timestamp) > 0)
-    {
-        // Backward wrap around
-        _wrapArounds--;
-    }
-}
-
-}
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/inter_frame_delay.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/inter_frame_delay.h
deleted file mode 100644
index 58b326ae96..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/inter_frame_delay.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
-#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class VCMInterFrameDelay
-{
-public:
-    VCMInterFrameDelay(int64_t currentWallClock);
-
-    // Resets the estimate. Zeros are given as parameters.
-    void Reset(int64_t currentWallClock);
-
-    // Calculates the delay of a frame with the given timestamp.
-    // This method is called when the frame is complete.
-    //
-    // Input:
-    //          - timestamp         : RTP timestamp of a received frame
-    //          - *delay            : Pointer to memory where the result should be stored
-    //          - currentWallClock  : The current time in milliseconds.
-    //                                Should be -1 for normal operation, only used for testing.
-    // Return value                 : true if OK, false when reordered timestamps
-    bool CalculateDelay(uint32_t timestamp,
-                        int64_t *delay,
-                        int64_t currentWallClock);
-
-    // Returns the current difference between incoming timestamps
-    //
-    // Return value                 : Wrap-around compensated difference between incoming
-    //                                timestamps.
-    uint32_t CurrentTimeStampDiffMs() const;
-
-private:
-    // Controls if the RTP timestamp counter has had a wrap around
-    // between the current and the previously received frame.
-    //
-    // Input:
-    //          - timestmap         : RTP timestamp of the current frame.
-    void CheckForWrapArounds(uint32_t timestamp);
-
-    int64_t         _zeroWallClock; // Local timestamp of the first video packet received
-    int32_t         _wrapArounds;   // Number of wrapArounds detected
-    // The previous timestamp passed to the delay estimate
-    uint32_t        _prevTimestamp;
-    // The previous wall clock timestamp used by the delay estimate
-    int64_t         _prevWallClock;
-    // Wrap-around compensated difference between incoming timestamps
-    int64_t         _dTS;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/internal_defines.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/internal_defines.h
deleted file mode 100644
index adc940f20d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/internal_defines.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
-
-inline uint32_t MaskWord64ToUWord32(int64_t w64)
-{
-    return static_cast<uint32_t>(MASK_32_BITS(w64));
-}
-
-#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
-#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
-
-#define VCM_DEFAULT_CODEC_WIDTH 352
-#define VCM_DEFAULT_CODEC_HEIGHT 288
-#define VCM_DEFAULT_FRAME_RATE 30
-#define VCM_MIN_BITRATE 30
-#define VCM_FLUSH_INDICATOR 4
-
-// Helper macros for creating the static codec list
-#define VCM_NO_CODEC_IDX -1
-#ifdef VIDEOCODEC_VP8
-  #define VCM_VP8_IDX (VCM_NO_CODEC_IDX + 1)
-#else
-  #define VCM_VP8_IDX VCM_NO_CODEC_IDX
-#endif
-#ifdef VIDEOCODEC_VP9
-  #define VCM_VP9_IDX (VCM_VP8_IDX + 1)
-#else
-  #define VCM_VP9_IDX VCM_VP8_IDX
-#endif
-#ifdef VIDEOCODEC_H264
-  #define VCM_H264_IDX (VCM_VP9_IDX + 1)
-#else
-  #define VCM_H264_IDX VCM_VP9_IDX
-#endif
-#ifdef VIDEOCODEC_I420
-  #define VCM_I420_IDX (VCM_H264_IDX + 1)
-#else
-  #define VCM_I420_IDX VCM_H264_IDX
-#endif
-#define VCM_NUM_VIDEO_CODECS_AVAILABLE (VCM_I420_IDX + 1)
-
-#define VCM_NO_RECEIVER_ID 0
-
-inline int32_t VCMId(const int32_t vcmId, const int32_t receiverId = 0)
-{
-    return static_cast<int32_t>((vcmId << 16) + receiverId);
-}
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer_common.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer_common.h
deleted file mode 100644
index 049e66c961..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_buffer_common.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
-#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Used to estimate rolling average of packets per frame.
-static const float kFastConvergeMultiplier = 0.4f;
-static const float kNormalConvergeMultiplier = 0.2f;
-
-enum { kMaxNumberOfFrames     = 300 };
-enum { kStartNumberOfFrames   = 6 };
-enum { kMaxVideoDelayMs       = 10000 };
-enum { kPacketsPerFrameMultiplier = 5 };
-enum { kFastConvergeThreshold = 5};
-
-enum VCMJitterBufferEnum {
-  kMaxConsecutiveOldFrames        = 60,
-  kMaxConsecutiveOldPackets       = 300,
-  kMaxPacketsInSession            = 800,
-  kBufferIncStepSizeBytes         = 30000,   // >20 packets.
-  kMaxJBFrameSizeBytes            = 4000000, // sanity don't go above 4Mbyte.
-  kBufferSafetyMargin             = 100      // enough for ~50 NALs in a STAP-A
-};
-
-enum VCMFrameBufferEnum {
-  kOutOfBoundsPacket    = -7,
-  kNotInitialized       = -6,
-  kOldPacket            = -5,
-  kGeneralError         = -4,
-  kFlushIndicator       = -3,   // Indicator that a flush has occurred.
-  kTimeStampError       = -2,
-  kSizeError            = -1,
-  kNoError              = 0,
-  kIncomplete           = 1,    // Frame incomplete.
-  kCompleteSession      = 3,    // at least one layer in the frame complete.
-  kDecodableSession     = 4,    // Frame incomplete, but ready to be decoded
-  kDuplicatePacket      = 5     // We're receiving a duplicate packet.
-};
-
-enum VCMFrameBufferStateEnum {
-  kStateEmpty,              // frame popped by the RTP receiver
-  kStateIncomplete,         // frame that have one or more packet(s) stored
-  kStateComplete,           // frame that have all packets
-  kStateDecodable           // Hybrid mode - frame can be decoded
-};
-
-enum { kH264StartCodeLengthBytes = 4};
-
-// Used to indicate if a received packet contain a complete NALU (or equivalent)
-enum VCMNaluCompleteness {
-  kNaluUnset = 0,       // Packet has not been filled.
-  kNaluComplete = 1,    // Packet can be decoded as is.
-  kNaluStart,           // Packet contain beginning of NALU
-  kNaluIncomplete,      // Packet is not beginning or end of NALU
-  kNaluEnd,             // Packet is the end of a NALU
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator.cc
deleted file mode 100644
index d0faf31d1c..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator.cc
+++ /dev/null
@@ -1,484 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
-#include "webrtc/modules/video_coding/main/source/rtt_filter.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/field_trial.h"
-
-#include <assert.h>
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-
-namespace webrtc {
-
-enum { kStartupDelaySamples = 30 };
-enum { kFsAccuStartupSamples = 5 };
-enum { kMaxFramerateEstimate = 200 };
-
-VCMJitterEstimator::VCMJitterEstimator(const Clock* clock,
-                                       int32_t vcmId,
-                                       int32_t receiverId)
-    : _vcmId(vcmId),
-      _receiverId(receiverId),
-      _phi(0.97),
-      _psi(0.9999),
-      _alphaCountMax(400),
-      _thetaLow(0.000001),
-      _nackLimit(3),
-      _numStdDevDelayOutlier(15),
-      _numStdDevFrameSizeOutlier(3),
-      _noiseStdDevs(2.33),       // ~Less than 1% chance
-                                 // (look up in normal distribution table)...
-      _noiseStdDevOffset(30.0),  // ...of getting 30 ms freezes
-      _rttFilter(),
-      fps_counter_(30),  // TODO(sprang): Use an estimator with limit based on
-                         // time, rather than number of samples.
-      low_rate_experiment_(kInit),
-      clock_(clock) {
-  Reset();
-}
-
-VCMJitterEstimator::~VCMJitterEstimator() {
-}
-
-VCMJitterEstimator&
-VCMJitterEstimator::operator=(const VCMJitterEstimator& rhs)
-{
-    if (this != &rhs)
-    {
-        memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
-        memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
-
-        _vcmId = rhs._vcmId;
-        _receiverId = rhs._receiverId;
-        _avgFrameSize = rhs._avgFrameSize;
-        _varFrameSize = rhs._varFrameSize;
-        _maxFrameSize = rhs._maxFrameSize;
-        _fsSum = rhs._fsSum;
-        _fsCount = rhs._fsCount;
-        _lastUpdateT = rhs._lastUpdateT;
-        _prevEstimate = rhs._prevEstimate;
-        _prevFrameSize = rhs._prevFrameSize;
-        _avgNoise = rhs._avgNoise;
-        _alphaCount = rhs._alphaCount;
-        _filterJitterEstimate = rhs._filterJitterEstimate;
-        _startupCount = rhs._startupCount;
-        _latestNackTimestamp = rhs._latestNackTimestamp;
-        _nackCount = rhs._nackCount;
-        _rttFilter = rhs._rttFilter;
-    }
-    return *this;
-}
-
-// Resets the JitterEstimate
-void
-VCMJitterEstimator::Reset()
-{
-    _theta[0] = 1/(512e3/8);
-    _theta[1] = 0;
-    _varNoise = 4.0;
-
-    _thetaCov[0][0] = 1e-4;
-    _thetaCov[1][1] = 1e2;
-    _thetaCov[0][1] = _thetaCov[1][0] = 0;
-    _Qcov[0][0] = 2.5e-10;
-    _Qcov[1][1] = 1e-10;
-    _Qcov[0][1] = _Qcov[1][0] = 0;
-    _avgFrameSize = 500;
-    _maxFrameSize = 500;
-    _varFrameSize = 100;
-    _lastUpdateT = -1;
-    _prevEstimate = -1.0;
-    _prevFrameSize = 0;
-    _avgNoise = 0.0;
-    _alphaCount = 1;
-    _filterJitterEstimate = 0.0;
-    _latestNackTimestamp = 0;
-    _nackCount = 0;
-    _fsSum = 0;
-    _fsCount = 0;
-    _startupCount = 0;
-    _rttFilter.Reset();
-    fps_counter_.Reset();
-}
-
-void
-VCMJitterEstimator::ResetNackCount()
-{
-    _nackCount = 0;
-}
-
-// Updates the estimates with the new measurements
-void
-VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS, uint32_t frameSizeBytes,
-                                            bool incompleteFrame /* = false */)
-{
-    if (frameSizeBytes == 0)
-    {
-        return;
-    }
-    int deltaFS = frameSizeBytes - _prevFrameSize;
-    if (_fsCount < kFsAccuStartupSamples)
-    {
-        _fsSum += frameSizeBytes;
-        _fsCount++;
-    }
-    else if (_fsCount == kFsAccuStartupSamples)
-    {
-        // Give the frame size filter
-        _avgFrameSize = static_cast<double>(_fsSum) /
-                        static_cast<double>(_fsCount);
-        _fsCount++;
-    }
-    if (!incompleteFrame || frameSizeBytes > _avgFrameSize)
-    {
-        double avgFrameSize = _phi * _avgFrameSize +
-                              (1 - _phi) * frameSizeBytes;
-        if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize))
-        {
-            // Only update the average frame size if this sample wasn't a
-            // key frame
-            _avgFrameSize = avgFrameSize;
-        }
-        // Update the variance anyway since we want to capture cases where we only get
-        // key frames.
-        _varFrameSize = VCM_MAX(_phi * _varFrameSize + (1 - _phi) *
-                                (frameSizeBytes - avgFrameSize) *
-                                (frameSizeBytes - avgFrameSize), 1.0);
-    }
-
-    // Update max frameSize estimate
-    _maxFrameSize = VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
-
-    if (_prevFrameSize == 0)
-    {
-        _prevFrameSize = frameSizeBytes;
-        return;
-    }
-    _prevFrameSize = frameSizeBytes;
-
-    // Only update the Kalman filter if the sample is not considered
-    // an extreme outlier. Even if it is an extreme outlier from a
-    // delay point of view, if the frame size also is large the
-    // deviation is probably due to an incorrect line slope.
-    double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
-
-    if (fabs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
-        frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize))
-    {
-        // Update the variance of the deviation from the
-        // line given by the Kalman filter
-        EstimateRandomJitter(deviation, incompleteFrame);
-        // Prevent updating with frames which have been congested by a large
-        // frame, and therefore arrives almost at the same time as that frame.
-        // This can occur when we receive a large frame (key frame) which
-        // has been delayed. The next frame is of normal size (delta frame),
-        // and thus deltaFS will be << 0. This removes all frame samples
-        // which arrives after a key frame.
-        if ((!incompleteFrame || deviation >= 0.0) &&
-            static_cast<double>(deltaFS) > - 0.25 * _maxFrameSize)
-        {
-            // Update the Kalman filter with the new data
-            KalmanEstimateChannel(frameDelayMS, deltaFS);
-        }
-    }
-    else
-    {
-        int nStdDev = (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
-        EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
-    }
-    // Post process the total estimated jitter
-    if (_startupCount >= kStartupDelaySamples)
-    {
-        PostProcessEstimate();
-    }
-    else
-    {
-        _startupCount++;
-    }
-}
-
-// Updates the nack/packet ratio
-void
-VCMJitterEstimator::FrameNacked()
-{
-    // Wait until _nackLimit retransmissions has been received,
-    // then always add ~1 RTT delay.
-    // TODO(holmer): Should we ever remove the additional delay if the
-    // the packet losses seem to have stopped? We could for instance scale
-    // the number of RTTs to add with the amount of retransmissions in a given
-    // time interval, or similar.
-    if (_nackCount < _nackLimit)
-    {
-        _nackCount++;
-    }
-}
-
-// Updates Kalman estimate of the channel
-// The caller is expected to sanity check the inputs.
-void
-VCMJitterEstimator::KalmanEstimateChannel(int64_t frameDelayMS,
-                                          int32_t deltaFSBytes)
-{
-    double Mh[2];
-    double hMh_sigma;
-    double kalmanGain[2];
-    double measureRes;
-    double t00, t01;
-
-    // Kalman filtering
-
-    // Prediction
-    // M = M + Q
-    _thetaCov[0][0] += _Qcov[0][0];
-    _thetaCov[0][1] += _Qcov[0][1];
-    _thetaCov[1][0] += _Qcov[1][0];
-    _thetaCov[1][1] += _Qcov[1][1];
-
-    // Kalman gain
-    // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
-    // h = [dFS 1]
-    // Mh = M*h'
-    // hMh_sigma = h*M*h' + R
-    Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
-    Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
-    // sigma weights measurements with a small deltaFS as noisy and
-    // measurements with large deltaFS as good
-    if (_maxFrameSize < 1.0)
-    {
-        return;
-    }
-    double sigma = (300.0 * exp(-fabs(static_cast<double>(deltaFSBytes)) /
-                   (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise);
-    if (sigma < 1.0)
-    {
-        sigma = 1.0;
-    }
-    hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
-    if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) || (hMh_sigma > -1e-9 && hMh_sigma <= 0))
-    {
-        assert(false);
-        return;
-    }
-    kalmanGain[0] = Mh[0] / hMh_sigma;
-    kalmanGain[1] = Mh[1] / hMh_sigma;
-
-    // Correction
-    // theta = theta + K*(dT - h*theta)
-    measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
-    _theta[0] += kalmanGain[0] * measureRes;
-    _theta[1] += kalmanGain[1] * measureRes;
-
-    if (_theta[0] < _thetaLow)
-    {
-        _theta[0] = _thetaLow;
-    }
-
-    // M = (I - K*h)*M
-    t00 = _thetaCov[0][0];
-    t01 = _thetaCov[0][1];
-    _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
-                      kalmanGain[0] * _thetaCov[1][0];
-    _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
-                      kalmanGain[0] * _thetaCov[1][1];
-    _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
-                      kalmanGain[1] * deltaFSBytes * t00;
-    _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
-                      kalmanGain[1] * deltaFSBytes * t01;
-
-    // Covariance matrix, must be positive semi-definite
-    assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
-           _thetaCov[0][0] * _thetaCov[1][1] - _thetaCov[0][1] * _thetaCov[1][0] >= 0 &&
-           _thetaCov[0][0] >= 0);
-}
-
-// Calculate difference in delay between a sample and the
-// expected delay estimated by the Kalman filter
-double
-VCMJitterEstimator::DeviationFromExpectedDelay(int64_t frameDelayMS,
-                                               int32_t deltaFSBytes) const
-{
-    return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
-}
-
-// Estimates the random jitter by calculating the variance of the
-// sample distance from the line given by theta.
-void VCMJitterEstimator::EstimateRandomJitter(double d_dT,
-                                              bool incompleteFrame) {
-  uint64_t now = clock_->TimeInMicroseconds();
-  if (_lastUpdateT != -1) {
-    fps_counter_.AddSample(now - _lastUpdateT);
-  }
-  _lastUpdateT = now;
-
-  if (_alphaCount == 0) {
-    assert(false);
-    return;
-  }
-  double alpha =
-      static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
-  _alphaCount++;
-  if (_alphaCount > _alphaCountMax)
-    _alphaCount = _alphaCountMax;
-
-  if (LowRateExperimentEnabled()) {
-    // In order to avoid a low frame rate stream to react slower to changes,
-    // scale the alpha weight relative a 30 fps stream.
-    double fps = GetFrameRate();
-    if (fps > 0.0) {
-      double rate_scale = 30.0 / fps;
-      // At startup, there can be a lot of noise in the fps estimate.
-      // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
-      // at sample #kStartupDelaySamples.
-      if (_alphaCount < kStartupDelaySamples) {
-        rate_scale =
-            (_alphaCount * rate_scale + (kStartupDelaySamples - _alphaCount)) /
-            kStartupDelaySamples;
-      }
-      alpha = pow(alpha, rate_scale);
-    }
-  }
-
-  double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
-  double varNoise =
-      alpha * _varNoise + (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
-  if (!incompleteFrame || varNoise > _varNoise) {
-    _avgNoise = avgNoise;
-    _varNoise = varNoise;
-  }
-  if (_varNoise < 1.0) {
-    // The variance should never be zero, since we might get
-    // stuck and consider all samples as outliers.
-    _varNoise = 1.0;
-  }
-}
-
-double
-VCMJitterEstimator::NoiseThreshold() const
-{
-    double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
-    if (noiseThreshold < 1.0)
-    {
-        noiseThreshold = 1.0;
-    }
-    return noiseThreshold;
-}
-
-// Calculates the current jitter estimate from the filtered estimates
-double
-VCMJitterEstimator::CalculateEstimate()
-{
-    double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
-
-    // A very low estimate (or negative) is neglected
-    if (ret < 1.0) {
-        if (_prevEstimate <= 0.01)
-        {
-            ret = 1.0;
-        }
-        else
-        {
-            ret = _prevEstimate;
-        }
-    }
-    if (ret > 10000.0) // Sanity
-    {
-        ret = 10000.0;
-    }
-    _prevEstimate = ret;
-    return ret;
-}
-
-void
-VCMJitterEstimator::PostProcessEstimate()
-{
-    _filterJitterEstimate = CalculateEstimate();
-}
-
-void
-VCMJitterEstimator::UpdateRtt(int64_t rttMs)
-{
-    _rttFilter.Update(rttMs);
-}
-
-void
-VCMJitterEstimator::UpdateMaxFrameSize(uint32_t frameSizeBytes)
-{
-    if (_maxFrameSize < frameSizeBytes)
-    {
-        _maxFrameSize = frameSizeBytes;
-    }
-}
-
-// Returns the current filtered estimate if available,
-// otherwise tries to calculate an estimate.
-int VCMJitterEstimator::GetJitterEstimate(double rttMultiplier) {
-  double jitterMS = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
-  if (_filterJitterEstimate > jitterMS)
-    jitterMS = _filterJitterEstimate;
-  if (_nackCount >= _nackLimit)
-    jitterMS += _rttFilter.RttMs() * rttMultiplier;
-
-  if (LowRateExperimentEnabled()) {
-    static const double kJitterScaleLowThreshold = 5.0;
-    static const double kJitterScaleHighThreshold = 10.0;
-    double fps = GetFrameRate();
-    // Ignore jitter for very low fps streams.
-    if (fps < kJitterScaleLowThreshold) {
-      if (fps == 0.0) {
-        return jitterMS;
-      }
-      return 0;
-    }
-
-    // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
-    // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
-    if (fps < kJitterScaleHighThreshold) {
-      jitterMS =
-          (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
-          (fps - kJitterScaleLowThreshold) * jitterMS;
-    }
-  }
-
-  return static_cast<uint32_t>(jitterMS + 0.5);
-}
-
-bool VCMJitterEstimator::LowRateExperimentEnabled() {
-#ifndef WEBRTC_MOZILLA_BUILD
-  if (low_rate_experiment_ == kInit) {
-    std::string group =
-        webrtc::field_trial::FindFullName("WebRTC-ReducedJitterDelay");
-    if (group == "Disabled") {
-      low_rate_experiment_ = kDisabled;
-    } else {
-      low_rate_experiment_ = kEnabled;
-    }
-  }
-#endif
-  return low_rate_experiment_ == kEnabled ? true : false;
-}
-
-double VCMJitterEstimator::GetFrameRate() const {
-  if (fps_counter_.count() == 0)
-    return 0;
-
-  double fps = 1000000.0 / fps_counter_.ComputeMean();
-  // Sanity check.
-  assert(fps >= 0.0);
-  if (fps > kMaxFramerateEstimate) {
-    fps = kMaxFramerateEstimate;
-  }
-  return fps;
-}
-
-}
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator.h
deleted file mode 100644
index 46ed67ba1d..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/jitter_estimator.h
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
-
-#include "webrtc/base/rollingaccumulator.h"
-#include "webrtc/modules/video_coding/main/source/rtt_filter.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class Clock;
-
-class VCMJitterEstimator
-{
-public:
-    VCMJitterEstimator(const Clock* clock,
-                       int32_t vcmId = 0,
-                       int32_t receiverId = 0);
-    virtual ~VCMJitterEstimator();
-    VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
-
-    // Resets the estimate to the initial state
-    void Reset();
-    void ResetNackCount();
-
-    // Updates the jitter estimate with the new data.
-    //
-    // Input:
-    //          - frameDelay      : Delay-delta calculated by UTILDelayEstimate in milliseconds
-    //          - frameSize       : Frame size of the current frame.
-    //          - incompleteFrame : Flags if the frame is used to update the estimate before it
-    //                              was complete. Default is false.
-    void UpdateEstimate(int64_t frameDelayMS,
-                        uint32_t frameSizeBytes,
-                        bool incompleteFrame = false);
-
-    // Returns the current jitter estimate in milliseconds and adds
-    // also adds an RTT dependent term in cases of retransmission.
-    //  Input:
-    //          - rttMultiplier  : RTT param multiplier (when applicable).
-    //
-    // Return value                   : Jitter estimate in milliseconds
-    int GetJitterEstimate(double rttMultiplier);
-
-    // Updates the nack counter.
-    void FrameNacked();
-
-    // Updates the RTT filter.
-    //
-    // Input:
-    //          - rttMs               : RTT in ms
-    void UpdateRtt(int64_t rttMs);
-
-    void UpdateMaxFrameSize(uint32_t frameSizeBytes);
-
-    // A constant describing the delay from the jitter buffer
-    // to the delay on the receiving side which is not accounted
-    // for by the jitter buffer nor the decoding delay estimate.
-    static const uint32_t OPERATING_SYSTEM_JITTER = 10;
-
-protected:
-    // These are protected for better testing possibilities
-    double              _theta[2]; // Estimated line parameters (slope, offset)
-    double              _varNoise; // Variance of the time-deviation from the line
-
-    virtual bool LowRateExperimentEnabled();
-
-private:
-    // Updates the Kalman filter for the line describing
-    // the frame size dependent jitter.
-    //
-    // Input:
-    //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in milliseconds
-    //          - deltaFSBytes    : Frame size delta, i.e.
-    //                            : frame size at time T minus frame size at time T-1
-    void KalmanEstimateChannel(int64_t frameDelayMS, int32_t deltaFSBytes);
-
-    // Updates the random jitter estimate, i.e. the variance
-    // of the time deviations from the line given by the Kalman filter.
-    //
-    // Input:
-    //          - d_dT              : The deviation from the kalman estimate
-    //          - incompleteFrame   : True if the frame used to update the estimate
-    //                                with was incomplete
-    void EstimateRandomJitter(double d_dT, bool incompleteFrame);
-
-    double NoiseThreshold() const;
-
-    // Calculates the current jitter estimate.
-    //
-    // Return value                 : The current jitter estimate in milliseconds
-    double CalculateEstimate();
-
-    // Post process the calculated estimate
-    void PostProcessEstimate();
-
-    // Calculates the difference in delay between a sample and the
-    // expected delay estimated by the Kalman filter.
-    //
-    // Input:
-    //          - frameDelayMS    : Delay-delta calculated by UTILDelayEstimate in milliseconds
-    //          - deltaFS         : Frame size delta, i.e. frame size at time
-    //                              T minus frame size at time T-1
-    //
-    // Return value                 : The difference in milliseconds
-    double DeviationFromExpectedDelay(int64_t frameDelayMS,
-                                      int32_t deltaFSBytes) const;
-
-    double GetFrameRate() const;
-
-    // Constants, filter parameters
-    int32_t         _vcmId;
-    int32_t         _receiverId;
-    const double          _phi;
-    const double          _psi;
-    const uint32_t  _alphaCountMax;
-    const double          _thetaLow;
-    const uint32_t  _nackLimit;
-    const int32_t   _numStdDevDelayOutlier;
-    const int32_t   _numStdDevFrameSizeOutlier;
-    const double          _noiseStdDevs;
-    const double          _noiseStdDevOffset;
-
-    double                _thetaCov[2][2]; // Estimate covariance
-    double                _Qcov[2][2];     // Process noise covariance
-    double                _avgFrameSize;   // Average frame size
-    double                _varFrameSize;   // Frame size variance
-    double                _maxFrameSize;   // Largest frame size received (descending
-                                           // with a factor _psi)
-    uint32_t        _fsSum;
-    uint32_t        _fsCount;
-
-    int64_t         _lastUpdateT;
-    double                _prevEstimate;         // The previously returned jitter estimate
-    uint32_t        _prevFrameSize;        // Frame size of the previous frame
-    double                _avgNoise;             // Average of the random jitter
-    uint32_t        _alphaCount;
-    double                _filterJitterEstimate; // The filtered sum of jitter estimates
-
-    uint32_t        _startupCount;
-
-    int64_t         _latestNackTimestamp;  // Timestamp in ms when the latest nack was seen
-    uint32_t        _nackCount;            // Keeps track of the number of nacks received,
-                                                 // but never goes above _nackLimit
-    VCMRttFilter          _rttFilter;
-
-    rtc::RollingAccumulator<uint64_t> fps_counter_;
-    enum ExperimentFlag { kInit, kEnabled, kDisabled };
-    ExperimentFlag low_rate_experiment_;
-    const Clock* clock_;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_opt_util.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_opt_util.cc
deleted file mode 100644
index d929cbc35a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_opt_util.cc
+++ /dev/null
@@ -1,784 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-
-#include <algorithm>
-#include <float.h>
-#include <limits.h>
-#include <math.h>
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/fec_tables_xor.h"
-#include "webrtc/modules/video_coding/main/source/nack_fec_tables.h"
-
-namespace webrtc {
-// Max value of loss rates in off-line model
-static const int kPacketLossMax = 129;
-
-namespace media_optimization {
-
-VCMProtectionMethod::VCMProtectionMethod()
-    : _effectivePacketLoss(0),
-      _protectionFactorK(0),
-      _protectionFactorD(0),
-      _scaleProtKey(2.0f),
-      _maxPayloadSize(1460),
-      _qmRobustness(new VCMQmRobustness()),
-      _useUepProtectionK(false),
-      _useUepProtectionD(true),
-      _corrFecCost(1.0),
-      _type(kNone) {
-}
-
-VCMProtectionMethod::~VCMProtectionMethod()
-{
-    delete _qmRobustness;
-}
-void
-VCMProtectionMethod::UpdateContentMetrics(const
-                                          VideoContentMetrics* contentMetrics)
-{
-    _qmRobustness->UpdateContent(contentMetrics);
-}
-
-VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
-                                   int64_t highRttNackThresholdMs)
-    : VCMFecMethod(),
-      _lowRttNackMs(lowRttNackThresholdMs),
-      _highRttNackMs(highRttNackThresholdMs),
-      _maxFramesFec(1) {
-  assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
-  assert(highRttNackThresholdMs == -1 ||
-         lowRttNackThresholdMs <= highRttNackThresholdMs);
-  assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
-  _type = kNackFec;
-}
-
-VCMNackFecMethod::~VCMNackFecMethod()
-{
-    //
-}
-bool
-VCMNackFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
-{
-    // Hybrid Nack FEC has three operational modes:
-    // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
-    //    (_protectionFactorD) to zero. -1 means no FEC.
-    // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
-    //    -1 means always allow NACK.
-    // 3. Medium RTT values - Hybrid mode: We will only nack the
-    //    residual following the decoding of the FEC (refer to JB logic). FEC
-    //    delta protection factor will be adjusted based on the RTT.
-
-    // Otherwise: we count on FEC; if the RTT is below a threshold, then we
-    // nack the residual, based on a decision made in the JB.
-
-    // Compute the protection factors
-    VCMFecMethod::ProtectionFactor(parameters);
-    if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs)
-    {
-        _protectionFactorD = 0;
-        VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
-    }
-
-    // When in Hybrid mode (RTT range), adjust FEC rates based on the
-    // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
-    else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs)
-    {
-        // TODO(mikhal): Disabling adjustment temporarily.
-        // uint16_t rttIndex = (uint16_t) parameters->rtt;
-        float adjustRtt = 1.0f;// (float)VCMNackFecTable[rttIndex] / 100.0f;
-
-        // Adjust FEC with NACK on (for delta frame only)
-        // table depends on RTT relative to rttMax (NACK Threshold)
-        _protectionFactorD = static_cast<uint8_t>
-                            (adjustRtt *
-                             static_cast<float>(_protectionFactorD));
-        // update FEC rates after applying adjustment
-        VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
-    }
-
-    return true;
-}
-
-int VCMNackFecMethod::ComputeMaxFramesFec(
-    const VCMProtectionParameters* parameters) {
-  if (parameters->numLayers > 2) {
-    // For more than 2 temporal layers we will only have FEC on the base layer,
-    // and the base layers will be pretty far apart. Therefore we force one
-    // frame FEC.
-    return 1;
-  }
-  // We set the max number of frames to base the FEC on so that on average
-  // we will have complete frames in one RTT. Note that this is an upper
-  // bound, and that the actual number of frames used for FEC is decided by the
-  // RTP module based on the actual number of packets and the protection factor.
-  float base_layer_framerate = parameters->frameRate /
-      static_cast<float>(1 << (parameters->numLayers - 1));
-  int max_frames_fec = std::max(static_cast<int>(
-      2.0f * base_layer_framerate * parameters->rtt /
-      1000.0f + 0.5f), 1);
-  // |kUpperLimitFramesFec| is the upper limit on how many frames we
-  // allow any FEC to be based on.
-  if (max_frames_fec > kUpperLimitFramesFec) {
-    max_frames_fec = kUpperLimitFramesFec;
-  }
-  return max_frames_fec;
-}
-
-int VCMNackFecMethod::MaxFramesFec() const {
-  return _maxFramesFec;
-}
-
-bool VCMNackFecMethod::BitRateTooLowForFec(
-    const VCMProtectionParameters* parameters) {
-  // Bitrate below which we turn off FEC, regardless of reported packet loss.
-  // The condition should depend on resolution and content. For now, use
-  // threshold on bytes per frame, with some effect for the frame size.
-  // The condition for turning off FEC is also based on other factors,
-  // such as |_numLayers|, |_maxFramesFec|, and |_rtt|.
-  int estimate_bytes_per_frame = 1000 * BitsPerFrame(parameters) / 8;
-  int max_bytes_per_frame = kMaxBytesPerFrameForFec;
-  int num_pixels = parameters->codecWidth * parameters->codecHeight;
-  if (num_pixels <= 352 * 288) {
-    max_bytes_per_frame = kMaxBytesPerFrameForFecLow;
-  } else if (num_pixels > 640 * 480) {
-    max_bytes_per_frame = kMaxBytesPerFrameForFecHigh;
-  }
-  // TODO (marpan): add condition based on maximum frames used for FEC,
-  // and expand condition based on frame size.
-  // Max round trip time threshold in ms.
-  const int64_t kMaxRttTurnOffFec = 200;
-  if (estimate_bytes_per_frame < max_bytes_per_frame &&
-      parameters->numLayers < 3 &&
-      parameters->rtt < kMaxRttTurnOffFec) {
-    return true;
-  }
-  return false;
-}
-
-bool
-VCMNackFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
-{
-    // Set the effective packet loss for encoder (based on FEC code).
-    // Compute the effective packet loss and residual packet loss due to FEC.
-    VCMFecMethod::EffectivePacketLoss(parameters);
-    return true;
-}
-
-bool
-VCMNackFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
-    ProtectionFactor(parameters);
-    EffectivePacketLoss(parameters);
-    _maxFramesFec = ComputeMaxFramesFec(parameters);
-    if (BitRateTooLowForFec(parameters)) {
-      _protectionFactorK = 0;
-      _protectionFactorD = 0;
-    }
-
-    // Protection/fec rates obtained above are defined relative to total number
-    // of packets (total rate: source + fec) FEC in RTP module assumes
-    // protection factor is defined relative to source number of packets so we
-    // should convert the factor to reduce mismatch between mediaOpt's rate and
-    // the actual one
-    _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
-    _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
-
-    return true;
-}
-
-VCMNackMethod::VCMNackMethod():
-VCMProtectionMethod()
-{
-    _type = kNack;
-}
-
-VCMNackMethod::~VCMNackMethod()
-{
-    //
-}
-
-bool
-VCMNackMethod::EffectivePacketLoss(const VCMProtectionParameters* parameter)
-{
-    // Effective Packet Loss, NA in current version.
-    _effectivePacketLoss = 0;
-    return true;
-}
-
-bool
-VCMNackMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
-    // Compute the effective packet loss
-    EffectivePacketLoss(parameters);
-
-    // nackCost  = (bitRate - nackCost) * (lossPr)
-    return true;
-}
-
-VCMFecMethod::VCMFecMethod():
-VCMProtectionMethod()
-{
-    _type = kFec;
-}
-VCMFecMethod::~VCMFecMethod()
-{
-    //
-}
-
-uint8_t
-VCMFecMethod::BoostCodeRateKey(uint8_t packetFrameDelta,
-                               uint8_t packetFrameKey) const
-{
-    uint8_t boostRateKey = 2;
-    // Default: ratio scales the FEC protection up for I frames
-    uint8_t ratio = 1;
-
-    if (packetFrameDelta > 0)
-    {
-        ratio = (int8_t) (packetFrameKey / packetFrameDelta);
-    }
-    ratio = VCM_MAX(boostRateKey, ratio);
-
-    return ratio;
-}
-
-uint8_t
-VCMFecMethod::ConvertFECRate(uint8_t codeRateRTP) const
-{
-    return static_cast<uint8_t> (VCM_MIN(255,(0.5 + 255.0 * codeRateRTP /
-                                      (float)(255 - codeRateRTP))));
-}
-
-// Update FEC with protectionFactorD
-void
-VCMFecMethod::UpdateProtectionFactorD(uint8_t protectionFactorD)
-{
-    _protectionFactorD = protectionFactorD;
-}
-
-// Update FEC with protectionFactorK
-void
-VCMFecMethod::UpdateProtectionFactorK(uint8_t protectionFactorK)
-{
-    _protectionFactorK = protectionFactorK;
-}
-
-bool
-VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
-{
-    // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
-
-    // No protection if (filtered) packetLoss is 0
-    uint8_t packetLoss = (uint8_t) (255 * parameters->lossPr);
-    if (packetLoss == 0)
-    {
-        _protectionFactorK = 0;
-        _protectionFactorD = 0;
-         return true;
-    }
-
-    // Parameters for FEC setting:
-    // first partition size, thresholds, table pars, spatial resoln fac.
-
-    // First partition protection: ~ 20%
-    uint8_t firstPartitionProt = (uint8_t) (255 * 0.20);
-
-    // Minimum protection level needed to generate one FEC packet for one
-    // source packet/frame (in RTP sender)
-    uint8_t minProtLevelFec = 85;
-
-    // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
-    // above which we allocate protection to cover at least first partition.
-    uint8_t lossThr = 0;
-    uint8_t packetNumThr = 1;
-
-    // Parameters for range of rate index of table.
-    const uint8_t ratePar1 = 5;
-    const uint8_t ratePar2 = 49;
-
-    // Spatial resolution size, relative to a reference size.
-    float spatialSizeToRef = static_cast<float>
-                           (parameters->codecWidth * parameters->codecHeight) /
-                           (static_cast<float>(704 * 576));
-    // resolnFac: This parameter will generally increase/decrease the FEC rate
-    // (for fixed bitRate and packetLoss) based on system size.
-    // Use a smaller exponent (< 1) to control/soften system size effect.
-    const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
-
-    const int bitRatePerFrame = BitsPerFrame(parameters);
-
-
-    // Average number of packets per frame (source and fec):
-    const uint8_t avgTotPackets = 1 + (uint8_t)
-                                        ((float) bitRatePerFrame * 1000.0
-                                       / (float) (8.0 * _maxPayloadSize) + 0.5);
-
-    // FEC rate parameters: for P and I frame
-    uint8_t codeRateDelta = 0;
-    uint8_t codeRateKey = 0;
-
-    // Get index for table: the FEC protection depends on an effective rate.
-    // The range on the rate index corresponds to rates (bps)
-    // from ~200k to ~8000k, for 30fps
-    const uint16_t effRateFecTable = static_cast<uint16_t>
-                                           (resolnFac * bitRatePerFrame);
-    uint8_t rateIndexTable =
-        (uint8_t) VCM_MAX(VCM_MIN((effRateFecTable - ratePar1) /
-                                         ratePar1, ratePar2), 0);
-
-    // Restrict packet loss range to 50:
-    // current tables defined only up to 50%
-    if (packetLoss >= kPacketLossMax)
-    {
-        packetLoss = kPacketLossMax - 1;
-    }
-    uint16_t indexTable = rateIndexTable * kPacketLossMax + packetLoss;
-
-    // Check on table index
-    assert(indexTable < kSizeCodeRateXORTable);
-
-    // Protection factor for P frame
-    codeRateDelta = kCodeRateXORTable[indexTable];
-
-    if (packetLoss > lossThr && avgTotPackets > packetNumThr)
-    {
-        // Set a minimum based on first partition size.
-        if (codeRateDelta < firstPartitionProt)
-        {
-            codeRateDelta = firstPartitionProt;
-        }
-    }
-
-    // Check limit on amount of protection for P frame; 50% is max.
-    if (codeRateDelta >= kPacketLossMax)
-    {
-        codeRateDelta = kPacketLossMax - 1;
-    }
-
-    float adjustFec = 1.0f;
-    // Avoid additional adjustments when layers are active.
-    // TODO(mikhal/marco): Update adjusmtent based on layer info.
-    if (parameters->numLayers == 1)
-    {
-        adjustFec = _qmRobustness->AdjustFecFactor(codeRateDelta,
-                                                   parameters->bitRate,
-                                                   parameters->frameRate,
-                                                   parameters->rtt,
-                                                   packetLoss);
-    }
-
-    codeRateDelta = static_cast<uint8_t>(codeRateDelta * adjustFec);
-
-    // For Key frame:
-    // Effectively at a higher rate, so we scale/boost the rate
-    // The boost factor may depend on several factors: ratio of packet
-    // number of I to P frames, how much protection placed on P frames, etc.
-    const uint8_t packetFrameDelta = (uint8_t)
-                                           (0.5 + parameters->packetsPerFrame);
-    const uint8_t packetFrameKey = (uint8_t)
-                                         (0.5 + parameters->packetsPerFrameKey);
-    const uint8_t boostKey = BoostCodeRateKey(packetFrameDelta,
-                                                    packetFrameKey);
-
-    rateIndexTable = (uint8_t) VCM_MAX(VCM_MIN(
-                      1 + (boostKey * effRateFecTable - ratePar1) /
-                      ratePar1,ratePar2),0);
-    uint16_t indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
-
-    indexTableKey = VCM_MIN(indexTableKey, kSizeCodeRateXORTable);
-
-    // Check on table index
-    assert(indexTableKey < kSizeCodeRateXORTable);
-
-    // Protection factor for I frame
-    codeRateKey = kCodeRateXORTable[indexTableKey];
-
-    // Boosting for Key frame.
-    int boostKeyProt = _scaleProtKey * codeRateDelta;
-    if (boostKeyProt >= kPacketLossMax)
-    {
-        boostKeyProt = kPacketLossMax - 1;
-    }
-
-    // Make sure I frame protection is at least larger than P frame protection,
-    // and at least as high as filtered packet loss.
-    codeRateKey = static_cast<uint8_t> (VCM_MAX(packetLoss,
-            VCM_MAX(boostKeyProt, codeRateKey)));
-
-    // Check limit on amount of protection for I frame: 50% is max.
-    if (codeRateKey >= kPacketLossMax)
-    {
-        codeRateKey = kPacketLossMax - 1;
-    }
-
-    _protectionFactorK = codeRateKey;
-    _protectionFactorD = codeRateDelta;
-
-    // Generally there is a rate mis-match between the FEC cost estimated
-    // in mediaOpt and the actual FEC cost sent out in RTP module.
-    // This is more significant at low rates (small # of source packets), where
-    // the granularity of the FEC decreases. In this case, non-zero protection
-    // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
-    // is based on rounding off protectionFactor on actual source packet number).
-    // The correction factor (_corrFecCost) attempts to corrects this, at least
-    // for cases of low rates (small #packets) and low protection levels.
-
-    float numPacketsFl = 1.0f + ((float) bitRatePerFrame * 1000.0
-                                / (float) (8.0 * _maxPayloadSize) + 0.5);
-
-    const float estNumFecGen = 0.5f + static_cast<float> (_protectionFactorD *
-                                                         numPacketsFl / 255.0f);
-
-
-    // We reduce cost factor (which will reduce overhead for FEC and
-    // hybrid method) and not the protectionFactor.
-    _corrFecCost = 1.0f;
-    if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec)
-    {
-        _corrFecCost = 0.5f;
-    }
-    if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec)
-    {
-        _corrFecCost = 0.0f;
-    }
-
-     // TODO (marpan): Set the UEP protection on/off for Key and Delta frames
-    _useUepProtectionK = _qmRobustness->SetUepProtection(codeRateKey,
-                                                         parameters->bitRate,
-                                                         packetLoss,
-                                                         0);
-
-    _useUepProtectionD = _qmRobustness->SetUepProtection(codeRateDelta,
-                                                         parameters->bitRate,
-                                                         packetLoss,
-                                                         1);
-
-    // DONE WITH FEC PROTECTION SETTINGS
-    return true;
-}
-
-int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
-  // When temporal layers are available FEC will only be applied on the base
-  // layer.
-  const float bitRateRatio =
-    kVp8LayerRateAlloction[parameters->numLayers - 1][0];
-  float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1);
-  float bitRate = parameters->bitRate * bitRateRatio;
-  float frameRate = parameters->frameRate * frameRateRatio;
-
-  // TODO(mikhal): Update factor following testing.
-  float adjustmentFactor = 1;
-
-  // Average bits per frame (units of kbits)
-  return static_cast<int>(adjustmentFactor * bitRate / frameRate);
-}
-
-bool
-VCMFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
-{
-    // Effective packet loss to encoder is based on RPL (residual packet loss)
-    // this is a soft setting based on degree of FEC protection
-    // RPL = received/input packet loss - average_FEC_recovery
-    // note: received/input packet loss may be filtered based on FilteredLoss
-
-    // Effective Packet Loss, NA in current version.
-    _effectivePacketLoss = 0;
-
-    return true;
-}
-
-bool
-VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
-    // Compute the protection factor
-    ProtectionFactor(parameters);
-
-    // Compute the effective packet loss
-    EffectivePacketLoss(parameters);
-
-    // Protection/fec rates obtained above is defined relative to total number
-    // of packets (total rate: source+fec) FEC in RTP module assumes protection
-    // factor is defined relative to source number of packets so we should
-    // convert the factor to reduce mismatch between mediaOpt suggested rate and
-    // the actual rate
-    _protectionFactorK = ConvertFECRate(_protectionFactorK);
-    _protectionFactorD = ConvertFECRate(_protectionFactorD);
-
-    return true;
-}
-VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs):
-_selectedMethod(NULL),
-_currentParameters(),
-_rtt(0),
-_lossPr(0.0f),
-_bitRate(0.0f),
-_frameRate(0.0f),
-_keyFrameSize(0.0f),
-_fecRateKey(0),
-_fecRateDelta(0),
-_lastPrUpdateT(0),
-_lossPr255(0.9999f),
-_lossPrHistory(),
-_shortMaxLossPr255(0),
-_packetsPerFrame(0.9999f),
-_packetsPerFrameKey(0.9999f),
-_codecWidth(0),
-_codecHeight(0),
-_numLayers(1)
-{
-    Reset(nowMs);
-}
-
-VCMLossProtectionLogic::~VCMLossProtectionLogic()
-{
-    Release();
-}
-
-void VCMLossProtectionLogic::SetMethod(
-    enum VCMProtectionMethodEnum newMethodType) {
-  if (_selectedMethod != nullptr) {
-    if (_selectedMethod->Type() == newMethodType)
-      return;
-    // Remove old method.
-    delete _selectedMethod;
-  }
-
-  switch(newMethodType) {
-    case kNack:
-      _selectedMethod = new VCMNackMethod();
-      break;
-    case kFec:
-      _selectedMethod = new VCMFecMethod();
-      break;
-    case kNackFec:
-      _selectedMethod = new VCMNackFecMethod(kLowRttNackMs, -1);
-      break;
-    case kNone:
-      _selectedMethod = nullptr;
-      break;
-  }
-  UpdateMethod();
-}
-
-void
-VCMLossProtectionLogic::UpdateRtt(int64_t rtt)
-{
-    _rtt = rtt;
-}
-
-void
-VCMLossProtectionLogic::UpdateMaxLossHistory(uint8_t lossPr255,
-                                             int64_t now)
-{
-    if (_lossPrHistory[0].timeMs >= 0 &&
-        now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs)
-    {
-        if (lossPr255 > _shortMaxLossPr255)
-        {
-            _shortMaxLossPr255 = lossPr255;
-        }
-    }
-    else
-    {
-        // Only add a new value to the history once a second
-        if (_lossPrHistory[0].timeMs == -1)
-        {
-            // First, no shift
-            _shortMaxLossPr255 = lossPr255;
-        }
-        else
-        {
-            // Shift
-            for (int32_t i = (kLossPrHistorySize - 2); i >= 0; i--)
-            {
-                _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
-                _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
-            }
-        }
-        if (_shortMaxLossPr255 == 0)
-        {
-            _shortMaxLossPr255 = lossPr255;
-        }
-
-        _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
-        _lossPrHistory[0].timeMs = now;
-        _shortMaxLossPr255 = 0;
-    }
-}
-
-uint8_t
-VCMLossProtectionLogic::MaxFilteredLossPr(int64_t nowMs) const
-{
-    uint8_t maxFound = _shortMaxLossPr255;
-    if (_lossPrHistory[0].timeMs == -1)
-    {
-        return maxFound;
-    }
-    for (int32_t i = 0; i < kLossPrHistorySize; i++)
-    {
-        if (_lossPrHistory[i].timeMs == -1)
-        {
-            break;
-        }
-        if (nowMs - _lossPrHistory[i].timeMs >
-            kLossPrHistorySize * kLossPrShortFilterWinMs)
-        {
-            // This sample (and all samples after this) is too old
-            break;
-        }
-        if (_lossPrHistory[i].lossPr255 > maxFound)
-        {
-            // This sample is the largest one this far into the history
-            maxFound = _lossPrHistory[i].lossPr255;
-        }
-    }
-    return maxFound;
-}
-
-uint8_t VCMLossProtectionLogic::FilteredLoss(
-    int64_t nowMs,
-    FilterPacketLossMode filter_mode,
-    uint8_t lossPr255) {
-
-  // Update the max window filter.
-  UpdateMaxLossHistory(lossPr255, nowMs);
-
-  // Update the recursive average filter.
-  _lossPr255.Apply(static_cast<float> (nowMs - _lastPrUpdateT),
-                   static_cast<float> (lossPr255));
-  _lastPrUpdateT = nowMs;
-
-  // Filtered loss: default is received loss (no filtering).
-  uint8_t filtered_loss = lossPr255;
-
-  switch (filter_mode) {
-    case kNoFilter:
-      break;
-    case kAvgFilter:
-      filtered_loss = static_cast<uint8_t>(_lossPr255.filtered() + 0.5);
-      break;
-    case kMaxFilter:
-      filtered_loss = MaxFilteredLossPr(nowMs);
-      break;
-  }
-
-  return filtered_loss;
-}
-
-void
-VCMLossProtectionLogic::UpdateFilteredLossPr(uint8_t packetLossEnc)
-{
-    _lossPr = (float) packetLossEnc / (float) 255.0;
-}
-
-void
-VCMLossProtectionLogic::UpdateBitRate(float bitRate)
-{
-    _bitRate = bitRate;
-}
-
-void
-VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets, int64_t nowMs)
-{
-    _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
-                           nPackets);
-    _lastPacketPerFrameUpdateT = nowMs;
-}
-
-void
-VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs)
-{
-    _packetsPerFrameKey.Apply(static_cast<float>(nowMs -
-                              _lastPacketPerFrameUpdateTKey), nPackets);
-    _lastPacketPerFrameUpdateTKey = nowMs;
-}
-
-void
-VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize)
-{
-    _keyFrameSize = keyFrameSize;
-}
-
-void
-VCMLossProtectionLogic::UpdateFrameSize(uint16_t width,
-                                        uint16_t height)
-{
-    _codecWidth = width;
-    _codecHeight = height;
-}
-
-void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
-  _numLayers = (numLayers == 0) ? 1 : numLayers;
-}
-
-bool
-VCMLossProtectionLogic::UpdateMethod()
-{
-    if (_selectedMethod == NULL)
-    {
-        return false;
-    }
-    _currentParameters.rtt = _rtt;
-    _currentParameters.lossPr = _lossPr;
-    _currentParameters.bitRate = _bitRate;
-    _currentParameters.frameRate = _frameRate; // rename actual frame rate?
-    _currentParameters.keyFrameSize = _keyFrameSize;
-    _currentParameters.fecRateDelta = _fecRateDelta;
-    _currentParameters.fecRateKey = _fecRateKey;
-    _currentParameters.packetsPerFrame = _packetsPerFrame.filtered();
-    _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.filtered();
-    _currentParameters.codecWidth = _codecWidth;
-    _currentParameters.codecHeight = _codecHeight;
-    _currentParameters.numLayers = _numLayers;
-    return _selectedMethod->UpdateParameters(&_currentParameters);
-}
-
-VCMProtectionMethod*
-VCMLossProtectionLogic::SelectedMethod() const
-{
-    return _selectedMethod;
-}
-
-VCMProtectionMethodEnum VCMLossProtectionLogic::SelectedType() const {
-  return _selectedMethod == nullptr ? kNone : _selectedMethod->Type();
-}
-
-void
-VCMLossProtectionLogic::Reset(int64_t nowMs)
-{
-    _lastPrUpdateT = nowMs;
-    _lastPacketPerFrameUpdateT = nowMs;
-    _lastPacketPerFrameUpdateTKey = nowMs;
-    _lossPr255.Reset(0.9999f);
-    _packetsPerFrame.Reset(0.9999f);
-    _fecRateDelta = _fecRateKey = 0;
-    for (int32_t i = 0; i < kLossPrHistorySize; i++)
-    {
-        _lossPrHistory[i].lossPr255 = 0;
-        _lossPrHistory[i].timeMs = -1;
-    }
-    _shortMaxLossPr255 = 0;
-    Release();
-}
-
-void
-VCMLossProtectionLogic::Release()
-{
-    delete _selectedMethod;
-    _selectedMethod = NULL;
-}
-
-}  // namespace media_optimization
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_opt_util.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_opt_util.h
deleted file mode 100644
index 498238768f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_opt_util.h
+++ /dev/null
@@ -1,363 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "webrtc/base/exp_filter.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace media_optimization {
-
-// Number of time periods used for (max) window filter for packet loss
-// TODO (marpan): set reasonable window size for filtered packet loss,
-// adjustment should be based on logged/real data of loss stats/correlation.
-enum { kLossPrHistorySize = 10 };
-
-// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
-enum { kLossPrShortFilterWinMs = 1000 };
-
-// The type of filter used on the received packet loss reports.
-enum FilterPacketLossMode {
-  kNoFilter,    // No filtering on received loss.
-  kAvgFilter,   // Recursive average filter.
-  kMaxFilter    // Max-window filter, over the time interval of:
-                // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
-};
-
-// Thresholds for hybrid NACK/FEC
-// common to media optimization and the jitter buffer.
-const int64_t kLowRttNackMs = 20;
-
-struct VCMProtectionParameters
-{
-    VCMProtectionParameters() : rtt(0), lossPr(0.0f), bitRate(0.0f),
-        packetsPerFrame(0.0f), packetsPerFrameKey(0.0f), frameRate(0.0f),
-        keyFrameSize(0.0f), fecRateDelta(0), fecRateKey(0),
-        codecWidth(0), codecHeight(0),
-        numLayers(1)
-        {}
-
-    int64_t             rtt;
-    float               lossPr;
-    float               bitRate;
-    float               packetsPerFrame;
-    float               packetsPerFrameKey;
-    float               frameRate;
-    float               keyFrameSize;
-    uint8_t       fecRateDelta;
-    uint8_t       fecRateKey;
-    uint16_t      codecWidth;
-    uint16_t      codecHeight;
-    int                 numLayers;
-};
-
-
-/******************************/
-/* VCMProtectionMethod class  */
-/******************************/
-
-enum VCMProtectionMethodEnum
-{
-    kNack,
-    kFec,
-    kNackFec,
-    kNone
-};
-
-class VCMLossProbabilitySample
-{
-public:
-    VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {};
-
-    uint8_t     lossPr255;
-    int64_t     timeMs;
-};
-
-
-class VCMProtectionMethod
-{
-public:
-    VCMProtectionMethod();
-    virtual ~VCMProtectionMethod();
-
-    // Updates the efficiency of the method using the parameters provided
-    //
-    // Input:
-    //         - parameters         : Parameters used to calculate efficiency
-    //
-    // Return value                 : True if this method is recommended in
-    //                                the given conditions.
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
-
-    // Returns the protection type
-    //
-    // Return value                 : The protection type
-    enum VCMProtectionMethodEnum Type() const { return _type; }
-
-    // Returns the effective packet loss for ER, required by this protection method
-    //
-    // Return value                 : Required effective packet loss
-    virtual uint8_t RequiredPacketLossER() { return _effectivePacketLoss; }
-
-    // Extracts the FEC protection factor for Key frame, required by this protection method
-    //
-    // Return value                 : Required protectionFactor for Key frame
-    virtual uint8_t RequiredProtectionFactorK() { return _protectionFactorK; }
-
-    // Extracts the FEC protection factor for Delta frame, required by this protection method
-    //
-    // Return value                 : Required protectionFactor for delta frame
-    virtual uint8_t RequiredProtectionFactorD() { return _protectionFactorD; }
-
-    // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
-    //
-    // Return value                 : Required Unequal protection on/off state.
-    virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
-
-    // Extracts whether the the FEC Unequal protection (UEP) is used for Delta frame.
-    //
-    // Return value                 : Required Unequal protection on/off state.
-    virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
-
-    virtual int MaxFramesFec() const { return 1; }
-
-    // Updates content metrics
-    void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
-
-protected:
-
-    uint8_t                        _effectivePacketLoss;
-    uint8_t                        _protectionFactorK;
-    uint8_t                        _protectionFactorD;
-    // Estimation of residual loss after the FEC
-    float                                _scaleProtKey;
-    int32_t                        _maxPayloadSize;
-
-    VCMQmRobustness*                     _qmRobustness;
-    bool                                 _useUepProtectionK;
-    bool                                 _useUepProtectionD;
-    float                                _corrFecCost;
-    enum VCMProtectionMethodEnum         _type;
-};
-
-class VCMNackMethod : public VCMProtectionMethod
-{
-public:
-    VCMNackMethod();
-    virtual ~VCMNackMethod();
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
-    // Get the effective packet loss
-    bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
-};
-
-class VCMFecMethod : public VCMProtectionMethod
-{
-public:
-    VCMFecMethod();
-    virtual ~VCMFecMethod();
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
-    // Get the effective packet loss for ER
-    bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
-    // Get the FEC protection factors
-    bool ProtectionFactor(const VCMProtectionParameters* parameters);
-    // Get the boost for key frame protection
-    uint8_t BoostCodeRateKey(uint8_t packetFrameDelta,
-                                   uint8_t packetFrameKey) const;
-    // Convert the rates: defined relative to total# packets or source# packets
-    uint8_t ConvertFECRate(uint8_t codeRate) const;
-    // Get the average effective recovery from FEC: for random loss model
-    float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
-    // Update FEC with protectionFactorD
-    void UpdateProtectionFactorD(uint8_t protectionFactorD);
-    // Update FEC with protectionFactorK
-    void UpdateProtectionFactorK(uint8_t protectionFactorK);
-    // Compute the bits per frame. Account for temporal layers when applicable.
-    int BitsPerFrame(const VCMProtectionParameters* parameters);
-
-protected:
-    enum { kUpperLimitFramesFec = 6 };
-    // Thresholds values for the bytes/frame and round trip time, below which we
-    // may turn off FEC, depending on |_numLayers| and |_maxFramesFec|.
-    // Max bytes/frame for VGA, corresponds to ~140k at 25fps.
-    enum { kMaxBytesPerFrameForFec = 700 };
-    // Max bytes/frame for CIF and lower: corresponds to ~80k at 25fps.
-    enum { kMaxBytesPerFrameForFecLow = 400 };
-    // Max bytes/frame for frame size larger than VGA, ~200k at 25fps.
-    enum { kMaxBytesPerFrameForFecHigh = 1000 };
-};
-
-
-class VCMNackFecMethod : public VCMFecMethod
-{
-public:
-    VCMNackFecMethod(int64_t lowRttNackThresholdMs,
-                     int64_t highRttNackThresholdMs);
-    virtual ~VCMNackFecMethod();
-    virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
-    // Get the effective packet loss for ER
-    bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
-    // Get the protection factors
-    bool ProtectionFactor(const VCMProtectionParameters* parameters);
-    // Get the max number of frames the FEC is allowed to be based on.
-    int MaxFramesFec() const;
-    // Turn off the FEC based on low bitrate and other factors.
-    bool BitRateTooLowForFec(const VCMProtectionParameters* parameters);
-private:
-    int ComputeMaxFramesFec(const VCMProtectionParameters* parameters);
-
-    int64_t _lowRttNackMs;
-    int64_t _highRttNackMs;
-    int _maxFramesFec;
-};
-
-class VCMLossProtectionLogic
-{
-public:
-    VCMLossProtectionLogic(int64_t nowMs);
-    ~VCMLossProtectionLogic();
-
-    // Set the protection method to be used
-    //
-    // Input:
-    //        - newMethodType    : New requested protection method type. If one
-    //                           is already set, it will be deleted and replaced
-    void SetMethod(VCMProtectionMethodEnum newMethodType);
-
-    // Update the round-trip time
-    //
-    // Input:
-    //          - rtt           : Round-trip time in seconds.
-    void UpdateRtt(int64_t rtt);
-
-    // Update the filtered packet loss.
-    //
-    // Input:
-    //          - packetLossEnc :  The reported packet loss filtered
-    //                             (max window or average)
-    void UpdateFilteredLossPr(uint8_t packetLossEnc);
-
-    // Update the current target bit rate.
-    //
-    // Input:
-    //          - bitRate          : The current target bit rate in kbits/s
-    void UpdateBitRate(float bitRate);
-
-    // Update the number of packets per frame estimate, for delta frames
-    //
-    // Input:
-    //          - nPackets         : Number of packets in the latest sent frame.
-    void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
-
-   // Update the number of packets per frame estimate, for key frames
-    //
-    // Input:
-    //          - nPackets         : umber of packets in the latest sent frame.
-    void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
-
-    // Update the keyFrameSize estimate
-    //
-    // Input:
-    //          - keyFrameSize     : The size of the latest sent key frame.
-    void UpdateKeyFrameSize(float keyFrameSize);
-
-    // Update the frame rate
-    //
-    // Input:
-    //          - frameRate        : The current target frame rate.
-    void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
-
-    // Update the frame size
-    //
-    // Input:
-    //          - width        : The codec frame width.
-    //          - height       : The codec frame height.
-    void UpdateFrameSize(uint16_t width, uint16_t height);
-
-    // Update the number of active layers
-    //
-    // Input:
-    //          - numLayers    : Number of layers used.
-    void UpdateNumLayers(int numLayers);
-
-    // The amount of packet loss to cover for with FEC.
-    //
-    // Input:
-    //          - fecRateKey      : Packet loss to cover for with FEC when
-    //                              sending key frames.
-    //          - fecRateDelta    : Packet loss to cover for with FEC when
-    //                              sending delta frames.
-    void UpdateFECRates(uint8_t fecRateKey, uint8_t fecRateDelta)
-                       { _fecRateKey = fecRateKey;
-                         _fecRateDelta = fecRateDelta; }
-
-    // Update the protection methods with the current VCMProtectionParameters
-    // and set the requested protection settings.
-    // Return value     : Returns true on update
-    bool UpdateMethod();
-
-    // Returns the method currently selected.
-    //
-    // Return value                 : The protection method currently selected.
-    VCMProtectionMethod* SelectedMethod() const;
-
-    // Return the protection type of the currently selected method
-    VCMProtectionMethodEnum SelectedType() const;
-
-    // Updates the filtered loss for the average and max window packet loss,
-    // and returns the filtered loss probability in the interval [0, 255].
-    // The returned filtered loss value depends on the parameter |filter_mode|.
-    // The input parameter |lossPr255| is the received packet loss.
-
-    // Return value                 : The filtered loss probability
-    uint8_t FilteredLoss(int64_t nowMs, FilterPacketLossMode filter_mode,
-                               uint8_t lossPr255);
-
-    void Reset(int64_t nowMs);
-
-    void Release();
-
-private:
-    // Sets the available loss protection methods.
-    void UpdateMaxLossHistory(uint8_t lossPr255, int64_t now);
-    uint8_t MaxFilteredLossPr(int64_t nowMs) const;
-    VCMProtectionMethod* _selectedMethod;
-    VCMProtectionParameters _currentParameters;
-    int64_t _rtt;
-    float _lossPr;
-    float _bitRate;
-    float _frameRate;
-    float _keyFrameSize;
-    uint8_t _fecRateKey;
-    uint8_t _fecRateDelta;
-    int64_t _lastPrUpdateT;
-    int64_t _lastPacketPerFrameUpdateT;
-    int64_t _lastPacketPerFrameUpdateTKey;
-    rtc::ExpFilter _lossPr255;
-    VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
-    uint8_t _shortMaxLossPr255;
-    rtc::ExpFilter _packetsPerFrame;
-    rtc::ExpFilter _packetsPerFrameKey;
-    uint16_t _codecWidth;
-    uint16_t _codecHeight;
-    int _numLayers;
-};
-
-}  // namespace media_optimization
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/nack_fec_tables.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/nack_fec_tables.h
deleted file mode 100644
index b82bb1b4ba..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/nack_fec_tables.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
-
-namespace webrtc
-{
-
-// Table for adjusting FEC rate for NACK/FEC protection method
-// Table values are built as a sigmoid function, ranging from 0 to 100, based on
-// the HybridNackTH values defined in media_opt_util.h.
-const uint16_t VCMNackFecTable[100] = {
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-1,
-1,
-1,
-1,
-1,
-2,
-2,
-2,
-3,
-3,
-4,
-5,
-6,
-7,
-9,
-10,
-12,
-15,
-18,
-21,
-24,
-28,
-32,
-37,
-41,
-46,
-51,
-56,
-61,
-66,
-70,
-74,
-78,
-81,
-84,
-86,
-89,
-90,
-92,
-93,
-95,
-95,
-96,
-97,
-97,
-98,
-98,
-99,
-99,
-99,
-99,
-99,
-99,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/packet.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/packet.h
deleted file mode 100644
index d98b6f65c2..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/packet.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class VCMPacket {
-public:
-    VCMPacket();
-    VCMPacket(const uint8_t* ptr,
-              const size_t size,
-              const WebRtcRTPHeader& rtpHeader);
-    VCMPacket(const uint8_t* ptr,
-              size_t size,
-              uint16_t seqNum,
-              uint32_t timestamp,
-              bool markerBit);
-
-    void Reset();
-
-    uint8_t           payloadType;
-    uint32_t          timestamp;
-    // NTP time of the capture time in local timebase in milliseconds.
-    int64_t ntp_time_ms_;
-    uint16_t          seqNum;
-    const uint8_t*    dataPtr;
-    size_t          sizeBytes;
-    bool                    markerBit;
-
-    FrameType               frameType;
-    webrtc::VideoCodecType  codec;
-
-    bool isFirstPacket;                 // Is this first packet in a frame.
-    VCMNaluCompleteness completeNALU;   // Default is kNaluIncomplete.
-    bool insertStartCode;               // True if a start code should be inserted before this
-                                        // packet.
-    int width;
-    int height;
-    RTPVideoHeader codecSpecificHeader;
-
-protected:
-    void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
-};
-
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver_unittest.cc
deleted file mode 100644
index e5b68047de..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver_unittest.cc
+++ /dev/null
@@ -1,335 +0,0 @@
-/*  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-
-#include <list>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/receiver.h"
-#include "webrtc/modules/video_coding/main/source/test/stream_generator.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-
-namespace webrtc {
-
-class TestVCMReceiver : public ::testing::Test {
- protected:
-  enum { kDataBufferSize = 10 };
-  enum { kWidth = 640 };
-  enum { kHeight = 480 };
-
-  TestVCMReceiver()
-      : clock_(new SimulatedClock(0)),
-        timing_(clock_.get()),
-        receiver_(&timing_, clock_.get(), &event_factory_, true) {
-    stream_generator_.reset(new
-        StreamGenerator(0, 0, clock_->TimeInMilliseconds()));
-    memset(data_buffer_, 0, kDataBufferSize);
-  }
-
-  virtual void SetUp() {
-    receiver_.Reset();
-  }
-
-  int32_t InsertPacket(int index) {
-    VCMPacket packet;
-    packet.dataPtr = data_buffer_;
-    bool packet_available = stream_generator_->GetPacket(&packet, index);
-    EXPECT_TRUE(packet_available);
-    if (!packet_available)
-      return kGeneralError;  // Return here to avoid crashes below.
-    // Arbitrary width and height.
-    return receiver_.InsertPacket(packet, 640, 480);
-  }
-
-  int32_t InsertPacketAndPop(int index) {
-    VCMPacket packet;
-    packet.dataPtr = data_buffer_;
-    bool packet_available = stream_generator_->PopPacket(&packet, index);
-    EXPECT_TRUE(packet_available);
-    if (!packet_available)
-      return kGeneralError;  // Return here to avoid crashes below.
-    return receiver_.InsertPacket(packet, kWidth, kHeight);
-  }
-
-  int32_t InsertFrame(FrameType frame_type, bool complete) {
-    int num_of_packets = complete ? 1 : 2;
-    stream_generator_->GenerateFrame(
-        frame_type,
-        (frame_type != kFrameEmpty) ? num_of_packets : 0,
-        (frame_type == kFrameEmpty) ? 1 : 0,
-        clock_->TimeInMilliseconds());
-    int32_t ret = InsertPacketAndPop(0);
-    if (!complete) {
-      // Drop the second packet.
-      VCMPacket packet;
-      stream_generator_->PopPacket(&packet, 0);
-    }
-    clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
-    return ret;
-  }
-
-  bool DecodeNextFrame() {
-    int64_t render_time_ms = 0;
-    VCMEncodedFrame* frame =
-        receiver_.FrameForDecoding(0, render_time_ms, false);
-    if (!frame)
-      return false;
-    receiver_.ReleaseFrame(frame);
-    return true;
-  }
-
-  rtc::scoped_ptr<SimulatedClock> clock_;
-  VCMTiming timing_;
-  NullEventFactory event_factory_;
-  VCMReceiver receiver_;
-  rtc::scoped_ptr<StreamGenerator> stream_generator_;
-  uint8_t data_buffer_[kDataBufferSize];
-};
-
-TEST_F(TestVCMReceiver, RenderBufferSize_AllComplete) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  int num_of_frames = 10;
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ(num_of_frames * kDefaultFramePeriodMs,
-            receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, RenderBufferSize_SkipToKeyFrame) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  const int kNumOfNonDecodableFrames = 2;
-  for (int i = 0; i < kNumOfNonDecodableFrames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  const int kNumOfFrames = 10;
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  for (int i = 0; i < kNumOfFrames - 1; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ((kNumOfFrames - 1) * kDefaultFramePeriodMs,
-      receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, RenderBufferSize_NotAllComplete) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  int num_of_frames = 10;
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  num_of_frames++;
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ((num_of_frames - 1) * kDefaultFramePeriodMs,
-      receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, RenderBufferSize_NoKeyFrame) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  int num_of_frames = 10;
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  int64_t next_render_time_ms = 0;
-  VCMEncodedFrame* frame = receiver_.FrameForDecoding(10, next_render_time_ms);
-  EXPECT_TRUE(frame == NULL);
-  receiver_.ReleaseFrame(frame);
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, NonDecodableDuration_Empty) {
-  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
-  receiver_.SetNackMode(kNack, -1, -1);
-  const size_t kMaxNackListSize = 1000;
-  const int kMaxPacketAgeToNack = 1000;
-  const int kMaxNonDecodableDuration = 500;
-  const int kMinDelayMs = 500;
-  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  // Advance time until it's time to decode the key frame.
-  clock_->AdvanceTimeMilliseconds(kMinDelayMs);
-  EXPECT_TRUE(DecodeNextFrame());
-  uint16_t nack_list[kMaxNackListSize];
-  uint16_t nack_list_length = 0;
-  VCMNackStatus ret = receiver_.NackList(nack_list, kMaxNackListSize,
-                                         &nack_list_length);
-  EXPECT_EQ(kNackOk, ret);
-}
-
-TEST_F(TestVCMReceiver, NonDecodableDuration_NoKeyFrame) {
-  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
-  receiver_.SetNackMode(kNack, -1, -1);
-  const size_t kMaxNackListSize = 1000;
-  const int kMaxPacketAgeToNack = 1000;
-  const int kMaxNonDecodableDuration = 500;
-  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
-  const int kNumFrames = kDefaultFrameRate * kMaxNonDecodableDuration / 1000;
-  for (int i = 0; i < kNumFrames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  uint16_t nack_list[kMaxNackListSize];
-  uint16_t nack_list_length = 0;
-  VCMNackStatus ret = receiver_.NackList(nack_list, kMaxNackListSize,
-                                         &nack_list_length);
-  EXPECT_EQ(kNackKeyFrameRequest, ret);
-}
-
-TEST_F(TestVCMReceiver, NonDecodableDuration_OneIncomplete) {
-  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
-  receiver_.SetNackMode(kNack, -1, -1);
-  const size_t kMaxNackListSize = 1000;
-  const int kMaxPacketAgeToNack = 1000;
-  const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
-  const int kMinDelayMs = 500;
-  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
-  receiver_.SetMinReceiverDelay(kMinDelayMs);
-  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  // Insert an incomplete frame.
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  // Insert enough frames to have too long non-decodable sequence.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames;
-       ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  // Advance time until it's time to decode the key frame.
-  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
-  EXPECT_TRUE(DecodeNextFrame());
-  // Make sure we get a key frame request.
-  uint16_t nack_list[kMaxNackListSize];
-  uint16_t nack_list_length = 0;
-  VCMNackStatus ret = receiver_.NackList(nack_list, kMaxNackListSize,
-                                         &nack_list_length);
-  EXPECT_EQ(kNackKeyFrameRequest, ret);
-}
-
-TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger) {
-  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
-  receiver_.SetNackMode(kNack, -1, -1);
-  const size_t kMaxNackListSize = 1000;
-  const int kMaxPacketAgeToNack = 1000;
-  const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
-  const int kMinDelayMs = 500;
-  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
-  receiver_.SetMinReceiverDelay(kMinDelayMs);
-  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  // Insert an incomplete frame.
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  // Insert all but one frame to not trigger a key frame request due to
-  // too long duration of non-decodable frames.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames - 1;
-       ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  // Advance time until it's time to decode the key frame.
-  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
-  EXPECT_TRUE(DecodeNextFrame());
-  // Make sure we don't get a key frame request since we haven't generated
-  // enough frames.
-  uint16_t nack_list[kMaxNackListSize];
-  uint16_t nack_list_length = 0;
-  VCMNackStatus ret = receiver_.NackList(nack_list, kMaxNackListSize,
-                                         &nack_list_length);
-  EXPECT_EQ(kNackOk, ret);
-}
-
-TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger2) {
-  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
-  receiver_.SetNackMode(kNack, -1, -1);
-  const size_t kMaxNackListSize = 1000;
-  const int kMaxPacketAgeToNack = 1000;
-  const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
-  const int kMinDelayMs = 500;
-  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
-  receiver_.SetMinReceiverDelay(kMinDelayMs);
-  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  // Insert enough frames to have too long non-decodable sequence, except that
-  // we don't have any losses.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames;
-       ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  // Insert an incomplete frame.
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  // Advance time until it's time to decode the key frame.
-  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
-  EXPECT_TRUE(DecodeNextFrame());
-  // Make sure we don't get a key frame request since the non-decodable duration
-  // is only one frame.
-  uint16_t nack_list[kMaxNackListSize];
-  uint16_t nack_list_length = 0;
-  VCMNackStatus ret = receiver_.NackList(nack_list, kMaxNackListSize,
-                                         &nack_list_length);
-  EXPECT_EQ(kNackOk, ret);
-}
-
-TEST_F(TestVCMReceiver, NonDecodableDuration_KeyFrameAfterIncompleteFrames) {
-  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
-  receiver_.SetNackMode(kNack, -1, -1);
-  const size_t kMaxNackListSize = 1000;
-  const int kMaxPacketAgeToNack = 1000;
-  const int kMaxNonDecodableDuration = 500;
-  const int kMaxNonDecodableDurationFrames = (kDefaultFrameRate *
-      kMaxNonDecodableDuration + 500) / 1000;
-  const int kMinDelayMs = 500;
-  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
-      kMaxNonDecodableDuration);
-  receiver_.SetMinReceiverDelay(kMinDelayMs);
-  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  // Insert an incomplete frame.
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  // Insert enough frames to have too long non-decodable sequence.
-  for (int i = 0; i < kMaxNonDecodableDurationFrames;
-       ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  // Advance time until it's time to decode the key frame.
-  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
-      key_frame_inserted);
-  EXPECT_TRUE(DecodeNextFrame());
-  // Make sure we don't get a key frame request since we have a key frame
-  // in the list.
-  uint16_t nack_list[kMaxNackListSize];
-  uint16_t nack_list_length = 0;
-  VCMNackStatus ret = receiver_.NackList(nack_list, kMaxNackListSize,
-                                         &nack_list_length);
-  EXPECT_EQ(kNackOk, ret);
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/rtt_filter.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/rtt_filter.h
deleted file mode 100644
index 9e14a1ab39..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/rtt_filter.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-class VCMRttFilter
-{
-public:
-    VCMRttFilter();
-
-    VCMRttFilter& operator=(const VCMRttFilter& rhs);
-
-    // Resets the filter.
-    void Reset();
-    // Updates the filter with a new sample.
-    void Update(int64_t rttMs);
-    // A getter function for the current RTT level in ms.
-    int64_t RttMs() const;
-
-private:
-    // The size of the drift and jump memory buffers
-    // and thus also the detection threshold for these
-    // detectors in number of samples.
-    enum { kMaxDriftJumpCount = 5 };
-    // Detects RTT jumps by comparing the difference between
-    // samples and average to the standard deviation.
-    // Returns true if the long time statistics should be updated
-    // and false otherwise
-    bool JumpDetection(int64_t rttMs);
-    // Detects RTT drifts by comparing the difference between
-    // max and average to the standard deviation.
-    // Returns true if the long time statistics should be updated
-    // and false otherwise
-    bool DriftDetection(int64_t rttMs);
-    // Computes the short time average and maximum of the vector buf.
-    void ShortRttFilter(int64_t* buf, uint32_t length);
-
-    bool                  _gotNonZeroUpdate;
-    double                _avgRtt;
-    double                _varRtt;
-    int64_t         _maxRtt;
-    uint32_t        _filtFactCount;
-    const uint32_t  _filtFactMax;
-    const double          _jumpStdDevs;
-    const double          _driftStdDevs;
-    int32_t         _jumpCount;
-    int32_t         _driftCount;
-    const int32_t   _detectThreshold;
-    int64_t         _jumpBuf[kMaxDriftJumpCount];
-    int64_t         _driftBuf[kMaxDriftJumpCount];
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timestamp_map.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timestamp_map.cc
deleted file mode 100644
index f3806bb87f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timestamp_map.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <stdlib.h>
-#include "webrtc/modules/video_coding/main/source/timestamp_map.h"
-
-namespace webrtc {
-
-// Constructor. Optional parameter specifies maximum number of
-// coexisting timers.
-VCMTimestampMap::VCMTimestampMap(int32_t length):
-    _nextAddIx(0),
-    _nextPopIx(0)
-{
-    if (length <= 0)
-    {
-        // default
-        length = 10;
-    }
-
-    _map = new VCMTimestampDataTuple[length];
-    _length = length;
-}
-
-// Destructor.
-VCMTimestampMap::~VCMTimestampMap()
-{
-    delete [] _map;
-}
-
-// Empty the list of timers.
-void
-VCMTimestampMap::Reset()
-{
-    _nextAddIx = 0;
-    _nextPopIx = 0;
-}
-
-int32_t
-VCMTimestampMap::Add(uint32_t timestamp, void* data)
-{
-    _map[_nextAddIx].timestamp = timestamp;
-    _map[_nextAddIx].data = data;
-    _nextAddIx = (_nextAddIx + 1) % _length;
-
-    if (_nextAddIx == _nextPopIx)
-    {
-        // Circular list full; forget oldest entry
-        _nextPopIx = (_nextPopIx + 1) % _length;
-        return -1;
-    }
-    return 0;
-}
-
-void*
-VCMTimestampMap::Pop(uint32_t timestamp)
-{
-    while (!IsEmpty())
-    {
-        if (_map[_nextPopIx].timestamp == timestamp)
-        {
-            // found start time for this timestamp
-            void* data = _map[_nextPopIx].data;
-            _map[_nextPopIx].data = NULL;
-            _nextPopIx = (_nextPopIx + 1) % _length;
-            return data;
-        }
-        else if (_map[_nextPopIx].timestamp > timestamp)
-        {
-            // the timestamp we are looking for is not in the list
-            assert(_nextPopIx < _length && _nextPopIx >= 0);
-            return NULL;
-        }
-
-        // not in this position, check next (and forget this position)
-        _nextPopIx = (_nextPopIx + 1) % _length;
-    }
-
-    // could not find matching timestamp in list
-    assert(_nextPopIx < _length && _nextPopIx >= 0);
-    return NULL;
-}
-
-// Check if no timers are currently running
-bool
-VCMTimestampMap::IsEmpty() const
-{
-    return (_nextAddIx == _nextPopIx);
-}
-
-}
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timestamp_map.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timestamp_map.h
deleted file mode 100644
index 14e06290ff..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timestamp_map.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-struct VCMTimestampDataTuple
-{
-    uint32_t    timestamp;
-    void*             data;
-};
-
-class VCMTimestampMap
-{
-public:
-    // Constructor. Optional parameter specifies maximum number of
-    // timestamps in map.
-    VCMTimestampMap(const int32_t length = 10);
-
-    // Destructor.
-    ~VCMTimestampMap();
-
-    // Empty the map
-    void Reset();
-
-    int32_t Add(uint32_t timestamp, void*  data);
-    void* Pop(uint32_t timestamp);
-
-private:
-    bool IsEmpty() const;
-
-    VCMTimestampDataTuple* _map;
-    int32_t                   _nextAddIx;
-    int32_t                   _nextPopIx;
-    int32_t                   _length;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.cc
deleted file mode 100644
index fbfa749dcd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.cc
+++ /dev/null
@@ -1,390 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-
-namespace webrtc {
-namespace vcm {
-
-int64_t
-VCMProcessTimer::Period() const {
-    return _periodMs;
-}
-
-int64_t
-VCMProcessTimer::TimeUntilProcess() const {
-    const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
-    const int64_t time_until_process = _periodMs - time_since_process;
-    return std::max<int64_t>(time_until_process, 0);
-}
-
-void
-VCMProcessTimer::Processed() {
-    _latestMs = _clock->TimeInMilliseconds();
-}
-}  // namespace vcm
-
-namespace {
-// This wrapper provides a way to modify the callback without the need to expose
-// a register method all the way down to the function calling it.
-class EncodedImageCallbackWrapper : public EncodedImageCallback {
- public:
-  EncodedImageCallbackWrapper()
-      : cs_(CriticalSectionWrapper::CreateCriticalSection()), callback_(NULL) {}
-
-  virtual ~EncodedImageCallbackWrapper() {}
-
-  void Register(EncodedImageCallback* callback) {
-    CriticalSectionScoped cs(cs_.get());
-    callback_ = callback;
-  }
-
-  // TODO(andresp): Change to void as return value is ignored.
-  virtual int32_t Encoded(const EncodedImage& encoded_image,
-                          const CodecSpecificInfo* codec_specific_info,
-                          const RTPFragmentationHeader* fragmentation) {
-    CriticalSectionScoped cs(cs_.get());
-    if (callback_)
-      return callback_->Encoded(
-          encoded_image, codec_specific_info, fragmentation);
-    return 0;
-  }
-
- private:
-  rtc::scoped_ptr<CriticalSectionWrapper> cs_;
-  EncodedImageCallback* callback_ GUARDED_BY(cs_);
-};
-
-class VideoCodingModuleImpl : public VideoCodingModule {
- public:
-  VideoCodingModuleImpl(Clock* clock,
-                        EventFactory* event_factory,
-                        bool owns_event_factory,
-                        VideoEncoderRateObserver* encoder_rate_observer)
-      : VideoCodingModule(),
-        sender_(new vcm::VideoSender(clock,
-                                     &post_encode_callback_,
-                                     encoder_rate_observer)),
-        receiver_(new vcm::VideoReceiver(clock, event_factory)),
-        own_event_factory_(owns_event_factory ? event_factory : NULL) {}
-
-  virtual ~VideoCodingModuleImpl() {
-    sender_.reset();
-    receiver_.reset();
-    own_event_factory_.reset();
-  }
-
-  int64_t TimeUntilNextProcess() override {
-    int64_t sender_time = sender_->TimeUntilNextProcess();
-    int64_t receiver_time = receiver_->TimeUntilNextProcess();
-    assert(sender_time >= 0);
-    assert(receiver_time >= 0);
-    return VCM_MIN(sender_time, receiver_time);
-  }
-
-  int32_t Process() override {
-    int32_t sender_return = sender_->Process();
-    int32_t receiver_return = receiver_->Process();
-    if (sender_return != VCM_OK)
-      return sender_return;
-    return receiver_return;
-  }
-
-  int32_t InitializeSender() override { return sender_->InitializeSender(); }
-
-  int32_t RegisterSendCodec(const VideoCodec* sendCodec,
-                            uint32_t numberOfCores,
-                            uint32_t maxPayloadSize) override {
-    return sender_->RegisterSendCodec(sendCodec, numberOfCores, maxPayloadSize);
-  }
-
-  const VideoCodec& GetSendCodec() const override {
-    return sender_->GetSendCodec();
-  }
-
-  // DEPRECATED.
-  int32_t SendCodec(VideoCodec* currentSendCodec) const override {
-    return sender_->SendCodecBlocking(currentSendCodec);
-  }
-
-  // DEPRECATED.
-  VideoCodecType SendCodec() const override {
-    return sender_->SendCodecBlocking();
-  }
-
-  int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                  uint8_t payloadType,
-                                  bool internalSource) override {
-    return sender_->RegisterExternalEncoder(
-        externalEncoder, payloadType, internalSource);
-  }
-
-  int32_t CodecConfigParameters(uint8_t* buffer, int32_t size) override {
-    return sender_->CodecConfigParameters(buffer, size);
-  }
-
-  int Bitrate(unsigned int* bitrate) const override {
-    return sender_->Bitrate(bitrate);
-  }
-
-  int FrameRate(unsigned int* framerate) const override {
-    return sender_->FrameRate(framerate);
-  }
-
-  int32_t SetChannelParameters(uint32_t target_bitrate,  // bits/s.
-                               uint8_t lossRate,
-                               int64_t rtt) override {
-    return sender_->SetChannelParameters(target_bitrate, lossRate, rtt);
-  }
-
-  int32_t RegisterTransportCallback(
-      VCMPacketizationCallback* transport) override {
-    return sender_->RegisterTransportCallback(transport);
-  }
-
-  int32_t RegisterSendStatisticsCallback(
-      VCMSendStatisticsCallback* sendStats) override {
-    return sender_->RegisterSendStatisticsCallback(sendStats);
-  }
-
-  int32_t RegisterVideoQMCallback(
-      VCMQMSettingsCallback* videoQMSettings) override {
-    return sender_->RegisterVideoQMCallback(videoQMSettings);
-  }
-
-  int32_t RegisterProtectionCallback(
-      VCMProtectionCallback* protection) override {
-    return sender_->RegisterProtectionCallback(protection);
-  }
-
-  int32_t SetVideoProtection(VCMVideoProtection videoProtection,
-                             bool enable) override {
-    sender_->SetVideoProtection(enable, videoProtection);
-    return receiver_->SetVideoProtection(videoProtection, enable);
-  }
-
-  int32_t AddVideoFrame(const I420VideoFrame& videoFrame,
-                        const VideoContentMetrics* contentMetrics,
-                        const CodecSpecificInfo* codecSpecificInfo) override {
-    return sender_->AddVideoFrame(
-        videoFrame, contentMetrics, codecSpecificInfo);
-  }
-
-  int32_t IntraFrameRequest(int stream_index) override {
-    return sender_->IntraFrameRequest(stream_index);
-  }
-
-  int32_t EnableFrameDropper(bool enable) override {
-    return sender_->EnableFrameDropper(enable);
-  }
-
-  int32_t SentFrameCount(VCMFrameCount& frameCount) const override {
-    return sender_->SentFrameCount(&frameCount);
-  }
-
-  int StartDebugRecording(const char* file_name_utf8) override {
-    return sender_->StartDebugRecording(file_name_utf8);
-  }
-
-  int StopDebugRecording() override {
-    sender_->StopDebugRecording();
-    return VCM_OK;
-  }
-
-  void SuspendBelowMinBitrate() override {
-    return sender_->SuspendBelowMinBitrate();
-  }
-
-  bool VideoSuspended() const override { return sender_->VideoSuspended(); }
-
-  int32_t InitializeReceiver() override {
-    return receiver_->InitializeReceiver();
-  }
-
-  int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
-                               int32_t numberOfCores,
-                               bool requireKeyFrame) override {
-    return receiver_->RegisterReceiveCodec(
-        receiveCodec, numberOfCores, requireKeyFrame);
-  }
-
-  int32_t RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                  uint8_t payloadType,
-                                  bool internalRenderTiming) override {
-    return receiver_->RegisterExternalDecoder(
-        externalDecoder, payloadType, internalRenderTiming);
-  }
-
-  int32_t RegisterReceiveCallback(
-      VCMReceiveCallback* receiveCallback) override {
-    return receiver_->RegisterReceiveCallback(receiveCallback);
-  }
-
-  int32_t RegisterReceiveStatisticsCallback(
-      VCMReceiveStatisticsCallback* receiveStats) override {
-    return receiver_->RegisterReceiveStatisticsCallback(receiveStats);
-  }
-
-  int32_t RegisterDecoderTimingCallback(
-      VCMDecoderTimingCallback* decoderTiming) override {
-    return receiver_->RegisterDecoderTimingCallback(decoderTiming);
-  }
-
-  int32_t RegisterFrameTypeCallback(
-      VCMFrameTypeCallback* frameTypeCallback) override {
-    return receiver_->RegisterFrameTypeCallback(frameTypeCallback);
-  }
-
-  int32_t RegisterPacketRequestCallback(
-      VCMPacketRequestCallback* callback) override {
-    return receiver_->RegisterPacketRequestCallback(callback);
-  }
-
-  virtual int32_t RegisterReceiveStateCallback(
-      VCMReceiveStateCallback* callback) override {
-    return receiver_->RegisterReceiveStateCallback(callback);
-  }
-
-  int RegisterRenderBufferSizeCallback(
-      VCMRenderBufferSizeCallback* callback) override {
-    return receiver_->RegisterRenderBufferSizeCallback(callback);
-  }
-
-  int32_t Decode(uint16_t maxWaitTimeMs) override {
-    return receiver_->Decode(maxWaitTimeMs);
-  }
-
-  int32_t ResetDecoder() override { return receiver_->ResetDecoder(); }
-
-  int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const override {
-    return receiver_->ReceiveCodec(currentReceiveCodec);
-  }
-
-  VideoCodecType ReceiveCodec() const override {
-    return receiver_->ReceiveCodec();
-  }
-
-  int32_t IncomingPacket(const uint8_t* incomingPayload,
-                         size_t payloadLength,
-                         const WebRtcRTPHeader& rtpInfo) override {
-    return receiver_->IncomingPacket(incomingPayload, payloadLength, rtpInfo);
-  }
-
-  int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) override {
-    return receiver_->SetMinimumPlayoutDelay(minPlayoutDelayMs);
-  }
-
-  int32_t SetRenderDelay(uint32_t timeMS) override {
-    return receiver_->SetRenderDelay(timeMS);
-  }
-
-  int32_t Delay() const override { return receiver_->Delay(); }
-
-  uint32_t DiscardedPackets() const override {
-    return receiver_->DiscardedPackets();
-  }
-
-  int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
-                                VCMDecodeErrorMode errorMode) override {
-    return receiver_->SetReceiverRobustnessMode(robustnessMode, errorMode);
-  }
-
-  void SetNackSettings(size_t max_nack_list_size,
-                       int max_packet_age_to_nack,
-                       int max_incomplete_time_ms) override {
-    return receiver_->SetNackSettings(
-        max_nack_list_size, max_packet_age_to_nack, max_incomplete_time_ms);
-  }
-
-  void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) override {
-    return receiver_->SetDecodeErrorMode(decode_error_mode);
-  }
-
-  int SetMinReceiverDelay(int desired_delay_ms) override {
-    return receiver_->SetMinReceiverDelay(desired_delay_ms);
-  }
-
-  virtual void SetCPULoadState(CPULoadState state) override {
-    return sender_->SetCPULoadState(state);
-  }
-
-  int32_t SetReceiveChannelParameters(int64_t rtt) override {
-    return receiver_->SetReceiveChannelParameters(rtt);
-  }
-
-  void RegisterPreDecodeImageCallback(EncodedImageCallback* observer) override {
-    receiver_->RegisterPreDecodeImageCallback(observer);
-  }
-
-  void RegisterPostEncodeImageCallback(
-      EncodedImageCallback* observer) override {
-    post_encode_callback_.Register(observer);
-  }
-
-  void TriggerDecoderShutdown() override {
-    receiver_->TriggerDecoderShutdown();
-  }
-
- private:
-  EncodedImageCallbackWrapper post_encode_callback_;
-  // TODO(tommi): Change sender_ and receiver_ to be non pointers
-  // (construction is 1 alloc instead of 3).
-  rtc::scoped_ptr<vcm::VideoSender> sender_;
-  rtc::scoped_ptr<vcm::VideoReceiver> receiver_;
-  rtc::scoped_ptr<EventFactory> own_event_factory_;
-};
-}  // namespace
-
-uint8_t VideoCodingModule::NumberOfCodecs() {
-  return VCMCodecDataBase::NumberOfCodecs();
-}
-
-int32_t VideoCodingModule::Codec(uint8_t listId, VideoCodec* codec) {
-  if (codec == NULL) {
-    return VCM_PARAMETER_ERROR;
-  }
-  return VCMCodecDataBase::Codec(listId, codec) ? 0 : -1;
-}
-
-int32_t VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec) {
-  if (codec == NULL) {
-    return VCM_PARAMETER_ERROR;
-  }
-  return VCMCodecDataBase::Codec(codecType, codec) ? 0 : -1;
-}
-
-VideoCodingModule* VideoCodingModule::Create(
-    VideoEncoderRateObserver* encoder_rate_observer) {
-  return new VideoCodingModuleImpl(Clock::GetRealTimeClock(),
-                                   new EventFactoryImpl, true,
-                                   encoder_rate_observer);
-}
-
-VideoCodingModule* VideoCodingModule::Create(
-    Clock* clock,
-    EventFactory* event_factory) {
-  assert(clock);
-  assert(event_factory);
-  return new VideoCodingModuleImpl(clock, event_factory, false, nullptr);
-}
-
-void VideoCodingModule::Destroy(VideoCodingModule* module) {
-  if (module != NULL) {
-    delete static_cast<VideoCodingModuleImpl*>(module);
-  }
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_source.h b/media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_source.h
deleted file mode 100644
index 05deb4a39b..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_source.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/typedefs.h"
-
-#include <string>
-
-enum VideoSize
-    {
-        kUndefined,
-        kSQCIF,     // 128*96       = 12 288
-        kQQVGA,     // 160*120      = 19 200
-        kQCIF,      // 176*144      = 25 344
-        kCGA,       // 320*200      = 64 000
-        kQVGA,      // 320*240      = 76 800
-        kSIF,       // 352*240      = 84 480
-        kWQVGA,     // 400*240      = 96 000
-        kCIF,       // 352*288      = 101 376
-        kW288p,     // 512*288      = 147 456 (WCIF)
-        k448p,      // 576*448      = 281 088
-        kVGA,       // 640*480      = 307 200
-        k432p,      // 720*432      = 311 040
-        kW432p,     // 768*432      = 331 776
-        k4SIF,      // 704*480      = 337 920
-        kW448p,     // 768*448      = 344 064
-        kNTSC,		// 720*480      = 345 600
-        kFW448p,    // 800*448      = 358 400
-        kWVGA,      // 800*480      = 384 000
-        k4CIF,      // 704*576      = 405 504
-        kSVGA,      // 800*600      = 480 000
-        kW544p,     // 960*544      = 522 240
-        kW576p,     // 1024*576     = 589 824 (W4CIF)
-        kHD,        // 960*720      = 691 200
-        kXGA,       // 1024*768     = 786 432
-        kWHD,       // 1280*720     = 921 600
-        kFullHD,   // 1440*1080    = 1 555 200
-        kWFullHD,  // 1920*1080    = 2 073 600
-
-        kNumberOfVideoSizes
-    };
-
-
-class VideoSource
-{
-public:
-  VideoSource();
-  VideoSource(std::string fileName, VideoSize size, float frameRate, webrtc::VideoType type = webrtc::kI420);
-  VideoSource(std::string fileName, uint16_t width, uint16_t height,
-      float frameRate = 30, webrtc::VideoType type = webrtc::kI420);
-
-    std::string GetFileName() const { return _fileName; }
-    uint16_t  GetWidth() const { return _width; }
-    uint16_t GetHeight() const { return _height; }
-    webrtc::VideoType GetType() const { return _type; }
-    float GetFrameRate() const { return _frameRate; }
-    int GetWidthHeight( VideoSize size);
-
-    // Returns the filename with the path (including the leading slash) removed.
-    std::string GetName() const;
-
-    size_t GetFrameLength() const;
-
-private:
-    std::string         _fileName;
-    uint16_t      _width;
-    uint16_t      _height;
-    webrtc::VideoType   _type;
-    float               _frameRate;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/media_opt_util.cc b/media/webrtc/trunk/webrtc/modules/video_coding/media_opt_util.cc
new file mode 100644
index 0000000000..d57e9c8dd2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/media_opt_util.cc
@@ -0,0 +1,682 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/media_opt_util.h"
+
+#include <float.h>
+#include <limits.h>
+#include <math.h>
+
+#include <algorithm>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/fec_tables_xor.h"
+#include "webrtc/modules/video_coding/nack_fec_tables.h"
+
+namespace webrtc {
+// Max value of loss rates in off-line model
+static const int kPacketLossMax = 129;
+
+namespace media_optimization {
+
+VCMProtectionMethod::VCMProtectionMethod()
+    : _effectivePacketLoss(0),
+      _protectionFactorK(0),
+      _protectionFactorD(0),
+      _scaleProtKey(2.0f),
+      _maxPayloadSize(1460),
+      _qmRobustness(new VCMQmRobustness()),
+      _useUepProtectionK(false),
+      _useUepProtectionD(true),
+      _corrFecCost(1.0),
+      _type(kNone) {}
+
+VCMProtectionMethod::~VCMProtectionMethod() {
+  delete _qmRobustness;
+}
+void VCMProtectionMethod::UpdateContentMetrics(
+    const VideoContentMetrics* contentMetrics) {
+  _qmRobustness->UpdateContent(contentMetrics);
+}
+
+VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
+                                   int64_t highRttNackThresholdMs)
+    : VCMFecMethod(),
+      _lowRttNackMs(lowRttNackThresholdMs),
+      _highRttNackMs(highRttNackThresholdMs),
+      _maxFramesFec(1) {
+  assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
+  assert(highRttNackThresholdMs == -1 ||
+         lowRttNackThresholdMs <= highRttNackThresholdMs);
+  assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
+  _type = kNackFec;
+}
+
+VCMNackFecMethod::~VCMNackFecMethod() {
+  //
+}
+bool VCMNackFecMethod::ProtectionFactor(
+    const VCMProtectionParameters* parameters) {
+  // Hybrid Nack FEC has three operational modes:
+  // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
+  //    (_protectionFactorD) to zero. -1 means no FEC.
+  // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
+  //    -1 means always allow NACK.
+  // 3. Medium RTT values - Hybrid mode: We will only nack the
+  //    residual following the decoding of the FEC (refer to JB logic). FEC
+  //    delta protection factor will be adjusted based on the RTT.
+
+  // Otherwise: we count on FEC; if the RTT is below a threshold, then we
+  // nack the residual, based on a decision made in the JB.
+
+  // Compute the protection factors
+  VCMFecMethod::ProtectionFactor(parameters);
+  if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs) {
+    _protectionFactorD = 0;
+    VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+
+    // When in Hybrid mode (RTT range), adjust FEC rates based on the
+    // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
+  } else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs) {
+    // TODO(mikhal): Disabling adjustment temporarily.
+    // uint16_t rttIndex = (uint16_t) parameters->rtt;
+    float adjustRtt = 1.0f;  // (float)VCMNackFecTable[rttIndex] / 100.0f;
+
+    // Adjust FEC with NACK on (for delta frame only)
+    // table depends on RTT relative to rttMax (NACK Threshold)
+    _protectionFactorD = static_cast<uint8_t>(
+        adjustRtt * static_cast<float>(_protectionFactorD));
+    // update FEC rates after applying adjustment
+    VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+  }
+
+  return true;
+}
+
+int VCMNackFecMethod::ComputeMaxFramesFec(
+    const VCMProtectionParameters* parameters) {
+  if (parameters->numLayers > 2) {
+    // For more than 2 temporal layers we will only have FEC on the base layer,
+    // and the base layers will be pretty far apart. Therefore we force one
+    // frame FEC.
+    return 1;
+  }
+  // We set the max number of frames to base the FEC on so that on average
+  // we will have complete frames in one RTT. Note that this is an upper
+  // bound, and that the actual number of frames used for FEC is decided by the
+  // RTP module based on the actual number of packets and the protection factor.
+  float base_layer_framerate =
+      parameters->frameRate /
+      static_cast<float>(1 << (parameters->numLayers - 1));
+  int max_frames_fec = std::max(
+      static_cast<int>(2.0f * base_layer_framerate * parameters->rtt / 1000.0f +
+                       0.5f),
+      1);
+  // |kUpperLimitFramesFec| is the upper limit on how many frames we
+  // allow any FEC to be based on.
+  if (max_frames_fec > kUpperLimitFramesFec) {
+    max_frames_fec = kUpperLimitFramesFec;
+  }
+  return max_frames_fec;
+}
+
+int VCMNackFecMethod::MaxFramesFec() const {
+  return _maxFramesFec;
+}
+
+bool VCMNackFecMethod::BitRateTooLowForFec(
+    const VCMProtectionParameters* parameters) {
+  // Bitrate below which we turn off FEC, regardless of reported packet loss.
+  // The condition should depend on resolution and content. For now, use
+  // threshold on bytes per frame, with some effect for the frame size.
+  // The condition for turning off FEC is also based on other factors,
+  // such as |_numLayers|, |_maxFramesFec|, and |_rtt|.
+  int estimate_bytes_per_frame = 1000 * BitsPerFrame(parameters) / 8;
+  int max_bytes_per_frame = kMaxBytesPerFrameForFec;
+  int num_pixels = parameters->codecWidth * parameters->codecHeight;
+  if (num_pixels <= 352 * 288) {
+    max_bytes_per_frame = kMaxBytesPerFrameForFecLow;
+  } else if (num_pixels > 640 * 480) {
+    max_bytes_per_frame = kMaxBytesPerFrameForFecHigh;
+  }
+  // TODO(marpan): add condition based on maximum frames used for FEC,
+  // and expand condition based on frame size.
+  // Max round trip time threshold in ms.
+  const int64_t kMaxRttTurnOffFec = 200;
+  if (estimate_bytes_per_frame < max_bytes_per_frame &&
+      parameters->numLayers < 3 && parameters->rtt < kMaxRttTurnOffFec) {
+    return true;
+  }
+  return false;
+}
+
+bool VCMNackFecMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameters) {
+  // Set the effective packet loss for encoder (based on FEC code).
+  // Compute the effective packet loss and residual packet loss due to FEC.
+  VCMFecMethod::EffectivePacketLoss(parameters);
+  return true;
+}
+
+bool VCMNackFecMethod::UpdateParameters(
+    const VCMProtectionParameters* parameters) {
+  ProtectionFactor(parameters);
+  EffectivePacketLoss(parameters);
+  _maxFramesFec = ComputeMaxFramesFec(parameters);
+  if (BitRateTooLowForFec(parameters)) {
+    _protectionFactorK = 0;
+    _protectionFactorD = 0;
+  }
+
+  // Protection/fec rates obtained above are defined relative to total number
+  // of packets (total rate: source + fec) FEC in RTP module assumes
+  // protection factor is defined relative to source number of packets so we
+  // should convert the factor to reduce mismatch between mediaOpt's rate and
+  // the actual one
+  _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
+  _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
+
+  return true;
+}
+
+VCMNackMethod::VCMNackMethod() : VCMProtectionMethod() {
+  _type = kNack;
+}
+
+VCMNackMethod::~VCMNackMethod() {
+  //
+}
+
+bool VCMNackMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameter) {
+  // Effective Packet Loss, NA in current version.
+  _effectivePacketLoss = 0;
+  return true;
+}
+
+bool VCMNackMethod::UpdateParameters(
+    const VCMProtectionParameters* parameters) {
+  // Compute the effective packet loss
+  EffectivePacketLoss(parameters);
+
+  // nackCost  = (bitRate - nackCost) * (lossPr)
+  return true;
+}
+
+VCMFecMethod::VCMFecMethod() : VCMProtectionMethod() {
+  _type = kFec;
+}
+VCMFecMethod::~VCMFecMethod() {
+  //
+}
+
+uint8_t VCMFecMethod::BoostCodeRateKey(uint8_t packetFrameDelta,
+                                       uint8_t packetFrameKey) const {
+  uint8_t boostRateKey = 2;
+  // Default: ratio scales the FEC protection up for I frames
+  uint8_t ratio = 1;
+
+  if (packetFrameDelta > 0) {
+    ratio = (int8_t)(packetFrameKey / packetFrameDelta);
+  }
+  ratio = VCM_MAX(boostRateKey, ratio);
+
+  return ratio;
+}
+
+uint8_t VCMFecMethod::ConvertFECRate(uint8_t codeRateRTP) const {
+  return static_cast<uint8_t>(VCM_MIN(
+      255,
+      (0.5 + 255.0 * codeRateRTP / static_cast<float>(255 - codeRateRTP))));
+}
+
+// Update FEC with protectionFactorD
+void VCMFecMethod::UpdateProtectionFactorD(uint8_t protectionFactorD) {
+  _protectionFactorD = protectionFactorD;
+}
+
+// Update FEC with protectionFactorK
+void VCMFecMethod::UpdateProtectionFactorK(uint8_t protectionFactorK) {
+  _protectionFactorK = protectionFactorK;
+}
+
+bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
+  // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
+
+  // No protection if (filtered) packetLoss is 0
+  uint8_t packetLoss = (uint8_t)(255 * parameters->lossPr);
+  if (packetLoss == 0) {
+    _protectionFactorK = 0;
+    _protectionFactorD = 0;
+    return true;
+  }
+
+  // Parameters for FEC setting:
+  // first partition size, thresholds, table pars, spatial resoln fac.
+
+  // First partition protection: ~ 20%
+  uint8_t firstPartitionProt = (uint8_t)(255 * 0.20);
+
+  // Minimum protection level needed to generate one FEC packet for one
+  // source packet/frame (in RTP sender)
+  uint8_t minProtLevelFec = 85;
+
+  // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
+  // above which we allocate protection to cover at least first partition.
+  uint8_t lossThr = 0;
+  uint8_t packetNumThr = 1;
+
+  // Parameters for range of rate index of table.
+  const uint8_t ratePar1 = 5;
+  const uint8_t ratePar2 = 49;
+
+  // Spatial resolution size, relative to a reference size.
+  float spatialSizeToRef =
+      static_cast<float>(parameters->codecWidth * parameters->codecHeight) /
+      (static_cast<float>(704 * 576));
+  // resolnFac: This parameter will generally increase/decrease the FEC rate
+  // (for fixed bitRate and packetLoss) based on system size.
+  // Use a smaller exponent (< 1) to control/soften system size effect.
+  const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
+
+  const int bitRatePerFrame = BitsPerFrame(parameters);
+
+  // Average number of packets per frame (source and fec):
+  const uint8_t avgTotPackets =
+      1 + (uint8_t)(static_cast<float>(bitRatePerFrame) * 1000.0 /
+                        static_cast<float>(8.0 * _maxPayloadSize) +
+                    0.5);
+
+  // FEC rate parameters: for P and I frame
+  uint8_t codeRateDelta = 0;
+  uint8_t codeRateKey = 0;
+
+  // Get index for table: the FEC protection depends on an effective rate.
+  // The range on the rate index corresponds to rates (bps)
+  // from ~200k to ~8000k, for 30fps
+  const uint16_t effRateFecTable =
+      static_cast<uint16_t>(resolnFac * bitRatePerFrame);
+  uint8_t rateIndexTable = (uint8_t)VCM_MAX(
+      VCM_MIN((effRateFecTable - ratePar1) / ratePar1, ratePar2), 0);
+
+  // Restrict packet loss range to 50:
+  // current tables defined only up to 50%
+  if (packetLoss >= kPacketLossMax) {
+    packetLoss = kPacketLossMax - 1;
+  }
+  uint16_t indexTable = rateIndexTable * kPacketLossMax + packetLoss;
+
+  // Check on table index
+  assert(indexTable < kSizeCodeRateXORTable);
+
+  // Protection factor for P frame
+  codeRateDelta = kCodeRateXORTable[indexTable];
+
+  if (packetLoss > lossThr && avgTotPackets > packetNumThr) {
+    // Set a minimum based on first partition size.
+    if (codeRateDelta < firstPartitionProt) {
+      codeRateDelta = firstPartitionProt;
+    }
+  }
+
+  // Check limit on amount of protection for P frame; 50% is max.
+  if (codeRateDelta >= kPacketLossMax) {
+    codeRateDelta = kPacketLossMax - 1;
+  }
+
+  float adjustFec = 1.0f;
+  // Avoid additional adjustments when layers are active.
+  // TODO(mikhal/marco): Update adjusmtent based on layer info.
+  if (parameters->numLayers == 1) {
+    adjustFec = _qmRobustness->AdjustFecFactor(
+        codeRateDelta, parameters->bitRate, parameters->frameRate,
+        parameters->rtt, packetLoss);
+  }
+
+  codeRateDelta = static_cast<uint8_t>(codeRateDelta * adjustFec);
+
+  // For Key frame:
+  // Effectively at a higher rate, so we scale/boost the rate
+  // The boost factor may depend on several factors: ratio of packet
+  // number of I to P frames, how much protection placed on P frames, etc.
+  const uint8_t packetFrameDelta = (uint8_t)(0.5 + parameters->packetsPerFrame);
+  const uint8_t packetFrameKey =
+      (uint8_t)(0.5 + parameters->packetsPerFrameKey);
+  const uint8_t boostKey = BoostCodeRateKey(packetFrameDelta, packetFrameKey);
+
+  rateIndexTable = (uint8_t)VCM_MAX(
+      VCM_MIN(1 + (boostKey * effRateFecTable - ratePar1) / ratePar1, ratePar2),
+      0);
+  uint16_t indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
+
+  indexTableKey = VCM_MIN(indexTableKey, kSizeCodeRateXORTable);
+
+  // Check on table index
+  assert(indexTableKey < kSizeCodeRateXORTable);
+
+  // Protection factor for I frame
+  codeRateKey = kCodeRateXORTable[indexTableKey];
+
+  // Boosting for Key frame.
+  int boostKeyProt = _scaleProtKey * codeRateDelta;
+  if (boostKeyProt >= kPacketLossMax) {
+    boostKeyProt = kPacketLossMax - 1;
+  }
+
+  // Make sure I frame protection is at least larger than P frame protection,
+  // and at least as high as filtered packet loss.
+  codeRateKey = static_cast<uint8_t>(
+      VCM_MAX(packetLoss, VCM_MAX(boostKeyProt, codeRateKey)));
+
+  // Check limit on amount of protection for I frame: 50% is max.
+  if (codeRateKey >= kPacketLossMax) {
+    codeRateKey = kPacketLossMax - 1;
+  }
+
+  _protectionFactorK = codeRateKey;
+  _protectionFactorD = codeRateDelta;
+
+  // Generally there is a rate mis-match between the FEC cost estimated
+  // in mediaOpt and the actual FEC cost sent out in RTP module.
+  // This is more significant at low rates (small # of source packets), where
+  // the granularity of the FEC decreases. In this case, non-zero protection
+  // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
+  // is based on rounding off protectionFactor on actual source packet number).
+  // The correction factor (_corrFecCost) attempts to corrects this, at least
+  // for cases of low rates (small #packets) and low protection levels.
+
+  float numPacketsFl = 1.0f + (static_cast<float>(bitRatePerFrame) * 1000.0 /
+                                   static_cast<float>(8.0 * _maxPayloadSize) +
+                               0.5);
+
+  const float estNumFecGen =
+      0.5f + static_cast<float>(_protectionFactorD * numPacketsFl / 255.0f);
+
+  // We reduce cost factor (which will reduce overhead for FEC and
+  // hybrid method) and not the protectionFactor.
+  _corrFecCost = 1.0f;
+  if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec) {
+    _corrFecCost = 0.5f;
+  }
+  if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec) {
+    _corrFecCost = 0.0f;
+  }
+
+  // TODO(marpan): Set the UEP protection on/off for Key and Delta frames
+  _useUepProtectionK = _qmRobustness->SetUepProtection(
+      codeRateKey, parameters->bitRate, packetLoss, 0);
+
+  _useUepProtectionD = _qmRobustness->SetUepProtection(
+      codeRateDelta, parameters->bitRate, packetLoss, 1);
+
+  // DONE WITH FEC PROTECTION SETTINGS
+  return true;
+}
+
+int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
+  // When temporal layers are available FEC will only be applied on the base
+  // layer.
+  const float bitRateRatio =
+      kVp8LayerRateAlloction[parameters->numLayers - 1][0];
+  float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1);
+  float bitRate = parameters->bitRate * bitRateRatio;
+  float frameRate = parameters->frameRate * frameRateRatio;
+
+  // TODO(mikhal): Update factor following testing.
+  float adjustmentFactor = 1;
+
+  // Average bits per frame (units of kbits)
+  return static_cast<int>(adjustmentFactor * bitRate / frameRate);
+}
+
+bool VCMFecMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameters) {
+  // Effective packet loss to encoder is based on RPL (residual packet loss)
+  // this is a soft setting based on degree of FEC protection
+  // RPL = received/input packet loss - average_FEC_recovery
+  // note: received/input packet loss may be filtered based on FilteredLoss
+
+  // Effective Packet Loss, NA in current version.
+  _effectivePacketLoss = 0;
+
+  return true;
+}
+
+bool VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters) {
+  // Compute the protection factor
+  ProtectionFactor(parameters);
+
+  // Compute the effective packet loss
+  EffectivePacketLoss(parameters);
+
+  // Protection/fec rates obtained above is defined relative to total number
+  // of packets (total rate: source+fec) FEC in RTP module assumes protection
+  // factor is defined relative to source number of packets so we should
+  // convert the factor to reduce mismatch between mediaOpt suggested rate and
+  // the actual rate
+  _protectionFactorK = ConvertFECRate(_protectionFactorK);
+  _protectionFactorD = ConvertFECRate(_protectionFactorD);
+
+  return true;
+}
+VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs)
+    : _currentParameters(),
+      _rtt(0),
+      _lossPr(0.0f),
+      _bitRate(0.0f),
+      _frameRate(0.0f),
+      _keyFrameSize(0.0f),
+      _fecRateKey(0),
+      _fecRateDelta(0),
+      _lastPrUpdateT(0),
+      _lossPr255(0.9999f),
+      _lossPrHistory(),
+      _shortMaxLossPr255(0),
+      _packetsPerFrame(0.9999f),
+      _packetsPerFrameKey(0.9999f),
+      _codecWidth(0),
+      _codecHeight(0),
+      _numLayers(1) {
+  Reset(nowMs);
+}
+
+VCMLossProtectionLogic::~VCMLossProtectionLogic() {
+  Release();
+}
+
+void VCMLossProtectionLogic::SetMethod(
+    enum VCMProtectionMethodEnum newMethodType) {
+  if (_selectedMethod && _selectedMethod->Type() == newMethodType)
+    return;
+
+  switch (newMethodType) {
+    case kNack:
+      _selectedMethod.reset(new VCMNackMethod());
+      break;
+    case kFec:
+      _selectedMethod.reset(new VCMFecMethod());
+      break;
+    case kNackFec:
+      _selectedMethod.reset(new VCMNackFecMethod(kLowRttNackMs, -1));
+      break;
+    case kNone:
+      _selectedMethod.reset();
+      break;
+  }
+  UpdateMethod();
+}
+
+void VCMLossProtectionLogic::UpdateRtt(int64_t rtt) {
+  _rtt = rtt;
+}
+
+void VCMLossProtectionLogic::UpdateMaxLossHistory(uint8_t lossPr255,
+                                                  int64_t now) {
+  if (_lossPrHistory[0].timeMs >= 0 &&
+      now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs) {
+    if (lossPr255 > _shortMaxLossPr255) {
+      _shortMaxLossPr255 = lossPr255;
+    }
+  } else {
+    // Only add a new value to the history once a second
+    if (_lossPrHistory[0].timeMs == -1) {
+      // First, no shift
+      _shortMaxLossPr255 = lossPr255;
+    } else {
+      // Shift
+      for (int32_t i = (kLossPrHistorySize - 2); i >= 0; i--) {
+        _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
+        _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
+      }
+    }
+    if (_shortMaxLossPr255 == 0) {
+      _shortMaxLossPr255 = lossPr255;
+    }
+
+    _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
+    _lossPrHistory[0].timeMs = now;
+    _shortMaxLossPr255 = 0;
+  }
+}
+
+uint8_t VCMLossProtectionLogic::MaxFilteredLossPr(int64_t nowMs) const {
+  uint8_t maxFound = _shortMaxLossPr255;
+  if (_lossPrHistory[0].timeMs == -1) {
+    return maxFound;
+  }
+  for (int32_t i = 0; i < kLossPrHistorySize; i++) {
+    if (_lossPrHistory[i].timeMs == -1) {
+      break;
+    }
+    if (nowMs - _lossPrHistory[i].timeMs >
+        kLossPrHistorySize * kLossPrShortFilterWinMs) {
+      // This sample (and all samples after this) is too old
+      break;
+    }
+    if (_lossPrHistory[i].lossPr255 > maxFound) {
+      // This sample is the largest one this far into the history
+      maxFound = _lossPrHistory[i].lossPr255;
+    }
+  }
+  return maxFound;
+}
+
+uint8_t VCMLossProtectionLogic::FilteredLoss(int64_t nowMs,
+                                             FilterPacketLossMode filter_mode,
+                                             uint8_t lossPr255) {
+  // Update the max window filter.
+  UpdateMaxLossHistory(lossPr255, nowMs);
+
+  // Update the recursive average filter.
+  _lossPr255.Apply(static_cast<float>(nowMs - _lastPrUpdateT),
+                   static_cast<float>(lossPr255));
+  _lastPrUpdateT = nowMs;
+
+  // Filtered loss: default is received loss (no filtering).
+  uint8_t filtered_loss = lossPr255;
+
+  switch (filter_mode) {
+    case kNoFilter:
+      break;
+    case kAvgFilter:
+      filtered_loss = static_cast<uint8_t>(_lossPr255.filtered() + 0.5);
+      break;
+    case kMaxFilter:
+      filtered_loss = MaxFilteredLossPr(nowMs);
+      break;
+  }
+
+  return filtered_loss;
+}
+
+void VCMLossProtectionLogic::UpdateFilteredLossPr(uint8_t packetLossEnc) {
+  _lossPr = static_cast<float>(packetLossEnc) / 255.0;
+}
+
+void VCMLossProtectionLogic::UpdateBitRate(float bitRate) {
+  _bitRate = bitRate;
+}
+
+void VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets,
+                                                   int64_t nowMs) {
+  _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
+                         nPackets);
+  _lastPacketPerFrameUpdateT = nowMs;
+}
+
+void VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets,
+                                                      int64_t nowMs) {
+  _packetsPerFrameKey.Apply(
+      static_cast<float>(nowMs - _lastPacketPerFrameUpdateTKey), nPackets);
+  _lastPacketPerFrameUpdateTKey = nowMs;
+}
+
+void VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize) {
+  _keyFrameSize = keyFrameSize;
+}
+
+void VCMLossProtectionLogic::UpdateFrameSize(uint16_t width, uint16_t height) {
+  _codecWidth = width;
+  _codecHeight = height;
+}
+
+void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
+  _numLayers = (numLayers == 0) ? 1 : numLayers;
+}
+
+bool VCMLossProtectionLogic::UpdateMethod() {
+  if (!_selectedMethod)
+    return false;
+  _currentParameters.rtt = _rtt;
+  _currentParameters.lossPr = _lossPr;
+  _currentParameters.bitRate = _bitRate;
+  _currentParameters.frameRate = _frameRate;  // rename actual frame rate?
+  _currentParameters.keyFrameSize = _keyFrameSize;
+  _currentParameters.fecRateDelta = _fecRateDelta;
+  _currentParameters.fecRateKey = _fecRateKey;
+  _currentParameters.packetsPerFrame = _packetsPerFrame.filtered();
+  _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.filtered();
+  _currentParameters.codecWidth = _codecWidth;
+  _currentParameters.codecHeight = _codecHeight;
+  _currentParameters.numLayers = _numLayers;
+  return _selectedMethod->UpdateParameters(&_currentParameters);
+}
+
+VCMProtectionMethod* VCMLossProtectionLogic::SelectedMethod() const {
+  return _selectedMethod.get();
+}
+
+VCMProtectionMethodEnum VCMLossProtectionLogic::SelectedType() const {
+  return _selectedMethod ? _selectedMethod->Type() : kNone;
+}
+
+void VCMLossProtectionLogic::Reset(int64_t nowMs) {
+  _lastPrUpdateT = nowMs;
+  _lastPacketPerFrameUpdateT = nowMs;
+  _lastPacketPerFrameUpdateTKey = nowMs;
+  _lossPr255.Reset(0.9999f);
+  _packetsPerFrame.Reset(0.9999f);
+  _fecRateDelta = _fecRateKey = 0;
+  for (int32_t i = 0; i < kLossPrHistorySize; i++) {
+    _lossPrHistory[i].lossPr255 = 0;
+    _lossPrHistory[i].timeMs = -1;
+  }
+  _shortMaxLossPr255 = 0;
+  Release();
+}
+
+void VCMLossProtectionLogic::Release() {
+  _selectedMethod.reset();
+}
+
+}  // namespace media_optimization
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/media_opt_util.h b/media/webrtc/trunk/webrtc/modules/video_coding/media_opt_util.h
new file mode 100644
index 0000000000..a016a03eab
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/media_opt_util.h
@@ -0,0 +1,361 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "webrtc/base/exp_filter.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/qm_select.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace media_optimization {
+
+// Number of time periods used for (max) window filter for packet loss
+// TODO(marpan): set reasonable window size for filtered packet loss,
+// adjustment should be based on logged/real data of loss stats/correlation.
+enum { kLossPrHistorySize = 10 };
+
+// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
+enum { kLossPrShortFilterWinMs = 1000 };
+
+// The type of filter used on the received packet loss reports.
+enum FilterPacketLossMode {
+  kNoFilter,   // No filtering on received loss.
+  kAvgFilter,  // Recursive average filter.
+  kMaxFilter   // Max-window filter, over the time interval of:
+               // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
+};
+
+// Thresholds for hybrid NACK/FEC
+// common to media optimization and the jitter buffer.
+const int64_t kLowRttNackMs = 20;
+
+struct VCMProtectionParameters {
+  VCMProtectionParameters()
+      : rtt(0),
+        lossPr(0.0f),
+        bitRate(0.0f),
+        packetsPerFrame(0.0f),
+        packetsPerFrameKey(0.0f),
+        frameRate(0.0f),
+        keyFrameSize(0.0f),
+        fecRateDelta(0),
+        fecRateKey(0),
+        codecWidth(0),
+        codecHeight(0),
+        numLayers(1) {}
+
+  int64_t rtt;
+  float lossPr;
+  float bitRate;
+  float packetsPerFrame;
+  float packetsPerFrameKey;
+  float frameRate;
+  float keyFrameSize;
+  uint8_t fecRateDelta;
+  uint8_t fecRateKey;
+  uint16_t codecWidth;
+  uint16_t codecHeight;
+  int numLayers;
+};
+
+/******************************/
+/* VCMProtectionMethod class  */
+/******************************/
+
+enum VCMProtectionMethodEnum { kNack, kFec, kNackFec, kNone };
+
+class VCMLossProbabilitySample {
+ public:
+  VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {}
+
+  uint8_t lossPr255;
+  int64_t timeMs;
+};
+
+class VCMProtectionMethod {
+ public:
+  VCMProtectionMethod();
+  virtual ~VCMProtectionMethod();
+
+  // Updates the efficiency of the method using the parameters provided
+  //
+  // Input:
+  //         - parameters         : Parameters used to calculate efficiency
+  //
+  // Return value                 : True if this method is recommended in
+  //                                the given conditions.
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
+
+  // Returns the protection type
+  //
+  // Return value                 : The protection type
+  enum VCMProtectionMethodEnum Type() const { return _type; }
+
+  // Returns the effective packet loss for ER, required by this protection
+  // method
+  //
+  // Return value                 : Required effective packet loss
+  virtual uint8_t RequiredPacketLossER() { return _effectivePacketLoss; }
+
+  // Extracts the FEC protection factor for Key frame, required by this
+  // protection method
+  //
+  // Return value                 : Required protectionFactor for Key frame
+  virtual uint8_t RequiredProtectionFactorK() { return _protectionFactorK; }
+
+  // Extracts the FEC protection factor for Delta frame, required by this
+  // protection method
+  //
+  // Return value                 : Required protectionFactor for delta frame
+  virtual uint8_t RequiredProtectionFactorD() { return _protectionFactorD; }
+
+  // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
+  //
+  // Return value                 : Required Unequal protection on/off state.
+  virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
+
+  // Extracts whether the the FEC Unequal protection (UEP) is used for Delta
+  // frame.
+  //
+  // Return value                 : Required Unequal protection on/off state.
+  virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
+
+  virtual int MaxFramesFec() const { return 1; }
+
+  // Updates content metrics
+  void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
+
+ protected:
+  uint8_t _effectivePacketLoss;
+  uint8_t _protectionFactorK;
+  uint8_t _protectionFactorD;
+  // Estimation of residual loss after the FEC
+  float _scaleProtKey;
+  int32_t _maxPayloadSize;
+
+  VCMQmRobustness* _qmRobustness;
+  bool _useUepProtectionK;
+  bool _useUepProtectionD;
+  float _corrFecCost;
+  enum VCMProtectionMethodEnum _type;
+};
+
+class VCMNackMethod : public VCMProtectionMethod {
+ public:
+  VCMNackMethod();
+  virtual ~VCMNackMethod();
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+  // Get the effective packet loss
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
+};
+
+class VCMFecMethod : public VCMProtectionMethod {
+ public:
+  VCMFecMethod();
+  virtual ~VCMFecMethod();
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+  // Get the effective packet loss for ER
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+  // Get the FEC protection factors
+  bool ProtectionFactor(const VCMProtectionParameters* parameters);
+  // Get the boost for key frame protection
+  uint8_t BoostCodeRateKey(uint8_t packetFrameDelta,
+                           uint8_t packetFrameKey) const;
+  // Convert the rates: defined relative to total# packets or source# packets
+  uint8_t ConvertFECRate(uint8_t codeRate) const;
+  // Get the average effective recovery from FEC: for random loss model
+  float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
+  // Update FEC with protectionFactorD
+  void UpdateProtectionFactorD(uint8_t protectionFactorD);
+  // Update FEC with protectionFactorK
+  void UpdateProtectionFactorK(uint8_t protectionFactorK);
+  // Compute the bits per frame. Account for temporal layers when applicable.
+  int BitsPerFrame(const VCMProtectionParameters* parameters);
+
+ protected:
+  enum { kUpperLimitFramesFec = 6 };
+  // Thresholds values for the bytes/frame and round trip time, below which we
+  // may turn off FEC, depending on |_numLayers| and |_maxFramesFec|.
+  // Max bytes/frame for VGA, corresponds to ~140k at 25fps.
+  enum { kMaxBytesPerFrameForFec = 700 };
+  // Max bytes/frame for CIF and lower: corresponds to ~80k at 25fps.
+  enum { kMaxBytesPerFrameForFecLow = 400 };
+  // Max bytes/frame for frame size larger than VGA, ~200k at 25fps.
+  enum { kMaxBytesPerFrameForFecHigh = 1000 };
+};
+
+class VCMNackFecMethod : public VCMFecMethod {
+ public:
+  VCMNackFecMethod(int64_t lowRttNackThresholdMs,
+                   int64_t highRttNackThresholdMs);
+  virtual ~VCMNackFecMethod();
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
+  // Get the effective packet loss for ER
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+  // Get the protection factors
+  bool ProtectionFactor(const VCMProtectionParameters* parameters);
+  // Get the max number of frames the FEC is allowed to be based on.
+  int MaxFramesFec() const;
+  // Turn off the FEC based on low bitrate and other factors.
+  bool BitRateTooLowForFec(const VCMProtectionParameters* parameters);
+
+ private:
+  int ComputeMaxFramesFec(const VCMProtectionParameters* parameters);
+
+  int64_t _lowRttNackMs;
+  int64_t _highRttNackMs;
+  int _maxFramesFec;
+};
+
+class VCMLossProtectionLogic {
+ public:
+  explicit VCMLossProtectionLogic(int64_t nowMs);
+  ~VCMLossProtectionLogic();
+
+  // Set the protection method to be used
+  //
+  // Input:
+  //        - newMethodType    : New requested protection method type. If one
+  //                           is already set, it will be deleted and replaced
+  void SetMethod(VCMProtectionMethodEnum newMethodType);
+
+  // Update the round-trip time
+  //
+  // Input:
+  //          - rtt           : Round-trip time in seconds.
+  void UpdateRtt(int64_t rtt);
+
+  // Update the filtered packet loss.
+  //
+  // Input:
+  //          - packetLossEnc :  The reported packet loss filtered
+  //                             (max window or average)
+  void UpdateFilteredLossPr(uint8_t packetLossEnc);
+
+  // Update the current target bit rate.
+  //
+  // Input:
+  //          - bitRate          : The current target bit rate in kbits/s
+  void UpdateBitRate(float bitRate);
+
+  // Update the number of packets per frame estimate, for delta frames
+  //
+  // Input:
+  //          - nPackets         : Number of packets in the latest sent frame.
+  void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
+
+  // Update the number of packets per frame estimate, for key frames
+  //
+  // Input:
+  //          - nPackets         : umber of packets in the latest sent frame.
+  void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
+
+  // Update the keyFrameSize estimate
+  //
+  // Input:
+  //          - keyFrameSize     : The size of the latest sent key frame.
+  void UpdateKeyFrameSize(float keyFrameSize);
+
+  // Update the frame rate
+  //
+  // Input:
+  //          - frameRate        : The current target frame rate.
+  void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
+
+  // Update the frame size
+  //
+  // Input:
+  //          - width        : The codec frame width.
+  //          - height       : The codec frame height.
+  void UpdateFrameSize(uint16_t width, uint16_t height);
+
+  // Update the number of active layers
+  //
+  // Input:
+  //          - numLayers    : Number of layers used.
+  void UpdateNumLayers(int numLayers);
+
+  // The amount of packet loss to cover for with FEC.
+  //
+  // Input:
+  //          - fecRateKey      : Packet loss to cover for with FEC when
+  //                              sending key frames.
+  //          - fecRateDelta    : Packet loss to cover for with FEC when
+  //                              sending delta frames.
+  void UpdateFECRates(uint8_t fecRateKey, uint8_t fecRateDelta) {
+    _fecRateKey = fecRateKey;
+    _fecRateDelta = fecRateDelta;
+  }
+
+  // Update the protection methods with the current VCMProtectionParameters
+  // and set the requested protection settings.
+  // Return value     : Returns true on update
+  bool UpdateMethod();
+
+  // Returns the method currently selected.
+  //
+  // Return value                 : The protection method currently selected.
+  VCMProtectionMethod* SelectedMethod() const;
+
+  // Return the protection type of the currently selected method
+  VCMProtectionMethodEnum SelectedType() const;
+
+  // Updates the filtered loss for the average and max window packet loss,
+  // and returns the filtered loss probability in the interval [0, 255].
+  // The returned filtered loss value depends on the parameter |filter_mode|.
+  // The input parameter |lossPr255| is the received packet loss.
+
+  // Return value                 : The filtered loss probability
+  uint8_t FilteredLoss(int64_t nowMs,
+                       FilterPacketLossMode filter_mode,
+                       uint8_t lossPr255);
+
+  void Reset(int64_t nowMs);
+
+  void Release();
+
+ private:
+  // Sets the available loss protection methods.
+  void UpdateMaxLossHistory(uint8_t lossPr255, int64_t now);
+  uint8_t MaxFilteredLossPr(int64_t nowMs) const;
+  rtc::scoped_ptr<VCMProtectionMethod> _selectedMethod;
+  VCMProtectionParameters _currentParameters;
+  int64_t _rtt;
+  float _lossPr;
+  float _bitRate;
+  float _frameRate;
+  float _keyFrameSize;
+  uint8_t _fecRateKey;
+  uint8_t _fecRateDelta;
+  int64_t _lastPrUpdateT;
+  int64_t _lastPacketPerFrameUpdateT;
+  int64_t _lastPacketPerFrameUpdateTKey;
+  rtc::ExpFilter _lossPr255;
+  VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
+  uint8_t _shortMaxLossPr255;
+  rtc::ExpFilter _packetsPerFrame;
+  rtc::ExpFilter _packetsPerFrameKey;
+  uint16_t _codecWidth;
+  uint16_t _codecHeight;
+  int _numLayers;
+};
+
+}  // namespace media_optimization
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.cc b/media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.cc
similarity index 85%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.cc
index 7040556f2d..aca8170935 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
 
-#include "webrtc/modules/video_coding/main/source/content_metrics_processing.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/content_metrics_processing.h"
+#include "webrtc/modules/video_coding/qm_select.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace media_optimization {
@@ -53,11 +53,9 @@ void UpdateProtectionCallback(
   key_fec_params.fec_mask_type = kFecMaskRandom;
 
   // TODO(Marco): Pass FEC protection values per layer.
-  video_protection_callback->ProtectionRequest(&delta_fec_params,
-                                               &key_fec_params,
-                                               video_rate_bps,
-                                               nack_overhead_rate_bps,
-                                               fec_overhead_rate_bps);
+  video_protection_callback->ProtectionRequest(
+      &delta_fec_params, &key_fec_params, video_rate_bps,
+      nack_overhead_rate_bps, fec_overhead_rate_bps);
 }
 }  // namespace
 
@@ -88,7 +86,7 @@ MediaOptimization::MediaOptimization(Clock* clock)
       fraction_lost_(0),
       send_statistics_zero_encode_(0),
       max_payload_size_(1460),
-      target_bit_rate_(0),
+      video_target_bitrate_(0),
       incoming_frame_rate_(0),
       enable_qm_(false),
       encoded_frame_samples_(),
@@ -128,7 +126,7 @@ void MediaOptimization::Reset() {
   loss_prot_logic_->UpdateFrameRate(incoming_frame_rate_);
   loss_prot_logic_->Reset(clock_->TimeInMilliseconds());
   send_statistics_zero_encode_ = 0;
-  target_bit_rate_ = 0;
+  video_target_bitrate_ = 0;
   codec_width_ = 0;
   codec_height_ = 0;
   min_width_ = 0;
@@ -164,15 +162,8 @@ void MediaOptimization::SetEncodingData(VideoCodecType send_codec_type,
                                         int num_layers,
                                         int32_t mtu) {
   CriticalSectionScoped lock(crit_sect_.get());
-  SetEncodingDataInternal(send_codec_type,
-                          max_bit_rate,
-                          target_bitrate,
-                          width,
-                          height,
-                          frame_rate,
-                          divisor,
-                          num_layers,
-                          mtu);
+  SetEncodingDataInternal(send_codec_type, max_bit_rate, target_bitrate,
+                          width, height, frame_rate, divisor, num_layers, mtu);
 }
 
 void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
@@ -194,7 +185,7 @@ void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
 
   max_bit_rate_ = max_bit_rate;
   send_codec_type_ = send_codec_type;
-  target_bit_rate_ = target_bitrate;
+  video_target_bitrate_ = target_bitrate;
   float target_bitrate_kbps = static_cast<float>(target_bitrate) / 1000.0f;
   loss_prot_logic_->UpdateBitRate(target_bitrate_kbps);
   loss_prot_logic_->UpdateFrameRate(static_cast<float>(frame_rate) / 1000.0f);
@@ -210,11 +201,8 @@ void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
   min_height_ = gcd ? (codec_height_/gcd * divisor) : 0;
   num_layers_ = (num_layers <= 1) ? 1 : num_layers;  // Can also be zero.
   max_payload_size_ = mtu;
-  qm_resolution_->Initialize(target_bitrate_kbps,
-                             user_frame_rate_,
-                             codec_width_,
-                             codec_height_,
-                             num_layers_);
+  qm_resolution_->Initialize(target_bitrate_kbps, user_frame_rate_,
+                             codec_width_, codec_height_, num_layers_);
 }
 
 uint32_t MediaOptimization::SetTargetRates(
@@ -223,16 +211,11 @@ uint32_t MediaOptimization::SetTargetRates(
     int64_t round_trip_time_ms,
     VCMProtectionCallback* protection_callback,
     VCMQMSettingsCallback* qmsettings_callback) {
-  LOG(LS_INFO) << "SetTargetRates: " <<  target_bitrate << " bps " << fraction_lost
-               << "% loss " << round_trip_time_ms << "ms RTT";
 
   CriticalSectionScoped lock(crit_sect_.get());
-  // TODO(holmer): Consider putting this threshold only on the video bitrate,
-  // and not on protection.
-  if (max_bit_rate_ > 0 &&
-      target_bitrate > static_cast<uint32_t>(max_bit_rate_)) {
-    target_bitrate = max_bit_rate_;
-  }
+  LOG(LS_INFO) << "SetTargetRates: " <<  target_bitrate << " bps " << (int) fraction_lost
+               << "% loss " << round_trip_time_ms << "ms RTT";
+
   VCMProtectionMethod* selected_method = loss_prot_logic_->SelectedMethod();
   float target_bitrate_kbps = static_cast<float>(target_bitrate) / 1000.0f;
   loss_prot_logic_->UpdateBitRate(target_bitrate_kbps);
@@ -264,11 +247,11 @@ uint32_t MediaOptimization::SetTargetRates(
   loss_prot_logic_->UpdateFilteredLossPr(packet_loss_enc);
 
   // Rate cost of the protection methods.
-  uint32_t protection_overhead_bps = 0;
+  float protection_overhead_rate = 0.0f;
 
   // Update protection settings, when applicable.
   float sent_video_rate_kbps = 0.0f;
-  if (selected_method) {
+  if (loss_prot_logic_->SelectedType() != kNone) {
     // Update protection method with content metrics.
     selected_method->UpdateContentMetrics(content_->ShortTermAvgData());
 
@@ -285,10 +268,8 @@ uint32_t MediaOptimization::SetTargetRates(
     // overhead data actually transmitted (including headers) the last
     // second.
     if (protection_callback) {
-      UpdateProtectionCallback(selected_method,
-                               &sent_video_rate_bps,
-                               &sent_nack_rate_bps,
-                               &sent_fec_rate_bps,
+      UpdateProtectionCallback(selected_method, &sent_video_rate_bps,
+                               &sent_nack_rate_bps, &sent_fec_rate_bps,
                                protection_callback);
     }
     uint32_t sent_total_rate_bps =
@@ -296,15 +277,13 @@ uint32_t MediaOptimization::SetTargetRates(
     // Estimate the overhead costs of the next second as staying the same
     // wrt the source bitrate.
     if (sent_total_rate_bps > 0) {
-      protection_overhead_bps = static_cast<uint32_t>(
-          target_bitrate *
-              static_cast<double>(sent_nack_rate_bps + sent_fec_rate_bps) /
-              sent_total_rate_bps +
-          0.5);
+      protection_overhead_rate =
+          static_cast<float>(sent_nack_rate_bps + sent_fec_rate_bps) /
+          sent_total_rate_bps;
     }
     // Cap the overhead estimate to 50%.
-    if (protection_overhead_bps > target_bitrate / 2)
-      protection_overhead_bps = target_bitrate / 2;
+    if (protection_overhead_rate > 0.5)
+      protection_overhead_rate = 0.5;
 
     // Get the effective packet loss for encoder ER when applicable. Should be
     // passed to encoder via fraction_lost.
@@ -313,11 +292,16 @@ uint32_t MediaOptimization::SetTargetRates(
   }
 
   // Source coding rate: total rate - protection overhead.
-  target_bit_rate_ = target_bitrate - protection_overhead_bps;
+  video_target_bitrate_ = target_bitrate * (1.0 - protection_overhead_rate);
+
+  // Cap target video bitrate to codec maximum.
+  if (max_bit_rate_ > 0 && video_target_bitrate_ > max_bit_rate_) {
+    video_target_bitrate_ = max_bit_rate_;
+  }
 
   // Update encoding rates following protection settings.
   float target_video_bitrate_kbps =
-      static_cast<float>(target_bit_rate_) / 1000.0f;
+      static_cast<float>(video_target_bitrate_) / 1000.0f;
   frame_dropper_->SetRates(target_video_bitrate_kbps, incoming_frame_rate_);
 
   if (enable_qm_ && qmsettings_callback) {
@@ -326,10 +310,8 @@ uint32_t MediaOptimization::SetTargetRates(
                << " fps, " << fraction_lost << " loss";
 
     // Update QM with rates.
-    qm_resolution_->UpdateRates(target_video_bitrate_kbps,
-                                sent_video_rate_kbps,
-                                incoming_frame_rate_,
-                                fraction_lost_);
+    qm_resolution_->UpdateRates(target_video_bitrate_kbps, sent_video_rate_kbps,
+                                incoming_frame_rate_, fraction_lost_);
     // Check for QM selection.
     bool select_qm = CheckStatusForQMchange();
     if (select_qm) {
@@ -341,16 +323,11 @@ uint32_t MediaOptimization::SetTargetRates(
 
   CheckSuspendConditions();
 
-  return target_bit_rate_;
+  return video_target_bitrate_;
 }
 
-void MediaOptimization::EnableProtectionMethod(bool enable,
-                                               VCMProtectionMethodEnum method) {
+void MediaOptimization::SetProtectionMethod(VCMProtectionMethodEnum method) {
   CriticalSectionScoped lock(crit_sect_.get());
-  if (!enable && loss_prot_logic_->SelectedType() != method)
-    return;
-  if (!enable)
-    method = kNone;
   loss_prot_logic_->SetMethod(method);
 }
 
@@ -383,14 +360,6 @@ uint32_t MediaOptimization::SentBitRate() {
   return avg_sent_bit_rate_bps_;
 }
 
-VCMFrameCount MediaOptimization::SentFrameCount() {
-  CriticalSectionScoped lock(crit_sect_.get());
-  VCMFrameCount count;
-  count.numDeltaFrames = delta_frame_cnt_;
-  count.numKeyFrames = key_frame_cnt_;
-  return count;
-}
-
 int32_t MediaOptimization::UpdateWithEncodedData(
     const EncodedImage& encoded_image) {
   size_t encoded_length = encoded_image._length;
@@ -415,7 +384,7 @@ int32_t MediaOptimization::UpdateWithEncodedData(
   UpdateSentBitrate(now_ms);
   UpdateSentFramerate();
   if (encoded_length > 0) {
-    const bool delta_frame = encoded_image._frameType != kKeyFrame;
+    const bool delta_frame = encoded_image._frameType != kVideoFrameKey;
 
     // XXX TODO(jesup): if same_frame is true, we should be considering it a single
     // frame here.
@@ -567,8 +536,7 @@ void MediaOptimization::UpdateSentBitrate(int64_t now_ms) {
   }
   size_t framesize_sum = 0;
   for (FrameSampleList::iterator it = encoded_frame_samples_.begin();
-       it != encoded_frame_samples_.end();
-       ++it) {
+       it != encoded_frame_samples_.end(); ++it) {
     framesize_sum += it->size_bytes;
   }
   float denom = static_cast<float>(
@@ -631,7 +599,7 @@ bool MediaOptimization::QMUpdate(
 
 
   LOG(LS_INFO) << "Media optimizer requests the video resolution to be changed "
-              "to " << qm->codec_width << " (" << codec_width_ << ") x " 
+              "to " << qm->codec_width << " (" << codec_width_ << ") x "
                << qm->codec_height << " (" << codec_height_ << ") @ "
                << qm->frame_rate;
 
@@ -641,11 +609,11 @@ bool MediaOptimization::QMUpdate(
   // will vary/fluctuate, and since we don't want to change the state of the
   // VPM frame dropper, unless a temporal action was selected, we use the
   // quantity |qm->frame_rate| for updating.
-  video_qmsettings_callback->SetVideoQMSettings(
-      qm->frame_rate, codec_width_, codec_height_);
+  video_qmsettings_callback->SetVideoQMSettings(qm->frame_rate, codec_width_,
+                                                codec_height_);
   content_->UpdateFrameRate(qm->frame_rate);
-  qm_resolution_->UpdateCodecParameters(
-      qm->frame_rate, codec_width_, codec_height_);
+  qm_resolution_->UpdateCodecParameters(qm->frame_rate, codec_width_,
+                                        codec_height_);
   return true;
 }
 
@@ -682,8 +650,9 @@ void MediaOptimization::ProcessIncomingFrameRate(int64_t now) {
     }
   }
   if (num > 1) {
-    const int64_t diff = now - incoming_frame_times_[num - 1];
-    incoming_frame_rate_ = 1.0;
+    const int64_t diff =
+        incoming_frame_times_[0] - incoming_frame_times_[num - 1];
+    incoming_frame_rate_ = 0.0;  // No frame rate estimate available.
     if (diff > 0) {
       incoming_frame_rate_ = nr_of_frames * 1000.0f / static_cast<float>(diff);
     }
@@ -696,17 +665,18 @@ void MediaOptimization::SetCPULoadState(CPULoadState state) {
 }
 
 void MediaOptimization::CheckSuspendConditions() {
-  // Check conditions for SuspendBelowMinBitrate. |target_bit_rate_| is in bps.
+  // Check conditions for SuspendBelowMinBitrate. |video_target_bitrate_| is in
+  // bps.
   if (suspension_enabled_) {
     if (!video_suspended_) {
       // Check if we just went below the threshold.
-      if (target_bit_rate_ < suspension_threshold_bps_) {
+      if (video_target_bitrate_ < suspension_threshold_bps_) {
         video_suspended_ = true;
       }
     } else {
       // Video is already suspended. Check if we just went over the threshold
       // with a margin.
-      if (target_bit_rate_ >
+      if (video_target_bitrate_ >
           suspension_threshold_bps_ + suspension_window_bps_) {
         video_suspended_ = false;
       }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.h b/media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.h
index fccd220025..d838c383e4 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/media_optimization.h
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MEDIA_OPTIMIZATION_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MEDIA_OPTIMIZATION_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
+#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
 
 #include <list>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/media_opt_util.h"
+#include "webrtc/modules/video_coding/qm_select.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -63,7 +63,7 @@ class MediaOptimization {
                           VCMProtectionCallback* protection_callback,
                           VCMQMSettingsCallback* qmsettings_callback);
 
-  void EnableProtectionMethod(bool enable, VCMProtectionMethodEnum method);
+  void SetProtectionMethod(VCMProtectionMethodEnum method);
   void EnableQM(bool enable);
   void EnableFrameDropper(bool enable);
 
@@ -83,21 +83,15 @@ class MediaOptimization {
   // Informs Media Optimization of CPU Load state
   void SetCPULoadState(CPULoadState state);
 
+  // InputFrameRate 0 = no frame rate estimate available.
   uint32_t InputFrameRate();
   uint32_t SentFrameRate();
   uint32_t SentBitRate();
-  VCMFrameCount SentFrameCount();
 
  private:
-  enum {
-    kFrameCountHistorySize = 90
-  };
-  enum {
-    kFrameHistoryWinMs = 2000
-  };
-  enum {
-    kBitrateAverageWinMs = 1000
-  };
+  enum { kFrameCountHistorySize = 90 };
+  enum { kFrameHistoryWinMs = 2000 };
+  enum { kBitrateAverageWinMs = 1000 };
 
   struct EncodedFrameSample;
   typedef std::list<EncodedFrameSample> FrameSampleList;
@@ -125,8 +119,8 @@ class MediaOptimization {
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Checks conditions for suspending the video. The method compares
-  // |target_bit_rate_| with the threshold values for suspension, and changes
-  // the state of |video_suspended_| accordingly.
+  // |video_target_bitrate_| with the threshold values for suspension, and
+  // changes the state of |video_suspended_| accordingly.
   void CheckSuspendConditions() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   void SetEncodingDataInternal(VideoCodecType send_codec_type,
@@ -162,7 +156,7 @@ class MediaOptimization {
   uint32_t send_statistics_[4] GUARDED_BY(crit_sect_);
   uint32_t send_statistics_zero_encode_ GUARDED_BY(crit_sect_);
   int32_t max_payload_size_ GUARDED_BY(crit_sect_);
-  int target_bit_rate_ GUARDED_BY(crit_sect_);
+  int video_target_bitrate_ GUARDED_BY(crit_sect_);
   float incoming_frame_rate_ GUARDED_BY(crit_sect_);
   int64_t incoming_frame_times_[kFrameCountHistorySize] GUARDED_BY(crit_sect_);
   bool enable_qm_ GUARDED_BY(crit_sect_);
@@ -185,4 +179,4 @@ class MediaOptimization {
 }  // namespace media_optimization
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MEDIA_OPTIMIZATION_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/media_optimization_unittest.cc
similarity index 59%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/media_optimization_unittest.cc
index 5031015d75..3f8ac5d075 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/media_optimization_unittest.cc
@@ -9,8 +9,8 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace media_optimization {
@@ -38,7 +38,7 @@ class TestMediaOptimization : public ::testing::Test {
       EncodedImage encoded_image;
       encoded_image._length = bytes_per_frame;
       encoded_image._timeStamp = next_timestamp_;
-      encoded_image._frameType = kKeyFrame;
+      encoded_image._frameType = kVideoFrameKey;
       ASSERT_EQ(VCM_OK, media_opt_.UpdateWithEncodedData(encoded_image));
     }
     next_timestamp_ += frame_time_ms_ * kSampleRate / 1000;
@@ -51,7 +51,6 @@ class TestMediaOptimization : public ::testing::Test {
   uint32_t next_timestamp_;
 };
 
-
 TEST_F(TestMediaOptimization, VerifyMuting) {
   // Enable video suspension with these limits.
   // Suspend the video when the rate is below 50 kbps and resume when it gets
@@ -65,10 +64,9 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
 
   uint32_t target_bitrate_kbps = 100;
   media_opt_.SetTargetRates(target_bitrate_kbps * 1000,
-                            0,  // Lossrate.
-                            100,
-                            NULL,
-                            NULL);  // RTT in ms.
+                            0,    // Lossrate.
+                            100,  // RTT in ms.
+                            nullptr, nullptr);
   media_opt_.EnableFrameDropper(true);
   for (int time = 0; time < 2000; time += frame_time_ms_) {
     ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, false));
@@ -76,10 +74,9 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
 
   // Set the target rate below the limit for muting.
   media_opt_.SetTargetRates(kThresholdBps - 1000,
-                            0,  // Lossrate.
-                            100,
-                            NULL,
-                            NULL);  // RTT in ms.
+                            0,    // Lossrate.
+                            100,  // RTT in ms.
+                            nullptr, nullptr);
   // Expect the muter to engage immediately and stay muted.
   // Test during 2 seconds.
   for (int time = 0; time < 2000; time += frame_time_ms_) {
@@ -90,11 +87,10 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
   // Set the target above the limit for muting, but not above the
   // limit + window.
   media_opt_.SetTargetRates(kThresholdBps + 1000,
-                            0,  // Lossrate.
-                            100,
-                            NULL,
-                            NULL);  // RTT in ms.
-                                    // Expect the muter to stay muted.
+                            0,    // Lossrate.
+                            100,  // RTT in ms.
+                            nullptr, nullptr);
+  // Expect the muter to stay muted.
   // Test during 2 seconds.
   for (int time = 0; time < 2000; time += frame_time_ms_) {
     EXPECT_TRUE(media_opt_.IsVideoSuspended());
@@ -103,10 +99,9 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
 
   // Set the target above limit + window.
   media_opt_.SetTargetRates(kThresholdBps + kWindowBps + 1000,
-                            0,  // Lossrate.
-                            100,
-                            NULL,
-                            NULL);  // RTT in ms.
+                            0,    // Lossrate.
+                            100,  // RTT in ms.
+                            nullptr, nullptr);
   // Expect the muter to disengage immediately.
   // Test during 2 seconds.
   for (int time = 0; time < 2000; time += frame_time_ms_) {
@@ -116,5 +111,44 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
   }
 }
 
+TEST_F(TestMediaOptimization, ProtectsUsingFecBitrateAboveCodecMax) {
+  static const int kCodecBitrateBps = 100000;
+  static const int kMaxBitrateBps = 130000;
+
+  class ProtectionCallback : public VCMProtectionCallback {
+    int ProtectionRequest(const FecProtectionParams* delta_params,
+                          const FecProtectionParams* key_params,
+                          uint32_t* sent_video_rate_bps,
+                          uint32_t* sent_nack_rate_bps,
+                          uint32_t* sent_fec_rate_bps) override {
+      *sent_video_rate_bps = kCodecBitrateBps;
+      *sent_nack_rate_bps = 0;
+      *sent_fec_rate_bps = fec_rate_bps_;
+      return 0;
+    }
+
+   public:
+    uint32_t fec_rate_bps_;
+  } protection_callback;
+
+  media_opt_.SetProtectionMethod(kFec);
+  media_opt_.SetEncodingData(kVideoCodecVP8, kCodecBitrateBps, kCodecBitrateBps,
+                             640, 480, 30, 1, 1000);
+
+  // Using 10% of codec bitrate for FEC, should still be able to use all of it.
+  protection_callback.fec_rate_bps_ = kCodecBitrateBps / 10;
+  uint32_t target_bitrate = media_opt_.SetTargetRates(
+      kMaxBitrateBps, 0, 0, &protection_callback, nullptr);
+
+  EXPECT_EQ(kCodecBitrateBps, static_cast<int>(target_bitrate));
+
+  // Using as much for codec bitrate as fec rate, new target rate should share
+  // both equally, but only be half of max (since that ceiling should be hit).
+  protection_callback.fec_rate_bps_ = kCodecBitrateBps;
+  target_bitrate = media_opt_.SetTargetRates(kMaxBitrateBps, 128, 100,
+                                             &protection_callback, nullptr);
+  EXPECT_EQ(kMaxBitrateBps / 2, static_cast<int>(target_bitrate));
+}
+
 }  // namespace media_optimization
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/nack_fec_tables.h b/media/webrtc/trunk/webrtc/modules/video_coding/nack_fec_tables.h
new file mode 100644
index 0000000000..f9f5ad97ac
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/nack_fec_tables.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_NACK_FEC_TABLES_H_
+#define WEBRTC_MODULES_VIDEO_CODING_NACK_FEC_TABLES_H_
+
+namespace webrtc {
+
+// Table for adjusting FEC rate for NACK/FEC protection method
+// Table values are built as a sigmoid function, ranging from 0 to 100, based on
+// the HybridNackTH values defined in media_opt_util.h.
+const uint16_t VCMNackFecTable[100] = {
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   1,   1,   1,   1,
+    1,   2,   2,   2,   3,   3,   4,   5,   6,   7,   9,   10,  12,  15,  18,
+    21,  24,  28,  32,  37,  41,  46,  51,  56,  61,  66,  70,  74,  78,  81,
+    84,  86,  89,  90,  92,  93,  95,  95,  96,  97,  97,  98,  98,  99,  99,
+    99,  99,  99,  99,  100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 100, 100, 100, 100,
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_NACK_FEC_TABLES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/packet.cc b/media/webrtc/trunk/webrtc/modules/video_coding/packet.cc
similarity index 56%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/packet.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/packet.cc
index 41ebdbb537..e063c6ee74 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/packet.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/packet.cc
@@ -8,78 +8,75 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 #include <assert.h>
 
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
+
 namespace webrtc {
 
 VCMPacket::VCMPacket()
-  :
-    payloadType(0),
-    timestamp(0),
-    ntp_time_ms_(0),
-    seqNum(0),
-    dataPtr(NULL),
-    sizeBytes(0),
-    markerBit(false),
-    frameType(kFrameEmpty),
-    codec(kVideoCodecUnknown),
-    isFirstPacket(false),
-    completeNALU(kNaluUnset),
-    insertStartCode(false),
-    width(0),
-    height(0),
-    codecSpecificHeader() {
-}
+    : payloadType(0),
+      timestamp(0),
+      ntp_time_ms_(0),
+      seqNum(0),
+      dataPtr(NULL),
+      sizeBytes(0),
+      markerBit(false),
+      frameType(kEmptyFrame),
+      codec(kVideoCodecUnknown),
+      isFirstPacket(false),
+      completeNALU(kNaluUnset),
+      insertStartCode(false),
+      width(0),
+      height(0),
+      codecSpecificHeader() {}
 
 VCMPacket::VCMPacket(const uint8_t* ptr,
                      const size_t size,
-                     const WebRtcRTPHeader& rtpHeader) :
-    payloadType(rtpHeader.header.payloadType),
-    timestamp(rtpHeader.header.timestamp),
-    ntp_time_ms_(rtpHeader.ntp_time_ms),
-    seqNum(rtpHeader.header.sequenceNumber),
-    dataPtr(ptr),
-    sizeBytes(size),
-    markerBit(rtpHeader.header.markerBit),
+                     const WebRtcRTPHeader& rtpHeader)
+    : payloadType(rtpHeader.header.payloadType),
+      timestamp(rtpHeader.header.timestamp),
+      ntp_time_ms_(rtpHeader.ntp_time_ms),
+      seqNum(rtpHeader.header.sequenceNumber),
+      dataPtr(ptr),
+      sizeBytes(size),
+      markerBit(rtpHeader.header.markerBit),
 
-    frameType(rtpHeader.frameType),
-    codec(kVideoCodecUnknown),
-    isFirstPacket(rtpHeader.type.Video.isFirstPacket),
-    completeNALU(kNaluComplete),
-    insertStartCode(false),
-    width(rtpHeader.type.Video.width),
-    height(rtpHeader.type.Video.height),
-    codecSpecificHeader(rtpHeader.type.Video)
-{
-    CopyCodecSpecifics(rtpHeader.type.Video);
+      frameType(rtpHeader.frameType),
+      codec(kVideoCodecUnknown),
+      isFirstPacket(rtpHeader.type.Video.isFirstPacket),
+      completeNALU(kNaluComplete),
+      insertStartCode(false),
+      width(rtpHeader.type.Video.width),
+      height(rtpHeader.type.Video.height),
+      codecSpecificHeader(rtpHeader.type.Video) {
+  CopyCodecSpecifics(rtpHeader.type.Video);
 }
 
 VCMPacket::VCMPacket(const uint8_t* ptr,
                      size_t size,
                      uint16_t seq,
                      uint32_t ts,
-                     bool mBit) :
-    payloadType(0),
-    timestamp(ts),
-    ntp_time_ms_(0),
-    seqNum(seq),
-    dataPtr(ptr),
-    sizeBytes(size),
-    markerBit(mBit),
+                     bool mBit)
+    : payloadType(0),
+      timestamp(ts),
+      ntp_time_ms_(0),
+      seqNum(seq),
+      dataPtr(ptr),
+      sizeBytes(size),
+      markerBit(mBit),
 
-    frameType(kVideoFrameDelta),
-    codec(kVideoCodecUnknown),
-    isFirstPacket(false),
-    completeNALU(kNaluComplete),
-    insertStartCode(false),
-    width(0),
-    height(0),
-    codecSpecificHeader()
-{}
+      frameType(kVideoFrameDelta),
+      codec(kVideoCodecUnknown),
+      isFirstPacket(false),
+      completeNALU(kNaluComplete),
+      insertStartCode(false),
+      width(0),
+      height(0),
+      codecSpecificHeader() {}
 
 void VCMPacket::Reset() {
   payloadType = 0;
@@ -89,7 +86,7 @@ void VCMPacket::Reset() {
   dataPtr = NULL;
   sizeBytes = 0;
   markerBit = false;
-  frameType = kFrameEmpty;
+  frameType = kEmptyFrame;
   codec = kVideoCodecUnknown;
   isFirstPacket = false;
   completeNALU = kNaluUnset;
@@ -105,7 +102,6 @@ void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
   }
   switch (videoHeader.codec) {
     case kRtpVideoVp8:
-    case kRtpVideoVp9:
       // Handle all packets within a frame as depending on the previous packet
       // TODO(holmer): This should be changed to make fragments independent
       // when the VP8 RTP receiver supports fragments.
@@ -118,15 +114,26 @@ void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
       else
         completeNALU = kNaluIncomplete;
 
-      codec = videoHeader.codec == kRtpVideoVp8 ? kVideoCodecVP8 : kVideoCodecVP9;
+      codec = kVideoCodecVP8;
+      return;
+    case kRtpVideoVp9:
+      if (isFirstPacket && markerBit)
+        completeNALU = kNaluComplete;
+      else if (isFirstPacket)
+        completeNALU = kNaluStart;
+      else if (markerBit)
+        completeNALU = kNaluEnd;
+      else
+        completeNALU = kNaluIncomplete;
+
+      codec = kVideoCodecVP9;
       return;
     case kRtpVideoH264:
       isFirstPacket = videoHeader.isFirstPacket;
-      if (isFirstPacket) {
+      if (isFirstPacket)
         insertStartCode = true;
-      }
 
-      if (videoHeader.codecHeader.H264.single_nalu) {
+      if (isFirstPacket && markerBit) {
         completeNALU = kNaluComplete;
       } else if (isFirstPacket) {
         completeNALU = kNaluStart;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/packet.h b/media/webrtc/trunk/webrtc/modules/video_coding/packet.h
new file mode 100644
index 0000000000..b77c1df039
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/packet.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMPacket {
+ public:
+  VCMPacket();
+  VCMPacket(const uint8_t* ptr,
+            const size_t size,
+            const WebRtcRTPHeader& rtpHeader);
+  VCMPacket(const uint8_t* ptr,
+            size_t size,
+            uint16_t seqNum,
+            uint32_t timestamp,
+            bool markerBit);
+
+  void Reset();
+
+  uint8_t payloadType;
+  uint32_t timestamp;
+  // NTP time of the capture time in local timebase in milliseconds.
+  int64_t ntp_time_ms_;
+  uint16_t seqNum;
+  const uint8_t* dataPtr;
+  size_t sizeBytes;
+  bool markerBit;
+
+  FrameType frameType;
+  VideoCodecType codec;
+
+  bool isFirstPacket;                // Is this first packet in a frame.
+  VCMNaluCompleteness completeNALU;  // Default is kNaluIncomplete.
+  bool insertStartCode;  // True if a start code should be inserted before this
+                         // packet.
+  int width;
+  int height;
+  RTPVideoHeader codecSpecificHeader;
+
+ protected:
+  void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.cc b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select.cc
similarity index 88%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/qm_select.cc
index 92f70d52e5..e5c04491b9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select.cc
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
+#include "webrtc/modules/video_coding/qm_select.h"
 
 #include <math.h>
 #ifdef ANDROID
 #include <android/log.h>
 #endif
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/qm_select_data.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/qm_select_data.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -39,8 +39,7 @@ VCMQmMethod::VCMQmMethod()
   ResetQM();
 }
 
-VCMQmMethod::~VCMQmMethod() {
-}
+VCMQmMethod::~VCMQmMethod() {}
 
 void VCMQmMethod::ResetQM() {
   aspect_ratio_ = 1.0f;
@@ -55,7 +54,7 @@ uint8_t VCMQmMethod::ComputeContentClass() {
   return content_class_ = 3 * motion_.level + spatial_.level;
 }
 
-void VCMQmMethod::UpdateContent(const VideoContentMetrics*  contentMetrics) {
+void VCMQmMethod::UpdateContent(const VideoContentMetrics* contentMetrics) {
   content_metrics_ = contentMetrics;
 }
 
@@ -71,7 +70,7 @@ void VCMQmMethod::ComputeMotionNFD() {
   if (motion_.value < kLowMotionNfd) {
     motion_.level = kLow;
   } else if (motion_.value > kHighMotionNfd) {
-    motion_.level  = kHigh;
+    motion_.level = kHigh;
   } else {
     motion_.level = kDefault;
   }
@@ -88,7 +87,7 @@ void VCMQmMethod::ComputeSpatial() {
   float spatial_err_h = 0.0;
   float spatial_err_v = 0.0;
   if (content_metrics_) {
-    spatial_err =  content_metrics_->spatial_pred_err;
+    spatial_err = content_metrics_->spatial_pred_err;
     spatial_err_h = content_metrics_->spatial_pred_err_h;
     spatial_err_v = content_metrics_->spatial_pred_err_v;
   }
@@ -108,8 +107,7 @@ void VCMQmMethod::ComputeSpatial() {
 #endif
 }
 
-ImageType VCMQmMethod::GetImageType(uint16_t width,
-                                    uint16_t height) {
+ImageType VCMQmMethod::GetImageType(uint16_t width, uint16_t height) {
   // Get the image type for the encoder frame size.
   uint32_t image_size = width * height;
   if (image_size == kSizeOfImageType[kQCIF]) {
@@ -156,7 +154,7 @@ FrameRateLevelClass VCMQmMethod::FrameRateLevel(float avg_framerate) {
   } else if (avg_framerate <= kMiddleFrameRate) {
     return kFrameRateMiddle1;
   } else if (avg_framerate <= kHighFrameRate) {
-     return kFrameRateMiddle2;
+    return kFrameRateMiddle2;
   } else {
     return kFrameRateHigh;
   }
@@ -164,8 +162,7 @@ FrameRateLevelClass VCMQmMethod::FrameRateLevel(float avg_framerate) {
 
 // RESOLUTION CLASS
 
-VCMQmResolution::VCMQmResolution()
-    :  qm_(new VCMResolutionScale()) {
+VCMQmResolution::VCMQmResolution() : qm_(new VCMResolutionScale()) {
   Reset();
 }
 
@@ -188,7 +185,7 @@ void VCMQmResolution::ResetRates() {
 
 void VCMQmResolution::ResetDownSamplingState() {
   state_dec_factor_spatial_ = 1.0;
-  state_dec_factor_temporal_  = 1.0;
+  state_dec_factor_temporal_ = 1.0;
   for (int i = 0; i < kDownActionHistorySize; i++) {
     down_action_history_[i].spatial = kNoChangeSpatial;
     down_action_history_[i].temporal = kNoChangeTemporal;
@@ -245,11 +242,12 @@ int VCMQmResolution::Initialize(float bitrate,
   buffer_level_ = kInitBufferLevel * target_bitrate_;
   // Per-frame bandwidth.
   per_frame_bandwidth_ = target_bitrate_ / user_framerate;
-  init_  = true;
+  init_ = true;
   return VCM_OK;
 }
 
-void VCMQmResolution::UpdateCodecParameters(float frame_rate, uint16_t width,
+void VCMQmResolution::UpdateCodecParameters(float frame_rate,
+                                            uint16_t width,
                                             uint16_t height) {
   width_ = width;
   height_ = height;
@@ -303,12 +301,12 @@ void VCMQmResolution::UpdateRates(float target_bitrate,
 
   // Update with the current new target and frame rate:
   // these values are ones the encoder will use for the current/next ~1sec.
-  target_bitrate_ =  target_bitrate;
+  target_bitrate_ = target_bitrate;
   incoming_framerate_ = incoming_framerate;
   sum_incoming_framerate_ += incoming_framerate_;
   // Update the per_frame_bandwidth:
   // this is the per_frame_bw for the current/next ~1sec.
-  per_frame_bandwidth_  = 0.0f;
+  per_frame_bandwidth_ = 0.0f;
   if (incoming_framerate_ > 0.0f) {
     per_frame_bandwidth_ = target_bitrate_ / incoming_framerate_;
   }
@@ -335,7 +333,7 @@ int VCMQmResolution::SelectResolution(VCMResolutionScale** qm) {
   }
   if (content_metrics_ == NULL) {
     Reset();
-    *qm =  qm_;
+    *qm = qm_;
     return VCM_OK;
   }
 
@@ -398,31 +396,31 @@ void VCMQmResolution::ComputeRatesForSelection() {
   avg_rate_mismatch_sgn_ = 0.0f;
   avg_packet_loss_ = 0.0f;
   if (frame_cnt_ > 0) {
-    avg_ratio_buffer_low_ = static_cast<float>(low_buffer_cnt_) /
-        static_cast<float>(frame_cnt_);
+    avg_ratio_buffer_low_ =
+        static_cast<float>(low_buffer_cnt_) / static_cast<float>(frame_cnt_);
   }
   if (update_rate_cnt_ > 0) {
-    avg_rate_mismatch_ = static_cast<float>(sum_rate_MM_) /
-        static_cast<float>(update_rate_cnt_);
+    avg_rate_mismatch_ =
+        static_cast<float>(sum_rate_MM_) / static_cast<float>(update_rate_cnt_);
     avg_rate_mismatch_sgn_ = static_cast<float>(sum_rate_MM_sgn_) /
-        static_cast<float>(update_rate_cnt_);
+                             static_cast<float>(update_rate_cnt_);
     avg_target_rate_ = static_cast<float>(sum_target_rate_) /
-        static_cast<float>(update_rate_cnt_);
+                       static_cast<float>(update_rate_cnt_);
     avg_incoming_framerate_ = static_cast<float>(sum_incoming_framerate_) /
-        static_cast<float>(update_rate_cnt_);
-    avg_packet_loss_ =  static_cast<float>(sum_packet_loss_) /
-        static_cast<float>(update_rate_cnt_);
+                              static_cast<float>(update_rate_cnt_);
+    avg_packet_loss_ = static_cast<float>(sum_packet_loss_) /
+                       static_cast<float>(update_rate_cnt_);
   }
   // For selection we may want to weight some quantities more heavily
   // with the current (i.e., next ~1sec) rate values.
-  avg_target_rate_ = kWeightRate * avg_target_rate_ +
-      (1.0 - kWeightRate) * target_bitrate_;
+  avg_target_rate_ =
+      kWeightRate * avg_target_rate_ + (1.0 - kWeightRate) * target_bitrate_;
   avg_incoming_framerate_ = kWeightRate * avg_incoming_framerate_ +
-      (1.0 - kWeightRate) * incoming_framerate_;
+                            (1.0 - kWeightRate) * incoming_framerate_;
   // Use base layer frame rate for temporal layers: this will favor spatial.
   assert(num_layers_ > 0);
-  framerate_level_ = FrameRateLevel(
-      avg_incoming_framerate_ / static_cast<float>(1 << (num_layers_ - 1)));
+  framerate_level_ = FrameRateLevel(avg_incoming_framerate_ /
+                                    static_cast<float>(1 << (num_layers_ - 1)));
 }
 
 void VCMQmResolution::ComputeEncoderState() {
@@ -434,7 +432,7 @@ void VCMQmResolution::ComputeEncoderState() {
   // 2) rate mis-match is high, and consistent over-shooting by encoder.
   if ((avg_ratio_buffer_low_ > kMaxBufferLow) ||
       ((avg_rate_mismatch_ > kMaxRateMisMatch) &&
-          (avg_rate_mismatch_sgn_ < -kRateOverShoot))) {
+       (avg_rate_mismatch_sgn_ < -kRateOverShoot))) {
     encoder_state_ = kStressedEncoding;
     WEBRTC_TRACE(webrtc::kTraceDebug,
                  webrtc::kTraceVideoCoding,
@@ -477,9 +475,9 @@ bool VCMQmResolution::GoingUpResolution() {
   // Modify the fac_width/height for this case.
   if (down_action_history_[0].spatial == kOneQuarterSpatialUniform) {
     fac_width = kFactorWidthSpatial[kOneQuarterSpatialUniform] /
-        kFactorWidthSpatial[kOneHalfSpatialUniform];
+                kFactorWidthSpatial[kOneHalfSpatialUniform];
     fac_height = kFactorHeightSpatial[kOneQuarterSpatialUniform] /
-        kFactorHeightSpatial[kOneHalfSpatialUniform];
+                 kFactorHeightSpatial[kOneHalfSpatialUniform];
   }
 
   // Check if we should go up both spatially and temporally.
@@ -501,8 +499,8 @@ bool VCMQmResolution::GoingUpResolution() {
                                               kTransRateScaleUpSpatial);
   }
   if (down_action_history_[0].temporal != kNoChangeTemporal) {
-    selected_up_temporal = ConditionForGoingUp(1.0f, 1.0f, fac_temp,
-                                               kTransRateScaleUpTemp);
+    selected_up_temporal =
+        ConditionForGoingUp(1.0f, 1.0f, fac_temp, kTransRateScaleUpTemp);
   }
   if (selected_up_spatial && !selected_up_temporal) {
     action_.spatial = down_action_history_[0].spatial;
@@ -526,13 +524,13 @@ bool VCMQmResolution::ConditionForGoingUp(float fac_width,
                                           float fac_height,
                                           float fac_temp,
                                           float scale_fac) {
-  float estimated_transition_rate_up = GetTransitionRate(fac_width, fac_height,
-                                                         fac_temp, scale_fac);
+  float estimated_transition_rate_up =
+      GetTransitionRate(fac_width, fac_height, fac_temp, scale_fac);
   // Go back up if:
   // 1) target rate is above threshold and current encoder state is stable, or
   // 2) encoder state is easy (encoder is significantly under-shooting target).
   if (((avg_target_rate_ > estimated_transition_rate_up) &&
-      (encoder_state_ == kStableEncoding)) ||
+       (encoder_state_ == kStableEncoding)) ||
       (encoder_state_ == kEasyEncoding)) {
     return true;
   } else {
@@ -580,9 +578,7 @@ bool VCMQmResolution::GoingDownResolution() {
         action_.spatial = kNoChangeSpatial;
         break;
       }
-      default: {
-        assert(false);
-      }
+      default: { assert(false); }
     }
     switch (temp_fact) {
       case 3: {
@@ -597,9 +593,7 @@ bool VCMQmResolution::GoingDownResolution() {
         action_.temporal = kNoChangeTemporal;
         break;
       }
-      default: {
-        assert(false);
-      }
+      default: { assert(false); }
     }
     // Only allow for one action (spatial or temporal) at a given time.
     assert(action_.temporal == kNoChangeTemporal ||
@@ -639,9 +633,9 @@ float VCMQmResolution::GetTransitionRate(float fac_width,
                                          float fac_height,
                                          float fac_temp,
                                          float scale_fac) {
-  ImageType image_type = GetImageType(
-      static_cast<uint16_t>(fac_width * width_),
-      static_cast<uint16_t>(fac_height * height_));
+  ImageType image_type =
+      GetImageType(static_cast<uint16_t>(fac_width * width_),
+                   static_cast<uint16_t>(fac_height * height_));
 
   FrameRateLevelClass framerate_level =
       FrameRateLevel(fac_temp * avg_incoming_framerate_);
@@ -656,13 +650,13 @@ float VCMQmResolution::GetTransitionRate(float fac_width,
   // Nominal values based on image format (frame size and frame rate).
   float max_rate = kFrameRateFac[framerate_level] * kMaxRateQm[image_type];
 
-  uint8_t image_class = image_type > kVGA ? 1: 0;
+  uint8_t image_class = image_type > kVGA ? 1 : 0;
   uint8_t table_index = image_class * 9 + content_class_;
   // Scale factor for down-sampling transition threshold:
   // factor based on the content class and the image size.
   float scaleTransRate = kScaleTransRateQm[table_index];
   // Threshold bitrate for resolution action.
-  return static_cast<float> (scale_fac * scaleTransRate * max_rate);
+  return static_cast<float>(scale_fac * scaleTransRate * max_rate);
 }
 
 void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
@@ -672,9 +666,9 @@ void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
     // If last spatial action was 1/2x1/2, we undo it in two steps, so the
     // spatial scale factor in this first step is modified as (4.0/3.0 / 2.0).
     if (action_.spatial == kOneQuarterSpatialUniform) {
-      qm_->spatial_width_fact =
-          1.0f * kFactorWidthSpatial[kOneHalfSpatialUniform] /
-          kFactorWidthSpatial[kOneQuarterSpatialUniform];
+      qm_->spatial_width_fact = 1.0f *
+                                kFactorWidthSpatial[kOneHalfSpatialUniform] /
+                                kFactorWidthSpatial[kOneQuarterSpatialUniform];
       qm_->spatial_height_fact =
           1.0f * kFactorHeightSpatial[kOneHalfSpatialUniform] /
           kFactorHeightSpatial[kOneQuarterSpatialUniform];
@@ -693,14 +687,14 @@ void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
     // has been selected.
     assert(false);
   }
-
   UpdateCodecResolution();
   state_dec_factor_spatial_ = state_dec_factor_spatial_ *
-      qm_->spatial_width_fact * qm_->spatial_height_fact;
+                              qm_->spatial_width_fact *
+                              qm_->spatial_height_fact;
   state_dec_factor_temporal_ = state_dec_factor_temporal_ * qm_->temporal_fact;
 }
 
-void  VCMQmResolution::UpdateCodecResolution() {
+void VCMQmResolution::UpdateCodecResolution() {
   if (action_.spatial != kNoChangeSpatial) {
     qm_->change_resolution_spatial = true;
     int old_width = qm_->codec_width;
@@ -777,13 +771,13 @@ void  VCMQmResolution::UpdateCodecResolution() {
                         old_rate,
                         qm_->frame_rate);
 #endif
-
   }
 }
 
 uint8_t VCMQmResolution::RateClass(float transition_rate) {
-  return avg_target_rate_ < (kFacLowRate * transition_rate) ? 0:
-  (avg_target_rate_ >= transition_rate ? 2 : 1);
+  return avg_target_rate_ < (kFacLowRate * transition_rate)
+             ? 0
+             : (avg_target_rate_ >= transition_rate ? 2 : 1);
 }
 
 // TODO(marpan): Would be better to capture these frame rate adjustments by
@@ -818,7 +812,7 @@ void VCMQmResolution::AdjustAction() {
   }
   // Never use temporal action if number of temporal layers is above 2.
   if (num_layers_ > 2) {
-    if (action_.temporal !=  kNoChangeTemporal) {
+    if (action_.temporal != kNoChangeTemporal) {
       action_.spatial = kOneHalfSpatialUniform;
     }
     action_.temporal = kNoChangeTemporal;
@@ -833,35 +827,36 @@ void VCMQmResolution::ConvertSpatialFractionalToWhole() {
     bool found = false;
     int isel = kDownActionHistorySize;
     for (int i = 0; i < kDownActionHistorySize; ++i) {
-      if (down_action_history_[i].spatial ==  kOneHalfSpatialUniform) {
+      if (down_action_history_[i].spatial == kOneHalfSpatialUniform) {
         isel = i;
         found = true;
         break;
       }
     }
     if (found) {
-       action_.spatial = kOneQuarterSpatialUniform;
-       state_dec_factor_spatial_ = state_dec_factor_spatial_ /
-           (kFactorWidthSpatial[kOneHalfSpatialUniform] *
-            kFactorHeightSpatial[kOneHalfSpatialUniform]);
-       // Check if switching to 1/2x1/2 (=1/4) spatial is allowed.
-       ConstrainAmountOfDownSampling();
-       if (action_.spatial == kNoChangeSpatial) {
-         // Not allowed. Go back to 3/4x3/4 spatial.
-         action_.spatial = kOneHalfSpatialUniform;
-         state_dec_factor_spatial_ = state_dec_factor_spatial_ *
-             kFactorWidthSpatial[kOneHalfSpatialUniform] *
-             kFactorHeightSpatial[kOneHalfSpatialUniform];
-       } else {
-         // Switching is allowed. Remove 3/4x3/4 from the history, and update
-         // the frame size.
-         for (int i = isel; i < kDownActionHistorySize - 1; ++i) {
-           down_action_history_[i].spatial =
-               down_action_history_[i + 1].spatial;
-         }
-         width_ = width_ * kFactorWidthSpatial[kOneHalfSpatialUniform];
-         height_ = height_ * kFactorHeightSpatial[kOneHalfSpatialUniform];
-       }
+      action_.spatial = kOneQuarterSpatialUniform;
+      state_dec_factor_spatial_ =
+          state_dec_factor_spatial_ /
+          (kFactorWidthSpatial[kOneHalfSpatialUniform] *
+           kFactorHeightSpatial[kOneHalfSpatialUniform]);
+      // Check if switching to 1/2x1/2 (=1/4) spatial is allowed.
+      ConstrainAmountOfDownSampling();
+      if (action_.spatial == kNoChangeSpatial) {
+        // Not allowed. Go back to 3/4x3/4 spatial.
+        action_.spatial = kOneHalfSpatialUniform;
+        state_dec_factor_spatial_ =
+            state_dec_factor_spatial_ *
+            kFactorWidthSpatial[kOneHalfSpatialUniform] *
+            kFactorHeightSpatial[kOneHalfSpatialUniform];
+      } else {
+        // Switching is allowed. Remove 3/4x3/4 from the history, and update
+        // the frame size.
+        for (int i = isel; i < kDownActionHistorySize - 1; ++i) {
+          down_action_history_[i].spatial = down_action_history_[i + 1].spatial;
+        }
+        width_ = width_ * kFactorWidthSpatial[kOneHalfSpatialUniform];
+        height_ = height_ * kFactorHeightSpatial[kOneHalfSpatialUniform];
+      }
     }
   }
 }
@@ -911,8 +906,8 @@ void VCMQmResolution::ConstrainAmountOfDownSampling() {
   float spatial_width_fact = kFactorWidthSpatial[action_.spatial];
   float spatial_height_fact = kFactorHeightSpatial[action_.spatial];
   float temporal_fact = kFactorTemporal[action_.temporal];
-  float new_dec_factor_spatial = state_dec_factor_spatial_ *
-      spatial_width_fact * spatial_height_fact;
+  float new_dec_factor_spatial =
+      state_dec_factor_spatial_ * spatial_width_fact * spatial_height_fact;
   float new_dec_factor_temp = state_dec_factor_temporal_ * temporal_fact;
 
   // No spatial sampling if current frame size is too small, or if the
@@ -1008,8 +1003,7 @@ VCMQmRobustness::VCMQmRobustness() {
   Reset();
 }
 
-VCMQmRobustness::~VCMQmRobustness() {
-}
+VCMQmRobustness::~VCMQmRobustness() {}
 
 void VCMQmRobustness::Reset() {
   prev_total_rate_ = 0.0f;
@@ -1028,7 +1022,7 @@ float VCMQmRobustness::AdjustFecFactor(uint8_t code_rate_delta,
                                        int64_t rtt_time,
                                        uint8_t packet_loss) {
   // Default: no adjustment
-  float adjust_fec =  1.0f;
+  float adjust_fec = 1.0f;
   if (content_metrics_ == NULL) {
     return adjust_fec;
   }
@@ -1055,5 +1049,4 @@ bool VCMQmRobustness::SetUepProtection(uint8_t code_rate_delta,
   // Default.
   return false;
 }
-
-}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.h b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/qm_select.h
index 96b3bb3f94..ba31af6fcd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select.h
@@ -30,8 +30,7 @@ struct VCMResolutionScale {
         spatial_height_fact(1.0f),
         temporal_fact(1.0f),
         change_resolution_spatial(false),
-        change_resolution_temporal(false) {
-  }
+        change_resolution_temporal(false) {}
   uint16_t codec_width;
   uint16_t codec_height;
   float frame_rate;
@@ -48,20 +47,20 @@ struct VCMResolutionScale {
 // k???     192x144
 // k???     256x192 (good step between 320x240 and 160x120)
 enum ImageType {
-  kQCIF = 0,            // 176x144
-  kHCIF,                // 264x216 = half(~3/4x3/4) CIF.
-  kQVGA,                // 320x240 = quarter VGA.
-  kCIF,                 // 352x288
-  kHVGA,                // 480x360 = half(~3/4x3/4) VGA.
-  kVGA,                 // 640x480
-  kQFULLHD,             // 960x540 = quarter FULLHD, and half(~3/4x3/4) WHD.
-  kWHD,                 // 1280x720
-  kFULLHD,              // 1920x1080
+  kQCIF = 0,  // 176x144
+  kHCIF,      // 264x216 = half(~3/4x3/4) CIF.
+  kQVGA,      // 320x240 = quarter VGA.
+  kCIF,       // 352x288
+  kHVGA,      // 480x360 = half(~3/4x3/4) VGA.
+  kVGA,       // 640x480
+  kQFULLHD,   // 960x540 = quarter FULLHD, and half(~3/4x3/4) WHD.
+  kWHD,       // 1280x720
+  kFULLHD,    // 1920x1080
   kNumImageTypes
 };
 
-const uint32_t kSizeOfImageType[kNumImageTypes] =
-{ 25344, 57024, 76800, 101376, 172800, 307200, 518400, 921600, 2073600 };
+const uint32_t kSizeOfImageType[kNumImageTypes] = {
+    25344, 57024, 76800, 101376, 172800, 307200, 518400, 921600, 2073600};
 
 enum FrameRateLevelClass {
   kFrameRateLow,
@@ -70,17 +69,10 @@ enum FrameRateLevelClass {
   kFrameRateHigh
 };
 
-enum ContentLevelClass {
-  kLow,
-  kHigh,
-  kDefault
-};
+enum ContentLevelClass { kLow, kHigh, kDefault };
 
 struct VCMContFeature {
-  VCMContFeature()
-      : value(0.0f),
-        level(kDefault) {
-  }
+  VCMContFeature() : value(0.0f), level(kDefault) {}
   void Reset() {
     value = 0.0f;
     level = kDefault;
@@ -89,43 +81,34 @@ struct VCMContFeature {
   ContentLevelClass level;
 };
 
-enum UpDownAction {
-  kUpResolution,
-  kDownResolution
-};
+enum UpDownAction { kUpResolution, kDownResolution };
 
 enum SpatialAction {
   kNoChangeSpatial,
-  kOneHalfSpatialUniform,        // 3/4 x 3/4: 9/6 ~1/2 pixel reduction.
-  kOneQuarterSpatialUniform,     // 1/2 x 1/2: 1/4 pixel reduction.
+  kOneHalfSpatialUniform,     // 3/4 x 3/4: 9/6 ~1/2 pixel reduction.
+  kOneQuarterSpatialUniform,  // 1/2 x 1/2: 1/4 pixel reduction.
   kNumModesSpatial
 };
 
 enum TemporalAction {
   kNoChangeTemporal,
-  kTwoThirdsTemporal,     // 2/3 frame rate reduction
-  kOneHalfTemporal,       // 1/2 frame rate reduction
+  kTwoThirdsTemporal,  // 2/3 frame rate reduction
+  kOneHalfTemporal,    // 1/2 frame rate reduction
   kNumModesTemporal
 };
 
 struct ResolutionAction {
-  ResolutionAction()
-      : spatial(kNoChangeSpatial),
-        temporal(kNoChangeTemporal) {
-  }
+  ResolutionAction() : spatial(kNoChangeSpatial), temporal(kNoChangeTemporal) {}
   SpatialAction spatial;
   TemporalAction temporal;
 };
 
 // Down-sampling factors for spatial (width and height), and temporal.
-const float kFactorWidthSpatial[kNumModesSpatial] =
-    { 1.0f, 4.0f / 3.0f, 2.0f };
+const float kFactorWidthSpatial[kNumModesSpatial] = {1.0f, 4.0f / 3.0f, 2.0f};
 
-const float kFactorHeightSpatial[kNumModesSpatial] =
-    { 1.0f, 4.0f / 3.0f, 2.0f };
+const float kFactorHeightSpatial[kNumModesSpatial] = {1.0f, 4.0f / 3.0f, 2.0f};
 
-const float kFactorTemporal[kNumModesTemporal] =
-    { 1.0f, 1.5f, 2.0f };
+const float kFactorTemporal[kNumModesTemporal] = {1.0f, 1.5f, 2.0f};
 
 enum EncoderState {
   kStableEncoding,    // Low rate mis-match, stable buffer levels.
@@ -301,7 +284,7 @@ class VCMQmResolution : public VCMQmMethod {
   // Select the directional (1x2 or 2x1) spatial down-sampling action.
   void SelectSpatialDirectionMode(float transition_rate);
 
-  enum { kDownActionHistorySize = 10};
+  enum { kDownActionHistorySize = 10 };
 
   VCMResolutionScale* qm_;
   // Encoder rate control parameters.
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_data.h b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select_data.h
similarity index 67%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_data.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/qm_select_data.h
index dc6bce4811..49190ef53b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_data.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select_data.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
+#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
 
 /***************************************************************
 *QMSelectData.h
@@ -69,36 +69,36 @@ const uint16_t kMaxRateQm[9] = {
 
 // Frame rate scale for maximum transition rate.
 const float kFrameRateFac[4] = {
-    0.5f,    // Low
-    0.7f,    // Middle level 1
-    0.85f,   // Middle level 2
-    1.0f,    // High
+    0.5f,   // Low
+    0.7f,   // Middle level 1
+    0.85f,  // Middle level 2
+    1.0f,   // High
 };
 
 // Scale for transitional rate: based on content class
 // motion=L/H/D,spatial==L/H/D: for low, high, middle levels
 const float kScaleTransRateQm[18] = {
     // VGA and lower
-    0.40f,       // L, L
-    0.50f,       // L, H
-    0.40f,       // L, D
-    0.60f,       // H ,L
-    0.60f,       // H, H
-    0.60f,       // H, D
-    0.50f,       // D, L
-    0.50f,       // D, D
-    0.50f,       // D, H
+    0.40f,  // L, L
+    0.50f,  // L, H
+    0.40f,  // L, D
+    0.60f,  // H ,L
+    0.60f,  // H, H
+    0.60f,  // H, D
+    0.50f,  // D, L
+    0.50f,  // D, D
+    0.50f,  // D, H
 
     // over VGA
-    0.40f,       // L, L
-    0.50f,       // L, H
-    0.40f,       // L, D
-    0.60f,       // H ,L
-    0.60f,       // H, H
-    0.60f,       // H, D
-    0.50f,       // D, L
-    0.50f,       // D, D
-    0.50f,       // D, H
+    0.40f,  // L, L
+    0.50f,  // L, H
+    0.40f,  // L, D
+    0.60f,  // H ,L
+    0.60f,  // H, H
+    0.60f,  // H, D
+    0.50f,  // D, L
+    0.50f,  // D, D
+    0.50f,  // D, H
 };
 
 // Threshold on the target rate relative to transitional rate.
@@ -108,73 +108,73 @@ const float kFacLowRate = 0.5f;
 // motion=L/H/D,spatial==L/H/D, for low, high, middle levels;
 // rate = 0/1/2, for target rate state relative to transition rate.
 const uint8_t kSpatialAction[27] = {
-// rateClass = 0:
-    1,       // L, L
-    1,       // L, H
-    1,       // L, D
-    4,       // H ,L
-    1,       // H, H
-    4,       // H, D
-    4,       // D, L
-    1,       // D, H
-    2,       // D, D
+    // rateClass = 0:
+    1,  // L, L
+    1,  // L, H
+    1,  // L, D
+    4,  // H ,L
+    1,  // H, H
+    4,  // H, D
+    4,  // D, L
+    1,  // D, H
+    2,  // D, D
 
-// rateClass = 1:
-    1,       // L, L
-    1,       // L, H
-    1,       // L, D
-    2,       // H ,L
-    1,       // H, H
-    2,       // H, D
-    2,       // D, L
-    1,       // D, H
-    2,       // D, D
+    // rateClass = 1:
+    1,  // L, L
+    1,  // L, H
+    1,  // L, D
+    2,  // H ,L
+    1,  // H, H
+    2,  // H, D
+    2,  // D, L
+    1,  // D, H
+    2,  // D, D
 
-// rateClass = 2:
-    1,       // L, L
-    1,       // L, H
-    1,       // L, D
-    2,       // H ,L
-    1,       // H, H
-    2,       // H, D
-    2,       // D, L
-    1,       // D, H
-    2,       // D, D
+    // rateClass = 2:
+    1,  // L, L
+    1,  // L, H
+    1,  // L, D
+    2,  // H ,L
+    1,  // H, H
+    2,  // H, D
+    2,  // D, L
+    1,  // D, H
+    2,  // D, D
 };
 
 const uint8_t kTemporalAction[27] = {
-// rateClass = 0:
-    3,       // L, L
-    2,       // L, H
-    2,       // L, D
-    1,       // H ,L
-    3,       // H, H
-    1,       // H, D
-    1,       // D, L
-    2,       // D, H
-    1,       // D, D
+    // rateClass = 0:
+    3,  // L, L
+    2,  // L, H
+    2,  // L, D
+    1,  // H ,L
+    3,  // H, H
+    1,  // H, D
+    1,  // D, L
+    2,  // D, H
+    1,  // D, D
 
-// rateClass = 1:
-    3,       // L, L
-    3,       // L, H
-    3,       // L, D
-    1,       // H ,L
-    3,       // H, H
-    1,       // H, D
-    1,       // D, L
-    3,       // D, H
-    1,       // D, D
+    // rateClass = 1:
+    3,  // L, L
+    3,  // L, H
+    3,  // L, D
+    1,  // H ,L
+    3,  // H, H
+    1,  // H, D
+    1,  // D, L
+    3,  // D, H
+    1,  // D, D
 
-// rateClass = 2:
-    1,       // L, L
-    3,       // L, H
-    3,       // L, D
-    1,       // H ,L
-    3,       // H, H
-    1,       // H, D
-    1,       // D, L
-    3,       // D, H
-    1,       // D, D
+    // rateClass = 2:
+    1,  // L, L
+    3,  // L, H
+    3,  // L, D
+    1,  // H ,L
+    3,  // H, H
+    1,  // H, D
+    1,  // D, L
+    3,  // D, H
+    1,  // D, D
 };
 
 // Control the total amount of down-sampling allowed.
@@ -224,4 +224,4 @@ const float kSpatialErrVertVsHoriz = 0.1f;  // percentage to favor H over V
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select_unittest.cc
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/qm_select_unittest.cc
index 6abc0d3099..f8542ec676 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/qm_select_unittest.cc
@@ -15,8 +15,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/qm_select.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/qm_select.h"
 
 namespace webrtc {
 
@@ -32,10 +32,9 @@ const float kTemporalHigh = 0.1f;
 class QmSelectTest : public ::testing::Test {
  protected:
   QmSelectTest()
-      :  qm_resolution_(new VCMQmResolution()),
-         content_metrics_(new VideoContentMetrics()),
-         qm_scale_(NULL) {
-  }
+      : qm_resolution_(new VCMQmResolution()),
+        content_metrics_(new VideoContentMetrics()),
+        qm_scale_(NULL) {}
   VCMQmResolution* qm_resolution_;
   VideoContentMetrics* content_metrics_;
   VCMResolutionScale* qm_scale_;
@@ -87,8 +86,8 @@ TEST_F(QmSelectTest, HandleInputs) {
   qm_resolution_->UpdateContent(content_metrics);
   // Content metrics are NULL: Expect success and no down-sampling action.
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0, 1.0, 1.0, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0, 1.0, 1.0, 640, 480, 30.0f));
 }
 
 // TODO(marpan): Add a test for number of temporal layers > 1.
@@ -118,8 +117,8 @@ TEST_F(QmSelectTest, NoActionHighRate) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 }
 
 // Rate is well below transition, down-sampling action is taken,
@@ -149,40 +148,40 @@ TEST_F(QmSelectTest, DownActionLowRate) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, low spatial: 2/3 temporal is expected.
   UpdateQmContentData(kTemporalLow, kSpatialLow, kSpatialLow, kSpatialLow);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Medium motion, low spatial: 2x2 spatial expected.
   UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // High motion, high spatial: 2/3 temporal expected.
   UpdateQmContentData(kTemporalHigh, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(4, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial: 1/2 temporal expected.
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Medium motion, high spatial: 1/2 temporal expected.
@@ -190,8 +189,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
                       kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // High motion, medium spatial: 2x2 spatial expected.
@@ -200,8 +199,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
   // Target frame rate for frame dropper should be the same as previous == 15.
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, medium spatial: high frame rate, so 1/2 temporal expected.
@@ -209,8 +208,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
                       kSpatialMedium);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   qm_resolution_->ResetDownSamplingState();
   // Medium motion, medium spatial: high frame rate, so 2/3 temporal expected.
@@ -218,8 +217,8 @@ TEST_F(QmSelectTest, DownActionLowRate) {
                       kSpatialMedium);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(8, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 }
 
 // Rate mis-match is high, and we have over-shooting.
@@ -249,16 +248,16 @@ TEST_F(QmSelectTest, DownActionHighRateMMOvershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 }
 
 // Rate mis-match is high, target rate is below max for down-sampling,
@@ -288,16 +287,16 @@ TEST_F(QmSelectTest, NoActionHighRateMMUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 }
 
 // Buffer is underflowing, and target rate is below max for down-sampling,
@@ -332,16 +331,16 @@ TEST_F(QmSelectTest, DownActionBufferUnderflow) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
 }
 
 // Target rate is below max for down-sampling, but buffer level is stable,
@@ -376,16 +375,16 @@ TEST_F(QmSelectTest, NoActionBufferStable) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 
   qm_resolution_->ResetDownSamplingState();
   // Low motion, high spatial
   UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
 }
 
 // Very low rate, but no spatial down-sampling below some size (QCIF).
@@ -414,8 +413,8 @@ TEST_F(QmSelectTest, LimitDownSpatialAction) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 176, 144,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 176, 144, 30.0f));
 }
 
 // Very low rate, but no frame reduction below some frame_rate (8fps).
@@ -445,8 +444,8 @@ TEST_F(QmSelectTest, LimitDownTemporalAction) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      8.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 8.0f));
 }
 
 // Two stages: spatial down-sample and then back up spatially,
@@ -468,7 +467,7 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -476,8 +475,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Reset and go up in rate: expected to go back up, in 2 stages of 3/4.
   qm_resolution_->ResetRates();
@@ -493,8 +492,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -522,7 +521,7 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -530,8 +529,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Reset rates and simulate under-shooting scenario.: expect to go back up.
   // Goes up spatially in two stages for 1/2x1/2 down-sampling.
@@ -548,8 +547,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -577,7 +576,7 @@ TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -585,8 +584,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Reset and simulate large rate mis-match: expect no action to go back up.
   qm_resolution_->ResetRates();
@@ -601,8 +600,8 @@ TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 320, 240, 30.0f));
 }
 
 // Two stages: temporally down-sample and then back up temporally,
@@ -632,8 +631,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   // Reset rates and go up in rate: expect to go back up.
   qm_resolution_->ResetRates();
@@ -646,8 +645,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480, 30.0f));
 }
 
 // Two stages: temporal down-sample and then back up temporally, since encoder
@@ -669,7 +668,7 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
   int incoming_frame_rate[] = {30, 30, 30};
   uint8_t fraction_lost[] = {10, 10, 10};
   UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                    fraction_lost, 3);
+                   fraction_lost, 3);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // Low motion, high spatial.
@@ -677,8 +676,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480,
-                                      15.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
 
   // Reset rates and simulate under-shooting scenario.: expect to go back up.
   qm_resolution_->ResetRates();
@@ -691,8 +690,8 @@ TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480, 30.0f));
 }
 
 // Two stages: temporal down-sample and then no action to go up,
@@ -736,8 +735,8 @@ TEST_F(QmSelectTest, 2StageDownTemporalNoActionUp) {
                    fraction_lost2, 5);
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480,
-                                      15.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 15.0f));
 }
 // 3 stages: spatial down-sample, followed by temporal down-sample,
 // and then go up to full state, as encoding rate has increased.
@@ -766,8 +765,8 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Change content data: expect temporal down-sample.
   qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
@@ -780,7 +779,7 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
   uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
   UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                    fraction_lost2, 5);
+                   fraction_lost2, 5);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // Low motion, high spatial.
@@ -788,8 +787,8 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
 
   // Reset rates and go high up in rate: expect to go back up both spatial
   // and temporally. The 1/2x1/2 spatial is undone in two stages.
@@ -806,8 +805,8 @@ TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f,
-                                      480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
+                                      360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -842,8 +841,8 @@ TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 640, 360,
-              30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 640, 360, 30.0f));
 
   // Reset and lower rates to get another spatial action (3/4x3/4).
   // Lower the frame rate for spatial to be selected again.
@@ -865,8 +864,8 @@ TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 270, 10.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 270, 10.0f));
 
   // Reset and go to very low rate: no action should be taken,
   // we went down too much already.
@@ -883,8 +882,8 @@ TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 480, 270,
-                                      10.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 480, 270, 10.0f));
 }
 
 // Multiple down-sampling stages and then undo all of them.
@@ -917,8 +916,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
   // Go down 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -936,8 +935,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360, 20.5f));
 
   // Go down 3/4x3/4 spatial:
   qm_resolution_->UpdateCodecParameters(20.0f, 480, 360);
@@ -947,7 +946,7 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   int incoming_frame_rate3[] = {20, 20, 20, 20, 20};
   uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
   UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                    fraction_lost3, 5);
+                   fraction_lost3, 5);
 
   // Update content: motion level, and 3 spatial prediction errors.
   // High motion, low spatial.
@@ -957,8 +956,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
   // The two spatial actions of 3/4x3/4 are converted to 1/2x1/2,
   // so scale factor is 2.0.
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      20.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 20.0f));
 
   // Reset rates and go high up in rate: expect to go up:
   // 1/2x1x2 spatial and 1/2 temporally.
@@ -1018,8 +1017,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 
   // Go down 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
@@ -1039,8 +1038,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
 
   // Go up 2/3 temporally.
   qm_resolution_->UpdateCodecParameters(20.0f, 320, 240);
@@ -1076,8 +1075,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
 
   // Go up spatial and temporal. Spatial undoing is done in 2 stages.
   qm_resolution_->UpdateCodecParameters(20.5f, 320, 240);
@@ -1092,8 +1091,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
 
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f,
-                                      480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
+                                      360, 30.0f));
 
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
   EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
@@ -1131,8 +1130,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   // Go down 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
@@ -1151,8 +1150,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360,
-                                      20.5f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360, 20.5f));
 
   // Go up 2/3 temporal.
   qm_resolution_->UpdateCodecParameters(20.5f, 480, 360);
@@ -1184,8 +1183,8 @@ TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
 
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f,
-                                      1.0f, 640, 480, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
+                                      640, 480, 30.0f));
 }
 
 // Two stages of 3/4x3/4 converted to one stage of 1/2x1/2.
@@ -1215,8 +1214,8 @@ TEST_F(QmSelectTest, ConvertThreeQuartersToOneHalf) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f,
-                                      1.0f, 480, 360, 30.0f));
+  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
+                                      480, 360, 30.0f));
 
   // Set rates to go down another 3/4 spatial. Should be converted ton 1/2.
   qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
@@ -1235,8 +1234,8 @@ TEST_F(QmSelectTest, ConvertThreeQuartersToOneHalf) {
   EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
   EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
   EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240,
-                                      30.0f));
+  EXPECT_TRUE(
+      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
 }
 
 void QmSelectTest::InitQmNativeData(float initial_bit_rate,
@@ -1244,11 +1243,9 @@ void QmSelectTest::InitQmNativeData(float initial_bit_rate,
                                     int native_width,
                                     int native_height,
                                     int num_layers) {
-  EXPECT_EQ(0, qm_resolution_->Initialize(initial_bit_rate,
-                                          user_frame_rate,
-                                          native_width,
-                                          native_height,
-                                          num_layers));
+  EXPECT_EQ(
+      0, qm_resolution_->Initialize(initial_bit_rate, user_frame_rate,
+                                    native_width, native_height, num_layers));
 }
 
 void QmSelectTest::UpdateQmContentData(float motion_metric,
@@ -1281,8 +1278,7 @@ void QmSelectTest::UpdateQmRateData(int* target_rate,
     float encoder_sent_rate_update = encoder_sent_rate[i];
     float incoming_frame_rate_update = incoming_frame_rate[i];
     uint8_t fraction_lost_update = fraction_lost[i];
-    qm_resolution_->UpdateRates(target_rate_update,
-                                encoder_sent_rate_update,
+    qm_resolution_->UpdateRates(target_rate_update, encoder_sent_rate_update,
                                 incoming_frame_rate_update,
                                 fraction_lost_update);
   }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.cc b/media/webrtc/trunk/webrtc/modules/video_coding/receiver.cc
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/receiver.cc
index 813d934de0..506376199e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/receiver.cc
@@ -8,18 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/receiver.h"
+#include "webrtc/modules/video_coding/receiver.h"
 
 #include <assert.h>
 
 #include <cstdlib>
+#include <utility>
+#include <vector>
 
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/media_opt_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -27,16 +29,26 @@ enum { kMaxReceiverDelayMs = 10000 };
 
 VCMReceiver::VCMReceiver(VCMTiming* timing,
                          Clock* clock,
-                         EventFactory* event_factory,
-                         bool master)
+                         EventFactory* event_factory)
+    : VCMReceiver(timing,
+                  clock,
+                  rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent()),
+                  rtc::scoped_ptr<EventWrapper>(event_factory->CreateEvent())) {
+}
+
+VCMReceiver::VCMReceiver(VCMTiming* timing,
+                         Clock* clock,
+                         rtc::scoped_ptr<EventWrapper> receiver_event,
+                         rtc::scoped_ptr<EventWrapper> jitter_buffer_event)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       clock_(clock),
-      jitter_buffer_(clock_, event_factory),
+      jitter_buffer_(clock_, std::move(jitter_buffer_event)),
       timing_(timing),
-      render_wait_event_(event_factory->CreateEvent()),
-      state_(kPassive),
+      render_wait_event_(std::move(receiver_event)),
       receiveState_(kReceiveStateInitial),
-      max_video_delay_ms_(kMaxVideoDelayMs) {}
+      max_video_delay_ms_(kMaxVideoDelayMs) {
+  Reset();
+}
 
 VCMReceiver::~VCMReceiver() {
   render_wait_event_->Set();
@@ -50,15 +62,9 @@ void VCMReceiver::Reset() {
   } else {
     jitter_buffer_.Flush();
   }
-  state_ = kReceiving;
   receiveState_ = kReceiveStateInitial;
 }
 
-int32_t VCMReceiver::Initialize() {
-  Reset();
-  return VCM_OK;
-}
-
 void VCMReceiver::UpdateRtt(int64_t rtt) {
   jitter_buffer_.UpdateRtt(rtt);
 }
@@ -69,8 +75,8 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
   // Insert the packet into the jitter buffer. The packet can either be empty or
   // contain media at this point.
   bool retransmitted = false;
-  const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
-                                                             &retransmitted);
+  const VCMFrameBufferEnum ret =
+      jitter_buffer_.InsertPacket(packet, &retransmitted);
   if (ret == kOldPacket) {
     return VCM_OK;
   } else if (ret == kFlushIndicator) {
@@ -93,13 +99,13 @@ void VCMReceiver::TriggerDecoderShutdown() {
 }
 
 VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
-                                               int64_t& next_render_time_ms,
-                                               bool render_timing) {
+                                               int64_t* next_render_time_ms,
+                                               bool prefer_late_decoding) {
   const int64_t start_time_ms = clock_->TimeInMilliseconds();
   uint32_t frame_timestamp = 0;
   // Exhaust wait time to get a complete frame for decoding.
-  bool found_frame = jitter_buffer_.NextCompleteTimestamp(
-      max_wait_time_ms, &frame_timestamp);
+  bool found_frame =
+      jitter_buffer_.NextCompleteTimestamp(max_wait_time_ms, &frame_timestamp);
 
   if (!found_frame)
     found_frame = jitter_buffer_.NextMaybeIncompleteTimestamp(&frame_timestamp);
@@ -111,15 +117,14 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
   timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
   const int64_t now_ms = clock_->TimeInMilliseconds();
   timing_->UpdateCurrentDelay(frame_timestamp);
-  next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
+  *next_render_time_ms = timing_->RenderTimeMs(frame_timestamp, now_ms);
   // Check render timing.
   bool timing_error = false;
   // Assume that render timing errors are due to changes in the video stream.
-  if (next_render_time_ms < 0) {
+  if (*next_render_time_ms < 0) {
     timing_error = true;
-  } else if (std::abs(static_cast<int>(next_render_time_ms - now_ms)) >
-             max_video_delay_ms_) {
-    int frame_delay = std::abs(static_cast<int>(next_render_time_ms - now_ms));
+  } else if (std::abs(*next_render_time_ms - now_ms) > max_video_delay_ms_) {
+    int frame_delay = std::abs(*next_render_time_ms - now_ms);
     LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
                     << "delay bounds (" << frame_delay << " > "
                     << max_video_delay_ms_
@@ -139,19 +144,20 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
     return NULL;
   }
 
-  if (!render_timing) {
+  if (prefer_late_decoding) {
     // Decode frame as close as possible to the render timestamp.
-    const int32_t available_wait_time = max_wait_time_ms -
+    const int32_t available_wait_time =
+        max_wait_time_ms -
         static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
-    uint16_t new_max_wait_time = static_cast<uint16_t>(
-        VCM_MAX(available_wait_time, 0));
+    uint16_t new_max_wait_time =
+        static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
     uint32_t wait_time_ms = timing_->MaxWaitingTime(
-        next_render_time_ms, clock_->TimeInMilliseconds());
+        *next_render_time_ms, clock_->TimeInMilliseconds());
     if (new_max_wait_time < wait_time_ms) {
       // We're not allowed to wait until the frame is supposed to be rendered,
       // waiting as long as we're allowed to avoid busy looping, and then return
       // NULL. Next call to this function might return the frame.
-      render_wait_event_->Wait(max_wait_time_ms);
+      render_wait_event_->Wait(new_max_wait_time);
       return NULL;
     }
     // Wait until it's time to render.
@@ -163,9 +169,9 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
   if (frame == NULL) {
     return NULL;
   }
-  frame->SetRenderTime(next_render_time_ms);
-  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(),
-                          "SetRenderTS", "render_time", next_render_time_ms);
+  frame->SetRenderTime(*next_render_time_ms);
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->TimeStamp(), "SetRenderTS",
+                          "render_time", *next_render_time_ms);
   UpdateReceiveState(*frame);
   if (!frame->Complete()) {
     // Update stats for incomplete frames.
@@ -186,8 +192,7 @@ void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
   jitter_buffer_.ReleaseFrame(frame);
 }
 
-void VCMReceiver::ReceiveStatistics(uint32_t* bitrate,
-                                    uint32_t* framerate) {
+void VCMReceiver::ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate) {
   assert(bitrate);
   assert(framerate);
   jitter_buffer_.IncomingRateStatistics(framerate, bitrate);
@@ -209,8 +214,7 @@ void VCMReceiver::SetNackMode(VCMNackMode nackMode,
 void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
                                   int max_packet_age_to_nack,
                                   int max_incomplete_time_ms) {
-  jitter_buffer_.SetNackSettings(max_nack_list_size,
-                                 max_packet_age_to_nack,
+  jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
                                  max_incomplete_time_ms);
 }
 
@@ -219,25 +223,8 @@ VCMNackMode VCMReceiver::NackMode() const {
   return jitter_buffer_.nack_mode();
 }
 
-VCMNackStatus VCMReceiver::NackList(uint16_t* nack_list,
-                                    uint16_t size,
-                                    uint16_t* nack_list_length) {
-  bool request_key_frame = false;
-  uint16_t* internal_nack_list = jitter_buffer_.GetNackList(
-      nack_list_length, &request_key_frame);
-  assert(*nack_list_length <= size);
-  if (internal_nack_list != NULL && *nack_list_length > 0) {
-    memcpy(nack_list, internal_nack_list, *nack_list_length * sizeof(uint16_t));
-  }
-  if (request_key_frame) {
-    return kNackKeyFrameRequest;
-  }
-  return kNackOk;
-}
-
-VCMReceiverState VCMReceiver::State() const {
-  CriticalSectionScoped cs(crit_sect_);
-  return state_;
+std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
+  return jitter_buffer_.GetNackList(request_key_frame);
 }
 
 VideoReceiveState VCMReceiver::ReceiveState() const {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.h b/media/webrtc/trunk/webrtc/modules/video_coding/receiver.h
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/receiver.h
index 2de5539600..03dc334e50 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/receiver.h
@@ -8,49 +8,46 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_RECEIVER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_RECEIVER_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
 
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include <vector>
+
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 
 class Clock;
 class VCMEncodedFrame;
 
-enum VCMNackStatus {
-  kNackOk,
-  kNackKeyFrameRequest
-};
-
-enum VCMReceiverState {
-  kReceiving,
-  kPassive,
-  kWaitForPrimaryDecode
-};
-
 class VCMReceiver {
  public:
+  VCMReceiver(VCMTiming* timing, Clock* clock, EventFactory* event_factory);
+
+  // Using this constructor, you can specify a different event factory for the
+  // jitter buffer. Useful for unit tests when you want to simulate incoming
+  // packets, in which case the jitter buffer's wait event is different from
+  // that of VCMReceiver itself.
   VCMReceiver(VCMTiming* timing,
               Clock* clock,
-              EventFactory* event_factory,
-              bool master);
+              rtc::scoped_ptr<EventWrapper> receiver_event,
+              rtc::scoped_ptr<EventWrapper> jitter_buffer_event);
+
   ~VCMReceiver();
 
   void Reset();
-  int32_t Initialize();
   void UpdateRtt(int64_t rtt);
   int32_t InsertPacket(const VCMPacket& packet,
                        uint16_t frame_width,
                        uint16_t frame_height);
   VCMEncodedFrame* FrameForDecoding(uint16_t max_wait_time_ms,
-                                    int64_t& next_render_time_ms,
-                                    bool render_timing = true);
+                                    int64_t* next_render_time_ms,
+                                    bool prefer_late_decoding);
   void ReleaseFrame(VCMEncodedFrame* frame);
   void ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate);
   uint32_t DiscardedPackets() const;
@@ -63,9 +60,8 @@ class VCMReceiver {
                        int max_packet_age_to_nack,
                        int max_incomplete_time_ms);
   VCMNackMode NackMode() const;
-  VCMNackStatus NackList(uint16_t* nackList, uint16_t size,
-                         uint16_t* nack_list_length);
-  VCMReceiverState State() const;
+  std::vector<uint16_t> NackList(bool* request_key_frame);
+
   VideoReceiveState ReceiveState() const;
 
   // Receiver video delay.
@@ -86,20 +82,16 @@ class VCMReceiver {
 
  private:
   void UpdateReceiveState(const VCMEncodedFrame& frame);
-  static int32_t GenerateReceiverId();
 
   CriticalSectionWrapper* crit_sect_;
   Clock* const clock_;
   VCMJitterBuffer jitter_buffer_;
   VCMTiming* timing_;
   rtc::scoped_ptr<EventWrapper> render_wait_event_;
-  VCMReceiverState state_;
   VideoReceiveState receiveState_;
   int max_video_delay_ms_;
-
-  static int32_t receiver_id_counter_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_RECEIVER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/receiver_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/receiver_unittest.cc
new file mode 100644
index 0000000000..1f3a144bad
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/receiver_unittest.cc
@@ -0,0 +1,575 @@
+/*  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include <list>
+#include <queue>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/receiver.h"
+#include "webrtc/modules/video_coding/test/stream_generator.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+class TestVCMReceiver : public ::testing::Test {
+ protected:
+  enum { kWidth = 640 };
+  enum { kHeight = 480 };
+
+  TestVCMReceiver()
+      : clock_(new SimulatedClock(0)),
+        timing_(clock_.get()),
+        receiver_(&timing_, clock_.get(), &event_factory_) {
+    stream_generator_.reset(
+        new StreamGenerator(0, clock_->TimeInMilliseconds()));
+  }
+
+  virtual void SetUp() { receiver_.Reset(); }
+
+  int32_t InsertPacket(int index) {
+    VCMPacket packet;
+    bool packet_available = stream_generator_->GetPacket(&packet, index);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return kGeneralError;  // Return here to avoid crashes below.
+    return receiver_.InsertPacket(packet, kWidth, kHeight);
+  }
+
+  int32_t InsertPacketAndPop(int index) {
+    VCMPacket packet;
+    bool packet_available = stream_generator_->PopPacket(&packet, index);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return kGeneralError;  // Return here to avoid crashes below.
+    return receiver_.InsertPacket(packet, kWidth, kHeight);
+  }
+
+  int32_t InsertFrame(FrameType frame_type, bool complete) {
+    int num_of_packets = complete ? 1 : 2;
+    stream_generator_->GenerateFrame(
+        frame_type, (frame_type != kEmptyFrame) ? num_of_packets : 0,
+        (frame_type == kEmptyFrame) ? 1 : 0, clock_->TimeInMilliseconds());
+    int32_t ret = InsertPacketAndPop(0);
+    if (!complete) {
+      // Drop the second packet.
+      VCMPacket packet;
+      stream_generator_->PopPacket(&packet, 0);
+    }
+    clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+    return ret;
+  }
+
+  bool DecodeNextFrame() {
+    int64_t render_time_ms = 0;
+    VCMEncodedFrame* frame =
+        receiver_.FrameForDecoding(0, &render_time_ms, false);
+    if (!frame)
+      return false;
+    receiver_.ReleaseFrame(frame);
+    return true;
+  }
+
+  rtc::scoped_ptr<SimulatedClock> clock_;
+  VCMTiming timing_;
+  NullEventFactory event_factory_;
+  VCMReceiver receiver_;
+  rtc::scoped_ptr<StreamGenerator> stream_generator_;
+};
+
+TEST_F(TestVCMReceiver, RenderBufferSize_AllComplete) {
+  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  int num_of_frames = 10;
+  for (int i = 0; i < num_of_frames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  EXPECT_EQ(num_of_frames * kDefaultFramePeriodMs,
+            receiver_.RenderBufferSizeMs());
+}
+
+TEST_F(TestVCMReceiver, RenderBufferSize_SkipToKeyFrame) {
+  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
+  const int kNumOfNonDecodableFrames = 2;
+  for (int i = 0; i < kNumOfNonDecodableFrames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  const int kNumOfFrames = 10;
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  for (int i = 0; i < kNumOfFrames - 1; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  EXPECT_EQ((kNumOfFrames - 1) * kDefaultFramePeriodMs,
+            receiver_.RenderBufferSizeMs());
+}
+
+TEST_F(TestVCMReceiver, RenderBufferSize_NotAllComplete) {
+  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  int num_of_frames = 10;
+  for (int i = 0; i < num_of_frames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  num_of_frames++;
+  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
+  for (int i = 0; i < num_of_frames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  EXPECT_EQ((num_of_frames - 1) * kDefaultFramePeriodMs,
+            receiver_.RenderBufferSizeMs());
+}
+
+TEST_F(TestVCMReceiver, RenderBufferSize_NoKeyFrame) {
+  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
+  int num_of_frames = 10;
+  for (int i = 0; i < num_of_frames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  int64_t next_render_time_ms = 0;
+  VCMEncodedFrame* frame =
+      receiver_.FrameForDecoding(10, &next_render_time_ms, false);
+  EXPECT_TRUE(frame == NULL);
+  receiver_.ReleaseFrame(frame);
+  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
+  for (int i = 0; i < num_of_frames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_Empty) {
+  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
+  receiver_.SetNackMode(kNack, -1, -1);
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  // Advance time until it's time to decode the key frame.
+  clock_->AdvanceTimeMilliseconds(kMinDelayMs);
+  EXPECT_TRUE(DecodeNextFrame());
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_NoKeyFrame) {
+  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
+  receiver_.SetNackMode(kNack, -1, -1);
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  const int kNumFrames = kDefaultFrameRate * kMaxNonDecodableDuration / 1000;
+  for (int i = 0; i < kNumFrames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_TRUE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_OneIncomplete) {
+  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
+  receiver_.SetNackMode(kNack, -1, -1);
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  receiver_.SetMinReceiverDelay(kMinDelayMs);
+  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
+  // Insert enough frames to have too long non-decodable sequence.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  // Advance time until it's time to decode the key frame.
+  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
+                                  key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we get a key frame request.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_TRUE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger) {
+  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
+  receiver_.SetNackMode(kNack, -1, -1);
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  receiver_.SetMinReceiverDelay(kMinDelayMs);
+  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
+  // Insert all but one frame to not trigger a key frame request due to
+  // too long duration of non-decodable frames.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames - 1; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  // Advance time until it's time to decode the key frame.
+  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
+                                  key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we don't get a key frame request since we haven't generated
+  // enough frames.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger2) {
+  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
+  receiver_.SetNackMode(kNack, -1, -1);
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  receiver_.SetMinReceiverDelay(kMinDelayMs);
+  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  // Insert enough frames to have too long non-decodable sequence, except that
+  // we don't have any losses.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
+  // Advance time until it's time to decode the key frame.
+  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
+                                  key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we don't get a key frame request since the non-decodable duration
+  // is only one frame.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_KeyFrameAfterIncompleteFrames) {
+  // Enable NACK and with no RTT thresholds for disabling retransmission delay.
+  receiver_.SetNackMode(kNack, -1, -1);
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  receiver_.SetMinReceiverDelay(kMinDelayMs);
+  int64_t key_frame_inserted = clock_->TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
+  // Insert enough frames to have too long non-decodable sequence.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
+    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
+  }
+  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
+  // Advance time until it's time to decode the key frame.
+  clock_->AdvanceTimeMilliseconds(kMinDelayMs - clock_->TimeInMilliseconds() -
+                                  key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we don't get a key frame request since we have a key frame
+  // in the list.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+// A simulated clock, when time elapses, will insert frames into the jitter
+// buffer, based on initial settings.
+class SimulatedClockWithFrames : public SimulatedClock {
+ public:
+  SimulatedClockWithFrames(StreamGenerator* stream_generator,
+                           VCMReceiver* receiver)
+      : SimulatedClock(0),
+        stream_generator_(stream_generator),
+        receiver_(receiver) {}
+  virtual ~SimulatedClockWithFrames() {}
+
+  // If |stop_on_frame| is true and next frame arrives between now and
+  // now+|milliseconds|, the clock will be advanced to the arrival time of next
+  // frame.
+  // Otherwise, the clock will be advanced by |milliseconds|.
+  //
+  // For both cases, a frame will be inserted into the jitter buffer at the
+  // instant when the clock time is timestamps_.front().arrive_time.
+  //
+  // Return true if some frame arrives between now and now+|milliseconds|.
+  bool AdvanceTimeMilliseconds(int64_t milliseconds, bool stop_on_frame) {
+    return AdvanceTimeMicroseconds(milliseconds * 1000, stop_on_frame);
+  }
+
+  bool AdvanceTimeMicroseconds(int64_t microseconds, bool stop_on_frame) {
+    int64_t start_time = TimeInMicroseconds();
+    int64_t end_time = start_time + microseconds;
+    bool frame_injected = false;
+    while (!timestamps_.empty() &&
+           timestamps_.front().arrive_time <= end_time) {
+      RTC_DCHECK(timestamps_.front().arrive_time >= start_time);
+
+      SimulatedClock::AdvanceTimeMicroseconds(timestamps_.front().arrive_time -
+                                              TimeInMicroseconds());
+      GenerateAndInsertFrame((timestamps_.front().render_time + 500) / 1000);
+      timestamps_.pop();
+      frame_injected = true;
+
+      if (stop_on_frame)
+        return frame_injected;
+    }
+
+    if (TimeInMicroseconds() < end_time) {
+      SimulatedClock::AdvanceTimeMicroseconds(end_time - TimeInMicroseconds());
+    }
+    return frame_injected;
+  }
+
+  // Input timestamps are in unit Milliseconds.
+  // And |arrive_timestamps| must be positive and in increasing order.
+  // |arrive_timestamps| determine when we are going to insert frames into the
+  // jitter buffer.
+  // |render_timestamps| are the timestamps on the frame.
+  void SetFrames(const int64_t* arrive_timestamps,
+                 const int64_t* render_timestamps,
+                 size_t size) {
+    int64_t previous_arrive_timestamp = 0;
+    for (size_t i = 0; i < size; i++) {
+      RTC_CHECK(arrive_timestamps[i] >= previous_arrive_timestamp);
+      timestamps_.push(TimestampPair(arrive_timestamps[i] * 1000,
+                                     render_timestamps[i] * 1000));
+      previous_arrive_timestamp = arrive_timestamps[i];
+    }
+  }
+
+ private:
+  struct TimestampPair {
+    TimestampPair(int64_t arrive_timestamp, int64_t render_timestamp)
+        : arrive_time(arrive_timestamp), render_time(render_timestamp) {}
+
+    int64_t arrive_time;
+    int64_t render_time;
+  };
+
+  void GenerateAndInsertFrame(int64_t render_timestamp_ms) {
+    VCMPacket packet;
+    stream_generator_->GenerateFrame(FrameType::kVideoFrameKey,
+                                     1,  // media packets
+                                     0,  // empty packets
+                                     render_timestamp_ms);
+
+    bool packet_available = stream_generator_->PopPacket(&packet, 0);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return;  // Return here to avoid crashes below.
+    receiver_->InsertPacket(packet, 640, 480);
+  }
+
+  std::queue<TimestampPair> timestamps_;
+  StreamGenerator* stream_generator_;
+  VCMReceiver* receiver_;
+};
+
+// Use a SimulatedClockWithFrames
+// Wait call will do either of these:
+// 1. If |stop_on_frame| is true, the clock will be turned to the exact instant
+// that the first frame comes and the frame will be inserted into the jitter
+// buffer, or the clock will be turned to now + |max_time| if no frame comes in
+// the window.
+// 2. If |stop_on_frame| is false, the clock will be turn to now + |max_time|,
+// and all the frames arriving between now and now + |max_time| will be
+// inserted into the jitter buffer.
+//
+// This is used to simulate the JitterBuffer getting packets from internet as
+// time elapses.
+
+class FrameInjectEvent : public EventWrapper {
+ public:
+  FrameInjectEvent(SimulatedClockWithFrames* clock, bool stop_on_frame)
+      : clock_(clock), stop_on_frame_(stop_on_frame) {}
+
+  bool Set() override { return true; }
+
+  EventTypeWrapper Wait(unsigned long max_time) override {  // NOLINT
+    if (clock_->AdvanceTimeMilliseconds(max_time, stop_on_frame_) &&
+        stop_on_frame_) {
+      return EventTypeWrapper::kEventSignaled;
+    } else {
+      return EventTypeWrapper::kEventTimeout;
+    }
+  }
+
+ private:
+  SimulatedClockWithFrames* clock_;
+  bool stop_on_frame_;
+};
+
+class VCMReceiverTimingTest : public ::testing::Test {
+ protected:
+  VCMReceiverTimingTest()
+
+      : clock_(&stream_generator_, &receiver_),
+        stream_generator_(0, clock_.TimeInMilliseconds()),
+        timing_(&clock_),
+        receiver_(
+            &timing_,
+            &clock_,
+            rtc::scoped_ptr<EventWrapper>(new FrameInjectEvent(&clock_, false)),
+            rtc::scoped_ptr<EventWrapper>(
+                new FrameInjectEvent(&clock_, true))) {}
+
+  virtual void SetUp() { receiver_.Reset(); }
+
+  SimulatedClockWithFrames clock_;
+  StreamGenerator stream_generator_;
+  VCMTiming timing_;
+  VCMReceiver receiver_;
+};
+
+// Test whether VCMReceiver::FrameForDecoding handles parameter
+// |max_wait_time_ms| correctly:
+// 1. The function execution should never take more than |max_wait_time_ms|.
+// 2. If the function exit before now + |max_wait_time_ms|, a frame must be
+//    returned.
+TEST_F(VCMReceiverTimingTest, FrameForDecoding) {
+  const size_t kNumFrames = 100;
+  const int kFramePeriod = 40;
+  int64_t arrive_timestamps[kNumFrames];
+  int64_t render_timestamps[kNumFrames];
+  int64_t next_render_time;
+
+  // Construct test samples.
+  // render_timestamps are the timestamps stored in the Frame;
+  // arrive_timestamps controls when the Frame packet got received.
+  for (size_t i = 0; i < kNumFrames; i++) {
+    // Preset frame rate to 25Hz.
+    // But we add a reasonable deviation to arrive_timestamps to mimic Internet
+    // fluctuation.
+    arrive_timestamps[i] =
+        (i + 1) * kFramePeriod + (i % 10) * ((i % 2) ? 1 : -1);
+    render_timestamps[i] = (i + 1) * kFramePeriod;
+  }
+
+  clock_.SetFrames(arrive_timestamps, render_timestamps, kNumFrames);
+
+  // Record how many frames we finally get out of the receiver.
+  size_t num_frames_return = 0;
+
+  const int64_t kMaxWaitTime = 30;
+
+  // Ideally, we should get all frames that we input in InitializeFrames.
+  // In the case that FrameForDecoding kills frames by error, we rely on the
+  // build bot to kill the test.
+  while (num_frames_return < kNumFrames) {
+    int64_t start_time = clock_.TimeInMilliseconds();
+    VCMEncodedFrame* frame =
+        receiver_.FrameForDecoding(kMaxWaitTime, &next_render_time, false);
+    int64_t end_time = clock_.TimeInMilliseconds();
+
+    // In any case the FrameForDecoding should not wait longer than
+    // max_wait_time.
+    // In the case that we did not get a frame, it should have been waiting for
+    // exactly max_wait_time. (By the testing samples we constructed above, we
+    // are sure there is no timing error, so the only case it returns with NULL
+    // is that it runs out of time.)
+    if (frame) {
+      receiver_.ReleaseFrame(frame);
+      ++num_frames_return;
+      EXPECT_GE(kMaxWaitTime, end_time - start_time);
+    } else {
+      EXPECT_EQ(kMaxWaitTime, end_time - start_time);
+    }
+  }
+}
+
+// Test whether VCMReceiver::FrameForDecoding handles parameter
+// |prefer_late_decoding| and |max_wait_time_ms| correctly:
+// 1. The function execution should never take more than |max_wait_time_ms|.
+// 2. If the function exit before now + |max_wait_time_ms|, a frame must be
+//    returned and the end time must be equal to the render timestamp - delay
+//    for decoding and rendering.
+TEST_F(VCMReceiverTimingTest, FrameForDecodingPreferLateDecoding) {
+  const size_t kNumFrames = 100;
+  const int kFramePeriod = 40;
+
+  int64_t arrive_timestamps[kNumFrames];
+  int64_t render_timestamps[kNumFrames];
+  int64_t next_render_time;
+
+  int render_delay_ms;
+  int max_decode_ms;
+  int dummy;
+  timing_.GetTimings(&dummy, &max_decode_ms, &dummy, &dummy, &dummy, &dummy,
+                     &render_delay_ms);
+
+  // Construct test samples.
+  // render_timestamps are the timestamps stored in the Frame;
+  // arrive_timestamps controls when the Frame packet got received.
+  for (size_t i = 0; i < kNumFrames; i++) {
+    // Preset frame rate to 25Hz.
+    // But we add a reasonable deviation to arrive_timestamps to mimic Internet
+    // fluctuation.
+    arrive_timestamps[i] =
+        (i + 1) * kFramePeriod + (i % 10) * ((i % 2) ? 1 : -1);
+    render_timestamps[i] = (i + 1) * kFramePeriod;
+  }
+
+  clock_.SetFrames(arrive_timestamps, render_timestamps, kNumFrames);
+
+  // Record how many frames we finally get out of the receiver.
+  size_t num_frames_return = 0;
+  const int64_t kMaxWaitTime = 30;
+  bool prefer_late_decoding = true;
+  while (num_frames_return < kNumFrames) {
+    int64_t start_time = clock_.TimeInMilliseconds();
+
+    VCMEncodedFrame* frame = receiver_.FrameForDecoding(
+        kMaxWaitTime, &next_render_time, prefer_late_decoding);
+    int64_t end_time = clock_.TimeInMilliseconds();
+    if (frame) {
+      EXPECT_EQ(frame->RenderTimeMs() - max_decode_ms - render_delay_ms,
+                end_time);
+      receiver_.ReleaseFrame(frame);
+      ++num_frames_return;
+    } else {
+      EXPECT_EQ(kMaxWaitTime, end_time - start_time);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.cc b/media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.cc
new file mode 100644
index 0000000000..742f70f1c1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.cc
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/rtt_filter.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "webrtc/modules/video_coding/internal_defines.h"
+
+namespace webrtc {
+
+VCMRttFilter::VCMRttFilter()
+    : _filtFactMax(35),
+      _jumpStdDevs(2.5),
+      _driftStdDevs(3.5),
+      _detectThreshold(kMaxDriftJumpCount) {
+  Reset();
+}
+
+VCMRttFilter& VCMRttFilter::operator=(const VCMRttFilter& rhs) {
+  if (this != &rhs) {
+    _gotNonZeroUpdate = rhs._gotNonZeroUpdate;
+    _avgRtt = rhs._avgRtt;
+    _varRtt = rhs._varRtt;
+    _maxRtt = rhs._maxRtt;
+    _filtFactCount = rhs._filtFactCount;
+    _jumpCount = rhs._jumpCount;
+    _driftCount = rhs._driftCount;
+    memcpy(_jumpBuf, rhs._jumpBuf, sizeof(_jumpBuf));
+    memcpy(_driftBuf, rhs._driftBuf, sizeof(_driftBuf));
+  }
+  return *this;
+}
+
+void VCMRttFilter::Reset() {
+  _gotNonZeroUpdate = false;
+  _avgRtt = 0;
+  _varRtt = 0;
+  _maxRtt = 0;
+  _filtFactCount = 1;
+  _jumpCount = 0;
+  _driftCount = 0;
+  memset(_jumpBuf, 0, kMaxDriftJumpCount);
+  memset(_driftBuf, 0, kMaxDriftJumpCount);
+}
+
+void VCMRttFilter::Update(int64_t rttMs) {
+  if (!_gotNonZeroUpdate) {
+    if (rttMs == 0) {
+      return;
+    }
+    _gotNonZeroUpdate = true;
+  }
+
+  // Sanity check
+  if (rttMs > 3000) {
+    rttMs = 3000;
+  }
+
+  double filtFactor = 0;
+  if (_filtFactCount > 1) {
+    filtFactor = static_cast<double>(_filtFactCount - 1) / _filtFactCount;
+  }
+  _filtFactCount++;
+  if (_filtFactCount > _filtFactMax) {
+    // This prevents filtFactor from going above
+    // (_filtFactMax - 1) / _filtFactMax,
+    // e.g., _filtFactMax = 50 => filtFactor = 49/50 = 0.98
+    _filtFactCount = _filtFactMax;
+  }
+  double oldAvg = _avgRtt;
+  double oldVar = _varRtt;
+  _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
+  _varRtt = filtFactor * _varRtt +
+            (1 - filtFactor) * (rttMs - _avgRtt) * (rttMs - _avgRtt);
+  _maxRtt = VCM_MAX(rttMs, _maxRtt);
+  if (!JumpDetection(rttMs) || !DriftDetection(rttMs)) {
+    // In some cases we don't want to update the statistics
+    _avgRtt = oldAvg;
+    _varRtt = oldVar;
+  }
+}
+
+bool VCMRttFilter::JumpDetection(int64_t rttMs) {
+  double diffFromAvg = _avgRtt - rttMs;
+  if (fabs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt)) {
+    int diffSign = (diffFromAvg >= 0) ? 1 : -1;
+    int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;
+    if (diffSign != jumpCountSign) {
+      // Since the signs differ the samples currently
+      // in the buffer is useless as they represent a
+      // jump in a different direction.
+      _jumpCount = 0;
+    }
+    if (abs(_jumpCount) < kMaxDriftJumpCount) {
+      // Update the buffer used for the short time
+      // statistics.
+      // The sign of the diff is used for updating the counter since
+      // we want to use the same buffer for keeping track of when
+      // the RTT jumps down and up.
+      _jumpBuf[abs(_jumpCount)] = rttMs;
+      _jumpCount += diffSign;
+    }
+    if (abs(_jumpCount) >= _detectThreshold) {
+      // Detected an RTT jump
+      ShortRttFilter(_jumpBuf, abs(_jumpCount));
+      _filtFactCount = _detectThreshold + 1;
+      _jumpCount = 0;
+    } else {
+      return false;
+    }
+  } else {
+    _jumpCount = 0;
+  }
+  return true;
+}
+
+bool VCMRttFilter::DriftDetection(int64_t rttMs) {
+  if (_maxRtt - _avgRtt > _driftStdDevs * sqrt(_varRtt)) {
+    if (_driftCount < kMaxDriftJumpCount) {
+      // Update the buffer used for the short time
+      // statistics.
+      _driftBuf[_driftCount] = rttMs;
+      _driftCount++;
+    }
+    if (_driftCount >= _detectThreshold) {
+      // Detected an RTT drift
+      ShortRttFilter(_driftBuf, _driftCount);
+      _filtFactCount = _detectThreshold + 1;
+      _driftCount = 0;
+    }
+  } else {
+    _driftCount = 0;
+  }
+  return true;
+}
+
+void VCMRttFilter::ShortRttFilter(int64_t* buf, uint32_t length) {
+  if (length == 0) {
+    return;
+  }
+  _maxRtt = 0;
+  _avgRtt = 0;
+  for (uint32_t i = 0; i < length; i++) {
+    if (buf[i] > _maxRtt) {
+      _maxRtt = buf[i];
+    }
+    _avgRtt += buf[i];
+  }
+  _avgRtt = _avgRtt / static_cast<double>(length);
+}
+
+int64_t VCMRttFilter::RttMs() const {
+  return static_cast<int64_t>(_maxRtt + 0.5);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.h b/media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.h
new file mode 100644
index 0000000000..f5de532cfc
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMRttFilter {
+ public:
+  VCMRttFilter();
+
+  VCMRttFilter& operator=(const VCMRttFilter& rhs);
+
+  // Resets the filter.
+  void Reset();
+  // Updates the filter with a new sample.
+  void Update(int64_t rttMs);
+  // A getter function for the current RTT level in ms.
+  int64_t RttMs() const;
+
+ private:
+  // The size of the drift and jump memory buffers
+  // and thus also the detection threshold for these
+  // detectors in number of samples.
+  enum { kMaxDriftJumpCount = 5 };
+  // Detects RTT jumps by comparing the difference between
+  // samples and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool JumpDetection(int64_t rttMs);
+  // Detects RTT drifts by comparing the difference between
+  // max and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool DriftDetection(int64_t rttMs);
+  // Computes the short time average and maximum of the vector buf.
+  void ShortRttFilter(int64_t* buf, uint32_t length);
+
+  bool _gotNonZeroUpdate;
+  double _avgRtt;
+  double _varRtt;
+  int64_t _maxRtt;
+  uint32_t _filtFactCount;
+  const uint32_t _filtFactMax;
+  const double _jumpStdDevs;
+  const double _driftStdDevs;
+  int32_t _jumpCount;
+  int32_t _driftCount;
+  const int32_t _detectThreshold;
+  int64_t _jumpBuf[kMaxDriftJumpCount];
+  int64_t _driftBuf[kMaxDriftJumpCount];
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.cc b/media/webrtc/trunk/webrtc/modules/video_coding/session_info.cc
similarity index 89%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/session_info.cc
index 811d107972..227100e940 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/session_info.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/session_info.h"
+#include "webrtc/modules/video_coding/session_info.h"
 
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/packet.h"
 
 namespace webrtc {
 
@@ -32,8 +32,7 @@ VCMSessionInfo::VCMSessionInfo()
       empty_seq_num_low_(-1),
       empty_seq_num_high_(-1),
       first_packet_seq_num_(-1),
-      last_packet_seq_num_(-1) {
-}
+      last_packet_seq_num_(-1) {}
 
 void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr,
                                         const uint8_t* new_base_ptr) {
@@ -88,8 +87,8 @@ bool VCMSessionInfo::LayerSync() const {
   if (packets_.front().codecSpecificHeader.codec == kRtpVideoVp8) {
     return packets_.front().codecSpecificHeader.codecHeader.VP8.layerSync;
   } else if (packets_.front().codecSpecificHeader.codec == kRtpVideoVp9) {
-    return
-        packets_.front().codecSpecificHeader.codecHeader.VP9.temporal_up_switch;
+    return packets_.front()
+        .codecSpecificHeader.codecHeader.VP9.temporal_up_switch;
   } else {
     return false;
   }
@@ -172,6 +171,8 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
 
   // We handle H.264 STAP-A packets in a special way as we need to remove the
   // two length bytes between each NAL unit, and potentially add start codes.
+  // TODO(pbos): Remove H264 parsing from this step and use a fragmentation
+  // header supplied by the H264 depacketizer.
   const size_t kH264NALHeaderLengthInBytes = 1;
   const size_t kLengthFieldLength = 2;
   if (packet.codecSpecificHeader.codec == kRtpVideoH264 &&
@@ -193,11 +194,6 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
         return 0;
       }
     }
-    if (required_length > packet.sizeBytes + kBufferSafetyMargin) {
-      LOG(LS_ERROR) << "Failed to insert packet due to too many NALs in a STAP-A";
-      return 0;
-    }
-
     ShiftSubsequentPackets(packet_it, required_length);
     nalu_ptr = packet_buffer + kH264NALHeaderLengthInBytes;
     uint8_t* frame_buffer_ptr = frame_buffer + offset;
@@ -205,9 +201,7 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
     while (nalu_ptr + kLengthFieldLength <= packet_buffer + packet.sizeBytes) {
       size_t length = BufferToUWord16(nalu_ptr);
       nalu_ptr += kLengthFieldLength;
-      frame_buffer_ptr += Insert(nalu_ptr,
-                                 length,
-                                 packet.insertStartCode,
+      frame_buffer_ptr += Insert(nalu_ptr, length, packet.insertStartCode,
                                  const_cast<uint8_t*>(frame_buffer_ptr));
       nalu_ptr += length;
     }
@@ -215,14 +209,12 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
     return packet.sizeBytes;
   }
   ShiftSubsequentPackets(
-      packet_it,
-      packet.sizeBytes +
-          (packet.insertStartCode ? kH264StartCodeLengthBytes : 0));
+      packet_it, packet.sizeBytes +
+                     (packet.insertStartCode ? kH264StartCodeLengthBytes : 0));
 
-  packet.sizeBytes = Insert(packet_buffer,
-                            packet.sizeBytes,
-                            packet.insertStartCode,
-                            const_cast<uint8_t*>(packet.dataPtr));
+  packet.sizeBytes =
+      Insert(packet_buffer, packet.sizeBytes, packet.insertStartCode,
+             const_cast<uint8_t*>(packet.dataPtr));
   return packet.sizeBytes;
 }
 
@@ -235,8 +227,7 @@ size_t VCMSessionInfo::Insert(const uint8_t* buffer,
     memcpy(frame_buffer, startCode, kH264StartCodeLengthBytes);
   }
   memcpy(frame_buffer + (insert_start_code ? kH264StartCodeLengthBytes : 0),
-         buffer,
-         length);
+         buffer, length);
   length += (insert_start_code ? kH264StartCodeLengthBytes : 0);
 
   return length;
@@ -288,13 +279,12 @@ void VCMSessionInfo::UpdateDecodableSession(const FrameData& frame_data) {
   // thresholds.
   const float kLowPacketPercentageThreshold = 0.2f;
   const float kHighPacketPercentageThreshold = 0.8f;
-  if (frame_data.rtt_ms < kRttThreshold
-      || frame_type_ == kVideoFrameKey
-      || !HaveFirstPacket()
-      || (NumPackets() <= kHighPacketPercentageThreshold
-                          * frame_data.rolling_average_packets_per_frame
-          && NumPackets() > kLowPacketPercentageThreshold
-                            * frame_data.rolling_average_packets_per_frame))
+  if (frame_data.rtt_ms < kRttThreshold || frame_type_ == kVideoFrameKey ||
+      !HaveFirstPacket() ||
+      (NumPackets() <= kHighPacketPercentageThreshold *
+                           frame_data.rolling_average_packets_per_frame &&
+       NumPackets() > kLowPacketPercentageThreshold *
+                          frame_data.rolling_average_packets_per_frame))
     return;
 
   decodable_ = true;
@@ -320,7 +310,7 @@ VCMSessionInfo::PacketIterator VCMSessionInfo::FindNaluEnd(
   // Find the end of the NAL unit.
   for (; packet_it != packets_.end(); ++packet_it) {
     if (((*packet_it).completeNALU == kNaluComplete &&
-        (*packet_it).sizeBytes > 0) ||
+         (*packet_it).sizeBytes > 0) ||
         // Found next NALU.
         (*packet_it).completeNALU == kNaluStart)
       return --packet_it;
@@ -360,7 +350,7 @@ size_t VCMSessionInfo::BuildVP8FragmentationHeader(
   memset(fragmentation->fragmentationLength, 0,
          kMaxVP8Partitions * sizeof(size_t));
   if (packets_.empty())
-      return new_length;
+    return new_length;
   PacketIterator it = FindNextPartitionBeginning(packets_.begin());
   while (it != packets_.end()) {
     const int partition_id =
@@ -383,7 +373,7 @@ size_t VCMSessionInfo::BuildVP8FragmentationHeader(
   // Set all empty fragments to start where the previous fragment ends,
   // and have zero length.
   if (fragmentation->fragmentationLength[0] == 0)
-      fragmentation->fragmentationOffset[0] = 0;
+    fragmentation->fragmentationOffset[0] = 0;
   for (int i = 1; i < fragmentation->fragmentationVectorSize; ++i) {
     if (fragmentation->fragmentationLength[i] == 0)
       fragmentation->fragmentationOffset[i] =
@@ -391,7 +381,7 @@ size_t VCMSessionInfo::BuildVP8FragmentationHeader(
           fragmentation->fragmentationLength[i - 1];
     assert(i == 0 ||
            fragmentation->fragmentationOffset[i] >=
-           fragmentation->fragmentationOffset[i - 1]);
+               fragmentation->fragmentationOffset[i - 1]);
   }
   assert(new_length <= frame_buffer_length);
   return new_length;
@@ -436,8 +426,8 @@ bool VCMSessionInfo::InSequence(const PacketIterator& packet_it,
   // If the two iterators are pointing to the same packet they are considered
   // to be in sequence.
   return (packet_it == prev_packet_it ||
-      (static_cast<uint16_t>((*prev_packet_it).seqNum + 1) ==
-          (*packet_it).seqNum));
+          (static_cast<uint16_t>((*prev_packet_it).seqNum + 1) ==
+           (*packet_it).seqNum));
 }
 
 size_t VCMSessionInfo::MakeDecodable() {
@@ -447,8 +437,7 @@ size_t VCMSessionInfo::MakeDecodable() {
   }
   PacketIterator it = packets_.begin();
   // Make sure we remove the first NAL unit if it's not decodable.
-  if ((*it).completeNALU == kNaluIncomplete ||
-      (*it).completeNALU == kNaluEnd) {
+  if ((*it).completeNALU == kNaluIncomplete || (*it).completeNALU == kNaluEnd) {
     PacketIterator nalu_end = FindNaluEnd(it);
     return_length += DeletePacketData(it, nalu_end);
     it = nalu_end;
@@ -457,7 +446,7 @@ size_t VCMSessionInfo::MakeDecodable() {
   // Take care of the rest of the NAL units.
   for (; it != packets_.end(); ++it) {
     bool start_of_nalu = ((*it).completeNALU == kNaluStart ||
-        (*it).completeNALU == kNaluComplete);
+                          (*it).completeNALU == kNaluComplete);
     if (!start_of_nalu && !InSequence(it, prev_it)) {
       // Found a sequence number gap due to packet loss.
       PacketIterator nalu_end = FindNaluEnd(it);
@@ -475,18 +464,15 @@ void VCMSessionInfo::SetNotDecodableIfIncomplete() {
   decodable_ = false;
 }
 
-bool
-VCMSessionInfo::HaveFirstPacket() const {
+bool VCMSessionInfo::HaveFirstPacket() const {
   return !packets_.empty() && (first_packet_seq_num_ != -1);
 }
 
-bool
-VCMSessionInfo::HaveLastPacket() const {
+bool VCMSessionInfo::HaveLastPacket() const {
   return !packets_.empty() && (last_packet_seq_num_ != -1);
 }
 
-bool
-VCMSessionInfo::session_nack() const {
+bool VCMSessionInfo::session_nack() const {
   return session_nack_;
 }
 
@@ -494,7 +480,7 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
                                  uint8_t* frame_buffer,
                                  VCMDecodeErrorMode decode_error_mode,
                                  const FrameData& frame_data) {
-  if (packet.frameType == kFrameEmpty) {
+  if (packet.frameType == kEmptyFrame) {
     // Update sequence number of an empty packet.
     // Only media packets are inserted into the packet list.
     InformOfEmptyPacket(packet.seqNum);
@@ -514,8 +500,8 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
       break;
 
   // Check for duplicate packets.
-  if (rit != packets_.rend() &&
-      (*rit).seqNum == packet.seqNum && (*rit).sizeBytes > 0)
+  if (rit != packets_.rend() && (*rit).seqNum == packet.seqNum &&
+      (*rit).sizeBytes > 0)
     return -2;
 
   if (packet.codec == kVideoCodecH264) {
@@ -557,7 +543,7 @@ int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
       LOG(LS_WARNING) << "Received packet with a sequence number which is out "
                          "of frame boundaries";
       return -3;
-    } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
+    } else if (frame_type_ == kEmptyFrame && packet.frameType != kEmptyFrame) {
       // Update the frame type with the type of the first media packet.
       // TODO(mikhal): Can this trigger?
       frame_type_ = packet.frameType;
@@ -598,8 +584,8 @@ void VCMSessionInfo::InformOfEmptyPacket(uint16_t seq_num) {
     empty_seq_num_high_ = seq_num;
   else
     empty_seq_num_high_ = LatestSequenceNumber(seq_num, empty_seq_num_high_);
-  if (empty_seq_num_low_ == -1 || IsNewerSequenceNumber(empty_seq_num_low_,
-                                                        seq_num))
+  if (empty_seq_num_low_ == -1 ||
+      IsNewerSequenceNumber(empty_seq_num_low_, seq_num))
     empty_seq_num_low_ = seq_num;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.h b/media/webrtc/trunk/webrtc/modules/video_coding/session_info.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/session_info.h
index 88071e19d5..e9ff25166d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/session_info.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_SESSION_INFO_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_SESSION_INFO_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
+#define WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
 
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -116,8 +116,7 @@ class VCMSessionInfo {
   PacketIterator FindPartitionEnd(PacketIterator it) const;
   static bool InSequence(const PacketIterator& it,
                          const PacketIterator& prev_it);
-  size_t InsertBuffer(uint8_t* frame_buffer,
-                      PacketIterator packetIterator);
+  size_t InsertBuffer(uint8_t* frame_buffer, PacketIterator packetIterator);
   size_t Insert(const uint8_t* buffer,
                 size_t length,
                 bool insert_start_code,
@@ -126,8 +125,7 @@ class VCMSessionInfo {
   PacketIterator FindNaluEnd(PacketIterator packet_iter) const;
   // Deletes the data of all packets between |start| and |end|, inclusively.
   // Note that this function doesn't delete the actual packets.
-  size_t DeletePacketData(PacketIterator start,
-                          PacketIterator end);
+  size_t DeletePacketData(PacketIterator start, PacketIterator end);
   void UpdateCompleteSession();
 
   // When enabled, determine if session is decodable, i.e. incomplete but
@@ -169,4 +167,4 @@ class VCMSessionInfo {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_SESSION_INFO_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/session_info_unittest.cc
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/session_info_unittest.cc
index fae55f4252..4019d63a5f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/session_info_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/session_info_unittest.cc
@@ -11,9 +11,9 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/session_info.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/session_info.h"
 
 namespace webrtc {
 
@@ -81,7 +81,7 @@ class TestVP8Partitions : public TestSessionInfo {
               fragmentation_.fragmentationLength[partition_id]);
     for (int i = 0; i < packets_expected; ++i) {
       size_t packet_index = fragmentation_.fragmentationOffset[partition_id] +
-          i * packet_buffer_size();
+                            i * packet_buffer_size();
       if (packet_index + packet_buffer_size() > frame_buffer_size())
         return false;
       VerifyPacket(frame_buffer_ + packet_index, start_value + i);
@@ -122,8 +122,7 @@ class TestNackList : public TestSessionInfo {
     memset(seq_num_list_, 0, sizeof(seq_num_list_));
   }
 
-  void BuildSeqNumList(uint16_t low,
-                       uint16_t high) {
+  void BuildSeqNumList(uint16_t low, uint16_t high) {
     size_t i = 0;
     while (low != high + 1) {
       EXPECT_LT(i, kMaxSeqNumListLength);
@@ -173,14 +172,11 @@ TEST_F(TestSessionInfo, TestSimpleAPIs) {
   // To make things more difficult we will make sure to have a wrap here.
   packet_.isFirstPacket = false;
   packet_.markerBit = true;
-  packet_.seqNum  = 2;
+  packet_.seqNum = 2;
   packet_.sizeBytes = 0;
-  packet_.frameType = kFrameEmpty;
-  EXPECT_EQ(0,
-            session_.InsertPacket(packet_,
-                                  frame_buffer_,
-                                  kNoErrors,
-                                  frame_data));
+  packet_.frameType = kEmptyFrame;
+  EXPECT_EQ(
+      0, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
 }
 
@@ -198,9 +194,8 @@ TEST_F(TestSessionInfo, NormalOperation) {
     packet_.seqNum += 1;
     FillPacket(i);
     ASSERT_EQ(packet_buffer_size(),
-              static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                        kNoErrors,
-                                                        frame_data)));
+              static_cast<size_t>(session_.InsertPacket(
+                  packet_, frame_buffer_, kNoErrors, frame_data)));
   }
 
   packet_.seqNum += 1;
@@ -223,9 +218,8 @@ TEST_F(TestSessionInfo, ErrorsEqualDecodableState) {
   packet_.markerBit = false;
   FillPacket(3);
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kWithErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kWithErrors, frame_data)));
   EXPECT_TRUE(session_.decodable());
 }
 
@@ -237,18 +231,16 @@ TEST_F(TestSessionInfo, SelectiveDecodableState) {
   frame_data.rolling_average_packets_per_frame = 11;
   frame_data.rtt_ms = 150;
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kSelectiveErrors, frame_data)));
   EXPECT_FALSE(session_.decodable());
 
   packet_.seqNum -= 1;
   FillPacket(0);
   packet_.isFirstPacket = true;
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kSelectiveErrors, frame_data)));
   EXPECT_TRUE(session_.decodable());
 
   packet_.isFirstPacket = false;
@@ -256,19 +248,17 @@ TEST_F(TestSessionInfo, SelectiveDecodableState) {
   for (int i = 2; i < 8; ++i) {
     packet_.seqNum += 1;
     FillPacket(i);
-  EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+    EXPECT_EQ(packet_buffer_size(),
+              static_cast<size_t>(session_.InsertPacket(
+                  packet_, frame_buffer_, kSelectiveErrors, frame_data)));
     EXPECT_TRUE(session_.decodable());
   }
 
   packet_.seqNum += 1;
   FillPacket(8);
   EXPECT_EQ(packet_buffer_size(),
-            static_cast<size_t>(session_.InsertPacket(packet_, frame_buffer_,
-                                                      kSelectiveErrors,
-                                                      frame_data)));
+            static_cast<size_t>(session_.InsertPacket(
+                packet_, frame_buffer_, kSelectiveErrors, frame_data)));
   EXPECT_TRUE(session_.decodable());
 }
 
@@ -285,18 +275,14 @@ TEST_F(TestSessionInfo, OutOfBoundsPackets1PacketFrame) {
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0000;
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, SetMarkerBitOnce) {
@@ -311,10 +297,8 @@ TEST_F(TestSessionInfo, SetMarkerBitOnce) {
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
@@ -331,10 +315,8 @@ TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0006;
   packet_.isFirstPacket = true;
   packet_.markerBit = true;
@@ -346,10 +328,8 @@ TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
   packet_.isFirstPacket = false;
   packet_.markerBit = true;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, OutOfBoundsPacketsWrap) {
@@ -379,20 +359,14 @@ TEST_F(TestSessionInfo, OutOfBoundsPacketsWrap) {
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3,
-            session_.InsertPacket(packet_,
-                                  frame_buffer_,
-                                  kNoErrors,
-                                  frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0006;
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3,
-            session_.InsertPacket(packet_,
-                                  frame_buffer_,
-                                  kNoErrors,
-                                  frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
@@ -417,10 +391,8 @@ TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
   packet_.seqNum = 0x0010;
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
@@ -440,10 +412,8 @@ TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
   packet_.isFirstPacket = false;
   packet_.markerBit = false;
   FillPacket(1);
-  EXPECT_EQ(-3, session_.InsertPacket(packet_,
-                                      frame_buffer_,
-                                      kNoErrors,
-                                      frame_data));
+  EXPECT_EQ(
+      -3, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 }
 
 TEST_F(TestVP8Partitions, TwoPartitionsOneLoss) {
@@ -455,8 +425,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsOneLoss) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -505,8 +475,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsOneLoss2) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 1;
   FillPacket(1);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -567,8 +537,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsNoLossWrap) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0xfffd;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -629,8 +599,8 @@ TEST_F(TestVP8Partitions, TwoPartitionsLossWrap) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0xfffd;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -682,7 +652,6 @@ TEST_F(TestVP8Partitions, TwoPartitionsLossWrap) {
   EXPECT_TRUE(VerifyPartition(1, 1, 2));
 }
 
-
 TEST_F(TestVP8Partitions, ThreePartitionsOneMissing) {
   // Partition 1  |Partition 2    | Partition 3
   // [ 1 ] [ 2 ]  |               | [ 5 ] | [ 6 ]
@@ -692,8 +661,8 @@ TEST_F(TestVP8Partitions, ThreePartitionsOneMissing) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 1;
   FillPacket(1);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -754,8 +723,8 @@ TEST_F(TestVP8Partitions, ThreePartitionsLossInSecond) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 1;
   FillPacket(1);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -767,8 +736,7 @@ TEST_F(TestVP8Partitions, ThreePartitionsLossInSecond) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber += 1;
   FillPacket(2);
-  packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                         packet_header_);
+  packet = new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -841,8 +809,8 @@ TEST_F(TestVP8Partitions, AggregationOverTwoPackets) {
   packet_header_.header.markerBit = false;
   packet_header_.header.sequenceNumber = 0;
   FillPacket(0);
-  VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(),
-                                    packet_header_);
+  VCMPacket* packet =
+      new VCMPacket(packet_buffer_, packet_buffer_size(), packet_header_);
   EXPECT_EQ(packet_buffer_size(),
             static_cast<size_t>(session_.InsertPacket(*packet, frame_buffer_,
                                                       kNoErrors, frame_data)));
@@ -888,14 +856,12 @@ TEST_F(TestVP8Partitions, AggregationOverTwoPackets) {
 TEST_F(TestNalUnits, OnlyReceivedEmptyPacket) {
   packet_.isFirstPacket = false;
   packet_.completeNALU = kNaluComplete;
-  packet_.frameType = kFrameEmpty;
+  packet_.frameType = kEmptyFrame;
   packet_.sizeBytes = 0;
   packet_.seqNum = 0;
   packet_.markerBit = false;
-  EXPECT_EQ(0, session_.InsertPacket(packet_,
-                                     frame_buffer_,
-                                     kNoErrors,
-                                     frame_data));
+  EXPECT_EQ(
+      0, session_.InsertPacket(packet_, frame_buffer_, kNoErrors, frame_data));
 
   EXPECT_EQ(0U, session_.MakeDecodable());
   EXPECT_EQ(0U, session_.SessionLength());
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotJitterEstimate.m b/media/webrtc/trunk/webrtc/modules/video_coding/test/plotJitterEstimate.m
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotJitterEstimate.m
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/plotJitterEstimate.m
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotReceiveTrace.m b/media/webrtc/trunk/webrtc/modules/video_coding/test/plotReceiveTrace.m
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotReceiveTrace.m
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/plotReceiveTrace.m
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotTimingTest.m b/media/webrtc/trunk/webrtc/modules/video_coding/test/plotTimingTest.m
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/plotTimingTest.m
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/plotTimingTest.m
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/receiver_tests.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/receiver_tests.h
similarity index 70%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/receiver_tests.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/receiver_tests.h
index 6d7b7beeb5..d6bac07392 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/receiver_tests.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/receiver_tests.h
@@ -11,20 +11,20 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
 #define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
 
-#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/modules/video_coding/main/test/video_source.h"
-#include "webrtc/typedefs.h"
-
 #include <stdio.h>
 #include <string>
 
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/modules/video_coding/test/video_source.h"
+#include "webrtc/typedefs.h"
+
 class RtpDataCallback : public webrtc::NullRtpData {
  public:
-  RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
+  explicit RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
   virtual ~RtpDataCallback() {}
 
   int32_t OnReceivedPayloadData(
@@ -40,4 +40,4 @@ class RtpDataCallback : public webrtc::NullRtpData {
 
 int RtpPlay(const CmdArgs& args);
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/release_test.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/release_test.h
similarity index 72%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/release_test.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/release_test.h
index 25781602c1..ab9b2159d9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/release_test.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/release_test.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef RELEASE_TEST_H
-#define RELEASE_TEST_H
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
 
 int ReleaseTest();
 int ReleaseTestPart2();
 
-#endif
\ No newline at end of file
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.cc b/media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.cc
similarity index 86%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.cc
index c7a2f660d7..9b6490618c 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.cc
@@ -8,27 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/rtp_player.h"
+#include "webrtc/modules/video_coding/test/rtp_player.h"
 
 #include <stdio.h>
 
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/test/rtp_file_reader.h"
 
 #if 1
-# define DEBUG_LOG1(text, arg)
+#define DEBUG_LOG1(text, arg)
 #else
-# define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
+#define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
 #endif
 
 namespace webrtc {
@@ -41,7 +41,9 @@ enum {
 
 class RawRtpPacket {
  public:
-  RawRtpPacket(const uint8_t* data, size_t length, uint32_t ssrc,
+  RawRtpPacket(const uint8_t* data,
+               size_t length,
+               uint32_t ssrc,
                uint16_t seq_num)
       : data_(new uint8_t[length]),
         length_(length),
@@ -66,7 +68,7 @@ class RawRtpPacket {
   uint32_t ssrc_;
   uint16_t seq_num_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RawRtpPacket);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RawRtpPacket);
 };
 
 class LostPackets {
@@ -140,7 +142,7 @@ class LostPackets {
     CriticalSectionScoped cs(crit_sect_.get());
     int count = 0;
     for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
-        ++it) {
+         ++it) {
       if ((*it)->resend_time_ms() >= 0) {
         count++;
       }
@@ -164,7 +166,7 @@ class LostPackets {
     printf("Packets still lost: %zd\n", packets_.size());
     printf("Sequence numbers:\n");
     for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
-        ++it) {
+         ++it) {
       printf("%u, ", (*it)->seq_num());
     }
     printf("\n");
@@ -182,7 +184,7 @@ class LostPackets {
   Clock* clock_;
   int64_t rtt_ms_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(LostPackets);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LostPackets);
 };
 
 class SsrcHandlers {
@@ -217,11 +219,10 @@ class SsrcHandlers {
 
     RtpRtcp::Configuration configuration;
     configuration.clock = clock;
-    configuration.id = 1;
     configuration.audio = false;
     handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver(
-        configuration.id, configuration.clock, handler->payload_sink_.get(),
-        NULL, handler->rtp_payload_registry_.get()));
+        configuration.clock, handler->payload_sink_.get(), NULL,
+        handler->rtp_payload_registry_.get()));
     if (handler->rtp_module_.get() == NULL) {
       return -1;
     }
@@ -232,17 +233,14 @@ class SsrcHandlers {
         kDefaultTransmissionTimeOffsetExtensionId);
 
     for (PayloadTypesIterator it = payload_types_.begin();
-        it != payload_types_.end(); ++it) {
+         it != payload_types_.end(); ++it) {
       VideoCodec codec;
       memset(&codec, 0, sizeof(codec));
-      strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName)-1);
+      strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName) - 1);
       codec.plType = it->payload_type();
       codec.codecType = it->codec_type();
-      if (handler->rtp_module_->RegisterReceivePayload(codec.plName,
-                                                       codec.plType,
-                                                       90000,
-                                                       0,
-                                                       codec.maxBitrate) < 0) {
+      if (handler->rtp_module_->RegisterReceivePayload(
+              codec.plName, codec.plType, 90000, 0, codec.maxBitrate) < 0) {
         return -1;
       }
     }
@@ -268,7 +266,8 @@ class SsrcHandlers {
  private:
   class Handler : public RtpStreamInterface {
    public:
-    Handler(uint32_t ssrc, const PayloadTypes& payload_types,
+    Handler(uint32_t ssrc,
+            const PayloadTypes& payload_types,
             LostPackets* lost_packets)
         : rtp_header_parser_(RtpHeaderParser::Create()),
           rtp_payload_registry_(new RTPPayloadRegistry(
@@ -291,9 +290,7 @@ class SsrcHandlers {
     }
 
     virtual uint32_t ssrc() const { return ssrc_; }
-    virtual const PayloadTypes& payload_types() const {
-      return payload_types_;
-    }
+    virtual const PayloadTypes& payload_types() const { return payload_types_; }
 
     rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
@@ -305,7 +302,7 @@ class SsrcHandlers {
     const PayloadTypes& payload_types_;
     LostPackets* lost_packets_;
 
-    DISALLOW_COPY_AND_ASSIGN(Handler);
+    RTC_DISALLOW_COPY_AND_ASSIGN(Handler);
   };
 
   typedef std::map<uint32_t, Handler*> HandlerMap;
@@ -315,7 +312,7 @@ class SsrcHandlers {
   PayloadTypes payload_types_;
   HandlerMap handlers_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(SsrcHandlers);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SsrcHandlers);
 };
 
 class RtpPlayerImpl : public RtpPlayerInterface {
@@ -352,8 +349,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
   virtual int NextPacket(int64_t time_now) {
     // Send any packets ready to be resent.
     for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
-         packet != NULL;
-         packet = lost_packets_.NextPacketToResend(time_now)) {
+         packet != NULL; packet = lost_packets_.NextPacketToResend(time_now)) {
       int ret = SendPacket(packet->data(), packet->length());
       if (ret > 0) {
         printf("Resend: %08x:%u\n", packet->ssrc(), packet->seq_num());
@@ -393,8 +389,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
       if (!packet_source_->NextPacket(&next_packet_)) {
         end_of_file_ = true;
         return 0;
-      }
-      else if (next_packet_.length == 0) {
+      } else if (next_packet_.length == 0) {
         return 0;
       }
     }
@@ -407,7 +402,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
   virtual uint32_t TimeUntilNextPacket() const {
     int64_t time_left = (next_rtp_time_ - first_packet_rtp_time_) -
-        (clock_->TimeInMilliseconds() - first_packet_time_ms_);
+                        (clock_->TimeInMilliseconds() - first_packet_time_ms_);
     if (time_left < 0) {
       return 0;
     }
@@ -439,7 +434,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
       if (no_loss_startup_ > 0) {
         no_loss_startup_--;
-      } else if ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate_) {
+      } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) {  // NOLINT
         uint16_t seq_num = header.sequenceNumber;
         lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
         DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
@@ -467,13 +462,16 @@ class RtpPlayerImpl : public RtpPlayerInterface {
   bool reordering_;
   rtc::scoped_ptr<RawRtpPacket> reorder_buffer_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPlayerImpl);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPlayerImpl);
 };
 
 RtpPlayerInterface* Create(const std::string& input_filename,
-    PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
-    const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
-    bool reordering) {
+                           PayloadSinkFactoryInterface* payload_sink_factory,
+                           Clock* clock,
+                           const PayloadTypes& payload_types,
+                           float loss_rate,
+                           int64_t rtt_ms,
+                           bool reordering) {
   rtc::scoped_ptr<test::RtpFileReader> packet_source(
       test::RtpFileReader::Create(test::RtpFileReader::kRtpDump,
                                   input_filename));
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.h
similarity index 80%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.h
index 7459231416..e50fb9ac70 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/rtp_player.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/rtp_player.h
@@ -14,8 +14,8 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 class Clock;
@@ -24,12 +24,12 @@ namespace rtpplayer {
 
 class PayloadCodecTuple {
  public:
-  PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
+  PayloadCodecTuple(uint8_t payload_type,
+                    const std::string& codec_name,
                     VideoCodecType codec_type)
       : name_(codec_name),
         payload_type_(payload_type),
-        codec_type_(codec_type) {
-  }
+        codec_type_(codec_type) {}
 
   const std::string& name() const { return name_; }
   uint8_t payload_type() const { return payload_type_; }
@@ -87,11 +87,14 @@ class RtpPlayerInterface {
 };
 
 RtpPlayerInterface* Create(const std::string& inputFilename,
-    PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
-    const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
-    bool reordering);
+                           PayloadSinkFactoryInterface* payloadSinkFactory,
+                           Clock* clock,
+                           const PayloadTypes& payload_types,
+                           float lossRate,
+                           int64_t rttMs,
+                           bool reordering);
 
 }  // namespace rtpplayer
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.cc b/media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.cc
similarity index 69%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.cc
index 4f85dffc50..167d55faff 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.cc
@@ -8,53 +8,45 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/test/stream_generator.h"
+#include "webrtc/modules/video_coding/test/stream_generator.h"
 
 #include <string.h>
 
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
-StreamGenerator::StreamGenerator(uint16_t start_seq_num,
-                                 uint32_t start_timestamp,
-                                 int64_t current_time)
-    : packets_(),
-      sequence_number_(start_seq_num),
-      timestamp_(start_timestamp),
-      start_time_(current_time) {}
+StreamGenerator::StreamGenerator(uint16_t start_seq_num, int64_t current_time)
+    : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
 
-void StreamGenerator::Init(uint16_t start_seq_num,
-                           uint32_t start_timestamp,
-                           int64_t current_time) {
+void StreamGenerator::Init(uint16_t start_seq_num, int64_t current_time) {
   packets_.clear();
   sequence_number_ = start_seq_num;
-  timestamp_ = start_timestamp;
   start_time_ = current_time;
-  memset(&packet_buffer, 0, sizeof(packet_buffer));
+  memset(packet_buffer_, 0, sizeof(packet_buffer_));
 }
 
 void StreamGenerator::GenerateFrame(FrameType type,
                                     int num_media_packets,
                                     int num_empty_packets,
-                                    int64_t current_time) {
-  timestamp_ = 90 * (current_time - start_time_);
+                                    int64_t time_ms) {
+  uint32_t timestamp = 90 * (time_ms - start_time_);
   for (int i = 0; i < num_media_packets; ++i) {
     const int packet_size =
         (kFrameSize + num_media_packets / 2) / num_media_packets;
     bool marker_bit = (i == num_media_packets - 1);
-    packets_.push_back(GeneratePacket(
-        sequence_number_, timestamp_, packet_size, (i == 0), marker_bit, type));
+    packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
+                                      (i == 0), marker_bit, type));
     ++sequence_number_;
   }
   for (int i = 0; i < num_empty_packets; ++i) {
-    packets_.push_back(GeneratePacket(
-        sequence_number_, timestamp_, 0, false, false, kFrameEmpty));
+    packets_.push_back(GeneratePacket(sequence_number_, timestamp, 0, false,
+                                      false, kEmptyFrame));
     ++sequence_number_;
   }
 }
@@ -73,7 +65,7 @@ VCMPacket StreamGenerator::GeneratePacket(uint16_t sequence_number,
   packet.isFirstPacket = first_packet;
   packet.markerBit = marker_bit;
   packet.sizeBytes = size;
-  packet.dataPtr = packet_buffer;
+  packet.dataPtr = packet_buffer_;
   if (packet.isFirstPacket)
     packet.completeNALU = kNaluStart;
   else if (packet.markerBit)
@@ -111,7 +103,9 @@ bool StreamGenerator::NextPacket(VCMPacket* packet) {
   return true;
 }
 
-void StreamGenerator::DropLastPacket() { packets_.pop_back(); }
+void StreamGenerator::DropLastPacket() {
+  packets_.pop_back();
+}
 
 uint16_t StreamGenerator::NextSequenceNumber() const {
   if (packets_.empty())
@@ -119,7 +113,9 @@ uint16_t StreamGenerator::NextSequenceNumber() const {
   return packets_.front().seqNum;
 }
 
-int StreamGenerator::PacketsRemaining() const { return packets_.size(); }
+int StreamGenerator::PacketsRemaining() const {
+  return packets_.size();
+}
 
 std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
   std::list<VCMPacket>::iterator it = packets_.begin();
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.h
similarity index 67%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.h
index 6565527db6..36b26db92e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/stream_generator.h
@@ -7,15 +7,14 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
-
-#include <string.h>
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
 
 #include <list>
 
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 
@@ -28,24 +27,16 @@ const int kDefaultFramePeriodMs = 1000 / kDefaultFrameRate;
 
 class StreamGenerator {
  public:
-  StreamGenerator(uint16_t start_seq_num,
-                  uint32_t start_timestamp,
-                  int64_t current_time);
-  void Init(uint16_t start_seq_num,
-            uint32_t start_timestamp,
-            int64_t current_time);
+  StreamGenerator(uint16_t start_seq_num, int64_t current_time);
+  void Init(uint16_t start_seq_num, int64_t current_time);
 
+  // |time_ms| denotes the timestamp you want to put on the frame, and the unit
+  // is millisecond. GenerateFrame will translate |time_ms| into a 90kHz
+  // timestamp and put it on the frame.
   void GenerateFrame(FrameType type,
                      int num_media_packets,
                      int num_empty_packets,
-                     int64_t current_time);
-
-  VCMPacket GeneratePacket(uint16_t sequence_number,
-                           uint32_t timestamp,
-                           unsigned int size,
-                           bool first_packet,
-                           bool marker_bit,
-                           FrameType type);
+                     int64_t time_ms);
 
   bool PopPacket(VCMPacket* packet, int index);
   void DropLastPacket();
@@ -59,17 +50,23 @@ class StreamGenerator {
   int PacketsRemaining() const;
 
  private:
+  VCMPacket GeneratePacket(uint16_t sequence_number,
+                           uint32_t timestamp,
+                           unsigned int size,
+                           bool first_packet,
+                           bool marker_bit,
+                           FrameType type);
+
   std::list<VCMPacket>::iterator GetPacketIterator(int index);
 
   std::list<VCMPacket> packets_;
   uint16_t sequence_number_;
-  uint32_t timestamp_;
   int64_t start_time_;
-  uint8_t packet_buffer[kMaxPacketSize];
+  uint8_t packet_buffer_[kMaxPacketSize];
 
-  DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TEST_STREAM_GENERATOR_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/subfigure.m b/media/webrtc/trunk/webrtc/modules/video_coding/test/subfigure.m
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/subfigure.m
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/subfigure.m
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.cc b/media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.cc
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.cc
index 6dad3773c1..7ff663e395 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/test_util.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
 
 #include <assert.h>
 #include <math.h>
@@ -17,7 +17,7 @@
 #include <sstream>
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 CmdArgs::CmdArgs()
@@ -28,12 +28,12 @@ CmdArgs::CmdArgs()
       rtt(0),
       inputFile(webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv"),
       outputFile(webrtc::test::OutputPath() +
-          "video_coding_test_output_352x288.yuv") {
-}
+                 "video_coding_test_output_352x288.yuv") {}
 
 namespace {
 
-void SplitFilename(const std::string& filename, std::string* basename,
+void SplitFilename(const std::string& filename,
+                   std::string* basename,
                    std::string* extension) {
   assert(basename);
   assert(extension);
@@ -41,7 +41,7 @@ void SplitFilename(const std::string& filename, std::string* basename,
   std::string::size_type idx;
   idx = filename.rfind('.');
 
-  if(idx != std::string::npos) {
+  if (idx != std::string::npos) {
     *basename = filename.substr(0, idx);
     *extension = filename.substr(idx + 1);
   } else {
@@ -50,21 +50,24 @@ void SplitFilename(const std::string& filename, std::string* basename,
   }
 }
 
-std::string AppendWidthHeightCount(const std::string& filename, int width,
-                                   int height, int count) {
+std::string AppendWidthHeightCount(const std::string& filename,
+                                   int width,
+                                   int height,
+                                   int count) {
   std::string basename;
   std::string extension;
   SplitFilename(filename, &basename, &extension);
   std::stringstream ss;
-  ss << basename << "_" << count << "." <<  width << "_" << height << "." <<
-      extension;
+  ss << basename << "_" << count << "." << width << "_" << height << "."
+     << extension;
   return ss.str();
 }
 
 }  // namespace
 
 FileOutputFrameReceiver::FileOutputFrameReceiver(
-    const std::string& base_out_filename, uint32_t ssrc)
+    const std::string& base_out_filename,
+    uint32_t ssrc)
     : out_filename_(),
       out_file_(NULL),
       timing_file_(NULL),
@@ -73,15 +76,15 @@ FileOutputFrameReceiver::FileOutputFrameReceiver(
       count_(0) {
   std::string basename;
   std::string extension;
-  if (base_out_filename == "") {
+  if (base_out_filename.empty()) {
     basename = webrtc::test::OutputPath() + "rtp_decoded";
     extension = "yuv";
   } else {
     SplitFilename(base_out_filename, &basename, &extension);
   }
   std::stringstream ss;
-  ss << basename << "_" << std::hex << std::setw(8) << std::setfill('0') <<
-      ssrc << "." << extension;
+  ss << basename << "_" << std::hex << std::setw(8) << std::setfill('0') << ssrc
+     << "." << extension;
   out_filename_ = ss.str();
 }
 
@@ -95,7 +98,7 @@ FileOutputFrameReceiver::~FileOutputFrameReceiver() {
 }
 
 int32_t FileOutputFrameReceiver::FrameToRender(
-    webrtc::I420VideoFrame& video_frame) {
+    webrtc::VideoFrame& video_frame) {
   if (timing_file_ == NULL) {
     std::string basename;
     std::string extension;
@@ -113,8 +116,8 @@ int32_t FileOutputFrameReceiver::FrameToRender(
     printf("New size: %dx%d\n", video_frame.width(), video_frame.height());
     width_ = video_frame.width();
     height_ = video_frame.height();
-    std::string filename_with_width_height = AppendWidthHeightCount(
-        out_filename_, width_, height_, count_);
+    std::string filename_with_width_height =
+        AppendWidthHeightCount(out_filename_, width_, height_, count_);
     ++count_;
     out_file_ = fopen(filename_with_width_height.c_str(), "wb");
     if (out_file_ == NULL) {
@@ -122,15 +125,15 @@ int32_t FileOutputFrameReceiver::FrameToRender(
     }
   }
   fprintf(timing_file_, "%u, %u\n", video_frame.timestamp(),
-      webrtc::MaskWord64ToUWord32(video_frame.render_time_ms()));
-  if (PrintI420VideoFrame(video_frame, out_file_) < 0) {
+          webrtc::MaskWord64ToUWord32(video_frame.render_time_ms()));
+  if (PrintVideoFrame(video_frame, out_file_) < 0) {
     return -1;
   }
   return 0;
 }
 
 webrtc::RtpVideoCodecTypes ConvertCodecType(const char* plname) {
-  if (strncmp(plname,"VP8" , 3) == 0) {
+  if (strncmp(plname, "VP8", 3) == 0) {
     return webrtc::kRtpVideoVp8;
   } else {
     // Default value.
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.h
similarity index 75%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.h
index b484353289..45b88b9b50 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/test_util.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/test_util.h
@@ -18,9 +18,9 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 
 enum { kMaxNackListSize = 250 };
 enum { kMaxPacketAgeToNack = 450 };
@@ -33,11 +33,13 @@ class NullEvent : public webrtc::EventWrapper {
 
   virtual bool Reset() { return true; }
 
-  virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {
+  virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {  // NOLINT
     return webrtc::kEventTimeout;
   }
 
-  virtual bool StartTimer(bool periodic, unsigned long time) { return true; }
+  virtual bool StartTimer(bool periodic, unsigned long time) {  // NOLINT
+    return true;
+  }
 
   virtual bool StopTimer() { return true; }
 };
@@ -46,9 +48,7 @@ class NullEventFactory : public webrtc::EventFactory {
  public:
   virtual ~NullEventFactory() {}
 
-  virtual webrtc::EventWrapper* CreateEvent() {
-    return new NullEvent;
-  }
+  virtual webrtc::EventWrapper* CreateEvent() { return new NullEvent; }
 };
 
 class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
@@ -57,18 +57,17 @@ class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
   virtual ~FileOutputFrameReceiver();
 
   // VCMReceiveCallback
-  virtual int32_t FrameToRender(webrtc::I420VideoFrame& video_frame);
+  virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame);  // NOLINT
 
  private:
   std::string out_filename_;
-  uint32_t ssrc_;
   FILE* out_file_;
   FILE* timing_file_;
   int width_;
   int height_;
   int count_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(FileOutputFrameReceiver);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FileOutputFrameReceiver);
 };
 
 class CmdArgs {
@@ -84,4 +83,4 @@ class CmdArgs {
   std::string outputFile;
 };
 
-#endif
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/tester_main.cc b/media/webrtc/trunk/webrtc/modules/video_coding/test/tester_main.cc
similarity index 54%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/tester_main.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/tester_main.cc
index 2885f00bd5..33ca82007d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/tester_main.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/tester_main.cc
@@ -8,25 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
 #include <stdlib.h>
 #include <string.h>
 
 #include "gflags/gflags.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/test/receiver_tests.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/test/receiver_tests.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 DEFINE_string(codec, "VP8", "Codec to use (VP8 or I420).");
 DEFINE_int32(width, 352, "Width in pixels of the frames in the input file.");
 DEFINE_int32(height, 288, "Height in pixels of the frames in the input file.");
 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
-DEFINE_string(input_filename, webrtc::test::ProjectRootPath() +
-              "/resources/foreman_cif.yuv", "Input file.");
-DEFINE_string(output_filename, webrtc::test::OutputPath() +
-              "video_coding_test_output_352x288.yuv", "Output file.");
+DEFINE_string(input_filename,
+              webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv",
+              "Input file.");
+DEFINE_string(output_filename,
+              webrtc::test::OutputPath() +
+                  "video_coding_test_output_352x288.yuv",
+              "Output file.");
 
-using namespace webrtc;
+namespace webrtc {
 
 /*
  * Build with EVENT_DEBUG defined
@@ -36,36 +38,37 @@ using namespace webrtc;
 int vcmMacrosTests = 0;
 int vcmMacrosErrors = 0;
 
-int ParseArguments(CmdArgs& args) {
-  args.width = FLAGS_width;
-  args.height = FLAGS_height;
-  if (args.width < 1 || args.height < 1) {
+int ParseArguments(CmdArgs* args) {
+  args->width = FLAGS_width;
+  args->height = FLAGS_height;
+  if (args->width < 1 || args->height < 1) {
     return -1;
   }
-  args.codecName = FLAGS_codec;
-  if (args.codecName == "VP8") {
-    args.codecType = kVideoCodecVP8;
-  } else if (args.codecName == "VP9") {
-    args.codecType = kVideoCodecVP9;
-  } else if (args.codecName == "I420") {
-    args.codecType = kVideoCodecI420;
+  args->codecName = FLAGS_codec;
+  if (args->codecName == "VP8") {
+    args->codecType = kVideoCodecVP8;
+  } else if (args->codecName == "VP9") {
+    args->codecType = kVideoCodecVP9;
+  } else if (args->codecName == "I420") {
+    args->codecType = kVideoCodecI420;
   } else {
-    printf("Invalid codec: %s\n", args.codecName.c_str());
+    printf("Invalid codec: %s\n", args->codecName.c_str());
     return -1;
   }
-  args.inputFile = FLAGS_input_filename;
-  args.outputFile = FLAGS_output_filename;
-  args.rtt = FLAGS_rtt;
+  args->inputFile = FLAGS_input_filename;
+  args->outputFile = FLAGS_output_filename;
+  args->rtt = FLAGS_rtt;
   return 0;
 }
+}  // namespace webrtc
 
-int main(int argc, char **argv) {
+int main(int argc, char** argv) {
   // Initialize WebRTC fileutils.h so paths to resources can be resolved.
   webrtc::test::SetExecutablePath(argv[0]);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   CmdArgs args;
-  if (ParseArguments(args) != 0) {
+  if (webrtc::ParseArguments(&args) != 0) {
     printf("Unable to parse input arguments\n");
     return -1;
   }
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc b/media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
index e8dbe8df01..c9ec372f41 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
@@ -8,23 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h"
+#include "webrtc/modules/video_coding/test/vcm_payload_sink_factory.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 namespace rtpplayer {
 
-class VcmPayloadSinkFactory::VcmPayloadSink
-    : public PayloadSinkInterface,
-      public VCMPacketRequestCallback {
+class VcmPayloadSinkFactory::VcmPayloadSink : public PayloadSinkInterface,
+                                              public VCMPacketRequestCallback {
  public:
   VcmPayloadSink(VcmPayloadSinkFactory* factory,
                  RtpStreamInterface* stream,
@@ -43,9 +42,7 @@ class VcmPayloadSinkFactory::VcmPayloadSink
     vcm_->RegisterReceiveCallback(frame_receiver_.get());
   }
 
-  virtual ~VcmPayloadSink() {
-    factory_->Remove(this);
-  }
+  virtual ~VcmPayloadSink() { factory_->Remove(this); }
 
   // PayloadSinkInterface
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
@@ -95,7 +92,7 @@ class VcmPayloadSinkFactory::VcmPayloadSink
   rtc::scoped_ptr<VideoCodingModule> vcm_;
   rtc::scoped_ptr<FileOutputFrameReceiver> frame_receiver_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(VcmPayloadSink);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VcmPayloadSink);
 };
 
 VcmPayloadSinkFactory::VcmPayloadSinkFactory(
@@ -134,19 +131,13 @@ PayloadSinkInterface* VcmPayloadSinkFactory::Create(
   if (vcm.get() == NULL) {
     return NULL;
   }
-  if (vcm->InitializeReceiver() < 0) {
-    return NULL;
-  }
 
   const PayloadTypes& plt = stream->payload_types();
-  for (PayloadTypesIterator it = plt.begin(); it != plt.end();
-      ++it) {
+  for (PayloadTypesIterator it = plt.begin(); it != plt.end(); ++it) {
     if (it->codec_type() != kVideoCodecULPFEC &&
         it->codec_type() != kVideoCodecRED) {
       VideoCodec codec;
-      if (VideoCodingModule::Codec(it->codec_type(), &codec) < 0) {
-        return NULL;
-      }
+      VideoCodingModule::Codec(it->codec_type(), &codec);
       codec.plType = it->payload_type();
       if (vcm->RegisterReceiveCodec(&codec, 1) < 0) {
         return NULL;
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
index ca0ed56054..dae53b0c08 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/vcm_payload_sink_factory.h
@@ -8,13 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VCM_PAYLOAD_SINK_FACTORY_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_VCM_PAYLOAD_SINK_FACTORY_H_
+
 #include <string>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/test/rtp_player.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/test/rtp_player.h"
 
 class NullEventFactory;
 
@@ -26,9 +29,11 @@ namespace rtpplayer {
 class VcmPayloadSinkFactory : public PayloadSinkFactoryInterface {
  public:
   VcmPayloadSinkFactory(const std::string& base_out_filename,
-                        Clock* clock, bool protection_enabled,
+                        Clock* clock,
+                        bool protection_enabled,
                         VCMVideoProtection protection_method,
-                        int64_t rtt_ms, uint32_t render_delay_ms,
+                        int64_t rtt_ms,
+                        uint32_t render_delay_ms,
                         uint32_t min_playout_delay_ms);
   virtual ~VcmPayloadSinkFactory();
 
@@ -57,7 +62,9 @@ class VcmPayloadSinkFactory : public PayloadSinkFactoryInterface {
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   Sinks sinks_;
 
-  DISALLOW_IMPLICIT_CONSTRUCTORS(VcmPayloadSinkFactory);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(VcmPayloadSinkFactory);
 };
 }  // namespace rtpplayer
 }  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_VCM_PAYLOAD_SINK_FACTORY_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_rtp_play.cc b/media/webrtc/trunk/webrtc/modules/video_coding/test/video_rtp_play.cc
similarity index 74%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_rtp_play.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/test/video_rtp_play.cc
index 1cf27c78e2..cb092e381e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/test/video_rtp_play.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/video_rtp_play.cc
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/test/receiver_tests.h"
-#include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/video_coding/test/receiver_tests.h"
+#include "webrtc/modules/video_coding/test/vcm_payload_sink_factory.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace {
@@ -24,7 +24,9 @@ const int64_t kConfigRttMs = 0;
 const uint32_t kConfigRenderDelayMs = 0;
 const uint32_t kConfigMinPlayoutDelayMs = 0;
 const int64_t kConfigMaxRuntimeMs = -1;
-
+const uint8_t kDefaultUlpFecPayloadType = 97;
+const uint8_t kDefaultRedPayloadType = 96;
+const uint8_t kDefaultVp8PayloadType = 100;
 }  // namespace
 
 int RtpPlay(const CmdArgs& args) {
@@ -35,21 +37,20 @@ int RtpPlay(const CmdArgs& args) {
 
   webrtc::rtpplayer::PayloadTypes payload_types;
   payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(
-      VCM_ULPFEC_PAYLOAD_TYPE, "ULPFEC", webrtc::kVideoCodecULPFEC));
+      kDefaultUlpFecPayloadType, "ULPFEC", webrtc::kVideoCodecULPFEC));
   payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(
-      VCM_RED_PAYLOAD_TYPE, "RED", webrtc::kVideoCodecRED));
+      kDefaultRedPayloadType, "RED", webrtc::kVideoCodecRED));
   payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple(
-      VCM_VP8_PAYLOAD_TYPE, "VP8", webrtc::kVideoCodecVP8));
+      kDefaultVp8PayloadType, "VP8", webrtc::kVideoCodecVP8));
 
   std::string output_file = args.outputFile;
-  if (output_file == "") {
+  if (output_file.empty())
     output_file = webrtc::test::OutputPath() + "RtpPlay_decoded.yuv";
-  }
 
   webrtc::SimulatedClock clock(0);
-  webrtc::rtpplayer::VcmPayloadSinkFactory factory(output_file, &clock,
-      kConfigProtectionEnabled, kConfigProtectionMethod, kConfigRttMs,
-      kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
+  webrtc::rtpplayer::VcmPayloadSinkFactory factory(
+      output_file, &clock, kConfigProtectionEnabled, kConfigProtectionMethod,
+      kConfigRttMs, kConfigRenderDelayMs, kConfigMinPlayoutDelayMs);
   rtc::scoped_ptr<webrtc::rtpplayer::RtpPlayerInterface> rtp_player(
       webrtc::rtpplayer::Create(args.inputFile, &factory, &clock, payload_types,
                                 kConfigLossRate, kConfigRttMs,
@@ -62,7 +63,7 @@ int RtpPlay(const CmdArgs& args) {
   while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) {
     ret = factory.DecodeAndProcessAll(true);
     if (ret < 0 || (kConfigMaxRuntimeMs > -1 &&
-        clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
+                    clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) {
       break;
     }
     clock.AdvanceTimeMilliseconds(1);
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/test/video_source.h b/media/webrtc/trunk/webrtc/modules/video_coding/test/video_source.h
new file mode 100644
index 0000000000..19d7f50b26
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/test/video_source.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+
+#include <string>
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/typedefs.h"
+
+enum VideoSize {
+  kUndefined,
+  kSQCIF,    // 128*96       = 12 288
+  kQQVGA,    // 160*120      = 19 200
+  kQCIF,     // 176*144      = 25 344
+  kCGA,      // 320*200      = 64 000
+  kQVGA,     // 320*240      = 76 800
+  kSIF,      // 352*240      = 84 480
+  kWQVGA,    // 400*240      = 96 000
+  kCIF,      // 352*288      = 101 376
+  kW288p,    // 512*288      = 147 456 (WCIF)
+  k448p,     // 576*448      = 281 088
+  kVGA,      // 640*480      = 307 200
+  k432p,     // 720*432      = 311 040
+  kW432p,    // 768*432      = 331 776
+  k4SIF,     // 704*480      = 337 920
+  kW448p,    // 768*448      = 344 064
+  kNTSC,     // 720*480      = 345 600
+  kFW448p,   // 800*448      = 358 400
+  kWVGA,     // 800*480      = 384 000
+  k4CIF,     // 704*576      = 405 504
+  kSVGA,     // 800*600      = 480 000
+  kW544p,    // 960*544      = 522 240
+  kW576p,    // 1024*576     = 589 824 (W4CIF)
+  kHD,       // 960*720      = 691 200
+  kXGA,      // 1024*768     = 786 432
+  kWHD,      // 1280*720     = 921 600
+  kFullHD,   // 1440*1080    = 1 555 200
+  kWFullHD,  // 1920*1080    = 2 073 600
+
+  kNumberOfVideoSizes
+};
+
+class VideoSource {
+ public:
+  VideoSource();
+  VideoSource(std::string fileName,
+              VideoSize size,
+              float frameRate,
+              webrtc::VideoType type = webrtc::kI420);
+  VideoSource(std::string fileName,
+              uint16_t width,
+              uint16_t height,
+              float frameRate = 30,
+              webrtc::VideoType type = webrtc::kI420);
+
+  std::string GetFileName() const { return _fileName; }
+  uint16_t GetWidth() const { return _width; }
+  uint16_t GetHeight() const { return _height; }
+  webrtc::VideoType GetType() const { return _type; }
+  float GetFrameRate() const { return _frameRate; }
+  int GetWidthHeight(VideoSize size);
+
+  // Returns the filename with the path (including the leading slash) removed.
+  std::string GetName() const;
+
+  size_t GetFrameLength() const;
+
+ private:
+  std::string _fileName;
+  uint16_t _width;
+  uint16_t _height;
+  webrtc::VideoType _type;
+  float _frameRate;
+};
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/timestamp_map.cc b/media/webrtc/trunk/webrtc/modules/video_coding/timestamp_map.cc
new file mode 100644
index 0000000000..97d2777658
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/timestamp_map.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <stdlib.h>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/timestamp_map.h"
+
+namespace webrtc {
+
+VCMTimestampMap::VCMTimestampMap(size_t capacity)
+    : ring_buffer_(new TimestampDataTuple[capacity]),
+      capacity_(capacity),
+      next_add_idx_(0),
+      next_pop_idx_(0) {}
+
+VCMTimestampMap::~VCMTimestampMap() {}
+
+void VCMTimestampMap::Add(uint32_t timestamp, VCMFrameInformation* data) {
+  ring_buffer_[next_add_idx_].timestamp = timestamp;
+  ring_buffer_[next_add_idx_].data = data;
+  next_add_idx_ = (next_add_idx_ + 1) % capacity_;
+
+  if (next_add_idx_ == next_pop_idx_) {
+    // Circular list full; forget oldest entry.
+    next_pop_idx_ = (next_pop_idx_ + 1) % capacity_;
+  }
+}
+
+VCMFrameInformation* VCMTimestampMap::Pop(uint32_t timestamp) {
+  while (!IsEmpty()) {
+    if (ring_buffer_[next_pop_idx_].timestamp == timestamp) {
+      // Found start time for this timestamp.
+      VCMFrameInformation* data = ring_buffer_[next_pop_idx_].data;
+      ring_buffer_[next_pop_idx_].data = nullptr;
+      next_pop_idx_ = (next_pop_idx_ + 1) % capacity_;
+      return data;
+    } else if (IsNewerTimestamp(ring_buffer_[next_pop_idx_].timestamp,
+                                timestamp)) {
+      // The timestamp we are looking for is not in the list.
+      return nullptr;
+    }
+
+    // Not in this position, check next (and forget this position).
+    next_pop_idx_ = (next_pop_idx_ + 1) % capacity_;
+  }
+
+  // Could not find matching timestamp in list.
+  return nullptr;
+}
+
+bool VCMTimestampMap::IsEmpty() const {
+  return (next_add_idx_ == next_pop_idx_);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/timestamp_map.h b/media/webrtc/trunk/webrtc/modules/video_coding/timestamp_map.h
new file mode 100644
index 0000000000..435d05895c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/timestamp_map.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct VCMFrameInformation;
+
+class VCMTimestampMap {
+ public:
+  explicit VCMTimestampMap(size_t capacity);
+  ~VCMTimestampMap();
+
+  // Empty the map.
+  void Reset();
+
+  void Add(uint32_t timestamp, VCMFrameInformation* data);
+  VCMFrameInformation* Pop(uint32_t timestamp);
+
+ private:
+  struct TimestampDataTuple {
+    uint32_t timestamp;
+    VCMFrameInformation* data;
+  };
+  bool IsEmpty() const;
+
+  rtc::scoped_ptr<TimestampDataTuple[]> ring_buffer_;
+  const size_t capacity_;
+  size_t next_add_idx_;
+  size_t next_pop_idx_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.cc b/media/webrtc/trunk/webrtc/modules/video_coding/timing.cc
similarity index 79%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/timing.cc
index 0e2fddf089..08dc307524 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/timing.cc
@@ -8,19 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/main/source/timing.h"
+#include "webrtc/modules/video_coding/timing.h"
 
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
-#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
+#include <algorithm>
 
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
 
 namespace webrtc {
 
-VCMTiming::VCMTiming(Clock* clock,
-                     VCMTiming* master_timing)
+VCMTiming::VCMTiming(Clock* clock, VCMTiming* master_timing)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       clock_(clock),
       master_(false),
@@ -62,14 +62,16 @@ void VCMTiming::UpdateHistograms() const {
   if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
     return;
   }
-  RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond",
+  RTC_HISTOGRAM_COUNTS_SPARSE_100(
+      "WebRTC.Video.DecodedFramesPerSecond",
       static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+      "WebRTC.Video.DelayedFramesToRenderer",
       num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
   if (num_delayed_decoded_frames_ > 0) {
-    RTC_HISTOGRAM_COUNTS_1000(
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
         "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
-            sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
+        sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
   }
 }
 
@@ -118,8 +120,8 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
     // Not initialized, set current delay to target.
     current_delay_ms_ = target_delay_ms;
   } else if (target_delay_ms != current_delay_ms_) {
-    int64_t delay_diff_ms = static_cast<int64_t>(target_delay_ms) -
-        current_delay_ms_;
+    int64_t delay_diff_ms =
+        static_cast<int64_t>(target_delay_ms) - current_delay_ms_;
     // Never change the delay with more than 100 ms every second. If we're
     // changing the delay in too large steps we will get noticeable freezes. By
     // limiting the change we can increase the delay in smaller steps, which
@@ -128,11 +130,13 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
     int64_t max_change_ms = 0;
     if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
       // wrap
-      max_change_ms = kDelayMaxChangeMsPerS * (frame_timestamp +
-          (static_cast<int64_t>(1) << 32) - prev_frame_timestamp_) / 90000;
+      max_change_ms = kDelayMaxChangeMsPerS *
+                      (frame_timestamp + (static_cast<int64_t>(1) << 32) -
+                       prev_frame_timestamp_) /
+                      90000;
     } else {
       max_change_ms = kDelayMaxChangeMsPerS *
-          (frame_timestamp - prev_frame_timestamp_) / 90000;
+                      (frame_timestamp - prev_frame_timestamp_) / 90000;
     }
     if (max_change_ms <= 0) {
       // Any changes less than 1 ms are truncated and
@@ -153,7 +157,7 @@ void VCMTiming::UpdateCurrentDelay(int64_t render_time_ms,
   CriticalSectionScoped cs(crit_sect_);
   uint32_t target_delay_ms = TargetDelayInternal();
   int64_t delayed_ms = actual_decode_time_ms -
-      (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
+                       (render_time_ms - MaxDecodeTimeMs() - render_delay_ms_);
   if (delayed_ms < 0) {
     return;
   }
@@ -165,13 +169,13 @@ void VCMTiming::UpdateCurrentDelay(int64_t render_time_ms,
 }
 
 int32_t VCMTiming::StopDecodeTimer(uint32_t time_stamp,
-                                   int64_t start_time_ms,
+                                   int32_t decode_time_ms,
                                    int64_t now_ms,
                                    int64_t render_time_ms) {
   CriticalSectionScoped cs(crit_sect_);
-  int32_t time_diff_ms = codec_timer_.StopTimer(start_time_ms, now_ms);
-  assert(time_diff_ms >= 0);
-  last_decode_ms_ = time_diff_ms;
+  codec_timer_.MaxFilter(decode_time_ms, now_ms);
+  assert(decode_time_ms >= 0);
+  last_decode_ms_ = decode_time_ms;
 
   // Update stats.
   ++num_decoded_frames_;
@@ -191,8 +195,8 @@ void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) {
   ts_extrapolator_->Update(now_ms, time_stamp);
 }
 
-int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms)
-    const {
+int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp,
+                                int64_t now_ms) const {
   CriticalSectionScoped cs(crit_sect_);
   const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms);
   return render_time_ms;
@@ -201,7 +205,7 @@ int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms)
 int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
                                         int64_t now_ms) const {
   int64_t estimated_complete_time_ms =
-    ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
+      ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
   if (estimated_complete_time_ms == -1) {
     estimated_complete_time_ms = now_ms;
   }
@@ -212,19 +216,19 @@ int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
 }
 
 // Must be called from inside a critical section.
-int32_t VCMTiming::MaxDecodeTimeMs(FrameType frame_type /*= kVideoFrameDelta*/)
-    const {
+int32_t VCMTiming::MaxDecodeTimeMs(
+    FrameType frame_type /*= kVideoFrameDelta*/) const {
   const int32_t decode_time_ms = codec_timer_.RequiredDecodeTimeMs(frame_type);
   assert(decode_time_ms >= 0);
   return decode_time_ms;
 }
 
-uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms)
-    const {
+uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms,
+                                   int64_t now_ms) const {
   CriticalSectionScoped cs(crit_sect_);
 
-  const int64_t max_wait_time_ms = render_time_ms - now_ms -
-      MaxDecodeTimeMs() - render_delay_ms_;
+  const int64_t max_wait_time_ms =
+      render_time_ms - now_ms - MaxDecodeTimeMs() - render_delay_ms_;
 
   if (max_wait_time_ms < 0) {
     return 0;
@@ -232,8 +236,8 @@ uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms, int64_t now_ms)
   return static_cast<uint32_t>(max_wait_time_ms);
 }
 
-bool VCMTiming::EnoughTimeToDecode(uint32_t available_processing_time_ms)
-    const {
+bool VCMTiming::EnoughTimeToDecode(
+    uint32_t available_processing_time_ms) const {
   CriticalSectionScoped cs(crit_sect_);
   int32_t max_decode_time_ms = MaxDecodeTimeMs();
   if (max_decode_time_ms < 0) {
@@ -246,7 +250,8 @@ bool VCMTiming::EnoughTimeToDecode(uint32_t available_processing_time_ms)
     max_decode_time_ms = 1;
   }
   return static_cast<int32_t>(available_processing_time_ms) -
-      max_decode_time_ms > 0;
+             max_decode_time_ms >
+         0;
 }
 
 uint32_t VCMTiming::TargetVideoDelay() const {
@@ -256,7 +261,7 @@ uint32_t VCMTiming::TargetVideoDelay() const {
 
 uint32_t VCMTiming::TargetDelayInternal() const {
   return std::max(min_playout_delay_ms_,
-      jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
+                  jitter_delay_ms_ + MaxDecodeTimeMs() + render_delay_ms_);
 }
 
 void VCMTiming::GetTimings(int* decode_ms,
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.h b/media/webrtc/trunk/webrtc/modules/video_coding/timing.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/timing.h
index 61c0273010..a4d0cf4543 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/timing.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/video_coding/main/source/codec_timer.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/video_coding/codec_timer.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -25,8 +25,7 @@ class VCMTiming {
  public:
   // The primary timing component should be passed
   // if this is the dual timing component.
-  VCMTiming(Clock* clock,
-            VCMTiming* master_timing = NULL);
+  explicit VCMTiming(Clock* clock, VCMTiming* master_timing = NULL);
   ~VCMTiming();
 
   // Resets the timing to the initial state.
@@ -58,7 +57,7 @@ class VCMTiming {
   // Stops the decoder timer, should be called when the decoder returns a frame
   // or when the decoded frame callback is called.
   int32_t StopDecodeTimer(uint32_t time_stamp,
-                          int64_t start_time_ms,
+                          int32_t decode_time_ms,
                           int64_t now_ms,
                           int64_t render_time_ms);
 
@@ -124,4 +123,4 @@ class VCMTiming {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/timing_unittest.cc
similarity index 84%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/timing_unittest.cc
index 0fa18d30ce..2e8df83683 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/timing_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/timing_unittest.cc
@@ -14,12 +14,12 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/internal_defines.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/internal_defines.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -55,8 +55,9 @@ TEST(ReceiverTiming, Tests) {
   clock.AdvanceTimeMilliseconds(1000);
   timing.SetJitterDelay(jitterDelayMs);
   timing.UpdateCurrentDelay(timeStamp);
-  waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(
-      timeStamp, clock.TimeInMilliseconds()), clock.TimeInMilliseconds());
+  waitTime = timing.MaxWaitingTime(
+      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
+      clock.TimeInMilliseconds());
   // Since we gradually increase the delay we only get 100 ms every second.
   EXPECT_EQ(jitterDelayMs - 10, waitTime);
 
@@ -85,9 +86,10 @@ TEST(ReceiverTiming, Tests) {
   for (int i = 0; i < 10; i++) {
     int64_t startTimeMs = clock.TimeInMilliseconds();
     clock.AdvanceTimeMilliseconds(10);
-    timing.StopDecodeTimer(timeStamp, startTimeMs,
-                           clock.TimeInMilliseconds(), timing.RenderTimeMs(
-                               timeStamp, clock.TimeInMilliseconds()));
+    timing.StopDecodeTimer(
+        timeStamp, clock.TimeInMilliseconds() - startTimeMs,
+        clock.TimeInMilliseconds(),
+        timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()));
     timeStamp += 90000 / 25;
     clock.AdvanceTimeMilliseconds(1000 / 25 - 10);
     timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
@@ -105,7 +107,7 @@ TEST(ReceiverTiming, Tests) {
   uint32_t minTotalDelayMs = 200;
   timing.set_min_playout_delay(minTotalDelayMs);
   clock.AdvanceTimeMilliseconds(5000);
-  timeStamp += 5*90000;
+  timeStamp += 5 * 90000;
   timing.UpdateCurrentDelay(timeStamp);
   const int kRenderDelayMs = 10;
   timing.set_render_delay(kRenderDelayMs);
@@ -121,7 +123,7 @@ TEST(ReceiverTiming, Tests) {
   // Reset playout delay.
   timing.set_min_playout_delay(0);
   clock.AdvanceTimeMilliseconds(5000);
-  timeStamp += 5*90000;
+  timeStamp += 5 * 90000;
   timing.UpdateCurrentDelay(timeStamp);
 }
 
@@ -135,8 +137,8 @@ TEST(ReceiverTiming, WrapAround) {
     timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
     clock.AdvanceTimeMilliseconds(1000 / kFramerate);
     timestamp += 90000 / kFramerate;
-    int64_t render_time = timing.RenderTimeMs(0xFFFFFFFFu,
-                                              clock.TimeInMilliseconds());
+    int64_t render_time =
+        timing.RenderTimeMs(0xFFFFFFFFu, clock.TimeInMilliseconds());
     EXPECT_EQ(3 * 1000 / kFramerate, render_time);
     render_time = timing.RenderTimeMs(89u,  // One second later in 90 kHz.
                                       clock.TimeInMilliseconds());
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.cc b/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.cc
index a684af7a39..a0aa67be4e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.cc
@@ -8,12 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
 const float kDefaultKeyFrameSizeAvgKBits = 0.9f;
 const float kDefaultKeyFrameRatio = 0.99f;
@@ -22,339 +21,266 @@ const float kDefaultDropRatioMax = 0.96f;
 const float kDefaultMaxTimeToDropFrames = 4.0f;  // In seconds.
 
 FrameDropper::FrameDropper()
-:
-_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
-_keyFrameRatio(kDefaultKeyFrameRatio),
-_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
-_enabled(true),
-_max_time_drops(kDefaultMaxTimeToDropFrames)
-{
-    Reset();
+    : _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
+      _keyFrameRatio(kDefaultKeyFrameRatio),
+      _dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
+      _enabled(true),
+      _max_time_drops(kDefaultMaxTimeToDropFrames) {
+  Reset();
 }
 
 FrameDropper::FrameDropper(float max_time_drops)
-:
-_keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
-_keyFrameRatio(kDefaultKeyFrameRatio),
-_dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
-_enabled(true),
-_max_time_drops(max_time_drops)
-{
-    Reset();
+    : _keyFrameSizeAvgKbits(kDefaultKeyFrameSizeAvgKBits),
+      _keyFrameRatio(kDefaultKeyFrameRatio),
+      _dropRatio(kDefaultDropRatioAlpha, kDefaultDropRatioMax),
+      _enabled(true),
+      _max_time_drops(max_time_drops) {
+  Reset();
 }
 
-void
-FrameDropper::Reset()
-{
-    _keyFrameRatio.Reset(0.99f);
-    _keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
-    _keyFrameSizeAvgKbits.Reset(0.9f);
-    _keyFrameCount = 0;
+void FrameDropper::Reset() {
+  _keyFrameRatio.Reset(0.99f);
+  _keyFrameRatio.Apply(
+      1.0f, 1.0f / 300.0f);  // 1 key frame every 10th second in 30 fps
+  _keyFrameSizeAvgKbits.Reset(0.9f);
+  _keyFrameCount = 0;
+  _accumulator = 0.0f;
+  _accumulatorMax = 150.0f;  // assume 300 kb/s and 0.5 s window
+  _targetBitRate = 300.0f;
+  _incoming_frame_rate = 30;
+  _keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
+  _dropNext = false;
+  _dropRatio.Reset(0.9f);
+  _dropRatio.Apply(0.0f, 0.0f);  // Initialize to 0
+  _dropCount = 0;
+  _windowSize = 0.5f;
+  _wasBelowMax = true;
+  _fastMode = false;  // start with normal (non-aggressive) mode
+  // Cap for the encoder buffer level/accumulator, in secs.
+  _cap_buffer_size = 3.0f;
+  // Cap on maximum amount of dropped frames between kept frames, in secs.
+  _max_time_drops = 4.0f;
+}
+
+void FrameDropper::Enable(bool enable) {
+  _enabled = enable;
+}
+
+void FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame) {
+  if (!_enabled) {
+    return;
+  }
+  float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
+  if (!deltaFrame &&
+      !_fastMode) {  // fast mode does not treat key-frames any different
+    _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
+    _keyFrameRatio.Apply(1.0, 1.0);
+    if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered()) {
+      // Remove the average key frame size since we
+      // compensate for key frames when adding delta
+      // frames.
+      frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
+    } else {
+      // Shouldn't be negative, so zero is the lower bound.
+      frameSizeKbits = 0;
+    }
+    if (_keyFrameRatio.filtered() > 1e-5 &&
+        1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
+      // We are sending key frames more often than our upper bound for
+      // how much we allow the key frame compensation to be spread
+      // out in time. Therefor we must use the key frame ratio rather
+      // than keyFrameSpreadFrames.
+      _keyFrameCount =
+          static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5);
+    } else {
+      // Compensate for the key frame the following frames
+      _keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
+    }
+  } else {
+    // Decrease the keyFrameRatio
+    _keyFrameRatio.Apply(1.0, 0.0);
+  }
+  // Change the level of the accumulator (bucket)
+  _accumulator += frameSizeKbits;
+  CapAccumulator();
+}
+
+void FrameDropper::Leak(uint32_t inputFrameRate) {
+  if (!_enabled) {
+    return;
+  }
+  if (inputFrameRate < 1) {
+    return;
+  }
+  if (_targetBitRate < 0.0f) {
+    return;
+  }
+  _keyFrameSpreadFrames = 0.5f * inputFrameRate;
+  // T is the expected bits per frame (target). If all frames were the same
+  // size,
+  // we would get T bits per frame. Notice that T is also weighted to be able to
+  // force a lower frame rate if wanted.
+  float T = _targetBitRate / inputFrameRate;
+  if (_keyFrameCount > 0) {
+    // Perform the key frame compensation
+    if (_keyFrameRatio.filtered() > 0 &&
+        1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames) {
+      T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
+    } else {
+      T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
+    }
+    _keyFrameCount--;
+  }
+  _accumulator -= T;
+  if (_accumulator < 0.0f) {
     _accumulator = 0.0f;
-    _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
-    _targetBitRate = 300.0f;
-    _incoming_frame_rate = 30;
-    _keyFrameSpreadFrames = 0.5f * _incoming_frame_rate;
-    _dropNext = false;
-    _dropRatio.Reset(0.9f);
-    _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
-    _dropCount = 0;
-    _windowSize = 0.5f;
-    _wasBelowMax = true;
-    _fastMode = false; // start with normal (non-aggressive) mode
-    // Cap for the encoder buffer level/accumulator, in secs.
-    _cap_buffer_size = 3.0f;
-    // Cap on maximum amount of dropped frames between kept frames, in secs.
-    _max_time_drops = 4.0f;
+  }
+  UpdateRatio();
 }
 
-void
-FrameDropper::Enable(bool enable)
-{
-    _enabled = enable;
+void FrameDropper::UpdateNack(uint32_t nackBytes) {
+  if (!_enabled) {
+    return;
+  }
+  _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
 }
 
-void
-FrameDropper::Fill(size_t frameSizeBytes, bool deltaFrame)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-    float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
-    if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
-    {
-        _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
-        _keyFrameRatio.Apply(1.0, 1.0);
-        if (frameSizeKbits > _keyFrameSizeAvgKbits.filtered())
-        {
-            // Remove the average key frame size since we
-            // compensate for key frames when adding delta
-            // frames.
-            frameSizeKbits -= _keyFrameSizeAvgKbits.filtered();
-        }
-        else
-        {
-            // Shouldn't be negative, so zero is the lower bound.
-            frameSizeKbits = 0;
-        }
-        if (_keyFrameRatio.filtered() > 1e-5 &&
-            1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
-        {
-            // We are sending key frames more often than our upper bound for
-            // how much we allow the key frame compensation to be spread
-            // out in time. Therefor we must use the key frame ratio rather
-            // than keyFrameSpreadFrames.
-            _keyFrameCount =
-                static_cast<int32_t>(1 / _keyFrameRatio.filtered() + 0.5);
-        }
-        else
-        {
-            // Compensate for the key frame the following frames
-            _keyFrameCount = static_cast<int32_t>(_keyFrameSpreadFrames + 0.5);
-        }
-    }
-    else
-    {
-        // Decrease the keyFrameRatio
-        _keyFrameRatio.Apply(1.0, 0.0);
-    }
-    // Change the level of the accumulator (bucket)
-    _accumulator += frameSizeKbits;
-    CapAccumulator();
+void FrameDropper::FillBucket(float inKbits, float outKbits) {
+  _accumulator += (inKbits - outKbits);
 }
 
-void
-FrameDropper::Leak(uint32_t inputFrameRate)
-{
-    if (!_enabled)
-    {
-        return;
+void FrameDropper::UpdateRatio() {
+  if (_accumulator > 1.3f * _accumulatorMax) {
+    // Too far above accumulator max, react faster
+    _dropRatio.UpdateBase(0.8f);
+  } else {
+    // Go back to normal reaction
+    _dropRatio.UpdateBase(0.9f);
+  }
+  if (_accumulator > _accumulatorMax) {
+    // We are above accumulator max, and should ideally
+    // drop a frame. Increase the dropRatio and drop
+    // the frame later.
+    if (_wasBelowMax) {
+      _dropNext = true;
     }
-    if (inputFrameRate < 1)
-    {
-        return;
+    if (_fastMode) {
+      // always drop in aggressive mode
+      _dropNext = true;
     }
-    if (_targetBitRate < 0.0f)
-    {
-        return;
-    }
-    _keyFrameSpreadFrames = 0.5f * inputFrameRate;
-    // T is the expected bits per frame (target). If all frames were the same size,
-    // we would get T bits per frame. Notice that T is also weighted to be able to
-    // force a lower frame rate if wanted.
-    float T = _targetBitRate / inputFrameRate;
-    if (_keyFrameCount > 0)
-    {
-        // Perform the key frame compensation
-        if (_keyFrameRatio.filtered() > 0 &&
-            1 / _keyFrameRatio.filtered() < _keyFrameSpreadFrames)
-        {
-            T -= _keyFrameSizeAvgKbits.filtered() * _keyFrameRatio.filtered();
-        }
-        else
-        {
-            T -= _keyFrameSizeAvgKbits.filtered() / _keyFrameSpreadFrames;
-        }
-        _keyFrameCount--;
-    }
-    _accumulator -= T;
-    if (_accumulator < 0.0f)
-    {
-        _accumulator = 0.0f;
-    }
-    UpdateRatio();
+
+    _dropRatio.Apply(1.0f, 1.0f);
+    _dropRatio.UpdateBase(0.9f);
+  } else {
+    _dropRatio.Apply(1.0f, 0.0f);
+  }
+  _wasBelowMax = _accumulator < _accumulatorMax;
 }
 
-void
-FrameDropper::UpdateNack(uint32_t nackBytes)
-{
-    if (!_enabled)
-    {
-        return;
-    }
-    _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
-}
-
-void
-FrameDropper::FillBucket(float inKbits, float outKbits)
-{
-    _accumulator += (inKbits - outKbits);
-}
-
-void
-FrameDropper::UpdateRatio()
-{
-    if (_accumulator > 1.3f * _accumulatorMax)
-    {
-        // Too far above accumulator max, react faster
-        _dropRatio.UpdateBase(0.8f);
-    }
-    else
-    {
-        // Go back to normal reaction
-        _dropRatio.UpdateBase(0.9f);
-    }
-    if (_accumulator > _accumulatorMax)
-    {
-        // We are above accumulator max, and should ideally
-        // drop a frame. Increase the dropRatio and drop
-        // the frame later.
-        if (_wasBelowMax)
-        {
-            _dropNext = true;
-        }
-        if (_fastMode)
-        {
-            // always drop in aggressive mode
-            _dropNext = true;
-        }
-
-        _dropRatio.Apply(1.0f, 1.0f);
-        _dropRatio.UpdateBase(0.9f);
-    }
-    else
-    {
-        _dropRatio.Apply(1.0f, 0.0f);
-    }
-    _wasBelowMax = _accumulator < _accumulatorMax;
-}
-
-// This function signals when to drop frames to the caller. It makes use of the dropRatio
+// This function signals when to drop frames to the caller. It makes use of the
+// dropRatio
 // to smooth out the drops over time.
-bool
-FrameDropper::DropFrame()
-{
-    if (!_enabled)
-    {
-        return false;
-    }
-    if (_dropNext)
-    {
-        _dropNext = false;
-        _dropCount = 0;
-    }
-
-    if (_dropRatio.filtered() >= 0.5f) // Drops per keep
-    {
-        // limit is the number of frames we should drop between each kept frame
-        // to keep our drop ratio. limit is positive in this case.
-        float denom = 1.0f - _dropRatio.filtered();
-        if (denom < 1e-5)
-        {
-            denom = (float)1e-5;
-        }
-        int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
-        // Put a bound on the max amount of dropped frames between each kept
-        // frame, in terms of frame rate and window size (secs).
-        int max_limit = static_cast<int>(_incoming_frame_rate *
-                                         _max_time_drops);
-        if (limit > max_limit) {
-          limit = max_limit;
-        }
-        if (_dropCount < 0)
-        {
-            // Reset the _dropCount since it was negative and should be positive.
-            if (_dropRatio.filtered() > 0.4f)
-            {
-                _dropCount = -_dropCount;
-            }
-            else
-            {
-                _dropCount = 0;
-            }
-        }
-        if (_dropCount < limit)
-        {
-            // As long we are below the limit we should drop frames.
-            _dropCount++;
-            return true;
-        }
-        else
-        {
-            // Only when we reset _dropCount a frame should be kept.
-            _dropCount = 0;
-            return false;
-        }
-    }
-    else if (_dropRatio.filtered() > 0.0f &&
-        _dropRatio.filtered() < 0.5f) // Keeps per drop
-    {
-        // limit is the number of frames we should keep between each drop
-        // in order to keep the drop ratio. limit is negative in this case,
-        // and the _dropCount is also negative.
-        float denom = _dropRatio.filtered();
-        if (denom < 1e-5)
-        {
-            denom = (float)1e-5;
-        }
-        int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
-        if (_dropCount > 0)
-        {
-            // Reset the _dropCount since we have a positive
-            // _dropCount, and it should be negative.
-            if (_dropRatio.filtered() < 0.6f)
-            {
-                _dropCount = -_dropCount;
-            }
-            else
-            {
-                _dropCount = 0;
-            }
-        }
-        if (_dropCount > limit)
-        {
-            if (_dropCount == 0)
-            {
-                // Drop frames when we reset _dropCount.
-                _dropCount--;
-                return true;
-            }
-            else
-            {
-                // Keep frames as long as we haven't reached limit.
-                _dropCount--;
-                return false;
-            }
-        }
-        else
-        {
-            _dropCount = 0;
-            return false;
-        }
-    }
-    _dropCount = 0;
+bool FrameDropper::DropFrame() {
+  if (!_enabled) {
     return false;
+  }
+  if (_dropNext) {
+    _dropNext = false;
+    _dropCount = 0;
+  }
 
-    // A simpler version, unfiltered and quicker
-    //bool dropNext = _dropNext;
-    //_dropNext = false;
-    //return dropNext;
+  if (_dropRatio.filtered() >= 0.5f) {  // Drops per keep
+    // limit is the number of frames we should drop between each kept frame
+    // to keep our drop ratio. limit is positive in this case.
+    float denom = 1.0f - _dropRatio.filtered();
+    if (denom < 1e-5) {
+      denom = 1e-5f;
+    }
+    int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+    // Put a bound on the max amount of dropped frames between each kept
+    // frame, in terms of frame rate and window size (secs).
+    int max_limit = static_cast<int>(_incoming_frame_rate * _max_time_drops);
+    if (limit > max_limit) {
+      limit = max_limit;
+    }
+    if (_dropCount < 0) {
+      // Reset the _dropCount since it was negative and should be positive.
+      if (_dropRatio.filtered() > 0.4f) {
+        _dropCount = -_dropCount;
+      } else {
+        _dropCount = 0;
+      }
+    }
+    if (_dropCount < limit) {
+      // As long we are below the limit we should drop frames.
+      _dropCount++;
+      return true;
+    } else {
+      // Only when we reset _dropCount a frame should be kept.
+      _dropCount = 0;
+      return false;
+    }
+  } else if (_dropRatio.filtered() > 0.0f &&
+             _dropRatio.filtered() < 0.5f) {  // Keeps per drop
+    // limit is the number of frames we should keep between each drop
+    // in order to keep the drop ratio. limit is negative in this case,
+    // and the _dropCount is also negative.
+    float denom = _dropRatio.filtered();
+    if (denom < 1e-5) {
+      denom = 1e-5f;
+    }
+    int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+    if (_dropCount > 0) {
+      // Reset the _dropCount since we have a positive
+      // _dropCount, and it should be negative.
+      if (_dropRatio.filtered() < 0.6f) {
+        _dropCount = -_dropCount;
+      } else {
+        _dropCount = 0;
+      }
+    }
+    if (_dropCount > limit) {
+      if (_dropCount == 0) {
+        // Drop frames when we reset _dropCount.
+        _dropCount--;
+        return true;
+      } else {
+        // Keep frames as long as we haven't reached limit.
+        _dropCount--;
+        return false;
+      }
+    } else {
+      _dropCount = 0;
+      return false;
+    }
+  }
+  _dropCount = 0;
+  return false;
+
+  // A simpler version, unfiltered and quicker
+  // bool dropNext = _dropNext;
+  // _dropNext = false;
+  // return dropNext;
 }
 
-void
-FrameDropper::SetRates(float bitRate, float incoming_frame_rate)
-{
-    // Bit rate of -1 means infinite bandwidth.
-    _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
-    if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
-    {
-        // Rescale the accumulator level if the accumulator max decreases
-        _accumulator = bitRate / _targetBitRate * _accumulator;
-    }
-    _targetBitRate = bitRate;
-    CapAccumulator();
-    _incoming_frame_rate = incoming_frame_rate;
+void FrameDropper::SetRates(float bitRate, float incoming_frame_rate) {
+  // Bit rate of -1 means infinite bandwidth.
+  _accumulatorMax = bitRate * _windowSize;  // bitRate * windowSize (in seconds)
+  if (_targetBitRate > 0.0f && bitRate < _targetBitRate &&
+      _accumulator > _accumulatorMax) {
+    // Rescale the accumulator level if the accumulator max decreases
+    _accumulator = bitRate / _targetBitRate * _accumulator;
+  }
+  _targetBitRate = bitRate;
+  CapAccumulator();
+  _incoming_frame_rate = incoming_frame_rate;
 }
 
-float
-FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const
-{
-    if (!_enabled)
-    {
-        return static_cast<float>(inputFrameRate);
-    }
-    return inputFrameRate * (1.0f - _dropRatio.filtered());
+float FrameDropper::ActualFrameRate(uint32_t inputFrameRate) const {
+  if (!_enabled) {
+    return static_cast<float>(inputFrameRate);
+  }
+  return inputFrameRate * (1.0f - _dropRatio.filtered());
 }
 
 // Put a cap on the accumulator, i.e., don't let it grow beyond some level.
@@ -366,5 +292,4 @@ void FrameDropper::CapAccumulator() {
     _accumulator = max_accumulator;
   }
 }
-
-}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.h
new file mode 100644
index 0000000000..7ec85ea880
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
+
+#include <cstddef>
+
+#include "webrtc/base/exp_filter.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// The Frame Dropper implements a variant of the leaky bucket algorithm
+// for keeping track of when to drop frames to avoid bit rate
+// over use when the encoder can't keep its bit rate.
+class FrameDropper {
+ public:
+  FrameDropper();
+  explicit FrameDropper(float max_time_drops);
+  virtual ~FrameDropper() {}
+
+  // Resets the FrameDropper to its initial state.
+  // This means that the frameRateWeight is set to its
+  // default value as well.
+  virtual void Reset();
+
+  virtual void Enable(bool enable);
+  // Answers the question if it's time to drop a frame
+  // if we want to reach a given frame rate. Must be
+  // called for every frame.
+  //
+  // Return value     : True if we should drop the current frame
+  virtual bool DropFrame();
+  // Updates the FrameDropper with the size of the latest encoded
+  // frame. The FrameDropper calculates a new drop ratio (can be
+  // seen as the probability to drop a frame) and updates its
+  // internal statistics.
+  //
+  // Input:
+  //          - frameSizeBytes    : The size of the latest frame
+  //                                returned from the encoder.
+  //          - deltaFrame        : True if the encoder returned
+  //                                a key frame.
+  virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
+
+  virtual void Leak(uint32_t inputFrameRate);
+
+  void UpdateNack(uint32_t nackBytes);
+
+  // Sets the target bit rate and the frame rate produced by
+  // the camera.
+  //
+  // Input:
+  //          - bitRate       : The target bit rate
+  virtual void SetRates(float bitRate, float incoming_frame_rate);
+
+  // Return value     : The current average frame rate produced
+  //                    if the DropFrame() function is used as
+  //                    instruction of when to drop frames.
+  virtual float ActualFrameRate(uint32_t inputFrameRate) const;
+
+ private:
+  void FillBucket(float inKbits, float outKbits);
+  void UpdateRatio();
+  void CapAccumulator();
+
+  rtc::ExpFilter _keyFrameSizeAvgKbits;
+  rtc::ExpFilter _keyFrameRatio;
+  float _keyFrameSpreadFrames;
+  int32_t _keyFrameCount;
+  float _accumulator;
+  float _accumulatorMax;
+  float _targetBitRate;
+  bool _dropNext;
+  rtc::ExpFilter _dropRatio;
+  int32_t _dropCount;
+  float _windowSize;
+  float _incoming_frame_rate;
+  bool _wasBelowMax;
+  bool _enabled;
+  bool _fastMode;
+  float _cap_buffer_size;
+  float _max_time_drops;
+};  // end of VCMFrameDropper class
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/include/frame_dropper.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/include/frame_dropper.h
deleted file mode 100644
index 2b78a7264f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/include/frame_dropper.h
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
-
-#include <cstddef>
-
-#include "webrtc/base/exp_filter.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc
-{
-
-// The Frame Dropper implements a variant of the leaky bucket algorithm
-// for keeping track of when to drop frames to avoid bit rate
-// over use when the encoder can't keep its bit rate.
-class FrameDropper
-{
-public:
-    FrameDropper();
-    explicit FrameDropper(float max_time_drops);
-    virtual ~FrameDropper() {}
-
-    // Resets the FrameDropper to its initial state.
-    // This means that the frameRateWeight is set to its
-    // default value as well.
-    virtual void Reset();
-
-    virtual void Enable(bool enable);
-    // Answers the question if it's time to drop a frame
-    // if we want to reach a given frame rate. Must be
-    // called for every frame.
-    //
-    // Return value     : True if we should drop the current frame
-    virtual bool DropFrame();
-    // Updates the FrameDropper with the size of the latest encoded
-    // frame. The FrameDropper calculates a new drop ratio (can be
-    // seen as the probability to drop a frame) and updates its
-    // internal statistics.
-    //
-    // Input:
-    //          - frameSizeBytes    : The size of the latest frame
-    //                                returned from the encoder.
-    //          - deltaFrame        : True if the encoder returned
-    //                                a key frame.
-    virtual void Fill(size_t frameSizeBytes, bool deltaFrame);
-
-    virtual void Leak(uint32_t inputFrameRate);
-
-    void UpdateNack(uint32_t nackBytes);
-
-    // Sets the target bit rate and the frame rate produced by
-    // the camera.
-    //
-    // Input:
-    //          - bitRate       : The target bit rate
-    virtual void SetRates(float bitRate, float incoming_frame_rate);
-
-    // Return value     : The current average frame rate produced
-    //                    if the DropFrame() function is used as
-    //                    instruction of when to drop frames.
-    virtual float ActualFrameRate(uint32_t inputFrameRate) const;
-
-private:
-    void FillBucket(float inKbits, float outKbits);
-    void UpdateRatio();
-    void CapAccumulator();
-
-    rtc::ExpFilter _keyFrameSizeAvgKbits;
-    rtc::ExpFilter _keyFrameRatio;
-    float _keyFrameSpreadFrames;
-    int32_t _keyFrameCount;
-    float _accumulator;
-    float _accumulatorMax;
-    float _targetBitRate;
-    bool _dropNext;
-    rtc::ExpFilter _dropRatio;
-    int32_t _dropCount;
-    float _windowSize;
-    float _incoming_frame_rate;
-    bool _wasBelowMax;
-    bool _enabled;
-    bool _fastMode;
-    float _cap_buffer_size;
-    float _max_time_drops;
-}; // end of VCMFrameDropper class
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_FRAME_DROPPER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h
deleted file mode 100644
index 1e31e5442a..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_MOCK_MOCK_FRAME_DROPPER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_MOCK_MOCK_FRAME_DROPPER_H_
-
-#include <string>
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/video_coding/utility/include/frame_dropper.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class MockFrameDropper : public FrameDropper {
- public:
-  MOCK_METHOD0(Reset,
-      void());
-  MOCK_METHOD1(Enable,
-      void(bool enable));
-  MOCK_METHOD0(DropFrame,
-      bool());
-  MOCK_METHOD2(Fill,
-      void(size_t frameSizeBytes, bool deltaFrame));
-  MOCK_METHOD1(Leak,
-      void(uint32_t inputFrameRate));
-  MOCK_METHOD2(SetRates,
-      void(float bitRate, float incoming_frame_rate));
-  MOCK_CONST_METHOD1(ActualFrameRate,
-      float(uint32_t inputFrameRate));
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_INCLUDE_MOCK_MOCK_FRAME_DROPPER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
new file mode 100644
index 0000000000..b68a4b8d5d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOCK_MOCK_FRAME_DROPPER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOCK_MOCK_FRAME_DROPPER_H_
+
+#include <string>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/video_coding/utility/frame_dropper.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class MockFrameDropper : public FrameDropper {
+ public:
+  MOCK_METHOD0(Reset, void());
+  MOCK_METHOD1(Enable, void(bool enable));
+  MOCK_METHOD0(DropFrame, bool());
+  MOCK_METHOD2(Fill, void(size_t frameSizeBytes, bool deltaFrame));
+  MOCK_METHOD1(Leak, void(uint32_t inputFrameRate));
+  MOCK_METHOD2(SetRates, void(float bitRate, float incoming_frame_rate));
+  MOCK_CONST_METHOD1(ActualFrameRate, float(uint32_t inputFrameRate));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOCK_MOCK_FRAME_DROPPER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/moving_average.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/moving_average.h
new file mode 100644
index 0000000000..494bfd51fb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/moving_average.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
+
+#include <list>
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+template <class T>
+class MovingAverage {
+ public:
+  MovingAverage();
+  void AddSample(T sample);
+  bool GetAverage(size_t num_samples, T* average);
+  void Reset();
+  int size();
+
+ private:
+  T sum_;
+  std::list<T> samples_;
+};
+
+template <class T>
+MovingAverage<T>::MovingAverage()
+    : sum_(static_cast<T>(0)) {}
+
+template <class T>
+void MovingAverage<T>::AddSample(T sample) {
+  samples_.push_back(sample);
+  sum_ += sample;
+}
+
+template <class T>
+bool MovingAverage<T>::GetAverage(size_t num_samples, T* avg) {
+  if (num_samples > samples_.size())
+    return false;
+
+  // Remove old samples.
+  while (num_samples < samples_.size()) {
+    sum_ -= samples_.front();
+    samples_.pop_front();
+  }
+
+  *avg = sum_ / static_cast<T>(num_samples);
+  return true;
+}
+
+template <class T>
+void MovingAverage<T>::Reset() {
+  sum_ = static_cast<T>(0);
+  samples_.clear();
+}
+
+template <class T>
+int MovingAverage<T>::size() {
+  return samples_.size();
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/qp_parser.cc b/media/webrtc/trunk/webrtc/modules/video_coding/utility/qp_parser.cc
new file mode 100644
index 0000000000..0916cb0094
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/qp_parser.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/utility/qp_parser.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
+
+namespace webrtc {
+
+bool QpParser::GetQp(const VCMEncodedFrame& frame, int* qp) {
+  switch (frame.CodecSpecific()->codecType) {
+    case kVideoCodecVP8:
+      // QP range: [0, 127].
+      return vp8::GetQp(frame.Buffer(), frame.Length(), qp);
+    default:
+      return false;
+  }
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/qp_parser.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/qp_parser.h
new file mode 100644
index 0000000000..0b644ef61c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/qp_parser.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
+
+#include "webrtc/modules/video_coding/encoded_frame.h"
+
+namespace webrtc {
+
+class QpParser {
+ public:
+  QpParser() {}
+  ~QpParser() {}
+
+  // Parses an encoded |frame| and extracts the |qp|.
+  // Returns true on success, false otherwise.
+  bool GetQp(const VCMEncodedFrame& frame, int* qp);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.cc b/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.cc
index 327748d224..76bf9f5b03 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.cc
@@ -7,7 +7,6 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
 #include "webrtc/modules/video_coding/utility/quality_scaler.h"
 
 namespace webrtc {
@@ -15,75 +14,117 @@ namespace webrtc {
 static const int kMinFps = 10;
 static const int kMeasureSeconds = 5;
 static const int kFramedropPercentThreshold = 60;
-static const int kLowQpThresholdDenominator = 3;
+
+const int QualityScaler::kDefaultLowQpDenominator = 3;
+// Note that this is the same for width and height to permit 120x90 in both
+// portrait and landscape mode.
+const int QualityScaler::kDefaultMinDownscaleDimension = 90;
 
 QualityScaler::QualityScaler()
-    : num_samples_(0), low_qp_threshold_(-1), downscale_shift_(0) {
-}
+    : num_samples_(0),
+      low_qp_threshold_(-1),
+      downscale_shift_(0),
+      framerate_down_(false),
+      min_width_(kDefaultMinDownscaleDimension),
+      min_height_(kDefaultMinDownscaleDimension) {}
 
-void QualityScaler::Init(int max_qp) {
+void QualityScaler::Init(int low_qp_threshold,
+                         int high_qp_threshold,
+                         bool use_framerate_reduction) {
   ClearSamples();
-  downscale_shift_ = 0;
-  low_qp_threshold_ = max_qp / kLowQpThresholdDenominator ;
+  low_qp_threshold_ = low_qp_threshold;
+  high_qp_threshold_ = high_qp_threshold;
+  use_framerate_reduction_ = use_framerate_reduction;
+  target_framerate_ = -1;
 }
 
+void QualityScaler::SetMinResolution(int min_width, int min_height) {
+  min_width_ = min_width;
+  min_height_ = min_height;
+}
+
+// Report framerate(fps) to estimate # of samples.
 void QualityScaler::ReportFramerate(int framerate) {
   num_samples_ = static_cast<size_t>(
       kMeasureSeconds * (framerate < kMinFps ? kMinFps : framerate));
+  framerate_ = framerate;
 }
 
-void QualityScaler::ReportEncodedFrame(int qp) {
-  average_qp_.AddSample(qp);
+void QualityScaler::ReportQP(int qp) {
   framedrop_percent_.AddSample(0);
+  average_qp_.AddSample(qp);
 }
 
 void QualityScaler::ReportDroppedFrame() {
   framedrop_percent_.AddSample(100);
 }
 
-QualityScaler::Resolution QualityScaler::GetScaledResolution(
-    const I420VideoFrame& frame) {
-  // Both of these should be set through InitEncode -> Should be set by now.
+void QualityScaler::OnEncodeFrame(const VideoFrame& frame) {
+  // Should be set through InitEncode -> Should be set by now.
   assert(low_qp_threshold_ >= 0);
   assert(num_samples_ > 0);
-  // Update scale factor.
-  int avg;
-  if (framedrop_percent_.GetAverage(num_samples_, &avg) &&
-      avg >= kFramedropPercentThreshold) {
-    AdjustScale(false);
-  } else if (average_qp_.GetAverage(num_samples_, &avg) &&
-             avg <= low_qp_threshold_) {
-    AdjustScale(true);
-  }
+  res_.width = frame.width();
+  res_.height = frame.height();
 
-  Resolution res;
-  res.width = frame.width();
-  res.height = frame.height();
+  // Update scale factor.
+  int avg_drop = 0;
+  int avg_qp = 0;
+
+  // When encoder consistently overshoots, framerate reduction and spatial
+  // resizing will be triggered to get a smoother video.
+  if ((framedrop_percent_.GetAverage(num_samples_, &avg_drop) &&
+       avg_drop >= kFramedropPercentThreshold) ||
+      (average_qp_.GetAverage(num_samples_, &avg_qp) &&
+       avg_qp > high_qp_threshold_)) {
+    // Reducing frame rate before spatial resolution change.
+    // Reduce frame rate only when it is above a certain number.
+    // Only one reduction is allowed for now.
+    // TODO(jackychen): Allow more than one framerate reduction.
+    if (use_framerate_reduction_ && !framerate_down_ && framerate_ >= 20) {
+      target_framerate_ = framerate_ / 2;
+      framerate_down_ = true;
+      // If frame rate has been updated, clear the buffer. We don't want
+      // spatial resolution to change right after frame rate change.
+      ClearSamples();
+    } else {
+      AdjustScale(false);
+    }
+  } else if (average_qp_.GetAverage(num_samples_, &avg_qp) &&
+             avg_qp <= low_qp_threshold_) {
+    if (use_framerate_reduction_ && framerate_down_) {
+      target_framerate_ = -1;
+      framerate_down_ = false;
+      ClearSamples();
+    } else {
+      AdjustScale(true);
+    }
+  }
 
   assert(downscale_shift_ >= 0);
   for (int shift = downscale_shift_;
-       shift > 0 && res.width > 1 && res.height > 1;
+       shift > 0 && (res_.width / 2 >= min_width_) &&
+       (res_.height / 2 >= min_height_);
        --shift) {
-    res.width >>= 1;
-    res.height >>= 1;
+    res_.width /= 2;
+    res_.height /= 2;
   }
-
-  return res;
 }
 
-const I420VideoFrame& QualityScaler::GetScaledFrame(
-    const I420VideoFrame& frame) {
-  Resolution res = GetScaledResolution(frame);
+QualityScaler::Resolution QualityScaler::GetScaledResolution() const {
+  return res_;
+}
+
+int QualityScaler::GetTargetFramerate() const {
+  return target_framerate_;
+}
+
+const VideoFrame& QualityScaler::GetScaledFrame(const VideoFrame& frame) {
+  Resolution res = GetScaledResolution();
   if (res.width == frame.width())
     return frame;
 
-  scaler_.Set(frame.width(),
-              frame.height(),
-              res.width,
-              res.height,
-              kI420,
-              kI420,
-              kScaleBox);
+  scaler_.Set(frame.width(), frame.height(), res.width, res.height, kI420,
+              kI420, kScaleBox);
   if (scaler_.Scale(frame, &scaled_frame_) != 0)
     return frame;
 
@@ -94,37 +135,9 @@ const I420VideoFrame& QualityScaler::GetScaledFrame(
   return scaled_frame_;
 }
 
-QualityScaler::MovingAverage::MovingAverage() : sum_(0) {
-}
-
-void QualityScaler::MovingAverage::AddSample(int sample) {
-  samples_.push_back(sample);
-  sum_ += sample;
-}
-
-bool QualityScaler::MovingAverage::GetAverage(size_t num_samples, int* avg) {
-  assert(num_samples > 0);
-  if (num_samples > samples_.size())
-    return false;
-
-  // Remove old samples.
-  while (num_samples < samples_.size()) {
-    sum_ -= samples_.front();
-    samples_.pop_front();
-  }
-
-  *avg = sum_ / static_cast<int>(num_samples);
-  return true;
-}
-
-void QualityScaler::MovingAverage::Reset() {
-  sum_ = 0;
-  samples_.clear();
-}
-
 void QualityScaler::ClearSamples() {
-  average_qp_.Reset();
   framedrop_percent_.Reset();
+  average_qp_.Reset();
 }
 
 void QualityScaler::AdjustScale(bool up) {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.h
index 47d6cb1c5e..a1233cca51 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler.h
@@ -11,53 +11,55 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
 
-#include <list>
-
 #include "webrtc/common_video/libyuv/include/scaler.h"
+#include "webrtc/modules/video_coding/utility/moving_average.h"
 
 namespace webrtc {
 class QualityScaler {
  public:
+  static const int kDefaultLowQpDenominator;
+  static const int kDefaultMinDownscaleDimension;
   struct Resolution {
     int width;
     int height;
   };
 
   QualityScaler();
-  void Init(int max_qp);
-
+  void Init(int low_qp_threshold,
+            int high_qp_threshold,
+            bool use_framerate_reduction);
+  void SetMinResolution(int min_width, int min_height);
   void ReportFramerate(int framerate);
-  void ReportEncodedFrame(int qp);
+  void ReportQP(int qp);
   void ReportDroppedFrame();
-
-  Resolution GetScaledResolution(const I420VideoFrame& frame);
-  const I420VideoFrame& GetScaledFrame(const I420VideoFrame& frame);
+  void Reset(int framerate, int bitrate, int width, int height);
+  void OnEncodeFrame(const VideoFrame& frame);
+  Resolution GetScaledResolution() const;
+  const VideoFrame& GetScaledFrame(const VideoFrame& frame);
+  int GetTargetFramerate() const;
+  int downscale_shift() const { return downscale_shift_; }
 
  private:
-  class MovingAverage {
-   public:
-    MovingAverage();
-    void AddSample(int sample);
-    bool GetAverage(size_t num_samples, int* average);
-    void Reset();
-
-   private:
-    int sum_;
-    std::list<int> samples_;
-  };
-
   void AdjustScale(bool up);
   void ClearSamples();
 
   Scaler scaler_;
-  I420VideoFrame scaled_frame_;
+  VideoFrame scaled_frame_;
 
   size_t num_samples_;
+  int framerate_;
+  int target_framerate_;
   int low_qp_threshold_;
-  MovingAverage average_qp_;
-  MovingAverage framedrop_percent_;
+  int high_qp_threshold_;
+  MovingAverage<int> framedrop_percent_;
+  MovingAverage<int> average_qp_;
+  Resolution res_;
 
   int downscale_shift_;
+  int framerate_down_;
+  bool use_framerate_reduction_;
+  int min_width_;
+  int min_height_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
index 381b959c95..bad73a748c 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
@@ -21,37 +21,60 @@ static const int kHeight = 1080;
 static const int kFramerate = 30;
 static const int kLowQp = 15;
 static const int kNormalQp = 30;
+static const int kHighQp = 40;
 static const int kMaxQp = 56;
 }  // namespace
 
 class QualityScalerTest : public ::testing::Test {
+ public:
+  // Temporal and spatial resolution.
+  struct Resolution {
+    int framerate;
+    int width;
+    int height;
+  };
+
  protected:
-  enum ScaleDirection { kScaleDown, kScaleUp };
+  enum ScaleDirection {
+    kKeepScaleAtHighQp,
+    kScaleDown,
+    kScaleDownAboveHighQp,
+    kScaleUp
+  };
+  enum BadQualityMetric { kDropFrame, kReportLowQP };
 
   QualityScalerTest() {
-    input_frame_.CreateEmptyFrame(
-        kWidth, kHeight, kWidth, kHalfWidth, kHalfWidth);
-    qs_.Init(kMaxQp);
+    input_frame_.CreateEmptyFrame(kWidth, kHeight, kWidth, kHalfWidth,
+                                  kHalfWidth);
+    qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator, kHighQp, false);
     qs_.ReportFramerate(kFramerate);
+    qs_.OnEncodeFrame(input_frame_);
   }
 
-  void TriggerScale(ScaleDirection scale_direction) {
-    int initial_width = qs_.GetScaledResolution(input_frame_).width;
+  bool TriggerScale(ScaleDirection scale_direction) {
+    qs_.OnEncodeFrame(input_frame_);
+    int initial_width = qs_.GetScaledResolution().width;
     for (int i = 0; i < kFramerate * kNumSeconds; ++i) {
       switch (scale_direction) {
         case kScaleUp:
-          qs_.ReportEncodedFrame(kLowQp);
+          qs_.ReportQP(kLowQp);
           break;
         case kScaleDown:
           qs_.ReportDroppedFrame();
           break;
+        case kKeepScaleAtHighQp:
+          qs_.ReportQP(kHighQp);
+          break;
+        case kScaleDownAboveHighQp:
+          qs_.ReportQP(kHighQp + 1);
+          break;
       }
-
-      if (qs_.GetScaledResolution(input_frame_).width != initial_width)
-        return;
+      qs_.OnEncodeFrame(input_frame_);
+      if (qs_.GetScaledResolution().width != initial_width)
+        return true;
     }
 
-    FAIL() << "No downscale within " << kNumSeconds << " seconds.";
+    return false;
   }
 
   void ExpectOriginalFrame() {
@@ -60,8 +83,9 @@ class QualityScalerTest : public ::testing::Test {
   }
 
   void ExpectScaleUsingReportedResolution() {
-    QualityScaler::Resolution res = qs_.GetScaledResolution(input_frame_);
-    const I420VideoFrame& scaled_frame = qs_.GetScaledFrame(input_frame_);
+    qs_.OnEncodeFrame(input_frame_);
+    QualityScaler::Resolution res = qs_.GetScaledResolution();
+    const VideoFrame& scaled_frame = qs_.GetScaledFrame(input_frame_);
     EXPECT_EQ(res.width, scaled_frame.width());
     EXPECT_EQ(res.height, scaled_frame.height());
   }
@@ -70,8 +94,22 @@ class QualityScalerTest : public ::testing::Test {
 
   void DoesNotDownscaleFrameDimensions(int width, int height);
 
+  Resolution TriggerResolutionChange(BadQualityMetric dropframe_lowqp,
+                                     int num_second,
+                                     int initial_framerate);
+
+  void VerifyQualityAdaptation(int initial_framerate,
+                               int seconds,
+                               bool expect_spatial_resize,
+                               bool expect_framerate_reduction);
+
+  void DownscaleEndsAt(int input_width,
+                       int input_height,
+                       int end_width,
+                       int end_height);
+
   QualityScaler qs_;
-  I420VideoFrame input_frame_;
+  VideoFrame input_frame_;
 };
 
 TEST_F(QualityScalerTest, UsesOriginalFrameInitially) {
@@ -79,24 +117,43 @@ TEST_F(QualityScalerTest, UsesOriginalFrameInitially) {
 }
 
 TEST_F(QualityScalerTest, ReportsOriginalResolutionInitially) {
-  QualityScaler::Resolution res = qs_.GetScaledResolution(input_frame_);
+  qs_.OnEncodeFrame(input_frame_);
+  QualityScaler::Resolution res = qs_.GetScaledResolution();
   EXPECT_EQ(input_frame_.width(), res.width);
   EXPECT_EQ(input_frame_.height(), res.height);
 }
 
 TEST_F(QualityScalerTest, DownscalesAfterContinuousFramedrop) {
-  TriggerScale(kScaleDown);
-  QualityScaler::Resolution res = qs_.GetScaledResolution(input_frame_);
+  EXPECT_TRUE(TriggerScale(kScaleDown)) << "No downscale within " << kNumSeconds
+                                        << " seconds.";
+  QualityScaler::Resolution res = qs_.GetScaledResolution();
+  EXPECT_LT(res.width, input_frame_.width());
+  EXPECT_LT(res.height, input_frame_.height());
+}
+
+TEST_F(QualityScalerTest, KeepsScaleAtHighQp) {
+  EXPECT_FALSE(TriggerScale(kKeepScaleAtHighQp))
+      << "Downscale at high threshold which should keep scale.";
+  QualityScaler::Resolution res = qs_.GetScaledResolution();
+  EXPECT_EQ(res.width, input_frame_.width());
+  EXPECT_EQ(res.height, input_frame_.height());
+}
+
+TEST_F(QualityScalerTest, DownscalesAboveHighQp) {
+  EXPECT_TRUE(TriggerScale(kScaleDownAboveHighQp))
+      << "No downscale within " << kNumSeconds << " seconds.";
+  QualityScaler::Resolution res = qs_.GetScaledResolution();
   EXPECT_LT(res.width, input_frame_.width());
   EXPECT_LT(res.height, input_frame_.height());
 }
 
 TEST_F(QualityScalerTest, DownscalesAfterTwoThirdsFramedrop) {
   for (int i = 0; i < kFramerate * kNumSeconds / 3; ++i) {
-    qs_.ReportEncodedFrame(kNormalQp);
+    qs_.ReportQP(kNormalQp);
     qs_.ReportDroppedFrame();
     qs_.ReportDroppedFrame();
-    if (qs_.GetScaledResolution(input_frame_).width < input_frame_.width())
+    qs_.OnEncodeFrame(input_frame_);
+    if (qs_.GetScaledResolution().width < input_frame_.width())
       return;
   }
 
@@ -105,34 +162,39 @@ TEST_F(QualityScalerTest, DownscalesAfterTwoThirdsFramedrop) {
 
 TEST_F(QualityScalerTest, DoesNotDownscaleOnNormalQp) {
   for (int i = 0; i < kFramerate * kNumSeconds; ++i) {
-    qs_.ReportEncodedFrame(kNormalQp);
-    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution(input_frame_).width)
+    qs_.ReportQP(kNormalQp);
+    qs_.OnEncodeFrame(input_frame_);
+    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution().width)
         << "Unexpected scale on half framedrop.";
   }
 }
 
 TEST_F(QualityScalerTest, DoesNotDownscaleAfterHalfFramedrop) {
   for (int i = 0; i < kFramerate * kNumSeconds / 2; ++i) {
-    qs_.ReportEncodedFrame(kNormalQp);
-    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution(input_frame_).width)
+    qs_.ReportQP(kNormalQp);
+    qs_.OnEncodeFrame(input_frame_);
+    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution().width)
         << "Unexpected scale on half framedrop.";
 
     qs_.ReportDroppedFrame();
-    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution(input_frame_).width)
+    qs_.OnEncodeFrame(input_frame_);
+    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution().width)
         << "Unexpected scale on half framedrop.";
   }
 }
 
 void QualityScalerTest::ContinuouslyDownscalesByHalfDimensionsAndBackUp() {
   const int initial_min_dimension = input_frame_.width() < input_frame_.height()
-                                  ? input_frame_.width()
-                                  : input_frame_.height();
+                                        ? input_frame_.width()
+                                        : input_frame_.height();
   int min_dimension = initial_min_dimension;
   int current_shift = 0;
   // Drop all frames to force-trigger downscaling.
-  while (min_dimension > 16) {
-    TriggerScale(kScaleDown);
-    QualityScaler::Resolution res = qs_.GetScaledResolution(input_frame_);
+  while (min_dimension >= 2 * QualityScaler::kDefaultMinDownscaleDimension) {
+    EXPECT_TRUE(TriggerScale(kScaleDown)) << "No downscale within "
+                                          << kNumSeconds << " seconds.";
+    qs_.OnEncodeFrame(input_frame_);
+    QualityScaler::Resolution res = qs_.GetScaledResolution();
     min_dimension = res.width < res.height ? res.width : res.height;
     ++current_shift;
     ASSERT_EQ(input_frame_.width() >> current_shift, res.width);
@@ -142,8 +204,10 @@ void QualityScalerTest::ContinuouslyDownscalesByHalfDimensionsAndBackUp() {
 
   // Make sure we can scale back with good-quality frames.
   while (min_dimension < initial_min_dimension) {
-    TriggerScale(kScaleUp);
-    QualityScaler::Resolution res = qs_.GetScaledResolution(input_frame_);
+    EXPECT_TRUE(TriggerScale(kScaleUp)) << "No upscale within " << kNumSeconds
+                                        << " seconds.";
+    qs_.OnEncodeFrame(input_frame_);
+    QualityScaler::Resolution res = qs_.GetScaledResolution();
     min_dimension = res.width < res.height ? res.width : res.height;
     --current_shift;
     ASSERT_EQ(input_frame_.width() >> current_shift, res.width);
@@ -153,7 +217,7 @@ void QualityScalerTest::ContinuouslyDownscalesByHalfDimensionsAndBackUp() {
 
   // Verify we don't start upscaling after further low use.
   for (int i = 0; i < kFramerate * kNumSeconds; ++i) {
-    qs_.ReportEncodedFrame(kLowQp);
+    qs_.ReportQP(kLowQp);
     ExpectOriginalFrame();
   }
 }
@@ -167,18 +231,19 @@ TEST_F(QualityScalerTest,
   const int kOddWidth = 517;
   const int kHalfOddWidth = (kOddWidth + 1) / 2;
   const int kOddHeight = 1239;
-  input_frame_.CreateEmptyFrame(
-      kOddWidth, kOddHeight, kOddWidth, kHalfOddWidth, kHalfOddWidth);
+  input_frame_.CreateEmptyFrame(kOddWidth, kOddHeight, kOddWidth, kHalfOddWidth,
+                                kHalfOddWidth);
   ContinuouslyDownscalesByHalfDimensionsAndBackUp();
 }
 
 void QualityScalerTest::DoesNotDownscaleFrameDimensions(int width, int height) {
-  input_frame_.CreateEmptyFrame(
-      width, height, width, (width + 1) / 2, (width + 1) / 2);
+  input_frame_.CreateEmptyFrame(width, height, width, (width + 1) / 2,
+                                (width + 1) / 2);
 
   for (int i = 0; i < kFramerate * kNumSeconds; ++i) {
     qs_.ReportDroppedFrame();
-    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution(input_frame_).width)
+    qs_.OnEncodeFrame(input_frame_);
+    ASSERT_EQ(input_frame_.width(), qs_.GetScaledResolution().width)
         << "Unexpected scale of minimal-size frame.";
   }
 }
@@ -195,4 +260,166 @@ TEST_F(QualityScalerTest, DoesNotDownscaleFrom1Px) {
   DoesNotDownscaleFrameDimensions(1, 1);
 }
 
+QualityScalerTest::Resolution QualityScalerTest::TriggerResolutionChange(
+    BadQualityMetric dropframe_lowqp,
+    int num_second,
+    int initial_framerate) {
+  QualityScalerTest::Resolution res;
+  res.framerate = initial_framerate;
+  qs_.OnEncodeFrame(input_frame_);
+  res.width = qs_.GetScaledResolution().width;
+  res.height = qs_.GetScaledResolution().height;
+  for (int i = 0; i < kFramerate * num_second; ++i) {
+    switch (dropframe_lowqp) {
+      case kReportLowQP:
+        qs_.ReportQP(kLowQp);
+        break;
+      case kDropFrame:
+        qs_.ReportDroppedFrame();
+        break;
+    }
+    qs_.OnEncodeFrame(input_frame_);
+    // Simulate the case when SetRates is called right after reducing
+    // framerate.
+    qs_.ReportFramerate(initial_framerate);
+    res.framerate = qs_.GetTargetFramerate();
+    if (res.framerate != -1)
+      qs_.ReportFramerate(res.framerate);
+    res.width = qs_.GetScaledResolution().width;
+    res.height = qs_.GetScaledResolution().height;
+  }
+  return res;
+}
+
+void QualityScalerTest::VerifyQualityAdaptation(
+    int initial_framerate,
+    int seconds,
+    bool expect_spatial_resize,
+    bool expect_framerate_reduction) {
+  const int kDisabledBadQpThreshold = kMaxQp + 1;
+  qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator,
+           kDisabledBadQpThreshold, true);
+  qs_.OnEncodeFrame(input_frame_);
+  int init_width = qs_.GetScaledResolution().width;
+  int init_height = qs_.GetScaledResolution().height;
+
+  // Test reducing framerate by dropping frame continuously.
+  QualityScalerTest::Resolution res =
+      TriggerResolutionChange(kDropFrame, seconds, initial_framerate);
+
+  if (expect_framerate_reduction) {
+    EXPECT_LT(res.framerate, initial_framerate);
+  } else {
+    // No framerate reduction, video decimator should be disabled.
+    EXPECT_EQ(-1, res.framerate);
+  }
+
+  if (expect_spatial_resize) {
+    EXPECT_LT(res.width, init_width);
+    EXPECT_LT(res.height, init_height);
+  } else {
+    EXPECT_EQ(init_width, res.width);
+    EXPECT_EQ(init_height, res.height);
+  }
+
+  // The "seconds * 1.5" is to ensure spatial resolution to recover.
+  // For example, in 10 seconds test, framerate reduction happens in the first
+  // 5 seconds from 30fps to 15fps and causes the buffer size to be half of the
+  // original one. Then it will take only 75 samples to downscale (twice in 150
+  // samples). So to recover the resolution changes, we need more than 10
+  // seconds (i.e, seconds * 1.5). This is because the framerate increases
+  // before spatial size recovers, so it will take 150 samples to recover
+  // spatial size (300 for twice).
+  res = TriggerResolutionChange(kReportLowQP, seconds * 1.5, initial_framerate);
+  EXPECT_EQ(-1, res.framerate);
+  EXPECT_EQ(init_width, res.width);
+  EXPECT_EQ(init_height, res.height);
+}
+
+// In 5 seconds test, only framerate adjusting should happen.
+TEST_F(QualityScalerTest, ChangeFramerateOnly) {
+  VerifyQualityAdaptation(kFramerate, 5, false, true);
+}
+
+// In 10 seconds test, framerate adjusting and scaling are both
+// triggered, it shows that scaling would happen after framerate
+// adjusting.
+TEST_F(QualityScalerTest, ChangeFramerateAndSpatialSize) {
+  VerifyQualityAdaptation(kFramerate, 10, true, true);
+}
+
+// When starting from a low framerate, only spatial size will be changed.
+TEST_F(QualityScalerTest, ChangeSpatialSizeOnly) {
+  qs_.ReportFramerate(kFramerate >> 1);
+  VerifyQualityAdaptation(kFramerate >> 1, 10, true, false);
+}
+
+TEST_F(QualityScalerTest, DoesNotDownscaleBelow2xDefaultMinDimensionsWidth) {
+  DoesNotDownscaleFrameDimensions(
+      2 * QualityScaler::kDefaultMinDownscaleDimension - 1, 1000);
+}
+
+TEST_F(QualityScalerTest, DoesNotDownscaleBelow2xDefaultMinDimensionsHeight) {
+  DoesNotDownscaleFrameDimensions(
+      1000, 2 * QualityScaler::kDefaultMinDownscaleDimension - 1);
+}
+
+void QualityScalerTest::DownscaleEndsAt(int input_width,
+                                        int input_height,
+                                        int end_width,
+                                        int end_height) {
+  // Create a frame with 2x expected end width/height to verify that we can
+  // scale down to expected end width/height.
+  input_frame_.CreateEmptyFrame(input_width, input_height, input_width,
+                                (input_width + 1) / 2, (input_width + 1) / 2);
+
+  int last_width = input_width;
+  int last_height = input_height;
+  // Drop all frames to force-trigger downscaling.
+  while (true) {
+    TriggerScale(kScaleDown);
+    QualityScaler::Resolution res = qs_.GetScaledResolution();
+    if (last_width == res.width) {
+      EXPECT_EQ(last_height, res.height);
+      EXPECT_EQ(end_width, res.width);
+      EXPECT_EQ(end_height, res.height);
+      break;
+    }
+    last_width = res.width;
+    last_height = res.height;
+  }
+}
+
+TEST_F(QualityScalerTest, DefaultDownscalesTo160x90) {
+  DownscaleEndsAt(320, 180, 160, 90);
+}
+
+TEST_F(QualityScalerTest, DefaultDownscalesTo90x160) {
+  DownscaleEndsAt(180, 320, 90, 160);
+}
+
+TEST_F(QualityScalerTest, DefaultDownscalesFrom1280x720To160x90) {
+  DownscaleEndsAt(1280, 720, 160, 90);
+}
+
+TEST_F(QualityScalerTest, DefaultDoesntDownscaleBelow160x90) {
+  DownscaleEndsAt(320 - 1, 180 - 1, 320 - 1, 180 - 1);
+}
+
+TEST_F(QualityScalerTest, DefaultDoesntDownscaleBelow90x160) {
+  DownscaleEndsAt(180 - 1, 320 - 1, 180 - 1, 320 - 1);
+}
+
+TEST_F(QualityScalerTest, RespectsMinResolutionWidth) {
+  // Should end at 200x100, as width can't go lower.
+  qs_.SetMinResolution(200, 10);
+  DownscaleEndsAt(1600, 800, 200, 100);
+}
+
+TEST_F(QualityScalerTest, RespectsMinResolutionHeight) {
+  // Should end at 100x200, as height can't go lower.
+  qs_.SetMinResolution(10, 200);
+  DownscaleEndsAt(800, 1600, 100, 200);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/video_coding_utility.gyp b/media/webrtc/trunk/webrtc/modules/video_coding/utility/video_coding_utility.gyp
index eccc4da563..42cbb3d4e0 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/utility/video_coding_utility.gyp
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/video_coding_utility.gyp
@@ -19,9 +19,14 @@
       ],
       'sources': [
         'frame_dropper.cc',
-        'include/frame_dropper.h',
+        'frame_dropper.h',
+        'moving_average.h',
+        'qp_parser.cc',
+        'qp_parser.h',
         'quality_scaler.cc',
         'quality_scaler.h',
+        'vp8_header_parser.cc',
+        'vp8_header_parser.h',
       ],
     },
   ], # targets
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/vp8_header_parser.cc b/media/webrtc/trunk/webrtc/modules/video_coding/utility/vp8_header_parser.cc
new file mode 100644
index 0000000000..631385d0f2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/vp8_header_parser.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
+
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+
+namespace vp8 {
+namespace {
+const size_t kCommonPayloadHeaderLength = 3;
+const size_t kKeyPayloadHeaderLength = 10;
+}  // namespace
+
+static uint32_t BSwap32(uint32_t x) {
+  return (x >> 24) | ((x >> 8) & 0xff00) | ((x << 8) & 0xff0000) | (x << 24);
+}
+
+static void VP8LoadFinalBytes(VP8BitReader* const br) {
+  // Only read 8bits at a time.
+  if (br->buf_ < br->buf_end_) {
+    br->bits_ += 8;
+    br->value_ = static_cast<uint32_t>(*br->buf_++) | (br->value_ << 8);
+  } else if (!br->eof_) {
+    br->value_ <<= 8;
+    br->bits_ += 8;
+    br->eof_ = 1;
+  }
+}
+
+static void VP8LoadNewBytes(VP8BitReader* const br) {
+  int BITS = 24;
+  // Read 'BITS' bits at a time.
+  if (br->buf_ + sizeof(uint32_t) <= br->buf_end_) {
+    uint32_t bits;
+    const uint32_t in_bits = *(const uint32_t*)(br->buf_);
+    br->buf_ += BITS >> 3;
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+    bits = static_cast<uint32_t>(in_bits);
+    if (BITS != 8 * sizeof(uint32_t))
+      bits >>= (8 * sizeof(uint32_t) - BITS);
+#else
+    bits = BSwap32(in_bits);
+    bits >>= 32 - BITS;
+#endif
+    br->value_ = bits | (br->value_ << BITS);
+    br->bits_ += BITS;
+  } else {
+    VP8LoadFinalBytes(br);
+  }
+}
+
+static void VP8InitBitReader(VP8BitReader* const br,
+                             const uint8_t* const start,
+                             const uint8_t* const end) {
+  br->range_ = 255 - 1;
+  br->buf_ = start;
+  br->buf_end_ = end;
+  br->value_ = 0;
+  br->bits_ = -8;  // To load the very first 8bits.
+  br->eof_ = 0;
+  VP8LoadNewBytes(br);
+}
+
+// Read a bit with proba 'prob'.
+static int VP8GetBit(VP8BitReader* const br, int prob) {
+  uint8_t range = br->range_;
+  if (br->bits_ < 0) {
+    VP8LoadNewBytes(br);
+  }
+
+  const int pos = br->bits_;
+  const uint8_t split = (range * prob) >> 8;
+  const uint8_t value = static_cast<uint8_t>(br->value_ >> pos);
+  int bit;
+  if (value > split) {
+    range -= split + 1;
+    br->value_ -= static_cast<uint32_t>(split + 1) << pos;
+    bit = 1;
+  } else {
+    range = split;
+    bit = 0;
+  }
+  if (range <= static_cast<uint8_t>(0x7e)) {
+    const int shift = kVP8Log2Range[range];
+    range = kVP8NewRange[range];
+    br->bits_ -= shift;
+  }
+  br->range_ = range;
+  return bit;
+}
+
+static uint32_t VP8GetValue(VP8BitReader* const br, int bits) {
+  uint32_t v = 0;
+  while (bits-- > 0) {
+    v |= VP8GetBit(br, 0x80) << bits;
+  }
+  return v;
+}
+
+static uint32_t VP8Get(VP8BitReader* const br) {
+  return VP8GetValue(br, 1);
+}
+
+static int32_t VP8GetSignedValue(VP8BitReader* const br, int bits) {
+  const int value = VP8GetValue(br, bits);
+  return VP8Get(br) ? -value : value;
+}
+
+static void ParseSegmentHeader(VP8BitReader* br) {
+  int use_segment = VP8Get(br);
+  if (use_segment) {
+    int update_map = VP8Get(br);
+    if (VP8Get(br)) {
+      int s;
+      VP8Get(br);
+      for (s = 0; s < NUM_MB_SEGMENTS; ++s) {
+        VP8Get(br) ? VP8GetSignedValue(br, 7) : 0;
+      }
+      for (s = 0; s < NUM_MB_SEGMENTS; ++s) {
+        VP8Get(br) ? VP8GetSignedValue(br, 6) : 0;
+      }
+    }
+    if (update_map) {
+      int s;
+      for (s = 0; s < MB_FEATURE_TREE_PROBS; ++s) {
+        VP8Get(br) ? VP8GetValue(br, 8) : 255;
+      }
+    }
+  }
+}
+
+static void ParseFilterHeader(VP8BitReader* br) {
+  VP8Get(br);
+  VP8GetValue(br, 6);
+  VP8GetValue(br, 3);
+  int use_lf_delta = VP8Get(br);
+  if (use_lf_delta) {
+    if (VP8Get(br)) {
+      int i;
+      for (i = 0; i < NUM_REF_LF_DELTAS; ++i) {
+        if (VP8Get(br)) {
+          VP8GetSignedValue(br, 6);
+        }
+      }
+      for (i = 0; i < NUM_MODE_LF_DELTAS; ++i) {
+        if (VP8Get(br)) {
+          VP8GetSignedValue(br, 6);
+        }
+      }
+    }
+  }
+}
+
+bool GetQp(const uint8_t* buf, size_t length, int* qp) {
+  if (length < kCommonPayloadHeaderLength) {
+    LOG(LS_WARNING) << "Failed to get QP, invalid length.";
+    return false;
+  }
+  VP8BitReader br;
+  const uint32_t bits = buf[0] | (buf[1] << 8) | (buf[2] << 16);
+  int key_frame = !(bits & 1);
+  // Size of first partition in bytes.
+  uint32_t partition_length = (bits >> 5);
+  size_t header_length = kCommonPayloadHeaderLength;
+  if (key_frame) {
+    header_length = kKeyPayloadHeaderLength;
+  }
+  if (header_length + partition_length > length) {
+    LOG(LS_WARNING) << "Failed to get QP, invalid length: " << length;
+    return false;
+  }
+  buf += header_length;
+
+  VP8InitBitReader(&br, buf, buf + partition_length);
+  if (key_frame) {
+    // Color space and pixel type.
+    VP8Get(&br);
+    VP8Get(&br);
+  }
+  ParseSegmentHeader(&br);
+  ParseFilterHeader(&br);
+  // Number of coefficient data partitions.
+  VP8GetValue(&br, 2);
+  // Base QP.
+  const int base_q0 = VP8GetValue(&br, 7);
+  if (br.eof_ == 1) {
+    LOG(LS_WARNING) << "Failed to get QP, end of file reached.";
+    return false;
+  }
+  *qp = base_q0;
+  return true;
+}
+
+}  // namespace vp8
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/utility/vp8_header_parser.h b/media/webrtc/trunk/webrtc/modules/video_coding/utility/vp8_header_parser.h
new file mode 100644
index 0000000000..b0c684c578
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/utility/vp8_header_parser.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
+
+#include <stdint.h>
+#include <stdio.h>
+
+namespace webrtc {
+
+namespace vp8 {
+
+enum {
+  MB_FEATURE_TREE_PROBS = 3,
+  NUM_MB_SEGMENTS = 4,
+  NUM_REF_LF_DELTAS = 4,
+  NUM_MODE_LF_DELTAS = 4,
+};
+
+typedef struct VP8BitReader VP8BitReader;
+struct VP8BitReader {
+  // Boolean decoder.
+  uint32_t value_;  // Current value.
+  uint32_t range_;  // Current range minus 1. In [127, 254] interval.
+  int bits_;        // Number of valid bits left.
+  // Read buffer.
+  const uint8_t* buf_;      // Next byte to be read.
+  const uint8_t* buf_end_;  // End of read buffer.
+  int eof_;                 // True if input is exhausted.
+};
+
+const uint8_t kVP8Log2Range[128] = {
+    7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, 3, 3,
+    3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+    2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0};
+
+// range = ((range - 1) << kVP8Log2Range[range]) + 1
+const uint8_t kVP8NewRange[128] = {
+    127, 127, 191, 127, 159, 191, 223, 127, 143, 159, 175, 191, 207, 223, 239,
+    127, 135, 143, 151, 159, 167, 175, 183, 191, 199, 207, 215, 223, 231, 239,
+    247, 127, 131, 135, 139, 143, 147, 151, 155, 159, 163, 167, 171, 175, 179,
+    183, 187, 191, 195, 199, 203, 207, 211, 215, 219, 223, 227, 231, 235, 239,
+    243, 247, 251, 127, 129, 131, 133, 135, 137, 139, 141, 143, 145, 147, 149,
+    151, 153, 155, 157, 159, 161, 163, 165, 167, 169, 171, 173, 175, 177, 179,
+    181, 183, 185, 187, 189, 191, 193, 195, 197, 199, 201, 203, 205, 207, 209,
+    211, 213, 215, 217, 219, 221, 223, 225, 227, 229, 231, 233, 235, 237, 239,
+    241, 243, 245, 247, 249, 251, 253, 127};
+
+// Gets the QP, QP range: [0, 127].
+// Returns true on success, false otherwise.
+bool GetQp(const uint8_t* buf, size_t length, int* qp);
+
+}  // namespace vp8
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/video_coding.gypi b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding.gypi
index 484b82a54a..438d8f1c1f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/video_coding.gypi
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding.gypi
@@ -12,6 +12,7 @@
       'target_name': 'webrtc_video_coding',
       'type': 'static_library',
       'dependencies': [
+        'webrtc_h264',
         'webrtc_i420',
         '<(webrtc_root)/common_video/common_video.gyp:common_video',
         '<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
@@ -21,62 +22,61 @@
       ],
       'sources': [
         # interfaces
-        'main/interface/video_coding.h',
-        'main/interface/video_coding_defines.h',
+        'include/video_coding.h',
+        'include/video_coding_defines.h',
 
         # headers
-        'main/source/codec_database.h',
-        'main/source/codec_timer.h',
-        'main/source/content_metrics_processing.h',
-        'main/source/decoding_state.h',
-        'main/source/encoded_frame.h',
-        'main/source/er_tables_xor.h',
-        'main/source/fec_tables_xor.h',
-        'main/source/frame_buffer.h',
-        'main/source/generic_decoder.h',
-        'main/source/generic_encoder.h',
-        'main/source/inter_frame_delay.h',
-        'main/source/internal_defines.h',
-        'main/source/jitter_buffer.h',
-        'main/source/jitter_buffer_common.h',
-        'main/source/jitter_estimator.h',
-        'main/source/media_opt_util.h',
-        'main/source/media_optimization.h',
-        'main/source/nack_fec_tables.h',
-        'main/source/packet.h',
-        'main/source/qm_select_data.h',
-        'main/source/qm_select.h',
-        'main/source/receiver.h',
-        'main/source/rtt_filter.h',
-        'main/source/session_info.h',
-        'main/source/timestamp_map.h',
-        'main/source/timing.h',
-        'main/source/video_coding_impl.h',
+        'codec_database.h',
+        'codec_timer.h',
+        'content_metrics_processing.h',
+        'decoding_state.h',
+        'encoded_frame.h',
+        'fec_tables_xor.h',
+        'frame_buffer.h',
+        'generic_decoder.h',
+        'generic_encoder.h',
+        'inter_frame_delay.h',
+        'internal_defines.h',
+        'jitter_buffer.h',
+        'jitter_buffer_common.h',
+        'jitter_estimator.h',
+        'media_opt_util.h',
+        'media_optimization.h',
+        'nack_fec_tables.h',
+        'packet.h',
+        'qm_select_data.h',
+        'qm_select.h',
+        'receiver.h',
+        'rtt_filter.h',
+        'session_info.h',
+        'timestamp_map.h',
+        'timing.h',
+        'video_coding_impl.h',
 
         # sources
-        'main/source/codec_database.cc',
-        'main/source/codec_timer.cc',
-        'main/source/content_metrics_processing.cc',
-        'main/source/decoding_state.cc',
-        'main/source/encoded_frame.cc',
-        'main/source/frame_buffer.cc',
-        'main/source/generic_decoder.cc',
-        'main/source/generic_encoder.cc',
-        'main/source/inter_frame_delay.cc',
-        'main/source/jitter_buffer.cc',
-        'main/source/jitter_estimator.cc',
-        'main/source/media_opt_util.cc',
-        'main/source/media_optimization.cc',
-        'main/source/packet.cc',
-        'main/source/qm_select.cc',
-        'main/source/receiver.cc',
-        'main/source/rtt_filter.cc',
-        'main/source/session_info.cc',
-        'main/source/timestamp_map.cc',
-        'main/source/timing.cc',
-        'main/source/video_coding_impl.cc',
-        'main/source/video_sender.cc',
-        'main/source/video_receiver.cc',
+        'codec_database.cc',
+        'codec_timer.cc',
+        'content_metrics_processing.cc',
+        'decoding_state.cc',
+        'encoded_frame.cc',
+        'frame_buffer.cc',
+        'generic_decoder.cc',
+        'generic_encoder.cc',
+        'inter_frame_delay.cc',
+        'jitter_buffer.cc',
+        'jitter_estimator.cc',
+        'media_opt_util.cc',
+        'media_optimization.cc',
+        'packet.cc',
+        'qm_select.cc',
+        'receiver.cc',
+        'rtt_filter.cc',
+        'session_info.cc',
+        'timestamp_map.cc',
+        'timing.cc',
+        'video_coding_impl.cc',
+        'video_sender.cc',
+        'video_receiver.cc',
       ], # source
       # TODO(jschuh): Bug 1348: fix size_t to int truncations.
       'msvs_disabled_warnings': [ 4267, ],
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.cc b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.cc
new file mode 100644
index 0000000000..d50b5da804
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.cc
@@ -0,0 +1,329 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+
+#include <algorithm>
+
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace vcm {
+
+int64_t VCMProcessTimer::Period() const {
+  return _periodMs;
+}
+
+int64_t VCMProcessTimer::TimeUntilProcess() const {
+  const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
+  const int64_t time_until_process = _periodMs - time_since_process;
+  return std::max<int64_t>(time_until_process, 0);
+}
+
+void VCMProcessTimer::Processed() {
+  _latestMs = _clock->TimeInMilliseconds();
+}
+}  // namespace vcm
+
+namespace {
+// This wrapper provides a way to modify the callback without the need to expose
+// a register method all the way down to the function calling it.
+class EncodedImageCallbackWrapper : public EncodedImageCallback {
+ public:
+  EncodedImageCallbackWrapper()
+      : cs_(CriticalSectionWrapper::CreateCriticalSection()), callback_(NULL) {}
+
+  virtual ~EncodedImageCallbackWrapper() {}
+
+  void Register(EncodedImageCallback* callback) {
+    CriticalSectionScoped cs(cs_.get());
+    callback_ = callback;
+  }
+
+  // TODO(andresp): Change to void as return value is ignored.
+  virtual int32_t Encoded(const EncodedImage& encoded_image,
+                          const CodecSpecificInfo* codec_specific_info,
+                          const RTPFragmentationHeader* fragmentation) {
+    CriticalSectionScoped cs(cs_.get());
+    if (callback_)
+      return callback_->Encoded(encoded_image, codec_specific_info,
+                                fragmentation);
+    return 0;
+  }
+
+ private:
+  rtc::scoped_ptr<CriticalSectionWrapper> cs_;
+  EncodedImageCallback* callback_ GUARDED_BY(cs_);
+};
+
+class VideoCodingModuleImpl : public VideoCodingModule {
+ public:
+  VideoCodingModuleImpl(Clock* clock,
+                        EventFactory* event_factory,
+                        bool owns_event_factory,
+                        VideoEncoderRateObserver* encoder_rate_observer,
+                        VCMQMSettingsCallback* qm_settings_callback)
+      : VideoCodingModule(),
+        sender_(clock,
+                &post_encode_callback_,
+                encoder_rate_observer,
+                qm_settings_callback),
+        receiver_(clock, event_factory),
+        own_event_factory_(owns_event_factory ? event_factory : NULL) {}
+
+  virtual ~VideoCodingModuleImpl() { own_event_factory_.reset(); }
+
+  int64_t TimeUntilNextProcess() override {
+    int64_t sender_time = sender_.TimeUntilNextProcess();
+    int64_t receiver_time = receiver_.TimeUntilNextProcess();
+    assert(sender_time >= 0);
+    assert(receiver_time >= 0);
+    return VCM_MIN(sender_time, receiver_time);
+  }
+
+  int32_t Process() override {
+    int32_t sender_return = sender_.Process();
+    int32_t receiver_return = receiver_.Process();
+    if (sender_return != VCM_OK)
+      return sender_return;
+    return receiver_return;
+  }
+
+  int32_t RegisterSendCodec(const VideoCodec* sendCodec,
+                            uint32_t numberOfCores,
+                            uint32_t maxPayloadSize) override {
+    return sender_.RegisterSendCodec(sendCodec, numberOfCores, maxPayloadSize);
+  }
+
+  int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                  uint8_t payloadType,
+                                  bool internalSource) override {
+    sender_.RegisterExternalEncoder(externalEncoder, payloadType,
+                                    internalSource);
+    return 0;
+  }
+
+  int Bitrate(unsigned int* bitrate) const override {
+    return sender_.Bitrate(bitrate);
+  }
+
+  int FrameRate(unsigned int* framerate) const override {
+    return sender_.FrameRate(framerate);
+  }
+
+  int32_t SetChannelParameters(uint32_t target_bitrate,  // bits/s.
+                               uint8_t lossRate,
+                               int64_t rtt) override {
+    return sender_.SetChannelParameters(target_bitrate, lossRate, rtt);
+  }
+
+  int32_t RegisterTransportCallback(
+      VCMPacketizationCallback* transport) override {
+    return sender_.RegisterTransportCallback(transport);
+  }
+
+  int32_t RegisterSendStatisticsCallback(
+      VCMSendStatisticsCallback* sendStats) override {
+    return sender_.RegisterSendStatisticsCallback(sendStats);
+  }
+
+  int32_t RegisterProtectionCallback(
+      VCMProtectionCallback* protection) override {
+    return sender_.RegisterProtectionCallback(protection);
+  }
+
+  int32_t SetVideoProtection(VCMVideoProtection videoProtection,
+                             bool enable) override {
+    // TODO(pbos): Remove enable from receive-side protection modes as well.
+    if (enable)
+      sender_.SetVideoProtection(videoProtection);
+    return receiver_.SetVideoProtection(videoProtection, enable);
+  }
+
+  int32_t AddVideoFrame(const VideoFrame& videoFrame,
+                        const VideoContentMetrics* contentMetrics,
+                        const CodecSpecificInfo* codecSpecificInfo) override {
+    return sender_.AddVideoFrame(videoFrame, contentMetrics, codecSpecificInfo);
+  }
+
+  int32_t IntraFrameRequest(int stream_index) override {
+    return sender_.IntraFrameRequest(stream_index);
+  }
+
+  int32_t EnableFrameDropper(bool enable) override {
+    return sender_.EnableFrameDropper(enable);
+  }
+
+  void SuspendBelowMinBitrate() override {
+    return sender_.SuspendBelowMinBitrate();
+  }
+
+  bool VideoSuspended() const override { return sender_.VideoSuspended(); }
+
+  int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
+                               int32_t numberOfCores,
+                               bool requireKeyFrame) override {
+    return receiver_.RegisterReceiveCodec(receiveCodec, numberOfCores,
+                                          requireKeyFrame);
+  }
+
+  void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                               uint8_t payloadType) override {
+    receiver_.RegisterExternalDecoder(externalDecoder, payloadType);
+  }
+
+  int32_t RegisterReceiveCallback(
+      VCMReceiveCallback* receiveCallback) override {
+    return receiver_.RegisterReceiveCallback(receiveCallback);
+  }
+
+  int32_t RegisterReceiveStatisticsCallback(
+      VCMReceiveStatisticsCallback* receiveStats) override {
+    return receiver_.RegisterReceiveStatisticsCallback(receiveStats);
+  }
+
+  int32_t RegisterDecoderTimingCallback(
+      VCMDecoderTimingCallback* decoderTiming) override {
+    return receiver_.RegisterDecoderTimingCallback(decoderTiming);
+  }
+
+  int32_t RegisterFrameTypeCallback(
+      VCMFrameTypeCallback* frameTypeCallback) override {
+    return receiver_.RegisterFrameTypeCallback(frameTypeCallback);
+  }
+
+  int32_t RegisterPacketRequestCallback(
+      VCMPacketRequestCallback* callback) override {
+    return receiver_.RegisterPacketRequestCallback(callback);
+  }
+
+  virtual int32_t RegisterReceiveStateCallback(
+      VCMReceiveStateCallback* callback) override {
+    return receiver_.RegisterReceiveStateCallback(callback);
+  }
+
+  int RegisterRenderBufferSizeCallback(
+      VCMRenderBufferSizeCallback* callback) override {
+    return receiver_.RegisterRenderBufferSizeCallback(callback);
+  }
+
+  int32_t Decode(uint16_t maxWaitTimeMs) override {
+    return receiver_.Decode(maxWaitTimeMs);
+  }
+
+  int32_t ResetDecoder() override { return receiver_.ResetDecoder(); }
+
+  int32_t ReceiveCodec(VideoCodec* currentReceiveCodec) const override {
+    return receiver_.ReceiveCodec(currentReceiveCodec);
+  }
+
+  VideoCodecType ReceiveCodec() const override {
+    return receiver_.ReceiveCodec();
+  }
+
+  int32_t IncomingPacket(const uint8_t* incomingPayload,
+                         size_t payloadLength,
+                         const WebRtcRTPHeader& rtpInfo) override {
+    return receiver_.IncomingPacket(incomingPayload, payloadLength, rtpInfo);
+  }
+
+  int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) override {
+    return receiver_.SetMinimumPlayoutDelay(minPlayoutDelayMs);
+  }
+
+  int32_t SetRenderDelay(uint32_t timeMS) override {
+    return receiver_.SetRenderDelay(timeMS);
+  }
+
+  int32_t Delay() const override { return receiver_.Delay(); }
+
+  uint32_t DiscardedPackets() const override {
+    return receiver_.DiscardedPackets();
+  }
+
+  int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
+                                VCMDecodeErrorMode errorMode) override {
+    return receiver_.SetReceiverRobustnessMode(robustnessMode, errorMode);
+  }
+
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack,
+                       int max_incomplete_time_ms) override {
+    return receiver_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                                     max_incomplete_time_ms);
+  }
+
+  void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) override {
+    return receiver_.SetDecodeErrorMode(decode_error_mode);
+  }
+
+  virtual void SetCPULoadState(CPULoadState state) override {
+    return sender_.SetCPULoadState(state);
+  }
+
+  int SetMinReceiverDelay(int desired_delay_ms) override {
+    return receiver_.SetMinReceiverDelay(desired_delay_ms);
+  }
+
+  int32_t SetReceiveChannelParameters(int64_t rtt) override {
+    return receiver_.SetReceiveChannelParameters(rtt);
+  }
+
+  void RegisterPreDecodeImageCallback(EncodedImageCallback* observer) override {
+    receiver_.RegisterPreDecodeImageCallback(observer);
+  }
+
+  void RegisterPostEncodeImageCallback(
+      EncodedImageCallback* observer) override {
+    post_encode_callback_.Register(observer);
+  }
+
+  void TriggerDecoderShutdown() override { receiver_.TriggerDecoderShutdown(); }
+
+ private:
+  EncodedImageCallbackWrapper post_encode_callback_;
+  vcm::VideoSender sender_;
+  vcm::VideoReceiver receiver_;
+  rtc::scoped_ptr<EventFactory> own_event_factory_;
+};
+}  // namespace
+
+void VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec) {
+  VCMCodecDataBase::Codec(codecType, codec);
+}
+
+VideoCodingModule* VideoCodingModule::Create(
+    Clock* clock,
+    VideoEncoderRateObserver* encoder_rate_observer,
+    VCMQMSettingsCallback* qm_settings_callback) {
+  return new VideoCodingModuleImpl(clock, new EventFactoryImpl, true,
+                                   encoder_rate_observer, qm_settings_callback);
+}
+
+VideoCodingModule* VideoCodingModule::Create(Clock* clock,
+                                             EventFactory* event_factory) {
+  assert(clock);
+  assert(event_factory);
+  return new VideoCodingModuleImpl(clock, event_factory, false, nullptr,
+                                   nullptr);
+}
+
+void VideoCodingModule::Destroy(VideoCodingModule* module) {
+  if (module != NULL) {
+    delete static_cast<VideoCodingModuleImpl*>(module);
+  }
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.h b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.h
similarity index 69%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.h
rename to media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.h
index f6190b00c2..7e149e2aab 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_impl.h
@@ -11,22 +11,23 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
 #define WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
 
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
 
 #include <vector>
 
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/video_coding/main/source/codec_database.h"
-#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
-#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
-#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/media_optimization.h"
-#include "webrtc/modules/video_coding/main/source/receiver.h"
-#include "webrtc/modules/video_coding/main/source/timing.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/video_coding/codec_database.h"
+#include "webrtc/modules/video_coding/frame_buffer.h"
+#include "webrtc/modules/video_coding/generic_decoder.h"
+#include "webrtc/modules/video_coding/generic_encoder.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/media_optimization.h"
+#include "webrtc/modules/video_coding/receiver.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/modules/video_coding/utility/qp_parser.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -34,8 +35,6 @@ class EncodedFrameObserver;
 
 namespace vcm {
 
-class DebugRecorder;
-
 class VCMProcessTimer {
  public:
   VCMProcessTimer(int64_t periodMs, Clock* clock)
@@ -58,38 +57,21 @@ class VideoSender {
 
   VideoSender(Clock* clock,
               EncodedImageCallback* post_encode_callback,
-              VideoEncoderRateObserver* encoder_rate_observer);
+              VideoEncoderRateObserver* encoder_rate_observer,
+              VCMQMSettingsCallback* qm_settings_callback);
 
   ~VideoSender();
 
-  int32_t InitializeSender();
-
   // Register the send codec to be used.
   // This method must be called on the construction thread.
   int32_t RegisterSendCodec(const VideoCodec* sendCodec,
                             uint32_t numberOfCores,
                             uint32_t maxPayloadSize);
-  // Non-blocking access to the currently active send codec configuration.
-  // Must be called from the same thread as the VideoSender instance was
-  // created on.
-  const VideoCodec& GetSendCodec() const;
 
-  // Get a copy of the currently configured send codec.
-  // This method acquires a lock to copy the current configuration out,
-  // so it can block and the returned information is not guaranteed to be
-  // accurate upon return.  Consider using GetSendCodec() instead and make
-  // decisions on that thread with regards to the current codec.
-  int32_t SendCodecBlocking(VideoCodec* currentSendCodec) const;
+  void RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                               uint8_t payloadType,
+                               bool internalSource);
 
-  // Same as SendCodecBlocking.  Try to use GetSendCodec() instead.
-  VideoCodecType SendCodecBlocking() const;
-
-  int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                  uint8_t payloadType,
-                                  bool internalSource);
-
-  int32_t CodecConfigParameters(uint8_t* buffer, int32_t size) const;
-  int32_t SentFrameCount(VCMFrameCount* frameCount);
   int Bitrate(unsigned int* bitrate) const;
   int FrameRate(unsigned int* framerate) const;
 
@@ -99,11 +81,10 @@ class VideoSender {
 
   int32_t RegisterTransportCallback(VCMPacketizationCallback* transport);
   int32_t RegisterSendStatisticsCallback(VCMSendStatisticsCallback* sendStats);
-  int32_t RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings);
   int32_t RegisterProtectionCallback(VCMProtectionCallback* protection);
-  void SetVideoProtection(bool enable, VCMVideoProtection videoProtection);
+  void SetVideoProtection(VCMVideoProtection videoProtection);
 
-  int32_t AddVideoFrame(const I420VideoFrame& videoFrame,
+  int32_t AddVideoFrame(const VideoFrame& videoFrame,
                         const VideoContentMetrics* _contentMetrics,
                         const CodecSpecificInfo* codecSpecificInfo);
 
@@ -112,9 +93,6 @@ class VideoSender {
 
   void SetCPULoadState(CPULoadState state);
 
-  int StartDebugRecording(const char* file_name_utf8);
-  void StopDebugRecording();
-
   void SuspendBelowMinBitrate();
   bool VideoSuspended() const;
 
@@ -122,27 +100,31 @@ class VideoSender {
   int32_t Process();
 
  private:
-  Clock* clock_;
+  void SetEncoderParameters(EncoderParameters params)
+      EXCLUSIVE_LOCKS_REQUIRED(send_crit_);
 
-  rtc::scoped_ptr<DebugRecorder> recorder_;
+  Clock* const clock_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> process_crit_sect_;
-  CriticalSectionWrapper* _sendCritSect;
+  mutable rtc::CriticalSection send_crit_;
   VCMGenericEncoder* _encoder;
   VCMEncodedFrameCallback _encodedFrameCallback;
   std::vector<FrameType> _nextFrameTypes;
   media_optimization::MediaOptimization _mediaOpt;
-  VCMSendStatisticsCallback* _sendStatsCallback;
-  VCMCodecDataBase _codecDataBase;
-  bool frame_dropper_enabled_;
+  VCMSendStatisticsCallback* _sendStatsCallback GUARDED_BY(process_crit_sect_);
+  VCMCodecDataBase _codecDataBase GUARDED_BY(send_crit_);
+  bool frame_dropper_enabled_ GUARDED_BY(send_crit_);
   VCMProcessTimer _sendStatsTimer;
 
   // Must be accessed on the construction thread of VideoSender.
   VideoCodec current_codec_;
   rtc::ThreadChecker main_thread_;
 
-  VCMQMSettingsCallback* qm_settings_callback_;
+  VCMQMSettingsCallback* const qm_settings_callback_;
   VCMProtectionCallback* protection_callback_;
+
+  rtc::CriticalSection params_lock_;
+  EncoderParameters encoder_params_ GUARDED_BY(params_lock_);
 };
 
 class VideoReceiver {
@@ -152,15 +134,13 @@ class VideoReceiver {
   VideoReceiver(Clock* clock, EventFactory* event_factory);
   ~VideoReceiver();
 
-  int32_t InitializeReceiver();
   void SetReceiveState(VideoReceiveState state);
   int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
                                int32_t numberOfCores,
                                bool requireKeyFrame);
 
-  int32_t RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                  uint8_t payloadType,
-                                  bool internalRenderTiming);
+  void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                               uint8_t payloadType);
   int32_t RegisterReceiveCallback(VCMReceiveCallback* receiveCallback);
   int32_t RegisterReceiveStatisticsCallback(
       VCMReceiveStatisticsCallback* receiveStats);
@@ -208,17 +188,8 @@ class VideoReceiver {
       EXCLUSIVE_LOCKS_REQUIRED(_receiveCritSect);
   int32_t RequestKeyFrame();
   int32_t RequestSliceLossIndication(const uint64_t pictureID) const;
-  int32_t NackList(uint16_t* nackList, uint16_t* size);
 
  private:
-  enum VCMKeyRequestMode {
-    kKeyOnError,    // Normal mode, request key frames on decoder error
-    kKeyOnKeyLoss,  // Request key frames on decoder error and on packet loss
-                    // in key frames.
-    kKeyOnLoss,     // Request key frames on decoder error and on packet loss
-                    // in any frame
-  };
-
   Clock* const clock_;
   rtc::scoped_ptr<CriticalSectionWrapper> process_crit_sect_;
   CriticalSectionWrapper* _receiveCritSect;
@@ -242,7 +213,6 @@ class VideoReceiver {
   FILE* _bitStreamBeforeDecoder;
 #endif
   VCMFrameBuffer _frameFromFile;
-  VCMKeyRequestMode _keyRequestMode;
   bool _scheduleKeyRequest GUARDED_BY(process_crit_sect_);
   size_t max_nack_list_size_ GUARDED_BY(process_crit_sect_);
   EncodedImageCallback* pre_decode_image_callback_ GUARDED_BY(_receiveCritSect);
@@ -251,6 +221,7 @@ class VideoReceiver {
   VCMProcessTimer _receiveStatsTimer;
   VCMProcessTimer _retransmissionTimer;
   VCMProcessTimer _keyRequestTimer;
+  QpParser qp_parser_;
 };
 
 }  // namespace vcm
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
index 40a754eab1..dd6565d505 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_robustness_unittest.cc
@@ -10,11 +10,11 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -37,22 +37,17 @@ class VCMRobustnessTest : public ::testing::Test {
     ASSERT_TRUE(clock_.get() != NULL);
     vcm_ = VideoCodingModule::Create(clock_.get(), &event_factory_);
     ASSERT_TRUE(vcm_ != NULL);
-    ASSERT_EQ(0, vcm_->InitializeReceiver());
     const size_t kMaxNackListSize = 250;
     const int kMaxPacketAgeToNack = 450;
     vcm_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
     ASSERT_EQ(0, vcm_->RegisterFrameTypeCallback(&frame_type_callback_));
     ASSERT_EQ(0, vcm_->RegisterPacketRequestCallback(&request_callback_));
-    ASSERT_EQ(VCM_OK, vcm_->Codec(kVideoCodecVP8, &video_codec_));
+    VideoCodingModule::Codec(kVideoCodecVP8, &video_codec_);
     ASSERT_EQ(VCM_OK, vcm_->RegisterReceiveCodec(&video_codec_, 1));
-    ASSERT_EQ(VCM_OK, vcm_->RegisterExternalDecoder(&decoder_,
-                                                    video_codec_.plType,
-                                                    true));
+    vcm_->RegisterExternalDecoder(&decoder_, video_codec_.plType);
   }
 
-  virtual void TearDown() {
-    VideoCodingModule::Destroy(vcm_);
-  }
+  virtual void TearDown() { VideoCodingModule::Destroy(vcm_); }
 
   void InsertPacket(uint32_t timestamp,
                     uint16_t seq_no,
@@ -90,19 +85,17 @@ TEST_F(VCMRobustnessTest, TestHardNack) {
       .With(Args<0, 1>(ElementsAre(6, 7)))
       .Times(1);
   for (int ts = 0; ts <= 6000; ts += 3000) {
-    EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
-                                       Field(&EncodedImage::_length,
-                                             kPayloadLen * 3),
-                                       Field(&EncodedImage::_completeFrame,
-                                             true)),
-                                 false, _, _, _))
+    EXPECT_CALL(decoder_,
+                Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
+                             Field(&EncodedImage::_length, kPayloadLen * 3),
+                             Field(&EncodedImage::_completeFrame, true)),
+                       false, _, _, _))
         .Times(1)
         .InSequence(s);
   }
 
   ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
-      VideoCodingModule::kHardNack,
-      kNoErrors));
+                        VideoCodingModule::kHardNack, kNoErrors));
 
   InsertPacket(0, 0, true, false, kVideoFrameKey);
   InsertPacket(0, 1, false, false, kVideoFrameKey);
@@ -139,14 +132,11 @@ TEST_F(VCMRobustnessTest, TestHardNack) {
 }
 
 TEST_F(VCMRobustnessTest, TestHardNackNoneDecoded) {
-  EXPECT_CALL(request_callback_, ResendPackets(_, _))
-      .Times(0);
-  EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
-        .Times(1);
+  EXPECT_CALL(request_callback_, ResendPackets(_, _)).Times(0);
+  EXPECT_CALL(frame_type_callback_, RequestKeyFrame()).Times(1);
 
   ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
-      VideoCodingModule::kHardNack,
-      kNoErrors));
+                        VideoCodingModule::kHardNack, kNoErrors));
 
   InsertPacket(3000, 3, true, false, kVideoFrameDelta);
   InsertPacket(3000, 4, false, false, kVideoFrameDelta);
@@ -169,46 +159,43 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
       .With(Args<0, 1>(ElementsAre(4)))
       .Times(0);
 
-  EXPECT_CALL(decoder_, Copy())
-      .Times(0);
-  EXPECT_CALL(decoderCopy_, Copy())
-      .Times(0);
+  EXPECT_CALL(decoder_, Copy()).Times(0);
+  EXPECT_CALL(decoderCopy_, Copy()).Times(0);
 
   // Decode operations
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
-                                     Field(&EncodedImage::_completeFrame,
-                                           true)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
-                                     Field(&EncodedImage::_completeFrame,
-                                           false)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
-                                     Field(&EncodedImage::_completeFrame,
-                                           true)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
-  EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
-                                     Field(&EncodedImage::_completeFrame,
-                                           true)),
-                               false, _, _, _))
-        .Times(1)
-        .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
+                           Field(&EncodedImage::_completeFrame, true)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
+                           Field(&EncodedImage::_completeFrame, false)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
+                           Field(&EncodedImage::_completeFrame, true)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
+  EXPECT_CALL(decoder_,
+              Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
+                           Field(&EncodedImage::_completeFrame, true)),
+                     false, _, _, _))
+      .Times(1)
+      .InSequence(s1);
 
-  ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
-      VideoCodingModule::kNone,
-      kWithErrors));
+  ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(VideoCodingModule::kNone,
+                                                    kWithErrors));
 
   InsertPacket(0, 0, true, false, kVideoFrameKey);
   InsertPacket(0, 1, false, false, kVideoFrameKey);
   InsertPacket(0, 2, false, true, kVideoFrameKey);
-  EXPECT_EQ(VCM_OK, vcm_->Decode(0));  // Decode timestamp 0.
-  EXPECT_EQ(VCM_OK, vcm_->Process());  // Expect no NACK list.
+  EXPECT_EQ(VCM_OK, vcm_->Decode(33));  // Decode timestamp 0.
+  EXPECT_EQ(VCM_OK, vcm_->Process());   // Expect no NACK list.
 
   clock_->AdvanceTimeMilliseconds(33);
   InsertPacket(3000, 3, true, false, kVideoFrameDelta);
@@ -225,8 +212,8 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
   EXPECT_EQ(VCM_OK, vcm_->Process());  // Expect no NACK list.
 
   clock_->AdvanceTimeMilliseconds(10);
-  EXPECT_EQ(VCM_OK, vcm_->Decode(0));  // Decode timestamp 6000 complete.
-  EXPECT_EQ(VCM_OK, vcm_->Process());  // Expect no NACK list.
+  EXPECT_EQ(VCM_OK, vcm_->Decode(23));  // Decode timestamp 6000 complete.
+  EXPECT_EQ(VCM_OK, vcm_->Process());   // Expect no NACK list.
 
   clock_->AdvanceTimeMilliseconds(23);
   InsertPacket(3000, 4, false, false, kVideoFrameDelta);
@@ -234,6 +221,6 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
   InsertPacket(9000, 9, true, false, kVideoFrameDelta);
   InsertPacket(9000, 10, false, false, kVideoFrameDelta);
   InsertPacket(9000, 11, false, true, kVideoFrameDelta);
-  EXPECT_EQ(VCM_OK, vcm_->Decode(0));  // Decode timestamp 9000 complete.
+  EXPECT_EQ(VCM_OK, vcm_->Decode(33));  // Decode timestamp 9000 complete.
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_test.gypi b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_test.gypi
index 5d720ebb63..fc2fec6c98 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_test.gypi
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_coding_test.gypi
@@ -19,16 +19,16 @@
       ],
       'sources': [
         # headers
-        'main/test/receiver_tests.h',
-        'main/test/rtp_player.h',
-        'main/test/vcm_payload_sink_factory.h',
+        'test/receiver_tests.h',
+        'test/rtp_player.h',
+        'test/vcm_payload_sink_factory.h',
 
         # sources
-        'main/test/rtp_player.cc',
-        'main/test/test_util.cc',
-        'main/test/tester_main.cc',
-        'main/test/vcm_payload_sink_factory.cc',
-        'main/test/video_rtp_play.cc',
+        'test/rtp_player.cc',
+        'test/test_util.cc',
+        'test/tester_main.cc',
+        'test/vcm_payload_sink_factory.cc',
+        'test/video_rtp_play.cc',
       ], # sources
     },
   ],
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver.cc b/media/webrtc/trunk/webrtc/modules/video_coding/video_receiver.cc
similarity index 65%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/video_receiver.cc
index 251893fdab..373fa6c21f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_receiver.cc
@@ -8,16 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
-#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 // #define DEBUG_DECODER_BIT_STREAM
 
@@ -30,8 +31,8 @@ VideoReceiver::VideoReceiver(Clock* clock, EventFactory* event_factory)
       _receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
       _receiveState(kReceiveStateInitial),
       _timing(clock_),
-      _receiver(&_timing, clock_, event_factory, true),
-      _decodedFrameCallback(_timing, clock_),
+      _receiver(&_timing, clock_, event_factory),
+      _decodedFrameCallback(&_timing, clock_),
       _frameTypeCallback(NULL),
       _receiveStatsCallback(NULL),
       _decoderTimingCallback(NULL),
@@ -43,11 +44,10 @@ VideoReceiver::VideoReceiver(Clock* clock, EventFactory* event_factory)
       _bitStreamBeforeDecoder(NULL),
 #endif
       _frameFromFile(),
-      _keyRequestMode(kKeyOnError),
       _scheduleKeyRequest(false),
       max_nack_list_size_(0),
       pre_decode_image_callback_(NULL),
-      _codecDataBase(NULL),
+      _codecDataBase(nullptr, nullptr),
       _receiveStatsTimer(1000, clock_),
       _retransmissionTimer(10, clock_),
       _keyRequestTimer(500, clock_) {
@@ -86,20 +86,12 @@ int32_t VideoReceiver::Process() {
       int jitter_buffer_ms;
       int min_playout_delay_ms;
       int render_delay_ms;
-      _timing.GetTimings(&decode_ms,
-                         &max_decode_ms,
-                         &current_delay_ms,
-                         &target_delay_ms,
-                         &jitter_buffer_ms,
-                         &min_playout_delay_ms,
-                         &render_delay_ms);
-      _decoderTimingCallback->OnDecoderTiming(decode_ms,
-                                              max_decode_ms,
-                                              current_delay_ms,
-                                              target_delay_ms,
-                                              jitter_buffer_ms,
-                                              min_playout_delay_ms,
-                                              render_delay_ms);
+      _timing.GetTimings(&decode_ms, &max_decode_ms, &current_delay_ms,
+                         &target_delay_ms, &jitter_buffer_ms,
+                         &min_playout_delay_ms, &render_delay_ms);
+      _decoderTimingCallback->OnDecoderTiming(
+          decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
+          jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
     }
 
     // Size of render buffer.
@@ -138,15 +130,20 @@ int32_t VideoReceiver::Process() {
       callback_registered = _packetRequestCallback != NULL;
     }
     if (callback_registered && length > 0) {
-      std::vector<uint16_t> nackList(length);
-      const int32_t ret = NackList(&nackList[0], &length);
-      if (ret != VCM_OK && returnValue == VCM_OK) {
-        returnValue = ret;
+      // Collect sequence numbers from the default receiver.
+      bool request_key_frame = false;
+      std::vector<uint16_t> nackList = _receiver.NackList(&request_key_frame);
+      int32_t ret = VCM_OK;
+      if (request_key_frame) {
+        ret = RequestKeyFrame();
+        if (ret != VCM_OK && returnValue == VCM_OK) {
+          returnValue = ret;
+        }
       }
-      if (ret == VCM_OK && length > 0) {
+      if (ret == VCM_OK && !nackList.empty()) {
         CriticalSectionScoped cs(process_crit_sect_.get());
         if (_packetRequestCallback != NULL) {
-          _packetRequestCallback->ResendPackets(&nackList[0], length);
+          _packetRequestCallback->ResendPackets(&nackList[0], nackList.size());
         }
       }
     }
@@ -203,98 +200,30 @@ int32_t VideoReceiver::SetVideoProtection(VCMVideoProtection videoProtection,
   // By default, do not decode with errors.
   _receiver.SetDecodeErrorMode(kNoErrors);
   switch (videoProtection) {
-    case kProtectionNack:
-    case kProtectionNackReceiver: {
-      CriticalSectionScoped cs(_receiveCritSect);
-      if (enable) {
-        // Enable NACK and always wait for retransmits.
-        _receiver.SetNackMode(kNack, -1, -1);
-      } else {
-        _receiver.SetNackMode(kNoNack, -1, -1);
-      }
-      break;
-    }
-
-    case kProtectionKeyOnLoss: {
-      CriticalSectionScoped cs(_receiveCritSect);
-      if (enable) {
-        _keyRequestMode = kKeyOnLoss;
-        _receiver.SetDecodeErrorMode(kWithErrors);
-      } else if (_keyRequestMode == kKeyOnLoss) {
-        _keyRequestMode = kKeyOnError;  // default mode
-      } else {
-        return VCM_PARAMETER_ERROR;
-      }
-      break;
-    }
-
-    case kProtectionKeyOnKeyLoss: {
-      CriticalSectionScoped cs(_receiveCritSect);
-      if (enable) {
-        _keyRequestMode = kKeyOnKeyLoss;
-      } else if (_keyRequestMode == kKeyOnKeyLoss) {
-        _keyRequestMode = kKeyOnError;  // default mode
-      } else {
-        return VCM_PARAMETER_ERROR;
-      }
+    case kProtectionNack: {
+      RTC_DCHECK(enable);
+      _receiver.SetNackMode(kNack, -1, -1);
       break;
     }
 
     case kProtectionNackFEC: {
       CriticalSectionScoped cs(_receiveCritSect);
-      if (enable) {
-        // Enable hybrid NACK/FEC. Always wait for retransmissions
-        // and don't add extra delay when RTT is above
-        // kLowRttNackMs.
-        _receiver.SetNackMode(kNack, media_optimization::kLowRttNackMs, -1);
-        _receiver.SetDecodeErrorMode(kNoErrors);
-        _receiver.SetDecodeErrorMode(kNoErrors);
-      } else {
-        _receiver.SetNackMode(kNoNack, -1, -1);
-      }
+      RTC_DCHECK(enable);
+      _receiver.SetNackMode(kNack, media_optimization::kLowRttNackMs, -1);
+      _receiver.SetDecodeErrorMode(kNoErrors);
       break;
     }
-    case kProtectionNackSender:
     case kProtectionFEC:
-      // Ignore encoder modes.
-      return VCM_OK;
     case kProtectionNone:
-      // TODO(pbos): Implement like sender and remove enable parameter. Ignored
-      // for now.
+      // No receiver-side protection.
+      RTC_DCHECK(enable);
+      _receiver.SetNackMode(kNoNack, -1, -1);
+      _receiver.SetDecodeErrorMode(kWithErrors);
       break;
   }
   return VCM_OK;
 }
 
-// Initialize receiver, resets codec database etc
-int32_t VideoReceiver::InitializeReceiver() {
-  int32_t ret = _receiver.Initialize();
-  if (ret < 0) {
-    return ret;
-  }
-
-  {
-    CriticalSectionScoped receive_cs(_receiveCritSect);
-    _codecDataBase.ResetReceiver();
-    _timing.Reset();
-  }
-
-  {
-    CriticalSectionScoped process_cs(process_crit_sect_.get());
-    _decoder = NULL;
-    _decodedFrameCallback.SetUserReceiveCallback(NULL);
-    _frameTypeCallback = NULL;
-    _receiveStatsCallback = NULL;
-    _decoderTimingCallback = NULL;
-    _packetRequestCallback = NULL;
-    _receiveStateCallback = NULL;
-    _keyRequestMode = kKeyOnError;
-    _scheduleKeyRequest = false;
-  }
-
-  return VCM_OK;
-}
-
 // Register a receive callback. Will be called whenever there is a new frame
 // ready for rendering.
 int32_t VideoReceiver::RegisterReceiveCallback(
@@ -319,21 +248,17 @@ int32_t VideoReceiver::RegisterDecoderTimingCallback(
   return VCM_OK;
 }
 
-// Register an externally defined decoder/render object.
-// Can be a decoder only or a decoder coupled with a renderer.
-int32_t VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
-                                               uint8_t payloadType,
-                                               bool internalRenderTiming) {
+// Register an externally defined decoder object.
+void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                            uint8_t payloadType) {
   CriticalSectionScoped cs(_receiveCritSect);
   if (externalDecoder == NULL) {
     // Make sure the VCM updates the decoder next time it decodes.
     _decoder = NULL;
-    return _codecDataBase.DeregisterExternalDecoder(payloadType) ? 0 : -1;
+    RTC_CHECK(_codecDataBase.DeregisterExternalDecoder(payloadType));
+    return;
   }
-  return _codecDataBase.RegisterExternalDecoder(
-             externalDecoder, payloadType, internalRenderTiming)
-             ? 0
-             : -1;
+  _codecDataBase.RegisterExternalDecoder(externalDecoder, payloadType);
 }
 
 // Register a frame type request callback.
@@ -373,47 +298,46 @@ void VideoReceiver::TriggerDecoderShutdown() {
 // Should be called as often as possible to get the most out of the decoder.
 int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
   int64_t nextRenderTimeMs;
-  bool supports_render_scheduling;
+  bool prefer_late_decoding = false;
   {
     CriticalSectionScoped cs(_receiveCritSect);
-    supports_render_scheduling = _codecDataBase.SupportsRenderScheduling();
+    prefer_late_decoding = _codecDataBase.PrefersLateDecoding();
   }
 
   VCMEncodedFrame* frame = _receiver.FrameForDecoding(
-      maxWaitTimeMs, nextRenderTimeMs, supports_render_scheduling);
+      maxWaitTimeMs, &nextRenderTimeMs, prefer_late_decoding);
 
-  if (frame == NULL) {
+  if (!frame)
     return VCM_FRAME_NOT_READY;
-  } else {
-    CriticalSectionScoped cs(_receiveCritSect);
 
-    // If this frame was too late, we should adjust the delay accordingly
-    _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
-                               clock_->TimeInMilliseconds());
+  CriticalSectionScoped cs(_receiveCritSect);
 
-    if (pre_decode_image_callback_) {
-      EncodedImage encoded_image(frame->EncodedImage());
-      pre_decode_image_callback_->Encoded(encoded_image, NULL, NULL);
+  // If this frame was too late, we should adjust the delay accordingly
+  _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
+                             clock_->TimeInMilliseconds());
+
+  if (pre_decode_image_callback_) {
+    EncodedImage encoded_image(frame->EncodedImage());
+    int qp = -1;
+    if (qp_parser_.GetQp(*frame, &qp)) {
+      encoded_image.qp_ = qp;
     }
+    pre_decode_image_callback_->Encoded(encoded_image, frame->CodecSpecific(),
+                                        NULL);
+  }
 
 #ifdef DEBUG_DECODER_BIT_STREAM
-    if (_bitStreamBeforeDecoder != NULL) {
-      // Write bit stream to file for debugging purposes
-      if (fwrite(
-              frame->Buffer(), 1, frame->Length(), _bitStreamBeforeDecoder) !=
-          frame->Length()) {
-        return -1;
-      }
-    }
-#endif
-    const int32_t ret = Decode(*frame);
-    _receiver.ReleaseFrame(frame);
-    frame = NULL;
-    if (ret != VCM_OK) {
-      return ret;
+  if (_bitStreamBeforeDecoder != NULL) {
+    // Write bit stream to file for debugging purposes
+    if (fwrite(frame->Buffer(), 1, frame->Length(), _bitStreamBeforeDecoder) !=
+        frame->Length()) {
+      return -1;
     }
   }
-  return VCM_OK;
+#endif
+  const int32_t ret = Decode(*frame);
+  _receiver.ReleaseFrame(frame);
+  return ret;
 }
 
 int32_t VideoReceiver::RequestSliceLossIndication(
@@ -449,21 +373,10 @@ int32_t VideoReceiver::RequestKeyFrame() {
 
 // Must be called from inside the receive side critical section.
 int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
-  TRACE_EVENT_ASYNC_STEP1("webrtc",
-                          "Video",
-                          frame.TimeStamp(),
-                          "Decode",
-                          "type",
-                          frame.FrameType());
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame.TimeStamp(), "Decode",
+                          "type", frame.FrameType());
   // Change decoder if payload type has changed
-  const bool renderTimingBefore = _codecDataBase.SupportsRenderScheduling();
-  _decoder =
-      _codecDataBase.GetDecoder(frame.PayloadType(), &_decodedFrameCallback);
-  if (renderTimingBefore != _codecDataBase.SupportsRenderScheduling()) {
-    // Make sure we reset the decode time estimate since it will
-    // be zero for codecs without render timing.
-    _timing.ResetDecodeTime();
-  }
+  _decoder = _codecDataBase.GetDecoder(frame, &_decodedFrameCallback);
   if (_decoder == NULL) {
     return VCM_NO_CODEC_REGISTERED;
   }
@@ -484,22 +397,8 @@ int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
         _decodedFrameCallback.LastReceivedPictureID() + 1);
   }
   if (!frame.Complete() || frame.MissingFrame()) {
-    switch (_keyRequestMode) {
-      case kKeyOnKeyLoss: {
-        if (frame.FrameType() == kVideoFrameKey) {
-          request_key_frame = true;
-          ret = VCM_OK;
-        }
-        break;
-      }
-      case kKeyOnLoss: {
-        request_key_frame = true;
-        ret = VCM_OK;
-        break;
-      }
-      default:
-        break;
-    }
+    request_key_frame = true;
+    ret = VCM_OK;
   }
   if (request_key_frame) {
     CriticalSectionScoped cs(process_crit_sect_.get());
@@ -514,12 +413,12 @@ int32_t VideoReceiver::ResetDecoder() {
   bool reset_key_request = false;
   {
     CriticalSectionScoped cs(_receiveCritSect);
-    _receiver.Initialize();
+    _receiver.Reset();
     _timing.Reset();
     reset_key_request = true;
     if (_decoder != NULL) {
-      //      _receiver.Initialize();
-      //      _timing.Reset();
+      //    _receiver.Reset();
+      //    _timing.Reset();
       _decoder->Reset();
     }
   }
@@ -538,8 +437,8 @@ int32_t VideoReceiver::RegisterReceiveCodec(const VideoCodec* receiveCodec,
   if (receiveCodec == NULL) {
     return VCM_PARAMETER_ERROR;
   }
-  if (!_codecDataBase.RegisterReceiveCodec(
-          receiveCodec, numberOfCores, requireKeyFrame)) {
+  if (!_codecDataBase.RegisterReceiveCodec(receiveCodec, numberOfCores,
+                                           requireKeyFrame)) {
     return -1;
   }
   return 0;
@@ -565,9 +464,7 @@ int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
                                       size_t payloadLength,
                                       const WebRtcRTPHeader& rtpInfo) {
   if (rtpInfo.frameType == kVideoFrameKey) {
-    TRACE_EVENT1("webrtc",
-                 "VCM::PacketKeyFrame",
-                 "seqnum",
+    TRACE_EVENT1("webrtc", "VCM::PacketKeyFrame", "seqnum",
                  rtpInfo.header.sequenceNumber);
   }
   if (incomingPayload == NULL) {
@@ -607,28 +504,8 @@ int32_t VideoReceiver::SetRenderDelay(uint32_t timeMS) {
 }
 
 // Current video delay
-int32_t VideoReceiver::Delay() const { return _timing.TargetVideoDelay(); }
-
-// Nack list
-int32_t VideoReceiver::NackList(uint16_t* nackList, uint16_t* size) {
-  VCMNackStatus nackStatus = kNackOk;
-  uint16_t nack_list_length = 0;
-  // Collect sequence numbers from the default receiver
-  // if in normal nack mode.
-  if (_receiver.NackMode() != kNoNack) {
-    nackStatus = _receiver.NackList(nackList, *size, &nack_list_length);
-  }
-  *size = nack_list_length;
-  if (nackStatus == kNackKeyFrameRequest) {
-      SetReceiveState(kReceiveStateWaitingKey);
-      return RequestKeyFrame();
-  }
-  if (*size != 0) {
-    // Note: not a valid transition from WaitingKey or DecodingWithErrors;
-    // will be ignored in that case
-    SetReceiveState(kReceiveStatePreemptiveNACK);
-  }
-  return VCM_OK;
+int32_t VideoReceiver::Delay() const {
+  return _timing.TargetVideoDelay();
 }
 
 uint32_t VideoReceiver::DiscardedPackets() const {
@@ -642,16 +519,10 @@ int VideoReceiver::SetReceiverRobustnessMode(
   switch (robustnessMode) {
     case VideoCodingModule::kNone:
       _receiver.SetNackMode(kNoNack, -1, -1);
-      if (decode_error_mode == kNoErrors) {
-        _keyRequestMode = kKeyOnLoss;
-      } else {
-        _keyRequestMode = kKeyOnError;
-      }
       break;
     case VideoCodingModule::kHardNack:
       // Always wait for retransmissions (except when decoding with errors).
       _receiver.SetNackMode(kNack, -1, -1);
-      _keyRequestMode = kKeyOnError;  // TODO(hlundin): On long NACK list?
       break;
     case VideoCodingModule::kSoftNack:
 #if 1
@@ -661,7 +532,6 @@ int VideoReceiver::SetReceiverRobustnessMode(
       // Enable hybrid NACK/FEC. Always wait for retransmissions and don't add
       // extra delay when RTT is above kLowRttNackMs.
       _receiver.SetNackMode(kNack, media_optimization::kLowRttNackMs, -1);
-      _keyRequestMode = kKeyOnError;
       break;
 #endif
     case VideoCodingModule::kReferenceSelection:
@@ -692,8 +562,8 @@ void VideoReceiver::SetNackSettings(size_t max_nack_list_size,
     CriticalSectionScoped process_cs(process_crit_sect_.get());
     max_nack_list_size_ = max_nack_list_size;
   }
-  _receiver.SetNackSettings(
-      max_nack_list_size, max_packet_age_to_nack, max_incomplete_time_ms);
+  _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                            max_incomplete_time_ms);
 }
 
 int VideoReceiver::SetMinReceiverDelay(int desired_delay_ms) {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/video_receiver_unittest.cc
similarity index 87%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/video_receiver_unittest.cc
index 209a45c4ef..820ce9ae2d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_receiver_unittest.cc
@@ -12,12 +12,12 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 using ::testing::_;
 using ::testing::NiceMock;
@@ -34,16 +34,12 @@ class TestVideoReceiver : public ::testing::Test {
 
   virtual void SetUp() {
     receiver_.reset(new VideoReceiver(&clock_, &event_factory_));
-    EXPECT_EQ(0, receiver_->InitializeReceiver());
-    EXPECT_EQ(0,
-              receiver_->RegisterExternalDecoder(
-                  &decoder_, kUnusedPayloadType, true));
+    receiver_->RegisterExternalDecoder(&decoder_, kUnusedPayloadType);
     const size_t kMaxNackListSize = 250;
     const int kMaxPacketAgeToNack = 450;
     receiver_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
 
-    memset(&settings_, 0, sizeof(settings_));
-    EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &settings_));
+    VideoCodingModule::Codec(kVideoCodecVP8, &settings_);
     settings_.plType = kUnusedPayloadType;  // Use the mocked encoder.
     EXPECT_EQ(0, receiver_->RegisterReceiveCodec(&settings_, 1, true));
   }
@@ -58,7 +54,7 @@ class TestVideoReceiver : public ::testing::Test {
     }
     EXPECT_EQ(0, receiver_->Process());
     EXPECT_CALL(decoder_, Decode(_, _, _, _, _)).Times(0);
-    EXPECT_EQ(VCM_FRAME_NOT_READY, receiver_->Decode(0));
+    EXPECT_EQ(VCM_FRAME_NOT_READY, receiver_->Decode(100));
   }
 
   void InsertAndVerifyDecodableFrame(const uint8_t* payload,
@@ -70,7 +66,7 @@ class TestVideoReceiver : public ::testing::Test {
     EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
     EXPECT_EQ(0, receiver_->Process());
     EXPECT_CALL(decoder_, Decode(_, _, _, _, _)).Times(1);
-    EXPECT_EQ(0, receiver_->Decode(0));
+    EXPECT_EQ(0, receiver_->Decode(100));
   }
 
   SimulatedClock clock_;
@@ -90,7 +86,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyFrames) {
   const uint8_t payload[kPaddingSize] = {0};
   WebRtcRTPHeader header;
   memset(&header, 0, sizeof(header));
-  header.frameType = kFrameEmpty;
+  header.frameType = kEmptyFrame;
   header.header.markerBit = false;
   header.header.paddingLength = kPaddingSize;
   header.header.payloadType = kUnusedPayloadType;
@@ -114,7 +110,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyFramesWithLosses) {
   const uint8_t payload[kFrameSize] = {0};
   WebRtcRTPHeader header;
   memset(&header, 0, sizeof(header));
-  header.frameType = kFrameEmpty;
+  header.frameType = kEmptyFrame;
   header.header.markerBit = false;
   header.header.paddingLength = kPaddingSize;
   header.header.payloadType = kUnusedPayloadType;
@@ -129,7 +125,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyFramesWithLosses) {
   clock_.AdvanceTimeMilliseconds(33);
   header.header.timestamp += 3000;
 
-  header.frameType = kFrameEmpty;
+  header.frameType = kEmptyFrame;
   header.type.Video.isFirstPacket = false;
   header.header.markerBit = false;
   // Insert padding frames.
@@ -165,7 +161,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
   const uint8_t payload[kFrameSize] = {0};
   WebRtcRTPHeader header;
   memset(&header, 0, sizeof(header));
-  header.frameType = kFrameEmpty;
+  header.frameType = kEmptyFrame;
   header.type.Video.isFirstPacket = false;
   header.header.markerBit = false;
   header.header.paddingLength = kPaddingSize;
@@ -190,7 +186,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
     }
 
     // Insert 2 padding only frames.
-    header.frameType = kFrameEmpty;
+    header.frameType = kEmptyFrame;
     header.type.Video.isFirstPacket = false;
     header.header.markerBit = false;
     for (int j = 0; j < 2; ++j) {
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender.cc b/media/webrtc/trunk/webrtc/modules/video_coding/video_sender.cc
similarity index 50%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/video_sender.cc
index 3b7463cc33..75a8d63844 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_sender.cc
@@ -8,84 +8,50 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/common_types.h"
 
 #include <algorithm>  // std::max
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/utility/quality_scaler.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace vcm {
 
-class DebugRecorder {
- public:
-  DebugRecorder()
-      : cs_(CriticalSectionWrapper::CreateCriticalSection()), file_(NULL) {}
-
-  ~DebugRecorder() { Stop(); }
-
-  int Start(const char* file_name_utf8) {
-    CriticalSectionScoped cs(cs_.get());
-    if (file_)
-      fclose(file_);
-    file_ = fopen(file_name_utf8, "wb");
-    if (!file_)
-      return VCM_GENERAL_ERROR;
-    return VCM_OK;
-  }
-
-  void Stop() {
-    CriticalSectionScoped cs(cs_.get());
-    if (file_) {
-      fclose(file_);
-      file_ = NULL;
-    }
-  }
-
-  void Add(const I420VideoFrame& frame) {
-    CriticalSectionScoped cs(cs_.get());
-    if (file_)
-      PrintI420VideoFrame(frame, file_);
-  }
-
- private:
-  rtc::scoped_ptr<CriticalSectionWrapper> cs_;
-  FILE* file_ GUARDED_BY(cs_);
-};
-
 VideoSender::VideoSender(Clock* clock,
                          EncodedImageCallback* post_encode_callback,
-                         VideoEncoderRateObserver* encoder_rate_observer)
+                         VideoEncoderRateObserver* encoder_rate_observer,
+                         VCMQMSettingsCallback* qm_settings_callback)
     : clock_(clock),
-      recorder_(new DebugRecorder()),
       process_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      _sendCritSect(CriticalSectionWrapper::CreateCriticalSection()),
-      _encoder(),
+      _encoder(nullptr),
       _encodedFrameCallback(post_encode_callback),
       _nextFrameTypes(1, kVideoFrameDelta),
       _mediaOpt(clock_),
-      _sendStatsCallback(NULL),
-      _codecDataBase(encoder_rate_observer),
+      _sendStatsCallback(nullptr),
+      _codecDataBase(encoder_rate_observer, &_encodedFrameCallback),
       frame_dropper_enabled_(true),
       _sendStatsTimer(1000, clock_),
       current_codec_(),
-      qm_settings_callback_(NULL),
-      protection_callback_(NULL) {
+      qm_settings_callback_(qm_settings_callback),
+      protection_callback_(nullptr),
+      encoder_params_({0, 0, 0, 0}) {
+  _encodedFrameCallback.SetCritSect(process_crit_sect_.get());
   // Allow VideoSender to be created on one thread but used on another, post
   // construction. This is currently how this class is being used by at least
   // one external project (diffractor).
+  _mediaOpt.EnableQM(qm_settings_callback_ != nullptr);
+  _mediaOpt.Reset();
   main_thread_.DetachFromThread();
 }
 
-VideoSender::~VideoSender() {
-  delete _sendCritSect;
-}
+VideoSender::~VideoSender() {}
 
 int32_t VideoSender::Process() {
   int32_t returnValue = VCM_OK;
@@ -93,26 +59,21 @@ int32_t VideoSender::Process() {
   if (_sendStatsTimer.TimeUntilProcess() == 0) {
     _sendStatsTimer.Processed();
     CriticalSectionScoped cs(process_crit_sect_.get());
-    if (_sendStatsCallback != NULL) {
+    if (_sendStatsCallback != nullptr) {
       uint32_t bitRate = _mediaOpt.SentBitRate();
       uint32_t frameRate = _mediaOpt.SentFrameRate();
       _sendStatsCallback->SendStatistics(bitRate, frameRate);
     }
   }
 
-  return returnValue;
-}
+  {
+    rtc::CritScope cs(&params_lock_);
+    // Force an encoder parameters update, so that incoming frame rate is
+    // updated even if bandwidth hasn't changed.
+    encoder_params_.input_frame_rate = _mediaOpt.InputFrameRate();
+  }
 
-// Reset send side to initial state - all components
-int32_t VideoSender::InitializeSender() {
-  DCHECK(main_thread_.CalledOnValidThread());
-  CriticalSectionScoped cs(_sendCritSect);
-  _codecDataBase.ResetSender();
-  _encoder = NULL;
-  _encodedFrameCallback.SetTransportCallback(NULL);
-  _encodedFrameCallback.SetCritSect(_sendCritSect);
-  _mediaOpt.Reset();  // Resetting frame dropper
-  return VCM_OK;
+  return returnValue;
 }
 
 int64_t VideoSender::TimeUntilNextProcess() {
@@ -123,14 +84,14 @@ int64_t VideoSender::TimeUntilNextProcess() {
 int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
                                        uint32_t numberOfCores,
                                        uint32_t maxPayloadSize) {
-  DCHECK(main_thread_.CalledOnValidThread());
-  CriticalSectionScoped cs(_sendCritSect);
-  if (sendCodec == NULL) {
+  RTC_DCHECK(main_thread_.CalledOnValidThread());
+  rtc::CritScope lock(&send_crit_);
+  if (sendCodec == nullptr) {
     return VCM_PARAMETER_ERROR;
   }
 
-  bool ret = _codecDataBase.SetSendCodec(
-      sendCodec, numberOfCores, maxPayloadSize, &_encodedFrameCallback);
+  bool ret =
+      _codecDataBase.SetSendCodec(sendCodec, numberOfCores, maxPayloadSize);
 
   // Update encoder regardless of result to make sure that we're not holding on
   // to a deleted instance.
@@ -145,10 +106,15 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
     return VCM_CODEC_ERROR;
   }
 
-  // XXX fix VP9 (bug 1138629)
-  int numLayers = (sendCodec->codecType != kVideoCodecVP8)
-                      ? 1
-                      : sendCodec->codecSpecific.VP8.numberOfTemporalLayers;
+  int numLayers;
+  if (sendCodec->codecType == kVideoCodecVP8) {
+    numLayers = sendCodec->codecSpecific.VP8.numberOfTemporalLayers;
+  } else if (sendCodec->codecType == kVideoCodecVP9) {
+    numLayers = sendCodec->codecSpecific.VP9.numberOfTemporalLayers;
+  } else {
+    numLayers = 1;
+  }
+
   // If we have screensharing and we have layers, we disable frame dropper.
   bool disable_frame_dropper =
       numLayers > 1 && sendCodec->mode == kScreensharing;
@@ -161,137 +127,92 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
   _nextFrameTypes.resize(VCM_MAX(sendCodec->numberOfSimulcastStreams, 1),
                          kVideoFrameDelta);
 
-  _mediaOpt.SetEncodingData(sendCodec->codecType,
-                            sendCodec->maxBitrate * 1000,
-                            sendCodec->startBitrate * 1000,
-                            sendCodec->width,
-                            sendCodec->height,
-                            sendCodec->maxFramerate * 1000,
+  _mediaOpt.SetEncodingData(sendCodec->codecType, sendCodec->maxBitrate * 1000,
+                            sendCodec->startBitrate * 1000, sendCodec->width,
+                            sendCodec->height, sendCodec->maxFramerate * 1000,
                             sendCodec->resolution_divisor,
-                            numLayers,
-                            maxPayloadSize);
+                            numLayers, maxPayloadSize);
   return VCM_OK;
 }
 
-const VideoCodec& VideoSender::GetSendCodec() const {
-  DCHECK(main_thread_.CalledOnValidThread());
-  return current_codec_;
-}
-
-int32_t VideoSender::SendCodecBlocking(VideoCodec* currentSendCodec) const {
-  CriticalSectionScoped cs(_sendCritSect);
-  if (currentSendCodec == NULL) {
-    return VCM_PARAMETER_ERROR;
-  }
-  return _codecDataBase.SendCodec(currentSendCodec) ? 0 : -1;
-}
-
-VideoCodecType VideoSender::SendCodecBlocking() const {
-  CriticalSectionScoped cs(_sendCritSect);
-  return _codecDataBase.SendCodec();
-}
-
 // Register an external decoder object.
 // This can not be used together with external decoder callbacks.
-int32_t VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
-                                             uint8_t payloadType,
-                                             bool internalSource /*= false*/) {
-  DCHECK(main_thread_.CalledOnValidThread());
+void VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
+                                          uint8_t payloadType,
+                                          bool internalSource /*= false*/) {
+  RTC_DCHECK(main_thread_.CalledOnValidThread());
 
-  CriticalSectionScoped cs(_sendCritSect);
+  rtc::CritScope lock(&send_crit_);
 
-  if (externalEncoder == NULL) {
+  if (externalEncoder == nullptr) {
     bool wasSendCodec = false;
-    const bool ret =
-        _codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec);
+    RTC_CHECK(
+        _codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec));
     if (wasSendCodec) {
       // Make sure the VCM doesn't use the de-registered codec
-      _encoder = NULL;
+      _encoder = nullptr;
     }
-    return ret ? 0 : -1;
+    return;
   }
-  _codecDataBase.RegisterExternalEncoder(
-      externalEncoder, payloadType, internalSource);
-  return 0;
-}
-
-// Get codec config parameters
-int32_t VideoSender::CodecConfigParameters(uint8_t* buffer,
-                                           int32_t size) const {
-  CriticalSectionScoped cs(_sendCritSect);
-  if (_encoder != NULL) {
-    return _encoder->CodecConfigParameters(buffer, size);
-  }
-  return VCM_UNINITIALIZED;
-}
-
-// TODO(andresp): Make const once media_opt is thread-safe and this has a
-// pointer to it.
-int32_t VideoSender::SentFrameCount(VCMFrameCount* frameCount) {
-  *frameCount = _mediaOpt.SentFrameCount();
-  return VCM_OK;
+  _codecDataBase.RegisterExternalEncoder(externalEncoder, payloadType,
+                                         internalSource);
 }
 
 // Get encode bitrate
 int VideoSender::Bitrate(unsigned int* bitrate) const {
-  DCHECK(main_thread_.CalledOnValidThread());
+  RTC_DCHECK(main_thread_.CalledOnValidThread());
   // Since we're running on the thread that's the only thread known to modify
   // the value of _encoder, we don't need to grab the lock here.
 
-  // return the bit rate which the encoder is set to
-  if (!_encoder) {
+  if (!_encoder)
     return VCM_UNINITIALIZED;
-  }
-  *bitrate = _encoder->BitRate();
+  *bitrate = _encoder->GetEncoderParameters().target_bitrate;
   return 0;
 }
 
 // Get encode frame rate
 int VideoSender::FrameRate(unsigned int* framerate) const {
-  DCHECK(main_thread_.CalledOnValidThread());
+  RTC_DCHECK(main_thread_.CalledOnValidThread());
   // Since we're running on the thread that's the only thread known to modify
   // the value of _encoder, we don't need to grab the lock here.
 
-  // input frame rate, not compensated
-  if (!_encoder) {
+  if (!_encoder)
     return VCM_UNINITIALIZED;
-  }
-  *framerate = _encoder->FrameRate();
+
+  *framerate = _encoder->GetEncoderParameters().input_frame_rate;
   return 0;
 }
 
 int32_t VideoSender::SetChannelParameters(uint32_t target_bitrate,
                                           uint8_t lossRate,
                                           int64_t rtt) {
-  // TODO(tommi,mflodman): This method is called on the network thread via the
-  // OnNetworkChanged event (ViEEncoder::OnNetworkChanged). Could we instead
-  // post the updated information to the encoding thread and not grab a lock
-  // here?  This effectively means that the network thread will be blocked for
-  // as much as frame encoding period.
+  uint32_t target_rate =
+      _mediaOpt.SetTargetRates(target_bitrate, lossRate, rtt,
+                               protection_callback_, qm_settings_callback_);
 
-  CriticalSectionScoped sendCs(_sendCritSect);
-  uint32_t target_rate = _mediaOpt.SetTargetRates(target_bitrate,
-                                                  lossRate,
-                                                  rtt,
-                                                  protection_callback_,
-                                                  qm_settings_callback_);
   uint32_t input_frame_rate = _mediaOpt.InputFrameRate();
 
-  int32_t ret = VCM_UNINITIALIZED;
-  static_assert(VCM_UNINITIALIZED < 0, "VCM_UNINITIALIZED must be negative.");
+  rtc::CritScope cs(&params_lock_);
+  encoder_params_ = {target_rate, lossRate, rtt, input_frame_rate};
 
-  if (_encoder != NULL) {
-    ret = _encoder->SetChannelParameters(lossRate, rtt);
-    if (ret >= 0) {
-      ret = _encoder->SetRates(target_rate, input_frame_rate);
-    }
+  return VCM_OK;
+}
+
+void VideoSender::SetEncoderParameters(EncoderParameters params) {
+  if (params.target_bitrate == 0)
+    return;
+
+  if (params.input_frame_rate == 0) {
+    // No frame rate estimate available, use default.
+    params.input_frame_rate = current_codec_.maxFramerate;
   }
-  return ret;
+  if (_encoder != nullptr)
+    _encoder->SetEncoderParameters(params);
 }
 
 int32_t VideoSender::RegisterTransportCallback(
     VCMPacketizationCallback* transport) {
-  CriticalSectionScoped cs(_sendCritSect);
+  rtc::CritScope lock(&send_crit_);
   _encodedFrameCallback.SetMediaOpt(&_mediaOpt);
   _encodedFrameCallback.SetTransportCallback(transport);
   return VCM_OK;
@@ -307,76 +228,60 @@ int32_t VideoSender::RegisterSendStatisticsCallback(
   return VCM_OK;
 }
 
-// Register a video quality settings callback which will be called when frame
-// rate/dimensions need to be updated for video quality optimization
-int32_t VideoSender::RegisterVideoQMCallback(
-    VCMQMSettingsCallback* qm_settings_callback) {
-  CriticalSectionScoped cs(_sendCritSect);
-  DCHECK(qm_settings_callback_ == qm_settings_callback ||
-         !qm_settings_callback_ ||
-         !qm_settings_callback) << "Overwriting the previous callback?";
-  qm_settings_callback_ = qm_settings_callback;
-  _mediaOpt.EnableQM(qm_settings_callback_ != NULL);
-  return VCM_OK;
-}
-
 // Register a video protection callback which will be called to deliver the
 // requested FEC rate and NACK status (on/off).
+// Note: this callback is assumed to only be registered once and before it is
+// used in this class.
 int32_t VideoSender::RegisterProtectionCallback(
     VCMProtectionCallback* protection_callback) {
-  CriticalSectionScoped cs(_sendCritSect);
-  DCHECK(protection_callback_ == protection_callback ||
-         !protection_callback_ ||
-         !protection_callback) << "Overwriting the previous callback?";
+  RTC_DCHECK(protection_callback == nullptr || protection_callback_ == nullptr);
   protection_callback_ = protection_callback;
   return VCM_OK;
 }
 
 // Enable or disable a video protection method.
-void VideoSender::SetVideoProtection(bool enable,
-                                     VCMVideoProtection videoProtection) {
-  CriticalSectionScoped cs(_sendCritSect);
+void VideoSender::SetVideoProtection(VCMVideoProtection videoProtection) {
+  rtc::CritScope lock(&send_crit_);
   switch (videoProtection) {
     case kProtectionNone:
-      _mediaOpt.EnableProtectionMethod(enable, media_optimization::kNone);
+      _mediaOpt.SetProtectionMethod(media_optimization::kNone);
       break;
     case kProtectionNack:
-    case kProtectionNackSender:
-      _mediaOpt.EnableProtectionMethod(enable, media_optimization::kNack);
+      _mediaOpt.SetProtectionMethod(media_optimization::kNack);
       break;
     case kProtectionNackFEC:
-      _mediaOpt.EnableProtectionMethod(enable, media_optimization::kNackFec);
+      _mediaOpt.SetProtectionMethod(media_optimization::kNackFec);
       break;
     case kProtectionFEC:
-      _mediaOpt.EnableProtectionMethod(enable, media_optimization::kFec);
+      _mediaOpt.SetProtectionMethod(media_optimization::kFec);
       break;
-    case kProtectionNackReceiver:
-    case kProtectionKeyOnLoss:
-    case kProtectionKeyOnKeyLoss:
-      // Ignore receiver modes.
-      return;
   }
 }
 // Add one raw video frame to the encoder, blocking.
-int32_t VideoSender::AddVideoFrame(const I420VideoFrame& videoFrame,
+int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
                                    const VideoContentMetrics* contentMetrics,
                                    const CodecSpecificInfo* codecSpecificInfo) {
-  CriticalSectionScoped cs(_sendCritSect);
-  if (_encoder == NULL) {
-    return VCM_UNINITIALIZED;
+  EncoderParameters encoder_params;
+  {
+    rtc::CritScope lock(&params_lock_);
+    encoder_params = encoder_params_;
   }
+  rtc::CritScope lock(&send_crit_);
+  if (_encoder == nullptr)
+    return VCM_UNINITIALIZED;
+  SetEncoderParameters(encoder_params);
   // TODO(holmer): Add support for dropping frames per stream. Currently we
   // only have one frame dropper for all streams.
-  if (_nextFrameTypes[0] == kFrameEmpty) {
+  if (_nextFrameTypes[0] == kEmptyFrame) {
     return VCM_OK;
   }
   if (_mediaOpt.DropFrame()) {
+    _encoder->OnDroppedFrame();
     return VCM_OK;
   }
-
   _mediaOpt.UpdateContentData(contentMetrics);
-#ifdef VERIFY_FRAME_SIZE_VS_DATABASE
   // TODO(pbos): Make sure setting send codec is synchronized with video
+#ifdef VERIFY_FRAME_SIZE_VS_DATABASE
   // processing so frame size always matches.
   if (!_codecDataBase.MatchesCurrentResolution(videoFrame.width(),
                                                videoFrame.height())) {
@@ -384,10 +289,16 @@ int32_t VideoSender::AddVideoFrame(const I420VideoFrame& videoFrame,
     return VCM_PARAMETER_ERROR;
   }
 #endif
+  VideoFrame converted_frame = videoFrame;
+  if (converted_frame.native_handle() && !_encoder->SupportsNativeHandle()) {
+    // This module only supports software encoding.
+    // TODO(pbos): Offload conversion from the encoder thread.
+    converted_frame = converted_frame.ConvertNativeToI420Frame();
+    RTC_CHECK(!converted_frame.IsZeroSize())
+        << "Frame conversion failed, won't be able to encode frame.";
+  }
   int32_t ret =
-      _encoder->Encode(videoFrame, codecSpecificInfo, _nextFrameTypes);
-  recorder_->Add(videoFrame);
-
+      _encoder->Encode(converted_frame, codecSpecificInfo, _nextFrameTypes);
   if (ret < 0) {
     LOG(LS_ERROR) << "Failed to encode frame. Error code: " << ret;
     return ret;
@@ -395,17 +306,19 @@ int32_t VideoSender::AddVideoFrame(const I420VideoFrame& videoFrame,
   for (size_t i = 0; i < _nextFrameTypes.size(); ++i) {
     _nextFrameTypes[i] = kVideoFrameDelta;  // Default frame type.
   }
+  if (qm_settings_callback_)
+    qm_settings_callback_->SetTargetFramerate(_encoder->GetTargetFramerate());
   return VCM_OK;
 }
 
 int32_t VideoSender::IntraFrameRequest(int stream_index) {
-  CriticalSectionScoped cs(_sendCritSect);
+  rtc::CritScope lock(&send_crit_);
   if (stream_index < 0 ||
       static_cast<unsigned int>(stream_index) >= _nextFrameTypes.size()) {
     return -1;
   }
   _nextFrameTypes[stream_index] = kVideoFrameKey;
-  if (_encoder != NULL && _encoder->InternalSource()) {
+  if (_encoder != nullptr && _encoder->InternalSource()) {
     // Try to request the frame if we have an external encoder with
     // internal source since AddVideoFrame never will be called.
     if (_encoder->RequestFrame(_nextFrameTypes) == WEBRTC_VIDEO_CODEC_OK) {
@@ -416,22 +329,14 @@ int32_t VideoSender::IntraFrameRequest(int stream_index) {
 }
 
 int32_t VideoSender::EnableFrameDropper(bool enable) {
-  CriticalSectionScoped cs(_sendCritSect);
+  rtc::CritScope lock(&send_crit_);
   frame_dropper_enabled_ = enable;
   _mediaOpt.EnableFrameDropper(enable);
   return VCM_OK;
 }
 
-int VideoSender::StartDebugRecording(const char* file_name_utf8) {
-  return recorder_->Start(file_name_utf8);
-}
-
-void VideoSender::StopDebugRecording() {
-  recorder_->Stop();
-}
-
 void VideoSender::SuspendBelowMinBitrate() {
-  DCHECK(main_thread_.CalledOnValidThread());
+  RTC_DCHECK(main_thread_.CalledOnValidThread());
   int threshold_bps;
   if (current_codec_.numberOfSimulcastStreams == 0) {
     threshold_bps = current_codec_.minBitrate * 1000;
@@ -445,12 +350,11 @@ void VideoSender::SuspendBelowMinBitrate() {
 }
 
 bool VideoSender::VideoSuspended() const {
-  CriticalSectionScoped cs(_sendCritSect);
   return _mediaOpt.IsVideoSuspended();
 }
 
 void VideoSender::SetCPULoadState(CPULoadState state) {
-  CriticalSectionScoped cs(_sendCritSect);
+  rtc::CritScope lock(&send_crit_);
   _mediaOpt.SetCPULoadState(state);
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_coding/video_sender_unittest.cc
similarity index 72%
rename from media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_coding/video_sender_unittest.cc
index f28b9dfba4..741c7b7a60 100644
--- a/media/webrtc/trunk/webrtc/modules/video_coding/main/source/video_sender_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_coding/video_sender_unittest.cc
@@ -13,17 +13,17 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
-#include "webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
-#include "webrtc/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/source/video_coding_impl.h"
-#include "webrtc/modules/video_coding/main/test/test_util.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/modules/video_coding/test/test_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 using ::testing::_;
 using ::testing::AllOf;
@@ -40,9 +40,7 @@ using webrtc::test::FrameGenerator;
 namespace webrtc {
 namespace vcm {
 namespace {
-enum {
-  kMaxNumberOfTemporalLayers = 3
-};
+enum { kMaxNumberOfTemporalLayers = 3 };
 
 struct Vp8StreamInfo {
   float framerate_fps[kMaxNumberOfTemporalLayers];
@@ -71,8 +69,8 @@ MATCHER_P(MatchesVp8StreamInfo, expected, "") {
 class EmptyFrameGenerator : public FrameGenerator {
  public:
   EmptyFrameGenerator(int width, int height) : width_(width), height_(height) {}
-  I420VideoFrame* NextFrame() override {
-    frame_.reset(new I420VideoFrame());
+  VideoFrame* NextFrame() override {
+    frame_.reset(new VideoFrame());
     frame_->CreateEmptyFrame(width_, height_, width_, (width_ + 1) / 2,
                              (width_ + 1) / 2);
     return frame_.get();
@@ -81,12 +79,12 @@ class EmptyFrameGenerator : public FrameGenerator {
  private:
   const int width_;
   const int height_;
-  rtc::scoped_ptr<I420VideoFrame> frame_;
+  rtc::scoped_ptr<VideoFrame> frame_;
 };
 
 class PacketizationCallback : public VCMPacketizationCallback {
  public:
-  PacketizationCallback(Clock* clock)
+  explicit PacketizationCallback(Clock* clock)
       : clock_(clock), start_time_ms_(clock_->TimeInMilliseconds()) {}
 
   virtual ~PacketizationCallback() {}
@@ -177,8 +175,8 @@ class TestVideoSender : public ::testing::Test {
   TestVideoSender() : clock_(1000), packetization_callback_(&clock_) {}
 
   void SetUp() override {
-    sender_.reset(new VideoSender(&clock_, &post_encode_callback_, nullptr));
-    EXPECT_EQ(0, sender_->InitializeSender());
+    sender_.reset(
+        new VideoSender(&clock_, &post_encode_callback_, nullptr, nullptr));
     EXPECT_EQ(0, sender_->RegisterTransportCallback(&packetization_callback_));
   }
 
@@ -190,6 +188,8 @@ class TestVideoSender : public ::testing::Test {
   SimulatedClock clock_;
   PacketizationCallback packetization_callback_;
   MockEncodedImageCallback post_encode_callback_;
+  // Used by subclassing tests, need to outlive sender_.
+  rtc::scoped_ptr<VideoEncoder> encoder_;
   rtc::scoped_ptr<VideoSender> sender_;
   rtc::scoped_ptr<FrameGenerator> generator_;
 };
@@ -204,22 +204,15 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
 
   void SetUp() override {
     TestVideoSender::SetUp();
-    EXPECT_EQ(
-        0,
-        sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, false));
-    memset(&settings_, 0, sizeof(settings_));
-    EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &settings_));
+    sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, false);
+    VideoCodingModule::Codec(kVideoCodecVP8, &settings_);
     settings_.numberOfSimulcastStreams = kNumberOfStreams;
-    ConfigureStream(kDefaultWidth / 4,
-                    kDefaultHeight / 4,
-                    100,
+    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, 100,
                     &settings_.simulcastStream[0]);
-    ConfigureStream(kDefaultWidth / 2,
-                    kDefaultHeight / 2,
-                    500,
+    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, 500,
                     &settings_.simulcastStream[1]);
-    ConfigureStream(
-        kDefaultWidth, kDefaultHeight, 1200, &settings_.simulcastStream[2]);
+    ConfigureStream(kDefaultWidth, kDefaultHeight, 1200,
+                    &settings_.simulcastStream[2]);
     settings_.plType = kUnusedPayloadType;  // Use the mocked encoder.
     generator_.reset(
         new EmptyFrameGenerator(settings_.width, settings_.height));
@@ -233,22 +226,21 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
       // No intra request expected.
       EXPECT_CALL(
           encoder_,
-          Encode(_,
-                 _,
-                 Pointee(ElementsAre(kDeltaFrame, kDeltaFrame, kDeltaFrame))))
-          .Times(1).WillRepeatedly(Return(0));
+          Encode(_, _, Pointee(ElementsAre(kVideoFrameDelta, kVideoFrameDelta,
+                                           kVideoFrameDelta))))
+          .Times(1)
+          .WillRepeatedly(Return(0));
       return;
     }
     assert(stream >= 0);
     assert(stream < kNumberOfStreams);
-    std::vector<VideoFrameType> frame_types(kNumberOfStreams, kDeltaFrame);
-    frame_types[stream] = kKeyFrame;
-    EXPECT_CALL(
-        encoder_,
-        Encode(_,
-               _,
-               Pointee(ElementsAreArray(&frame_types[0], frame_types.size()))))
-        .Times(1).WillRepeatedly(Return(0));
+    std::vector<FrameType> frame_types(kNumberOfStreams, kVideoFrameDelta);
+    frame_types[stream] = kVideoFrameKey;
+    EXPECT_CALL(encoder_,
+                Encode(_, _, Pointee(ElementsAreArray(&frame_types[0],
+                                                      frame_types.size()))))
+        .Times(1)
+        .WillRepeatedly(Return(0));
   }
 
   static void ConfigureStream(int width,
@@ -297,11 +289,9 @@ TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequests) {
 
 TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequestsInternalCapture) {
   // De-register current external encoder.
-  EXPECT_EQ(0,
-            sender_->RegisterExternalEncoder(NULL, kUnusedPayloadType, false));
+  sender_->RegisterExternalEncoder(nullptr, kUnusedPayloadType, false);
   // Register encoder with internal capture.
-  EXPECT_EQ(
-      0, sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true));
+  sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true);
   EXPECT_EQ(0, sender_->RegisterSendCodec(&settings_, 1, 1200));
   ExpectIntraRequest(0);
   EXPECT_EQ(0, sender_->IntraFrameRequest(0));
@@ -314,6 +304,53 @@ TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequestsInternalCapture) {
   EXPECT_EQ(-1, sender_->IntraFrameRequest(-1));
 }
 
+TEST_F(TestVideoSenderWithMockEncoder, EncoderFramerateUpdatedViaProcess) {
+  sender_->SetChannelParameters(settings_.startBitrate * 1000, 0, 200);
+  const int64_t kRateStatsWindowMs = 2000;
+  const uint32_t kInputFps = 20;
+  int64_t start_time = clock_.TimeInMilliseconds();
+  while (clock_.TimeInMilliseconds() < start_time + kRateStatsWindowMs) {
+    AddFrame();
+    clock_.AdvanceTimeMilliseconds(1000 / kInputFps);
+  }
+  EXPECT_CALL(encoder_, SetRates(_, kInputFps)).Times(1).WillOnce(Return(0));
+  sender_->Process();
+  AddFrame();
+}
+
+TEST_F(TestVideoSenderWithMockEncoder,
+       NoRedundantSetChannelParameterOrSetRatesCalls) {
+  const uint8_t kLossRate = 4;
+  const uint8_t kRtt = 200;
+  const int64_t kRateStatsWindowMs = 2000;
+  const uint32_t kInputFps = 20;
+  int64_t start_time = clock_.TimeInMilliseconds();
+  // Expect initial call to SetChannelParameters. Rates are initialized through
+  // InitEncode and expects no additional call before the framerate (or bitrate)
+  // updates.
+  EXPECT_CALL(encoder_, SetChannelParameters(kLossRate, kRtt))
+      .Times(1)
+      .WillOnce(Return(0));
+  sender_->SetChannelParameters(settings_.startBitrate * 1000, kLossRate, kRtt);
+  while (clock_.TimeInMilliseconds() < start_time + kRateStatsWindowMs) {
+    AddFrame();
+    clock_.AdvanceTimeMilliseconds(1000 / kInputFps);
+  }
+  // After process, input framerate should be updated but not ChannelParameters
+  // as they are the same as before.
+  EXPECT_CALL(encoder_, SetRates(_, kInputFps)).Times(1).WillOnce(Return(0));
+  sender_->Process();
+  AddFrame();
+  // Call to SetChannelParameters with changed bitrate should call encoder
+  // SetRates but not encoder SetChannelParameters (that are unchanged).
+  EXPECT_CALL(encoder_, SetRates(2 * settings_.startBitrate, kInputFps))
+      .Times(1)
+      .WillOnce(Return(0));
+  sender_->SetChannelParameters(2 * settings_.startBitrate * 1000, kLossRate,
+                                kRtt);
+  AddFrame();
+}
+
 class TestVideoSenderWithVp8 : public TestVideoSender {
  public:
   TestVideoSenderWithVp8()
@@ -333,6 +370,8 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
     codec_.minBitrate = 10;
     codec_.startBitrate = codec_bitrate_kbps_;
     codec_.maxBitrate = codec_bitrate_kbps_;
+    encoder_.reset(VP8Encoder::Create());
+    sender_->RegisterExternalEncoder(encoder_.get(), codec_.plType, false);
     EXPECT_EQ(0, sender_->RegisterSendCodec(&codec_, 1, 1200));
   }
 
@@ -340,8 +379,7 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
                                       int height,
                                       int temporal_layers) {
     VideoCodec codec;
-    memset(&codec, 0, sizeof(codec));
-    EXPECT_EQ(0, VideoCodingModule::Codec(kVideoCodecVP8, &codec));
+    VideoCodingModule::Codec(kVideoCodecVP8, &codec);
     codec.width = width;
     codec.height = height;
     codec.codecSpecific.VP8.numberOfTemporalLayers = temporal_layers;
@@ -354,15 +392,13 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
       EXPECT_CALL(post_encode_callback_, Encoded(_, NULL, NULL))
           .WillOnce(Return(0));
       AddFrame();
-
       // SetChannelParameters needs to be called frequently to propagate
       // framerate from the media optimization into the encoder.
       // Note: SetChannelParameters fails if less than 2 frames are in the
       // buffer since it will fail to calculate the framerate.
       if (i != 0) {
-        EXPECT_EQ(VCM_OK,
-                  sender_->SetChannelParameters(
-                      available_bitrate_kbps_ * 1000, 0, 200));
+        EXPECT_EQ(VCM_OK, sender_->SetChannelParameters(
+                              available_bitrate_kbps_ * 1000, 0, 200));
       }
     }
   }
@@ -385,8 +421,12 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
   int available_bitrate_kbps_;
 };
 
-TEST_F(TestVideoSenderWithVp8,
-       DISABLED_ON_ANDROID(FixedTemporalLayersStrategy)) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_FixedTemporalLayersStrategy DISABLED_FixedTemporalLayersStrategy
+#else
+#define MAYBE_FixedTemporalLayersStrategy FixedTemporalLayersStrategy
+#endif
+TEST_F(TestVideoSenderWithVp8, MAYBE_FixedTemporalLayersStrategy) {
   const int low_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][0];
   const int mid_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][1];
   const int high_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][2];
@@ -400,8 +440,13 @@ TEST_F(TestVideoSenderWithVp8,
   }
 }
 
-TEST_F(TestVideoSenderWithVp8,
-       DISABLED_ON_ANDROID(RealTimeTemporalLayersStrategy)) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_RealTimeTemporalLayersStrategy \
+  DISABLED_RealTimeTemporalLayersStrategy
+#else
+#define MAYBE_RealTimeTemporalLayersStrategy RealTimeTemporalLayersStrategy
+#endif
+TEST_F(TestVideoSenderWithVp8, MAYBE_RealTimeTemporalLayersStrategy) {
   Config extra_options;
   extra_options.Set<TemporalLayers::Factory>(
       new RealTimeTemporalLayersFactory());
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/BUILD.gn b/media/webrtc/trunk/webrtc/modules/video_processing/BUILD.gn
index 22bb7c52d9..6d411edda1 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/BUILD.gn
@@ -6,33 +6,37 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
+import("//build/config/arm.gni")
 import("../../build/webrtc.gni")
 
 build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"
 
 source_set("video_processing") {
   sources = [
-    "main/interface/video_processing.h",
-    "main/interface/video_processing_defines.h",
-    "main/source/brighten.cc",
-    "main/source/brighten.h",
-    "main/source/brightness_detection.cc",
-    "main/source/brightness_detection.h",
-    "main/source/color_enhancement.cc",
-    "main/source/color_enhancement.h",
-    "main/source/color_enhancement_private.h",
-    "main/source/content_analysis.cc",
-    "main/source/content_analysis.h",
-    "main/source/deflickering.cc",
-    "main/source/deflickering.h",
-    "main/source/frame_preprocessor.cc",
-    "main/source/frame_preprocessor.h",
-    "main/source/spatial_resampler.cc",
-    "main/source/spatial_resampler.h",
-    "main/source/video_decimator.cc",
-    "main/source/video_decimator.h",
-    "main/source/video_processing_impl.cc",
-    "main/source/video_processing_impl.h",
+    "brightness_detection.cc",
+    "brightness_detection.h",
+    "content_analysis.cc",
+    "content_analysis.h",
+    "deflickering.cc",
+    "deflickering.h",
+    "frame_preprocessor.cc",
+    "frame_preprocessor.h",
+    "include/video_processing.h",
+    "include/video_processing_defines.h",
+    "spatial_resampler.cc",
+    "spatial_resampler.h",
+    "util/denoiser_filter.cc",
+    "util/denoiser_filter.h",
+    "util/denoiser_filter_c.cc",
+    "util/denoiser_filter_c.h",
+    "util/skin_detection.cc",
+    "util/skin_detection.h",
+    "video_decimator.cc",
+    "video_decimator.h",
+    "video_denoiser.cc",
+    "video_denoiser.h",
+    "video_processing_impl.cc",
+    "video_processing_impl.h",
   ]
 
   deps = [
@@ -44,6 +48,9 @@ source_set("video_processing") {
   if (build_video_processing_sse2) {
     deps += [ ":video_processing_sse2" ]
   }
+  if (rtc_build_with_neon) {
+    deps += [ ":video_processing_neon" ]
+  }
 
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
@@ -57,7 +64,11 @@ source_set("video_processing") {
 
 if (build_video_processing_sse2) {
   source_set("video_processing_sse2") {
-    sources = [ "main/source/content_analysis_sse2.cc" ]
+    sources = [
+      "content_analysis_sse2.cc",
+      "util/denoiser_filter_sse2.cc",
+      "util/denoiser_filter_sse2.h",
+    ]
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
@@ -73,3 +84,18 @@ if (build_video_processing_sse2) {
     }
   }
 }
+
+if (rtc_build_with_neon) {
+  source_set("video_processing_neon") {
+    sources = [
+      "util/denoiser_filter_neon.cc",
+      "util/denoiser_filter_neon.h",
+    ]
+    if (current_cpu != "arm64") {
+      configs -= [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
+    }
+    configs += [ "../..:common_config" ]
+    public_configs = [ "../..:common_inherited_config" ]
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/OWNERS b/media/webrtc/trunk/webrtc/modules/video_processing/OWNERS
index f452c9ed83..389d632dfd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/OWNERS
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/OWNERS
@@ -1,4 +1,9 @@
 stefan@webrtc.org
 marpan@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brightness_detection.cc b/media/webrtc/trunk/webrtc/modules/video_processing/brightness_detection.cc
similarity index 62%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/brightness_detection.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/brightness_detection.cc
index 77cdc52cee..7455cf9759 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brightness_detection.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/brightness_detection.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/brightness_detection.h"
+#include "webrtc/modules/video_processing/brightness_detection.h"
 
 #include <math.h>
 
+#include "webrtc/modules/video_processing/include/video_processing.h"
+
 namespace webrtc {
 
 VPMBrightnessDetection::VPMBrightnessDetection() {
@@ -27,15 +28,15 @@ void VPMBrightnessDetection::Reset() {
 }
 
 int32_t VPMBrightnessDetection::ProcessFrame(
-    const I420VideoFrame& frame,
-    const VideoProcessingModule::FrameStats& stats) {
+    const VideoFrame& frame,
+    const VideoProcessing::FrameStats& stats) {
   if (frame.IsZeroSize()) {
     return VPM_PARAMETER_ERROR;
   }
   int width = frame.width();
   int height = frame.height();
 
-  if (!VideoProcessingModule::ValidFrameStats(stats)) {
+  if (!VideoProcessing::ValidFrameStats(stats)) {
     return VPM_PARAMETER_ERROR;
   }
 
@@ -62,11 +63,11 @@ int32_t VPMBrightnessDetection::ProcessFrame(
       // Standard deviation of Y
       const uint8_t* buffer = frame.buffer(kYPlane);
       float std_y = 0;
-      for (int h = 0; h < height; h += (1 << stats.subSamplHeight)) {
-        int row = h*width;
-        for (int w = 0; w < width; w += (1 << stats.subSamplWidth)) {
-          std_y += (buffer[w + row] - stats.mean) * (buffer[w + row] -
-              stats.mean);
+      for (int h = 0; h < height; h += (1 << stats.sub_sampling_factor)) {
+        int row = h * width;
+        for (int w = 0; w < width; w += (1 << stats.sub_sampling_factor)) {
+          std_y +=
+              (buffer[w + row] - stats.mean) * (buffer[w + row] - stats.mean);
         }
       }
       std_y = sqrt(std_y / stats.num_pixels);
@@ -81,37 +82,39 @@ int32_t VPMBrightnessDetection::ProcessFrame(
       float posPerc95 = stats.num_pixels * 0.95f;
       for (uint32_t i = 0; i < 256; i++) {
         sum += stats.hist[i];
-        if (sum < pos_perc05) perc05 = i;     // 5th perc.
-        if (sum < pos_median) median_y = i;    // 50th perc.
+        if (sum < pos_perc05)
+          perc05 = i;  // 5th perc.
+        if (sum < pos_median)
+          median_y = i;  // 50th perc.
         if (sum < posPerc95)
-          perc95 = i;     // 95th perc.
+          perc95 = i;  // 95th perc.
         else
           break;
       }
 
-        // Check if image is too dark
-        if ((std_y < 55) && (perc05 < 50))  {
-          if (median_y < 60 || stats.mean < 80 ||  perc95 < 130 ||
-              prop_low > 0.20) {
-            frame_cnt_dark_++;
-          } else {
-            frame_cnt_dark_ = 0;
-          }
+      // Check if image is too dark
+      if ((std_y < 55) && (perc05 < 50)) {
+        if (median_y < 60 || stats.mean < 80 || perc95 < 130 ||
+            prop_low > 0.20) {
+          frame_cnt_dark_++;
         } else {
           frame_cnt_dark_ = 0;
         }
+      } else {
+        frame_cnt_dark_ = 0;
+      }
 
-        // Check if image is too bright
-        if ((std_y < 52) && (perc95 > 200) && (median_y > 160)) {
-          if (median_y > 185 || stats.mean > 185 || perc05 > 140 ||
-              prop_high > 0.25) {
-            frame_cnt_bright_++;
-          } else {
-            frame_cnt_bright_ = 0;
-          }
+      // Check if image is too bright
+      if ((std_y < 52) && (perc95 > 200) && (median_y > 160)) {
+        if (median_y > 185 || stats.mean > 185 || perc05 > 140 ||
+            prop_high > 0.25) {
+          frame_cnt_bright_++;
         } else {
           frame_cnt_bright_ = 0;
         }
+      } else {
+        frame_cnt_bright_ = 0;
+      }
     } else {
       frame_cnt_dark_ = 0;
       frame_cnt_bright_ = 0;
@@ -122,11 +125,11 @@ int32_t VPMBrightnessDetection::ProcessFrame(
   }
 
   if (frame_cnt_dark_ > frame_cnt_alarm) {
-    return VideoProcessingModule::kDarkWarning;
+    return VideoProcessing::kDarkWarning;
   } else if (frame_cnt_bright_ > frame_cnt_alarm) {
-    return VideoProcessingModule::kBrightWarning;
+    return VideoProcessing::kBrightWarning;
   } else {
-    return VideoProcessingModule::kNoWarning;
+    return VideoProcessing::kNoWarning;
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brightness_detection.h b/media/webrtc/trunk/webrtc/modules/video_processing/brightness_detection.h
similarity index 60%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/brightness_detection.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/brightness_detection.h
index f2600c0bfe..78a7ac5e0b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brightness_detection.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/brightness_detection.h
@@ -8,12 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * brightness_detection.h
- */
-#ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTNESS_DETECTION_H
-#define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTNESS_DETECTION_H
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
+
+#include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -24,8 +22,8 @@ class VPMBrightnessDetection {
   ~VPMBrightnessDetection();
 
   void Reset();
-  int32_t ProcessFrame(const I420VideoFrame& frame,
-                       const VideoProcessingModule::FrameStats& stats);
+  int32_t ProcessFrame(const VideoFrame& frame,
+                       const VideoProcessing::FrameStats& stats);
 
  private:
   uint32_t frame_cnt_bright_;
@@ -34,4 +32,4 @@ class VPMBrightnessDetection {
 
 }  // namespace webrtc
 
-#endif // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTNESS_DETECTION_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.cc b/media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.cc
similarity index 73%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.cc
index f837f5c026..441b94258d 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.cc
@@ -7,13 +7,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
 
 #include <math.h>
 #include <stdlib.h>
 
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -47,9 +47,8 @@ VPMContentAnalysis::~VPMContentAnalysis() {
   Release();
 }
 
-
 VideoContentMetrics* VPMContentAnalysis::ComputeContentMetrics(
-  const I420VideoFrame& inputFrame) {
+    const VideoFrame& inputFrame) {
   if (inputFrame.IsZeroSize())
     return NULL;
 
@@ -75,7 +74,7 @@ VideoContentMetrics* VPMContentAnalysis::ComputeContentMetrics(
     memcpy(prev_frame_.get(), orig_frame_, width_ * height_);
 
     first_frame_ =  false;
-  }
+ }
 #endif
 
   return ContentMetrics();
@@ -103,11 +102,11 @@ int32_t VPMContentAnalysis::Initialize(int width, int height) {
   skip_num_ = 1;
 
   // use skipNum = 2 for 4CIF, WHD
-  if ( (height_ >=  576) && (width_ >= 704) ) {
+  if ((height_ >= 576) && (width_ >= 704)) {
     skip_num_ = 2;
   }
   // use skipNum = 4 for FULLL_HD images
-  if ( (height_ >=  1080) && (width_ >= 1920) ) {
+  if ((height_ >= 1080) && (width_ >= 1920)) {
     skip_num_ = 4;
   }
 
@@ -131,13 +130,12 @@ int32_t VPMContentAnalysis::Initialize(int width, int height) {
     return VPM_MEMORY;
   }
 #endif
-
+ 
   // ok, all initialized
   ca_Init_ = true;
   return VPM_OK;
 }
 
-
 // Compute motion metrics: magnitude over non-zero motion vectors,
 //  and size of zero cluster
 int32_t VPMContentAnalysis::ComputeMotionMetrics() {
@@ -160,37 +158,42 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_C() {
   uint64_t pixelSqSum = 0;
 
   uint32_t num_pixels = 0;  // Counter for # of pixels.
-  const int width_end = ((width_ - 2*border_) & -16) + border_;
+  const int width_end = ((width_ - 2 * border_) & -16) + border_;
   uint8_t *prev_frame = prev_frame_.get();
 
   for (int i = border_; i < sizei - border_; i += skip_num_) {
     for (int j = border_; j < width_end; j++) {
       num_pixels += 1;
-      int ssn =  i * sizej + j;
+      int ssn = i * sizej + j;
 
-      uint8_t currPixel  = orig_frame_[ssn];
-      uint8_t prevPixel  = prev_frame[ssn];
+      uint8_t currPixel = orig_frame_[ssn];
+      uint8_t prevPixel = prev_frame[ssn];
 
-      tempDiffSum += (uint32_t)abs((int16_t)(currPixel - prevPixel));
-      pixelSum += (uint32_t) currPixel;
-      pixelSqSum += (uint64_t) (currPixel * currPixel);
+      tempDiffSum +=
+          static_cast<uint32_t>(abs((int16_t)(currPixel - prevPixel)));
+      pixelSum += static_cast<uint32_t>(currPixel);
+      pixelSqSum += static_cast<uint64_t>(currPixel * currPixel);
     }
   }
 
   // Default.
   motion_magnitude_ = 0.0f;
 
-  if (tempDiffSum == 0) return VPM_OK;
+  if (tempDiffSum == 0)
+    return VPM_OK;
 
   // Normalize over all pixels.
-  float const tempDiffAvg = (float)tempDiffSum / (float)(num_pixels);
-  float const pixelSumAvg = (float)pixelSum / (float)(num_pixels);
-  float const pixelSqSumAvg = (float)pixelSqSum / (float)(num_pixels);
+  float const tempDiffAvg =
+      static_cast<float>(tempDiffSum) / static_cast<float>(num_pixels);
+  float const pixelSumAvg =
+      static_cast<float>(pixelSum) / static_cast<float>(num_pixels);
+  float const pixelSqSumAvg =
+      static_cast<float>(pixelSqSum) / static_cast<float>(num_pixels);
   float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
 
   if (contrast > 0.0) {
     contrast = sqrt(contrast);
-    motion_magnitude_ = tempDiffAvg/contrast;
+    motion_magnitude_ = tempDiffAvg / contrast;
   }
   return VPM_OK;
 }
@@ -214,39 +217,40 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_C() {
   uint32_t spatialErrHSum = 0;
 
   // make sure work section is a multiple of 16
-  const int width_end = ((sizej - 2*border_) & -16) + border_;
+  const int width_end = ((sizej - 2 * border_) & -16) + border_;
 
   for (int i = border_; i < sizei - border_; i += skip_num_) {
     for (int j = border_; j < width_end; j++) {
-      int ssn1=  i * sizej + j;
-      int ssn2 = (i + 1) * sizej + j; // bottom
-      int ssn3 = (i - 1) * sizej + j; // top
-      int ssn4 = i * sizej + j + 1;   // right
-      int ssn5 = i * sizej + j - 1;   // left
+      int ssn1 = i * sizej + j;
+      int ssn2 = (i + 1) * sizej + j;  // bottom
+      int ssn3 = (i - 1) * sizej + j;  // top
+      int ssn4 = i * sizej + j + 1;    // right
+      int ssn5 = i * sizej + j - 1;    // left
 
-      uint16_t refPixel1  = orig_frame_[ssn1] << 1;
-      uint16_t refPixel2  = orig_frame_[ssn1] << 2;
+      uint16_t refPixel1 = orig_frame_[ssn1] << 1;
+      uint16_t refPixel2 = orig_frame_[ssn1] << 2;
 
       uint8_t bottPixel = orig_frame_[ssn2];
       uint8_t topPixel = orig_frame_[ssn3];
       uint8_t rightPixel = orig_frame_[ssn4];
       uint8_t leftPixel = orig_frame_[ssn5];
 
-      spatialErrSum  += (uint32_t) abs((int16_t)(refPixel2
-          - (uint16_t)(bottPixel + topPixel + leftPixel + rightPixel)));
-      spatialErrVSum += (uint32_t) abs((int16_t)(refPixel1
-          - (uint16_t)(bottPixel + topPixel)));
-      spatialErrHSum += (uint32_t) abs((int16_t)(refPixel1
-          - (uint16_t)(leftPixel + rightPixel)));
+      spatialErrSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
+          refPixel2 - static_cast<uint16_t>(bottPixel + topPixel + leftPixel +
+                                            rightPixel))));
+      spatialErrVSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
+          refPixel1 - static_cast<uint16_t>(bottPixel + topPixel))));
+      spatialErrHSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
+          refPixel1 - static_cast<uint16_t>(leftPixel + rightPixel))));
       pixelMSA += orig_frame_[ssn1];
     }
   }
 
   // Normalize over all pixels.
-  const float spatialErr = (float)(spatialErrSum >> 2);
-  const float spatialErrH = (float)(spatialErrHSum >> 1);
-  const float spatialErrV = (float)(spatialErrVSum >> 1);
-  const float norm = (float)pixelMSA;
+  const float spatialErr = static_cast<float>(spatialErrSum >> 2);
+  const float spatialErrH = static_cast<float>(spatialErrHSum >> 1);
+  const float spatialErrV = static_cast<float>(spatialErrVSum >> 1);
+  const float norm = static_cast<float>(pixelMSA);
 
   // 2X2:
   spatial_pred_err_ = spatialErr / norm;
@@ -258,7 +262,8 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_C() {
 }
 
 VideoContentMetrics* VPMContentAnalysis::ContentMetrics() {
-  if (ca_Init_ == false) return NULL;
+  if (ca_Init_ == false)
+    return NULL;
 
   if (content_metrics_) {
     content_metrics_->spatial_pred_err = spatial_pred_err_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.h b/media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.h
similarity index 79%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.h
index 32a63a5879..a8ef9309e9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/content_analysis.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
@@ -35,8 +35,7 @@ class VPMContentAnalysis {
   // Input:           new frame
   // Return value:    pointer to structure containing content Analysis
   //                  metrics or NULL value upon error
-  VideoContentMetrics* ComputeContentMetrics(const I420VideoFrame&
-                                             inputFrame);
+  VideoContentMetrics* ComputeContentMetrics(const VideoFrame& inputFrame);
 
   // Release all allocated memory
   // Output: 0 if OK, negative value upon error
@@ -73,8 +72,8 @@ class VPMContentAnalysis {
   int border_;
 
   // Content Metrics: Stores the local average of the metrics.
-  float motion_magnitude_;   // motion class
-  float spatial_pred_err_;   // spatial class
+  float motion_magnitude_;    // motion class
+  float spatial_pred_err_;    // spatial class
   float spatial_pred_err_h_;  // spatial class
   float spatial_pred_err_v_;  // spatial class
   bool first_frame_;
@@ -85,4 +84,4 @@ class VPMContentAnalysis {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis_sse2.cc b/media/webrtc/trunk/webrtc/modules/video_processing/content_analysis_sse2.cc
similarity index 54%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis_sse2.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/content_analysis_sse2.cc
index 6a92687c40..355c72def0 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/content_analysis_sse2.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/content_analysis_sse2.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
 
 #include <emmintrin.h>
 #include <math.h>
@@ -16,22 +16,22 @@
 namespace webrtc {
 
 int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
-  uint32_t num_pixels = 0;       // counter for # of pixels
-  const uint8_t* imgBufO = orig_frame_ + border_*width_ + border_;
+  uint32_t num_pixels = 0;  // counter for # of pixels
+  const uint8_t* imgBufO = orig_frame_ + border_ * width_ + border_;
   const uint8_t* imgBufP = prev_frame_.get() + border_*width_ + border_;
 
-  const int32_t width_end = ((width_ - 2*border_) & -16) + border_;
+  const int32_t width_end = ((width_ - 2 * border_) & -16) + border_;
 
-  __m128i sad_64   = _mm_setzero_si128();
-  __m128i sum_64   = _mm_setzero_si128();
+  __m128i sad_64 = _mm_setzero_si128();
+  __m128i sum_64 = _mm_setzero_si128();
   __m128i sqsum_64 = _mm_setzero_si128();
-  const __m128i z  = _mm_setzero_si128();
+  const __m128i z = _mm_setzero_si128();
 
-  for (uint16_t i = 0; i < (height_ - 2*border_); i += skip_num_) {
-    __m128i sqsum_32  = _mm_setzero_si128();
+  for (uint16_t i = 0; i < (height_ - 2 * border_); i += skip_num_) {
+    __m128i sqsum_32 = _mm_setzero_si128();
 
-    const uint8_t *lineO = imgBufO;
-    const uint8_t *lineP = imgBufP;
+    const uint8_t* lineO = imgBufO;
+    const uint8_t* lineP = imgBufP;
 
     // Work on 16 pixels at a time.  For HD content with a width of 1920
     // this loop will run ~67 times (depending on border).  Maximum for
@@ -49,14 +49,14 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
       lineP += 16;
 
       // Abs pixel difference between frames.
-      sad_64 = _mm_add_epi64 (sad_64, _mm_sad_epu8(o, p));
+      sad_64 = _mm_add_epi64(sad_64, _mm_sad_epu8(o, p));
 
       // sum of all pixels in frame
-      sum_64 = _mm_add_epi64 (sum_64, _mm_sad_epu8(o, z));
+      sum_64 = _mm_add_epi64(sum_64, _mm_sad_epu8(o, z));
 
       // Squared sum of all pixels in frame.
-      const __m128i olo = _mm_unpacklo_epi8(o,z);
-      const __m128i ohi = _mm_unpackhi_epi8(o,z);
+      const __m128i olo = _mm_unpacklo_epi8(o, z);
+      const __m128i ohi = _mm_unpackhi_epi8(o, z);
 
       const __m128i sqsum_32_lo = _mm_madd_epi16(olo, olo);
       const __m128i sqsum_32_hi = _mm_madd_epi16(ohi, ohi);
@@ -66,9 +66,9 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
     }
 
     // Add to 64 bit running sum as to not roll over.
-    sqsum_64 = _mm_add_epi64(sqsum_64,
-        _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32,z),
-        _mm_unpacklo_epi32(sqsum_32,z)));
+    sqsum_64 =
+        _mm_add_epi64(sqsum_64, _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32, z),
+                                              _mm_unpacklo_epi32(sqsum_32, z)));
 
     imgBufO += width_ * skip_num_;
     imgBufP += width_ * skip_num_;
@@ -81,13 +81,13 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
 
   // Bring sums out of vector registers and into integer register
   // domain, summing them along the way.
-  _mm_store_si128 (&sad_final_128, sad_64);
-  _mm_store_si128 (&sum_final_128, sum_64);
-  _mm_store_si128 (&sqsum_final_128, sqsum_64);
+  _mm_store_si128(&sad_final_128, sad_64);
+  _mm_store_si128(&sum_final_128, sum_64);
+  _mm_store_si128(&sqsum_final_128, sqsum_64);
 
-  uint64_t *sad_final_64 = reinterpret_cast<uint64_t*>(&sad_final_128);
-  uint64_t *sum_final_64 = reinterpret_cast<uint64_t*>(&sum_final_128);
-  uint64_t *sqsum_final_64 = reinterpret_cast<uint64_t*>(&sqsum_final_128);
+  uint64_t* sad_final_64 = reinterpret_cast<uint64_t*>(&sad_final_128);
+  uint64_t* sum_final_64 = reinterpret_cast<uint64_t*>(&sum_final_128);
+  uint64_t* sqsum_final_64 = reinterpret_cast<uint64_t*>(&sqsum_final_128);
 
   const uint32_t pixelSum = sum_final_64[0] + sum_final_64[1];
   const uint64_t pixelSqSum = sqsum_final_64[0] + sqsum_final_64[1];
@@ -96,27 +96,31 @@ int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
   // Default.
   motion_magnitude_ = 0.0f;
 
-  if (tempDiffSum == 0) return VPM_OK;
+  if (tempDiffSum == 0)
+    return VPM_OK;
 
   // Normalize over all pixels.
-  const float tempDiffAvg = (float)tempDiffSum / (float)(num_pixels);
-  const float pixelSumAvg = (float)pixelSum / (float)(num_pixels);
-  const float pixelSqSumAvg = (float)pixelSqSum / (float)(num_pixels);
+  const float tempDiffAvg =
+      static_cast<float>(tempDiffSum) / static_cast<float>(num_pixels);
+  const float pixelSumAvg =
+      static_cast<float>(pixelSum) / static_cast<float>(num_pixels);
+  const float pixelSqSumAvg =
+      static_cast<float>(pixelSqSum) / static_cast<float>(num_pixels);
   float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
 
   if (contrast > 0.0) {
     contrast = sqrt(contrast);
-    motion_magnitude_ = tempDiffAvg/contrast;
+    motion_magnitude_ = tempDiffAvg / contrast;
   }
 
   return VPM_OK;
 }
 
 int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
-  const uint8_t* imgBuf = orig_frame_ + border_*width_;
+  const uint8_t* imgBuf = orig_frame_ + border_ * width_;
   const int32_t width_end = ((width_ - 2 * border_) & -16) + border_;
 
-  __m128i se_32  = _mm_setzero_si128();
+  __m128i se_32 = _mm_setzero_si128();
   __m128i sev_32 = _mm_setzero_si128();
   __m128i seh_32 = _mm_setzero_si128();
   __m128i msa_32 = _mm_setzero_si128();
@@ -127,8 +131,8 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
   // value is maxed out at 65529 for every row, 65529*1080 = 70777800, which
   // will not roll over a 32 bit accumulator.
   // skip_num_ is also used to reduce the number of rows
-  for (int32_t i = 0; i < (height_ - 2*border_); i += skip_num_) {
-    __m128i se_16  = _mm_setzero_si128();
+  for (int32_t i = 0; i < (height_ - 2 * border_); i += skip_num_) {
+    __m128i se_16 = _mm_setzero_si128();
     __m128i sev_16 = _mm_setzero_si128();
     __m128i seh_16 = _mm_setzero_si128();
     __m128i msa_16 = _mm_setzero_si128();
@@ -143,9 +147,9 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
     // border_ could also be adjusted to concentrate on just the center of
     // the images for an HD capture in order to reduce the possiblity of
     // rollover.
-    const uint8_t *lineTop = imgBuf - width_ + border_;
-    const uint8_t *lineCen = imgBuf + border_;
-    const uint8_t *lineBot = imgBuf + width_ + border_;
+    const uint8_t* lineTop = imgBuf - width_ + border_;
+    const uint8_t* lineCen = imgBuf + border_;
+    const uint8_t* lineBot = imgBuf + width_ + border_;
 
     for (int32_t j = 0; j < width_end - border_; j += 16) {
       const __m128i t = _mm_loadu_si128((__m128i*)(lineTop));
@@ -159,20 +163,20 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
       lineBot += 16;
 
       // center pixel unpacked
-      __m128i clo = _mm_unpacklo_epi8(c,z);
-      __m128i chi = _mm_unpackhi_epi8(c,z);
+      __m128i clo = _mm_unpacklo_epi8(c, z);
+      __m128i chi = _mm_unpackhi_epi8(c, z);
 
       // left right pixels unpacked and added together
-      const __m128i lrlo = _mm_add_epi16(_mm_unpacklo_epi8(l,z),
-                                        _mm_unpacklo_epi8(r,z));
-      const __m128i lrhi = _mm_add_epi16(_mm_unpackhi_epi8(l,z),
-                                         _mm_unpackhi_epi8(r,z));
+      const __m128i lrlo =
+          _mm_add_epi16(_mm_unpacklo_epi8(l, z), _mm_unpacklo_epi8(r, z));
+      const __m128i lrhi =
+          _mm_add_epi16(_mm_unpackhi_epi8(l, z), _mm_unpackhi_epi8(r, z));
 
       // top & bottom pixels unpacked and added together
-      const __m128i tblo = _mm_add_epi16(_mm_unpacklo_epi8(t,z),
-                                         _mm_unpacklo_epi8(b,z));
-      const __m128i tbhi = _mm_add_epi16(_mm_unpackhi_epi8(t,z),
-                                         _mm_unpackhi_epi8(b,z));
+      const __m128i tblo =
+          _mm_add_epi16(_mm_unpacklo_epi8(t, z), _mm_unpacklo_epi8(b, z));
+      const __m128i tbhi =
+          _mm_add_epi16(_mm_unpackhi_epi8(t, z), _mm_unpackhi_epi8(b, z));
 
       // running sum of all pixels
       msa_16 = _mm_add_epi16(msa_16, _mm_add_epi16(chi, clo));
@@ -190,29 +194,32 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
       const __m128i sethi = _mm_subs_epi16(chi, _mm_add_epi16(lrhi, tbhi));
 
       // Add to 16 bit running sum
-      se_16  = _mm_add_epi16(se_16, _mm_max_epi16(setlo,
-          _mm_subs_epi16(z, setlo)));
-      se_16  = _mm_add_epi16(se_16, _mm_max_epi16(sethi,
-          _mm_subs_epi16(z, sethi)));
-      sev_16 = _mm_add_epi16(sev_16, _mm_max_epi16(sevtlo,
-                                             _mm_subs_epi16(z, sevtlo)));
-      sev_16 = _mm_add_epi16(sev_16, _mm_max_epi16(sevthi,
-          _mm_subs_epi16(z, sevthi)));
-      seh_16 = _mm_add_epi16(seh_16, _mm_max_epi16(sehtlo,
-            _mm_subs_epi16(z, sehtlo)));
-      seh_16 = _mm_add_epi16(seh_16, _mm_max_epi16(sehthi,
-            _mm_subs_epi16(z, sehthi)));
+      se_16 =
+          _mm_add_epi16(se_16, _mm_max_epi16(setlo, _mm_subs_epi16(z, setlo)));
+      se_16 =
+          _mm_add_epi16(se_16, _mm_max_epi16(sethi, _mm_subs_epi16(z, sethi)));
+      sev_16 = _mm_add_epi16(sev_16,
+                             _mm_max_epi16(sevtlo, _mm_subs_epi16(z, sevtlo)));
+      sev_16 = _mm_add_epi16(sev_16,
+                             _mm_max_epi16(sevthi, _mm_subs_epi16(z, sevthi)));
+      seh_16 = _mm_add_epi16(seh_16,
+                             _mm_max_epi16(sehtlo, _mm_subs_epi16(z, sehtlo)));
+      seh_16 = _mm_add_epi16(seh_16,
+                             _mm_max_epi16(sehthi, _mm_subs_epi16(z, sehthi)));
     }
 
     // Add to 32 bit running sum as to not roll over.
-    se_32  = _mm_add_epi32(se_32, _mm_add_epi32(_mm_unpackhi_epi16(se_16,z),
-        _mm_unpacklo_epi16(se_16,z)));
-    sev_32 = _mm_add_epi32(sev_32, _mm_add_epi32(_mm_unpackhi_epi16(sev_16,z),
-        _mm_unpacklo_epi16(sev_16,z)));
-    seh_32 = _mm_add_epi32(seh_32, _mm_add_epi32(_mm_unpackhi_epi16(seh_16,z),
-        _mm_unpacklo_epi16(seh_16,z)));
-    msa_32 = _mm_add_epi32(msa_32, _mm_add_epi32(_mm_unpackhi_epi16(msa_16,z),
-        _mm_unpacklo_epi16(msa_16,z)));
+    se_32 = _mm_add_epi32(se_32, _mm_add_epi32(_mm_unpackhi_epi16(se_16, z),
+                                               _mm_unpacklo_epi16(se_16, z)));
+    sev_32 =
+        _mm_add_epi32(sev_32, _mm_add_epi32(_mm_unpackhi_epi16(sev_16, z),
+                                            _mm_unpacklo_epi16(sev_16, z)));
+    seh_32 =
+        _mm_add_epi32(seh_32, _mm_add_epi32(_mm_unpackhi_epi16(seh_16, z),
+                                            _mm_unpacklo_epi16(seh_16, z)));
+    msa_32 =
+        _mm_add_epi32(msa_32, _mm_add_epi32(_mm_unpackhi_epi16(msa_16, z),
+                                            _mm_unpacklo_epi16(msa_16, z)));
 
     imgBuf += width_ * skip_num_;
   }
@@ -224,30 +231,30 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
 
   // Bring sums out of vector registers and into integer register
   // domain, summing them along the way.
-  _mm_store_si128 (&se_128, _mm_add_epi64(_mm_unpackhi_epi32(se_32,z),
-        _mm_unpacklo_epi32(se_32,z)));
-  _mm_store_si128 (&sev_128, _mm_add_epi64(_mm_unpackhi_epi32(sev_32,z),
-      _mm_unpacklo_epi32(sev_32,z)));
-  _mm_store_si128 (&seh_128, _mm_add_epi64(_mm_unpackhi_epi32(seh_32,z),
-      _mm_unpacklo_epi32(seh_32,z)));
-  _mm_store_si128 (&msa_128, _mm_add_epi64(_mm_unpackhi_epi32(msa_32,z),
-      _mm_unpacklo_epi32(msa_32,z)));
+  _mm_store_si128(&se_128, _mm_add_epi64(_mm_unpackhi_epi32(se_32, z),
+                                         _mm_unpacklo_epi32(se_32, z)));
+  _mm_store_si128(&sev_128, _mm_add_epi64(_mm_unpackhi_epi32(sev_32, z),
+                                          _mm_unpacklo_epi32(sev_32, z)));
+  _mm_store_si128(&seh_128, _mm_add_epi64(_mm_unpackhi_epi32(seh_32, z),
+                                          _mm_unpacklo_epi32(seh_32, z)));
+  _mm_store_si128(&msa_128, _mm_add_epi64(_mm_unpackhi_epi32(msa_32, z),
+                                          _mm_unpacklo_epi32(msa_32, z)));
 
-  uint64_t *se_64 = reinterpret_cast<uint64_t*>(&se_128);
-  uint64_t *sev_64 = reinterpret_cast<uint64_t*>(&sev_128);
-  uint64_t *seh_64 = reinterpret_cast<uint64_t*>(&seh_128);
-  uint64_t *msa_64 = reinterpret_cast<uint64_t*>(&msa_128);
+  uint64_t* se_64 = reinterpret_cast<uint64_t*>(&se_128);
+  uint64_t* sev_64 = reinterpret_cast<uint64_t*>(&sev_128);
+  uint64_t* seh_64 = reinterpret_cast<uint64_t*>(&seh_128);
+  uint64_t* msa_64 = reinterpret_cast<uint64_t*>(&msa_128);
 
-  const uint32_t spatialErrSum  = se_64[0] + se_64[1];
+  const uint32_t spatialErrSum = se_64[0] + se_64[1];
   const uint32_t spatialErrVSum = sev_64[0] + sev_64[1];
   const uint32_t spatialErrHSum = seh_64[0] + seh_64[1];
   const uint32_t pixelMSA = msa_64[0] + msa_64[1];
 
   // Normalize over all pixels.
-  const float spatialErr  = (float)(spatialErrSum >> 2);
-  const float spatialErrH = (float)(spatialErrHSum >> 1);
-  const float spatialErrV = (float)(spatialErrVSum >> 1);
-  const float norm = (float)pixelMSA;
+  const float spatialErr = static_cast<float>(spatialErrSum >> 2);
+  const float spatialErrH = static_cast<float>(spatialErrHSum >> 1);
+  const float spatialErrV = static_cast<float>(spatialErrVSum >> 1);
+  const float norm = static_cast<float>(pixelMSA);
 
   // 2X2:
   spatial_pred_err_ = spatialErr / norm;
@@ -258,7 +265,7 @@ int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
   // 2X1:
   spatial_pred_err_v_ = spatialErrV / norm;
 
-    return VPM_OK;
+  return VPM_OK;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.cc b/media/webrtc/trunk/webrtc/modules/video_processing/deflickering.cc
similarity index 83%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/deflickering.cc
index d845dbbea0..0e936ce9b7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/deflickering.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/deflickering.h"
+#include "webrtc/modules/video_processing/deflickering.h"
 
 #include <math.h>
 #include <stdlib.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/sort.h"
+#include "webrtc/system_wrappers/include/sort.h"
 
 namespace webrtc {
 
@@ -40,16 +40,17 @@ enum { kLog2OfDownsamplingFactor = 3 };
 // >> fprintf('%d, ', probUW16)
 // Resolution reduced to avoid overflow when multiplying with the
 // (potentially) large number of pixels.
-const uint16_t VPMDeflickering::prob_uw16_[kNumProbs] = {102, 205, 410, 614,
-    819, 1024, 1229, 1434, 1638, 1843, 1946, 1987}; // <Q11>
+const uint16_t VPMDeflickering::prob_uw16_[kNumProbs] = {
+    102,  205,  410,  614,  819,  1024,
+    1229, 1434, 1638, 1843, 1946, 1987};  // <Q11>
 
 // To generate in Matlab:
 // >> numQuants = 14; maxOnlyLength = 5;
 // >> weightUW16 = round(2^15 *
 //    [linspace(0.5, 1.0, numQuants - maxOnlyLength)]);
 // >> fprintf('%d, %d,\n ', weightUW16);
-const uint16_t VPMDeflickering::weight_uw16_[kNumQuants - kMaxOnlyLength] =
-    {16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768}; // <Q15>
+const uint16_t VPMDeflickering::weight_uw16_[kNumQuants - kMaxOnlyLength] = {
+    16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768};  // <Q15>
 
 VPMDeflickering::VPMDeflickering() {
   Reset();
@@ -70,8 +71,8 @@ void VPMDeflickering::Reset() {
   quant_hist_uw8_[0][kNumQuants - 1] = 255;
   for (int32_t i = 0; i < kNumProbs; i++) {
     // Unsigned round. <Q0>
-    quant_hist_uw8_[0][i + 1] = static_cast<uint8_t>(
-        (prob_uw16_[i] * 255 + (1 << 10)) >> 11);
+    quant_hist_uw8_[0][i + 1] =
+        static_cast<uint8_t>((prob_uw16_[i] * 255 + (1 << 10)) >> 11);
   }
 
   for (int32_t i = 1; i < kFrameHistory_size; i++) {
@@ -80,8 +81,8 @@ void VPMDeflickering::Reset() {
   }
 }
 
-int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
-    VideoProcessingModule::FrameStats* stats) {
+int32_t VPMDeflickering::ProcessFrame(VideoFrame* frame,
+                                      VideoProcessing::FrameStats* stats) {
   assert(frame);
   uint32_t frame_memory;
   uint8_t quant_uw8[kNumQuants];
@@ -106,11 +107,12 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
     return VPM_GENERAL_ERROR;
   }
 
-  if (!VideoProcessingModule::ValidFrameStats(*stats)) {
+  if (!VideoProcessing::ValidFrameStats(*stats)) {
     return VPM_GENERAL_ERROR;
   }
 
-  if (PreDetection(frame->timestamp(), *stats) == -1) return VPM_GENERAL_ERROR;
+  if (PreDetection(frame->timestamp(), *stats) == -1)
+    return VPM_GENERAL_ERROR;
 
   // Flicker detection
   int32_t det_flicker = DetectFlicker();
@@ -123,13 +125,13 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
   // Size of luminance component.
   const uint32_t y_size = height * width;
 
-  const uint32_t y_sub_size = width * (((height - 1) >>
-      kLog2OfDownsamplingFactor) + 1);
+  const uint32_t y_sub_size =
+      width * (((height - 1) >> kLog2OfDownsamplingFactor) + 1);
   uint8_t* y_sorted = new uint8_t[y_sub_size];
   uint32_t sort_row_idx = 0;
   for (int i = 0; i < height; i += kDownsamplingFactor) {
-    memcpy(y_sorted + sort_row_idx * width,
-        frame->buffer(kYPlane) + i * width, width);
+    memcpy(y_sorted + sort_row_idx * width, frame->buffer(kYPlane) + i * width,
+           width);
     sort_row_idx++;
   }
 
@@ -152,12 +154,12 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
     quant_uw8[i + 1] = y_sorted[prob_idx_uw32];
   }
 
-  delete [] y_sorted;
+  delete[] y_sorted;
   y_sorted = NULL;
 
   // Shift history for new frame.
   memmove(quant_hist_uw8_[1], quant_hist_uw8_[0],
-      (kFrameHistory_size - 1) * kNumQuants * sizeof(uint8_t));
+          (kFrameHistory_size - 1) * kNumQuants * sizeof(uint8_t));
   // Store current frame in history.
   memcpy(quant_hist_uw8_[0], quant_uw8, kNumQuants * sizeof(uint8_t));
 
@@ -189,9 +191,10 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
     // target = w * maxquant_uw8 + (1 - w) * minquant_uw8
     // Weights w = |weight_uw16_| are in Q15, hence the final output has to be
     // right shifted by 8 to end up in Q7.
-    target_quant_uw16[i] = static_cast<uint16_t>((
-        weight_uw16_[i] * maxquant_uw8[i] +
-        ((1 << 15) - weight_uw16_[i]) * minquant_uw8[i]) >> 8);  // <Q7>
+    target_quant_uw16[i] = static_cast<uint16_t>(
+        (weight_uw16_[i] * maxquant_uw8[i] +
+         ((1 << 15) - weight_uw16_[i]) * minquant_uw8[i]) >>
+        8);  // <Q7>
   }
 
   for (int32_t i = kNumQuants - kMaxOnlyLength; i < kNumQuants; i++) {
@@ -202,13 +205,14 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
   uint16_t mapUW16;  // <Q7>
   for (int32_t i = 1; i < kNumQuants; i++) {
     // As quant and targetQuant are limited to UWord8, it's safe to use Q7 here.
-    tmp_uw32 = static_cast<uint32_t>(target_quant_uw16[i] -
-        target_quant_uw16[i - 1]);
+    tmp_uw32 =
+        static_cast<uint32_t>(target_quant_uw16[i] - target_quant_uw16[i - 1]);
     tmp_uw16 = static_cast<uint16_t>(quant_uw8[i] - quant_uw8[i - 1]);  // <Q0>
 
     if (tmp_uw16 > 0) {
-      increment_uw16 = static_cast<uint16_t>(WebRtcSpl_DivU32U16(tmp_uw32,
-          tmp_uw16)); // <Q7>
+      increment_uw16 =
+          static_cast<uint16_t>(WebRtcSpl_DivU32U16(tmp_uw32,
+                                                    tmp_uw16));  // <Q7>
     } else {
       // The value is irrelevant; the loop below will only iterate once.
       increment_uw16 = 0;
@@ -229,7 +233,7 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
   }
 
   // Frame was altered, so reset stats.
-  VideoProcessingModule::ClearFrameStats(stats);
+  VideoProcessing::ClearFrameStats(stats);
 
   return VPM_OK;
 }
@@ -246,8 +250,9 @@ int32_t VPMDeflickering::ProcessFrame(I420VideoFrame* frame,
               zero.\n
           -1: Error
 */
-int32_t VPMDeflickering::PreDetection(const uint32_t timestamp,
-  const VideoProcessingModule::FrameStats& stats) {
+int32_t VPMDeflickering::PreDetection(
+    const uint32_t timestamp,
+    const VideoProcessing::FrameStats& stats) {
   int32_t mean_val;  // Mean value of frame (Q4)
   uint32_t frame_rate = 0;
   int32_t meanBufferLength;  // Temp variable.
@@ -256,16 +261,16 @@ int32_t VPMDeflickering::PreDetection(const uint32_t timestamp,
   // Update mean value buffer.
   // This should be done even though we might end up in an unreliable detection.
   memmove(mean_buffer_ + 1, mean_buffer_,
-      (kMeanBufferLength - 1) * sizeof(int32_t));
+          (kMeanBufferLength - 1) * sizeof(int32_t));
   mean_buffer_[0] = mean_val;
 
   // Update timestamp buffer.
   // This should be done even though we might end up in an unreliable detection.
-  memmove(timestamp_buffer_ + 1, timestamp_buffer_, (kMeanBufferLength - 1) *
-      sizeof(uint32_t));
+  memmove(timestamp_buffer_ + 1, timestamp_buffer_,
+          (kMeanBufferLength - 1) * sizeof(uint32_t));
   timestamp_buffer_[0] = timestamp;
 
-/* Compute current frame rate (Q4) */
+  /* Compute current frame rate (Q4) */
   if (timestamp_buffer_[kMeanBufferLength - 1] != 0) {
     frame_rate = ((90000 << 4) * (kMeanBufferLength - 1));
     frame_rate /=
@@ -314,22 +319,22 @@ int32_t VPMDeflickering::PreDetection(const uint32_t timestamp,
           -1: Error
 */
 int32_t VPMDeflickering::DetectFlicker() {
-  uint32_t  i;
-  int32_t  freqEst;       // (Q4) Frequency estimate to base detection upon
-  int32_t  ret_val = -1;
+  uint32_t i;
+  int32_t freqEst;  // (Q4) Frequency estimate to base detection upon
+  int32_t ret_val = -1;
 
   /* Sanity check for mean_buffer_length_ */
   if (mean_buffer_length_ < 2) {
     /* Not possible to estimate frequency */
-    return(2);
+    return 2;
   }
   // Count zero crossings with a dead zone to be robust against noise. If the
   // noise std is 2 pixel this corresponds to about 95% confidence interval.
   int32_t deadzone = (kZeroCrossingDeadzone << kmean_valueScaling);  // Q4
   int32_t meanOfBuffer = 0;  // Mean value of mean value buffer.
-  int32_t numZeros     = 0;  // Number of zeros that cross the dead-zone.
-  int32_t cntState     = 0;  // State variable for zero crossing regions.
-  int32_t cntStateOld  = 0;  // Previous state for zero crossing regions.
+  int32_t numZeros = 0;      // Number of zeros that cross the dead-zone.
+  int32_t cntState = 0;      // State variable for zero crossing regions.
+  int32_t cntStateOld = 0;   // Previous state for zero crossing regions.
 
   for (i = 0; i < mean_buffer_length_; i++) {
     meanOfBuffer += mean_buffer_[i];
@@ -370,7 +375,7 @@ int32_t VPMDeflickering::DetectFlicker() {
   int32_t freqAlias = freqEst;
   if (freqEst > kMinFrequencyToDetect) {
     uint8_t aliasState = 1;
-    while(freqState == 0) {
+    while (freqState == 0) {
       /* Increase frequency */
       freqAlias += (aliasState * frame_rate_);
       freqAlias += ((freqEst << 1) * (1 - (aliasState << 1)));
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.h b/media/webrtc/trunk/webrtc/modules/video_processing/deflickering.h
similarity index 57%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/deflickering.h
index aeb4fe6a78..3ff2723aba 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/deflickering.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/deflickering.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCEdeflickering__H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCEdeflickering__H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
 
 #include <string.h>  // NULL
 
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -24,12 +24,11 @@ class VPMDeflickering {
   ~VPMDeflickering();
 
   void Reset();
-  int32_t ProcessFrame(I420VideoFrame* frame,
-                       VideoProcessingModule::FrameStats* stats);
+  int32_t ProcessFrame(VideoFrame* frame, VideoProcessing::FrameStats* stats);
 
  private:
   int32_t PreDetection(uint32_t timestamp,
-                       const VideoProcessingModule::FrameStats& stats);
+                       const VideoProcessing::FrameStats& stats);
 
   int32_t DetectFlicker();
 
@@ -39,13 +38,13 @@ class VPMDeflickering {
   enum { kNumQuants = kNumProbs + 2 };
   enum { kMaxOnlyLength = 5 };
 
-  uint32_t  mean_buffer_length_;
-  uint8_t   detection_state_;    // 0: No flickering
-                                // 1: Flickering detected
-                                // 2: In flickering
-  int32_t    mean_buffer_[kMeanBufferLength];
-  uint32_t   timestamp_buffer_[kMeanBufferLength];
-  uint32_t   frame_rate_;
+  uint32_t mean_buffer_length_;
+  uint8_t detection_state_;  // 0: No flickering
+                             // 1: Flickering detected
+                             // 2: In flickering
+  int32_t mean_buffer_[kMeanBufferLength];
+  uint32_t timestamp_buffer_[kMeanBufferLength];
+  uint32_t frame_rate_;
   static const uint16_t prob_uw16_[kNumProbs];
   static const uint16_t weight_uw16_[kNumQuants - kMaxOnlyLength];
   uint8_t quant_hist_uw8_[kFrameHistory_size][kNumQuants];
@@ -53,4 +52,4 @@ class VPMDeflickering {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCEdeflickering__H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/frame_preprocessor.cc b/media/webrtc/trunk/webrtc/modules/video_processing/frame_preprocessor.cc
new file mode 100644
index 0000000000..6778a597be
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/frame_preprocessor.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_processing/frame_preprocessor.h"
+
+#include "webrtc/modules/video_processing/video_denoiser.h"
+
+namespace webrtc {
+
+VPMFramePreprocessor::VPMFramePreprocessor()
+    : content_metrics_(nullptr),
+      resampled_frame_(),
+      enable_ca_(false),
+      frame_cnt_(0) {
+  spatial_resampler_ = new VPMSimpleSpatialResampler();
+  ca_ = new VPMContentAnalysis(true);
+  vd_ = new VPMVideoDecimator();
+}
+
+VPMFramePreprocessor::~VPMFramePreprocessor() {
+  Reset();
+  delete ca_;
+  delete vd_;
+  delete spatial_resampler_;
+}
+
+void VPMFramePreprocessor::Reset() {
+  ca_->Release();
+  vd_->Reset();
+  content_metrics_ = nullptr;
+  spatial_resampler_->Reset();
+  enable_ca_ = false;
+  frame_cnt_ = 0;
+}
+
+void VPMFramePreprocessor::EnableTemporalDecimation(bool enable) {
+  vd_->EnableTemporalDecimation(enable);
+}
+
+void VPMFramePreprocessor::EnableContentAnalysis(bool enable) {
+  enable_ca_ = enable;
+}
+
+void VPMFramePreprocessor::SetInputFrameResampleMode(
+    VideoFrameResampling resampling_mode) {
+  spatial_resampler_->SetInputFrameResampleMode(resampling_mode);
+}
+
+int32_t VPMFramePreprocessor::SetTargetResolution(uint32_t width,
+                                                  uint32_t height,
+                                                  uint32_t frame_rate) {
+  if ((width == 0) || (height == 0) || (frame_rate == 0)) {
+    return VPM_PARAMETER_ERROR;
+  }
+  int32_t ret_val = 0;
+  ret_val = spatial_resampler_->SetTargetFrameSize(width, height);
+
+  if (ret_val < 0)
+    return ret_val;
+
+  vd_->SetTargetFramerate(frame_rate);
+  return VPM_OK;
+}
+
+void VPMFramePreprocessor::SetTargetFramerate(int frame_rate) {
+  if (frame_rate == -1) {
+    vd_->EnableTemporalDecimation(false);
+  } else {
+    vd_->EnableTemporalDecimation(true);
+    vd_->SetTargetFramerate(frame_rate);
+  }
+}
+
+void VPMFramePreprocessor::UpdateIncomingframe_rate() {
+  vd_->UpdateIncomingframe_rate();
+}
+
+uint32_t VPMFramePreprocessor::GetDecimatedFrameRate() {
+  return vd_->GetDecimatedFrameRate();
+}
+
+uint32_t VPMFramePreprocessor::GetDecimatedWidth() const {
+  return spatial_resampler_->TargetWidth();
+}
+
+uint32_t VPMFramePreprocessor::GetDecimatedHeight() const {
+  return spatial_resampler_->TargetHeight();
+}
+
+void VPMFramePreprocessor::EnableDenosing(bool enable) {
+  denoiser_.reset(new VideoDenoiser(true));
+}
+
+const VideoFrame* VPMFramePreprocessor::PreprocessFrame(
+    const VideoFrame& frame) {
+  if (frame.IsZeroSize()) {
+    return nullptr;
+  }
+
+  vd_->UpdateIncomingframe_rate();
+  if (vd_->DropFrame()) {
+    return nullptr;
+  }
+
+  const VideoFrame* current_frame = &frame;
+  if (denoiser_) {
+    denoiser_->DenoiseFrame(*current_frame, &denoised_frame_);
+    current_frame = &denoised_frame_;
+  }
+
+  if (spatial_resampler_->ApplyResample(current_frame->width(),
+                                        current_frame->height())) {
+    if (spatial_resampler_->ResampleFrame(*current_frame, &resampled_frame_) !=
+        VPM_OK) {
+      return nullptr;
+    }
+    current_frame = &resampled_frame_;
+  }
+
+  // Perform content analysis on the frame to be encoded.
+  if (enable_ca_ && frame_cnt_ % kSkipFrameCA == 0) {
+    // Compute new metrics every |kSkipFramesCA| frames, starting with
+    // the first frame.
+    content_metrics_ = ca_->ComputeContentMetrics(*current_frame);
+  }
+  ++frame_cnt_;
+  return current_frame;
+}
+
+VideoContentMetrics* VPMFramePreprocessor::GetContentMetrics() const {
+  return content_metrics_;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/frame_preprocessor.h b/media/webrtc/trunk/webrtc/modules/video_processing/frame_preprocessor.h
similarity index 56%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/frame_preprocessor.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/frame_preprocessor.h
index 44c057b56a..5bdc576f37 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/frame_preprocessor.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/frame_preprocessor.h
@@ -8,20 +8,23 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * frame_preprocessor.h
- */
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_FRAME_PREPROCESSOR_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_FRAME_PREPROCESSOR_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_FRAME_PREPROCESSOR_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_FRAME_PREPROCESSOR_H_
 
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
-#include "webrtc/modules/video_processing/main/source/spatial_resampler.h"
-#include "webrtc/modules/video_processing/main/source/video_decimator.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
+#include "webrtc/modules/video_processing/spatial_resampler.h"
+#include "webrtc/modules/video_processing/video_decimator.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 
+class VideoDenoiser;
+
+// All pointers/members in this class are assumed to be protected by the class
+// owner.
 class VPMFramePreprocessor {
  public:
   VPMFramePreprocessor();
@@ -38,23 +41,27 @@ class VPMFramePreprocessor {
   void EnableContentAnalysis(bool enable);
 
   // Set target resolution: frame rate and dimension.
-  int32_t SetTargetResolution(uint32_t width, uint32_t height,
+  int32_t SetTargetResolution(uint32_t width,
+                              uint32_t height,
                               uint32_t frame_rate);
 
+  // Set target frame rate.
+  void SetTargetFramerate(int frame_rate);
+
   // Update incoming frame rate/dimension.
   void UpdateIncomingframe_rate();
 
   int32_t updateIncomingFrameSize(uint32_t width, uint32_t height);
 
   // Set decimated values: frame rate/dimension.
-  uint32_t Decimatedframe_rate();
-  uint32_t DecimatedWidth() const;
-  uint32_t DecimatedHeight() const;
+  uint32_t GetDecimatedFrameRate();
+  uint32_t GetDecimatedWidth() const;
+  uint32_t GetDecimatedHeight() const;
 
   // Preprocess output:
-  int32_t PreprocessFrame(const I420VideoFrame& frame,
-                          I420VideoFrame** processed_frame);
-  VideoContentMetrics* ContentMetrics() const;
+  void EnableDenosing(bool enable);
+  const VideoFrame* PreprocessFrame(const VideoFrame& frame);
+  VideoContentMetrics* GetContentMetrics() const;
 
  private:
   // The content does not change so much every frame, so to reduce complexity
@@ -62,15 +69,16 @@ class VPMFramePreprocessor {
   enum { kSkipFrameCA = 2 };
 
   VideoContentMetrics* content_metrics_;
-  I420VideoFrame resampled_frame_;
+  VideoFrame denoised_frame_;
+  VideoFrame resampled_frame_;
   VPMSpatialResampler* spatial_resampler_;
   VPMContentAnalysis* ca_;
   VPMVideoDecimator* vd_;
+  rtc::scoped_ptr<VideoDenoiser> denoiser_;
   bool enable_ca_;
-  int frame_cnt_;
-
+  uint32_t frame_cnt_;
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_FRAME_PREPROCESSOR_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_FRAME_PREPROCESSOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/include/video_processing.h b/media/webrtc/trunk/webrtc/modules/video_processing/include/video_processing.h
new file mode 100644
index 0000000000..a8d6358887
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/include/video_processing.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
+#include "webrtc/video_frame.h"
+
+// The module is largely intended to process video streams, except functionality
+// provided by static functions which operate independent of previous frames. It
+// is recommended, but not required that a unique instance be used for each
+// concurrently processed stream. Similarly, it is recommended to call Reset()
+// before switching to a new stream, but this is not absolutely required.
+//
+// The module provides basic thread safety by permitting only a single function
+// to execute concurrently.
+
+namespace webrtc {
+
+class VideoProcessing {
+ public:
+  struct FrameStats {
+    uint32_t hist[256];  // Frame histogram.
+    uint32_t mean;
+    uint32_t sum;
+    uint32_t num_pixels;
+    uint32_t sub_sampling_factor;  // Sub-sampling factor, in powers of 2.
+  };
+
+  enum BrightnessWarning { kNoWarning, kDarkWarning, kBrightWarning };
+
+  static VideoProcessing* Create();
+  virtual ~VideoProcessing() {}
+
+  // Retrieves statistics for the input frame. This function must be used to
+  // prepare a FrameStats struct for use in certain VPM functions.
+  static void GetFrameStats(const VideoFrame& frame, FrameStats* stats);
+
+  // Checks the validity of a FrameStats struct. Currently, valid implies only
+  // that is had changed from its initialized state.
+  static bool ValidFrameStats(const FrameStats& stats);
+
+  static void ClearFrameStats(FrameStats* stats);
+
+  // Increases/decreases the luminance value. 'delta' can be in the range {}
+  static void Brighten(int delta, VideoFrame* frame);
+
+  // Detects and removes camera flicker from a video stream. Every frame from
+  // the stream must be passed in. A frame will only be altered if flicker has
+  // been detected. Has a fixed-point implementation.
+  // Frame statistics provided by GetFrameStats(). On return the stats will
+  // be reset to zero if the frame was altered. Call GetFrameStats() again
+  // if the statistics for the altered frame are required.
+  virtual int32_t Deflickering(VideoFrame* frame, FrameStats* stats) = 0;
+
+  // Detects if a video frame is excessively bright or dark. Returns a
+  // warning if this is the case. Multiple frames should be passed in before
+  // expecting a warning. Has a floating-point implementation.
+  virtual int32_t BrightnessDetection(const VideoFrame& frame,
+                                      const FrameStats& stats) = 0;
+
+  // The following functions refer to the pre-processor unit within VPM. The
+  // pre-processor perfoms spatial/temporal decimation and content analysis on
+  // the frames prior to encoding.
+
+  // Enable/disable temporal decimation
+  virtual void EnableTemporalDecimation(bool enable) = 0;
+
+  virtual int32_t SetTargetResolution(uint32_t width,
+                                      uint32_t height,
+                                      uint32_t frame_rate) = 0;
+
+  virtual void SetTargetFramerate(int frame_rate) = 0;
+
+  virtual uint32_t GetDecimatedFrameRate() = 0;
+  virtual uint32_t GetDecimatedWidth() const = 0;
+  virtual uint32_t GetDecimatedHeight() const = 0;
+
+  // Set the spatial resampling settings of the VPM according to
+  // VideoFrameResampling.
+  virtual void SetInputFrameResampleMode(
+      VideoFrameResampling resampling_mode) = 0;
+
+  virtual void EnableDenosing(bool enable) = 0;
+  virtual const VideoFrame* PreprocessFrame(const VideoFrame& frame) = 0;
+
+  virtual VideoContentMetrics* GetContentMetrics() const = 0;
+  virtual void EnableContentAnalysis(bool enable) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/interface/video_processing_defines.h b/media/webrtc/trunk/webrtc/modules/video_processing/include/video_processing_defines.h
similarity index 51%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/interface/video_processing_defines.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/include/video_processing_defines.h
index 93a0658966..9cc71bde27 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/interface/video_processing_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/include/video_processing_defines.h
@@ -13,29 +13,29 @@
  * This header file includes the definitions used in the video processor module
  */
 
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
-#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_DEFINES_H_
 
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Error codes
-#define VPM_OK                   0
-#define VPM_GENERAL_ERROR       -1
-#define VPM_MEMORY              -2
-#define VPM_PARAMETER_ERROR     -3
-#define VPM_SCALE_ERROR         -4
-#define VPM_UNINITIALIZED       -5
-#define VPM_UNIMPLEMENTED       -6
+#define VPM_OK 0
+#define VPM_GENERAL_ERROR -1
+#define VPM_MEMORY -2
+#define VPM_PARAMETER_ERROR -3
+#define VPM_SCALE_ERROR -4
+#define VPM_UNINITIALIZED -5
+#define VPM_UNIMPLEMENTED -6
 
 enum VideoFrameResampling {
-  kNoRescaling,         // Disables rescaling.
-  kFastRescaling,       // Point filter.
-  kBiLinear,            // Bi-linear interpolation.
-  kBox,                 // Box inteprolation.
+  kNoRescaling,    // Disables rescaling.
+  kFastRescaling,  // Point filter.
+  kBiLinear,       // Bi-linear interpolation.
+  kBox,            // Box inteprolation.
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/interface/video_processing.h b/media/webrtc/trunk/webrtc/modules/video_processing/main/interface/video_processing.h
deleted file mode 100644
index fb6770b7a4..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/interface/video_processing.h
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_processing.h
- * This header file contains the API required for the video
- * processing module class.
- */
-
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
-
-/**
-   The module is largely intended to process video streams, except functionality
-   provided by static functions which operate independent of previous frames. It
-   is recommended, but not required that a unique instance be used for each
-   concurrently processed stream. Similarly, it is recommended to call Reset()
-   before switching to a new stream, but this is not absolutely required.
-
-   The module provides basic thread safety by permitting only a single function
-   to execute concurrently.
-*/
-
-namespace webrtc {
-
-class VideoProcessingModule : public Module {
- public:
-  /**
-     Structure to hold frame statistics. Populate it with GetFrameStats().
-  */
-  struct FrameStats {
-      FrameStats() :
-          mean(0),
-          sum(0),
-          num_pixels(0),
-          subSamplWidth(0),
-          subSamplHeight(0) {
-    memset(hist, 0, sizeof(hist));
-  }
-
-  uint32_t hist[256];       // FRame histogram.
-  uint32_t mean;            // Frame Mean value.
-  uint32_t sum;             // Sum of frame.
-  uint32_t num_pixels;       // Number of pixels.
-  uint8_t  subSamplWidth;   // Subsampling rate of width in powers of 2.
-  uint8_t  subSamplHeight;  // Subsampling rate of height in powers of 2.
-};
-
-  /**
-     Specifies the warning types returned by BrightnessDetection().
-  */
-  enum BrightnessWarning {
-    kNoWarning,     // Frame has acceptable brightness.
-    kDarkWarning,   // Frame is too dark.
-    kBrightWarning  // Frame is too bright.
-  };
-
-  /*
-     Creates a VPM object.
-
-     \param[in] id
-         Unique identifier of this object.
-
-     \return Pointer to a VPM object.
-  */
-  static VideoProcessingModule* Create(int32_t id);
-
-  /**
-     Destroys a VPM object.
-
-     \param[in] module
-         Pointer to the VPM object to destroy.
-  */
-  static void Destroy(VideoProcessingModule* module);
-
-  /**
-     Not supported.
-  */
-  int64_t TimeUntilNextProcess() override { return -1; }
-
-  /**
-     Not supported.
-  */
-  int32_t Process() override { return -1; }
-
-  /**
-     Resets all processing components to their initial states. This should be
-     called whenever a new video stream is started.
-  */
-  virtual void Reset() = 0;
-
-  /**
-     Retrieves statistics for the input frame. This function must be used to
-     prepare a FrameStats struct for use in certain VPM functions.
-
-     \param[out] stats
-         The frame statistics will be stored here on return.
-
-     \param[in]  frame
-         Reference to the video frame.
-
-     \return 0 on success, -1 on failure.
-  */
-  static int32_t GetFrameStats(FrameStats* stats,
-                               const I420VideoFrame& frame);
-
-  /**
-     Checks the validity of a FrameStats struct. Currently, valid implies only
-     that is had changed from its initialized state.
-
-     \param[in] stats
-         Frame statistics.
-
-     \return True on valid stats, false on invalid stats.
-  */
-  static bool ValidFrameStats(const FrameStats& stats);
-
-  /**
-     Returns a FrameStats struct to its intialized state.
-
-     \param[in,out] stats
-         Frame statistics.
-  */
-  static void ClearFrameStats(FrameStats* stats);
-
-  /**
-     Enhances the color of an image through a constant mapping. Only the
-     chrominance is altered. Has a fixed-point implementation.
-
-     \param[in,out] frame
-         Pointer to the video frame.
-  */
-  static int32_t ColorEnhancement(I420VideoFrame* frame);
-
-  /**
-     Increases/decreases the luminance value.
-
-     \param[in,out] frame
-         Pointer to the video frame.
-
-    \param[in] delta
-         The amount to change the chrominance value of every single pixel.
-         Can be < 0 also.
-
-     \return 0 on success, -1 on failure.
-  */
-  static int32_t Brighten(I420VideoFrame* frame, int delta);
-
-  /**
-     Detects and removes camera flicker from a video stream. Every frame from
-     the stream must be passed in. A frame will only be altered if flicker has
-     been detected. Has a fixed-point implementation.
-
-     \param[in,out] frame
-         Pointer to the video frame.
-
-     \param[in,out] stats
-         Frame statistics provided by GetFrameStats(). On return the stats will
-         be reset to zero if the frame was altered. Call GetFrameStats() again
-         if the statistics for the altered frame are required.
-
-     \return 0 on success, -1 on failure.
-  */
-  virtual int32_t Deflickering(I420VideoFrame* frame, FrameStats* stats) = 0;
-
-  /**
-     Detects if a video frame is excessively bright or dark. Returns a
-     warning if this is the case. Multiple frames should be passed in before
-     expecting a warning. Has a floating-point implementation.
-
-     \param[in] frame
-         Pointer to the video frame.
-
-     \param[in] stats
-         Frame statistics provided by GetFrameStats().
-
-     \return A member of BrightnessWarning on success, -1 on error
-  */
-  virtual int32_t BrightnessDetection(const I420VideoFrame& frame,
-                                      const FrameStats& stats) = 0;
-
-  /**
-  The following functions refer to the pre-processor unit within VPM. The
-  pre-processor perfoms spatial/temporal decimation and content analysis on
-  the frames prior to encoding.
-  */
-
-  /**
-  Enable/disable temporal decimation
-
-  \param[in] enable when true, temporal decimation is enabled
-  */
-  virtual void EnableTemporalDecimation(bool enable) = 0;
-
-  /**
- Set target resolution
-
- \param[in] width
- Target width
-
- \param[in] height
- Target height
-
-  \param[in] frame_rate
-  Target frame_rate
-
-  \return VPM_OK on success, a negative value on error (see error codes)
-
-  */
-  virtual int32_t SetTargetResolution(uint32_t width,
-                                      uint32_t height,
-                                      uint32_t frame_rate) = 0;
-
-  /**
-  Get decimated(target) frame rate
-  */
-  virtual uint32_t Decimatedframe_rate() = 0;
-
-  /**
-  Get decimated(target) frame width
-  */
-  virtual uint32_t DecimatedWidth() const = 0;
-
-  /**
-  Get decimated(target) frame height
-  */
-  virtual uint32_t DecimatedHeight() const = 0 ;
-
-  /**
-  Set the spatial resampling settings of the VPM: The resampler may either be
-  disabled or one of the following:
-  scaling to a close to target dimension followed by crop/pad
-
-  \param[in] resampling_mode
-  Set resampling mode (a member of VideoFrameResampling)
-  */
-  virtual void SetInputFrameResampleMode(VideoFrameResampling
-                                         resampling_mode) = 0;
-
-  /**
-  Get Processed (decimated) frame
-
-  \param[in] frame pointer to the video frame.
-  \param[in] processed_frame pointer (double) to the processed frame. If no
-             processing is required, processed_frame will be NULL.
-
-  \return VPM_OK on success, a negative value on error (see error codes)
-  */
-  virtual int32_t PreprocessFrame(const I420VideoFrame& frame,
-                                  I420VideoFrame** processed_frame) = 0;
-
-  /**
-  Return content metrics for the last processed frame
-  */
-  virtual VideoContentMetrics* ContentMetrics() const = 0 ;
-
-  /**
-  Enable content analysis
-  */
-  virtual void EnableContentAnalysis(bool enable) = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/OWNERS b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brighten.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brighten.cc
deleted file mode 100644
index 907a549064..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brighten.cc
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_processing/main/source/brighten.h"
-
-#include <stdlib.h>
-
-namespace webrtc {
-namespace VideoProcessing {
-
-int32_t Brighten(I420VideoFrame* frame, int delta) {
-  assert(frame);
-  if (frame->IsZeroSize()) {
-    return VPM_PARAMETER_ERROR;
-  }
-  if (frame->width() <= 0 || frame->height() <= 0) {
-    return VPM_PARAMETER_ERROR;
-  }
-
-  int num_pixels = frame->width() * frame->height();
-
-  int look_up[256];
-  for (int i = 0; i < 256; i++) {
-    int val = i + delta;
-    look_up[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
-  }
-
-  uint8_t* temp_ptr = frame->buffer(kYPlane);
-
-  for (int i = 0; i < num_pixels; i++) {
-    *temp_ptr = static_cast<uint8_t>(look_up[*temp_ptr]);
-    temp_ptr++;
-  }
-  return VPM_OK;
-}
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brighten.h b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brighten.h
deleted file mode 100644
index 4d4c0f6dfd..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/brighten.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
-#define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
-
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-int32_t Brighten(I420VideoFrame* frame, int delta);
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
-
-#endif  // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement.cc
deleted file mode 100644
index f4812016ff..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement.cc
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>  // NULL
-
-#include "webrtc/modules/video_processing/main/source/color_enhancement.h"
-#include "webrtc/modules/video_processing/main/source/color_enhancement_private.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-int32_t ColorEnhancement(I420VideoFrame* frame) {
-  // MOZILLA: we don't use this function and by stubbing it out we can avoid
-  // storing colorTable[], which is 64 KiB of static data.
-  assert(false);
-  return VPM_GENERAL_ERROR;
-#if 0
-  assert(frame);
-  // Pointers to U and V color pixels.
-  uint8_t* ptr_u;
-  uint8_t* ptr_v;
-  uint8_t temp_chroma;
-  if (frame->IsZeroSize()) {
-    return VPM_GENERAL_ERROR;
-  }
-  if (frame->width() == 0 || frame->height() == 0) {
-    return VPM_GENERAL_ERROR;
-  }
-
-  // Set pointers to first U and V pixels (skip luminance).
-  ptr_u = frame->buffer(kUPlane);
-  ptr_v = frame->buffer(kVPlane);
-  int size_uv = ((frame->width() + 1) / 2) * ((frame->height() + 1) / 2);
-
-  // Loop through all chrominance pixels and modify color.
-  for (int ix = 0; ix < size_uv; ix++) {
-    temp_chroma = colorTable[*ptr_u][*ptr_v];
-    *ptr_v = colorTable[*ptr_v][*ptr_u];
-    *ptr_u = temp_chroma;
-
-    ptr_u++;
-    ptr_v++;
-  }
-  return VPM_OK;
-#endif
-}
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement.h b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement.h
deleted file mode 100644
index 233a47fe83..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * color_enhancement.h
- */
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_COLOR_ENHANCEMENT_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_COLOR_ENHANCEMENT_H
-
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-int32_t ColorEnhancement(I420VideoFrame* frame);
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_COLOR_ENHANCEMENT_H
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement_private.h b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement_private.h
deleted file mode 100644
index d36390ec83..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/color_enhancement_private.h
+++ /dev/null
@@ -1,290 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_COLOR_ENHANCEMENT_PRIVATE_H_
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_COLOR_ENHANCEMENT_PRIVATE_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-// MOZILLA: comment this out because it's 64 KiB of static data and we don't
-// used the function it's used by.
-#if 0
-// Table created with Matlab script createTable.m
-// Usage:
-//    Umod=colorTable[U][V]
-//    Vmod=colorTable[V][U]
-static const uint8_t colorTable[256][256] = {
-    {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
-    {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
-    {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
-    {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3},
-    {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4},
-    {5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5},
-    {6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6},
-    {7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7},
-    {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8},
-    {9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9},
-    {10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10},
-    {11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11},
-    {12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12},
-    {13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13},
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-    {248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248, 248},
-    {249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249},
-    {250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250},
-    {251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251},
-    {252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252},
-    {253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253},
-    {254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254},
-    {255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255}
-};
-#endif
-
-
-}  // namespace VideoProcessing
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_COLOR_ENHANCEMENT_PRIVATE_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/frame_preprocessor.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/frame_preprocessor.cc
deleted file mode 100644
index 6eff69c041..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/frame_preprocessor.cc
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_processing/main/source/frame_preprocessor.h"
-
-namespace webrtc {
-
-VPMFramePreprocessor::VPMFramePreprocessor()
-    : content_metrics_(NULL),
-      resampled_frame_(),
-      enable_ca_(false),
-      frame_cnt_(0) {
-  spatial_resampler_ = new VPMSimpleSpatialResampler();
-  ca_ = new VPMContentAnalysis(true);
-  vd_ = new VPMVideoDecimator();
-}
-
-VPMFramePreprocessor::~VPMFramePreprocessor() {
-  Reset();
-  delete spatial_resampler_;
-  delete ca_;
-  delete vd_;
-}
-
-void  VPMFramePreprocessor::Reset() {
-  ca_->Release();
-  vd_->Reset();
-  content_metrics_ = NULL;
-  spatial_resampler_->Reset();
-  enable_ca_ = false;
-  frame_cnt_ = 0;
-}
-
-
-void  VPMFramePreprocessor::EnableTemporalDecimation(bool enable) {
-  vd_->EnableTemporalDecimation(enable);
-}
-
-void VPMFramePreprocessor::EnableContentAnalysis(bool enable) {
-  enable_ca_ = enable;
-}
-
-void  VPMFramePreprocessor::SetInputFrameResampleMode(
-    VideoFrameResampling resampling_mode) {
-  spatial_resampler_->SetInputFrameResampleMode(resampling_mode);
-}
-
-int32_t VPMFramePreprocessor::SetTargetResolution(
-    uint32_t width, uint32_t height, uint32_t frame_rate) {
-  if ( (width == 0) || (height == 0) || (frame_rate == 0)) {
-    return VPM_PARAMETER_ERROR;
-  }
-  int32_t ret_val = 0;
-  ret_val = spatial_resampler_->SetTargetFrameSize(width, height);
-
-  if (ret_val < 0) return ret_val;
-
-  ret_val = vd_->SetTargetFramerate(frame_rate);
-  if (ret_val < 0) return ret_val;
-
-  return VPM_OK;
-}
-
-void VPMFramePreprocessor::UpdateIncomingframe_rate() {
-  vd_->UpdateIncomingframe_rate();
-}
-
-uint32_t VPMFramePreprocessor::Decimatedframe_rate() {
-  return vd_->Decimatedframe_rate();
-}
-
-
-uint32_t VPMFramePreprocessor::DecimatedWidth() const {
-  return spatial_resampler_->TargetWidth();
-}
-
-
-uint32_t VPMFramePreprocessor::DecimatedHeight() const {
-  return spatial_resampler_->TargetHeight();
-}
-
-
-int32_t VPMFramePreprocessor::PreprocessFrame(const I420VideoFrame& frame,
-    I420VideoFrame** processed_frame) {
-  if (frame.IsZeroSize()) {
-    return VPM_PARAMETER_ERROR;
-  }
-
-  vd_->UpdateIncomingframe_rate();
-
-  if (vd_->DropFrame()) {
-    return 1;  // drop 1 frame
-  }
-
-  // Resizing incoming frame if needed. Otherwise, remains NULL.
-  // We are not allowed to resample the input frame (must make a copy of it).
-  *processed_frame = NULL;
-  if (spatial_resampler_->ApplyResample(frame.width(), frame.height()))  {
-    int32_t ret = spatial_resampler_->ResampleFrame(frame, &resampled_frame_);
-    if (ret != VPM_OK) return ret;
-    *processed_frame = &resampled_frame_;
-  }
-
-  // Perform content analysis on the frame to be encoded.
-  if (enable_ca_) {
-    // Compute new metrics every |kSkipFramesCA| frames, starting with
-    // the first frame.
-    if (frame_cnt_ % kSkipFrameCA == 0) {
-      if (*processed_frame == NULL)  {
-        content_metrics_ = ca_->ComputeContentMetrics(frame);
-      } else {
-        content_metrics_ = ca_->ComputeContentMetrics(resampled_frame_);
-      }
-    }
-    ++frame_cnt_;
-  }
-  return VPM_OK;
-}
-
-VideoContentMetrics* VPMFramePreprocessor::ContentMetrics() const {
-  return content_metrics_;
-}
-
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_processing_impl.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_processing_impl.cc
deleted file mode 100644
index 6e7808ede3..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_processing_impl.cc
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "webrtc/modules/video_processing/main/source/video_processing_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-
-#include <assert.h>
-
-namespace webrtc {
-
-namespace {
-void  SetSubSampling(VideoProcessingModule::FrameStats* stats,
-                     const int32_t width,
-                     const int32_t height) {
-  if (width * height >= 640 * 480) {
-    stats->subSamplWidth = 3;
-    stats->subSamplHeight = 3;
-  } else if (width * height >= 352 * 288) {
-    stats->subSamplWidth = 2;
-    stats->subSamplHeight = 2;
-  } else if (width * height >= 176 * 144) {
-    stats->subSamplWidth = 1;
-    stats->subSamplHeight = 1;
-  } else {
-    stats->subSamplWidth = 0;
-    stats->subSamplHeight = 0;
-  }
-}
-}  // namespace
-
-VideoProcessingModule* VideoProcessingModule::Create(const int32_t id) {
-  return new VideoProcessingModuleImpl(id);
-}
-
-void VideoProcessingModule::Destroy(VideoProcessingModule* module) {
-  if (module)
-    delete static_cast<VideoProcessingModuleImpl*>(module);
-}
-
-VideoProcessingModuleImpl::VideoProcessingModuleImpl(const int32_t id)
-    : mutex_(*CriticalSectionWrapper::CreateCriticalSection()) {
-}
-
-VideoProcessingModuleImpl::~VideoProcessingModuleImpl() {
-  delete &mutex_;
-}
-
-void VideoProcessingModuleImpl::Reset() {
-  CriticalSectionScoped mutex(&mutex_);
-  deflickering_.Reset();
-  brightness_detection_.Reset();
-  frame_pre_processor_.Reset();
-}
-
-int32_t VideoProcessingModule::GetFrameStats(FrameStats* stats,
-                                             const I420VideoFrame& frame) {
-  if (frame.IsZeroSize()) {
-    LOG(LS_ERROR) << "Zero size frame.";
-    return VPM_PARAMETER_ERROR;
-  }
-
-  int width = frame.width();
-  int height = frame.height();
-
-  ClearFrameStats(stats);  // The histogram needs to be zeroed out.
-  SetSubSampling(stats, width, height);
-
-  const uint8_t* buffer = frame.buffer(kYPlane);
-  // Compute histogram and sum of frame
-  for (int i = 0; i < height; i += (1 << stats->subSamplHeight)) {
-    int k = i * width;
-    for (int j = 0; j < width; j += (1 << stats->subSamplWidth)) {
-      stats->hist[buffer[k + j]]++;
-      stats->sum += buffer[k + j];
-    }
-  }
-
-  stats->num_pixels = (width * height) / ((1 << stats->subSamplWidth) *
-                     (1 << stats->subSamplHeight));
-  assert(stats->num_pixels > 0);
-
-  // Compute mean value of frame
-  stats->mean = stats->sum / stats->num_pixels;
-
-  return VPM_OK;
-}
-
-bool VideoProcessingModule::ValidFrameStats(const FrameStats& stats) {
-  if (stats.num_pixels == 0) {
-    LOG(LS_WARNING) << "Invalid frame stats.";
-    return false;
-  }
-  return true;
-}
-
-void VideoProcessingModule::ClearFrameStats(FrameStats* stats) {
-  stats->mean = 0;
-  stats->sum = 0;
-  stats->num_pixels = 0;
-  stats->subSamplWidth = 0;
-  stats->subSamplHeight = 0;
-  memset(stats->hist, 0, sizeof(stats->hist));
-}
-
-int32_t VideoProcessingModule::ColorEnhancement(I420VideoFrame* frame) {
-  return VideoProcessing::ColorEnhancement(frame);
-}
-
-int32_t VideoProcessingModule::Brighten(I420VideoFrame* frame, int delta) {
-  return VideoProcessing::Brighten(frame, delta);
-}
-
-int32_t VideoProcessingModuleImpl::Deflickering(I420VideoFrame* frame,
-                                                FrameStats* stats) {
-  CriticalSectionScoped mutex(&mutex_);
-  return deflickering_.ProcessFrame(frame, stats);
-}
-
-int32_t VideoProcessingModuleImpl::BrightnessDetection(
-  const I420VideoFrame& frame,
-  const FrameStats& stats) {
-  CriticalSectionScoped mutex(&mutex_);
-  return brightness_detection_.ProcessFrame(frame, stats);
-}
-
-
-void VideoProcessingModuleImpl::EnableTemporalDecimation(bool enable) {
-  CriticalSectionScoped mutex(&mutex_);
-  frame_pre_processor_.EnableTemporalDecimation(enable);
-}
-
-
-void VideoProcessingModuleImpl::SetInputFrameResampleMode(VideoFrameResampling
-                                                          resampling_mode) {
-  CriticalSectionScoped cs(&mutex_);
-  frame_pre_processor_.SetInputFrameResampleMode(resampling_mode);
-}
-
-int32_t VideoProcessingModuleImpl::SetTargetResolution(uint32_t width,
-                                                       uint32_t height,
-                                                       uint32_t frame_rate) {
-  CriticalSectionScoped cs(&mutex_);
-  return frame_pre_processor_.SetTargetResolution(width, height, frame_rate);
-}
-
-uint32_t VideoProcessingModuleImpl::Decimatedframe_rate() {
-  CriticalSectionScoped cs(&mutex_);
-  return  frame_pre_processor_.Decimatedframe_rate();
-}
-
-uint32_t VideoProcessingModuleImpl::DecimatedWidth() const {
-  CriticalSectionScoped cs(&mutex_);
-  return frame_pre_processor_.DecimatedWidth();
-}
-
-uint32_t VideoProcessingModuleImpl::DecimatedHeight() const {
-  CriticalSectionScoped cs(&mutex_);
-  return frame_pre_processor_.DecimatedHeight();
-}
-
-int32_t VideoProcessingModuleImpl::PreprocessFrame(
-    const I420VideoFrame& frame,
-    I420VideoFrame **processed_frame) {
-  CriticalSectionScoped mutex(&mutex_);
-  return frame_pre_processor_.PreprocessFrame(frame, processed_frame);
-}
-
-VideoContentMetrics* VideoProcessingModuleImpl::ContentMetrics() const {
-  CriticalSectionScoped mutex(&mutex_);
-  return frame_pre_processor_.ContentMetrics();
-}
-
-void VideoProcessingModuleImpl::EnableContentAnalysis(bool enable) {
-  CriticalSectionScoped mutex(&mutex_);
-  frame_pre_processor_.EnableContentAnalysis(enable);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_processing_impl.h b/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_processing_impl.h
deleted file mode 100644
index 14a9e54831..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_processing_impl.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
-#define WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
-
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/brighten.h"
-#include "webrtc/modules/video_processing/main/source/brightness_detection.h"
-#include "webrtc/modules/video_processing/main/source/color_enhancement.h"
-#include "webrtc/modules/video_processing/main/source/deflickering.h"
-#include "webrtc/modules/video_processing/main/source/frame_preprocessor.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoProcessingModuleImpl : public VideoProcessingModule {
- public:
-  VideoProcessingModuleImpl(int32_t id);
-
-  virtual ~VideoProcessingModuleImpl();
-
-  void Reset() override;
-
-  int32_t Deflickering(I420VideoFrame* frame, FrameStats* stats) override;
-
-  int32_t BrightnessDetection(const I420VideoFrame& frame,
-                              const FrameStats& stats) override;
-
-  // Frame pre-processor functions
-
-  // Enable temporal decimation
-  void EnableTemporalDecimation(bool enable) override;
-
-  void SetInputFrameResampleMode(VideoFrameResampling resampling_mode) override;
-
-  // Enable content analysis
-  void EnableContentAnalysis(bool enable) override;
-
-  // Set Target Resolution: frame rate and dimension
-  int32_t SetTargetResolution(uint32_t width,
-                              uint32_t height,
-                              uint32_t frame_rate) override;
-
-  // Get decimated values: frame rate/dimension
-  uint32_t Decimatedframe_rate() override;
-  uint32_t DecimatedWidth() const override;
-  uint32_t DecimatedHeight() const override;
-
-  // Preprocess:
-  // Pre-process incoming frame: Sample when needed and compute content
-  // metrics when enabled.
-  // If no resampling takes place - processed_frame is set to NULL.
-  int32_t PreprocessFrame(const I420VideoFrame& frame,
-                          I420VideoFrame** processed_frame) override;
-  VideoContentMetrics* ContentMetrics() const override;
-
- private:
-  CriticalSectionWrapper& mutex_;
-  VPMDeflickering deflickering_;
-  VPMBrightnessDetection brightness_detection_;
-  VPMFramePreprocessor frame_pre_processor_;
-};
-
-}  // namespace
-
-#endif
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/brightness_detection_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
deleted file mode 100644
index 8e15d64393..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-
-using namespace webrtc;
-
-TEST_F(VideoProcessingModuleTest, BrightnessDetection)
-{
-    uint32_t frameNum = 0;
-    int32_t brightnessWarning = 0;
-    uint32_t warningCount = 0;
-    rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-           frame_length_)
-    {
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-        VideoProcessingModule::FrameStats stats;
-        ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-        ASSERT_GE(brightnessWarning = vpm_->BrightnessDetection(video_frame_,
-                                                                stats), 0);
-        if (brightnessWarning != VideoProcessingModule::kNoWarning)
-        {
-            warningCount++;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Expect few warnings
-    float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-    printf("\nWarning proportions:\n");
-    printf("Stock foreman: %.1f %%\n", warningProportion);
-    EXPECT_LT(warningProportion, 10);
-
-    rewind(source_file_);
-    frameNum = 0;
-    warningCount = 0;
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-        frame_length_ &&
-        frameNum < 300)
-    {
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-
-        uint8_t* frame = video_frame_.buffer(kYPlane);
-        uint32_t yTmp = 0;
-        for (int yIdx = 0; yIdx < width_ * height_; yIdx++)
-        {
-            yTmp = frame[yIdx] << 1;
-            if (yTmp > 255)
-            {
-                yTmp = 255;
-            }
-            frame[yIdx] = static_cast<uint8_t>(yTmp);
-        }
-
-        VideoProcessingModule::FrameStats stats;
-        ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-        ASSERT_GE(brightnessWarning = vpm_->BrightnessDetection(video_frame_,
-                                                                stats), 0);
-        EXPECT_NE(VideoProcessingModule::kDarkWarning, brightnessWarning);
-        if (brightnessWarning == VideoProcessingModule::kBrightWarning)
-        {
-            warningCount++;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Expect many brightness warnings
-    warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-    printf("Bright foreman: %.1f %%\n", warningProportion);
-    EXPECT_GT(warningProportion, 95);
-
-    rewind(source_file_);
-    frameNum = 0;
-    warningCount = 0;
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-        frame_length_ && frameNum < 300)
-    {
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-
-        uint8_t* y_plane = video_frame_.buffer(kYPlane);
-        int32_t yTmp = 0;
-        for (int yIdx = 0; yIdx < width_ * height_; yIdx++)
-        {
-            yTmp = y_plane[yIdx] >> 1;
-            y_plane[yIdx] = static_cast<uint8_t>(yTmp);
-        }
-
-        VideoProcessingModule::FrameStats stats;
-        ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-        ASSERT_GE(brightnessWarning = vpm_->BrightnessDetection(video_frame_,
-                                                                stats), 0);
-        EXPECT_NE(VideoProcessingModule::kBrightWarning, brightnessWarning);
-        if (brightnessWarning == VideoProcessingModule::kDarkWarning)
-        {
-            warningCount++;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Expect many darkness warnings
-    warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-    printf("Dark foreman: %.1f %%\n\n", warningProportion);
-    EXPECT_GT(warningProportion, 90);
-}
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/color_enhancement_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
deleted file mode 100644
index 4307be3f3e..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, ColorEnhancement)
-{
-    TickTime t0;
-    TickTime t1;
-    TickInterval acc_ticks;
-
-    // Use a shorter version of the Foreman clip for this test.
-    fclose(source_file_);
-    const std::string video_file =
-      webrtc::test::ResourcePath("foreman_cif_short", "yuv");
-    source_file_  = fopen(video_file.c_str(), "rb");
-    ASSERT_TRUE(source_file_ != NULL) <<
-        "Cannot read source file: " + video_file + "\n";
-
-    std::string output_file = webrtc::test::OutputPath() +
-        "foremanColorEnhancedVPM_cif_short.yuv";
-    FILE* modFile = fopen(output_file.c_str(), "w+b");
-    ASSERT_TRUE(modFile != NULL) << "Could not open output file.\n";
-
-    uint32_t frameNum = 0;
-    rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-        frame_length_)
-    {
-        // Using ConvertToI420 to add stride to the image.
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        frameNum++;
-        t0 = TickTime::Now();
-        ASSERT_EQ(0, VideoProcessingModule::ColorEnhancement(&video_frame_));
-        t1 = TickTime::Now();
-        acc_ticks += t1 - t0;
-        if (PrintI420VideoFrame(video_frame_, modFile) < 0) {
-          return;
-        }
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    printf("\nTime per frame: %d us \n",
-        static_cast<int>(acc_ticks.Microseconds() / frameNum));
-    rewind(modFile);
-
-    printf("Comparing files...\n\n");
-    std::string reference_filename =
-        webrtc::test::ResourcePath("foremanColorEnhanced_cif_short", "yuv");
-    FILE* refFile = fopen(reference_filename.c_str(), "rb");
-    ASSERT_TRUE(refFile != NULL) << "Cannot open reference file: " <<
-        reference_filename << "\n"
-        "Create the reference by running Matlab script createTable.m.";
-
-    // get file lenghts
-    ASSERT_EQ(0, fseek(refFile, 0L, SEEK_END));
-    long refLen = ftell(refFile);
-    ASSERT_NE(-1L, refLen);
-    rewind(refFile);
-    ASSERT_EQ(0, fseek(modFile, 0L, SEEK_END));
-    long testLen = ftell(modFile);
-    ASSERT_NE(-1L, testLen);
-    rewind(modFile);
-    ASSERT_EQ(refLen, testLen) << "File lengths differ.";
-
-    I420VideoFrame refVideoFrame;
-    refVideoFrame.CreateEmptyFrame(width_, height_,
-                                   width_, half_width_, half_width_);
-
-    // Compare frame-by-frame.
-    rtc::scoped_ptr<uint8_t[]> ref_buffer(new uint8_t[frame_length_]);
-    while (fread(video_buffer.get(), 1, frame_length_, modFile) ==
-        frame_length_)
-    {
-        // Using ConvertToI420 to add stride to the image.
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-        ASSERT_EQ(frame_length_, fread(ref_buffer.get(), 1, frame_length_,
-                                       refFile));
-        EXPECT_EQ(
-            0, ConvertToI420(kI420, ref_buffer.get(), 0, 0, width_, height_, 0,
-                             kVideoRotation_0, &refVideoFrame));
-        EXPECT_EQ(0, memcmp(video_frame_.buffer(kYPlane),
-                            refVideoFrame.buffer(kYPlane),
-                            size_y_));
-        EXPECT_EQ(0, memcmp(video_frame_.buffer(kUPlane),
-                            refVideoFrame.buffer(kUPlane),
-                            size_uv_));
-        EXPECT_EQ(0, memcmp(video_frame_.buffer(kVPlane),
-                            refVideoFrame.buffer(kVPlane),
-                            size_uv_));
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    // Verify that all color pixels are enhanced, and no luminance values are
-    // altered.
-
-    rtc::scoped_ptr<uint8_t[]> testFrame(new uint8_t[frame_length_]);
-
-    // Use value 128 as probe value, since we know that this will be changed
-    // in the enhancement.
-    memset(testFrame.get(), 128, frame_length_);
-
-    I420VideoFrame testVideoFrame;
-    testVideoFrame.CreateEmptyFrame(width_, height_,
-                                    width_, half_width_, half_width_);
-    EXPECT_EQ(0, ConvertToI420(kI420, testFrame.get(), 0, 0, width_, height_, 0,
-                               kVideoRotation_0, &testVideoFrame));
-
-    ASSERT_EQ(0, VideoProcessingModule::ColorEnhancement(&testVideoFrame));
-
-    EXPECT_EQ(0, memcmp(testVideoFrame.buffer(kYPlane), testFrame.get(),
-                        size_y_))
-      << "Function is modifying the luminance.";
-
-    EXPECT_NE(0, memcmp(testVideoFrame.buffer(kUPlane),
-                        testFrame.get() + size_y_, size_uv_)) <<
-                        "Function is not modifying all chrominance pixels";
-    EXPECT_NE(0, memcmp(testVideoFrame.buffer(kVPlane),
-                        testFrame.get() + size_y_ + size_uv_, size_uv_)) <<
-                        "Function is not modifying all chrominance pixels";
-
-    ASSERT_EQ(0, fclose(refFile));
-    ASSERT_EQ(0, fclose(modFile));
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc
deleted file mode 100644
index cba1dfc4f8..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/deflickering_test.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, Deflickering)
-{
-    enum { NumRuns = 30 };
-    uint32_t frameNum = 0;
-    const uint32_t frame_rate = 15;
-
-    int64_t min_runtime = 0;
-    int64_t avg_runtime = 0;
-
-    // Close automatically opened Foreman.
-    fclose(source_file_);
-    const std::string input_file =
-        webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
-    source_file_  = fopen(input_file.c_str(), "rb");
-    ASSERT_TRUE(source_file_ != NULL) <<
-        "Cannot read input file: " << input_file << "\n";
-
-    const std::string output_file =
-        webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
-    FILE* deflickerFile = fopen(output_file.c_str(), "wb");
-    ASSERT_TRUE(deflickerFile != NULL) <<
-        "Could not open output file: " << output_file << "\n";
-
-    printf("\nRun time [us / frame]:\n");
-    rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-    for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++)
-    {
-        TickTime t0;
-        TickTime t1;
-        TickInterval acc_ticks;
-        uint32_t timeStamp = 1;
-
-        frameNum = 0;
-        while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-               frame_length_)
-        {
-            frameNum++;
-            EXPECT_EQ(
-                0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-            video_frame_.set_timestamp(timeStamp);
-
-            t0 = TickTime::Now();
-            VideoProcessingModule::FrameStats stats;
-            ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-            ASSERT_EQ(0, vpm_->Deflickering(&video_frame_, &stats));
-            t1 = TickTime::Now();
-            acc_ticks += (t1 - t0);
-
-            if (run_idx == 0)
-            {
-              if (PrintI420VideoFrame(video_frame_, deflickerFile) < 0) {
-                return;
-              }
-            }
-            timeStamp += (90000 / frame_rate);
-        }
-        ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-        printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum));
-        if (acc_ticks.Microseconds() < min_runtime || run_idx == 0)
-        {
-            min_runtime = acc_ticks.Microseconds();
-        }
-        avg_runtime += acc_ticks.Microseconds();
-
-        rewind(source_file_);
-    }
-    ASSERT_EQ(0, fclose(deflickerFile));
-    // TODO(kjellander): Add verification of deflicker output file.
-
-    printf("\nAverage run time = %d us / frame\n",
-        static_cast<int>(avg_runtime / frameNum / NumRuns));
-    printf("Min run time = %d us / frame\n\n",
-        static_cast<int>(min_runtime / frameNum));
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.cc b/media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.cc
similarity index 73%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.cc
index fd90c8f76a..cdbe0efac1 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.cc
@@ -8,8 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/source/spatial_resampler.h"
-
+#include "webrtc/modules/video_processing/spatial_resampler.h"
 
 namespace webrtc {
 
@@ -21,12 +20,13 @@ VPMSimpleSpatialResampler::VPMSimpleSpatialResampler()
 
 VPMSimpleSpatialResampler::~VPMSimpleSpatialResampler() {}
 
-
 int32_t VPMSimpleSpatialResampler::SetTargetFrameSize(int32_t width,
                                                       int32_t height) {
-  if (resampling_mode_ == kNoRescaling) return VPM_OK;
+  if (resampling_mode_ == kNoRescaling)
+    return VPM_OK;
 
-  if (width < 1 || height < 1) return VPM_PARAMETER_ERROR;
+  if (width < 1 || height < 1)
+    return VPM_PARAMETER_ERROR;
 
   target_width_ = width;
   target_height_ = height;
@@ -45,14 +45,14 @@ void VPMSimpleSpatialResampler::Reset() {
   target_height_ = 0;
 }
 
-int32_t VPMSimpleSpatialResampler::ResampleFrame(const I420VideoFrame& inFrame,
-                                                 I420VideoFrame* outFrame) {
+int32_t VPMSimpleSpatialResampler::ResampleFrame(const VideoFrame& inFrame,
+                                                 VideoFrame* outFrame) {
   // Don't copy if frame remains as is.
-  if (resampling_mode_ == kNoRescaling)
-     return VPM_OK;
+  if (resampling_mode_ == kNoRescaling) {
+    return VPM_OK;
   // Check if re-sampling is needed
-  else if ((inFrame.width() == target_width_) &&
-    (inFrame.height() == target_height_))  {
+  } else if ((inFrame.width() == target_width_) &&
+             (inFrame.height() == target_height_)) {
     return VPM_OK;
   }
 
@@ -60,8 +60,8 @@ int32_t VPMSimpleSpatialResampler::ResampleFrame(const I420VideoFrame& inFrame,
   // TODO(mikhal/marpan): Should we allow for setting the filter mode in
   // _scale.Set() with |resampling_mode_|?
   int ret_val = 0;
-  ret_val = scaler_.Set(inFrame.width(), inFrame.height(),
-                       target_width_, target_height_, kI420, kI420, kScaleBox);
+  ret_val = scaler_.Set(inFrame.width(), inFrame.height(), target_width_,
+                        target_height_, kI420, kI420, kScaleBox);
   if (ret_val < 0)
     return ret_val;
 
@@ -86,10 +86,9 @@ int32_t VPMSimpleSpatialResampler::TargetWidth() {
   return target_width_;
 }
 
-bool VPMSimpleSpatialResampler::ApplyResample(int32_t width,
-                                              int32_t height) {
+bool VPMSimpleSpatialResampler::ApplyResample(int32_t width, int32_t height) {
   if ((width == target_width_ && height == target_height_) ||
-       resampling_mode_ == kNoRescaling)
+      resampling_mode_ == kNoRescaling)
     return false;
   else
     return true;
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.h b/media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.h
similarity index 58%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.h
index 05247341d5..51820e24e5 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/spatial_resampler.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/spatial_resampler.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_SPATIAL_RESAMPLER_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_SPATIAL_RESAMPLER_H_
 
 #include "webrtc/typedefs.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
 
 #include "webrtc/common_video/libyuv/include/scaler.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -23,13 +23,13 @@ namespace webrtc {
 
 class VPMSpatialResampler {
  public:
-  virtual ~VPMSpatialResampler() {};
+  virtual ~VPMSpatialResampler() {}
   virtual int32_t SetTargetFrameSize(int32_t width, int32_t height) = 0;
-  virtual void SetInputFrameResampleMode(VideoFrameResampling
-                                         resampling_mode) = 0;
+  virtual void SetInputFrameResampleMode(
+      VideoFrameResampling resampling_mode) = 0;
   virtual void Reset() = 0;
-  virtual int32_t ResampleFrame(const I420VideoFrame& inFrame,
-                                I420VideoFrame* outFrame) = 0;
+  virtual int32_t ResampleFrame(const VideoFrame& inFrame,
+                                VideoFrame* outFrame) = 0;
   virtual int32_t TargetWidth() = 0;
   virtual int32_t TargetHeight() = 0;
   virtual bool ApplyResample(int32_t width, int32_t height) = 0;
@@ -42,20 +42,19 @@ class VPMSimpleSpatialResampler : public VPMSpatialResampler {
   virtual int32_t SetTargetFrameSize(int32_t width, int32_t height);
   virtual void SetInputFrameResampleMode(VideoFrameResampling resampling_mode);
   virtual void Reset();
-  virtual int32_t ResampleFrame(const I420VideoFrame& inFrame,
-                                I420VideoFrame* outFrame);
+  virtual int32_t ResampleFrame(const VideoFrame& inFrame,
+                                VideoFrame* outFrame);
   virtual int32_t TargetWidth();
   virtual int32_t TargetHeight();
   virtual bool ApplyResample(int32_t width, int32_t height);
 
  private:
-
-  VideoFrameResampling        resampling_mode_;
-  int32_t                     target_width_;
-  int32_t                     target_height_;
-  Scaler                      scaler_;
+  VideoFrameResampling resampling_mode_;
+  int32_t target_width_;
+  int32_t target_height_;
+  Scaler scaler_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_SPATIAL_RESAMPLER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/test/brightness_detection_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/test/brightness_detection_test.cc
new file mode 100644
index 0000000000..669bb183e5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/brightness_detection_test.cc
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_BrightnessDetection DISABLED_BrightnessDetection
+#else
+#define MAYBE_BrightnessDetection BrightnessDetection
+#endif
+TEST_F(VideoProcessingTest, MAYBE_BrightnessDetection) {
+  uint32_t frameNum = 0;
+  int32_t brightnessWarning = 0;
+  uint32_t warningCount = 0;
+  rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+         frame_length_) {
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+    frameNum++;
+    VideoProcessing::FrameStats stats;
+    vp_->GetFrameStats(video_frame_, &stats);
+    EXPECT_GT(stats.num_pixels, 0u);
+    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
+              0);
+    if (brightnessWarning != VideoProcessing::kNoWarning) {
+      warningCount++;
+    }
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+  // Expect few warnings
+  float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+  printf("\nWarning proportions:\n");
+  printf("Stock foreman: %.1f %%\n", warningProportion);
+  EXPECT_LT(warningProportion, 10);
+
+  rewind(source_file_);
+  frameNum = 0;
+  warningCount = 0;
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+             frame_length_ &&
+         frameNum < 300) {
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+    frameNum++;
+
+    uint8_t* frame = video_frame_.buffer(kYPlane);
+    uint32_t yTmp = 0;
+    for (int yIdx = 0; yIdx < width_ * height_; yIdx++) {
+      yTmp = frame[yIdx] << 1;
+      if (yTmp > 255) {
+        yTmp = 255;
+      }
+      frame[yIdx] = static_cast<uint8_t>(yTmp);
+    }
+
+    VideoProcessing::FrameStats stats;
+    vp_->GetFrameStats(video_frame_, &stats);
+    EXPECT_GT(stats.num_pixels, 0u);
+    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
+              0);
+    EXPECT_NE(VideoProcessing::kDarkWarning, brightnessWarning);
+    if (brightnessWarning == VideoProcessing::kBrightWarning) {
+      warningCount++;
+    }
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+  // Expect many brightness warnings
+  warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+  printf("Bright foreman: %.1f %%\n", warningProportion);
+  EXPECT_GT(warningProportion, 95);
+
+  rewind(source_file_);
+  frameNum = 0;
+  warningCount = 0;
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+             frame_length_ &&
+         frameNum < 300) {
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+    frameNum++;
+
+    uint8_t* y_plane = video_frame_.buffer(kYPlane);
+    int32_t yTmp = 0;
+    for (int yIdx = 0; yIdx < width_ * height_; yIdx++) {
+      yTmp = y_plane[yIdx] >> 1;
+      y_plane[yIdx] = static_cast<uint8_t>(yTmp);
+    }
+
+    VideoProcessing::FrameStats stats;
+    vp_->GetFrameStats(video_frame_, &stats);
+    EXPECT_GT(stats.num_pixels, 0u);
+    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
+              0);
+    EXPECT_NE(VideoProcessing::kBrightWarning, brightnessWarning);
+    if (brightnessWarning == VideoProcessing::kDarkWarning) {
+      warningCount++;
+    }
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+  // Expect many darkness warnings
+  warningProportion = static_cast<float>(warningCount) / frameNum * 100;
+  printf("Dark foreman: %.1f %%\n\n", warningProportion);
+  EXPECT_GT(warningProportion, 90);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/content_metrics_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/test/content_metrics_test.cc
similarity index 66%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/content_metrics_test.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/test/content_metrics_test.cc
index 8a2404f8e6..782f9cff59 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/content_metrics_test.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/content_metrics_test.cc
@@ -9,27 +9,32 @@
  */
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/content_analysis.h"
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/content_analysis.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
 
 namespace webrtc {
 
-TEST_F(VideoProcessingModuleTest, ContentAnalysis) {
-  VPMContentAnalysis    ca__c(false);
-  VPMContentAnalysis    ca__sse(true);
-  VideoContentMetrics  *_cM_c, *_cM_SSE;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_ContentAnalysis) {
+#else
+TEST_F(VideoProcessingTest, ContentAnalysis) {
+#endif
+  VPMContentAnalysis ca__c(false);
+  VPMContentAnalysis ca__sse(true);
+  VideoContentMetrics* _cM_c;
+  VideoContentMetrics* _cM_SSE;
 
-  ca__c.Initialize(width_,height_);
-  ca__sse.Initialize(width_,height_);
+  ca__c.Initialize(width_, height_);
+  ca__sse.Initialize(width_, height_);
 
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  while (fread(video_buffer.get(), 1, frame_length_, source_file_)
-       == frame_length_) {
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+         frame_length_) {
     // Using ConvertToI420 to add stride to the image.
     EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                                0, kVideoRotation_0, &video_frame_));
-    _cM_c   = ca__c.ComputeContentMetrics(video_frame_);
+    _cM_c = ca__c.ComputeContentMetrics(video_frame_);
     _cM_SSE = ca__sse.ComputeContentMetrics(video_frame_);
 
     ASSERT_EQ(_cM_c->spatial_pred_err, _cM_SSE->spatial_pred_err);
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/createTable.m b/media/webrtc/trunk/webrtc/modules/video_processing/test/createTable.m
similarity index 99%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/createTable.m
rename to media/webrtc/trunk/webrtc/modules/video_processing/test/createTable.m
index 2c7fb522f6..fe8777ee71 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/createTable.m
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/createTable.m
@@ -31,7 +31,7 @@ A=(1-B)/r0;
 f0=A*x0.^2+B*x0; % compander function in zone 1
 
 % equation system for finding second zone parameters
-M=[r0^3 r0^2 r0 1; 
+M=[r0^3 r0^2 r0 1;
     3*r0^2 2*r0 1 0;
     3*r1^2 2*r1 1 0;
     r1^3 r1^2 r1 1];
@@ -173,7 +173,7 @@ for k=1:size(y,3)
         end
     end
 end
-      
+
 fprintf('\nWriting modified test file...')
 writeYUV420file('../out/Debug/foremanColorEnhanced.yuv',y,unew,vnew);
 fprintf(' done\n');
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/test/deflickering_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/test/deflickering_test.cc
new file mode 100644
index 0000000000..5410015b06
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/deflickering_test.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_Deflickering) {
+#else
+TEST_F(VideoProcessingTest, Deflickering) {
+#endif
+  enum { NumRuns = 30 };
+  uint32_t frameNum = 0;
+  const uint32_t frame_rate = 15;
+
+  int64_t min_runtime = 0;
+  int64_t avg_runtime = 0;
+
+  // Close automatically opened Foreman.
+  fclose(source_file_);
+  const std::string input_file =
+      webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
+  source_file_ = fopen(input_file.c_str(), "rb");
+  ASSERT_TRUE(source_file_ != NULL) << "Cannot read input file: " << input_file
+                                    << "\n";
+
+  const std::string output_file =
+      webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
+  FILE* deflickerFile = fopen(output_file.c_str(), "wb");
+  ASSERT_TRUE(deflickerFile != NULL)
+      << "Could not open output file: " << output_file << "\n";
+
+  printf("\nRun time [us / frame]:\n");
+  rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
+  for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++) {
+    TickTime t0;
+    TickTime t1;
+    TickInterval acc_ticks;
+    uint32_t timeStamp = 1;
+
+    frameNum = 0;
+    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+           frame_length_) {
+      frameNum++;
+      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
+                                 height_, 0, kVideoRotation_0, &video_frame_));
+      video_frame_.set_timestamp(timeStamp);
+
+      t0 = TickTime::Now();
+      VideoProcessing::FrameStats stats;
+      vp_->GetFrameStats(video_frame_, &stats);
+      EXPECT_GT(stats.num_pixels, 0u);
+      ASSERT_EQ(0, vp_->Deflickering(&video_frame_, &stats));
+      t1 = TickTime::Now();
+      acc_ticks += (t1 - t0);
+
+      if (run_idx == 0) {
+        if (PrintVideoFrame(video_frame_, deflickerFile) < 0) {
+          return;
+        }
+      }
+      timeStamp += (90000 / frame_rate);
+    }
+    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+
+    printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum));
+    if (acc_ticks.Microseconds() < min_runtime || run_idx == 0) {
+      min_runtime = acc_ticks.Microseconds();
+    }
+    avg_runtime += acc_ticks.Microseconds();
+
+    rewind(source_file_);
+  }
+  ASSERT_EQ(0, fclose(deflickerFile));
+  // TODO(kjellander): Add verification of deflicker output file.
+
+  printf("\nAverage run time = %d us / frame\n",
+         static_cast<int>(avg_runtime / frameNum / NumRuns));
+  printf("Min run time = %d us / frame\n\n",
+         static_cast<int>(min_runtime / frameNum));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/test/denoiser_test.cc b/media/webrtc/trunk/webrtc/modules/video_processing/test/denoiser_test.cc
new file mode 100644
index 0000000000..551a77617d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/denoiser_test.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+#include "webrtc/modules/video_processing/video_denoiser.h"
+
+namespace webrtc {
+
+TEST_F(VideoProcessingTest, CopyMem) {
+  rtc::scoped_ptr<DenoiserFilter> df_c(DenoiserFilter::Create(false));
+  rtc::scoped_ptr<DenoiserFilter> df_sse_neon(DenoiserFilter::Create(true));
+  uint8_t src[16 * 16], dst[16 * 16];
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      src[i * 16 + j] = i * 16 + j;
+    }
+  }
+
+  memset(dst, 0, 8 * 8);
+  df_c->CopyMem8x8(src, 8, dst, 8);
+  EXPECT_EQ(0, memcmp(src, dst, 8 * 8));
+
+  memset(dst, 0, 16 * 16);
+  df_c->CopyMem16x16(src, 16, dst, 16);
+  EXPECT_EQ(0, memcmp(src, dst, 16 * 16));
+
+  memset(dst, 0, 8 * 8);
+  df_sse_neon->CopyMem16x16(src, 8, dst, 8);
+  EXPECT_EQ(0, memcmp(src, dst, 8 * 8));
+
+  memset(dst, 0, 16 * 16);
+  df_sse_neon->CopyMem16x16(src, 16, dst, 16);
+  EXPECT_EQ(0, memcmp(src, dst, 16 * 16));
+}
+
+TEST_F(VideoProcessingTest, Variance) {
+  rtc::scoped_ptr<DenoiserFilter> df_c(DenoiserFilter::Create(false));
+  rtc::scoped_ptr<DenoiserFilter> df_sse_neon(DenoiserFilter::Create(true));
+  uint8_t src[16 * 16], dst[16 * 16];
+  uint32_t sum = 0, sse = 0, var;
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      src[i * 16 + j] = i * 16 + j;
+    }
+  }
+  // Compute the 16x8 variance of the 16x16 block.
+  for (int i = 0; i < 8; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      sum += (i * 32 + j);
+      sse += (i * 32 + j) * (i * 32 + j);
+    }
+  }
+  var = sse - ((sum * sum) >> 7);
+  memset(dst, 0, 16 * 16);
+  EXPECT_EQ(var, df_c->Variance16x8(src, 16, dst, 16, &sse));
+  EXPECT_EQ(var, df_sse_neon->Variance16x8(src, 16, dst, 16, &sse));
+}
+
+TEST_F(VideoProcessingTest, MbDenoise) {
+  rtc::scoped_ptr<DenoiserFilter> df_c(DenoiserFilter::Create(false));
+  rtc::scoped_ptr<DenoiserFilter> df_sse_neon(DenoiserFilter::Create(true));
+  uint8_t running_src[16 * 16], src[16 * 16], dst[16 * 16], dst_ref[16 * 16];
+
+  // Test case: |diff| <= |3 + shift_inc1|
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 2;
+      dst_ref[i * 16 + j] = running_src[i * 16 + j];
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+
+  // Test case: |diff| >= |4 + shift_inc1|
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 5;
+      dst_ref[i * 16 + j] = src[i * 16 + j] - 2;
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+  memset(dst, 0, 16 * 16);
+  df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+
+  // Test case: |diff| >= 8
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 8;
+      dst_ref[i * 16 + j] = src[i * 16 + j] - 6;
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+  memset(dst, 0, 16 * 16);
+  df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(0, memcmp(dst, dst_ref, 16 * 16));
+
+  // Test case: |diff| > 15
+  for (int i = 0; i < 16; ++i) {
+    for (int j = 0; j < 16; ++j) {
+      running_src[i * 16 + j] = i * 11 + j;
+      src[i * 16 + j] = i * 11 + j + 16;
+    }
+  }
+  memset(dst, 0, 16 * 16);
+  DenoiserDecision decision =
+      df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(COPY_BLOCK, decision);
+  decision = df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
+  EXPECT_EQ(COPY_BLOCK, decision);
+}
+
+TEST_F(VideoProcessingTest, Denoiser) {
+  // Create pure C denoiser.
+  VideoDenoiser denoiser_c(false);
+  // Create SSE or NEON denoiser.
+  VideoDenoiser denoiser_sse_neon(true);
+  VideoFrame denoised_frame_c;
+  VideoFrame denoised_frame_sse_neon;
+
+  rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
+  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
+         frame_length_) {
+    // Using ConvertToI420 to add stride to the image.
+    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
+                               0, kVideoRotation_0, &video_frame_));
+
+    denoiser_c.DenoiseFrame(video_frame_, &denoised_frame_c);
+    denoiser_sse_neon.DenoiseFrame(video_frame_, &denoised_frame_sse_neon);
+
+    // Denoising results should be the same for C and SSE/NEON denoiser.
+    ASSERT_EQ(true, denoised_frame_c.EqualsFrame(denoised_frame_sse_neon));
+  }
+  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/readYUV420file.m b/media/webrtc/trunk/webrtc/modules/video_processing/test/readYUV420file.m
similarity index 95%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/readYUV420file.m
rename to media/webrtc/trunk/webrtc/modules/video_processing/test/readYUV420file.m
index 03013efd3a..f409820283 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/readYUV420file.m
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/readYUV420file.m
@@ -10,7 +10,7 @@ end
 nPx=width*height;
 
 % nPx bytes luminance, nPx/4 bytes U, nPx/4 bytes V
-frameSizeBytes = nPx*1.5; 
+frameSizeBytes = nPx*1.5;
 
 % calculate number of frames
 fseek(fid,0,'eof'); % move to end of file
@@ -27,19 +27,19 @@ V=uint8(zeros(height/2,width/2,numFrames));
 [X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
 
 for k=1:numFrames
-    
+
     % Store luminance
     Y(:,:,k)=uint8(reshape(X(1:nPx), width, height).');
-    
+
     % Store U channel
     U(:,:,k)=uint8(reshape(X(nPx + (1:nPx/4)), width/2, height/2).');
 
     % Store V channel
     V(:,:,k)=uint8(reshape(X(nPx + nPx/4 + (1:nPx/4)), width/2, height/2).');
-    
+
     % Read next frame
     [X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
 end
 
-    
+
 fclose(fid);
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc b/media/webrtc/trunk/webrtc/modules/video_processing/test/video_processing_unittest.cc
similarity index 58%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/test/video_processing_unittest.cc
index 8d3fcd6531..2fd8fb6673 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/video_processing_unittest.cc
@@ -8,21 +8,30 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h"
+#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
+
+#include <gflags/gflags.h>
 
 #include <string>
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
-static void PreprocessFrameAndVerify(const I420VideoFrame& source,
+namespace {
+
+// Define command line flag 'gen_files' (default value: false).
+DEFINE_bool(gen_files, false, "Output files for visual inspection.");
+
+}  // namespace
+
+static void PreprocessFrameAndVerify(const VideoFrame& source,
                                      int target_width,
                                      int target_height,
-                                     VideoProcessingModule* vpm,
-                                     I420VideoFrame** out_frame);
+                                     VideoProcessing* vpm,
+                                     const VideoFrame* out_frame);
 static void CropFrame(const uint8_t* source_data,
                       int source_width,
                       int source_height,
@@ -30,25 +39,24 @@ static void CropFrame(const uint8_t* source_data,
                       int offset_y,
                       int cropped_width,
                       int cropped_height,
-                      I420VideoFrame* cropped_frame);
+                      VideoFrame* cropped_frame);
 // The |source_data| is cropped and scaled to |target_width| x |target_height|,
 // and then scaled back to the expected cropped size. |expected_psnr| is used to
 // verify basic quality, and is set to be ~0.1/0.05dB lower than actual PSNR
 // verified under the same conditions.
-static void TestSize(const I420VideoFrame& source_frame,
-                     const I420VideoFrame& cropped_source_frame,
+static void TestSize(const VideoFrame& source_frame,
+                     const VideoFrame& cropped_source_frame,
                      int target_width,
                      int target_height,
                      double expected_psnr,
-                     VideoProcessingModule* vpm);
-bool CompareFrames(const webrtc::I420VideoFrame& frame1,
-                   const webrtc::I420VideoFrame& frame2);
-static void WriteProcessedFrameForVisualInspection(
-    const I420VideoFrame& source,
-    const I420VideoFrame& processed);
+                     VideoProcessing* vpm);
+static bool CompareFrames(const webrtc::VideoFrame& frame1,
+                          const webrtc::VideoFrame& frame2);
+static void WriteProcessedFrameForVisualInspection(const VideoFrame& source,
+                                                   const VideoFrame& processed);
 
-VideoProcessingModuleTest::VideoProcessingModuleTest()
-    : vpm_(NULL),
+VideoProcessingTest::VideoProcessingTest()
+    : vp_(NULL),
       source_file_(NULL),
       width_(352),
       half_width_((width_ + 1) / 2),
@@ -57,162 +65,184 @@ VideoProcessingModuleTest::VideoProcessingModuleTest()
       size_uv_(half_width_ * ((height_ + 1) / 2)),
       frame_length_(CalcBufferSize(kI420, width_, height_)) {}
 
-void VideoProcessingModuleTest::SetUp() {
-  vpm_ = VideoProcessingModule::Create(0);
-  ASSERT_TRUE(vpm_ != NULL);
+void VideoProcessingTest::SetUp() {
+  vp_ = VideoProcessing::Create();
+  ASSERT_TRUE(vp_ != NULL);
 
   ASSERT_EQ(0, video_frame_.CreateEmptyFrame(width_, height_, width_,
-                                            half_width_, half_width_));
+                                             half_width_, half_width_));
   // Clear video frame so DrMemory/Valgrind will allow reads of the buffer.
   memset(video_frame_.buffer(kYPlane), 0, video_frame_.allocated_size(kYPlane));
   memset(video_frame_.buffer(kUPlane), 0, video_frame_.allocated_size(kUPlane));
   memset(video_frame_.buffer(kVPlane), 0, video_frame_.allocated_size(kVPlane));
   const std::string video_file =
       webrtc::test::ResourcePath("foreman_cif", "yuv");
-  source_file_  = fopen(video_file.c_str(),"rb");
-  ASSERT_TRUE(source_file_ != NULL) <<
-      "Cannot read source file: " + video_file + "\n";
+  source_file_ = fopen(video_file.c_str(), "rb");
+  ASSERT_TRUE(source_file_ != NULL)
+      << "Cannot read source file: " + video_file + "\n";
 }
 
-void VideoProcessingModuleTest::TearDown() {
-  if (source_file_ != NULL)  {
+void VideoProcessingTest::TearDown() {
+  if (source_file_ != NULL) {
     ASSERT_EQ(0, fclose(source_file_));
   }
   source_file_ = NULL;
-
-  if (vpm_ != NULL)  {
-    VideoProcessingModule::Destroy(vpm_);
-  }
-  vpm_ = NULL;
+  delete vp_;
+  vp_ = NULL;
 }
 
-TEST_F(VideoProcessingModuleTest, HandleNullBuffer) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_HandleNullBuffer) {
+#else
+TEST_F(VideoProcessingTest, HandleNullBuffer) {
+#endif
   // TODO(mikhal/stefan): Do we need this one?
-  VideoProcessingModule::FrameStats stats;
+  VideoProcessing::FrameStats stats;
   // Video frame with unallocated buffer.
-  I420VideoFrame videoFrame;
+  VideoFrame videoFrame;
 
-  EXPECT_EQ(-3, vpm_->GetFrameStats(&stats, videoFrame));
+  vp_->GetFrameStats(videoFrame, &stats);
+  EXPECT_EQ(stats.num_pixels, 0u);
 
-  EXPECT_EQ(-1, vpm_->ColorEnhancement(&videoFrame));
+  EXPECT_EQ(-1, vp_->Deflickering(&videoFrame, &stats));
 
-  EXPECT_EQ(-1, vpm_->Deflickering(&videoFrame, &stats));
-
-  EXPECT_EQ(-3, vpm_->BrightnessDetection(videoFrame, stats));
+  EXPECT_EQ(-3, vp_->BrightnessDetection(videoFrame, stats));
 }
 
-TEST_F(VideoProcessingModuleTest, HandleBadStats) {
-  VideoProcessingModule::FrameStats stats;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_HandleBadStats) {
+#else
+TEST_F(VideoProcessingTest, HandleBadStats) {
+#endif
+  VideoProcessing::FrameStats stats;
+  vp_->ClearFrameStats(&stats);
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
 
-  EXPECT_EQ(-1, vpm_->Deflickering(&video_frame_, &stats));
+  EXPECT_EQ(-1, vp_->Deflickering(&video_frame_, &stats));
 
-  EXPECT_EQ(-3, vpm_->BrightnessDetection(video_frame_, stats));
+  EXPECT_EQ(-3, vp_->BrightnessDetection(video_frame_, stats));
 }
 
-TEST_F(VideoProcessingModuleTest, IdenticalResultsAfterReset) {
-  I420VideoFrame video_frame2;
-  VideoProcessingModule::FrameStats stats;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_IdenticalResultsAfterReset) {
+#else
+TEST_F(VideoProcessingTest, IdenticalResultsAfterReset) {
+#endif
+  VideoFrame video_frame2;
+  VideoProcessing::FrameStats stats;
   // Only testing non-static functions here.
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
-  ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
+  vp_->GetFrameStats(video_frame_, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
   ASSERT_EQ(0, video_frame2.CopyFrame(video_frame_));
-  ASSERT_EQ(0, vpm_->Deflickering(&video_frame_, &stats));
-  vpm_->Reset();
+  ASSERT_EQ(0, vp_->Deflickering(&video_frame_, &stats));
+
   // Retrieve frame stats again in case Deflickering() has zeroed them.
-  ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame2));
-  ASSERT_EQ(0, vpm_->Deflickering(&video_frame2, &stats));
+  vp_->GetFrameStats(video_frame2, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
+  ASSERT_EQ(0, vp_->Deflickering(&video_frame2, &stats));
   EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
 
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
-  ASSERT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
+  vp_->GetFrameStats(video_frame_, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
   video_frame2.CopyFrame(video_frame_);
-  ASSERT_EQ(0, vpm_->BrightnessDetection(video_frame_, stats));
-  vpm_->Reset();
-  ASSERT_EQ(0, vpm_->BrightnessDetection(video_frame2, stats));
+  ASSERT_EQ(0, vp_->BrightnessDetection(video_frame_, stats));
+
+  ASSERT_EQ(0, vp_->BrightnessDetection(video_frame2, stats));
   EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
 }
 
-TEST_F(VideoProcessingModuleTest, FrameStats) {
-  VideoProcessingModule::FrameStats stats;
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_FrameStats) {
+#else
+TEST_F(VideoProcessingTest, FrameStats) {
+#endif
+  VideoProcessing::FrameStats stats;
+  vp_->ClearFrameStats(&stats);
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
 
-  EXPECT_FALSE(vpm_->ValidFrameStats(stats));
-  EXPECT_EQ(0, vpm_->GetFrameStats(&stats, video_frame_));
-  EXPECT_TRUE(vpm_->ValidFrameStats(stats));
+  EXPECT_FALSE(vp_->ValidFrameStats(stats));
+  vp_->GetFrameStats(video_frame_, &stats);
+  EXPECT_GT(stats.num_pixels, 0u);
+  EXPECT_TRUE(vp_->ValidFrameStats(stats));
 
   printf("\nFrameStats\n");
-  printf("mean: %u\nnum_pixels: %u\nsubSamplWidth: "
-         "%u\nsumSamplHeight: %u\nsum: %u\n\n",
+  printf("mean: %u\nnum_pixels: %u\nsubSamplFactor: %u\nsum: %u\n\n",
          static_cast<unsigned int>(stats.mean),
          static_cast<unsigned int>(stats.num_pixels),
-         static_cast<unsigned int>(stats.subSamplHeight),
-         static_cast<unsigned int>(stats.subSamplWidth),
+         static_cast<unsigned int>(stats.sub_sampling_factor),
          static_cast<unsigned int>(stats.sum));
 
-  vpm_->ClearFrameStats(&stats);
-  EXPECT_FALSE(vpm_->ValidFrameStats(stats));
+  vp_->ClearFrameStats(&stats);
+  EXPECT_FALSE(vp_->ValidFrameStats(stats));
 }
 
-TEST_F(VideoProcessingModuleTest, PreprocessorLogic) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_PreprocessorLogic) {
+#else
+TEST_F(VideoProcessingTest, PreprocessorLogic) {
+#endif
   // Disable temporal sampling (frame dropping).
-  vpm_->EnableTemporalDecimation(false);
+  vp_->EnableTemporalDecimation(false);
   int resolution = 100;
-  EXPECT_EQ(VPM_OK, vpm_->SetTargetResolution(resolution, resolution, 15));
-  EXPECT_EQ(VPM_OK, vpm_->SetTargetResolution(resolution, resolution, 30));
+  EXPECT_EQ(VPM_OK, vp_->SetTargetResolution(resolution, resolution, 15));
+  EXPECT_EQ(VPM_OK, vp_->SetTargetResolution(resolution, resolution, 30));
   // Disable spatial sampling.
-  vpm_->SetInputFrameResampleMode(kNoRescaling);
-  EXPECT_EQ(VPM_OK, vpm_->SetTargetResolution(resolution, resolution, 30));
-  I420VideoFrame* out_frame = NULL;
+  vp_->SetInputFrameResampleMode(kNoRescaling);
+  EXPECT_EQ(VPM_OK, vp_->SetTargetResolution(resolution, resolution, 30));
+  VideoFrame* out_frame = NULL;
   // Set rescaling => output frame != NULL.
-  vpm_->SetInputFrameResampleMode(kFastRescaling);
-  PreprocessFrameAndVerify(video_frame_, resolution, resolution, vpm_,
-                           &out_frame);
+  vp_->SetInputFrameResampleMode(kFastRescaling);
+  PreprocessFrameAndVerify(video_frame_, resolution, resolution, vp_,
+                           out_frame);
   // No rescaling=> output frame = NULL.
-  vpm_->SetInputFrameResampleMode(kNoRescaling);
-  EXPECT_EQ(VPM_OK, vpm_->PreprocessFrame(video_frame_, &out_frame));
-  EXPECT_TRUE(out_frame == NULL);
+  vp_->SetInputFrameResampleMode(kNoRescaling);
+  EXPECT_TRUE(vp_->PreprocessFrame(video_frame_) != nullptr);
 }
 
-TEST_F(VideoProcessingModuleTest, Resampler) {
+#if defined(WEBRTC_IOS)
+TEST_F(VideoProcessingTest, DISABLED_Resampler) {
+#else
+TEST_F(VideoProcessingTest, Resampler) {
+#endif
   enum { NumRuns = 1 };
 
   int64_t min_runtime = 0;
   int64_t total_runtime = 0;
 
   rewind(source_file_);
-  ASSERT_TRUE(source_file_ != NULL) <<
-      "Cannot read input file \n";
+  ASSERT_TRUE(source_file_ != NULL) << "Cannot read input file \n";
 
   // CA not needed here
-  vpm_->EnableContentAnalysis(false);
+  vp_->EnableContentAnalysis(false);
   // no temporal decimation
-  vpm_->EnableTemporalDecimation(false);
+  vp_->EnableTemporalDecimation(false);
 
   // Reading test frame
   rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
-                                 source_file_));
+  ASSERT_EQ(frame_length_,
+            fread(video_buffer.get(), 1, frame_length_, source_file_));
   // Using ConvertToI420 to add stride to the image.
   EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                              0, kVideoRotation_0, &video_frame_));
   // Cropped source frame that will contain the expected visible region.
-  I420VideoFrame cropped_source_frame;
+  VideoFrame cropped_source_frame;
   cropped_source_frame.CopyFrame(video_frame_);
 
   for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++) {
@@ -225,43 +255,43 @@ TEST_F(VideoProcessingModuleTest, Resampler) {
 
     // Test scaling to different sizes: source is of |width|/|height| = 352/288.
     // Pure scaling:
-    TestSize(video_frame_, video_frame_, width_ / 4, height_ / 4, 25.2, vpm_);
-    TestSize(video_frame_, video_frame_, width_ / 2, height_ / 2, 28.1, vpm_);
+    TestSize(video_frame_, video_frame_, width_ / 4, height_ / 4, 25.2, vp_);
+    TestSize(video_frame_, video_frame_, width_ / 2, height_ / 2, 28.1, vp_);
     // No resampling:
-    TestSize(video_frame_, video_frame_, width_, height_, -1, vpm_);
-    TestSize(video_frame_, video_frame_, 2 * width_, 2 * height_, 32.2, vpm_);
+    TestSize(video_frame_, video_frame_, width_, height_, -1, vp_);
+    TestSize(video_frame_, video_frame_, 2 * width_, 2 * height_, 32.2, vp_);
 
     // Scaling and cropping. The cropped source frame is the largest center
     // aligned region that can be used from the source while preserving aspect
     // ratio.
     CropFrame(video_buffer.get(), width_, height_, 0, 56, 352, 176,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 100, 50, 24.0, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 100, 50, 24.0, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 0, 30, 352, 225,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 400, 256, 31.3, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 400, 256, 31.3, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 68, 0, 216, 288,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 480, 640, 32.15, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 480, 640, 32.15, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 0, 12, 352, 264,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 960, 720, 32.2, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 960, 720, 32.2, vp_);
 
     CropFrame(video_buffer.get(), width_, height_, 0, 44, 352, 198,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 1280, 720, 32.15, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 1280, 720, 32.15, vp_);
 
     // Upsampling to odd size.
     CropFrame(video_buffer.get(), width_, height_, 0, 26, 352, 233,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 501, 333, 32.05, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 501, 333, 32.05, vp_);
     // Downsample to odd size.
     CropFrame(video_buffer.get(), width_, height_, 0, 34, 352, 219,
               &cropped_source_frame);
-    TestSize(video_frame_, cropped_source_frame, 281, 175, 29.3, vpm_);
+    TestSize(video_frame_, cropped_source_frame, 281, 175, 29.3, vp_);
 
     // Stop timer.
     const int64_t runtime = (TickTime::Now() - time_start).Microseconds();
@@ -273,30 +303,30 @@ TEST_F(VideoProcessingModuleTest, Resampler) {
 
   printf("\nAverage run time = %d us / frame\n",
          static_cast<int>(total_runtime));
-  printf("Min run time = %d us / frame\n\n",
-         static_cast<int>(min_runtime));
+  printf("Min run time = %d us / frame\n\n", static_cast<int>(min_runtime));
 }
 
-void PreprocessFrameAndVerify(const I420VideoFrame& source,
+void PreprocessFrameAndVerify(const VideoFrame& source,
                               int target_width,
                               int target_height,
-                              VideoProcessingModule* vpm,
-                              I420VideoFrame** out_frame) {
+                              VideoProcessing* vpm,
+                              const VideoFrame* out_frame) {
   ASSERT_EQ(VPM_OK, vpm->SetTargetResolution(target_width, target_height, 30));
-  ASSERT_EQ(VPM_OK, vpm->PreprocessFrame(source, out_frame));
+  out_frame = vpm->PreprocessFrame(source);
+  EXPECT_TRUE(out_frame != nullptr);
 
-  // If no resizing is needed, expect NULL.
+  // If no resizing is needed, expect the original frame.
   if (target_width == source.width() && target_height == source.height()) {
-    EXPECT_EQ(NULL, *out_frame);
+    EXPECT_EQ(&source, out_frame);
     return;
   }
 
   // Verify the resampled frame.
-  EXPECT_TRUE(*out_frame != NULL);
-  EXPECT_EQ(source.render_time_ms(), (*out_frame)->render_time_ms());
-  EXPECT_EQ(source.timestamp(), (*out_frame)->timestamp());
-  EXPECT_EQ(target_width, (*out_frame)->width());
-  EXPECT_EQ(target_height, (*out_frame)->height());
+  EXPECT_TRUE(out_frame != NULL);
+  EXPECT_EQ(source.render_time_ms(), (out_frame)->render_time_ms());
+  EXPECT_EQ(source.timestamp(), (out_frame)->timestamp());
+  EXPECT_EQ(target_width, (out_frame)->width());
+  EXPECT_EQ(target_height, (out_frame)->height());
 }
 
 void CropFrame(const uint8_t* source_data,
@@ -306,7 +336,7 @@ void CropFrame(const uint8_t* source_data,
                int offset_y,
                int cropped_width,
                int cropped_height,
-               I420VideoFrame* cropped_frame) {
+               VideoFrame* cropped_frame) {
   cropped_frame->CreateEmptyFrame(cropped_width, cropped_height, cropped_width,
                                   (cropped_width + 1) / 2,
                                   (cropped_width + 1) / 2);
@@ -315,40 +345,41 @@ void CropFrame(const uint8_t* source_data,
                           source_height, 0, kVideoRotation_0, cropped_frame));
 }
 
-void TestSize(const I420VideoFrame& source_frame,
-              const I420VideoFrame& cropped_source_frame,
+void TestSize(const VideoFrame& source_frame,
+              const VideoFrame& cropped_source_frame,
               int target_width,
               int target_height,
               double expected_psnr,
-              VideoProcessingModule* vpm) {
+              VideoProcessing* vpm) {
   // Resample source_frame to out_frame.
-  I420VideoFrame* out_frame = NULL;
+  VideoFrame* out_frame = NULL;
   vpm->SetInputFrameResampleMode(kBox);
   PreprocessFrameAndVerify(source_frame, target_width, target_height, vpm,
-                           &out_frame);
+                           out_frame);
   if (out_frame == NULL)
     return;
   WriteProcessedFrameForVisualInspection(source_frame, *out_frame);
 
   // Scale |resampled_source_frame| back to the source scale.
-  I420VideoFrame resampled_source_frame;
+  VideoFrame resampled_source_frame;
   resampled_source_frame.CopyFrame(*out_frame);
   PreprocessFrameAndVerify(resampled_source_frame, cropped_source_frame.width(),
-                           cropped_source_frame.height(), vpm, &out_frame);
+                           cropped_source_frame.height(), vpm, out_frame);
   WriteProcessedFrameForVisualInspection(resampled_source_frame, *out_frame);
 
   // Compute PSNR against the cropped source frame and check expectation.
   double psnr = I420PSNR(&cropped_source_frame, out_frame);
   EXPECT_GT(psnr, expected_psnr);
-  printf("PSNR: %f. PSNR is between source of size %d %d, and a modified "
-         "source which is scaled down/up to: %d %d, and back to source size \n",
-         psnr, source_frame.width(), source_frame.height(),
-         target_width, target_height);
+  printf(
+      "PSNR: %f. PSNR is between source of size %d %d, and a modified "
+      "source which is scaled down/up to: %d %d, and back to source size \n",
+      psnr, source_frame.width(), source_frame.height(), target_width,
+      target_height);
 }
 
-bool CompareFrames(const webrtc::I420VideoFrame& frame1,
-                   const webrtc::I420VideoFrame& frame2) {
-  for (int plane = 0; plane < webrtc::kNumOfPlanes; plane ++) {
+bool CompareFrames(const webrtc::VideoFrame& frame1,
+                   const webrtc::VideoFrame& frame2) {
+  for (int plane = 0; plane < webrtc::kNumOfPlanes; plane++) {
     webrtc::PlaneType plane_type = static_cast<webrtc::PlaneType>(plane);
     int allocated_size1 = frame1.allocated_size(plane_type);
     int allocated_size2 = frame2.allocated_size(plane_type);
@@ -362,8 +393,11 @@ bool CompareFrames(const webrtc::I420VideoFrame& frame1,
   return true;
 }
 
-void WriteProcessedFrameForVisualInspection(const I420VideoFrame& source,
-                                            const I420VideoFrame& processed) {
+void WriteProcessedFrameForVisualInspection(const VideoFrame& source,
+                                            const VideoFrame& processed) {
+  // Skip if writing to files is not enabled.
+  if (!FLAGS_gen_files)
+    return;
   // Write the processed frame to file for visual inspection.
   std::ostringstream filename;
   filename << webrtc::test::OutputPath() << "Resampler_from_" << source.width()
@@ -372,7 +406,7 @@ void WriteProcessedFrameForVisualInspection(const I420VideoFrame& source,
   std::cout << "Watch " << filename.str() << " and verify that it is okay."
             << std::endl;
   FILE* stand_alone_file = fopen(filename.str().c_str(), "wb");
-  if (PrintI420VideoFrame(processed, stand_alone_file) < 0)
+  if (PrintVideoFrame(processed, stand_alone_file) < 0)
     std::cerr << "Failed to write: " << filename.str() << std::endl;
   if (stand_alone_file)
     fclose(stand_alone_file);
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h b/media/webrtc/trunk/webrtc/modules/video_processing/test/video_processing_unittest.h
similarity index 61%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/test/video_processing_unittest.h
index 37e2c02b94..3433c6ca86 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/video_processing_unittest.h
@@ -8,19 +8,21 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VIDEO_PROCESSING_UNITTEST_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VIDEO_PROCESSING_UNITTEST_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_TEST_VIDEO_PROCESSING_UNITTEST_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_TEST_VIDEO_PROCESSING_UNITTEST_H_
+
+#include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
-class VideoProcessingModuleTest : public ::testing::Test {
+class VideoProcessingTest : public ::testing::Test {
  protected:
-  VideoProcessingModuleTest();
+  VideoProcessingTest();
   virtual void SetUp();
   virtual void TearDown();
   static void SetUpTestCase() {
@@ -28,12 +30,10 @@ class VideoProcessingModuleTest : public ::testing::Test {
     std::string trace_file = webrtc::test::OutputPath() + "VPMTrace.txt";
     ASSERT_EQ(0, Trace::SetTraceFile(trace_file.c_str()));
   }
-  static void TearDownTestCase() {
-    Trace::ReturnTrace();
-  }
-  VideoProcessingModule* vpm_;
+  static void TearDownTestCase() { Trace::ReturnTrace(); }
+  VideoProcessing* vp_;
   FILE* source_file_;
-  I420VideoFrame video_frame_;
+  VideoFrame video_frame_;
   const int width_;
   const int half_width_;
   const int height_;
@@ -44,4 +44,4 @@ class VideoProcessingModuleTest : public ::testing::Test {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VIDEO_PROCESSING_UNITTEST_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_TEST_VIDEO_PROCESSING_UNITTEST_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m b/media/webrtc/trunk/webrtc/modules/video_processing/test/writeYUV420file.m
similarity index 98%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m
rename to media/webrtc/trunk/webrtc/modules/video_processing/test/writeYUV420file.m
index 69a8808338..359445009b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/writeYUV420file.m
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/test/writeYUV420file.m
@@ -11,10 +11,10 @@ numFrames=size(Y,3);
 for k=1:numFrames
    % Write luminance
    fwrite(fid,uint8(Y(:,:,k).'), 'uchar');
-   
+
    % Write U channel
    fwrite(fid,uint8(U(:,:,k).'), 'uchar');
-   
+
    % Write V channel
    fwrite(fid,uint8(V(:,:,k).'), 'uchar');
 end
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter.cc b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter.cc
new file mode 100644
index 0000000000..fbc2435cb5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter_c.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter_neon.h"
+#include "webrtc/modules/video_processing/util/denoiser_filter_sse2.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+
+namespace webrtc {
+
+const int kMotionMagnitudeThreshold = 8 * 3;
+const int kSumDiffThreshold = 16 * 16 * 2;
+const int kSumDiffThresholdHigh = 600;
+
+rtc::scoped_ptr<DenoiserFilter> DenoiserFilter::Create(
+    bool runtime_cpu_detection) {
+  rtc::scoped_ptr<DenoiserFilter> filter;
+
+  if (runtime_cpu_detection) {
+// If we know the minimum architecture at compile time, avoid CPU detection.
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    // x86 CPU detection required.
+    if (WebRtc_GetCPUInfo(kSSE2)) {
+      filter.reset(new DenoiserFilterSSE2());
+    } else {
+      filter.reset(new DenoiserFilterC());
+    }
+#elif defined(WEBRTC_DETECT_NEON)
+    if (WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) {
+      filter.reset(new DenoiserFilterNEON());
+    } else {
+      filter.reset(new DenoiserFilterC());
+    }
+#else
+    filter.reset(new DenoiserFilterC());
+#endif
+  } else {
+    filter.reset(new DenoiserFilterC());
+  }
+
+  RTC_DCHECK(filter.get() != nullptr);
+  return filter;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter.h b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter.h
new file mode 100644
index 0000000000..5d5a61c59c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
+
+namespace webrtc {
+
+extern const int kMotionMagnitudeThreshold;
+extern const int kSumDiffThreshold;
+extern const int kSumDiffThresholdHigh;
+
+enum DenoiserDecision { COPY_BLOCK, FILTER_BLOCK };
+struct DenoiseMetrics {
+  uint32_t var;
+  uint32_t sad;
+  uint8_t denoise;
+  bool is_skin;
+};
+
+class DenoiserFilter {
+ public:
+  static rtc::scoped_ptr<DenoiserFilter> Create(bool runtime_cpu_detection);
+
+  virtual ~DenoiserFilter() {}
+
+  virtual void CopyMem16x16(const uint8_t* src,
+                            int src_stride,
+                            uint8_t* dst,
+                            int dst_stride) = 0;
+  virtual void CopyMem8x8(const uint8_t* src,
+                          int src_stride,
+                          uint8_t* dst,
+                          int dst_stride) = 0;
+  virtual uint32_t Variance16x8(const uint8_t* a,
+                                int a_stride,
+                                const uint8_t* b,
+                                int b_stride,
+                                unsigned int* sse) = 0;
+  virtual DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                                     int mc_avg_y_stride,
+                                     uint8_t* running_avg_y,
+                                     int avg_y_stride,
+                                     const uint8_t* sig,
+                                     int sig_stride,
+                                     uint8_t motion_magnitude,
+                                     int increase_denoising) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_c.cc b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_c.cc
new file mode 100644
index 0000000000..6323980e18
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_c.cc
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include "webrtc/modules/video_processing/util/denoiser_filter_c.h"
+
+namespace webrtc {
+
+void DenoiserFilterC::CopyMem16x16(const uint8_t* src,
+                                   int src_stride,
+                                   uint8_t* dst,
+                                   int dst_stride) {
+  for (int i = 0; i < 16; i++) {
+    memcpy(dst, src, 16);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+void DenoiserFilterC::CopyMem8x8(const uint8_t* src,
+                                 int src_stride,
+                                 uint8_t* dst,
+                                 int dst_stride) {
+  for (int i = 0; i < 8; i++) {
+    memcpy(dst, src, 8);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+uint32_t DenoiserFilterC::Variance16x8(const uint8_t* a,
+                                       int a_stride,
+                                       const uint8_t* b,
+                                       int b_stride,
+                                       uint32_t* sse) {
+  int sum = 0;
+  *sse = 0;
+  a_stride <<= 1;
+  b_stride <<= 1;
+
+  for (int i = 0; i < 8; i++) {
+    for (int j = 0; j < 16; j++) {
+      const int diff = a[j] - b[j];
+      sum += diff;
+      *sse += diff * diff;
+    }
+
+    a += a_stride;
+    b += b_stride;
+  }
+  return *sse - ((static_cast<int64_t>(sum) * sum) >> 7);
+}
+
+DenoiserDecision DenoiserFilterC::MbDenoise(uint8_t* mc_running_avg_y,
+                                            int mc_avg_y_stride,
+                                            uint8_t* running_avg_y,
+                                            int avg_y_stride,
+                                            const uint8_t* sig,
+                                            int sig_stride,
+                                            uint8_t motion_magnitude,
+                                            int increase_denoising) {
+  int sum_diff_thresh = 0;
+  int sum_diff = 0;
+  int adj_val[3] = {3, 4, 6};
+  int shift_inc1 = 0;
+  int shift_inc2 = 1;
+  int col_sum[16] = {0};
+  if (motion_magnitude <= kMotionMagnitudeThreshold) {
+    if (increase_denoising) {
+      shift_inc1 = 1;
+      shift_inc2 = 2;
+    }
+    adj_val[0] += shift_inc2;
+    adj_val[1] += shift_inc2;
+    adj_val[2] += shift_inc2;
+  }
+
+  for (int r = 0; r < 16; ++r) {
+    for (int c = 0; c < 16; ++c) {
+      int diff = 0;
+      int adjustment = 0;
+      int absdiff = 0;
+
+      diff = mc_running_avg_y[c] - sig[c];
+      absdiff = abs(diff);
+
+      // When |diff| <= |3 + shift_inc1|, use pixel value from
+      // last denoised raw.
+      if (absdiff <= 3 + shift_inc1) {
+        running_avg_y[c] = mc_running_avg_y[c];
+        col_sum[c] += diff;
+      } else {
+        if (absdiff >= 4 + shift_inc1 && absdiff <= 7)
+          adjustment = adj_val[0];
+        else if (absdiff >= 8 && absdiff <= 15)
+          adjustment = adj_val[1];
+        else
+          adjustment = adj_val[2];
+
+        if (diff > 0) {
+          if ((sig[c] + adjustment) > 255)
+            running_avg_y[c] = 255;
+          else
+            running_avg_y[c] = sig[c] + adjustment;
+
+          col_sum[c] += adjustment;
+        } else {
+          if ((sig[c] - adjustment) < 0)
+            running_avg_y[c] = 0;
+          else
+            running_avg_y[c] = sig[c] - adjustment;
+
+          col_sum[c] -= adjustment;
+        }
+      }
+    }
+
+    // Update pointers for next iteration.
+    sig += sig_stride;
+    mc_running_avg_y += mc_avg_y_stride;
+    running_avg_y += avg_y_stride;
+  }
+
+  for (int c = 0; c < 16; ++c) {
+    if (col_sum[c] >= 128) {
+      col_sum[c] = 127;
+    }
+    sum_diff += col_sum[c];
+  }
+
+  sum_diff_thresh = kSumDiffThreshold;
+  if (increase_denoising)
+    sum_diff_thresh = kSumDiffThresholdHigh;
+  if (abs(sum_diff) > sum_diff_thresh) {
+    int delta = ((abs(sum_diff) - sum_diff_thresh) >> 8) + 1;
+    // Only apply the adjustment for max delta up to 3.
+    if (delta < 4) {
+      sig -= sig_stride * 16;
+      mc_running_avg_y -= mc_avg_y_stride * 16;
+      running_avg_y -= avg_y_stride * 16;
+      for (int r = 0; r < 16; ++r) {
+        for (int c = 0; c < 16; ++c) {
+          int diff = mc_running_avg_y[c] - sig[c];
+          int adjustment = abs(diff);
+          if (adjustment > delta)
+            adjustment = delta;
+          if (diff > 0) {
+            // Bring denoised signal down.
+            if (running_avg_y[c] - adjustment < 0)
+              running_avg_y[c] = 0;
+            else
+              running_avg_y[c] = running_avg_y[c] - adjustment;
+            col_sum[c] -= adjustment;
+          } else if (diff < 0) {
+            // Bring denoised signal up.
+            if (running_avg_y[c] + adjustment > 255)
+              running_avg_y[c] = 255;
+            else
+              running_avg_y[c] = running_avg_y[c] + adjustment;
+            col_sum[c] += adjustment;
+          }
+        }
+        sig += sig_stride;
+        mc_running_avg_y += mc_avg_y_stride;
+        running_avg_y += avg_y_stride;
+      }
+
+      sum_diff = 0;
+      for (int c = 0; c < 16; ++c) {
+        if (col_sum[c] >= 128) {
+          col_sum[c] = 127;
+        }
+        sum_diff += col_sum[c];
+      }
+
+      if (abs(sum_diff) > sum_diff_thresh)
+        return COPY_BLOCK;
+    } else {
+      return COPY_BLOCK;
+    }
+  }
+
+  return FILTER_BLOCK;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_c.h b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_c.h
new file mode 100644
index 0000000000..fe46ac38ec
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_c.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_C_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_C_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+
+namespace webrtc {
+
+class DenoiserFilterC : public DenoiserFilter {
+ public:
+  DenoiserFilterC() {}
+  void CopyMem16x16(const uint8_t* src,
+                    int src_stride,
+                    uint8_t* dst,
+                    int dst_stride) override;
+  void CopyMem8x8(const uint8_t* src,
+                  int src_stride,
+                  uint8_t* dst,
+                  int dst_stride) override;
+  uint32_t Variance16x8(const uint8_t* a,
+                        int a_stride,
+                        const uint8_t* b,
+                        int b_stride,
+                        unsigned int* sse) override;
+  DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                             int mc_avg_y_stride,
+                             uint8_t* running_avg_y,
+                             int avg_y_stride,
+                             const uint8_t* sig,
+                             int sig_stride,
+                             uint8_t motion_magnitude,
+                             int increase_denoising) override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_C_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_neon.cc b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
new file mode 100644
index 0000000000..b522bf002b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <arm_neon.h>
+
+#include "webrtc/modules/video_processing/util/denoiser_filter_neon.h"
+
+namespace webrtc {
+
+static int HorizontalAddS16x8(const int16x8_t v_16x8) {
+  const int32x4_t a = vpaddlq_s16(v_16x8);
+  const int64x2_t b = vpaddlq_s32(a);
+  const int32x2_t c = vadd_s32(vreinterpret_s32_s64(vget_low_s64(b)),
+                               vreinterpret_s32_s64(vget_high_s64(b)));
+  return vget_lane_s32(c, 0);
+}
+
+static int HorizontalAddS32x4(const int32x4_t v_32x4) {
+  const int64x2_t b = vpaddlq_s32(v_32x4);
+  const int32x2_t c = vadd_s32(vreinterpret_s32_s64(vget_low_s64(b)),
+                               vreinterpret_s32_s64(vget_high_s64(b)));
+  return vget_lane_s32(c, 0);
+}
+
+static void VarianceNeonW8(const uint8_t* a,
+                           int a_stride,
+                           const uint8_t* b,
+                           int b_stride,
+                           int w,
+                           int h,
+                           uint32_t* sse,
+                           int64_t* sum) {
+  int16x8_t v_sum = vdupq_n_s16(0);
+  int32x4_t v_sse_lo = vdupq_n_s32(0);
+  int32x4_t v_sse_hi = vdupq_n_s32(0);
+
+  for (int i = 0; i < h; ++i) {
+    for (int j = 0; j < w; j += 8) {
+      const uint8x8_t v_a = vld1_u8(&a[j]);
+      const uint8x8_t v_b = vld1_u8(&b[j]);
+      const uint16x8_t v_diff = vsubl_u8(v_a, v_b);
+      const int16x8_t sv_diff = vreinterpretq_s16_u16(v_diff);
+      v_sum = vaddq_s16(v_sum, sv_diff);
+      v_sse_lo =
+          vmlal_s16(v_sse_lo, vget_low_s16(sv_diff), vget_low_s16(sv_diff));
+      v_sse_hi =
+          vmlal_s16(v_sse_hi, vget_high_s16(sv_diff), vget_high_s16(sv_diff));
+    }
+    a += a_stride;
+    b += b_stride;
+  }
+
+  *sum = HorizontalAddS16x8(v_sum);
+  *sse =
+      static_cast<uint32_t>(HorizontalAddS32x4(vaddq_s32(v_sse_lo, v_sse_hi)));
+}
+
+void DenoiserFilterNEON::CopyMem16x16(const uint8_t* src,
+                                      int src_stride,
+                                      uint8_t* dst,
+                                      int dst_stride) {
+  uint8x16_t qtmp;
+  for (int r = 0; r < 16; r++) {
+    qtmp = vld1q_u8(src);
+    vst1q_u8(dst, qtmp);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+void DenoiserFilterNEON::CopyMem8x8(const uint8_t* src,
+                                    int src_stride,
+                                    uint8_t* dst,
+                                    int dst_stride) {
+  uint8x8_t vtmp;
+
+  for (int r = 0; r < 8; r++) {
+    vtmp = vld1_u8(src);
+    vst1_u8(dst, vtmp);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+uint32_t DenoiserFilterNEON::Variance16x8(const uint8_t* a,
+                                          int a_stride,
+                                          const uint8_t* b,
+                                          int b_stride,
+                                          uint32_t* sse) {
+  int64_t sum = 0;
+  VarianceNeonW8(a, a_stride << 1, b, b_stride << 1, 16, 8, sse, &sum);
+  return *sse - ((sum * sum) >> 7);
+}
+
+DenoiserDecision DenoiserFilterNEON::MbDenoise(uint8_t* mc_running_avg_y,
+                                               int mc_running_avg_y_stride,
+                                               uint8_t* running_avg_y,
+                                               int running_avg_y_stride,
+                                               const uint8_t* sig,
+                                               int sig_stride,
+                                               uint8_t motion_magnitude,
+                                               int increase_denoising) {
+  // If motion_magnitude is small, making the denoiser more aggressive by
+  // increasing the adjustment for each level, level1 adjustment is
+  // increased, the deltas stay the same.
+  int shift_inc =
+      (increase_denoising && motion_magnitude <= kMotionMagnitudeThreshold) ? 1
+                                                                            : 0;
+  const uint8x16_t v_level1_adjustment = vmovq_n_u8(
+      (motion_magnitude <= kMotionMagnitudeThreshold) ? 4 + shift_inc : 3);
+  const uint8x16_t v_delta_level_1_and_2 = vdupq_n_u8(1);
+  const uint8x16_t v_delta_level_2_and_3 = vdupq_n_u8(2);
+  const uint8x16_t v_level1_threshold = vmovq_n_u8(4 + shift_inc);
+  const uint8x16_t v_level2_threshold = vdupq_n_u8(8);
+  const uint8x16_t v_level3_threshold = vdupq_n_u8(16);
+  int64x2_t v_sum_diff_total = vdupq_n_s64(0);
+
+  // Go over lines.
+  for (int r = 0; r < 16; ++r) {
+    // Load inputs.
+    const uint8x16_t v_sig = vld1q_u8(sig);
+    const uint8x16_t v_mc_running_avg_y = vld1q_u8(mc_running_avg_y);
+
+    // Calculate absolute difference and sign masks.
+    const uint8x16_t v_abs_diff = vabdq_u8(v_sig, v_mc_running_avg_y);
+    const uint8x16_t v_diff_pos_mask = vcltq_u8(v_sig, v_mc_running_avg_y);
+    const uint8x16_t v_diff_neg_mask = vcgtq_u8(v_sig, v_mc_running_avg_y);
+
+    // Figure out which level that put us in.
+    const uint8x16_t v_level1_mask = vcleq_u8(v_level1_threshold, v_abs_diff);
+    const uint8x16_t v_level2_mask = vcleq_u8(v_level2_threshold, v_abs_diff);
+    const uint8x16_t v_level3_mask = vcleq_u8(v_level3_threshold, v_abs_diff);
+
+    // Calculate absolute adjustments for level 1, 2 and 3.
+    const uint8x16_t v_level2_adjustment =
+        vandq_u8(v_level2_mask, v_delta_level_1_and_2);
+    const uint8x16_t v_level3_adjustment =
+        vandq_u8(v_level3_mask, v_delta_level_2_and_3);
+    const uint8x16_t v_level1and2_adjustment =
+        vaddq_u8(v_level1_adjustment, v_level2_adjustment);
+    const uint8x16_t v_level1and2and3_adjustment =
+        vaddq_u8(v_level1and2_adjustment, v_level3_adjustment);
+
+    // Figure adjustment absolute value by selecting between the absolute
+    // difference if in level0 or the value for level 1, 2 and 3.
+    const uint8x16_t v_abs_adjustment =
+        vbslq_u8(v_level1_mask, v_level1and2and3_adjustment, v_abs_diff);
+
+    // Calculate positive and negative adjustments. Apply them to the signal
+    // and accumulate them. Adjustments are less than eight and the maximum
+    // sum of them (7 * 16) can fit in a signed char.
+    const uint8x16_t v_pos_adjustment =
+        vandq_u8(v_diff_pos_mask, v_abs_adjustment);
+    const uint8x16_t v_neg_adjustment =
+        vandq_u8(v_diff_neg_mask, v_abs_adjustment);
+
+    uint8x16_t v_running_avg_y = vqaddq_u8(v_sig, v_pos_adjustment);
+    v_running_avg_y = vqsubq_u8(v_running_avg_y, v_neg_adjustment);
+
+    // Store results.
+    vst1q_u8(running_avg_y, v_running_avg_y);
+
+    // Sum all the accumulators to have the sum of all pixel differences
+    // for this macroblock.
+    {
+      const int8x16_t v_sum_diff =
+          vqsubq_s8(vreinterpretq_s8_u8(v_pos_adjustment),
+                    vreinterpretq_s8_u8(v_neg_adjustment));
+      const int16x8_t fe_dc_ba_98_76_54_32_10 = vpaddlq_s8(v_sum_diff);
+      const int32x4_t fedc_ba98_7654_3210 =
+          vpaddlq_s16(fe_dc_ba_98_76_54_32_10);
+      const int64x2_t fedcba98_76543210 = vpaddlq_s32(fedc_ba98_7654_3210);
+
+      v_sum_diff_total = vqaddq_s64(v_sum_diff_total, fedcba98_76543210);
+    }
+
+    // Update pointers for next iteration.
+    sig += sig_stride;
+    mc_running_avg_y += mc_running_avg_y_stride;
+    running_avg_y += running_avg_y_stride;
+  }
+
+  // Too much adjustments => copy block.
+  {
+    int64x1_t x = vqadd_s64(vget_high_s64(v_sum_diff_total),
+                            vget_low_s64(v_sum_diff_total));
+    int sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
+    int sum_diff_thresh = kSumDiffThreshold;
+
+    if (increase_denoising)
+      sum_diff_thresh = kSumDiffThresholdHigh;
+    if (sum_diff > sum_diff_thresh) {
+      // Before returning to copy the block (i.e., apply no denoising),
+      // checK if we can still apply some (weaker) temporal filtering to
+      // this block, that would otherwise not be denoised at all. Simplest
+      // is to apply an additional adjustment to running_avg_y to bring it
+      // closer to sig. The adjustment is capped by a maximum delta, and
+      // chosen such that in most cases the resulting sum_diff will be
+      // within the accceptable range given by sum_diff_thresh.
+
+      // The delta is set by the excess of absolute pixel diff over the
+      // threshold.
+      int delta = ((sum_diff - sum_diff_thresh) >> 8) + 1;
+      // Only apply the adjustment for max delta up to 3.
+      if (delta < 4) {
+        const uint8x16_t k_delta = vmovq_n_u8(delta);
+        sig -= sig_stride * 16;
+        mc_running_avg_y -= mc_running_avg_y_stride * 16;
+        running_avg_y -= running_avg_y_stride * 16;
+        for (int r = 0; r < 16; ++r) {
+          uint8x16_t v_running_avg_y = vld1q_u8(running_avg_y);
+          const uint8x16_t v_sig = vld1q_u8(sig);
+          const uint8x16_t v_mc_running_avg_y = vld1q_u8(mc_running_avg_y);
+
+          // Calculate absolute difference and sign masks.
+          const uint8x16_t v_abs_diff = vabdq_u8(v_sig, v_mc_running_avg_y);
+          const uint8x16_t v_diff_pos_mask =
+              vcltq_u8(v_sig, v_mc_running_avg_y);
+          const uint8x16_t v_diff_neg_mask =
+              vcgtq_u8(v_sig, v_mc_running_avg_y);
+          // Clamp absolute difference to delta to get the adjustment.
+          const uint8x16_t v_abs_adjustment = vminq_u8(v_abs_diff, (k_delta));
+
+          const uint8x16_t v_pos_adjustment =
+              vandq_u8(v_diff_pos_mask, v_abs_adjustment);
+          const uint8x16_t v_neg_adjustment =
+              vandq_u8(v_diff_neg_mask, v_abs_adjustment);
+
+          v_running_avg_y = vqsubq_u8(v_running_avg_y, v_pos_adjustment);
+          v_running_avg_y = vqaddq_u8(v_running_avg_y, v_neg_adjustment);
+
+          // Store results.
+          vst1q_u8(running_avg_y, v_running_avg_y);
+
+          {
+            const int8x16_t v_sum_diff =
+                vqsubq_s8(vreinterpretq_s8_u8(v_neg_adjustment),
+                          vreinterpretq_s8_u8(v_pos_adjustment));
+
+            const int16x8_t fe_dc_ba_98_76_54_32_10 = vpaddlq_s8(v_sum_diff);
+            const int32x4_t fedc_ba98_7654_3210 =
+                vpaddlq_s16(fe_dc_ba_98_76_54_32_10);
+            const int64x2_t fedcba98_76543210 =
+                vpaddlq_s32(fedc_ba98_7654_3210);
+
+            v_sum_diff_total = vqaddq_s64(v_sum_diff_total, fedcba98_76543210);
+          }
+          // Update pointers for next iteration.
+          sig += sig_stride;
+          mc_running_avg_y += mc_running_avg_y_stride;
+          running_avg_y += running_avg_y_stride;
+        }
+        {
+          // Update the sum of all pixel differences of this MB.
+          x = vqadd_s64(vget_high_s64(v_sum_diff_total),
+                        vget_low_s64(v_sum_diff_total));
+          sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
+
+          if (sum_diff > sum_diff_thresh) {
+            return COPY_BLOCK;
+          }
+        }
+      } else {
+        return COPY_BLOCK;
+      }
+    }
+  }
+
+  // Tell above level that block was filtered.
+  running_avg_y -= running_avg_y_stride * 16;
+  sig -= sig_stride * 16;
+
+  return FILTER_BLOCK;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_neon.h b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_neon.h
new file mode 100644
index 0000000000..bc87ba788e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_neon.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_NEON_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_NEON_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+
+namespace webrtc {
+
+class DenoiserFilterNEON : public DenoiserFilter {
+ public:
+  DenoiserFilterNEON() {}
+  void CopyMem16x16(const uint8_t* src,
+                    int src_stride,
+                    uint8_t* dst,
+                    int dst_stride) override;
+  void CopyMem8x8(const uint8_t* src,
+                  int src_stride,
+                  uint8_t* dst,
+                  int dst_stride) override;
+  uint32_t Variance16x8(const uint8_t* a,
+                        int a_stride,
+                        const uint8_t* b,
+                        int b_stride,
+                        unsigned int* sse) override;
+  DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                             int mc_avg_y_stride,
+                             uint8_t* running_avg_y,
+                             int avg_y_stride,
+                             const uint8_t* sig,
+                             int sig_stride,
+                             uint8_t motion_magnitude,
+                             int increase_denoising) override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_NEON_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
new file mode 100644
index 0000000000..903d7b1ec6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
@@ -0,0 +1,280 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <emmintrin.h>
+
+#include "webrtc/modules/video_processing/util/denoiser_filter_sse2.h"
+
+namespace webrtc {
+
+static void Get8x8varSse2(const uint8_t* src,
+                          int src_stride,
+                          const uint8_t* ref,
+                          int ref_stride,
+                          unsigned int* sse,
+                          int* sum) {
+  const __m128i zero = _mm_setzero_si128();
+  __m128i vsum = _mm_setzero_si128();
+  __m128i vsse = _mm_setzero_si128();
+
+  for (int i = 0; i < 8; i += 2) {
+    const __m128i src0 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(src + i * src_stride)), zero);
+    const __m128i ref0 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(ref + i * ref_stride)), zero);
+    const __m128i diff0 = _mm_sub_epi16(src0, ref0);
+
+    const __m128i src1 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(src + (i + 1) * src_stride)), zero);
+    const __m128i ref1 = _mm_unpacklo_epi8(
+        _mm_loadl_epi64((const __m128i*)(ref + (i + 1) * ref_stride)), zero);
+    const __m128i diff1 = _mm_sub_epi16(src1, ref1);
+
+    vsum = _mm_add_epi16(vsum, diff0);
+    vsum = _mm_add_epi16(vsum, diff1);
+    vsse = _mm_add_epi32(vsse, _mm_madd_epi16(diff0, diff0));
+    vsse = _mm_add_epi32(vsse, _mm_madd_epi16(diff1, diff1));
+  }
+
+  // sum
+  vsum = _mm_add_epi16(vsum, _mm_srli_si128(vsum, 8));
+  vsum = _mm_add_epi16(vsum, _mm_srli_si128(vsum, 4));
+  vsum = _mm_add_epi16(vsum, _mm_srli_si128(vsum, 2));
+  *sum = static_cast<int16_t>(_mm_extract_epi16(vsum, 0));
+
+  // sse
+  vsse = _mm_add_epi32(vsse, _mm_srli_si128(vsse, 8));
+  vsse = _mm_add_epi32(vsse, _mm_srli_si128(vsse, 4));
+  *sse = _mm_cvtsi128_si32(vsse);
+}
+
+static void VarianceSSE2(const unsigned char* src,
+                         int src_stride,
+                         const unsigned char* ref,
+                         int ref_stride,
+                         int w,
+                         int h,
+                         uint32_t* sse,
+                         int64_t* sum,
+                         int block_size) {
+  *sse = 0;
+  *sum = 0;
+
+  for (int i = 0; i < h; i += block_size) {
+    for (int j = 0; j < w; j += block_size) {
+      uint32_t sse0 = 0;
+      int32_t sum0 = 0;
+
+      Get8x8varSse2(src + src_stride * i + j, src_stride,
+                    ref + ref_stride * i + j, ref_stride, &sse0, &sum0);
+      *sse += sse0;
+      *sum += sum0;
+    }
+  }
+}
+
+// Compute the sum of all pixel differences of this MB.
+static uint32_t AbsSumDiff16x1(__m128i acc_diff) {
+  const __m128i k_1 = _mm_set1_epi16(1);
+  const __m128i acc_diff_lo =
+      _mm_srai_epi16(_mm_unpacklo_epi8(acc_diff, acc_diff), 8);
+  const __m128i acc_diff_hi =
+      _mm_srai_epi16(_mm_unpackhi_epi8(acc_diff, acc_diff), 8);
+  const __m128i acc_diff_16 = _mm_add_epi16(acc_diff_lo, acc_diff_hi);
+  const __m128i hg_fe_dc_ba = _mm_madd_epi16(acc_diff_16, k_1);
+  const __m128i hgfe_dcba =
+      _mm_add_epi32(hg_fe_dc_ba, _mm_srli_si128(hg_fe_dc_ba, 8));
+  const __m128i hgfedcba =
+      _mm_add_epi32(hgfe_dcba, _mm_srli_si128(hgfe_dcba, 4));
+  unsigned int sum_diff = abs(_mm_cvtsi128_si32(hgfedcba));
+
+  return sum_diff;
+}
+
+// TODO(jackychen): Optimize this function using SSE2.
+void DenoiserFilterSSE2::CopyMem16x16(const uint8_t* src,
+                                      int src_stride,
+                                      uint8_t* dst,
+                                      int dst_stride) {
+  for (int i = 0; i < 16; i++) {
+    memcpy(dst, src, 16);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+// TODO(jackychen): Optimize this function using SSE2.
+void DenoiserFilterSSE2::CopyMem8x8(const uint8_t* src,
+                                    int src_stride,
+                                    uint8_t* dst,
+                                    int dst_stride) {
+  for (int i = 0; i < 8; i++) {
+    memcpy(dst, src, 8);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+uint32_t DenoiserFilterSSE2::Variance16x8(const uint8_t* src,
+                                          int src_stride,
+                                          const uint8_t* ref,
+                                          int ref_stride,
+                                          uint32_t* sse) {
+  int64_t sum = 0;
+  VarianceSSE2(src, src_stride << 1, ref, ref_stride << 1, 16, 8, sse, &sum, 8);
+  return *sse - ((sum * sum) >> 7);
+}
+
+DenoiserDecision DenoiserFilterSSE2::MbDenoise(uint8_t* mc_running_avg_y,
+                                               int mc_avg_y_stride,
+                                               uint8_t* running_avg_y,
+                                               int avg_y_stride,
+                                               const uint8_t* sig,
+                                               int sig_stride,
+                                               uint8_t motion_magnitude,
+                                               int increase_denoising) {
+  int shift_inc =
+      (increase_denoising && motion_magnitude <= kMotionMagnitudeThreshold) ? 1
+                                                                            : 0;
+  __m128i acc_diff = _mm_setzero_si128();
+  const __m128i k_0 = _mm_setzero_si128();
+  const __m128i k_4 = _mm_set1_epi8(4 + shift_inc);
+  const __m128i k_8 = _mm_set1_epi8(8);
+  const __m128i k_16 = _mm_set1_epi8(16);
+  // Modify each level's adjustment according to motion_magnitude.
+  const __m128i l3 = _mm_set1_epi8(
+      (motion_magnitude <= kMotionMagnitudeThreshold) ? 7 + shift_inc : 6);
+  // Difference between level 3 and level 2 is 2.
+  const __m128i l32 = _mm_set1_epi8(2);
+  // Difference between level 2 and level 1 is 1.
+  const __m128i l21 = _mm_set1_epi8(1);
+
+  for (int r = 0; r < 16; ++r) {
+    // Calculate differences.
+    const __m128i v_sig =
+        _mm_loadu_si128(reinterpret_cast<const __m128i*>(&sig[0]));
+    const __m128i v_mc_running_avg_y =
+        _mm_loadu_si128(reinterpret_cast<__m128i*>(&mc_running_avg_y[0]));
+    __m128i v_running_avg_y;
+    const __m128i pdiff = _mm_subs_epu8(v_mc_running_avg_y, v_sig);
+    const __m128i ndiff = _mm_subs_epu8(v_sig, v_mc_running_avg_y);
+    // Obtain the sign. FF if diff is negative.
+    const __m128i diff_sign = _mm_cmpeq_epi8(pdiff, k_0);
+    // Clamp absolute difference to 16 to be used to get mask. Doing this
+    // allows us to use _mm_cmpgt_epi8, which operates on signed byte.
+    const __m128i clamped_absdiff =
+        _mm_min_epu8(_mm_or_si128(pdiff, ndiff), k_16);
+    // Get masks for l2 l1 and l0 adjustments.
+    const __m128i mask2 = _mm_cmpgt_epi8(k_16, clamped_absdiff);
+    const __m128i mask1 = _mm_cmpgt_epi8(k_8, clamped_absdiff);
+    const __m128i mask0 = _mm_cmpgt_epi8(k_4, clamped_absdiff);
+    // Get adjustments for l2, l1, and l0.
+    __m128i adj2 = _mm_and_si128(mask2, l32);
+    const __m128i adj1 = _mm_and_si128(mask1, l21);
+    const __m128i adj0 = _mm_and_si128(mask0, clamped_absdiff);
+    __m128i adj, padj, nadj;
+
+    // Combine the adjustments and get absolute adjustments.
+    adj2 = _mm_add_epi8(adj2, adj1);
+    adj = _mm_sub_epi8(l3, adj2);
+    adj = _mm_andnot_si128(mask0, adj);
+    adj = _mm_or_si128(adj, adj0);
+
+    // Restore the sign and get positive and negative adjustments.
+    padj = _mm_andnot_si128(diff_sign, adj);
+    nadj = _mm_and_si128(diff_sign, adj);
+
+    // Calculate filtered value.
+    v_running_avg_y = _mm_adds_epu8(v_sig, padj);
+    v_running_avg_y = _mm_subs_epu8(v_running_avg_y, nadj);
+    _mm_storeu_si128(reinterpret_cast<__m128i*>(running_avg_y),
+                     v_running_avg_y);
+
+    // Adjustments <=7, and each element in acc_diff can fit in signed
+    // char.
+    acc_diff = _mm_adds_epi8(acc_diff, padj);
+    acc_diff = _mm_subs_epi8(acc_diff, nadj);
+
+    // Update pointers for next iteration.
+    sig += sig_stride;
+    mc_running_avg_y += mc_avg_y_stride;
+    running_avg_y += avg_y_stride;
+  }
+
+  {
+    // Compute the sum of all pixel differences of this MB.
+    unsigned int abs_sum_diff = AbsSumDiff16x1(acc_diff);
+    unsigned int sum_diff_thresh = kSumDiffThreshold;
+    if (increase_denoising)
+      sum_diff_thresh = kSumDiffThresholdHigh;
+    if (abs_sum_diff > sum_diff_thresh) {
+      // Before returning to copy the block (i.e., apply no denoising),
+      // check if we can still apply some (weaker) temporal filtering to
+      // this block, that would otherwise not be denoised at all. Simplest
+      // is to apply an additional adjustment to running_avg_y to bring it
+      // closer to sig. The adjustment is capped by a maximum delta, and
+      // chosen such that in most cases the resulting sum_diff will be
+      // within the acceptable range given by sum_diff_thresh.
+
+      // The delta is set by the excess of absolute pixel diff over the
+      // threshold.
+      int delta = ((abs_sum_diff - sum_diff_thresh) >> 8) + 1;
+      // Only apply the adjustment for max delta up to 3.
+      if (delta < 4) {
+        const __m128i k_delta = _mm_set1_epi8(delta);
+        sig -= sig_stride * 16;
+        mc_running_avg_y -= mc_avg_y_stride * 16;
+        running_avg_y -= avg_y_stride * 16;
+        for (int r = 0; r < 16; ++r) {
+          __m128i v_running_avg_y =
+              _mm_loadu_si128(reinterpret_cast<__m128i*>(&running_avg_y[0]));
+          // Calculate differences.
+          const __m128i v_sig =
+              _mm_loadu_si128(reinterpret_cast<const __m128i*>(&sig[0]));
+          const __m128i v_mc_running_avg_y =
+              _mm_loadu_si128(reinterpret_cast<__m128i*>(&mc_running_avg_y[0]));
+          const __m128i pdiff = _mm_subs_epu8(v_mc_running_avg_y, v_sig);
+          const __m128i ndiff = _mm_subs_epu8(v_sig, v_mc_running_avg_y);
+          // Obtain the sign. FF if diff is negative.
+          const __m128i diff_sign = _mm_cmpeq_epi8(pdiff, k_0);
+          // Clamp absolute difference to delta to get the adjustment.
+          const __m128i adj = _mm_min_epu8(_mm_or_si128(pdiff, ndiff), k_delta);
+          // Restore the sign and get positive and negative adjustments.
+          __m128i padj, nadj;
+          padj = _mm_andnot_si128(diff_sign, adj);
+          nadj = _mm_and_si128(diff_sign, adj);
+          // Calculate filtered value.
+          v_running_avg_y = _mm_subs_epu8(v_running_avg_y, padj);
+          v_running_avg_y = _mm_adds_epu8(v_running_avg_y, nadj);
+          _mm_storeu_si128(reinterpret_cast<__m128i*>(running_avg_y),
+                           v_running_avg_y);
+
+          // Accumulate the adjustments.
+          acc_diff = _mm_subs_epi8(acc_diff, padj);
+          acc_diff = _mm_adds_epi8(acc_diff, nadj);
+
+          // Update pointers for next iteration.
+          sig += sig_stride;
+          mc_running_avg_y += mc_avg_y_stride;
+          running_avg_y += avg_y_stride;
+        }
+        abs_sum_diff = AbsSumDiff16x1(acc_diff);
+        if (abs_sum_diff > sum_diff_thresh) {
+          return COPY_BLOCK;
+        }
+      } else {
+        return COPY_BLOCK;
+      }
+    }
+  }
+  return FILTER_BLOCK;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_sse2.h b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
new file mode 100644
index 0000000000..31d8510902
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_SSE2_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_SSE2_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+
+namespace webrtc {
+
+class DenoiserFilterSSE2 : public DenoiserFilter {
+ public:
+  DenoiserFilterSSE2() {}
+  void CopyMem16x16(const uint8_t* src,
+                    int src_stride,
+                    uint8_t* dst,
+                    int dst_stride) override;
+  void CopyMem8x8(const uint8_t* src,
+                  int src_stride,
+                  uint8_t* dst,
+                  int dst_stride) override;
+  uint32_t Variance16x8(const uint8_t* a,
+                        int a_stride,
+                        const uint8_t* b,
+                        int b_stride,
+                        unsigned int* sse) override;
+  DenoiserDecision MbDenoise(uint8_t* mc_running_avg_y,
+                             int mc_avg_y_stride,
+                             uint8_t* running_avg_y,
+                             int avg_y_stride,
+                             const uint8_t* sig,
+                             int sig_stride,
+                             uint8_t motion_magnitude,
+                             int increase_denoising) override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_DENOISER_FILTER_SSE2_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/skin_detection.cc b/media/webrtc/trunk/webrtc/modules/video_processing/util/skin_detection.cc
new file mode 100644
index 0000000000..bf631ce2f6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/skin_detection.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <limits.h>
+#include <math.h>
+
+#include "webrtc/modules/video_processing/util/skin_detection.h"
+
+namespace webrtc {
+
+// Fixed-point skin color model parameters.
+static const int skin_mean[2] = {7463, 9614};                 // q6
+static const int skin_inv_cov[4] = {4107, 1663, 1663, 2157};  // q16
+static const int skin_threshold = 1570636;                    // q18
+
+// Thresholds on luminance.
+static const int y_low = 20;
+static const int y_high = 220;
+
+// Evaluates the Mahalanobis distance measure for the input CbCr values.
+static int EvaluateSkinColorDifference(int cb, int cr) {
+  const int cb_q6 = cb << 6;
+  const int cr_q6 = cr << 6;
+  const int cb_diff_q12 = (cb_q6 - skin_mean[0]) * (cb_q6 - skin_mean[0]);
+  const int cbcr_diff_q12 = (cb_q6 - skin_mean[0]) * (cr_q6 - skin_mean[1]);
+  const int cr_diff_q12 = (cr_q6 - skin_mean[1]) * (cr_q6 - skin_mean[1]);
+  const int cb_diff_q2 = (cb_diff_q12 + (1 << 9)) >> 10;
+  const int cbcr_diff_q2 = (cbcr_diff_q12 + (1 << 9)) >> 10;
+  const int cr_diff_q2 = (cr_diff_q12 + (1 << 9)) >> 10;
+  const int skin_diff =
+      skin_inv_cov[0] * cb_diff_q2 + skin_inv_cov[1] * cbcr_diff_q2 +
+      skin_inv_cov[2] * cbcr_diff_q2 + skin_inv_cov[3] * cr_diff_q2;
+  return skin_diff;
+}
+
+bool MbHasSkinColor(const uint8_t* y_src,
+                    const uint8_t* u_src,
+                    const uint8_t* v_src,
+                    const int stride_y,
+                    const int stride_u,
+                    const int stride_v,
+                    const int mb_row,
+                    const int mb_col) {
+  const uint8_t* y = y_src + ((mb_row << 4) + 8) * stride_y + (mb_col << 4) + 8;
+  const uint8_t* u = u_src + ((mb_row << 3) + 4) * stride_u + (mb_col << 3) + 4;
+  const uint8_t* v = v_src + ((mb_row << 3) + 4) * stride_v + (mb_col << 3) + 4;
+  // Use 2x2 average of center pixel to compute skin area.
+  uint8_t y_avg = (*y + *(y + 1) + *(y + stride_y) + *(y + stride_y + 1)) >> 2;
+  uint8_t u_avg = (*u + *(u + 1) + *(u + stride_u) + *(u + stride_u + 1)) >> 2;
+  uint8_t v_avg = (*v + *(v + 1) + *(v + stride_v) + *(v + stride_v + 1)) >> 2;
+  // Ignore MB with too high or low brightness.
+  if (y_avg < y_low || y_avg > y_high)
+    return false;
+  else
+    return (EvaluateSkinColorDifference(u_avg, v_avg) < skin_threshold);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/util/skin_detection.h b/media/webrtc/trunk/webrtc/modules/video_processing/util/skin_detection.h
new file mode 100644
index 0000000000..561c03c425
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/util/skin_detection.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_SKIN_DETECTION_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_SKIN_DETECTION_H_
+
+namespace webrtc {
+
+typedef unsigned char uint8_t;
+bool MbHasSkinColor(const uint8_t* y_src,
+                    const uint8_t* u_src,
+                    const uint8_t* v_src,
+                    const int stride_y,
+                    const int stride_u,
+                    const int stride_v,
+                    const int mb_row,
+                    const int mb_col);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_SKIN_DETECTION_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.cc b/media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.cc
similarity index 70%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.cc
rename to media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.cc
index bf05bd7154..63e347b026 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.cc
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_processing/main/source/video_decimator.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/video_decimator.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 #define VD_MIN(a, b) ((a) < (b)) ? (a) : (b)
 
@@ -22,7 +23,7 @@ VPMVideoDecimator::VPMVideoDecimator() {
 
 VPMVideoDecimator::~VPMVideoDecimator() {}
 
-void VPMVideoDecimator::Reset()  {
+void VPMVideoDecimator::Reset() {
   overshoot_modifier_ = 0;
   drop_count_ = 0;
   keep_count_ = 0;
@@ -36,22 +37,23 @@ void VPMVideoDecimator::EnableTemporalDecimation(bool enable) {
   enable_temporal_decimation_ = enable;
 }
 
-int32_t VPMVideoDecimator::SetTargetFramerate(uint32_t frame_rate) {
-  if (frame_rate == 0) return VPM_PARAMETER_ERROR;
-
+void VPMVideoDecimator::SetTargetFramerate(int frame_rate) {
+  RTC_DCHECK(frame_rate);
   target_frame_rate_ = frame_rate;
-  return VPM_OK;
 }
 
 bool VPMVideoDecimator::DropFrame() {
-  if (!enable_temporal_decimation_) return false;
+  if (!enable_temporal_decimation_)
+    return false;
 
-  if (incoming_frame_rate_ <= 0) return false;
+  if (incoming_frame_rate_ <= 0)
+    return false;
 
   const uint32_t incomingframe_rate =
       static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
 
-  if (target_frame_rate_ == 0) return true;
+  if (target_frame_rate_ == 0)
+    return true;
 
   bool drop = false;
   if (incomingframe_rate > target_frame_rate_) {
@@ -62,44 +64,43 @@ bool VPMVideoDecimator::DropFrame() {
       overshoot_modifier_ = 0;
     }
 
-    if (overshoot && 2 * overshoot < (int32_t) incomingframe_rate) {
+    if (overshoot && 2 * overshoot < (int32_t)incomingframe_rate) {
       if (drop_count_) {  // Just got here so drop to be sure.
-          drop_count_ = 0;
-          return true;
+        drop_count_ = 0;
+        return true;
       }
       const uint32_t dropVar = incomingframe_rate / overshoot;
 
       if (keep_count_ >= dropVar) {
-          drop = true;
-          overshoot_modifier_ = -((int32_t) incomingframe_rate % overshoot) / 3;
-          keep_count_ = 1;
+        drop = true;
+        overshoot_modifier_ = -((int32_t)incomingframe_rate % overshoot) / 3;
+        keep_count_ = 1;
       } else {
-          keep_count_++;
+        keep_count_++;
       }
     } else {
       keep_count_ = 0;
       const uint32_t dropVar = overshoot / target_frame_rate_;
       if (drop_count_ < dropVar) {
-          drop = true;
-          drop_count_++;
+        drop = true;
+        drop_count_++;
       } else {
-          overshoot_modifier_ = overshoot % target_frame_rate_;
-          drop = false;
-          drop_count_ = 0;
+        overshoot_modifier_ = overshoot % target_frame_rate_;
+        drop = false;
+        drop_count_ = 0;
       }
     }
   }
   return drop;
 }
 
-
-uint32_t VPMVideoDecimator::Decimatedframe_rate() {
-ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
+uint32_t VPMVideoDecimator::GetDecimatedFrameRate() {
+  ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
   if (!enable_temporal_decimation_) {
     return static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
   }
   return VD_MIN(target_frame_rate_,
-      static_cast<uint32_t>(incoming_frame_rate_ + 0.5f));
+                static_cast<uint32_t>(incoming_frame_rate_ + 0.5f));
 }
 
 uint32_t VPMVideoDecimator::Inputframe_rate() {
@@ -114,7 +115,7 @@ void VPMVideoDecimator::UpdateIncomingframe_rate() {
   } else {
     // Shift.
     for (int i = kFrameCountHistory_size - 2; i >= 0; i--) {
-        incoming_frame_times_[i+1] = incoming_frame_times_[i];
+      incoming_frame_times_[i + 1] = incoming_frame_times_[i];
     }
   }
   incoming_frame_times_[0] = now;
@@ -134,7 +135,7 @@ void VPMVideoDecimator::ProcessIncomingframe_rate(int64_t now) {
     }
   }
   if (num > 1) {
-    int64_t diff = now - incoming_frame_times_[num-1];
+    int64_t diff = now - incoming_frame_times_[num - 1];
     incoming_frame_rate_ = 1.0;
     if (diff > 0) {
       incoming_frame_rate_ = nrOfFrames * 1000.0f / static_cast<float>(diff);
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.h b/media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.h
similarity index 71%
rename from media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.h
rename to media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.h
index fca74aeae1..1b871df8c3 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/main/source/video_decimator.h
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_decimator.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DECIMATOR_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DECIMATOR_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -25,14 +25,14 @@ class VPMVideoDecimator {
 
   void EnableTemporalDecimation(bool enable);
 
-  int32_t SetTargetFramerate(uint32_t frame_rate);
+  void SetTargetFramerate(int frame_rate);
 
   bool DropFrame();
 
   void UpdateIncomingframe_rate();
 
   // Get Decimated Frame Rate/Dimensions.
-  uint32_t Decimatedframe_rate();
+  uint32_t GetDecimatedFrameRate();
 
   // Get input frame rate.
   uint32_t Inputframe_rate();
@@ -40,8 +40,8 @@ class VPMVideoDecimator {
  private:
   void ProcessIncomingframe_rate(int64_t now);
 
-  enum { kFrameCountHistory_size = 90};
-  enum { kFrameHistoryWindowMs = 2000};
+  enum { kFrameCountHistory_size = 90 };
+  enum { kFrameHistoryWindowMs = 2000 };
 
   // Temporal decimation.
   int32_t overshoot_modifier_;
@@ -55,4 +55,4 @@ class VPMVideoDecimator {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DECIMATOR_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/video_denoiser.cc b/media/webrtc/trunk/webrtc/modules/video_processing/video_denoiser.cc
new file mode 100644
index 0000000000..4902a89491
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_denoiser.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/common_video/libyuv/include/scaler.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/video_processing/video_denoiser.h"
+
+namespace webrtc {
+
+VideoDenoiser::VideoDenoiser(bool runtime_cpu_detection)
+    : width_(0),
+      height_(0),
+      filter_(DenoiserFilter::Create(runtime_cpu_detection)) {}
+
+void VideoDenoiser::TrailingReduction(int mb_rows,
+                                      int mb_cols,
+                                      const uint8_t* y_src,
+                                      int stride_y,
+                                      uint8_t* y_dst) {
+  for (int mb_row = 1; mb_row < mb_rows - 1; ++mb_row) {
+    for (int mb_col = 1; mb_col < mb_cols - 1; ++mb_col) {
+      int mb_index = mb_row * mb_cols + mb_col;
+      uint8_t* mb_dst = y_dst + (mb_row << 4) * stride_y + (mb_col << 4);
+      const uint8_t* mb_src = y_src + (mb_row << 4) * stride_y + (mb_col << 4);
+      // If the number of denoised neighbors is less than a threshold,
+      // do NOT denoise for the block. Set different threshold for skin MB.
+      // The change of denoising status will not propagate.
+      if (metrics_[mb_index].is_skin) {
+        // The threshold is high (more strict) for non-skin MB where the
+        // trailing usually happen.
+        if (metrics_[mb_index].denoise &&
+            metrics_[mb_index + 1].denoise + metrics_[mb_index - 1].denoise +
+                    metrics_[mb_index + mb_cols].denoise +
+                    metrics_[mb_index - mb_cols].denoise <=
+                2) {
+          metrics_[mb_index].denoise = 0;
+          filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+        }
+      } else if (metrics_[mb_index].denoise &&
+                 metrics_[mb_index + 1].denoise +
+                         metrics_[mb_index - 1].denoise +
+                         metrics_[mb_index + mb_cols + 1].denoise +
+                         metrics_[mb_index + mb_cols - 1].denoise +
+                         metrics_[mb_index - mb_cols + 1].denoise +
+                         metrics_[mb_index - mb_cols - 1].denoise +
+                         metrics_[mb_index + mb_cols].denoise +
+                         metrics_[mb_index - mb_cols].denoise <=
+                     7) {
+        filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+      }
+    }
+  }
+}
+
+void VideoDenoiser::DenoiseFrame(const VideoFrame& frame,
+                                 VideoFrame* denoised_frame) {
+  int stride_y = frame.stride(kYPlane);
+  int stride_u = frame.stride(kUPlane);
+  int stride_v = frame.stride(kVPlane);
+  // If previous width and height are different from current frame's, then no
+  // denoising for the current frame.
+  if (width_ != frame.width() || height_ != frame.height()) {
+    width_ = frame.width();
+    height_ = frame.height();
+    denoised_frame->CreateFrame(frame.buffer(kYPlane), frame.buffer(kUPlane),
+                                frame.buffer(kVPlane), width_, height_,
+                                stride_y, stride_u, stride_v);
+    // Setting time parameters to the output frame.
+    denoised_frame->set_timestamp(frame.timestamp());
+    denoised_frame->set_render_time_ms(frame.render_time_ms());
+    return;
+  }
+  // For 16x16 block.
+  int mb_cols = width_ >> 4;
+  int mb_rows = height_ >> 4;
+  if (metrics_.get() == nullptr)
+    metrics_.reset(new DenoiseMetrics[mb_cols * mb_rows]());
+  // Denoise on Y plane.
+  uint8_t* y_dst = denoised_frame->buffer(kYPlane);
+  uint8_t* u_dst = denoised_frame->buffer(kUPlane);
+  uint8_t* v_dst = denoised_frame->buffer(kVPlane);
+  const uint8_t* y_src = frame.buffer(kYPlane);
+  const uint8_t* u_src = frame.buffer(kUPlane);
+  const uint8_t* v_src = frame.buffer(kVPlane);
+  // Temporary buffer to store denoising result.
+  uint8_t y_tmp[16 * 16] = {0};
+  for (int mb_row = 0; mb_row < mb_rows; ++mb_row) {
+    for (int mb_col = 0; mb_col < mb_cols; ++mb_col) {
+      const uint8_t* mb_src = y_src + (mb_row << 4) * stride_y + (mb_col << 4);
+      uint8_t* mb_dst = y_dst + (mb_row << 4) * stride_y + (mb_col << 4);
+      int mb_index = mb_row * mb_cols + mb_col;
+      // Denoise each MB at the very start and save the result to a temporary
+      // buffer.
+      if (filter_->MbDenoise(mb_dst, stride_y, y_tmp, 16, mb_src, stride_y, 0,
+                             1) == FILTER_BLOCK) {
+        uint32_t thr_var = 0;
+        // Save var and sad to the buffer.
+        metrics_[mb_index].var = filter_->Variance16x8(
+            mb_dst, stride_y, y_tmp, 16, &metrics_[mb_index].sad);
+        // Get skin map.
+        metrics_[mb_index].is_skin = MbHasSkinColor(
+            y_src, u_src, v_src, stride_y, stride_u, stride_v, mb_row, mb_col);
+        // Variance threshold for skin/non-skin MB is different.
+        // Skin MB use a small threshold to reduce blockiness.
+        thr_var = metrics_[mb_index].is_skin ? 128 : 12 * 128;
+        if (metrics_[mb_index].var > thr_var) {
+          metrics_[mb_index].denoise = 0;
+          // Use the source MB.
+          filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+        } else {
+          metrics_[mb_index].denoise = 1;
+          // Use the denoised MB.
+          filter_->CopyMem16x16(y_tmp, 16, mb_dst, stride_y);
+        }
+      } else {
+        metrics_[mb_index].denoise = 0;
+        filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+      }
+      // Copy source U/V plane.
+      const uint8_t* mb_src_u =
+          u_src + (mb_row << 3) * stride_u + (mb_col << 3);
+      const uint8_t* mb_src_v =
+          v_src + (mb_row << 3) * stride_v + (mb_col << 3);
+      uint8_t* mb_dst_u = u_dst + (mb_row << 3) * stride_u + (mb_col << 3);
+      uint8_t* mb_dst_v = v_dst + (mb_row << 3) * stride_v + (mb_col << 3);
+      filter_->CopyMem8x8(mb_src_u, stride_u, mb_dst_u, stride_u);
+      filter_->CopyMem8x8(mb_src_v, stride_v, mb_dst_v, stride_v);
+    }
+  }
+  // Second round.
+  // This is to reduce the trailing artifact and blockiness by referring
+  // neighbors' denoising status.
+  TrailingReduction(mb_rows, mb_cols, y_src, stride_y, y_dst);
+
+  // Setting time parameters to the output frame.
+  denoised_frame->set_timestamp(frame.timestamp());
+  denoised_frame->set_render_time_ms(frame.render_time_ms());
+  return;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/video_denoiser.h b/media/webrtc/trunk/webrtc/modules/video_processing/video_denoiser.h
new file mode 100644
index 0000000000..107a15ca07
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_denoiser.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DENOISER_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DENOISER_H_
+
+#include "webrtc/modules/video_processing/util/denoiser_filter.h"
+#include "webrtc/modules/video_processing/util/skin_detection.h"
+
+namespace webrtc {
+
+class VideoDenoiser {
+ public:
+  explicit VideoDenoiser(bool runtime_cpu_detection);
+  void DenoiseFrame(const VideoFrame& frame, VideoFrame* denoised_frame);
+
+ private:
+  void TrailingReduction(int mb_rows,
+                         int mb_cols,
+                         const uint8_t* y_src,
+                         int stride_y,
+                         uint8_t* y_dst);
+  int width_;
+  int height_;
+  rtc::scoped_ptr<DenoiseMetrics[]> metrics_;
+  rtc::scoped_ptr<DenoiserFilter> filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_DENOISER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/video_processing.gypi b/media/webrtc/trunk/webrtc/modules/video_processing/video_processing.gypi
index 84fc2b5458..42f1811c1f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_processing/video_processing.gypi
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_processing.gypi
@@ -18,32 +18,38 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
-        'main/interface/video_processing.h',
-        'main/interface/video_processing_defines.h',
-        'main/source/brighten.cc',
-        'main/source/brighten.h',
-        'main/source/brightness_detection.cc',
-        'main/source/brightness_detection.h',
-        'main/source/color_enhancement.cc',
-        'main/source/color_enhancement.h',
-        'main/source/color_enhancement_private.h',
-        'main/source/content_analysis.cc',
-        'main/source/content_analysis.h',
-        'main/source/deflickering.cc',
-        'main/source/deflickering.h',
-        'main/source/frame_preprocessor.cc',
-        'main/source/frame_preprocessor.h',
-        'main/source/spatial_resampler.cc',
-        'main/source/spatial_resampler.h',
-        'main/source/video_decimator.cc',
-        'main/source/video_decimator.h',
-        'main/source/video_processing_impl.cc',
-        'main/source/video_processing_impl.h',
+        'include/video_processing.h',
+        'include/video_processing_defines.h',
+        'brightness_detection.cc',
+        'brightness_detection.h',
+        'content_analysis.cc',
+        'content_analysis.h',
+        'deflickering.cc',
+        'deflickering.h',
+        'frame_preprocessor.cc',
+        'frame_preprocessor.h',
+        'spatial_resampler.cc',
+        'spatial_resampler.h',
+        'video_decimator.cc',
+        'video_decimator.h',
+        'video_processing_impl.cc',
+        'video_processing_impl.h',
+        'video_denoiser.cc',
+        'video_denoiser.h',
+        'util/denoiser_filter.cc',
+        'util/denoiser_filter.h',
+        'util/denoiser_filter_c.cc',
+        'util/denoiser_filter_c.h',
+        'util/skin_detection.cc',
+        'util/skin_detection.h',
       ],
       'conditions': [
         ['target_arch=="ia32" or target_arch=="x64"', {
           'dependencies': [ 'video_processing_sse2', ],
         }],
+        ['target_arch=="arm" or target_arch == "arm64"', {
+          'dependencies': [ 'video_processing_neon', ],
+        }],
       ],
     },
   ],
@@ -54,7 +60,9 @@
           'target_name': 'video_processing_sse2',
           'type': 'static_library',
           'sources': [
-            'main/source/content_analysis_sse2.cc',
+            'content_analysis_sse2.cc',
+            'util/denoiser_filter_sse2.cc',
+            'util/denoiser_filter_sse2.h',
           ],
           'conditions': [
             ['os_posix==1 and OS!="mac"', {
@@ -70,6 +78,19 @@
         },
       ],
     }],
+    ['target_arch=="arm" or target_arch == "arm64"', {
+      'targets': [
+        {
+          'target_name': 'video_processing_neon',
+          'type': 'static_library',
+          'includes': [ '../../build/arm_neon.gypi', ],
+          'sources': [
+            'util/denoiser_filter_neon.cc',
+            'util/denoiser_filter_neon.h',
+          ],
+        },
+      ],
+    }],
   ],
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/video_processing_impl.cc b/media/webrtc/trunk/webrtc/modules/video_processing/video_processing_impl.cc
new file mode 100644
index 0000000000..f34886f10f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_processing_impl.cc
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_processing/video_processing_impl.h"
+
+#include <assert.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+namespace {
+
+int GetSubSamplingFactor(int width, int height) {
+  if (width * height >= 640 * 480) {
+    return 3;
+  } else if (width * height >= 352 * 288) {
+    return 2;
+  } else if (width * height >= 176 * 144) {
+    return 1;
+  } else {
+    return 0;
+  }
+}
+}  // namespace
+
+VideoProcessing* VideoProcessing::Create() {
+  return new VideoProcessingImpl();
+}
+
+VideoProcessingImpl::VideoProcessingImpl() {}
+VideoProcessingImpl::~VideoProcessingImpl() {}
+
+void VideoProcessing::GetFrameStats(const VideoFrame& frame,
+                                    FrameStats* stats) {
+  ClearFrameStats(stats);  // The histogram needs to be zeroed out.
+  if (frame.IsZeroSize()) {
+    return;
+  }
+
+  int width = frame.width();
+  int height = frame.height();
+  stats->sub_sampling_factor = GetSubSamplingFactor(width, height);
+
+  const uint8_t* buffer = frame.buffer(kYPlane);
+  // Compute histogram and sum of frame
+  for (int i = 0; i < height; i += (1 << stats->sub_sampling_factor)) {
+    int k = i * width;
+    for (int j = 0; j < width; j += (1 << stats->sub_sampling_factor)) {
+      stats->hist[buffer[k + j]]++;
+      stats->sum += buffer[k + j];
+    }
+  }
+
+  stats->num_pixels = (width * height) / ((1 << stats->sub_sampling_factor) *
+                                          (1 << stats->sub_sampling_factor));
+  assert(stats->num_pixels > 0);
+
+  // Compute mean value of frame
+  stats->mean = stats->sum / stats->num_pixels;
+}
+
+bool VideoProcessing::ValidFrameStats(const FrameStats& stats) {
+  if (stats.num_pixels == 0) {
+    LOG(LS_WARNING) << "Invalid frame stats.";
+    return false;
+  }
+  return true;
+}
+
+void VideoProcessing::ClearFrameStats(FrameStats* stats) {
+  stats->mean = 0;
+  stats->sum = 0;
+  stats->num_pixels = 0;
+  stats->sub_sampling_factor = 0;
+  memset(stats->hist, 0, sizeof(stats->hist));
+}
+
+void VideoProcessing::Brighten(int delta, VideoFrame* frame) {
+  RTC_DCHECK(!frame->IsZeroSize());
+  RTC_DCHECK(frame->width() > 0);
+  RTC_DCHECK(frame->height() > 0);
+
+  int num_pixels = frame->width() * frame->height();
+
+  int look_up[256];
+  for (int i = 0; i < 256; i++) {
+    int val = i + delta;
+    look_up[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
+  }
+
+  uint8_t* temp_ptr = frame->buffer(kYPlane);
+  for (int i = 0; i < num_pixels; i++) {
+    *temp_ptr = static_cast<uint8_t>(look_up[*temp_ptr]);
+    temp_ptr++;
+  }
+}
+
+int32_t VideoProcessingImpl::Deflickering(VideoFrame* frame,
+                                          FrameStats* stats) {
+  rtc::CritScope mutex(&mutex_);
+  return deflickering_.ProcessFrame(frame, stats);
+}
+
+int32_t VideoProcessingImpl::BrightnessDetection(const VideoFrame& frame,
+                                                 const FrameStats& stats) {
+  rtc::CritScope mutex(&mutex_);
+  return brightness_detection_.ProcessFrame(frame, stats);
+}
+
+void VideoProcessingImpl::EnableTemporalDecimation(bool enable) {
+  rtc::CritScope mutex(&mutex_);
+  frame_pre_processor_.EnableTemporalDecimation(enable);
+}
+
+void VideoProcessingImpl::SetInputFrameResampleMode(
+    VideoFrameResampling resampling_mode) {
+  rtc::CritScope cs(&mutex_);
+  frame_pre_processor_.SetInputFrameResampleMode(resampling_mode);
+}
+
+int32_t VideoProcessingImpl::SetTargetResolution(uint32_t width,
+                                                 uint32_t height,
+                                                 uint32_t frame_rate) {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.SetTargetResolution(width, height, frame_rate);
+}
+
+void VideoProcessingImpl::SetTargetFramerate(int frame_rate) {
+  rtc::CritScope cs(&mutex_);
+  frame_pre_processor_.SetTargetFramerate(frame_rate);
+}
+
+uint32_t VideoProcessingImpl::GetDecimatedFrameRate() {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.GetDecimatedFrameRate();
+}
+
+uint32_t VideoProcessingImpl::GetDecimatedWidth() const {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.GetDecimatedWidth();
+}
+
+uint32_t VideoProcessingImpl::GetDecimatedHeight() const {
+  rtc::CritScope cs(&mutex_);
+  return frame_pre_processor_.GetDecimatedHeight();
+}
+
+void VideoProcessingImpl::EnableDenosing(bool enable) {
+  rtc::CritScope cs(&mutex_);
+  frame_pre_processor_.EnableDenosing(enable);
+}
+
+const VideoFrame* VideoProcessingImpl::PreprocessFrame(
+    const VideoFrame& frame) {
+  rtc::CritScope mutex(&mutex_);
+  return frame_pre_processor_.PreprocessFrame(frame);
+}
+
+VideoContentMetrics* VideoProcessingImpl::GetContentMetrics() const {
+  rtc::CritScope mutex(&mutex_);
+  return frame_pre_processor_.GetContentMetrics();
+}
+
+void VideoProcessingImpl::EnableContentAnalysis(bool enable) {
+  rtc::CritScope mutex(&mutex_);
+  frame_pre_processor_.EnableContentAnalysis(enable);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_processing/video_processing_impl.h b/media/webrtc/trunk/webrtc/modules/video_processing/video_processing_impl.h
new file mode 100644
index 0000000000..edbaba12fa
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/modules/video_processing/video_processing_impl.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_PROCESSING_IMPL_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_PROCESSING_IMPL_H_
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_processing/brightness_detection.h"
+#include "webrtc/modules/video_processing/deflickering.h"
+#include "webrtc/modules/video_processing/frame_preprocessor.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+
+class VideoProcessingImpl : public VideoProcessing {
+ public:
+  VideoProcessingImpl();
+  ~VideoProcessingImpl() override;
+
+  // Implements VideoProcessing.
+  int32_t Deflickering(VideoFrame* frame, FrameStats* stats) override;
+  int32_t BrightnessDetection(const VideoFrame& frame,
+                              const FrameStats& stats) override;
+  void EnableTemporalDecimation(bool enable) override;
+  void SetInputFrameResampleMode(VideoFrameResampling resampling_mode) override;
+  void EnableContentAnalysis(bool enable) override;
+  int32_t SetTargetResolution(uint32_t width,
+                              uint32_t height,
+                              uint32_t frame_rate) override;
+  void SetTargetFramerate(int frame_rate) override;
+  uint32_t GetDecimatedFrameRate() override;
+  uint32_t GetDecimatedWidth() const override;
+  uint32_t GetDecimatedHeight() const override;
+  void EnableDenosing(bool enable) override;
+  const VideoFrame* PreprocessFrame(const VideoFrame& frame) override;
+  VideoContentMetrics* GetContentMetrics() const override;
+
+ private:
+  mutable rtc::CriticalSection mutex_;
+  VPMDeflickering deflickering_ GUARDED_BY(mutex_);
+  VPMBrightnessDetection brightness_detection_;
+  VPMFramePreprocessor frame_pre_processor_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_VIDEO_PROCESSING_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/BUILD.gn b/media/webrtc/trunk/webrtc/modules/video_render/BUILD.gn
index 1e3e3e5d3d..0771bd7080 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/video_render/BUILD.gn
@@ -13,12 +13,8 @@ source_set("video_render_module") {
     "external/video_render_external_impl.cc",
     "external/video_render_external_impl.h",
     "i_video_render.h",
-    "include/video_render.h",
-    "include/video_render_defines.h",
-    "incoming_video_stream.cc",
-    "incoming_video_stream.h",
-    "video_render_frames.cc",
-    "video_render_frames.h",
+    "video_render.h",
+    "video_render_defines.h",
     "video_render_impl.h",
   ]
 
@@ -123,17 +119,17 @@ if (!build_with_chromium) {
       ]
 
       directxsdk_exists =
-        (exec_script("//build/dir_exists.py",
-                     [ rebase_path("//third_party/directxsdk/files",
-                                   root_build_dir) ],
-                     "trim string") == "True")
+          exec_script("//build/dir_exists.py",
+                      [ rebase_path("//third_party/directxsdk/files",
+                                    root_build_dir) ],
+                      "trim string") == "True"
       if (directxsdk_exists) {
         directxsdk_path = "//third_party/directxsdk/files"
       } else {
         directxsdk_path =
             exec_script("../../build/find_directx_sdk.py", [], "trim string")
       }
-      include_dirs = [ directxsdk_path +  "/Include" ]
+      include_dirs = [ directxsdk_path + "/Include" ]
     }
     if (is_android) {
       sources += [
@@ -165,10 +161,10 @@ if (!build_with_chromium) {
 
       deps += [ "../..:webrtc_common" ]
 
-      cflags += [ "-fobjc-arc" ]  # CLANG_ENABLE_OBJC_ARC = YES.
+      cflags = [ "-fobjc-arc" ]  # CLANG_ENABLE_OBJC_ARC = YES.
     }
 
-    all_dependent_configs = [ ":video_render_internal_impl_config"]
+    all_dependent_configs = [ ":video_render_internal_impl_config" ]
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java b/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java
index b1097504a2..290635f088 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java
@@ -23,6 +23,7 @@ import android.content.Context;
 import android.content.pm.ConfigurationInfo;
 import android.graphics.PixelFormat;
 import android.opengl.GLSurfaceView;
+
 import android.util.Log;
 
 import org.mozilla.gecko.annotation.WebRTCJNITarget;
@@ -31,15 +32,12 @@ public class ViEAndroidGLES20 extends GLSurfaceView
         implements GLSurfaceView.Renderer {
     static final String TAG = "WEBRTC-JR";
     private static final boolean DEBUG = false;
-
     // True if onSurfaceCreated has been called.
     private boolean surfaceCreated;
     private boolean openGLCreated;
-
     // True if NativeFunctionsRegistered has been called.
     private boolean nativeFunctionsRegisted;
     private ReentrantLock nativeFunctionLock = new ReentrantLock();
-
     // Address of Native object that will do the drawing.
     private long nativeObject;
     private int viewWidth;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java b/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java
index 9f46255bfb..4757038976 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java
@@ -21,11 +21,12 @@ import java.nio.ByteBuffer;
 import android.graphics.Bitmap;
 import android.graphics.Canvas;
 import android.graphics.Rect;
-import android.util.Log;
 import android.view.SurfaceHolder;
 import android.view.SurfaceView;
 import android.view.SurfaceHolder.Callback;
 
+import android.util.Log;
+
 import org.mozilla.gecko.annotation.WebRTCJNITarget;
 
 public class ViESurfaceRenderer implements Callback {
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.cc
index 15b52d7739..9affb23d99 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.cc
@@ -11,9 +11,9 @@
 #include "webrtc/modules/video_render/android/video_render_android_impl.h"
 
 #include "webrtc/modules/video_render/video_render_internal.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 #ifdef ANDROID
 #include <android/log.h>
@@ -22,7 +22,7 @@
 #undef WEBRTC_TRACE
 #define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
 #else
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #endif
 
 namespace webrtc {
@@ -141,18 +141,13 @@ int32_t VideoRenderAndroid::StartRender() {
     return 0;
   }
 
-  _javaRenderThread = ThreadWrapper::CreateThread(JavaRenderThreadFun, this,
-                                                  "AndroidRenderThread");
+  _javaRenderThread.reset(new rtc::PlatformThread(JavaRenderThreadFun, this,
+                                                  "AndroidRenderThread"));
 
-  if (_javaRenderThread->Start())
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-                 "%s: thread started", __FUNCTION__);
-  else {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Could not start send thread", __FUNCTION__);
-    return -1;
-  }
-  _javaRenderThread->SetPriority(kRealtimePriority);
+  _javaRenderThread->Start();
+  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: thread started",
+               __FUNCTION__);
+  _javaRenderThread->SetPriority(rtc::kRealtimePriority);
   return 0;
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.h b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.h
index fb32acf08a..e5b7de4643 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_impl.h
@@ -15,8 +15,8 @@
 
 #include <map>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 
 namespace webrtc {
@@ -144,7 +144,8 @@ class VideoRenderAndroid: IVideoRender {
   EventWrapper& _javaRenderEvent;
   int64_t _lastJavaRenderEvent;
   JNIEnv* _javaRenderJniEnv; // JNIEnv for the java render thread.
-  rtc::scoped_ptr<ThreadWrapper> _javaRenderThread;
+  // TODO(pbos): Remove scoped_ptr and use the member directly.
+  rtc::scoped_ptr<rtc::PlatformThread> _javaRenderThread;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc
index 170b73376f..286776e317 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc
@@ -9,8 +9,8 @@
  */
 
 #include "webrtc/modules/video_render/android/video_render_android_native_opengl2.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 #ifdef ANDROID_LOG
 #include <android/log.h>
@@ -19,7 +19,7 @@
 #undef WEBRTC_TRACE
 #define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
 #else
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #endif
 
 namespace webrtc {
@@ -381,9 +381,8 @@ int32_t AndroidNativeOpenGl2Channel::Init(int32_t zOrder,
   return 0;
 }
 
-int32_t AndroidNativeOpenGl2Channel::RenderFrame(
-    const uint32_t /*streamId*/,
-    const I420VideoFrame& videoFrame) {
+int32_t AndroidNativeOpenGl2Channel::RenderFrame(const uint32_t /*streamId*/,
+                                                 const VideoFrame& videoFrame) {
   //   WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
   _renderCritSect.Enter();
   _bufferToRender = videoFrame;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.h b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
index a006f2e2e2..8be247b834 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
@@ -15,7 +15,7 @@
 
 #include "webrtc/modules/video_render/android/video_render_android_impl.h"
 #include "webrtc/modules/video_render/android/video_render_opengles20.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 namespace webrtc {
 
@@ -33,9 +33,8 @@ class AndroidNativeOpenGl2Channel: public AndroidStream {
                const float right, const float bottom);
 
   //Implement VideoRenderCallback
-  virtual int32_t RenderFrame(
-      const uint32_t streamId,
-      const I420VideoFrame& videoFrame);
+  virtual int32_t RenderFrame(const uint32_t streamId,
+                              const VideoFrame& videoFrame);
 
   //Implements AndroidStream
   virtual void DeliverFrame(JNIEnv* jniEnv);
@@ -54,7 +53,7 @@ class AndroidNativeOpenGl2Channel: public AndroidStream {
   uint32_t _id;
   CriticalSectionWrapper& _renderCritSect;
 
-  I420VideoFrame _bufferToRender;
+  VideoFrame _bufferToRender;
   VideoRenderAndroid& _renderer;
   JavaVM*     _jvm;
   jobject     _javaRenderObj;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.cc b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.cc
index eca8a1a8cd..ea3b106b1e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.cc
@@ -10,8 +10,8 @@
 
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_render/android/video_render_android_surface_view.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 #ifdef ANDROID_LOG
 #include <android/log.h>
@@ -20,7 +20,7 @@
 #undef WEBRTC_TRACE
 #define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
 #else
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #endif
 
 namespace webrtc {
@@ -409,10 +409,8 @@ int32_t AndroidSurfaceViewChannel::Init(
   return 0;
 }
 
-
-int32_t AndroidSurfaceViewChannel::RenderFrame(
-    const uint32_t /*streamId*/,
-    const I420VideoFrame& videoFrame) {
+int32_t AndroidSurfaceViewChannel::RenderFrame(const uint32_t /*streamId*/,
+                                               const VideoFrame& videoFrame) {
   // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
   _renderCritSect.Enter();
   _bufferToRender = videoFrame;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.h b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.h
index acafa464db..0f029b54f3 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_android_surface_view.h
@@ -14,7 +14,7 @@
 #include <jni.h>
 
 #include "webrtc/modules/video_render/android/video_render_android_impl.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 namespace webrtc {
 
@@ -33,7 +33,7 @@ class AndroidSurfaceViewChannel : public AndroidStream {
 
   //Implement VideoRenderCallback
   virtual int32_t RenderFrame(const uint32_t streamId,
-                              const I420VideoFrame& videoFrame);
+                              const VideoFrame& videoFrame);
 
   //Implements AndroidStream
   virtual void DeliverFrame(JNIEnv* jniEnv);
@@ -42,7 +42,7 @@ class AndroidSurfaceViewChannel : public AndroidStream {
   uint32_t _id;
   CriticalSectionWrapper& _renderCritSect;
 
-  I420VideoFrame _bufferToRender;
+  VideoFrame _bufferToRender;
   VideoRenderAndroid& _renderer;
   JavaVM* _jvm;
   jobject _javaRenderObj;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.cc b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.cc
index 046de68428..45db56a4f6 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.cc
@@ -25,7 +25,7 @@
 #undef WEBRTC_TRACE
 #define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
 #else
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #endif
 
 namespace webrtc {
@@ -214,8 +214,7 @@ int32_t VideoRenderOpenGles20::SetCoordinates(int32_t zOrder,
   return 0;
 }
 
-int32_t VideoRenderOpenGles20::Render(const I420VideoFrame& frameToRender) {
-
+int32_t VideoRenderOpenGles20::Render(const VideoFrame& frameToRender) {
   if (frameToRender.IsZeroSize()) {
     return -1;
   }
@@ -335,7 +334,7 @@ static void InitializeTexture(int name, int id, int width, int height) {
                GL_LUMINANCE, GL_UNSIGNED_BYTE, NULL);
 }
 
-void VideoRenderOpenGles20::SetupTextures(const I420VideoFrame& frameToRender) {
+void VideoRenderOpenGles20::SetupTextures(const VideoFrame& frameToRender) {
   WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
                "%s: width %d, height %d", __FUNCTION__,
                frameToRender.width(), frameToRender.height());
@@ -373,8 +372,7 @@ static void GlTexSubImage2D(GLsizei width, GLsizei height, int stride,
   }
 }
 
-void VideoRenderOpenGles20::UpdateTextures(const
-                                           I420VideoFrame& frameToRender) {
+void VideoRenderOpenGles20::UpdateTextures(const VideoFrame& frameToRender) {
   const GLsizei width = frameToRender.width();
   const GLsizei height = frameToRender.height();
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.h b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.h
index f20f61477c..57e2a10d42 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/android/video_render_opengles20.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #include <GLES2/gl2.h>
 #include <GLES2/gl2ext.h>
@@ -25,7 +25,7 @@ class VideoRenderOpenGles20 {
   ~VideoRenderOpenGles20();
 
   int32_t Setup(int32_t widht, int32_t height);
-  int32_t Render(const I420VideoFrame& frameToRender);
+  int32_t Render(const VideoFrame& frameToRender);
   int32_t SetCoordinates(int32_t zOrder, const float left, const float top,
                          const float right, const float bottom);
 
@@ -35,8 +35,8 @@ class VideoRenderOpenGles20 {
   GLuint loadShader(GLenum shaderType, const char* pSource);
   GLuint createProgram(const char* pVertexSource,
                        const char* pFragmentSource);
-  void SetupTextures(const I420VideoFrame& frameToRender);
-  void UpdateTextures(const I420VideoFrame& frameToRender);
+  void SetupTextures(const VideoFrame& frameToRender);
+  void UpdateTextures(const VideoFrame& frameToRender);
 
   int32_t _id;
   GLuint _textureIds[3]; // Texture id of Y,U and V texture.
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.cc
index d37b30f47c..58df07875e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.cc
@@ -189,8 +189,7 @@ int32_t VideoRenderExternalImpl::SetBitmap(const void* bitMap,
 
 // VideoRenderCallback
 int32_t VideoRenderExternalImpl::RenderFrame(const uint32_t streamId,
-                                             const I420VideoFrame& videoFrame)
-{
+                                             const VideoFrame& videoFrame) {
     return 0;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.h b/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.h
index 43182b1c5a..a8b663fff7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/external/video_render_external_impl.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -115,7 +115,7 @@ public:
 
     // VideoRenderCallback
     virtual int32_t RenderFrame(const uint32_t streamId,
-                                const I420VideoFrame& videoFrame);
+                                const VideoFrame& videoFrame);
 
 private:
     CriticalSectionWrapper& _critSect;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/i_video_render.h b/media/webrtc/trunk/webrtc/modules/video_render/i_video_render.h
index ff1cce782e..e6ec7a4680 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/i_video_render.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/i_video_render.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
 
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/modules/video_render/video_render.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/incoming_video_stream.cc b/media/webrtc/trunk/webrtc/modules/video_render/incoming_video_stream.cc
deleted file mode 100644
index 19659658f0..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_render/incoming_video_stream.cc
+++ /dev/null
@@ -1,329 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/incoming_video_stream.h"
-
-#include <assert.h>
-
-#if defined(_WIN32)
-#include <windows.h>
-#elif defined(WEBRTC_LINUX)
-#include <sys/time.h>
-#include <time.h>
-#else
-#include <sys/time.h>
-#endif
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/video_render_frames.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-IncomingVideoStream::IncomingVideoStream(const int32_t module_id,
-                                         const uint32_t stream_id)
-    : module_id_(module_id),
-      stream_id_(stream_id),
-      stream_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
-      thread_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
-      buffer_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
-      incoming_render_thread_(),
-      deliver_buffer_event_(*EventWrapper::Create()),
-      running_(false),
-      external_callback_(NULL),
-      render_callback_(NULL),
-      render_buffers_(*(new VideoRenderFrames)),
-      callbackVideoType_(kVideoI420),
-      callbackWidth_(0),
-      callbackHeight_(0),
-      incoming_rate_(0),
-      last_rate_calculation_time_ms_(0),
-      num_frames_since_last_calculation_(0),
-      last_render_time_ms_(0),
-      temp_frame_(),
-      start_image_(),
-      timeout_image_(),
-      timeout_time_() {
-  WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, module_id_,
-               "%s created for stream %d", __FUNCTION__, stream_id);
-}
-
-IncomingVideoStream::~IncomingVideoStream() {
-  WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, module_id_,
-               "%s deleted for stream %d", __FUNCTION__, stream_id_);
-
-  Stop();
-
-  // incoming_render_thread_ - Delete in stop
-  delete &render_buffers_;
-  delete &stream_critsect_;
-  delete &buffer_critsect_;
-  delete &thread_critsect_;
-  delete &deliver_buffer_event_;
-}
-
-int32_t IncomingVideoStream::ChangeModuleId(const int32_t id) {
-  CriticalSectionScoped cs(&stream_critsect_);
-  module_id_ = id;
-  return 0;
-}
-
-VideoRenderCallback* IncomingVideoStream::ModuleCallback() {
-  CriticalSectionScoped cs(&stream_critsect_);
-  return this;
-}
-
-int32_t IncomingVideoStream::RenderFrame(const uint32_t stream_id,
-                                         const I420VideoFrame& video_frame) {
-  CriticalSectionScoped csS(&stream_critsect_);
-  WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, module_id_,
-               "%s for stream %d, render time: %u", __FUNCTION__, stream_id_,
-               video_frame.render_time_ms());
-
-  if (!running_) {
-    WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, module_id_,
-                 "%s: Not running", __FUNCTION__);
-    return -1;
-  }
-
-  // Rate statistics.
-  num_frames_since_last_calculation_++;
-  int64_t now_ms = TickTime::MillisecondTimestamp();
-  if (now_ms >= last_rate_calculation_time_ms_ + KFrameRatePeriodMs) {
-    incoming_rate_ =
-        static_cast<uint32_t>(1000 * num_frames_since_last_calculation_ /
-                              (now_ms - last_rate_calculation_time_ms_));
-    num_frames_since_last_calculation_ = 0;
-    last_rate_calculation_time_ms_ = now_ms;
-  }
-
-  // Insert frame.
-  CriticalSectionScoped csB(&buffer_critsect_);
-  if (render_buffers_.AddFrame(video_frame) == 1)
-    deliver_buffer_event_.Set();
-
-  return 0;
-}
-
-int32_t IncomingVideoStream::SetStartImage(
-    const I420VideoFrame& video_frame) {
-  CriticalSectionScoped csS(&thread_critsect_);
-  return start_image_.CopyFrame(video_frame);
-}
-
-int32_t IncomingVideoStream::SetTimeoutImage(
-    const I420VideoFrame& video_frame, const uint32_t timeout) {
-  CriticalSectionScoped csS(&thread_critsect_);
-  timeout_time_ = timeout;
-  return timeout_image_.CopyFrame(video_frame);
-}
-
-int32_t IncomingVideoStream::SetRenderCallback(
-    VideoRenderCallback* render_callback) {
-  CriticalSectionScoped cs(&stream_critsect_);
-
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, module_id_,
-               "%s(%x) for stream %d", __FUNCTION__, render_callback,
-               stream_id_);
-  render_callback_ = render_callback;
-  return 0;
-}
-
-int32_t IncomingVideoStream::SetExpectedRenderDelay(
-    int32_t delay_ms) {
-  CriticalSectionScoped csS(&stream_critsect_);
-  if (running_) {
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, module_id_,
-                 "%s(%d) for stream %d", __FUNCTION__, delay_ms, stream_id_);
-    return -1;
-  }
-  CriticalSectionScoped cs(&buffer_critsect_);
-  return render_buffers_.SetRenderDelay(delay_ms);
-}
-
-int32_t IncomingVideoStream::SetExternalCallback(
-    VideoRenderCallback* external_callback) {
-  CriticalSectionScoped cs(&stream_critsect_);
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, module_id_,
-               "%s(%x) for stream %d", __FUNCTION__, external_callback,
-               stream_id_);
-  external_callback_ = external_callback;
-  callbackVideoType_ = kVideoI420;
-  callbackWidth_ = 0;
-  callbackHeight_ = 0;
-  return 0;
-}
-
-int32_t IncomingVideoStream::Start() {
-  CriticalSectionScoped csS(&stream_critsect_);
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, module_id_,
-               "%s for stream %d", __FUNCTION__, stream_id_);
-  if (running_) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, module_id_,
-                 "%s: Already running", __FUNCTION__);
-    return 0;
-  }
-
-  CriticalSectionScoped csT(&thread_critsect_);
-  assert(incoming_render_thread_ == NULL);
-
-  incoming_render_thread_ = ThreadWrapper::CreateThread(
-      IncomingVideoStreamThreadFun, this, "IncomingVideoStreamThread");
-  if (!incoming_render_thread_) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, module_id_,
-                 "%s: No thread", __FUNCTION__);
-    return -1;
-  }
-
-  if (incoming_render_thread_->Start()) {
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, module_id_,
-                 "%s: thread started", __FUNCTION__);
-  } else {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, module_id_,
-                 "%s: Could not start send thread", __FUNCTION__);
-    return -1;
-  }
-  incoming_render_thread_->SetPriority(kRealtimePriority);
-  deliver_buffer_event_.StartTimer(false, KEventStartupTimeMS);
-
-  running_ = true;
-  return 0;
-}
-
-int32_t IncomingVideoStream::Stop() {
-  CriticalSectionScoped cs_stream(&stream_critsect_);
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, module_id_,
-               "%s for stream %d", __FUNCTION__, stream_id_);
-
-  if (!running_) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, module_id_,
-                 "%s: Not running", __FUNCTION__);
-    return 0;
-  }
-
-  ThreadWrapper* thread = NULL;
-  {
-    CriticalSectionScoped cs_thread(&thread_critsect_);
-    if (incoming_render_thread_) {
-      // Setting the incoming render thread to NULL marks that we're performing
-      // a shutdown and will make IncomingVideoStreamProcess abort after wakeup.
-      thread = incoming_render_thread_.release();
-      deliver_buffer_event_.StopTimer();
-      // Set the event to allow the thread to wake up and shut down without
-      // waiting for a timeout.
-      deliver_buffer_event_.Set();
-    }
-  }
-  if (thread) {
-    if (thread->Stop()) {
-      delete thread;
-    } else {
-      assert(false);
-      WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, module_id_,
-                   "%s: Not able to stop thread, leaking", __FUNCTION__);
-    }
-  }
-  running_ = false;
-  return 0;
-}
-
-int32_t IncomingVideoStream::Reset() {
-  CriticalSectionScoped cs_stream(&stream_critsect_);
-  CriticalSectionScoped cs_buffer(&buffer_critsect_);
-  render_buffers_.ReleaseAllFrames();
-  return 0;
-}
-
-uint32_t IncomingVideoStream::StreamId() const {
-  CriticalSectionScoped cs_stream(&stream_critsect_);
-  return stream_id_;
-}
-
-uint32_t IncomingVideoStream::IncomingRate() const {
-  CriticalSectionScoped cs(&stream_critsect_);
-  return incoming_rate_;
-}
-
-bool IncomingVideoStream::IncomingVideoStreamThreadFun(void* obj) {
-  return static_cast<IncomingVideoStream*>(obj)->IncomingVideoStreamProcess();
-}
-
-bool IncomingVideoStream::IncomingVideoStreamProcess() {
-  if (kEventError != deliver_buffer_event_.Wait(KEventMaxWaitTimeMs)) {
-    thread_critsect_.Enter();
-    if (incoming_render_thread_ == NULL) {
-      // Terminating
-      thread_critsect_.Leave();
-      return false;
-    }
-    // Get a new frame to render and the time for the frame after this one.
-    buffer_critsect_.Enter();
-    I420VideoFrame frame_to_render = render_buffers_.FrameToRender();
-    uint32_t wait_time = render_buffers_.TimeToNextFrameRelease();
-    buffer_critsect_.Leave();
-
-    // Set timer for next frame to render.
-    if (wait_time > KEventMaxWaitTimeMs) {
-      wait_time = KEventMaxWaitTimeMs;
-    }
-    deliver_buffer_event_.StartTimer(false, wait_time);
-
-    if (frame_to_render.IsZeroSize()) {
-      if (render_callback_) {
-        if (last_render_time_ms_ == 0 && !start_image_.IsZeroSize()) {
-          // We have not rendered anything and have a start image.
-          temp_frame_.CopyFrame(start_image_);
-          render_callback_->RenderFrame(stream_id_, temp_frame_);
-        } else if (!timeout_image_.IsZeroSize() &&
-                   last_render_time_ms_ + timeout_time_ <
-                       TickTime::MillisecondTimestamp()) {
-          // Render a timeout image.
-          temp_frame_.CopyFrame(timeout_image_);
-          render_callback_->RenderFrame(stream_id_, temp_frame_);
-        }
-      }
-
-      // No frame.
-      thread_critsect_.Leave();
-      return true;
-    }
-
-    // Send frame for rendering.
-    if (external_callback_) {
-      WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, module_id_,
-                   "%s: executing external renderer callback to deliver frame",
-                   __FUNCTION__, frame_to_render.render_time_ms());
-      external_callback_->RenderFrame(stream_id_, frame_to_render);
-    } else {
-      if (render_callback_) {
-        WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, module_id_,
-                     "%s: Render frame, time: ", __FUNCTION__,
-                     frame_to_render.render_time_ms());
-        render_callback_->RenderFrame(stream_id_, frame_to_render);
-      }
-    }
-
-    // Release critsect before calling the module user.
-    thread_critsect_.Leave();
-
-    // We're done with this frame.
-    if (!frame_to_render.IsZeroSize()) {
-      CriticalSectionScoped cs(&buffer_critsect_);
-      last_render_time_ms_= frame_to_render.render_time_ms();
-    }
-  }
-  return true;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/incoming_video_stream.h b/media/webrtc/trunk/webrtc/modules/video_render/incoming_video_stream.h
deleted file mode 100644
index c5a7d5f7c1..0000000000
--- a/media/webrtc/trunk/webrtc/modules/video_render/incoming_video_stream.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
-
-#include "webrtc/modules/video_render/include/video_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class VideoRenderCallback;
-class VideoRenderFrames;
-
-class IncomingVideoStream : public VideoRenderCallback {
- public:
-  IncomingVideoStream(const int32_t module_id,
-                      const uint32_t stream_id);
-  ~IncomingVideoStream();
-
-  int32_t ChangeModuleId(const int32_t id);
-
-  // Get callback to deliver frames to the module.
-  VideoRenderCallback* ModuleCallback();
-  virtual int32_t RenderFrame(const uint32_t stream_id,
-                              const I420VideoFrame& video_frame);
-
-  // Set callback to the platform dependent code.
-  int32_t SetRenderCallback(VideoRenderCallback* render_callback);
-
-  // Callback for file recording, snapshot, ...
-  int32_t SetExternalCallback(VideoRenderCallback* render_object);
-
-  // Start/Stop.
-  int32_t Start();
-  int32_t Stop();
-
-  // Clear all buffers.
-  int32_t Reset();
-
-  // Properties.
-  uint32_t StreamId() const;
-  uint32_t IncomingRate() const;
-
-  int32_t SetStartImage(const I420VideoFrame& video_frame);
-
-  int32_t SetTimeoutImage(const I420VideoFrame& video_frame,
-                          const uint32_t timeout);
-
-  int32_t SetExpectedRenderDelay(int32_t delay_ms);
-
- protected:
-  static bool IncomingVideoStreamThreadFun(void* obj);
-  bool IncomingVideoStreamProcess();
-
- private:
-  enum { KEventStartupTimeMS = 10 };
-  enum { KEventMaxWaitTimeMs = 100 };
-  enum { KFrameRatePeriodMs = 1000 };
-
-  int32_t module_id_;
-  uint32_t stream_id_;
-  // Critsects in allowed to enter order.
-  CriticalSectionWrapper& stream_critsect_;
-  CriticalSectionWrapper& thread_critsect_;
-  CriticalSectionWrapper& buffer_critsect_;
-  rtc::scoped_ptr<ThreadWrapper> incoming_render_thread_;
-  EventWrapper& deliver_buffer_event_;
-  bool running_;
-
-  VideoRenderCallback* external_callback_;
-  VideoRenderCallback* render_callback_;
-  VideoRenderFrames& render_buffers_;
-
-  RawVideoType callbackVideoType_;
-  uint32_t callbackWidth_;
-  uint32_t callbackHeight_;
-
-  uint32_t incoming_rate_;
-  int64_t last_rate_calculation_time_ms_;
-  uint16_t num_frames_since_last_calculation_;
-  int64_t last_render_time_ms_;
-  I420VideoFrame temp_frame_;
-  I420VideoFrame start_image_;
-  I420VideoFrame timeout_image_;
-  uint32_t timeout_time_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.h b/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.h
index f4235379c2..880ddb5231 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.h
@@ -13,10 +13,10 @@
 
 #include <OpenGLES/ES2/glext.h>
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 /*
- * This OpenGles20 is the class of renderer for I420VideoFrame into a GLES 2.0
+ * This OpenGles20 is the class of renderer for VideoFrame into a GLES 2.0
  * windows used in the VideoRenderIosView class.
  */
 namespace webrtc {
@@ -26,7 +26,7 @@ class OpenGles20 {
   ~OpenGles20();
 
   bool Setup(int32_t width, int32_t height);
-  bool Render(const I420VideoFrame& frame);
+  bool Render(const VideoFrame& frame);
 
   // SetCoordinates
   // Sets the coordinates where the stream shall be rendered.
@@ -45,10 +45,10 @@ class OpenGles20 {
   GLuint CreateProgram(const char* vertex_source, const char* fragment_source);
 
   // Initialize the textures by the frame width and height
-  void SetupTextures(const I420VideoFrame& frame);
+  void SetupTextures(const VideoFrame& frame);
 
   // Update the textures by the YUV data from the frame
-  void UpdateTextures(const I420VideoFrame& frame);
+  void UpdateTextures(const VideoFrame& frame);
 
   GLuint texture_ids_[3];  // Texture id of Y,U and V texture.
   GLuint program_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.mm b/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.mm
index 1b7a6d1754..d1735280f2 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/open_gles20.mm
@@ -17,7 +17,7 @@
 
 // TODO(sjlee): unify this copy with the android one.
 #include "webrtc/modules/video_render/ios/open_gles20.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
@@ -151,7 +151,7 @@ bool OpenGles20::SetCoordinates(const float z_order,
   return true;
 }
 
-bool OpenGles20::Render(const I420VideoFrame& frame) {
+bool OpenGles20::Render(const VideoFrame& frame) {
   if (texture_width_ != (GLsizei)frame.width() ||
       texture_height_ != (GLsizei)frame.height()) {
     SetupTextures(frame);
@@ -261,7 +261,7 @@ static void InitializeTexture(int name, int id, int width, int height) {
                NULL);
 }
 
-void OpenGles20::SetupTextures(const I420VideoFrame& frame) {
+void OpenGles20::SetupTextures(const VideoFrame& frame) {
   const GLsizei width = frame.width();
   const GLsizei height = frame.height();
 
@@ -310,7 +310,7 @@ static void GlTexSubImage2D(GLsizei width,
   }
 }
 
-void OpenGles20::UpdateTextures(const I420VideoFrame& frame) {
+void OpenGles20::UpdateTextures(const VideoFrame& frame) {
   const GLsizei width = frame.width();
   const GLsizei height = frame.height();
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.h b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.h
index 97f1efdd8f..a15ba393dc 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_view.h"
 
 namespace webrtc {
@@ -25,7 +25,7 @@ class VideoRenderIosChannel : public VideoRenderCallback {
 
   // Implementation of VideoRenderCallback.
   int32_t RenderFrame(const uint32_t stream_id,
-                      const I420VideoFrame& video_frame) override;
+                      const VideoFrame& video_frame) override;
 
   int SetStreamSettings(const float z_order,
                         const float left,
@@ -37,7 +37,7 @@ class VideoRenderIosChannel : public VideoRenderCallback {
 
  private:
   VideoRenderIosView* view_;
-  I420VideoFrame* current_frame_;
+  VideoFrame* current_frame_;
   bool buffer_is_updated_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.mm b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.mm
index 33f142367e..b2b15857f9 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_channel.mm
@@ -17,14 +17,13 @@
 using namespace webrtc;
 
 VideoRenderIosChannel::VideoRenderIosChannel(VideoRenderIosView* view)
-    : view_(view),
-      current_frame_(new I420VideoFrame()),
-      buffer_is_updated_(false) {}
+    : view_(view), current_frame_(new VideoFrame()), buffer_is_updated_(false) {
+}
 
 VideoRenderIosChannel::~VideoRenderIosChannel() { delete current_frame_; }
 
 int32_t VideoRenderIosChannel::RenderFrame(const uint32_t stream_id,
-                                           const I420VideoFrame& video_frame) {
+                                           const VideoFrame& video_frame) {
   current_frame_->CopyFrame(video_frame);
   current_frame_->set_render_time_ms(0);
   buffer_is_updated_ = true;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.h b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.h
index e0353aaea1..d703630d92 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.h
@@ -14,15 +14,15 @@
 #include <list>
 #include <map>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_channel.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_view.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
-class EventWrapper;
+class EventTimerWrapper;
 
 class VideoRenderIosGles20 {
  public:
@@ -63,8 +63,9 @@ class VideoRenderIosGles20 {
 
  private:
   rtc::scoped_ptr<CriticalSectionWrapper> gles_crit_sec_;
-  EventWrapper* screen_update_event_;
-  rtc::scoped_ptr<ThreadWrapper> screen_update_thread_;
+  EventTimerWrapper* screen_update_event_;
+  // TODO(pbos): Remove scoped_ptr and use member directly.
+  rtc::scoped_ptr<rtc::PlatformThread> screen_update_thread_;
 
   VideoRenderIosView* view_;
   Rect window_rect_;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.mm b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
index 35382ea814..6ad5db8b8c 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
@@ -13,8 +13,8 @@
 #endif
 
 #include "webrtc/modules/video_render/ios/video_render_ios_gles20.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 
 using namespace webrtc;
 
@@ -32,15 +32,15 @@ VideoRenderIosGles20::VideoRenderIosGles20(VideoRenderIosView* view,
       z_order_to_channel_(),
       gles_context_([view context]),
       is_rendering_(true) {
-  screen_update_thread_ = ThreadWrapper::CreateThread(
-      ScreenUpdateThreadProc, this, "ScreenUpdateGles20");
-  screen_update_event_ = EventWrapper::Create();
+  screen_update_thread_.reset(new rtc::PlatformThread(
+      ScreenUpdateThreadProc, this, "ScreenUpdateGles20"));
+  screen_update_event_ = EventTimerWrapper::Create();
   GetWindowRect(window_rect_);
 }
 
 VideoRenderIosGles20::~VideoRenderIosGles20() {
   // Signal event to exit thread, then delete it
-  ThreadWrapper* thread_wrapper = screen_update_thread_.release();
+  rtc::PlatformThread* thread_wrapper = screen_update_thread_.release();
 
   if (thread_wrapper) {
     screen_update_event_->Set();
@@ -83,7 +83,7 @@ int VideoRenderIosGles20::Init() {
   }
 
   screen_update_thread_->Start();
-  screen_update_thread_->SetPriority(kRealtimePriority);
+  screen_update_thread_->SetPriority(rtc::kRealtimePriority);
 
   // Start the event triggering the render process
   unsigned int monitor_freq = 60;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_impl.mm b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_impl.mm
index 49bae089a7..0ef411d56f 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_impl.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_impl.mm
@@ -14,8 +14,8 @@
 
 #include "webrtc/modules/video_render/ios/video_render_ios_impl.h"
 #include "webrtc/modules/video_render/ios/video_render_ios_gles20.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.h b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.h
index 915c0f71e2..d110bc78bd 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.h
@@ -20,7 +20,7 @@
 
 - (BOOL)createContext;
 - (BOOL)presentFramebuffer;
-- (BOOL)renderFrame:(webrtc::I420VideoFrame*)frameToRender;
+- (BOOL)renderFrame:(webrtc::VideoFrame*)frameToRender;
 - (BOOL)setCoordinatesForZOrder:(const float)zOrder
                            Left:(const float)left
                             Top:(const float)top
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.mm b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.mm
index 6c1ae314e8..6ffe976c99 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/ios/video_render_ios_view.mm
@@ -13,7 +13,7 @@
 #endif
 
 #include "webrtc/modules/video_render/ios/video_render_ios_view.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
@@ -142,7 +142,7 @@ using namespace webrtc;
   return YES;
 }
 
-- (BOOL)renderFrame:(I420VideoFrame*)frameToRender {
+- (BOOL)renderFrame:(VideoFrame*)frameToRender {
   if (![EAGLContext setCurrentContext:_context]) {
     return NO;
   }
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_render_linux_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_render_linux_impl.cc
index af0e9acb9b..7e53dfdf80 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_render_linux_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_render_linux_impl.cc
@@ -11,8 +11,8 @@
 #include "webrtc/modules/video_render/linux/video_render_linux_impl.h"
 
 #include "webrtc/modules/video_render/linux/video_x11_render.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #include <X11/Xlib.h>
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.cc b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.cc
index 92b990d6ab..8d86b7c72a 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.cc
@@ -10,8 +10,8 @@
 
 #include "webrtc/modules/video_render/linux/video_x11_channel.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -44,7 +44,7 @@ VideoX11Channel::~VideoX11Channel()
 }
 
 int32_t VideoX11Channel::RenderFrame(const uint32_t streamId,
-                                     const I420VideoFrame& videoFrame) {
+                                     const VideoFrame& videoFrame) {
   CriticalSectionScoped cs(&_crit);
   if (_width != videoFrame.width() || _height
       != videoFrame.height()) {
@@ -72,7 +72,7 @@ int32_t VideoX11Channel::FrameSizeChange(int32_t width,
     return 0;
 }
 
-int32_t VideoX11Channel::DeliverFrame(const I420VideoFrame& videoFrame) {
+int32_t VideoX11Channel::DeliverFrame(const VideoFrame& videoFrame) {
   CriticalSectionScoped cs(&_crit);
   if (!_prepared) {
     return 0;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.h b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.h
index 4a83a60884..6eb402e12e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_channel.h
@@ -13,7 +13,7 @@
 
 #include <sys/shm.h>
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #include <X11/Xlib.h>
 #include <X11/Xutil.h>
@@ -34,11 +34,11 @@ public:
     virtual ~VideoX11Channel();
 
     virtual int32_t RenderFrame(const uint32_t streamId,
-                                const I420VideoFrame& videoFrame);
+                                const VideoFrame& videoFrame);
 
     int32_t FrameSizeChange(int32_t width, int32_t height,
                             int32_t numberOfStreams);
-    int32_t DeliverFrame(const I420VideoFrame& videoFrame);
+    int32_t DeliverFrame(const VideoFrame& videoFrame);
     int32_t GetFrameSize(int32_t& width, int32_t& height);
     int32_t Init(Window window, float left, float top, float right,
                  float bottom);
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.cc b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.cc
index 4bccb3ccc6..5eb4f36f95 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.cc
@@ -11,8 +11,8 @@
 #include "webrtc/modules/video_render/linux/video_x11_channel.h"
 #include "webrtc/modules/video_render/linux/video_x11_render.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.h b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.h
index 265ef7cfab..23b83bd67b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/linux/video_x11_render.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #include <X11/Xlib.h>
 #include <map>
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm b/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm
index 31f6b64031..b57223b4df 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm
@@ -9,7 +9,7 @@
  */
 
 #include "webrtc/modules/video_render/mac/cocoa_full_screen_window.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_render_view.mm b/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_render_view.mm
index 86320bee1c..4631ff31a4 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_render_view.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/cocoa_render_view.mm
@@ -12,7 +12,7 @@
 #import <AppKit/AppKit.h>
 
 #include "webrtc/modules/video_render/mac/cocoa_render_view.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.cc b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.cc
index 6cd4173d8a..3243563b2b 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.cc
@@ -16,9 +16,9 @@
 
 //  includes
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -80,7 +80,7 @@ VideoChannelAGL::~VideoChannelAGL()
 }
 
 int32_t VideoChannelAGL::RenderFrame(const uint32_t streamId,
-                                     I420VideoFrame& videoFrame) {
+                                     VideoFrame& videoFrame) {
   _owner->LockAGLCntx();
   if (_width != videoFrame.width() ||
       _height != videoFrame.height()) {
@@ -219,7 +219,7 @@ int VideoChannelAGL::FrameSizeChange(int width, int height, int numberOfStreams)
 }
 
 // Called from video engine when a new frame should be rendered.
-int VideoChannelAGL::DeliverFrame(const I420VideoFrame& videoFrame) {
+int VideoChannelAGL::DeliverFrame(const VideoFrame& videoFrame) {
   _owner->LockAGLCntx();
 
   if (_texture == 0) {
@@ -395,8 +395,8 @@ _renderingIsPaused( false),
 {
     //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s");
 
-    _screenUpdateThread = ThreadWrapper::CreateThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdate");
+    _screenUpdateThread.reset(
+        new rtc::PlatformThread(ScreenUpdateThreadProc, this, "ScreenUpdate"));
     _screenUpdateEvent = EventWrapper::Create();
 
     if(!IsValidWindowPtr(_windowRef))
@@ -512,8 +512,8 @@ _renderingIsPaused( false),
     //WEBRTC_TRACE(kTraceDebug, "%s:%d Constructor", __FUNCTION__, __LINE__);
     //    _renderCritSec = CriticalSectionWrapper::CreateCriticalSection();
 
-    _screenUpdateThread = ThreadWrapper::CreateThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdateThread");
+    _screenUpdateThread.reset(new rtc::PlatformThread(
+        ScreenUpdateThreadProc, this, "ScreenUpdateThread"));
     _screenUpdateEvent = EventWrapper::Create();
 
     GetWindowRect(_windowRect);
@@ -677,7 +677,7 @@ VideoRenderAGL::~VideoRenderAGL()
 #endif
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
@@ -739,7 +739,7 @@ int VideoRenderAGL::Init()
         return -1;
     }
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
@@ -856,7 +856,7 @@ int VideoRenderAGL::DeleteAGLChannel(int channel)
 int VideoRenderAGL::StopThread()
 {
     CriticalSectionScoped cs(&_renderCritSec);
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
@@ -1880,7 +1880,7 @@ int32_t VideoRenderAGL::StartRender()
             UnlockAGLCntx();
             return -1;
         }
-        _screenUpdateThread->SetPriority(kRealtimePriority);
+        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
         if(FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
         {
             //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateEvent", __FUNCTION__, __LINE__);
@@ -1891,8 +1891,8 @@ int32_t VideoRenderAGL::StartRender()
         return 0;
     }
 
-    _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
-        this, "ScreenUpdate");
+    _screenUpdateThread.reset(
+        new rtc::PlatformThread(ScreenUpdateThreadProc, this, "ScreenUpdate"));
     _screenUpdateEvent = EventWrapper::Create();
 
     if (!_screenUpdateThread)
@@ -1903,14 +1903,13 @@ int32_t VideoRenderAGL::StartRender()
     }
 
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
     _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ);
 
     //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Started screenUpdateThread", __FUNCTION__, __LINE__);
 
     UnlockAGLCntx();
     return 0;
-
 }
 
 int32_t VideoRenderAGL::StopRender()
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.h b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.h
index effd334e54..e1da8faf83 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_agl.h
@@ -15,8 +15,8 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #define NEW_HIVIEW_PARENT_EVENT_HANDLER 1
 #define NEW_HIVIEW_EVENT_HANDLER 1
@@ -42,7 +42,7 @@ class VideoChannelAGL : public VideoRenderCallback {
   VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner);
   virtual ~VideoChannelAGL();
   virtual int FrameSizeChange(int width, int height, int numberOfStreams);
-  virtual int DeliverFrame(const I420VideoFrame& videoFrame);
+  virtual int DeliverFrame(const VideoFrame& videoFrame);
   virtual int UpdateSize(int width, int height);
   int SetStreamSettings(int streamId, float startWidth, float startHeight,
                         float stopWidth, float stopHeight);
@@ -51,8 +51,7 @@ class VideoChannelAGL : public VideoRenderCallback {
   int RenderOffScreenBuffer();
   int IsUpdated(bool& isUpdated);
   virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
-  virtual int32_t RenderFrame(const uint32_t streamId,
-                              I420VideoFrame& videoFrame);
+  virtual int32_t RenderFrame(const uint32_t streamId, VideoFrame& videoFrame);
 
  private:
 
@@ -143,7 +142,8 @@ class VideoRenderAGL {
   bool _fullScreen;
   int _id;
   webrtc::CriticalSectionWrapper& _renderCritSec;
-  rtc::scoped_ptr<webrtc::ThreadWrapper> _screenUpdateThread;
+  // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+  rtc::scoped_ptr<rtc::PlatformThread> _screenUpdateThread;
   webrtc::EventWrapper* _screenUpdateEvent;
   bool _isHIViewRef;
   AGLContext _aglContext;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc
index dbb09a35f5..f85be5fb5e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc
@@ -14,8 +14,8 @@
 #include <AGL/agl.h>
 #include "webrtc/modules/video_render/mac/video_render_agl.h"
 #include "webrtc/modules/video_render/mac/video_render_mac_carbon_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm
index 561d71fc37..5b017fecc0 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm
@@ -14,8 +14,8 @@
 #include "webrtc/modules/video_render/mac/cocoa_render_view.h"
 #include "webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.h"
 #include "webrtc/modules/video_render/mac/video_render_nsopengl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.h b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.h
index 867b4a572d..a888b68a97 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.h
@@ -23,29 +23,29 @@
 #include <map>
 
 #include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #import "webrtc/modules/video_render/mac/cocoa_full_screen_window.h"
 #import "webrtc/modules/video_render/mac/cocoa_render_view.h"
 
 class Trace;
 
+namespace rtc {
+class PlatformThread;
+}  // namespace rtc
+
 namespace webrtc {
-class EventWrapper;
-class ThreadWrapper;
+class EventTimerWrapper;
 class VideoRenderNSOpenGL;
 class CriticalSectionWrapper;
 
-class VideoChannelNSOpenGL : public VideoRenderCallback
-{
-
+class VideoChannelNSOpenGL : public VideoRenderCallback {
 public:
-
     VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner);
     virtual ~VideoChannelNSOpenGL();
 
     // A new frame is delivered
-    virtual int DeliverFrame(const I420VideoFrame& videoFrame);
+    virtual int DeliverFrame(const VideoFrame& videoFrame);
 
     // Called when the incoming frame size and/or number of streams in mix
     // changes.
@@ -66,7 +66,7 @@ public:
 
     // ********** new module functions ************ //
     virtual int32_t RenderFrame(const uint32_t streamId,
-                                const I420VideoFrame& videoFrame);
+                                const VideoFrame& videoFrame);
 
     // ********** new module helper functions ***** //
     int ChangeContext(NSOpenGLContext *nsglContext);
@@ -169,8 +169,9 @@ private: // variables
     bool _fullScreen;
     int _id;
     CriticalSectionWrapper& _nsglContextCritSec;
-    rtc::scoped_ptr<ThreadWrapper> _screenUpdateThread;
-    EventWrapper* _screenUpdateEvent;
+    // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+    rtc::scoped_ptr<rtc::PlatformThread> _screenUpdateThread;
+    EventTimerWrapper* _screenUpdateEvent;
     NSOpenGLContext* _nsglContext;
     NSOpenGLContext* _nsglFullScreenContext;
     CocoaFullScreenWindow* _fullScreenWindow;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.mm b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.mm
index d9cc8a4226..b7683a96af 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/mac/video_render_nsopengl.mm
@@ -11,12 +11,12 @@
 #include "webrtc/engine_configurations.h"
 #if defined(COCOA_RENDERING)
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_render/mac/video_render_nsopengl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -89,9 +89,8 @@ int32_t VideoChannelNSOpenGL::GetChannelProperties(float& left, float& top,
     return 0;
 }
 
-int32_t VideoChannelNSOpenGL::RenderFrame(
-  const uint32_t /*streamId*/, const I420VideoFrame& videoFrame) {
-
+int32_t VideoChannelNSOpenGL::RenderFrame(const uint32_t /*streamId*/,
+                                          const VideoFrame& videoFrame) {
   _owner->LockAGLCntx();
 
   if(_width != videoFrame.width() ||
@@ -206,8 +205,7 @@ int VideoChannelNSOpenGL::FrameSizeChange(int width, int height, int numberOfStr
     return 0;
 }
 
-int VideoChannelNSOpenGL::DeliverFrame(const I420VideoFrame& videoFrame) {
-
+int VideoChannelNSOpenGL::DeliverFrame(const VideoFrame& videoFrame) {
   _owner->LockAGLCntx();
 
   if (_texture == 0) {
@@ -221,7 +219,7 @@ int VideoChannelNSOpenGL::DeliverFrame(const I420VideoFrame& videoFrame) {
     return -1;
   }
 
-  // Using the I420VideoFrame for YV12: YV12 is YVU; I420 assumes
+  // Using the VideoFrame for YV12: YV12 is YVU; I420 assumes
   // YUV.
   // TODO(mikhal) : Use appropriate functionality.
   // TODO(wu): See if we are using glTexSubImage2D correctly.
@@ -367,7 +365,7 @@ _windowRef( (CocoaRenderView*)windowRef),
 _fullScreen( fullScreen),
 _id( iId),
 _nsglContextCritSec( *CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateEvent( 0),
+_screenUpdateEvent(EventTimerWrapper::Create()),
 _nsglContext( 0),
 _nsglFullScreenContext( 0),
 _fullScreenWindow( nil),
@@ -380,9 +378,8 @@ _renderingIsPaused (FALSE),
 _windowRefSuperView(NULL),
 _windowRefSuperViewFrame(NSMakeRect(0,0,0,0))
 {
-    _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
-            this, "ScreenUpdateNSOpenGL");
-    _screenUpdateEvent = EventWrapper::Create();
+  _screenUpdateThread.reset(new rtc::PlatformThread(
+      ScreenUpdateThreadProc, this, "ScreenUpdateNSOpenGL"));
 }
 
 int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef)
@@ -430,15 +427,15 @@ int32_t VideoRenderNSOpenGL::StartRender()
         WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Restarting screenUpdateThread");
 
         // we already have the thread. Most likely StopRender() was called and they were paused
-        if(FALSE == _screenUpdateThread->Start() ||
-                FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
-        {
+        _screenUpdateThread->Start();
+        if (FALSE ==
+            _screenUpdateEvent->StartTimer(true, 1000 / MONITOR_FREQ)) {
             WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "Failed to restart screenUpdateThread or screenUpdateEvent");
             UnlockAGLCntx();
             return -1;
         }
 
-        _screenUpdateThread->SetPriority(kRealtimePriority);
+        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
         UnlockAGLCntx();
         return 0;
@@ -474,8 +471,8 @@ int32_t VideoRenderNSOpenGL::StopRender()
         return 0;
     }
 
-    if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
-    {
+    _screenUpdateThread->Stop();
+    if (FALSE == _screenUpdateEvent->StopTimer()) {
         _renderingIsPaused = FALSE;
 
         UnlockAGLCntx();
@@ -660,17 +657,15 @@ VideoRenderNSOpenGL::~VideoRenderNSOpenGL()
     }
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
 
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
         _screenUpdateEvent->StopTimer();
 
-        if (tmpPtr->Stop())
-        {
-            delete tmpPtr;
-        }
+        tmpPtr->Stop();
+        delete tmpPtr;
         delete _screenUpdateEvent;
         _screenUpdateEvent = NULL;
     }
@@ -719,7 +714,7 @@ int VideoRenderNSOpenGL::Init()
     }
 
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
@@ -867,17 +862,15 @@ int32_t VideoRenderNSOpenGL::GetChannelProperties(const uint16_t streamId,
 int VideoRenderNSOpenGL::StopThread()
 {
 
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
     WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
                  "%s Stopping thread ", __FUNCTION__, tmpPtr);
 
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
-        if (tmpPtr->Stop())
-        {
-            delete tmpPtr;
-        }
+        tmpPtr->Stop();
+        delete tmpPtr;
     }
 
     delete _screenUpdateEvent;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.cc b/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.cc
index 14f9791f83..06ea00b5ad 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.cc
@@ -32,18 +32,17 @@
 #endif
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_render/video_render.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
-void GetTestVideoFrame(I420VideoFrame* frame,
-                       uint8_t startColor);
+void GetTestVideoFrame(VideoFrame* frame, uint8_t startColor);
 int TestSingleStream(VideoRender* renderModule);
 int TestFullscreenStream(VideoRender* &renderModule,
                          void* window,
@@ -262,8 +261,7 @@ public:
     }
     ;
     virtual int32_t RenderFrame(const uint32_t streamId,
-                                const I420VideoFrame& videoFrame)
-    {
+                                const VideoFrame& videoFrame) {
         _cnt++;
         if (_cnt % 100 == 0)
         {
@@ -274,8 +272,7 @@ public:
     int32_t _cnt;
 };
 
-void GetTestVideoFrame(I420VideoFrame* frame,
-                       uint8_t startColor) {
+void GetTestVideoFrame(VideoFrame* frame, uint8_t startColor) {
     // changing color
     static uint8_t color = startColor;
 
@@ -308,7 +305,7 @@ int TestSingleStream(VideoRender* renderModule) {
     const int half_width = (width + 1) / 2;
     const int height = 288;
 
-    I420VideoFrame videoFrame0;
+    VideoFrame videoFrame0;
     videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
 
     const uint32_t renderDelayMs = 500;
@@ -382,7 +379,7 @@ int TestBitmapText(VideoRender* renderModule) {
     const int half_width = (width + 1) / 2;
     const int height = 288;
 
-    I420VideoFrame videoFrame0;
+    VideoFrame videoFrame0;
     videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
 
     const uint32_t renderDelayMs = 500;
@@ -460,13 +457,13 @@ int TestMultipleStreams(VideoRender* renderModule) {
     const int half_width = (width + 1) / 2;
     const int height = 288;
 
-    I420VideoFrame videoFrame0;
+    VideoFrame videoFrame0;
     videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
-    I420VideoFrame videoFrame1;
+    VideoFrame videoFrame1;
     videoFrame1.CreateEmptyFrame(width, height, width, half_width, half_width);
-    I420VideoFrame videoFrame2;
+    VideoFrame videoFrame2;
     videoFrame2.CreateEmptyFrame(width, height, width, half_width, half_width);
-    I420VideoFrame videoFrame3;
+    VideoFrame videoFrame3;
     videoFrame3.CreateEmptyFrame(width, height, width, half_width, half_width);
 
     const uint32_t renderDelayMs = 500;
@@ -542,7 +539,7 @@ int TestExternalRender(VideoRender* renderModule) {
     const int width = 352;
     const int half_width = (width + 1) / 2;
     const int height = 288;
-    I420VideoFrame videoFrame0;
+    VideoFrame videoFrame0;
     videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
 
     const uint32_t renderDelayMs = 500;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.h b/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.h
index 8b14e84931..0655a5b434 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
 
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 void RunVideoRenderTests(void* window, webrtc::VideoRenderType windowType);
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm b/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
index 8ebb9ce609..dfee4c7298 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
+++ b/media/webrtc/trunk/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
@@ -20,12 +20,12 @@
 
 #import "webrtc/modules/video_render/mac/cocoa_render_view.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_render/video_render.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 using namespace webrtc;
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render.gypi b/media/webrtc/trunk/webrtc/modules/video_render/video_render.gypi
index 12ef354ee5..4bd73a0159 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/video_render.gypi
+++ b/media/webrtc/trunk/webrtc/modules/video_render/video_render.gypi
@@ -25,12 +25,8 @@
         'external/video_render_external_impl.cc',
         'external/video_render_external_impl.h',
         'i_video_render.h',
-        'include/video_render.h',
-        'include/video_render_defines.h',
-        'incoming_video_stream.cc',
-        'incoming_video_stream.h',
-        'video_render_frames.cc',
-        'video_render_frames.h',
+        'video_render.h',
+        'video_render_defines.h',
         'video_render_impl.h',
       ],
     },
@@ -49,7 +45,7 @@
   ], # targets
 
   'conditions': [
-    ['build_with_chromium==0', {
+    ['build_with_chromium==0 and build_with_mozilla==0', {
       'targets': [
         {
           # video_render_module implementation that supports the internal
@@ -150,16 +146,31 @@
                 'windows/video_render_windows_impl.cc',
               ],
               'include_dirs': [
-                '<(directx_sdk_path)/Include',
+#                '<(directx_sdk_path)/Include',
               ],
             }],
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-comment',
+                    '-Wno-reorder',
+                    '-Wno-unused-value',
+                    '-Wno-unused-private-field',
+                  ],
+                },
+              },
+            }],
           ] # conditions
         },
       ],
     }], # build_with_chromium==0
-    ['include_tests==1', {
+    ['include_tests==1 and OS!="ios"', {
       'targets': [
         {
+          # Does not compile on iOS: webrtc:4755.
           'target_name': 'video_render_tests',
           'type': 'executable',
           'dependencies': [
@@ -201,26 +212,7 @@
           ] # conditions
         }, # video_render_module_test
       ], # targets
-      'conditions': [
-        ['test_isolation_mode != "noop"', {
-          'targets': [
-            {
-              'target_name': 'video_render_tests_run',
-              'type': 'none',
-              'dependencies': [
-                'video_render_tests',
-              ],
-              'includes': [
-                '../../build/isolate.gypi',
-              ],
-              'sources': [
-                'video_render_tests.isolate',
-              ],
-            },
-          ],
-        }],
-      ],
-    }], # include_tests==1
+    }], # include_tests==1 and OS!=ios
   ], # conditions
 }
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/include/video_render.h b/media/webrtc/trunk/webrtc/modules/video_render/video_render.h
similarity index 94%
rename from media/webrtc/trunk/webrtc/modules/video_render/include/video_render.h
rename to media/webrtc/trunk/webrtc/modules/video_render/video_render.h
index a70c720ca6..a193a187e7 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/include/video_render.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/video_render.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
 
 /*
  * video_render.h
@@ -20,8 +20,8 @@
  *
  */
 
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 namespace webrtc {
 
@@ -254,16 +254,15 @@ public:
     /*
      * Set a start image. The image is rendered before the first image has been delivered
      */
-    virtual int32_t
-            SetStartImage(const uint32_t streamId,
-                          const I420VideoFrame& videoFrame) = 0;
+    virtual int32_t SetStartImage(const uint32_t streamId,
+                                  const VideoFrame& videoFrame) = 0;
 
     /*
      * Set a timout image. The image is rendered if no videoframe has been delivered
      */
     virtual int32_t SetTimeoutImage(const uint32_t streamId,
-                                    const I420VideoFrame& videoFrame,
-                                    const uint32_t timeout)= 0;
+                                    const VideoFrame& videoFrame,
+                                    const uint32_t timeout) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
+#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/include/video_render_defines.h b/media/webrtc/trunk/webrtc/modules/video_render/video_render_defines.h
similarity index 66%
rename from media/webrtc/trunk/webrtc/modules/video_render/include/video_render_defines.h
rename to media/webrtc/trunk/webrtc/modules/video_render/video_render_defines.h
index 4eb8409fca..999707cb6e 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/include/video_render_defines.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/video_render_defines.h
@@ -8,13 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
+#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
+#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
 
-// Includes
 #include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc
 {
@@ -43,20 +42,6 @@ enum VideoRenderError
     kRenderPerformanceAlarm = 1
 };
 
-// The object a module user uses to send new frames to the renderer
-// One object is used for each incoming stream
-class VideoRenderCallback
-{
-public:
-    virtual int32_t RenderFrame(const uint32_t streamId,
-                                const I420VideoFrame& videoFrame) = 0;
-
-protected:
-    virtual ~VideoRenderCallback()
-    {
-    }
-};
-
 // Feedback class to be implemented by module user
 class VideoRenderFeedback
 {
@@ -82,4 +67,4 @@ enum StretchMode
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
+#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.cc
index 283a6d2ae3..d2a074b4c4 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.cc
@@ -10,14 +10,14 @@
 
 #include <assert.h>
 
+#include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/video_render/external/video_render_external_impl.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/modules/video_render/incoming_video_stream.h"
 #include "webrtc/modules/video_render/i_video_render.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/modules/video_render/video_render_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -197,27 +197,10 @@ ModuleVideoRenderImpl::AddIncomingRenderStream(const uint32_t streamId,
     }
 
     // Create platform independant code
-    IncomingVideoStream* ptrIncomingStream = new IncomingVideoStream(_id,
-                                                                     streamId);
-    if (ptrIncomingStream == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Can't create incoming stream", __FUNCTION__);
-        return NULL;
-    }
-
-
-    if (ptrIncomingStream->SetRenderCallback(ptrRenderCallback) == -1)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Can't set render callback", __FUNCTION__);
-        delete ptrIncomingStream;
-        _ptrRenderer->DeleteIncomingRenderStream(streamId);
-        return NULL;
-    }
-
-    VideoRenderCallback* moduleCallback =
-            ptrIncomingStream->ModuleCallback();
+    IncomingVideoStream* ptrIncomingStream =
+        new IncomingVideoStream(streamId, false);
+    ptrIncomingStream->SetRenderCallback(ptrRenderCallback);
+    VideoRenderCallback* moduleCallback = ptrIncomingStream->ModuleCallback();
 
     // Store the stream
     _streamRenderMap[streamId] = ptrIncomingStream;
@@ -273,7 +256,8 @@ int32_t ModuleVideoRenderImpl::AddExternalRenderCallback(
                      "%s: could not get stream", __FUNCTION__);
         return -1;
     }
-    return item->second->SetExternalCallback(renderObject);
+    item->second->SetExternalCallback(renderObject);
+    return 0;
 }
 
 int32_t ModuleVideoRenderImpl::GetIncomingRenderStreamProperties(
@@ -567,10 +551,8 @@ int32_t ModuleVideoRenderImpl::ConfigureRenderer(
                                            bottom);
 }
 
-int32_t ModuleVideoRenderImpl::SetStartImage(
-    const uint32_t streamId,
-    const I420VideoFrame& videoFrame)
-{
+int32_t ModuleVideoRenderImpl::SetStartImage(const uint32_t streamId,
+                                             const VideoFrame& videoFrame) {
     CriticalSectionScoped cs(&_moduleCrit);
 
     if (!_ptrRenderer)
@@ -594,11 +576,9 @@ int32_t ModuleVideoRenderImpl::SetStartImage(
 
 }
 
-int32_t ModuleVideoRenderImpl::SetTimeoutImage(
-    const uint32_t streamId,
-    const I420VideoFrame& videoFrame,
-    const uint32_t timeout)
-{
+int32_t ModuleVideoRenderImpl::SetTimeoutImage(const uint32_t streamId,
+                                               const VideoFrame& videoFrame,
+                                               const uint32_t timeout) {
     CriticalSectionScoped cs(&_moduleCrit);
 
     if (!_ptrRenderer)
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.h b/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.h
index 278f543012..ce93cea6b5 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/video_render_impl.h
@@ -14,7 +14,7 @@
 #include <map>
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/modules/video_render/video_render.h"
 
 namespace webrtc {
 class CriticalSectionWrapper;
@@ -193,10 +193,10 @@ public:
                             const float right, const float bottom);
 
     virtual int32_t SetStartImage(const uint32_t streamId,
-                                  const I420VideoFrame& videoFrame);
+                                  const VideoFrame& videoFrame);
 
     virtual int32_t SetTimeoutImage(const uint32_t streamId,
-                                    const I420VideoFrame& videoFrame,
+                                    const VideoFrame& videoFrame,
                                     const uint32_t timeout);
 
 private:
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render_internal_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/video_render_internal_impl.cc
index 60934b7f6c..1fed26e9c4 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/video_render_internal_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/video_render_internal_impl.cc
@@ -10,13 +10,13 @@
 
 #include <assert.h>
 
+#include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/modules/video_render/incoming_video_stream.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/modules/video_render/video_render_impl.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #if defined (_WIN32)
 #include "webrtc/modules/video_render/windows/video_render_windows_impl.h"
@@ -420,27 +420,10 @@ ModuleVideoRenderImpl::AddIncomingRenderStream(const uint32_t streamId,
     }
 
     // Create platform independant code
-    IncomingVideoStream* ptrIncomingStream = new IncomingVideoStream(_id,
-                                                                     streamId);
-    if (ptrIncomingStream == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Can't create incoming stream", __FUNCTION__);
-        return NULL;
-    }
-
-
-    if (ptrIncomingStream->SetRenderCallback(ptrRenderCallback) == -1)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Can't set render callback", __FUNCTION__);
-        delete ptrIncomingStream;
-        _ptrRenderer->DeleteIncomingRenderStream(streamId);
-        return NULL;
-    }
-
-    VideoRenderCallback* moduleCallback =
-            ptrIncomingStream->ModuleCallback();
+    IncomingVideoStream* ptrIncomingStream =
+        new IncomingVideoStream(streamId, false);
+    ptrIncomingStream->SetRenderCallback(ptrRenderCallback);
+    VideoRenderCallback* moduleCallback = ptrIncomingStream->ModuleCallback();
 
     // Store the stream
     _streamRenderMap[streamId] = ptrIncomingStream;
@@ -496,7 +479,8 @@ int32_t ModuleVideoRenderImpl::AddExternalRenderCallback(
                      "%s: could not get stream", __FUNCTION__);
         return -1;
     }
-    return item->second->SetExternalCallback(renderObject);
+    item->second->SetExternalCallback(renderObject);
+    return 0;
 }
 
 int32_t ModuleVideoRenderImpl::GetIncomingRenderStreamProperties(
@@ -790,10 +774,8 @@ int32_t ModuleVideoRenderImpl::ConfigureRenderer(
                                            bottom);
 }
 
-int32_t ModuleVideoRenderImpl::SetStartImage(
-    const uint32_t streamId,
-    const I420VideoFrame& videoFrame)
-{
+int32_t ModuleVideoRenderImpl::SetStartImage(const uint32_t streamId,
+                                             const VideoFrame& videoFrame) {
     CriticalSectionScoped cs(&_moduleCrit);
 
     if (!_ptrRenderer)
@@ -817,11 +799,9 @@ int32_t ModuleVideoRenderImpl::SetStartImage(
 
 }
 
-int32_t ModuleVideoRenderImpl::SetTimeoutImage(
-    const uint32_t streamId,
-    const I420VideoFrame& videoFrame,
-    const uint32_t timeout)
-{
+int32_t ModuleVideoRenderImpl::SetTimeoutImage(const uint32_t streamId,
+                                               const VideoFrame& videoFrame,
+                                               const uint32_t timeout) {
     CriticalSectionScoped cs(&_moduleCrit);
 
     if (!_ptrRenderer)
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/windows/i_video_render_win.h b/media/webrtc/trunk/webrtc/modules/video_render/windows/i_video_render_win.h
index 56731e3770..6dbb4fd3cb 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/windows/i_video_render_win.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/windows/i_video_render_win.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
 #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
 
-#include "webrtc/modules/video_render/include/video_render.h"
+#include "webrtc/modules/video_render/video_render.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.cc b/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.cc
index 25f4df0c53..83835aebb8 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.cc
@@ -16,9 +16,9 @@
 
 // WebRtc include files
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -142,8 +142,7 @@ int D3D9Channel::FrameSizeChange(int width, int height, int numberOfStreams)
 }
 
 int32_t D3D9Channel::RenderFrame(const uint32_t streamId,
-                                 const I420VideoFrame& videoFrame)
-{
+                                 const VideoFrame& videoFrame) {
     CriticalSectionScoped cs(_critSect);
     if (_width != videoFrame.width() || _height != videoFrame.height())
     {
@@ -156,7 +155,7 @@ int32_t D3D9Channel::RenderFrame(const uint32_t streamId,
 }
 
 // Called from video engine when a new frame should be rendered.
-int D3D9Channel::DeliverFrame(const I420VideoFrame& videoFrame) {
+int D3D9Channel::DeliverFrame(const VideoFrame& videoFrame) {
   WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
                "DeliverFrame to D3D9Channel");
 
@@ -295,9 +294,9 @@ VideoRenderDirect3D9::VideoRenderDirect3D9(Trace* trace,
     _totalMemory(0),
     _availableMemory(0)
 {
-    _screenUpdateThread = ThreadWrapper::CreateThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdateThread");
-    _screenUpdateEvent = EventWrapper::Create();
+    _screenUpdateThread.reset(new rtc::PlatformThread(
+        ScreenUpdateThreadProc, this, "ScreenUpdateThread"));
+    _screenUpdateEvent = EventTimerWrapper::Create();
     SetRect(&_originalHwndRect, 0, 0, 0, 0);
 }
 
@@ -306,7 +305,7 @@ VideoRenderDirect3D9::~VideoRenderDirect3D9()
     //NOTE: we should not enter CriticalSection in here!
 
     // Signal event to exit thread, then delete it
-    ThreadWrapper* tmpPtr = _screenUpdateThread.release();
+    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
     if (tmpPtr)
     {
         _screenUpdateEvent->Set();
@@ -547,7 +546,7 @@ int32_t VideoRenderDirect3D9::Init()
         return -1;
     }
     _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(kRealtimePriority);
+    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
 
     // Start the event triggering the render process
     unsigned int monitorFreq = 60;
@@ -603,9 +602,6 @@ int VideoRenderDirect3D9::UpdateRenderSurface()
         _pd3dDevice->SetStreamSource(0, _pVB, 0, sizeof(CUSTOMVERTEX));
         _pd3dDevice->SetFVF(D3DFVF_CUSTOMVERTEX);
 
-        D3DXMATRIX matWorld;
-        D3DXMATRIX matWorldTemp;
-
         //draw all the channels
         //get texture from the channels
         LPDIRECT3DTEXTURE9 textureFromChannel = NULL;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.h b/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.h
index 6a631e9b3e..5a1f207934 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.h
+++ b/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_direct3d9.h
@@ -14,20 +14,19 @@
 #include "webrtc/modules/video_render/windows/i_video_render_win.h"
 
 #include <d3d9.h>
-#include <d3dx9.h>
 #include <ddraw.h>
 
 #include <Map>
 
 // Added
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
 
 #pragma comment(lib, "d3d9.lib")       // located in DirectX SDK
 
 namespace webrtc {
 class CriticalSectionWrapper;
-class EventWrapper;
+class EventTimerWrapper;
 class Trace;
 
 class D3D9Channel: public VideoRenderCallback
@@ -43,9 +42,9 @@ public:
     virtual int FrameSizeChange(int width, int height, int numberOfStreams);
 
     // A new frame is delivered.
-    virtual int DeliverFrame(const I420VideoFrame& videoFrame);
+    virtual int DeliverFrame(const VideoFrame& videoFrame);
     virtual int32_t RenderFrame(const uint32_t streamId,
-                                const I420VideoFrame& videoFrame);
+                                const VideoFrame& videoFrame);
 
     // Called to check if the video frame is updated.
     int IsUpdated(bool& isUpdated);
@@ -204,8 +203,9 @@ private:
 
     CriticalSectionWrapper& _refD3DCritsect;
     Trace* _trace;
-    rtc::scoped_ptr<ThreadWrapper> _screenUpdateThread;
-    EventWrapper* _screenUpdateEvent;
+    // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+    rtc::scoped_ptr<rtc::PlatformThread> _screenUpdateThread;
+    EventTimerWrapper* _screenUpdateEvent;
 
     HWND _hWnd;
     bool _fullScreen;
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_windows_impl.cc b/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_windows_impl.cc
index 38d897c0d3..042d7fdfa3 100644
--- a/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_windows_impl.cc
+++ b/media/webrtc/trunk/webrtc/modules/video_render/windows/video_render_windows_impl.cc
@@ -11,8 +11,8 @@
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/video_render/windows/video_render_windows_impl.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #ifdef DIRECT3D9_RENDERING
 #include "webrtc/modules/video_render/windows/video_render_direct3d9.h"
 #endif
diff --git a/media/webrtc/trunk/webrtc/modules/video_render/video_render_tests.isolate b/media/webrtc/trunk/webrtc/modules/video_render_tests.isolate
similarity index 100%
rename from media/webrtc/trunk/webrtc/modules/video_render/video_render_tests.isolate
rename to media/webrtc/trunk/webrtc/modules/video_render_tests.isolate
diff --git a/media/webrtc/trunk/webrtc/overrides/OWNERS b/media/webrtc/trunk/webrtc/overrides/OWNERS
deleted file mode 100644
index 9a527df143..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/OWNERS
+++ /dev/null
@@ -1,12 +0,0 @@
-henrika@webrtc.org
-henrikg@webrtc.org
-hta@webrtc.org
-jiayl@webrtc.org
-juberti@webrtc.org
-mflodman@webrtc.org
-perkj@webrtc.org
-pthatcher@webrtc.org
-sergeyu@chromium.org
-tommi@webrtc.org
-
-per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/basictypes.h b/media/webrtc/trunk/webrtc/overrides/webrtc/base/basictypes.h
deleted file mode 100644
index c32d8b6502..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/basictypes.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file overrides the inclusion of webrtc/base/basictypes.h to remove
-// collisions with Chromium's base/basictypes.h.   We then add back a few
-// items that Chromium's version doesn't provide, but libjingle expects.
-
-#ifndef OVERRIDES_WEBRTC_BASE_BASICTYPES_H__
-#define OVERRIDES_WEBRTC_BASE_BASICTYPES_H__
-
-#include "base/basictypes.h"
-#include "build/build_config.h"
-
-#ifndef INT_TYPES_DEFINED
-#define INT_TYPES_DEFINED
-
-#ifdef COMPILER_MSVC
-#if _MSC_VER >= 1600
-#include <stdint.h>
-#else
-typedef unsigned __int64 uint64;
-typedef __int64 int64;
-#endif
-#ifndef INT64_C
-#define INT64_C(x) x ## I64
-#endif
-#ifndef UINT64_C
-#define UINT64_C(x) x ## UI64
-#endif
-#define INT64_F "I64"
-#else  // COMPILER_MSVC
-#ifndef INT64_C
-#define INT64_C(x) x ## LL
-#endif
-#ifndef UINT64_C
-#define UINT64_C(x) x ## ULL
-#endif
-#ifndef INT64_F
-#define INT64_F "ll"
-#endif
-#endif  // COMPILER_MSVC
-#endif  // INT_TYPES_DEFINED
-
-// Detect compiler is for x86 or x64.
-#if defined(__x86_64__) || defined(_M_X64) || \
-    defined(__i386__) || defined(_M_IX86)
-#define CPU_X86 1
-#endif
-// Detect compiler is for arm.
-#if defined(__arm__) || defined(_M_ARM)
-#define CPU_ARM 1
-#endif
-#if defined(CPU_X86) && defined(CPU_ARM)
-#error CPU_X86 and CPU_ARM both defined.
-#endif
-#if !defined(ARCH_CPU_BIG_ENDIAN) && !defined(ARCH_CPU_LITTLE_ENDIAN)
-// x86, arm or GCC provided __BYTE_ORDER__ macros
-#if CPU_X86 || CPU_ARM ||  \
-  (defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
-#define ARCH_CPU_LITTLE_ENDIAN
-#elif defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
-#define ARCH_CPU_BIG_ENDIAN
-#else
-#error ARCH_CPU_BIG_ENDIAN or ARCH_CPU_LITTLE_ENDIAN should be defined.
-#endif
-#endif
-#if defined(ARCH_CPU_BIG_ENDIAN) && defined(ARCH_CPU_LITTLE_ENDIAN)
-#error ARCH_CPU_BIG_ENDIAN and ARCH_CPU_LITTLE_ENDIAN both defined.
-#endif
-
-#if defined(WEBRTC_WIN)
-typedef int socklen_t;
-#endif
-
-#if defined(WEBRTC_WIN)
-#if _MSC_VER < 1700
-  #define alignof(t) __alignof(t)
-#endif
-#else  // !WEBRTC_WIN
-#define alignof(t) __alignof__(t)
-#endif  // !WEBRTC_WIN
-#define RTC_IS_ALIGNED(p, a) (0==(reinterpret_cast<uintptr_t>(p) & ((a)-1)))
-#define ALIGNP(p, t) \
-  (reinterpret_cast<uint8*>(((reinterpret_cast<uintptr_t>(p) + \
-  ((t)-1)) & ~((t)-1))))
-
-// LIBJINGLE_DEFINE_STATIC_LOCAL() is a libjingle's copy
-// of CR_DEFINE_STATIC_LOCAL().
-#define LIBJINGLE_DEFINE_STATIC_LOCAL(type, name, arguments) \
-  CR_DEFINE_STATIC_LOCAL(type, name, arguments)
-
-#endif // OVERRIDES_WEBRTC_BASE_BASICTYPES_H__
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/constructormagic.h b/media/webrtc/trunk/webrtc/overrides/webrtc/base/constructormagic.h
deleted file mode 100644
index 72b334c18e..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/constructormagic.h
+++ /dev/null
@@ -1,20 +0,0 @@
-/*
- *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file overrides the inclusion of webrtc/base/constructormagic.h
-// We do this because constructor magic defines DISALLOW_EVIL_CONSTRUCTORS,
-// but we want to use the version from Chromium.
-
-#ifndef OVERRIDES_WEBRTC_BASE_CONSTRUCTORMAGIC_H__
-#define OVERRIDES_WEBRTC_BASE_CONSTRUCTORMAGIC_H__
-
-#include "base/macros.h"
-
-#endif  // OVERRIDES_WEBRTC_BASE_CONSTRUCTORMAGIC_H__
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/diagnostic_logging.h b/media/webrtc/trunk/webrtc/overrides/webrtc/base/diagnostic_logging.h
deleted file mode 100644
index 403bfc90be..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/diagnostic_logging.h
+++ /dev/null
@@ -1,156 +0,0 @@
-/*
- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef THIRD_PARTY_LIBJINGLE_OVERRIDES_WEBRTC_BASE_DIAGNOSTIC_LOGGING_H_
-#define THIRD_PARTY_LIBJINGLE_OVERRIDES_WEBRTC_BASE_DIAGNOSTIC_LOGGING_H_
-
-#include <sstream>
-#include <string>
-
-#include "base/logging.h"
-#include "third_party/webrtc/base/scoped_ref_ptr.h"
-
-namespace rtc {
-
-///////////////////////////////////////////////////////////////////////////////
-// ConstantLabel can be used to easily generate string names from constant
-// values.  This can be useful for logging descriptive names of error messages.
-// Usage:
-//   const ConstantLabel LIBRARY_ERRORS[] = {
-//     KLABEL(SOME_ERROR),
-//     KLABEL(SOME_OTHER_ERROR),
-//     ...
-//     LASTLABEL
-//   }
-//
-//   int err = LibraryFunc();
-//   LOG(LS_ERROR) << "LibraryFunc returned: "
-//                 << ErrorName(err, LIBRARY_ERRORS);
-
-struct ConstantLabel {
-  int value;
-  const char* label;
-};
-#define KLABEL(x) { x, #x }
-#define LASTLABEL { 0, 0 }
-
-const char* FindLabel(int value, const ConstantLabel entries[]);
-std::string ErrorName(int err, const ConstantLabel* err_table);
-
-//////////////////////////////////////////////////////////////////////
-// Note that the non-standard LoggingSeverity aliases exist because they are
-// still in broad use.  The meanings of the levels are:
-//  LS_SENSITIVE: Information which should only be logged with the consent
-//   of the user, due to privacy concerns.
-//  LS_VERBOSE: This level is for data which we do not want to appear in the
-//   normal debug log, but should appear in diagnostic logs.
-//  LS_INFO: Chatty level used in debugging for all sorts of things, the default
-//   in debug builds.
-//  LS_WARNING: Something that may warrant investigation.
-//  LS_ERROR: Something that should not have occurred.
-// Note that LoggingSeverity is mapped over to chromiums verbosity levels where
-// anything lower than or equal to the current verbosity level is written to
-// file which is the opposite of logging severity in libjingle where higher
-// severity numbers than or equal to the current severity level are written to
-// file. Also, note that the values are explicitly defined here for convenience
-// since the command line flag must be set using numerical values.
-enum LoggingSeverity { LS_ERROR = 1,
-                       LS_WARNING = 2,
-                       LS_INFO = 3,
-                       LS_VERBOSE = 4,
-                       LS_SENSITIVE = 5,
-                       INFO = LS_INFO,
-                       WARNING = LS_WARNING,
-                       LERROR = LS_ERROR };
-
-// LogErrorContext assists in interpreting the meaning of an error value.
-enum LogErrorContext {
-  ERRCTX_NONE,
-  ERRCTX_ERRNO,     // System-local errno
-  ERRCTX_HRESULT,   // Windows HRESULT
-  ERRCTX_OSSTATUS,  // MacOS OSStatus
-
-  // Abbreviations for LOG_E macro
-  ERRCTX_EN = ERRCTX_ERRNO,     // LOG_E(sev, EN, x)
-  ERRCTX_HR = ERRCTX_HRESULT,   // LOG_E(sev, HR, x)
-  ERRCTX_OS = ERRCTX_OSSTATUS,  // LOG_E(sev, OS, x)
-};
-
-// Class that writes a log message to the logging delegate ("WebRTC logging
-// stream" in Chrome) and to Chrome's logging stream.
-class DiagnosticLogMessage {
- public:
-  DiagnosticLogMessage(const char* file, int line, LoggingSeverity severity,
-                       bool log_to_chrome, LogErrorContext err_ctx, int err);
-  DiagnosticLogMessage(const char* file, int line, LoggingSeverity severity,
-                       bool log_to_chrome, LogErrorContext err_ctx, int err,
-                       const char* module);
-  ~DiagnosticLogMessage();
-
-  void CreateTimestamp();
-
-  std::ostream& stream() { return print_stream_; }
-
- private:
-  const char* file_name_;
-  const int line_;
-  const LoggingSeverity severity_;
-  const bool log_to_chrome_;
-
-  std::string extra_;
-
-  std::ostringstream print_stream_;
-};
-
-// This class is used to explicitly ignore values in the conditional
-// logging macros.  This avoids compiler warnings like "value computed
-// is not used" and "statement has no effect".
-class LogMessageVoidify {
- public:
-  LogMessageVoidify() { }
-  // This has to be an operator with a precedence lower than << but
-  // higher than ?:
-  void operator&(std::ostream&) { }
-};
-
-//////////////////////////////////////////////////////////////////////
-// Logging Helpers
-//////////////////////////////////////////////////////////////////////
-
-class LogMultilineState {
- public:
-  size_t unprintable_count_[2];
-  LogMultilineState() {
-    unprintable_count_[0] = unprintable_count_[1] = 0;
-  }
-};
-
-class LogMessage {
- public:
-  static void LogToDebug(int min_sev);
-};
-
-// When possible, pass optional state variable to track various data across
-// multiple calls to LogMultiline.  Otherwise, pass NULL.
-void LogMultiline(LoggingSeverity level, const char* label, bool input,
-                  const void* data, size_t len, bool hex_mode,
-                  LogMultilineState* state);
-
-// TODO(grunell): Change name to InitDiagnosticLoggingDelegate or
-// InitDiagnosticLogging. Change also in init_webrtc.h/cc.
-// TODO(grunell): typedef the delegate function.
-void InitDiagnosticLoggingDelegateFunction(
-    void (*delegate)(const std::string&));
-
-void SetExtraLoggingInit(
-    void (*function)(void (*delegate)(const std::string&)));
-}  // namespace rtc
-
-#endif  // THIRD_PARTY_LIBJINGLE_OVERRIDES_WEBRTC_BASE_DIAGNOSTIC_LOGGING_H_
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/logging.cc b/media/webrtc/trunk/webrtc/overrides/webrtc/base/logging.cc
deleted file mode 100644
index 20b3ba380f..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/logging.cc
+++ /dev/null
@@ -1,333 +0,0 @@
-/*
- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// IMPORTANT
-// Since this file includes Chromium source files, it must not include
-// logging.h since logging.h defines some of the same macros as Chrome does
-// and we'll run into conflict.
-
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-#include <CoreServices/CoreServices.h>
-#endif  // OS_MACOSX
-
-#include <algorithm>
-#include <iomanip>
-
-#include "base/atomicops.h"
-#include "base/strings/string_util.h"
-#include "base/threading/platform_thread.h"
-#include "third_party/webrtc/base/ipaddress.h"
-#include "third_party/webrtc/base/stream.h"
-#include "third_party/webrtc/base/stringencode.h"
-#include "third_party/webrtc/base/stringutils.h"
-#include "third_party/webrtc/base/timeutils.h"
-#include "third_party/webrtc/overrides/webrtc/base/diagnostic_logging.h"
-
-// From this file we can't use VLOG since it expands into usage of the __FILE__
-// macro (for correct filtering). The actual logging call from DIAGNOSTIC_LOG in
-// ~DiagnosticLogMessage. Note that the second parameter to the LAZY_STREAM
-// macro is true since the filter check has already been done for
-// DIAGNOSTIC_LOG.
-#define LOG_LAZY_STREAM_DIRECT(file_name, line_number, sev) \
-  LAZY_STREAM(logging::LogMessage(file_name, line_number, \
-                                  -sev).stream(), true)
-
-namespace rtc {
-
-void (*g_logging_delegate_function)(const std::string&) = NULL;
-void (*g_extra_logging_init_function)(
-    void (*logging_delegate_function)(const std::string&)) = NULL;
-#ifndef NDEBUG
-static_assert(sizeof(base::subtle::Atomic32) == sizeof(base::PlatformThreadId),
-              "Atomic32 not same size as PlatformThreadId");
-base::subtle::Atomic32 g_init_logging_delegate_thread_id = 0;
-#endif
-
-/////////////////////////////////////////////////////////////////////////////
-// Constant Labels
-/////////////////////////////////////////////////////////////////////////////
-
-const char* FindLabel(int value, const ConstantLabel entries[]) {
-  for (int i = 0; entries[i].label; ++i) {
-    if (value == entries[i].value) return entries[i].label;
-  }
-  return 0;
-}
-
-std::string ErrorName(int err, const ConstantLabel* err_table) {
-  if (err == 0)
-    return "No error";
-
-  if (err_table != 0) {
-    if (const char * value = FindLabel(err, err_table))
-      return value;
-  }
-
-  char buffer[16];
-  base::snprintf(buffer, sizeof(buffer), "0x%08x", err);
-  return buffer;
-}
-
-/////////////////////////////////////////////////////////////////////////////
-// Log helper functions
-/////////////////////////////////////////////////////////////////////////////
-
-// Generates extra information for LOG_E.
-static std::string GenerateExtra(LogErrorContext err_ctx,
-                                 int err,
-                                 const char* module) {
-  if (err_ctx != ERRCTX_NONE) {
-    std::ostringstream tmp;
-    tmp << ": ";
-    tmp << "[0x" << std::setfill('0') << std::hex << std::setw(8) << err << "]";
-    switch (err_ctx) {
-      case ERRCTX_ERRNO:
-        tmp << " " << strerror(err);
-        break;
-#if defined(WEBRTC_WIN)
-      case ERRCTX_HRESULT: {
-        char msgbuf[256];
-        DWORD flags = FORMAT_MESSAGE_FROM_SYSTEM;
-        HMODULE hmod = GetModuleHandleA(module);
-        if (hmod)
-          flags |= FORMAT_MESSAGE_FROM_HMODULE;
-        if (DWORD len = FormatMessageA(
-            flags, hmod, err,
-            MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT),
-            msgbuf, sizeof(msgbuf) / sizeof(msgbuf[0]), NULL)) {
-          while ((len > 0) &&
-              isspace(static_cast<unsigned char>(msgbuf[len-1]))) {
-            msgbuf[--len] = 0;
-          }
-          tmp << " " << msgbuf;
-        }
-        break;
-      }
-#endif  // OS_WIN
-#if defined(WEBRTC_IOS)
-      case ERRCTX_OSSTATUS:
-        tmp << " " << "Unknown LibJingle error: " << err;
-        break;
-#elif defined(WEBRTC_MAC)
-      case ERRCTX_OSSTATUS: {
-        tmp << " " << nonnull(GetMacOSStatusErrorString(err), "Unknown error");
-        if (const char* desc = GetMacOSStatusCommentString(err)) {
-          tmp << ": " << desc;
-        }
-        break;
-      }
-#endif  // OS_MACOSX
-      default:
-        break;
-    }
-    return tmp.str();
-  }
-  return "";
-}
-
-DiagnosticLogMessage::DiagnosticLogMessage(const char* file,
-                                           int line,
-                                           LoggingSeverity severity,
-                                           bool log_to_chrome,
-                                           LogErrorContext err_ctx,
-                                           int err)
-    : file_name_(file),
-      line_(line),
-      severity_(severity),
-      log_to_chrome_(log_to_chrome) {
-  extra_ = GenerateExtra(err_ctx, err, NULL);
-}
-
-DiagnosticLogMessage::DiagnosticLogMessage(const char* file,
-                                           int line,
-                                           LoggingSeverity severity,
-                                           bool log_to_chrome,
-                                           LogErrorContext err_ctx,
-                                           int err,
-                                           const char* module)
-    : file_name_(file),
-      line_(line),
-      severity_(severity),
-      log_to_chrome_(log_to_chrome) {
-  extra_ = GenerateExtra(err_ctx, err, module);
-}
-
-DiagnosticLogMessage::~DiagnosticLogMessage() {
-  const bool call_delegate =
-      g_logging_delegate_function && severity_ <= LS_INFO;
-
-  if (call_delegate || log_to_chrome_) {
-    print_stream_ << extra_;
-    const std::string& str = print_stream_.str();
-    if (log_to_chrome_)
-      LOG_LAZY_STREAM_DIRECT(file_name_, line_, severity_) << str;
-    if (g_logging_delegate_function && severity_ <= LS_INFO) {
-      g_logging_delegate_function(str);
-    }
-  }
-}
-
-// static
-void LogMessage::LogToDebug(int min_sev) {
-  logging::SetMinLogLevel(min_sev);
-}
-
-// Note: this function is a copy from the overriden libjingle implementation.
-void LogMultiline(LoggingSeverity level, const char* label, bool input,
-                  const void* data, size_t len, bool hex_mode,
-                  LogMultilineState* state) {
-  if (!LOG_CHECK_LEVEL_V(level))
-    return;
-
-  const char * direction = (input ? " << " : " >> ");
-
-  // NULL data means to flush our count of unprintable characters.
-  if (!data) {
-    if (state && state->unprintable_count_[input]) {
-      LOG_V(level) << label << direction << "## "
-                   << state->unprintable_count_[input]
-                   << " consecutive unprintable ##";
-      state->unprintable_count_[input] = 0;
-    }
-    return;
-  }
-
-  // The ctype classification functions want unsigned chars.
-  const unsigned char* udata = static_cast<const unsigned char*>(data);
-
-  if (hex_mode) {
-    const size_t LINE_SIZE = 24;
-    char hex_line[LINE_SIZE * 9 / 4 + 2], asc_line[LINE_SIZE + 1];
-    while (len > 0) {
-      memset(asc_line, ' ', sizeof(asc_line));
-      memset(hex_line, ' ', sizeof(hex_line));
-      size_t line_len = std::min(len, LINE_SIZE);
-      for (size_t i = 0; i < line_len; ++i) {
-        unsigned char ch = udata[i];
-        asc_line[i] = isprint(ch) ? ch : '.';
-        hex_line[i*2 + i/4] = hex_encode(ch >> 4);
-        hex_line[i*2 + i/4 + 1] = hex_encode(ch & 0xf);
-      }
-      asc_line[sizeof(asc_line)-1] = 0;
-      hex_line[sizeof(hex_line)-1] = 0;
-      LOG_V(level) << label << direction
-                   << asc_line << " " << hex_line << " ";
-      udata += line_len;
-      len -= line_len;
-    }
-    return;
-  }
-
-  size_t consecutive_unprintable = state ? state->unprintable_count_[input] : 0;
-
-  const unsigned char* end = udata + len;
-  while (udata < end) {
-    const unsigned char* line = udata;
-    const unsigned char* end_of_line = strchrn<unsigned char>(udata,
-                                                              end - udata,
-                                                              '\n');
-    if (!end_of_line) {
-      udata = end_of_line = end;
-    } else {
-      udata = end_of_line + 1;
-    }
-
-    bool is_printable = true;
-
-    // If we are in unprintable mode, we need to see a line of at least
-    // kMinPrintableLine characters before we'll switch back.
-    const ptrdiff_t kMinPrintableLine = 4;
-    if (consecutive_unprintable && ((end_of_line - line) < kMinPrintableLine)) {
-      is_printable = false;
-    } else {
-      // Determine if the line contains only whitespace and printable
-      // characters.
-      bool is_entirely_whitespace = true;
-      for (const unsigned char* pos = line; pos < end_of_line; ++pos) {
-        if (isspace(*pos))
-          continue;
-        is_entirely_whitespace = false;
-        if (!isprint(*pos)) {
-          is_printable = false;
-          break;
-        }
-      }
-      // Treat an empty line following unprintable data as unprintable.
-      if (consecutive_unprintable && is_entirely_whitespace) {
-        is_printable = false;
-      }
-    }
-    if (!is_printable) {
-      consecutive_unprintable += (udata - line);
-      continue;
-    }
-    // Print out the current line, but prefix with a count of prior unprintable
-    // characters.
-    if (consecutive_unprintable) {
-      LOG_V(level) << label << direction << "## " << consecutive_unprintable
-                  << " consecutive unprintable ##";
-      consecutive_unprintable = 0;
-    }
-    // Strip off trailing whitespace.
-    while ((end_of_line > line) && isspace(*(end_of_line-1))) {
-      --end_of_line;
-    }
-    // Filter out any private data
-    std::string substr(reinterpret_cast<const char*>(line), end_of_line - line);
-    std::string::size_type pos_private = substr.find("Email");
-    if (pos_private == std::string::npos) {
-      pos_private = substr.find("Passwd");
-    }
-    if (pos_private == std::string::npos) {
-      LOG_V(level) << label << direction << substr;
-    } else {
-      LOG_V(level) << label << direction << "## omitted for privacy ##";
-    }
-  }
-
-  if (state) {
-    state->unprintable_count_[input] = consecutive_unprintable;
-  }
-}
-
-void InitDiagnosticLoggingDelegateFunction(
-    void (*delegate)(const std::string&)) {
-#ifndef NDEBUG
-  // Ensure that this function is always called from the same thread.
-  base::subtle::NoBarrier_CompareAndSwap(&g_init_logging_delegate_thread_id, 0,
-      static_cast<base::subtle::Atomic32>(base::PlatformThread::CurrentId()));
-  DCHECK_EQ(
-      g_init_logging_delegate_thread_id,
-      static_cast<base::subtle::Atomic32>(base::PlatformThread::CurrentId()));
-#endif
-  CHECK(delegate);
-  // This function may be called with the same argument several times if the
-  // page is reloaded or there are several PeerConnections on one page with
-  // logging enabled. This is OK, we simply don't have to do anything.
-  if (delegate == g_logging_delegate_function)
-    return;
-  CHECK(!g_logging_delegate_function);
-#ifdef NDEBUG
-  IPAddress::set_strip_sensitive(true);
-#endif
-  g_logging_delegate_function = delegate;
-
-  if (g_extra_logging_init_function)
-    g_extra_logging_init_function(delegate);
-}
-
-void SetExtraLoggingInit(
-    void (*function)(void (*delegate)(const std::string&))) {
-  CHECK(function);
-  CHECK(!g_extra_logging_init_function);
-  g_extra_logging_init_function = function;
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/logging.h b/media/webrtc/trunk/webrtc/overrides/webrtc/base/logging.h
deleted file mode 100644
index 78f48f4359..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/logging.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file overrides the logging macros in libjingle (webrtc/base/logging.h).
-// Instead of using libjingle's logging implementation, the libjingle macros are
-// mapped to the corresponding base/logging.h macro (chromium's VLOG).
-// If this file is included outside of libjingle (e.g. in wrapper code) it
-// should be included after base/logging.h (if any) or compiler error or
-// unexpected behavior may occur (macros that have the same name in libjingle as
-// in chromium will use the libjingle definition if this file is included
-// first).
-
-// Setting the LoggingSeverity (and lower) that should be written to file should
-// be done via command line by specifying the flags:
-// --vmodule or --v please see base/logging.h for details on how to use them.
-// Specifying what file to write to is done using InitLogging also in
-// base/logging.h.
-
-// The macros and classes declared in here are not described as they are
-// NOT TO BE USED outside of libjingle.
-
-#ifndef THIRD_PARTY_LIBJINGLE_OVERRIDES_WEBRTC_BASE_LOGGING_H_
-#define THIRD_PARTY_LIBJINGLE_OVERRIDES_WEBRTC_BASE_LOGGING_H_
-
-#include "third_party/webrtc/overrides/webrtc/base/diagnostic_logging.h"
-
-//////////////////////////////////////////////////////////////////////
-// Libjingle macros which are mapped over to their VLOG equivalent in
-// base/logging.h
-//////////////////////////////////////////////////////////////////////
-
-#if defined(LOGGING_INSIDE_WEBRTC)
-
-#define DIAGNOSTIC_LOG(sev, ctx, err, ...) \
-  rtc::DiagnosticLogMessage( \
-      __FILE__, __LINE__, sev, VLOG_IS_ON(sev), \
-      rtc::ERRCTX_ ## ctx, err, ##__VA_ARGS__).stream()
-
-#define LOG_CHECK_LEVEL(sev) VLOG_IS_ON(rtc::sev)
-#define LOG_CHECK_LEVEL_V(sev) VLOG_IS_ON(sev)
-
-#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
-#undef LOG
-#define LOG(sev) DIAGNOSTIC_LOG(rtc::sev, NONE, 0)
-
-// The _F version prefixes the message with the current function name.
-#if defined(__GNUC__) && defined(_DEBUG)
-#define LOG_F(sev) LOG(sev) << __PRETTY_FUNCTION__ << ": "
-#else
-#define LOG_F(sev) LOG(sev) << __FUNCTION__ << ": "
-#endif
-
-#define LOG_E(sev, ctx, err, ...) \
-  DIAGNOSTIC_LOG(rtc::sev, ctx, err, ##__VA_ARGS__)
-
-#undef LOG_ERRNO_EX
-#define LOG_ERRNO_EX(sev, err) LOG_E(sev, ERRNO, err)
-#undef LOG_ERRNO
-#define LOG_ERRNO(sev) LOG_ERRNO_EX(sev, errno)
-
-#if defined(WEBRTC_WIN)
-#define LOG_GLE_EX(sev, err) LOG_E(sev, HRESULT, err)
-#define LOG_GLE(sev) LOG_GLE_EX(sev, GetLastError())
-#define LOG_GLEM(sev, mod) LOG_E(sev, HRESULT, GetLastError(), mod)
-#define LOG_ERR_EX(sev, err) LOG_GLE_EX(sev, err)
-#define LOG_ERR(sev) LOG_GLE(sev)
-#define LAST_SYSTEM_ERROR (::GetLastError())
-#else
-#define LOG_ERR_EX(sev, err) LOG_ERRNO_EX(sev, err)
-#define LOG_ERR(sev) LOG_ERRNO(sev)
-#define LAST_SYSTEM_ERROR (errno)
-#endif  // OS_WIN
-
-#undef PLOG
-#define PLOG(sev, err) LOG_ERR_EX(sev, err)
-
-#endif  // LOGGING_INSIDE_WEBRTC
-
-#endif  // THIRD_PARTY_LIBJINGLE_OVERRIDES_WEBRTC_BASE_LOGGING_H_
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/win32socketinit.cc b/media/webrtc/trunk/webrtc/overrides/webrtc/base/win32socketinit.cc
deleted file mode 100644
index 929ce8d363..0000000000
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/win32socketinit.cc
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- *  Copyright 2006 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Redirect Libjingle's winsock initialization activity into Chromium's
-// singleton object that managest precisely that for the browser.
-
-#include "webrtc/base/win32socketinit.h"
-
-#include "net/base/winsock_init.h"
-
-#if !defined(WEBRTC_WIN)
-#error "Only compile this on Windows"
-#endif
-
-namespace rtc {
-
-void EnsureWinsockInit() {
-  net::EnsureWinsockInit();
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/p2p/OWNERS b/media/webrtc/trunk/webrtc/p2p/OWNERS
index 9a527df143..0f00d1aa48 100644
--- a/media/webrtc/trunk/webrtc/p2p/OWNERS
+++ b/media/webrtc/trunk/webrtc/p2p/OWNERS
@@ -9,4 +9,9 @@ pthatcher@webrtc.org
 sergeyu@chromium.org
 tommi@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.cc b/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.cc
index 2b1b693588..444f06146a 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.cc
@@ -20,13 +20,13 @@ namespace cricket {
 
 static const size_t kMaxPacketSize = 64 * 1024;
 
-typedef uint16 PacketLength;
+typedef uint16_t PacketLength;
 static const size_t kPacketLenSize = sizeof(PacketLength);
 static const size_t kPacketLenOffset = 2;
 static const size_t kBufSize = kMaxPacketSize + kStunHeaderSize;
 static const size_t kTurnChannelDataHdrSize = 4;
 
-inline bool IsStunMessage(uint16 msg_type) {
+inline bool IsStunMessage(uint16_t msg_type) {
   // The first two bits of a channel data message are 0b01.
   return (msg_type & 0xC000) ? false : true;
 }
@@ -129,7 +129,7 @@ size_t AsyncStunTCPSocket::GetExpectedLength(const void* data, size_t len,
   PacketLength pkt_len =
       rtc::GetBE16(static_cast<const char*>(data) + kPacketLenOffset);
   size_t expected_pkt_len;
-  uint16 msg_type = rtc::GetBE16(data);
+  uint16_t msg_type = rtc::GetBE16(data);
   if (IsStunMessage(msg_type)) {
     // STUN message.
     expected_pkt_len = kStunHeaderSize + pkt_len;
diff --git a/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.h b/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.h
index 4f53b0311d..3a15d4a399 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/asyncstuntcpsocket.h
@@ -42,7 +42,7 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
   size_t GetExpectedLength(const void* data, size_t len,
                            int* pad_bytes);
 
-  DISALLOW_EVIL_CONSTRUCTORS(AsyncStunTCPSocket);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncStunTCPSocket);
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.cc b/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.cc
index 9b12e78d87..697518da9d 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.cc
@@ -44,7 +44,9 @@ BasicPacketSocketFactory::~BasicPacketSocketFactory() {
 }
 
 AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
-    const SocketAddress& address, uint16 min_port, uint16 max_port) {
+    const SocketAddress& address,
+    uint16_t min_port,
+    uint16_t max_port) {
   // UDP sockets are simple.
   rtc::AsyncSocket* socket =
       socket_factory()->CreateAsyncSocket(
@@ -62,9 +64,10 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
 }
 
 AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
-    const SocketAddress& local_address, uint16 min_port, uint16 max_port,
+    const SocketAddress& local_address,
+    uint16_t min_port,
+    uint16_t max_port,
     int opts) {
-
   // Fail if TLS is required.
   if (opts & PacketSocketFactory::OPT_TLS) {
     LOG(LS_ERROR) << "TLS support currently is not available.";
@@ -176,9 +179,10 @@ AsyncResolverInterface* BasicPacketSocketFactory::CreateAsyncResolver() {
   return new rtc::AsyncResolver();
 }
 
-int BasicPacketSocketFactory::BindSocket(
-    AsyncSocket* socket, const SocketAddress& local_address,
-    uint16 min_port, uint16 max_port) {
+int BasicPacketSocketFactory::BindSocket(AsyncSocket* socket,
+                                         const SocketAddress& local_address,
+                                         uint16_t min_port,
+                                         uint16_t max_port) {
   int ret = -1;
   if (min_port == 0 && max_port == 0) {
     // If there's no port range, let the OS pick a port for us.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.h b/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.h
index b23a67729e..5046e0f518 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/basicpacketsocketfactory.h
@@ -27,11 +27,11 @@ class BasicPacketSocketFactory : public PacketSocketFactory {
   ~BasicPacketSocketFactory() override;
 
   AsyncPacketSocket* CreateUdpSocket(const SocketAddress& local_address,
-                                     uint16 min_port,
-                                     uint16 max_port) override;
+                                     uint16_t min_port,
+                                     uint16_t max_port) override;
   AsyncPacketSocket* CreateServerTcpSocket(const SocketAddress& local_address,
-                                           uint16 min_port,
-                                           uint16 max_port,
+                                           uint16_t min_port,
+                                           uint16_t max_port,
                                            int opts) override;
   AsyncPacketSocket* CreateClientTcpSocket(const SocketAddress& local_address,
                                            const SocketAddress& remote_address,
@@ -44,8 +44,8 @@ class BasicPacketSocketFactory : public PacketSocketFactory {
  private:
   int BindSocket(AsyncSocket* socket,
                  const SocketAddress& local_address,
-                 uint16 min_port,
-                 uint16 max_port);
+                 uint16_t min_port,
+                 uint16_t max_port);
 
   SocketFactory* socket_factory();
 
diff --git a/media/webrtc/trunk/webrtc/p2p/base/candidate.h b/media/webrtc/trunk/webrtc/p2p/base/candidate.h
index c2b889541e..ac7acabf05 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/candidate.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/candidate.h
@@ -43,11 +43,11 @@ class Candidate {
   Candidate(int component,
             const std::string& protocol,
             const rtc::SocketAddress& address,
-            uint32 priority,
+            uint32_t priority,
             const std::string& username,
             const std::string& password,
             const std::string& type,
-            uint32 generation,
+            uint32_t generation,
             const std::string& foundation)
       : id_(rtc::CreateRandomString(8)),
         component_(component),
@@ -70,18 +70,22 @@ class Candidate {
   const std::string & protocol() const { return protocol_; }
   void set_protocol(const std::string & protocol) { protocol_ = protocol; }
 
+  // The protocol used to talk to relay.
+  const std::string& relay_protocol() const { return relay_protocol_; }
+  void set_relay_protocol(const std::string& protocol) {
+    relay_protocol_ = protocol;
+  }
+
   const rtc::SocketAddress & address() const { return address_; }
   void set_address(const rtc::SocketAddress & address) {
     address_ = address;
   }
 
-  uint32 priority() const { return priority_; }
-  void set_priority(const uint32 priority) { priority_ = priority; }
-
-//  void set_type_preference(uint32 type_preference) {
-//    priority_ = GetPriority(type_preference);
-//  }
+  uint32_t priority() const { return priority_; }
+  void set_priority(const uint32_t priority) { priority_ = priority; }
 
+  // TODO(pthatcher): Remove once Chromium's jingle/glue/utils.cc
+  // doesn't use it.
   // Maps old preference (which was 0.0-1.0) to match priority (which
   // is 0-2^32-1) to to match RFC 5245, section 4.1.2.1.  Also see
   // https://docs.google.com/a/google.com/document/d/
@@ -91,14 +95,17 @@ class Candidate {
     return static_cast<float>(((priority_ >> 24) * 100 / 127) / 100.0);
   }
 
+  // TODO(pthatcher): Remove once Chromium's jingle/glue/utils.cc
+  // doesn't use it.
   void set_preference(float preference) {
-    // Limiting priority to UINT_MAX when value exceeds uint32 max.
+    // Limiting priority to UINT_MAX when value exceeds uint32_t max.
     // This can happen for e.g. when preference = 3.
-    uint64 prio_val = static_cast<uint64>(preference * 127) << 24;
-    priority_ =
-        static_cast<uint32>(std::min(prio_val, static_cast<uint64>(UINT_MAX)));
+    uint64_t prio_val = static_cast<uint64_t>(preference * 127) << 24;
+    priority_ = static_cast<uint32_t>(
+        std::min(prio_val, static_cast<uint64_t>(UINT_MAX)));
   }
 
+  // TODO(honghaiz): Change to usernameFragment or ufrag.
   const std::string & username() const { return username_; }
   void set_username(const std::string & username) { username_ = username; }
 
@@ -119,8 +126,8 @@ class Candidate {
   }
 
   // Candidates in a new generation replace those in the old generation.
-  uint32 generation() const { return generation_; }
-  void set_generation(uint32 generation) { generation_ = generation; }
+  uint32_t generation() const { return generation_; }
+  void set_generation(uint32_t generation) { generation_ = generation; }
   const std::string generation_str() const {
     std::ostringstream ost;
     ost << generation_;
@@ -171,9 +178,9 @@ class Candidate {
     return ToStringInternal(true);
   }
 
-  uint32 GetPriority(uint32 type_preference,
-                     int network_adapter_preference,
-                     int relay_preference) const {
+  uint32_t GetPriority(uint32_t type_preference,
+                       int network_adapter_preference,
+                       int relay_preference) const {
     // RFC 5245 - 4.1.2.1.
     // priority = (2^24)*(type preference) +
     //            (2^8)*(local preference) +
@@ -214,14 +221,15 @@ class Candidate {
   std::string id_;
   int component_;
   std::string protocol_;
+  std::string relay_protocol_;
   rtc::SocketAddress address_;
-  uint32 priority_;
+  uint32_t priority_;
   std::string username_;
   std::string password_;
   std::string type_;
   std::string network_name_;
   rtc::AdapterType network_type_;
-  uint32 generation_;
+  uint32_t generation_;
   std::string foundation_;
   rtc::SocketAddress related_address_;
   std::string tcptype_;
diff --git a/media/webrtc/trunk/webrtc/p2p/base/constants.cc b/media/webrtc/trunk/webrtc/p2p/base/constants.cc
index 614cbc845a..2a258718f4 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/constants.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/constants.cc
@@ -21,12 +21,6 @@ const char CN_OTHER[] = "main";
 
 const char GROUP_TYPE_BUNDLE[] = "BUNDLE";
 
-const char NS_JINGLE_ICE_UDP[] = "urn:xmpp:jingle:transports:ice-udp:1";
-const char NS_GINGLE_P2P[] = "http://www.google.com/transport/p2p";
-const char NS_GINGLE_RAW[] = "http://www.google.com/transport/raw-udp";
-
-const char ICE_OPTION_GICE[] = "google-ice";
-
 // Minimum ufrag length is 4 characters as per RFC5245. We chose 16 because
 // some internal systems expect username to be 16 bytes.
 const int ICE_UFRAG_LENGTH = 16;
@@ -37,7 +31,6 @@ const size_t ICE_UFRAG_MIN_LENGTH = 4;
 const size_t ICE_PWD_MIN_LENGTH = 22;
 const size_t ICE_UFRAG_MAX_LENGTH = 255;
 const size_t ICE_PWD_MAX_LENGTH = 256;
-const size_t GICE_UFRAG_MAX_LENGTH = 16;
 
 // TODO: This is media-specific, so might belong
 // somewhere like media/base/constants.h
@@ -48,14 +41,6 @@ const int ICE_CANDIDATE_COMPONENT_DEFAULT = 1;
 const char NS_JINGLE_RTP[] = "urn:xmpp:jingle:apps:rtp:1";
 const char NS_JINGLE_DRAFT_SCTP[] = "google:jingle:sctp";
 
-const char GICE_CHANNEL_NAME_RTP[] = "rtp";
-const char GICE_CHANNEL_NAME_RTCP[] = "rtcp";
-const char GICE_CHANNEL_NAME_VIDEO_RTP[] = "video_rtp";
-const char GICE_CHANNEL_NAME_VIDEO_RTCP[] = "video_rtcp";
-const char GICE_CHANNEL_NAME_DATA_RTP[] = "data_rtp";
-const char GICE_CHANNEL_NAME_DATA_RTCP[] = "data_rtcp";
-
-
 // From RFC 4145, SDP setup attribute values.
 const char CONNECTIONROLE_ACTIVE_STR[] = "active";
 const char CONNECTIONROLE_PASSIVE_STR[] = "passive";
diff --git a/media/webrtc/trunk/webrtc/p2p/base/constants.h b/media/webrtc/trunk/webrtc/p2p/base/constants.h
index 90a7816239..c3e1b781dc 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/constants.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/constants.h
@@ -28,19 +28,12 @@ extern const char CN_OTHER[];
 // GN stands for group name
 extern const char GROUP_TYPE_BUNDLE[];
 
-extern const char NS_JINGLE_ICE_UDP[];
-extern const char NS_GINGLE_P2P[];
-extern const char NS_GINGLE_RAW[];
-
-extern const char ICE_OPTION_GICE[];
-
 extern const int ICE_UFRAG_LENGTH;
 extern const int ICE_PWD_LENGTH;
 extern const size_t ICE_UFRAG_MIN_LENGTH;
 extern const size_t ICE_PWD_MIN_LENGTH;
 extern const size_t ICE_UFRAG_MAX_LENGTH;
 extern const size_t ICE_PWD_MAX_LENGTH;
-extern const size_t GICE_UFRAG_MAX_LENGTH;
 
 extern const int ICE_CANDIDATE_COMPONENT_RTP;
 extern const int ICE_CANDIDATE_COMPONENT_RTCP;
@@ -49,13 +42,6 @@ extern const int ICE_CANDIDATE_COMPONENT_DEFAULT;
 extern const char NS_JINGLE_RTP[];
 extern const char NS_JINGLE_DRAFT_SCTP[];
 
-extern const char GICE_CHANNEL_NAME_RTP[];
-extern const char GICE_CHANNEL_NAME_RTCP[];
-extern const char GICE_CHANNEL_NAME_VIDEO_RTP[];
-extern const char GICE_CHANNEL_NAME_VIDEO_RTCP[];
-extern const char GICE_CHANNEL_NAME_DATA_RTP[];
-extern const char GICE_CHANNEL_NAME_DATA_RTCP[];
-
 // RFC 4145, SDP setup attribute values.
 extern const char CONNECTIONROLE_ACTIVE_STR[];
 extern const char CONNECTIONROLE_PASSIVE_STR[];
diff --git a/media/webrtc/trunk/webrtc/p2p/base/dtlstransport.h b/media/webrtc/trunk/webrtc/p2p/base/dtlstransport.h
index 8e17ea6609..9f2903e1d7 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/dtlstransport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/dtlstransport.h
@@ -22,45 +22,54 @@ namespace cricket {
 
 class PortAllocator;
 
-// Base should be a descendant of cricket::Transport
+// Base should be a descendant of cricket::Transport and have a constructor
+// that takes a transport name and PortAllocator.
+//
+// Everything in this class should be called on the worker thread.
 template<class Base>
 class DtlsTransport : public Base {
  public:
-  DtlsTransport(rtc::Thread* signaling_thread,
-                rtc::Thread* worker_thread,
-                const std::string& content_name,
+  DtlsTransport(const std::string& name,
                 PortAllocator* allocator,
-                rtc::SSLIdentity* identity)
-      : Base(signaling_thread, worker_thread, content_name, allocator),
-        identity_(identity),
-        secure_role_(rtc::SSL_CLIENT) {
-  }
+                const rtc::scoped_refptr<rtc::RTCCertificate>& certificate)
+      : Base(name, allocator),
+        certificate_(certificate),
+        secure_role_(rtc::SSL_CLIENT),
+        ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12) {}
 
   ~DtlsTransport() {
     Base::DestroyAllChannels();
   }
-  virtual void SetIdentity_w(rtc::SSLIdentity* identity) {
-    identity_ = identity;
+
+  void SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override {
+    certificate_ = certificate;
   }
-  virtual bool GetIdentity_w(rtc::SSLIdentity** identity) {
-    if (!identity_)
+  bool GetLocalCertificate(
+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override {
+    if (!certificate_)
       return false;
 
-    *identity = identity_->GetReference();
+    *certificate = certificate_;
     return true;
   }
 
-  virtual bool ApplyLocalTransportDescription_w(TransportChannelImpl* channel,
-                                                std::string* error_desc) {
+  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override {
+    ssl_max_version_ = version;
+    return true;
+  }
+
+  bool ApplyLocalTransportDescription(TransportChannelImpl* channel,
+                                      std::string* error_desc) override {
     rtc::SSLFingerprint* local_fp =
         Base::local_description()->identity_fingerprint.get();
 
     if (local_fp) {
       // Sanity check local fingerprint.
-      if (identity_) {
+      if (certificate_) {
         rtc::scoped_ptr<rtc::SSLFingerprint> local_fp_tmp(
             rtc::SSLFingerprint::Create(local_fp->algorithm,
-                                              identity_));
+                                        certificate_->identity()));
         ASSERT(local_fp_tmp.get() != NULL);
         if (!(*local_fp_tmp == *local_fp)) {
           std::ostringstream desc;
@@ -75,20 +84,20 @@ class DtlsTransport : public Base {
             error_desc);
       }
     } else {
-      identity_ = NULL;
+      certificate_ = nullptr;
     }
 
-    if (!channel->SetLocalIdentity(identity_)) {
+    if (!channel->SetLocalCertificate(certificate_)) {
       return BadTransportDescription("Failed to set local identity.",
                                      error_desc);
     }
 
     // Apply the description in the base class.
-    return Base::ApplyLocalTransportDescription_w(channel, error_desc);
+    return Base::ApplyLocalTransportDescription(channel, error_desc);
   }
 
-  virtual bool NegotiateTransportDescription_w(ContentAction local_role,
-                                               std::string* error_desc) {
+  bool NegotiateTransportDescription(ContentAction local_role,
+                                     std::string* error_desc) override {
     if (!Base::local_description() || !Base::remote_description()) {
       const std::string msg = "Local and Remote description must be set before "
                               "transport descriptions are negotiated";
@@ -185,15 +194,17 @@ class DtlsTransport : public Base {
     }
 
     // Now run the negotiation for the base class.
-    return Base::NegotiateTransportDescription_w(local_role, error_desc);
+    return Base::NegotiateTransportDescription(local_role, error_desc);
   }
 
-  virtual DtlsTransportChannelWrapper* CreateTransportChannel(int component) {
-    return new DtlsTransportChannelWrapper(
+  DtlsTransportChannelWrapper* CreateTransportChannel(int component) override {
+    DtlsTransportChannelWrapper* channel = new DtlsTransportChannelWrapper(
         this, Base::CreateTransportChannel(component));
+    channel->SetSslMaxProtocolVersion(ssl_max_version_);
+    return channel;
   }
 
-  virtual void DestroyTransportChannel(TransportChannelImpl* channel) {
+  void DestroyTransportChannel(TransportChannelImpl* channel) override {
     // Kind of ugly, but this lets us do the exact inverse of the create.
     DtlsTransportChannelWrapper* dtls_channel =
         static_cast<DtlsTransportChannelWrapper*>(channel);
@@ -202,16 +213,15 @@ class DtlsTransport : public Base {
     Base::DestroyTransportChannel(base_channel);
   }
 
-  virtual bool GetSslRole_w(rtc::SSLRole* ssl_role) const {
+  bool GetSslRole(rtc::SSLRole* ssl_role) const override {
     ASSERT(ssl_role != NULL);
     *ssl_role = secure_role_;
     return true;
   }
 
  private:
-  virtual bool ApplyNegotiatedTransportDescription_w(
-      TransportChannelImpl* channel,
-      std::string* error_desc) {
+  bool ApplyNegotiatedTransportDescription(TransportChannelImpl* channel,
+                                           std::string* error_desc) override {
     // Set ssl role. Role must be set before fingerprint is applied, which
     // initiates DTLS setup.
     if (!channel->SetSslRole(secure_role_)) {
@@ -219,18 +229,19 @@ class DtlsTransport : public Base {
                                      error_desc);
     }
     // Apply remote fingerprint.
-    if (!channel->SetRemoteFingerprint(
-            remote_fingerprint_->algorithm,
-            reinterpret_cast<const uint8*>(remote_fingerprint_->digest.data()),
-            remote_fingerprint_->digest.size())) {
+    if (!channel->SetRemoteFingerprint(remote_fingerprint_->algorithm,
+                                       reinterpret_cast<const uint8_t*>(
+                                           remote_fingerprint_->digest.data()),
+                                       remote_fingerprint_->digest.size())) {
       return BadTransportDescription("Failed to apply remote fingerprint.",
                                      error_desc);
     }
-    return Base::ApplyNegotiatedTransportDescription_w(channel, error_desc);
+    return Base::ApplyNegotiatedTransportDescription(channel, error_desc);
   }
 
-  rtc::SSLIdentity* identity_;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
   rtc::SSLRole secure_role_;
+  rtc::SSLProtocolVersion ssl_max_version_;
   rtc::scoped_ptr<rtc::SSLFingerprint> remote_fingerprint_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.cc b/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.cc
index ca561a0898..d6b5bce723 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.cc
@@ -8,10 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <utility>
+
 #include "webrtc/p2p/base/dtlstransportchannel.h"
 
 #include "webrtc/p2p/base/common.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/sslstreamadapter.h"
@@ -25,15 +28,25 @@ static const size_t kDtlsRecordHeaderLen = 13;
 static const size_t kMaxDtlsPacketLen = 2048;
 static const size_t kMinRtpPacketLen = 12;
 
+// Maximum number of pending packets in the queue. Packets are read immediately
+// after they have been written, so a capacity of "1" is sufficient.
+static const size_t kMaxPendingPackets = 1;
+
 static bool IsDtlsPacket(const char* data, size_t len) {
-  const uint8* u = reinterpret_cast<const uint8*>(data);
+  const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
   return (len >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64));
 }
 static bool IsRtpPacket(const char* data, size_t len) {
-  const uint8* u = reinterpret_cast<const uint8*>(data);
+  const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
   return (len >= kMinRtpPacketLen && (u[0] & 0xC0) == 0x80);
 }
 
+StreamInterfaceChannel::StreamInterfaceChannel(TransportChannel* channel)
+    : channel_(channel),
+      state_(rtc::SS_OPEN),
+      packets_(kMaxPendingPackets, kMaxDtlsPacketLen) {
+}
+
 rtc::StreamResult StreamInterfaceChannel::Read(void* buffer,
                                                      size_t buffer_len,
                                                      size_t* read,
@@ -43,7 +56,11 @@ rtc::StreamResult StreamInterfaceChannel::Read(void* buffer,
   if (state_ == rtc::SS_OPENING)
     return rtc::SR_BLOCK;
 
-  return fifo_.Read(buffer, buffer_len, read, error);
+  if (!packets_.ReadFront(buffer, buffer_len, read)) {
+    return rtc::SR_BLOCK;
+  }
+
+  return rtc::SR_SUCCESS;
 }
 
 rtc::StreamResult StreamInterfaceChannel::Write(const void* data,
@@ -62,52 +79,45 @@ rtc::StreamResult StreamInterfaceChannel::Write(const void* data,
 }
 
 bool StreamInterfaceChannel::OnPacketReceived(const char* data, size_t size) {
-  // We force a read event here to ensure that we don't overflow our FIFO.
-  // Under high packet rate this can occur if we wait for the FIFO to post its
-  // own SE_READ.
-  bool ret = (fifo_.WriteAll(data, size, NULL, NULL) == rtc::SR_SUCCESS);
+  // We force a read event here to ensure that we don't overflow our queue.
+  bool ret = packets_.WriteBack(data, size, NULL);
+  RTC_CHECK(ret) << "Failed to write packet to queue.";
   if (ret) {
     SignalEvent(this, rtc::SE_READ, 0);
   }
   return ret;
 }
 
-void StreamInterfaceChannel::OnEvent(rtc::StreamInterface* stream,
-                                     int sig, int err) {
-  SignalEvent(this, sig, err);
-}
-
 DtlsTransportChannelWrapper::DtlsTransportChannelWrapper(
-                                           Transport* transport,
-                                           TransportChannelImpl* channel)
-    : TransportChannelImpl(channel->content_name(), channel->component()),
+    Transport* transport,
+    TransportChannelImpl* channel)
+    : TransportChannelImpl(channel->transport_name(), channel->component()),
       transport_(transport),
       worker_thread_(rtc::Thread::Current()),
       channel_(channel),
       downward_(NULL),
-      dtls_state_(STATE_NONE),
-      local_identity_(NULL),
-      ssl_role_(rtc::SSL_CLIENT) {
-  channel_->SignalReadableState.connect(this,
-      &DtlsTransportChannelWrapper::OnReadableState);
+      ssl_role_(rtc::SSL_CLIENT),
+      ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12) {
   channel_->SignalWritableState.connect(this,
       &DtlsTransportChannelWrapper::OnWritableState);
   channel_->SignalReadPacket.connect(this,
       &DtlsTransportChannelWrapper::OnReadPacket);
+  channel_->SignalSentPacket.connect(
+      this, &DtlsTransportChannelWrapper::OnSentPacket);
   channel_->SignalReadyToSend.connect(this,
       &DtlsTransportChannelWrapper::OnReadyToSend);
-  channel_->SignalRequestSignaling.connect(this,
-      &DtlsTransportChannelWrapper::OnRequestSignaling);
-  channel_->SignalCandidateReady.connect(this,
-      &DtlsTransportChannelWrapper::OnCandidateReady);
-  channel_->SignalCandidatesAllocationDone.connect(this,
-      &DtlsTransportChannelWrapper::OnCandidatesAllocationDone);
+  channel_->SignalGatheringState.connect(
+      this, &DtlsTransportChannelWrapper::OnGatheringState);
+  channel_->SignalCandidateGathered.connect(
+      this, &DtlsTransportChannelWrapper::OnCandidateGathered);
   channel_->SignalRoleConflict.connect(this,
       &DtlsTransportChannelWrapper::OnRoleConflict);
   channel_->SignalRouteChange.connect(this,
       &DtlsTransportChannelWrapper::OnRouteChange);
   channel_->SignalConnectionRemoved.connect(this,
       &DtlsTransportChannelWrapper::OnConnectionRemoved);
+  channel_->SignalReceivingState.connect(this,
+      &DtlsTransportChannelWrapper::OnReceivingState);
 }
 
 DtlsTransportChannelWrapper::~DtlsTransportChannelWrapper() {
@@ -115,31 +125,14 @@ DtlsTransportChannelWrapper::~DtlsTransportChannelWrapper() {
 
 void DtlsTransportChannelWrapper::Connect() {
   // We should only get a single call to Connect.
-  ASSERT(dtls_state_ == STATE_NONE ||
-         dtls_state_ == STATE_OFFERED ||
-         dtls_state_ == STATE_ACCEPTED);
+  ASSERT(dtls_state() == DTLS_TRANSPORT_NEW);
   channel_->Connect();
 }
 
-void DtlsTransportChannelWrapper::Reset() {
-  channel_->Reset();
-  set_writable(false);
-  set_readable(false);
-
-  // Re-call SetupDtls()
-  if (!SetupDtls()) {
-    LOG_J(LS_ERROR, this) << "Error re-initializing DTLS";
-    dtls_state_ = STATE_CLOSED;
-    return;
-  }
-
-  dtls_state_ = STATE_ACCEPTED;
-}
-
-bool DtlsTransportChannelWrapper::SetLocalIdentity(
-    rtc::SSLIdentity* identity) {
-  if (dtls_state_ != STATE_NONE) {
-    if (identity == local_identity_) {
+bool DtlsTransportChannelWrapper::SetLocalCertificate(
+    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+  if (dtls_active_) {
+    if (certificate == local_certificate_) {
       // This may happen during renegotiation.
       LOG_J(LS_INFO, this) << "Ignoring identical DTLS identity";
       return true;
@@ -149,9 +142,9 @@ bool DtlsTransportChannelWrapper::SetLocalIdentity(
     }
   }
 
-  if (identity) {
-    local_identity_ = identity;
-    dtls_state_ = STATE_OFFERED;
+  if (certificate) {
+    local_certificate_ = certificate;
+    dtls_active_ = true;
   } else {
     LOG_J(LS_INFO, this) << "NULL DTLS identity supplied. Not doing DTLS";
   }
@@ -159,17 +152,25 @@ bool DtlsTransportChannelWrapper::SetLocalIdentity(
   return true;
 }
 
-bool DtlsTransportChannelWrapper::GetLocalIdentity(
-    rtc::SSLIdentity** identity) const {
-  if (!local_identity_)
-    return false;
+rtc::scoped_refptr<rtc::RTCCertificate>
+DtlsTransportChannelWrapper::GetLocalCertificate() const {
+  return local_certificate_;
+}
 
-  *identity = local_identity_->GetReference();
+bool DtlsTransportChannelWrapper::SetSslMaxProtocolVersion(
+    rtc::SSLProtocolVersion version) {
+  if (dtls_active_) {
+    LOG(LS_ERROR) << "Not changing max. protocol version "
+                  << "while DTLS is negotiating";
+    return false;
+  }
+
+  ssl_max_version_ = version;
   return true;
 }
 
 bool DtlsTransportChannelWrapper::SetSslRole(rtc::SSLRole role) {
-  if (dtls_state_ == STATE_OPEN) {
+  if (dtls_state() == DTLS_TRANSPORT_CONNECTED) {
     if (ssl_role_ != role) {
       LOG(LS_ERROR) << "SSL Role can't be reversed after the session is setup.";
       return false;
@@ -186,67 +187,73 @@ bool DtlsTransportChannelWrapper::GetSslRole(rtc::SSLRole* role) const {
   return true;
 }
 
-bool DtlsTransportChannelWrapper::GetSslCipher(std::string* cipher) {
-  if (dtls_state_ != STATE_OPEN) {
+bool DtlsTransportChannelWrapper::GetSslCipherSuite(int* cipher) {
+  if (dtls_state() != DTLS_TRANSPORT_CONNECTED) {
     return false;
   }
 
-  return dtls_->GetSslCipher(cipher);
+  return dtls_->GetSslCipherSuite(cipher);
 }
 
 bool DtlsTransportChannelWrapper::SetRemoteFingerprint(
     const std::string& digest_alg,
-    const uint8* digest,
+    const uint8_t* digest,
     size_t digest_len) {
-
   rtc::Buffer remote_fingerprint_value(digest, digest_len);
 
-  if (dtls_state_ != STATE_NONE &&
-      remote_fingerprint_value_ == remote_fingerprint_value &&
+  // Once we have the local certificate, the same remote fingerprint can be set
+  // multiple times.
+  if (dtls_active_ && remote_fingerprint_value_ == remote_fingerprint_value &&
       !digest_alg.empty()) {
     // This may happen during renegotiation.
     LOG_J(LS_INFO, this) << "Ignoring identical remote DTLS fingerprint";
     return true;
   }
 
-  // Allow SetRemoteFingerprint with a NULL digest even if SetLocalIdentity
-  // hasn't been called.
-  if (dtls_state_ > STATE_OFFERED ||
-      (dtls_state_ == STATE_NONE && !digest_alg.empty())) {
+  // If the other side doesn't support DTLS, turn off |dtls_active_|.
+  if (digest_alg.empty()) {
+    RTC_DCHECK(!digest_len);
+    LOG_J(LS_INFO, this) << "Other side didn't support DTLS.";
+    dtls_active_ = false;
+    return true;
+  }
+
+  // Otherwise, we must have a local certificate before setting remote
+  // fingerprint.
+  if (!dtls_active_) {
     LOG_J(LS_ERROR, this) << "Can't set DTLS remote settings in this state.";
     return false;
   }
 
-  if (digest_alg.empty()) {
-    LOG_J(LS_INFO, this) << "Other side didn't support DTLS.";
-    dtls_state_ = STATE_NONE;
-    return true;
-  }
-
   // At this point we know we are doing DTLS
-  remote_fingerprint_value.TransferTo(&remote_fingerprint_value_);
+  remote_fingerprint_value_ = std::move(remote_fingerprint_value);
   remote_fingerprint_algorithm_ = digest_alg;
 
+  bool reconnect = dtls_;
+
   if (!SetupDtls()) {
-    dtls_state_ = STATE_CLOSED;
+    set_dtls_state(DTLS_TRANSPORT_FAILED);
     return false;
   }
 
-  dtls_state_ = STATE_ACCEPTED;
+  if (reconnect) {
+    Reconnect();
+  }
+
   return true;
 }
 
-bool DtlsTransportChannelWrapper::GetRemoteCertificate(
+bool DtlsTransportChannelWrapper::GetRemoteSSLCertificate(
     rtc::SSLCertificate** cert) const {
-  if (!dtls_)
+  if (!dtls_) {
     return false;
+  }
 
   return dtls_->GetPeerCertificate(cert);
 }
 
 bool DtlsTransportChannelWrapper::SetupDtls() {
-  StreamInterfaceChannel* downward =
-      new StreamInterfaceChannel(worker_thread_, channel_);
+  StreamInterfaceChannel* downward = new StreamInterfaceChannel(channel_);
 
   dtls_.reset(rtc::SSLStreamAdapter::Create(downward));
   if (!dtls_) {
@@ -257,8 +264,9 @@ bool DtlsTransportChannelWrapper::SetupDtls() {
 
   downward_ = downward;
 
-  dtls_->SetIdentity(local_identity_->GetReference());
+  dtls_->SetIdentity(local_certificate_->identity()->GetReference());
   dtls_->SetMode(rtc::SSL_MODE_DTLS);
+  dtls_->SetMaxProtocolVersion(ssl_max_version_);
   dtls_->SetServerRole(ssl_role_);
   dtls_->SignalEvent.connect(this, &DtlsTransportChannelWrapper::OnDtlsEvent);
   if (!dtls_->SetPeerCertificateDigest(
@@ -271,44 +279,44 @@ bool DtlsTransportChannelWrapper::SetupDtls() {
 
   // Set up DTLS-SRTP, if it's been enabled.
   if (!srtp_ciphers_.empty()) {
-    if (!dtls_->SetDtlsSrtpCiphers(srtp_ciphers_)) {
+    if (!dtls_->SetDtlsSrtpCryptoSuites(srtp_ciphers_)) {
       LOG_J(LS_ERROR, this) << "Couldn't set DTLS-SRTP ciphers.";
       return false;
     }
   } else {
-    LOG_J(LS_INFO, this) << "Not using DTLS.";
+    LOG_J(LS_INFO, this) << "Not using DTLS-SRTP.";
   }
 
   LOG_J(LS_INFO, this) << "DTLS setup complete.";
   return true;
 }
 
-bool DtlsTransportChannelWrapper::SetSrtpCiphers(
-    const std::vector<std::string>& ciphers) {
+bool DtlsTransportChannelWrapper::SetSrtpCryptoSuites(
+    const std::vector<int>& ciphers) {
   if (srtp_ciphers_ == ciphers)
     return true;
 
-  if (dtls_state_ == STATE_STARTED) {
+  if (dtls_state() == DTLS_TRANSPORT_CONNECTING) {
     LOG(LS_WARNING) << "Ignoring new SRTP ciphers while DTLS is negotiating";
     return true;
   }
 
-  if (dtls_state_ == STATE_OPEN) {
+  if (dtls_state() == DTLS_TRANSPORT_CONNECTED) {
     // We don't support DTLS renegotiation currently. If new set of srtp ciphers
     // are different than what's being used currently, we will not use it.
     // So for now, let's be happy (or sad) with a warning message.
-    std::string current_srtp_cipher;
-    if (!dtls_->GetDtlsSrtpCipher(&current_srtp_cipher)) {
+    int current_srtp_cipher;
+    if (!dtls_->GetDtlsSrtpCryptoSuite(&current_srtp_cipher)) {
       LOG(LS_ERROR) << "Failed to get the current SRTP cipher for DTLS channel";
       return false;
     }
-    const std::vector<std::string>::const_iterator iter =
+    const std::vector<int>::const_iterator iter =
         std::find(ciphers.begin(), ciphers.end(), current_srtp_cipher);
     if (iter == ciphers.end()) {
       std::string requested_str;
       for (size_t i = 0; i < ciphers.size(); ++i) {
         requested_str.append(" ");
-        requested_str.append(ciphers[i]);
+        requested_str.append(rtc::SrtpCryptoSuiteToName(ciphers[i]));
         requested_str.append(" ");
       }
       LOG(LS_WARNING) << "Ignoring new set of SRTP ciphers, as DTLS "
@@ -319,10 +327,7 @@ bool DtlsTransportChannelWrapper::SetSrtpCiphers(
     return true;
   }
 
-  if (dtls_state_ != STATE_NONE &&
-      dtls_state_ != STATE_OFFERED &&
-      dtls_state_ != STATE_ACCEPTED) {
-    ASSERT(false);
+  if (!VERIFY(dtls_state() == DTLS_TRANSPORT_NEW)) {
     return false;
   }
 
@@ -330,12 +335,12 @@ bool DtlsTransportChannelWrapper::SetSrtpCiphers(
   return true;
 }
 
-bool DtlsTransportChannelWrapper::GetSrtpCipher(std::string* cipher) {
-  if (dtls_state_ != STATE_OPEN) {
+bool DtlsTransportChannelWrapper::GetSrtpCryptoSuite(int* cipher) {
+  if (dtls_state() != DTLS_TRANSPORT_CONNECTED) {
     return false;
   }
 
-  return dtls_->GetDtlsSrtpCipher(cipher);
+  return dtls_->GetDtlsSrtpCryptoSuite(cipher);
 }
 
 
@@ -343,107 +348,101 @@ bool DtlsTransportChannelWrapper::GetSrtpCipher(std::string* cipher) {
 int DtlsTransportChannelWrapper::SendPacket(
     const char* data, size_t size,
     const rtc::PacketOptions& options, int flags) {
-  int result = -1;
+  if (!dtls_active_) {
+    // Not doing DTLS.
+    return channel_->SendPacket(data, size, options);
+  }
 
-  switch (dtls_state_) {
-    case STATE_OFFERED:
-      // We don't know if we are doing DTLS yet, so we can't send a packet.
-      // TODO(ekr@rtfm.com): assert here?
-      result = -1;
-      break;
-
-    case STATE_STARTED:
-    case STATE_ACCEPTED:
-      // Can't send data until the connection is active
-      result = -1;
-      break;
-
-    case STATE_OPEN:
+  switch (dtls_state()) {
+    case DTLS_TRANSPORT_NEW:
+      // Can't send data until the connection is active.
+      // TODO(ekr@rtfm.com): assert here if dtls_ is NULL?
+      return -1;
+    case DTLS_TRANSPORT_CONNECTING:
+      // Can't send data until the connection is active.
+      return -1;
+    case DTLS_TRANSPORT_CONNECTED:
       if (flags & PF_SRTP_BYPASS) {
         ASSERT(!srtp_ciphers_.empty());
         if (!IsRtpPacket(data, size)) {
-          result = -1;
-          break;
+          return -1;
         }
 
-        result = channel_->SendPacket(data, size, options);
+        return channel_->SendPacket(data, size, options);
       } else {
-        result = (dtls_->WriteAll(data, size, NULL, NULL) ==
-          rtc::SR_SUCCESS) ? static_cast<int>(size) : -1;
+        return (dtls_->WriteAll(data, size, NULL, NULL) == rtc::SR_SUCCESS)
+                   ? static_cast<int>(size)
+                   : -1;
       }
-      break;
-      // Not doing DTLS.
-    case STATE_NONE:
-      result = channel_->SendPacket(data, size, options);
-      break;
-
-    case STATE_CLOSED:  // Can't send anything when we're closed.
+    case DTLS_TRANSPORT_FAILED:
+    case DTLS_TRANSPORT_CLOSED:
+      // Can't send anything when we're closed.
+      return -1;
+    default:
+      ASSERT(false);
       return -1;
   }
-
-  return result;
 }
 
 // The state transition logic here is as follows:
 // (1) If we're not doing DTLS-SRTP, then the state is just the
 //     state of the underlying impl()
 // (2) If we're doing DTLS-SRTP:
-//     - Prior to the DTLS handshake, the state is neither readable or
+//     - Prior to the DTLS handshake, the state is neither receiving nor
 //       writable
 //     - When the impl goes writable for the first time we
 //       start the DTLS handshake
 //     - Once the DTLS handshake completes, the state is that of the
 //       impl again
-void DtlsTransportChannelWrapper::OnReadableState(TransportChannel* channel) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == channel_);
-  LOG_J(LS_VERBOSE, this)
-      << "DTLSTransportChannelWrapper: channel readable state changed.";
-
-  if (dtls_state_ == STATE_NONE || dtls_state_ == STATE_OPEN) {
-    set_readable(channel_->readable());
-    // Note: SignalReadableState fired by set_readable.
-  }
-}
-
 void DtlsTransportChannelWrapper::OnWritableState(TransportChannel* channel) {
   ASSERT(rtc::Thread::Current() == worker_thread_);
   ASSERT(channel == channel_);
   LOG_J(LS_VERBOSE, this)
-      << "DTLSTransportChannelWrapper: channel writable state changed.";
+      << "DTLSTransportChannelWrapper: channel writable state changed to "
+      << channel_->writable();
 
-  switch (dtls_state_) {
-    case STATE_NONE:
-    case STATE_OPEN:
-      set_writable(channel_->writable());
+  if (!dtls_active_) {
+    // Not doing DTLS.
+    // Note: SignalWritableState fired by set_writable.
+    set_writable(channel_->writable());
+    return;
+  }
+
+  switch (dtls_state()) {
+    case DTLS_TRANSPORT_NEW:
+      // This should never fail:
+      // Because we are operating in a nonblocking mode and all
+      // incoming packets come in via OnReadPacket(), which rejects
+      // packets in this state, the incoming queue must be empty. We
+      // ignore write errors, thus any errors must be because of
+      // configuration and therefore are our fault.
+      // Note that in non-debug configurations, failure in
+      // MaybeStartDtls() changes the state to DTLS_TRANSPORT_FAILED.
+      VERIFY(MaybeStartDtls());
+      break;
+    case DTLS_TRANSPORT_CONNECTED:
       // Note: SignalWritableState fired by set_writable.
+      set_writable(channel_->writable());
       break;
+    case DTLS_TRANSPORT_CONNECTING:
+      // Do nothing.
+      break;
+    case DTLS_TRANSPORT_FAILED:
+    case DTLS_TRANSPORT_CLOSED:
+      // Should not happen. Do nothing.
+      break;
+  }
+}
 
-    case STATE_OFFERED:
-      // Do nothing
-      break;
-
-    case STATE_ACCEPTED:
-      if (!MaybeStartDtls()) {
-        // This should never happen:
-        // Because we are operating in a nonblocking mode and all
-        // incoming packets come in via OnReadPacket(), which rejects
-        // packets in this state, the incoming queue must be empty. We
-        // ignore write errors, thus any errors must be because of
-        // configuration and therefore are our fault.
-        // Note that in non-debug configurations, failure in
-        // MaybeStartDtls() changes the state to STATE_CLOSED.
-        ASSERT(false);
-      }
-      break;
-
-    case STATE_STARTED:
-      // Do nothing
-      break;
-
-    case STATE_CLOSED:
-      // Should not happen. Do nothing
-      break;
+void DtlsTransportChannelWrapper::OnReceivingState(TransportChannel* channel) {
+  ASSERT(rtc::Thread::Current() == worker_thread_);
+  ASSERT(channel == channel_);
+  LOG_J(LS_VERBOSE, this)
+      << "DTLSTransportChannelWrapper: channel receiving state changed to "
+      << channel_->receiving();
+  if (!dtls_active_ || dtls_state() == DTLS_TRANSPORT_CONNECTED) {
+    // Note: SignalReceivingState fired by set_receiving.
+    set_receiving(channel_->receiving());
   }
 }
 
@@ -454,28 +453,29 @@ void DtlsTransportChannelWrapper::OnReadPacket(
   ASSERT(channel == channel_);
   ASSERT(flags == 0);
 
-  switch (dtls_state_) {
-    case STATE_NONE:
-      // We are not doing DTLS
-      SignalReadPacket(this, data, size, packet_time, 0);
+  if (!dtls_active_) {
+    // Not doing DTLS.
+    SignalReadPacket(this, data, size, packet_time, 0);
+    return;
+  }
+
+  switch (dtls_state()) {
+    case DTLS_TRANSPORT_NEW:
+      if (dtls_) {
+        // Drop packets received before DTLS has actually started.
+        LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started.";
+      } else {
+        // Currently drop the packet, but we might in future
+        // decide to take this as evidence that the other
+        // side is ready to do DTLS and start the handshake
+        // on our end.
+        LOG_J(LS_WARNING, this) << "Received packet before we know if we are "
+                                << "doing DTLS or not; dropping.";
+      }
       break;
 
-    case STATE_OFFERED:
-      // Currently drop the packet, but we might in future
-      // decide to take this as evidence that the other
-      // side is ready to do DTLS and start the handshake
-      // on our end
-      LOG_J(LS_WARNING, this) << "Received packet before we know if we are "
-                              << "doing DTLS or not; dropping.";
-      break;
-
-    case STATE_ACCEPTED:
-      // Drop packets received before DTLS has actually started
-      LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started.";
-      break;
-
-    case STATE_STARTED:
-    case STATE_OPEN:
+    case DTLS_TRANSPORT_CONNECTING:
+    case DTLS_TRANSPORT_CONNECTED:
       // We should only get DTLS or SRTP packets; STUN's already been demuxed.
       // Is this potentially a DTLS packet?
       if (IsDtlsPacket(data, size)) {
@@ -485,7 +485,7 @@ void DtlsTransportChannelWrapper::OnReadPacket(
         }
       } else {
         // Not a DTLS packet; our handshake should be complete by now.
-        if (dtls_state_ != STATE_OPEN) {
+        if (dtls_state() != DTLS_TRANSPORT_CONNECTED) {
           LOG_J(LS_ERROR, this) << "Received non-DTLS packet before DTLS "
                                 << "complete.";
           return;
@@ -504,12 +504,21 @@ void DtlsTransportChannelWrapper::OnReadPacket(
         SignalReadPacket(this, data, size, packet_time, PF_SRTP_BYPASS);
       }
       break;
-    case STATE_CLOSED:
-      // This shouldn't be happening. Drop the packet
+    case DTLS_TRANSPORT_FAILED:
+    case DTLS_TRANSPORT_CLOSED:
+      // This shouldn't be happening. Drop the packet.
       break;
   }
 }
 
+void DtlsTransportChannelWrapper::OnSentPacket(
+    TransportChannel* channel,
+    const rtc::SentPacket& sent_packet) {
+  ASSERT(rtc::Thread::Current() == worker_thread_);
+
+  SignalSentPacket(this, sent_packet);
+}
+
 void DtlsTransportChannelWrapper::OnReadyToSend(TransportChannel* channel) {
   if (writable()) {
     SignalReadyToSend(this);
@@ -526,9 +535,7 @@ void DtlsTransportChannelWrapper::OnDtlsEvent(rtc::StreamInterface* dtls,
     if (dtls_->GetState() == rtc::SS_OPEN) {
       // The check for OPEN shouldn't be necessary but let's make
       // sure we don't accidentally frob the state if it's closed.
-      dtls_state_ = STATE_OPEN;
-
-      set_readable(true);
+      set_dtls_state(DTLS_TRANSPORT_CONNECTED);
       set_writable(true);
     }
   }
@@ -541,29 +548,27 @@ void DtlsTransportChannelWrapper::OnDtlsEvent(rtc::StreamInterface* dtls,
   }
   if (sig & rtc::SE_CLOSE) {
     ASSERT(sig == rtc::SE_CLOSE);  // SE_CLOSE should be by itself.
+    set_writable(false);
     if (!err) {
       LOG_J(LS_INFO, this) << "DTLS channel closed";
+      set_dtls_state(DTLS_TRANSPORT_CLOSED);
     } else {
       LOG_J(LS_INFO, this) << "DTLS channel error, code=" << err;
+      set_dtls_state(DTLS_TRANSPORT_FAILED);
     }
-
-    set_readable(false);
-    set_writable(false);
-    dtls_state_ = STATE_CLOSED;
   }
 }
 
 bool DtlsTransportChannelWrapper::MaybeStartDtls() {
-  if (channel_->writable()) {
+  if (dtls_ && channel_->writable()) {
     if (dtls_->StartSSLWithPeer()) {
       LOG_J(LS_ERROR, this) << "Couldn't start DTLS handshake";
-      dtls_state_ = STATE_CLOSED;
+      set_dtls_state(DTLS_TRANSPORT_FAILED);
       return false;
     }
     LOG_J(LS_INFO, this)
       << "DtlsTransportChannelWrapper: Started DTLS handshake";
-
-    dtls_state_ = STATE_STARTED;
+    set_dtls_state(DTLS_TRANSPORT_CONNECTING);
   }
   return true;
 }
@@ -573,7 +578,7 @@ bool DtlsTransportChannelWrapper::HandleDtlsPacket(const char* data,
                                                    size_t size) {
   // Sanity check we're not passing junk that
   // just looks like DTLS.
-  const uint8* tmp_data = reinterpret_cast<const uint8* >(data);
+  const uint8_t* tmp_data = reinterpret_cast<const uint8_t*>(data);
   size_t tmp_size = size;
   while (tmp_size > 0) {
     if (tmp_size < kDtlsRecordHeaderLen)
@@ -592,22 +597,17 @@ bool DtlsTransportChannelWrapper::HandleDtlsPacket(const char* data,
   return downward_->OnPacketReceived(data, size);
 }
 
-void DtlsTransportChannelWrapper::OnRequestSignaling(
+void DtlsTransportChannelWrapper::OnGatheringState(
     TransportChannelImpl* channel) {
   ASSERT(channel == channel_);
-  SignalRequestSignaling(this);
+  SignalGatheringState(this);
 }
 
-void DtlsTransportChannelWrapper::OnCandidateReady(
-    TransportChannelImpl* channel, const Candidate& c) {
+void DtlsTransportChannelWrapper::OnCandidateGathered(
+    TransportChannelImpl* channel,
+    const Candidate& c) {
   ASSERT(channel == channel_);
-  SignalCandidateReady(this, c);
-}
-
-void DtlsTransportChannelWrapper::OnCandidatesAllocationDone(
-    TransportChannelImpl* channel) {
-  ASSERT(channel == channel_);
-  SignalCandidatesAllocationDone(this);
+  SignalCandidateGathered(this, c);
 }
 
 void DtlsTransportChannelWrapper::OnRoleConflict(
@@ -628,4 +628,12 @@ void DtlsTransportChannelWrapper::OnConnectionRemoved(
   SignalConnectionRemoved(this);
 }
 
+void DtlsTransportChannelWrapper::Reconnect() {
+  set_dtls_state(DTLS_TRANSPORT_NEW);
+  set_writable(false);
+  if (channel_->writable()) {
+    OnWritableState(channel_);
+  }
+}
+
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.h b/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.h
index 03a916b767..955b963a36 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel.h
@@ -16,6 +16,7 @@
 
 #include "webrtc/p2p/base/transportchannelimpl.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/bufferqueue.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stream.h"
@@ -24,38 +25,31 @@ namespace cricket {
 
 // A bridge between a packet-oriented/channel-type interface on
 // the bottom and a StreamInterface on the top.
-class StreamInterfaceChannel : public rtc::StreamInterface,
-                               public sigslot::has_slots<> {
+class StreamInterfaceChannel : public rtc::StreamInterface {
  public:
-  StreamInterfaceChannel(rtc::Thread* owner, TransportChannel* channel)
-      : channel_(channel),
-        state_(rtc::SS_OPEN),
-        fifo_(kFifoSize, owner) {
-    fifo_.SignalEvent.connect(this, &StreamInterfaceChannel::OnEvent);
-  }
+  explicit StreamInterfaceChannel(TransportChannel* channel);
 
   // Push in a packet; this gets pulled out from Read().
   bool OnPacketReceived(const char* data, size_t size);
 
   // Implementations of StreamInterface
-  virtual rtc::StreamState GetState() const { return state_; }
-  virtual void Close() { state_ = rtc::SS_CLOSED; }
-  virtual rtc::StreamResult Read(void* buffer, size_t buffer_len,
-                                       size_t* read, int* error);
-  virtual rtc::StreamResult Write(const void* data, size_t data_len,
-                                        size_t* written, int* error);
+  rtc::StreamState GetState() const override { return state_; }
+  void Close() override { state_ = rtc::SS_CLOSED; }
+  rtc::StreamResult Read(void* buffer,
+                         size_t buffer_len,
+                         size_t* read,
+                         int* error) override;
+  rtc::StreamResult Write(const void* data,
+                          size_t data_len,
+                          size_t* written,
+                          int* error) override;
 
  private:
-  static const size_t kFifoSize = 8192;
-
-  // Forward events
-  virtual void OnEvent(rtc::StreamInterface* stream, int sig, int err);
-
   TransportChannel* channel_;  // owned by DtlsTransportChannelWrapper
   rtc::StreamState state_;
-  rtc::FifoBuffer fifo_;
+  rtc::BufferQueue packets_;
 
-  DISALLOW_COPY_AND_ASSIGN(StreamInterfaceChannel);
+  RTC_DISALLOW_COPY_AND_ASSIGN(StreamInterfaceChannel);
 };
 
 
@@ -87,86 +81,76 @@ class StreamInterfaceChannel : public rtc::StreamInterface,
 //     which translates it into packet writes on channel_.
 class DtlsTransportChannelWrapper : public TransportChannelImpl {
  public:
-    enum State {
-      STATE_NONE,      // No state or rejected.
-      STATE_OFFERED,   // Our identity has been set.
-      STATE_ACCEPTED,  // The other side sent a fingerprint.
-      STATE_STARTED,   // We are negotiating.
-      STATE_OPEN,      // Negotiation complete.
-      STATE_CLOSED     // Connection closed.
-    };
-
   // The parameters here are:
   // transport -- the DtlsTransport that created us
   // channel -- the TransportChannel we are wrapping
   DtlsTransportChannelWrapper(Transport* transport,
                               TransportChannelImpl* channel);
-  virtual ~DtlsTransportChannelWrapper();
+  ~DtlsTransportChannelWrapper() override;
 
-  virtual void SetIceRole(IceRole role) {
-    channel_->SetIceRole(role);
-  }
-  virtual IceRole GetIceRole() const {
-    return channel_->GetIceRole();
-  }
-  virtual bool SetLocalIdentity(rtc::SSLIdentity *identity);
-  virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const;
+  void SetIceRole(IceRole role) override { channel_->SetIceRole(role); }
+  IceRole GetIceRole() const override { return channel_->GetIceRole(); }
+  bool SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override;
+  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override;
 
-  virtual bool SetRemoteFingerprint(const std::string& digest_alg,
-                                    const uint8* digest,
-                                    size_t digest_len);
-  virtual bool IsDtlsActive() const { return dtls_state_ != STATE_NONE; }
+  bool SetRemoteFingerprint(const std::string& digest_alg,
+                            const uint8_t* digest,
+                            size_t digest_len) override;
+
+  // Returns false if no local certificate was set, or if the peer doesn't
+  // support DTLS.
+  bool IsDtlsActive() const override { return dtls_active_; }
 
   // Called to send a packet (via DTLS, if turned on).
-  virtual int SendPacket(const char* data, size_t size,
-                         const rtc::PacketOptions& options,
-                         int flags);
+  int SendPacket(const char* data,
+                 size_t size,
+                 const rtc::PacketOptions& options,
+                 int flags) override;
 
   // TransportChannel calls that we forward to the wrapped transport.
-  virtual int SetOption(rtc::Socket::Option opt, int value) {
+  int SetOption(rtc::Socket::Option opt, int value) override {
     return channel_->SetOption(opt, value);
   }
-  virtual bool GetOption(rtc::Socket::Option opt, int* value) {
+  bool GetOption(rtc::Socket::Option opt, int* value) override {
     return channel_->GetOption(opt, value);
   }
-  virtual int GetError() {
-    return channel_->GetError();
-  }
-  virtual bool GetStats(ConnectionInfos* infos) {
+  int GetError() override { return channel_->GetError(); }
+  bool GetStats(ConnectionInfos* infos) override {
     return channel_->GetStats(infos);
   }
-  virtual const std::string SessionId() const {
-    return channel_->SessionId();
-  }
+  const std::string SessionId() const override { return channel_->SessionId(); }
+
+  virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version);
 
   // Set up the ciphers to use for DTLS-SRTP. If this method is not called
   // before DTLS starts, or |ciphers| is empty, SRTP keys won't be negotiated.
   // This method should be called before SetupDtls.
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers);
+  bool SetSrtpCryptoSuites(const std::vector<int>& ciphers) override;
 
   // Find out which DTLS-SRTP cipher was negotiated
-  virtual bool GetSrtpCipher(std::string* cipher);
+  bool GetSrtpCryptoSuite(int* cipher) override;
 
-  virtual bool GetSslRole(rtc::SSLRole* role) const;
-  virtual bool SetSslRole(rtc::SSLRole role);
+  bool GetSslRole(rtc::SSLRole* role) const override;
+  bool SetSslRole(rtc::SSLRole role) override;
 
   // Find out which DTLS cipher was negotiated
-  virtual bool GetSslCipher(std::string* cipher);
+  bool GetSslCipherSuite(int* cipher) override;
 
   // Once DTLS has been established, this method retrieves the certificate in
   // use by the remote peer, for use in external identity verification.
-  virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const;
+  bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert) const override;
 
   // Once DTLS has established (i.e., this channel is writable), this method
   // extracts the keys negotiated during the DTLS handshake, for use in external
   // encryption. DTLS-SRTP uses this to extract the needed SRTP keys.
   // See the SSLStreamAdapter documentation for info on the specific parameters.
-  virtual bool ExportKeyingMaterial(const std::string& label,
-                                    const uint8* context,
-                                    size_t context_len,
-                                    bool use_context,
-                                    uint8* result,
-                                    size_t result_len) {
+  bool ExportKeyingMaterial(const std::string& label,
+                            const uint8_t* context,
+                            size_t context_len,
+                            bool use_context,
+                            uint8_t* result,
+                            size_t result_len) override {
     return (dtls_.get()) ? dtls_->ExportKeyingMaterial(label, context,
                                                        context_len,
                                                        use_context,
@@ -175,42 +159,40 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   }
 
   // TransportChannelImpl calls.
-  virtual Transport* GetTransport() {
-    return transport_;
-  }
+  Transport* GetTransport() override { return transport_; }
 
-  virtual TransportChannelState GetState() const {
+  TransportChannelState GetState() const override {
     return channel_->GetState();
   }
-  virtual void SetIceTiebreaker(uint64 tiebreaker) {
+  void SetIceTiebreaker(uint64_t tiebreaker) override {
     channel_->SetIceTiebreaker(tiebreaker);
   }
-  virtual bool GetIceProtocolType(IceProtocolType* type) const {
-    return channel_->GetIceProtocolType(type);
-  }
-  virtual void SetIceProtocolType(IceProtocolType type) {
-    channel_->SetIceProtocolType(type);
-  }
-  virtual void SetIceCredentials(const std::string& ice_ufrag,
-                                 const std::string& ice_pwd) {
+  void SetIceCredentials(const std::string& ice_ufrag,
+                         const std::string& ice_pwd) override {
     channel_->SetIceCredentials(ice_ufrag, ice_pwd);
   }
-  virtual void SetRemoteIceCredentials(const std::string& ice_ufrag,
-                                       const std::string& ice_pwd) {
+  void SetRemoteIceCredentials(const std::string& ice_ufrag,
+                               const std::string& ice_pwd) override {
     channel_->SetRemoteIceCredentials(ice_ufrag, ice_pwd);
   }
-  virtual void SetRemoteIceMode(IceMode mode) {
+  void SetRemoteIceMode(IceMode mode) override {
     channel_->SetRemoteIceMode(mode);
   }
 
-  virtual void Connect();
-  virtual void Reset();
+  void Connect() override;
 
-  virtual void OnSignalingReady() {
-    channel_->OnSignalingReady();
+  void MaybeStartGathering() override { channel_->MaybeStartGathering(); }
+
+  IceGatheringState gathering_state() const override {
+    return channel_->gathering_state();
   }
-  virtual void OnCandidate(const Candidate& candidate) {
-    channel_->OnCandidate(candidate);
+
+  void AddRemoteCandidate(const Candidate& candidate) override {
+    channel_->AddRemoteCandidate(candidate);
+  }
+
+  void SetIceConfig(const IceConfig& config) override {
+    channel_->SetIceConfig(config);
   }
 
   // Needed by DtlsTransport.
@@ -221,31 +203,36 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   void OnWritableState(TransportChannel* channel);
   void OnReadPacket(TransportChannel* channel, const char* data, size_t size,
                     const rtc::PacketTime& packet_time, int flags);
+  void OnSentPacket(TransportChannel* channel,
+                    const rtc::SentPacket& sent_packet);
   void OnReadyToSend(TransportChannel* channel);
+  void OnReceivingState(TransportChannel* channel);
   void OnDtlsEvent(rtc::StreamInterface* stream_, int sig, int err);
   bool SetupDtls();
   bool MaybeStartDtls();
   bool HandleDtlsPacket(const char* data, size_t size);
-  void OnRequestSignaling(TransportChannelImpl* channel);
-  void OnCandidateReady(TransportChannelImpl* channel, const Candidate& c);
-  void OnCandidatesAllocationDone(TransportChannelImpl* channel);
+  void OnGatheringState(TransportChannelImpl* channel);
+  void OnCandidateGathered(TransportChannelImpl* channel, const Candidate& c);
   void OnRoleConflict(TransportChannelImpl* channel);
   void OnRouteChange(TransportChannel* channel, const Candidate& candidate);
   void OnConnectionRemoved(TransportChannelImpl* channel);
+  void Reconnect();
 
   Transport* transport_;  // The transport_ that created us.
   rtc::Thread* worker_thread_;  // Everything should occur on this thread.
-  TransportChannelImpl* channel_;  // Underlying channel, owned by transport_.
+  // Underlying channel, owned by transport_.
+  TransportChannelImpl* const channel_;
   rtc::scoped_ptr<rtc::SSLStreamAdapter> dtls_;  // The DTLS stream
   StreamInterfaceChannel* downward_;  // Wrapper for channel_, owned by dtls_.
-  std::vector<std::string> srtp_ciphers_;  // SRTP ciphers to use with DTLS.
-  State dtls_state_;
-  rtc::SSLIdentity* local_identity_;
+  std::vector<int> srtp_ciphers_;     // SRTP ciphers to use with DTLS.
+  bool dtls_active_ = false;
+  rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
   rtc::SSLRole ssl_role_;
+  rtc::SSLProtocolVersion ssl_max_version_;
   rtc::Buffer remote_fingerprint_value_;
   std::string remote_fingerprint_algorithm_;
 
-  DISALLOW_COPY_AND_ASSIGN(DtlsTransportChannelWrapper);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransportChannelWrapper);
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel_unittest.cc
index f3086bb3a0..3791893442 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/dtlstransportchannel_unittest.cc
@@ -11,7 +11,7 @@
 #include <set>
 
 #include "webrtc/p2p/base/dtlstransport.h"
-#include "webrtc/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/gunit.h"
@@ -21,7 +21,6 @@
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
-#include "webrtc/base/thread.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \
   if (!(rtc::SSLStreamAdapter::feature())) {  \
@@ -29,13 +28,13 @@
     return;                                         \
   }
 
-static const char AES_CM_128_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80";
 static const char kIceUfrag1[] = "TESTICEUFRAG0001";
 static const char kIcePwd1[] = "TESTICEPWD00000000000001";
 static const size_t kPacketNumOffset = 8;
 static const size_t kPacketHeaderLen = 12;
+static const int kFakePacketId = 0x1234;
 
-static bool IsRtpLeadByte(uint8 b) {
+static bool IsRtpLeadByte(uint8_t b) {
   return ((b & 0xC0) == 0x80);
 }
 
@@ -45,50 +44,50 @@ enum Flags { NF_REOFFER = 0x1, NF_EXPECT_FAILURE = 0x2 };
 
 class DtlsTestClient : public sigslot::has_slots<> {
  public:
-  DtlsTestClient(const std::string& name,
-                 rtc::Thread* signaling_thread,
-                 rtc::Thread* worker_thread) :
-      name_(name),
-      signaling_thread_(signaling_thread),
-      worker_thread_(worker_thread),
-      protocol_(cricket::ICEPROTO_GOOGLE),
-      packet_size_(0),
-      use_dtls_srtp_(false),
-      negotiated_dtls_(false),
-      received_dtls_client_hello_(false),
-      received_dtls_server_hello_(false) {
+  DtlsTestClient(const std::string& name)
+      : name_(name),
+        packet_size_(0),
+        use_dtls_srtp_(false),
+        ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
+        negotiated_dtls_(false),
+        received_dtls_client_hello_(false),
+        received_dtls_server_hello_(false) {}
+  void CreateCertificate(rtc::KeyType key_type) {
+    certificate_ =
+        rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+            rtc::SSLIdentity::Generate(name_, key_type)));
   }
-  void SetIceProtocol(cricket::TransportProtocol proto) {
-    protocol_ = proto;
+  const rtc::scoped_refptr<rtc::RTCCertificate>& certificate() {
+    return certificate_;
   }
-  void CreateIdentity() {
-    identity_.reset(rtc::SSLIdentity::Generate(name_));
-  }
-  rtc::SSLIdentity* identity() { return identity_.get(); }
   void SetupSrtp() {
-    ASSERT(identity_.get() != NULL);
+    ASSERT(certificate_);
     use_dtls_srtp_ = true;
   }
+  void SetupMaxProtocolVersion(rtc::SSLProtocolVersion version) {
+    ASSERT(!transport_);
+    ssl_max_version_ = version;
+  }
   void SetupChannels(int count, cricket::IceRole role) {
     transport_.reset(new cricket::DtlsTransport<cricket::FakeTransport>(
-        signaling_thread_, worker_thread_, "dtls content name", NULL,
-        identity_.get()));
+        "dtls content name", nullptr, certificate_));
     transport_->SetAsync(true);
     transport_->SetIceRole(role);
     transport_->SetIceTiebreaker(
         (role == cricket::ICEROLE_CONTROLLING) ? 1 : 2);
-    transport_->SignalWritableState.connect(this,
-        &DtlsTestClient::OnTransportWritableState);
 
     for (int i = 0; i < count; ++i) {
       cricket::DtlsTransportChannelWrapper* channel =
           static_cast<cricket::DtlsTransportChannelWrapper*>(
               transport_->CreateChannel(i));
       ASSERT_TRUE(channel != NULL);
+      channel->SetSslMaxProtocolVersion(ssl_max_version_);
       channel->SignalWritableState.connect(this,
         &DtlsTestClient::OnTransportChannelWritableState);
       channel->SignalReadPacket.connect(this,
         &DtlsTestClient::OnTransportChannelReadPacket);
+      channel->SignalSentPacket.connect(
+          this, &DtlsTestClient::OnTransportChannelSentPacket);
       channels_.push_back(channel);
 
       // Hook the raw packets so that we can verify they are encrypted.
@@ -112,55 +111,63 @@ class DtlsTestClient : public sigslot::has_slots<> {
   void Negotiate(DtlsTestClient* peer, cricket::ContentAction action,
                  ConnectionRole local_role, ConnectionRole remote_role,
                  int flags) {
-    Negotiate(identity_.get(), (identity_) ? peer->identity_.get() : NULL,
-              action, local_role, remote_role, flags);
+    Negotiate(certificate_, certificate_ ? peer->certificate_ : nullptr, action,
+              local_role, remote_role, flags);
   }
 
   // Allow any DTLS configuration to be specified (including invalid ones).
-  void Negotiate(rtc::SSLIdentity* local_identity,
-                 rtc::SSLIdentity* remote_identity,
+  void Negotiate(const rtc::scoped_refptr<rtc::RTCCertificate>& local_cert,
+                 const rtc::scoped_refptr<rtc::RTCCertificate>& remote_cert,
                  cricket::ContentAction action,
                  ConnectionRole local_role,
                  ConnectionRole remote_role,
                  int flags) {
     rtc::scoped_ptr<rtc::SSLFingerprint> local_fingerprint;
     rtc::scoped_ptr<rtc::SSLFingerprint> remote_fingerprint;
-    if (local_identity) {
+    if (local_cert) {
+      std::string digest_algorithm;
+      ASSERT_TRUE(local_cert->ssl_certificate().GetSignatureDigestAlgorithm(
+          &digest_algorithm));
+      ASSERT_FALSE(digest_algorithm.empty());
       local_fingerprint.reset(rtc::SSLFingerprint::Create(
-          rtc::DIGEST_SHA_1, local_identity));
+          digest_algorithm, local_cert->identity()));
       ASSERT_TRUE(local_fingerprint.get() != NULL);
+      EXPECT_EQ(rtc::DIGEST_SHA_256, digest_algorithm);
     }
-    if (remote_identity) {
+    if (remote_cert) {
+      std::string digest_algorithm;
+      ASSERT_TRUE(remote_cert->ssl_certificate().GetSignatureDigestAlgorithm(
+          &digest_algorithm));
+      ASSERT_FALSE(digest_algorithm.empty());
       remote_fingerprint.reset(rtc::SSLFingerprint::Create(
-          rtc::DIGEST_SHA_1, remote_identity));
+          digest_algorithm, remote_cert->identity()));
       ASSERT_TRUE(remote_fingerprint.get() != NULL);
+      EXPECT_EQ(rtc::DIGEST_SHA_256, digest_algorithm);
     }
 
     if (use_dtls_srtp_ && !(flags & NF_REOFFER)) {
       // SRTP ciphers will be set only in the beginning.
       for (std::vector<cricket::DtlsTransportChannelWrapper*>::iterator it =
            channels_.begin(); it != channels_.end(); ++it) {
-        std::vector<std::string> ciphers;
-        ciphers.push_back(AES_CM_128_HMAC_SHA1_80);
-        ASSERT_TRUE((*it)->SetSrtpCiphers(ciphers));
+        std::vector<int> ciphers;
+        ciphers.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+        ASSERT_TRUE((*it)->SetSrtpCryptoSuites(ciphers));
       }
     }
 
-    std::string transport_type = (protocol_ == cricket::ICEPROTO_GOOGLE) ?
-        cricket::NS_GINGLE_P2P : cricket::NS_JINGLE_ICE_UDP;
     cricket::TransportDescription local_desc(
-        transport_type, std::vector<std::string>(), kIceUfrag1, kIcePwd1,
-        cricket::ICEMODE_FULL, local_role,
-         // If remote if the offerer and has no DTLS support, answer will be
+        std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+        local_role,
+        // If remote if the offerer and has no DTLS support, answer will be
         // without any fingerprint.
-        (action == cricket::CA_ANSWER && !remote_identity) ?
-            NULL : local_fingerprint.get(),
+        (action == cricket::CA_ANSWER && !remote_cert)
+            ? nullptr
+            : local_fingerprint.get(),
         cricket::Candidates());
 
     cricket::TransportDescription remote_desc(
-        transport_type, std::vector<std::string>(), kIceUfrag1, kIcePwd1,
-        cricket::ICEMODE_FULL, remote_role, remote_fingerprint.get(),
-        cricket::Candidates());
+        std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+        remote_role, remote_fingerprint.get(), cricket::Candidates());
 
     bool expect_success = (flags & NF_EXPECT_FAILURE) ? false : true;
     // If |expect_success| is false, expect SRTD or SLTD to fail when
@@ -176,7 +183,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
       ASSERT_EQ(expect_success, transport_->SetLocalTransportDescription(
           local_desc, cricket::CA_ANSWER, NULL));
     }
-    negotiated_dtls_ = (local_identity && remote_identity);
+    negotiated_dtls_ = (local_cert && remote_cert);
   }
 
   bool Connect(DtlsTestClient* peer) {
@@ -185,7 +192,17 @@ class DtlsTestClient : public sigslot::has_slots<> {
     return true;
   }
 
-  bool writable() const { return transport_->writable(); }
+  bool all_channels_writable() const {
+    if (channels_.empty()) {
+      return false;
+    }
+    for (cricket::DtlsTransportChannelWrapper* channel : channels_) {
+      if (!channel->writable()) {
+        return false;
+      }
+    }
+    return true;
+  }
 
   void CheckRole(rtc::SSLRole role) {
     if (role == rtc::SSL_CLIENT) {
@@ -197,29 +214,29 @@ class DtlsTestClient : public sigslot::has_slots<> {
     }
   }
 
-  void CheckSrtp(const std::string& expected_cipher) {
+  void CheckSrtp(int expected_crypto_suite) {
     for (std::vector<cricket::DtlsTransportChannelWrapper*>::iterator it =
            channels_.begin(); it != channels_.end(); ++it) {
-      std::string cipher;
+      int crypto_suite;
 
-      bool rv = (*it)->GetSrtpCipher(&cipher);
-      if (negotiated_dtls_ && !expected_cipher.empty()) {
+      bool rv = (*it)->GetSrtpCryptoSuite(&crypto_suite);
+      if (negotiated_dtls_ && expected_crypto_suite) {
         ASSERT_TRUE(rv);
 
-        ASSERT_EQ(cipher, expected_cipher);
+        ASSERT_EQ(crypto_suite, expected_crypto_suite);
       } else {
         ASSERT_FALSE(rv);
       }
     }
   }
 
-  void CheckSsl(const std::string& expected_cipher) {
+  void CheckSsl(int expected_cipher) {
     for (std::vector<cricket::DtlsTransportChannelWrapper*>::iterator it =
            channels_.begin(); it != channels_.end(); ++it) {
-      std::string cipher;
+      int cipher;
 
-      bool rv = (*it)->GetSslCipher(&cipher);
-      if (negotiated_dtls_ && !expected_cipher.empty()) {
+      bool rv = (*it)->GetSslCipherSuite(&cipher);
+      if (negotiated_dtls_ && expected_cipher) {
         ASSERT_TRUE(rv);
 
         ASSERT_EQ(cipher, expected_cipher);
@@ -239,11 +256,12 @@ class DtlsTestClient : public sigslot::has_slots<> {
       memset(packet.get(), sent & 0xff, size);
       packet[0] = (srtp) ? 0x80 : 0x00;
       rtc::SetBE32(packet.get() + kPacketNumOffset,
-                         static_cast<uint32>(sent));
+                   static_cast<uint32_t>(sent));
 
       // Only set the bypass flag if we've activated DTLS.
-      int flags = (identity_.get() && srtp) ? cricket::PF_SRTP_BYPASS : 0;
+      int flags = (certificate_ && srtp) ? cricket::PF_SRTP_BYPASS : 0;
       rtc::PacketOptions packet_options;
+      packet_options.packet_id = kFakePacketId;
       int rv = channels_[channel]->SendPacket(
           packet.get(), size, packet_options, flags);
       ASSERT_GT(rv, 0);
@@ -272,14 +290,14 @@ class DtlsTestClient : public sigslot::has_slots<> {
     return received_.size();
   }
 
-  bool VerifyPacket(const char* data, size_t size, uint32* out_num) {
+  bool VerifyPacket(const char* data, size_t size, uint32_t* out_num) {
     if (size != packet_size_ ||
-        (data[0] != 0 && static_cast<uint8>(data[0]) != 0x80)) {
+        (data[0] != 0 && static_cast<uint8_t>(data[0]) != 0x80)) {
       return false;
     }
-    uint32 packet_num = rtc::GetBE32(data + kPacketNumOffset);
+    uint32_t packet_num = rtc::GetBE32(data + kPacketNumOffset);
     for (size_t i = kPacketHeaderLen; i < size; ++i) {
-      if (static_cast<uint8>(data[i]) != (packet_num & 0xff)) {
+      if (static_cast<uint8_t>(data[i]) != (packet_num & 0xff)) {
         return false;
       }
     }
@@ -294,21 +312,16 @@ class DtlsTestClient : public sigslot::has_slots<> {
     if (size <= packet_size_) {
       return false;
     }
-    uint32 packet_num = rtc::GetBE32(data + kPacketNumOffset);
+    uint32_t packet_num = rtc::GetBE32(data + kPacketNumOffset);
     int num_matches = 0;
     for (size_t i = kPacketNumOffset; i < size; ++i) {
-      if (static_cast<uint8>(data[i]) == (packet_num & 0xff)) {
+      if (static_cast<uint8_t>(data[i]) == (packet_num & 0xff)) {
         ++num_matches;
       }
     }
     return (num_matches < ((static_cast<int>(size) - 5) / 10));
   }
 
-  // Transport callbacks
-  void OnTransportWritableState(cricket::Transport* transport) {
-    LOG(LS_INFO) << name_ << ": is writable";
-  }
-
   // Transport channel callbacks
   void OnTransportChannelWritableState(cricket::TransportChannel* channel) {
     LOG(LS_INFO) << name_ << ": Channel '" << channel->component()
@@ -319,15 +332,22 @@ class DtlsTestClient : public sigslot::has_slots<> {
                                     const char* data, size_t size,
                                     const rtc::PacketTime& packet_time,
                                     int flags) {
-    uint32 packet_num = 0;
+    uint32_t packet_num = 0;
     ASSERT_TRUE(VerifyPacket(data, size, &packet_num));
     received_.insert(packet_num);
     // Only DTLS-SRTP packets should have the bypass flag set.
-    int expected_flags = (identity_.get() && IsRtpLeadByte(data[0])) ?
-        cricket::PF_SRTP_BYPASS : 0;
+    int expected_flags =
+        (certificate_ && IsRtpLeadByte(data[0])) ? cricket::PF_SRTP_BYPASS : 0;
     ASSERT_EQ(expected_flags, flags);
   }
 
+  void OnTransportChannelSentPacket(cricket::TransportChannel* channel,
+                                    const rtc::SentPacket& sent_packet) {
+    sent_packet_ = sent_packet;
+  }
+
+  rtc::SentPacket sent_packet() const { return sent_packet_; }
+
   // Hook into the raw packet stream to make sure DTLS packets are encrypted.
   void OnFakeTransportChannelReadPacket(cricket::TransportChannel* channel,
                                         const char* data, size_t size,
@@ -358,42 +378,45 @@ class DtlsTestClient : public sigslot::has_slots<> {
 
  private:
   std::string name_;
-  rtc::Thread* signaling_thread_;
-  rtc::Thread* worker_thread_;
-  cricket::TransportProtocol protocol_;
-  rtc::scoped_ptr<rtc::SSLIdentity> identity_;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
   rtc::scoped_ptr<cricket::FakeTransport> transport_;
   std::vector<cricket::DtlsTransportChannelWrapper*> channels_;
   size_t packet_size_;
   std::set<int> received_;
   bool use_dtls_srtp_;
+  rtc::SSLProtocolVersion ssl_max_version_;
   bool negotiated_dtls_;
   bool received_dtls_client_hello_;
   bool received_dtls_server_hello_;
+  rtc::SentPacket sent_packet_;
 };
 
 
 class DtlsTransportChannelTest : public testing::Test {
  public:
-  DtlsTransportChannelTest() :
-      client1_("P1", rtc::Thread::Current(),
-               rtc::Thread::Current()),
-      client2_("P2", rtc::Thread::Current(),
-               rtc::Thread::Current()),
-      channel_ct_(1),
-      use_dtls_(false),
-      use_dtls_srtp_(false) {
-  }
+  DtlsTransportChannelTest()
+      : client1_("P1"),
+        client2_("P2"),
+        channel_ct_(1),
+        use_dtls_(false),
+        use_dtls_srtp_(false),
+        ssl_expected_version_(rtc::SSL_PROTOCOL_DTLS_12) {}
 
   void SetChannelCount(size_t channel_ct) {
     channel_ct_ = static_cast<int>(channel_ct);
   }
-  void PrepareDtls(bool c1, bool c2) {
+  void SetMaxProtocolVersions(rtc::SSLProtocolVersion c1,
+                              rtc::SSLProtocolVersion c2) {
+    client1_.SetupMaxProtocolVersion(c1);
+    client2_.SetupMaxProtocolVersion(c2);
+    ssl_expected_version_ = std::min(c1, c2);
+  }
+  void PrepareDtls(bool c1, bool c2, rtc::KeyType key_type) {
     if (c1) {
-      client1_.CreateIdentity();
+      client1_.CreateCertificate(key_type);
     }
     if (c2) {
-      client2_.CreateIdentity();
+      client2_.CreateCertificate(key_type);
     }
     if (c1 && c2)
       use_dtls_ = true;
@@ -419,8 +442,10 @@ class DtlsTransportChannelTest : public testing::Test {
     if (!rv)
       return false;
 
-    EXPECT_TRUE_WAIT(client1_.writable() && client2_.writable(), 10000);
-    if (!client1_.writable() || !client2_.writable())
+    EXPECT_TRUE_WAIT(
+        client1_.all_channels_writable() && client2_.all_channels_writable(),
+        10000);
+    if (!client1_.all_channels_writable() || !client2_.all_channels_writable())
       return false;
 
     // Check that we used the right roles.
@@ -443,14 +468,16 @@ class DtlsTransportChannelTest : public testing::Test {
 
     // Check that we negotiated the right ciphers.
     if (use_dtls_srtp_) {
-      client1_.CheckSrtp(AES_CM_128_HMAC_SHA1_80);
-      client2_.CheckSrtp(AES_CM_128_HMAC_SHA1_80);
+      client1_.CheckSrtp(rtc::SRTP_AES128_CM_SHA1_80);
+      client2_.CheckSrtp(rtc::SRTP_AES128_CM_SHA1_80);
     } else {
-      client1_.CheckSrtp("");
-      client2_.CheckSrtp("");
+      client1_.CheckSrtp(rtc::SRTP_INVALID_CRYPTO_SUITE);
+      client2_.CheckSrtp(rtc::SRTP_INVALID_CRYPTO_SUITE);
     }
-    client1_.CheckSsl(rtc::SSLStreamAdapter::GetDefaultSslCipher());
-    client2_.CheckSsl(rtc::SSLStreamAdapter::GetDefaultSslCipher());
+    client1_.CheckSsl(rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
+        ssl_expected_version_, rtc::KT_DEFAULT));
+    client2_.CheckSsl(rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
+        ssl_expected_version_, rtc::KT_DEFAULT));
 
     return true;
   }
@@ -518,12 +545,11 @@ class DtlsTransportChannelTest : public testing::Test {
   int channel_ct_;
   bool use_dtls_;
   bool use_dtls_srtp_;
+  rtc::SSLProtocolVersion ssl_expected_version_;
 };
 
 // Test that transport negotiation of ICE, no DTLS works properly.
 TEST_F(DtlsTransportChannelTest, TestChannelSetupIce) {
-  client1_.SetIceProtocol(cricket::ICEPROTO_RFC5245);
-  client2_.SetIceProtocol(cricket::ICEPROTO_RFC5245);
   Negotiate();
   cricket::FakeTransportChannel* channel1 = client1_.GetFakeChannel(0);
   cricket::FakeTransportChannel* channel2 = client2_.GetFakeChannel(0);
@@ -531,31 +557,10 @@ TEST_F(DtlsTransportChannelTest, TestChannelSetupIce) {
   ASSERT_TRUE(channel2 != NULL);
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel1->GetIceRole());
   EXPECT_EQ(1U, channel1->IceTiebreaker());
-  EXPECT_EQ(cricket::ICEPROTO_RFC5245, channel1->protocol());
   EXPECT_EQ(kIceUfrag1, channel1->ice_ufrag());
   EXPECT_EQ(kIcePwd1, channel1->ice_pwd());
   EXPECT_EQ(cricket::ICEROLE_CONTROLLED, channel2->GetIceRole());
   EXPECT_EQ(2U, channel2->IceTiebreaker());
-  EXPECT_EQ(cricket::ICEPROTO_RFC5245, channel2->protocol());
-}
-
-// Test that transport negotiation of GICE, no DTLS works properly.
-TEST_F(DtlsTransportChannelTest, TestChannelSetupGice) {
-  client1_.SetIceProtocol(cricket::ICEPROTO_GOOGLE);
-  client2_.SetIceProtocol(cricket::ICEPROTO_GOOGLE);
-  Negotiate();
-  cricket::FakeTransportChannel* channel1 = client1_.GetFakeChannel(0);
-  cricket::FakeTransportChannel* channel2 = client2_.GetFakeChannel(0);
-  ASSERT_TRUE(channel1 != NULL);
-  ASSERT_TRUE(channel2 != NULL);
-  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel1->GetIceRole());
-  EXPECT_EQ(1U, channel1->IceTiebreaker());
-  EXPECT_EQ(cricket::ICEPROTO_GOOGLE, channel1->protocol());
-  EXPECT_EQ(kIceUfrag1, channel1->ice_ufrag());
-  EXPECT_EQ(kIcePwd1, channel1->ice_pwd());
-  EXPECT_EQ(cricket::ICEROLE_CONTROLLED, channel2->GetIceRole());
-  EXPECT_EQ(2U, channel2->IceTiebreaker());
-  EXPECT_EQ(cricket::ICEPROTO_GOOGLE, channel2->protocol());
 }
 
 // Connect without DTLS, and transfer some data.
@@ -564,6 +569,15 @@ TEST_F(DtlsTransportChannelTest, TestTransfer) {
   TestTransfer(0, 1000, 100, false);
 }
 
+// Connect without DTLS, and transfer some data.
+TEST_F(DtlsTransportChannelTest, TestOnSentPacket) {
+  ASSERT_TRUE(Connect());
+  EXPECT_EQ(client1_.sent_packet().send_time_ms, -1);
+  TestTransfer(0, 1000, 100, false);
+  EXPECT_EQ(kFakePacketId, client1_.sent_packet().packet_id);
+  EXPECT_GE(client1_.sent_packet().send_time_ms, 0);
+}
+
 // Create two channels without DTLS, and transfer some data.
 TEST_F(DtlsTransportChannelTest, TestTransferTwoChannels) {
   SetChannelCount(2);
@@ -586,19 +600,33 @@ TEST_F(DtlsTransportChannelTest, TestTransferSrtpTwoChannels) {
   TestTransfer(1, 1000, 100, true);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtls DISABLED_TestTransferDtls
+#else
+#define MAYBE_TestTransferDtls TestTransferDtls
+#endif
 // Connect with DTLS, and transfer some data.
-TEST_F(DtlsTransportChannelTest, TestTransferDtls) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtls) {
   MAYBE_SKIP_TEST(HaveDtls);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, false);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsTwoChannels DISABLED_TestTransferDtlsTwoChannels
+#else
+#define MAYBE_TestTransferDtlsTwoChannels TestTransferDtlsTwoChannels
+#endif
 // Create two channels with DTLS, and transfer some data.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsTwoChannels) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsTwoChannels) {
   MAYBE_SKIP_TEST(HaveDtls);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, false);
   TestTransfer(1, 1000, 100, false);
@@ -606,80 +634,171 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsTwoChannels) {
 
 // Connect with A doing DTLS and B not, and transfer some data.
 TEST_F(DtlsTransportChannelTest, TestTransferDtlsRejected) {
-  PrepareDtls(true, false);
+  PrepareDtls(true, false, rtc::KT_DEFAULT);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, false);
 }
 
 // Connect with B doing DTLS and A not, and transfer some data.
 TEST_F(DtlsTransportChannelTest, TestTransferDtlsNotOffered) {
-  PrepareDtls(false, true);
+  PrepareDtls(false, true, rtc::KT_DEFAULT);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, false);
 }
 
+// Create two channels with DTLS 1.0 and check ciphers.
+TEST_F(DtlsTransportChannelTest, TestDtls12None) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetChannelCount(2);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10);
+  ASSERT_TRUE(Connect());
+}
+
+// Create two channels with DTLS 1.2 and check ciphers.
+TEST_F(DtlsTransportChannelTest, TestDtls12Both) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetChannelCount(2);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12);
+  ASSERT_TRUE(Connect());
+}
+
+// Create two channels with DTLS 1.0 / DTLS 1.2 and check ciphers.
+TEST_F(DtlsTransportChannelTest, TestDtls12Client1) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetChannelCount(2);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10);
+  ASSERT_TRUE(Connect());
+}
+
+// Create two channels with DTLS 1.2 / DTLS 1.0 and check ciphers.
+TEST_F(DtlsTransportChannelTest, TestDtls12Client2) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  SetChannelCount(2);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12);
+  ASSERT_TRUE(Connect());
+}
+
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsSrtp DISABLED_TestTransferDtlsSrtp
+#else
+#define MAYBE_TestTransferDtlsSrtp TestTransferDtlsSrtp
+#endif
 // Connect with DTLS, negotiate DTLS-SRTP, and transfer SRTP using bypass.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsSrtp) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, true);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsInvalidSrtpPacket \
+  DISABLED_TestTransferDtlsInvalidSrtpPacket
+#else
+#define MAYBE_TestTransferDtlsInvalidSrtpPacket \
+  TestTransferDtlsInvalidSrtpPacket
+#endif
 // Connect with DTLS-SRTP, transfer an invalid SRTP packet, and expects -1
 // returned.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsInvalidSrtpPacket) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsInvalidSrtpPacket) {
   MAYBE_SKIP_TEST(HaveDtls);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect());
   int result = client1_.SendInvalidSrtpPacket(0, 100);
   ASSERT_EQ(-1, result);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsSrtpRejected DISABLED_TestTransferDtlsSrtpRejected
+#else
+#define MAYBE_TestTransferDtlsSrtpRejected TestTransferDtlsSrtpRejected
+#endif
 // Connect with DTLS. A does DTLS-SRTP but B does not.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpRejected) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsSrtpRejected) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, false);
   ASSERT_TRUE(Connect());
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsSrtpNotOffered \
+  DISABLED_TestTransferDtlsSrtpNotOffered
+#else
+#define MAYBE_TestTransferDtlsSrtpNotOffered TestTransferDtlsSrtpNotOffered
+#endif
 // Connect with DTLS. B does DTLS-SRTP but A does not.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpNotOffered) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsSrtpNotOffered) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(false, true);
   ASSERT_TRUE(Connect());
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsSrtpTwoChannels \
+  DISABLED_TestTransferDtlsSrtpTwoChannels
+#else
+#define MAYBE_TestTransferDtlsSrtpTwoChannels TestTransferDtlsSrtpTwoChannels
+#endif
 // Create two channels with DTLS, negotiate DTLS-SRTP, and transfer bypass SRTP.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpTwoChannels) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsSrtpTwoChannels) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, true);
   TestTransfer(1, 1000, 100, true);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsSrtpDemux DISABLED_TestTransferDtlsSrtpDemux
+#else
+#define MAYBE_TestTransferDtlsSrtpDemux TestTransferDtlsSrtpDemux
+#endif
 // Create a single channel with DTLS, and send normal data and SRTP data on it.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsSrtpDemux) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect());
   TestTransfer(0, 1000, 100, false);
   TestTransfer(0, 1000, 100, true);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestTransferDtlsAnswererIsPassive \
+  DISABLED_TestTransferDtlsAnswererIsPassive
+#else
+#define MAYBE_TestTransferDtlsAnswererIsPassive \
+  TestTransferDtlsAnswererIsPassive
+#endif
 // Testing when the remote is passive.
-TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestTransferDtlsAnswererIsPassive) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
                       cricket::CONNECTIONROLE_PASSIVE));
@@ -691,7 +810,7 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) {
 // In this case legacy is the answerer.
 TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   NegotiateWithLegacy();
   rtc::SSLRole channel1_role;
   rtc::SSLRole channel2_role;
@@ -701,12 +820,19 @@ TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) {
   EXPECT_EQ(rtc::SSL_CLIENT, channel2_role);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestDtlsReOfferFromOfferer DISABLED_TestDtlsReOfferFromOfferer
+#else
+#define MAYBE_TestDtlsReOfferFromOfferer TestDtlsReOfferFromOfferer
+#endif
 // Testing re offer/answer after the session is estbalished. Roles will be
 // kept same as of the previous negotiation.
-TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestDtlsReOfferFromOfferer) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   // Initial role for client1 is ACTPASS and client2 is ACTIVE.
   ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
@@ -720,10 +846,17 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) {
   TestTransfer(1, 1000, 100, true);
 }
 
-TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) {
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestDtlsReOfferFromAnswerer DISABLED_TestDtlsReOfferFromAnswerer
+#else
+#define MAYBE_TestDtlsReOfferFromAnswerer TestDtlsReOfferFromAnswerer
+#endif
+TEST_F(DtlsTransportChannelTest, MAYBE_TestDtlsReOfferFromAnswerer) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   // Initial role for client1 is ACTPASS and client2 is ACTIVE.
   ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
@@ -737,11 +870,18 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) {
   TestTransfer(1, 1000, 100, true);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestDtlsRoleReversal DISABLED_TestDtlsRoleReversal
+#else
+#define MAYBE_TestDtlsRoleReversal TestDtlsRoleReversal
+#endif
 // Test that any change in role after the intial setup will result in failure.
-TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestDtlsRoleReversal) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
                       cricket::CONNECTIONROLE_PASSIVE));
@@ -752,12 +892,21 @@ TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) {
               NF_REOFFER | NF_EXPECT_FAILURE);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestDtlsReOfferWithDifferentSetupAttr \
+  DISABLED_TestDtlsReOfferWithDifferentSetupAttr
+#else
+#define MAYBE_TestDtlsReOfferWithDifferentSetupAttr \
+  TestDtlsReOfferWithDifferentSetupAttr
+#endif
 // Test that using different setup attributes which results in similar ssl
 // role as the initial negotiation will result in success.
-TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestDtlsReOfferWithDifferentSetupAttr) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
                       cricket::CONNECTIONROLE_PASSIVE));
@@ -773,7 +922,7 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) {
 TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   SetChannelCount(2);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   PrepareDtlsSrtp(true, true);
   Negotiate();
 
@@ -781,7 +930,9 @@ TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
               cricket::CONNECTIONROLE_ACTIVE, NF_REOFFER);
   bool rv = client1_.Connect(&client2_);
   EXPECT_TRUE(rv);
-  EXPECT_TRUE_WAIT(client1_.writable() && client2_.writable(), 10000);
+  EXPECT_TRUE_WAIT(
+      client1_.all_channels_writable() && client2_.all_channels_writable(),
+      10000);
 
   TestTransfer(0, 1000, 100, true);
   TestTransfer(1, 1000, 100, true);
@@ -790,52 +941,59 @@ TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
 // Test Certificates state after negotiation but before connection.
 TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) {
   MAYBE_SKIP_TEST(HaveDtls);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   Negotiate();
 
-  rtc::scoped_ptr<rtc::SSLIdentity> identity1;
-  rtc::scoped_ptr<rtc::SSLIdentity> identity2;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate1;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate2;
   rtc::scoped_ptr<rtc::SSLCertificate> remote_cert1;
   rtc::scoped_ptr<rtc::SSLCertificate> remote_cert2;
 
   // After negotiation, each side has a distinct local certificate, but still no
   // remote certificate, because connection has not yet occurred.
-  ASSERT_TRUE(client1_.transport()->GetIdentity(identity1.accept()));
-  ASSERT_TRUE(client2_.transport()->GetIdentity(identity2.accept()));
-  ASSERT_NE(identity1->certificate().ToPEMString(),
-            identity2->certificate().ToPEMString());
+  ASSERT_TRUE(client1_.transport()->GetLocalCertificate(&certificate1));
+  ASSERT_TRUE(client2_.transport()->GetLocalCertificate(&certificate2));
+  ASSERT_NE(certificate1->ssl_certificate().ToPEMString(),
+            certificate2->ssl_certificate().ToPEMString());
   ASSERT_FALSE(
-      client1_.transport()->GetRemoteCertificate(remote_cert1.accept()));
+      client1_.transport()->GetRemoteSSLCertificate(remote_cert1.accept()));
   ASSERT_FALSE(remote_cert1 != NULL);
   ASSERT_FALSE(
-      client2_.transport()->GetRemoteCertificate(remote_cert2.accept()));
+      client2_.transport()->GetRemoteSSLCertificate(remote_cert2.accept()));
   ASSERT_FALSE(remote_cert2 != NULL);
 }
 
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_TestCertificatesAfterConnect DISABLED_TestCertificatesAfterConnect
+#else
+#define MAYBE_TestCertificatesAfterConnect TestCertificatesAfterConnect
+#endif
 // Test Certificates state after connection.
-TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
+TEST_F(DtlsTransportChannelTest, MAYBE_TestCertificatesAfterConnect) {
   MAYBE_SKIP_TEST(HaveDtls);
-  PrepareDtls(true, true);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
   ASSERT_TRUE(Connect());
 
-  rtc::scoped_ptr<rtc::SSLIdentity> identity1;
-  rtc::scoped_ptr<rtc::SSLIdentity> identity2;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate1;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate2;
   rtc::scoped_ptr<rtc::SSLCertificate> remote_cert1;
   rtc::scoped_ptr<rtc::SSLCertificate> remote_cert2;
 
   // After connection, each side has a distinct local certificate.
-  ASSERT_TRUE(client1_.transport()->GetIdentity(identity1.accept()));
-  ASSERT_TRUE(client2_.transport()->GetIdentity(identity2.accept()));
-  ASSERT_NE(identity1->certificate().ToPEMString(),
-            identity2->certificate().ToPEMString());
+  ASSERT_TRUE(client1_.transport()->GetLocalCertificate(&certificate1));
+  ASSERT_TRUE(client2_.transport()->GetLocalCertificate(&certificate2));
+  ASSERT_NE(certificate1->ssl_certificate().ToPEMString(),
+            certificate2->ssl_certificate().ToPEMString());
 
   // Each side's remote certificate is the other side's local certificate.
   ASSERT_TRUE(
-      client1_.transport()->GetRemoteCertificate(remote_cert1.accept()));
+      client1_.transport()->GetRemoteSSLCertificate(remote_cert1.accept()));
   ASSERT_EQ(remote_cert1->ToPEMString(),
-            identity2->certificate().ToPEMString());
+            certificate2->ssl_certificate().ToPEMString());
   ASSERT_TRUE(
-      client2_.transport()->GetRemoteCertificate(remote_cert2.accept()));
+      client2_.transport()->GetRemoteSSLCertificate(remote_cert2.accept()));
   ASSERT_EQ(remote_cert2->ToPEMString(),
-            identity1->certificate().ToPEMString());
+            certificate1->ssl_certificate().ToPEMString());
 }
diff --git a/media/webrtc/trunk/webrtc/p2p/base/fakesession.h b/media/webrtc/trunk/webrtc/p2p/base/fakesession.h
deleted file mode 100644
index 5d07d2558f..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/fakesession.h
+++ /dev/null
@@ -1,508 +0,0 @@
-/*
- *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_BASE_FAKESESSION_H_
-#define WEBRTC_P2P_BASE_FAKESESSION_H_
-
-#include <map>
-#include <string>
-#include <vector>
-
-#include "webrtc/p2p/base/session.h"
-#include "webrtc/p2p/base/transport.h"
-#include "webrtc/p2p/base/transportchannel.h"
-#include "webrtc/p2p/base/transportchannelimpl.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/fakesslidentity.h"
-#include "webrtc/base/messagequeue.h"
-#include "webrtc/base/sigslot.h"
-#include "webrtc/base/sslfingerprint.h"
-
-namespace cricket {
-
-class FakeTransport;
-
-struct PacketMessageData : public rtc::MessageData {
-  PacketMessageData(const char* data, size_t len) : packet(data, len) {
-  }
-  rtc::Buffer packet;
-};
-
-// Fake transport channel class, which can be passed to anything that needs a
-// transport channel. Can be informed of another FakeTransportChannel via
-// SetDestination.
-class FakeTransportChannel : public TransportChannelImpl,
-                             public rtc::MessageHandler {
- public:
-  explicit FakeTransportChannel(Transport* transport,
-                                const std::string& content_name,
-                                int component)
-      : TransportChannelImpl(content_name, component),
-        transport_(transport),
-        dest_(NULL),
-        state_(STATE_INIT),
-        async_(false),
-        identity_(NULL),
-        do_dtls_(false),
-        role_(ICEROLE_UNKNOWN),
-        tiebreaker_(0),
-        ice_proto_(ICEPROTO_HYBRID),
-        remote_ice_mode_(ICEMODE_FULL),
-        dtls_fingerprint_("", NULL, 0),
-        ssl_role_(rtc::SSL_CLIENT),
-        connection_count_(0) {
-  }
-  ~FakeTransportChannel() {
-    Reset();
-  }
-
-  uint64 IceTiebreaker() const { return tiebreaker_; }
-  TransportProtocol protocol() const { return ice_proto_; }
-  IceMode remote_ice_mode() const { return remote_ice_mode_; }
-  const std::string& ice_ufrag() const { return ice_ufrag_; }
-  const std::string& ice_pwd() const { return ice_pwd_; }
-  const std::string& remote_ice_ufrag() const { return remote_ice_ufrag_; }
-  const std::string& remote_ice_pwd() const { return remote_ice_pwd_; }
-  const rtc::SSLFingerprint& dtls_fingerprint() const {
-    return dtls_fingerprint_;
-  }
-
-  void SetAsync(bool async) {
-    async_ = async;
-  }
-
-  virtual Transport* GetTransport() {
-    return transport_;
-  }
-
-  virtual TransportChannelState GetState() const {
-    if (connection_count_ == 0) {
-      return TransportChannelState::STATE_FAILED;
-    }
-
-    if (connection_count_ == 1) {
-      return TransportChannelState::STATE_COMPLETED;
-    }
-
-    return TransportChannelState::STATE_FAILED;
-  }
-
-  virtual void SetIceRole(IceRole role) { role_ = role; }
-  virtual IceRole GetIceRole() const { return role_; }
-  virtual void SetIceTiebreaker(uint64 tiebreaker) { tiebreaker_ = tiebreaker; }
-  virtual bool GetIceProtocolType(IceProtocolType* type) const {
-    *type = ice_proto_;
-    return true;
-  }
-  virtual void SetIceProtocolType(IceProtocolType type) { ice_proto_ = type; }
-  virtual void SetIceCredentials(const std::string& ice_ufrag,
-                                 const std::string& ice_pwd) {
-    ice_ufrag_ = ice_ufrag;
-    ice_pwd_ = ice_pwd;
-  }
-  virtual void SetRemoteIceCredentials(const std::string& ice_ufrag,
-                                       const std::string& ice_pwd) {
-    remote_ice_ufrag_ = ice_ufrag;
-    remote_ice_pwd_ = ice_pwd;
-  }
-
-  virtual void SetRemoteIceMode(IceMode mode) { remote_ice_mode_ = mode; }
-  virtual bool SetRemoteFingerprint(const std::string& alg, const uint8* digest,
-                                    size_t digest_len) {
-    dtls_fingerprint_ = rtc::SSLFingerprint(alg, digest, digest_len);
-    return true;
-  }
-  virtual bool SetSslRole(rtc::SSLRole role) {
-    ssl_role_ = role;
-    return true;
-  }
-  virtual bool GetSslRole(rtc::SSLRole* role) const {
-    *role = ssl_role_;
-    return true;
-  }
-
-  virtual void Connect() {
-    if (state_ == STATE_INIT) {
-      state_ = STATE_CONNECTING;
-    }
-  }
-  virtual void Reset() {
-    if (state_ != STATE_INIT) {
-      state_ = STATE_INIT;
-      if (dest_) {
-        dest_->state_ = STATE_INIT;
-        dest_->dest_ = NULL;
-        dest_ = NULL;
-      }
-    }
-  }
-
-  void SetWritable(bool writable) {
-    set_writable(writable);
-  }
-
-  void SetDestination(FakeTransportChannel* dest) {
-    if (state_ == STATE_CONNECTING && dest) {
-      // This simulates the delivery of candidates.
-      dest_ = dest;
-      dest_->dest_ = this;
-      if (identity_ && dest_->identity_) {
-        do_dtls_ = true;
-        dest_->do_dtls_ = true;
-        NegotiateSrtpCiphers();
-      }
-      state_ = STATE_CONNECTED;
-      dest_->state_ = STATE_CONNECTED;
-      set_writable(true);
-      dest_->set_writable(true);
-    } else if (state_ == STATE_CONNECTED && !dest) {
-      // Simulates loss of connectivity, by asymmetrically forgetting dest_.
-      dest_ = NULL;
-      state_ = STATE_CONNECTING;
-      set_writable(false);
-    }
-  }
-
-  void SetConnectionCount(size_t connection_count) {
-    size_t old_connection_count = connection_count_;
-    connection_count_ = connection_count;
-    if (connection_count_ < old_connection_count)
-      SignalConnectionRemoved(this);
-  }
-
-  virtual int SendPacket(const char* data, size_t len,
-                         const rtc::PacketOptions& options, int flags) {
-    if (state_ != STATE_CONNECTED) {
-      return -1;
-    }
-
-    if (flags != PF_SRTP_BYPASS && flags != 0) {
-      return -1;
-    }
-
-    PacketMessageData* packet = new PacketMessageData(data, len);
-    if (async_) {
-      rtc::Thread::Current()->Post(this, 0, packet);
-    } else {
-      rtc::Thread::Current()->Send(this, 0, packet);
-    }
-    return static_cast<int>(len);
-  }
-  virtual int SetOption(rtc::Socket::Option opt, int value) {
-    return true;
-  }
-  virtual bool GetOption(rtc::Socket::Option opt, int* value) {
-    return true;
-  }
-  virtual int GetError() {
-    return 0;
-  }
-
-  virtual void OnSignalingReady() {
-  }
-  virtual void OnCandidate(const Candidate& candidate) {
-  }
-
-  virtual void OnMessage(rtc::Message* msg) {
-    PacketMessageData* data = static_cast<PacketMessageData*>(
-        msg->pdata);
-    dest_->SignalReadPacket(dest_, data->packet.data(), data->packet.size(),
-                            rtc::CreatePacketTime(0), 0);
-    delete data;
-  }
-
-  bool SetLocalIdentity(rtc::SSLIdentity* identity) {
-    identity_ = identity;
-    return true;
-  }
-
-
-  void SetRemoteCertificate(rtc::FakeSSLCertificate* cert) {
-    remote_cert_ = cert;
-  }
-
-  virtual bool IsDtlsActive() const {
-    return do_dtls_;
-  }
-
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
-    srtp_ciphers_ = ciphers;
-    return true;
-  }
-
-  virtual bool GetSrtpCipher(std::string* cipher) {
-    if (!chosen_srtp_cipher_.empty()) {
-      *cipher = chosen_srtp_cipher_;
-      return true;
-    }
-    return false;
-  }
-
-  virtual bool GetSslCipher(std::string* cipher) {
-    return false;
-  }
-
-  virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const {
-    if (!identity_)
-      return false;
-
-    *identity = identity_->GetReference();
-    return true;
-  }
-
-  virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const {
-    if (!remote_cert_)
-      return false;
-
-    *cert = remote_cert_->GetReference();
-    return true;
-  }
-
-  virtual bool ExportKeyingMaterial(const std::string& label,
-                                    const uint8* context,
-                                    size_t context_len,
-                                    bool use_context,
-                                    uint8* result,
-                                    size_t result_len) {
-    if (!chosen_srtp_cipher_.empty()) {
-      memset(result, 0xff, result_len);
-      return true;
-    }
-
-    return false;
-  }
-
-  virtual void NegotiateSrtpCiphers() {
-    for (std::vector<std::string>::const_iterator it1 = srtp_ciphers_.begin();
-        it1 != srtp_ciphers_.end(); ++it1) {
-      for (std::vector<std::string>::const_iterator it2 =
-              dest_->srtp_ciphers_.begin();
-          it2 != dest_->srtp_ciphers_.end(); ++it2) {
-        if (*it1 == *it2) {
-          chosen_srtp_cipher_ = *it1;
-          dest_->chosen_srtp_cipher_ = *it2;
-          return;
-        }
-      }
-    }
-  }
-
-  bool GetStats(ConnectionInfos* infos) override {
-    ConnectionInfo info;
-    infos->clear();
-    infos->push_back(info);
-    return true;
-  }
-
- private:
-  enum State { STATE_INIT, STATE_CONNECTING, STATE_CONNECTED };
-  Transport* transport_;
-  FakeTransportChannel* dest_;
-  State state_;
-  bool async_;
-  rtc::SSLIdentity* identity_;
-  rtc::FakeSSLCertificate* remote_cert_;
-  bool do_dtls_;
-  std::vector<std::string> srtp_ciphers_;
-  std::string chosen_srtp_cipher_;
-  IceRole role_;
-  uint64 tiebreaker_;
-  IceProtocolType ice_proto_;
-  std::string ice_ufrag_;
-  std::string ice_pwd_;
-  std::string remote_ice_ufrag_;
-  std::string remote_ice_pwd_;
-  IceMode remote_ice_mode_;
-  rtc::SSLFingerprint dtls_fingerprint_;
-  rtc::SSLRole ssl_role_;
-  size_t connection_count_;
-};
-
-// Fake transport class, which can be passed to anything that needs a Transport.
-// Can be informed of another FakeTransport via SetDestination (low-tech way
-// of doing candidates)
-class FakeTransport : public Transport {
- public:
-  typedef std::map<int, FakeTransportChannel*> ChannelMap;
-  FakeTransport(rtc::Thread* signaling_thread,
-                rtc::Thread* worker_thread,
-                const std::string& content_name,
-                PortAllocator* alllocator = NULL)
-      : Transport(signaling_thread, worker_thread,
-                  content_name, "test_type", NULL),
-      dest_(NULL),
-      async_(false),
-      identity_(NULL) {
-  }
-  ~FakeTransport() {
-    DestroyAllChannels();
-  }
-
-  const ChannelMap& channels() const { return channels_; }
-
-  void SetAsync(bool async) { async_ = async; }
-  void SetDestination(FakeTransport* dest) {
-    dest_ = dest;
-    for (ChannelMap::iterator it = channels_.begin(); it != channels_.end();
-         ++it) {
-      it->second->SetLocalIdentity(identity_);
-      SetChannelDestination(it->first, it->second);
-    }
-  }
-
-  void SetWritable(bool writable) {
-    for (ChannelMap::iterator it = channels_.begin(); it != channels_.end();
-         ++it) {
-      it->second->SetWritable(writable);
-    }
-  }
-
-  void set_identity(rtc::SSLIdentity* identity) {
-    identity_ = identity;
-  }
-
-  using Transport::local_description;
-  using Transport::remote_description;
-
- protected:
-  virtual TransportChannelImpl* CreateTransportChannel(int component) {
-    if (channels_.find(component) != channels_.end()) {
-      return NULL;
-    }
-    FakeTransportChannel* channel =
-        new FakeTransportChannel(this, content_name(), component);
-    channel->SetAsync(async_);
-    SetChannelDestination(component, channel);
-    channels_[component] = channel;
-    return channel;
-  }
-  virtual void DestroyTransportChannel(TransportChannelImpl* channel) {
-    channels_.erase(channel->component());
-    delete channel;
-  }
-  virtual void SetIdentity_w(rtc::SSLIdentity* identity) {
-    identity_ = identity;
-  }
-  virtual bool GetIdentity_w(rtc::SSLIdentity** identity) {
-    if (!identity_)
-      return false;
-
-    *identity = identity_->GetReference();
-    return true;
-  }
-
- private:
-  FakeTransportChannel* GetFakeChannel(int component) {
-    ChannelMap::iterator it = channels_.find(component);
-    return (it != channels_.end()) ? it->second : NULL;
-  }
-  void SetChannelDestination(int component,
-                             FakeTransportChannel* channel) {
-    FakeTransportChannel* dest_channel = NULL;
-    if (dest_) {
-      dest_channel = dest_->GetFakeChannel(component);
-      if (dest_channel) {
-        dest_channel->SetLocalIdentity(dest_->identity_);
-      }
-    }
-    channel->SetDestination(dest_channel);
-  }
-
-  // Note, this is distinct from the Channel map owned by Transport.
-  // This map just tracks the FakeTransportChannels created by this class.
-  ChannelMap channels_;
-  FakeTransport* dest_;
-  bool async_;
-  rtc::SSLIdentity* identity_;
-};
-
-// Fake session class, which can be passed into a BaseChannel object for
-// test purposes. Can be connected to other FakeSessions via Connect().
-class FakeSession : public BaseSession {
- public:
-  explicit FakeSession()
-      : BaseSession(rtc::Thread::Current(),
-                    rtc::Thread::Current(),
-                    NULL, "", "", true),
-      fail_create_channel_(false) {
-  }
-  explicit FakeSession(bool initiator)
-      : BaseSession(rtc::Thread::Current(),
-                    rtc::Thread::Current(),
-                    NULL, "", "", initiator),
-      fail_create_channel_(false) {
-  }
-  FakeSession(rtc::Thread* worker_thread, bool initiator)
-      : BaseSession(rtc::Thread::Current(),
-                    worker_thread,
-                    NULL, "", "", initiator),
-      fail_create_channel_(false) {
-  }
-
-  FakeTransport* GetTransport(const std::string& content_name) {
-    return static_cast<FakeTransport*>(
-        BaseSession::GetTransport(content_name));
-  }
-
-  void Connect(FakeSession* dest) {
-    // Simulate the exchange of candidates.
-    CompleteNegotiation();
-    dest->CompleteNegotiation();
-    for (TransportMap::const_iterator it = transport_proxies().begin();
-        it != transport_proxies().end(); ++it) {
-      static_cast<FakeTransport*>(it->second->impl())->SetDestination(
-          dest->GetTransport(it->first));
-    }
-  }
-
-  virtual TransportChannel* CreateChannel(
-      const std::string& content_name,
-      int component) {
-    if (fail_create_channel_) {
-      return NULL;
-    }
-    return BaseSession::CreateChannel(content_name, component);
-  }
-
-  void set_fail_channel_creation(bool fail_channel_creation) {
-    fail_create_channel_ = fail_channel_creation;
-  }
-
-  // TODO: Hoist this into Session when we re-work the Session code.
-  void set_ssl_identity(rtc::SSLIdentity* identity) {
-    for (TransportMap::const_iterator it = transport_proxies().begin();
-        it != transport_proxies().end(); ++it) {
-      // We know that we have a FakeTransport*
-
-      static_cast<FakeTransport*>(it->second->impl())->set_identity
-          (identity);
-    }
-  }
-
- protected:
-  virtual Transport* CreateTransport(const std::string& content_name) {
-    return new FakeTransport(signaling_thread(), worker_thread(), content_name);
-  }
-
-  void CompleteNegotiation() {
-    for (TransportMap::const_iterator it = transport_proxies().begin();
-        it != transport_proxies().end(); ++it) {
-      it->second->CompleteNegotiation();
-      it->second->ConnectChannels();
-    }
-  }
-
- private:
-  bool fail_create_channel_;
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_P2P_BASE_FAKESESSION_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/base/faketransportcontroller.h b/media/webrtc/trunk/webrtc/p2p/base/faketransportcontroller.h
new file mode 100644
index 0000000000..65c59be98d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/base/faketransportcontroller.h
@@ -0,0 +1,543 @@
+/*
+ *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_P2P_BASE_FAKETRANSPORTCONTROLLER_H_
+#define WEBRTC_P2P_BASE_FAKETRANSPORTCONTROLLER_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
+#include "webrtc/base/bind.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/fakesslidentity.h"
+#include "webrtc/base/messagequeue.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/sslfingerprint.h"
+#include "webrtc/base/thread.h"
+
+namespace cricket {
+
+class FakeTransport;
+
+namespace {
+struct PacketMessageData : public rtc::MessageData {
+  PacketMessageData(const char* data, size_t len) : packet(data, len) {}
+  rtc::Buffer packet;
+};
+}  // namespace
+
+// Fake transport channel class, which can be passed to anything that needs a
+// transport channel. Can be informed of another FakeTransportChannel via
+// SetDestination.
+// TODO(hbos): Move implementation to .cc file, this and other classes in file.
+class FakeTransportChannel : public TransportChannelImpl,
+                             public rtc::MessageHandler {
+ public:
+  explicit FakeTransportChannel(Transport* transport,
+                                const std::string& name,
+                                int component)
+      : TransportChannelImpl(name, component),
+        transport_(transport),
+        dtls_fingerprint_("", nullptr, 0) {}
+  ~FakeTransportChannel() { Reset(); }
+
+  uint64_t IceTiebreaker() const { return tiebreaker_; }
+  IceMode remote_ice_mode() const { return remote_ice_mode_; }
+  const std::string& ice_ufrag() const { return ice_ufrag_; }
+  const std::string& ice_pwd() const { return ice_pwd_; }
+  const std::string& remote_ice_ufrag() const { return remote_ice_ufrag_; }
+  const std::string& remote_ice_pwd() const { return remote_ice_pwd_; }
+  const rtc::SSLFingerprint& dtls_fingerprint() const {
+    return dtls_fingerprint_;
+  }
+
+  // If async, will send packets by "Post"-ing to message queue instead of
+  // synchronously "Send"-ing.
+  void SetAsync(bool async) { async_ = async; }
+
+  Transport* GetTransport() override { return transport_; }
+
+  TransportChannelState GetState() const override {
+    if (connection_count_ == 0) {
+      return had_connection_ ? TransportChannelState::STATE_FAILED
+                             : TransportChannelState::STATE_INIT;
+    }
+
+    if (connection_count_ == 1) {
+      return TransportChannelState::STATE_COMPLETED;
+    }
+
+    return TransportChannelState::STATE_CONNECTING;
+  }
+
+  void SetIceRole(IceRole role) override { role_ = role; }
+  IceRole GetIceRole() const override { return role_; }
+  void SetIceTiebreaker(uint64_t tiebreaker) override {
+    tiebreaker_ = tiebreaker;
+  }
+  void SetIceCredentials(const std::string& ice_ufrag,
+                         const std::string& ice_pwd) override {
+    ice_ufrag_ = ice_ufrag;
+    ice_pwd_ = ice_pwd;
+  }
+  void SetRemoteIceCredentials(const std::string& ice_ufrag,
+                               const std::string& ice_pwd) override {
+    remote_ice_ufrag_ = ice_ufrag;
+    remote_ice_pwd_ = ice_pwd;
+  }
+
+  void SetRemoteIceMode(IceMode mode) override { remote_ice_mode_ = mode; }
+  bool SetRemoteFingerprint(const std::string& alg,
+                            const uint8_t* digest,
+                            size_t digest_len) override {
+    dtls_fingerprint_ = rtc::SSLFingerprint(alg, digest, digest_len);
+    return true;
+  }
+  bool SetSslRole(rtc::SSLRole role) override {
+    ssl_role_ = role;
+    return true;
+  }
+  bool GetSslRole(rtc::SSLRole* role) const override {
+    *role = ssl_role_;
+    return true;
+  }
+
+  void Connect() override {
+    if (state_ == STATE_INIT) {
+      state_ = STATE_CONNECTING;
+    }
+  }
+
+  void MaybeStartGathering() override {
+    if (gathering_state_ == kIceGatheringNew) {
+      gathering_state_ = kIceGatheringGathering;
+      SignalGatheringState(this);
+    }
+  }
+
+  IceGatheringState gathering_state() const override {
+    return gathering_state_;
+  }
+
+  void Reset() {
+    if (state_ != STATE_INIT) {
+      state_ = STATE_INIT;
+      if (dest_) {
+        dest_->state_ = STATE_INIT;
+        dest_->dest_ = nullptr;
+        dest_ = nullptr;
+      }
+    }
+  }
+
+  void SetWritable(bool writable) { set_writable(writable); }
+
+  void SetDestination(FakeTransportChannel* dest) {
+    if (state_ == STATE_CONNECTING && dest) {
+      // This simulates the delivery of candidates.
+      dest_ = dest;
+      dest_->dest_ = this;
+      if (local_cert_ && dest_->local_cert_) {
+        do_dtls_ = true;
+        dest_->do_dtls_ = true;
+        NegotiateSrtpCiphers();
+      }
+      state_ = STATE_CONNECTED;
+      dest_->state_ = STATE_CONNECTED;
+      set_writable(true);
+      dest_->set_writable(true);
+    } else if (state_ == STATE_CONNECTED && !dest) {
+      // Simulates loss of connectivity, by asymmetrically forgetting dest_.
+      dest_ = nullptr;
+      state_ = STATE_CONNECTING;
+      set_writable(false);
+    }
+  }
+
+  void SetConnectionCount(size_t connection_count) {
+    size_t old_connection_count = connection_count_;
+    connection_count_ = connection_count;
+    if (connection_count)
+      had_connection_ = true;
+    if (connection_count_ < old_connection_count)
+      SignalConnectionRemoved(this);
+  }
+
+  void SetCandidatesGatheringComplete() {
+    if (gathering_state_ != kIceGatheringComplete) {
+      gathering_state_ = kIceGatheringComplete;
+      SignalGatheringState(this);
+    }
+  }
+
+  void SetReceiving(bool receiving) { set_receiving(receiving); }
+
+  void SetIceConfig(const IceConfig& config) override {
+    receiving_timeout_ = config.receiving_timeout_ms;
+    gather_continually_ = config.gather_continually;
+  }
+
+  int receiving_timeout() const { return receiving_timeout_; }
+  bool gather_continually() const { return gather_continually_; }
+
+  int SendPacket(const char* data,
+                 size_t len,
+                 const rtc::PacketOptions& options,
+                 int flags) override {
+    if (state_ != STATE_CONNECTED) {
+      return -1;
+    }
+
+    if (flags != PF_SRTP_BYPASS && flags != 0) {
+      return -1;
+    }
+
+    PacketMessageData* packet = new PacketMessageData(data, len);
+    if (async_) {
+      rtc::Thread::Current()->Post(this, 0, packet);
+    } else {
+      rtc::Thread::Current()->Send(this, 0, packet);
+    }
+    rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+    SignalSentPacket(this, sent_packet);
+    return static_cast<int>(len);
+  }
+  int SetOption(rtc::Socket::Option opt, int value) override { return true; }
+  bool GetOption(rtc::Socket::Option opt, int* value) override { return true; }
+  int GetError() override { return 0; }
+
+  void AddRemoteCandidate(const Candidate& candidate) override {
+    remote_candidates_.push_back(candidate);
+  }
+  const Candidates& remote_candidates() const { return remote_candidates_; }
+
+  void OnMessage(rtc::Message* msg) override {
+    PacketMessageData* data = static_cast<PacketMessageData*>(msg->pdata);
+    dest_->SignalReadPacket(dest_, data->packet.data<char>(),
+                            data->packet.size(), rtc::CreatePacketTime(0), 0);
+    delete data;
+  }
+
+  bool SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+    local_cert_ = certificate;
+    return true;
+  }
+
+  void SetRemoteSSLCertificate(rtc::FakeSSLCertificate* cert) {
+    remote_cert_ = cert;
+  }
+
+  bool IsDtlsActive() const override { return do_dtls_; }
+
+  bool SetSrtpCryptoSuites(const std::vector<int>& ciphers) override {
+    srtp_ciphers_ = ciphers;
+    return true;
+  }
+
+  bool GetSrtpCryptoSuite(int* crypto_suite) override {
+    if (chosen_crypto_suite_ != rtc::SRTP_INVALID_CRYPTO_SUITE) {
+      *crypto_suite = chosen_crypto_suite_;
+      return true;
+    }
+    return false;
+  }
+
+  bool GetSslCipherSuite(int* cipher_suite) override { return false; }
+
+  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const {
+    return local_cert_;
+  }
+
+  bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert) const override {
+    if (!remote_cert_)
+      return false;
+
+    *cert = remote_cert_->GetReference();
+    return true;
+  }
+
+  bool ExportKeyingMaterial(const std::string& label,
+                            const uint8_t* context,
+                            size_t context_len,
+                            bool use_context,
+                            uint8_t* result,
+                            size_t result_len) override {
+    if (chosen_crypto_suite_ != rtc::SRTP_INVALID_CRYPTO_SUITE) {
+      memset(result, 0xff, result_len);
+      return true;
+    }
+
+    return false;
+  }
+
+  void NegotiateSrtpCiphers() {
+    for (std::vector<int>::const_iterator it1 = srtp_ciphers_.begin();
+         it1 != srtp_ciphers_.end(); ++it1) {
+      for (std::vector<int>::const_iterator it2 = dest_->srtp_ciphers_.begin();
+           it2 != dest_->srtp_ciphers_.end(); ++it2) {
+        if (*it1 == *it2) {
+          chosen_crypto_suite_ = *it1;
+          dest_->chosen_crypto_suite_ = *it2;
+          return;
+        }
+      }
+    }
+  }
+
+  bool GetStats(ConnectionInfos* infos) override {
+    ConnectionInfo info;
+    infos->clear();
+    infos->push_back(info);
+    return true;
+  }
+
+  void set_ssl_max_protocol_version(rtc::SSLProtocolVersion version) {
+    ssl_max_version_ = version;
+  }
+  rtc::SSLProtocolVersion ssl_max_protocol_version() const {
+    return ssl_max_version_;
+  }
+
+ private:
+  enum State { STATE_INIT, STATE_CONNECTING, STATE_CONNECTED };
+  Transport* transport_;
+  FakeTransportChannel* dest_ = nullptr;
+  State state_ = STATE_INIT;
+  bool async_ = false;
+  Candidates remote_candidates_;
+  rtc::scoped_refptr<rtc::RTCCertificate> local_cert_;
+  rtc::FakeSSLCertificate* remote_cert_ = nullptr;
+  bool do_dtls_ = false;
+  std::vector<int> srtp_ciphers_;
+  int chosen_crypto_suite_ = rtc::SRTP_INVALID_CRYPTO_SUITE;
+  int receiving_timeout_ = -1;
+  bool gather_continually_ = false;
+  IceRole role_ = ICEROLE_UNKNOWN;
+  uint64_t tiebreaker_ = 0;
+  std::string ice_ufrag_;
+  std::string ice_pwd_;
+  std::string remote_ice_ufrag_;
+  std::string remote_ice_pwd_;
+  IceMode remote_ice_mode_ = ICEMODE_FULL;
+  rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
+  rtc::SSLFingerprint dtls_fingerprint_;
+  rtc::SSLRole ssl_role_ = rtc::SSL_CLIENT;
+  size_t connection_count_ = 0;
+  IceGatheringState gathering_state_ = kIceGatheringNew;
+  bool had_connection_ = false;
+};
+
+// Fake transport class, which can be passed to anything that needs a Transport.
+// Can be informed of another FakeTransport via SetDestination (low-tech way
+// of doing candidates)
+class FakeTransport : public Transport {
+ public:
+  typedef std::map<int, FakeTransportChannel*> ChannelMap;
+
+  explicit FakeTransport(const std::string& name) : Transport(name, nullptr) {}
+
+  // Note that we only have a constructor with the allocator parameter so it can
+  // be wrapped by a DtlsTransport.
+  FakeTransport(const std::string& name, PortAllocator* allocator)
+      : Transport(name, nullptr) {}
+
+  ~FakeTransport() { DestroyAllChannels(); }
+
+  const ChannelMap& channels() const { return channels_; }
+
+  // If async, will send packets by "Post"-ing to message queue instead of
+  // synchronously "Send"-ing.
+  void SetAsync(bool async) { async_ = async; }
+  void SetDestination(FakeTransport* dest) {
+    dest_ = dest;
+    for (const auto& kv : channels_) {
+      kv.second->SetLocalCertificate(certificate_);
+      SetChannelDestination(kv.first, kv.second);
+    }
+  }
+
+  void SetWritable(bool writable) {
+    for (const auto& kv : channels_) {
+      kv.second->SetWritable(writable);
+    }
+  }
+
+  void SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override {
+    certificate_ = certificate;
+  }
+  bool GetLocalCertificate(
+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override {
+    if (!certificate_)
+      return false;
+
+    *certificate = certificate_;
+    return true;
+  }
+
+  bool GetSslRole(rtc::SSLRole* role) const override {
+    if (channels_.empty()) {
+      return false;
+    }
+    return channels_.begin()->second->GetSslRole(role);
+  }
+
+  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override {
+    ssl_max_version_ = version;
+    for (const auto& kv : channels_) {
+      kv.second->set_ssl_max_protocol_version(ssl_max_version_);
+    }
+    return true;
+  }
+  rtc::SSLProtocolVersion ssl_max_protocol_version() const {
+    return ssl_max_version_;
+  }
+
+  using Transport::local_description;
+  using Transport::remote_description;
+
+ protected:
+  TransportChannelImpl* CreateTransportChannel(int component) override {
+    if (channels_.find(component) != channels_.end()) {
+      return nullptr;
+    }
+    FakeTransportChannel* channel =
+        new FakeTransportChannel(this, name(), component);
+    channel->set_ssl_max_protocol_version(ssl_max_version_);
+    channel->SetAsync(async_);
+    SetChannelDestination(component, channel);
+    channels_[component] = channel;
+    return channel;
+  }
+
+  void DestroyTransportChannel(TransportChannelImpl* channel) override {
+    channels_.erase(channel->component());
+    delete channel;
+  }
+
+ private:
+  FakeTransportChannel* GetFakeChannel(int component) {
+    auto it = channels_.find(component);
+    return (it != channels_.end()) ? it->second : nullptr;
+  }
+
+  void SetChannelDestination(int component, FakeTransportChannel* channel) {
+    FakeTransportChannel* dest_channel = nullptr;
+    if (dest_) {
+      dest_channel = dest_->GetFakeChannel(component);
+      if (dest_channel) {
+        dest_channel->SetLocalCertificate(dest_->certificate_);
+      }
+    }
+    channel->SetDestination(dest_channel);
+  }
+
+  // Note, this is distinct from the Channel map owned by Transport.
+  // This map just tracks the FakeTransportChannels created by this class.
+  // It's mainly needed so that we can access a FakeTransportChannel directly,
+  // even if wrapped by a DtlsTransportChannelWrapper.
+  ChannelMap channels_;
+  FakeTransport* dest_ = nullptr;
+  bool async_ = false;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
+  rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
+};
+
+// Fake TransportController class, which can be passed into a BaseChannel object
+// for test purposes. Can be connected to other FakeTransportControllers via
+// Connect().
+//
+// This fake is unusual in that for the most part, it's implemented with the
+// real TransportController code, but with fake TransportChannels underneath.
+class FakeTransportController : public TransportController {
+ public:
+  FakeTransportController()
+      : TransportController(rtc::Thread::Current(),
+                            rtc::Thread::Current(),
+                            nullptr),
+        fail_create_channel_(false) {}
+
+  explicit FakeTransportController(IceRole role)
+      : TransportController(rtc::Thread::Current(),
+                            rtc::Thread::Current(),
+                            nullptr),
+        fail_create_channel_(false) {
+    SetIceRole(role);
+  }
+
+  explicit FakeTransportController(rtc::Thread* worker_thread)
+      : TransportController(rtc::Thread::Current(), worker_thread, nullptr),
+        fail_create_channel_(false) {}
+
+  FakeTransportController(rtc::Thread* worker_thread, IceRole role)
+      : TransportController(rtc::Thread::Current(), worker_thread, nullptr),
+        fail_create_channel_(false) {
+    SetIceRole(role);
+  }
+
+  FakeTransport* GetTransport_w(const std::string& transport_name) {
+    return static_cast<FakeTransport*>(
+        TransportController::GetTransport_w(transport_name));
+  }
+
+  void Connect(FakeTransportController* dest) {
+    worker_thread()->Invoke<void>(
+        rtc::Bind(&FakeTransportController::Connect_w, this, dest));
+  }
+
+  TransportChannel* CreateTransportChannel_w(const std::string& transport_name,
+                                             int component) override {
+    if (fail_create_channel_) {
+      return nullptr;
+    }
+    return TransportController::CreateTransportChannel_w(transport_name,
+                                                         component);
+  }
+
+  void set_fail_channel_creation(bool fail_channel_creation) {
+    fail_create_channel_ = fail_channel_creation;
+  }
+
+ protected:
+  Transport* CreateTransport_w(const std::string& transport_name) override {
+    return new FakeTransport(transport_name);
+  }
+
+  void Connect_w(FakeTransportController* dest) {
+    // Simulate the exchange of candidates.
+    ConnectChannels_w();
+    dest->ConnectChannels_w();
+    for (auto& kv : transports()) {
+      FakeTransport* transport = static_cast<FakeTransport*>(kv.second);
+      transport->SetDestination(dest->GetTransport_w(kv.first));
+    }
+  }
+
+  void ConnectChannels_w() {
+    for (auto& kv : transports()) {
+      FakeTransport* transport = static_cast<FakeTransport*>(kv.second);
+      transport->ConnectChannels();
+      transport->MaybeStartGathering();
+    }
+  }
+
+ private:
+  bool fail_create_channel_;
+};
+
+}  // namespace cricket
+
+#endif  // WEBRTC_P2P_BASE_FAKETRANSPORTCONTROLLER_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.cc b/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.cc
index 89586f9f3d..abc4c14504 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.cc
@@ -20,21 +20,15 @@
 
 namespace cricket {
 
-P2PTransport::P2PTransport(rtc::Thread* signaling_thread,
-                           rtc::Thread* worker_thread,
-                           const std::string& content_name,
-                           PortAllocator* allocator)
-    : Transport(signaling_thread, worker_thread,
-                content_name, NS_GINGLE_P2P, allocator) {
-}
+P2PTransport::P2PTransport(const std::string& name, PortAllocator* allocator)
+    : Transport(name, allocator) {}
 
 P2PTransport::~P2PTransport() {
   DestroyAllChannels();
 }
 
 TransportChannelImpl* P2PTransport::CreateTransportChannel(int component) {
-  return new P2PTransportChannel(content_name(), component, this,
-                                 port_allocator());
+  return new P2PTransportChannel(name(), component, this, port_allocator());
 }
 
 void P2PTransport::DestroyTransportChannel(TransportChannelImpl* channel) {
diff --git a/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.h b/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.h
index c0e3952d80..0f965b4cdc 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/p2ptransport.h
@@ -16,12 +16,10 @@
 
 namespace cricket {
 
+// Everything in this class should be called on the worker thread.
 class P2PTransport : public Transport {
  public:
-  P2PTransport(rtc::Thread* signaling_thread,
-               rtc::Thread* worker_thread,
-               const std::string& content_name,
-               PortAllocator* allocator);
+  P2PTransport(const std::string& name, PortAllocator* allocator);
   virtual ~P2PTransport();
 
  protected:
@@ -31,7 +29,7 @@ class P2PTransport : public Transport {
 
   friend class P2PTransportChannel;
 
-  DISALLOW_EVIL_CONSTRUCTORS(P2PTransport);
+  RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransport);
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.cc b/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.cc
index 031a64ce0b..952cfab747 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/p2p/base/p2ptransportchannel.h"
 
+#include <algorithm>
 #include <set>
 #include "webrtc/p2p/base/common.h"
 #include "webrtc/p2p/base/relayport.h"  // For RELAY_PORT_TYPE.
@@ -18,27 +19,12 @@
 #include "webrtc/base/crc32.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 
 // messages for queuing up work for ourselves
-enum {
-  MSG_SORT = 1,
-  MSG_PING,
-};
-
-// When the socket is unwritable, we will use 10 Kbps (ignoring IP+UDP headers)
-// for pinging.  When the socket is writable, we will use only 1 Kbps because
-// we don't want to degrade the quality on a modem.  These numbers should work
-// well on a 28.8K modem, which is the slowest connection on which the voice
-// quality is reasonable at all.
-static const uint32 PING_PACKET_SIZE = 60 * 8;
-static const uint32 WRITABLE_DELAY = 1000 * PING_PACKET_SIZE / 1000;  // 480ms
-static const uint32 UNWRITABLE_DELAY = 1000 * PING_PACKET_SIZE / 10000;  // 50ms
-
-// If there is a current writable connection, then we will also try hard to
-// make sure it is pinged at this rate.
-static const uint32 MAX_CURRENT_WRITABLE_DELAY = 900;  // 2*WRITABLE_DELAY - bit
+enum { MSG_SORT = 1, MSG_CHECK_AND_PING };
 
 // The minimum improvement in RTT that justifies a switch.
 static const double kMinImprovement = 10;
@@ -67,14 +53,62 @@ int CompareConnectionCandidates(cricket::Connection* a,
          (b->remote_candidate().generation() + b->port()->generation());
 }
 
-// Compare two connections based on their writability and static preferences.
-int CompareConnections(cricket::Connection *a, cricket::Connection *b) {
+// Compare two connections based on their writing, receiving, and connected
+// states.
+int CompareConnectionStates(cricket::Connection* a, cricket::Connection* b) {
   // Sort based on write-state.  Better states have lower values.
   if (a->write_state() < b->write_state())
     return 1;
   if (a->write_state() > b->write_state())
     return -1;
 
+  // We prefer a receiving connection to a non-receiving, higher-priority
+  // connection when sorting connections and choosing which connection to
+  // switch to.
+  if (a->receiving() && !b->receiving())
+    return 1;
+  if (!a->receiving() && b->receiving())
+    return -1;
+
+  // WARNING: Some complexity here about TCP reconnecting.
+  // When a TCP connection fails because of a TCP socket disconnecting, the
+  // active side of the connection will attempt to reconnect for 5 seconds while
+  // pretending to be writable (the connection is not set to the unwritable
+  // state).  On the passive side, the connection also remains writable even
+  // though it is disconnected, and a new connection is created when the active
+  // side connects.  At that point, there are two TCP connections on the passive
+  // side: 1. the old, disconnected one that is pretending to be writable, and
+  // 2.  the new, connected one that is maybe not yet writable.  For purposes of
+  // pruning, pinging, and selecting the best connection, we want to treat the
+  // new connection as "better" than the old one.  We could add a method called
+  // something like Connection::ImReallyBadEvenThoughImWritable, but that is
+  // equivalent to the existing Connection::connected(), which we already have.
+  // So, in code throughout this file, we'll check whether the connection is
+  // connected() or not, and if it is not, treat it as "worse" than a connected
+  // one, even though it's writable.  In the code below, we're doing so to make
+  // sure we treat a new writable connection as better than an old disconnected
+  // connection.
+
+  // In the case where we reconnect TCP connections, the original best
+  // connection is disconnected without changing to WRITE_TIMEOUT. In this case,
+  // the new connection, when it becomes writable, should have higher priority.
+  if (a->write_state() == cricket::Connection::STATE_WRITABLE &&
+      b->write_state() == cricket::Connection::STATE_WRITABLE) {
+    if (a->connected() && !b->connected()) {
+      return 1;
+    }
+    if (!a->connected() && b->connected()) {
+      return -1;
+    }
+  }
+  return 0;
+}
+
+int CompareConnections(cricket::Connection* a, cricket::Connection* b) {
+  int state_cmp = CompareConnectionStates(a, b);
+  if (state_cmp != 0) {
+    return state_cmp;
+  }
   // Compare the candidate information.
   return CompareConnectionCandidates(a, b);
 }
@@ -88,14 +122,6 @@ class ConnectionCompare {
     cricket::Connection* a = const_cast<cricket::Connection*>(ca);
     cricket::Connection* b = const_cast<cricket::Connection*>(cb);
 
-    // The IceProtocol is initialized to ICEPROTO_HYBRID and can be updated to
-    // GICE or RFC5245 when an answer SDP is set, or when a STUN message is
-    // received. So the port receiving the STUN message may have a different
-    // IceProtocol if the answer SDP is not set yet.
-    ASSERT(a->port()->IceProtocol() == b->port()->IceProtocol() ||
-           a->port()->IceProtocol() == cricket::ICEPROTO_HYBRID ||
-           b->port()->IceProtocol() == cricket::ICEPROTO_HYBRID);
-
     // Compare first on writability and static preferences.
     int cmp = CompareConnections(a, b);
     if (cmp > 0)
@@ -120,19 +146,33 @@ class ConnectionCompare {
 };
 
 // Determines whether we should switch between two connections, based first on
-// static preferences and then (if those are equal) on latency estimates.
-bool ShouldSwitch(cricket::Connection* a_conn, cricket::Connection* b_conn) {
+// connection states, static preferences, and then (if those are equal) on
+// latency estimates.
+bool ShouldSwitch(cricket::Connection* a_conn,
+                  cricket::Connection* b_conn,
+                  cricket::IceRole ice_role) {
   if (a_conn == b_conn)
     return false;
 
   if (!a_conn || !b_conn)  // don't think the latter should happen
     return true;
 
-  int prefs_cmp = CompareConnections(a_conn, b_conn);
-  if (prefs_cmp < 0)
-    return true;
-  if (prefs_cmp > 0)
+  // We prefer to switch to a writable and receiving connection over a
+  // non-writable or non-receiving connection, even if the latter has
+  // been nominated by the controlling side.
+  int state_cmp = CompareConnectionStates(a_conn, b_conn);
+  if (state_cmp != 0) {
+    return state_cmp < 0;
+  }
+  if (ice_role == cricket::ICEROLE_CONTROLLED && a_conn->nominated()) {
+    LOG(LS_VERBOSE) << "Controlled side did not switch due to nominated status";
     return false;
+  }
+
+  int prefs_cmp = CompareConnectionCandidates(a_conn, b_conn);
+  if (prefs_cmp != 0) {
+    return prefs_cmp < 0;
+  }
 
   return b_conn->rtt() <= a_conn->rtt() + kMinImprovement;
 }
@@ -141,39 +181,68 @@ bool ShouldSwitch(cricket::Connection* a_conn, cricket::Connection* b_conn) {
 
 namespace cricket {
 
-P2PTransportChannel::P2PTransportChannel(const std::string& content_name,
+// When the socket is unwritable, we will use 10 Kbps (ignoring IP+UDP headers)
+// for pinging.  When the socket is writable, we will use only 1 Kbps because
+// we don't want to degrade the quality on a modem.  These numbers should work
+// well on a 28.8K modem, which is the slowest connection on which the voice
+// quality is reasonable at all.
+static const uint32_t PING_PACKET_SIZE = 60 * 8;
+// TODO(honghaiz): Change the word DELAY to INTERVAL whenever appropriate.
+// STRONG_PING_DELAY (480ms) is applied when the best connection is both
+// writable and receiving.
+static const uint32_t STRONG_PING_DELAY = 1000 * PING_PACKET_SIZE / 1000;
+// WEAK_PING_DELAY (48ms) is applied when the best connection is either not
+// writable or not receiving.
+const uint32_t WEAK_PING_DELAY = 1000 * PING_PACKET_SIZE / 10000;
+
+// If the current best connection is both writable and receiving, then we will
+// also try hard to make sure it is pinged at this rate (a little less than
+// 2 * STRONG_PING_DELAY).
+static const uint32_t MAX_CURRENT_STRONG_DELAY = 900;
+
+static const int MIN_CHECK_RECEIVING_DELAY = 50;  // ms
+
+P2PTransportChannel::P2PTransportChannel(const std::string& transport_name,
                                          int component,
                                          P2PTransport* transport,
-                                         PortAllocator *allocator) :
-    TransportChannelImpl(content_name, component),
-    transport_(transport),
-    allocator_(allocator),
-    worker_thread_(rtc::Thread::Current()),
-    incoming_only_(false),
-    waiting_for_signaling_(false),
-    error_(0),
-    best_connection_(NULL),
-    pending_best_connection_(NULL),
-    sort_dirty_(false),
-    was_writable_(false),
-    protocol_type_(ICEPROTO_HYBRID),
-    remote_ice_mode_(ICEMODE_FULL),
-    ice_role_(ICEROLE_UNKNOWN),
-    tiebreaker_(0),
-    remote_candidate_generation_(0) {
+                                         PortAllocator* allocator)
+    : TransportChannelImpl(transport_name, component),
+      transport_(transport),
+      allocator_(allocator),
+      worker_thread_(rtc::Thread::Current()),
+      incoming_only_(false),
+      error_(0),
+      best_connection_(NULL),
+      pending_best_connection_(NULL),
+      sort_dirty_(false),
+      remote_ice_mode_(ICEMODE_FULL),
+      ice_role_(ICEROLE_UNKNOWN),
+      tiebreaker_(0),
+      gathering_state_(kIceGatheringNew),
+      check_receiving_delay_(MIN_CHECK_RECEIVING_DELAY * 5),
+      receiving_timeout_(MIN_CHECK_RECEIVING_DELAY * 50),
+      backup_connection_ping_interval_(0) {
+  uint32_t weak_ping_delay = ::strtoul(
+      webrtc::field_trial::FindFullName("WebRTC-StunInterPacketDelay").c_str(),
+      nullptr, 10);
+  if (weak_ping_delay) {
+    weak_ping_delay_ =  weak_ping_delay;
+  }
 }
 
 P2PTransportChannel::~P2PTransportChannel() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
-  for (uint32 i = 0; i < allocator_sessions_.size(); ++i)
+  for (size_t i = 0; i < allocator_sessions_.size(); ++i)
     delete allocator_sessions_[i];
 }
 
 // Add the allocator session to our list so that we know which sessions
 // are still active.
 void P2PTransportChannel::AddAllocatorSession(PortAllocatorSession* session) {
-  session->set_generation(static_cast<uint32>(allocator_sessions_.size()));
+  ASSERT(worker_thread_ == rtc::Thread::Current());
+
+  session->set_generation(static_cast<uint32_t>(allocator_sessions_.size()));
   allocator_sessions_.push_back(session);
 
   // We now only want to apply new candidates that we receive to the ports
@@ -192,6 +261,7 @@ void P2PTransportChannel::AddAllocatorSession(PortAllocatorSession* session) {
 void P2PTransportChannel::AddConnection(Connection* connection) {
   connections_.push_back(connection);
   connection->set_remote_ice_mode(remote_ice_mode_);
+  connection->set_receiving_timeout(receiving_timeout_);
   connection->SignalReadPacket.connect(
       this, &P2PTransportChannel::OnReadPacket);
   connection->SignalReadyToSend.connect(
@@ -200,8 +270,8 @@ void P2PTransportChannel::AddConnection(Connection* connection) {
       this, &P2PTransportChannel::OnConnectionStateChange);
   connection->SignalDestroyed.connect(
       this, &P2PTransportChannel::OnConnectionDestroyed);
-  connection->SignalUseCandidate.connect(
-      this, &P2PTransportChannel::OnUseCandidate);
+  connection->SignalNominated.connect(this, &P2PTransportChannel::OnNominated);
+  had_connection_ = true;
 }
 
 void P2PTransportChannel::SetIceRole(IceRole ice_role) {
@@ -215,7 +285,7 @@ void P2PTransportChannel::SetIceRole(IceRole ice_role) {
   }
 }
 
-void P2PTransportChannel::SetIceTiebreaker(uint64 tiebreaker) {
+void P2PTransportChannel::SetIceTiebreaker(uint64_t tiebreaker) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   if (!ports_.empty()) {
     LOG(LS_ERROR)
@@ -226,22 +296,30 @@ void P2PTransportChannel::SetIceTiebreaker(uint64 tiebreaker) {
   tiebreaker_ = tiebreaker;
 }
 
-// Currently a channel is considered ICE completed once there is no
-// more than one connection per Network. This works for a single NIC
-// with both IPv4 and IPv6 enabled. However, this condition won't
-// happen when there are multiple NICs and all of them have
-// connectivity.
-// TODO(guoweis): Change Completion to be driven by a channel level
-// timer.
 TransportChannelState P2PTransportChannel::GetState() const {
-  std::set<rtc::Network*> networks;
+  return state_;
+}
 
-  if (connections_.size() == 0) {
+// A channel is considered ICE completed once there is at most one active
+// connection per network and at least one active connection.
+TransportChannelState P2PTransportChannel::ComputeState() const {
+  if (!had_connection_) {
+    return TransportChannelState::STATE_INIT;
+  }
+
+  std::vector<Connection*> active_connections;
+  for (Connection* connection : connections_) {
+    if (connection->active()) {
+      active_connections.push_back(connection);
+    }
+  }
+  if (active_connections.empty()) {
     return TransportChannelState::STATE_FAILED;
   }
 
-  for (uint32 i = 0; i < connections_.size(); ++i) {
-    rtc::Network* network = connections_[i]->port()->Network();
+  std::set<rtc::Network*> networks;
+  for (Connection* connection : active_connections) {
+    rtc::Network* network = connection->port()->Network();
     if (networks.find(network) == networks.end()) {
       networks.insert(network);
     } else {
@@ -251,69 +329,40 @@ TransportChannelState P2PTransportChannel::GetState() const {
       return TransportChannelState::STATE_CONNECTING;
     }
   }
+
   LOG_J(LS_VERBOSE, this) << "Ice is completed for this channel.";
-
   return TransportChannelState::STATE_COMPLETED;
 }
 
-bool P2PTransportChannel::GetIceProtocolType(IceProtocolType* type) const {
-  *type = protocol_type_;
-  return true;
-}
-
-void P2PTransportChannel::SetIceProtocolType(IceProtocolType type) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-
-  protocol_type_ = type;
-  for (std::vector<PortInterface *>::iterator it = ports_.begin();
-       it != ports_.end(); ++it) {
-    (*it)->SetIceProtocolType(protocol_type_);
-  }
-}
-
 void P2PTransportChannel::SetIceCredentials(const std::string& ice_ufrag,
                                             const std::string& ice_pwd) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
-  bool ice_restart = false;
-  if (!ice_ufrag_.empty() && !ice_pwd_.empty()) {
-    // Restart candidate allocation if there is any change in either
-    // ice ufrag or password.
-    ice_restart =
-        IceCredentialsChanged(ice_ufrag_, ice_pwd_, ice_ufrag, ice_pwd);
-  }
-
   ice_ufrag_ = ice_ufrag;
   ice_pwd_ = ice_pwd;
-
-  if (ice_restart) {
-    // Restart candidate gathering.
-    Allocate();
-  }
+  // Note: Candidate gathering will restart when MaybeStartGathering is next
+  // called.
 }
 
 void P2PTransportChannel::SetRemoteIceCredentials(const std::string& ice_ufrag,
                                                   const std::string& ice_pwd) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
-  bool ice_restart = false;
-  if (!remote_ice_ufrag_.empty() && !remote_ice_pwd_.empty()) {
-    ice_restart = (remote_ice_ufrag_ != ice_ufrag) ||
-                  (remote_ice_pwd_!= ice_pwd);
+  IceParameters* current_ice = remote_ice();
+  IceParameters new_ice(ice_ufrag, ice_pwd);
+  if (!current_ice || *current_ice != new_ice) {
+    // Keep the ICE credentials so that newer connections
+    // are prioritized over the older ones.
+    remote_ice_parameters_.push_back(new_ice);
   }
 
-  remote_ice_ufrag_ = ice_ufrag;
-  remote_ice_pwd_ = ice_pwd;
-
+  // Update the pwd of remote candidate if needed.
+  for (RemoteCandidate& candidate : remote_candidates_) {
+    if (candidate.username() == ice_ufrag && candidate.password().empty()) {
+      candidate.set_password(ice_pwd);
+    }
+  }
   // We need to update the credentials for any peer reflexive candidates.
-  std::vector<Connection*>::iterator it = connections_.begin();
-  for (; it != connections_.end(); ++it) {
-    (*it)->MaybeSetRemoteIceCredentials(ice_ufrag, ice_pwd);
-  }
-
-  if (ice_restart) {
-    // |candidate.generation()| is not signaled in ICEPROTO_RFC5245.
-    // Therefore we need to keep track of the remote ice restart so
-    // newer connections are prioritized over the older.
-    ++remote_candidate_generation_;
+  for (Connection* conn : connections_) {
+    conn->MaybeSetRemoteIceCredentials(ice_ufrag, ice_pwd);
   }
 }
 
@@ -321,6 +370,32 @@ void P2PTransportChannel::SetRemoteIceMode(IceMode mode) {
   remote_ice_mode_ = mode;
 }
 
+void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
+  gather_continually_ = config.gather_continually;
+  LOG(LS_INFO) << "Set gather_continually to " << gather_continually_;
+
+  if (config.backup_connection_ping_interval >= 0 &&
+      backup_connection_ping_interval_ !=
+          config.backup_connection_ping_interval) {
+    backup_connection_ping_interval_ = config.backup_connection_ping_interval;
+    LOG(LS_INFO) << "Set backup connection ping interval to "
+                 << backup_connection_ping_interval_ << " milliseconds.";
+  }
+
+  if (config.receiving_timeout_ms >= 0 &&
+      receiving_timeout_ != config.receiving_timeout_ms) {
+    receiving_timeout_ = config.receiving_timeout_ms;
+    check_receiving_delay_ =
+        std::max(MIN_CHECK_RECEIVING_DELAY, receiving_timeout_ / 10);
+
+    for (Connection* connection : connections_) {
+      connection->set_receiving_timeout(receiving_timeout_);
+    }
+    LOG(LS_INFO) << "Set ICE receiving timeout to " << receiving_timeout_
+                 << " milliseconds";
+  }
+}
+
 // Go into the state of processing candidates, and running in general
 void P2PTransportChannel::Connect() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
@@ -331,44 +406,24 @@ void P2PTransportChannel::Connect() {
     return;
   }
 
-  // Kick off an allocator session
-  Allocate();
-
-  // Start pinging as the ports come in.
-  thread()->Post(this, MSG_PING);
+  // Start checking and pinging as the ports come in.
+  thread()->Post(this, MSG_CHECK_AND_PING);
 }
 
-// Reset the socket, clear up any previous allocations and start over
-void P2PTransportChannel::Reset() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-
-  // Get rid of all the old allocators.  This should clean up everything.
-  for (uint32 i = 0; i < allocator_sessions_.size(); ++i)
-    delete allocator_sessions_[i];
-
-  allocator_sessions_.clear();
-  ports_.clear();
-  connections_.clear();
-  best_connection_ = NULL;
-
-  // Forget about all of the candidates we got before.
-  remote_candidates_.clear();
-
-  // Revert to the initial state.
-  set_readable(false);
-  set_writable(false);
-
-  // Reinitialize the rest of our state.
-  waiting_for_signaling_ = false;
-  sort_dirty_ = false;
-
-  // If we allocated before, start a new one now.
-  if (transport_->connect_requested())
-    Allocate();
-
-  // Start pinging as the ports come in.
-  thread()->Clear(this);
-  thread()->Post(this, MSG_PING);
+void P2PTransportChannel::MaybeStartGathering() {
+  // Start gathering if we never started before, or if an ICE restart occurred.
+  if (allocator_sessions_.empty() ||
+      IceCredentialsChanged(allocator_sessions_.back()->ice_ufrag(),
+                            allocator_sessions_.back()->ice_pwd(), ice_ufrag_,
+                            ice_pwd_)) {
+    if (gathering_state_ != kIceGatheringGathering) {
+      gathering_state_ = kIceGatheringGathering;
+      SignalGatheringState(this);
+    }
+    // Time for a new allocator
+    AddAllocatorSession(allocator_->CreateSession(
+        SessionId(), transport_name(), component(), ice_ufrag_, ice_pwd_));
+  }
 }
 
 // A new port is available, attempt to make connections for it
@@ -392,7 +447,6 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
   // The session will handle this, and send an initiate/accept/modify message
   // if one is pending.
 
-  port->SetIceProtocolType(protocol_type_);
   port->SetIceRole(ice_role_);
   port->SetIceTiebreaker(tiebreaker_);
   ports_.push_back(port);
@@ -401,6 +455,7 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
   port->SignalDestroyed.connect(this, &P2PTransportChannel::OnPortDestroyed);
   port->SignalRoleConflict.connect(
       this, &P2PTransportChannel::OnRoleConflict);
+  port->SignalSentPacket.connect(this, &P2PTransportChannel::OnSentPacket);
 
   // Attempt to create a connection from this new port to all of the remote
   // candidates that we were given so far.
@@ -408,7 +463,7 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
   std::vector<RemoteCandidate>::iterator iter;
   for (iter = remote_candidates_.begin(); iter != remote_candidates_.end();
        ++iter) {
-    CreateConnection(port, *iter, iter->origin_port(), false);
+    CreateConnection(port, *iter, iter->origin_port());
   }
 
   SortConnections();
@@ -416,17 +471,21 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
 
 // A new candidate is available, let listeners know
 void P2PTransportChannel::OnCandidatesReady(
-    PortAllocatorSession *session, const std::vector<Candidate>& candidates) {
+    PortAllocatorSession* session,
+    const std::vector<Candidate>& candidates) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   for (size_t i = 0; i < candidates.size(); ++i) {
-    SignalCandidateReady(this, candidates[i]);
+    SignalCandidateGathered(this, candidates[i]);
   }
 }
 
 void P2PTransportChannel::OnCandidatesAllocationDone(
     PortAllocatorSession* session) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
-  SignalCandidatesAllocationDone(this);
+  gathering_state_ = kIceGatheringComplete;
+  LOG(LS_INFO) << "P2PTransportChannel: " << transport_name() << ", component "
+               << component() << " gathering complete";
+  SignalGatheringState(this);
 }
 
 // Handle stun packets
@@ -471,55 +530,29 @@ void P2PTransportChannel::OnUnknownAddress(
     }
   }
 
+  uint32_t remote_generation = 0;
   // The STUN binding request may arrive after setRemoteDescription and before
   // adding remote candidate, so we need to set the password to the shared
   // password if the user name matches.
-  if (remote_password.empty() && remote_username == remote_ice_ufrag_) {
-    remote_password = remote_ice_pwd_;
+  if (remote_password.empty()) {
+    const IceParameters* ice_param =
+        FindRemoteIceFromUfrag(remote_username, &remote_generation);
+    // Note: if not found, the remote_generation will still be 0.
+    if (ice_param != nullptr) {
+      remote_password = ice_param->pwd;
+    }
   }
 
-  Candidate new_remote_candidate;
-  if (candidate != NULL) {
-    new_remote_candidate = *candidate;
+  Candidate remote_candidate;
+  bool remote_candidate_is_new = (candidate == nullptr);
+  if (!remote_candidate_is_new) {
+    remote_candidate = *candidate;
     if (ufrag_per_port) {
-      new_remote_candidate.set_address(address);
+      remote_candidate.set_address(address);
     }
   } else {
     // Create a new candidate with this address.
-    std::string type;
-    if (port->IceProtocol() == ICEPROTO_RFC5245) {
-      type = PRFLX_PORT_TYPE;
-    } else {
-      // G-ICE doesn't support prflx candidate.
-      // We set candidate type to STUN_PORT_TYPE if the binding request comes
-      // from a relay port or the shared socket is used. Otherwise we use the
-      // port's type as the candidate type.
-      if (port->Type() == RELAY_PORT_TYPE || port->SharedSocket()) {
-        type = STUN_PORT_TYPE;
-      } else {
-        type = port->Type();
-      }
-    }
-
-    new_remote_candidate =
-        Candidate(component(), ProtoToString(proto), address, 0,
-                  remote_username, remote_password, type, 0U, "");
-
-    // From RFC 5245, section-7.2.1.3:
-    // The foundation of the candidate is set to an arbitrary value, different
-    // from the foundation for all other remote candidates.
-    new_remote_candidate.set_foundation(
-        rtc::ToString<uint32>(rtc::ComputeCrc32(new_remote_candidate.id())));
-
-    new_remote_candidate.set_priority(new_remote_candidate.GetPriority(
-        ICE_TYPE_PREFERENCE_PRFLX, port->Network()->preference(), 0));
-  }
-
-  if (port->IceProtocol() == ICEPROTO_RFC5245) {
-    // RFC 5245
-    // If the source transport address of the request does not match any
-    // existing remote candidates, it represents a new peer reflexive remote
-    // candidate.
+    int remote_candidate_priority;
 
     // The priority of the candidate is set to the PRIORITY attribute
     // from the request.
@@ -529,79 +562,71 @@ void P2PTransportChannel::OnUnknownAddress(
       LOG(LS_WARNING) << "P2PTransportChannel::OnUnknownAddress - "
                       << "No STUN_ATTR_PRIORITY found in the "
                       << "stun request message";
-      port->SendBindingErrorResponse(stun_msg, address,
-                                     STUN_ERROR_BAD_REQUEST,
+      port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_BAD_REQUEST,
                                      STUN_ERROR_REASON_BAD_REQUEST);
       return;
     }
-    new_remote_candidate.set_priority(priority_attr->value());
+    remote_candidate_priority = priority_attr->value();
 
-    // RFC5245, the agent constructs a pair whose local candidate is equal to
-    // the transport address on which the STUN request was received, and a
-    // remote candidate equal to the source transport address where the
-    // request came from.
+    // RFC 5245
+    // If the source transport address of the request does not match any
+    // existing remote candidates, it represents a new peer reflexive remote
+    // candidate.
+    remote_candidate = Candidate(component(), ProtoToString(proto), address, 0,
+                                 remote_username, remote_password,
+                                 PRFLX_PORT_TYPE, remote_generation, "");
 
-    // There shouldn't be an existing connection with this remote address.
-    // When ports are muxed, this channel might get multiple unknown address
-    // signals. In that case if the connection is already exists, we should
-    // simply ignore the signal othewise send server error.
-    if (port->GetConnection(new_remote_candidate.address())) {
-      if (port_muxed) {
-        LOG(LS_INFO) << "Connection already exists for peer reflexive "
-                     << "candidate: " << new_remote_candidate.ToString();
-        return;
-      } else {
-        ASSERT(false);
-        port->SendBindingErrorResponse(stun_msg, address,
-                                       STUN_ERROR_SERVER_ERROR,
-                                       STUN_ERROR_REASON_SERVER_ERROR);
-        return;
-      }
-    }
+    // From RFC 5245, section-7.2.1.3:
+    // The foundation of the candidate is set to an arbitrary value, different
+    // from the foundation for all other remote candidates.
+    remote_candidate.set_foundation(
+        rtc::ToString<uint32_t>(rtc::ComputeCrc32(remote_candidate.id())));
 
-    Connection* connection = port->CreateConnection(
-        new_remote_candidate, cricket::PortInterface::ORIGIN_THIS_PORT);
-    if (!connection) {
+    remote_candidate.set_priority(remote_candidate_priority);
+  }
+
+  // RFC5245, the agent constructs a pair whose local candidate is equal to
+  // the transport address on which the STUN request was received, and a
+  // remote candidate equal to the source transport address where the
+  // request came from.
+
+  // There shouldn't be an existing connection with this remote address.
+  // When ports are muxed, this channel might get multiple unknown address
+  // signals. In that case if the connection is already exists, we should
+  // simply ignore the signal otherwise send server error.
+  if (port->GetConnection(remote_candidate.address())) {
+    if (port_muxed) {
+      LOG(LS_INFO) << "Connection already exists for peer reflexive "
+                   << "candidate: " << remote_candidate.ToString();
+      return;
+    } else {
       ASSERT(false);
       port->SendBindingErrorResponse(stun_msg, address,
                                      STUN_ERROR_SERVER_ERROR,
                                      STUN_ERROR_REASON_SERVER_ERROR);
       return;
     }
-
-    LOG(LS_INFO) << "Adding connection from peer reflexive candidate: "
-                 << new_remote_candidate.ToString();
-    AddConnection(connection);
-    connection->ReceivedPing();
-
-    // Send the pinger a successful stun response.
-    port->SendBindingResponse(stun_msg, address);
-
-    // Update the list of connections since we just added another.  We do this
-    // after sending the response since it could (in principle) delete the
-    // connection in question.
-    SortConnections();
-  } else {
-    // Check for connectivity to this address. Create connections
-    // to this address across all local ports. First, add this as a new remote
-    // address
-    if (!CreateConnections(new_remote_candidate, port, true)) {
-      // Hopefully this won't occur, because changing a destination address
-      // shouldn't cause a new connection to fail
-      ASSERT(false);
-      port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_SERVER_ERROR,
-          STUN_ERROR_REASON_SERVER_ERROR);
-      return;
-    }
-
-    // Send the pinger a successful stun response.
-    port->SendBindingResponse(stun_msg, address);
-
-    // Update the list of connections since we just added another.  We do this
-    // after sending the response since it could (in principle) delete the
-    // connection in question.
-    SortConnections();
   }
+
+  Connection* connection = port->CreateConnection(
+      remote_candidate, cricket::PortInterface::ORIGIN_THIS_PORT);
+  if (!connection) {
+    ASSERT(false);
+    port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_SERVER_ERROR,
+                                   STUN_ERROR_REASON_SERVER_ERROR);
+    return;
+  }
+
+  LOG(LS_INFO) << "Adding connection from "
+               << (remote_candidate_is_new ? "peer reflexive" : "resurrected")
+               << " candidate: " << remote_candidate.ToString();
+  AddConnection(connection);
+  connection->HandleBindingRequest(stun_msg);
+
+  // Update the list of connections since we just added another.  We do this
+  // after sending the response since it could (in principle) delete the
+  // connection in question.
+  SortConnections();
 }
 
 void P2PTransportChannel::OnRoleConflict(PortInterface* port) {
@@ -609,38 +634,78 @@ void P2PTransportChannel::OnRoleConflict(PortInterface* port) {
                              // from Transport.
 }
 
-// When the signalling channel is ready, we can really kick off the allocator
-void P2PTransportChannel::OnSignalingReady() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-  if (waiting_for_signaling_) {
-    waiting_for_signaling_ = false;
-    AddAllocatorSession(allocator_->CreateSession(
-        SessionId(), content_name(), component(), ice_ufrag_, ice_pwd_));
+const IceParameters* P2PTransportChannel::FindRemoteIceFromUfrag(
+    const std::string& ufrag,
+    uint32_t* generation) {
+  const auto& params = remote_ice_parameters_;
+  auto it = std::find_if(
+      params.rbegin(), params.rend(),
+      [ufrag](const IceParameters& param) { return param.ufrag == ufrag; });
+  if (it == params.rend()) {
+    // Not found.
+    return nullptr;
   }
+  *generation = params.rend() - it - 1;
+  return &(*it);
 }
 
-void P2PTransportChannel::OnUseCandidate(Connection* conn) {
+void P2PTransportChannel::OnNominated(Connection* conn) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   ASSERT(ice_role_ == ICEROLE_CONTROLLED);
-  ASSERT(protocol_type_ == ICEPROTO_RFC5245);
+
   if (conn->write_state() == Connection::STATE_WRITABLE) {
     if (best_connection_ != conn) {
       pending_best_connection_ = NULL;
+      LOG(LS_INFO) << "Switching best connection on controlled side: "
+                   << conn->ToString();
       SwitchBestConnectionTo(conn);
       // Now we have selected the best connection, time to prune other existing
       // connections and update the read/write state of the channel.
       RequestSort();
     }
   } else {
+    LOG(LS_INFO) << "Not switching the best connection on controlled side yet,"
+                 << " because it's not writable: " << conn->ToString();
     pending_best_connection_ = conn;
   }
 }
 
-void P2PTransportChannel::OnCandidate(const Candidate& candidate) {
+void P2PTransportChannel::AddRemoteCandidate(const Candidate& candidate) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
+  uint32_t generation = GetRemoteCandidateGeneration(candidate);
+  // If a remote candidate with a previous generation arrives, drop it.
+  if (generation < remote_ice_generation()) {
+    LOG(LS_WARNING) << "Dropping a remote candidate because its ufrag "
+                    << candidate.username()
+                    << " indicates it was for a previous generation.";
+    return;
+  }
+
+  Candidate new_remote_candidate(candidate);
+  new_remote_candidate.set_generation(generation);
+  // ICE candidates don't need to have username and password set, but
+  // the code below this (specifically, ConnectionRequest::Prepare in
+  // port.cc) uses the remote candidates's username.  So, we set it
+  // here.
+  if (remote_ice()) {
+    if (candidate.username().empty()) {
+      new_remote_candidate.set_username(remote_ice()->ufrag);
+    }
+    if (new_remote_candidate.username() == remote_ice()->ufrag) {
+      if (candidate.password().empty()) {
+        new_remote_candidate.set_password(remote_ice()->pwd);
+      }
+    } else {
+      // The candidate belongs to the next generation. Its pwd will be set
+      // when the new remote ICE credentials arrive.
+      LOG(LS_WARNING) << "A remote candidate arrives with an unknown ufrag: "
+                      << candidate.username();
+    }
+  }
+
   // Create connections to this remote candidate.
-  CreateConnections(candidate, NULL, false);
+  CreateConnections(new_remote_candidate, NULL);
 
   // Resort the connections list, which may have new elements.
   SortConnections();
@@ -650,24 +715,9 @@ void P2PTransportChannel::OnCandidate(const Candidate& candidate) {
 // remote candidate.  The return value is true if we created a connection from
 // the origin port.
 bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate,
-                                            PortInterface* origin_port,
-                                            bool readable) {
+                                            PortInterface* origin_port) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
-  Candidate new_remote_candidate(remote_candidate);
-  new_remote_candidate.set_generation(
-      GetRemoteCandidateGeneration(remote_candidate));
-  // ICE candidates don't need to have username and password set, but
-  // the code below this (specifically, ConnectionRequest::Prepare in
-  // port.cc) uses the remote candidates's username.  So, we set it
-  // here.
-  if (remote_candidate.username().empty()) {
-    new_remote_candidate.set_username(remote_ice_ufrag_);
-  }
-  if (remote_candidate.password().empty()) {
-    new_remote_candidate.set_password(remote_ice_pwd_);
-  }
-
   // If we've already seen the new remote candidate (in the current candidate
   // generation), then we shouldn't try creating connections for it.
   // We either already have a connection for it, or we previously created one
@@ -676,7 +726,7 @@ bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate,
   // immediately be re-pruned, churning the network for no purpose.
   // This only applies to candidates received over signaling (i.e. origin_port
   // is NULL).
-  if (!origin_port && IsDuplicateRemoteCandidate(new_remote_candidate)) {
+  if (!origin_port && IsDuplicateRemoteCandidate(remote_candidate)) {
     // return true to indicate success, without creating any new connections.
     return true;
   }
@@ -689,7 +739,7 @@ bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate,
   bool created = false;
   std::vector<PortInterface *>::reverse_iterator it;
   for (it = ports_.rbegin(); it != ports_.rend(); ++it) {
-    if (CreateConnection(*it, new_remote_candidate, origin_port, readable)) {
+    if (CreateConnection(*it, remote_candidate, origin_port)) {
       if (*it == origin_port)
         created = true;
     }
@@ -697,13 +747,12 @@ bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate,
 
   if ((origin_port != NULL) &&
       std::find(ports_.begin(), ports_.end(), origin_port) == ports_.end()) {
-    if (CreateConnection(
-        origin_port, new_remote_candidate, origin_port, readable))
+    if (CreateConnection(origin_port, remote_candidate, origin_port))
       created = true;
   }
 
   // Remember this remote candidate so that we can add it to future ports.
-  RememberRemoteCandidate(new_remote_candidate, origin_port);
+  RememberRemoteCandidate(remote_candidate, origin_port);
 
   return created;
 }
@@ -712,8 +761,10 @@ bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate,
 // And then listen to connection object for changes.
 bool P2PTransportChannel::CreateConnection(PortInterface* port,
                                            const Candidate& remote_candidate,
-                                           PortInterface* origin_port,
-                                           bool readable) {
+                                           PortInterface* origin_port) {
+  if (!port->SupportsProtocol(remote_candidate.protocol())) {
+    return false;
+  }
   // Look for an existing connection with this remote address.  If one is not
   // found, then we can create a new connection for this address.
   Connection* connection = port->GetConnection(remote_candidate.address());
@@ -748,11 +799,6 @@ bool P2PTransportChannel::CreateConnection(PortInterface* port,
                          << connections_.size() << " total)";
   }
 
-  // If we are readable, it is because we are creating this in response to a
-  // ping from the other side.  This will cause the state to become readable.
-  if (readable)
-    connection->ReceivedPing();
-
   return true;
 }
 
@@ -763,24 +809,29 @@ bool P2PTransportChannel::FindConnection(
   return citer != connections_.end();
 }
 
-uint32 P2PTransportChannel::GetRemoteCandidateGeneration(
+uint32_t P2PTransportChannel::GetRemoteCandidateGeneration(
     const Candidate& candidate) {
-  if (protocol_type_ == ICEPROTO_GOOGLE) {
-    // The Candidate.generation() can be trusted. Nothing needs to be done.
+  // If the candidate has a ufrag, use it to find the generation.
+  if (!candidate.username().empty()) {
+    uint32_t generation = 0;
+    if (!FindRemoteIceFromUfrag(candidate.username(), &generation)) {
+      // If the ufrag is not found, assume the next/future generation.
+      generation = static_cast<uint32_t>(remote_ice_parameters_.size());
+    }
+    return generation;
+  }
+  // If candidate generation is set, use that.
+  if (candidate.generation() > 0) {
     return candidate.generation();
   }
-  // |candidate.generation()| is not signaled in ICEPROTO_RFC5245.
-  // Therefore we need to keep track of the remote ice restart so
-  // newer connections are prioritized over the older.
-  ASSERT(candidate.generation() == 0 ||
-         candidate.generation() == remote_candidate_generation_);
-  return remote_candidate_generation_;
+  // Otherwise, assume the generation from remote ice parameters.
+  return remote_ice_generation();
 }
 
 // Check if remote candidate is already cached.
 bool P2PTransportChannel::IsDuplicateRemoteCandidate(
     const Candidate& candidate) {
-  for (uint32 i = 0; i < remote_candidates_.size(); ++i) {
+  for (size_t i = 0; i < remote_candidates_.size(); ++i) {
     if (remote_candidates_[i].IsEquivalent(candidate)) {
       return true;
     }
@@ -793,7 +844,7 @@ void P2PTransportChannel::RememberRemoteCandidate(
     const Candidate& remote_candidate, PortInterface* origin_port) {
   // Remove any candidates whose generation is older than this one.  The
   // presence of a new generation indicates that the old ones are not useful.
-  uint32 i = 0;
+  size_t i = 0;
   while (i < remote_candidates_.size()) {
     if (remote_candidates_[i].generation() < remote_candidate.generation()) {
       LOG(INFO) << "Pruning candidate from old generation: "
@@ -827,7 +878,7 @@ int P2PTransportChannel::SetOption(rtc::Socket::Option opt, int value) {
     it->second = value;
   }
 
-  for (uint32 i = 0; i < ports_.size(); ++i) {
+  for (size_t i = 0; i < ports_.size(); ++i) {
     int val = ports_[i]->SetOption(opt, value);
     if (val < 0) {
       // Because this also occurs deferred, probably no point in reporting an
@@ -878,12 +929,10 @@ bool P2PTransportChannel::GetStats(ConnectionInfos *infos) {
   infos->clear();
 
   std::vector<Connection *>::const_iterator it;
-  for (it = connections_.begin(); it != connections_.end(); ++it) {
-    Connection *connection = *it;
+  for (Connection* connection : connections_) {
     ConnectionInfo info;
     info.best_connection = (best_connection_ == connection);
-    info.readable =
-        (connection->read_state() == Connection::STATE_READABLE);
+    info.receiving = connection->receiving();
     info.writable =
         (connection->write_state() == Connection::STATE_WRITABLE);
     info.timeout =
@@ -914,21 +963,13 @@ rtc::DiffServCodePoint P2PTransportChannel::DefaultDscpValue() const {
   return static_cast<rtc::DiffServCodePoint> (it->second);
 }
 
-// Begin allocate (or immediately re-allocate, if MSG_ALLOCATE pending)
-void P2PTransportChannel::Allocate() {
-  // Time for a new allocator, lets make sure we have a signalling channel
-  // to communicate candidates through first.
-  waiting_for_signaling_ = true;
-  SignalRequestSignaling(this);
-}
-
 // Monitor connection states.
 void P2PTransportChannel::UpdateConnectionStates() {
-  uint32 now = rtc::Time();
+  uint32_t now = rtc::Time();
 
   // We need to copy the list of connections since some may delete themselves
   // when we call UpdateState.
-  for (uint32 i = 0; i < connections_.size(); ++i)
+  for (size_t i = 0; i < connections_.size(); ++i)
     connections_[i]->UpdateState(now);
 }
 
@@ -949,73 +990,42 @@ void P2PTransportChannel::SortConnections() {
   // will be sorted.
   UpdateConnectionStates();
 
-  if (protocol_type_ == ICEPROTO_HYBRID) {
-    // If we are in hybrid mode, we are not sending any ping requests, so there
-    // is no point in sorting the connections. In hybrid state, ports can have
-    // different protocol than hybrid and protocol may differ from one another.
-    // Instead just update the state of this channel
-    UpdateChannelState();
-    return;
-  }
-
   // Any changes after this point will require a re-sort.
   sort_dirty_ = false;
 
-  // Get a list of the networks that we are using.
-  std::set<rtc::Network*> networks;
-  for (uint32 i = 0; i < connections_.size(); ++i)
-    networks.insert(connections_[i]->port()->Network());
-
   // Find the best alternative connection by sorting.  It is important to note
   // that amongst equal preference, writable connections, this will choose the
   // one whose estimated latency is lowest.  So it is the only one that we
   // need to consider switching to.
-
   ConnectionCompare cmp;
   std::stable_sort(connections_.begin(), connections_.end(), cmp);
-  LOG(LS_VERBOSE) << "Sorting available connections:";
-  for (uint32 i = 0; i < connections_.size(); ++i) {
+  LOG(LS_VERBOSE) << "Sorting " << connections_.size()
+                  << " available connections:";
+  for (size_t i = 0; i < connections_.size(); ++i) {
     LOG(LS_VERBOSE) << connections_[i]->ToString();
   }
 
-  Connection* top_connection = NULL;
-  if (connections_.size() > 0)
-    top_connection = connections_[0];
-
-  // We don't want to pick the best connections if channel is using RFC5245
-  // and it's mode is CONTROLLED, as connections will be selected by the
-  // CONTROLLING agent.
+  Connection* top_connection =
+      (connections_.size() > 0) ? connections_[0] : nullptr;
 
   // If necessary, switch to the new choice.
-  if (protocol_type_ != ICEPROTO_RFC5245 || ice_role_ == ICEROLE_CONTROLLING) {
-    if (ShouldSwitch(best_connection_, top_connection))
-      SwitchBestConnectionTo(top_connection);
+  // Note that |top_connection| doesn't have to be writable to become the best
+  // connection although it will have higher priority if it is writable.
+  if (ShouldSwitch(best_connection_, top_connection, ice_role_)) {
+    LOG(LS_INFO) << "Switching best connection: " << top_connection->ToString();
+    SwitchBestConnectionTo(top_connection);
   }
 
-  // We can prune any connection for which there is a writable connection on
-  // the same network with better or equal priority.  We leave those with
-  // better priority just in case they become writable later (at which point,
-  // we would prune out the current best connection).  We leave connections on
-  // other networks because they may not be using the same resources and they
-  // may represent very distinct paths over which we can switch.
-  std::set<rtc::Network*>::iterator network;
-  for (network = networks.begin(); network != networks.end(); ++network) {
-    Connection* primier = GetBestConnectionOnNetwork(*network);
-    if (!primier || (primier->write_state() != Connection::STATE_WRITABLE))
-      continue;
-
-    for (uint32 i = 0; i < connections_.size(); ++i) {
-      if ((connections_[i] != primier) &&
-          (connections_[i]->port()->Network() == *network) &&
-          (CompareConnectionCandidates(primier, connections_[i]) >= 0)) {
-        connections_[i]->Prune();
-      }
-    }
+  // Controlled side can prune only if the best connection has been nominated.
+  // because otherwise it may delete the connection that will be selected by
+  // the controlling side.
+  if (ice_role_ == ICEROLE_CONTROLLING || best_nominated_connection()) {
+    PruneConnections();
   }
 
   // Check if all connections are timedout.
   bool all_connections_timedout = true;
-  for (uint32 i = 0; i < connections_.size(); ++i) {
+  for (size_t i = 0; i < connections_.size(); ++i) {
     if (connections_[i]->write_state() != Connection::STATE_WRITE_TIMEOUT) {
       all_connections_timedout = false;
       break;
@@ -1024,19 +1034,52 @@ void P2PTransportChannel::SortConnections() {
 
   // Now update the writable state of the channel with the information we have
   // so far.
-  if (best_connection_ && best_connection_->writable()) {
-    HandleWritable();
-  } else if (all_connections_timedout) {
+  if (all_connections_timedout) {
     HandleAllTimedOut();
-  } else {
-    HandleNotWritable();
   }
 
   // Update the state of this channel.  This method is called whenever the
   // state of any connection changes, so this is a good place to do this.
-  UpdateChannelState();
+  UpdateState();
 }
 
+Connection* P2PTransportChannel::best_nominated_connection() const {
+  return (best_connection_ && best_connection_->nominated()) ? best_connection_
+                                                             : nullptr;
+}
+
+void P2PTransportChannel::PruneConnections() {
+  // We can prune any connection for which there is a connected, writable
+  // connection on the same network with better or equal priority.  We leave
+  // those with better priority just in case they become writable later (at
+  // which point, we would prune out the current best connection).  We leave
+  // connections on other networks because they may not be using the same
+  // resources and they may represent very distinct paths over which we can
+  // switch. If the |premier| connection is not connected, we may be
+  // reconnecting a TCP connection and temporarily do not prune connections in
+  // this network. See the big comment in CompareConnections.
+
+  // Get a list of the networks that we are using.
+  std::set<rtc::Network*> networks;
+  for (const Connection* conn : connections_) {
+    networks.insert(conn->port()->Network());
+  }
+  for (rtc::Network* network : networks) {
+    Connection* premier = GetBestConnectionOnNetwork(network);
+    // Do not prune connections if the current best connection is weak on this
+    // network. Otherwise, it may delete connections prematurely.
+    if (!premier || premier->weak()) {
+      continue;
+    }
+
+    for (Connection* conn : connections_) {
+      if ((conn != premier) && (conn->port()->Network() == network) &&
+          (CompareConnectionCandidates(premier, conn) >= 0)) {
+        conn->Prune();
+      }
+    }
+  }
+}
 
 // Track the best connection, and let listeners know
 void P2PTransportChannel::SwitchBestConnectionTo(Connection* conn) {
@@ -1057,54 +1100,56 @@ void P2PTransportChannel::SwitchBestConnectionTo(Connection* conn) {
   }
 }
 
-void P2PTransportChannel::UpdateChannelState() {
-  // The Handle* functions already set the writable state.  We'll just double-
-  // check it here.
-  bool writable = ((best_connection_ != NULL)  &&
-      (best_connection_->write_state() ==
-      Connection::STATE_WRITABLE));
-  ASSERT(writable == this->writable());
-  if (writable != this->writable())
-    LOG(LS_ERROR) << "UpdateChannelState: writable state mismatch";
+// Warning: UpdateState should eventually be called whenever a connection
+// is added, deleted, or the write state of any connection changes so that the
+// transport controller will get the up-to-date channel state. However it
+// should not be called too often; in the case that multiple connection states
+// change, it should be called after all the connection states have changed. For
+// example, we call this at the end of SortConnections.
+void P2PTransportChannel::UpdateState() {
+  state_ = ComputeState();
 
-  bool readable = false;
-  for (uint32 i = 0; i < connections_.size(); ++i) {
-    if (connections_[i]->read_state() == Connection::STATE_READABLE) {
-      readable = true;
+  bool writable = best_connection_ && best_connection_->writable();
+  set_writable(writable);
+
+  bool receiving = false;
+  for (const Connection* connection : connections_) {
+    if (connection->receiving()) {
+      receiving = true;
       break;
     }
   }
-  set_readable(readable);
+  set_receiving(receiving);
 }
 
-// We checked the status of our connections and we had at least one that
-// was writable, go into the writable state.
-void P2PTransportChannel::HandleWritable() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-  if (!writable()) {
-    for (uint32 i = 0; i < allocator_sessions_.size(); ++i) {
-      if (allocator_sessions_[i]->IsGettingPorts()) {
-        allocator_sessions_[i]->StopGettingPorts();
-      }
+void P2PTransportChannel::MaybeStopPortAllocatorSessions() {
+  if (!IsGettingPorts()) {
+    return;
+  }
+
+  for (PortAllocatorSession* session : allocator_sessions_) {
+    if (!session->IsGettingPorts()) {
+      continue;
     }
-  }
-
-  was_writable_ = true;
-  set_writable(true);
-}
-
-// Notify upper layer about channel not writable state, if it was before.
-void P2PTransportChannel::HandleNotWritable() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-  if (was_writable_) {
-    was_writable_ = false;
-    set_writable(false);
+    // If gathering continually, keep the last session running so that it
+    // will gather candidates if the networks change.
+    if (gather_continually_ && session == allocator_sessions_.back()) {
+      session->ClearGettingPorts();
+      break;
+    }
+    session->StopGettingPorts();
   }
 }
 
+// If all connections timed out, delete them all.
 void P2PTransportChannel::HandleAllTimedOut() {
-  // Currently we are treating this as channel not writable.
-  HandleNotWritable();
+  for (Connection* connection : connections_) {
+    connection->Destroy();
+  }
+}
+
+bool P2PTransportChannel::weak() const {
+  return !best_connection_ || best_connection_->weak();
 }
 
 // If we have a best connection, return it, otherwise return top one in the
@@ -1116,7 +1161,7 @@ Connection* P2PTransportChannel::GetBestConnectionOnNetwork(
     return best_connection_;
 
   // Otherwise, we return the top-most in sorted order.
-  for (uint32 i = 0; i < connections_.size(); ++i) {
+  for (size_t i = 0; i < connections_.size(); ++i) {
     if (connections_[i]->port()->Network() == network)
       return connections_[i];
   }
@@ -1130,8 +1175,8 @@ void P2PTransportChannel::OnMessage(rtc::Message *pmsg) {
     case MSG_SORT:
       OnSort();
       break;
-    case MSG_PING:
-      OnPing();
+    case MSG_CHECK_AND_PING:
+      OnCheckAndPing();
       break;
     default:
       ASSERT(false);
@@ -1145,24 +1190,35 @@ void P2PTransportChannel::OnSort() {
   SortConnections();
 }
 
-// Handle queued up ping request
-void P2PTransportChannel::OnPing() {
+// Handle queued up check-and-ping request
+void P2PTransportChannel::OnCheckAndPing() {
   // Make sure the states of the connections are up-to-date (since this affects
   // which ones are pingable).
   UpdateConnectionStates();
+  // When the best connection is either not receiving or not writable,
+  // switch to weak ping delay.
+  int ping_delay = weak() ? weak_ping_delay_ : STRONG_PING_DELAY;
+  if (rtc::Time() >= last_ping_sent_ms_ + ping_delay) {
+    Connection* conn = FindNextPingableConnection();
+    if (conn) {
+      PingConnection(conn);
+    }
+  }
+  int check_delay = std::min(ping_delay, check_receiving_delay_);
+  thread()->PostDelayed(check_delay, this, MSG_CHECK_AND_PING);
+}
 
-  // Find the oldest pingable connection and have it do a ping.
-  Connection* conn = FindNextPingableConnection();
-  if (conn)
-    PingConnection(conn);
-
-  // Post ourselves a message to perform the next ping.
-  uint32 delay = writable() ? WRITABLE_DELAY : UNWRITABLE_DELAY;
-  thread()->PostDelayed(delay, this, MSG_PING);
+// A connection is considered a backup connection if the channel state
+// is completed, the connection is not the best connection and it is active.
+bool P2PTransportChannel::IsBackupConnection(Connection* conn) const {
+  return state_ == STATE_COMPLETED && conn != best_connection_ &&
+         conn->active();
 }
 
 // Is the connection in a state for us to even consider pinging the other side?
-bool P2PTransportChannel::IsPingable(Connection* conn) {
+// We consider a connection pingable even if it's not connected because that's
+// how a TCP connection is kicked into reconnecting on the active side.
+bool P2PTransportChannel::IsPingable(Connection* conn, uint32_t now) {
   const Candidate& remote = conn->remote_candidate();
   // We should never get this far with an empty remote ufrag.
   ASSERT(!remote.username().empty());
@@ -1171,49 +1227,71 @@ bool P2PTransportChannel::IsPingable(Connection* conn) {
     return false;
   }
 
-  // An unconnected connection cannot be written to at all, so pinging is out
-  // of the question.
-  if (!conn->connected())
+  // An never connected connection cannot be written to at all, so pinging is
+  // out of the question. However, if it has become WRITABLE, it is in the
+  // reconnecting state so ping is needed.
+  if (!conn->connected() && !conn->writable()) {
     return false;
-
-  if (writable()) {
-    // If we are writable, then we only want to ping connections that could be
-    // better than this one, i.e., the ones that were not pruned.
-    return (conn->write_state() != Connection::STATE_WRITE_TIMEOUT);
-  } else {
-    // If we are not writable, then we need to try everything that might work.
-    // This includes both connections that do not have write timeout as well as
-    // ones that do not have read timeout.  A connection could be readable but
-    // be in write-timeout if we pruned it before.  Since the other side is
-    // still pinging it, it very well might still work.
-    return (conn->write_state() != Connection::STATE_WRITE_TIMEOUT) ||
-           (conn->read_state() != Connection::STATE_READ_TIMEOUT);
   }
+
+  // If the channel is weakly connected, ping all connections.
+  if (weak()) {
+    return true;
+  }
+
+  // Always ping active connections regardless whether the channel is completed
+  // or not, but backup connections are pinged at a slower rate.
+  if (IsBackupConnection(conn)) {
+    return (now >= conn->last_ping_response_received() +
+                       backup_connection_ping_interval_);
+  }
+  return conn->active();
 }
 
 // Returns the next pingable connection to ping.  This will be the oldest
-// pingable connection unless we have a writable connection that is past the
-// maximum acceptable ping delay.
+// pingable connection unless we have a connected, writable connection that is
+// past the maximum acceptable ping delay. When reconnecting a TCP connection,
+// the best connection is disconnected, although still WRITABLE while
+// reconnecting. The newly created connection should be selected as the ping
+// target to become writable instead. See the big comment in CompareConnections.
 Connection* P2PTransportChannel::FindNextPingableConnection() {
-  uint32 now = rtc::Time();
-  if (best_connection_ &&
-      (best_connection_->write_state() == Connection::STATE_WRITABLE) &&
-      (best_connection_->last_ping_sent()
-       + MAX_CURRENT_WRITABLE_DELAY <= now)) {
+  uint32_t now = rtc::Time();
+  if (best_connection_ && best_connection_->connected() &&
+      best_connection_->writable() &&
+      (best_connection_->last_ping_sent() + MAX_CURRENT_STRONG_DELAY <= now)) {
     return best_connection_;
   }
 
-  Connection* oldest_conn = NULL;
-  uint32 oldest_time = 0xFFFFFFFF;
-  for (uint32 i = 0; i < connections_.size(); ++i) {
-    if (IsPingable(connections_[i])) {
-      if (connections_[i]->last_ping_sent() < oldest_time) {
-        oldest_time = connections_[i]->last_ping_sent();
-        oldest_conn = connections_[i];
-      }
+  // First, find "triggered checks".  We ping first those connections
+  // that have received a ping but have not sent a ping since receiving
+  // it (last_received_ping > last_sent_ping).  But we shouldn't do
+  // triggered checks if the connection is already writable.
+  Connection* oldest_needing_triggered_check = nullptr;
+  Connection* oldest = nullptr;
+  for (Connection* conn : connections_) {
+    if (!IsPingable(conn, now)) {
+      continue;
+    }
+    bool needs_triggered_check =
+        (!conn->writable() &&
+         conn->last_ping_received() > conn->last_ping_sent());
+    if (needs_triggered_check &&
+        (!oldest_needing_triggered_check ||
+         (conn->last_ping_received() <
+          oldest_needing_triggered_check->last_ping_received()))) {
+      oldest_needing_triggered_check = conn;
+    }
+    if (!oldest || (conn->last_ping_sent() < oldest->last_ping_sent())) {
+      oldest = conn;
     }
   }
-  return oldest_conn;
+
+  if (oldest_needing_triggered_check) {
+    LOG(LS_INFO) << "Selecting connection for triggered check: " <<
+        oldest_needing_triggered_check->ToString();
+    return oldest_needing_triggered_check;
+  }
+  return oldest;
 }
 
 // Apart from sending ping from |conn| this method also updates
@@ -1230,18 +1308,16 @@ Connection* P2PTransportChannel::FindNextPingableConnection() {
 //      b.2) |conn| is writable.
 void P2PTransportChannel::PingConnection(Connection* conn) {
   bool use_candidate = false;
-  if (protocol_type_ == ICEPROTO_RFC5245) {
-    if (remote_ice_mode_ == ICEMODE_FULL && ice_role_ == ICEROLE_CONTROLLING) {
-      use_candidate = (conn == best_connection_) ||
-                      (best_connection_ == NULL) ||
-                      (!best_connection_->writable()) ||
-                      (conn->priority() > best_connection_->priority());
-    } else if (remote_ice_mode_ == ICEMODE_LITE && conn == best_connection_) {
-      use_candidate = best_connection_->writable();
-    }
+  if (remote_ice_mode_ == ICEMODE_FULL && ice_role_ == ICEROLE_CONTROLLING) {
+    use_candidate = (conn == best_connection_) || (best_connection_ == NULL) ||
+                    (!best_connection_->writable()) ||
+                    (conn->priority() > best_connection_->priority());
+  } else if (remote_ice_mode_ == ICEMODE_LITE && conn == best_connection_) {
+    use_candidate = best_connection_->writable();
   }
   conn->set_use_candidate_attr(use_candidate);
-  conn->Ping(rtc::Time());
+  last_ping_sent_ms_ = rtc::Time();
+  conn->Ping(last_ping_sent_ms_);
 }
 
 // When a connection's state changes, we need to figure out who to use as
@@ -1251,13 +1327,23 @@ void P2PTransportChannel::OnConnectionStateChange(Connection* connection) {
 
   // Update the best connection if the state change is from pending best
   // connection and role is controlled.
-  if (protocol_type_ == ICEPROTO_RFC5245 && ice_role_ == ICEROLE_CONTROLLED) {
+  if (ice_role_ == ICEROLE_CONTROLLED) {
     if (connection == pending_best_connection_ && connection->writable()) {
       pending_best_connection_ = NULL;
+      LOG(LS_INFO) << "Switching best connection on controlled side"
+                   << " because it's now writable: " << connection->ToString();
       SwitchBestConnectionTo(connection);
     }
   }
 
+  // May stop the allocator session when at least one connection becomes
+  // strongly connected after starting to get ports. It is not enough to check
+  // that the connection becomes weakly connected because the connection may be
+  // changing from (writable, receiving) to (writable, not receiving).
+  if (!connection->weak()) {
+    MaybeStopPortAllocatorSessions();
+  }
+
   // We have to unroll the stack before doing this because we may be changing
   // the state of connections while sorting.
   RequestSort();
@@ -1290,10 +1376,14 @@ void P2PTransportChannel::OnConnectionDestroyed(Connection* connection) {
   // Since this connection is no longer an option, we can just set best to NULL
   // and re-choose a best assuming that there was no best connection.
   if (best_connection_ == connection) {
+    LOG(LS_INFO) << "Best connection destroyed.  Will choose a new one.";
     SwitchBestConnectionTo(NULL);
     RequestSort();
   }
 
+  UpdateState();
+  // SignalConnectionRemoved should be called after the channel state is
+  // updated because the receiver of the event may access the channel state.
   SignalConnectionRemoved(this);
 }
 
@@ -1313,9 +1403,10 @@ void P2PTransportChannel::OnPortDestroyed(PortInterface* port) {
 }
 
 // We data is available, let listeners know
-void P2PTransportChannel::OnReadPacket(
-    Connection *connection, const char *data, size_t len,
-    const rtc::PacketTime& packet_time) {
+void P2PTransportChannel::OnReadPacket(Connection* connection,
+                                       const char* data,
+                                       size_t len,
+                                       const rtc::PacketTime& packet_time) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
   // Do not deliver, if packet doesn't belong to the correct transport channel.
@@ -1324,6 +1415,19 @@ void P2PTransportChannel::OnReadPacket(
 
   // Let the client know of an incoming packet
   SignalReadPacket(this, data, len, packet_time, 0);
+
+  // May need to switch the sending connection based on the receiving media path
+  // if this is the controlled side.
+  if (ice_role_ == ICEROLE_CONTROLLED && !best_nominated_connection() &&
+      connection->writable() && best_connection_ != connection) {
+    SwitchBestConnectionTo(connection);
+  }
+}
+
+void P2PTransportChannel::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  ASSERT(worker_thread_ == rtc::Thread::Current());
+
+  SignalSentPacket(this, sent_packet);
 }
 
 void P2PTransportChannel::OnReadyToSend(Connection* connection) {
diff --git a/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.h b/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.h
index 78c9528eb0..f2e9315343 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel.h
@@ -34,6 +34,20 @@
 
 namespace cricket {
 
+extern const uint32_t WEAK_PING_DELAY;
+
+struct IceParameters {
+  std::string ufrag;
+  std::string pwd;
+  IceParameters(const std::string& ice_ufrag, const std::string& ice_pwd)
+      : ufrag(ice_ufrag), pwd(ice_pwd) {}
+
+  bool operator==(const IceParameters& other) {
+    return ufrag == other.ufrag && pwd == other.pwd;
+  }
+  bool operator!=(const IceParameters& other) { return !(*this == other); }
+};
+
 // Adds the port on which the candidate originated.
 class RemoteCandidate : public Candidate {
  public:
@@ -51,140 +65,153 @@ class RemoteCandidate : public Candidate {
 class P2PTransportChannel : public TransportChannelImpl,
                             public rtc::MessageHandler {
  public:
-  P2PTransportChannel(const std::string& content_name,
+  P2PTransportChannel(const std::string& transport_name,
                       int component,
                       P2PTransport* transport,
-                      PortAllocator *allocator);
+                      PortAllocator* allocator);
   virtual ~P2PTransportChannel();
 
   // From TransportChannelImpl:
-  virtual Transport* GetTransport() { return transport_; }
-  virtual TransportChannelState GetState() const;
-  virtual void SetIceRole(IceRole role);
-  virtual IceRole GetIceRole() const { return ice_role_; }
-  virtual void SetIceTiebreaker(uint64 tiebreaker);
-  virtual bool GetIceProtocolType(IceProtocolType* type) const;
-  virtual void SetIceProtocolType(IceProtocolType type);
-  virtual void SetIceCredentials(const std::string& ice_ufrag,
-                                 const std::string& ice_pwd);
-  virtual void SetRemoteIceCredentials(const std::string& ice_ufrag,
-                                       const std::string& ice_pwd);
-  virtual void SetRemoteIceMode(IceMode mode);
-  virtual void Connect();
-  virtual void Reset();
-  virtual void OnSignalingReady();
-  virtual void OnCandidate(const Candidate& candidate);
+  Transport* GetTransport() override { return transport_; }
+  TransportChannelState GetState() const override;
+  void SetIceRole(IceRole role) override;
+  IceRole GetIceRole() const override { return ice_role_; }
+  void SetIceTiebreaker(uint64_t tiebreaker) override;
+  void SetIceCredentials(const std::string& ice_ufrag,
+                         const std::string& ice_pwd) override;
+  void SetRemoteIceCredentials(const std::string& ice_ufrag,
+                               const std::string& ice_pwd) override;
+  void SetRemoteIceMode(IceMode mode) override;
+  void Connect() override;
+  void MaybeStartGathering() override;
+  IceGatheringState gathering_state() const override {
+    return gathering_state_;
+  }
+  void AddRemoteCandidate(const Candidate& candidate) override;
+  // Sets the receiving timeout and gather_continually.
+  // This also sets the check_receiving_delay proportionally.
+  void SetIceConfig(const IceConfig& config) override;
 
   // From TransportChannel:
-  virtual int SendPacket(const char *data, size_t len,
-                         const rtc::PacketOptions& options, int flags);
-  virtual int SetOption(rtc::Socket::Option opt, int value);
-  virtual bool GetOption(rtc::Socket::Option opt, int* value);
-  virtual int GetError() { return error_; }
-  virtual bool GetStats(std::vector<ConnectionInfo>* stats);
+  int SendPacket(const char* data,
+                 size_t len,
+                 const rtc::PacketOptions& options,
+                 int flags) override;
+  int SetOption(rtc::Socket::Option opt, int value) override;
+  bool GetOption(rtc::Socket::Option opt, int* value) override;
+  int GetError() override { return error_; }
+  bool GetStats(std::vector<ConnectionInfo>* stats) override;
 
   const Connection* best_connection() const { return best_connection_; }
   void set_incoming_only(bool value) { incoming_only_ = value; }
 
   // Note: This is only for testing purpose.
   // |ports_| should not be changed from outside.
-  const std::vector<PortInterface *>& ports() { return ports_; }
+  const std::vector<PortInterface*>& ports() { return ports_; }
 
   IceMode remote_ice_mode() const { return remote_ice_mode_; }
 
   // DTLS methods.
-  virtual bool IsDtlsActive() const { return false; }
+  bool IsDtlsActive() const override { return false; }
 
   // Default implementation.
-  virtual bool GetSslRole(rtc::SSLRole* role) const {
-    return false;
-  }
+  bool GetSslRole(rtc::SSLRole* role) const override { return false; }
 
-  virtual bool SetSslRole(rtc::SSLRole role) {
-    return false;
-  }
+  bool SetSslRole(rtc::SSLRole role) override { return false; }
 
   // Set up the ciphers to use for DTLS-SRTP.
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
+  bool SetSrtpCryptoSuites(const std::vector<int>& ciphers) override {
     return false;
   }
 
   // Find out which DTLS-SRTP cipher was negotiated.
-  virtual bool GetSrtpCipher(std::string* cipher) {
-    return false;
-  }
+  bool GetSrtpCryptoSuite(int* cipher) override { return false; }
 
   // Find out which DTLS cipher was negotiated.
-  virtual bool GetSslCipher(std::string* cipher) {
-    return false;
+  bool GetSslCipherSuite(int* cipher) override { return false; }
+
+  // Returns null because the channel is not encrypted by default.
+  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override {
+    return nullptr;
   }
 
-  // Returns false because the channel is not encrypted by default.
-  virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const {
-    return false;
-  }
-
-  virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const {
+  bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert) const override {
     return false;
   }
 
   // Allows key material to be extracted for external encryption.
-  virtual bool ExportKeyingMaterial(
-      const std::string& label,
-      const uint8* context,
-      size_t context_len,
-      bool use_context,
-      uint8* result,
-      size_t result_len) {
+  bool ExportKeyingMaterial(const std::string& label,
+                            const uint8_t* context,
+                            size_t context_len,
+                            bool use_context,
+                            uint8_t* result,
+                            size_t result_len) override {
     return false;
   }
 
-  virtual bool SetLocalIdentity(rtc::SSLIdentity* identity) {
+  bool SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override {
     return false;
   }
 
   // Set DTLS Remote fingerprint. Must be after local identity set.
-  virtual bool SetRemoteFingerprint(
-    const std::string& digest_alg,
-    const uint8* digest,
-    size_t digest_len) {
+  bool SetRemoteFingerprint(const std::string& digest_alg,
+                            const uint8_t* digest,
+                            size_t digest_len) override {
     return false;
   }
 
+  int receiving_timeout() const { return receiving_timeout_; }
+  int check_receiving_delay() const { return check_receiving_delay_; }
+
   // Helper method used only in unittest.
   rtc::DiffServCodePoint DefaultDscpValue() const;
 
   // Public for unit tests.
   Connection* FindNextPingableConnection();
 
- private:
-  rtc::Thread* thread() { return worker_thread_; }
+  // Public for unit tests.
+  const std::vector<Connection*>& connections() const { return connections_; }
+
+  // Public for unit tests.
   PortAllocatorSession* allocator_session() {
     return allocator_sessions_.back();
   }
 
-  void Allocate();
+  // Public for unit tests.
+  const std::vector<RemoteCandidate>& remote_candidates() const {
+    return remote_candidates_;
+  }
+
+ private:
+  rtc::Thread* thread() { return worker_thread_; }
+  bool IsGettingPorts() { return allocator_session()->IsGettingPorts(); }
+
+  // A transport channel is weak if the current best connection is either
+  // not receiving or not writable, or if there is no best connection at all.
+  bool weak() const;
   void UpdateConnectionStates();
   void RequestSort();
   void SortConnections();
   void SwitchBestConnectionTo(Connection* conn);
-  void UpdateChannelState();
-  void HandleWritable();
-  void HandleNotWritable();
+  void UpdateState();
   void HandleAllTimedOut();
+  void MaybeStopPortAllocatorSessions();
+  TransportChannelState ComputeState() const;
 
   Connection* GetBestConnectionOnNetwork(rtc::Network* network) const;
-  bool CreateConnections(const Candidate &remote_candidate,
-                         PortInterface* origin_port, bool readable);
-  bool CreateConnection(PortInterface* port, const Candidate& remote_candidate,
-                        PortInterface* origin_port, bool readable);
+  bool CreateConnections(const Candidate& remote_candidate,
+                         PortInterface* origin_port);
+  bool CreateConnection(PortInterface* port,
+                        const Candidate& remote_candidate,
+                        PortInterface* origin_port);
   bool FindConnection(cricket::Connection* connection) const;
 
-  uint32 GetRemoteCandidateGeneration(const Candidate& candidate);
+  uint32_t GetRemoteCandidateGeneration(const Candidate& candidate);
   bool IsDuplicateRemoteCandidate(const Candidate& candidate);
   void RememberRemoteCandidate(const Candidate& remote_candidate,
                                PortInterface* origin_port);
-  bool IsPingable(Connection* conn);
+  bool IsPingable(Connection* conn, uint32_t now);
   void PingConnection(Connection* conn);
   void AddAllocatorSession(PortAllocatorSession* session);
   void AddConnection(Connection* connection);
@@ -205,20 +232,42 @@ class P2PTransportChannel : public TransportChannelImpl,
   void OnConnectionStateChange(Connection* connection);
   void OnReadPacket(Connection *connection, const char *data, size_t len,
                     const rtc::PacketTime& packet_time);
+  void OnSentPacket(const rtc::SentPacket& sent_packet);
   void OnReadyToSend(Connection* connection);
   void OnConnectionDestroyed(Connection *connection);
 
-  void OnUseCandidate(Connection* conn);
+  void OnNominated(Connection* conn);
 
-  virtual void OnMessage(rtc::Message *pmsg);
+  void OnMessage(rtc::Message* pmsg) override;
   void OnSort();
-  void OnPing();
+  void OnCheckAndPing();
+
+  void PruneConnections();
+  Connection* best_nominated_connection() const;
+  bool IsBackupConnection(Connection* conn) const;
+
+  // Returns the latest remote ICE parameters or nullptr if there are no remote
+  // ICE parameters yet.
+  IceParameters* remote_ice() {
+    return remote_ice_parameters_.empty() ? nullptr
+                                          : &remote_ice_parameters_.back();
+  }
+  // Returns the remote IceParameters and generation that match |ufrag|
+  // if found, and returns nullptr otherwise.
+  const IceParameters* FindRemoteIceFromUfrag(const std::string& ufrag,
+                                              uint32_t* generation);
+  // Returns the index of the latest remote ICE parameters, or 0 if no remote
+  // ICE parameters have been received.
+  uint32_t remote_ice_generation() {
+    return remote_ice_parameters_.empty()
+               ? 0
+               : static_cast<uint32_t>(remote_ice_parameters_.size() - 1);
+  }
 
   P2PTransport* transport_;
-  PortAllocator *allocator_;
-  rtc::Thread *worker_thread_;
+  PortAllocator* allocator_;
+  rtc::Thread* worker_thread_;
   bool incoming_only_;
-  bool waiting_for_signaling_;
   int error_;
   std::vector<PortAllocatorSession*> allocator_sessions_;
   std::vector<PortInterface *> ports_;
@@ -229,20 +278,26 @@ class P2PTransportChannel : public TransportChannelImpl,
   Connection* pending_best_connection_;
   std::vector<RemoteCandidate> remote_candidates_;
   bool sort_dirty_;  // indicates whether another sort is needed right now
-  bool was_writable_;
+  bool had_connection_ = false;  // if connections_ has ever been nonempty
   typedef std::map<rtc::Socket::Option, int> OptionMap;
   OptionMap options_;
   std::string ice_ufrag_;
   std::string ice_pwd_;
-  std::string remote_ice_ufrag_;
-  std::string remote_ice_pwd_;
-  IceProtocolType protocol_type_;
+  std::vector<IceParameters> remote_ice_parameters_;
   IceMode remote_ice_mode_;
   IceRole ice_role_;
-  uint64 tiebreaker_;
-  uint32 remote_candidate_generation_;
+  uint64_t tiebreaker_;
+  IceGatheringState gathering_state_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(P2PTransportChannel);
+  int check_receiving_delay_;
+  int receiving_timeout_;
+  int backup_connection_ping_interval_;
+  uint32_t last_ping_sent_ms_ = 0;
+  bool gather_continually_ = false;
+  int weak_ping_delay_ = WEAK_PING_DELAY;
+  TransportChannelState state_ = TransportChannelState::STATE_INIT;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel_unittest.cc
index d70ecd86d7..90ddd43714 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/p2ptransportchannel_unittest.cc
@@ -13,6 +13,7 @@
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/fakenetwork.h"
 #include "webrtc/base/firewallsocketserver.h"
@@ -31,8 +32,6 @@
 using cricket::kDefaultPortAllocatorFlags;
 using cricket::kMinimumStepDelay;
 using cricket::kDefaultStepDelay;
-using cricket::PORTALLOCATOR_ENABLE_BUNDLE;
-using cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG;
 using cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET;
 using cricket::ServerAddresses;
 using rtc::SocketAddress;
@@ -94,13 +93,23 @@ static const char* kIcePwd[4] = {"TESTICEPWD00000000000000",
                                  "TESTICEPWD00000000000002",
                                  "TESTICEPWD00000000000003"};
 
-static const uint64 kTiebreaker1 = 11111;
-static const uint64 kTiebreaker2 = 22222;
+static const uint64_t kTiebreaker1 = 11111;
+static const uint64_t kTiebreaker2 = 22222;
 
 enum {
   MSG_CANDIDATE
 };
 
+static cricket::IceConfig CreateIceConfig(int receiving_timeout_ms,
+                                          bool gather_continually,
+                                          int backup_ping_interval = -1) {
+  cricket::IceConfig config;
+  config.receiving_timeout_ms = receiving_timeout_ms;
+  config.gather_continually = gather_continually;
+  config.backup_connection_ping_interval = backup_ping_interval;
+  return config;
+}
+
 // This test simulates 2 P2P endpoints that want to establish connectivity
 // with each other over various network topologies and conditions, which can be
 // specified in each individial test.
@@ -177,6 +186,7 @@ class P2PTransportChannelTestBase : public testing::Test,
           local_type2(lt2), local_proto2(lp2), remote_type2(rt2),
           remote_proto2(rp2), connect_wait(wait) {
     }
+
     std::string local_type;
     std::string local_proto;
     std::string remote_type;
@@ -217,8 +227,7 @@ class P2PTransportChannelTestBase : public testing::Test,
         : role_(cricket::ICEROLE_UNKNOWN),
           tiebreaker_(0),
           role_conflict_(false),
-          save_candidates_(false),
-          protocol_type_(cricket::ICEPROTO_GOOGLE) {}
+          save_candidates_(false) {}
     bool HasChannel(cricket::TransportChannel* ch) {
       return (ch == cd1_.ch_.get() || ch == cd2_.ch_.get());
     }
@@ -232,15 +241,11 @@ class P2PTransportChannelTestBase : public testing::Test,
 
     void SetIceRole(cricket::IceRole role) { role_ = role; }
     cricket::IceRole ice_role() { return role_; }
-    void SetIceProtocolType(cricket::IceProtocolType type) {
-      protocol_type_ = type;
-    }
-    cricket::IceProtocolType protocol_type() { return protocol_type_; }
-    void SetIceTiebreaker(uint64 tiebreaker) { tiebreaker_ = tiebreaker; }
-    uint64 GetIceTiebreaker() { return tiebreaker_; }
+    void SetIceTiebreaker(uint64_t tiebreaker) { tiebreaker_ = tiebreaker; }
+    uint64_t GetIceTiebreaker() { return tiebreaker_; }
     void OnRoleConflict(bool role_conflict) { role_conflict_ = role_conflict; }
     bool role_conflict() { return role_conflict_; }
-    void SetAllocationStepDelay(uint32 delay) {
+    void SetAllocationStepDelay(uint32_t delay) {
       allocator_->set_step_delay(delay);
     }
     void SetAllowTcpListen(bool allow_tcp_listen) {
@@ -252,10 +257,9 @@ class P2PTransportChannelTestBase : public testing::Test,
     ChannelData cd1_;
     ChannelData cd2_;
     cricket::IceRole role_;
-    uint64 tiebreaker_;
+    uint64_t tiebreaker_;
     bool role_conflict_;
     bool save_candidates_;
-    cricket::IceProtocolType protocol_type_;
     std::vector<CandidateData*> saved_candidates_;
   };
 
@@ -284,15 +288,6 @@ class P2PTransportChannelTestBase : public testing::Test,
       std::string ice_pwd_ep1_cd2_ch = kIcePwd[2];
       std::string ice_ufrag_ep2_cd2_ch = kIceUfrag[3];
       std::string ice_pwd_ep2_cd2_ch = kIcePwd[3];
-      // In BUNDLE each endpoint must share common ICE credentials.
-      if (ep1_.allocator_->flags() & PORTALLOCATOR_ENABLE_BUNDLE) {
-        ice_ufrag_ep1_cd2_ch = ice_ufrag_ep1_cd1_ch;
-        ice_pwd_ep1_cd2_ch = ice_pwd_ep1_cd1_ch;
-      }
-      if (ep2_.allocator_->flags() & PORTALLOCATOR_ENABLE_BUNDLE) {
-        ice_ufrag_ep2_cd2_ch = ice_ufrag_ep2_cd1_ch;
-        ice_pwd_ep2_cd2_ch = ice_pwd_ep2_cd1_ch;
-      }
       ep1_.cd2_.ch_.reset(CreateChannel(
           0, cricket::ICE_CANDIDATE_COMPONENT_DEFAULT,
           ice_ufrag_ep1_cd2_ch, ice_pwd_ep1_cd2_ch,
@@ -312,15 +307,12 @@ class P2PTransportChannelTestBase : public testing::Test,
       const std::string& remote_ice_pwd) {
     cricket::P2PTransportChannel* channel = new cricket::P2PTransportChannel(
         "test content name", component, NULL, GetAllocator(endpoint));
-    channel->SignalRequestSignaling.connect(
-        this, &P2PTransportChannelTestBase::OnChannelRequestSignaling);
-    channel->SignalCandidateReady.connect(this,
-        &P2PTransportChannelTestBase::OnCandidate);
+    channel->SignalCandidateGathered.connect(
+        this, &P2PTransportChannelTestBase::OnCandidate);
     channel->SignalReadPacket.connect(
         this, &P2PTransportChannelTestBase::OnReadPacket);
     channel->SignalRoleConflict.connect(
         this, &P2PTransportChannelTestBase::OnRoleConflict);
-    channel->SetIceProtocolType(GetEndpoint(endpoint)->protocol_type());
     channel->SetIceCredentials(local_ice_ufrag, local_ice_pwd);
     if (clear_remote_candidates_ufrag_pwd_) {
       // This only needs to be set if we're clearing them from the
@@ -330,6 +322,7 @@ class P2PTransportChannelTestBase : public testing::Test,
     channel->SetIceRole(GetEndpoint(endpoint)->ice_role());
     channel->SetIceTiebreaker(GetEndpoint(endpoint)->GetIceTiebreaker());
     channel->Connect();
+    channel->MaybeStartGathering();
     return channel;
   }
   void DestroyChannels() {
@@ -388,35 +381,127 @@ class P2PTransportChannelTestBase : public testing::Test,
   void SetAllocatorFlags(int endpoint, int flags) {
     GetAllocator(endpoint)->set_flags(flags);
   }
-  void SetIceProtocol(int endpoint, cricket::IceProtocolType type) {
-    GetEndpoint(endpoint)->SetIceProtocolType(type);
-  }
   void SetIceRole(int endpoint, cricket::IceRole role) {
     GetEndpoint(endpoint)->SetIceRole(role);
   }
-  void SetIceTiebreaker(int endpoint, uint64 tiebreaker) {
+  void SetIceTiebreaker(int endpoint, uint64_t tiebreaker) {
     GetEndpoint(endpoint)->SetIceTiebreaker(tiebreaker);
   }
   bool GetRoleConflict(int endpoint) {
     return GetEndpoint(endpoint)->role_conflict();
   }
-  void SetAllocationStepDelay(int endpoint, uint32 delay) {
+  void SetAllocationStepDelay(int endpoint, uint32_t delay) {
     return GetEndpoint(endpoint)->SetAllocationStepDelay(delay);
   }
   void SetAllowTcpListen(int endpoint, bool allow_tcp_listen) {
     return GetEndpoint(endpoint)->SetAllowTcpListen(allow_tcp_listen);
   }
+  bool IsLocalToPrflxOrTheReverse(const Result& expected) {
+    return (
+        (expected.local_type == "local" && expected.remote_type == "prflx") ||
+        (expected.local_type == "prflx" && expected.remote_type == "local"));
+  }
+
+  // Return true if the approprite parts of the expected Result, based
+  // on the local and remote candidate of ep1_ch1, match.  This can be
+  // used in an EXPECT_TRUE_WAIT.
+  bool CheckCandidate1(const Result& expected) {
+    const std::string& local_type = LocalCandidate(ep1_ch1())->type();
+    const std::string& local_proto = LocalCandidate(ep1_ch1())->protocol();
+    const std::string& remote_type = RemoteCandidate(ep1_ch1())->type();
+    const std::string& remote_proto = RemoteCandidate(ep1_ch1())->protocol();
+    return ((local_proto == expected.local_proto &&
+             remote_proto == expected.remote_proto) &&
+            ((local_type == expected.local_type &&
+              remote_type == expected.remote_type) ||
+             // Sometimes we expect local -> prflx or prflx -> local
+             // and instead get prflx -> local or local -> prflx, and
+             // that's OK.
+             (IsLocalToPrflxOrTheReverse(expected) &&
+              local_type == expected.remote_type &&
+              remote_type == expected.local_type)));
+  }
+
+  // EXPECT_EQ on the approprite parts of the expected Result, based
+  // on the local and remote candidate of ep1_ch1.  This is like
+  // CheckCandidate1, except that it will provide more detail about
+  // what didn't match.
+  void ExpectCandidate1(const Result& expected) {
+    if (CheckCandidate1(expected)) {
+      return;
+    }
+
+    const std::string& local_type = LocalCandidate(ep1_ch1())->type();
+    const std::string& local_proto = LocalCandidate(ep1_ch1())->protocol();
+    const std::string& remote_type = RemoteCandidate(ep1_ch1())->type();
+    const std::string& remote_proto = RemoteCandidate(ep1_ch1())->protocol();
+    EXPECT_EQ(expected.local_type, local_type);
+    EXPECT_EQ(expected.remote_type, remote_type);
+    EXPECT_EQ(expected.local_proto, local_proto);
+    EXPECT_EQ(expected.remote_proto, remote_proto);
+  }
+
+  // Return true if the approprite parts of the expected Result, based
+  // on the local and remote candidate of ep2_ch1, match.  This can be
+  // used in an EXPECT_TRUE_WAIT.
+  bool CheckCandidate2(const Result& expected) {
+    const std::string& local_type = LocalCandidate(ep2_ch1())->type();
+    // const std::string& remote_type = RemoteCandidate(ep2_ch1())->type();
+    const std::string& local_proto = LocalCandidate(ep2_ch1())->protocol();
+    const std::string& remote_proto = RemoteCandidate(ep2_ch1())->protocol();
+    // Removed remote_type comparision aginst best connection remote
+    // candidate. This is done to handle remote type discrepancy from
+    // local to stun based on the test type.
+    // For example in case of Open -> NAT, ep2 channels will have LULU
+    // and in other cases like NAT -> NAT it will be LUSU. To avoid these
+    // mismatches and we are doing comparision in different way.
+    // i.e. when don't match its remote type is either local or stun.
+    // TODO(ronghuawu): Refine the test criteria.
+    // https://code.google.com/p/webrtc/issues/detail?id=1953
+    return ((local_proto == expected.local_proto2 &&
+             remote_proto == expected.remote_proto2) &&
+            (local_type == expected.local_type2 ||
+             // Sometimes we expect local -> prflx or prflx -> local
+             // and instead get prflx -> local or local -> prflx, and
+             // that's OK.
+             (IsLocalToPrflxOrTheReverse(expected) &&
+              local_type == expected.remote_type2)));
+  }
+
+  // EXPECT_EQ on the approprite parts of the expected Result, based
+  // on the local and remote candidate of ep2_ch1.  This is like
+  // CheckCandidate2, except that it will provide more detail about
+  // what didn't match.
+  void ExpectCandidate2(const Result& expected) {
+    if (CheckCandidate2(expected)) {
+      return;
+    }
+
+    const std::string& local_type = LocalCandidate(ep2_ch1())->type();
+    const std::string& local_proto = LocalCandidate(ep2_ch1())->protocol();
+    const std::string& remote_type = RemoteCandidate(ep2_ch1())->type();
+    EXPECT_EQ(expected.local_proto2, local_proto);
+    EXPECT_EQ(expected.remote_proto2, remote_type);
+    EXPECT_EQ(expected.local_type2, local_type);
+    if (remote_type != expected.remote_type2) {
+      EXPECT_TRUE(expected.remote_type2 == cricket::LOCAL_PORT_TYPE ||
+                  expected.remote_type2 == cricket::STUN_PORT_TYPE);
+      EXPECT_TRUE(remote_type == cricket::LOCAL_PORT_TYPE ||
+                  remote_type == cricket::STUN_PORT_TYPE ||
+                  remote_type == cricket::PRFLX_PORT_TYPE);
+    }
+  }
 
   void Test(const Result& expected) {
-    int32 connect_start = rtc::Time(), connect_time;
+    int32_t connect_start = rtc::Time(), connect_time;
 
     // Create the channels and wait for them to connect.
     CreateChannels(1);
     EXPECT_TRUE_WAIT_MARGIN(ep1_ch1() != NULL &&
                             ep2_ch1() != NULL &&
-                            ep1_ch1()->readable() &&
+                            ep1_ch1()->receiving() &&
                             ep1_ch1()->writable() &&
-                            ep2_ch1()->readable() &&
+                            ep2_ch1()->receiving() &&
                             ep2_ch1()->writable(),
                             expected.connect_wait,
                             1000);
@@ -432,57 +517,22 @@ class P2PTransportChannelTestBase : public testing::Test,
     // This may take up to 2 seconds.
     if (ep1_ch1()->best_connection() &&
         ep2_ch1()->best_connection()) {
-      int32 converge_start = rtc::Time(), converge_time;
+      int32_t converge_start = rtc::Time(), converge_time;
       int converge_wait = 2000;
-      EXPECT_TRUE_WAIT_MARGIN(
-          LocalCandidate(ep1_ch1())->type() == expected.local_type &&
-          LocalCandidate(ep1_ch1())->protocol() == expected.local_proto &&
-          RemoteCandidate(ep1_ch1())->type() == expected.remote_type &&
-          RemoteCandidate(ep1_ch1())->protocol() == expected.remote_proto,
-          converge_wait,
-          converge_wait);
-
+      EXPECT_TRUE_WAIT_MARGIN(CheckCandidate1(expected), converge_wait,
+                              converge_wait);
       // Also do EXPECT_EQ on each part so that failures are more verbose.
-      EXPECT_EQ(expected.local_type, LocalCandidate(ep1_ch1())->type());
-      EXPECT_EQ(expected.local_proto, LocalCandidate(ep1_ch1())->protocol());
-      EXPECT_EQ(expected.remote_type, RemoteCandidate(ep1_ch1())->type());
-      EXPECT_EQ(expected.remote_proto, RemoteCandidate(ep1_ch1())->protocol());
+      ExpectCandidate1(expected);
 
       // Verifying remote channel best connection information. This is done
       // only for the RFC 5245 as controlled agent will use USE-CANDIDATE
       // from controlling (ep1) agent. We can easily predict from EP1 result
       // matrix.
-      if (ep2_.protocol_type_ == cricket::ICEPROTO_RFC5245) {
-        // Checking for best connection candidates information at remote.
-        EXPECT_TRUE_WAIT(
-            LocalCandidate(ep2_ch1())->type() == expected.local_type2 &&
-            LocalCandidate(ep2_ch1())->protocol() == expected.local_proto2 &&
-            RemoteCandidate(ep2_ch1())->protocol() == expected.remote_proto2,
-            kDefaultTimeout);
 
-        // For verbose
-        EXPECT_EQ(expected.local_type2, LocalCandidate(ep2_ch1())->type());
-        EXPECT_EQ(expected.local_proto2, LocalCandidate(ep2_ch1())->protocol());
-        EXPECT_EQ(expected.remote_proto2,
-                  RemoteCandidate(ep2_ch1())->protocol());
-        // Removed remote_type comparision aginst best connection remote
-        // candidate. This is done to handle remote type discrepancy from
-        // local to stun based on the test type.
-        // For example in case of Open -> NAT, ep2 channels will have LULU
-        // and in other cases like NAT -> NAT it will be LUSU. To avoid these
-        // mismatches and we are doing comparision in different way.
-        // i.e. when don't match its remote type is either local or stun.
-        // TODO(ronghuawu): Refine the test criteria.
-        // https://code.google.com/p/webrtc/issues/detail?id=1953
-        if (expected.remote_type2 != RemoteCandidate(ep2_ch1())->type()) {
-          EXPECT_TRUE(expected.remote_type2 == cricket::LOCAL_PORT_TYPE ||
-                      expected.remote_type2 == cricket::STUN_PORT_TYPE);
-          EXPECT_TRUE(
-              RemoteCandidate(ep2_ch1())->type() == cricket::LOCAL_PORT_TYPE ||
-              RemoteCandidate(ep2_ch1())->type() == cricket::STUN_PORT_TYPE ||
-              RemoteCandidate(ep2_ch1())->type() == cricket::PRFLX_PORT_TYPE);
-        }
-      }
+      // Checking for best connection candidates information at remote.
+      EXPECT_TRUE_WAIT(CheckCandidate2(expected), kDefaultTimeout);
+      // For verbose
+      ExpectCandidate2(expected);
 
       converge_time = rtc::TimeSince(converge_start);
       if (converge_time < converge_wait) {
@@ -518,16 +568,16 @@ class P2PTransportChannelTestBase : public testing::Test,
     }
   }
 
-  // This test waits for the transport to become readable and writable on both
-  // end points. Once they are, the end points set new local ice credentials to
+  // This test waits for the transport to become receiving and writable on both
+  // end points. Once they are, the end points set new local ice credentials and
   // restart the ice gathering. Finally it waits for the transport to select a
   // new connection using the newly generated ice candidates.
   // Before calling this function the end points must be configured.
   void TestHandleIceUfragPasswordChanged() {
     ep1_ch1()->SetRemoteIceCredentials(kIceUfrag[1], kIcePwd[1]);
     ep2_ch1()->SetRemoteIceCredentials(kIceUfrag[0], kIcePwd[0]);
-    EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                            ep2_ch1()->readable() && ep2_ch1()->writable(),
+    EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                            ep2_ch1()->receiving() && ep2_ch1()->writable(),
                             1000, 1000);
 
     const cricket::Candidate* old_local_candidate1 = LocalCandidate(ep1_ch1());
@@ -539,8 +589,10 @@ class P2PTransportChannelTestBase : public testing::Test,
 
     ep1_ch1()->SetIceCredentials(kIceUfrag[2], kIcePwd[2]);
     ep1_ch1()->SetRemoteIceCredentials(kIceUfrag[3], kIcePwd[3]);
+    ep1_ch1()->MaybeStartGathering();
     ep2_ch1()->SetIceCredentials(kIceUfrag[3], kIcePwd[3]);
     ep2_ch1()->SetRemoteIceCredentials(kIceUfrag[2], kIcePwd[2]);
+    ep2_ch1()->MaybeStartGathering();
 
     EXPECT_TRUE_WAIT_MARGIN(LocalCandidate(ep1_ch1())->generation() !=
                             old_local_candidate1->generation(),
@@ -559,10 +611,8 @@ class P2PTransportChannelTestBase : public testing::Test,
   }
 
   void TestSignalRoleConflict() {
-    SetIceProtocol(0, cricket::ICEPROTO_RFC5245);
     SetIceTiebreaker(0, kTiebreaker1);  // Default EP1 is in controlling state.
 
-    SetIceProtocol(1, cricket::ICEPROTO_RFC5245);
     SetIceRole(1, cricket::ICEROLE_CONTROLLING);
     SetIceTiebreaker(1, kTiebreaker2);
 
@@ -573,9 +623,9 @@ class P2PTransportChannelTestBase : public testing::Test,
     EXPECT_TRUE_WAIT(GetRoleConflict(0), 1000);
     EXPECT_FALSE(GetRoleConflict(1));
 
-    EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
+    EXPECT_TRUE_WAIT(ep1_ch1()->receiving() &&
                      ep1_ch1()->writable() &&
-                     ep2_ch1()->readable() &&
+                     ep2_ch1()->receiving() &&
                      ep2_ch1()->writable(),
                      1000);
 
@@ -585,49 +635,6 @@ class P2PTransportChannelTestBase : public testing::Test,
     TestSendRecv(1);
   }
 
-  void TestHybridConnectivity(cricket::IceProtocolType proto) {
-    AddAddress(0, kPublicAddrs[0]);
-    AddAddress(1, kPublicAddrs[1]);
-
-    SetAllocationStepDelay(0, kMinimumStepDelay);
-    SetAllocationStepDelay(1, kMinimumStepDelay);
-
-    SetIceRole(0, cricket::ICEROLE_CONTROLLING);
-    SetIceProtocol(0, cricket::ICEPROTO_HYBRID);
-    SetIceTiebreaker(0, kTiebreaker1);
-    SetIceRole(1, cricket::ICEROLE_CONTROLLED);
-    SetIceProtocol(1, proto);
-    SetIceTiebreaker(1, kTiebreaker2);
-
-    CreateChannels(1);
-    // When channel is in hybrid and it's controlling agent, channel will
-    // receive ping request from the remote. Hence connection is readable.
-    // Since channel is in hybrid, it will not send any pings, so no writable
-    // connection. Since channel2 is in controlled state, it will not have
-    // any connections which are readable or writable, as it didn't received
-    // pings (or none) with USE-CANDIDATE attribute.
-    EXPECT_TRUE_WAIT(ep1_ch1()->readable(), 1000);
-
-    // Set real protocol type.
-    ep1_ch1()->SetIceProtocolType(proto);
-
-    // Channel should able to send ping requests and connections become writable
-    // in both directions.
-    EXPECT_TRUE_WAIT(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                     ep2_ch1()->readable() && ep2_ch1()->writable(),
-                     1000);
-    EXPECT_TRUE(
-        ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
-        LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
-        RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
-
-    TestSendRecv(1);
-    DestroyChannels();
-  }
-
-  void OnChannelRequestSignaling(cricket::TransportChannelImpl* channel) {
-    channel->OnSignalingReady();
-  }
   // We pass the candidates directly to the other side.
   void OnCandidate(cricket::TransportChannelImpl* ch,
                    const cricket::Candidate& c) {
@@ -645,6 +652,21 @@ class P2PTransportChannelTestBase : public testing::Test,
     GetEndpoint(endpoint)->save_candidates_ = true;
   }
 
+  // Tcp candidate verification has to be done when they are generated.
+  void VerifySavedTcpCandidates(int endpoint, const std::string& tcptype) {
+    for (auto& data : GetEndpoint(endpoint)->saved_candidates_) {
+      EXPECT_EQ(data->candidate.protocol(), cricket::TCP_PROTOCOL_NAME);
+      EXPECT_EQ(data->candidate.tcptype(), tcptype);
+      if (data->candidate.tcptype() == cricket::TCPTYPE_ACTIVE_STR) {
+        EXPECT_EQ(data->candidate.address().port(), cricket::DISCARD_PORT);
+      } else if (data->candidate.tcptype() == cricket::TCPTYPE_PASSIVE_STR) {
+        EXPECT_NE(data->candidate.address().port(), cricket::DISCARD_PORT);
+      } else {
+        FAIL() << "Unknown tcptype: " << data->candidate.tcptype();
+      }
+    }
+  }
+
   void ResumeCandidates(int endpoint) {
     Endpoint* ed = GetEndpoint(endpoint);
     std::vector<CandidateData*>::iterator it = ed->saved_candidates_.begin();
@@ -668,7 +690,7 @@ class P2PTransportChannelTestBase : public testing::Test,
         }
         LOG(LS_INFO) << "Candidate(" << data->channel->component() << "->"
                      << rch->component() << "): " << c.ToString();
-        rch->OnCandidate(c);
+        rch->AddRemoteCandidate(c);
         break;
       }
     }
@@ -803,13 +825,10 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
   static const Result* kMatrixSharedUfrag[NUM_CONFIGS][NUM_CONFIGS];
   static const Result* kMatrixSharedSocketAsGice[NUM_CONFIGS][NUM_CONFIGS];
   static const Result* kMatrixSharedSocketAsIce[NUM_CONFIGS][NUM_CONFIGS];
-  void ConfigureEndpoints(Config config1, Config config2,
-      int allocator_flags1, int allocator_flags2,
-      int delay1, int delay2,
-      cricket::IceProtocolType type) {
-    // Ideally we want to use TURN server for both GICE and ICE, but in case
-    // of GICE, TURN server usage is not producing results reliabally.
-    // TODO(mallinath): Remove Relay and use TURN server for all tests.
+  void ConfigureEndpoints(Config config1,
+                          Config config2,
+                          int allocator_flags1,
+                          int allocator_flags2) {
     ServerAddresses stun_servers;
     stun_servers.insert(kStunAddr);
     GetEndpoint(0)->allocator_.reset(
@@ -823,35 +842,22 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
         rtc::SocketAddress(), rtc::SocketAddress(),
         rtc::SocketAddress()));
 
-    cricket::RelayServerConfig relay_server(cricket::RELAY_GTURN);
-    if (type == cricket::ICEPROTO_RFC5245) {
-      relay_server.type = cricket::RELAY_TURN;
-      relay_server.credentials = kRelayCredentials;
-      relay_server.ports.push_back(cricket::ProtocolAddress(
-          kTurnUdpIntAddr, cricket::PROTO_UDP, false));
-    } else {
-      relay_server.ports.push_back(cricket::ProtocolAddress(
-          kRelayUdpIntAddr, cricket::PROTO_UDP, false));
-      relay_server.ports.push_back(cricket::ProtocolAddress(
-          kRelayTcpIntAddr, cricket::PROTO_TCP, false));
-      relay_server.ports.push_back(cricket::ProtocolAddress(
-          kRelaySslTcpIntAddr, cricket::PROTO_SSLTCP, false));
-    }
-    GetEndpoint(0)->allocator_->AddRelay(relay_server);
-    GetEndpoint(1)->allocator_->AddRelay(relay_server);
+    cricket::RelayServerConfig turn_server(cricket::RELAY_TURN);
+    turn_server.credentials = kRelayCredentials;
+    turn_server.ports.push_back(
+        cricket::ProtocolAddress(kTurnUdpIntAddr, cricket::PROTO_UDP, false));
+    GetEndpoint(0)->allocator_->AddTurnServer(turn_server);
+    GetEndpoint(1)->allocator_->AddTurnServer(turn_server);
 
+    int delay = kMinimumStepDelay;
     ConfigureEndpoint(0, config1);
-    SetIceProtocol(0, type);
     SetAllocatorFlags(0, allocator_flags1);
-    SetAllocationStepDelay(0, delay1);
+    SetAllocationStepDelay(0, delay);
     ConfigureEndpoint(1, config2);
-    SetIceProtocol(1, type);
     SetAllocatorFlags(1, allocator_flags2);
-    SetAllocationStepDelay(1, delay2);
+    SetAllocationStepDelay(1, delay);
 
-    if (type == cricket::ICEPROTO_RFC5245) {
-      set_clear_remote_candidates_ufrag_pwd(true);
-    }
+    set_clear_remote_candidates_ufrag_pwd(true);
   }
   void ConfigureEndpoint(int endpoint, Config config) {
     switch (config) {
@@ -1027,7 +1033,7 @@ const P2PTransportChannelTest::Result*
     P2PTransportChannelTest::kMatrixSharedSocketAsIce
         [NUM_CONFIGS][NUM_CONFIGS] = {
 //      OPEN  CONE  ADDR  PORT  SYMM  2CON  SCON  !UDP  !TCP  HTTP  PRXH  PRXS
-/*OP*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, PTLT, LTPT, LSRS, NULL, PTLT},
+/*OP*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, PTLT, LTPT, LSRS, NULL, LTPT},
 /*CO*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, NULL, NULL, LSRS, NULL, LTRT},
 /*AD*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, NULL, NULL, LSRS, NULL, LTRT},
 /*PO*/ {LULU, LUSU, LUSU, LUSU, LURU, LUSU, LURU, NULL, NULL, LSRS, NULL, LTRT},
@@ -1043,89 +1049,14 @@ const P2PTransportChannelTest::Result*
 
 // The actual tests that exercise all the various configurations.
 // Test names are of the form P2PTransportChannelTest_TestOPENToNAT_FULL_CONE
-// Same test case is run in both GICE and ICE mode.
-// kDefaultStepDelay - is used for all Gice cases.
-// kMinimumStepDelay - is used when both end points have
-//                     PORTALLOCATOR_ENABLE_SHARED_UFRAG flag enabled.
-// Technically we should be able to use kMinimumStepDelay irrespective of
-// protocol type. But which might need modifications to current result matrices
-// for tests in this file.
-#define P2P_TEST_DECLARATION(x, y, z) \
-  TEST_F(P2PTransportChannelTest, z##Test##x##To##y##AsGiceNoneSharedUfrag) { \
-    ConfigureEndpoints(x, y, kDefaultPortAllocatorFlags, \
-                       kDefaultPortAllocatorFlags, \
-                       kDefaultStepDelay, kDefaultStepDelay, \
-                       cricket::ICEPROTO_GOOGLE); \
-    if (kMatrix[x][y] != NULL) \
-      Test(*kMatrix[x][y]); \
-    else \
-      LOG(LS_WARNING) << "Not yet implemented"; \
-  } \
-  TEST_F(P2PTransportChannelTest, z##Test##x##To##y##AsGiceP0SharedUfrag) { \
-    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_UFRAG, \
-                       kDefaultPortAllocatorFlags, \
-                       kDefaultStepDelay, kDefaultStepDelay, \
-                       cricket::ICEPROTO_GOOGLE); \
-    if (kMatrix[x][y] != NULL) \
-      Test(*kMatrix[x][y]); \
-    else \
-      LOG(LS_WARNING) << "Not yet implemented"; \
-  } \
-  TEST_F(P2PTransportChannelTest, z##Test##x##To##y##AsGiceP1SharedUfrag) { \
-    ConfigureEndpoints(x, y, kDefaultPortAllocatorFlags, \
-                       PORTALLOCATOR_ENABLE_SHARED_UFRAG, \
-                       kDefaultStepDelay, kDefaultStepDelay, \
-                       cricket::ICEPROTO_GOOGLE); \
-    if (kMatrixSharedUfrag[x][y] != NULL) \
-      Test(*kMatrixSharedUfrag[x][y]); \
-    else \
-      LOG(LS_WARNING) << "Not yet implemented"; \
-  } \
-  TEST_F(P2PTransportChannelTest, z##Test##x##To##y##AsGiceBothSharedUfrag) { \
-    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_UFRAG, \
-                       PORTALLOCATOR_ENABLE_SHARED_UFRAG, \
-                       kDefaultStepDelay, kDefaultStepDelay, \
-                       cricket::ICEPROTO_GOOGLE); \
-    if (kMatrixSharedUfrag[x][y] != NULL) \
-      Test(*kMatrixSharedUfrag[x][y]); \
-    else \
-      LOG(LS_WARNING) << "Not yet implemented"; \
-  } \
-  TEST_F(P2PTransportChannelTest, \
-         z##Test##x##To##y##AsGiceBothSharedUfragWithMinimumStepDelay) { \
-    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_UFRAG, \
-                       PORTALLOCATOR_ENABLE_SHARED_UFRAG, \
-                       kMinimumStepDelay, kMinimumStepDelay, \
-                       cricket::ICEPROTO_GOOGLE); \
-    if (kMatrixSharedUfrag[x][y] != NULL) \
-      Test(*kMatrixSharedUfrag[x][y]); \
-    else \
-      LOG(LS_WARNING) << "Not yet implemented"; \
-  } \
-  TEST_F(P2PTransportChannelTest, \
-         z##Test##x##To##y##AsGiceBothSharedUfragSocket) { \
-    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_UFRAG | \
-                       PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
-                       PORTALLOCATOR_ENABLE_SHARED_UFRAG | \
-                       PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
-                       kMinimumStepDelay, kMinimumStepDelay, \
-                       cricket::ICEPROTO_GOOGLE); \
-    if (kMatrixSharedSocketAsGice[x][y] != NULL) \
-      Test(*kMatrixSharedSocketAsGice[x][y]); \
-    else \
-    LOG(LS_WARNING) << "Not yet implemented"; \
-  } \
-  TEST_F(P2PTransportChannelTest, z##Test##x##To##y##AsIce) { \
-    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_UFRAG | \
-                       PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
-                       PORTALLOCATOR_ENABLE_SHARED_UFRAG | \
-                       PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
-                       kMinimumStepDelay, kMinimumStepDelay, \
-                       cricket::ICEPROTO_RFC5245); \
-    if (kMatrixSharedSocketAsIce[x][y] != NULL) \
-      Test(*kMatrixSharedSocketAsIce[x][y]); \
-    else \
-    LOG(LS_WARNING) << "Not yet implemented"; \
+#define P2P_TEST_DECLARATION(x, y, z)                            \
+  TEST_F(P2PTransportChannelTest, z##Test##x##To##y) {           \
+    ConfigureEndpoints(x, y, PORTALLOCATOR_ENABLE_SHARED_SOCKET, \
+                       PORTALLOCATOR_ENABLE_SHARED_SOCKET);      \
+    if (kMatrixSharedSocketAsIce[x][y] != NULL)                  \
+      Test(*kMatrixSharedSocketAsIce[x][y]);                     \
+    else                                                         \
+      LOG(LS_WARNING) << "Not yet implemented";                  \
   }
 
 #define P2P_TEST(x, y) \
@@ -1179,94 +1110,51 @@ P2P_TEST_SET(PROXY_SOCKS)
 
 // Test that we restart candidate allocation when local ufrag&pwd changed.
 // Standard Ice protocol is used.
-TEST_F(P2PTransportChannelTest, HandleUfragPwdChangeAsIce) {
-  ConfigureEndpoints(OPEN, OPEN,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kMinimumStepDelay, kMinimumStepDelay,
-                     cricket::ICEPROTO_RFC5245);
+TEST_F(P2PTransportChannelTest, HandleUfragPwdChange) {
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   CreateChannels(1);
   TestHandleIceUfragPasswordChanged();
   DestroyChannels();
 }
 
-// Test that we restart candidate allocation when local ufrag&pwd changed.
-// Standard Ice protocol is used.
-TEST_F(P2PTransportChannelTest, HandleUfragPwdChangeBundleAsIce) {
-  ConfigureEndpoints(
-      OPEN, OPEN,
-      PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-      PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-      kMinimumStepDelay, kMinimumStepDelay,
-      cricket::ICEPROTO_RFC5245);
-  CreateChannels(2);
-  TestHandleIceUfragPasswordChanged();
-  DestroyChannels();
-}
-
-// Test that we restart candidate allocation when local ufrag&pwd changed.
-// Google Ice protocol is used.
-TEST_F(P2PTransportChannelTest, HandleUfragPwdChangeAsGice) {
-  ConfigureEndpoints(OPEN, OPEN,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_GOOGLE);
-  CreateChannels(1);
-  TestHandleIceUfragPasswordChanged();
-  DestroyChannels();
-}
-
-// Test that ICE restart works when bundle is enabled.
-// Google Ice protocol is used.
-TEST_F(P2PTransportChannelTest, HandleUfragPwdChangeBundleAsGice) {
-  ConfigureEndpoints(
-      OPEN, OPEN,
-      PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-      PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-      kDefaultStepDelay, kDefaultStepDelay,
-      cricket::ICEPROTO_GOOGLE);
-  CreateChannels(2);
-  TestHandleIceUfragPasswordChanged();
-  DestroyChannels();
-}
-
 // Test the operation of GetStats.
 TEST_F(P2PTransportChannelTest, GetStats) {
-  ConfigureEndpoints(OPEN, OPEN,
-                     kDefaultPortAllocatorFlags,
-                     kDefaultPortAllocatorFlags,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_GOOGLE);
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   CreateChannels(1);
-  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                          ep2_ch1()->readable() && ep2_ch1()->writable(),
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                          ep2_ch1()->receiving() && ep2_ch1()->writable(),
                           1000, 1000);
   TestSendRecv(1);
   cricket::ConnectionInfos infos;
   ASSERT_TRUE(ep1_ch1()->GetStats(&infos));
-  ASSERT_EQ(1U, infos.size());
-  EXPECT_TRUE(infos[0].new_connection);
-  EXPECT_TRUE(infos[0].best_connection);
-  EXPECT_TRUE(infos[0].readable);
-  EXPECT_TRUE(infos[0].writable);
-  EXPECT_FALSE(infos[0].timeout);
-  EXPECT_EQ(10U, infos[0].sent_total_packets);
-  EXPECT_EQ(0U, infos[0].sent_discarded_packets);
-  EXPECT_EQ(10 * 36U, infos[0].sent_total_bytes);
-  EXPECT_EQ(10 * 36U, infos[0].recv_total_bytes);
-  EXPECT_GT(infos[0].rtt, 0U);
+  ASSERT_TRUE(infos.size() >= 1);
+  cricket::ConnectionInfo* best_conn_info = nullptr;
+  for (cricket::ConnectionInfo& info : infos) {
+    if (info.best_connection) {
+      best_conn_info = &info;
+      break;
+    }
+  }
+  ASSERT_TRUE(best_conn_info != nullptr);
+  EXPECT_TRUE(best_conn_info->new_connection);
+  EXPECT_TRUE(best_conn_info->receiving);
+  EXPECT_TRUE(best_conn_info->writable);
+  EXPECT_FALSE(best_conn_info->timeout);
+  EXPECT_EQ(10U, best_conn_info->sent_total_packets);
+  EXPECT_EQ(0U, best_conn_info->sent_discarded_packets);
+  EXPECT_EQ(10 * 36U, best_conn_info->sent_total_bytes);
+  EXPECT_EQ(10 * 36U, best_conn_info->recv_total_bytes);
+  EXPECT_GT(best_conn_info->rtt, 0U);
   DestroyChannels();
 }
 
 // Test that we properly create a connection on a STUN ping from unknown address
 // when the signaling is slow.
 TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
-  ConfigureEndpoints(OPEN, OPEN,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_RFC5245);
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   // Emulate no remote credentials coming in.
   set_clear_remote_candidates_ufrag_pwd(false);
   CreateChannels(1);
@@ -1309,11 +1197,8 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
 // Test that we properly create a connection on a STUN ping from unknown address
 // when the signaling is slow and the end points are behind NAT.
 TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
-  ConfigureEndpoints(OPEN, NAT_SYMMETRIC,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_RFC5245);
+  ConfigureEndpoints(OPEN, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   // Emulate no remote credentials coming in.
   set_clear_remote_candidates_ufrag_pwd(false);
   CreateChannels(1);
@@ -1354,11 +1239,8 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
 // Test that if remote candidates don't have ufrag and pwd, we still work.
 TEST_F(P2PTransportChannelTest, RemoteCandidatesWithoutUfragPwd) {
   set_clear_remote_candidates_ufrag_pwd(true);
-  ConfigureEndpoints(OPEN, OPEN,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kMinimumStepDelay, kMinimumStepDelay,
-                     cricket::ICEPROTO_GOOGLE);
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
   CreateChannels(1);
   const cricket::Connection* best_connection = NULL;
   // Wait until the callee's connections are created.
@@ -1372,11 +1254,8 @@ TEST_F(P2PTransportChannelTest, RemoteCandidatesWithoutUfragPwd) {
 // Test that a host behind NAT cannot be reached when incoming_only
 // is set to true.
 TEST_F(P2PTransportChannelTest, IncomingOnlyBlocked) {
-  ConfigureEndpoints(NAT_FULL_CONE, OPEN,
-                     kDefaultPortAllocatorFlags,
-                     kDefaultPortAllocatorFlags,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_GOOGLE);
+  ConfigureEndpoints(NAT_FULL_CONE, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
 
   SetAllocatorFlags(0, kOnlyLocalPorts);
   CreateChannels(1);
@@ -1385,9 +1264,9 @@ TEST_F(P2PTransportChannelTest, IncomingOnlyBlocked) {
   // Pump for 1 second and verify that the channels are not connected.
   rtc::Thread::Current()->ProcessMessages(1000);
 
-  EXPECT_FALSE(ep1_ch1()->readable());
+  EXPECT_FALSE(ep1_ch1()->receiving());
   EXPECT_FALSE(ep1_ch1()->writable());
-  EXPECT_FALSE(ep2_ch1()->readable());
+  EXPECT_FALSE(ep2_ch1()->receiving());
   EXPECT_FALSE(ep2_ch1()->writable());
 
   DestroyChannels();
@@ -1396,19 +1275,16 @@ TEST_F(P2PTransportChannelTest, IncomingOnlyBlocked) {
 // Test that a peer behind NAT can connect to a peer that has
 // incoming_only flag set.
 TEST_F(P2PTransportChannelTest, IncomingOnlyOpen) {
-  ConfigureEndpoints(OPEN, NAT_FULL_CONE,
-                     kDefaultPortAllocatorFlags,
-                     kDefaultPortAllocatorFlags,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_GOOGLE);
+  ConfigureEndpoints(OPEN, NAT_FULL_CONE, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
 
   SetAllocatorFlags(0, kOnlyLocalPorts);
   CreateChannels(1);
   ep1_ch1()->set_incoming_only(true);
 
   EXPECT_TRUE_WAIT_MARGIN(ep1_ch1() != NULL && ep2_ch1() != NULL &&
-                          ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                          ep2_ch1()->readable() && ep2_ch1()->writable(),
+                          ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                          ep2_ch1()->receiving() && ep2_ch1()->writable(),
                           1000, 1000);
 
   DestroyChannels();
@@ -1423,8 +1299,7 @@ TEST_F(P2PTransportChannelTest, TestTcpConnectionsFromActiveToPassive) {
 
   int kOnlyLocalTcpPorts = cricket::PORTALLOCATOR_DISABLE_UDP |
                            cricket::PORTALLOCATOR_DISABLE_STUN |
-                           cricket::PORTALLOCATOR_DISABLE_RELAY |
-                           cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG;
+                           cricket::PORTALLOCATOR_DISABLE_RELAY;
   // Disable all protocols except TCP.
   SetAllocatorFlags(0, kOnlyLocalTcpPorts);
   SetAllocatorFlags(1, kOnlyLocalTcpPorts);
@@ -1432,95 +1307,34 @@ TEST_F(P2PTransportChannelTest, TestTcpConnectionsFromActiveToPassive) {
   SetAllowTcpListen(0, true);   // actpass.
   SetAllowTcpListen(1, false);  // active.
 
+  // Pause candidate so we could verify the candidate properties.
+  PauseCandidates(0);
+  PauseCandidates(1);
   CreateChannels(1);
 
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() && ep2_ch1()->writable(),
+  // Verify tcp candidates.
+  VerifySavedTcpCandidates(0, cricket::TCPTYPE_PASSIVE_STR);
+  VerifySavedTcpCandidates(1, cricket::TCPTYPE_ACTIVE_STR);
+
+  // Resume candidates.
+  ResumeCandidates(0);
+  ResumeCandidates(1);
+
+  EXPECT_TRUE_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                   ep2_ch1()->receiving() && ep2_ch1()->writable(),
                    1000);
   EXPECT_TRUE(
       ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
       LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
       RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
 
-  std::string kTcpProtocol = "tcp";
-  EXPECT_EQ(kTcpProtocol, RemoteCandidate(ep1_ch1())->protocol());
-  EXPECT_EQ(kTcpProtocol, LocalCandidate(ep1_ch1())->protocol());
-  EXPECT_EQ(kTcpProtocol, RemoteCandidate(ep2_ch1())->protocol());
-  EXPECT_EQ(kTcpProtocol, LocalCandidate(ep2_ch1())->protocol());
-
   TestSendRecv(1);
   DestroyChannels();
 }
 
-TEST_F(P2PTransportChannelTest, TestBundleAllocatorToBundleAllocator) {
+TEST_F(P2PTransportChannelTest, TestIceRoleConflict) {
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
-  SetAllocatorFlags(
-      0, PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  SetAllocatorFlags(
-      1, PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-
-  CreateChannels(2);
-
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
-                   ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
-                   ep2_ch1()->writable(),
-                   1000);
-  EXPECT_TRUE(ep1_ch1()->best_connection() &&
-              ep2_ch1()->best_connection());
-
-  EXPECT_FALSE(ep1_ch2()->readable());
-  EXPECT_FALSE(ep1_ch2()->writable());
-  EXPECT_FALSE(ep2_ch2()->readable());
-  EXPECT_FALSE(ep2_ch2()->writable());
-
-  TestSendRecv(1);  // Only 1 channel is writable per Endpoint.
-  DestroyChannels();
-}
-
-TEST_F(P2PTransportChannelTest, TestBundleAllocatorToNonBundleAllocator) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  // Enable BUNDLE flag at one side.
-  SetAllocatorFlags(
-      0, PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-
-  CreateChannels(2);
-
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
-                   ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
-                   ep2_ch1()->writable(),
-                   1000);
-  EXPECT_TRUE_WAIT(ep1_ch2()->readable() &&
-                   ep1_ch2()->writable() &&
-                   ep2_ch2()->readable() &&
-                   ep2_ch2()->writable(),
-                   1000);
-
-  EXPECT_TRUE(ep1_ch1()->best_connection() &&
-              ep2_ch1()->best_connection());
-  EXPECT_TRUE(ep1_ch2()->best_connection() &&
-              ep2_ch2()->best_connection());
-
-  TestSendRecv(2);
-  DestroyChannels();
-}
-
-TEST_F(P2PTransportChannelTest, TestIceRoleConflictWithoutBundle) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  TestSignalRoleConflict();
-}
-
-TEST_F(P2PTransportChannelTest, TestIceRoleConflictWithBundle) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  SetAllocatorFlags(
-      0, PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  SetAllocatorFlags(
-      1, PORTALLOCATOR_ENABLE_BUNDLE | PORTALLOCATOR_ENABLE_SHARED_UFRAG);
   TestSignalRoleConflict();
 }
 
@@ -1531,10 +1345,8 @@ TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
   AddAddress(1, kPublicAddrs[1]);
 
   SetIceRole(0, cricket::ICEROLE_CONTROLLING);
-  SetIceProtocol(0, cricket::ICEPROTO_GOOGLE);
   SetIceTiebreaker(0, kTiebreaker1);
   SetIceRole(1, cricket::ICEROLE_CONTROLLING);
-  SetIceProtocol(1, cricket::ICEPROTO_RFC5245);
   SetIceTiebreaker(1, kTiebreaker2);
 
   CreateChannels(1);
@@ -1544,26 +1356,23 @@ TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
   const std::vector<cricket::PortInterface *> ports_before = ep1_ch1()->ports();
   for (size_t i = 0; i < ports_before.size(); ++i) {
     EXPECT_EQ(cricket::ICEROLE_CONTROLLING, ports_before[i]->GetIceRole());
-    EXPECT_EQ(cricket::ICEPROTO_GOOGLE, ports_before[i]->IceProtocol());
     EXPECT_EQ(kTiebreaker1, ports_before[i]->IceTiebreaker());
   }
 
   ep1_ch1()->SetIceRole(cricket::ICEROLE_CONTROLLED);
-  ep1_ch1()->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
   ep1_ch1()->SetIceTiebreaker(kTiebreaker2);
 
   const std::vector<cricket::PortInterface *> ports_after = ep1_ch1()->ports();
   for (size_t i = 0; i < ports_after.size(); ++i) {
     EXPECT_EQ(cricket::ICEROLE_CONTROLLED, ports_before[i]->GetIceRole());
-    EXPECT_EQ(cricket::ICEPROTO_RFC5245, ports_before[i]->IceProtocol());
     // SetIceTiebreaker after Connect() has been called will fail. So expect the
     // original value.
     EXPECT_EQ(kTiebreaker1, ports_before[i]->IceTiebreaker());
   }
 
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
+  EXPECT_TRUE_WAIT(ep1_ch1()->receiving() &&
                    ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
+                   ep2_ch1()->receiving() &&
                    ep2_ch1()->writable(),
                    1000);
 
@@ -1574,18 +1383,6 @@ TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
   DestroyChannels();
 }
 
-// This test verifies channel can handle ice messages when channel is in
-// hybrid mode.
-TEST_F(P2PTransportChannelTest, TestConnectivityBetweenHybridandIce) {
-  TestHybridConnectivity(cricket::ICEPROTO_RFC5245);
-}
-
-// This test verifies channel can handle Gice messages when channel is in
-// hybrid mode.
-TEST_F(P2PTransportChannelTest, TestConnectivityBetweenHybridandGice) {
-  TestHybridConnectivity(cricket::ICEPROTO_GOOGLE);
-}
-
 // Verify that we can set DSCP value and retrieve properly from P2PTC.
 TEST_F(P2PTransportChannelTest, TestDefaultDscpValue) {
   AddAddress(0, kPublicAddrs[0]);
@@ -1630,8 +1427,8 @@ TEST_F(P2PTransportChannelTest, TestIPv6Connections) {
 
   CreateChannels(1);
 
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() && ep2_ch1()->writable(),
+  EXPECT_TRUE_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                   ep2_ch1()->receiving() && ep2_ch1()->writable(),
                    1000);
   EXPECT_TRUE(
       ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
@@ -1644,15 +1441,10 @@ TEST_F(P2PTransportChannelTest, TestIPv6Connections) {
 
 // Testing forceful TURN connections.
 TEST_F(P2PTransportChannelTest, TestForceTurn) {
-  ConfigureEndpoints(NAT_PORT_RESTRICTED, NAT_SYMMETRIC,
-                     kDefaultPortAllocatorFlags |
-                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
-                         cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kDefaultPortAllocatorFlags |
-                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
-                         cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG,
-                     kDefaultStepDelay, kDefaultStepDelay,
-                     cricket::ICEPROTO_RFC5245);
+  ConfigureEndpoints(
+      NAT_PORT_RESTRICTED, NAT_SYMMETRIC,
+      kDefaultPortAllocatorFlags | cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET,
+      kDefaultPortAllocatorFlags | cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   set_force_relay(true);
 
   SetAllocationStepDelay(0, kMinimumStepDelay);
@@ -1660,11 +1452,9 @@ TEST_F(P2PTransportChannelTest, TestForceTurn) {
 
   CreateChannels(1);
 
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
-                   ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
-                   ep2_ch1()->writable(),
-                   1000);
+  EXPECT_TRUE_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                       ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                   2000);
 
   EXPECT_TRUE(ep1_ch1()->best_connection() &&
               ep2_ch1()->best_connection());
@@ -1678,6 +1468,34 @@ TEST_F(P2PTransportChannelTest, TestForceTurn) {
   DestroyChannels();
 }
 
+// Test that if continual gathering is set to true, ICE gathering state will
+// not change to "Complete", and vice versa.
+TEST_F(P2PTransportChannelTest, TestContinualGathering) {
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
+  SetAllocationStepDelay(0, kDefaultStepDelay);
+  SetAllocationStepDelay(1, kDefaultStepDelay);
+  CreateChannels(1);
+  cricket::IceConfig config = CreateIceConfig(1000, true);
+  ep1_ch1()->SetIceConfig(config);
+  // By default, ep2 does not gather continually.
+
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1() != NULL && ep2_ch1() != NULL &&
+                              ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                          1000, 1000);
+  WAIT(cricket::IceGatheringState::kIceGatheringComplete ==
+           ep1_ch1()->gathering_state(),
+       1000);
+  EXPECT_EQ(cricket::IceGatheringState::kIceGatheringGathering,
+            ep1_ch1()->gathering_state());
+  // By now, ep2 should have completed gathering.
+  EXPECT_EQ(cricket::IceGatheringState::kIceGatheringComplete,
+            ep2_ch1()->gathering_state());
+
+  DestroyChannels();
+}
+
 // Test what happens when we have 2 users behind the same NAT. This can lead
 // to interesting behavior because the STUN server will only give out the
 // address of the outermost NAT.
@@ -1708,7 +1526,8 @@ class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
 
 TEST_F(P2PTransportChannelSameNatTest, TestConesBehindSameCone) {
   ConfigureEndpoints(NAT_FULL_CONE, NAT_FULL_CONE, NAT_FULL_CONE);
-  Test(kLocalUdpToStunUdp);
+  Test(P2PTransportChannelTestBase::Result(
+      "prflx", "udp", "stun", "udp", "stun", "udp", "prflx", "udp", 1000));
 }
 
 // Test what happens when we have multiple available pathways.
@@ -1727,7 +1546,8 @@ TEST_F(P2PTransportChannelMultihomedTest, DISABLED_TestBasic) {
 }
 
 // Test that we can quickly switch links if an interface goes down.
-TEST_F(P2PTransportChannelMultihomedTest, TestFailover) {
+// The controlled side has two interfaces and one will die.
+TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControlledSide) {
   AddAddress(0, kPublicAddrs[0]);
   // Adding alternate address will make sure |kPublicAddrs| has the higher
   // priority than others. This is due to FakeNetwork::AddInterface method.
@@ -1740,33 +1560,154 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailover) {
 
   // Create channels and let them go writable, as usual.
   CreateChannels(1);
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() && ep2_ch1()->writable(),
-                   1000);
+
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                          1000, 1000);
   EXPECT_TRUE(
       ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
       LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
       RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
 
+  // Make the receiving timeout shorter for testing.
+  cricket::IceConfig config = CreateIceConfig(1000, false);
+  ep1_ch1()->SetIceConfig(config);
+  ep2_ch1()->SetIceConfig(config);
+
   // Blackhole any traffic to or from the public addrs.
   LOG(LS_INFO) << "Failing over...";
-  fw()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY,
-                kPublicAddrs[1]);
-
-  // We should detect loss of connectivity within 5 seconds or so.
-  EXPECT_TRUE_WAIT(!ep1_ch1()->writable(), 7000);
-
-  // We should switch over to use the alternate addr immediately
-  // when we lose writability.
+  fw()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, kPublicAddrs[1]);
+  // The best connections will switch, so keep references to them.
+  const cricket::Connection* best_connection1 = ep1_ch1()->best_connection();
+  const cricket::Connection* best_connection2 = ep2_ch1()->best_connection();
+  // We should detect loss of receiving within 1 second or so.
   EXPECT_TRUE_WAIT(
-      ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
-      LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
-      RemoteCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[1]),
-      3000);
+      !best_connection1->receiving() && !best_connection2->receiving(), 3000);
+
+  // We should switch over to use the alternate addr immediately on both sides
+  // when we are not receiving.
+  EXPECT_TRUE_WAIT(
+      ep1_ch1()->best_connection()->receiving() &&
+      ep2_ch1()->best_connection()->receiving(), 1000);
+  EXPECT_TRUE(LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]));
+  EXPECT_TRUE(
+      RemoteCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[1]));
+  EXPECT_TRUE(
+      LocalCandidate(ep2_ch1())->address().EqualIPs(kAlternateAddrs[1]));
 
   DestroyChannels();
 }
 
+// Test that we can quickly switch links if an interface goes down.
+// The controlling side has two interfaces and one will die.
+TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControllingSide) {
+  // Adding alternate address will make sure |kPublicAddrs| has the higher
+  // priority than others. This is due to FakeNetwork::AddInterface method.
+  AddAddress(0, kAlternateAddrs[0]);
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+
+  // Use only local ports for simplicity.
+  SetAllocatorFlags(0, kOnlyLocalPorts);
+  SetAllocatorFlags(1, kOnlyLocalPorts);
+
+  // Create channels and let them go writable, as usual.
+  CreateChannels(1);
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                          1000, 1000);
+  EXPECT_TRUE(
+      ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
+      LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
+      RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
+
+  // Make the receiving timeout shorter for testing.
+  cricket::IceConfig config = CreateIceConfig(1000, false);
+  ep1_ch1()->SetIceConfig(config);
+  ep2_ch1()->SetIceConfig(config);
+
+  // Blackhole any traffic to or from the public addrs.
+  LOG(LS_INFO) << "Failing over...";
+  fw()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, kPublicAddrs[0]);
+  // The best connections will switch, so keep references to them.
+  const cricket::Connection* best_connection1 = ep1_ch1()->best_connection();
+  const cricket::Connection* best_connection2 = ep2_ch1()->best_connection();
+  // We should detect loss of receiving within 1 second or so.
+  EXPECT_TRUE_WAIT(
+      !best_connection1->receiving() && !best_connection2->receiving(), 3000);
+
+  // We should switch over to use the alternate addr immediately on both sides
+  // when we are not receiving.
+  EXPECT_TRUE_WAIT(
+      ep1_ch1()->best_connection()->receiving() &&
+      ep2_ch1()->best_connection()->receiving(), 1000);
+  EXPECT_TRUE(
+    LocalCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[0]));
+  EXPECT_TRUE(RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
+  EXPECT_TRUE(
+      RemoteCandidate(ep2_ch1())->address().EqualIPs(kAlternateAddrs[0]));
+
+  DestroyChannels();
+}
+
+// Test that the backup connection is pinged at a rate no faster than
+// what was configured.
+TEST_F(P2PTransportChannelMultihomedTest, TestPingBackupConnectionRate) {
+  AddAddress(0, kPublicAddrs[0]);
+  // Adding alternate address will make sure |kPublicAddrs| has the higher
+  // priority than others. This is due to FakeNetwork::AddInterface method.
+  AddAddress(1, kAlternateAddrs[1]);
+  AddAddress(1, kPublicAddrs[1]);
+
+  // Use only local ports for simplicity.
+  SetAllocatorFlags(0, kOnlyLocalPorts);
+  SetAllocatorFlags(1, kOnlyLocalPorts);
+
+  // Create channels and let them go writable, as usual.
+  CreateChannels(1);
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                          1000, 1000);
+  int backup_ping_interval = 2000;
+  ep2_ch1()->SetIceConfig(CreateIceConfig(2000, false, backup_ping_interval));
+  // After the state becomes COMPLETED, the backup connection will be pinged
+  // once every |backup_ping_interval| milliseconds.
+  ASSERT_TRUE_WAIT(ep2_ch1()->GetState() == cricket::STATE_COMPLETED, 1000);
+  const std::vector<cricket::Connection*>& connections =
+      ep2_ch1()->connections();
+  ASSERT_EQ(2U, connections.size());
+  cricket::Connection* backup_conn = connections[1];
+  EXPECT_TRUE_WAIT(backup_conn->writable(), 3000);
+  uint32_t last_ping_response_ms = backup_conn->last_ping_response_received();
+  EXPECT_TRUE_WAIT(
+      last_ping_response_ms < backup_conn->last_ping_response_received(), 5000);
+  int time_elapsed =
+      backup_conn->last_ping_response_received() - last_ping_response_ms;
+  LOG(LS_INFO) << "Time elapsed: " << time_elapsed;
+  EXPECT_GE(time_elapsed, backup_ping_interval);
+}
+
+TEST_F(P2PTransportChannelMultihomedTest, TestGetState) {
+  AddAddress(0, kAlternateAddrs[0]);
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+  // Create channels and let them go writable, as usual.
+  CreateChannels(1);
+
+  // Both transport channels will reach STATE_COMPLETED quickly.
+  EXPECT_EQ_WAIT(cricket::TransportChannelState::STATE_COMPLETED,
+                 ep1_ch1()->GetState(), 1000);
+  EXPECT_EQ_WAIT(cricket::TransportChannelState::STATE_COMPLETED,
+                 ep2_ch1()->GetState(), 1000);
+}
+
+/*
+
+TODO(pthatcher): Once have a way to handle network interfaces changes
+without signalling an ICE restart, put a test like this back.  In the
+mean time, this test only worked for GICE.  With ICE, it's currently
+not possible without an ICE restart.
+
 // Test that we can switch links in a coordinated fashion.
 TEST_F(P2PTransportChannelMultihomedTest, TestDrain) {
   AddAddress(0, kPublicAddrs[0]);
@@ -1777,20 +1718,22 @@ TEST_F(P2PTransportChannelMultihomedTest, TestDrain) {
 
   // Create channels and let them go writable, as usual.
   CreateChannels(1);
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() && ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() && ep2_ch1()->writable(),
+  EXPECT_TRUE_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                   ep2_ch1()->receiving() && ep2_ch1()->writable(),
                    1000);
   EXPECT_TRUE(
       ep1_ch1()->best_connection() && ep2_ch1()->best_connection() &&
       LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
       RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
 
+
   // Remove the public interface, add the alternate interface, and allocate
-  // a new generation of candidates for the new interface (via Connect()).
+  // a new generation of candidates for the new interface (via
+  // MaybeStartGathering()).
   LOG(LS_INFO) << "Draining...";
   AddAddress(1, kAlternateAddrs[1]);
   RemoveAddress(1, kPublicAddrs[1]);
-  ep2_ch1()->Connect();
+  ep2_ch1()->MaybeStartGathering();
 
   // We should switch over to use the alternate address after
   // an exchange of pings.
@@ -1802,3 +1745,614 @@ TEST_F(P2PTransportChannelMultihomedTest, TestDrain) {
 
   DestroyChannels();
 }
+
+*/
+
+// A collection of tests which tests a single P2PTransportChannel by sending
+// pings.
+class P2PTransportChannelPingTest : public testing::Test,
+                                    public sigslot::has_slots<> {
+ public:
+  P2PTransportChannelPingTest()
+      : pss_(new rtc::PhysicalSocketServer),
+        vss_(new rtc::VirtualSocketServer(pss_.get())),
+        ss_scope_(vss_.get()) {}
+
+ protected:
+  void PrepareChannel(cricket::P2PTransportChannel* ch) {
+    ch->SetIceRole(cricket::ICEROLE_CONTROLLING);
+    ch->SetIceCredentials(kIceUfrag[0], kIcePwd[0]);
+    ch->SetRemoteIceCredentials(kIceUfrag[1], kIcePwd[1]);
+  }
+
+  cricket::Candidate CreateCandidate(const std::string& ip,
+                                     int port,
+                                     int priority,
+                                     const std::string& ufrag = "") {
+    cricket::Candidate c;
+    c.set_address(rtc::SocketAddress(ip, port));
+    c.set_component(1);
+    c.set_protocol(cricket::UDP_PROTOCOL_NAME);
+    c.set_priority(priority);
+    c.set_username(ufrag);
+    return c;
+  }
+
+  cricket::Connection* WaitForConnectionTo(cricket::P2PTransportChannel* ch,
+                                           const std::string& ip,
+                                           int port_num) {
+    EXPECT_TRUE_WAIT(GetConnectionTo(ch, ip, port_num) != nullptr, 3000);
+    return GetConnectionTo(ch, ip, port_num);
+  }
+
+  cricket::Port* GetPort(cricket::P2PTransportChannel* ch) {
+    if (ch->ports().empty()) {
+      return nullptr;
+    }
+    return static_cast<cricket::Port*>(ch->ports()[0]);
+  }
+
+  cricket::Connection* GetConnectionTo(cricket::P2PTransportChannel* ch,
+                                       const std::string& ip,
+                                       int port_num) {
+    cricket::Port* port = GetPort(ch);
+    if (!port) {
+      return nullptr;
+    }
+    return port->GetConnection(rtc::SocketAddress(ip, port_num));
+  }
+
+ private:
+  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
+  rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
+  rtc::SocketServerScope ss_scope_;
+};
+
+TEST_F(P2PTransportChannelPingTest, TestTriggeredChecks) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("trigger checks", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1));
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 2));
+
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn1 != nullptr);
+  ASSERT_TRUE(conn2 != nullptr);
+
+  // Before a triggered check, the first connection to ping is the
+  // highest priority one.
+  EXPECT_EQ(conn2, ch.FindNextPingableConnection());
+
+  // Receiving a ping causes a triggered check which should make conn1
+  // be pinged first instead of conn2, even though conn2 has a higher
+  // priority.
+  conn1->ReceivedPing();
+  EXPECT_EQ(conn1, ch.FindNextPingableConnection());
+}
+
+TEST_F(P2PTransportChannelPingTest, TestNoTriggeredChecksWhenWritable) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("trigger checks", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1));
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 2));
+
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn1 != nullptr);
+  ASSERT_TRUE(conn2 != nullptr);
+
+  EXPECT_EQ(conn2, ch.FindNextPingableConnection());
+  conn1->ReceivedPingResponse();
+  ASSERT_TRUE(conn1->writable());
+  conn1->ReceivedPing();
+
+  // Ping received, but the connection is already writable, so no
+  // "triggered check" and conn2 is pinged before conn1 because it has
+  // a higher priority.
+  EXPECT_EQ(conn2, ch.FindNextPingableConnection());
+}
+
+// Test adding remote candidates with different ufrags. If a remote candidate
+// is added with an old ufrag, it will be discarded. If it is added with a
+// ufrag that was not seen before, it will be used to create connections
+// although the ICE pwd in the remote candidate will be set when the ICE
+// credentials arrive. If a remote candidate is added with the current ICE
+// ufrag, its pwd and generation will be set properly.
+TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithVariousUfrags) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("add candidate", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  // Add a candidate with a future ufrag.
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1, kIceUfrag[2]));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  const cricket::Candidate& candidate = conn1->remote_candidate();
+  EXPECT_EQ(kIceUfrag[2], candidate.username());
+  EXPECT_TRUE(candidate.password().empty());
+  EXPECT_TRUE(ch.FindNextPingableConnection() == nullptr);
+
+  // Set the remote credentials with the "future" ufrag.
+  // This should set the ICE pwd in the remote candidate of |conn1|, making
+  // it pingable.
+  ch.SetRemoteIceCredentials(kIceUfrag[2], kIcePwd[2]);
+  EXPECT_EQ(kIceUfrag[2], candidate.username());
+  EXPECT_EQ(kIcePwd[2], candidate.password());
+  EXPECT_EQ(conn1, ch.FindNextPingableConnection());
+
+  // Add a candidate with an old ufrag. No connection will be created.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 2, kIceUfrag[1]));
+  rtc::Thread::Current()->ProcessMessages(500);
+  EXPECT_TRUE(GetConnectionTo(&ch, "2.2.2.2", 2) == nullptr);
+
+  // Add a candidate with the current ufrag, its pwd and generation will be
+  // assigned, even if the generation is not set.
+  ch.AddRemoteCandidate(CreateCandidate("3.3.3.3", 3, 0, kIceUfrag[2]));
+  cricket::Connection* conn3 = nullptr;
+  ASSERT_TRUE_WAIT((conn3 = GetConnectionTo(&ch, "3.3.3.3", 3)) != nullptr,
+                   3000);
+  const cricket::Candidate& new_candidate = conn3->remote_candidate();
+  EXPECT_EQ(kIcePwd[2], new_candidate.password());
+  EXPECT_EQ(1U, new_candidate.generation());
+
+  // Check that the pwd of all remote candidates are properly assigned.
+  for (const cricket::RemoteCandidate& candidate : ch.remote_candidates()) {
+    EXPECT_TRUE(candidate.username() == kIceUfrag[1] ||
+                candidate.username() == kIceUfrag[2]);
+    if (candidate.username() == kIceUfrag[1]) {
+      EXPECT_EQ(kIcePwd[1], candidate.password());
+    } else if (candidate.username() == kIceUfrag[2]) {
+      EXPECT_EQ(kIcePwd[2], candidate.password());
+    }
+  }
+}
+
+TEST_F(P2PTransportChannelPingTest, ConnectionResurrection) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("connection resurrection", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.Connect();
+  ch.MaybeStartGathering();
+
+  // Create conn1 and keep track of original candidate priority.
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  uint32_t remote_priority = conn1->remote_candidate().priority();
+
+  // Create a higher priority candidate and make the connection
+  // receiving/writable. This will prune conn1.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 2));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  conn2->ReceivedPing();
+  conn2->ReceivedPingResponse();
+
+  // Wait for conn1 to be pruned.
+  EXPECT_TRUE_WAIT(conn1->pruned(), 3000);
+  // Destroy the connection to test SignalUnknownAddress.
+  conn1->Destroy();
+  EXPECT_TRUE_WAIT(GetConnectionTo(&ch, "1.1.1.1", 1) == nullptr, 1000);
+
+  // Create a minimal STUN message with prflx priority.
+  cricket::IceMessage request;
+  request.SetType(cricket::STUN_BINDING_REQUEST);
+  request.AddAttribute(new cricket::StunByteStringAttribute(
+      cricket::STUN_ATTR_USERNAME, kIceUfrag[1]));
+  uint32_t prflx_priority = cricket::ICE_TYPE_PREFERENCE_PRFLX << 24;
+  request.AddAttribute(new cricket::StunUInt32Attribute(
+      cricket::STUN_ATTR_PRIORITY, prflx_priority));
+  EXPECT_NE(prflx_priority, remote_priority);
+
+  cricket::Port* port = GetPort(&ch);
+  // conn1 should be resurrected with original priority.
+  port->SignalUnknownAddress(port, rtc::SocketAddress("1.1.1.1", 1),
+                             cricket::PROTO_UDP, &request, kIceUfrag[1], false);
+  conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  EXPECT_EQ(conn1->remote_candidate().priority(), remote_priority);
+
+  // conn3, a real prflx connection, should have prflx priority.
+  port->SignalUnknownAddress(port, rtc::SocketAddress("3.3.3.3", 1),
+                             cricket::PROTO_UDP, &request, kIceUfrag[1], false);
+  cricket::Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 1);
+  ASSERT_TRUE(conn3 != nullptr);
+  EXPECT_EQ(conn3->remote_candidate().priority(), prflx_priority);
+}
+
+TEST_F(P2PTransportChannelPingTest, TestReceivingStateChange) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("receiving state change", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  // Default receiving timeout and checking receiving delay should not be too
+  // small.
+  EXPECT_LE(1000, ch.receiving_timeout());
+  EXPECT_LE(200, ch.check_receiving_delay());
+  ch.SetIceConfig(CreateIceConfig(500, false));
+  EXPECT_EQ(500, ch.receiving_timeout());
+  EXPECT_EQ(50, ch.check_receiving_delay());
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+
+  conn1->ReceivedPing();
+  conn1->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
+  EXPECT_TRUE_WAIT(ch.best_connection() != nullptr, 1000);
+  EXPECT_TRUE_WAIT(ch.receiving(), 1000);
+  EXPECT_TRUE_WAIT(!ch.receiving(), 1000);
+}
+
+// The controlled side will select a connection as the "best connection" based
+// on priority until the controlling side nominates a connection, at which
+// point the controlled side will select that connection as the
+// "best connection".
+TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBeforeNomination) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("receiving state change", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.SetIceRole(cricket::ICEROLE_CONTROLLED);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  EXPECT_EQ(conn1, ch.best_connection());
+
+  // When a higher priority candidate comes in, the new connection is chosen
+  // as the best connection.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 10));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  EXPECT_EQ(conn2, ch.best_connection());
+
+  // If a stun request with use-candidate attribute arrives, the receiving
+  // connection will be set as the best connection, even though
+  // its priority is lower.
+  ch.AddRemoteCandidate(CreateCandidate("3.3.3.3", 3, 1));
+  cricket::Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
+  ASSERT_TRUE(conn3 != nullptr);
+  // Because it has a lower priority, the best connection is still conn2.
+  EXPECT_EQ(conn2, ch.best_connection());
+  conn3->ReceivedPingResponse();  // Become writable.
+  // But if it is nominated via use_candidate, it is chosen as the best
+  // connection.
+  conn3->set_nominated(true);
+  conn3->SignalNominated(conn3);
+  EXPECT_EQ(conn3, ch.best_connection());
+
+  // Even if another higher priority candidate arrives,
+  // it will not be set as the best connection because the best connection
+  // is nominated by the controlling side.
+  ch.AddRemoteCandidate(CreateCandidate("4.4.4.4", 4, 100));
+  cricket::Connection* conn4 = WaitForConnectionTo(&ch, "4.4.4.4", 4);
+  ASSERT_TRUE(conn4 != nullptr);
+  EXPECT_EQ(conn3, ch.best_connection());
+  // But if it is nominated via use_candidate and writable, it will be set as
+  // the best connection.
+  conn4->set_nominated(true);
+  conn4->SignalNominated(conn4);
+  // Not switched yet because conn4 is not writable.
+  EXPECT_EQ(conn3, ch.best_connection());
+  // The best connection switches after conn4 becomes writable.
+  conn4->ReceivedPingResponse();
+  EXPECT_EQ(conn4, ch.best_connection());
+}
+
+// The controlled side will select a connection as the "best connection" based
+// on requests from an unknown address before the controlling side nominates
+// a connection, and will nominate a connection from an unknown address if the
+// request contains the use_candidate attribute. Plus, it will also sends back
+// a ping response and set the ICE pwd in the remote candidate appropriately.
+TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("receiving state change", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.SetIceRole(cricket::ICEROLE_CONTROLLED);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  // A minimal STUN message with prflx priority.
+  cricket::IceMessage request;
+  request.SetType(cricket::STUN_BINDING_REQUEST);
+  request.AddAttribute(new cricket::StunByteStringAttribute(
+      cricket::STUN_ATTR_USERNAME, kIceUfrag[1]));
+  uint32_t prflx_priority = cricket::ICE_TYPE_PREFERENCE_PRFLX << 24;
+  request.AddAttribute(new cricket::StunUInt32Attribute(
+      cricket::STUN_ATTR_PRIORITY, prflx_priority));
+  cricket::TestUDPPort* port = static_cast<cricket::TestUDPPort*>(GetPort(&ch));
+  port->SignalUnknownAddress(port, rtc::SocketAddress("1.1.1.1", 1),
+                             cricket::PROTO_UDP, &request, kIceUfrag[1], false);
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  EXPECT_TRUE(port->sent_binding_response());
+  EXPECT_EQ(conn1, ch.best_connection());
+  conn1->ReceivedPingResponse();
+  EXPECT_EQ(conn1, ch.best_connection());
+  port->set_sent_binding_response(false);
+
+  // Another connection is nominated via use_candidate.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 1));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  // Because it has a lower priority, the best connection is still conn1.
+  EXPECT_EQ(conn1, ch.best_connection());
+  // When it is nominated via use_candidate and writable, it is chosen as the
+  // best connection.
+  conn2->ReceivedPingResponse();  // Become writable.
+  conn2->set_nominated(true);
+  conn2->SignalNominated(conn2);
+  EXPECT_EQ(conn2, ch.best_connection());
+
+  // Another request with unknown address, it will not be set as the best
+  // connection because the best connection was nominated by the controlling
+  // side.
+  port->SignalUnknownAddress(port, rtc::SocketAddress("3.3.3.3", 3),
+                             cricket::PROTO_UDP, &request, kIceUfrag[1], false);
+  cricket::Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
+  ASSERT_TRUE(conn3 != nullptr);
+  EXPECT_TRUE(port->sent_binding_response());
+  conn3->ReceivedPingResponse();  // Become writable.
+  EXPECT_EQ(conn2, ch.best_connection());
+  port->set_sent_binding_response(false);
+
+  // However if the request contains use_candidate attribute, it will be
+  // selected as the best connection.
+  request.AddAttribute(
+      new cricket::StunByteStringAttribute(cricket::STUN_ATTR_USE_CANDIDATE));
+  port->SignalUnknownAddress(port, rtc::SocketAddress("4.4.4.4", 4),
+                             cricket::PROTO_UDP, &request, kIceUfrag[1], false);
+  cricket::Connection* conn4 = WaitForConnectionTo(&ch, "4.4.4.4", 4);
+  ASSERT_TRUE(conn4 != nullptr);
+  EXPECT_TRUE(port->sent_binding_response());
+  // conn4 is not the best connection yet because it is not writable.
+  EXPECT_EQ(conn2, ch.best_connection());
+  conn4->ReceivedPingResponse();  // Become writable.
+  EXPECT_EQ(conn4, ch.best_connection());
+
+  // Test that the request from an unknown address contains a ufrag from an old
+  // generation.
+  port->set_sent_binding_response(false);
+  ch.SetRemoteIceCredentials(kIceUfrag[2], kIcePwd[2]);
+  ch.SetRemoteIceCredentials(kIceUfrag[3], kIcePwd[3]);
+  port->SignalUnknownAddress(port, rtc::SocketAddress("5.5.5.5", 5),
+                             cricket::PROTO_UDP, &request, kIceUfrag[2], false);
+  cricket::Connection* conn5 = WaitForConnectionTo(&ch, "5.5.5.5", 5);
+  ASSERT_TRUE(conn5 != nullptr);
+  EXPECT_TRUE(port->sent_binding_response());
+  EXPECT_EQ(kIcePwd[2], conn5->remote_candidate().password());
+}
+
+// The controlled side will select a connection as the "best connection"
+// based on media received until the controlling side nominates a connection,
+// at which point the controlled side will select that connection as
+// the "best connection".
+TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("receiving state change", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.SetIceRole(cricket::ICEROLE_CONTROLLED);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 10));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  EXPECT_EQ(conn1, ch.best_connection());
+
+  // If a data packet is received on conn2, the best connection should
+  // switch to conn2 because the controlled side must mirror the media path
+  // chosen by the controlling side.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 1));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  conn2->ReceivedPing();  // Start receiving.
+  // Do not switch because it is not writable.
+  conn2->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
+  EXPECT_EQ(conn1, ch.best_connection());
+
+  conn2->ReceivedPingResponse();  // Become writable.
+  // Switch because it is writable.
+  conn2->OnReadPacket("DEF", 3, rtc::CreatePacketTime(0));
+  EXPECT_EQ(conn2, ch.best_connection());
+
+  // Now another STUN message with an unknown address and use_candidate will
+  // nominate the best connection.
+  cricket::IceMessage request;
+  request.SetType(cricket::STUN_BINDING_REQUEST);
+  request.AddAttribute(new cricket::StunByteStringAttribute(
+      cricket::STUN_ATTR_USERNAME, kIceUfrag[1]));
+  uint32_t prflx_priority = cricket::ICE_TYPE_PREFERENCE_PRFLX << 24;
+  request.AddAttribute(new cricket::StunUInt32Attribute(
+      cricket::STUN_ATTR_PRIORITY, prflx_priority));
+  request.AddAttribute(
+      new cricket::StunByteStringAttribute(cricket::STUN_ATTR_USE_CANDIDATE));
+  cricket::Port* port = GetPort(&ch);
+  port->SignalUnknownAddress(port, rtc::SocketAddress("3.3.3.3", 3),
+                             cricket::PROTO_UDP, &request, kIceUfrag[1], false);
+  cricket::Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
+  ASSERT_TRUE(conn3 != nullptr);
+  EXPECT_EQ(conn2, ch.best_connection());  // Not writable yet.
+  conn3->ReceivedPingResponse();           // Become writable.
+  EXPECT_EQ(conn3, ch.best_connection());
+
+  // Now another data packet will not switch the best connection because the
+  // best connection was nominated by the controlling side.
+  conn2->ReceivedPing();
+  conn2->ReceivedPingResponse();
+  conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
+  EXPECT_EQ(conn3, ch.best_connection());
+}
+
+// When the current best connection is strong, lower-priority connections will
+// be pruned. Otherwise, lower-priority connections are kept.
+TEST_F(P2PTransportChannelPingTest, TestDontPruneWhenWeak) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("test channel", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.SetIceRole(cricket::ICEROLE_CONTROLLED);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 1));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  EXPECT_EQ(conn1, ch.best_connection());
+  conn1->ReceivedPingResponse();  // Becomes writable and receiving
+
+  // When a higher-priority, nominated candidate comes in, the connections with
+  // lower-priority are pruned.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 10));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  conn2->ReceivedPingResponse();  // Becomes writable and receiving
+  conn2->set_nominated(true);
+  conn2->SignalNominated(conn2);
+  EXPECT_TRUE_WAIT(conn1->pruned(), 3000);
+
+  ch.SetIceConfig(CreateIceConfig(500, false));
+  // Wait until conn2 becomes not receiving.
+  EXPECT_TRUE_WAIT(!conn2->receiving(), 3000);
+
+  ch.AddRemoteCandidate(CreateCandidate("3.3.3.3", 3, 1));
+  cricket::Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
+  ASSERT_TRUE(conn3 != nullptr);
+  // The best connection should still be conn2. Even through conn3 has lower
+  // priority and is not receiving/writable, it is not pruned because the best
+  // connection is not receiving.
+  WAIT(conn3->pruned(), 1000);
+  EXPECT_FALSE(conn3->pruned());
+}
+
+// Test that GetState returns the state correctly.
+TEST_F(P2PTransportChannelPingTest, TestGetState) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("test channel", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  EXPECT_EQ(cricket::TransportChannelState::STATE_INIT, ch.GetState());
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 100));
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 1));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn1 != nullptr);
+  ASSERT_TRUE(conn2 != nullptr);
+  // Now there are two connections, so the transport channel is connecting.
+  EXPECT_EQ(cricket::TransportChannelState::STATE_CONNECTING, ch.GetState());
+  // |conn1| becomes writable and receiving; it then should prune |conn2|.
+  conn1->ReceivedPingResponse();
+  EXPECT_TRUE_WAIT(conn2->pruned(), 1000);
+  EXPECT_EQ(cricket::TransportChannelState::STATE_COMPLETED, ch.GetState());
+  conn1->Prune();  // All connections are pruned.
+  // Need to wait until the channel state is updated.
+  EXPECT_EQ_WAIT(cricket::TransportChannelState::STATE_FAILED, ch.GetState(),
+                 1000);
+}
+
+// Test that when a low-priority connection is pruned, it is not deleted
+// right away, and it can become active and be pruned again.
+TEST_F(P2PTransportChannelPingTest, TestConnectionPrunedAgain) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("test channel", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.SetIceConfig(CreateIceConfig(1000, false));
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 100));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  EXPECT_EQ(conn1, ch.best_connection());
+  conn1->ReceivedPingResponse();  // Becomes writable and receiving
+
+  // Add a low-priority connection |conn2|, which will be pruned, but it will
+  // not be deleted right away. Once the current best connection becomes not
+  // receiving, |conn2| will start to ping and upon receiving the ping response,
+  // it will become the best connection.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 1));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  EXPECT_TRUE_WAIT(!conn2->active(), 1000);
+  // |conn2| should not send a ping yet.
+  EXPECT_EQ(cricket::Connection::STATE_WAITING, conn2->state());
+  EXPECT_EQ(cricket::TransportChannelState::STATE_COMPLETED, ch.GetState());
+  // Wait for |conn1| becoming not receiving.
+  EXPECT_TRUE_WAIT(!conn1->receiving(), 3000);
+  // Make sure conn2 is not deleted.
+  conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  EXPECT_EQ_WAIT(cricket::Connection::STATE_INPROGRESS, conn2->state(), 1000);
+  conn2->ReceivedPingResponse();
+  EXPECT_EQ_WAIT(conn2, ch.best_connection(), 1000);
+  EXPECT_EQ(cricket::TransportChannelState::STATE_CONNECTING, ch.GetState());
+
+  // When |conn1| comes back again, |conn2| will be pruned again.
+  conn1->ReceivedPingResponse();
+  EXPECT_EQ_WAIT(conn1, ch.best_connection(), 1000);
+  EXPECT_TRUE_WAIT(!conn2->active(), 1000);
+  EXPECT_EQ(cricket::TransportChannelState::STATE_COMPLETED, ch.GetState());
+}
+
+// Test that if all connections in a channel has timed out on writing, they
+// will all be deleted. We use Prune to simulate write_time_out.
+TEST_F(P2PTransportChannelPingTest, TestDeleteConnectionsIfAllWriteTimedout) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("test channel", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.Connect();
+  ch.MaybeStartGathering();
+  // Have one connection only but later becomes write-time-out.
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 100));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  conn1->ReceivedPing();  // Becomes receiving
+  conn1->Prune();
+  EXPECT_TRUE_WAIT(ch.connections().empty(), 1000);
+
+  // Have two connections but both become write-time-out later.
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 1));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  conn2->ReceivedPing();  // Becomes receiving
+  ch.AddRemoteCandidate(CreateCandidate("3.3.3.3", 3, 2));
+  cricket::Connection* conn3 = WaitForConnectionTo(&ch, "3.3.3.3", 3);
+  ASSERT_TRUE(conn3 != nullptr);
+  conn3->ReceivedPing();  // Becomes receiving
+  // Now prune both conn2 and conn3; they will be deleted soon.
+  conn2->Prune();
+  conn3->Prune();
+  EXPECT_TRUE_WAIT(ch.connections().empty(), 1000);
+}
+
+// Test that after a port allocator session is started, it will be stopped
+// when a new connection becomes writable and receiving. Also test that this
+// holds even if the transport channel did not lose the writability.
+TEST_F(P2PTransportChannelPingTest, TestStopPortAllocatorSessions) {
+  cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
+  cricket::P2PTransportChannel ch("test channel", 1, nullptr, &pa);
+  PrepareChannel(&ch);
+  ch.SetIceConfig(CreateIceConfig(2000, false));
+  ch.Connect();
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("1.1.1.1", 1, 100));
+  cricket::Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1);
+  ASSERT_TRUE(conn1 != nullptr);
+  conn1->ReceivedPingResponse();  // Becomes writable and receiving
+  EXPECT_TRUE(!ch.allocator_session()->IsGettingPorts());
+
+  // Restart gathering even if the transport channel is still writable.
+  // It should stop getting ports after a new connection becomes strongly
+  // connected.
+  ch.SetIceCredentials(kIceUfrag[1], kIcePwd[1]);
+  ch.MaybeStartGathering();
+  ch.AddRemoteCandidate(CreateCandidate("2.2.2.2", 2, 100));
+  cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
+  ASSERT_TRUE(conn2 != nullptr);
+  conn2->ReceivedPingResponse();  // Becomes writable and receiving
+  EXPECT_TRUE(!ch.allocator_session()->IsGettingPorts());
+}
diff --git a/media/webrtc/trunk/webrtc/p2p/base/packetsocketfactory.h b/media/webrtc/trunk/webrtc/p2p/base/packetsocketfactory.h
index d2d7b1b1c9..54037241b0 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/packetsocketfactory.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/packetsocketfactory.h
@@ -30,12 +30,12 @@ class PacketSocketFactory {
   virtual ~PacketSocketFactory() { }
 
   virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
-                                             uint16 min_port,
-                                             uint16 max_port) = 0;
+                                             uint16_t min_port,
+                                             uint16_t max_port) = 0;
   virtual AsyncPacketSocket* CreateServerTcpSocket(
       const SocketAddress& local_address,
-      uint16 min_port,
-      uint16 max_port,
+      uint16_t min_port,
+      uint16_t max_port,
       int opts) = 0;
 
   // TODO: |proxy_info| and |user_agent| should be set
@@ -50,7 +50,7 @@ class PacketSocketFactory {
   virtual AsyncResolverInterface* CreateAsyncResolver() = 0;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(PacketSocketFactory);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PacketSocketFactory);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/p2p/base/port.cc b/media/webrtc/trunk/webrtc/p2p/base/port.cc
index c321f83e80..9dd5c83fed 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/port.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/port.cc
@@ -29,64 +29,44 @@ namespace {
 // Determines whether we have seen at least the given maximum number of
 // pings fail to have a response.
 inline bool TooManyFailures(
-    const std::vector<uint32>& pings_since_last_response,
-    uint32 maximum_failures,
-    uint32 rtt_estimate,
-    uint32 now) {
-
+    const std::vector<cricket::Connection::SentPing>& pings_since_last_response,
+    uint32_t maximum_failures,
+    uint32_t rtt_estimate,
+    uint32_t now) {
   // If we haven't sent that many pings, then we can't have failed that many.
   if (pings_since_last_response.size() < maximum_failures)
     return false;
 
   // Check if the window in which we would expect a response to the ping has
   // already elapsed.
-  return pings_since_last_response[maximum_failures - 1] + rtt_estimate < now;
+  uint32_t expected_response_time =
+      pings_since_last_response[maximum_failures - 1].sent_time + rtt_estimate;
+  return now > expected_response_time;
 }
 
 // Determines whether we have gone too long without seeing any response.
 inline bool TooLongWithoutResponse(
-    const std::vector<uint32>& pings_since_last_response,
-    uint32 maximum_time,
-    uint32 now) {
-
+    const std::vector<cricket::Connection::SentPing>& pings_since_last_response,
+    uint32_t maximum_time,
+    uint32_t now) {
   if (pings_since_last_response.size() == 0)
     return false;
 
-  return pings_since_last_response[0] + maximum_time < now;
-}
-
-// GICE(ICEPROTO_GOOGLE) requires different username for RTP and RTCP.
-// This function generates a different username by +1 on the last character of
-// the given username (|rtp_ufrag|).
-std::string GetRtcpUfragFromRtpUfrag(const std::string& rtp_ufrag) {
-  ASSERT(!rtp_ufrag.empty());
-  if (rtp_ufrag.empty()) {
-    return rtp_ufrag;
-  }
-  // Change the last character to the one next to it in the base64 table.
-  char new_last_char;
-  if (!rtc::Base64::GetNextBase64Char(rtp_ufrag[rtp_ufrag.size() - 1],
-                                            &new_last_char)) {
-    // Should not be here.
-    ASSERT(false);
-  }
-  std::string rtcp_ufrag = rtp_ufrag;
-  rtcp_ufrag[rtcp_ufrag.size() - 1] = new_last_char;
-  ASSERT(rtcp_ufrag != rtp_ufrag);
-  return rtcp_ufrag;
+  auto first = pings_since_last_response[0];
+  return now > (first.sent_time + maximum_time);
 }
 
 // We will restrict RTT estimates (when used for determining state) to be
 // within a reasonable range.
-const uint32 MINIMUM_RTT = 100;   // 0.1 seconds
-const uint32 MAXIMUM_RTT = 3000;  // 3 seconds
+const uint32_t MINIMUM_RTT = 100;   // 0.1 seconds
+const uint32_t MAXIMUM_RTT = 3000;  // 3 seconds
 
 // When we don't have any RTT data, we have to pick something reasonable.  We
 // use a large value just in case the connection is really slow.
-const uint32 DEFAULT_RTT = MAXIMUM_RTT;
+const uint32_t DEFAULT_RTT = MAXIMUM_RTT;
 
 // Computes our estimate of the RTT given the current estimate.
-inline uint32 ConservativeRTTEstimate(uint32 rtt) {
+inline uint32_t ConservativeRTTEstimate(uint32_t rtt) {
   return std::max(MINIMUM_RTT, std::min(MAXIMUM_RTT, 2 * rtt));
 }
 
@@ -95,9 +75,6 @@ const int RTT_RATIO = 3;  // 3 : 1
 
 // The delay before we begin checking if this port is useless.
 const int kPortTimeoutDelay = 30 * 1000;  // 30 seconds
-
-// Used by the Connection.
-const uint32 MSG_DELETE = 1;
 }
 
 namespace cricket {
@@ -149,7 +126,7 @@ static std::string ComputeFoundation(
     const rtc::SocketAddress& base_address) {
   std::ostringstream ost;
   ost << type << base_address.ipaddr().ToString() << protocol;
-  return rtc::ToString<uint32>(rtc::ComputeCrc32(ost.str()));
+  return rtc::ToString<uint32_t>(rtc::ComputeCrc32(ost.str()));
 }
 
 Port::Port(rtc::Thread* thread,
@@ -171,7 +148,6 @@ Port::Port(rtc::Thread* thread,
       password_(password),
       timeout_delay_(kPortTimeoutDelay),
       enable_port_packets_(false),
-      ice_protocol_(ICEPROTO_HYBRID),
       ice_role_(ICEROLE_UNKNOWN),
       tiebreaker_(0),
       shared_socket_(true),
@@ -184,8 +160,8 @@ Port::Port(rtc::Thread* thread,
            rtc::PacketSocketFactory* factory,
            rtc::Network* network,
            const rtc::IPAddress& ip,
-           uint16 min_port,
-           uint16 max_port,
+           uint16_t min_port,
+           uint16_t max_port,
            const std::string& username_fragment,
            const std::string& password)
     : thread_(thread),
@@ -202,7 +178,6 @@ Port::Port(rtc::Thread* thread,
       password_(password),
       timeout_delay_(kPortTimeoutDelay),
       enable_port_packets_(false),
-      ice_protocol_(ICEPROTO_HYBRID),
       ice_role_(ICEROLE_UNKNOWN),
       tiebreaker_(0),
       shared_socket_(false),
@@ -212,7 +187,9 @@ Port::Port(rtc::Thread* thread,
 }
 
 void Port::Construct() {
-  // If the username_fragment and password are empty, we should just create one.
+  // TODO(pthatcher): Remove this old behavior once we're sure no one
+  // relies on it.  If the username_fragment and password are empty,
+  // we should just create one.
   if (ice_username_fragment_.empty()) {
     ASSERT(password_.empty());
     ice_username_fragment_ = rtc::CreateRandomString(ICE_UFRAG_LENGTH);
@@ -233,7 +210,7 @@ Port::~Port() {
     ++iter;
   }
 
-  for (uint32 i = 0; i < list.size(); i++)
+  for (uint32_t i = 0; i < list.size(); i++)
     delete list[i];
 }
 
@@ -249,10 +226,11 @@ void Port::AddAddress(const rtc::SocketAddress& address,
                       const rtc::SocketAddress& base_address,
                       const rtc::SocketAddress& related_address,
                       const std::string& protocol,
+                      const std::string& relay_protocol,
                       const std::string& tcptype,
                       const std::string& type,
-                      uint32 type_preference,
-                      uint32 relay_preference,
+                      uint32_t type_preference,
+                      uint32_t relay_preference,
                       bool final) {
   if (protocol == TCP_PROTOCOL_NAME && type == LOCAL_PORT_TYPE) {
     ASSERT(!tcptype.empty());
@@ -263,6 +241,7 @@ void Port::AddAddress(const rtc::SocketAddress& address,
   c.set_component(component_);
   c.set_type(type);
   c.set_protocol(protocol);
+  c.set_relay_protocol(relay_protocol);
   c.set_tcptype(tcptype);
   c.set_address(address);
   c.set_priority(c.GetPriority(type_preference, network_->preference(),
@@ -307,9 +286,12 @@ void Port::OnReadPacket(
   } else if (!msg) {
     // STUN message handled already
   } else if (msg->type() == STUN_BINDING_REQUEST) {
+    LOG(LS_INFO) << "Received STUN ping "
+                 << " id=" << rtc::hex_encode(msg->transaction_id())
+                 << " from unknown address " << addr.ToSensitiveString();
+
     // Check for role conflicts.
-    if (IsStandardIce() &&
-        !MaybeIceRoleConflict(addr, msg.get(), remote_username)) {
+    if (!MaybeIceRoleConflict(addr, msg.get(), remote_username)) {
       LOG(LS_INFO) << "Received conflicting role from the peer.";
       return;
     }
@@ -340,18 +322,6 @@ size_t Port::AddPrflxCandidate(const Candidate& local) {
   return (candidates_.size() - 1);
 }
 
-bool Port::IsStandardIce() const {
-  return (ice_protocol_ == ICEPROTO_RFC5245);
-}
-
-bool Port::IsGoogleIce() const {
-  return (ice_protocol_ == ICEPROTO_GOOGLE);
-}
-
-bool Port::IsHybridIce() const {
-  return (ice_protocol_ == ICEPROTO_HYBRID);
-}
-
 bool Port::GetStunMessage(const char* data, size_t size,
                           const rtc::SocketAddress& addr,
                           IceMessage** out_msg, std::string* out_username) {
@@ -365,7 +335,7 @@ bool Port::GetStunMessage(const char* data, size_t size,
 
   // Don't bother parsing the packet if we can tell it's not STUN.
   // In ICE mode, all STUN packets will have a valid fingerprint.
-  if (IsStandardIce() && !StunMessage::ValidateFingerprint(data, size)) {
+  if (!StunMessage::ValidateFingerprint(data, size)) {
     return false;
   }
 
@@ -381,8 +351,7 @@ bool Port::GetStunMessage(const char* data, size_t size,
     // Check for the presence of USERNAME and MESSAGE-INTEGRITY (if ICE) first.
     // If not present, fail with a 400 Bad Request.
     if (!stun_msg->GetByteString(STUN_ATTR_USERNAME) ||
-        (IsStandardIce() &&
-         !stun_msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY))) {
+        !stun_msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY)) {
       LOG_J(LS_ERROR, this) << "Received STUN request without username/M-I "
                             << "from " << addr.ToSensitiveString();
       SendBindingErrorResponse(stun_msg.get(), addr, STUN_ERROR_BAD_REQUEST,
@@ -393,9 +362,7 @@ bool Port::GetStunMessage(const char* data, size_t size,
     // If the username is bad or unknown, fail with a 401 Unauthorized.
     std::string local_ufrag;
     std::string remote_ufrag;
-    IceProtocolType remote_protocol_type;
-    if (!ParseStunUsername(stun_msg.get(), &local_ufrag, &remote_ufrag,
-                           &remote_protocol_type) ||
+    if (!ParseStunUsername(stun_msg.get(), &local_ufrag, &remote_ufrag) ||
         local_ufrag != username_fragment()) {
       LOG_J(LS_ERROR, this) << "Received STUN request with bad local username "
                             << local_ufrag << " from "
@@ -405,18 +372,8 @@ bool Port::GetStunMessage(const char* data, size_t size,
       return true;
     }
 
-    // Port is initialized to GOOGLE-ICE protocol type. If pings from remote
-    // are received before the signal message, protocol type may be different.
-    // Based on the STUN username, we can determine what's the remote protocol.
-    // This also enables us to send the response back using the same protocol
-    // as the request.
-    if (IsHybridIce()) {
-      SetIceProtocolType(remote_protocol_type);
-    }
-
     // If ICE, and the MESSAGE-INTEGRITY is bad, fail with a 401 Unauthorized
-    if (IsStandardIce() &&
-        !stun_msg->ValidateMessageIntegrity(data, size, password_)) {
+    if (!stun_msg->ValidateMessageIntegrity(data, size, password_)) {
       LOG_J(LS_ERROR, this) << "Received STUN request with bad M-I "
                             << "from " << addr.ToSensitiveString()
                             << ", password_=" << password_;
@@ -468,7 +425,8 @@ bool Port::IsCompatibleAddress(const rtc::SocketAddress& addr) {
     return false;
   }
   // Link-local IPv6 ports can only connect to other link-local IPv6 ports.
-  if (family == AF_INET6 && (IPIsPrivate(ip()) != IPIsPrivate(addr.ipaddr()))) {
+  if (family == AF_INET6 &&
+      (IPIsLinkLocal(ip()) != IPIsLinkLocal(addr.ipaddr()))) {
     return false;
   }
   return true;
@@ -476,8 +434,7 @@ bool Port::IsCompatibleAddress(const rtc::SocketAddress& addr) {
 
 bool Port::ParseStunUsername(const StunMessage* stun_msg,
                              std::string* local_ufrag,
-                             std::string* remote_ufrag,
-                             IceProtocolType* remote_protocol_type) const {
+                             std::string* remote_ufrag) const {
   // The packet must include a username that either begins or ends with our
   // fragment.  It should begin with our fragment if it is a request and it
   // should end with our fragment if it is a response.
@@ -488,34 +445,15 @@ bool Port::ParseStunUsername(const StunMessage* stun_msg,
   if (username_attr == NULL)
     return false;
 
-  const std::string username_attr_str = username_attr->GetString();
-  size_t colon_pos = username_attr_str.find(":");
-  // If we are in hybrid mode set the appropriate ice protocol type based on
-  // the username argument style.
-  if (IsHybridIce()) {
-    *remote_protocol_type = (colon_pos != std::string::npos) ?
-        ICEPROTO_RFC5245 : ICEPROTO_GOOGLE;
-  } else {
-    *remote_protocol_type = ice_protocol_;
+  // RFRAG:LFRAG
+  const std::string username = username_attr->GetString();
+  size_t colon_pos = username.find(":");
+  if (colon_pos == std::string::npos) {
+    return false;
   }
-  if (*remote_protocol_type == ICEPROTO_RFC5245) {
-    if (colon_pos != std::string::npos) {  // RFRAG:LFRAG
-      *local_ufrag = username_attr_str.substr(0, colon_pos);
-      *remote_ufrag = username_attr_str.substr(
-          colon_pos + 1, username_attr_str.size());
-    } else {
-      return false;
-    }
-  } else if (*remote_protocol_type == ICEPROTO_GOOGLE) {
-    int remote_frag_len = static_cast<int>(username_attr_str.size());
-    remote_frag_len -= static_cast<int>(username_fragment().size());
-    if (remote_frag_len < 0)
-      return false;
 
-    *local_ufrag = username_attr_str.substr(0, username_fragment().size());
-    *remote_ufrag = username_attr_str.substr(
-        username_fragment().size(), username_attr_str.size());
-  }
+  *local_ufrag = username.substr(0, colon_pos);
+  *remote_ufrag = username.substr(colon_pos + 1, username.size());
   return true;
 }
 
@@ -525,7 +463,7 @@ bool Port::MaybeIceRoleConflict(
   // Validate ICE_CONTROLLING or ICE_CONTROLLED attributes.
   bool ret = true;
   IceRole remote_ice_role = ICEROLE_UNKNOWN;
-  uint64 remote_tiebreaker = 0;
+  uint64_t remote_tiebreaker = 0;
   const StunUInt64Attribute* stun_attr =
       stun_msg->GetUInt64(STUN_ATTR_ICE_CONTROLLING);
   if (stun_attr) {
@@ -584,10 +522,7 @@ void Port::CreateStunUsername(const std::string& remote_username,
                               std::string* stun_username_attr_str) const {
   stun_username_attr_str->clear();
   *stun_username_attr_str = remote_username;
-  if (IsStandardIce()) {
-    // Connectivity checks from L->R will have username RFRAG:LFRAG.
-    stun_username_attr_str->append(":");
-  }
+  stun_username_attr_str->append(":");
   stun_username_attr_str->append(username_fragment());
 }
 
@@ -623,35 +558,33 @@ void Port::SendBindingResponse(StunMessage* request,
     }
   }
 
-  // Only GICE messages have USERNAME and MAPPED-ADDRESS in the response.
-  // ICE messages use XOR-MAPPED-ADDRESS, and add MESSAGE-INTEGRITY.
-  if (IsStandardIce()) {
-    response.AddAttribute(
-        new StunXorAddressAttribute(STUN_ATTR_XOR_MAPPED_ADDRESS, addr));
-    response.AddMessageIntegrity(password_);
-    response.AddFingerprint();
-  } else if (IsGoogleIce()) {
-    response.AddAttribute(
-        new StunAddressAttribute(STUN_ATTR_MAPPED_ADDRESS, addr));
-    response.AddAttribute(new StunByteStringAttribute(
-        STUN_ATTR_USERNAME, username_attr->GetString()));
-  }
+  response.AddAttribute(
+      new StunXorAddressAttribute(STUN_ATTR_XOR_MAPPED_ADDRESS, addr));
+  response.AddMessageIntegrity(password_);
+  response.AddFingerprint();
 
   // Send the response message.
   rtc::ByteBuffer buf;
   response.Write(&buf);
   rtc::PacketOptions options(DefaultDscpValue());
-  if (SendTo(buf.Data(), buf.Length(), addr, options, false) < 0) {
-    LOG_J(LS_ERROR, this) << "Failed to send STUN ping response to "
-                          << addr.ToSensitiveString();
+  auto err = SendTo(buf.Data(), buf.Length(), addr, options, false);
+  if (err < 0) {
+    LOG_J(LS_ERROR, this)
+        << "Failed to send STUN ping response"
+        << ", to=" << addr.ToSensitiveString()
+        << ", err=" << err
+        << ", id=" << rtc::hex_encode(response.transaction_id());
+  } else {
+    // Log at LS_INFO if we send a stun ping response on an unwritable
+    // connection.
+    Connection* conn = GetConnection(addr);
+    rtc::LoggingSeverity sev = (conn && !conn->writable()) ?
+        rtc::LS_INFO : rtc::LS_VERBOSE;
+    LOG_JV(sev, this)
+        << "Sent STUN ping response"
+        << ", to=" << addr.ToSensitiveString()
+        << ", id=" << rtc::hex_encode(response.transaction_id());
   }
-
-  // The fact that we received a successful request means that this connection
-  // (if one exists) should now be readable.
-  Connection* conn = GetConnection(addr);
-  ASSERT(conn != NULL);
-  if (conn)
-    conn->ReceivedPing();
 }
 
 void Port::SendBindingErrorResponse(StunMessage* request,
@@ -667,30 +600,16 @@ void Port::SendBindingErrorResponse(StunMessage* request,
   // When doing GICE, we need to write out the error code incorrectly to
   // maintain backwards compatiblility.
   StunErrorCodeAttribute* error_attr = StunAttribute::CreateErrorCode();
-  if (IsStandardIce()) {
-    error_attr->SetCode(error_code);
-  } else if (IsGoogleIce()) {
-    error_attr->SetClass(error_code / 256);
-    error_attr->SetNumber(error_code % 256);
-  }
+  error_attr->SetCode(error_code);
   error_attr->SetReason(reason);
   response.AddAttribute(error_attr);
 
-  if (IsStandardIce()) {
-    // Per Section 10.1.2, certain error cases don't get a MESSAGE-INTEGRITY,
-    // because we don't have enough information to determine the shared secret.
-    if (error_code != STUN_ERROR_BAD_REQUEST &&
-        error_code != STUN_ERROR_UNAUTHORIZED)
-      response.AddMessageIntegrity(password_);
-    response.AddFingerprint();
-  } else if (IsGoogleIce()) {
-    // GICE responses include a username, if one exists.
-    const StunByteStringAttribute* username_attr =
-        request->GetByteString(STUN_ATTR_USERNAME);
-    if (username_attr)
-      response.AddAttribute(new StunByteStringAttribute(
-          STUN_ATTR_USERNAME, username_attr->GetString()));
-  }
+  // Per Section 10.1.2, certain error cases don't get a MESSAGE-INTEGRITY,
+  // because we don't have enough information to determine the shared secret.
+  if (error_code != STUN_ERROR_BAD_REQUEST &&
+      error_code != STUN_ERROR_UNAUTHORIZED)
+    response.AddMessageIntegrity(password_);
+  response.AddFingerprint();
 
   // Send the response message.
   rtc::ByteBuffer buf;
@@ -702,8 +621,10 @@ void Port::SendBindingErrorResponse(StunMessage* request,
 }
 
 void Port::OnMessage(rtc::Message *pmsg) {
-  ASSERT(pmsg->message_id == MSG_CHECKTIMEOUT);
-  CheckTimeout();
+  ASSERT(pmsg->message_id == MSG_DEAD);
+  if (dead()) {
+    Destroy();
+  }
 }
 
 std::string Port::ToString() const {
@@ -724,12 +645,13 @@ void Port::OnConnectionDestroyed(Connection* conn) {
   ASSERT(iter != connections_.end());
   connections_.erase(iter);
 
-  // On the controlled side, ports time out, but only after all connections
-  // fail.  Note: If a new connection is added after this message is posted,
-  // but it fails and is removed before kPortTimeoutDelay, then this message
-  //  will still cause the Port to be destroyed.
-  if (ice_role_ == ICEROLE_CONTROLLED)
-    thread_->PostDelayed(timeout_delay_, this, MSG_CHECKTIMEOUT);
+  // On the controlled side, ports time out after all connections fail.
+  // Note: If a new connection is added after this message is posted, but it
+  // fails and is removed before kPortTimeoutDelay, then this message will
+  // still cause the Port to be destroyed.
+  if (dead()) {
+    thread_->PostDelayed(timeout_delay_, this, MSG_DEAD);
+  }
 }
 
 void Port::Destroy() {
@@ -739,24 +661,8 @@ void Port::Destroy() {
   delete this;
 }
 
-void Port::CheckTimeout() {
-  ASSERT(ice_role_ == ICEROLE_CONTROLLED);
-  // If this port has no connections, then there's no reason to keep it around.
-  // When the connections time out (both read and write), they will delete
-  // themselves, so if we have any connections, they are either readable or
-  // writable (or still connecting).
-  if (connections_.empty())
-    Destroy();
-}
-
 const std::string Port::username_fragment() const {
-  if (!IsStandardIce() &&
-      component_ == ICE_CANDIDATE_COMPONENT_RTCP) {
-    // In GICE mode, we should adjust username fragment for rtcp component.
-    return GetRtcpUfragFromRtpUfrag(ice_username_fragment_);
-  } else {
-    return ice_username_fragment_;
-  }
+  return ice_username_fragment_;
 }
 
 // A ConnectionRequest is a simple STUN ping used to determine writability.
@@ -770,7 +676,7 @@ class ConnectionRequest : public StunRequest {
   virtual ~ConnectionRequest() {
   }
 
-  virtual void Prepare(StunMessage* request) {
+  void Prepare(StunMessage* request) override {
     request->SetType(STUN_BINDING_REQUEST);
     std::string username;
     connection_->port()->CreateStunUsername(
@@ -782,66 +688,68 @@ class ConnectionRequest : public StunRequest {
     if (connection_->port()->send_retransmit_count_attribute()) {
       request->AddAttribute(new StunUInt32Attribute(
           STUN_ATTR_RETRANSMIT_COUNT,
-          static_cast<uint32>(
-              connection_->pings_since_last_response_.size() - 1)));
+          static_cast<uint32_t>(connection_->pings_since_last_response_.size() -
+                                1)));
     }
 
-    // Adding ICE-specific attributes to the STUN request message.
-    if (connection_->port()->IsStandardIce()) {
-      // Adding ICE_CONTROLLED or ICE_CONTROLLING attribute based on the role.
-      if (connection_->port()->GetIceRole() == ICEROLE_CONTROLLING) {
-        request->AddAttribute(new StunUInt64Attribute(
-            STUN_ATTR_ICE_CONTROLLING, connection_->port()->IceTiebreaker()));
-        // Since we are trying aggressive nomination, sending USE-CANDIDATE
-        // attribute in every ping.
-        // If we are dealing with a ice-lite end point, nomination flag
-        // in Connection will be set to false by default. Once the connection
-        // becomes "best connection", nomination flag will be turned on.
-        if (connection_->use_candidate_attr()) {
-          request->AddAttribute(new StunByteStringAttribute(
-              STUN_ATTR_USE_CANDIDATE));
-        }
-      } else if (connection_->port()->GetIceRole() == ICEROLE_CONTROLLED) {
-        request->AddAttribute(new StunUInt64Attribute(
-            STUN_ATTR_ICE_CONTROLLED, connection_->port()->IceTiebreaker()));
-      } else {
-        ASSERT(false);
+    // Adding ICE_CONTROLLED or ICE_CONTROLLING attribute based on the role.
+    if (connection_->port()->GetIceRole() == ICEROLE_CONTROLLING) {
+      request->AddAttribute(new StunUInt64Attribute(
+          STUN_ATTR_ICE_CONTROLLING, connection_->port()->IceTiebreaker()));
+      // Since we are trying aggressive nomination, sending USE-CANDIDATE
+      // attribute in every ping.
+      // If we are dealing with a ice-lite end point, nomination flag
+      // in Connection will be set to false by default. Once the connection
+      // becomes "best connection", nomination flag will be turned on.
+      if (connection_->use_candidate_attr()) {
+        request->AddAttribute(new StunByteStringAttribute(
+            STUN_ATTR_USE_CANDIDATE));
       }
-
-      // Adding PRIORITY Attribute.
-      // Changing the type preference to Peer Reflexive and local preference
-      // and component id information is unchanged from the original priority.
-      // priority = (2^24)*(type preference) +
-      //           (2^8)*(local preference) +
-      //           (2^0)*(256 - component ID)
-      uint32 prflx_priority = ICE_TYPE_PREFERENCE_PRFLX << 24 |
-          (connection_->local_candidate().priority() & 0x00FFFFFF);
-      request->AddAttribute(
-          new StunUInt32Attribute(STUN_ATTR_PRIORITY, prflx_priority));
-
-      // Adding Message Integrity attribute.
-      request->AddMessageIntegrity(connection_->remote_candidate().password());
-      // Adding Fingerprint.
-      request->AddFingerprint();
+    } else if (connection_->port()->GetIceRole() == ICEROLE_CONTROLLED) {
+      request->AddAttribute(new StunUInt64Attribute(
+          STUN_ATTR_ICE_CONTROLLED, connection_->port()->IceTiebreaker()));
+    } else {
+      ASSERT(false);
     }
+
+    // Adding PRIORITY Attribute.
+    // Changing the type preference to Peer Reflexive and local preference
+    // and component id information is unchanged from the original priority.
+    // priority = (2^24)*(type preference) +
+    //           (2^8)*(local preference) +
+    //           (2^0)*(256 - component ID)
+    uint32_t prflx_priority =
+        ICE_TYPE_PREFERENCE_PRFLX << 24 |
+        (connection_->local_candidate().priority() & 0x00FFFFFF);
+    request->AddAttribute(
+        new StunUInt32Attribute(STUN_ATTR_PRIORITY, prflx_priority));
+
+    // Adding Message Integrity attribute.
+    request->AddMessageIntegrity(connection_->remote_candidate().password());
+    // Adding Fingerprint.
+    request->AddFingerprint();
   }
 
-  virtual void OnResponse(StunMessage* response) {
+  void OnResponse(StunMessage* response) override {
     connection_->OnConnectionRequestResponse(this, response);
   }
 
-  virtual void OnErrorResponse(StunMessage* response) {
+  void OnErrorResponse(StunMessage* response) override {
     connection_->OnConnectionRequestErrorResponse(this, response);
   }
 
-  virtual void OnTimeout() {
+  void OnTimeout() override {
     connection_->OnConnectionRequestTimeout(this);
   }
 
-  virtual int GetNextDelay() {
+  void OnSent() override {
+    connection_->OnConnectionRequestSent(this);
     // Each request is sent only once.  After a single delay , the request will
     // time out.
     timeout_ = true;
+  }
+
+  int resend_delay() override {
     return CONNECTION_RESPONSE_TIMEOUT;
   }
 
@@ -859,11 +767,12 @@ Connection::Connection(Port* port,
     : port_(port),
       local_candidate_index_(index),
       remote_candidate_(remote_candidate),
-      read_state_(STATE_READ_INIT),
       write_state_(STATE_WRITE_INIT),
+      receiving_(false),
       connected_(true),
       pruned_(false),
       use_candidate_attr_(false),
+      nominated_(false),
       remote_ice_mode_(ICEMODE_FULL),
       requests_(port->thread()),
       rtt_(DEFAULT_RTT),
@@ -871,10 +780,14 @@ Connection::Connection(Port* port,
       last_ping_received_(0),
       last_data_received_(0),
       last_ping_response_received_(0),
+      recv_rate_tracker_(100u, 10u),
+      send_rate_tracker_(100u, 10u),
       sent_packets_discarded_(0),
       sent_packets_total_(0),
       reported_(false),
-      state_(STATE_WAITING) {
+      state_(STATE_WAITING),
+      receiving_timeout_(WEAK_CONNECTION_RECEIVE_TIMEOUT),
+      time_created_ms_(rtc::Time()) {
   // All of our connections start in WAITING state.
   // TODO(mallinath) - Start connections from STATE_FROZEN.
   // Wire up to send stun packets
@@ -890,8 +803,8 @@ const Candidate& Connection::local_candidate() const {
   return port_->Candidates()[local_candidate_index_];
 }
 
-uint64 Connection::priority() const {
-  uint64 priority = 0;
+uint64_t Connection::priority() const {
+  uint64_t priority = 0;
   // RFC 5245 - 5.7.2.  Computing Pair Priority and Ordering Pairs
   // Let G be the priority for the candidate provided by the controlling
   // agent.  Let D be the priority for the candidate provided by the
@@ -899,8 +812,8 @@ uint64 Connection::priority() const {
   // pair priority = 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0)
   IceRole role = port_->GetIceRole();
   if (role != ICEROLE_UNKNOWN) {
-    uint32 g = 0;
-    uint32 d = 0;
+    uint32_t g = 0;
+    uint32_t d = 0;
     if (role == ICEROLE_CONTROLLING) {
       g = local_candidate().priority();
       d = remote_candidate_.priority();
@@ -915,16 +828,6 @@ uint64 Connection::priority() const {
   return priority;
 }
 
-void Connection::set_read_state(ReadState value) {
-  ReadState old_value = read_state_;
-  read_state_ = value;
-  if (value != old_value) {
-    LOG_J(LS_VERBOSE, this) << "set_read_state";
-    SignalStateChange(this);
-    CheckTimeout();
-  }
-}
-
 void Connection::set_write_state(WriteState value) {
   WriteState old_value = write_state_;
   write_state_ = value;
@@ -932,7 +835,14 @@ void Connection::set_write_state(WriteState value) {
     LOG_J(LS_VERBOSE, this) << "set_write_state from: " << old_value << " to "
                             << value;
     SignalStateChange(this);
-    CheckTimeout();
+  }
+}
+
+void Connection::set_receiving(bool value) {
+  if (value != receiving_) {
+    LOG_J(LS_VERBOSE, this) << "set_receiving to " << value;
+    receiving_ = value;
+    SignalStateChange(this);
   }
 }
 
@@ -948,7 +858,8 @@ void Connection::set_connected(bool value) {
   bool old_value = connected_;
   connected_ = value;
   if (value != old_value) {
-    LOG_J(LS_VERBOSE, this) << "set_connected";
+    LOG_J(LS_VERBOSE, this) << "set_connected from: " << old_value << " to "
+                            << value;
   }
 }
 
@@ -959,9 +870,12 @@ void Connection::set_use_candidate_attr(bool enable) {
 void Connection::OnSendStunPacket(const void* data, size_t size,
                                   StunRequest* req) {
   rtc::PacketOptions options(port_->DefaultDscpValue());
-  if (port_->SendTo(data, size, remote_candidate_.address(),
-                    options, false) < 0) {
-    LOG_J(LS_WARNING, this) << "Failed to send STUN ping " << req->id();
+  auto err = port_->SendTo(
+      data, size, remote_candidate_.address(), options, false);
+  if (err < 0) {
+    LOG_J(LS_WARNING, this) << "Failed to send STUN ping "
+                            << " err=" << err
+                            << " id=" << rtc::hex_encode(req->id());
   }
 }
 
@@ -972,62 +886,34 @@ void Connection::OnReadPacket(
   const rtc::SocketAddress& addr(remote_candidate_.address());
   if (!port_->GetStunMessage(data, size, addr, msg.accept(), &remote_ufrag)) {
     // The packet did not parse as a valid STUN message
+    // This is a data packet, pass it along.
+    set_receiving(true);
+    last_data_received_ = rtc::Time();
+    recv_rate_tracker_.AddSamples(size);
+    SignalReadPacket(this, data, size, packet_time);
 
-    // If this connection is readable, then pass along the packet.
-    if (read_state_ == STATE_READABLE) {
-      // readable means data from this address is acceptable
-      // Send it on!
-
-      last_data_received_ = rtc::Time();
-      recv_rate_tracker_.Update(size);
-      SignalReadPacket(this, data, size, packet_time);
-
-      // If timed out sending writability checks, start up again
-      if (!pruned_ && (write_state_ == STATE_WRITE_TIMEOUT)) {
-        LOG(LS_WARNING) << "Received a data packet on a timed-out Connection. "
-                        << "Resetting state to STATE_WRITE_INIT.";
-        set_write_state(STATE_WRITE_INIT);
-      }
-    } else {
-      // Not readable means the remote address hasn't sent a valid
-      // binding request yet.
-
-      LOG_J(LS_WARNING, this)
-        << "Received non-STUN packet from an unreadable connection.";
+    // If timed out sending writability checks, start up again
+    if (!pruned_ && (write_state_ == STATE_WRITE_TIMEOUT)) {
+      LOG(LS_WARNING) << "Received a data packet on a timed-out Connection. "
+                      << "Resetting state to STATE_WRITE_INIT.";
+      set_write_state(STATE_WRITE_INIT);
     }
   } else if (!msg) {
     // The packet was STUN, but failed a check and was handled internally.
   } else {
     // The packet is STUN and passed the Port checks.
     // Perform our own checks to ensure this packet is valid.
-    // If this is a STUN request, then update the readable bit and respond.
+    // If this is a STUN request, then update the receiving bit and respond.
     // If this is a STUN response, then update the writable bit.
+    // Log at LS_INFO if we receive a ping on an unwritable connection.
+    rtc::LoggingSeverity sev = (!writable() ? rtc::LS_INFO : rtc::LS_VERBOSE);
     switch (msg->type()) {
       case STUN_BINDING_REQUEST:
+        LOG_JV(sev, this) << "Received STUN ping"
+                          << ", id=" << rtc::hex_encode(msg->transaction_id());
+
         if (remote_ufrag == remote_candidate_.username()) {
-          // Check for role conflicts.
-          if (port_->IsStandardIce() &&
-              !port_->MaybeIceRoleConflict(addr, msg.get(), remote_ufrag)) {
-            // Received conflicting role from the peer.
-            LOG(LS_INFO) << "Received conflicting role from the peer.";
-            return;
-          }
-
-          // Incoming, validated stun request from remote peer.
-          // This call will also set the connection readable.
-          port_->SendBindingResponse(msg.get(), addr);
-
-          // If timed out sending writability checks, start up again
-          if (!pruned_ && (write_state_ == STATE_WRITE_TIMEOUT))
-            set_write_state(STATE_WRITE_INIT);
-
-          if ((port_->IsStandardIce()) &&
-              (port_->GetIceRole() == ICEROLE_CONTROLLED)) {
-            const StunByteStringAttribute* use_candidate_attr =
-                msg->GetByteString(STUN_ATTR_USE_CANDIDATE);
-            if (use_candidate_attr)
-              SignalUseCandidate(this);
-          }
+          HandleBindingRequest(msg.get());
         } else {
           // The packet had the right local username, but the remote username
           // was not the right one for the remote address.
@@ -1046,25 +932,18 @@ void Connection::OnReadPacket(
       // id's match.
       case STUN_BINDING_RESPONSE:
       case STUN_BINDING_ERROR_RESPONSE:
-        if (port_->IsGoogleIce() ||
-            msg->ValidateMessageIntegrity(
+        if (msg->ValidateMessageIntegrity(
                 data, size, remote_candidate().password())) {
           requests_.CheckResponse(msg.get());
         }
         // Otherwise silently discard the response message.
         break;
 
-      // Remote end point sent an STUN indication instead of regular
-      // binding request. In this case |last_ping_received_| will be updated.
-      // Otherwise we can mark connection to read timeout. No response will be
-      // sent in this scenario.
+      // Remote end point sent an STUN indication instead of regular binding
+      // request. In this case |last_ping_received_| will be updated but no
+      // response will be sent.
       case STUN_BINDING_INDICATION:
-        if (port_->IsStandardIce() && read_state_ == STATE_READABLE) {
-          ReceivedPing();
-        } else {
-          LOG_J(LS_WARNING, this) << "Received STUN binding indication "
-                                  << "from an unreadable connection.";
-        }
+        ReceivedPing();
         break;
 
       default:
@@ -1074,6 +953,37 @@ void Connection::OnReadPacket(
   }
 }
 
+void Connection::HandleBindingRequest(IceMessage* msg) {
+  // This connection should now be receiving.
+  ReceivedPing();
+
+  const rtc::SocketAddress& remote_addr = remote_candidate_.address();
+  const std::string& remote_ufrag = remote_candidate_.username();
+  // Check for role conflicts.
+  if (!port_->MaybeIceRoleConflict(remote_addr, msg, remote_ufrag)) {
+    // Received conflicting role from the peer.
+    LOG(LS_INFO) << "Received conflicting role from the peer.";
+    return;
+  }
+
+  // This is a validated stun request from remote peer.
+  port_->SendBindingResponse(msg, remote_addr);
+
+  // If it timed out on writing check, start up again
+  if (!pruned_ && write_state_ == STATE_WRITE_TIMEOUT) {
+    set_write_state(STATE_WRITE_INIT);
+  }
+
+  if (port_->GetIceRole() == ICEROLE_CONTROLLED) {
+    const StunByteStringAttribute* use_candidate_attr =
+        msg->GetByteString(STUN_ATTR_USE_CANDIDATE);
+    if (use_candidate_attr) {
+      set_nominated(true);
+      SignalNominated(this);
+    }
+  }
+}
+
 void Connection::OnReadyToSend() {
   if (write_state_ == STATE_WRITABLE) {
     SignalReadyToSend(this);
@@ -1081,7 +991,7 @@ void Connection::OnReadyToSend() {
 }
 
 void Connection::Prune() {
-  if (!pruned_) {
+  if (!pruned_ || active()) {
     LOG_J(LS_VERBOSE, this) << "Connection pruned";
     pruned_ = true;
     requests_.Clear();
@@ -1091,48 +1001,44 @@ void Connection::Prune() {
 
 void Connection::Destroy() {
   LOG_J(LS_VERBOSE, this) << "Connection destroyed";
-  set_read_state(STATE_READ_TIMEOUT);
-  set_write_state(STATE_WRITE_TIMEOUT);
+  port_->thread()->Post(this, MSG_DELETE);
 }
 
-void Connection::UpdateState(uint32 now) {
-  uint32 rtt = ConservativeRTTEstimate(rtt_);
+void Connection::FailAndDestroy() {
+  set_state(Connection::STATE_FAILED);
+  Destroy();
+}
 
-  std::string pings;
-  for (size_t i = 0; i < pings_since_last_response_.size(); ++i) {
-    char buf[32];
-    rtc::sprintfn(buf, sizeof(buf), "%u",
-        pings_since_last_response_[i]);
-    pings.append(buf).append(" ");
+void Connection::PrintPingsSinceLastResponse(std::string* s, size_t max) {
+  std::ostringstream oss;
+  oss << std::boolalpha;
+  if (pings_since_last_response_.size() > max) {
+    for (size_t i = 0; i < max; i++) {
+      const SentPing& ping = pings_since_last_response_[i];
+      oss << rtc::hex_encode(ping.id) << " ";
+    }
+    oss << "... " << (pings_since_last_response_.size() - max) << " more";
+  } else {
+    for (const SentPing& ping : pings_since_last_response_) {
+      oss << rtc::hex_encode(ping.id) << " ";
+    }
   }
-  LOG_J(LS_VERBOSE, this) << "UpdateState(): pings_since_last_response_="
-                          << pings << ", rtt=" << rtt << ", now=" << now
-                          << ", last ping received: " << last_ping_received_
-                          << ", last data_received: " << last_data_received_;
+  *s = oss.str();
+}
 
-  // Check the readable state.
-  //
-  // Since we don't know how many pings the other side has attempted, the best
-  // test we can do is a simple window.
-  // If other side has not sent ping after connection has become readable, use
-  // |last_data_received_| as the indication.
-  // If remote endpoint is doing RFC 5245, it's not required to send ping
-  // after connection is established. If this connection is serving a data
-  // channel, it may not be in a position to send media continuously. Do not
-  // mark connection timeout if it's in RFC5245 mode.
-  // Below check will be performed with end point if it's doing google-ice.
-  if (port_->IsGoogleIce() && (read_state_ == STATE_READABLE) &&
-      (last_ping_received_ + CONNECTION_READ_TIMEOUT <= now) &&
-      (last_data_received_ + CONNECTION_READ_TIMEOUT <= now)) {
-    LOG_J(LS_INFO, this) << "Unreadable after "
-                         << now - last_ping_received_
-                         << " ms without a ping,"
-                         << " ms since last received response="
-                         << now - last_ping_response_received_
-                         << " ms since last received data="
-                         << now - last_data_received_
-                         << " rtt=" << rtt;
-    set_read_state(STATE_READ_TIMEOUT);
+void Connection::UpdateState(uint32_t now) {
+  uint32_t rtt = ConservativeRTTEstimate(rtt_);
+
+  if (LOG_CHECK_LEVEL(LS_VERBOSE)) {
+    std::string pings;
+    PrintPingsSinceLastResponse(&pings, 5);
+    LOG_J(LS_VERBOSE, this) << "UpdateState()"
+                            << ", ms since last received response="
+                            << now - last_ping_response_received_
+                            << ", ms since last received data="
+                            << now - last_data_received_
+                            << ", rtt=" << rtt
+                            << ", pings_since_last_response=" << pings;
   }
 
   // Check the writable state.  (The order of these checks is important.)
@@ -1152,10 +1058,10 @@ void Connection::UpdateState(uint32 now) {
       TooLongWithoutResponse(pings_since_last_response_,
                              CONNECTION_WRITE_CONNECT_TIMEOUT,
                              now)) {
-    uint32 max_pings = CONNECTION_WRITE_CONNECT_FAILURES;
+    uint32_t max_pings = CONNECTION_WRITE_CONNECT_FAILURES;
     LOG_J(LS_INFO, this) << "Unwritable after " << max_pings
                          << " ping failures and "
-                         << now - pings_since_last_response_[0]
+                         << now - pings_since_last_response_[0].sent_time
                          << " ms without a response,"
                          << " ms since last received ping="
                          << now - last_ping_received_
@@ -1164,32 +1070,78 @@ void Connection::UpdateState(uint32 now) {
                          << " rtt=" << rtt;
     set_write_state(STATE_WRITE_UNRELIABLE);
   }
-
   if ((write_state_ == STATE_WRITE_UNRELIABLE ||
        write_state_ == STATE_WRITE_INIT) &&
       TooLongWithoutResponse(pings_since_last_response_,
                              CONNECTION_WRITE_TIMEOUT,
                              now)) {
     LOG_J(LS_INFO, this) << "Timed out after "
-                         << now - pings_since_last_response_[0]
-                         << " ms without a response, rtt=" << rtt;
+                         << now - pings_since_last_response_[0].sent_time
+                         << " ms without a response"
+                         << ", rtt=" << rtt;
     set_write_state(STATE_WRITE_TIMEOUT);
   }
+
+  // Check the receiving state.
+  uint32_t last_recv_time = last_received();
+  bool receiving = now <= last_recv_time + receiving_timeout_;
+  set_receiving(receiving);
+  if (dead(now)) {
+    Destroy();
+  }
 }
 
-void Connection::Ping(uint32 now) {
-  ASSERT(connected_);
+void Connection::Ping(uint32_t now) {
   last_ping_sent_ = now;
-  pings_since_last_response_.push_back(now);
   ConnectionRequest *req = new ConnectionRequest(this);
-  LOG_J(LS_VERBOSE, this) << "Sending STUN ping " << req->id() << " at " << now;
+  pings_since_last_response_.push_back(SentPing(req->id(), now));
+  LOG_J(LS_VERBOSE, this) << "Sending STUN ping "
+                          << ", id=" << rtc::hex_encode(req->id());
   requests_.Send(req);
   state_ = STATE_INPROGRESS;
 }
 
 void Connection::ReceivedPing() {
+  set_receiving(true);
   last_ping_received_ = rtc::Time();
-  set_read_state(STATE_READABLE);
+}
+
+void Connection::ReceivedPingResponse() {
+  // We've already validated that this is a STUN binding response with
+  // the correct local and remote username for this connection.
+  // So if we're not already, become writable. We may be bringing a pruned
+  // connection back to life, but if we don't really want it, we can always
+  // prune it again.
+  set_receiving(true);
+  set_write_state(STATE_WRITABLE);
+  set_state(STATE_SUCCEEDED);
+  pings_since_last_response_.clear();
+  last_ping_response_received_ = rtc::Time();
+}
+
+bool Connection::dead(uint32_t now) const {
+  if (last_received() > 0) {
+    // If it has ever received anything, we keep it alive until it hasn't
+    // received anything for DEAD_CONNECTION_RECEIVE_TIMEOUT. This covers the
+    // normal case of a successfully used connection that stops working. This
+    // also allows a remote peer to continue pinging over a locally inactive
+    // (pruned) connection.
+    return (now > (last_received() + DEAD_CONNECTION_RECEIVE_TIMEOUT));
+  }
+
+  if (active()) {
+    // If it has never received anything, keep it alive as long as it is
+    // actively pinging and not pruned. Otherwise, the connection might be
+    // deleted before it has a chance to ping. This is the normal case for a
+    // new connection that is pinging but hasn't received anything yet.
+    return false;
+  }
+
+  // If it has never received anything and is not actively pinging (pruned), we
+  // keep it around for at least MIN_CONNECTION_LIFETIME to prevent connections
+  // from being pruned too quickly during a network change event when two
+  // networks would be up simultaneously but only for a brief period.
+  return now > (time_created_ms_ + MIN_CONNECTION_LIFETIME);
 }
 
 std::string Connection::ToDebugId() const {
@@ -1203,10 +1155,9 @@ std::string Connection::ToString() const {
     '-',  // not connected (false)
     'C',  // connected (true)
   };
-  const char READ_STATE_ABBREV[3] = {
-    '-',  // STATE_READ_INIT
-    'R',  // STATE_READABLE
-    'x',  // STATE_READ_TIMEOUT
+  const char RECEIVE_STATE_ABBREV[2] = {
+    '-',  // not receiving (false)
+    'R',  // receiving (true)
   };
   const char WRITE_STATE_ABBREV[4] = {
     'W',  // STATE_WRITABLE
@@ -1234,7 +1185,7 @@ std::string Connection::ToString() const {
      << ":" << remote.type() << ":"
      << remote.protocol() << ":" << remote.address().ToSensitiveString() << "|"
      << CONNECT_STATE_ABBREV[connected()]
-     << READ_STATE_ABBREV[read_state()]
+     << RECEIVE_STATE_ABBREV[receiving()]
      << WRITE_STATE_ABBREV[write_state()]
      << ICESTATE[state()] << "|"
      << priority() << "|";
@@ -1252,45 +1203,30 @@ std::string Connection::ToSensitiveString() const {
 
 void Connection::OnConnectionRequestResponse(ConnectionRequest* request,
                                              StunMessage* response) {
-  // We've already validated that this is a STUN binding response with
-  // the correct local and remote username for this connection.
-  // So if we're not already, become writable. We may be bringing a pruned
-  // connection back to life, but if we don't really want it, we can always
-  // prune it again.
-  uint32 rtt = request->Elapsed();
-  set_write_state(STATE_WRITABLE);
-  set_state(STATE_SUCCEEDED);
+  // Log at LS_INFO if we receive a ping response on an unwritable
+  // connection.
+  rtc::LoggingSeverity sev = !writable() ? rtc::LS_INFO : rtc::LS_VERBOSE;
 
-  if (remote_ice_mode_ == ICEMODE_LITE) {
-    // A ice-lite end point never initiates ping requests. This will allow
-    // us to move to STATE_READABLE.
-    ReceivedPing();
+  uint32_t rtt = request->Elapsed();
+
+  ReceivedPingResponse();
+
+  if (LOG_CHECK_LEVEL_V(sev)) {
+    bool use_candidate = (
+        response->GetByteString(STUN_ATTR_USE_CANDIDATE) != nullptr);
+    std::string pings;
+    PrintPingsSinceLastResponse(&pings, 5);
+    LOG_JV(sev, this) << "Received STUN ping response"
+                      << ", id=" << rtc::hex_encode(request->id())
+                      << ", code=0"  // Makes logging easier to parse.
+                      << ", rtt=" << rtt
+                      << ", use_candidate=" << use_candidate
+                      << ", pings_since_last_response=" << pings;
   }
 
-  std::string pings;
-  for (size_t i = 0; i < pings_since_last_response_.size(); ++i) {
-    char buf[32];
-    rtc::sprintfn(buf, sizeof(buf), "%u",
-        pings_since_last_response_[i]);
-    pings.append(buf).append(" ");
-  }
-
-  rtc::LoggingSeverity level =
-      (pings_since_last_response_.size() > CONNECTION_WRITE_CONNECT_FAILURES) ?
-          rtc::LS_INFO : rtc::LS_VERBOSE;
-
-  LOG_JV(level, this) << "Received STUN ping response " << request->id()
-                      << ", pings_since_last_response_=" << pings
-                      << ", rtt=" << rtt;
-
-  pings_since_last_response_.clear();
-  last_ping_response_received_ = rtc::Time();
   rtt_ = (RTT_RATIO * rtt_ + rtt) / (RTT_RATIO + 1);
 
-  // Peer reflexive candidate is only for RFC 5245 ICE.
-  if (port_->IsStandardIce()) {
-    MaybeAddPrflxCandidate(request, response);
-  }
+  MaybeAddPrflxCandidate(request, response);
 }
 
 void Connection::OnConnectionRequestErrorResponse(ConnectionRequest* request,
@@ -1298,15 +1234,14 @@ void Connection::OnConnectionRequestErrorResponse(ConnectionRequest* request,
   const StunErrorCodeAttribute* error_attr = response->GetErrorCode();
   int error_code = STUN_ERROR_GLOBAL_FAILURE;
   if (error_attr) {
-    if (port_->IsGoogleIce()) {
-      // When doing GICE, the error code is written out incorrectly, so we need
-      // to unmunge it here.
-      error_code = error_attr->eclass() * 256 + error_attr->number();
-    } else {
-      error_code = error_attr->code();
-    }
+    error_code = error_attr->code();
   }
 
+  LOG_J(LS_INFO, this) << "Received STUN error response"
+                       << " id=" << rtc::hex_encode(request->id())
+                       << " code=" << error_code
+                       << " rtt=" << request->Elapsed();
+
   if (error_code == STUN_ERROR_UNKNOWN_ATTRIBUTE ||
       error_code == STUN_ERROR_SERVER_ERROR ||
       error_code == STUN_ERROR_UNAUTHORIZED) {
@@ -1319,30 +1254,25 @@ void Connection::OnConnectionRequestErrorResponse(ConnectionRequest* request,
     // This is not a valid connection.
     LOG_J(LS_ERROR, this) << "Received STUN error response, code="
                           << error_code << "; killing connection";
-    set_state(STATE_FAILED);
-    set_write_state(STATE_WRITE_TIMEOUT);
+    FailAndDestroy();
   }
 }
 
 void Connection::OnConnectionRequestTimeout(ConnectionRequest* request) {
   // Log at LS_INFO if we miss a ping on a writable connection.
-  rtc::LoggingSeverity sev = (write_state_ == STATE_WRITABLE) ?
-      rtc::LS_INFO : rtc::LS_VERBOSE;
-  LOG_JV(sev, this) << "Timing-out STUN ping " << request->id()
+  rtc::LoggingSeverity sev = writable() ? rtc::LS_INFO : rtc::LS_VERBOSE;
+  LOG_JV(sev, this) << "Timing-out STUN ping "
+                    << rtc::hex_encode(request->id())
                     << " after " << request->Elapsed() << " ms";
 }
 
-void Connection::CheckTimeout() {
-  // If both read and write have timed out or read has never initialized, then
-  // this connection can contribute no more to p2p socket unless at some later
-  // date readability were to come back.  However, we gave readability a long
-  // time to timeout, so at this point, it seems fair to get rid of this
-  // connection.
-  if ((read_state_ == STATE_READ_TIMEOUT ||
-       read_state_ == STATE_READ_INIT) &&
-      write_state_ == STATE_WRITE_TIMEOUT) {
-    port_->thread()->Post(this, MSG_DELETE);
-  }
+void Connection::OnConnectionRequestSent(ConnectionRequest* request) {
+  // Log at LS_INFO if we send a ping on an unwritable connection.
+  rtc::LoggingSeverity sev = !writable() ? rtc::LS_INFO : rtc::LS_VERBOSE;
+  bool use_candidate = use_candidate_attr();
+  LOG_JV(sev, this) << "Sent STUN ping"
+                    << ", id=" << rtc::hex_encode(request->id())
+                    << ", use_candidate=" << use_candidate;
 }
 
 void Connection::HandleRoleConflictFromPeer() {
@@ -1372,26 +1302,30 @@ void Connection::MaybeUpdatePeerReflexiveCandidate(
 
 void Connection::OnMessage(rtc::Message *pmsg) {
   ASSERT(pmsg->message_id == MSG_DELETE);
-
-  LOG_J(LS_INFO, this) << "Connection deleted due to read or write timeout";
+  LOG_J(LS_INFO, this) << "Connection deleted";
   SignalDestroyed(this);
   delete this;
 }
 
+uint32_t Connection::last_received() const {
+  return std::max(last_data_received_,
+             std::max(last_ping_received_, last_ping_response_received_));
+}
+
 size_t Connection::recv_bytes_second() {
-  return recv_rate_tracker_.units_second();
+  return round(recv_rate_tracker_.ComputeRate());
 }
 
 size_t Connection::recv_total_bytes() {
-  return recv_rate_tracker_.total_units();
+  return recv_rate_tracker_.TotalSampleCount();
 }
 
 size_t Connection::sent_bytes_second() {
-  return send_rate_tracker_.units_second();
+  return round(send_rate_tracker_.ComputeRate());
 }
 
 size_t Connection::sent_total_bytes() {
-  return send_rate_tracker_.total_units();
+  return send_rate_tracker_.TotalSampleCount();
 }
 
 size_t Connection::sent_discarded_packets() {
@@ -1440,7 +1374,7 @@ void Connection::MaybeAddPrflxCandidate(ConnectionRequest* request,
                     << "stun response message";
     return;
   }
-  const uint32 priority = priority_attr->value();
+  const uint32_t priority = priority_attr->value();
   std::string id = rtc::CreateRandomString(8);
 
   Candidate new_local_candidate;
@@ -1467,10 +1401,10 @@ void Connection::MaybeAddPrflxCandidate(ConnectionRequest* request,
   SignalStateChange(this);
 }
 
-ProxyConnection::ProxyConnection(Port* port, size_t index,
-                                 const Candidate& candidate)
-  : Connection(port, index, candidate), error_(0) {
-}
+ProxyConnection::ProxyConnection(Port* port,
+                                 size_t index,
+                                 const Candidate& remote_candidate)
+    : Connection(port, index, remote_candidate) {}
 
 int ProxyConnection::Send(const void* data, size_t size,
                           const rtc::PacketOptions& options) {
@@ -1486,7 +1420,7 @@ int ProxyConnection::Send(const void* data, size_t size,
     error_ = port_->GetError();
     sent_packets_discarded_++;
   } else {
-    send_rate_tracker_.Update(sent);
+    send_rate_tracker_.AddSamples(sent);
   }
   return sent;
 }
diff --git a/media/webrtc/trunk/webrtc/p2p/base/port.h b/media/webrtc/trunk/webrtc/p2p/base/port.h
index 6d7d6d5437..436b1e7faa 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/port.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/port.h
@@ -50,17 +50,25 @@ extern const char TCPTYPE_ACTIVE_STR[];
 extern const char TCPTYPE_PASSIVE_STR[];
 extern const char TCPTYPE_SIMOPEN_STR[];
 
-// The length of time we wait before timing out readability on a connection.
-const uint32 CONNECTION_READ_TIMEOUT = 30 * 1000;   // 30 seconds
+// The minimum time we will wait before destroying a connection after creating
+// it.
+const uint32_t MIN_CONNECTION_LIFETIME = 10 * 1000;  // 10 seconds.
+
+// A connection will be declared dead if it has not received anything for this
+// long.
+const uint32_t DEAD_CONNECTION_RECEIVE_TIMEOUT = 30 * 1000;  // 30 seconds.
+
+// The timeout duration when a connection does not receive anything.
+const uint32_t WEAK_CONNECTION_RECEIVE_TIMEOUT = 2500;  // 2.5 seconds
 
 // The length of time we wait before timing out writability on a connection.
-const uint32 CONNECTION_WRITE_TIMEOUT = 15 * 1000;  // 15 seconds
+const uint32_t CONNECTION_WRITE_TIMEOUT = 15 * 1000;  // 15 seconds
 
 // The length of time we wait before we become unwritable.
-const uint32 CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000;  // 5 seconds
+const uint32_t CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000;  // 5 seconds
 
 // The number of pings that must fail to respond before we become unwritable.
-const uint32 CONNECTION_WRITE_CONNECT_FAILURES = 5;
+const uint32_t CONNECTION_WRITE_CONNECT_FAILURES = 5;
 
 // This is the length of time that we wait for a ping response to come back.
 const int CONNECTION_RESPONSE_TIMEOUT = 5 * 1000;   // 5 seconds
@@ -118,8 +126,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
        rtc::PacketSocketFactory* factory,
        rtc::Network* network,
        const rtc::IPAddress& ip,
-       uint16 min_port,
-       uint16 max_port,
+       uint16_t min_port,
+       uint16_t max_port,
        const std::string& username_fragment,
        const std::string& password);
   virtual ~Port();
@@ -127,22 +135,12 @@ class Port : public PortInterface, public rtc::MessageHandler,
   virtual const std::string& Type() const { return type_; }
   virtual rtc::Network* Network() const { return network_; }
 
-  // This method will set the flag which enables standard ICE/STUN procedures
-  // in STUN connectivity checks. Currently this method does
-  // 1. Add / Verify MI attribute in STUN binding requests.
-  // 2. Username attribute in STUN binding request will be RFRAF:LFRAG,
-  // as opposed to RFRAGLFRAG.
-  virtual void SetIceProtocolType(IceProtocolType protocol) {
-    ice_protocol_ = protocol;
-  }
-  virtual IceProtocolType IceProtocol() const { return ice_protocol_; }
-
   // Methods to set/get ICE role and tiebreaker values.
   IceRole GetIceRole() const { return ice_role_; }
   void SetIceRole(IceRole role) { ice_role_ = role; }
 
-  void SetIceTiebreaker(uint64 tiebreaker) { tiebreaker_ = tiebreaker; }
-  uint64 IceTiebreaker() const { return tiebreaker_; }
+  void SetIceTiebreaker(uint64_t tiebreaker) { tiebreaker_ = tiebreaker; }
+  uint64_t IceTiebreaker() const { return tiebreaker_; }
 
   virtual bool SharedSocket() const { return shared_socket_; }
   void ResetSharedSocket() { shared_socket_ = false; }
@@ -173,8 +171,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
   }
 
   // Identifies the generation that this port was created in.
-  uint32 generation() { return generation_; }
-  void set_generation(uint32 generation) { generation_ = generation; }
+  uint32_t generation() { return generation_; }
+  void set_generation(uint32_t generation) { generation_ = generation; }
 
   // ICE requires a single username/password per content/media line. So the
   // |ice_username_fragment_| of the ports that belongs to the same content will
@@ -263,8 +261,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
   // Debugging description of this port
   virtual std::string ToString() const;
   const rtc::IPAddress& ip() const { return ip_; }
-  uint16 min_port() { return min_port_; }
-  uint16 max_port() { return max_port_; }
+  uint16_t min_port() { return min_port_; }
+  uint16_t max_port() { return max_port_; }
 
   // Timeout shortening function to speed up unit tests.
   void set_timeout_delay(int delay) { timeout_delay_ = delay; }
@@ -273,8 +271,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
   // stun username attribute if present.
   bool ParseStunUsername(const StunMessage* stun_msg,
                          std::string* local_username,
-                         std::string* remote_username,
-                         IceProtocolType* remote_protocol_type) const;
+                         std::string* remote_username) const;
   void CreateStunUsername(const std::string& remote_username,
                           std::string* stun_username_attr_str) const;
 
@@ -282,6 +279,13 @@ class Port : public PortInterface, public rtc::MessageHandler,
                             IceMessage* stun_msg,
                             const std::string& remote_ufrag);
 
+  // Called when a packet has been sent to the socket.
+  // This is made pure virtual to notify subclasses of Port that they MUST
+  // listen to AsyncPacketSocket::SignalSentPacket and then call
+  // PortInterface::OnSentPacket.
+  virtual void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                            const rtc::SentPacket& sent_packet) = 0;
+
   // Called when the socket is currently able to send.
   void OnReadyToSend();
 
@@ -289,22 +293,13 @@ class Port : public PortInterface, public rtc::MessageHandler,
   // Returns the index of the new local candidate.
   size_t AddPrflxCandidate(const Candidate& local);
 
-  // Returns if RFC 5245 ICE protocol is used.
-  bool IsStandardIce() const;
-
-  // Returns if Google ICE protocol is used.
-  bool IsGoogleIce() const;
-
-  // Returns if Hybrid ICE protocol is used.
-  bool IsHybridIce() const;
-
-  void set_candidate_filter(uint32 candidate_filter) {
+  void set_candidate_filter(uint32_t candidate_filter) {
     candidate_filter_ = candidate_filter;
   }
 
  protected:
   enum {
-    MSG_CHECKTIMEOUT = 0,
+    MSG_DEAD = 0,
     MSG_FIRST_AVAILABLE
   };
 
@@ -313,9 +308,13 @@ class Port : public PortInterface, public rtc::MessageHandler,
   void AddAddress(const rtc::SocketAddress& address,
                   const rtc::SocketAddress& base_address,
                   const rtc::SocketAddress& related_address,
-                  const std::string& protocol, const std::string& tcptype,
-                  const std::string& type, uint32 type_preference,
-                  uint32 relay_preference, bool final);
+                  const std::string& protocol,
+                  const std::string& relay_protocol,
+                  const std::string& tcptype,
+                  const std::string& type,
+                  uint32_t type_preference,
+                  uint32_t relay_preference,
+                  bool final);
 
   // Adds the given connection to the list.  (Deleting removes them.)
   void AddConnection(Connection* conn);
@@ -345,15 +344,18 @@ class Port : public PortInterface, public rtc::MessageHandler,
     return rtc::DSCP_NO_CHANGE;
   }
 
-  uint32 candidate_filter() { return candidate_filter_; }
+  uint32_t candidate_filter() { return candidate_filter_; }
 
  private:
   void Construct();
   // Called when one of our connections deletes itself.
   void OnConnectionDestroyed(Connection* conn);
 
-  // Checks if this port is useless, and hence, should be destroyed.
-  void CheckTimeout();
+  // Whether this port is dead, and hence, should be destroyed on the controlled
+  // side.
+  bool dead() const {
+    return ice_role_ == ICEROLE_CONTROLLED && connections_.empty();
+  }
 
   rtc::Thread* thread_;
   rtc::PacketSocketFactory* factory_;
@@ -361,11 +363,11 @@ class Port : public PortInterface, public rtc::MessageHandler,
   bool send_retransmit_count_attribute_;
   rtc::Network* network_;
   rtc::IPAddress ip_;
-  uint16 min_port_;
-  uint16 max_port_;
+  uint16_t min_port_;
+  uint16_t max_port_;
   std::string content_name_;
   int component_;
-  uint32 generation_;
+  uint32_t generation_;
   // In order to establish a connection to this Port (so that real data can be
   // sent through), the other side must send us a STUN binding request that is
   // authenticated with this username_fragment and password.
@@ -380,9 +382,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
   AddressMap connections_;
   int timeout_delay_;
   bool enable_port_packets_;
-  IceProtocolType ice_protocol_;
   IceRole ice_role_;
-  uint64 tiebreaker_;
+  uint64_t tiebreaker_;
   bool shared_socket_;
   // Information to use when going through a proxy.
   std::string user_agent_;
@@ -392,7 +393,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
   // make its own decision on how to create candidates. For example,
   // when IceTransportsType is set to relay, both RelayPort and
   // TurnPort will hide raddr to avoid local address leakage.
-  uint32 candidate_filter_;
+  uint32_t candidate_filter_;
 
   friend class Connection;
 };
@@ -402,6 +403,14 @@ class Port : public PortInterface, public rtc::MessageHandler,
 class Connection : public rtc::MessageHandler,
     public sigslot::has_slots<> {
  public:
+  struct SentPing {
+    SentPing(const std::string id, uint32_t sent_time)
+        : id(id), sent_time(sent_time) {}
+
+    std::string id;
+    uint32_t sent_time;
+  };
+
   // States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
   enum State {
     STATE_WAITING = 0,  // Check has not been performed, Waiting pair on CL.
@@ -423,16 +432,7 @@ class Connection : public rtc::MessageHandler,
   const Candidate& remote_candidate() const { return remote_candidate_; }
 
   // Returns the pair priority.
-  uint64 priority() const;
-
-  enum ReadState {
-    STATE_READ_INIT    = 0,  // we have yet to receive a ping
-    STATE_READABLE     = 1,  // we have received pings recently
-    STATE_READ_TIMEOUT = 2,  // we haven't received pings in a while
-  };
-
-  ReadState read_state() const { return read_state_; }
-  bool readable() const { return read_state_ == STATE_READABLE; }
+  uint64_t priority() const;
 
   enum WriteState {
     STATE_WRITABLE          = 0,  // we have received ping responses recently
@@ -443,13 +443,20 @@ class Connection : public rtc::MessageHandler,
 
   WriteState write_state() const { return write_state_; }
   bool writable() const { return write_state_ == STATE_WRITABLE; }
+  bool receiving() const { return receiving_; }
 
   // Determines whether the connection has finished connecting.  This can only
   // be false for TCP connections.
   bool connected() const { return connected_; }
+  bool weak() const { return !(writable() && receiving() && connected()); }
+  bool active() const {
+    return write_state_ != STATE_WRITE_TIMEOUT;
+  }
+  // A connection is dead if it can be safely deleted.
+  bool dead(uint32_t now) const;
 
   // Estimate of the round-trip time over this connection.
-  uint32 rtt() const { return rtt_; }
+  uint32_t rtt() const { return rtt_; }
 
   size_t sent_total_bytes();
   size_t sent_bytes_second();
@@ -474,8 +481,8 @@ class Connection : public rtc::MessageHandler,
   // Error if Send() returns < 0
   virtual int GetError() = 0;
 
-  sigslot::signal4<Connection*, const char*, size_t,
-                   const rtc::PacketTime&> SignalReadPacket;
+  sigslot::signal4<Connection*, const char*, size_t, const rtc::PacketTime&>
+      SignalReadPacket;
 
   sigslot::signal1<Connection*> SignalReadyToSend;
 
@@ -496,38 +503,56 @@ class Connection : public rtc::MessageHandler,
   bool use_candidate_attr() const { return use_candidate_attr_; }
   void set_use_candidate_attr(bool enable);
 
+  bool nominated() const { return nominated_; }
+  void set_nominated(bool nominated) { nominated_ = nominated; }
+
   void set_remote_ice_mode(IceMode mode) {
     remote_ice_mode_ = mode;
   }
 
+  void set_receiving_timeout(uint32_t receiving_timeout_ms) {
+    receiving_timeout_ = receiving_timeout_ms;
+  }
+
   // Makes the connection go away.
   void Destroy();
 
+  // Makes the connection go away, in a failed state.
+  void FailAndDestroy();
+
   // Checks that the state of this connection is up-to-date.  The argument is
   // the current time, which is compared against various timeouts.
-  void UpdateState(uint32 now);
+  void UpdateState(uint32_t now);
 
   // Called when this connection should try checking writability again.
-  uint32 last_ping_sent() const { return last_ping_sent_; }
-  void Ping(uint32 now);
+  uint32_t last_ping_sent() const { return last_ping_sent_; }
+  void Ping(uint32_t now);
+  void ReceivedPingResponse();
+  uint32_t last_ping_response_received() const {
+    return last_ping_response_received_;
+  }
 
   // Called whenever a valid ping is received on this connection.  This is
   // public because the connection intercepts the first ping for us.
-  uint32 last_ping_received() const { return last_ping_received_; }
+  uint32_t last_ping_received() const { return last_ping_received_; }
   void ReceivedPing();
+  // Handles the binding request; sends a response if this is a valid request.
+  void HandleBindingRequest(IceMessage* msg);
 
   // Debugging description of this connection
   std::string ToDebugId() const;
   std::string ToString() const;
   std::string ToSensitiveString() const;
+  // Prints pings_since_last_response_ into a string.
+  void PrintPingsSinceLastResponse(std::string* pings, size_t max);
 
   bool reported() const { return reported_; }
   void set_reported(bool reported) { reported_ = reported;}
 
-  // This flag will be set if this connection is the chosen one for media
-  // transmission. This connection will send STUN ping with USE-CANDIDATE
-  // attribute.
-  sigslot::signal1<Connection*> SignalUseCandidate;
+  // This signal will be fired if this connection is nominated by the
+  // controlling side.
+  sigslot::signal1<Connection*> SignalNominated;
+
   // Invoked when Connection receives STUN error response with 487 code.
   void HandleRoleConflictFromPeer();
 
@@ -545,7 +570,13 @@ class Connection : public rtc::MessageHandler,
   // |new_candidate|.
   void MaybeUpdatePeerReflexiveCandidate(const Candidate& new_candidate);
 
+  // Returns the last received time of any data, stun request, or stun
+  // response in milliseconds
+  uint32_t last_received() const;
+
  protected:
+  enum { MSG_DELETE = 0, MSG_FIRST_AVAILABLE };
+
   // Constructs a new connection to the given remote port.
   Connection(Port* port, size_t index, const Candidate& candidate);
 
@@ -553,49 +584,50 @@ class Connection : public rtc::MessageHandler,
   void OnSendStunPacket(const void* data, size_t size, StunRequest* req);
 
   // Callbacks from ConnectionRequest
-  void OnConnectionRequestResponse(ConnectionRequest* req,
-                                   StunMessage* response);
+  virtual void OnConnectionRequestResponse(ConnectionRequest* req,
+                                           StunMessage* response);
   void OnConnectionRequestErrorResponse(ConnectionRequest* req,
                                         StunMessage* response);
   void OnConnectionRequestTimeout(ConnectionRequest* req);
+  void OnConnectionRequestSent(ConnectionRequest* req);
 
   // Changes the state and signals if necessary.
-  void set_read_state(ReadState value);
   void set_write_state(WriteState value);
+  void set_receiving(bool value);
   void set_state(State state);
   void set_connected(bool value);
 
-  // Checks if this connection is useless, and hence, should be destroyed.
-  void CheckTimeout();
-
   void OnMessage(rtc::Message *pmsg);
 
   Port* port_;
   size_t local_candidate_index_;
   Candidate remote_candidate_;
-  ReadState read_state_;
   WriteState write_state_;
+  bool receiving_;
   bool connected_;
   bool pruned_;
   // By default |use_candidate_attr_| flag will be true,
-  // as we will be using agrressive nomination.
+  // as we will be using aggressive nomination.
   // But when peer is ice-lite, this flag "must" be initialized to false and
   // turn on when connection becomes "best connection".
   bool use_candidate_attr_;
+  // Whether this connection has been nominated by the controlling side via
+  // the use_candidate attribute.
+  bool nominated_;
   IceMode remote_ice_mode_;
   StunRequestManager requests_;
-  uint32 rtt_;
-  uint32 last_ping_sent_;      // last time we sent a ping to the other side
-  uint32 last_ping_received_;  // last time we received a ping from the other
-                               // side
-  uint32 last_data_received_;
-  uint32 last_ping_response_received_;
-  std::vector<uint32> pings_since_last_response_;
+  uint32_t rtt_;
+  uint32_t last_ping_sent_;      // last time we sent a ping to the other side
+  uint32_t last_ping_received_;  // last time we received a ping from the other
+                                 // side
+  uint32_t last_data_received_;
+  uint32_t last_ping_response_received_;
+  std::vector<SentPing> pings_since_last_response_;
 
   rtc::RateTracker recv_rate_tracker_;
   rtc::RateTracker send_rate_tracker_;
-  uint32 sent_packets_discarded_;
-  uint32 sent_packets_total_;
+  uint32_t sent_packets_discarded_;
+  uint32_t sent_packets_total_;
 
  private:
   void MaybeAddPrflxCandidate(ConnectionRequest* request,
@@ -603,22 +635,26 @@ class Connection : public rtc::MessageHandler,
 
   bool reported_;
   State state_;
+  // Time duration to switch from receiving to not receiving.
+  uint32_t receiving_timeout_;
+  uint32_t time_created_ms_;
 
   friend class Port;
   friend class ConnectionRequest;
 };
 
-// ProxyConnection defers all the interesting work to the port
+// ProxyConnection defers all the interesting work to the port.
 class ProxyConnection : public Connection {
  public:
-  ProxyConnection(Port* port, size_t index, const Candidate& candidate);
+  ProxyConnection(Port* port, size_t index, const Candidate& remote_candidate);
 
-  virtual int Send(const void* data, size_t size,
-                   const rtc::PacketOptions& options);
-  virtual int GetError() { return error_; }
+  int Send(const void* data,
+           size_t size,
+           const rtc::PacketOptions& options) override;
+  int GetError() override { return error_; }
 
  private:
-  int error_;
+  int error_ = 0;
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/port_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/port_unittest.cc
index 26e46a6995..449021ad9f 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/port_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/port_unittest.cc
@@ -9,7 +9,6 @@
  */
 
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
-#include "webrtc/p2p/base/portproxy.h"
 #include "webrtc/p2p/base/relayport.h"
 #include "webrtc/p2p/base/stunport.h"
 #include "webrtc/p2p/base/tcpport.h"
@@ -18,6 +17,7 @@
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/transport.h"
 #include "webrtc/p2p/base/turnport.h"
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/crc32.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
@@ -48,8 +48,8 @@ using namespace cricket;
 static const int kTimeout = 1000;
 static const SocketAddress kLocalAddr1("192.168.1.2", 0);
 static const SocketAddress kLocalAddr2("192.168.1.3", 0);
-static const SocketAddress kNatAddr1("77.77.77.77", rtc::NAT_SERVER_PORT);
-static const SocketAddress kNatAddr2("88.88.88.88", rtc::NAT_SERVER_PORT);
+static const SocketAddress kNatAddr1("77.77.77.77", rtc::NAT_SERVER_UDP_PORT);
+static const SocketAddress kNatAddr2("88.88.88.88", rtc::NAT_SERVER_UDP_PORT);
 static const SocketAddress kStunAddr("99.99.99.1", STUN_SERVER_PORT);
 static const SocketAddress kRelayUdpIntAddr("99.99.99.2", 5000);
 static const SocketAddress kRelayUdpExtAddr("99.99.99.3", 5001);
@@ -63,20 +63,17 @@ static const RelayCredentials kRelayCredentials("test", "test");
 
 // TODO: Update these when RFC5245 is completely supported.
 // Magic value of 30 is from RFC3484, for IPv4 addresses.
-static const uint32 kDefaultPrflxPriority = ICE_TYPE_PREFERENCE_PRFLX << 24 |
-             30 << 8 | (256 - ICE_CANDIDATE_COMPONENT_DEFAULT);
-static const int STUN_ERROR_BAD_REQUEST_AS_GICE =
-    STUN_ERROR_BAD_REQUEST / 256 * 100 + STUN_ERROR_BAD_REQUEST % 256;
-static const int STUN_ERROR_UNAUTHORIZED_AS_GICE =
-    STUN_ERROR_UNAUTHORIZED / 256 * 100 + STUN_ERROR_UNAUTHORIZED % 256;
-static const int STUN_ERROR_SERVER_ERROR_AS_GICE =
-    STUN_ERROR_SERVER_ERROR / 256 * 100 + STUN_ERROR_SERVER_ERROR % 256;
+static const uint32_t kDefaultPrflxPriority =
+    ICE_TYPE_PREFERENCE_PRFLX << 24 | 30 << 8 |
+    (256 - ICE_CANDIDATE_COMPONENT_DEFAULT);
 
 static const int kTiebreaker1 = 11111;
 static const int kTiebreaker2 = 22222;
 
+static const char* data = "ABCDEFGHIJKLMNOPQRSTUVWXYZ1234567890";
+
 static Candidate GetCandidate(Port* port) {
-  assert(port->Candidates().size() == 1);
+  assert(port->Candidates().size() >= 1);
   return port->Candidates()[0];
 }
 
@@ -105,13 +102,19 @@ class TestPort : public Port {
            rtc::PacketSocketFactory* factory,
            rtc::Network* network,
            const rtc::IPAddress& ip,
-           uint16 min_port,
-           uint16 max_port,
+           uint16_t min_port,
+           uint16_t max_port,
            const std::string& username_fragment,
            const std::string& password)
-      : Port(thread, type, factory, network, ip, min_port, max_port,
-             username_fragment, password) {
-  }
+      : Port(thread,
+             type,
+             factory,
+             network,
+             ip,
+             min_port,
+             max_port,
+             username_fragment,
+             password) {}
   ~TestPort() {}
 
   // Expose GetStunMessage so that we can test it.
@@ -134,13 +137,17 @@ class TestPort : public Port {
 
   virtual void PrepareAddress() {
     rtc::SocketAddress addr(ip(), min_port());
-    AddAddress(addr, addr, rtc::SocketAddress(), "udp", "", Type(),
+    AddAddress(addr, addr, rtc::SocketAddress(), "udp", "", "", Type(),
                ICE_TYPE_PREFERENCE_HOST, 0, true);
   }
 
+  virtual bool SupportsProtocol(const std::string& protocol) const {
+    return true;
+  }
+
   // Exposed for testing candidate building.
   void AddCandidateAddress(const rtc::SocketAddress& addr) {
-    AddAddress(addr, addr, rtc::SocketAddress(), "udp", "", Type(),
+    AddAddress(addr, addr, rtc::SocketAddress(), "udp", "", "", Type(),
                type_preference_, 0, false);
   }
   void AddCandidateAddress(const rtc::SocketAddress& addr,
@@ -148,7 +155,7 @@ class TestPort : public Port {
                            const std::string& type,
                            int type_preference,
                            bool final) {
-    AddAddress(addr, base_address, rtc::SocketAddress(), "udp", "", type,
+    AddAddress(addr, base_address, rtc::SocketAddress(), "udp", "", "", type,
                type_preference, 0, final);
   }
 
@@ -197,21 +204,28 @@ class TestPort : public Port {
   }
 
  private:
+  void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                    const rtc::SentPacket& sent_packet) {
+    PortInterface::SignalSentPacket(sent_packet);
+  }
   rtc::scoped_ptr<ByteBuffer> last_stun_buf_;
   rtc::scoped_ptr<IceMessage> last_stun_msg_;
-  int type_preference_;
+  int type_preference_ = 0;
 };
 
 class TestChannel : public sigslot::has_slots<> {
  public:
   // Takes ownership of |p1| (but not |p2|).
-  TestChannel(Port* p1, Port* p2)
-      : ice_mode_(ICEMODE_FULL), src_(p1), dst_(p2), complete_count_(0),
-        conn_(NULL), remote_request_(), nominated_(false) {
-    src_->SignalPortComplete.connect(
-        this, &TestChannel::OnPortComplete);
-    src_->SignalUnknownAddress.connect(this, &TestChannel::OnUnknownAddress);
-    src_->SignalDestroyed.connect(this, &TestChannel::OnSrcPortDestroyed);
+  TestChannel(Port* p1)
+      : ice_mode_(ICEMODE_FULL),
+        port_(p1),
+        complete_count_(0),
+        conn_(NULL),
+        remote_request_(),
+        nominated_(false) {
+    port_->SignalPortComplete.connect(this, &TestChannel::OnPortComplete);
+    port_->SignalUnknownAddress.connect(this, &TestChannel::OnUnknownAddress);
+    port_->SignalDestroyed.connect(this, &TestChannel::OnSrcPortDestroyed);
   }
 
   int complete_count() { return complete_count_; }
@@ -219,17 +233,19 @@ class TestChannel : public sigslot::has_slots<> {
   const SocketAddress& remote_address() { return remote_address_; }
   const std::string remote_fragment() { return remote_frag_; }
 
-  void Start() {
-    src_->PrepareAddress();
-  }
-  void CreateConnection() {
-    conn_ = src_->CreateConnection(GetCandidate(dst_), Port::ORIGIN_MESSAGE);
+  void Start() { port_->PrepareAddress(); }
+  void CreateConnection(const Candidate& remote_candidate) {
+    conn_ = port_->CreateConnection(remote_candidate, Port::ORIGIN_MESSAGE);
     IceMode remote_ice_mode =
         (ice_mode_ == ICEMODE_FULL) ? ICEMODE_LITE : ICEMODE_FULL;
     conn_->set_remote_ice_mode(remote_ice_mode);
     conn_->set_use_candidate_attr(remote_ice_mode == ICEMODE_FULL);
     conn_->SignalStateChange.connect(
         this, &TestChannel::OnConnectionStateChange);
+    conn_->SignalDestroyed.connect(this, &TestChannel::OnDestroyed);
+    conn_->SignalReadyToSend.connect(this,
+                                     &TestChannel::OnConnectionReadyToSend);
+    connection_ready_to_send_ = false;
   }
   void OnConnectionStateChange(Connection* conn) {
     if (conn->write_state() == Connection::STATE_WRITABLE) {
@@ -237,23 +253,23 @@ class TestChannel : public sigslot::has_slots<> {
       nominated_ = true;
     }
   }
-  void AcceptConnection() {
+  void AcceptConnection(const Candidate& remote_candidate) {
     ASSERT_TRUE(remote_request_.get() != NULL);
-    Candidate c = GetCandidate(dst_);
+    Candidate c = remote_candidate;
     c.set_address(remote_address_);
-    conn_ = src_->CreateConnection(c, Port::ORIGIN_MESSAGE);
-    src_->SendBindingResponse(remote_request_.get(), remote_address_);
+    conn_ = port_->CreateConnection(c, Port::ORIGIN_MESSAGE);
+    conn_->SignalDestroyed.connect(this, &TestChannel::OnDestroyed);
+    port_->SendBindingResponse(remote_request_.get(), remote_address_);
     remote_request_.reset();
   }
   void Ping() {
     Ping(0);
   }
-  void Ping(uint32 now) {
-    conn_->Ping(now);
-  }
+  void Ping(uint32_t now) { conn_->Ping(now); }
   void Stop() {
-    conn_->SignalDestroyed.connect(this, &TestChannel::OnDestroyed);
-    conn_->Destroy();
+    if (conn_) {
+      conn_->Destroy();
+    }
   }
 
   void OnPortComplete(Port* port) {
@@ -263,31 +279,28 @@ class TestChannel : public sigslot::has_slots<> {
     ice_mode_ = ice_mode;
   }
 
+  int SendData(const char* data, size_t len) {
+    rtc::PacketOptions options;
+    return conn_->Send(data, len, options);
+  }
+
   void OnUnknownAddress(PortInterface* port, const SocketAddress& addr,
                         ProtocolType proto,
                         IceMessage* msg, const std::string& rf,
                         bool /*port_muxed*/) {
-    ASSERT_EQ(src_.get(), port);
+    ASSERT_EQ(port_.get(), port);
     if (!remote_address_.IsNil()) {
       ASSERT_EQ(remote_address_, addr);
     }
-    // MI and PRIORITY attribute should be present in ping requests when port
-    // is in ICEPROTO_RFC5245 mode.
     const cricket::StunUInt32Attribute* priority_attr =
         msg->GetUInt32(STUN_ATTR_PRIORITY);
     const cricket::StunByteStringAttribute* mi_attr =
         msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY);
     const cricket::StunUInt32Attribute* fingerprint_attr =
         msg->GetUInt32(STUN_ATTR_FINGERPRINT);
-    if (src_->IceProtocol() == cricket::ICEPROTO_RFC5245) {
-      EXPECT_TRUE(priority_attr != NULL);
-      EXPECT_TRUE(mi_attr != NULL);
-      EXPECT_TRUE(fingerprint_attr != NULL);
-    } else {
-      EXPECT_TRUE(priority_attr == NULL);
-      EXPECT_TRUE(mi_attr == NULL);
-      EXPECT_TRUE(fingerprint_attr == NULL);
-    }
+    EXPECT_TRUE(priority_attr != NULL);
+    EXPECT_TRUE(mi_attr != NULL);
+    EXPECT_TRUE(fingerprint_attr != NULL);
     remote_address_ = addr;
     remote_request_.reset(CopyStunMessage(msg));
     remote_frag_ = rf;
@@ -295,20 +308,39 @@ class TestChannel : public sigslot::has_slots<> {
 
   void OnDestroyed(Connection* conn) {
     ASSERT_EQ(conn_, conn);
+    LOG(INFO) << "OnDestroy connection " << conn << " deleted";
     conn_ = NULL;
+    // When the connection is destroyed, also clear these fields so future
+    // connections are possible.
+    remote_request_.reset();
+    remote_address_.Clear();
   }
 
   void OnSrcPortDestroyed(PortInterface* port) {
-    Port* destroyed_src = src_.release();
+    Port* destroyed_src = port_.release();
     ASSERT_EQ(destroyed_src, port);
   }
 
+  Port* port() { return port_.get(); }
+
   bool nominated() const { return nominated_; }
 
+  void set_connection_ready_to_send(bool ready) {
+    connection_ready_to_send_ = ready;
+  }
+  bool connection_ready_to_send() const {
+    return connection_ready_to_send_;
+  }
+
  private:
+  // ReadyToSend will only issue after a Connection recovers from EWOULDBLOCK.
+  void OnConnectionReadyToSend(Connection* conn) {
+    ASSERT_EQ(conn, conn_);
+    connection_ready_to_send_ = true;
+  }
+
   IceMode ice_mode_;
-  rtc::scoped_ptr<Port> src_;
-  Port* dst_;
+  rtc::scoped_ptr<Port> port_;
 
   int complete_count_;
   Connection* conn_;
@@ -316,6 +348,7 @@ class TestChannel : public sigslot::has_slots<> {
   rtc::scoped_ptr<StunMessage> remote_request_;
   std::string remote_frag_;
   bool nominated_;
+  bool connection_ready_to_send_ = false;
 };
 
 class PortTest : public testing::Test, public sigslot::has_slots<> {
@@ -327,18 +360,21 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
         ss_scope_(ss_.get()),
         network_("unittest", "unittest", rtc::IPAddress(INADDR_ANY), 32),
         socket_factory_(rtc::Thread::Current()),
-        nat_factory1_(ss_.get(), kNatAddr1),
-        nat_factory2_(ss_.get(), kNatAddr2),
+        nat_factory1_(ss_.get(), kNatAddr1, SocketAddress()),
+        nat_factory2_(ss_.get(), kNatAddr2, SocketAddress()),
         nat_socket_factory1_(&nat_factory1_),
         nat_socket_factory2_(&nat_factory2_),
         stun_server_(TestStunServer::Create(main_, kStunAddr)),
         turn_server_(main_, kTurnUdpIntAddr, kTurnUdpExtAddr),
-        relay_server_(main_, kRelayUdpIntAddr, kRelayUdpExtAddr,
-                      kRelayTcpIntAddr, kRelayTcpExtAddr,
-                      kRelaySslTcpIntAddr, kRelaySslTcpExtAddr),
+        relay_server_(main_,
+                      kRelayUdpIntAddr,
+                      kRelayUdpExtAddr,
+                      kRelayTcpIntAddr,
+                      kRelayTcpExtAddr,
+                      kRelaySslTcpIntAddr,
+                      kRelaySslTcpExtAddr),
         username_(rtc::CreateRandomString(ICE_UFRAG_LENGTH)),
         password_(rtc::CreateRandomString(ICE_PWD_LENGTH)),
-        ice_protocol_(cricket::ICEPROTO_GOOGLE),
         role_conflict_(false),
         destroyed_(false) {
     network_.AddIP(rtc::IPAddress(INADDR_ANY));
@@ -347,35 +383,45 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
  protected:
   void TestLocalToLocal() {
     Port* port1 = CreateUdpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateUdpPort(kLocalAddr2);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity("udp", port1, "udp", port2, true, true, true, true);
   }
   void TestLocalToStun(NATType ntype) {
     Port* port1 = CreateUdpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     nat_server2_.reset(CreateNatServer(kNatAddr2, ntype));
     Port* port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity("udp", port1, StunName(ntype), port2,
                      ntype == NAT_OPEN_CONE, true,
                      ntype != NAT_SYMMETRIC, true);
   }
   void TestLocalToRelay(RelayType rtype, ProtocolType proto) {
     Port* port1 = CreateUdpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_UDP);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity("udp", port1, RelayName(rtype, proto), port2,
                      rtype == RELAY_GTURN, true, true, true);
   }
   void TestStunToLocal(NATType ntype) {
     nat_server1_.reset(CreateNatServer(kNatAddr1, ntype));
     Port* port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateUdpPort(kLocalAddr2);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity(StunName(ntype), port1, "udp", port2,
                      true, ntype != NAT_SYMMETRIC, true, true);
   }
   void TestStunToStun(NATType ntype1, NATType ntype2) {
     nat_server1_.reset(CreateNatServer(kNatAddr1, ntype1));
     Port* port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     nat_server2_.reset(CreateNatServer(kNatAddr2, ntype2));
     Port* port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity(StunName(ntype1), port1, StunName(ntype2), port2,
                      ntype2 == NAT_OPEN_CONE,
                      ntype1 != NAT_SYMMETRIC, ntype2 != NAT_SYMMETRIC,
@@ -384,63 +430,61 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
   void TestStunToRelay(NATType ntype, RelayType rtype, ProtocolType proto) {
     nat_server1_.reset(CreateNatServer(kNatAddr1, ntype));
     Port* port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_UDP);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity(StunName(ntype), port1, RelayName(rtype, proto), port2,
                      rtype == RELAY_GTURN, ntype != NAT_SYMMETRIC, true, true);
   }
   void TestTcpToTcp() {
     Port* port1 = CreateTcpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateTcpPort(kLocalAddr2);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity("tcp", port1, "tcp", port2, true, false, true, true);
   }
   void TestTcpToRelay(RelayType rtype, ProtocolType proto) {
     Port* port1 = CreateTcpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_TCP);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity("tcp", port1, RelayName(rtype, proto), port2,
                      rtype == RELAY_GTURN, false, true, true);
   }
   void TestSslTcpToRelay(RelayType rtype, ProtocolType proto) {
     Port* port1 = CreateTcpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
     Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_SSLTCP);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
     TestConnectivity("ssltcp", port1, RelayName(rtype, proto), port2,
                      rtype == RELAY_GTURN, false, true, true);
   }
-
   // helpers for above functions
   UDPPort* CreateUdpPort(const SocketAddress& addr) {
     return CreateUdpPort(addr, &socket_factory_);
   }
   UDPPort* CreateUdpPort(const SocketAddress& addr,
                          PacketSocketFactory* socket_factory) {
-    UDPPort* port = UDPPort::Create(main_, socket_factory, &network_,
-                                    addr.ipaddr(), 0, 0, username_, password_,
-                                    std::string());
-    port->SetIceProtocolType(ice_protocol_);
-    return port;
+    return UDPPort::Create(main_, socket_factory, &network_, addr.ipaddr(), 0,
+                           0, username_, password_, std::string(), true);
   }
   TCPPort* CreateTcpPort(const SocketAddress& addr) {
-    TCPPort* port = CreateTcpPort(addr, &socket_factory_);
-    port->SetIceProtocolType(ice_protocol_);
-    return port;
+    return CreateTcpPort(addr, &socket_factory_);
   }
   TCPPort* CreateTcpPort(const SocketAddress& addr,
                         PacketSocketFactory* socket_factory) {
-    TCPPort* port = TCPPort::Create(main_, socket_factory, &network_,
-                                    addr.ipaddr(), 0, 0, username_, password_,
-                                    true);
-    port->SetIceProtocolType(ice_protocol_);
-    return port;
+    return TCPPort::Create(main_, socket_factory, &network_,
+                           addr.ipaddr(), 0, 0, username_, password_,
+                           true);
   }
   StunPort* CreateStunPort(const SocketAddress& addr,
                            rtc::PacketSocketFactory* factory) {
     ServerAddresses stun_servers;
     stun_servers.insert(kStunAddr);
-    StunPort* port = StunPort::Create(main_, factory, &network_,
-                                      addr.ipaddr(), 0, 0,
-                                      username_, password_, stun_servers,
-                                      std::string());
-    port->SetIceProtocolType(ice_protocol_);
-    return port;
+    return StunPort::Create(main_, factory, &network_,
+                            addr.ipaddr(), 0, 0,
+                            username_, password_, stun_servers,
+                            std::string());
   }
   Port* CreateRelayPort(const SocketAddress& addr, RelayType rtype,
                         ProtocolType int_proto, ProtocolType ext_proto) {
@@ -460,14 +504,12 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
                            PacketSocketFactory* socket_factory,
                            ProtocolType int_proto, ProtocolType ext_proto,
                            const rtc::SocketAddress& server_addr) {
-    TurnPort* port = TurnPort::Create(main_, socket_factory, &network_,
-                                      addr.ipaddr(), 0, 0,
-                                      username_, password_, ProtocolAddress(
-                                         server_addr, PROTO_UDP),
-                                      kRelayCredentials, 0,
-                                      std::string());
-    port->SetIceProtocolType(ice_protocol_);
-    return port;
+    return TurnPort::Create(main_, socket_factory, &network_,
+                            addr.ipaddr(), 0, 0,
+                            username_, password_, ProtocolAddress(
+                                server_addr, PROTO_UDP),
+                            kRelayCredentials, 0,
+                            std::string());
   }
   RelayPort* CreateGturnPort(const SocketAddress& addr,
                              ProtocolType int_proto, ProtocolType ext_proto) {
@@ -478,17 +520,16 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
     return port;
   }
   RelayPort* CreateGturnPort(const SocketAddress& addr) {
-    RelayPort* port = RelayPort::Create(main_, &socket_factory_, &network_,
-                                        addr.ipaddr(), 0, 0,
-                                        username_, password_);
+    // TODO(pthatcher):  Remove GTURN.
+    return RelayPort::Create(main_, &socket_factory_, &network_,
+                             addr.ipaddr(), 0, 0,
+                             username_, password_);
     // TODO: Add an external address for ext_proto, so that the
     // other side can connect to this port using a non-UDP protocol.
-    port->SetIceProtocolType(ice_protocol_);
-    return port;
   }
   rtc::NATServer* CreateNatServer(const SocketAddress& addr,
                                         rtc::NATType type) {
-    return new rtc::NATServer(type, ss_.get(), addr, ss_.get(), addr);
+    return new rtc::NATServer(type, ss_.get(), addr, addr, ss_.get(), addr);
   }
   static const char* StunName(NATType type) {
     switch (type) {
@@ -519,19 +560,145 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
 
   void TestCrossFamilyPorts(int type);
 
+  void ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2);
+
   // This does all the work and then deletes |port1| and |port2|.
   void TestConnectivity(const char* name1, Port* port1,
                         const char* name2, Port* port2,
                         bool accept, bool same_addr1,
                         bool same_addr2, bool possible);
 
+  // This connects the provided channels which have already started.  |ch1|
+  // should have its Connection created (either through CreateConnection() or
+  // TCP reconnecting mechanism before entering this function.
+  void ConnectStartedChannels(TestChannel* ch1, TestChannel* ch2) {
+    ASSERT_TRUE(ch1->conn());
+    EXPECT_TRUE_WAIT(ch1->conn()->connected(), kTimeout);  // for TCP connect
+    ch1->Ping();
+    WAIT(!ch2->remote_address().IsNil(), kTimeout);
+
+    // Send a ping from dst to src.
+    ch2->AcceptConnection(GetCandidate(ch1->port()));
+    ch2->Ping();
+    EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch2->conn()->write_state(),
+                   kTimeout);
+  }
+
   // This connects and disconnects the provided channels in the same sequence as
   // TestConnectivity with all options set to |true|.  It does not delete either
   // channel.
-  void ConnectAndDisconnectChannels(TestChannel* ch1, TestChannel* ch2);
+  void StartConnectAndStopChannels(TestChannel* ch1, TestChannel* ch2) {
+    // Acquire addresses.
+    ch1->Start();
+    ch2->Start();
 
-  void SetIceProtocolType(cricket::IceProtocolType protocol) {
-    ice_protocol_ = protocol;
+    ch1->CreateConnection(GetCandidate(ch2->port()));
+    ConnectStartedChannels(ch1, ch2);
+
+    // Destroy the connections.
+    ch1->Stop();
+    ch2->Stop();
+  }
+
+  // This disconnects both end's Connection and make sure ch2 ready for new
+  // connection.
+  void DisconnectTcpTestChannels(TestChannel* ch1, TestChannel* ch2) {
+    TCPConnection* tcp_conn1 = static_cast<TCPConnection*>(ch1->conn());
+    TCPConnection* tcp_conn2 = static_cast<TCPConnection*>(ch2->conn());
+    ASSERT_TRUE(
+        ss_->CloseTcpConnections(tcp_conn1->socket()->GetLocalAddress(),
+                                 tcp_conn2->socket()->GetLocalAddress()));
+
+    // Wait for both OnClose are delivered.
+    EXPECT_TRUE_WAIT(!ch1->conn()->connected(), kTimeout);
+    EXPECT_TRUE_WAIT(!ch2->conn()->connected(), kTimeout);
+
+    // Ensure redundant SignalClose events on TcpConnection won't break tcp
+    // reconnection. Chromium will fire SignalClose for all outstanding IPC
+    // packets during reconnection.
+    tcp_conn1->socket()->SignalClose(tcp_conn1->socket(), 0);
+    tcp_conn2->socket()->SignalClose(tcp_conn2->socket(), 0);
+
+    // Speed up destroying ch2's connection such that the test is ready to
+    // accept a new connection from ch1 before ch1's connection destroys itself.
+    ch2->conn()->Destroy();
+    EXPECT_TRUE_WAIT(ch2->conn() == NULL, kTimeout);
+  }
+
+  void TestTcpReconnect(bool ping_after_disconnected,
+                        bool send_after_disconnected) {
+    Port* port1 = CreateTcpPort(kLocalAddr1);
+    port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
+    Port* port2 = CreateTcpPort(kLocalAddr2);
+    port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
+
+    port1->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
+    port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
+
+    // Set up channels and ensure both ports will be deleted.
+    TestChannel ch1(port1);
+    TestChannel ch2(port2);
+    EXPECT_EQ(0, ch1.complete_count());
+    EXPECT_EQ(0, ch2.complete_count());
+
+    ch1.Start();
+    ch2.Start();
+    ASSERT_EQ_WAIT(1, ch1.complete_count(), kTimeout);
+    ASSERT_EQ_WAIT(1, ch2.complete_count(), kTimeout);
+
+    // Initial connecting the channel, create connection on channel1.
+    ch1.CreateConnection(GetCandidate(port2));
+    ConnectStartedChannels(&ch1, &ch2);
+
+    // Shorten the timeout period.
+    const int kTcpReconnectTimeout = kTimeout;
+    static_cast<TCPConnection*>(ch1.conn())
+        ->set_reconnection_timeout(kTcpReconnectTimeout);
+    static_cast<TCPConnection*>(ch2.conn())
+        ->set_reconnection_timeout(kTcpReconnectTimeout);
+
+    EXPECT_FALSE(ch1.connection_ready_to_send());
+    EXPECT_FALSE(ch2.connection_ready_to_send());
+
+    // Once connected, disconnect them.
+    DisconnectTcpTestChannels(&ch1, &ch2);
+
+    if (send_after_disconnected || ping_after_disconnected) {
+      if (send_after_disconnected) {
+        // First SendData after disconnect should fail but will trigger
+        // reconnect.
+        EXPECT_EQ(-1, ch1.SendData(data, static_cast<int>(strlen(data))));
+      }
+
+      if (ping_after_disconnected) {
+        // Ping should trigger reconnect.
+        ch1.Ping();
+      }
+
+      // Wait for channel's outgoing TCPConnection connected.
+      EXPECT_TRUE_WAIT(ch1.conn()->connected(), kTimeout);
+
+      // Verify that we could still connect channels.
+      ConnectStartedChannels(&ch1, &ch2);
+      EXPECT_TRUE_WAIT(ch1.connection_ready_to_send(),
+                       kTcpReconnectTimeout);
+      // Channel2 is the passive one so a new connection is created during
+      // reconnect. This new connection should never have issued EWOULDBLOCK
+      // hence the connection_ready_to_send() should be false.
+      EXPECT_FALSE(ch2.connection_ready_to_send());
+    } else {
+      EXPECT_EQ(ch1.conn()->write_state(), Connection::STATE_WRITABLE);
+      // Since the reconnection never happens, the connections should have been
+      // destroyed after the timeout.
+      EXPECT_TRUE_WAIT(!ch1.conn(), kTcpReconnectTimeout + kTimeout);
+      EXPECT_TRUE(!ch2.conn());
+    }
+
+    // Tear down and ensure that goes smoothly.
+    ch1.Stop();
+    ch2.Stop();
+    EXPECT_TRUE_WAIT(ch1.conn() == NULL, kTimeout);
+    EXPECT_TRUE_WAIT(ch2.conn() == NULL, kTimeout);
   }
 
   IceMessage* CreateStunMessage(int type) {
@@ -558,11 +725,9 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
   TestPort* CreateTestPort(const rtc::SocketAddress& addr,
                            const std::string& username,
                            const std::string& password,
-                           cricket::IceProtocolType type,
                            cricket::IceRole role,
                            int tiebreaker) {
     TestPort* port = CreateTestPort(addr, username, password);
-    port->SetIceProtocolType(type);
     port->SetIceRole(role);
     port->SetIceTiebreaker(tiebreaker);
     return port;
@@ -586,6 +751,9 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
     return &nat_socket_factory1_;
   }
 
+ protected:
+  rtc::VirtualSocketServer* vss() { return ss_.get(); }
+
  private:
   rtc::Thread* main_;
   rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
@@ -604,7 +772,6 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
   TestRelayServer relay_server_;
   std::string username_;
   std::string password_;
-  cricket::IceProtocolType ice_protocol_;
   bool role_conflict_;
   bool destroyed_;
 };
@@ -618,8 +785,8 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
   port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
 
   // Set up channels and ensure both ports will be deleted.
-  TestChannel ch1(port1, port2);
-  TestChannel ch2(port2, port1);
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
   EXPECT_EQ(0, ch1.complete_count());
   EXPECT_EQ(0, ch2.complete_count());
 
@@ -630,7 +797,7 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
   ASSERT_EQ_WAIT(1, ch2.complete_count(), kTimeout);
 
   // Send a ping from src to dst. This may or may not make it.
-  ch1.CreateConnection();
+  ch1.CreateConnection(GetCandidate(port2));
   ASSERT_TRUE(ch1.conn() != NULL);
   EXPECT_TRUE_WAIT(ch1.conn()->connected(), kTimeout);  // for TCP connect
   ch1.Ping();
@@ -648,7 +815,7 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
     EXPECT_TRUE(same_addr2);
 
     // Send a ping from dst to src.
-    ch2.AcceptConnection();
+    ch2.AcceptConnection(GetCandidate(port1));
     ASSERT_TRUE(ch2.conn() != NULL);
     ch2.Ping();
     EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch2.conn()->write_state(),
@@ -660,14 +827,14 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
     EXPECT_TRUE(ch2.remote_address().IsNil());
 
     // Send a ping from dst to src. Again, this may or may not make it.
-    ch2.CreateConnection();
+    ch2.CreateConnection(GetCandidate(port1));
     ASSERT_TRUE(ch2.conn() != NULL);
     ch2.Ping();
     WAIT(ch2.conn()->write_state() == Connection::STATE_WRITABLE, kTimeout);
 
     if (same_addr1 && same_addr2) {
       // The new ping got back to the source.
-      EXPECT_EQ(Connection::STATE_READABLE, ch1.conn()->read_state());
+      EXPECT_TRUE(ch1.conn()->receiving());
       EXPECT_EQ(Connection::STATE_WRITABLE, ch2.conn()->write_state());
 
       // First connection may not be writable if the first ping did not get
@@ -687,11 +854,11 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
       // able to get a ping from it. This gives us the real source address.
       ch1.Ping();
       EXPECT_TRUE_WAIT(!ch2.remote_address().IsNil(), kTimeout);
-      EXPECT_EQ(Connection::STATE_READ_INIT, ch2.conn()->read_state());
+      EXPECT_FALSE(ch2.conn()->receiving());
       EXPECT_TRUE(ch1.remote_address().IsNil());
 
       // Pick up the actual address and establish the connection.
-      ch2.AcceptConnection();
+      ch2.AcceptConnection(GetCandidate(port1));
       ASSERT_TRUE(ch2.conn() != NULL);
       ch2.Ping();
       EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch2.conn()->write_state(),
@@ -700,10 +867,10 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
       // The new ping came in, but from an unexpected address. This will happen
       // when the destination NAT is symmetric.
       EXPECT_FALSE(ch1.remote_address().IsNil());
-      EXPECT_EQ(Connection::STATE_READ_INIT, ch1.conn()->read_state());
+      EXPECT_FALSE(ch1.conn()->receiving());
 
       // Update our address and complete the connection.
-      ch1.AcceptConnection();
+      ch1.AcceptConnection(GetCandidate(port2));
       ch1.Ping();
       EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch1.conn()->write_state(),
                      kTimeout);
@@ -722,14 +889,14 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
   ASSERT_TRUE(ch1.conn() != NULL);
   ASSERT_TRUE(ch2.conn() != NULL);
   if (possible) {
-    EXPECT_EQ(Connection::STATE_READABLE, ch1.conn()->read_state());
+    EXPECT_TRUE(ch1.conn()->receiving());
     EXPECT_EQ(Connection::STATE_WRITABLE, ch1.conn()->write_state());
-    EXPECT_EQ(Connection::STATE_READABLE, ch2.conn()->read_state());
+    EXPECT_TRUE(ch2.conn()->receiving());
     EXPECT_EQ(Connection::STATE_WRITABLE, ch2.conn()->write_state());
   } else {
-    EXPECT_NE(Connection::STATE_READABLE, ch1.conn()->read_state());
+    EXPECT_FALSE(ch1.conn()->receiving());
     EXPECT_NE(Connection::STATE_WRITABLE, ch1.conn()->write_state());
-    EXPECT_NE(Connection::STATE_READABLE, ch2.conn()->read_state());
+    EXPECT_FALSE(ch2.conn()->receiving());
     EXPECT_NE(Connection::STATE_WRITABLE, ch2.conn()->write_state());
   }
 
@@ -740,29 +907,6 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
   EXPECT_TRUE_WAIT(ch2.conn() == NULL, kTimeout);
 }
 
-void PortTest::ConnectAndDisconnectChannels(TestChannel* ch1,
-                                            TestChannel* ch2) {
-  // Acquire addresses.
-  ch1->Start();
-  ch2->Start();
-
-  // Send a ping from src to dst.
-  ch1->CreateConnection();
-  EXPECT_TRUE_WAIT(ch1->conn()->connected(), kTimeout);  // for TCP connect
-  ch1->Ping();
-  WAIT(!ch2->remote_address().IsNil(), kTimeout);
-
-  // Send a ping from dst to src.
-  ch2->AcceptConnection();
-  ch2->Ping();
-  EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch2->conn()->write_state(),
-                 kTimeout);
-
-  // Destroy the connections.
-  ch1->Stop();
-  ch2->Stop();
-}
-
 class FakePacketSocketFactory : public rtc::PacketSocketFactory {
  public:
   FakePacketSocketFactory()
@@ -773,8 +917,8 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
   ~FakePacketSocketFactory() override { }
 
   AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
-                                     uint16 min_port,
-                                     uint16 max_port) override {
+                                     uint16_t min_port,
+                                     uint16_t max_port) override {
     EXPECT_TRUE(next_udp_socket_ != NULL);
     AsyncPacketSocket* result = next_udp_socket_;
     next_udp_socket_ = NULL;
@@ -782,8 +926,8 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
   }
 
   AsyncPacketSocket* CreateServerTcpSocket(const SocketAddress& local_address,
-                                           uint16 min_port,
-                                           uint16 max_port,
+                                           uint16_t min_port,
+                                           uint16_t max_port,
                                            int opts) override {
     EXPECT_TRUE(next_server_tcp_socket_ != NULL);
     AsyncPacketSocket* result = next_server_tcp_socket_;
@@ -1039,6 +1183,45 @@ TEST_F(PortTest, TestTcpToTcp) {
   TestTcpToTcp();
 }
 
+TEST_F(PortTest, TestTcpReconnectOnSendPacket) {
+  TestTcpReconnect(false /* ping */, true /* send */);
+}
+
+TEST_F(PortTest, TestTcpReconnectOnPing) {
+  TestTcpReconnect(true /* ping */, false /* send */);
+}
+
+TEST_F(PortTest, TestTcpReconnectTimeout) {
+  TestTcpReconnect(false /* ping */, false /* send */);
+}
+
+// Test when TcpConnection never connects, the OnClose() will be called to
+// destroy the connection.
+TEST_F(PortTest, TestTcpNeverConnect) {
+  Port* port1 = CreateTcpPort(kLocalAddr1);
+  port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  port1->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
+
+  // Set up a channel and ensure the port will be deleted.
+  TestChannel ch1(port1);
+  EXPECT_EQ(0, ch1.complete_count());
+
+  ch1.Start();
+  ASSERT_EQ_WAIT(1, ch1.complete_count(), kTimeout);
+
+  rtc::scoped_ptr<rtc::AsyncSocket> server(
+      vss()->CreateAsyncSocket(kLocalAddr2.family(), SOCK_STREAM));
+  // Bind but not listen.
+  EXPECT_EQ(0, server->Bind(kLocalAddr2));
+
+  Candidate c = GetCandidate(port1);
+  c.set_address(server->GetLocalAddress());
+
+  ch1.CreateConnection(c);
+  EXPECT_TRUE(ch1.conn());
+  EXPECT_TRUE_WAIT(!ch1.conn(), kTimeout);  // for TCP connect
+}
+
 /* TODO: Enable these once testrelayserver can accept external TCP.
 TEST_F(PortTest, TestTcpToTcpRelay) {
   TestTcpToRelay(PROTO_TCP);
@@ -1060,19 +1243,68 @@ TEST_F(PortTest, TestSslTcpToSslTcpRelay) {
 }
 */
 
+// Test that a connection will be dead and deleted if
+// i) it has never received anything for MIN_CONNECTION_LIFETIME milliseconds
+//    since it was created, or
+// ii) it has not received anything for DEAD_CONNECTION_RECEIVE_TIMEOUT
+//     milliseconds since last receiving.
+TEST_F(PortTest, TestConnectionDead) {
+  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
+  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
+  // Acquire address.
+  ch1.Start();
+  ch2.Start();
+  ASSERT_EQ_WAIT(1, ch1.complete_count(), kTimeout);
+  ASSERT_EQ_WAIT(1, ch2.complete_count(), kTimeout);
+
+  // Test case that the connection has never received anything.
+  uint32_t before_created = rtc::Time();
+  ch1.CreateConnection(GetCandidate(port2));
+  uint32_t after_created = rtc::Time();
+  Connection* conn = ch1.conn();
+  ASSERT(conn != nullptr);
+  // It is not dead if it is after MIN_CONNECTION_LIFETIME but not pruned.
+  conn->UpdateState(after_created + MIN_CONNECTION_LIFETIME + 1);
+  rtc::Thread::Current()->ProcessMessages(0);
+  EXPECT_TRUE(ch1.conn() != nullptr);
+  // It is not dead if it is before MIN_CONNECTION_LIFETIME and pruned.
+  conn->UpdateState(before_created + MIN_CONNECTION_LIFETIME - 1);
+  conn->Prune();
+  rtc::Thread::Current()->ProcessMessages(0);
+  EXPECT_TRUE(ch1.conn() != nullptr);
+  // It will be dead after MIN_CONNECTION_LIFETIME and pruned.
+  conn->UpdateState(after_created + MIN_CONNECTION_LIFETIME + 1);
+  EXPECT_TRUE_WAIT(ch1.conn() == nullptr, kTimeout);
+
+  // Test case that the connection has received something.
+  // Create a connection again and receive a ping.
+  ch1.CreateConnection(GetCandidate(port2));
+  conn = ch1.conn();
+  ASSERT(conn != nullptr);
+  uint32_t before_last_receiving = rtc::Time();
+  conn->ReceivedPing();
+  uint32_t after_last_receiving = rtc::Time();
+  // The connection will be dead after DEAD_CONNECTION_RECEIVE_TIMEOUT
+  conn->UpdateState(
+      before_last_receiving + DEAD_CONNECTION_RECEIVE_TIMEOUT - 1);
+  rtc::Thread::Current()->ProcessMessages(100);
+  EXPECT_TRUE(ch1.conn() != nullptr);
+  conn->UpdateState(after_last_receiving + DEAD_CONNECTION_RECEIVE_TIMEOUT + 1);
+  EXPECT_TRUE_WAIT(ch1.conn() == nullptr, kTimeout);
+}
+
 // This test case verifies standard ICE features in STUN messages. Currently it
 // verifies Message Integrity attribute in STUN messages and username in STUN
 // binding request will have colon (":") between remote and local username.
-TEST_F(PortTest, TestLocalToLocalAsIce) {
-  SetIceProtocolType(cricket::ICEPROTO_RFC5245);
+TEST_F(PortTest, TestLocalToLocalStandard) {
   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
   port1->SetIceTiebreaker(kTiebreaker1);
-  ASSERT_EQ(cricket::ICEPROTO_RFC5245, port1->IceProtocol());
   UDPPort* port2 = CreateUdpPort(kLocalAddr2);
   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
   port2->SetIceTiebreaker(kTiebreaker2);
-  ASSERT_EQ(cricket::ICEPROTO_RFC5245, port2->IceProtocol());
   // Same parameters as TestLocalToLocal above.
   TestConnectivity("udp", port1, "udp", port2, true, true, true, true);
 }
@@ -1081,10 +1313,9 @@ TEST_F(PortTest, TestLocalToLocalAsIce) {
 // loopback test when protocol is RFC5245. For success IceTiebreaker, username
 // should remain equal to the request generated by the port and role of port
 // must be in controlling.
-TEST_F(PortTest, TestLoopbackCallAsIce) {
+TEST_F(PortTest, TestLoopbackCal) {
   rtc::scoped_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
-  lport->SetIceProtocolType(ICEPROTO_RFC5245);
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
   lport->PrepareAddress();
@@ -1107,7 +1338,7 @@ TEST_F(PortTest, TestLoopbackCallAsIce) {
   // response.
   lport->Reset();
   lport->AddCandidateAddress(kLocalAddr2);
-  // Creating a different connection as |conn| is in STATE_READABLE.
+  // Creating a different connection as |conn| is receiving.
   Connection* conn1 = lport->CreateConnection(lport->Candidates()[1],
                                               Port::ORIGIN_MESSAGE);
   conn1->Ping(0);
@@ -1145,12 +1376,10 @@ TEST_F(PortTest, TestLoopbackCallAsIce) {
 TEST_F(PortTest, TestIceRoleConflict) {
   rtc::scoped_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
-  lport->SetIceProtocolType(ICEPROTO_RFC5245);
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
   rtc::scoped_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  rport->SetIceProtocolType(ICEPROTO_RFC5245);
   rport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   rport->SetIceTiebreaker(kTiebreaker2);
 
@@ -1179,6 +1408,7 @@ TEST_F(PortTest, TestIceRoleConflict) {
 
 TEST_F(PortTest, TestTcpNoDelay) {
   TCPPort* port1 = CreateTcpPort(kLocalAddr1);
+  port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
   int option_value = -1;
   int success = port1->GetOption(rtc::Socket::OPT_NODELAY,
                                  &option_value);
@@ -1279,6 +1509,49 @@ TEST_F(PortTest, TestSkipCrossFamilyUdp) {
   TestCrossFamilyPorts(SOCK_DGRAM);
 }
 
+void PortTest::ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2) {
+  Connection* c = p1->CreateConnection(GetCandidate(p2),
+                                       Port::ORIGIN_MESSAGE);
+  if (can_connect) {
+    EXPECT_FALSE(NULL == c);
+    EXPECT_EQ(1U, p1->connections().size());
+  } else {
+    EXPECT_TRUE(NULL == c);
+    EXPECT_EQ(0U, p1->connections().size());
+  }
+}
+
+TEST_F(PortTest, TestUdpV6CrossTypePorts) {
+  FakePacketSocketFactory factory;
+  scoped_ptr<Port> ports[4];
+  SocketAddress addresses[4] = {SocketAddress("2001:db8::1", 0),
+                                SocketAddress("fe80::1", 0),
+                                SocketAddress("fe80::2", 0),
+                                SocketAddress("::1", 0)};
+  for (int i = 0; i < 4; i++) {
+    FakeAsyncPacketSocket *socket = new FakeAsyncPacketSocket();
+    factory.set_next_udp_socket(socket);
+    ports[i].reset(CreateUdpPort(addresses[i], &factory));
+    socket->set_state(AsyncPacketSocket::STATE_BINDING);
+    socket->SignalAddressReady(socket, addresses[i]);
+    ports[i]->PrepareAddress();
+  }
+
+  Port* standard = ports[0].get();
+  Port* link_local1 = ports[1].get();
+  Port* link_local2 = ports[2].get();
+  Port* localhost = ports[3].get();
+
+  ExpectPortsCanConnect(false, link_local1, standard);
+  ExpectPortsCanConnect(false, standard, link_local1);
+  ExpectPortsCanConnect(false, link_local1, localhost);
+  ExpectPortsCanConnect(false, localhost, link_local1);
+
+  ExpectPortsCanConnect(true, link_local1, link_local2);
+  ExpectPortsCanConnect(true, localhost, standard);
+  ExpectPortsCanConnect(true, standard, localhost);
+}
+
 // This test verifies DSCP value set through SetOption interface can be
 // get through DefaultDscpValue.
 TEST_F(PortTest, TestDefaultDscpValue) {
@@ -1315,93 +1588,14 @@ TEST_F(PortTest, TestDefaultDscpValue) {
   EXPECT_EQ(rtc::DSCP_CS6, dscp);
 }
 
-// Test sending STUN messages in GICE format.
-TEST_F(PortTest, TestSendStunMessageAsGice) {
+// Test sending STUN messages.
+TEST_F(PortTest, TestSendStunMessage) {
   rtc::scoped_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
   rtc::scoped_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  lport->SetIceProtocolType(ICEPROTO_GOOGLE);
-  rport->SetIceProtocolType(ICEPROTO_GOOGLE);
-
-  // Send a fake ping from lport to rport.
-  lport->PrepareAddress();
-  rport->PrepareAddress();
-  ASSERT_FALSE(rport->Candidates().empty());
-  Connection* conn = lport->CreateConnection(rport->Candidates()[0],
-      Port::ORIGIN_MESSAGE);
-  rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
-  conn->Ping(0);
-
-  // Check that it's a proper BINDING-REQUEST.
-  ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, 1000);
-  IceMessage* msg = lport->last_stun_msg();
-  EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
-  EXPECT_FALSE(msg->IsLegacy());
-  const StunByteStringAttribute* username_attr = msg->GetByteString(
-      STUN_ATTR_USERNAME);
-  ASSERT_TRUE(username_attr != NULL);
-  EXPECT_EQ("rfraglfrag", username_attr->GetString());
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_FINGERPRINT) == NULL);
-
-  // Save a copy of the BINDING-REQUEST for use below.
-  rtc::scoped_ptr<IceMessage> request(CopyStunMessage(msg));
-
-  // Respond with a BINDING-RESPONSE.
-  rport->SendBindingResponse(request.get(), lport->Candidates()[0].address());
-  msg = rport->last_stun_msg();
-  ASSERT_TRUE(msg != NULL);
-  EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
-  EXPECT_FALSE(msg->IsLegacy());
-  username_attr = msg->GetByteString(STUN_ATTR_USERNAME);
-  ASSERT_TRUE(username_attr != NULL);  // GICE has a username in the response.
-  EXPECT_EQ("rfraglfrag", username_attr->GetString());
-  const StunAddressAttribute* addr_attr = msg->GetAddress(
-      STUN_ATTR_MAPPED_ADDRESS);
-  ASSERT_TRUE(addr_attr != NULL);
-  EXPECT_EQ(lport->Candidates()[0].address(), addr_attr->GetAddress());
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_XOR_MAPPED_ADDRESS) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_FINGERPRINT) == NULL);
-
-  // Respond with a BINDING-ERROR-RESPONSE. This wouldn't happen in real life,
-  // but we can do it here.
-  rport->SendBindingErrorResponse(request.get(),
-                                  rport->Candidates()[0].address(),
-                                  STUN_ERROR_SERVER_ERROR,
-                                  STUN_ERROR_REASON_SERVER_ERROR);
-  msg = rport->last_stun_msg();
-  ASSERT_TRUE(msg != NULL);
-  EXPECT_EQ(STUN_BINDING_ERROR_RESPONSE, msg->type());
-  EXPECT_FALSE(msg->IsLegacy());
-  username_attr = msg->GetByteString(STUN_ATTR_USERNAME);
-  ASSERT_TRUE(username_attr != NULL);  // GICE has a username in the response.
-  EXPECT_EQ("rfraglfrag", username_attr->GetString());
-  const StunErrorCodeAttribute* error_attr = msg->GetErrorCode();
-  ASSERT_TRUE(error_attr != NULL);
-  // The GICE wire format for error codes is incorrect.
-  EXPECT_EQ(STUN_ERROR_SERVER_ERROR_AS_GICE, error_attr->code());
-  EXPECT_EQ(STUN_ERROR_SERVER_ERROR / 256, error_attr->eclass());
-  EXPECT_EQ(STUN_ERROR_SERVER_ERROR % 256, error_attr->number());
-  EXPECT_EQ(std::string(STUN_ERROR_REASON_SERVER_ERROR), error_attr->reason());
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) == NULL);
-  EXPECT_TRUE(msg->GetByteString(STUN_ATTR_FINGERPRINT) == NULL);
-}
-
-// Test sending STUN messages in ICE format.
-TEST_F(PortTest, TestSendStunMessageAsIce) {
-  rtc::scoped_ptr<TestPort> lport(
-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
-  rtc::scoped_ptr<TestPort> rport(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  lport->SetIceProtocolType(ICEPROTO_RFC5245);
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
-  rport->SetIceProtocolType(ICEPROTO_RFC5245);
   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
   rport->SetIceTiebreaker(kTiebreaker2);
 
@@ -1540,10 +1734,8 @@ TEST_F(PortTest, TestUseCandidateAttribute) {
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
   rtc::scoped_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  lport->SetIceProtocolType(ICEPROTO_RFC5245);
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
-  rport->SetIceProtocolType(ICEPROTO_RFC5245);
   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
   rport->SetIceTiebreaker(kTiebreaker2);
 
@@ -1564,79 +1756,11 @@ TEST_F(PortTest, TestUseCandidateAttribute) {
   ASSERT_TRUE(use_candidate_attr != NULL);
 }
 
-// Test handling STUN messages in GICE format.
-TEST_F(PortTest, TestHandleStunMessageAsGice) {
+// Test handling STUN messages.
+TEST_F(PortTest, TestHandleStunMessage) {
   // Our port will act as the "remote" port.
   rtc::scoped_ptr<TestPort> port(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_GOOGLE);
-
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
-  rtc::SocketAddress addr(kLocalAddr1);
-  std::string username;
-
-  // BINDING-REQUEST from local to remote with valid GICE username and no M-I.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
-                                             "rfraglfrag"));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() != NULL);  // Succeeds, since this is GICE.
-  EXPECT_EQ("lfrag", username);
-
-  // Add M-I; should be ignored and rest of message parsed normally.
-  in_msg->AddMessageIntegrity("password");
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() != NULL);
-  EXPECT_EQ("lfrag", username);
-
-  // BINDING-RESPONSE with username, as done in GICE. Should succeed.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_RESPONSE,
-                                             "rfraglfrag"));
-  in_msg->AddAttribute(
-      new StunAddressAttribute(STUN_ATTR_MAPPED_ADDRESS, kLocalAddr2));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() != NULL);
-  EXPECT_EQ("", username);
-
-  // BINDING-RESPONSE without username. Should be tolerated as well.
-  in_msg.reset(CreateStunMessage(STUN_BINDING_RESPONSE));
-  in_msg->AddAttribute(
-      new StunAddressAttribute(STUN_ATTR_MAPPED_ADDRESS, kLocalAddr2));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() != NULL);
-  EXPECT_EQ("", username);
-
-  // BINDING-ERROR-RESPONSE with username and error code.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_ERROR_RESPONSE,
-                                             "rfraglfrag"));
-  in_msg->AddAttribute(new StunErrorCodeAttribute(STUN_ATTR_ERROR_CODE,
-      STUN_ERROR_SERVER_ERROR_AS_GICE, STUN_ERROR_REASON_SERVER_ERROR));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  ASSERT_TRUE(out_msg.get() != NULL);
-  EXPECT_EQ("", username);
-  ASSERT_TRUE(out_msg->GetErrorCode() != NULL);
-  // GetStunMessage doesn't unmunge the GICE error code (happens downstream).
-  EXPECT_EQ(STUN_ERROR_SERVER_ERROR_AS_GICE, out_msg->GetErrorCode()->code());
-  EXPECT_EQ(std::string(STUN_ERROR_REASON_SERVER_ERROR),
-      out_msg->GetErrorCode()->reason());
-}
-
-// Test handling STUN messages in ICE format.
-TEST_F(PortTest, TestHandleStunMessageAsIce) {
-  // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_RFC5245);
 
   rtc::scoped_ptr<IceMessage> in_msg, out_msg;
   rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
@@ -1683,145 +1807,10 @@ TEST_F(PortTest, TestHandleStunMessageAsIce) {
       out_msg->GetErrorCode()->reason());
 }
 
-// This test verifies port can handle ICE messages in Hybrid mode and switches
-// ICEPROTO_RFC5245 mode after successfully handling the message.
-TEST_F(PortTest, TestHandleStunMessageAsIceInHybridMode) {
-  // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_HYBRID);
-
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
-  rtc::SocketAddress addr(kLocalAddr1);
-  std::string username;
-
-  // BINDING-REQUEST from local to remote with valid ICE username,
-  // MESSAGE-INTEGRITY, and FINGERPRINT.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
-                                             "rfrag:lfrag"));
-  in_msg->AddMessageIntegrity("rpass");
-  in_msg->AddFingerprint();
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() != NULL);
-  EXPECT_EQ("lfrag", username);
-  EXPECT_EQ(ICEPROTO_RFC5245, port->IceProtocol());
-}
-
-// This test verifies port can handle GICE messages in Hybrid mode and switches
-// ICEPROTO_GOOGLE mode after successfully handling the message.
-TEST_F(PortTest, TestHandleStunMessageAsGiceInHybridMode) {
-  // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_HYBRID);
-
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
-  rtc::SocketAddress addr(kLocalAddr1);
-  std::string username;
-
-  // BINDING-REQUEST from local to remote with valid GICE username and no M-I.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
-                                             "rfraglfrag"));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() != NULL);  // Succeeds, since this is GICE.
-  EXPECT_EQ("lfrag", username);
-  EXPECT_EQ(ICEPROTO_GOOGLE, port->IceProtocol());
-}
-
-// Verify port is not switched out of RFC5245 mode if GICE message is received
-// in that mode.
-TEST_F(PortTest, TestHandleStunMessageAsGiceInIceMode) {
-  // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_RFC5245);
-
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
-  rtc::SocketAddress addr(kLocalAddr1);
-  std::string username;
-
-  // BINDING-REQUEST from local to remote with valid GICE username and no M-I.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
-                                             "rfraglfrag"));
-  WriteStunMessage(in_msg.get(), buf.get());
-  // Should fail as there is no MI and fingerprint.
-  EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                    out_msg.accept(), &username));
-  EXPECT_EQ(ICEPROTO_RFC5245, port->IceProtocol());
-}
-
-
-// Tests handling of GICE binding requests with missing or incorrect usernames.
-TEST_F(PortTest, TestHandleStunMessageAsGiceBadUsername) {
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_GOOGLE);
-
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
-  rtc::SocketAddress addr(kLocalAddr1);
-  std::string username;
-
-  // BINDING-REQUEST with no username.
-  in_msg.reset(CreateStunMessage(STUN_BINDING_REQUEST));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() == NULL);
-  EXPECT_EQ("", username);
-  EXPECT_EQ(STUN_ERROR_BAD_REQUEST_AS_GICE, port->last_stun_error_code());
-
-  // BINDING-REQUEST with empty username.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST, ""));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() == NULL);
-  EXPECT_EQ("", username);
-  EXPECT_EQ(STUN_ERROR_UNAUTHORIZED_AS_GICE, port->last_stun_error_code());
-
-  // BINDING-REQUEST with too-short username.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "lfra"));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() == NULL);
-  EXPECT_EQ("", username);
-  EXPECT_EQ(STUN_ERROR_UNAUTHORIZED_AS_GICE, port->last_stun_error_code());
-
-  // BINDING-REQUEST with reversed username.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
-                                             "lfragrfrag"));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() == NULL);
-  EXPECT_EQ("", username);
-  EXPECT_EQ(STUN_ERROR_UNAUTHORIZED_AS_GICE, port->last_stun_error_code());
-
-  // BINDING-REQUEST with garbage username.
-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
-                                             "abcdefgh"));
-  WriteStunMessage(in_msg.get(), buf.get());
-  EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr,
-                                   out_msg.accept(), &username));
-  EXPECT_TRUE(out_msg.get() == NULL);
-  EXPECT_EQ("", username);
-  EXPECT_EQ(STUN_ERROR_UNAUTHORIZED_AS_GICE, port->last_stun_error_code());
-}
-
 // Tests handling of ICE binding requests with missing or incorrect usernames.
-TEST_F(PortTest, TestHandleStunMessageAsIceBadUsername) {
+TEST_F(PortTest, TestHandleStunMessageBadUsername) {
   rtc::scoped_ptr<TestPort> port(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_RFC5245);
 
   rtc::scoped_ptr<IceMessage> in_msg, out_msg;
   rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
@@ -1886,12 +1875,11 @@ TEST_F(PortTest, TestHandleStunMessageAsIceBadUsername) {
   EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code());
 }
 
-// Test handling STUN messages (as ICE) with missing or malformed M-I.
-TEST_F(PortTest, TestHandleStunMessageAsIceBadMessageIntegrity) {
+// Test handling STUN messages with missing or malformed M-I.
+TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
   // Our port will act as the "remote" port.
   rtc::scoped_ptr<TestPort> port(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_RFC5245);
 
   rtc::scoped_ptr<IceMessage> in_msg, out_msg;
   rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
@@ -1928,12 +1916,11 @@ TEST_F(PortTest, TestHandleStunMessageAsIceBadMessageIntegrity) {
   // Change this test to pass in data via Connection::OnReadPacket instead.
 }
 
-// Test handling STUN messages (as ICE) with missing or malformed FINGERPRINT.
-TEST_F(PortTest, TestHandleStunMessageAsIceBadFingerprint) {
+// Test handling STUN messages with missing or malformed FINGERPRINT.
+TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
   // Our port will act as the "remote" port.
   rtc::scoped_ptr<TestPort> port(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  port->SetIceProtocolType(ICEPROTO_RFC5245);
 
   rtc::scoped_ptr<IceMessage> in_msg, out_msg;
   rtc::scoped_ptr<ByteBuffer> buf(new ByteBuffer());
@@ -1995,12 +1982,11 @@ TEST_F(PortTest, TestHandleStunMessageAsIceBadFingerprint) {
   EXPECT_EQ(0, port->last_stun_error_code());
 }
 
-// Test handling of STUN binding indication messages (as ICE). STUN binding
+// Test handling of STUN binding indication messages . STUN binding
 // indications are allowed only to the connection which is in read mode.
 TEST_F(PortTest, TestHandleStunBindingIndication) {
   rtc::scoped_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr2, "lfrag", "lpass"));
-  lport->SetIceProtocolType(ICEPROTO_RFC5245);
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
 
@@ -2023,7 +2009,6 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
   // last_ping_received.
   rtc::scoped_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
-  rport->SetIceProtocolType(ICEPROTO_RFC5245);
   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
   rport->SetIceTiebreaker(kTiebreaker2);
 
@@ -2047,13 +2032,13 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
                       rtc::PacketTime());
   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, 1000);
   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
-  uint32 last_ping_received1 = lconn->last_ping_received();
+  uint32_t last_ping_received1 = lconn->last_ping_received();
 
   // Adding a delay of 100ms.
   rtc::Thread::Current()->ProcessMessages(100);
   // Pinging lconn using stun indication message.
   lconn->OnReadPacket(buf->Data(), buf->Length(), rtc::PacketTime());
-  uint32 last_ping_received2 = lconn->last_ping_received();
+  uint32_t last_ping_received2 = lconn->last_ping_received();
   EXPECT_GT(last_ping_received2, last_ping_received1);
 }
 
@@ -2073,15 +2058,15 @@ TEST_F(PortTest, TestComputeCandidatePriority) {
   port->AddCandidateAddress(SocketAddress("3ffe::1234:5678", 1234));
   // These should all be:
   // (90 << 24) | ([rfc3484 pref value] << 8) | (256 - 177)
-  uint32 expected_priority_v4 = 1509957199U;
-  uint32 expected_priority_v6 = 1509959759U;
-  uint32 expected_priority_ula = 1509962319U;
-  uint32 expected_priority_v4mapped = expected_priority_v4;
-  uint32 expected_priority_v4compat = 1509949775U;
-  uint32 expected_priority_6to4 = 1509954639U;
-  uint32 expected_priority_teredo = 1509952079U;
-  uint32 expected_priority_sitelocal = 1509949775U;
-  uint32 expected_priority_6bone = 1509949775U;
+  uint32_t expected_priority_v4 = 1509957199U;
+  uint32_t expected_priority_v6 = 1509959759U;
+  uint32_t expected_priority_ula = 1509962319U;
+  uint32_t expected_priority_v4mapped = expected_priority_v4;
+  uint32_t expected_priority_v4compat = 1509949775U;
+  uint32_t expected_priority_6to4 = 1509954639U;
+  uint32_t expected_priority_teredo = 1509952079U;
+  uint32_t expected_priority_sitelocal = 1509949775U;
+  uint32_t expected_priority_6bone = 1509949775U;
   ASSERT_EQ(expected_priority_v4, port->Candidates()[0].priority());
   ASSERT_EQ(expected_priority_v6, port->Candidates()[1].priority());
   ASSERT_EQ(expected_priority_ula, port->Candidates()[2].priority());
@@ -2093,21 +2078,6 @@ TEST_F(PortTest, TestComputeCandidatePriority) {
   ASSERT_EQ(expected_priority_6bone, port->Candidates()[8].priority());
 }
 
-TEST_F(PortTest, TestPortProxyProperties) {
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr1, "name", "pass"));
-  port->SetIceRole(cricket::ICEROLE_CONTROLLING);
-  port->SetIceTiebreaker(kTiebreaker1);
-
-  // Create a proxy port.
-  rtc::scoped_ptr<PortProxy> proxy(new PortProxy());
-  proxy->set_impl(port.get());
-  EXPECT_EQ(port->Type(), proxy->Type());
-  EXPECT_EQ(port->Network(), proxy->Network());
-  EXPECT_EQ(port->GetIceRole(), proxy->GetIceRole());
-  EXPECT_EQ(port->IceTiebreaker(), proxy->IceTiebreaker());
-}
-
 // In the case of shared socket, one port may be shared by local and stun.
 // Test that candidates with different types will have different foundation.
 TEST_F(PortTest, TestFoundation) {
@@ -2241,12 +2211,12 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
 }
 
 // Test priority value overflow handling when preference is set to 3.
-TEST_F(PortTest, TestCandidatePreference) {
+TEST_F(PortTest, TestCandidatePriority) {
   cricket::Candidate cand1;
-  cand1.set_preference(3);
+  cand1.set_priority(3);
   cricket::Candidate cand2;
-  cand2.set_preference(1);
-  EXPECT_TRUE(cand1.preference() > cand2.preference());
+  cand2.set_priority(1);
+  EXPECT_TRUE(cand1.priority() > cand2.priority());
 }
 
 // Test the Connection priority is calculated correctly.
@@ -2290,11 +2260,13 @@ TEST_F(PortTest, TestConnectionPriority) {
 
 TEST_F(PortTest, TestWritableState) {
   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
+  port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
   UDPPort* port2 = CreateUdpPort(kLocalAddr2);
+  port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
 
   // Set up channels.
-  TestChannel ch1(port1, port2);
-  TestChannel ch2(port2, port1);
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
 
   // Acquire addresses.
   ch1.Start();
@@ -2303,7 +2275,7 @@ TEST_F(PortTest, TestWritableState) {
   ASSERT_EQ_WAIT(1, ch2.complete_count(), kTimeout);
 
   // Send a ping from src to dst.
-  ch1.CreateConnection();
+  ch1.CreateConnection(GetCandidate(port2));
   ASSERT_TRUE(ch1.conn() != NULL);
   EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state());
   EXPECT_TRUE_WAIT(ch1.conn()->connected(), kTimeout);  // for TCP connect
@@ -2312,13 +2284,13 @@ TEST_F(PortTest, TestWritableState) {
 
   // Data should be unsendable until the connection is accepted.
   char data[] = "abcd";
-  int data_size = ARRAY_SIZE(data);
+  int data_size = arraysize(data);
   rtc::PacketOptions options;
   EXPECT_EQ(SOCKET_ERROR, ch1.conn()->Send(data, data_size, options));
 
   // Accept the connection to return the binding response, transition to
   // writable, and allow data to be sent.
-  ch2.AcceptConnection();
+  ch2.AcceptConnection(GetCandidate(port1));
   EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch1.conn()->write_state(),
                  kTimeout);
   EXPECT_EQ(data_size, ch1.conn()->Send(data, data_size, options));
@@ -2326,10 +2298,10 @@ TEST_F(PortTest, TestWritableState) {
   // Ask the connection to update state as if enough time has passed to lose
   // full writability and 5 pings went unresponded to. We'll accomplish the
   // latter by sending pings but not pumping messages.
-  for (uint32 i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
+  for (uint32_t i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
     ch1.Ping(i);
   }
-  uint32 unreliable_timeout_delay = CONNECTION_WRITE_CONNECT_TIMEOUT + 500u;
+  uint32_t unreliable_timeout_delay = CONNECTION_WRITE_CONNECT_TIMEOUT + 500u;
   ch1.conn()->UpdateState(unreliable_timeout_delay);
   EXPECT_EQ(Connection::STATE_WRITE_UNRELIABLE, ch1.conn()->write_state());
 
@@ -2343,7 +2315,7 @@ TEST_F(PortTest, TestWritableState) {
 
   // Wait long enough for a full timeout (past however long we've already
   // waited).
-  for (uint32 i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
+  for (uint32_t i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
     ch1.Ping(unreliable_timeout_delay + i);
   }
   ch1.conn()->UpdateState(unreliable_timeout_delay + CONNECTION_WRITE_TIMEOUT +
@@ -2359,22 +2331,24 @@ TEST_F(PortTest, TestWritableState) {
 
 TEST_F(PortTest, TestTimeoutForNeverWritable) {
   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
+  port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
   UDPPort* port2 = CreateUdpPort(kLocalAddr2);
+  port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
 
   // Set up channels.
-  TestChannel ch1(port1, port2);
-  TestChannel ch2(port2, port1);
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
 
   // Acquire addresses.
   ch1.Start();
   ch2.Start();
 
-  ch1.CreateConnection();
+  ch1.CreateConnection(GetCandidate(port2));
   ASSERT_TRUE(ch1.conn() != NULL);
   EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state());
 
   // Attempt to go directly to write timeout.
-  for (uint32 i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
+  for (uint32_t i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
     ch1.Ping(i);
   }
   ch1.conn()->UpdateState(CONNECTION_WRITE_TIMEOUT + 500u);
@@ -2387,14 +2361,14 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) {
 // port which responds to the ping message just like LITE client.
 TEST_F(PortTest, TestIceLiteConnectivity) {
   TestPort* ice_full_port = CreateTestPort(
-      kLocalAddr1, "lfrag", "lpass", cricket::ICEPROTO_RFC5245,
+      kLocalAddr1, "lfrag", "lpass",
       cricket::ICEROLE_CONTROLLING, kTiebreaker1);
 
   rtc::scoped_ptr<TestPort> ice_lite_port(CreateTestPort(
-      kLocalAddr2, "rfrag", "rpass", cricket::ICEPROTO_RFC5245,
+      kLocalAddr2, "rfrag", "rpass",
       cricket::ICEROLE_CONTROLLED, kTiebreaker2));
   // Setup TestChannel. This behaves like FULL mode client.
-  TestChannel ch1(ice_full_port, ice_lite_port.get());
+  TestChannel ch1(ice_full_port);
   ch1.SetIceMode(ICEMODE_FULL);
 
   // Start gathering candidates.
@@ -2404,7 +2378,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
   ASSERT_EQ_WAIT(1, ch1.complete_count(), kTimeout);
   ASSERT_FALSE(ice_lite_port->Candidates().empty());
 
-  ch1.CreateConnection();
+  ch1.CreateConnection(GetCandidate(ice_lite_port.get()));
   ASSERT_TRUE(ch1.conn() != NULL);
   EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state());
 
@@ -2450,7 +2424,6 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
 
 // This test case verifies that the CONTROLLING port does not time out.
 TEST_F(PortTest, TestControllingNoTimeout) {
-  SetIceProtocolType(cricket::ICEPROTO_RFC5245);
   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
   ConnectToSignalDestroyed(port1);
   port1->set_timeout_delay(10);  // milliseconds
@@ -2462,11 +2435,11 @@ TEST_F(PortTest, TestControllingNoTimeout) {
   port2->SetIceTiebreaker(kTiebreaker2);
 
   // Set up channels and ensure both ports will be deleted.
-  TestChannel ch1(port1, port2);
-  TestChannel ch2(port2, port1);
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
 
   // Simulate a connection that succeeds, and then is destroyed.
-  ConnectAndDisconnectChannels(&ch1, &ch2);
+  StartConnectAndStopChannels(&ch1, &ch2);
 
   // After the connection is destroyed, the port should not be destroyed.
   rtc::Thread::Current()->ProcessMessages(kTimeout);
@@ -2476,7 +2449,6 @@ TEST_F(PortTest, TestControllingNoTimeout) {
 // This test case verifies that the CONTROLLED port does time out, but only
 // after connectivity is lost.
 TEST_F(PortTest, TestControlledTimeout) {
-  SetIceProtocolType(cricket::ICEPROTO_RFC5245);
   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
   port1->SetIceTiebreaker(kTiebreaker1);
@@ -2494,12 +2466,68 @@ TEST_F(PortTest, TestControlledTimeout) {
   port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
 
   // Set up channels and ensure both ports will be deleted.
-  TestChannel ch1(port1, port2);
-  TestChannel ch2(port2, port1);
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
 
   // Simulate a connection that succeeds, and then is destroyed.
-  ConnectAndDisconnectChannels(&ch1, &ch2);
+  StartConnectAndStopChannels(&ch1, &ch2);
 
   // The controlled port should be destroyed after 10 milliseconds.
   EXPECT_TRUE_WAIT(destroyed(), kTimeout);
 }
+
+// This test case verifies that if the role of a port changes from controlled
+// to controlling after all connections fail, the port will not be destroyed.
+TEST_F(PortTest, TestControlledToControllingNotDestroyed) {
+  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
+  port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  port1->SetIceTiebreaker(kTiebreaker1);
+
+  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
+  ConnectToSignalDestroyed(port2);
+  port2->set_timeout_delay(10);  // milliseconds
+  port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
+  port2->SetIceTiebreaker(kTiebreaker2);
+
+  // The connection must not be destroyed before a connection is attempted.
+  EXPECT_FALSE(destroyed());
+
+  port1->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
+  port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
+
+  // Set up channels and ensure both ports will be deleted.
+  TestChannel ch1(port1);
+  TestChannel ch2(port2);
+
+  // Simulate a connection that succeeds, and then is destroyed.
+  StartConnectAndStopChannels(&ch1, &ch2);
+  // Switch the role after all connections are destroyed.
+  EXPECT_TRUE_WAIT(ch2.conn() == nullptr, kTimeout);
+  port1->SetIceRole(cricket::ICEROLE_CONTROLLED);
+  port2->SetIceRole(cricket::ICEROLE_CONTROLLING);
+
+  // After the connection is destroyed, the port should not be destroyed.
+  rtc::Thread::Current()->ProcessMessages(kTimeout);
+  EXPECT_FALSE(destroyed());
+}
+
+TEST_F(PortTest, TestSupportsProtocol) {
+  rtc::scoped_ptr<Port> udp_port(CreateUdpPort(kLocalAddr1));
+  EXPECT_TRUE(udp_port->SupportsProtocol(UDP_PROTOCOL_NAME));
+  EXPECT_FALSE(udp_port->SupportsProtocol(TCP_PROTOCOL_NAME));
+
+  rtc::scoped_ptr<Port> stun_port(
+      CreateStunPort(kLocalAddr1, nat_socket_factory1()));
+  EXPECT_TRUE(stun_port->SupportsProtocol(UDP_PROTOCOL_NAME));
+  EXPECT_FALSE(stun_port->SupportsProtocol(TCP_PROTOCOL_NAME));
+
+  rtc::scoped_ptr<Port> tcp_port(CreateTcpPort(kLocalAddr1));
+  EXPECT_TRUE(tcp_port->SupportsProtocol(TCP_PROTOCOL_NAME));
+  EXPECT_TRUE(tcp_port->SupportsProtocol(SSLTCP_PROTOCOL_NAME));
+  EXPECT_FALSE(tcp_port->SupportsProtocol(UDP_PROTOCOL_NAME));
+
+  rtc::scoped_ptr<Port> turn_port(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  EXPECT_TRUE(turn_port->SupportsProtocol(UDP_PROTOCOL_NAME));
+  EXPECT_FALSE(turn_port->SupportsProtocol(TCP_PROTOCOL_NAME));
+}
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portallocator.cc b/media/webrtc/trunk/webrtc/p2p/base/portallocator.cc
index 5ac58ea9de..5c4243abf6 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/portallocator.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/portallocator.cc
@@ -8,36 +8,24 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/checks.h"
 #include "webrtc/p2p/base/portallocator.h"
 
-#include "webrtc/p2p/base/portallocatorsessionproxy.h"
-
 namespace cricket {
 
 PortAllocatorSession::PortAllocatorSession(const std::string& content_name,
                                            int component,
                                            const std::string& ice_ufrag,
                                            const std::string& ice_pwd,
-                                           uint32 flags)
+                                           uint32_t flags)
     : content_name_(content_name),
       component_(component),
       flags_(flags),
       generation_(0),
-      // If PORTALLOCATOR_ENABLE_SHARED_UFRAG flag is not enabled, ignore the
-      // incoming ufrag and pwd, which will cause each Port to generate one
-      // by itself.
-      username_(flags_ & PORTALLOCATOR_ENABLE_SHARED_UFRAG ? ice_ufrag : ""),
-      password_(flags_ & PORTALLOCATOR_ENABLE_SHARED_UFRAG ? ice_pwd : "") {
-  // If bundle is enabled, shared ufrag must be enabled too.
-  ASSERT((!(flags_ & PORTALLOCATOR_ENABLE_BUNDLE)) ||
-         (flags_ & PORTALLOCATOR_ENABLE_SHARED_UFRAG));
-}
-
-PortAllocator::~PortAllocator() {
-  for (SessionMuxerMap::iterator iter = muxers_.begin();
-       iter != muxers_.end(); ++iter) {
-    delete iter->second;
-  }
+      ice_ufrag_(ice_ufrag),
+      ice_pwd_(ice_pwd) {
+  RTC_DCHECK(!ice_ufrag.empty());
+  RTC_DCHECK(!ice_pwd.empty());
 }
 
 PortAllocatorSession* PortAllocator::CreateSession(
@@ -46,50 +34,7 @@ PortAllocatorSession* PortAllocator::CreateSession(
     int component,
     const std::string& ice_ufrag,
     const std::string& ice_pwd) {
-  if (flags_ & PORTALLOCATOR_ENABLE_BUNDLE) {
-    // If we just use |sid| as key in identifying PortAllocatorSessionMuxer,
-    // ICE restart will not result in different candidates, as |sid| will
-    // be same. To yield different candiates we are using combination of
-    // |ice_ufrag| and |ice_pwd|.
-    // Ideally |ice_ufrag| and |ice_pwd| should change together, but
-    // there can be instances where only ice_pwd will be changed.
-    std::string key_str = ice_ufrag + ":" + ice_pwd;
-    PortAllocatorSessionMuxer* muxer = GetSessionMuxer(key_str);
-    if (!muxer) {
-      PortAllocatorSession* session_impl = CreateSessionInternal(
-          content_name, component, ice_ufrag, ice_pwd);
-      // Create PortAllocatorSessionMuxer object for |session_impl|.
-      muxer = new PortAllocatorSessionMuxer(session_impl);
-      muxer->SignalDestroyed.connect(
-          this, &PortAllocator::OnSessionMuxerDestroyed);
-      // Add PortAllocatorSession to the map.
-      muxers_[key_str] = muxer;
-    }
-    PortAllocatorSessionProxy* proxy =
-        new PortAllocatorSessionProxy(content_name, component, flags_);
-    muxer->RegisterSessionProxy(proxy);
-    return proxy;
-  }
   return CreateSessionInternal(content_name, component, ice_ufrag, ice_pwd);
 }
 
-PortAllocatorSessionMuxer* PortAllocator::GetSessionMuxer(
-    const std::string& key) const {
-  SessionMuxerMap::const_iterator iter = muxers_.find(key);
-  if (iter != muxers_.end())
-    return iter->second;
-  return NULL;
-}
-
-void PortAllocator::OnSessionMuxerDestroyed(
-    PortAllocatorSessionMuxer* session) {
-  SessionMuxerMap::iterator iter;
-  for (iter = muxers_.begin(); iter != muxers_.end(); ++iter) {
-    if (iter->second == session)
-      break;
-  }
-  if (iter != muxers_.end())
-    muxers_.erase(iter);
-}
-
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portallocator.h b/media/webrtc/trunk/webrtc/p2p/base/portallocator.h
index 654358504b..6fb79b065e 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/portallocator.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/portallocator.h
@@ -14,6 +14,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/p2p/base/port.h"
 #include "webrtc/p2p/base/portinterface.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/proxyinfo.h"
@@ -28,25 +29,43 @@ namespace cricket {
 // what kinds of ports are allocated.
 
 enum {
+  // Disable local UDP ports. This doesn't impact how we connect to relay
+  // servers.
   PORTALLOCATOR_DISABLE_UDP = 0x01,
   PORTALLOCATOR_DISABLE_STUN = 0x02,
   PORTALLOCATOR_DISABLE_RELAY = 0x04,
+  // Disable local TCP ports. This doesn't impact how we connect to relay
+  // servers.
   PORTALLOCATOR_DISABLE_TCP = 0x08,
   PORTALLOCATOR_ENABLE_SHAKER = 0x10,
-  PORTALLOCATOR_ENABLE_BUNDLE = 0x20,
   PORTALLOCATOR_ENABLE_IPV6 = 0x40,
+  // TODO(pthatcher): Remove this once it's no longer used in:
+  // remoting/client/plugin/pepper_port_allocator.cc
+  // remoting/protocol/chromium_port_allocator.cc
+  // remoting/test/fake_port_allocator.cc
+  // It's a no-op and is no longer needed.
   PORTALLOCATOR_ENABLE_SHARED_UFRAG = 0x80,
   PORTALLOCATOR_ENABLE_SHARED_SOCKET = 0x100,
   PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE = 0x200,
+  // When specified, we'll only allocate the STUN candidate for the public
+  // interface as seen by regular http traffic and the HOST candidate associated
+  // with the default local interface.
   PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION = 0x400,
+  // When specified along with PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION, the
+  // default local candidate mentioned above will not be allocated. Only the
+  // STUN candidate will be.
+  PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE = 0x800,
+  // Disallow use of UDP when connecting to a relay server. Since proxy servers
+  // usually don't handle UDP, using UDP will leak the IP address.
+  PORTALLOCATOR_DISABLE_UDP_RELAY = 0x1000,
 };
 
-const uint32 kDefaultPortAllocatorFlags = 0;
+const uint32_t kDefaultPortAllocatorFlags = 0;
 
-const uint32 kDefaultStepDelay = 1000;  // 1 sec step delay.
+const uint32_t kDefaultStepDelay = 1000;  // 1 sec step delay.
 // As per RFC 5245 Appendix B.1, STUN transactions need to be paced at certain
 // internal. Less than 20ms is not acceptable. We choose 50ms as our default.
-const uint32 kMinimumStepDelay = 50;
+const uint32_t kMinimumStepDelay = 50;
 
 // CF = CANDIDATE FILTER
 enum {
@@ -57,29 +76,61 @@ enum {
   CF_ALL = 0x7,
 };
 
-class PortAllocatorSessionMuxer;
+// TODO(deadbeef): Rename to TurnCredentials (and username to ufrag).
+struct RelayCredentials {
+  RelayCredentials() {}
+  RelayCredentials(const std::string& username, const std::string& password)
+      : username(username), password(password) {}
+
+  std::string username;
+  std::string password;
+};
+
+typedef std::vector<ProtocolAddress> PortList;
+// TODO(deadbeef): Rename to TurnServerConfig.
+struct RelayServerConfig {
+  RelayServerConfig(RelayType type) : type(type), priority(0) {}
+
+  RelayServerConfig(const std::string& address,
+                    int port,
+                    const std::string& username,
+                    const std::string& password,
+                    ProtocolType proto,
+                    bool secure)
+      : type(RELAY_TURN), credentials(username, password) {
+    ports.push_back(
+        ProtocolAddress(rtc::SocketAddress(address, port), proto, secure));
+  }
+
+  RelayType type;
+  PortList ports;
+  RelayCredentials credentials;
+  int priority;
+};
 
 class PortAllocatorSession : public sigslot::has_slots<> {
  public:
   // Content name passed in mostly for logging and debugging.
-  // TODO(mallinath) - Change username and password to ice_ufrag and ice_pwd.
   PortAllocatorSession(const std::string& content_name,
                        int component,
-                       const std::string& username,
-                       const std::string& password,
-                       uint32 flags);
+                       const std::string& ice_ufrag,
+                       const std::string& ice_pwd,
+                       uint32_t flags);
 
   // Subclasses should clean up any ports created.
   virtual ~PortAllocatorSession() {}
 
-  uint32 flags() const { return flags_; }
-  void set_flags(uint32 flags) { flags_ = flags; }
+  uint32_t flags() const { return flags_; }
+  void set_flags(uint32_t flags) { flags_ = flags; }
   std::string content_name() const { return content_name_; }
   int component() const { return component_; }
 
   // Starts gathering STUN and Relay configurations.
   virtual void StartGettingPorts() = 0;
   virtual void StopGettingPorts() = 0;
+  // Only stop the existing gathering process but may start new ones if needed.
+  virtual void ClearGettingPorts() = 0;
+  // Whether the process of getting ports has been stopped.
   virtual bool IsGettingPorts() = 0;
 
   sigslot::signal2<PortAllocatorSession*, PortInterface*> SignalPortReady;
@@ -87,22 +138,27 @@ class PortAllocatorSession : public sigslot::has_slots<> {
                    const std::vector<Candidate>&> SignalCandidatesReady;
   sigslot::signal1<PortAllocatorSession*> SignalCandidatesAllocationDone;
 
-  virtual uint32 generation() { return generation_; }
-  virtual void set_generation(uint32 generation) { generation_ = generation; }
+  virtual uint32_t generation() { return generation_; }
+  virtual void set_generation(uint32_t generation) { generation_ = generation; }
   sigslot::signal1<PortAllocatorSession*> SignalDestroyed;
 
+  const std::string& ice_ufrag() const { return ice_ufrag_; }
+  const std::string& ice_pwd() const { return ice_pwd_; }
+
  protected:
-  const std::string& username() const { return username_; }
-  const std::string& password() const { return password_; }
+  // TODO(deadbeef): Get rid of these when everyone switches to ice_ufrag and
+  // ice_pwd.
+  const std::string& username() const { return ice_ufrag_; }
+  const std::string& password() const { return ice_pwd_; }
 
   std::string content_name_;
   int component_;
 
  private:
-  uint32 flags_;
-  uint32 generation_;
-  std::string username_;
-  std::string password_;
+  uint32_t flags_;
+  uint32_t generation_;
+  std::string ice_ufrag_;
+  std::string ice_pwd_;
 };
 
 class PortAllocator : public sigslot::has_slots<> {
@@ -116,7 +172,19 @@ class PortAllocator : public sigslot::has_slots<> {
       candidate_filter_(CF_ALL) {
     // This will allow us to have old behavior on non webrtc clients.
   }
-  virtual ~PortAllocator();
+  virtual ~PortAllocator() {}
+
+  // Set STUN and TURN servers to be used in future sessions.
+  virtual void SetIceServers(
+      const ServerAddresses& stun_servers,
+      const std::vector<RelayServerConfig>& turn_servers) = 0;
+
+  // Sets the network types to ignore.
+  // Values are defined by the AdapterType enum.
+  // For instance, calling this with
+  // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
+  // loopback interfaces.
+  virtual void SetNetworkIgnoreMask(int network_ignore_mask) = 0;
 
   PortAllocatorSession* CreateSession(
       const std::string& sid,
@@ -125,11 +193,8 @@ class PortAllocator : public sigslot::has_slots<> {
       const std::string& ice_ufrag,
       const std::string& ice_pwd);
 
-  PortAllocatorSessionMuxer* GetSessionMuxer(const std::string& key) const;
-  void OnSessionMuxerDestroyed(PortAllocatorSessionMuxer* session);
-
-  uint32 flags() const { return flags_; }
-  void set_flags(uint32 flags) { flags_ = flags; }
+  uint32_t flags() const { return flags_; }
+  void set_flags(uint32_t flags) { flags_ = flags; }
 
   const std::string& user_agent() const { return agent_; }
   const rtc::ProxyInfo& proxy() const { return proxy_; }
@@ -151,18 +216,16 @@ class PortAllocator : public sigslot::has_slots<> {
     return true;
   }
 
-  uint32 step_delay() const { return step_delay_; }
-  void set_step_delay(uint32 delay) {
-    step_delay_ = delay;
-  }
+  uint32_t step_delay() const { return step_delay_; }
+  void set_step_delay(uint32_t delay) { step_delay_ = delay; }
 
   bool allow_tcp_listen() const { return allow_tcp_listen_; }
   void set_allow_tcp_listen(bool allow_tcp_listen) {
     allow_tcp_listen_ = allow_tcp_listen;
   }
 
-  uint32 candidate_filter() { return candidate_filter_; }
-  bool set_candidate_filter(uint32 filter) {
+  uint32_t candidate_filter() { return candidate_filter_; }
+  bool set_candidate_filter(uint32_t filter) {
     // TODO(mallinath) - Do transition check?
     candidate_filter_ = filter;
     return true;
@@ -179,17 +242,14 @@ class PortAllocator : public sigslot::has_slots<> {
       const std::string& ice_ufrag,
       const std::string& ice_pwd) = 0;
 
-  typedef std::map<std::string, PortAllocatorSessionMuxer*> SessionMuxerMap;
-
-  uint32 flags_;
+  uint32_t flags_;
   std::string agent_;
   rtc::ProxyInfo proxy_;
   int min_port_;
   int max_port_;
-  uint32 step_delay_;
-  SessionMuxerMap muxers_;
+  uint32_t step_delay_;
   bool allow_tcp_listen_;
-  uint32 candidate_filter_;
+  uint32_t candidate_filter_;
   std::string origin_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy.cc b/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy.cc
deleted file mode 100644
index f5ce9a4a63..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy.cc
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/p2p/base/portallocatorsessionproxy.h"
-
-#include "webrtc/p2p/base/portallocator.h"
-#include "webrtc/p2p/base/portproxy.h"
-#include "webrtc/base/thread.h"
-
-namespace cricket {
-
-enum {
-  MSG_SEND_ALLOCATION_DONE = 1,
-  MSG_SEND_ALLOCATED_PORTS,
-};
-
-typedef rtc::TypedMessageData<PortAllocatorSessionProxy*> ProxyObjData;
-
-PortAllocatorSessionMuxer::PortAllocatorSessionMuxer(
-    PortAllocatorSession* session)
-    : worker_thread_(rtc::Thread::Current()),
-      session_(session),
-      candidate_done_signal_received_(false) {
-  session_->SignalPortReady.connect(
-      this, &PortAllocatorSessionMuxer::OnPortReady);
-  session_->SignalCandidatesAllocationDone.connect(
-      this, &PortAllocatorSessionMuxer::OnCandidatesAllocationDone);
-}
-
-PortAllocatorSessionMuxer::~PortAllocatorSessionMuxer() {
-  for (size_t i = 0; i < session_proxies_.size(); ++i)
-    delete session_proxies_[i];
-
-  SignalDestroyed(this);
-}
-
-void PortAllocatorSessionMuxer::RegisterSessionProxy(
-    PortAllocatorSessionProxy* session_proxy) {
-  session_proxies_.push_back(session_proxy);
-  session_proxy->SignalDestroyed.connect(
-      this, &PortAllocatorSessionMuxer::OnSessionProxyDestroyed);
-  session_proxy->set_impl(session_.get());
-
-  // Populate new proxy session with the information available in the actual
-  // implementation.
-  if (!ports_.empty()) {
-    worker_thread_->Post(
-        this, MSG_SEND_ALLOCATED_PORTS, new ProxyObjData(session_proxy));
-  }
-
-  if (candidate_done_signal_received_) {
-    worker_thread_->Post(
-        this, MSG_SEND_ALLOCATION_DONE, new ProxyObjData(session_proxy));
-  }
-}
-
-void PortAllocatorSessionMuxer::OnCandidatesAllocationDone(
-    PortAllocatorSession* session) {
-  candidate_done_signal_received_ = true;
-}
-
-void PortAllocatorSessionMuxer::OnPortReady(PortAllocatorSession* session,
-                                            PortInterface* port) {
-  ASSERT(session == session_.get());
-  ports_.push_back(port);
-  port->SignalDestroyed.connect(
-      this, &PortAllocatorSessionMuxer::OnPortDestroyed);
-}
-
-void PortAllocatorSessionMuxer::OnPortDestroyed(PortInterface* port) {
-  std::vector<PortInterface*>::iterator it =
-      std::find(ports_.begin(), ports_.end(), port);
-  if (it != ports_.end())
-    ports_.erase(it);
-}
-
-void PortAllocatorSessionMuxer::OnSessionProxyDestroyed(
-    PortAllocatorSession* proxy) {
-
-  std::vector<PortAllocatorSessionProxy*>::iterator it =
-      std::find(session_proxies_.begin(), session_proxies_.end(), proxy);
-  if (it != session_proxies_.end()) {
-    session_proxies_.erase(it);
-  }
-
-  if (session_proxies_.empty()) {
-    // Destroy PortAllocatorSession and its associated muxer object if all
-    // proxies belonging to this session are already destroyed.
-    delete this;
-  }
-}
-
-void PortAllocatorSessionMuxer::OnMessage(rtc::Message *pmsg) {
-  ProxyObjData* proxy = static_cast<ProxyObjData*>(pmsg->pdata);
-  switch (pmsg->message_id) {
-    case MSG_SEND_ALLOCATION_DONE:
-      SendAllocationDone_w(proxy->data());
-      delete proxy;
-      break;
-    case MSG_SEND_ALLOCATED_PORTS:
-      SendAllocatedPorts_w(proxy->data());
-      delete proxy;
-      break;
-    default:
-      ASSERT(false);
-      break;
-  }
-}
-
-void PortAllocatorSessionMuxer::SendAllocationDone_w(
-    PortAllocatorSessionProxy* proxy) {
-  std::vector<PortAllocatorSessionProxy*>::iterator iter =
-      std::find(session_proxies_.begin(), session_proxies_.end(), proxy);
-  if (iter != session_proxies_.end()) {
-    proxy->OnCandidatesAllocationDone(session_.get());
-  }
-}
-
-void PortAllocatorSessionMuxer::SendAllocatedPorts_w(
-    PortAllocatorSessionProxy* proxy) {
-  std::vector<PortAllocatorSessionProxy*>::iterator iter =
-      std::find(session_proxies_.begin(), session_proxies_.end(), proxy);
-  if (iter != session_proxies_.end()) {
-    for (size_t i = 0; i < ports_.size(); ++i) {
-      PortInterface* port = ports_[i];
-      proxy->OnPortReady(session_.get(), port);
-      // If port already has candidates, send this to the clients of proxy
-      // session. This can happen if proxy is created later than the actual
-      // implementation.
-      if (!port->Candidates().empty()) {
-        proxy->OnCandidatesReady(session_.get(), port->Candidates());
-      }
-    }
-  }
-}
-
-PortAllocatorSessionProxy::~PortAllocatorSessionProxy() {
-  std::map<PortInterface*, PortProxy*>::iterator it;
-  for (it = proxy_ports_.begin(); it != proxy_ports_.end(); it++)
-    delete it->second;
-
-  SignalDestroyed(this);
-}
-
-void PortAllocatorSessionProxy::set_impl(
-    PortAllocatorSession* session) {
-  impl_ = session;
-
-  impl_->SignalCandidatesReady.connect(
-      this, &PortAllocatorSessionProxy::OnCandidatesReady);
-  impl_->SignalPortReady.connect(
-      this, &PortAllocatorSessionProxy::OnPortReady);
-  impl_->SignalCandidatesAllocationDone.connect(
-      this, &PortAllocatorSessionProxy::OnCandidatesAllocationDone);
-}
-
-void PortAllocatorSessionProxy::StartGettingPorts() {
-  ASSERT(impl_ != NULL);
-  // Since all proxies share a common PortAllocatorSession, this check will
-  // prohibit sending multiple STUN ping messages to the stun server, which
-  // is a problem on Chrome. GetInitialPorts() and StartGetAllPorts() called
-  // from the worker thread and are called together from TransportChannel,
-  // checking for IsGettingAllPorts() for GetInitialPorts() will not be a
-  // problem.
-  if (!impl_->IsGettingPorts()) {
-    impl_->StartGettingPorts();
-  }
-}
-
-void PortAllocatorSessionProxy::StopGettingPorts() {
-  ASSERT(impl_ != NULL);
-  if (impl_->IsGettingPorts()) {
-    impl_->StopGettingPorts();
-  }
-}
-
-bool PortAllocatorSessionProxy::IsGettingPorts() {
-  ASSERT(impl_ != NULL);
-  return impl_->IsGettingPorts();
-}
-
-void PortAllocatorSessionProxy::OnPortReady(PortAllocatorSession* session,
-                                            PortInterface* port) {
-  ASSERT(session == impl_);
-
-  PortProxy* proxy_port = new PortProxy();
-  proxy_port->set_impl(port);
-  proxy_ports_[port] = proxy_port;
-  SignalPortReady(this, proxy_port);
-}
-
-void PortAllocatorSessionProxy::OnCandidatesReady(
-    PortAllocatorSession* session,
-    const std::vector<Candidate>& candidates) {
-  ASSERT(session == impl_);
-
-  // Since all proxy sessions share a common PortAllocatorSession,
-  // all Candidates will have name associated with the common PAS.
-  // Change Candidate name with the PortAllocatorSessionProxy name.
-  std::vector<Candidate> our_candidates;
-  for (size_t i = 0; i < candidates.size(); ++i) {
-    Candidate new_local_candidate = candidates[i];
-    new_local_candidate.set_component(component_);
-    our_candidates.push_back(new_local_candidate);
-  }
-  SignalCandidatesReady(this, our_candidates);
-}
-
-void PortAllocatorSessionProxy::OnCandidatesAllocationDone(
-    PortAllocatorSession* session) {
-  ASSERT(session == impl_);
-  SignalCandidatesAllocationDone(this);
-}
-
-}  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy.h b/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy.h
deleted file mode 100644
index 94ae19d9c4..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
-#define WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
-
-#include <string>
-
-#include "webrtc/p2p/base/candidate.h"
-#include "webrtc/p2p/base/portallocator.h"
-
-namespace cricket {
-class PortAllocator;
-class PortAllocatorSessionProxy;
-class PortProxy;
-
-// This class maintains the list of cricket::Port* objects. Ports will be
-// deleted upon receiving SignalDestroyed signal. This class is used when
-// PORTALLOCATOR_ENABLE_BUNDLE flag is set.
-
-class PortAllocatorSessionMuxer : public rtc::MessageHandler,
-                                  public sigslot::has_slots<> {
- public:
-  explicit PortAllocatorSessionMuxer(PortAllocatorSession* session);
-  virtual ~PortAllocatorSessionMuxer();
-
-  void RegisterSessionProxy(PortAllocatorSessionProxy* session_proxy);
-
-  void OnPortReady(PortAllocatorSession* session, PortInterface* port);
-  void OnPortDestroyed(PortInterface* port);
-  void OnCandidatesAllocationDone(PortAllocatorSession* session);
-
-  const std::vector<PortInterface*>& ports() { return ports_; }
-
-  sigslot::signal1<PortAllocatorSessionMuxer*> SignalDestroyed;
-
- private:
-  virtual void OnMessage(rtc::Message *pmsg);
-  void OnSessionProxyDestroyed(PortAllocatorSession* proxy);
-  void SendAllocationDone_w(PortAllocatorSessionProxy* proxy);
-  void SendAllocatedPorts_w(PortAllocatorSessionProxy* proxy);
-
-  // Port will be deleted when SignalDestroyed received, otherwise delete
-  // happens when PortAllocatorSession dtor is called.
-  rtc::Thread* worker_thread_;
-  std::vector<PortInterface*> ports_;
-  rtc::scoped_ptr<PortAllocatorSession> session_;
-  std::vector<PortAllocatorSessionProxy*> session_proxies_;
-  bool candidate_done_signal_received_;
-};
-
-class PortAllocatorSessionProxy : public PortAllocatorSession {
- public:
-  PortAllocatorSessionProxy(const std::string& content_name,
-                            int component,
-                            uint32 flags)
-        // Use empty string as the ufrag and pwd because the proxy always uses
-        // the ufrag and pwd from the underlying implementation.
-      : PortAllocatorSession(content_name, component, "", "", flags),
-        impl_(NULL) {
-  }
-
-  virtual ~PortAllocatorSessionProxy();
-
-  PortAllocatorSession* impl() { return impl_; }
-  void set_impl(PortAllocatorSession* session);
-
-  // Forwards call to the actual PortAllocatorSession.
-  virtual void StartGettingPorts();
-  virtual void StopGettingPorts();
-  virtual bool IsGettingPorts();
-
-  virtual void set_generation(uint32 generation) {
-    ASSERT(impl_ != NULL);
-    impl_->set_generation(generation);
-  }
-
-  virtual uint32 generation() {
-    ASSERT(impl_ != NULL);
-    return impl_->generation();
-  }
-
- private:
-  void OnPortReady(PortAllocatorSession* session, PortInterface* port);
-  void OnCandidatesReady(PortAllocatorSession* session,
-                         const std::vector<Candidate>& candidates);
-  void OnPortDestroyed(PortInterface* port);
-  void OnCandidatesAllocationDone(PortAllocatorSession* session);
-
-  // This is the actual PortAllocatorSession, owned by PortAllocator.
-  PortAllocatorSession* impl_;
-  std::map<PortInterface*, PortProxy*> proxy_ports_;
-
-  friend class PortAllocatorSessionMuxer;
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy_unittest.cc
deleted file mode 100644
index 61a9e9896b..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/portallocatorsessionproxy_unittest.cc
+++ /dev/null
@@ -1,146 +0,0 @@
-/*
- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <vector>
-
-#include "webrtc/p2p/base/basicpacketsocketfactory.h"
-#include "webrtc/p2p/base/portallocatorsessionproxy.h"
-#include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
-#include "webrtc/base/fakenetwork.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/thread.h"
-
-using cricket::Candidate;
-using cricket::PortAllocatorSession;
-using cricket::PortAllocatorSessionMuxer;
-using cricket::PortAllocatorSessionProxy;
-
-// Based on ICE_UFRAG_LENGTH
-static const char kIceUfrag0[] = "TESTICEUFRAG0000";
-// Based on ICE_PWD_LENGTH
-static const char kIcePwd0[] = "TESTICEPWD00000000000000";
-
-class TestSessionChannel : public sigslot::has_slots<> {
- public:
-  explicit TestSessionChannel(PortAllocatorSessionProxy* proxy)
-      : proxy_session_(proxy),
-        candidates_count_(0),
-        allocation_complete_(false),
-        ports_count_(0) {
-    proxy_session_->SignalCandidatesAllocationDone.connect(
-        this, &TestSessionChannel::OnCandidatesAllocationDone);
-    proxy_session_->SignalCandidatesReady.connect(
-        this, &TestSessionChannel::OnCandidatesReady);
-    proxy_session_->SignalPortReady.connect(
-        this, &TestSessionChannel::OnPortReady);
-  }
-  virtual ~TestSessionChannel() {
-    delete proxy_session_;
-  }
-  void OnCandidatesReady(PortAllocatorSession* session,
-                         const std::vector<Candidate>& candidates) {
-    EXPECT_EQ(proxy_session_, session);
-    candidates_count_ += static_cast<int>(candidates.size());
-  }
-  void OnCandidatesAllocationDone(PortAllocatorSession* session) {
-    EXPECT_EQ(proxy_session_, session);
-    allocation_complete_ = true;
-  }
-  void OnPortReady(PortAllocatorSession* session,
-                   cricket::PortInterface* port) {
-    EXPECT_EQ(proxy_session_, session);
-    ++ports_count_;
-  }
-  int candidates_count() { return candidates_count_; }
-  bool allocation_complete() { return allocation_complete_; }
-  int ports_count() { return ports_count_; }
-
-  void StartGettingPorts() {
-    proxy_session_->StartGettingPorts();
-  }
-
-  void StopGettingPorts() {
-    proxy_session_->StopGettingPorts();
-  }
-
-  bool IsGettingPorts() {
-    return proxy_session_->IsGettingPorts();
-  }
-
- private:
-  PortAllocatorSessionProxy* proxy_session_;
-  int candidates_count_;
-  bool allocation_complete_;
-  int ports_count_;
-};
-
-class PortAllocatorSessionProxyTest : public testing::Test {
- public:
-  PortAllocatorSessionProxyTest()
-      : socket_factory_(rtc::Thread::Current()),
-        allocator_(rtc::Thread::Current(), NULL),
-        session_(new cricket::FakePortAllocatorSession(
-                     rtc::Thread::Current(), &socket_factory_,
-                     "test content", 1,
-                     kIceUfrag0, kIcePwd0)),
-        session_muxer_(new PortAllocatorSessionMuxer(session_)) {
-  }
-  virtual ~PortAllocatorSessionProxyTest() {}
-  void RegisterSessionProxy(PortAllocatorSessionProxy* proxy) {
-    session_muxer_->RegisterSessionProxy(proxy);
-  }
-
-  TestSessionChannel* CreateChannel() {
-    PortAllocatorSessionProxy* proxy =
-        new PortAllocatorSessionProxy("test content", 1, 0);
-    TestSessionChannel* channel = new TestSessionChannel(proxy);
-    session_muxer_->RegisterSessionProxy(proxy);
-    channel->StartGettingPorts();
-    return channel;
-  }
-
- protected:
-  rtc::BasicPacketSocketFactory socket_factory_;
-  cricket::FakePortAllocator allocator_;
-  cricket::FakePortAllocatorSession* session_;
-  // Muxer object will be delete itself after all registered session proxies
-  // are deleted.
-  PortAllocatorSessionMuxer* session_muxer_;
-};
-
-TEST_F(PortAllocatorSessionProxyTest, TestBasic) {
-  TestSessionChannel* channel = CreateChannel();
-  EXPECT_EQ_WAIT(1, channel->candidates_count(), 1000);
-  EXPECT_EQ(1, channel->ports_count());
-  EXPECT_TRUE(channel->allocation_complete());
-  delete channel;
-}
-
-TEST_F(PortAllocatorSessionProxyTest, TestLateBinding) {
-  TestSessionChannel* channel1 = CreateChannel();
-  EXPECT_EQ_WAIT(1, channel1->candidates_count(), 1000);
-  EXPECT_EQ(1, channel1->ports_count());
-  EXPECT_TRUE(channel1->allocation_complete());
-  EXPECT_EQ(1, session_->port_config_count());
-  // Creating another PortAllocatorSessionProxy and it also should receive
-  // already happened events.
-  PortAllocatorSessionProxy* proxy =
-      new PortAllocatorSessionProxy("test content", 2, 0);
-  TestSessionChannel* channel2 = new TestSessionChannel(proxy);
-  session_muxer_->RegisterSessionProxy(proxy);
-  EXPECT_TRUE(channel2->IsGettingPorts());
-  EXPECT_EQ_WAIT(1, channel2->candidates_count(), 1000);
-  EXPECT_EQ(1, channel2->ports_count());
-  EXPECT_TRUE_WAIT(channel2->allocation_complete(), 1000);
-  EXPECT_EQ(1, session_->port_config_count());
-  delete channel1;
-  delete channel2;
-}
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portinterface.h b/media/webrtc/trunk/webrtc/p2p/base/portinterface.h
index ee6835ebf3..e83879f3b7 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/portinterface.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/portinterface.h
@@ -14,6 +14,7 @@
 #include <string>
 
 #include "webrtc/p2p/base/transport.h"
+#include "webrtc/base/asyncpacketsocket.h"
 #include "webrtc/base/socketaddress.h"
 
 namespace rtc {
@@ -43,18 +44,17 @@ class PortInterface {
   virtual const std::string& Type() const = 0;
   virtual rtc::Network* Network() const = 0;
 
-  virtual void SetIceProtocolType(IceProtocolType protocol) = 0;
-  virtual IceProtocolType IceProtocol() const = 0;
-
   // Methods to set/get ICE role and tiebreaker values.
   virtual void SetIceRole(IceRole role) = 0;
   virtual IceRole GetIceRole() const = 0;
 
-  virtual void SetIceTiebreaker(uint64 tiebreaker) = 0;
-  virtual uint64 IceTiebreaker() const = 0;
+  virtual void SetIceTiebreaker(uint64_t tiebreaker) = 0;
+  virtual uint64_t IceTiebreaker() const = 0;
 
   virtual bool SharedSocket() const = 0;
 
+  virtual bool SupportsProtocol(const std::string& protocol) const = 0;
+
   // PrepareAddress will attempt to get an address for this port that other
   // clients can send to.  It may take some time before the address is ready.
   // Once it is ready, we will send SignalAddressReady.  If errors are
@@ -115,6 +115,9 @@ class PortInterface {
   sigslot::signal4<PortInterface*, const char*, size_t,
                    const rtc::SocketAddress&> SignalReadPacket;
 
+  // Emitted each time a packet is sent on this port.
+  sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
+
   virtual std::string ToString() const = 0;
 
  protected:
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portproxy.cc b/media/webrtc/trunk/webrtc/p2p/base/portproxy.cc
deleted file mode 100644
index e28af279fd..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/portproxy.cc
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/p2p/base/portproxy.h"
-
-namespace cricket {
-
-void PortProxy::set_impl(PortInterface* port) {
-  impl_ = port;
-  impl_->SignalUnknownAddress.connect(
-      this, &PortProxy::OnUnknownAddress);
-  impl_->SignalDestroyed.connect(this, &PortProxy::OnPortDestroyed);
-  impl_->SignalRoleConflict.connect(this, &PortProxy::OnRoleConflict);
-}
-
-const std::string& PortProxy::Type() const {
-  ASSERT(impl_ != NULL);
-  return impl_->Type();
-}
-
-rtc::Network* PortProxy::Network() const {
-  ASSERT(impl_ != NULL);
-  return impl_->Network();
-}
-
-void PortProxy::SetIceProtocolType(IceProtocolType protocol) {
-  ASSERT(impl_ != NULL);
-  impl_->SetIceProtocolType(protocol);
-}
-
-IceProtocolType PortProxy::IceProtocol() const {
-  ASSERT(impl_ != NULL);
-  return impl_->IceProtocol();
-}
-
-// Methods to set/get ICE role and tiebreaker values.
-void PortProxy::SetIceRole(IceRole role) {
-  ASSERT(impl_ != NULL);
-  impl_->SetIceRole(role);
-}
-
-IceRole PortProxy::GetIceRole() const {
-  ASSERT(impl_ != NULL);
-  return impl_->GetIceRole();
-}
-
-void PortProxy::SetIceTiebreaker(uint64 tiebreaker) {
-  ASSERT(impl_ != NULL);
-  impl_->SetIceTiebreaker(tiebreaker);
-}
-
-uint64 PortProxy::IceTiebreaker() const {
-  ASSERT(impl_ != NULL);
-  return impl_->IceTiebreaker();
-}
-
-bool PortProxy::SharedSocket() const {
-  ASSERT(impl_ != NULL);
-  return impl_->SharedSocket();
-}
-
-void PortProxy::PrepareAddress() {
-  ASSERT(impl_ != NULL);
-  impl_->PrepareAddress();
-}
-
-Connection* PortProxy::CreateConnection(const Candidate& remote_candidate,
-                                        CandidateOrigin origin) {
-  ASSERT(impl_ != NULL);
-  return impl_->CreateConnection(remote_candidate, origin);
-}
-
-int PortProxy::SendTo(const void* data,
-                      size_t size,
-                      const rtc::SocketAddress& addr,
-                      const rtc::PacketOptions& options,
-                      bool payload) {
-  ASSERT(impl_ != NULL);
-  return impl_->SendTo(data, size, addr, options, payload);
-}
-
-int PortProxy::SetOption(rtc::Socket::Option opt,
-                         int value) {
-  ASSERT(impl_ != NULL);
-  return impl_->SetOption(opt, value);
-}
-
-int PortProxy::GetOption(rtc::Socket::Option opt,
-                         int* value) {
-  ASSERT(impl_ != NULL);
-  return impl_->GetOption(opt, value);
-}
-
-int PortProxy::GetError() {
-  ASSERT(impl_ != NULL);
-  return impl_->GetError();
-}
-
-const std::vector<Candidate>& PortProxy::Candidates() const {
-  ASSERT(impl_ != NULL);
-  return impl_->Candidates();
-}
-
-void PortProxy::SendBindingResponse(
-    StunMessage* request, const rtc::SocketAddress& addr) {
-  ASSERT(impl_ != NULL);
-  impl_->SendBindingResponse(request, addr);
-}
-
-Connection* PortProxy::GetConnection(
-    const rtc::SocketAddress& remote_addr) {
-  ASSERT(impl_ != NULL);
-  return impl_->GetConnection(remote_addr);
-}
-
-void PortProxy::SendBindingErrorResponse(
-    StunMessage* request, const rtc::SocketAddress& addr,
-    int error_code, const std::string& reason) {
-  ASSERT(impl_ != NULL);
-  impl_->SendBindingErrorResponse(request, addr, error_code, reason);
-}
-
-void PortProxy::EnablePortPackets() {
-  ASSERT(impl_ != NULL);
-  impl_->EnablePortPackets();
-}
-
-std::string PortProxy::ToString() const {
-  ASSERT(impl_ != NULL);
-  return impl_->ToString();
-}
-
-void PortProxy::OnUnknownAddress(
-    PortInterface *port,
-    const rtc::SocketAddress &addr,
-    ProtocolType proto,
-    IceMessage *stun_msg,
-    const std::string &remote_username,
-    bool port_muxed) {
-  ASSERT(port == impl_);
-  ASSERT(!port_muxed);
-  SignalUnknownAddress(this, addr, proto, stun_msg, remote_username, true);
-}
-
-void PortProxy::OnRoleConflict(PortInterface* port) {
-  ASSERT(port == impl_);
-  SignalRoleConflict(this);
-}
-
-void PortProxy::OnPortDestroyed(PortInterface* port) {
-  ASSERT(port == impl_);
-  // |port| will be destroyed in PortAllocatorSessionMuxer.
-  SignalDestroyed(this);
-}
-
-}  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/portproxy.h b/media/webrtc/trunk/webrtc/p2p/base/portproxy.h
deleted file mode 100644
index 79507fea8e..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/portproxy.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_BASE_PORTPROXY_H_
-#define WEBRTC_P2P_BASE_PORTPROXY_H_
-
-#include "webrtc/p2p/base/portinterface.h"
-#include "webrtc/base/sigslot.h"
-
-namespace rtc {
-class Network;
-}
-
-namespace cricket {
-
-class PortProxy : public PortInterface, public sigslot::has_slots<> {
- public:
-  PortProxy() {}
-  virtual ~PortProxy() {}
-
-  PortInterface* impl() { return impl_; }
-  void set_impl(PortInterface* port);
-
-  virtual const std::string& Type() const;
-  virtual rtc::Network* Network() const;
-
-  virtual void SetIceProtocolType(IceProtocolType protocol);
-  virtual IceProtocolType IceProtocol() const;
-
-  // Methods to set/get ICE role and tiebreaker values.
-  virtual void SetIceRole(IceRole role);
-  virtual IceRole GetIceRole() const;
-
-  virtual void SetIceTiebreaker(uint64 tiebreaker);
-  virtual uint64 IceTiebreaker() const;
-
-  virtual bool SharedSocket() const;
-
-  // Forwards call to the actual Port.
-  virtual void PrepareAddress();
-  virtual Connection* CreateConnection(const Candidate& remote_candidate,
-                                       CandidateOrigin origin);
-  virtual Connection* GetConnection(
-      const rtc::SocketAddress& remote_addr);
-
-  virtual int SendTo(const void* data, size_t size,
-                     const rtc::SocketAddress& addr,
-                     const rtc::PacketOptions& options,
-                     bool payload);
-  virtual int SetOption(rtc::Socket::Option opt, int value);
-  virtual int GetOption(rtc::Socket::Option opt, int* value);
-  virtual int GetError();
-
-  virtual const std::vector<Candidate>& Candidates() const;
-
-  virtual void SendBindingResponse(StunMessage* request,
-                                   const rtc::SocketAddress& addr);
-  virtual void SendBindingErrorResponse(
-        StunMessage* request, const rtc::SocketAddress& addr,
-        int error_code, const std::string& reason);
-
-  virtual void EnablePortPackets();
-  virtual std::string ToString() const;
-
- private:
-  void OnUnknownAddress(PortInterface *port,
-                        const rtc::SocketAddress &addr,
-                        ProtocolType proto,
-                        IceMessage *stun_msg,
-                        const std::string &remote_username,
-                        bool port_muxed);
-  void OnRoleConflict(PortInterface* port);
-  void OnPortDestroyed(PortInterface* port);
-
-  PortInterface* impl_;
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_P2P_BASE_PORTPROXY_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.cc b/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.cc
index a54127bbb4..6281315dc1 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.cc
@@ -16,6 +16,7 @@
 #include <algorithm>
 #include <set>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/bytebuffer.h"
 #include "webrtc/base/byteorder.h"
@@ -39,40 +40,40 @@ namespace cricket {
 //////////////////////////////////////////////////////////////////////
 
 // Standard MTUs
-const uint16 PACKET_MAXIMUMS[] = {
-  65535,    // Theoretical maximum, Hyperchannel
-  32000,    // Nothing
-  17914,    // 16Mb IBM Token Ring
-  8166,   // IEEE 802.4
-  //4464,   // IEEE 802.5 (4Mb max)
-  4352,   // FDDI
-  //2048,   // Wideband Network
-  2002,   // IEEE 802.5 (4Mb recommended)
-  //1536,   // Expermental Ethernet Networks
-  //1500,   // Ethernet, Point-to-Point (default)
-  1492,   // IEEE 802.3
-  1006,   // SLIP, ARPANET
-  //576,    // X.25 Networks
-  //544,    // DEC IP Portal
-  //512,    // NETBIOS
-  508,    // IEEE 802/Source-Rt Bridge, ARCNET
-  296,    // Point-to-Point (low delay)
-  //68,     // Official minimum
-  0,      // End of list marker
+const uint16_t PACKET_MAXIMUMS[] = {
+    65535,  // Theoretical maximum, Hyperchannel
+    32000,  // Nothing
+    17914,  // 16Mb IBM Token Ring
+    8166,   // IEEE 802.4
+    // 4464,   // IEEE 802.5 (4Mb max)
+    4352,  // FDDI
+    // 2048,   // Wideband Network
+    2002,  // IEEE 802.5 (4Mb recommended)
+    // 1536,   // Expermental Ethernet Networks
+    // 1500,   // Ethernet, Point-to-Point (default)
+    1492,  // IEEE 802.3
+    1006,  // SLIP, ARPANET
+    // 576,    // X.25 Networks
+    // 544,    // DEC IP Portal
+    // 512,    // NETBIOS
+    508,  // IEEE 802/Source-Rt Bridge, ARCNET
+    296,  // Point-to-Point (low delay)
+    // 68,     // Official minimum
+    0,  // End of list marker
 };
 
-const uint32 MAX_PACKET = 65535;
+const uint32_t MAX_PACKET = 65535;
 // Note: we removed lowest level because packet overhead was larger!
-const uint32 MIN_PACKET = 296;
+const uint32_t MIN_PACKET = 296;
 
-const uint32 IP_HEADER_SIZE = 20; // (+ up to 40 bytes of options?)
-const uint32 UDP_HEADER_SIZE = 8;
+const uint32_t IP_HEADER_SIZE = 20;  // (+ up to 40 bytes of options?)
+const uint32_t UDP_HEADER_SIZE = 8;
 // TODO: Make JINGLE_HEADER_SIZE transparent to this code?
-const uint32 JINGLE_HEADER_SIZE = 64; // when relay framing is in use
+const uint32_t JINGLE_HEADER_SIZE = 64;  // when relay framing is in use
 
 // Default size for receive and send buffer.
-const uint32 DEFAULT_RCV_BUF_SIZE = 60 * 1024;
-const uint32 DEFAULT_SND_BUF_SIZE = 90 * 1024;
+const uint32_t DEFAULT_RCV_BUF_SIZE = 60 * 1024;
+const uint32_t DEFAULT_SND_BUF_SIZE = 90 * 1024;
 
 //////////////////////////////////////////////////////////////////////
 // Global Constants and Functions
@@ -102,55 +103,59 @@ const uint32 DEFAULT_SND_BUF_SIZE = 90 * 1024;
 
 #define PSEUDO_KEEPALIVE 0
 
-const uint32 HEADER_SIZE = 24;
-const uint32 PACKET_OVERHEAD = HEADER_SIZE + UDP_HEADER_SIZE + IP_HEADER_SIZE + JINGLE_HEADER_SIZE;
+const uint32_t HEADER_SIZE = 24;
+const uint32_t PACKET_OVERHEAD =
+    HEADER_SIZE + UDP_HEADER_SIZE + IP_HEADER_SIZE + JINGLE_HEADER_SIZE;
 
-const uint32 MIN_RTO   =   250; // 250 ms (RFC1122, Sec 4.2.3.1 "fractions of a second")
-const uint32 DEF_RTO   =  3000; // 3 seconds (RFC1122, Sec 4.2.3.1)
-const uint32 MAX_RTO   = 60000; // 60 seconds
-const uint32 DEF_ACK_DELAY = 100; // 100 milliseconds
+const uint32_t MIN_RTO =
+    250;  // 250 ms (RFC1122, Sec 4.2.3.1 "fractions of a second")
+const uint32_t DEF_RTO = 3000;       // 3 seconds (RFC1122, Sec 4.2.3.1)
+const uint32_t MAX_RTO = 60000;      // 60 seconds
+const uint32_t DEF_ACK_DELAY = 100;  // 100 milliseconds
 
-const uint8 FLAG_CTL = 0x02;
-const uint8 FLAG_RST = 0x04;
+const uint8_t FLAG_CTL = 0x02;
+const uint8_t FLAG_RST = 0x04;
 
-const uint8 CTL_CONNECT = 0;
+const uint8_t CTL_CONNECT = 0;
 
 // TCP options.
-const uint8 TCP_OPT_EOL = 0;  // End of list.
-const uint8 TCP_OPT_NOOP = 1;  // No-op.
-const uint8 TCP_OPT_MSS = 2;  // Maximum segment size.
-const uint8 TCP_OPT_WND_SCALE = 3;  // Window scale factor.
+const uint8_t TCP_OPT_EOL = 0;        // End of list.
+const uint8_t TCP_OPT_NOOP = 1;       // No-op.
+const uint8_t TCP_OPT_MSS = 2;        // Maximum segment size.
+const uint8_t TCP_OPT_WND_SCALE = 3;  // Window scale factor.
 
 const long DEFAULT_TIMEOUT = 4000; // If there are no pending clocks, wake up every 4 seconds
 const long CLOSED_TIMEOUT = 60 * 1000; // If the connection is closed, once per minute
 
 #if PSEUDO_KEEPALIVE
 // !?! Rethink these times
-const uint32 IDLE_PING = 20 * 1000; // 20 seconds (note: WinXP SP2 firewall udp timeout is 90 seconds)
-const uint32 IDLE_TIMEOUT = 90 * 1000; // 90 seconds;
+const uint32_t IDLE_PING =
+    20 *
+    1000;  // 20 seconds (note: WinXP SP2 firewall udp timeout is 90 seconds)
+const uint32_t IDLE_TIMEOUT = 90 * 1000;  // 90 seconds;
 #endif // PSEUDO_KEEPALIVE
 
 //////////////////////////////////////////////////////////////////////
 // Helper Functions
 //////////////////////////////////////////////////////////////////////
 
-inline void long_to_bytes(uint32 val, void* buf) {
-  *static_cast<uint32*>(buf) = rtc::HostToNetwork32(val);
+inline void long_to_bytes(uint32_t val, void* buf) {
+  *static_cast<uint32_t*>(buf) = rtc::HostToNetwork32(val);
 }
 
-inline void short_to_bytes(uint16 val, void* buf) {
-  *static_cast<uint16*>(buf) = rtc::HostToNetwork16(val);
+inline void short_to_bytes(uint16_t val, void* buf) {
+  *static_cast<uint16_t*>(buf) = rtc::HostToNetwork16(val);
 }
 
-inline uint32 bytes_to_long(const void* buf) {
-  return rtc::NetworkToHost32(*static_cast<const uint32*>(buf));
+inline uint32_t bytes_to_long(const void* buf) {
+  return rtc::NetworkToHost32(*static_cast<const uint32_t*>(buf));
 }
 
-inline uint16 bytes_to_short(const void* buf) {
-  return rtc::NetworkToHost16(*static_cast<const uint16*>(buf));
+inline uint16_t bytes_to_short(const void* buf) {
+  return rtc::NetworkToHost16(*static_cast<const uint16_t*>(buf));
 }
 
-uint32 bound(uint32 lower, uint32 middle, uint32 upper) {
+uint32_t bound(uint32_t lower, uint32_t middle, uint32_t upper) {
   return std::min(std::max(lower, middle), upper);
 }
 
@@ -183,7 +188,7 @@ void ReportStats() {
   char buffer[256];
   size_t len = 0;
   for (int i = 0; i < S_NUM_STATS; ++i) {
-    len += rtc::sprintfn(buffer, ARRAY_SIZE(buffer), "%s%s:%d",
+    len += rtc::sprintfn(buffer, arraysize(buffer), "%s%s:%d",
                                (i == 0) ? "" : ",", STAT_NAMES[i], g_stats[i]);
     g_stats[i] = 0;
   }
@@ -196,7 +201,7 @@ void ReportStats() {
 // PseudoTcp
 //////////////////////////////////////////////////////////////////////
 
-uint32 PseudoTcp::Now() {
+uint32_t PseudoTcp::Now() {
 #if 0  // Use this to synchronize timers with logging timestamps (easier debug)
   return rtc::TimeSince(StartTime());
 #else
@@ -204,7 +209,7 @@ uint32 PseudoTcp::Now() {
 #endif
 }
 
-PseudoTcp::PseudoTcp(IPseudoTcpNotify* notify, uint32 conv)
+PseudoTcp::PseudoTcp(IPseudoTcpNotify* notify, uint32_t conv)
     : m_notify(notify),
       m_shutdown(SD_NONE),
       m_error(0),
@@ -212,11 +217,10 @@ PseudoTcp::PseudoTcp(IPseudoTcpNotify* notify, uint32 conv)
       m_rbuf(m_rbuf_len),
       m_sbuf_len(DEFAULT_SND_BUF_SIZE),
       m_sbuf(m_sbuf_len) {
-
   // Sanity check on buffer sizes (needed for OnTcpWriteable notification logic)
   ASSERT(m_rbuf_len + MIN_PACKET < m_sbuf_len);
 
-  uint32 now = Now();
+  uint32_t now = Now();
 
   m_state = TCP_LISTEN;
   m_conv = conv;
@@ -273,14 +277,14 @@ int PseudoTcp::Connect() {
   return 0;
 }
 
-void PseudoTcp::NotifyMTU(uint16 mtu) {
+void PseudoTcp::NotifyMTU(uint16_t mtu) {
   m_mtu_advise = mtu;
   if (m_state == TCP_ESTABLISHED) {
     adjustMTU();
   }
 }
 
-void PseudoTcp::NotifyClock(uint32 now) {
+void PseudoTcp::NotifyClock(uint32_t now) {
   if (m_state == TCP_CLOSED)
     return;
 
@@ -303,13 +307,13 @@ void PseudoTcp::NotifyClock(uint32 now) {
         return;
       }
 
-      uint32 nInFlight = m_snd_nxt - m_snd_una;
+      uint32_t nInFlight = m_snd_nxt - m_snd_una;
       m_ssthresh = std::max(nInFlight / 2, 2 * m_mss);
       //LOG(LS_INFO) << "m_ssthresh: " << m_ssthresh << "  nInFlight: " << nInFlight << "  m_mss: " << m_mss;
       m_cwnd = m_mss;
 
       // Back off retransmit timer.  Note: the limit is lower when connecting.
-      uint32 rto_limit = (m_state < TCP_ESTABLISHED) ? DEF_RTO : MAX_RTO;
+      uint32_t rto_limit = (m_state < TCP_ESTABLISHED) ? DEF_RTO : MAX_RTO;
       m_rx_rto = std::min(rto_limit, m_rx_rto * 2);
       m_rto_base = now;
     }
@@ -355,10 +359,10 @@ bool PseudoTcp::NotifyPacket(const char* buffer, size_t len) {
     LOG_F(WARNING) << "packet too large";
     return false;
   }
-  return parse(reinterpret_cast<const uint8 *>(buffer), uint32(len));
+  return parse(reinterpret_cast<const uint8_t*>(buffer), uint32_t(len));
 }
 
-bool PseudoTcp::GetNextClock(uint32 now, long& timeout) {
+bool PseudoTcp::GetNextClock(uint32_t now, long& timeout) {
   return clock_check(now, timeout);
 }
 
@@ -391,21 +395,21 @@ void PseudoTcp::SetOption(Option opt, int value) {
   }
 }
 
-uint32 PseudoTcp::GetCongestionWindow() const {
+uint32_t PseudoTcp::GetCongestionWindow() const {
   return m_cwnd;
 }
 
-uint32 PseudoTcp::GetBytesInFlight() const {
+uint32_t PseudoTcp::GetBytesInFlight() const {
   return m_snd_nxt - m_snd_una;
 }
 
-uint32 PseudoTcp::GetBytesBufferedNotSent() const {
+uint32_t PseudoTcp::GetBytesBufferedNotSent() const {
   size_t buffered_bytes = 0;
   m_sbuf.GetBuffered(&buffered_bytes);
-  return static_cast<uint32>(m_snd_una + buffered_bytes - m_snd_nxt);
+  return static_cast<uint32_t>(m_snd_una + buffered_bytes - m_snd_nxt);
 }
 
-uint32 PseudoTcp::GetRoundTripTimeEstimateMs() const {
+uint32_t PseudoTcp::GetRoundTripTimeEstimateMs() const {
   return m_rx_srtt;
 }
 
@@ -433,11 +437,11 @@ int PseudoTcp::Recv(char* buffer, size_t len) {
   size_t available_space = 0;
   m_rbuf.GetWriteRemaining(&available_space);
 
-  if (uint32(available_space) - m_rcv_wnd >=
-      std::min<uint32>(m_rbuf_len / 2, m_mss)) {
+  if (uint32_t(available_space) - m_rcv_wnd >=
+      std::min<uint32_t>(m_rbuf_len / 2, m_mss)) {
     // TODO(jbeda): !?! Not sure about this was closed business
     bool bWasClosed = (m_rcv_wnd == 0);
-    m_rcv_wnd = static_cast<uint32>(available_space);
+    m_rcv_wnd = static_cast<uint32_t>(available_space);
 
     if (bWasClosed) {
       attemptSend(sfImmediateAck);
@@ -462,7 +466,7 @@ int PseudoTcp::Send(const char* buffer, size_t len) {
     return SOCKET_ERROR;
   }
 
-  int written = queue(buffer, uint32(len), false);
+  int written = queue(buffer, uint32_t(len), false);
   attemptSend();
   return written;
 }
@@ -480,13 +484,13 @@ int PseudoTcp::GetError() {
 // Internal Implementation
 //
 
-uint32 PseudoTcp::queue(const char* data, uint32 len, bool bCtrl) {
+uint32_t PseudoTcp::queue(const char* data, uint32_t len, bool bCtrl) {
   size_t available_space = 0;
   m_sbuf.GetWriteRemaining(&available_space);
 
-  if (len > static_cast<uint32>(available_space)) {
+  if (len > static_cast<uint32_t>(available_space)) {
     ASSERT(!bCtrl);
-    len = static_cast<uint32>(available_space);
+    len = static_cast<uint32_t>(available_space);
   }
 
   // We can concatenate data if the last segment is the same type
@@ -497,29 +501,31 @@ uint32 PseudoTcp::queue(const char* data, uint32 len, bool bCtrl) {
   } else {
     size_t snd_buffered = 0;
     m_sbuf.GetBuffered(&snd_buffered);
-    SSegment sseg(static_cast<uint32>(m_snd_una + snd_buffered), len, bCtrl);
+    SSegment sseg(static_cast<uint32_t>(m_snd_una + snd_buffered), len, bCtrl);
     m_slist.push_back(sseg);
   }
 
   size_t written = 0;
   m_sbuf.Write(data, len, &written, NULL);
-  return static_cast<uint32>(written);
+  return static_cast<uint32_t>(written);
 }
 
-IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32 seq, uint8 flags,
-                                                uint32 offset, uint32 len) {
+IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32_t seq,
+                                                uint8_t flags,
+                                                uint32_t offset,
+                                                uint32_t len) {
   ASSERT(HEADER_SIZE + len <= MAX_PACKET);
 
-  uint32 now = Now();
+  uint32_t now = Now();
 
-  rtc::scoped_ptr<uint8[]> buffer(new uint8[MAX_PACKET]);
+  rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[MAX_PACKET]);
   long_to_bytes(m_conv, buffer.get());
   long_to_bytes(seq, buffer.get() + 4);
   long_to_bytes(m_rcv_nxt, buffer.get() + 8);
   buffer[12] = 0;
   buffer[13] = flags;
-  short_to_bytes(
-      static_cast<uint16>(m_rcv_wnd >> m_rwnd_scale), buffer.get() + 14);
+  short_to_bytes(static_cast<uint16_t>(m_rcv_wnd >> m_rwnd_scale),
+                 buffer.get() + 14);
 
   // Timestamp computations
   long_to_bytes(now, buffer.get() + 16);
@@ -532,7 +538,7 @@ IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32 seq, uint8 flags,
         buffer.get() + HEADER_SIZE, len, offset, &bytes_read);
     RTC_UNUSED(result);
     ASSERT(result == rtc::SR_SUCCESS);
-    ASSERT(static_cast<uint32>(bytes_read) == len);
+    ASSERT(static_cast<uint32_t>(bytes_read) == len);
   }
 
 #if _DEBUGMSG >= _DBG_VERBOSE
@@ -564,7 +570,7 @@ IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32 seq, uint8 flags,
   return IPseudoTcpNotify::WR_SUCCESS;
 }
 
-bool PseudoTcp::parse(const uint8* buffer, uint32 size) {
+bool PseudoTcp::parse(const uint8_t* buffer, uint32_t size) {
   if (size < 12)
     return false;
 
@@ -595,7 +601,7 @@ bool PseudoTcp::parse(const uint8* buffer, uint32 size) {
   return process(seg);
 }
 
-bool PseudoTcp::clock_check(uint32 now, long& nTimeout) {
+bool PseudoTcp::clock_check(uint32_t now, long& nTimeout) {
   if (m_shutdown == SD_FORCEFUL)
     return false;
 
@@ -616,19 +622,19 @@ bool PseudoTcp::clock_check(uint32 now, long& nTimeout) {
 
   if (m_t_ack) {
     nTimeout =
-        std::min<int32>(nTimeout, rtc::TimeDiff(m_t_ack + m_ack_delay, now));
+        std::min<int32_t>(nTimeout, rtc::TimeDiff(m_t_ack + m_ack_delay, now));
   }
   if (m_rto_base) {
     nTimeout =
-        std::min<int32>(nTimeout, rtc::TimeDiff(m_rto_base + m_rx_rto, now));
+        std::min<int32_t>(nTimeout, rtc::TimeDiff(m_rto_base + m_rx_rto, now));
   }
   if (m_snd_wnd == 0) {
     nTimeout =
-        std::min<int32>(nTimeout, rtc::TimeDiff(m_lastsend + m_rx_rto, now));
+        std::min<int32_t>(nTimeout, rtc::TimeDiff(m_lastsend + m_rx_rto, now));
   }
 #if PSEUDO_KEEPALIVE
   if (m_state == TCP_ESTABLISHED) {
-    nTimeout = std::min<int32>(
+    nTimeout = std::min<int32_t>(
         nTimeout, rtc::TimeDiff(m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3 / 2
                                                              : IDLE_PING),
                                 now));
@@ -647,7 +653,7 @@ bool PseudoTcp::process(Segment& seg) {
     return false;
   }
 
-  uint32 now = Now();
+  uint32_t now = Now();
   m_lasttraffic = m_lastrecv = now;
   m_bOutgoing = false;
 
@@ -704,20 +710,22 @@ bool PseudoTcp::process(Segment& seg) {
   if ((seg.ack > m_snd_una) && (seg.ack <= m_snd_nxt)) {
     // Calculate round-trip time
     if (seg.tsecr) {
-      int32 rtt = rtc::TimeDiff(now, seg.tsecr);
+      int32_t rtt = rtc::TimeDiff(now, seg.tsecr);
       if (rtt >= 0) {
         if (m_rx_srtt == 0) {
           m_rx_srtt = rtt;
           m_rx_rttvar = rtt / 2;
         } else {
-          uint32 unsigned_rtt = static_cast<uint32>(rtt);
-          uint32 abs_err = unsigned_rtt > m_rx_srtt ? unsigned_rtt - m_rx_srtt
-                                                    : m_rx_srtt - unsigned_rtt;
+          uint32_t unsigned_rtt = static_cast<uint32_t>(rtt);
+          uint32_t abs_err = unsigned_rtt > m_rx_srtt
+                                 ? unsigned_rtt - m_rx_srtt
+                                 : m_rx_srtt - unsigned_rtt;
           m_rx_rttvar = (3 * m_rx_rttvar + abs_err) / 4;
           m_rx_srtt = (7 * m_rx_srtt + rtt) / 8;
         }
-        m_rx_rto = bound(
-            MIN_RTO, m_rx_srtt + std::max<uint32>(1, 4 * m_rx_rttvar), MAX_RTO);
+        m_rx_rto =
+            bound(MIN_RTO, m_rx_srtt + std::max<uint32_t>(1, 4 * m_rx_rttvar),
+                  MAX_RTO);
 #if _DEBUGMSG >= _DBG_VERBOSE
         LOG(LS_INFO) << "rtt: " << rtt
                      << "  srtt: " << m_rx_srtt
@@ -728,16 +736,16 @@ bool PseudoTcp::process(Segment& seg) {
       }
     }
 
-    m_snd_wnd = static_cast<uint32>(seg.wnd) << m_swnd_scale;
+    m_snd_wnd = static_cast<uint32_t>(seg.wnd) << m_swnd_scale;
 
-    uint32 nAcked = seg.ack - m_snd_una;
+    uint32_t nAcked = seg.ack - m_snd_una;
     m_snd_una = seg.ack;
 
     m_rto_base = (m_snd_una == m_snd_nxt) ? 0 : now;
 
     m_sbuf.ConsumeReadData(nAcked);
 
-    for (uint32 nFree = nAcked; nFree > 0; ) {
+    for (uint32_t nFree = nAcked; nFree > 0;) {
       ASSERT(!m_slist.empty());
       if (nFree < m_slist.front().len) {
         m_slist.front().len -= nFree;
@@ -753,7 +761,7 @@ bool PseudoTcp::process(Segment& seg) {
 
     if (m_dup_acks >= 3) {
       if (m_snd_una >= m_recover) { // NewReno
-        uint32 nInFlight = m_snd_nxt - m_snd_una;
+        uint32_t nInFlight = m_snd_nxt - m_snd_una;
         m_cwnd = std::min(m_ssthresh, nInFlight + m_mss);  // (Fast Retransmit)
 #if _DEBUGMSG >= _DBG_NORMAL
         LOG(LS_INFO) << "exit recovery";
@@ -775,12 +783,12 @@ bool PseudoTcp::process(Segment& seg) {
       if (m_cwnd < m_ssthresh) {
         m_cwnd += m_mss;
       } else {
-        m_cwnd += std::max<uint32>(1, m_mss * m_mss / m_cwnd);
+        m_cwnd += std::max<uint32_t>(1, m_mss * m_mss / m_cwnd);
       }
     }
   } else if (seg.ack == m_snd_una) {
     // !?! Note, tcp says don't do this... but otherwise how does a closed window become open?
-    m_snd_wnd = static_cast<uint32>(seg.wnd) << m_swnd_scale;
+    m_snd_wnd = static_cast<uint32_t>(seg.wnd) << m_swnd_scale;
 
     // Check duplicate acks
     if (seg.len > 0) {
@@ -797,7 +805,7 @@ bool PseudoTcp::process(Segment& seg) {
           return false;
         }
         m_recover = m_snd_nxt;
-        uint32 nInFlight = m_snd_nxt - m_snd_una;
+        uint32_t nInFlight = m_snd_nxt - m_snd_una;
         m_ssthresh = std::max(nInFlight / 2, 2 * m_mss);
         //LOG(LS_INFO) << "m_ssthresh: " << m_ssthresh << "  nInFlight: " << nInFlight << "  m_mss: " << m_mss;
         m_cwnd = m_ssthresh + 3 * m_mss;
@@ -823,10 +831,11 @@ bool PseudoTcp::process(Segment& seg) {
   // If we make room in the send queue, notify the user
   // The goal it to make sure we always have at least enough data to fill the
   // window.  We'd like to notify the app when we are halfway to that point.
-  const uint32 kIdealRefillSize = (m_sbuf_len + m_rbuf_len) / 2;
+  const uint32_t kIdealRefillSize = (m_sbuf_len + m_rbuf_len) / 2;
   size_t snd_buffered = 0;
   m_sbuf.GetBuffered(&snd_buffered);
-  if (m_bWriteEnable && static_cast<uint32>(snd_buffered) < kIdealRefillSize) {
+  if (m_bWriteEnable &&
+      static_cast<uint32_t>(snd_buffered) < kIdealRefillSize) {
     m_bWriteEnable = false;
     if (m_notify) {
       m_notify->OnTcpWriteable(this);
@@ -862,7 +871,7 @@ bool PseudoTcp::process(Segment& seg) {
 
   // Adjust the incoming segment to fit our receive buffer
   if (seg.seq < m_rcv_nxt) {
-    uint32 nAdjust = m_rcv_nxt - seg.seq;
+    uint32_t nAdjust = m_rcv_nxt - seg.seq;
     if (nAdjust < seg.len) {
       seg.seq += nAdjust;
       seg.data += nAdjust;
@@ -875,8 +884,10 @@ bool PseudoTcp::process(Segment& seg) {
   size_t available_space = 0;
   m_rbuf.GetWriteRemaining(&available_space);
 
-  if ((seg.seq + seg.len - m_rcv_nxt) > static_cast<uint32>(available_space)) {
-    uint32 nAdjust = seg.seq + seg.len - m_rcv_nxt - static_cast<uint32>(available_space);
+  if ((seg.seq + seg.len - m_rcv_nxt) >
+      static_cast<uint32_t>(available_space)) {
+    uint32_t nAdjust =
+        seg.seq + seg.len - m_rcv_nxt - static_cast<uint32_t>(available_space);
     if (nAdjust < seg.len) {
       seg.len -= nAdjust;
     } else {
@@ -893,7 +904,7 @@ bool PseudoTcp::process(Segment& seg) {
         m_rcv_nxt += seg.len;
       }
     } else {
-      uint32 nOffset = seg.seq - m_rcv_nxt;
+      uint32_t nOffset = seg.seq - m_rcv_nxt;
 
       rtc::StreamResult result = m_rbuf.WriteOffset(seg.data, seg.len,
                                                           nOffset, NULL);
@@ -910,7 +921,7 @@ bool PseudoTcp::process(Segment& seg) {
         while ((it != m_rlist.end()) && (it->seq <= m_rcv_nxt)) {
           if (it->seq + it->len > m_rcv_nxt) {
             sflags = sfImmediateAck; // (Fast Recovery)
-            uint32 nAdjust = (it->seq + it->len) - m_rcv_nxt;
+            uint32_t nAdjust = (it->seq + it->len) - m_rcv_nxt;
 #if _DEBUGMSG >= _DBG_NORMAL
             LOG(LS_INFO) << "Recovered " << nAdjust << " bytes (" << m_rcv_nxt << " -> " << m_rcv_nxt + nAdjust << ")";
 #endif // _DEBUGMSG
@@ -950,17 +961,17 @@ bool PseudoTcp::process(Segment& seg) {
   return true;
 }
 
-bool PseudoTcp::transmit(const SList::iterator& seg, uint32 now) {
+bool PseudoTcp::transmit(const SList::iterator& seg, uint32_t now) {
   if (seg->xmit >= ((m_state == TCP_ESTABLISHED) ? 15 : 30)) {
     LOG_F(LS_VERBOSE) << "too many retransmits";
     return false;
   }
 
-  uint32 nTransmit = std::min(seg->len, m_mss);
+  uint32_t nTransmit = std::min(seg->len, m_mss);
 
   while (true) {
-    uint32 seq = seg->seq;
-    uint8 flags = (seg->bCtrl ? FLAG_CTL : 0);
+    uint32_t seq = seg->seq;
+    uint8_t flags = (seg->bCtrl ? FLAG_CTL : 0);
     IPseudoTcpNotify::WriteResult wres = packet(seq,
                                                 flags,
                                                 seg->seq - m_snd_una,
@@ -1020,7 +1031,7 @@ bool PseudoTcp::transmit(const SList::iterator& seg, uint32 now) {
 }
 
 void PseudoTcp::attemptSend(SendFlags sflags) {
-  uint32 now = Now();
+  uint32_t now = Now();
 
   if (rtc::TimeDiff(now, m_lastsend) > static_cast<long>(m_rx_rto)) {
     m_cwnd = m_mss;
@@ -1032,18 +1043,18 @@ void PseudoTcp::attemptSend(SendFlags sflags) {
 #endif // _DEBUGMSG
 
   while (true) {
-    uint32 cwnd = m_cwnd;
+    uint32_t cwnd = m_cwnd;
     if ((m_dup_acks == 1) || (m_dup_acks == 2)) { // Limited Transmit
       cwnd += m_dup_acks * m_mss;
     }
-    uint32 nWindow = std::min(m_snd_wnd, cwnd);
-    uint32 nInFlight = m_snd_nxt - m_snd_una;
-    uint32 nUseable = (nInFlight < nWindow) ? (nWindow - nInFlight) : 0;
+    uint32_t nWindow = std::min(m_snd_wnd, cwnd);
+    uint32_t nInFlight = m_snd_nxt - m_snd_una;
+    uint32_t nUseable = (nInFlight < nWindow) ? (nWindow - nInFlight) : 0;
 
     size_t snd_buffered = 0;
     m_sbuf.GetBuffered(&snd_buffered);
-    uint32 nAvailable =
-        std::min(static_cast<uint32>(snd_buffered) - nInFlight, m_mss);
+    uint32_t nAvailable =
+        std::min(static_cast<uint32_t>(snd_buffered) - nInFlight, m_mss);
 
     if (nAvailable > nUseable) {
       if (nUseable * 4 < nWindow) {
@@ -1116,8 +1127,7 @@ void PseudoTcp::attemptSend(SendFlags sflags) {
   }
 }
 
-void
-PseudoTcp::closedown(uint32 err) {
+void PseudoTcp::closedown(uint32_t err) {
   LOG(LS_INFO) << "State: TCP_CLOSED";
   m_state = TCP_CLOSED;
   if (m_notify) {
@@ -1130,7 +1140,7 @@ void
 PseudoTcp::adjustMTU() {
   // Determine our current mss level, so that we can adjust appropriately later
   for (m_msslevel = 0; PACKET_MAXIMUMS[m_msslevel + 1] > 0; ++m_msslevel) {
-    if (static_cast<uint16>(PACKET_MAXIMUMS[m_msslevel]) <= m_mtu_advise) {
+    if (static_cast<uint16_t>(PACKET_MAXIMUMS[m_msslevel]) <= m_mtu_advise) {
       break;
     }
   }
@@ -1166,19 +1176,18 @@ PseudoTcp::queueConnectMessage() {
     buf.WriteUInt8(1);
     buf.WriteUInt8(m_rwnd_scale);
   }
-  m_snd_wnd = static_cast<uint32>(buf.Length());
-  queue(buf.Data(), static_cast<uint32>(buf.Length()), true);
+  m_snd_wnd = static_cast<uint32_t>(buf.Length());
+  queue(buf.Data(), static_cast<uint32_t>(buf.Length()), true);
 }
 
-void
-PseudoTcp::parseOptions(const char* data, uint32 len) {
-  std::set<uint8> options_specified;
+void PseudoTcp::parseOptions(const char* data, uint32_t len) {
+  std::set<uint8_t> options_specified;
 
   // See http://www.freesoft.org/CIE/Course/Section4/8.htm for
   // parsing the options list.
   rtc::ByteBuffer buf(data, len);
   while (buf.Length()) {
-    uint8 kind = TCP_OPT_EOL;
+    uint8_t kind = TCP_OPT_EOL;
     buf.ReadUInt8(&kind);
 
     if (kind == TCP_OPT_EOL) {
@@ -1192,7 +1201,7 @@ PseudoTcp::parseOptions(const char* data, uint32 len) {
     // Length of this option.
     ASSERT(len != 0);
     RTC_UNUSED(len);
-    uint8 opt_len = 0;
+    uint8_t opt_len = 0;
     buf.ReadUInt8(&opt_len);
 
     // Content of this option.
@@ -1218,8 +1227,7 @@ PseudoTcp::parseOptions(const char* data, uint32 len) {
   }
 }
 
-void
-PseudoTcp::applyOption(char kind, const char* data, uint32 len) {
+void PseudoTcp::applyOption(char kind, const char* data, uint32_t len) {
   if (kind == TCP_OPT_MSS) {
     LOG(LS_WARNING) << "Peer specified MSS option which is not supported.";
     // TODO: Implement.
@@ -1234,20 +1242,17 @@ PseudoTcp::applyOption(char kind, const char* data, uint32 len) {
   }
 }
 
-void
-PseudoTcp::applyWindowScaleOption(uint8 scale_factor) {
+void PseudoTcp::applyWindowScaleOption(uint8_t scale_factor) {
   m_swnd_scale = scale_factor;
 }
 
-void
-PseudoTcp::resizeSendBuffer(uint32 new_size) {
+void PseudoTcp::resizeSendBuffer(uint32_t new_size) {
   m_sbuf_len = new_size;
   m_sbuf.SetCapacity(new_size);
 }
 
-void
-PseudoTcp::resizeReceiveBuffer(uint32 new_size) {
-  uint8 scale_factor = 0;
+void PseudoTcp::resizeReceiveBuffer(uint32_t new_size) {
+  uint8_t scale_factor = 0;
 
   // Determine the scale factor such that the scaled window size can fit
   // in a 16-bit unsigned integer.
@@ -1272,7 +1277,7 @@ PseudoTcp::resizeReceiveBuffer(uint32 new_size) {
 
   size_t available_space = 0;
   m_rbuf.GetWriteRemaining(&available_space);
-  m_rcv_wnd = static_cast<uint32>(available_space);
+  m_rcv_wnd = static_cast<uint32_t>(available_space);
 }
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.h b/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.h
index b2cfcb79ef..6d402daa6f 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/pseudotcp.h
@@ -30,7 +30,7 @@ class IPseudoTcpNotify {
   virtual void OnTcpOpen(PseudoTcp* tcp) = 0;
   virtual void OnTcpReadable(PseudoTcp* tcp) = 0;
   virtual void OnTcpWriteable(PseudoTcp* tcp) = 0;
-  virtual void OnTcpClosed(PseudoTcp* tcp, uint32 error) = 0;
+  virtual void OnTcpClosed(PseudoTcp* tcp, uint32_t error) = 0;
 
   // Write the packet onto the network
   enum WriteResult { WR_SUCCESS, WR_TOO_LARGE, WR_FAIL };
@@ -47,9 +47,9 @@ class IPseudoTcpNotify {
 
 class PseudoTcp {
  public:
-  static uint32 Now();
+  static uint32_t Now();
 
-  PseudoTcp(IPseudoTcpNotify* notify, uint32 conv);
+  PseudoTcp(IPseudoTcpNotify* notify, uint32_t conv);
   virtual ~PseudoTcp();
 
   int Connect();
@@ -64,11 +64,11 @@ class PseudoTcp {
   TcpState State() const { return m_state; }
 
   // Call this when the PMTU changes.
-  void NotifyMTU(uint16 mtu);
+  void NotifyMTU(uint16_t mtu);
 
   // Call this based on timeout value returned from GetNextClock.
   // It's ok to call this too frequently.
-  void NotifyClock(uint32 now);
+  void NotifyClock(uint32_t now);
 
   // Call this whenever a packet arrives.
   // Returns true if the packet was processed successfully.
@@ -76,7 +76,7 @@ class PseudoTcp {
 
   // Call this to determine the next time NotifyClock should be called.
   // Returns false if the socket is ready to be destroyed.
-  bool GetNextClock(uint32 now, long& timeout);
+  bool GetNextClock(uint32_t now, long& timeout);
 
   // Call these to get/set option values to tailor this PseudoTcp
   // instance's behaviour for the kind of data it will carry.
@@ -94,47 +94,46 @@ class PseudoTcp {
   void SetOption(Option opt, int value);
 
   // Returns current congestion window in bytes.
-  uint32 GetCongestionWindow() const;
+  uint32_t GetCongestionWindow() const;
 
   // Returns amount of data in bytes that has been sent, but haven't
   // been acknowledged.
-  uint32 GetBytesInFlight() const;
+  uint32_t GetBytesInFlight() const;
 
   // Returns number of bytes that were written in buffer and haven't
   // been sent.
-  uint32 GetBytesBufferedNotSent() const;
+  uint32_t GetBytesBufferedNotSent() const;
 
   // Returns current round-trip time estimate in milliseconds.
-  uint32 GetRoundTripTimeEstimateMs() const;
+  uint32_t GetRoundTripTimeEstimateMs() const;
 
  protected:
   enum SendFlags { sfNone, sfDelayedAck, sfImmediateAck };
 
   struct Segment {
-    uint32 conv, seq, ack;
-    uint8 flags;
-    uint16 wnd;
+    uint32_t conv, seq, ack;
+    uint8_t flags;
+    uint16_t wnd;
     const char * data;
-    uint32 len;
-    uint32 tsval, tsecr;
+    uint32_t len;
+    uint32_t tsval, tsecr;
   };
 
   struct SSegment {
-    SSegment(uint32 s, uint32 l, bool c)
-        : seq(s), len(l), /*tstamp(0),*/ xmit(0), bCtrl(c) {
-    }
-    uint32 seq, len;
-    //uint32 tstamp;
-    uint8 xmit;
+    SSegment(uint32_t s, uint32_t l, bool c)
+        : seq(s), len(l), /*tstamp(0),*/ xmit(0), bCtrl(c) {}
+    uint32_t seq, len;
+    // uint32_t tstamp;
+    uint8_t xmit;
     bool bCtrl;
   };
   typedef std::list<SSegment> SList;
 
   struct RSegment {
-    uint32 seq, len;
+    uint32_t seq, len;
   };
 
-  uint32 queue(const char* data, uint32 len, bool bCtrl);
+  uint32_t queue(const char* data, uint32_t len, bool bCtrl);
 
   // Creates a packet and submits it to the network. This method can either
   // send payload or just an ACK packet.
@@ -144,18 +143,20 @@ class PseudoTcp {
   // |offset| is the offset to read from |m_sbuf|.
   // |len| is the number of bytes to read from |m_sbuf| as payload. If this
   // value is 0 then this is an ACK packet, otherwise this packet has payload.
-  IPseudoTcpNotify::WriteResult packet(uint32 seq, uint8 flags,
-                                       uint32 offset, uint32 len);
-  bool parse(const uint8* buffer, uint32 size);
+  IPseudoTcpNotify::WriteResult packet(uint32_t seq,
+                                       uint8_t flags,
+                                       uint32_t offset,
+                                       uint32_t len);
+  bool parse(const uint8_t* buffer, uint32_t size);
 
   void attemptSend(SendFlags sflags = sfNone);
 
-  void closedown(uint32 err = 0);
+  void closedown(uint32_t err = 0);
 
-  bool clock_check(uint32 now, long& nTimeout);
+  bool clock_check(uint32_t now, long& nTimeout);
 
   bool process(Segment& seg);
-  bool transmit(const SList::iterator& seg, uint32 now);
+  bool transmit(const SList::iterator& seg, uint32_t now);
 
   void adjustMTU();
 
@@ -172,20 +173,20 @@ class PseudoTcp {
   void queueConnectMessage();
 
   // Parse TCP options in the header.
-  void parseOptions(const char* data, uint32 len);
+  void parseOptions(const char* data, uint32_t len);
 
   // Apply a TCP option that has been read from the header.
-  void applyOption(char kind, const char* data, uint32 len);
+  void applyOption(char kind, const char* data, uint32_t len);
 
   // Apply window scale option.
-  void applyWindowScaleOption(uint8 scale_factor);
+  void applyWindowScaleOption(uint8_t scale_factor);
 
   // Resize the send buffer with |new_size| in bytes.
-  void resizeSendBuffer(uint32 new_size);
+  void resizeSendBuffer(uint32_t new_size);
 
   // Resize the receive buffer with |new_size| in bytes. This call adjusts
   // window scale factor |m_swnd_scale| accordingly.
-  void resizeReceiveBuffer(uint32 new_size);
+  void resizeReceiveBuffer(uint32_t new_size);
 
   IPseudoTcpNotify* m_notify;
   enum Shutdown { SD_NONE, SD_GRACEFUL, SD_FORCEFUL } m_shutdown;
@@ -193,43 +194,43 @@ class PseudoTcp {
 
   // TCB data
   TcpState m_state;
-  uint32 m_conv;
+  uint32_t m_conv;
   bool m_bReadEnable, m_bWriteEnable, m_bOutgoing;
-  uint32 m_lasttraffic;
+  uint32_t m_lasttraffic;
 
   // Incoming data
   typedef std::list<RSegment> RList;
   RList m_rlist;
-  uint32 m_rbuf_len, m_rcv_nxt, m_rcv_wnd, m_lastrecv;
-  uint8 m_rwnd_scale;  // Window scale factor.
+  uint32_t m_rbuf_len, m_rcv_nxt, m_rcv_wnd, m_lastrecv;
+  uint8_t m_rwnd_scale;  // Window scale factor.
   rtc::FifoBuffer m_rbuf;
 
   // Outgoing data
   SList m_slist;
-  uint32 m_sbuf_len, m_snd_nxt, m_snd_wnd, m_lastsend, m_snd_una;
-  uint8 m_swnd_scale;  // Window scale factor.
+  uint32_t m_sbuf_len, m_snd_nxt, m_snd_wnd, m_lastsend, m_snd_una;
+  uint8_t m_swnd_scale;  // Window scale factor.
   rtc::FifoBuffer m_sbuf;
 
   // Maximum segment size, estimated protocol level, largest segment sent
-  uint32 m_mss, m_msslevel, m_largest, m_mtu_advise;
+  uint32_t m_mss, m_msslevel, m_largest, m_mtu_advise;
   // Retransmit timer
-  uint32 m_rto_base;
+  uint32_t m_rto_base;
 
   // Timestamp tracking
-  uint32 m_ts_recent, m_ts_lastack;
+  uint32_t m_ts_recent, m_ts_lastack;
 
   // Round-trip calculation
-  uint32 m_rx_rttvar, m_rx_srtt, m_rx_rto;
+  uint32_t m_rx_rttvar, m_rx_srtt, m_rx_rto;
 
   // Congestion avoidance, Fast retransmit/recovery, Delayed ACKs
-  uint32 m_ssthresh, m_cwnd;
-  uint8 m_dup_acks;
-  uint32 m_recover;
-  uint32 m_t_ack;
+  uint32_t m_ssthresh, m_cwnd;
+  uint8_t m_dup_acks;
+  uint32_t m_recover;
+  uint32_t m_t_ack;
 
   // Configuration options
   bool m_use_nagling;
-  uint32 m_ack_delay;
+  uint32_t m_ack_delay;
 
   // This is used by unit tests to test backward compatibility of
   // PseudoTcp implementations that don't support window scaling.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/pseudotcp_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/pseudotcp_unittest.cc
index 03e72932a0..c9ccbca1d9 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/pseudotcp_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/pseudotcp_unittest.cc
@@ -27,9 +27,8 @@ static const int kBlockSize = 4096;
 
 class PseudoTcpForTest : public cricket::PseudoTcp {
  public:
-  PseudoTcpForTest(cricket::IPseudoTcpNotify* notify, uint32 conv)
-      : PseudoTcp(notify, conv) {
-  }
+  PseudoTcpForTest(cricket::IPseudoTcpNotify* notify, uint32_t conv)
+      : PseudoTcp(notify, conv) {}
 
   bool isReceiveBufferFull() const {
     return PseudoTcp::isReceiveBufferFull();
@@ -127,7 +126,7 @@ class PseudoTcpTestBase : public testing::Test,
   //   virtual void OnTcpReadable(PseudoTcp* tcp)
   // and
   //   virtual void OnTcpWritable(PseudoTcp* tcp)
-  virtual void OnTcpClosed(PseudoTcp* tcp, uint32 error) {
+  virtual void OnTcpClosed(PseudoTcp* tcp, uint32_t error) {
     // Consider ourselves closed when the remote side gets OnTcpClosed.
     // TODO: OnTcpClosed is only ever notified in case of error in
     // the current implementation.  Solicited close is not (yet) supported.
@@ -141,7 +140,7 @@ class PseudoTcpTestBase : public testing::Test,
                                      const char* buffer, size_t len) {
     // Randomly drop the desired percentage of packets.
     // Also drop packets that are larger than the configured MTU.
-    if (rtc::CreateRandomId() % 100 < static_cast<uint32>(loss_)) {
+    if (rtc::CreateRandomId() % 100 < static_cast<uint32_t>(loss_)) {
       LOG(LS_VERBOSE) << "Randomly dropping packet, size=" << len;
     } else if (len > static_cast<size_t>(std::min(local_mtu_, remote_mtu_))) {
       LOG(LS_VERBOSE) << "Dropping packet that exceeds path MTU, size=" << len;
@@ -156,7 +155,7 @@ class PseudoTcpTestBase : public testing::Test,
 
   void UpdateLocalClock() { UpdateClock(&local_, MSG_LCLOCK); }
   void UpdateRemoteClock() { UpdateClock(&remote_, MSG_RCLOCK); }
-  void UpdateClock(PseudoTcp* tcp, uint32 message) {
+  void UpdateClock(PseudoTcp* tcp, uint32_t message) {
     long interval = 0;  // NOLINT
     tcp->GetNextClock(PseudoTcp::Now(), interval);
     interval = std::max<int>(interval, 0L);  // sometimes interval is < 0
@@ -209,7 +208,7 @@ class PseudoTcpTestBase : public testing::Test,
 class PseudoTcpTest : public PseudoTcpTestBase {
  public:
   void TestTransfer(int size) {
-    uint32 start, elapsed;
+    uint32_t start, elapsed;
     size_t received;
     // Create some dummy data to send.
     send_stream_.ReserveSize(size);
@@ -326,7 +325,7 @@ class PseudoTcpTestPingPong : public PseudoTcpTestBase {
     bytes_per_send_ = bytes;
   }
   void TestPingPong(int size, int iterations) {
-    uint32 start, elapsed;
+    uint32_t start, elapsed;
     iterations_remaining_ = iterations;
     receiver_ = &remote_;
     sender_ = &local_;
@@ -489,12 +488,12 @@ class PseudoTcpTestReceiveWindow : public PseudoTcpTestBase {
     }
   }
 
-  uint32 EstimateReceiveWindowSize() const {
-    return static_cast<uint32>(recv_position_[0]);
+  uint32_t EstimateReceiveWindowSize() const {
+    return static_cast<uint32_t>(recv_position_[0]);
   }
 
-  uint32 EstimateSendWindowSize() const {
-    return static_cast<uint32>(send_position_[0] - recv_position_[0]);
+  uint32_t EstimateSendWindowSize() const {
+    return static_cast<uint32_t>(send_position_[0] - recv_position_[0]);
   }
 
  private:
diff --git a/media/webrtc/trunk/webrtc/p2p/base/rawtransport.cc b/media/webrtc/trunk/webrtc/p2p/base/rawtransport.cc
index 8ff00cd3e0..cb700ae4a0 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/rawtransport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/rawtransport.cc
@@ -1,43 +1,2 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-
-#include "webrtc/p2p/base/rawtransport.h"
-#include "webrtc/p2p/base/rawtransportchannel.h"
-#include "webrtc/base/common.h"
-
-#if defined(FEATURE_ENABLE_PSTN)
-namespace cricket {
-
-RawTransport::RawTransport(rtc::Thread* signaling_thread,
-                           rtc::Thread* worker_thread,
-                           const std::string& content_name,
-                           PortAllocator* allocator)
-    : Transport(signaling_thread, worker_thread,
-                content_name, NS_GINGLE_RAW, allocator) {
-}
-
-RawTransport::~RawTransport() {
-  DestroyAllChannels();
-}
-
-TransportChannelImpl* RawTransport::CreateTransportChannel(int component) {
-  return new RawTransportChannel(content_name(), component, this,
-                                 worker_thread(),
-                                 port_allocator());
-}
-
-void RawTransport::DestroyTransportChannel(TransportChannelImpl* channel) {
-  delete channel;
-}
-
-}  // namespace cricket
-#endif  // defined(FEATURE_ENABLE_PSTN)
+// TODO(pthatcher): Remove this file once Chrome's build files no
+// longer refer to it.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/rawtransport.h b/media/webrtc/trunk/webrtc/p2p/base/rawtransport.h
index bf0560b172..cb700ae4a0 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/rawtransport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/rawtransport.h
@@ -1,46 +1,2 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_BASE_RAWTRANSPORT_H_
-#define WEBRTC_P2P_BASE_RAWTRANSPORT_H_
-
-#include <string>
-#include "webrtc/p2p/base/transport.h"
-
-#if defined(FEATURE_ENABLE_PSTN)
-namespace cricket {
-
-// Implements a transport that only sends raw packets, no STUN.  As a result,
-// it cannot do pings to determine connectivity, so it only uses a single port
-// that it thinks will work.
-class RawTransport : public Transport {
- public:
-  RawTransport(rtc::Thread* signaling_thread,
-               rtc::Thread* worker_thread,
-               const std::string& content_name,
-               PortAllocator* allocator);
-  virtual ~RawTransport();
-
- protected:
-  // Creates and destroys raw channels.
-  virtual TransportChannelImpl* CreateTransportChannel(int component);
-  virtual void DestroyTransportChannel(TransportChannelImpl* channel);
-
- private:
-  friend class RawTransportChannel;  // For ParseAddress.
-
-  DISALLOW_EVIL_CONSTRUCTORS(RawTransport);
-};
-
-}  // namespace cricket
-
-#endif  // defined(FEATURE_ENABLE_PSTN)
-
-#endif  // WEBRTC_P2P_BASE_RAWTRANSPORT_H_
+// TODO(pthatcher): Remove this file once Chrome's build files no
+// longer refer to it.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.cc b/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.cc
index b032e63cda..cb700ae4a0 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.cc
@@ -1,260 +1,2 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/p2p/base/rawtransportchannel.h"
-
-#include <string>
-#include <vector>
-#include "webrtc/p2p/base/constants.h"
-#include "webrtc/p2p/base/portallocator.h"
-#include "webrtc/p2p/base/portinterface.h"
-#include "webrtc/p2p/base/rawtransport.h"
-#include "webrtc/p2p/base/relayport.h"
-#include "webrtc/p2p/base/stunport.h"
-#include "webrtc/base/common.h"
-
-#if defined(FEATURE_ENABLE_PSTN)
-
-namespace {
-
-const uint32 MSG_DESTROY_RTC_UNUSED_PORTS = 1;
-
-}  // namespace
-
-namespace cricket {
-
-RawTransportChannel::RawTransportChannel(const std::string& content_name,
-                                         int component,
-                                         RawTransport* transport,
-                                         rtc::Thread *worker_thread,
-                                         PortAllocator *allocator)
-  : TransportChannelImpl(content_name, component),
-    raw_transport_(transport),
-    allocator_(allocator),
-    allocator_session_(NULL),
-    stun_port_(NULL),
-    relay_port_(NULL),
-    port_(NULL),
-    use_relay_(false) {
-  if (worker_thread == NULL)
-    worker_thread_ = raw_transport_->worker_thread();
-  else
-    worker_thread_ = worker_thread;
-}
-
-RawTransportChannel::~RawTransportChannel() {
-  delete allocator_session_;
-}
-
-int RawTransportChannel::SendPacket(const char *data, size_t size,
-                                    const rtc::PacketOptions& options,
-                                    int flags) {
-  if (port_ == NULL)
-    return -1;
-  if (remote_address_.IsNil())
-    return -1;
-  if (flags != 0)
-    return -1;
-  return port_->SendTo(data, size, remote_address_, options, true);
-}
-
-int RawTransportChannel::SetOption(rtc::Socket::Option opt, int value) {
-  // TODO: allow these to be set before we have a port
-  if (port_ == NULL)
-    return -1;
-  return port_->SetOption(opt, value);
-}
-
-bool RawTransportChannel::GetOption(rtc::Socket::Option opt, int* value) {
-  return false;
-}
-
-int RawTransportChannel::GetError() {
-  return (port_ != NULL) ? port_->GetError() : 0;
-}
-
-void RawTransportChannel::Connect() {
-  // Create an allocator that only returns stun and relay ports.
-  // Use empty string for ufrag and pwd here. There won't be any STUN or relay
-  // interactions when using RawTC.
-  // TODO: Change raw to only use local udp ports.
-  allocator_session_ = allocator_->CreateSession(
-      SessionId(), content_name(), component(), "", "");
-
-  uint32 flags = PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_TCP;
-
-#if !defined(FEATURE_ENABLE_STUN_CLASSIFICATION)
-  flags |= PORTALLOCATOR_DISABLE_RELAY;
-#endif
-  allocator_session_->set_flags(flags);
-  allocator_session_->SignalPortReady.connect(
-      this, &RawTransportChannel::OnPortReady);
-  allocator_session_->SignalCandidatesReady.connect(
-      this, &RawTransportChannel::OnCandidatesReady);
-
-  // The initial ports will include stun.
-  allocator_session_->StartGettingPorts();
-}
-
-void RawTransportChannel::Reset() {
-  set_readable(false);
-  set_writable(false);
-
-  delete allocator_session_;
-
-  allocator_session_ = NULL;
-  stun_port_ = NULL;
-  relay_port_ = NULL;
-  port_ = NULL;
-  remote_address_ = rtc::SocketAddress();
-}
-
-void RawTransportChannel::OnCandidate(const Candidate& candidate) {
-  remote_address_ = candidate.address();
-  ASSERT(!remote_address_.IsNil());
-  set_readable(true);
-
-  // We can write once we have a port and a remote address.
-  if (port_ != NULL)
-    SetWritable();
-}
-
-void RawTransportChannel::OnRemoteAddress(
-    const rtc::SocketAddress& remote_address) {
-  remote_address_ = remote_address;
-  set_readable(true);
-
-  if (port_ != NULL)
-    SetWritable();
-}
-
-// Note about stun classification
-// Code to classify our NAT type and use the relay port if we are behind an
-// asymmetric NAT is under a FEATURE_ENABLE_STUN_CLASSIFICATION #define.
-// To turn this one we will have to enable a second stun address and make sure
-// that the relay server works for raw UDP.
-//
-// Another option is to classify the NAT type early and not offer the raw
-// transport type at all if we can't support it.
-
-void RawTransportChannel::OnPortReady(
-    PortAllocatorSession* session, PortInterface* port) {
-  ASSERT(session == allocator_session_);
-
-  if (port->Type() == STUN_PORT_TYPE) {
-    stun_port_ = static_cast<StunPort*>(port);
-  } else if (port->Type() == RELAY_PORT_TYPE) {
-    relay_port_ = static_cast<RelayPort*>(port);
-  } else {
-    ASSERT(false);
-  }
-}
-
-void RawTransportChannel::OnCandidatesReady(
-    PortAllocatorSession *session, const std::vector<Candidate>& candidates) {
-  ASSERT(session == allocator_session_);
-  ASSERT(candidates.size() >= 1);
-
-  // The most recent candidate is the one we haven't seen yet.
-  Candidate c = candidates[candidates.size() - 1];
-
-  if (c.type() == STUN_PORT_TYPE) {
-    ASSERT(stun_port_ != NULL);
-
-#if defined(FEATURE_ENABLE_STUN_CLASSIFICATION)
-    // We need to wait until we have two addresses.
-    if (stun_port_->candidates().size() < 2)
-      return;
-
-    // This is the second address.  If these addresses are the same, then we
-    // are not behind a symmetric NAT.  Hence, a stun port should be sufficient.
-    if (stun_port_->candidates()[0].address() ==
-        stun_port_->candidates()[1].address()) {
-      SetPort(stun_port_);
-      return;
-    }
-
-    // We will need to use relay.
-    use_relay_ = true;
-
-    // If we already have a relay address, we're good.  Otherwise, we will need
-    // to wait until one arrives.
-    if (relay_port_->candidates().size() > 0)
-      SetPort(relay_port_);
-#else  // defined(FEATURE_ENABLE_STUN_CLASSIFICATION)
-    // Always use the stun port.  We don't classify right now so just assume it
-    // will work fine.
-    SetPort(stun_port_);
-#endif
-  } else if (c.type() == RELAY_PORT_TYPE) {
-    if (use_relay_)
-      SetPort(relay_port_);
-  } else {
-    ASSERT(false);
-  }
-}
-
-void RawTransportChannel::SetPort(PortInterface* port) {
-  ASSERT(port_ == NULL);
-  port_ = port;
-
-  // We don't need any ports other than the one we picked.
-  allocator_session_->StopGettingPorts();
-  worker_thread_->Post(
-      this, MSG_DESTROY_RTC_UNUSED_PORTS, NULL);
-
-  // Send a message to the other client containing our address.
-
-  ASSERT(port_->Candidates().size() >= 1);
-  ASSERT(port_->Candidates()[0].protocol() == "udp");
-  SignalCandidateReady(this, port_->Candidates()[0]);
-
-  // Read all packets from this port.
-  port_->EnablePortPackets();
-  port_->SignalReadPacket.connect(this, &RawTransportChannel::OnReadPacket);
-
-  // We can write once we have a port and a remote address.
-  if (!remote_address_.IsAny())
-    SetWritable();
-}
-
-void RawTransportChannel::SetWritable() {
-  ASSERT(port_ != NULL);
-  ASSERT(!remote_address_.IsAny());
-
-  set_writable(true);
-
-  Candidate remote_candidate;
-  remote_candidate.set_address(remote_address_);
-  SignalRouteChange(this, remote_candidate);
-}
-
-void RawTransportChannel::OnReadPacket(
-    PortInterface* port, const char* data, size_t size,
-    const rtc::SocketAddress& addr) {
-  ASSERT(port_ == port);
-  SignalReadPacket(this, data, size, rtc::CreatePacketTime(0), 0);
-}
-
-void RawTransportChannel::OnMessage(rtc::Message* msg) {
-  ASSERT(msg->message_id == MSG_DESTROY_RTC_UNUSED_PORTS);
-  ASSERT(port_ != NULL);
-  if (port_ != stun_port_) {
-    stun_port_->Destroy();
-    stun_port_ = NULL;
-  }
-  if (port_ != relay_port_ && relay_port_ != NULL) {
-    relay_port_->Destroy();
-    relay_port_ = NULL;
-  }
-}
-
-}  // namespace cricket
-#endif  // defined(FEATURE_ENABLE_PSTN)
+// TODO(pthatcher): Remove this file once Chrome's build files no
+// longer refer to it.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.h b/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.h
index 3455325c51..cb700ae4a0 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/rawtransportchannel.h
@@ -1,198 +1,2 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
-#define WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
-
-#include <string>
-#include <vector>
-#include "webrtc/p2p/base/candidate.h"
-#include "webrtc/p2p/base/rawtransport.h"
-#include "webrtc/p2p/base/transportchannelimpl.h"
-#include "webrtc/base/messagequeue.h"
-
-#if defined(FEATURE_ENABLE_PSTN)
-
-namespace rtc {
-class Thread;
-}
-
-namespace cricket {
-
-class Connection;
-class PortAllocator;
-class PortAllocatorSession;
-class PortInterface;
-class RelayPort;
-class StunPort;
-
-// Implements a channel that just sends bare packets once we have received the
-// address of the other side.  We pick a single address to send them based on
-// a simple investigation of NAT type.
-class RawTransportChannel : public TransportChannelImpl,
-    public rtc::MessageHandler {
- public:
-  RawTransportChannel(const std::string& content_name,
-                      int component,
-                      RawTransport* transport,
-                      rtc::Thread *worker_thread,
-                      PortAllocator *allocator);
-  virtual ~RawTransportChannel();
-
-  // Implementation of normal channel packet sending.
-  virtual int SendPacket(const char *data, size_t len,
-                         const rtc::PacketOptions& options, int flags);
-  virtual int SetOption(rtc::Socket::Option opt, int value);
-  virtual bool GetOption(rtc::Socket::Option opt, int* value);
-  virtual int GetError();
-
-  // Implements TransportChannelImpl.
-  virtual Transport* GetTransport() { return raw_transport_; }
-  virtual TransportChannelState GetState() const {
-    return TransportChannelState::STATE_COMPLETED;
-  }
-  virtual void SetIceCredentials(const std::string& ice_ufrag,
-                                 const std::string& ice_pwd) {}
-  virtual void SetRemoteIceCredentials(const std::string& ice_ufrag,
-                                       const std::string& ice_pwd) {}
-
-  // Creates an allocator session to start figuring out which type of
-  // port we should send to the other client.  This will send
-  // SignalAvailableCandidate once we have decided.
-  virtual void Connect();
-
-  // Resets state back to unconnected.
-  virtual void Reset();
-
-  // We don't actually worry about signaling since we can't send new candidates.
-  virtual void OnSignalingReady() {}
-
-  // Handles a message setting the remote address.  We are writable once we
-  // have this since we now know where to send.
-  virtual void OnCandidate(const Candidate& candidate);
-
-  void OnRemoteAddress(const rtc::SocketAddress& remote_address);
-
-  // Below ICE specific virtual methods not implemented.
-  virtual IceRole GetIceRole() const { return ICEROLE_UNKNOWN; }
-  virtual void SetIceRole(IceRole role) {}
-  virtual void SetIceTiebreaker(uint64 tiebreaker) {}
-
-  virtual bool GetIceProtocolType(IceProtocolType* type) const { return false; }
-  virtual void SetIceProtocolType(IceProtocolType type) {}
-
-  virtual void SetIceUfrag(const std::string& ice_ufrag) {}
-  virtual void SetIcePwd(const std::string& ice_pwd) {}
-  virtual void SetRemoteIceMode(IceMode mode) {}
-  virtual size_t GetConnectionCount() const { return 1; }
-
-  virtual bool GetStats(ConnectionInfos* infos) {
-    return false;
-  }
-
-  // DTLS methods.
-  virtual bool IsDtlsActive() const { return false; }
-
-  // Default implementation.
-  virtual bool GetSslRole(rtc::SSLRole* role) const {
-    return false;
-  }
-
-  virtual bool SetSslRole(rtc::SSLRole role) {
-    return false;
-  }
-
-  // Set up the ciphers to use for DTLS-SRTP.
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
-    return false;
-  }
-
-  // Find out which DTLS-SRTP cipher was negotiated.
-  virtual bool GetSrtpCipher(std::string* cipher) {
-    return false;
-  }
-
-  // Find out which DTLS cipher was negotiated.
-  virtual bool GetSslCipher(std::string* cipher) {
-    return false;
-  }
-
-  // Returns false because the channel is not DTLS.
-  virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const {
-    return false;
-  }
-
-  virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const {
-    return false;
-  }
-
-  // Allows key material to be extracted for external encryption.
-  virtual bool ExportKeyingMaterial(
-      const std::string& label,
-      const uint8* context,
-      size_t context_len,
-      bool use_context,
-      uint8* result,
-      size_t result_len) {
-    return false;
-  }
-
-  virtual bool SetLocalIdentity(rtc::SSLIdentity* identity) {
-    return false;
-  }
-
-  // Set DTLS Remote fingerprint. Must be after local identity set.
-  virtual bool SetRemoteFingerprint(
-    const std::string& digest_alg,
-    const uint8* digest,
-    size_t digest_len) {
-    return false;
-  }
-
- private:
-  RawTransport* raw_transport_;
-  rtc::Thread *worker_thread_;
-  PortAllocator* allocator_;
-  PortAllocatorSession* allocator_session_;
-  StunPort* stun_port_;
-  RelayPort* relay_port_;
-  PortInterface* port_;
-  bool use_relay_;
-  rtc::SocketAddress remote_address_;
-
-  // Called when the allocator creates another port.
-  void OnPortReady(PortAllocatorSession* session, PortInterface* port);
-
-  // Called when one of the ports we are using has determined its address.
-  void OnCandidatesReady(PortAllocatorSession *session,
-                         const std::vector<Candidate>& candidates);
-
-  // Called once we have chosen the port to use for communication with the
-  // other client.  This will send its address and prepare the port for use.
-  void SetPort(PortInterface* port);
-
-  // Called once we have a port and a remote address.  This will set mark the
-  // channel as writable and signal the route to the client.
-  void SetWritable();
-
-  // Called when we receive a packet from the other client.
-  void OnReadPacket(PortInterface* port, const char* data, size_t size,
-                    const rtc::SocketAddress& addr);
-
-  // Handles a message to destroy unused ports.
-  virtual void OnMessage(rtc::Message *msg);
-
-  DISALLOW_EVIL_CONSTRUCTORS(RawTransportChannel);
-};
-
-}  // namespace cricket
-
-#endif  // defined(FEATURE_ENABLE_PSTN)
-#endif  // WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
+// TODO(pthatcher): Remove this file once Chrome's build files no
+// longer refer to it.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/relayport.cc b/media/webrtc/trunk/webrtc/p2p/base/relayport.cc
index 8e74ee3682..19883a3121 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/relayport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/relayport.cc
@@ -16,7 +16,7 @@
 
 namespace cricket {
 
-static const uint32 kMessageConnectTimeout = 1;
+static const uint32_t kMessageConnectTimeout = 1;
 static const int kKeepAliveDelay           = 10 * 60 * 1000;
 static const int kRetryTimeout             = 50 * 1000;  // ICE says 50 secs
 // How long to wait for a socket to connect to remote host in milliseconds
@@ -144,6 +144,10 @@ class RelayEntry : public rtc::MessageHandler,
     const char* data, size_t size,
     const rtc::SocketAddress& remote_addr,
     const rtc::PacketTime& packet_time);
+
+  void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                    const rtc::SentPacket& sent_packet);
+
   // Called when the socket is currently able to send.
   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
 
@@ -159,30 +163,38 @@ class AllocateRequest : public StunRequest {
   AllocateRequest(RelayEntry* entry, RelayConnection* connection);
   virtual ~AllocateRequest() {}
 
-  virtual void Prepare(StunMessage* request);
+  void Prepare(StunMessage* request) override;
 
-  virtual int GetNextDelay();
+  void OnSent() override;
+  int resend_delay() override;
 
-  virtual void OnResponse(StunMessage* response);
-  virtual void OnErrorResponse(StunMessage* response);
-  virtual void OnTimeout();
+  void OnResponse(StunMessage* response) override;
+  void OnErrorResponse(StunMessage* response) override;
+  void OnTimeout() override;
 
  private:
   RelayEntry* entry_;
   RelayConnection* connection_;
-  uint32 start_time_;
+  uint32_t start_time_;
 };
 
 RelayPort::RelayPort(rtc::Thread* thread,
                      rtc::PacketSocketFactory* factory,
                      rtc::Network* network,
                      const rtc::IPAddress& ip,
-                     uint16 min_port,
-                     uint16 max_port,
+                     uint16_t min_port,
+                     uint16_t max_port,
                      const std::string& username,
                      const std::string& password)
-    : Port(thread, RELAY_PORT_TYPE, factory, network, ip, min_port, max_port,
-           username, password),
+    : Port(thread,
+           RELAY_PORT_TYPE,
+           factory,
+           network,
+           ip,
+           min_port,
+           max_port,
+           username,
+           password),
       ready_(false),
       error_(0) {
   entries_.push_back(
@@ -230,9 +242,9 @@ void RelayPort::SetReady() {
       // In case of Gturn, related address is set to null socket address.
       // This is due to as mapped address stun attribute is used for allocated
       // address.
-      AddAddress(iter->address, iter->address, rtc::SocketAddress(),
-                 proto_name, "", RELAY_PORT_TYPE,
-                 ICE_TYPE_PREFERENCE_RELAY, 0, false);
+      AddAddress(iter->address, iter->address, rtc::SocketAddress(), proto_name,
+                 proto_name, "", RELAY_PORT_TYPE, ICE_TYPE_PREFERENCE_RELAY, 0,
+                 false);
     }
     ready_ = true;
     SignalPortComplete(this);
@@ -500,6 +512,7 @@ void RelayEntry::Connect() {
 
   // Otherwise, create the new connection and configure any socket options.
   socket->SignalReadPacket.connect(this, &RelayEntry::OnReadPacket);
+  socket->SignalSentPacket.connect(this, &RelayEntry::OnSentPacket);
   socket->SignalReadyToSend.connect(this, &RelayEntry::OnReadyToSend);
   current_connection_ = new RelayConnection(ra, socket, port()->thread());
   for (size_t i = 0; i < port_->options().size(); ++i) {
@@ -739,6 +752,11 @@ void RelayEntry::OnReadPacket(
                       PROTO_UDP, packet_time);
 }
 
+void RelayEntry::OnSentPacket(rtc::AsyncPacketSocket* socket,
+                              const rtc::SentPacket& sent_packet) {
+  port_->OnSentPacket(socket, sent_packet);
+}
+
 void RelayEntry::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
   if (connected()) {
     port_->OnReadyToSend();
@@ -775,14 +793,20 @@ void AllocateRequest::Prepare(StunMessage* request) {
   VERIFY(request->AddAttribute(username_attr));
 }
 
-int AllocateRequest::GetNextDelay() {
-  int delay = 100 * std::max(1 << count_, 2);
+void AllocateRequest::OnSent() {
   count_ += 1;
   if (count_ == 5)
     timeout_ = true;
-  return delay;
 }
 
+int AllocateRequest::resend_delay() {
+  if (count_ == 0) {
+    return 0;
+  }
+  return 100 * std::max(1 << (count_-1), 2);
+}
+
+
 void AllocateRequest::OnResponse(StunMessage* response) {
   const StunAddressAttribute* addr_attr =
       response->GetAddress(STUN_ATTR_MAPPED_ADDRESS);
diff --git a/media/webrtc/trunk/webrtc/p2p/base/relayport.h b/media/webrtc/trunk/webrtc/p2p/base/relayport.h
index 629714267d..402736c34d 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/relayport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/relayport.h
@@ -29,25 +29,24 @@ class RelayConnection;
 // is created. The RelayEntry will try to reach the remote destination
 // by connecting to all available server addresses in a pre defined
 // order with a small delay in between. When a connection is
-// successful all other connection attemts are aborted.
+// successful all other connection attempts are aborted.
 class RelayPort : public Port {
  public:
   typedef std::pair<rtc::Socket::Option, int> OptionValue;
 
   // RelayPort doesn't yet do anything fancy in the ctor.
-  static RelayPort* Create(
-      rtc::Thread* thread,
-      rtc::PacketSocketFactory* factory,
-      rtc::Network* network,
-      const rtc::IPAddress& ip,
-      uint16 min_port,
-      uint16 max_port,
-      const std::string& username,
-      const std::string& password) {
+  static RelayPort* Create(rtc::Thread* thread,
+                           rtc::PacketSocketFactory* factory,
+                           rtc::Network* network,
+                           const rtc::IPAddress& ip,
+                           uint16_t min_port,
+                           uint16_t max_port,
+                           const std::string& username,
+                           const std::string& password) {
     return new RelayPort(thread, factory, network, ip, min_port, max_port,
                          username, password);
   }
-  virtual ~RelayPort();
+  ~RelayPort() override;
 
   void AddServerAddress(const ProtocolAddress& addr);
   void AddExternalAddress(const ProtocolAddress& addr);
@@ -55,12 +54,16 @@ class RelayPort : public Port {
   const std::vector<OptionValue>& options() const { return options_; }
   bool HasMagicCookie(const char* data, size_t size);
 
-  virtual void PrepareAddress();
-  virtual Connection* CreateConnection(const Candidate& address,
-                                       CandidateOrigin origin);
-  virtual int SetOption(rtc::Socket::Option opt, int value);
-  virtual int GetOption(rtc::Socket::Option opt, int* value);
-  virtual int GetError();
+  void PrepareAddress() override;
+  Connection* CreateConnection(const Candidate& address,
+                               CandidateOrigin origin) override;
+  int SetOption(rtc::Socket::Option opt, int value) override;
+  int GetOption(rtc::Socket::Option opt, int* value) override;
+  int GetError() override;
+  bool SupportsProtocol(const std::string& protocol) const override {
+    // Relay port may create both TCP and UDP connections.
+    return true;
+  }
 
   const ProtocolAddress * ServerAddress(size_t index) const;
   bool IsReady() { return ready_; }
@@ -74,18 +77,19 @@ class RelayPort : public Port {
             rtc::PacketSocketFactory* factory,
             rtc::Network*,
             const rtc::IPAddress& ip,
-            uint16 min_port,
-            uint16 max_port,
+            uint16_t min_port,
+            uint16_t max_port,
             const std::string& username,
             const std::string& password);
   bool Init();
 
   void SetReady();
 
-  virtual int SendTo(const void* data, size_t size,
-                     const rtc::SocketAddress& addr,
-                     const rtc::PacketOptions& options,
-                     bool payload);
+  int SendTo(const void* data,
+             size_t size,
+             const rtc::SocketAddress& addr,
+             const rtc::PacketOptions& options,
+             bool payload) override;
 
   // Dispatches the given packet to the port or connection as appropriate.
   void OnReadPacket(const char* data, size_t size,
@@ -93,6 +97,11 @@ class RelayPort : public Port {
                     ProtocolType proto,
                     const rtc::PacketTime& packet_time);
 
+  // The OnSentPacket callback is left empty here since they are handled by
+  // RelayEntry.
+  void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                    const rtc::SentPacket& sent_packet) override {}
+
  private:
   friend class RelayEntry;
 
diff --git a/media/webrtc/trunk/webrtc/p2p/base/relayserver.cc b/media/webrtc/trunk/webrtc/p2p/base/relayserver.cc
index 19e9268277..e208d70d0f 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/relayserver.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/relayserver.cc
@@ -27,7 +27,7 @@ namespace cricket {
 const int MAX_LIFETIME = 15 * 60 * 1000;
 
 // The number of bytes in each of the usernames we use.
-const uint32 USERNAME_LENGTH = 16;
+const uint32_t USERNAME_LENGTH = 16;
 
 // Calls SendTo on the given socket and logs any bad results.
 void Send(rtc::AsyncPacketSocket* socket, const char* bytes, size_t size,
@@ -263,8 +263,8 @@ void RelayServer::OnExternalPacket(
     return;
   }
 
-  uint32 length =
-      std::min(static_cast<uint32>(username_attr->length()), USERNAME_LENGTH);
+  uint32_t length =
+      std::min(static_cast<uint32_t>(username_attr->length()), USERNAME_LENGTH);
   std::string username(username_attr->bytes(), length);
   // TODO: Check the HMAC.
 
@@ -355,7 +355,7 @@ void RelayServer::HandleStunAllocate(
     //       else-branch will then disappear.
 
     // Compute the appropriate lifetime for this binding.
-    uint32 lifetime = MAX_LIFETIME;
+    uint32_t lifetime = MAX_LIFETIME;
     const StunUInt32Attribute* lifetime_attr =
         request.GetUInt32(STUN_ATTR_LIFETIME);
     if (lifetime_attr)
@@ -530,7 +530,7 @@ void RelayServer::RemoveBinding(RelayServerBinding* binding) {
 
 void RelayServer::OnMessage(rtc::Message *pmsg) {
 #if ENABLE_DEBUG
-  static const uint32 kMessageAcceptConnection = 1;
+  static const uint32_t kMessageAcceptConnection = 1;
   ASSERT(pmsg->message_id == kMessageAcceptConnection);
 #endif
   rtc::MessageData* data = pmsg->pdata;
@@ -616,7 +616,7 @@ void RelayServerConnection::Send(
   StunByteStringAttribute* data_attr =
       StunAttribute::CreateByteString(STUN_ATTR_DATA);
   ASSERT(size <= 65536);
-  data_attr->CopyBytes(data, uint16(size));
+  data_attr->CopyBytes(data, uint16_t(size));
   msg.AddAttribute(data_attr);
 
   SendStun(msg);
@@ -648,13 +648,16 @@ void RelayServerConnection::Unlock() {
 }
 
 // IDs used for posted messages:
-const uint32 MSG_LIFETIME_TIMER = 1;
+const uint32_t MSG_LIFETIME_TIMER = 1;
 
-RelayServerBinding::RelayServerBinding(
-    RelayServer* server, const std::string& username,
-    const std::string& password, uint32 lifetime)
-  : server_(server), username_(username), password_(password),
-    lifetime_(lifetime) {
+RelayServerBinding::RelayServerBinding(RelayServer* server,
+                                       const std::string& username,
+                                       const std::string& password,
+                                       uint32_t lifetime)
+    : server_(server),
+      username_(username),
+      password_(password),
+      lifetime_(lifetime) {
   // For now, every connection uses the standard magic cookie value.
   magic_cookie_.append(
       reinterpret_cast<const char*>(TURN_MAGIC_COOKIE_VALUE),
diff --git a/media/webrtc/trunk/webrtc/p2p/base/relayserver.h b/media/webrtc/trunk/webrtc/p2p/base/relayserver.h
index e0e45d5254..f1109f1ce4 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/relayserver.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/relayserver.h
@@ -181,13 +181,14 @@ class RelayServerConnection {
 // or in other words, that are "bound" together.
 class RelayServerBinding : public rtc::MessageHandler {
  public:
-  RelayServerBinding(
-      RelayServer* server, const std::string& username,
-      const std::string& password, uint32 lifetime);
+  RelayServerBinding(RelayServer* server,
+                     const std::string& username,
+                     const std::string& password,
+                     uint32_t lifetime);
   virtual ~RelayServerBinding();
 
   RelayServer* server() { return server_; }
-  uint32 lifetime() { return lifetime_; }
+  uint32_t lifetime() { return lifetime_; }
   const std::string& username() { return username_; }
   const std::string& password() { return password_; }
   const std::string& magic_cookie() { return magic_cookie_; }
@@ -225,8 +226,8 @@ class RelayServerBinding : public rtc::MessageHandler {
   std::vector<RelayServerConnection*> internal_connections_;
   std::vector<RelayServerConnection*> external_connections_;
 
-  uint32 lifetime_;
-  uint32 last_used_;
+  uint32_t lifetime_;
+  uint32_t last_used_;
   // TODO: bandwidth
 };
 
diff --git a/media/webrtc/trunk/webrtc/p2p/base/relayserver_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/relayserver_unittest.cc
index 4f1164acc6..83e5353fc9 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/relayserver_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/relayserver_unittest.cc
@@ -24,7 +24,7 @@
 using rtc::SocketAddress;
 using namespace cricket;
 
-static const uint32 LIFETIME = 4;  // seconds
+static const uint32_t LIFETIME = 4;  // seconds
 static const SocketAddress server_int_addr("127.0.0.1", 5000);
 static const SocketAddress server_ext_addr("127.0.0.1", 5001);
 static const SocketAddress client1_addr("127.0.0.1", 6000 + (rand() % 1000));
diff --git a/media/webrtc/trunk/webrtc/p2p/base/session.cc b/media/webrtc/trunk/webrtc/p2p/base/session.cc
index 136f3919ab..1a23f8363f 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/session.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/session.cc
@@ -8,842 +8,5 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/p2p/base/session.h"
-
-#include "webrtc/p2p/base/dtlstransport.h"
-#include "webrtc/p2p/base/p2ptransport.h"
-#include "webrtc/p2p/base/transport.h"
-#include "webrtc/p2p/base/transportchannelproxy.h"
-#include "webrtc/p2p/base/transportinfo.h"
-#include "webrtc/base/bind.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/stringencode.h"
-#include "webrtc/base/sslstreamadapter.h"
-
-#include "webrtc/p2p/base/constants.h"
-
-namespace cricket {
-
-using rtc::Bind;
-
-TransportProxy::~TransportProxy() {
-  for (ChannelMap::iterator iter = channels_.begin();
-       iter != channels_.end(); ++iter) {
-    iter->second->SignalDestroyed(iter->second);
-    delete iter->second;
-  }
-}
-
-const std::string& TransportProxy::type() const {
-  return transport_->get()->type();
-}
-
-TransportChannel* TransportProxy::GetChannel(int component) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  return GetChannelProxy(component);
-}
-
-TransportChannel* TransportProxy::CreateChannel(int component) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(GetChannel(component) == NULL);
-  ASSERT(!transport_->get()->HasChannel(component));
-
-  // We always create a proxy in case we need to change out the transport later.
-  TransportChannelProxy* channel_proxy =
-      new TransportChannelProxy(content_name(), component);
-  channels_[component] = channel_proxy;
-
-  // If we're already negotiated, create an impl and hook it up to the proxy
-  // channel. If we're connecting, create an impl but don't hook it up yet.
-  if (negotiated_) {
-    CreateChannelImpl_w(component);
-    SetChannelImplFromTransport_w(channel_proxy, component);
-  } else if (connecting_) {
-    CreateChannelImpl_w(component);
-  }
-  return channel_proxy;
-}
-
-bool TransportProxy::HasChannel(int component) {
-  return transport_->get()->HasChannel(component);
-}
-
-void TransportProxy::DestroyChannel(int component) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  TransportChannelProxy* channel_proxy = GetChannelProxy(component);
-  if (channel_proxy) {
-    // If the state of TransportProxy is not NEGOTIATED then
-    // TransportChannelProxy and its impl are not connected. Both must
-    // be connected before deletion.
-    //
-    // However, if we haven't entered the connecting state then there
-    // is no implementation to hook up.
-    if (connecting_ && !negotiated_) {
-      SetChannelImplFromTransport_w(channel_proxy, component);
-    }
-
-    channels_.erase(component);
-    channel_proxy->SignalDestroyed(channel_proxy);
-    delete channel_proxy;
-  }
-}
-
-void TransportProxy::ConnectChannels() {
-  if (!connecting_) {
-    if (!negotiated_) {
-      for (auto& iter : channels_) {
-        CreateChannelImpl(iter.first);
-      }
-    }
-    connecting_ = true;
-  }
-  // TODO(juberti): Right now Transport::ConnectChannels doesn't work if we
-  // don't have any channels yet, so we need to allow this method to be called
-  // multiple times. Once we fix Transport, we can move this call inside the
-  // if (!connecting_) block.
-  transport_->get()->ConnectChannels();
-}
-
-void TransportProxy::CompleteNegotiation() {
-  if (!negotiated_) {
-    // Negotiating assumes connecting_ has happened and
-    // implementations exist. If not we need to create the
-    // implementations.
-    for (auto& iter : channels_) {
-      if (!connecting_) {
-        CreateChannelImpl(iter.first);
-      }
-      SetChannelImplFromTransport(iter.second, iter.first);
-    }
-    negotiated_ = true;
-  }
-}
-
-void TransportProxy::AddSentCandidates(const Candidates& candidates) {
-  for (Candidates::const_iterator cand = candidates.begin();
-       cand != candidates.end(); ++cand) {
-    sent_candidates_.push_back(*cand);
-  }
-}
-
-void TransportProxy::AddUnsentCandidates(const Candidates& candidates) {
-  for (Candidates::const_iterator cand = candidates.begin();
-       cand != candidates.end(); ++cand) {
-    unsent_candidates_.push_back(*cand);
-  }
-}
-
-TransportChannelProxy* TransportProxy::GetChannelProxy(int component) const {
-  ChannelMap::const_iterator iter = channels_.find(component);
-  return (iter != channels_.end()) ? iter->second : NULL;
-}
-
-void TransportProxy::CreateChannelImpl(int component) {
-  worker_thread_->Invoke<void>(Bind(
-      &TransportProxy::CreateChannelImpl_w, this, component));
-}
-
-void TransportProxy::CreateChannelImpl_w(int component) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  transport_->get()->CreateChannel(component);
-}
-
-void TransportProxy::SetChannelImplFromTransport(TransportChannelProxy* proxy,
-                                                 int component) {
-  worker_thread_->Invoke<void>(Bind(
-      &TransportProxy::SetChannelImplFromTransport_w, this, proxy, component));
-}
-
-void TransportProxy::SetChannelImplFromTransport_w(TransportChannelProxy* proxy,
-                                                   int component) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  TransportChannelImpl* impl = transport_->get()->GetChannel(component);
-  ASSERT(impl != NULL);
-  ReplaceChannelImpl_w(proxy, impl);
-}
-
-void TransportProxy::ReplaceChannelImpl(TransportChannelProxy* proxy,
-                                        TransportChannelImpl* impl) {
-  worker_thread_->Invoke<void>(Bind(
-      &TransportProxy::ReplaceChannelImpl_w, this, proxy, impl));
-}
-
-void TransportProxy::ReplaceChannelImpl_w(TransportChannelProxy* proxy,
-                                          TransportChannelImpl* impl) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(proxy != NULL);
-  proxy->SetImplementation(impl);
-}
-
-// This function muxes |this| onto |target| by repointing |this| at
-// |target|'s transport and setting our TransportChannelProxies
-// to point to |target|'s underlying implementations.
-bool TransportProxy::SetupMux(TransportProxy* target) {
-  // Bail out if there's nothing to do.
-  if (transport_ == target->transport_) {
-    return true;
-  }
-
-  // Run through all channels and remove any non-rtp transport channels before
-  // setting target transport channels.
-  for (ChannelMap::const_iterator iter = channels_.begin();
-       iter != channels_.end(); ++iter) {
-    if (!target->transport_->get()->HasChannel(iter->first)) {
-      // Remove if channel doesn't exist in |transport_|.
-      ReplaceChannelImpl(iter->second, NULL);
-    } else {
-      // Replace the impl for all the TransportProxyChannels with the channels
-      // from |target|'s transport. Fail if there's not an exact match.
-      ReplaceChannelImpl(
-          iter->second, target->transport_->get()->CreateChannel(iter->first));
-    }
-  }
-
-  // Now replace our transport. Must happen afterwards because
-  // it deletes all impls as a side effect.
-  transport_ = target->transport_;
-  transport_->get()->SignalCandidatesReady.connect(
-      this, &TransportProxy::OnTransportCandidatesReady);
-  set_candidates_allocated(target->candidates_allocated());
-  return true;
-}
-
-void TransportProxy::SetIceRole(IceRole role) {
-  transport_->get()->SetIceRole(role);
-}
-
-bool TransportProxy::SetLocalTransportDescription(
-    const TransportDescription& description,
-    ContentAction action,
-    std::string* error_desc) {
-  // If this is an answer, finalize the negotiation.
-  if (action == CA_ANSWER) {
-    CompleteNegotiation();
-  }
-  bool result = transport_->get()->SetLocalTransportDescription(description,
-                                                                action,
-                                                                error_desc);
-  if (result)
-    local_description_set_ = true;
-  return result;
-}
-
-bool TransportProxy::SetRemoteTransportDescription(
-    const TransportDescription& description,
-    ContentAction action,
-    std::string* error_desc) {
-  // If this is an answer, finalize the negotiation.
-  if (action == CA_ANSWER) {
-    CompleteNegotiation();
-  }
-  bool result = transport_->get()->SetRemoteTransportDescription(description,
-                                                                 action,
-                                                                 error_desc);
-  if (result)
-    remote_description_set_ = true;
-  return result;
-}
-
-void TransportProxy::OnSignalingReady() {
-  // If we're starting a new allocation sequence, reset our state.
-  set_candidates_allocated(false);
-  transport_->get()->OnSignalingReady();
-}
-
-bool TransportProxy::OnRemoteCandidates(const Candidates& candidates,
-                                        std::string* error) {
-  // Ensure the transport is negotiated before handling candidates.
-  // TODO(juberti): Remove this once everybody calls SetLocalTD.
-  CompleteNegotiation();
-
-  // Verify each candidate before passing down to transport layer.
-  for (Candidates::const_iterator cand = candidates.begin();
-       cand != candidates.end(); ++cand) {
-    if (!transport_->get()->VerifyCandidate(*cand, error))
-      return false;
-    if (!HasChannel(cand->component())) {
-      *error = "Candidate has unknown component: " + cand->ToString() +
-               " for content: " + content_name_;
-      return false;
-    }
-  }
-  transport_->get()->OnRemoteCandidates(candidates);
-  return true;
-}
-
-void TransportProxy::SetIdentity(
-    rtc::SSLIdentity* identity) {
-  transport_->get()->SetIdentity(identity);
-}
-
-std::string BaseSession::StateToString(State state) {
-  switch (state) {
-    case STATE_INIT:
-      return "STATE_INIT";
-    case STATE_SENTINITIATE:
-      return "STATE_SENTINITIATE";
-    case STATE_RECEIVEDINITIATE:
-      return "STATE_RECEIVEDINITIATE";
-    case STATE_SENTPRACCEPT:
-      return "STATE_SENTPRACCEPT";
-    case STATE_SENTACCEPT:
-      return "STATE_SENTACCEPT";
-    case STATE_RECEIVEDPRACCEPT:
-      return "STATE_RECEIVEDPRACCEPT";
-    case STATE_RECEIVEDACCEPT:
-      return "STATE_RECEIVEDACCEPT";
-    case STATE_SENTMODIFY:
-      return "STATE_SENTMODIFY";
-    case STATE_RECEIVEDMODIFY:
-      return "STATE_RECEIVEDMODIFY";
-    case STATE_SENTREJECT:
-      return "STATE_SENTREJECT";
-    case STATE_RECEIVEDREJECT:
-      return "STATE_RECEIVEDREJECT";
-    case STATE_SENTREDIRECT:
-      return "STATE_SENTREDIRECT";
-    case STATE_SENTTERMINATE:
-      return "STATE_SENTTERMINATE";
-    case STATE_RECEIVEDTERMINATE:
-      return "STATE_RECEIVEDTERMINATE";
-    case STATE_INPROGRESS:
-      return "STATE_INPROGRESS";
-    case STATE_DEINIT:
-      return "STATE_DEINIT";
-    default:
-      break;
-  }
-  return "STATE_" + rtc::ToString(state);
-}
-
-BaseSession::BaseSession(rtc::Thread* signaling_thread,
-                         rtc::Thread* worker_thread,
-                         PortAllocator* port_allocator,
-                         const std::string& sid,
-                         const std::string& content_type,
-                         bool initiator)
-    : state_(STATE_INIT),
-      error_(ERROR_NONE),
-      signaling_thread_(signaling_thread),
-      worker_thread_(worker_thread),
-      port_allocator_(port_allocator),
-      sid_(sid),
-      content_type_(content_type),
-      transport_type_(NS_GINGLE_P2P),
-      initiator_(initiator),
-      identity_(NULL),
-      ice_tiebreaker_(rtc::CreateRandomId64()),
-      role_switch_(false) {
-  ASSERT(signaling_thread->IsCurrent());
-}
-
-BaseSession::~BaseSession() {
-  ASSERT(signaling_thread()->IsCurrent());
-
-  ASSERT(state_ != STATE_DEINIT);
-  LogState(state_, STATE_DEINIT);
-  state_ = STATE_DEINIT;
-  SignalState(this, state_);
-
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    delete iter->second;
-  }
-}
-
-const SessionDescription* BaseSession::local_description() const {
-  // TODO(tommi): Assert on thread correctness.
-  return local_description_.get();
-}
-
-const SessionDescription* BaseSession::remote_description() const {
-  // TODO(tommi): Assert on thread correctness.
-  return remote_description_.get();
-}
-
-SessionDescription* BaseSession::remote_description() {
-  // TODO(tommi): Assert on thread correctness.
-  return remote_description_.get();
-}
-
-void BaseSession::set_local_description(const SessionDescription* sdesc) {
-  // TODO(tommi): Assert on thread correctness.
-  if (sdesc != local_description_.get())
-    local_description_.reset(sdesc);
-}
-
-void BaseSession::set_remote_description(SessionDescription* sdesc) {
-  // TODO(tommi): Assert on thread correctness.
-  if (sdesc != remote_description_)
-    remote_description_.reset(sdesc);
-}
-
-const SessionDescription* BaseSession::initiator_description() const {
-  // TODO(tommi): Assert on thread correctness.
-  return initiator_ ? local_description_.get() : remote_description_.get();
-}
-
-bool BaseSession::SetIdentity(rtc::SSLIdentity* identity) {
-  if (identity_)
-    return false;
-  identity_ = identity;
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    iter->second->SetIdentity(identity_);
-  }
-  return true;
-}
-
-bool BaseSession::PushdownTransportDescription(ContentSource source,
-                                               ContentAction action,
-                                               std::string* error_desc) {
-  if (source == CS_LOCAL) {
-    return PushdownLocalTransportDescription(local_description(),
-                                             action,
-                                             error_desc);
-  }
-  return PushdownRemoteTransportDescription(remote_description(),
-                                            action,
-                                            error_desc);
-}
-
-bool BaseSession::PushdownLocalTransportDescription(
-    const SessionDescription* sdesc,
-    ContentAction action,
-    std::string* error_desc) {
-  // Update the Transports with the right information, and trigger them to
-  // start connecting.
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    // If no transport info was in this session description, ret == false
-    // and we just skip this one.
-    TransportDescription tdesc;
-    bool ret = GetTransportDescription(
-        sdesc, iter->second->content_name(), &tdesc);
-    if (ret) {
-      if (!iter->second->SetLocalTransportDescription(tdesc, action,
-                                                      error_desc)) {
-        return false;
-      }
-
-      iter->second->ConnectChannels();
-    }
-  }
-
-  return true;
-}
-
-bool BaseSession::PushdownRemoteTransportDescription(
-    const SessionDescription* sdesc,
-    ContentAction action,
-    std::string* error_desc) {
-  // Update the Transports with the right information.
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    TransportDescription tdesc;
-
-    // If no transport info was in this session description, ret == false
-    // and we just skip this one.
-    bool ret = GetTransportDescription(
-        sdesc, iter->second->content_name(), &tdesc);
-    if (ret) {
-      if (!iter->second->SetRemoteTransportDescription(tdesc, action,
-                                                       error_desc)) {
-        return false;
-      }
-    }
-  }
-
-  return true;
-}
-
-TransportChannel* BaseSession::CreateChannel(const std::string& content_name,
-                                             int component) {
-  // We create the proxy "on demand" here because we need to support
-  // creating channels at any time, even before we send or receive
-  // initiate messages, which is before we create the transports.
-  TransportProxy* transproxy = GetOrCreateTransportProxy(content_name);
-  return transproxy->CreateChannel(component);
-}
-
-TransportChannel* BaseSession::GetChannel(const std::string& content_name,
-                                          int component) {
-  TransportProxy* transproxy = GetTransportProxy(content_name);
-  if (transproxy == NULL)
-    return NULL;
-
-  return transproxy->GetChannel(component);
-}
-
-void BaseSession::DestroyChannel(const std::string& content_name,
-                                 int component) {
-  TransportProxy* transproxy = GetTransportProxy(content_name);
-  ASSERT(transproxy != NULL);
-  transproxy->DestroyChannel(component);
-}
-
-TransportProxy* BaseSession::GetOrCreateTransportProxy(
-    const std::string& content_name) {
-  TransportProxy* transproxy = GetTransportProxy(content_name);
-  if (transproxy)
-    return transproxy;
-
-  Transport* transport = CreateTransport(content_name);
-  transport->SetIceRole(initiator_ ? ICEROLE_CONTROLLING : ICEROLE_CONTROLLED);
-  transport->SetIceTiebreaker(ice_tiebreaker_);
-  // TODO: Connect all the Transport signals to TransportProxy
-  // then to the BaseSession.
-  transport->SignalConnecting.connect(
-      this, &BaseSession::OnTransportConnecting);
-  transport->SignalWritableState.connect(
-      this, &BaseSession::OnTransportWritable);
-  transport->SignalRequestSignaling.connect(
-      this, &BaseSession::OnTransportRequestSignaling);
-  transport->SignalRouteChange.connect(
-      this, &BaseSession::OnTransportRouteChange);
-  transport->SignalCandidatesAllocationDone.connect(
-      this, &BaseSession::OnTransportCandidatesAllocationDone);
-  transport->SignalRoleConflict.connect(
-      this, &BaseSession::OnRoleConflict);
-  transport->SignalCompleted.connect(
-      this, &BaseSession::OnTransportCompleted);
-  transport->SignalFailed.connect(
-      this, &BaseSession::OnTransportFailed);
-
-  transproxy = new TransportProxy(worker_thread_, sid_, content_name,
-                                  new TransportWrapper(transport));
-  transproxy->SignalCandidatesReady.connect(
-      this, &BaseSession::OnTransportProxyCandidatesReady);
-  if (identity_)
-    transproxy->SetIdentity(identity_);
-  transports_[content_name] = transproxy;
-
-  return transproxy;
-}
-
-Transport* BaseSession::GetTransport(const std::string& content_name) {
-  TransportProxy* transproxy = GetTransportProxy(content_name);
-  if (transproxy == NULL)
-    return NULL;
-  return transproxy->impl();
-}
-
-TransportProxy* BaseSession::GetTransportProxy(
-    const std::string& content_name) {
-  TransportMap::iterator iter = transports_.find(content_name);
-  return (iter != transports_.end()) ? iter->second : NULL;
-}
-
-void BaseSession::DestroyTransportProxy(
-    const std::string& content_name) {
-  TransportMap::iterator iter = transports_.find(content_name);
-  if (iter != transports_.end()) {
-    delete iter->second;
-    transports_.erase(content_name);
-  }
-}
-
-cricket::Transport* BaseSession::CreateTransport(
-    const std::string& content_name) {
-  ASSERT(transport_type_ == NS_GINGLE_P2P);
-  return new cricket::DtlsTransport<P2PTransport>(
-      signaling_thread(), worker_thread(), content_name,
-      port_allocator(), identity_);
-}
-
-void BaseSession::SetState(State state) {
-  ASSERT(signaling_thread_->IsCurrent());
-  if (state != state_) {
-    LogState(state_, state);
-    state_ = state;
-    SignalState(this, state_);
-    signaling_thread_->Post(this, MSG_STATE);
-  }
-  SignalNewDescription();
-}
-
-void BaseSession::SetError(Error error, const std::string& error_desc) {
-  ASSERT(signaling_thread_->IsCurrent());
-  if (error != error_) {
-    error_ = error;
-    error_desc_ = error_desc;
-    SignalError(this, error);
-  }
-}
-
-void BaseSession::OnSignalingReady() {
-  ASSERT(signaling_thread()->IsCurrent());
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    iter->second->OnSignalingReady();
-  }
-}
-
-// TODO(juberti): Since PushdownLocalTD now triggers the connection process to
-// start, remove this method once everyone calls PushdownLocalTD.
-void BaseSession::SpeculativelyConnectAllTransportChannels() {
-  // Put all transports into the connecting state.
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    iter->second->ConnectChannels();
-  }
-}
-
-bool BaseSession::OnRemoteCandidates(const std::string& content_name,
-                                     const Candidates& candidates,
-                                     std::string* error) {
-  // Give candidates to the appropriate transport, and tell that transport
-  // to start connecting, if it's not already doing so.
-  TransportProxy* transproxy = GetTransportProxy(content_name);
-  if (!transproxy) {
-    *error = "Unknown content name " + content_name;
-    return false;
-  }
-  if (!transproxy->OnRemoteCandidates(candidates, error)) {
-    return false;
-  }
-  // TODO(juberti): Remove this call once we can be sure that we always have
-  // a local transport description (which will trigger the connection).
-  transproxy->ConnectChannels();
-  return true;
-}
-
-bool BaseSession::MaybeEnableMuxingSupport() {
-  // We need both a local and remote description to decide if we should mux.
-  if ((state_ == STATE_SENTINITIATE ||
-      state_ == STATE_RECEIVEDINITIATE) &&
-      ((local_description_ == NULL) ||
-      (remote_description_ == NULL))) {
-    return false;
-  }
-
-  // In order to perform the multiplexing, we need all proxies to be in the
-  // negotiated state, i.e. to have implementations underneath.
-  // Ensure that this is the case, regardless of whether we are going to mux.
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    ASSERT(iter->second->negotiated());
-    if (!iter->second->negotiated()) {
-      return false;
-    }
-  }
-
-  // If both sides agree to BUNDLE, mux all the specified contents onto the
-  // transport belonging to the first content name in the BUNDLE group.
-  // If the contents are already muxed, this will be a no-op.
-  // TODO(juberti): Should this check that local and remote have configured
-  // BUNDLE the same way?
-  bool candidates_allocated = IsCandidateAllocationDone();
-  const ContentGroup* local_bundle_group =
-      local_description_->GetGroupByName(GROUP_TYPE_BUNDLE);
-  const ContentGroup* remote_bundle_group =
-      remote_description_->GetGroupByName(GROUP_TYPE_BUNDLE);
-  if (local_bundle_group && remote_bundle_group) {
-    if (!BundleContentGroup(local_bundle_group)) {
-      LOG(LS_WARNING) << "Failed to set up BUNDLE";
-      return false;
-    }
-
-    // If we weren't done gathering before, we might be done now, as a result
-    // of enabling mux.
-    if (!candidates_allocated) {
-      MaybeCandidateAllocationDone();
-    }
-  } else {
-    LOG(LS_INFO) << "BUNDLE group missing from remote or local description.";
-  }
-  return true;
-}
-
-bool BaseSession::BundleContentGroup(const ContentGroup* bundle_group) {
-  const std::string* content_name = bundle_group->FirstContentName();
-  if (!content_name) {
-    LOG(LS_INFO) << "No content names specified in BUNDLE group.";
-    return true;
-  }
-
-  const ContentInfo* content =
-      local_description_->GetContentByName(*content_name);
-  if (!content) {
-    LOG(LS_WARNING) << "Content \"" << *content_name
-                    << "\" referenced in BUNDLE group"
-                    << " not present in local description";
-    return false;
-  }
-
-  TransportProxy* selected_proxy = GetTransportProxy(*content_name);
-  if (!selected_proxy) {
-    LOG(LS_WARNING) << "No transport found for content \""
-                    << *content_name << "\".";
-    return false;
-  }
-
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    // If content is part of the mux group, then repoint its proxy at the
-    // transport object that we have chosen to mux onto. If the proxy
-    // is already pointing at the right object, it will be a no-op.
-    if (bundle_group->HasContentName(iter->first) &&
-        !iter->second->SetupMux(selected_proxy)) {
-      LOG(LS_WARNING) << "Failed to bundle " << iter->first << " to "
-                      << *content_name;
-      return false;
-    }
-    LOG(LS_INFO) << "Bundling " << iter->first << " to " << *content_name;
-  }
-
-  return true;
-}
-
-void BaseSession::OnTransportCandidatesAllocationDone(Transport* transport) {
-  // TODO(juberti): This is a clunky way of processing the done signal. Instead,
-  // TransportProxy should receive the done signal directly, set its allocated
-  // flag internally, and then reissue the done signal to Session.
-  // Overall we should make TransportProxy receive *all* the signals from
-  // Transport, since this removes the need to manually iterate over all
-  // the transports, as is needed to make sure signals are handled properly
-  // when BUNDLEing.
-  // TODO(juberti): Per b/7998978, devs and QA are hitting this assert in ways
-  // that make it prohibitively difficult to run dbg builds. Disabled for now.
-  //ASSERT(!IsCandidateAllocationDone());
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    if (iter->second->impl() == transport) {
-      iter->second->set_candidates_allocated(true);
-    }
-  }
-  MaybeCandidateAllocationDone();
-}
-
-bool BaseSession::IsCandidateAllocationDone() const {
-  for (TransportMap::const_iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    if (!iter->second->candidates_allocated()) {
-      LOG(LS_INFO) << "Candidate allocation not done for "
-                   << iter->second->content_name();
-      return false;
-    }
-  }
-  return true;
-}
-
-void BaseSession::MaybeCandidateAllocationDone() {
-  if (IsCandidateAllocationDone()) {
-    LOG(LS_INFO) << "Candidate gathering is complete.";
-    OnCandidatesAllocationDone();
-  }
-}
-
-void BaseSession::OnRoleConflict() {
-  if (role_switch_) {
-    LOG(LS_WARNING) << "Repeat of role conflict signal from Transport.";
-    return;
-  }
-
-  role_switch_ = true;
-  for (TransportMap::iterator iter = transports_.begin();
-       iter != transports_.end(); ++iter) {
-    // Role will be reverse of initial role setting.
-    IceRole role = initiator_ ? ICEROLE_CONTROLLED : ICEROLE_CONTROLLING;
-    iter->second->SetIceRole(role);
-  }
-}
-
-void BaseSession::LogState(State old_state, State new_state) {
-  LOG(LS_INFO) << "Session:" << id()
-               << " Old state:" << StateToString(old_state)
-               << " New state:" << StateToString(new_state)
-               << " Type:" << content_type()
-               << " Transport:" << transport_type();
-}
-
-// static
-bool BaseSession::GetTransportDescription(const SessionDescription* description,
-                                          const std::string& content_name,
-                                          TransportDescription* tdesc) {
-  if (!description || !tdesc) {
-    return false;
-  }
-  const TransportInfo* transport_info =
-      description->GetTransportInfoByName(content_name);
-  if (!transport_info) {
-    return false;
-  }
-  *tdesc = transport_info->description;
-  return true;
-}
-
-void BaseSession::SignalNewDescription() {
-  ContentAction action;
-  ContentSource source;
-  if (!GetContentAction(&action, &source)) {
-    return;
-  }
-  if (source == CS_LOCAL) {
-    SignalNewLocalDescription(this, action);
-  } else {
-    SignalNewRemoteDescription(this, action);
-  }
-}
-
-bool BaseSession::GetContentAction(ContentAction* action,
-                                   ContentSource* source) {
-  switch (state_) {
-    // new local description
-    case STATE_SENTINITIATE:
-      *action = CA_OFFER;
-      *source = CS_LOCAL;
-      break;
-    case STATE_SENTPRACCEPT:
-      *action = CA_PRANSWER;
-      *source = CS_LOCAL;
-      break;
-    case STATE_SENTACCEPT:
-      *action = CA_ANSWER;
-      *source = CS_LOCAL;
-      break;
-    // new remote description
-    case STATE_RECEIVEDINITIATE:
-      *action = CA_OFFER;
-      *source = CS_REMOTE;
-      break;
-    case STATE_RECEIVEDPRACCEPT:
-      *action = CA_PRANSWER;
-      *source = CS_REMOTE;
-      break;
-    case STATE_RECEIVEDACCEPT:
-      *action = CA_ANSWER;
-      *source = CS_REMOTE;
-      break;
-    default:
-      return false;
-  }
-  return true;
-}
-
-void BaseSession::OnMessage(rtc::Message *pmsg) {
-  switch (pmsg->message_id) {
-  case MSG_TIMEOUT:
-    // Session timeout has occured.
-    SetError(ERROR_TIME, "Session timeout has occured.");
-    break;
-
-  case MSG_STATE:
-    switch (state_) {
-    case STATE_SENTACCEPT:
-    case STATE_RECEIVEDACCEPT:
-      SetState(STATE_INPROGRESS);
-      break;
-
-    default:
-      // Explicitly ignoring some states here.
-      break;
-    }
-    break;
-  }
-}
-
-}  // namespace cricket
+// TODO(deadbeef): Remove this file when Chrome build files no longer reference
+// it.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/session.h b/media/webrtc/trunk/webrtc/p2p/base/session.h
index ba4206aeec..a98a5efe13 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/session.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/session.h
@@ -8,452 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_P2P_BASE_SESSION_H_
-#define WEBRTC_P2P_BASE_SESSION_H_
-
-#include <list>
-#include <map>
-#include <string>
-#include <vector>
-
-#include "webrtc/p2p/base/candidate.h"
-#include "webrtc/p2p/base/port.h"
-#include "webrtc/p2p/base/transport.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/socketaddress.h"
-
-namespace cricket {
-
-class BaseSession;
-class P2PTransportChannel;
-class Transport;
-class TransportChannel;
-class TransportChannelProxy;
-class TransportChannelImpl;
-
-typedef rtc::RefCountedObject<rtc::scoped_ptr<Transport> >
-TransportWrapper;
-
-// Bundles a Transport and ChannelMap together. ChannelMap is used to
-// create transport channels before receiving or sending a session
-// initiate, and for speculatively connecting channels.  Previously, a
-// session had one ChannelMap and transport.  Now, with multiple
-// transports per session, we need multiple ChannelMaps as well.
-
-typedef std::map<int, TransportChannelProxy*> ChannelMap;
-
-class TransportProxy : public sigslot::has_slots<> {
- public:
-  TransportProxy(
-      rtc::Thread* worker_thread,
-      const std::string& sid,
-      const std::string& content_name,
-      TransportWrapper* transport)
-      : worker_thread_(worker_thread),
-        sid_(sid),
-        content_name_(content_name),
-        transport_(transport),
-        connecting_(false),
-        negotiated_(false),
-        sent_candidates_(false),
-        candidates_allocated_(false),
-        local_description_set_(false),
-        remote_description_set_(false) {
-    transport_->get()->SignalCandidatesReady.connect(
-        this, &TransportProxy::OnTransportCandidatesReady);
-  }
-  ~TransportProxy();
-
-  const std::string& content_name() const { return content_name_; }
-  // TODO(juberti): It's not good form to expose the object you're wrapping,
-  // since callers can mutate it. Can we make this return a const Transport*?
-  Transport* impl() const { return transport_->get(); }
-
-  const std::string& type() const;
-  bool negotiated() const { return negotiated_; }
-  const Candidates& sent_candidates() const { return sent_candidates_; }
-  const Candidates& unsent_candidates() const { return unsent_candidates_; }
-  bool candidates_allocated() const { return candidates_allocated_; }
-  void set_candidates_allocated(bool allocated) {
-    candidates_allocated_ = allocated;
-  }
-
-  TransportChannel* GetChannel(int component);
-  TransportChannel* CreateChannel(int component);
-  bool HasChannel(int component);
-  void DestroyChannel(int component);
-
-  void AddSentCandidates(const Candidates& candidates);
-  void AddUnsentCandidates(const Candidates& candidates);
-  void ClearSentCandidates() { sent_candidates_.clear(); }
-  void ClearUnsentCandidates() { unsent_candidates_.clear(); }
-
-  // Start the connection process for any channels, creating impls if needed.
-  void ConnectChannels();
-  // Hook up impls to the proxy channels. Doesn't change connect state.
-  void CompleteNegotiation();
-
-  // Mux this proxy onto the specified proxy's transport.
-  bool SetupMux(TransportProxy* proxy);
-
-  // Simple functions that thunk down to the same functions on Transport.
-  void SetIceRole(IceRole role);
-  void SetIdentity(rtc::SSLIdentity* identity);
-  bool SetLocalTransportDescription(const TransportDescription& description,
-                                    ContentAction action,
-                                    std::string* error_desc);
-  bool SetRemoteTransportDescription(const TransportDescription& description,
-                                     ContentAction action,
-                                     std::string* error_desc);
-  void OnSignalingReady();
-  bool OnRemoteCandidates(const Candidates& candidates, std::string* error);
-
-  // Called when a transport signals that it has new candidates.
-  void OnTransportCandidatesReady(cricket::Transport* transport,
-                                  const Candidates& candidates) {
-    SignalCandidatesReady(this, candidates);
-  }
-
-  bool local_description_set() const {
-    return local_description_set_;
-  }
-  bool remote_description_set() const {
-    return remote_description_set_;
-  }
-
-  // Handles sending of ready candidates and receiving of remote candidates.
-  sigslot::signal2<TransportProxy*,
-                         const std::vector<Candidate>&> SignalCandidatesReady;
-
- private:
-  TransportChannelProxy* GetChannelProxy(int component) const;
-
-  // Creates a new channel on the Transport which causes the reference
-  // count to increment.
-  void CreateChannelImpl(int component);
-  void CreateChannelImpl_w(int component);
-
-  // Manipulators of transportchannelimpl in channel proxy.
-  void SetChannelImplFromTransport(TransportChannelProxy* proxy, int component);
-  void SetChannelImplFromTransport_w(TransportChannelProxy* proxy,
-                                     int component);
-  void ReplaceChannelImpl(TransportChannelProxy* proxy,
-                          TransportChannelImpl* impl);
-  void ReplaceChannelImpl_w(TransportChannelProxy* proxy,
-                            TransportChannelImpl* impl);
-
-  rtc::Thread* const worker_thread_;
-  const std::string sid_;
-  const std::string content_name_;
-  rtc::scoped_refptr<TransportWrapper> transport_;
-  bool connecting_;
-  bool negotiated_;
-  ChannelMap channels_;
-  Candidates sent_candidates_;
-  Candidates unsent_candidates_;
-  bool candidates_allocated_;
-  bool local_description_set_;
-  bool remote_description_set_;
-};
-
-typedef std::map<std::string, TransportProxy*> TransportMap;
-
-// Statistics for all the transports of this session.
-typedef std::map<std::string, TransportStats> TransportStatsMap;
-typedef std::map<std::string, std::string> ProxyTransportMap;
-
-// TODO(pthatcher): Think of a better name for this.  We already have
-// a TransportStats in transport.h.  Perhaps TransportsStats?
-struct SessionStats {
-  ProxyTransportMap proxy_to_transport;
-  TransportStatsMap transport_stats;
-};
-
-// A BaseSession manages general session state. This includes negotiation
-// of both the application-level and network-level protocols:  the former
-// defines what will be sent and the latter defines how it will be sent.  Each
-// network-level protocol is represented by a Transport object.  Each Transport
-// participates in the network-level negotiation.  The individual streams of
-// packets are represented by TransportChannels.  The application-level protocol
-// is represented by SessionDecription objects.
-class BaseSession : public sigslot::has_slots<>,
-                    public rtc::MessageHandler {
- public:
-  enum {
-    MSG_TIMEOUT = 0,
-    MSG_ERROR,
-    MSG_STATE,
-  };
-
-  enum State {
-    STATE_INIT = 0,
-    STATE_SENTINITIATE,       // sent initiate, waiting for Accept or Reject
-    STATE_RECEIVEDINITIATE,   // received an initiate. Call Accept or Reject
-    STATE_SENTPRACCEPT,       // sent provisional Accept
-    STATE_SENTACCEPT,         // sent accept. begin connecting transport
-    STATE_RECEIVEDPRACCEPT,   // received provisional Accept, waiting for Accept
-    STATE_RECEIVEDACCEPT,     // received accept. begin connecting transport
-    STATE_SENTMODIFY,         // sent modify, waiting for Accept or Reject
-    STATE_RECEIVEDMODIFY,     // received modify, call Accept or Reject
-    STATE_SENTREJECT,         // sent reject after receiving initiate
-    STATE_RECEIVEDREJECT,     // received reject after sending initiate
-    STATE_SENTREDIRECT,       // sent direct after receiving initiate
-    STATE_SENTTERMINATE,      // sent terminate (any time / either side)
-    STATE_RECEIVEDTERMINATE,  // received terminate (any time / either side)
-    STATE_INPROGRESS,         // session accepted and in progress
-    STATE_DEINIT,             // session is being destroyed
-  };
-
-  enum Error {
-    ERROR_NONE = 0,       // no error
-    ERROR_TIME = 1,       // no response to signaling
-    ERROR_RESPONSE = 2,   // error during signaling
-    ERROR_NETWORK = 3,    // network error, could not allocate network resources
-    ERROR_CONTENT = 4,    // channel errors in SetLocalContent/SetRemoteContent
-    ERROR_TRANSPORT = 5,  // transport error of some kind
-  };
-
-  // Convert State to a readable string.
-  static std::string StateToString(State state);
-
-  BaseSession(rtc::Thread* signaling_thread,
-              rtc::Thread* worker_thread,
-              PortAllocator* port_allocator,
-              const std::string& sid,
-              const std::string& content_type,
-              bool initiator);
-  virtual ~BaseSession();
-
-  // These are const to allow them to be called from const methods.
-  rtc::Thread* signaling_thread() const { return signaling_thread_; }
-  rtc::Thread* worker_thread() const { return worker_thread_; }
-  PortAllocator* port_allocator() const { return port_allocator_; }
-
-  // The ID of this session.
-  const std::string& id() const { return sid_; }
-
-  // TODO(juberti): This data is largely redundant, as it can now be obtained
-  // from local/remote_description(). Remove these functions and members.
-  // Returns the XML namespace identifying the type of this session.
-  const std::string& content_type() const { return content_type_; }
-  // Returns the XML namespace identifying the transport used for this session.
-  const std::string& transport_type() const { return transport_type_; }
-
-  // Indicates whether we initiated this session.
-  bool initiator() const { return initiator_; }
-
-  // Returns the application-level description given by our client.
-  // If we are the recipient, this will be NULL until we send an accept.
-  const SessionDescription* local_description() const;
-
-  // Returns the application-level description given by the other client.
-  // If we are the initiator, this will be NULL until we receive an accept.
-  const SessionDescription* remote_description() const;
-
-  SessionDescription* remote_description();
-
-  // Takes ownership of SessionDescription*
-  void set_local_description(const SessionDescription* sdesc);
-
-  // Takes ownership of SessionDescription*
-  void set_remote_description(SessionDescription* sdesc);
-
-  const SessionDescription* initiator_description() const;
-
-  // Returns the current state of the session.  See the enum above for details.
-  // Each time the state changes, we will fire this signal.
-  State state() const { return state_; }
-  sigslot::signal2<BaseSession* , State> SignalState;
-
-  // Returns the last error in the session.  See the enum above for details.
-  // Each time the an error occurs, we will fire this signal.
-  Error error() const { return error_; }
-  const std::string& error_desc() const { return error_desc_; }
-  sigslot::signal2<BaseSession* , Error> SignalError;
-
-  // Updates the state, signaling if necessary.
-  virtual void SetState(State state);
-
-  // Updates the error state, signaling if necessary.
-  // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
-  virtual void SetError(Error error, const std::string& error_desc);
-
-  // Fired when the remote description is updated, with the updated
-  // contents.
-  sigslot::signal2<BaseSession* , const ContentInfos&>
-      SignalRemoteDescriptionUpdate;
-
-  // Fired when SetState is called (regardless if there's a state change), which
-  // indicates the session description might have be updated.
-  sigslot::signal2<BaseSession*, ContentAction> SignalNewLocalDescription;
-
-  // Fired when SetState is called (regardless if there's a state change), which
-  // indicates the session description might have be updated.
-  sigslot::signal2<BaseSession*, ContentAction> SignalNewRemoteDescription;
-
-  // Returns the transport that has been negotiated or NULL if
-  // negotiation is still in progress.
-  virtual Transport* GetTransport(const std::string& content_name);
-
-  // Creates a new channel with the given names.  This method may be called
-  // immediately after creating the session.  However, the actual
-  // implementation may not be fixed until transport negotiation completes.
-  // This will usually be called from the worker thread, but that
-  // shouldn't be an issue since the main thread will be blocked in
-  // Send when doing so.
-  virtual TransportChannel* CreateChannel(const std::string& content_name,
-                                          int component);
-
-  // Returns the channel with the given names.
-  virtual TransportChannel* GetChannel(const std::string& content_name,
-                                       int component);
-
-  // Destroys the channel with the given names.
-  // This will usually be called from the worker thread, but that
-  // shouldn't be an issue since the main thread will be blocked in
-  // Send when doing so.
-  virtual void DestroyChannel(const std::string& content_name,
-                              int component);
-
-  rtc::SSLIdentity* identity() { return identity_; }
-
- protected:
-  // Specifies the identity to use in this session.
-  bool SetIdentity(rtc::SSLIdentity* identity);
-
-  bool PushdownTransportDescription(ContentSource source,
-                                    ContentAction action,
-                                    std::string* error_desc);
-  void set_initiator(bool initiator) { initiator_ = initiator; }
-
-  const TransportMap& transport_proxies() const { return transports_; }
-  // Get a TransportProxy by content_name or transport. NULL if not found.
-  TransportProxy* GetTransportProxy(const std::string& content_name);
-  void DestroyTransportProxy(const std::string& content_name);
-  // TransportProxy is owned by session.  Return proxy just for convenience.
-  TransportProxy* GetOrCreateTransportProxy(const std::string& content_name);
-  // Creates the actual transport object. Overridable for testing.
-  virtual Transport* CreateTransport(const std::string& content_name);
-
-  void OnSignalingReady();
-  void SpeculativelyConnectAllTransportChannels();
-  // Helper method to provide remote candidates to the transport.
-  bool OnRemoteCandidates(const std::string& content_name,
-                          const Candidates& candidates,
-                          std::string* error);
-
-  // This method will mux transport channels by content_name.
-  // First content is used for muxing.
-  bool MaybeEnableMuxingSupport();
-
-  // Called when a transport requests signaling.
-  virtual void OnTransportRequestSignaling(Transport* transport) {
-  }
-
-  // Called when the first channel of a transport begins connecting.  We use
-  // this to start a timer, to make sure that the connection completes in a
-  // reasonable amount of time.
-  virtual void OnTransportConnecting(Transport* transport) {
-  }
-
-  // Called when a transport changes its writable state.  We track this to make
-  // sure that the transport becomes writable within a reasonable amount of
-  // time.  If this does not occur, we signal an error.
-  virtual void OnTransportWritable(Transport* transport) {
-  }
-  virtual void OnTransportReadable(Transport* transport) {
-  }
-
-  // Called when a transport has found its steady-state connections.
-  virtual void OnTransportCompleted(Transport* transport) {
-  }
-
-  // Called when a transport has failed permanently.
-  virtual void OnTransportFailed(Transport* transport) {
-  }
-
-  // Called when a transport signals that it has new candidates.
-  virtual void OnTransportProxyCandidatesReady(TransportProxy* proxy,
-                                               const Candidates& candidates) {
-  }
-
-  virtual void OnTransportRouteChange(
-      Transport* transport,
-      int component,
-      const cricket::Candidate& remote_candidate) {
-  }
-
-  virtual void OnTransportCandidatesAllocationDone(Transport* transport);
-
-  // Called when all transport channels allocated required candidates.
-  // This method should be used as an indication of candidates gathering process
-  // is completed and application can now send local candidates list to remote.
-  virtual void OnCandidatesAllocationDone() {
-  }
-
-  // Handles the ice role change callback from Transport. This must be
-  // propagated to all the transports.
-  virtual void OnRoleConflict();
-
-  // Handles messages posted to us.
-  virtual void OnMessage(rtc::Message *pmsg);
-
- protected:
-  bool IsCandidateAllocationDone() const;
-
-  State state_;
-  Error error_;
-  std::string error_desc_;
-
-  // Fires the new description signal according to the current state.
-  virtual void SignalNewDescription();
-  // This method will delete the Transport and TransportChannelImpls
-  // and replace those with the Transport object of the first
-  // MediaContent in bundle_group.
-  bool BundleContentGroup(const ContentGroup* bundle_group);
-
- private:
-  // Helper methods to push local and remote transport descriptions.
-  bool PushdownLocalTransportDescription(
-      const SessionDescription* sdesc, ContentAction action,
-      std::string* error_desc);
-  bool PushdownRemoteTransportDescription(
-      const SessionDescription* sdesc, ContentAction action,
-      std::string* error_desc);
-
-  void MaybeCandidateAllocationDone();
-
-  // Log session state.
-  void LogState(State old_state, State new_state);
-
-  // Returns true and the TransportInfo of the given |content_name|
-  // from |description|. Returns false if it's not available.
-  static bool GetTransportDescription(const SessionDescription* description,
-                                      const std::string& content_name,
-                                      TransportDescription* info);
-
-  // Gets the ContentAction and ContentSource according to the session state.
-  bool GetContentAction(ContentAction* action, ContentSource* source);
-
-  rtc::Thread* const signaling_thread_;
-  rtc::Thread* const worker_thread_;
-  PortAllocator* const port_allocator_;
-  const std::string sid_;
-  const std::string content_type_;
-  const std::string transport_type_;
-  bool initiator_;
-  rtc::SSLIdentity* identity_;
-  rtc::scoped_ptr<const SessionDescription> local_description_;
-  rtc::scoped_ptr<SessionDescription> remote_description_;
-  uint64 ice_tiebreaker_;
-  // This flag will be set to true after the first role switch. This flag
-  // will enable us to stop any role switch during the call.
-  bool role_switch_;
-  TransportMap transports_;
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_P2P_BASE_SESSION_H_
+// TODO(deadbeef): Remove this file when Chrome build files no longer reference
+// it.
+#error "DONT INCLUDE THIS"
diff --git a/media/webrtc/trunk/webrtc/p2p/base/sessiondescription.h b/media/webrtc/trunk/webrtc/p2p/base/sessiondescription.h
index 1182a67740..7880167569 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/sessiondescription.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/sessiondescription.h
@@ -160,10 +160,15 @@ class SessionDescription {
   // Remove the first group with the same semantics specified by |name|.
   void RemoveGroupByName(const std::string& name);
 
+  // Global attributes.
+  void set_msid_supported(bool supported) { msid_supported_ = supported; }
+  bool msid_supported() const { return msid_supported_; }
+
  private:
   ContentInfos contents_;
   TransportInfos transport_infos_;
   ContentGroups content_groups_;
+  bool msid_supported_ = true;
 };
 
 // Indicates whether a ContentDescription was an offer or an answer, as
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stun.cc b/media/webrtc/trunk/webrtc/p2p/base/stun.cc
index 866621f75d..9c22995755 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stun.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/stun.cc
@@ -38,7 +38,7 @@ const char STUN_ERROR_REASON_SERVER_ERROR[] = "Server Error";
 
 const char TURN_MAGIC_COOKIE_VALUE[] = { '\x72', '\xC6', '\x4B', '\xC6' };
 const char EMPTY_TRANSACTION_ID[] = "0000000000000000";
-const uint32 STUN_FINGERPRINT_XOR_VALUE = 0x5354554E;
+const uint32_t STUN_FINGERPRINT_XOR_VALUE = 0x5354554E;
 
 // StunMessage
 
@@ -82,7 +82,7 @@ bool StunMessage::AddAttribute(StunAttribute* attr) {
   if (attr_length % 4 != 0) {
     attr_length += (4 - (attr_length % 4));
   }
-  length_ += static_cast<uint16>(attr_length + 4);
+  length_ += static_cast<uint16_t>(attr_length + 4);
   return true;
 }
 
@@ -135,7 +135,7 @@ bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
   }
 
   // Getting the message length from the STUN header.
-  uint16 msg_length = rtc::GetBE16(&data[2]);
+  uint16_t msg_length = rtc::GetBE16(&data[2]);
   if (size != (msg_length + kStunHeaderSize)) {
     return false;
   }
@@ -144,7 +144,7 @@ bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
   size_t current_pos = kStunHeaderSize;
   bool has_message_integrity_attr = false;
   while (current_pos < size) {
-    uint16 attr_type, attr_length;
+    uint16_t attr_type, attr_length;
     // Getting attribute type and length.
     attr_type = rtc::GetBE16(&data[current_pos]);
     attr_length = rtc::GetBE16(&data[current_pos + sizeof(attr_type)]);
@@ -187,8 +187,7 @@ bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
     //     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     //     |0 0|     STUN Message Type     |         Message Length        |
     //     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-    rtc::SetBE16(temp_data.get() + 2,
-                       static_cast<uint16>(new_adjusted_len));
+    rtc::SetBE16(temp_data.get() + 2, static_cast<uint16_t>(new_adjusted_len));
   }
 
   char hmac[kStunMessageIntegritySize];
@@ -262,12 +261,12 @@ bool StunMessage::ValidateFingerprint(const char* data, size_t size) {
   // Check the fingerprint type and length.
   const char* fingerprint_attr_data = data + size - fingerprint_attr_size;
   if (rtc::GetBE16(fingerprint_attr_data) != STUN_ATTR_FINGERPRINT ||
-      rtc::GetBE16(fingerprint_attr_data + sizeof(uint16)) !=
+      rtc::GetBE16(fingerprint_attr_data + sizeof(uint16_t)) !=
           StunUInt32Attribute::SIZE)
     return false;
 
   // Check the fingerprint value.
-  uint32 fingerprint =
+  uint32_t fingerprint =
       rtc::GetBE32(fingerprint_attr_data + kStunAttributeHeaderSize);
   return ((fingerprint ^ STUN_FINGERPRINT_XOR_VALUE) ==
       rtc::ComputeCrc32(data, size - fingerprint_attr_size));
@@ -287,7 +286,7 @@ bool StunMessage::AddFingerprint() {
 
   int msg_len_for_crc32 = static_cast<int>(
       buf.Length() - kStunAttributeHeaderSize - fingerprint_attr->length());
-  uint32 c = rtc::ComputeCrc32(buf.Data(), msg_len_for_crc32);
+  uint32_t c = rtc::ComputeCrc32(buf.Data(), msg_len_for_crc32);
 
   // Insert the correct CRC-32, XORed with a constant, into the attribute.
   fingerprint_attr->SetValue(c ^ STUN_FINGERPRINT_XOR_VALUE);
@@ -315,8 +314,8 @@ bool StunMessage::Read(ByteBuffer* buf) {
   if (!buf->ReadString(&transaction_id, kStunTransactionIdLength))
     return false;
 
-  uint32 magic_cookie_int =
-      *reinterpret_cast<const uint32*>(magic_cookie.data());
+  uint32_t magic_cookie_int =
+      *reinterpret_cast<const uint32_t*>(magic_cookie.data());
   if (rtc::NetworkToHost32(magic_cookie_int) != kStunMagicCookie) {
     // If magic cookie is invalid it means that the peer implements
     // RFC3489 instead of RFC5389.
@@ -332,7 +331,7 @@ bool StunMessage::Read(ByteBuffer* buf) {
 
   size_t rest = buf->Length() - length_;
   while (buf->Length() > rest) {
-    uint16 attr_type, attr_length;
+    uint16_t attr_type, attr_length;
     if (!buf->ReadUInt16(&attr_type))
       return false;
     if (!buf->ReadUInt16(&attr_length))
@@ -366,7 +365,7 @@ bool StunMessage::Write(ByteBuffer* buf) const {
 
   for (size_t i = 0; i < attrs_->size(); ++i) {
     buf->WriteUInt16((*attrs_)[i]->type());
-    buf->WriteUInt16(static_cast<uint16>((*attrs_)[i]->length()));
+    buf->WriteUInt16(static_cast<uint16_t>((*attrs_)[i]->length()));
     if (!(*attrs_)[i]->Write(buf))
       return false;
   }
@@ -395,8 +394,8 @@ StunAttributeValueType StunMessage::GetAttributeValueType(int type) const {
 
 StunAttribute* StunMessage::CreateAttribute(int type, size_t length) /*const*/ {
   StunAttributeValueType value_type = GetAttributeValueType(type);
-  return StunAttribute::Create(value_type, type,
-                               static_cast<uint16>(length), this);
+  return StunAttribute::Create(value_type, type, static_cast<uint16_t>(length),
+                               this);
 }
 
 const StunAttribute* StunMessage::GetAttribute(int type) const {
@@ -414,7 +413,7 @@ bool StunMessage::IsValidTransactionId(const std::string& transaction_id) {
 
 // StunAttribute
 
-StunAttribute::StunAttribute(uint16 type, uint16 length)
+StunAttribute::StunAttribute(uint16_t type, uint16_t length)
     : type_(type), length_(length) {
 }
 
@@ -434,7 +433,8 @@ void StunAttribute::WritePadding(rtc::ByteBuffer* buf) const {
 }
 
 StunAttribute* StunAttribute::Create(StunAttributeValueType value_type,
-                                     uint16 type, uint16 length,
+                                     uint16_t type,
+                                     uint16_t length,
                                      StunMessage* owner) {
   switch (value_type) {
     case STUN_VALUE_ADDRESS:
@@ -456,23 +456,23 @@ StunAttribute* StunAttribute::Create(StunAttributeValueType value_type,
   }
 }
 
-StunAddressAttribute* StunAttribute::CreateAddress(uint16 type) {
+StunAddressAttribute* StunAttribute::CreateAddress(uint16_t type) {
   return new StunAddressAttribute(type, 0);
 }
 
-StunXorAddressAttribute* StunAttribute::CreateXorAddress(uint16 type) {
+StunXorAddressAttribute* StunAttribute::CreateXorAddress(uint16_t type) {
   return new StunXorAddressAttribute(type, 0, NULL);
 }
 
-StunUInt64Attribute* StunAttribute::CreateUInt64(uint16 type) {
+StunUInt64Attribute* StunAttribute::CreateUInt64(uint16_t type) {
   return new StunUInt64Attribute(type);
 }
 
-StunUInt32Attribute* StunAttribute::CreateUInt32(uint16 type) {
+StunUInt32Attribute* StunAttribute::CreateUInt32(uint16_t type) {
   return new StunUInt32Attribute(type);
 }
 
-StunByteStringAttribute* StunAttribute::CreateByteString(uint16 type) {
+StunByteStringAttribute* StunAttribute::CreateByteString(uint16_t type) {
   return new StunByteStringAttribute(type, 0);
 }
 
@@ -485,26 +485,26 @@ StunUInt16ListAttribute* StunAttribute::CreateUnknownAttributes() {
   return new StunUInt16ListAttribute(STUN_ATTR_UNKNOWN_ATTRIBUTES, 0);
 }
 
-StunAddressAttribute::StunAddressAttribute(uint16 type,
-   const rtc::SocketAddress& addr)
-   : StunAttribute(type, 0) {
+StunAddressAttribute::StunAddressAttribute(uint16_t type,
+                                           const rtc::SocketAddress& addr)
+    : StunAttribute(type, 0) {
   SetAddress(addr);
 }
 
-StunAddressAttribute::StunAddressAttribute(uint16 type, uint16 length)
+StunAddressAttribute::StunAddressAttribute(uint16_t type, uint16_t length)
     : StunAttribute(type, length) {
 }
 
 bool StunAddressAttribute::Read(ByteBuffer* buf) {
-  uint8 dummy;
+  uint8_t dummy;
   if (!buf->ReadUInt8(&dummy))
     return false;
 
-  uint8 stun_family;
+  uint8_t stun_family;
   if (!buf->ReadUInt8(&stun_family)) {
     return false;
   }
-  uint16 port;
+  uint16_t port;
   if (!buf->ReadUInt16(&port))
     return false;
   if (stun_family == STUN_ADDRESS_IPV4) {
@@ -557,15 +557,16 @@ bool StunAddressAttribute::Write(ByteBuffer* buf) const {
   return true;
 }
 
-StunXorAddressAttribute::StunXorAddressAttribute(uint16 type,
-    const rtc::SocketAddress& addr)
+StunXorAddressAttribute::StunXorAddressAttribute(uint16_t type,
+                                                 const rtc::SocketAddress& addr)
     : StunAddressAttribute(type, addr), owner_(NULL) {
 }
 
-StunXorAddressAttribute::StunXorAddressAttribute(uint16 type,
-                                                 uint16 length,
+StunXorAddressAttribute::StunXorAddressAttribute(uint16_t type,
+                                                 uint16_t length,
                                                  StunMessage* owner)
-    : StunAddressAttribute(type, length), owner_(owner) {}
+    : StunAddressAttribute(type, length), owner_(owner) {
+}
 
 rtc::IPAddress StunXorAddressAttribute::GetXoredIP() const {
   if (owner_) {
@@ -581,10 +582,10 @@ rtc::IPAddress StunXorAddressAttribute::GetXoredIP() const {
         in6_addr v6addr = ip.ipv6_address();
         const std::string& transaction_id = owner_->transaction_id();
         if (transaction_id.length() == kStunTransactionIdLength) {
-          uint32 transactionid_as_ints[3];
+          uint32_t transactionid_as_ints[3];
           memcpy(&transactionid_as_ints[0], transaction_id.c_str(),
                  transaction_id.length());
-          uint32* ip_as_ints = reinterpret_cast<uint32*>(&v6addr.s6_addr);
+          uint32_t* ip_as_ints = reinterpret_cast<uint32_t*>(&v6addr.s6_addr);
           // Transaction ID is in network byte order, but magic cookie
           // is stored in host byte order.
           ip_as_ints[0] =
@@ -606,7 +607,7 @@ rtc::IPAddress StunXorAddressAttribute::GetXoredIP() const {
 bool StunXorAddressAttribute::Read(ByteBuffer* buf) {
   if (!StunAddressAttribute::Read(buf))
     return false;
-  uint16 xoredport = port() ^ (kStunMagicCookie >> 16);
+  uint16_t xoredport = port() ^ (kStunMagicCookie >> 16);
   rtc::IPAddress xored_ip = GetXoredIP();
   SetAddress(rtc::SocketAddress(xored_ip, xoredport));
   return true;
@@ -640,11 +641,11 @@ bool StunXorAddressAttribute::Write(ByteBuffer* buf) const {
   return true;
 }
 
-StunUInt32Attribute::StunUInt32Attribute(uint16 type, uint32 value)
+StunUInt32Attribute::StunUInt32Attribute(uint16_t type, uint32_t value)
     : StunAttribute(type, SIZE), bits_(value) {
 }
 
-StunUInt32Attribute::StunUInt32Attribute(uint16 type)
+StunUInt32Attribute::StunUInt32Attribute(uint16_t type)
     : StunAttribute(type, SIZE), bits_(0) {
 }
 
@@ -670,11 +671,11 @@ bool StunUInt32Attribute::Write(ByteBuffer* buf) const {
   return true;
 }
 
-StunUInt64Attribute::StunUInt64Attribute(uint16 type, uint64 value)
+StunUInt64Attribute::StunUInt64Attribute(uint16_t type, uint64_t value)
     : StunAttribute(type, SIZE), bits_(value) {
 }
 
-StunUInt64Attribute::StunUInt64Attribute(uint16 type)
+StunUInt64Attribute::StunUInt64Attribute(uint16_t type)
     : StunAttribute(type, SIZE), bits_(0) {
 }
 
@@ -689,24 +690,24 @@ bool StunUInt64Attribute::Write(ByteBuffer* buf) const {
   return true;
 }
 
-StunByteStringAttribute::StunByteStringAttribute(uint16 type)
+StunByteStringAttribute::StunByteStringAttribute(uint16_t type)
     : StunAttribute(type, 0), bytes_(NULL) {
 }
 
-StunByteStringAttribute::StunByteStringAttribute(uint16 type,
+StunByteStringAttribute::StunByteStringAttribute(uint16_t type,
                                                  const std::string& str)
     : StunAttribute(type, 0), bytes_(NULL) {
   CopyBytes(str.c_str(), str.size());
 }
 
-StunByteStringAttribute::StunByteStringAttribute(uint16 type,
+StunByteStringAttribute::StunByteStringAttribute(uint16_t type,
                                                  const void* bytes,
                                                  size_t length)
     : StunAttribute(type, 0), bytes_(NULL) {
   CopyBytes(bytes, length);
 }
 
-StunByteStringAttribute::StunByteStringAttribute(uint16 type, uint16 length)
+StunByteStringAttribute::StunByteStringAttribute(uint16_t type, uint16_t length)
     : StunAttribute(type, length), bytes_(NULL) {
 }
 
@@ -724,13 +725,13 @@ void StunByteStringAttribute::CopyBytes(const void* bytes, size_t length) {
   SetBytes(new_bytes, length);
 }
 
-uint8 StunByteStringAttribute::GetByte(size_t index) const {
+uint8_t StunByteStringAttribute::GetByte(size_t index) const {
   ASSERT(bytes_ != NULL);
   ASSERT(index < length());
-  return static_cast<uint8>(bytes_[index]);
+  return static_cast<uint8_t>(bytes_[index]);
 }
 
-void StunByteStringAttribute::SetByte(size_t index, uint8 value) {
+void StunByteStringAttribute::SetByte(size_t index, uint8_t value) {
   ASSERT(bytes_ != NULL);
   ASSERT(index < length());
   bytes_[index] = value;
@@ -755,17 +756,18 @@ bool StunByteStringAttribute::Write(ByteBuffer* buf) const {
 void StunByteStringAttribute::SetBytes(char* bytes, size_t length) {
   delete [] bytes_;
   bytes_ = bytes;
-  SetLength(static_cast<uint16>(length));
+  SetLength(static_cast<uint16_t>(length));
 }
 
-StunErrorCodeAttribute::StunErrorCodeAttribute(uint16 type, int code,
+StunErrorCodeAttribute::StunErrorCodeAttribute(uint16_t type,
+                                               int code,
                                                const std::string& reason)
     : StunAttribute(type, 0) {
   SetCode(code);
   SetReason(reason);
 }
 
-StunErrorCodeAttribute::StunErrorCodeAttribute(uint16 type, uint16 length)
+StunErrorCodeAttribute::StunErrorCodeAttribute(uint16_t type, uint16_t length)
     : StunAttribute(type, length), class_(0), number_(0) {
 }
 
@@ -777,17 +779,17 @@ int StunErrorCodeAttribute::code() const {
 }
 
 void StunErrorCodeAttribute::SetCode(int code) {
-  class_ = static_cast<uint8>(code / 100);
-  number_ = static_cast<uint8>(code % 100);
+  class_ = static_cast<uint8_t>(code / 100);
+  number_ = static_cast<uint8_t>(code % 100);
 }
 
 void StunErrorCodeAttribute::SetReason(const std::string& reason) {
-  SetLength(MIN_SIZE + static_cast<uint16>(reason.size()));
+  SetLength(MIN_SIZE + static_cast<uint16_t>(reason.size()));
   reason_ = reason;
 }
 
 bool StunErrorCodeAttribute::Read(ByteBuffer* buf) {
-  uint32 val;
+  uint32_t val;
   if (length() < MIN_SIZE || !buf->ReadUInt32(&val))
     return false;
 
@@ -811,9 +813,9 @@ bool StunErrorCodeAttribute::Write(ByteBuffer* buf) const {
   return true;
 }
 
-StunUInt16ListAttribute::StunUInt16ListAttribute(uint16 type, uint16 length)
+StunUInt16ListAttribute::StunUInt16ListAttribute(uint16_t type, uint16_t length)
     : StunAttribute(type, length) {
-  attr_types_ = new std::vector<uint16>();
+  attr_types_ = new std::vector<uint16_t>();
 }
 
 StunUInt16ListAttribute::~StunUInt16ListAttribute() {
@@ -824,17 +826,17 @@ size_t StunUInt16ListAttribute::Size() const {
   return attr_types_->size();
 }
 
-uint16 StunUInt16ListAttribute::GetType(int index) const {
+uint16_t StunUInt16ListAttribute::GetType(int index) const {
   return (*attr_types_)[index];
 }
 
-void StunUInt16ListAttribute::SetType(int index, uint16 value) {
+void StunUInt16ListAttribute::SetType(int index, uint16_t value) {
   (*attr_types_)[index] = value;
 }
 
-void StunUInt16ListAttribute::AddType(uint16 value) {
+void StunUInt16ListAttribute::AddType(uint16_t value) {
   attr_types_->push_back(value);
-  SetLength(static_cast<uint16>(attr_types_->size() * 2));
+  SetLength(static_cast<uint16_t>(attr_types_->size() * 2));
 }
 
 bool StunUInt16ListAttribute::Read(ByteBuffer* buf) {
@@ -842,7 +844,7 @@ bool StunUInt16ListAttribute::Read(ByteBuffer* buf) {
     return false;
 
   for (size_t i = 0; i < length() / 2; i++) {
-    uint16 attr;
+    uint16_t attr;
     if (!buf->ReadUInt16(&attr))
       return false;
     attr_types_->push_back(attr);
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stun.h b/media/webrtc/trunk/webrtc/p2p/base/stun.h
index 4bf6547885..75b89afb8a 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stun.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/stun.h
@@ -97,7 +97,7 @@ extern const char STUN_ERROR_REASON_STALE_NONCE[];
 extern const char STUN_ERROR_REASON_SERVER_ERROR[];
 
 // The mask used to determine whether a STUN message is a request/response etc.
-const uint32 kStunTypeMask = 0x0110;
+const uint32_t kStunTypeMask = 0x0110;
 
 // STUN Attribute header length.
 const size_t kStunAttributeHeaderSize = 4;
@@ -106,7 +106,7 @@ const size_t kStunAttributeHeaderSize = 4;
 const size_t kStunHeaderSize = 20;
 const size_t kStunTransactionIdOffset = 8;
 const size_t kStunTransactionIdLength = 12;
-const uint32 kStunMagicCookie = 0x2112A442;
+const uint32_t kStunMagicCookie = 0x2112A442;
 const size_t kStunMagicCookieLength = sizeof(kStunMagicCookie);
 
 // Following value corresponds to an earlier version of STUN from
@@ -145,7 +145,7 @@ class StunMessage {
   // is determined by the lengths of the transaction ID.
   bool IsLegacy() const;
 
-  void SetType(int type) { type_ = static_cast<uint16>(type); }
+  void SetType(int type) { type_ = static_cast<uint16_t>(type); }
   bool SetTransactionID(const std::string& str);
 
   // Gets the desired attribute value, or NULL if no such attribute type exists.
@@ -198,8 +198,8 @@ class StunMessage {
   const StunAttribute* GetAttribute(int type) const;
   static bool IsValidTransactionId(const std::string& transaction_id);
 
-  uint16 type_;
-  uint16 length_;
+  uint16_t type_;
+  uint16_t length_;
   std::string transaction_id_;
   std::vector<StunAttribute*>* attrs_;
 };
@@ -228,37 +228,39 @@ class StunAttribute {
   virtual bool Write(rtc::ByteBuffer* buf) const = 0;
 
   // Creates an attribute object with the given type and smallest length.
-  static StunAttribute* Create(StunAttributeValueType value_type, uint16 type,
-                               uint16 length, StunMessage* owner);
+  static StunAttribute* Create(StunAttributeValueType value_type,
+                               uint16_t type,
+                               uint16_t length,
+                               StunMessage* owner);
   // TODO: Allow these create functions to take parameters, to reduce
   // the amount of work callers need to do to initialize attributes.
-  static StunAddressAttribute* CreateAddress(uint16 type);
-  static StunXorAddressAttribute* CreateXorAddress(uint16 type);
-  static StunUInt32Attribute* CreateUInt32(uint16 type);
-  static StunUInt64Attribute* CreateUInt64(uint16 type);
-  static StunByteStringAttribute* CreateByteString(uint16 type);
+  static StunAddressAttribute* CreateAddress(uint16_t type);
+  static StunXorAddressAttribute* CreateXorAddress(uint16_t type);
+  static StunUInt32Attribute* CreateUInt32(uint16_t type);
+  static StunUInt64Attribute* CreateUInt64(uint16_t type);
+  static StunByteStringAttribute* CreateByteString(uint16_t type);
   static StunErrorCodeAttribute* CreateErrorCode();
   static StunUInt16ListAttribute* CreateUnknownAttributes();
 
  protected:
-  StunAttribute(uint16 type, uint16 length);
-  void SetLength(uint16 length) { length_ = length; }
+  StunAttribute(uint16_t type, uint16_t length);
+  void SetLength(uint16_t length) { length_ = length; }
   void WritePadding(rtc::ByteBuffer* buf) const;
   void ConsumePadding(rtc::ByteBuffer* buf) const;
 
  private:
-  uint16 type_;
-  uint16 length_;
+  uint16_t type_;
+  uint16_t length_;
 };
 
 // Implements STUN attributes that record an Internet address.
 class StunAddressAttribute : public StunAttribute {
  public:
-  static const uint16 SIZE_UNDEF = 0;
-  static const uint16 SIZE_IP4 = 8;
-  static const uint16 SIZE_IP6 = 20;
-  StunAddressAttribute(uint16 type, const rtc::SocketAddress& addr);
-  StunAddressAttribute(uint16 type, uint16 length);
+  static const uint16_t SIZE_UNDEF = 0;
+  static const uint16_t SIZE_IP4 = 8;
+  static const uint16_t SIZE_IP6 = 20;
+  StunAddressAttribute(uint16_t type, const rtc::SocketAddress& addr);
+  StunAddressAttribute(uint16_t type, uint16_t length);
 
   virtual StunAttributeValueType value_type() const {
     return STUN_VALUE_ADDRESS;
@@ -276,7 +278,7 @@ class StunAddressAttribute : public StunAttribute {
 
   const rtc::SocketAddress& GetAddress() const { return address_; }
   const rtc::IPAddress& ipaddr() const { return address_.ipaddr(); }
-  uint16 port() const { return address_.port(); }
+  uint16_t port() const { return address_.port(); }
 
   void SetAddress(const rtc::SocketAddress& addr) {
     address_ = addr;
@@ -286,7 +288,7 @@ class StunAddressAttribute : public StunAttribute {
     address_.SetIP(ip);
     EnsureAddressLength();
   }
-  void SetPort(uint16 port) { address_.SetPort(port); }
+  void SetPort(uint16_t port) { address_.SetPort(port); }
 
   virtual bool Read(rtc::ByteBuffer* buf);
   virtual bool Write(rtc::ByteBuffer* buf) const;
@@ -316,9 +318,8 @@ class StunAddressAttribute : public StunAttribute {
 // transaction ID of the message.
 class StunXorAddressAttribute : public StunAddressAttribute {
  public:
-  StunXorAddressAttribute(uint16 type, const rtc::SocketAddress& addr);
-  StunXorAddressAttribute(uint16 type, uint16 length,
-                          StunMessage* owner);
+  StunXorAddressAttribute(uint16_t type, const rtc::SocketAddress& addr);
+  StunXorAddressAttribute(uint16_t type, uint16_t length, StunMessage* owner);
 
   virtual StunAttributeValueType value_type() const {
     return STUN_VALUE_XOR_ADDRESS;
@@ -337,16 +338,16 @@ class StunXorAddressAttribute : public StunAddressAttribute {
 // Implements STUN attributes that record a 32-bit integer.
 class StunUInt32Attribute : public StunAttribute {
  public:
-  static const uint16 SIZE = 4;
-  StunUInt32Attribute(uint16 type, uint32 value);
-  explicit StunUInt32Attribute(uint16 type);
+  static const uint16_t SIZE = 4;
+  StunUInt32Attribute(uint16_t type, uint32_t value);
+  explicit StunUInt32Attribute(uint16_t type);
 
   virtual StunAttributeValueType value_type() const {
     return STUN_VALUE_UINT32;
   }
 
-  uint32 value() const { return bits_; }
-  void SetValue(uint32 bits) { bits_ = bits; }
+  uint32_t value() const { return bits_; }
+  void SetValue(uint32_t bits) { bits_ = bits; }
 
   bool GetBit(size_t index) const;
   void SetBit(size_t index, bool value);
@@ -355,36 +356,36 @@ class StunUInt32Attribute : public StunAttribute {
   virtual bool Write(rtc::ByteBuffer* buf) const;
 
  private:
-  uint32 bits_;
+  uint32_t bits_;
 };
 
 class StunUInt64Attribute : public StunAttribute {
  public:
-  static const uint16 SIZE = 8;
-  StunUInt64Attribute(uint16 type, uint64 value);
-  explicit StunUInt64Attribute(uint16 type);
+  static const uint16_t SIZE = 8;
+  StunUInt64Attribute(uint16_t type, uint64_t value);
+  explicit StunUInt64Attribute(uint16_t type);
 
   virtual StunAttributeValueType value_type() const {
     return STUN_VALUE_UINT64;
   }
 
-  uint64 value() const { return bits_; }
-  void SetValue(uint64 bits) { bits_ = bits; }
+  uint64_t value() const { return bits_; }
+  void SetValue(uint64_t bits) { bits_ = bits; }
 
   virtual bool Read(rtc::ByteBuffer* buf);
   virtual bool Write(rtc::ByteBuffer* buf) const;
 
  private:
-  uint64 bits_;
+  uint64_t bits_;
 };
 
 // Implements STUN attributes that record an arbitrary byte string.
 class StunByteStringAttribute : public StunAttribute {
  public:
-  explicit StunByteStringAttribute(uint16 type);
-  StunByteStringAttribute(uint16 type, const std::string& str);
-  StunByteStringAttribute(uint16 type, const void* bytes, size_t length);
-  StunByteStringAttribute(uint16 type, uint16 length);
+  explicit StunByteStringAttribute(uint16_t type);
+  StunByteStringAttribute(uint16_t type, const std::string& str);
+  StunByteStringAttribute(uint16_t type, const void* bytes, size_t length);
+  StunByteStringAttribute(uint16_t type, uint16_t length);
   ~StunByteStringAttribute();
 
   virtual StunAttributeValueType value_type() const {
@@ -397,8 +398,8 @@ class StunByteStringAttribute : public StunAttribute {
   void CopyBytes(const char* bytes);  // uses strlen
   void CopyBytes(const void* bytes, size_t length);
 
-  uint8 GetByte(size_t index) const;
-  void SetByte(size_t index, uint8 value);
+  uint8_t GetByte(size_t index) const;
+  void SetByte(size_t index, uint8_t value);
 
   virtual bool Read(rtc::ByteBuffer* buf);
   virtual bool Write(rtc::ByteBuffer* buf) const;
@@ -412,9 +413,9 @@ class StunByteStringAttribute : public StunAttribute {
 // Implements STUN attributes that record an error code.
 class StunErrorCodeAttribute : public StunAttribute {
  public:
-  static const uint16 MIN_SIZE = 4;
-  StunErrorCodeAttribute(uint16 type, int code, const std::string& reason);
-  StunErrorCodeAttribute(uint16 type, uint16 length);
+  static const uint16_t MIN_SIZE = 4;
+  StunErrorCodeAttribute(uint16_t type, int code, const std::string& reason);
+  StunErrorCodeAttribute(uint16_t type, uint16_t length);
   ~StunErrorCodeAttribute();
 
   virtual StunAttributeValueType value_type() const {
@@ -429,23 +430,23 @@ class StunErrorCodeAttribute : public StunAttribute {
   int eclass() const { return class_; }
   int number() const { return number_; }
   const std::string& reason() const { return reason_; }
-  void SetClass(uint8 eclass) { class_ = eclass; }
-  void SetNumber(uint8 number) { number_ = number; }
+  void SetClass(uint8_t eclass) { class_ = eclass; }
+  void SetNumber(uint8_t number) { number_ = number; }
   void SetReason(const std::string& reason);
 
   bool Read(rtc::ByteBuffer* buf);
   bool Write(rtc::ByteBuffer* buf) const;
 
  private:
-  uint8 class_;
-  uint8 number_;
+  uint8_t class_;
+  uint8_t number_;
   std::string reason_;
 };
 
 // Implements STUN attributes that record a list of attribute names.
 class StunUInt16ListAttribute : public StunAttribute {
  public:
-  StunUInt16ListAttribute(uint16 type, uint16 length);
+  StunUInt16ListAttribute(uint16_t type, uint16_t length);
   ~StunUInt16ListAttribute();
 
   virtual StunAttributeValueType value_type() const {
@@ -453,15 +454,15 @@ class StunUInt16ListAttribute : public StunAttribute {
   }
 
   size_t Size() const;
-  uint16 GetType(int index) const;
-  void SetType(int index, uint16 value);
-  void AddType(uint16 value);
+  uint16_t GetType(int index) const;
+  void SetType(int index, uint16_t value);
+  void AddType(uint16_t value);
 
   bool Read(rtc::ByteBuffer* buf);
   bool Write(rtc::ByteBuffer* buf) const;
 
  private:
-  std::vector<uint16>* attr_types_;
+  std::vector<uint16_t>* attr_types_;
 };
 
 // Returns the (successful) response type for the given request type.
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stun_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/stun_unittest.cc
index 9d5779d7d6..12492570c4 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stun_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/stun_unittest.cc
@@ -11,6 +11,7 @@
 #include <string>
 
 #include "webrtc/p2p/base/stun.h"
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/bytebuffer.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
@@ -165,7 +166,7 @@ static const unsigned char kStunMessageWithPaddedByteStringAttribute[] = {
   0x61, 0x62, 0x63, 0xcc   // abc
 };
 
-// Message with an Unknown Attributes (uint16 list) attribute.
+// Message with an Unknown Attributes (uint16_t list) attribute.
 static const unsigned char kStunMessageWithUInt16ListAttribute[] = {
   0x00, 0x01, 0x00, 0x0c,
   0x21, 0x12, 0xa4, 0x42,
@@ -515,11 +516,11 @@ TEST_F(StunTest, MessageTypes) {
     STUN_BINDING_REQUEST, STUN_BINDING_INDICATION,
     STUN_BINDING_RESPONSE, STUN_BINDING_ERROR_RESPONSE
   };
-  for (int i = 0; i < ARRAY_SIZE(types); ++i) {
-    EXPECT_EQ(i == 0, IsStunRequestType(types[i]));
-    EXPECT_EQ(i == 1, IsStunIndicationType(types[i]));
-    EXPECT_EQ(i == 2, IsStunSuccessResponseType(types[i]));
-    EXPECT_EQ(i == 3, IsStunErrorResponseType(types[i]));
+  for (size_t i = 0; i < arraysize(types); ++i) {
+    EXPECT_EQ(i == 0U, IsStunRequestType(types[i]));
+    EXPECT_EQ(i == 1U, IsStunIndicationType(types[i]));
+    EXPECT_EQ(i == 2U, IsStunSuccessResponseType(types[i]));
+    EXPECT_EQ(i == 3U, IsStunErrorResponseType(types[i]));
     EXPECT_EQ(1, types[i] & 0xFEEF);
   }
 }
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stunport.cc b/media/webrtc/trunk/webrtc/p2p/base/stunport.cc
index fe125ec2bf..8f37dd5218 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stunport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/stunport.cc
@@ -13,8 +13,10 @@
 #include "webrtc/p2p/base/common.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/stun.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
+#include "webrtc/base/ipaddress.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/nethelpers.h"
 
@@ -22,15 +24,19 @@ namespace cricket {
 
 // TODO: Move these to a common place (used in relayport too)
 const int KEEPALIVE_DELAY = 10 * 1000;  // 10 seconds - sort timeouts
-const int RETRY_DELAY = 50;             // 50ms, from ICE spec
 const int RETRY_TIMEOUT = 50 * 1000;    // ICE says 50 secs
+// Stop sending STUN binding requests after this amount of time
+// (in milliseconds) because the connection binding requests should keep
+// the NAT binding alive.
+const int KEEP_ALIVE_TIMEOUT = 2 * 60 * 1000;  // 2 minutes
 
 // Handles a binding request sent to the STUN server.
 class StunBindingRequest : public StunRequest {
  public:
-  StunBindingRequest(UDPPort* port, bool keep_alive,
-                     const rtc::SocketAddress& addr)
-    : port_(port), keep_alive_(keep_alive), server_addr_(addr) {
+  StunBindingRequest(UDPPort* port,
+                     const rtc::SocketAddress& addr,
+                     uint32_t deadline)
+      : port_(port), server_addr_(addr), deadline_(deadline) {
     start_time_ = rtc::Time();
   }
 
@@ -39,11 +45,11 @@ class StunBindingRequest : public StunRequest {
 
   const rtc::SocketAddress& server_addr() const { return server_addr_; }
 
-  virtual void Prepare(StunMessage* request) {
+  virtual void Prepare(StunMessage* request) override {
     request->SetType(STUN_BINDING_REQUEST);
   }
 
-  virtual void OnResponse(StunMessage* response) {
+  virtual void OnResponse(StunMessage* response) override {
     const StunAddressAttribute* addr_attr =
         response->GetAddress(STUN_ATTR_MAPPED_ADDRESS);
     if (!addr_attr) {
@@ -57,15 +63,15 @@ class StunBindingRequest : public StunRequest {
     }
 
     // We will do a keep-alive regardless of whether this request succeeds.
-    // This should have almost no impact on network usage.
-    if (keep_alive_) {
+    // It will be stopped after |deadline_| mostly to conserve the battery life.
+    if (rtc::Time() <= deadline_) {
       port_->requests_.SendDelayed(
-          new StunBindingRequest(port_, true, server_addr_),
+          new StunBindingRequest(port_, server_addr_, deadline_),
           port_->stun_keepalive_delay());
     }
   }
 
-  virtual void OnErrorResponse(StunMessage* response) {
+  virtual void OnErrorResponse(StunMessage* response) override {
     const StunErrorCodeAttribute* attr = response->GetErrorCode();
     if (!attr) {
       LOG(LS_ERROR) << "Bad allocate response error code";
@@ -78,34 +84,27 @@ class StunBindingRequest : public StunRequest {
 
     port_->OnStunBindingOrResolveRequestFailed(server_addr_);
 
-    if (keep_alive_
-        && (rtc::TimeSince(start_time_) <= RETRY_TIMEOUT)) {
+    uint32_t now = rtc::Time();
+    if (now <= deadline_ && rtc::TimeDiff(now, start_time_) <= RETRY_TIMEOUT) {
       port_->requests_.SendDelayed(
-          new StunBindingRequest(port_, true, server_addr_),
+          new StunBindingRequest(port_, server_addr_, deadline_),
           port_->stun_keepalive_delay());
     }
   }
 
-  virtual void OnTimeout() {
+  virtual void OnTimeout() override {
     LOG(LS_ERROR) << "Binding request timed out from "
       << port_->GetLocalAddress().ToSensitiveString()
       << " (" << port_->Network()->name() << ")";
 
     port_->OnStunBindingOrResolveRequestFailed(server_addr_);
-
-    if (keep_alive_
-        && (rtc::TimeSince(start_time_) <= RETRY_TIMEOUT)) {
-      port_->requests_.SendDelayed(
-          new StunBindingRequest(port_, true, server_addr_),
-          RETRY_DELAY);
-    }
   }
 
  private:
   UDPPort* port_;
-  bool keep_alive_;
   const rtc::SocketAddress server_addr_;
-  uint32 start_time_;
+  uint32_t start_time_;
+  uint32_t deadline_;
 };
 
 UDPPort::AddressResolver::AddressResolver(
@@ -115,7 +114,10 @@ UDPPort::AddressResolver::AddressResolver(
 UDPPort::AddressResolver::~AddressResolver() {
   for (ResolverMap::iterator it = resolvers_.begin();
        it != resolvers_.end(); ++it) {
-    it->second->Destroy(true);
+    // TODO(guoweis): Change to asynchronous DNS resolution to prevent the hang
+    // when passing true to the Destroy() which is a safer way to avoid the code
+    // unloaded before the thread exits. Please see webrtc bug 5139.
+    it->second->Destroy(false);
   }
 }
 
@@ -164,14 +166,20 @@ UDPPort::UDPPort(rtc::Thread* thread,
                  rtc::AsyncPacketSocket* socket,
                  const std::string& username,
                  const std::string& password,
-                 const std::string& origin)
-    : Port(thread, factory, network, socket->GetLocalAddress().ipaddr(),
-           username, password),
+                 const std::string& origin,
+                 bool emit_local_for_anyaddress)
+    : Port(thread,
+           factory,
+           network,
+           socket->GetLocalAddress().ipaddr(),
+           username,
+           password),
       requests_(thread),
       socket_(socket),
       error_(0),
       ready_(false),
-      stun_keepalive_delay_(KEEPALIVE_DELAY) {
+      stun_keepalive_delay_(KEEPALIVE_DELAY),
+      emit_local_for_anyaddress_(emit_local_for_anyaddress) {
   requests_.set_origin(origin);
 }
 
@@ -179,18 +187,27 @@ UDPPort::UDPPort(rtc::Thread* thread,
                  rtc::PacketSocketFactory* factory,
                  rtc::Network* network,
                  const rtc::IPAddress& ip,
-                 uint16 min_port,
-                 uint16 max_port,
+                 uint16_t min_port,
+                 uint16_t max_port,
                  const std::string& username,
                  const std::string& password,
-                 const std::string& origin)
-    : Port(thread, LOCAL_PORT_TYPE, factory, network, ip, min_port, max_port,
-           username, password),
+                 const std::string& origin,
+                 bool emit_local_for_anyaddress)
+    : Port(thread,
+           LOCAL_PORT_TYPE,
+           factory,
+           network,
+           ip,
+           min_port,
+           max_port,
+           username,
+           password),
       requests_(thread),
       socket_(NULL),
       error_(0),
       ready_(false),
-      stun_keepalive_delay_(KEEPALIVE_DELAY) {
+      stun_keepalive_delay_(KEEPALIVE_DELAY),
+      emit_local_for_anyaddress_(emit_local_for_anyaddress) {
   requests_.set_origin(origin);
 }
 
@@ -205,6 +222,7 @@ bool UDPPort::Init() {
     }
     socket_->SignalReadPacket.connect(this, &UDPPort::OnReadPacket);
   }
+  socket_->SignalSentPacket.connect(this, &UDPPort::OnSentPacket);
   socket_->SignalReadyToSend.connect(this, &UDPPort::OnReadyToSend);
   socket_->SignalAddressReady.connect(this, &UDPPort::OnLocalAddressReady);
   requests_.SignalSendPacket.connect(this, &UDPPort::OnSendPacket);
@@ -235,9 +253,10 @@ void UDPPort::MaybePrepareStunCandidate() {
 }
 
 Connection* UDPPort::CreateConnection(const Candidate& address,
-                                       CandidateOrigin origin) {
-  if (address.protocol() != "udp")
+                                      CandidateOrigin origin) {
+  if (!SupportsProtocol(address.protocol())) {
     return NULL;
+  }
 
   if (!IsCompatibleAddress(address.address())) {
     return NULL;
@@ -280,18 +299,28 @@ int UDPPort::GetError() {
 
 void UDPPort::OnLocalAddressReady(rtc::AsyncPacketSocket* socket,
                                   const rtc::SocketAddress& address) {
-  AddAddress(address, address, rtc::SocketAddress(),
-             UDP_PROTOCOL_NAME, "", LOCAL_PORT_TYPE,
-             ICE_TYPE_PREFERENCE_HOST, 0, false);
+  // When adapter enumeration is disabled and binding to the any address, the
+  // default local address will be issued as a candidate instead if
+  // |emit_local_for_anyaddress| is true. This is to allow connectivity for
+  // applications which absolutely requires a HOST candidate.
+  rtc::SocketAddress addr = address;
+
+  // If MaybeSetDefaultLocalAddress fails, we keep the "any" IP so that at
+  // least the port is listening.
+  MaybeSetDefaultLocalAddress(&addr);
+
+  AddAddress(addr, addr, rtc::SocketAddress(), UDP_PROTOCOL_NAME, "", "",
+             LOCAL_PORT_TYPE, ICE_TYPE_PREFERENCE_HOST, 0, false);
   MaybePrepareStunCandidate();
 }
 
-void UDPPort::OnReadPacket(
-  rtc::AsyncPacketSocket* socket, const char* data, size_t size,
-  const rtc::SocketAddress& remote_addr,
-  const rtc::PacketTime& packet_time) {
+void UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
+                           const char* data,
+                           size_t size,
+                           const rtc::SocketAddress& remote_addr,
+                           const rtc::PacketTime& packet_time) {
   ASSERT(socket == socket_);
-  ASSERT(!remote_addr.IsUnresolved());
+  ASSERT(!remote_addr.IsUnresolvedIP());
 
   // Look for a response from the STUN server.
   // Even if the response doesn't match one of our outstanding requests, we
@@ -309,13 +338,18 @@ void UDPPort::OnReadPacket(
   }
 }
 
+void UDPPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
+                           const rtc::SentPacket& sent_packet) {
+  PortInterface::SignalSentPacket(sent_packet);
+}
+
 void UDPPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
   Port::OnReadyToSend();
 }
 
 void UDPPort::SendStunBindingRequests() {
   // We will keep pinging the stun server to make sure our NAT pin-hole stays
-  // open during the call.
+  // open until the deadline (specified in SendStunBindingRequest).
   ASSERT(requests_.empty());
 
   for (ServerAddresses::const_iterator it = server_addresses_.begin();
@@ -330,6 +364,8 @@ void UDPPort::ResolveStunAddress(const rtc::SocketAddress& stun_addr) {
     resolver_->SignalDone.connect(this, &UDPPort::OnResolveResult);
   }
 
+  LOG_J(LS_INFO, this) << "Starting STUN host lookup for "
+                       << stun_addr.ToSensitiveString();
   resolver_->Resolve(stun_addr);
 }
 
@@ -354,15 +390,15 @@ void UDPPort::OnResolveResult(const rtc::SocketAddress& input,
   }
 }
 
-void UDPPort::SendStunBindingRequest(
-    const rtc::SocketAddress& stun_addr) {
-  if (stun_addr.IsUnresolved()) {
+void UDPPort::SendStunBindingRequest(const rtc::SocketAddress& stun_addr) {
+  if (stun_addr.IsUnresolvedIP()) {
     ResolveStunAddress(stun_addr);
 
   } else if (socket_->GetState() == rtc::AsyncPacketSocket::STATE_BOUND) {
     // Check if |server_addr_| is compatible with the port's ip.
     if (IsCompatibleAddress(stun_addr)) {
-      requests_.Send(new StunBindingRequest(this, true, stun_addr));
+      requests_.Send(new StunBindingRequest(this, stun_addr,
+                                            rtc::Time() + KEEP_ALIVE_TIMEOUT));
     } else {
       // Since we can't send stun messages to the server, we should mark this
       // port ready.
@@ -372,6 +408,23 @@ void UDPPort::SendStunBindingRequest(
   }
 }
 
+bool UDPPort::MaybeSetDefaultLocalAddress(rtc::SocketAddress* addr) const {
+  if (!addr->IsAnyIP() || !emit_local_for_anyaddress_ ||
+      !Network()->default_local_address_provider()) {
+    return true;
+  }
+  rtc::IPAddress default_address;
+  bool result =
+      Network()->default_local_address_provider()->GetDefaultLocalAddress(
+          addr->family(), &default_address);
+  if (!result || default_address.IsNil()) {
+    return false;
+  }
+
+  addr->SetIP(default_address);
+  return true;
+}
+
 void UDPPort::OnStunBindingRequestSucceeded(
     const rtc::SocketAddress& stun_server_addr,
     const rtc::SocketAddress& stun_reflected_addr) {
@@ -389,16 +442,18 @@ void UDPPort::OnStunBindingRequestSucceeded(
       !HasCandidateWithAddress(stun_reflected_addr)) {
 
     rtc::SocketAddress related_address = socket_->GetLocalAddress();
-    if (!(candidate_filter() & CF_HOST)) {
+    // If we can't stamp the related address correctly, empty it to avoid leak.
+    if (!MaybeSetDefaultLocalAddress(&related_address) ||
+        !(candidate_filter() & CF_HOST)) {
       // If candidate filter doesn't have CF_HOST specified, empty raddr to
       // avoid local address leakage.
       related_address = rtc::EmptySocketAddressWithFamily(
           related_address.family());
     }
 
-    AddAddress(stun_reflected_addr, socket_->GetLocalAddress(),
-               related_address, UDP_PROTOCOL_NAME, "",
-               STUN_PORT_TYPE, ICE_TYPE_PREFERENCE_SRFLX, 0, false);
+    AddAddress(stun_reflected_addr, socket_->GetLocalAddress(), related_address,
+               UDP_PROTOCOL_NAME, "", "", STUN_PORT_TYPE,
+               ICE_TYPE_PREFERENCE_SRFLX, 0, false);
   }
   MaybeSetPortCompleteOrError();
 }
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stunport.h b/media/webrtc/trunk/webrtc/p2p/base/stunport.h
index d840a97126..ecf61a782d 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stunport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/stunport.h
@@ -34,9 +34,10 @@ class UDPPort : public Port {
                          rtc::AsyncPacketSocket* socket,
                          const std::string& username,
                          const std::string& password,
-                         const std::string& origin) {
-    UDPPort* port = new UDPPort(thread, factory, network, socket,
-                                username, password, origin);
+                         const std::string& origin,
+                         bool emit_local_for_anyaddress) {
+    UDPPort* port = new UDPPort(thread, factory, network, socket, username,
+                                password, origin, emit_local_for_anyaddress);
     if (!port->Init()) {
       delete port;
       port = NULL;
@@ -48,14 +49,15 @@ class UDPPort : public Port {
                          rtc::PacketSocketFactory* factory,
                          rtc::Network* network,
                          const rtc::IPAddress& ip,
-                         uint16 min_port,
-                         uint16 max_port,
+                         uint16_t min_port,
+                         uint16_t max_port,
                          const std::string& username,
                          const std::string& password,
-                         const std::string& origin) {
-    UDPPort* port = new UDPPort(thread, factory, network,
-                                ip, min_port, max_port,
-                                username, password, origin);
+                         const std::string& origin,
+                         bool emit_local_for_anyaddress) {
+    UDPPort* port =
+        new UDPPort(thread, factory, network, ip, min_port, max_port, username,
+                    password, origin, emit_local_for_anyaddress);
     if (!port->Init()) {
       delete port;
       port = NULL;
@@ -93,6 +95,9 @@ class UDPPort : public Port {
     OnReadPacket(socket, data, size, remote_addr, packet_time);
     return true;
   }
+  virtual bool SupportsProtocol(const std::string& protocol) const {
+    return protocol == UDP_PROTOCOL_NAME;
+  }
 
   void set_stun_keepalive_delay(int delay) {
     stun_keepalive_delay_ = delay;
@@ -106,11 +111,12 @@ class UDPPort : public Port {
           rtc::PacketSocketFactory* factory,
           rtc::Network* network,
           const rtc::IPAddress& ip,
-          uint16 min_port,
-          uint16 max_port,
+          uint16_t min_port,
+          uint16_t max_port,
           const std::string& username,
           const std::string& password,
-          const std::string& origin);
+          const std::string& origin,
+          bool emit_local_for_anyaddress);
 
   UDPPort(rtc::Thread* thread,
           rtc::PacketSocketFactory* factory,
@@ -118,7 +124,8 @@ class UDPPort : public Port {
           rtc::AsyncPacketSocket* socket,
           const std::string& username,
           const std::string& password,
-          const std::string& origin);
+          const std::string& origin,
+          bool emit_local_for_anyaddress);
 
   bool Init();
 
@@ -134,6 +141,9 @@ class UDPPort : public Port {
                     const rtc::SocketAddress& remote_addr,
                     const rtc::PacketTime& packet_time);
 
+  void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                    const rtc::SentPacket& sent_packet);
+
   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
 
   // This method will send STUN binding request if STUN server address is set.
@@ -141,6 +151,12 @@ class UDPPort : public Port {
 
   void SendStunBindingRequests();
 
+  // Helper function which will set |addr|'s IP to the default local address if
+  // |addr| is the "any" address and |emit_local_for_anyaddress_| is true. When
+  // returning false, it indicates that the operation has failed and the
+  // address shouldn't be used by any candidate.
+  bool MaybeSetDefaultLocalAddress(rtc::SocketAddress* addr) const;
+
  private:
   // A helper class which can be called repeatedly to resolve multiple
   // addresses, as opposed to rtc::AsyncResolverInterface, which can only
@@ -202,6 +218,10 @@ class UDPPort : public Port {
   bool ready_;
   int stun_keepalive_delay_;
 
+  // This is true by default and false when
+  // PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE is specified.
+  bool emit_local_for_anyaddress_;
+
   friend class StunBindingRequest;
 };
 
@@ -211,7 +231,8 @@ class StunPort : public UDPPort {
                           rtc::PacketSocketFactory* factory,
                           rtc::Network* network,
                           const rtc::IPAddress& ip,
-                          uint16 min_port, uint16 max_port,
+                          uint16_t min_port,
+                          uint16_t max_port,
                           const std::string& username,
                           const std::string& password,
                           const ServerAddresses& servers,
@@ -238,14 +259,22 @@ class StunPort : public UDPPort {
            rtc::PacketSocketFactory* factory,
            rtc::Network* network,
            const rtc::IPAddress& ip,
-           uint16 min_port,
-           uint16 max_port,
+           uint16_t min_port,
+           uint16_t max_port,
            const std::string& username,
            const std::string& password,
            const ServerAddresses& servers,
            const std::string& origin)
-     : UDPPort(thread, factory, network, ip, min_port, max_port, username,
-               password, origin) {
+      : UDPPort(thread,
+                factory,
+                network,
+                ip,
+                min_port,
+                max_port,
+                username,
+                password,
+                origin,
+                false) {
     // UDPPort will set these to local udp, updating these to STUN.
     set_type(STUN_PORT_TYPE);
     set_server_addresses(servers);
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stunport_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/stunport_unittest.cc
index 8b6e81b6d0..037d448b9e 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stunport_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/stunport_unittest.cc
@@ -30,7 +30,7 @@ static const SocketAddress kStunHostnameAddr("localhost", 5000);
 static const SocketAddress kBadHostnameAddr("not-a-real-hostname", 5000);
 static const int kTimeoutMs = 10000;
 // stun prio = 100 << 24 | 30 (IPV4) << 8 | 256 - 0
-static const uint32 kStunCandidatePriority = 1677729535;
+static const uint32_t kStunCandidatePriority = 1677729535;
 
 // Tests connecting a StunPort to a fake STUN server (cricket::StunServer)
 // TODO: Use a VirtualSocketServer here. We have to use a
@@ -82,7 +82,7 @@ class StunPortTest : public testing::Test,
         rtc::Thread::Current(), &socket_factory_,
         &network_, socket_.get(),
         rtc::CreateRandomString(16), rtc::CreateRandomString(22),
-        std::string()));
+        std::string(), false));
     ASSERT_TRUE(stun_port_ != NULL);
     ServerAddresses stun_servers;
     stun_servers.insert(server_addr);
@@ -250,6 +250,7 @@ TEST_F(StunPortTest, TestNoDuplicatedAddressWithTwoStunServers) {
   PrepareAddress();
   EXPECT_TRUE_WAIT(done(), kTimeoutMs);
   EXPECT_EQ(1U, port()->Candidates().size());
+  EXPECT_EQ(port()->Candidates()[0].relay_protocol(), "");
 }
 
 // Test that candidates can be allocated for multiple STUN servers, one of which
@@ -281,4 +282,6 @@ TEST_F(StunPortTest, TestTwoCandidatesWithTwoStunServersAcrossNat) {
   PrepareAddress();
   EXPECT_TRUE_WAIT(done(), kTimeoutMs);
   EXPECT_EQ(2U, port()->Candidates().size());
+  EXPECT_EQ(port()->Candidates()[0].relay_protocol(), "");
+  EXPECT_EQ(port()->Candidates()[1].relay_protocol(), "");
 }
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stunrequest.cc b/media/webrtc/trunk/webrtc/p2p/base/stunrequest.cc
index 1f124ee917..ce0364e8db 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stunrequest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/stunrequest.cc
@@ -14,10 +14,11 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/stringencode.h"
 
 namespace cricket {
 
-const uint32 MSG_STUN_SEND = 1;
+const uint32_t MSG_STUN_SEND = 1;
 
 const int MAX_SENDS = 9;
 const int DELAY_UNIT = 100;  // 100 milliseconds
@@ -52,6 +53,16 @@ void StunRequestManager::SendDelayed(StunRequest* request, int delay) {
   }
 }
 
+void StunRequestManager::Flush(int msg_type) {
+  for (const auto kv : requests_) {
+    StunRequest* request = kv.second;
+    if (msg_type == kAllRequests || msg_type == request->type()) {
+      thread_->Clear(request, MSG_STUN_SEND);
+      thread_->Send(request, MSG_STUN_SEND, NULL);
+    }
+  }
+}
+
 void StunRequestManager::Remove(StunRequest* request) {
   ASSERT(request->manager() == this);
   RequestMap::iterator iter = requests_.find(request->id());
@@ -67,7 +78,7 @@ void StunRequestManager::Clear() {
   for (RequestMap::iterator i = requests_.begin(); i != requests_.end(); ++i)
     requests.push_back(i->second);
 
-  for (uint32 i = 0; i < requests.size(); ++i) {
+  for (uint32_t i = 0; i < requests.size(); ++i) {
     // StunRequest destructor calls Remove() which deletes requests
     // from |requests_|.
     delete requests[i];
@@ -76,8 +87,11 @@ void StunRequestManager::Clear() {
 
 bool StunRequestManager::CheckResponse(StunMessage* msg) {
   RequestMap::iterator iter = requests_.find(msg->transaction_id());
-  if (iter == requests_.end())
+  if (iter == requests_.end()) {
+    // TODO(pthatcher): Log unknown responses without being too spammy
+    // in the logs.
     return false;
+  }
 
   StunRequest* request = iter->second;
   if (msg->type() == GetStunSuccessResponseType(request->type())) {
@@ -106,15 +120,20 @@ bool StunRequestManager::CheckResponse(const char* data, size_t size) {
   id.append(data + kStunTransactionIdOffset, kStunTransactionIdLength);
 
   RequestMap::iterator iter = requests_.find(id);
-  if (iter == requests_.end())
+  if (iter == requests_.end()) {
+    // TODO(pthatcher): Log unknown responses without being too spammy
+    // in the logs.
     return false;
+  }
 
   // Parse the STUN message and continue processing as usual.
 
   rtc::ByteBuffer buf(data, size);
   rtc::scoped_ptr<StunMessage> response(iter->second->msg_->CreateNew());
-  if (!response->Read(&buf))
+  if (!response->Read(&buf)) {
+    LOG(LS_WARNING) << "Failed to read STUN response " << rtc::hex_encode(id);
     return false;
+  }
 
   return CheckResponse(response.get());
 }
@@ -162,7 +181,7 @@ const StunMessage* StunRequest::msg() const {
   return msg_;
 }
 
-uint32 StunRequest::Elapsed() const {
+uint32_t StunRequest::Elapsed() const {
   return rtc::TimeSince(tstamp_);
 }
 
@@ -188,16 +207,21 @@ void StunRequest::OnMessage(rtc::Message* pmsg) {
   msg_->Write(&buf);
   manager_->SignalSendPacket(buf.Data(), buf.Length(), this);
 
-  int delay = GetNextDelay();
-  manager_->thread_->PostDelayed(delay, this, MSG_STUN_SEND, NULL);
+  OnSent();
+  manager_->thread_->PostDelayed(resend_delay(), this, MSG_STUN_SEND, NULL);
 }
 
-int StunRequest::GetNextDelay() {
-  int delay = DELAY_UNIT * std::min(1 << count_, DELAY_MAX_FACTOR);
+void StunRequest::OnSent() {
   count_ += 1;
   if (count_ == MAX_SENDS)
     timeout_ = true;
-  return delay;
+}
+
+int StunRequest::resend_delay() {
+  if (count_ == 0) {
+    return 0;
+  }
+  return DELAY_UNIT * std::min(1 << (count_-1), DELAY_MAX_FACTOR);
 }
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stunrequest.h b/media/webrtc/trunk/webrtc/p2p/base/stunrequest.h
index 6a4bdc0971..44c1ebff56 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stunrequest.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/stunrequest.h
@@ -21,6 +21,8 @@ namespace cricket {
 
 class StunRequest;
 
+const int kAllRequests = 0;
+
 // Manages a set of STUN requests, sending and resending until we receive a
 // response or determine that the request has timed out.
 class StunRequestManager {
@@ -32,6 +34,11 @@ class StunRequestManager {
   void Send(StunRequest* request);
   void SendDelayed(StunRequest* request, int delay);
 
+  // If |msg_type| is kAllRequests, sends all pending requests right away.
+  // Otherwise, sends those that have a matching type right away.
+  // Only for testing.
+  void Flush(int msg_type);
+
   // Removes a stun request that was added previously.  This will happen
   // automatically when a request succeeds, fails, or times out.
   void Remove(StunRequest* request);
@@ -90,7 +97,7 @@ class StunRequest : public rtc::MessageHandler {
   const StunMessage* msg() const;
 
   // Time elapsed since last send (in ms)
-  uint32 Elapsed() const;
+  uint32_t Elapsed() const;
 
  protected:
   int count_;
@@ -105,7 +112,10 @@ class StunRequest : public rtc::MessageHandler {
   virtual void OnResponse(StunMessage* response) {}
   virtual void OnErrorResponse(StunMessage* response) {}
   virtual void OnTimeout() {}
-  virtual int GetNextDelay();
+  // Called when the message is sent.
+  virtual void OnSent();
+  // Returns the next delay for resends.
+  virtual int resend_delay();
 
  private:
   void set_manager(StunRequestManager* manager);
@@ -115,7 +125,7 @@ class StunRequest : public rtc::MessageHandler {
 
   StunRequestManager* manager_;
   StunMessage* msg_;
-  uint32 tstamp_;
+  uint32_t tstamp_;
 
   friend class StunRequestManager;
 };
diff --git a/media/webrtc/trunk/webrtc/p2p/base/stunrequest_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/stunrequest_unittest.cc
index 3ff6cbaf72..8a23834891 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/stunrequest_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/stunrequest_unittest.cc
@@ -146,13 +146,13 @@ TEST_F(StunRequestTest, TestUnexpected) {
 TEST_F(StunRequestTest, TestBackoff) {
   StunMessage* req = CreateStunMessage(STUN_BINDING_REQUEST, NULL);
 
-  uint32 start = rtc::Time();
+  uint32_t start = rtc::Time();
   manager_.Send(new StunRequestThunker(req, this));
   StunMessage* res = CreateStunMessage(STUN_BINDING_RESPONSE, req);
   for (int i = 0; i < 9; ++i) {
     while (request_count_ == i)
       rtc::Thread::Current()->ProcessMessages(1);
-    int32 elapsed = rtc::TimeSince(start);
+    int32_t elapsed = rtc::TimeSince(start);
     LOG(LS_INFO) << "STUN request #" << (i + 1)
                  << " sent at " << elapsed << " ms";
     EXPECT_GE(TotalDelay(i + 1), elapsed);
diff --git a/media/webrtc/trunk/webrtc/p2p/base/tcpport.cc b/media/webrtc/trunk/webrtc/p2p/base/tcpport.cc
index 89265d7b11..cd3c9192e4 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/tcpport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/tcpport.cc
@@ -8,6 +8,62 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+/*
+ *  This is a diagram of how TCP reconnect works for the active side. The
+ *  passive side just waits for an incoming connection.
+ *
+ *  - Connected: Indicate whether the TCP socket is connected.
+ *
+ *  - Writable: Whether the stun binding is completed. Sending a data packet
+ *    before stun binding completed will trigger IPC socket layer to shutdown
+ *    the connection.
+ *
+ *  - PendingTCP: |connection_pending_| indicates whether there is an
+ *    outstanding TCP connection in progress.
+ *
+ *  - PretendWri: Tracked by |pretending_to_be_writable_|. Marking connection as
+ *    WRITE_TIMEOUT will cause the connection be deleted. Instead, we're
+ *    "pretending" we're still writable for a period of time such that reconnect
+ *    could work.
+ *
+ *  Data could only be sent in state 3. Sening data during state 2 & 6 will get
+ *  EWOULDBLOCK, 4 & 5 EPIPE.
+ *
+ *         OS Timeout         7 -------------+
+ *   +----------------------->|Connected: N  |
+ *   |                        |Writable:  N  |     Timeout
+ *   |       Timeout          |Connection is |<----------------+
+ *   |   +------------------->|Dead          |                 |
+ *   |   |                    +--------------+                 |
+ *   |   |                               ^                     |
+ *   |   |            OnClose            |                     |
+ *   |   |    +-----------------------+  |                     |
+ *   |   |    |                       |  |Timeout              |
+ *   |   |    v                       |  |                     |
+ *   | 4 +----------+          5 -----+--+--+           6 -----+-----+
+ *   | |Connected: N|Send() or |Connected: N|           |Connected: Y|
+ *   | |Writable:  Y|Ping()    |Writable:  Y|OnConnect  |Writable:  Y|
+ *   | |PendingTCP:N+--------> |PendingTCP:Y+---------> |PendingTCP:N|
+ *   | |PretendWri:Y|          |PretendWri:Y|           |PretendWri:Y|
+ *   | +-----+------+          +------------+           +---+--+-----+
+ *   |   ^   ^                                              |  |
+ *   |   |   |                     OnClose                  |  |
+ *   |   |   +----------------------------------------------+  |
+ *   |   |                                                     |
+ *   |   |                              Stun Binding Completed |
+ *   |   |                                                     |
+ *   |   |                    OnClose                          |
+ *   |   +------------------------------------------------+    |
+ *   |                                                    |    v
+ *  1 -----------+           2 -----------+Stun      3 -----------+
+ *  |Connected: N|           |Connected: Y|Binding   |Connected: Y|
+ *  |Writable:  N|OnConnect  |Writable:  N|Completed |Writable:  Y|
+ *  |PendingTCP:Y+---------> |PendingTCP:N+--------> |PendingTCP:N|
+ *  |PretendWri:N|           |PretendWri:N|          |PretendWri:N|
+ *  +------------+           +------------+          +------------+
+ *
+ */
+
 #include "webrtc/p2p/base/tcpport.h"
 
 #include "webrtc/p2p/base/common.h"
@@ -20,13 +76,20 @@ TCPPort::TCPPort(rtc::Thread* thread,
                  rtc::PacketSocketFactory* factory,
                  rtc::Network* network,
                  const rtc::IPAddress& ip,
-                 uint16 min_port,
-                 uint16 max_port,
+                 uint16_t min_port,
+                 uint16_t max_port,
                  const std::string& username,
                  const std::string& password,
                  bool allow_listen)
-    : Port(thread, LOCAL_PORT_TYPE, factory, network, ip, min_port, max_port,
-           username, password),
+    : Port(thread,
+           LOCAL_PORT_TYPE,
+           factory,
+           network,
+           ip,
+           min_port,
+           max_port,
+           username,
+           password),
       incoming_only_(false),
       allow_listen_(allow_listen),
       socket_(NULL),
@@ -62,9 +125,7 @@ TCPPort::~TCPPort() {
 
 Connection* TCPPort::CreateConnection(const Candidate& address,
                                       CandidateOrigin origin) {
-  // We only support TCP protocols
-  if ((address.protocol() != TCP_PROTOCOL_NAME) &&
-      (address.protocol() != SSLTCP_PROTOCOL_NAME)) {
+  if (!SupportsProtocol(address.protocol())) {
     return NULL;
   }
 
@@ -115,16 +176,18 @@ void TCPPort::PrepareAddress() {
     if (socket_->GetState() == rtc::AsyncPacketSocket::STATE_BOUND ||
         socket_->GetState() == rtc::AsyncPacketSocket::STATE_CLOSED)
       AddAddress(socket_->GetLocalAddress(), socket_->GetLocalAddress(),
-                 rtc::SocketAddress(),
-                 TCP_PROTOCOL_NAME, TCPTYPE_PASSIVE_STR, LOCAL_PORT_TYPE,
+                 rtc::SocketAddress(), TCP_PROTOCOL_NAME, "",
+                 TCPTYPE_PASSIVE_STR, LOCAL_PORT_TYPE,
                  ICE_TYPE_PREFERENCE_HOST_TCP, 0, true);
   } else {
     LOG_J(LS_INFO, this) << "Not listening due to firewall restrictions.";
     // Note: We still add the address, since otherwise the remote side won't
-    // recognize our incoming TCP connections.
-    AddAddress(rtc::SocketAddress(ip(), 0),
+    // recognize our incoming TCP connections. According to
+    // https://tools.ietf.org/html/rfc6544#section-4.5, for active candidate,
+    // the port must be set to the discard port, i.e. 9.
+    AddAddress(rtc::SocketAddress(ip(), DISCARD_PORT),
                rtc::SocketAddress(ip(), 0), rtc::SocketAddress(),
-               TCP_PROTOCOL_NAME, TCPTYPE_ACTIVE_STR, LOCAL_PORT_TYPE,
+               TCP_PROTOCOL_NAME, "", TCPTYPE_ACTIVE_STR, LOCAL_PORT_TYPE,
                ICE_TYPE_PREFERENCE_HOST_TCP, 0, true);
   }
 }
@@ -134,7 +197,16 @@ int TCPPort::SendTo(const void* data, size_t size,
                     const rtc::PacketOptions& options,
                     bool payload) {
   rtc::AsyncPacketSocket * socket = NULL;
-  if (TCPConnection * conn = static_cast<TCPConnection*>(GetConnection(addr))) {
+  TCPConnection* conn = static_cast<TCPConnection*>(GetConnection(addr));
+
+  // For Connection, this is the code path used by Ping() to establish
+  // WRITABLE. It has to send through the socket directly as TCPConnection::Send
+  // checks writability.
+  if (conn) {
+    if (!conn->connected()) {
+      conn->MaybeReconnect();
+      return SOCKET_ERROR;
+    }
     socket = conn->socket();
   } else {
     socket = GetIncoming(addr);
@@ -142,12 +214,15 @@ int TCPPort::SendTo(const void* data, size_t size,
   if (!socket) {
     LOG_J(LS_ERROR, this) << "Attempted to send to an unknown destination, "
                           << addr.ToSensitiveString();
-    return -1;  // TODO: Set error_
+    return SOCKET_ERROR;  // TODO(tbd): Set error_
   }
 
   int sent = socket->Send(data, size, options);
   if (sent < 0) {
     error_ = socket->GetError();
+    // Error from this code path for a Connection (instead of from a bare
+    // socket) will not trigger reconnecting. In theory, this shouldn't matter
+    // as OnClose should always be called and set connected to false.
     LOG_J(LS_ERROR, this) << "TCP send of " << size
                           << " bytes failed with error " << error_;
   }
@@ -183,6 +258,7 @@ void TCPPort::OnNewConnection(rtc::AsyncPacketSocket* socket,
   incoming.socket = new_socket;
   incoming.socket->SignalReadPacket.connect(this, &TCPPort::OnReadPacket);
   incoming.socket->SignalReadyToSend.connect(this, &TCPPort::OnReadyToSend);
+  incoming.socket->SignalSentPacket.connect(this, &TCPPort::OnSentPacket);
 
   LOG_J(LS_VERBOSE, this) << "Accepted connection from "
                           << incoming.addr.ToSensitiveString();
@@ -211,53 +287,45 @@ void TCPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
   Port::OnReadPacket(data, size, remote_addr, PROTO_TCP);
 }
 
+void TCPPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
+                           const rtc::SentPacket& sent_packet) {
+  PortInterface::SignalSentPacket(sent_packet);
+}
+
 void TCPPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
   Port::OnReadyToSend();
 }
 
 void TCPPort::OnAddressReady(rtc::AsyncPacketSocket* socket,
                              const rtc::SocketAddress& address) {
-  AddAddress(address, address, rtc::SocketAddress(),
-             TCP_PROTOCOL_NAME, TCPTYPE_PASSIVE_STR, LOCAL_PORT_TYPE,
-             ICE_TYPE_PREFERENCE_HOST_TCP, 0, true);
+  AddAddress(address, address, rtc::SocketAddress(), TCP_PROTOCOL_NAME, "",
+             TCPTYPE_PASSIVE_STR, LOCAL_PORT_TYPE, ICE_TYPE_PREFERENCE_HOST_TCP,
+             0, true);
 }
 
-TCPConnection::TCPConnection(TCPPort* port, const Candidate& candidate,
+TCPConnection::TCPConnection(TCPPort* port,
+                             const Candidate& candidate,
                              rtc::AsyncPacketSocket* socket)
-    : Connection(port, 0, candidate), socket_(socket), error_(0) {
-  bool outgoing = (socket_ == NULL);
-  if (outgoing) {
-    // TODO: Handle failures here (unlikely since TCP).
-    int opts = (candidate.protocol() == SSLTCP_PROTOCOL_NAME) ?
-        rtc::PacketSocketFactory::OPT_SSLTCP : 0;
-    socket_ = port->socket_factory()->CreateClientTcpSocket(
-        rtc::SocketAddress(port->ip(), 0),
-        candidate.address(), port->proxy(), port->user_agent(), opts);
-    if (socket_) {
-      LOG_J(LS_VERBOSE, this) << "Connecting from "
-                              << socket_->GetLocalAddress().ToSensitiveString()
-                              << " to "
-                              << candidate.address().ToSensitiveString();
-      set_connected(false);
-      socket_->SignalConnect.connect(this, &TCPConnection::OnConnect);
-    } else {
-      LOG_J(LS_WARNING, this) << "Failed to create connection to "
-                              << candidate.address().ToSensitiveString();
-    }
+    : Connection(port, 0, candidate),
+      socket_(socket),
+      error_(0),
+      outgoing_(socket == NULL),
+      connection_pending_(false),
+      pretending_to_be_writable_(false),
+      reconnection_timeout_(cricket::CONNECTION_WRITE_CONNECT_TIMEOUT) {
+  if (outgoing_) {
+    CreateOutgoingTcpSocket();
   } else {
     // Incoming connections should match the network address.
+    LOG_J(LS_VERBOSE, this)
+        << "socket ipaddr: " << socket_->GetLocalAddress().ToString()
+        << ",port() ip:" << port->ip().ToString();
     ASSERT(socket_->GetLocalAddress().ipaddr() == port->ip());
-  }
-
-  if (socket_) {
-    socket_->SignalReadPacket.connect(this, &TCPConnection::OnReadPacket);
-    socket_->SignalReadyToSend.connect(this, &TCPConnection::OnReadyToSend);
-    socket_->SignalClose.connect(this, &TCPConnection::OnClose);
+    ConnectSocketSignals(socket);
   }
 }
 
 TCPConnection::~TCPConnection() {
-  delete socket_;
 }
 
 int TCPConnection::Send(const void* data, size_t size,
@@ -267,7 +335,18 @@ int TCPConnection::Send(const void* data, size_t size,
     return SOCKET_ERROR;
   }
 
-  if (write_state() != STATE_WRITABLE) {
+  // Sending after OnClose on active side will trigger a reconnect for a
+  // outgoing connection. Note that the write state is still WRITABLE as we want
+  // to spend a few seconds attempting a reconnect before saying we're
+  // unwritable.
+  if (!connected()) {
+    MaybeReconnect();
+    return SOCKET_ERROR;
+  }
+
+  // Note that this is important to put this after the previous check to give
+  // the connection a chance to reconnect.
+  if (pretending_to_be_writable_ || write_state() != STATE_WRITABLE) {
     // TODO: Should STATE_WRITE_TIMEOUT return a non-blocking error?
     error_ = EWOULDBLOCK;
     return SOCKET_ERROR;
@@ -278,7 +357,7 @@ int TCPConnection::Send(const void* data, size_t size,
     sent_packets_discarded_++;
     error_ = socket_->GetError();
   } else {
-    send_rate_tracker_.Update(sent);
+    send_rate_tracker_.AddSamples(sent);
   }
   return sent;
 }
@@ -287,6 +366,21 @@ int TCPConnection::GetError() {
   return error_;
 }
 
+void TCPConnection::OnConnectionRequestResponse(ConnectionRequest* req,
+                                                StunMessage* response) {
+  // Process the STUN response before we inform upper layer ready to send.
+  Connection::OnConnectionRequestResponse(req, response);
+
+  // If we're in the state of pretending to be writeable, we should inform the
+  // upper layer it's ready to send again as previous EWOULDLBLOCK from socket
+  // would have stopped the outgoing stream.
+  if (pretending_to_be_writable_) {
+    Connection::OnReadyToSend();
+  }
+  pretending_to_be_writable_ = false;
+  ASSERT(write_state() == STATE_WRITABLE);
+}
+
 void TCPConnection::OnConnect(rtc::AsyncPacketSocket* socket) {
   ASSERT(socket == socket_);
   // Do not use this connection if the socket bound to a different address than
@@ -294,24 +388,83 @@ void TCPConnection::OnConnect(rtc::AsyncPacketSocket* socket) {
   // given a binding address, and the platform is expected to pick the
   // correct local address.
   const rtc::IPAddress& socket_ip = socket->GetLocalAddress().ipaddr();
-  if (socket_ip == port()->ip()) {
-    LOG_J(LS_VERBOSE, this) << "Connection established to "
-                            << socket->GetRemoteAddress().ToSensitiveString();
+  if (socket_ip == port()->ip() || IPIsAny(port()->ip())) {
+    if (socket_ip == port()->ip()) {
+      LOG_J(LS_VERBOSE, this) << "Connection established to "
+                              << socket->GetRemoteAddress().ToSensitiveString();
+    } else {
+      LOG(LS_WARNING) << "Socket is bound to a different address:"
+                      << socket->GetLocalAddress().ipaddr().ToString()
+                      << ", rather then the local port:"
+                      << port()->ip().ToString()
+                      << ". Still allowing it since it's any address"
+                      << ", possibly caused by multi-routes being disabled.";
+    }
     set_connected(true);
+    connection_pending_ = false;
   } else {
     LOG_J(LS_WARNING, this) << "Dropping connection as TCP socket bound to IP "
                             << socket_ip.ToSensitiveString()
                             << ", different from the local candidate IP "
                             << port()->ip().ToSensitiveString();
-    socket_->Close();
+    OnClose(socket, 0);
   }
 }
 
 void TCPConnection::OnClose(rtc::AsyncPacketSocket* socket, int error) {
   ASSERT(socket == socket_);
   LOG_J(LS_INFO, this) << "Connection closed with error " << error;
-  set_connected(false);
-  set_write_state(STATE_WRITE_TIMEOUT);
+
+  // Guard against the condition where IPC socket will call OnClose for every
+  // packet it can't send.
+  if (connected()) {
+    set_connected(false);
+
+    // Prevent the connection from being destroyed by redundant SignalClose
+    // events.
+    pretending_to_be_writable_ = true;
+
+    // We don't attempt reconnect right here. This is to avoid a case where the
+    // shutdown is intentional and reconnect is not necessary. We only reconnect
+    // when the connection is used to Send() or Ping().
+    port()->thread()->PostDelayed(reconnection_timeout(), this,
+                                  MSG_TCPCONNECTION_DELAYED_ONCLOSE);
+  } else if (!pretending_to_be_writable_) {
+    // OnClose could be called when the underneath socket times out during the
+    // initial connect() (i.e. |pretending_to_be_writable_| is false) . We have
+    // to manually destroy here as this connection, as never connected, will not
+    // be scheduled for ping to trigger destroy.
+    Destroy();
+  }
+}
+
+void TCPConnection::OnMessage(rtc::Message* pmsg) {
+  switch (pmsg->message_id) {
+    case MSG_TCPCONNECTION_DELAYED_ONCLOSE:
+      // If this connection can't become connected and writable again in 5
+      // seconds, it's time to tear this down. This is the case for the original
+      // TCP connection on passive side during a reconnect.
+      if (pretending_to_be_writable_) {
+        Destroy();
+      }
+      break;
+    default:
+      Connection::OnMessage(pmsg);
+  }
+}
+
+void TCPConnection::MaybeReconnect() {
+  // Only reconnect for an outgoing TCPConnection when OnClose was signaled and
+  // no outstanding reconnect is pending.
+  if (connected() || connection_pending_ || !outgoing_) {
+    return;
+  }
+
+  LOG_J(LS_INFO, this) << "TCP Connection with remote is closed, "
+                       << "trying to reconnect";
+
+  CreateOutgoingTcpSocket();
+  error_ = EPIPE;
 }
 
 void TCPConnection::OnReadPacket(
@@ -327,4 +480,35 @@ void TCPConnection::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
   Connection::OnReadyToSend();
 }
 
+void TCPConnection::CreateOutgoingTcpSocket() {
+  ASSERT(outgoing_);
+  // TODO(guoweis): Handle failures here (unlikely since TCP).
+  int opts = (remote_candidate().protocol() == SSLTCP_PROTOCOL_NAME)
+                 ? rtc::PacketSocketFactory::OPT_SSLTCP
+                 : 0;
+  socket_.reset(port()->socket_factory()->CreateClientTcpSocket(
+      rtc::SocketAddress(port()->ip(), 0), remote_candidate().address(),
+      port()->proxy(), port()->user_agent(), opts));
+  if (socket_) {
+    LOG_J(LS_VERBOSE, this)
+        << "Connecting from " << socket_->GetLocalAddress().ToSensitiveString()
+        << " to " << remote_candidate().address().ToSensitiveString();
+    set_connected(false);
+    connection_pending_ = true;
+    ConnectSocketSignals(socket_.get());
+  } else {
+    LOG_J(LS_WARNING, this) << "Failed to create connection to "
+                            << remote_candidate().address().ToSensitiveString();
+  }
+}
+
+void TCPConnection::ConnectSocketSignals(rtc::AsyncPacketSocket* socket) {
+  if (outgoing_) {
+    socket->SignalConnect.connect(this, &TCPConnection::OnConnect);
+  }
+  socket->SignalReadPacket.connect(this, &TCPConnection::OnReadPacket);
+  socket->SignalReadyToSend.connect(this, &TCPConnection::OnReadyToSend);
+  socket->SignalClose.connect(this, &TCPConnection::OnClose);
+}
+
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/tcpport.h b/media/webrtc/trunk/webrtc/p2p/base/tcpport.h
index b3655a8067..cfc6245601 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/tcpport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/tcpport.h
@@ -32,8 +32,8 @@ class TCPPort : public Port {
                          rtc::PacketSocketFactory* factory,
                          rtc::Network* network,
                          const rtc::IPAddress& ip,
-                         uint16 min_port,
-                         uint16 max_port,
+                         uint16_t min_port,
+                         uint16_t max_port,
                          const std::string& username,
                          const std::string& password,
                          bool allow_listen) {
@@ -45,34 +45,38 @@ class TCPPort : public Port {
     }
     return port;
   }
-  virtual ~TCPPort();
+  ~TCPPort() override;
 
-  virtual Connection* CreateConnection(const Candidate& address,
-                                       CandidateOrigin origin);
+  Connection* CreateConnection(const Candidate& address,
+                               CandidateOrigin origin) override;
 
-  virtual void PrepareAddress();
+  void PrepareAddress() override;
 
-  virtual int GetOption(rtc::Socket::Option opt, int* value);
-  virtual int SetOption(rtc::Socket::Option opt, int value);
-  virtual int GetError();
+  int GetOption(rtc::Socket::Option opt, int* value) override;
+  int SetOption(rtc::Socket::Option opt, int value) override;
+  int GetError() override;
+  bool SupportsProtocol(const std::string& protocol) const override {
+    return protocol == TCP_PROTOCOL_NAME || protocol == SSLTCP_PROTOCOL_NAME;
+  }
 
  protected:
   TCPPort(rtc::Thread* thread,
           rtc::PacketSocketFactory* factory,
           rtc::Network* network,
           const rtc::IPAddress& ip,
-          uint16 min_port,
-          uint16 max_port,
+          uint16_t min_port,
+          uint16_t max_port,
           const std::string& username,
           const std::string& password,
           bool allow_listen);
   bool Init();
 
   // Handles sending using the local TCP socket.
-  virtual int SendTo(const void* data, size_t size,
-                     const rtc::SocketAddress& addr,
-                     const rtc::PacketOptions& options,
-                     bool payload);
+  int SendTo(const void* data,
+             size_t size,
+             const rtc::SocketAddress& addr,
+             const rtc::PacketOptions& options,
+             bool payload) override;
 
   // Accepts incoming TCP connection.
   void OnNewConnection(rtc::AsyncPacketSocket* socket,
@@ -93,6 +97,9 @@ class TCPPort : public Port {
                     const rtc::SocketAddress& remote_addr,
                     const rtc::PacketTime& packet_time);
 
+  void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                    const rtc::SentPacket& sent_packet) override;
+
   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
 
   void OnAddressReady(rtc::AsyncPacketSocket* socket,
@@ -113,15 +120,42 @@ class TCPConnection : public Connection {
   // Connection is outgoing unless socket is specified
   TCPConnection(TCPPort* port, const Candidate& candidate,
                 rtc::AsyncPacketSocket* socket = 0);
-  virtual ~TCPConnection();
+  ~TCPConnection() override;
 
-  virtual int Send(const void* data, size_t size,
-                   const rtc::PacketOptions& options);
-  virtual int GetError();
+  int Send(const void* data,
+           size_t size,
+           const rtc::PacketOptions& options) override;
+  int GetError() override;
 
-  rtc::AsyncPacketSocket* socket() { return socket_; }
+  rtc::AsyncPacketSocket* socket() { return socket_.get(); }
+
+  void OnMessage(rtc::Message* pmsg) override;
+
+  // Allow test cases to overwrite the default timeout period.
+  int reconnection_timeout() const { return reconnection_timeout_; }
+  void set_reconnection_timeout(int timeout_in_ms) {
+    reconnection_timeout_ = timeout_in_ms;
+  }
+
+ protected:
+  enum {
+    MSG_TCPCONNECTION_DELAYED_ONCLOSE = Connection::MSG_FIRST_AVAILABLE,
+  };
+
+  // Set waiting_for_stun_binding_complete_ to false to allow data packets in
+  // addition to what Port::OnConnectionRequestResponse does.
+  void OnConnectionRequestResponse(ConnectionRequest* req,
+                                   StunMessage* response) override;
 
  private:
+  // Helper function to handle the case when Ping or Send fails with error
+  // related to socket close.
+  void MaybeReconnect();
+
+  void CreateOutgoingTcpSocket();
+
+  void ConnectSocketSignals(rtc::AsyncPacketSocket* socket);
+
   void OnConnect(rtc::AsyncPacketSocket* socket);
   void OnClose(rtc::AsyncPacketSocket* socket, int error);
   void OnReadPacket(rtc::AsyncPacketSocket* socket,
@@ -130,8 +164,23 @@ class TCPConnection : public Connection {
                     const rtc::PacketTime& packet_time);
   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
 
-  rtc::AsyncPacketSocket* socket_;
+  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
   int error_;
+  bool outgoing_;
+
+  // Guard against multiple outgoing tcp connection during a reconnect.
+  bool connection_pending_;
+
+  // Guard against data packets sent when we reconnect a TCP connection. During
+  // reconnecting, when a new tcp connection has being made, we can't send data
+  // packets out until the STUN binding is completed (i.e. the write state is
+  // set to WRITABLE again by Connection::OnConnectionRequestResponse). IPC
+  // socket, when receiving data packets before that, will trigger OnError which
+  // will terminate the newly created connection.
+  bool pretending_to_be_writable_;
+
+  // Allow test case to overwrite the default timeout period.
+  int reconnection_timeout_;
 
   friend class TCPPort;
 };
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transport.cc b/media/webrtc/trunk/webrtc/p2p/base/transport.cc
index 10a069683c..eff10aa0a9 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/transport.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <utility>  // for std::pair
+
 #include "webrtc/p2p/base/transport.h"
 
 #include "webrtc/p2p/base/candidate.h"
@@ -15,65 +17,11 @@
 #include "webrtc/p2p/base/port.h"
 #include "webrtc/p2p/base/transportchannelimpl.h"
 #include "webrtc/base/bind.h"
-#include "webrtc/base/common.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 
 namespace cricket {
 
-using rtc::Bind;
-
-enum {
-  MSG_ONSIGNALINGREADY = 1,
-  MSG_ONREMOTECANDIDATE,
-  MSG_READSTATE,
-  MSG_WRITESTATE,
-  MSG_REQUESTSIGNALING,
-  MSG_CANDIDATEREADY,
-  MSG_ROUTECHANGE,
-  MSG_CONNECTING,
-  MSG_CANDIDATEALLOCATIONCOMPLETE,
-  MSG_ROLECONFLICT,
-  MSG_COMPLETED,
-  MSG_FAILED,
-};
-
-struct ChannelParams : public rtc::MessageData {
-  ChannelParams() : channel(NULL), candidate(NULL) {}
-  explicit ChannelParams(int component)
-      : component(component), channel(NULL), candidate(NULL) {}
-  explicit ChannelParams(Candidate* candidate)
-      : channel(NULL), candidate(candidate) {
-  }
-
-  ~ChannelParams() {
-    delete candidate;
-  }
-
-  std::string name;
-  int component;
-  TransportChannelImpl* channel;
-  Candidate* candidate;
-};
-
-static std::string IceProtoToString(TransportProtocol proto) {
-  std::string proto_str;
-  switch (proto) {
-    case ICEPROTO_GOOGLE:
-      proto_str = "gice";
-      break;
-    case ICEPROTO_HYBRID:
-      proto_str = "hybrid";
-      break;
-    case ICEPROTO_RFC5245:
-      proto_str = "ice";
-      break;
-    default:
-      ASSERT(false);
-      break;
-  }
-  return proto_str;
-}
-
 static bool VerifyIceParams(const TransportDescription& desc) {
   // For legacy protocols.
   if (desc.ice_ufrag.empty() && desc.ice_pwd.empty())
@@ -115,215 +63,169 @@ static bool IceCredentialsChanged(const TransportDescription& old_desc,
                                new_desc.ice_ufrag, new_desc.ice_pwd);
 }
 
-Transport::Transport(rtc::Thread* signaling_thread,
-                     rtc::Thread* worker_thread,
-                     const std::string& content_name,
-                     const std::string& type,
-                     PortAllocator* allocator)
-  : signaling_thread_(signaling_thread),
-    worker_thread_(worker_thread),
-    content_name_(content_name),
-    type_(type),
-    allocator_(allocator),
-    destroyed_(false),
-    readable_(TRANSPORT_STATE_NONE),
-    writable_(TRANSPORT_STATE_NONE),
-    was_writable_(false),
-    connect_requested_(false),
-    ice_role_(ICEROLE_UNKNOWN),
-    tiebreaker_(0),
-    protocol_(ICEPROTO_HYBRID),
-    remote_ice_mode_(ICEMODE_FULL) {
-}
+Transport::Transport(const std::string& name, PortAllocator* allocator)
+    : name_(name), allocator_(allocator) {}
 
 Transport::~Transport() {
-  ASSERT(signaling_thread_->IsCurrent());
-  ASSERT(destroyed_);
+  RTC_DCHECK(channels_destroyed_);
 }
 
 void Transport::SetIceRole(IceRole role) {
-  worker_thread_->Invoke<void>(Bind(&Transport::SetIceRole_w, this, role));
+  ice_role_ = role;
+  for (const auto& kv : channels_) {
+    kv.second->SetIceRole(ice_role_);
+  }
 }
 
-void Transport::SetIdentity(rtc::SSLIdentity* identity) {
-  worker_thread_->Invoke<void>(Bind(&Transport::SetIdentity_w, this, identity));
-}
-
-bool Transport::GetIdentity(rtc::SSLIdentity** identity) {
-  // The identity is set on the worker thread, so for safety it must also be
-  // acquired on the worker thread.
-  return worker_thread_->Invoke<bool>(
-      Bind(&Transport::GetIdentity_w, this, identity));
-}
-
-bool Transport::GetRemoteCertificate(rtc::SSLCertificate** cert) {
-  // Channels can be deleted on the worker thread, so for safety the remote
-  // certificate is acquired on the worker thread.
-  return worker_thread_->Invoke<bool>(
-      Bind(&Transport::GetRemoteCertificate_w, this, cert));
-}
-
-bool Transport::GetRemoteCertificate_w(rtc::SSLCertificate** cert) {
-  ASSERT(worker_thread()->IsCurrent());
-  if (channels_.empty())
+bool Transport::GetRemoteSSLCertificate(rtc::SSLCertificate** cert) {
+  if (channels_.empty()) {
     return false;
+  }
 
-  ChannelMap::iterator iter = channels_.begin();
-  return iter->second->GetRemoteCertificate(cert);
+  auto iter = channels_.begin();
+  return iter->second->GetRemoteSSLCertificate(cert);
+}
+
+void Transport::SetIceConfig(const IceConfig& config) {
+  ice_config_ = config;
+  for (const auto& kv : channels_) {
+    kv.second->SetIceConfig(ice_config_);
+  }
 }
 
 bool Transport::SetLocalTransportDescription(
     const TransportDescription& description,
     ContentAction action,
     std::string* error_desc) {
-  return worker_thread_->Invoke<bool>(Bind(
-      &Transport::SetLocalTransportDescription_w, this,
-      description, action, error_desc));
+  bool ret = true;
+
+  if (!VerifyIceParams(description)) {
+    return BadTransportDescription("Invalid ice-ufrag or ice-pwd length",
+                                   error_desc);
+  }
+
+  if (local_description_ &&
+      IceCredentialsChanged(*local_description_, description)) {
+    IceRole new_ice_role =
+        (action == CA_OFFER) ? ICEROLE_CONTROLLING : ICEROLE_CONTROLLED;
+
+    // It must be called before ApplyLocalTransportDescription, which may
+    // trigger an ICE restart and depends on the new ICE role.
+    SetIceRole(new_ice_role);
+  }
+
+  local_description_.reset(new TransportDescription(description));
+
+  for (const auto& kv : channels_) {
+    ret &= ApplyLocalTransportDescription(kv.second, error_desc);
+  }
+  if (!ret) {
+    return false;
+  }
+
+  // If PRANSWER/ANSWER is set, we should decide transport protocol type.
+  if (action == CA_PRANSWER || action == CA_ANSWER) {
+    ret &= NegotiateTransportDescription(action, error_desc);
+  }
+  if (ret) {
+    local_description_set_ = true;
+    ConnectChannels();
+  }
+
+  return ret;
 }
 
 bool Transport::SetRemoteTransportDescription(
     const TransportDescription& description,
     ContentAction action,
     std::string* error_desc) {
-  return worker_thread_->Invoke<bool>(Bind(
-      &Transport::SetRemoteTransportDescription_w, this,
-      description, action, error_desc));
+  bool ret = true;
+
+  if (!VerifyIceParams(description)) {
+    return BadTransportDescription("Invalid ice-ufrag or ice-pwd length",
+                                   error_desc);
+  }
+
+  remote_description_.reset(new TransportDescription(description));
+  for (const auto& kv : channels_) {
+    ret &= ApplyRemoteTransportDescription(kv.second, error_desc);
+  }
+
+  // If PRANSWER/ANSWER is set, we should decide transport protocol type.
+  if (action == CA_PRANSWER || action == CA_ANSWER) {
+    ret = NegotiateTransportDescription(CA_OFFER, error_desc);
+  }
+  if (ret) {
+    remote_description_set_ = true;
+  }
+
+  return ret;
 }
 
 TransportChannelImpl* Transport::CreateChannel(int component) {
-  return worker_thread_->Invoke<TransportChannelImpl*>(Bind(
-      &Transport::CreateChannel_w, this, component));
-}
-
-TransportChannelImpl* Transport::CreateChannel_w(int component) {
-  ASSERT(worker_thread()->IsCurrent());
-  TransportChannelImpl* impl;
-  // TODO(tommi): We don't really need to grab the lock until the actual call
-  // to insert() below and presumably hold it throughout initialization of
-  // |impl| after the impl_exists check.  Maybe we can factor that out to
-  // a separate function and not grab the lock in this function.
-  // Actually, we probably don't need to hold the lock while initializing
-  // |impl| since we can just do the insert when that's done.
-  rtc::CritScope cs(&crit_);
+  TransportChannelImpl* channel;
 
   // Create the entry if it does not exist.
-  bool impl_exists = false;
-  auto iterator = channels_.find(component);
-  if (iterator == channels_.end()) {
-    impl = CreateTransportChannel(component);
-    iterator = channels_.insert(std::pair<int, ChannelMapEntry>(
-        component, ChannelMapEntry(impl))).first;
+  bool channel_exists = false;
+  auto iter = channels_.find(component);
+  if (iter == channels_.end()) {
+    channel = CreateTransportChannel(component);
+    channels_.insert(std::pair<int, TransportChannelImpl*>(component, channel));
   } else {
-    impl = iterator->second.get();
-    impl_exists = true;
+    channel = iter->second;
+    channel_exists = true;
   }
 
-  // Increase the ref count.
-  iterator->second.AddRef();
-  destroyed_ = false;
+  channels_destroyed_ = false;
 
-  if (impl_exists) {
-    // If this is an existing channel, we should just return it without
-    // connecting to all the signal again.
-    return impl;
+  if (channel_exists) {
+    // If this is an existing channel, we should just return it.
+    return channel;
   }
 
   // Push down our transport state to the new channel.
-  impl->SetIceRole(ice_role_);
-  impl->SetIceTiebreaker(tiebreaker_);
-  // TODO(ronghuawu): Change CreateChannel_w to be able to return error since
-  // below Apply**Description_w calls can fail.
+  channel->SetIceRole(ice_role_);
+  channel->SetIceTiebreaker(tiebreaker_);
+  channel->SetIceConfig(ice_config_);
+  // TODO(ronghuawu): Change CreateChannel to be able to return error since
+  // below Apply**Description calls can fail.
   if (local_description_)
-    ApplyLocalTransportDescription_w(impl, NULL);
+    ApplyLocalTransportDescription(channel, nullptr);
   if (remote_description_)
-    ApplyRemoteTransportDescription_w(impl, NULL);
+    ApplyRemoteTransportDescription(channel, nullptr);
   if (local_description_ && remote_description_)
-    ApplyNegotiatedTransportDescription_w(impl, NULL);
-
-  impl->SignalReadableState.connect(this, &Transport::OnChannelReadableState);
-  impl->SignalWritableState.connect(this, &Transport::OnChannelWritableState);
-  impl->SignalRequestSignaling.connect(
-      this, &Transport::OnChannelRequestSignaling);
-  impl->SignalCandidateReady.connect(this, &Transport::OnChannelCandidateReady);
-  impl->SignalRouteChange.connect(this, &Transport::OnChannelRouteChange);
-  impl->SignalCandidatesAllocationDone.connect(
-      this, &Transport::OnChannelCandidatesAllocationDone);
-  impl->SignalRoleConflict.connect(this, &Transport::OnRoleConflict);
-  impl->SignalConnectionRemoved.connect(
-      this, &Transport::OnChannelConnectionRemoved);
+    ApplyNegotiatedTransportDescription(channel, nullptr);
 
   if (connect_requested_) {
-    impl->Connect();
-    if (channels_.size() == 1) {
-      // If this is the first channel, then indicate that we have started
-      // connecting.
-      signaling_thread()->Post(this, MSG_CONNECTING, NULL);
-    }
+    channel->Connect();
   }
-  return impl;
+  return channel;
 }
 
 TransportChannelImpl* Transport::GetChannel(int component) {
-  // TODO(tommi,pthatcher): Since we're returning a pointer from the channels_
-  // map, shouldn't we assume that we're on the worker thread? (The pointer
-  // will be used outside of the lock).
-  // And if we're on the worker thread, which is the only thread that modifies
-  // channels_, can we skip grabbing the lock?
-  rtc::CritScope cs(&crit_);
-  ChannelMap::iterator iter = channels_.find(component);
-  return (iter != channels_.end()) ? iter->second.get() : NULL;
+  auto iter = channels_.find(component);
+  return (iter != channels_.end()) ? iter->second : nullptr;
 }
 
 bool Transport::HasChannels() {
-  rtc::CritScope cs(&crit_);
   return !channels_.empty();
 }
 
 void Transport::DestroyChannel(int component) {
-  worker_thread_->Invoke<void>(Bind(
-      &Transport::DestroyChannel_w, this, component));
-}
-
-void Transport::DestroyChannel_w(int component) {
-  ASSERT(worker_thread()->IsCurrent());
-
-  ChannelMap::iterator iter = channels_.find(component);
+  auto iter = channels_.find(component);
   if (iter == channels_.end())
     return;
 
-  TransportChannelImpl* impl = NULL;
-
-  iter->second.DecRef();
-  if (!iter->second.ref()) {
-    impl = iter->second.get();
-    rtc::CritScope cs(&crit_);
-    channels_.erase(iter);
-  }
-
-  if (connect_requested_ && channels_.empty()) {
-    // We're no longer attempting to connect.
-    signaling_thread()->Post(this, MSG_CONNECTING, NULL);
-  }
-
-  if (impl) {
-    // Check in case the deleted channel was the only non-writable channel.
-    OnChannelWritableState(impl);
-    DestroyTransportChannel(impl);
-  }
+  TransportChannelImpl* channel = iter->second;
+  channels_.erase(iter);
+  DestroyTransportChannel(channel);
 }
 
 void Transport::ConnectChannels() {
-  ASSERT(signaling_thread()->IsCurrent());
-  worker_thread_->Invoke<void>(Bind(&Transport::ConnectChannels_w, this));
-}
-
-void Transport::ConnectChannels_w() {
-  ASSERT(worker_thread()->IsCurrent());
   if (connect_requested_ || channels_.empty())
     return;
 
   connect_requested_ = true;
-  signaling_thread()->Post(this, MSG_CANDIDATEREADY, NULL);
 
   if (!local_description_) {
     // TOOD(mallinath) : TransportDescription(TD) shouldn't be generated here.
@@ -332,93 +234,41 @@ void Transport::ConnectChannels_w() {
     // Session.
     // Session must generate local TD before remote candidates pushed when
     // initiate request initiated by the remote.
-    LOG(LS_INFO) << "Transport::ConnectChannels_w: No local description has "
+    LOG(LS_INFO) << "Transport::ConnectChannels: No local description has "
                  << "been set. Will generate one.";
-    TransportDescription desc(NS_GINGLE_P2P, std::vector<std::string>(),
-                              rtc::CreateRandomString(ICE_UFRAG_LENGTH),
-                              rtc::CreateRandomString(ICE_PWD_LENGTH),
-                              ICEMODE_FULL, CONNECTIONROLE_NONE, NULL,
-                              Candidates());
-    SetLocalTransportDescription_w(desc, CA_OFFER, NULL);
+    TransportDescription desc(
+        std::vector<std::string>(), rtc::CreateRandomString(ICE_UFRAG_LENGTH),
+        rtc::CreateRandomString(ICE_PWD_LENGTH), ICEMODE_FULL,
+        CONNECTIONROLE_NONE, nullptr, Candidates());
+    SetLocalTransportDescription(desc, CA_OFFER, nullptr);
   }
 
-  CallChannels_w(&TransportChannelImpl::Connect);
-  if (!channels_.empty()) {
-    signaling_thread()->Post(this, MSG_CONNECTING, NULL);
-  }
+  CallChannels(&TransportChannelImpl::Connect);
 }
 
-void Transport::OnConnecting_s() {
-  ASSERT(signaling_thread()->IsCurrent());
-  SignalConnecting(this);
+void Transport::MaybeStartGathering() {
+  if (connect_requested_) {
+    CallChannels(&TransportChannelImpl::MaybeStartGathering);
+  }
 }
 
 void Transport::DestroyAllChannels() {
-  ASSERT(signaling_thread()->IsCurrent());
-  worker_thread_->Invoke<void>(Bind(&Transport::DestroyAllChannels_w, this));
-  worker_thread()->Clear(this);
-  signaling_thread()->Clear(this);
-  destroyed_ = true;
-}
-
-void Transport::DestroyAllChannels_w() {
-  ASSERT(worker_thread()->IsCurrent());
-
-  std::vector<TransportChannelImpl*> impls;
-  for (auto& iter : channels_) {
-    iter.second.DecRef();
-    if (!iter.second.ref())
-      impls.push_back(iter.second.get());
+  for (const auto& kv : channels_) {
+    DestroyTransportChannel(kv.second);
   }
-
-  {
-    rtc::CritScope cs(&crit_);
-    channels_.clear();
-  }
-
-  for (size_t i = 0; i < impls.size(); ++i)
-    DestroyTransportChannel(impls[i]);
+  channels_.clear();
+  channels_destroyed_ = true;
 }
 
-void Transport::ResetChannels() {
-  ASSERT(signaling_thread()->IsCurrent());
-  worker_thread_->Invoke<void>(Bind(&Transport::ResetChannels_w, this));
-}
-
-void Transport::ResetChannels_w() {
-  ASSERT(worker_thread()->IsCurrent());
-
-  // We are no longer attempting to connect
-  connect_requested_ = false;
-
-  // Clear out the old messages, they aren't relevant
-  rtc::CritScope cs(&crit_);
-  ready_candidates_.clear();
-
-  // Reset all of the channels
-  CallChannels_w(&TransportChannelImpl::Reset);
-}
-
-void Transport::OnSignalingReady() {
-  ASSERT(signaling_thread()->IsCurrent());
-  if (destroyed_) return;
-
-  worker_thread()->Post(this, MSG_ONSIGNALINGREADY, NULL);
-
-  // Notify the subclass.
-  OnTransportSignalingReady();
-}
-
-void Transport::CallChannels_w(TransportChannelFunc func) {
-  ASSERT(worker_thread()->IsCurrent());
-  for (const auto& iter : channels_) {
-    ((iter.second.get())->*func)();
+void Transport::CallChannels(TransportChannelFunc func) {
+  for (const auto& kv : channels_) {
+    (kv.second->*func)();
   }
 }
 
 bool Transport::VerifyCandidate(const Candidate& cand, std::string* error) {
   // No address zero.
-  if (cand.address().IsNil() || cand.address().IsAny()) {
+  if (cand.address().IsNil() || cand.address().IsAnyIP()) {
     *error = "candidate has address of zero";
     return false;
   }
@@ -449,22 +299,15 @@ bool Transport::VerifyCandidate(const Candidate& cand, std::string* error) {
 
 
 bool Transport::GetStats(TransportStats* stats) {
-  ASSERT(signaling_thread()->IsCurrent());
-  return worker_thread_->Invoke<bool>(Bind(
-      &Transport::GetStats_w, this, stats));
-}
-
-bool Transport::GetStats_w(TransportStats* stats) {
-  ASSERT(worker_thread()->IsCurrent());
-  stats->content_name = content_name();
+  stats->transport_name = name();
   stats->channel_stats.clear();
-  for (auto iter : channels_) {
-    ChannelMapEntry& entry = iter.second;
+  for (auto kv : channels_) {
+    TransportChannelImpl* channel = kv.second;
     TransportChannelStats substats;
-    substats.component = entry->component();
-    entry->GetSrtpCipher(&substats.srtp_cipher);
-    entry->GetSslCipher(&substats.ssl_cipher);
-    if (!entry->GetStats(&substats.connection_infos)) {
+    substats.component = channel->component();
+    channel->GetSrtpCryptoSuite(&substats.srtp_crypto_suite);
+    channel->GetSslCipherSuite(&substats.ssl_cipher_suite);
+    if (!channel->GetStats(&substats.connection_infos)) {
       return false;
     }
     stats->channel_stats.push_back(substats);
@@ -472,405 +315,61 @@ bool Transport::GetStats_w(TransportStats* stats) {
   return true;
 }
 
-bool Transport::GetSslRole(rtc::SSLRole* ssl_role) const {
-  return worker_thread_->Invoke<bool>(Bind(
-      &Transport::GetSslRole_w, this, ssl_role));
-}
-
-void Transport::OnRemoteCandidates(const std::vector<Candidate>& candidates) {
-  for (std::vector<Candidate>::const_iterator iter = candidates.begin();
-       iter != candidates.end();
-       ++iter) {
-    OnRemoteCandidate(*iter);
-  }
-}
-
-void Transport::OnRemoteCandidate(const Candidate& candidate) {
-  ASSERT(signaling_thread()->IsCurrent());
-  if (destroyed_) return;
-
-  if (!HasChannel(candidate.component())) {
-    LOG(LS_WARNING) << "Ignoring candidate for unknown component "
-                    << candidate.component();
-    return;
-  }
-
-  ChannelParams* params = new ChannelParams(new Candidate(candidate));
-  worker_thread()->Post(this, MSG_ONREMOTECANDIDATE, params);
-}
-
-void Transport::OnRemoteCandidate_w(const Candidate& candidate) {
-  ASSERT(worker_thread()->IsCurrent());
-  ChannelMap::iterator iter = channels_.find(candidate.component());
-  // It's ok for a channel to go away while this message is in transit.
-  if (iter != channels_.end()) {
-    iter->second->OnCandidate(candidate);
-  }
-}
-
-void Transport::OnChannelReadableState(TransportChannel* channel) {
-  ASSERT(worker_thread()->IsCurrent());
-  signaling_thread()->Post(this, MSG_READSTATE, NULL);
-}
-
-void Transport::OnChannelReadableState_s() {
-  ASSERT(signaling_thread()->IsCurrent());
-  TransportState readable = GetTransportState_s(true);
-  if (readable_ != readable) {
-    readable_ = readable;
-    SignalReadableState(this);
-  }
-}
-
-void Transport::OnChannelWritableState(TransportChannel* channel) {
-  ASSERT(worker_thread()->IsCurrent());
-  signaling_thread()->Post(this, MSG_WRITESTATE, NULL);
-
-  MaybeCompleted_w();
-}
-
-void Transport::OnChannelWritableState_s() {
-  ASSERT(signaling_thread()->IsCurrent());
-  TransportState writable = GetTransportState_s(false);
-  if (writable_ != writable) {
-    was_writable_ = (writable_ == TRANSPORT_STATE_ALL);
-    writable_ = writable;
-    SignalWritableState(this);
-  }
-}
-
-TransportState Transport::GetTransportState_s(bool read) {
-  ASSERT(signaling_thread()->IsCurrent());
-
-  rtc::CritScope cs(&crit_);
-  bool any = false;
-  bool all = !channels_.empty();
-  for (const auto iter : channels_) {
-    bool b = (read ? iter.second->readable() :
-                     iter.second->writable());
-    any |= b;
-    all &=  b;
-  }
-
-  if (all) {
-    return TRANSPORT_STATE_ALL;
-  } else if (any) {
-    return TRANSPORT_STATE_SOME;
-  }
-
-  return TRANSPORT_STATE_NONE;
-}
-
-void Transport::OnChannelRequestSignaling(TransportChannelImpl* channel) {
-  ASSERT(worker_thread()->IsCurrent());
-  // Resetting ICE state for the channel.
-  ChannelMap::iterator iter = channels_.find(channel->component());
-  if (iter != channels_.end())
-    iter->second.set_candidates_allocated(false);
-  signaling_thread()->Post(this, MSG_REQUESTSIGNALING, nullptr);
-}
-
-void Transport::OnChannelRequestSignaling_s() {
-  ASSERT(signaling_thread()->IsCurrent());
-  LOG(LS_INFO) << "Transport: " << content_name_ << ", allocating candidates";
-  SignalRequestSignaling(this);
-}
-
-void Transport::OnChannelCandidateReady(TransportChannelImpl* channel,
-                                        const Candidate& candidate) {
-  ASSERT(worker_thread()->IsCurrent());
-  rtc::CritScope cs(&crit_);
-  ready_candidates_.push_back(candidate);
-
-  // We hold any messages until the client lets us connect.
-  if (connect_requested_) {
-    signaling_thread()->Post(
-        this, MSG_CANDIDATEREADY, NULL);
-  }
-}
-
-void Transport::OnChannelCandidateReady_s() {
-  ASSERT(signaling_thread()->IsCurrent());
-  ASSERT(connect_requested_);
-
-  std::vector<Candidate> candidates;
-  {
-    rtc::CritScope cs(&crit_);
-    candidates.swap(ready_candidates_);
-  }
-
-  // we do the deleting of Candidate* here to keep the new above and
-  // delete below close to each other
-  if (!candidates.empty()) {
-    SignalCandidatesReady(this, candidates);
-  }
-}
-
-void Transport::OnChannelRouteChange(TransportChannel* channel,
-                                     const Candidate& remote_candidate) {
-  ASSERT(worker_thread()->IsCurrent());
-  ChannelParams* params = new ChannelParams(new Candidate(remote_candidate));
-  params->channel = static_cast<cricket::TransportChannelImpl*>(channel);
-  signaling_thread()->Post(this, MSG_ROUTECHANGE, params);
-}
-
-void Transport::OnChannelRouteChange_s(const TransportChannel* channel,
-                                       const Candidate& remote_candidate) {
-  ASSERT(signaling_thread()->IsCurrent());
-  SignalRouteChange(this, remote_candidate.component(), remote_candidate);
-}
-
-void Transport::OnChannelCandidatesAllocationDone(
-    TransportChannelImpl* channel) {
-  ASSERT(worker_thread()->IsCurrent());
-  ChannelMap::iterator iter = channels_.find(channel->component());
-  ASSERT(iter != channels_.end());
-  LOG(LS_INFO) << "Transport: " << content_name_ << ", component "
-               << channel->component() << " allocation complete";
-
-  iter->second.set_candidates_allocated(true);
-
-  // If all channels belonging to this Transport got signal, then
-  // forward this signal to upper layer.
-  // Can this signal arrive before all transport channels are created?
-  for (auto& iter : channels_) {
-    if (!iter.second.candidates_allocated())
-      return;
-  }
-  signaling_thread_->Post(this, MSG_CANDIDATEALLOCATIONCOMPLETE);
-
-  MaybeCompleted_w();
-}
-
-void Transport::OnChannelCandidatesAllocationDone_s() {
-  ASSERT(signaling_thread()->IsCurrent());
-  LOG(LS_INFO) << "Transport: " << content_name_ << " allocation complete";
-  SignalCandidatesAllocationDone(this);
-}
-
-void Transport::OnRoleConflict(TransportChannelImpl* channel) {
-  signaling_thread_->Post(this, MSG_ROLECONFLICT);
-}
-
-void Transport::OnChannelConnectionRemoved(TransportChannelImpl* channel) {
-  ASSERT(worker_thread()->IsCurrent());
-  MaybeCompleted_w();
-
-  // Check if the state is now Failed.
-  // Failed is only available in the Controlling ICE role.
-  if (channel->GetIceRole() != ICEROLE_CONTROLLING) {
-    return;
-  }
-
-  ChannelMap::iterator iter = channels_.find(channel->component());
-  ASSERT(iter != channels_.end());
-  // Failed can only occur after candidate allocation has stopped.
-  if (!iter->second.candidates_allocated()) {
-    return;
-  }
-
-  if (channel->GetState() == TransportChannelState::STATE_FAILED) {
-    // A Transport has failed if any of its channels have no remaining
-    // connections.
-    signaling_thread_->Post(this, MSG_FAILED);
-  }
-}
-
-void Transport::MaybeCompleted_w() {
-  ASSERT(worker_thread()->IsCurrent());
-
-  // When there is no channel created yet, calling this function could fire an
-  // IceConnectionCompleted event prematurely.
-  if (channels_.empty()) {
-    return;
-  }
-
-  // A Transport's ICE process is completed if all of its channels are writable,
-  // have finished allocating candidates, and have pruned all but one of their
-  // connections.
-  for (const auto& iter : channels_) {
-    const TransportChannelImpl* channel = iter.second.get();
-    if (!(channel->writable() &&
-          channel->GetState() == TransportChannelState::STATE_COMPLETED &&
-          channel->GetIceRole() == ICEROLE_CONTROLLING &&
-          iter.second.candidates_allocated())) {
-      return;
+bool Transport::AddRemoteCandidates(const std::vector<Candidate>& candidates,
+                                    std::string* error) {
+  ASSERT(!channels_destroyed_);
+  // Verify each candidate before passing down to transport layer.
+  for (const Candidate& cand : candidates) {
+    if (!VerifyCandidate(cand, error)) {
+      return false;
+    }
+    if (!HasChannel(cand.component())) {
+      *error = "Candidate has unknown component: " + cand.ToString() +
+               " for content: " + name();
+      return false;
     }
   }
 
-  signaling_thread_->Post(this, MSG_COMPLETED);
+  for (const Candidate& candidate : candidates) {
+    TransportChannelImpl* channel = GetChannel(candidate.component());
+    if (channel != nullptr) {
+      channel->AddRemoteCandidate(candidate);
+    }
+  }
+  return true;
 }
 
-void Transport::SetIceRole_w(IceRole role) {
-  ASSERT(worker_thread()->IsCurrent());
-  rtc::CritScope cs(&crit_);
-  ice_role_ = role;
-  for (auto& iter : channels_) {
-    iter.second->SetIceRole(ice_role_);
-  }
-}
-
-void Transport::SetRemoteIceMode_w(IceMode mode) {
-  ASSERT(worker_thread()->IsCurrent());
-  remote_ice_mode_ = mode;
-  // Shouldn't channels be created after this method executed?
-  for (auto& iter : channels_) {
-    iter.second->SetRemoteIceMode(remote_ice_mode_);
-  }
-}
-
-bool Transport::SetLocalTransportDescription_w(
-    const TransportDescription& desc,
-    ContentAction action,
-    std::string* error_desc) {
-  ASSERT(worker_thread()->IsCurrent());
-  bool ret = true;
-
-  if (!VerifyIceParams(desc)) {
-    return BadTransportDescription("Invalid ice-ufrag or ice-pwd length",
-                                   error_desc);
-  }
-
-  // TODO(tommi,pthatcher): I'm not sure why we need to grab this lock at this
-  // point. |local_description_| seems to always be modified on the worker
-  // thread, so we should be able to use it here without grabbing the lock.
-  // However, we _might_ need it before the call to reset() below?
-  // Raw access to |local_description_| is granted to derived transports outside
-  // of locking (see local_description() in the header file).
-  // The contract is that the derived implementations must be aware of when the
-  // description might change and do appropriate synchronization.
-  rtc::CritScope cs(&crit_);
-  if (local_description_ && IceCredentialsChanged(*local_description_, desc)) {
-    IceRole new_ice_role = (action == CA_OFFER) ? ICEROLE_CONTROLLING
-                                                : ICEROLE_CONTROLLED;
-
-    // It must be called before ApplyLocalTransportDescription_w, which may
-    // trigger an ICE restart and depends on the new ICE role.
-    SetIceRole_w(new_ice_role);
-  }
-
-  local_description_.reset(new TransportDescription(desc));
-
-  for (auto& iter : channels_) {
-    ret &= ApplyLocalTransportDescription_w(iter.second.get(), error_desc);
-  }
-  if (!ret)
-    return false;
-
-  // If PRANSWER/ANSWER is set, we should decide transport protocol type.
-  if (action == CA_PRANSWER || action == CA_ANSWER) {
-    ret &= NegotiateTransportDescription_w(action, error_desc);
-  }
-  return ret;
-}
-
-bool Transport::SetRemoteTransportDescription_w(
-    const TransportDescription& desc,
-    ContentAction action,
-    std::string* error_desc) {
-  bool ret = true;
-
-  if (!VerifyIceParams(desc)) {
-    return BadTransportDescription("Invalid ice-ufrag or ice-pwd length",
-                                   error_desc);
-  }
-
-  // TODO(tommi,pthatcher): See todo for local_description_ above.
-  rtc::CritScope cs(&crit_);
-  remote_description_.reset(new TransportDescription(desc));
-  for (auto& iter : channels_) {
-    ret &= ApplyRemoteTransportDescription_w(iter.second.get(), error_desc);
-  }
-
-  // If PRANSWER/ANSWER is set, we should decide transport protocol type.
-  if (action == CA_PRANSWER || action == CA_ANSWER) {
-    ret = NegotiateTransportDescription_w(CA_OFFER, error_desc);
-  }
-  return ret;
-}
-
-bool Transport::ApplyLocalTransportDescription_w(TransportChannelImpl* ch,
-                                                 std::string* error_desc) {
-  ASSERT(worker_thread()->IsCurrent());
-  // If existing protocol_type is HYBRID, we may have not chosen the final
-  // protocol type, so update the channel protocol type from the
-  // local description. Otherwise, skip updating the protocol type.
-  // We check for HYBRID to avoid accidental changes; in the case of a
-  // session renegotiation, the new offer will have the google-ice ICE option,
-  // so we need to make sure we don't switch back from ICE mode to HYBRID
-  // when this happens.
-  // There are some other ways we could have solved this, but this is the
-  // simplest. The ultimate solution will be to get rid of GICE altogether.
-  IceProtocolType protocol_type;
-  if (ch->GetIceProtocolType(&protocol_type) &&
-      protocol_type == ICEPROTO_HYBRID) {
-    ch->SetIceProtocolType(
-        TransportProtocolFromDescription(local_description()));
-  }
+bool Transport::ApplyLocalTransportDescription(TransportChannelImpl* ch,
+                                               std::string* error_desc) {
   ch->SetIceCredentials(local_description_->ice_ufrag,
                         local_description_->ice_pwd);
   return true;
 }
 
-bool Transport::ApplyRemoteTransportDescription_w(TransportChannelImpl* ch,
-                                                  std::string* error_desc) {
+bool Transport::ApplyRemoteTransportDescription(TransportChannelImpl* ch,
+                                                std::string* error_desc) {
   ch->SetRemoteIceCredentials(remote_description_->ice_ufrag,
                               remote_description_->ice_pwd);
   return true;
 }
 
-bool Transport::ApplyNegotiatedTransportDescription_w(
-    TransportChannelImpl* channel, std::string* error_desc) {
-  ASSERT(worker_thread()->IsCurrent());
-  channel->SetIceProtocolType(protocol_);
+bool Transport::ApplyNegotiatedTransportDescription(
+    TransportChannelImpl* channel,
+    std::string* error_desc) {
   channel->SetRemoteIceMode(remote_ice_mode_);
   return true;
 }
 
-bool Transport::NegotiateTransportDescription_w(ContentAction local_role,
-                                                std::string* error_desc) {
-  ASSERT(worker_thread()->IsCurrent());
+bool Transport::NegotiateTransportDescription(ContentAction local_role,
+                                              std::string* error_desc) {
   // TODO(ekr@rtfm.com): This is ICE-specific stuff. Refactor into
   // P2PTransport.
-  const TransportDescription* offer;
-  const TransportDescription* answer;
-
-  if (local_role == CA_OFFER) {
-    offer = local_description_.get();
-    answer = remote_description_.get();
-  } else {
-    offer = remote_description_.get();
-    answer = local_description_.get();
-  }
-
-  TransportProtocol offer_proto = TransportProtocolFromDescription(offer);
-  TransportProtocol answer_proto = TransportProtocolFromDescription(answer);
-
-  // If offered protocol is gice/ice, then we expect to receive matching
-  // protocol in answer, anything else is treated as an error.
-  // HYBRID is not an option when offered specific protocol.
-  // If offered protocol is HYBRID and answered protocol is HYBRID then
-  // gice is preferred protocol.
-  // TODO(mallinath) - Answer from local or remote should't have both ice
-  // and gice support. It should always pick which protocol it wants to use.
-  // Once WebRTC stops supporting gice (for backward compatibility), HYBRID in
-  // answer must be treated as error.
-  if ((offer_proto == ICEPROTO_GOOGLE || offer_proto == ICEPROTO_RFC5245) &&
-      (offer_proto != answer_proto)) {
-    std::ostringstream desc;
-    desc << "Offer and answer protocol mismatch: "
-         << IceProtoToString(offer_proto)
-         << " vs "
-         << IceProtoToString(answer_proto);
-    return BadTransportDescription(desc.str(), error_desc);
-  }
-  protocol_ = answer_proto == ICEPROTO_HYBRID ? ICEPROTO_GOOGLE : answer_proto;
 
   // If transport is in ICEROLE_CONTROLLED and remote end point supports only
   // ice_lite, this local end point should take CONTROLLING role.
   if (ice_role_ == ICEROLE_CONTROLLED &&
       remote_description_->ice_mode == ICEMODE_LITE) {
-    SetIceRole_w(ICEROLE_CONTROLLING);
+    SetIceRole(ICEROLE_CONTROLLING);
   }
 
   // Update remote ice_mode to all existing channels.
@@ -881,70 +380,12 @@ bool Transport::NegotiateTransportDescription_w(ContentAction local_role,
   // between future SetRemote/SetLocal invocations and new channel
   // creation, we have the negotiation state saved until a new
   // negotiation happens.
-  for (auto& iter : channels_) {
-    if (!ApplyNegotiatedTransportDescription_w(iter.second.get(), error_desc))
+  for (const auto& kv : channels_) {
+    if (!ApplyNegotiatedTransportDescription(kv.second, error_desc)) {
       return false;
+    }
   }
   return true;
 }
 
-void Transport::OnMessage(rtc::Message* msg) {
-  switch (msg->message_id) {
-    case MSG_ONSIGNALINGREADY:
-      CallChannels_w(&TransportChannelImpl::OnSignalingReady);
-      break;
-    case MSG_ONREMOTECANDIDATE: {
-        ChannelParams* params = static_cast<ChannelParams*>(msg->pdata);
-        OnRemoteCandidate_w(*params->candidate);
-        delete params;
-      }
-      break;
-    case MSG_CONNECTING:
-      OnConnecting_s();
-      break;
-    case MSG_READSTATE:
-      OnChannelReadableState_s();
-      break;
-    case MSG_WRITESTATE:
-      OnChannelWritableState_s();
-      break;
-    case MSG_REQUESTSIGNALING:
-      OnChannelRequestSignaling_s();
-      break;
-    case MSG_CANDIDATEREADY:
-      OnChannelCandidateReady_s();
-      break;
-    case MSG_ROUTECHANGE: {
-        ChannelParams* params = static_cast<ChannelParams*>(msg->pdata);
-        OnChannelRouteChange_s(params->channel, *params->candidate);
-        delete params;
-      }
-      break;
-    case MSG_CANDIDATEALLOCATIONCOMPLETE:
-      OnChannelCandidatesAllocationDone_s();
-      break;
-    case MSG_ROLECONFLICT:
-      SignalRoleConflict();
-      break;
-    case MSG_COMPLETED:
-      SignalCompleted(this);
-      break;
-    case MSG_FAILED:
-      SignalFailed(this);
-      break;
-  }
-}
-
-// We're GICE if the namespace is NS_GOOGLE_P2P, or if NS_JINGLE_ICE_UDP is
-// used and the GICE ice-option is set.
-TransportProtocol TransportProtocolFromDescription(
-    const TransportDescription* desc) {
-  ASSERT(desc != NULL);
-  if (desc->transport_type == NS_JINGLE_ICE_UDP) {
-    return (desc->HasOption(ICE_OPTION_GICE)) ?
-        ICEPROTO_HYBRID : ICEPROTO_RFC5245;
-  }
-  return ICEPROTO_GOOGLE;
-}
-
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transport.h b/media/webrtc/trunk/webrtc/p2p/base/transport.h
index eda112bb7b..6b4b37d4c5 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/transport.h
@@ -15,15 +15,11 @@
 // state changes (in order to update the manager's state), and forwards
 // requests to begin connecting or to reset to each of the channels.
 //
-// On Threading:  Transport performs work on both the signaling and worker
-// threads.  For subclasses, the rule is that all signaling related calls will
-// be made on the signaling thread and all channel related calls (including
-// signaling for a channel) will be made on the worker thread.  When
-// information needs to be sent between the two threads, this class should do
-// the work (e.g., OnRemoteCandidate).
+// On Threading:  Transport performs work solely on the worker thread, and so
+// its methods should only be called on the worker thread.
 //
-// Note: Subclasses must call DestroyChannels() in their own constructors.
-// It is not possible to do so here because the subclass constructor will
+// Note: Subclasses must call DestroyChannels() in their own destructors.
+// It is not possible to do so here because the subclass destructor will
 // already have run.
 
 #ifndef WEBRTC_P2P_BASE_TRANSPORT_H_
@@ -36,15 +32,11 @@
 #include "webrtc/p2p/base/constants.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/p2p/base/transportinfo.h"
-#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagequeue.h"
+#include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslstreamadapter.h"
 
-namespace rtc {
-class Thread;
-}
-
 namespace cricket {
 
 class PortAllocator;
@@ -53,12 +45,37 @@ class TransportChannelImpl;
 
 typedef std::vector<Candidate> Candidates;
 
-// For "writable" and "readable", we need to differentiate between
-// none, all, and some.
-enum TransportState {
-  TRANSPORT_STATE_NONE = 0,
-  TRANSPORT_STATE_SOME,
-  TRANSPORT_STATE_ALL
+// TODO(deadbeef): Unify with PeerConnectionInterface::IceConnectionState
+// once /talk/ and /webrtc/ are combined, and also switch to ENUM_NAME naming
+// style.
+enum IceConnectionState {
+  kIceConnectionConnecting = 0,
+  kIceConnectionFailed,
+  kIceConnectionConnected,  // Writable, but still checking one or more
+                            // connections
+  kIceConnectionCompleted,
+};
+
+enum DtlsTransportState {
+  // Haven't started negotiating.
+  DTLS_TRANSPORT_NEW = 0,
+  // Have started negotiating.
+  DTLS_TRANSPORT_CONNECTING,
+  // Negotiated, and has a secure connection.
+  DTLS_TRANSPORT_CONNECTED,
+  // Transport is closed.
+  DTLS_TRANSPORT_CLOSED,
+  // Failed due to some error in the handshake process.
+  DTLS_TRANSPORT_FAILED,
+};
+
+// TODO(deadbeef): Unify with PeerConnectionInterface::IceConnectionState
+// once /talk/ and /webrtc/ are combined, and also switch to ENUM_NAME naming
+// style.
+enum IceGatheringState {
+  kIceGatheringNew = 0,
+  kIceGatheringGathering,
+  kIceGatheringComplete,
 };
 
 // Stats that we can return about the connections for a transport channel.
@@ -67,7 +84,7 @@ struct ConnectionInfo {
   ConnectionInfo()
       : best_connection(false),
         writable(false),
-        readable(false),
+        receiving(false),
         timeout(false),
         new_connection(false),
         rtt(0),
@@ -81,7 +98,7 @@ struct ConnectionInfo {
 
   bool best_connection;        // Is this the best connection we have?
   bool writable;               // Has this connection received a STUN response?
-  bool readable;               // Has this connection received a STUN request?
+  bool receiving;              // Has this connection received anything?
   bool timeout;                // Has this connection timed out?
   bool new_connection;         // Is this a newly created connection?
   size_t rtt;                  // The STUN RTT for this connection.
@@ -104,10 +121,10 @@ typedef std::vector<ConnectionInfo> ConnectionInfos;
 
 // Information about a specific channel
 struct TransportChannelStats {
-  int component;
+  int component = 0;
   ConnectionInfos connection_infos;
-  std::string srtp_cipher;
-  std::string ssl_cipher;
+  int srtp_crypto_suite = rtc::SRTP_INVALID_CRYPTO_SUITE;
+  int ssl_cipher_suite = rtc::TLS_NULL_WITH_NULL_NULL;
 };
 
 // Information about all the channels of a transport.
@@ -116,10 +133,22 @@ typedef std::vector<TransportChannelStats> TransportChannelStatsList;
 
 // Information about the stats of a transport.
 struct TransportStats {
-  std::string content_name;
+  std::string transport_name;
   TransportChannelStatsList channel_stats;
 };
 
+// Information about ICE configuration.
+struct IceConfig {
+  // The ICE connection receiving timeout value.
+  // TODO(honghaiz): Remove suffix _ms to be consistent.
+  int receiving_timeout_ms = -1;
+  // Time interval in milliseconds to ping a backup connection when the ICE
+  // channel is strongly connected.
+  int backup_connection_ping_interval = -1;
+  // If true, the most recent port allocator session will keep on running.
+  bool gather_continually = false;
+};
+
 bool BadTransportDescription(const std::string& desc, std::string* err_desc);
 
 bool IceCredentialsChanged(const std::string& old_ufrag,
@@ -127,55 +156,20 @@ bool IceCredentialsChanged(const std::string& old_ufrag,
                            const std::string& new_ufrag,
                            const std::string& new_pwd);
 
-class Transport : public rtc::MessageHandler,
-                  public sigslot::has_slots<> {
+class Transport : public sigslot::has_slots<> {
  public:
-  Transport(rtc::Thread* signaling_thread,
-            rtc::Thread* worker_thread,
-            const std::string& content_name,
-            const std::string& type,
-            PortAllocator* allocator);
+  Transport(const std::string& name, PortAllocator* allocator);
   virtual ~Transport();
 
-  // Returns the signaling thread. The app talks to Transport on this thread.
-  rtc::Thread* signaling_thread() { return signaling_thread_; }
-  // Returns the worker thread. The actual networking is done on this thread.
-  rtc::Thread* worker_thread() { return worker_thread_; }
-
-  // Returns the content_name of this transport.
-  const std::string& content_name() const { return content_name_; }
-  // Returns the type of this transport.
-  const std::string& type() const { return type_; }
+  // Returns the name of this transport.
+  const std::string& name() const { return name_; }
 
   // Returns the port allocator object for this transport.
   PortAllocator* port_allocator() { return allocator_; }
 
-  // Returns the readable and states of this manager.  These bits are the ORs
-  // of the corresponding bits on the managed channels.  Each time one of these
-  // states changes, a signal is raised.
-  // TODO: Replace uses of readable() and writable() with
-  // any_channels_readable() and any_channels_writable().
-  bool readable() const { return any_channels_readable(); }
-  bool writable() const { return any_channels_writable(); }
-  bool was_writable() const { return was_writable_; }
-  bool any_channels_readable() const {
-    return (readable_ == TRANSPORT_STATE_SOME ||
-            readable_ == TRANSPORT_STATE_ALL);
+  bool ready_for_remote_candidates() const {
+    return local_description_set_ && remote_description_set_;
   }
-  bool any_channels_writable() const {
-    return (writable_ == TRANSPORT_STATE_SOME ||
-            writable_ == TRANSPORT_STATE_ALL);
-  }
-  bool all_channels_readable() const {
-    return (readable_ == TRANSPORT_STATE_ALL);
-  }
-  bool all_channels_writable() const {
-    return (writable_ == TRANSPORT_STATE_ALL);
-  }
-  sigslot::signal1<Transport*> SignalReadableState;
-  sigslot::signal1<Transport*> SignalWritableState;
-  sigslot::signal1<Transport*> SignalCompleted;
-  sigslot::signal1<Transport*> SignalFailed;
 
   // Returns whether the client has requested the channels to connect.
   bool connect_requested() const { return connect_requested_; }
@@ -183,34 +177,37 @@ class Transport : public rtc::MessageHandler,
   void SetIceRole(IceRole role);
   IceRole ice_role() const { return ice_role_; }
 
-  void SetIceTiebreaker(uint64 IceTiebreaker) { tiebreaker_ = IceTiebreaker; }
-  uint64 IceTiebreaker() { return tiebreaker_; }
+  void SetIceTiebreaker(uint64_t IceTiebreaker) { tiebreaker_ = IceTiebreaker; }
+  uint64_t IceTiebreaker() { return tiebreaker_; }
+
+  void SetIceConfig(const IceConfig& config);
 
   // Must be called before applying local session description.
-  void SetIdentity(rtc::SSLIdentity* identity);
+  virtual void SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {}
 
-  // Get a copy of the local identity provided by SetIdentity.
-  bool GetIdentity(rtc::SSLIdentity** identity);
+  // Get a copy of the local certificate provided by SetLocalCertificate.
+  virtual bool GetLocalCertificate(
+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
+    return false;
+  }
 
   // Get a copy of the remote certificate in use by the specified channel.
-  bool GetRemoteCertificate(rtc::SSLCertificate** cert);
-
-  TransportProtocol protocol() const { return protocol_; }
+  bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert);
 
   // Create, destroy, and lookup the channels of this type by their components.
   TransportChannelImpl* CreateChannel(int component);
-  // Note: GetChannel may lead to race conditions, since the mutex is not held
-  // after the pointer is returned.
+
   TransportChannelImpl* GetChannel(int component);
-  // Note: HasChannel does not lead to race conditions, unlike GetChannel.
+
   bool HasChannel(int component) {
     return (NULL != GetChannel(component));
   }
   bool HasChannels();
+
   void DestroyChannel(int component);
 
   // Set the local TransportDescription to be used by TransportChannels.
-  // This should be called before ConnectChannels().
   bool SetLocalTransportDescription(const TransportDescription& description,
                                     ContentAction action,
                                     std::string* error_desc);
@@ -220,10 +217,13 @@ class Transport : public rtc::MessageHandler,
                                      ContentAction action,
                                      std::string* error_desc);
 
-  // Tells all current and future channels to start connecting.  When the first
-  // channel begins connecting, the following signal is raised.
+  // Tells all current and future channels to start connecting.
   void ConnectChannels();
-  sigslot::signal1<Transport*> SignalConnecting;
+
+  // Tells channels to start gathering candidates if necessary.
+  // Should be called after ConnectChannels() has been called at least once,
+  // which will happen in SetLocalTransportDescription.
+  void MaybeStartGathering();
 
   // Resets all of the channels back to their initial state.  They are no
   // longer connecting.
@@ -234,35 +234,21 @@ class Transport : public rtc::MessageHandler,
 
   bool GetStats(TransportStats* stats);
 
-  // Before any stanza is sent, the manager will request signaling.  Once
-  // signaling is available, the client should call OnSignalingReady.  Once
-  // this occurs, the transport (or its channels) can send any waiting stanzas.
-  // OnSignalingReady invokes OnTransportSignalingReady and then forwards this
-  // signal to each channel.
-  sigslot::signal1<Transport*> SignalRequestSignaling;
-  void OnSignalingReady();
-
-  // Handles sending of ready candidates and receiving of remote candidates.
-  sigslot::signal2<Transport*,
-                   const std::vector<Candidate>&> SignalCandidatesReady;
-
-  sigslot::signal1<Transport*> SignalCandidatesAllocationDone;
-  void OnRemoteCandidates(const std::vector<Candidate>& candidates);
+  // Called when one or more candidates are ready from the remote peer.
+  bool AddRemoteCandidates(const std::vector<Candidate>& candidates,
+                           std::string* error);
 
   // If candidate is not acceptable, returns false and sets error.
   // Call this before calling OnRemoteCandidates.
   virtual bool VerifyCandidate(const Candidate& candidate,
                                std::string* error);
 
-  // Signals when the best connection for a channel changes.
-  sigslot::signal3<Transport*,
-                   int,  // component
-                   const Candidate&> SignalRouteChange;
+  virtual bool GetSslRole(rtc::SSLRole* ssl_role) const { return false; }
 
-  // Forwards the signal from TransportChannel to BaseSession.
-  sigslot::signal0<> SignalRoleConflict;
-
-  virtual bool GetSslRole(rtc::SSLRole* ssl_role) const;
+  // Must be called before channel is starting to connect.
+  virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
+    return false;
+  }
 
  protected:
   // These are called by Create/DestroyChannel above in order to create or
@@ -270,9 +256,6 @@ class Transport : public rtc::MessageHandler,
   virtual TransportChannelImpl* CreateTransportChannel(int component) = 0;
   virtual void DestroyTransportChannel(TransportChannelImpl* channel) = 0;
 
-  // Informs the subclass that we received the signaling ready message.
-  virtual void OnTransportSignalingReady() {}
-
   // The current local transport description, for use by derived classes
   // when performing transport description negotiation.
   const TransportDescription* local_description() const {
@@ -285,167 +268,58 @@ class Transport : public rtc::MessageHandler,
     return remote_description_.get();
   }
 
-  virtual void SetIdentity_w(rtc::SSLIdentity* identity) {}
-
-  virtual bool GetIdentity_w(rtc::SSLIdentity** identity) {
-    return false;
-  }
-
   // Pushes down the transport parameters from the local description, such
   // as the ICE ufrag and pwd.
   // Derived classes can override, but must call the base as well.
-  virtual bool ApplyLocalTransportDescription_w(TransportChannelImpl* channel,
-                                                std::string* error_desc);
+  virtual bool ApplyLocalTransportDescription(TransportChannelImpl* channel,
+                                              std::string* error_desc);
 
   // Pushes down remote ice credentials from the remote description to the
   // transport channel.
-  virtual bool ApplyRemoteTransportDescription_w(TransportChannelImpl* ch,
-                                                 std::string* error_desc);
+  virtual bool ApplyRemoteTransportDescription(TransportChannelImpl* ch,
+                                               std::string* error_desc);
 
   // Negotiates the transport parameters based on the current local and remote
-  // transport description, such at the version of ICE to use, and whether DTLS
+  // transport description, such as the ICE role to use, and whether DTLS
   // should be activated.
   // Derived classes can negotiate their specific parameters here, but must call
   // the base as well.
-  virtual bool NegotiateTransportDescription_w(ContentAction local_role,
-                                               std::string* error_desc);
+  virtual bool NegotiateTransportDescription(ContentAction local_role,
+                                             std::string* error_desc);
 
   // Pushes down the transport parameters obtained via negotiation.
   // Derived classes can set their specific parameters here, but must call the
   // base as well.
-  virtual bool ApplyNegotiatedTransportDescription_w(
-      TransportChannelImpl* channel, std::string* error_desc);
-
-  virtual bool GetSslRole_w(rtc::SSLRole* ssl_role) const {
-    return false;
-  }
+  virtual bool ApplyNegotiatedTransportDescription(
+      TransportChannelImpl* channel,
+      std::string* error_desc);
 
  private:
-  struct ChannelMapEntry {
-    ChannelMapEntry() : impl_(NULL), candidates_allocated_(false), ref_(0) {}
-    explicit ChannelMapEntry(TransportChannelImpl *impl)
-        : impl_(impl),
-          candidates_allocated_(false),
-          ref_(0) {
-    }
-
-    void AddRef() { ++ref_; }
-    void DecRef() {
-      ASSERT(ref_ > 0);
-      --ref_;
-    }
-    int ref() const { return ref_; }
-
-    TransportChannelImpl* get() const { return impl_; }
-    TransportChannelImpl* operator->() const  { return impl_; }
-    void set_candidates_allocated(bool status) {
-      candidates_allocated_ = status;
-    }
-    bool candidates_allocated() const { return candidates_allocated_; }
-
-  private:
-    TransportChannelImpl *impl_;
-    bool candidates_allocated_;
-    int ref_;
-  };
-
-  // Candidate component => ChannelMapEntry
-  typedef std::map<int, ChannelMapEntry> ChannelMap;
-
-  // Called when the state of a channel changes.
-  void OnChannelReadableState(TransportChannel* channel);
-  void OnChannelWritableState(TransportChannel* channel);
-
-  // Called when a channel requests signaling.
-  void OnChannelRequestSignaling(TransportChannelImpl* channel);
-
-  // Called when a candidate is ready from remote peer.
-  void OnRemoteCandidate(const Candidate& candidate);
-  // Called when a candidate is ready from channel.
-  void OnChannelCandidateReady(TransportChannelImpl* channel,
-                               const Candidate& candidate);
-  void OnChannelRouteChange(TransportChannel* channel,
-                            const Candidate& remote_candidate);
-  void OnChannelCandidatesAllocationDone(TransportChannelImpl* channel);
-  // Called when there is ICE role change.
-  void OnRoleConflict(TransportChannelImpl* channel);
-  // Called when the channel removes a connection.
-  void OnChannelConnectionRemoved(TransportChannelImpl* channel);
-
-  // Dispatches messages to the appropriate handler (below).
-  void OnMessage(rtc::Message* msg);
-
-  // These are versions of the above methods that are called only on a
-  // particular thread (s = signaling, w = worker).  The above methods post or
-  // send a message to invoke this version.
-  TransportChannelImpl* CreateChannel_w(int component);
-  void DestroyChannel_w(int component);
-  void ConnectChannels_w();
-  void ResetChannels_w();
-  void DestroyAllChannels_w();
-  void OnRemoteCandidate_w(const Candidate& candidate);
-  void OnChannelReadableState_s();
-  void OnChannelWritableState_s();
-  void OnChannelRequestSignaling_s();
-  void OnConnecting_s();
-  void OnChannelRouteChange_s(const TransportChannel* channel,
-                              const Candidate& remote_candidate);
-  void OnChannelCandidatesAllocationDone_s();
+  // Candidate component => TransportChannelImpl*
+  typedef std::map<int, TransportChannelImpl*> ChannelMap;
 
   // Helper function that invokes the given function on every channel.
   typedef void (TransportChannelImpl::* TransportChannelFunc)();
-  void CallChannels_w(TransportChannelFunc func);
+  void CallChannels(TransportChannelFunc func);
 
-  // Computes the OR of the channel's read or write state (argument picks).
-  TransportState GetTransportState_s(bool read);
-
-  void OnChannelCandidateReady_s();
-
-  void SetIceRole_w(IceRole role);
-  void SetRemoteIceMode_w(IceMode mode);
-  bool SetLocalTransportDescription_w(const TransportDescription& desc,
-                                      ContentAction action,
-                                      std::string* error_desc);
-  bool SetRemoteTransportDescription_w(const TransportDescription& desc,
-                                       ContentAction action,
-                                       std::string* error_desc);
-  bool GetStats_w(TransportStats* infos);
-  bool GetRemoteCertificate_w(rtc::SSLCertificate** cert);
-
-  // Sends SignalCompleted if we are now in that state.
-  void MaybeCompleted_w();
-
-  rtc::Thread* const signaling_thread_;
-  rtc::Thread* const worker_thread_;
-  const std::string content_name_;
-  const std::string type_;
+  const std::string name_;
   PortAllocator* const allocator_;
-  bool destroyed_;
-  TransportState readable_;
-  TransportState writable_;
-  bool was_writable_;
-  bool connect_requested_;
-  IceRole ice_role_;
-  uint64 tiebreaker_;
-  TransportProtocol protocol_;
-  IceMode remote_ice_mode_;
+  bool channels_destroyed_ = false;
+  bool connect_requested_ = false;
+  IceRole ice_role_ = ICEROLE_UNKNOWN;
+  uint64_t tiebreaker_ = 0;
+  IceMode remote_ice_mode_ = ICEMODE_FULL;
+  IceConfig ice_config_;
   rtc::scoped_ptr<TransportDescription> local_description_;
   rtc::scoped_ptr<TransportDescription> remote_description_;
+  bool local_description_set_ = false;
+  bool remote_description_set_ = false;
 
-  // TODO(tommi): Make sure we only use this on the worker thread.
   ChannelMap channels_;
-  // Buffers the ready_candidates so that SignalCanidatesReady can
-  // provide them in multiples.
-  std::vector<Candidate> ready_candidates_;
-  // Protects changes to channels and messages
-  rtc::CriticalSection crit_;
 
-  DISALLOW_EVIL_CONSTRUCTORS(Transport);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Transport);
 };
 
-// Extract a TransportProtocol from a TransportDescription.
-TransportProtocol TransportProtocolFromDescription(
-    const TransportDescription* desc);
 
 }  // namespace cricket
 
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transport_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/transport_unittest.cc
index ba11678ece..1f66a47c99 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transport_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/transport_unittest.cc
@@ -11,8 +11,7 @@
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/network.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/p2p/base/p2ptransport.h"
 
 using cricket::Candidate;
@@ -35,17 +34,7 @@ class TransportTest : public testing::Test,
                       public sigslot::has_slots<> {
  public:
   TransportTest()
-      : thread_(rtc::Thread::Current()),
-        transport_(new FakeTransport(
-            thread_, thread_, "test content name", NULL)),
-        channel_(NULL),
-        connecting_signalled_(false),
-        completed_(false),
-        failed_(false) {
-    transport_->SignalConnecting.connect(this, &TransportTest::OnConnecting);
-    transport_->SignalCompleted.connect(this, &TransportTest::OnCompleted);
-    transport_->SignalFailed.connect(this, &TransportTest::OnFailed);
-  }
+      : transport_(new FakeTransport("test content name")), channel_(NULL) {}
   ~TransportTest() {
     transport_->DestroyAllChannels();
   }
@@ -63,52 +52,17 @@ class TransportTest : public testing::Test,
   }
 
  protected:
-  void OnConnecting(Transport* transport) {
-    connecting_signalled_ = true;
-  }
-  void OnCompleted(Transport* transport) {
-    completed_ = true;
-  }
-  void OnFailed(Transport* transport) {
-    failed_ = true;
-  }
-
-  rtc::Thread* thread_;
   rtc::scoped_ptr<FakeTransport> transport_;
   FakeTransportChannel* channel_;
-  bool connecting_signalled_;
-  bool completed_;
-  bool failed_;
 };
 
-// Test that calling ConnectChannels triggers an OnConnecting signal.
-TEST_F(TransportTest, TestConnectChannelsDoesSignal) {
-  EXPECT_TRUE(SetupChannel());
-  transport_->ConnectChannels();
-  EXPECT_FALSE(connecting_signalled_);
-
-  EXPECT_TRUE_WAIT(connecting_signalled_, 100);
-}
-
-// Test that DestroyAllChannels kills any pending OnConnecting signals.
-TEST_F(TransportTest, TestDestroyAllClearsPosts) {
-  EXPECT_TRUE(transport_->CreateChannel(1) != NULL);
-
-  transport_->ConnectChannels();
-  transport_->DestroyAllChannels();
-
-  thread_->ProcessMessages(0);
-  EXPECT_FALSE(connecting_signalled_);
-}
-
 // This test verifies channels are created with proper ICE
 // role, tiebreaker and remote ice mode and credentials after offer and
 // answer negotiations.
 TEST_F(TransportTest, TestChannelIceParameters) {
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
   transport_->SetIceTiebreaker(99U);
-  cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription local_desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
                                                        cricket::CA_OFFER,
                                                        NULL));
@@ -119,8 +73,7 @@ TEST_F(TransportTest, TestChannelIceParameters) {
   EXPECT_EQ(kIceUfrag1, channel_->ice_ufrag());
   EXPECT_EQ(kIcePwd1, channel_->ice_pwd());
 
-  cricket::TransportDescription remote_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription remote_desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
                                                         cricket::CA_ANSWER,
                                                         NULL));
@@ -150,8 +103,7 @@ TEST_F(TransportTest, TestIceControlledToControllingOnIceRestart) {
   EXPECT_TRUE(SetupChannel());
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLED);
 
-  cricket::TransportDescription desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(desc,
                                                         cricket::CA_OFFER,
                                                         NULL));
@@ -160,8 +112,7 @@ TEST_F(TransportTest, TestIceControlledToControllingOnIceRestart) {
                                                        NULL));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLED, transport_->ice_role());
 
-  cricket::TransportDescription new_local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag2, kIcePwd2);
+  cricket::TransportDescription new_local_desc(kIceUfrag2, kIcePwd2);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(new_local_desc,
                                                        cricket::CA_OFFER,
                                                        NULL));
@@ -175,8 +126,7 @@ TEST_F(TransportTest, TestIceControllingToControlledOnIceRestart) {
   EXPECT_TRUE(SetupChannel());
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
 
-  cricket::TransportDescription desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(desc,
                                                        cricket::CA_OFFER,
                                                        NULL));
@@ -185,8 +135,7 @@ TEST_F(TransportTest, TestIceControllingToControlledOnIceRestart) {
                                                         NULL));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, transport_->ice_role());
 
-  cricket::TransportDescription new_local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag2, kIcePwd2);
+  cricket::TransportDescription new_local_desc(kIceUfrag2, kIcePwd2);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(new_local_desc,
                                                        cricket::CA_ANSWER,
                                                        NULL));
@@ -200,14 +149,13 @@ TEST_F(TransportTest, TestIceControllingOnIceRestartIfRemoteIsIceLite) {
   EXPECT_TRUE(SetupChannel());
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
 
-  cricket::TransportDescription desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(desc,
                                                        cricket::CA_OFFER,
                                                        NULL));
 
   cricket::TransportDescription remote_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
+      std::vector<std::string>(),
       kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE,
       cricket::CONNECTIONROLE_NONE, NULL, cricket::Candidates());
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
@@ -216,8 +164,7 @@ TEST_F(TransportTest, TestIceControllingOnIceRestartIfRemoteIsIceLite) {
 
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, transport_->ice_role());
 
-  cricket::TransportDescription new_local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag2, kIcePwd2);
+  cricket::TransportDescription new_local_desc(kIceUfrag2, kIcePwd2);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(new_local_desc,
                                                        cricket::CA_ANSWER,
                                                        NULL));
@@ -225,55 +172,17 @@ TEST_F(TransportTest, TestIceControllingOnIceRestartIfRemoteIsIceLite) {
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel_->GetIceRole());
 }
 
-// This test verifies that the Completed and Failed states can be reached.
-TEST_F(TransportTest, TestChannelCompletedAndFailed) {
-  transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
-  cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
-  ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
-                                                       cricket::CA_OFFER,
-                                                       NULL));
-  EXPECT_TRUE(SetupChannel());
-
-  cricket::TransportDescription remote_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
-                                                        cricket::CA_ANSWER,
-                                                        NULL));
-
-  channel_->SetConnectionCount(2);
-  channel_->SignalCandidatesAllocationDone(channel_);
-  channel_->SetWritable(true);
-  EXPECT_TRUE_WAIT(transport_->all_channels_writable(), 100);
-  // ICE is not yet completed because there is still more than one connection.
-  EXPECT_FALSE(completed_);
-  EXPECT_FALSE(failed_);
-
-  // When the connection count drops to 1, SignalCompleted should be emitted,
-  // and completed() should be true.
-  channel_->SetConnectionCount(1);
-  EXPECT_TRUE_WAIT(completed_, 100);
-  completed_ = false;
-
-  // When the connection count drops to 0, SignalFailed should be emitted, and
-  // completed() should be false.
-  channel_->SetConnectionCount(0);
-  EXPECT_TRUE_WAIT(failed_, 100);
-  EXPECT_FALSE(completed_);
-}
-
 // Tests channel role is reversed after receiving ice-lite from remote.
 TEST_F(TransportTest, TestSetRemoteIceLiteInOffer) {
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLED);
   cricket::TransportDescription remote_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
+      std::vector<std::string>(),
       kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE,
       cricket::CONNECTIONROLE_ACTPASS, NULL, cricket::Candidates());
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
                                                         cricket::CA_OFFER,
                                                         NULL));
-  cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription local_desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
                                                        cricket::CA_ANSWER,
                                                        NULL));
@@ -286,8 +195,7 @@ TEST_F(TransportTest, TestSetRemoteIceLiteInOffer) {
 // Tests ice-lite in remote answer.
 TEST_F(TransportTest, TestSetRemoteIceLiteInAnswer) {
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
-  cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
+  cricket::TransportDescription local_desc(kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
                                                        cricket::CA_OFFER,
                                                        NULL));
@@ -297,7 +205,7 @@ TEST_F(TransportTest, TestSetRemoteIceLiteInAnswer) {
   // Channels will be created in ICEFULL_MODE.
   EXPECT_EQ(cricket::ICEMODE_FULL, channel_->remote_ice_mode());
   cricket::TransportDescription remote_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
+      std::vector<std::string>(),
       kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE,
       cricket::CONNECTIONROLE_NONE, NULL, cricket::Candidates());
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
@@ -321,3 +229,4 @@ TEST_F(TransportTest, TestGetStats) {
   ASSERT_EQ(1U, stats.channel_stats.size());
   EXPECT_EQ(1, stats.channel_stats[0].component);
 }
+
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportchannel.cc b/media/webrtc/trunk/webrtc/p2p/base/transportchannel.cc
index 16ae27d164..6cbe2b7583 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportchannel.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportchannel.cc
@@ -9,35 +9,62 @@
  */
 
 #include <sstream>
+#include "webrtc/p2p/base/common.h"
 #include "webrtc/p2p/base/transportchannel.h"
 
 namespace cricket {
 
 std::string TransportChannel::ToString() const {
-  const char READABLE_ABBREV[2] = { '_', 'R' };
+  const char RECEIVING_ABBREV[2] = { '_', 'R' };
   const char WRITABLE_ABBREV[2] = { '_', 'W' };
   std::stringstream ss;
-  ss << "Channel[" << content_name_
-     << "|" << component_
-     << "|" << READABLE_ABBREV[readable_] << WRITABLE_ABBREV[writable_] << "]";
+  ss << "Channel[" << transport_name_ << "|" << component_ << "|"
+     << RECEIVING_ABBREV[receiving_] << WRITABLE_ABBREV[writable_] << "]";
   return ss.str();
 }
 
-void TransportChannel::set_readable(bool readable) {
-  if (readable_ != readable) {
-    readable_ = readable;
-    SignalReadableState(this);
+void TransportChannel::set_receiving(bool receiving) {
+  if (receiving_ == receiving) {
+    return;
   }
+  receiving_ = receiving;
+  SignalReceivingState(this);
 }
 
 void TransportChannel::set_writable(bool writable) {
-  if (writable_ != writable) {
-    writable_ = writable;
-    if (writable_) {
-      SignalReadyToSend(this);
-    }
-    SignalWritableState(this);
+  if (writable_ == writable) {
+    return;
   }
+  LOG_J(LS_VERBOSE, this) << "set_writable from:" << writable_ << " to "
+                          << writable;
+  writable_ = writable;
+  if (writable_) {
+    SignalReadyToSend(this);
+  }
+  SignalWritableState(this);
+}
+
+void TransportChannel::set_dtls_state(DtlsTransportState state) {
+  if (dtls_state_ == state) {
+    return;
+  }
+  LOG_J(LS_VERBOSE, this) << "set_dtls_state from:" << dtls_state_ << " to "
+                          << state;
+  dtls_state_ = state;
+  SignalDtlsState(this, state);
+}
+
+bool TransportChannel::SetSrtpCryptoSuites(const std::vector<int>& ciphers) {
+  return false;
+}
+
+// TODO(guoweis): Remove this function once everything is moved away.
+bool TransportChannel::SetSrtpCiphers(const std::vector<std::string>& ciphers) {
+  std::vector<int> crypto_suites;
+  for (const auto cipher : ciphers) {
+    crypto_suites.push_back(rtc::SrtpCryptoSuiteFromName(cipher));
+  }
+  return SetSrtpCryptoSuites(crypto_suites);
 }
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportchannel.h b/media/webrtc/trunk/webrtc/p2p/base/transportchannel.h
index 3d32b63267..b91af139b7 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportchannel.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportchannel.h
@@ -37,16 +37,24 @@ enum PacketFlags {
 };
 
 // Used to indicate channel's connection state.
-enum TransportChannelState { STATE_CONNECTING, STATE_COMPLETED, STATE_FAILED };
+enum TransportChannelState {
+  STATE_INIT,
+  STATE_CONNECTING,  // Will enter this state once a connection is created
+  STATE_COMPLETED,
+  STATE_FAILED
+};
 
 // A TransportChannel represents one logical stream of packets that are sent
 // between the two sides of a session.
+// TODO(deadbeef): This interface currently represents the unity of an ICE
+// transport and a DTLS transport. They need to be separated apart.
 class TransportChannel : public sigslot::has_slots<> {
  public:
-  explicit TransportChannel(const std::string& content_name, int component)
-      : content_name_(content_name),
+  TransportChannel(const std::string& transport_name, int component)
+      : transport_name_(transport_name),
         component_(component),
-        readable_(false), writable_(false) {}
+        writable_(false),
+        receiving_(false) {}
   virtual ~TransportChannel() {}
 
   // TODO(guoweis) - Make this pure virtual once all subclasses of
@@ -59,18 +67,21 @@ class TransportChannel : public sigslot::has_slots<> {
   // Returns the session id of this channel.
   virtual const std::string SessionId() const { return std::string(); }
 
-  const std::string& content_name() const { return content_name_; }
+  const std::string& transport_name() const { return transport_name_; }
   int component() const { return component_; }
 
-  // Returns the readable and states of this channel.  Each time one of these
-  // states changes, a signal is raised.  These states are aggregated by the
-  // TransportManager.
-  bool readable() const { return readable_; }
+  // Returns the states of this channel.  Each time one of these states changes,
+  // a signal is raised.  These states are aggregated by the TransportManager.
   bool writable() const { return writable_; }
-  sigslot::signal1<TransportChannel*> SignalReadableState;
+  bool receiving() const { return receiving_; }
+  DtlsTransportState dtls_state() const { return dtls_state_; }
   sigslot::signal1<TransportChannel*> SignalWritableState;
   // Emitted when the TransportChannel's ability to send has changed.
   sigslot::signal1<TransportChannel*> SignalReadyToSend;
+  sigslot::signal1<TransportChannel*> SignalReceivingState;
+  // Emitted whenever DTLS-SRTP is setup which will require setting up a new
+  // SRTP context.
+  sigslot::signal2<TransportChannel*, DtlsTransportState> SignalDtlsState;
 
   // Attempts to send the given packet.  The return value is < 0 on failure.
   // TODO: Remove the default argument once channel code is updated.
@@ -97,33 +108,44 @@ class TransportChannel : public sigslot::has_slots<> {
   // Default implementation.
   virtual bool GetSslRole(rtc::SSLRole* role) const = 0;
 
-  // Sets up the ciphers to use for DTLS-SRTP.
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) = 0;
+  // Sets up the ciphers to use for DTLS-SRTP. TODO(guoweis): Make this pure
+  // virtual once all dependencies have implementation.
+  virtual bool SetSrtpCryptoSuites(const std::vector<int>& ciphers);
+
+  // Keep the original one for backward compatibility until all dependencies
+  // move away. TODO(guoweis): Remove this function.
+  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers);
 
   // Finds out which DTLS-SRTP cipher was negotiated.
-  virtual bool GetSrtpCipher(std::string* cipher) = 0;
+  // TODO(guoweis): Remove this once all dependencies implement this.
+  virtual bool GetSrtpCryptoSuite(int* cipher) { return false; }
 
   // Finds out which DTLS cipher was negotiated.
-  virtual bool GetSslCipher(std::string* cipher) = 0;
+  // TODO(guoweis): Remove this once all dependencies implement this.
+  virtual bool GetSslCipherSuite(int* cipher) { return false; }
 
-  // Gets a copy of the local SSL identity, owned by the caller.
-  virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const = 0;
+  // Gets the local RTCCertificate used for DTLS.
+  virtual rtc::scoped_refptr<rtc::RTCCertificate>
+  GetLocalCertificate() const = 0;
 
   // Gets a copy of the remote side's SSL certificate, owned by the caller.
-  virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const = 0;
+  virtual bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert) const = 0;
 
   // Allows key material to be extracted for external encryption.
   virtual bool ExportKeyingMaterial(const std::string& label,
-      const uint8* context,
-      size_t context_len,
-      bool use_context,
-      uint8* result,
-      size_t result_len) = 0;
+                                    const uint8_t* context,
+                                    size_t context_len,
+                                    bool use_context,
+                                    uint8_t* result,
+                                    size_t result_len) = 0;
 
   // Signalled each time a packet is received on this channel.
   sigslot::signal5<TransportChannel*, const char*,
                    size_t, const rtc::PacketTime&, int> SignalReadPacket;
 
+  // Signalled each time a packet is sent on this channel.
+  sigslot::signal2<TransportChannel*, const rtc::SentPacket&> SignalSentPacket;
+
   // This signal occurs when there is a change in the way that packets are
   // being routed, i.e. to a different remote location. The candidate
   // indicates where and how we are currently sending media.
@@ -136,21 +158,24 @@ class TransportChannel : public sigslot::has_slots<> {
   std::string ToString() const;
 
  protected:
-  // Sets the readable state, signaling if necessary.
-  void set_readable(bool readable);
-
   // Sets the writable state, signaling if necessary.
   void set_writable(bool writable);
 
+  // Sets the receiving state, signaling if necessary.
+  void set_receiving(bool receiving);
+
+  // Sets the DTLS state, signaling if necessary.
+  void set_dtls_state(DtlsTransportState state);
 
  private:
   // Used mostly for debugging.
-  std::string content_name_;
+  std::string transport_name_;
   int component_;
-  bool readable_;
   bool writable_;
+  bool receiving_;
+  DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW;
 
-  DISALLOW_EVIL_CONSTRUCTORS(TransportChannel);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TransportChannel);
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportchannelimpl.h b/media/webrtc/trunk/webrtc/p2p/base/transportchannelimpl.h
index 6c2eac8ce2..8d4d4bb728 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportchannelimpl.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportchannelimpl.h
@@ -21,13 +21,20 @@ namespace cricket {
 
 class Candidate;
 
+// TODO(pthatcher): Remove this once it's no longer used in
+// remoting/protocol/libjingle_transport_factory.cc
+enum IceProtocolType {
+  ICEPROTO_RFC5245  // Standard RFC 5245 version of ICE.
+};
+
 // Base class for real implementations of TransportChannel.  This includes some
 // methods called only by Transport, which do not need to be exposed to the
 // client.
 class TransportChannelImpl : public TransportChannel {
  public:
-  explicit TransportChannelImpl(const std::string& content_name, int component)
-      : TransportChannel(content_name, component) {}
+  explicit TransportChannelImpl(const std::string& transport_name,
+                                int component)
+      : TransportChannel(transport_name, component) {}
 
   // Returns the transport that created this channel.
   virtual Transport* GetTransport() = 0;
@@ -35,10 +42,10 @@ class TransportChannelImpl : public TransportChannel {
   // For ICE channels.
   virtual IceRole GetIceRole() const = 0;
   virtual void SetIceRole(IceRole role) = 0;
-  virtual void SetIceTiebreaker(uint64 tiebreaker) = 0;
-  // To toggle G-ICE/ICE.
-  virtual bool GetIceProtocolType(IceProtocolType* type) const = 0;
-  virtual void SetIceProtocolType(IceProtocolType type) = 0;
+  virtual void SetIceTiebreaker(uint64_t tiebreaker) = 0;
+  // TODO(pthatcher): Remove this once it's no longer called in
+  // remoting/protocol/libjingle_transport_factory.cc
+  virtual void SetIceProtocolType(IceProtocolType type) {}
   // SetIceCredentials only need to be implemented by the ICE
   // transport channels. Non-ICE transport channels can just ignore.
   // The ufrag and pwd should be set before the Connect() is called.
@@ -52,17 +59,16 @@ class TransportChannelImpl : public TransportChannel {
   // SetRemoteIceMode must be implemented only by the ICE transport channels.
   virtual void SetRemoteIceMode(IceMode mode) = 0;
 
+  virtual void SetIceConfig(const IceConfig& config) = 0;
+
   // Begins the process of attempting to make a connection to the other client.
   virtual void Connect() = 0;
 
-  // Resets this channel back to the initial state (i.e., not connecting).
-  virtual void Reset() = 0;
+  // Start gathering candidates if not already started, or if an ICE restart
+  // occurred.
+  virtual void MaybeStartGathering() = 0;
 
-  // Allows an individual channel to request signaling and be notified when it
-  // is ready.  This is useful if the individual named channels have need to
-  // send their own transport-info stanzas.
-  sigslot::signal1<TransportChannelImpl*> SignalRequestSignaling;
-  virtual void OnSignalingReady() = 0;
+  sigslot::signal1<TransportChannelImpl*> SignalGatheringState;
 
   // Handles sending and receiving of candidates.  The Transport
   // receives the candidates and may forward them to the relevant
@@ -72,27 +78,23 @@ class TransportChannelImpl : public TransportChannel {
   // channel, they cannot return an error if the message is invalid.
   // It is assumed that the Transport will have checked validity
   // before forwarding.
-  sigslot::signal2<TransportChannelImpl*,
-                   const Candidate&> SignalCandidateReady;
-  virtual void OnCandidate(const Candidate& candidate) = 0;
+  sigslot::signal2<TransportChannelImpl*, const Candidate&>
+      SignalCandidateGathered;
+  virtual void AddRemoteCandidate(const Candidate& candidate) = 0;
+
+  virtual IceGatheringState gathering_state() const = 0;
 
   // DTLS methods
-  // Set DTLS local identity.  The identity object is not copied, but the caller
-  // retains ownership and must delete it after this TransportChannelImpl is
-  // destroyed.
-  // TODO(bemasc): Fix the ownership semantics of this method.
-  virtual bool SetLocalIdentity(rtc::SSLIdentity* identity) = 0;
+  virtual bool SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) = 0;
 
   // Set DTLS Remote fingerprint. Must be after local identity set.
   virtual bool SetRemoteFingerprint(const std::string& digest_alg,
-    const uint8* digest,
-    size_t digest_len) = 0;
+                                    const uint8_t* digest,
+                                    size_t digest_len) = 0;
 
   virtual bool SetSslRole(rtc::SSLRole role) = 0;
 
-  // TransportChannel is forwarding this signal from PortAllocatorSession.
-  sigslot::signal1<TransportChannelImpl*> SignalCandidatesAllocationDone;
-
   // Invoked when there is conflict in the ICE role between local and remote
   // agents.
   sigslot::signal1<TransportChannelImpl*> SignalRoleConflict;
@@ -102,7 +104,7 @@ class TransportChannelImpl : public TransportChannel {
   sigslot::signal1<TransportChannelImpl*> SignalConnectionRemoved;
 
  private:
-  DISALLOW_EVIL_CONSTRUCTORS(TransportChannelImpl);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TransportChannelImpl);
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportchannelproxy.cc b/media/webrtc/trunk/webrtc/p2p/base/transportchannelproxy.cc
deleted file mode 100644
index 79772514c6..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/transportchannelproxy.cc
+++ /dev/null
@@ -1,276 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/p2p/base/transport.h"
-#include "webrtc/p2p/base/transportchannelimpl.h"
-#include "webrtc/p2p/base/transportchannelproxy.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/thread.h"
-
-namespace cricket {
-
-enum {
-  MSG_UPDATESTATE,
-};
-
-TransportChannelProxy::TransportChannelProxy(const std::string& content_name,
-                                             int component)
-    : TransportChannel(content_name, component),
-      impl_(NULL) {
-  worker_thread_ = rtc::Thread::Current();
-}
-
-TransportChannelProxy::~TransportChannelProxy() {
-  // Clearing any pending signal.
-  worker_thread_->Clear(this);
-  if (impl_) {
-    impl_->GetTransport()->DestroyChannel(impl_->component());
-  }
-}
-
-void TransportChannelProxy::SetImplementation(TransportChannelImpl* impl) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-
-  if (impl == impl_) {
-    // Ignore if the |impl| has already been set.
-    LOG(LS_WARNING) << "Ignored TransportChannelProxy::SetImplementation call "
-                    << "with a same impl as the existing one.";
-    return;
-  }
-
-  // Destroy any existing impl_.
-  if (impl_) {
-    impl_->GetTransport()->DestroyChannel(impl_->component());
-  }
-
-  // Adopt the supplied impl, and connect to its signals.
-  impl_ = impl;
-
-  if (impl_) {
-    impl_->SignalReadableState.connect(
-        this, &TransportChannelProxy::OnReadableState);
-    impl_->SignalWritableState.connect(
-        this, &TransportChannelProxy::OnWritableState);
-    impl_->SignalReadPacket.connect(
-        this, &TransportChannelProxy::OnReadPacket);
-    impl_->SignalReadyToSend.connect(
-        this, &TransportChannelProxy::OnReadyToSend);
-    impl_->SignalRouteChange.connect(
-        this, &TransportChannelProxy::OnRouteChange);
-    for (const auto& pair : options_) {
-      impl_->SetOption(pair.first, pair.second);
-    }
-
-    // Push down the SRTP ciphers, if any were set.
-    if (!pending_srtp_ciphers_.empty()) {
-      impl_->SetSrtpCiphers(pending_srtp_ciphers_);
-    }
-  }
-
-  // Post ourselves a message to see if we need to fire state callbacks.
-  worker_thread_->Post(this, MSG_UPDATESTATE);
-}
-
-int TransportChannelProxy::SendPacket(const char* data, size_t len,
-                                      const rtc::PacketOptions& options,
-                                      int flags) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  // Fail if we don't have an impl yet.
-  if (!impl_) {
-    return -1;
-  }
-  return impl_->SendPacket(data, len, options, flags);
-}
-
-int TransportChannelProxy::SetOption(rtc::Socket::Option opt, int value) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  options_.push_back(OptionPair(opt, value));
-  if (!impl_) {
-    return 0;
-  }
-  return impl_->SetOption(opt, value);
-}
-
-bool TransportChannelProxy::GetOption(rtc::Socket::Option opt, int* value) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (impl_) {
-    return impl_->GetOption(opt, value);
-  }
-
-  for (const auto& pair : options_) {
-    if (pair.first == opt) {
-      *value = pair.second;
-      return true;
-    }
-  }
-  return false;
-}
-
-int TransportChannelProxy::GetError() {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return 0;
-  }
-  return impl_->GetError();
-}
-
-TransportChannelState TransportChannelProxy::GetState() const {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return TransportChannelState::STATE_CONNECTING;
-  }
-  return impl_->GetState();
-}
-
-bool TransportChannelProxy::GetStats(ConnectionInfos* infos) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->GetStats(infos);
-}
-
-bool TransportChannelProxy::IsDtlsActive() const {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->IsDtlsActive();
-}
-
-bool TransportChannelProxy::GetSslRole(rtc::SSLRole* role) const {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->GetSslRole(role);
-}
-
-bool TransportChannelProxy::SetSslRole(rtc::SSLRole role) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->SetSslRole(role);
-}
-
-bool TransportChannelProxy::SetSrtpCiphers(const std::vector<std::string>&
-                                           ciphers) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  pending_srtp_ciphers_ = ciphers;  // Cache so we can send later, but always
-                                    // set so it stays consistent.
-  if (impl_) {
-    return impl_->SetSrtpCiphers(ciphers);
-  }
-  return true;
-}
-
-bool TransportChannelProxy::GetSrtpCipher(std::string* cipher) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->GetSrtpCipher(cipher);
-}
-
-bool TransportChannelProxy::GetSslCipher(std::string* cipher) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->GetSslCipher(cipher);
-}
-
-bool TransportChannelProxy::GetLocalIdentity(
-    rtc::SSLIdentity** identity) const {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->GetLocalIdentity(identity);
-}
-
-bool TransportChannelProxy::GetRemoteCertificate(
-    rtc::SSLCertificate** cert) const {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->GetRemoteCertificate(cert);
-}
-
-bool TransportChannelProxy::ExportKeyingMaterial(const std::string& label,
-                                                 const uint8* context,
-                                                 size_t context_len,
-                                                 bool use_context,
-                                                 uint8* result,
-                                                 size_t result_len) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return false;
-  }
-  return impl_->ExportKeyingMaterial(label, context, context_len, use_context,
-                                     result, result_len);
-}
-
-IceRole TransportChannelProxy::GetIceRole() const {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (!impl_) {
-    return ICEROLE_UNKNOWN;
-  }
-  return impl_->GetIceRole();
-}
-
-void TransportChannelProxy::OnReadableState(TransportChannel* channel) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == impl_);
-  set_readable(impl_->readable());
-  // Note: SignalReadableState fired by set_readable.
-}
-
-void TransportChannelProxy::OnWritableState(TransportChannel* channel) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == impl_);
-  set_writable(impl_->writable());
-  // Note: SignalWritableState fired by set_readable.
-}
-
-void TransportChannelProxy::OnReadPacket(
-    TransportChannel* channel, const char* data, size_t size,
-    const rtc::PacketTime& packet_time, int flags) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == impl_);
-  SignalReadPacket(this, data, size, packet_time, flags);
-}
-
-void TransportChannelProxy::OnReadyToSend(TransportChannel* channel) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == impl_);
-  SignalReadyToSend(this);
-}
-
-void TransportChannelProxy::OnRouteChange(TransportChannel* channel,
-                                          const Candidate& candidate) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == impl_);
-  SignalRouteChange(this, candidate);
-}
-
-void TransportChannelProxy::OnMessage(rtc::Message* msg) {
-  ASSERT(rtc::Thread::Current() == worker_thread_);
-  if (msg->message_id == MSG_UPDATESTATE) {
-     // If impl_ is already readable or writable, push up those signals.
-     set_readable(impl_ ? impl_->readable() : false);
-     set_writable(impl_ ? impl_->writable() : false);
-  }
-}
-
-}  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportchannelproxy.h b/media/webrtc/trunk/webrtc/p2p/base/transportchannelproxy.h
deleted file mode 100644
index 23cd20b90e..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/base/transportchannelproxy.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_
-#define WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_
-
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "webrtc/p2p/base/transportchannel.h"
-#include "webrtc/base/messagehandler.h"
-
-namespace rtc {
-class Thread;
-}
-
-namespace cricket {
-
-class TransportChannelImpl;
-
-// Proxies calls between the client and the transport channel implementation.
-// This is needed because clients are allowed to create channels before the
-// network negotiation is complete.  Hence, we create a proxy up front, and
-// when negotiation completes, connect the proxy to the implementaiton.
-class TransportChannelProxy : public TransportChannel,
-                              public rtc::MessageHandler {
- public:
-  TransportChannelProxy(const std::string& content_name,
-                        int component);
-  virtual ~TransportChannelProxy();
-
-  TransportChannelImpl* impl() { return impl_; }
-
-  virtual TransportChannelState GetState() const;
-
-  // Sets the implementation to which we will proxy.
-  void SetImplementation(TransportChannelImpl* impl);
-
-  // Implementation of the TransportChannel interface.  These simply forward to
-  // the implementation.
-  virtual int SendPacket(const char* data, size_t len,
-                         const rtc::PacketOptions& options,
-                         int flags);
-  virtual int SetOption(rtc::Socket::Option opt, int value);
-  virtual bool GetOption(rtc::Socket::Option opt, int* value);
-  virtual int GetError();
-  virtual IceRole GetIceRole() const;
-  virtual bool GetStats(ConnectionInfos* infos);
-  virtual bool IsDtlsActive() const;
-  virtual bool GetSslRole(rtc::SSLRole* role) const;
-  virtual bool SetSslRole(rtc::SSLRole role);
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers);
-  virtual bool GetSrtpCipher(std::string* cipher);
-  virtual bool GetSslCipher(std::string* cipher);
-  virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const;
-  virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const;
-  virtual bool ExportKeyingMaterial(const std::string& label,
-                            const uint8* context,
-                            size_t context_len,
-                            bool use_context,
-                            uint8* result,
-                            size_t result_len);
-
- private:
-  // Catch signals from the implementation channel.  These just forward to the
-  // client (after updating our state to match).
-  void OnReadableState(TransportChannel* channel);
-  void OnWritableState(TransportChannel* channel);
-  void OnReadPacket(TransportChannel* channel, const char* data, size_t size,
-                    const rtc::PacketTime& packet_time, int flags);
-  void OnReadyToSend(TransportChannel* channel);
-  void OnRouteChange(TransportChannel* channel, const Candidate& candidate);
-
-  void OnMessage(rtc::Message* message);
-
-  typedef std::pair<rtc::Socket::Option, int> OptionPair;
-  typedef std::vector<OptionPair> OptionList;
-  rtc::Thread* worker_thread_;
-  TransportChannelImpl* impl_;
-  OptionList options_;
-  std::vector<std::string> pending_srtp_ciphers_;
-
-  DISALLOW_EVIL_CONSTRUCTORS(TransportChannelProxy);
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportcontroller.cc b/media/webrtc/trunk/webrtc/p2p/base/transportcontroller.cc
new file mode 100644
index 0000000000..053388eeb8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportcontroller.cc
@@ -0,0 +1,609 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/p2p/base/transportcontroller.h"
+
+#include <algorithm>
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/dtlstransport.h"
+#include "webrtc/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/port.h"
+
+namespace cricket {
+
+enum {
+  MSG_ICECONNECTIONSTATE,
+  MSG_RECEIVING,
+  MSG_ICEGATHERINGSTATE,
+  MSG_CANDIDATESGATHERED,
+};
+
+struct CandidatesData : public rtc::MessageData {
+  CandidatesData(const std::string& transport_name,
+                 const Candidates& candidates)
+      : transport_name(transport_name), candidates(candidates) {}
+
+  std::string transport_name;
+  Candidates candidates;
+};
+
+TransportController::TransportController(rtc::Thread* signaling_thread,
+                                         rtc::Thread* worker_thread,
+                                         PortAllocator* port_allocator)
+    : signaling_thread_(signaling_thread),
+      worker_thread_(worker_thread),
+      port_allocator_(port_allocator) {}
+
+TransportController::~TransportController() {
+  worker_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::DestroyAllTransports_w, this));
+  signaling_thread_->Clear(this);
+}
+
+bool TransportController::SetSslMaxProtocolVersion(
+    rtc::SSLProtocolVersion version) {
+  return worker_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::SetSslMaxProtocolVersion_w, this, version));
+}
+
+void TransportController::SetIceConfig(const IceConfig& config) {
+  worker_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::SetIceConfig_w, this, config));
+}
+
+void TransportController::SetIceRole(IceRole ice_role) {
+  worker_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::SetIceRole_w, this, ice_role));
+}
+
+bool TransportController::GetSslRole(const std::string& transport_name,
+                                     rtc::SSLRole* role) {
+  return worker_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::GetSslRole_w, this, transport_name, role));
+}
+
+bool TransportController::SetLocalCertificate(
+    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+  return worker_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::SetLocalCertificate_w, this, certificate));
+}
+
+bool TransportController::GetLocalCertificate(
+    const std::string& transport_name,
+    rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::GetLocalCertificate_w, this,
+                transport_name, certificate));
+}
+
+bool TransportController::GetRemoteSSLCertificate(
+    const std::string& transport_name,
+    rtc::SSLCertificate** cert) {
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::GetRemoteSSLCertificate_w, this,
+                transport_name, cert));
+}
+
+bool TransportController::SetLocalTransportDescription(
+    const std::string& transport_name,
+    const TransportDescription& tdesc,
+    ContentAction action,
+    std::string* err) {
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::SetLocalTransportDescription_w, this,
+                transport_name, tdesc, action, err));
+}
+
+bool TransportController::SetRemoteTransportDescription(
+    const std::string& transport_name,
+    const TransportDescription& tdesc,
+    ContentAction action,
+    std::string* err) {
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::SetRemoteTransportDescription_w, this,
+                transport_name, tdesc, action, err));
+}
+
+void TransportController::MaybeStartGathering() {
+  worker_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::MaybeStartGathering_w, this));
+}
+
+bool TransportController::AddRemoteCandidates(const std::string& transport_name,
+                                              const Candidates& candidates,
+                                              std::string* err) {
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::AddRemoteCandidates_w, this,
+                transport_name, candidates, err));
+}
+
+bool TransportController::ReadyForRemoteCandidates(
+    const std::string& transport_name) {
+  return worker_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::ReadyForRemoteCandidates_w, this, transport_name));
+}
+
+bool TransportController::GetStats(const std::string& transport_name,
+                                   TransportStats* stats) {
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::GetStats_w, this, transport_name, stats));
+}
+
+TransportChannel* TransportController::CreateTransportChannel_w(
+    const std::string& transport_name,
+    int component) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  auto it = FindChannel_w(transport_name, component);
+  if (it != channels_.end()) {
+    // Channel already exists; increment reference count and return.
+    it->AddRef();
+    return it->get();
+  }
+
+  // Need to create a new channel.
+  Transport* transport = GetOrCreateTransport_w(transport_name);
+  TransportChannelImpl* channel = transport->CreateChannel(component);
+  channel->SignalWritableState.connect(
+      this, &TransportController::OnChannelWritableState_w);
+  channel->SignalReceivingState.connect(
+      this, &TransportController::OnChannelReceivingState_w);
+  channel->SignalGatheringState.connect(
+      this, &TransportController::OnChannelGatheringState_w);
+  channel->SignalCandidateGathered.connect(
+      this, &TransportController::OnChannelCandidateGathered_w);
+  channel->SignalRoleConflict.connect(
+      this, &TransportController::OnChannelRoleConflict_w);
+  channel->SignalConnectionRemoved.connect(
+      this, &TransportController::OnChannelConnectionRemoved_w);
+  channels_.insert(channels_.end(), RefCountedChannel(channel))->AddRef();
+  // Adding a channel could cause aggregate state to change.
+  UpdateAggregateStates_w();
+  return channel;
+}
+
+void TransportController::DestroyTransportChannel_w(
+    const std::string& transport_name,
+    int component) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  auto it = FindChannel_w(transport_name, component);
+  if (it == channels_.end()) {
+    LOG(LS_WARNING) << "Attempting to delete " << transport_name
+                    << " TransportChannel " << component
+                    << ", which doesn't exist.";
+    return;
+  }
+
+  it->DecRef();
+  if (it->ref() > 0) {
+    return;
+  }
+
+  channels_.erase(it);
+  Transport* transport = GetTransport_w(transport_name);
+  transport->DestroyChannel(component);
+  // Just as we create a Transport when its first channel is created,
+  // we delete it when its last channel is deleted.
+  if (!transport->HasChannels()) {
+    DestroyTransport_w(transport_name);
+  }
+  // Removing a channel could cause aggregate state to change.
+  UpdateAggregateStates_w();
+}
+
+const rtc::scoped_refptr<rtc::RTCCertificate>&
+TransportController::certificate_for_testing() {
+  return certificate_;
+}
+
+Transport* TransportController::CreateTransport_w(
+    const std::string& transport_name) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  Transport* transport = new DtlsTransport<P2PTransport>(
+      transport_name, port_allocator(), certificate_);
+  return transport;
+}
+
+Transport* TransportController::GetTransport_w(
+    const std::string& transport_name) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  auto iter = transports_.find(transport_name);
+  return (iter != transports_.end()) ? iter->second : nullptr;
+}
+
+void TransportController::OnMessage(rtc::Message* pmsg) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+
+  switch (pmsg->message_id) {
+    case MSG_ICECONNECTIONSTATE: {
+      rtc::TypedMessageData<IceConnectionState>* data =
+          static_cast<rtc::TypedMessageData<IceConnectionState>*>(pmsg->pdata);
+      SignalConnectionState(data->data());
+      delete data;
+      break;
+    }
+    case MSG_RECEIVING: {
+      rtc::TypedMessageData<bool>* data =
+          static_cast<rtc::TypedMessageData<bool>*>(pmsg->pdata);
+      SignalReceiving(data->data());
+      delete data;
+      break;
+    }
+    case MSG_ICEGATHERINGSTATE: {
+      rtc::TypedMessageData<IceGatheringState>* data =
+          static_cast<rtc::TypedMessageData<IceGatheringState>*>(pmsg->pdata);
+      SignalGatheringState(data->data());
+      delete data;
+      break;
+    }
+    case MSG_CANDIDATESGATHERED: {
+      CandidatesData* data = static_cast<CandidatesData*>(pmsg->pdata);
+      SignalCandidatesGathered(data->transport_name, data->candidates);
+      delete data;
+      break;
+    }
+    default:
+      ASSERT(false);
+  }
+}
+
+std::vector<TransportController::RefCountedChannel>::iterator
+TransportController::FindChannel_w(const std::string& transport_name,
+                                   int component) {
+  return std::find_if(
+      channels_.begin(), channels_.end(),
+      [transport_name, component](const RefCountedChannel& channel) {
+        return channel->transport_name() == transport_name &&
+               channel->component() == component;
+      });
+}
+
+Transport* TransportController::GetOrCreateTransport_w(
+    const std::string& transport_name) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  Transport* transport = GetTransport_w(transport_name);
+  if (transport) {
+    return transport;
+  }
+
+  transport = CreateTransport_w(transport_name);
+  // The stuff below happens outside of CreateTransport_w so that unit tests
+  // can override CreateTransport_w to return a different type of transport.
+  transport->SetSslMaxProtocolVersion(ssl_max_version_);
+  transport->SetIceConfig(ice_config_);
+  transport->SetIceRole(ice_role_);
+  transport->SetIceTiebreaker(ice_tiebreaker_);
+  if (certificate_) {
+    transport->SetLocalCertificate(certificate_);
+  }
+  transports_[transport_name] = transport;
+
+  return transport;
+}
+
+void TransportController::DestroyTransport_w(
+    const std::string& transport_name) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  auto iter = transports_.find(transport_name);
+  if (iter != transports_.end()) {
+    delete iter->second;
+    transports_.erase(transport_name);
+  }
+}
+
+void TransportController::DestroyAllTransports_w() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  for (const auto& kv : transports_) {
+    delete kv.second;
+  }
+  transports_.clear();
+}
+
+bool TransportController::SetSslMaxProtocolVersion_w(
+    rtc::SSLProtocolVersion version) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  // Max SSL version can only be set before transports are created.
+  if (!transports_.empty()) {
+    return false;
+  }
+
+  ssl_max_version_ = version;
+  return true;
+}
+
+void TransportController::SetIceConfig_w(const IceConfig& config) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  ice_config_ = config;
+  for (const auto& kv : transports_) {
+    kv.second->SetIceConfig(ice_config_);
+  }
+}
+
+void TransportController::SetIceRole_w(IceRole ice_role) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  ice_role_ = ice_role;
+  for (const auto& kv : transports_) {
+    kv.second->SetIceRole(ice_role_);
+  }
+}
+
+bool TransportController::GetSslRole_w(const std::string& transport_name,
+                                       rtc::SSLRole* role) {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+
+  Transport* t = GetTransport_w(transport_name);
+  if (!t) {
+    return false;
+  }
+
+  return t->GetSslRole(role);
+}
+
+bool TransportController::SetLocalCertificate_w(
+    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  if (certificate_) {
+    return false;
+  }
+  if (!certificate) {
+    return false;
+  }
+  certificate_ = certificate;
+
+  for (const auto& kv : transports_) {
+    kv.second->SetLocalCertificate(certificate_);
+  }
+  return true;
+}
+
+bool TransportController::GetLocalCertificate_w(
+    const std::string& transport_name,
+    rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  Transport* t = GetTransport_w(transport_name);
+  if (!t) {
+    return false;
+  }
+
+  return t->GetLocalCertificate(certificate);
+}
+
+bool TransportController::GetRemoteSSLCertificate_w(
+    const std::string& transport_name,
+    rtc::SSLCertificate** cert) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  Transport* t = GetTransport_w(transport_name);
+  if (!t) {
+    return false;
+  }
+
+  return t->GetRemoteSSLCertificate(cert);
+}
+
+bool TransportController::SetLocalTransportDescription_w(
+    const std::string& transport_name,
+    const TransportDescription& tdesc,
+    ContentAction action,
+    std::string* err) {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+
+  Transport* transport = GetTransport_w(transport_name);
+  if (!transport) {
+    // If we didn't find a transport, that's not an error;
+    // it could have been deleted as a result of bundling.
+    // TODO(deadbeef): Make callers smarter so they won't attempt to set a
+    // description on a deleted transport.
+    return true;
+  }
+
+  return transport->SetLocalTransportDescription(tdesc, action, err);
+}
+
+bool TransportController::SetRemoteTransportDescription_w(
+    const std::string& transport_name,
+    const TransportDescription& tdesc,
+    ContentAction action,
+    std::string* err) {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+
+  Transport* transport = GetTransport_w(transport_name);
+  if (!transport) {
+    // If we didn't find a transport, that's not an error;
+    // it could have been deleted as a result of bundling.
+    // TODO(deadbeef): Make callers smarter so they won't attempt to set a
+    // description on a deleted transport.
+    return true;
+  }
+
+  return transport->SetRemoteTransportDescription(tdesc, action, err);
+}
+
+void TransportController::MaybeStartGathering_w() {
+  for (const auto& kv : transports_) {
+    kv.second->MaybeStartGathering();
+  }
+}
+
+bool TransportController::AddRemoteCandidates_w(
+    const std::string& transport_name,
+    const Candidates& candidates,
+    std::string* err) {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+
+  Transport* transport = GetTransport_w(transport_name);
+  if (!transport) {
+    // If we didn't find a transport, that's not an error;
+    // it could have been deleted as a result of bundling.
+    return true;
+  }
+
+  return transport->AddRemoteCandidates(candidates, err);
+}
+
+bool TransportController::ReadyForRemoteCandidates_w(
+    const std::string& transport_name) {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+
+  Transport* transport = GetTransport_w(transport_name);
+  if (!transport) {
+    return false;
+  }
+  return transport->ready_for_remote_candidates();
+}
+
+bool TransportController::GetStats_w(const std::string& transport_name,
+                                     TransportStats* stats) {
+  RTC_DCHECK(worker_thread()->IsCurrent());
+
+  Transport* transport = GetTransport_w(transport_name);
+  if (!transport) {
+    return false;
+  }
+  return transport->GetStats(stats);
+}
+
+void TransportController::OnChannelWritableState_w(TransportChannel* channel) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  LOG(LS_INFO) << channel->transport_name() << " TransportChannel "
+               << channel->component() << " writability changed to "
+               << channel->writable() << ".";
+  UpdateAggregateStates_w();
+}
+
+void TransportController::OnChannelReceivingState_w(TransportChannel* channel) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  UpdateAggregateStates_w();
+}
+
+void TransportController::OnChannelGatheringState_w(
+    TransportChannelImpl* channel) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  UpdateAggregateStates_w();
+}
+
+void TransportController::OnChannelCandidateGathered_w(
+    TransportChannelImpl* channel,
+    const Candidate& candidate) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  // We should never signal peer-reflexive candidates.
+  if (candidate.type() == PRFLX_PORT_TYPE) {
+    RTC_DCHECK(false);
+    return;
+  }
+  std::vector<Candidate> candidates;
+  candidates.push_back(candidate);
+  CandidatesData* data =
+      new CandidatesData(channel->transport_name(), candidates);
+  signaling_thread_->Post(this, MSG_CANDIDATESGATHERED, data);
+}
+
+void TransportController::OnChannelRoleConflict_w(
+    TransportChannelImpl* channel) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  if (ice_role_switch_) {
+    LOG(LS_WARNING)
+        << "Repeat of role conflict signal from TransportChannelImpl.";
+    return;
+  }
+
+  ice_role_switch_ = true;
+  IceRole reversed_role = (ice_role_ == ICEROLE_CONTROLLING)
+                              ? ICEROLE_CONTROLLED
+                              : ICEROLE_CONTROLLING;
+  for (const auto& kv : transports_) {
+    kv.second->SetIceRole(reversed_role);
+  }
+}
+
+void TransportController::OnChannelConnectionRemoved_w(
+    TransportChannelImpl* channel) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  LOG(LS_INFO) << channel->transport_name() << " TransportChannel "
+               << channel->component()
+               << " connection removed. Check if state is complete.";
+  UpdateAggregateStates_w();
+}
+
+void TransportController::UpdateAggregateStates_w() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  IceConnectionState new_connection_state = kIceConnectionConnecting;
+  IceGatheringState new_gathering_state = kIceGatheringNew;
+  bool any_receiving = false;
+  bool any_failed = false;
+  bool all_connected = !channels_.empty();
+  bool all_completed = !channels_.empty();
+  bool any_gathering = false;
+  bool all_done_gathering = !channels_.empty();
+  for (const auto& channel : channels_) {
+    any_receiving = any_receiving || channel->receiving();
+    any_failed = any_failed ||
+                 channel->GetState() == TransportChannelState::STATE_FAILED;
+    all_connected = all_connected && channel->writable();
+    all_completed =
+        all_completed && channel->writable() &&
+        channel->GetState() == TransportChannelState::STATE_COMPLETED &&
+        channel->GetIceRole() == ICEROLE_CONTROLLING &&
+        channel->gathering_state() == kIceGatheringComplete;
+    any_gathering =
+        any_gathering || channel->gathering_state() != kIceGatheringNew;
+    all_done_gathering = all_done_gathering &&
+                         channel->gathering_state() == kIceGatheringComplete;
+  }
+
+  if (any_failed) {
+    new_connection_state = kIceConnectionFailed;
+  } else if (all_completed) {
+    new_connection_state = kIceConnectionCompleted;
+  } else if (all_connected) {
+    new_connection_state = kIceConnectionConnected;
+  }
+  if (connection_state_ != new_connection_state) {
+    connection_state_ = new_connection_state;
+    signaling_thread_->Post(
+        this, MSG_ICECONNECTIONSTATE,
+        new rtc::TypedMessageData<IceConnectionState>(new_connection_state));
+  }
+
+  if (receiving_ != any_receiving) {
+    receiving_ = any_receiving;
+    signaling_thread_->Post(this, MSG_RECEIVING,
+                            new rtc::TypedMessageData<bool>(any_receiving));
+  }
+
+  if (all_done_gathering) {
+    new_gathering_state = kIceGatheringComplete;
+  } else if (any_gathering) {
+    new_gathering_state = kIceGatheringGathering;
+  }
+  if (gathering_state_ != new_gathering_state) {
+    gathering_state_ = new_gathering_state;
+    signaling_thread_->Post(
+        this, MSG_ICEGATHERINGSTATE,
+        new rtc::TypedMessageData<IceGatheringState>(new_gathering_state));
+  }
+}
+
+}  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportcontroller.h b/media/webrtc/trunk/webrtc/p2p/base/transportcontroller.h
new file mode 100644
index 0000000000..450e6b391f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportcontroller.h
@@ -0,0 +1,219 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_P2P_BASE_TRANSPORTCONTROLLER_H_
+#define WEBRTC_P2P_BASE_TRANSPORTCONTROLLER_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/sslstreamadapter.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/transport.h"
+
+namespace rtc {
+class Thread;
+}
+
+namespace cricket {
+
+class TransportController : public sigslot::has_slots<>,
+                            public rtc::MessageHandler {
+ public:
+  TransportController(rtc::Thread* signaling_thread,
+                      rtc::Thread* worker_thread,
+                      PortAllocator* port_allocator);
+
+  virtual ~TransportController();
+
+  rtc::Thread* signaling_thread() const { return signaling_thread_; }
+  rtc::Thread* worker_thread() const { return worker_thread_; }
+
+  PortAllocator* port_allocator() const { return port_allocator_; }
+
+  // Can only be set before transports are created.
+  // TODO(deadbeef): Make this an argument to the constructor once BaseSession
+  // and WebRtcSession are combined
+  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version);
+
+  void SetIceConfig(const IceConfig& config);
+  void SetIceRole(IceRole ice_role);
+
+  bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
+
+  // Specifies the identity to use in this session.
+  // Can only be called once.
+  bool SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
+  bool GetLocalCertificate(
+      const std::string& transport_name,
+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
+  // Caller owns returned certificate
+  bool GetRemoteSSLCertificate(const std::string& transport_name,
+                               rtc::SSLCertificate** cert);
+  bool SetLocalTransportDescription(const std::string& transport_name,
+                                    const TransportDescription& tdesc,
+                                    ContentAction action,
+                                    std::string* err);
+  bool SetRemoteTransportDescription(const std::string& transport_name,
+                                     const TransportDescription& tdesc,
+                                     ContentAction action,
+                                     std::string* err);
+  // Start gathering candidates for any new transports, or transports doing an
+  // ICE restart.
+  void MaybeStartGathering();
+  bool AddRemoteCandidates(const std::string& transport_name,
+                           const Candidates& candidates,
+                           std::string* err);
+  bool ReadyForRemoteCandidates(const std::string& transport_name);
+  bool GetStats(const std::string& transport_name, TransportStats* stats);
+
+  // Creates a channel if it doesn't exist. Otherwise, increments a reference
+  // count and returns an existing channel.
+  virtual TransportChannel* CreateTransportChannel_w(
+      const std::string& transport_name,
+      int component);
+
+  // Decrements a channel's reference count, and destroys the channel if
+  // nothing is referencing it.
+  virtual void DestroyTransportChannel_w(const std::string& transport_name,
+                                         int component);
+
+  // All of these signals are fired on the signalling thread.
+
+  // If any transport failed => failed,
+  // Else if all completed => completed,
+  // Else if all connected => connected,
+  // Else => connecting
+  sigslot::signal1<IceConnectionState> SignalConnectionState;
+
+  // Receiving if any transport is receiving
+  sigslot::signal1<bool> SignalReceiving;
+
+  // If all transports done gathering => complete,
+  // Else if any are gathering => gathering,
+  // Else => new
+  sigslot::signal1<IceGatheringState> SignalGatheringState;
+
+  // (transport_name, candidates)
+  sigslot::signal2<const std::string&, const Candidates&>
+      SignalCandidatesGathered;
+
+  // for unit test
+  const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
+
+ protected:
+  // Protected and virtual so we can override it in unit tests.
+  virtual Transport* CreateTransport_w(const std::string& transport_name);
+
+  // For unit tests
+  const std::map<std::string, Transport*>& transports() { return transports_; }
+  Transport* GetTransport_w(const std::string& transport_name);
+
+ private:
+  void OnMessage(rtc::Message* pmsg) override;
+
+  // It's the Transport that's currently responsible for creating/destroying
+  // channels, but the TransportController keeps track of how many external
+  // objects (BaseChannels) reference each channel.
+  struct RefCountedChannel {
+    RefCountedChannel() : impl_(nullptr), ref_(0) {}
+    explicit RefCountedChannel(TransportChannelImpl* impl)
+        : impl_(impl), ref_(0) {}
+
+    void AddRef() { ++ref_; }
+    void DecRef() {
+      ASSERT(ref_ > 0);
+      --ref_;
+    }
+    int ref() const { return ref_; }
+
+    TransportChannelImpl* get() const { return impl_; }
+    TransportChannelImpl* operator->() const { return impl_; }
+
+   private:
+    TransportChannelImpl* impl_;
+    int ref_;
+  };
+
+  std::vector<RefCountedChannel>::iterator FindChannel_w(
+      const std::string& transport_name,
+      int component);
+
+  Transport* GetOrCreateTransport_w(const std::string& transport_name);
+  void DestroyTransport_w(const std::string& transport_name);
+  void DestroyAllTransports_w();
+
+  bool SetSslMaxProtocolVersion_w(rtc::SSLProtocolVersion version);
+  void SetIceConfig_w(const IceConfig& config);
+  void SetIceRole_w(IceRole ice_role);
+  bool GetSslRole_w(const std::string& transport_name, rtc::SSLRole* role);
+  bool SetLocalCertificate_w(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
+  bool GetLocalCertificate_w(
+      const std::string& transport_name,
+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
+  bool GetRemoteSSLCertificate_w(const std::string& transport_name,
+                                 rtc::SSLCertificate** cert);
+  bool SetLocalTransportDescription_w(const std::string& transport_name,
+                                      const TransportDescription& tdesc,
+                                      ContentAction action,
+                                      std::string* err);
+  bool SetRemoteTransportDescription_w(const std::string& transport_name,
+                                       const TransportDescription& tdesc,
+                                       ContentAction action,
+                                       std::string* err);
+  void MaybeStartGathering_w();
+  bool AddRemoteCandidates_w(const std::string& transport_name,
+                             const Candidates& candidates,
+                             std::string* err);
+  bool ReadyForRemoteCandidates_w(const std::string& transport_name);
+  bool GetStats_w(const std::string& transport_name, TransportStats* stats);
+
+  // Handlers for signals from Transport.
+  void OnChannelWritableState_w(TransportChannel* channel);
+  void OnChannelReceivingState_w(TransportChannel* channel);
+  void OnChannelGatheringState_w(TransportChannelImpl* channel);
+  void OnChannelCandidateGathered_w(TransportChannelImpl* channel,
+                                    const Candidate& candidate);
+  void OnChannelRoleConflict_w(TransportChannelImpl* channel);
+  void OnChannelConnectionRemoved_w(TransportChannelImpl* channel);
+
+  void UpdateAggregateStates_w();
+
+  rtc::Thread* const signaling_thread_ = nullptr;
+  rtc::Thread* const worker_thread_ = nullptr;
+  typedef std::map<std::string, Transport*> TransportMap;
+  TransportMap transports_;
+
+  std::vector<RefCountedChannel> channels_;
+
+  PortAllocator* const port_allocator_ = nullptr;
+  rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
+
+  // Aggregate state for TransportChannelImpls.
+  IceConnectionState connection_state_ = kIceConnectionConnecting;
+  bool receiving_ = false;
+  IceGatheringState gathering_state_ = kIceGatheringNew;
+
+  // TODO(deadbeef): Move the fields below down to the transports themselves
+  IceConfig ice_config_;
+  IceRole ice_role_ = ICEROLE_CONTROLLING;
+  // Flag which will be set to true after the first role switch
+  bool ice_role_switch_ = false;
+  uint64_t ice_tiebreaker_ = rtc::CreateRandomId64();
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
+};
+
+}  // namespace cricket
+
+#endif  // WEBRTC_P2P_BASE_TRANSPORTCONTROLLER_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportcontroller_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/transportcontroller_unittest.cc
new file mode 100644
index 0000000000..6ff158e8fc
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportcontroller_unittest.cc
@@ -0,0 +1,686 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <map>
+
+#include "webrtc/base/fakesslidentity.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/helpers.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/sslidentity.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/dtlstransportchannel.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
+
+static const int kTimeout = 100;
+static const char kIceUfrag1[] = "TESTICEUFRAG0001";
+static const char kIcePwd1[] = "TESTICEPWD00000000000001";
+static const char kIceUfrag2[] = "TESTICEUFRAG0002";
+static const char kIcePwd2[] = "TESTICEPWD00000000000002";
+
+using cricket::Candidate;
+using cricket::Candidates;
+using cricket::FakeTransportChannel;
+using cricket::FakeTransportController;
+using cricket::IceConnectionState;
+using cricket::IceGatheringState;
+using cricket::TransportChannel;
+using cricket::TransportController;
+using cricket::TransportDescription;
+using cricket::TransportStats;
+
+// Only subclassing from FakeTransportController because currently that's the
+// only way to have a TransportController with fake TransportChannels.
+//
+// TODO(deadbeef): Change this once the Transport/TransportChannel class
+// heirarchy is cleaned up, and we can pass a "TransportChannelFactory" or
+// something similar into TransportController.
+typedef FakeTransportController TransportControllerForTest;
+
+class TransportControllerTest : public testing::Test,
+                                public sigslot::has_slots<> {
+ public:
+  TransportControllerTest()
+      : transport_controller_(new TransportControllerForTest()),
+        signaling_thread_(rtc::Thread::Current()) {
+    ConnectTransportControllerSignals();
+  }
+
+  void CreateTransportControllerWithWorkerThread() {
+    if (!worker_thread_) {
+      worker_thread_.reset(new rtc::Thread());
+      worker_thread_->Start();
+    }
+    transport_controller_.reset(
+        new TransportControllerForTest(worker_thread_.get()));
+    ConnectTransportControllerSignals();
+  }
+
+  void ConnectTransportControllerSignals() {
+    transport_controller_->SignalConnectionState.connect(
+        this, &TransportControllerTest::OnConnectionState);
+    transport_controller_->SignalReceiving.connect(
+        this, &TransportControllerTest::OnReceiving);
+    transport_controller_->SignalGatheringState.connect(
+        this, &TransportControllerTest::OnGatheringState);
+    transport_controller_->SignalCandidatesGathered.connect(
+        this, &TransportControllerTest::OnCandidatesGathered);
+  }
+
+  FakeTransportChannel* CreateChannel(const std::string& content,
+                                      int component) {
+    TransportChannel* channel =
+        transport_controller_->CreateTransportChannel_w(content, component);
+    return static_cast<FakeTransportChannel*>(channel);
+  }
+
+  void DestroyChannel(const std::string& content, int component) {
+    transport_controller_->DestroyTransportChannel_w(content, component);
+  }
+
+  Candidate CreateCandidate(int component) {
+    Candidate c;
+    c.set_address(rtc::SocketAddress("192.168.1.1", 8000));
+    c.set_component(1);
+    c.set_protocol(cricket::UDP_PROTOCOL_NAME);
+    c.set_priority(1);
+    return c;
+  }
+
+  // Used for thread hopping test.
+  void CreateChannelsAndCompleteConnectionOnWorkerThread() {
+    worker_thread_->Invoke<void>(rtc::Bind(
+        &TransportControllerTest::CreateChannelsAndCompleteConnection_w, this));
+  }
+
+  void CreateChannelsAndCompleteConnection_w() {
+    transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+    FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+    ASSERT_NE(nullptr, channel1);
+    FakeTransportChannel* channel2 = CreateChannel("video", 1);
+    ASSERT_NE(nullptr, channel2);
+
+    TransportDescription local_desc(
+        std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+        cricket::CONNECTIONROLE_ACTPASS, nullptr, Candidates());
+    std::string err;
+    transport_controller_->SetLocalTransportDescription(
+        "audio", local_desc, cricket::CA_OFFER, &err);
+    transport_controller_->SetLocalTransportDescription(
+        "video", local_desc, cricket::CA_OFFER, &err);
+    transport_controller_->MaybeStartGathering();
+    channel1->SignalCandidateGathered(channel1, CreateCandidate(1));
+    channel2->SignalCandidateGathered(channel2, CreateCandidate(1));
+    channel1->SetCandidatesGatheringComplete();
+    channel2->SetCandidatesGatheringComplete();
+    channel1->SetConnectionCount(2);
+    channel2->SetConnectionCount(2);
+    channel1->SetReceiving(true);
+    channel2->SetReceiving(true);
+    channel1->SetWritable(true);
+    channel2->SetWritable(true);
+    channel1->SetConnectionCount(1);
+    channel2->SetConnectionCount(1);
+  }
+
+  cricket::IceConfig CreateIceConfig(int receiving_timeout_ms,
+                                     bool gather_continually) {
+    cricket::IceConfig config;
+    config.receiving_timeout_ms = receiving_timeout_ms;
+    config.gather_continually = gather_continually;
+    return config;
+  }
+
+ protected:
+  void OnConnectionState(IceConnectionState state) {
+    if (!signaling_thread_->IsCurrent()) {
+      signaled_on_non_signaling_thread_ = true;
+    }
+    connection_state_ = state;
+    ++connection_state_signal_count_;
+  }
+
+  void OnReceiving(bool receiving) {
+    if (!signaling_thread_->IsCurrent()) {
+      signaled_on_non_signaling_thread_ = true;
+    }
+    receiving_ = receiving;
+    ++receiving_signal_count_;
+  }
+
+  void OnGatheringState(IceGatheringState state) {
+    if (!signaling_thread_->IsCurrent()) {
+      signaled_on_non_signaling_thread_ = true;
+    }
+    gathering_state_ = state;
+    ++gathering_state_signal_count_;
+  }
+
+  void OnCandidatesGathered(const std::string& transport_name,
+                            const Candidates& candidates) {
+    if (!signaling_thread_->IsCurrent()) {
+      signaled_on_non_signaling_thread_ = true;
+    }
+    candidates_[transport_name].insert(candidates_[transport_name].end(),
+                                       candidates.begin(), candidates.end());
+    ++candidates_signal_count_;
+  }
+
+  rtc::scoped_ptr<rtc::Thread> worker_thread_;  // Not used for most tests.
+  rtc::scoped_ptr<TransportControllerForTest> transport_controller_;
+
+  // Information received from signals from transport controller.
+  IceConnectionState connection_state_ = cricket::kIceConnectionConnecting;
+  bool receiving_ = false;
+  IceGatheringState gathering_state_ = cricket::kIceGatheringNew;
+  // transport_name => candidates
+  std::map<std::string, Candidates> candidates_;
+  // Counts of each signal emitted.
+  int connection_state_signal_count_ = 0;
+  int receiving_signal_count_ = 0;
+  int gathering_state_signal_count_ = 0;
+  int candidates_signal_count_ = 0;
+
+  // Used to make sure signals only come on signaling thread.
+  rtc::Thread* const signaling_thread_ = nullptr;
+  bool signaled_on_non_signaling_thread_ = false;
+};
+
+TEST_F(TransportControllerTest, TestSetIceConfig) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+
+  transport_controller_->SetIceConfig(CreateIceConfig(1000, true));
+  EXPECT_EQ(1000, channel1->receiving_timeout());
+  EXPECT_TRUE(channel1->gather_continually());
+
+  // Test that value stored in controller is applied to new channels.
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+  EXPECT_EQ(1000, channel2->receiving_timeout());
+  EXPECT_TRUE(channel2->gather_continually());
+}
+
+TEST_F(TransportControllerTest, TestSetSslMaxProtocolVersion) {
+  EXPECT_TRUE(transport_controller_->SetSslMaxProtocolVersion(
+      rtc::SSL_PROTOCOL_DTLS_12));
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+
+  ASSERT_NE(nullptr, channel);
+  EXPECT_EQ(rtc::SSL_PROTOCOL_DTLS_12, channel->ssl_max_protocol_version());
+
+  // Setting max version after transport is created should fail.
+  EXPECT_FALSE(transport_controller_->SetSslMaxProtocolVersion(
+      rtc::SSL_PROTOCOL_DTLS_10));
+}
+
+TEST_F(TransportControllerTest, TestSetIceRole) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel1->GetIceRole());
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED, channel1->GetIceRole());
+
+  // Test that value stored in controller is applied to new channels.
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED, channel2->GetIceRole());
+}
+
+// Test that when one channel encounters a role conflict, the ICE role is
+// swapped on every channel.
+TEST_F(TransportControllerTest, TestIceRoleConflict) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel1->GetIceRole());
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel2->GetIceRole());
+
+  channel1->SignalRoleConflict(channel1);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED, channel1->GetIceRole());
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED, channel2->GetIceRole());
+}
+
+TEST_F(TransportControllerTest, TestGetSslRole) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+  ASSERT_TRUE(channel->SetSslRole(rtc::SSL_CLIENT));
+  rtc::SSLRole role;
+  EXPECT_FALSE(transport_controller_->GetSslRole("video", &role));
+  EXPECT_TRUE(transport_controller_->GetSslRole("audio", &role));
+  EXPECT_EQ(rtc::SSL_CLIENT, role);
+}
+
+TEST_F(TransportControllerTest, TestSetAndGetLocalCertificate) {
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
+      rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+          rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
+      rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+          rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
+  rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
+
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+
+  EXPECT_TRUE(transport_controller_->SetLocalCertificate(certificate1));
+  EXPECT_TRUE(transport_controller_->GetLocalCertificate(
+      "audio", &returned_certificate));
+  EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
+            returned_certificate->identity()->certificate().ToPEMString());
+
+  // Should fail if called for a nonexistant transport.
+  EXPECT_FALSE(transport_controller_->GetLocalCertificate(
+      "video", &returned_certificate));
+
+  // Test that identity stored in controller is applied to new channels.
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+  EXPECT_TRUE(transport_controller_->GetLocalCertificate(
+      "video", &returned_certificate));
+  EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
+            returned_certificate->identity()->certificate().ToPEMString());
+
+  // Shouldn't be able to change the identity once set.
+  EXPECT_FALSE(transport_controller_->SetLocalCertificate(certificate2));
+}
+
+TEST_F(TransportControllerTest, TestGetRemoteSSLCertificate) {
+  rtc::FakeSSLCertificate fake_certificate("fake_data");
+  rtc::scoped_ptr<rtc::SSLCertificate> returned_certificate;
+
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+
+  channel->SetRemoteSSLCertificate(&fake_certificate);
+  EXPECT_TRUE(transport_controller_->GetRemoteSSLCertificate(
+      "audio", returned_certificate.accept()));
+  EXPECT_EQ(fake_certificate.ToPEMString(),
+            returned_certificate->ToPEMString());
+
+  // Should fail if called for a nonexistant transport.
+  EXPECT_FALSE(transport_controller_->GetRemoteSSLCertificate(
+      "video", returned_certificate.accept()));
+}
+
+TEST_F(TransportControllerTest, TestSetLocalTransportDescription) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+  TransportDescription local_desc(
+      std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+      cricket::CONNECTIONROLE_ACTPASS, nullptr, Candidates());
+  std::string err;
+  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
+      "audio", local_desc, cricket::CA_OFFER, &err));
+  // Check that ICE ufrag and pwd were propagated to channel.
+  EXPECT_EQ(kIceUfrag1, channel->ice_ufrag());
+  EXPECT_EQ(kIcePwd1, channel->ice_pwd());
+  // After setting local description, we should be able to start gathering
+  // candidates.
+  transport_controller_->MaybeStartGathering();
+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
+  EXPECT_EQ(1, gathering_state_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSetRemoteTransportDescription) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+  TransportDescription remote_desc(
+      std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+      cricket::CONNECTIONROLE_ACTPASS, nullptr, Candidates());
+  std::string err;
+  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
+      "audio", remote_desc, cricket::CA_OFFER, &err));
+  // Check that ICE ufrag and pwd were propagated to channel.
+  EXPECT_EQ(kIceUfrag1, channel->remote_ice_ufrag());
+  EXPECT_EQ(kIcePwd1, channel->remote_ice_pwd());
+}
+
+TEST_F(TransportControllerTest, TestAddRemoteCandidates) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+  Candidates candidates;
+  candidates.push_back(CreateCandidate(1));
+  std::string err;
+  EXPECT_TRUE(
+      transport_controller_->AddRemoteCandidates("audio", candidates, &err));
+  EXPECT_EQ(1U, channel->remote_candidates().size());
+}
+
+TEST_F(TransportControllerTest, TestReadyForRemoteCandidates) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+  // We expect to be ready for remote candidates only after local and remote
+  // descriptions are set.
+  EXPECT_FALSE(transport_controller_->ReadyForRemoteCandidates("audio"));
+
+  std::string err;
+  TransportDescription remote_desc(
+      std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+      cricket::CONNECTIONROLE_ACTPASS, nullptr, Candidates());
+  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
+      "audio", remote_desc, cricket::CA_OFFER, &err));
+  EXPECT_FALSE(transport_controller_->ReadyForRemoteCandidates("audio"));
+
+  TransportDescription local_desc(
+      std::vector<std::string>(), kIceUfrag2, kIcePwd2, cricket::ICEMODE_FULL,
+      cricket::CONNECTIONROLE_ACTPASS, nullptr, Candidates());
+  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
+      "audio", local_desc, cricket::CA_ANSWER, &err));
+  EXPECT_TRUE(transport_controller_->ReadyForRemoteCandidates("audio"));
+}
+
+TEST_F(TransportControllerTest, TestGetStats) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("audio", 2);
+  ASSERT_NE(nullptr, channel2);
+  FakeTransportChannel* channel3 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel3);
+
+  TransportStats stats;
+  EXPECT_TRUE(transport_controller_->GetStats("audio", &stats));
+  EXPECT_EQ("audio", stats.transport_name);
+  EXPECT_EQ(2U, stats.channel_stats.size());
+}
+
+// Test that transport gets destroyed when it has no more channels.
+TEST_F(TransportControllerTest, TestCreateAndDestroyChannel) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel2);
+  ASSERT_EQ(channel1, channel2);
+  FakeTransportChannel* channel3 = CreateChannel("audio", 2);
+  ASSERT_NE(nullptr, channel3);
+
+  // Using GetStats to check if transport is destroyed from an outside class's
+  // perspective.
+  TransportStats stats;
+  EXPECT_TRUE(transport_controller_->GetStats("audio", &stats));
+  DestroyChannel("audio", 2);
+  DestroyChannel("audio", 1);
+  EXPECT_TRUE(transport_controller_->GetStats("audio", &stats));
+  DestroyChannel("audio", 1);
+  EXPECT_FALSE(transport_controller_->GetStats("audio", &stats));
+}
+
+TEST_F(TransportControllerTest, TestSignalConnectionStateFailed) {
+  // Need controlling ICE role to get in failed state.
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+
+  // Should signal "failed" if any channel failed; channel is considered failed
+  // if it previously had a connection but now has none, and gathering is
+  // complete.
+  channel1->SetCandidatesGatheringComplete();
+  channel1->SetConnectionCount(1);
+  channel1->SetConnectionCount(0);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
+  EXPECT_EQ(1, connection_state_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSignalConnectionStateConnected) {
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+  FakeTransportChannel* channel3 = CreateChannel("video", 2);
+  ASSERT_NE(nullptr, channel3);
+
+  // First, have one channel connect, and another fail, to ensure that
+  // the first channel connecting didn't trigger a "connected" state signal.
+  // We should only get a signal when all are connected.
+  channel1->SetConnectionCount(2);
+  channel1->SetWritable(true);
+  channel3->SetCandidatesGatheringComplete();
+  channel3->SetConnectionCount(1);
+  channel3->SetConnectionCount(0);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
+  // Signal count of 1 means that the only signal emitted was "failed".
+  EXPECT_EQ(1, connection_state_signal_count_);
+
+  // Destroy the failed channel to return to "connecting" state.
+  DestroyChannel("video", 2);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
+                 kTimeout);
+  EXPECT_EQ(2, connection_state_signal_count_);
+
+  // Make the remaining channel reach a connected state.
+  channel2->SetConnectionCount(2);
+  channel2->SetWritable(true);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
+  EXPECT_EQ(3, connection_state_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSignalConnectionStateComplete) {
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+  FakeTransportChannel* channel3 = CreateChannel("video", 2);
+  ASSERT_NE(nullptr, channel3);
+
+  // Similar to above test, but we're now reaching the completed state, which
+  // means only one connection per FakeTransportChannel.
+  channel1->SetCandidatesGatheringComplete();
+  channel1->SetConnectionCount(1);
+  channel1->SetWritable(true);
+  channel3->SetCandidatesGatheringComplete();
+  channel3->SetConnectionCount(1);
+  channel3->SetConnectionCount(0);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
+  // Signal count of 1 means that the only signal emitted was "failed".
+  EXPECT_EQ(1, connection_state_signal_count_);
+
+  // Destroy the failed channel to return to "connecting" state.
+  DestroyChannel("video", 2);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
+                 kTimeout);
+  EXPECT_EQ(2, connection_state_signal_count_);
+
+  // Make the remaining channel reach a connected state.
+  channel2->SetCandidatesGatheringComplete();
+  channel2->SetConnectionCount(2);
+  channel2->SetWritable(true);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
+  EXPECT_EQ(3, connection_state_signal_count_);
+
+  // Finally, transition to completed state.
+  channel2->SetConnectionCount(1);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
+  EXPECT_EQ(4, connection_state_signal_count_);
+}
+
+// Make sure that if we're "connected" and remove a transport, we stay in the
+// "connected" state.
+TEST_F(TransportControllerTest, TestDestroyTransportAndStayConnected) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+
+  channel1->SetCandidatesGatheringComplete();
+  channel1->SetConnectionCount(2);
+  channel1->SetWritable(true);
+  channel2->SetCandidatesGatheringComplete();
+  channel2->SetConnectionCount(2);
+  channel2->SetWritable(true);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
+  EXPECT_EQ(1, connection_state_signal_count_);
+
+  // Destroy one channel, then "complete" the other one, so we reach
+  // a known state.
+  DestroyChannel("video", 1);
+  channel1->SetConnectionCount(1);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
+  // Signal count of 2 means the deletion didn't cause any unexpected signals
+  EXPECT_EQ(2, connection_state_signal_count_);
+}
+
+// If we destroy the last/only transport, we should simply transition to
+// "connecting".
+TEST_F(TransportControllerTest, TestDestroyLastTransportWhileConnected) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+
+  channel->SetCandidatesGatheringComplete();
+  channel->SetConnectionCount(2);
+  channel->SetWritable(true);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
+  EXPECT_EQ(1, connection_state_signal_count_);
+
+  DestroyChannel("audio", 1);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
+                 kTimeout);
+  // Signal count of 2 means the deletion didn't cause any unexpected signals
+  EXPECT_EQ(2, connection_state_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSignalReceiving) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+
+  // Should signal receiving as soon as any channel is receiving.
+  channel1->SetReceiving(true);
+  EXPECT_TRUE_WAIT(receiving_, kTimeout);
+  EXPECT_EQ(1, receiving_signal_count_);
+
+  channel2->SetReceiving(true);
+  channel1->SetReceiving(false);
+  channel2->SetReceiving(false);
+  EXPECT_TRUE_WAIT(!receiving_, kTimeout);
+  EXPECT_EQ(2, receiving_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSignalGatheringStateGathering) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+  channel->Connect();
+  channel->MaybeStartGathering();
+  // Should be in the gathering state as soon as any transport starts gathering.
+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
+  EXPECT_EQ(1, gathering_state_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSignalGatheringStateComplete) {
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+  FakeTransportChannel* channel3 = CreateChannel("data", 1);
+  ASSERT_NE(nullptr, channel3);
+
+  channel3->Connect();
+  channel3->MaybeStartGathering();
+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
+  EXPECT_EQ(1, gathering_state_signal_count_);
+
+  // Have one channel finish gathering, then destroy it, to make sure gathering
+  // completion wasn't signalled if only one transport finished gathering.
+  channel3->SetCandidatesGatheringComplete();
+  DestroyChannel("data", 1);
+  EXPECT_EQ_WAIT(cricket::kIceGatheringNew, gathering_state_, kTimeout);
+  EXPECT_EQ(2, gathering_state_signal_count_);
+
+  // Make remaining channels start and then finish gathering.
+  channel1->Connect();
+  channel1->MaybeStartGathering();
+  channel2->Connect();
+  channel2->MaybeStartGathering();
+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
+  EXPECT_EQ(3, gathering_state_signal_count_);
+
+  channel1->SetCandidatesGatheringComplete();
+  channel2->SetCandidatesGatheringComplete();
+  EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
+  EXPECT_EQ(4, gathering_state_signal_count_);
+}
+
+// Test that when the last transport that hasn't finished connecting and/or
+// gathering is destroyed, the aggregate state jumps to "completed". This can
+// happen if, for example, we have an audio and video transport, the audio
+// transport completes, then we start bundling video on the audio transport.
+TEST_F(TransportControllerTest,
+       TestSignalingWhenLastIncompleteTransportDestroyed) {
+  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
+  FakeTransportChannel* channel1 = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel1);
+  FakeTransportChannel* channel2 = CreateChannel("video", 1);
+  ASSERT_NE(nullptr, channel2);
+
+  channel1->SetCandidatesGatheringComplete();
+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
+  EXPECT_EQ(1, gathering_state_signal_count_);
+
+  channel1->SetConnectionCount(1);
+  channel1->SetWritable(true);
+  DestroyChannel("video", 1);
+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
+  EXPECT_EQ(1, connection_state_signal_count_);
+  EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
+  EXPECT_EQ(2, gathering_state_signal_count_);
+}
+
+TEST_F(TransportControllerTest, TestSignalCandidatesGathered) {
+  FakeTransportChannel* channel = CreateChannel("audio", 1);
+  ASSERT_NE(nullptr, channel);
+
+  // Transport won't signal candidates until it has a local description.
+  TransportDescription local_desc(
+      std::vector<std::string>(), kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL,
+      cricket::CONNECTIONROLE_ACTPASS, nullptr, Candidates());
+  std::string err;
+  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
+      "audio", local_desc, cricket::CA_OFFER, &err));
+  transport_controller_->MaybeStartGathering();
+
+  channel->SignalCandidateGathered(channel, CreateCandidate(1));
+  EXPECT_EQ_WAIT(1, candidates_signal_count_, kTimeout);
+  EXPECT_EQ(1U, candidates_["audio"].size());
+}
+
+TEST_F(TransportControllerTest, TestSignalingOccursOnSignalingThread) {
+  CreateTransportControllerWithWorkerThread();
+  CreateChannelsAndCompleteConnectionOnWorkerThread();
+
+  // connecting --> connected --> completed
+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
+  EXPECT_EQ(2, connection_state_signal_count_);
+
+  EXPECT_TRUE_WAIT(receiving_, kTimeout);
+  EXPECT_EQ(1, receiving_signal_count_);
+
+  // new --> gathering --> complete
+  EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
+  EXPECT_EQ(2, gathering_state_signal_count_);
+
+  EXPECT_EQ_WAIT(1U, candidates_["audio"].size(), kTimeout);
+  EXPECT_EQ_WAIT(1U, candidates_["video"].size(), kTimeout);
+  EXPECT_EQ(2, candidates_signal_count_);
+
+  EXPECT_TRUE(!signaled_on_non_signaling_thread_);
+}
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportdescription.cc b/media/webrtc/trunk/webrtc/p2p/base/transportdescription.cc
index 01c6a8f071..b8f14eaa98 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportdescription.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportdescription.cc
@@ -10,8 +10,9 @@
 
 #include "webrtc/p2p/base/transportdescription.h"
 
-#include "webrtc/p2p/base/constants.h"
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/p2p/base/constants.h"
 
 namespace cricket {
 
@@ -23,7 +24,7 @@ bool StringToConnectionRole(const std::string& role_str, ConnectionRole* role) {
       CONNECTIONROLE_HOLDCONN_STR
   };
 
-  for (size_t i = 0; i < ARRAY_SIZE(roles); ++i) {
+  for (size_t i = 0; i < arraysize(roles); ++i) {
     if (_stricmp(roles[i], role_str.c_str()) == 0) {
       *role = static_cast<ConnectionRole>(CONNECTIONROLE_ACTIVE + i);
       return true;
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportdescription.h b/media/webrtc/trunk/webrtc/p2p/base/transportdescription.h
index 5ab1cd6a12..8ea1f4bc2e 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportdescription.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportdescription.h
@@ -35,16 +35,6 @@ enum SecurePolicy {
   SEC_REQUIRED
 };
 
-// The transport protocol we've elected to use.
-enum TransportProtocol {
-  ICEPROTO_GOOGLE,  // Google version of ICE protocol.
-  ICEPROTO_HYBRID,  // ICE, but can fall back to the Google version.
-  ICEPROTO_RFC5245  // Standard RFC 5245 version of ICE.
-};
-// The old name for TransportProtocol.
-// TODO(juberti): remove this.
-typedef TransportProtocol IceProtocolType;
-
 // Whether our side of the call is driving the negotiation, or the other side.
 enum IceRole {
   ICEROLE_CONTROLLING = 0,
@@ -86,33 +76,28 @@ struct TransportDescription {
       : ice_mode(ICEMODE_FULL),
         connection_role(CONNECTIONROLE_NONE) {}
 
-  TransportDescription(const std::string& transport_type,
-                       const std::vector<std::string>& transport_options,
+  TransportDescription(const std::vector<std::string>& transport_options,
                        const std::string& ice_ufrag,
                        const std::string& ice_pwd,
                        IceMode ice_mode,
                        ConnectionRole role,
                        const rtc::SSLFingerprint* identity_fingerprint,
                        const Candidates& candidates)
-      : transport_type(transport_type),
-        transport_options(transport_options),
+      : transport_options(transport_options),
         ice_ufrag(ice_ufrag),
         ice_pwd(ice_pwd),
         ice_mode(ice_mode),
         connection_role(role),
         identity_fingerprint(CopyFingerprint(identity_fingerprint)),
         candidates(candidates) {}
-  TransportDescription(const std::string& transport_type,
-                       const std::string& ice_ufrag,
+  TransportDescription(const std::string& ice_ufrag,
                        const std::string& ice_pwd)
-      : transport_type(transport_type),
-        ice_ufrag(ice_ufrag),
+      : ice_ufrag(ice_ufrag),
         ice_pwd(ice_pwd),
         ice_mode(ICEMODE_FULL),
         connection_role(CONNECTIONROLE_NONE) {}
   TransportDescription(const TransportDescription& from)
-      : transport_type(from.transport_type),
-        transport_options(from.transport_options),
+      : transport_options(from.transport_options),
         ice_ufrag(from.ice_ufrag),
         ice_pwd(from.ice_pwd),
         ice_mode(from.ice_mode),
@@ -125,7 +110,6 @@ struct TransportDescription {
     if (this == &from)
       return *this;
 
-    transport_type = from.transport_type;
     transport_options = from.transport_options;
     ice_ufrag = from.ice_ufrag;
     ice_pwd = from.ice_pwd;
@@ -155,7 +139,6 @@ struct TransportDescription {
     return new rtc::SSLFingerprint(*from);
   }
 
-  std::string transport_type;  // xmlns of <transport>
   std::vector<std::string> transport_options;
   std::string ice_ufrag;
   std::string ice_pwd;
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.cc b/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.cc
index 1230ba52c1..1ddf55d4a1 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.cc
@@ -14,17 +14,12 @@
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslfingerprint.h"
 
 namespace cricket {
 
-static TransportProtocol kDefaultProtocol = ICEPROTO_RFC5245;
-
 TransportDescriptionFactory::TransportDescriptionFactory()
-    : protocol_(kDefaultProtocol),
-      secure_(SEC_DISABLED),
-      identity_(NULL) {
+    : secure_(SEC_DISABLED) {
 }
 
 TransportDescription* TransportDescriptionFactory::CreateOffer(
@@ -32,16 +27,6 @@ TransportDescription* TransportDescriptionFactory::CreateOffer(
     const TransportDescription* current_description) const {
   rtc::scoped_ptr<TransportDescription> desc(new TransportDescription());
 
-  // Set the transport type depending on the selected protocol.
-  if (protocol_ == ICEPROTO_RFC5245) {
-    desc->transport_type = NS_JINGLE_ICE_UDP;
-  } else if (protocol_ == ICEPROTO_HYBRID) {
-    desc->transport_type = NS_JINGLE_ICE_UDP;
-    desc->AddOption(ICE_OPTION_GICE);
-  } else if (protocol_ == ICEPROTO_GOOGLE) {
-    desc->transport_type = NS_GINGLE_P2P;
-  }
-
   // Generate the ICE credentials if we don't already have them.
   if (!current_description || options.ice_restart) {
     desc->ice_ufrag = rtc::CreateRandomString(ICE_UFRAG_LENGTH);
@@ -67,33 +52,14 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
     const TransportDescription* offer,
     const TransportOptions& options,
     const TransportDescription* current_description) const {
-  // A NULL offer is treated as a GICE transport description.
   // TODO(juberti): Figure out why we get NULL offers, and fix this upstream.
-  rtc::scoped_ptr<TransportDescription> desc(new TransportDescription());
-
-  // Figure out which ICE variant to negotiate; prefer RFC 5245 ICE, but fall
-  // back to G-ICE if needed. Note that we never create a hybrid answer, since
-  // we know what the other side can support already.
-  if (offer && offer->transport_type == NS_JINGLE_ICE_UDP &&
-      (protocol_ == ICEPROTO_RFC5245 || protocol_ == ICEPROTO_HYBRID)) {
-    // Offer is ICE or hybrid, we support ICE or hybrid: use ICE.
-    desc->transport_type = NS_JINGLE_ICE_UDP;
-  } else if (offer && offer->transport_type == NS_JINGLE_ICE_UDP &&
-             offer->HasOption(ICE_OPTION_GICE) &&
-             protocol_ == ICEPROTO_GOOGLE) {
-    desc->transport_type = NS_GINGLE_P2P;
-    // Offer is hybrid, we support GICE: use GICE.
-  } else if ((!offer || offer->transport_type == NS_GINGLE_P2P) &&
-             (protocol_ == ICEPROTO_HYBRID || protocol_ == ICEPROTO_GOOGLE)) {
-    // Offer is GICE, we support hybrid or GICE: use GICE.
-    desc->transport_type = NS_GINGLE_P2P;
-  } else {
-    // Mismatch.
-    LOG(LS_WARNING) << "Failed to create TransportDescription answer "
-                       "because of incompatible transport types";
+  if (!offer) {
+    LOG(LS_WARNING) << "Failed to create TransportDescription answer " <<
+        "because offer is NULL";
     return NULL;
   }
 
+  rtc::scoped_ptr<TransportDescription> desc(new TransportDescription());
   // Generate the ICE credentials if we don't already have them or ice is
   // being restarted.
   if (!current_description || options.ice_restart) {
@@ -129,8 +95,8 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
 
 bool TransportDescriptionFactory::SetSecurityInfo(
     TransportDescription* desc, ConnectionRole role) const {
-  if (!identity_) {
-    LOG(LS_ERROR) << "Cannot create identity digest with no identity";
+  if (!certificate_) {
+    LOG(LS_ERROR) << "Cannot create identity digest with no certificate";
     return false;
   }
 
@@ -138,13 +104,14 @@ bool TransportDescriptionFactory::SetSecurityInfo(
   // RFC 4572 Section 5 requires that those lines use the same hash function as
   // the certificate's signature.
   std::string digest_alg;
-  if (!identity_->certificate().GetSignatureDigestAlgorithm(&digest_alg)) {
+  if (!certificate_->ssl_certificate().GetSignatureDigestAlgorithm(
+          &digest_alg)) {
     LOG(LS_ERROR) << "Failed to retrieve the certificate's digest algorithm";
     return false;
   }
 
   desc->identity_fingerprint.reset(
-      rtc::SSLFingerprint::Create(digest_alg, identity_));
+      rtc::SSLFingerprint::Create(digest_alg, certificate_->identity()));
   if (!desc->identity_fingerprint.get()) {
     LOG(LS_ERROR) << "Failed to create identity fingerprint, alg="
                   << digest_alg;
@@ -157,4 +124,3 @@ bool TransportDescriptionFactory::SetSecurityInfo(
 }
 
 }  // namespace cricket
-
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.h b/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.h
index a137f72115..828aa6d22c 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
 #define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
 
+#include "webrtc/base/rtccertificate.h"
 #include "webrtc/p2p/base/transportdescription.h"
 
 namespace rtc {
@@ -33,15 +34,18 @@ class TransportDescriptionFactory {
   // Default ctor; use methods below to set configuration.
   TransportDescriptionFactory();
   SecurePolicy secure() const { return secure_; }
-  // The identity to use when setting up DTLS.
-  rtc::SSLIdentity* identity() const { return identity_; }
+  // The certificate to use when setting up DTLS.
+  const rtc::scoped_refptr<rtc::RTCCertificate>& certificate() const {
+    return certificate_;
+  }
 
-  // Specifies the transport protocol to be use.
-  void set_protocol(TransportProtocol protocol) { protocol_ = protocol; }
   // Specifies the transport security policy to use.
   void set_secure(SecurePolicy s) { secure_ = s; }
-  // Specifies the identity to use (only used when secure is not SEC_DISABLED).
-  void set_identity(rtc::SSLIdentity* identity) { identity_ = identity; }
+  // Specifies the certificate to use (only used when secure != SEC_DISABLED).
+  void set_certificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+    certificate_ = certificate;
+  }
 
   // Creates a transport description suitable for use in an offer.
   TransportDescription* CreateOffer(const TransportOptions& options,
@@ -56,9 +60,8 @@ class TransportDescriptionFactory {
   bool SetSecurityInfo(TransportDescription* description,
                        ConnectionRole role) const;
 
-  TransportProtocol protocol_;
   SecurePolicy secure_;
-  rtc::SSLIdentity* identity_;
+  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
index 48267b57bb..a52d9ed95a 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
@@ -26,15 +26,15 @@ using cricket::TransportOptions;
 class TransportDescriptionFactoryTest : public testing::Test {
  public:
   TransportDescriptionFactoryTest()
-      : id1_(new rtc::FakeSSLIdentity("User1")),
-        id2_(new rtc::FakeSSLIdentity("User2")) {
-  }
+      : cert1_(rtc::RTCCertificate::Create(
+            scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("User1")))),
+        cert2_(rtc::RTCCertificate::Create(
+            scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("User2")))) {}
 
-  void CheckDesc(const TransportDescription* desc, const std::string& type,
+  void CheckDesc(const TransportDescription* desc,
                  const std::string& opt, const std::string& ice_ufrag,
                  const std::string& ice_pwd, const std::string& dtls_alg) {
     ASSERT_TRUE(desc != NULL);
-    EXPECT_EQ(type, desc->transport_type);
     EXPECT_EQ(!opt.empty(), desc->HasOption(opt));
     if (ice_ufrag.empty() && ice_pwd.empty()) {
       EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
@@ -62,8 +62,8 @@ class TransportDescriptionFactoryTest : public testing::Test {
     if (dtls) {
       f1_.set_secure(cricket::SEC_ENABLED);
       f2_.set_secure(cricket::SEC_ENABLED);
-      f1_.set_identity(id1_.get());
-      f2_.set_identity(id2_.get());
+      f1_.set_certificate(cert1_);
+      f2_.set_certificate(cert2_);
     } else {
       f1_.set_secure(cricket::SEC_DISABLED);
       f2_.set_secure(cricket::SEC_DISABLED);
@@ -114,170 +114,71 @@ class TransportDescriptionFactoryTest : public testing::Test {
  protected:
   TransportDescriptionFactory f1_;
   TransportDescriptionFactory f2_;
-  scoped_ptr<rtc::SSLIdentity> id1_;
-  scoped_ptr<rtc::SSLIdentity> id2_;
+
+  rtc::scoped_refptr<rtc::RTCCertificate> cert1_;
+  rtc::scoped_refptr<rtc::RTCCertificate> cert2_;
 };
 
-// Test that in the default case, we generate the expected G-ICE offer.
-TEST_F(TransportDescriptionFactoryTest, TestOfferGice) {
-  f1_.set_protocol(cricket::ICEPROTO_GOOGLE);
+TEST_F(TransportDescriptionFactoryTest, TestOfferDefault) {
   scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
       TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_GINGLE_P2P, "", "", "", "");
+  CheckDesc(desc.get(), "", "", "", "");
 }
 
-// Test generating a hybrid offer.
-TEST_F(TransportDescriptionFactoryTest, TestOfferHybrid) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
-  scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
-      TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "google-ice", "", "", "");
-}
-
-// Test generating an ICE-only offer.
-TEST_F(TransportDescriptionFactoryTest, TestOfferIce) {
-  f1_.set_protocol(cricket::ICEPROTO_RFC5245);
-  scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
-      TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
-}
-
-// Test generating a hybrid offer with DTLS.
-TEST_F(TransportDescriptionFactoryTest, TestOfferHybridDtls) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
+TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) {
   f1_.set_secure(cricket::SEC_ENABLED);
-  f1_.set_identity(id1_.get());
+  f1_.set_certificate(cert1_);
   std::string digest_alg;
-  ASSERT_TRUE(id1_->certificate().GetSignatureDigestAlgorithm(&digest_alg));
+  ASSERT_TRUE(cert1_->ssl_certificate().GetSignatureDigestAlgorithm(
+      &digest_alg));
   scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
       TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "google-ice", "", "",
-            digest_alg);
+  CheckDesc(desc.get(), "", "", "", digest_alg);
   // Ensure it also works with SEC_REQUIRED.
   f1_.set_secure(cricket::SEC_REQUIRED);
   desc.reset(f1_.CreateOffer(TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "google-ice", "", "",
-            digest_alg);
+  CheckDesc(desc.get(), "", "", "", digest_alg);
 }
 
-// Test generating a hybrid offer with DTLS fails with no identity.
-TEST_F(TransportDescriptionFactoryTest, TestOfferHybridDtlsWithNoIdentity) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
+// Test generating an offer with DTLS fails with no identity.
+TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsWithNoIdentity) {
   f1_.set_secure(cricket::SEC_ENABLED);
   scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
       TransportOptions(), NULL));
   ASSERT_TRUE(desc.get() == NULL);
 }
 
-// Test updating a hybrid offer with DTLS to pick ICE.
+// Test updating an offer with DTLS to pick ICE.
 // The ICE credentials should stay the same in the new offer.
-TEST_F(TransportDescriptionFactoryTest, TestOfferHybridDtlsReofferIceDtls) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
+TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsReofferDtls) {
   f1_.set_secure(cricket::SEC_ENABLED);
-  f1_.set_identity(id1_.get());
+  f1_.set_certificate(cert1_);
   std::string digest_alg;
-  ASSERT_TRUE(id1_->certificate().GetSignatureDigestAlgorithm(&digest_alg));
+  ASSERT_TRUE(cert1_->ssl_certificate().GetSignatureDigestAlgorithm(
+      &digest_alg));
   scoped_ptr<TransportDescription> old_desc(f1_.CreateOffer(
       TransportOptions(), NULL));
   ASSERT_TRUE(old_desc.get() != NULL);
-  f1_.set_protocol(cricket::ICEPROTO_RFC5245);
   scoped_ptr<TransportDescription> desc(
       f1_.CreateOffer(TransportOptions(), old_desc.get()));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "",
+  CheckDesc(desc.get(), "",
             old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg);
 }
 
-// Test that we can answer a GICE offer with GICE.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerGiceToGice) {
-  f1_.set_protocol(cricket::ICEPROTO_GOOGLE);
-  f2_.set_protocol(cricket::ICEPROTO_GOOGLE);
+TEST_F(TransportDescriptionFactoryTest, TestAnswerDefault) {
   scoped_ptr<TransportDescription> offer(f1_.CreateOffer(
       TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
   scoped_ptr<TransportDescription> desc(f2_.CreateAnswer(
       offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_GINGLE_P2P, "", "", "", "");
-  // Should get the same result when answering as hybrid.
-  f2_.set_protocol(cricket::ICEPROTO_HYBRID);
+  CheckDesc(desc.get(), "", "", "", "");
   desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(),
                               NULL));
-  CheckDesc(desc.get(), cricket::NS_GINGLE_P2P, "", "", "", "");
-}
-
-// Test that we can answer a hybrid offer with GICE.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerGiceToHybrid) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
-  f2_.set_protocol(cricket::ICEPROTO_GOOGLE);
-  scoped_ptr<TransportDescription> offer(f1_.CreateOffer(
-      TransportOptions(), NULL));
-  ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
-      f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_GINGLE_P2P, "", "", "", "");
-}
-
-// Test that we can answer a hybrid offer with ICE.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerIceToHybrid) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
-  f2_.set_protocol(cricket::ICEPROTO_RFC5245);
-  scoped_ptr<TransportDescription> offer(f1_.CreateOffer(
-      TransportOptions(), NULL));
-  ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
-      f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
-  // Should get the same result when answering as hybrid.
-  f2_.set_protocol(cricket::ICEPROTO_HYBRID);
-  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(),
-                              NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
-}
-
-// Test that we can answer an ICE offer with ICE.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerIceToIce) {
-  f1_.set_protocol(cricket::ICEPROTO_RFC5245);
-  f2_.set_protocol(cricket::ICEPROTO_RFC5245);
-  scoped_ptr<TransportDescription> offer(f1_.CreateOffer(
-      TransportOptions(), NULL));
-  ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(f2_.CreateAnswer(
-      offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
-  // Should get the same result when answering as hybrid.
-  f2_.set_protocol(cricket::ICEPROTO_HYBRID);
-  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(),
-                              NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
-}
-
-// Test that we can't answer a GICE offer with ICE.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerIceToGice) {
-  f1_.set_protocol(cricket::ICEPROTO_GOOGLE);
-  f2_.set_protocol(cricket::ICEPROTO_RFC5245);
-  scoped_ptr<TransportDescription> offer(
-      f1_.CreateOffer(TransportOptions(), NULL));
-  ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
-      f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
-  ASSERT_TRUE(desc.get() == NULL);
-}
-
-// Test that we can't answer an ICE offer with GICE.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerGiceToIce) {
-  f1_.set_protocol(cricket::ICEPROTO_RFC5245);
-  f2_.set_protocol(cricket::ICEPROTO_GOOGLE);
-  scoped_ptr<TransportDescription> offer(
-      f1_.CreateOffer(TransportOptions(), NULL));
-  ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(f2_.CreateAnswer(
-      offer.get(), TransportOptions(), NULL));
-  ASSERT_TRUE(desc.get() == NULL);
+  CheckDesc(desc.get(), "", "", "", "");
 }
 
 // Test that we can update an answer properly; ICE credentials shouldn't change.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerIceToIceReanswer) {
-  f1_.set_protocol(cricket::ICEPROTO_RFC5245);
-  f2_.set_protocol(cricket::ICEPROTO_RFC5245);
+TEST_F(TransportDescriptionFactoryTest, TestReanswer) {
   scoped_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
@@ -288,37 +189,33 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerIceToIceReanswer) {
       f2_.CreateAnswer(offer.get(), TransportOptions(),
                        old_desc.get()));
   ASSERT_TRUE(desc.get() != NULL);
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "",
+  CheckDesc(desc.get(), "",
             old_desc->ice_ufrag, old_desc->ice_pwd, "");
 }
 
 // Test that we handle answering an offer with DTLS with no DTLS.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerHybridToHybridDtls) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
+TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToNoDtls) {
   f1_.set_secure(cricket::SEC_ENABLED);
-  f1_.set_identity(id1_.get());
-  f2_.set_protocol(cricket::ICEPROTO_HYBRID);
+  f1_.set_certificate(cert1_);
   scoped_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
   scoped_ptr<TransportDescription> desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
+  CheckDesc(desc.get(), "", "", "", "");
 }
 
 // Test that we handle answering an offer without DTLS if we have DTLS enabled,
 // but fail if we require DTLS.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerHybridDtlsToHybrid) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
-  f2_.set_protocol(cricket::ICEPROTO_HYBRID);
+TEST_F(TransportDescriptionFactoryTest, TestAnswerNoDtlsToDtls) {
   f2_.set_secure(cricket::SEC_ENABLED);
-  f2_.set_identity(id2_.get());
+  f2_.set_certificate(cert2_);
   scoped_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
   scoped_ptr<TransportDescription> desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", "");
+  CheckDesc(desc.get(), "", "", "", "");
   f2_.set_secure(cricket::SEC_REQUIRED);
   desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(),
                               NULL));
@@ -327,29 +224,28 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerHybridDtlsToHybrid) {
 
 // Test that we handle answering an DTLS offer with DTLS, both if we have
 // DTLS enabled and required.
-TEST_F(TransportDescriptionFactoryTest, TestAnswerHybridDtlsToHybridDtls) {
-  f1_.set_protocol(cricket::ICEPROTO_HYBRID);
+TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToDtls) {
   f1_.set_secure(cricket::SEC_ENABLED);
-  f1_.set_identity(id1_.get());
+  f1_.set_certificate(cert1_);
 
-  f2_.set_protocol(cricket::ICEPROTO_HYBRID);
   f2_.set_secure(cricket::SEC_ENABLED);
-  f2_.set_identity(id2_.get());
+  f2_.set_certificate(cert2_);
   // f2_ produces the answer that is being checked in this test, so the
-  // answer must contain fingerprint lines with id2_'s digest algorithm.
+  // answer must contain fingerprint lines with cert2_'s digest algorithm.
   std::string digest_alg2;
-  ASSERT_TRUE(id2_->certificate().GetSignatureDigestAlgorithm(&digest_alg2));
+  ASSERT_TRUE(cert2_->ssl_certificate().GetSignatureDigestAlgorithm(
+      &digest_alg2));
 
   scoped_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
   scoped_ptr<TransportDescription> desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", digest_alg2);
+  CheckDesc(desc.get(), "", "", "", digest_alg2);
   f2_.set_secure(cricket::SEC_REQUIRED);
   desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(),
                               NULL));
-  CheckDesc(desc.get(), cricket::NS_JINGLE_ICE_UDP, "", "", "", digest_alg2);
+  CheckDesc(desc.get(), "", "", "", digest_alg2);
 }
 
 // Test that ice ufrag and password is changed in an updated offer and answer
diff --git a/media/webrtc/trunk/webrtc/p2p/base/turnport.cc b/media/webrtc/trunk/webrtc/p2p/base/turnport.cc
index a3e355e60a..5ed93dd1d8 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/turnport.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/turnport.cc
@@ -38,7 +38,9 @@ static const size_t TURN_CHANNEL_HEADER_SIZE = 4U;
 // STUN_ERROR_ALLOCATION_MISMATCH error per rfc5766.
 static const size_t MAX_ALLOCATE_MISMATCH_RETRIES = 2;
 
-inline bool IsTurnChannelData(uint16 msg_type) {
+static const int TURN_SUCCESS_RESULT_CODE = 0;
+
+inline bool IsTurnChannelData(uint16_t msg_type) {
   return ((msg_type & 0xC000) == 0x4000);  // MSB are 0b01
 }
 
@@ -57,10 +59,11 @@ static int GetRelayPreference(cricket::ProtocolType proto, bool secure) {
 class TurnAllocateRequest : public StunRequest {
  public:
   explicit TurnAllocateRequest(TurnPort* port);
-  virtual void Prepare(StunMessage* request);
-  virtual void OnResponse(StunMessage* response);
-  virtual void OnErrorResponse(StunMessage* response);
-  virtual void OnTimeout();
+  void Prepare(StunMessage* request) override;
+  void OnSent() override;
+  void OnResponse(StunMessage* response) override;
+  void OnErrorResponse(StunMessage* response) override;
+  void OnTimeout() override;
 
  private:
   // Handles authentication challenge from the server.
@@ -74,10 +77,11 @@ class TurnAllocateRequest : public StunRequest {
 class TurnRefreshRequest : public StunRequest {
  public:
   explicit TurnRefreshRequest(TurnPort* port);
-  virtual void Prepare(StunMessage* request);
-  virtual void OnResponse(StunMessage* response);
-  virtual void OnErrorResponse(StunMessage* response);
-  virtual void OnTimeout();
+  void Prepare(StunMessage* request) override;
+  void OnSent() override;
+  void OnResponse(StunMessage* response) override;
+  void OnErrorResponse(StunMessage* response) override;
+  void OnTimeout() override;
   void set_lifetime(int lifetime) { lifetime_ = lifetime; }
 
  private:
@@ -90,10 +94,11 @@ class TurnCreatePermissionRequest : public StunRequest,
  public:
   TurnCreatePermissionRequest(TurnPort* port, TurnEntry* entry,
                               const rtc::SocketAddress& ext_addr);
-  virtual void Prepare(StunMessage* request);
-  virtual void OnResponse(StunMessage* response);
-  virtual void OnErrorResponse(StunMessage* response);
-  virtual void OnTimeout();
+  void Prepare(StunMessage* request) override;
+  void OnSent() override;
+  void OnResponse(StunMessage* response) override;
+  void OnErrorResponse(StunMessage* response) override;
+  void OnTimeout() override;
 
  private:
   void OnEntryDestroyed(TurnEntry* entry);
@@ -108,10 +113,11 @@ class TurnChannelBindRequest : public StunRequest,
  public:
   TurnChannelBindRequest(TurnPort* port, TurnEntry* entry, int channel_id,
                          const rtc::SocketAddress& ext_addr);
-  virtual void Prepare(StunMessage* request);
-  virtual void OnResponse(StunMessage* response);
-  virtual void OnErrorResponse(StunMessage* response);
-  virtual void OnTimeout();
+  void Prepare(StunMessage* request) override;
+  void OnSent() override;
+  void OnResponse(StunMessage* response) override;
+  void OnErrorResponse(StunMessage* response) override;
+  void OnTimeout() override;
 
  private:
   void OnEntryDestroyed(TurnEntry* entry);
@@ -133,11 +139,19 @@ class TurnEntry : public sigslot::has_slots<> {
   TurnPort* port() { return port_; }
 
   int channel_id() const { return channel_id_; }
+  // For testing only.
+  void set_channel_id(int channel_id) { channel_id_ = channel_id; }
+
   const rtc::SocketAddress& address() const { return ext_addr_; }
   BindState state() const { return state_; }
 
+  uint32_t destruction_timestamp() { return destruction_timestamp_; }
+  void set_destruction_timestamp(uint32_t destruction_timestamp) {
+    destruction_timestamp_ = destruction_timestamp;
+  }
+
   // Helper methods to send permission and channel bind requests.
-  void SendCreatePermissionRequest();
+  void SendCreatePermissionRequest(int delay);
   void SendChannelBindRequest(int delay);
   // Sends a packet to the given destination address.
   // This will wrap the packet in STUN if necessary.
@@ -146,8 +160,10 @@ class TurnEntry : public sigslot::has_slots<> {
 
   void OnCreatePermissionSuccess();
   void OnCreatePermissionError(StunMessage* response, int code);
+  void OnCreatePermissionTimeout();
   void OnChannelBindSuccess();
   void OnChannelBindError(StunMessage* response, int code);
+  void OnChannelBindTimeout();
   // Signal sent when TurnEntry is destroyed.
   sigslot::signal1<TurnEntry*> SignalDestroyed;
 
@@ -156,6 +172,11 @@ class TurnEntry : public sigslot::has_slots<> {
   int channel_id_;
   rtc::SocketAddress ext_addr_;
   BindState state_;
+  // A non-zero value indicates that this entry is scheduled to be destroyed.
+  // It is also used as an ID of the event scheduling. When the destruction
+  // event actually fires, the TurnEntry will be destroyed only if the
+  // timestamp here matches the one in the firing event.
+  uint32_t destruction_timestamp_ = 0;
 };
 
 TurnPort::TurnPort(rtc::Thread* thread,
@@ -168,8 +189,12 @@ TurnPort::TurnPort(rtc::Thread* thread,
                    const RelayCredentials& credentials,
                    int server_priority,
                    const std::string& origin)
-    : Port(thread, factory, network, socket->GetLocalAddress().ipaddr(),
-           username, password),
+    : Port(thread,
+           factory,
+           network,
+           socket->GetLocalAddress().ipaddr(),
+           username,
+           password),
       server_address_(server_address),
       credentials_(credentials),
       socket_(socket),
@@ -177,7 +202,7 @@ TurnPort::TurnPort(rtc::Thread* thread,
       error_(0),
       request_manager_(thread),
       next_channel_number_(TURN_CHANNEL_NUMBER_START),
-      connected_(false),
+      state_(STATE_CONNECTING),
       server_priority_(server_priority),
       allocate_mismatch_retries_(0) {
   request_manager_.SignalSendPacket.connect(this, &TurnPort::OnSendStunPacket);
@@ -188,16 +213,23 @@ TurnPort::TurnPort(rtc::Thread* thread,
                    rtc::PacketSocketFactory* factory,
                    rtc::Network* network,
                    const rtc::IPAddress& ip,
-                   uint16 min_port,
-                   uint16 max_port,
+                   uint16_t min_port,
+                   uint16_t max_port,
                    const std::string& username,
                    const std::string& password,
                    const ProtocolAddress& server_address,
                    const RelayCredentials& credentials,
                    int server_priority,
                    const std::string& origin)
-    : Port(thread, RELAY_PORT_TYPE, factory, network, ip, min_port, max_port,
-           username, password),
+    : Port(thread,
+           RELAY_PORT_TYPE,
+           factory,
+           network,
+           ip,
+           min_port,
+           max_port,
+           username,
+           password),
       server_address_(server_address),
       credentials_(credentials),
       socket_(NULL),
@@ -205,7 +237,7 @@ TurnPort::TurnPort(rtc::Thread* thread,
       error_(0),
       request_manager_(thread),
       next_channel_number_(TURN_CHANNEL_NUMBER_START),
-      connected_(false),
+      state_(STATE_CONNECTING),
       server_priority_(server_priority),
       allocate_mismatch_retries_(0) {
   request_manager_.SignalSendPacket.connect(this, &TurnPort::OnSendStunPacket);
@@ -217,14 +249,14 @@ TurnPort::~TurnPort() {
 
   // release the allocation by sending a refresh with
   // lifetime 0.
-  if (connected_) {
+  if (ready()) {
     TurnRefreshRequest bye(this);
     bye.set_lifetime(0);
     SendRequest(&bye, 0);
   }
 
   while (!entries_.empty()) {
-    DestroyEntry(entries_.front()->address());
+    DestroyEntry(entries_.front());
   }
   if (resolver_) {
     resolver_->Destroy(false);
@@ -252,13 +284,14 @@ void TurnPort::PrepareAddress() {
     server_address_.address.SetPort(TURN_DEFAULT_PORT);
   }
 
-  if (server_address_.address.IsUnresolved()) {
+  if (server_address_.address.IsUnresolvedIP()) {
     ResolveTurnAddress(server_address_.address);
   } else {
     // If protocol family of server address doesn't match with local, return.
     if (!IsCompatibleAddress(server_address_.address)) {
-      LOG(LS_ERROR) << "Server IP address family does not match with "
-                    << "local host address family type";
+      LOG(LS_ERROR) << "IP address family does not match: "
+                    << "server: " << server_address_.address.family()
+                    << "local: " << ip().family();
       OnAllocateError();
       return;
     }
@@ -270,8 +303,11 @@ void TurnPort::PrepareAddress() {
                          << ProtoToString(server_address_.proto) << " @ "
                          << server_address_.address.ToSensitiveString();
     if (!CreateTurnClientSocket()) {
+      LOG(LS_ERROR) << "Failed to create TURN client socket";
       OnAllocateError();
-    } else if (server_address_.proto == PROTO_UDP) {
+      return;
+    }
+    if (server_address_.proto == PROTO_UDP) {
       // If its UDP, send AllocateRequest now.
       // For TCP and TLS AllcateRequest will be sent by OnSocketConnect.
       SendRequest(new TurnAllocateRequest(this), 0);
@@ -315,9 +351,15 @@ bool TurnPort::CreateTurnClientSocket() {
 
   socket_->SignalReadyToSend.connect(this, &TurnPort::OnReadyToSend);
 
+  socket_->SignalSentPacket.connect(this, &TurnPort::OnSentPacket);
+
+  // TCP port is ready to send stun requests after the socket is connected,
+  // while UDP port is ready to do so once the socket is created.
   if (server_address_.proto == PROTO_TCP) {
     socket_->SignalConnect.connect(this, &TurnPort::OnSocketConnect);
     socket_->SignalClose.connect(this, &TurnPort::OnSocketClose);
+  } else {
+    state_ = STATE_CONNECTED;
   }
   return true;
 }
@@ -329,15 +371,23 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
   // given a binding address, and the platform is expected to pick the
   // correct local address.
 
-  // Further, to workaround issue 3927 in which a proxy is forcing TCP bound to
-  // localhost only, we're allowing Loopback IP even if it's not the same as the
-  // local Turn port.
+  // However, there are two situations in which we allow the bound address to
+  // differ from the requested address: 1. The bound address is the loopback
+  // address.  This happens when a proxy forces TCP to bind to only the
+  // localhost address (see issue 3927). 2. The bound address is the "any
+  // address".  This happens when multiple_routes is disabled (see issue 4780).
   if (socket->GetLocalAddress().ipaddr() != ip()) {
     if (socket->GetLocalAddress().IsLoopbackIP()) {
       LOG(LS_WARNING) << "Socket is bound to a different address:"
                       << socket->GetLocalAddress().ipaddr().ToString()
                       << ", rather then the local port:" << ip().ToString()
                       << ". Still allowing it since it's localhost.";
+    } else if (IPIsAny(ip())) {
+      LOG(LS_WARNING) << "Socket is bound to a different address:"
+                      << socket->GetLocalAddress().ipaddr().ToString()
+                      << ", rather then the local port:" << ip().ToString()
+                      << ". Still allowing it since it's any address"
+                      << ", possibly caused by multiple_routes being disabled.";
     } else {
       LOG(LS_WARNING) << "Socket is bound to a different address:"
                       << socket->GetLocalAddress().ipaddr().ToString()
@@ -348,7 +398,8 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
     }
   }
 
-  if (server_address_.address.IsUnresolved()) {
+  state_ = STATE_CONNECTED;  // It is ready to send stun requests.
+  if (server_address_.address.IsUnresolvedIP()) {
     server_address_.address = socket_->GetRemoteAddress();
   }
 
@@ -360,10 +411,7 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
 void TurnPort::OnSocketClose(rtc::AsyncPacketSocket* socket, int error) {
   LOG_J(LS_WARNING, this) << "Connection with server failed, error=" << error;
   ASSERT(socket == socket_);
-  if (!connected_) {
-    OnAllocateError();
-  }
-  connected_ = false;
+  Close();
 }
 
 void TurnPort::OnAllocateMismatch() {
@@ -392,7 +440,7 @@ void TurnPort::OnAllocateMismatch() {
 Connection* TurnPort::CreateConnection(const Candidate& address,
                                        CandidateOrigin origin) {
   // TURN-UDP can only connect to UDP candidates.
-  if (address.protocol() != UDP_PROTOCOL_NAME) {
+  if (!SupportsProtocol(address.protocol())) {
     return NULL;
   }
 
@@ -400,8 +448,12 @@ Connection* TurnPort::CreateConnection(const Candidate& address,
     return NULL;
   }
 
+  if (state_ == STATE_DISCONNECTED) {
+    return NULL;
+  }
+
   // Create an entry, if needed, so we can get our permissions set up correctly.
-  CreateEntry(address.address());
+  CreateOrRefreshEntry(address.address());
 
   // A TURN port will have two candiates, STUN and TURN. STUN may not
   // present in all cases. If present stun candidate will be added first
@@ -417,6 +469,15 @@ Connection* TurnPort::CreateConnection(const Candidate& address,
   return NULL;
 }
 
+bool TurnPort::DestroyConnection(const rtc::SocketAddress& address) {
+  Connection* conn = GetConnection(address);
+  if (conn != nullptr) {
+    conn->Destroy();
+    return true;
+  }
+  return false;
+}
+
 int TurnPort::SetOption(rtc::Socket::Option opt, int value) {
   if (!socket_) {
     // If socket is not created yet, these options will be applied during socket
@@ -450,12 +511,12 @@ int TurnPort::SendTo(const void* data, size_t size,
                      bool payload) {
   // Try to find an entry for this specific address; we should have one.
   TurnEntry* entry = FindEntry(addr);
-  ASSERT(entry != NULL);
   if (!entry) {
+    LOG(LS_ERROR) << "Did not find the TurnEntry for address " << addr;
     return 0;
   }
 
-  if (!connected()) {
+  if (!ready()) {
     error_ = EWOULDBLOCK;
     return SOCKET_ERROR;
   }
@@ -497,7 +558,7 @@ void TurnPort::OnReadPacket(
   // Check the message type, to see if is a Channel Data message.
   // The message will either be channel data, a TURN data indication, or
   // a response to a previous request.
-  uint16 msg_type = rtc::GetBE16(data);
+  uint16_t msg_type = rtc::GetBE16(data);
   if (IsTurnChannelData(msg_type)) {
     HandleChannelData(msg_type, data, size, packet_time);
   } else if (msg_type == TURN_DATA_INDICATION) {
@@ -523,8 +584,13 @@ void TurnPort::OnReadPacket(
   }
 }
 
+void TurnPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
+                            const rtc::SentPacket& sent_packet) {
+  PortInterface::SignalSentPacket(sent_packet);
+}
+
 void TurnPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
-  if (connected_) {
+  if (ready()) {
     Port::OnReadyToSend();
   }
 }
@@ -565,6 +631,8 @@ void TurnPort::ResolveTurnAddress(const rtc::SocketAddress& address) {
   if (resolver_)
     return;
 
+  LOG_J(LS_INFO, this) << "Starting TURN host lookup for "
+                       << address.ToSensitiveString();
   resolver_ = socket_factory()->CreateAsyncResolver();
   resolver_->SignalDone.connect(this, &TurnPort::OnResolveResult);
   resolver_->Start(address);
@@ -604,6 +672,7 @@ void TurnPort::OnResolveResult(rtc::AsyncResolverInterface* resolver) {
 
 void TurnPort::OnSendStunPacket(const void* data, size_t size,
                                 StunRequest* request) {
+  ASSERT(connected());
   rtc::PacketOptions options(DefaultDscpValue());
   if (Send(data, size, options) < 0) {
     LOG_J(LS_ERROR, this) << "Failed to send TURN message, err="
@@ -623,7 +692,7 @@ void TurnPort::OnStunAddress(const rtc::SocketAddress& address) {
 
 void TurnPort::OnAllocateSuccess(const rtc::SocketAddress& address,
                                  const rtc::SocketAddress& stun_address) {
-  connected_ = true;
+  state_ = STATE_READY;
 
   rtc::SocketAddress related_address = stun_address;
     if (!(candidate_filter() & CF_REFLEXIVE)) {
@@ -637,46 +706,70 @@ void TurnPort::OnAllocateSuccess(const rtc::SocketAddress& address,
              address,          // Base address.
              related_address,  // Related address.
              UDP_PROTOCOL_NAME,
+             ProtoToString(server_address_.proto),  // The first hop protocol.
              "",  // TCP canddiate type, empty for turn candidates.
              RELAY_PORT_TYPE,
              GetRelayPreference(server_address_.proto, server_address_.secure),
-             server_priority_,
-             true);
+             server_priority_, true);
 }
 
 void TurnPort::OnAllocateError() {
   // We will send SignalPortError asynchronously as this can be sent during
   // port initialization. This way it will not be blocking other port
   // creation.
-  thread()->Post(this, MSG_ERROR);
+  thread()->Post(this, MSG_ALLOCATE_ERROR);
+}
+
+void TurnPort::OnTurnRefreshError() {
+  // Need to Close the port asynchronously because otherwise, the refresh
+  // request may be deleted twice: once at the end of the message processing
+  // and the other in Close().
+  thread()->Post(this, MSG_REFRESH_ERROR);
+}
+
+void TurnPort::Close() {
+  if (!ready()) {
+    OnAllocateError();
+  }
+  request_manager_.Clear();
+  // Stop the port from creating new connections.
+  state_ = STATE_DISCONNECTED;
+  // Delete all existing connections; stop sending data.
+  for (auto kv : connections()) {
+    kv.second->Destroy();
+  }
 }
 
 void TurnPort::OnMessage(rtc::Message* message) {
-  if (message->message_id == MSG_ERROR) {
-    SignalPortError(this);
-    return;
-  } else if (message->message_id == MSG_ALLOCATE_MISMATCH) {
-    OnAllocateMismatch();
-    return;
-  } else if (message->message_id == MSG_TRY_ALTERNATE_SERVER) {
-    if (server_address().proto == PROTO_UDP) {
-      // Send another allocate request to alternate server, with the received
-      // realm and nonce values.
-      SendRequest(new TurnAllocateRequest(this), 0);
-    } else {
-      // Since it's TCP, we have to delete the connected socket and reconnect
-      // with the alternate server. PrepareAddress will send stun binding once
-      // the new socket is connected.
-      ASSERT(server_address().proto == PROTO_TCP);
-      ASSERT(!SharedSocket());
-      delete socket_;
-      socket_ = NULL;
-      PrepareAddress();
-    }
-    return;
+  switch (message->message_id) {
+    case MSG_ALLOCATE_ERROR:
+      SignalPortError(this);
+      break;
+    case MSG_ALLOCATE_MISMATCH:
+      OnAllocateMismatch();
+      break;
+    case MSG_REFRESH_ERROR:
+      Close();
+      break;
+    case MSG_TRY_ALTERNATE_SERVER:
+      if (server_address().proto == PROTO_UDP) {
+        // Send another allocate request to alternate server, with the received
+        // realm and nonce values.
+        SendRequest(new TurnAllocateRequest(this), 0);
+      } else {
+        // Since it's TCP, we have to delete the connected socket and reconnect
+        // with the alternate server. PrepareAddress will send stun binding once
+        // the new socket is connected.
+        ASSERT(server_address().proto == PROTO_TCP);
+        ASSERT(!SharedSocket());
+        delete socket_;
+        socket_ = NULL;
+        PrepareAddress();
+      }
+      break;
+    default:
+      Port::OnMessage(message);
   }
-
-  Port::OnMessage(message);
 }
 
 void TurnPort::OnAllocateRequestTimeout() {
@@ -741,7 +834,7 @@ void TurnPort::HandleChannelData(int channel_id, const char* data,
   //   +-------------------------------+
 
   // Extract header fields from the message.
-  uint16 len = rtc::GetBE16(data + 2);
+  uint16_t len = rtc::GetBE16(data + 2);
   if (len > size - TURN_CHANNEL_HEADER_SIZE) {
     LOG_J(LS_WARNING, this) << "Received TURN channel data message with "
                             << "incorrect length, len=" << len;
@@ -778,7 +871,9 @@ bool TurnPort::ScheduleRefresh(int lifetime) {
     return false;
   }
 
-  SendRequest(new TurnRefreshRequest(this), (lifetime - 60) * 1000);
+  int delay = (lifetime - 60) * 1000;
+  SendRequest(new TurnRefreshRequest(this), delay);
+  LOG_J(LS_INFO, this) << "Scheduled refresh in " << delay << "ms.";
   return true;
 }
 
@@ -858,24 +953,73 @@ TurnEntry* TurnPort::FindEntry(int channel_id) const {
   return (it != entries_.end()) ? *it : NULL;
 }
 
-TurnEntry* TurnPort::CreateEntry(const rtc::SocketAddress& addr) {
-  ASSERT(FindEntry(addr) == NULL);
-  TurnEntry* entry = new TurnEntry(this, next_channel_number_++, addr);
-  entries_.push_back(entry);
-  return entry;
+bool TurnPort::EntryExists(TurnEntry* e) {
+  auto it = std::find(entries_.begin(), entries_.end(), e);
+  return it != entries_.end();
 }
 
-void TurnPort::DestroyEntry(const rtc::SocketAddress& addr) {
+void TurnPort::CreateOrRefreshEntry(const rtc::SocketAddress& addr) {
   TurnEntry* entry = FindEntry(addr);
+  if (entry == nullptr) {
+    entry = new TurnEntry(this, next_channel_number_++, addr);
+    entries_.push_back(entry);
+  } else {
+    // The channel binding request for the entry will be refreshed automatically
+    // until the entry is destroyed.
+    CancelEntryDestruction(entry);
+  }
+}
+
+void TurnPort::DestroyEntry(TurnEntry* entry) {
   ASSERT(entry != NULL);
   entry->SignalDestroyed(entry);
   entries_.remove(entry);
   delete entry;
 }
 
+void TurnPort::DestroyEntryIfNotCancelled(TurnEntry* entry,
+                                          uint32_t timestamp) {
+  if (!EntryExists(entry)) {
+    return;
+  }
+  bool cancelled = timestamp != entry->destruction_timestamp();
+  if (!cancelled) {
+    DestroyEntry(entry);
+  }
+}
+
 void TurnPort::OnConnectionDestroyed(Connection* conn) {
-  // Destroying TurnEntry for the connection, which is already destroyed.
-  DestroyEntry(conn->remote_candidate().address());
+  // Schedule an event to destroy TurnEntry for the connection, which is
+  // already destroyed.
+  const rtc::SocketAddress& remote_address = conn->remote_candidate().address();
+  TurnEntry* entry = FindEntry(remote_address);
+  ASSERT(entry != NULL);
+  ScheduleEntryDestruction(entry);
+}
+
+void TurnPort::ScheduleEntryDestruction(TurnEntry* entry) {
+  ASSERT(entry->destruction_timestamp() == 0);
+  uint32_t timestamp = rtc::Time();
+  entry->set_destruction_timestamp(timestamp);
+  invoker_.AsyncInvokeDelayed<void>(
+      thread(),
+      rtc::Bind(&TurnPort::DestroyEntryIfNotCancelled, this, entry, timestamp),
+      TURN_PERMISSION_TIMEOUT);
+}
+
+void TurnPort::CancelEntryDestruction(TurnEntry* entry) {
+  ASSERT(entry->destruction_timestamp() != 0);
+  entry->set_destruction_timestamp(0);
+}
+
+bool TurnPort::SetEntryChannelId(const rtc::SocketAddress& address,
+                                 int channel_id) {
+  TurnEntry* entry = FindEntry(address);
+  if (!entry) {
+    return false;
+  }
+  entry->set_channel_id(channel_id);
+  return true;
 }
 
 TurnAllocateRequest::TurnAllocateRequest(TurnPort* port)
@@ -895,7 +1039,18 @@ void TurnAllocateRequest::Prepare(StunMessage* request) {
   }
 }
 
+void TurnAllocateRequest::OnSent() {
+  LOG_J(LS_INFO, port_) << "TURN allocate request sent"
+                        << ", id=" << rtc::hex_encode(id());
+  StunRequest::OnSent();
+}
+
 void TurnAllocateRequest::OnResponse(StunMessage* response) {
+  LOG_J(LS_INFO, port_) << "TURN allocate requested successfully"
+                        << ", id=" << rtc::hex_encode(id())
+                        << ", code=0"  // Makes logging easier to parse.
+                        << ", rtt=" << Elapsed();
+
   // Check mandatory attributes as indicated in RFC5766, Section 6.3.
   const StunAddressAttribute* mapped_attr =
       response->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
@@ -931,6 +1086,12 @@ void TurnAllocateRequest::OnResponse(StunMessage* response) {
 void TurnAllocateRequest::OnErrorResponse(StunMessage* response) {
   // Process error response according to RFC5766, Section 6.4.
   const StunErrorCodeAttribute* error_code = response->GetErrorCode();
+
+  LOG_J(LS_INFO, port_) << "Received TURN allocate error response"
+                        << ", id=" << rtc::hex_encode(id())
+                        << ", code=" << error_code->code()
+                        << ", rtt=" << Elapsed();
+
   switch (error_code->code()) {
     case STUN_ERROR_UNAUTHORIZED:       // Unauthrorized.
       OnAuthChallenge(response, error_code->code());
@@ -944,14 +1105,17 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) {
       port_->thread()->Post(port_, TurnPort::MSG_ALLOCATE_MISMATCH);
       break;
     default:
-      LOG_J(LS_WARNING, port_) << "Allocate response error, code="
-                               << error_code->code();
+      LOG_J(LS_WARNING, port_) << "Received TURN allocate error response"
+                               << ", id=" << rtc::hex_encode(id())
+                               << ", code=" << error_code->code()
+                               << ", rtt=" << Elapsed();
       port_->OnAllocateError();
   }
 }
 
 void TurnAllocateRequest::OnTimeout() {
-  LOG_J(LS_WARNING, port_) << "Allocate request timeout";
+  LOG_J(LS_WARNING, port_) << "TURN allocate request "
+                           << rtc::hex_encode(id()) << " timout";
   port_->OnAllocateRequestTimeout();
 }
 
@@ -1048,7 +1212,18 @@ void TurnRefreshRequest::Prepare(StunMessage* request) {
   port_->AddRequestAuthInfo(request);
 }
 
+void TurnRefreshRequest::OnSent() {
+  LOG_J(LS_INFO, port_) << "TURN refresh request sent"
+                        << ", id=" << rtc::hex_encode(id());
+  StunRequest::OnSent();
+}
+
 void TurnRefreshRequest::OnResponse(StunMessage* response) {
+  LOG_J(LS_INFO, port_) << "TURN refresh requested successfully"
+                        << ", id=" << rtc::hex_encode(id())
+                        << ", code=0"  // Makes logging easier to parse.
+                        << ", rtt=" << Elapsed();
+
   // Check mandatory attributes as indicated in RFC5766, Section 7.3.
   const StunUInt32Attribute* lifetime_attr =
       response->GetUInt32(STUN_ATTR_TURN_LIFETIME);
@@ -1060,22 +1235,30 @@ void TurnRefreshRequest::OnResponse(StunMessage* response) {
 
   // Schedule a refresh based on the returned lifetime value.
   port_->ScheduleRefresh(lifetime_attr->value());
+  port_->SignalTurnRefreshResult(port_, TURN_SUCCESS_RESULT_CODE);
 }
 
 void TurnRefreshRequest::OnErrorResponse(StunMessage* response) {
   const StunErrorCodeAttribute* error_code = response->GetErrorCode();
-  LOG_J(LS_WARNING, port_) << "Refresh response error, code="
-                           << error_code->code();
 
   if (error_code->code() == STUN_ERROR_STALE_NONCE) {
     if (port_->UpdateNonce(response)) {
       // Send RefreshRequest immediately.
       port_->SendRequest(new TurnRefreshRequest(port_), 0);
     }
+  } else {
+    LOG_J(LS_WARNING, port_) << "Received TURN refresh error response"
+                             << ", id=" << rtc::hex_encode(id())
+                             << ", code=" << error_code->code()
+                             << ", rtt=" << Elapsed();
+    port_->OnTurnRefreshError();
+    port_->SignalTurnRefreshResult(port_, error_code->code());
   }
 }
 
 void TurnRefreshRequest::OnTimeout() {
+  LOG_J(LS_WARNING, port_) << "TURN refresh timeout " << rtc::hex_encode(id());
+  port_->OnTurnRefreshError();
 }
 
 TurnCreatePermissionRequest::TurnCreatePermissionRequest(
@@ -1097,21 +1280,40 @@ void TurnCreatePermissionRequest::Prepare(StunMessage* request) {
   port_->AddRequestAuthInfo(request);
 }
 
+void TurnCreatePermissionRequest::OnSent() {
+  LOG_J(LS_INFO, port_) << "TURN create permission request sent"
+                        << ", id=" << rtc::hex_encode(id());
+  StunRequest::OnSent();
+}
+
 void TurnCreatePermissionRequest::OnResponse(StunMessage* response) {
+  LOG_J(LS_INFO, port_) << "TURN permission requested successfully"
+                        << ", id=" << rtc::hex_encode(id())
+                        << ", code=0"  // Makes logging easier to parse.
+                        << ", rtt=" << Elapsed();
+
   if (entry_) {
     entry_->OnCreatePermissionSuccess();
   }
 }
 
 void TurnCreatePermissionRequest::OnErrorResponse(StunMessage* response) {
+  const StunErrorCodeAttribute* error_code = response->GetErrorCode();
+  LOG_J(LS_WARNING, port_) << "Received TURN create permission error response"
+                           << ", id=" << rtc::hex_encode(id())
+                           << ", code=" << error_code->code()
+                           << ", rtt=" << Elapsed();
   if (entry_) {
-    const StunErrorCodeAttribute* error_code = response->GetErrorCode();
     entry_->OnCreatePermissionError(response, error_code->code());
   }
 }
 
 void TurnCreatePermissionRequest::OnTimeout() {
-  LOG_J(LS_WARNING, port_) << "Create permission timeout";
+  LOG_J(LS_WARNING, port_) << "TURN create permission timeout "
+                           << rtc::hex_encode(id());
+  if (entry_) {
+    entry_->OnCreatePermissionTimeout();
+  }
 }
 
 void TurnCreatePermissionRequest::OnEntryDestroyed(TurnEntry* entry) {
@@ -1141,26 +1343,47 @@ void TurnChannelBindRequest::Prepare(StunMessage* request) {
   port_->AddRequestAuthInfo(request);
 }
 
+void TurnChannelBindRequest::OnSent() {
+  LOG_J(LS_INFO, port_) << "TURN channel bind request sent"
+                        << ", id=" << rtc::hex_encode(id());
+  StunRequest::OnSent();
+}
+
 void TurnChannelBindRequest::OnResponse(StunMessage* response) {
+  LOG_J(LS_INFO, port_) << "TURN channel bind requested successfully"
+                        << ", id=" << rtc::hex_encode(id())
+                        << ", code=0"  // Makes logging easier to parse.
+                        << ", rtt=" << Elapsed();
+
   if (entry_) {
     entry_->OnChannelBindSuccess();
     // Refresh the channel binding just under the permission timeout
     // threshold. The channel binding has a longer lifetime, but
     // this is the easiest way to keep both the channel and the
     // permission from expiring.
-    entry_->SendChannelBindRequest(TURN_PERMISSION_TIMEOUT - 60 * 1000);
+    int delay = TURN_PERMISSION_TIMEOUT - 60000;
+    entry_->SendChannelBindRequest(delay);
+    LOG_J(LS_INFO, port_) << "Scheduled channel bind in " << delay << "ms.";
   }
 }
 
 void TurnChannelBindRequest::OnErrorResponse(StunMessage* response) {
+  const StunErrorCodeAttribute* error_code = response->GetErrorCode();
+  LOG_J(LS_WARNING, port_) << "Received TURN channel bind error response"
+                           << ", id=" << rtc::hex_encode(id())
+                           << ", code=" << error_code->code()
+                           << ", rtt=" << Elapsed();
   if (entry_) {
-    const StunErrorCodeAttribute* error_code = response->GetErrorCode();
     entry_->OnChannelBindError(response, error_code->code());
   }
 }
 
 void TurnChannelBindRequest::OnTimeout() {
-  LOG_J(LS_WARNING, port_) << "Channel bind timeout";
+  LOG_J(LS_WARNING, port_) << "TURN channel bind timeout "
+                           << rtc::hex_encode(id());
+  if (entry_) {
+    entry_->OnChannelBindTimeout();
+  }
 }
 
 void TurnChannelBindRequest::OnEntryDestroyed(TurnEntry* entry) {
@@ -1175,12 +1398,12 @@ TurnEntry::TurnEntry(TurnPort* port, int channel_id,
       ext_addr_(ext_addr),
       state_(STATE_UNBOUND) {
   // Creating permission for |ext_addr_|.
-  SendCreatePermissionRequest();
+  SendCreatePermissionRequest(0);
 }
 
-void TurnEntry::SendCreatePermissionRequest() {
-  port_->SendRequest(new TurnCreatePermissionRequest(
-      port_, this, ext_addr_), 0);
+void TurnEntry::SendCreatePermissionRequest(int delay) {
+  port_->SendRequest(new TurnCreatePermissionRequest(port_, this, ext_addr_),
+                     delay);
 }
 
 void TurnEntry::SendChannelBindRequest(int delay) {
@@ -1211,7 +1434,7 @@ int TurnEntry::Send(const void* data, size_t size, bool payload,
   } else {
     // If the channel is bound, we can send the data as a Channel Message.
     buf.WriteUInt16(channel_id_);
-    buf.WriteUInt16(static_cast<uint16>(size));
+    buf.WriteUInt16(static_cast<uint16_t>(size));
     buf.WriteBytes(reinterpret_cast<const char*>(data), size);
   }
   return port_->Send(buf.Data(), buf.Length(), options);
@@ -1221,24 +1444,43 @@ void TurnEntry::OnCreatePermissionSuccess() {
   LOG_J(LS_INFO, port_) << "Create permission for "
                         << ext_addr_.ToSensitiveString()
                         << " succeeded";
-  // For success result code will be 0.
-  port_->SignalCreatePermissionResult(port_, ext_addr_, 0);
+  port_->SignalCreatePermissionResult(port_, ext_addr_,
+                                      TURN_SUCCESS_RESULT_CODE);
+
+  // If |state_| is STATE_BOUND, the permission will be refreshed
+  // by ChannelBindRequest.
+  if (state_ != STATE_BOUND) {
+    // Refresh the permission request about 1 minute before the permission
+    // times out.
+    int delay = TURN_PERMISSION_TIMEOUT - 60000;
+    SendCreatePermissionRequest(delay);
+    LOG_J(LS_INFO, port_) << "Scheduled create-permission-request in "
+                          << delay << "ms.";
+  }
 }
 
 void TurnEntry::OnCreatePermissionError(StunMessage* response, int code) {
-  LOG_J(LS_WARNING, port_) << "Create permission for "
-                           << ext_addr_.ToSensitiveString()
-                           << " failed, code=" << code;
   if (code == STUN_ERROR_STALE_NONCE) {
     if (port_->UpdateNonce(response)) {
-      SendCreatePermissionRequest();
+      SendCreatePermissionRequest(0);
     }
   } else {
+    port_->DestroyConnection(ext_addr_);
     // Send signal with error code.
     port_->SignalCreatePermissionResult(port_, ext_addr_, code);
+    Connection* c = port_->GetConnection(ext_addr_);
+    if (c) {
+      LOG_J(LS_ERROR, c) << "Received TURN CreatePermission error response, "
+                         << "code=" << code << "; killing connection.";
+      c->FailAndDestroy();
+    }
   }
 }
 
+void TurnEntry::OnCreatePermissionTimeout() {
+  port_->DestroyConnection(ext_addr_);
+}
+
 void TurnEntry::OnChannelBindSuccess() {
   LOG_J(LS_INFO, port_) << "Channel bind for " << ext_addr_.ToSensitiveString()
                         << " succeeded";
@@ -1247,17 +1489,21 @@ void TurnEntry::OnChannelBindSuccess() {
 }
 
 void TurnEntry::OnChannelBindError(StunMessage* response, int code) {
-  // TODO(mallinath) - Implement handling of error response for channel
-  // bind request as per http://tools.ietf.org/html/rfc5766#section-11.3
-  LOG_J(LS_WARNING, port_) << "Channel bind for "
-                           << ext_addr_.ToSensitiveString()
-                           << " failed, code=" << code;
+  // If the channel bind fails due to errors other than STATE_NONCE,
+  // we just destroy the connection and rely on ICE restart to re-establish
+  // the connection.
   if (code == STUN_ERROR_STALE_NONCE) {
     if (port_->UpdateNonce(response)) {
       // Send channel bind request with fresh nonce.
       SendChannelBindRequest(0);
     }
+  } else {
+    state_ = STATE_UNBOUND;
+    port_->DestroyConnection(ext_addr_);
   }
 }
-
+void TurnEntry::OnChannelBindTimeout() {
+  state_ = STATE_UNBOUND;
+  port_->DestroyConnection(ext_addr_);
+}
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/base/turnport.h b/media/webrtc/trunk/webrtc/p2p/base/turnport.h
index 5bb7558598..4d83806a37 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/turnport.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/turnport.h
@@ -16,9 +16,10 @@
 #include <set>
 #include <string>
 
+#include "webrtc/base/asyncinvoker.h"
+#include "webrtc/base/asyncpacketsocket.h"
 #include "webrtc/p2p/base/port.h"
 #include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/base/asyncpacketsocket.h"
 
 namespace rtc {
 class AsyncResolver;
@@ -33,6 +34,12 @@ class TurnEntry;
 
 class TurnPort : public Port {
  public:
+  enum PortState {
+    STATE_CONNECTING,    // Initial state, cannot send any packets.
+    STATE_CONNECTED,     // Socket connected, ready to send stun requests.
+    STATE_READY,         // Received allocate success, can send any packets.
+    STATE_DISCONNECTED,  // TCP connection died, cannot send any packets.
+  };
   static TurnPort* Create(rtc::Thread* thread,
                           rtc::PacketSocketFactory* factory,
                           rtc::Network* network,
@@ -51,8 +58,8 @@ class TurnPort : public Port {
                           rtc::PacketSocketFactory* factory,
                           rtc::Network* network,
                           const rtc::IPAddress& ip,
-                          uint16 min_port,
-                          uint16 max_port,
+                          uint16_t min_port,
+                          uint16_t max_port,
                           const std::string& username,  // ice username.
                           const std::string& password,  // ice password.
                           const ProtocolAddress& server_address,
@@ -70,7 +77,10 @@ class TurnPort : public Port {
   // Returns an empty address if the local address has not been assigned.
   rtc::SocketAddress GetLocalAddress() const;
 
-  bool connected() const { return connected_; }
+  bool ready() const { return state_ == STATE_READY; }
+  bool connected() const {
+    return state_ == STATE_READY || state_ == STATE_CONNECTED;
+  }
   const RelayCredentials& credentials() const { return credentials_; }
 
   virtual void PrepareAddress();
@@ -96,7 +106,13 @@ class TurnPort : public Port {
                             const rtc::SocketAddress& remote_addr,
                             const rtc::PacketTime& packet_time);
 
+  virtual void OnSentPacket(rtc::AsyncPacketSocket* socket,
+                            const rtc::SentPacket& sent_packet);
   virtual void OnReadyToSend(rtc::AsyncPacketSocket* socket);
+  virtual bool SupportsProtocol(const std::string& protocol) const {
+    // Turn port only connects to UDP candidates.
+    return protocol == UDP_PROTOCOL_NAME;
+  }
 
   void OnSocketConnect(rtc::AsyncPacketSocket* socket);
   void OnSocketClose(rtc::AsyncPacketSocket* socket, int error);
@@ -113,6 +129,9 @@ class TurnPort : public Port {
     return socket_;
   }
 
+  // For testing only.
+  rtc::AsyncInvoker* invoker() { return &invoker_; }
+
   // Signal with resolved server address.
   // Parameters are port, server address and resolved server address.
   // This signal will be sent only if server address is resolved successfully.
@@ -120,9 +139,18 @@ class TurnPort : public Port {
                    const rtc::SocketAddress&,
                    const rtc::SocketAddress&> SignalResolvedServerAddress;
 
-  // This signal is only for testing purpose.
+  // All public methods/signals below are for testing only.
+  sigslot::signal2<TurnPort*, int> SignalTurnRefreshResult;
   sigslot::signal3<TurnPort*, const rtc::SocketAddress&, int>
       SignalCreatePermissionResult;
+  void FlushRequests(int msg_type) { request_manager_.Flush(msg_type); }
+  bool HasRequests() { return !request_manager_.empty(); }
+  void set_credentials(RelayCredentials& credentials) {
+    credentials_ = credentials;
+  }
+  // Finds the turn entry with |address| and sets its channel id.
+  // Returns true if the entry is found.
+  bool SetEntryChannelId(const rtc::SocketAddress& address, int channel_id);
 
  protected:
   TurnPort(rtc::Thread* thread,
@@ -140,8 +168,8 @@ class TurnPort : public Port {
            rtc::PacketSocketFactory* factory,
            rtc::Network* network,
            const rtc::IPAddress& ip,
-           uint16 min_port,
-           uint16 max_port,
+           uint16_t min_port,
+           uint16_t max_port,
            const std::string& username,
            const std::string& password,
            const ProtocolAddress& server_address,
@@ -151,9 +179,10 @@ class TurnPort : public Port {
 
  private:
   enum {
-    MSG_ERROR = MSG_FIRST_AVAILABLE,
+    MSG_ALLOCATE_ERROR = MSG_FIRST_AVAILABLE,
     MSG_ALLOCATE_MISMATCH,
-    MSG_TRY_ALTERNATE_SERVER
+    MSG_TRY_ALTERNATE_SERVER,
+    MSG_REFRESH_ERROR
   };
 
   typedef std::list<TurnEntry*> EntryList;
@@ -172,6 +201,9 @@ class TurnPort : public Port {
     }
   }
 
+  // Shuts down the turn port, usually because of some fatal errors.
+  void Close();
+  void OnTurnRefreshError();
   bool SetAlternateServer(const rtc::SocketAddress& address);
   void ResolveTurnAddress(const rtc::SocketAddress& address);
   void OnResolveResult(rtc::AsyncResolverInterface* resolver);
@@ -204,10 +236,20 @@ class TurnPort : public Port {
   bool HasPermission(const rtc::IPAddress& ipaddr) const;
   TurnEntry* FindEntry(const rtc::SocketAddress& address) const;
   TurnEntry* FindEntry(int channel_id) const;
-  TurnEntry* CreateEntry(const rtc::SocketAddress& address);
-  void DestroyEntry(const rtc::SocketAddress& address);
+  bool EntryExists(TurnEntry* e);
+  void CreateOrRefreshEntry(const rtc::SocketAddress& address);
+  void DestroyEntry(TurnEntry* entry);
+  // Destroys the entry only if |timestamp| matches the destruction timestamp
+  // in |entry|.
+  void DestroyEntryIfNotCancelled(TurnEntry* entry, uint32_t timestamp);
+  void ScheduleEntryDestruction(TurnEntry* entry);
+  void CancelEntryDestruction(TurnEntry* entry);
   void OnConnectionDestroyed(Connection* conn);
 
+  // Destroys the connection with remote address |address|. Returns true if
+  // a connection is found and destroyed.
+  bool DestroyConnection(const rtc::SocketAddress& address);
+
   ProtocolAddress server_address_;
   RelayCredentials credentials_;
   AttemptedServerSet attempted_server_addresses_;
@@ -225,7 +267,7 @@ class TurnPort : public Port {
   int next_channel_number_;
   EntryList entries_;
 
-  bool connected_;
+  PortState state_;
   // By default the value will be set to 0. This value will be used in
   // calculating the candidate priority.
   int server_priority_;
@@ -233,6 +275,8 @@ class TurnPort : public Port {
   // The number of retries made due to allocate mismatch error.
   size_t allocate_mismatch_retries_;
 
+  rtc::AsyncInvoker invoker_;
+
   friend class TurnEntry;
   friend class TurnAllocateRequest;
   friend class TurnRefreshRequest;
diff --git a/media/webrtc/trunk/webrtc/p2p/base/turnport_unittest.cc b/media/webrtc/trunk/webrtc/p2p/base/turnport_unittest.cc
index 3172ba252f..916162575f 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/turnport_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/turnport_unittest.cc
@@ -13,6 +13,7 @@
 
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/tcpport.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/turnport.h"
@@ -100,6 +101,24 @@ class TurnPortTestVirtualSocketServer : public rtc::VirtualSocketServer {
   using rtc::VirtualSocketServer::LookupBinding;
 };
 
+class TestConnectionWrapper : public sigslot::has_slots<> {
+ public:
+  TestConnectionWrapper(Connection* conn) : connection_(conn) {
+    conn->SignalDestroyed.connect(
+        this, &TestConnectionWrapper::OnConnectionDestroyed);
+  }
+
+  Connection* connection() { return connection_; }
+
+ private:
+  void OnConnectionDestroyed(Connection* conn) {
+    ASSERT_TRUE(conn == connection_);
+    connection_ = nullptr;
+  }
+
+  Connection* connection_;
+};
+
 class TurnPortTest : public testing::Test,
                      public sigslot::has_slots<>,
                      public rtc::MessageHandler {
@@ -154,12 +173,15 @@ class TurnPortTest : public testing::Test,
                             bool /*port_muxed*/) {
     turn_unknown_address_ = true;
   }
-  void OnTurnCreatePermissionResult(TurnPort* port, const SocketAddress& addr,
-                                     int code) {
+  void OnTurnCreatePermissionResult(TurnPort* port,
+                                    const SocketAddress& addr,
+                                    int code) {
     // Ignoring the address.
-    if (code == 0) {
-      turn_create_permission_success_ = true;
-    }
+    turn_create_permission_success_ = (code == 0);
+  }
+
+  void OnTurnRefreshResult(TurnPort* port, int code) {
+    turn_refresh_success_ = (code == 0);
   }
   void OnTurnReadPacket(Connection* conn, const char* data, size_t size,
                         const rtc::PacketTime& packet_time) {
@@ -172,6 +194,7 @@ class TurnPortTest : public testing::Test,
                        const rtc::PacketTime& packet_time) {
     udp_packets_.push_back(rtc::Buffer(data, size));
   }
+  void OnConnectionDestroyed(Connection* conn) { connection_destroyed_ = true; }
   void OnSocketReadPacket(rtc::AsyncPacketSocket* socket,
                           const char* data, size_t size,
                           const rtc::SocketAddress& remote_addr,
@@ -201,11 +224,7 @@ class TurnPortTest : public testing::Test,
                                  kIceUfrag1, kIcePwd1,
                                  server_address, credentials, 0,
                                  std::string()));
-    // Set ICE protocol type to ICEPROTO_RFC5245, as port by default will be
-    // in Hybrid mode. Protocol type is necessary to send correct type STUN ping
-    // messages.
     // This TURN port will be the controlling.
-    turn_port_->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
     turn_port_->SetIceRole(cricket::ICEROLE_CONTROLLING);
     ConnectSignals();
   }
@@ -223,11 +242,7 @@ class TurnPortTest : public testing::Test,
                                  kIceUfrag1, kIcePwd1,
                                  server_address, credentials, 0,
                                  origin));
-    // Set ICE protocol type to ICEPROTO_RFC5245, as port by default will be
-    // in Hybrid mode. Protocol type is necessary to send correct type STUN ping
-    // messages.
     // This TURN port will be the controlling.
-    turn_port_->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
     turn_port_->SetIceRole(cricket::ICEROLE_CONTROLLING);
     ConnectSignals();
   }
@@ -249,11 +264,7 @@ class TurnPortTest : public testing::Test,
     turn_port_.reset(cricket::TurnPort::Create(
         main_, &socket_factory_, &network_, socket_.get(),
         kIceUfrag1, kIcePwd1, server_address, credentials, 0, std::string()));
-    // Set ICE protocol type to ICEPROTO_RFC5245, as port by default will be
-    // in Hybrid mode. Protocol type is necessary to send correct type STUN ping
-    // messages.
     // This TURN port will be the controlling.
-    turn_port_->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
     turn_port_->SetIceRole(cricket::ICEROLE_CONTROLLING);
     ConnectSignals();
   }
@@ -267,20 +278,42 @@ class TurnPortTest : public testing::Test,
         &TurnPortTest::OnTurnUnknownAddress);
     turn_port_->SignalCreatePermissionResult.connect(this,
         &TurnPortTest::OnTurnCreatePermissionResult);
+    turn_port_->SignalTurnRefreshResult.connect(
+        this, &TurnPortTest::OnTurnRefreshResult);
   }
-  void CreateUdpPort() {
+  void ConnectConnectionDestroyedSignal(Connection* conn) {
+    conn->SignalDestroyed.connect(this, &TurnPortTest::OnConnectionDestroyed);
+  }
+
+  void CreateUdpPort() { CreateUdpPort(kLocalAddr2); }
+
+  void CreateUdpPort(const SocketAddress& address) {
     udp_port_.reset(UDPPort::Create(main_, &socket_factory_, &network_,
-                                    kLocalAddr2.ipaddr(), 0, 0,
-                                    kIceUfrag2, kIcePwd2,
-                                    std::string()));
-    // Set protocol type to RFC5245, as turn port is also in same mode.
+                                    address.ipaddr(), 0, 0, kIceUfrag2,
+                                    kIcePwd2, std::string(), false));
     // UDP port will be controlled.
-    udp_port_->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
     udp_port_->SetIceRole(cricket::ICEROLE_CONTROLLED);
     udp_port_->SignalPortComplete.connect(
         this, &TurnPortTest::OnUdpPortComplete);
   }
 
+  void PrepareTurnAndUdpPorts() {
+    // turn_port_ should have been created.
+    ASSERT_TRUE(turn_port_ != nullptr);
+    turn_port_->PrepareAddress();
+    ASSERT_TRUE_WAIT(turn_ready_, kTimeout);
+
+    CreateUdpPort();
+    udp_port_->PrepareAddress();
+    ASSERT_TRUE_WAIT(udp_ready_, kTimeout);
+  }
+
+  bool CheckConnectionDestroyed() {
+    turn_port_->FlushRequests(cricket::kAllRequests);
+    rtc::Thread::Current()->ProcessMessages(50);
+    return connection_destroyed_;
+  }
+
   void TestTurnAlternateServer(cricket::ProtocolType protocol_type) {
     std::vector<rtc::SocketAddress> redirect_addresses;
     redirect_addresses.push_back(kTurnAlternateIntAddr);
@@ -364,12 +397,7 @@ class TurnPortTest : public testing::Test,
 
   void TestTurnConnection() {
     // Create ports and prepare addresses.
-    ASSERT_TRUE(turn_port_ != NULL);
-    turn_port_->PrepareAddress();
-    ASSERT_TRUE_WAIT(turn_ready_, kTimeout);
-    CreateUdpPort();
-    udp_port_->PrepareAddress();
-    ASSERT_TRUE_WAIT(udp_ready_, kTimeout);
+    PrepareTurnAndUdpPorts();
 
     // Send ping from UDP to TURN.
     Connection* conn1 = udp_port_->CreateConnection(
@@ -378,7 +406,7 @@ class TurnPortTest : public testing::Test,
     conn1->Ping(0);
     WAIT(!turn_unknown_address_, kTimeout);
     EXPECT_FALSE(turn_unknown_address_);
-    EXPECT_EQ(Connection::STATE_READ_INIT, conn1->read_state());
+    EXPECT_FALSE(conn1->receiving());
     EXPECT_EQ(Connection::STATE_WRITE_INIT, conn1->write_state());
 
     // Send ping from TURN to UDP.
@@ -389,22 +417,56 @@ class TurnPortTest : public testing::Test,
     conn2->Ping(0);
 
     EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, conn2->write_state(), kTimeout);
-    EXPECT_EQ(Connection::STATE_READABLE, conn1->read_state());
-    EXPECT_EQ(Connection::STATE_READ_INIT, conn2->read_state());
+    EXPECT_TRUE(conn1->receiving());
+    EXPECT_TRUE(conn2->receiving());
     EXPECT_EQ(Connection::STATE_WRITE_INIT, conn1->write_state());
 
     // Send another ping from UDP to TURN.
     conn1->Ping(0);
     EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, conn1->write_state(), kTimeout);
-    EXPECT_EQ(Connection::STATE_READABLE, conn2->read_state());
+    EXPECT_TRUE(conn2->receiving());
+  }
+
+  void TestDestroyTurnConnection() {
+    PrepareTurnAndUdpPorts();
+
+    // Create connections on both ends.
+    Connection* conn1 = udp_port_->CreateConnection(turn_port_->Candidates()[0],
+                                                    Port::ORIGIN_MESSAGE);
+    Connection* conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                                                     Port::ORIGIN_MESSAGE);
+    ASSERT_TRUE(conn2 != NULL);
+    ASSERT_TRUE_WAIT(turn_create_permission_success_, kTimeout);
+    // Make sure turn connection can receive.
+    conn1->Ping(0);
+    EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, conn1->write_state(), kTimeout);
+    EXPECT_FALSE(turn_unknown_address_);
+
+    // Destroy the connection on the turn port. The TurnEntry is still
+    // there. So the turn port gets ping from unknown address if it is pinged.
+    conn2->Destroy();
+    conn1->Ping(0);
+    EXPECT_TRUE_WAIT(turn_unknown_address_, kTimeout);
+
+    // Flush all requests in the invoker to destroy the TurnEntry.
+    // Now the turn port cannot receive the ping.
+    turn_unknown_address_ = false;
+    turn_port_->invoker()->Flush(rtc::Thread::Current());
+    conn1->Ping(0);
+    rtc::Thread::Current()->ProcessMessages(500);
+    EXPECT_FALSE(turn_unknown_address_);
+
+    // If the connection is created again, it will start to receive pings.
+    conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                                         Port::ORIGIN_MESSAGE);
+    conn1->Ping(0);
+    EXPECT_TRUE_WAIT(conn2->receiving(), kTimeout);
+    EXPECT_FALSE(turn_unknown_address_);
   }
 
   void TestTurnSendData() {
-    turn_port_->PrepareAddress();
-    EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
-    CreateUdpPort();
-    udp_port_->PrepareAddress();
-    EXPECT_TRUE_WAIT(udp_ready_, kTimeout);
+    PrepareTurnAndUdpPorts();
+
     // Create connections and send pings.
     Connection* conn1 = turn_port_->CreateConnection(
         udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE);
@@ -460,6 +522,8 @@ class TurnPortTest : public testing::Test,
   bool turn_create_permission_success_;
   bool udp_ready_;
   bool test_finish_;
+  bool turn_refresh_success_ = false;
+  bool connection_destroyed_ = false;
   std::vector<rtc::Buffer> turn_packets_;
   std::vector<rtc::Buffer> udp_packets_;
   rtc::PacketOptions options;
@@ -627,6 +691,45 @@ TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) {
   EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
 }
 
+TEST_F(TurnPortTest, TestRefreshRequestGetsErrorResponse) {
+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+  PrepareTurnAndUdpPorts();
+  turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                               Port::ORIGIN_MESSAGE);
+  // Set bad credentials.
+  cricket::RelayCredentials bad_credentials("bad_user", "bad_pwd");
+  turn_port_->set_credentials(bad_credentials);
+  turn_refresh_success_ = false;
+  // This sends out the first RefreshRequest with correct credentials.
+  // When this succeeds, it will schedule a new RefreshRequest with the bad
+  // credential.
+  turn_port_->FlushRequests(cricket::TURN_REFRESH_REQUEST);
+  EXPECT_TRUE_WAIT(turn_refresh_success_, kTimeout);
+  // Flush it again, it will receive a bad response.
+  turn_port_->FlushRequests(cricket::TURN_REFRESH_REQUEST);
+  EXPECT_TRUE_WAIT(!turn_refresh_success_, kTimeout);
+  EXPECT_TRUE_WAIT(!turn_port_->connected(), kTimeout);
+  EXPECT_TRUE_WAIT(turn_port_->connections().empty(), kTimeout);
+  EXPECT_FALSE(turn_port_->HasRequests());
+}
+
+// Test that CreateConnection will return null if port becomes disconnected.
+TEST_F(TurnPortTest, TestCreateConnectionWhenSocketClosed) {
+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
+  PrepareTurnAndUdpPorts();
+  // Create a connection.
+  Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                                                   Port::ORIGIN_MESSAGE);
+  ASSERT_TRUE(conn1 != NULL);
+
+  // Close the socket and create a connection again.
+  turn_port_->OnSocketClose(turn_port_->socket(), 1);
+  conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                                       Port::ORIGIN_MESSAGE);
+  ASSERT_TRUE(conn1 == NULL);
+}
+
 // Test try-alternate-server feature.
 TEST_F(TurnPortTest, TestTurnAlternateServerUDP) {
   TestTurnAlternateServer(cricket::PROTO_UDP);
@@ -686,6 +789,20 @@ TEST_F(TurnPortTest, TestTurnTcpConnection) {
   TestTurnConnection();
 }
 
+// Test that if a connection on a TURN port is destroyed, the TURN port can
+// still receive ping on that connection as if it is from an unknown address.
+// If the connection is created again, it will be used to receive ping.
+TEST_F(TurnPortTest, TestDestroyTurnConnection) {
+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+  TestDestroyTurnConnection();
+}
+
+// Similar to above, except that this test will use the shared socket.
+TEST_F(TurnPortTest, TestDestroyTurnConnectionUsingSharedSocket) {
+  CreateSharedTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+  TestDestroyTurnConnection();
+}
+
 // Test that we fail to create a connection when we want to use TLS over TCP.
 // This test should be removed once we have TLS support.
 TEST_F(TurnPortTest, TestTurnTlsTcpConnectionFails) {
@@ -707,11 +824,61 @@ TEST_F(TurnPortTest, TestTurnConnectionUsingOTUNonce) {
   TestTurnConnection();
 }
 
+// Test that CreatePermissionRequest will be scheduled after the success
+// of the first create permission request and the request will get an
+// ErrorResponse if the ufrag and pwd are incorrect.
+TEST_F(TurnPortTest, TestRefreshCreatePermissionRequest) {
+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+  PrepareTurnAndUdpPorts();
+
+  Connection* conn = turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                                                  Port::ORIGIN_MESSAGE);
+  ConnectConnectionDestroyedSignal(conn);
+  ASSERT_TRUE(conn != NULL);
+  ASSERT_TRUE_WAIT(turn_create_permission_success_, kTimeout);
+  turn_create_permission_success_ = false;
+  // A create-permission-request should be pending.
+  // After the next create-permission-response is received, it will schedule
+  // another request with bad_ufrag and bad_pwd.
+  cricket::RelayCredentials bad_credentials("bad_user", "bad_pwd");
+  turn_port_->set_credentials(bad_credentials);
+  turn_port_->FlushRequests(cricket::kAllRequests);
+  ASSERT_TRUE_WAIT(turn_create_permission_success_, kTimeout);
+  // Flush the requests again; the create-permission-request will fail.
+  turn_port_->FlushRequests(cricket::kAllRequests);
+  EXPECT_TRUE_WAIT(!turn_create_permission_success_, kTimeout);
+  EXPECT_TRUE_WAIT(connection_destroyed_, kTimeout);
+}
+
+TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) {
+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+  PrepareTurnAndUdpPorts();
+  Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
+                                                   Port::ORIGIN_MESSAGE);
+  ASSERT_TRUE(conn1 != nullptr);
+  Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0],
+                                                  Port::ORIGIN_MESSAGE);
+  ASSERT_TRUE(conn2 != nullptr);
+  ConnectConnectionDestroyedSignal(conn1);
+  conn1->Ping(0);
+  ASSERT_TRUE_WAIT(conn1->writable(), kTimeout);
+
+  std::string data = "ABC";
+  conn1->Send(data.data(), data.length(), options);
+  bool success =
+      turn_port_->SetEntryChannelId(udp_port_->Candidates()[0].address(), -1);
+  ASSERT_TRUE(success);
+  // Next time when the binding request is sent, it will get an ErrorResponse.
+  EXPECT_TRUE_WAIT(CheckConnectionDestroyed(), kTimeout);
+}
+
 // Do a TURN allocation, establish a UDP connection, and send some data.
 TEST_F(TurnPortTest, TestTurnSendDataTurnUdpToUdp) {
   // Create ports and prepare addresses.
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
   TestTurnSendData();
+  EXPECT_EQ(cricket::UDP_PROTOCOL_NAME,
+            turn_port_->Candidates()[0].relay_protocol());
 }
 
 // Do a TURN allocation, establish a TCP connection, and send some data.
@@ -720,6 +887,8 @@ TEST_F(TurnPortTest, TestTurnSendDataTurnTcpToUdp) {
   // Create ports and prepare addresses.
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
   TestTurnSendData();
+  EXPECT_EQ(cricket::TCP_PROTOCOL_NAME,
+            turn_port_->Candidates()[0].relay_protocol());
 }
 
 // Test TURN fails to make a connection from IPv6 address to a server which has
@@ -759,6 +928,29 @@ TEST_F(TurnPortTest, TestOriginHeader) {
   EXPECT_EQ(kTestOrigin, turn_server_.FindAllocation(local_address)->origin());
 }
 
+// Test that a CreatePermission failure will result in the connection being
+// destroyed.
+TEST_F(TurnPortTest, TestConnectionDestroyedOnCreatePermissionFailure) {
+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+  turn_server_.server()->set_reject_private_addresses(true);
+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
+  turn_port_->PrepareAddress();
+  ASSERT_TRUE_WAIT(turn_ready_, kTimeout);
+
+  CreateUdpPort(SocketAddress("10.0.0.10", 0));
+  udp_port_->PrepareAddress();
+  ASSERT_TRUE_WAIT(udp_ready_, kTimeout);
+  // Create a connection.
+  TestConnectionWrapper conn(turn_port_->CreateConnection(
+      udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE));
+  ASSERT_TRUE(conn.connection() != nullptr);
+
+  // Asynchronously, CreatePermission request should be sent and fail, closing
+  // the connection.
+  EXPECT_TRUE_WAIT(conn.connection() == nullptr, kTimeout);
+  EXPECT_FALSE(turn_create_permission_success_);
+}
+
 // Test that a TURN allocation is released when the port is closed.
 TEST_F(TurnPortTest, TestTurnReleaseAllocation) {
   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
@@ -785,15 +977,19 @@ TEST_F(TurnPortTest, DISABLED_TestTurnTCPReleaseAllocation) {
 // This test verifies any FD's are not leaked after TurnPort is destroyed.
 // https://code.google.com/p/webrtc/issues/detail?id=2651
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+// 1 second is not always enough for getaddrinfo().
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5191
+static const unsigned int kResolverTimeout = 10000;
+
 TEST_F(TurnPortTest, TestResolverShutdown) {
   turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, cricket::PROTO_UDP);
   int last_fd_count = GetFDCount();
   // Need to supply unresolved address to kick off resolver.
   CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword,
                  cricket::ProtocolAddress(rtc::SocketAddress(
-                    "stun.l.google.com", 3478), cricket::PROTO_UDP));
+                    "www.google.invalid", 3478), cricket::PROTO_UDP));
   turn_port_->PrepareAddress();
-  ASSERT_TRUE_WAIT(turn_error_, kTimeout);
+  ASSERT_TRUE_WAIT(turn_error_, kResolverTimeout);
   EXPECT_TRUE(turn_port_->Candidates().empty());
   turn_port_.reset();
   rtc::Thread::Current()->Post(this, MSG_TESTFINISH);
diff --git a/media/webrtc/trunk/webrtc/p2p/base/turnserver.cc b/media/webrtc/trunk/webrtc/p2p/base/turnserver.cc
index 7d82d55c5d..1502cdd52e 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/turnserver.cc
+++ b/media/webrtc/trunk/webrtc/p2p/base/turnserver.cc
@@ -40,7 +40,7 @@ static const size_t kNonceSize = 40;
 static const size_t TURN_CHANNEL_HEADER_SIZE = 4U;
 
 // TODO(mallinath) - Move these to a common place.
-inline bool IsTurnChannelData(uint16 msg_type) {
+inline bool IsTurnChannelData(uint16_t msg_type) {
   // The first two bits of a channel data message are 0b01.
   return ((msg_type & 0xC000) == 0x4000);
 }
@@ -200,7 +200,7 @@ void TurnServer::OnInternalPacket(rtc::AsyncPacketSocket* socket,
   InternalSocketMap::iterator iter = server_sockets_.find(socket);
   ASSERT(iter != server_sockets_.end());
   TurnServerConnection conn(addr, iter->second, socket);
-  uint16 msg_type = rtc::GetBE16(data);
+  uint16_t msg_type = rtc::GetBE16(data);
   if (!IsTurnChannelData(msg_type)) {
     // This is a STUN message.
     HandleStunMessage(&conn, data, size);
@@ -394,7 +394,7 @@ void TurnServer::HandleAllocateRequest(TurnServerConnection* conn,
 
 std::string TurnServer::GenerateNonce() const {
   // Generate a nonce of the form hex(now + HMAC-MD5(nonce_key_, now))
-  uint32 now = rtc::Time();
+  uint32_t now = rtc::Time();
   std::string input(reinterpret_cast<const char*>(&now), sizeof(now));
   std::string nonce = rtc::hex_encode(input.c_str(), input.size());
   nonce += rtc::ComputeHmac(rtc::DIGEST_MD5, nonce_key_, input);
@@ -409,7 +409,7 @@ bool TurnServer::ValidateNonce(const std::string& nonce) const {
   }
 
   // Decode the timestamp.
-  uint32 then;
+  uint32_t then;
   char* p = reinterpret_cast<char*>(&then);
   size_t len = rtc::hex_decode(p, sizeof(then),
       nonce.substr(0, sizeof(then) * 2));
@@ -698,6 +698,12 @@ void TurnServerAllocation::HandleCreatePermissionRequest(
     return;
   }
 
+  if (server_->reject_private_addresses_ &&
+      rtc::IPIsPrivate(peer_attr->GetAddress().ipaddr())) {
+    SendErrorResponse(msg, STUN_ERROR_FORBIDDEN, STUN_ERROR_REASON_FORBIDDEN);
+    return;
+  }
+
   // Add this permission.
   AddPermission(peer_attr->GetAddress().ipaddr());
 
@@ -761,7 +767,7 @@ void TurnServerAllocation::HandleChannelBindRequest(const TurnMessage* msg) {
 
 void TurnServerAllocation::HandleChannelData(const char* data, size_t size) {
   // Extract the channel number from the data.
-  uint16 channel_id = rtc::GetBE16(data);
+  uint16_t channel_id = rtc::GetBE16(data);
   Channel* channel = FindChannel(channel_id);
   if (channel) {
     // Send the data to the peer address.
@@ -784,7 +790,7 @@ void TurnServerAllocation::OnExternalPacket(
     // There is a channel bound to this address. Send as a channel message.
     rtc::ByteBuffer buf;
     buf.WriteUInt16(channel->id());
-    buf.WriteUInt16(static_cast<uint16>(size));
+    buf.WriteUInt16(static_cast<uint16_t>(size));
     buf.WriteBytes(data, size);
     server_->Send(&conn_, buf);
   } else if (HasPermission(addr.ipaddr())) {
@@ -806,7 +812,7 @@ void TurnServerAllocation::OnExternalPacket(
 
 int TurnServerAllocation::ComputeLifetime(const TurnMessage* msg) {
   // Return the smaller of our default lifetime and the requested lifetime.
-  uint32 lifetime = kDefaultAllocationTimeout / 1000;  // convert to seconds
+  uint32_t lifetime = kDefaultAllocationTimeout / 1000;  // convert to seconds
   const StunUInt32Attribute* lifetime_attr = msg->GetUInt32(STUN_ATTR_LIFETIME);
   if (lifetime_attr && lifetime_attr->value() < lifetime) {
     lifetime = lifetime_attr->value();
diff --git a/media/webrtc/trunk/webrtc/p2p/base/turnserver.h b/media/webrtc/trunk/webrtc/p2p/base/turnserver.h
index d3bd77a866..113bd4c462 100644
--- a/media/webrtc/trunk/webrtc/p2p/base/turnserver.h
+++ b/media/webrtc/trunk/webrtc/p2p/base/turnserver.h
@@ -183,6 +183,11 @@ class TurnServer : public sigslot::has_slots<> {
 
   void set_enable_otu_nonce(bool enable) { enable_otu_nonce_ = enable; }
 
+  // If set to true, reject CreatePermission requests to RFC1918 addresses.
+  void set_reject_private_addresses(bool filter) {
+    reject_private_addresses_ = filter;
+  }
+
   // Starts listening for packets from internal clients.
   void AddInternalSocket(rtc::AsyncPacketSocket* socket,
                          ProtocolType proto);
@@ -255,6 +260,7 @@ class TurnServer : public sigslot::has_slots<> {
   // otu - one-time-use. Server will respond with 438 if it's
   // sees the same nonce in next transaction.
   bool enable_otu_nonce_;
+  bool reject_private_addresses_ = false;
 
   InternalSocketMap server_sockets_;
   ServerSocketMap server_listen_sockets_;
diff --git a/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.cc b/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.cc
index 5c322da59e..e45d2c8f0f 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.cc
+++ b/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/p2p/client/basicportallocator.h"
 
+#include <algorithm>
 #include <string>
 #include <vector>
 
@@ -21,6 +22,7 @@
 #include "webrtc/p2p/base/tcpport.h"
 #include "webrtc/p2p/base/turnport.h"
 #include "webrtc/p2p/base/udpport.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
@@ -58,109 +60,27 @@ int ShakeDelay() {
 }  // namespace
 
 namespace cricket {
-
-const uint32 DISABLE_ALL_PHASES =
-  PORTALLOCATOR_DISABLE_UDP
-  | PORTALLOCATOR_DISABLE_TCP
-  | PORTALLOCATOR_DISABLE_STUN
-  | PORTALLOCATOR_DISABLE_RELAY;
-
-// Performs the allocation of ports, in a sequenced (timed) manner, for a given
-// network and IP address.
-class AllocationSequence : public rtc::MessageHandler,
-                           public sigslot::has_slots<> {
- public:
-  enum State {
-    kInit,       // Initial state.
-    kRunning,    // Started allocating ports.
-    kStopped,    // Stopped from running.
-    kCompleted,  // All ports are allocated.
-
-    // kInit --> kRunning --> {kCompleted|kStopped}
-  };
-
-  AllocationSequence(BasicPortAllocatorSession* session,
-                     rtc::Network* network,
-                     PortConfiguration* config,
-                     uint32 flags);
-  ~AllocationSequence();
-  bool Init();
-  void Clear();
-
-  State state() const { return state_; }
-
-  // Disables the phases for a new sequence that this one already covers for an
-  // equivalent network setup.
-  void DisableEquivalentPhases(rtc::Network* network,
-      PortConfiguration* config, uint32* flags);
-
-  // Starts and stops the sequence.  When started, it will continue allocating
-  // new ports on its own timed schedule.
-  void Start();
-  void Stop();
-
-  // MessageHandler
-  void OnMessage(rtc::Message* msg);
-
-  void EnableProtocol(ProtocolType proto);
-  bool ProtocolEnabled(ProtocolType proto) const;
-
-  // Signal from AllocationSequence, when it's done with allocating ports.
-  // This signal is useful, when port allocation fails which doesn't result
-  // in any candidates. Using this signal BasicPortAllocatorSession can send
-  // its candidate discovery conclusion signal. Without this signal,
-  // BasicPortAllocatorSession doesn't have any event to trigger signal. This
-  // can also be achieved by starting timer in BPAS.
-  sigslot::signal1<AllocationSequence*> SignalPortAllocationComplete;
-
- private:
-  typedef std::vector<ProtocolType> ProtocolList;
-
-  bool IsFlagSet(uint32 flag) {
-    return ((flags_ & flag) != 0);
-  }
-  void CreateUDPPorts();
-  void CreateTCPPorts();
-  void CreateStunPorts();
-  void CreateRelayPorts();
-  void CreateGturnPort(const RelayServerConfig& config);
-  void CreateTurnPort(const RelayServerConfig& config);
-
-  void OnReadPacket(rtc::AsyncPacketSocket* socket,
-                    const char* data, size_t size,
-                    const rtc::SocketAddress& remote_addr,
-                    const rtc::PacketTime& packet_time);
-
-  void OnPortDestroyed(PortInterface* port);
-
-  BasicPortAllocatorSession* session_;
-  rtc::Network* network_;
-  rtc::IPAddress ip_;
-  PortConfiguration* config_;
-  State state_;
-  uint32 flags_;
-  ProtocolList protocols_;
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> udp_socket_;
-  // There will be only one udp port per AllocationSequence.
-  UDPPort* udp_port_;
-  std::vector<TurnPort*> turn_ports_;
-  int phase_;
-};
+const uint32_t DISABLE_ALL_PHASES =
+    PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_TCP |
+    PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY;
 
 // BasicPortAllocator
 BasicPortAllocator::BasicPortAllocator(
     rtc::NetworkManager* network_manager,
     rtc::PacketSocketFactory* socket_factory)
     : network_manager_(network_manager),
-      socket_factory_(socket_factory) {
-  ASSERT(socket_factory_ != NULL);
+      socket_factory_(socket_factory),
+      stun_servers_() {
+  ASSERT(network_manager_ != nullptr);
+  ASSERT(socket_factory_ != nullptr);
   Construct();
 }
 
-BasicPortAllocator::BasicPortAllocator(
-    rtc::NetworkManager* network_manager)
+BasicPortAllocator::BasicPortAllocator(rtc::NetworkManager* network_manager)
     : network_manager_(network_manager),
-      socket_factory_(NULL) {
+      socket_factory_(nullptr),
+      stun_servers_() {
+  ASSERT(network_manager_ != nullptr);
   Construct();
 }
 
@@ -186,15 +106,19 @@ BasicPortAllocator::BasicPortAllocator(
       stun_servers_(stun_servers) {
 
   RelayServerConfig config(RELAY_GTURN);
-  if (!relay_address_udp.IsNil())
+  if (!relay_address_udp.IsNil()) {
     config.ports.push_back(ProtocolAddress(relay_address_udp, PROTO_UDP));
-  if (!relay_address_tcp.IsNil())
+  }
+  if (!relay_address_tcp.IsNil()) {
     config.ports.push_back(ProtocolAddress(relay_address_tcp, PROTO_TCP));
-  if (!relay_address_ssl.IsNil())
+  }
+  if (!relay_address_ssl.IsNil()) {
     config.ports.push_back(ProtocolAddress(relay_address_ssl, PROTO_SSLTCP));
+  }
 
-  if (!config.ports.empty())
-    AddRelay(config);
+  if (!config.ports.empty()) {
+    AddTurnServer(config);
+  }
 
   Construct();
 }
@@ -206,7 +130,7 @@ void BasicPortAllocator::Construct() {
 BasicPortAllocator::~BasicPortAllocator() {
 }
 
-PortAllocatorSession *BasicPortAllocator::CreateSessionInternal(
+PortAllocatorSession* BasicPortAllocator::CreateSessionInternal(
     const std::string& content_name, int component,
     const std::string& ice_ufrag, const std::string& ice_pwd) {
   return new BasicPortAllocatorSession(
@@ -239,7 +163,7 @@ BasicPortAllocatorSession::~BasicPortAllocatorSession() {
   if (network_thread_ != NULL)
     network_thread_->Clear(this);
 
-  for (uint32 i = 0; i < sequences_.size(); ++i) {
+  for (uint32_t i = 0; i < sequences_.size(); ++i) {
     // AllocationSequence should clear it's map entry for turn ports before
     // ports are destroyed.
     sequences_[i]->Clear();
@@ -249,10 +173,10 @@ BasicPortAllocatorSession::~BasicPortAllocatorSession() {
   for (it = ports_.begin(); it != ports_.end(); it++)
     delete it->port();
 
-  for (uint32 i = 0; i < configs_.size(); ++i)
+  for (uint32_t i = 0; i < configs_.size(); ++i)
     delete configs_[i];
 
-  for (uint32 i = 0; i < sequences_.size(); ++i)
+  for (uint32_t i = 0; i < sequences_.size(); ++i)
     delete sequences_[i];
 }
 
@@ -274,10 +198,14 @@ void BasicPortAllocatorSession::StartGettingPorts() {
 void BasicPortAllocatorSession::StopGettingPorts() {
   ASSERT(rtc::Thread::Current() == network_thread_);
   running_ = false;
-  network_thread_->Clear(this, MSG_ALLOCATE);
-  for (uint32 i = 0; i < sequences_.size(); ++i)
-    sequences_[i]->Stop();
   network_thread_->Post(this, MSG_CONFIG_STOP);
+  ClearGettingPorts();
+}
+
+void BasicPortAllocatorSession::ClearGettingPorts() {
+  network_thread_->Clear(this, MSG_ALLOCATE);
+  for (uint32_t i = 0; i < sequences_.size(); ++i)
+    sequences_[i]->Stop();
 }
 
 void BasicPortAllocatorSession::OnMessage(rtc::Message *message) {
@@ -319,8 +247,8 @@ void BasicPortAllocatorSession::GetPortConfigurations() {
                                                     username(),
                                                     password());
 
-  for (size_t i = 0; i < allocator_->relays().size(); ++i) {
-    config->AddRelay(allocator_->relays()[i]);
+  for (const RelayServerConfig& turn_server : allocator_->turn_servers()) {
+    config->AddRelay(turn_server);
   }
   ConfigReady(config);
 }
@@ -331,8 +259,9 @@ void BasicPortAllocatorSession::ConfigReady(PortConfiguration* config) {
 
 // Adds a configuration to the list.
 void BasicPortAllocatorSession::OnConfigReady(PortConfiguration* config) {
-  if (config)
+  if (config) {
     configs_.push_back(config);
+  }
 
   AllocatePorts();
 }
@@ -380,31 +309,51 @@ void BasicPortAllocatorSession::OnAllocate() {
   allocation_started_ = true;
 }
 
-// For each network, see if we have a sequence that covers it already.  If not,
-// create a new sequence to create the appropriate ports.
-void BasicPortAllocatorSession::DoAllocate() {
-  bool done_signal_needed = false;
-  std::vector<rtc::Network*> networks;
-
+void BasicPortAllocatorSession::GetNetworks(
+    std::vector<rtc::Network*>* networks) {
+  networks->clear();
+  rtc::NetworkManager* network_manager = allocator_->network_manager();
+  ASSERT(network_manager != nullptr);
+  // If the network permission state is BLOCKED, we just act as if the flag has
+  // been passed in.
+  if (network_manager->enumeration_permission() ==
+      rtc::NetworkManager::ENUMERATION_BLOCKED) {
+    set_flags(flags() | PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
+  }
   // If the adapter enumeration is disabled, we'll just bind to any address
   // instead of specific NIC. This is to ensure the same routing for http
   // traffic by OS is also used here to avoid any local or public IP leakage
   // during stun process.
   if (flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) {
-    allocator_->network_manager()->GetAnyAddressNetworks(&networks);
+    network_manager->GetAnyAddressNetworks(networks);
   } else {
-    allocator_->network_manager()->GetNetworks(&networks);
+    network_manager->GetNetworks(networks);
   }
+  networks->erase(std::remove_if(networks->begin(), networks->end(),
+                                 [this](rtc::Network* network) {
+                                   return allocator_->network_ignore_mask() &
+                                          network->type();
+                                 }),
+                  networks->end());
+}
+
+// For each network, see if we have a sequence that covers it already.  If not,
+// create a new sequence to create the appropriate ports.
+void BasicPortAllocatorSession::DoAllocate() {
+  bool done_signal_needed = false;
+  std::vector<rtc::Network*> networks;
+  GetNetworks(&networks);
+
   if (networks.empty()) {
     LOG(LS_WARNING) << "Machine has no networks; no ports will be allocated";
     done_signal_needed = true;
   } else {
-    for (uint32 i = 0; i < networks.size(); ++i) {
+    for (uint32_t i = 0; i < networks.size(); ++i) {
       PortConfiguration* config = NULL;
       if (configs_.size() > 0)
         config = configs_.back();
 
-      uint32 sequence_flags = flags();
+      uint32_t sequence_flags = flags();
       if ((sequence_flags & DISABLE_ALL_PHASES) == DISABLE_ALL_PHASES) {
         // If all the ports are disabled we should just fire the allocation
         // done event and return.
@@ -412,11 +361,6 @@ void BasicPortAllocatorSession::DoAllocate() {
         break;
       }
 
-      // Disables phases that are not specified in this config.
-      if (!config || config->StunServers().empty()) {
-        // No STUN ports specified in this config.
-        sequence_flags |= PORTALLOCATOR_DISABLE_STUN;
-      }
       if (!config || config->relays.empty()) {
         // No relay ports specified in this config.
         sequence_flags |= PORTALLOCATOR_DISABLE_RELAY;
@@ -457,15 +401,30 @@ void BasicPortAllocatorSession::DoAllocate() {
 }
 
 void BasicPortAllocatorSession::OnNetworksChanged() {
+  std::vector<rtc::Network*> networks;
+  GetNetworks(&networks);
+  for (AllocationSequence* sequence : sequences_) {
+    // Remove the network from the allocation sequence if it is not in
+    // |networks|.
+    if (!sequence->network_removed() &&
+        std::find(networks.begin(), networks.end(), sequence->network()) ==
+            networks.end()) {
+      sequence->OnNetworkRemoved();
+    }
+  }
+
   network_manager_started_ = true;
   if (allocation_started_)
     DoAllocate();
 }
 
 void BasicPortAllocatorSession::DisableEquivalentPhases(
-    rtc::Network* network, PortConfiguration* config, uint32* flags) {
-  for (uint32 i = 0; i < sequences_.size() &&
-      (*flags & DISABLE_ALL_PHASES) != DISABLE_ALL_PHASES; ++i) {
+    rtc::Network* network,
+    PortConfiguration* config,
+    uint32_t* flags) {
+  for (uint32_t i = 0; i < sequences_.size() &&
+                           (*flags & DISABLE_ALL_PHASES) != DISABLE_ALL_PHASES;
+       ++i) {
     sequences_[i]->DisableEquivalentPhases(network, config, flags);
   }
 }
@@ -486,11 +445,12 @@ void BasicPortAllocatorSession::AddAllocatedPort(Port* port,
       PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0);
 
   // Push down the candidate_filter to individual port.
-  uint32 candidate_filter = allocator_->candidate_filter();
+  uint32_t candidate_filter = allocator_->candidate_filter();
 
   // When adapter enumeration is disabled, disable CF_HOST at port level so
   // local address is not leaked by stunport in the candidate's related address.
-  if (flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) {
+  if ((flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) &&
+      (flags() & PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE)) {
     candidate_filter &= ~CF_HOST;
   }
   port->set_candidate_filter(candidate_filter);
@@ -528,26 +488,44 @@ void BasicPortAllocatorSession::OnCandidateReady(
   if (data->complete())
     return;
 
-  // Send candidates whose protocol is enabled.
-  std::vector<Candidate> candidates;
   ProtocolType pvalue;
-  bool candidate_allowed_to_send = CheckCandidateFilter(c);
-  if (StringToProto(c.protocol().c_str(), &pvalue) &&
-      data->sequence()->ProtocolEnabled(pvalue) &&
-      candidate_allowed_to_send) {
-    candidates.push_back(c);
-  }
+  bool candidate_signalable = CheckCandidateFilter(c);
 
-  if (!candidates.empty()) {
+  // When device enumeration is disabled (to prevent non-default IP addresses
+  // from leaking), we ping from some local candidates even though we don't
+  // signal them. However, if host candidates are also disabled (for example, to
+  // prevent even default IP addresses from leaking), we still don't want to
+  // ping from them, even if device enumeration is disabled.  Thus, we check for
+  // both device enumeration and host candidates being disabled.
+  bool network_enumeration_disabled = c.address().IsAnyIP();
+  bool can_ping_from_candidate =
+      (port->SharedSocket() || c.protocol() == TCP_PROTOCOL_NAME);
+  bool host_canidates_disabled = !(allocator_->candidate_filter() & CF_HOST);
+
+  bool candidate_pairable =
+      candidate_signalable ||
+      (network_enumeration_disabled && can_ping_from_candidate &&
+       !host_canidates_disabled);
+  bool candidate_protocol_enabled =
+      StringToProto(c.protocol().c_str(), &pvalue) &&
+      data->sequence()->ProtocolEnabled(pvalue);
+
+  if (candidate_signalable && candidate_protocol_enabled) {
+    std::vector<Candidate> candidates;
+    candidates.push_back(c);
     SignalCandidatesReady(this, candidates);
   }
 
-  // Moving to READY state as we have atleast one candidate from the port.
-  // Since this port has atleast one candidate we should forward this port
-  // to listners, to allow connections from this port.
-  // Also we should make sure that candidate gathered from this port is allowed
-  // to send outside.
-  if (!data->ready() && candidate_allowed_to_send) {
+  // Port has been made ready. Nothing to do here.
+  if (data->ready()) {
+    return;
+  }
+
+  // Move the port to the READY state, either because we have a usable candidate
+  // from the port, or simply because the port is bound to the any address and
+  // therefore has no host candidate. This will trigger the port to start
+  // creating candidate pairs (connections) and issue connectivity checks.
+  if (candidate_pairable) {
     data->set_ready();
     SignalPortReady(this, port);
   }
@@ -596,9 +574,10 @@ void BasicPortAllocatorSession::OnProtocolEnabled(AllocationSequence* seq,
       if (!CheckCandidateFilter(potentials[i]))
         continue;
       ProtocolType pvalue;
-      if (!StringToProto(potentials[i].protocol().c_str(), &pvalue))
-        continue;
-      if (pvalue == proto) {
+      bool candidate_protocol_enabled =
+          StringToProto(potentials[i].protocol().c_str(), &pvalue) &&
+          pvalue == proto;
+      if (candidate_protocol_enabled) {
         candidates.push_back(potentials[i]);
       }
     }
@@ -610,7 +589,7 @@ void BasicPortAllocatorSession::OnProtocolEnabled(AllocationSequence* seq,
 }
 
 bool BasicPortAllocatorSession::CheckCandidateFilter(const Candidate& c) {
-  uint32 filter = allocator_->candidate_filter();
+  uint32_t filter = allocator_->candidate_filter();
 
   // When binding to any address, before sending packets out, the getsockname
   // returns all 0s, but after sending packets, it'll be the NIC used to
@@ -635,17 +614,6 @@ bool BasicPortAllocatorSession::CheckCandidateFilter(const Candidate& c) {
       return true;
     }
 
-    // This is just to prevent the case when binding to any address (all 0s), if
-    // somehow the host candidate address is not all 0s. Either because local
-    // installed proxy changes the address or a packet has been sent for any
-    // reason before getsockname is called.
-    if (flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) {
-      LOG(LS_WARNING) << "Received non-0 host address: "
-                      << c.address().ToString()
-                      << " when adapter enumeration is disabled";
-      return false;
-    }
-
     return ((filter & CF_HOST) != 0);
   }
   return false;
@@ -744,7 +712,7 @@ BasicPortAllocatorSession::PortData* BasicPortAllocatorSession::FindPort(
 AllocationSequence::AllocationSequence(BasicPortAllocatorSession* session,
                                        rtc::Network* network,
                                        PortConfiguration* config,
-                                       uint32 flags)
+                                       uint32_t flags)
     : session_(session),
       network_(network),
       ip_(network->GetBestIP()),
@@ -757,14 +725,6 @@ AllocationSequence::AllocationSequence(BasicPortAllocatorSession* session,
 }
 
 bool AllocationSequence::Init() {
-  if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) &&
-      !IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_UFRAG)) {
-    LOG(LS_ERROR) << "Shared socket option can't be set without "
-                  << "shared ufrag.";
-    ASSERT(false);
-    return false;
-  }
-
   if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET)) {
     udp_socket_.reset(session_->socket_factory()->CreateUdpSocket(
         rtc::SocketAddress(ip_, 0), session_->allocator()->min_port(),
@@ -784,12 +744,24 @@ void AllocationSequence::Clear() {
   turn_ports_.clear();
 }
 
+void AllocationSequence::OnNetworkRemoved() {
+  // Stop the allocation sequence if its network is gone.
+  Stop();
+  network_removed_ = true;
+}
+
 AllocationSequence::~AllocationSequence() {
   session_->network_thread()->Clear(this);
 }
 
 void AllocationSequence::DisableEquivalentPhases(rtc::Network* network,
-    PortConfiguration* config, uint32* flags) {
+    PortConfiguration* config, uint32_t* flags) {
+  if (network_removed_) {
+    // If the network of this allocation sequence has ever gone away,
+    // it won't be equivalent to the new network.
+    return;
+  }
+
   if (!((network == network_) && (ip_ == network->GetBestIP()))) {
     // Different network setup; nothing is equivalent.
     return;
@@ -905,20 +877,19 @@ void AllocationSequence::CreateUDPPorts() {
   // TODO(mallinath) - Remove UDPPort creating socket after shared socket
   // is enabled completely.
   UDPPort* port = NULL;
+  bool emit_local_candidate_for_anyaddress =
+      !IsFlagSet(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
   if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) && udp_socket_) {
-    port = UDPPort::Create(session_->network_thread(),
-                           session_->socket_factory(), network_,
-                           udp_socket_.get(),
-                           session_->username(), session_->password(),
-                           session_->allocator()->origin());
+    port = UDPPort::Create(
+        session_->network_thread(), session_->socket_factory(), network_,
+        udp_socket_.get(), session_->username(), session_->password(),
+        session_->allocator()->origin(), emit_local_candidate_for_anyaddress);
   } else {
-    port = UDPPort::Create(session_->network_thread(),
-                           session_->socket_factory(),
-                           network_, ip_,
-                           session_->allocator()->min_port(),
-                           session_->allocator()->max_port(),
-                           session_->username(), session_->password(),
-                           session_->allocator()->origin());
+    port = UDPPort::Create(
+        session_->network_thread(), session_->socket_factory(), network_, ip_,
+        session_->allocator()->min_port(), session_->allocator()->max_port(),
+        session_->username(), session_->password(),
+        session_->allocator()->origin(), emit_local_candidate_for_anyaddress);
   }
 
   if (port) {
@@ -930,18 +901,10 @@ void AllocationSequence::CreateUDPPorts() {
 
       // If STUN is not disabled, setting stun server address to port.
       if (!IsFlagSet(PORTALLOCATOR_DISABLE_STUN)) {
-        // If config has stun_servers, use it to get server reflexive candidate
-        // otherwise use first TURN server which supports UDP.
         if (config_ && !config_->StunServers().empty()) {
           LOG(LS_INFO) << "AllocationSequence: UDPPort will be handling the "
                        <<  "STUN candidate generation.";
           port->set_server_addresses(config_->StunServers());
-        } else if (config_ &&
-                   config_->SupportsProtocol(RELAY_TURN, PROTO_UDP)) {
-          port->set_server_addresses(config_->GetRelayServerAddresses(
-              RELAY_TURN, PROTO_UDP));
-          LOG(LS_INFO) << "AllocationSequence: TURN Server address will be "
-                       << " used for generating STUN candidate.";
         }
       }
     }
@@ -980,9 +943,6 @@ void AllocationSequence::CreateStunPorts() {
     return;
   }
 
-  // If BasicPortAllocatorSession::OnAllocate left STUN ports enabled then we
-  // ought to have an address for them here.
-  ASSERT(config_ && !config_->StunServers().empty());
   if (!(config_ && !config_->StunServers().empty())) {
     LOG(LS_WARNING)
         << "AllocationSequence: No STUN server configured, skipping.";
@@ -1068,12 +1028,19 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config) {
   for (relay_port = config.ports.begin();
        relay_port != config.ports.end(); ++relay_port) {
     TurnPort* port = NULL;
+
+    // Skip UDP connections to relay servers if it's disallowed.
+    if (IsFlagSet(PORTALLOCATOR_DISABLE_UDP_RELAY) &&
+        relay_port->proto == PROTO_UDP) {
+      continue;
+    }
+
     // Shared socket mode must be enabled only for UDP based ports. Hence
     // don't pass shared socket for ports which will create TCP sockets.
     // TODO(mallinath) - Enable shared socket mode for TURN ports. Disabled
     // due to webrtc bug https://code.google.com/p/webrtc/issues/detail?id=3537
     if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) &&
-        relay_port->proto == PROTO_UDP) {
+        relay_port->proto == PROTO_UDP && udp_socket_) {
       port = TurnPort::Create(session_->network_thread(),
                               session_->socket_factory(),
                               network_, udp_socket_.get(),
@@ -1177,6 +1144,13 @@ ServerAddresses PortConfiguration::StunServers() {
       stun_servers.find(stun_address) == stun_servers.end()) {
     stun_servers.insert(stun_address);
   }
+  // Every UDP TURN server should also be used as a STUN server.
+  ServerAddresses turn_servers = GetRelayServerAddresses(RELAY_TURN, PROTO_UDP);
+  for (const rtc::SocketAddress& turn_server : turn_servers) {
+    if (stun_servers.find(turn_server) == stun_servers.end()) {
+      stun_servers.insert(turn_server);
+    }
+  }
   return stun_servers;
 }
 
diff --git a/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.h b/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.h
index 96468d32cc..ca1a23aaf2 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.h
+++ b/media/webrtc/trunk/webrtc/p2p/client/basicportallocator.h
@@ -14,7 +14,6 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/p2p/base/port.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/network.h"
@@ -23,28 +22,6 @@
 
 namespace cricket {
 
-struct RelayCredentials {
-  RelayCredentials() {}
-  RelayCredentials(const std::string& username,
-                   const std::string& password)
-      : username(username),
-        password(password) {
-  }
-
-  std::string username;
-  std::string password;
-};
-
-typedef std::vector<ProtocolAddress> PortList;
-struct RelayServerConfig {
-  RelayServerConfig(RelayType type) : type(type), priority(0) {}
-
-  RelayType type;
-  PortList ports;
-  RelayCredentials credentials;
-  int priority;
-};
-
 class BasicPortAllocator : public PortAllocator {
  public:
   BasicPortAllocator(rtc::NetworkManager* network_manager,
@@ -60,6 +37,23 @@ class BasicPortAllocator : public PortAllocator {
                      const rtc::SocketAddress& relay_server_ssl);
   virtual ~BasicPortAllocator();
 
+  void SetIceServers(
+      const ServerAddresses& stun_servers,
+      const std::vector<RelayServerConfig>& turn_servers) override {
+    stun_servers_ = stun_servers;
+    turn_servers_ = turn_servers;
+  }
+
+  // Set to kDefaultNetworkIgnoreMask by default.
+  void SetNetworkIgnoreMask(int network_ignore_mask) override {
+    // TODO(phoglund): implement support for other types than loopback.
+    // See https://code.google.com/p/webrtc/issues/detail?id=4288.
+    // Then remove set_network_ignore_list from NetworkManager.
+    network_ignore_mask_ = network_ignore_mask;
+  }
+
+  int network_ignore_mask() const { return network_ignore_mask_; }
+
   rtc::NetworkManager* network_manager() { return network_manager_; }
 
   // If socket_factory() is set to NULL each PortAllocatorSession
@@ -70,27 +64,28 @@ class BasicPortAllocator : public PortAllocator {
     return stun_servers_;
   }
 
-  const std::vector<RelayServerConfig>& relays() const {
-    return relays_;
+  const std::vector<RelayServerConfig>& turn_servers() const {
+    return turn_servers_;
   }
-  virtual void AddRelay(const RelayServerConfig& relay) {
-    relays_.push_back(relay);
+  virtual void AddTurnServer(const RelayServerConfig& turn_server) {
+    turn_servers_.push_back(turn_server);
   }
 
-  virtual PortAllocatorSession* CreateSessionInternal(
+  PortAllocatorSession* CreateSessionInternal(
       const std::string& content_name,
       int component,
       const std::string& ice_ufrag,
-      const std::string& ice_pwd);
+      const std::string& ice_pwd) override;
 
  private:
   void Construct();
 
   rtc::NetworkManager* network_manager_;
   rtc::PacketSocketFactory* socket_factory_;
-  const ServerAddresses stun_servers_;
-  std::vector<RelayServerConfig> relays_;
+  ServerAddresses stun_servers_;
+  std::vector<RelayServerConfig> turn_servers_;
   bool allow_tcp_listen_;
+  int network_ignore_mask_ = rtc::kDefaultNetworkIgnoreMask;
 };
 
 struct PortConfiguration;
@@ -110,9 +105,10 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
   rtc::Thread* network_thread() { return network_thread_; }
   rtc::PacketSocketFactory* socket_factory() { return socket_factory_; }
 
-  virtual void StartGettingPorts();
-  virtual void StopGettingPorts();
-  virtual bool IsGettingPorts() { return running_; }
+  void StartGettingPorts() override;
+  void StopGettingPorts() override;
+  void ClearGettingPorts() override;
+  bool IsGettingPorts() override { return running_; }
 
  protected:
   // Starts the process of getting the port configurations.
@@ -123,7 +119,7 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
   virtual void ConfigReady(PortConfiguration* config);
 
   // MessageHandler.  Can be overriden if message IDs do not conflict.
-  virtual void OnMessage(rtc::Message *message);
+  void OnMessage(rtc::Message* message) override;
 
  private:
   class PortData {
@@ -170,7 +166,8 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
   void OnNetworksChanged();
   void OnAllocationSequenceObjectsCreated();
   void DisableEquivalentPhases(rtc::Network* network,
-                               PortConfiguration* config, uint32* flags);
+                               PortConfiguration* config,
+                               uint32_t* flags);
   void AddAllocatedPort(Port* port, AllocationSequence* seq,
                         bool prepare_address);
   void OnCandidateReady(Port* port, const Candidate& c);
@@ -182,6 +179,7 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
   void MaybeSignalCandidatesAllocationDone();
   void OnPortAllocationComplete(AllocationSequence* seq);
   PortData* FindPort(Port* port);
+  void GetNetworks(std::vector<rtc::Network*>* networks);
 
   bool CheckCandidateFilter(const Candidate& c);
 
@@ -201,6 +199,7 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
 };
 
 // Records configuration information useful in creating ports.
+// TODO(deadbeef): Rename "relay" to "turn_server" in this struct.
 struct PortConfiguration : public rtc::MessageData {
   // TODO(jiayl): remove |stun_address| when Chrome is updated.
   rtc::SocketAddress stun_address;
@@ -220,7 +219,8 @@ struct PortConfiguration : public rtc::MessageData {
                     const std::string& username,
                     const std::string& password);
 
-  // TODO(jiayl): remove when |stun_address| is removed.
+  // Returns addresses of both the explicitly configured STUN servers,
+  // and TURN servers that should be used as STUN servers.
   ServerAddresses StunServers();
 
   // Adds another relay server, with the given ports and modifier, to the list.
@@ -236,6 +236,97 @@ struct PortConfiguration : public rtc::MessageData {
       RelayType turn_type, ProtocolType type) const;
 };
 
+class UDPPort;
+class TurnPort;
+
+// Performs the allocation of ports, in a sequenced (timed) manner, for a given
+// network and IP address.
+class AllocationSequence : public rtc::MessageHandler,
+                           public sigslot::has_slots<> {
+ public:
+  enum State {
+    kInit,       // Initial state.
+    kRunning,    // Started allocating ports.
+    kStopped,    // Stopped from running.
+    kCompleted,  // All ports are allocated.
+
+    // kInit --> kRunning --> {kCompleted|kStopped}
+  };
+  AllocationSequence(BasicPortAllocatorSession* session,
+                     rtc::Network* network,
+                     PortConfiguration* config,
+                     uint32_t flags);
+  ~AllocationSequence();
+  bool Init();
+  void Clear();
+  void OnNetworkRemoved();
+
+  State state() const { return state_; }
+  const rtc::Network* network() const { return network_; }
+  bool network_removed() const { return network_removed_; }
+
+  // Disables the phases for a new sequence that this one already covers for an
+  // equivalent network setup.
+  void DisableEquivalentPhases(rtc::Network* network,
+                               PortConfiguration* config,
+                               uint32_t* flags);
+
+  // Starts and stops the sequence.  When started, it will continue allocating
+  // new ports on its own timed schedule.
+  void Start();
+  void Stop();
+
+  // MessageHandler
+  void OnMessage(rtc::Message* msg);
+
+  void EnableProtocol(ProtocolType proto);
+  bool ProtocolEnabled(ProtocolType proto) const;
+
+  // Signal from AllocationSequence, when it's done with allocating ports.
+  // This signal is useful, when port allocation fails which doesn't result
+  // in any candidates. Using this signal BasicPortAllocatorSession can send
+  // its candidate discovery conclusion signal. Without this signal,
+  // BasicPortAllocatorSession doesn't have any event to trigger signal. This
+  // can also be achieved by starting timer in BPAS.
+  sigslot::signal1<AllocationSequence*> SignalPortAllocationComplete;
+
+ protected:
+  // For testing.
+  void CreateTurnPort(const RelayServerConfig& config);
+
+ private:
+  typedef std::vector<ProtocolType> ProtocolList;
+
+  bool IsFlagSet(uint32_t flag) { return ((flags_ & flag) != 0); }
+  void CreateUDPPorts();
+  void CreateTCPPorts();
+  void CreateStunPorts();
+  void CreateRelayPorts();
+  void CreateGturnPort(const RelayServerConfig& config);
+
+  void OnReadPacket(rtc::AsyncPacketSocket* socket,
+                    const char* data,
+                    size_t size,
+                    const rtc::SocketAddress& remote_addr,
+                    const rtc::PacketTime& packet_time);
+
+  void OnPortDestroyed(PortInterface* port);
+
+  BasicPortAllocatorSession* session_;
+  bool network_removed_ = false;
+  rtc::Network* network_;
+  rtc::IPAddress ip_;
+  PortConfiguration* config_;
+  State state_;
+  uint32_t flags_;
+  ProtocolList protocols_;
+  rtc::scoped_ptr<rtc::AsyncPacketSocket> udp_socket_;
+  // There will be only one udp port per AllocationSequence.
+  UDPPort* udp_port_;
+  std::vector<TurnPort*> turn_ports_;
+  int phase_;
+};
+
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker.cc b/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker.cc
deleted file mode 100644
index 1fb9165670..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker.cc
+++ /dev/null
@@ -1,532 +0,0 @@
-/*
- *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-
-#include "webrtc/p2p/client/connectivitychecker.h"
-
-#include "webrtc/p2p/base/candidate.h"
-#include "webrtc/p2p/base/common.h"
-#include "webrtc/p2p/base/constants.h"
-#include "webrtc/p2p/base/port.h"
-#include "webrtc/p2p/base/relayport.h"
-#include "webrtc/p2p/base/stunport.h"
-#include "webrtc/base/asynchttprequest.h"
-#include "webrtc/base/autodetectproxy.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/httpcommon-inl.h"
-#include "webrtc/base/httpcommon.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/proxydetect.h"
-#include "webrtc/base/thread.h"
-
-namespace cricket {
-
-static const char kDefaultStunHostname[] = "stun.l.google.com";
-static const int kDefaultStunPort = 19302;
-
-// Default maximum time in milliseconds we will wait for connections.
-static const uint32 kDefaultTimeoutMs = 3000;
-
-enum {
-  MSG_START = 1,
-  MSG_STOP = 2,
-  MSG_TIMEOUT = 3,
-  MSG_SIGNAL_RESULTS = 4
-};
-
-class TestHttpPortAllocator : public HttpPortAllocator {
- public:
-  TestHttpPortAllocator(rtc::NetworkManager* network_manager,
-                        const std::string& user_agent,
-                        const std::string& relay_token) :
-      HttpPortAllocator(network_manager, user_agent) {
-    SetRelayToken(relay_token);
-  }
-  PortAllocatorSession* CreateSessionInternal(
-      const std::string& content_name,
-      int component,
-      const std::string& ice_ufrag,
-      const std::string& ice_pwd) {
-    return new TestHttpPortAllocatorSession(this, content_name, component,
-                                            ice_ufrag, ice_pwd,
-                                            stun_hosts(), relay_hosts(),
-                                            relay_token(), user_agent());
-  }
-};
-
-void TestHttpPortAllocatorSession::ConfigReady(PortConfiguration* config) {
-  SignalConfigReady(username(), password(), config, proxy_);
-  delete config;
-}
-
-void TestHttpPortAllocatorSession::OnRequestDone(
-    rtc::SignalThread* data) {
-  rtc::AsyncHttpRequest* request =
-      static_cast<rtc::AsyncHttpRequest*>(data);
-
-  // Tell the checker that the request is complete.
-  SignalRequestDone(request);
-
-  // Pass on the response to super class.
-  HttpPortAllocatorSession::OnRequestDone(data);
-}
-
-ConnectivityChecker::ConnectivityChecker(
-    rtc::Thread* worker,
-    const std::string& jid,
-    const std::string& session_id,
-    const std::string& user_agent,
-    const std::string& relay_token,
-    const std::string& connection)
-    : worker_(worker),
-      jid_(jid),
-      session_id_(session_id),
-      user_agent_(user_agent),
-      relay_token_(relay_token),
-      connection_(connection),
-      proxy_detect_(NULL),
-      timeout_ms_(kDefaultTimeoutMs),
-      stun_address_(kDefaultStunHostname, kDefaultStunPort),
-      started_(false) {
-}
-
-ConnectivityChecker::~ConnectivityChecker() {
-  if (started_) {
-    // We try to clear the TIMEOUT below. But worker may still handle it and
-    // cause SignalCheckDone to happen on main-thread. So we finally clear any
-    // pending SIGNAL_RESULTS.
-    worker_->Clear(this, MSG_TIMEOUT);
-    worker_->Send(this, MSG_STOP);
-    nics_.clear();
-    main_->Clear(this, MSG_SIGNAL_RESULTS);
-  }
-}
-
-bool ConnectivityChecker::Initialize() {
-  network_manager_.reset(CreateNetworkManager());
-  socket_factory_.reset(CreateSocketFactory(worker_));
-  port_allocator_.reset(CreatePortAllocator(network_manager_.get(),
-                                            user_agent_, relay_token_));
-  uint32 new_allocator_flags = port_allocator_->flags();
-  new_allocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG;
-  port_allocator_->set_flags(new_allocator_flags);
-  return true;
-}
-
-void ConnectivityChecker::Start() {
-  main_ = rtc::Thread::Current();
-  worker_->Post(this, MSG_START);
-  started_ = true;
-}
-
-void ConnectivityChecker::CleanUp() {
-  ASSERT(worker_ == rtc::Thread::Current());
-  if (proxy_detect_) {
-    proxy_detect_->Release();
-    proxy_detect_ = NULL;
-  }
-
-  for (uint32 i = 0; i < sessions_.size(); ++i) {
-    delete sessions_[i];
-  }
-  sessions_.clear();
-  for (uint32 i = 0; i < ports_.size(); ++i) {
-    delete ports_[i];
-  }
-  ports_.clear();
-}
-
-bool ConnectivityChecker::AddNic(const rtc::IPAddress& ip,
-                                 const rtc::SocketAddress& proxy_addr) {
-  NicMap::iterator i = nics_.find(NicId(ip, proxy_addr));
-  if (i != nics_.end()) {
-    // Already have it.
-    return false;
-  }
-  uint32 now = rtc::Time();
-  NicInfo info;
-  info.ip = ip;
-  info.proxy_info = GetProxyInfo();
-  info.stun.start_time_ms = now;
-  nics_.insert(std::pair<NicId, NicInfo>(NicId(ip, proxy_addr), info));
-  return true;
-}
-
-void ConnectivityChecker::SetProxyInfo(const rtc::ProxyInfo& proxy_info) {
-  port_allocator_->set_proxy(user_agent_, proxy_info);
-  AllocatePorts();
-}
-
-rtc::ProxyInfo ConnectivityChecker::GetProxyInfo() const {
-  rtc::ProxyInfo proxy_info;
-  if (proxy_detect_) {
-    proxy_info = proxy_detect_->proxy();
-  }
-  return proxy_info;
-}
-
-void ConnectivityChecker::CheckNetworks() {
-  network_manager_->SignalNetworksChanged.connect(
-      this, &ConnectivityChecker::OnNetworksChanged);
-  network_manager_->StartUpdating();
-}
-
-void ConnectivityChecker::OnMessage(rtc::Message *msg) {
-  switch (msg->message_id) {
-    case MSG_START:
-      ASSERT(worker_ == rtc::Thread::Current());
-      worker_->PostDelayed(timeout_ms_, this, MSG_TIMEOUT);
-      CheckNetworks();
-      break;
-    case MSG_STOP:
-      // We're being stopped, free resources.
-      CleanUp();
-      break;
-    case MSG_TIMEOUT:
-      // We need to signal results on the main thread.
-      main_->Post(this, MSG_SIGNAL_RESULTS);
-      break;
-    case MSG_SIGNAL_RESULTS:
-      ASSERT(main_ == rtc::Thread::Current());
-      SignalCheckDone(this);
-      break;
-    default:
-      LOG(LS_ERROR) << "Unknown message: " << msg->message_id;
-  }
-}
-
-void ConnectivityChecker::OnProxyDetect(rtc::SignalThread* thread) {
-  ASSERT(worker_ == rtc::Thread::Current());
-  if (proxy_detect_->proxy().type != rtc::PROXY_NONE) {
-    SetProxyInfo(proxy_detect_->proxy());
-  }
-}
-
-void ConnectivityChecker::OnRequestDone(rtc::AsyncHttpRequest* request) {
-  ASSERT(worker_ == rtc::Thread::Current());
-  // Since we don't know what nic were actually used for the http request,
-  // for now, just use the first one.
-  std::vector<rtc::Network*> networks;
-  network_manager_->GetNetworks(&networks);
-  if (networks.empty()) {
-    LOG(LS_ERROR) << "No networks while registering http start.";
-    return;
-  }
-  rtc::ProxyInfo proxy_info = request->proxy();
-  NicMap::iterator i =
-      nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
-  if (i != nics_.end()) {
-    int port = request->port();
-    uint32 now = rtc::Time();
-    NicInfo* nic_info = &i->second;
-    if (port == rtc::HTTP_SECURE_PORT) {
-      nic_info->https.rtt = now - nic_info->https.start_time_ms;
-    } else {
-      LOG(LS_ERROR) << "Got response with unknown port: " << port;
-    }
-  } else {
-    LOG(LS_ERROR) << "No nic info found while receiving response.";
-  }
-}
-
-void ConnectivityChecker::OnConfigReady(
-    const std::string& username, const std::string& password,
-    const PortConfiguration* config, const rtc::ProxyInfo& proxy_info) {
-  ASSERT(worker_ == rtc::Thread::Current());
-
-  // Since we send requests on both HTTP and HTTPS we will get two
-  // configs per nic. Results from the second will overwrite the
-  // result from the first.
-  // TODO: Handle multiple pings on one nic.
-  CreateRelayPorts(username, password, config, proxy_info);
-}
-
-void ConnectivityChecker::OnRelayPortComplete(Port* port) {
-  ASSERT(worker_ == rtc::Thread::Current());
-  RelayPort* relay_port = reinterpret_cast<RelayPort*>(port);
-  const ProtocolAddress* address = relay_port->ServerAddress(0);
-  rtc::IPAddress ip = port->Network()->GetBestIP();
-  NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
-  if (i != nics_.end()) {
-    // We have it already, add the new information.
-    NicInfo* nic_info = &i->second;
-    ConnectInfo* connect_info = NULL;
-    if (address) {
-      switch (address->proto) {
-        case PROTO_UDP:
-          connect_info = &nic_info->udp;
-          break;
-        case PROTO_TCP:
-          connect_info = &nic_info->tcp;
-          break;
-        case PROTO_SSLTCP:
-          connect_info = &nic_info->ssltcp;
-          break;
-        default:
-          LOG(LS_ERROR) << " relay address with bad protocol added";
-      }
-      if (connect_info) {
-        connect_info->rtt =
-            rtc::TimeSince(connect_info->start_time_ms);
-      }
-    }
-  } else {
-    LOG(LS_ERROR) << " got relay address for non-existing nic";
-  }
-}
-
-void ConnectivityChecker::OnStunPortComplete(Port* port) {
-  ASSERT(worker_ == rtc::Thread::Current());
-  const std::vector<Candidate> candidates = port->Candidates();
-  Candidate c = candidates[0];
-  rtc::IPAddress ip = port->Network()->GetBestIP();
-  NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
-  if (i != nics_.end()) {
-    // We have it already, add the new information.
-    uint32 now = rtc::Time();
-    NicInfo* nic_info = &i->second;
-    nic_info->external_address = c.address();
-
-    nic_info->stun_server_addresses =
-        static_cast<StunPort*>(port)->server_addresses();
-    nic_info->stun.rtt = now - nic_info->stun.start_time_ms;
-  } else {
-    LOG(LS_ERROR) << "Got stun address for non-existing nic";
-  }
-}
-
-void ConnectivityChecker::OnStunPortError(Port* port) {
-  ASSERT(worker_ == rtc::Thread::Current());
-  LOG(LS_ERROR) << "Stun address error.";
-  rtc::IPAddress ip = port->Network()->GetBestIP();
-  NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
-  if (i != nics_.end()) {
-    // We have it already, add the new information.
-    NicInfo* nic_info = &i->second;
-
-    nic_info->stun_server_addresses =
-        static_cast<StunPort*>(port)->server_addresses();
-  }
-}
-
-void ConnectivityChecker::OnRelayPortError(Port* port) {
-  ASSERT(worker_ == rtc::Thread::Current());
-  LOG(LS_ERROR) << "Relay address error.";
-}
-
-void ConnectivityChecker::OnNetworksChanged() {
-  ASSERT(worker_ == rtc::Thread::Current());
-  std::vector<rtc::Network*> networks;
-  network_manager_->GetNetworks(&networks);
-  if (networks.empty()) {
-    LOG(LS_ERROR) << "Machine has no networks; nothing to do";
-    return;
-  }
-  AllocatePorts();
-}
-
-HttpPortAllocator* ConnectivityChecker::CreatePortAllocator(
-    rtc::NetworkManager* network_manager,
-    const std::string& user_agent,
-    const std::string& relay_token) {
-  return new TestHttpPortAllocator(network_manager, user_agent, relay_token);
-}
-
-StunPort* ConnectivityChecker::CreateStunPort(
-    const std::string& username, const std::string& password,
-    const PortConfiguration* config, rtc::Network* network) {
-  return StunPort::Create(worker_,
-                          socket_factory_.get(),
-                          network,
-                          network->GetBestIP(),
-                          0,
-                          0,
-                          username,
-                          password,
-                          config->stun_servers,
-                          std::string());
-}
-
-RelayPort* ConnectivityChecker::CreateRelayPort(
-    const std::string& username, const std::string& password,
-    const PortConfiguration* config, rtc::Network* network) {
-  return RelayPort::Create(worker_,
-                           socket_factory_.get(),
-                           network,
-                           network->GetBestIP(),
-                           port_allocator_->min_port(),
-                           port_allocator_->max_port(),
-                           username,
-                           password);
-}
-
-void ConnectivityChecker::CreateRelayPorts(
-    const std::string& username, const std::string& password,
-    const PortConfiguration* config, const rtc::ProxyInfo& proxy_info) {
-  PortConfiguration::RelayList::const_iterator relay;
-  std::vector<rtc::Network*> networks;
-  network_manager_->GetNetworks(&networks);
-  if (networks.empty()) {
-    LOG(LS_ERROR) << "Machine has no networks; no relay ports created.";
-    return;
-  }
-  for (relay = config->relays.begin();
-       relay != config->relays.end(); ++relay) {
-    for (uint32 i = 0; i < networks.size(); ++i) {
-      NicMap::iterator iter =
-          nics_.find(NicId(networks[i]->GetBestIP(), proxy_info.address));
-      if (iter != nics_.end()) {
-        // TODO: Now setting the same start time for all protocols.
-        // This might affect accuracy, but since we are mainly looking for
-        // connect failures or number that stick out, this is good enough.
-        uint32 now = rtc::Time();
-        NicInfo* nic_info = &iter->second;
-        nic_info->udp.start_time_ms = now;
-        nic_info->tcp.start_time_ms = now;
-        nic_info->ssltcp.start_time_ms = now;
-
-        // Add the addresses of this protocol.
-        PortList::const_iterator relay_port;
-        for (relay_port = relay->ports.begin();
-             relay_port != relay->ports.end();
-             ++relay_port) {
-          RelayPort* port = CreateRelayPort(username, password,
-                                            config, networks[i]);
-          port->AddServerAddress(*relay_port);
-          port->AddExternalAddress(*relay_port);
-
-          nic_info->media_server_address = port->ServerAddress(0)->address;
-
-          // Listen to network events.
-          port->SignalPortComplete.connect(
-              this, &ConnectivityChecker::OnRelayPortComplete);
-          port->SignalPortError.connect(
-              this, &ConnectivityChecker::OnRelayPortError);
-
-          port->set_proxy(user_agent_, proxy_info);
-
-          // Start fetching an address for this port.
-          port->PrepareAddress();
-          ports_.push_back(port);
-        }
-      } else {
-        LOG(LS_ERROR) << "Failed to find nic info when creating relay ports.";
-      }
-    }
-  }
-}
-
-void ConnectivityChecker::AllocatePorts() {
-  const std::string username = rtc::CreateRandomString(ICE_UFRAG_LENGTH);
-  const std::string password = rtc::CreateRandomString(ICE_PWD_LENGTH);
-  ServerAddresses stun_servers;
-  stun_servers.insert(stun_address_);
-  PortConfiguration config(stun_servers, username, password);
-  std::vector<rtc::Network*> networks;
-  network_manager_->GetNetworks(&networks);
-  if (networks.empty()) {
-    LOG(LS_ERROR) << "Machine has no networks; no ports will be allocated";
-    return;
-  }
-  rtc::ProxyInfo proxy_info = GetProxyInfo();
-  bool allocate_relay_ports = false;
-  for (uint32 i = 0; i < networks.size(); ++i) {
-    if (AddNic(networks[i]->GetBestIP(), proxy_info.address)) {
-      Port* port = CreateStunPort(username, password, &config, networks[i]);
-      if (port) {
-
-        // Listen to network events.
-        port->SignalPortComplete.connect(
-            this, &ConnectivityChecker::OnStunPortComplete);
-        port->SignalPortError.connect(
-            this, &ConnectivityChecker::OnStunPortError);
-
-        port->set_proxy(user_agent_, proxy_info);
-        port->PrepareAddress();
-        ports_.push_back(port);
-        allocate_relay_ports = true;
-      }
-    }
-  }
-
-  // If any new ip/proxy combinations were added, send a relay allocate.
-  if (allocate_relay_ports) {
-    AllocateRelayPorts();
-  }
-
-  // Initiate proxy detection.
-  InitiateProxyDetection();
-}
-
-void ConnectivityChecker::InitiateProxyDetection() {
-  // Only start if we haven't been started before.
-  if (!proxy_detect_) {
-    proxy_detect_ = new rtc::AutoDetectProxy(user_agent_);
-    rtc::Url<char> host_url("/", "relay.google.com",
-                                  rtc::HTTP_SECURE_PORT);
-    host_url.set_secure(true);
-    proxy_detect_->set_server_url(host_url.url());
-    proxy_detect_->SignalWorkDone.connect(
-        this, &ConnectivityChecker::OnProxyDetect);
-    proxy_detect_->Start();
-  }
-}
-
-void ConnectivityChecker::AllocateRelayPorts() {
-  // Currently we are using the 'default' nic for http(s) requests.
-  TestHttpPortAllocatorSession* allocator_session =
-      reinterpret_cast<TestHttpPortAllocatorSession*>(
-          port_allocator_->CreateSessionInternal(
-              "connectivity checker test content",
-              ICE_CANDIDATE_COMPONENT_RTP,
-              rtc::CreateRandomString(ICE_UFRAG_LENGTH),
-              rtc::CreateRandomString(ICE_PWD_LENGTH)));
-  allocator_session->set_proxy(port_allocator_->proxy());
-  allocator_session->SignalConfigReady.connect(
-      this, &ConnectivityChecker::OnConfigReady);
-  allocator_session->SignalRequestDone.connect(
-      this, &ConnectivityChecker::OnRequestDone);
-
-  // Try https only since using http would result in credentials being sent
-  // over the network unprotected.
-  RegisterHttpStart(rtc::HTTP_SECURE_PORT);
-  allocator_session->SendSessionRequest("relay.l.google.com",
-                                        rtc::HTTP_SECURE_PORT);
-
-  sessions_.push_back(allocator_session);
-}
-
-void ConnectivityChecker::RegisterHttpStart(int port) {
-  // Since we don't know what nic were actually used for the http request,
-  // for now, just use the first one.
-  std::vector<rtc::Network*> networks;
-  network_manager_->GetNetworks(&networks);
-  if (networks.empty()) {
-    LOG(LS_ERROR) << "No networks while registering http start.";
-    return;
-  }
-  rtc::ProxyInfo proxy_info = GetProxyInfo();
-  NicMap::iterator i =
-      nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
-  if (i != nics_.end()) {
-    uint32 now = rtc::Time();
-    NicInfo* nic_info = &i->second;
-    if (port == rtc::HTTP_SECURE_PORT) {
-      nic_info->https.start_time_ms = now;
-    } else {
-      LOG(LS_ERROR) << "Registering start time for unknown port: " << port;
-    }
-  } else {
-    LOG(LS_ERROR) << "Error, no nic info found while registering http start.";
-  }
-}
-
-}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker.h b/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker.h
deleted file mode 100644
index 427749e5b5..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker.h
+++ /dev/null
@@ -1,281 +0,0 @@
-/*
- *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_
-#define WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_
-
-#include <map>
-#include <string>
-
-#include "webrtc/p2p/base/basicpacketsocketfactory.h"
-#include "webrtc/p2p/client/httpportallocator.h"
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/messagehandler.h"
-#include "webrtc/base/network.h"
-#include "webrtc/base/proxyinfo.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/sigslot.h"
-#include "webrtc/base/socketaddress.h"
-
-namespace rtc {
-class AsyncHttpRequest;
-class AutoDetectProxy;
-class BasicPacketSocketFactory;
-class NetworkManager;
-class PacketSocketFactory;
-class SignalThread;
-class TestHttpPortAllocatorSession;
-class Thread;
-}
-
-namespace cricket {
-class HttpPortAllocator;
-class Port;
-class PortAllocatorSession;
-struct PortConfiguration;
-class RelayPort;
-class StunPort;
-
-// Contains details about a discovered firewall that are of interest
-// when debugging call failures.
-struct FirewallInfo {
-  std::string brand;
-  std::string model;
-
-  // TODO: List of current port mappings.
-};
-
-// Contains details about a specific connect attempt.
-struct ConnectInfo {
-  ConnectInfo()
-      : rtt(-1), error(0) {}
-  // Time when the connection was initiated. Needed for calculating
-  // the round trip time.
-  uint32 start_time_ms;
-  // Round trip time in milliseconds or -1 for failed connection.
-  int32 rtt;
-  // Error code representing low level errors like socket errors.
-  int error;
-};
-
-// Identifier for a network interface and proxy address pair.
-struct NicId {
-  NicId(const rtc::IPAddress& ip,
-        const rtc::SocketAddress& proxy_address)
-      : ip(ip),
-        proxy_address(proxy_address) {
-  }
-  rtc::IPAddress ip;
-  rtc::SocketAddress proxy_address;
-};
-
-// Comparator implementation identifying unique network interface and
-// proxy address pairs.
-class NicIdComparator {
- public:
-  int compare(const NicId &first, const NicId &second) const {
-    if (first.ip == second.ip) {
-      // Compare proxy address.
-      if (first.proxy_address == second.proxy_address) {
-        return 0;
-      } else {
-        return first.proxy_address < second.proxy_address? -1 : 1;
-      }
-    }
-    return first.ip < second.ip ? -1 : 1;
-  }
-
-  bool operator()(const NicId &first, const NicId &second) const {
-    return (compare(first, second) < 0);
-  }
-};
-
-// Contains information of a network interface and proxy address pair.
-struct NicInfo {
-  NicInfo() {}
-  rtc::IPAddress ip;
-  rtc::ProxyInfo proxy_info;
-  rtc::SocketAddress external_address;
-  ServerAddresses stun_server_addresses;
-  rtc::SocketAddress media_server_address;
-  ConnectInfo stun;
-  ConnectInfo http;
-  ConnectInfo https;
-  ConnectInfo udp;
-  ConnectInfo tcp;
-  ConnectInfo ssltcp;
-  FirewallInfo firewall;
-};
-
-// Holds the result of the connectivity check.
-class NicMap : public std::map<NicId, NicInfo, NicIdComparator> {
-};
-
-class TestHttpPortAllocatorSession : public HttpPortAllocatorSession {
- public:
-  TestHttpPortAllocatorSession(
-      HttpPortAllocator* allocator,
-      const std::string& content_name,
-      int component,
-      const std::string& ice_ufrag,
-      const std::string& ice_pwd,
-      const std::vector<rtc::SocketAddress>& stun_hosts,
-      const std::vector<std::string>& relay_hosts,
-      const std::string& relay_token,
-      const std::string& user_agent)
-      : HttpPortAllocatorSession(
-          allocator, content_name, component, ice_ufrag, ice_pwd, stun_hosts,
-          relay_hosts, relay_token, user_agent) {
-  }
-  void set_proxy(const rtc::ProxyInfo& proxy) {
-    proxy_ = proxy;
-  }
-
-  void ConfigReady(PortConfiguration* config);
-
-  void OnRequestDone(rtc::SignalThread* data);
-
-  sigslot::signal4<const std::string&, const std::string&,
-                   const PortConfiguration*,
-                   const rtc::ProxyInfo&> SignalConfigReady;
-  sigslot::signal1<rtc::AsyncHttpRequest*> SignalRequestDone;
-
- private:
-  rtc::ProxyInfo proxy_;
-};
-
-// Runs a request/response check on all network interface and proxy
-// address combinations. The check is considered done either when all
-// checks has been successful or when the check times out.
-class ConnectivityChecker
-    : public rtc::MessageHandler, public sigslot::has_slots<> {
- public:
-  ConnectivityChecker(rtc::Thread* worker,
-                      const std::string& jid,
-                      const std::string& session_id,
-                      const std::string& user_agent,
-                      const std::string& relay_token,
-                      const std::string& connection);
-  virtual ~ConnectivityChecker();
-
-  // Virtual for gMock.
-  virtual bool Initialize();
-  virtual void Start();
-
-  // MessageHandler implementation.
-  virtual void OnMessage(rtc::Message *msg);
-
-  // Instruct checker to stop and wait until that's done.
-  // Virtual for gMock.
-  virtual void Stop() {
-    worker_->Stop();
-  }
-
-  const NicMap& GetResults() const {
-    return nics_;
-  }
-
-  void set_timeout_ms(uint32 timeout) {
-    timeout_ms_ = timeout;
-  }
-
-  void set_stun_address(const rtc::SocketAddress& stun_address) {
-    stun_address_ = stun_address;
-  }
-
-  const std::string& connection() const {
-    return connection_;
-  }
-
-  const std::string& jid() const {
-    return jid_;
-  }
-
-  const std::string& session_id() const {
-    return session_id_;
-  }
-
-  // Context: Main Thread. Signalled when the connectivity check is complete.
-  sigslot::signal1<ConnectivityChecker*> SignalCheckDone;
-
- protected:
-  // Can be overridden for test.
-  virtual rtc::NetworkManager* CreateNetworkManager() {
-    return new rtc::BasicNetworkManager();
-  }
-  virtual rtc::BasicPacketSocketFactory* CreateSocketFactory(
-      rtc::Thread* thread) {
-    return new rtc::BasicPacketSocketFactory(thread);
-  }
-  virtual HttpPortAllocator* CreatePortAllocator(
-      rtc::NetworkManager* network_manager,
-      const std::string& user_agent,
-      const std::string& relay_token);
-  virtual StunPort* CreateStunPort(
-      const std::string& username, const std::string& password,
-      const PortConfiguration* config, rtc::Network* network);
-  virtual RelayPort* CreateRelayPort(
-      const std::string& username, const std::string& password,
-      const PortConfiguration* config, rtc::Network* network);
-  virtual void InitiateProxyDetection();
-  virtual void SetProxyInfo(const rtc::ProxyInfo& info);
-  virtual rtc::ProxyInfo GetProxyInfo() const;
-
-  rtc::Thread* worker() {
-    return worker_;
-  }
-
- private:
-  bool AddNic(const rtc::IPAddress& ip,
-              const rtc::SocketAddress& proxy_address);
-  void AllocatePorts();
-  void AllocateRelayPorts();
-  void CheckNetworks();
-  void CreateRelayPorts(
-      const std::string& username, const std::string& password,
-      const PortConfiguration* config, const rtc::ProxyInfo& proxy_info);
-
-  // Must be called by the worker thread.
-  void CleanUp();
-
-  void OnRequestDone(rtc::AsyncHttpRequest* request);
-  void OnRelayPortComplete(Port* port);
-  void OnStunPortComplete(Port* port);
-  void OnRelayPortError(Port* port);
-  void OnStunPortError(Port* port);
-  void OnNetworksChanged();
-  void OnProxyDetect(rtc::SignalThread* thread);
-  void OnConfigReady(
-      const std::string& username, const std::string& password,
-      const PortConfiguration* config, const rtc::ProxyInfo& proxy);
-  void OnConfigWithProxyReady(const PortConfiguration*);
-  void RegisterHttpStart(int port);
-  rtc::Thread* worker_;
-  std::string jid_;
-  std::string session_id_;
-  std::string user_agent_;
-  std::string relay_token_;
-  std::string connection_;
-  rtc::AutoDetectProxy* proxy_detect_;
-  rtc::scoped_ptr<rtc::NetworkManager> network_manager_;
-  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> socket_factory_;
-  rtc::scoped_ptr<HttpPortAllocator> port_allocator_;
-  NicMap nics_;
-  std::vector<Port*> ports_;
-  std::vector<PortAllocatorSession*> sessions_;
-  uint32 timeout_ms_;
-  rtc::SocketAddress stun_address_;
-  rtc::Thread* main_;
-  bool started_;
-};
-
-}  // namespace cricket
-
-#endif  // WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker_unittest.cc b/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker_unittest.cc
deleted file mode 100644
index 71c848397d..0000000000
--- a/media/webrtc/trunk/webrtc/p2p/client/connectivitychecker_unittest.cc
+++ /dev/null
@@ -1,367 +0,0 @@
-/*
- *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-
-#include "webrtc/p2p/base/basicpacketsocketfactory.h"
-#include "webrtc/p2p/base/relayport.h"
-#include "webrtc/p2p/base/stunport.h"
-#include "webrtc/p2p/client/connectivitychecker.h"
-#include "webrtc/p2p/client/httpportallocator.h"
-#include "webrtc/base/asynchttprequest.h"
-#include "webrtc/base/fakenetwork.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/socketaddress.h"
-
-namespace cricket {
-
-static const rtc::SocketAddress kClientAddr1("11.11.11.11", 0);
-static const rtc::SocketAddress kClientAddr2("22.22.22.22", 0);
-static const rtc::SocketAddress kExternalAddr("33.33.33.33", 3333);
-static const rtc::SocketAddress kStunAddr("44.44.44.44", 4444);
-static const rtc::SocketAddress kRelayAddr("55.55.55.55", 5555);
-static const rtc::SocketAddress kProxyAddr("66.66.66.66", 6666);
-static const rtc::ProxyType kProxyType = rtc::PROXY_HTTPS;
-static const char kRelayHost[] = "relay.google.com";
-static const char kRelayToken[] =
-    "CAESFwoOb2phQGdvb2dsZS5jb20Q043h47MmGhBTB1rbfIXkhuarDCZe+xF6";
-static const char kBrowserAgent[] = "browser_test";
-static const char kJid[] = "a.b@c";
-static const char kUserName[] = "testuser";
-static const char kPassword[] = "testpassword";
-static const char kMagicCookie[] = "testcookie";
-static const char kRelayUdpPort[] = "4444";
-static const char kRelayTcpPort[] = "5555";
-static const char kRelaySsltcpPort[] = "6666";
-static const char kSessionId[] = "testsession";
-static const char kConnection[] = "testconnection";
-static const int kMinPort = 1000;
-static const int kMaxPort = 2000;
-
-// Fake implementation to mock away real network usage.
-class FakeRelayPort : public RelayPort {
- public:
-  FakeRelayPort(rtc::Thread* thread,
-                rtc::PacketSocketFactory* factory,
-                rtc::Network* network, const rtc::IPAddress& ip,
-                int min_port, int max_port,
-                const std::string& username, const std::string& password)
-      : RelayPort(thread, factory, network, ip, min_port, max_port,
-                  username, password) {
-  }
-
-  // Just signal that we are done.
-  virtual void PrepareAddress() {
-    SignalPortComplete(this);
-  }
-};
-
-// Fake implementation to mock away real network usage.
-class FakeStunPort : public StunPort {
- public:
-  FakeStunPort(rtc::Thread* thread,
-               rtc::PacketSocketFactory* factory,
-               rtc::Network* network,
-               const rtc::IPAddress& ip,
-               int min_port, int max_port,
-               const std::string& username, const std::string& password,
-               const ServerAddresses& server_addr)
-      : StunPort(thread, factory, network, ip, min_port, max_port,
-                 username, password, server_addr, std::string()) {
-  }
-
-  // Just set external address and signal that we are done.
-  virtual void PrepareAddress() {
-    AddAddress(kExternalAddr, kExternalAddr, rtc::SocketAddress(), "udp", "",
-               STUN_PORT_TYPE, ICE_TYPE_PREFERENCE_SRFLX, 0, true);
-    SignalPortComplete(this);
-  }
-};
-
-// Fake implementation to mock away real network usage by responding
-// to http requests immediately.
-class FakeHttpPortAllocatorSession : public TestHttpPortAllocatorSession {
- public:
-  FakeHttpPortAllocatorSession(
-      HttpPortAllocator* allocator,
-      const std::string& content_name,
-      int component,
-      const std::string& ice_ufrag, const std::string& ice_pwd,
-      const std::vector<rtc::SocketAddress>& stun_hosts,
-      const std::vector<std::string>& relay_hosts,
-      const std::string& relay_token,
-      const std::string& agent)
-      : TestHttpPortAllocatorSession(allocator,
-                                     content_name,
-                                     component,
-                                     ice_ufrag,
-                                     ice_pwd,
-                                     stun_hosts,
-                                     relay_hosts,
-                                     relay_token,
-                                     agent) {
-  }
-  virtual void SendSessionRequest(const std::string& host, int port) {
-    FakeReceiveSessionResponse(host, port);
-  }
-
-  // Pass results to the real implementation.
-  void FakeReceiveSessionResponse(const std::string& host, int port) {
-    rtc::AsyncHttpRequest* response = CreateAsyncHttpResponse(port);
-    TestHttpPortAllocatorSession::OnRequestDone(response);
-    response->Destroy(true);
-  }
-
- private:
-  // Helper method for creating a response to a relay session request.
-  rtc::AsyncHttpRequest* CreateAsyncHttpResponse(int port) {
-    rtc::AsyncHttpRequest* request =
-        new rtc::AsyncHttpRequest(kBrowserAgent);
-    std::stringstream ss;
-    ss << "username=" << kUserName << std::endl
-       << "password=" << kPassword << std::endl
-       << "magic_cookie=" << kMagicCookie << std::endl
-       << "relay.ip=" << kRelayAddr.ipaddr().ToString() << std::endl
-       << "relay.udp_port=" << kRelayUdpPort << std::endl
-       << "relay.tcp_port=" << kRelayTcpPort << std::endl
-       << "relay.ssltcp_port=" << kRelaySsltcpPort << std::endl;
-    request->response().document.reset(
-        new rtc::MemoryStream(ss.str().c_str()));
-    request->response().set_success();
-    request->set_port(port);
-    request->set_secure(port == rtc::HTTP_SECURE_PORT);
-    return request;
-  }
-};
-
-// Fake implementation for creating fake http sessions.
-class FakeHttpPortAllocator : public HttpPortAllocator {
- public:
-  FakeHttpPortAllocator(rtc::NetworkManager* network_manager,
-                        const std::string& user_agent)
-      : HttpPortAllocator(network_manager, user_agent) {
-  }
-
-  virtual PortAllocatorSession* CreateSessionInternal(
-      const std::string& content_name, int component,
-      const std::string& ice_ufrag, const std::string& ice_pwd) {
-    std::vector<rtc::SocketAddress> stun_hosts;
-    stun_hosts.push_back(kStunAddr);
-    std::vector<std::string> relay_hosts;
-    relay_hosts.push_back(kRelayHost);
-    return new FakeHttpPortAllocatorSession(this,
-                                            content_name,
-                                            component,
-                                            ice_ufrag,
-                                            ice_pwd,
-                                            stun_hosts,
-                                            relay_hosts,
-                                            kRelayToken,
-                                            kBrowserAgent);
-  }
-};
-
-class ConnectivityCheckerForTest : public ConnectivityChecker {
- public:
-  ConnectivityCheckerForTest(rtc::Thread* worker,
-                             const std::string& jid,
-                             const std::string& session_id,
-                             const std::string& user_agent,
-                             const std::string& relay_token,
-                             const std::string& connection)
-      : ConnectivityChecker(worker,
-                            jid,
-                            session_id,
-                            user_agent,
-                            relay_token,
-                            connection),
-        proxy_initiated_(false) {
-  }
-
-  rtc::FakeNetworkManager* network_manager() const {
-    return network_manager_;
-  }
-
-  FakeHttpPortAllocator* port_allocator() const {
-    return fake_port_allocator_;
-  }
-
- protected:
-  // Overridden methods for faking a real network.
-  virtual rtc::NetworkManager* CreateNetworkManager() {
-    network_manager_ = new rtc::FakeNetworkManager();
-    return network_manager_;
-  }
-  virtual rtc::BasicPacketSocketFactory* CreateSocketFactory(
-      rtc::Thread* thread) {
-    // Create socket factory, for simplicity, let it run on the current thread.
-    socket_factory_ =
-        new rtc::BasicPacketSocketFactory(rtc::Thread::Current());
-    return socket_factory_;
-  }
-  virtual HttpPortAllocator* CreatePortAllocator(
-      rtc::NetworkManager* network_manager,
-      const std::string& user_agent,
-      const std::string& relay_token) {
-    fake_port_allocator_ =
-        new FakeHttpPortAllocator(network_manager, user_agent);
-    return fake_port_allocator_;
-  }
-  virtual StunPort* CreateStunPort(
-      const std::string& username, const std::string& password,
-      const PortConfiguration* config, rtc::Network* network) {
-    return new FakeStunPort(worker(),
-                            socket_factory_,
-                            network,
-                            network->GetBestIP(),
-                            kMinPort,
-                            kMaxPort,
-                            username,
-                            password,
-                            config->stun_servers);
-  }
-  virtual RelayPort* CreateRelayPort(
-      const std::string& username, const std::string& password,
-      const PortConfiguration* config, rtc::Network* network) {
-    return new FakeRelayPort(worker(),
-                             socket_factory_,
-                             network,
-                             network->GetBestIP(),
-                             kMinPort,
-                             kMaxPort,
-                             username,
-                             password);
-  }
-  virtual void InitiateProxyDetection() {
-    if (!proxy_initiated_) {
-      proxy_initiated_ = true;
-      proxy_info_.address = kProxyAddr;
-      proxy_info_.type = kProxyType;
-      SetProxyInfo(proxy_info_);
-    }
-  }
-
-  virtual rtc::ProxyInfo GetProxyInfo() const {
-    return proxy_info_;
-  }
-
- private:
-  rtc::BasicPacketSocketFactory* socket_factory_;
-  FakeHttpPortAllocator* fake_port_allocator_;
-  rtc::FakeNetworkManager* network_manager_;
-  rtc::ProxyInfo proxy_info_;
-  bool proxy_initiated_;
-};
-
-class ConnectivityCheckerTest : public testing::Test {
- protected:
-  void VerifyNic(const NicInfo& info,
-                 const rtc::SocketAddress& local_address) {
-    // Verify that the external address has been set.
-    EXPECT_EQ(kExternalAddr, info.external_address);
-
-    // Verify that the stun server address has been set.
-    EXPECT_EQ(1U, info.stun_server_addresses.size());
-    EXPECT_EQ(kStunAddr, *(info.stun_server_addresses.begin()));
-
-    // Verify that the media server address has been set. Don't care
-    // about port since it is different for different protocols.
-    EXPECT_EQ(kRelayAddr.ipaddr(), info.media_server_address.ipaddr());
-
-    // Verify that local ip matches.
-    EXPECT_EQ(local_address.ipaddr(), info.ip);
-
-    // Verify that we have received responses for our
-    // pings. Unsuccessful ping has rtt value -1, successful >= 0.
-    EXPECT_GE(info.stun.rtt, 0);
-    EXPECT_GE(info.udp.rtt, 0);
-    EXPECT_GE(info.tcp.rtt, 0);
-    EXPECT_GE(info.ssltcp.rtt, 0);
-
-    // If proxy has been set, verify address and type.
-    if (!info.proxy_info.address.IsNil()) {
-      EXPECT_EQ(kProxyAddr, info.proxy_info.address);
-      EXPECT_EQ(kProxyType, info.proxy_info.type);
-    }
-  }
-};
-
-// Tests a configuration with two network interfaces. Verifies that 4
-// combinations of ip/proxy are created and that all protocols are
-// tested on each combination.
-TEST_F(ConnectivityCheckerTest, TestStart) {
-  ConnectivityCheckerForTest connectivity_checker(rtc::Thread::Current(),
-                                                  kJid,
-                                                  kSessionId,
-                                                  kBrowserAgent,
-                                                  kRelayToken,
-                                                  kConnection);
-  connectivity_checker.Initialize();
-  connectivity_checker.set_stun_address(kStunAddr);
-  connectivity_checker.network_manager()->AddInterface(kClientAddr1);
-  connectivity_checker.network_manager()->AddInterface(kClientAddr2);
-
-  connectivity_checker.Start();
-  rtc::Thread::Current()->ProcessMessages(1000);
-
-  NicMap nics = connectivity_checker.GetResults();
-
-  // There should be 4 nics in our map. 2 for each interface added,
-  // one with proxy set and one without.
-  EXPECT_EQ(4U, nics.size());
-
-  // First verify interfaces without proxy.
-  rtc::SocketAddress nilAddress;
-
-  // First lookup the address of the first nic combined with no proxy.
-  NicMap::iterator i = nics.find(NicId(kClientAddr1.ipaddr(), nilAddress));
-  ASSERT(i != nics.end());
-  NicInfo info = i->second;
-  VerifyNic(info, kClientAddr1);
-
-  // Then make sure the second device has been tested without proxy.
-  i = nics.find(NicId(kClientAddr2.ipaddr(), nilAddress));
-  ASSERT(i != nics.end());
-  info = i->second;
-  VerifyNic(info, kClientAddr2);
-
-  // Now verify both interfaces with proxy.
-  i = nics.find(NicId(kClientAddr1.ipaddr(), kProxyAddr));
-  ASSERT(i != nics.end());
-  info = i->second;
-  VerifyNic(info, kClientAddr1);
-
-  i = nics.find(NicId(kClientAddr2.ipaddr(), kProxyAddr));
-  ASSERT(i != nics.end());
-  info = i->second;
-  VerifyNic(info, kClientAddr2);
-};
-
-// Tests that nothing bad happens if thera are no network interfaces
-// available to check.
-TEST_F(ConnectivityCheckerTest, TestStartNoNetwork) {
-  ConnectivityCheckerForTest connectivity_checker(rtc::Thread::Current(),
-                                                  kJid,
-                                                  kSessionId,
-                                                  kBrowserAgent,
-                                                  kRelayToken,
-                                                  kConnection);
-  connectivity_checker.Initialize();
-  connectivity_checker.Start();
-  rtc::Thread::Current()->ProcessMessages(1000);
-
-  NicMap nics = connectivity_checker.GetResults();
-
-  // Verify that no nics where checked.
-  EXPECT_EQ(0U, nics.size());
-}
-
-}  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/client/fakeportallocator.h b/media/webrtc/trunk/webrtc/p2p/client/fakeportallocator.h
index 89093cf4d5..fb188261a2 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/fakeportallocator.h
+++ b/media/webrtc/trunk/webrtc/p2p/client/fakeportallocator.h
@@ -24,6 +24,62 @@ class Thread;
 
 namespace cricket {
 
+class TestUDPPort : public UDPPort {
+ public:
+  static TestUDPPort* Create(rtc::Thread* thread,
+                             rtc::PacketSocketFactory* factory,
+                             rtc::Network* network,
+                             const rtc::IPAddress& ip,
+                             uint16_t min_port,
+                             uint16_t max_port,
+                             const std::string& username,
+                             const std::string& password,
+                             const std::string& origin,
+                             bool emit_localhost_for_anyaddress) {
+    TestUDPPort* port = new TestUDPPort(thread, factory, network, ip, min_port,
+                                        max_port, username, password, origin,
+                                        emit_localhost_for_anyaddress);
+    if (!port->Init()) {
+      delete port;
+      port = nullptr;
+    }
+    return port;
+  }
+  void SendBindingResponse(StunMessage* request,
+                           const rtc::SocketAddress& addr) override {
+    UDPPort::SendBindingResponse(request, addr);
+    sent_binding_response_ = true;
+  }
+  bool sent_binding_response() { return sent_binding_response_; }
+  void set_sent_binding_response(bool response) {
+    sent_binding_response_ = response;
+  }
+
+ protected:
+  TestUDPPort(rtc::Thread* thread,
+              rtc::PacketSocketFactory* factory,
+              rtc::Network* network,
+              const rtc::IPAddress& ip,
+              uint16_t min_port,
+              uint16_t max_port,
+              const std::string& username,
+              const std::string& password,
+              const std::string& origin,
+              bool emit_localhost_for_anyaddress)
+      : UDPPort(thread,
+                factory,
+                network,
+                ip,
+                min_port,
+                max_port,
+                username,
+                password,
+                origin,
+                emit_localhost_for_anyaddress) {}
+
+  bool sent_binding_response_ = false;
+};
+
 class FakePortAllocatorSession : public PortAllocatorSession {
  public:
   FakePortAllocatorSession(rtc::Thread* worker_thread,
@@ -33,7 +89,7 @@ class FakePortAllocatorSession : public PortAllocatorSession {
                            const std::string& ice_ufrag,
                            const std::string& ice_pwd)
       : PortAllocatorSession(content_name, component, ice_ufrag, ice_pwd,
-                             cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG),
+                             cricket::kDefaultPortAllocatorFlags),
         worker_thread_(worker_thread),
         factory_(factory),
         network_("network", "unittest",
@@ -45,15 +101,9 @@ class FakePortAllocatorSession : public PortAllocatorSession {
 
   virtual void StartGettingPorts() {
     if (!port_) {
-      port_.reset(cricket::UDPPort::Create(worker_thread_,
-                                           factory_,
-                                           &network_,
-                                           network_.GetBestIP(),
-                                           0,
-                                           0,
-                                           username(),
-                                           password(),
-                                           std::string()));
+      port_.reset(TestUDPPort::Create(worker_thread_, factory_, &network_,
+                                      network_.GetBestIP(), 0, 0, username(),
+                                      password(), std::string(), false));
       AddPort(port_.get());
     }
     ++port_config_count_;
@@ -62,6 +112,8 @@ class FakePortAllocatorSession : public PortAllocatorSession {
 
   virtual void StopGettingPorts() { running_ = false; }
   virtual bool IsGettingPorts() { return running_; }
+  virtual void ClearGettingPorts() {}
+
   int port_config_count() { return port_config_count_; }
 
   void AddPort(cricket::Port* port) {
@@ -98,11 +150,26 @@ class FakePortAllocator : public cricket::PortAllocator {
     }
   }
 
+  void SetIceServers(
+      const ServerAddresses& stun_servers,
+      const std::vector<RelayServerConfig>& turn_servers) override {
+    stun_servers_ = stun_servers;
+    turn_servers_ = turn_servers;
+  }
+
+  void SetNetworkIgnoreMask(int network_ignore_mask) override {}
+
+  const ServerAddresses& stun_servers() const { return stun_servers_; }
+
+  const std::vector<RelayServerConfig>& turn_servers() const {
+    return turn_servers_;
+  }
+
   virtual cricket::PortAllocatorSession* CreateSessionInternal(
       const std::string& content_name,
       int component,
       const std::string& ice_ufrag,
-      const std::string& ice_pwd) {
+      const std::string& ice_pwd) override {
     return new FakePortAllocatorSession(
         worker_thread_, factory_, content_name, component, ice_ufrag, ice_pwd);
   }
@@ -111,6 +178,8 @@ class FakePortAllocator : public cricket::PortAllocator {
   rtc::Thread* worker_thread_;
   rtc::PacketSocketFactory* factory_;
   rtc::scoped_ptr<rtc::BasicPacketSocketFactory> owned_factory_;
+  ServerAddresses stun_servers_;
+  std::vector<RelayServerConfig> turn_servers_;
 };
 
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.cc b/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.cc
index c072da27e1..1342cf70e9 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.cc
+++ b/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.cc
@@ -13,10 +13,9 @@
 #include <algorithm>
 #include <map>
 
-#include "webrtc/base/asynchttprequest.h"
-#include "webrtc/base/basicdefs.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
+#include "webrtc/base/httpcommon.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/signalthread.h"
@@ -144,7 +143,7 @@ void HttpPortAllocatorSessionBase::TryCreateRelaySession() {
     return;
   }
 
-  if (attempts_ == HttpPortAllocator::kNumRetries) {
+  if (attempts_ == HttpPortAllocatorBase::kNumRetries) {
     LOG(LS_ERROR) << "HttpPortAllocator: maximum number of requests reached; "
                   << "giving up on relay.";
     return;
@@ -167,13 +166,11 @@ void HttpPortAllocatorSessionBase::TryCreateRelaySession() {
 }
 
 std::string HttpPortAllocatorSessionBase::GetSessionRequestUrl() {
-  std::string url = std::string(HttpPortAllocator::kCreateSessionURL);
-  if (allocator()->flags() & PORTALLOCATOR_ENABLE_SHARED_UFRAG) {
-    ASSERT(!username().empty());
-    ASSERT(!password().empty());
-    url = url + "?username=" + rtc::s_url_encode(username()) +
-        "&password=" + rtc::s_url_encode(password());
-  }
+  std::string url = std::string(HttpPortAllocatorBase::kCreateSessionURL);
+  ASSERT(!username().empty());
+  ASSERT(!password().empty());
+  url = url + "?username=" + rtc::s_url_encode(username()) +
+      "&password=" + rtc::s_url_encode(password());
   return url;
 }
 
@@ -222,105 +219,4 @@ void HttpPortAllocatorSessionBase::ReceiveSessionResponse(
   ConfigReady(config);
 }
 
-// HttpPortAllocator
-
-HttpPortAllocator::HttpPortAllocator(
-    rtc::NetworkManager* network_manager,
-    rtc::PacketSocketFactory* socket_factory,
-    const std::string &user_agent)
-    : HttpPortAllocatorBase(network_manager, socket_factory, user_agent) {
-}
-
-HttpPortAllocator::HttpPortAllocator(
-    rtc::NetworkManager* network_manager,
-    const std::string &user_agent)
-    : HttpPortAllocatorBase(network_manager, user_agent) {
-}
-HttpPortAllocator::~HttpPortAllocator() {}
-
-PortAllocatorSession* HttpPortAllocator::CreateSessionInternal(
-    const std::string& content_name,
-    int component,
-    const std::string& ice_ufrag, const std::string& ice_pwd) {
-  return new HttpPortAllocatorSession(this, content_name, component,
-                                      ice_ufrag, ice_pwd, stun_hosts(),
-                                      relay_hosts(), relay_token(),
-                                      user_agent());
-}
-
-// HttpPortAllocatorSession
-
-HttpPortAllocatorSession::HttpPortAllocatorSession(
-    HttpPortAllocator* allocator,
-    const std::string& content_name,
-    int component,
-    const std::string& ice_ufrag,
-    const std::string& ice_pwd,
-    const std::vector<rtc::SocketAddress>& stun_hosts,
-    const std::vector<std::string>& relay_hosts,
-    const std::string& relay,
-    const std::string& agent)
-    : HttpPortAllocatorSessionBase(allocator, content_name, component,
-                                   ice_ufrag, ice_pwd, stun_hosts,
-                                   relay_hosts, relay, agent) {
-}
-
-HttpPortAllocatorSession::~HttpPortAllocatorSession() {
-  for (std::list<rtc::AsyncHttpRequest*>::iterator it = requests_.begin();
-       it != requests_.end(); ++it) {
-    (*it)->Destroy(true);
-  }
-}
-
-void HttpPortAllocatorSession::SendSessionRequest(const std::string& host,
-                                                  int port) {
-  // Initiate an HTTP request to create a session through the chosen host.
-  rtc::AsyncHttpRequest* request =
-      new rtc::AsyncHttpRequest(user_agent());
-  request->SignalWorkDone.connect(this,
-      &HttpPortAllocatorSession::OnRequestDone);
-
-  request->set_secure(port == rtc::HTTP_SECURE_PORT);
-  request->set_proxy(allocator()->proxy());
-  request->response().document.reset(new rtc::MemoryStream);
-  request->request().verb = rtc::HV_GET;
-  request->request().path = GetSessionRequestUrl();
-  request->request().addHeader("X-Talk-Google-Relay-Auth", relay_token(), true);
-  request->request().addHeader("X-Stream-Type", "video_rtp", true);
-  request->set_host(host);
-  request->set_port(port);
-  request->Start();
-  request->Release();
-
-  requests_.push_back(request);
-}
-
-void HttpPortAllocatorSession::OnRequestDone(rtc::SignalThread* data) {
-  rtc::AsyncHttpRequest* request =
-      static_cast<rtc::AsyncHttpRequest*>(data);
-
-  // Remove the request from the list of active requests.
-  std::list<rtc::AsyncHttpRequest*>::iterator it =
-      std::find(requests_.begin(), requests_.end(), request);
-  if (it != requests_.end()) {
-    requests_.erase(it);
-  }
-
-  if (request->response().scode != 200) {
-    LOG(LS_WARNING) << "HTTPPortAllocator: request "
-                    << " received error " << request->response().scode;
-    TryCreateRelaySession();
-    return;
-  }
-  LOG(LS_INFO) << "HTTPPortAllocator: request succeeded";
-
-  rtc::MemoryStream* stream =
-      static_cast<rtc::MemoryStream*>(request->response().document.get());
-  stream->Rewind();
-  size_t length;
-  stream->GetSize(&length);
-  std::string resp = std::string(stream->GetBuffer(), length);
-  ReceiveSessionResponse(resp);
-}
-
 }  // namespace cricket
diff --git a/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.h b/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.h
index e2fa74354d..e52765901f 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.h
+++ b/media/webrtc/trunk/webrtc/p2p/client/httpportallocator.h
@@ -26,6 +26,12 @@ class SignalThread;
 
 namespace cricket {
 
+// TODO(pthatcher): Remove this.  It's only used by chromoting, so we
+// should just move this code there.  It's used in these places in
+// chromium:
+// src/remoting/protocol/chromium_port_allocator.cc
+// src/remoting/client/plugin/pepper_port_allocator.cc
+// src/remoting/protocol/libjingle_transport_factory.cc
 class HttpPortAllocatorBase : public BasicPortAllocator {
  public:
   // The number of HTTP requests we should attempt before giving up.
@@ -130,44 +136,6 @@ class HttpPortAllocatorSessionBase : public BasicPortAllocatorSession {
   int attempts_;
 };
 
-class HttpPortAllocator : public HttpPortAllocatorBase {
- public:
-  HttpPortAllocator(rtc::NetworkManager* network_manager,
-                    const std::string& user_agent);
-  HttpPortAllocator(rtc::NetworkManager* network_manager,
-                    rtc::PacketSocketFactory* socket_factory,
-                    const std::string& user_agent);
-  virtual ~HttpPortAllocator();
-  virtual PortAllocatorSession* CreateSessionInternal(
-      const std::string& content_name,
-      int component,
-      const std::string& ice_ufrag, const std::string& ice_pwd);
-};
-
-class HttpPortAllocatorSession : public HttpPortAllocatorSessionBase {
- public:
-  HttpPortAllocatorSession(
-      HttpPortAllocator* allocator,
-      const std::string& content_name,
-      int component,
-      const std::string& ice_ufrag,
-      const std::string& ice_pwd,
-      const std::vector<rtc::SocketAddress>& stun_hosts,
-      const std::vector<std::string>& relay_hosts,
-      const std::string& relay,
-      const std::string& agent);
-  virtual ~HttpPortAllocatorSession();
-
-  virtual void SendSessionRequest(const std::string& host, int port);
-
- protected:
-  // Protected for diagnostics.
-  virtual void OnRequestDone(rtc::SignalThread* request);
-
- private:
-  std::list<rtc::AsyncHttpRequest*> requests_;
-};
-
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/client/portallocator_unittest.cc b/media/webrtc/trunk/webrtc/p2p/client/portallocator_unittest.cc
index b32d3124d0..5fce3b5762 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/portallocator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/p2p/client/portallocator_unittest.cc
@@ -11,7 +11,6 @@
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/constants.h"
 #include "webrtc/p2p/base/p2ptransportchannel.h"
-#include "webrtc/p2p/base/portallocatorsessionproxy.h"
 #include "webrtc/p2p/base/testrelayserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
@@ -21,6 +20,7 @@
 #include "webrtc/base/firewallsocketserver.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
+#include "webrtc/base/ipaddress.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/natserver.h"
 #include "webrtc/base/natsocketfactory.h"
@@ -32,16 +32,19 @@
 #include "webrtc/base/virtualsocketserver.h"
 
 using cricket::ServerAddresses;
+using rtc::IPAddress;
 using rtc::SocketAddress;
 using rtc::Thread;
 
 static const SocketAddress kClientAddr("11.11.11.11", 0);
+static const SocketAddress kLoopbackAddr("127.0.0.1", 0);
 static const SocketAddress kPrivateAddr("192.168.1.11", 0);
 static const SocketAddress kPrivateAddr2("192.168.1.12", 0);
 static const SocketAddress kClientIPv6Addr(
     "2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
 static const SocketAddress kClientAddr2("22.22.22.22", 0);
-static const SocketAddress kNatAddr("77.77.77.77", rtc::NAT_SERVER_PORT);
+static const SocketAddress kNatUdpAddr("77.77.77.77", rtc::NAT_SERVER_UDP_PORT);
+static const SocketAddress kNatTcpAddr("77.77.77.77", rtc::NAT_SERVER_TCP_PORT);
 static const SocketAddress kRemoteClientAddr("22.22.22.22", 0);
 static const SocketAddress kStunAddr("99.99.99.1", cricket::STUN_SERVER_PORT);
 static const SocketAddress kRelayUdpIntAddr("99.99.99.2", 5000);
@@ -86,8 +89,8 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
         vss_(new rtc::VirtualSocketServer(pss_.get())),
         fss_(new rtc::FirewallSocketServer(vss_.get())),
         ss_scope_(fss_.get()),
-        nat_factory_(vss_.get(), kNatAddr),
-        nat_socket_factory_(&nat_factory_),
+        nat_factory_(vss_.get(), kNatUdpAddr, kNatTcpAddr),
+        nat_socket_factory_(new rtc::BasicPacketSocketFactory(&nat_factory_)),
         stun_server_(cricket::TestStunServer::Create(Thread::Current(),
                                                      kStunAddr)),
         relay_server_(Thread::Current(), kRelayUdpIntAddr, kRelayUdpExtAddr,
@@ -109,43 +112,67 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
   void AddInterface(const SocketAddress& addr) {
     network_manager_.AddInterface(addr);
   }
+  void AddInterface(const SocketAddress& addr, const std::string& if_name) {
+    network_manager_.AddInterface(addr, if_name);
+  }
+  void AddInterface(const SocketAddress& addr,
+                    const std::string& if_name,
+                    rtc::AdapterType type) {
+    network_manager_.AddInterface(addr, if_name, type);
+  }
+  // The default route is the public address that STUN server will observe when
+  // the endpoint is sitting on the public internet and the local port is bound
+  // to the "any" address. This may be different from the default local address
+  // which the endpoint observes. This can occur if the route to the public
+  // endpoint like 8.8.8.8 (specified as the default local address) is
+  // different from the route to the STUN server (the default route).
+  void AddInterfaceAsDefaultRoute(const SocketAddress& addr) {
+    AddInterface(addr);
+    // When a binding comes from the any address, the |addr| will be used as the
+    // srflx address.
+    vss_->SetDefaultRoute(addr.ipaddr());
+  }
+  void RemoveInterface(const SocketAddress& addr) {
+    network_manager_.RemoveInterface(addr);
+  }
   bool SetPortRange(int min_port, int max_port) {
     return allocator_->SetPortRange(min_port, max_port);
   }
-  void ResetWithNatServer(const rtc::SocketAddress& stun_server) {
-    nat_server_.reset(new rtc::NATServer(
-        rtc::NAT_OPEN_CONE, vss_.get(), kNatAddr, vss_.get(), kNatAddr));
-
-    ServerAddresses stun_servers;
-    stun_servers.insert(stun_server);
-    allocator_.reset(new cricket::BasicPortAllocator(
-        &network_manager_, &nat_socket_factory_, stun_servers));
-    allocator().set_step_delay(cricket::kMinimumStepDelay);
-  }
-
-  // Create a BasicPortAllocator without GTURN and add the TURN servers.
-  void ResetWithTurnServers(const rtc::SocketAddress& udp_turn,
-                            const rtc::SocketAddress& tcp_turn) {
+  // Endpoint is on the public network. No STUN or TURN.
+  void ResetWithNoServersOrNat() {
     allocator_.reset(new cricket::BasicPortAllocator(&network_manager_));
-    allocator().set_step_delay(cricket::kMinimumStepDelay);
+    allocator_->set_step_delay(cricket::kMinimumStepDelay);
+  }
+  // Endpoint is behind a NAT, with STUN specified.
+  void ResetWithStunServerAndNat(const rtc::SocketAddress& stun_server) {
+    ResetWithStunServer(stun_server, true);
+  }
+  // Endpoint is on the public network, with STUN specified.
+  void ResetWithStunServerNoNat(const rtc::SocketAddress& stun_server) {
+    ResetWithStunServer(stun_server, false);
+  }
+  // Endpoint is on the public network, with TURN specified.
+  void ResetWithTurnServersNoNat(const rtc::SocketAddress& udp_turn,
+                                 const rtc::SocketAddress& tcp_turn) {
+    ResetWithNoServersOrNat();
     AddTurnServers(udp_turn, tcp_turn);
   }
 
   void AddTurnServers(const rtc::SocketAddress& udp_turn,
                       const rtc::SocketAddress& tcp_turn) {
-    cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
+    cricket::RelayServerConfig turn_server(cricket::RELAY_TURN);
     cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
-    relay_server.credentials = credentials;
+    turn_server.credentials = credentials;
 
     if (!udp_turn.IsNil()) {
-      relay_server.ports.push_back(cricket::ProtocolAddress(
-          kTurnUdpIntAddr, cricket::PROTO_UDP, false));
+      turn_server.ports.push_back(
+          cricket::ProtocolAddress(kTurnUdpIntAddr, cricket::PROTO_UDP, false));
     }
     if (!tcp_turn.IsNil()) {
-      relay_server.ports.push_back(cricket::ProtocolAddress(
-          kTurnTcpIntAddr, cricket::PROTO_TCP, false));
+      turn_server.ports.push_back(
+          cricket::ProtocolAddress(kTurnTcpIntAddr, cricket::PROTO_TCP, false));
     }
-    allocator_->AddRelay(relay_server);
+    allocator_->AddTurnServer(turn_server);
   }
 
   bool CreateSession(int component) {
@@ -228,27 +255,68 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
     }
   }
 
-  void CheckDisableAdapterEnumeration() {
-    EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
-    session_->set_flags(cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
+  // This function starts the port/address gathering and check the existence of
+  // candidates as specified. When |expect_stun_candidate| is true,
+  // |stun_candidate_addr| carries the expected reflective address, which is
+  // also the related address for TURN candidate if it is expected. Otherwise,
+  // it should be ignore.
+  void CheckDisableAdapterEnumeration(
+      uint32_t total_ports,
+      const rtc::IPAddress& host_candidate_addr,
+      const rtc::IPAddress& stun_candidate_addr,
+      const rtc::IPAddress& relay_candidate_udp_transport_addr,
+      const rtc::IPAddress& relay_candidate_tcp_transport_addr) {
+    network_manager_.set_default_local_addresses(kPrivateAddr.ipaddr(),
+                                                 rtc::IPAddress());
+    if (!session_) {
+      EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+    }
+    session_->set_flags(session_->flags() |
+                        cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION |
+                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+    allocator().set_allow_tcp_listen(false);
     session_->StartGettingPorts();
     EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
 
-    // Only 2 candidates as local UDP/TCP are all 0s and get trimmed out.
-    EXPECT_EQ(2U, candidates_.size());
-    EXPECT_EQ(2U, ports_.size());  // One stunport and one turnport.
+    uint32_t total_candidates = 0;
+    if (!host_candidate_addr.IsNil()) {
+      EXPECT_PRED5(CheckCandidate, candidates_[total_candidates],
+                   cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+                   rtc::SocketAddress(kPrivateAddr.ipaddr(), 0));
+      ++total_candidates;
+    }
+    if (!stun_candidate_addr.IsNil()) {
+      EXPECT_PRED5(CheckCandidate, candidates_[total_candidates],
+                   cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+                   rtc::SocketAddress(stun_candidate_addr, 0));
+      rtc::IPAddress related_address = host_candidate_addr;
+      if (host_candidate_addr.IsNil()) {
+        related_address =
+            rtc::GetAnyIP(candidates_[total_candidates].address().family());
+      }
+      EXPECT_EQ(related_address,
+                candidates_[total_candidates].related_address().ipaddr());
+      ++total_candidates;
+    }
+    if (!relay_candidate_udp_transport_addr.IsNil()) {
+      EXPECT_PRED5(CheckCandidate, candidates_[total_candidates],
+                   cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+                   rtc::SocketAddress(relay_candidate_udp_transport_addr, 0));
+      EXPECT_EQ(stun_candidate_addr,
+                candidates_[total_candidates].related_address().ipaddr());
+      ++total_candidates;
+    }
+    if (!relay_candidate_tcp_transport_addr.IsNil()) {
+      EXPECT_PRED5(CheckCandidate, candidates_[total_candidates],
+                   cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+                   rtc::SocketAddress(relay_candidate_tcp_transport_addr, 0));
+      EXPECT_EQ(stun_candidate_addr,
+                candidates_[total_candidates].related_address().ipaddr());
+      ++total_candidates;
+    }
 
-    EXPECT_PRED5(CheckCandidate, candidates_[0],
-                 cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-                 rtc::SocketAddress(kNatAddr.ipaddr(), 0));
-    EXPECT_EQ(
-        rtc::EmptySocketAddressWithFamily(candidates_[0].address().family()),
-        candidates_[0].related_address());
-
-    EXPECT_PRED5(CheckCandidate, candidates_[1],
-                 cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
-                 rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
-    EXPECT_EQ(kNatAddr.ipaddr(), candidates_[1].related_address().ipaddr());
+    EXPECT_EQ(total_candidates, candidates_.size());
+    EXPECT_EQ(total_ports, ports_.size());
   }
 
  protected:
@@ -270,8 +338,8 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
   }
 
   bool HasRelayAddress(const cricket::ProtocolAddress& proto_addr) {
-    for (size_t i = 0; i < allocator_->relays().size(); ++i) {
-      cricket::RelayServerConfig server_config = allocator_->relays()[i];
+    for (size_t i = 0; i < allocator_->turn_servers().size(); ++i) {
+      cricket::RelayServerConfig server_config = allocator_->turn_servers()[i];
       cricket::PortList::const_iterator relay_port;
       for (relay_port = server_config.ports.begin();
           relay_port != server_config.ports.end(); ++relay_port) {
@@ -283,13 +351,32 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
     return false;
   }
 
+  void ResetWithStunServer(const rtc::SocketAddress& stun_server,
+                           bool with_nat) {
+    if (with_nat) {
+      nat_server_.reset(new rtc::NATServer(
+          rtc::NAT_OPEN_CONE, vss_.get(), kNatUdpAddr, kNatTcpAddr, vss_.get(),
+          rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0)));
+    } else {
+      nat_socket_factory_.reset(new rtc::BasicPacketSocketFactory());
+    }
+
+    ServerAddresses stun_servers;
+    if (!stun_server.IsNil()) {
+      stun_servers.insert(stun_server);
+    }
+    allocator_.reset(new cricket::BasicPortAllocator(
+        &network_manager_, nat_socket_factory_.get(), stun_servers));
+    allocator().set_step_delay(cricket::kMinimumStepDelay);
+  }
+
   rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
   rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
   rtc::scoped_ptr<rtc::FirewallSocketServer> fss_;
   rtc::SocketServerScope ss_scope_;
   rtc::scoped_ptr<rtc::NATServer> nat_server_;
   rtc::NATSocketFactory nat_factory_;
-  rtc::BasicPacketSocketFactory nat_socket_factory_;
+  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> nat_socket_factory_;
   rtc::scoped_ptr<cricket::TestStunServer> stun_server_;
   cricket::TestRelayServer relay_server_;
   cricket::TestTurnServer turn_server_;
@@ -305,11 +392,11 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
 TEST_F(PortAllocatorTest, TestBasic) {
   EXPECT_EQ(&network_manager_, allocator().network_manager());
   EXPECT_EQ(kStunAddr, *allocator().stun_servers().begin());
-  ASSERT_EQ(1u, allocator().relays().size());
-  EXPECT_EQ(cricket::RELAY_GTURN, allocator().relays()[0].type);
+  ASSERT_EQ(1u, allocator().turn_servers().size());
+  EXPECT_EQ(cricket::RELAY_GTURN, allocator().turn_servers()[0].type);
   // Empty relay credentials are used for GTURN.
-  EXPECT_TRUE(allocator().relays()[0].credentials.username.empty());
-  EXPECT_TRUE(allocator().relays()[0].credentials.password.empty());
+  EXPECT_TRUE(allocator().turn_servers()[0].credentials.username.empty());
+  EXPECT_TRUE(allocator().turn_servers()[0].credentials.password.empty());
   EXPECT_TRUE(HasRelayAddress(cricket::ProtocolAddress(
       kRelayUdpIntAddr, cricket::PROTO_UDP)));
   EXPECT_TRUE(HasRelayAddress(cricket::ProtocolAddress(
@@ -319,6 +406,50 @@ TEST_F(PortAllocatorTest, TestBasic) {
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
 }
 
+// Tests that our network filtering works properly.
+TEST_F(PortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) {
+  AddInterface(SocketAddress(IPAddress(0x12345600U), 0), "test_eth0",
+               rtc::ADAPTER_TYPE_ETHERNET);
+  AddInterface(SocketAddress(IPAddress(0x12345601U), 0), "test_wlan0",
+               rtc::ADAPTER_TYPE_WIFI);
+  AddInterface(SocketAddress(IPAddress(0x12345602U), 0), "test_cell0",
+               rtc::ADAPTER_TYPE_CELLULAR);
+  AddInterface(SocketAddress(IPAddress(0x12345603U), 0), "test_vpn0",
+               rtc::ADAPTER_TYPE_VPN);
+  AddInterface(SocketAddress(IPAddress(0x12345604U), 0), "test_lo",
+               rtc::ADAPTER_TYPE_LOOPBACK);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_STUN |
+                      cricket::PORTALLOCATOR_DISABLE_RELAY |
+                      cricket::PORTALLOCATOR_DISABLE_TCP);
+  session_->StartGettingPorts();
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+  EXPECT_EQ(4U, candidates_.size());
+  for (cricket::Candidate candidate : candidates_) {
+    EXPECT_LT(candidate.address().ip(), 0x12345604U);
+  }
+}
+
+TEST_F(PortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) {
+  AddInterface(SocketAddress(IPAddress(0x12345600U), 0), "test_eth0",
+               rtc::ADAPTER_TYPE_ETHERNET);
+  AddInterface(SocketAddress(IPAddress(0x12345601U), 0), "test_wlan0",
+               rtc::ADAPTER_TYPE_WIFI);
+  AddInterface(SocketAddress(IPAddress(0x12345602U), 0), "test_cell0",
+               rtc::ADAPTER_TYPE_CELLULAR);
+  allocator_->SetNetworkIgnoreMask(rtc::ADAPTER_TYPE_ETHERNET |
+                                   rtc::ADAPTER_TYPE_LOOPBACK |
+                                   rtc::ADAPTER_TYPE_WIFI);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_STUN |
+                      cricket::PORTALLOCATOR_DISABLE_RELAY |
+                      cricket::PORTALLOCATOR_DISABLE_TCP);
+  session_->StartGettingPorts();
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+  EXPECT_EQ(1U, candidates_.size());
+  EXPECT_EQ(0x12345602U, candidates_[0].address().ip());
+}
+
 // Tests that we allocator session not trying to allocate ports for every 250ms.
 TEST_F(PortAllocatorTest, TestNoNetworkInterface) {
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -332,6 +463,19 @@ TEST_F(PortAllocatorTest, TestNoNetworkInterface) {
   EXPECT_EQ(0U, candidates_.size());
 }
 
+// Test that we could use loopback interface as host candidate.
+TEST_F(PortAllocatorTest, TestLoopbackNetworkInterface) {
+  AddInterface(kLoopbackAddr, "test_loopback", rtc::ADAPTER_TYPE_LOOPBACK);
+  allocator_->SetNetworkIgnoreMask(0);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_STUN |
+                      cricket::PORTALLOCATOR_DISABLE_RELAY |
+                      cricket::PORTALLOCATOR_DISABLE_TCP);
+  session_->StartGettingPorts();
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+  EXPECT_EQ(1U, candidates_.size());
+}
+
 // Tests that we can get all the desired addresses successfully.
 TEST_F(PortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
   AddInterface(kClientAddr);
@@ -357,6 +501,61 @@ TEST_F(PortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
+// Test that when the same network interface is brought down and up, the
+// port allocator session will restart a new allocation sequence if
+// it is not stopped.
+TEST_F(PortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
+  std::string if_name("test_net0");
+  AddInterface(kClientAddr, if_name);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+  ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(4U, ports_.size());
+  EXPECT_TRUE(candidate_allocation_done_);
+  candidate_allocation_done_ = false;
+  candidates_.clear();
+  ports_.clear();
+
+  RemoveInterface(kClientAddr);
+  ASSERT_EQ_WAIT(0U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(0U, ports_.size());
+  EXPECT_FALSE(candidate_allocation_done_);
+
+  // When the same interfaces are added again, new candidates/ports should be
+  // generated.
+  AddInterface(kClientAddr, if_name);
+  ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(4U, ports_.size());
+  EXPECT_TRUE(candidate_allocation_done_);
+}
+
+// Test that when the same network interface is brought down and up, the
+// port allocator session will not restart a new allocation sequence if
+// it is stopped.
+TEST_F(PortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
+  std::string if_name("test_net0");
+  AddInterface(kClientAddr, if_name);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+  ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(4U, ports_.size());
+  EXPECT_TRUE(candidate_allocation_done_);
+  session_->StopGettingPorts();
+  candidates_.clear();
+  ports_.clear();
+
+  RemoveInterface(kClientAddr);
+  ASSERT_EQ_WAIT(0U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(0U, ports_.size());
+
+  // When the same interfaces are added again, new candidates/ports should not
+  // be generated because the session has stopped.
+  AddInterface(kClientAddr, if_name);
+  ASSERT_EQ_WAIT(0U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(0U, ports_.size());
+  EXPECT_TRUE(candidate_allocation_done_);
+}
+
 // Verify candidates with default step delay of 1sec.
 TEST_F(PortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
   AddInterface(kClientAddr);
@@ -449,28 +648,143 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoAdapters) {
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
-// Test that we should only get STUN and TURN candidates when adapter
-// enumeration is disabled.
-TEST_F(PortAllocatorTest, TestDisableAdapterEnumeration) {
-  AddInterface(kClientAddr);
-  // GTURN is not configured here.
-  ResetWithNatServer(kStunAddr);
-  AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
-
-  CheckDisableAdapterEnumeration();
+// Test that when enumeration is disabled, we should not have any ports when
+// candidate_filter() is set to CF_RELAY and no relay is specified.
+TEST_F(PortAllocatorTest,
+       TestDisableAdapterEnumerationWithoutNatRelayTransportOnly) {
+  ResetWithStunServerNoNat(kStunAddr);
+  allocator().set_candidate_filter(cricket::CF_RELAY);
+  // Expect to see no ports and no candidates.
+  CheckDisableAdapterEnumeration(0U, rtc::IPAddress(), rtc::IPAddress(),
+                                 rtc::IPAddress(), rtc::IPAddress());
 }
 
-// Test that even with multiple interfaces, the result should be only 1 Stun
-// candidate since we bind to any address (i.e. all 0s).
-TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationMultipleInterfaces) {
+// Test that even with multiple interfaces, the result should still be a single
+// default private, one STUN and one TURN candidate since we bind to any address
+// (i.e. all 0s).
+TEST_F(PortAllocatorTest,
+       TestDisableAdapterEnumerationBehindNatMultipleInterfaces) {
   AddInterface(kPrivateAddr);
   AddInterface(kPrivateAddr2);
-  ResetWithNatServer(kStunAddr);
+  ResetWithStunServerAndNat(kStunAddr);
   AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
 
-  CheckDisableAdapterEnumeration();
+  // Enable IPv6 here. Since the network_manager doesn't have IPv6 default
+  // address set and we have no IPv6 STUN server, there should be no IPv6
+  // candidates.
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_ENABLE_IPV6);
+
+  // Expect to see 3 ports for IPv4: HOST/STUN, TURN/UDP and TCP ports, 2 ports
+  // for IPv6: HOST, and TCP. Only IPv4 candidates: a default private, STUN and
+  // TURN/UDP candidates.
+  CheckDisableAdapterEnumeration(5U, kPrivateAddr.ipaddr(),
+                                 kNatUdpAddr.ipaddr(), kTurnUdpExtAddr.ipaddr(),
+                                 rtc::IPAddress());
 }
 
+// Test that we should get a default private, STUN, TURN/UDP and TURN/TCP
+// candidates when both TURN/UDP and TURN/TCP servers are specified.
+TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationBehindNatWithTcp) {
+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+  AddInterface(kPrivateAddr);
+  ResetWithStunServerAndNat(kStunAddr);
+  AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr);
+  // Expect to see 4 ports - STUN, TURN/UDP, TURN/TCP and TCP port. A default
+  // private, STUN, TURN/UDP, and TURN/TCP candidates.
+  CheckDisableAdapterEnumeration(4U, kPrivateAddr.ipaddr(),
+                                 kNatUdpAddr.ipaddr(), kTurnUdpExtAddr.ipaddr(),
+                                 kTurnUdpExtAddr.ipaddr());
+}
+
+// Test that when adapter enumeration is disabled, for endpoints without
+// STUN/TURN specified, a default private candidate is still generated.
+TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationWithoutNatOrServers) {
+  ResetWithNoServersOrNat();
+  // Expect to see 2 ports: STUN and TCP ports, one default private candidate.
+  CheckDisableAdapterEnumeration(2U, kPrivateAddr.ipaddr(), rtc::IPAddress(),
+                                 rtc::IPAddress(), rtc::IPAddress());
+}
+
+// Test that when adapter enumeration is disabled, with
+// PORTALLOCATOR_DISABLE_LOCALHOST_CANDIDATE specified, for endpoints not behind
+// a NAT, there is no local candidate.
+TEST_F(PortAllocatorTest,
+       TestDisableAdapterEnumerationWithoutNatLocalhostCandidateDisabled) {
+  ResetWithStunServerNoNat(kStunAddr);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
+  // Expect to see 2 ports: STUN and TCP ports, localhost candidate and STUN
+  // candidate.
+  CheckDisableAdapterEnumeration(2U, rtc::IPAddress(), rtc::IPAddress(),
+                                 rtc::IPAddress(), rtc::IPAddress());
+}
+
+// Test that when adapter enumeration is disabled, with
+// PORTALLOCATOR_DISABLE_LOCALHOST_CANDIDATE specified, for endpoints not behind
+// a NAT, there is no local candidate. However, this specified default route
+// (kClientAddr) which was discovered when sending STUN requests, will become
+// the srflx addresses.
+TEST_F(
+    PortAllocatorTest,
+    TestDisableAdapterEnumerationWithoutNatLocalhostCandidateDisabledWithDifferentDefaultRoute) {
+  ResetWithStunServerNoNat(kStunAddr);
+  AddInterfaceAsDefaultRoute(kClientAddr);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
+  // Expect to see 2 ports: STUN and TCP ports, localhost candidate and STUN
+  // candidate.
+  CheckDisableAdapterEnumeration(2U, rtc::IPAddress(), kClientAddr.ipaddr(),
+                                 rtc::IPAddress(), rtc::IPAddress());
+}
+
+// Test that when adapter enumeration is disabled, with
+// PORTALLOCATOR_DISABLE_LOCALHOST_CANDIDATE specified, for endpoints behind a
+// NAT, there is only one STUN candidate.
+TEST_F(PortAllocatorTest,
+       TestDisableAdapterEnumerationWithNatLocalhostCandidateDisabled) {
+  ResetWithStunServerAndNat(kStunAddr);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
+  // Expect to see 2 ports: STUN and TCP ports, and single STUN candidate.
+  CheckDisableAdapterEnumeration(2U, rtc::IPAddress(), kNatUdpAddr.ipaddr(),
+                                 rtc::IPAddress(), rtc::IPAddress());
+}
+
+// Test that we disable relay over UDP, and only TCP is used when connecting to
+// the relay server.
+TEST_F(PortAllocatorTest, TestDisableUdpTurn) {
+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+  AddInterface(kClientAddr);
+  ResetWithStunServerAndNat(kStunAddr);
+  AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->set_flags(cricket::PORTALLOCATOR_DISABLE_UDP_RELAY |
+                      cricket::PORTALLOCATOR_DISABLE_UDP |
+                      cricket::PORTALLOCATOR_DISABLE_STUN |
+                      cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+
+  session_->StartGettingPorts();
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+
+  // Expect to see 2 ports and 2 candidates - TURN/TCP and TCP ports, TCP and
+  // TURN/TCP candidates.
+  EXPECT_EQ(2U, ports_.size());
+  EXPECT_EQ(2U, candidates_.size());
+  EXPECT_PRED5(CheckCandidate, candidates_[0],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kTurnUdpExtAddr);
+  // The TURN candidate should use TCP to contact the TURN server.
+  EXPECT_EQ(cricket::TCP_PROTOCOL_NAME, candidates_[0].relay_protocol());
+  EXPECT_PRED5(CheckCandidate, candidates_[1],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
+}
+
+// Disable for asan, see
+// https://code.google.com/p/webrtc/issues/detail?id=4743 for details.
+#if !defined(ADDRESS_SANITIZER)
+
 // Test that we can get OnCandidatesAllocationDone callback when all the ports
 // are disabled.
 TEST_F(PortAllocatorTest, TestDisableAllPorts) {
@@ -508,6 +822,8 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpSockets) {
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
+#endif // if !defined(ADDRESS_SANITIZER)
+
 // Test that we don't crash or malfunction if we can't create UDP sockets or
 // listen on TCP sockets. We still give out a local TCP address, since
 // apparently this is needed for the remote side to accept our connection.
@@ -611,7 +927,7 @@ TEST_F(PortAllocatorTest, TestGetAllPortsRestarts) {
 TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
   AddInterface(kClientAddr);
   // GTURN is not configured here.
-  ResetWithTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
+  ResetWithTurnServersNoNat(kTurnUdpIntAddr, rtc::SocketAddress());
   allocator().set_candidate_filter(cricket::CF_RELAY);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -635,8 +951,7 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
 
 TEST_F(PortAllocatorTest, TestCandidateFilterWithHostOnly) {
   AddInterface(kClientAddr);
-  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   allocator().set_candidate_filter(cricket::CF_HOST);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -651,10 +966,9 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithHostOnly) {
 // Host is behind the NAT.
 TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
   AddInterface(kPrivateAddr);
-  ResetWithNatServer(kStunAddr);
+  ResetWithStunServerAndNat(kStunAddr);
 
-  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   allocator().set_candidate_filter(cricket::CF_REFLEXIVE);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -674,8 +988,7 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
 // Host is not behind the NAT.
 TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
   AddInterface(kClientAddr);
-  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   allocator().set_candidate_filter(cricket::CF_REFLEXIVE);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -688,94 +1001,8 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
   }
 }
 
-TEST_F(PortAllocatorTest, TestBasicMuxFeatures) {
-  AddInterface(kClientAddr);
-  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_BUNDLE |
-      cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  // Session ID - session1.
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session1(
-      CreateSession("session1", cricket::ICE_CANDIDATE_COMPONENT_RTP));
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session2(
-      CreateSession("session1", cricket::ICE_CANDIDATE_COMPONENT_RTCP));
-  session1->StartGettingPorts();
-  session2->StartGettingPorts();
-  // Each session should receive two proxy ports of local and stun.
-  ASSERT_EQ_WAIT(14U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(8U, ports_.size());
-
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session3(
-      CreateSession("session1", cricket::ICE_CANDIDATE_COMPONENT_RTP));
-  session3->StartGettingPorts();
-  // Already allocated candidates and ports will be sent to the newly
-  // allocated proxy session.
-  ASSERT_EQ_WAIT(21U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(12U, ports_.size());
-}
-
-// This test verifies by changing ice_ufrag and/or ice_pwd
-// will result in different set of candidates when BUNDLE is enabled.
-// If BUNDLE is disabled, CreateSession will always allocate new
-// set of candidates.
-TEST_F(PortAllocatorTest, TestBundleIceRestart) {
-  AddInterface(kClientAddr);
-  allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_BUNDLE |
-      cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  // Session ID - session1.
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session1(
-      CreateSession("session1", kContentName,
-                    cricket::ICE_CANDIDATE_COMPONENT_RTP,
-                    kIceUfrag0, kIcePwd0));
-  session1->StartGettingPorts();
-  ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(4U, ports_.size());
-
-  // Allocate a different session with sid |session1| and different ice_ufrag.
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session2(
-      CreateSession("session1", kContentName,
-                    cricket::ICE_CANDIDATE_COMPONENT_RTP,
-                    "TestIceUfrag", kIcePwd0));
-  session2->StartGettingPorts();
-  ASSERT_EQ_WAIT(14U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(8U, ports_.size());
-  // Verifying the candidate address different from previously allocated
-  // address.
-  // Skipping verification of component id and candidate type.
-  EXPECT_NE(candidates_[0].address(), candidates_[7].address());
-  EXPECT_NE(candidates_[1].address(), candidates_[8].address());
-
-  // Allocating a different session with sid |session1| and
-  // different ice_pwd.
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session3(
-      CreateSession("session1", kContentName,
-                    cricket::ICE_CANDIDATE_COMPONENT_RTP,
-                    kIceUfrag0, "TestIcePwd"));
-  session3->StartGettingPorts();
-  ASSERT_EQ_WAIT(21U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(12U, ports_.size());
-  // Verifying the candidate address different from previously
-  // allocated address.
-  EXPECT_NE(candidates_[7].address(), candidates_[14].address());
-  EXPECT_NE(candidates_[8].address(), candidates_[15].address());
-
-  // Allocating a session with by changing both ice_ufrag and ice_pwd.
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session4(
-      CreateSession("session1", kContentName,
-                    cricket::ICE_CANDIDATE_COMPONENT_RTP,
-                    "TestIceUfrag", "TestIcePwd"));
-  session4->StartGettingPorts();
-  ASSERT_EQ_WAIT(28U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(16U, ports_.size());
-  // Verifying the candidate address different from previously
-  // allocated address.
-  EXPECT_NE(candidates_[14].address(), candidates_[21].address());
-  EXPECT_NE(candidates_[15].address(), candidates_[22].address());
-}
-
-// Test that when the PORTALLOCATOR_ENABLE_SHARED_UFRAG is enabled we got same
-// ufrag and pwd for the collected candidates.
+// Test that we get the same ufrag and pwd for all candidates.
 TEST_F(PortAllocatorTest, TestEnableSharedUfrag) {
-  allocator().set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG);
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -795,30 +1022,6 @@ TEST_F(PortAllocatorTest, TestEnableSharedUfrag) {
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
-// Test that when the PORTALLOCATOR_ENABLE_SHARED_UFRAG isn't enabled we got
-// different ufrag and pwd for the collected candidates.
-TEST_F(PortAllocatorTest, TestDisableSharedUfrag) {
-  allocator().set_flags(allocator().flags() &
-                        ~cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  AddInterface(kClientAddr);
-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
-  session_->StartGettingPorts();
-  ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
-  EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp", kClientAddr);
-  EXPECT_EQ(4U, ports_.size());
-  // Port should generate random ufrag and pwd.
-  EXPECT_NE(kIceUfrag0, candidates_[0].username());
-  EXPECT_NE(kIceUfrag0, candidates_[1].username());
-  EXPECT_NE(candidates_[0].username(), candidates_[1].username());
-  EXPECT_NE(kIcePwd0, candidates_[0].password());
-  EXPECT_NE(kIcePwd0, candidates_[1].password());
-  EXPECT_NE(candidates_[0].password(), candidates_[1].password());
-  EXPECT_TRUE(candidate_allocation_done_);
-}
-
 // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port
 // is allocated for udp and stun. Also verify there is only one candidate
 // (local) if stun candidate is same as local candidate, which will be the case
@@ -826,7 +1029,6 @@ TEST_F(PortAllocatorTest, TestDisableSharedUfrag) {
 TEST_F(PortAllocatorTest, TestSharedSocketWithoutNat) {
   AddInterface(kClientAddr);
   allocator_->set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -842,10 +1044,9 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithoutNat) {
 // local candidates as client behind a nat.
 TEST_F(PortAllocatorTest, TestSharedSocketWithNat) {
   AddInterface(kClientAddr);
-  ResetWithNatServer(kStunAddr);
+  ResetWithStunServerAndNat(kStunAddr);
 
   allocator_->set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -855,12 +1056,12 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNat) {
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-      rtc::SocketAddress(kNatAddr.ipaddr(), 0));
+      rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   EXPECT_EQ(3U, candidates_.size());
 }
 
-// Test TURN port in shared socket mode with UDP and TCP TURN server adderesses.
+// Test TURN port in shared socket mode with UDP and TCP TURN server addresses.
 TEST_F(PortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
   turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
   AddInterface(kClientAddr);
@@ -870,7 +1071,6 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
 
   allocator_->set_step_delay(cricket::kMinimumStepDelay);
   allocator_->set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
                         cricket::PORTALLOCATOR_DISABLE_TCP);
 
@@ -898,17 +1098,15 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithServerAddressResolve) {
                                  cricket::PROTO_UDP);
   AddInterface(kClientAddr);
   allocator_.reset(new cricket::BasicPortAllocator(&network_manager_));
-  cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
+  cricket::RelayServerConfig turn_server(cricket::RELAY_TURN);
   cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
-  relay_server.credentials = credentials;
-  relay_server.ports.push_back(cricket::ProtocolAddress(
-      rtc::SocketAddress("localhost", 3478),
-      cricket::PROTO_UDP, false));
-  allocator_->AddRelay(relay_server);
+  turn_server.credentials = credentials;
+  turn_server.ports.push_back(cricket::ProtocolAddress(
+      rtc::SocketAddress("localhost", 3478), cricket::PROTO_UDP, false));
+  allocator_->AddTurnServer(turn_server);
 
   allocator_->set_step_delay(cricket::kMinimumStepDelay);
   allocator_->set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
                         cricket::PORTALLOCATOR_DISABLE_TCP);
 
@@ -923,12 +1121,11 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithServerAddressResolve) {
 // stun and turn candidates.
 TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
   AddInterface(kClientAddr);
-  ResetWithNatServer(kStunAddr);
+  ResetWithStunServerAndNat(kStunAddr);
 
   AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
 
   allocator_->set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
                         cricket::PORTALLOCATOR_DISABLE_TCP);
 
@@ -941,7 +1138,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-      rtc::SocketAddress(kNatAddr.ipaddr(), 0));
+      rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
   EXPECT_PRED5(CheckCandidate, candidates_[2],
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
       rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
@@ -957,7 +1154,8 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
 // 'relay' candidates.
 TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
   AddInterface(kClientAddr);
-  ResetWithNatServer(kTurnUdpIntAddr);
+  // Use an empty SocketAddress to add a NAT without STUN server.
+  ResetWithStunServerAndNat(SocketAddress());
   AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
 
   // Must set the step delay to 0 to make sure the relay allocation phase is
@@ -966,7 +1164,6 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
   // webrtc issue 3537.
   allocator_->set_step_delay(0);
   allocator_->set_flags(allocator().flags() |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
                         cricket::PORTALLOCATOR_DISABLE_TCP);
 
@@ -978,7 +1175,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-      rtc::SocketAddress(kNatAddr.ipaddr(), 0));
+      rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
   EXPECT_PRED5(CheckCandidate, candidates_[2],
       cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
       rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
@@ -991,6 +1188,108 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
   EXPECT_EQ(1U, ports_[1]->Candidates().size());
 }
 
+// Test that when only a TCP TURN server is available, we do NOT use it as
+// a UDP STUN server, as this could leak our IP address. Thus we should only
+// expect two ports, a UDPPort and TurnPort.
+TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) {
+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+  AddInterface(kClientAddr);
+  ResetWithStunServerAndNat(rtc::SocketAddress());
+  AddTurnServers(rtc::SocketAddress(), kTurnTcpIntAddr);
+
+  allocator_->set_flags(allocator().flags() |
+                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
+                        cricket::PORTALLOCATOR_DISABLE_TCP);
+
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+
+  ASSERT_EQ_WAIT(2U, candidates_.size(), kDefaultAllocationTimeout);
+  ASSERT_EQ(2U, ports_.size());
+  EXPECT_PRED5(CheckCandidate, candidates_[0],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[1],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+  EXPECT_EQ(2U, candidates_.size());
+  EXPECT_EQ(1U, ports_[0]->Candidates().size());
+  EXPECT_EQ(1U, ports_[1]->Candidates().size());
+}
+
+// Test that even when PORTALLOCATOR_ENABLE_SHARED_SOCKET is NOT enabled, the
+// TURN server is used as the STUN server and we get 'local', 'stun', and
+// 'relay' candidates.
+// TODO(deadbeef): Remove this test when support for non-shared socket mode
+// is removed.
+TEST_F(PortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) {
+  AddInterface(kClientAddr);
+  // Use an empty SocketAddress to add a NAT without STUN server.
+  ResetWithStunServerAndNat(SocketAddress());
+  AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
+
+  allocator_->set_flags(allocator().flags() |
+                        cricket::PORTALLOCATOR_DISABLE_TCP);
+
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+
+  ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
+  ASSERT_EQ(3U, ports_.size());
+  EXPECT_PRED5(CheckCandidate, candidates_[0],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[1],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
+  EXPECT_PRED5(CheckCandidate, candidates_[2],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+  // Not using shared socket, so the STUN request's server reflexive address
+  // should be different than the TURN request's server reflexive address.
+  EXPECT_NE(candidates_[2].related_address(), candidates_[1].address());
+
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+  EXPECT_EQ(3U, candidates_.size());
+  EXPECT_EQ(1U, ports_[0]->Candidates().size());
+  EXPECT_EQ(1U, ports_[1]->Candidates().size());
+  EXPECT_EQ(1U, ports_[2]->Candidates().size());
+}
+
+// Test that even when both a STUN and TURN server are configured, the TURN
+// server is used as a STUN server and we get a 'stun' candidate.
+TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) {
+  AddInterface(kClientAddr);
+  // Configure with STUN server but destroy it, so we can ensure that it's
+  // the TURN server actually being used as a STUN server.
+  ResetWithStunServerAndNat(kStunAddr);
+  stun_server_.reset();
+  AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
+
+  allocator_->set_flags(allocator().flags() |
+                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
+                        cricket::PORTALLOCATOR_DISABLE_TCP);
+
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+
+  ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
+  EXPECT_PRED5(CheckCandidate, candidates_[0],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[1],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
+  EXPECT_PRED5(CheckCandidate, candidates_[2],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+  EXPECT_EQ(candidates_[2].related_address(), candidates_[1].address());
+
+  // Don't bother waiting for STUN timeout, since we already verified
+  // that we got a STUN candidate from the TURN server.
+}
+
 // This test verifies when PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is enabled
 // and fail to generate STUN candidate, local UDP candidate is generated
 // properly.
@@ -998,7 +1297,6 @@ TEST_F(PortAllocatorTest, TestSharedSocketNoUdpAllowed) {
   allocator().set_flags(allocator().flags() |
                         cricket::PORTALLOCATOR_DISABLE_RELAY |
                         cricket::PORTALLOCATOR_DISABLE_TCP |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr);
   AddInterface(kClientAddr);
@@ -1013,12 +1311,38 @@ TEST_F(PortAllocatorTest, TestSharedSocketNoUdpAllowed) {
   EXPECT_EQ(1U, candidates_.size());
 }
 
+// Test that when the NetworkManager doesn't have permission to enumerate
+// adapters, the PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION is specified
+// automatically.
+TEST_F(PortAllocatorTest, TestNetworkPermissionBlocked) {
+  network_manager_.set_default_local_addresses(kPrivateAddr.ipaddr(),
+                                               rtc::IPAddress());
+  network_manager_.set_enumeration_permission(
+      rtc::NetworkManager::ENUMERATION_BLOCKED);
+  allocator().set_flags(allocator().flags() |
+                        cricket::PORTALLOCATOR_DISABLE_RELAY |
+                        cricket::PORTALLOCATOR_DISABLE_TCP |
+                        cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+  EXPECT_EQ(0U, allocator_->flags() &
+                    cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  EXPECT_EQ(0U, session_->flags() &
+                    cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
+  session_->StartGettingPorts();
+  EXPECT_EQ_WAIT(1U, ports_.size(), kDefaultAllocationTimeout);
+  EXPECT_EQ(1U, candidates_.size());
+  EXPECT_PRED5(CheckCandidate, candidates_[0],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kPrivateAddr);
+  EXPECT_TRUE((session_->flags() &
+               cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) != 0);
+}
+
 // This test verifies allocator can use IPv6 addresses along with IPv4.
 TEST_F(PortAllocatorTest, TestEnableIPv6Addresses) {
   allocator().set_flags(allocator().flags() |
                         cricket::PORTALLOCATOR_DISABLE_RELAY |
                         cricket::PORTALLOCATOR_ENABLE_IPV6 |
-                        cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   AddInterface(kClientIPv6Addr);
   AddInterface(kClientAddr);
@@ -1043,87 +1367,44 @@ TEST_F(PortAllocatorTest, TestEnableIPv6Addresses) {
   EXPECT_EQ(4U, candidates_.size());
 }
 
-// Test that the httpportallocator correctly maintains its lists of stun and
-// relay servers, by never allowing an empty list.
-TEST(HttpPortAllocatorTest, TestHttpPortAllocatorHostLists) {
-  rtc::FakeNetworkManager network_manager;
-  cricket::HttpPortAllocator alloc(&network_manager, "unit test agent");
-  EXPECT_EQ(1U, alloc.relay_hosts().size());
-  EXPECT_EQ(1U, alloc.stun_hosts().size());
-
-  std::vector<std::string> relay_servers;
-  std::vector<rtc::SocketAddress> stun_servers;
-
-  alloc.SetRelayHosts(relay_servers);
-  alloc.SetStunHosts(stun_servers);
-  EXPECT_EQ(1U, alloc.relay_hosts().size());
-  EXPECT_EQ(1U, alloc.stun_hosts().size());
-
-  relay_servers.push_back("1.unittest.corp.google.com");
-  relay_servers.push_back("2.unittest.corp.google.com");
-  stun_servers.push_back(
-      rtc::SocketAddress("1.unittest.corp.google.com", 0));
-  stun_servers.push_back(
-      rtc::SocketAddress("2.unittest.corp.google.com", 0));
-
-  alloc.SetRelayHosts(relay_servers);
-  alloc.SetStunHosts(stun_servers);
-  EXPECT_EQ(2U, alloc.relay_hosts().size());
-  EXPECT_EQ(2U, alloc.stun_hosts().size());
-}
-
-// Test that the HttpPortAllocator uses correct URL to create sessions.
-TEST(HttpPortAllocatorTest, TestSessionRequestUrl) {
-  rtc::FakeNetworkManager network_manager;
-  cricket::HttpPortAllocator alloc(&network_manager, "unit test agent");
-
-  // Disable PORTALLOCATOR_ENABLE_SHARED_UFRAG.
-  alloc.set_flags(alloc.flags() & ~cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  rtc::scoped_ptr<cricket::HttpPortAllocatorSessionBase> session(
-      static_cast<cricket::HttpPortAllocatorSession*>(
-          alloc.CreateSessionInternal(
-              "test content", 0, kIceUfrag0, kIcePwd0)));
-  std::string url = session->GetSessionRequestUrl();
-  LOG(LS_INFO) << "url: " << url;
-  EXPECT_EQ(std::string(cricket::HttpPortAllocator::kCreateSessionURL), url);
-
-  // Enable PORTALLOCATOR_ENABLE_SHARED_UFRAG.
-  alloc.set_flags(alloc.flags() | cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG);
-  session.reset(static_cast<cricket::HttpPortAllocatorSession*>(
-      alloc.CreateSessionInternal("test content", 0, kIceUfrag0, kIcePwd0)));
-  url = session->GetSessionRequestUrl();
-  LOG(LS_INFO) << "url: " << url;
-  std::vector<std::string> parts;
-  rtc::split(url, '?', &parts);
-  ASSERT_EQ(2U, parts.size());
-
-  std::vector<std::string> args_parts;
-  rtc::split(parts[1], '&', &args_parts);
-
-  std::map<std::string, std::string> args;
-  for (std::vector<std::string>::iterator it = args_parts.begin();
-       it != args_parts.end(); ++it) {
-    std::vector<std::string> parts;
-    rtc::split(*it, '=', &parts);
-    ASSERT_EQ(2U, parts.size());
-    args[rtc::s_url_decode(parts[0])] = rtc::s_url_decode(parts[1]);
-  }
-
-  EXPECT_EQ(kIceUfrag0, args["username"]);
-  EXPECT_EQ(kIcePwd0, args["password"]);
-}
-
-// Tests that destroying ports with non-shared sockets does not crash.
-// b/19074679.
-TEST_F(PortAllocatorTest, TestDestroyPortsNonSharedSockets) {
+TEST_F(PortAllocatorTest, TestStopGettingPorts) {
   AddInterface(kClientAddr);
+  allocator_->set_step_delay(cricket::kDefaultStepDelay);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
-  ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
-  EXPECT_EQ(4U, ports_.size());
+  ASSERT_EQ_WAIT(2U, candidates_.size(), 1000);
+  EXPECT_EQ(2U, ports_.size());
+  session_->StopGettingPorts();
+  EXPECT_TRUE_WAIT(candidate_allocation_done_, 1000);
 
-  auto it = ports_.begin();
-  for (; it != ports_.end(); ++it) {
-    (reinterpret_cast<cricket::Port*>(*it))->Destroy();
-  }
+  // After stopping getting ports, adding a new interface will not start
+  // getting ports again.
+  candidates_.clear();
+  ports_.clear();
+  candidate_allocation_done_ = false;
+  network_manager_.AddInterface(kClientAddr2);
+  rtc::Thread::Current()->ProcessMessages(1000);
+  EXPECT_EQ(0U, candidates_.size());
+  EXPECT_EQ(0U, ports_.size());
+}
+
+TEST_F(PortAllocatorTest, TestClearGettingPorts) {
+  AddInterface(kClientAddr);
+  allocator_->set_step_delay(cricket::kDefaultStepDelay);
+  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  session_->StartGettingPorts();
+  ASSERT_EQ_WAIT(2U, candidates_.size(), 1000);
+  EXPECT_EQ(2U, ports_.size());
+  session_->ClearGettingPorts();
+  WAIT(candidate_allocation_done_, 1000);
+  EXPECT_FALSE(candidate_allocation_done_);
+
+  // After clearing getting ports, adding a new interface will start getting
+  // ports again.
+  candidates_.clear();
+  ports_.clear();
+  candidate_allocation_done_ = false;
+  network_manager_.AddInterface(kClientAddr2);
+  ASSERT_EQ_WAIT(2U, candidates_.size(), 1000);
+  EXPECT_EQ(2U, ports_.size());
 }
diff --git a/media/webrtc/trunk/webrtc/p2p/client/socketmonitor.h b/media/webrtc/trunk/webrtc/p2p/client/socketmonitor.h
index e0dd81ef6c..eb11516002 100644
--- a/media/webrtc/trunk/webrtc/p2p/client/socketmonitor.h
+++ b/media/webrtc/trunk/webrtc/p2p/client/socketmonitor.h
@@ -53,7 +53,7 @@ public:
   rtc::Thread* worker_thread_;
   rtc::Thread* monitoring_thread_;
   rtc::CriticalSection crit_;
-  uint32 rate_;
+  uint32_t rate_;
   bool monitoring_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/p2p/p2p.gyp b/media/webrtc/trunk/webrtc/p2p/p2p.gyp
index d33efef0f8..490cfbf087 100644
--- a/media/webrtc/trunk/webrtc/p2p/p2p.gyp
+++ b/media/webrtc/trunk/webrtc/p2p/p2p.gyp
@@ -39,23 +39,13 @@
         'base/port.h',
         'base/portallocator.cc',
         'base/portallocator.h',
-        'base/portallocatorsessionproxy.cc',
-        'base/portallocatorsessionproxy.h',
         'base/portinterface.h',
-        'base/portproxy.cc',
-        'base/portproxy.h',
         'base/pseudotcp.cc',
         'base/pseudotcp.h',
-        'base/rawtransport.cc',
-        'base/rawtransport.h',
-        'base/rawtransportchannel.cc',
-        'base/rawtransportchannel.h',
         'base/relayport.cc',
         'base/relayport.h',
         'base/relayserver.cc',
         'base/relayserver.h',
-        'base/session.cc',
-        'base/session.h',
         'base/sessiondescription.cc',
         'base/sessiondescription.h',
         'base/sessionid.h',
@@ -74,8 +64,8 @@
         'base/transportchannel.cc',
         'base/transportchannel.h',
         'base/transportchannelimpl.h',
-        'base/transportchannelproxy.cc',
-        'base/transportchannelproxy.h',
+        'base/transportcontroller.cc',
+        'base/transportcontroller.h',
         'base/transportdescription.cc',
         'base/transportdescription.h',
         'base/transportdescriptionfactory.cc',
@@ -86,11 +76,8 @@
         'base/turnserver.cc',
         'base/turnserver.h',
         'base/udpport.h',
-        'client/autoportallocator.h',
         'client/basicportallocator.cc',
         'client/basicportallocator.h',
-        'client/connectivitychecker.cc',
-        'client/connectivitychecker.h',
         'client/httpportallocator.cc',
         'client/httpportallocator.h',
         'client/socketmonitor.cc',
@@ -112,6 +99,34 @@
           ],
         }],
       ],
+    },
+    {
+      'target_name': 'libstunprober',
+      'type': 'static_library',
+      'dependencies': [
+        '<(webrtc_root)/base/base.gyp:rtc_base',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+      ],
+      'cflags_cc!': [
+        '-Wnon-virtual-dtor',
+      ],
+      'sources': [
+        'stunprober/stunprober.cc',
+      ],
+    },
+    {
+      'target_name': 'stun_prober',
+      'type': 'executable',
+      'dependencies': [
+        'libstunprober',
+        'rtc_p2p'
+      ],
+      'cflags_cc!': [
+        '-Wnon-virtual-dtor',
+      ],
+      'sources': [
+        'stunprober/main.cc',
+      ],
     }],
 }
 
diff --git a/media/webrtc/trunk/webrtc/p2p/p2p_tests.gypi b/media/webrtc/trunk/webrtc/p2p/p2p_tests.gypi
index afab33bfaf..ba7f553bba 100644
--- a/media/webrtc/trunk/webrtc/p2p/p2p_tests.gypi
+++ b/media/webrtc/trunk/webrtc/p2p/p2p_tests.gypi
@@ -15,10 +15,9 @@
       'direct_dependent_settings': {
         'sources': [
           'base/dtlstransportchannel_unittest.cc',
-          'base/fakesession.h',
+          'base/faketransportcontroller.h',
           'base/p2ptransportchannel_unittest.cc',
           'base/port_unittest.cc',
-          'base/portallocatorsessionproxy_unittest.cc',
           'base/pseudotcp_unittest.cc',
           'base/relayport_unittest.cc',
           'base/relayserver_unittest.cc',
@@ -30,11 +29,12 @@
           'base/teststunserver.h',
           'base/testturnserver.h',
           'base/transport_unittest.cc',
+          'base/transportcontroller_unittest.cc',
           'base/transportdescriptionfactory_unittest.cc',
           'base/turnport_unittest.cc',
-          'client/connectivitychecker_unittest.cc',
           'client/fakeportallocator.h',
           'client/portallocator_unittest.cc',
+          'stunprober/stunprober_unittest.cc',
         ],
       },
     },
diff --git a/media/webrtc/trunk/webrtc/p2p/stunprober/main.cc b/media/webrtc/trunk/webrtc/p2p/stunprober/main.cc
new file mode 100644
index 0000000000..076113ce68
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/stunprober/main.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <iostream>
+#include <map>
+#include "webrtc/base/checks.h"
+#include "webrtc/base/flags.h"
+#include "webrtc/base/helpers.h"
+#include "webrtc/base/nethelpers.h"
+#include "webrtc/base/network.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/ssladapter.h"
+#include "webrtc/base/stringutils.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.cc"
+#include "webrtc/p2p/stunprober/stunprober.h"
+
+using stunprober::StunProber;
+using stunprober::AsyncCallback;
+
+DEFINE_bool(help, false, "Prints this message");
+DEFINE_int(interval, 10, "Interval of consecutive stun pings in milliseconds");
+DEFINE_bool(shared_socket, false, "Share socket mode for different remote IPs");
+DEFINE_int(pings_per_ip,
+           10,
+           "Number of consecutive stun pings to send for each IP");
+DEFINE_int(timeout,
+           1000,
+           "Milliseconds of wait after the last ping sent before exiting");
+DEFINE_string(
+    servers,
+    "stun.l.google.com:19302,stun1.l.google.com:19302,stun2.l.google.com:19302",
+    "Comma separated STUN server addresses with ports");
+
+namespace {
+
+const char* PrintNatType(stunprober::NatType type) {
+  switch (type) {
+    case stunprober::NATTYPE_NONE:
+      return "Not behind a NAT";
+    case stunprober::NATTYPE_UNKNOWN:
+      return "Unknown NAT type";
+    case stunprober::NATTYPE_SYMMETRIC:
+      return "Symmetric NAT";
+    case stunprober::NATTYPE_NON_SYMMETRIC:
+      return "Non-Symmetric NAT";
+    default:
+      return "Invalid";
+  }
+}
+
+void PrintStats(StunProber* prober) {
+  StunProber::Stats stats;
+  if (!prober->GetStats(&stats)) {
+    LOG(LS_WARNING) << "Results are inconclusive.";
+    return;
+  }
+
+  LOG(LS_INFO) << "Shared Socket Mode: " << stats.shared_socket_mode;
+  LOG(LS_INFO) << "Requests sent: " << stats.num_request_sent;
+  LOG(LS_INFO) << "Responses received: " << stats.num_response_received;
+  LOG(LS_INFO) << "Target interval (ns): " << stats.target_request_interval_ns;
+  LOG(LS_INFO) << "Actual interval (ns): " << stats.actual_request_interval_ns;
+  LOG(LS_INFO) << "NAT Type: " << PrintNatType(stats.nat_type);
+  LOG(LS_INFO) << "Host IP: " << stats.host_ip;
+  LOG(LS_INFO) << "Server-reflexive ips: ";
+  for (auto& ip : stats.srflx_addrs) {
+    LOG(LS_INFO) << "\t" << ip;
+  }
+
+  LOG(LS_INFO) << "Success Precent: " << stats.success_percent;
+  LOG(LS_INFO) << "Response Latency:" << stats.average_rtt_ms;
+}
+
+void StopTrial(rtc::Thread* thread, StunProber* prober, int result) {
+  thread->Quit();
+  if (prober) {
+    LOG(LS_INFO) << "Result: " << result;
+    if (result == StunProber::SUCCESS) {
+      PrintStats(prober);
+    }
+  }
+}
+
+}  // namespace
+
+int main(int argc, char** argv) {
+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+
+  std::vector<rtc::SocketAddress> server_addresses;
+  std::istringstream servers(FLAG_servers);
+  std::string server;
+  while (getline(servers, server, ',')) {
+    rtc::SocketAddress addr;
+    if (!addr.FromString(server)) {
+      LOG(LS_ERROR) << "Parsing " << server << " failed.";
+      return -1;
+    }
+    server_addresses.push_back(addr);
+  }
+
+  rtc::InitializeSSL();
+  rtc::InitRandom(rtc::Time());
+  rtc::Thread* thread = rtc::ThreadManager::Instance()->WrapCurrentThread();
+  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> socket_factory(
+      new rtc::BasicPacketSocketFactory());
+  rtc::scoped_ptr<rtc::BasicNetworkManager> network_manager(
+      new rtc::BasicNetworkManager());
+  rtc::NetworkManager::NetworkList networks;
+  network_manager->GetNetworks(&networks);
+  StunProber* prober =
+      new StunProber(socket_factory.get(), rtc::Thread::Current(), networks);
+  auto finish_callback = [thread](StunProber* prober, int result) {
+    StopTrial(thread, prober, result);
+  };
+  prober->Start(server_addresses, FLAG_shared_socket, FLAG_interval,
+                FLAG_pings_per_ip, FLAG_timeout,
+                AsyncCallback(finish_callback));
+  thread->Run();
+  delete prober;
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober.cc b/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober.cc
new file mode 100644
index 0000000000..9316ea89bd
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober.cc
@@ -0,0 +1,570 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <map>
+#include <set>
+#include <string>
+
+#include "webrtc/base/asyncpacketsocket.h"
+#include "webrtc/base/asyncresolverinterface.h"
+#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/helpers.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/packetsocketfactory.h"
+#include "webrtc/p2p/base/stun.h"
+#include "webrtc/p2p/stunprober/stunprober.h"
+
+namespace stunprober {
+
+namespace {
+
+const int THREAD_WAKE_UP_INTERVAL_MS = 5;
+
+template <typename T>
+void IncrementCounterByAddress(std::map<T, int>* counter_per_ip, const T& ip) {
+  counter_per_ip->insert(std::make_pair(ip, 0)).first->second++;
+}
+
+}  // namespace
+
+// A requester tracks the requests and responses from a single socket to many
+// STUN servers
+class StunProber::Requester : public sigslot::has_slots<> {
+ public:
+  // Each Request maps to a request and response.
+  struct Request {
+    // Actual time the STUN bind request was sent.
+    int64_t sent_time_ms = 0;
+    // Time the response was received.
+    int64_t received_time_ms = 0;
+
+    // Server reflexive address from STUN response for this given request.
+    rtc::SocketAddress srflx_addr;
+
+    rtc::IPAddress server_addr;
+
+    int64_t rtt() { return received_time_ms - sent_time_ms; }
+    void ProcessResponse(const char* buf, size_t buf_len);
+  };
+
+  // StunProber provides |server_ips| for Requester to probe. For shared
+  // socket mode, it'll be all the resolved IP addresses. For non-shared mode,
+  // it'll just be a single address.
+  Requester(StunProber* prober,
+            rtc::AsyncPacketSocket* socket,
+            const std::vector<rtc::SocketAddress>& server_ips);
+  virtual ~Requester();
+
+  // There is no callback for SendStunRequest as the underneath socket send is
+  // expected to be completed immediately. Otherwise, it'll skip this request
+  // and move to the next one.
+  void SendStunRequest();
+
+  void OnStunResponseReceived(rtc::AsyncPacketSocket* socket,
+                              const char* buf,
+                              size_t size,
+                              const rtc::SocketAddress& addr,
+                              const rtc::PacketTime& time);
+
+  const std::vector<Request*>& requests() { return requests_; }
+
+  // Whether this Requester has completed all requests.
+  bool Done() {
+    return static_cast<size_t>(num_request_sent_) == server_ips_.size();
+  }
+
+ private:
+  Request* GetRequestByAddress(const rtc::IPAddress& ip);
+
+  StunProber* prober_;
+
+  // The socket for this session.
+  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+
+  // Temporary SocketAddress and buffer for RecvFrom.
+  rtc::SocketAddress addr_;
+  rtc::scoped_ptr<rtc::ByteBuffer> response_packet_;
+
+  std::vector<Request*> requests_;
+  std::vector<rtc::SocketAddress> server_ips_;
+  int16_t num_request_sent_ = 0;
+  int16_t num_response_received_ = 0;
+
+  rtc::ThreadChecker& thread_checker_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Requester);
+};
+
+StunProber::Requester::Requester(
+    StunProber* prober,
+    rtc::AsyncPacketSocket* socket,
+    const std::vector<rtc::SocketAddress>& server_ips)
+    : prober_(prober),
+      socket_(socket),
+      response_packet_(new rtc::ByteBuffer(nullptr, kMaxUdpBufferSize)),
+      server_ips_(server_ips),
+      thread_checker_(prober->thread_checker_) {
+  socket_->SignalReadPacket.connect(
+      this, &StunProber::Requester::OnStunResponseReceived);
+}
+
+StunProber::Requester::~Requester() {
+  if (socket_) {
+    socket_->Close();
+  }
+  for (auto req : requests_) {
+    if (req) {
+      delete req;
+    }
+  }
+}
+
+void StunProber::Requester::SendStunRequest() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  requests_.push_back(new Request());
+  Request& request = *(requests_.back());
+  cricket::StunMessage message;
+
+  // Random transaction ID, STUN_BINDING_REQUEST
+  message.SetTransactionID(
+      rtc::CreateRandomString(cricket::kStunTransactionIdLength));
+  message.SetType(cricket::STUN_BINDING_REQUEST);
+
+  rtc::scoped_ptr<rtc::ByteBuffer> request_packet(
+      new rtc::ByteBuffer(nullptr, kMaxUdpBufferSize));
+  if (!message.Write(request_packet.get())) {
+    prober_->ReportOnFinished(WRITE_FAILED);
+    return;
+  }
+
+  auto addr = server_ips_[num_request_sent_];
+  request.server_addr = addr.ipaddr();
+
+  // The write must succeed immediately. Otherwise, the calculating of the STUN
+  // request timing could become too complicated. Callback is ignored by passing
+  // empty AsyncCallback.
+  rtc::PacketOptions options;
+  int rv = socket_->SendTo(const_cast<char*>(request_packet->Data()),
+                           request_packet->Length(), addr, options);
+  if (rv < 0) {
+    prober_->ReportOnFinished(WRITE_FAILED);
+    return;
+  }
+
+  request.sent_time_ms = rtc::Time();
+
+  num_request_sent_++;
+  RTC_DCHECK(static_cast<size_t>(num_request_sent_) <= server_ips_.size());
+}
+
+void StunProber::Requester::Request::ProcessResponse(const char* buf,
+                                                     size_t buf_len) {
+  int64_t now = rtc::Time();
+  rtc::ByteBuffer message(buf, buf_len);
+  cricket::StunMessage stun_response;
+  if (!stun_response.Read(&message)) {
+    // Invalid or incomplete STUN packet.
+    received_time_ms = 0;
+    return;
+  }
+
+  // Get external address of the socket.
+  const cricket::StunAddressAttribute* addr_attr =
+      stun_response.GetAddress(cricket::STUN_ATTR_MAPPED_ADDRESS);
+  if (addr_attr == nullptr) {
+    // Addresses not available to detect whether or not behind a NAT.
+    return;
+  }
+
+  if (addr_attr->family() != cricket::STUN_ADDRESS_IPV4 &&
+      addr_attr->family() != cricket::STUN_ADDRESS_IPV6) {
+    return;
+  }
+
+  received_time_ms = now;
+
+  srflx_addr = addr_attr->GetAddress();
+}
+
+void StunProber::Requester::OnStunResponseReceived(
+    rtc::AsyncPacketSocket* socket,
+    const char* buf,
+    size_t size,
+    const rtc::SocketAddress& addr,
+    const rtc::PacketTime& time) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(socket_);
+  Request* request = GetRequestByAddress(addr.ipaddr());
+  if (!request) {
+    // Something is wrong, finish the test.
+    prober_->ReportOnFinished(GENERIC_FAILURE);
+    return;
+  }
+
+  num_response_received_++;
+  request->ProcessResponse(buf, size);
+}
+
+StunProber::Requester::Request* StunProber::Requester::GetRequestByAddress(
+    const rtc::IPAddress& ipaddr) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  for (auto request : requests_) {
+    if (request->server_addr == ipaddr) {
+      return request;
+    }
+  }
+
+  return nullptr;
+}
+
+StunProber::StunProber(rtc::PacketSocketFactory* socket_factory,
+                       rtc::Thread* thread,
+                       const rtc::NetworkManager::NetworkList& networks)
+    : interval_ms_(0),
+      socket_factory_(socket_factory),
+      thread_(thread),
+      networks_(networks) {
+}
+
+StunProber::~StunProber() {
+  for (auto req : requesters_) {
+    if (req) {
+      delete req;
+    }
+  }
+  for (auto s : sockets_) {
+    if (s) {
+      delete s;
+    }
+  }
+}
+
+bool StunProber::Start(const std::vector<rtc::SocketAddress>& servers,
+                       bool shared_socket_mode,
+                       int interval_ms,
+                       int num_request_per_ip,
+                       int timeout_ms,
+                       const AsyncCallback callback) {
+  observer_adapter_.set_callback(callback);
+  return Prepare(servers, shared_socket_mode, interval_ms, num_request_per_ip,
+                 timeout_ms, &observer_adapter_);
+}
+
+bool StunProber::Prepare(const std::vector<rtc::SocketAddress>& servers,
+                         bool shared_socket_mode,
+                         int interval_ms,
+                         int num_request_per_ip,
+                         int timeout_ms,
+                         StunProber::Observer* observer) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  interval_ms_ = interval_ms;
+  shared_socket_mode_ = shared_socket_mode;
+
+  requests_per_ip_ = num_request_per_ip;
+  if (requests_per_ip_ == 0 || servers.size() == 0) {
+    return false;
+  }
+
+  timeout_ms_ = timeout_ms;
+  servers_ = servers;
+  observer_ = observer;
+  return ResolveServerName(servers_.back());
+}
+
+bool StunProber::Start(StunProber::Observer* observer) {
+  observer_ = observer;
+  if (total_ready_sockets_ != total_socket_required()) {
+    return false;
+  }
+  MaybeScheduleStunRequests();
+  return true;
+}
+
+bool StunProber::ResolveServerName(const rtc::SocketAddress& addr) {
+  rtc::AsyncResolverInterface* resolver =
+      socket_factory_->CreateAsyncResolver();
+  if (!resolver) {
+    return false;
+  }
+  resolver->SignalDone.connect(this, &StunProber::OnServerResolved);
+  resolver->Start(addr);
+  return true;
+}
+
+void StunProber::OnSocketReady(rtc::AsyncPacketSocket* socket,
+                               const rtc::SocketAddress& addr) {
+  total_ready_sockets_++;
+  if (total_ready_sockets_ == total_socket_required()) {
+    ReportOnPrepared(SUCCESS);
+  }
+}
+
+void StunProber::OnServerResolved(rtc::AsyncResolverInterface* resolver) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+
+  if (resolver->GetError() == 0) {
+    rtc::SocketAddress addr(resolver->address().ipaddr(),
+                            resolver->address().port());
+    all_servers_addrs_.push_back(addr);
+  }
+
+  // Deletion of AsyncResolverInterface can't be done in OnResolveResult which
+  // handles SignalDone.
+  invoker_.AsyncInvoke<void>(
+      thread_,
+      rtc::Bind(&rtc::AsyncResolverInterface::Destroy, resolver, false));
+  servers_.pop_back();
+
+  if (servers_.size()) {
+    if (!ResolveServerName(servers_.back())) {
+      ReportOnPrepared(RESOLVE_FAILED);
+    }
+    return;
+  }
+
+  if (all_servers_addrs_.size() == 0) {
+    ReportOnPrepared(RESOLVE_FAILED);
+    return;
+  }
+
+  // Dedupe.
+  std::set<rtc::SocketAddress> addrs(all_servers_addrs_.begin(),
+                                     all_servers_addrs_.end());
+  all_servers_addrs_.assign(addrs.begin(), addrs.end());
+
+  // Prepare all the sockets beforehand. All of them will bind to "any" address.
+  while (sockets_.size() < total_socket_required()) {
+    rtc::scoped_ptr<rtc::AsyncPacketSocket> socket(
+        socket_factory_->CreateUdpSocket(rtc::SocketAddress(INADDR_ANY, 0), 0,
+                                         0));
+    if (!socket) {
+      ReportOnPrepared(GENERIC_FAILURE);
+      return;
+    }
+    // Chrome and WebRTC behave differently in terms of the state of a socket
+    // once returned from PacketSocketFactory::CreateUdpSocket.
+    if (socket->GetState() == rtc::AsyncPacketSocket::STATE_BINDING) {
+      socket->SignalAddressReady.connect(this, &StunProber::OnSocketReady);
+    } else {
+      OnSocketReady(socket.get(), rtc::SocketAddress(INADDR_ANY, 0));
+    }
+    sockets_.push_back(socket.release());
+  }
+}
+
+StunProber::Requester* StunProber::CreateRequester() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!sockets_.size()) {
+    return nullptr;
+  }
+  StunProber::Requester* requester;
+  if (shared_socket_mode_) {
+    requester = new Requester(this, sockets_.back(), all_servers_addrs_);
+  } else {
+    std::vector<rtc::SocketAddress> server_ip;
+    server_ip.push_back(
+        all_servers_addrs_[(num_request_sent_ % all_servers_addrs_.size())]);
+    requester = new Requester(this, sockets_.back(), server_ip);
+  }
+
+  sockets_.pop_back();
+  return requester;
+}
+
+bool StunProber::SendNextRequest() {
+  if (!current_requester_ || current_requester_->Done()) {
+    current_requester_ = CreateRequester();
+    requesters_.push_back(current_requester_);
+  }
+  if (!current_requester_) {
+    return false;
+  }
+  current_requester_->SendStunRequest();
+  num_request_sent_++;
+  return true;
+}
+
+bool StunProber::should_send_next_request(uint32_t now) {
+  if (interval_ms_ < THREAD_WAKE_UP_INTERVAL_MS) {
+    return now >= next_request_time_ms_;
+  } else {
+    return (now + (THREAD_WAKE_UP_INTERVAL_MS / 2)) >= next_request_time_ms_;
+  }
+}
+
+int StunProber::get_wake_up_interval_ms() {
+  if (interval_ms_ < THREAD_WAKE_UP_INTERVAL_MS) {
+    return 1;
+  } else {
+    return THREAD_WAKE_UP_INTERVAL_MS;
+  }
+}
+
+void StunProber::MaybeScheduleStunRequests() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  uint32_t now = rtc::Time();
+
+  if (Done()) {
+    invoker_.AsyncInvokeDelayed<void>(
+        thread_, rtc::Bind(&StunProber::ReportOnFinished, this, SUCCESS),
+        timeout_ms_);
+    return;
+  }
+  if (should_send_next_request(now)) {
+    if (!SendNextRequest()) {
+      ReportOnFinished(GENERIC_FAILURE);
+      return;
+    }
+    next_request_time_ms_ = now + interval_ms_;
+  }
+  invoker_.AsyncInvokeDelayed<void>(
+      thread_, rtc::Bind(&StunProber::MaybeScheduleStunRequests, this),
+      get_wake_up_interval_ms());
+}
+
+bool StunProber::GetStats(StunProber::Stats* prob_stats) const {
+  // No need to be on the same thread.
+  if (!prob_stats) {
+    return false;
+  }
+
+  StunProber::Stats stats;
+
+  int rtt_sum = 0;
+  int64_t first_sent_time = 0;
+  int64_t last_sent_time = 0;
+  NatType nat_type = NATTYPE_INVALID;
+
+  // Track of how many srflx IP that we have seen.
+  std::set<rtc::IPAddress> srflx_ips;
+
+  // If we're not receiving any response on a given IP, all requests sent to
+  // that IP should be ignored as this could just be an DNS error.
+  std::map<rtc::IPAddress, int> num_response_per_server;
+  std::map<rtc::IPAddress, int> num_request_per_server;
+
+  for (auto* requester : requesters_) {
+    std::map<rtc::SocketAddress, int> num_response_per_srflx_addr;
+    for (auto request : requester->requests()) {
+      if (request->sent_time_ms <= 0) {
+        continue;
+      }
+
+      ++stats.raw_num_request_sent;
+      IncrementCounterByAddress(&num_request_per_server, request->server_addr);
+
+      if (!first_sent_time) {
+        first_sent_time = request->sent_time_ms;
+      }
+      last_sent_time = request->sent_time_ms;
+
+      if (request->received_time_ms < request->sent_time_ms) {
+        continue;
+      }
+
+      IncrementCounterByAddress(&num_response_per_server, request->server_addr);
+      IncrementCounterByAddress(&num_response_per_srflx_addr,
+                                request->srflx_addr);
+      rtt_sum += request->rtt();
+      stats.srflx_addrs.insert(request->srflx_addr.ToString());
+      srflx_ips.insert(request->srflx_addr.ipaddr());
+    }
+
+    // If we're using shared mode and seeing >1 srflx addresses for a single
+    // requester, it's symmetric NAT.
+    if (shared_socket_mode_ && num_response_per_srflx_addr.size() > 1) {
+      nat_type = NATTYPE_SYMMETRIC;
+    }
+  }
+
+  // We're probably not behind a regular NAT. We have more than 1 distinct
+  // server reflexive IPs.
+  if (srflx_ips.size() > 1) {
+    return false;
+  }
+
+  int num_sent = 0;
+  int num_received = 0;
+  int num_server_ip_with_response = 0;
+
+  for (const auto& kv : num_response_per_server) {
+    RTC_DCHECK_GT(kv.second, 0);
+    num_server_ip_with_response++;
+    num_received += kv.second;
+    num_sent += num_request_per_server[kv.first];
+  }
+
+  // Shared mode is only true if we use the shared socket and there are more
+  // than 1 responding servers.
+  stats.shared_socket_mode =
+      shared_socket_mode_ && (num_server_ip_with_response > 1);
+
+  if (stats.shared_socket_mode && nat_type == NATTYPE_INVALID) {
+    nat_type = NATTYPE_NON_SYMMETRIC;
+  }
+
+  // If we could find a local IP matching srflx, we're not behind a NAT.
+  rtc::SocketAddress srflx_addr;
+  if (stats.srflx_addrs.size() &&
+      !srflx_addr.FromString(*(stats.srflx_addrs.begin()))) {
+    return false;
+  }
+  for (const auto& net : networks_) {
+    if (srflx_addr.ipaddr() == net->GetBestIP()) {
+      nat_type = stunprober::NATTYPE_NONE;
+      stats.host_ip = net->GetBestIP().ToString();
+      break;
+    }
+  }
+
+  // Finally, we know we're behind a NAT but can't determine which type it is.
+  if (nat_type == NATTYPE_INVALID) {
+    nat_type = NATTYPE_UNKNOWN;
+  }
+
+  stats.nat_type = nat_type;
+  stats.num_request_sent = num_sent;
+  stats.num_response_received = num_received;
+  stats.target_request_interval_ns = interval_ms_ * 1000;
+
+  if (num_sent) {
+    stats.success_percent = static_cast<int>(100 * num_received / num_sent);
+  }
+
+  if (stats.raw_num_request_sent > 1) {
+    stats.actual_request_interval_ns =
+        (1000 * (last_sent_time - first_sent_time)) /
+        (stats.raw_num_request_sent - 1);
+  }
+
+  if (num_received) {
+    stats.average_rtt_ms = static_cast<int>((rtt_sum / num_received));
+  }
+
+  *prob_stats = stats;
+  return true;
+}
+
+void StunProber::ReportOnPrepared(StunProber::Status status) {
+  if (observer_) {
+    observer_->OnPrepared(this, status);
+  }
+}
+
+void StunProber::ReportOnFinished(StunProber::Status status) {
+  if (observer_) {
+    observer_->OnFinished(this, status);
+  }
+}
+
+}  // namespace stunprober
diff --git a/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober.h b/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober.h
new file mode 100644
index 0000000000..b725cbef0a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober.h
@@ -0,0 +1,254 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_P2P_STUNPROBER_STUNPROBER_H_
+#define WEBRTC_P2P_STUNPROBER_STUNPROBER_H_
+
+#include <set>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/asyncinvoker.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/callback.h"
+#include "webrtc/base/ipaddress.h"
+#include "webrtc/base/network.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/socketaddress.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/typedefs.h"
+
+namespace rtc {
+class AsyncPacketSocket;
+class PacketSocketFactory;
+class Thread;
+class NetworkManager;
+class AsyncResolverInterface;
+}  // namespace rtc
+
+namespace stunprober {
+
+class StunProber;
+
+static const int kMaxUdpBufferSize = 1200;
+
+typedef rtc::Callback2<void, StunProber*, int> AsyncCallback;
+
+enum NatType {
+  NATTYPE_INVALID,
+  NATTYPE_NONE,          // Not behind a NAT.
+  NATTYPE_UNKNOWN,       // Behind a NAT but type can't be determine.
+  NATTYPE_SYMMETRIC,     // Behind a symmetric NAT.
+  NATTYPE_NON_SYMMETRIC  // Behind a non-symmetric NAT.
+};
+
+class StunProber : public sigslot::has_slots<> {
+ public:
+  enum Status {       // Used in UMA_HISTOGRAM_ENUMERATION.
+    SUCCESS,          // Successfully received bytes from the server.
+    GENERIC_FAILURE,  // Generic failure.
+    RESOLVE_FAILED,   // Host resolution failed.
+    WRITE_FAILED,     // Sending a message to the server failed.
+    READ_FAILED,      // Reading the reply from the server failed.
+  };
+
+  class Observer {
+   public:
+    virtual ~Observer() = default;
+    virtual void OnPrepared(StunProber* prober, StunProber::Status status) = 0;
+    virtual void OnFinished(StunProber* prober, StunProber::Status status) = 0;
+  };
+
+  struct Stats {
+    Stats() {}
+
+    // |raw_num_request_sent| is the total number of requests
+    // sent. |num_request_sent| is the count of requests against a server where
+    // we see at least one response. |num_request_sent| is designed to protect
+    // against DNS resolution failure or the STUN server is not responsive
+    // which could skew the result.
+    int raw_num_request_sent = 0;
+    int num_request_sent = 0;
+
+    int num_response_received = 0;
+    NatType nat_type = NATTYPE_INVALID;
+    int average_rtt_ms = -1;
+    int success_percent = 0;
+    int target_request_interval_ns = 0;
+    int actual_request_interval_ns = 0;
+
+    // Also report whether this trial can't be considered truly as shared
+    // mode. Share mode only makes sense when we have multiple IP resolved and
+    // successfully probed.
+    bool shared_socket_mode = false;
+
+    std::string host_ip;
+
+    // If the srflx_addrs has more than 1 element, the NAT is symmetric.
+    std::set<std::string> srflx_addrs;
+  };
+
+  StunProber(rtc::PacketSocketFactory* socket_factory,
+             rtc::Thread* thread,
+             const rtc::NetworkManager::NetworkList& networks);
+  virtual ~StunProber();
+
+  // Begin performing the probe test against the |servers|. If
+  // |shared_socket_mode| is false, each request will be done with a new socket.
+  // Otherwise, a unique socket will be used for a single round of requests
+  // against all resolved IPs. No single socket will be used against a given IP
+  // more than once.  The interval of requests will be as close to the requested
+  // inter-probe interval |stun_ta_interval_ms| as possible. After sending out
+  // the last scheduled request, the probe will wait |timeout_ms| for request
+  // responses and then call |finish_callback|.  |requests_per_ip| indicates how
+  // many requests should be tried for each resolved IP address. In shared mode,
+  // (the number of sockets to be created) equals to |requests_per_ip|. In
+  // non-shared mode, (the number of sockets) equals to requests_per_ip * (the
+  // number of resolved IP addresses). TODO(guoweis): Remove this once
+  // everything moved to Prepare() and Run().
+  bool Start(const std::vector<rtc::SocketAddress>& servers,
+             bool shared_socket_mode,
+             int stun_ta_interval_ms,
+             int requests_per_ip,
+             int timeout_ms,
+             const AsyncCallback finish_callback);
+
+  // TODO(guoweis): The combination of Prepare() and Run() are equivalent to the
+  // Start() above. Remove Start() once everything is migrated.
+  bool Prepare(const std::vector<rtc::SocketAddress>& servers,
+               bool shared_socket_mode,
+               int stun_ta_interval_ms,
+               int requests_per_ip,
+               int timeout_ms,
+               StunProber::Observer* observer);
+
+  // Start to send out the STUN probes.
+  bool Start(StunProber::Observer* observer);
+
+  // Method to retrieve the Stats once |finish_callback| is invoked. Returning
+  // false when the result is inconclusive, for example, whether it's behind a
+  // NAT or not.
+  bool GetStats(Stats* stats) const;
+
+  int estimated_execution_time() {
+    return static_cast<int>(requests_per_ip_ * all_servers_addrs_.size() *
+                            interval_ms_);
+  }
+
+ private:
+  // A requester tracks the requests and responses from a single socket to many
+  // STUN servers.
+  class Requester;
+
+  // TODO(guoweis): Remove this once all dependencies move away from
+  // AsyncCallback.
+  class ObserverAdapter : public Observer {
+   public:
+    void set_callback(AsyncCallback callback) { callback_ = callback; }
+    void OnPrepared(StunProber* stunprober, Status status) {
+      if (status == SUCCESS) {
+        stunprober->Start(this);
+      } else {
+        callback_(stunprober, status);
+      }
+    }
+    void OnFinished(StunProber* stunprober, Status status) {
+      callback_(stunprober, status);
+    }
+
+   private:
+    AsyncCallback callback_;
+  };
+
+  bool ResolveServerName(const rtc::SocketAddress& addr);
+  void OnServerResolved(rtc::AsyncResolverInterface* resolver);
+
+  void OnSocketReady(rtc::AsyncPacketSocket* socket,
+                     const rtc::SocketAddress& addr);
+
+  bool Done() {
+    return num_request_sent_ >= requests_per_ip_ * all_servers_addrs_.size();
+  }
+
+  size_t total_socket_required() {
+    return (shared_socket_mode_ ? 1 : all_servers_addrs_.size()) *
+           requests_per_ip_;
+  }
+
+  bool should_send_next_request(uint32_t now);
+  int get_wake_up_interval_ms();
+
+  bool SendNextRequest();
+
+  // Will be invoked in 1ms intervals and schedule the next request from the
+  // |current_requester_| if the time has passed for another request.
+  void MaybeScheduleStunRequests();
+
+  void ReportOnPrepared(StunProber::Status status);
+  void ReportOnFinished(StunProber::Status status);
+
+  Requester* CreateRequester();
+
+  Requester* current_requester_ = nullptr;
+
+  // The time when the next request should go out.
+  uint64_t next_request_time_ms_ = 0;
+
+  // Total requests sent so far.
+  uint32_t num_request_sent_ = 0;
+
+  bool shared_socket_mode_ = false;
+
+  // How many requests should be done against each resolved IP.
+  uint32_t requests_per_ip_ = 0;
+
+  // Milliseconds to pause between each STUN request.
+  int interval_ms_;
+
+  // Timeout period after the last request is sent.
+  int timeout_ms_;
+
+  // STUN server name to be resolved.
+  std::vector<rtc::SocketAddress> servers_;
+
+  // Weak references.
+  rtc::PacketSocketFactory* socket_factory_;
+  rtc::Thread* thread_;
+
+  // Accumulate all resolved addresses.
+  std::vector<rtc::SocketAddress> all_servers_addrs_;
+
+  // The set of STUN probe sockets and their state.
+  std::vector<Requester*> requesters_;
+
+  rtc::ThreadChecker thread_checker_;
+
+  // Temporary storage for created sockets.
+  std::vector<rtc::AsyncPacketSocket*> sockets_;
+  // This tracks how many of the sockets are ready.
+  size_t total_ready_sockets_ = 0;
+
+  rtc::AsyncInvoker invoker_;
+
+  Observer* observer_ = nullptr;
+  // TODO(guoweis): Remove this once all dependencies move away from
+  // AsyncCallback.
+  ObserverAdapter observer_adapter_;
+
+  rtc::NetworkManager::NetworkList networks_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(StunProber);
+};
+
+}  // namespace stunprober
+
+#endif  // WEBRTC_P2P_STUNPROBER_STUNPROBER_H_
diff --git a/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober_unittest.cc b/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober_unittest.cc
new file mode 100644
index 0000000000..cdcc14a36f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/p2p/stunprober/stunprober_unittest.cc
@@ -0,0 +1,140 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/asyncresolverinterface.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/physicalsocketserver.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/ssladapter.h"
+#include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/stunprober/stunprober.h"
+
+using stunprober::StunProber;
+using stunprober::AsyncCallback;
+
+namespace stunprober {
+
+namespace {
+
+const rtc::SocketAddress kLocalAddr("192.168.0.1", 0);
+const rtc::SocketAddress kStunAddr1("1.1.1.1", 3478);
+const rtc::SocketAddress kStunAddr2("1.1.1.2", 3478);
+const rtc::SocketAddress kFailedStunAddr("1.1.1.3", 3478);
+const rtc::SocketAddress kStunMappedAddr("77.77.77.77", 0);
+
+}  // namespace
+
+class StunProberTest : public testing::Test {
+ public:
+  StunProberTest()
+      : main_(rtc::Thread::Current()),
+        pss_(new rtc::PhysicalSocketServer),
+        ss_(new rtc::VirtualSocketServer(pss_.get())),
+        ss_scope_(ss_.get()),
+        result_(StunProber::SUCCESS),
+        stun_server_1_(cricket::TestStunServer::Create(rtc::Thread::Current(),
+                                                       kStunAddr1)),
+        stun_server_2_(cricket::TestStunServer::Create(rtc::Thread::Current(),
+                                                       kStunAddr2)) {
+    stun_server_1_->set_fake_stun_addr(kStunMappedAddr);
+    stun_server_2_->set_fake_stun_addr(kStunMappedAddr);
+    rtc::InitializeSSL();
+  }
+
+  void set_expected_result(int result) { result_ = result; }
+
+  void StartProbing(rtc::PacketSocketFactory* socket_factory,
+                    const std::vector<rtc::SocketAddress>& addrs,
+                    const rtc::NetworkManager::NetworkList& networks,
+                    bool shared_socket,
+                    uint16_t interval,
+                    uint16_t pings_per_ip) {
+    prober.reset(
+        new StunProber(socket_factory, rtc::Thread::Current(), networks));
+    prober->Start(addrs, shared_socket, interval, pings_per_ip,
+                  100 /* timeout_ms */, [this](StunProber* prober, int result) {
+                    this->StopCallback(prober, result);
+                  });
+  }
+
+  void RunProber(bool shared_mode) {
+    const int pings_per_ip = 3;
+    std::vector<rtc::SocketAddress> addrs;
+    addrs.push_back(kStunAddr1);
+    addrs.push_back(kStunAddr2);
+    // Add a non-existing server. This shouldn't pollute the result.
+    addrs.push_back(kFailedStunAddr);
+
+    rtc::Network ipv4_network1("test_eth0", "Test Network Adapter 1",
+                               rtc::IPAddress(0x12345600U), 24);
+    ipv4_network1.AddIP(rtc::IPAddress(0x12345678));
+    rtc::NetworkManager::NetworkList networks;
+    networks.push_back(&ipv4_network1);
+
+    rtc::scoped_ptr<rtc::BasicPacketSocketFactory> socket_factory(
+        new rtc::BasicPacketSocketFactory());
+
+    // Set up the expected results for verification.
+    std::set<std::string> srflx_addresses;
+    srflx_addresses.insert(kStunMappedAddr.ToString());
+    const uint32_t total_pings_tried =
+        static_cast<uint32_t>(pings_per_ip * addrs.size());
+
+    // The reported total_pings should not count for pings sent to the
+    // kFailedStunAddr.
+    const uint32_t total_pings_reported = total_pings_tried - pings_per_ip;
+
+    StartProbing(socket_factory.get(), addrs, networks, shared_mode, 3,
+                 pings_per_ip);
+
+    WAIT(stopped_, 1000);
+
+    StunProber::Stats stats;
+    EXPECT_TRUE(prober->GetStats(&stats));
+    EXPECT_EQ(stats.success_percent, 100);
+    EXPECT_TRUE(stats.nat_type > stunprober::NATTYPE_NONE);
+    EXPECT_EQ(stats.srflx_addrs, srflx_addresses);
+    EXPECT_EQ(static_cast<uint32_t>(stats.num_request_sent),
+              total_pings_reported);
+    EXPECT_EQ(static_cast<uint32_t>(stats.num_response_received),
+              total_pings_reported);
+  }
+
+ private:
+  void StopCallback(StunProber* prober, int result) {
+    EXPECT_EQ(result, result_);
+    stopped_ = true;
+  }
+
+  rtc::Thread* main_;
+  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
+  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  rtc::SocketServerScope ss_scope_;
+  rtc::scoped_ptr<StunProber> prober;
+  int result_ = 0;
+  bool stopped_ = false;
+  rtc::scoped_ptr<cricket::TestStunServer> stun_server_1_;
+  rtc::scoped_ptr<cricket::TestStunServer> stun_server_2_;
+};
+
+TEST_F(StunProberTest, NonSharedMode) {
+  RunProber(false);
+}
+
+TEST_F(StunProberTest, SharedMode) {
+  RunProber(true);
+}
+
+}  // namespace stunprober
diff --git a/media/webrtc/trunk/webrtc/sound/OWNERS b/media/webrtc/trunk/webrtc/sound/OWNERS
index b6d6626934..0f00d1aa48 100644
--- a/media/webrtc/trunk/webrtc/sound/OWNERS
+++ b/media/webrtc/trunk/webrtc/sound/OWNERS
@@ -9,5 +9,9 @@ pthatcher@webrtc.org
 sergeyu@chromium.org
 tommi@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
 
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/sound/alsasoundsystem.cc b/media/webrtc/trunk/webrtc/sound/alsasoundsystem.cc
index 867af440a0..696ff1e450 100644
--- a/media/webrtc/trunk/webrtc/sound/alsasoundsystem.cc
+++ b/media/webrtc/trunk/webrtc/sound/alsasoundsystem.cc
@@ -11,15 +11,18 @@
 #include "webrtc/sound/alsasoundsystem.h"
 
 #include <algorithm>
-#include "webrtc/sound/sounddevicelocator.h"
-#include "webrtc/sound/soundinputstreaminterface.h"
-#include "webrtc/sound/soundoutputstreaminterface.h"
+#include <string>
+
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/worker.h"
+#include "webrtc/sound/sounddevicelocator.h"
+#include "webrtc/sound/soundinputstreaminterface.h"
+#include "webrtc/sound/soundoutputstreaminterface.h"
 
 namespace rtc {
 
@@ -62,7 +65,7 @@ class AlsaDeviceLocator : public SoundDeviceLocator {
                                &name_);
   }
 
-  virtual SoundDeviceLocator *Copy() const {
+  SoundDeviceLocator *Copy() const override {
     return new AlsaDeviceLocator(*this);
   }
 };
@@ -223,7 +226,7 @@ class AlsaStream {
   int flags_;
   int freq_;
 
-  DISALLOW_COPY_AND_ASSIGN(AlsaStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaStream);
 };
 
 // Implementation of an input stream. See soundinputstreaminterface.h regarding
@@ -242,46 +245,46 @@ class AlsaInputStream :
         buffer_size_(0) {
   }
 
-  virtual ~AlsaInputStream() {
+  ~AlsaInputStream() override {
     bool success = StopReading();
     // We need that to live.
     VERIFY(success);
   }
 
-  virtual bool StartReading() {
+  bool StartReading() override {
     return StartWork();
   }
 
-  virtual bool StopReading() {
+  bool StopReading() override {
     return StopWork();
   }
 
-  virtual bool GetVolume(int *volume) {
-    // TODO: Implement this.
+  bool GetVolume(int *volume) override {
+    // TODO(henrika): Implement this.
     return false;
   }
 
-  virtual bool SetVolume(int volume) {
-    // TODO: Implement this.
+  bool SetVolume(int volume) override {
+    // TODO(henrika): Implement this.
     return false;
   }
 
-  virtual bool Close() {
+  bool Close() override {
     return StopReading() && stream_.Close();
   }
 
-  virtual int LatencyUsecs() {
+  int LatencyUsecs() override {
     return stream_.CurrentDelayUsecs();
   }
 
  private:
   // Inherited from Worker.
-  virtual void OnStart() {
+  void OnStart() override {
     HaveWork();
   }
 
   // Inherited from Worker.
-  virtual void OnHaveWork() {
+  void OnHaveWork() override {
     // Block waiting for data.
     snd_pcm_uframes_t avail = stream_.Wait();
     if (avail > 0) {
@@ -317,7 +320,7 @@ class AlsaInputStream :
   }
 
   // Inherited from Worker.
-  virtual void OnStop() {
+  void OnStop() override {
     // Nothing to do.
   }
 
@@ -329,14 +332,13 @@ class AlsaInputStream :
   rtc::scoped_ptr<char[]> buffer_;
   size_t buffer_size_;
 
-  DISALLOW_COPY_AND_ASSIGN(AlsaInputStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaInputStream);
 };
 
 // Implementation of an output stream. See soundoutputstreaminterface.h
 // regarding thread-safety.
-class AlsaOutputStream :
-    public SoundOutputStreamInterface,
-    private rtc::Worker {
+class AlsaOutputStream : public SoundOutputStreamInterface,
+                         private rtc::Worker {
  public:
   AlsaOutputStream(AlsaSoundSystem *alsa,
                    snd_pcm_t *handle,
@@ -347,22 +349,21 @@ class AlsaOutputStream :
       : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq) {
   }
 
-  virtual ~AlsaOutputStream() {
+  ~AlsaOutputStream() override {
     bool success = DisableBufferMonitoring();
     // We need that to live.
     VERIFY(success);
   }
 
-  virtual bool EnableBufferMonitoring() {
+  bool EnableBufferMonitoring() override {
     return StartWork();
   }
 
-  virtual bool DisableBufferMonitoring() {
+  bool DisableBufferMonitoring() override {
     return StopWork();
   }
 
-  virtual bool WriteSamples(const void *sample_data,
-                            size_t size) {
+  bool WriteSamples(const void *sample_data, size_t size) override {
     if (size % stream_.frame_size() != 0) {
       // No client of SoundSystemInterface does this, so let's not support it.
       // (If we wanted to support it, we'd basically just buffer the fractional
@@ -389,32 +390,32 @@ class AlsaOutputStream :
     return true;
   }
 
-  virtual bool GetVolume(int *volume) {
-    // TODO: Implement this.
+  bool GetVolume(int *volume) override {
+    // TODO(henrika): Implement this.
     return false;
   }
 
-  virtual bool SetVolume(int volume) {
-    // TODO: Implement this.
+  bool SetVolume(int volume) override {
+    // TODO(henrika): Implement this.
     return false;
   }
 
-  virtual bool Close() {
+  bool Close() override {
     return DisableBufferMonitoring() && stream_.Close();
   }
 
-  virtual int LatencyUsecs() {
+  int LatencyUsecs() override {
     return stream_.CurrentDelayUsecs();
   }
 
  private:
   // Inherited from Worker.
-  virtual void OnStart() {
+  void OnStart() override {
     HaveWork();
   }
 
   // Inherited from Worker.
-  virtual void OnHaveWork() {
+  void OnHaveWork() override {
     snd_pcm_uframes_t avail = stream_.Wait();
     if (avail > 0) {
       size_t space = avail * stream_.frame_size();
@@ -424,7 +425,7 @@ class AlsaOutputStream :
   }
 
   // Inherited from Worker.
-  virtual void OnStop() {
+  void OnStop() override {
     // Nothing to do.
   }
 
@@ -434,7 +435,7 @@ class AlsaOutputStream :
 
   AlsaStream stream_;
 
-  DISALLOW_COPY_AND_ASSIGN(AlsaOutputStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaOutputStream);
 };
 
 AlsaSoundSystem::AlsaSoundSystem() : initialized_(false) {}
@@ -569,7 +570,6 @@ bool AlsaSoundSystem::EnumerateDevices(
         strcmp(name, ignore_null) != 0 &&
         strcmp(name, ignore_pulse) != 0 &&
         !rtc::starts_with(name, ignore_prefix)) {
-
       // Yes, we do.
       char *desc = symbol_table_.snd_device_name_get_hint()(*list, "DESC");
       if (!desc) {
@@ -608,8 +608,6 @@ bool AlsaSoundSystem::GetDefaultDevice(SoundDeviceLocator **device) {
 }
 
 inline size_t AlsaSoundSystem::FrameSize(const OpenParams &params) {
-  ASSERT(static_cast<int>(params.format) <
-         ARRAY_SIZE(kCricketFormatToSampleSizeTable));
   return kCricketFormatToSampleSizeTable[params.format] * params.channels;
 }
 
@@ -624,7 +622,6 @@ StreamInterface *AlsaSoundSystem::OpenDevice(
         int wait_timeout_ms,
         int flags,
         int freq)) {
-
   if (!IsInitialized()) {
     return NULL;
   }
@@ -664,8 +661,7 @@ StreamInterface *AlsaSoundSystem::OpenDevice(
     latency = std::max(latency, kMinimumLatencyUsecs);
   }
 
-  ASSERT(static_cast<int>(params.format) <
-         ARRAY_SIZE(kCricketFormatToAlsaFormatTable));
+  ASSERT(params.format < arraysize(kCricketFormatToAlsaFormatTable));
 
   err = symbol_table_.snd_pcm_set_params()(
       handle,
diff --git a/media/webrtc/trunk/webrtc/sound/alsasoundsystem.h b/media/webrtc/trunk/webrtc/sound/alsasoundsystem.h
index f95e68618d..dbf34d178b 100644
--- a/media/webrtc/trunk/webrtc/sound/alsasoundsystem.h
+++ b/media/webrtc/trunk/webrtc/sound/alsasoundsystem.h
@@ -34,25 +34,25 @@ class AlsaSoundSystem : public SoundSystemInterface {
 
   AlsaSoundSystem();
 
-  virtual ~AlsaSoundSystem();
+  ~AlsaSoundSystem() override;
 
-  virtual bool Init();
-  virtual void Terminate();
+  bool Init() override;
+  void Terminate() override;
 
-  virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
-  virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
+  bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices) override;
+  bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices) override;
 
-  virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
-  virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
+  bool GetDefaultPlaybackDevice(SoundDeviceLocator **device) override;
+  bool GetDefaultCaptureDevice(SoundDeviceLocator **device) override;
 
-  virtual SoundOutputStreamInterface *OpenPlaybackDevice(
+  SoundOutputStreamInterface *OpenPlaybackDevice(
       const SoundDeviceLocator *device,
-      const OpenParams &params);
-  virtual SoundInputStreamInterface *OpenCaptureDevice(
+      const OpenParams &params) override;
+  SoundInputStreamInterface *OpenCaptureDevice(
       const SoundDeviceLocator *device,
-      const OpenParams &params);
+      const OpenParams &params) override;
 
-  virtual const char *GetName() const;
+  const char *GetName() const override;
 
  private:
   bool IsInitialized() { return initialized_; }
@@ -95,7 +95,7 @@ class AlsaSoundSystem : public SoundSystemInterface {
   bool initialized_;
   AlsaSymbolTable symbol_table_;
 
-  DISALLOW_COPY_AND_ASSIGN(AlsaSoundSystem);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AlsaSoundSystem);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/automaticallychosensoundsystem_unittest.cc b/media/webrtc/trunk/webrtc/sound/automaticallychosensoundsystem_unittest.cc
index 5cfd7c6fc7..318385c651 100644
--- a/media/webrtc/trunk/webrtc/sound/automaticallychosensoundsystem_unittest.cc
+++ b/media/webrtc/trunk/webrtc/sound/automaticallychosensoundsystem_unittest.cc
@@ -9,8 +9,10 @@
  */
 
 #include "webrtc/sound/automaticallychosensoundsystem.h"
-#include "webrtc/sound/nullsoundsystem.h"
+
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/sound/nullsoundsystem.h"
 
 namespace rtc {
 
@@ -112,7 +114,7 @@ extern const SoundSystemCreator kSingleSystemFailingCreators[] = {
 TEST(AutomaticallyChosenSoundSystem, SingleSystemFailing) {
   AutomaticallyChosenSoundSystem<
       kSingleSystemFailingCreators,
-      ARRAY_SIZE(kSingleSystemFailingCreators)> sound_system;
+      arraysize(kSingleSystemFailingCreators)> sound_system;
   EXPECT_FALSE(sound_system.Init());
 }
 
@@ -123,7 +125,7 @@ extern const SoundSystemCreator kSingleSystemSucceedingCreators[] = {
 TEST(AutomaticallyChosenSoundSystem, SingleSystemSucceeding) {
   AutomaticallyChosenSoundSystem<
       kSingleSystemSucceedingCreators,
-      ARRAY_SIZE(kSingleSystemSucceedingCreators)> sound_system;
+      arraysize(kSingleSystemSucceedingCreators)> sound_system;
   EXPECT_TRUE(sound_system.Init());
 }
 
@@ -136,7 +138,7 @@ extern const SoundSystemCreator
 TEST(AutomaticallyChosenSoundSystem, FailedFirstSystemResultsInUsingSecond) {
   AutomaticallyChosenSoundSystem<
       kFailedFirstSystemResultsInUsingSecondCreators,
-      ARRAY_SIZE(kFailedFirstSystemResultsInUsingSecondCreators)> sound_system;
+      arraysize(kFailedFirstSystemResultsInUsingSecondCreators)> sound_system;
   EXPECT_TRUE(sound_system.Init());
 }
 
@@ -148,7 +150,7 @@ extern const SoundSystemCreator kEarlierEntriesHavePriorityCreators[] = {
 TEST(AutomaticallyChosenSoundSystem, EarlierEntriesHavePriority) {
   AutomaticallyChosenSoundSystem<
       kEarlierEntriesHavePriorityCreators,
-      ARRAY_SIZE(kEarlierEntriesHavePriorityCreators)> sound_system;
+      arraysize(kEarlierEntriesHavePriorityCreators)> sound_system;
   InitCheckingSoundSystem1::created_ = false;
   InitCheckingSoundSystem2::created_ = false;
   EXPECT_TRUE(sound_system.Init());
@@ -169,7 +171,7 @@ extern const SoundSystemCreator kManySoundSystemsCreators[] = {
 TEST(AutomaticallyChosenSoundSystem, ManySoundSystems) {
   AutomaticallyChosenSoundSystem<
       kManySoundSystemsCreators,
-      ARRAY_SIZE(kManySoundSystemsCreators)> sound_system;
+      arraysize(kManySoundSystemsCreators)> sound_system;
   EXPECT_TRUE(sound_system.Init());
 }
 
@@ -182,7 +184,7 @@ extern const SoundSystemCreator kDeletesAllCreatedSoundSystemsCreators[] = {
 TEST(AutomaticallyChosenSoundSystem, DeletesAllCreatedSoundSystems) {
   typedef AutomaticallyChosenSoundSystem<
       kDeletesAllCreatedSoundSystemsCreators,
-      ARRAY_SIZE(kDeletesAllCreatedSoundSystemsCreators)> TestSoundSystem;
+      arraysize(kDeletesAllCreatedSoundSystemsCreators)> TestSoundSystem;
   TestSoundSystem *sound_system = new TestSoundSystem();
   DeletionCheckingSoundSystem1::deleted_ = false;
   DeletionCheckingSoundSystem2::deleted_ = false;
diff --git a/media/webrtc/trunk/webrtc/sound/linuxsoundsystem.h b/media/webrtc/trunk/webrtc/sound/linuxsoundsystem.h
index 0016f8a428..56721a1faf 100644
--- a/media/webrtc/trunk/webrtc/sound/linuxsoundsystem.h
+++ b/media/webrtc/trunk/webrtc/sound/linuxsoundsystem.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_SOUND_LINUXSOUNDSYSTEM_H_
 #define WEBRTC_SOUND_LINUXSOUNDSYSTEM_H_
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/sound/automaticallychosensoundsystem.h"
 
 namespace rtc {
@@ -34,7 +35,7 @@ extern const SoundSystemCreator kLinuxSoundSystemCreators[
 // initializes then we choose that. Otherwise we choose ALSA.
 typedef AutomaticallyChosenSoundSystem<
     kLinuxSoundSystemCreators,
-    ARRAY_SIZE(kLinuxSoundSystemCreators)> LinuxSoundSystem;
+    arraysize(kLinuxSoundSystemCreators)> LinuxSoundSystem;
 
 }  // namespace rtc
 
diff --git a/media/webrtc/trunk/webrtc/sound/nullsoundsystem.cc b/media/webrtc/trunk/webrtc/sound/nullsoundsystem.cc
index 962f410572..6f908c9fc3 100644
--- a/media/webrtc/trunk/webrtc/sound/nullsoundsystem.cc
+++ b/media/webrtc/trunk/webrtc/sound/nullsoundsystem.cc
@@ -16,9 +16,7 @@
 #include "webrtc/base/logging.h"
 
 namespace rtc {
-
 class Thread;
-
 }
 
 namespace rtc {
@@ -30,69 +28,68 @@ class NullSoundDeviceLocator : public SoundDeviceLocator {
  public:
   NullSoundDeviceLocator() : SoundDeviceLocator(kNullName, kNullName) {}
 
-  virtual SoundDeviceLocator *Copy() const {
+  SoundDeviceLocator *Copy() const override {
     return new NullSoundDeviceLocator();
   }
 };
 
 class NullSoundInputStream : public SoundInputStreamInterface {
  public:
-  virtual bool StartReading() {
+  bool StartReading() override {
     return true;
   }
 
-  virtual bool StopReading() {
+  bool StopReading() override {
     return true;
   }
 
-  virtual bool GetVolume(int *volume) {
+  bool GetVolume(int *volume) override {
     *volume = SoundSystemInterface::kMinVolume;
     return true;
   }
 
-  virtual bool SetVolume(int volume) {
+  bool SetVolume(int volume) override {
     return false;
   }
 
-  virtual bool Close() {
+  bool Close() override {
     return true;
   }
 
-  virtual int LatencyUsecs() {
+  int LatencyUsecs() override {
     return 0;
   }
 };
 
 class NullSoundOutputStream : public SoundOutputStreamInterface {
  public:
-  virtual bool EnableBufferMonitoring() {
+  bool EnableBufferMonitoring() override {
     return true;
   }
 
-  virtual bool DisableBufferMonitoring() {
+  bool DisableBufferMonitoring() override {
     return true;
   }
 
-  virtual bool WriteSamples(const void *sample_data,
-                            size_t size) {
+  bool WriteSamples(const void *sample_data, size_t size) override {
     LOG(LS_VERBOSE) << "Got " << size << " bytes of playback samples";
     return true;
   }
 
-  virtual bool GetVolume(int *volume) {
+  bool GetVolume(int *volume) override {
     *volume = SoundSystemInterface::kMinVolume;
     return true;
   }
 
-  virtual bool SetVolume(int volume) {
+  bool SetVolume(int volume) override {
     return false;
   }
 
-  virtual bool Close() {
+  bool Close() override {
     return true;
   }
 
-  virtual int LatencyUsecs() {
+  int LatencyUsecs() override {
     return 0;
   }
 };
diff --git a/media/webrtc/trunk/webrtc/sound/nullsoundsystem.h b/media/webrtc/trunk/webrtc/sound/nullsoundsystem.h
index 6b74997665..08ffad1015 100644
--- a/media/webrtc/trunk/webrtc/sound/nullsoundsystem.h
+++ b/media/webrtc/trunk/webrtc/sound/nullsoundsystem.h
@@ -27,25 +27,25 @@ class NullSoundSystem : public SoundSystemInterface {
     return new NullSoundSystem();
   }
 
-  virtual ~NullSoundSystem();
+  ~NullSoundSystem() override;
 
-  virtual bool Init();
-  virtual void Terminate();
+  bool Init() override;
+  void Terminate() override;
 
-  virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
-  virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
+  bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices) override;
+  bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices) override;
 
-  virtual SoundOutputStreamInterface *OpenPlaybackDevice(
+  SoundOutputStreamInterface *OpenPlaybackDevice(
       const SoundDeviceLocator *device,
-      const OpenParams &params);
-  virtual SoundInputStreamInterface *OpenCaptureDevice(
+      const OpenParams &params) override;
+  SoundInputStreamInterface *OpenCaptureDevice(
       const SoundDeviceLocator *device,
-      const OpenParams &params);
+      const OpenParams &params) override;
 
-  virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
-  virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
+  bool GetDefaultPlaybackDevice(SoundDeviceLocator **device) override;
+  bool GetDefaultCaptureDevice(SoundDeviceLocator **device) override;
 
-  virtual const char *GetName() const;
+  const char *GetName() const override;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/nullsoundsystemfactory.h b/media/webrtc/trunk/webrtc/sound/nullsoundsystemfactory.h
index 8bdb46394b..a14490cfe9 100644
--- a/media/webrtc/trunk/webrtc/sound/nullsoundsystemfactory.h
+++ b/media/webrtc/trunk/webrtc/sound/nullsoundsystemfactory.h
@@ -20,12 +20,12 @@ namespace rtc {
 class NullSoundSystemFactory : public SoundSystemFactory {
  public:
   NullSoundSystemFactory();
-  virtual ~NullSoundSystemFactory();
+  ~NullSoundSystemFactory() override;
 
  protected:
   // Inherited from SoundSystemFactory.
-  virtual bool SetupInstance();
-  virtual void CleanupInstance();
+  bool SetupInstance() override;
+  void CleanupInstance() override;
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/platformsoundsystemfactory.h b/media/webrtc/trunk/webrtc/sound/platformsoundsystemfactory.h
index c5105ef051..5319fa50b6 100644
--- a/media/webrtc/trunk/webrtc/sound/platformsoundsystemfactory.h
+++ b/media/webrtc/trunk/webrtc/sound/platformsoundsystemfactory.h
@@ -20,16 +20,14 @@ namespace rtc {
 class PlatformSoundSystemFactory : public SoundSystemFactory {
  public:
   PlatformSoundSystemFactory();
-  virtual ~PlatformSoundSystemFactory();
+  ~PlatformSoundSystemFactory() override;
 
  protected:
   // Inherited from SoundSystemFactory.
-  virtual bool SetupInstance();
-  virtual void CleanupInstance();
+  bool SetupInstance() override;
+  void CleanupInstance() override;
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_SOUND_PLATFORMSOUNDSYSTEMFACTORY_H_
-
-
diff --git a/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.cc b/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.cc
index abc6d500cb..15da76c583 100644
--- a/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.cc
+++ b/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.cc
@@ -13,14 +13,17 @@
 #ifdef HAVE_LIBPULSE
 
 #include <algorithm>
-#include "webrtc/sound/sounddevicelocator.h"
-#include "webrtc/sound/soundinputstreaminterface.h"
-#include "webrtc/sound/soundoutputstreaminterface.h"
+#include <string>
+
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/fileutils.h"  // for GetApplicationName()
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/worker.h"
+#include "webrtc/sound/sounddevicelocator.h"
+#include "webrtc/sound/soundinputstreaminterface.h"
+#include "webrtc/sound/soundoutputstreaminterface.h"
 
 namespace rtc {
 
@@ -206,7 +209,7 @@ class PulseAudioStream {
   pa_stream *stream_;
   int flags_;
 
-  DISALLOW_COPY_AND_ASSIGN(PulseAudioStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PulseAudioStream);
 };
 
 // Implementation of an input stream. See soundinputstreaminterface.h regarding
@@ -214,17 +217,6 @@ class PulseAudioStream {
 class PulseAudioInputStream :
     public SoundInputStreamInterface,
     private rtc::Worker {
-
-  struct GetVolumeCallbackData {
-    PulseAudioInputStream *instance;
-    pa_cvolume *channel_volumes;
-  };
-
-  struct GetSourceChannelCountCallbackData {
-    PulseAudioInputStream *instance;
-    uint8_t *channels;
-  };
-
  public:
   PulseAudioInputStream(PulseAudioSoundSystem *pulse,
                         pa_stream *stream,
@@ -384,6 +376,16 @@ class PulseAudioInputStream :
   }
 
  private:
+  struct GetVolumeCallbackData {
+    PulseAudioInputStream* instance;
+    pa_cvolume* channel_volumes;
+  };
+
+  struct GetSourceChannelCountCallbackData {
+    PulseAudioInputStream* instance;
+    uint8_t* channels;
+  };
+
   void Lock() {
     stream_.Lock();
   }
@@ -570,7 +572,7 @@ class PulseAudioInputStream :
   const void *temp_sample_data_;
   size_t temp_sample_data_size_;
 
-  DISALLOW_COPY_AND_ASSIGN(PulseAudioInputStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PulseAudioInputStream);
 };
 
 // Implementation of an output stream. See soundoutputstreaminterface.h
@@ -578,12 +580,6 @@ class PulseAudioInputStream :
 class PulseAudioOutputStream :
     public SoundOutputStreamInterface,
     private rtc::Worker {
-
-  struct GetVolumeCallbackData {
-    PulseAudioOutputStream *instance;
-    pa_cvolume *channel_volumes;
-  };
-
  public:
   PulseAudioOutputStream(PulseAudioSoundSystem *pulse,
                          pa_stream *stream,
@@ -731,7 +727,7 @@ class PulseAudioOutputStream :
   }
 
 #if 0
-  // TODO: Versions 0.9.16 and later of Pulse have a new API for
+  // TODO(henrika): Versions 0.9.16 and later of Pulse have a new API for
   // zero-copy writes, but Hardy is not new enough to have that so we can't
   // rely on it. Perhaps auto-detect if it's present or not and use it if we
   // can?
@@ -775,6 +771,11 @@ class PulseAudioOutputStream :
 #endif
 
  private:
+  struct GetVolumeCallbackData {
+    PulseAudioOutputStream* instance;
+    pa_cvolume* channel_volumes;
+  };
+
   void Lock() {
     stream_.Lock();
   }
@@ -954,7 +955,7 @@ class PulseAudioOutputStream :
   // Temporary storage for passing data between threads.
   size_t temp_buffer_space_;
 
-  DISALLOW_COPY_AND_ASSIGN(PulseAudioOutputStream);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PulseAudioOutputStream);
 };
 
 PulseAudioSoundSystem::PulseAudioSoundSystem()
@@ -1163,7 +1164,7 @@ bool PulseAudioSoundSystem::ConnectToPulse(pa_context *context) {
 pa_context *PulseAudioSoundSystem::CreateNewConnection() {
   // Create connection context.
   std::string app_name;
-  // TODO: Pulse etiquette says this name should be localized. Do
+  // TODO(henrika): Pulse etiquette says this name should be localized. Do
   // we care?
   rtc::Filesystem::GetApplicationName(&app_name);
   pa_context *context = symbol_table_.pa_context_new()(
@@ -1373,7 +1374,7 @@ StreamInterface *PulseAudioSoundSystem::OpenDevice(
 
   StreamInterface *stream_interface = NULL;
 
-  ASSERT(params.format < ARRAY_SIZE(kCricketFormatToPulseFormatTable));
+  ASSERT(params.format < arraysize(kCricketFormatToPulseFormatTable));
 
   pa_sample_spec spec;
   spec.format = kCricketFormatToPulseFormatTable[params.format];
diff --git a/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.h b/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.h
index 4e67acc0ac..895b784552 100644
--- a/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.h
+++ b/media/webrtc/trunk/webrtc/sound/pulseaudiosoundsystem.h
@@ -167,7 +167,7 @@ class PulseAudioSoundSystem : public SoundSystemInterface {
   pa_context *context_;
   PulseAudioSymbolTable symbol_table_;
 
-  DISALLOW_COPY_AND_ASSIGN(PulseAudioSoundSystem);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PulseAudioSoundSystem);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/sound.gyp b/media/webrtc/trunk/webrtc/sound/sound.gyp
index a7d929b25d..e09215371b 100644
--- a/media/webrtc/trunk/webrtc/sound/sound.gyp
+++ b/media/webrtc/trunk/webrtc/sound/sound.gyp
@@ -26,7 +26,9 @@
         'platformsoundsystemfactory.cc',
         'platformsoundsystemfactory.h',
         'sounddevicelocator.h',
+        'soundinputstreaminterface.cc',
         'soundinputstreaminterface.h',
+        'soundoutputstreaminterface.cc',
         'soundoutputstreaminterface.h',
         'soundsystemfactory.h',
         'soundsysteminterface.cc',
diff --git a/media/webrtc/trunk/webrtc/sound/sounddevicelocator.h b/media/webrtc/trunk/webrtc/sound/sounddevicelocator.h
index 4e8e1485ab..f8a6cf8802 100644
--- a/media/webrtc/trunk/webrtc/sound/sounddevicelocator.h
+++ b/media/webrtc/trunk/webrtc/sound/sounddevicelocator.h
@@ -46,7 +46,7 @@ class SoundDeviceLocator {
   std::string device_name_;
 
  private:
-  DISALLOW_ASSIGN(SoundDeviceLocator);
+  RTC_DISALLOW_ASSIGN(SoundDeviceLocator);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/overrides/webrtc/base/arraysize.h b/media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.cc
similarity index 55%
rename from media/webrtc/trunk/webrtc/overrides/webrtc/base/arraysize.h
rename to media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.cc
index abc1240fe5..b7bf0744ed 100644
--- a/media/webrtc/trunk/webrtc/overrides/webrtc/base/arraysize.h
+++ b/media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.cc
@@ -8,13 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// This file overrides the inclusion of webrtc/base/arraysize.h
-// We do this because in Chromium it redefines arraysize, which is already
-// defined in base/macros.h.
+#include "webrtc/sound/soundinputstreaminterface.h"
+namespace rtc {
 
-#ifndef OVERRIDES_WEBRTC_BASE_ARRAYSIZE_H__
-#define OVERRIDES_WEBRTC_BASE_ARRAYSIZE_H__
+SoundInputStreamInterface::~SoundInputStreamInterface() {}
 
-#include "base/macros.h"
+SoundInputStreamInterface::SoundInputStreamInterface() {}
 
-#endif  // OVERRIDES_WEBRTC_BASE_ARRAYSIZE_H__
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.h b/media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.h
index 6ce9446d8d..576ff71dd2 100644
--- a/media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.h
+++ b/media/webrtc/trunk/webrtc/sound/soundinputstreaminterface.h
@@ -21,7 +21,7 @@ namespace rtc {
 // for rtc::Worker.
 class SoundInputStreamInterface {
  public:
-  virtual ~SoundInputStreamInterface() {}
+  virtual ~SoundInputStreamInterface();
 
   // Starts the reading of samples on the current thread.
   virtual bool StartReading() = 0;
@@ -57,12 +57,12 @@ class SoundInputStreamInterface {
       SoundInputStreamInterface *> SignalSamplesRead;
 
  protected:
-  SoundInputStreamInterface() {}
+  SoundInputStreamInterface();
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(SoundInputStreamInterface);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SoundInputStreamInterface);
 };
 
 }  // namespace rtc
 
-#endif  // WEBRTC_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
+#endif  // WEBRTC_SOUND_SOUNDINPUTSTREAMINTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event_tracer.cc b/media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.cc
similarity index 54%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/event_tracer.cc
rename to media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.cc
index 9328e80036..c40f1d70d9 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event_tracer.cc
+++ b/media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,5 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// This file has moved.
-// TODO(tommi): Delete after removing dependencies and updating Chromium.
+#include "webrtc/sound/soundoutputstreaminterface.h"
+
+namespace rtc {
+
+SoundOutputStreamInterface::~SoundOutputStreamInterface() {}
+
+SoundOutputStreamInterface::SoundOutputStreamInterface() {}
+
+}  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.h b/media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.h
index 2b501d658e..a94147b838 100644
--- a/media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.h
+++ b/media/webrtc/trunk/webrtc/sound/soundoutputstreaminterface.h
@@ -21,7 +21,7 @@ namespace rtc {
 // DisableBufferMonitoring() are the same as for rtc::Worker.
 class SoundOutputStreamInterface {
  public:
-  virtual ~SoundOutputStreamInterface() {}
+  virtual ~SoundOutputStreamInterface();
 
   // Enables monitoring the available buffer space on the current thread.
   virtual bool EnableBufferMonitoring() = 0;
@@ -61,10 +61,10 @@ class SoundOutputStreamInterface {
   sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace;
 
  protected:
-  SoundOutputStreamInterface() {}
+  SoundOutputStreamInterface();
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/soundsysteminterface.h b/media/webrtc/trunk/webrtc/sound/soundsysteminterface.h
index aa9a53a93c..c386b07a4e 100644
--- a/media/webrtc/trunk/webrtc/sound/soundsysteminterface.h
+++ b/media/webrtc/trunk/webrtc/sound/soundsysteminterface.h
@@ -66,7 +66,7 @@ class SoundSystemInterface {
   // a slightly higher one in the event that the true minimum requires an
   // undesirable trade-off.
   static const int kLowLatency = 0;
- 
+
   // Max value for the volume parameters for Sound(Input|Output)StreamInterface.
   static const int kMaxVolume = 255;
   // Min value for the volume parameters for Sound(Input|Output)StreamInterface.
@@ -104,7 +104,7 @@ class SoundSystemInterface {
   SoundSystemInterface() {}
 
  private:
-  DISALLOW_COPY_AND_ASSIGN(SoundSystemInterface);
+  RTC_DISALLOW_COPY_AND_ASSIGN(SoundSystemInterface);
 };
 
 }  // namespace rtc
diff --git a/media/webrtc/trunk/webrtc/sound/soundsystemproxy.h b/media/webrtc/trunk/webrtc/sound/soundsystemproxy.h
index d13cf15b74..19696b1fef 100644
--- a/media/webrtc/trunk/webrtc/sound/soundsystemproxy.h
+++ b/media/webrtc/trunk/webrtc/sound/soundsystemproxy.h
@@ -25,18 +25,18 @@ class SoundSystemProxy : public SoundSystemInterface {
 
   // Each of these methods simply defers to wrapped_ if non-NULL, else fails.
 
-  virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
-  virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
+  bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices) override;
+  bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices) override;
 
-  virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
-  virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
+  bool GetDefaultPlaybackDevice(SoundDeviceLocator **device) override;
+  bool GetDefaultCaptureDevice(SoundDeviceLocator **device) override;
 
-  virtual SoundOutputStreamInterface *OpenPlaybackDevice(
+  SoundOutputStreamInterface *OpenPlaybackDevice(
       const SoundDeviceLocator *device,
-      const OpenParams &params);
-  virtual SoundInputStreamInterface *OpenCaptureDevice(
+      const OpenParams &params) override;
+  SoundInputStreamInterface *OpenCaptureDevice(
       const SoundDeviceLocator *device,
-      const OpenParams &params);
+      const OpenParams &params) override;
 
  protected:
   SoundSystemInterface *wrapped_;
diff --git a/media/webrtc/trunk/webrtc/stream.h b/media/webrtc/trunk/webrtc/stream.h
new file mode 100644
index 0000000000..5afab0f200
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/stream.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_STREAM_H_
+#define WEBRTC_STREAM_H_
+
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+enum NetworkState {
+  kNetworkUp,
+  kNetworkDown,
+};
+
+// Common base class for streams.
+class Stream {
+ public:
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+  // Called to notify that network state has changed, so that the stream can
+  // respond, e.g. by pausing or resuming activity.
+  virtual void SignalNetworkState(NetworkState state) = 0;
+  // Called when a RTCP packet is received.
+  virtual bool DeliverRtcp(const uint8_t* packet, size_t length) = 0;
+
+ protected:
+  virtual ~Stream() {}
+};
+
+// Common base class for receive streams.
+class ReceiveStream : public Stream {
+ public:
+  // Called when a RTP packet is received.
+  virtual bool DeliverRtp(const uint8_t* packet,
+                          size_t length,
+                          const PacketTime& packet_time) = 0;
+};
+
+// Common base class for send streams.
+// A tag class that denotes send stream type.
+class SendStream : public Stream {};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_STREAM_H_
diff --git a/media/webrtc/trunk/webrtc/supplement.gypi b/media/webrtc/trunk/webrtc/supplement.gypi
index cf9e6d4b07..3691a54427 100644
--- a/media/webrtc/trunk/webrtc/supplement.gypi
+++ b/media/webrtc/trunk/webrtc/supplement.gypi
@@ -1,10 +1,36 @@
 {
   'variables': {
     'variables': {
-      'webrtc_root%': '<(DEPTH)/webrtc',
+      'webrtc_root%': '<(DEPTH)', # '<(DEPTH)/webrtc',
+      # Override the default (10.6) in Chromium's build/common.gypi.
+      # Needed for ARC and libc++.
+      'mac_deployment_target%': '10.7',
+      # Disable use of sysroot for Linux. It's enabled by default in Chromium,
+      # but it currently lacks the libudev-dev package.
+      # TODO(kjellander): Remove when crbug.com/561584 is fixed.
+      'use_sysroot': 0,
     },
     'webrtc_root%': '<(webrtc_root)',
+    'mac_deployment_target%': '<(mac_deployment_target)',
+    'use_sysroot%': '<(use_sysroot)',
     'build_with_chromium': 0,
+    'conditions': [
+      ['OS=="ios"', {
+        # Default to using BoringSSL on iOS.
+        'use_openssl%': 1,
+
+        # Set target_subarch for if not already set. This is needed because the
+        # Chromium iOS toolchain relies on target_subarch being set.
+        'conditions': [
+          ['target_arch=="arm" or target_arch=="ia32"', {
+            'target_subarch%': 'arm32',
+          }],
+          ['target_arch=="arm64" or target_arch=="x64"', {
+            'target_subarch%': 'arm64',
+          }],
+        ],
+      }],
+    ],
   },
   'target_defaults': {
     'target_conditions': [
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/BUILD.gn b/media/webrtc/trunk/webrtc/system_wrappers/BUILD.gn
index e179020028..5e0e41e832 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/system_wrappers/BUILD.gn
@@ -11,53 +11,49 @@ import("../build/webrtc.gni")
 
 static_library("system_wrappers") {
   sources = [
-    "interface/aligned_array.h",
-    "interface/aligned_malloc.h",
-    "interface/atomic32.h",
-    "interface/clock.h",
-    "interface/condition_variable_wrapper.h",
-    "interface/cpu_info.h",
-    "interface/cpu_features_wrapper.h",
-    "interface/critical_section_wrapper.h",
-    "interface/data_log.h",
-    "interface/data_log_c.h",
-    "interface/data_log_impl.h",
-    "interface/event_tracer.h",
-    "interface/event_wrapper.h",
-    "interface/field_trial.h",
-    "interface/file_wrapper.h",
-    "interface/fix_interlocked_exchange_pointer_win.h",
-    "interface/logging.h",
-    "interface/metrics.h",
-    "interface/ref_count.h",
-    "interface/rtp_to_ntp.h",
-    "interface/rw_lock_wrapper.h",
-    "interface/scoped_refptr.h",
-    "interface/scoped_vector.h",
-    "interface/sleep.h",
-    "interface/sort.h",
-    "interface/static_instance.h",
-    "interface/stl_util.h",
-    "interface/stringize_macros.h",
-    "interface/thread_wrapper.h",
-    "interface/tick_util.h",
-    "interface/timestamp_extrapolator.h",
-    "interface/trace.h",
-    "interface/trace_event.h",
-    "interface/utf_util_win.h",
+    "include/aligned_array.h",
+    "include/aligned_malloc.h",
+    "include/atomic32.h",
+    "include/clock.h",
+    "include/condition_variable_wrapper.h",
+    "include/cpu_features_wrapper.h",
+    "include/cpu_info.h",
+    "include/critical_section_wrapper.h",
+    "include/data_log.h",
+    "include/data_log_c.h",
+    "include/data_log_impl.h",
+    "include/event_wrapper.h",
+    "include/field_trial.h",
+    "include/file_wrapper.h",
+    "include/fix_interlocked_exchange_pointer_win.h",
+    "include/logging.h",
+    "include/metrics.h",
+    "include/ref_count.h",
+    "include/rtp_to_ntp.h",
+    "include/rw_lock_wrapper.h",
+    "include/scoped_vector.h",
+    "include/sleep.h",
+    "include/sort.h",
+    "include/static_instance.h",
+    "include/stl_util.h",
+    "include/stringize_macros.h",
+    "include/tick_util.h",
+    "include/timestamp_extrapolator.h",
+    "include/trace.h",
+    "include/utf_util_win.h",
     "source/aligned_malloc.cc",
     "source/atomic32_mac.cc",
     "source/atomic32_win.cc",
     "source/clock.cc",
     "source/condition_variable.cc",
-    "source/condition_variable_posix.cc",
-    "source/condition_variable_posix.h",
     "source/condition_variable_event_win.cc",
     "source/condition_variable_event_win.h",
     "source/condition_variable_native_win.cc",
     "source/condition_variable_native_win.h",
-    "source/cpu_info.cc",
+    "source/condition_variable_posix.cc",
+    "source/condition_variable_posix.h",
     "source/cpu_features.cc",
+    "source/cpu_info.cc",
     "source/critical_section.cc",
     "source/critical_section_posix.cc",
     "source/critical_section_posix.h",
@@ -65,11 +61,10 @@ static_library("system_wrappers") {
     "source/critical_section_win.h",
     "source/data_log_c.cc",
     "source/event.cc",
-    "source/event_posix.cc",
-    "source/event_posix.h",
-    "source/event_tracer.cc",
-    "source/event_win.cc",
-    "source/event_win.h",
+    "source/event_timer_posix.cc",
+    "source/event_timer_posix.h",
+    "source/event_timer_win.cc",
+    "source/event_timer_win.h",
     "source/file_impl.cc",
     "source/file_impl.h",
     "source/logging.cc",
@@ -84,11 +79,6 @@ static_library("system_wrappers") {
     "source/sleep.cc",
     "source/sort.cc",
     "source/tick_util.cc",
-    "source/thread.cc",
-    "source/thread_posix.cc",
-    "source/thread_posix.h",
-    "source/thread_win.cc",
-    "source/thread_win.h",
     "source/timestamp_extrapolator.cc",
     "source/trace_impl.cc",
     "source/trace_impl.h",
@@ -100,9 +90,7 @@ static_library("system_wrappers") {
 
   configs += [ "..:common_config" ]
 
-  public_configs = [
-    "..:common_inherited_config",
-  ]
+  public_configs = [ "..:common_inherited_config" ]
 
   if (rtc_enable_data_logging) {
     sources += [ "source/data_log.cc" ]
@@ -112,16 +100,19 @@ static_library("system_wrappers") {
 
   defines = []
   libs = []
-  deps = [ "..:webrtc_common" ]
+  deps = [
+    "..:webrtc_common",
+  ]
 
   if (is_android) {
     sources += [
-      "interface/logcat_trace_context.h",
+      "include/logcat_trace_context.h",
       "source/logcat_trace_context.cc",
     ]
 
     defines += [
       "WEBRTC_THREAD_RR",
+
       # TODO(leozwang): Investigate CLOCK_REALTIME and CLOCK_MONOTONIC
       # support on Android. Keep WEBRTC_CLOCK_TYPE_REALTIME for now,
       # remove it after I verify that CLOCK_MONOTONIC is fully functional
@@ -146,9 +137,7 @@ static_library("system_wrappers") {
   }
 
   if (!is_mac && !is_ios) {
-    sources += [
-      "source/atomic32_posix.cc",
-    ]
+    sources += [ "source/atomic32_posix.cc" ]
   }
 
   if (is_ios || is_mac) {
@@ -159,9 +148,7 @@ static_library("system_wrappers") {
   }
 
   if (is_ios) {
-    sources += [
-      "source/atomic32_mac.cc",
-    ]
+    sources += [ "source/atomic32_mac.cc" ]
   }
 
   if (is_win) {
@@ -173,23 +160,17 @@ static_library("system_wrappers") {
     ]
   }
 
-  deps += [
-    "../base:rtc_base_approved",
-  ]
+  deps += [ "../base:rtc_base_approved" ]
 }
 
 source_set("field_trial_default") {
   sources = [
-    "interface/field_trial_default.h",
+    "include/field_trial_default.h",
     "source/field_trial_default.cc",
   ]
 
   configs += [ "..:common_config" ]
   public_configs = [ "..:common_inherited_config" ]
-
-  deps = [
-    ":system_wrappers",
-  ]
 }
 
 source_set("metrics_default") {
@@ -199,20 +180,16 @@ source_set("metrics_default") {
 
   configs += [ "..:common_config" ]
   public_configs = [ "..:common_inherited_config" ]
-
-  deps = [
-    ":system_wrappers",
-  ]
 }
 
 source_set("system_wrappers_default") {
-
   configs += [ "..:common_config" ]
   public_configs = [ "..:common_inherited_config" ]
 
   deps = [
     ":field_trial_default",
     ":metrics_default",
+    ":system_wrappers",
   ]
 }
 
@@ -224,11 +201,8 @@ if (is_android) {
 
     configs += [ "..:common_config" ]
     public_configs = [ "..:common_inherited_config" ]
-
-    if (is_android_webview_build) {
-      libs += [ "cpufeatures.a" ]
-    } else {
-      deps = [ "//third_party/android_tools:cpu_features" ]
-    }
+    deps = [
+      "//third_party/android_tools:cpu_features",
+    ]
   }
 }
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/OWNERS b/media/webrtc/trunk/webrtc/system_wrappers/OWNERS
index 76fdda254e..f55277e8ea 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/OWNERS
+++ b/media/webrtc/trunk/webrtc/system_wrappers/OWNERS
@@ -4,4 +4,9 @@ henrikg@webrtc.org
 mflodman@webrtc.org
 niklas.enbom@webrtc.org
 
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
 per-file BUILD.gn=kjellander@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_chromium.gyp b/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_chromium.gyp
index 96bb6d9c60..519fe44119 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_chromium.gyp
+++ b/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_chromium.gyp
@@ -16,8 +16,8 @@
           'sources': [
             'source/cpu_features_android.c',
           ],
-          'includes': [
-            '../../../build/android/cpufeatures.gypi',
+          'dependencies': [
+            '../../../build/android/ndk.gyp:cpu_features',
           ],
         },
       ],
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_webrtc.gyp b/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_webrtc.gyp
index 8064f280e5..afec6ed3f6 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_webrtc.gyp
+++ b/media/webrtc/trunk/webrtc/system_wrappers/cpu_features_webrtc.gyp
@@ -21,8 +21,8 @@
           ],
           'conditions': [
             ['include_ndk_cpu_features==1', {
-               'includes': [
-                  '../../build/android/cpufeatures.gypi',
+              'includes': [
+                 '../../build/android/cpufeatures.gypi',
                ],
             }, {
               'sources': [
@@ -31,8 +31,7 @@
               ],
             }],
           ],
-        },
-      ],
+      }],
     }],
   ], # conditions
 }
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_array.h b/media/webrtc/trunk/webrtc/system_wrappers/include/aligned_array.h
similarity index 58%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_array.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/aligned_array.h
index 4b5c276d43..a2ffe99c14 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_array.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/aligned_array.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_ARRAY_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_ARRAY_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_ARRAY_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_ARRAY_
 
 #include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 namespace webrtc {
 
@@ -20,21 +20,20 @@ namespace webrtc {
 // aligned to the given byte alignment.
 template<typename T> class AlignedArray {
  public:
-  AlignedArray(int rows, int cols, int alignment)
+  AlignedArray(size_t rows, size_t cols, size_t alignment)
       : rows_(rows),
-        cols_(cols),
-        alignment_(alignment) {
-    CHECK_GT(alignment_, 0);
+        cols_(cols) {
+    RTC_CHECK_GT(alignment, 0u);
     head_row_ = static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_),
-                                               alignment_));
-    for (int i = 0; i < rows_; ++i) {
+                                               alignment));
+    for (size_t i = 0; i < rows_; ++i) {
       head_row_[i] = static_cast<T*>(AlignedMalloc(cols_ * sizeof(**head_row_),
-                                                   alignment_));
+                                                   alignment));
     }
   }
 
   ~AlignedArray() {
-    for (int i = 0; i < rows_; ++i) {
+    for (size_t i = 0; i < rows_; ++i) {
       AlignedFree(head_row_[i]);
     }
     AlignedFree(head_row_);
@@ -48,42 +47,40 @@ template<typename T> class AlignedArray {
     return head_row_;
   }
 
-  T* Row(int row) {
-    CHECK_LE(row, rows_);
+  T* Row(size_t row) {
+    RTC_CHECK_LE(row, rows_);
     return head_row_[row];
   }
 
-  const T* Row(int row) const {
-    CHECK_LE(row, rows_);
+  const T* Row(size_t row) const {
+    RTC_CHECK_LE(row, rows_);
     return head_row_[row];
   }
 
-  T& At(int row, int col) {
-    CHECK_LE(col, cols_);
+  T& At(size_t row, size_t col) {
+    RTC_CHECK_LE(col, cols_);
     return Row(row)[col];
   }
 
-  const T& At(int row, int col) const {
-    CHECK_LE(col, cols_);
+  const T& At(size_t row, size_t col) const {
+    RTC_CHECK_LE(col, cols_);
     return Row(row)[col];
   }
 
-  int rows() const {
+  size_t rows() const {
     return rows_;
   }
 
-  int cols() const {
+  size_t cols() const {
     return cols_;
   }
 
  private:
-  int rows_;
-  int cols_;
-  int alignment_;
+  size_t rows_;
+  size_t cols_;
   T** head_row_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_ARRAY_
-
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_ARRAY_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_malloc.h b/media/webrtc/trunk/webrtc/system_wrappers/include/aligned_malloc.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_malloc.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/aligned_malloc.h
index 5d343cde7c..277abec020 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/aligned_malloc.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/aligned_malloc.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_MALLOC_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_MALLOC_H_
 
 // The functions declared here
 // 1) Allocates block of aligned memory.
@@ -56,4 +56,4 @@ struct AlignedFreeDeleter {
 
 }  // namespace webrtc
 
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_MALLOC_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/asm_defines.h b/media/webrtc/trunk/webrtc/system_wrappers/include/asm_defines.h
similarity index 85%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/asm_defines.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/asm_defines.h
index 2e2bd7a812..2c3c9699e8 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/asm_defines.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/asm_defines.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ASM_DEFINES_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ASM_DEFINES_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ASM_DEFINES_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ASM_DEFINES_H_
 
 #if (defined(__linux__) || defined(__FreeBSD__)) && defined(__ELF__)
 .section .note.GNU-stack,"",%progbits
@@ -38,6 +38,9 @@ bl _\name
 .hidden \name
 .endm
 .macro DEFINE_FUNCTION name
+#if defined(__linux__) && defined(__ELF__)
+.type \name,%function
+#endif
 \name:
 .endm
 .macro CALL_FUNCTION name
@@ -60,4 +63,4 @@ strheq \reg1, \reg2, \num
 
 .text
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ASM_DEFINES_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ASM_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/atomic32.h b/media/webrtc/trunk/webrtc/system_wrappers/include/atomic32.h
similarity index 90%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/atomic32.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/atomic32.h
index 8633e26362..78e649d8b6 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/atomic32.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/atomic32.h
@@ -12,8 +12,8 @@
 // doing, use locks instead! :-)
 //
 // Note: assumes 32-bit (or higher) system
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ATOMIC32_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ATOMIC32_H_
 
 #include <stddef.h>
 
@@ -56,11 +56,11 @@ class Atomic32 {
     return (reinterpret_cast<ptrdiff_t>(&value_) & 3) == 0;
   }
 
-  DISALLOW_COPY_AND_ASSIGN(Atomic32);
+  RTC_DISALLOW_COPY_AND_ASSIGN(Atomic32);
 
   int32_t value_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ATOMIC32_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/clock.h b/media/webrtc/trunk/webrtc/system_wrappers/include/clock.h
similarity index 92%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/clock.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/clock.h
index a103d95155..f443057bea 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/clock.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/clock.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CLOCK_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CLOCK_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -81,4 +81,4 @@ class SimulatedClock : public Clock {
 
 };  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CLOCK_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/compile_assert_c.h b/media/webrtc/trunk/webrtc/system_wrappers/include/compile_assert_c.h
similarity index 80%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/compile_assert_c.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/compile_assert_c.h
index dbb5292d97..b402d7192d 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/compile_assert_c.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/compile_assert_c.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_COMPILE_ASSERT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_COMPILE_ASSERT_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_COMPILE_ASSERT_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_COMPILE_ASSERT_H_
 
 #ifdef __cplusplus
 #error "Only use this for C files. For C++, use static_assert."
@@ -21,4 +21,4 @@
 //   COMPILE_ASSERT(sizeof(foo) < 128);
 #define COMPILE_ASSERT(expression) switch (0) {case 0: case expression:;}
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_COMPILE_ASSERT_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_COMPILE_ASSERT_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/condition_variable_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/include/condition_variable_wrapper.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/condition_variable_wrapper.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/condition_variable_wrapper.h
index 151f00ece1..37ca30f036 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/condition_variable_wrapper.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/condition_variable_wrapper.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CONDITION_VARIABLE_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CONDITION_VARIABLE_WRAPPER_H_
 
 namespace webrtc {
 
@@ -39,4 +39,4 @@ class ConditionVariableWrapper {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CONDITION_VARIABLE_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_features_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/include/cpu_features_wrapper.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_features_wrapper.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/cpu_features_wrapper.h
index 5697c49164..9838d94e58 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_features_wrapper.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/cpu_features_wrapper.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
 
 #if defined(__cplusplus) || defined(c_plusplus)
 extern "C" {
@@ -48,4 +48,4 @@ extern uint64_t WebRtc_GetCPUFeaturesARM(void);
 }  // extern "C"
 #endif
 
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_info.h b/media/webrtc/trunk/webrtc/system_wrappers/include/cpu_info.h
similarity index 74%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_info.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/cpu_info.h
index fa8c38810d..3c00d33ed3 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/cpu_info.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/cpu_info.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_
 
 #include "webrtc/typedefs.h"
 
@@ -21,9 +21,8 @@ class CpuInfo {
 
  private:
   CpuInfo() {}
-  static uint32_t number_of_cores_;
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
+#endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/critical_section_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/include/critical_section_wrapper.h
similarity index 88%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/critical_section_wrapper.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/critical_section_wrapper.h
index e93a249e25..7dd217e40d 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/critical_section_wrapper.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/critical_section_wrapper.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CRITICAL_SECTION_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CRITICAL_SECTION_WRAPPER_H_
 
 // If the critical section is heavily contended it may be beneficial to use
 // read/write locks instead.
@@ -51,4 +51,4 @@ class SCOPED_LOCKABLE CriticalSectionScoped {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CRITICAL_SECTION_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/data_log.h b/media/webrtc/trunk/webrtc/system_wrappers/include/data_log.h
similarity index 95%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/data_log.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/data_log.h
index 9608f2c420..f6cad88e96 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/data_log.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/data_log.h
@@ -28,12 +28,12 @@
 //
 // Table names and column names are case sensitive.
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_H_
 
 #include <string>
 
-#include "webrtc/system_wrappers/interface/data_log_impl.h"
+#include "webrtc/system_wrappers/include/data_log_impl.h"
 
 namespace webrtc {
 
@@ -116,4 +116,4 @@ class DataLog {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_c.h b/media/webrtc/trunk/webrtc/system_wrappers/include/data_log_c.h
similarity index 95%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_c.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/data_log_c.h
index 4ff8329c85..d31e4d972e 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_c.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/data_log_c.h
@@ -12,8 +12,8 @@
 // mapped here except for InsertCell as C does not support templates.
 // See data_log.h for a description of the functions.
 
-#ifndef SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
-#define SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
+#ifndef SRC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_C_H_
+#define SRC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_C_H_
 
 #include <stddef.h>  // size_t
 
@@ -82,4 +82,4 @@ int WebRtcDataLog_NextRow(const char* table_name);
 }  // end of extern "C"
 #endif
 
-#endif  // SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_  // NOLINT
+#endif  // SRC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_C_H_  // NOLINT
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_impl.h b/media/webrtc/trunk/webrtc/system_wrappers/include/data_log_impl.h
similarity index 92%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_impl.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/data_log_impl.h
index 61a4e29754..35519609b9 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/data_log_impl.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/data_log_impl.h
@@ -14,16 +14,16 @@
 // These classes are helper classes used for logging data for offline
 // processing. Data logged with these classes can conveniently be parsed and
 // processed with e.g. Matlab.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_IMPL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_IMPL_H_
 
 #include <map>
 #include <sstream>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -146,10 +146,12 @@ class DataLogImpl {
   int                       counter_;
   TableMap                  tables_;
   EventWrapper*             flush_event_;
-  rtc::scoped_ptr<ThreadWrapper> file_writer_thread_;
+  // This is a scoped_ptr so that we don't have to create threads in the no-op
+  // impl.
+  rtc::scoped_ptr<rtc::PlatformThread> file_writer_thread_;
   RWLockWrapper*            tables_lock_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/event_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/include/event_wrapper.h
similarity index 78%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/event_wrapper.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/event_wrapper.h
index ea686174f2..cc3722bd6b 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/event_wrapper.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/event_wrapper.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_EVENT_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_EVENT_WRAPPER_H_
 
 namespace webrtc {
 enum EventTypeWrapper {
@@ -18,13 +18,15 @@ enum EventTypeWrapper {
   kEventTimeout = 3
 };
 
-#define WEBRTC_EVENT_10_SEC   10000
 #define WEBRTC_EVENT_INFINITE 0xffffffff
 
+class EventTimerWrapper;
+
 class EventWrapper {
  public:
   // Factory method. Constructor disabled.
   static EventWrapper* Create();
+
   virtual ~EventWrapper() {}
 
   // Releases threads who are calling Wait() and has started waiting. Please
@@ -37,21 +39,32 @@ class EventWrapper {
 
   // Puts the calling thread into a wait state. The thread may be released
   // by a Set() call depending on if other threads are waiting and if so on
-  // timing. The thread that was released will call Reset() before leaving
+  // timing. The thread that was released will reset the event before leaving
   // preventing more threads from being released. If multiple threads
   // are waiting for the same Set(), only one (random) thread is guaranteed to
   // be released. It is possible that multiple (random) threads are released
   // Depending on timing.
+  //
+  // |max_time| is the maximum time to wait in milliseconds or
+  // WEBRTC_EVENT_INFINITE to wait infinitely.
   virtual EventTypeWrapper Wait(unsigned long max_time) = 0;
+};
+
+class EventTimerWrapper : public EventWrapper {
+ public:
+  static EventTimerWrapper* Create();
 
   // Starts a timer that will call a non-sticky version of Set() either once
   // or periodically. If the timer is periodic it ensures that there is no
   // drift over time relative to the system clock.
+  //
+  // |time| is in milliseconds.
   virtual bool StartTimer(bool periodic, unsigned long time) = 0;
 
   virtual bool StopTimer() = 0;
 
 };
+
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_EVENT_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial.h b/media/webrtc/trunk/webrtc/system_wrappers/include/field_trial.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/field_trial.h
index 2f116adec0..62fbfd1a50 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/field_trial.h
@@ -8,13 +8,11 @@
 // be found in the AUTHORS file in the root of the source tree.
 //
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FIELD_TRIAL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FIELD_TRIAL_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FIELD_TRIAL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FIELD_TRIAL_H_
 
 #include <string>
 
-#include "webrtc/common_types.h"
-
 // Field trials allow webrtc clients (such as Chrome) to turn on feature code
 // in binaries out in the field and gather information with that.
 //
@@ -67,4 +65,4 @@ std::string FindFullName(const std::string& name);
 }  // namespace field_trial
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FIELD_TRIAL_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FIELD_TRIAL_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial_default.h b/media/webrtc/trunk/webrtc/system_wrappers/include/field_trial_default.h
similarity index 78%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial_default.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/field_trial_default.h
index fafe550dcc..7417ced39d 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/field_trial_default.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/field_trial_default.h
@@ -8,8 +8,8 @@
 // be found in the AUTHORS file in the root of the source tree.
 //
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FIELD_TRIAL_DEFAULT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FIELD_TRIAL_DEFAULT_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FIELD_TRIAL_DEFAULT_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FIELD_TRIAL_DEFAULT_H_
 
 namespace webrtc {
 namespace field_trial {
@@ -20,7 +20,9 @@ namespace field_trial {
 // Note: trials_string must never be destroyed.
 void InitFieldTrialsFromString(const char* trials_string);
 
+const char* GetFieldTrialString();
+
 }  // namespace field_trial
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FIELD_TRIAL_DEFAULT_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FIELD_TRIAL_DEFAULT_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/file_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/include/file_wrapper.h
similarity index 93%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/file_wrapper.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/file_wrapper.h
index 8f4e09f9c9..b32a62f2f9 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/file_wrapper.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/file_wrapper.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FILE_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FILE_WRAPPER_H_
 
 #include <stddef.h>
 #include <stdio.h>
@@ -75,4 +75,4 @@ class FileWrapper : public InStream, public OutStream {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_FILE_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h b/media/webrtc/trunk/webrtc/system_wrappers/include/fix_interlocked_exchange_pointer_win.h
similarity index 100%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/fix_interlocked_exchange_pointer_win.h
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/logcat_trace_context.h b/media/webrtc/trunk/webrtc/system_wrappers/include/logcat_trace_context.h
similarity index 78%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/logcat_trace_context.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/logcat_trace_context.h
index d23e451f2e..8bb01d8102 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/logcat_trace_context.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/logcat_trace_context.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LOGCAT_TRACE_CONTEXT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LOGCAT_TRACE_CONTEXT_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_LOGCAT_TRACE_CONTEXT_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_LOGCAT_TRACE_CONTEXT_H_
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 #ifndef ANDROID
 #error This file only makes sense to include on Android!
@@ -32,4 +32,4 @@ class LogcatTraceContext : public webrtc::TraceCallback {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LOGCAT_TRACE_CONTEXT_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_LOGCAT_TRACE_CONTEXT_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/logging.h b/media/webrtc/trunk/webrtc/system_wrappers/include/logging.h
similarity index 79%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/logging.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/logging.h
index 41c436b1f3..0089841d4e 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/logging.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/logging.h
@@ -36,20 +36,8 @@
 //     type (basically, it just doesn't prepend the namespace).
 // LOG_F(sev) Like LOG(), but includes the name of the current function.
 
-// Additional helper macros added by WebRTC:
-// LOG_API is a shortcut for API call logging. Pass in the input parameters of
-// the method. For example:
-//   Foo(int bar, int baz) {
-//     LOG_API2(bar, baz);
-//   }
-//
-// LOG_FERR is a shortcut for logging a failed function call. For example:
-//   if (!Foo(bar)) {
-//     LOG_FERR1(LS_WARNING, Foo, bar);
-//   }
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LOGGING_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LOGGING_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_LOGGING_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_LOGGING_H_
 
 #include <sstream>
 
@@ -131,31 +119,14 @@ class LogMessageVoidify {
     webrtc::LogMessage(__FILE__, __LINE__, sev).stream()
 
 // The _F version prefixes the message with the current function name.
-#if (defined(__GNUC__) && defined(_DEBUG)) || defined(WANT_PRETTY_LOG_F)
+#if (defined(__GNUC__) && !defined(NDEBUG)) || defined(WANT_PRETTY_LOG_F)
 #define LOG_F(sev) LOG(sev) << __PRETTY_FUNCTION__ << ": "
 #else
 #define LOG_F(sev) LOG(sev) << __FUNCTION__ << ": "
 #endif
 
-#define LOG_API0() LOG_F(LS_VERBOSE)
-#define LOG_API1(v1) LOG_API0() << #v1 << "=" << v1
-#define LOG_API2(v1, v2) LOG_API1(v1) \
-    << ", " << #v2 << "=" << v2
-#define LOG_API3(v1, v2, v3) LOG_API2(v1, v2) \
-    << ", " << #v3 << "=" << v3
-
-#define LOG_FERR0(sev, func) LOG(sev) << #func << " failed"
-#define LOG_FERR1(sev, func, v1) LOG_FERR0(sev, func) \
-    << ": " << #v1 << "=" << v1
-#define LOG_FERR2(sev, func, v1, v2) LOG_FERR1(sev, func, v1) \
-    << ", " << #v2 << "=" << v2
-#define LOG_FERR3(sev, func, v1, v2, v3) LOG_FERR2(sev, func, v1, v2) \
-    << ", " << #v3 << "=" << v3
-#define LOG_FERR4(sev, func, v1, v2, v3, v4) LOG_FERR3(sev, func, v1, v2, v3) \
-    << ", " << #v4 << "=" << v4
-
 #endif  // LOG
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LOGGING_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_LOGGING_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/include/metrics.h b/media/webrtc/trunk/webrtc/system_wrappers/include/metrics.h
new file mode 100644
index 0000000000..4cd74c5e84
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/metrics.h
@@ -0,0 +1,185 @@
+//
+// Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+//
+// Use of this source code is governed by a BSD-style license
+// that can be found in the LICENSE file in the root of the source
+// tree. An additional intellectual property rights grant can be found
+// in the file PATENTS.  All contributing project authors may
+// be found in the AUTHORS file in the root of the source tree.
+//
+
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_METRICS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_METRICS_H_
+
+#include <string>
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
+
+// Macros for allowing WebRTC clients (e.g. Chrome) to gather and aggregate
+// statistics.
+//
+// Histogram for counters.
+// RTC_HISTOGRAM_COUNTS(name, sample, min, max, bucket_count);
+//
+// Histogram for enumerators.
+// The boundary should be above the max enumerator sample.
+// RTC_HISTOGRAM_ENUMERATION(name, sample, boundary);
+//
+//
+// The macros use the methods HistogramFactoryGetCounts,
+// HistogramFactoryGetEnumeration and HistogramAdd.
+//
+// Therefore, WebRTC clients must either:
+//
+// - provide implementations of
+//   Histogram* webrtc::metrics::HistogramFactoryGetCounts(
+//       const std::string& name, int sample, int min, int max,
+//       int bucket_count);
+//   Histogram* webrtc::metrics::HistogramFactoryGetEnumeration(
+//       const std::string& name, int sample, int boundary);
+//   void webrtc::metrics::HistogramAdd(
+//       Histogram* histogram_pointer, const std::string& name, int sample);
+//
+// - or link with the default implementations (i.e.
+//   system_wrappers/system_wrappers.gyp:metrics_default).
+//
+//
+// Example usage:
+//
+// RTC_HISTOGRAM_COUNTS("WebRTC.Video.NacksSent", nacks_sent, 1, 100000, 100);
+//
+// enum Types {
+//   kTypeX,
+//   kTypeY,
+//   kBoundary,
+// };
+//
+// RTC_HISTOGRAM_ENUMERATION("WebRTC.Types", kTypeX, kBoundary);
+
+
+// Macros for adding samples to a named histogram.
+
+// Histogram for counters (exponentially spaced buckets).
+#define RTC_HISTOGRAM_COUNTS_100(name, sample) \
+  RTC_HISTOGRAM_COUNTS(name, sample, 1, 100, 50)
+
+#define RTC_HISTOGRAM_COUNTS_200(name, sample) \
+  RTC_HISTOGRAM_COUNTS(name, sample, 1, 200, 50)
+
+#define RTC_HISTOGRAM_COUNTS_1000(name, sample) \
+  RTC_HISTOGRAM_COUNTS(name, sample, 1, 1000, 50)
+
+#define RTC_HISTOGRAM_COUNTS_10000(name, sample) \
+  RTC_HISTOGRAM_COUNTS(name, sample, 1, 10000, 50)
+
+#define RTC_HISTOGRAM_COUNTS_100000(name, sample) \
+  RTC_HISTOGRAM_COUNTS(name, sample, 1, 100000, 50)
+
+#define RTC_HISTOGRAM_COUNTS(name, sample, min, max, bucket_count) \
+  RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
+      webrtc::metrics::HistogramFactoryGetCounts(name, min, max, bucket_count))
+
+// Deprecated.
+// TODO(asapersson): Remove.
+#define RTC_HISTOGRAM_COUNTS_SPARSE_100(name, sample) \
+  RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 100, 50)
+
+#define RTC_HISTOGRAM_COUNTS_SPARSE_200(name, sample) \
+  RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 200, 50)
+
+#define RTC_HISTOGRAM_COUNTS_SPARSE_1000(name, sample) \
+  RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 1000, 50)
+
+#define RTC_HISTOGRAM_COUNTS_SPARSE_10000(name, sample) \
+  RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 10000, 50)
+
+#define RTC_HISTOGRAM_COUNTS_SPARSE_100000(name, sample) \
+  RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 100000, 50)
+
+#define RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, min, max, bucket_count) \
+  RTC_HISTOGRAM_COMMON_BLOCK_SLOW(name, sample, \
+      webrtc::metrics::HistogramFactoryGetCounts(name, min, max, bucket_count))
+
+// Histogram for percentage (evenly spaced buckets).
+#define RTC_HISTOGRAM_PERCENTAGE(name, sample) \
+  RTC_HISTOGRAM_ENUMERATION(name, sample, 101)
+
+// Deprecated.
+// TODO(asapersson): Remove.
+#define RTC_HISTOGRAM_PERCENTAGE_SPARSE(name, sample) \
+  RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, 101)
+
+// Histogram for enumerators (evenly spaced buckets).
+// |boundary| should be above the max enumerator sample.
+#define RTC_HISTOGRAM_ENUMERATION(name, sample, boundary) \
+  RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
+      webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
+
+// Deprecated.
+// TODO(asapersson): Remove.
+#define RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, boundary) \
+  RTC_HISTOGRAM_COMMON_BLOCK_SLOW(name, sample, \
+      webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
+
+// The name of the histogram should not vary.
+// TODO(asapersson): Consider changing string to const char*.
+#define RTC_HISTOGRAM_COMMON_BLOCK(constant_name, sample, \
+                                   factory_get_invocation) \
+  do { \
+    static webrtc::metrics::Histogram* atomic_histogram_pointer = nullptr; \
+    webrtc::metrics::Histogram* histogram_pointer = \
+        rtc::AtomicOps::AcquireLoadPtr(&atomic_histogram_pointer); \
+    if (!histogram_pointer) { \
+      histogram_pointer = factory_get_invocation; \
+      webrtc::metrics::Histogram* prev_pointer = \
+          rtc::AtomicOps::CompareAndSwapPtr( \
+              &atomic_histogram_pointer, \
+              static_cast<webrtc::metrics::Histogram*>(nullptr), \
+              histogram_pointer); \
+      RTC_DCHECK(prev_pointer == nullptr || \
+                 prev_pointer == histogram_pointer); \
+    } \
+    webrtc::metrics::HistogramAdd(histogram_pointer, constant_name, sample); \
+  } while (0)
+
+// Deprecated.
+// The histogram is constructed/found for each call.
+// May be used for histograms with infrequent updates.
+#define RTC_HISTOGRAM_COMMON_BLOCK_SLOW(name, sample, factory_get_invocation) \
+  do { \
+    webrtc::metrics::Histogram* histogram_pointer = factory_get_invocation; \
+    webrtc::metrics::HistogramAdd(histogram_pointer, name, sample); \
+  } while (0)
+
+namespace webrtc {
+namespace metrics {
+
+// Time that should have elapsed for stats that are gathered once per call.
+enum { kMinRunTimeInSeconds = 10 };
+
+class Histogram;
+
+// Functions for getting pointer to histogram (constructs or finds the named
+// histogram).
+
+// Get histogram for counters.
+Histogram* HistogramFactoryGetCounts(
+    const std::string& name, int min, int max, int bucket_count);
+
+// Get histogram for enumerators.
+// |boundary| should be above the max enumerator sample.
+Histogram* HistogramFactoryGetEnumeration(
+    const std::string& name, int boundary);
+
+// Function for adding a |sample| to a histogram.
+// |name| can be used to verify that it matches the histogram name.
+void HistogramAdd(
+    Histogram* histogram_pointer, const std::string& name, int sample);
+
+}  // namespace metrics
+}  // namespace webrtc
+
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_METRICS_H_
+
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/include/ntp_time.h b/media/webrtc/trunk/webrtc/system_wrappers/include/ntp_time.h
new file mode 100644
index 0000000000..229666e8dd
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/ntp_time.h
@@ -0,0 +1,63 @@
+/*
+*  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+*
+*  Use of this source code is governed by a BSD-style license
+*  that can be found in the LICENSE file in the root of the source
+*  tree. An additional intellectual property rights grant can be found
+*  in the file PATENTS.  All contributing project authors may
+*  be found in the AUTHORS file in the root of the source tree.
+*/
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_NTP_TIME_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_NTP_TIME_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class NtpTime {
+ public:
+  NtpTime() : seconds_(0), fractions_(0) {}
+  explicit NtpTime(const Clock& clock) {
+    clock.CurrentNtp(seconds_, fractions_);
+  }
+  NtpTime(uint32_t seconds, uint32_t fractions)
+      : seconds_(seconds), fractions_(fractions) {}
+
+  NtpTime(const NtpTime&) = default;
+  NtpTime& operator=(const NtpTime&) = default;
+
+  void SetCurrent(const Clock& clock) {
+    clock.CurrentNtp(seconds_, fractions_);
+  }
+  void Set(uint32_t seconds, uint32_t fractions) {
+    seconds_ = seconds;
+    fractions_ = fractions;
+  }
+  void Reset() {
+    seconds_ = 0;
+    fractions_ = 0;
+  }
+
+  int64_t ToMs() const { return Clock::NtpToMs(seconds_, fractions_); }
+
+  // NTP standard (RFC1305, section 3.1) explicitly state value 0/0 is invalid.
+  bool Valid() const { return !(seconds_ == 0 && fractions_ == 0); }
+
+  uint32_t seconds() const { return seconds_; }
+  uint32_t fractions() const { return fractions_; }
+
+ private:
+  uint32_t seconds_;
+  uint32_t fractions_;
+};
+
+inline bool operator==(const NtpTime& n1, const NtpTime& n2) {
+  return n1.seconds() == n2.seconds() && n1.fractions() == n2.fractions();
+}
+inline bool operator!=(const NtpTime& n1, const NtpTime& n2) {
+  return !(n1 == n2);
+}
+
+}  // namespace webrtc
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_NTP_TIME_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/ref_count.h b/media/webrtc/trunk/webrtc/system_wrappers/include/ref_count.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/ref_count.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/ref_count.h
index 68616662e9..3dd335a8da 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/ref_count.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/ref_count.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
-#define SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
+#ifndef SYSTEM_WRAPPERS_INCLUDE_REF_COUNT_H_
+#define SYSTEM_WRAPPERS_INCLUDE_REF_COUNT_H_
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
@@ -61,11 +61,11 @@ class RefCountImpl : public T {
   RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5)
       : T(p1, p2, p3, p4, p5), ref_count_(0) {}
 
-  virtual int32_t AddRef() {
+  int32_t AddRef() const override {
     return ++ref_count_;
   }
 
-  virtual int32_t Release() {
+  int32_t Release() const override {
     int32_t ref_count;
     ref_count = --ref_count_;
     if (ref_count == 0)
@@ -74,9 +74,9 @@ class RefCountImpl : public T {
   }
 
  protected:
-  Atomic32 ref_count_;
+  mutable Atomic32 ref_count_;
 };
 
 }  // namespace webrtc
 
-#endif  // SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
+#endif  // SYSTEM_WRAPPERS_INCLUDE_REF_COUNT_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/rtp_to_ntp.h b/media/webrtc/trunk/webrtc/system_wrappers/include/rtp_to_ntp.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/rtp_to_ntp.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/rtp_to_ntp.h
index dfc25cd9e9..0c91928626 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/rtp_to_ntp.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/rtp_to_ntp.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef SYSTEM_WRAPPERS_INTERFACE_RTP_TO_NTP_H_
-#define SYSTEM_WRAPPERS_INTERFACE_RTP_TO_NTP_H_
+#ifndef SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_H_
+#define SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_H_
 
 #include <list>
 
@@ -47,4 +47,4 @@ int CheckForWrapArounds(uint32_t rtp_timestamp, uint32_t rtcp_rtp_timestamp);
 
 }  // namespace webrtc
 
-#endif  // SYSTEM_WRAPPERS_INTERFACE_RTP_TO_NTP_H_
+#endif  // SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/rw_lock_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/include/rw_lock_wrapper.h
similarity index 90%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/rw_lock_wrapper.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/rw_lock_wrapper.h
index dbe6d6c7c0..751b6a1df5 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/rw_lock_wrapper.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/rw_lock_wrapper.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_RW_LOCK_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_RW_LOCK_WRAPPER_H_
 
 #include "webrtc/base/thread_annotations.h"
 
@@ -65,4 +65,4 @@ class SCOPED_LOCKABLE WriteLockScoped {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_RW_LOCK_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_vector.h b/media/webrtc/trunk/webrtc/system_wrappers/include/scoped_vector.h
similarity index 80%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_vector.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/scoped_vector.h
index a3409b635e..15c3380c8c 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_vector.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/scoped_vector.h
@@ -10,14 +10,14 @@
 
 // Borrowed from Chromium's src/base/memory/scoped_vector.h.
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_VECTOR_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_VECTOR_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SCOPED_VECTOR_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SCOPED_VECTOR_H_
 
 #include <vector>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/base/move.h"
-#include "webrtc/system_wrappers/interface/stl_util.h"
+#include "webrtc/base/deprecation.h"
+#include "webrtc/system_wrappers/include/stl_util.h"
 
 namespace webrtc {
 
@@ -25,8 +25,6 @@ namespace webrtc {
 // destructor.
 template <class T>
 class ScopedVector {
-  RTC_MOVE_ONLY_TYPE_FOR_CPP_03(ScopedVector, RValue)
-
  public:
   typedef typename std::vector<T*>::allocator_type allocator_type;
   typedef typename std::vector<T*>::size_type size_type;
@@ -44,13 +42,27 @@ class ScopedVector {
 
   ScopedVector() {}
   ~ScopedVector() { clear(); }
-  ScopedVector(RValue other) { swap(*other.object); }
 
-  ScopedVector& operator=(RValue rhs) {
-    swap(*rhs.object);
+  // Move construction and assignment.
+  ScopedVector(ScopedVector&& other) { *this = std::move(other); }
+  ScopedVector& operator=(ScopedVector&& other) {
+    std::swap(v_, other.v_);  // The arguments are std::vectors, so std::swap
+                              // is the one that we want.
+    other.clear();
     return *this;
   }
 
+  // Deleted copy constructor and copy assignment, to make the type move-only.
+  ScopedVector(const ScopedVector& other) = delete;
+  ScopedVector& operator=(const ScopedVector& other) = delete;
+
+  // Get an rvalue reference. (sv.Pass() does the same thing as std::move(sv).)
+  // Deprecated; remove in March 2016 (bug 5373).
+  RTC_DEPRECATED ScopedVector&& Pass() { return DEPRECATED_Pass(); }
+  ScopedVector&& DEPRECATED_Pass() {
+    return std::move(*this);
+  }
+
   reference operator[](size_t index) { return v_[index]; }
   const_reference operator[](size_t index) const { return v_[index]; }
 
@@ -75,7 +87,7 @@ class ScopedVector {
   void push_back(T* elem) { v_.push_back(elem); }
 
   void pop_back() {
-    DCHECK(!empty());
+    RTC_DCHECK(!empty());
     delete v_.back();
     v_.pop_back();
   }
@@ -145,4 +157,4 @@ class ScopedVector {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_VECTOR_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SCOPED_VECTOR_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/sleep.h b/media/webrtc/trunk/webrtc/system_wrappers/include/sleep.h
similarity index 82%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/sleep.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/sleep.h
index c0205bf085..e7ed8b32b8 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/sleep.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/sleep.h
@@ -9,8 +9,8 @@
  */
 // An OS-independent sleep function.
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SLEEP_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SLEEP_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_
 
 namespace webrtc {
 
@@ -21,4 +21,4 @@ void SleepMs(int msecs);
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SLEEP_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/sort.h b/media/webrtc/trunk/webrtc/system_wrappers/include/sort.h
similarity index 93%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/sort.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/sort.h
index da6ff8d52e..5bf2afa8a5 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/sort.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/sort.h
@@ -10,8 +10,8 @@
 
 // Generic unstable sorting routines.
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SORT_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SORT_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
@@ -62,4 +62,4 @@ int32_t KeySort(void* data, void* key, uint32_t num_of_elements,
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_SORT_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/static_instance.h b/media/webrtc/trunk/webrtc/system_wrappers/include/static_instance.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/static_instance.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/static_instance.h
index 071edabfa0..d6df05c92c 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/static_instance.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/static_instance.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATIC_INSTANCE_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATIC_INSTANCE_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STATIC_INSTANCE_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STATIC_INSTANCE_H_
 
 #include <assert.h>
 
@@ -38,6 +38,7 @@ static T* GetStaticInstance(CountOperation count_operation) {
   // Simple solution since we don't use this for large objects anymore
   return Singleton<T>::get();
 }
+
 }  // namspace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATIC_INSTANCE_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STATIC_INSTANCE_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/stl_util.h b/media/webrtc/trunk/webrtc/system_wrappers/include/stl_util.h
similarity index 98%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/stl_util.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/stl_util.h
index ebe855fb10..b7a702113f 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/stl_util.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/stl_util.h
@@ -10,8 +10,8 @@
 
 // Borrowed from Chromium's src/base/stl_util.h.
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STL_UTIL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STL_UTIL_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STL_UTIL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STL_UTIL_H_
 
 #include <assert.h>
 #include <algorithm>
@@ -262,4 +262,4 @@ bool STLIncludes(const Arg1& a1, const Arg2& a2) {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STL_UTIL_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STL_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/stringize_macros.h b/media/webrtc/trunk/webrtc/system_wrappers/include/stringize_macros.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/stringize_macros.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/stringize_macros.h
index ab8c43d4e2..9c8e7e9120 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/stringize_macros.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/stringize_macros.h
@@ -15,8 +15,8 @@
 // symbols (or their output) and manipulating preprocessor symbols
 // that define strings.
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STRINGIZE_MACROS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STRINGIZE_MACROS_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
 
 // This is not very useful as it does not expand defined symbols if
 // called directly. Use its counterpart without the _NO_EXPANSION
@@ -35,4 +35,4 @@
 //   STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
 #define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STRINGIZE_MACROS_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/tick_util.h b/media/webrtc/trunk/webrtc/system_wrappers/include/tick_util.h
similarity index 61%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/tick_util.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/tick_util.h
index 5b32055006..ad46866342 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/tick_util.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/tick_util.h
@@ -11,14 +11,14 @@
 // System independant wrapper for polling elapsed time in ms and us.
 // The implementation works in the tick domain which can be mapped over to the
 // time domain.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TICK_UTIL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TICK_UTIL_H_
 
 #if _WIN32
 // Note: The Windows header must always be included before mmsystem.h
 #include <windows.h>
 #include <mmsystem.h>
-#elif WEBRTC_LINUX
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
 #include <time.h>
 #elif WEBRTC_MAC
 #include <mach/mach_time.h>
@@ -56,6 +56,8 @@ class TickTime {
 
   static int64_t TicksToMilliseconds(const int64_t ticks);
 
+  static int64_t TicksToMicroseconds(const int64_t ticks);
+
   // Returns a TickTime that is ticks later than the passed TickTime.
   friend TickTime operator+(const TickTime lhs, const int64_t ticks);
   TickTime& operator+=(const int64_t& ticks);
@@ -63,19 +65,9 @@ class TickTime {
   // Returns a TickInterval that is the difference in ticks beween rhs and lhs.
   friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
 
-  // Call to engage the fake clock. This is useful for tests since relying on
-  // a real clock often makes the test flaky.
-  static void UseFakeClock(int64_t start_millisecond);
-
-  // Advance the fake clock. Must be called after UseFakeClock.
-  static void AdvanceFakeClock(int64_t milliseconds);
-
  private:
   static int64_t QueryOsForTicks();
 
-  static bool use_fake_clock_;
-  static int64_t fake_ticks_;
-
   int64_t ticks_;
 };
 
@@ -83,6 +75,7 @@ class TickTime {
 class TickInterval {
  public:
   TickInterval();
+  explicit TickInterval(int64_t interval);
 
   int64_t Milliseconds() const;
   int64_t Microseconds() const;
@@ -103,8 +96,6 @@ class TickInterval {
   friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
 
  private:
-  explicit TickInterval(int64_t interval);
-
   friend class TickTime;
   friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
 
@@ -112,6 +103,14 @@ class TickInterval {
   int64_t interval_;
 };
 
+inline int64_t TickInterval::Milliseconds() const {
+  return TickTime::TicksToMilliseconds(interval_);
+}
+
+inline int64_t TickInterval::Microseconds() const {
+  return TickTime::TicksToMicroseconds(interval_);
+}
+
 inline TickInterval operator+(const TickInterval& lhs,
                               const TickInterval& rhs) {
   return TickInterval(lhs.interval_ + rhs.interval_);
@@ -157,82 +156,13 @@ inline TickTime::TickTime(int64_t ticks)
 }
 
 inline TickTime TickTime::Now() {
-  if (use_fake_clock_)
-    return TickTime(fake_ticks_);
-  else
-    return TickTime(QueryOsForTicks());
-}
-
-inline int64_t TickTime::MillisecondTimestamp() {
-  int64_t ticks = TickTime::Now().Ticks();
-#if _WIN32
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  LARGE_INTEGER qpfreq;
-  QueryPerformanceFrequency(&qpfreq);
-  return (ticks * 1000) / qpfreq.QuadPart;
-#else
-  return ticks;
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-  return ticks / 1000000LL;
-#else
-  return ticks / 1000LL;
-#endif
-}
-
-inline int64_t TickTime::MicrosecondTimestamp() {
-  int64_t ticks = TickTime::Now().Ticks();
-#if _WIN32
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  LARGE_INTEGER qpfreq;
-  QueryPerformanceFrequency(&qpfreq);
-  return (ticks * 1000) / (qpfreq.QuadPart / 1000);
-#else
-  return ticks * 1000LL;
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-  return ticks / 1000LL;
-#else
-  return ticks;
-#endif
+  return TickTime(QueryOsForTicks());
 }
 
 inline int64_t TickTime::Ticks() const {
   return ticks_;
 }
 
-inline int64_t TickTime::MillisecondsToTicks(const int64_t ms) {
-#if _WIN32
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  LARGE_INTEGER qpfreq;
-  QueryPerformanceFrequency(&qpfreq);
-  return (qpfreq.QuadPart * ms) / 1000;
-#else
-  return ms;
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-  return ms * 1000000LL;
-#else
-  return ms * 1000LL;
-#endif
-}
-
-inline int64_t TickTime::TicksToMilliseconds(const int64_t ticks) {
-#if _WIN32
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  LARGE_INTEGER qpfreq;
-  QueryPerformanceFrequency(&qpfreq);
-  return (ticks * 1000) / qpfreq.QuadPart;
-#else
-  return ticks;
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-  return ticks / 1000000LL;
-#else
-  return ticks / 1000LL;
-#endif
-}
-
 inline TickTime& TickTime::operator+=(const int64_t& ticks) {
   ticks_ += ticks;
   return *this;
@@ -245,44 +175,6 @@ inline TickInterval::TickInterval(const int64_t interval)
   : interval_(interval) {
 }
 
-inline int64_t TickInterval::Milliseconds() const {
-#if _WIN32
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  LARGE_INTEGER qpfreq;
-  QueryPerformanceFrequency(&qpfreq);
-  return (interval_ * 1000) / qpfreq.QuadPart;
-#else
-  // interval_ is in ms
-  return interval_;
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-  // interval_ is in ns
-  return interval_ / 1000000;
-#else
-  // interval_ is usecs
-  return interval_ / 1000;
-#endif
-}
-
-inline int64_t TickInterval::Microseconds() const {
-#if _WIN32
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  LARGE_INTEGER qpfreq;
-  QueryPerformanceFrequency(&qpfreq);
-  return (interval_ * 1000000) / qpfreq.QuadPart;
-#else
-  // interval_ is in ms
-  return interval_ * 1000LL;
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-  // interval_ is in ns
-  return interval_ / 1000;
-#else
-  // interval_ is usecs
-  return interval_;
-#endif
-}
-
 inline TickInterval& TickInterval::operator+=(const TickInterval& rhs) {
   interval_ += rhs.interval_;
   return *this;
@@ -295,4 +187,4 @@ inline TickInterval& TickInterval::operator-=(const TickInterval& rhs) {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TICK_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/timestamp_extrapolator.h b/media/webrtc/trunk/webrtc/system_wrappers/include/timestamp_extrapolator.h
similarity index 86%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/timestamp_extrapolator.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/timestamp_extrapolator.h
index b78cf64be1..d9c5c6fb37 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/timestamp_extrapolator.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/timestamp_extrapolator.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef SYSTEM_WRAPPERS_INTERFACE_TIMESTAMP_EXTRAPOLATOR_H_
-#define SYSTEM_WRAPPERS_INTERFACE_TIMESTAMP_EXTRAPOLATOR_H_
+#ifndef SYSTEM_WRAPPERS_INCLUDE_TIMESTAMP_EXTRAPOLATOR_H_
+#define SYSTEM_WRAPPERS_INCLUDE_TIMESTAMP_EXTRAPOLATOR_H_
 
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc
@@ -53,4 +53,4 @@ private:
 
 }  // namespace webrtc
 
-#endif // SYSTEM_WRAPPERS_INTERFACE_TIMESTAMP_EXTRAPOLATOR_H_
+#endif // SYSTEM_WRAPPERS_INCLUDE_TIMESTAMP_EXTRAPOLATOR_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/trace.h b/media/webrtc/trunk/webrtc/system_wrappers/include/trace.h
similarity index 96%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/trace.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/trace.h
index 8531458541..68e401fc48 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/trace.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/trace.h
@@ -13,8 +13,8 @@
  *  messages. Apply filtering to avoid that.
  */
 
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TRACE_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TRACE_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
@@ -110,4 +110,4 @@ extern "C" {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TRACE_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/utf_util_win.h b/media/webrtc/trunk/webrtc/system_wrappers/include/utf_util_win.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/system_wrappers/interface/utf_util_win.h
rename to media/webrtc/trunk/webrtc/system_wrappers/include/utf_util_win.h
index cc48fd254d..0e3f2d01c6 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/utf_util_win.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/include/utf_util_win.h
@@ -10,8 +10,8 @@
 
 // Conversion functions for UTF-8 and UTF-16 strings on Windows.
 // Duplicated from talk/base/win32.h.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_UTF_UTIL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_UTF_UTIL_H_
+#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_UTF_UTIL_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_UTF_UTIL_H_
 
 #ifdef WIN32
 #include <windows.h>
@@ -54,4 +54,4 @@ inline std::string ToUtf8(const std::wstring& wstr) {
 }  // namespace webrtc
 
 #endif  // WIN32
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_UTF_UTIL_H_
+#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_UTF_UTIL_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/event_tracer.h b/media/webrtc/trunk/webrtc/system_wrappers/interface/event_tracer.h
deleted file mode 100644
index 9b1eb1eb92..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/event_tracer.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file defines the interface for event tracing in WebRTC.
-//
-// Event log handlers are set through SetupEventTracer(). User of this API will
-// provide two function pointers to handle event tracing calls.
-//
-// * GetCategoryEnabledPtr
-//   Event tracing system calls this function to determine if a particular
-//   event category is enabled.
-//
-// * AddTraceEventPtr
-//   Adds a tracing event. It is the user's responsibility to log the data
-//   provided.
-//
-// Parameters for the above two functions are described in trace_event.h.
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_TRACER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_TRACER_H_
-
-// This file has moved.
-// TODO(tommi): Delete after removing dependencies and updating Chromium.
-#include "webrtc/base/event_tracer.h"
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_TRACER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/metrics.h b/media/webrtc/trunk/webrtc/system_wrappers/interface/metrics.h
deleted file mode 100644
index cb641c0674..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/metrics.h
+++ /dev/null
@@ -1,136 +0,0 @@
-//
-// Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
-//
-// Use of this source code is governed by a BSD-style license
-// that can be found in the LICENSE file in the root of the source
-// tree. An additional intellectual property rights grant can be found
-// in the file PATENTS.  All contributing project authors may
-// be found in the AUTHORS file in the root of the source tree.
-//
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_METRICS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_METRICS_H_
-
-#include <string>
-
-#include "webrtc/common_types.h"
-
-// Macros for allowing WebRTC clients (e.g. Chrome) to gather and aggregate
-// statistics.
-//
-// Histogram for counters.
-// RTC_HISTOGRAM_COUNTS(name, sample, min, max, bucket_count);
-//
-// Histogram for enumerators.
-// The boundary should be above the max enumerator sample.
-// RTC_HISTOGRAM_ENUMERATION(name, sample, boundary);
-//
-//
-// The macros use the methods HistogramFactoryGetCounts,
-// HistogramFactoryGetEnumeration and HistogramAdd.
-//
-// Therefore, WebRTC clients must either:
-//
-// - provide implementations of
-//   Histogram* webrtc::metrics::HistogramFactoryGetCounts(
-//       const std::string& name, int sample, int min, int max,
-//       int bucket_count);
-//   Histogram* webrtc::metrics::HistogramFactoryGetEnumeration(
-//       const std::string& name, int sample, int boundary);
-//   void webrtc::metrics::HistogramAdd(
-//       Histogram* histogram_pointer, const std::string& name, int sample);
-//
-// - or link with the default implementations (i.e.
-//   system_wrappers/system_wrappers.gyp:metrics_default).
-//
-//
-// Example usage:
-//
-// RTC_HISTOGRAM_COUNTS("WebRTC.Video.NacksSent", nacks_sent, 1, 100000, 100);
-//
-// enum Types {
-//   kTypeX,
-//   kTypeY,
-//   kBoundary,
-// };
-//
-// RTC_HISTOGRAM_ENUMERATION("WebRTC.Types", kTypeX, kBoundary);
-
-
-// Macros for adding samples to a named histogram.
-//
-// NOTE: this is a temporary solution.
-// The aim is to mimic the behaviour in Chromium's src/base/metrics/histograms.h
-// However as atomics are not supported in webrtc, this is for now a modified
-// and temporary solution. Note that the histogram is constructed/found for
-// each call. Therefore, for now only use this implementation for metrics
-// that do not need to be updated frequently.
-// TODO(asapersson): Change implementation when atomics are supported.
-// Also consider changing string to const char* when switching to atomics.
-
-// Histogram for counters.
-#define RTC_HISTOGRAM_COUNTS_100(name, sample) RTC_HISTOGRAM_COUNTS( \
-    name, sample, 1, 100, 50)
-
-#define RTC_HISTOGRAM_COUNTS_1000(name, sample) RTC_HISTOGRAM_COUNTS( \
-    name, sample, 1, 1000, 50)
-
-#define RTC_HISTOGRAM_COUNTS_10000(name, sample) RTC_HISTOGRAM_COUNTS( \
-    name, sample, 1, 10000, 50)
-
-#define RTC_HISTOGRAM_COUNTS_100000(name, sample) RTC_HISTOGRAM_COUNTS( \
-    name, sample, 1, 100000, 50)
-
-#define RTC_HISTOGRAM_COUNTS(name, sample, min, max, bucket_count) \
-    RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
-        webrtc::metrics::HistogramFactoryGetCounts( \
-            name, min, max, bucket_count))
-
-// Histogram for percentage.
-#define RTC_HISTOGRAM_PERCENTAGE(name, sample) \
-    RTC_HISTOGRAM_ENUMERATION(name, sample, 101)
-
-// Histogram for enumerators.
-// |boundary| should be above the max enumerator sample.
-#define RTC_HISTOGRAM_ENUMERATION(name, sample, boundary) \
-    RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
-        webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
-
-#define RTC_HISTOGRAM_COMMON_BLOCK(constant_name, sample, \
-                                   factory_get_invocation) \
-  do { \
-    webrtc::metrics::Histogram* histogram_pointer = factory_get_invocation; \
-    webrtc::metrics::HistogramAdd(histogram_pointer, constant_name, sample); \
-  } while (0)
-
-
-namespace webrtc {
-namespace metrics {
-
-// Time that should have elapsed for stats that are gathered once per call.
-enum { kMinRunTimeInSeconds = 10 };
-
-class Histogram;
-
-// Functions for getting pointer to histogram (constructs or finds the named
-// histogram).
-
-// Get histogram for counters.
-Histogram* HistogramFactoryGetCounts(
-    const std::string& name, int min, int max, int bucket_count);
-
-// Get histogram for enumerators.
-// |boundary| should be above the max enumerator sample.
-Histogram* HistogramFactoryGetEnumeration(
-    const std::string& name, int boundary);
-
-// Function for adding a |sample| to a histogram.
-// |name| can be used to verify that it matches the histogram name.
-void HistogramAdd(
-    Histogram* histogram_pointer, const std::string& name, int sample);
-
-}  // namespace metrics
-}  // namespace webrtc
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_METRICS_H_
-
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_refptr.h b/media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_refptr.h
deleted file mode 100644
index b344d211b1..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/scoped_refptr.h
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
-#define SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
-
-#include <stddef.h>
-
-namespace webrtc {
-
-// Extracted from Chromium's src/base/memory/ref_counted.h.
-
-//
-// A smart pointer class for reference counted objects.  Use this class instead
-// of calling AddRef and Release manually on a reference counted object to
-// avoid common memory leaks caused by forgetting to Release an object
-// reference.  Sample usage:
-//
-//   class MyFoo : public RefCounted<MyFoo> {
-//    ...
-//   };
-//
-//   void some_function() {
-//     scoped_refptr<MyFoo> foo = new MyFoo();
-//     foo->Method(param);
-//     // |foo| is released when this function returns
-//   }
-//
-//   void some_other_function() {
-//     scoped_refptr<MyFoo> foo = new MyFoo();
-//     ...
-//     foo = NULL;  // explicitly releases |foo|
-//     ...
-//     if (foo)
-//       foo->Method(param);
-//   }
-//
-// The above examples show how scoped_refptr<T> acts like a pointer to T.
-// Given two scoped_refptr<T> classes, it is also possible to exchange
-// references between the two objects, like so:
-//
-//   {
-//     scoped_refptr<MyFoo> a = new MyFoo();
-//     scoped_refptr<MyFoo> b;
-//
-//     b.swap(a);
-//     // now, |b| references the MyFoo object, and |a| references NULL.
-//   }
-//
-// To make both |a| and |b| in the above example reference the same MyFoo
-// object, simply use the assignment operator:
-//
-//   {
-//     scoped_refptr<MyFoo> a = new MyFoo();
-//     scoped_refptr<MyFoo> b;
-//
-//     b = a;
-//     // now, |a| and |b| each own a reference to the same MyFoo object.
-//   }
-//
-template <class T>
-class scoped_refptr {
- public:
-  scoped_refptr() : ptr_(NULL) {
-  }
-
-  scoped_refptr(T* p) : ptr_(p) {
-    if (ptr_)
-      ptr_->AddRef();
-  }
-
-  scoped_refptr(const scoped_refptr<T>& r) : ptr_(r.ptr_) {
-    if (ptr_)
-      ptr_->AddRef();
-  }
-
-  template <typename U>
-  scoped_refptr(const scoped_refptr<U>& r) : ptr_(r.get()) {
-    if (ptr_)
-      ptr_->AddRef();
-  }
-
-  ~scoped_refptr() {
-    if (ptr_)
-      ptr_->Release();
-  }
-
-  T* get() const { return ptr_; }
-  operator T*() const { return ptr_; }
-  T* operator->() const { return ptr_; }
-
-  // Release a pointer.
-  // The return value is the current pointer held by this object.
-  // If this object holds a NULL pointer, the return value is NULL.
-  // After this operation, this object will hold a NULL pointer,
-  // and will not own the object any more.
-  T* release() {
-    T* retVal = ptr_;
-    ptr_ = NULL;
-    return retVal;
-  }
-
-  scoped_refptr<T>& operator=(T* p) {
-    // AddRef first so that self assignment should work
-    if (p)
-      p->AddRef();
-    if (ptr_ )
-      ptr_->Release();
-    ptr_ = p;
-    return *this;
-  }
-
-  scoped_refptr<T>& operator=(const scoped_refptr<T>& r) {
-    return *this = r.ptr_;
-  }
-
-  template <typename U>
-  scoped_refptr<T>& operator=(const scoped_refptr<U>& r) {
-    return *this = r.get();
-  }
-
-  void swap(T** pp) {
-    T* p = ptr_;
-    ptr_ = *pp;
-    *pp = p;
-  }
-
-  void swap(scoped_refptr<T>& r) {
-    swap(&r.ptr_);
-  }
-
- protected:
-  T* ptr_;
-};
-}  // namespace webrtc
-
-#endif  // SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/thread_wrapper.h b/media/webrtc/trunk/webrtc/system_wrappers/interface/thread_wrapper.h
deleted file mode 100644
index d18b6b21ff..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/thread_wrapper.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for spawning threads
-// Note: the spawned thread will loop over the callback function until stopped.
-// Note: The callback function is expected to return every 2 seconds or more
-// often.
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
-
-#if defined(WEBRTC_WIN)
-#include <windows.h>
-#endif
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Callback function that the spawned thread will enter once spawned.
-// A return value of false is interpreted as that the function has no
-// more work to do and that the thread can be released.
-typedef bool(*ThreadRunFunction)(void*);
-
-enum ThreadPriority {
-#ifdef WEBRTC_WIN
-  kLowPriority = THREAD_PRIORITY_BELOW_NORMAL,
-  kNormalPriority = THREAD_PRIORITY_NORMAL,
-  kHighPriority = THREAD_PRIORITY_ABOVE_NORMAL,
-  kHighestPriority = THREAD_PRIORITY_HIGHEST,
-  kRealtimePriority = THREAD_PRIORITY_TIME_CRITICAL
-#else
-  kLowPriority = 1,
-  kNormalPriority = 2,
-  kHighPriority = 3,
-  kHighestPriority = 4,
-  kRealtimePriority = 5
-#endif
-};
-
-// Represents a simple worker thread.  The implementation must be assumed
-// to be single threaded, meaning that all methods of the class, must be
-// called from the same thread, including instantiation.
-// TODO(tommi): There's no need for this to be a virtual interface since there's
-// only ever a single implementation of it.
-class ThreadWrapper {
- public:
-  virtual ~ThreadWrapper() {}
-
-  // Factory method. Constructor disabled.
-  //
-  // func        Pointer to a, by user, specified callback function.
-  // obj         Object associated with the thread. Passed in the callback
-  //             function.
-  // prio        Thread priority. May require root/admin rights.
-  // thread_name  NULL terminated thread name, will be visable in the Windows
-  //             debugger.
-  static rtc::scoped_ptr<ThreadWrapper> CreateThread(ThreadRunFunction func,
-      void* obj, const char* thread_name);
-
-  static rtc::scoped_ptr<ThreadWrapper> CreateUIThread(ThreadRunFunction func,
-      void* obj,  const char* thread_name);
-
-  // Get the current thread's thread ID.
-  // NOTE: This is a static method. It returns the id of the calling thread,
-  // *not* the id of the worker thread that a ThreadWrapper instance represents.
-  // TODO(tommi): Move outside of the ThreadWrapper class to avoid confusion.
-  static uint32_t GetThreadId();
-
-  // Tries to spawns a thread and returns true if that was successful.
-  // Additionally, it tries to set thread priority according to the priority
-  // from when CreateThread was called. However, failure to set priority will
-  // not result in a false return value.
-  virtual bool Start() = 0;
-
-  // Stops the spawned thread and waits for it to be reclaimed with a timeout
-  // of two seconds. Will return false if the thread was not reclaimed.
-  // Multiple tries to Stop are allowed (e.g. to wait longer than 2 seconds).
-  // It's ok to call Stop() even if the spawned thread has been reclaimed.
-  virtual bool Stop() = 0;
-
-  // Request a timed callback for ThreadRunFunction. Currently only
-  // implemented for a specific type of thread on Windows.
-  virtual bool RequestCallbackTimer(unsigned int milliseconds);
-
-  // Set the priority of the worker thread.  Must be called when thread
-  // is running.
-  virtual bool SetPriority(ThreadPriority priority) = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/interface/trace_event.h b/media/webrtc/trunk/webrtc/system_wrappers/interface/trace_event.h
deleted file mode 100644
index e82afbabf5..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/interface/trace_event.h
+++ /dev/null
@@ -1,13 +0,0 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file under third_party_mods/chromium or at:
-// http://src.chromium.org/svn/trunk/src/LICENSE
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_EVENT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_EVENT_H_
-
-// This file has moved.
-// TODO(tommi): Delete after removing dependencies and updating Chromium.
-#include "webrtc/base/trace_event.h"
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_EVENT_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_array_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_array_unittest.cc
index e5e556dff5..01238f8342 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_array_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_array_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/aligned_array.h"
+#include "webrtc/system_wrappers/include/aligned_array.h"
 
 #include <stdint.h>
 
@@ -16,7 +16,7 @@
 
 namespace {
 
-bool IsAligned(const void* ptr, int alignment) {
+bool IsAligned(const void* ptr, size_t alignment) {
   return reinterpret_cast<uintptr_t>(ptr) % alignment == 0;
 }
 
@@ -27,23 +27,23 @@ namespace webrtc {
 TEST(AlignedArrayTest, CheckAlignment) {
   AlignedArray<bool> arr(10, 7, 128);
   ASSERT_TRUE(IsAligned(arr.Array(), 128));
-  for (int i = 0; i < 10; ++i) {
+  for (size_t i = 0; i < 10; ++i) {
     ASSERT_TRUE(IsAligned(arr.Row(i), 128));
     ASSERT_EQ(arr.Row(i), arr.Array()[i]);
   }
 }
 
 TEST(AlignedArrayTest, CheckOverlap) {
-  AlignedArray<int> arr(10, 7, 128);
+  AlignedArray<size_t> arr(10, 7, 128);
 
-  for (int i = 0; i < 10; ++i) {
-    for (int j = 0; j < 7; ++j) {
+  for (size_t i = 0; i < 10; ++i) {
+    for (size_t j = 0; j < 7; ++j) {
       arr.At(i, j) = 20 * i + j;
     }
   }
 
-  for (int i = 0; i < 10; ++i) {
-    for (int j = 0; j < 7; ++j) {
+  for (size_t i = 0; i < 10; ++i) {
+    for (size_t j = 0; j < 7; ++j) {
       ASSERT_EQ(arr.At(i, j), 20 * i + j);
       ASSERT_EQ(arr.Row(i)[j], 20 * i + j);
       ASSERT_EQ(arr.Array()[i][j], 20 * i + j);
@@ -51,5 +51,10 @@ TEST(AlignedArrayTest, CheckOverlap) {
   }
 }
 
-}  // namespace webrtc
+TEST(AlignedArrayTest, CheckRowsCols) {
+  AlignedArray<bool> arr(10, 7, 128);
+  ASSERT_EQ(arr.rows(), 10u);
+  ASSERT_EQ(arr.cols(), 7u);
+}
 
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc.cc
index 258b6be92c..a654e97e75 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 #include <memory.h>
 #include <stdlib.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
index 57c083be2c..3933c2ac05 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 #if _WIN32
 #include <windows.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_mac.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_mac.cc
index d3728465d1..7c77d092b5 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_mac.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_mac.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 #include <assert.h>
 #include <libkern/OSAtomic.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_posix.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_posix.cc
index 3c2bd8508a..1477938bd8 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_posix.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_posix.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 #include <assert.h>
 #include <inttypes.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_win.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_win.cc
index f3c10f6b7c..cd4ce08580 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_win.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_win.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 #include <assert.h>
 #include <windows.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/clock.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/clock.cc
index bad4e0b1a0..d89dae4a4b 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/clock.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/clock.cc
@@ -8,20 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 #if defined(_WIN32)
 // Windows needs to be included before mmsystem.h
 #include "webrtc/base/win32.h"
 #include <MMSystem.h>
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_BSD) || (defined WEBRTC_MAC))
+#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC) || (defined WEBRTC_BSD))
 #include <sys/time.h>
 #include <time.h>
 #endif
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -155,9 +155,9 @@ class WindowsRealTimeClock : public RealTimeClock {
   }
 
   static ReferencePoint GetSystemReferencePoint() {
-    ReferencePoint ref = {0};
-    FILETIME ft0 = {0};
-    FILETIME ft1 = {0};
+    ReferencePoint ref = {};
+    FILETIME ft0 = {};
+    FILETIME ft1 = {};
     // Spin waiting for a change in system time. As soon as this change happens,
     // get the matching call for timeGetTime() as soon as possible. This is
     // assumed to be the most accurate offset that we can get between
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/clock_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/clock_unittest.cc
index 8672d3966e..9cb8ec73d0 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/clock_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/clock_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable.cc
index d57542cd2d..6719340cd4 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
 
 #if defined(_WIN32)
 #include <windows.h>
 #include "webrtc/system_wrappers/source/condition_variable_event_win.h"
 #include "webrtc/system_wrappers/source/condition_variable_native_win.h"
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_BSD)
 #include <pthread.h>
 #include "webrtc/system_wrappers/source/condition_variable_posix.h"
 #endif
@@ -31,7 +31,7 @@ ConditionVariableWrapper* ConditionVariableWrapper::CreateConditionVariable() {
     ret_val = new ConditionVariableEventWin();
   }
   return ret_val;
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_BSD)
   return ConditionVariablePosix::Create();
 #else
   return NULL;
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_event_win.h b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_event_win.h
index fce45d3daa..cdcef7dcb8 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_event_win.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_event_win.h
@@ -13,7 +13,7 @@
 
 #include <windows.h>
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.cc
index b8c19a4270..e44ca8f865 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/system_wrappers/source/condition_variable_native_win.h"
 #include "webrtc/system_wrappers/source/critical_section_win.h"
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.h b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.h
index 1fbce37387..c22787f2f2 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_native_win.h
@@ -13,7 +13,7 @@
 
 #include <windows.h>
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_posix.h b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_posix.h
index b29e116d7f..0aab1f03de 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_posix.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_posix.h
@@ -13,7 +13,7 @@
 
 #include <pthread.h>
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_unittest.cc
index c34c4ea0d8..5a8dd0b36e 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/condition_variable_unittest.cc
@@ -8,12 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -21,6 +23,7 @@ namespace {
 
 const int kLongWaitMs = 100 * 1000; // A long time in testing terms
 const int kShortWaitMs = 2 * 1000; // Long enough for process switches to happen
+const int kVeryShortWaitMs = 20; // Used when we want a timeout
 
 // A Baton is one possible control structure one can build using
 // conditional variables.
@@ -141,12 +144,10 @@ bool WaitingRunFunction(void* obj) {
 
 class CondVarTest : public ::testing::Test {
  public:
-  CondVarTest() {}
+  CondVarTest() : thread_(&WaitingRunFunction, &baton_, "CondVarTest") {}
 
   virtual void SetUp() {
-    thread_ = ThreadWrapper::CreateThread(&WaitingRunFunction,
-                                          &baton_, "CondVarTest");
-    ASSERT_TRUE(thread_->Start());
+    thread_.Start();
   }
 
   virtual void TearDown() {
@@ -157,14 +158,14 @@ class CondVarTest : public ::testing::Test {
     // and Pass).
     ASSERT_TRUE(baton_.Pass(kShortWaitMs));
     ASSERT_TRUE(baton_.Grab(kShortWaitMs));
-    ASSERT_TRUE(thread_->Stop());
+    thread_.Stop();
   }
 
  protected:
   Baton baton_;
 
  private:
-  rtc::scoped_ptr<ThreadWrapper> thread_;
+  rtc::PlatformThread thread_;
 };
 
 // The SetUp and TearDown functions use condition variables.
@@ -184,6 +185,19 @@ TEST_F(CondVarTest, DISABLED_PassBatonMultipleTimes) {
   EXPECT_EQ(2 * kNumberOfRounds, baton_.PassCount());
 }
 
+TEST(CondVarWaitTest, WaitingWaits) {
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect(
+      CriticalSectionWrapper::CreateCriticalSection());
+  rtc::scoped_ptr<ConditionVariableWrapper> cond_var(
+      ConditionVariableWrapper::CreateConditionVariable());
+  CriticalSectionScoped cs(crit_sect.get());
+  int64_t start_ms = TickTime::MillisecondTimestamp();
+  EXPECT_FALSE(cond_var->SleepCS(*(crit_sect), kVeryShortWaitMs));
+  int64_t end_ms = TickTime::MillisecondTimestamp();
+  EXPECT_LE(start_ms + kVeryShortWaitMs, end_ms)
+      << "actual elapsed:" << end_ms - start_ms;
+}
+
 }  // anonymous namespace
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_features.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_features.cc
index af29d08e72..51f24b6900 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_features.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_features.cc
@@ -10,7 +10,7 @@
 
 // Parts of this file derived from Chromium's base/cpu.cc.
 
-#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
 #if defined(WEBRTC_ARCH_X86_FAMILY) && defined(_MSC_VER)
 #include <intrin.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_info.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_info.cc
index 16bf20b56f..40231b65af 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_info.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_info.cc
@@ -8,68 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/cpu_info.h"
+#include "webrtc/system_wrappers/include/cpu_info.h"
 
-#if defined(_WIN32)
-#include <Windows.h>
-#elif defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-#include <sys/types.h>
-#include <sys/sysctl.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
-#include <unistd.h>
-#else // defined(_SC_NPROCESSORS_ONLN)
-#include <unistd.h>
-#endif
-
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/systeminfo.h"
 
 namespace webrtc {
 
-uint32_t CpuInfo::number_of_cores_ = 0;
-
 uint32_t CpuInfo::DetectNumberOfCores() {
-  if (!number_of_cores_) {
-#if defined(_WIN32)
-    SYSTEM_INFO si;
-    GetSystemInfo(&si);
-    number_of_cores_ = static_cast<uint32_t>(si.dwNumberOfProcessors);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                 "Available number of cores:%d", number_of_cores_);
-
-#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID) && !defined(WEBRTC_GONK)
-    number_of_cores_ = static_cast<uint32_t>(sysconf(_SC_NPROCESSORS_ONLN));
-    WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                 "Available number of cores:%d", number_of_cores_);
-
-#elif defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-    int name[] = {
-      CTL_HW,
-#ifdef HW_AVAILCPU
-      HW_AVAILCPU,
-#else
-      HW_NCPU,
-#endif
-    };
-    int ncpu;
-    size_t size = sizeof(ncpu);
-    if (0 == sysctl(name, 2, &ncpu, &size, NULL, 0)) {
-      number_of_cores_ = static_cast<uint32_t>(ncpu);
-      WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                   "Available number of cores:%d", number_of_cores_);
-    } else {
-      WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
-                   "Failed to get number of cores");
-      number_of_cores_ = 1;
-    }
-#elif defined(_SC_NPROCESSORS_ONLN)
-    number_of_cores_ = sysconf(_SC_NPROCESSORS_ONLN);
-#else
-    WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
-                 "No function to get number of cores");
-    number_of_cores_ = 1;
-#endif
-  }
-  return number_of_cores_;
+  return static_cast<uint32_t>(rtc::SystemInfo::GetMaxCpus());
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.cc
index 36b9f13735..41b77327a3 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.cc
@@ -10,8 +10,7 @@
 
 // General note: return values for the various pthread synchronization APIs
 // are explicitly ignored here. In Chromium, the same thing is done for release.
-// However, in debugging, failure in these APIs are logged. There is currently
-// no equivalent to DCHECK_EQ in WebRTC code so this is the best we can do here.
+// However, in debugging, failure in these APIs are logged.
 // TODO(henrike): add logging when pthread synchronization APIs are failing.
 
 #include "webrtc/system_wrappers/source/critical_section_posix.h"
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.h b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.h
index d71c93de0a..099f74c2df 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_posix.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #include <pthread.h>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_unittest.cc
index ec639eb673..9abf8b8017 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_unittest.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -78,10 +78,10 @@ TEST_F(CritSectTest, ThreadWakesOnce) NO_THREAD_SAFETY_ANALYSIS {
   CriticalSectionWrapper* crit_sect =
       CriticalSectionWrapper::CreateCriticalSection();
   ProtectedCount count(crit_sect);
-  rtc::scoped_ptr<ThreadWrapper> thread = ThreadWrapper::CreateThread(
+  rtc::PlatformThread thread(
       &LockUnlockThenStopRunFunction, &count, "ThreadWakesOnce");
   crit_sect->Enter();
-  ASSERT_TRUE(thread->Start());
+  thread.Start();
   SwitchProcess();
   // The critical section is of reentrant mode, so this should not release
   // the lock, even though count.Count() locks and unlocks the critical section
@@ -90,7 +90,7 @@ TEST_F(CritSectTest, ThreadWakesOnce) NO_THREAD_SAFETY_ANALYSIS {
   ASSERT_EQ(0, count.Count());
   crit_sect->Leave();  // This frees the thread to act.
   EXPECT_TRUE(WaitForCount(1, &count));
-  EXPECT_TRUE(thread->Stop());
+  thread.Stop();
   delete crit_sect;
 }
 
@@ -105,10 +105,10 @@ TEST_F(CritSectTest, ThreadWakesTwice) NO_THREAD_SAFETY_ANALYSIS {
   CriticalSectionWrapper* crit_sect =
       CriticalSectionWrapper::CreateCriticalSection();
   ProtectedCount count(crit_sect);
-  rtc::scoped_ptr<ThreadWrapper> thread = ThreadWrapper::CreateThread(
+  rtc::PlatformThread thread(
       &LockUnlockRunFunction, &count, "ThreadWakesTwice");
   crit_sect->Enter();  // Make sure counter stays 0 until we wait for it.
-  ASSERT_TRUE(thread->Start());
+  thread.Start();
   crit_sect->Leave();
 
   // The thread is capable of grabbing the lock multiple times,
@@ -128,7 +128,7 @@ TEST_F(CritSectTest, ThreadWakesTwice) NO_THREAD_SAFETY_ANALYSIS {
 
   SwitchProcess();
   EXPECT_TRUE(WaitForCount(count_before + 1, &count));
-  EXPECT_TRUE(thread->Stop());
+  thread.Stop();
   delete crit_sect;
 }
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_win.h b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_win.h
index be237accaf..8268bc3017 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_win.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/critical_section_win.h
@@ -12,7 +12,7 @@
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WIN_H_
 
 #include <windows.h>
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log.cc
index 653af657dd..778769603b 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log.cc
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 
 #include <assert.h>
 
 #include <algorithm>
 #include <list>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 namespace webrtc {
 
@@ -318,11 +318,12 @@ int DataLog::NextRow(const std::string& table_name) {
 }
 
 DataLogImpl::DataLogImpl()
-  : counter_(1),
-    tables_(),
-    flush_event_(EventWrapper::Create()),
-    tables_lock_(RWLockWrapper::CreateRWLock()) {
-}
+    : counter_(1),
+      tables_(),
+      flush_event_(EventWrapper::Create()),
+      file_writer_thread_(
+          new rtc::PlatformThread(DataLogImpl::Run, instance_, "DataLog")),
+      tables_lock_(RWLockWrapper::CreateRWLock()) {}
 
 DataLogImpl::~DataLogImpl() {
   StopThread();
@@ -348,12 +349,8 @@ int DataLogImpl::CreateLog() {
 }
 
 int DataLogImpl::Init() {
-  file_writer_thread_ = ThreadWrapper::CreateThread(
-      DataLogImpl::Run, instance_, "DataLog");
-  bool success = file_writer_thread_->Start();
-  if (!success)
-    return -1;
-  file_writer_thread_->SetPriority(kHighestPriority);
+  file_writer_thread_->Start();
+  file_writer_thread_->SetPriority(rtc::kHighestPriority);
   return 0;
 }
 
@@ -406,13 +403,8 @@ int DataLogImpl::NextRow(const std::string& table_name) {
   if (tables_.count(table_name) == 0)
     return -1;
   tables_[table_name]->NextRow();
-  if (!file_writer_thread_) {
-    // Write every row to file as they get complete.
-    tables_[table_name]->Flush();
-  } else {
-    // Signal a complete row
-    flush_event_->Set();
-  }
+  // Signal a complete row
+  flush_event_->Set();
   return 0;
 }
 
@@ -435,10 +427,8 @@ void DataLogImpl::Process() {
 }
 
 void DataLogImpl::StopThread() {
-  if (file_writer_thread_) {
-    flush_event_->Set();
-    file_writer_thread_->Stop();
-  }
+  flush_event_->Set();
+  file_writer_thread_->Stop();
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c.cc
index a11d545fc0..12a0d3f61a 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c.cc
@@ -10,11 +10,11 @@
 
 // This is the pure C wrapper of the DataLog class.
 
-#include "webrtc/system_wrappers/interface/data_log_c.h"
+#include "webrtc/system_wrappers/include/data_log_c.h"
 
 #include <string>
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 
 extern "C" int WebRtcDataLog_CreateLog() {
   return webrtc::DataLog::CreateLog();
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c
index 524bd24e1c..0b05e224eb 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_c_helpers_unittest.c
@@ -14,7 +14,7 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/system_wrappers/interface/data_log_c.h"
+#include "webrtc/system_wrappers/include/data_log_c.h"
 
 enum { kTestArrayLen = 4 };
 static const char kTableName[] = "c_wrapper_table";
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_helpers_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_helpers_unittest.cc
index 820d8cb4f3..25e1827ddb 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_helpers_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_helpers_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 
 #include <string>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_no_op.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_no_op.cc
index f5b0ea855f..bdd2b0a403 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_no_op.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_no_op.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 
 #include <string>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest.cc
index e36226992f..53f201ca5f 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 
 #include <map>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/data_log_c.h"
+#include "webrtc/system_wrappers/include/data_log_c.h"
 #include "webrtc/system_wrappers/source/data_log_c_helpers_unittest.h"
 
 using ::webrtc::DataLog;
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest_disabled.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest_disabled.cc
index 02d3cdb68c..1855a3e74f 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest_disabled.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/data_log_unittest_disabled.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/include/data_log.h"
 
 #include <stdio.h>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/event.cc
index 75571b55fb..05f918ffc2 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/event.cc
@@ -8,26 +8,47 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 
 #if defined(_WIN32)
 #include <windows.h>
-#include "webrtc/system_wrappers/source/event_win.h"
+#include "webrtc/system_wrappers/source/event_timer_win.h"
 #elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 #include <ApplicationServices/ApplicationServices.h>
 #include <pthread.h>
-#include "webrtc/system_wrappers/source/event_posix.h"
+#include "webrtc/system_wrappers/source/event_timer_posix.h"
 #else
 #include <pthread.h>
-#include "webrtc/system_wrappers/source/event_posix.h"
+#include "webrtc/system_wrappers/source/event_timer_posix.h"
 #endif
 
+#include "webrtc/base/event.h"
+
 namespace webrtc {
+
+class EventWrapperImpl : public EventWrapper {
+ public:
+  EventWrapperImpl() : event_(false, false) {}
+  ~EventWrapperImpl() override {}
+
+  bool Set() override {
+    event_.Set();
+    return true;
+  }
+
+  EventTypeWrapper Wait(unsigned long max_time) override {
+    int to_wait = max_time == WEBRTC_EVENT_INFINITE ?
+        rtc::Event::kForever : static_cast<int>(max_time);
+    return event_.Wait(to_wait) ? kEventSignaled : kEventTimeout;
+  }
+
+ private:
+  rtc::Event event_;
+};
+
+// static
 EventWrapper* EventWrapper::Create() {
-#if defined(_WIN32)
-  return new EventWindows();
-#else
-  return EventPosix::Create();
-#endif
+  return new EventWrapperImpl();
 }
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.cc
similarity index 80%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.cc
rename to media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.cc
index 6833e0e8af..9f9a324bcb 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/source/event_posix.h"
+#include "webrtc/system_wrappers/source/event_timer_posix.h"
 
 #include <errno.h>
 #include <pthread.h>
@@ -22,17 +22,17 @@
 
 namespace webrtc {
 
+// static
+EventTimerWrapper* EventTimerWrapper::Create() {
+  return new EventTimerPosix();
+}
+
 const long int E6 = 1000000;
 const long int E9 = 1000 * E6;
 
-EventWrapper* EventPosix::Create() {
-  return new EventPosix();
-}
-
-EventPosix::EventPosix()
+EventTimerPosix::EventTimerPosix()
     : event_set_(false),
       timer_thread_(nullptr),
-      timer_event_(0),
       created_at_(),
       periodic_(false),
       time_(0),
@@ -52,24 +52,24 @@ EventPosix::EventPosix()
 #endif
 }
 
-EventPosix::~EventPosix() {
+EventTimerPosix::~EventTimerPosix() {
   StopTimer();
   pthread_cond_destroy(&cond_);
   pthread_mutex_destroy(&mutex_);
 }
 
 // TODO(pbos): Make this void.
-bool EventPosix::Set() {
-  CHECK_EQ(0, pthread_mutex_lock(&mutex_));
+bool EventTimerPosix::Set() {
+  RTC_CHECK_EQ(0, pthread_mutex_lock(&mutex_));
   event_set_ = true;
   pthread_cond_signal(&cond_);
   pthread_mutex_unlock(&mutex_);
   return true;
 }
 
-EventTypeWrapper EventPosix::Wait(unsigned long timeout) {
+EventTypeWrapper EventTimerPosix::Wait(unsigned long timeout) {
   int ret_val = 0;
-  CHECK_EQ(0, pthread_mutex_lock(&mutex_));
+  RTC_CHECK_EQ(0, pthread_mutex_lock(&mutex_));
 
   if (!event_set_) {
     if (WEBRTC_EVENT_INFINITE != timeout) {
@@ -103,7 +103,7 @@ EventTypeWrapper EventPosix::Wait(unsigned long timeout) {
     }
   }
 
-  DCHECK(ret_val == 0 || ret_val == ETIMEDOUT);
+  RTC_DCHECK(ret_val == 0 || ret_val == ETIMEDOUT);
 
   // Reset and signal if set, regardless of why the thread woke up.
   if (event_set_) {
@@ -115,14 +115,14 @@ EventTypeWrapper EventPosix::Wait(unsigned long timeout) {
   return ret_val == 0 ? kEventSignaled : kEventTimeout;
 }
 
-EventTypeWrapper EventPosix::Wait(timespec* end_at) {
+EventTypeWrapper EventTimerPosix::Wait(timespec* end_at) {
   int ret_val = 0;
-  CHECK_EQ(0, pthread_mutex_lock(&mutex_));
+  RTC_CHECK_EQ(0, pthread_mutex_lock(&mutex_));
 
   while (ret_val == 0 && !event_set_)
     ret_val = pthread_cond_timedwait(&cond_, &mutex_, end_at);
 
-  DCHECK(ret_val == 0 || ret_val == ETIMEDOUT);
+  RTC_DCHECK(ret_val == 0 || ret_val == ETIMEDOUT);
 
   // Reset and signal if set, regardless of why the thread woke up.
   if (event_set_) {
@@ -134,7 +134,7 @@ EventTypeWrapper EventPosix::Wait(timespec* end_at) {
   return ret_val == 0 ? kEventSignaled : kEventTimeout;
 }
 
-bool EventPosix::StartTimer(bool periodic, unsigned long time) {
+bool EventTimerPosix::StartTimer(bool periodic, unsigned long time) {
   pthread_mutex_lock(&mutex_);
   if (timer_thread_) {
     if (periodic_) {
@@ -152,23 +152,23 @@ bool EventPosix::StartTimer(bool periodic, unsigned long time) {
   }
 
   // Start the timer thread
-  timer_event_ = static_cast<EventPosix*>(EventWrapper::Create());
+  timer_event_.reset(new EventTimerPosix());
   const char* thread_name = "WebRtc_event_timer_thread";
-  timer_thread_ = ThreadWrapper::CreateThread(Run, this, thread_name);
+  timer_thread_.reset(new rtc::PlatformThread(Run, this, thread_name));
   periodic_ = periodic;
   time_ = time;
-  bool started = timer_thread_->Start();
-  timer_thread_->SetPriority(kRealtimePriority);
+  timer_thread_->Start();
+  timer_thread_->SetPriority(rtc::kRealtimePriority);
   pthread_mutex_unlock(&mutex_);
 
-  return started;
+  return true;
 }
 
-bool EventPosix::Run(void* obj) {
-  return static_cast<EventPosix*>(obj)->Process();
+bool EventTimerPosix::Run(void* obj) {
+  return static_cast<EventTimerPosix*>(obj)->Process();
 }
 
-bool EventPosix::Process() {
+bool EventTimerPosix::Process() {
   pthread_mutex_lock(&mutex_);
   if (created_at_.tv_sec == 0) {
 #ifndef WEBRTC_MAC
@@ -210,20 +210,15 @@ bool EventPosix::Process() {
   return true;
 }
 
-bool EventPosix::StopTimer() {
+bool EventTimerPosix::StopTimer() {
   if (timer_event_) {
     timer_event_->Set();
   }
   if (timer_thread_) {
-    if (!timer_thread_->Stop()) {
-      return false;
-    }
+    timer_thread_->Stop();
     timer_thread_.reset();
   }
-  if (timer_event_) {
-    delete timer_event_;
-    timer_event_ = 0;
-  }
+  timer_event_.reset();
 
   // Set time to zero to force new reference time for the timer.
   memset(&created_at_, 0, sizeof(created_at_));
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.h b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.h
similarity index 76%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.h
rename to media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.h
index f1105ec02f..bbf51f72db 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event_posix.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_posix.h
@@ -11,12 +11,12 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 
 #include <pthread.h>
 #include <time.h>
 
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
 
 namespace webrtc {
 
@@ -25,11 +25,10 @@ enum State {
   kDown = 2
 };
 
-class EventPosix : public EventWrapper {
+class EventTimerPosix : public EventTimerWrapper {
  public:
-  static EventWrapper* Create();
-
-  ~EventPosix() override;
+  EventTimerPosix();
+  ~EventTimerPosix() override;
 
   EventTypeWrapper Wait(unsigned long max_time) override;
   bool Set() override;
@@ -38,8 +37,6 @@ class EventPosix : public EventWrapper {
   bool StopTimer() override;
 
  private:
-  EventPosix();
-
   static bool Run(void* obj);
   bool Process();
   EventTypeWrapper Wait(timespec* end_at);
@@ -49,8 +46,9 @@ class EventPosix : public EventWrapper {
   pthread_mutex_t mutex_;
   bool event_set_;
 
-  rtc::scoped_ptr<ThreadWrapper> timer_thread_;
-  EventPosix*    timer_event_;
+  // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+  rtc::scoped_ptr<rtc::PlatformThread> timer_thread_;
+  rtc::scoped_ptr<EventTimerPosix> timer_event_;
   timespec       created_at_;
 
   bool          periodic_;
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event_win.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.cc
similarity index 53%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/event_win.cc
rename to media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.cc
index 425f3f2b2a..4c586988df 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event_win.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.cc
@@ -8,37 +8,36 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/source/event_win.h"
+#include "webrtc/system_wrappers/source/event_timer_win.h"
 
 #include "Mmsystem.h"
 
 namespace webrtc {
 
-EventWindows::EventWindows()
+// static
+EventTimerWrapper* EventTimerWrapper::Create() {
+  return new EventTimerWin();
+}
+
+EventTimerWin::EventTimerWin()
     : event_(::CreateEvent(NULL,    // security attributes
                            FALSE,   // manual reset
                            FALSE,   // initial state
                            NULL)),  // name of event
-#ifdef WIN32_USE_TIMER_QUEUES
-    timerHandle_(NULL),
-    pulse_(false)
-#else
-    timerID_(NULL)
-#endif
-{
+    timerID_(NULL) {
 }
 
-EventWindows::~EventWindows() {
+EventTimerWin::~EventTimerWin() {
   StopTimer();
   CloseHandle(event_);
 }
 
-bool EventWindows::Set() {
+bool EventTimerWin::Set() {
   // Note: setting an event that is already set has no effect.
   return SetEvent(event_) == 1;
 }
 
-EventTypeWrapper EventWindows::Wait(unsigned long max_time) {
+EventTypeWrapper EventTimerWin::Wait(unsigned long max_time) {
   unsigned long res = WaitForSingleObject(event_, max_time);
   switch (res) {
     case WAIT_OBJECT_0:
@@ -50,35 +49,7 @@ EventTypeWrapper EventWindows::Wait(unsigned long max_time) {
   }
 }
 
-#ifdef WIN32_USE_TIMER_QUEUES
-// static
-void CALLBACK EventWindows::TimerRoutine(PVOID lpParam, BOOLEAN TimerOrWaitFired)
-{
-  EventWindows *eventwin = (EventWindows*) lpParam;
-  if (eventwin->pulse_) {
-    PulseEvent(eventwin->event_);
-  } else {
-    SetEvent(eventwin->event_);
-  }
-}
-#endif
-
-bool EventWindows::StartTimer(bool periodic, unsigned long time) {
-#ifdef WIN32_USE_TIMER_QUEUES
-  if (timerHandle_) {
-    // Wait for running timer callbacks to finish
-    DeleteTimerQueueTimer(NULL, timerHandle_, INVALID_HANDLE_VALUE);
-    timerHandle_ = NULL;
-  }
-  pulse_ = periodic;
-  if (!CreateTimerQueueTimer(&timerHandle_, NULL,
-                             (WAITORTIMERCALLBACK) TimerRoutine,
-                             (PVOID) this, time, periodic ? time : 0,
-                             WT_EXECUTEINTIMERTHREAD)) {
-    return false;
-  }
-  return true;
-#else
+bool EventTimerWin::StartTimer(bool periodic, unsigned long time) {
   if (timerID_ != NULL) {
     timeKillEvent(timerID_);
     timerID_ = NULL;
@@ -93,22 +64,13 @@ bool EventWindows::StartTimer(bool periodic, unsigned long time) {
   }
 
   return timerID_ != NULL;
-#endif
 }
 
-bool EventWindows::StopTimer() {
-#ifdef WIN32_USE_TIMER_QUEUES
-  if (timerHandle_) {
-    // Wait for running timer callbacks to finish
-    DeleteTimerQueueTimer(NULL, timerHandle_, INVALID_HANDLE_VALUE);
-    timerHandle_ = NULL;
-  }
-#else
+bool EventTimerWin::StopTimer() {
   if (timerID_ != NULL) {
     timeKillEvent(timerID_);
     timerID_ = NULL;
   }
-#endif
 
   return true;
 }
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/event_win.h b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.h
similarity index 85%
rename from media/webrtc/trunk/webrtc/system_wrappers/source/event_win.h
rename to media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.h
index c0915c4dff..163cddeda9 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/event_win.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/event_timer_win.h
@@ -13,16 +13,16 @@
 
 #include <windows.h>
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-class EventWindows : public EventWrapper {
+class EventTimerWin : public EventTimerWrapper {
  public:
-  EventWindows();
-  virtual ~EventWindows();
+  EventTimerWin();
+  virtual ~EventTimerWin();
 
   virtual EventTypeWrapper Wait(unsigned long max_time);
   virtual bool Set();
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/field_trial_default.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/field_trial_default.cc
index 97b703f6b1..0e2c286117 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/field_trial_default.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/field_trial_default.cc
@@ -7,8 +7,8 @@
 // be found in the AUTHORS file in the root of the source tree.
 //
 
-#include "webrtc/system_wrappers/interface/field_trial.h"
-#include "webrtc/system_wrappers/interface/field_trial_default.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/system_wrappers/include/field_trial_default.h"
 
 #include <string>
 
@@ -58,5 +58,9 @@ void InitFieldTrialsFromString(const char* trials_string) {
   trials_init_string = trials_string;
 }
 
+const char* GetFieldTrialString() {
+  return trials_init_string;
+}
+
 }  // namespace field_trial
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.cc
index dfb138897f..0ee0deab6c 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.cc
@@ -20,7 +20,7 @@
 #endif
 
 #include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 namespace webrtc {
 
@@ -271,7 +271,7 @@ int FileWrapperImpl::FlushImpl() {
 }
 
 int FileWrapper::Rewind() {
-  DCHECK(false);
+  RTC_DCHECK(false);
   return -1;
 }
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.h b/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.h
index e6679aa8e0..06ba58200b 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/file_impl.h
@@ -14,7 +14,7 @@
 #include <stdio.h>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/logcat_trace_context.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/logcat_trace_context.cc
index 313acc74e8..cc2e45b28e 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/logcat_trace_context.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/logcat_trace_context.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/logcat_trace_context.h"
+#include "webrtc/system_wrappers/include/logcat_trace_context.h"
 
 #include <android/log.h>
 #include <assert.h>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/logging.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/logging.cc
index da1c1a57d8..6b50d6acf8 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/logging.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/logging.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 #include <string.h>
 
 #include <sstream>
 
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace {
@@ -32,7 +32,8 @@ TraceLevel WebRtcSeverity(LoggingSeverity sev) {
   }
 }
 
-const char* DescribeFile(const char* file) {
+// Return the filename portion of the string (that following the last slash).
+const char* FilenameFromPath(const char* file) {
   const char* end1 = ::strrchr(file, '/');
   const char* end2 = ::strrchr(file, '\\');
   if (!end1 && !end2)
@@ -45,7 +46,7 @@ const char* DescribeFile(const char* file) {
 
 LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev)
     : severity_(sev) {
-  print_stream_ << "(" << DescribeFile(file) << ":" << line << "): ";
+  print_stream_ << "(" << FilenameFromPath(file) << ":" << line << "): ";
 }
 
 bool LogMessage::Loggable(LoggingSeverity sev) {
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/logging_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/logging_unittest.cc
index 39bca65689..2da24b26f4 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/logging_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/logging_unittest.cc
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace {
@@ -72,18 +72,5 @@ TEST_F(LoggingTest, LogStream) {
   }
 }
 
-TEST_F(LoggingTest, LogFunctionError) {
-  {
-    CriticalSectionScoped cs(crit_.get());
-    int bar = 42;
-    int baz = 99;
-    level_ = kTraceError;
-    expected_log_ << "(logging_unittest.cc:" << __LINE__ + 2
-                  << "): Foo failed: bar=" << bar << ", baz=" << baz;
-    LOG_FERR2(LS_ERROR, Foo, bar, baz);
-    cv_->SleepCS(*crit_.get(), 2000);
-  }
-}
-
 }  // namespace
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_default.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_default.cc
index af950b4d91..48c9111e1b 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_default.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_default.cc
@@ -7,7 +7,7 @@
 // be found in the AUTHORS file in the root of the source tree.
 //
 
-#include "webrtc/system_wrappers/interface/metrics.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 // Default implementation of histogram methods for WebRTC clients that do not
 // want to provide their own implementation.
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_unittest.cc
new file mode 100644
index 0000000000..8319b78ee0
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/metrics_unittest.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/test/histogram.h"
+
+namespace webrtc {
+namespace {
+const int kSample = 22;
+const std::string kName = "Name";
+
+void AddSparseSample(const std::string& name, int sample) {
+  RTC_HISTOGRAM_COUNTS_SPARSE_100(name, sample);
+}
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+void AddSample(const std::string& name, int sample) {
+  RTC_HISTOGRAM_COUNTS_100(name, sample);
+}
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+}  // namespace
+
+TEST(MetricsTest, InitiallyNoSamples) {
+  test::ClearHistograms();
+  EXPECT_EQ(0, test::NumHistogramSamples(kName));
+  EXPECT_EQ(-1, test::LastHistogramSample(kName));
+}
+
+TEST(MetricsTest, RtcHistogramPercent_AddSample) {
+  test::ClearHistograms();
+  RTC_HISTOGRAM_PERCENTAGE(kName, kSample);
+  EXPECT_EQ(1, test::NumHistogramSamples(kName));
+  EXPECT_EQ(kSample, test::LastHistogramSample(kName));
+}
+
+TEST(MetricsTest, RtcHistogramEnumeration_AddSample) {
+  test::ClearHistograms();
+  RTC_HISTOGRAM_ENUMERATION(kName, kSample, kSample + 1);
+  EXPECT_EQ(1, test::NumHistogramSamples(kName));
+  EXPECT_EQ(kSample, test::LastHistogramSample(kName));
+}
+
+TEST(MetricsTest, RtcHistogramCountsSparse_AddSample) {
+  test::ClearHistograms();
+  RTC_HISTOGRAM_COUNTS_SPARSE_100(kName, kSample);
+  EXPECT_EQ(1, test::NumHistogramSamples(kName));
+  EXPECT_EQ(kSample, test::LastHistogramSample(kName));
+}
+
+TEST(MetricsTest, RtcHistogramCounts_AddSample) {
+  test::ClearHistograms();
+  RTC_HISTOGRAM_COUNTS_100(kName, kSample);
+  EXPECT_EQ(1, test::NumHistogramSamples(kName));
+  EXPECT_EQ(kSample, test::LastHistogramSample(kName));
+}
+
+TEST(MetricsTest, RtcHistogramCounts_AddMultipleSamples) {
+  test::ClearHistograms();
+  const int kNumSamples = 10;
+  for (int i = 0; i < kNumSamples; ++i) {
+    RTC_HISTOGRAM_COUNTS_100(kName, i);
+  }
+  EXPECT_EQ(kNumSamples, test::NumHistogramSamples(kName));
+  EXPECT_EQ(kNumSamples - 1, test::LastHistogramSample(kName));
+}
+
+TEST(MetricsTest, RtcHistogramSparse_NonConstantNameWorks) {
+  test::ClearHistograms();
+  AddSparseSample("Name1", kSample);
+  AddSparseSample("Name2", kSample);
+  EXPECT_EQ(1, test::NumHistogramSamples("Name1"));
+  EXPECT_EQ(1, test::NumHistogramSamples("Name2"));
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(MetricsTest, RtcHistogram_FailsForNonConstantName) {
+  test::ClearHistograms();
+  AddSample("Name1", kSample);
+  EXPECT_DEATH(AddSample("Name2", kSample), "");
+}
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/ntp_time_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/ntp_time_unittest.cc
new file mode 100644
index 0000000000..ff11288c1b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/ntp_time_unittest.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace {
+
+const uint32_t kNtpSec = 0x12345678;
+const uint32_t kNtpFrac = 0x23456789;
+
+TEST(NtpTimeTest, NoValueMeansInvalid) {
+  NtpTime ntp;
+  EXPECT_FALSE(ntp.Valid());
+}
+
+TEST(NtpTimeTest, CanResetValue) {
+  NtpTime ntp(kNtpSec, kNtpFrac);
+  EXPECT_TRUE(ntp.Valid());
+  ntp.Reset();
+  EXPECT_FALSE(ntp.Valid());
+}
+
+TEST(NtpTimeTest, CanGetWhatIsSet) {
+  NtpTime ntp;
+  ntp.Set(kNtpSec, kNtpFrac);
+  EXPECT_EQ(kNtpSec, ntp.seconds());
+  EXPECT_EQ(kNtpFrac, ntp.fractions());
+}
+
+TEST(NtpTimeTest, SetIsSameAs2ParameterConstructor) {
+  NtpTime ntp1(kNtpSec, kNtpFrac);
+  NtpTime ntp2;
+  EXPECT_NE(ntp1, ntp2);
+
+  ntp2.Set(kNtpSec, kNtpFrac);
+  EXPECT_EQ(ntp1, ntp2);
+}
+
+TEST(NtpTimeTest, SetCurrentIsSameAs1ParameterConstructor) {
+  SimulatedClock clock(0x0123456789abcdef);
+
+  NtpTime ntp1(clock);
+  NtpTime ntp2;
+  EXPECT_NE(ntp1, ntp2);
+
+  ntp2.SetCurrent(clock);
+  EXPECT_EQ(ntp1, ntp2);
+}
+
+TEST(NtpTimeTest, ToMsMeansToNtpMilliseconds) {
+  SimulatedClock clock(0x123456789abc);
+
+  NtpTime ntp(clock);
+  EXPECT_EQ(ntp.ToMs(), Clock::NtpToMs(ntp.seconds(), ntp.fractions()));
+  EXPECT_EQ(ntp.ToMs(), clock.CurrentNtpInMilliseconds());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp.cc
index d6b7b14084..0aceb0625f 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp.cc
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 #include <assert.h>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
index a4d75aed04..4c166774a4 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
@@ -9,7 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock.cc
index 02296b6d1f..3cb2f56897 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 #include <assert.h>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.cc
index 0ca9518747..9786155a63 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.cc
@@ -10,8 +10,8 @@
 
 #include "webrtc/system_wrappers/source/rw_lock_generic.h"
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.h b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.h
index 653564c186..f0d445692e 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_generic.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
 
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_posix.h b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_posix.h
index bec3c2de55..0ce7305b60 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_posix.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_posix.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
 
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
 #include <pthread.h>
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.cc
index a1d4b58195..d29ec35626 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/system_wrappers/source/rw_lock_win.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.h b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.h
index 6f7cd3344e..c279eaba44 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/rw_lock_win.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_
 
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 
 #include <Windows.h>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/scoped_vector_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/scoped_vector_unittest.cc
index 9d7c811229..6e38f01f0b 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/scoped_vector_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/scoped_vector_unittest.cc
@@ -10,7 +10,7 @@
 
 // Borrowed from Chromium's src/base/memory/scoped_vector_unittest.cc
 
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -44,7 +44,7 @@ class LifeCycleObject {
 
   Observer* observer_;
 
-  DISALLOW_COPY_AND_ASSIGN(LifeCycleObject);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LifeCycleObject);
 };
 
 // The life cycle states we care about for the purposes of testing ScopedVector
@@ -107,7 +107,7 @@ class LifeCycleWatcher : public LifeCycleObject::Observer {
   LifeCycleState life_cycle_state_;
   rtc::scoped_ptr<LifeCycleObject> constructed_life_cycle_object_;
 
-  DISALLOW_COPY_AND_ASSIGN(LifeCycleWatcher);
+  RTC_DISALLOW_COPY_AND_ASSIGN(LifeCycleWatcher);
 };
 
 TEST(ScopedVectorTest, LifeCycleWatcher) {
@@ -221,7 +221,8 @@ TEST(ScopedVectorTest, MoveConstruct) {
     EXPECT_FALSE(scoped_vector.empty());
     EXPECT_TRUE(watcher.IsWatching(scoped_vector.back()));
 
-    ScopedVector<LifeCycleObject> scoped_vector_copy(scoped_vector.Pass());
+    ScopedVector<LifeCycleObject> scoped_vector_copy(
+        scoped_vector.DEPRECATED_Pass());
     EXPECT_TRUE(scoped_vector.empty());
     EXPECT_FALSE(scoped_vector_copy.empty());
     EXPECT_TRUE(watcher.IsWatching(scoped_vector_copy.back()));
@@ -241,7 +242,7 @@ TEST(ScopedVectorTest, MoveAssign) {
     EXPECT_FALSE(scoped_vector.empty());
     EXPECT_TRUE(watcher.IsWatching(scoped_vector.back()));
 
-    scoped_vector_assign = scoped_vector.Pass();
+    scoped_vector_assign = scoped_vector.DEPRECATED_Pass();
     EXPECT_TRUE(scoped_vector.empty());
     EXPECT_FALSE(scoped_vector_assign.empty());
     EXPECT_TRUE(watcher.IsWatching(scoped_vector_assign.back()));
@@ -266,17 +267,18 @@ class DeleteCounter {
  private:
   int* const deletes_;
 
-  DISALLOW_COPY_AND_ASSIGN(DeleteCounter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(DeleteCounter);
 };
 
 // This class is used in place of Chromium's base::Callback.
 template <typename T>
 class PassThru  {
  public:
-  explicit PassThru(ScopedVector<T> scoper) : scoper_(scoper.Pass()) {}
+  explicit PassThru(ScopedVector<T> scoper)
+      : scoper_(scoper.DEPRECATED_Pass()) {}
 
   ScopedVector<T> Run() {
-    return scoper_.Pass();
+    return scoper_.DEPRECATED_Pass();
   }
 
  private:
@@ -288,7 +290,7 @@ TEST(ScopedVectorTest, Passed) {
   ScopedVector<DeleteCounter> deleter_vector;
   deleter_vector.push_back(new DeleteCounter(&deletes));
   EXPECT_EQ(0, deletes);
-  PassThru<DeleteCounter> pass_thru(deleter_vector.Pass());
+  PassThru<DeleteCounter> pass_thru(deleter_vector.DEPRECATED_Pass());
   EXPECT_EQ(0, deletes);
   ScopedVector<DeleteCounter> result = pass_thru.Run();
   EXPECT_EQ(0, deletes);
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/sleep.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/sleep.cc
index a916477e37..181381fd53 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/sleep.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/sleep.cc
@@ -9,7 +9,7 @@
  */
 // An OS-independent sleep function.
 
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 
 #ifdef _WIN32
 // For Sleep()
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/sort.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/sort.cc
index 9144a58c02..f166f95311 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/sort.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/sort.cc
@@ -13,7 +13,7 @@
 // part of the Boost C++ library collection. Otherwise, the C standard library's
 // qsort() will be used.
 
-#include "webrtc/system_wrappers/interface/sort.h"
+#include "webrtc/system_wrappers/include/sort.h"
 
 #include <assert.h>
 #include <string.h>  // memcpy
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/stl_util_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/stl_util_unittest.cc
index e60a913cfc..ed5c1d9590 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/stl_util_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/stl_util_unittest.cc
@@ -9,7 +9,7 @@
  */
 
 // Borrowed from Chromium's src/base/stl_util_unittest.cc
-#include "webrtc/system_wrappers/interface/stl_util.h"
+#include "webrtc/system_wrappers/include/stl_util.h"
 
 #include <set>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/stringize_macros_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/stringize_macros_unittest.cc
index 8d953dd540..c2f312bf90 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/stringize_macros_unittest.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/stringize_macros_unittest.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/stringize_macros.h"
+#include "webrtc/system_wrappers/include/stringize_macros.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/thread.cc
deleted file mode 100644
index 18e31ea91d..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread.cc
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-#if defined(_WIN32)
-#include "webrtc/system_wrappers/source/thread_win.h"
-#else
-#include "webrtc/system_wrappers/source/thread_posix.h"
-#endif
-
-namespace webrtc {
-
-#if defined(_WIN32)
-typedef ThreadWindows ThreadType;
-#else
-typedef ThreadPosix ThreadType;
-#endif
-
-rtc::scoped_ptr<ThreadWrapper> ThreadWrapper::CreateThread(
-    ThreadRunFunction func, void* obj, const char* thread_name) {
-  return rtc::scoped_ptr<ThreadWrapper>(
-      new ThreadType(func, obj, thread_name)).Pass();
-}
-
-rtc::scoped_ptr<ThreadWrapper> ThreadWrapper::CreateUIThread(
-    ThreadRunFunction func, void* obj, const char* thread_name) {
-#if defined(_WIN32)
-  return rtc::scoped_ptr<ThreadWrapper>(
-    new ThreadWindowsUI(func, obj, thread_name)).Pass();
-#else
-  return rtc::scoped_ptr<ThreadWrapper>(
-    new ThreadType(func, obj, thread_name)).Pass();
-#endif
-}
-
-bool ThreadWrapper::RequestCallbackTimer(unsigned int milliseconds) {
-  return false;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix.cc
deleted file mode 100644
index 5ed77e87e6..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix.cc
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/source/thread_posix.h"
-
-#include <algorithm>
-
-#include <errno.h>
-#include <unistd.h>
-#ifdef WEBRTC_LINUX
-#include <linux/unistd.h>
-#include <sched.h>
-#include <sys/prctl.h>
-#include <sys/syscall.h>
-#include <sys/types.h>
-#endif
-
-#if defined(WEBRTC_BSD) && !defined(__NetBSD__)
-#include <pthread_np.h>
-#endif
-
-#include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-namespace {
-struct ThreadAttributes {
-  ThreadAttributes() { pthread_attr_init(&attr); }
-  ~ThreadAttributes() { pthread_attr_destroy(&attr); }
-  pthread_attr_t* operator&() { return &attr; }
-  pthread_attr_t attr;
-};
-}  // namespace
-
-int ConvertToSystemPriority(ThreadPriority priority, int min_prio,
-                            int max_prio) {
-  DCHECK(max_prio - min_prio > 2);
-  const int top_prio = max_prio - 1;
-  const int low_prio = min_prio + 1;
-
-  switch (priority) {
-    case kLowPriority:
-      return low_prio;
-    case kNormalPriority:
-      // The -1 ensures that the kHighPriority is always greater or equal to
-      // kNormalPriority.
-      return (low_prio + top_prio - 1) / 2;
-    case kHighPriority:
-      return std::max(top_prio - 2, low_prio);
-    case kHighestPriority:
-      return std::max(top_prio - 1, low_prio);
-    case kRealtimePriority:
-      return top_prio;
-  }
-  DCHECK(false);
-  return low_prio;
-}
-
-// static
-void* ThreadPosix::StartThread(void* param) {
-  static_cast<ThreadPosix*>(param)->Run();
-  return 0;
-}
-
-ThreadPosix::ThreadPosix(ThreadRunFunction func, void* obj,
-                         const char* thread_name)
-    : run_function_(func),
-      obj_(obj),
-      stop_event_(false, false),
-      name_(thread_name ? thread_name : "webrtc"),
-      thread_(0) {
-  DCHECK(name_.length() < 64);
-}
-
-uint32_t ThreadWrapper::GetThreadId() {
-  return rtc::CurrentThreadId();
-}
-
-ThreadPosix::~ThreadPosix() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-// TODO(pbos): Make Start void, calling code really doesn't support failures
-// here.
-bool ThreadPosix::Start() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!thread_) << "Thread already started?";
-
-  ThreadAttributes attr;
-  // Set the stack stack size to 1M.
-  pthread_attr_setstacksize(&attr, 1024 * 1024);
-  CHECK_EQ(0, pthread_create(&thread_, &attr, &StartThread, this));
-  return true;
-}
-
-bool ThreadPosix::Stop() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (!thread_)
-    return true;
-
-  stop_event_.Set();
-  CHECK_EQ(0, pthread_join(thread_, nullptr));
-  thread_ = 0;
-
-  return true;
-}
-
-bool ThreadPosix::SetPriority(ThreadPriority priority) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (!thread_)
-    return false;
-#if defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_LINUX)
-  // TODO(tommi): Switch to the same mechanism as Chromium uses for
-  // changing thread priorities.
-  return true;
-#else
-#ifdef WEBRTC_THREAD_RR
-  const int policy = SCHED_RR;
-#else
-  const int policy = SCHED_FIFO;
-#endif
-  const int min_prio = sched_get_priority_min(policy);
-  const int max_prio = sched_get_priority_max(policy);
-  if (min_prio == -1 || max_prio == -1) {
-    WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
-                 "unable to retreive min or max priority for threads");
-    return false;
-  }
-
-  if (max_prio - min_prio <= 2)
-    return false;
-
-  sched_param param;
-  param.sched_priority = ConvertToSystemPriority(priority, min_prio, max_prio);
-  if (pthread_setschedparam(thread_, policy, &param) != 0) {
-    WEBRTC_TRACE(
-        kTraceError, kTraceUtility, -1, "unable to set thread priority");
-    return false;
-  }
-
-  return true;
-#endif  // defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_LINUX)
-}
-
-void ThreadPosix::Run() {
-  if (!name_.empty()) {
-    // Setting the thread name may fail (harmlessly) if running inside a
-    // sandbox. Ignore failures if they happen.
-#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID) || defined(WEBRTC_GONK))
-    prctl(PR_SET_NAME, reinterpret_cast<unsigned long>(name_.c_str()));
-#elif defined(__NetBSD__)
-    pthread_setname_np(pthread_self(), "%s", (void *)name_.c_str());
-#elif defined(WEBRTC_BSD)
-    pthread_set_name_np(pthread_self(), name_.c_str());
-#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
-    pthread_setname_np(name_.substr(0, 63).c_str());
-#endif
-  }
-
-  // It's a requirement that for successful thread creation that the run
-  // function be called at least once (see RunFunctionIsCalled unit test),
-  // so to fullfill that requirement, we use a |do| loop and not |while|.
-  do {
-    if (!run_function_(obj_))
-      break;
-  } while (!stop_event_.Wait(0));
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix.h b/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix.h
deleted file mode 100644
index c726e480cf..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
-
-#include "webrtc/base/event.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-
-int ConvertToSystemPriority(ThreadPriority priority, int min_prio,
-                            int max_prio);
-
-class ThreadPosix : public ThreadWrapper {
- public:
-  ThreadPosix(ThreadRunFunction func, void* obj, const char* thread_name);
-  ~ThreadPosix() override;
-
-  // From ThreadWrapper.
-  bool Start() override;
-  bool Stop() override;
-
-  bool SetPriority(ThreadPriority priority) override;
-
- private:
-  static void* StartThread(void* param);
-
-  void Run();
-
-  rtc::ThreadChecker thread_checker_;
-  ThreadRunFunction const run_function_;
-  void* const obj_;
-  rtc::Event stop_event_;
-  const std::string name_;
-
-  pthread_t thread_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix_unittest.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix_unittest.cc
deleted file mode 100644
index edfb14502e..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_posix_unittest.cc
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/source/thread_posix.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-TEST(ThreadTestPosix, PrioritySettings) {
-  // API assumes that max_prio - min_prio > 2. Test the extreme case.
-  const int kMinPrio = -1;
-  const int kMaxPrio = 2;
-
-  int last_priority = kMinPrio;
-  for (int priority = webrtc::kLowPriority;
-       priority <= webrtc::kRealtimePriority; ++priority) {
-    int system_priority = webrtc::ConvertToSystemPriority(
-        static_cast<webrtc::ThreadPriority>(priority), kMinPrio, kMaxPrio);
-    EXPECT_GT(system_priority, kMinPrio);
-    EXPECT_LT(system_priority, kMaxPrio);
-    EXPECT_GE(system_priority, last_priority);
-    last_priority = system_priority;
-  }
-}
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_win.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/thread_win.cc
deleted file mode 100644
index a40e5b0db0..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_win.cc
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/source/thread_win.h"
-
-#include <process.h>
-#include <stdio.h>
-#include <windows.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-namespace {
-void CALLBACK RaiseFlag(ULONG_PTR param) {
-  *reinterpret_cast<bool*>(param) = true;
-}
-
-// TODO(tommi): This is borrowed from webrtc/base/thread.cc, but we can't
-// include thread.h from here since thread.h pulls in libjingle dependencies.
-// Would be good to consolidate.
-
-// As seen on MSDN.
-// http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx
-#define MSDEV_SET_THREAD_NAME  0x406D1388
-typedef struct tagTHREADNAME_INFO {
-  DWORD dwType;
-  LPCSTR szName;
-  DWORD dwThreadID;
-  DWORD dwFlags;
-} THREADNAME_INFO;
-
-void SetThreadName(DWORD dwThreadID, LPCSTR szThreadName) {
-  THREADNAME_INFO info;
-  info.dwType = 0x1000;
-  info.szName = szThreadName;
-  info.dwThreadID = dwThreadID;
-  info.dwFlags = 0;
-
-  __try {
-    RaiseException(MSDEV_SET_THREAD_NAME, 0, sizeof(info) / sizeof(DWORD),
-                   reinterpret_cast<ULONG_PTR*>(&info));
-  }
-  __except(EXCEPTION_CONTINUE_EXECUTION) {
-  }
-}
-
-}
-
-// For use in ThreadWindowsUI callbacks
-static UINT static_reg_windows_msg = RegisterWindowMessageW(L"WebrtcWindowsUIThreadEvent");
-// timer id used in delayed callbacks
-static const UINT_PTR kTimerId = 1;
-static const wchar_t kThisProperty[] = L"ThreadWindowsUIPtr";
-static const wchar_t kThreadWindow[] = L"WebrtcWindowsUIThread";
-
-ThreadWindows::ThreadWindows(ThreadRunFunction func, void* obj,
-                             const char* thread_name)
-    : run_function_(func),
-      obj_(obj),
-      stop_(false),
-      thread_(NULL),
-      name_(thread_name ? thread_name : "webrtc") {
-  DCHECK(func);
-}
-
-ThreadWindows::~ThreadWindows() {
-  DCHECK(main_thread_.CalledOnValidThread());
-  DCHECK(!thread_);
-}
-
-// static
-uint32_t ThreadWrapper::GetThreadId() {
-  return GetCurrentThreadId();
-}
-
-// static
-DWORD WINAPI ThreadWindows::StartThread(void* param) {
-  static_cast<ThreadWindows*>(param)->Run();
-  return 0;
-}
-
-bool ThreadWindows::Start() {
-  DCHECK(main_thread_.CalledOnValidThread());
-  DCHECK(!thread_);
-
-  stop_ = false;
-
-  // See bug 2902 for background on STACK_SIZE_PARAM_IS_A_RESERVATION.
-  // Set the reserved stack stack size to 1M, which is the default on Windows
-  // and Linux.
-  DWORD thread_id;
-  thread_ = ::CreateThread(NULL, 1024 * 1024, &StartThread, this,
-      STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
-  if (!thread_ ) {
-    DCHECK(false) << "CreateThread failed";
-    return false;
-  }
-
-  return true;
-}
-
-bool ThreadWindows::Stop() {
-  DCHECK(main_thread_.CalledOnValidThread());
-  if (thread_) {
-    // Set stop_ to |true| on the worker thread.
-    QueueUserAPC(&RaiseFlag, thread_, reinterpret_cast<ULONG_PTR>(&stop_));
-    WaitForSingleObject(thread_, INFINITE);
-    CloseHandle(thread_);
-    thread_ = nullptr;
-  }
-
-  return true;
-}
-
-bool ThreadWindows::SetPriority(ThreadPriority priority) {
-  DCHECK(main_thread_.CalledOnValidThread());
-  return thread_ && SetThreadPriority(thread_, priority);
-}
-
-void ThreadWindows::Run() {
-  if (!name_.empty())
-    SetThreadName(static_cast<DWORD>(-1), name_.c_str());
-
-  do {
-    // The interface contract of Start/Stop is that for a successfull call to
-    // Start, there should be at least one call to the run function.  So we
-    // call the function before checking |stop_|.
-    if (!run_function_(obj_))
-      break;
-    // Alertable sleep to permit RaiseFlag to run and update |stop_|.
-    SleepEx(0, true);
-  } while (!stop_);
-}
-
-bool ThreadWindowsUI::Stop() {
-  DCHECK(main_thread_.CalledOnValidThread());
-
-  // Shut down the dispatch loop and let the background thread exit.
-  if (timerid_) {
-    KillTimer(hwnd_, timerid_);
-    timerid_ = 0;
-  }
-
-  PostMessage(hwnd_, WM_CLOSE, 0, 0);
-
-  return ThreadWindows::Stop();
-}
-
-bool ThreadWindowsUI::InternalInit() {
-  // Create an event window for use in generating callbacks to capture
-  // objects.
-  if (hwnd_ == NULL) {
-    WNDCLASSW wc;
-    HMODULE hModule = GetModuleHandle(NULL);
-    if (!GetClassInfoW(hModule, kThreadWindow, &wc)) {
-      ZeroMemory(&wc, sizeof(WNDCLASSW));
-      wc.hInstance = hModule;
-      wc.lpfnWndProc = EventWindowProc;
-      wc.lpszClassName = kThreadWindow;
-      RegisterClassW(&wc);
-    }
-    hwnd_ = CreateWindowW(kThreadWindow, L"",
-                          0, 0, 0, 0, 0,
-                          NULL, NULL, hModule, NULL);
-    assert(hwnd_);
-    SetPropW(hwnd_, kThisProperty, this);
-
-    if (timeout_) {
-      // if someone set the timer before we started
-      RequestCallbackTimer(timeout_);
-    }
-  }
-  return !!hwnd_;
-}
-
-void ThreadWindowsUI::RequestCallback() {
-  assert(hwnd_);
-  assert(static_reg_windows_msg);
-  PostMessage(hwnd_, static_reg_windows_msg, 0, 0);
-}
-
-bool ThreadWindowsUI::RequestCallbackTimer(unsigned int milliseconds) {
-  if (!hwnd_) {
-    assert(!thread_);
-    // set timer once thread starts
-  } else {
-    if (timerid_) {
-      KillTimer(hwnd_, timerid_);
-    }
-    timerid_ = SetTimer(hwnd_, kTimerId, milliseconds, NULL);
-  }
-  timeout_ = milliseconds;
-  return !!timerid_;
-}
-
-void ThreadWindowsUI::Run() {
-  if (!InternalInit()) {
-    assert(false);
-  }
-
-  if (!name_.empty())
-    SetThreadName(static_cast<DWORD>(-1), name_.c_str());
-
-  do {
-    // The interface contract of Start/Stop is that for a successful call to
-    // Start, there should be at least one call to the run function.  So we
-    // call the function before checking |stop_|.
-    if (!run_function_(obj_))
-      break;
-
-    // Alertable sleep to permit RaiseFlag to run and update |stop_|.
-    if (MsgWaitForMultipleObjectsEx(0, nullptr, INFINITE, QS_ALLINPUT,
-                                    MWMO_ALERTABLE | MWMO_INPUTAVAILABLE) ==
-        WAIT_OBJECT_0) {
-      MSG msg;
-      if (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
-        if (msg.message == WM_QUIT) {
-          stop_ = true;
-          break;
-        }
-        TranslateMessage(&msg);
-        DispatchMessage(&msg);
-      }
-    }
-
-  } while (!stop_);
-
-  // Don't need to DestroyWindow(hwnd_) due to WM_CLOSE->WM_DESTROY handling
-};
-
-void
-ThreadWindowsUI::NativeEventCallback() {
-  if (!run_function_) {
-    stop_ = true;
-    return;
-  }
-  stop_ = !run_function_(obj_);
-}
-
-/* static */
-LRESULT CALLBACK
-ThreadWindowsUI::EventWindowProc(HWND hwnd, UINT uMsg, WPARAM wParam, LPARAM lParam) {
-  if (uMsg == WM_DESTROY) {
-    RemovePropW(hwnd, kThisProperty);
-    PostQuitMessage(0);
-    return 0;
-  }
-
-  ThreadWindowsUI *twui = static_cast<ThreadWindowsUI*>(GetPropW(hwnd, kThisProperty));
-  if (!twui) {
-    return DefWindowProc(hwnd, uMsg, wParam, lParam);
-  }
-
-  if ((uMsg == static_reg_windows_msg && uMsg != WM_NULL) ||
-      (uMsg == WM_TIMER && wParam == kTimerId)) {
-    twui->NativeEventCallback();
-    return 0;
-  }
-
-  return DefWindowProc(hwnd, uMsg, wParam, lParam);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_win.h b/media/webrtc/trunk/webrtc/system_wrappers/source/thread_win.h
deleted file mode 100644
index 232dad6625..0000000000
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/thread_win.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WIN_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WIN_H_
-
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-#include <windows.h>
-
-#include "webrtc/base/thread_checker.h"
-
-namespace webrtc {
-
-class ThreadWindows : public ThreadWrapper {
- public:
-  ThreadWindows(ThreadRunFunction func, void* obj, const char* thread_name);
-  ~ThreadWindows() override;
-
-  virtual bool Start() override;
-  virtual bool Stop() override;
-
-  bool SetPriority(ThreadPriority priority) override;
-
- protected:
-  virtual void Run();
-
-  static DWORD WINAPI StartThread(void* param);
-
-  ThreadRunFunction const run_function_;
-  void* const obj_;
-  bool stop_;
-  HANDLE thread_;
-  const std::string name_;
-  rtc::ThreadChecker main_thread_;
-};
-
-class ThreadWindowsUI : public ThreadWindows {
- public:
-  ThreadWindowsUI(ThreadRunFunction func, void* obj,
-		  const char* thread_name) :
-  ThreadWindows(func, obj, thread_name),
-  hwnd_(nullptr),
-  timerid_(0),
-  timeout_(0) {
- }
-
- virtual bool Stop() override;
-
- /**
-  * Request an async callback soon.
-  */
- void RequestCallback();
-
- /**
-  * Request a recurring callback.
-  */
- bool RequestCallbackTimer(unsigned int milliseconds);
-
- protected:
-  virtual void Run() override;
-
- private:
-  static LRESULT CALLBACK EventWindowProc(HWND, UINT, WPARAM, LPARAM);
-  void NativeEventCallback();
-  bool InternalInit();
-
-  HWND hwnd_;
-  UINT_PTR timerid_;
-  unsigned int timeout_;
-};
-
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WIN_H_
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/tick_util.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/tick_util.cc
index 4b5f71aa37..0485e42921 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/tick_util.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/tick_util.cc
@@ -8,91 +8,35 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
-#include <assert.h>
+#include "webrtc/base/timeutils.h"
 
 namespace webrtc {
 
-bool TickTime::use_fake_clock_ = false;
-int64_t TickTime::fake_ticks_ = 0;
-
-void TickTime::UseFakeClock(int64_t start_millisecond) {
-  use_fake_clock_ = true;
-  fake_ticks_ = MillisecondsToTicks(start_millisecond);
+int64_t TickTime::MillisecondTimestamp() {
+  return TicksToMilliseconds(TickTime::Now().Ticks());
 }
 
-void TickTime::AdvanceFakeClock(int64_t milliseconds) {
-  assert(use_fake_clock_);
-  fake_ticks_ += MillisecondsToTicks(milliseconds);
+int64_t TickTime::MicrosecondTimestamp() {
+  return TicksToMicroseconds(TickTime::Now().Ticks());
 }
 
+int64_t TickTime::MillisecondsToTicks(const int64_t ms) {
+  return ms * rtc::kNumNanosecsPerMillisec;
+}
+
+int64_t TickTime::TicksToMilliseconds(const int64_t ticks) {
+  return ticks / rtc::kNumNanosecsPerMillisec;
+}
+
+int64_t TickTime::TicksToMicroseconds(const int64_t ticks) {
+  return ticks / rtc::kNumNanosecsPerMicrosec;
+}
+
+// Gets the native system tick count, converted to nanoseconds.
 int64_t TickTime::QueryOsForTicks() {
-  TickTime result;
-#if _WIN32
-  // TODO(wu): Remove QueryPerformanceCounter implementation.
-#ifdef USE_QUERY_PERFORMANCE_COUNTER
-  // QueryPerformanceCounter returns the value from the TSC which is
-  // incremented at the CPU frequency. The algorithm used requires
-  // the CPU frequency to be constant. Technology like speed stepping
-  // which has variable CPU frequency will therefore yield unpredictable,
-  // incorrect time estimations.
-  LARGE_INTEGER qpcnt;
-  QueryPerformanceCounter(&qpcnt);
-  result.ticks_ = qpcnt.QuadPart;
-#else
-  static volatile LONG last_time_get_time = 0;
-  static volatile int64_t num_wrap_time_get_time = 0;
-  volatile LONG* last_time_get_time_ptr = &last_time_get_time;
-  DWORD now = timeGetTime();
-  // Atomically update the last gotten time
-  DWORD old = InterlockedExchange(last_time_get_time_ptr, now);
-  if (now < old) {
-    // If now is earlier than old, there may have been a race between
-    // threads.
-    // 0x0fffffff ~3.1 days, the code will not take that long to execute
-    // so it must have been a wrap around.
-    if (old > 0xf0000000 && now < 0x0fffffff) {
-      num_wrap_time_get_time++;
-    }
-  }
-  result.ticks_ = now + (num_wrap_time_get_time << 32);
-#endif
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
-  struct timespec ts;
-  // TODO(wu): Remove CLOCK_REALTIME implementation.
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
-  clock_gettime(CLOCK_REALTIME, &ts);
-#else
-  clock_gettime(CLOCK_MONOTONIC, &ts);
-#endif
-  result.ticks_ = 1000000000LL * static_cast<int64_t>(ts.tv_sec) +
-      static_cast<int64_t>(ts.tv_nsec);
-#elif defined(WEBRTC_MAC)
-  static mach_timebase_info_data_t timebase;
-  if (timebase.denom == 0) {
-    // Get the timebase if this is the first time we run.
-    // Recommended by Apple's QA1398.
-    kern_return_t retval = mach_timebase_info(&timebase);
-    if (retval != KERN_SUCCESS) {
-      // TODO(wu): Implement CHECK similar to chrome for all the platforms.
-      // Then replace this with a CHECK(retval == KERN_SUCCESS);
-#ifndef WEBRTC_IOS
-      asm("int3");
-#else
-      __builtin_trap();
-#endif  // WEBRTC_IOS
-    }
-  }
-  // Use timebase to convert absolute time tick units into nanoseconds.
-  result.ticks_ = mach_absolute_time() * timebase.numer / timebase.denom;
-#else
-  struct timeval tv;
-  gettimeofday(&tv, NULL);
-  result.ticks_ = 1000000LL * static_cast<int64_t>(tv.tv_sec) +
-      static_cast<int64_t>(tv.tv_usec);
-#endif
-  return result.ticks_;
+  return rtc::TimeNanos();
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/timestamp_extrapolator.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/timestamp_extrapolator.cc
index f2b7092686..c7ed856a54 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/timestamp_extrapolator.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/timestamp_extrapolator.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
+#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
 
 #include <algorithm>
 
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.cc
index af6b3d5739..671435b389 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.cc
@@ -14,8 +14,9 @@
 #include <stdarg.h>
 #include <stdio.h>
 #include <string.h>
-#include "base/singleton.h"
 
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/platform_thread.h"
 #ifdef _WIN32
 #include "webrtc/system_wrappers/source/trace_win.h"
 #else
@@ -69,7 +70,6 @@ TraceImpl* TraceImpl::StaticInstance(CountOperation count_operation,
 #else
     GetStaticInstance<TracePosix>(count_operation);
 #endif
-
   return impl;
 }
 
@@ -77,14 +77,6 @@ TraceImpl* TraceImpl::GetTrace(const TraceLevel level) {
   return StaticInstance(kAddRefNoCreate, level);
 }
 
-TraceImpl* TraceImpl::CreateInstance() {
-#if defined(_WIN32)
-  return new TraceWindows();
-#else
-  return new TracePosix();
-#endif
-}
-
 TraceImpl::TraceImpl()
     : callback_(NULL),
       row_count_text_(0),
@@ -98,7 +90,7 @@ TraceImpl::~TraceImpl() {
 }
 
 int32_t TraceImpl::AddThreadId(char* trace_message) const {
-  uint32_t thread_id = ThreadWrapper::GetThreadId();
+  uint32_t thread_id = rtc::CurrentThreadId();
   // Messages is 12 characters.
   return sprintf(trace_message, "%10u; ", thread_id);
 }
@@ -423,7 +415,12 @@ void TraceImpl::WriteToFile(const char* msg, uint16_t length) {
       row_count_text_++;
     }
   }
-  trace_file_->Write(msg, length);
+
+  char trace_message[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
+  memcpy(trace_message, msg, length);
+  trace_message[length] = 0;
+  trace_message[length - 1] = '\n';
+  trace_file_->Write(trace_message, length);
   row_count_text_++;
 }
 
@@ -563,12 +560,12 @@ int32_t Trace::TraceFile(char file_name[FileWrapper::kMaxFileNameSize]) {
 
 // static
 void Trace::set_level_filter(int filter) {
-  rtc::AtomicOps::Store(&level_filter_, filter);
+  rtc::AtomicOps::ReleaseStore(&level_filter_, filter);
 }
 
 // static
 int Trace::level_filter() {
-  return rtc::AtomicOps::Load(&level_filter_);
+  return rtc::AtomicOps::AcquireLoad(&level_filter_);
 }
 
 // static
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.h b/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.h
index da5af72c48..a4135eafc7 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/trace_impl.h
@@ -13,11 +13,11 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/static_instance.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/static_instance.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -36,7 +36,6 @@ class TraceImpl : public Trace {
  public:
   virtual ~TraceImpl();
 
-  static TraceImpl* CreateInstance();
   static TraceImpl* GetTrace(const TraceLevel level = kTraceAll);
 
   int32_t SetTraceFileImpl(const char* file_name, const bool add_file_counter);
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/trace_posix.h b/media/webrtc/trunk/webrtc/system_wrappers/source/trace_posix.h
index 89420c650f..25dfeec079 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/source/trace_posix.h
+++ b/media/webrtc/trunk/webrtc/system_wrappers/source/trace_posix.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/source/trace_impl.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers.gyp b/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers.gyp
index fb76f34a59..700b673ef9 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers.gyp
+++ b/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers.gyp
@@ -17,41 +17,38 @@
         '../base/base.gyp:rtc_base_approved',
       ],
       'sources': [
-        'interface/aligned_array.h',
-        'interface/aligned_malloc.h',
-        'interface/atomic32.h',
-        'interface/clock.h',
-        'interface/condition_variable_wrapper.h',
-        'interface/cpu_info.h',
-        'interface/cpu_features_wrapper.h',
-        'interface/critical_section_wrapper.h',
-        'interface/data_log.h',
-        'interface/data_log_c.h',
-        'interface/data_log_impl.h',
-        'interface/event_tracer.h',
-        'interface/event_wrapper.h',
-        'interface/field_trial.h',
-        'interface/file_wrapper.h',
-        'interface/fix_interlocked_exchange_pointer_win.h',
-        'interface/logcat_trace_context.h',
-        'interface/logging.h',
-        'interface/metrics.h',
-        'interface/ref_count.h',
-        'interface/rtp_to_ntp.h',
-        'interface/rw_lock_wrapper.h',
-        'interface/scoped_refptr.h',
-        'interface/scoped_vector.h',
-        'interface/sleep.h',
-        'interface/sort.h',
-        'interface/static_instance.h',
-        'interface/stl_util.h',
-        'interface/stringize_macros.h',
-        'interface/thread_wrapper.h',
-        'interface/tick_util.h',
-        'interface/timestamp_extrapolator.h',
-        'interface/trace.h',
-        'interface/trace_event.h',
-        'interface/utf_util_win.h',
+        'include/aligned_array.h',
+        'include/aligned_malloc.h',
+        'include/atomic32.h',
+        'include/clock.h',
+        'include/condition_variable_wrapper.h',
+        'include/cpu_info.h',
+        'include/cpu_features_wrapper.h',
+        'include/critical_section_wrapper.h',
+        'include/data_log.h',
+        'include/data_log_c.h',
+        'include/data_log_impl.h',
+        'include/event_wrapper.h',
+        'include/field_trial.h',
+        'include/file_wrapper.h',
+        'include/fix_interlocked_exchange_pointer_win.h',
+        'include/logcat_trace_context.h',
+        'include/logging.h',
+        'include/metrics.h',
+        'include/ntp_time.h',
+        'include/ref_count.h',
+        'include/rtp_to_ntp.h',
+        'include/rw_lock_wrapper.h',
+        'include/scoped_vector.h',
+        'include/sleep.h',
+        'include/sort.h',
+        'include/static_instance.h',
+        'include/stl_util.h',
+        'include/stringize_macros.h',
+        'include/tick_util.h',
+        'include/timestamp_extrapolator.h',
+        'include/trace.h',
+        'include/utf_util_win.h',
         'source/aligned_malloc.cc',
         'source/atomic32_mac.cc',
         'source/atomic32_posix.cc',
@@ -75,11 +72,10 @@
         'source/data_log_c.cc',
         'source/data_log_no_op.cc',
         'source/event.cc',
-        'source/event_posix.cc',
-        'source/event_posix.h',
-        'source/event_tracer.cc',
-        'source/event_win.cc',
-        'source/event_win.h',
+        'source/event_timer_posix.cc',
+        'source/event_timer_posix.h',
+        'source/event_timer_win.cc',
+        'source/event_timer_win.h',
         'source/file_impl.cc',
         'source/file_impl.h',
         'source/logcat_trace_context.cc',
@@ -95,11 +91,6 @@
         'source/sleep.cc',
         'source/sort.cc',
         'source/tick_util.cc',
-        'source/thread.cc',
-        'source/thread_posix.cc',
-        'source/thread_posix.h',
-        'source/thread_win.cc',
-        'source/thread_win.h',
         'source/timestamp_extrapolator.cc',
         'source/trace_impl.cc',
         'source/trace_impl.h',
@@ -151,7 +142,7 @@
           },
         }, {  # OS!="android"
           'sources!': [
-            'interface/logcat_trace_context.h',
+            'include/logcat_trace_context.h',
             'source/logcat_trace_context.cc',
           ],
         }],
@@ -209,11 +200,8 @@
       'target_name': 'field_trial_default',
       'type': 'static_library',
       'sources': [
-        'interface/field_trial_default.h',
+        'include/field_trial_default.h',
         'source/field_trial_default.cc',
-      ],
-      'dependencies': [
-        'system_wrappers',
       ]
     }, {
       'target_name': 'metrics_default',
@@ -221,13 +209,11 @@
       'sources': [
         'source/metrics_default.cc',
       ],
-      'dependencies': [
-        'system_wrappers',
-      ]
     }, {
       'target_name': 'system_wrappers_default',
       'type': 'static_library',
       'dependencies': [
+        'system_wrappers',
         'field_trial_default',
         'metrics_default',
       ]
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers_tests.gyp b/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers_tests.gyp
index da2fe7432f..a0ae14d6cf 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers_tests.gyp
+++ b/media/webrtc/trunk/webrtc/system_wrappers/system_wrappers_tests.gyp
@@ -15,6 +15,7 @@
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/test/test.gyp:histogram',
         '<(webrtc_root)/test/test.gyp:test_support_main',
       ],
       'sources': [
@@ -23,19 +24,18 @@
         'source/clock_unittest.cc',
         'source/condition_variable_unittest.cc',
         'source/critical_section_unittest.cc',
-        'source/event_tracer_unittest.cc',
         'source/logging_unittest.cc',
         'source/data_log_unittest.cc',
         'source/data_log_unittest_disabled.cc',
         'source/data_log_helpers_unittest.cc',
         'source/data_log_c_helpers_unittest.c',
         'source/data_log_c_helpers_unittest.h',
+        'source/metrics_unittest.cc',
+        'source/ntp_time_unittest.cc',
         'source/rtp_to_ntp_unittest.cc',
         'source/scoped_vector_unittest.cc',
         'source/stringize_macros_unittest.cc',
         'source/stl_util_unittest.cc',
-        'source/thread_unittest.cc',
-        'source/thread_posix_unittest.cc',
       ],
       'conditions': [
         ['enable_data_logging==1', {
@@ -43,9 +43,6 @@
         }, {
           'sources!': [ 'source/data_log_unittest.cc', ],
         }],
-        ['os_posix==0', {
-          'sources!': [ 'source/thread_posix_unittest.cc', ],
-        }],
         ['OS=="android"', {
           'dependencies': [
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
diff --git a/media/webrtc/trunk/webrtc/system_wrappers/test/TestSort/TestSort.cc b/media/webrtc/trunk/webrtc/system_wrappers/test/TestSort/TestSort.cc
index cfa47cfb95..b2b9f85755 100644
--- a/media/webrtc/trunk/webrtc/system_wrappers/test/TestSort/TestSort.cc
+++ b/media/webrtc/trunk/webrtc/system_wrappers/test/TestSort/TestSort.cc
@@ -13,8 +13,8 @@
 
 #include <algorithm>
 
-#include "webrtc/system_wrappers/interface/sort.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/sort.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 // Excellent work polluting the global namespace Visual Studio...
 #undef max
diff --git a/media/webrtc/trunk/webrtc/test/BUILD.gn b/media/webrtc/trunk/webrtc/test/BUILD.gn
index 337dcf6db6..3ecd903522 100644
--- a/media/webrtc/trunk/webrtc/test/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/test/BUILD.gn
@@ -27,10 +27,11 @@ source_set("field_trial") {
   deps = [
     "..:webrtc_common",
     "../system_wrappers",
+    "../system_wrappers:field_trial_default",
   ]
 
   configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config"]
+  public_configs = [ "..:common_inherited_config" ]
 }
 
 source_set("histogram") {
@@ -45,7 +46,7 @@ source_set("histogram") {
   ]
 
   configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config"]
+  public_configs = [ "..:common_inherited_config" ]
 }
 
 source_set("test_support") {
@@ -58,7 +59,6 @@ source_set("test_support") {
     "testsupport/frame_reader.h",
     "testsupport/frame_writer.cc",
     "testsupport/frame_writer.h",
-    "testsupport/gtest_disable.h",
     "testsupport/mock/mock_frame_reader.h",
     "testsupport/mock/mock_frame_writer.h",
     "testsupport/packet_reader.cc",
@@ -70,10 +70,10 @@ source_set("test_support") {
   ]
 
   deps = [
-    "//testing/gmock",
-    "//testing/gtest",
     "..:gtest_prod",
     "../system_wrappers",
+    "//testing/gmock",
+    "//testing/gtest",
   ]
 
   if (is_android) {
@@ -81,7 +81,7 @@ source_set("test_support") {
   }
 
   configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config"]
+  public_configs = [ "..:common_inherited_config" ]
 }
 
 source_set("test_support_main") {
@@ -103,5 +103,5 @@ source_set("test_support_main") {
   ]
 
   configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config"]
+  public_configs = [ "..:common_inherited_config" ]
 }
diff --git a/media/webrtc/trunk/webrtc/test/call_test.cc b/media/webrtc/trunk/webrtc/test/call_test.cc
index 9e78e82d7f..850e487caf 100644
--- a/media/webrtc/trunk/webrtc/test/call_test.cc
+++ b/media/webrtc/trunk/webrtc/test/call_test.cc
@@ -7,8 +7,15 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#include "webrtc/base/checks.h"
+#include "webrtc/common.h"
+#include "webrtc/config.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/encoder_settings.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_network.h"
 
 namespace webrtc {
 namespace test {
@@ -19,49 +26,109 @@ const int kVideoRotationRtpExtensionId = 4;
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
-      send_stream_(NULL),
-      fake_encoder_(clock_) {
-}
+      video_send_config_(nullptr),
+      video_send_stream_(nullptr),
+      audio_send_config_(nullptr),
+      audio_send_stream_(nullptr),
+      fake_encoder_(clock_),
+      num_video_streams_(0),
+      num_audio_streams_(0),
+      fake_send_audio_device_(nullptr),
+      fake_recv_audio_device_(nullptr) {}
 
 CallTest::~CallTest() {
 }
 
 void CallTest::RunBaseTest(BaseTest* test) {
-  CreateSenderCall(test->GetSenderCallConfig());
-  if (test->ShouldCreateReceivers())
-    CreateReceiverCall(test->GetReceiverCallConfig());
+  num_video_streams_ = test->GetNumVideoStreams();
+  num_audio_streams_ = test->GetNumAudioStreams();
+  RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
+  Call::Config send_config(test->GetSenderCallConfig());
+  if (num_audio_streams_ > 0) {
+    CreateVoiceEngines();
+    AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = voe_send_.voice_engine;
+    send_config.audio_state = AudioState::Create(audio_state_config);
+  }
+  CreateSenderCall(send_config);
+  if (test->ShouldCreateReceivers()) {
+    Call::Config recv_config(test->GetReceiverCallConfig());
+    if (num_audio_streams_ > 0) {
+      AudioState::Config audio_state_config;
+      audio_state_config.voice_engine = voe_recv_.voice_engine;
+      recv_config.audio_state = AudioState::Create(audio_state_config);
+    }
+    CreateReceiverCall(recv_config);
+  }
+  send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
+  receive_transport_.reset(test->CreateReceiveTransport());
   test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
 
   if (test->ShouldCreateReceivers()) {
-    test->SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
+    send_transport_->SetReceiver(receiver_call_->Receiver());
+    receive_transport_->SetReceiver(sender_call_->Receiver());
   } else {
     // Sender-only call delivers to itself.
-    test->SetReceivers(sender_call_->Receiver(), NULL);
+    send_transport_->SetReceiver(sender_call_->Receiver());
+    receive_transport_->SetReceiver(nullptr);
   }
 
-  CreateSendConfig(test->GetNumStreams());
+  CreateSendConfig(num_video_streams_, num_audio_streams_,
+                   send_transport_.get());
   if (test->ShouldCreateReceivers()) {
-    CreateMatchingReceiveConfigs();
+    CreateMatchingReceiveConfigs(receive_transport_.get());
+  }
+  if (num_audio_streams_ > 0)
+    SetupVoiceEngineTransports(send_transport_.get(), receive_transport_.get());
+
+  if (num_video_streams_ > 0) {
+    test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
+                             &video_encoder_config_);
+  }
+  if (num_audio_streams_ > 0)
+    test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
+
+  if (num_video_streams_ > 0) {
+    CreateVideoStreams();
+    test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
+  }
+  if (num_audio_streams_ > 0) {
+    CreateAudioStreams();
+    test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
   }
-  test->ModifyConfigs(&send_config_, &receive_configs_, &encoder_config_);
-  CreateStreams();
-  test->OnStreamsCreated(send_stream_, receive_streams_);
 
   CreateFrameGeneratorCapturer();
   test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
 
   Start();
   test->PerformTest();
-  test->StopSending();
+  send_transport_->StopSending();
+  receive_transport_->StopSending();
   Stop();
 
   DestroyStreams();
+  DestroyCalls();
+  if (num_audio_streams_ > 0)
+    DestroyVoiceEngines();
 }
 
 void CallTest::Start() {
-  send_stream_->Start();
-  for (size_t i = 0; i < receive_streams_.size(); ++i)
-    receive_streams_[i]->Start();
+  if (video_send_stream_)
+    video_send_stream_->Start();
+  for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
+    video_recv_stream->Start();
+  if (audio_send_stream_) {
+    fake_send_audio_device_->Start();
+    audio_send_stream_->Start();
+    EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id));
+  }
+  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+    audio_recv_stream->Start();
+  if (!audio_receive_streams_.empty()) {
+    fake_recv_audio_device_->Start();
+    EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id));
+    EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id));
+  }
   if (frame_generator_capturer_.get() != NULL)
     frame_generator_capturer_->Start();
 }
@@ -69,9 +136,22 @@ void CallTest::Start() {
 void CallTest::Stop() {
   if (frame_generator_capturer_.get() != NULL)
     frame_generator_capturer_->Stop();
-  for (size_t i = 0; i < receive_streams_.size(); ++i)
-    receive_streams_[i]->Stop();
-  send_stream_->Stop();
+  if (!audio_receive_streams_.empty()) {
+    fake_recv_audio_device_->Stop();
+    EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id));
+    EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id));
+  }
+  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+    audio_recv_stream->Stop();
+  if (audio_send_stream_) {
+    fake_send_audio_device_->Stop();
+    EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id));
+    audio_send_stream_->Stop();
+  }
+  for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
+    video_recv_stream->Stop();
+  if (video_send_stream_)
+    video_send_stream_->Stop();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
@@ -88,117 +168,252 @@ void CallTest::CreateReceiverCall(const Call::Config& config) {
   receiver_call_.reset(Call::Create(config));
 }
 
-void CallTest::CreateSendConfig(size_t num_streams) {
-  assert(num_streams <= kNumSsrcs);
-  send_config_ = VideoSendStream::Config();
-  send_config_.encoder_settings.encoder = &fake_encoder_;
-  send_config_.encoder_settings.payload_name = "FAKE";
-  send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
-  encoder_config_.streams = test::CreateVideoStreams(num_streams);
-  for (size_t i = 0; i < num_streams; ++i)
-    send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
-  send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+void CallTest::DestroyCalls() {
+  sender_call_.reset();
+  receiver_call_.reset();
 }
 
-void CallTest::CreateMatchingReceiveConfigs() {
-  assert(!send_config_.rtp.ssrcs.empty());
-  assert(receive_configs_.empty());
-  assert(allocated_decoders_.empty());
-  VideoReceiveStream::Config config;
-  config.rtp.local_ssrc = kReceiverLocalSsrc;
-  for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) {
+void CallTest::CreateSendConfig(size_t num_video_streams,
+                                size_t num_audio_streams,
+                                Transport* send_transport) {
+  RTC_DCHECK(num_video_streams <= kNumSsrcs);
+  RTC_DCHECK_LE(num_audio_streams, 1u);
+  RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
+  video_send_config_ = VideoSendStream::Config(send_transport);
+  video_send_config_.encoder_settings.encoder = &fake_encoder_;
+  video_send_config_.encoder_settings.payload_name = "FAKE";
+  video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+  video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
+  for (size_t i = 0; i < num_video_streams; ++i)
+    video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
+  video_send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+
+  if (num_audio_streams > 0) {
+    audio_send_config_ = AudioSendStream::Config(send_transport);
+    audio_send_config_.voe_channel_id = voe_send_.channel_id;
+    audio_send_config_.rtp.ssrc = kAudioSendSsrc;
+  }
+}
+
+void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
+  RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
+  RTC_DCHECK(video_receive_configs_.empty());
+  RTC_DCHECK(allocated_decoders_.empty());
+  RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
+  VideoReceiveStream::Config video_config(rtcp_send_transport);
+  video_config.rtp.remb = true;
+  video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+  for (const RtpExtension& extension : video_send_config_.rtp.extensions)
+    video_config.rtp.extensions.push_back(extension);
+  for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
     VideoReceiveStream::Decoder decoder =
-        test::CreateMatchingDecoder(send_config_.encoder_settings);
+        test::CreateMatchingDecoder(video_send_config_.encoder_settings);
     allocated_decoders_.push_back(decoder.decoder);
-    config.decoders.clear();
-    config.decoders.push_back(decoder);
-    config.rtp.remote_ssrc = send_config_.rtp.ssrcs[i];
-    receive_configs_.push_back(config);
+    video_config.decoders.clear();
+    video_config.decoders.push_back(decoder);
+    video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
+    video_receive_configs_.push_back(video_config);
+  }
+
+  RTC_DCHECK(num_audio_streams_ <= 1);
+  if (num_audio_streams_ == 1) {
+    AudioReceiveStream::Config audio_config;
+    audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
+    audio_config.rtcp_send_transport = rtcp_send_transport;
+    audio_config.voe_channel_id = voe_recv_.channel_id;
+    audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
+    audio_receive_configs_.push_back(audio_config);
   }
 }
 
 void CallTest::CreateFrameGeneratorCapturer() {
-  VideoStream stream = encoder_config_.streams.back();
-  frame_generator_capturer_.reset(
-      test::FrameGeneratorCapturer::Create(send_stream_->Input(),
-                                           stream.width,
-                                           stream.height,
-                                           stream.max_framerate,
-                                           clock_));
+  VideoStream stream = video_encoder_config_.streams.back();
+  frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
+      video_send_stream_->Input(), stream.width, stream.height,
+      stream.max_framerate, clock_));
 }
-void CallTest::CreateStreams() {
-  assert(send_stream_ == NULL);
-  assert(receive_streams_.empty());
 
-  send_stream_ =
-      sender_call_->CreateVideoSendStream(send_config_, encoder_config_);
+void CallTest::CreateFakeAudioDevices() {
+  fake_send_audio_device_.reset(new FakeAudioDevice(
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
+  fake_recv_audio_device_.reset(new FakeAudioDevice(
+      clock_, test::ResourcePath("voice_engine/audio_long16", "pcm")));
+}
 
-  for (size_t i = 0; i < receive_configs_.size(); ++i) {
-    receive_streams_.push_back(
-        receiver_call_->CreateVideoReceiveStream(receive_configs_[i]));
+void CallTest::CreateVideoStreams() {
+  RTC_DCHECK(video_send_stream_ == nullptr);
+  RTC_DCHECK(video_receive_streams_.empty());
+  RTC_DCHECK(audio_send_stream_ == nullptr);
+  RTC_DCHECK(audio_receive_streams_.empty());
+
+  video_send_stream_ = sender_call_->CreateVideoSendStream(
+      video_send_config_, video_encoder_config_);
+  for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
+    video_receive_streams_.push_back(
+        receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
   }
 }
 
+void CallTest::CreateAudioStreams() {
+  audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
+  for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
+    audio_receive_streams_.push_back(
+        receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
+  }
+  CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
+  EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
+}
+
 void CallTest::DestroyStreams() {
-  if (send_stream_ != NULL)
-    sender_call_->DestroyVideoSendStream(send_stream_);
-  send_stream_ = NULL;
-  for (size_t i = 0; i < receive_streams_.size(); ++i)
-    receiver_call_->DestroyVideoReceiveStream(receive_streams_[i]);
-  receive_streams_.clear();
+  if (video_send_stream_)
+    sender_call_->DestroyVideoSendStream(video_send_stream_);
+  video_send_stream_ = nullptr;
+  for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
+    receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
+
+  if (audio_send_stream_)
+    sender_call_->DestroyAudioSendStream(audio_send_stream_);
+  audio_send_stream_ = nullptr;
+  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+    receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
+  video_receive_streams_.clear();
+
   allocated_decoders_.clear();
 }
 
-const unsigned int CallTest::kDefaultTimeoutMs = 30 * 1000;
-const unsigned int CallTest::kLongTimeoutMs = 120 * 1000;
-const uint8_t CallTest::kSendPayloadType = 100;
-const uint8_t CallTest::kFakeSendPayloadType = 125;
+void CallTest::CreateVoiceEngines() {
+  CreateFakeAudioDevices();
+  voe_send_.voice_engine = VoiceEngine::Create();
+  voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
+  voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine);
+  voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
+  EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
+  Config voe_config;
+  voe_config.Set<VoicePacing>(new VoicePacing(true));
+  voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
+  EXPECT_GE(voe_send_.channel_id, 0);
+
+  voe_recv_.voice_engine = VoiceEngine::Create();
+  voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
+  voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine);
+  voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
+  EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
+  voe_recv_.channel_id = voe_recv_.base->CreateChannel();
+  EXPECT_GE(voe_recv_.channel_id, 0);
+}
+
+void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport,
+                                          PacketTransport* recv_transport) {
+  voe_send_.transport_adapter.reset(
+      new internal::TransportAdapter(send_transport));
+  voe_send_.transport_adapter->Enable();
+  EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport(
+                   voe_send_.channel_id, *voe_send_.transport_adapter.get()));
+
+  voe_recv_.transport_adapter.reset(
+      new internal::TransportAdapter(recv_transport));
+  voe_recv_.transport_adapter->Enable();
+  EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport(
+                   voe_recv_.channel_id, *voe_recv_.transport_adapter.get()));
+}
+
+void CallTest::DestroyVoiceEngines() {
+  voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
+  voe_recv_.channel_id = -1;
+  voe_recv_.base->Release();
+  voe_recv_.base = nullptr;
+  voe_recv_.network->Release();
+  voe_recv_.network = nullptr;
+  voe_recv_.codec->Release();
+  voe_recv_.codec = nullptr;
+
+  voe_send_.base->DeleteChannel(voe_send_.channel_id);
+  voe_send_.channel_id = -1;
+  voe_send_.base->Release();
+  voe_send_.base = nullptr;
+  voe_send_.network->Release();
+  voe_send_.network = nullptr;
+  voe_send_.codec->Release();
+  voe_send_.codec = nullptr;
+
+  VoiceEngine::Delete(voe_send_.voice_engine);
+  voe_send_.voice_engine = nullptr;
+  VoiceEngine::Delete(voe_recv_.voice_engine);
+  voe_recv_.voice_engine = nullptr;
+}
+
+const int CallTest::kDefaultTimeoutMs = 30 * 1000;
+const int CallTest::kLongTimeoutMs = 120 * 1000;
+const uint8_t CallTest::kVideoSendPayloadType = 100;
+const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
 const uint8_t CallTest::kSendRtxPayloadType = 98;
 const uint8_t CallTest::kRedPayloadType = 118;
+const uint8_t CallTest::kRtxRedPayloadType = 99;
 const uint8_t CallTest::kUlpfecPayloadType = 119;
+const uint8_t CallTest::kAudioSendPayloadType = 103;
 const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
                                                      0xBADCAFF};
-const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF};
-const uint32_t CallTest::kReceiverLocalSsrc = 0x123456;
+const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
+                                                       0xC0FFEF};
+const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
+const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
+const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
 const int CallTest::kNackRtpHistoryMs = 1000;
 
 BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
 }
 
-BaseTest::BaseTest(unsigned int timeout_ms,
-                   const FakeNetworkPipe::Config& config)
-    : RtpRtcpObserver(timeout_ms, config) {
-}
-
 BaseTest::~BaseTest() {
 }
 
 Call::Config BaseTest::GetSenderCallConfig() {
-  return Call::Config(SendTransport());
+  return Call::Config();
 }
 
 Call::Config BaseTest::GetReceiverCallConfig() {
-  return Call::Config(ReceiveTransport());
+  return Call::Config();
 }
 
 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 }
 
-size_t BaseTest::GetNumStreams() const {
+test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
+  return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
+                             FakeNetworkPipe::Config());
+}
+
+test::PacketTransport* BaseTest::CreateReceiveTransport() {
+  return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
+                             FakeNetworkPipe::Config());
+}
+
+size_t BaseTest::GetNumVideoStreams() const {
   return 1;
 }
 
-void BaseTest::ModifyConfigs(
-    VideoSendStream::Config* send_config,
-    std::vector<VideoReceiveStream::Config>* receive_configs,
-    VideoEncoderConfig* encoder_config) {
+size_t BaseTest::GetNumAudioStreams() const {
+  return 0;
 }
 
-void BaseTest::OnStreamsCreated(
+void BaseTest::ModifyVideoConfigs(
+    VideoSendStream::Config* send_config,
+    std::vector<VideoReceiveStream::Config>* receive_configs,
+    VideoEncoderConfig* encoder_config) {}
+
+void BaseTest::OnVideoStreamsCreated(
     VideoSendStream* send_stream,
-    const std::vector<VideoReceiveStream*>& receive_streams) {
-}
+    const std::vector<VideoReceiveStream*>& receive_streams) {}
+
+void BaseTest::ModifyAudioConfigs(
+    AudioSendStream::Config* send_config,
+    std::vector<AudioReceiveStream::Config>* receive_configs) {}
+
+void BaseTest::OnAudioStreamsCreated(
+    AudioSendStream* send_stream,
+    const std::vector<AudioReceiveStream*>& receive_streams) {}
 
 void BaseTest::OnFrameGeneratorCapturerCreated(
     FrameGeneratorCapturer* frame_generator_capturer) {
@@ -207,11 +422,6 @@ void BaseTest::OnFrameGeneratorCapturerCreated(
 SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
-SendTest::SendTest(unsigned int timeout_ms,
-                   const FakeNetworkPipe::Config& config)
-    : BaseTest(timeout_ms, config) {
-}
-
 bool SendTest::ShouldCreateReceivers() const {
   return false;
 }
@@ -219,11 +429,6 @@ bool SendTest::ShouldCreateReceivers() const {
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
-EndToEndTest::EndToEndTest(unsigned int timeout_ms,
-                           const FakeNetworkPipe::Config& config)
-    : BaseTest(timeout_ms, config) {
-}
-
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
diff --git a/media/webrtc/trunk/webrtc/test/call_test.h b/media/webrtc/trunk/webrtc/test/call_test.h
index 4771dee59d..251d7f6044 100644
--- a/media/webrtc/trunk/webrtc/test/call_test.h
+++ b/media/webrtc/trunk/webrtc/test/call_test.h
@@ -7,19 +7,26 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_TEST_COMMON_CALL_TEST_H_
-#define WEBRTC_TEST_COMMON_CALL_TEST_H_
+#ifndef WEBRTC_TEST_CALL_TEST_H_
+#define WEBRTC_TEST_CALL_TEST_H_
 
 #include <vector>
 
 #include "webrtc/call.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
+#include "webrtc/call/transport_adapter.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
+#include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 
 namespace webrtc {
+
+class VoEBase;
+class VoECodec;
+class VoENetwork;
+
 namespace test {
 
 class BaseTest;
@@ -27,36 +34,48 @@ class BaseTest;
 class CallTest : public ::testing::Test {
  public:
   CallTest();
-  ~CallTest();
+  virtual ~CallTest();
 
   static const size_t kNumSsrcs = 3;
 
-  static const unsigned int kDefaultTimeoutMs;
-  static const unsigned int kLongTimeoutMs;
-  static const uint8_t kSendPayloadType;
+  static const int kDefaultTimeoutMs;
+  static const int kLongTimeoutMs;
+  static const uint8_t kVideoSendPayloadType;
   static const uint8_t kSendRtxPayloadType;
-  static const uint8_t kFakeSendPayloadType;
+  static const uint8_t kFakeVideoSendPayloadType;
   static const uint8_t kRedPayloadType;
+  static const uint8_t kRtxRedPayloadType;
   static const uint8_t kUlpfecPayloadType;
+  static const uint8_t kAudioSendPayloadType;
   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
-  static const uint32_t kSendSsrcs[kNumSsrcs];
-  static const uint32_t kReceiverLocalSsrc;
+  static const uint32_t kVideoSendSsrcs[kNumSsrcs];
+  static const uint32_t kAudioSendSsrc;
+  static const uint32_t kReceiverLocalVideoSsrc;
+  static const uint32_t kReceiverLocalAudioSsrc;
   static const int kNackRtpHistoryMs;
 
  protected:
+  // RunBaseTest overwrites the audio_state and the voice_engine of the send and
+  // receive Call configs to simplify test code and avoid having old VoiceEngine
+  // APIs in the tests.
   void RunBaseTest(BaseTest* test);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
+  void DestroyCalls();
 
-  void CreateSendConfig(size_t num_streams);
-  void CreateMatchingReceiveConfigs();
+  void CreateSendConfig(size_t num_video_streams,
+                        size_t num_audio_streams,
+                        Transport* send_transport);
+  void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturer();
+  void CreateFakeAudioDevices();
 
-  void CreateStreams();
+  void CreateVideoStreams();
+  void CreateAudioStreams();
   void Start();
   void Stop();
   void DestroyStreams();
@@ -64,42 +83,93 @@ class CallTest : public ::testing::Test {
   Clock* const clock_;
 
   rtc::scoped_ptr<Call> sender_call_;
-  VideoSendStream::Config send_config_;
-  VideoEncoderConfig encoder_config_;
-  VideoSendStream* send_stream_;
+  rtc::scoped_ptr<PacketTransport> send_transport_;
+  VideoSendStream::Config video_send_config_;
+  VideoEncoderConfig video_encoder_config_;
+  VideoSendStream* video_send_stream_;
+  AudioSendStream::Config audio_send_config_;
+  AudioSendStream* audio_send_stream_;
 
   rtc::scoped_ptr<Call> receiver_call_;
-  std::vector<VideoReceiveStream::Config> receive_configs_;
-  std::vector<VideoReceiveStream*> receive_streams_;
+  rtc::scoped_ptr<PacketTransport> receive_transport_;
+  std::vector<VideoReceiveStream::Config> video_receive_configs_;
+  std::vector<VideoReceiveStream*> video_receive_streams_;
+  std::vector<AudioReceiveStream::Config> audio_receive_configs_;
+  std::vector<AudioReceiveStream*> audio_receive_streams_;
 
   rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
   ScopedVector<VideoDecoder> allocated_decoders_;
+  size_t num_video_streams_;
+  size_t num_audio_streams_;
+
+ private:
+  // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
+  // These methods are used to set up legacy voice engines and channels which is
+  // necessary while voice engine is being refactored to the new stream API.
+  struct VoiceEngineState {
+    VoiceEngineState()
+        : voice_engine(nullptr),
+          base(nullptr),
+          network(nullptr),
+          codec(nullptr),
+          channel_id(-1),
+          transport_adapter(nullptr) {}
+
+    VoiceEngine* voice_engine;
+    VoEBase* base;
+    VoENetwork* network;
+    VoECodec* codec;
+    int channel_id;
+    rtc::scoped_ptr<internal::TransportAdapter> transport_adapter;
+  };
+
+  void CreateVoiceEngines();
+  void SetupVoiceEngineTransports(PacketTransport* send_transport,
+                                  PacketTransport* recv_transport);
+  void DestroyVoiceEngines();
+
+  VoiceEngineState voe_send_;
+  VoiceEngineState voe_recv_;
+
+  // The audio devices must outlive the voice engines.
+  rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_;
+  rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
 };
 
 class BaseTest : public RtpRtcpObserver {
  public:
   explicit BaseTest(unsigned int timeout_ms);
-  BaseTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
   virtual ~BaseTest();
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
-  virtual size_t GetNumStreams() const;
+  virtual size_t GetNumVideoStreams() const;
+  virtual size_t GetNumAudioStreams() const;
 
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
-  virtual void ModifyConfigs(
+  virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
+  virtual test::PacketTransport* CreateReceiveTransport();
+
+  virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
-  virtual void OnStreamsCreated(
+  virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);
 
+  virtual void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStream::Config>* receive_configs);
+  virtual void OnAudioStreamsCreated(
+      AudioSendStream* send_stream,
+      const std::vector<AudioReceiveStream*>& receive_streams);
+
   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);
 };
@@ -107,7 +177,6 @@ class BaseTest : public RtpRtcpObserver {
 class SendTest : public BaseTest {
  public:
   explicit SendTest(unsigned int timeout_ms);
-  SendTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
 
   bool ShouldCreateReceivers() const override;
 };
@@ -115,7 +184,6 @@ class SendTest : public BaseTest {
 class EndToEndTest : public BaseTest {
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
-  EndToEndTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
 
   bool ShouldCreateReceivers() const override;
 };
@@ -123,4 +191,4 @@ class EndToEndTest : public BaseTest {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_COMMON_CALL_TEST_H_
+#endif  // WEBRTC_TEST_CALL_TEST_H_
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.cc b/media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.cc
index 725a090641..38eefe54a2 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 
 #include <stdio.h>
 
@@ -16,8 +16,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #endif
 #include "webrtc/test/channel_transport/udp_transport.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
@@ -67,10 +65,11 @@ void VoiceChannelTransport::IncomingRTCPPacket(
 }
 
 int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) {
+  static const int kNumReceiveSocketBuffers = 500;
   int return_value = socket_transport_->InitializeReceiveSockets(this,
                                                                  rtp_port);
   if (return_value == 0) {
-    return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers);
+    return socket_transport_->StartReceiving(kNumReceiveSocketBuffers);
   }
   return return_value;
 }
@@ -80,58 +79,5 @@ int VoiceChannelTransport::SetSendDestination(const char* ip_address,
   return socket_transport_->InitializeSendSockets(ip_address, rtp_port);
 }
 
-
-VideoChannelTransport::VideoChannelTransport(ViENetwork* vie_network,
-                                             int channel)
-    : channel_(channel),
-      vie_network_(vie_network) {
-  uint8_t socket_threads = 1;
-  socket_transport_ = UdpTransport::Create(channel, socket_threads);
-  int registered = vie_network_->RegisterSendTransport(channel,
-                                                       *socket_transport_);
-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
-  EXPECT_EQ(0, registered);
-#else
-  assert(registered == 0);
-#endif
-}
-
-VideoChannelTransport::~VideoChannelTransport() {
-  vie_network_->DeregisterSendTransport(channel_);
-  UdpTransport::Destroy(socket_transport_);
-}
-
-void VideoChannelTransport::IncomingRTPPacket(
-    const int8_t* incoming_rtp_packet,
-    const size_t packet_length,
-    const char* /*from_ip*/,
-    const uint16_t /*from_port*/) {
-  vie_network_->ReceivedRTPPacket(
-      channel_, incoming_rtp_packet, packet_length, PacketTime());
-}
-
-void VideoChannelTransport::IncomingRTCPPacket(
-    const int8_t* incoming_rtcp_packet,
-    const size_t packet_length,
-    const char* /*from_ip*/,
-    const uint16_t /*from_port*/) {
-  vie_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet,
-                                   packet_length);
-}
-
-int VideoChannelTransport::SetLocalReceiver(uint16_t rtp_port) {
-  int return_value = socket_transport_->InitializeReceiveSockets(this,
-                                                                 rtp_port);
-  if (return_value == 0) {
-    return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers);
-  }
-  return return_value;
-}
-
-int VideoChannelTransport::SetSendDestination(const char* ip_address,
-                                              uint16_t rtp_port) {
-  return socket_transport_->InitializeSendSockets(ip_address, rtp_port);
-}
-
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/include/channel_transport.h b/media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.h
similarity index 55%
rename from media/webrtc/trunk/webrtc/test/channel_transport/include/channel_transport.h
rename to media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.h
index 77107d94d5..bab7c59181 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/include/channel_transport.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.h
@@ -8,14 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TEST_CHANNEL_TRANSPORT_INCLUDE_CHANNEL_TRANSPORT_H_
-#define WEBRTC_TEST_CHANNEL_TRANSPORT_INCLUDE_CHANNEL_TRANSPORT_H_
+#ifndef WEBRTC_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_
+#define WEBRTC_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_
 
 #include "webrtc/test/channel_transport/udp_transport.h"
 
 namespace webrtc {
 
-class ViENetwork;
 class VoENetwork;
 
 namespace test {
@@ -51,38 +50,7 @@ class VoiceChannelTransport : public UdpTransportData {
   UdpTransport* socket_transport_;
 };
 
-// Helper class for VideoEngine tests.
-class VideoChannelTransport : public UdpTransportData {
- public:
-  VideoChannelTransport(ViENetwork* vie_network, int channel);
-
-  virtual  ~VideoChannelTransport();
-
-  // Start implementation of UdpTransportData.
-  void IncomingRTPPacket(const int8_t* incoming_rtp_packet,
-                         const size_t packet_length,
-                         const char* /*from_ip*/,
-                         const uint16_t /*from_port*/) override;
-
-  void IncomingRTCPPacket(const int8_t* incoming_rtcp_packet,
-                          const size_t packet_length,
-                          const char* /*from_ip*/,
-                          const uint16_t /*from_port*/) override;
-  // End implementation of UdpTransportData.
-
-  // Specifies the ports to receive RTP packets on.
-  int SetLocalReceiver(uint16_t rtp_port);
-
-  // Specifies the destination port and IP address for a specified channel.
-  int SetSendDestination(const char* ip_address, uint16_t rtp_port);
-
- private:
-  int channel_;
-  ViENetwork* vie_network_;
-  UdpTransport* socket_transport_;
-};
-
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_TEST_CHANNEL_TRANSPORT_INCLUDE_CHANNEL_TRANSPORT_H_
+#endif  // WEBRTC_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc b/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc
index 9aeda75b03..3584359758 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc
@@ -12,7 +12,7 @@
 
 #include <assert.h>
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace test {
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h b/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h
index acd625ffdc..1197b94b20 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h
@@ -23,7 +23,7 @@
 #include <ntddndis.h>
 #include <traffic.h>
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace test {
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc
index 55ddaee068..9f40350287 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc
@@ -13,7 +13,7 @@
 #include <assert.h>
 #include <stdio.h>
 
-#include "webrtc/system_wrappers/interface/aligned_malloc.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
 #include "webrtc/test/channel_transport/udp_socket2_win.h"
 
 namespace webrtc {
@@ -520,8 +520,8 @@ int32_t UdpSocket2WorkerWindows::_numOfWorkers = 0;
 
 UdpSocket2WorkerWindows::UdpSocket2WorkerWindows(HANDLE ioCompletionHandle)
     : _ioCompletionHandle(ioCompletionHandle),
-      _init(false)
-{
+      _pThread(Run, this, "UdpSocket2ManagerWindows_thread"),
+      _init(false) {
     _workerNumber = _numOfWorkers++;
     WEBRTC_TRACE(kTraceMemory,  kTraceTransport, -1,
                  "UdpSocket2WorkerWindows created");
@@ -537,10 +537,9 @@ bool UdpSocket2WorkerWindows::Start()
 {
     WEBRTC_TRACE(kTraceStateInfo,  kTraceTransport, -1,
                  "Start UdpSocket2WorkerWindows");
-    if (!_pThread->Start())
-        return false;
+    _pThread.Start();
 
-    _pThread->SetPriority(kRealtimePriority);
+    _pThread.SetPriority(rtc::kRealtimePriority);
     return true;
 }
 
@@ -548,18 +547,14 @@ bool UdpSocket2WorkerWindows::Stop()
 {
     WEBRTC_TRACE(kTraceStateInfo,  kTraceTransport, -1,
                  "Stop UdpSocket2WorkerWindows");
-    return _pThread->Stop();
+    _pThread.Stop();
+    return true;
 }
 
 int32_t UdpSocket2WorkerWindows::Init()
 {
-    if(!_init)
-    {
-        const char* threadName = "UdpSocket2ManagerWindows_thread";
-        _pThread = ThreadWrapper::CreateThread(Run, this, threadName);
-        _init = true;
-    }
-    return 0;
+  _init = true;
+  return 0;
 }
 
 bool UdpSocket2WorkerWindows::Run(void* obj)
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h
index 7e4e805d0c..e762dccd0d 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h
@@ -14,10 +14,10 @@
 #include <winsock2.h>
 #include <list>
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/test/channel_transport/udp_socket2_win.h"
 #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h"
 #include "webrtc/test/channel_transport/udp_transport.h"
@@ -47,7 +47,7 @@ struct PerIoContext {
     int fromLen;
     // Should be set to true if the I/O context was passed to the system by
     // a thread not controlled by the socket implementation.
-    bool ioInitiatedByThreadWrapper;
+    bool ioInitiatedByPlatformThread;
     // TODO (hellner): Not used. Delete it.
     PerIoContext* pNextFree;
 };
@@ -105,7 +105,7 @@ protected:
     bool Process();
 private:
     HANDLE _ioCompletionHandle;
-    rtc::scoped_ptr<ThreadWrapper> _pThread;
+    rtc::PlatformThread _pThread;
     static int32_t _numOfWorkers;
     int32_t _workerNumber;
     volatile bool _stop;
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc
index 6bcb551012..adeb46a9d2 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc
@@ -15,7 +15,7 @@
 #include <winsock2.h>
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/channel_transport/traffic_control_win.h"
 #include "webrtc/test/channel_transport/udp_socket2_manager_win.h"
 
@@ -432,13 +432,13 @@ void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext,
     if(pIOContext == NULL || error == ERROR_OPERATION_ABORTED)
     {
         if ((pIOContext != NULL) &&
-            !pIOContext->ioInitiatedByThreadWrapper &&
+            !pIOContext->ioInitiatedByPlatformThread &&
             (error == ERROR_OPERATION_ABORTED) &&
             (pIOContext->ioOperation == OP_READ) &&
             _outstandingCallsDisabled)
         {
-            // !pIOContext->initiatedIOByThreadWrapper indicate that the I/O
-            // was not initiated by a ThreadWrapper thread.
+            // !pIOContext->initiatedIOByPlatformThread indicate that the I/O
+            // was not initiated by a PlatformThread thread.
             // This may happen if the thread that initiated receiving (e.g.
             // by calling StartListen())) is deleted before any packets have
             // been received.
@@ -519,7 +519,7 @@ void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext,
         {
             // The PerIoContext was posted by a thread controlled by the socket
             // implementation.
-            pIOContext->ioInitiatedByThreadWrapper = true;
+            pIOContext->ioInitiatedByPlatformThread = true;
         }
         OutstandingCallCompleted();
         return;
@@ -546,7 +546,7 @@ int32_t UdpSocket2Windows::PostRecv()
     }
     // This function may have been called by thread not controlled by the socket
     // implementation.
-    pIoContext->ioInitiatedByThreadWrapper = false;
+    pIoContext->ioInitiatedByPlatformThread = false;
     return PostRecv(pIoContext);
 }
 
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.h
index 5d6b212b93..ea37ac47c5 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.h
@@ -19,12 +19,12 @@
 #include <Ntddndis.h>
 #include <traffic.h>
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_socket2_manager_win.h"
 #include "webrtc/test/channel_transport/udp_socket_wrapper.h"
 
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc
index 9748400f11..6b1a466bf2 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc
@@ -17,8 +17,8 @@
 #include <time.h>
 #include <unistd.h>
 
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_socket_posix.h"
 
 namespace webrtc {
@@ -184,12 +184,11 @@ bool UdpSocketManagerPosix::RemoveSocket(UdpSocketWrapper* s)
     return retVal;
 }
 
-
 UdpSocketManagerPosixImpl::UdpSocketManagerPosixImpl()
-{
-    _critSectList = CriticalSectionWrapper::CreateCriticalSection();
-    _thread = ThreadWrapper::CreateThread(UdpSocketManagerPosixImpl::Run, this,
-                                          "UdpSocketManagerPosixImplThread");
+    : _thread(UdpSocketManagerPosixImpl::Run,
+              this,
+              "UdpSocketManagerPosixImplThread"),
+      _critSectList(CriticalSectionWrapper::CreateCriticalSection()) {
     FD_ZERO(&_readFds);
     WEBRTC_TRACE(kTraceMemory,  kTraceTransport, -1,
                  "UdpSocketManagerPosix created");
@@ -220,29 +219,19 @@ UdpSocketManagerPosixImpl::~UdpSocketManagerPosixImpl()
 
 bool UdpSocketManagerPosixImpl::Start()
 {
-    if (!_thread)
-    {
-        return false;
-    }
-
     WEBRTC_TRACE(kTraceStateInfo,  kTraceTransport, -1,
                  "Start UdpSocketManagerPosix");
-    if (!_thread->Start())
-        return false;
-    _thread->SetPriority(kRealtimePriority);
+    _thread.Start();
+    _thread.SetPriority(rtc::kRealtimePriority);
     return true;
 }
 
 bool UdpSocketManagerPosixImpl::Stop()
 {
-    if (!_thread)
-    {
-        return true;
-    }
-
     WEBRTC_TRACE(kTraceStateInfo,  kTraceTransport, -1,
                  "Stop UdpSocketManagerPosix");
-    return _thread->Stop();
+    _thread.Stop();
+    return true;
 }
 
 bool UdpSocketManagerPosixImpl::Process()
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.h
index e1fad9cf3d..45e55af99a 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.h
@@ -17,8 +17,8 @@
 #include <list>
 #include <map>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h"
 #include "webrtc/test/channel_transport/udp_socket_wrapper.h"
 
@@ -75,7 +75,7 @@ protected:
 private:
     typedef std::list<UdpSocketWrapper*> SocketList;
     typedef std::list<SOCKET> FdList;
-    rtc::scoped_ptr<ThreadWrapper> _thread;
+    rtc::PlatformThread _thread;
     CriticalSectionWrapper* _critSectList;
 
     fd_set _readFds;
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_unittest.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_unittest.cc
index d8e66b9395..b49021bffe 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_unittest.cc
@@ -16,7 +16,7 @@
 // The most important property of these tests is that they do not leak memory.
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h"
 #include "webrtc/test/channel_transport/udp_socket_wrapper.h"
 
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.cc
index 2e84c28c84..3127767cbc 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.cc
@@ -13,7 +13,7 @@
 #include <assert.h>
 
 #ifdef _WIN32
-#include "webrtc/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h"
+#include "webrtc/system_wrappers/include/fix_interlocked_exchange_pointer_win.h"
 #include "webrtc/test/channel_transport/udp_socket2_manager_win.h"
 #else
 #include "webrtc/test/channel_transport/udp_socket_manager_posix.h"
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.h
index 123db42983..0c3c3850d9 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_TEST_CHANNEL_TRANSPORT_UDP_SOCKET_MANAGER_WRAPPER_H_
 #define WEBRTC_TEST_CHANNEL_TRANSPORT_UDP_SOCKET_MANAGER_WRAPPER_H_
 
-#include "webrtc/system_wrappers/interface/static_instance.h"
+#include "webrtc/system_wrappers/include/static_instance.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc
index a2c84e5651..639d444f55 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc
@@ -20,7 +20,7 @@
 #include <time.h>
 #include <unistd.h>
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h"
 #include "webrtc/test/channel_transport/udp_socket_wrapper.h"
 
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.h
index 6ddf7e5408..c391b2e397 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.h
@@ -16,8 +16,8 @@
 #include <sys/socket.h>
 #include <sys/types.h>
 
-#include "webrtc/system_wrappers/interface/condition_variable_wrapper.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/condition_variable_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/test/channel_transport/udp_socket_wrapper.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc
index 7dad0cffea..f4fa3e950b 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc
@@ -13,8 +13,8 @@
 #include <stdlib.h>
 #include <string.h>
 
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h"
 
 #if defined(_WIN32)
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport.h
index a923835685..0c5079e69f 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport.h
@@ -12,6 +12,7 @@
 #define WEBRTC_TEST_CHANNEL_TRANSPORT_UDP_TRANSPORT_H_
 
 #include "webrtc/common_types.h"
+#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 /*
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc b/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc
index bae6b4cacf..eb33740a8e 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc
@@ -24,32 +24,30 @@
 #include <fcntl.h>
 #include <net/if.h>
 #include <netdb.h>
-#include <sys/socket.h>
 #include <netinet/in.h>
 #include <stdlib.h>
 #include <sys/ioctl.h>
+#include <sys/socket.h>
 #include <sys/time.h>
 #include <unistd.h>
 #ifndef WEBRTC_IOS
 #include <net/if_arp.h>
 #endif
-#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
 
 #if defined(WEBRTC_MAC)
+#include <ifaddrs.h>
 #include <machine/types.h>
 #endif
-#if defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
-#include <ifaddrs.h>
-#endif
-#if defined(WEBRTC_LINUX)
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
 #include <linux/netlink.h>
 #include <linux/rtnetlink.h>
 #endif
 
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h"
 #include "webrtc/typedefs.h"
 
@@ -1001,7 +999,7 @@ int32_t UdpTransportImpl::SetPCP(int32_t PCP)
         return -1;
     }
 
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     if (!rtpSock->SetSockopt(SOL_SOCKET, SO_PRIORITY, (int8_t*) &PCP,
                              sizeof(PCP)))
     {
@@ -1933,21 +1931,20 @@ int32_t UdpTransportImpl::SendRTCPPacketTo(const int8_t* data,
     return -1;
 }
 
-int UdpTransportImpl::SendPacket(int /*channel*/,
-                                 const void* data,
-                                 size_t length)
-{
+bool UdpTransportImpl::SendRtp(const uint8_t* data,
+                               size_t length,
+                               const PacketOptions& packet_options) {
     WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
 
     CriticalSectionScoped cs(_crit);
 
     if(_destIP[0] == 0)
     {
-        return -1;
+        return false;
     }
     if(_destPort == 0)
     {
-        return -1;
+        return false;
     }
 
     // Create socket if it hasn't been set up already.
@@ -1985,35 +1982,32 @@ int UdpTransportImpl::SendPacket(int /*channel*/,
                          "SendPacket() failed to bind RTP socket");
             _lastError = retVal;
             CloseReceiveSockets();
-            return -1;
+            return false;
         }
     }
 
     if(_ptrSendRtpSocket)
     {
         return _ptrSendRtpSocket->SendTo((const int8_t*)data, length,
-                                         _remoteRTPAddr);
+                                         _remoteRTPAddr) >= 0;
 
     } else if(_ptrRtpSocket)
     {
         return _ptrRtpSocket->SendTo((const int8_t*)data, length,
-                                     _remoteRTPAddr);
+                                     _remoteRTPAddr) >= 0;
     }
-    return -1;
+    return false;
 }
 
-int UdpTransportImpl::SendRTCPPacket(int /*channel*/, const void* data,
-                                     size_t length)
-{
-
+bool UdpTransportImpl::SendRtcp(const uint8_t* data, size_t length) {
     CriticalSectionScoped cs(_crit);
     if(_destIP[0] == 0)
     {
-        return -1;
+        return false;
     }
     if(_destPortRTCP == 0)
     {
-        return -1;
+        return false;
     }
 
     // Create socket if it hasn't been set up already.
@@ -2049,22 +2043,22 @@ int UdpTransportImpl::SendRTCPPacket(int /*channel*/, const void* data,
         {
             _lastError = retVal;
             WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
-                         "SendRTCPPacket() failed to bind RTCP socket");
+                         "SendRtcp() failed to bind RTCP socket");
             CloseReceiveSockets();
-            return -1;
+            return false;
         }
     }
 
     if(_ptrSendRtcpSocket)
     {
         return _ptrSendRtcpSocket->SendTo((const int8_t*)data, length,
-                                          _remoteRTCPAddr);
+                                          _remoteRTCPAddr) >= 0;
     } else if(_ptrRtcpSocket)
     {
         return _ptrRtcpSocket->SendTo((const int8_t*)data, length,
-                                      _remoteRTCPAddr);
+                                      _remoteRTCPAddr) >= 0;
     }
-    return -1;
+    return false;
 }
 
 int32_t UdpTransportImpl::SetSendIP(const char* ipaddr)
@@ -2485,7 +2479,7 @@ int32_t UdpTransport::LocalHostAddressIPV6(char n_localIP[16])
     return -1;
 #elif defined(WEBRTC_ANDROID)
     return -1;
-#else // WEBRTC_LINUX
+#else // WEBRTC_LINUX || WEBRTC_BSD
     struct
     {
         struct nlmsghdr n;
@@ -2671,7 +2665,7 @@ int32_t UdpTransport::LocalHostAddress(uint32_t& localIP)
     }
     WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1, "gethostname failed");
     return -1;
-#else // WEBRTC_LINUX
+#else // WEBRTC_LINUX || WEBRTC_BSD
     int sockfd, size  = 1;
     struct ifreq* ifr;
     struct ifconf ifc;
diff --git a/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.h b/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.h
index 5dbf5d8913..f80ee02d71 100644
--- a/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.h
+++ b/media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.h
@@ -116,8 +116,10 @@ public:
                              size_t length,
                              uint16_t rtcpPort) override;
     // Transport functions
-    int SendPacket(int channel, const void* data, size_t length) override;
-    int SendRTCPPacket(int channel, const void* data, size_t length) override;
+    bool SendRtp(const uint8_t* data,
+                 size_t length,
+                 const PacketOptions& packet_options) override;
+    bool SendRtcp(const uint8_t* data, size_t length) override;
 
     // UdpTransport functions continue.
     int32_t SetSendIP(const char* ipaddr) override;
diff --git a/media/webrtc/trunk/webrtc/test/common_unittest.cc b/media/webrtc/trunk/webrtc/test/common_unittest.cc
index 082c18c2c7..a239dade73 100644
--- a/media/webrtc/trunk/webrtc/test/common_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/common_unittest.cc
@@ -15,6 +15,7 @@ namespace webrtc {
 namespace {
 
 struct MyExperiment {
+  static const ConfigOptionID identifier = ConfigOptionID::kMyExperimentForTest;
   static const int kDefaultFactor;
   static const int kDefaultOffset;
 
@@ -56,6 +57,8 @@ TEST(Config, SetNullSetsTheOptionBackToDefault) {
 }
 
 struct Algo1_CostFunction {
+  static const ConfigOptionID identifier =
+      ConfigOptionID::kAlgo1CostFunctionForTest;
   Algo1_CostFunction() {}
 
   virtual int cost(int x) const {
diff --git a/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.cc b/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.cc
index 7d13be6897..831e481bd8 100644
--- a/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.cc
+++ b/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.cc
@@ -14,8 +14,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/common_video/include/video_image.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 namespace test {
@@ -39,15 +39,15 @@ int32_t ConfigurableFrameSizeEncoder::InitEncode(
 }
 
 int32_t ConfigurableFrameSizeEncoder::Encode(
-    const I420VideoFrame& inputImage,
+    const VideoFrame& inputImage,
     const CodecSpecificInfo* codecSpecificInfo,
-    const std::vector<VideoFrameType>* frame_types) {
+    const std::vector<FrameType>* frame_types) {
   EncodedImage encodedImage(
       buffer_.get(), current_frame_size_, max_frame_size_);
   encodedImage._completeFrame = true;
   encodedImage._encodedHeight = inputImage.height();
   encodedImage._encodedWidth = inputImage.width();
-  encodedImage._frameType = kKeyFrame;
+  encodedImage._frameType = kVideoFrameKey;
   encodedImage._timeStamp = inputImage.timestamp();
   encodedImage.capture_time_ms_ = inputImage.render_time_ms();
   RTPFragmentationHeader* fragmentation = NULL;
@@ -82,11 +82,6 @@ int32_t ConfigurableFrameSizeEncoder::SetPeriodicKeyFrames(bool enable) {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int32_t ConfigurableFrameSizeEncoder::CodecConfigParameters(uint8_t* buffer,
-                                                            int32_t size) {
-  return WEBRTC_VIDEO_CODEC_OK;
-}
-
 int32_t ConfigurableFrameSizeEncoder::SetFrameSize(size_t size) {
   assert(size <= max_frame_size_);
   current_frame_size_ = size;
diff --git a/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.h b/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.h
index eed44982f9..3794e8db08 100644
--- a/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.h
+++ b/media/webrtc/trunk/webrtc/test/configurable_frame_size_encoder.h
@@ -28,9 +28,9 @@ class ConfigurableFrameSizeEncoder : public VideoEncoder {
                      int32_t number_of_cores,
                      size_t max_payload_size) override;
 
-  int32_t Encode(const I420VideoFrame& input_image,
+  int32_t Encode(const VideoFrame& input_image,
                  const CodecSpecificInfo* codec_specific_info,
-                 const std::vector<VideoFrameType>* frame_types) override;
+                 const std::vector<FrameType>* frame_types) override;
 
   int32_t RegisterEncodeCompleteCallback(
       EncodedImageCallback* callback) override;
@@ -43,8 +43,6 @@ class ConfigurableFrameSizeEncoder : public VideoEncoder {
 
   int32_t SetPeriodicKeyFrames(bool enable) override;
 
-  int32_t CodecConfigParameters(uint8_t* buffer, int32_t size) override;
-
   int32_t SetFrameSize(size_t size);
 
  private:
diff --git a/media/webrtc/trunk/webrtc/test/constants.cc b/media/webrtc/trunk/webrtc/test/constants.cc
new file mode 100644
index 0000000000..7e94fe58fb
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/constants.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/test/constants.h"
+
+namespace webrtc {
+namespace test {
+
+const int kTOffsetExtensionId = 6;
+const int kAbsSendTimeExtensionId = 7;
+const int kTransportSequenceNumberExtensionId = 8;
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/ensure_initialized.h b/media/webrtc/trunk/webrtc/test/constants.h
similarity index 57%
rename from media/webrtc/trunk/webrtc/modules/video_capture/ensure_initialized.h
rename to media/webrtc/trunk/webrtc/test/constants.h
index 429879537c..14b2ba65bc 100644
--- a/media/webrtc/trunk/webrtc/modules/video_capture/ensure_initialized.h
+++ b/media/webrtc/trunk/webrtc/test/constants.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -9,11 +9,10 @@
  */
 
 namespace webrtc {
-namespace videocapturemodule {
+namespace test {
 
-// Ensure any necessary initialization of webrtc::videocapturemodule has
-// completed.
-void EnsureInitialized();
-
-}  // namespace videocapturemodule.
-}  // namespace webrtc.
+extern const int kTOffsetExtensionId;
+extern const int kAbsSendTimeExtensionId;
+extern const int kTransportSequenceNumberExtensionId;
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/direct_transport.cc b/media/webrtc/trunk/webrtc/test/direct_transport.cc
index 871bcb06c3..591e154b14 100644
--- a/media/webrtc/trunk/webrtc/test/direct_transport.cc
+++ b/media/webrtc/trunk/webrtc/test/direct_transport.cc
@@ -12,32 +12,23 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/call.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace test {
 
-DirectTransport::DirectTransport()
-    : lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      packet_event_(EventWrapper::Create()),
-      thread_(ThreadWrapper::CreateThread(
-          NetworkProcess, this, "NetworkProcess")),
-      clock_(Clock::GetRealTimeClock()),
-      shutting_down_(false),
-      fake_network_(FakeNetworkPipe::Config()) {
-  EXPECT_TRUE(thread_->Start());
-}
+DirectTransport::DirectTransport(Call* send_call)
+    : DirectTransport(FakeNetworkPipe::Config(), send_call) {}
 
-DirectTransport::DirectTransport(
-    const FakeNetworkPipe::Config& config)
-    : lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      packet_event_(EventWrapper::Create()),
-      thread_(ThreadWrapper::CreateThread(
-          NetworkProcess, this, "NetworkProcess")),
+DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
+                                 Call* send_call)
+    : send_call_(send_call),
+      packet_event_(false, false),
+      thread_(NetworkProcess, this, "NetworkProcess"),
       clock_(Clock::GetRealTimeClock()),
       shutting_down_(false),
-      fake_network_(config) {
-  EXPECT_TRUE(thread_->Start());
+      fake_network_(clock_, config) {
+  thread_.Start();
 }
 
 DirectTransport::~DirectTransport() { StopSending(); }
@@ -48,30 +39,41 @@ void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
 
 void DirectTransport::StopSending() {
   {
-    CriticalSectionScoped crit_(lock_.get());
+    rtc::CritScope crit(&lock_);
     shutting_down_ = true;
   }
 
-  packet_event_->Set();
-  EXPECT_TRUE(thread_->Stop());
+  packet_event_.Set();
+  thread_.Stop();
 }
 
 void DirectTransport::SetReceiver(PacketReceiver* receiver) {
   fake_network_.SetReceiver(receiver);
 }
 
-bool DirectTransport::SendRtp(const uint8_t* data, size_t length) {
+bool DirectTransport::SendRtp(const uint8_t* data,
+                              size_t length,
+                              const PacketOptions& options) {
+  if (send_call_) {
+    rtc::SentPacket sent_packet(options.packet_id,
+                                clock_->TimeInMilliseconds());
+    send_call_->OnSentPacket(sent_packet);
+  }
   fake_network_.SendPacket(data, length);
-  packet_event_->Set();
+  packet_event_.Set();
   return true;
 }
 
 bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
   fake_network_.SendPacket(data, length);
-  packet_event_->Set();
+  packet_event_.Set();
   return true;
 }
 
+int DirectTransport::GetAverageDelayMs() {
+  return fake_network_.AverageDelay();
+}
+
 bool DirectTransport::NetworkProcess(void* transport) {
   return static_cast<DirectTransport*>(transport)->SendPackets();
 }
@@ -80,17 +82,9 @@ bool DirectTransport::SendPackets() {
   fake_network_.Process();
   int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
   if (wait_time_ms > 0) {
-    switch (packet_event_->Wait(static_cast<unsigned long>(wait_time_ms))) {
-      case kEventSignaled:
-        break;
-      case kEventTimeout:
-        break;
-      case kEventError:
-        // TODO(pbos): Log a warning here?
-        return true;
-    }
+    packet_event_.Wait(static_cast<int>(wait_time_ms));
   }
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   return shutting_down_ ? false : true;
 }
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/test/direct_transport.h b/media/webrtc/trunk/webrtc/test/direct_transport.h
index 338be75e53..d68bc7184e 100644
--- a/media/webrtc/trunk/webrtc/test/direct_transport.h
+++ b/media/webrtc/trunk/webrtc/test/direct_transport.h
@@ -7,48 +7,55 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_
+#ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_
+#define WEBRTC_TEST_DIRECT_TRANSPORT_H_
 
 #include <assert.h>
 
 #include <deque>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/test/fake_network_pipe.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
+class Call;
 class Clock;
 class PacketReceiver;
 
 namespace test {
 
-class DirectTransport : public newapi::Transport {
+class DirectTransport : public Transport {
  public:
-  DirectTransport();
-  explicit DirectTransport(const FakeNetworkPipe::Config& config);
+  explicit DirectTransport(Call* send_call);
+  DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call);
   ~DirectTransport();
 
   void SetConfig(const FakeNetworkPipe::Config& config);
 
   virtual void StopSending();
+  // TODO(holmer): Look into moving this to the constructor.
   virtual void SetReceiver(PacketReceiver* receiver);
 
-  bool SendRtp(const uint8_t* data, size_t length) override;
+  bool SendRtp(const uint8_t* data,
+               size_t length,
+               const PacketOptions& options) override;
   bool SendRtcp(const uint8_t* data, size_t length) override;
 
+  int GetAverageDelayMs();
+
  private:
   static bool NetworkProcess(void* transport);
   bool SendPackets();
 
-  rtc::scoped_ptr<CriticalSectionWrapper> lock_;
-  rtc::scoped_ptr<EventWrapper> packet_event_;
-  rtc::scoped_ptr<ThreadWrapper> thread_;
+  rtc::CriticalSection lock_;
+  Call* const send_call_;
+  rtc::Event packet_event_;
+  rtc::PlatformThread thread_;
   Clock* const clock_;
 
   bool shutting_down_;
@@ -58,4 +65,4 @@ class DirectTransport : public newapi::Transport {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_
+#endif  // WEBRTC_TEST_DIRECT_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/test/fake_audio_device.cc b/media/webrtc/trunk/webrtc/test/fake_audio_device.cc
index a55be4a2fc..31cebda652 100644
--- a/media/webrtc/trunk/webrtc/test/fake_audio_device.cc
+++ b/media/webrtc/trunk/webrtc/test/fake_audio_device.cc
@@ -13,12 +13,11 @@
 #include <algorithm>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/media_file/source/media_file_utility.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/modules/media_file/media_file_utility.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace webrtc {
 namespace test {
@@ -30,8 +29,8 @@ FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
       playout_buffer_(),
       last_playout_ms_(-1),
       clock_(clock),
-      tick_(EventWrapper::Create()),
-      lock_(CriticalSectionWrapper::CreateCriticalSection()),
+      tick_(EventTimerWrapper::Create()),
+      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
       file_utility_(new ModuleFileUtility(0)),
       input_stream_(FileWrapper::Create()) {
   memset(captured_audio_, 0, sizeof(captured_audio_));
@@ -44,37 +43,29 @@ FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
 FakeAudioDevice::~FakeAudioDevice() {
   Stop();
 
-  if (thread_.get() != NULL)
-    thread_->Stop();
+  thread_.Stop();
 }
 
 int32_t FakeAudioDevice::Init() {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
     return -1;
 
   if (!tick_->StartTimer(true, 10))
     return -1;
-  thread_ = ThreadWrapper::CreateThread(FakeAudioDevice::Run, this,
-                                        "FakeAudioDevice");
-  if (thread_.get() == NULL)
-    return -1;
-  if (!thread_->Start()) {
-    thread_.reset();
-    return -1;
-  }
-  thread_->SetPriority(webrtc::kHighPriority);
+  thread_.Start();
+  thread_.SetPriority(rtc::kHighPriority);
   return 0;
 }
 
 int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   audio_callback_ = callback;
   return 0;
 }
 
 bool FakeAudioDevice::Playing() const {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   return capturing_;
 }
 
@@ -84,7 +75,7 @@ int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
 }
 
 bool FakeAudioDevice::Recording() const {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   return capturing_;
 }
 
@@ -95,13 +86,14 @@ bool FakeAudioDevice::Run(void* obj) {
 
 void FakeAudioDevice::CaptureAudio() {
   {
-    CriticalSectionScoped cs(lock_.get());
+    rtc::CritScope cs(&lock_);
     if (capturing_) {
       int bytes_read = file_utility_->ReadPCMData(
           *input_stream_.get(), captured_audio_, kBufferSizeBytes);
       if (bytes_read <= 0)
         return;
-      int num_samples = bytes_read / 2;  // 2 bytes per sample.
+      // 2 bytes per sample.
+      size_t num_samples = static_cast<size_t>(bytes_read / 2);
       uint32_t new_mic_level;
       EXPECT_EQ(0,
                 audio_callback_->RecordedDataIsAvailable(captured_audio_,
@@ -114,13 +106,15 @@ void FakeAudioDevice::CaptureAudio() {
                                                          0,
                                                          false,
                                                          new_mic_level));
-      uint32_t samples_needed = kFrequencyHz / 100;
+      size_t samples_needed = kFrequencyHz / 100;
       int64_t now_ms = clock_->TimeInMilliseconds();
       uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
-      if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0)
-        samples_needed = std::min(kFrequencyHz / time_since_last_playout_ms,
-                                  kBufferSizeBytes / 2);
-      uint32_t samples_out = 0;
+      if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
+        samples_needed = std::min(
+            static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
+            kBufferSizeBytes / 2);
+      }
+      size_t samples_out = 0;
       int64_t elapsed_time_ms = -1;
       int64_t ntp_time_ms = -1;
       EXPECT_EQ(0,
@@ -138,12 +132,12 @@ void FakeAudioDevice::CaptureAudio() {
 }
 
 void FakeAudioDevice::Start() {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   capturing_ = true;
 }
 
 void FakeAudioDevice::Stop() {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   capturing_ = false;
 }
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/test/fake_audio_device.h b/media/webrtc/trunk/webrtc/test/fake_audio_device.h
index 6df53e98b4..7ca657bbb6 100644
--- a/media/webrtc/trunk/webrtc/test/fake_audio_device.h
+++ b/media/webrtc/trunk/webrtc/test/fake_audio_device.h
@@ -12,6 +12,8 @@
 
 #include <string>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/typedefs.h"
@@ -19,11 +21,9 @@
 namespace webrtc {
 
 class Clock;
-class CriticalSectionWrapper;
-class EventWrapper;
+class EventTimerWrapper;
 class FileWrapper;
 class ModuleFileUtility;
-class ThreadWrapper;
 
 namespace test {
 
@@ -48,7 +48,7 @@ class FakeAudioDevice : public FakeAudioDeviceModule {
   void CaptureAudio();
 
   static const uint32_t kFrequencyHz = 16000;
-  static const uint32_t kBufferSizeBytes = 2 * kFrequencyHz;
+  static const size_t kBufferSizeBytes = 2 * kFrequencyHz;
 
   AudioTransport* audio_callback_;
   bool capturing_;
@@ -57,9 +57,9 @@ class FakeAudioDevice : public FakeAudioDeviceModule {
   int64_t last_playout_ms_;
 
   Clock* clock_;
-  rtc::scoped_ptr<EventWrapper> tick_;
-  rtc::scoped_ptr<CriticalSectionWrapper> lock_;
-  rtc::scoped_ptr<ThreadWrapper> thread_;
+  rtc::scoped_ptr<EventTimerWrapper> tick_;
+  mutable rtc::CriticalSection lock_;
+  rtc::PlatformThread thread_;
   rtc::scoped_ptr<ModuleFileUtility> file_utility_;
   rtc::scoped_ptr<FileWrapper> input_stream_;
 };
diff --git a/media/webrtc/trunk/webrtc/test/fake_common.h b/media/webrtc/trunk/webrtc/test/fake_common.h
deleted file mode 100644
index ec81798760..0000000000
--- a/media/webrtc/trunk/webrtc/test/fake_common.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_FAKE_COMMON_H_
-#define WEBRTC_TEST_FAKE_COMMON_H_
-
-// Borrowed from libjingle's talk/media/webrtc/fakewebrtccommon.h.
-
-#include "webrtc/typedefs.h"
-
-#define WEBRTC_STUB(method, args) \
-  int method args override { return 0; }
-
-#define WEBRTC_STUB_CONST(method, args) \
-  int method args const override { return 0; }
-
-#define WEBRTC_BOOL_STUB(method, args) \
-  bool method args override { return true; }
-
-#define WEBRTC_VOID_STUB(method, args) \
-  void method args override {}
-
-#define WEBRTC_FUNC(method, args) int method args override
-
-#define WEBRTC_FUNC_CONST(method, args) int method args const override
-
-#define WEBRTC_BOOL_FUNC(method, args) bool method args override
-
-#define WEBRTC_VOID_FUNC(method, args) void method args override
-
-#define WEBRTC_CHECK_CHANNEL(channel) \
-  if (channels_.find(channel) == channels_.end()) return -1;
-
-#define WEBRTC_ASSERT_CHANNEL(channel) \
-  ASSERT(channels_.find(channel) != channels_.end());
-
-#endif  // WEBRTC_TEST_FAKE_COMMON_H_
diff --git a/media/webrtc/trunk/webrtc/test/fake_decoder.cc b/media/webrtc/trunk/webrtc/test/fake_decoder.cc
index 63316e0dab..dbdd580e88 100644
--- a/media/webrtc/trunk/webrtc/test/fake_decoder.cc
+++ b/media/webrtc/trunk/webrtc/test/fake_decoder.cc
@@ -53,10 +53,16 @@ int32_t FakeDecoder::RegisterDecodeCompleteCallback(
 int32_t FakeDecoder::Release() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
+
 int32_t FakeDecoder::Reset() {
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
+const char* FakeDecoder::kImplementationName = "fake_decoder";
+const char* FakeDecoder::ImplementationName() const {
+  return kImplementationName;
+}
+
 int32_t FakeH264Decoder::Decode(const EncodedImage& input,
                                 bool missing_frames,
                                 const RTPFragmentationHeader* fragmentation,
diff --git a/media/webrtc/trunk/webrtc/test/fake_decoder.h b/media/webrtc/trunk/webrtc/test/fake_decoder.h
index 2031c676bf..0da961d9a0 100644
--- a/media/webrtc/trunk/webrtc/test/fake_decoder.h
+++ b/media/webrtc/trunk/webrtc/test/fake_decoder.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_
+#ifndef WEBRTC_TEST_FAKE_DECODER_H_
+#define WEBRTC_TEST_FAKE_DECODER_H_
 
 #include <vector>
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace test {
@@ -39,9 +39,13 @@ class FakeDecoder : public VideoDecoder {
   int32_t Release() override;
   int32_t Reset() override;
 
+  const char* ImplementationName() const override;
+
+  static const char* kImplementationName;
+
  private:
   VideoCodec config_;
-  I420VideoFrame frame_;
+  VideoFrame frame_;
   DecodedImageCallback* callback_;
 };
 
@@ -71,4 +75,4 @@ class FakeNullDecoder : public FakeDecoder {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_
+#endif  // WEBRTC_TEST_FAKE_DECODER_H_
diff --git a/media/webrtc/trunk/webrtc/test/fake_encoder.cc b/media/webrtc/trunk/webrtc/test/fake_encoder.cc
index 4c8d768383..72df40f9a5 100644
--- a/media/webrtc/trunk/webrtc/test/fake_encoder.cc
+++ b/media/webrtc/trunk/webrtc/test/fake_encoder.cc
@@ -12,8 +12,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 
 namespace webrtc {
 namespace test {
@@ -45,10 +45,9 @@ int32_t FakeEncoder::InitEncode(const VideoCodec* config,
   return 0;
 }
 
-int32_t FakeEncoder::Encode(
-    const I420VideoFrame& input_image,
-    const CodecSpecificInfo* codec_specific_info,
-    const std::vector<VideoFrameType>* frame_types) {
+int32_t FakeEncoder::Encode(const VideoFrame& input_image,
+                            const CodecSpecificInfo* codec_specific_info,
+                            const std::vector<FrameType>* frame_types) {
   assert(config_.maxFramerate > 0);
   int64_t time_since_last_encode_ms = 1000 / config_.maxFramerate;
   int64_t time_now_ms = clock_->TimeInMilliseconds();
@@ -58,6 +57,11 @@ int32_t FakeEncoder::Encode(
     // at the display time of the previous frame.
     time_since_last_encode_ms = time_now_ms - last_encode_time_ms_;
   }
+  if (time_since_last_encode_ms > 3 * 1000 / config_.maxFramerate) {
+    // Rudimentary check to make sure we don't widely overshoot bitrate target
+    // when resuming encoding after a suspension.
+    time_since_last_encode_ms = 3 * 1000 / config_.maxFramerate;
+  }
 
   size_t bits_available =
       static_cast<size_t>(target_bitrate_kbps_ * time_since_last_encode_ms);
@@ -98,11 +102,11 @@ int32_t FakeEncoder::Encode(
     encoded._timeStamp = input_image.timestamp();
     encoded.capture_time_ms_ = input_image.render_time_ms();
     encoded._frameType = (*frame_types)[i];
+    encoded._encodedWidth = config_.simulcastStream[i].width;
+    encoded._encodedHeight = config_.simulcastStream[i].height;
     // Always encode something on the first frame.
-    if (min_stream_bits > bits_available && i > 0) {
-      encoded._length = 0;
-      encoded._frameType = kSkipFrame;
-    }
+    if (min_stream_bits > bits_available && i > 0)
+      continue;
     assert(callback_ != NULL);
     if (callback_->Encoded(encoded, &specifics, NULL) != 0)
       return -1;
@@ -128,6 +132,11 @@ int32_t FakeEncoder::SetRates(uint32_t new_target_bitrate, uint32_t framerate) {
   return 0;
 }
 
+const char* FakeEncoder::kImplementationName = "fake_encoder";
+const char* FakeEncoder::ImplementationName() const {
+  return kImplementationName;
+}
+
 FakeH264Encoder::FakeH264Encoder(Clock* clock)
     : FakeEncoder(clock), callback_(NULL), idr_counter_(0) {
   FakeEncoder::RegisterEncodeCompleteCallback(this);
@@ -188,9 +197,9 @@ DelayedEncoder::DelayedEncoder(Clock* clock, int delay_ms)
     : test::FakeEncoder(clock),
       delay_ms_(delay_ms) {}
 
-int32_t DelayedEncoder::Encode(const I420VideoFrame& input_image,
+int32_t DelayedEncoder::Encode(const VideoFrame& input_image,
                                const CodecSpecificInfo* codec_specific_info,
-                               const std::vector<VideoFrameType>* frame_types) {
+                               const std::vector<FrameType>* frame_types) {
   SleepMs(delay_ms_);
   return FakeEncoder::Encode(input_image, codec_specific_info, frame_types);
 }
diff --git a/media/webrtc/trunk/webrtc/test/fake_encoder.h b/media/webrtc/trunk/webrtc/test/fake_encoder.h
index 096b9c22ee..6bff00e2a3 100644
--- a/media/webrtc/trunk/webrtc/test/fake_encoder.h
+++ b/media/webrtc/trunk/webrtc/test/fake_encoder.h
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_
+#ifndef WEBRTC_TEST_FAKE_ENCODER_H_
+#define WEBRTC_TEST_FAKE_ENCODER_H_
 
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video_encoder.h"
 
 namespace webrtc {
@@ -31,14 +31,17 @@ class FakeEncoder : public VideoEncoder {
   int32_t InitEncode(const VideoCodec* config,
                      int32_t number_of_cores,
                      size_t max_payload_size) override;
-  int32_t Encode(const I420VideoFrame& input_image,
+  int32_t Encode(const VideoFrame& input_image,
                  const CodecSpecificInfo* codec_specific_info,
-                 const std::vector<VideoFrameType>* frame_types) override;
+                 const std::vector<FrameType>* frame_types) override;
   int32_t RegisterEncodeCompleteCallback(
       EncodedImageCallback* callback) override;
   int32_t Release() override;
   int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
   int32_t SetRates(uint32_t new_target_bitrate, uint32_t framerate) override;
+  const char* ImplementationName() const override;
+
+  static const char* kImplementationName;
 
  protected:
   Clock* const clock_;
@@ -72,9 +75,9 @@ class DelayedEncoder : public test::FakeEncoder {
   DelayedEncoder(Clock* clock, int delay_ms);
   virtual ~DelayedEncoder() {}
 
-  int32_t Encode(const I420VideoFrame& input_image,
+  int32_t Encode(const VideoFrame& input_image,
                  const CodecSpecificInfo* codec_specific_info,
-                 const std::vector<VideoFrameType>* frame_types) override;
+                 const std::vector<FrameType>* frame_types) override;
 
  private:
   const int delay_ms_;
@@ -82,4 +85,4 @@ class DelayedEncoder : public test::FakeEncoder {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_
+#endif  // WEBRTC_TEST_FAKE_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/test/fake_network_pipe.cc b/media/webrtc/trunk/webrtc/test/fake_network_pipe.cc
index 93a4f6e1c6..491a0526b9 100644
--- a/media/webrtc/trunk/webrtc/test/fake_network_pipe.cc
+++ b/media/webrtc/trunk/webrtc/test/fake_network_pipe.cc
@@ -16,8 +16,7 @@
 #include <algorithm>
 
 #include "webrtc/call.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -71,16 +70,15 @@ class NetworkPacket {
   int64_t arrival_time_;
 };
 
-FakeNetworkPipe::FakeNetworkPipe(
-    const FakeNetworkPipe::Config& config)
-    : lock_(CriticalSectionWrapper::CreateCriticalSection()),
+FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
+                                 const FakeNetworkPipe::Config& config)
+    : clock_(clock),
       packet_receiver_(NULL),
       config_(config),
       dropped_packets_(0),
       sent_packets_(0),
       total_packet_delay_(0),
-      next_process_time_(TickTime::MillisecondTimestamp()) {
-}
+      next_process_time_(clock_->TimeInMilliseconds()) {}
 
 FakeNetworkPipe::~FakeNetworkPipe() {
   while (!capacity_link_.empty()) {
@@ -98,7 +96,7 @@ void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
 }
 
 void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   config_ = config;  // Shallow copy of the struct.
 }
 
@@ -107,7 +105,7 @@ void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
   // packets.
   if (packet_receiver_ == NULL)
     return;
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   if (config_.queue_length_packets > 0 &&
       capacity_link_.size() >= config_.queue_length_packets) {
     // Too many packet on the link, drop this one.
@@ -115,7 +113,7 @@ void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
     return;
   }
 
-  int64_t time_now = TickTime::MillisecondTimestamp();
+  int64_t time_now = clock_->TimeInMilliseconds();
 
   // Delay introduced by the link capacity.
   int64_t capacity_delay_ms = 0;
@@ -135,7 +133,7 @@ void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
 }
 
 float FakeNetworkPipe::PercentageLoss() {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   if (sent_packets_ == 0)
     return 0;
 
@@ -144,18 +142,19 @@ float FakeNetworkPipe::PercentageLoss() {
 }
 
 int FakeNetworkPipe::AverageDelay() {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   if (sent_packets_ == 0)
     return 0;
 
-  return total_packet_delay_ / static_cast<int>(sent_packets_);
+  return static_cast<int>(total_packet_delay_ /
+                          static_cast<int64_t>(sent_packets_));
 }
 
 void FakeNetworkPipe::Process() {
-  int64_t time_now = TickTime::MillisecondTimestamp();
+  int64_t time_now = clock_->TimeInMilliseconds();
   std::queue<NetworkPacket*> packets_to_deliver;
   {
-    CriticalSectionScoped crit(lock_.get());
+    rtc::CritScope crit(&lock_);
     // Check the capacity link first.
     while (capacity_link_.size() > 0 &&
            time_now >= capacity_link_.front()->arrival_time()) {
@@ -202,18 +201,19 @@ void FakeNetworkPipe::Process() {
   while (!packets_to_deliver.empty()) {
     NetworkPacket* packet = packets_to_deliver.front();
     packets_to_deliver.pop();
-    packet_receiver_->DeliverPacket(packet->data(), packet->data_length());
+    packet_receiver_->DeliverPacket(MediaType::ANY, packet->data(),
+                                    packet->data_length(), PacketTime());
     delete packet;
   }
 }
 
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   const int64_t kDefaultProcessIntervalMs = 30;
   if (capacity_link_.size() == 0 || delay_link_.size() == 0)
     return kDefaultProcessIntervalMs;
-  return std::max<int64_t>(
-      next_process_time_ - TickTime::MillisecondTimestamp(), 0);
+  return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
+                           0);
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fake_network_pipe.h b/media/webrtc/trunk/webrtc/test/fake_network_pipe.h
index b3b691ca2d..5d589d86f0 100644
--- a/media/webrtc/trunk/webrtc/test/fake_network_pipe.h
+++ b/media/webrtc/trunk/webrtc/test/fake_network_pipe.h
@@ -14,12 +14,13 @@
 #include <queue>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
+class Clock;
 class CriticalSectionWrapper;
 class NetworkPacket;
 class PacketReceiver;
@@ -32,26 +33,20 @@ class PacketReceiver;
 class FakeNetworkPipe {
  public:
   struct Config {
-    Config()
-        : queue_length_packets(0),
-          queue_delay_ms(0),
-          delay_standard_deviation_ms(0),
-          link_capacity_kbps(0),
-          loss_percent(0) {
-    }
+    Config() {}
     // Queue length in number of packets.
-    size_t queue_length_packets;
+    size_t queue_length_packets = 0;
     // Delay in addition to capacity induced delay.
-    int queue_delay_ms;
+    int queue_delay_ms = 0;
     // Standard deviation of the extra delay.
-    int delay_standard_deviation_ms;
+    int delay_standard_deviation_ms = 0;
     // Link capacity in kbps.
-    int link_capacity_kbps;
+    int link_capacity_kbps = 0;
     // Random packet loss.
-    int loss_percent;
+    int loss_percent = 0;
   };
 
-  explicit FakeNetworkPipe(const FakeNetworkPipe::Config& config);
+  FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config);
   ~FakeNetworkPipe();
 
   // Must not be called in parallel with SendPacket or Process.
@@ -75,7 +70,8 @@ class FakeNetworkPipe {
   size_t sent_packets() { return sent_packets_; }
 
  private:
-  rtc::scoped_ptr<CriticalSectionWrapper> lock_;
+  Clock* const clock_;
+  mutable rtc::CriticalSection lock_;
   PacketReceiver* packet_receiver_;
   std::queue<NetworkPacket*> capacity_link_;
   std::queue<NetworkPacket*> delay_link_;
@@ -86,11 +82,11 @@ class FakeNetworkPipe {
   // Statistics.
   size_t dropped_packets_;
   size_t sent_packets_;
-  int total_packet_delay_;
+  int64_t total_packet_delay_;
 
   int64_t next_process_time_;
 
-  DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
+  RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc b/media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc
index 4e5ec03563..ff18993829 100644
--- a/media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc
@@ -13,7 +13,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/fake_network_pipe.h"
 
 using ::testing::_;
@@ -29,19 +29,23 @@ class MockReceiver : public PacketReceiver {
   virtual ~MockReceiver() {}
 
   void IncomingPacket(const uint8_t* data, size_t length) {
-    DeliverPacket(data, length);
+    DeliverPacket(MediaType::ANY, data, length, PacketTime());
     delete [] data;
   }
 
-  MOCK_METHOD2(DeliverPacket, DeliveryStatus(const uint8_t*, size_t));
+  MOCK_METHOD4(
+      DeliverPacket,
+      DeliveryStatus(MediaType, const uint8_t*, size_t, const PacketTime&));
 };
 
 class FakeNetworkPipeTest : public ::testing::Test {
+ public:
+  FakeNetworkPipeTest() : fake_clock_(12345) {}
+
  protected:
   virtual void SetUp() {
-    TickTime::UseFakeClock(12345);
     receiver_.reset(new MockReceiver());
-    ON_CALL(*receiver_, DeliverPacket(_, _))
+    ON_CALL(*receiver_, DeliverPacket(_, _, _, _))
         .WillByDefault(Return(PacketReceiver::DELIVERY_OK));
   }
 
@@ -59,6 +63,7 @@ class FakeNetworkPipeTest : public ::testing::Test {
     return 8 * kPacketSize / capacity_kbps;
   }
 
+  SimulatedClock fake_clock_;
   rtc::scoped_ptr<MockReceiver> receiver_;
 };
 
@@ -69,7 +74,8 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config));
+  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+      new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
   // Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
@@ -83,26 +89,22 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) {
                                          kPacketSize);
 
   // Time haven't increased yet, so we souldn't get any packets.
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(0);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 
   // Advance enough time to release one packet.
-  TickTime::AdvanceFakeClock(kPacketTimeMs);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
   pipe->Process();
 
   // Release all but one packet
-  TickTime::AdvanceFakeClock(9 * kPacketTimeMs - 1);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(8);
+  fake_clock_.AdvanceTimeMilliseconds(9 * kPacketTimeMs - 1);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(8);
   pipe->Process();
 
   // And the last one.
-  TickTime::AdvanceFakeClock(1);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
   pipe->Process();
 }
 
@@ -112,7 +114,8 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
   config.queue_length_packets = 20;
   config.queue_delay_ms = 100;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config));
+  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+      new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
   const int kNumPackets = 2;
@@ -124,21 +127,18 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
                                          kPacketSize);
 
   // Increase more than kPacketTimeMs, but not more than the extra delay.
-  TickTime::AdvanceFakeClock(kPacketTimeMs);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(0);
+  fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 
   // Advance the network delay to get the first packet.
-  TickTime::AdvanceFakeClock(config.queue_delay_ms);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(config.queue_delay_ms);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
   pipe->Process();
 
   // Advance one more kPacketTimeMs to get the last packet.
-  TickTime::AdvanceFakeClock(kPacketTimeMs);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
   pipe->Process();
 }
 
@@ -148,7 +148,8 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 2;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config));
+  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+      new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
   const int kPacketSize = 1000;
@@ -160,9 +161,8 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
 
   // Increase time enough to deliver all three packets, verify only two are
   // delivered.
-  TickTime::AdvanceFakeClock(3 * kPacketTimeMs);
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(2);
+  fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2);
   pipe->Process();
 }
 
@@ -172,7 +172,8 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) {
   config.queue_length_packets = 2;
   config.queue_delay_ms = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config));
+  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+      new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
   const int kPacketSize = 1000;
@@ -181,10 +182,10 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) {
 
   // Send three packets and verify only 2 are delivered.
   SendPackets(pipe.get(), 3, kPacketSize);
-  TickTime::AdvanceFakeClock(3 * kPacketTimeMs + config.queue_delay_ms);
+  fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs +
+                                      config.queue_delay_ms);
 
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _))
-      .Times(2);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2);
   pipe->Process();
 
   // Packet 1: kPacketTimeMs + config.queue_delay_ms,
@@ -201,7 +202,8 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config));
+  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+      new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
   // Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
@@ -214,13 +216,13 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
   int packet_time_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
 
   // Time hasn't increased yet, so we souldn't get any packets.
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(0);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 
   // Advance time in steps to release one packet at a time.
   for (int i = 0; i < kNumPackets; ++i) {
-    TickTime::AdvanceFakeClock(packet_time_ms);
-    EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(1);
+    fake_clock_.AdvanceTimeMilliseconds(packet_time_ms);
+    EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
     pipe->Process();
   }
 
@@ -236,20 +238,20 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
   packet_time_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
 
   // Time hasn't increased yet, so we souldn't get any packets.
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(0);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 
   // Advance time in steps to release one packet at a time.
   for (int i = 0; i < kNumPackets; ++i) {
-    TickTime::AdvanceFakeClock(packet_time_ms);
-    EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(1);
+    fake_clock_.AdvanceTimeMilliseconds(packet_time_ms);
+    EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
     pipe->Process();
   }
 
   // Check that all the packets were sent.
   EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
-  TickTime::AdvanceFakeClock(pipe->TimeUntilNextProcess());
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(0);
+  fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess());
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 }
 
@@ -259,7 +261,8 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config));
+  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+      new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
   // Add 10 packets of 1000 bytes, = 80 kb.
@@ -282,27 +285,27 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
   int packet_time_2_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
 
   // Time hasn't increased yet, so we souldn't get any packets.
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(0);
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 
   // Advance time in steps to release one packet at a time.
   for (int i = 0; i < kNumPackets; ++i) {
-    TickTime::AdvanceFakeClock(packet_time_1_ms);
-    EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(1);
+    fake_clock_.AdvanceTimeMilliseconds(packet_time_1_ms);
+    EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
     pipe->Process();
   }
 
   // Advance time in steps to release one packet at a time.
   for (int i = 0; i < kNumPackets; ++i) {
-    TickTime::AdvanceFakeClock(packet_time_2_ms);
-    EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(1);
+    fake_clock_.AdvanceTimeMilliseconds(packet_time_2_ms);
+    EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1);
     pipe->Process();
   }
 
   // Check that all the packets were sent.
   EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
-  TickTime::AdvanceFakeClock(pipe->TimeUntilNextProcess());
-  EXPECT_CALL(*receiver_, DeliverPacket(_, _)).Times(0);
+  fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess());
+  EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0);
   pipe->Process();
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fake_texture_frame.cc b/media/webrtc/trunk/webrtc/test/fake_texture_frame.cc
new file mode 100644
index 0000000000..5d46eec4b6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fake_texture_frame.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/test/fake_texture_frame.h"
+
+namespace webrtc {
+namespace test {
+
+VideoFrame FakeNativeHandle::CreateFrame(FakeNativeHandle* native_handle,
+                                         int width,
+                                         int height,
+                                         uint32_t timestamp,
+                                         int64_t render_time_ms,
+                                         VideoRotation rotation) {
+  return VideoFrame(new rtc::RefCountedObject<FakeNativeHandleBuffer>(
+                        native_handle, width, height),
+                    timestamp, render_time_ms, rotation);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fake_texture_frame.h b/media/webrtc/trunk/webrtc/test/fake_texture_frame.h
new file mode 100644
index 0000000000..9575fae469
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fake_texture_frame.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_TEST_FAKE_TEXTURE_FRAME_H_
+#define WEBRTC_TEST_FAKE_TEXTURE_FRAME_H_
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeNativeHandle {
+ public:
+  static VideoFrame CreateFrame(FakeNativeHandle* native_handle,
+                                int width,
+                                int height,
+                                uint32_t timestamp,
+                                int64_t render_time_ms,
+                                VideoRotation rotation);
+};
+
+class FakeNativeHandleBuffer : public NativeHandleBuffer {
+ public:
+  FakeNativeHandleBuffer(void* native_handle, int width, int height)
+      : NativeHandleBuffer(native_handle, width, height) {}
+
+  ~FakeNativeHandleBuffer() {
+    delete reinterpret_cast<FakeNativeHandle*>(native_handle_);
+  }
+
+ private:
+  rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override {
+    rtc::scoped_refptr<VideoFrameBuffer> buffer(
+        new rtc::RefCountedObject<I420Buffer>(width_, height_));
+    int half_height = (height_ + 1) / 2;
+    int half_width = (width_ + 1) / 2;
+    memset(buffer->MutableData(kYPlane), 0, height_ * width_);
+    memset(buffer->MutableData(kUPlane), 0, half_height * half_width);
+    memset(buffer->MutableData(kVPlane), 0, half_height * half_width);
+    return buffer;
+  }
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  //  WEBRTC_TEST_FAKE_TEXTURE_FRAME_H_
diff --git a/media/webrtc/trunk/webrtc/test/field_trial.cc b/media/webrtc/trunk/webrtc/test/field_trial.cc
index 6b3d83cf56..c40d0783d8 100644
--- a/media/webrtc/trunk/webrtc/test/field_trial.cc
+++ b/media/webrtc/trunk/webrtc/test/field_trial.cc
@@ -17,28 +17,14 @@
 #include <map>
 #include <string>
 
-#include "webrtc/system_wrappers/interface/field_trial.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/system_wrappers/include/field_trial_default.h"
 
 namespace webrtc {
 namespace {
-// Clients of this library have show a clear intent to setup field trials by
-// linking with it. As so try to crash if they forget to call
-// InitFieldTrialsFromString before webrtc tries to access a field trial.
 bool field_trials_initiated_ = false;
-std::map<std::string, std::string> field_trials_;
 }  // namespace
 
-namespace field_trial {
-std::string FindFullName(const std::string& trial_name) {
-  assert(field_trials_initiated_);
-  std::map<std::string, std::string>::const_iterator it =
-      field_trials_.find(trial_name);
-  if (it == field_trials_.end())
-    return std::string();
-  return it->second;
-}
-}  // namespace field_trial
-
 namespace test {
 // Note: this code is copied from src/base/metrics/field_trial.cc since the aim
 // is to mimic chromium --force-fieldtrials.
@@ -46,12 +32,14 @@ void InitFieldTrialsFromString(const std::string& trials_string) {
   static const char kPersistentStringSeparator = '/';
 
   // Catch an error if this is called more than once.
-  assert(field_trials_initiated_ == false);
+  assert(!field_trials_initiated_);
   field_trials_initiated_ = true;
 
-  if (trials_string.empty()) return;
+  if (trials_string.empty())
+    return;
 
   size_t next_item = 0;
+  std::map<std::string, std::string> field_trials;
   while (next_item < trials_string.length()) {
     size_t name_end = trials_string.find(kPersistentStringSeparator, next_item);
     if (name_end == trials_string.npos || next_item == name_end)
@@ -66,21 +54,40 @@ void InitFieldTrialsFromString(const std::string& trials_string) {
     next_item = group_name_end + 1;
 
     // Fail if duplicate with different group name.
-    if (field_trials_.find(name) != field_trials_.end() &&
-        field_trials_.find(name)->second != group_name)
+    if (field_trials.find(name) != field_trials.end() &&
+        field_trials.find(name)->second != group_name) {
       break;
+    }
 
-    field_trials_[name] = group_name;
+    field_trials[name] = group_name;
 
     // Successfully parsed all field trials from the string.
-    if (next_item == trials_string.length())
+    if (next_item == trials_string.length()) {
+      webrtc::field_trial::InitFieldTrialsFromString(trials_string.c_str());
       return;
+    }
   }
-  // LOG does not prints when this is called early on main.
+  // Using fprintf as LOG does not print when this is called early in main.
   fprintf(stderr, "Invalid field trials string.\n");
 
-  // Using abort so it crashs both in debug and release mode.
+  // Using abort so it crashes in both debug and release mode.
   abort();
 }
+
+ScopedFieldTrials::ScopedFieldTrials(const std::string& config)
+  : previous_field_trials_(webrtc::field_trial::GetFieldTrialString()) {
+  assert(field_trials_initiated_);
+  field_trials_initiated_ = false;
+  current_field_trials_ = config;
+  InitFieldTrialsFromString(current_field_trials_);
+}
+
+ScopedFieldTrials::~ScopedFieldTrials() {
+  // Should still be initialized, since InitFieldTrials is called from ctor.
+  // That's why we don't restore the flag.
+  assert(field_trials_initiated_);
+  webrtc::field_trial::InitFieldTrialsFromString(previous_field_trials_);
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/field_trial.h b/media/webrtc/trunk/webrtc/test/field_trial.h
index 6503254729..735aa1f833 100644
--- a/media/webrtc/trunk/webrtc/test/field_trial.h
+++ b/media/webrtc/trunk/webrtc/test/field_trial.h
@@ -12,6 +12,7 @@
 #define WEBRTC_TEST_FIELD_TRIAL_H_
 
 #include <string>
+#include <map>
 
 namespace webrtc {
 namespace test {
@@ -31,6 +32,17 @@ namespace test {
 // passed to it. That can be used to find out if a binary is parsing the flags.
 void InitFieldTrialsFromString(const std::string& config);
 
+// This class is used to override field-trial configs within specific tests.
+// After this class goes out of scope previous field trials will be restored.
+class ScopedFieldTrials {
+ public:
+  explicit ScopedFieldTrials(const std::string& config);
+  ~ScopedFieldTrials();
+ private:
+  std::string current_field_trials_;
+  const char* previous_field_trials_;
+};
+
 }  // namespace test
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/test/frame_generator.cc b/media/webrtc/trunk/webrtc/test/frame_generator.cc
index 5152f1a874..589dde4bad 100644
--- a/media/webrtc/trunk/webrtc/test/frame_generator.cc
+++ b/media/webrtc/trunk/webrtc/test/frame_generator.cc
@@ -15,6 +15,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace test {
@@ -28,7 +29,7 @@ class ChromaGenerator : public FrameGenerator {
     assert(height > 0);
   }
 
-  I420VideoFrame* NextFrame() override {
+  VideoFrame* NextFrame() override {
     frame_.CreateEmptyFrame(static_cast<int>(width_),
                             static_cast<int>(height_),
                             static_cast<int>(width_),
@@ -48,7 +49,7 @@ class ChromaGenerator : public FrameGenerator {
   double angle_;
   size_t width_;
   size_t height_;
-  I420VideoFrame frame_;
+  VideoFrame frame_;
 };
 
 class YuvFileGenerator : public FrameGenerator {
@@ -77,7 +78,7 @@ class YuvFileGenerator : public FrameGenerator {
       fclose(file);
   }
 
-  I420VideoFrame* NextFrame() override {
+  VideoFrame* NextFrame() override {
     if (current_display_count_ == 0)
       ReadNextFrame();
     if (++current_display_count_ >= frame_display_count_)
@@ -123,9 +124,113 @@ class YuvFileGenerator : public FrameGenerator {
   const rtc::scoped_ptr<uint8_t[]> frame_buffer_;
   const int frame_display_count_;
   int current_display_count_;
-  I420VideoFrame last_read_frame_;
-  I420VideoFrame temp_frame_copy_;
+  VideoFrame last_read_frame_;
+  VideoFrame temp_frame_copy_;
 };
+
+class ScrollingImageFrameGenerator : public FrameGenerator {
+ public:
+  ScrollingImageFrameGenerator(Clock* clock,
+                               const std::vector<FILE*>& files,
+                               size_t source_width,
+                               size_t source_height,
+                               size_t target_width,
+                               size_t target_height,
+                               int64_t scroll_time_ms,
+                               int64_t pause_time_ms)
+      : clock_(clock),
+        start_time_(clock->TimeInMilliseconds()),
+        scroll_time_(scroll_time_ms),
+        pause_time_(pause_time_ms),
+        num_frames_(files.size()),
+        current_frame_num_(num_frames_ - 1),
+        current_source_frame_(nullptr),
+        file_generator_(files, source_width, source_height, 1) {
+    RTC_DCHECK(clock_ != nullptr);
+    RTC_DCHECK_GT(num_frames_, 0u);
+    RTC_DCHECK_GE(source_height, target_height);
+    RTC_DCHECK_GE(source_width, target_width);
+    RTC_DCHECK_GE(scroll_time_ms, 0);
+    RTC_DCHECK_GE(pause_time_ms, 0);
+    RTC_DCHECK_GT(scroll_time_ms + pause_time_ms, 0);
+    current_frame_.CreateEmptyFrame(static_cast<int>(target_width),
+                                    static_cast<int>(target_height),
+                                    static_cast<int>(target_width),
+                                    static_cast<int>((target_width + 1) / 2),
+                                    static_cast<int>((target_width + 1) / 2));
+  }
+
+  virtual ~ScrollingImageFrameGenerator() {}
+
+  VideoFrame* NextFrame() override {
+    const int64_t kFrameDisplayTime = scroll_time_ + pause_time_;
+    const int64_t now = clock_->TimeInMilliseconds();
+    int64_t ms_since_start = now - start_time_;
+
+    size_t frame_num = (ms_since_start / kFrameDisplayTime) % num_frames_;
+    UpdateSourceFrame(frame_num);
+
+    double scroll_factor;
+    int64_t time_into_frame = ms_since_start % kFrameDisplayTime;
+    if (time_into_frame < scroll_time_) {
+      scroll_factor = static_cast<double>(time_into_frame) / scroll_time_;
+    } else {
+      scroll_factor = 1.0;
+    }
+    CropSourceToScrolledImage(scroll_factor);
+
+    return &current_frame_;
+  }
+
+  void UpdateSourceFrame(size_t frame_num) {
+    while (current_frame_num_ != frame_num) {
+      current_source_frame_ = file_generator_.NextFrame();
+      current_frame_num_ = (current_frame_num_ + 1) % num_frames_;
+    }
+    RTC_DCHECK(current_source_frame_ != nullptr);
+  }
+
+  void CropSourceToScrolledImage(double scroll_factor) {
+    const int kTargetWidth = current_frame_.width();
+    const int kTargetHeight = current_frame_.height();
+    int scroll_margin_x = current_source_frame_->width() - kTargetWidth;
+    int pixels_scrolled_x =
+        static_cast<int>(scroll_margin_x * scroll_factor + 0.5);
+    int scroll_margin_y = current_source_frame_->height() - kTargetHeight;
+    int pixels_scrolled_y =
+        static_cast<int>(scroll_margin_y * scroll_factor + 0.5);
+
+    int offset_y = (current_source_frame_->stride(PlaneType::kYPlane) *
+                    pixels_scrolled_y) +
+                   pixels_scrolled_x;
+    int offset_u = (current_source_frame_->stride(PlaneType::kUPlane) *
+                    (pixels_scrolled_y / 2)) +
+                   (pixels_scrolled_x / 2);
+    int offset_v = (current_source_frame_->stride(PlaneType::kVPlane) *
+                    (pixels_scrolled_y / 2)) +
+                   (pixels_scrolled_x / 2);
+
+    current_frame_.CreateFrame(
+        &current_source_frame_->buffer(PlaneType::kYPlane)[offset_y],
+        &current_source_frame_->buffer(PlaneType::kUPlane)[offset_u],
+        &current_source_frame_->buffer(PlaneType::kVPlane)[offset_v],
+        kTargetWidth, kTargetHeight,
+        current_source_frame_->stride(PlaneType::kYPlane),
+        current_source_frame_->stride(PlaneType::kUPlane),
+        current_source_frame_->stride(PlaneType::kVPlane));
+  }
+
+  Clock* const clock_;
+  const int64_t start_time_;
+  const int64_t scroll_time_;
+  const int64_t pause_time_;
+  const size_t num_frames_;
+  size_t current_frame_num_;
+  VideoFrame* current_source_frame_;
+  VideoFrame current_frame_;
+  YuvFileGenerator file_generator_;
+};
+
 }  // namespace
 
 FrameGenerator* FrameGenerator::CreateChromaGenerator(size_t width,
@@ -142,12 +247,34 @@ FrameGenerator* FrameGenerator::CreateFromYuvFile(
   std::vector<FILE*> files;
   for (const std::string& filename : filenames) {
     FILE* file = fopen(filename.c_str(), "rb");
-    DCHECK(file != nullptr);
+    RTC_DCHECK(file != nullptr);
     files.push_back(file);
   }
 
   return new YuvFileGenerator(files, width, height, frame_repeat_count);
 }
 
+FrameGenerator* FrameGenerator::CreateScrollingInputFromYuvFiles(
+    Clock* clock,
+    std::vector<std::string> filenames,
+    size_t source_width,
+    size_t source_height,
+    size_t target_width,
+    size_t target_height,
+    int64_t scroll_time_ms,
+    int64_t pause_time_ms) {
+  assert(!filenames.empty());
+  std::vector<FILE*> files;
+  for (const std::string& filename : filenames) {
+    FILE* file = fopen(filename.c_str(), "rb");
+    RTC_DCHECK(file != nullptr);
+    files.push_back(file);
+  }
+
+  return new ScrollingImageFrameGenerator(
+      clock, files, source_width, source_height, target_width, target_height,
+      scroll_time_ms, pause_time_ms);
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/frame_generator.h b/media/webrtc/trunk/webrtc/test/frame_generator.h
index 969a358ffb..7f20c749e8 100644
--- a/media/webrtc/trunk/webrtc/test/frame_generator.h
+++ b/media/webrtc/trunk/webrtc/test/frame_generator.h
@@ -13,10 +13,11 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
+class Clock;
 namespace test {
 
 class FrameGenerator {
@@ -25,7 +26,7 @@ class FrameGenerator {
   virtual ~FrameGenerator() {}
 
   // Returns video frame that remains valid until next call.
-  virtual I420VideoFrame* NextFrame() = 0;
+  virtual VideoFrame* NextFrame() = 0;
 
   // Creates a test frame generator that creates fully saturated frames with
   // varying U, V values over time.
@@ -38,6 +39,24 @@ class FrameGenerator {
                                            size_t width,
                                            size_t height,
                                            int frame_repeat_count);
+
+  // Creates a frame generator which takes a set of yuv files (wrapping a
+  // frame generator created by CreateFromYuvFile() above), but outputs frames
+  // that have been cropped to specified resolution: source_width/source_height
+  // is the size of the source images, target_width/target_height is the size of
+  // the cropped output. For each source image read, the cropped viewport will
+  // be scrolled top to bottom/left to right for scroll_tim_ms milliseconds.
+  // After that the image will stay in place for pause_time_ms milliseconds,
+  // and then this will be repeated with the next file from the input set.
+  static FrameGenerator* CreateScrollingInputFromYuvFiles(
+      Clock* clock,
+      std::vector<std::string> filenames,
+      size_t source_width,
+      size_t source_height,
+      size_t target_width,
+      size_t target_height,
+      int64_t scroll_time_ms,
+      int64_t pause_time_ms);
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc b/media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc
index 664ed6b19a..35ce6168a2 100644
--- a/media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc
+++ b/media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc
@@ -10,23 +10,22 @@
 
 #include "webrtc/test/frame_generator_capturer.h"
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/frame_generator.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 namespace test {
 
-FrameGeneratorCapturer* FrameGeneratorCapturer::Create(
-    VideoSendStreamInput* input,
-    size_t width,
-    size_t height,
-    int target_fps,
-    Clock* clock) {
+FrameGeneratorCapturer* FrameGeneratorCapturer::Create(VideoCaptureInput* input,
+                                                       size_t width,
+                                                       size_t height,
+                                                       int target_fps,
+                                                       Clock* clock) {
   FrameGeneratorCapturer* capturer = new FrameGeneratorCapturer(
       clock, input, FrameGenerator::CreateChromaGenerator(width, height),
       target_fps);
@@ -39,7 +38,7 @@ FrameGeneratorCapturer* FrameGeneratorCapturer::Create(
 }
 
 FrameGeneratorCapturer* FrameGeneratorCapturer::CreateFromYuvFile(
-    VideoSendStreamInput* input,
+    VideoCaptureInput* input,
     const std::string& file_name,
     size_t width,
     size_t height,
@@ -59,14 +58,14 @@ FrameGeneratorCapturer* FrameGeneratorCapturer::CreateFromYuvFile(
 }
 
 FrameGeneratorCapturer::FrameGeneratorCapturer(Clock* clock,
-                                               VideoSendStreamInput* input,
+                                               VideoCaptureInput* input,
                                                FrameGenerator* frame_generator,
                                                int target_fps)
     : VideoCapturer(input),
       clock_(clock),
       sending_(false),
-      tick_(EventWrapper::Create()),
-      lock_(CriticalSectionWrapper::CreateCriticalSection()),
+      tick_(EventTimerWrapper::Create()),
+      thread_(FrameGeneratorCapturer::Run, this, "FrameGeneratorCapturer"),
       frame_generator_(frame_generator),
       target_fps_(target_fps),
       first_frame_capture_time_(-1) {
@@ -78,8 +77,7 @@ FrameGeneratorCapturer::FrameGeneratorCapturer(Clock* clock,
 FrameGeneratorCapturer::~FrameGeneratorCapturer() {
   Stop();
 
-  if (thread_.get() != NULL)
-    thread_->Stop();
+  thread_.Stop();
 }
 
 bool FrameGeneratorCapturer::Init() {
@@ -90,15 +88,8 @@ bool FrameGeneratorCapturer::Init() {
 
   if (!tick_->StartTimer(true, 1000 / target_fps_))
     return false;
-  thread_ = ThreadWrapper::CreateThread(FrameGeneratorCapturer::Run, this,
-                                        "FrameGeneratorCapturer");
-  if (thread_.get() == NULL)
-    return false;
-  if (!thread_->Start()) {
-    thread_.reset();
-    return false;
-  }
-  thread_->SetPriority(webrtc::kHighPriority);
+  thread_.Start();
+  thread_.SetPriority(rtc::kHighPriority);
   return true;
 }
 
@@ -109,9 +100,9 @@ bool FrameGeneratorCapturer::Run(void* obj) {
 
 void FrameGeneratorCapturer::InsertFrame() {
   {
-    CriticalSectionScoped cs(lock_.get());
+    rtc::CritScope cs(&lock_);
     if (sending_) {
-      I420VideoFrame* frame = frame_generator_->NextFrame();
+      VideoFrame* frame = frame_generator_->NextFrame();
       frame->set_ntp_time_ms(clock_->CurrentNtpInMilliseconds());
       if (first_frame_capture_time_ == -1) {
         first_frame_capture_time_ = frame->ntp_time_ms();
@@ -123,13 +114,17 @@ void FrameGeneratorCapturer::InsertFrame() {
 }
 
 void FrameGeneratorCapturer::Start() {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   sending_ = true;
 }
 
 void FrameGeneratorCapturer::Stop() {
-  CriticalSectionScoped cs(lock_.get());
+  rtc::CritScope cs(&lock_);
   sending_ = false;
 }
+
+void FrameGeneratorCapturer::ForceFrame() {
+  tick_->Set();
+}
 }  // test
 }  // webrtc
diff --git a/media/webrtc/trunk/webrtc/test/frame_generator_capturer.h b/media/webrtc/trunk/webrtc/test/frame_generator_capturer.h
index 1064a5f3de..6bd0e0b327 100644
--- a/media/webrtc/trunk/webrtc/test/frame_generator_capturer.h
+++ b/media/webrtc/trunk/webrtc/test/frame_generator_capturer.h
@@ -7,11 +7,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_
+#ifndef WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
+#define WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
 
 #include <string>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/video_capturer.h"
 #include "webrtc/typedefs.h"
@@ -19,8 +21,7 @@
 namespace webrtc {
 
 class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
+class EventTimerWrapper;
 
 namespace test {
 
@@ -28,13 +29,13 @@ class FrameGenerator;
 
 class FrameGeneratorCapturer : public VideoCapturer {
  public:
-  static FrameGeneratorCapturer* Create(VideoSendStreamInput* input,
+  static FrameGeneratorCapturer* Create(VideoCaptureInput* input,
                                         size_t width,
                                         size_t height,
                                         int target_fps,
                                         Clock* clock);
 
-  static FrameGeneratorCapturer* CreateFromYuvFile(VideoSendStreamInput* input,
+  static FrameGeneratorCapturer* CreateFromYuvFile(VideoCaptureInput* input,
                                                    const std::string& file_name,
                                                    size_t width,
                                                    size_t height,
@@ -44,11 +45,12 @@ class FrameGeneratorCapturer : public VideoCapturer {
 
   void Start() override;
   void Stop() override;
+  void ForceFrame();
 
   int64_t first_frame_capture_time() const { return first_frame_capture_time_; }
 
   FrameGeneratorCapturer(Clock* clock,
-                         VideoSendStreamInput* input,
+                         VideoCaptureInput* input,
                          FrameGenerator* frame_generator,
                          int target_fps);
   bool Init();
@@ -60,9 +62,9 @@ class FrameGeneratorCapturer : public VideoCapturer {
   Clock* const clock_;
   bool sending_;
 
-  rtc::scoped_ptr<EventWrapper> tick_;
-  rtc::scoped_ptr<CriticalSectionWrapper> lock_;
-  rtc::scoped_ptr<ThreadWrapper> thread_;
+  rtc::scoped_ptr<EventTimerWrapper> tick_;
+  rtc::CriticalSection lock_;
+  rtc::PlatformThread thread_;
   rtc::scoped_ptr<FrameGenerator> frame_generator_;
 
   int target_fps_;
@@ -72,4 +74,4 @@ class FrameGeneratorCapturer : public VideoCapturer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_
+#endif  // WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
diff --git a/media/webrtc/trunk/webrtc/test/frame_generator_unittest.cc b/media/webrtc/trunk/webrtc/test/frame_generator_unittest.cc
index 7d9a3a434d..6376e2c221 100644
--- a/media/webrtc/trunk/webrtc/test/frame_generator_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/frame_generator_unittest.cc
@@ -55,10 +55,7 @@ class FrameGeneratorTest : public ::testing::Test {
     fwrite(plane_buffer.get(), 1, uv_size, file);
   }
 
-  void CheckFrameAndMutate(I420VideoFrame* frame,
-                           uint8_t y,
-                           uint8_t u,
-                           uint8_t v) {
+  void CheckFrameAndMutate(VideoFrame* frame, uint8_t y, uint8_t u, uint8_t v) {
     // Check that frame is valid, has the correct color and timestamp are clean.
     ASSERT_NE(nullptr, frame);
     uint8_t* buffer;
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/BUILD.gn b/media/webrtc/trunk/webrtc/test/fuzzers/BUILD.gn
new file mode 100644
index 0000000000..6a43548ec9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/BUILD.gn
@@ -0,0 +1,115 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/features.gni")
+import("webrtc_fuzzer.gni")
+
+static_library("webrtc_fuzzer_main") {
+  public_configs = [ "../..:common_inherited_config" ]
+  sources = [
+    "webrtc_fuzzer_main.cc",
+  ]
+  deps = [
+    "../../system_wrappers:field_trial_default",
+    "../../system_wrappers:metrics_default",
+    "//testing/libfuzzer:libfuzzer_main",
+  ]
+}
+
+webrtc_fuzzer_test("h264_depacketizer_fuzzer") {
+  sources = [
+    "h264_depacketizer_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp",
+  ]
+}
+
+webrtc_fuzzer_test("vp8_depacketizer_fuzzer") {
+  sources = [
+    "vp8_depacketizer_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp",
+  ]
+}
+
+webrtc_fuzzer_test("vp9_depacketizer_fuzzer") {
+  sources = [
+    "vp9_depacketizer_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp",
+  ]
+}
+
+webrtc_fuzzer_test("vp8_qp_parser_fuzzer") {
+  sources = [
+    "vp8_qp_parser_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/video_coding/",
+  ]
+}
+
+webrtc_fuzzer_test("producer_fec_fuzzer") {
+  sources = [
+    "producer_fec_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp/",
+  ]
+}
+
+source_set("audio_decoder_fuzzer") {
+  public_configs = [ "../..:common_inherited_config" ]
+  sources = [
+    "audio_decoder_fuzzer.cc",
+    "audio_decoder_fuzzer.h",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") {
+  sources = [
+    "audio_decoder_ilbc_fuzzer.cc",
+  ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:ilbc",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_isac_fuzzer") {
+  sources = [
+    "audio_decoder_isac_fuzzer.cc",
+  ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:isac",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") {
+  sources = [
+    "audio_decoder_isacfix_fuzzer.cc",
+  ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:isac_fix",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_opus_fuzzer") {
+  sources = [
+    "audio_decoder_opus_fuzzer.cc",
+  ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:webrtc_opus",
+  ]
+}
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/OWNERS b/media/webrtc/trunk/webrtc/test/fuzzers/OWNERS
new file mode 100644
index 0000000000..6782b61fca
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/OWNERS
@@ -0,0 +1 @@
+pbos@webrtc.org
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
new file mode 100644
index 0000000000..fb5adb6cd8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+namespace webrtc {
+namespace {
+size_t PacketSizeFromTwoBytes(const uint8_t* data, size_t size) {
+  if (size < 2)
+    return 0;
+  return static_cast<size_t>((data[0] << 8) + data[1]);
+}
+}  // namespace
+
+// This function reads two bytes from the beginning of |data|, interprets them
+// as the first packet length, and reads this many bytes if available. The
+// payload is inserted into the decoder, and the process continues until no more
+// data is available.
+void FuzzAudioDecoder(const uint8_t* data,
+                      size_t size,
+                      AudioDecoder* decoder,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded) {
+  const uint8_t* data_ptr = data;
+  size_t remaining_size = size;
+  size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
+  while (packet_len != 0 && packet_len <= remaining_size - 2) {
+    data_ptr += 2;
+    remaining_size -= 2;
+    AudioDecoder::SpeechType speech_type;
+    decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes,
+                    decoded, &speech_type);
+    data_ptr += packet_len;
+    remaining_size -= packet_len;
+    packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
+  }
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_fuzzer.h b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_fuzzer.h
new file mode 100644
index 0000000000..cdd8574300
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_fuzzer.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
+#define WEBRTC_TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
+
+#include <stddef.h>
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDecoder;
+
+void FuzzAudioDecoder(const uint8_t* data,
+                      size_t size,
+                      AudioDecoder* decoder,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
new file mode 100644
index 0000000000..d2a87f0cb6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  AudioDecoderIlbc dec;
+  static const int kSampleRateHz = 8000;
+  static const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc
new file mode 100644
index 0000000000..984cfda398
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  AudioDecoderIsac dec(nullptr);
+  const int sample_rate_hz = size % 2 == 0 ? 16000 : 32000;     // 16 or 32 kHz.
+  static const size_t kAllocatedOuputSizeSamples = 32000 / 10;  // 100 ms.
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(data, size, &dec, sample_rate_hz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc
new file mode 100644
index 0000000000..83fb8c2d62
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  AudioDecoderIsacFix dec(nullptr);
+  static const int kSampleRateHz = 16000;
+  static const size_t kAllocatedOuputSizeSamples = 16000 / 10;  // 100 ms.
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc
new file mode 100644
index 0000000000..3d70ec507d
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
+  AudioDecoderOpus dec(channels);
+  const int kSampleRateHz = 48000;
+  const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/h264_depacketizer_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/h264_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..ca73d9495f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/h264_depacketizer_fuzzer.cc
@@ -0,0 +1,18 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RtpDepacketizerH264 depacketizer;
+  RtpDepacketizer::ParsedPayload parsed_payload;
+  depacketizer.Parse(&parsed_payload, data, size);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/producer_fec_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/producer_fec_fuzzer.cc
new file mode 100644
index 0000000000..7322fed4bf
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/producer_fec_fuzzer.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  ForwardErrorCorrection fec;
+  ProducerFec producer(&fec);
+  size_t i = 0;
+  if (size < 4)
+    return;
+  FecProtectionParams params = {data[i++] % 128, data[i++] % 1,
+                                static_cast<int>(data[i++] % 10),
+                                kFecMaskBursty};
+  producer.SetFecParameters(&params, 0);
+  uint16_t seq_num = data[i++];
+
+  while (i + 3 < size) {
+    size_t rtp_header_length = data[i++] % 10 + 12;
+    size_t payload_size = data[i++] % 10;
+    if (i + payload_size + rtp_header_length + 2 > size)
+      break;
+    rtc::scoped_ptr<uint8_t[]> packet(
+        new uint8_t[payload_size + rtp_header_length]);
+    memcpy(packet.get(), &data[i], payload_size + rtp_header_length);
+    ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
+    i += payload_size + rtp_header_length;
+    // Make sure sequence numbers are increasing.
+    const int kRedPayloadType = 98;
+    rtc::scoped_ptr<RedPacket> red_packet(producer.BuildRedPacket(
+        packet.get(), payload_size, rtp_header_length, kRedPayloadType));
+    bool protect = static_cast<bool>(data[i++] % 2);
+    if (protect) {
+      producer.AddRtpPacketAndGenerateFec(packet.get(), payload_size,
+                                          rtp_header_length);
+    }
+    uint16_t num_fec_packets = producer.NumAvailableFecPackets();
+    std::vector<RedPacket*> fec_packets;
+    if (num_fec_packets > 0) {
+      fec_packets =
+          producer.GetFecPackets(kRedPayloadType, 99, 100, rtp_header_length);
+      RTC_CHECK_EQ(num_fec_packets, fec_packets.size());
+    }
+    for (RedPacket* fec_packet : fec_packets) {
+      delete fec_packet;
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..d048372456
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc
@@ -0,0 +1,18 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RtpDepacketizerVp8 depacketizer;
+  RtpDepacketizer::ParsedPayload parsed_payload;
+  depacketizer.Parse(&parsed_payload, data, size);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.cc b/media/webrtc/trunk/webrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc
similarity index 63%
rename from media/webrtc/trunk/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.cc
rename to media/webrtc/trunk/webrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc
index 5c7fa4f1de..5135f1a471 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.cc
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -7,9 +7,11 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h"
+#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
 
 namespace webrtc {
-  int32_t AudioDeviceUtilityDummy::Init() { return 0; }
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  int qp;
+  vp8::GetQp(data, size, &qp);
+}
 }  // namespace webrtc
-
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc b/media/webrtc/trunk/webrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..02a7cc0f81
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc
@@ -0,0 +1,18 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RtpDepacketizerVp9 depacketizer;
+  RtpDepacketizer::ParsedPayload parsed_payload;
+  depacketizer.Parse(&parsed_payload, data, size);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/webrtc_fuzzer.gni b/media/webrtc/trunk/webrtc/test/fuzzers/webrtc_fuzzer.gni
new file mode 100644
index 0000000000..d264392c07
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/webrtc_fuzzer.gni
@@ -0,0 +1,28 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//testing/test.gni")
+
+template("webrtc_fuzzer_test") {
+  assert(defined(invoker.sources), "Need sources in $target_name.")
+
+  test(target_name) {
+    forward_variables_from(invoker, [ "sources" ])
+    deps = [
+      ":webrtc_fuzzer_main",
+    ]
+    if (defined(invoker.deps)) {
+      deps += invoker.deps
+    }
+    if (is_clang) {
+      # Suppress warnings from Chrome's Clang plugins.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}
diff --git a/media/webrtc/trunk/webrtc/test/fuzzers/webrtc_fuzzer_main.cc b/media/webrtc/trunk/webrtc/test/fuzzers/webrtc_fuzzer_main.cc
new file mode 100644
index 0000000000..50a513c094
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/fuzzers/webrtc_fuzzer_main.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is intended to provide a common interface for fuzzing functions.
+// It's intended to set sane defaults, such as removing logging for further
+// fuzzing efficiency.
+
+#include "webrtc/base/logging.h"
+
+namespace {
+bool g_initialized = false;
+void InitializeWebRtcFuzzDefaults() {
+  if (g_initialized)
+    return;
+
+  // Remove default logging to prevent huge slowdowns.
+  // TODO(pbos): Disable in Chromium: http://crbug.com/561667
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+  rtc::LogMessage::LogToDebug(rtc::LS_NONE);
+#endif  // !defined(WEBRTC_CHROMIUM_BUILD)
+
+  g_initialized = true;
+}
+}
+
+namespace webrtc {
+extern void FuzzOneInput(const uint8_t* data, size_t size);
+}  // namespace webrtc
+
+extern "C" int LLVMFuzzerTestOneInput(const unsigned char *data, size_t size) {
+  InitializeWebRtcFuzzDefaults();
+  webrtc::FuzzOneInput(data, size);
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc b/media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc
index f2c7acd6c0..ff87d9999a 100644
--- a/media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc
+++ b/media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc
@@ -69,7 +69,7 @@ void GlRenderer::ResizeVideo(size_t width, size_t height) {
                GL_UNSIGNED_INT_8_8_8_8, static_cast<GLvoid*>(buffer_));
 }
 
-void GlRenderer::RenderFrame(const webrtc::I420VideoFrame& frame,
+void GlRenderer::RenderFrame(const webrtc::VideoFrame& frame,
                              int /*render_delay_ms*/) {
   assert(is_init_);
 
diff --git a/media/webrtc/trunk/webrtc/test/gl/gl_renderer.h b/media/webrtc/trunk/webrtc/test/gl/gl_renderer.h
index b712f25695..7682d3c918 100644
--- a/media/webrtc/trunk/webrtc/test/gl/gl_renderer.h
+++ b/media/webrtc/trunk/webrtc/test/gl/gl_renderer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_
+#ifndef WEBRTC_TEST_GL_GL_RENDERER_H_
+#define WEBRTC_TEST_GL_GL_RENDERER_H_
 
 #ifdef WEBRTC_MAC
 #include <OpenGL/gl.h>
@@ -26,7 +26,7 @@ namespace test {
 
 class GlRenderer : public VideoRenderer {
  public:
-  void RenderFrame(const webrtc::I420VideoFrame& frame,
+  void RenderFrame(const webrtc::VideoFrame& frame,
                    int time_to_render_ms) override;
 
  protected:
@@ -48,4 +48,4 @@ class GlRenderer : public VideoRenderer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_
+#endif  // WEBRTC_TEST_GL_GL_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/test/histogram.cc b/media/webrtc/trunk/webrtc/test/histogram.cc
index 42307151f7..2893e4389a 100644
--- a/media/webrtc/trunk/webrtc/test/histogram.cc
+++ b/media/webrtc/trunk/webrtc/test/histogram.cc
@@ -12,38 +12,86 @@
 
 #include <map>
 
-#include "webrtc/system_wrappers/interface/metrics.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 // Test implementation of histogram methods in
-// webrtc/system_wrappers/interface/metrics.h.
+// webrtc/system_wrappers/include/metrics.h.
 
 namespace webrtc {
 namespace {
-// Map holding the last added sample to a histogram (mapped by histogram name).
-std::map<std::string, int> histograms_;
+struct SampleInfo {
+  SampleInfo(const std::string& name) : name_(name), last_(-1), total_(0) {}
+  const std::string name_;
+  int last_;   // Last added sample.
+  int total_;  // Total number of added samples.
+};
+
+rtc::CriticalSection histogram_crit_;
+// Map holding info about added samples to a histogram (mapped by the histogram
+// name).
+std::map<std::string, SampleInfo> histograms_ GUARDED_BY(histogram_crit_);
 }  // namespace
 
 namespace metrics {
 Histogram* HistogramFactoryGetCounts(const std::string& name, int min, int max,
-    int bucket_count) { return NULL; }
+                                     int bucket_count) {
+  rtc::CritScope cs(&histogram_crit_);
+  if (histograms_.find(name) == histograms_.end()) {
+    histograms_.insert(std::make_pair(name, SampleInfo(name)));
+  }
+  auto it = histograms_.find(name);
+  return reinterpret_cast<Histogram*>(&it->second);
+}
 
 Histogram* HistogramFactoryGetEnumeration(const std::string& name,
-    int boundary) { return NULL; }
+                                          int boundary) {
+  rtc::CritScope cs(&histogram_crit_);
+  if (histograms_.find(name) == histograms_.end()) {
+    histograms_.insert(std::make_pair(name, SampleInfo(name)));
+  }
+  auto it = histograms_.find(name);
+  return reinterpret_cast<Histogram*>(&it->second);
+}
 
 void HistogramAdd(
     Histogram* histogram_pointer, const std::string& name, int sample) {
-  histograms_[name] = sample;
+  rtc::CritScope cs(&histogram_crit_);
+  SampleInfo* ptr = reinterpret_cast<SampleInfo*>(histogram_pointer);
+  // The name should not vary.
+  RTC_CHECK(ptr->name_ == name);
+  ptr->last_ = sample;
+  ++ptr->total_;
 }
 }  // namespace metrics
 
 namespace test {
 int LastHistogramSample(const std::string& name) {
-  std::map<std::string, int>::const_iterator it = histograms_.find(name);
+  rtc::CritScope cs(&histogram_crit_);
+  const auto it = histograms_.find(name);
   if (it == histograms_.end()) {
     return -1;
   }
-  return it->second;
+  return it->second.last_;
+}
+
+int NumHistogramSamples(const std::string& name) {
+  rtc::CritScope cs(&histogram_crit_);
+  const auto it = histograms_.find(name);
+  if (it == histograms_.end()) {
+    return 0;
+  }
+  return it->second.total_;
+}
+
+void ClearHistograms() {
+  rtc::CritScope cs(&histogram_crit_);
+  for (auto& it : histograms_) {
+    it.second.last_ = -1;
+    it.second.total_ = 0;
+  }
 }
 }  // namespace test
 }  // namespace webrtc
-
diff --git a/media/webrtc/trunk/webrtc/test/histogram.h b/media/webrtc/trunk/webrtc/test/histogram.h
index 213fc8c22f..3c8e743aa1 100644
--- a/media/webrtc/trunk/webrtc/test/histogram.h
+++ b/media/webrtc/trunk/webrtc/test/histogram.h
@@ -20,6 +20,12 @@ namespace test {
 // found).
 int LastHistogramSample(const std::string& name);
 
+// Returns the number of added samples to a histogram.
+int NumHistogramSamples(const std::string& name);
+
+// Removes all histogram samples.
+void ClearHistograms();
+
 }  // namespace test
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/test/layer_filtering_transport.cc b/media/webrtc/trunk/webrtc/test/layer_filtering_transport.cc
new file mode 100644
index 0000000000..41d63ad6e7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/layer_filtering_transport.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/layer_filtering_transport.h"
+
+namespace webrtc {
+namespace test {
+
+LayerFilteringTransport::LayerFilteringTransport(
+    const FakeNetworkPipe::Config& config,
+    Call* send_call,
+    uint8_t vp8_video_payload_type,
+    uint8_t vp9_video_payload_type,
+    int selected_tl,
+    int selected_sl)
+    : test::DirectTransport(config, send_call),
+      vp8_video_payload_type_(vp8_video_payload_type),
+      vp9_video_payload_type_(vp9_video_payload_type),
+      selected_tl_(selected_tl),
+      selected_sl_(selected_sl),
+      discarded_last_packet_(false) {}
+
+bool LayerFilteringTransport::DiscardedLastPacket() const {
+  return discarded_last_packet_;
+}
+
+bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
+                                      size_t length,
+                                      const PacketOptions& options) {
+  if (selected_tl_ == -1 && selected_sl_ == -1) {
+    // Nothing to change, forward the packet immediately.
+    return test::DirectTransport::SendRtp(packet, length, options);
+  }
+
+  bool set_marker_bit = false;
+  RtpUtility::RtpHeaderParser parser(packet, length);
+  RTPHeader header;
+  parser.Parse(&header);
+
+  RTC_DCHECK_LE(length, static_cast<size_t>(IP_PACKET_SIZE));
+  uint8_t temp_buffer[IP_PACKET_SIZE];
+  memcpy(temp_buffer, packet, length);
+
+  if (header.payloadType == vp8_video_payload_type_ ||
+      header.payloadType == vp9_video_payload_type_) {
+    const uint8_t* payload = packet + header.headerLength;
+    RTC_DCHECK_GT(length, header.headerLength);
+    const size_t payload_length = length - header.headerLength;
+    RTC_DCHECK_GT(payload_length, header.paddingLength);
+    const size_t payload_data_length = payload_length - header.paddingLength;
+
+    const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
+    rtc::scoped_ptr<RtpDepacketizer> depacketizer(
+        RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
+    RtpDepacketizer::ParsedPayload parsed_payload;
+    if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
+      const int temporal_idx = static_cast<int>(
+          is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
+                 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
+      const int spatial_idx = static_cast<int>(
+          is_vp8 ? kNoSpatialIdx
+                 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
+      if (selected_sl_ >= 0 && spatial_idx == selected_sl_ &&
+          parsed_payload.type.Video.codecHeader.VP9.end_of_frame) {
+        // This layer is now the last in the superframe.
+        set_marker_bit = true;
+      } else if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
+                  temporal_idx > selected_tl_) ||
+                 (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
+                  spatial_idx > selected_sl_)) {
+        // Truncate packet to a padding packet.
+        length = header.headerLength + 1;
+        temp_buffer[0] |= (1 << 5);  // P = 1.
+        temp_buffer[1] &= 0x7F;      // M = 0.
+        discarded_last_packet_ = true;
+        temp_buffer[header.headerLength] = 1;  // One byte of padding.
+      }
+    } else {
+      RTC_NOTREACHED() << "Parse error";
+    }
+  }
+
+  // We are discarding some of the packets (specifically, whole layers), so
+  // make sure the marker bit is set properly, and that sequence numbers are
+  // continuous.
+  if (set_marker_bit)
+    temp_buffer[1] |= kRtpMarkerBitMask;
+
+  return test::DirectTransport::SendRtp(temp_buffer, length, options);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/layer_filtering_transport.h b/media/webrtc/trunk/webrtc/test/layer_filtering_transport.h
new file mode 100644
index 0000000000..d453556235
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/layer_filtering_transport.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_TEST_LAYER_FILTERING_TRANSPORT_H_
+#define WEBRTC_TEST_LAYER_FILTERING_TRANSPORT_H_
+
+#include "webrtc/call.h"
+#include "webrtc/test/direct_transport.h"
+#include "webrtc/test/fake_network_pipe.h"
+
+#include <map>
+
+namespace webrtc {
+
+namespace test {
+
+class LayerFilteringTransport : public test::DirectTransport {
+ public:
+  LayerFilteringTransport(const FakeNetworkPipe::Config& config,
+                          Call* send_call,
+                          uint8_t vp8_video_payload_type,
+                          uint8_t vp9_video_payload_type,
+                          int selected_tl,
+                          int selected_sl);
+  bool DiscardedLastPacket() const;
+  bool SendRtp(const uint8_t* data,
+               size_t length,
+               const PacketOptions& options) override;
+
+ private:
+  // Used to distinguish between VP8 and VP9.
+  const uint8_t vp8_video_payload_type_;
+  const uint8_t vp9_video_payload_type_;
+  // Discard or invalidate all temporal/spatial layers with id greater than the
+  // selected one. -1 to disable filtering.
+  const int selected_tl_;
+  const int selected_sl_;
+  bool discarded_last_packet_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_LAYER_FILTERING_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc b/media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc
index c0b6130c25..450c6bd8a5 100644
--- a/media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc
+++ b/media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc
@@ -143,7 +143,7 @@ void GlxRenderer::Resize(size_t width, size_t height) {
   XConfigureWindow(display_, window_, CWWidth | CWHeight, &wc);
 }
 
-void GlxRenderer::RenderFrame(const webrtc::I420VideoFrame& frame,
+void GlxRenderer::RenderFrame(const webrtc::VideoFrame& frame,
                               int /*render_delay_ms*/) {
   if (static_cast<size_t>(frame.width()) != width_ ||
       static_cast<size_t>(frame.height()) != height_) {
diff --git a/media/webrtc/trunk/webrtc/test/linux/glx_renderer.h b/media/webrtc/trunk/webrtc/test/linux/glx_renderer.h
index 6d9e4e6dd9..c117281cf1 100644
--- a/media/webrtc/trunk/webrtc/test/linux/glx_renderer.h
+++ b/media/webrtc/trunk/webrtc/test/linux/glx_renderer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_
+#ifndef WEBRTC_TEST_LINUX_GLX_RENDERER_H_
+#define WEBRTC_TEST_LINUX_GLX_RENDERER_H_
 
 #include <GL/glx.h>
 #include <X11/Xlib.h>
@@ -26,7 +26,7 @@ class GlxRenderer : public GlRenderer {
                              size_t height);
   virtual ~GlxRenderer();
 
-  void RenderFrame(const webrtc::I420VideoFrame& frame, int delta) override;
+  void RenderFrame(const webrtc::VideoFrame& frame, int delta) override;
   bool IsTextureSupported() const override { return false; }
 
  private:
@@ -45,4 +45,4 @@ class GlxRenderer : public GlRenderer {
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_
+#endif  // WEBRTC_TEST_LINUX_GLX_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/test/mac/run_test.mm b/media/webrtc/trunk/webrtc/test/mac/run_test.mm
index e8f38401c3..4e0093a9b6 100644
--- a/media/webrtc/trunk/webrtc/test/mac/run_test.mm
+++ b/media/webrtc/trunk/webrtc/test/mac/run_test.mm
@@ -65,7 +65,7 @@ void RunTest(void(*test)()) {
   NSRunLoop *runLoop = [NSRunLoop currentRunLoop];
   while ([testRunner running] &&
          [runLoop runMode:NSDefaultRunLoopMode
-               beforeDate:[NSDate distantFuture]]);
+               beforeDate:[NSDate dateWithTimeIntervalSinceNow:0.1]]);
 
   [testRunner release];
   [pool release];
diff --git a/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h b/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h
index 6237709f93..7baf794744 100644
--- a/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h
+++ b/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_
+#ifndef WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_
+#define WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/test/gl/gl_renderer.h"
@@ -27,15 +27,15 @@ class MacRenderer : public GlRenderer {
   bool Init(const char* window_title, int width, int height);
 
   // Implements GlRenderer.
-  void RenderFrame(const I420VideoFrame& frame, int delta) override;
+  void RenderFrame(const VideoFrame& frame, int delta) override;
   bool IsTextureSupported() const override { return false; }
 
  private:
   CocoaWindow* window_;
 
-  DISALLOW_COPY_AND_ASSIGN(MacRenderer);
+  RTC_DISALLOW_COPY_AND_ASSIGN(MacRenderer);
 };
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_
+#endif  // WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_
diff --git a/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm b/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm
index 1bc05a190e..9cde95a982 100644
--- a/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm
+++ b/media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm
@@ -125,7 +125,7 @@ bool MacRenderer::Init(const char* window_title, int width, int height) {
   return true;
 }
 
-void MacRenderer::RenderFrame(const I420VideoFrame& frame, int /*delta*/) {
+void MacRenderer::RenderFrame(const VideoFrame& frame, int /*delta*/) {
   [window_ makeCurrentContext];
   GlRenderer::RenderFrame(frame, 0);
 }
diff --git a/media/webrtc/trunk/webrtc/test/mock_transport.h b/media/webrtc/trunk/webrtc/test/mock_transport.h
index 5b1cb8dfb6..4937134512 100644
--- a/media/webrtc/trunk/webrtc/test/mock_transport.h
+++ b/media/webrtc/trunk/webrtc/test/mock_transport.h
@@ -16,12 +16,13 @@
 
 namespace webrtc {
 
-class MockTransport : public webrtc::Transport {
+class MockTransport : public Transport {
  public:
-  MOCK_METHOD3(SendPacket,
-      int(int channel, const void* data, size_t len));
-  MOCK_METHOD3(SendRTCPPacket,
-      int(int channel, const void* data, size_t len));
+  MOCK_METHOD3(SendRtp,
+               bool(const uint8_t* data,
+                    size_t len,
+                    const PacketOptions& options));
+  MOCK_METHOD2(SendRtcp, bool(const uint8_t* data, size_t len));
 };
 }  // namespace webrtc
 #endif  // WEBRTC_TEST_MOCK_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/test/mock_voe_channel_proxy.h b/media/webrtc/trunk/webrtc/test/mock_voe_channel_proxy.h
new file mode 100644
index 0000000000..b5d79c18ea
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/mock_voe_channel_proxy.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
+#define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
+
+#include <string>
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/voice_engine/channel_proxy.h"
+
+namespace webrtc {
+namespace test {
+
+class MockVoEChannelProxy : public voe::ChannelProxy {
+ public:
+  MOCK_METHOD1(SetRTCPStatus, void(bool enable));
+  MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
+  MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
+  MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
+  MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
+  MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
+  MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
+  MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
+  MOCK_METHOD3(SetCongestionControlObjects,
+               void(RtpPacketSender* rtp_packet_sender,
+                    TransportFeedbackObserver* transport_feedback_observer,
+                    PacketRouter* seq_num_allocator));
+  MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
+  MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
+  MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
+  MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
+  MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
+  MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
+  MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
+  MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event,
+                                               uint32_t duration_ms));
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
diff --git a/media/webrtc/trunk/webrtc/test/mock_voice_engine.h b/media/webrtc/trunk/webrtc/test/mock_voice_engine.h
new file mode 100644
index 0000000000..28a75f8063
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/mock_voice_engine.h
@@ -0,0 +1,337 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_
+#define WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/test/mock_voe_channel_proxy.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
+
+namespace webrtc {
+namespace test {
+
+// NOTE: This class inherits from VoiceEngineImpl so that its clients will be
+// able to get the various interfaces as usual, via T::GetInterface().
+class MockVoiceEngine : public VoiceEngineImpl {
+ public:
+  MockVoiceEngine() : VoiceEngineImpl(new Config(), true) {
+    // Increase ref count so this object isn't automatically deleted whenever
+    // interfaces are Release():d.
+    ++_ref_count;
+    // We add this default behavior to make the mock easier to use in tests. It
+    // will create a NiceMock of a voe::ChannelProxy.
+    ON_CALL(*this, ChannelProxyFactory(testing::_))
+        .WillByDefault(
+            testing::Invoke([](int channel_id) {
+              return new testing::NiceMock<MockVoEChannelProxy>();
+            }));
+  }
+  ~MockVoiceEngine() override {
+    // Decrease ref count before base class d-tor is called; otherwise it will
+    // trigger an assertion.
+    --_ref_count;
+  }
+  // Allows injecting a ChannelProxy factory.
+  MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id));
+
+  // VoiceEngineImpl
+  rtc::scoped_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id) override {
+    return rtc::scoped_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id));
+  }
+
+  // VoEAudioProcessing
+  MOCK_METHOD2(SetNsStatus, int(bool enable, NsModes mode));
+  MOCK_METHOD2(GetNsStatus, int(bool& enabled, NsModes& mode));
+  MOCK_METHOD2(SetAgcStatus, int(bool enable, AgcModes mode));
+  MOCK_METHOD2(GetAgcStatus, int(bool& enabled, AgcModes& mode));
+  MOCK_METHOD1(SetAgcConfig, int(AgcConfig config));
+  MOCK_METHOD1(GetAgcConfig, int(AgcConfig& config));
+  MOCK_METHOD2(SetEcStatus, int(bool enable, EcModes mode));
+  MOCK_METHOD2(GetEcStatus, int(bool& enabled, EcModes& mode));
+  MOCK_METHOD1(EnableDriftCompensation, int(bool enable));
+  MOCK_METHOD0(DriftCompensationEnabled, bool());
+  MOCK_METHOD1(SetDelayOffsetMs, void(int offset));
+  MOCK_METHOD0(DelayOffsetMs, int());
+  MOCK_METHOD2(SetAecmMode, int(AecmModes mode, bool enableCNG));
+  MOCK_METHOD2(GetAecmMode, int(AecmModes& mode, bool& enabledCNG));
+  MOCK_METHOD1(EnableHighPassFilter, int(bool enable));
+  MOCK_METHOD0(IsHighPassFilterEnabled, bool());
+  MOCK_METHOD3(SetRxNsStatus, int(int channel, bool enable, NsModes mode));
+  MOCK_METHOD3(GetRxNsStatus, int(int channel, bool& enabled, NsModes& mode));
+  MOCK_METHOD3(SetRxAgcStatus, int(int channel, bool enable, AgcModes mode));
+  MOCK_METHOD3(GetRxAgcStatus, int(int channel, bool& enabled, AgcModes& mode));
+  MOCK_METHOD2(SetRxAgcConfig, int(int channel, AgcConfig config));
+  MOCK_METHOD2(GetRxAgcConfig, int(int channel, AgcConfig& config));
+  MOCK_METHOD2(RegisterRxVadObserver,
+               int(int channel, VoERxVadCallback& observer));
+  MOCK_METHOD1(DeRegisterRxVadObserver, int(int channel));
+  MOCK_METHOD1(VoiceActivityIndicator, int(int channel));
+  MOCK_METHOD1(SetEcMetricsStatus, int(bool enable));
+  MOCK_METHOD1(GetEcMetricsStatus, int(bool& enabled));
+  MOCK_METHOD4(GetEchoMetrics, int(int& ERL, int& ERLE, int& RERL, int& A_NLP));
+  MOCK_METHOD3(GetEcDelayMetrics,
+               int(int& delay_median,
+                   int& delay_std,
+                   float& fraction_poor_delays));
+  MOCK_METHOD1(StartDebugRecording, int(const char* fileNameUTF8));
+  MOCK_METHOD1(StartDebugRecording, int(FILE* file_handle));
+  MOCK_METHOD0(StopDebugRecording, int());
+  MOCK_METHOD1(SetTypingDetectionStatus, int(bool enable));
+  MOCK_METHOD1(GetTypingDetectionStatus, int(bool& enabled));
+  MOCK_METHOD1(TimeSinceLastTyping, int(int& seconds));
+  MOCK_METHOD5(SetTypingDetectionParameters,
+               int(int timeWindow,
+                   int costPerTyping,
+                   int reportingThreshold,
+                   int penaltyDecay,
+                   int typeEventDelay));
+  MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable));
+  MOCK_METHOD0(IsStereoChannelSwappingEnabled, bool());
+
+  // VoEBase
+  MOCK_METHOD1(RegisterVoiceEngineObserver, int(VoiceEngineObserver& observer));
+  MOCK_METHOD0(DeRegisterVoiceEngineObserver, int());
+  MOCK_METHOD2(Init,
+               int(AudioDeviceModule* external_adm,
+                   AudioProcessing* audioproc));
+  MOCK_METHOD0(audio_processing, AudioProcessing*());
+  MOCK_METHOD0(Terminate, int());
+  MOCK_METHOD0(CreateChannel, int());
+  MOCK_METHOD1(CreateChannel, int(const Config& config));
+  MOCK_METHOD1(DeleteChannel, int(int channel));
+  MOCK_METHOD1(StartReceive, int(int channel));
+  MOCK_METHOD1(StopReceive, int(int channel));
+  MOCK_METHOD1(StartPlayout, int(int channel));
+  MOCK_METHOD1(StopPlayout, int(int channel));
+  MOCK_METHOD1(StartSend, int(int channel));
+  MOCK_METHOD1(StopSend, int(int channel));
+  MOCK_METHOD1(GetVersion, int(char version[1024]));
+  MOCK_METHOD0(LastError, int());
+  MOCK_METHOD0(audio_transport, AudioTransport*());
+  MOCK_METHOD2(AssociateSendChannel,
+               int(int channel, int accociate_send_channel));
+
+  // VoECodec
+  MOCK_METHOD0(NumOfCodecs, int());
+  MOCK_METHOD2(GetCodec, int(int index, CodecInst& codec));
+  MOCK_METHOD2(SetSendCodec, int(int channel, const CodecInst& codec));
+  MOCK_METHOD2(GetSendCodec, int(int channel, CodecInst& codec));
+  MOCK_METHOD2(SetBitRate, int(int channel, int bitrate_bps));
+  MOCK_METHOD2(GetRecCodec, int(int channel, CodecInst& codec));
+  MOCK_METHOD2(SetRecPayloadType, int(int channel, const CodecInst& codec));
+  MOCK_METHOD2(GetRecPayloadType, int(int channel, CodecInst& codec));
+  MOCK_METHOD3(SetSendCNPayloadType,
+               int(int channel, int type, PayloadFrequencies frequency));
+  MOCK_METHOD2(SetFECStatus, int(int channel, bool enable));
+  MOCK_METHOD2(GetFECStatus, int(int channel, bool& enabled));
+  MOCK_METHOD4(SetVADStatus,
+               int(int channel, bool enable, VadModes mode, bool disableDTX));
+  MOCK_METHOD4(
+      GetVADStatus,
+      int(int channel, bool& enabled, VadModes& mode, bool& disabledDTX));
+  MOCK_METHOD2(SetOpusMaxPlaybackRate, int(int channel, int frequency_hz));
+  MOCK_METHOD2(SetOpusDtx, int(int channel, bool enable_dtx));
+  MOCK_METHOD0(GetEventLog, RtcEventLog*());
+
+  // VoEDtmf
+  MOCK_METHOD5(SendTelephoneEvent,
+               int(int channel,
+                   int eventCode,
+                   bool outOfBand,
+                   int lengthMs,
+                   int attenuationDb));
+  MOCK_METHOD2(SetSendTelephoneEventPayloadType,
+               int(int channel, unsigned char type));
+  MOCK_METHOD2(GetSendTelephoneEventPayloadType,
+               int(int channel, unsigned char& type));
+  MOCK_METHOD2(SetDtmfFeedbackStatus, int(bool enable, bool directFeedback));
+  MOCK_METHOD2(GetDtmfFeedbackStatus, int(bool& enabled, bool& directFeedback));
+  MOCK_METHOD3(PlayDtmfTone,
+               int(int eventCode, int lengthMs, int attenuationDb));
+
+  // VoEExternalMedia
+  MOCK_METHOD3(RegisterExternalMediaProcessing,
+               int(int channel,
+                   ProcessingTypes type,
+                   VoEMediaProcess& processObject));
+  MOCK_METHOD2(DeRegisterExternalMediaProcessing,
+               int(int channel, ProcessingTypes type));
+  MOCK_METHOD3(GetAudioFrame,
+               int(int channel, int desired_sample_rate_hz, AudioFrame* frame));
+  MOCK_METHOD2(SetExternalMixing, int(int channel, bool enable));
+
+  // VoEFile
+  MOCK_METHOD7(StartPlayingFileLocally,
+               int(int channel,
+                   const char fileNameUTF8[1024],
+                   bool loop,
+                   FileFormats format,
+                   float volumeScaling,
+                   int startPointMs,
+                   int stopPointMs));
+  MOCK_METHOD6(StartPlayingFileLocally,
+               int(int channel,
+                   InStream* stream,
+                   FileFormats format,
+                   float volumeScaling,
+                   int startPointMs,
+                   int stopPointMs));
+  MOCK_METHOD1(StopPlayingFileLocally, int(int channel));
+  MOCK_METHOD1(IsPlayingFileLocally, int(int channel));
+  MOCK_METHOD6(StartPlayingFileAsMicrophone,
+               int(int channel,
+                   const char fileNameUTF8[1024],
+                   bool loop,
+                   bool mixWithMicrophone,
+                   FileFormats format,
+                   float volumeScaling));
+  MOCK_METHOD5(StartPlayingFileAsMicrophone,
+               int(int channel,
+                   InStream* stream,
+                   bool mixWithMicrophone,
+                   FileFormats format,
+                   float volumeScaling));
+  MOCK_METHOD1(StopPlayingFileAsMicrophone, int(int channel));
+  MOCK_METHOD1(IsPlayingFileAsMicrophone, int(int channel));
+  MOCK_METHOD4(StartRecordingPlayout,
+               int(int channel,
+                   const char* fileNameUTF8,
+                   CodecInst* compression,
+                   int maxSizeBytes));
+  MOCK_METHOD1(StopRecordingPlayout, int(int channel));
+  MOCK_METHOD3(StartRecordingPlayout,
+               int(int channel, OutStream* stream, CodecInst* compression));
+  MOCK_METHOD3(StartRecordingMicrophone,
+               int(const char* fileNameUTF8,
+                   CodecInst* compression,
+                   int maxSizeBytes));
+  MOCK_METHOD2(StartRecordingMicrophone,
+               int(OutStream* stream, CodecInst* compression));
+  MOCK_METHOD0(StopRecordingMicrophone, int());
+
+  // VoEHardware
+  MOCK_METHOD1(GetNumOfRecordingDevices, int(int& devices));
+  MOCK_METHOD1(GetNumOfPlayoutDevices, int(int& devices));
+  MOCK_METHOD3(GetRecordingDeviceName,
+               int(int index, char strNameUTF8[128], char strGuidUTF8[128]));
+  MOCK_METHOD3(GetPlayoutDeviceName,
+               int(int index, char strNameUTF8[128], char strGuidUTF8[128]));
+  MOCK_METHOD2(SetRecordingDevice,
+               int(int index, StereoChannel recordingChannel));
+  MOCK_METHOD1(SetPlayoutDevice, int(int index));
+  MOCK_METHOD1(SetAudioDeviceLayer, int(AudioLayers audioLayer));
+  MOCK_METHOD1(GetAudioDeviceLayer, int(AudioLayers& audioLayer));
+  MOCK_METHOD1(SetRecordingSampleRate, int(unsigned int samples_per_sec));
+  MOCK_CONST_METHOD1(RecordingSampleRate, int(unsigned int* samples_per_sec));
+  MOCK_METHOD1(SetPlayoutSampleRate, int(unsigned int samples_per_sec));
+  MOCK_CONST_METHOD1(PlayoutSampleRate, int(unsigned int* samples_per_sec));
+  MOCK_CONST_METHOD0(BuiltInAECIsAvailable, bool());
+  MOCK_METHOD1(EnableBuiltInAEC, int(bool enable));
+  MOCK_CONST_METHOD0(BuiltInAGCIsAvailable, bool());
+  MOCK_METHOD1(EnableBuiltInAGC, int(bool enable));
+  MOCK_CONST_METHOD0(BuiltInNSIsAvailable, bool());
+  MOCK_METHOD1(EnableBuiltInNS, int(bool enable));
+
+  // VoENetEqStats
+  MOCK_METHOD2(GetNetworkStatistics,
+               int(int channel, NetworkStatistics& stats));
+  MOCK_CONST_METHOD2(GetDecodingCallStatistics,
+                     int(int channel, AudioDecodingCallStats* stats));
+
+  // VoENetwork
+  MOCK_METHOD2(RegisterExternalTransport,
+               int(int channel, Transport& transport));
+  MOCK_METHOD1(DeRegisterExternalTransport, int(int channel));
+  MOCK_METHOD3(ReceivedRTPPacket,
+               int(int channel, const void* data, size_t length));
+  MOCK_METHOD4(ReceivedRTPPacket,
+               int(int channel,
+                   const void* data,
+                   size_t length,
+                   const PacketTime& packet_time));
+  MOCK_METHOD3(ReceivedRTCPPacket,
+               int(int channel, const void* data, size_t length));
+
+  // VoERTP_RTCP
+  MOCK_METHOD2(SetLocalSSRC, int(int channel, unsigned int ssrc));
+  MOCK_METHOD2(GetLocalSSRC, int(int channel, unsigned int& ssrc));
+  MOCK_METHOD2(GetRemoteSSRC, int(int channel, unsigned int& ssrc));
+  MOCK_METHOD3(SetSendAudioLevelIndicationStatus,
+               int(int channel, bool enable, unsigned char id));
+  MOCK_METHOD3(SetReceiveAudioLevelIndicationStatus,
+               int(int channel, bool enable, unsigned char id));
+  MOCK_METHOD3(SetSendAbsoluteSenderTimeStatus,
+               int(int channel, bool enable, unsigned char id));
+  MOCK_METHOD3(SetReceiveAbsoluteSenderTimeStatus,
+               int(int channel, bool enable, unsigned char id));
+  MOCK_METHOD2(SetRTCPStatus, int(int channel, bool enable));
+  MOCK_METHOD2(GetRTCPStatus, int(int channel, bool& enabled));
+  MOCK_METHOD2(SetRTCP_CNAME, int(int channel, const char cName[256]));
+  MOCK_METHOD2(GetRTCP_CNAME, int(int channel, char cName[256]));
+  MOCK_METHOD2(GetRemoteRTCP_CNAME, int(int channel, char cName[256]));
+  MOCK_METHOD7(GetRemoteRTCPData,
+               int(int channel,
+                   unsigned int& NTPHigh,
+                   unsigned int& NTPLow,
+                   unsigned int& timestamp,
+                   unsigned int& playoutTimestamp,
+                   unsigned int* jitter,
+                   unsigned short* fractionLost));
+  MOCK_METHOD4(GetRTPStatistics,
+               int(int channel,
+                   unsigned int& averageJitterMs,
+                   unsigned int& maxJitterMs,
+                   unsigned int& discardedPackets));
+  MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics& stats));
+  MOCK_METHOD2(GetRemoteRTCPReportBlocks,
+               int(int channel, std::vector<ReportBlock>* receive_blocks));
+  MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype));
+  MOCK_METHOD3(GetREDStatus,
+               int(int channel, bool& enable, int& redPayloadtype));
+  MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets));
+
+  // VoEVideoSync
+  MOCK_METHOD1(GetPlayoutBufferSize, int(int& buffer_ms));
+  MOCK_METHOD2(SetMinimumPlayoutDelay, int(int channel, int delay_ms));
+  MOCK_METHOD3(GetDelayEstimate,
+               int(int channel,
+                   int* jitter_buffer_delay_ms,
+                   int* playout_buffer_delay_ms));
+  MOCK_CONST_METHOD1(GetLeastRequiredDelayMs, int(int channel));
+  MOCK_METHOD2(SetInitTimestamp, int(int channel, unsigned int timestamp));
+  MOCK_METHOD2(SetInitSequenceNumber, int(int channel, short sequenceNumber));
+  MOCK_METHOD2(GetPlayoutTimestamp, int(int channel, unsigned int& timestamp));
+  MOCK_METHOD3(GetRtpRtcp,
+               int(int channel,
+                   RtpRtcp** rtpRtcpModule,
+                   RtpReceiver** rtp_receiver));
+
+  // VoEVolumeControl
+  MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
+  MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
+  MOCK_METHOD1(SetMicVolume, int(unsigned int volume));
+  MOCK_METHOD1(GetMicVolume, int(unsigned int& volume));
+  MOCK_METHOD2(SetInputMute, int(int channel, bool enable));
+  MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled));
+  MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level));
+  MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level));
+  MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level));
+  MOCK_METHOD2(GetSpeechOutputLevelFullRange,
+               int(int channel, unsigned& level));
+  MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling));
+  MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
+  MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right));
+  MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_
diff --git a/media/webrtc/trunk/webrtc/test/null_transport.cc b/media/webrtc/trunk/webrtc/test/null_transport.cc
index 3cba6386cd..7fa36d1246 100644
--- a/media/webrtc/trunk/webrtc/test/null_transport.cc
+++ b/media/webrtc/trunk/webrtc/test/null_transport.cc
@@ -12,7 +12,9 @@
 namespace webrtc {
 namespace test {
 
-bool NullTransport::SendRtp(const uint8_t* packet, size_t length) {
+bool NullTransport::SendRtp(const uint8_t* packet,
+                            size_t length,
+                            const PacketOptions& options) {
   return true;
 }
 
diff --git a/media/webrtc/trunk/webrtc/test/null_transport.h b/media/webrtc/trunk/webrtc/test/null_transport.h
index b80f751b1e..c49883e1dc 100644
--- a/media/webrtc/trunk/webrtc/test/null_transport.h
+++ b/media/webrtc/trunk/webrtc/test/null_transport.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
+#ifndef WEBRTC_TEST_NULL_TRANSPORT_H_
+#define WEBRTC_TEST_NULL_TRANSPORT_H_
 
 #include "webrtc/transport.h"
 
@@ -17,12 +17,14 @@ namespace webrtc {
 class PacketReceiver;
 
 namespace test {
-class NullTransport : public newapi::Transport {
+class NullTransport : public Transport {
  public:
-  bool SendRtp(const uint8_t* packet, size_t length) override;
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override;
   bool SendRtcp(const uint8_t* packet, size_t length) override;
 };
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
+#endif  // WEBRTC_TEST_NULL_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc b/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc
index b9e430fe9f..8ce249e0b6 100644
--- a/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc
+++ b/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc
@@ -15,6 +15,8 @@
 namespace webrtc {
 namespace test {
 
+using namespace RTCPUtility;
+
 RtcpPacketParser::RtcpPacketParser() {}
 
 RtcpPacketParser::~RtcpPacketParser() {}
@@ -24,98 +26,97 @@ void RtcpPacketParser::Parse(const void *data, size_t len) {
   RTCPUtility::RTCPParserV2 parser(packet, len, true);
   EXPECT_TRUE(parser.IsValid());
   for (RTCPUtility::RTCPPacketTypes type = parser.Begin();
-      type != RTCPUtility::kRtcpNotValidCode;
-      type = parser.Iterate()) {
+       type != RTCPPacketTypes::kInvalid; type = parser.Iterate()) {
     switch (type) {
-      case RTCPUtility::kRtcpSrCode:
+      case RTCPPacketTypes::kSr:
         sender_report_.Set(parser.Packet().SR);
         break;
-      case RTCPUtility::kRtcpRrCode:
+      case RTCPPacketTypes::kRr:
         receiver_report_.Set(parser.Packet().RR);
         break;
-      case RTCPUtility::kRtcpReportBlockItemCode:
+      case RTCPPacketTypes::kReportBlockItem:
         report_block_.Set(parser.Packet().ReportBlockItem);
         ++report_blocks_per_ssrc_[parser.Packet().ReportBlockItem.SSRC];
         break;
-      case RTCPUtility::kRtcpSdesCode:
+      case RTCPPacketTypes::kSdes:
         sdes_.Set();
         break;
-      case RTCPUtility::kRtcpSdesChunkCode:
+      case RTCPPacketTypes::kSdesChunk:
         sdes_chunk_.Set(parser.Packet().CName);
         break;
-      case RTCPUtility::kRtcpByeCode:
+      case RTCPPacketTypes::kBye:
         bye_.Set(parser.Packet().BYE);
         break;
-      case RTCPUtility::kRtcpAppCode:
+      case RTCPPacketTypes::kApp:
         app_.Set(parser.Packet().APP);
         break;
-      case RTCPUtility::kRtcpAppItemCode:
+      case RTCPPacketTypes::kAppItem:
         app_item_.Set(parser.Packet().APP);
         break;
-      case RTCPUtility::kRtcpExtendedIjCode:
+      case RTCPPacketTypes::kExtendedIj:
         ij_.Set();
         break;
-      case RTCPUtility::kRtcpExtendedIjItemCode:
+      case RTCPPacketTypes::kExtendedIjItem:
         ij_item_.Set(parser.Packet().ExtendedJitterReportItem);
         break;
-      case RTCPUtility::kRtcpPsfbPliCode:
+      case RTCPPacketTypes::kPsfbPli:
         pli_.Set(parser.Packet().PLI);
         break;
-      case RTCPUtility::kRtcpPsfbSliCode:
+      case RTCPPacketTypes::kPsfbSli:
         sli_.Set(parser.Packet().SLI);
         break;
-      case RTCPUtility::kRtcpPsfbSliItemCode:
+      case RTCPPacketTypes::kPsfbSliItem:
         sli_item_.Set(parser.Packet().SLIItem);
         break;
-      case RTCPUtility::kRtcpPsfbRpsiCode:
+      case RTCPPacketTypes::kPsfbRpsi:
         rpsi_.Set(parser.Packet().RPSI);
         break;
-      case RTCPUtility::kRtcpPsfbFirCode:
+      case RTCPPacketTypes::kPsfbFir:
         fir_.Set(parser.Packet().FIR);
         break;
-      case RTCPUtility::kRtcpPsfbFirItemCode:
+      case RTCPPacketTypes::kPsfbFirItem:
         fir_item_.Set(parser.Packet().FIRItem);
         break;
-      case RTCPUtility::kRtcpRtpfbNackCode:
+      case RTCPPacketTypes::kRtpfbNack:
         nack_.Set(parser.Packet().NACK);
         nack_item_.Clear();
         break;
-      case RTCPUtility::kRtcpRtpfbNackItemCode:
+      case RTCPPacketTypes::kRtpfbNackItem:
         nack_item_.Set(parser.Packet().NACKItem);
         break;
-      case RTCPUtility::kRtcpPsfbAppCode:
+      case RTCPPacketTypes::kPsfbApp:
         psfb_app_.Set(parser.Packet().PSFBAPP);
         break;
-      case RTCPUtility::kRtcpPsfbRembItemCode:
+      case RTCPPacketTypes::kPsfbRembItem:
         remb_item_.Set(parser.Packet().REMBItem);
         break;
-      case RTCPUtility::kRtcpRtpfbTmmbrCode:
+      case RTCPPacketTypes::kRtpfbTmmbr:
         tmmbr_.Set(parser.Packet().TMMBR);
         break;
-      case RTCPUtility::kRtcpRtpfbTmmbrItemCode:
+      case RTCPPacketTypes::kRtpfbTmmbrItem:
         tmmbr_item_.Set(parser.Packet().TMMBRItem);
         break;
-      case RTCPUtility::kRtcpRtpfbTmmbnCode:
+      case RTCPPacketTypes::kRtpfbTmmbn:
         tmmbn_.Set(parser.Packet().TMMBN);
         tmmbn_items_.Clear();
         break;
-      case RTCPUtility::kRtcpRtpfbTmmbnItemCode:
+      case RTCPPacketTypes::kRtpfbTmmbnItem:
         tmmbn_items_.Set(parser.Packet().TMMBNItem);
         break;
-      case RTCPUtility::kRtcpXrHeaderCode:
+      case RTCPPacketTypes::kXrHeader:
         xr_header_.Set(parser.Packet().XR);
         dlrr_items_.Clear();
         break;
-      case RTCPUtility::kRtcpXrReceiverReferenceTimeCode:
+      case RTCPPacketTypes::kXrReceiverReferenceTime:
         rrtr_.Set(parser.Packet().XRReceiverReferenceTimeItem);
         break;
-      case RTCPUtility::kRtcpXrDlrrReportBlockCode:
+      case RTCPPacketTypes::kXrDlrrReportBlock:
         dlrr_.Set();
         break;
-      case RTCPUtility::kRtcpXrDlrrReportBlockItemCode:
+      case RTCPPacketTypes::kXrDlrrReportBlockItem:
         dlrr_items_.Set(parser.Packet().XRDLRRReportBlockItem);
         break;
-      case RTCPUtility::kRtcpXrVoipMetricCode:
+      case RTCPPacketTypes::kXrVoipMetric:
         voip_metric_.Set(parser.Packet().XRVOIPMetricItem);
         break;
       default:
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc b/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc
index 26151bba1f..1413f00797 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc
@@ -69,7 +69,8 @@ bool ReadUint16(uint16_t* out, FILE* file) {
 
 class RtpFileReaderImpl : public RtpFileReader {
  public:
-  virtual bool Init(const std::string& filename) = 0;
+  virtual bool Init(const std::string& filename,
+                    const std::set<uint32_t>& ssrc_filter) = 0;
 };
 
 class InterleavedRtpFileReader : public RtpFileReaderImpl {
@@ -81,7 +82,8 @@ class InterleavedRtpFileReader : public RtpFileReaderImpl {
     }
   }
 
-  virtual bool Init(const std::string& filename) {
+  virtual bool Init(const std::string& filename,
+                    const std::set<uint32_t>& ssrc_filter) {
     file_ = fopen(filename.c_str(), "rb");
     if (file_ == NULL) {
       printf("ERROR: Can't open file: %s\n", filename.c_str());
@@ -127,7 +129,8 @@ class RtpDumpReader : public RtpFileReaderImpl {
     }
   }
 
-  bool Init(const std::string& filename) {
+  bool Init(const std::string& filename,
+            const std::set<uint32_t>& ssrc_filter) {
     file_ = fopen(filename.c_str(), "rb");
     if (file_ == NULL) {
       printf("ERROR: Can't open file: %s\n", filename.c_str());
@@ -200,7 +203,7 @@ class RtpDumpReader : public RtpFileReaderImpl {
  private:
   FILE* file_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
 };
 
 enum {
@@ -265,11 +268,13 @@ class PcapReader : public RtpFileReaderImpl {
     }
   }
 
-  bool Init(const std::string& filename) override {
-    return Initialize(filename) == kResultSuccess;
+  bool Init(const std::string& filename,
+            const std::set<uint32_t>& ssrc_filter) override {
+    return Initialize(filename, ssrc_filter) == kResultSuccess;
   }
 
-  int Initialize(const std::string& filename) {
+  int Initialize(const std::string& filename,
+                 const std::set<uint32_t>& ssrc_filter) {
     file_ = fopen(filename.c_str(), "rb");
     if (file_ == NULL) {
       printf("ERROR: Can't open file: %s\n", filename.c_str());
@@ -286,7 +291,7 @@ class PcapReader : public RtpFileReaderImpl {
     for (;;) {
       TRY_PCAP(fseek(file_, next_packet_pos, SEEK_SET));
       int result = ReadPacket(&next_packet_pos, stream_start_ms,
-                              ++total_packet_count);
+                              ++total_packet_count, ssrc_filter);
       if (result == kResultFail) {
         break;
       } else if (result == kResultSuccess && packets_.size() == 1) {
@@ -308,10 +313,10 @@ class PcapReader : public RtpFileReaderImpl {
     for (SsrcMapIterator mit = packets_by_ssrc_.begin();
         mit != packets_by_ssrc_.end(); ++mit) {
       uint32_t ssrc = mit->first;
-      const std::vector<uint32_t>& packet_numbers = mit->second;
-      uint8_t pt = packets_[packet_numbers[0]].rtp_header.payloadType;
+      const std::vector<uint32_t>& packet_indices = mit->second;
+      uint8_t pt = packets_[packet_indices[0]].rtp_header.payloadType;
       printf("SSRC: %08x, %" PRIuS " packets, pt=%d\n", ssrc,
-             packet_numbers.size(), pt);
+             packet_indices.size(), pt);
     }
 
     // TODO(solenberg): Better validation of identified SSRC streams.
@@ -419,8 +424,10 @@ class PcapReader : public RtpFileReaderImpl {
     return kResultSuccess;
   }
 
-  int ReadPacket(int32_t* next_packet_pos, uint32_t stream_start_ms,
-                 uint32_t number) {
+  int ReadPacket(int32_t* next_packet_pos,
+                 uint32_t stream_start_ms,
+                 uint32_t number,
+                 const std::set<uint32_t>& ssrc_filter) {
     assert(next_packet_pos);
 
     uint32_t ts_sec;    // Timestamp seconds.
@@ -451,14 +458,19 @@ class PcapReader : public RtpFileReaderImpl {
       rtp_parser.ParseRtcp(&marker.rtp_header);
       packets_.push_back(marker);
     } else {
-      if (!rtp_parser.Parse(marker.rtp_header, NULL)) {
+      if (!rtp_parser.Parse(&marker.rtp_header, nullptr)) {
         DEBUG_LOG("Not recognized as RTP/RTCP");
         return kResultSkip;
       }
 
       uint32_t ssrc = marker.rtp_header.ssrc;
-      packets_by_ssrc_[ssrc].push_back(marker.packet_number);
-      packets_.push_back(marker);
+      if (ssrc_filter.empty() || ssrc_filter.find(ssrc) != ssrc_filter.end()) {
+        packets_by_ssrc_[ssrc].push_back(
+            static_cast<uint32_t>(packets_.size()));
+        packets_.push_back(marker);
+      } else {
+        return kResultSkip;
+      }
     }
 
     return kResultSuccess;
@@ -628,11 +640,12 @@ class PcapReader : public RtpFileReaderImpl {
   std::vector<RtpPacketMarker> packets_;
   PacketIterator next_packet_it_;
 
-  DISALLOW_COPY_AND_ASSIGN(PcapReader);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PcapReader);
 };
 
 RtpFileReader* RtpFileReader::Create(FileFormat format,
-                                     const std::string& filename) {
+                                     const std::string& filename,
+                                     const std::set<uint32_t>& ssrc_filter) {
   RtpFileReaderImpl* reader = NULL;
   switch (format) {
     case kPcap:
@@ -645,12 +658,17 @@ RtpFileReader* RtpFileReader::Create(FileFormat format,
       reader = new InterleavedRtpFileReader();
       break;
   }
-  if (!reader->Init(filename)) {
+  if (!reader->Init(filename, ssrc_filter)) {
     delete reader;
     return NULL;
   }
   return reader;
 }
 
+RtpFileReader* RtpFileReader::Create(FileFormat format,
+                                     const std::string& filename) {
+  return RtpFileReader::Create(format, filename, std::set<uint32_t>());
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_reader.h b/media/webrtc/trunk/webrtc/test/rtp_file_reader.h
index c302d4fbde..c132d318fd 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_reader.h
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_reader.h
@@ -10,6 +10,7 @@
 #ifndef WEBRTC_TEST_RTP_FILE_READER_H_
 #define WEBRTC_TEST_RTP_FILE_READER_H_
 
+#include <set>
 #include <string>
 
 #include "webrtc/common_types.h"
@@ -37,6 +38,9 @@ class RtpFileReader {
   virtual ~RtpFileReader() {}
   static RtpFileReader* Create(FileFormat format,
                                const std::string& filename);
+  static RtpFileReader* Create(FileFormat format,
+                               const std::string& filename,
+                               const std::set<uint32_t>& ssrc_filter);
 
   virtual bool NextPacket(RtpPacket* packet) = 0;
 };
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc b/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc
index 929813f999..15a456ccf6 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc
@@ -85,7 +85,8 @@ class TestPcapFileReader : public ::testing::Test {
     while (rtp_packet_source_->NextPacket(&packet)) {
       RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
       webrtc::RTPHeader header;
-      if (!rtp_header_parser.RTCP() && rtp_header_parser.Parse(header, NULL)) {
+      if (!rtp_header_parser.RTCP() &&
+          rtp_header_parser.Parse(&header, nullptr)) {
         pps[header.ssrc]++;
       }
     }
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc b/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc
index 4acaa27815..d9e0586468 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc
@@ -28,7 +28,7 @@ static const char kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
 class RtpDumpWriter : public RtpFileWriter {
  public:
   explicit RtpDumpWriter(FILE* file) : file_(file) {
-    CHECK(file_ != NULL);
+    RTC_CHECK(file_ != NULL);
     Init();
   }
   virtual ~RtpDumpWriter() {
@@ -40,12 +40,11 @@ class RtpDumpWriter : public RtpFileWriter {
 
   bool WritePacket(const RtpPacket* packet) override {
     uint16_t len = static_cast<uint16_t>(packet->length + kPacketHeaderSize);
-    CHECK_GE(packet->original_length, packet->length);
     uint16_t plen = static_cast<uint16_t>(packet->original_length);
     uint32_t offset = packet->time_ms;
-    CHECK(WriteUint16(len));
-    CHECK(WriteUint16(plen));
-    CHECK(WriteUint32(offset));
+    RTC_CHECK(WriteUint16(len));
+    RTC_CHECK(WriteUint16(plen));
+    RTC_CHECK(WriteUint32(offset));
     return fwrite(packet->data, sizeof(uint8_t), packet->length, file_) ==
            packet->length;
   }
@@ -54,11 +53,11 @@ class RtpDumpWriter : public RtpFileWriter {
   bool Init() {
     fprintf(file_, "%s", kFirstLine);
 
-    CHECK(WriteUint32(0));
-    CHECK(WriteUint32(0));
-    CHECK(WriteUint32(0));
-    CHECK(WriteUint16(0));
-    CHECK(WriteUint16(0));
+    RTC_CHECK(WriteUint32(0));
+    RTC_CHECK(WriteUint32(0));
+    RTC_CHECK(WriteUint32(0));
+    RTC_CHECK(WriteUint16(0));
+    RTC_CHECK(WriteUint16(0));
 
     return true;
   }
@@ -87,7 +86,7 @@ class RtpDumpWriter : public RtpFileWriter {
 
   FILE* file_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
 };
 
 RtpFileWriter* RtpFileWriter::Create(FileFormat format,
diff --git a/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h b/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
index dd731f04db..5eb88d3f0d 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
+++ b/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
@@ -7,15 +7,18 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
+#ifndef WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
+#define WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
 
 #include <map>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/test/constants.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_send_stream.h"
@@ -23,161 +26,114 @@
 namespace webrtc {
 namespace test {
 
+class PacketTransport;
+
 class RtpRtcpObserver {
  public:
-  virtual ~RtpRtcpObserver() {}
-  newapi::Transport* SendTransport() {
-    return &send_transport_;
-  }
-
-  newapi::Transport* ReceiveTransport() {
-    return &receive_transport_;
-  }
-
-  virtual void SetReceivers(PacketReceiver* send_transport_receiver,
-                            PacketReceiver* receive_transport_receiver) {
-    send_transport_.SetReceiver(send_transport_receiver);
-    receive_transport_.SetReceiver(receive_transport_receiver);
-  }
-
-  void StopSending() {
-    send_transport_.StopSending();
-    receive_transport_.StopSending();
-  }
-
-  virtual EventTypeWrapper Wait() {
-    EventTypeWrapper result = observation_complete_->Wait(timeout_ms_);
-    return result;
-  }
-
- protected:
-  RtpRtcpObserver(unsigned int event_timeout_ms,
-      const FakeNetworkPipe::Config& configuration)
-      : crit_(CriticalSectionWrapper::CreateCriticalSection()),
-        observation_complete_(EventWrapper::Create()),
-        parser_(RtpHeaderParser::Create()),
-        send_transport_(crit_.get(),
-                        this,
-                        &RtpRtcpObserver::OnSendRtp,
-                        &RtpRtcpObserver::OnSendRtcp,
-                        configuration),
-        receive_transport_(crit_.get(),
-                           this,
-                           &RtpRtcpObserver::OnReceiveRtp,
-                           &RtpRtcpObserver::OnReceiveRtcp,
-                           configuration),
-        timeout_ms_(event_timeout_ms) {}
-
-  explicit RtpRtcpObserver(unsigned int event_timeout_ms)
-      : crit_(CriticalSectionWrapper::CreateCriticalSection()),
-        observation_complete_(EventWrapper::Create()),
-        parser_(RtpHeaderParser::Create()),
-        send_transport_(crit_.get(),
-                        this,
-                        &RtpRtcpObserver::OnSendRtp,
-                        &RtpRtcpObserver::OnSendRtcp,
-                        FakeNetworkPipe::Config()),
-        receive_transport_(crit_.get(),
-                           this,
-                           &RtpRtcpObserver::OnReceiveRtp,
-                           &RtpRtcpObserver::OnReceiveRtcp,
-                           FakeNetworkPipe::Config()),
-        timeout_ms_(event_timeout_ms) {}
-
   enum Action {
     SEND_PACKET,
     DROP_PACKET,
   };
 
-  virtual Action OnSendRtp(const uint8_t* packet, size_t length)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+  virtual ~RtpRtcpObserver() {}
+
+  virtual bool Wait() { return observation_complete_.Wait(timeout_ms_); }
+
+  virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
     return SEND_PACKET;
   }
 
-  virtual Action OnSendRtcp(const uint8_t* packet, size_t length)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+  virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
     return SEND_PACKET;
   }
 
-  virtual Action OnReceiveRtp(const uint8_t* packet, size_t length)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+  virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
     return SEND_PACKET;
   }
 
-  virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+  virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
     return SEND_PACKET;
   }
 
- private:
-  class PacketTransport : public test::DirectTransport {
-   public:
-    typedef Action (RtpRtcpObserver::*PacketTransportAction)(const uint8_t*,
-                                                             size_t);
-
-    PacketTransport(CriticalSectionWrapper* lock,
-                    RtpRtcpObserver* observer,
-                    PacketTransportAction on_rtp,
-                    PacketTransportAction on_rtcp,
-                    const FakeNetworkPipe::Config& configuration)
-        : test::DirectTransport(configuration),
-          crit_(lock),
-          observer_(observer),
-          on_rtp_(on_rtp),
-          on_rtcp_(on_rtcp) {}
-
-  private:
-   bool SendRtp(const uint8_t* packet, size_t length) override {
-      EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
-      Action action;
-      {
-        CriticalSectionScoped lock(crit_);
-        action = (observer_->*on_rtp_)(packet, length);
-      }
-      switch (action) {
-        case DROP_PACKET:
-          // Drop packet silently.
-          return true;
-        case SEND_PACKET:
-          return test::DirectTransport::SendRtp(packet, length);
-      }
-      return true;  // Will never happen, makes compiler happy.
-    }
-
-    bool SendRtcp(const uint8_t* packet, size_t length) override {
-      EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
-      Action action;
-      {
-        CriticalSectionScoped lock(crit_);
-        action = (observer_->*on_rtcp_)(packet, length);
-      }
-      switch (action) {
-        case DROP_PACKET:
-          // Drop packet silently.
-          return true;
-        case SEND_PACKET:
-          return test::DirectTransport::SendRtcp(packet, length);
-      }
-      return true;  // Will never happen, makes compiler happy.
-    }
-
-    // Pointer to shared lock instance protecting on_rtp_/on_rtcp_ calls.
-    CriticalSectionWrapper* const crit_;
-
-    RtpRtcpObserver* const observer_;
-    const PacketTransportAction on_rtp_, on_rtcp_;
-  };
-
  protected:
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-  const rtc::scoped_ptr<EventWrapper> observation_complete_;
+  explicit RtpRtcpObserver(int event_timeout_ms)
+      : observation_complete_(false, false),
+        parser_(RtpHeaderParser::Create()),
+        timeout_ms_(event_timeout_ms) {
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
+                                        kTOffsetExtensionId);
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                        kAbsSendTimeExtensionId);
+    parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                        kTransportSequenceNumberExtensionId);
+  }
+
+  rtc::Event observation_complete_;
   const rtc::scoped_ptr<RtpHeaderParser> parser_;
 
  private:
-  PacketTransport send_transport_, receive_transport_;
-  unsigned int timeout_ms_;
+  const int timeout_ms_;
+};
+
+class PacketTransport : public test::DirectTransport {
+ public:
+  enum TransportType { kReceiver, kSender };
+
+  PacketTransport(Call* send_call,
+                  RtpRtcpObserver* observer,
+                  TransportType transport_type,
+                  const FakeNetworkPipe::Config& configuration)
+      : test::DirectTransport(configuration, send_call),
+        observer_(observer),
+        transport_type_(transport_type) {}
+
+ private:
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override {
+    EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
+    RtpRtcpObserver::Action action;
+    {
+      if (transport_type_ == kSender) {
+        action = observer_->OnSendRtp(packet, length);
+      } else {
+        action = observer_->OnReceiveRtp(packet, length);
+      }
+    }
+    switch (action) {
+      case RtpRtcpObserver::DROP_PACKET:
+        // Drop packet silently.
+        return true;
+      case RtpRtcpObserver::SEND_PACKET:
+        return test::DirectTransport::SendRtp(packet, length, options);
+    }
+    return true;  // Will never happen, makes compiler happy.
+  }
+
+  bool SendRtcp(const uint8_t* packet, size_t length) override {
+    EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
+    RtpRtcpObserver::Action action;
+    {
+      if (transport_type_ == kSender) {
+        action = observer_->OnSendRtcp(packet, length);
+      } else {
+        action = observer_->OnReceiveRtcp(packet, length);
+      }
+    }
+    switch (action) {
+      case RtpRtcpObserver::DROP_PACKET:
+        // Drop packet silently.
+        return true;
+      case RtpRtcpObserver::SEND_PACKET:
+        return test::DirectTransport::SendRtcp(packet, length);
+    }
+    return true;  // Will never happen, makes compiler happy.
+  }
+
+  RtpRtcpObserver* const observer_;
+  TransportType transport_type_;
 };
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
+#endif  // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
diff --git a/media/webrtc/trunk/webrtc/test/run_loop.h b/media/webrtc/trunk/webrtc/test/run_loop.h
index 31012525e2..238e2dc282 100644
--- a/media/webrtc/trunk/webrtc/test/run_loop.h
+++ b/media/webrtc/trunk/webrtc/test/run_loop.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_
+#ifndef WEBRTC_TEST_RUN_LOOP_H_
+#define WEBRTC_TEST_RUN_LOOP_H_
 
 namespace webrtc {
 namespace test {
@@ -19,4 +19,4 @@ void PressEnterToContinue();
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_
+#endif  // WEBRTC_TEST_RUN_LOOP_H_
diff --git a/media/webrtc/trunk/webrtc/test/statistics.h b/media/webrtc/trunk/webrtc/test/statistics.h
index 0fc3a04ea9..d4a111e061 100644
--- a/media/webrtc/trunk/webrtc/test/statistics.h
+++ b/media/webrtc/trunk/webrtc/test/statistics.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_
+#ifndef WEBRTC_TEST_STATISTICS_H_
+#define WEBRTC_TEST_STATISTICS_H_
 
 #include "webrtc/typedefs.h"
 
@@ -33,4 +33,4 @@ class Statistics {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_
+#endif  // WEBRTC_TEST_STATISTICS_H_
diff --git a/media/webrtc/trunk/webrtc/test/test.gyp b/media/webrtc/trunk/webrtc/test/test.gyp
index ea831a1471..5bb7793842 100644
--- a/media/webrtc/trunk/webrtc/test/test.gyp
+++ b/media/webrtc/trunk/webrtc/test/test.gyp
@@ -22,7 +22,7 @@
       ],
       'sources': [
         'channel_transport/channel_transport.cc',
-        'channel_transport/include/channel_transport.h',
+        'channel_transport/channel_transport.h',
         'channel_transport/traffic_control_win.cc',
         'channel_transport/traffic_control_win.h',
         'channel_transport/udp_socket_manager_posix.cc',
@@ -41,11 +41,29 @@
         'channel_transport/udp_transport_impl.cc',
         'channel_transport/udp_transport_impl.h',
       ],
+      'conditions': [
+        ['OS=="win" and clang==1', {
+          'msvs_settings': {
+            'VCCLCompilerTool': {
+              'AdditionalOptions': [
+                # Disable warnings failing when compiling with Clang on Windows.
+                # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                '-Wno-parentheses-equality',
+                '-Wno-reorder',
+                '-Wno-tautological-constant-out-of-range-compare',
+                '-Wno-unused-private-field',
+              ],
+            },
+          },
+        }],
+      ],  # conditions.
     },
     {
-      'target_name': 'frame_generator',
+      'target_name': 'fake_video_frames',
       'type': 'static_library',
       'sources': [
+        'fake_texture_frame.cc',
+        'fake_texture_frame.h',
         'frame_generator.cc',
         'frame_generator.h',
       ],
@@ -79,6 +97,7 @@
       ],
       'dependencies': [
         '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
     },
@@ -103,6 +122,7 @@
       'dependencies': [
         'field_trial',
         'histogram',
+        'test_support',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
       ],
@@ -123,7 +143,7 @@
         'testsupport/frame_reader.h',
         'testsupport/frame_writer.cc',
         'testsupport/frame_writer.h',
-        'testsupport/gtest_disable.h',
+        'testsupport/iosfileutils.mm',
         'testsupport/mock/mock_frame_reader.h',
         'testsupport/mock/mock_frame_writer.h',
         'testsupport/packet_reader.cc',
@@ -133,6 +153,18 @@
         'testsupport/trace_to_stderr.cc',
         'testsupport/trace_to_stderr.h',
       ],
+      'conditions': [
+        ['OS=="ios"', {
+          'xcode_settings': {
+            'CLANG_ENABLE_OBJC_ARC': 'YES',
+          },
+        }],
+        ['use_x11==1', {
+          'dependencies': [
+            '<(DEPTH)/tools/xdisplaycheck/xdisplaycheck.gyp:xdisplaycheck',
+          ],
+        }],
+      ],
     },
     {
       # Depend on this target when you want to have test_support but also the
diff --git a/media/webrtc/trunk/webrtc/test/test_main.cc b/media/webrtc/trunk/webrtc/test/test_main.cc
index 733831f5be..a435575f88 100644
--- a/media/webrtc/trunk/webrtc/test/test_main.cc
+++ b/media/webrtc/trunk/webrtc/test/test_main.cc
@@ -10,6 +10,7 @@
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/test/field_trial.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -21,6 +22,11 @@ DEFINE_string(force_fieldtrials, "",
 int main(int argc, char* argv[]) {
   ::testing::InitGoogleTest(&argc, argv);
 
+  // Default to LS_INFO, even for release builds to provide better test logging.
+  // TODO(pbos): Consider adding a command-line override.
+  if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO)
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+
   // AllowCommandLineParsing allows us to ignore flags passed on to us by
   // Chromium build bots without having to explicitly disable them.
   google::AllowCommandLineReparsing();
diff --git a/media/webrtc/trunk/webrtc/test/test_suite.cc b/media/webrtc/trunk/webrtc/test/test_suite.cc
index e88b0301a2..2900f0eddb 100644
--- a/media/webrtc/trunk/webrtc/test/test_suite.cc
+++ b/media/webrtc/trunk/webrtc/test/test_suite.cc
@@ -13,6 +13,7 @@
 #include "gflags/gflags.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 #include "webrtc/test/field_trial.h"
@@ -49,6 +50,7 @@ int TestSuite::Run() {
 }
 
 void TestSuite::Initialize() {
+  rtc::LogMessage::SetLogToStderr(FLAGS_logs);
   if (FLAGS_logs)
     trace_to_stderr_.reset(new TraceToStderr);
 }
diff --git a/media/webrtc/trunk/webrtc/test/test_suite.h b/media/webrtc/trunk/webrtc/test/test_suite.h
index c166f283a5..dab2acd388 100644
--- a/media/webrtc/trunk/webrtc/test/test_suite.h
+++ b/media/webrtc/trunk/webrtc/test/test_suite.h
@@ -38,7 +38,7 @@ class TestSuite {
   virtual void Initialize();
   virtual void Shutdown();
 
-  DISALLOW_COPY_AND_ASSIGN(TestSuite);
+  RTC_DISALLOW_COPY_AND_ASSIGN(TestSuite);
 
  private:
   rtc::scoped_ptr<TraceToStderr> trace_to_stderr_;
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc b/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc
index 8301e77165..15abf5c517 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc
@@ -18,7 +18,7 @@
 #include <windows.h>
 #include <algorithm>
 
-#include "webrtc/system_wrappers/interface/utf_util_win.h"
+#include "webrtc/system_wrappers/include/utf_util_win.h"
 #define GET_CURRENT_DIR _getcwd
 #else
 #include <unistd.h>
@@ -41,6 +41,11 @@
 namespace webrtc {
 namespace test {
 
+#if defined(WEBRTC_IOS)
+// Defined in iosfileutils.mm.  No header file to discourage use elsewhere.
+std::string IOSResourcePath(std::string name, std::string extension);
+#endif
+
 namespace {
 
 #ifdef WIN32
@@ -57,7 +62,9 @@ const char* kProjectRootFileName = "DEPS";
 const char* kOutputDirName = "out";
 const char* kFallbackPath = "./";
 #endif
+#if !defined(WEBRTC_IOS)
 const char* kResourcesDirName = "resources";
+#endif
 
 char relative_dir_path[FILENAME_MAX];
 bool relative_dir_path_set = false;
@@ -205,6 +212,9 @@ bool CreateDir(std::string directory_name) {
 }
 
 std::string ResourcePath(std::string name, std::string extension) {
+#if defined(WEBRTC_IOS)
+  return IOSResourcePath(name, extension);
+#else
   std::string platform = "win";
 #ifdef WEBRTC_LINUX
   platform = "linux";
@@ -239,6 +249,7 @@ std::string ResourcePath(std::string name, std::string extension) {
 
   // Fall back on name without architecture or platform.
   return resources_path + name + "." + extension;
+#endif  // defined (WEBRTC_IOS)
 }
 
 size_t GetFileSize(std::string filename) {
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/fileutils_unittest.cc b/media/webrtc/trunk/webrtc/test/testsupport/fileutils_unittest.cc
index dff7f2249b..e205db3ecf 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/fileutils_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/fileutils_unittest.cc
@@ -16,7 +16,6 @@
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 
 #ifdef WIN32
 #define chdir _chdir
@@ -66,7 +65,14 @@ TEST_F(FileUtilsTest, ProjectRootPath) {
 }
 
 // Similar to the above test, but for the output dir
-TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutputPathFromUnchangedWorkingDir \
+  DISABLED_OutputPathFromUnchangedWorkingDir
+#else
+#define MAYBE_OutputPathFromUnchangedWorkingDir \
+  OutputPathFromUnchangedWorkingDir
+#endif
+TEST_F(FileUtilsTest, MAYBE_OutputPathFromUnchangedWorkingDir) {
   std::string path = webrtc::test::OutputPath();
   std::string expected_end = "out";
   expected_end = kPathDelimiter + expected_end + kPathDelimiter;
@@ -75,7 +81,12 @@ TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) {
 
 // Tests with current working directory set to a directory higher up in the
 // directory tree than the project root dir.
-TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromRootWorkingDir)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OutputPathFromRootWorkingDir DISABLED_OutputPathFromRootWorkingDir
+#else
+#define MAYBE_OutputPathFromRootWorkingDir OutputPathFromRootWorkingDir
+#endif
+TEST_F(FileUtilsTest, MAYBE_OutputPathFromRootWorkingDir) {
   ASSERT_EQ(0, chdir(kPathDelimiter));
   ASSERT_EQ("./", webrtc::test::OutputPath());
 }
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/gtest_disable.h b/media/webrtc/trunk/webrtc/test/testsupport/gtest_disable.h
deleted file mode 100644
index 257d836121..0000000000
--- a/media/webrtc/trunk/webrtc/test/testsupport/gtest_disable.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_
-#define TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_
-
-// Helper macros for platform disables. These can be chained. Example use:
-// TEST_F(ViEStandardIntegrationTest,
-//        DISABLED_ON_LINUX(RunsBaseTestWithoutErrors)) {  // ...
-//
-// Or, you can disable a whole test class by wrapping all mentions of the test
-// class name inside one of these macros.
-//
-// The platform #defines we are looking at here are set by the build system.
-#ifdef WEBRTC_LINUX
-#define DISABLED_ON_LINUX(test) DISABLED_##test
-#else
-#define DISABLED_ON_LINUX(test) test
-#endif
-
-#ifdef WEBRTC_MAC
-#define DISABLED_ON_MAC(test) DISABLED_##test
-#else
-#define DISABLED_ON_MAC(test) test
-#endif
-
-#ifdef _WIN32
-#define DISABLED_ON_WIN(test) DISABLED_##test
-#else
-#define DISABLED_ON_WIN(test) test
-#endif
-
-#ifdef WEBRTC_ANDROID
-#define DISABLED_ON_ANDROID(test) DISABLED_##test
-#else
-#define DISABLED_ON_ANDROID(test) test
-#endif
-
-#endif  // TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/iosfileutils.mm b/media/webrtc/trunk/webrtc/test/testsupport/iosfileutils.mm
new file mode 100644
index 0000000000..f3615ed681
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/testsupport/iosfileutils.mm
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if defined(WEBRTC_IOS)
+
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
+
+#import <Foundation/Foundation.h>
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// TODO(henrika): move to shared location.
+// See https://code.google.com/p/webrtc/issues/detail?id=4773 for details.
+NSString* NSStringFromStdString(const std::string& stdString) {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+std::string StdStringFromNSString(NSString* nsString) {
+  NSData* charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char*>([charData bytes]),
+                     [charData length]);
+}
+
+// For iOS, resource files are added to the application bundle in the root
+// and not in separate folders as is the case for other platforms. This method
+// therefore removes any prepended folders and uses only the actual file name.
+std::string IOSResourcePath(std::string name, std::string extension) {
+  @autoreleasepool {
+    NSString* path = NSStringFromStdString(name);
+    NSString* fileName = path.lastPathComponent;
+    NSString* fileType = NSStringFromStdString(extension);
+    // Get full pathname for the resource identified by the name and extension.
+    NSString* pathString = [[NSBundle mainBundle] pathForResource:fileName
+                                                           ofType:fileType];
+    return StdStringFromNSString(pathString);
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_IOS)
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc b/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc
index 0202a71ebe..947b81d442 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc
@@ -15,8 +15,8 @@
 
 #include <algorithm>  // min_element, max_element
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 namespace test {
@@ -34,8 +34,8 @@ enum VideoMetricsType { kPSNR, kSSIM, kBoth };
 
 // Calculates metrics for a frame and adds statistics to the result for it.
 void CalculateFrame(VideoMetricsType video_metrics_type,
-                    const I420VideoFrame* ref,
-                    const I420VideoFrame* test,
+                    const VideoFrame* ref,
+                    const VideoFrame* test,
                     int frame_number,
                     QualityMetricsResult* result) {
   FrameResult frame_result = {0, 0};
@@ -109,8 +109,8 @@ int CalculateMetrics(VideoMetricsType video_metrics_type,
 
   // Read reference and test frames.
   const size_t frame_length = 3 * width * height >> 1;
-  I420VideoFrame ref_frame;
-  I420VideoFrame test_frame;
+  VideoFrame ref_frame;
+  VideoFrame test_frame;
   rtc::scoped_ptr<uint8_t[]> ref_buffer(new uint8_t[frame_length]);
   rtc::scoped_ptr<uint8_t[]> test_buffer(new uint8_t[frame_length]);
 
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h b/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h
index 88f1811740..a713b798c5 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h
+++ b/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_TEST_TEST_SUPPORT_TRACE_TO_STDERR_H_
 #define WEBRTC_TEST_TEST_SUPPORT_TRACE_TO_STDERR_H_
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 namespace test {
diff --git a/media/webrtc/trunk/webrtc/test/vcm_capturer.cc b/media/webrtc/trunk/webrtc/test/vcm_capturer.cc
index f9976d21c0..0a82236c98 100644
--- a/media/webrtc/trunk/webrtc/test/vcm_capturer.cc
+++ b/media/webrtc/trunk/webrtc/test/vcm_capturer.cc
@@ -10,14 +10,15 @@
 
 #include "webrtc/test/vcm_capturer.h"
 
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 namespace test {
 
-VcmCapturer::VcmCapturer(webrtc::VideoSendStreamInput* input)
-    : VideoCapturer(input), started_(false), vcm_(NULL) {}
+VcmCapturer::VcmCapturer(webrtc::VideoCaptureInput* input)
+    : VideoCapturer(input), started_(false), vcm_(NULL) {
+}
 
 bool VcmCapturer::Init(size_t width, size_t height, size_t target_fps) {
   VideoCaptureModule::DeviceInfo* device_info =
@@ -53,22 +54,29 @@ bool VcmCapturer::Init(size_t width, size_t height, size_t target_fps) {
   return true;
 }
 
-VcmCapturer* VcmCapturer::Create(VideoSendStreamInput* input,
-                                 size_t width, size_t height,
+VcmCapturer* VcmCapturer::Create(VideoCaptureInput* input,
+                                 size_t width,
+                                 size_t height,
                                  size_t target_fps) {
-  VcmCapturer* vcm__capturer = new VcmCapturer(input);
-  if (!vcm__capturer->Init(width, height, target_fps)) {
+  VcmCapturer* vcm_capturer = new VcmCapturer(input);
+  if (!vcm_capturer->Init(width, height, target_fps)) {
     // TODO(pbos): Log a warning that this failed.
-    delete vcm__capturer;
+    delete vcm_capturer;
     return NULL;
   }
-  return vcm__capturer;
+  return vcm_capturer;
 }
 
 
-void VcmCapturer::Start() { started_ = true; }
+void VcmCapturer::Start() {
+  rtc::CritScope lock(&crit_);
+  started_ = true;
+}
 
-void VcmCapturer::Stop() { started_ = false; }
+void VcmCapturer::Stop() {
+  rtc::CritScope lock(&crit_);
+  started_ = false;
+}
 
 void VcmCapturer::Destroy() {
   if (vcm_ == NULL) {
@@ -87,7 +95,8 @@ void VcmCapturer::Destroy() {
 VcmCapturer::~VcmCapturer() { Destroy(); }
 
 void VcmCapturer::OnIncomingCapturedFrame(const int32_t id,
-                                          const I420VideoFrame& frame) {
+                                          const VideoFrame& frame) {
+  rtc::CritScope lock(&crit_);
   if (started_)
     input_->IncomingCapturedFrame(frame);
 }
diff --git a/media/webrtc/trunk/webrtc/test/vcm_capturer.h b/media/webrtc/trunk/webrtc/test/vcm_capturer.h
index c73eeb1ce9..6c30dd50e0 100644
--- a/media/webrtc/trunk/webrtc/test/vcm_capturer.h
+++ b/media/webrtc/trunk/webrtc/test/vcm_capturer.h
@@ -7,12 +7,13 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_
+#ifndef WEBRTC_TEST_VCM_CAPTURER_H_
+#define WEBRTC_TEST_VCM_CAPTURER_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/test/video_capturer.h"
 
 namespace webrtc {
@@ -20,27 +21,30 @@ namespace test {
 
 class VcmCapturer : public VideoCapturer, public VideoCaptureDataCallback {
  public:
-  static VcmCapturer* Create(VideoSendStreamInput* input, size_t width,
-                             size_t height, size_t target_fps);
+  static VcmCapturer* Create(VideoCaptureInput* input,
+                             size_t width,
+                             size_t height,
+                             size_t target_fps);
   virtual ~VcmCapturer();
 
   void Start() override;
   void Stop() override;
 
   void OnIncomingCapturedFrame(const int32_t id,
-                               const I420VideoFrame& frame) override;  // NOLINT
+                               const VideoFrame& frame) override;  // NOLINT
   void OnCaptureDelayChanged(const int32_t id, const int32_t delay) override;
 
  private:
-  explicit VcmCapturer(VideoSendStreamInput* input);
+  explicit VcmCapturer(VideoCaptureInput* input);
   bool Init(size_t width, size_t height, size_t target_fps);
   void Destroy();
 
-  bool started_;
+  rtc::CriticalSection crit_;
+  bool started_ GUARDED_BY(crit_);
   VideoCaptureModule* vcm_;
   VideoCaptureCapability capability_;
 };
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_
+#endif  // WEBRTC_TEST_VCM_CAPTURER_H_
diff --git a/media/webrtc/trunk/webrtc/test/video_capturer.cc b/media/webrtc/trunk/webrtc/test/video_capturer.cc
index fc37648bc5..840378f013 100644
--- a/media/webrtc/trunk/webrtc/test/video_capturer.cc
+++ b/media/webrtc/trunk/webrtc/test/video_capturer.cc
@@ -26,10 +26,10 @@ class NullCapturer : public VideoCapturer {
   virtual void Stop() {}
 };
 
-VideoCapturer::VideoCapturer(VideoSendStreamInput* input)
-    : input_(input) {}
+VideoCapturer::VideoCapturer(VideoCaptureInput* input) : input_(input) {
+}
 
-VideoCapturer* VideoCapturer::Create(VideoSendStreamInput* input,
+VideoCapturer* VideoCapturer::Create(VideoCaptureInput* input,
                                      size_t width,
                                      size_t height,
                                      int fps,
diff --git a/media/webrtc/trunk/webrtc/test/video_capturer.h b/media/webrtc/trunk/webrtc/test/video_capturer.h
index ec576a0fd9..169fd7151d 100644
--- a/media/webrtc/trunk/webrtc/test/video_capturer.h
+++ b/media/webrtc/trunk/webrtc/test/video_capturer.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_
+#ifndef WEBRTC_TEST_VIDEO_CAPTURER_H_
+#define WEBRTC_TEST_VIDEO_CAPTURER_H_
 
 #include <stddef.h>
 
@@ -16,13 +16,13 @@ namespace webrtc {
 
 class Clock;
 
-class VideoSendStreamInput;
+class VideoCaptureInput;
 
 namespace test {
 
 class VideoCapturer {
  public:
-  static VideoCapturer* Create(VideoSendStreamInput* input,
+  static VideoCapturer* Create(VideoCaptureInput* input,
                                size_t width,
                                size_t height,
                                int fps,
@@ -33,10 +33,10 @@ class VideoCapturer {
   virtual void Stop() = 0;
 
  protected:
-  explicit VideoCapturer(VideoSendStreamInput* input);
-  VideoSendStreamInput* input_;
+  explicit VideoCapturer(VideoCaptureInput* input);
+  VideoCaptureInput* input_;
 };
 }  // test
 }  // webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_
+#endif  // WEBRTC_TEST_VIDEO_CAPTURER_H_
diff --git a/media/webrtc/trunk/webrtc/test/video_renderer.cc b/media/webrtc/trunk/webrtc/test/video_renderer.cc
index 03c4948910..c7b60e5949 100644
--- a/media/webrtc/trunk/webrtc/test/video_renderer.cc
+++ b/media/webrtc/trunk/webrtc/test/video_renderer.cc
@@ -17,7 +17,7 @@ namespace webrtc {
 namespace test {
 
 class NullRenderer : public VideoRenderer {
-  void RenderFrame(const I420VideoFrame& video_frame,
+  void RenderFrame(const VideoFrame& video_frame,
                    int time_to_render_ms) override {}
   bool IsTextureSupported() const override { return false; }
 };
diff --git a/media/webrtc/trunk/webrtc/test/video_renderer.h b/media/webrtc/trunk/webrtc/test/video_renderer.h
index c8623270a7..3739522d7a 100644
--- a/media/webrtc/trunk/webrtc/test/video_renderer.h
+++ b/media/webrtc/trunk/webrtc/test/video_renderer.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_
+#ifndef WEBRTC_TEST_VIDEO_RENDERER_H_
+#define WEBRTC_TEST_VIDEO_RENDERER_H_
 
 #include <stddef.h>
 
@@ -36,4 +36,4 @@ class VideoRenderer : public webrtc::VideoRenderer {
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_
+#endif  // WEBRTC_TEST_VIDEO_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp b/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp
index ea6e4052db..07ea2c9b74 100644
--- a/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp
+++ b/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp
@@ -18,6 +18,8 @@
         'call_test.h',
         'configurable_frame_size_encoder.cc',
         'configurable_frame_size_encoder.h',
+        'constants.cc',
+        'constants.h',
         'direct_transport.cc',
         'direct_transport.h',
         'encoder_settings.cc',
@@ -32,7 +34,11 @@
         'fake_network_pipe.h',
         'frame_generator_capturer.cc',
         'frame_generator_capturer.h',
+        'layer_filtering_transport.cc',
+        'layer_filtering_transport.h',
         'mock_transport.h',
+        'mock_voe_channel_proxy.h',
+        'mock_voice_engine.h',
         'null_transport.cc',
         'null_transport.h',
         'rtp_rtcp_observer.h',
@@ -54,13 +60,14 @@
         }],
       ],
       'dependencies': [
+        '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/base/base.gyp:rtc_base',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:media_file',
         '<(webrtc_root)/modules/modules.gyp:video_render',
-        '<(webrtc_root)/test/test.gyp:frame_generator',
+        '<(webrtc_root)/test/test.gyp:fake_video_frames',
         '<(webrtc_root)/test/test.gyp:test_support',
         '<(webrtc_root)/test/test.gyp:rtp_test_utils',
         '<(webrtc_root)/webrtc.gyp:webrtc',
@@ -108,11 +115,24 @@
             '<(directx_sdk_path)/Include',
           ],
         }],
+        ['OS=="win" and clang==1', {
+          'msvs_settings': {
+            'VCCLCompilerTool': {
+              'AdditionalOptions': [
+                # Disable warnings failing when compiling with Clang on Windows.
+                # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                '-Wno-bool-conversion',
+                '-Wno-comment',
+                '-Wno-delete-non-virtual-dtor',
+              ],
+            },
+          },
+        }],
       ],
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/modules/modules.gyp:media_file',
-        '<(webrtc_root)/test/test.gyp:frame_generator',
+        '<(webrtc_root)/test/test.gyp:fake_video_frames',
         '<(webrtc_root)/test/test.gyp:test_support',
       ],
       'direct_dependent_settings': {
diff --git a/media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc b/media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc
index 7da8f445b0..86900e93dd 100644
--- a/media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc
+++ b/media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc
@@ -191,7 +191,7 @@ void D3dRenderer::Resize(size_t width, size_t height) {
   vertex_buffer_->Unlock();
 }
 
-void D3dRenderer::RenderFrame(const webrtc::I420VideoFrame& frame,
+void D3dRenderer::RenderFrame(const webrtc::VideoFrame& frame,
                               int /*render_delay_ms*/) {
   if (static_cast<size_t>(frame.width()) != width_ ||
       static_cast<size_t>(frame.height()) != height_) {
diff --git a/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h b/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
index 4fa6c7ef46..cf2319edc4 100644
--- a/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
+++ b/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
@@ -7,14 +7,14 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_
+#ifndef WEBRTC_TEST_WIN_D3D_RENDERER_H_
+#define WEBRTC_TEST_WIN_D3D_RENDERER_H_
 
 #include <Windows.h>
 #include <d3d9.h>
 #pragma comment(lib, "d3d9.lib")       // located in DirectX SDK
 
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/typedefs.h"
 
@@ -27,7 +27,7 @@ class D3dRenderer : public VideoRenderer {
                              size_t height);
   virtual ~D3dRenderer();
 
-  void RenderFrame(const webrtc::I420VideoFrame& frame, int delta) override;
+  void RenderFrame(const webrtc::VideoFrame& frame, int delta) override;
   bool IsTextureSupported() const override { return false; }
 
  private:
@@ -42,13 +42,13 @@ class D3dRenderer : public VideoRenderer {
   size_t width_, height_;
 
   HWND hwnd_;
-  scoped_refptr<IDirect3D9> d3d_;
-  scoped_refptr<IDirect3DDevice9> d3d_device_;
+  rtc::scoped_refptr<IDirect3D9> d3d_;
+  rtc::scoped_refptr<IDirect3DDevice9> d3d_device_;
 
-  scoped_refptr<IDirect3DTexture9> texture_;
-  scoped_refptr<IDirect3DVertexBuffer9> vertex_buffer_;
+  rtc::scoped_refptr<IDirect3DTexture9> texture_;
+  rtc::scoped_refptr<IDirect3DVertexBuffer9> vertex_buffer_;
 };
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_
+#endif  // WEBRTC_TEST_WIN_D3D_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/tools/BUILD.gn b/media/webrtc/trunk/webrtc/tools/BUILD.gn
index 882a16e306..0daba4d6c8 100644
--- a/media/webrtc/trunk/webrtc/tools/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/tools/BUILD.gn
@@ -16,12 +16,49 @@ source_set("tools") {
 
 source_set("command_line_parser") {
   sources = [
-    "simple_command_line_parser.h",
     "simple_command_line_parser.cc",
+    "simple_command_line_parser.h",
+  ]
+  deps = [
+    "..:gtest_prod",
   ]
-
   configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config"]
+  public_configs = [ "..:common_inherited_config" ]
+}
+
+source_set("video_quality_analysis") {
+  sources = [
+    "frame_analyzer/video_quality_analysis.cc",
+    "frame_analyzer/video_quality_analysis.h",
+  ]
+  deps = [
+    "../common_video",
+  ]
+  public_deps = [
+    "../common_video",
+  ]
+}
+
+executable("frame_analyzer") {
+  sources = [
+    "frame_analyzer/frame_analyzer.cc",
+  ]
+  deps = [
+    ":command_line_parser",
+    ":video_quality_analysis",
+  ]
+}
+
+executable("rgba_to_i420_converter") {
+  sources = [
+    "converter/converter.cc",
+    "converter/converter.h",
+    "converter/rgba_to_i420_converter.cc",
+  ]
+  deps = [
+    ":command_line_parser",
+    "../common_video",
+  ]
 }
 
 # TODO(kjellander): Convert all of tools.gyp into GN here.
@@ -35,7 +72,7 @@ if (!build_with_chromium) {
     ]
 
     configs += [ "..:common_config" ]
-    public_configs = [ "..:common_inherited_config"]
+    public_configs = [ "..:common_inherited_config" ]
 
     deps = [
       ":command_line_parser",
diff --git a/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc b/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc
index a51216acd9..2cb0a1b2df 100644
--- a/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc
+++ b/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc
@@ -18,13 +18,13 @@
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
 #include "webrtc/modules/audio_processing/agc/histogram.h"
-#include "webrtc/modules/audio_processing/agc/pitch_based_vad.h"
-#include "webrtc/modules/audio_processing/agc/standalone_vad.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h"
+#include "webrtc/modules/audio_processing/vad/common.h"
+#include "webrtc/modules/audio_processing/vad/pitch_based_vad.h"
+#include "webrtc/modules/audio_processing/vad/standalone_vad.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 static const int kAgcAnalWindowSamples = 100;
 static const double kDefaultActivityThreshold = 0.3;
@@ -56,16 +56,16 @@ namespace webrtc {
 // silence frame. Otherwise true VAD would drift with respect to the audio.
 // We only consider mono inputs.
 static void DitherSilence(AudioFrame* frame) {
-  ASSERT_EQ(1, frame->num_channels_);
+  ASSERT_EQ(1u, frame->num_channels_);
   const double kRmsSilence = 5;
   const double sum_squared_silence = kRmsSilence * kRmsSilence *
       frame->samples_per_channel_;
   double sum_squared = 0;
-  for (int n = 0; n < frame->samples_per_channel_; n++)
+  for (size_t n = 0; n < frame->samples_per_channel_; n++)
     sum_squared += frame->data_[n] * frame->data_[n];
   if (sum_squared <= sum_squared_silence) {
-    for (int n = 0; n < frame->samples_per_channel_; n++)
-      frame->data_[n] = (rand() & 0xF) - 8;
+    for (size_t n = 0; n < frame->samples_per_channel_; n++)
+      frame->data_[n] = (rand() & 0xF) - 8;  // NOLINT: ignore non-threadsafe.
   }
 }
 
@@ -75,11 +75,11 @@ class AgcStat {
       : video_index_(0),
         activity_threshold_(kDefaultActivityThreshold),
         audio_content_(Histogram::Create(kAgcAnalWindowSamples)),
-        audio_processing_(new AgcAudioProc()),
+        audio_processing_(new VadAudioProc()),
         vad_(new PitchBasedVad()),
         standalone_vad_(StandaloneVad::Create()),
         audio_content_fid_(NULL) {
-    for (int n = 0; n < kMaxNumFrames; n++)
+    for (size_t n = 0; n < kMaxNumFrames; n++)
       video_vad_[n] = 0.5;
   }
 
@@ -116,7 +116,7 @@ class AgcStat {
       // TODO(turajs) combining and limiting are used in the source files as
       // well they can be moved to utility.
       // Combine Video and stand-alone VAD.
-      for (int n = 0; n < features.num_frames; n++) {
+      for (size_t n = 0; n < features.num_frames; n++) {
         double p_active = p[n] * video_vad_[n];
         double p_passive = (1 - p[n]) * (1 - video_vad_[n]);
         p[n]  = p_active / (p_active + p_passive);
@@ -125,7 +125,7 @@ class AgcStat {
       }
       if (vad_->VoicingProbability(features, p) < 0)
         return -1;
-      for (int n = 0; n < features.num_frames; n++) {
+      for (size_t n = 0; n < features.num_frames; n++) {
         audio_content_->Update(features.rms[n], p[n]);
         double ac = audio_content_->AudioContent();
         if (audio_content_fid_ != NULL) {
@@ -139,7 +139,7 @@ class AgcStat {
       }
       video_index_ = 0;
     }
-    return features.num_frames;
+    return static_cast<int>(features.num_frames);
   }
 
   void Reset() {
@@ -155,7 +155,7 @@ class AgcStat {
   double activity_threshold_;
   double video_vad_[kMaxNumFrames];
   rtc::scoped_ptr<Histogram> audio_content_;
-  rtc::scoped_ptr<AgcAudioProc> audio_processing_;
+  rtc::scoped_ptr<VadAudioProc> audio_processing_;
   rtc::scoped_ptr<PitchBasedVad> vad_;
   rtc::scoped_ptr<StandaloneVad> standalone_vad_;
 
@@ -246,7 +246,7 @@ void void_main(int argc, char* argv[]) {
   bool onset = false;
   uint8_t previous_true_vad = 0;
   int num_not_adapted = 0;
-  int true_vad_index = 0;
+  size_t true_vad_index = 0;
   bool in_false_positive_region = false;
   int total_false_positive_duration = 0;
   bool video_adapted = false;
@@ -292,7 +292,7 @@ void void_main(int argc, char* argv[]) {
     ASSERT_GE(ret_val, 0);
 
     if (ret_val > 0) {
-      ASSERT_TRUE(ret_val == true_vad_index);
+      ASSERT_EQ(true_vad_index, static_cast<size_t>(ret_val));
       for (int n = 0; n < ret_val; n++) {
         if (true_vad[n] == 1) {
           total_active++;
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc
index ae8d942295..0d35d4b56a 100644
--- a/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc
@@ -11,13 +11,14 @@
 // Refer to kUsage below for a description.
 
 #include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
-#include "webrtc/tools/agc/agc_manager.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
@@ -28,20 +29,26 @@
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 
 DEFINE_bool(codecs, false, "print out available codecs");
-DEFINE_int32(pt, 103, "codec payload type (defaults to ISAC/16000/1)");
-DEFINE_bool(internal, true, "use the internal AGC in 'serial' mode, or as the "
-                            "first voice engine's AGC in parallel mode");
-DEFINE_bool(parallel, false, "run internal and public AGCs in parallel, with "
-    "left- and right-panning respectively. Not compatible with -aec.");
+DEFINE_int32(pt, 120, "codec payload type (defaults to opus/48000/2)");
+DEFINE_bool(legacy_agc,
+            false,
+            "use the legacy AGC in 'serial' mode, or as the first voice "
+            "engine's AGC in parallel mode");
+DEFINE_bool(parallel,
+            false,
+            "run new and legacy AGCs in parallel, with left- and right-panning "
+            "respectively. Not compatible with -aec.");
 DEFINE_bool(devices, false, "print out capture devices and indexes to be used "
                             "with the capture flags");
 DEFINE_int32(capture1, 0, "capture device index for the first voice engine");
 DEFINE_int32(capture2, 0, "capture device index for second voice engine");
 DEFINE_int32(render1, 0, "render device index for first voice engine");
 DEFINE_int32(render2, 0, "render device index for second voice engine");
-DEFINE_bool(aec, false, "runs two voice engines in parallel, with the first "
-    "playing out a file and sending its captured signal to the second voice "
-    "engine. Also enables echo cancellation.");
+DEFINE_bool(aec,
+            false,
+            "runs two voice engines in parallel, with the first playing out a "
+            "file and sending its captured signal to the second voice engine. "
+            "Also enables echo cancellation.");
 DEFINE_bool(ns, true, "enable noise suppression");
 DEFINE_bool(highpass, true, "enable high pass filter");
 DEFINE_string(filename, "", "filename for the -aec mode");
@@ -51,10 +58,9 @@ namespace {
 
 const char kUsage[] =
     "\nWithout additional flags, sets up a simple VoiceEngine loopback call\n"
-    "with the default audio devices and runs forever. The internal AGC is\n"
-    "enabled and the public disabled.\n\n"
+    "with the default audio devices and runs forever.\n"
 
-    "It can also run the public AGC in parallel with the internal, panned to\n"
+    "It can also run the new and legacy AGCs in parallel, panned to\n"
     "opposite stereo channels on the default render device. The capture\n"
     "devices for each can be selected (recommended, because otherwise they\n"
     "will fight for the level on the same device).\n\n"
@@ -75,63 +81,63 @@ class AgcVoiceEngine {
     PanRight
   };
 
-  AgcVoiceEngine(bool internal, int tx_port, int rx_port, int capture_idx,
+  AgcVoiceEngine(bool legacy_agc,
+                 int tx_port,
+                 int rx_port,
+                 int capture_idx,
                  int render_idx)
       : voe_(VoiceEngine::Create()),
         base_(VoEBase::GetInterface(voe_)),
         hardware_(VoEHardware::GetInterface(voe_)),
         codec_(VoECodec::GetInterface(voe_)),
-        manager_(new AgcManager(voe_)),
         channel_(-1),
         capture_idx_(capture_idx),
         render_idx_(render_idx) {
-    SetUp(internal, tx_port, rx_port);
+    SetUp(legacy_agc, tx_port, rx_port);
   }
 
   ~AgcVoiceEngine() {
     TearDown();
   }
 
-  void SetUp(bool internal, int tx_port, int rx_port) {
-    ASSERT_TRUE(voe_ != NULL);
-    ASSERT_TRUE(base_ != NULL);
-    ASSERT_TRUE(hardware_ != NULL);
-    ASSERT_TRUE(codec_ != NULL);
+  void SetUp(bool legacy_agc, int tx_port, int rx_port) {
     VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe_);
-    ASSERT_TRUE(audio != NULL);
     VoENetwork* network = VoENetwork::GetInterface(voe_);
-    ASSERT_TRUE(network != NULL);
-
-    ASSERT_EQ(0, base_->Init());
+    {
+      webrtc::Config config;
+      config.Set<ExperimentalAgc>(new ExperimentalAgc(!legacy_agc));
+      AudioProcessing* audioproc = AudioProcessing::Create(config);
+      RTC_CHECK_EQ(0, base_->Init(nullptr, audioproc));
+      // Set this stuff after Init, to override the default voice engine
+      // settings.
+      audioproc->gain_control()->Enable(true);
+      audioproc->high_pass_filter()->Enable(FLAGS_highpass);
+      audioproc->noise_suppression()->Enable(FLAGS_ns);
+      audioproc->echo_cancellation()->Enable(FLAGS_aec);
+    }
     channel_ = base_->CreateChannel();
-    ASSERT_NE(-1, channel_);
+    RTC_CHECK_NE(-1, channel_);
 
     channel_transport_.reset(
         new test::VoiceChannelTransport(network, channel_));
-    ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", tx_port));
-    ASSERT_EQ(0, channel_transport_->SetLocalReceiver(rx_port));
+    RTC_CHECK_EQ(0,
+                 channel_transport_->SetSendDestination("127.0.0.1", tx_port));
+    RTC_CHECK_EQ(0, channel_transport_->SetLocalReceiver(rx_port));
 
-    ASSERT_EQ(0, hardware_->SetRecordingDevice(capture_idx_));
-    ASSERT_EQ(0, hardware_->SetPlayoutDevice(render_idx_));
+    RTC_CHECK_EQ(0, hardware_->SetRecordingDevice(capture_idx_));
+    RTC_CHECK_EQ(0, hardware_->SetPlayoutDevice(render_idx_));
 
-    CodecInst codec_params = {0};
+    CodecInst codec_params = {};
     bool codec_found = false;
     for (int i = 0; i < codec_->NumOfCodecs(); i++) {
-      ASSERT_EQ(0, codec_->GetCodec(i, codec_params));
+      RTC_CHECK_EQ(0, codec_->GetCodec(i, codec_params));
       if (FLAGS_pt == codec_params.pltype) {
         codec_found = true;
         break;
       }
     }
-    ASSERT_TRUE(codec_found);
-    ASSERT_EQ(0, codec_->SetSendCodec(channel_, codec_params));
-
-    ASSERT_EQ(0, audio->EnableHighPassFilter(FLAGS_highpass));
-    ASSERT_EQ(0, audio->SetNsStatus(FLAGS_ns));
-    ASSERT_EQ(0, audio->SetEcStatus(FLAGS_aec));
-
-    ASSERT_EQ(0, manager_->Enable(internal));
-    ASSERT_EQ(0, audio->SetAgcStatus(!internal));
+    RTC_CHECK(codec_found);
+    RTC_CHECK_EQ(0, codec_->SetSendCodec(channel_, codec_params));
 
     audio->Release();
     network->Release();
@@ -139,31 +145,29 @@ class AgcVoiceEngine {
 
   void TearDown() {
     Stop();
-    channel_transport_.reset(NULL);
-    ASSERT_EQ(0, base_->DeleteChannel(channel_));
-    ASSERT_EQ(0, base_->Terminate());
-    // Don't test; the manager hasn't released its interfaces.
+    channel_transport_.reset(nullptr);
+    RTC_CHECK_EQ(0, base_->DeleteChannel(channel_));
+    RTC_CHECK_EQ(0, base_->Terminate());
     hardware_->Release();
     base_->Release();
     codec_->Release();
-    delete manager_;
-    ASSERT_TRUE(VoiceEngine::Delete(voe_));
+    RTC_CHECK(VoiceEngine::Delete(voe_));
   }
 
   void PrintDevices() {
     int num_devices = 0;
     char device_name[128] = {0};
     char guid[128] = {0};
-    ASSERT_EQ(0, hardware_->GetNumOfRecordingDevices(num_devices));
+    RTC_CHECK_EQ(0, hardware_->GetNumOfRecordingDevices(num_devices));
     printf("Capture devices:\n");
     for (int i = 0; i < num_devices; i++) {
-      ASSERT_EQ(0, hardware_->GetRecordingDeviceName(i, device_name, guid));
+      RTC_CHECK_EQ(0, hardware_->GetRecordingDeviceName(i, device_name, guid));
       printf("%d: %s\n", i, device_name);
     }
-    ASSERT_EQ(0, hardware_->GetNumOfPlayoutDevices(num_devices));
+    RTC_CHECK_EQ(0, hardware_->GetNumOfPlayoutDevices(num_devices));
     printf("Render devices:\n");
     for (int i = 0; i < num_devices; i++) {
-      ASSERT_EQ(0, hardware_->GetPlayoutDeviceName(i, device_name, guid));
+      RTC_CHECK_EQ(0, hardware_->GetPlayoutDeviceName(i, device_name, guid));
       printf("%d: %s\n", i, device_name);
     }
   }
@@ -172,21 +176,17 @@ class AgcVoiceEngine {
     CodecInst params = {0};
     printf("Codecs:\n");
     for (int i = 0; i < codec_->NumOfCodecs(); i++) {
-      ASSERT_EQ(0, codec_->GetCodec(i, params));
-      printf("%d %s/%d/%d\n", params.pltype, params.plname, params.plfreq,
-             params.channels);
+      RTC_CHECK_EQ(0, codec_->GetCodec(i, params));
+      printf("%d %s/%d/%" PRIuS "\n", params.pltype, params.plname,
+             params.plfreq, params.channels);
     }
   }
 
-  void StartSending() {
-    ASSERT_EQ(0, base_->StartSend(channel_));
-  }
+  void StartSending() { RTC_CHECK_EQ(0, base_->StartSend(channel_)); }
 
   void StartPlaying(Pan pan, const std::string& filename) {
     VoEVolumeControl* volume = VoEVolumeControl::GetInterface(voe_);
     VoEFile* file = VoEFile::GetInterface(voe_);
-    ASSERT_TRUE(volume != NULL);
-    ASSERT_TRUE(file != NULL);
     if (pan == PanLeft) {
       volume->SetOutputVolumePan(channel_, 1, 0);
     } else if (pan == PanRight) {
@@ -194,18 +194,19 @@ class AgcVoiceEngine {
     }
     if (filename != "") {
       printf("playing file\n");
-      ASSERT_EQ(0, file->StartPlayingFileLocally(channel_, filename.c_str(),
-          true, kFileFormatPcm16kHzFile, 1.0, 0, 0));
+      RTC_CHECK_EQ(
+          0, file->StartPlayingFileLocally(channel_, filename.c_str(), true,
+                                           kFileFormatPcm16kHzFile, 1.0, 0, 0));
     }
-    ASSERT_EQ(0, base_->StartReceive(channel_));
-    ASSERT_EQ(0, base_->StartPlayout(channel_));
+    RTC_CHECK_EQ(0, base_->StartReceive(channel_));
+    RTC_CHECK_EQ(0, base_->StartPlayout(channel_));
     volume->Release();
     file->Release();
   }
 
   void Stop() {
-    ASSERT_EQ(0, base_->StopSend(channel_));
-    ASSERT_EQ(0, base_->StopPlayout(channel_));
+    RTC_CHECK_EQ(0, base_->StopSend(channel_));
+    RTC_CHECK_EQ(0, base_->StopPlayout(channel_));
   }
 
  private:
@@ -213,7 +214,6 @@ class AgcVoiceEngine {
   VoEBase* base_;
   VoEHardware* hardware_;
   VoECodec* codec_;
-  AgcManager* manager_;
   int channel_;
   int capture_idx_;
   int render_idx_;
@@ -222,19 +222,19 @@ class AgcVoiceEngine {
 
 void RunHarness() {
   rtc::scoped_ptr<AgcVoiceEngine> voe1(new AgcVoiceEngine(
-      FLAGS_internal, 2000, 2000, FLAGS_capture1, FLAGS_render1));
+      FLAGS_legacy_agc, 2000, 2000, FLAGS_capture1, FLAGS_render1));
   rtc::scoped_ptr<AgcVoiceEngine> voe2;
   if (FLAGS_parallel) {
-    voe2.reset(new AgcVoiceEngine(!FLAGS_internal, 3000, 3000, FLAGS_capture2,
+    voe2.reset(new AgcVoiceEngine(!FLAGS_legacy_agc, 3000, 3000, FLAGS_capture2,
                                   FLAGS_render2));
     voe1->StartPlaying(AgcVoiceEngine::PanLeft, "");
     voe1->StartSending();
     voe2->StartPlaying(AgcVoiceEngine::PanRight, "");
     voe2->StartSending();
   } else if (FLAGS_aec) {
-    voe1.reset(new AgcVoiceEngine(FLAGS_internal, 2000, 4242, FLAGS_capture1,
+    voe1.reset(new AgcVoiceEngine(FLAGS_legacy_agc, 2000, 4242, FLAGS_capture1,
                                   FLAGS_render1));
-    voe2.reset(new AgcVoiceEngine(!FLAGS_internal, 4242, 2000, FLAGS_capture2,
+    voe2.reset(new AgcVoiceEngine(!FLAGS_legacy_agc, 4242, 2000, FLAGS_capture2,
                                   FLAGS_render2));
     voe1->StartPlaying(AgcVoiceEngine::NoPan, FLAGS_filename);
     voe1->StartSending();
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_manager.cc
deleted file mode 100644
index 83c0d0075b..0000000000
--- a/media/webrtc/trunk/webrtc/tools/agc/agc_manager.cc
+++ /dev/null
@@ -1,252 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/tools/agc/agc_manager.h"
-
-#include <assert.h>
-
-#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-namespace webrtc {
-
-class AgcManagerVolume : public VolumeCallbacks {
- public:
-  // AgcManagerVolume acquires ownership of |volume|.
-  explicit AgcManagerVolume(VoEVolumeControl* volume)
-      : volume_(volume) {
-  }
-
-  ~AgcManagerVolume() {
-    if (volume_) {
-      volume_->Release();
-    }
-  }
-
-  virtual void SetMicVolume(int volume) {
-    if (volume_->SetMicVolume(volume) != 0) {
-      LOG_FERR1(LS_WARNING, SetMicVolume, volume);
-    }
-  }
-
-  int GetMicVolume() {
-    unsigned int volume = 0;
-    if (volume_->GetMicVolume(volume) != 0) {
-      LOG_FERR0(LS_WARNING, GetMicVolume);
-      return -1;
-    }
-    return volume;
-  }
-
- private:
-  VoEVolumeControl* volume_;
-};
-
-class MediaCallback : public VoEMediaProcess {
- public:
-  MediaCallback(AgcManagerDirect* direct, AudioProcessing* audioproc,
-                CriticalSectionWrapper* crit)
-      : direct_(direct),
-        audioproc_(audioproc),
-        crit_(crit),
-        frame_() {
-  }
-
- protected:
-  virtual void Process(const int channel, const ProcessingTypes type,
-                       int16_t audio[], const int samples_per_channel,
-                       const int sample_rate_hz, const bool is_stereo) {
-    CriticalSectionScoped cs(crit_);
-    if (direct_->capture_muted()) {
-      return;
-    }
-
-    // Extract the first channel.
-    const int kMaxSampleRateHz = 48000;
-    const int kMaxSamplesPerChannel = kMaxSampleRateHz / 100;
-    assert(samples_per_channel < kMaxSamplesPerChannel &&
-           sample_rate_hz < kMaxSampleRateHz);
-    int16_t mono[kMaxSamplesPerChannel];
-    int16_t* mono_ptr = audio;
-    if (is_stereo) {
-      for (int n = 0; n < samples_per_channel; n++) {
-        mono[n] = audio[n * 2];
-      }
-      mono_ptr = mono;
-    }
-
-    direct_->Process(mono_ptr, samples_per_channel, sample_rate_hz);
-
-    // TODO(ajm): It's unfortunate we have to memcpy to this frame here, but
-    // it's needed for use with AudioProcessing.
-    frame_.num_channels_ = is_stereo ? 2 : 1;
-    frame_.samples_per_channel_ = samples_per_channel;
-    frame_.sample_rate_hz_ = sample_rate_hz;
-    const int length_samples = frame_.num_channels_ * samples_per_channel;
-    memcpy(frame_.data_, audio, length_samples * sizeof(int16_t));
-
-    // Apply compression to the audio.
-    if (audioproc_->ProcessStream(&frame_) != 0) {
-      LOG_FERR0(LS_ERROR, ProcessStream);
-    }
-
-    // Copy the compressed audio back to voice engine's array.
-    memcpy(audio, frame_.data_, length_samples * sizeof(int16_t));
-  }
-
- private:
-  AgcManagerDirect* direct_;
-  AudioProcessing* audioproc_;
-  CriticalSectionWrapper* crit_;
-  AudioFrame frame_;
-};
-
-class PreprocCallback : public VoEMediaProcess {
- public:
-  PreprocCallback(AgcManagerDirect* direct, CriticalSectionWrapper* crit)
-      : direct_(direct),
-        crit_(crit) {
-  }
-
- protected:
-  virtual void Process(const int channel, const ProcessingTypes type,
-                       int16_t audio[], const int samples_per_channel,
-                       const int sample_rate_hz, const bool is_stereo) {
-    CriticalSectionScoped cs(crit_);
-    if (direct_->capture_muted()) {
-      return;
-    }
-    direct_->AnalyzePreProcess(audio, is_stereo ? 2 : 1, samples_per_channel);
-  }
-
- private:
-  AgcManagerDirect* direct_;
-  CriticalSectionWrapper* crit_;
-};
-
-AgcManager::AgcManager(VoiceEngine* voe)
-    : media_(VoEExternalMedia::GetInterface(voe)),
-      volume_callbacks_(new AgcManagerVolume(VoEVolumeControl::GetInterface(
-          voe))),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      enabled_(false),
-      initialized_(false) {
-    Config config;
-    config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
-    audioproc_.reset(AudioProcessing::Create(config));
-    direct_.reset(new AgcManagerDirect(audioproc_->gain_control(),
-                                       volume_callbacks_.get()));
-    media_callback_.reset(new MediaCallback(direct_.get(),
-                                            audioproc_.get(),
-                                            crit_.get()));
-    preproc_callback_.reset(new PreprocCallback(direct_.get(), crit_.get()));
-}
-
-AgcManager::AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume,
-                       Agc* agc, AudioProcessing* audioproc)
-    : media_(media),
-      volume_callbacks_(new AgcManagerVolume(volume)),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      audioproc_(audioproc),
-      direct_(new AgcManagerDirect(agc,
-                                   audioproc_->gain_control(),
-                                   volume_callbacks_.get())),
-      media_callback_(new MediaCallback(direct_.get(),
-                                        audioproc_.get(),
-                                        crit_.get())),
-      preproc_callback_(new PreprocCallback(direct_.get(), crit_.get())),
-      enabled_(false),
-      initialized_(false) {
-}
-
-AgcManager::AgcManager()
-    : media_(NULL),
-      enabled_(false),
-      initialized_(false) {
-}
-
-AgcManager::~AgcManager() {
-  if (media_) {
-    if (enabled_) {
-      DeregisterCallbacks();
-    }
-    media_->Release();
-  }
-}
-
-int AgcManager::Enable(bool enable) {
-  if (enable == enabled_) {
-    return 0;
-  }
-  if (!initialized_) {
-    CriticalSectionScoped cs(crit_.get());
-    if (audioproc_->gain_control()->Enable(true) != 0) {
-      LOG_FERR1(LS_ERROR, gain_control()->Enable, true);
-      return -1;
-    }
-    if (direct_->Initialize() != 0) {
-      assert(false);
-      return -1;
-    }
-    initialized_ = true;
-  }
-
-  if (enable) {
-    if (media_->RegisterExternalMediaProcessing(0, kRecordingAllChannelsMixed,
-                                                *media_callback_) != 0) {
-      LOG(LS_ERROR) << "Failed to register postproc callback";
-      return -1;
-    }
-    if (media_->RegisterExternalMediaProcessing(0, kRecordingPreprocessing,
-                                                *preproc_callback_) != 0) {
-      LOG(LS_ERROR) << "Failed to register preproc callback";
-      return -1;
-    }
-  } else {
-    if (DeregisterCallbacks() != 0)
-      return -1;
-  }
-  enabled_ = enable;
-  return 0;
-}
-
-void AgcManager::CaptureDeviceChanged() {
-  CriticalSectionScoped cs(crit_.get());
-  direct_->Initialize();
-}
-
-void AgcManager::SetCaptureMuted(bool muted) {
-  CriticalSectionScoped cs(crit_.get());
-  direct_->SetCaptureMuted(muted);
-}
-
-int AgcManager::DeregisterCallbacks() {
-  // DeRegister shares a lock with the Process() callback. This call will block
-  // until the callback is finished and it's safe to continue teardown.
-  int err = 0;
-  if (media_->DeRegisterExternalMediaProcessing(0,
-          kRecordingAllChannelsMixed) != 0) {
-    LOG(LS_ERROR) << "Failed to deregister postproc callback";
-    err = -1;
-  }
-  if (media_->DeRegisterExternalMediaProcessing(0,
-          kRecordingPreprocessing) != 0) {
-    LOG(LS_ERROR) << "Failed to deregister preproc callback";
-    err = -1;
-  }
-  return err;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager.h b/media/webrtc/trunk/webrtc/tools/agc/agc_manager.h
deleted file mode 100644
index 4f79f102d6..0000000000
--- a/media/webrtc/trunk/webrtc/tools/agc/agc_manager.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
-#define WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
-
-namespace webrtc {
-
-class Agc;
-class AudioProcessing;
-class CriticalSectionWrapper;
-class MediaCallback;
-class PreprocCallback;
-class VoEExternalMedia;
-class VoEVolumeControl;
-class VoiceEngine;
-class VolumeCallbacks;
-
-// Handles the interaction between VoiceEngine and the internal AGC. It hooks
-// into the capture stream through VoiceEngine's external media interface and
-// sends the audio to the AGC for analysis. It forwards requests for a capture
-// volume change from the AGC to the VoiceEngine volume interface.
-class AgcManager {
- public:
-  explicit AgcManager(VoiceEngine* voe);
-  // Dependency injection for testing. Don't delete |agc| or |audioproc| as the
-  // memory is owned by the manager. If |media| or |volume| are non-fake
-  // reference counted classes, don't release them as this is handled by the
-  // manager.
-  AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume, Agc* agc,
-             AudioProcessing* audioproc);
-  virtual ~AgcManager();
-
-  // When enabled, registers external media processing callbacks with
-  // VoiceEngine to hook into the capture stream. Disabling deregisters the
-  // callbacks.
-  virtual int Enable(bool enable);
-  virtual bool enabled() const { return enabled_; }
-
-  // Call when the capture device has changed. This will trigger a retrieval of
-  // the initial capture volume on the next audio frame.
-  virtual void CaptureDeviceChanged();
-
-  // Call when the capture stream has been muted/unmuted. This causes the
-  // manager to disregard all incoming audio; chances are good it's background
-  // noise to which we'd like to avoid adapting.
-  virtual void SetCaptureMuted(bool muted);
-  virtual bool capture_muted() const { return direct_->capture_muted(); }
-
- protected:
-  // Provide a default constructor for testing.
-  AgcManager();
-
- private:
-  int DeregisterCallbacks();
-  int CheckVolumeAndReset();
-
-  VoEExternalMedia* media_;
-  rtc::scoped_ptr<VolumeCallbacks> volume_callbacks_;
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-  rtc::scoped_ptr<AudioProcessing> audioproc_;
-  rtc::scoped_ptr<AgcManagerDirect> direct_;
-  rtc::scoped_ptr<MediaCallback> media_callback_;
-  rtc::scoped_ptr<PreprocCallback> preproc_callback_;
-  bool enabled_;
-  bool initialized_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager_integrationtest.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_manager_integrationtest.cc
deleted file mode 100644
index 4179e8c1c2..0000000000
--- a/media/webrtc/trunk/webrtc/tools/agc/agc_manager_integrationtest.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/tools/agc/agc_manager.h"
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_processing/agc/mock_agc.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-using ::testing::_;
-using ::testing::AtLeast;
-using ::testing::Mock;
-using ::testing::Return;
-
-namespace webrtc {
-
-class AgcManagerTest : public ::testing::Test {
- protected:
-  AgcManagerTest()
-      : voe_(VoiceEngine::Create()),
-        base_(VoEBase::GetInterface(voe_)),
-        agc_(new MockAgc()),
-        manager_(new AgcManager(VoEExternalMedia::GetInterface(voe_),
-                                VoEVolumeControl::GetInterface(voe_),
-                                agc_,
-                                AudioProcessing::Create())),
-        channel_(-1) {
-  }
-
-  virtual void SetUp() {
-    ASSERT_TRUE(voe_ != NULL);
-    ASSERT_TRUE(base_ != NULL);
-    ASSERT_EQ(0, base_->Init());
-    channel_ = base_->CreateChannel();
-    ASSERT_NE(-1, channel_);
-
-    VoENetwork* network = VoENetwork::GetInterface(voe_);
-    ASSERT_TRUE(network != NULL);
-    channel_transport_.reset(
-        new test::VoiceChannelTransport(network, channel_));
-    ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", 1234));
-    network->Release();
-  }
-
-  virtual void TearDown() {
-    channel_transport_.reset(NULL);
-    ASSERT_EQ(0, base_->DeleteChannel(channel_));
-    ASSERT_EQ(0, base_->Terminate());
-    delete manager_;
-    // Test that the manager has released all VoE interfaces. The last
-    // reference is released in VoiceEngine::Delete.
-    EXPECT_EQ(1, base_->Release());
-    ASSERT_TRUE(VoiceEngine::Delete(voe_));
-  }
-
-  VoiceEngine* voe_;
-  VoEBase* base_;
-  MockAgc* agc_;
-  rtc::scoped_ptr<test::VoiceChannelTransport> channel_transport_;
-  // We use a pointer for the manager, so we can tear it down and test
-  // base_->Release() in the destructor.
-  AgcManager* manager_;
-  int channel_;
-};
-
-TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(EnableSucceeds)) {
-  EXPECT_EQ(0, manager_->Enable(true));
-  EXPECT_TRUE(manager_->enabled());
-  EXPECT_EQ(0, manager_->Enable(false));
-  EXPECT_FALSE(manager_->enabled());
-}
-
-TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(ProcessIsNotCalledByDefault)) {
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
-  EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
-  ASSERT_EQ(0, base_->StartSend(channel_));
-  SleepMs(100);
-  ASSERT_EQ(0, base_->StopSend(channel_));
-}
-
-TEST_F(AgcManagerTest, DISABLED_ProcessIsCalledOnlyWhenEnabled) {
-  EXPECT_CALL(*agc_, Reset());
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .Times(AtLeast(1))
-      .WillRepeatedly(Return(0));
-  EXPECT_CALL(*agc_, Process(_, _, _))
-      .Times(AtLeast(1))
-      .WillRepeatedly(Return(0));
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .Times(AtLeast(1))
-      .WillRepeatedly(Return(false));
-  EXPECT_EQ(0, manager_->Enable(true));
-  ASSERT_EQ(0, base_->StartSend(channel_));
-  SleepMs(100);
-  EXPECT_EQ(0, manager_->Enable(false));
-  SleepMs(100);
-  Mock::VerifyAndClearExpectations(agc_);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
-  EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
-  SleepMs(100);
-  ASSERT_EQ(0, base_->StopSend(channel_));
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager_unittest.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_manager_unittest.cc
deleted file mode 100644
index fca8decc85..0000000000
--- a/media/webrtc/trunk/webrtc/tools/agc/agc_manager_unittest.cc
+++ /dev/null
@@ -1,736 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/tools/agc/agc_manager.h"
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_processing/agc/mock_agc.h"
-#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
-#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-
-using ::testing::_;
-using ::testing::DoAll;
-using ::testing::Eq;
-using ::testing::Mock;
-using ::testing::Return;
-using ::testing::SetArgPointee;
-using ::testing::SetArgReferee;
-
-namespace webrtc {
-namespace {
-
-const int kSampleRateHz = 32000;
-const int kNumChannels = 1;
-const int kSamplesPerChannel = kSampleRateHz / 100;
-const float kAboveClippedThreshold = 0.2f;
-
-}  // namespace
-
-class AgcManagerUnitTest : public ::testing::Test {
- protected:
-  AgcManagerUnitTest()
-      : media_(),
-        volume_(),
-        agc_(new MockAgc),
-        audioproc_(new MockAudioProcessing),
-        gctrl_(audioproc_->gain_control()),
-        manager_(&media_, &volume_, agc_, audioproc_) {
-    EXPECT_CALL(*gctrl_, Enable(true));
-    ExpectInitialize();
-    manager_.Enable(true);
-    EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-        .WillOnce(Return(false));
-    // TODO(bjornv): Find a better solution that adds an initial volume here
-    // instead of applying SetVolumeAndProcess(128u) in each test, but at the
-    // same time can test a too low initial value.
-  }
-
-  void SetInitialVolume(unsigned int volume) {
-    ExpectInitialize();
-    manager_.CaptureDeviceChanged();
-    ExpectCheckVolumeAndReset(volume);
-    EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
-    PostProcCallback(1);
-  }
-
-  void SetVolumeAndProcess(unsigned int volume) {
-    // Volume is checked on first process call.
-    ExpectCheckVolumeAndReset(volume);
-    PostProcCallback(1);
-  }
-
-  void ExpectCheckVolumeAndReset(unsigned int volume) {
-    EXPECT_CALL(volume_, GetMicVolume(_))
-        .WillOnce(DoAll(SetArgReferee<0>(volume), Return(0)));
-    EXPECT_CALL(*agc_, Reset());
-  }
-
-  void ExpectVolumeChange(unsigned int current_volume,
-                          unsigned int new_volume) {
-    EXPECT_CALL(volume_, GetMicVolume(_))
-        .WillOnce(DoAll(SetArgReferee<0>(current_volume), Return(0)));
-    EXPECT_CALL(volume_, SetMicVolume(Eq(new_volume))).WillOnce(Return(0));
-  }
-
-  void ExpectInitialize() {
-    EXPECT_CALL(*gctrl_, set_mode(GainControl::kFixedDigital));
-    EXPECT_CALL(*gctrl_, set_target_level_dbfs(2));
-    EXPECT_CALL(*gctrl_, set_compression_gain_db(7));
-    EXPECT_CALL(*gctrl_, enable_limiter(true));
-  }
-
-  void PreProcCallback(int num_calls) {
-    for (int i = 0; i < num_calls; ++i) {
-      media_.CallProcess(kRecordingPreprocessing, NULL, kSamplesPerChannel,
-                         kSampleRateHz, kNumChannels);
-    }
-  }
-
-  void PostProcCallback(int num_calls) {
-    for (int i = 0; i < num_calls; ++i) {
-      EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return(0));
-      EXPECT_CALL(*audioproc_, ProcessStream(_)).WillOnce(Return(0));
-      media_.CallProcess(kRecordingAllChannelsMixed, NULL, kSamplesPerChannel,
-                         kSampleRateHz, kNumChannels);
-    }
-  }
-
-  ~AgcManagerUnitTest() {
-    EXPECT_CALL(volume_, Release()).WillOnce(Return(0));
-  }
-
-  FakeVoEExternalMedia media_;
-  MockVoEVolumeControl volume_;
-  MockAgc* agc_;
-  MockAudioProcessing* audioproc_;
-  MockGainControl* gctrl_;
-  AgcManager manager_;
-  test::TraceToStderr trace_to_stderr;
-};
-
-TEST_F(AgcManagerUnitTest, MicVolumeResponseToRmsError) {
-  SetVolumeAndProcess(128u);
-  // Compressor default; no residual error.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
-  PostProcCallback(1);
-
-  // Inside the compressor's window; no change of volume.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)));
-  PostProcCallback(1);
-
-  // Above the compressor's window; volume should be increased.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
-  ExpectVolumeChange(128u, 130u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
-  ExpectVolumeChange(130u, 168u);
-  PostProcCallback(1);
-
-  // Inside the compressor's window; no change of volume.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
-  PostProcCallback(1);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)));
-  PostProcCallback(1);
-
-  // Below the compressor's window; volume should be decreased.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  ExpectVolumeChange(168u, 167u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  ExpectVolumeChange(167u, 163u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-9), Return(true)));
-  ExpectVolumeChange(163u, 129u);
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, MicVolumeIsLimited) {
-  SetVolumeAndProcess(128u);
-  // Maximum upwards change is limited.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
-  ExpectVolumeChange(128u, 183u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
-  ExpectVolumeChange(183u, 243u);
-  PostProcCallback(1);
-
-  // Won't go higher than the maximum.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
-  ExpectVolumeChange(243u, 255u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  ExpectVolumeChange(255u, 254u);
-  PostProcCallback(1);
-
-  // Maximum downwards change is limited.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(254u, 194u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(194u, 137u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(137u, 88u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(88u, 54u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(54u, 33u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(33u, 18u);
-  PostProcCallback(1);
-
-  // Won't go lower than the minimum.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
-  ExpectVolumeChange(18u, 12u);
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorStepsTowardsTarget) {
-  SetVolumeAndProcess(128u);
-  // Compressor default; no call to set_compression_gain_db.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
-      .WillRepeatedly(Return(false));
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(20);
-
-  // Moves slowly upwards.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
-      .WillRepeatedly(Return(false));
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
-  PostProcCallback(1);
-
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
-  PostProcCallback(1);
-
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(20);
-
-  // Moves slowly downward, then reverses before reaching the original target.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
-      .WillRepeatedly(Return(false));
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
-      .WillRepeatedly(Return(false));
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
-  PostProcCallback(1);
-
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(20);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorErrorIsDeemphasized) {
-  SetVolumeAndProcess(128u);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
-      .WillRepeatedly(Return(false));
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
-  PostProcCallback(1);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(20);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
-      .WillRepeatedly(Return(false));
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(7)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
-  PostProcCallback(1);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
-  PostProcCallback(20);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorReachesMaximum) {
-  SetVolumeAndProcess(128u);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
-      .WillRepeatedly(Return(false));
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorReachesMinimum) {
-  SetVolumeAndProcess(128u);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
-      .WillRepeatedly(Return(false));
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(5)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(4)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(3)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(2)).WillOnce(Return(0));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, NoActionWhileMuted) {
-  SetVolumeAndProcess(128u);
-  manager_.SetCaptureMuted(true);
-  media_.CallProcess(kRecordingAllChannelsMixed, NULL, kSamplesPerChannel,
-                     kSampleRateHz, kNumChannels);
-}
-
-TEST_F(AgcManagerUnitTest, UnmutingChecksVolumeWithoutRaising) {
-  SetVolumeAndProcess(128u);
-  manager_.SetCaptureMuted(true);
-  manager_.SetCaptureMuted(false);
-  ExpectCheckVolumeAndReset(127u);
-  // SetMicVolume should not be called.
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(Return(false));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, UnmutingRaisesTooLowVolume) {
-  SetVolumeAndProcess(128u);
-  manager_.SetCaptureMuted(true);
-  manager_.SetCaptureMuted(false);
-  ExpectCheckVolumeAndReset(11u);
-  EXPECT_CALL(volume_, SetMicVolume(Eq(12u))).WillOnce(Return(0));
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(Return(false));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ChangingDevicesChecksVolume) {
-  SetVolumeAndProcess(128u);
-  ExpectInitialize();
-  manager_.CaptureDeviceChanged();
-  ExpectCheckVolumeAndReset(128u);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(Return(false));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, LowInitialVolumeIsRaised) {
-  ExpectCheckVolumeAndReset(11u);
-  // Should set MicVolume to kMinInitMicLevel = 85.
-  EXPECT_CALL(volume_, SetMicVolume(Eq(85u))).WillOnce(Return(0));
-  PostProcCallback(1);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(Return(false));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ManualLevelChangeResultsInNoSetMicCall) {
-  SetVolumeAndProcess(128u);
-  // Change outside of compressor's range, which would normally trigger a call
-  // to SetMicVolume.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
-  // GetMicVolume returns a value outside of the quantization slack, indicating
-  // a manual volume change.
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillOnce(DoAll(SetArgReferee<0>(154u), Return(0)));
-  // SetMicVolume should not be called.
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  PostProcCallback(1);
-
-  // Do the same thing, except downwards now.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillOnce(DoAll(SetArgReferee<0>(100u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  PostProcCallback(1);
-
-  // And finally verify the AGC continues working without a manual change.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  ExpectVolumeChange(100u, 99u);
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, RecoveryAfterManualLevelChangeFromMax) {
-  SetVolumeAndProcess(128u);
-  // Force the mic up to max volume. Takes a few steps due to the residual
-  // gain limitation.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
-  ExpectVolumeChange(128u, 183u);
-  PostProcCallback(1);
-  ExpectVolumeChange(183u, 243u);
-  PostProcCallback(1);
-  ExpectVolumeChange(243u, 255u);
-  PostProcCallback(1);
-
-  // Manual change does not result in SetMicVolume call.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillOnce(DoAll(SetArgReferee<0>(50u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  PostProcCallback(1);
-
-  // Continues working as usual afterwards.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
-  ExpectVolumeChange(50u, 69u);
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, RecoveryAfterManualLevelChangeBelowMin) {
-  SetVolumeAndProcess(128u);
-  // Manual change below min.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
-  // Don't set to zero, which will cause AGC to take no action.
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillOnce(DoAll(SetArgReferee<0>(1u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  PostProcCallback(1);
-
-  // Continues working as usual afterwards.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
-  ExpectVolumeChange(1u, 2u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
-  ExpectVolumeChange(2u, 11u);
-  PostProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
-  ExpectVolumeChange(11u, 18u);
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, NoClippingHasNoImpact) {
-  SetVolumeAndProcess(128u);
-  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(0);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillRepeatedly(Return(0));
-  PreProcCallback(100);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingUnderThresholdHasNoImpact) {
-  SetVolumeAndProcess(128u);
-  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(0);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.099));
-  PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingLowersVolume) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(255u);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.101));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(255u, 240u);
-  PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, WaitingPeriodBetweenClippingChecks) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(255u);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(255u, 240u);
-  PreProcCallback(1);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillRepeatedly(Return(kAboveClippedThreshold));
-  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(0);
-  PreProcCallback(300);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(240u, 225u);
-  PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingLoweringIsLimited) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(180u);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(180u, 170u);
-  PreProcCallback(1);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillRepeatedly(Return(kAboveClippedThreshold));
-  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(0);
-  PreProcCallback(1000);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingMaxIsRespectedWhenEqualToLevel) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(255u);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(255u, 240u);
-  PreProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillRepeatedly(DoAll(SetArgReferee<0>(240u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  PostProcCallback(10);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingMaxIsRespectedWhenHigherThanLevel) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(200u);
-
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(200u, 185u);
-  PreProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-    .WillRepeatedly(DoAll(SetArgPointee<0>(40), Return(true)));
-  ExpectVolumeChange(185u, 240u);
-  PostProcCallback(1);
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillRepeatedly(DoAll(SetArgReferee<0>(240u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  PostProcCallback(10);
-}
-
-TEST_F(AgcManagerUnitTest, MaxCompressionIsIncreasedAfterClipping) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(210u);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(210u, 195u);
-  PreProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
-      .WillRepeatedly(Return(false));
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(13)).WillOnce(Return(0));
-  PostProcCallback(1);
-
-  // Continue clipping until we hit the maximum surplus compression.
-  PreProcCallback(300);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(195u, 180u);
-  PreProcCallback(1);
-
-  PreProcCallback(300);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(180u, 170u);
-  PreProcCallback(1);
-
-  // Current level is now at the minimum, but the maximum allowed level still
-  // has more to decrease.
-  PreProcCallback(300);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  PreProcCallback(1);
-
-  PreProcCallback(300);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  PreProcCallback(1);
-
-  PreProcCallback(300);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  PreProcCallback(1);
-
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
-      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
-      .WillRepeatedly(Return(false));
-  PostProcCallback(19);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(14)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(15)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(16)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(17)).WillOnce(Return(0));
-  PostProcCallback(20);
-  EXPECT_CALL(*gctrl_, set_compression_gain_db(18)).WillOnce(Return(0));
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, UserCanRaiseVolumeAfterClipping) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(225u);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  ExpectVolumeChange(225u, 210u);
-  PreProcCallback(1);
-
-  // High enough error to trigger a volume check.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(14), Return(true)));
-  // User changed the volume.
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillOnce(DoAll(SetArgReferee<0>(250u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(1);
-  PostProcCallback(1);
-
-  // Move down...
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(-10), Return(true)));
-  ExpectVolumeChange(250u, 210u);
-  PostProcCallback(1);
-  // And back up to the new max established by the user.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(40), Return(true)));
-  ExpectVolumeChange(210u, 250u);
-  PostProcCallback(1);
-  // Will not move above new maximum.
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillRepeatedly(DoAll(SetArgReferee<0>(250u), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingDoesNotPullLowVolumeBackUp) {
-  SetVolumeAndProcess(128u);
-  SetInitialVolume(80u);
-  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
-      .WillOnce(Return(kAboveClippedThreshold));
-  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  EXPECT_CALL(*agc_, Reset()).Times(0);
-  PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, TakesNoActionOnZeroMicVolume) {
-  SetVolumeAndProcess(128u);
-  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
-      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
-  EXPECT_CALL(volume_, GetMicVolume(_))
-      .WillRepeatedly(DoAll(SetArgReferee<0>(0), Return(0)));
-  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
-  PostProcCallback(10);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_test.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_test.cc
deleted file mode 100644
index 29769488c1..0000000000
--- a/media/webrtc/trunk/webrtc/tools/agc/agc_test.cc
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cmath>
-#include <cstdio>
-
-#include <algorithm>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-#include "webrtc/tools/agc/agc_manager.h"
-#include "webrtc/tools/agc/test_utils.h"
-#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
-#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
-
-DEFINE_string(in, "in.pcm", "input filename");
-DEFINE_string(out, "out.pcm", "output filename");
-DEFINE_int32(rate, 16000, "sample rate in Hz");
-DEFINE_int32(channels, 1, "number of channels");
-DEFINE_int32(level, -18, "target level in RMS dBFs [-100, 0]");
-DEFINE_bool(limiter, true, "enable a limiter for the compression stage");
-DEFINE_int32(cmp_level, 2, "target level in dBFs for the compression stage");
-DEFINE_int32(mic_gain, 80, "range of gain provided by the virtual mic in dB");
-DEFINE_int32(gain_offset, 0,
-             "an amount (in dB) to add to every entry in the gain map");
-DEFINE_string(gain_file, "",
-    "filename providing a mic gain mapping. The file should be text containing "
-    "a (floating-point) gain entry in dBFs per line corresponding to levels "
-    "from 0 to 255.");
-
-using ::testing::_;
-using ::testing::ByRef;
-using ::testing::DoAll;
-using ::testing::Mock;
-using ::testing::Return;
-using ::testing::SaveArg;
-using ::testing::SetArgReferee;
-
-namespace webrtc {
-namespace {
-
-const char kUsage[] = "\nProcess an audio file to simulate an analog agc.";
-
-void ReadGainMapFromFile(FILE* file, int offset, int gain_map[256]) {
-  for (int i = 0; i < 256; ++i) {
-    float gain = 0;
-    ASSERT_EQ(1, fscanf(file, "%f", &gain));
-    gain_map[i] = std::floor(gain + 0.5);
-  }
-
-  // Adjust from dBFs to gain in dB. We assume that level 127 provides 0 dB
-  // gain. This corresponds to the interpretation in MicLevel2Gain().
-  const int midpoint = gain_map[127];
-  printf("Gain map\n");
-  for (int i = 0; i < 256; ++i) {
-    gain_map[i] += offset - midpoint;
-    if (i % 5 == 0) {
-      printf("%d: %d dB\n", i, gain_map[i]);
-    }
-  }
-}
-
-void CalculateGainMap(int gain_range_db, int offset, int gain_map[256]) {
-  printf("Gain map\n");
-  for (int i = 0; i < 256; ++i) {
-    gain_map[i] = std::floor(MicLevel2Gain(gain_range_db, i) + 0.5) + offset;
-    if (i % 5 == 0) {
-      printf("%d: %d dB\n", i, gain_map[i]);
-    }
-  }
-}
-
-void RunAgc() {
-  test::TraceToStderr trace_to_stderr(true);
-  FILE* in_file = fopen(FLAGS_in.c_str(), "rb");
-  ASSERT_TRUE(in_file != NULL);
-  FILE* out_file = fopen(FLAGS_out.c_str(), "wb");
-  ASSERT_TRUE(out_file != NULL);
-
-  int gain_map[256];
-  if (FLAGS_gain_file != "") {
-    FILE* gain_file = fopen(FLAGS_gain_file.c_str(), "rt");
-    ASSERT_TRUE(gain_file != NULL);
-    ReadGainMapFromFile(gain_file, FLAGS_gain_offset, gain_map);
-    fclose(gain_file);
-  } else {
-    CalculateGainMap(FLAGS_mic_gain, FLAGS_gain_offset, gain_map);
-  }
-
-  FakeVoEExternalMedia media;
-  MockVoEVolumeControl volume;
-  Agc* agc = new Agc;
-  AudioProcessing* audioproc = AudioProcessing::Create();
-  ASSERT_TRUE(audioproc != NULL);
-  AgcManager manager(&media, &volume, agc, audioproc);
-
-  int mic_level = 128;
-  int last_mic_level = mic_level;
-  EXPECT_CALL(volume, GetMicVolume(_))
-      .WillRepeatedly(DoAll(SetArgReferee<0>(ByRef(mic_level)), Return(0)));
-  EXPECT_CALL(volume, SetMicVolume(_))
-      .WillRepeatedly(DoAll(SaveArg<0>(&mic_level), Return(0)));
-
-  manager.Enable(true);
-  ASSERT_EQ(0, agc->set_target_level_dbfs(FLAGS_level));
-  const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
-  GainControl* gctrl = audioproc->gain_control();
-  ASSERT_EQ(kNoErr, gctrl->set_target_level_dbfs(FLAGS_cmp_level));
-  ASSERT_EQ(kNoErr, gctrl->enable_limiter(FLAGS_limiter));
-
-  AudioFrame frame;
-  frame.num_channels_ = FLAGS_channels;
-  frame.sample_rate_hz_ = FLAGS_rate;
-  frame.samples_per_channel_ = FLAGS_rate / 100;
-  const size_t frame_length = frame.samples_per_channel_ * FLAGS_channels;
-  size_t sample_count = 0;
-  while (fread(frame.data_, sizeof(int16_t), frame_length, in_file) ==
-      frame_length) {
-    SimulateMic(gain_map, mic_level, last_mic_level, &frame);
-    last_mic_level = mic_level;
-    media.CallProcess(kRecordingAllChannelsMixed, frame.data_,
-                      frame.samples_per_channel_, FLAGS_rate, FLAGS_channels);
-    ASSERT_EQ(frame_length,
-              fwrite(frame.data_, sizeof(int16_t), frame_length, out_file));
-    sample_count += frame_length;
-    trace_to_stderr.SetTimeSeconds(static_cast<float>(sample_count) /
-                                   FLAGS_channels / FLAGS_rate);
-  }
-  fclose(in_file);
-  fclose(out_file);
-  EXPECT_CALL(volume, Release());
-}
-
-}  // namespace
-}  // namespace webrtc
-
-int main(int argc, char* argv[]) {
-  google::SetUsageMessage(webrtc::kUsage);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::RunAgc();
-  return 0;
-}
diff --git a/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc b/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc
index 3a26cb9ac7..a0ed74732d 100644
--- a/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc
+++ b/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc
@@ -14,7 +14,7 @@
 
 #include <algorithm>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
@@ -27,11 +27,12 @@ float Db2Linear(float db) {
 }
 
 void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
-  const int frame_length = frame->samples_per_channel_ * frame->num_channels_;
+  const size_t frame_length =
+      frame->samples_per_channel_ * frame->num_channels_;
   // Smooth the transition between gain levels across the frame.
   float smoothed_gain = last_gain;
   float gain_step = (gain - last_gain) / (frame_length - 1);
-  for (int i = 0; i < frame_length; ++i) {
+  for (size_t i = 0; i < frame_length; ++i) {
     smoothed_gain += gain_step;
     float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
     sample = std::max(std::min(32767.0f, sample), -32768.0f);
diff --git a/media/webrtc/trunk/webrtc/tools/barcode_tools/barcode_decoder.py b/media/webrtc/trunk/webrtc/tools/barcode_tools/barcode_decoder.py
index b7b7ddd4a6..e615fa83b8 100644
--- a/media/webrtc/trunk/webrtc/tools/barcode_tools/barcode_decoder.py
+++ b/media/webrtc/trunk/webrtc/tools/barcode_tools/barcode_decoder.py
@@ -56,7 +56,7 @@ def convert_yuv_to_png_files(yuv_file_name, yuv_frame_width, yuv_frame_height,
     print 'Error executing command: %s. Error: %s' % (command, err)
     return False
   except OSError:
-    print ('Did not find %s. Have you installed it?' % ffmpeg_path)
+    print 'Did not find %s. Have you installed it?' % ffmpeg_path
     return False
   return True
 
@@ -111,7 +111,7 @@ def _decode_barcode_in_file(file_name, command_line_decoder):
     print err
     return False
   except OSError:
-    print ('Did not find %s. Have you installed it?' % command_line_decoder)
+    print 'Did not find %s. Have you installed it?' % command_line_decoder
     return False
   return True
 
@@ -200,7 +200,7 @@ def _check_barcode(barcode):
   return dsum == int(barcode[11])
 
 
-def _count_frames_in(input_directory = '.'):
+def _count_frames_in(input_directory='.'):
   """Calculates the number of frames in the input directory.
 
   The function calculates the number of frames in the input directory. The
@@ -252,7 +252,7 @@ def _parse_args():
                           'decoded. If using Windows and a Cygwin-compiled '
                           'zxing.exe, you should keep the default value to '
                           'avoid problems. Default: %default'))
-  options, _args = parser.parse_args()
+  options, _ = parser.parse_args()
   return options
 
 
diff --git a/media/webrtc/trunk/webrtc/tools/barcode_tools/helper_functions.py b/media/webrtc/trunk/webrtc/tools/barcode_tools/helper_functions.py
index bb0b167334..fb9854f8a3 100644
--- a/media/webrtc/trunk/webrtc/tools/barcode_tools/helper_functions.py
+++ b/media/webrtc/trunk/webrtc/tools/barcode_tools/helper_functions.py
@@ -7,6 +7,7 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
+import multiprocessing
 import os
 import subprocess
 import sys
@@ -64,8 +65,7 @@ def perform_action_on_all_files(directory, file_pattern, file_extension,
   """Function that performs a given action on all files matching a pattern.
 
   It is assumed that the files are named file_patternxxxx.file_extension, where
-  xxxx are digits. The file names start from
-  file_patern0..start_number>.file_extension.
+  xxxx are digits starting from start_number.
 
   Args:
     directory(string): The directory where the files live.
@@ -73,24 +73,33 @@ def perform_action_on_all_files(directory, file_pattern, file_extension,
     file_extension(string): The files' extension.
     start_number(int): From where to start to count frames.
     action(function): The action to be performed over the files. Must return
-      False if the action failed, True otherwise.
+        False if the action failed, True otherwise. It should take a file name
+        as the first argument and **kwargs as arguments. The function must be
+        possible to pickle, so it cannot be a bound function (for instance).
 
   Return:
     (bool): Whether performing the action over all files was successful or not.
   """
   file_prefix = os.path.join(directory, file_pattern)
-  file_exists = True
   file_number = start_number
-  errors = False
 
-  while file_exists:
+  process_pool = multiprocessing.Pool(processes=multiprocessing.cpu_count())
+  results = []
+  while True:
     zero_padded_file_number = zero_pad(file_number)
     file_name = file_prefix + zero_padded_file_number + '.' + file_extension
-    if os.path.isfile(file_name):
-      if not action(file_name=file_name, **kwargs):
-        errors = True
-        break
-      file_number += 1
-    else:
-      file_exists = False
-  return not errors
+    if not os.path.isfile(file_name):
+      break
+    future = process_pool.apply_async(action, args=(file_name,), kwds=kwargs)
+    results.append(future)
+    file_number += 1
+
+  successful = True
+  for result in results:
+    if not result.get():
+      print "At least one action %s failed for files %sxxxx.%s." % (
+          action, file_pattern, file_extension)
+      successful = False
+
+  process_pool.close()
+  return successful
diff --git a/media/webrtc/trunk/webrtc/tools/barcode_tools/yuv_cropper.py b/media/webrtc/trunk/webrtc/tools/barcode_tools/yuv_cropper.py
index 9652c168c7..c57a90db08 100644
--- a/media/webrtc/trunk/webrtc/tools/barcode_tools/yuv_cropper.py
+++ b/media/webrtc/trunk/webrtc/tools/barcode_tools/yuv_cropper.py
@@ -122,4 +122,4 @@ def _main():
 
 
 if __name__ == '__main__':
-  sys.exit(_main())
\ No newline at end of file
+  sys.exit(_main())
diff --git a/media/webrtc/trunk/webrtc/tools/compare_videos.py b/media/webrtc/trunk/webrtc/tools/compare_videos.py
index f6275a67d3..6aa659be36 100644
--- a/media/webrtc/trunk/webrtc/tools/compare_videos.py
+++ b/media/webrtc/trunk/webrtc/tools/compare_videos.py
@@ -55,7 +55,7 @@ def _ParseArgs():
                     help='Width of the YUV file\'s frames. Default: %default')
   parser.add_option('--yuv_frame_height', type='int', default=480,
                     help='Height of the YUV file\'s frames. Default: %default')
-  options, _args = parser.parse_args()
+  options, _ = parser.parse_args()
 
   if not options.ref_video:
     parser.error('You must provide a path to the reference video!')
diff --git a/media/webrtc/trunk/webrtc/tools/converter/converter.cc b/media/webrtc/trunk/webrtc/tools/converter/converter.cc
index 6c9154c7da..a9b453d509 100644
--- a/media/webrtc/trunk/webrtc/tools/converter/converter.cc
+++ b/media/webrtc/trunk/webrtc/tools/converter/converter.cc
@@ -45,13 +45,13 @@ bool Converter::ConvertRGBAToI420Video(std::string frames_dir,
   }
 
   int input_frame_size = InputFrameSize();
-  uint8* rgba_buffer = new uint8[input_frame_size];
+  uint8_t* rgba_buffer = new uint8_t[input_frame_size];
   int y_plane_size = YPlaneSize();
-  uint8* dst_y = new uint8[y_plane_size];
+  uint8_t* dst_y = new uint8_t[y_plane_size];
   int u_plane_size = UPlaneSize();
-  uint8* dst_u = new uint8[u_plane_size];
+  uint8_t* dst_u = new uint8_t[u_plane_size];
   int v_plane_size = VPlaneSize();
-  uint8* dst_v = new uint8[v_plane_size];
+  uint8_t* dst_v = new uint8_t[v_plane_size];
 
   int counter = 0;  // Counter to form frame names.
   bool success = false;  // Is conversion successful.
@@ -106,9 +106,12 @@ bool Converter::ConvertRGBAToI420Video(std::string frames_dir,
   return success;
 }
 
-bool Converter::AddYUVToFile(uint8* y_plane, int y_plane_size,
-                             uint8* u_plane, int u_plane_size,
-                             uint8* v_plane, int v_plane_size,
+bool Converter::AddYUVToFile(uint8_t* y_plane,
+                             int y_plane_size,
+                             uint8_t* u_plane,
+                             int u_plane_size,
+                             uint8_t* v_plane,
+                             int v_plane_size,
                              FILE* output_file) {
   bool success = AddYUVPlaneToFile(y_plane, y_plane_size, output_file) &&
                  AddYUVPlaneToFile(u_plane, u_plane_size, output_file) &&
@@ -116,7 +119,8 @@ bool Converter::AddYUVToFile(uint8* y_plane, int y_plane_size,
   return success;
 }
 
-bool Converter::AddYUVPlaneToFile(uint8* yuv_plane, int yuv_plane_size,
+bool Converter::AddYUVPlaneToFile(uint8_t* yuv_plane,
+                                  int yuv_plane_size,
                                   FILE* file) {
   size_t bytes_written = fwrite(yuv_plane, 1, yuv_plane_size, file);
 
diff --git a/media/webrtc/trunk/webrtc/tools/converter/converter.h b/media/webrtc/trunk/webrtc/tools/converter/converter.h
index a23d5a14d4..f7641ff60d 100644
--- a/media/webrtc/trunk/webrtc/tools/converter/converter.h
+++ b/media/webrtc/trunk/webrtc/tools/converter/converter.h
@@ -75,13 +75,16 @@ class Converter {
 
   // Writes the Y, U and V (in this order) planes to the file, thus adding a
   // raw YUV frame to the file.
-  bool AddYUVToFile(uint8* y_plane, int y_plane_size,
-                    uint8* u_plane, int u_plane_size,
-                    uint8* v_plane, int v_plane_size,
+  bool AddYUVToFile(uint8_t* y_plane,
+                    int y_plane_size,
+                    uint8_t* u_plane,
+                    int u_plane_size,
+                    uint8_t* v_plane,
+                    int v_plane_size,
                     FILE* output_file);
 
   // Adds the Y, U or V plane to the file.
-  bool AddYUVPlaneToFile(uint8* yuv_plane, int yuv_plane_size, FILE* file);
+  bool AddYUVPlaneToFile(uint8_t* yuv_plane, int yuv_plane_size, FILE* file);
 
   // Reads a RGBA frame from input_file_name with input_frame_size size in bytes
   // into the buffer.
diff --git a/media/webrtc/trunk/webrtc/tools/converter/rgba_to_i420_converter.cc b/media/webrtc/trunk/webrtc/tools/converter/rgba_to_i420_converter.cc
index 126a31c7a3..6b1056fae4 100644
--- a/media/webrtc/trunk/webrtc/tools/converter/rgba_to_i420_converter.cc
+++ b/media/webrtc/trunk/webrtc/tools/converter/rgba_to_i420_converter.cc
@@ -61,6 +61,7 @@ int main(int argc, char** argv) {
   parser.ProcessFlags();
   if (parser.GetFlag("help") == "true") {
     parser.PrintUsageMessage();
+    exit(EXIT_SUCCESS);
   }
   parser.PrintEnteredFlags();
 
diff --git a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
index 1eb8925537..2594fd1317 100644
--- a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
+++ b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
@@ -16,7 +16,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
diff --git a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py
index 77eda1e7a3..3d3cb10fb6 100644
--- a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py
+++ b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py
@@ -1,6 +1,12 @@
-# Copyright (c) 2012 The Chromium Authors. All rights reserved.
-# Use of this source code is governed by a BSD-style license that can be
-# found in the LICENSE file.
+#!/usr/bin/env python
+#
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
 
 # Copied from /src/chrome/test/pyautolib/pyauto_utils.py in Chromium.
 
@@ -28,4 +34,4 @@ def PrintPerfResult(graph_name, series_name, data_point, units,
   print '%sRESULT %s: %s= %s %s' % (
       waterfall_indicator, graph_name, series_name,
       str(data_point).replace(' ', ''), units)
-  sys.stdout.flush()
\ No newline at end of file
+  sys.stdout.flush()
diff --git a/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc b/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
index 570fa0ad24..b6b1596866 100644
--- a/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
+++ b/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
@@ -13,7 +13,7 @@
 #include <stdio.h>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
diff --git a/media/webrtc/trunk/webrtc/tools/frame_analyzer/frame_analyzer.cc b/media/webrtc/trunk/webrtc/tools/frame_analyzer/frame_analyzer.cc
index aed956562b..8020109327 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_analyzer/frame_analyzer.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_analyzer/frame_analyzer.cc
@@ -74,6 +74,7 @@ int main(int argc, char** argv) {
   parser.ProcessFlags();
   if (parser.GetFlag("help") == "true") {
     parser.PrintUsageMessage();
+    exit(EXIT_SUCCESS);
   }
   parser.PrintEnteredFlags();
 
diff --git a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc
index f5608c8343..dfd57f1961 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc
@@ -26,6 +26,9 @@ namespace test {
 
 using std::string;
 
+ResultsContainer::ResultsContainer() {}
+ResultsContainer::~ResultsContainer() {}
+
 int GetI420FrameSize(int width, int height) {
   int half_width = (width + 1) >> 1;
   int half_height = (height + 1) >> 1;
@@ -87,8 +90,11 @@ bool GetNextStatsLine(FILE* stats_file, char* line) {
   return true;
 }
 
-bool ExtractFrameFromYuvFile(const char* i420_file_name, int width, int height,
-                             int frame_number, uint8* result_frame) {
+bool ExtractFrameFromYuvFile(const char* i420_file_name,
+                             int width,
+                             int height,
+                             int frame_number,
+                             uint8_t* result_frame) {
   int frame_size = GetI420FrameSize(width, height);
   int offset = frame_number * frame_size;  // Calculate offset for the frame.
   bool errors = false;
@@ -114,8 +120,11 @@ bool ExtractFrameFromYuvFile(const char* i420_file_name, int width, int height,
   return !errors;
 }
 
-bool ExtractFrameFromY4mFile(const char* y4m_file_name, int width, int height,
-                             int frame_number, uint8* result_frame) {
+bool ExtractFrameFromY4mFile(const char* y4m_file_name,
+                             int width,
+                             int height,
+                             int frame_number,
+                             uint8_t* result_frame) {
   int frame_size = GetI420FrameSize(width, height);
   int frame_offset = frame_number * frame_size;
   bool errors = false;
@@ -167,20 +176,22 @@ bool ExtractFrameFromY4mFile(const char* y4m_file_name, int width, int height,
 }
 
 double CalculateMetrics(VideoAnalysisMetricsType video_metrics_type,
-                        const uint8* ref_frame,  const uint8* test_frame,
-                        int width, int height) {
+                        const uint8_t* ref_frame,
+                        const uint8_t* test_frame,
+                        int width,
+                        int height) {
   if (!ref_frame || !test_frame)
     return -1;
   else if (height < 0 || width < 0)
     return -1;
   int half_width = (width + 1) >> 1;
   int half_height = (height + 1) >> 1;
-  const uint8* src_y_a = ref_frame;
-  const uint8* src_u_a = src_y_a + width * height;
-  const uint8* src_v_a = src_u_a + half_width * half_height;
-  const uint8* src_y_b = test_frame;
-  const uint8* src_u_b = src_y_b + width * height;
-  const uint8* src_v_b = src_u_b + half_width * half_height;
+  const uint8_t* src_y_a = ref_frame;
+  const uint8_t* src_u_a = src_y_a + width * height;
+  const uint8_t* src_v_a = src_u_a + half_width * half_height;
+  const uint8_t* src_y_b = test_frame;
+  const uint8_t* src_u_b = src_y_b + width * height;
+  const uint8_t* src_v_b = src_u_b + half_width * half_height;
 
   int stride_y = width;
   int stride_uv = half_width;
@@ -216,7 +227,7 @@ void RunAnalysis(const char* reference_file_name, const char* test_file_name,
                  ResultsContainer* results) {
   // Check if the reference_file_name ends with "y4m".
   bool y4m_mode = false;
-  if (std::string(reference_file_name).find("y4m") != std::string::npos){
+  if (std::string(reference_file_name).find("y4m") != std::string::npos) {
     y4m_mode = true;
   }
 
@@ -227,8 +238,8 @@ void RunAnalysis(const char* reference_file_name, const char* test_file_name,
   char line[STATS_LINE_LENGTH];
 
   // Allocate buffers for test and reference frames.
-  uint8* test_frame = new uint8[size];
-  uint8* reference_frame = new uint8[size];
+  uint8_t* test_frame = new uint8_t[size];
+  uint8_t* reference_frame = new uint8_t[size];
   int previous_frame_number = -1;
 
   // While there are entries in the stats file.
diff --git a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h
index 4704a8cde9..475b2fa197 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h
+++ b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h
@@ -32,6 +32,9 @@ struct AnalysisResult {
 };
 
 struct ResultsContainer {
+  ResultsContainer();
+  ~ResultsContainer();
+
   std::vector<AnalysisResult> frames;
 };
 
@@ -59,8 +62,10 @@ void RunAnalysis(const char* reference_file_name, const char* test_file_name,
 // frames are exactly the same) will be 48. In the case of SSIM the max return
 // value will be 1.
 double CalculateMetrics(VideoAnalysisMetricsType video_metrics_type,
-                        const uint8* ref_frame,  const uint8* test_frame,
-                        int width, int height);
+                        const uint8_t* ref_frame,
+                        const uint8_t* test_frame,
+                        int width,
+                        int height);
 
 // Prints the result from the analysis in Chromium performance
 // numbers compatible format to stdout. If the results object contains no frames
@@ -98,14 +103,19 @@ bool IsThereBarcodeError(std::string line);
 int ExtractDecodedFrameNumber(std::string line);
 
 // Extracts an I420 frame at position frame_number from the raw YUV file.
-bool ExtractFrameFromYuvFile(const char* i420_file_name, int width, int height,
-                             int frame_number, uint8* result_frame);
+bool ExtractFrameFromYuvFile(const char* i420_file_name,
+                             int width,
+                             int height,
+                             int frame_number,
+                             uint8_t* result_frame);
 
 // Extracts an I420 frame at position frame_number from the Y4M file. The first
 // frame has corresponded |frame_number| 0.
-bool ExtractFrameFromY4mFile(const char* i420_file_name, int width, int height,
-                             int frame_number, uint8* result_frame);
-
+bool ExtractFrameFromY4mFile(const char* i420_file_name,
+                             int width,
+                             int height,
+                             int frame_number,
+                             uint8_t* result_frame);
 
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing.cc b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing.cc
index fa234ef8e2..015d5d99ae 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing.cc
@@ -77,6 +77,7 @@ int main(int argc, char** argv) {
   parser.ProcessFlags();
   if (parser.GetFlag("help") == "true") {
     parser.PrintUsageMessage();
+    exit(EXIT_SUCCESS);
   }
   parser.PrintEnteredFlags();
 
diff --git a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc
index 79c6033a30..90855a354c 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc
@@ -15,6 +15,7 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/tools/frame_editing/frame_editing_lib.h"
 #include "webrtc/typedefs.h"
 
 using std::string;
diff --git a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.h b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.h
index 245d60f376..94595c43bb 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.h
+++ b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_
-#define WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_
+#ifndef WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_
+#define WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_
 
 #include <string>
 
@@ -36,4 +36,4 @@ int EditFrames(const std::string& in_path, int width, int height,
                 int last_frame_to_process, const std::string& out_path);
 }  // namespace webrtc
 
-#endif  // WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_H_
+#endif  // WEBRTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_
diff --git a/media/webrtc/trunk/webrtc/tools/internal_tools.gyp b/media/webrtc/trunk/webrtc/tools/internal_tools.gyp
index 9d5fe25dac..61000d1bba 100644
--- a/media/webrtc/trunk/webrtc/tools/internal_tools.gyp
+++ b/media/webrtc/trunk/webrtc/tools/internal_tools.gyp
@@ -20,6 +20,9 @@
         'simple_command_line_parser.h',
         'simple_command_line_parser.cc',
       ],
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:gtest_prod',
+      ],
     }, # command_line_parser
   ],
-}
\ No newline at end of file
+}
diff --git a/media/webrtc/trunk/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc b/media/webrtc/trunk/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc
index 9c9b131a21..737661c3df 100644
--- a/media/webrtc/trunk/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc
+++ b/media/webrtc/trunk/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc
@@ -25,7 +25,7 @@ void CompareFiles(const char* reference_file_name, const char* test_file_name,
                   const char* results_file_name, int width, int height) {
   // Check if the reference_file_name ends with "y4m".
   bool y4m_mode = false;
-  if (std::string(reference_file_name).find("y4m") != std::string::npos){
+  if (std::string(reference_file_name).find("y4m") != std::string::npos) {
     y4m_mode = true;
   }
 
@@ -34,12 +34,12 @@ void CompareFiles(const char* reference_file_name, const char* test_file_name,
   int size = webrtc::test::GetI420FrameSize(width, height);
 
   // Allocate buffers for test and reference frames.
-  uint8* test_frame = new uint8[size];
-  uint8* ref_frame = new uint8[size];
+  uint8_t* test_frame = new uint8_t[size];
+  uint8_t* ref_frame = new uint8_t[size];
 
   bool read_result = true;
-  for(int frame_counter = 0; frame_counter < MAX_NUM_FRAMES_PER_FILE;
-      ++frame_counter){
+  for (int frame_counter = 0; frame_counter < MAX_NUM_FRAMES_PER_FILE;
+      ++frame_counter) {
     read_result &= (y4m_mode) ? webrtc::test::ExtractFrameFromY4mFile(
         reference_file_name, width, height, frame_counter, ref_frame):
         webrtc::test::ExtractFrameFromYuvFile(reference_file_name, width,
@@ -115,6 +115,7 @@ int main(int argc, char** argv) {
   parser.ProcessFlags();
   if (parser.GetFlag("help") == "true") {
     parser.PrintUsageMessage();
+    exit(EXIT_SUCCESS);
   }
   parser.PrintEnteredFlags();
 
diff --git a/media/webrtc/trunk/webrtc/tools/rtcbot/OWNERS b/media/webrtc/trunk/webrtc/tools/rtcbot/OWNERS
index efdce51ca6..296f71fffc 100644
--- a/media/webrtc/trunk/webrtc/tools/rtcbot/OWNERS
+++ b/media/webrtc/trunk/webrtc/tools/rtcbot/OWNERS
@@ -1,2 +1 @@
 andresp@webrtc.org
-houssainy@google.com
diff --git a/media/webrtc/trunk/webrtc/tools/simple_command_line_parser.h b/media/webrtc/trunk/webrtc/tools/simple_command_line_parser.h
index 6bb33137f5..c7bed5cd3a 100644
--- a/media/webrtc/trunk/webrtc/tools/simple_command_line_parser.h
+++ b/media/webrtc/trunk/webrtc/tools/simple_command_line_parser.h
@@ -91,7 +91,7 @@ class CommandLineParser {
   FRIEND_TEST_ALL_PREFIXES(CommandLineParserTest, GetCommandLineFlagName);
   FRIEND_TEST_ALL_PREFIXES(CommandLineParserTest, GetCommandLineFlagValue);
 
-  DISALLOW_COPY_AND_ASSIGN(CommandLineParser);
+  RTC_DISALLOW_COPY_AND_ASSIGN(CommandLineParser);
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/tools/tools.gyp b/media/webrtc/trunk/webrtc/tools/tools.gyp
index e2a54212ec..b69f7cb5b8 100644
--- a/media/webrtc/trunk/webrtc/tools/tools.gyp
+++ b/media/webrtc/trunk/webrtc/tools/tools.gyp
@@ -101,19 +101,6 @@
   'conditions': [
     ['include_tests==1', {
       'targets' : [
-        {
-          'target_name': 'agc_manager',
-          'type': 'static_library',
-          'dependencies': [
-            '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
-            '<(webrtc_root)/modules/modules.gyp:audio_processing',
-            '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-          ],
-          'sources': [
-            'agc/agc_manager.cc',
-            'agc/agc_manager.h',
-          ],
-        },
         {
           'target_name': 'agc_test_utils',
           'type': 'static_library',
@@ -131,35 +118,19 @@
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
-            'agc_manager',
+            '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
           ],
           'sources': [
             'agc/agc_harness.cc',
           ],
         },  # agc_harness
-        {
-          'target_name': 'agc_proc',
-          'type': 'executable',
-          'dependencies': [
-            '<(DEPTH)/testing/gmock.gyp:gmock',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            '<(webrtc_root)/test/test.gyp:test_support',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            'agc_manager',
-            'agc_test_utils',
-          ],
-          'sources': [
-            'agc/agc_test.cc',
-          ],
-        },  # agc_proc
         {
           'target_name': 'activity_metric',
           'type': 'executable',
           'dependencies': [
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            'agc_manager',
+            '<(webrtc_root)/modules/modules.gyp:audio_processing',
           ],
           'sources': [
             'agc/activity_metric.cc',
diff --git a/media/webrtc/trunk/webrtc/transport.h b/media/webrtc/trunk/webrtc/transport.h
index c44c5b2cc5..4e329de93f 100644
--- a/media/webrtc/trunk/webrtc/transport.h
+++ b/media/webrtc/trunk/webrtc/transport.h
@@ -8,25 +8,34 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_TRANSPORT_H_
-#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_TRANSPORT_H_
+#ifndef WEBRTC_TRANSPORT_H_
+#define WEBRTC_TRANSPORT_H_
 
 #include <stddef.h>
 
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-namespace newapi {
+
+// TODO(holmer): Look into unifying this with the PacketOptions in
+// asyncpacketsocket.h.
+struct PacketOptions {
+  // A 16 bits positive id. Negative ids are invalid and should be interpreted
+  // as packet_id not being set.
+  int packet_id = -1;
+};
 
 class Transport {
  public:
-  virtual bool SendRtp(const uint8_t* packet, size_t length) = 0;
+  virtual bool SendRtp(const uint8_t* packet,
+                       size_t length,
+                       const PacketOptions& options) = 0;
   virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
 
  protected:
   virtual ~Transport() {}
 };
-}  // namespace newapi
+
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_TRANSPORT_H_
+#endif  // WEBRTC_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/typedefs.h b/media/webrtc/trunk/webrtc/typedefs.h
index fa669b8864..a236e87d6c 100644
--- a/media/webrtc/trunk/webrtc/typedefs.h
+++ b/media/webrtc/trunk/webrtc/typedefs.h
@@ -24,6 +24,7 @@
 #define WEBRTC_ARCH_64_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
 #elif defined(__aarch64__)
+#define WEBRTC_ARCH_ARM_FAMILY
 #define WEBRTC_ARCH_64_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
 #elif defined(_M_IX86) || defined(__i386__)
@@ -32,13 +33,7 @@
 #define WEBRTC_ARCH_32_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
 #elif defined(__ARMEL__)
-// TODO(ajm): We'd prefer to control platform defines here, but this is
-// currently provided by the Android makefiles. Commented to avoid duplicate
-// definition warnings.
-//#define WEBRTC_ARCH_ARM
-// TODO(ajm): Chromium uses the following two defines. Should we switch?
-//#define WEBRTC_ARCH_ARM_FAMILY
-//#define WEBRTC_ARCH_ARMEL
+#define WEBRTC_ARCH_ARM_FAMILY
 #define WEBRTC_ARCH_32_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
 #elif defined(__powerpc64__)
@@ -61,7 +56,7 @@
 #define WEBRTC_ARCH_BIG_ENDIAN
 #define WEBRTC_BIG_ENDIAN
 #endif
-#elif defined(__sparc__) && defined(__arch64__)
+#elif defined(__sparc64__)
 #define WEBRTC_ARCH_SPARC 1
 #define WEBRTC_ARCH_64_BITS 1
 #define WEBRTC_ARCH_BIG_ENDIAN
@@ -136,33 +131,25 @@
 #error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
 #endif
 
-#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)) ||  \
-    (defined(WEBRTC_ARCH_ARM_V7) && !defined(WEBRTC_ARCH_ARM_NEON))
+// TODO(zhongwei.yao): WEBRTC_CPU_DETECTION is only used in one place; we should
+// probably just remove it.
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)) || \
+    defined(WEBRTC_DETECT_NEON)
 #define WEBRTC_CPU_DETECTION
 #endif
 
-#if !defined(_MSC_VER)
+// TODO(pbos): Use webrtc/base/basictypes.h instead to include fixed-size ints.
 #include <stdint.h>
-#else
-// Define C99 equivalent types, since pre-2010 MSVC doesn't provide stdint.h.
-typedef signed char         int8_t;
-typedef signed short        int16_t;
-typedef signed int          int32_t;
-typedef __int64             int64_t;
-typedef unsigned char       uint8_t;
-typedef unsigned short      uint16_t;
-typedef unsigned int        uint32_t;
-typedef unsigned __int64    uint64_t;
-#endif
 
 // Annotate a function indicating the caller must examine the return value.
 // Use like:
 //   int foo() WARN_UNUSED_RESULT;
+// To explicitly ignore a result, see |ignore_result()| in <base/macros.h>.
 // TODO(ajm): Hack to avoid multiple definitions until the base/ of webrtc and
 // libjingle are merged.
 #if !defined(WARN_UNUSED_RESULT)
-#if defined(__GNUC__)
-#define WARN_UNUSED_RESULT __attribute__((warn_unused_result))
+#if defined(__GNUC__) || defined(__clang__)
+#define WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
 #else
 #define WARN_UNUSED_RESULT
 #endif
@@ -173,7 +160,7 @@ typedef unsigned __int64    uint64_t;
 //   assert(result == 17);
 #ifndef ATTRIBUTE_UNUSED
 #if defined(__GNUC__) || defined(__clang__)
-#define ATTRIBUTE_UNUSED __attribute__((unused))
+#define ATTRIBUTE_UNUSED __attribute__ ((__unused__))
 #else
 #define ATTRIBUTE_UNUSED
 #endif
@@ -193,7 +180,7 @@ typedef unsigned __int64    uint64_t;
 #if defined(_MSC_VER)
 #define NO_RETURN __declspec(noreturn)
 #elif defined(__GNUC__)
-#define NO_RETURN __attribute__((noreturn))
+#define NO_RETURN __attribute__ ((__noreturn__))
 #else
 #define NO_RETURN
 #endif
diff --git a/media/webrtc/trunk/webrtc/video/BUILD.gn b/media/webrtc/trunk/webrtc/video/BUILD.gn
index ee55105798..e35772e22c 100644
--- a/media/webrtc/trunk/webrtc/video/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/video/BUILD.gn
@@ -10,19 +10,42 @@ import("../build/webrtc.gni")
 
 source_set("video") {
   sources = [
-    "call.cc",
+    "call_stats.cc",
+    "call_stats.h",
     "encoded_frame_callback_adapter.cc",
     "encoded_frame_callback_adapter.h",
+    "encoder_state_feedback.cc",
+    "encoder_state_feedback.h",
+    "overuse_frame_detector.cc",
+    "overuse_frame_detector.h",
+    "payload_router.cc",
+    "payload_router.h",
     "receive_statistics_proxy.cc",
     "receive_statistics_proxy.h",
+    "report_block_stats.cc",
+    "report_block_stats.h",
     "send_statistics_proxy.cc",
     "send_statistics_proxy.h",
-    "transport_adapter.cc",
-    "transport_adapter.h",
+    "stream_synchronization.cc",
+    "stream_synchronization.h",
+    "video_capture_input.cc",
+    "video_capture_input.h",
+    "video_decoder.cc",
+    "video_encoder.cc",
     "video_receive_stream.cc",
     "video_receive_stream.h",
     "video_send_stream.cc",
     "video_send_stream.h",
+    "vie_channel.cc",
+    "vie_channel.h",
+    "vie_encoder.cc",
+    "vie_encoder.h",
+    "vie_receiver.cc",
+    "vie_receiver.h",
+    "vie_remb.cc",
+    "vie_remb.h",
+    "vie_sync_module.cc",
+    "vie_sync_module.h",
   ]
 
   configs += [ "..:common_config" ]
@@ -35,8 +58,18 @@ source_set("video") {
   }
 
   deps = [
+    "..:rtc_event_log",
     "..:webrtc_common",
-    "../video_engine:video_engine_core",
+    "../common_video",
+    "../modules/bitrate_controller",
+    "../modules/pacing",
+    "../modules/rtp_rtcp",
+    "../modules/utility",
+    "../modules/video_capture:video_capture_module",
+    "../modules/video_coding",
+    "../modules/video_processing",
+    "../modules/video_render:video_render_module",
+    "../system_wrappers",
+    "../voice_engine",
   ]
 }
-
diff --git a/media/webrtc/trunk/webrtc/video/OWNERS b/media/webrtc/trunk/webrtc/video/OWNERS
index b5f9aeba7e..3f5e1653ec 100644
--- a/media/webrtc/trunk/webrtc/video/OWNERS
+++ b/media/webrtc/trunk/webrtc/video/OWNERS
@@ -1,6 +1,6 @@
 mflodman@webrtc.org
-stefan@webrtc.org
 pbos@webrtc.org
+stefan@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
diff --git a/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc b/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc
deleted file mode 100644
index c968b71b6b..0000000000
--- a/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc
+++ /dev/null
@@ -1,327 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#include <functional>
-#include <list>
-#include <string>
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/call.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/test/call_test.h"
-#include "webrtc/test/direct_transport.h"
-#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_decoder.h"
-#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/frame_generator_capturer.h"
-
-namespace webrtc {
-namespace {
-// Note: consider to write tests that don't depend on the trace system instead
-// of re-using this class.
-class TraceObserver {
- public:
-  TraceObserver() {
-    Trace::set_level_filter(kTraceTerseInfo);
-
-    Trace::CreateTrace();
-    Trace::SetTraceCallback(&callback_);
-
-    // Call webrtc trace to initialize the tracer that would otherwise trigger a
-    // data-race if left to be initialized by multiple threads (i.e. threads
-    // spawned by test::DirectTransport members in BitrateEstimatorTest).
-    WEBRTC_TRACE(kTraceStateInfo,
-                 kTraceUtility,
-                 -1,
-                 "Instantiate without data races.");
-  }
-
-  ~TraceObserver() {
-    Trace::SetTraceCallback(nullptr);
-    Trace::ReturnTrace();
-  }
-
-  void PushExpectedLogLine(const std::string& expected_log_line) {
-    callback_.PushExpectedLogLine(expected_log_line);
-  }
-
-  EventTypeWrapper Wait() {
-    return callback_.Wait();
-  }
-
- private:
-  class Callback : public TraceCallback {
-   public:
-    Callback()
-        : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-          done_(EventWrapper::Create()) {}
-
-    void Print(TraceLevel level, const char* message, int length) override {
-      CriticalSectionScoped lock(crit_sect_.get());
-      std::string msg(message);
-      if (msg.find("BitrateEstimator") != std::string::npos) {
-        received_log_lines_.push_back(msg);
-      }
-      int num_popped = 0;
-      while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
-        std::string a = received_log_lines_.front();
-        std::string b = expected_log_lines_.front();
-        received_log_lines_.pop_front();
-        expected_log_lines_.pop_front();
-        num_popped++;
-        EXPECT_TRUE(a.find(b) != std::string::npos);
-      }
-      if (expected_log_lines_.size() <= 0) {
-        if (num_popped > 0) {
-          done_->Set();
-        }
-        return;
-      }
-    }
-
-    EventTypeWrapper Wait() {
-      return done_->Wait(test::CallTest::kDefaultTimeoutMs);
-    }
-
-    void PushExpectedLogLine(const std::string& expected_log_line) {
-      CriticalSectionScoped lock(crit_sect_.get());
-      expected_log_lines_.push_back(expected_log_line);
-    }
-
-   private:
-    typedef std::list<std::string> Strings;
-    const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-    Strings received_log_lines_ GUARDED_BY(crit_sect_);
-    Strings expected_log_lines_ GUARDED_BY(crit_sect_);
-    rtc::scoped_ptr<EventWrapper> done_;
-  };
-
-  Callback callback_;
-};
-}  // namespace
-
-static const int kTOFExtensionId = 4;
-static const int kASTExtensionId = 5;
-
-class BitrateEstimatorTest : public test::CallTest {
- public:
-  BitrateEstimatorTest()
-      : receiver_trace_(),
-        send_transport_(),
-        receive_transport_(),
-        sender_call_(),
-        receiver_call_(),
-        receive_config_(),
-        streams_() {
-  }
-
-  virtual ~BitrateEstimatorTest() {
-    EXPECT_TRUE(streams_.empty());
-  }
-
-  virtual void SetUp() {
-    Call::Config receiver_call_config(&receive_transport_);
-    receiver_call_.reset(Call::Create(receiver_call_config));
-
-    Call::Config sender_call_config(&send_transport_);
-    sender_call_.reset(Call::Create(sender_call_config));
-
-    send_transport_.SetReceiver(receiver_call_->Receiver());
-    receive_transport_.SetReceiver(sender_call_->Receiver());
-
-    send_config_ = VideoSendStream::Config();
-    send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
-    // Encoders will be set separately per stream.
-    send_config_.encoder_settings.encoder = nullptr;
-    send_config_.encoder_settings.payload_name = "FAKE";
-    send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
-    encoder_config_.streams = test::CreateVideoStreams(1);
-
-    receive_config_ = VideoReceiveStream::Config();
-    // receive_config_.decoders will be set by every stream separately.
-    receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
-    receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
-    receive_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-    receive_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  }
-
-  virtual void TearDown() {
-    std::for_each(streams_.begin(), streams_.end(),
-        std::mem_fun(&Stream::StopSending));
-
-    send_transport_.StopSending();
-    receive_transport_.StopSending();
-
-    while (!streams_.empty()) {
-      delete streams_.back();
-      streams_.pop_back();
-    }
-
-    receiver_call_.reset();
-  }
-
- protected:
-  friend class Stream;
-
-  class Stream {
-   public:
-    explicit Stream(BitrateEstimatorTest* test)
-        : test_(test),
-          is_sending_receiving_(false),
-          send_stream_(nullptr),
-          receive_stream_(nullptr),
-          frame_generator_capturer_(),
-          fake_encoder_(Clock::GetRealTimeClock()),
-          fake_decoder_() {
-      test_->send_config_.rtp.ssrcs[0]++;
-      test_->send_config_.encoder_settings.encoder = &fake_encoder_;
-      send_stream_ = test_->sender_call_->CreateVideoSendStream(
-          test_->send_config_, test_->encoder_config_);
-      DCHECK_EQ(1u, test_->encoder_config_.streams.size());
-      frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
-          send_stream_->Input(),
-          test_->encoder_config_.streams[0].width,
-          test_->encoder_config_.streams[0].height,
-          30,
-          Clock::GetRealTimeClock()));
-      send_stream_->Start();
-      frame_generator_capturer_->Start();
-
-      VideoReceiveStream::Decoder decoder;
-      decoder.decoder = &fake_decoder_;
-      decoder.payload_type = test_->send_config_.encoder_settings.payload_type;
-      decoder.payload_name = test_->send_config_.encoder_settings.payload_name;
-      test_->receive_config_.decoders.push_back(decoder);
-      test_->receive_config_.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
-      test_->receive_config_.rtp.local_ssrc++;
-      receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
-          test_->receive_config_);
-      receive_stream_->Start();
-
-      is_sending_receiving_ = true;
-    }
-
-    ~Stream() {
-      frame_generator_capturer_.reset(nullptr);
-      test_->sender_call_->DestroyVideoSendStream(send_stream_);
-      send_stream_ = nullptr;
-      test_->receiver_call_->DestroyVideoReceiveStream(receive_stream_);
-      receive_stream_ = nullptr;
-    }
-
-    void StopSending() {
-      if (is_sending_receiving_) {
-        frame_generator_capturer_->Stop();
-        send_stream_->Stop();
-        receive_stream_->Stop();
-        is_sending_receiving_ = false;
-      }
-    }
-
-   private:
-    BitrateEstimatorTest* test_;
-    bool is_sending_receiving_;
-    VideoSendStream* send_stream_;
-    VideoReceiveStream* receive_stream_;
-    rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
-    test::FakeEncoder fake_encoder_;
-    test::FakeDecoder fake_decoder_;
-  };
-
-  TraceObserver receiver_trace_;
-  test::DirectTransport send_transport_;
-  test::DirectTransport receive_transport_;
-  rtc::scoped_ptr<Call> sender_call_;
-  rtc::scoped_ptr<Call> receiver_call_;
-  VideoReceiveStream::Config receive_config_;
-  std::vector<Stream*> streams_;
-};
-
-TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefault) {
-  send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-}
-
-TEST_F(BitrateEstimatorTest, ImmediatelySwitchToAST) {
-  send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(
-      "AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-}
-
-TEST_F(BitrateEstimatorTest, SwitchesToAST) {
-  send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-
-  send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(
-      "AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-}
-
-TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOF) {
-  send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-
-  send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
-  receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
-  receiver_trace_.PushExpectedLogLine(
-      "AbsoluteSendTimeRemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-
-  send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
-  receiver_trace_.PushExpectedLogLine(
-      "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
-  receiver_trace_.PushExpectedLogLine(
-      "RemoteBitrateEstimatorFactory: Instantiating.");
-  streams_.push_back(new Stream(this));
-  streams_[0]->StopSending();
-  streams_[1]->StopSending();
-  EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/call.cc b/media/webrtc/trunk/webrtc/video/call.cc
deleted file mode 100644
index fe9f807326..0000000000
--- a/media/webrtc/trunk/webrtc/video/call.cc
+++ /dev/null
@@ -1,495 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-
-#include <map>
-#include <vector>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/call.h"
-#include "webrtc/common.h"
-#include "webrtc/config.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
-#include "webrtc/video/video_receive_stream.h"
-#include "webrtc/video/video_send_stream.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-
-namespace webrtc {
-const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
-const char* RtpExtension::kAbsSendTime =
-    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
-const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
-const char* RtpExtension::kRtpStreamId =
-  "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
-
-bool RtpExtension::IsSupported(const std::string& name) {
-  return name == webrtc::RtpExtension::kTOffset ||
-         name == webrtc::RtpExtension::kAbsSendTime ||
-         name == webrtc::RtpExtension::kVideoRotation ||
-         name == webrtc::RtpExtension::kRtpStreamId;
-}
-
-VideoEncoder* VideoEncoder::Create(VideoEncoder::EncoderType codec_type) {
-  switch (codec_type) {
-    case kVp8:
-      return VP8Encoder::Create();
-    case kVp9:
-      return VP9Encoder::Create();
-  }
-  RTC_NOTREACHED();
-  return nullptr;
-}
-
-VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) {
-  switch (codec_type) {
-    case kVp8:
-      return VP8Decoder::Create();
-    case kVp9:
-      return VP9Decoder::Create();
-  }
-  RTC_NOTREACHED();
-  return nullptr;
-}
-
-const int Call::Config::kDefaultStartBitrateBps = 300000;
-
-namespace internal {
-
-class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
- public:
-  explicit CpuOveruseObserverProxy(LoadObserver* overuse_callback)
-      : crit_(CriticalSectionWrapper::CreateCriticalSection()),
-        overuse_callback_(overuse_callback) {
-    DCHECK(overuse_callback != nullptr);
-  }
-
-  virtual ~CpuOveruseObserverProxy() {}
-
-  void OveruseDetected() override {
-    CriticalSectionScoped lock(crit_.get());
-    overuse_callback_->OnLoadUpdate(LoadObserver::kOveruse);
-  }
-
-  void NormalUsage() override {
-    CriticalSectionScoped lock(crit_.get());
-    overuse_callback_->OnLoadUpdate(LoadObserver::kUnderuse);
-  }
-
- private:
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-  LoadObserver* overuse_callback_ GUARDED_BY(crit_);
-};
-
-class Call : public webrtc::Call, public PacketReceiver {
- public:
-  Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
-  virtual ~Call();
-
-  PacketReceiver* Receiver() override;
-
-  VideoSendStream* CreateVideoSendStream(
-      const VideoSendStream::Config& config,
-      const VideoEncoderConfig& encoder_config) override;
-
-  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
-
-  VideoReceiveStream* CreateVideoReceiveStream(
-      const VideoReceiveStream::Config& config) override;
-
-  void DestroyVideoReceiveStream(
-      webrtc::VideoReceiveStream* receive_stream) override;
-
-  Stats GetStats() const override;
-
-  DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) override;
-
-  void SetBitrateConfig(
-      const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
-  void SignalNetworkState(NetworkState state) override;
-
- private:
-  DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length);
-  DeliveryStatus DeliverRtp(const uint8_t* packet, size_t length);
-
-  Call::Config config_;
-
-  // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
-  // ensures that we have a consistent network state signalled to all senders
-  // and receivers.
-  rtc::scoped_ptr<CriticalSectionWrapper> network_enabled_crit_;
-  bool network_enabled_ GUARDED_BY(network_enabled_crit_);
-
-  rtc::scoped_ptr<RWLockWrapper> receive_crit_;
-  std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_
-      GUARDED_BY(receive_crit_);
-
-  rtc::scoped_ptr<RWLockWrapper> send_crit_;
-  std::map<uint32_t, VideoSendStream*> send_ssrcs_ GUARDED_BY(send_crit_);
-
-  rtc::scoped_ptr<CpuOveruseObserverProxy> overuse_observer_proxy_;
-
-  VideoSendStream::RtpStateMap suspended_send_ssrcs_;
-
-  VideoEngine* video_engine_;
-  ViERTP_RTCP* rtp_rtcp_;
-  ViECodec* codec_;
-  ViERender* render_;
-  ViEBase* base_;
-  ViENetwork* network_;
-  int base_channel_id_;
-
-  rtc::scoped_ptr<VideoRender> external_render_;
-
-  DISALLOW_COPY_AND_ASSIGN(Call);
-};
-}  // namespace internal
-
-Call* Call::Create(const Call::Config& config) {
-  VideoEngine* video_engine = config.webrtc_config != nullptr
-                                  ? VideoEngine::Create(*config.webrtc_config)
-                                  : VideoEngine::Create();
-  DCHECK(video_engine != nullptr);
-
-  return new internal::Call(video_engine, config);
-}
-
-namespace internal {
-
-Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
-    : config_(config),
-      network_enabled_crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      network_enabled_(true),
-      receive_crit_(RWLockWrapper::CreateRWLock()),
-      send_crit_(RWLockWrapper::CreateRWLock()),
-      video_engine_(video_engine),
-      base_channel_id_(-1),
-      external_render_(
-          VideoRender::CreateVideoRender(42, nullptr, false, kRenderExternal)) {
-  DCHECK(video_engine != nullptr);
-  DCHECK(config.send_transport != nullptr);
-
-  DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
-  DCHECK_GE(config.bitrate_config.start_bitrate_bps,
-            config.bitrate_config.min_bitrate_bps);
-  if (config.bitrate_config.max_bitrate_bps != -1) {
-    DCHECK_GE(config.bitrate_config.max_bitrate_bps,
-              config.bitrate_config.start_bitrate_bps);
-  }
-
-  if (config.overuse_callback) {
-    overuse_observer_proxy_.reset(
-        new CpuOveruseObserverProxy(config.overuse_callback));
-  }
-
-  render_ = ViERender::GetInterface(video_engine_);
-  DCHECK(render_ != nullptr);
-
-  render_->RegisterVideoRenderModule(*external_render_.get());
-
-  rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_);
-  DCHECK(rtp_rtcp_ != nullptr);
-
-  codec_ = ViECodec::GetInterface(video_engine_);
-  DCHECK(codec_ != nullptr);
-
-  network_ = ViENetwork::GetInterface(video_engine_);
-
-  // As a workaround for non-existing calls in the old API, create a base
-  // channel used as default channel when creating send and receive streams.
-  base_ = ViEBase::GetInterface(video_engine_);
-  DCHECK(base_ != nullptr);
-
-  base_->CreateChannel(base_channel_id_);
-  DCHECK(base_channel_id_ != -1);
-
-  network_->SetBitrateConfig(base_channel_id_,
-                             config_.bitrate_config.min_bitrate_bps,
-                             config_.bitrate_config.start_bitrate_bps,
-                             config_.bitrate_config.max_bitrate_bps);
-}
-
-Call::~Call() {
-  CHECK_EQ(0u, send_ssrcs_.size());
-  CHECK_EQ(0u, receive_ssrcs_.size());
-  base_->DeleteChannel(base_channel_id_);
-
-  render_->DeRegisterVideoRenderModule(*external_render_.get());
-
-  base_->Release();
-  network_->Release();
-  codec_->Release();
-  render_->Release();
-  rtp_rtcp_->Release();
-  CHECK(webrtc::VideoEngine::Delete(video_engine_));
-}
-
-PacketReceiver* Call::Receiver() { return this; }
-
-VideoSendStream* Call::CreateVideoSendStream(
-    const VideoSendStream::Config& config,
-    const VideoEncoderConfig& encoder_config) {
-  TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
-  LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
-  DCHECK(!config.rtp.ssrcs.empty());
-
-  // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
-  // the call has already started.
-  VideoSendStream* send_stream = new VideoSendStream(
-      config_.send_transport, overuse_observer_proxy_.get(), video_engine_,
-      config, encoder_config, suspended_send_ssrcs_, base_channel_id_);
-
-  // This needs to be taken before send_crit_ as both locks need to be held
-  // while changing network state.
-  CriticalSectionScoped lock(network_enabled_crit_.get());
-  WriteLockScoped write_lock(*send_crit_);
-  for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
-    DCHECK(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
-    send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
-  }
-  if (!network_enabled_)
-    send_stream->SignalNetworkState(kNetworkDown);
-  return send_stream;
-}
-
-void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
-  TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
-  DCHECK(send_stream != nullptr);
-
-  send_stream->Stop();
-
-  VideoSendStream* send_stream_impl = nullptr;
-  {
-    WriteLockScoped write_lock(*send_crit_);
-    std::map<uint32_t, VideoSendStream*>::iterator it = send_ssrcs_.begin();
-    while (it != send_ssrcs_.end()) {
-      if (it->second == static_cast<VideoSendStream*>(send_stream)) {
-        send_stream_impl = it->second;
-        send_ssrcs_.erase(it++);
-      } else {
-        ++it;
-      }
-    }
-  }
-  CHECK(send_stream_impl != nullptr);
-
-  VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates();
-
-  for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
-       it != rtp_state.end();
-       ++it) {
-    suspended_send_ssrcs_[it->first] = it->second;
-  }
-
-  delete send_stream_impl;
-}
-
-VideoReceiveStream* Call::CreateVideoReceiveStream(
-    const VideoReceiveStream::Config& config) {
-  TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
-  LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
-  VideoReceiveStream* receive_stream =
-      new VideoReceiveStream(video_engine_,
-                             config,
-                             config_.send_transport,
-                             config_.voice_engine,
-                             base_channel_id_);
-
-  // This needs to be taken before receive_crit_ as both locks need to be held
-  // while changing network state.
-  CriticalSectionScoped lock(network_enabled_crit_.get());
-  WriteLockScoped write_lock(*receive_crit_);
-  DCHECK(receive_ssrcs_.find(config.rtp.remote_ssrc) == receive_ssrcs_.end());
-  receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
-  // TODO(pbos): Configure different RTX payloads per receive payload.
-  VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
-      config.rtp.rtx.begin();
-  if (it != config.rtp.rtx.end())
-    receive_ssrcs_[it->second.ssrc] = receive_stream;
-
-  if (!network_enabled_)
-    receive_stream->SignalNetworkState(kNetworkDown);
-  return receive_stream;
-}
-
-void Call::DestroyVideoReceiveStream(
-    webrtc::VideoReceiveStream* receive_stream) {
-  TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
-  DCHECK(receive_stream != nullptr);
-
-  VideoReceiveStream* receive_stream_impl = nullptr;
-  {
-    WriteLockScoped write_lock(*receive_crit_);
-    // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
-    // separate SSRC there can be either one or two.
-    std::map<uint32_t, VideoReceiveStream*>::iterator it =
-        receive_ssrcs_.begin();
-    while (it != receive_ssrcs_.end()) {
-      if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
-        if (receive_stream_impl != nullptr)
-          DCHECK(receive_stream_impl == it->second);
-        receive_stream_impl = it->second;
-        receive_ssrcs_.erase(it++);
-      } else {
-        ++it;
-      }
-    }
-  }
-  CHECK(receive_stream_impl != nullptr);
-  delete receive_stream_impl;
-}
-
-Call::Stats Call::GetStats() const {
-  Stats stats;
-  // Ignoring return values.
-  uint32_t send_bandwidth = 0;
-  rtp_rtcp_->GetEstimatedSendBandwidth(base_channel_id_, &send_bandwidth);
-  stats.send_bandwidth_bps = send_bandwidth;
-  uint32_t recv_bandwidth = 0;
-  rtp_rtcp_->GetEstimatedReceiveBandwidth(base_channel_id_, &recv_bandwidth);
-  stats.recv_bandwidth_bps = recv_bandwidth;
-  {
-    ReadLockScoped read_lock(*send_crit_);
-    for (std::map<uint32_t, VideoSendStream*>::const_iterator it =
-             send_ssrcs_.begin();
-         it != send_ssrcs_.end();
-         ++it) {
-      stats.pacer_delay_ms =
-          std::max(it->second->GetPacerQueuingDelayMs(), stats.pacer_delay_ms);
-      int rtt_ms = it->second->GetRtt();
-      if (rtt_ms > 0)
-        stats.rtt_ms = rtt_ms;
-    }
-  }
-  return stats;
-}
-
-void Call::SetBitrateConfig(
-    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
-  TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
-  DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
-  if (bitrate_config.max_bitrate_bps != -1)
-    DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
-  if (config_.bitrate_config.min_bitrate_bps ==
-          bitrate_config.min_bitrate_bps &&
-      (bitrate_config.start_bitrate_bps <= 0 ||
-       config_.bitrate_config.start_bitrate_bps ==
-           bitrate_config.start_bitrate_bps) &&
-      config_.bitrate_config.max_bitrate_bps ==
-          bitrate_config.max_bitrate_bps) {
-    // Nothing new to set, early abort to avoid encoder reconfigurations.
-    return;
-  }
-  config_.bitrate_config = bitrate_config;
-  network_->SetBitrateConfig(base_channel_id_, bitrate_config.min_bitrate_bps,
-                             bitrate_config.start_bitrate_bps,
-                             bitrate_config.max_bitrate_bps);
-}
-
-void Call::SignalNetworkState(NetworkState state) {
-  // Take crit for entire function, it needs to be held while updating streams
-  // to guarantee a consistent state across streams.
-  CriticalSectionScoped lock(network_enabled_crit_.get());
-  network_enabled_ = state == kNetworkUp;
-  {
-    ReadLockScoped write_lock(*send_crit_);
-    for (std::map<uint32_t, VideoSendStream*>::iterator it =
-             send_ssrcs_.begin();
-         it != send_ssrcs_.end();
-         ++it) {
-      it->second->SignalNetworkState(state);
-    }
-  }
-  {
-    ReadLockScoped write_lock(*receive_crit_);
-    for (std::map<uint32_t, VideoReceiveStream*>::iterator it =
-             receive_ssrcs_.begin();
-         it != receive_ssrcs_.end();
-         ++it) {
-      it->second->SignalNetworkState(state);
-    }
-  }
-}
-
-PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet,
-                                                       size_t length) {
-  // TODO(pbos): Figure out what channel needs it actually.
-  //             Do NOT broadcast! Also make sure it's a valid packet.
-  //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
-  //             there's no receiver of the packet.
-  bool rtcp_delivered = false;
-  {
-    ReadLockScoped read_lock(*receive_crit_);
-    for (std::map<uint32_t, VideoReceiveStream*>::iterator it =
-             receive_ssrcs_.begin();
-         it != receive_ssrcs_.end();
-         ++it) {
-      if (it->second->DeliverRtcp(packet, length))
-        rtcp_delivered = true;
-    }
-  }
-
-  {
-    ReadLockScoped read_lock(*send_crit_);
-    for (std::map<uint32_t, VideoSendStream*>::iterator it =
-             send_ssrcs_.begin();
-         it != send_ssrcs_.end();
-         ++it) {
-      if (it->second->DeliverRtcp(packet, length))
-        rtcp_delivered = true;
-    }
-  }
-  return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
-}
-
-PacketReceiver::DeliveryStatus Call::DeliverRtp(const uint8_t* packet,
-                                                size_t length) {
-  // Minimum RTP header size.
-  if (length < 12)
-    return DELIVERY_PACKET_ERROR;
-
-  uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
-
-  ReadLockScoped read_lock(*receive_crit_);
-  std::map<uint32_t, VideoReceiveStream*>::iterator it =
-      receive_ssrcs_.find(ssrc);
-
-  if (it == receive_ssrcs_.end())
-    return DELIVERY_UNKNOWN_SSRC;
-
-  return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
-                                                : DELIVERY_PACKET_ERROR;
-}
-
-PacketReceiver::DeliveryStatus Call::DeliverPacket(const uint8_t* packet,
-                                                   size_t length) {
-  if (RtpHeaderParser::IsRtcp(packet, length))
-    return DeliverRtcp(packet, length);
-
-  return DeliverRtp(packet, length);
-}
-
-}  // namespace internal
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/call_stats.cc b/media/webrtc/trunk/webrtc/video/call_stats.cc
similarity index 84%
rename from media/webrtc/trunk/webrtc/video_engine/call_stats.cc
rename to media/webrtc/trunk/webrtc/video/call_stats.cc
index 6e51eebe5a..69ea1a3d78 100644
--- a/media/webrtc/trunk/webrtc/video_engine/call_stats.cc
+++ b/media/webrtc/trunk/webrtc/video/call_stats.cc
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/call_stats.h"
+#include "webrtc/video/call_stats.h"
 
 #include <assert.h>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include <algorithm>
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
@@ -86,29 +88,28 @@ class RtcpObserver : public RtcpRttStats {
  private:
   CallStats* owner_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
 };
 
-CallStats::CallStats()
-    : crit_(CriticalSectionWrapper::CreateCriticalSection()),
+CallStats::CallStats(Clock* clock)
+    : clock_(clock),
+      crit_(CriticalSectionWrapper::CreateCriticalSection()),
       rtcp_rtt_stats_(new RtcpObserver(this)),
-      last_process_time_(TickTime::MillisecondTimestamp()),
+      last_process_time_(clock_->TimeInMilliseconds()),
       max_rtt_ms_(0),
-      avg_rtt_ms_(0) {
-}
+      avg_rtt_ms_(0) {}
 
 CallStats::~CallStats() {
   assert(observers_.empty());
 }
 
 int64_t CallStats::TimeUntilNextProcess() {
-  return last_process_time_ + kUpdateIntervalMs -
-      TickTime::MillisecondTimestamp();
+  return last_process_time_ + kUpdateIntervalMs - clock_->TimeInMilliseconds();
 }
 
 int32_t CallStats::Process() {
   CriticalSectionScoped cs(crit_.get());
-  int64_t now = TickTime::MillisecondTimestamp();
+  int64_t now = clock_->TimeInMilliseconds();
   if (now < last_process_time_ + kUpdateIntervalMs)
     return 0;
 
@@ -123,7 +124,7 @@ int32_t CallStats::Process() {
   if (max_rtt_ms_ > 0) {
     for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
          it != observers_.end(); ++it) {
-      (*it)->OnRttUpdate(max_rtt_ms_);
+      (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_);
     }
   }
   return 0;
@@ -161,7 +162,7 @@ void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) {
 
 void CallStats::OnRttUpdate(int64_t rtt) {
   CriticalSectionScoped cs(crit_.get());
-  reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp()));
+  reports_.push_back(RttTime(rtt, clock_->TimeInMilliseconds()));
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/call_stats.h b/media/webrtc/trunk/webrtc/video/call_stats.h
similarity index 87%
rename from media/webrtc/trunk/webrtc/video_engine/call_stats.h
rename to media/webrtc/trunk/webrtc/video/call_stats.h
index 79a8af6267..4ecd911b07 100644
--- a/media/webrtc/trunk/webrtc/video_engine/call_stats.h
+++ b/media/webrtc/trunk/webrtc/video/call_stats.h
@@ -8,14 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
-#define WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
+#ifndef WEBRTC_VIDEO_CALL_STATS_H_
+#define WEBRTC_VIDEO_CALL_STATS_H_
 
 #include <list>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -28,7 +29,7 @@ class CallStats : public Module {
  public:
   friend class RtcpObserver;
 
-  CallStats();
+  explicit CallStats(Clock* clock);
   ~CallStats();
 
   // Implements Module, to use the process thread.
@@ -57,6 +58,7 @@ class CallStats : public Module {
   int64_t avg_rtt_ms() const;
 
  private:
+  Clock* const clock_;
   // Protecting all members.
   rtc::scoped_ptr<CriticalSectionWrapper> crit_;
   // Observer receiving statistics updates.
@@ -73,9 +75,9 @@ class CallStats : public Module {
   // Observers getting stats reports.
   std::list<CallStatsObserver*> observers_;
 
-  DISALLOW_COPY_AND_ASSIGN(CallStats);
+  RTC_DISALLOW_COPY_AND_ASSIGN(CallStats);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_CALL_STATS_H_
+#endif  // WEBRTC_VIDEO_CALL_STATS_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc b/media/webrtc/trunk/webrtc/video/call_stats_unittest.cc
similarity index 68%
rename from media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc
rename to media/webrtc/trunk/webrtc/video/call_stats_unittest.cc
index 0febbd0198..6226a5bf6e 100644
--- a/media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/call_stats_unittest.cc
@@ -12,9 +12,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/call_stats.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video/call_stats.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
@@ -27,15 +27,16 @@ class MockStatsObserver : public CallStatsObserver {
   MockStatsObserver() {}
   virtual ~MockStatsObserver() {}
 
-  MOCK_METHOD1(OnRttUpdate, void(int64_t));
+  MOCK_METHOD2(OnRttUpdate, void(int64_t, int64_t));
 };
 
 class CallStatsTest : public ::testing::Test {
+ public:
+  CallStatsTest() : fake_clock_(12345) {}
+
  protected:
-  virtual void SetUp() {
-    TickTime::UseFakeClock(12345);
-    call_stats_.reset(new CallStats());
-  }
+  virtual void SetUp() { call_stats_.reset(new CallStats(&fake_clock_)); }
+  SimulatedClock fake_clock_;
   rtc::scoped_ptr<CallStats> call_stats_;
 };
 
@@ -43,20 +44,18 @@ TEST_F(CallStatsTest, AddAndTriggerCallback) {
   MockStatsObserver stats_observer;
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
   call_stats_->RegisterStatsObserver(&stats_observer);
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
 
   const int64_t kRtt = 25;
   rtcp_rtt_stats->OnRttUpdate(kRtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt))
-      .Times(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt)).Times(1);
   call_stats_->Process();
   EXPECT_EQ(kRtt, rtcp_rtt_stats->LastProcessedRtt());
 
   const int64_t kRttTimeOutMs = 1500 + 10;
-  TickTime::AdvanceFakeClock(kRttTimeOutMs);
-  EXPECT_CALL(stats_observer, OnRttUpdate(_))
-      .Times(0);
+  fake_clock_.AdvanceTimeMilliseconds(kRttTimeOutMs);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
   call_stats_->Process();
   EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
 
@@ -70,27 +69,23 @@ TEST_F(CallStatsTest, ProcessTime) {
   rtcp_rtt_stats->OnRttUpdate(100);
 
   // Time isn't updated yet.
-  EXPECT_CALL(stats_observer, OnRttUpdate(_))
-      .Times(0);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
   call_stats_->Process();
 
   // Advance clock and verify we get an update.
-  TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer, OnRttUpdate(_))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(1);
   call_stats_->Process();
 
   // Advance clock just too little to get an update.
-  TickTime::AdvanceFakeClock(999);
+  fake_clock_.AdvanceTimeMilliseconds(999);
   rtcp_rtt_stats->OnRttUpdate(100);
-  EXPECT_CALL(stats_observer, OnRttUpdate(_))
-      .Times(0);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
   call_stats_->Process();
 
   // Advance enough to trigger a new update.
-  TickTime::AdvanceFakeClock(1);
-  EXPECT_CALL(stats_observer, OnRttUpdate(_))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(1);
   call_stats_->Process();
 
   call_stats_->DeregisterStatsObserver(&stats_observer);
@@ -112,32 +107,26 @@ TEST_F(CallStatsTest, MultipleObservers) {
   rtcp_rtt_stats->OnRttUpdate(kRtt);
 
   // Verify both observers are updated.
-  TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt))
-      .Times(1);
-  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(1);
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(1);
   call_stats_->Process();
 
   // Deregister the second observer and verify update is only sent to the first
   // observer.
   call_stats_->DeregisterStatsObserver(&stats_observer_2);
   rtcp_rtt_stats->OnRttUpdate(kRtt);
-  TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt))
-      .Times(1);
-  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt))
-      .Times(0);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(1);
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(0);
   call_stats_->Process();
 
   // Deregister the first observer.
   call_stats_->DeregisterStatsObserver(&stats_observer_1);
   rtcp_rtt_stats->OnRttUpdate(kRtt);
-  TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt))
-      .Times(0);
-  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt))
-      .Times(0);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(0);
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(0);
   call_stats_->Process();
 }
 
@@ -148,38 +137,37 @@ TEST_F(CallStatsTest, ChangeRtt) {
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
 
   // Advance clock to be ready for an update.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
 
   // Set a first value and verify the callback is triggered.
   const int64_t kFirstRtt = 100;
   rtcp_rtt_stats->OnRttUpdate(kFirstRtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kFirstRtt))
-      .Times(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kFirstRtt, kFirstRtt)).Times(1);
   call_stats_->Process();
 
   // Increase rtt and verify the new value is reported.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   const int64_t kHighRtt = kFirstRtt + 20;
+  const int64_t kAvgRtt1 = 103;
   rtcp_rtt_stats->OnRttUpdate(kHighRtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kHighRtt))
-      .Times(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt1, kHighRtt)).Times(1);
   call_stats_->Process();
 
   // Increase time enough for a new update, but not too much to make the
   // rtt invalid. Report a lower rtt and verify the old/high value still is sent
   // in the callback.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   const int64_t kLowRtt = kFirstRtt - 20;
+  const int64_t kAvgRtt2 = 102;
   rtcp_rtt_stats->OnRttUpdate(kLowRtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kHighRtt))
-      .Times(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt2, kHighRtt)).Times(1);
   call_stats_->Process();
 
   // Advance time to make the high report invalid, the lower rtt should now be
   // in the callback.
-  TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kLowRtt))
-      .Times(1);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
+  const int64_t kAvgRtt3 = 95;
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt3, kLowRtt)).Times(1);
   call_stats_->Process();
 
   call_stats_->DeregisterStatsObserver(&stats_observer);
@@ -189,7 +177,7 @@ TEST_F(CallStatsTest, LastProcessedRtt) {
   MockStatsObserver stats_observer;
   call_stats_->RegisterStatsObserver(&stats_observer);
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
 
   // Set a first values and verify that LastProcessedRtt initially returns the
   // average rtt.
@@ -198,17 +186,15 @@ TEST_F(CallStatsTest, LastProcessedRtt) {
   const int64_t kAvgRtt = 20;
   rtcp_rtt_stats->OnRttUpdate(kRttLow);
   rtcp_rtt_stats->OnRttUpdate(kRttHigh);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kRttHigh))
-      .Times(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
   call_stats_->Process();
   EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
 
   // Update values and verify LastProcessedRtt.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   rtcp_rtt_stats->OnRttUpdate(kRttLow);
   rtcp_rtt_stats->OnRttUpdate(kRttHigh);
-  EXPECT_CALL(stats_observer, OnRttUpdate(kRttHigh))
-      .Times(1);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
   call_stats_->Process();
   EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
 
diff --git a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc
index 1261ad5123..4c6823fa47 100644
--- a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc
+++ b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc
@@ -11,7 +11,7 @@
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
 
 namespace webrtc {
 namespace internal {
@@ -26,12 +26,9 @@ int32_t EncodedFrameCallbackAdapter::Encoded(
     const EncodedImage& encodedImage,
     const CodecSpecificInfo* codecSpecificInfo,
     const RTPFragmentationHeader* fragmentation) {
-  DCHECK(observer_ != nullptr);
-  FrameType frame_type =
-        VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
-  const EncodedFrame frame(encodedImage._buffer,
-                           encodedImage._length,
-                           frame_type);
+  RTC_DCHECK(observer_ != nullptr);
+  const EncodedFrame frame(encodedImage._buffer, encodedImage._length,
+                           encodedImage._frameType);
   observer_->EncodedFrameCallback(frame);
   return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h
index b39a8e2167..b10c4f1645 100644
--- a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h
+++ b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_VIDEO_ENCODED_FRAME_CALLBACK_ADAPTER_H_
 #define WEBRTC_VIDEO_ENCODED_FRAME_CALLBACK_ADAPTER_H_
 
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/frame_callback.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.cc b/media/webrtc/trunk/webrtc/video/encoder_state_feedback.cc
similarity index 86%
rename from media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.cc
rename to media/webrtc/trunk/webrtc/video/encoder_state_feedback.cc
index e5d6bd69de..c0c4b67dbd 100644
--- a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.cc
+++ b/media/webrtc/trunk/webrtc/video/encoder_state_feedback.cc
@@ -8,13 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/encoder_state_feedback.h"
+#include "webrtc/video/encoder_state_feedback.h"
 
 #include <assert.h>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/video_engine/vie_encoder.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/video/vie_encoder.h"
 
 namespace webrtc {
 
@@ -53,15 +54,14 @@ EncoderStateFeedback::~EncoderStateFeedback() {
   assert(encoders_.empty());
 }
 
-bool EncoderStateFeedback::AddEncoder(uint32_t ssrc, ViEEncoder* encoder)  {
+void EncoderStateFeedback::AddEncoder(const std::vector<uint32_t>& ssrcs,
+                                      ViEEncoder* encoder) {
+  RTC_DCHECK(!ssrcs.empty());
   CriticalSectionScoped lock(crit_.get());
-  if (encoders_.find(ssrc) != encoders_.end()) {
-    // Two encoders must not have the same ssrc.
-    return false;
+  for (uint32_t ssrc : ssrcs) {
+    RTC_DCHECK(encoders_.find(ssrc) == encoders_.end());
+    encoders_[ssrc] = encoder;
   }
-
-  encoders_[ssrc] = encoder;
-  return true;
 }
 
 void EncoderStateFeedback::RemoveEncoder(const ViEEncoder* encoder)  {
diff --git a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h b/media/webrtc/trunk/webrtc/video/encoder_state_feedback.h
similarity index 84%
rename from media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h
rename to media/webrtc/trunk/webrtc/video/encoder_state_feedback.h
index 998793a254..620e382d89 100644
--- a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h
+++ b/media/webrtc/trunk/webrtc/video/encoder_state_feedback.h
@@ -11,10 +11,11 @@
 // TODO(mflodman) ViEEncoder has a time check to not send key frames too often,
 // move the logic to this class.
 
-#ifndef WEBRTC_VIDEO_ENGINE_ENCODER_STATE_FEEDBACK_H_
-#define WEBRTC_VIDEO_ENGINE_ENCODER_STATE_FEEDBACK_H_
+#ifndef WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
+#define WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
 
 #include <map>
+#include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -34,8 +35,8 @@ class EncoderStateFeedback {
   EncoderStateFeedback();
   ~EncoderStateFeedback();
 
-  // Adds an encoder to receive feedback for a unique ssrc.
-  bool AddEncoder(uint32_t ssrc, ViEEncoder* encoder);
+  // Adds an encoder to receive feedback for a set of SSRCs.
+  void AddEncoder(const std::vector<uint32_t>& ssrc, ViEEncoder* encoder);
 
   // Removes a registered ViEEncoder.
   void RemoveEncoder(const ViEEncoder* encoder);
@@ -62,9 +63,9 @@ class EncoderStateFeedback {
   // Maps a unique ssrc to the given encoder.
   SsrcEncoderMap encoders_;
 
-  DISALLOW_COPY_AND_ASSIGN(EncoderStateFeedback);
+  RTC_DISALLOW_COPY_AND_ASSIGN(EncoderStateFeedback);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_ENCODER_STATE_FEEDBACK_H_
+#endif  // WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc b/media/webrtc/trunk/webrtc/video/encoder_state_feedback_unittest.cc
similarity index 81%
rename from media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc
rename to media/webrtc/trunk/webrtc/video/encoder_state_feedback_unittest.cc
index eab33d407a..834447e513 100644
--- a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/encoder_state_feedback_unittest.cc
@@ -10,19 +10,20 @@
 
 
 // This file includes unit tests for EncoderStateFeedback.
-#include "webrtc/video_engine/encoder_state_feedback.h"
+#include "webrtc/video/encoder_state_feedback.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
-#include "webrtc/video_engine/payload_router.h"
-#include "webrtc/video_engine/vie_encoder.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/vie_encoder.h"
 
 using ::testing::NiceMock;
 
@@ -31,7 +32,7 @@ namespace webrtc {
 class MockVieEncoder : public ViEEncoder {
  public:
   explicit MockVieEncoder(ProcessThread* process_thread, PacedSender* pacer)
-      : ViEEncoder(1, 1, config_, *process_thread, pacer, NULL, NULL, false) {}
+      : ViEEncoder(1, process_thread, nullptr, nullptr, pacer, nullptr) {}
   ~MockVieEncoder() {}
 
   MOCK_METHOD1(OnReceivedIntraFrameRequest,
@@ -42,8 +43,6 @@ class MockVieEncoder : public ViEEncoder {
                void(uint32_t ssrc, uint64_t picture_id));
   MOCK_METHOD2(OnLocalSsrcChanged,
                void(uint32_t old_ssrc, uint32_t new_ssrc));
-
-  const Config config_;
 };
 
 class VieKeyRequestTest : public ::testing::Test {
@@ -68,7 +67,7 @@ class VieKeyRequestTest : public ::testing::Test {
 TEST_F(VieKeyRequestTest, CreateAndTriggerRequests) {
   const int ssrc = 1234;
   MockVieEncoder encoder(process_thread_.get(), &pacer_);
-  EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
+  encoder_state_feedback_->AddEncoder(std::vector<uint32_t>(1, ssrc), &encoder);
 
   EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(ssrc))
       .Times(1);
@@ -97,8 +96,10 @@ TEST_F(VieKeyRequestTest, MultipleEncoders) {
   const int ssrc_2 = 5678;
   MockVieEncoder encoder_1(process_thread_.get(), &pacer_);
   MockVieEncoder encoder_2(process_thread_.get(), &pacer_);
-  EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc_1, &encoder_1));
-  EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc_2, &encoder_2));
+  encoder_state_feedback_->AddEncoder(std::vector<uint32_t>(1, ssrc_1),
+                                      &encoder_1);
+  encoder_state_feedback_->AddEncoder(std::vector<uint32_t>(1, ssrc_2),
+                                      &encoder_2);
 
   EXPECT_CALL(encoder_1, OnReceivedIntraFrameRequest(ssrc_1))
       .Times(1);
@@ -139,12 +140,4 @@ TEST_F(VieKeyRequestTest, MultipleEncoders) {
   encoder_state_feedback_->RemoveEncoder(&encoder_2);
 }
 
-TEST_F(VieKeyRequestTest, AddTwiceError) {
-  const int ssrc = 1234;
-  MockVieEncoder encoder(process_thread_.get(), &pacer_);
-  EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
-  EXPECT_FALSE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
-  encoder_state_feedback_->RemoveEncoder(&encoder);
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc b/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc
index 956e8219fa..51d1d2c3fd 100644
--- a/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc
@@ -15,21 +15,22 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
+#include "webrtc/call/transport_adapter.h"
 #include "webrtc/frame_callback.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
@@ -39,28 +40,28 @@
 #include "webrtc/test/rtcp_packet_parser.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video/transport_adapter.h"
 #include "webrtc/video_encoder.h"
 
 namespace webrtc {
 
-static const unsigned long kSilenceTimeoutMs = 2000;
+static const int kSilenceTimeoutMs = 2000;
 
 class EndToEndTest : public test::CallTest {
  public:
   EndToEndTest() {}
 
   virtual ~EndToEndTest() {
-    EXPECT_EQ(nullptr, send_stream_);
-    EXPECT_TRUE(receive_streams_.empty());
+    EXPECT_EQ(nullptr, video_send_stream_);
+    EXPECT_TRUE(video_receive_streams_.empty());
   }
 
  protected:
-  class UnusedTransport : public newapi::Transport {
+  class UnusedTransport : public Transport {
    private:
-    bool SendRtp(const uint8_t* packet, size_t length) override {
+    bool SendRtp(const uint8_t* packet,
+                 size_t length,
+                 const PacketOptions& options) override {
       ADD_FAILURE() << "Unexpected RTP sent.";
       return false;
     }
@@ -71,41 +72,41 @@ class EndToEndTest : public test::CallTest {
     }
   };
 
-  void DecodesRetransmittedFrame(bool retransmit_over_rtx);
+  void DecodesRetransmittedFrame(bool use_rtx, bool use_red);
   void ReceivesPliAndRecovers(int rtp_history_ms);
-  void RespectsRtcpMode(newapi::RtcpMode rtcp_mode);
+  void RespectsRtcpMode(RtcpMode rtcp_mode);
   void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
   void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
   void TestRtpStatePreservation(bool use_rtx);
-  void TestReceivedFecPacketsNotNacked(const FakeNetworkPipe::Config& config);
+  void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
 };
 
 TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
+  CreateCalls(Call::Config(), Call::Config());
+
   test::NullTransport transport;
-  CreateCalls(Call::Config(&transport), Call::Config(&transport));
+  CreateSendConfig(1, 0, &transport);
+  CreateMatchingReceiveConfigs(&transport);
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateVideoStreams();
 
-  CreateStreams();
-
-  receive_streams_[0]->Start();
-  receive_streams_[0]->Start();
+  video_receive_streams_[0]->Start();
+  video_receive_streams_[0]->Start();
 
   DestroyStreams();
 }
 
 TEST_F(EndToEndTest, ReceiverCanBeStoppedTwice) {
+  CreateCalls(Call::Config(), Call::Config());
+
   test::NullTransport transport;
-  CreateCalls(Call::Config(&transport), Call::Config(&transport));
+  CreateSendConfig(1, 0, &transport);
+  CreateMatchingReceiveConfigs(&transport);
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateVideoStreams();
 
-  CreateStreams();
-
-  receive_streams_[0]->Stop();
-  receive_streams_[0]->Stop();
+  video_receive_streams_[0]->Stop();
+  video_receive_streams_[0]->Stop();
 
   DestroyStreams();
 }
@@ -120,61 +121,61 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
 
   class Renderer : public VideoRenderer {
    public:
-    Renderer() : event_(EventWrapper::Create()) {}
+    Renderer() : event_(false, false) {}
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int /*time_to_render_ms*/) override {
-      event_->Set();
+      event_.Set();
     }
 
     bool IsTextureSupported() const override { return false; }
 
-    EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
 
-    rtc::scoped_ptr<EventWrapper> event_;
+    rtc::Event event_;
   } renderer;
 
   class TestFrameCallback : public I420FrameCallback {
    public:
-    TestFrameCallback() : event_(EventWrapper::Create()) {}
+    TestFrameCallback() : event_(false, false) {}
 
-    EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
 
    private:
-    void FrameCallback(I420VideoFrame* frame) override {
+    void FrameCallback(VideoFrame* frame) override {
       SleepMs(kDelayRenderCallbackMs);
-      event_->Set();
+      event_.Set();
     }
 
-    rtc::scoped_ptr<EventWrapper> event_;
+    rtc::Event event_;
   };
 
-  test::DirectTransport sender_transport, receiver_transport;
-
-  CreateCalls(Call::Config(&sender_transport),
-              Call::Config(&receiver_transport));
+  CreateCalls(Call::Config(), Call::Config());
 
+  test::DirectTransport sender_transport(sender_call_.get());
+  test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(&receiver_transport);
 
   TestFrameCallback pre_render_callback;
-  receive_configs_[0].pre_render_callback = &pre_render_callback;
-  receive_configs_[0].renderer = &renderer;
+  video_receive_configs_[0].pre_render_callback = &pre_render_callback;
+  video_receive_configs_[0].renderer = &renderer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(kWidth, kHeight));
-  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
-  EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
+  video_send_stream_->Input()->IncomingCapturedFrame(
+      *frame_generator->NextFrame());
+  EXPECT_TRUE(pre_render_callback.Wait())
       << "Timed out while waiting for pre-render callback.";
-  EXPECT_EQ(kEventSignaled, renderer.Wait())
+  EXPECT_TRUE(renderer.Wait())
       << "Timed out while waiting for the frame to render.";
 
   Stop();
@@ -188,40 +189,41 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
 TEST_F(EndToEndTest, TransmitsFirstFrame) {
   class Renderer : public VideoRenderer {
    public:
-    Renderer() : event_(EventWrapper::Create()) {}
+    Renderer() : event_(false, false) {}
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int /*time_to_render_ms*/) override {
-      event_->Set();
+      event_.Set();
     }
     bool IsTextureSupported() const override { return false; }
 
-    EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
 
-    rtc::scoped_ptr<EventWrapper> event_;
+    rtc::Event event_;
   } renderer;
 
-  test::DirectTransport sender_transport, receiver_transport;
-
-  CreateCalls(Call::Config(&sender_transport),
-              Call::Config(&receiver_transport));
+  CreateCalls(Call::Config(), Call::Config());
 
+  test::DirectTransport sender_transport(sender_call_.get());
+  test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
-  receive_configs_[0].renderer = &renderer;
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(&receiver_transport);
+  video_receive_configs_[0].renderer = &renderer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(
-          encoder_config_.streams[0].width, encoder_config_.streams[0].height));
-  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
+          video_encoder_config_.streams[0].width,
+          video_encoder_config_.streams[0].height));
+  video_send_stream_->Input()->IncomingCapturedFrame(
+      *frame_generator->NextFrame());
 
-  EXPECT_EQ(kEventSignaled, renderer.Wait())
+  EXPECT_TRUE(renderer.Wait())
       << "Timed out while waiting for the frame to render.";
 
   Stop();
@@ -242,16 +244,17 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) {
           frame_counter_(0) {}
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for enough frames to be decoded.";
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "VP9";
-      send_config->encoder_settings.payload_type = VCM_VP9_PAYLOAD_TYPE;
+      send_config->encoder_settings.payload_type = 124;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
           encoder_config->streams[0].max_bitrate_bps = 2000000;
@@ -265,11 +268,11 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) {
       (*receive_configs)[0].decoders[0].decoder = decoder_.get();
     }
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
       const int kRequiredFrames = 500;
       if (++frame_counter_ == kRequiredFrames)
-        observation_complete_->Set();
+        observation_complete_.Set();
     }
 
     bool IsTextureSupported() const override { return false; }
@@ -292,18 +295,19 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
           frame_counter_(0) {}
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for enough frames to be decoded.";
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->encoder_settings.encoder = &fake_encoder_;
       send_config->encoder_settings.payload_name = "H264";
-      send_config->encoder_settings.payload_type = kFakeSendPayloadType;
+      send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
           encoder_config->streams[0].max_bitrate_bps = 2000000;
@@ -317,11 +321,11 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
       (*receive_configs)[0].decoders[0].decoder = &fake_decoder_;
     }
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
       const int kRequiredFrames = 500;
       if (++frame_counter_ == kRequiredFrames)
-        observation_complete_->Set();
+        observation_complete_.Set();
     }
 
     bool IsTextureSupported() const override { return false; }
@@ -348,14 +352,14 @@ TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
       ssrc |= static_cast<uint32_t>(packet[5]) << 16;
       ssrc |= static_cast<uint32_t>(packet[6]) << 8;
       ssrc |= static_cast<uint32_t>(packet[7]) << 0;
-      EXPECT_EQ(kReceiverLocalSsrc, ssrc);
-      observation_complete_->Set();
+      EXPECT_EQ(kReceiverLocalVideoSsrc, ssrc);
+      observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for a receiver RTCP packet to be sent.";
     }
   } test;
@@ -371,23 +375,23 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
    public:
     NackObserver()
         : EndToEndTest(kLongTimeoutMs),
-          rtp_parser_(RtpHeaderParser::Create()),
           sent_rtp_packets_(0),
           packets_left_to_drop_(0),
           nacks_left_(kNumberOfNacksToObserve) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
-      EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       // Never drop retransmitted packets.
       if (dropped_packets_.find(header.sequenceNumber) !=
           dropped_packets_.end()) {
         retransmitted_packets_.insert(header.sequenceNumber);
-        if (nacks_left_ == 0 &&
+        if (nacks_left_ <= 0 &&
             retransmitted_packets_.size() == dropped_packets_.size()) {
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
         return SEND_PACKET;
       }
@@ -395,14 +399,15 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
       ++sent_rtp_packets_;
 
       // Enough NACKs received, stop dropping packets.
-      if (nacks_left_ == 0)
+      if (nacks_left_ <= 0)
         return SEND_PACKET;
 
       // Check if it's time for a new loss burst.
       if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
         packets_left_to_drop_ = kLossBurstSize;
 
-      if (packets_left_to_drop_ > 0) {
+      // Never drop padding packets as those won't be retransmitted.
+      if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
         --packets_left_to_drop_;
         dropped_packets_.insert(header.sequenceNumber);
         return DROP_PACKET;
@@ -412,12 +417,13 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kRtpfbNack) {
           --nacks_left_;
           break;
         }
@@ -426,25 +432,26 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out waiting for packets to be NACKed, retransmitted and "
              "rendered.";
     }
 
-    rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
+    rtc::CriticalSection crit_;
     std::set<uint16_t> dropped_packets_;
     std::set<uint16_t> retransmitted_packets_;
     uint64_t sent_rtp_packets_;
     int packets_left_to_drop_;
-    int nacks_left_;
+    int nacks_left_ GUARDED_BY(&crit_);
   } test;
 
   RunBaseTest(&test);
@@ -457,16 +464,20 @@ TEST_F(EndToEndTest, CanReceiveFec) {
         : EndToEndTest(kDefaultTimeoutMs), state_(kFirstPacket) {}
 
    private:
-    Action OnSendRtp(const uint8_t* packet, size_t length) override
-        EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
-      EXPECT_EQ(kRedPayloadType, header.payloadType);
-      int encapsulated_payload_type =
-          static_cast<int>(packet[header.headerLength]);
-      if (encapsulated_payload_type != kFakeSendPayloadType)
-        EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
+      int encapsulated_payload_type = -1;
+      if (header.payloadType == kRedPayloadType) {
+        encapsulated_payload_type =
+            static_cast<int>(packet[header.headerLength]);
+        if (encapsulated_payload_type != kFakeVideoSendPayloadType)
+          EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
+      } else {
+        EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
+      }
 
       if (protected_sequence_numbers_.count(header.sequenceNumber) != 0) {
         // Retransmitted packet, should not count.
@@ -489,7 +500,7 @@ TEST_F(EndToEndTest, CanReceiveFec) {
             return DROP_PACKET;
           break;
         case kDropNextMediaPacket:
-          if (encapsulated_payload_type == kFakeSendPayloadType) {
+          if (encapsulated_payload_type == kFakeVideoSendPayloadType) {
             protected_sequence_numbers_.insert(header.sequenceNumber);
             protected_timestamps_.insert(header.timestamp);
             state_ = kDropEveryOtherPacketUntilFec;
@@ -501,13 +512,13 @@ TEST_F(EndToEndTest, CanReceiveFec) {
       return SEND_PACKET;
     }
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       // Rendering frame with timestamp of packet that was dropped -> FEC
       // protection worked.
       if (protected_timestamps_.count(video_frame.timestamp()) != 0)
-        observation_complete_->Set();
+        observation_complete_.Set();
     }
 
     bool IsTextureSupported() const override { return false; }
@@ -518,9 +529,10 @@ TEST_F(EndToEndTest, CanReceiveFec) {
       kDropNextMediaPacket,
     } state_;
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       // TODO(pbos): Run this test with combined NACK/FEC enabled as well.
       // int rtp_history_ms = 1000;
       // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms;
@@ -534,10 +546,11 @@ TEST_F(EndToEndTest, CanReceiveFec) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out waiting for dropped frames frames to be rendered.";
     }
 
+    rtc::CriticalSection crit_;
     std::set<uint32_t> protected_sequence_numbers_ GUARDED_BY(crit_);
     std::set<uint32_t> protected_timestamps_ GUARDED_BY(crit_);
   } test;
@@ -547,20 +560,10 @@ TEST_F(EndToEndTest, CanReceiveFec) {
 
 // Flacky on all platforms. See webrtc:4328.
 TEST_F(EndToEndTest, DISABLED_ReceivedFecPacketsNotNacked) {
-  // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
-  // Configure some network delay.
-  const int kNetworkDelayMs = 50;
-  FakeNetworkPipe::Config config;
-  config.queue_delay_ms = kNetworkDelayMs;
-  TestReceivedFecPacketsNotNacked(config);
-}
-
-void EndToEndTest::TestReceivedFecPacketsNotNacked(
-    const FakeNetworkPipe::Config& config) {
   class FecNackObserver : public test::EndToEndTest {
    public:
-    explicit FecNackObserver(const FakeNetworkPipe::Config& config)
-        : EndToEndTest(kDefaultTimeoutMs, config),
+    FecNackObserver()
+        : EndToEndTest(kDefaultTimeoutMs),
           state_(kFirstPacket),
           fec_sequence_number_(0),
           has_last_sequence_number_(false),
@@ -568,14 +571,19 @@ void EndToEndTest::TestReceivedFecPacketsNotNacked(
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock_(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
-      EXPECT_EQ(kRedPayloadType, header.payloadType);
 
-      int encapsulated_payload_type =
-          static_cast<int>(packet[header.headerLength]);
-      if (encapsulated_payload_type != kFakeSendPayloadType)
-        EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
+      int encapsulated_payload_type = -1;
+      if (header.payloadType == kRedPayloadType) {
+        encapsulated_payload_type =
+            static_cast<int>(packet[header.headerLength]);
+        if (encapsulated_payload_type != kFakeVideoSendPayloadType)
+          EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
+      } else {
+        EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
+      }
 
       if (has_last_sequence_number_ &&
           !IsNewerSequenceNumber(header.sequenceNumber,
@@ -614,6 +622,7 @@ void EndToEndTest::TestReceivedFecPacketsNotNacked(
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock_(&crit_);
       if (state_ == kVerifyFecPacketNotInNackList) {
         test::RtcpPacketParser rtcp_parser;
         rtcp_parser.Parse(packet, length);
@@ -622,24 +631,35 @@ void EndToEndTest::TestReceivedFecPacketsNotNacked(
             IsNewerSequenceNumber(nacks.back(), fec_sequence_number_)) {
           EXPECT_TRUE(std::find(
               nacks.begin(), nacks.end(), fec_sequence_number_) == nacks.end());
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
       }
       return SEND_PACKET;
     }
 
+    test::PacketTransport* CreateSendTransport(Call* sender_call) override {
+      // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
+      // Configure some network delay.
+      const int kNetworkDelayMs = 50;
+      FakeNetworkPipe::Config config;
+      config.queue_delay_ms = kNetworkDelayMs;
+      return new test::PacketTransport(sender_call, this,
+                                       test::PacketTransport::kSender, config);
+    }
+
     // TODO(holmer): Investigate why we don't send FEC packets when the bitrate
     // is 10 kbps.
     Call::Config GetSenderCallConfig() override {
-      Call::Config config(SendTransport());
+      Call::Config config;
       const int kMinBitrateBps = 30000;
       config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
       return config;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       // Configure hybrid NACK/FEC.
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->rtp.fec.red_payload_type = kRedPayloadType;
@@ -650,7 +670,7 @@ void EndToEndTest::TestReceivedFecPacketsNotNacked(
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for FEC packets to be received.";
     }
 
@@ -661,35 +681,48 @@ void EndToEndTest::TestReceivedFecPacketsNotNacked(
       kVerifyFecPacketNotInNackList,
     } state_;
 
-    uint16_t fec_sequence_number_;
+    rtc::CriticalSection crit_;
+    uint16_t fec_sequence_number_ GUARDED_BY(&crit_);
     bool has_last_sequence_number_;
     uint16_t last_sequence_number_;
-  } test(config);
+  } test;
 
   RunBaseTest(&test);
 }
 
 // This test drops second RTP packet with a marker bit set, makes sure it's
 // retransmitted and renders. Retransmission SSRCs are also checked.
-void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
-  static const int kDroppedFrameNumber = 2;
+void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {
+  // Must be set high enough to allow the bitrate probing to finish.
+  static const int kMinProbePackets = 30;
+  static const int kDroppedFrameNumber = kMinProbePackets + 1;
   class RetransmissionObserver : public test::EndToEndTest,
                                  public I420FrameCallback {
    public:
-    explicit RetransmissionObserver(bool expect_rtx)
+    explicit RetransmissionObserver(bool use_rtx, bool use_red)
         : EndToEndTest(kDefaultTimeoutMs),
-          retransmission_ssrc_(expect_rtx ? kSendRtxSsrcs[0] : kSendSsrcs[0]),
-          retransmission_payload_type_(expect_rtx ? kSendRtxPayloadType
-                                                  : kFakeSendPayloadType),
+          payload_type_(GetPayloadType(false, use_red)),
+          retransmission_ssrc_(use_rtx ? kSendRtxSsrcs[0] : kVideoSendSsrcs[0]),
+          retransmission_payload_type_(GetPayloadType(use_rtx, use_red)),
           marker_bits_observed_(0),
+          num_packets_observed_(0),
           retransmitted_timestamp_(0),
           frame_retransmitted_(false) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
+      // We accept some padding or RTX packets in the beginning to enable
+      // bitrate probing.
+      if (num_packets_observed_++ < kMinProbePackets &&
+          header.payloadType != payload_type_) {
+        EXPECT_TRUE(retransmission_payload_type_ == header.payloadType ||
+                    length == header.headerLength + header.paddingLength);
+        return SEND_PACKET;
+      }
       if (header.timestamp == retransmitted_timestamp_) {
         EXPECT_EQ(retransmission_ssrc_, header.ssrc);
         EXPECT_EQ(retransmission_payload_type_, header.payloadType);
@@ -697,11 +730,11 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
         return SEND_PACKET;
       }
 
-      EXPECT_EQ(kSendSsrcs[0], header.ssrc);
-      EXPECT_EQ(kFakeSendPayloadType, header.payloadType);
+      EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc);
+      EXPECT_EQ(payload_type_, header.payloadType);
 
-      // Found the second frame's final packet, drop this and expect a
-      // retransmission.
+      // Found the final packet of the frame to inflict loss to, drop this and
+      // expect a retransmission.
       if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) {
         retransmitted_timestamp_ = header.timestamp;
         return DROP_PACKET;
@@ -710,51 +743,76 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
       return SEND_PACKET;
     }
 
-    void FrameCallback(I420VideoFrame* frame) override {
-      CriticalSectionScoped lock(crit_.get());
+    void FrameCallback(VideoFrame* frame) override {
+      rtc::CritScope lock(&crit_);
       if (frame->timestamp() == retransmitted_timestamp_) {
         EXPECT_TRUE(frame_retransmitted_);
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].pre_render_callback = this;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+
+      if (payload_type_ == kRedPayloadType) {
+        send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+        send_config->rtp.fec.red_payload_type = kRedPayloadType;
+        (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
+        (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+      }
+
       if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
-        (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].ssrc =
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
             kSendRtxSsrcs[0];
-        (*receive_configs)[0].rtp.rtx[kSendRtxPayloadType].payload_type =
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
             kSendRtxPayloadType;
       }
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for retransmission to render.";
     }
 
+    int GetPayloadType(bool use_rtx, bool use_red) {
+      return use_rtx ? kSendRtxPayloadType
+                     : (use_red ? kRedPayloadType : kFakeVideoSendPayloadType);
+    }
+
+    rtc::CriticalSection crit_;
+    const int payload_type_;
     const uint32_t retransmission_ssrc_;
     const int retransmission_payload_type_;
     int marker_bits_observed_;
-    uint32_t retransmitted_timestamp_;
+    int num_packets_observed_;
+    uint32_t retransmitted_timestamp_ GUARDED_BY(&crit_);
     bool frame_retransmitted_;
-  } test(retransmit_over_rtx);
+  } test(use_rtx, use_red);
 
   RunBaseTest(&test);
 }
 
 TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
-  DecodesRetransmittedFrame(false);
+  DecodesRetransmittedFrame(false, false);
 }
 
 TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
-  DecodesRetransmittedFrame(true);
+  DecodesRetransmittedFrame(true, false);
+}
+
+TEST_F(EndToEndTest, DecodesRetransmittedFrameByRed) {
+  DecodesRetransmittedFrame(false, true);
+}
+
+TEST_F(EndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
+  DecodesRetransmittedFrame(true, true);
 }
 
 TEST_F(EndToEndTest, UsesFrameCallbacks) {
@@ -763,33 +821,33 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
 
   class Renderer : public VideoRenderer {
    public:
-    Renderer() : event_(EventWrapper::Create()) {}
+    Renderer() : event_(false, false) {}
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int /*time_to_render_ms*/) override {
       EXPECT_EQ(0, *video_frame.buffer(kYPlane))
           << "Rendered frame should have zero luma which is applied by the "
              "pre-render callback.";
-      event_->Set();
+      event_.Set();
     }
 
     bool IsTextureSupported() const override { return false; }
 
-    EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
-    rtc::scoped_ptr<EventWrapper> event_;
+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
+    rtc::Event event_;
   } renderer;
 
   class TestFrameCallback : public I420FrameCallback {
    public:
     TestFrameCallback(int expected_luma_byte, int next_luma_byte)
-        : event_(EventWrapper::Create()),
+        : event_(false, false),
           expected_luma_byte_(expected_luma_byte),
           next_luma_byte_(next_luma_byte) {}
 
-    EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
 
    private:
-    virtual void FrameCallback(I420VideoFrame* frame) {
+    virtual void FrameCallback(VideoFrame* frame) {
       EXPECT_EQ(kWidth, frame->width())
           << "Width not as expected, callback done before resize?";
       EXPECT_EQ(kHeight, frame->height())
@@ -804,10 +862,10 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
              next_luma_byte_,
              frame->allocated_size(kYPlane));
 
-      event_->Set();
+      event_.Set();
     }
 
-    rtc::scoped_ptr<EventWrapper> event_;
+    rtc::Event event_;
     int expected_luma_byte_;
     int next_luma_byte_;
   };
@@ -815,42 +873,42 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
   TestFrameCallback pre_encode_callback(-1, 255);  // Changes luma to 255.
   TestFrameCallback pre_render_callback(255, 0);  // Changes luma from 255 to 0.
 
-  test::DirectTransport sender_transport, receiver_transport;
-
-  CreateCalls(Call::Config(&sender_transport),
-              Call::Config(&receiver_transport));
+  CreateCalls(Call::Config(), Call::Config());
 
+  test::DirectTransport sender_transport(sender_call_.get());
+  test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1);
+  CreateSendConfig(1, 0, &sender_transport);
   rtc::scoped_ptr<VideoEncoder> encoder(
       VideoEncoder::Create(VideoEncoder::kVp8));
-  send_config_.encoder_settings.encoder = encoder.get();
-  send_config_.encoder_settings.payload_name = "VP8";
-  ASSERT_EQ(1u, encoder_config_.streams.size()) << "Test setup error.";
-  encoder_config_.streams[0].width = kWidth;
-  encoder_config_.streams[0].height = kHeight;
-  send_config_.pre_encode_callback = &pre_encode_callback;
+  video_send_config_.encoder_settings.encoder = encoder.get();
+  video_send_config_.encoder_settings.payload_name = "VP8";
+  ASSERT_EQ(1u, video_encoder_config_.streams.size()) << "Test setup error.";
+  video_encoder_config_.streams[0].width = kWidth;
+  video_encoder_config_.streams[0].height = kHeight;
+  video_send_config_.pre_encode_callback = &pre_encode_callback;
 
-  CreateMatchingReceiveConfigs();
-  receive_configs_[0].pre_render_callback = &pre_render_callback;
-  receive_configs_[0].renderer = &renderer;
+  CreateMatchingReceiveConfigs(&receiver_transport);
+  video_receive_configs_[0].pre_render_callback = &pre_render_callback;
+  video_receive_configs_[0].renderer = &renderer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(kWidth / 2, kHeight / 2));
-  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
+  video_send_stream_->Input()->IncomingCapturedFrame(
+      *frame_generator->NextFrame());
 
-  EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait())
+  EXPECT_TRUE(pre_encode_callback.Wait())
       << "Timed out while waiting for pre-encode callback.";
-  EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
+  EXPECT_TRUE(pre_render_callback.Wait())
       << "Timed out while waiting for pre-render callback.";
-  EXPECT_EQ(kEventSignaled, renderer.Wait())
+  EXPECT_TRUE(renderer.Wait())
       << "Timed out while waiting for the frame to render.";
 
   Stop();
@@ -876,6 +934,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -893,16 +952,17 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-           packet_type != RTCPUtility::kRtcpNotValidCode;
+           packet_type != RTCPUtility::RTCPPacketTypes::kInvalid;
            packet_type = parser.Iterate()) {
         if (!nack_enabled_)
-          EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode);
+          EXPECT_NE(packet_type, RTCPUtility::RTCPPacketTypes::kRtpfbNack);
 
-        if (packet_type == RTCPUtility::kRtcpPsfbPliCode) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbPli) {
           received_pli_ = true;
           break;
         }
@@ -910,12 +970,12 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
       return SEND_PACKET;
     }
 
-    void RenderFrame(const I420VideoFrame& video_frame,
+    void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       if (received_pli_ &&
           video_frame.timestamp() > highest_dropped_timestamp_) {
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
       if (!received_pli_)
         frames_to_drop_ = kPacketsToDrop;
@@ -923,25 +983,27 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
 
     bool IsTextureSupported() const override { return false; }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
       (*receive_configs)[0].renderer = this;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be "
-                                           "received and a frame to be "
-                                           "rendered afterwards.";
+      EXPECT_TRUE(Wait()) << "Timed out waiting for PLI to be "
+                             "received and a frame to be "
+                             "rendered afterwards.";
     }
 
+    rtc::CriticalSection crit_;
     int rtp_history_ms_;
     bool nack_enabled_;
-    uint32_t highest_dropped_timestamp_;
-    int frames_to_drop_;
-    bool received_pli_;
+    uint32_t highest_dropped_timestamp_ GUARDED_BY(&crit_);
+    int frames_to_drop_ GUARDED_BY(&crit_);
+    bool received_pli_ GUARDED_BY(&crit_);
   } test(rtp_history_ms);
 
   RunBaseTest(&test);
@@ -951,8 +1013,7 @@ TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) {
   ReceivesPliAndRecovers(1000);
 }
 
-// TODO(pbos): Enable this when 2250 is resolved.
-TEST_F(EndToEndTest, DISABLED_ReceivesPliAndRecoversWithoutNack) {
+TEST_F(EndToEndTest, ReceivesPliAndRecoversWithoutNack) {
   ReceivesPliAndRecovers(0);
 }
 
@@ -960,50 +1021,51 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
   class PacketInputObserver : public PacketReceiver {
    public:
     explicit PacketInputObserver(PacketReceiver* receiver)
-        : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
+        : receiver_(receiver), delivered_packet_(false, false) {}
 
-    EventTypeWrapper Wait() {
-      return delivered_packet_->Wait(kDefaultTimeoutMs);
-    }
+    bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
 
    private:
-    DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                 size_t length) override {
+    DeliveryStatus DeliverPacket(MediaType media_type,
+                                 const uint8_t* packet,
+                                 size_t length,
+                                 const PacketTime& packet_time) override {
       if (RtpHeaderParser::IsRtcp(packet, length)) {
-        return receiver_->DeliverPacket(packet, length);
+        return receiver_->DeliverPacket(media_type, packet, length,
+                                        packet_time);
       } else {
         DeliveryStatus delivery_status =
-            receiver_->DeliverPacket(packet, length);
+            receiver_->DeliverPacket(media_type, packet, length, packet_time);
         EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
-        delivered_packet_->Set();
+        delivered_packet_.Set();
         return delivery_status;
       }
     }
 
     PacketReceiver* receiver_;
-    rtc::scoped_ptr<EventWrapper> delivered_packet_;
+    rtc::Event delivered_packet_;
   };
 
-  test::DirectTransport send_transport, receive_transport;
+  CreateCalls(Call::Config(), Call::Config());
 
-  CreateCalls(Call::Config(&send_transport), Call::Config(&receive_transport));
+  test::DirectTransport send_transport(sender_call_.get());
+  test::DirectTransport receive_transport(receiver_call_.get());
   PacketInputObserver input_observer(receiver_call_->Receiver());
-
   send_transport.SetReceiver(&input_observer);
   receive_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateSendConfig(1, 0, &send_transport);
+  CreateMatchingReceiveConfigs(&receive_transport);
 
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
   Start();
 
-  receiver_call_->DestroyVideoReceiveStream(receive_streams_[0]);
-  receive_streams_.clear();
+  receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
+  video_receive_streams_.clear();
 
   // Wait() waits for a received packet.
-  EXPECT_EQ(kEventSignaled, input_observer.Wait());
+  EXPECT_TRUE(input_observer.Wait());
 
   Stop();
 
@@ -1013,11 +1075,11 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
   receive_transport.StopSending();
 }
 
-void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
+void EndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
   static const int kNumCompoundRtcpPacketsToObserve = 10;
   class RtcpModeObserver : public test::EndToEndTest {
    public:
-    explicit RtcpModeObserver(newapi::RtcpMode rtcp_mode)
+    explicit RtcpModeObserver(RtcpMode rtcp_mode)
         : EndToEndTest(kDefaultTimeoutMs),
           rtcp_mode_(rtcp_mode),
           sent_rtp_(0),
@@ -1038,9 +1100,9 @@ void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
       bool has_report_block = false;
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        EXPECT_NE(RTCPUtility::kRtcpSrCode, packet_type);
-        if (packet_type == RTCPUtility::kRtcpRrCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        EXPECT_NE(RTCPUtility::RTCPPacketTypes::kSr, packet_type);
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kRr) {
           has_report_block = true;
           break;
         }
@@ -1048,42 +1110,46 @@ void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
       }
 
       switch (rtcp_mode_) {
-        case newapi::kRtcpCompound:
+        case RtcpMode::kCompound:
           if (!has_report_block) {
             ADD_FAILURE() << "Received RTCP packet without receiver report for "
-                             "kRtcpCompound.";
-            observation_complete_->Set();
+                             "RtcpMode::kCompound.";
+            observation_complete_.Set();
           }
 
           if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
-            observation_complete_->Set();
+            observation_complete_.Set();
 
           break;
-        case newapi::kRtcpReducedSize:
+        case RtcpMode::kReducedSize:
           if (!has_report_block)
-            observation_complete_->Set();
+            observation_complete_.Set();
+          break;
+        case RtcpMode::kOff:
+          RTC_NOTREACHED();
           break;
       }
 
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << (rtcp_mode_ == newapi::kRtcpCompound
+      EXPECT_TRUE(Wait())
+          << (rtcp_mode_ == RtcpMode::kCompound
                   ? "Timed out before observing enough compound packets."
                   : "Timed out before receiving a non-compound RTCP packet.");
     }
 
-    newapi::RtcpMode rtcp_mode_;
+    RtcpMode rtcp_mode_;
     int sent_rtp_;
     int sent_rtcp_;
   } test(rtcp_mode);
@@ -1092,138 +1158,473 @@ void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
 }
 
 TEST_F(EndToEndTest, UsesRtcpCompoundMode) {
-  RespectsRtcpMode(newapi::kRtcpCompound);
+  RespectsRtcpMode(RtcpMode::kCompound);
 }
 
 TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) {
-  RespectsRtcpMode(newapi::kRtcpReducedSize);
+  RespectsRtcpMode(RtcpMode::kReducedSize);
 }
 
 // Test sets up a Call multiple senders with different resolutions and SSRCs.
 // Another is set up to receive all three of these with different renderers.
-// Each renderer verifies that it receives the expected resolution, and as soon
-// as every renderer has received a frame, the test finishes.
-TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
+class MultiStreamTest {
+ public:
   static const size_t kNumStreams = 3;
-
-  class VideoOutputObserver : public VideoRenderer {
-   public:
-    VideoOutputObserver(test::FrameGeneratorCapturer** capturer,
-                        int width,
-                        int height)
-        : capturer_(capturer),
-          width_(width),
-          height_(height),
-          done_(EventWrapper::Create()) {}
-
-    void RenderFrame(const I420VideoFrame& video_frame,
-                     int time_to_render_ms) override {
-      EXPECT_EQ(width_, video_frame.width());
-      EXPECT_EQ(height_, video_frame.height());
-      (*capturer_)->Stop();
-      done_->Set();
-    }
-
-    bool IsTextureSupported() const override { return false; }
-
-    EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
-
-   private:
-    test::FrameGeneratorCapturer** capturer_;
-    int width_;
-    int height_;
-    rtc::scoped_ptr<EventWrapper> done_;
-  };
-
-  struct {
+  struct CodecSettings {
     uint32_t ssrc;
     int width;
     int height;
-  } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}};
+  } codec_settings[kNumStreams];
 
-  test::DirectTransport sender_transport, receiver_transport;
-  rtc::scoped_ptr<Call> sender_call(
-      Call::Create(Call::Config(&sender_transport)));
-  rtc::scoped_ptr<Call> receiver_call(
-      Call::Create(Call::Config(&receiver_transport)));
-  sender_transport.SetReceiver(receiver_call->Receiver());
-  receiver_transport.SetReceiver(sender_call->Receiver());
-
-  VideoSendStream* send_streams[kNumStreams];
-  VideoReceiveStream* receive_streams[kNumStreams];
-
-  VideoOutputObserver* observers[kNumStreams];
-  test::FrameGeneratorCapturer* frame_generators[kNumStreams];
-
-  rtc::scoped_ptr<VideoEncoder> encoders[kNumStreams];
-  for (size_t i = 0; i < kNumStreams; ++i)
-    encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8));
-
-  ScopedVector<VideoDecoder> allocated_decoders;
-  for (size_t i = 0; i < kNumStreams; ++i) {
-    uint32_t ssrc = codec_settings[i].ssrc;
-    int width = codec_settings[i].width;
-    int height = codec_settings[i].height;
-    observers[i] = new VideoOutputObserver(&frame_generators[i], width, height);
-
-    VideoSendStream::Config send_config;
-    send_config.rtp.ssrcs.push_back(ssrc);
-    send_config.encoder_settings.encoder = encoders[i].get();
-    send_config.encoder_settings.payload_name = "VP8";
-    send_config.encoder_settings.payload_type = 124;
-    VideoEncoderConfig encoder_config;
-    encoder_config.streams = test::CreateVideoStreams(1);
-    VideoStream* stream = &encoder_config.streams[0];
-    stream->width = width;
-    stream->height = height;
-    stream->max_framerate = 5;
-    stream->min_bitrate_bps = stream->target_bitrate_bps =
-        stream->max_bitrate_bps = 100000;
-    send_streams[i] =
-        sender_call->CreateVideoSendStream(send_config, encoder_config);
-    send_streams[i]->Start();
-
-    VideoReceiveStream::Config receive_config;
-    receive_config.renderer = observers[i];
-    receive_config.rtp.remote_ssrc = ssrc;
-    receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
-    VideoReceiveStream::Decoder decoder =
-        test::CreateMatchingDecoder(send_config.encoder_settings);
-    allocated_decoders.push_back(decoder.decoder);
-    receive_config.decoders.push_back(decoder);
-    receive_streams[i] =
-        receiver_call->CreateVideoReceiveStream(receive_config);
-    receive_streams[i]->Start();
-
-    frame_generators[i] = test::FrameGeneratorCapturer::Create(
-        send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock());
-    frame_generators[i]->Start();
+  MultiStreamTest() {
+    // TODO(sprang): Cleanup when msvc supports explicit initializers for array.
+    codec_settings[0] = {1, 640, 480};
+    codec_settings[1] = {2, 320, 240};
+    codec_settings[2] = {3, 240, 160};
   }
 
-  for (size_t i = 0; i < kNumStreams; ++i) {
-    EXPECT_EQ(kEventSignaled, observers[i]->Wait())
-        << "Timed out while waiting for observer " << i << " to render.";
+  virtual ~MultiStreamTest() {}
+
+  void RunTest() {
+    rtc::scoped_ptr<Call> sender_call(Call::Create(Call::Config()));
+    rtc::scoped_ptr<Call> receiver_call(Call::Create(Call::Config()));
+    rtc::scoped_ptr<test::DirectTransport> sender_transport(
+        CreateSendTransport(sender_call.get()));
+    rtc::scoped_ptr<test::DirectTransport> receiver_transport(
+        CreateReceiveTransport(receiver_call.get()));
+    sender_transport->SetReceiver(receiver_call->Receiver());
+    receiver_transport->SetReceiver(sender_call->Receiver());
+
+    rtc::scoped_ptr<VideoEncoder> encoders[kNumStreams];
+    for (size_t i = 0; i < kNumStreams; ++i)
+      encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8));
+
+    VideoSendStream* send_streams[kNumStreams];
+    VideoReceiveStream* receive_streams[kNumStreams];
+
+    test::FrameGeneratorCapturer* frame_generators[kNumStreams];
+    ScopedVector<VideoDecoder> allocated_decoders;
+    for (size_t i = 0; i < kNumStreams; ++i) {
+      uint32_t ssrc = codec_settings[i].ssrc;
+      int width = codec_settings[i].width;
+      int height = codec_settings[i].height;
+
+      VideoSendStream::Config send_config(sender_transport.get());
+      send_config.rtp.ssrcs.push_back(ssrc);
+      send_config.encoder_settings.encoder = encoders[i].get();
+      send_config.encoder_settings.payload_name = "VP8";
+      send_config.encoder_settings.payload_type = 124;
+      VideoEncoderConfig encoder_config;
+      encoder_config.streams = test::CreateVideoStreams(1);
+      VideoStream* stream = &encoder_config.streams[0];
+      stream->width = width;
+      stream->height = height;
+      stream->max_framerate = 5;
+      stream->min_bitrate_bps = stream->target_bitrate_bps =
+          stream->max_bitrate_bps = 100000;
+
+      UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
+
+      send_streams[i] =
+          sender_call->CreateVideoSendStream(send_config, encoder_config);
+      send_streams[i]->Start();
+
+      VideoReceiveStream::Config receive_config(receiver_transport.get());
+      receive_config.rtp.remote_ssrc = ssrc;
+      receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
+      VideoReceiveStream::Decoder decoder =
+          test::CreateMatchingDecoder(send_config.encoder_settings);
+      allocated_decoders.push_back(decoder.decoder);
+      receive_config.decoders.push_back(decoder);
+
+      UpdateReceiveConfig(i, &receive_config);
+
+      receive_streams[i] =
+          receiver_call->CreateVideoReceiveStream(receive_config);
+      receive_streams[i]->Start();
+
+      frame_generators[i] = test::FrameGeneratorCapturer::Create(
+          send_streams[i]->Input(), width, height, 30,
+          Clock::GetRealTimeClock());
+      frame_generators[i]->Start();
+    }
+
+    Wait();
+
+    for (size_t i = 0; i < kNumStreams; ++i) {
+      frame_generators[i]->Stop();
+      sender_call->DestroyVideoSendStream(send_streams[i]);
+      receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
+      delete frame_generators[i];
+    }
+
+    sender_transport->StopSending();
+    receiver_transport->StopSending();
   }
 
-  for (size_t i = 0; i < kNumStreams; ++i) {
-    frame_generators[i]->Stop();
-    sender_call->DestroyVideoSendStream(send_streams[i]);
-    receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
-    delete frame_generators[i];
-    delete observers[i];
+ protected:
+  virtual void Wait() = 0;
+  // Note: frame_generator is a point-to-pointer, since the actual instance
+  // hasn't been created at the time of this call. Only when packets/frames
+  // start flowing should this be dereferenced.
+  virtual void UpdateSendConfig(
+      size_t stream_index,
+      VideoSendStream::Config* send_config,
+      VideoEncoderConfig* encoder_config,
+      test::FrameGeneratorCapturer** frame_generator) {}
+  virtual void UpdateReceiveConfig(size_t stream_index,
+                                   VideoReceiveStream::Config* receive_config) {
   }
+  virtual test::DirectTransport* CreateSendTransport(Call* sender_call) {
+    return new test::DirectTransport(sender_call);
+  }
+  virtual test::DirectTransport* CreateReceiveTransport(Call* receiver_call) {
+    return new test::DirectTransport(receiver_call);
+  }
+};
 
-  sender_transport.StopSending();
-  receiver_transport.StopSending();
+// Each renderer verifies that it receives the expected resolution, and as soon
+// as every renderer has received a frame, the test finishes.
+TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
+  class VideoOutputObserver : public VideoRenderer {
+   public:
+    VideoOutputObserver(const MultiStreamTest::CodecSettings& settings,
+                        uint32_t ssrc,
+                        test::FrameGeneratorCapturer** frame_generator)
+        : settings_(settings),
+          ssrc_(ssrc),
+          frame_generator_(frame_generator),
+          done_(false, false) {}
+
+    void RenderFrame(const VideoFrame& video_frame,
+                     int time_to_render_ms) override {
+      EXPECT_EQ(settings_.width, video_frame.width());
+      EXPECT_EQ(settings_.height, video_frame.height());
+      (*frame_generator_)->Stop();
+      done_.Set();
+    }
+
+    uint32_t Ssrc() { return ssrc_; }
+
+    bool IsTextureSupported() const override { return false; }
+
+    bool Wait() { return done_.Wait(kDefaultTimeoutMs); }
+
+   private:
+    const MultiStreamTest::CodecSettings& settings_;
+    const uint32_t ssrc_;
+    test::FrameGeneratorCapturer** const frame_generator_;
+    rtc::Event done_;
+  };
+
+  class Tester : public MultiStreamTest {
+   public:
+    Tester() {}
+    virtual ~Tester() {}
+
+   protected:
+    void Wait() override {
+      for (const auto& observer : observers_) {
+        EXPECT_TRUE(observer->Wait()) << "Time out waiting for from on ssrc "
+                                      << observer->Ssrc();
+      }
+    }
+
+    void UpdateSendConfig(
+        size_t stream_index,
+        VideoSendStream::Config* send_config,
+        VideoEncoderConfig* encoder_config,
+        test::FrameGeneratorCapturer** frame_generator) override {
+      observers_[stream_index].reset(new VideoOutputObserver(
+          codec_settings[stream_index], send_config->rtp.ssrcs.front(),
+          frame_generator));
+    }
+
+    void UpdateReceiveConfig(
+        size_t stream_index,
+        VideoReceiveStream::Config* receive_config) override {
+      receive_config->renderer = observers_[stream_index].get();
+    }
+
+   private:
+    rtc::scoped_ptr<VideoOutputObserver> observers_[kNumStreams];
+  } tester;
+
+  tester.RunTest();
+}
+
+TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {
+  static const int kExtensionId = 5;
+
+  class RtpExtensionHeaderObserver : public test::DirectTransport {
+   public:
+    RtpExtensionHeaderObserver(Call* sender_call,
+                               const uint32_t& first_media_ssrc,
+                               const std::map<uint32_t, uint32_t>& ssrc_map)
+        : DirectTransport(sender_call),
+          done_(false, false),
+          parser_(RtpHeaderParser::Create()),
+          first_media_ssrc_(first_media_ssrc),
+          rtx_to_media_ssrcs_(ssrc_map),
+          padding_observed_(false),
+          rtx_padding_observed_(false),
+          retransmit_observed_(false),
+          started_(false) {
+      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                          kExtensionId);
+    }
+    virtual ~RtpExtensionHeaderObserver() {}
+
+    bool SendRtp(const uint8_t* data,
+                 size_t length,
+                 const PacketOptions& options) override {
+      {
+        rtc::CritScope cs(&lock_);
+
+        if (IsDone())
+          return false;
+
+        if (started_) {
+          RTPHeader header;
+          EXPECT_TRUE(parser_->Parse(data, length, &header));
+          bool drop_packet = false;
+
+          EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+          EXPECT_EQ(options.packet_id,
+                    header.extension.transportSequenceNumber);
+          if (!streams_observed_.empty()) {
+            // Unwrap packet id and verify uniqueness.
+            int64_t packet_id = unwrapper_.Unwrap(options.packet_id);
+            EXPECT_TRUE(received_packed_ids_.insert(packet_id).second);
+          }
+
+          // Drop (up to) every 17th packet, so we get retransmits.
+          // Only drop media, and not on the first stream (otherwise it will be
+          // hard to distinguish from padding, which is always sent on the first
+          // stream).
+          if (header.payloadType != kSendRtxPayloadType &&
+              header.ssrc != first_media_ssrc_ &&
+              header.extension.transportSequenceNumber % 17 == 0) {
+            dropped_seq_[header.ssrc].insert(header.sequenceNumber);
+            drop_packet = true;
+          }
+
+          size_t payload_length =
+              length - (header.headerLength + header.paddingLength);
+          if (payload_length == 0) {
+            padding_observed_ = true;
+          } else if (header.payloadType == kSendRtxPayloadType) {
+            uint16_t original_sequence_number =
+                ByteReader<uint16_t>::ReadBigEndian(&data[header.headerLength]);
+            uint32_t original_ssrc =
+                rtx_to_media_ssrcs_.find(header.ssrc)->second;
+            std::set<uint16_t>* seq_no_map = &dropped_seq_[original_ssrc];
+            auto it = seq_no_map->find(original_sequence_number);
+            if (it != seq_no_map->end()) {
+              retransmit_observed_ = true;
+              seq_no_map->erase(it);
+            } else {
+              rtx_padding_observed_ = true;
+            }
+          } else {
+            streams_observed_.insert(header.ssrc);
+          }
+
+          if (IsDone())
+            done_.Set();
+
+          if (drop_packet)
+            return true;
+        }
+      }
+
+      return test::DirectTransport::SendRtp(data, length, options);
+    }
+
+    bool IsDone() {
+      bool observed_types_ok =
+          streams_observed_.size() == MultiStreamTest::kNumStreams &&
+          padding_observed_ && retransmit_observed_ && rtx_padding_observed_;
+      if (!observed_types_ok)
+        return false;
+      // We should not have any gaps in the sequence number range.
+      size_t seqno_range =
+          *received_packed_ids_.rbegin() - *received_packed_ids_.begin() + 1;
+      return seqno_range == received_packed_ids_.size();
+    }
+
+    bool Wait() {
+      {
+        // Can't be sure until this point that rtx_to_media_ssrcs_ etc have
+        // been initialized and are OK to read.
+        rtc::CritScope cs(&lock_);
+        started_ = true;
+      }
+      return done_.Wait(kDefaultTimeoutMs);
+    }
+
+    rtc::CriticalSection lock_;
+    rtc::Event done_;
+    rtc::scoped_ptr<RtpHeaderParser> parser_;
+    SequenceNumberUnwrapper unwrapper_;
+    std::set<int64_t> received_packed_ids_;
+    std::set<uint32_t> streams_observed_;
+    std::map<uint32_t, std::set<uint16_t>> dropped_seq_;
+    const uint32_t& first_media_ssrc_;
+    const std::map<uint32_t, uint32_t>& rtx_to_media_ssrcs_;
+    bool padding_observed_;
+    bool rtx_padding_observed_;
+    bool retransmit_observed_;
+    bool started_;
+  };
+
+  class TransportSequenceNumberTester : public MultiStreamTest {
+   public:
+    TransportSequenceNumberTester()
+        : first_media_ssrc_(0), observer_(nullptr) {}
+    virtual ~TransportSequenceNumberTester() {}
+
+   protected:
+    void Wait() override {
+      RTC_DCHECK(observer_ != nullptr);
+      EXPECT_TRUE(observer_->Wait());
+    }
+
+    void UpdateSendConfig(
+        size_t stream_index,
+        VideoSendStream::Config* send_config,
+        VideoEncoderConfig* encoder_config,
+        test::FrameGeneratorCapturer** frame_generator) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+
+      // Force some padding to be sent.
+      const int kPaddingBitrateBps = 50000;
+      int total_target_bitrate = 0;
+      for (const VideoStream& stream : encoder_config->streams)
+        total_target_bitrate += stream.target_bitrate_bps;
+      encoder_config->min_transmit_bitrate_bps =
+          total_target_bitrate + kPaddingBitrateBps;
+
+      // Configure RTX for redundant payload padding.
+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]);
+      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
+      rtx_to_media_ssrcs_[kSendRtxSsrcs[stream_index]] =
+          send_config->rtp.ssrcs[0];
+
+      if (stream_index == 0)
+        first_media_ssrc_ = send_config->rtp.ssrcs[0];
+    }
+
+    void UpdateReceiveConfig(
+        size_t stream_index,
+        VideoReceiveStream::Config* receive_config) override {
+      receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      receive_config->rtp.extensions.clear();
+      receive_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    test::DirectTransport* CreateSendTransport(Call* sender_call) override {
+      observer_ = new RtpExtensionHeaderObserver(sender_call, first_media_ssrc_,
+                                                 rtx_to_media_ssrcs_);
+      return observer_;
+    }
+
+   private:
+    uint32_t first_media_ssrc_;
+    std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
+    RtpExtensionHeaderObserver* observer_;
+  } tester;
+
+  tester.RunTest();
+}
+
+void TransportFeedbackTest(bool feedback_enabled) {
+  static const int kExtensionId = 5;
+  class TransportFeedbackObserver : public test::DirectTransport {
+   public:
+    TransportFeedbackObserver(Call* receiver_call, rtc::Event* done_event)
+        : DirectTransport(receiver_call), done_(done_event) {}
+    virtual ~TransportFeedbackObserver() {}
+
+    bool SendRtcp(const uint8_t* data, size_t length) override {
+      RTCPUtility::RTCPParserV2 parser(data, length, true);
+      EXPECT_TRUE(parser.IsValid());
+
+      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback) {
+          done_->Set();
+          break;
+        }
+        packet_type = parser.Iterate();
+      }
+
+      return test::DirectTransport::SendRtcp(data, length);
+    }
+
+    rtc::Event* done_;
+  };
+
+  class TransportFeedbackTester : public MultiStreamTest {
+   public:
+    explicit TransportFeedbackTester(bool feedback_enabled)
+        : feedback_enabled_(feedback_enabled), done_(false, false) {}
+    virtual ~TransportFeedbackTester() {}
+
+   protected:
+    void Wait() override {
+      const int64_t kDisabledFeedbackTimeoutMs = 5000;
+      EXPECT_EQ(feedback_enabled_, done_.Wait(feedback_enabled_
+                                  ? test::CallTest::kDefaultTimeoutMs
+                                  : kDisabledFeedbackTimeoutMs));
+    }
+
+    void UpdateSendConfig(
+        size_t stream_index,
+        VideoSendStream::Config* send_config,
+        VideoEncoderConfig* encoder_config,
+        test::FrameGeneratorCapturer** frame_generator) override {
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    void UpdateReceiveConfig(
+        size_t stream_index,
+        VideoReceiveStream::Config* receive_config) override {
+      receive_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      receive_config->rtp.transport_cc = feedback_enabled_;
+    }
+
+    test::DirectTransport* CreateReceiveTransport(
+        Call* receiver_call) override {
+      return new TransportFeedbackObserver(receiver_call, &done_);
+    }
+
+   private:
+    const bool feedback_enabled_;
+    rtc::Event done_;
+  } tester(feedback_enabled);
+  tester.RunTest();
+}
+
+TEST_F(EndToEndTest, ReceivesTransportFeedback) {
+  TransportFeedbackTest(true);
+}
+
+TEST_F(EndToEndTest, TransportFeedbackNotConfigured) {
+  TransportFeedbackTest(false);
 }
 
 TEST_F(EndToEndTest, ObserversEncodedFrames) {
   class EncodedFrameTestObserver : public EncodedFrameObserver {
    public:
     EncodedFrameTestObserver()
-        : length_(0),
-          frame_type_(kFrameEmpty),
-          called_(EventWrapper::Create()) {}
+        : length_(0), frame_type_(kEmptyFrame), called_(false, false) {}
     virtual ~EncodedFrameTestObserver() {}
 
     virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
@@ -1231,10 +1632,10 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) {
       length_ = encoded_frame.length_;
       buffer_.reset(new uint8_t[length_]);
       memcpy(buffer_.get(), encoded_frame.data_, length_);
-      called_->Set();
+      called_.Set();
     }
 
-    EventTypeWrapper Wait() { return called_->Wait(kDefaultTimeoutMs); }
+    bool Wait() { return called_.Wait(kDefaultTimeoutMs); }
 
     void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
       ASSERT_EQ(length_, observer.length_)
@@ -1249,37 +1650,38 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) {
     rtc::scoped_ptr<uint8_t[]> buffer_;
     size_t length_;
     FrameType frame_type_;
-    rtc::scoped_ptr<EventWrapper> called_;
+    rtc::Event called_;
   };
 
   EncodedFrameTestObserver post_encode_observer;
   EncodedFrameTestObserver pre_decode_observer;
 
-  test::DirectTransport sender_transport, receiver_transport;
-
-  CreateCalls(Call::Config(&sender_transport),
-              Call::Config(&receiver_transport));
+  CreateCalls(Call::Config(), Call::Config());
 
+  test::DirectTransport sender_transport(sender_call_.get());
+  test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
-  send_config_.post_encode_callback = &post_encode_observer;
-  receive_configs_[0].pre_decode_callback = &pre_decode_observer;
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(&receiver_transport);
+  video_send_config_.post_encode_callback = &post_encode_observer;
+  video_receive_configs_[0].pre_decode_callback = &pre_decode_observer;
 
-  CreateStreams();
+  CreateVideoStreams();
   Start();
 
   rtc::scoped_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(
-          encoder_config_.streams[0].width, encoder_config_.streams[0].height));
-  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
+          video_encoder_config_.streams[0].width,
+          video_encoder_config_.streams[0].height));
+  video_send_stream_->Input()->IncomingCapturedFrame(
+      *frame_generator->NextFrame());
 
-  EXPECT_EQ(kEventSignaled, post_encode_observer.Wait())
+  EXPECT_TRUE(post_encode_observer.Wait())
       << "Timed out while waiting for send-side encoded-frame callback.";
 
-  EXPECT_EQ(kEventSignaled, pre_decode_observer.Wait())
+  EXPECT_TRUE(pre_decode_observer.Wait())
       << "Timed out while waiting for pre-decode encoded-frame callback.";
 
   post_encode_observer.ExpectEqualFrames(pre_decode_observer);
@@ -1304,28 +1706,28 @@ TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
       bool received_psfb = false;
       bool received_remb = false;
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        if (packet_type == RTCPUtility::kRtcpPsfbRembCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRemb) {
           const RTCPUtility::RTCPPacket& packet = parser.Packet();
-          EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalSsrc);
+          EXPECT_EQ(packet.PSFBAPP.SenderSSRC, kReceiverLocalVideoSsrc);
           received_psfb = true;
-        } else if (packet_type == RTCPUtility::kRtcpPsfbRembItemCode) {
+        } else if (packet_type == RTCPUtility::RTCPPacketTypes::kPsfbRembItem) {
           const RTCPUtility::RTCPPacket& packet = parser.Packet();
           EXPECT_GT(packet.REMBItem.BitRate, 0u);
           EXPECT_EQ(packet.REMBItem.NumberOfSSRCs, 1u);
-          EXPECT_EQ(packet.REMBItem.SSRCs[0], kSendSsrcs[0]);
+          EXPECT_EQ(packet.REMBItem.SSRCs[0], kVideoSendSsrcs[0]);
           received_remb = true;
         }
         packet_type = parser.Iterate();
       }
       if (received_psfb && received_remb)
-        observation_complete_->Set();
+        observation_complete_.Set();
       return SEND_PACKET;
     }
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a "
-                                           "receiver RTCP REMB packet to be "
-                                           "sent.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
+                             "receiver RTCP REMB packet to be "
+                             "sent.";
     }
   } test;
 
@@ -1333,7 +1735,7 @@ TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
 }
 
 TEST_F(EndToEndTest, VerifyBandwidthStats) {
-  class RtcpObserver : public test::EndToEndTest, public PacketReceiver {
+  class RtcpObserver : public test::EndToEndTest {
    public:
     RtcpObserver()
         : EndToEndTest(kDefaultTimeoutMs),
@@ -1341,17 +1743,16 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {
           receiver_call_(nullptr),
           has_seen_pacer_delay_(false) {}
 
-    DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                 size_t length) override {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       Call::Stats sender_stats = sender_call_->GetStats();
       Call::Stats receiver_stats = receiver_call_->GetStats();
       if (!has_seen_pacer_delay_)
         has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
       if (sender_stats.send_bandwidth_bps > 0 &&
           receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_) {
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
-      return receiver_call_->Receiver()->DeliverPacket(packet, length);
+      return SEND_PACKET;
     }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
@@ -1360,13 +1761,8 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
-                                           "non-zero bandwidth stats.";
-    }
-
-    void SetReceivers(PacketReceiver* send_transport_receiver,
-                      PacketReceiver* receive_transport_receiver) override {
-      test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for "
+                             "non-zero bandwidth stats.";
     }
 
    private:
@@ -1392,6 +1788,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       if (++sent_rtp_packets_ == kPacketNumberToDrop) {
         rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
         RTPHeader header;
@@ -1404,6 +1801,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       test::RtcpPacketParser rtcp_parser;
       rtcp_parser.Parse(packet, length);
       std::vector<uint16_t> nacks = rtcp_parser.nack_item()->last_nack_list();
@@ -1414,7 +1812,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {
       return SEND_PACKET;
     }
 
-    void VerifyStats() {
+    void VerifyStats() EXCLUSIVE_LOCKS_REQUIRED(&crit_) {
       if (!dropped_rtp_packet_requested_)
         return;
       int send_stream_nack_packets = 0;
@@ -1434,7 +1832,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {
       if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
         // NACK packet sent on receive stream and received on sent stream.
         if (MinMetricRunTimePassed())
-          observation_complete_->Set();
+          observation_complete_.Set();
       }
     }
 
@@ -1448,14 +1846,15 @@ TEST_F(EndToEndTest, VerifyNackStats) {
       return elapsed_sec > metrics::kMinRunTimeInSeconds;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
@@ -1463,23 +1862,24 @@ TEST_F(EndToEndTest, VerifyNackStats) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out waiting for packet to be NACKed.";
+      EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
     }
 
+    rtc::CriticalSection crit_;
     uint64_t sent_rtp_packets_;
-    uint16_t dropped_rtp_packet_;
-    bool dropped_rtp_packet_requested_;
+    uint16_t dropped_rtp_packet_ GUARDED_BY(&crit_);
+    bool dropped_rtp_packet_requested_ GUARDED_BY(&crit_);
     std::vector<VideoReceiveStream*> receive_streams_;
     VideoSendStream* send_stream_;
     int64_t start_runtime_ms_;
   } test;
 
+  test::ClearHistograms();
   RunBaseTest(&test);
 
-  EXPECT_NE(-1, test::LastHistogramSample(
+  EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.UniqueNackRequestsSentInPercent"));
-  EXPECT_NE(-1, test::LastHistogramSample(
+  EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
   EXPECT_GT(test::LastHistogramSample(
       "WebRTC.Video.NackPacketsSentPerMinute"), 0);
@@ -1487,6 +1887,224 @@ TEST_F(EndToEndTest, VerifyNackStats) {
       "WebRTC.Video.NackPacketsReceivedPerMinute"), 0);
 }
 
+void EndToEndTest::VerifyHistogramStats(bool use_rtx,
+                                        bool use_red,
+                                        bool screenshare) {
+  class StatsObserver : public test::EndToEndTest {
+   public:
+    StatsObserver(bool use_rtx, bool use_red, bool screenshare)
+        : EndToEndTest(kLongTimeoutMs),
+          use_rtx_(use_rtx),
+          use_red_(use_red),
+          screenshare_(screenshare),
+          sender_call_(nullptr),
+          receiver_call_(nullptr),
+          start_runtime_ms_(-1) {}
+
+   private:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      if (MinMetricRunTimePassed())
+        observation_complete_.Set();
+
+      // GetStats calls GetSendChannelRtcpStatistics
+      // (via VideoSendStream::GetRtt) which updates ReportBlockStats used by
+      // WebRTC.Video.SentPacketsLostInPercent.
+      // TODO(asapersson): Remove dependency on calling GetStats.
+      sender_call_->GetStats();
+
+      return SEND_PACKET;
+    }
+
+    bool MinMetricRunTimePassed() {
+      int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
+      if (start_runtime_ms_ == -1) {
+        start_runtime_ms_ = now;
+        return false;
+      }
+      int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
+      return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
+    }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      // NACK
+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      // FEC
+      if (use_red_) {
+        send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+        send_config->rtp.fec.red_payload_type = kRedPayloadType;
+        (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
+        (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+      }
+      // RTX
+      if (use_rtx_) {
+        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
+        send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
+            kSendRtxSsrcs[0];
+        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
+            kSendRtxPayloadType;
+      }
+      encoder_config->content_type =
+          screenshare_ ? VideoEncoderConfig::ContentType::kScreen
+                       : VideoEncoderConfig::ContentType::kRealtimeVideo;
+    }
+
+    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
+      sender_call_ = sender_call;
+      receiver_call_ = receiver_call;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
+    }
+
+    const bool use_rtx_;
+    const bool use_red_;
+    const bool screenshare_;
+    Call* sender_call_;
+    Call* receiver_call_;
+    int64_t start_runtime_ms_;
+  } test(use_rtx, use_red, screenshare);
+
+  test::ClearHistograms();
+  RunBaseTest(&test);
+
+  // Delete the call for Call stats to be reported.
+  sender_call_.reset();
+  receiver_call_.reset();
+
+  std::string video_prefix =
+      screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
+
+  // Verify that stats have been updated once.
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
+  EXPECT_EQ(1,
+            test::NumHistogramSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.BitrateReceivedInKbps"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.PacerBitrateInKbps"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.NackPacketsSentPerMinute"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.NackPacketsReceivedPerMinute"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.FirPacketsSentPerMinute"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.FirPacketsReceivedPerMinute"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.PliPacketsSentPerMinute"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.PliPacketsReceivedPerMinute"));
+
+  EXPECT_EQ(
+      1, test::NumHistogramSamples(video_prefix + "KeyFramesSentInPermille"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.KeyFramesReceivedInPermille"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.SentPacketsLostInPercent"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.ReceivedPacketsLostInPercent"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "InputWidthInPixels"));
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "InputHeightInPixels"));
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentWidthInPixels"));
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentHeightInPixels"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.ReceivedWidthInPixels"));
+  EXPECT_EQ(1,
+            test::NumHistogramSamples("WebRTC.Video.ReceivedHeightInPixels"));
+
+  EXPECT_EQ(static_cast<int>(video_encoder_config_.streams[0].width),
+            test::LastHistogramSample(video_prefix + "InputWidthInPixels"));
+  EXPECT_EQ(static_cast<int>(video_encoder_config_.streams[0].height),
+            test::LastHistogramSample(video_prefix + "InputHeightInPixels"));
+  EXPECT_EQ(static_cast<int>(video_encoder_config_.streams[0].width),
+            test::LastHistogramSample(video_prefix + "SentWidthInPixels"));
+  EXPECT_EQ(static_cast<int>(video_encoder_config_.streams[0].height),
+            test::LastHistogramSample(video_prefix + "SentHeightInPixels"));
+  EXPECT_EQ(static_cast<int>(video_encoder_config_.streams[0].width),
+            test::LastHistogramSample("WebRTC.Video.ReceivedWidthInPixels"));
+  EXPECT_EQ(static_cast<int>(video_encoder_config_.streams[0].height),
+            test::LastHistogramSample("WebRTC.Video.ReceivedHeightInPixels"));
+
+  EXPECT_EQ(1,
+            test::NumHistogramSamples(video_prefix + "InputFramesPerSecond"));
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentFramesPerSecond"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.DecodedFramesPerSecond"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.RenderFramesPerSecond"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.OnewayDelayInMs"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "EncodeTimeInMs"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.DecodeTimeInMs"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.BitrateSentInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.BitrateReceivedInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.MediaBitrateSentInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.MediaBitrateReceivedInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.PaddingBitrateSentInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.PaddingBitrateReceivedInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.RetransmittedBitrateSentInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples(
+      "WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
+
+  EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SendSideDelayInMs"));
+  EXPECT_EQ(1,
+            test::NumHistogramSamples(video_prefix + "SendSideDelayMaxInMs"));
+
+  int num_rtx_samples = use_rtx ? 1 : 0;
+  EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
+      "WebRTC.Video.RtxBitrateSentInKbps"));
+  EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
+      "WebRTC.Video.RtxBitrateReceivedInKbps"));
+
+  int num_red_samples = use_red ? 1 : 0;
+  EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
+      "WebRTC.Video.FecBitrateSentInKbps"));
+  EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
+      "WebRTC.Video.FecBitrateReceivedInKbps"));
+  EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
+      "WebRTC.Video.ReceivedFecPacketsInPercent"));
+}
+
+TEST_F(EndToEndTest, VerifyHistogramStatsWithRtx) {
+  const bool kEnabledRtx = true;
+  const bool kEnabledRed = false;
+  const bool kScreenshare = false;
+  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
+}
+
+TEST_F(EndToEndTest, VerifyHistogramStatsWithRed) {
+  const bool kEnabledRtx = false;
+  const bool kEnabledRed = true;
+  const bool kScreenshare = false;
+  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
+}
+
+TEST_F(EndToEndTest, VerifyHistogramStatsWithScreenshare) {
+  const bool kEnabledRtx = false;
+  const bool kEnabledRed = false;
+  const bool kScreenshare = true;
+  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
+}
+
 void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
   static const int kNumRtcpReportPacketsToObserve = 5;
   class RtcpXrObserver : public test::EndToEndTest {
@@ -1502,36 +2120,41 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
    private:
     // Receive stream should send RR packets (and RRTR packets if enabled).
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        if (packet_type == RTCPUtility::kRtcpRrCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kRr) {
           ++sent_rtcp_rr_;
         } else if (packet_type ==
-                   RTCPUtility::kRtcpXrReceiverReferenceTimeCode) {
+                   RTCPUtility::RTCPPacketTypes::kXrReceiverReferenceTime) {
           ++sent_rtcp_rrtr_;
         }
-        EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode);
-        EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode);
+        EXPECT_NE(packet_type, RTCPUtility::RTCPPacketTypes::kSr);
+        EXPECT_NE(packet_type,
+                  RTCPUtility::RTCPPacketTypes::kXrDlrrReportBlockItem);
         packet_type = parser.Iterate();
       }
       return SEND_PACKET;
     }
     // Send stream should send SR packets (and DLRR packets if enabled).
     virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
+      rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        if (packet_type == RTCPUtility::kRtcpSrCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
           ++sent_rtcp_sr_;
-        } else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) {
+        } else if (packet_type ==
+                   RTCPUtility::RTCPPacketTypes::kXrDlrrReportBlockItem) {
           ++sent_rtcp_dlrr_;
         }
-        EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode);
+        EXPECT_NE(packet_type,
+                  RTCPUtility::RTCPPacketTypes::kXrReceiverReferenceTime);
         packet_type = parser.Iterate();
       }
       if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
@@ -1543,28 +2166,30 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
           EXPECT_EQ(0, sent_rtcp_rrtr_);
           EXPECT_EQ(0, sent_rtcp_dlrr_);
         }
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
-      (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize;
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
           enable_rrtr_;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for RTCP SR/RR packets to be sent.";
     }
 
+    rtc::CriticalSection crit_;
     bool enable_rrtr_;
     int sent_rtcp_sr_;
-    int sent_rtcp_rr_;
-    int sent_rtcp_rrtr_;
+    int sent_rtcp_rr_ GUARDED_BY(&crit_);
+    int sent_rtcp_rrtr_ GUARDED_BY(&crit_);
     int sent_rtcp_dlrr_;
   } test(enable_rrtr);
 
@@ -1582,7 +2207,8 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
           num_ssrcs_(num_ssrcs),
           send_single_ssrc_first_(send_single_ssrc_first),
           ssrcs_to_observe_(num_ssrcs),
-          expect_single_ssrc_(send_single_ssrc_first) {
+          expect_single_ssrc_(send_single_ssrc_first),
+          send_stream_(nullptr) {
       for (size_t i = 0; i < num_ssrcs; ++i)
         valid_ssrcs_[ssrcs[i]] = true;
     }
@@ -1596,28 +2222,29 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
           << "Received unknown SSRC: " << header.ssrc;
 
       if (!valid_ssrcs_[header.ssrc])
-        observation_complete_->Set();
+        observation_complete_.Set();
 
       if (!is_observed_[header.ssrc]) {
         is_observed_[header.ssrc] = true;
         --ssrcs_to_observe_;
         if (expect_single_ssrc_) {
           expect_single_ssrc_ = false;
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
       }
 
       if (ssrcs_to_observe_ == 0)
-        observation_complete_->Set();
+        observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
-    size_t GetNumStreams() const override { return num_ssrcs_; }
+    size_t GetNumVideoStreams() const override { return num_ssrcs_; }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       if (num_ssrcs_ > 1) {
         // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
         for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
@@ -1627,27 +2254,27 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
         }
       }
 
-      encoder_config_all_streams_ = *encoder_config;
+      video_encoder_config_all_streams_ = *encoder_config;
       if (send_single_ssrc_first_)
         encoder_config->streams.resize(1);
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for "
-          << (send_single_ssrc_first_ ? "first SSRC." : "SSRCs.");
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for "
+                          << (send_single_ssrc_first_ ? "first SSRC."
+                                                      : "SSRCs.");
 
       if (send_single_ssrc_first_) {
         // Set full simulcast and continue with the rest of the SSRCs.
-        send_stream_->ReconfigureVideoEncoder(encoder_config_all_streams_);
-        EXPECT_EQ(kEventSignaled, Wait())
-            << "Timed out while waiting on additional SSRCs.";
+        send_stream_->ReconfigureVideoEncoder(
+            video_encoder_config_all_streams_);
+        EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
       }
     }
 
@@ -1662,8 +2289,8 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
     bool expect_single_ssrc_;
 
     VideoSendStream* send_stream_;
-    VideoEncoderConfig encoder_config_all_streams_;
-  } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);
+    VideoEncoderConfig video_encoder_config_all_streams_;
+  } test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);
 
   RunBaseTest(&test);
 }
@@ -1674,17 +2301,20 @@ TEST_F(EndToEndTest, ReportsSetEncoderRates) {
    public:
     EncoderRateStatsTest()
         : EndToEndTest(kDefaultTimeoutMs),
-          FakeEncoder(Clock::GetRealTimeClock()) {}
+          FakeEncoder(Clock::GetRealTimeClock()),
+          send_stream_(nullptr),
+          bitrate_kbps_(0) {}
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
     }
 
@@ -1692,20 +2322,20 @@ TEST_F(EndToEndTest, ReportsSetEncoderRates) {
       // Make sure not to trigger on any default zero bitrates.
       if (new_target_bitrate == 0)
         return 0;
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       bitrate_kbps_ = new_target_bitrate;
-      observation_complete_->Set();
+      observation_complete_.Set();
       return 0;
     }
 
     void PerformTest() override {
-      ASSERT_EQ(kEventSignaled, Wait())
+      ASSERT_TRUE(Wait())
           << "Timed out while waiting for encoder SetRates() call.";
       // Wait for GetStats to report a corresponding bitrate.
-      for (unsigned int i = 0; i < kDefaultTimeoutMs; ++i) {
+      for (int i = 0; i < kDefaultTimeoutMs; ++i) {
         VideoSendStream::Stats stats = send_stream_->GetStats();
         {
-          CriticalSectionScoped lock(crit_.get());
+          rtc::CritScope lock(&crit_);
           if ((stats.target_media_bitrate_bps + 500) / 1000 ==
               static_cast<int>(bitrate_kbps_)) {
             return;
@@ -1718,6 +2348,7 @@ TEST_F(EndToEndTest, ReportsSetEncoderRates) {
     }
 
    private:
+    rtc::CriticalSection crit_;
     VideoSendStream* send_stream_;
     uint32_t bitrate_kbps_ GUARDED_BY(crit_);
   } test;
@@ -1727,36 +2358,37 @@ TEST_F(EndToEndTest, ReportsSetEncoderRates) {
 
 TEST_F(EndToEndTest, GetStats) {
   static const int kStartBitrateBps = 3000000;
+  static const int kExpectedRenderDelayMs = 20;
   class StatsObserver : public test::EndToEndTest, public I420FrameCallback {
    public:
-    explicit StatsObserver(const FakeNetworkPipe::Config& config)
-        : EndToEndTest(kLongTimeoutMs, config),
+    StatsObserver()
+        : EndToEndTest(kLongTimeoutMs),
           send_stream_(nullptr),
           expected_send_ssrcs_(),
-          check_stats_event_(EventWrapper::Create()) {}
+          check_stats_event_(false, false) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      check_stats_event_->Set();
+      check_stats_event_.Set();
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
-      check_stats_event_->Set();
+      check_stats_event_.Set();
       return SEND_PACKET;
     }
 
     Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
-      check_stats_event_->Set();
+      check_stats_event_.Set();
       return SEND_PACKET;
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
-      check_stats_event_->Set();
+      check_stats_event_.Set();
       return SEND_PACKET;
     }
 
-    void FrameCallback(I420VideoFrame* video_frame) override {
+    void FrameCallback(VideoFrame* video_frame) override {
       // Ensure that we have at least 5ms send side delay.
       int64_t render_time = video_frame->render_time_ms();
       if (render_time > 0)
@@ -1774,6 +2406,12 @@ TEST_F(EndToEndTest, GetStats) {
         receive_stats_filled_["IncomingRate"] |=
             stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
 
+        send_stats_filled_["DecoderImplementationName"] |=
+            stats.decoder_implementation_name ==
+            test::FakeDecoder::kImplementationName;
+        receive_stats_filled_["RenderDelayAsHighAsExpected"] |=
+            stats.render_delay_ms >= kExpectedRenderDelayMs;
+
         receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
 
         receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
@@ -1798,7 +2436,7 @@ TEST_F(EndToEndTest, GetStats) {
             stats.frame_counts.key_frames != 0 ||
             stats.frame_counts.delta_frames != 0;
 
-        receive_stats_filled_["CName"] |= stats.c_name != "";
+        receive_stats_filled_["CName"] |= !stats.c_name.empty();
 
         receive_stats_filled_["RtcpPacketTypeCount"] |=
             stats.rtcp_packet_type_counts.fir_packets != 0 ||
@@ -1806,13 +2444,18 @@ TEST_F(EndToEndTest, GetStats) {
             stats.rtcp_packet_type_counts.pli_packets != 0 ||
             stats.rtcp_packet_type_counts.nack_requests != 0 ||
             stats.rtcp_packet_type_counts.unique_nack_requests != 0;
+
+        assert(stats.current_payload_type == -1 ||
+               stats.current_payload_type == kFakeVideoSendPayloadType);
+        receive_stats_filled_["IncomingPayloadType"] |=
+            stats.current_payload_type == kFakeVideoSendPayloadType;
       }
 
       return AllStatsFilled(receive_stats_filled_);
     }
 
     bool CheckSendStats() {
-      DCHECK(send_stream_ != nullptr);
+      RTC_DCHECK(send_stream_ != nullptr);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       send_stats_filled_["NumStreams"] |=
@@ -1821,6 +2464,10 @@ TEST_F(EndToEndTest, GetStats) {
       send_stats_filled_["CpuOveruseMetrics"] |=
           stats.avg_encode_time_ms != 0 || stats.encode_usage_percent != 0;
 
+      send_stats_filled_["EncoderImplementationName"] |=
+          stats.encoder_implementation_name ==
+          test::FakeEncoder::kImplementationName;
+
       for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
                stats.substreams.begin();
            it != stats.substreams.end(); ++it) {
@@ -1886,15 +2533,23 @@ TEST_F(EndToEndTest, GetStats) {
       return true;
     }
 
+    test::PacketTransport* CreateSendTransport(Call* sender_call) override {
+      FakeNetworkPipe::Config network_config;
+      network_config.loss_percent = 5;
+      return new test::PacketTransport(
+          sender_call, this, test::PacketTransport::kSender, network_config);
+    }
+
     Call::Config GetSenderCallConfig() override {
       Call::Config config = EndToEndTest::GetSenderCallConfig();
       config.bitrate_config.start_bitrate_bps = kStartBitrateBps;
       return config;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->pre_encode_callback = this;  // Used to inject delay.
       expected_cname_ = send_config->rtp.c_name = "SomeCName";
 
@@ -1903,12 +2558,13 @@ TEST_F(EndToEndTest, GetStats) {
         expected_send_ssrcs_.insert(ssrcs[i]);
         expected_receive_ssrcs_.push_back(
             (*receive_configs)[i].rtp.remote_ssrc);
+        (*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
       }
     }
 
-    size_t GetNumStreams() const override { return kNumSsrcs; }
+    size_t GetNumVideoStreams() const override { return kNumSsrcs; }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
@@ -1933,7 +2589,7 @@ TEST_F(EndToEndTest, GetStats) {
 
         int64_t time_until_timout_ = stop_time - now;
         if (time_until_timout_ > 0)
-          check_stats_event_->Wait(time_until_timout_);
+          check_stats_event_.Wait(time_until_timout_);
         now = clock->TimeInMilliseconds();
       }
 
@@ -1967,13 +2623,9 @@ TEST_F(EndToEndTest, GetStats) {
     std::set<uint32_t> expected_send_ssrcs_;
     std::string expected_cname_;
 
-    rtc::scoped_ptr<EventWrapper> check_stats_event_;
-  };
+    rtc::Event check_stats_event_;
+  } test;
 
-  FakeNetworkPipe::Config network_config;
-  network_config.loss_percent = 5;
-
-  StatsObserver test(network_config);
   RunBaseTest(&test);
 }
 
@@ -1995,7 +2647,7 @@ TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
           sent_rtp_(0) {}
 
    private:
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       receive_stream_ = receive_streams[0];
@@ -2005,7 +2657,7 @@ TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
       if (sent_rtp_ >= kNumRtpPacketsToSend) {
         VideoReceiveStream::Stats stats = receive_stream_->GetStats();
         if (kNumRtpPacketsToSend == stats.rtp_stats.transmitted.packets) {
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
         return DROP_PACKET;
       }
@@ -2014,7 +2666,7 @@ TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while verifying number of received RTP packets.";
     }
 
@@ -2063,17 +2715,18 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
       if (!observed_redundant_retransmission_[header.ssrc]) {
         observed_redundant_retransmission_[header.ssrc] = true;
         if (--ssrcs_to_observe_ == 0)
-          observation_complete_->Set();
+          observation_complete_.Set();
       }
 
       return SEND_PACKET;
     }
 
-    size_t GetNumStreams() const override { return kNumSsrcs; }
+    size_t GetNumVideoStreams() const override { return kNumSsrcs; }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
       for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
         encoder_config->streams[i].min_bitrate_bps = 10000;
@@ -2092,7 +2745,7 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for redundant payloads on all SSRCs.";
     }
 
@@ -2112,17 +2765,16 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
    public:
     explicit RtpSequenceObserver(bool use_rtx)
         : test::RtpRtcpObserver(kDefaultTimeoutMs),
-          crit_(CriticalSectionWrapper::CreateCriticalSection()),
           ssrcs_to_observe_(kNumSsrcs) {
       for (size_t i = 0; i < kNumSsrcs; ++i) {
-        configured_ssrcs_[kSendSsrcs[i]] = true;
+        configured_ssrcs_[kVideoSendSsrcs[i]] = true;
         if (use_rtx)
           configured_ssrcs_[kSendRtxSsrcs[i]] = true;
       }
     }
 
     void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       ssrc_observed_.clear();
       ssrcs_to_observe_ = num_expected_ssrcs;
     }
@@ -2177,12 +2829,12 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
         last_observed_timestamp_[ssrc] = timestamp;
       }
 
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       // Wait for media packets on all ssrcs.
       if (!ssrc_observed_[ssrc] && !only_padding) {
         ssrc_observed_[ssrc] = true;
         if (--ssrcs_to_observe_ == 0)
-          observation_complete_->Set();
+          observation_complete_.Set();
       }
 
       return SEND_PACKET;
@@ -2192,90 +2844,96 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
     std::map<uint32_t, uint32_t> last_observed_timestamp_;
     std::map<uint32_t, bool> configured_ssrcs_;
 
-    rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+    rtc::CriticalSection crit_;
     size_t ssrcs_to_observe_ GUARDED_BY(crit_);
     std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
   } observer(use_rtx);
 
-  CreateCalls(Call::Config(observer.SendTransport()),
-              Call::Config(observer.ReceiveTransport()));
-  observer.SetReceivers(sender_call_->Receiver(), nullptr);
+  CreateCalls(Call::Config(), Call::Config());
 
-  CreateSendConfig(kNumSsrcs);
+  test::PacketTransport send_transport(sender_call_.get(), &observer,
+                                       test::PacketTransport::kSender,
+                                       FakeNetworkPipe::Config());
+  test::PacketTransport receive_transport(nullptr, &observer,
+                                          test::PacketTransport::kReceiver,
+                                          FakeNetworkPipe::Config());
+  send_transport.SetReceiver(receiver_call_->Receiver());
+  receive_transport.SetReceiver(sender_call_->Receiver());
+
+  CreateSendConfig(kNumSsrcs, 0, &send_transport);
 
   if (use_rtx) {
     for (size_t i = 0; i < kNumSsrcs; ++i) {
-      send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
+      video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
     }
-    send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
+    video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
   }
 
   // Lower bitrates so that all streams send initially.
-  for (size_t i = 0; i < encoder_config_.streams.size(); ++i) {
-    encoder_config_.streams[i].min_bitrate_bps = 10000;
-    encoder_config_.streams[i].target_bitrate_bps = 15000;
-    encoder_config_.streams[i].max_bitrate_bps = 20000;
+  for (size_t i = 0; i < video_encoder_config_.streams.size(); ++i) {
+    video_encoder_config_.streams[i].min_bitrate_bps = 10000;
+    video_encoder_config_.streams[i].target_bitrate_bps = 15000;
+    video_encoder_config_.streams[i].max_bitrate_bps = 20000;
   }
 
   // Use the same total bitrates when sending a single stream to avoid lowering
   // the bitrate estimate and requiring a subsequent rampup.
-  VideoEncoderConfig one_stream = encoder_config_;
+  VideoEncoderConfig one_stream = video_encoder_config_;
   one_stream.streams.resize(1);
-  for (size_t i = 1; i < encoder_config_.streams.size(); ++i) {
+  for (size_t i = 1; i < video_encoder_config_.streams.size(); ++i) {
     one_stream.streams.front().min_bitrate_bps +=
-        encoder_config_.streams[i].min_bitrate_bps;
+        video_encoder_config_.streams[i].min_bitrate_bps;
     one_stream.streams.front().target_bitrate_bps +=
-        encoder_config_.streams[i].target_bitrate_bps;
+        video_encoder_config_.streams[i].target_bitrate_bps;
     one_stream.streams.front().max_bitrate_bps +=
-        encoder_config_.streams[i].max_bitrate_bps;
+        video_encoder_config_.streams[i].max_bitrate_bps;
   }
 
-  CreateMatchingReceiveConfigs();
+  CreateMatchingReceiveConfigs(&receive_transport);
 
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
-  EXPECT_EQ(kEventSignaled, observer.Wait())
+  EXPECT_TRUE(observer.Wait())
       << "Timed out waiting for all SSRCs to send packets.";
 
   // Test stream resetting more than once to make sure that the state doesn't
   // get set once (this could be due to using std::map::insert for instance).
   for (size_t i = 0; i < 3; ++i) {
     frame_generator_capturer_->Stop();
-    sender_call_->DestroyVideoSendStream(send_stream_);
+    sender_call_->DestroyVideoSendStream(video_send_stream_);
 
     // Re-create VideoSendStream with only one stream.
-    send_stream_ =
-        sender_call_->CreateVideoSendStream(send_config_, one_stream);
-    send_stream_->Start();
+    video_send_stream_ =
+        sender_call_->CreateVideoSendStream(video_send_config_, one_stream);
+    video_send_stream_->Start();
     CreateFrameGeneratorCapturer();
     frame_generator_capturer_->Start();
 
     observer.ResetExpectedSsrcs(1);
-    EXPECT_EQ(kEventSignaled, observer.Wait())
-        << "Timed out waiting for single RTP packet.";
+    EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
 
     // Reconfigure back to use all streams.
-    send_stream_->ReconfigureVideoEncoder(encoder_config_);
+    video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
     observer.ResetExpectedSsrcs(kNumSsrcs);
-    EXPECT_EQ(kEventSignaled, observer.Wait())
+    EXPECT_TRUE(observer.Wait())
         << "Timed out waiting for all SSRCs to send packets.";
 
     // Reconfigure down to one stream.
-    send_stream_->ReconfigureVideoEncoder(one_stream);
+    video_send_stream_->ReconfigureVideoEncoder(one_stream);
     observer.ResetExpectedSsrcs(1);
-    EXPECT_EQ(kEventSignaled, observer.Wait())
-        << "Timed out waiting for single RTP packet.";
+    EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
 
     // Reconfigure back to use all streams.
-    send_stream_->ReconfigureVideoEncoder(encoder_config_);
+    video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
     observer.ResetExpectedSsrcs(kNumSsrcs);
-    EXPECT_EQ(kEventSignaled, observer.Wait())
+    EXPECT_TRUE(observer.Wait())
         << "Timed out waiting for all SSRCs to send packets.";
   }
 
-  observer.StopSending();
+  send_transport.StopSending();
+  receive_transport.StopSending();
 
   Stop();
   DestroyStreams();
@@ -2304,26 +2962,27 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
     NetworkStateTest()
         : EndToEndTest(kDefaultTimeoutMs),
           FakeEncoder(Clock::GetRealTimeClock()),
-          test_crit_(CriticalSectionWrapper::CreateCriticalSection()),
-          encoded_frames_(EventWrapper::Create()),
-          packet_event_(EventWrapper::Create()),
-          sender_state_(Call::kNetworkUp),
+          encoded_frames_(false, false),
+          packet_event_(false, false),
+          sender_call_(nullptr),
+          receiver_call_(nullptr),
+          sender_state_(kNetworkUp),
           sender_rtp_(0),
           sender_rtcp_(0),
           receiver_rtcp_(0),
           down_frames_(0) {}
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      CriticalSectionScoped lock(test_crit_.get());
+      rtc::CritScope lock(&test_crit_);
       ++sender_rtp_;
-      packet_event_->Set();
+      packet_event_.Set();
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
-      CriticalSectionScoped lock(test_crit_.get());
+      rtc::CritScope lock(&test_crit_);
       ++sender_rtcp_;
-      packet_event_->Set();
+      packet_event_.Set();
       return SEND_PACKET;
     }
 
@@ -2333,9 +2992,9 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
     }
 
     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
-      CriticalSectionScoped lock(test_crit_.get());
+      rtc::CritScope lock(&test_crit_);
       ++receiver_rtcp_;
-      packet_event_->Set();
+      packet_event_.Set();
       return SEND_PACKET;
     }
 
@@ -2344,57 +3003,58 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
       receiver_call_ = receiver_call;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, encoded_frames_->Wait(kDefaultTimeoutMs))
+      EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
           << "No frames received by the encoder.";
       // Wait for packets from both sender/receiver.
       WaitForPacketsOrSilence(false, false);
 
       // Sender-side network down.
-      sender_call_->SignalNetworkState(Call::kNetworkDown);
+      sender_call_->SignalNetworkState(kNetworkDown);
       {
-        CriticalSectionScoped lock(test_crit_.get());
+        rtc::CritScope lock(&test_crit_);
         // After network goes down we shouldn't be encoding more frames.
-        sender_state_ = Call::kNetworkDown;
+        sender_state_ = kNetworkDown;
       }
       // Wait for receiver-packets and no sender packets.
       WaitForPacketsOrSilence(true, false);
 
       // Receiver-side network down.
-      receiver_call_->SignalNetworkState(Call::kNetworkDown);
+      receiver_call_->SignalNetworkState(kNetworkDown);
       WaitForPacketsOrSilence(true, true);
 
       // Network back up again for both.
       {
-        CriticalSectionScoped lock(test_crit_.get());
+        rtc::CritScope lock(&test_crit_);
         // It's OK to encode frames again, as we're about to bring up the
         // network.
-        sender_state_ = Call::kNetworkUp;
+        sender_state_ = kNetworkUp;
       }
-      sender_call_->SignalNetworkState(Call::kNetworkUp);
-      receiver_call_->SignalNetworkState(Call::kNetworkUp);
+      sender_call_->SignalNetworkState(kNetworkUp);
+      receiver_call_->SignalNetworkState(kNetworkUp);
       WaitForPacketsOrSilence(false, false);
     }
 
-    int32_t Encode(const I420VideoFrame& input_image,
+    int32_t Encode(const VideoFrame& input_image,
                    const CodecSpecificInfo* codec_specific_info,
-                   const std::vector<VideoFrameType>* frame_types) override {
+                   const std::vector<FrameType>* frame_types) override {
       {
-        CriticalSectionScoped lock(test_crit_.get());
-        if (sender_state_ == Call::kNetworkDown) {
+        rtc::CritScope lock(&test_crit_);
+        if (sender_state_ == kNetworkDown) {
           ++down_frames_;
           EXPECT_LE(down_frames_, 1)
               << "Encoding more than one frame while network is down.";
           if (down_frames_ > 1)
-            encoded_frames_->Set();
+            encoded_frames_.Set();
         } else {
-          encoded_frames_->Set();
+          encoded_frames_.Set();
         }
       }
       return test::FakeEncoder::Encode(
@@ -2408,17 +3068,17 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
       int initial_sender_rtcp;
       int initial_receiver_rtcp;
       {
-        CriticalSectionScoped lock(test_crit_.get());
+        rtc::CritScope lock(&test_crit_);
         initial_sender_rtp = sender_rtp_;
         initial_sender_rtcp = sender_rtcp_;
         initial_receiver_rtcp = receiver_rtcp_;
       }
       bool sender_done = false;
       bool receiver_done = false;
-      while(!sender_done || !receiver_done) {
-        packet_event_->Wait(kSilenceTimeoutMs);
+      while (!sender_done || !receiver_done) {
+        packet_event_.Wait(kSilenceTimeoutMs);
         int64_t time_now_ms = clock_->TimeInMilliseconds();
-        CriticalSectionScoped lock(test_crit_.get());
+        rtc::CritScope lock(&test_crit_);
         if (sender_down) {
           ASSERT_LE(sender_rtp_ - initial_sender_rtp, kNumAcceptedDowntimeRtp)
               << "RTP sent during sender-side downtime.";
@@ -2448,12 +3108,12 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
       }
     }
 
-    const rtc::scoped_ptr<CriticalSectionWrapper> test_crit_;
-    const rtc::scoped_ptr<EventWrapper> encoded_frames_;
-    const rtc::scoped_ptr<EventWrapper> packet_event_;
+    rtc::CriticalSection test_crit_;
+    rtc::Event encoded_frames_;
+    rtc::Event packet_event_;
     Call* sender_call_;
     Call* receiver_call_;
-    Call::NetworkState sender_state_ GUARDED_BY(test_crit_);
+    NetworkState sender_state_ GUARDED_BY(test_crit_);
     int sender_rtp_ GUARDED_BY(test_crit_);
     int sender_rtcp_ GUARDED_BY(test_crit_);
     int receiver_rtcp_ GUARDED_BY(test_crit_);
@@ -2467,22 +3127,20 @@ TEST_F(EndToEndTest, CallReportsRttForSender) {
   static const int kSendDelayMs = 30;
   static const int kReceiveDelayMs = 70;
 
+  CreateCalls(Call::Config(), Call::Config());
+
   FakeNetworkPipe::Config config;
   config.queue_delay_ms = kSendDelayMs;
-  test::DirectTransport sender_transport(config);
+  test::DirectTransport sender_transport(config, sender_call_.get());
   config.queue_delay_ms = kReceiveDelayMs;
-  test::DirectTransport receiver_transport(config);
-
-  CreateCalls(Call::Config(&sender_transport),
-              Call::Config(&receiver_transport));
-
+  test::DirectTransport receiver_transport(config, receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateSendConfig(1, 0, &sender_transport);
+  CreateMatchingReceiveConfigs(&receiver_transport);
 
-  CreateStreams();
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
   Start();
 
@@ -2505,25 +3163,25 @@ TEST_F(EndToEndTest, CallReportsRttForSender) {
 
 TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
   class UnusedEncoder : public test::FakeEncoder {
-    public:
+   public:
      UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
-     int32_t Encode(const I420VideoFrame& input_image,
+     int32_t Encode(const VideoFrame& input_image,
                     const CodecSpecificInfo* codec_specific_info,
-                    const std::vector<VideoFrameType>* frame_types) override {
+                    const std::vector<FrameType>* frame_types) override {
       ADD_FAILURE() << "Unexpected frame encode.";
       return test::FakeEncoder::Encode(
           input_image, codec_specific_info, frame_types);
     }
   };
 
-  UnusedTransport transport;
-  CreateSenderCall(Call::Config(&transport));
-  sender_call_->SignalNetworkState(Call::kNetworkDown);
+  CreateSenderCall(Call::Config());
+  sender_call_->SignalNetworkState(kNetworkDown);
 
-  CreateSendConfig(1);
+  UnusedTransport transport;
+  CreateSendConfig(1, 0, &transport);
   UnusedEncoder unused_encoder;
-  send_config_.encoder_settings.encoder = &unused_encoder;
-  CreateStreams();
+  video_send_config_.encoder_settings.encoder = &unused_encoder;
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
@@ -2534,17 +3192,15 @@ TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
 }
 
 TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
-  test::DirectTransport sender_transport;
-  CreateSenderCall(Call::Config(&sender_transport));
-  UnusedTransport transport;
-  CreateReceiverCall(Call::Config(&transport));
+  CreateCalls(Call::Config(), Call::Config());
+  receiver_call_->SignalNetworkState(kNetworkDown);
+
+  test::DirectTransport sender_transport(sender_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
-
-  receiver_call_->SignalNetworkState(Call::kNetworkDown);
-
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
-  CreateStreams();
+  CreateSendConfig(1, 0, &sender_transport);
+  UnusedTransport transport;
+  CreateMatchingReceiveConfigs(&transport);
+  CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
@@ -2556,38 +3212,121 @@ TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
   DestroyStreams();
 }
 
-// TODO(pbos): Remove this regression test when VideoEngine is no longer used as
-// a backend. This is to test that we hand channels back properly.
-TEST_F(EndToEndTest, CanCreateAndDestroyManyVideoStreams) {
-  test::NullTransport transport;
-  rtc::scoped_ptr<Call> call(Call::Create(Call::Config(&transport)));
-  test::FakeDecoder fake_decoder;
-  test::FakeEncoder fake_encoder(Clock::GetRealTimeClock());
-  for (size_t i = 0; i < 100; ++i) {
-    VideoSendStream::Config send_config;
-    send_config.encoder_settings.encoder = &fake_encoder;
-    send_config.encoder_settings.payload_name = "FAKE";
-    send_config.encoder_settings.payload_type = 123;
-
-    VideoEncoderConfig encoder_config;
-    encoder_config.streams = test::CreateVideoStreams(1);
-    send_config.rtp.ssrcs.push_back(1);
-    VideoSendStream* send_stream =
-        call->CreateVideoSendStream(send_config, encoder_config);
-    call->DestroyVideoSendStream(send_stream);
-
-    VideoReceiveStream::Config receive_config;
-    receive_config.rtp.remote_ssrc = 1;
-    receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
-    VideoReceiveStream::Decoder decoder;
-    decoder.decoder = &fake_decoder;
-    decoder.payload_type = 123;
-    decoder.payload_name = "FAKE";
-    receive_config.decoders.push_back(decoder);
-    VideoReceiveStream* receive_stream =
-        call->CreateVideoReceiveStream(receive_config);
-    call->DestroyVideoReceiveStream(receive_stream);
-  }
+void VerifyEmptyNackConfig(const NackConfig& config) {
+  EXPECT_EQ(0, config.rtp_history_ms)
+      << "Enabling NACK requires rtcp-fb: nack negotiation.";
 }
 
+void VerifyEmptyFecConfig(const FecConfig& config) {
+  EXPECT_EQ(-1, config.ulpfec_payload_type)
+      << "Enabling FEC requires rtpmap: ulpfec negotiation.";
+  EXPECT_EQ(-1, config.red_payload_type)
+      << "Enabling FEC requires rtpmap: red negotiation.";
+  EXPECT_EQ(-1, config.red_rtx_payload_type)
+      << "Enabling RTX in FEC requires rtpmap: rtx negotiation.";
+}
+
+TEST_F(EndToEndTest, VerifyDefaultSendConfigParameters) {
+  VideoSendStream::Config default_send_config(nullptr);
+  EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
+      << "Enabling NACK require rtcp-fb: nack negotiation.";
+  EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
+      << "Enabling RTX requires rtpmap: rtx negotiation.";
+  EXPECT_TRUE(default_send_config.rtp.extensions.empty())
+      << "Enabling RTP extensions require negotiation.";
+
+  VerifyEmptyNackConfig(default_send_config.rtp.nack);
+  VerifyEmptyFecConfig(default_send_config.rtp.fec);
+}
+
+TEST_F(EndToEndTest, VerifyDefaultReceiveConfigParameters) {
+  VideoReceiveStream::Config default_receive_config(nullptr);
+  EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode)
+      << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
+  EXPECT_FALSE(default_receive_config.rtp.remb)
+      << "REMB require rtcp-fb: goog-remb to be negotiated.";
+  EXPECT_FALSE(
+      default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
+      << "RTCP XR settings require rtcp-xr to be negotiated.";
+  EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
+      << "Enabling RTX requires rtpmap: rtx negotiation.";
+  EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
+      << "Enabling RTP extensions require negotiation.";
+
+  VerifyEmptyNackConfig(default_receive_config.rtp.nack);
+  VerifyEmptyFecConfig(default_receive_config.rtp.fec);
+}
+
+TEST_F(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
+  static const int kExtensionId = 8;
+  class TransportSequenceNumberTest : public test::EndToEndTest {
+   public:
+    TransportSequenceNumberTest()
+        : EndToEndTest(kDefaultTimeoutMs),
+          video_observed_(false),
+          audio_observed_(false) {
+      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                          kExtensionId);
+    }
+
+    size_t GetNumVideoStreams() const override { return 1; }
+    size_t GetNumAudioStreams() const override { return 1; }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+    }
+
+    void ModifyAudioConfigs(
+        AudioSendStream::Config* send_config,
+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();
+      (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+    }
+
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+      // Unwrap packet id and verify uniqueness.
+      int64_t packet_id =
+          unwrapper_.Unwrap(header.extension.transportSequenceNumber);
+      EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
+
+      if (header.ssrc == kVideoSendSsrcs[0])
+        video_observed_ = true;
+      if (header.ssrc == kAudioSendSsrc)
+        audio_observed_ = true;
+      if (audio_observed_ && video_observed_ &&
+          received_packet_ids_.size() == 50) {
+        size_t packet_id_range =
+            *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
+        EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
+        observation_complete_.Set();
+      }
+      return SEND_PACKET;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
+                             "packets with transport sequence number.";
+    }
+
+   private:
+    bool video_observed_;
+    bool audio_observed_;
+    SequenceNumberUnwrapper unwrapper_;
+    std::set<int64_t> received_packet_ids_;
+  } test;
+
+  RunBaseTest(&test);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/full_stack.cc b/media/webrtc/trunk/webrtc/video/full_stack.cc
index 7f98f38a79..e870c1ff14 100644
--- a/media/webrtc/trunk/webrtc/video/full_stack.cc
+++ b/media/webrtc/trunk/webrtc/video/full_stack.cc
@@ -9,652 +9,175 @@
  */
 #include <stdio.h>
 
-#include <deque>
-#include <map>
-
 #include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/call.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/cpu_info.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/test/call_test.h"
-#include "webrtc/test/direct_transport.h"
-#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/frame_generator.h"
-#include "webrtc/test/frame_generator_capturer.h"
-#include "webrtc/test/statistics.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "webrtc/video/video_quality_test.h"
 
 namespace webrtc {
 
 static const int kFullStackTestDurationSecs = 60;
 
-struct FullStackTestParams {
-  const char* test_label;
-  struct {
-    const char* name;
-    size_t width, height;
-    int fps;
-  } clip;
-  bool screenshare;
-  int min_bitrate_bps;
-  int target_bitrate_bps;
-  int max_bitrate_bps;
-  double avg_psnr_threshold;
-  double avg_ssim_threshold;
-  int test_durations_secs;
-  FakeNetworkPipe::Config link;
-};
-
-class FullStackTest : public test::CallTest {
- protected:
-  void RunTest(const FullStackTestParams& params);
-};
-
-class VideoAnalyzer : public PacketReceiver,
-                      public newapi::Transport,
-                      public VideoRenderer,
-                      public VideoSendStreamInput {
+class FullStackTest : public VideoQualityTest {
  public:
-  VideoAnalyzer(VideoSendStreamInput* input,
-                Transport* transport,
-                const char* test_label,
-                double avg_psnr_threshold,
-                double avg_ssim_threshold,
-                int duration_frames)
-      : input_(input),
-        transport_(transport),
-        receiver_(nullptr),
-        test_label_(test_label),
-        frames_to_process_(duration_frames),
-        frames_recorded_(0),
-        frames_processed_(0),
-        dropped_frames_(0),
-        last_render_time_(0),
-        rtp_timestamp_delta_(0),
-        crit_(CriticalSectionWrapper::CreateCriticalSection()),
-        avg_psnr_threshold_(avg_psnr_threshold),
-        avg_ssim_threshold_(avg_ssim_threshold),
-        comparison_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-        comparison_available_event_(EventWrapper::Create()),
-        done_(EventWrapper::Create()) {
-    // Create thread pool for CPU-expensive PSNR/SSIM calculations.
-
-    // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
-    // so that we don't accidentally starve "real" worker threads (codec etc).
-    // Also, don't allocate more than kMaxComparisonThreads, even if there are
-    // spare cores.
-
-    uint32_t num_cores = CpuInfo::DetectNumberOfCores();
-    DCHECK_GE(num_cores, 1u);
-    static const uint32_t kMinCoresLeft = 4;
-    static const uint32_t kMaxComparisonThreads = 8;
-
-    if (num_cores <= kMinCoresLeft) {
-      num_cores = 1;
-    } else {
-      num_cores -= kMinCoresLeft;
-      num_cores = std::min(num_cores, kMaxComparisonThreads);
-    }
-
-    for (uint32_t i = 0; i < num_cores; ++i) {
-      rtc::scoped_ptr<ThreadWrapper> thread =
-          ThreadWrapper::CreateThread(&FrameComparisonThread, this, "Analyzer");
-      EXPECT_TRUE(thread->Start());
-      comparison_thread_pool_.push_back(thread.release());
-    }
+  void RunTest(const VideoQualityTest::Params &params) {
+    RunWithAnalyzer(params);
   }
-
-  ~VideoAnalyzer() {
-    for (ThreadWrapper* thread : comparison_thread_pool_) {
-      EXPECT_TRUE(thread->Stop());
-      delete thread;
-    }
-  }
-
-  virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
-
-  DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) override {
-    rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-    RTPHeader header;
-    parser->Parse(packet, length, &header);
-    {
-      CriticalSectionScoped lock(crit_.get());
-      recv_times_[header.timestamp - rtp_timestamp_delta_] =
-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
-    }
-
-    return receiver_->DeliverPacket(packet, length);
-  }
-
-  void IncomingCapturedFrame(const I420VideoFrame& video_frame) override {
-    I420VideoFrame copy = video_frame;
-    copy.set_timestamp(copy.ntp_time_ms() * 90);
-
-    {
-      CriticalSectionScoped lock(crit_.get());
-      if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
-        first_send_frame_ = copy;
-
-      frames_.push_back(copy);
-    }
-
-    input_->IncomingCapturedFrame(video_frame);
-  }
-
-  bool SendRtp(const uint8_t* packet, size_t length) override {
-    rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-    RTPHeader header;
-    parser->Parse(packet, length, &header);
-
-    {
-      CriticalSectionScoped lock(crit_.get());
-      if (rtp_timestamp_delta_ == 0) {
-        rtp_timestamp_delta_ =
-            header.timestamp - first_send_frame_.timestamp();
-        first_send_frame_.Reset();
-      }
-      send_times_[header.timestamp - rtp_timestamp_delta_] =
-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
-    }
-
-    return transport_->SendRtp(packet, length);
-  }
-
-  bool SendRtcp(const uint8_t* packet, size_t length) override {
-    return transport_->SendRtcp(packet, length);
-  }
-
-  void RenderFrame(const I420VideoFrame& video_frame,
-                   int time_to_render_ms) override {
-    int64_t render_time_ms =
-        Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
-    uint32_t send_timestamp = video_frame.timestamp() - rtp_timestamp_delta_;
-
-    CriticalSectionScoped lock(crit_.get());
-
-    while (frames_.front().timestamp() < send_timestamp) {
-      AddFrameComparison(frames_.front(), last_rendered_frame_, true,
-                         render_time_ms);
-      frames_.pop_front();
-    }
-
-    I420VideoFrame reference_frame = frames_.front();
-    frames_.pop_front();
-    assert(!reference_frame.IsZeroSize());
-    EXPECT_EQ(reference_frame.timestamp(), send_timestamp);
-    assert(reference_frame.timestamp() == send_timestamp);
-
-    AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
-
-    last_rendered_frame_ = video_frame;
-  }
-
-  bool IsTextureSupported() const override { return false; }
-
-  void Wait() {
-    // Frame comparisons can be very expensive. Wait for test to be done, but
-    // at time-out check if frames_processed is going up. If so, give it more
-    // time, otherwise fail. Hopefully this will reduce test flakiness.
-
-    int last_frames_processed = -1;
-    EventTypeWrapper eventType;
-    while ((eventType = done_->Wait(FullStackTest::kDefaultTimeoutMs)) !=
-           kEventSignaled) {
-      int frames_processed;
-      {
-        CriticalSectionScoped crit(comparison_lock_.get());
-        frames_processed = frames_processed_;
-      }
-      if (last_frames_processed == -1) {
-        last_frames_processed = frames_processed;
-        continue;
-      }
-      ASSERT_GT(frames_processed, last_frames_processed)
-          << "Analyzer stalled while waiting for test to finish.";
-      last_frames_processed = frames_processed;
-    }
-  }
-
-  VideoSendStreamInput* input_;
-  Transport* transport_;
-  PacketReceiver* receiver_;
-
- private:
-  struct FrameComparison {
-    FrameComparison()
-        : dropped(false), send_time_ms(0), recv_time_ms(0), render_time_ms(0) {}
-
-    FrameComparison(const I420VideoFrame& reference,
-                    const I420VideoFrame& render,
-                    bool dropped,
-                    int64_t send_time_ms,
-                    int64_t recv_time_ms,
-                    int64_t render_time_ms)
-        : reference(reference),
-          render(render),
-          dropped(dropped),
-          send_time_ms(send_time_ms),
-          recv_time_ms(recv_time_ms),
-          render_time_ms(render_time_ms) {}
-
-    I420VideoFrame reference;
-    I420VideoFrame render;
-    bool dropped;
-    int64_t send_time_ms;
-    int64_t recv_time_ms;
-    int64_t render_time_ms;
-  };
-
-  void AddFrameComparison(const I420VideoFrame& reference,
-                          const I420VideoFrame& render,
-                          bool dropped,
-                          int64_t render_time_ms)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_) {
-    int64_t send_time_ms = send_times_[reference.timestamp()];
-    send_times_.erase(reference.timestamp());
-    int64_t recv_time_ms = recv_times_[reference.timestamp()];
-    recv_times_.erase(reference.timestamp());
-
-    CriticalSectionScoped crit(comparison_lock_.get());
-    comparisons_.push_back(FrameComparison(reference,
-                                           render,
-                                           dropped,
-                                           send_time_ms,
-                                           recv_time_ms,
-                                           render_time_ms));
-    comparison_available_event_->Set();
-  }
-
-  static bool FrameComparisonThread(void* obj) {
-    return static_cast<VideoAnalyzer*>(obj)->CompareFrames();
-  }
-
-  bool CompareFrames() {
-    if (AllFramesRecorded())
-      return false;
-
-    I420VideoFrame reference;
-    I420VideoFrame render;
-    FrameComparison comparison;
-
-    if (!PopComparison(&comparison)) {
-      // Wait until new comparison task is available, or test is done.
-      // If done, wake up remaining threads waiting.
-      comparison_available_event_->Wait(1000);
-      if (AllFramesRecorded()) {
-        comparison_available_event_->Set();
-        return false;
-      }
-      return true;  // Try again.
-    }
-
-    PerformFrameComparison(comparison);
-
-    if (FrameProcessed()) {
-      PrintResults();
-      done_->Set();
-      comparison_available_event_->Set();
-      return false;
-    }
-
-    return true;
-  }
-
-  bool PopComparison(FrameComparison* comparison) {
-    CriticalSectionScoped crit(comparison_lock_.get());
-    // If AllFramesRecorded() is true, it means we have already popped
-    // frames_to_process_ frames from comparisons_, so there is no more work
-    // for this thread to be done. frames_processed_ might still be lower if
-    // all comparisons are not done, but those frames are currently being
-    // worked on by other threads.
-    if (comparisons_.empty() || AllFramesRecorded())
-      return false;
-
-    *comparison = comparisons_.front();
-    comparisons_.pop_front();
-
-    FrameRecorded();
-    return true;
-  }
-
-  // Increment counter for number of frames received for comparison.
-  void FrameRecorded() {
-    CriticalSectionScoped crit(comparison_lock_.get());
-    ++frames_recorded_;
-  }
-
-  // Returns true if all frames to be compared have been taken from the queue.
-  bool AllFramesRecorded() {
-    CriticalSectionScoped crit(comparison_lock_.get());
-    assert(frames_recorded_ <= frames_to_process_);
-    return frames_recorded_ == frames_to_process_;
-  }
-
-  // Increase count of number of frames processed. Returns true if this was the
-  // last frame to be processed.
-  bool FrameProcessed() {
-    CriticalSectionScoped crit(comparison_lock_.get());
-    ++frames_processed_;
-    assert(frames_processed_ <= frames_to_process_);
-    return frames_processed_ == frames_to_process_;
-  }
-
-  void PrintResults() {
-    CriticalSectionScoped crit(comparison_lock_.get());
-    PrintResult("psnr", psnr_, " dB");
-    PrintResult("ssim", ssim_, "");
-    PrintResult("sender_time", sender_time_, " ms");
-    printf("RESULT dropped_frames: %s = %d frames\n", test_label_,
-           dropped_frames_);
-    PrintResult("receiver_time", receiver_time_, " ms");
-    PrintResult("total_delay_incl_network", end_to_end_, " ms");
-    PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
-    EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
-    EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
-  }
-
-  void PerformFrameComparison(const FrameComparison& comparison) {
-    // Perform expensive psnr and ssim calculations while not holding lock.
-    double psnr = I420PSNR(&comparison.reference, &comparison.render);
-    double ssim = I420SSIM(&comparison.reference, &comparison.render);
-
-    CriticalSectionScoped crit(comparison_lock_.get());
-    psnr_.AddSample(psnr);
-    ssim_.AddSample(ssim);
-    if (comparison.dropped) {
-      ++dropped_frames_;
-      return;
-    }
-    if (last_render_time_ != 0)
-      rendered_delta_.AddSample(comparison.render_time_ms - last_render_time_);
-    last_render_time_ = comparison.render_time_ms;
-
-    int64_t input_time_ms = comparison.reference.ntp_time_ms();
-    sender_time_.AddSample(comparison.send_time_ms - input_time_ms);
-    receiver_time_.AddSample(comparison.render_time_ms -
-                             comparison.recv_time_ms);
-    end_to_end_.AddSample(comparison.render_time_ms - input_time_ms);
-  }
-
-  void PrintResult(const char* result_type,
-                   test::Statistics stats,
-                   const char* unit) {
-    printf("RESULT %s: %s = {%f, %f}%s\n",
-           result_type,
-           test_label_,
-           stats.Mean(),
-           stats.StandardDeviation(),
-           unit);
-  }
-
-  const char* const test_label_;
-  test::Statistics sender_time_;
-  test::Statistics receiver_time_;
-  test::Statistics psnr_;
-  test::Statistics ssim_;
-  test::Statistics end_to_end_;
-  test::Statistics rendered_delta_;
-  const int frames_to_process_;
-  int frames_recorded_;
-  int frames_processed_;
-  int dropped_frames_;
-  int64_t last_render_time_;
-  uint32_t rtp_timestamp_delta_;
-
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-  std::deque<I420VideoFrame> frames_ GUARDED_BY(crit_);
-  I420VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
-  std::map<uint32_t, int64_t> send_times_ GUARDED_BY(crit_);
-  std::map<uint32_t, int64_t> recv_times_ GUARDED_BY(crit_);
-  I420VideoFrame first_send_frame_ GUARDED_BY(crit_);
-  const double avg_psnr_threshold_;
-  const double avg_ssim_threshold_;
-
-  const rtc::scoped_ptr<CriticalSectionWrapper> comparison_lock_;
-  std::vector<ThreadWrapper*> comparison_thread_pool_;
-  const rtc::scoped_ptr<EventWrapper> comparison_available_event_;
-  std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
-  const rtc::scoped_ptr<EventWrapper> done_;
 };
 
-void FullStackTest::RunTest(const FullStackTestParams& params) {
-  test::DirectTransport send_transport(params.link);
-  test::DirectTransport recv_transport(params.link);
-  VideoAnalyzer analyzer(nullptr, &send_transport, params.test_label,
-                         params.avg_psnr_threshold, params.avg_ssim_threshold,
-                         params.test_durations_secs * params.clip.fps);
-
-  CreateCalls(Call::Config(&analyzer), Call::Config(&recv_transport));
-
-  analyzer.SetReceiver(receiver_call_->Receiver());
-  send_transport.SetReceiver(&analyzer);
-  recv_transport.SetReceiver(sender_call_->Receiver());
-
-  CreateSendConfig(1);
-
-  rtc::scoped_ptr<VideoEncoder> encoder(
-      VideoEncoder::Create(VideoEncoder::kVp8));
-  send_config_.encoder_settings.encoder = encoder.get();
-  send_config_.encoder_settings.payload_name = "VP8";
-  send_config_.encoder_settings.payload_type = 124;
-  send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-  send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
-  send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
-
-  VideoStream* stream = &encoder_config_.streams[0];
-  stream->width = params.clip.width;
-  stream->height = params.clip.height;
-  stream->min_bitrate_bps = params.min_bitrate_bps;
-  stream->target_bitrate_bps = params.target_bitrate_bps;
-  stream->max_bitrate_bps = params.max_bitrate_bps;
-  stream->max_framerate = params.clip.fps;
-
-  if (params.screenshare) {
-    encoder_config_.content_type = VideoEncoderConfig::kScreenshare;
-    encoder_config_.min_transmit_bitrate_bps = 400 * 1000;
-    VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
-    vp8_settings.denoisingOn = false;
-    vp8_settings.frameDroppingOn = false;
-    vp8_settings.numberOfTemporalLayers = 2;
-    encoder_config_.encoder_specific_settings = &vp8_settings;
-
-    stream->temporal_layer_thresholds_bps.clear();
-    stream->temporal_layer_thresholds_bps.push_back(stream->target_bitrate_bps);
-  }
-
-  CreateMatchingReceiveConfigs();
-  receive_configs_[0].renderer = &analyzer;
-  receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-  receive_configs_[0].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrcs[0];
-  receive_configs_[0].rtp.rtx[kSendRtxPayloadType].payload_type =
-      kSendRtxPayloadType;
-
-  CreateStreams();
-  analyzer.input_ = send_stream_->Input();
-
-  if (params.screenshare) {
-    std::vector<std::string> slides;
-    slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
-    slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
-    slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
-    slides.push_back(test::ResourcePath("difficult_photo_1850_1110", "yuv"));
-
-    rtc::scoped_ptr<test::FrameGenerator> frame_generator(
-        test::FrameGenerator::CreateFromYuvFile(
-            slides, 1850, 1110,
-            10 * params.clip.fps)  // Cycle image every 10 seconds.
-        );
-    frame_generator_capturer_.reset(new test::FrameGeneratorCapturer(
-        Clock::GetRealTimeClock(), &analyzer, frame_generator.release(),
-        params.clip.fps));
-    ASSERT_TRUE(frame_generator_capturer_->Init());
-  } else {
-    frame_generator_capturer_.reset(
-        test::FrameGeneratorCapturer::CreateFromYuvFile(
-            &analyzer, test::ResourcePath(params.clip.name, "yuv"),
-            params.clip.width, params.clip.height, params.clip.fps,
-            Clock::GetRealTimeClock()));
-
-    ASSERT_TRUE(frame_generator_capturer_.get() != nullptr)
-        << "Could not create capturer for " << params.clip.name
-        << ".yuv. Is this resource file present?";
-  }
-
-  Start();
-
-  analyzer.Wait();
-
-  send_transport.StopSending();
-  recv_transport.StopSending();
-
-  Stop();
-
-  DestroyStreams();
-}
+// VideoQualityTest::Params params = {
+//   { ... },      // Common.
+//   { ... },      // Video-specific settings.
+//   { ... },      // Screenshare-specific settings.
+//   { ... },      // Analyzer settings.
+//   pipe,         // FakeNetworkPipe::Config
+//   { ... },      // Spatial scalability.
+//   logs          // bool
+// };
 
 TEST_F(FullStackTest, ParisQcifWithoutPacketLoss) {
-  FullStackTestParams paris_qcif = {"net_delay_0_0_plr_0",
-                                    {"paris_qcif", 176, 144, 30},
-                                    false,
-                                    300000,
-                                    300000,
-                                    300000,
-                                    36.0,
-                                    0.96,
-                                    kFullStackTestDurationSecs};
+  VideoQualityTest::Params paris_qcif = {
+      {176, 144, 30, 300000, 300000, 300000, "VP8", 1},
+      {"paris_qcif"},
+      {},
+      {"net_delay_0_0_plr_0", 36.0, 0.96, kFullStackTestDurationSecs}};
   RunTest(paris_qcif);
 }
 
 TEST_F(FullStackTest, ForemanCifWithoutPacketLoss) {
   // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
-  FullStackTestParams foreman_cif = {"foreman_cif_net_delay_0_0_plr_0",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     700000,
-                                     700000,
-                                     700000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 700000, 700000, 700000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_net_delay_0_0_plr_0", 0.0, 0.0, kFullStackTestDurationSecs}
+  };
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCifPlr5) {
-  FullStackTestParams foreman_cif = {"foreman_cif_delay_50_0_plr_5",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     30000,
-                                     500000,
-                                     2000000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
-  foreman_cif.link.loss_percent = 5;
-  foreman_cif.link.queue_delay_ms = 50;
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 30000, 500000, 2000000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_delay_50_0_plr_5", 0.0, 0.0, kFullStackTestDurationSecs}};
+  foreman_cif.pipe.loss_percent = 5;
+  foreman_cif.pipe.queue_delay_ms = 50;
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCif500kbps) {
-  FullStackTestParams foreman_cif = {"foreman_cif_500kbps",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     30000,
-                                     500000,
-                                     2000000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
-  foreman_cif.link.queue_length_packets = 0;
-  foreman_cif.link.queue_delay_ms = 0;
-  foreman_cif.link.link_capacity_kbps = 500;
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 30000, 500000, 2000000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_500kbps", 0.0, 0.0, kFullStackTestDurationSecs}};
+  foreman_cif.pipe.queue_length_packets = 0;
+  foreman_cif.pipe.queue_delay_ms = 0;
+  foreman_cif.pipe.link_capacity_kbps = 500;
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCif500kbpsLimitedQueue) {
-  FullStackTestParams foreman_cif = {"foreman_cif_500kbps_32pkts_queue",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     30000,
-                                     500000,
-                                     2000000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
-  foreman_cif.link.queue_length_packets = 32;
-  foreman_cif.link.queue_delay_ms = 0;
-  foreman_cif.link.link_capacity_kbps = 500;
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 30000, 500000, 2000000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_500kbps_32pkts_queue", 0.0, 0.0, kFullStackTestDurationSecs}
+  };
+  foreman_cif.pipe.queue_length_packets = 32;
+  foreman_cif.pipe.queue_delay_ms = 0;
+  foreman_cif.pipe.link_capacity_kbps = 500;
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCif500kbps100ms) {
-  FullStackTestParams foreman_cif = {"foreman_cif_500kbps_100ms",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     30000,
-                                     500000,
-                                     2000000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
-  foreman_cif.link.queue_length_packets = 0;
-  foreman_cif.link.queue_delay_ms = 100;
-  foreman_cif.link.link_capacity_kbps = 500;
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 30000, 500000, 2000000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_500kbps_100ms", 0.0, 0.0, kFullStackTestDurationSecs}};
+  foreman_cif.pipe.queue_length_packets = 0;
+  foreman_cif.pipe.queue_delay_ms = 100;
+  foreman_cif.pipe.link_capacity_kbps = 500;
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCif500kbps100msLimitedQueue) {
-  FullStackTestParams foreman_cif = {"foreman_cif_500kbps_100ms_32pkts_queue",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     30000,
-                                     500000,
-                                     2000000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
-  foreman_cif.link.queue_length_packets = 32;
-  foreman_cif.link.queue_delay_ms = 100;
-  foreman_cif.link.link_capacity_kbps = 500;
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 30000, 500000, 2000000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_500kbps_100ms_32pkts_queue", 0.0, 0.0,
+        kFullStackTestDurationSecs}};
+  foreman_cif.pipe.queue_length_packets = 32;
+  foreman_cif.pipe.queue_delay_ms = 100;
+  foreman_cif.pipe.link_capacity_kbps = 500;
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCif1000kbps100msLimitedQueue) {
-  FullStackTestParams foreman_cif = {"foreman_cif_1000kbps_100ms_32pkts_queue",
-                                     {"foreman_cif", 352, 288, 30},
-                                     false,
-                                     30000,
-                                     2000000,
-                                     2000000,
-                                     0.0,
-                                     0.0,
-                                     kFullStackTestDurationSecs};
-  foreman_cif.link.queue_length_packets = 32;
-  foreman_cif.link.queue_delay_ms = 100;
-  foreman_cif.link.link_capacity_kbps = 1000;
+  VideoQualityTest::Params foreman_cif = {
+      {352, 288, 30, 30000, 2000000, 2000000, "VP8", 1},
+      {"foreman_cif"},
+      {},
+      {"foreman_cif_1000kbps_100ms_32pkts_queue", 0.0, 0.0,
+        kFullStackTestDurationSecs}};
+  foreman_cif.pipe.queue_length_packets = 32;
+  foreman_cif.pipe.queue_delay_ms = 100;
+  foreman_cif.pipe.link_capacity_kbps = 1000;
   RunTest(foreman_cif);
 }
 
-TEST_F(FullStackTest, ScreenshareSlides) {
-  FullStackTestParams screenshare_params = {
-      "screenshare_slides",
-      {"screenshare_slides", 1850, 1110, 5},
-      true,
-      50000,
-      100000,
-      1000000,
-      0.0,
-      0.0,
-      kFullStackTestDurationSecs};
-  RunTest(screenshare_params);
+TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL) {
+  VideoQualityTest::Params screenshare = {
+      {1850, 1110, 5, 50000, 200000, 2000000, "VP8", 2, 1, 400000},
+      {},
+      {true, 10},
+      {"screenshare_slides", 0.0, 0.0, kFullStackTestDurationSecs}};
+  RunTest(screenshare);
+}
+
+TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_Scroll) {
+  VideoQualityTest::Params config = {
+      {1850, 1110 / 2, 5, 50000, 200000, 2000000, "VP8", 2, 1, 400000},
+      {},
+      {true, 10, 2},
+      {"screenshare_slides_scrolling", 0.0, 0.0, kFullStackTestDurationSecs}};
+  RunTest(config);
+}
+
+TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNet) {
+  VideoQualityTest::Params screenshare = {
+      {1850, 1110, 5, 50000, 200000, 2000000, "VP8", 2, 1, 400000},
+      {},          // Video-specific.
+      {true, 10},  // Screenshare-specific.
+      {"screenshare_slides_lossy_net", 0.0, 0.0, kFullStackTestDurationSecs}};
+  screenshare.pipe.loss_percent = 5;
+  screenshare.pipe.queue_delay_ms = 200;
+  screenshare.pipe.link_capacity_kbps = 500;
+  RunTest(screenshare);
+}
+
+TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_VeryLossyNet) {
+  VideoQualityTest::Params screenshare = {
+      {1850, 1110, 5, 50000, 200000, 2000000, "VP8", 2, 1, 400000},
+      {},          // Video-specific.
+      {true, 10},  // Screenshare-specific.
+      {"screenshare_slides_very_lossy", 0.0, 0.0, kFullStackTestDurationSecs}};
+  screenshare.pipe.loss_percent = 10;
+  screenshare.pipe.queue_delay_ms = 200;
+  screenshare.pipe.link_capacity_kbps = 500;
+  RunTest(screenshare);
+}
+
+TEST_F(FullStackTest, ScreenshareSlidesVP9_2SL) {
+  VideoQualityTest::Params screenshare = {
+      {1850, 1110, 5, 50000, 200000, 2000000, "VP9", 1, 0, 400000},
+      {},
+      {true, 10},
+      {"screenshare_slides_vp9_2sl", 0.0, 0.0, kFullStackTestDurationSecs},
+      {},
+      false,
+      {std::vector<VideoStream>(), 0, 2, 1}};
+  RunTest(screenshare);
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/full_stack_plot.py b/media/webrtc/trunk/webrtc/video/full_stack_plot.py
new file mode 100644
index 0000000000..4e09b4192a
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/full_stack_plot.py
@@ -0,0 +1,414 @@
+#!/usr/bin/env python
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+"""Generate graphs for data generated by loopback tests.
+
+Usage examples:
+  Show end to end time for a single full stack test.
+  ./full_stack_plot.py -df end_to_end -o 600 --frames 1000 vp9_data.txt
+
+  Show simultaneously PSNR and encoded frame size for two different runs of
+  full stack test. Averaged over a cycle of 200 frames. Used e.g. for
+  screenshare slide test.
+  ./full_stack_plot.py -c 200 -df psnr -drf encoded_frame_size \\
+                       before.txt after.txt
+
+  Similar to the previous test, but multiple graphs.
+  ./full_stack_plot.py -c 200 -df psnr vp8.txt vp9.txt --next \\
+                       -c 200 -df sender_time vp8.txt vp9.txt --next \\
+                       -c 200 -df end_to_end vp8.txt vp9.txt
+"""
+
+import argparse
+from collections import defaultdict
+import itertools
+import sys
+import matplotlib.pyplot as plt
+import numpy
+
+# Fields
+DROPPED = 0
+INPUT_TIME = 1              # ms (timestamp)
+SEND_TIME = 2               # ms (timestamp)
+RECV_TIME = 3               # ms (timestamp)
+RENDER_TIME = 4             # ms (timestamp)
+ENCODED_FRAME_SIZE = 5      # bytes
+PSNR = 6
+SSIM = 7
+ENCODE_TIME = 8             # ms (time interval)
+
+TOTAL_RAW_FIELDS = 9
+
+SENDER_TIME = TOTAL_RAW_FIELDS + 0
+RECEIVER_TIME = TOTAL_RAW_FIELDS + 1
+END_TO_END = TOTAL_RAW_FIELDS + 2
+RENDERED_DELTA = TOTAL_RAW_FIELDS + 3
+
+FIELD_MASK = 255
+
+# Options
+HIDE_DROPPED = 256
+RIGHT_Y_AXIS = 512
+
+# internal field id, field name, title
+_fields = [
+    # Raw
+    (DROPPED, "dropped", "dropped"),
+    (INPUT_TIME, "input_time_ms", "input time"),
+    (SEND_TIME, "send_time_ms", "send time"),
+    (RECV_TIME, "recv_time_ms", "recv time"),
+    (ENCODED_FRAME_SIZE, "encoded_frame_size", "encoded frame size"),
+    (PSNR, "psnr", "PSNR"),
+    (SSIM, "ssim", "SSIM"),
+    (RENDER_TIME, "render_time_ms", "render time"),
+    (ENCODE_TIME, "encode_time_ms", "encode time"),
+    # Auto-generated
+    (SENDER_TIME, "sender_time", "sender time"),
+    (RECEIVER_TIME, "receiver_time", "receiver time"),
+    (END_TO_END, "end_to_end", "end to end"),
+    (RENDERED_DELTA, "rendered_delta", "rendered delta"),
+]
+
+name_to_id = {field[1]: field[0] for field in _fields}
+id_to_title = {field[0]: field[2] for field in _fields}
+
+def field_arg_to_id(arg):
+  if arg == "none":
+    return None
+  if arg in name_to_id:
+    return name_to_id[arg]
+  if arg + "_ms" in name_to_id:
+    return name_to_id[arg + "_ms"]
+  raise Exception("Unrecognized field name \"{}\"".format(arg))
+
+
+class PlotLine(object):
+  """Data for a single graph line."""
+
+  def __init__(self, label, values, flags):
+    self.label = label
+    self.values = values
+    self.flags = flags
+
+
+class Data(object):
+  """Object representing one full stack test."""
+
+  def __init__(self, filename):
+    self.title = ""
+    self.length = 0
+    self.samples = defaultdict(list)
+
+    self._read_samples(filename)
+
+  def _read_samples(self, filename):
+    """Reads graph data from the given file."""
+    f = open(filename)
+    it = iter(f)
+
+    self.title = it.next().strip()
+    self.length = int(it.next())
+    field_names = [name.strip() for name in it.next().split()]
+    field_ids = [name_to_id[name] for name in field_names]
+
+    for field_id in field_ids:
+      self.samples[field_id] = [0.0] * self.length
+
+    for sample_id in xrange(self.length):
+      for col, value in enumerate(it.next().split()):
+        self.samples[field_ids[col]][sample_id] = float(value)
+
+    self._subtract_first_input_time()
+    self._generate_additional_data()
+
+    f.close()
+
+  def _subtract_first_input_time(self):
+    offset = self.samples[INPUT_TIME][0]
+    for field in [INPUT_TIME, SEND_TIME, RECV_TIME, RENDER_TIME]:
+      if field in self.samples:
+        self.samples[field] = [x - offset for x in self.samples[field]]
+
+  def _generate_additional_data(self):
+    """Calculates sender time, receiver time etc. from the raw data."""
+    s = self.samples
+    last_render_time = 0
+    for field_id in [SENDER_TIME, RECEIVER_TIME, END_TO_END, RENDERED_DELTA]:
+      s[field_id] = [0] * self.length
+
+    for k in range(self.length):
+      s[SENDER_TIME][k] = s[SEND_TIME][k] - s[INPUT_TIME][k]
+
+      decoded_time = s[RENDER_TIME][k]
+      s[RECEIVER_TIME][k] = decoded_time - s[RECV_TIME][k]
+      s[END_TO_END][k] = decoded_time - s[INPUT_TIME][k]
+      if not s[DROPPED][k]:
+        if k > 0:
+          s[RENDERED_DELTA][k] = decoded_time - last_render_time
+        last_render_time = decoded_time
+
+  def _hide(self, values):
+    """
+    Replaces values for dropped frames with None.
+    These values are then skipped by the plot() method.
+    """
+
+    return [None if self.samples[DROPPED][k] else values[k]
+            for k in range(len(values))]
+
+  def add_samples(self, config, target_lines_list):
+    """Creates graph lines from the current data set with given config."""
+    for field in config.fields:
+      # field is None means the user wants just to skip the color.
+      if field is None:
+        target_lines_list.append(None)
+        continue
+
+      field_id = field & FIELD_MASK
+      values = self.samples[field_id]
+
+      if field & HIDE_DROPPED:
+        values = self._hide(values)
+
+      target_lines_list.append(PlotLine(
+          self.title + " " + id_to_title[field_id],
+          values, field & ~FIELD_MASK))
+
+
+def average_over_cycle(values, length):
+  """
+  Returns the list:
+    [
+        avg(values[0], values[length], ...),
+        avg(values[1], values[length + 1], ...),
+        ...
+        avg(values[length - 1], values[2 * length - 1], ...),
+    ]
+
+  Skips None values when calculating the average value.
+  """
+
+  total = [0.0] * length
+  count = [0] * length
+  for k in range(len(values)):
+    if values[k] is not None:
+      total[k % length] += values[k]
+      count[k % length] += 1
+
+  result = [0.0] * length
+  for k in range(length):
+    result[k] = total[k] / count[k] if count[k] else None
+  return result
+
+
+class PlotConfig(object):
+  """Object representing a single graph."""
+
+  def __init__(self, fields, data_list, cycle_length=None, frames=None,
+               offset=0, output_filename=None, title="Graph"):
+    self.fields = fields
+    self.data_list = data_list
+    self.cycle_length = cycle_length
+    self.frames = frames
+    self.offset = offset
+    self.output_filename = output_filename
+    self.title = title
+
+  def plot(self, ax1):
+    lines = []
+    for data in self.data_list:
+      if not data:
+        # Add None lines to skip the colors.
+        lines.extend([None] * len(self.fields))
+      else:
+        data.add_samples(self, lines)
+
+    def _slice_values(values):
+      if self.offset:
+        values = values[self.offset:]
+      if self.frames:
+        values = values[:self.frames]
+      return values
+
+    length = None
+    for line in lines:
+      if line is None:
+        continue
+
+      line.values = _slice_values(line.values)
+      if self.cycle_length:
+        line.values = average_over_cycle(line.values, self.cycle_length)
+
+      if length is None:
+        length = len(line.values)
+      elif length != len(line.values):
+        raise Exception("All arrays should have the same length!")
+
+    ax1.set_xlabel("Frame", fontsize="large")
+    if any(line.flags & RIGHT_Y_AXIS for line in lines if line):
+      ax2 = ax1.twinx()
+      ax2.set_xlabel("Frame", fontsize="large")
+    else:
+      ax2 = None
+
+    # Have to implement color_cycle manually, due to two scales in a graph.
+    color_cycle = ["b", "r", "g", "c", "m", "y", "k"]
+    color_iter = itertools.cycle(color_cycle)
+
+    for line in lines:
+      if not line:
+        color_iter.next()
+        continue
+
+      if self.cycle_length:
+        x = numpy.array(range(self.cycle_length))
+      else:
+        x = numpy.array(range(self.offset, self.offset + len(line.values)))
+      y = numpy.array(line.values)
+      ax = ax2 if line.flags & RIGHT_Y_AXIS else ax1
+      ax.plot(x, y, "o-", label=line.label, markersize=3.0, linewidth=1.0,
+              color=color_iter.next())
+
+    ax1.grid(True)
+    if ax2:
+      ax1.legend(loc="upper left", shadow=True, fontsize="large")
+      ax2.legend(loc="upper right", shadow=True, fontsize="large")
+    else:
+      ax1.legend(loc="best", shadow=True, fontsize="large")
+
+
+def load_files(filenames):
+  result = []
+  for filename in filenames:
+    if filename in load_files.cache:
+      result.append(load_files.cache[filename])
+    else:
+      data = Data(filename)
+      load_files.cache[filename] = data
+      result.append(data)
+  return result
+load_files.cache = {}
+
+
+def get_parser():
+  class CustomAction(argparse.Action):
+    def __call__(self, parser, namespace, values, option_string=None):
+      if "ordered_args" not in namespace:
+        namespace.ordered_args = []
+      namespace.ordered_args.append((self.dest, values))
+
+  parser = argparse.ArgumentParser(
+      description=__doc__, formatter_class=argparse.RawDescriptionHelpFormatter)
+
+  parser.add_argument(
+      "-c", "--cycle_length", nargs=1, action=CustomAction,
+      type=int, help="Cycle length over which to average the values.")
+  parser.add_argument(
+      "-f", "--field", nargs=1, action=CustomAction,
+      help="Name of the field to show. Use 'none' to skip a color.")
+  parser.add_argument("-r", "--right", nargs=0, action=CustomAction,
+                      help="Use right Y axis for given field.")
+  parser.add_argument("-d", "--drop", nargs=0, action=CustomAction,
+                      help="Hide values for dropped frames.")
+  parser.add_argument("-o", "--offset", nargs=1, action=CustomAction, type=int,
+                      help="Frame offset.")
+  parser.add_argument("-n", "--next", nargs=0, action=CustomAction,
+                      help="Separator for multiple graphs.")
+  parser.add_argument(
+      "--frames", nargs=1, action=CustomAction, type=int,
+      help="Frame count to show or take into account while averaging.")
+  parser.add_argument("-t", "--title", nargs=1, action=CustomAction,
+                      help="Title of the graph.")
+  parser.add_argument(
+      "-O", "--output_filename", nargs=1, action=CustomAction,
+      help="Use to save the graph into a file. "
+           "Otherwise, a window will be shown.")
+  parser.add_argument(
+      "files", nargs="+", action=CustomAction,
+      help="List of text-based files generated by loopback tests.")
+  return parser
+
+
+def _plot_config_from_args(args, graph_num):
+  # Pylint complains about using kwargs, so have to do it this way.
+  cycle_length = None
+  frames = None
+  offset = 0
+  output_filename = None
+  title = "Graph"
+
+  fields = []
+  files = []
+  mask = 0
+  for key, values in args:
+    if key == "cycle_length":
+      cycle_length = values[0]
+    elif key == "frames":
+      frames = values[0]
+    elif key == "offset":
+      offset = values[0]
+    elif key == "output_filename":
+      output_filename = values[0]
+    elif key == "title":
+      title = values[0]
+    elif key == "drop":
+      mask |= HIDE_DROPPED
+    elif key == "right":
+      mask |= RIGHT_Y_AXIS
+    elif key == "field":
+      field_id = field_arg_to_id(values[0])
+      fields.append(field_id | mask if field_id is not None else None)
+      mask = 0  # Reset mask after the field argument.
+    elif key == "files":
+      files.extend(values)
+
+  if not files:
+    raise Exception("Missing file argument(s) for graph #{}".format(graph_num))
+  if not fields:
+    raise Exception("Missing field argument(s) for graph #{}".format(graph_num))
+
+  return PlotConfig(fields, load_files(files), cycle_length=cycle_length,
+      frames=frames, offset=offset, output_filename=output_filename,
+      title=title)
+
+
+def plot_configs_from_args(args):
+  """Generates plot configs for given command line arguments."""
+  # The way it works:
+  #   First we detect separators -n/--next and split arguments into groups, one
+  #   for each plot. For each group, we partially parse it with
+  #   argparse.ArgumentParser, modified to remember the order of arguments.
+  #   Then we traverse the argument list and fill the PlotConfig.
+  args = itertools.groupby(args, lambda x: x in ["-n", "--next"])
+  args = list(list(group) for match, group in args if not match)
+
+  parser = get_parser()
+  plot_configs = []
+  for index, raw_args in enumerate(args):
+    graph_args = parser.parse_args(raw_args).ordered_args
+    plot_configs.append(_plot_config_from_args(graph_args, index))
+  return plot_configs
+
+
+def show_or_save_plots(plot_configs):
+  for config in plot_configs:
+    fig = plt.figure(figsize=(14.0, 10.0))
+    ax = fig.add_subplot(1, 1, 1)
+
+    plt.title(config.title)
+    config.plot(ax)
+    if config.output_filename:
+      print "Saving to", config.output_filename
+      fig.savefig(config.output_filename)
+      plt.close(fig)
+
+  plt.show()
+
+if __name__ == "__main__":
+  show_or_save_plots(plot_configs_from_args(sys.argv[1:]))
diff --git a/media/webrtc/trunk/webrtc/video/loopback.cc b/media/webrtc/trunk/webrtc/video/loopback.cc
deleted file mode 100644
index 9ef0ec4224..0000000000
--- a/media/webrtc/trunk/webrtc/video/loopback.cc
+++ /dev/null
@@ -1,168 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include <map>
-
-#include "webrtc/video/loopback.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/call.h"
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/test/direct_transport.h"
-#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/run_loop.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-#include "webrtc/test/video_capturer.h"
-#include "webrtc/test/video_renderer.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-static const int kAbsSendTimeExtensionId = 7;
-
-static const uint32_t kSendSsrc = 0x654321;
-static const uint32_t kSendRtxSsrc = 0x654322;
-static const uint32_t kReceiverLocalSsrc = 0x123456;
-
-static const uint8_t kRtxPayloadType = 96;
-
-Loopback::Loopback(const Config& config)
-    : config_(config), clock_(Clock::GetRealTimeClock()) {
-}
-
-Loopback::~Loopback() {
-}
-
-void Loopback::Run() {
-  rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_;
-  if (config_.logs)
-    trace_to_stderr_.reset(new test::TraceToStderr);
-
-  rtc::scoped_ptr<test::VideoRenderer> local_preview(
-      test::VideoRenderer::Create("Local Preview", config_.width,
-                                  config_.height));
-  rtc::scoped_ptr<test::VideoRenderer> loopback_video(
-      test::VideoRenderer::Create("Loopback Video", config_.width,
-                                  config_.height));
-
-  FakeNetworkPipe::Config pipe_config;
-  pipe_config.loss_percent = config_.loss_percent;
-  pipe_config.link_capacity_kbps = config_.link_capacity_kbps;
-  pipe_config.queue_length_packets = config_.queue_size;
-  pipe_config.queue_delay_ms = config_.avg_propagation_delay_ms;
-  pipe_config.delay_standard_deviation_ms = config_.std_propagation_delay_ms;
-  test::DirectTransport transport(pipe_config);
-  Call::Config call_config(&transport);
-
-  call_config.bitrate_config.min_bitrate_bps =
-      static_cast<int>(config_.min_bitrate_kbps) * 1000;
-  call_config.bitrate_config.start_bitrate_bps =
-      static_cast<int>(config_.start_bitrate_kbps) * 1000;
-  call_config.bitrate_config.max_bitrate_bps =
-      static_cast<int>(config_.max_bitrate_kbps) * 1000;
-  rtc::scoped_ptr<Call> call(Call::Create(call_config));
-
-  // Loopback, call sends to itself.
-  transport.SetReceiver(call->Receiver());
-
-  VideoSendStream::Config send_config;
-  send_config.rtp.ssrcs.push_back(kSendSsrc);
-  send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc);
-  send_config.rtp.rtx.payload_type = kRtxPayloadType;
-  send_config.rtp.nack.rtp_history_ms = 1000;
-  send_config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
-
-  send_config.local_renderer = local_preview.get();
-  rtc::scoped_ptr<VideoEncoder> encoder;
-  if (config_.codec == "VP8") {
-    encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
-  } else if (config_.codec == "VP9") {
-    encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
-  } else {
-    // Codec not supported.
-    RTC_NOTREACHED() << "Codec not supported!";
-    return;
-  }
-  send_config.encoder_settings.encoder = encoder.get();
-  send_config.encoder_settings.payload_name = config_.codec;
-  send_config.encoder_settings.payload_type = 124;
-
-  VideoEncoderConfig encoder_config(CreateEncoderConfig());
-
-  VideoSendStream* send_stream =
-      call->CreateVideoSendStream(send_config, encoder_config);
-
-  rtc::scoped_ptr<test::VideoCapturer> capturer(CreateCapturer(send_stream));
-
-  VideoReceiveStream::Config receive_config;
-  receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
-  receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
-  receive_config.rtp.nack.rtp_history_ms = 1000;
-  receive_config.rtp.rtx[kRtxPayloadType].ssrc = kSendRtxSsrc;
-  receive_config.rtp.rtx[kRtxPayloadType].payload_type = kRtxPayloadType;
-  receive_config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
-  receive_config.renderer = loopback_video.get();
-  VideoReceiveStream::Decoder decoder =
-      test::CreateMatchingDecoder(send_config.encoder_settings);
-  receive_config.decoders.push_back(decoder);
-
-  VideoReceiveStream* receive_stream =
-      call->CreateVideoReceiveStream(receive_config);
-
-  receive_stream->Start();
-  send_stream->Start();
-  capturer->Start();
-
-  test::PressEnterToContinue();
-
-  capturer->Stop();
-  send_stream->Stop();
-  receive_stream->Stop();
-
-  call->DestroyVideoReceiveStream(receive_stream);
-  call->DestroyVideoSendStream(send_stream);
-
-  delete decoder.decoder;
-
-  transport.StopSending();
-}
-
-VideoEncoderConfig Loopback::CreateEncoderConfig() {
-  VideoEncoderConfig encoder_config;
-  encoder_config.streams = test::CreateVideoStreams(1);
-  VideoStream* stream = &encoder_config.streams[0];
-  stream->width = config_.width;
-  stream->height = config_.height;
-  stream->min_bitrate_bps = static_cast<int>(config_.min_bitrate_kbps) * 1000;
-  stream->max_bitrate_bps = static_cast<int>(config_.max_bitrate_kbps) * 1000;
-  stream->target_bitrate_bps =
-      static_cast<int>(config_.max_bitrate_kbps) * 1000;
-  stream->max_framerate = config_.fps;
-  stream->max_qp = 56;
-  return encoder_config;
-}
-
-test::VideoCapturer* Loopback::CreateCapturer(VideoSendStream* send_stream) {
-  return test::VideoCapturer::Create(send_stream->Input(), config_.width,
-                                     config_.height, config_.fps, clock_);
-}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/loopback.h b/media/webrtc/trunk/webrtc/video/loopback.h
deleted file mode 100644
index d0aa591b4c..0000000000
--- a/media/webrtc/trunk/webrtc/video/loopback.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-
-#include "webrtc/config.h"
-
-namespace webrtc {
-
-class VideoSendStream;
-class Clock;
-
-namespace test {
-
-class VideoCapturer;
-
-class Loopback {
- public:
-  struct Config {
-    size_t width;
-    size_t height;
-    int32_t fps;
-    size_t min_bitrate_kbps;
-    size_t start_bitrate_kbps;
-    size_t max_bitrate_kbps;
-    int32_t min_transmit_bitrate_kbps;
-    std::string codec;
-    int32_t loss_percent;
-    int32_t link_capacity_kbps;
-    int32_t queue_size;
-    int32_t avg_propagation_delay_ms;
-    int32_t std_propagation_delay_ms;
-    bool logs;
-  };
-
-  explicit Loopback(const Config& config);
-  virtual ~Loopback();
-
-  void Run();
-
- protected:
-  virtual VideoEncoderConfig CreateEncoderConfig();
-  virtual VideoCapturer* CreateCapturer(VideoSendStream* send_stream);
-
-  const Config config_;
-  Clock* const clock_;
-};
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc b/media/webrtc/trunk/webrtc/video/overuse_frame_detector.cc
similarity index 55%
rename from media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc
rename to media/webrtc/trunk/webrtc/video/overuse_frame_detector.cc
index 88d5d052bc..d971ad9d3e 100644
--- a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc
+++ b/media/webrtc/trunk/webrtc/video/overuse_frame_detector.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/overuse_frame_detector.h"
+#include "webrtc/video/overuse_frame_detector.h"
 
 #include <assert.h>
 #include <math.h>
@@ -19,21 +19,14 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/exp_filter.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
-// TODO(mflodman) Test different values for all of these to trigger correctly,
-// avoid fluctuations etc.
 namespace {
 const int64_t kProcessIntervalMs = 5000;
 
-// Weight factor to apply to the standard deviation.
-const float kWeightFactor = 0.997f;
-// Weight factor to apply to the average.
-const float kWeightFactorMean = 0.98f;
-
 // Delay between consecutive rampups. (Used for quick recovery.)
 const int kQuickRampUpDelayMs = 10 * 1000;
 // Delay between rampup attempts. Initially uses standard, scales up to max.
@@ -51,114 +44,24 @@ const float kMaxExp = 7.0f;
 
 }  // namespace
 
-// TODO(asapersson): Remove this class. Not used.
-Statistics::Statistics() :
-    sum_(0.0),
-    count_(0),
-    filtered_samples_(new rtc::ExpFilter(kWeightFactorMean)),
-    filtered_variance_(new rtc::ExpFilter(kWeightFactor)) {
-  Reset();
-}
-
-void Statistics::SetOptions(const CpuOveruseOptions& options) {
-  options_ = options;
-}
-
-void Statistics::Reset() {
-  sum_ =  0.0;
-  count_ = 0;
-  filtered_variance_->Reset(kWeightFactor);
-  filtered_variance_->Apply(1.0f, InitialVariance());
-}
-
-void Statistics::AddSample(float sample_ms) {
-  sum_ += sample_ms;
-  ++count_;
-
-  if (count_ < static_cast<uint32_t>(options_.min_frame_samples)) {
-    // Initialize filtered samples.
-    filtered_samples_->Reset(kWeightFactorMean);
-    filtered_samples_->Apply(1.0f, InitialMean());
-    return;
-  }
-
-  float exp = sample_ms / kSampleDiffMs;
-  exp = std::min(exp, kMaxExp);
-  filtered_samples_->Apply(exp, sample_ms);
-  filtered_variance_->Apply(exp, (sample_ms - filtered_samples_->filtered()) *
-                                 (sample_ms - filtered_samples_->filtered()));
-}
-
-float Statistics::InitialMean() const {
-  if (count_ == 0)
-    return 0.0;
-  return sum_ / count_;
-}
-
-float Statistics::InitialVariance() const {
-  // Start in between the underuse and overuse threshold.
-  float average_stddev = (options_.low_capture_jitter_threshold_ms +
-                          options_.high_capture_jitter_threshold_ms) / 2.0f;
-  return average_stddev * average_stddev;
-}
-
-float Statistics::Mean() const { return filtered_samples_->filtered(); }
-
-float Statistics::StdDev() const {
-  return sqrt(std::max(filtered_variance_->filtered(), 0.0f));
-}
-
-uint64_t Statistics::Count() const { return count_; }
-
-
-// Class for calculating the average encode time.
-class OveruseFrameDetector::EncodeTimeAvg {
- public:
-  EncodeTimeAvg()
-      : kWeightFactor(0.5f),
-        kInitialAvgEncodeTimeMs(5.0f),
-        filtered_encode_time_ms_(new rtc::ExpFilter(kWeightFactor)) {
-    filtered_encode_time_ms_->Apply(1.0f, kInitialAvgEncodeTimeMs);
-  }
-  ~EncodeTimeAvg() {}
-
-  void AddSample(float encode_time_ms, int64_t diff_last_sample_ms) {
-    float exp =  diff_last_sample_ms / kSampleDiffMs;
-    exp = std::min(exp, kMaxExp);
-    filtered_encode_time_ms_->Apply(exp, encode_time_ms);
-  }
-
-  int Value() const {
-    return static_cast<int>(filtered_encode_time_ms_->filtered() + 0.5);
-  }
-
- private:
-  const float kWeightFactor;
-  const float kInitialAvgEncodeTimeMs;
-  rtc::scoped_ptr<rtc::ExpFilter> filtered_encode_time_ms_;
-};
-
 // Class for calculating the processing usage on the send-side (the average
 // processing time of a frame divided by the average time difference between
 // captured frames).
 class OveruseFrameDetector::SendProcessingUsage {
  public:
-  SendProcessingUsage()
+  explicit SendProcessingUsage(const CpuOveruseOptions& options)
       : kWeightFactorFrameDiff(0.998f),
         kWeightFactorProcessing(0.995f),
         kInitialSampleDiffMs(40.0f),
         kMaxSampleDiffMs(45.0f),
         count_(0),
+        options_(options),
         filtered_processing_ms_(new rtc::ExpFilter(kWeightFactorProcessing)),
         filtered_frame_diff_ms_(new rtc::ExpFilter(kWeightFactorFrameDiff)) {
     Reset();
   }
   ~SendProcessingUsage() {}
 
-  void SetOptions(const CpuOveruseOptions& options) {
-    options_ = options;
-  }
-
   void Reset() {
     count_ = 0;
     filtered_frame_diff_ms_->Reset(kWeightFactorFrameDiff);
@@ -207,7 +110,7 @@ class OveruseFrameDetector::SendProcessingUsage {
   const float kInitialSampleDiffMs;
   const float kMaxSampleDiffMs;
   uint64_t count_;
-  CpuOveruseOptions options_;
+  const CpuOveruseOptions options_;
   rtc::scoped_ptr<rtc::ExpFilter> filtered_processing_ms_;
   rtc::scoped_ptr<rtc::ExpFilter> filtered_frame_diff_ms_;
 };
@@ -250,7 +153,7 @@ class OveruseFrameDetector::FrameQueue {
   }
 
   void Reset() { frame_times_.clear(); }
-  int NumFrames() const { return frame_times_.size(); }
+  int NumFrames() const { return static_cast<int>(frame_times_.size()); }
   int last_processing_time_ms() const { return last_processing_time_ms_; }
 
  private:
@@ -259,115 +162,40 @@ class OveruseFrameDetector::FrameQueue {
   int last_processing_time_ms_;
 };
 
-// TODO(asapersson): Remove this class. Not used.
-// Class for calculating the capture queue delay change.
-class OveruseFrameDetector::CaptureQueueDelay {
- public:
-  CaptureQueueDelay()
-      : kWeightFactor(0.5f),
-        delay_ms_(0),
-        filtered_delay_ms_per_s_(new rtc::ExpFilter(kWeightFactor)) {
-    filtered_delay_ms_per_s_->Apply(1.0f, 0.0f);
-  }
-  ~CaptureQueueDelay() {}
-
-  void FrameCaptured(int64_t now) {
-    const size_t kMaxSize = 200;
-    if (frames_.size() > kMaxSize) {
-      frames_.pop_front();
-    }
-    frames_.push_back(now);
-  }
-
-  void FrameProcessingStarted(int64_t now) {
-    if (frames_.empty()) {
-      return;
-    }
-    delay_ms_ = now - frames_.front();
-    frames_.pop_front();
-  }
-
-  void CalculateDelayChange(int64_t diff_last_sample_ms) {
-    if (diff_last_sample_ms <= 0) {
-      return;
-    }
-    float exp = static_cast<float>(diff_last_sample_ms) / kProcessIntervalMs;
-    exp = std::min(exp, kMaxExp);
-    filtered_delay_ms_per_s_->Apply(exp,
-                                    delay_ms_ * 1000.0f / diff_last_sample_ms);
-    ClearFrames();
-  }
-
-  void ClearFrames() {
-    frames_.clear();
-  }
-
-  int delay_ms() const {
-    return delay_ms_;
-  }
-
-  int Value() const {
-    return static_cast<int>(filtered_delay_ms_per_s_->filtered() + 0.5);
-  }
-
- private:
-  const float kWeightFactor;
-  std::list<int64_t> frames_;
-  int delay_ms_;
-  rtc::scoped_ptr<rtc::ExpFilter> filtered_delay_ms_per_s_;
-};
 
 OveruseFrameDetector::OveruseFrameDetector(
     Clock* clock,
+    const CpuOveruseOptions& options,
+    CpuOveruseObserver* observer,
     CpuOveruseMetricsObserver* metrics_observer)
-    : observer_(NULL),
+    : options_(options),
+      observer_(observer),
       metrics_observer_(metrics_observer),
       clock_(clock),
-      next_process_time_(clock_->TimeInMilliseconds()),
       num_process_times_(0),
       last_capture_time_(0),
+      num_pixels_(0),
+      next_process_time_(clock_->TimeInMilliseconds()),
       last_overuse_time_(0),
       checks_above_threshold_(0),
       num_overuse_detections_(0),
       last_rampup_time_(0),
       in_quick_rampup_(false),
       current_rampup_delay_ms_(kStandardRampUpDelayMs),
-      num_pixels_(0),
-      last_encode_sample_ms_(0),
-      encode_time_(new EncodeTimeAvg()),
-      usage_(new SendProcessingUsage()),
-      frame_queue_(new FrameQueue()),
       last_sample_time_ms_(0),
-      capture_queue_delay_(new CaptureQueueDelay()) {
-  DCHECK(metrics_observer != nullptr);
+      usage_(new SendProcessingUsage(options)),
+      frame_queue_(new FrameQueue()) {
+  RTC_DCHECK(metrics_observer != nullptr);
+  // Make sure stats are initially up-to-date. This simplifies unit testing
+  // since we don't have to trigger an update using one of the methods which
+  // would also alter the overuse state.
+  UpdateCpuOveruseMetrics();
   processing_thread_.DetachFromThread();
 }
 
 OveruseFrameDetector::~OveruseFrameDetector() {
 }
 
-void OveruseFrameDetector::SetObserver(CpuOveruseObserver* observer) {
-  rtc::CritScope cs(&crit_);
-  observer_ = observer;
-}
-
-void OveruseFrameDetector::SetOptions(const CpuOveruseOptions& options) {
-  assert(options.min_frame_samples > 0);
-  rtc::CritScope cs(&crit_);
-  if (options_.Equals(options)) {
-    return;
-  }
-  options_ = options;
-  capture_deltas_.SetOptions(options);
-  usage_->SetOptions(options);
-  ResetAll(num_pixels_);
-}
-
-int OveruseFrameDetector::CaptureQueueDelayMsPerS() const {
-  rtc::CritScope cs(&crit_);
-  return capture_queue_delay_->delay_ms();
-}
-
 int OveruseFrameDetector::LastProcessingTimeMs() const {
   rtc::CritScope cs(&crit_);
   return frame_queue_->last_processing_time_ms();
@@ -379,16 +207,13 @@ int OveruseFrameDetector::FramesInQueue() const {
 }
 
 void OveruseFrameDetector::UpdateCpuOveruseMetrics() {
-  metrics_.capture_jitter_ms = static_cast<int>(capture_deltas_.StdDev() + 0.5);
-  metrics_.avg_encode_time_ms = encode_time_->Value();
   metrics_.encode_usage_percent = usage_->Value();
-  metrics_.capture_queue_delay_ms_per_s = capture_queue_delay_->Value();
 
   metrics_observer_->CpuOveruseMetricsUpdated(metrics_);
 }
 
 int64_t OveruseFrameDetector::TimeUntilNextProcess() {
-  DCHECK(processing_thread_.CalledOnValidThread());
+  RTC_DCHECK(processing_thread_.CalledOnValidThread());
   return next_process_time_ - clock_->TimeInMilliseconds();
 }
 
@@ -408,10 +233,8 @@ bool OveruseFrameDetector::FrameTimeoutDetected(int64_t now) const {
 
 void OveruseFrameDetector::ResetAll(int num_pixels) {
   num_pixels_ = num_pixels;
-  capture_deltas_.Reset();
   usage_->Reset();
   frame_queue_->Reset();
-  capture_queue_delay_->ClearFrames();
   last_capture_time_ = 0;
   num_process_times_ = 0;
   UpdateCpuOveruseMetrics();
@@ -427,51 +250,21 @@ void OveruseFrameDetector::FrameCaptured(int width,
     ResetAll(width * height);
   }
 
-  if (last_capture_time_ != 0) {
-    capture_deltas_.AddSample(now - last_capture_time_);
+  if (last_capture_time_ != 0)
     usage_->AddCaptureSample(now - last_capture_time_);
-  }
+
   last_capture_time_ = now;
 
-  capture_queue_delay_->FrameCaptured(now);
-
-  if (options_.enable_extended_processing_usage) {
-    frame_queue_->Start(capture_time_ms, now);
-  }
-  UpdateCpuOveruseMetrics();
-}
-
-void OveruseFrameDetector::FrameProcessingStarted() {
-  rtc::CritScope cs(&crit_);
-  capture_queue_delay_->FrameProcessingStarted(clock_->TimeInMilliseconds());
-}
-
-void OveruseFrameDetector::FrameEncoded(int encode_time_ms) {
-  rtc::CritScope cs(&crit_);
-  int64_t now = clock_->TimeInMilliseconds();
-  if (last_encode_sample_ms_ != 0) {
-    int64_t diff_ms = now - last_encode_sample_ms_;
-    encode_time_->AddSample(encode_time_ms, diff_ms);
-  }
-  last_encode_sample_ms_ = now;
-
-  if (!options_.enable_extended_processing_usage) {
-    AddProcessingTime(encode_time_ms);
-  }
-  UpdateCpuOveruseMetrics();
+  frame_queue_->Start(capture_time_ms, now);
 }
 
 void OveruseFrameDetector::FrameSent(int64_t capture_time_ms) {
   rtc::CritScope cs(&crit_);
-  if (!options_.enable_extended_processing_usage) {
-    return;
-  }
   int delay_ms = frame_queue_->End(capture_time_ms,
                                    clock_->TimeInMilliseconds());
   if (delay_ms > 0) {
     AddProcessingTime(delay_ms);
   }
-  UpdateCpuOveruseMetrics();
 }
 
 void OveruseFrameDetector::AddProcessingTime(int elapsed_ms) {
@@ -481,10 +274,11 @@ void OveruseFrameDetector::AddProcessingTime(int elapsed_ms) {
     usage_->AddSample(elapsed_ms, diff_ms);
   }
   last_sample_time_ms_ = now;
+  UpdateCpuOveruseMetrics();
 }
 
 int32_t OveruseFrameDetector::Process() {
-  DCHECK(processing_thread_.CalledOnValidThread());
+  RTC_DCHECK(processing_thread_.CalledOnValidThread());
 
   int64_t now = clock_->TimeInMilliseconds();
 
@@ -492,20 +286,19 @@ int32_t OveruseFrameDetector::Process() {
   if (now < next_process_time_)
     return 0;
 
-  int64_t diff_ms = now - next_process_time_ + kProcessIntervalMs;
   next_process_time_ = now + kProcessIntervalMs;
 
-  rtc::CritScope cs(&crit_);
-  ++num_process_times_;
+  CpuOveruseMetrics current_metrics;
+  {
+    rtc::CritScope cs(&crit_);
+    ++num_process_times_;
 
-  capture_queue_delay_->CalculateDelayChange(diff_ms);
-  UpdateCpuOveruseMetrics();
-
-  if (num_process_times_ <= options_.min_process_count) {
-    return 0;
+    current_metrics = metrics_;
+    if (num_process_times_ <= options_.min_process_count)
+      return 0;
   }
 
-  if (IsOverusing()) {
+  if (IsOverusing(current_metrics)) {
     // If the last thing we did was going up, and now have to back down, we need
     // to check if this peak was short. If so we should back off to avoid going
     // back and forth between this load, the system doesn't seem to handle it.
@@ -530,7 +323,7 @@ int32_t OveruseFrameDetector::Process() {
 
     if (observer_ != NULL)
       observer_->OveruseDetected();
-  } else if (IsUnderusing(now)) {
+  } else if (IsUnderusing(current_metrics, now)) {
     last_rampup_time_ = now;
     in_quick_rampup_ = true;
 
@@ -540,25 +333,18 @@ int32_t OveruseFrameDetector::Process() {
 
   int rampup_delay =
       in_quick_rampup_ ? kQuickRampUpDelayMs : current_rampup_delay_ms_;
-  LOG(LS_VERBOSE) << " Frame stats: capture avg: " << capture_deltas_.Mean()
-                  << " capture stddev " << capture_deltas_.StdDev()
-                  << " encode usage " << usage_->Value()
+
+  LOG(LS_VERBOSE) << " Frame stats: "
+                  << " encode usage " << current_metrics.encode_usage_percent
                   << " overuse detections " << num_overuse_detections_
                   << " rampup delay " << rampup_delay;
 
   return 0;
 }
 
-bool OveruseFrameDetector::IsOverusing() {
-  bool overusing = false;
-  if (options_.enable_capture_jitter_method) {
-    overusing = capture_deltas_.StdDev() >=
-        options_.high_capture_jitter_threshold_ms;
-  } else if (options_.enable_encode_usage_method) {
-    overusing = usage_->Value() >= options_.high_encode_usage_threshold_percent;
-  }
-
-  if (overusing) {
+bool OveruseFrameDetector::IsOverusing(const CpuOveruseMetrics& metrics) {
+  if (metrics.encode_usage_percent >=
+      options_.high_encode_usage_threshold_percent) {
     ++checks_above_threshold_;
   } else {
     checks_above_threshold_ = 0;
@@ -566,18 +352,13 @@ bool OveruseFrameDetector::IsOverusing() {
   return checks_above_threshold_ >= options_.high_threshold_consecutive_count;
 }
 
-bool OveruseFrameDetector::IsUnderusing(int64_t time_now) {
+bool OveruseFrameDetector::IsUnderusing(const CpuOveruseMetrics& metrics,
+                                        int64_t time_now) {
   int delay = in_quick_rampup_ ? kQuickRampUpDelayMs : current_rampup_delay_ms_;
   if (time_now < last_rampup_time_ + delay)
     return false;
 
-  bool underusing = false;
-  if (options_.enable_capture_jitter_method) {
-    underusing = capture_deltas_.StdDev() <
-        options_.low_capture_jitter_threshold_ms;
-  } else if (options_.enable_encode_usage_method) {
-    underusing = usage_->Value() < options_.low_encode_usage_threshold_percent;
-  }
-  return underusing;
+  return metrics.encode_usage_percent <
+         options_.low_encode_usage_threshold_percent;
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h b/media/webrtc/trunk/webrtc/video/overuse_frame_detector.h
similarity index 51%
rename from media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h
rename to media/webrtc/trunk/webrtc/video/overuse_frame_detector.h
index 7c04cf7752..d2606c19e6 100644
--- a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h
+++ b/media/webrtc/trunk/webrtc/video/overuse_frame_detector.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_
-#define WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_
+#ifndef WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_
+#define WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
@@ -17,66 +17,78 @@
 #include "webrtc/base/exp_filter.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/video_engine/include/vie_base.h"
+#include "webrtc/modules/include/module.h"
 
 namespace webrtc {
 
 class Clock;
-class CpuOveruseObserver;
 
-// TODO(pbos): Move this somewhere appropriate.
-class Statistics {
+// CpuOveruseObserver is called when a system overuse is detected and
+// VideoEngine cannot keep up the encoding frequency.
+class CpuOveruseObserver {
  public:
-  Statistics();
+  // Called as soon as an overuse is detected.
+  virtual void OveruseDetected() = 0;
+  // Called periodically when the system is not overused any longer.
+  virtual void NormalUsage() = 0;
 
-  void AddSample(float sample_ms);
-  void Reset();
-  void SetOptions(const CpuOveruseOptions& options);
-
-  float Mean() const;
-  float StdDev() const;
-  uint64_t Count() const;
-
- private:
-  float InitialMean() const;
-  float InitialVariance() const;
-
-  float sum_;
-  uint64_t count_;
-  CpuOveruseOptions options_;
-  rtc::scoped_ptr<rtc::ExpFilter> filtered_samples_;
-  rtc::scoped_ptr<rtc::ExpFilter> filtered_variance_;
+ protected:
+  virtual ~CpuOveruseObserver() {}
 };
 
-// Use to detect system overuse based on jitter in incoming frames.
+struct CpuOveruseOptions {
+  CpuOveruseOptions()
+      : low_encode_usage_threshold_percent(55),
+        high_encode_usage_threshold_percent(85),
+        frame_timeout_interval_ms(1500),
+        min_frame_samples(120),
+        min_process_count(3),
+        high_threshold_consecutive_count(2) {}
+
+  int low_encode_usage_threshold_percent;  // Threshold for triggering underuse.
+  int high_encode_usage_threshold_percent;  // Threshold for triggering overuse.
+  // General settings.
+  int frame_timeout_interval_ms;  // The maximum allowed interval between two
+                                  // frames before resetting estimations.
+  int min_frame_samples;  // The minimum number of frames required.
+  int min_process_count;  // The number of initial process times required before
+                          // triggering an overuse/underuse.
+  int high_threshold_consecutive_count;  // The number of consecutive checks
+                                         // above the high threshold before
+                                         // triggering an overuse.
+};
+
+struct CpuOveruseMetrics {
+  CpuOveruseMetrics() : encode_usage_percent(-1) {}
+
+  int encode_usage_percent;  // Average encode time divided by the average time
+                             // difference between incoming captured frames.
+};
+
+class CpuOveruseMetricsObserver {
+ public:
+  virtual ~CpuOveruseMetricsObserver() {}
+  virtual void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) = 0;
+};
+
+
+// Use to detect system overuse based on the send-side processing time of
+// incoming frames.
 class OveruseFrameDetector : public Module {
  public:
   OveruseFrameDetector(Clock* clock,
+                       const CpuOveruseOptions& options,
+                       CpuOveruseObserver* overuse_observer,
                        CpuOveruseMetricsObserver* metrics_observer);
   ~OveruseFrameDetector();
 
-  // Registers an observer receiving overuse and underuse callbacks. Set
-  // 'observer' to NULL to disable callbacks.
-  void SetObserver(CpuOveruseObserver* observer);
-
-  // Sets options for overuse detection.
-  void SetOptions(const CpuOveruseOptions& options);
-
   // Called for each captured frame.
   void FrameCaptured(int width, int height, int64_t capture_time_ms);
 
-  // Called when the processing of a captured frame is started.
-  void FrameProcessingStarted();
-
-  // Called for each encoded frame.
-  void FrameEncoded(int encode_time_ms);
-
   // Called for each sent frame.
   void FrameSent(int64_t capture_time_ms);
 
   // Only public for testing.
-  int CaptureQueueDelayMsPerS() const;
   int LastProcessingTimeMs() const;
   int FramesInQueue() const;
 
@@ -85,9 +97,7 @@ class OveruseFrameDetector : public Module {
   int32_t Process() override;
 
  private:
-  class EncodeTimeAvg;
   class SendProcessingUsage;
-  class CaptureQueueDelay;
   class FrameQueue;
 
   void UpdateCpuOveruseMetrics() EXCLUSIVE_LOCKS_REQUIRED(crit_);
@@ -96,12 +106,9 @@ class OveruseFrameDetector : public Module {
   // need a guard.
   void AddProcessingTime(int elapsed_ms) EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
-  // TODO(asapersson): This method is always called on the processing thread.
-  // If locking is required, consider doing that locking inside the
-  // implementation and reduce scope as much as possible.  We should also
-  // see if we can avoid calling out to other methods while holding the lock.
-  bool IsOverusing() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  bool IsUnderusing(int64_t time_now) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // Only called on the processing thread.
+  bool IsOverusing(const CpuOveruseMetrics& metrics);
+  bool IsUnderusing(const CpuOveruseMetrics& metrics, int64_t time_now);
 
   bool FrameTimeoutDetected(int64_t now) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
   bool FrameSizeChanged(int num_pixels) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
@@ -114,52 +121,44 @@ class OveruseFrameDetector : public Module {
   // processing contends with reading stats and the processing thread.
   mutable rtc::CriticalSection crit_;
 
-  // Observer getting overuse reports.
-  CpuOveruseObserver* observer_ GUARDED_BY(crit_);
+  const CpuOveruseOptions options_;
 
-  CpuOveruseOptions options_ GUARDED_BY(crit_);
+  // Observer getting overuse reports.
+  CpuOveruseObserver* const observer_;
 
   // Stats metrics.
   CpuOveruseMetricsObserver* const metrics_observer_;
   CpuOveruseMetrics metrics_ GUARDED_BY(crit_);
 
   Clock* const clock_;
-  int64_t next_process_time_;  // Only accessed on the processing thread.
   int64_t num_process_times_ GUARDED_BY(crit_);
 
-  Statistics capture_deltas_ GUARDED_BY(crit_);
   int64_t last_capture_time_ GUARDED_BY(crit_);
 
-  // These six members are only accessed on the processing thread.
-  int64_t last_overuse_time_;
-  int checks_above_threshold_;
-  int num_overuse_detections_;
-
-  int64_t last_rampup_time_;
-  bool in_quick_rampup_;
-  int current_rampup_delay_ms_;
-
   // Number of pixels of last captured frame.
   int num_pixels_ GUARDED_BY(crit_);
 
-  int64_t last_encode_sample_ms_;  // Only accessed by one thread.
+  // These seven members are only accessed on the processing thread.
+  int64_t next_process_time_;
+  int64_t last_overuse_time_;
+  int checks_above_threshold_;
+  int num_overuse_detections_;
+  int64_t last_rampup_time_;
+  bool in_quick_rampup_;
+  int current_rampup_delay_ms_;
+
+  int64_t last_sample_time_ms_;    // Only accessed by one thread.
 
   // TODO(asapersson): Can these be regular members (avoid separate heap
   // allocs)?
-  const rtc::scoped_ptr<EncodeTimeAvg> encode_time_ GUARDED_BY(crit_);
   const rtc::scoped_ptr<SendProcessingUsage> usage_ GUARDED_BY(crit_);
   const rtc::scoped_ptr<FrameQueue> frame_queue_ GUARDED_BY(crit_);
 
-  int64_t last_sample_time_ms_;  // Only accessed by one thread.
-
-  const rtc::scoped_ptr<CaptureQueueDelay> capture_queue_delay_
-      GUARDED_BY(crit_);
-
   rtc::ThreadChecker processing_thread_;
 
-  DISALLOW_COPY_AND_ASSIGN(OveruseFrameDetector);
+  RTC_DISALLOW_COPY_AND_ASSIGN(OveruseFrameDetector);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_
+#endif  // WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_
diff --git a/media/webrtc/trunk/webrtc/video/overuse_frame_detector_unittest.cc b/media/webrtc/trunk/webrtc/video/overuse_frame_detector_unittest.cc
new file mode 100644
index 0000000000..65e006b485
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/overuse_frame_detector_unittest.cc
@@ -0,0 +1,310 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/overuse_frame_detector.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+  const int kWidth = 640;
+  const int kHeight = 480;
+  const int kFrameInterval33ms = 33;
+  const int kProcessIntervalMs = 5000;
+  const int kProcessTime5ms = 5;
+}  // namespace
+
+class MockCpuOveruseObserver : public CpuOveruseObserver {
+ public:
+  MockCpuOveruseObserver() {}
+  virtual ~MockCpuOveruseObserver() {}
+
+  MOCK_METHOD0(OveruseDetected, void());
+  MOCK_METHOD0(NormalUsage, void());
+};
+
+class CpuOveruseObserverImpl : public CpuOveruseObserver {
+ public:
+  CpuOveruseObserverImpl() :
+    overuse_(0),
+    normaluse_(0) {}
+  virtual ~CpuOveruseObserverImpl() {}
+
+  void OveruseDetected() { ++overuse_; }
+  void NormalUsage() { ++normaluse_; }
+
+  int overuse_;
+  int normaluse_;
+};
+
+class OveruseFrameDetectorTest : public ::testing::Test,
+                                 public CpuOveruseMetricsObserver {
+ protected:
+  virtual void SetUp() {
+    clock_.reset(new SimulatedClock(1234));
+    observer_.reset(new MockCpuOveruseObserver());
+    options_.min_process_count = 0;
+    ReinitializeOveruseDetector();
+  }
+
+  void ReinitializeOveruseDetector() {
+    overuse_detector_.reset(new OveruseFrameDetector(clock_.get(), options_,
+                                                     observer_.get(), this));
+  }
+
+  void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override {
+    metrics_ = metrics;
+  }
+
+  int InitialUsage() {
+    return ((options_.low_encode_usage_threshold_percent +
+             options_.high_encode_usage_threshold_percent) / 2.0f) + 0.5;
+  }
+
+  void InsertAndSendFramesWithInterval(
+      int num_frames, int interval_ms, int width, int height, int delay_ms) {
+    while (num_frames-- > 0) {
+      int64_t capture_time_ms = clock_->TimeInMilliseconds();
+      overuse_detector_->FrameCaptured(width, height, capture_time_ms);
+      clock_->AdvanceTimeMilliseconds(delay_ms);
+      overuse_detector_->FrameSent(capture_time_ms);
+      clock_->AdvanceTimeMilliseconds(interval_ms - delay_ms);
+    }
+  }
+
+  void TriggerOveruse(int num_times) {
+    const int kDelayMs = 32;
+    for (int i = 0; i < num_times; ++i) {
+      InsertAndSendFramesWithInterval(
+          1000, kFrameInterval33ms, kWidth, kHeight, kDelayMs);
+      overuse_detector_->Process();
+    }
+  }
+
+  void TriggerUnderuse() {
+    const int kDelayMs1 = 5;
+    const int kDelayMs2 = 6;
+    InsertAndSendFramesWithInterval(
+        1300, kFrameInterval33ms, kWidth, kHeight, kDelayMs1);
+    InsertAndSendFramesWithInterval(
+        1, kFrameInterval33ms, kWidth, kHeight, kDelayMs2);
+    overuse_detector_->Process();
+  }
+
+  int UsagePercent() { return metrics_.encode_usage_percent; }
+
+  CpuOveruseOptions options_;
+  rtc::scoped_ptr<SimulatedClock> clock_;
+  rtc::scoped_ptr<MockCpuOveruseObserver> observer_;
+  rtc::scoped_ptr<OveruseFrameDetector> overuse_detector_;
+  CpuOveruseMetrics metrics_;
+};
+
+
+// UsagePercent() > high_encode_usage_threshold_percent => overuse.
+// UsagePercent() < low_encode_usage_threshold_percent => underuse.
+TEST_F(OveruseFrameDetectorTest, TriggerOveruse) {
+  // usage > high => overuse
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
+  TriggerOveruse(options_.high_threshold_consecutive_count);
+}
+
+TEST_F(OveruseFrameDetectorTest, OveruseAndRecover) {
+  // usage > high => overuse
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
+  TriggerOveruse(options_.high_threshold_consecutive_count);
+  // usage < low => underuse
+  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(testing::AtLeast(1));
+  TriggerUnderuse();
+}
+
+TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithNoObserver) {
+  overuse_detector_.reset(
+      new OveruseFrameDetector(clock_.get(), options_, nullptr, this));
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
+  TriggerOveruse(options_.high_threshold_consecutive_count);
+  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
+  TriggerUnderuse();
+}
+
+TEST_F(OveruseFrameDetectorTest, DoubleOveruseAndRecover) {
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(2);
+  TriggerOveruse(options_.high_threshold_consecutive_count);
+  TriggerOveruse(options_.high_threshold_consecutive_count);
+  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(testing::AtLeast(1));
+  TriggerUnderuse();
+}
+
+TEST_F(OveruseFrameDetectorTest, TriggerUnderuseWithMinProcessCount) {
+  options_.min_process_count = 1;
+  CpuOveruseObserverImpl overuse_observer;
+  overuse_detector_.reset(new OveruseFrameDetector(clock_.get(), options_,
+                                                   &overuse_observer, this));
+  InsertAndSendFramesWithInterval(
+      1200, kFrameInterval33ms, kWidth, kHeight, kProcessTime5ms);
+  overuse_detector_->Process();
+  EXPECT_EQ(0, overuse_observer.normaluse_);
+  clock_->AdvanceTimeMilliseconds(kProcessIntervalMs);
+  overuse_detector_->Process();
+  EXPECT_EQ(1, overuse_observer.normaluse_);
+}
+
+TEST_F(OveruseFrameDetectorTest, ConstantOveruseGivesNoNormalUsage) {
+  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(64);
+  for (size_t i = 0; i < 64; ++i) {
+    TriggerOveruse(options_.high_threshold_consecutive_count);
+  }
+}
+
+TEST_F(OveruseFrameDetectorTest, ConsecutiveCountTriggersOveruse) {
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
+  options_.high_threshold_consecutive_count = 2;
+  ReinitializeOveruseDetector();
+  TriggerOveruse(2);
+}
+
+TEST_F(OveruseFrameDetectorTest, IncorrectConsecutiveCountTriggersNoOveruse) {
+  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
+  options_.high_threshold_consecutive_count = 2;
+  ReinitializeOveruseDetector();
+  TriggerOveruse(1);
+}
+
+TEST_F(OveruseFrameDetectorTest, ProcessingUsage) {
+  InsertAndSendFramesWithInterval(
+      1000, kFrameInterval33ms, kWidth, kHeight, kProcessTime5ms);
+  EXPECT_EQ(kProcessTime5ms * 100 / kFrameInterval33ms, UsagePercent());
+}
+
+TEST_F(OveruseFrameDetectorTest, ResetAfterResolutionChange) {
+  EXPECT_EQ(InitialUsage(), UsagePercent());
+  InsertAndSendFramesWithInterval(
+      1000, kFrameInterval33ms, kWidth, kHeight, kProcessTime5ms);
+  EXPECT_NE(InitialUsage(), UsagePercent());
+  // Verify reset.
+  InsertAndSendFramesWithInterval(
+      1, kFrameInterval33ms, kWidth, kHeight + 1, kProcessTime5ms);
+  EXPECT_EQ(InitialUsage(), UsagePercent());
+}
+
+TEST_F(OveruseFrameDetectorTest, ResetAfterFrameTimeout) {
+  EXPECT_EQ(InitialUsage(), UsagePercent());
+  InsertAndSendFramesWithInterval(
+      1000, kFrameInterval33ms, kWidth, kHeight, kProcessTime5ms);
+  EXPECT_NE(InitialUsage(), UsagePercent());
+  InsertAndSendFramesWithInterval(
+      2, options_.frame_timeout_interval_ms, kWidth, kHeight, kProcessTime5ms);
+  EXPECT_NE(InitialUsage(), UsagePercent());
+  // Verify reset.
+  InsertAndSendFramesWithInterval(
+      2, options_.frame_timeout_interval_ms + 1, kWidth, kHeight,
+      kProcessTime5ms);
+  EXPECT_EQ(InitialUsage(), UsagePercent());
+}
+
+TEST_F(OveruseFrameDetectorTest, MinFrameSamplesBeforeUpdating) {
+  options_.min_frame_samples = 40;
+  ReinitializeOveruseDetector();
+  InsertAndSendFramesWithInterval(
+      40, kFrameInterval33ms, kWidth, kHeight, kProcessTime5ms);
+  EXPECT_EQ(InitialUsage(), UsagePercent());
+  InsertAndSendFramesWithInterval(
+      1, kFrameInterval33ms, kWidth, kHeight, kProcessTime5ms);
+  EXPECT_NE(InitialUsage(), UsagePercent());
+}
+
+TEST_F(OveruseFrameDetectorTest, InitialProcessingUsage) {
+  EXPECT_EQ(InitialUsage(), UsagePercent());
+}
+
+TEST_F(OveruseFrameDetectorTest, FrameDelay_OneFrame) {
+  const int kProcessingTimeMs = 100;
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
+  overuse_detector_->FrameSent(33);
+  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
+  EXPECT_EQ(0, overuse_detector_->FramesInQueue());
+}
+
+TEST_F(OveruseFrameDetectorTest, FrameDelay_TwoFrames) {
+  const int kProcessingTimeMs1 = 100;
+  const int kProcessingTimeMs2 = 50;
+  const int kTimeBetweenFramesMs = 200;
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs1);
+  overuse_detector_->FrameSent(33);
+  EXPECT_EQ(kProcessingTimeMs1, overuse_detector_->LastProcessingTimeMs());
+  clock_->AdvanceTimeMilliseconds(kTimeBetweenFramesMs);
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 66);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs2);
+  overuse_detector_->FrameSent(66);
+  EXPECT_EQ(kProcessingTimeMs2, overuse_detector_->LastProcessingTimeMs());
+}
+
+TEST_F(OveruseFrameDetectorTest, FrameDelay_MaxQueueSize) {
+  const int kMaxQueueSize = 91;
+  for (int i = 0; i < kMaxQueueSize * 2; ++i) {
+    overuse_detector_->FrameCaptured(kWidth, kHeight, i);
+  }
+  EXPECT_EQ(kMaxQueueSize, overuse_detector_->FramesInQueue());
+}
+
+TEST_F(OveruseFrameDetectorTest, FrameDelay_NonProcessedFramesRemoved) {
+  const int kProcessingTimeMs = 100;
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 35);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 66);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 99);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
+  EXPECT_EQ(4, overuse_detector_->FramesInQueue());
+  overuse_detector_->FrameSent(66);
+  // Frame 33, 35 removed, 66 processed, 99 not processed.
+  EXPECT_EQ(2 * kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
+  EXPECT_EQ(1, overuse_detector_->FramesInQueue());
+  overuse_detector_->FrameSent(99);
+  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
+  EXPECT_EQ(0, overuse_detector_->FramesInQueue());
+}
+
+TEST_F(OveruseFrameDetectorTest, FrameDelay_ResetClearsFrames) {
+  const int kProcessingTimeMs = 100;
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
+  EXPECT_EQ(1, overuse_detector_->FramesInQueue());
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  // Verify reset (resolution changed).
+  overuse_detector_->FrameCaptured(kWidth, kHeight + 1, 66);
+  EXPECT_EQ(1, overuse_detector_->FramesInQueue());
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  overuse_detector_->FrameSent(66);
+  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
+  EXPECT_EQ(0, overuse_detector_->FramesInQueue());
+}
+
+TEST_F(OveruseFrameDetectorTest, FrameDelay_NonMatchingSendFrameIgnored) {
+  const int kProcessingTimeMs = 100;
+  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
+  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
+  overuse_detector_->FrameSent(34);
+  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
+  overuse_detector_->FrameSent(33);
+  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/payload_router.cc b/media/webrtc/trunk/webrtc/video/payload_router.cc
similarity index 93%
rename from media/webrtc/trunk/webrtc/video_engine/payload_router.cc
rename to media/webrtc/trunk/webrtc/video/payload_router.cc
index 958cdd869c..177f2dd4e8 100644
--- a/media/webrtc/trunk/webrtc/video_engine/payload_router.cc
+++ b/media/webrtc/trunk/webrtc/video/payload_router.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video/payload_router.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/payload_router.h b/media/webrtc/trunk/webrtc/video/payload_router.h
similarity index 91%
rename from media/webrtc/trunk/webrtc/video_engine/payload_router.h
rename to media/webrtc/trunk/webrtc/video/payload_router.h
index b96defd5e3..881145976d 100644
--- a/media/webrtc/trunk/webrtc/video_engine/payload_router.h
+++ b/media/webrtc/trunk/webrtc/video/payload_router.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
-#define WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+#ifndef WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
+#define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
 
 #include <list>
 #include <vector>
@@ -18,7 +18,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
 
@@ -77,9 +77,9 @@ class PayloadRouter {
 
   Atomic32 ref_count_;
 
-  DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
+  RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+#endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc b/media/webrtc/trunk/webrtc/video/payload_router_unittest.cc
similarity index 98%
rename from media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc
rename to media/webrtc/trunk/webrtc/video/payload_router_unittest.cc
index de391576d8..8c22f2fd5c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/payload_router_unittest.cc
@@ -14,9 +14,9 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video/payload_router.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
@@ -182,7 +182,7 @@ TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
 
   const uint32_t bitrate_1 = 10000;
   const uint32_t bitrate_2 = 76543;
-  std::vector<uint32_t> bitrates (2, bitrate_1);
+  std::vector<uint32_t> bitrates(2, bitrate_1);
   bitrates[1] = bitrate_2;
   EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
       .Times(1);
diff --git a/media/webrtc/trunk/webrtc/video/rampup_tests.cc b/media/webrtc/trunk/webrtc/video/rampup_tests.cc
deleted file mode 100644
index 35d3297f39..0000000000
--- a/media/webrtc/trunk/webrtc/video/rampup_tests.cc
+++ /dev/null
@@ -1,525 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/common.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video/rampup_tests.h"
-
-namespace webrtc {
-namespace {
-
-static const int kMaxPacketSize = 1500;
-
-std::vector<uint32_t> GenerateSsrcs(size_t num_streams,
-                                    uint32_t ssrc_offset) {
-  std::vector<uint32_t> ssrcs;
-  for (size_t i = 0; i != num_streams; ++i)
-    ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
-  return ssrcs;
-}
-}  // namespace
-
-StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs,
-                               newapi::Transport* feedback_transport,
-                               Clock* clock,
-                               RemoteBitrateEstimatorFactory* rbe_factory,
-                               RateControlType control_type)
-    : clock_(clock),
-      test_done_(EventWrapper::Create()),
-      rtp_parser_(RtpHeaderParser::Create()),
-      feedback_transport_(feedback_transport),
-      receive_stats_(ReceiveStatistics::Create(clock)),
-      payload_registry_(
-          new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      expected_bitrate_bps_(0),
-      start_bitrate_bps_(0),
-      rtx_media_ssrcs_(rtx_media_ssrcs),
-      total_sent_(0),
-      padding_sent_(0),
-      rtx_media_sent_(0),
-      total_packets_sent_(0),
-      padding_packets_sent_(0),
-      rtx_media_packets_sent_(0),
-      test_start_ms_(clock_->TimeInMilliseconds()),
-      ramp_up_finished_ms_(0) {
-  // Ideally we would only have to instantiate an RtcpSender, an
-  // RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current
-  // state of the RTP module we need a full module and receive statistics to
-  // be able to produce an RTCP with REMB.
-  RtpRtcp::Configuration config;
-  config.receive_statistics = receive_stats_.get();
-  feedback_transport_.Enable();
-  config.outgoing_transport = &feedback_transport_;
-  rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
-  rtp_rtcp_->SetREMBStatus(true);
-  rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
-  rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
-                                          kAbsSendTimeExtensionId);
-  rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
-                                          kTransmissionTimeOffsetExtensionId);
-  const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
-  remote_bitrate_estimator_.reset(
-      rbe_factory->Create(this, clock, control_type,
-                          kRemoteBitrateEstimatorMinBitrateBps));
-}
-
-void StreamObserver::set_expected_bitrate_bps(
-    unsigned int expected_bitrate_bps) {
-  CriticalSectionScoped lock(crit_.get());
-  expected_bitrate_bps_ = expected_bitrate_bps;
-}
-
-void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) {
-  CriticalSectionScoped lock(crit_.get());
-  start_bitrate_bps_ = start_bitrate_bps;
-}
-
-void StreamObserver::OnReceiveBitrateChanged(
-    const std::vector<unsigned int>& ssrcs, unsigned int bitrate) {
-  CriticalSectionScoped lock(crit_.get());
-  DCHECK_GT(expected_bitrate_bps_, 0u);
-  if (start_bitrate_bps_ != 0) {
-    // For tests with an explicitly set start bitrate, verify the first
-    // bitrate estimate is close to the start bitrate and lower than the
-    // test target bitrate. This is to verify a call respects the configured
-    // start bitrate, but due to the BWE implementation we can't guarantee the
-    // first estimate really is as high as the start bitrate.
-    EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_);
-    start_bitrate_bps_ = 0;
-  }
-  if (bitrate >= expected_bitrate_bps_) {
-    ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
-    // Just trigger if there was any rtx padding packet.
-    if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) {
-      TriggerTestDone();
-    }
-  }
-  rtp_rtcp_->SetREMBData(bitrate, ssrcs);
-  rtp_rtcp_->Process();
-}
-
-bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
-  CriticalSectionScoped lock(crit_.get());
-  RTPHeader header;
-  EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
-  receive_stats_->IncomingPacket(header, length, false);
-  payload_registry_->SetIncomingPayloadType(header);
-  remote_bitrate_estimator_->IncomingPacket(
-      clock_->TimeInMilliseconds(), length - 12, header);
-  if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
-    remote_bitrate_estimator_->Process();
-  }
-  total_sent_ += length;
-  padding_sent_ += header.paddingLength;
-  ++total_packets_sent_;
-  if (header.paddingLength > 0)
-    ++padding_packets_sent_;
-  if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) {
-    rtx_media_sent_ += length - header.headerLength - header.paddingLength;
-    if (header.paddingLength == 0)
-      ++rtx_media_packets_sent_;
-    uint8_t restored_packet[kMaxPacketSize];
-    uint8_t* restored_packet_ptr = restored_packet;
-    size_t restored_length = length;
-    payload_registry_->RestoreOriginalPacket(&restored_packet_ptr,
-                                             packet,
-                                             &restored_length,
-                                             rtx_media_ssrcs_[header.ssrc],
-                                             header);
-    length = restored_length;
-    EXPECT_TRUE(rtp_parser_->Parse(
-        restored_packet, static_cast<int>(length), &header));
-  } else {
-    rtp_rtcp_->SetRemoteSSRC(header.ssrc);
-  }
-  return true;
-}
-
-bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
-  return true;
-}
-
-EventTypeWrapper StreamObserver::Wait() {
-  return test_done_->Wait(test::CallTest::kLongTimeoutMs);
-}
-
-void StreamObserver::ReportResult(const std::string& measurement,
-                  size_t value,
-                  const std::string& units) {
-  webrtc::test::PrintResult(
-      measurement, "",
-      ::testing::UnitTest::GetInstance()->current_test_info()->name(),
-      value, units, false);
-}
-
-void StreamObserver::TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
-  ReportResult("ramp-up-total-sent", total_sent_, "bytes");
-  ReportResult("ramp-up-padding-sent", padding_sent_, "bytes");
-  ReportResult("ramp-up-rtx-media-sent", rtx_media_sent_, "bytes");
-  ReportResult("ramp-up-total-packets-sent", total_packets_sent_, "packets");
-  ReportResult("ramp-up-padding-packets-sent",
-               padding_packets_sent_,
-               "packets");
-  ReportResult("ramp-up-rtx-packets-sent",
-               rtx_media_packets_sent_,
-               "packets");
-  ReportResult("ramp-up-time",
-               ramp_up_finished_ms_ - test_start_ms_,
-               "milliseconds");
-  test_done_->Set();
-}
-
-LowRateStreamObserver::LowRateStreamObserver(
-    newapi::Transport* feedback_transport,
-    Clock* clock,
-    size_t number_of_streams,
-    bool rtx_used)
-    : clock_(clock),
-      number_of_streams_(number_of_streams),
-      rtx_used_(rtx_used),
-      test_done_(EventWrapper::Create()),
-      rtp_parser_(RtpHeaderParser::Create()),
-      feedback_transport_(feedback_transport),
-      receive_stats_(ReceiveStatistics::Create(clock)),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      send_stream_(nullptr),
-      test_state_(kFirstRampup),
-      state_start_ms_(clock_->TimeInMilliseconds()),
-      interval_start_ms_(state_start_ms_),
-      last_remb_bps_(0),
-      sent_bytes_(0),
-      total_overuse_bytes_(0),
-      suspended_in_stats_(false) {
-  RtpRtcp::Configuration config;
-  config.receive_statistics = receive_stats_.get();
-  feedback_transport_.Enable();
-  config.outgoing_transport = &feedback_transport_;
-  rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
-  rtp_rtcp_->SetREMBStatus(true);
-  rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
-  rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
-                                          kAbsSendTimeExtensionId);
-  AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
-  const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000;
-  remote_bitrate_estimator_.reset(
-      rbe_factory.Create(this, clock, kAimdControl,
-                         kRemoteBitrateEstimatorMinBitrateBps));
-  forward_transport_config_.link_capacity_kbps =
-      kHighBandwidthLimitBps / 1000;
-  forward_transport_config_.queue_length_packets = 100;  // Something large.
-  test::DirectTransport::SetConfig(forward_transport_config_);
-  test::DirectTransport::SetReceiver(this);
-}
-
-void LowRateStreamObserver::SetSendStream(VideoSendStream* send_stream) {
-  CriticalSectionScoped lock(crit_.get());
-  send_stream_ = send_stream;
-}
-
-void LowRateStreamObserver::OnReceiveBitrateChanged(
-    const std::vector<unsigned int>& ssrcs,
-    unsigned int bitrate) {
-  CriticalSectionScoped lock(crit_.get());
-  rtp_rtcp_->SetREMBData(bitrate, ssrcs);
-  rtp_rtcp_->Process();
-  last_remb_bps_ = bitrate;
-}
-
-bool LowRateStreamObserver::SendRtp(const uint8_t* data, size_t length) {
-  CriticalSectionScoped lock(crit_.get());
-  sent_bytes_ += length;
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  if (now_ms > interval_start_ms_ + 1000) {  // Let at least 1 second pass.
-    // Verify that the send rate was about right.
-    unsigned int average_rate_bps = static_cast<unsigned int>(sent_bytes_) *
-                                    8 * 1000 / (now_ms - interval_start_ms_);
-    // TODO(holmer): Why is this failing?
-    // EXPECT_LT(average_rate_bps, last_remb_bps_ * 1.1);
-    if (average_rate_bps > last_remb_bps_ * 1.1) {
-      total_overuse_bytes_ +=
-          sent_bytes_ -
-          last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000;
-    }
-    EvolveTestState(average_rate_bps);
-    interval_start_ms_ = now_ms;
-    sent_bytes_ = 0;
-  }
-  return test::DirectTransport::SendRtp(data, length);
-}
-
-PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
-    const uint8_t* packet, size_t length) {
-  CriticalSectionScoped lock(crit_.get());
-  RTPHeader header;
-  EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
-  receive_stats_->IncomingPacket(header, length, false);
-  remote_bitrate_estimator_->IncomingPacket(
-      clock_->TimeInMilliseconds(), length - 12, header);
-  if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
-    remote_bitrate_estimator_->Process();
-  }
-  suspended_in_stats_ = send_stream_->GetStats().suspended;
-  return DELIVERY_OK;
-}
-
-bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) {
-  return true;
-}
-
-std::string LowRateStreamObserver::GetModifierString() {
-  std::string str("_");
-  char temp_str[5];
-  sprintf(temp_str, "%i",
-      static_cast<int>(number_of_streams_));
-  str += std::string(temp_str);
-  str += "stream";
-  str += (number_of_streams_ > 1 ? "s" : "");
-  str += "_";
-  str += (rtx_used_ ? "" : "no");
-  str += "rtx";
-  return str;
-}
-
-void LowRateStreamObserver::EvolveTestState(unsigned int bitrate_bps) {
-  int64_t now = clock_->TimeInMilliseconds();
-  CriticalSectionScoped lock(crit_.get());
-  DCHECK(send_stream_ != nullptr);
-  switch (test_state_) {
-    case kFirstRampup: {
-      EXPECT_FALSE(suspended_in_stats_);
-      if (bitrate_bps > kExpectedHighBitrateBps) {
-        // The first ramp-up has reached the target bitrate. Change the
-        // channel limit, and move to the next test state.
-        forward_transport_config_.link_capacity_kbps =
-            kLowBandwidthLimitBps / 1000;
-        test::DirectTransport::SetConfig(forward_transport_config_);
-        test_state_ = kLowRate;
-        webrtc::test::PrintResult("ramp_up_down_up",
-                                  GetModifierString(),
-                                  "first_rampup",
-                                  now - state_start_ms_,
-                                  "ms",
-                                  false);
-        state_start_ms_ = now;
-        interval_start_ms_ = now;
-        sent_bytes_ = 0;
-      }
-      break;
-    }
-    case kLowRate: {
-      if (bitrate_bps < kExpectedLowBitrateBps && suspended_in_stats_) {
-        // The ramp-down was successful. Change the channel limit back to a
-        // high value, and move to the next test state.
-        forward_transport_config_.link_capacity_kbps =
-            kHighBandwidthLimitBps / 1000;
-        test::DirectTransport::SetConfig(forward_transport_config_);
-        test_state_ = kSecondRampup;
-        webrtc::test::PrintResult("ramp_up_down_up",
-                                  GetModifierString(),
-                                  "rampdown",
-                                  now - state_start_ms_,
-                                  "ms",
-                                  false);
-        state_start_ms_ = now;
-        interval_start_ms_ = now;
-        sent_bytes_ = 0;
-      }
-      break;
-    }
-    case kSecondRampup: {
-      if (bitrate_bps > kExpectedHighBitrateBps && !suspended_in_stats_) {
-        webrtc::test::PrintResult("ramp_up_down_up",
-                                  GetModifierString(),
-                                  "second_rampup",
-                                  now - state_start_ms_,
-                                  "ms",
-                                  false);
-        webrtc::test::PrintResult("ramp_up_down_up",
-                                  GetModifierString(),
-                                  "total_overuse",
-                                  total_overuse_bytes_,
-                                  "bytes",
-                                  false);
-        test_done_->Set();
-      }
-      break;
-    }
-  }
-}
-
-EventTypeWrapper LowRateStreamObserver::Wait() {
-  return test_done_->Wait(test::CallTest::kLongTimeoutMs);
-}
-
-void RampUpTest::RunRampUpTest(bool rtx,
-                               size_t num_streams,
-                               unsigned int start_bitrate_bps,
-                               const std::string& extension_type) {
-  std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100));
-  std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200));
-  StreamObserver::SsrcMap rtx_ssrc_map;
-  if (rtx) {
-    for (size_t i = 0; i < ssrcs.size(); ++i)
-      rtx_ssrc_map[rtx_ssrcs[i]] = ssrcs[i];
-  }
-
-  CreateSendConfig(num_streams);
-
-  rtc::scoped_ptr<RemoteBitrateEstimatorFactory> rbe_factory;
-  RateControlType control_type;
-  if (extension_type == RtpExtension::kAbsSendTime) {
-    control_type = kAimdControl;
-    rbe_factory.reset(new AbsoluteSendTimeRemoteBitrateEstimatorFactory);
-    send_config_.rtp.extensions.push_back(RtpExtension(
-        extension_type.c_str(), kAbsSendTimeExtensionId));
-  } else {
-    control_type = kMimdControl;
-    rbe_factory.reset(new RemoteBitrateEstimatorFactory);
-    send_config_.rtp.extensions.push_back(RtpExtension(
-        extension_type.c_str(), kTransmissionTimeOffsetExtensionId));
-  }
-
-  test::DirectTransport receiver_transport;
-  StreamObserver stream_observer(rtx_ssrc_map,
-                                 &receiver_transport,
-                                 Clock::GetRealTimeClock(),
-                                 rbe_factory.get(),
-                                 control_type);
-
-  Call::Config call_config(&stream_observer);
-  if (start_bitrate_bps != 0) {
-    call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps;
-    stream_observer.set_start_bitrate_bps(start_bitrate_bps);
-  }
-
-  CreateSenderCall(call_config);
-
-  receiver_transport.SetReceiver(sender_call_->Receiver());
-
-  if (num_streams == 1) {
-    encoder_config_.streams[0].target_bitrate_bps = 2000000;
-    encoder_config_.streams[0].max_bitrate_bps = 2000000;
-  }
-
-  send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-  send_config_.rtp.ssrcs = ssrcs;
-  if (rtx) {
-    send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
-    send_config_.rtp.rtx.ssrcs = rtx_ssrcs;
-  }
-
-  if (num_streams == 1) {
-    // For single stream rampup until 1mbps
-    stream_observer.set_expected_bitrate_bps(kSingleStreamTargetBps);
-  } else {
-    // For multi stream rampup until all streams are being sent. That means
-    // enough birate to send all the target streams plus the min bitrate of
-    // the last one.
-    int expected_bitrate_bps = encoder_config_.streams.back().min_bitrate_bps;
-    for (size_t i = 0; i < encoder_config_.streams.size() - 1; ++i) {
-      expected_bitrate_bps += encoder_config_.streams[i].target_bitrate_bps;
-    }
-    stream_observer.set_expected_bitrate_bps(expected_bitrate_bps);
-  }
-
-  CreateStreams();
-  CreateFrameGeneratorCapturer();
-
-  Start();
-
-  EXPECT_EQ(kEventSignaled, stream_observer.Wait());
-
-  Stop();
-  DestroyStreams();
-}
-
-void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
-  test::DirectTransport receiver_transport;
-  LowRateStreamObserver stream_observer(
-      &receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx);
-
-  Call::Config call_config(&stream_observer);
-  CreateSenderCall(call_config);
-  receiver_transport.SetReceiver(sender_call_->Receiver());
-
-  CreateSendConfig(number_of_streams);
-
-  send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-  send_config_.rtp.extensions.push_back(RtpExtension(
-      RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
-  send_config_.suspend_below_min_bitrate = true;
-  if (rtx) {
-    send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
-    send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200);
-  }
-
-  CreateStreams();
-  stream_observer.SetSendStream(send_stream_);
-
-  CreateFrameGeneratorCapturer();
-
-  Start();
-
-  EXPECT_EQ(kEventSignaled, stream_observer.Wait());
-
-  Stop();
-  DestroyStreams();
-}
-
-TEST_F(RampUpTest, SingleStream) {
-  RunRampUpTest(false, 1, 0, RtpExtension::kTOffset);
-}
-
-TEST_F(RampUpTest, Simulcast) {
-  RunRampUpTest(false, 3, 0, RtpExtension::kTOffset);
-}
-
-TEST_F(RampUpTest, SimulcastWithRtx) {
-  RunRampUpTest(true, 3, 0, RtpExtension::kTOffset);
-}
-
-TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
-  RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset);
-}
-
-TEST_F(RampUpTest, UpDownUpOneStream) { RunRampUpDownUpTest(1, false); }
-
-TEST_F(RampUpTest, UpDownUpThreeStreams) { RunRampUpDownUpTest(3, false); }
-
-TEST_F(RampUpTest, UpDownUpOneStreamRtx) { RunRampUpDownUpTest(1, true); }
-
-TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) { RunRampUpDownUpTest(3, true); }
-
-TEST_F(RampUpTest, AbsSendTimeSingleStream) {
-  RunRampUpTest(false, 1, 0, RtpExtension::kAbsSendTime);
-}
-
-TEST_F(RampUpTest, AbsSendTimeSimulcast) {
-  RunRampUpTest(false, 3, 0, RtpExtension::kAbsSendTime);
-}
-
-TEST_F(RampUpTest, AbsSendTimeSimulcastWithRtx) {
-  RunRampUpTest(true, 3, 0, RtpExtension::kAbsSendTime);
-}
-
-TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
-  RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps,
-                RtpExtension::kAbsSendTime);
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/rampup_tests.h b/media/webrtc/trunk/webrtc/video/rampup_tests.h
deleted file mode 100644
index 4335fc13a8..0000000000
--- a/media/webrtc/trunk/webrtc/video/rampup_tests.h
+++ /dev/null
@@ -1,158 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_RAMPUP_TESTS_H_
-#define WEBRTC_VIDEO_RAMPUP_TESTS_H_
-
-#include <map>
-#include <string>
-#include <vector>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/call.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/test/call_test.h"
-#include "webrtc/video/transport_adapter.h"
-
-namespace webrtc {
-
-static const int kTransmissionTimeOffsetExtensionId = 6;
-static const int kAbsSendTimeExtensionId = 7;
-static const unsigned int kSingleStreamTargetBps = 1000000;
-
-class Clock;
-class CriticalSectionWrapper;
-class ReceiveStatistics;
-class RtpHeaderParser;
-class RTPPayloadRegistry;
-class RtpRtcp;
-
-class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
- public:
-  typedef std::map<uint32_t, int> BytesSentMap;
-  typedef std::map<uint32_t, uint32_t> SsrcMap;
-  StreamObserver(const SsrcMap& rtx_media_ssrcs,
-                 newapi::Transport* feedback_transport,
-                 Clock* clock,
-                 RemoteBitrateEstimatorFactory* rbe_factory,
-                 RateControlType control_type);
-
-  void set_expected_bitrate_bps(unsigned int expected_bitrate_bps);
-
-  void set_start_bitrate_bps(unsigned int start_bitrate_bps);
-
-  void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
-                               unsigned int bitrate) override;
-
-  bool SendRtp(const uint8_t* packet, size_t length) override;
-
-  bool SendRtcp(const uint8_t* packet, size_t length) override;
-
-  EventTypeWrapper Wait();
-
- private:
-  void ReportResult(const std::string& measurement,
-                    size_t value,
-                    const std::string& units);
-  void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-
-  Clock* const clock_;
-  const rtc::scoped_ptr<EventWrapper> test_done_;
-  const rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
-  rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
-  internal::TransportAdapter feedback_transport_;
-  const rtc::scoped_ptr<ReceiveStatistics> receive_stats_;
-  const rtc::scoped_ptr<RTPPayloadRegistry> payload_registry_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
-
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-  unsigned int expected_bitrate_bps_ GUARDED_BY(crit_);
-  unsigned int start_bitrate_bps_ GUARDED_BY(crit_);
-  SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_);
-  size_t total_sent_ GUARDED_BY(crit_);
-  size_t padding_sent_ GUARDED_BY(crit_);
-  size_t rtx_media_sent_ GUARDED_BY(crit_);
-  int total_packets_sent_ GUARDED_BY(crit_);
-  int padding_packets_sent_ GUARDED_BY(crit_);
-  int rtx_media_packets_sent_ GUARDED_BY(crit_);
-  int64_t test_start_ms_ GUARDED_BY(crit_);
-  int64_t ramp_up_finished_ms_ GUARDED_BY(crit_);
-};
-
-class LowRateStreamObserver : public test::DirectTransport,
-                              public RemoteBitrateObserver,
-                              public PacketReceiver {
- public:
-  LowRateStreamObserver(newapi::Transport* feedback_transport,
-                        Clock* clock,
-                        size_t number_of_streams,
-                        bool rtx_used);
-
-  virtual void SetSendStream(VideoSendStream* send_stream);
-
-  virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
-                                       unsigned int bitrate);
-
-  bool SendRtp(const uint8_t* data, size_t length) override;
-
-  DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) override;
-
-  bool SendRtcp(const uint8_t* packet, size_t length) override;
-
-  // Produces a string similar to "1stream_nortx", depending on the values of
-  // number_of_streams_ and rtx_used_;
-  std::string GetModifierString();
-
-  // This method defines the state machine for the ramp up-down-up test.
-  void EvolveTestState(unsigned int bitrate_bps);
-
-  EventTypeWrapper Wait();
-
- private:
-  static const unsigned int kHighBandwidthLimitBps = 80000;
-  static const unsigned int kExpectedHighBitrateBps = 60000;
-  static const unsigned int kLowBandwidthLimitBps = 20000;
-  static const unsigned int kExpectedLowBitrateBps = 20000;
-  enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
-
-  Clock* const clock_;
-  const size_t number_of_streams_;
-  const bool rtx_used_;
-  const rtc::scoped_ptr<EventWrapper> test_done_;
-  const rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
-  rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
-  internal::TransportAdapter feedback_transport_;
-  const rtc::scoped_ptr<ReceiveStatistics> receive_stats_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
-
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-  VideoSendStream* send_stream_ GUARDED_BY(crit_);
-  FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_);
-  TestStates test_state_ GUARDED_BY(crit_);
-  int64_t state_start_ms_ GUARDED_BY(crit_);
-  int64_t interval_start_ms_ GUARDED_BY(crit_);
-  unsigned int last_remb_bps_ GUARDED_BY(crit_);
-  size_t sent_bytes_ GUARDED_BY(crit_);
-  size_t total_overuse_bytes_ GUARDED_BY(crit_);
-  bool suspended_in_stats_ GUARDED_BY(crit_);
-};
-
-class RampUpTest : public test::CallTest {
- protected:
-  void RunRampUpTest(bool rtx,
-                     size_t num_streams,
-                     unsigned int start_bitrate_bps,
-                     const std::string& extension_type);
-
-  void RunRampUpDownUpTest(size_t number_of_streams, bool rtx);
-};
-}  // namespace webrtc
-#endif  // WEBRTC_VIDEO_RAMPUP_TESTS_H_
diff --git a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc
index b7a38535a4..e3298a1e51 100644
--- a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc
+++ b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc
@@ -10,50 +10,106 @@
 
 #include "webrtc/video/receive_statistics_proxy.h"
 
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include <cmath>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
 ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
     : clock_(clock),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
       renders_fps_estimator_(1000, 1000),
-      receive_state_(kReceiveStateInitial) {
+      render_fps_tracker_(100u, 10u),
+      render_pixel_tracker_(100u, 10u),
+      receive_state_(kReceiveStateInitial) {      
   stats_.ssrc = ssrc;
 }
 
-ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {}
+ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
+  UpdateHistograms();
+}
+
+void ReceiveStatisticsProxy::UpdateHistograms() {
+  int fraction_lost = report_block_stats_.FractionLostInPercent();
+  if (fraction_lost != -1) {
+    RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.ReceivedPacketsLostInPercent",
+                                    fraction_lost);
+  }
+  const int kMinRequiredSamples = 200;
+  int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
+  if (samples > kMinRequiredSamples) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100("WebRTC.Video.RenderFramesPerSecond",
+        round(render_fps_tracker_.ComputeTotalRate()));
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Video.RenderSqrtPixelsPerSecond",
+        round(render_pixel_tracker_.ComputeTotalRate()));
+  }
+  int width = render_width_counter_.Avg(kMinRequiredSamples);
+  int height = render_height_counter_.Avg(kMinRequiredSamples);
+  if (width != -1) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.ReceivedWidthInPixels",
+                                      width);
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.ReceivedHeightInPixels",
+                                      height);
+  }
+  int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
+  if (qp != -1)
+    RTC_HISTOGRAM_COUNTS_SPARSE_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
+
+  // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
+  // not per frame. Change decode time to include every frame.
+  const int kMinRequiredDecodeSamples = 5;
+  int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
+  if (decode_ms != -1)
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
+
+  int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (delay_ms != -1)
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
+}
 
 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   return stats_;
 }
 
-void ReceiveStatisticsProxy::IncomingRate(const int video_channel,
-                                          const unsigned int framerate,
-                                          const unsigned int bitrate_bps) {
-  CriticalSectionScoped lock(crit_.get());
+void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
+  rtc::CritScope lock(&crit_);
+  stats_.current_payload_type = payload_type;
+}
+
+void ReceiveStatisticsProxy::OnDecoderImplementationName(
+    const char* implementation_name) {
+  rtc::CritScope lock(&crit_);
+  stats_.decoder_implementation_name = implementation_name;
+}
+
+void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
+                                            unsigned int bitrate_bps) {
+  rtc::CritScope lock(&crit_);
   stats_.network_frame_rate = framerate;
   stats_.total_bitrate_bps = bitrate_bps;
 }
 
-void ReceiveStatisticsProxy::ReceiveStateChange(const int video_channel,
-                                                VideoReceiveState state) {
-  CriticalSectionScoped cs(lock_.get());
+void ReceiveStatisticsProxy::ReceiveStateChange(VideoReceiveState state) {
+  rtc::CritScope lock(&crit_);
   receive_state_ = state;
 }
 
-void ReceiveStatisticsProxy::DecoderTiming(int decode_ms,
-                                           int max_decode_ms,
-                                           int current_delay_ms,
-                                           int target_delay_ms,
-                                           int jitter_buffer_ms,
-                                           int min_playout_delay_ms,
-                                           int render_delay_ms) {
-  CriticalSectionScoped lock(crit_.get());
+void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
+                                             int max_decode_ms,
+                                             int current_delay_ms,
+                                             int target_delay_ms,
+                                             int jitter_buffer_ms,
+                                             int min_playout_delay_ms,
+                                             int render_delay_ms,
+                                             int64_t rtt_ms) {
+  rtc::CritScope lock(&crit_);
   stats_.decode_ms = decode_ms;
   stats_.max_decode_ms = max_decode_ms;
   stats_.current_delay_ms = current_delay_ms;
@@ -61,12 +117,16 @@ void ReceiveStatisticsProxy::DecoderTiming(int decode_ms,
   stats_.jitter_buffer_ms = jitter_buffer_ms;
   stats_.min_playout_delay_ms = min_playout_delay_ms;
   stats_.render_delay_ms = render_delay_ms;
+  decode_time_counter_.Add(decode_ms);
+  // Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
+  // render delay).
+  delay_counter_.Add(target_delay_ms + rtt_ms / 2);
 }
 
 void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
     uint32_t ssrc,
     const RtcpPacketTypeCounter& packet_counter) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   if (stats_.ssrc != ssrc)
     return;
   stats_.rtcp_packet_type_counts = packet_counter;
@@ -75,17 +135,18 @@ void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
 void ReceiveStatisticsProxy::StatisticsUpdated(
     const webrtc::RtcpStatistics& statistics,
     uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
-  // TODO(pbos): Handle both local and remote ssrcs here and DCHECK that we
+  rtc::CritScope lock(&crit_);
+  // TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
   // receive stats from one of them.
   if (stats_.ssrc != ssrc)
     return;
   stats_.rtcp_stats = statistics;
+  report_block_stats_.Store(statistics, ssrc, 0);
 }
 
 void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
-  // TODO(pbos): Handle both local and remote ssrcs here and DCHECK that we
+  rtc::CritScope lock(&crit_);
+  // TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
   // receive stats from one of them.
   if (stats_.ssrc != ssrc)
     return;
@@ -95,7 +156,7 @@ void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
 void ReceiveStatisticsProxy::DataCountersUpdated(
     const webrtc::StreamDataCounters& counters,
     uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   if (stats_.ssrc != ssrc)
     return;
   stats_.rtp_stats = counters;
@@ -104,17 +165,23 @@ void ReceiveStatisticsProxy::DataCountersUpdated(
 void ReceiveStatisticsProxy::OnDecodedFrame() {
   uint64_t now = clock_->TimeInMilliseconds();
 
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   decode_fps_estimator_.Update(1, now);
   stats_.decode_frame_rate = decode_fps_estimator_.Rate(now);
 }
 
-void ReceiveStatisticsProxy::OnRenderedFrame() {
+void ReceiveStatisticsProxy::OnRenderedFrame(int width, int height) {
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
   uint64_t now = clock_->TimeInMilliseconds();
 
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   renders_fps_estimator_.Update(1, now);
   stats_.render_frame_rate = renders_fps_estimator_.Rate(now);
+  render_width_counter_.Add(width);
+  render_height_counter_.Add(height);
+  render_fps_tracker_.AddSamples(1);
+  render_pixel_tracker_.AddSamples(sqrt(width * height));
 }
 
 void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
@@ -123,13 +190,35 @@ void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
 
 void ReceiveStatisticsProxy::OnFrameCountsUpdated(
     const FrameCounts& frame_counts) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   stats_.frame_counts = frame_counts;
 }
 
 void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   stats_.discarded_packets = discarded_packets;
 }
 
+void ReceiveStatisticsProxy::OnPreDecode(
+    const EncodedImage& encoded_image,
+    const CodecSpecificInfo* codec_specific_info) {
+  if (codec_specific_info == nullptr || encoded_image.qp_ == -1) {
+    return;
+  }
+  if (codec_specific_info->codecType == kVideoCodecVP8) {
+    qp_counters_.vp8.Add(encoded_image.qp_);
+  }
+}
+
+void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
+  sum += sample;
+  ++num_samples;
+}
+
+int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
+  if (num_samples < min_required_samples || num_samples == 0)
+    return -1;
+  return sum / num_samples;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h
index aaecb02a38..f1a9cbd54f 100644
--- a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h
+++ b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h
@@ -13,25 +13,26 @@
 
 #include <string>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/ratetracker.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/video/report_block_stats.h"
+#include "webrtc/video/vie_channel.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_renderer.h"
 
 namespace webrtc {
 
 class Clock;
-class CriticalSectionWrapper;
 class ViECodec;
 class ViEDecoderObserver;
+struct CodecSpecificInfo;
 
-class ReceiveStatisticsProxy : public ViEDecoderObserver,
-                               public VCMReceiveStatisticsCallback,
+class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
                                public RtcpStatisticsCallback,
                                public RtcpPacketTypeCounterObserver,
                                public StreamDataCountersCallback {
@@ -42,35 +43,34 @@ class ReceiveStatisticsProxy : public ViEDecoderObserver,
   VideoReceiveStream::Stats GetStats() const;
 
   void OnDecodedFrame();
-  void OnRenderedFrame();
+  void OnRenderedFrame(int width, int height);
+  void OnIncomingPayloadType(int payload_type);
+  void OnDecoderImplementationName(const char* implementation_name);
+  void OnIncomingRate(unsigned int framerate, unsigned int bitrate_bps);
+  void OnDecoderTiming(int decode_ms,
+                       int max_decode_ms,
+                       int current_delay_ms,
+                       int target_delay_ms,
+                       int jitter_buffer_ms,
+                       int min_playout_delay_ms,
+                       int render_delay_ms,
+                       int64_t rtt_ms);
+  void ReceiveStateChange(VideoReceiveState state);
 
-  // Overrides VCMReceiveStatisticsCallback
+  void OnPreDecode(const EncodedImage& encoded_image,
+                   const CodecSpecificInfo* codec_specific_info);
+
+  // Overrides VCMReceiveStatisticsCallback.
   void OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) override;
   void OnFrameCountsUpdated(const FrameCounts& frame_counts) override;
   void OnDiscardedPacketsUpdated(int discarded_packets) override;
 
-  // Overrides ViEDecoderObserver.
-  void IncomingCodecChanged(const int video_channel,
-                            const VideoCodec& video_codec) override {}
-  void IncomingRate(const int video_channel,
-                    const unsigned int framerate,
-                    const unsigned int bitrate_bps) override;
-  void DecoderTiming(int decode_ms,
-                     int max_decode_ms,
-                     int current_delay_ms,
-                     int target_delay_ms,
-                     int jitter_buffer_ms,
-                     int min_playout_delay_ms,
-                     int render_delay_ms) override;
-  void RequestNewKeyFrame(const int video_channel) override {}
-  virtual void ReceiveStateChange(const int video_channel, VideoReceiveState state) override;
-
   // Overrides RtcpStatisticsCallback.
   void StatisticsUpdated(const webrtc::RtcpStatistics& statistics,
                          uint32_t ssrc) override;
   void CNameChanged(const char* cname, uint32_t ssrc) override;
 
-  // Overrides RtcpPacketTypeCounterObserver
+  // Overrides RtcpPacketTypeCounterObserver.
   void RtcpPacketTypesCounterUpdated(
       uint32_t ssrc,
       const RtcpPacketTypeCounter& packet_counter) override;
@@ -79,12 +79,35 @@ class ReceiveStatisticsProxy : public ViEDecoderObserver,
                            uint32_t ssrc) override;
 
  private:
+  struct SampleCounter {
+    SampleCounter() : sum(0), num_samples(0) {}
+    void Add(int sample);
+    int Avg(int min_required_samples) const;
+
+   private:
+    int sum;
+    int num_samples;
+  };
+  struct QpCounters {
+    SampleCounter vp8;
+  };
+
+  void UpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
   Clock* const clock_;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
   VideoReceiveStream::Stats stats_ GUARDED_BY(crit_);
   RateStatistics decode_fps_estimator_ GUARDED_BY(crit_);
   RateStatistics renders_fps_estimator_ GUARDED_BY(crit_);
+  rtc::RateTracker render_fps_tracker_ GUARDED_BY(crit_);
+  rtc::RateTracker render_pixel_tracker_ GUARDED_BY(crit_);
+  SampleCounter render_width_counter_ GUARDED_BY(crit_);
+  SampleCounter render_height_counter_ GUARDED_BY(crit_);
+  SampleCounter decode_time_counter_ GUARDED_BY(crit_);
+  SampleCounter delay_counter_ GUARDED_BY(crit_);
+  ReportBlockStats report_block_stats_ GUARDED_BY(crit_);
+  QpCounters qp_counters_;  // Only accessed on the decoding thread.
   VideoReceiveState receive_state_ GUARDED_BY(crit_);
 };
 
diff --git a/media/webrtc/trunk/webrtc/video/replay.cc b/media/webrtc/trunk/webrtc/video/replay.cc
index 740d149be8..484924872b 100644
--- a/media/webrtc/trunk/webrtc/video/replay.cc
+++ b/media/webrtc/trunk/webrtc/video/replay.cc
@@ -20,9 +20,9 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/fake_decoder.h"
@@ -106,7 +106,7 @@ static const bool timestamp_offset_dummy =
 // Flag for rtpdump input file.
 bool ValidateInputFilenameNotEmpty(const char* flagname,
                                    const std::string& string) {
-  return string != "";
+  return !string.empty();
 }
 
 DEFINE_string(input_file, "", "input file");
@@ -152,11 +152,11 @@ class FileRenderPassthrough : public VideoRenderer {
   }
 
  private:
-  void RenderFrame(const I420VideoFrame& video_frame,
+  void RenderFrame(const VideoFrame& video_frame,
                    int time_to_render_ms) override {
     if (renderer_ != nullptr)
       renderer_->RenderFrame(video_frame, time_to_render_ms);
-    if (basename_ == "")
+    if (basename_.empty())
       return;
     if (last_width_ != video_frame.width() ||
         last_height_ != video_frame.height()) {
@@ -179,7 +179,7 @@ class FileRenderPassthrough : public VideoRenderer {
     last_height_ = video_frame.height();
     if (file_ == nullptr)
       return;
-    PrintI420VideoFrame(video_frame, file_);
+    PrintVideoFrame(video_frame, file_);
   }
 
   bool IsTextureSupported() const override { return false; }
@@ -196,7 +196,7 @@ class DecoderBitstreamFileWriter : public EncodedFrameObserver {
  public:
   explicit DecoderBitstreamFileWriter(const char* filename)
       : file_(fopen(filename, "wb")) {
-    DCHECK(file_ != nullptr);
+    RTC_DCHECK(file_ != nullptr);
   }
   ~DecoderBitstreamFileWriter() { fclose(file_); }
 
@@ -214,13 +214,10 @@ void RtpReplay() {
   FileRenderPassthrough file_passthrough(flags::OutBase(),
                                          playback_video.get());
 
-  // TODO(pbos): Might be good to have a transport that prints keyframe requests
-  //             etc.
-  test::NullTransport transport;
-  Call::Config call_config(&transport);
-  rtc::scoped_ptr<Call> call(Call::Create(call_config));
+  rtc::scoped_ptr<Call> call(Call::Create(Call::Config()));
 
-  VideoReceiveStream::Config receive_config;
+  test::NullTransport transport;
+  VideoReceiveStream::Config receive_config(&transport);
   receive_config.rtp.remote_ssrc = flags::Ssrc();
   receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
   receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType();
@@ -241,13 +238,13 @@ void RtpReplay() {
   encoder_settings.payload_type = flags::PayloadType();
   VideoReceiveStream::Decoder decoder;
   rtc::scoped_ptr<DecoderBitstreamFileWriter> bitstream_writer;
-  if (flags::DecoderBitstreamFilename() != "") {
+  if (!flags::DecoderBitstreamFilename().empty()) {
     bitstream_writer.reset(new DecoderBitstreamFileWriter(
         flags::DecoderBitstreamFilename().c_str()));
     receive_config.pre_decode_callback = bitstream_writer.get();
   }
   decoder = test::CreateMatchingDecoder(encoder_settings);
-  if (flags::DecoderBitstreamFilename() != "") {
+  if (!flags::DecoderBitstreamFilename().empty()) {
     // Replace with a null decoder if we're writing the bitstream to a file
     // instead.
     delete decoder.decoder;
@@ -287,7 +284,8 @@ void RtpReplay() {
     if (!rtp_reader->NextPacket(&packet))
       break;
     ++num_packets;
-    switch (call->Receiver()->DeliverPacket(packet.data, packet.length)) {
+    switch (call->Receiver()->DeliverPacket(webrtc::MediaType::ANY, packet.data,
+                                            packet.length, PacketTime())) {
       case PacketReceiver::DELIVERY_OK:
         break;
       case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
diff --git a/media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc b/media/webrtc/trunk/webrtc/video/report_block_stats.cc
similarity index 97%
rename from media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc
rename to media/webrtc/trunk/webrtc/video/report_block_stats.cc
index ea18e830e3..dee5662c3c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc
+++ b/media/webrtc/trunk/webrtc/video/report_block_stats.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/report_block_stats.h"
+#include "webrtc/video/report_block_stats.h"
 
 namespace webrtc {
 
@@ -70,8 +70,8 @@ RTCPReportBlock ReportBlockStats::AggregateAndStore(
   // Fraction lost since previous report block.
   aggregate.fractionLost =
       FractionLost(num_lost_sequence_numbers, num_sequence_numbers);
-  aggregate.jitter =
-      (aggregate.jitter + report_blocks.size() / 2) / report_blocks.size();
+  aggregate.jitter = static_cast<uint32_t>(
+      (aggregate.jitter + report_blocks.size() / 2) / report_blocks.size());
   return aggregate;
 }
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/report_block_stats.h b/media/webrtc/trunk/webrtc/video/report_block_stats.h
similarity index 89%
rename from media/webrtc/trunk/webrtc/video_engine/report_block_stats.h
rename to media/webrtc/trunk/webrtc/video/report_block_stats.h
index dadcc9d410..c54e4677f4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/report_block_stats.h
+++ b/media/webrtc/trunk/webrtc/video/report_block_stats.h
@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
-#define WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
+#ifndef WEBRTC_VIDEO_REPORT_BLOCK_STATS_H_
+#define WEBRTC_VIDEO_REPORT_BLOCK_STATS_H_
 
 #include <map>
 #include <vector>
 
 #include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -58,5 +58,5 @@ class ReportBlockStats {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
+#endif  // WEBRTC_VIDEO_REPORT_BLOCK_STATS_H_
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc b/media/webrtc/trunk/webrtc/video/report_block_stats_unittest.cc
similarity index 98%
rename from media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc
rename to media/webrtc/trunk/webrtc/video/report_block_stats_unittest.cc
index 13b7af5ba2..5cde9004b1 100644
--- a/media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/report_block_stats_unittest.cc
@@ -10,7 +10,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/video_engine/report_block_stats.h"
+#include "webrtc/video/report_block_stats.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc b/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc
index 66bdf6ebc5..6479aa4ebb 100644
--- a/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc
+++ b/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc
@@ -10,29 +10,25 @@
 
 #include <stdio.h>
 
-#include <map>
-
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/test/field_trial.h"
-#include "webrtc/test/frame_generator.h"
-#include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/run_test.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video/loopback.h"
-#include "webrtc/video/video_send_stream.h"
+#include "webrtc/video/video_quality_test.h"
 
 namespace webrtc {
 namespace flags {
 
-// Fixed for prerecorded screenshare content.
+// Flags common with video loopback, with different default values.
+DEFINE_int32(width, 1850, "Video width (crops source).");
 size_t Width() {
-  return 1850;
+  return static_cast<size_t>(FLAGS_width);
 }
+
+DEFINE_int32(height, 1110, "Video height (crops source).");
 size_t Height() {
-  return 1110;
+  return static_cast<size_t>(FLAGS_height);
 }
 
 DEFINE_int32(fps, 5, "Frames per second.");
@@ -40,31 +36,64 @@ int Fps() {
   return static_cast<int>(FLAGS_fps);
 }
 
-DEFINE_int32(min_bitrate, 50, "Minimum video bitrate.");
-size_t MinBitrate() {
-  return static_cast<size_t>(FLAGS_min_bitrate);
+DEFINE_int32(min_bitrate, 50, "Call and stream min bitrate in kbps.");
+int MinBitrateKbps() {
+  return static_cast<int>(FLAGS_min_bitrate);
 }
 
-DEFINE_int32(tl0_bitrate, 100, "Temporal layer 0 target bitrate.");
-size_t StartBitrate() {
-  return static_cast<size_t>(FLAGS_tl0_bitrate);
+DEFINE_int32(start_bitrate, 200, "Call start bitrate in kbps.");
+int StartBitrateKbps() {
+  return static_cast<int>(FLAGS_start_bitrate);
 }
 
-DEFINE_int32(tl1_bitrate, 1000, "Temporal layer 1 target bitrate.");
-size_t MaxBitrate() {
-  return static_cast<size_t>(FLAGS_tl1_bitrate);
+DEFINE_int32(target_bitrate, 2000, "Stream target bitrate in kbps.");
+int TargetBitrateKbps() {
+  return static_cast<int>(FLAGS_target_bitrate);
 }
 
-DEFINE_int32(min_transmit_bitrate, 400, "Min transmit bitrate incl. padding.");
-int MinTransmitBitrate() {
-  return FLAGS_min_transmit_bitrate;
+DEFINE_int32(max_bitrate, 2000, "Call and stream max bitrate in kbps.");
+int MaxBitrateKbps() {
+  return static_cast<int>(FLAGS_max_bitrate);
 }
 
+DEFINE_int32(num_temporal_layers, 2, "Number of temporal layers to use.");
+int NumTemporalLayers() {
+  return static_cast<int>(FLAGS_num_temporal_layers);
+}
+
+// Flags common with video loopback, with equal default values.
 DEFINE_string(codec, "VP8", "Video codec to use.");
 std::string Codec() {
   return static_cast<std::string>(FLAGS_codec);
 }
 
+DEFINE_int32(selected_tl,
+             -1,
+             "Temporal layer to show or analyze. -1 to disable filtering.");
+int SelectedTL() {
+  return static_cast<int>(FLAGS_selected_tl);
+}
+
+DEFINE_int32(
+    duration,
+    0,
+    "Duration of the test in seconds. If 0, rendered will be shown instead.");
+int DurationSecs() {
+  return static_cast<int>(FLAGS_duration);
+}
+
+DEFINE_string(output_filename, "", "Target graph data filename.");
+std::string OutputFilename() {
+  return static_cast<std::string>(FLAGS_output_filename);
+}
+
+DEFINE_string(graph_title,
+              "",
+              "If empty, title will be generated automatically.");
+std::string GraphTitle() {
+  return static_cast<std::string>(FLAGS_graph_title);
+}
+
 DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
 int LossPercent() {
   return static_cast<int>(FLAGS_loss_percent);
@@ -73,7 +102,7 @@ int LossPercent() {
 DEFINE_int32(link_capacity,
              0,
              "Capacity (kbps) of the fake link. 0 means infinite.");
-int LinkCapacity() {
+int LinkCapacityKbps() {
   return static_cast<int>(FLAGS_link_capacity);
 }
 
@@ -96,8 +125,55 @@ int StdPropagationDelayMs() {
   return static_cast<int>(FLAGS_std_propagation_delay_ms);
 }
 
+DEFINE_int32(selected_stream, 0, "ID of the stream to show or analyze.");
+int SelectedStream() {
+  return static_cast<int>(FLAGS_selected_stream);
+}
+
+DEFINE_int32(num_spatial_layers, 1, "Number of spatial layers to use.");
+int NumSpatialLayers() {
+  return static_cast<int>(FLAGS_num_spatial_layers);
+}
+
+DEFINE_int32(selected_sl,
+             -1,
+             "Spatial layer to show or analyze. -1 to disable filtering.");
+int SelectedSL() {
+  return static_cast<int>(FLAGS_selected_sl);
+}
+
+DEFINE_string(stream0,
+              "",
+              "Comma separated values describing VideoStream for stream #0.");
+std::string Stream0() {
+  return static_cast<std::string>(FLAGS_stream0);
+}
+
+DEFINE_string(stream1,
+              "",
+              "Comma separated values describing VideoStream for stream #1.");
+std::string Stream1() {
+  return static_cast<std::string>(FLAGS_stream1);
+}
+
+DEFINE_string(sl0,
+              "",
+              "Comma separated values describing SpatialLayer for layer #0.");
+std::string SL0() {
+  return static_cast<std::string>(FLAGS_sl0);
+}
+
+DEFINE_string(sl1,
+              "",
+              "Comma separated values describing SpatialLayer for layer #1.");
+std::string SL1() {
+  return static_cast<std::string>(FLAGS_sl1);
+}
+
 DEFINE_bool(logs, false, "print logs to stderr");
 
+DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
+
 DEFINE_string(
     force_fieldtrials,
     "",
@@ -105,65 +181,73 @@ DEFINE_string(
     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
     " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
     "trials are separated by \"/\"");
+
+// Screenshare-specific flags.
+DEFINE_int32(min_transmit_bitrate, 400, "Min transmit bitrate incl. padding.");
+int MinTransmitBitrateKbps() {
+  return FLAGS_min_transmit_bitrate;
+}
+
+DEFINE_int32(slide_change_interval,
+             10,
+             "Interval (in seconds) between simulated slide changes.");
+int SlideChangeInterval() {
+  return static_cast<int>(FLAGS_slide_change_interval);
+}
+
+DEFINE_int32(
+    scroll_duration,
+    0,
+    "Duration (in seconds) during which a slide will be scrolled into place.");
+int ScrollDuration() {
+  return static_cast<int>(FLAGS_scroll_duration);
+}
+
 }  // namespace flags
 
-class ScreenshareLoopback : public test::Loopback {
- public:
-  explicit ScreenshareLoopback(const Config& config) : Loopback(config) {}
-  virtual ~ScreenshareLoopback() {}
-
- protected:
-  VideoEncoderConfig CreateEncoderConfig() override {
-    VideoEncoderConfig encoder_config(test::Loopback::CreateEncoderConfig());
-    VideoStream* stream = &encoder_config.streams[0];
-    encoder_config.content_type = VideoEncoderConfig::kScreenshare;
-    encoder_config.min_transmit_bitrate_bps = flags::MinTransmitBitrate();
-    VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
-    vp8_settings.denoisingOn = false;
-    vp8_settings.frameDroppingOn = false;
-    vp8_settings.numberOfTemporalLayers = 2;
-    encoder_config.encoder_specific_settings = &vp8_settings;
-    stream->temporal_layer_thresholds_bps.clear();
-    stream->target_bitrate_bps =
-        static_cast<int>(config_.start_bitrate_kbps) * 1000;
-    stream->temporal_layer_thresholds_bps.push_back(stream->target_bitrate_bps);
-    return encoder_config;
-  }
-
-  test::VideoCapturer* CreateCapturer(VideoSendStream* send_stream) override {
-    std::vector<std::string> slides;
-    slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
-    slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
-    slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
-    slides.push_back(test::ResourcePath("difficult_photo_1850_1110", "yuv"));
-
-    test::FrameGenerator* frame_generator =
-        test::FrameGenerator::CreateFromYuvFile(
-            slides, flags::Width(), flags::Height(), 10 * flags::Fps());
-    test::FrameGeneratorCapturer* capturer(new test::FrameGeneratorCapturer(
-        clock_, send_stream->Input(), frame_generator, flags::Fps()));
-    EXPECT_TRUE(capturer->Init());
-    return capturer;
-  }
-};
-
 void Loopback() {
-  test::Loopback::Config config{flags::Width(),
-                                flags::Height(),
-                                flags::Fps(),
-                                flags::MinBitrate(),
-                                flags::StartBitrate(),
-                                flags::MaxBitrate(),
-                                flags::MinTransmitBitrate(),
-                                flags::Codec(),
-                                flags::LossPercent(),
-                                flags::LinkCapacity(),
-                                flags::QueueSize(),
-                                flags::AvgPropagationDelayMs(),
-                                flags::StdPropagationDelayMs(),
-                                flags::FLAGS_logs};
-  ScreenshareLoopback loopback(config);
-  loopback.Run();
+  FakeNetworkPipe::Config pipe_config;
+  pipe_config.loss_percent = flags::LossPercent();
+  pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
+  pipe_config.queue_length_packets = flags::QueueSize();
+  pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
+  pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
+
+  Call::Config::BitrateConfig call_bitrate_config;
+  call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
+  call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
+  call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
+
+  VideoQualityTest::Params params{
+      {flags::Width(), flags::Height(), flags::Fps(),
+       flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
+       flags::MaxBitrateKbps() * 1000, flags::Codec(),
+       flags::NumTemporalLayers(), flags::SelectedTL(),
+       flags::MinTransmitBitrateKbps() * 1000, call_bitrate_config,
+       flags::FLAGS_send_side_bwe},
+      {},  // Video specific.
+      {true, flags::SlideChangeInterval(), flags::ScrollDuration()},
+      {"screenshare", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
+       flags::GraphTitle()},
+      pipe_config,
+      flags::FLAGS_logs};
+
+  std::vector<std::string> stream_descriptors;
+  stream_descriptors.push_back(flags::Stream0());
+  stream_descriptors.push_back(flags::Stream1());
+  std::vector<std::string> SL_descriptors;
+  SL_descriptors.push_back(flags::SL0());
+  SL_descriptors.push_back(flags::SL1());
+  VideoQualityTest::FillScalabilitySettings(
+      &params, stream_descriptors, flags::SelectedStream(),
+      flags::NumSpatialLayers(), flags::SelectedSL(), SL_descriptors);
+
+  VideoQualityTest test;
+  if (flags::DurationSecs()) {
+    test.RunWithAnalyzer(params);
+  } else {
+    test.RunWithVideoRenderer(params);
+  }
 }
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc
index 41a8a99727..d2964b21da 100644
--- a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc
+++ b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc
@@ -10,66 +10,204 @@
 
 #include "webrtc/video/send_statistics_proxy.h"
 
+#include <algorithm>
+#include <cmath>
 #include <map>
 
 #include "webrtc/base/checks.h"
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
+namespace {
+const float kEncodeTimeWeigthFactor = 0.5f;
+
+// Used by histograms. Values of entries should not be changed.
+enum HistogramCodecType {
+  kVideoUnknown = 0,
+  kVideoVp8 = 1,
+  kVideoVp9 = 2,
+  kVideoH264 = 3,
+  kVideoMax = 64,
+};
+
+const char* GetUmaPrefix(VideoEncoderConfig::ContentType content_type) {
+  switch (content_type) {
+    case VideoEncoderConfig::ContentType::kRealtimeVideo:
+      return "WebRTC.Video.";
+    case VideoEncoderConfig::ContentType::kScreen:
+      return "WebRTC.Video.Screenshare.";
+  }
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+HistogramCodecType PayloadNameToHistogramCodecType(
+    const std::string& payload_name) {
+  if (payload_name == "VP8") {
+    return kVideoVp8;
+  } else if (payload_name == "VP9") {
+    return kVideoVp9;
+  } else if (payload_name == "H264") {
+    return kVideoH264;
+  } else {
+    return kVideoUnknown;
+  }
+}
+
+void UpdateCodecTypeHistogram(const std::string& payload_name) {
+  RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Video.Encoder.CodecType",
+      PayloadNameToHistogramCodecType(payload_name), kVideoMax);
+}
+}  // namespace
+
 
 const int SendStatisticsProxy::kStatsTimeoutMs = 5000;
 
-SendStatisticsProxy::SendStatisticsProxy(Clock* clock,
-                                         const VideoSendStream::Config& config)
+SendStatisticsProxy::SendStatisticsProxy(
+    Clock* clock,
+    const VideoSendStream::Config& config,
+    VideoEncoderConfig::ContentType content_type)
     : clock_(clock),
       config_(config),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()) {
+      content_type_(content_type),
+      last_sent_frame_timestamp_(0),
+      encode_time_(kEncodeTimeWeigthFactor),
+      uma_container_(new UmaSamplesContainer(GetUmaPrefix(content_type_))) {
+  UpdateCodecTypeHistogram(config_.encoder_settings.payload_name);
 }
 
 SendStatisticsProxy::~SendStatisticsProxy() {}
 
-void SendStatisticsProxy::OutgoingRate(const int video_channel,
-                                       const unsigned int framerate,
-                                       const unsigned int bitrate) {
-  CriticalSectionScoped lock(crit_.get());
+SendStatisticsProxy::UmaSamplesContainer::UmaSamplesContainer(
+    const char* prefix)
+    : uma_prefix_(prefix),
+      max_sent_width_per_timestamp_(0),
+      max_sent_height_per_timestamp_(0),
+      input_frame_rate_tracker_(100u, 10u),
+      sent_frame_rate_tracker_(100u, 10u) {}
+
+SendStatisticsProxy::UmaSamplesContainer::~UmaSamplesContainer() {
+  UpdateHistograms();
+}
+
+void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms() {
+  const int kMinRequiredSamples = 200;
+  int in_width = input_width_counter_.Avg(kMinRequiredSamples);
+  int in_height = input_height_counter_.Avg(kMinRequiredSamples);
+  int in_fps = round(input_frame_rate_tracker_.ComputeTotalRate());
+  if (in_width != -1) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "InputWidthInPixels",
+                                      in_width);
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "InputHeightInPixels",
+                                      in_height);
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(uma_prefix_ + "InputFramesPerSecond",
+                                    in_fps);
+  }
+  int sent_width = sent_width_counter_.Avg(kMinRequiredSamples);
+  int sent_height = sent_height_counter_.Avg(kMinRequiredSamples);
+  int sent_fps = round(sent_frame_rate_tracker_.ComputeTotalRate());
+  if (sent_width != -1) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "SentWidthInPixels",
+                                      sent_width);
+    RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "SentHeightInPixels",
+                                      sent_height);
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(uma_prefix_ + "SentFramesPerSecond",
+                                    sent_fps);
+  }
+  int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
+  if (encode_ms != -1)
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(uma_prefix_ + "EncodeTimeInMs", encode_ms);
+
+  int key_frames_permille = key_frame_counter_.Permille(kMinRequiredSamples);
+  if (key_frames_permille != -1) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(uma_prefix_ + "KeyFramesSentInPermille",
+                                     key_frames_permille);
+  }
+  int quality_limited =
+      quality_limited_frame_counter_.Percent(kMinRequiredSamples);
+  if (quality_limited != -1) {
+    RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+        uma_prefix_ + "QualityLimitedResolutionInPercent", quality_limited);
+  }
+  int downscales = quality_downscales_counter_.Avg(kMinRequiredSamples);
+  if (downscales != -1) {
+    RTC_HISTOGRAM_ENUMERATION_SPARSE(
+        uma_prefix_ + "QualityLimitedResolutionDownscales", downscales, 20);
+  }
+  int bw_limited = bw_limited_frame_counter_.Percent(kMinRequiredSamples);
+  if (bw_limited != -1) {
+    RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+        uma_prefix_ + "BandwidthLimitedResolutionInPercent", bw_limited);
+  }
+  int num_disabled = bw_resolutions_disabled_counter_.Avg(kMinRequiredSamples);
+  if (num_disabled != -1) {
+    RTC_HISTOGRAM_ENUMERATION_SPARSE(
+        uma_prefix_ + "BandwidthLimitedResolutionsDisabled", num_disabled, 10);
+  }
+  int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
+  if (delay_ms != -1)
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix_ + "SendSideDelayInMs",
+                                       delay_ms);
+
+  int max_delay_ms = max_delay_counter_.Avg(kMinRequiredSamples);
+  if (max_delay_ms != -1) {
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix_ + "SendSideDelayMaxInMs",
+                                       max_delay_ms);
+  }
+}
+
+void SendStatisticsProxy::SetContentType(
+    VideoEncoderConfig::ContentType content_type) {
+  rtc::CritScope lock(&crit_);
+  if (content_type_ != content_type) {
+    uma_container_.reset(new UmaSamplesContainer(GetUmaPrefix(content_type)));
+    content_type_ = content_type;
+  }
+}
+
+void SendStatisticsProxy::OnEncoderImplementationName(
+    const char* implementation_name) {
+  rtc::CritScope lock(&crit_);
+  stats_.encoder_implementation_name = implementation_name;
+}
+
+void SendStatisticsProxy::OnOutgoingRate(uint32_t framerate, uint32_t bitrate) {
+  rtc::CritScope lock(&crit_);
   stats_.encode_frame_rate = framerate;
   stats_.media_bitrate_bps = bitrate;
 }
 
 void SendStatisticsProxy::CpuOveruseMetricsUpdated(
     const CpuOveruseMetrics& metrics) {
-  CriticalSectionScoped lock(crit_.get());
-  stats_.avg_encode_time_ms = metrics.avg_encode_time_ms;
+  rtc::CritScope lock(&crit_);
   stats_.encode_usage_percent = metrics.encode_usage_percent;
 }
 
-void SendStatisticsProxy::SuspendChange(int video_channel, bool is_suspended) {
-  CriticalSectionScoped lock(crit_.get());
+void SendStatisticsProxy::OnSuspendChange(bool is_suspended) {
+  rtc::CritScope lock(&crit_);
   stats_.suspended = is_suspended;
 }
 
 VideoSendStream::Stats SendStatisticsProxy::GetStats() {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   PurgeOldStats();
   stats_.input_frame_rate =
-      static_cast<int>(input_frame_rate_tracker_.units_second());
+      round(uma_container_->input_frame_rate_tracker_.ComputeRate());
   return stats_;
 }
 
 void SendStatisticsProxy::PurgeOldStats() {
-  int64_t current_time_ms = clock_->TimeInMilliseconds();
+  int64_t old_stats_ms = clock_->TimeInMilliseconds() - kStatsTimeoutMs;
   for (std::map<uint32_t, VideoSendStream::StreamStats>::iterator it =
            stats_.substreams.begin();
        it != stats_.substreams.end(); ++it) {
     uint32_t ssrc = it->first;
-    if (update_times_[ssrc].resolution_update_ms + kStatsTimeoutMs >
-        current_time_ms)
-      continue;
-
-    it->second.width = 0;
-    it->second.height = 0;
+    if (update_times_[ssrc].resolution_update_ms <= old_stats_ms) {
+      it->second.width = 0;
+      it->second.height = 0;
+    }
   }
 }
 
@@ -91,8 +229,20 @@ VideoSendStream::StreamStats* SendStatisticsProxy::GetStatsEntry(
   return &stats_.substreams[ssrc];  // Insert new entry and return ptr.
 }
 
+void SendStatisticsProxy::OnInactiveSsrc(uint32_t ssrc) {
+  rtc::CritScope lock(&crit_);
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
+    return;
+
+  stats->total_bitrate_bps = 0;
+  stats->retransmit_bitrate_bps = 0;
+  stats->height = 0;
+  stats->width = 0;
+}
+
 void SendStatisticsProxy::OnSetRates(uint32_t bitrate_bps, int framerate) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   stats_.target_media_bitrate_bps = bitrate_bps;
 }
 
@@ -108,7 +258,7 @@ void SendStatisticsProxy::OnSendEncodedImage(
   }
   uint32_t ssrc = config_.rtp.ssrcs[simulcast_idx];
 
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
@@ -116,17 +266,72 @@ void SendStatisticsProxy::OnSendEncodedImage(
   stats->width = encoded_image._encodedWidth;
   stats->height = encoded_image._encodedHeight;
   update_times_[ssrc].resolution_update_ms = clock_->TimeInMilliseconds();
+
+  uma_container_->key_frame_counter_.Add(encoded_image._frameType ==
+                                         kVideoFrameKey);
+
+  stats_.bw_limited_resolution =
+      encoded_image.adapt_reason_.quality_resolution_downscales > 0 ||
+      encoded_image.adapt_reason_.bw_resolutions_disabled > 0;
+
+  if (encoded_image.adapt_reason_.quality_resolution_downscales != -1) {
+    bool downscaled =
+        encoded_image.adapt_reason_.quality_resolution_downscales > 0;
+    uma_container_->quality_limited_frame_counter_.Add(downscaled);
+    if (downscaled) {
+      uma_container_->quality_downscales_counter_.Add(
+          encoded_image.adapt_reason_.quality_resolution_downscales);
+    }
+  }
+  if (encoded_image.adapt_reason_.bw_resolutions_disabled != -1) {
+    bool bw_limited = encoded_image.adapt_reason_.bw_resolutions_disabled > 0;
+    uma_container_->bw_limited_frame_counter_.Add(bw_limited);
+    if (bw_limited) {
+      uma_container_->bw_resolutions_disabled_counter_.Add(
+          encoded_image.adapt_reason_.bw_resolutions_disabled);
+    }
+  }
+
+  // TODO(asapersson): This is incorrect if simulcast layers are encoded on
+  // different threads and there is no guarantee that one frame of all layers
+  // are encoded before the next start.
+  if (last_sent_frame_timestamp_ > 0 &&
+      encoded_image._timeStamp != last_sent_frame_timestamp_) {
+    uma_container_->sent_frame_rate_tracker_.AddSamples(1);
+    uma_container_->sent_width_counter_.Add(
+        uma_container_->max_sent_width_per_timestamp_);
+    uma_container_->sent_height_counter_.Add(
+        uma_container_->max_sent_height_per_timestamp_);
+    uma_container_->max_sent_width_per_timestamp_ = 0;
+    uma_container_->max_sent_height_per_timestamp_ = 0;
+  }
+  last_sent_frame_timestamp_ = encoded_image._timeStamp;
+  uma_container_->max_sent_width_per_timestamp_ =
+      std::max(uma_container_->max_sent_width_per_timestamp_,
+               static_cast<int>(encoded_image._encodedWidth));
+  uma_container_->max_sent_height_per_timestamp_ =
+      std::max(uma_container_->max_sent_height_per_timestamp_,
+               static_cast<int>(encoded_image._encodedHeight));
 }
 
-void SendStatisticsProxy::OnIncomingFrame() {
-  CriticalSectionScoped lock(crit_.get());
-  input_frame_rate_tracker_.Update(1);
+void SendStatisticsProxy::OnIncomingFrame(int width, int height) {
+  rtc::CritScope lock(&crit_);
+  uma_container_->input_frame_rate_tracker_.AddSamples(1);
+  uma_container_->input_width_counter_.Add(width);
+  uma_container_->input_height_counter_.Add(height);
+}
+
+void SendStatisticsProxy::OnEncodedFrame(int encode_time_ms) {
+  rtc::CritScope lock(&crit_);
+  uma_container_->encode_time_counter_.Add(encode_time_ms);
+  encode_time_.Apply(1.0f, encode_time_ms);
+  stats_.avg_encode_time_ms = round(encode_time_.filtered());
 }
 
 void SendStatisticsProxy::RtcpPacketTypesCounterUpdated(
     uint32_t ssrc,
     const RtcpPacketTypeCounter& packet_counter) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
@@ -136,7 +341,7 @@ void SendStatisticsProxy::RtcpPacketTypesCounterUpdated(
 
 void SendStatisticsProxy::StatisticsUpdated(const RtcpStatistics& statistics,
                                             uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
@@ -150,10 +355,10 @@ void SendStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
 void SendStatisticsProxy::DataCountersUpdated(
     const StreamDataCounters& counters,
     uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
-  DCHECK(stats != nullptr) << "DataCountersUpdated reported for unknown ssrc: "
-                           << ssrc;
+  RTC_DCHECK(stats != nullptr)
+      << "DataCountersUpdated reported for unknown ssrc: " << ssrc;
 
   stats->rtp_stats = counters;
 }
@@ -161,7 +366,7 @@ void SendStatisticsProxy::DataCountersUpdated(
 void SendStatisticsProxy::Notify(const BitrateStatistics& total_stats,
                                  const BitrateStatistics& retransmit_stats,
                                  uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
@@ -172,7 +377,7 @@ void SendStatisticsProxy::Notify(const BitrateStatistics& total_stats,
 
 void SendStatisticsProxy::FrameCountUpdated(const FrameCounts& frame_counts,
                                             uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
@@ -183,12 +388,48 @@ void SendStatisticsProxy::FrameCountUpdated(const FrameCounts& frame_counts,
 void SendStatisticsProxy::SendSideDelayUpdated(int avg_delay_ms,
                                                int max_delay_ms,
                                                uint32_t ssrc) {
-  CriticalSectionScoped lock(crit_.get());
+  rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
   stats->avg_delay_ms = avg_delay_ms;
   stats->max_delay_ms = max_delay_ms;
+
+  uma_container_->delay_counter_.Add(avg_delay_ms);
+  uma_container_->max_delay_counter_.Add(max_delay_ms);
 }
 
+void SendStatisticsProxy::SampleCounter::Add(int sample) {
+  sum += sample;
+  ++num_samples;
+}
+
+int SendStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
+  if (num_samples < min_required_samples || num_samples == 0)
+    return -1;
+  return (sum + (num_samples / 2)) / num_samples;
+}
+
+void SendStatisticsProxy::BoolSampleCounter::Add(bool sample) {
+  if (sample)
+    ++sum;
+  ++num_samples;
+}
+
+int SendStatisticsProxy::BoolSampleCounter::Percent(
+    int min_required_samples) const {
+  return Fraction(min_required_samples, 100.0f);
+}
+
+int SendStatisticsProxy::BoolSampleCounter::Permille(
+    int min_required_samples) const {
+  return Fraction(min_required_samples, 1000.0f);
+}
+
+int SendStatisticsProxy::BoolSampleCounter::Fraction(
+    int min_required_samples, float multiplier) const {
+  if (num_samples < min_required_samples || num_samples == 0)
+    return -1;
+  return static_cast<int>((sum * multiplier / num_samples) + 0.5f);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h
index 0a048a59c2..7f6df06ad8 100644
--- a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h
+++ b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h
@@ -11,37 +11,38 @@
 #ifndef WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_
 #define WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_
 
+#include <map>
 #include <string>
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/exp_filter.h"
 #include "webrtc/base/ratetracker.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/video/overuse_frame_detector.h"
+#include "webrtc/video/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 class SendStatisticsProxy : public CpuOveruseMetricsObserver,
                             public RtcpStatisticsCallback,
                             public RtcpPacketTypeCounterObserver,
                             public StreamDataCountersCallback,
                             public BitrateStatisticsObserver,
                             public FrameCountObserver,
-                            public ViEEncoderObserver,
                             public VideoEncoderRateObserver,
                             public SendSideDelayObserver {
  public:
   static const int kStatsTimeoutMs;
 
-  SendStatisticsProxy(Clock* clock, const VideoSendStream::Config& config);
+  SendStatisticsProxy(Clock* clock,
+                      const VideoSendStream::Config& config,
+                      VideoEncoderConfig::ContentType content_type);
   virtual ~SendStatisticsProxy();
 
   VideoSendStream::Stats GetStats();
@@ -49,11 +50,23 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
   virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
                                   const RTPVideoHeader* rtp_video_header);
   // Used to update incoming frame rate.
-  void OnIncomingFrame();
+  void OnIncomingFrame(int width, int height);
+
+  // Used to update encode time of frames.
+  void OnEncodedFrame(int encode_time_ms);
 
   // From VideoEncoderRateObserver.
   void OnSetRates(uint32_t bitrate_bps, int framerate) override;
 
+  void OnEncoderImplementationName(const char* implementation_name);
+  void OnOutgoingRate(uint32_t framerate, uint32_t bitrate);
+  void OnSuspendChange(bool is_suspended);
+  void OnInactiveSsrc(uint32_t ssrc);
+
+  // Used to indicate change in content type, which may require a change in
+  // how stats are collected.
+  void SetContentType(VideoEncoderConfig::ContentType content_type);
+
  protected:
   // From CpuOveruseMetricsObserver.
   void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override;
@@ -61,7 +74,7 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
   void StatisticsUpdated(const RtcpStatistics& statistics,
                          uint32_t ssrc) override;
   void CNameChanged(const char* cname, uint32_t ssrc) override;
-  // From RtcpPacketTypeCounterObserver
+  // From RtcpPacketTypeCounterObserver.
   void RtcpPacketTypesCounterUpdated(
       uint32_t ssrc,
       const RtcpPacketTypeCounter& packet_counter) override;
@@ -78,21 +91,39 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
   void FrameCountUpdated(const FrameCounts& frame_counts,
                          uint32_t ssrc) override;
 
-  // From ViEEncoderObserver.
-  void OutgoingRate(const int video_channel,
-                    const unsigned int framerate,
-                    const unsigned int bitrate) override;
-
-  void SuspendChange(int video_channel, bool is_suspended) override;
-
   void SendSideDelayUpdated(int avg_delay_ms,
                             int max_delay_ms,
                             uint32_t ssrc) override;
 
  private:
+  class SampleCounter {
+   public:
+    SampleCounter() : sum(0), num_samples(0) {}
+    ~SampleCounter() {}
+    void Add(int sample);
+    int Avg(int min_required_samples) const;
+
+   private:
+    int sum;
+    int num_samples;
+  };
+  class BoolSampleCounter {
+   public:
+    BoolSampleCounter() : sum(0), num_samples(0) {}
+    ~BoolSampleCounter() {}
+    void Add(bool sample);
+    int Percent(int min_required_samples) const;
+    int Permille(int min_required_samples) const;
+
+   private:
+    int Fraction(int min_required_samples, float multiplier) const;
+    int sum;
+    int num_samples;
+  };
   struct StatsUpdateTimes {
-    StatsUpdateTimes() : resolution_update_ms(0) {}
+    StatsUpdateTimes() : resolution_update_ms(0), bitrate_update_ms(0) {}
     int64_t resolution_update_ms;
+    int64_t bitrate_update_ms;
   };
   void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
@@ -100,10 +131,42 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
 
   Clock* const clock_;
   const VideoSendStream::Config config_;
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
+  VideoEncoderConfig::ContentType content_type_ GUARDED_BY(crit_);
   VideoSendStream::Stats stats_ GUARDED_BY(crit_);
-  rtc::RateTracker input_frame_rate_tracker_ GUARDED_BY(crit_);
+  uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_);
   std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_);
+  rtc::ExpFilter encode_time_ GUARDED_BY(crit_);
+
+  // Contains stats used for UMA histograms. These stats will be reset if
+  // content type changes between real-time video and screenshare, since these
+  // will be reported separately.
+  struct UmaSamplesContainer {
+    explicit UmaSamplesContainer(const char* prefix);
+    ~UmaSamplesContainer();
+
+    void UpdateHistograms();
+
+    const std::string uma_prefix_;
+    int max_sent_width_per_timestamp_;
+    int max_sent_height_per_timestamp_;
+    SampleCounter input_width_counter_;
+    SampleCounter input_height_counter_;
+    SampleCounter sent_width_counter_;
+    SampleCounter sent_height_counter_;
+    SampleCounter encode_time_counter_;
+    BoolSampleCounter key_frame_counter_;
+    BoolSampleCounter quality_limited_frame_counter_;
+    SampleCounter quality_downscales_counter_;
+    BoolSampleCounter bw_limited_frame_counter_;
+    SampleCounter bw_resolutions_disabled_counter_;
+    SampleCounter delay_counter_;
+    SampleCounter max_delay_counter_;
+    rtc::RateTracker input_frame_rate_tracker_;
+    rtc::RateTracker sent_frame_rate_tracker_;
+  };
+
+  rtc::scoped_ptr<UmaSamplesContainer> uma_container_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc b/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc
index 0243add3b0..fc1f3fdbde 100644
--- a/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc
@@ -16,25 +16,27 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/histogram.h"
 
 namespace webrtc {
 
 class SendStatisticsProxyTest : public ::testing::Test {
  public:
   SendStatisticsProxyTest()
-      : fake_clock_(1234), avg_delay_ms_(0), max_delay_ms_(0) {}
+      : fake_clock_(1234), config_(GetTestConfig()), avg_delay_ms_(0),
+        max_delay_ms_(0) {}
   virtual ~SendStatisticsProxyTest() {}
 
  protected:
   virtual void SetUp() {
-    statistics_proxy_.reset(
-        new SendStatisticsProxy(&fake_clock_, GetTestConfig()));
-    config_ = GetTestConfig();
+    statistics_proxy_.reset(new SendStatisticsProxy(
+        &fake_clock_, GetTestConfig(),
+        VideoEncoderConfig::ContentType::kRealtimeVideo));
     expected_ = VideoSendStream::Stats();
   }
 
   VideoSendStream::Config GetTestConfig() {
-    VideoSendStream::Config config;
+    VideoSendStream::Config config(nullptr);
     config.rtp.ssrcs.push_back(17);
     config.rtp.ssrcs.push_back(42);
     config.rtp.rtx.ssrcs.push_back(18);
@@ -129,11 +131,10 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
 }
 
 TEST_F(SendStatisticsProxyTest, EncodedBitrateAndFramerate) {
-  const int media_bitrate_bps = 500;
-  const int encode_fps = 29;
+  int media_bitrate_bps = 500;
+  int encode_fps = 29;
 
-  ViEEncoderObserver* encoder_observer = statistics_proxy_.get();
-  encoder_observer->OutgoingRate(0, encode_fps, media_bitrate_bps);
+  statistics_proxy_->OnOutgoingRate(encode_fps, media_bitrate_bps);
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   EXPECT_EQ(media_bitrate_bps, stats.media_bitrate_bps);
@@ -145,12 +146,11 @@ TEST_F(SendStatisticsProxyTest, Suspended) {
   EXPECT_FALSE(statistics_proxy_->GetStats().suspended);
 
   // Verify that we can set it to true.
-  ViEEncoderObserver* encoder_observer = statistics_proxy_.get();
-  encoder_observer->SuspendChange(0, true);
+  statistics_proxy_->OnSuspendChange(true);
   EXPECT_TRUE(statistics_proxy_->GetStats().suspended);
 
   // Verify that we can set it back to false again.
-  encoder_observer->SuspendChange(0, false);
+  statistics_proxy_->OnSuspendChange(false);
   EXPECT_FALSE(statistics_proxy_->GetStats().suspended);
 }
 
@@ -289,6 +289,33 @@ TEST_F(SendStatisticsProxyTest, SendSideDelay) {
   ExpectEqual(expected_, stats);
 }
 
+TEST_F(SendStatisticsProxyTest, OnEncodedFrame) {
+  const int kEncodeTimeMs = 11;
+  statistics_proxy_->OnEncodedFrame(kEncodeTimeMs);
+
+  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(kEncodeTimeMs, stats.avg_encode_time_ms);
+}
+
+TEST_F(SendStatisticsProxyTest, SwitchContentTypeUpdatesHistograms) {
+  test::ClearHistograms();
+  const int kMinRequiredSamples = 200;
+  const int kWidth = 640;
+  const int kHeight = 480;
+
+  for (int i = 0; i < kMinRequiredSamples; ++i)
+    statistics_proxy_->OnIncomingFrame(kWidth, kHeight);
+
+  // No switch, stats not should be updated.
+  statistics_proxy_->SetContentType(
+      VideoEncoderConfig::ContentType::kRealtimeVideo);
+  EXPECT_EQ(0, test::NumHistogramSamples("WebRTC.Video.InputWidthInPixels"));
+
+  // Switch to screenshare, real-time stats should be updated.
+  statistics_proxy_->SetContentType(VideoEncoderConfig::ContentType::kScreen);
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.InputWidthInPixels"));
+}
+
 TEST_F(SendStatisticsProxyTest, NoSubstreams) {
   uint32_t excluded_ssrc =
       std::max(
@@ -364,4 +391,44 @@ TEST_F(SendStatisticsProxyTest, EncodedResolutionTimesOut) {
   EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].height);
 }
 
+TEST_F(SendStatisticsProxyTest, ClearsResolutionFromInactiveSsrcs) {
+  static const int kEncodedWidth = 123;
+  static const int kEncodedHeight = 81;
+  EncodedImage encoded_image;
+  encoded_image._encodedWidth = kEncodedWidth;
+  encoded_image._encodedHeight = kEncodedHeight;
+
+  RTPVideoHeader rtp_video_header;
+
+  rtp_video_header.simulcastIdx = 0;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+  rtp_video_header.simulcastIdx = 1;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+
+  statistics_proxy_->OnInactiveSsrc(config_.rtp.ssrcs[1]);
+  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].width);
+  EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].height);
+  EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].width);
+  EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].height);
+}
+
+TEST_F(SendStatisticsProxyTest, ClearsBitratesFromInactiveSsrcs) {
+  BitrateStatistics bitrate;
+  bitrate.bitrate_bps = 42;
+  BitrateStatisticsObserver* observer = statistics_proxy_.get();
+  observer->Notify(bitrate, bitrate, config_.rtp.ssrcs[0]);
+  observer->Notify(bitrate, bitrate, config_.rtp.ssrcs[1]);
+
+  statistics_proxy_->OnInactiveSsrc(config_.rtp.ssrcs[1]);
+
+  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps),
+            stats.substreams[config_.rtp.ssrcs[0]].total_bitrate_bps);
+  EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps),
+            stats.substreams[config_.rtp.ssrcs[0]].retransmit_bitrate_bps);
+  EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].total_bitrate_bps);
+  EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].retransmit_bitrate_bps);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/stream_synchronization.cc b/media/webrtc/trunk/webrtc/video/stream_synchronization.cc
similarity index 95%
rename from media/webrtc/trunk/webrtc/video_engine/stream_synchronization.cc
rename to media/webrtc/trunk/webrtc/video/stream_synchronization.cc
index 8f72fa93cd..cb37d80ef5 100644
--- a/media/webrtc/trunk/webrtc/video_engine/stream_synchronization.cc
+++ b/media/webrtc/trunk/webrtc/video/stream_synchronization.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/stream_synchronization.h"
+#include "webrtc/video/stream_synchronization.h"
 
 #include <assert.h>
 #include <math.h>
@@ -16,7 +16,7 @@
 
 #include <algorithm>
 
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
@@ -42,13 +42,14 @@ struct ViESyncDelay {
   int network_delay;
 };
 
-StreamSynchronization::StreamSynchronization(int audio_channel_id,
-                                             int video_channel_id)
+StreamSynchronization::StreamSynchronization(uint32_t video_primary_ssrc,
+                                             int audio_channel_id)
     : channel_delay_(new ViESyncDelay),
+      video_primary_ssrc_(video_primary_ssrc),
       audio_channel_id_(audio_channel_id),
-      video_channel_id_(video_channel_id),
       base_target_delay_ms_(0),
-      avg_diff_ms_(0) {}
+      avg_diff_ms_(0) {
+}
 
 StreamSynchronization::~StreamSynchronization() {
   delete channel_delay_;
@@ -193,8 +194,8 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
   channel_delay_->last_audio_delay_ms = new_audio_delay_ms;
 
   LOG(LS_VERBOSE) << "Sync video delay " << new_video_delay_ms
+                  << " for video primary SSRC " << video_primary_ssrc_
                   << " and audio delay " << channel_delay_->extra_audio_delay_ms
-                  << " for video channel " << video_channel_id_
                   << " for audio channel " << audio_channel_id_;
 
   // Return values.
diff --git a/media/webrtc/trunk/webrtc/video_engine/stream_synchronization.h b/media/webrtc/trunk/webrtc/video/stream_synchronization.h
similarity index 82%
rename from media/webrtc/trunk/webrtc/video_engine/stream_synchronization.h
rename to media/webrtc/trunk/webrtc/video/stream_synchronization.h
index 5fa9536d17..cb7c110f44 100644
--- a/media/webrtc/trunk/webrtc/video_engine/stream_synchronization.h
+++ b/media/webrtc/trunk/webrtc/video/stream_synchronization.h
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
-#define WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
+#ifndef WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
+#define WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
 
 #include <list>
 
-#include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,7 +29,7 @@ class StreamSynchronization {
     uint32_t latest_timestamp;
   };
 
-  StreamSynchronization(int audio_channel_id, int video_channel_id);
+  StreamSynchronization(uint32_t video_primary_ssrc, int audio_channel_id);
   ~StreamSynchronization();
 
   bool ComputeDelays(int relative_delay_ms,
@@ -49,11 +49,11 @@ class StreamSynchronization {
 
  private:
   ViESyncDelay* channel_delay_;
-  int audio_channel_id_;
-  int video_channel_id_;
+  const uint32_t video_primary_ssrc_;
+  const int audio_channel_id_;
   int base_target_delay_ms_;
   int avg_diff_ms_;
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_STREAM_SYNCHRONIZATION_H_
+#endif  // WEBRTC_VIDEO_STREAM_SYNCHRONIZATION_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/stream_synchronization_unittest.cc b/media/webrtc/trunk/webrtc/video/stream_synchronization_unittest.cc
similarity index 99%
rename from media/webrtc/trunk/webrtc/video_engine/stream_synchronization_unittest.cc
rename to media/webrtc/trunk/webrtc/video/stream_synchronization_unittest.cc
index 7136f1e1c7..2834dfe1b2 100644
--- a/media/webrtc/trunk/webrtc/video_engine/stream_synchronization_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/stream_synchronization_unittest.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <algorithm>
 #include <math.h>
 
+#include <algorithm>
+
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video_engine/stream_synchronization.h"
+#include "webrtc/video/stream_synchronization.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/video/video_capture_input.cc b/media/webrtc/trunk/webrtc/video/video_capture_input.cc
new file mode 100644
index 0000000000..1c5f299291
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_capture_input.cc
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/video_capture_input.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video/overuse_frame_detector.h"
+#include "webrtc/video/send_statistics_proxy.h"
+#include "webrtc/video/vie_encoder.h"
+
+namespace webrtc {
+
+namespace internal {
+VideoCaptureInput::VideoCaptureInput(
+    ProcessThread* module_process_thread,
+    VideoCaptureCallback* frame_callback,
+    VideoRenderer* local_renderer,
+    SendStatisticsProxy* stats_proxy,
+    CpuOveruseObserver* overuse_observer,
+    EncodingTimeObserver* encoding_time_observer)
+    : capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      module_process_thread_(module_process_thread),
+      frame_callback_(frame_callback),
+      local_renderer_(local_renderer),
+      stats_proxy_(stats_proxy),
+      incoming_frame_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      encoder_thread_(EncoderThreadFunction, this, "EncoderThread"),
+      capture_event_(false, false),
+      stop_(0),
+      last_captured_timestamp_(0),
+      delta_ntp_internal_ms_(
+          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() -
+          TickTime::MillisecondTimestamp()),
+      overuse_detector_(new OveruseFrameDetector(Clock::GetRealTimeClock(),
+                                                 CpuOveruseOptions(),
+                                                 overuse_observer,
+                                                 stats_proxy)),
+      encoding_time_observer_(encoding_time_observer) {
+  encoder_thread_.Start();
+  encoder_thread_.SetPriority(rtc::kHighPriority);
+  module_process_thread_->RegisterModule(overuse_detector_.get());
+}
+
+VideoCaptureInput::~VideoCaptureInput() {
+  module_process_thread_->DeRegisterModule(overuse_detector_.get());
+
+  // Stop the thread.
+  rtc::AtomicOps::ReleaseStore(&stop_, 1);
+  capture_event_.Set();
+  encoder_thread_.Stop();
+}
+
+void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) {
+  // TODO(pbos): Remove local rendering, it should be handled by the client code
+  // if required.
+  if (local_renderer_)
+    local_renderer_->RenderFrame(video_frame, 0);
+
+  stats_proxy_->OnIncomingFrame(video_frame.width(), video_frame.height());
+
+  VideoFrame incoming_frame = video_frame;
+
+  if (incoming_frame.ntp_time_ms() != 0) {
+    // If a NTP time stamp is set, this is the time stamp we will use.
+    incoming_frame.set_render_time_ms(incoming_frame.ntp_time_ms() -
+                                      delta_ntp_internal_ms_);
+  } else {  // NTP time stamp not set.
+    int64_t render_time = incoming_frame.render_time_ms() != 0
+                              ? incoming_frame.render_time_ms()
+                              : TickTime::MillisecondTimestamp();
+
+    incoming_frame.set_render_time_ms(render_time);
+    incoming_frame.set_ntp_time_ms(render_time + delta_ntp_internal_ms_);
+  }
+
+  // Convert NTP time, in ms, to RTP timestamp.
+  const int kMsToRtpTimestamp = 90;
+  incoming_frame.set_timestamp(
+      kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
+
+  CriticalSectionScoped cs(capture_cs_.get());
+  if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
+    // We don't allow the same capture time for two frames, drop this one.
+    LOG(LS_WARNING) << "Same/old NTP timestamp ("
+                    << incoming_frame.ntp_time_ms()
+                    << " <= " << last_captured_timestamp_
+                    << ") for incoming frame. Dropping.";
+    return;
+  }
+
+  captured_frame_.ShallowCopy(incoming_frame);
+  last_captured_timestamp_ = incoming_frame.ntp_time_ms();
+
+  overuse_detector_->FrameCaptured(captured_frame_.width(),
+                                   captured_frame_.height(),
+                                   captured_frame_.render_time_ms());
+
+  TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
+                           "render_time", video_frame.render_time_ms());
+
+  capture_event_.Set();
+}
+
+bool VideoCaptureInput::EncoderThreadFunction(void* obj) {
+  return static_cast<VideoCaptureInput*>(obj)->EncoderProcess();
+}
+
+bool VideoCaptureInput::EncoderProcess() {
+  static const int kThreadWaitTimeMs = 100;
+  int64_t capture_time = -1;
+  if (capture_event_.Wait(kThreadWaitTimeMs)) {
+    if (rtc::AtomicOps::AcquireLoad(&stop_))
+      return false;
+
+    int64_t encode_start_time = -1;
+    VideoFrame deliver_frame;
+    {
+      CriticalSectionScoped cs(capture_cs_.get());
+      if (!captured_frame_.IsZeroSize()) {
+        deliver_frame = captured_frame_;
+        captured_frame_.Reset();
+      }
+    }
+    if (!deliver_frame.IsZeroSize()) {
+      capture_time = deliver_frame.render_time_ms();
+      encode_start_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
+      frame_callback_->DeliverFrame(deliver_frame);
+    }
+    // Update the overuse detector with the duration.
+    if (encode_start_time != -1) {
+      int encode_time_ms = static_cast<int>(
+          Clock::GetRealTimeClock()->TimeInMilliseconds() - encode_start_time);
+      stats_proxy_->OnEncodedFrame(encode_time_ms);
+      if (encoding_time_observer_) {
+        encoding_time_observer_->OnReportEncodedTime(
+            deliver_frame.ntp_time_ms(), encode_time_ms);
+      }
+    }
+  }
+  // We're done!
+  if (capture_time != -1) {
+    overuse_detector_->FrameSent(capture_time);
+  }
+  return true;
+}
+
+}  // namespace internal
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_capture_input.h b/media/webrtc/trunk/webrtc/video/video_capture_input.h
new file mode 100644
index 0000000000..d44907cd0e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_capture_input.h
@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_VIDEO_CAPTURE_INPUT_H_
+#define WEBRTC_VIDEO_VIDEO_CAPTURE_INPUT_H_
+
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/video_capture/video_capture.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_send_stream.h"
+
+namespace webrtc {
+
+class Config;
+class CpuOveruseMetricsObserver;
+class CpuOveruseObserver;
+class CriticalSectionWrapper;
+class OveruseFrameDetector;
+class ProcessThread;
+class RegistrableCpuOveruseMetricsObserver;
+class SendStatisticsProxy;
+class VideoRenderer;
+
+class VideoCaptureCallback {
+ public:
+  virtual ~VideoCaptureCallback() {}
+
+  virtual void DeliverFrame(VideoFrame video_frame) = 0;
+};
+
+namespace internal {
+class VideoCaptureInput : public webrtc::VideoCaptureInput {
+ public:
+  VideoCaptureInput(ProcessThread* module_process_thread,
+                    VideoCaptureCallback* frame_callback,
+                    VideoRenderer* local_renderer,
+                    SendStatisticsProxy* send_stats_proxy,
+                    CpuOveruseObserver* overuse_observer,
+                    EncodingTimeObserver* encoding_time_observer);
+  ~VideoCaptureInput();
+
+  void IncomingCapturedFrame(const VideoFrame& video_frame) override;
+
+ private:
+  // Thread functions for deliver captured frames to receivers.
+  static bool EncoderThreadFunction(void* obj);
+  bool EncoderProcess();
+
+  rtc::scoped_ptr<CriticalSectionWrapper> capture_cs_;
+  ProcessThread* const module_process_thread_;
+
+  VideoCaptureCallback* const frame_callback_;
+  VideoRenderer* const local_renderer_;
+  SendStatisticsProxy* const stats_proxy_;
+
+  // Frame used in IncomingFrameI420.
+  rtc::scoped_ptr<CriticalSectionWrapper> incoming_frame_cs_;
+  VideoFrame incoming_frame_;
+
+  rtc::PlatformThread encoder_thread_;
+  rtc::Event capture_event_;
+
+  volatile int stop_;
+
+  VideoFrame captured_frame_ GUARDED_BY(capture_cs_.get());
+  // Used to make sure incoming time stamp is increasing for every frame.
+  int64_t last_captured_timestamp_;
+  // Delta used for translating between NTP and internal timestamps.
+  const int64_t delta_ntp_internal_ms_;
+
+  rtc::scoped_ptr<OveruseFrameDetector> overuse_detector_;
+  EncodingTimeObserver* const encoding_time_observer_;
+};
+
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIDEO_CAPTURE_INPUT_H_
diff --git a/media/webrtc/trunk/webrtc/video/video_capture_input_unittest.cc b/media/webrtc/trunk/webrtc/video/video_capture_input_unittest.cc
new file mode 100644
index 0000000000..9d720e2294
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_capture_input_unittest.cc
@@ -0,0 +1,305 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/video/video_capture_input.h"
+
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/scoped_vector.h"
+#include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/video/send_statistics_proxy.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::WithArg;
+
+// If an output frame does not arrive in 500ms, the test will fail.
+#define FRAME_TIMEOUT_MS 500
+
+namespace webrtc {
+
+class MockVideoCaptureCallback : public VideoCaptureCallback {
+ public:
+  MOCK_METHOD1(DeliverFrame, void(VideoFrame video_frame));
+};
+
+bool EqualFrames(const VideoFrame& frame1, const VideoFrame& frame2);
+bool EqualTextureFrames(const VideoFrame& frame1, const VideoFrame& frame2);
+bool EqualBufferFrames(const VideoFrame& frame1, const VideoFrame& frame2);
+bool EqualFramesVector(const ScopedVector<VideoFrame>& frames1,
+                       const ScopedVector<VideoFrame>& frames2);
+VideoFrame* CreateVideoFrame(uint8_t length);
+
+class VideoCaptureInputTest : public ::testing::Test {
+ protected:
+  VideoCaptureInputTest()
+      : mock_process_thread_(new NiceMock<MockProcessThread>),
+        mock_frame_callback_(new NiceMock<MockVideoCaptureCallback>),
+        output_frame_event_(false, false),
+        stats_proxy_(Clock::GetRealTimeClock(),
+                     webrtc::VideoSendStream::Config(nullptr),
+                     webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo) {}
+
+  virtual void SetUp() {
+    EXPECT_CALL(*mock_frame_callback_, DeliverFrame(_))
+        .WillRepeatedly(
+            WithArg<0>(Invoke(this, &VideoCaptureInputTest::AddOutputFrame)));
+
+    Config config;
+    input_.reset(new internal::VideoCaptureInput(
+        mock_process_thread_.get(), mock_frame_callback_.get(), nullptr,
+        &stats_proxy_, nullptr, nullptr));
+  }
+
+  virtual void TearDown() {
+    // VideoCaptureInput accesses |mock_process_thread_| in destructor and
+    // should
+    // be deleted first.
+    input_.reset();
+  }
+
+  void AddInputFrame(VideoFrame* frame) {
+    input_->IncomingCapturedFrame(*frame);
+  }
+
+  void AddOutputFrame(const VideoFrame& frame) {
+    if (frame.native_handle() == NULL)
+      output_frame_ybuffers_.push_back(frame.buffer(kYPlane));
+    output_frames_.push_back(new VideoFrame(frame));
+    output_frame_event_.Set();
+  }
+
+  void WaitOutputFrame() {
+    EXPECT_TRUE(output_frame_event_.Wait(FRAME_TIMEOUT_MS));
+  }
+
+  rtc::scoped_ptr<MockProcessThread> mock_process_thread_;
+  rtc::scoped_ptr<MockVideoCaptureCallback> mock_frame_callback_;
+
+  // Used to send input capture frames to VideoCaptureInput.
+  rtc::scoped_ptr<internal::VideoCaptureInput> input_;
+
+  // Input capture frames of VideoCaptureInput.
+  ScopedVector<VideoFrame> input_frames_;
+
+  // Indicate an output frame has arrived.
+  rtc::Event output_frame_event_;
+
+  // Output delivered frames of VideoCaptureInput.
+  ScopedVector<VideoFrame> output_frames_;
+
+  // The pointers of Y plane buffers of output frames. This is used to verify
+  // the frame are swapped and not copied.
+  std::vector<const uint8_t*> output_frame_ybuffers_;
+  SendStatisticsProxy stats_proxy_;
+};
+
+TEST_F(VideoCaptureInputTest, DoesNotRetainHandleNorCopyBuffer) {
+  // Indicate an output frame has arrived.
+  rtc::Event frame_destroyed_event(false, false);
+  class TestBuffer : public webrtc::I420Buffer {
+   public:
+    explicit TestBuffer(rtc::Event* event) : I420Buffer(5, 5), event_(event) {}
+
+   private:
+    friend class rtc::RefCountedObject<TestBuffer>;
+    ~TestBuffer() override { event_->Set(); }
+    rtc::Event* const event_;
+  };
+
+  VideoFrame frame(
+      new rtc::RefCountedObject<TestBuffer>(&frame_destroyed_event), 1, 1,
+      kVideoRotation_0);
+
+  AddInputFrame(&frame);
+  WaitOutputFrame();
+
+  EXPECT_EQ(output_frames_[0]->video_frame_buffer().get(),
+            frame.video_frame_buffer().get());
+  output_frames_.clear();
+  frame.Reset();
+  EXPECT_TRUE(frame_destroyed_event.Wait(FRAME_TIMEOUT_MS));
+}
+
+TEST_F(VideoCaptureInputTest, TestNtpTimeStampSetIfRenderTimeSet) {
+  input_frames_.push_back(CreateVideoFrame(0));
+  input_frames_[0]->set_render_time_ms(5);
+  input_frames_[0]->set_ntp_time_ms(0);
+
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_GT(output_frames_[0]->ntp_time_ms(),
+            input_frames_[0]->render_time_ms());
+}
+
+TEST_F(VideoCaptureInputTest, TestRtpTimeStampSet) {
+  input_frames_.push_back(CreateVideoFrame(0));
+  input_frames_[0]->set_render_time_ms(0);
+  input_frames_[0]->set_ntp_time_ms(1);
+  input_frames_[0]->set_timestamp(0);
+
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_EQ(output_frames_[0]->timestamp(),
+            input_frames_[0]->ntp_time_ms() * 90);
+}
+
+TEST_F(VideoCaptureInputTest, DropsFramesWithSameOrOldNtpTimestamp) {
+  input_frames_.push_back(CreateVideoFrame(0));
+
+  input_frames_[0]->set_ntp_time_ms(17);
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_EQ(output_frames_[0]->timestamp(),
+            input_frames_[0]->ntp_time_ms() * 90);
+
+  // Repeat frame with the same NTP timestamp should drop.
+  AddInputFrame(input_frames_[0]);
+  EXPECT_FALSE(output_frame_event_.Wait(FRAME_TIMEOUT_MS));
+
+  // As should frames with a decreased NTP timestamp.
+  input_frames_[0]->set_ntp_time_ms(input_frames_[0]->ntp_time_ms() - 1);
+  AddInputFrame(input_frames_[0]);
+  EXPECT_FALSE(output_frame_event_.Wait(FRAME_TIMEOUT_MS));
+
+  // But delivering with an increased NTP timestamp should succeed.
+  input_frames_[0]->set_ntp_time_ms(4711);
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_EQ(output_frames_[1]->timestamp(),
+            input_frames_[0]->ntp_time_ms() * 90);
+}
+
+TEST_F(VideoCaptureInputTest, TestTextureFrames) {
+  const int kNumFrame = 3;
+  for (int i = 0 ; i < kNumFrame; ++i) {
+    test::FakeNativeHandle* dummy_handle = new test::FakeNativeHandle();
+    // Add one to |i| so that width/height > 0.
+    input_frames_.push_back(new VideoFrame(test::FakeNativeHandle::CreateFrame(
+        dummy_handle, i + 1, i + 1, i + 1, i + 1, webrtc::kVideoRotation_0)));
+    AddInputFrame(input_frames_[i]);
+    WaitOutputFrame();
+    EXPECT_EQ(dummy_handle, output_frames_[i]->native_handle());
+  }
+
+  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
+}
+
+TEST_F(VideoCaptureInputTest, TestI420Frames) {
+  const int kNumFrame = 4;
+  std::vector<const uint8_t*> ybuffer_pointers;
+  for (int i = 0; i < kNumFrame; ++i) {
+    input_frames_.push_back(CreateVideoFrame(static_cast<uint8_t>(i + 1)));
+    const VideoFrame* const_input_frame = input_frames_[i];
+    ybuffer_pointers.push_back(const_input_frame->buffer(kYPlane));
+    AddInputFrame(input_frames_[i]);
+    WaitOutputFrame();
+  }
+
+  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
+  // Make sure the buffer is not copied.
+  for (int i = 0; i < kNumFrame; ++i)
+    EXPECT_EQ(ybuffer_pointers[i], output_frame_ybuffers_[i]);
+}
+
+TEST_F(VideoCaptureInputTest, TestI420FrameAfterTextureFrame) {
+  test::FakeNativeHandle* dummy_handle = new test::FakeNativeHandle();
+  input_frames_.push_back(new VideoFrame(test::FakeNativeHandle::CreateFrame(
+      dummy_handle, 1, 1, 1, 1, webrtc::kVideoRotation_0)));
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_EQ(dummy_handle, output_frames_[0]->native_handle());
+
+  input_frames_.push_back(CreateVideoFrame(2));
+  AddInputFrame(input_frames_[1]);
+  WaitOutputFrame();
+
+  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
+}
+
+TEST_F(VideoCaptureInputTest, TestTextureFrameAfterI420Frame) {
+  input_frames_.push_back(CreateVideoFrame(1));
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+
+  test::FakeNativeHandle* dummy_handle = new test::FakeNativeHandle();
+  input_frames_.push_back(new VideoFrame(test::FakeNativeHandle::CreateFrame(
+      dummy_handle, 1, 1, 2, 2, webrtc::kVideoRotation_0)));
+  AddInputFrame(input_frames_[1]);
+  WaitOutputFrame();
+
+  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
+}
+
+bool EqualFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
+  if (frame1.native_handle() != NULL || frame2.native_handle() != NULL)
+    return EqualTextureFrames(frame1, frame2);
+  return EqualBufferFrames(frame1, frame2);
+}
+
+bool EqualTextureFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
+  return ((frame1.native_handle() == frame2.native_handle()) &&
+          (frame1.width() == frame2.width()) &&
+          (frame1.height() == frame2.height()) &&
+          (frame1.render_time_ms() == frame2.render_time_ms()));
+}
+
+bool EqualBufferFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
+  return ((frame1.width() == frame2.width()) &&
+          (frame1.height() == frame2.height()) &&
+          (frame1.stride(kYPlane) == frame2.stride(kYPlane)) &&
+          (frame1.stride(kUPlane) == frame2.stride(kUPlane)) &&
+          (frame1.stride(kVPlane) == frame2.stride(kVPlane)) &&
+          (frame1.render_time_ms() == frame2.render_time_ms()) &&
+          (frame1.allocated_size(kYPlane) == frame2.allocated_size(kYPlane)) &&
+          (frame1.allocated_size(kUPlane) == frame2.allocated_size(kUPlane)) &&
+          (frame1.allocated_size(kVPlane) == frame2.allocated_size(kVPlane)) &&
+          (memcmp(frame1.buffer(kYPlane), frame2.buffer(kYPlane),
+                  frame1.allocated_size(kYPlane)) == 0) &&
+          (memcmp(frame1.buffer(kUPlane), frame2.buffer(kUPlane),
+                  frame1.allocated_size(kUPlane)) == 0) &&
+          (memcmp(frame1.buffer(kVPlane), frame2.buffer(kVPlane),
+                  frame1.allocated_size(kVPlane)) == 0));
+}
+
+bool EqualFramesVector(const ScopedVector<VideoFrame>& frames1,
+                       const ScopedVector<VideoFrame>& frames2) {
+  if (frames1.size() != frames2.size())
+    return false;
+  for (size_t i = 0; i < frames1.size(); ++i) {
+    if (!EqualFrames(*frames1[i], *frames2[i]))
+      return false;
+  }
+  return true;
+}
+
+VideoFrame* CreateVideoFrame(uint8_t data) {
+  VideoFrame* frame = new VideoFrame();
+  const int width = 36;
+  const int height = 24;
+  const int kSizeY = width * height * 2;
+  uint8_t buffer[kSizeY];
+  memset(buffer, data, kSizeY);
+  frame->CreateFrame(buffer, buffer, buffer, width, height, width, width / 2,
+                     width / 2);
+  frame->set_render_time_ms(data);
+  return frame;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_decoder.cc b/media/webrtc/trunk/webrtc/video/video_decoder.cc
new file mode 100644
index 0000000000..d699175274
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_decoder.cc
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_decoder.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
+
+namespace webrtc {
+VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) {
+  switch (codec_type) {
+    case kH264:
+      RTC_DCHECK(H264Decoder::IsSupported());
+      return H264Decoder::Create();
+    case kVp8:
+      return VP8Decoder::Create();
+    case kVp9:
+      return VP9Decoder::Create();
+    case kUnsupportedCodec:
+      RTC_NOTREACHED();
+      return nullptr;
+  }
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+VideoDecoder::DecoderType CodecTypeToDecoderType(VideoCodecType codec_type) {
+  switch (codec_type) {
+    case kVideoCodecH264:
+      return VideoDecoder::kH264;
+    case kVideoCodecVP8:
+      return VideoDecoder::kVp8;
+    case kVideoCodecVP9:
+      return VideoDecoder::kVp9;
+    default:
+      return VideoDecoder::kUnsupportedCodec;
+  }
+}
+
+VideoDecoderSoftwareFallbackWrapper::VideoDecoderSoftwareFallbackWrapper(
+    VideoCodecType codec_type,
+    VideoDecoder* decoder)
+    : decoder_type_(CodecTypeToDecoderType(codec_type)),
+      decoder_(decoder),
+      callback_(nullptr) {
+}
+
+int32_t VideoDecoderSoftwareFallbackWrapper::InitDecode(
+    const VideoCodec* codec_settings,
+    int32_t number_of_cores) {
+  codec_settings_ = *codec_settings;
+  number_of_cores_ = number_of_cores;
+  return decoder_->InitDecode(codec_settings, number_of_cores);
+}
+
+bool VideoDecoderSoftwareFallbackWrapper::InitFallbackDecoder() {
+  RTC_CHECK(decoder_type_ != kUnsupportedCodec)
+      << "Decoder requesting fallback to codec not supported in software.";
+  LOG(LS_WARNING) << "Decoder falling back to software decoding.";
+  fallback_decoder_.reset(VideoDecoder::Create(decoder_type_));
+  if (fallback_decoder_->InitDecode(&codec_settings_, number_of_cores_) !=
+      WEBRTC_VIDEO_CODEC_OK) {
+    LOG(LS_ERROR) << "Failed to initialize software-decoder fallback.";
+    fallback_decoder_.reset();
+    return false;
+  }
+  if (callback_ != nullptr)
+    fallback_decoder_->RegisterDecodeCompleteCallback(callback_);
+  fallback_implementation_name_ =
+      std::string(fallback_decoder_->ImplementationName()) +
+      " (fallback from: " + decoder_->ImplementationName() + ")";
+  return true;
+}
+
+int32_t VideoDecoderSoftwareFallbackWrapper::Decode(
+    const EncodedImage& input_image,
+    bool missing_frames,
+    const RTPFragmentationHeader* fragmentation,
+    const CodecSpecificInfo* codec_specific_info,
+    int64_t render_time_ms) {
+  // Try decoding with the provided decoder on every keyframe or when there's no
+  // fallback decoder. This is the normal case.
+  if (!fallback_decoder_ || input_image._frameType == kVideoFrameKey) {
+    int32_t ret = decoder_->Decode(input_image, missing_frames, fragmentation,
+                                   codec_specific_info, render_time_ms);
+    if (ret == WEBRTC_VIDEO_CODEC_OK) {
+      if (fallback_decoder_) {
+        // Decode OK -> stop using fallback decoder.
+        fallback_decoder_->Release();
+        fallback_decoder_.reset();
+        return WEBRTC_VIDEO_CODEC_OK;
+      }
+    }
+    if (ret != WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE)
+      return ret;
+    if (!fallback_decoder_) {
+      // Try to initialize fallback decoder.
+      if (!InitFallbackDecoder())
+        return ret;
+    }
+  }
+  return fallback_decoder_->Decode(input_image, missing_frames, fragmentation,
+                                   codec_specific_info, render_time_ms);
+}
+
+int32_t VideoDecoderSoftwareFallbackWrapper::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  callback_ = callback;
+  int32_t ret = decoder_->RegisterDecodeCompleteCallback(callback);
+  if (fallback_decoder_)
+    return fallback_decoder_->RegisterDecodeCompleteCallback(callback);
+  return ret;
+}
+
+int32_t VideoDecoderSoftwareFallbackWrapper::Release() {
+  if (fallback_decoder_)
+    fallback_decoder_->Release();
+  return decoder_->Release();
+}
+
+int32_t VideoDecoderSoftwareFallbackWrapper::Reset() {
+  if (fallback_decoder_)
+    fallback_decoder_->Reset();
+  return decoder_->Reset();
+}
+
+bool VideoDecoderSoftwareFallbackWrapper::PrefersLateDecoding() const {
+  if (fallback_decoder_)
+    return fallback_decoder_->PrefersLateDecoding();
+  return decoder_->PrefersLateDecoding();
+}
+
+const char* VideoDecoderSoftwareFallbackWrapper::ImplementationName() const {
+  if (fallback_decoder_)
+    return fallback_implementation_name_.c_str();
+  return decoder_->ImplementationName();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_decoder_unittest.cc b/media/webrtc/trunk/webrtc/video/video_decoder_unittest.cc
new file mode 100644
index 0000000000..4d54a3e53f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_decoder_unittest.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_decoder.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
+
+namespace webrtc {
+
+class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
+ protected:
+  VideoDecoderSoftwareFallbackWrapperTest()
+      : fallback_wrapper_(kVideoCodecVP8, &fake_decoder_) {}
+
+  class CountingFakeDecoder : public VideoDecoder {
+   public:
+    int32_t InitDecode(const VideoCodec* codec_settings,
+                       int32_t number_of_cores) override {
+      ++init_decode_count_;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    int32_t Decode(const EncodedImage& input_image,
+                   bool missing_frames,
+                   const RTPFragmentationHeader* fragmentation,
+                   const CodecSpecificInfo* codec_specific_info,
+                   int64_t render_time_ms) override {
+      ++decode_count_;
+      return decode_return_code_;
+    }
+
+    int32_t RegisterDecodeCompleteCallback(
+        DecodedImageCallback* callback) override {
+      decode_complete_callback_ = callback;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    int32_t Release() override {
+      ++release_count_;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    int32_t Reset() override {
+      ++reset_count_;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    const char* ImplementationName() const override {
+      return "fake-decoder";
+    }
+
+    int init_decode_count_ = 0;
+    int decode_count_ = 0;
+    int32_t decode_return_code_ = WEBRTC_VIDEO_CODEC_OK;
+    DecodedImageCallback* decode_complete_callback_ = nullptr;
+    int release_count_ = 0;
+    int reset_count_ = 0;
+  };
+  CountingFakeDecoder fake_decoder_;
+  VideoDecoderSoftwareFallbackWrapper fallback_wrapper_;
+};
+
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest, InitializesDecoder) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+  EXPECT_EQ(1, fake_decoder_.init_decode_count_);
+}
+
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
+       CanRecoverFromSoftwareFallback) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+  // Unfortunately faking a VP8 frame is hard. Rely on no Decode -> using SW
+  // decoder.
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  EncodedImage encoded_image;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  EXPECT_EQ(1, fake_decoder_.decode_count_);
+
+  // Fail -> fake_decoder shouldn't be used anymore.
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  EXPECT_EQ(1, fake_decoder_.decode_count_)
+      << "Decoder used even though fallback should be active.";
+
+  // Should be able to recover on a keyframe.
+  encoded_image._frameType = kVideoFrameKey;
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_OK;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  EXPECT_EQ(2, fake_decoder_.decode_count_)
+      << "Wrapper did not try to decode a keyframe using registered decoder.";
+
+  encoded_image._frameType = kVideoFrameDelta;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  EXPECT_EQ(3, fake_decoder_.decode_count_)
+      << "Decoder not used on future delta frames.";
+}
+
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest, DoesNotFallbackOnEveryError) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_ERROR;
+  EncodedImage encoded_image;
+  EXPECT_EQ(
+      fake_decoder_.decode_return_code_,
+      fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1));
+  EXPECT_EQ(1, fake_decoder_.decode_count_);
+
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  EXPECT_EQ(2, fake_decoder_.decode_count_)
+      << "Decoder should be active even though previous decode failed.";
+}
+
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest, ForwardsReleaseCall) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+  fallback_wrapper_.Release();
+  EXPECT_EQ(1, fake_decoder_.release_count_);
+
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  EncodedImage encoded_image;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  EXPECT_EQ(1, fake_decoder_.release_count_)
+      << "Decoder should not be released during fallback.";
+  fallback_wrapper_.Release();
+  EXPECT_EQ(2, fake_decoder_.release_count_);
+}
+
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest, ForwardsResetCall) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+  fallback_wrapper_.Reset();
+  EXPECT_EQ(1, fake_decoder_.reset_count_);
+
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  EncodedImage encoded_image;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  fallback_wrapper_.Reset();
+  EXPECT_EQ(2, fake_decoder_.reset_count_)
+      << "Reset not forwarded during fallback.";
+}
+
+// TODO(pbos): Fake a VP8 frame well enough to actually receive a callback from
+// the software decoder.
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
+       ForwardsRegisterDecodeCompleteCallback) {
+  class FakeDecodedImageCallback : public DecodedImageCallback {
+    int32_t Decoded(VideoFrame& decodedImage) override { return 0; }
+    int32_t Decoded(
+        webrtc::VideoFrame& decodedImage, int64_t decode_time_ms) override {
+      RTC_NOTREACHED();
+      return -1;
+    }
+  } callback, callback2;
+
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+  fallback_wrapper_.RegisterDecodeCompleteCallback(&callback);
+  EXPECT_EQ(&callback, fake_decoder_.decode_complete_callback_);
+
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  EncodedImage encoded_image;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  fallback_wrapper_.RegisterDecodeCompleteCallback(&callback2);
+  EXPECT_EQ(&callback2, fake_decoder_.decode_complete_callback_);
+}
+
+TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
+       ReportsFallbackImplementationName) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitDecode(&codec, 2);
+
+  fake_decoder_.decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  EncodedImage encoded_image;
+  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
+  // Hard coded expected value since libvpx is the software implementation name
+  // for VP8. Change accordingly if the underlying implementation does.
+  EXPECT_STREQ("libvpx (fallback from: fake-decoder)",
+               fallback_wrapper_.ImplementationName());
+  fallback_wrapper_.Release();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_encoder.cc b/media/webrtc/trunk/webrtc/video/video_encoder.cc
new file mode 100644
index 0000000000..1a69259bd4
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_encoder.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_encoder.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
+#include "webrtc/modules/video_coding/codecs/vp8/vp8_factory.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
+
+namespace webrtc {
+  VideoEncoder* VideoEncoder::Create(VideoEncoder::EncoderType codec_type,
+                                     bool enable_simulcast) {
+  switch (codec_type) {
+    case kH264:
+      RTC_DCHECK(H264Encoder::IsSupported());
+      return H264Encoder::Create();
+    case kVp8:
+      VP8EncoderFactoryConfig::set_use_simulcast_adapter(enable_simulcast);
+      return VP8Encoder::Create();
+    case kVp9:
+      return VP9Encoder::Create();
+    case kUnsupportedCodec:
+      RTC_NOTREACHED();
+      return nullptr;
+  }
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+VideoEncoder::EncoderType CodecToEncoderType(VideoCodecType codec_type) {
+  switch (codec_type) {
+    case kVideoCodecH264:
+      return VideoEncoder::kH264;
+    case kVideoCodecVP8:
+      return VideoEncoder::kVp8;
+    case kVideoCodecVP9:
+      return VideoEncoder::kVp9;
+    default:
+      return VideoEncoder::kUnsupportedCodec;
+  }
+}
+
+VideoEncoderSoftwareFallbackWrapper::VideoEncoderSoftwareFallbackWrapper(
+    VideoCodecType codec_type,
+    webrtc::VideoEncoder* encoder)
+    : rates_set_(false),
+      channel_parameters_set_(false),
+      encoder_type_(CodecToEncoderType(codec_type)),
+      encoder_(encoder),
+      callback_(nullptr) {}
+
+bool VideoEncoderSoftwareFallbackWrapper::InitFallbackEncoder() {
+  RTC_CHECK(encoder_type_ != kUnsupportedCodec)
+      << "Encoder requesting fallback to codec not supported in software.";
+  fallback_encoder_.reset(VideoEncoder::Create(encoder_type_));
+  if (fallback_encoder_->InitEncode(&codec_settings_, number_of_cores_,
+                                    max_payload_size_) !=
+      WEBRTC_VIDEO_CODEC_OK) {
+    LOG(LS_ERROR) << "Failed to initialize software-encoder fallback.";
+    fallback_encoder_->Release();
+    fallback_encoder_.reset();
+    return false;
+  }
+  // Replay callback, rates, and channel parameters.
+  if (callback_)
+    fallback_encoder_->RegisterEncodeCompleteCallback(callback_);
+  if (rates_set_)
+    fallback_encoder_->SetRates(bitrate_, framerate_);
+  if (channel_parameters_set_)
+    fallback_encoder_->SetChannelParameters(packet_loss_, rtt_);
+
+  fallback_implementation_name_ =
+      std::string(fallback_encoder_->ImplementationName()) +
+      " (fallback from: " + encoder_->ImplementationName() + ")";
+  // Since we're switching to the fallback encoder, Release the real encoder. It
+  // may be re-initialized via InitEncode later, and it will continue to get
+  // Set calls for rates and channel parameters in the meantime.
+  encoder_->Release();
+  return true;
+}
+
+int32_t VideoEncoderSoftwareFallbackWrapper::InitEncode(
+    const VideoCodec* codec_settings,
+    int32_t number_of_cores,
+    size_t max_payload_size) {
+  // Store settings, in case we need to dynamically switch to the fallback
+  // encoder after a failed Encode call.
+  codec_settings_ = *codec_settings;
+  number_of_cores_ = number_of_cores;
+  max_payload_size_ = max_payload_size;
+  // Clear stored rate/channel parameters.
+  rates_set_ = false;
+  channel_parameters_set_ = false;
+
+  int32_t ret =
+      encoder_->InitEncode(codec_settings, number_of_cores, max_payload_size);
+  if (ret == WEBRTC_VIDEO_CODEC_OK || encoder_type_ == kUnsupportedCodec) {
+    if (fallback_encoder_)
+      fallback_encoder_->Release();
+    fallback_encoder_.reset();
+    if (callback_)
+      encoder_->RegisterEncodeCompleteCallback(callback_);
+    return ret;
+  }
+  // Try to instantiate software codec.
+  if (InitFallbackEncoder()) {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+  // Software encoder failed, use original return code.
+  return ret;
+}
+
+int32_t VideoEncoderSoftwareFallbackWrapper::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  callback_ = callback;
+  int32_t ret = encoder_->RegisterEncodeCompleteCallback(callback);
+  if (fallback_encoder_)
+    return fallback_encoder_->RegisterEncodeCompleteCallback(callback);
+  return ret;
+}
+
+int32_t VideoEncoderSoftwareFallbackWrapper::Release() {
+  // If the fallback_encoder_ is non-null, it means it was created via
+  // InitFallbackEncoder which has Release()d encoder_, so we should only ever
+  // need to Release() whichever one is active.
+  if (fallback_encoder_)
+    return fallback_encoder_->Release();
+  return encoder_->Release();
+}
+
+int32_t VideoEncoderSoftwareFallbackWrapper::Encode(
+    const VideoFrame& frame,
+    const CodecSpecificInfo* codec_specific_info,
+    const std::vector<FrameType>* frame_types) {
+  if (fallback_encoder_)
+    return fallback_encoder_->Encode(frame, codec_specific_info, frame_types);
+  int32_t ret = encoder_->Encode(frame, codec_specific_info, frame_types);
+  // If requested, try a software fallback.
+  if (ret == WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE && InitFallbackEncoder()) {
+    // Fallback was successful, so start using it with this frame.
+    return fallback_encoder_->Encode(frame, codec_specific_info, frame_types);
+  }
+  return ret;
+}
+
+int32_t VideoEncoderSoftwareFallbackWrapper::SetChannelParameters(
+    uint32_t packet_loss,
+    int64_t rtt) {
+  channel_parameters_set_ = true;
+  packet_loss_ = packet_loss;
+  rtt_ = rtt;
+  int32_t ret = encoder_->SetChannelParameters(packet_loss, rtt);
+  if (fallback_encoder_)
+    return fallback_encoder_->SetChannelParameters(packet_loss, rtt);
+  return ret;
+}
+
+int32_t VideoEncoderSoftwareFallbackWrapper::SetRates(uint32_t bitrate,
+                                                      uint32_t framerate) {
+  rates_set_ = true;
+  bitrate_ = bitrate;
+  framerate_ = framerate;
+  int32_t ret = encoder_->SetRates(bitrate, framerate);
+  if (fallback_encoder_)
+    return fallback_encoder_->SetRates(bitrate, framerate);
+  return ret;
+}
+
+void VideoEncoderSoftwareFallbackWrapper::OnDroppedFrame() {
+  if (fallback_encoder_)
+    return fallback_encoder_->OnDroppedFrame();
+  return encoder_->OnDroppedFrame();
+}
+
+bool VideoEncoderSoftwareFallbackWrapper::SupportsNativeHandle() const {
+  if (fallback_encoder_)
+    return fallback_encoder_->SupportsNativeHandle();
+  return encoder_->SupportsNativeHandle();
+}
+
+const char* VideoEncoderSoftwareFallbackWrapper::ImplementationName() const {
+  if (fallback_encoder_)
+    return fallback_implementation_name_.c_str();
+  return encoder_->ImplementationName();
+}
+
+int VideoEncoderSoftwareFallbackWrapper::GetTargetFramerate() {
+  if (fallback_encoder_)
+    return fallback_encoder_->GetTargetFramerate();
+  return encoder_->GetTargetFramerate();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_encoder_unittest.cc b/media/webrtc/trunk/webrtc/video/video_encoder_unittest.cc
new file mode 100644
index 0000000000..0f28f89163
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_encoder_unittest.cc
@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_encoder.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
+
+namespace webrtc {
+
+const int kWidth = 320;
+const int kHeight = 240;
+const size_t kMaxPayloadSize = 800;
+
+class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
+ protected:
+  VideoEncoderSoftwareFallbackWrapperTest()
+      : fallback_wrapper_(kVideoCodecVP8, &fake_encoder_) {}
+
+  class CountingFakeEncoder : public VideoEncoder {
+   public:
+    int32_t InitEncode(const VideoCodec* codec_settings,
+                       int32_t number_of_cores,
+                       size_t max_payload_size) override {
+      ++init_encode_count_;
+      return init_encode_return_code_;
+    }
+    int32_t Encode(const VideoFrame& frame,
+                   const CodecSpecificInfo* codec_specific_info,
+                   const std::vector<FrameType>* frame_types) override {
+      ++encode_count_;
+      return encode_return_code_;
+    }
+
+    int32_t RegisterEncodeCompleteCallback(
+        EncodedImageCallback* callback) override {
+      encode_complete_callback_ = callback;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    int32_t Release() override {
+      ++release_count_;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override {
+      ++set_channel_parameters_count_;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    int32_t SetRates(uint32_t bitrate, uint32_t framerate) override {
+      ++set_rates_count_;
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+
+    void OnDroppedFrame() override { ++on_dropped_frame_count_; }
+
+    bool SupportsNativeHandle() const override {
+      ++supports_native_handle_count_;
+      return false;
+    }
+
+    const char* ImplementationName() const override {
+      return "fake-encoder";
+    }
+
+    int init_encode_count_ = 0;
+    int32_t init_encode_return_code_ = WEBRTC_VIDEO_CODEC_OK;
+    int32_t encode_return_code_ = WEBRTC_VIDEO_CODEC_OK;
+    int encode_count_ = 0;
+    EncodedImageCallback* encode_complete_callback_ = nullptr;
+    int release_count_ = 0;
+    int set_channel_parameters_count_ = 0;
+    int set_rates_count_ = 0;
+    int on_dropped_frame_count_ = 0;
+    mutable int supports_native_handle_count_ = 0;
+  };
+
+  class FakeEncodedImageCallback : public EncodedImageCallback {
+   public:
+    int32_t Encoded(const EncodedImage& encoded_image,
+                    const CodecSpecificInfo* codec_specific_info,
+                    const RTPFragmentationHeader* fragmentation) override {
+      return ++callback_count_;
+    }
+    int callback_count_ = 0;
+  };
+
+  void UtilizeFallbackEncoder();
+  void FallbackFromEncodeRequest();
+  void EncodeFrame();
+
+  FakeEncodedImageCallback callback_;
+  CountingFakeEncoder fake_encoder_;
+  VideoEncoderSoftwareFallbackWrapper fallback_wrapper_;
+  VideoCodec codec_ = {};
+  VideoFrame frame_;
+};
+
+void VideoEncoderSoftwareFallbackWrapperTest::EncodeFrame() {
+  frame_.CreateEmptyFrame(kWidth, kHeight, kWidth, (kWidth + 1) / 2,
+                          (kWidth + 1) / 2);
+  memset(frame_.buffer(webrtc::kYPlane), 16,
+         frame_.allocated_size(webrtc::kYPlane));
+  memset(frame_.buffer(webrtc::kUPlane), 128,
+         frame_.allocated_size(webrtc::kUPlane));
+  memset(frame_.buffer(webrtc::kVPlane), 128,
+         frame_.allocated_size(webrtc::kVPlane));
+
+  std::vector<FrameType> types(1, kVideoFrameKey);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            fallback_wrapper_.Encode(frame_, nullptr, &types));
+}
+
+void VideoEncoderSoftwareFallbackWrapperTest::UtilizeFallbackEncoder() {
+  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback_);
+  EXPECT_EQ(&callback_, fake_encoder_.encode_complete_callback_);
+
+  // Register with failing fake encoder. Should succeed with VP8 fallback.
+  codec_.codecType = kVideoCodecVP8;
+  codec_.maxFramerate = 30;
+  codec_.width = kWidth;
+  codec_.height = kHeight;
+  fake_encoder_.init_encode_return_code_ = WEBRTC_VIDEO_CODEC_ERROR;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            fallback_wrapper_.InitEncode(&codec_, 2, kMaxPayloadSize));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.SetRates(300, 30));
+
+  int callback_count = callback_.callback_count_;
+  int encode_count = fake_encoder_.encode_count_;
+  EncodeFrame();
+  EXPECT_EQ(encode_count, fake_encoder_.encode_count_);
+  EXPECT_EQ(callback_count + 1, callback_.callback_count_);
+}
+
+void VideoEncoderSoftwareFallbackWrapperTest::FallbackFromEncodeRequest() {
+  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback_);
+  codec_.codecType = kVideoCodecVP8;
+  codec_.maxFramerate = 30;
+  codec_.width = kWidth;
+  codec_.height = kHeight;
+  fallback_wrapper_.InitEncode(&codec_, 2, kMaxPayloadSize);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.SetRates(300, 30));
+  EXPECT_EQ(1, fake_encoder_.init_encode_count_);
+
+  // Have the non-fallback encoder request a software fallback.
+  fake_encoder_.encode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  int callback_count = callback_.callback_count_;
+  int encode_count = fake_encoder_.encode_count_;
+  EncodeFrame();
+  // Single encode request, which returned failure.
+  EXPECT_EQ(encode_count + 1, fake_encoder_.encode_count_);
+  EXPECT_EQ(callback_count + 1, callback_.callback_count_);
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest, InitializesEncoder) {
+  VideoCodec codec = {};
+  fallback_wrapper_.InitEncode(&codec, 2, kMaxPayloadSize);
+  EXPECT_EQ(1, fake_encoder_.init_encode_count_);
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest, EncodeRequestsFallback) {
+  FallbackFromEncodeRequest();
+  // After fallback, further encodes shouldn't hit the fake encoder.
+  int encode_count = fake_encoder_.encode_count_;
+  EncodeFrame();
+  EXPECT_EQ(encode_count, fake_encoder_.encode_count_);
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest, CanUtilizeFallbackEncoder) {
+  UtilizeFallbackEncoder();
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       InternalEncoderReleasedDuringFallback) {
+  EXPECT_EQ(0, fake_encoder_.release_count_);
+  UtilizeFallbackEncoder();
+  EXPECT_EQ(1, fake_encoder_.release_count_);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+  // No extra release when the fallback is released.
+  EXPECT_EQ(1, fake_encoder_.release_count_);
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       InternalEncoderNotEncodingDuringFallback) {
+  UtilizeFallbackEncoder();
+  int encode_count = fake_encoder_.encode_count_;
+  EncodeFrame();
+  EXPECT_EQ(encode_count, fake_encoder_.encode_count_);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       CanRegisterCallbackWhileUsingFallbackEncoder) {
+  UtilizeFallbackEncoder();
+  // Registering an encode-complete callback should still work when fallback
+  // encoder is being used.
+  FakeEncodedImageCallback callback2;
+  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback2);
+  EXPECT_EQ(&callback2, fake_encoder_.encode_complete_callback_);
+
+  // Encoding a frame using the fallback should arrive at the new callback.
+  std::vector<FrameType> types(1, kVideoFrameKey);
+  frame_.set_timestamp(frame_.timestamp() + 1000);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            fallback_wrapper_.Encode(frame_, nullptr, &types));
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       SetChannelParametersForwardedDuringFallback) {
+  UtilizeFallbackEncoder();
+  EXPECT_EQ(0, fake_encoder_.set_channel_parameters_count_);
+  fallback_wrapper_.SetChannelParameters(1, 1);
+  EXPECT_EQ(1, fake_encoder_.set_channel_parameters_count_);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       SetRatesForwardedDuringFallback) {
+  UtilizeFallbackEncoder();
+  EXPECT_EQ(1, fake_encoder_.set_rates_count_);
+  fallback_wrapper_.SetRates(1, 1);
+  EXPECT_EQ(2, fake_encoder_.set_rates_count_);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       OnDroppedFrameForwardedWithoutFallback) {
+  fallback_wrapper_.OnDroppedFrame();
+  EXPECT_EQ(1, fake_encoder_.on_dropped_frame_count_);
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       OnDroppedFrameNotForwardedDuringFallback) {
+  UtilizeFallbackEncoder();
+  fallback_wrapper_.OnDroppedFrame();
+  EXPECT_EQ(0, fake_encoder_.on_dropped_frame_count_);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       SupportsNativeHandleForwardedWithoutFallback) {
+  fallback_wrapper_.SupportsNativeHandle();
+  EXPECT_EQ(1, fake_encoder_.supports_native_handle_count_);
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       SupportsNativeHandleNotForwardedDuringFallback) {
+  UtilizeFallbackEncoder();
+  fallback_wrapper_.SupportsNativeHandle();
+  EXPECT_EQ(0, fake_encoder_.supports_native_handle_count_);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
+}
+
+TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
+       ReportsFallbackImplementationName) {
+  UtilizeFallbackEncoder();
+  // Hard coded expected value since libvpx is the software implementation name
+  // for VP8. Change accordingly if the underlying implementation does.
+  EXPECT_STREQ("libvpx (fallback from: fake-encoder)",
+               fallback_wrapper_.ImplementationName());
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_loopback.cc b/media/webrtc/trunk/webrtc/video/video_loopback.cc
index d183920b16..2338a84a43 100644
--- a/media/webrtc/trunk/webrtc/video/video_loopback.cc
+++ b/media/webrtc/trunk/webrtc/video/video_loopback.cc
@@ -10,20 +10,17 @@
 
 #include <stdio.h>
 
-#include <map>
-
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/test/field_trial.h"
 #include "webrtc/test/run_test.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video/loopback.h"
+#include "webrtc/video/video_quality_test.h"
 
 namespace webrtc {
-
 namespace flags {
 
+// Flags common with screenshare loopback, with different default values.
 DEFINE_int32(width, 640, "Video width.");
 size_t Width() {
   return static_cast<size_t>(FLAGS_width);
@@ -39,30 +36,66 @@ int Fps() {
   return static_cast<int>(FLAGS_fps);
 }
 
-DEFINE_int32(min_bitrate, 50, "Minimum video bitrate.");
-size_t MinBitrate() {
-  return static_cast<size_t>(FLAGS_min_bitrate);
+DEFINE_int32(min_bitrate, 50, "Call and stream min bitrate in kbps.");
+int MinBitrateKbps() {
+  return static_cast<int>(FLAGS_min_bitrate);
 }
 
-DEFINE_int32(start_bitrate, 300, "Video starting bitrate.");
-size_t StartBitrate() {
-  return static_cast<size_t>(FLAGS_start_bitrate);
+DEFINE_int32(start_bitrate, 300, "Call start bitrate in kbps.");
+int StartBitrateKbps() {
+  return static_cast<int>(FLAGS_start_bitrate);
 }
 
-DEFINE_int32(max_bitrate, 800, "Maximum video bitrate.");
-size_t MaxBitrate() {
-  return static_cast<size_t>(FLAGS_max_bitrate);
+DEFINE_int32(target_bitrate, 800, "Stream target bitrate in kbps.");
+int TargetBitrateKbps() {
+  return static_cast<int>(FLAGS_target_bitrate);
 }
 
-int MinTransmitBitrate() {
-  return 0;
-}  // No min padding for regular video.
+DEFINE_int32(max_bitrate, 800, "Call and stream max bitrate in kbps.");
+int MaxBitrateKbps() {
+  return static_cast<int>(FLAGS_max_bitrate);
+}
 
+DEFINE_int32(num_temporal_layers,
+             1,
+             "Number of temporal layers. Set to 1-4 to override.");
+int NumTemporalLayers() {
+  return static_cast<int>(FLAGS_num_temporal_layers);
+}
+
+// Flags common with screenshare loopback, with equal default values.
 DEFINE_string(codec, "VP8", "Video codec to use.");
 std::string Codec() {
   return static_cast<std::string>(FLAGS_codec);
 }
 
+DEFINE_int32(selected_tl,
+             -1,
+             "Temporal layer to show or analyze. -1 to disable filtering.");
+int SelectedTL() {
+  return static_cast<int>(FLAGS_selected_tl);
+}
+
+DEFINE_int32(
+    duration,
+    0,
+    "Duration of the test in seconds. If 0, rendered will be shown instead.");
+int DurationSecs() {
+  return static_cast<int>(FLAGS_duration);
+}
+
+DEFINE_string(output_filename, "", "Target graph data filename.");
+std::string OutputFilename() {
+  return static_cast<std::string>(FLAGS_output_filename);
+}
+
+DEFINE_string(graph_title,
+              "",
+              "If empty, title will be generated automatically.");
+std::string GraphTitle() {
+  return static_cast<std::string>(FLAGS_graph_title);
+}
+
 DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
 int LossPercent() {
   return static_cast<int>(FLAGS_loss_percent);
@@ -71,7 +104,7 @@ int LossPercent() {
 DEFINE_int32(link_capacity,
              0,
              "Capacity (kbps) of the fake link. 0 means infinite.");
-int LinkCapacity() {
+int LinkCapacityKbps() {
   return static_cast<int>(FLAGS_link_capacity);
 }
 
@@ -94,8 +127,55 @@ int StdPropagationDelayMs() {
   return static_cast<int>(FLAGS_std_propagation_delay_ms);
 }
 
+DEFINE_int32(selected_stream, 0, "ID of the stream to show or analyze.");
+int SelectedStream() {
+  return static_cast<int>(FLAGS_selected_stream);
+}
+
+DEFINE_int32(num_spatial_layers, 1, "Number of spatial layers to use.");
+int NumSpatialLayers() {
+  return static_cast<int>(FLAGS_num_spatial_layers);
+}
+
+DEFINE_int32(selected_sl,
+             -1,
+             "Spatial layer to show or analyze. -1 to disable filtering.");
+int SelectedSL() {
+  return static_cast<int>(FLAGS_selected_sl);
+}
+
+DEFINE_string(stream0,
+              "",
+              "Comma separated values describing VideoStream for stream #0.");
+std::string Stream0() {
+  return static_cast<std::string>(FLAGS_stream0);
+}
+
+DEFINE_string(stream1,
+              "",
+              "Comma separated values describing VideoStream for stream #1.");
+std::string Stream1() {
+  return static_cast<std::string>(FLAGS_stream1);
+}
+
+DEFINE_string(sl0,
+              "",
+              "Comma separated values describing SpatialLayer for layer #0.");
+std::string SL0() {
+  return static_cast<std::string>(FLAGS_sl0);
+}
+
+DEFINE_string(sl1,
+              "",
+              "Comma separated values describing SpatialLayer for layer #1.");
+std::string SL1() {
+  return static_cast<std::string>(FLAGS_sl1);
+}
+
 DEFINE_bool(logs, false, "print logs to stderr");
 
+DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
+
 DEFINE_string(
     force_fieldtrials,
     "",
@@ -103,25 +183,60 @@ DEFINE_string(
     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
     " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
     "trials are separated by \"/\"");
+
+// Video-specific flags.
+DEFINE_string(clip,
+              "",
+              "Name of the clip to show. If empty, using chroma generator.");
+std::string Clip() {
+  return static_cast<std::string>(FLAGS_clip);
+}
+
 }  // namespace flags
 
 void Loopback() {
-  test::Loopback::Config config{flags::Width(),
-                                flags::Height(),
-                                flags::Fps(),
-                                flags::MinBitrate(),
-                                flags::StartBitrate(),
-                                flags::MaxBitrate(),
-                                0,  // No min transmit bitrate.
-                                flags::Codec(),
-                                flags::LossPercent(),
-                                flags::LinkCapacity(),
-                                flags::QueueSize(),
-                                flags::AvgPropagationDelayMs(),
-                                flags::StdPropagationDelayMs(),
-                                flags::FLAGS_logs};
-  test::Loopback loopback(config);
-  loopback.Run();
+  FakeNetworkPipe::Config pipe_config;
+  pipe_config.loss_percent = flags::LossPercent();
+  pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
+  pipe_config.queue_length_packets = flags::QueueSize();
+  pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
+  pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
+
+  Call::Config::BitrateConfig call_bitrate_config;
+  call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
+  call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
+  call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
+
+  VideoQualityTest::Params params{
+      {flags::Width(), flags::Height(), flags::Fps(),
+       flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
+       flags::MaxBitrateKbps() * 1000, flags::Codec(),
+       flags::NumTemporalLayers(), flags::SelectedTL(),
+       0,  // No min transmit bitrate.
+       call_bitrate_config, flags::FLAGS_send_side_bwe},
+      {flags::Clip()},
+      {},  // Screenshare specific.
+      {"video", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
+       flags::GraphTitle()},
+      pipe_config,
+      flags::FLAGS_logs};
+
+  std::vector<std::string> stream_descriptors;
+  stream_descriptors.push_back(flags::Stream0());
+  stream_descriptors.push_back(flags::Stream1());
+  std::vector<std::string> SL_descriptors;
+  SL_descriptors.push_back(flags::SL0());
+  SL_descriptors.push_back(flags::SL1());
+  VideoQualityTest::FillScalabilitySettings(
+      &params, stream_descriptors, flags::SelectedStream(),
+      flags::NumSpatialLayers(), flags::SelectedSL(), SL_descriptors);
+
+  VideoQualityTest test;
+  if (flags::DurationSecs()) {
+    test.RunWithAnalyzer(params);
+  } else {
+    test.RunWithVideoRenderer(params);
+  }
 }
 }  // namespace webrtc
 
diff --git a/media/webrtc/trunk/webrtc/video/video_quality_test.cc b/media/webrtc/trunk/webrtc/video/video_quality_test.cc
new file mode 100644
index 0000000000..08ae0a9cee
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_quality_test.cc
@@ -0,0 +1,1073 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <stdio.h>
+
+#include <algorithm>
+#include <deque>
+#include <map>
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/include/cpu_info.h"
+#include "webrtc/test/layer_filtering_transport.h"
+#include "webrtc/test/run_loop.h"
+#include "webrtc/test/statistics.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/video_renderer.h"
+#include "webrtc/video/video_quality_test.h"
+
+namespace webrtc {
+
+static const int kSendStatsPollingIntervalMs = 1000;
+static const int kPayloadTypeVP8 = 123;
+static const int kPayloadTypeVP9 = 124;
+
+class VideoAnalyzer : public PacketReceiver,
+                      public Transport,
+                      public VideoRenderer,
+                      public VideoCaptureInput,
+                      public EncodedFrameObserver,
+                      public EncodingTimeObserver {
+ public:
+  VideoAnalyzer(test::LayerFilteringTransport* transport,
+                const std::string& test_label,
+                double avg_psnr_threshold,
+                double avg_ssim_threshold,
+                int duration_frames,
+                FILE* graph_data_output_file,
+                const std::string& graph_title,
+                uint32_t ssrc_to_analyze)
+      : input_(nullptr),
+        transport_(transport),
+        receiver_(nullptr),
+        send_stream_(nullptr),
+        test_label_(test_label),
+        graph_data_output_file_(graph_data_output_file),
+        graph_title_(graph_title),
+        ssrc_to_analyze_(ssrc_to_analyze),
+        frames_to_process_(duration_frames),
+        frames_recorded_(0),
+        frames_processed_(0),
+        dropped_frames_(0),
+        last_render_time_(0),
+        rtp_timestamp_delta_(0),
+        avg_psnr_threshold_(avg_psnr_threshold),
+        avg_ssim_threshold_(avg_ssim_threshold),
+        stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
+        comparison_available_event_(false, false),
+        done_(false, false) {
+    // Create thread pool for CPU-expensive PSNR/SSIM calculations.
+
+    // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
+    // so that we don't accidentally starve "real" worker threads (codec etc).
+    // Also, don't allocate more than kMaxComparisonThreads, even if there are
+    // spare cores.
+
+    uint32_t num_cores = CpuInfo::DetectNumberOfCores();
+    RTC_DCHECK_GE(num_cores, 1u);
+    static const uint32_t kMinCoresLeft = 4;
+    static const uint32_t kMaxComparisonThreads = 8;
+
+    if (num_cores <= kMinCoresLeft) {
+      num_cores = 1;
+    } else {
+      num_cores -= kMinCoresLeft;
+      num_cores = std::min(num_cores, kMaxComparisonThreads);
+    }
+
+    for (uint32_t i = 0; i < num_cores; ++i) {
+      rtc::PlatformThread* thread =
+          new rtc::PlatformThread(&FrameComparisonThread, this, "Analyzer");
+      thread->Start();
+      comparison_thread_pool_.push_back(thread);
+    }
+  }
+
+  ~VideoAnalyzer() {
+    for (rtc::PlatformThread* thread : comparison_thread_pool_) {
+      thread->Stop();
+      delete thread;
+    }
+  }
+
+  virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
+
+  DeliveryStatus DeliverPacket(MediaType media_type,
+                               const uint8_t* packet,
+                               size_t length,
+                               const PacketTime& packet_time) override {
+    RtpUtility::RtpHeaderParser parser(packet, length);
+    RTPHeader header;
+    parser.Parse(&header);
+    {
+      rtc::CritScope lock(&crit_);
+      recv_times_[header.timestamp - rtp_timestamp_delta_] =
+          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
+    }
+
+    return receiver_->DeliverPacket(media_type, packet, length, packet_time);
+  }
+
+  // EncodingTimeObserver.
+  void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) override {
+    rtc::CritScope crit(&comparison_lock_);
+    samples_encode_time_ms_[ntp_time_ms] = encode_time_ms;
+  }
+
+  void IncomingCapturedFrame(const VideoFrame& video_frame) override {
+    VideoFrame copy = video_frame;
+    copy.set_timestamp(copy.ntp_time_ms() * 90);
+
+    {
+      rtc::CritScope lock(&crit_);
+      if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
+        first_send_frame_ = copy;
+
+      frames_.push_back(copy);
+    }
+
+    input_->IncomingCapturedFrame(video_frame);
+  }
+
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override {
+    RtpUtility::RtpHeaderParser parser(packet, length);
+    RTPHeader header;
+    parser.Parse(&header);
+
+    int64_t current_time =
+        Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
+    bool result = transport_->SendRtp(packet, length, options);
+    {
+      rtc::CritScope lock(&crit_);
+      if (rtp_timestamp_delta_ == 0) {
+        rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
+        first_send_frame_.Reset();
+      }
+      uint32_t timestamp = header.timestamp - rtp_timestamp_delta_;
+      send_times_[timestamp] = current_time;
+      if (!transport_->DiscardedLastPacket() &&
+          header.ssrc == ssrc_to_analyze_) {
+        encoded_frame_sizes_[timestamp] +=
+            length - (header.headerLength + header.paddingLength);
+      }
+    }
+    return result;
+  }
+
+  bool SendRtcp(const uint8_t* packet, size_t length) override {
+    return transport_->SendRtcp(packet, length);
+  }
+
+  void EncodedFrameCallback(const EncodedFrame& frame) override {
+    rtc::CritScope lock(&comparison_lock_);
+    if (frames_recorded_ < frames_to_process_)
+      encoded_frame_size_.AddSample(frame.length_);
+  }
+
+  void RenderFrame(const VideoFrame& video_frame,
+                   int time_to_render_ms) override {
+    int64_t render_time_ms =
+        Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
+    uint32_t send_timestamp = video_frame.timestamp() - rtp_timestamp_delta_;
+
+    rtc::CritScope lock(&crit_);
+
+    while (frames_.front().timestamp() < send_timestamp) {
+      AddFrameComparison(frames_.front(), last_rendered_frame_, true,
+                         render_time_ms);
+      frames_.pop_front();
+    }
+
+    VideoFrame reference_frame = frames_.front();
+    frames_.pop_front();
+    assert(!reference_frame.IsZeroSize());
+    if (send_timestamp == reference_frame.timestamp() - 1) {
+      // TODO(ivica): Make this work for > 2 streams.
+      // Look at rtp_sender.c:RTPSender::BuildRTPHeader.
+      ++send_timestamp;
+    }
+    EXPECT_EQ(reference_frame.timestamp(), send_timestamp);
+    assert(reference_frame.timestamp() == send_timestamp);
+
+    AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
+
+    last_rendered_frame_ = video_frame;
+  }
+
+  bool IsTextureSupported() const override { return false; }
+
+  void Wait() {
+    // Frame comparisons can be very expensive. Wait for test to be done, but
+    // at time-out check if frames_processed is going up. If so, give it more
+    // time, otherwise fail. Hopefully this will reduce test flakiness.
+
+    stats_polling_thread_.Start();
+
+    int last_frames_processed = -1;
+    int iteration = 0;
+    while (!done_.Wait(VideoQualityTest::kDefaultTimeoutMs)) {
+      int frames_processed;
+      {
+        rtc::CritScope crit(&comparison_lock_);
+        frames_processed = frames_processed_;
+      }
+
+      // Print some output so test infrastructure won't think we've crashed.
+      const char* kKeepAliveMessages[3] = {
+          "Uh, I'm-I'm not quite dead, sir.",
+          "Uh, I-I think uh, I could pull through, sir.",
+          "Actually, I think I'm all right to come with you--"};
+      printf("- %s\n", kKeepAliveMessages[iteration++ % 3]);
+
+      if (last_frames_processed == -1) {
+        last_frames_processed = frames_processed;
+        continue;
+      }
+      ASSERT_GT(frames_processed, last_frames_processed)
+          << "Analyzer stalled while waiting for test to finish.";
+      last_frames_processed = frames_processed;
+    }
+
+    if (iteration > 0)
+      printf("- Farewell, sweet Concorde!\n");
+
+    // Signal stats polling thread if that is still waiting and stop it now,
+    // since it uses the send_stream_ reference that might be reclaimed after
+    // returning from this method.
+    done_.Set();
+    stats_polling_thread_.Stop();
+  }
+
+  VideoCaptureInput* input_;
+  test::LayerFilteringTransport* const transport_;
+  PacketReceiver* receiver_;
+  VideoSendStream* send_stream_;
+
+ private:
+  struct FrameComparison {
+    FrameComparison()
+        : dropped(false),
+          send_time_ms(0),
+          recv_time_ms(0),
+          render_time_ms(0),
+          encoded_frame_size(0) {}
+
+    FrameComparison(const VideoFrame& reference,
+                    const VideoFrame& render,
+                    bool dropped,
+                    int64_t send_time_ms,
+                    int64_t recv_time_ms,
+                    int64_t render_time_ms,
+                    size_t encoded_frame_size)
+        : reference(reference),
+          render(render),
+          dropped(dropped),
+          send_time_ms(send_time_ms),
+          recv_time_ms(recv_time_ms),
+          render_time_ms(render_time_ms),
+          encoded_frame_size(encoded_frame_size) {}
+
+    VideoFrame reference;
+    VideoFrame render;
+    bool dropped;
+    int64_t send_time_ms;
+    int64_t recv_time_ms;
+    int64_t render_time_ms;
+    size_t encoded_frame_size;
+  };
+
+  struct Sample {
+    Sample(int dropped,
+           int64_t input_time_ms,
+           int64_t send_time_ms,
+           int64_t recv_time_ms,
+           int64_t render_time_ms,
+           size_t encoded_frame_size,
+           double psnr,
+           double ssim)
+        : dropped(dropped),
+          input_time_ms(input_time_ms),
+          send_time_ms(send_time_ms),
+          recv_time_ms(recv_time_ms),
+          render_time_ms(render_time_ms),
+          encoded_frame_size(encoded_frame_size),
+          psnr(psnr),
+          ssim(ssim) {}
+
+    int dropped;
+    int64_t input_time_ms;
+    int64_t send_time_ms;
+    int64_t recv_time_ms;
+    int64_t render_time_ms;
+    size_t encoded_frame_size;
+    double psnr;
+    double ssim;
+  };
+
+  void AddFrameComparison(const VideoFrame& reference,
+                          const VideoFrame& render,
+                          bool dropped,
+                          int64_t render_time_ms)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+    int64_t send_time_ms = send_times_[reference.timestamp()];
+    send_times_.erase(reference.timestamp());
+    int64_t recv_time_ms = recv_times_[reference.timestamp()];
+    recv_times_.erase(reference.timestamp());
+
+    // TODO(ivica): Make this work for > 2 streams.
+    auto it = encoded_frame_sizes_.find(reference.timestamp());
+    if (it == encoded_frame_sizes_.end())
+      it = encoded_frame_sizes_.find(reference.timestamp() - 1);
+    size_t encoded_size = it == encoded_frame_sizes_.end() ? 0 : it->second;
+    if (it != encoded_frame_sizes_.end())
+      encoded_frame_sizes_.erase(it);
+
+    VideoFrame reference_copy;
+    VideoFrame render_copy;
+    reference_copy.CopyFrame(reference);
+    render_copy.CopyFrame(render);
+
+    rtc::CritScope crit(&comparison_lock_);
+    comparisons_.push_back(FrameComparison(reference_copy, render_copy, dropped,
+                                           send_time_ms, recv_time_ms,
+                                           render_time_ms, encoded_size));
+    comparison_available_event_.Set();
+  }
+
+  static bool PollStatsThread(void* obj) {
+    return static_cast<VideoAnalyzer*>(obj)->PollStats();
+  }
+
+  bool PollStats() {
+    if (done_.Wait(kSendStatsPollingIntervalMs)) {
+      // Set event again to make sure main thread is also signaled, then we're
+      // done.
+      done_.Set();
+      return false;
+    }
+
+    VideoSendStream::Stats stats = send_stream_->GetStats();
+
+    rtc::CritScope crit(&comparison_lock_);
+    encode_frame_rate_.AddSample(stats.encode_frame_rate);
+    encode_time_ms.AddSample(stats.avg_encode_time_ms);
+    encode_usage_percent.AddSample(stats.encode_usage_percent);
+    media_bitrate_bps.AddSample(stats.media_bitrate_bps);
+
+    return true;
+  }
+
+  static bool FrameComparisonThread(void* obj) {
+    return static_cast<VideoAnalyzer*>(obj)->CompareFrames();
+  }
+
+  bool CompareFrames() {
+    if (AllFramesRecorded())
+      return false;
+
+    VideoFrame reference;
+    VideoFrame render;
+    FrameComparison comparison;
+
+    if (!PopComparison(&comparison)) {
+      // Wait until new comparison task is available, or test is done.
+      // If done, wake up remaining threads waiting.
+      comparison_available_event_.Wait(1000);
+      if (AllFramesRecorded()) {
+        comparison_available_event_.Set();
+        return false;
+      }
+      return true;  // Try again.
+    }
+
+    PerformFrameComparison(comparison);
+
+    if (FrameProcessed()) {
+      PrintResults();
+      if (graph_data_output_file_)
+        PrintSamplesToFile();
+      done_.Set();
+      comparison_available_event_.Set();
+      return false;
+    }
+
+    return true;
+  }
+
+  bool PopComparison(FrameComparison* comparison) {
+    rtc::CritScope crit(&comparison_lock_);
+    // If AllFramesRecorded() is true, it means we have already popped
+    // frames_to_process_ frames from comparisons_, so there is no more work
+    // for this thread to be done. frames_processed_ might still be lower if
+    // all comparisons are not done, but those frames are currently being
+    // worked on by other threads.
+    if (comparisons_.empty() || AllFramesRecorded())
+      return false;
+
+    *comparison = comparisons_.front();
+    comparisons_.pop_front();
+
+    FrameRecorded();
+    return true;
+  }
+
+  // Increment counter for number of frames received for comparison.
+  void FrameRecorded() {
+    rtc::CritScope crit(&comparison_lock_);
+    ++frames_recorded_;
+  }
+
+  // Returns true if all frames to be compared have been taken from the queue.
+  bool AllFramesRecorded() {
+    rtc::CritScope crit(&comparison_lock_);
+    assert(frames_recorded_ <= frames_to_process_);
+    return frames_recorded_ == frames_to_process_;
+  }
+
+  // Increase count of number of frames processed. Returns true if this was the
+  // last frame to be processed.
+  bool FrameProcessed() {
+    rtc::CritScope crit(&comparison_lock_);
+    ++frames_processed_;
+    assert(frames_processed_ <= frames_to_process_);
+    return frames_processed_ == frames_to_process_;
+  }
+
+  void PrintResults() {
+    rtc::CritScope crit(&comparison_lock_);
+    PrintResult("psnr", psnr_, " dB");
+    PrintResult("ssim", ssim_, "");
+    PrintResult("sender_time", sender_time_, " ms");
+    printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(),
+           dropped_frames_);
+    PrintResult("receiver_time", receiver_time_, " ms");
+    PrintResult("total_delay_incl_network", end_to_end_, " ms");
+    PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
+    PrintResult("encoded_frame_size", encoded_frame_size_, " bytes");
+    PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
+    PrintResult("encode_time", encode_time_ms, " ms");
+    PrintResult("encode_usage_percent", encode_usage_percent, " percent");
+    PrintResult("media_bitrate", media_bitrate_bps, " bps");
+
+    EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
+    EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
+  }
+
+  void PerformFrameComparison(const FrameComparison& comparison) {
+    // Perform expensive psnr and ssim calculations while not holding lock.
+    double psnr = I420PSNR(&comparison.reference, &comparison.render);
+    double ssim = I420SSIM(&comparison.reference, &comparison.render);
+
+    int64_t input_time_ms = comparison.reference.ntp_time_ms();
+
+    rtc::CritScope crit(&comparison_lock_);
+    if (graph_data_output_file_) {
+      samples_.push_back(
+          Sample(comparison.dropped, input_time_ms, comparison.send_time_ms,
+                 comparison.recv_time_ms, comparison.render_time_ms,
+                 comparison.encoded_frame_size, psnr, ssim));
+    }
+    psnr_.AddSample(psnr);
+    ssim_.AddSample(ssim);
+
+    if (comparison.dropped) {
+      ++dropped_frames_;
+      return;
+    }
+    if (last_render_time_ != 0)
+      rendered_delta_.AddSample(comparison.render_time_ms - last_render_time_);
+    last_render_time_ = comparison.render_time_ms;
+
+    sender_time_.AddSample(comparison.send_time_ms - input_time_ms);
+    receiver_time_.AddSample(comparison.render_time_ms -
+                             comparison.recv_time_ms);
+    end_to_end_.AddSample(comparison.render_time_ms - input_time_ms);
+    encoded_frame_size_.AddSample(comparison.encoded_frame_size);
+  }
+
+  void PrintResult(const char* result_type,
+                   test::Statistics stats,
+                   const char* unit) {
+    printf("RESULT %s: %s = {%f, %f}%s\n",
+           result_type,
+           test_label_.c_str(),
+           stats.Mean(),
+           stats.StandardDeviation(),
+           unit);
+  }
+
+  void PrintSamplesToFile(void) {
+    FILE* out = graph_data_output_file_;
+    rtc::CritScope crit(&comparison_lock_);
+    std::sort(samples_.begin(), samples_.end(),
+              [](const Sample& A, const Sample& B) -> bool {
+                return A.input_time_ms < B.input_time_ms;
+              });
+
+    fprintf(out, "%s\n", graph_title_.c_str());
+    fprintf(out, "%" PRIuS "\n", samples_.size());
+    fprintf(out,
+            "dropped "
+            "input_time_ms "
+            "send_time_ms "
+            "recv_time_ms "
+            "render_time_ms "
+            "encoded_frame_size "
+            "psnr "
+            "ssim "
+            "encode_time_ms\n");
+    int missing_encode_time_samples = 0;
+    for (const Sample& sample : samples_) {
+      auto it = samples_encode_time_ms_.find(sample.input_time_ms);
+      int encode_time_ms;
+      if (it != samples_encode_time_ms_.end()) {
+        encode_time_ms = it->second;
+      } else {
+        ++missing_encode_time_samples;
+        encode_time_ms = -1;
+      }
+      fprintf(out, "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" PRIuS
+                   " %lf %lf %d\n",
+              sample.dropped, sample.input_time_ms, sample.send_time_ms,
+              sample.recv_time_ms, sample.render_time_ms,
+              sample.encoded_frame_size, sample.psnr, sample.ssim,
+              encode_time_ms);
+    }
+    if (missing_encode_time_samples) {
+      fprintf(stderr,
+              "Warning: Missing encode_time_ms samples for %d frame(s).\n",
+              missing_encode_time_samples);
+    }
+  }
+
+  const std::string test_label_;
+  FILE* const graph_data_output_file_;
+  const std::string graph_title_;
+  const uint32_t ssrc_to_analyze_;
+  std::vector<Sample> samples_ GUARDED_BY(comparison_lock_);
+  std::map<int64_t, int> samples_encode_time_ms_ GUARDED_BY(comparison_lock_);
+  test::Statistics sender_time_ GUARDED_BY(comparison_lock_);
+  test::Statistics receiver_time_ GUARDED_BY(comparison_lock_);
+  test::Statistics psnr_ GUARDED_BY(comparison_lock_);
+  test::Statistics ssim_ GUARDED_BY(comparison_lock_);
+  test::Statistics end_to_end_ GUARDED_BY(comparison_lock_);
+  test::Statistics rendered_delta_ GUARDED_BY(comparison_lock_);
+  test::Statistics encoded_frame_size_ GUARDED_BY(comparison_lock_);
+  test::Statistics encode_frame_rate_ GUARDED_BY(comparison_lock_);
+  test::Statistics encode_time_ms GUARDED_BY(comparison_lock_);
+  test::Statistics encode_usage_percent GUARDED_BY(comparison_lock_);
+  test::Statistics media_bitrate_bps GUARDED_BY(comparison_lock_);
+
+  const int frames_to_process_;
+  int frames_recorded_;
+  int frames_processed_;
+  int dropped_frames_;
+  int64_t last_render_time_;
+  uint32_t rtp_timestamp_delta_;
+
+  rtc::CriticalSection crit_;
+  std::deque<VideoFrame> frames_ GUARDED_BY(crit_);
+  VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
+  std::map<uint32_t, int64_t> send_times_ GUARDED_BY(crit_);
+  std::map<uint32_t, int64_t> recv_times_ GUARDED_BY(crit_);
+  std::map<uint32_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
+  VideoFrame first_send_frame_ GUARDED_BY(crit_);
+  const double avg_psnr_threshold_;
+  const double avg_ssim_threshold_;
+
+  rtc::CriticalSection comparison_lock_;
+  std::vector<rtc::PlatformThread*> comparison_thread_pool_;
+  rtc::PlatformThread stats_polling_thread_;
+  rtc::Event comparison_available_event_;
+  std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
+  rtc::Event done_;
+};
+
+VideoQualityTest::VideoQualityTest() : clock_(Clock::GetRealTimeClock()) {}
+
+void VideoQualityTest::TestBody() {}
+
+std::string VideoQualityTest::GenerateGraphTitle() const {
+  std::stringstream ss;
+  ss << params_.common.codec;
+  ss << " (" << params_.common.target_bitrate_bps / 1000 << "kbps";
+  ss << ", " << params_.common.fps << " FPS";
+  if (params_.screenshare.scroll_duration)
+    ss << ", " << params_.screenshare.scroll_duration << "s scroll";
+  if (params_.ss.streams.size() > 1)
+    ss << ", Stream #" << params_.ss.selected_stream;
+  if (params_.ss.num_spatial_layers > 1)
+    ss << ", Layer #" << params_.ss.selected_sl;
+  ss << ")";
+  return ss.str();
+}
+
+void VideoQualityTest::CheckParams() {
+  // Add a default stream in none specified.
+  if (params_.ss.streams.empty())
+    params_.ss.streams.push_back(VideoQualityTest::DefaultVideoStream(params_));
+  if (params_.ss.num_spatial_layers == 0)
+    params_.ss.num_spatial_layers = 1;
+
+  if (params_.pipe.loss_percent != 0 ||
+      params_.pipe.queue_length_packets != 0) {
+    // Since LayerFilteringTransport changes the sequence numbers, we can't
+    // use that feature with pack loss, since the NACK request would end up
+    // retransmitting the wrong packets.
+    RTC_CHECK(params_.ss.selected_sl == -1 ||
+              params_.ss.selected_sl == params_.ss.num_spatial_layers - 1);
+    RTC_CHECK(params_.common.selected_tl == -1 ||
+              params_.common.selected_tl ==
+                  params_.common.num_temporal_layers - 1);
+  }
+
+  // TODO(ivica): Should max_bitrate_bps == -1 represent inf max bitrate, as it
+  // does in some parts of the code?
+  RTC_CHECK_GE(params_.common.max_bitrate_bps,
+               params_.common.target_bitrate_bps);
+  RTC_CHECK_GE(params_.common.target_bitrate_bps,
+               params_.common.min_bitrate_bps);
+  RTC_CHECK_LT(params_.common.selected_tl, params_.common.num_temporal_layers);
+  RTC_CHECK_LT(params_.ss.selected_stream, params_.ss.streams.size());
+  for (const VideoStream& stream : params_.ss.streams) {
+    RTC_CHECK_GE(stream.min_bitrate_bps, 0);
+    RTC_CHECK_GE(stream.target_bitrate_bps, stream.min_bitrate_bps);
+    RTC_CHECK_GE(stream.max_bitrate_bps, stream.target_bitrate_bps);
+    RTC_CHECK_EQ(static_cast<int>(stream.temporal_layer_thresholds_bps.size()),
+                 params_.common.num_temporal_layers - 1);
+  }
+  // TODO(ivica): Should we check if the sum of all streams/layers is equal to
+  // the total bitrate? We anyway have to update them in the case bitrate
+  // estimator changes the total bitrates.
+  RTC_CHECK_GE(params_.ss.num_spatial_layers, 1);
+  RTC_CHECK_LE(params_.ss.selected_sl, params_.ss.num_spatial_layers);
+  RTC_CHECK(params_.ss.spatial_layers.empty() ||
+            params_.ss.spatial_layers.size() ==
+                static_cast<size_t>(params_.ss.num_spatial_layers));
+  if (params_.common.codec == "VP8") {
+    RTC_CHECK_EQ(params_.ss.num_spatial_layers, 1);
+  } else if (params_.common.codec == "VP9") {
+    RTC_CHECK_EQ(params_.ss.streams.size(), 1u);
+  }
+}
+
+// Static.
+std::vector<int> VideoQualityTest::ParseCSV(const std::string& str) {
+  // Parse comma separated nonnegative integers, where some elements may be
+  // empty. The empty values are replaced with -1.
+  // E.g. "10,-20,,30,40" --> {10, 20, -1, 30,40}
+  // E.g. ",,10,,20," --> {-1, -1, 10, -1, 20, -1}
+  std::vector<int> result;
+  if (str.empty())
+    return result;
+
+  const char* p = str.c_str();
+  int value = -1;
+  int pos;
+  while (*p) {
+    if (*p == ',') {
+      result.push_back(value);
+      value = -1;
+      ++p;
+      continue;
+    }
+    RTC_CHECK_EQ(sscanf(p, "%d%n", &value, &pos), 1)
+        << "Unexpected non-number value.";
+    p += pos;
+  }
+  result.push_back(value);
+  return result;
+}
+
+// Static.
+VideoStream VideoQualityTest::DefaultVideoStream(const Params& params) {
+  VideoStream stream;
+  stream.width = params.common.width;
+  stream.height = params.common.height;
+  stream.max_framerate = params.common.fps;
+  stream.min_bitrate_bps = params.common.min_bitrate_bps;
+  stream.target_bitrate_bps = params.common.target_bitrate_bps;
+  stream.max_bitrate_bps = params.common.max_bitrate_bps;
+  stream.max_qp = 52;
+  if (params.common.num_temporal_layers == 2)
+    stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
+  return stream;
+}
+
+// Static.
+void VideoQualityTest::FillScalabilitySettings(
+    Params* params,
+    const std::vector<std::string>& stream_descriptors,
+    size_t selected_stream,
+    int num_spatial_layers,
+    int selected_sl,
+    const std::vector<std::string>& sl_descriptors) {
+  // Read VideoStream and SpatialLayer elements from a list of comma separated
+  // lists. To use a default value for an element, use -1 or leave empty.
+  // Validity checks performed in CheckParams.
+
+  RTC_CHECK(params->ss.streams.empty());
+  for (auto descriptor : stream_descriptors) {
+    if (descriptor.empty())
+      continue;
+    VideoStream stream = VideoQualityTest::DefaultVideoStream(*params);
+    std::vector<int> v = VideoQualityTest::ParseCSV(descriptor);
+    if (v[0] != -1)
+      stream.width = static_cast<size_t>(v[0]);
+    if (v[1] != -1)
+      stream.height = static_cast<size_t>(v[1]);
+    if (v[2] != -1)
+      stream.max_framerate = v[2];
+    if (v[3] != -1)
+      stream.min_bitrate_bps = v[3];
+    if (v[4] != -1)
+      stream.target_bitrate_bps = v[4];
+    if (v[5] != -1)
+      stream.max_bitrate_bps = v[5];
+    if (v.size() > 6 && v[6] != -1)
+      stream.max_qp = v[6];
+    if (v.size() > 7) {
+      stream.temporal_layer_thresholds_bps.clear();
+      stream.temporal_layer_thresholds_bps.insert(
+          stream.temporal_layer_thresholds_bps.end(), v.begin() + 7, v.end());
+    } else {
+      // Automatic TL thresholds for more than two layers not supported.
+      RTC_CHECK_LE(params->common.num_temporal_layers, 2);
+    }
+    params->ss.streams.push_back(stream);
+  }
+  params->ss.selected_stream = selected_stream;
+
+  params->ss.num_spatial_layers = num_spatial_layers ? num_spatial_layers : 1;
+  params->ss.selected_sl = selected_sl;
+  RTC_CHECK(params->ss.spatial_layers.empty());
+  for (auto descriptor : sl_descriptors) {
+    if (descriptor.empty())
+      continue;
+    std::vector<int> v = VideoQualityTest::ParseCSV(descriptor);
+    RTC_CHECK_GT(v[2], 0);
+
+    SpatialLayer layer;
+    layer.scaling_factor_num = v[0] == -1 ? 1 : v[0];
+    layer.scaling_factor_den = v[1] == -1 ? 1 : v[1];
+    layer.target_bitrate_bps = v[2];
+    params->ss.spatial_layers.push_back(layer);
+  }
+}
+
+void VideoQualityTest::SetupCommon(Transport* send_transport,
+                                   Transport* recv_transport) {
+  if (params_.logs)
+    trace_to_stderr_.reset(new test::TraceToStderr);
+
+  size_t num_streams = params_.ss.streams.size();
+  CreateSendConfig(num_streams, 0, send_transport);
+
+  int payload_type;
+  if (params_.common.codec == "VP8") {
+    encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp8));
+    payload_type = kPayloadTypeVP8;
+  } else if (params_.common.codec == "VP9") {
+    encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp9));
+    payload_type = kPayloadTypeVP9;
+  } else {
+    RTC_NOTREACHED() << "Codec not supported!";
+    return;
+  }
+  video_send_config_.encoder_settings.encoder = encoder_.get();
+  video_send_config_.encoder_settings.payload_name = params_.common.codec;
+  video_send_config_.encoder_settings.payload_type = payload_type;
+  video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+  video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
+  for (size_t i = 0; i < num_streams; ++i)
+    video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
+
+  video_send_config_.rtp.extensions.clear();
+  if (params_.common.send_side_bwe) {
+    video_send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTransportSequenceNumber,
+                     test::kTransportSequenceNumberExtensionId));
+  } else {
+    video_send_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
+  }
+
+  video_encoder_config_.min_transmit_bitrate_bps =
+      params_.common.min_transmit_bps;
+  video_encoder_config_.streams = params_.ss.streams;
+  video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
+
+  CreateMatchingReceiveConfigs(recv_transport);
+
+  for (size_t i = 0; i < num_streams; ++i) {
+    video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+    video_receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc =
+        kSendRtxSsrcs[i];
+    video_receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type =
+        kSendRtxPayloadType;
+    video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe;
+  }
+}
+
+void VideoQualityTest::SetupScreenshare() {
+  RTC_CHECK(params_.screenshare.enabled);
+
+  // Fill out codec settings.
+  video_encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
+  if (params_.common.codec == "VP8") {
+    codec_settings_.VP8 = VideoEncoder::GetDefaultVp8Settings();
+    codec_settings_.VP8.denoisingOn = false;
+    codec_settings_.VP8.frameDroppingOn = false;
+    codec_settings_.VP8.numberOfTemporalLayers =
+        static_cast<unsigned char>(params_.common.num_temporal_layers);
+    video_encoder_config_.encoder_specific_settings = &codec_settings_.VP8;
+  } else if (params_.common.codec == "VP9") {
+    codec_settings_.VP9 = VideoEncoder::GetDefaultVp9Settings();
+    codec_settings_.VP9.denoisingOn = false;
+    codec_settings_.VP9.frameDroppingOn = false;
+    codec_settings_.VP9.numberOfTemporalLayers =
+        static_cast<unsigned char>(params_.common.num_temporal_layers);
+    video_encoder_config_.encoder_specific_settings = &codec_settings_.VP9;
+    codec_settings_.VP9.numberOfSpatialLayers =
+        static_cast<unsigned char>(params_.ss.num_spatial_layers);
+  }
+
+  // Setup frame generator.
+  const size_t kWidth = 1850;
+  const size_t kHeight = 1110;
+  std::vector<std::string> slides;
+  slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
+  slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
+  slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
+  slides.push_back(test::ResourcePath("difficult_photo_1850_1110", "yuv"));
+
+  if (params_.screenshare.scroll_duration == 0) {
+    // Cycle image every slide_change_interval seconds.
+    frame_generator_.reset(test::FrameGenerator::CreateFromYuvFile(
+        slides, kWidth, kHeight,
+        params_.screenshare.slide_change_interval * params_.common.fps));
+  } else {
+    RTC_CHECK_LE(params_.common.width, kWidth);
+    RTC_CHECK_LE(params_.common.height, kHeight);
+    RTC_CHECK_GT(params_.screenshare.slide_change_interval, 0);
+    const int kPauseDurationMs = (params_.screenshare.slide_change_interval -
+                                  params_.screenshare.scroll_duration) *
+                                 1000;
+    RTC_CHECK_LE(params_.screenshare.scroll_duration,
+                 params_.screenshare.slide_change_interval);
+
+    frame_generator_.reset(
+        test::FrameGenerator::CreateScrollingInputFromYuvFiles(
+            clock_, slides, kWidth, kHeight, params_.common.width,
+            params_.common.height, params_.screenshare.scroll_duration * 1000,
+            kPauseDurationMs));
+  }
+}
+
+void VideoQualityTest::CreateCapturer(VideoCaptureInput* input) {
+  if (params_.screenshare.enabled) {
+    test::FrameGeneratorCapturer* frame_generator_capturer =
+        new test::FrameGeneratorCapturer(
+            clock_, input, frame_generator_.release(), params_.common.fps);
+    EXPECT_TRUE(frame_generator_capturer->Init());
+    capturer_.reset(frame_generator_capturer);
+  } else {
+    if (params_.video.clip_name.empty()) {
+      capturer_.reset(test::VideoCapturer::Create(input, params_.common.width,
+                                                  params_.common.height,
+                                                  params_.common.fps, clock_));
+    } else {
+      capturer_.reset(test::FrameGeneratorCapturer::CreateFromYuvFile(
+          input, test::ResourcePath(params_.video.clip_name, "yuv"),
+          params_.common.width, params_.common.height, params_.common.fps,
+          clock_));
+      ASSERT_TRUE(capturer_.get() != nullptr)
+          << "Could not create capturer for " << params_.video.clip_name
+          << ".yuv. Is this resource file present?";
+    }
+  }
+}
+
+void VideoQualityTest::RunWithAnalyzer(const Params& params) {
+  params_ = params;
+
+  // TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
+  // differentiate between the analyzer and the renderer case.
+  CheckParams();
+
+  FILE* graph_data_output_file = nullptr;
+  if (!params_.analyzer.graph_data_output_filename.empty()) {
+    graph_data_output_file =
+        fopen(params_.analyzer.graph_data_output_filename.c_str(), "w");
+    RTC_CHECK(graph_data_output_file != nullptr)
+        << "Can't open the file " << params_.analyzer.graph_data_output_filename
+        << "!";
+  }
+
+  Call::Config call_config;
+  call_config.bitrate_config = params.common.call_bitrate_config;
+  CreateCalls(call_config, call_config);
+
+  test::LayerFilteringTransport send_transport(
+      params.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
+      params.common.selected_tl, params_.ss.selected_sl);
+  test::DirectTransport recv_transport(params.pipe, receiver_call_.get());
+
+  std::string graph_title = params_.analyzer.graph_title;
+  if (graph_title.empty())
+    graph_title = VideoQualityTest::GenerateGraphTitle();
+
+  // In the case of different resolutions, the functions calculating PSNR and
+  // SSIM return -1.0, instead of a positive value as usual. VideoAnalyzer
+  // aborts if the average psnr/ssim are below the given threshold, which is
+  // 0.0 by default. Setting the thresholds to -1.1 prevents the unnecessary
+  // abort.
+  VideoStream& selected_stream = params_.ss.streams[params_.ss.selected_stream];
+  int selected_sl = params_.ss.selected_sl != -1
+                        ? params_.ss.selected_sl
+                        : params_.ss.num_spatial_layers - 1;
+  bool disable_quality_check =
+      selected_stream.width != params_.common.width ||
+      selected_stream.height != params_.common.height ||
+      (!params_.ss.spatial_layers.empty() &&
+       params_.ss.spatial_layers[selected_sl].scaling_factor_num !=
+           params_.ss.spatial_layers[selected_sl].scaling_factor_den);
+  if (disable_quality_check) {
+    fprintf(stderr,
+            "Warning: Calculating PSNR and SSIM for downsized resolution "
+            "not implemented yet! Skipping PSNR and SSIM calculations!");
+  }
+
+  VideoAnalyzer analyzer(
+      &send_transport, params_.analyzer.test_label,
+      disable_quality_check ? -1.1 : params_.analyzer.avg_psnr_threshold,
+      disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
+      params_.analyzer.test_durations_secs * params_.common.fps,
+      graph_data_output_file, graph_title,
+      kVideoSendSsrcs[params_.ss.selected_stream]);
+
+  analyzer.SetReceiver(receiver_call_->Receiver());
+  send_transport.SetReceiver(&analyzer);
+  recv_transport.SetReceiver(sender_call_->Receiver());
+
+  SetupCommon(&analyzer, &recv_transport);
+  video_send_config_.encoding_time_observer = &analyzer;
+  video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
+  for (auto& config : video_receive_configs_)
+    config.pre_decode_callback = &analyzer;
+
+  if (params_.screenshare.enabled)
+    SetupScreenshare();
+
+  CreateVideoStreams();
+  analyzer.input_ = video_send_stream_->Input();
+  analyzer.send_stream_ = video_send_stream_;
+
+  CreateCapturer(&analyzer);
+
+  video_send_stream_->Start();
+  for (VideoReceiveStream* receive_stream : video_receive_streams_)
+    receive_stream->Start();
+  capturer_->Start();
+
+  analyzer.Wait();
+
+  send_transport.StopSending();
+  recv_transport.StopSending();
+
+  capturer_->Stop();
+  for (VideoReceiveStream* receive_stream : video_receive_streams_)
+    receive_stream->Stop();
+  video_send_stream_->Stop();
+
+  DestroyStreams();
+
+  if (graph_data_output_file)
+    fclose(graph_data_output_file);
+}
+
+void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
+  params_ = params;
+  CheckParams();
+
+  rtc::scoped_ptr<test::VideoRenderer> local_preview(
+      test::VideoRenderer::Create("Local Preview", params_.common.width,
+                                  params_.common.height));
+  size_t stream_id = params_.ss.selected_stream;
+  std::string title = "Loopback Video";
+  if (params_.ss.streams.size() > 1) {
+    std::ostringstream s;
+    s << stream_id;
+    title += " - Stream #" + s.str();
+  }
+
+  rtc::scoped_ptr<test::VideoRenderer> loopback_video(
+      test::VideoRenderer::Create(title.c_str(),
+                                  params_.ss.streams[stream_id].width,
+                                  params_.ss.streams[stream_id].height));
+
+  // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
+  // match the full stack tests.
+  Call::Config call_config;
+  call_config.bitrate_config = params_.common.call_bitrate_config;
+  rtc::scoped_ptr<Call> call(Call::Create(call_config));
+
+  test::LayerFilteringTransport transport(
+      params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
+      params.common.selected_tl, params_.ss.selected_sl);
+  // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
+  // least share as much code as possible. That way this test would also match
+  // the full stack tests better.
+  transport.SetReceiver(call->Receiver());
+
+  SetupCommon(&transport, &transport);
+
+  video_send_config_.local_renderer = local_preview.get();
+  video_receive_configs_[stream_id].renderer = loopback_video.get();
+
+  if (params_.screenshare.enabled)
+    SetupScreenshare();
+
+  video_send_stream_ =
+      call->CreateVideoSendStream(video_send_config_, video_encoder_config_);
+  VideoReceiveStream* receive_stream =
+      call->CreateVideoReceiveStream(video_receive_configs_[stream_id]);
+  CreateCapturer(video_send_stream_->Input());
+
+  receive_stream->Start();
+  video_send_stream_->Start();
+  capturer_->Start();
+
+  test::PressEnterToContinue();
+
+  capturer_->Stop();
+  video_send_stream_->Stop();
+  receive_stream->Stop();
+
+  call->DestroyVideoReceiveStream(receive_stream);
+  call->DestroyVideoSendStream(video_send_stream_);
+
+  transport.StopSending();
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_quality_test.h b/media/webrtc/trunk/webrtc/video/video_quality_test.h
new file mode 100644
index 0000000000..dd2b011cc3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_quality_test.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
+#define WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/test/call_test.h"
+#include "webrtc/test/frame_generator.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+
+namespace webrtc {
+
+class VideoQualityTest : public test::CallTest {
+ public:
+  // Parameters are grouped into smaller structs to make it easier to set
+  // the desired elements and skip unused, using aggregate initialization.
+  // Unfortunately, C++11 (as opposed to C11) doesn't support unnamed structs,
+  // which makes the implementation of VideoQualityTest a bit uglier.
+  struct Params {
+    struct {
+      size_t width;
+      size_t height;
+      int32_t fps;
+      int min_bitrate_bps;
+      int target_bitrate_bps;
+      int max_bitrate_bps;
+      std::string codec;
+      int num_temporal_layers;
+      int selected_tl;
+      int min_transmit_bps;
+
+      Call::Config::BitrateConfig call_bitrate_config;
+      bool send_side_bwe;
+    } common;
+    struct {  // Video-specific settings.
+      std::string clip_name;
+    } video;
+    struct {  // Screenshare-specific settings.
+      bool enabled;
+      int32_t slide_change_interval;
+      int32_t scroll_duration;
+    } screenshare;
+    struct {  // Analyzer settings.
+      std::string test_label;
+      double avg_psnr_threshold;  // (*)
+      double avg_ssim_threshold;  // (*)
+      int test_durations_secs;
+      std::string graph_data_output_filename;
+      std::string graph_title;
+    } analyzer;
+    FakeNetworkPipe::Config pipe;
+    bool logs;
+    struct {                             // Spatial scalability.
+      std::vector<VideoStream> streams;  // If empty, one stream is assumed.
+      size_t selected_stream;
+      int num_spatial_layers;
+      int selected_sl;
+      // If empty, bitrates are generated in VP9Impl automatically.
+      std::vector<SpatialLayer> spatial_layers;
+    } ss;
+  };
+  // (*) Set to -1.1 if generating graph data for simulcast or SVC and the
+  // selected stream/layer doesn't have the same resolution as the largest
+  // stream/layer (to ignore the PSNR and SSIM calculation errors).
+
+  VideoQualityTest();
+  void RunWithAnalyzer(const Params& params);
+  void RunWithVideoRenderer(const Params& params);
+
+  static void FillScalabilitySettings(
+      Params* params,
+      const std::vector<std::string>& stream_descriptors,
+      size_t selected_stream,
+      int num_spatial_layers,
+      int selected_sl,
+      const std::vector<std::string>& sl_descriptors);
+
+ protected:
+  // No-op implementation to be able to instantiate this class from non-TEST_F
+  // locations.
+  void TestBody() override;
+
+  // Helper methods accessing only params_.
+  std::string GenerateGraphTitle() const;
+  void CheckParams();
+
+  // Helper static methods.
+  static VideoStream DefaultVideoStream(const Params& params);
+  static std::vector<int> ParseCSV(const std::string& str);
+
+  // Helper methods for setting up the call.
+  void CreateCapturer(VideoCaptureInput* input);
+  void SetupCommon(Transport* send_transport, Transport* recv_transport);
+  void SetupScreenshare();
+
+  // We need a more general capturer than the FrameGeneratorCapturer.
+  rtc::scoped_ptr<test::VideoCapturer> capturer_;
+  rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_;
+  rtc::scoped_ptr<test::FrameGenerator> frame_generator_;
+  rtc::scoped_ptr<VideoEncoder> encoder_;
+  VideoCodecUnion codec_settings_;
+  Clock* const clock_;
+
+  Params params_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
diff --git a/media/webrtc/trunk/webrtc/video/video_receive_stream.cc b/media/webrtc/trunk/webrtc/video/video_receive_stream.cc
index 126d485e58..03a4acf9e0 100644
--- a/media/webrtc/trunk/webrtc/video/video_receive_stream.cc
+++ b/media/webrtc/trunk/webrtc/video/video_receive_stream.cc
@@ -12,32 +12,33 @@
 
 #include <stdlib.h>
 
+#include <set>
 #include <string>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video_encoder.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_external_codec.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
+
+static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
+  for (const auto& extension : extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber)
+      return true;
+  }
+  return false;
+}
+
 std::string VideoReceiveStream::Decoder::ToString() const {
   std::stringstream ss;
   ss << "{decoder: " << (decoder != nullptr ? "(VideoDecoder)" : "nullptr");
   ss << ", payload_type: " << payload_type;
   ss << ", payload_name: " << payload_name;
-  ss << ", is_renderer: " << (is_renderer ? "yes" : "no");
-  ss << ", expected_delay_ms: " << expected_delay_ms;
   ss << '}';
 
   return ss.str();
@@ -55,7 +56,8 @@ std::string VideoReceiveStream::Config::ToString() const {
   ss << ", rtp: " << rtp.ToString();
   ss << ", renderer: " << (renderer != nullptr ? "(renderer)" : "nullptr");
   ss << ", render_delay_ms: " << render_delay_ms;
-  ss << ", audio_channel_id: " << audio_channel_id;
+  if (!sync_group.empty())
+    ss << ", sync_group: " << sync_group;
   ss << ", pre_decode_callback: "
      << (pre_decode_callback != nullptr ? "(EncodedFrameObserver)" : "nullptr");
   ss << ", pre_render_callback: "
@@ -70,14 +72,15 @@ std::string VideoReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
-  ss << ", rtcp_mode: " << (rtcp_mode == newapi::kRtcpCompound
-                                ? "kRtcpCompound"
-                                : "kRtcpReducedSize");
+  ss << ", rtcp_mode: "
+     << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
+                                          : "RtcpMode::kReducedSize");
   ss << ", rtcp_xr: ";
   ss << "{receiver_reference_time_report: "
      << (rtcp_xr.receiver_reference_time_report ? "on" : "off");
   ss << '}';
   ss << ", remb: " << (remb ? "on" : "off");
+  ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
   ss << ", fec: " << fec.ToString();
   ss << ", rtx: {";
@@ -107,9 +110,11 @@ VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
   memset(&codec, 0, sizeof(codec));
 
   codec.plType = decoder.payload_type;
-  strcpy(codec.plName, decoder.payload_name.c_str());
+  strncpy(codec.plName, decoder.payload_name.c_str(), sizeof(codec.plName));
   if (decoder.payload_name == "VP8") {
     codec.codecType = kVideoCodecVP8;
+  } else if (decoder.payload_name == "VP9") {
+    codec.codecType = kVideoCodecVP9;
   } else if (decoder.payload_name == "H264") {
     codec.codecType = kVideoCodecH264;
   } else {
@@ -118,6 +123,8 @@ VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
 
   if (codec.codecType == kVideoCodecVP8) {
     codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
+  } else if (codec.codecType == kVideoCodecVP9) {
+    codec.codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
   } else if (codec.codecType == kVideoCodecH264) {
     codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
   }
@@ -131,253 +138,274 @@ VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
 }
 }  // namespace
 
-VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
-                                       const VideoReceiveStream::Config& config,
-                                       newapi::Transport* transport,
-                                       webrtc::VoiceEngine* voice_engine,
-                                       int base_channel)
-    : transport_adapter_(transport),
+VideoReceiveStream::VideoReceiveStream(
+    int num_cpu_cores,
+    CongestionController* congestion_controller,
+    const VideoReceiveStream::Config& config,
+    webrtc::VoiceEngine* voice_engine,
+    ProcessThread* process_thread,
+    CallStats* call_stats)
+    : transport_adapter_(config.rtcp_send_transport),
       encoded_frame_proxy_(config.pre_decode_callback),
       config_(config),
       clock_(Clock::GetRealTimeClock()),
-      channel_(-1) {
-  video_engine_base_ = ViEBase::GetInterface(video_engine);
-  video_engine_base_->CreateReceiveChannel(channel_, base_channel);
-  DCHECK(channel_ != -1);
+      congestion_controller_(congestion_controller),
+      call_stats_(call_stats) {
+  LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
-  rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
-  DCHECK(rtp_rtcp_ != nullptr);
+  bool send_side_bwe =
+      config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions);
+
+  RemoteBitrateEstimator* bitrate_estimator =
+      congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe);
+
+  vie_channel_.reset(new ViEChannel(
+      num_cpu_cores, &transport_adapter_, process_thread, nullptr,
+      nullptr, nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(),
+      congestion_controller_->pacer(), congestion_controller_->packet_router(),
+      1, false));
+
+  RTC_CHECK(vie_channel_->Init() == 0);
+
+  // Register the channel to receive stats updates.
+  call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver());
 
   // TODO(pbos): This is not fine grained enough...
-  rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
-  rtp_rtcp_->SetKeyFrameRequestMethod(channel_, kViEKeyFrameRequestPliRtcp);
-  SetRtcpMode(config_.rtp.rtcp_mode);
+  vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false,
+                                  -1, -1);
+  RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
+      << "A stream should not be configured with RTCP disabled. This value is "
+         "reserved for internal usage.";
+  vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
 
-  DCHECK(config_.rtp.remote_ssrc != 0);
+  RTC_DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
-  DCHECK(config_.rtp.local_ssrc != 0);
-  DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
+  RTC_DCHECK(config_.rtp.local_ssrc != 0);
+  RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
-  rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
+  vie_channel_->SetSSRC(config_.rtp.local_ssrc, kViEStreamTypeNormal, 0);
   // TODO(pbos): Support multiple RTX, per video payload.
   Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
-  if (it != config_.rtp.rtx.end()) {
-    DCHECK(it->second.ssrc != 0);
-    DCHECK(it->second.payload_type != 0);
+  for (; it != config_.rtp.rtx.end(); ++it) {
+    RTC_DCHECK(it->second.ssrc != 0);
+    RTC_DCHECK(it->second.payload_type != 0);
 
-    rtp_rtcp_->SetRemoteSSRCType(channel_, kViEStreamTypeRtx, it->second.ssrc);
-    rtp_rtcp_->SetRtxReceivePayloadType(channel_, it->second.payload_type);
+    vie_channel_->SetRemoteSSRCType(kViEStreamTypeRtx, it->second.ssrc);
+    vie_channel_->SetRtxReceivePayloadType(it->second.payload_type, it->first);
   }
+  // TODO(holmer): When Chrome no longer depends on this being false by default,
+  // always use the mapping and remove this whole codepath.
+  vie_channel_->SetUseRtxPayloadMappingOnRestore(
+      config_.rtp.use_rtx_payload_mapping_on_restore);
 
-  rtp_rtcp_->SetRembStatus(channel_, false, config_.rtp.remb);
+  congestion_controller_->SetChannelRembStatus(false, config_.rtp.remb,
+                                               vie_channel_->rtp_rtcp());
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
-    DCHECK_GE(id, 1);
-    DCHECK_LE(id, 14);
+    RTC_DCHECK_GE(id, 1);
+    RTC_DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
-      CHECK_EQ(0,
-               rtp_rtcp_->SetReceiveTimestampOffsetStatus(channel_, true, id));
+      RTC_CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
-      CHECK_EQ(0,
-               rtp_rtcp_->SetReceiveAbsoluteSendTimeStatus(channel_, true, id));
+      RTC_CHECK_EQ(0, vie_channel_->SetReceiveAbsoluteSendTimeStatus(true, id));
     } else if (extension == RtpExtension::kVideoRotation) {
-      CHECK_EQ(0, rtp_rtcp_->SetReceiveVideoRotationStatus(channel_, true, id));
+      RTC_CHECK_EQ(0, vie_channel_->SetReceiveVideoRotationStatus(true, id));
+    } else if (extension == RtpExtension::kTransportSequenceNumber) {
+      RTC_CHECK_EQ(0,
+                   vie_channel_->SetReceiveTransportSequenceNumber(true, id));
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 
-  network_ = ViENetwork::GetInterface(video_engine);
-  DCHECK(network_ != nullptr);
-
-  network_->RegisterSendTransport(channel_, transport_adapter_);
-
-  codec_ = ViECodec::GetInterface(video_engine);
-
   if (config_.rtp.fec.ulpfec_payload_type != -1) {
     // ULPFEC without RED doesn't make sense.
-    DCHECK(config_.rtp.fec.red_payload_type != -1);
+    RTC_DCHECK(config_.rtp.fec.red_payload_type != -1);
     VideoCodec codec;
     memset(&codec, 0, sizeof(codec));
     codec.codecType = kVideoCodecULPFEC;
-    strcpy(codec.plName, "ulpfec");
+    strncpy(codec.plName, "ulpfec", sizeof(codec.plName));
     codec.plType = config_.rtp.fec.ulpfec_payload_type;
-    CHECK_EQ(0, codec_->SetReceiveCodec(channel_, codec));
+    RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
   }
   if (config_.rtp.fec.red_payload_type != -1) {
     VideoCodec codec;
     memset(&codec, 0, sizeof(codec));
     codec.codecType = kVideoCodecRED;
-    strcpy(codec.plName, "red");
+    strncpy(codec.plName, "red", sizeof(codec.plName));
     codec.plType = config_.rtp.fec.red_payload_type;
-    CHECK_EQ(0, codec_->SetReceiveCodec(channel_, codec));
+    RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
+    if (config_.rtp.fec.red_rtx_payload_type != -1) {
+      vie_channel_->SetRtxReceivePayloadType(
+          config_.rtp.fec.red_rtx_payload_type,
+          config_.rtp.fec.red_payload_type);
+    }
   }
 
+  vie_channel_->EnableTMMBR(config.rtp.tmmbr);
+
+  vie_channel_->rtp_rtcp()->SetKeyFrameRequestMethod(config.rtp.keyframe_method);
+
+  if (config.rtp.rtcp_xr.receiver_reference_time_report)
+    vie_channel_->SetRtcpXrRrtrStatus(true);
+
   stats_proxy_.reset(
       new ReceiveStatisticsProxy(config_.rtp.remote_ssrc, clock_));
 
-  CHECK_EQ(0, rtp_rtcp_->RegisterReceiveChannelRtcpStatisticsCallback(
-                  channel_, stats_proxy_.get()));
-  CHECK_EQ(0, rtp_rtcp_->RegisterReceiveChannelRtpStatisticsCallback(
-                  channel_, stats_proxy_.get()));
-  CHECK_EQ(0, rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(
-                  channel_, stats_proxy_.get()));
-  CHECK_EQ(0, codec_->RegisterDecoderObserver(channel_, *stats_proxy_));
+  vie_channel_->RegisterReceiveStatisticsProxy(stats_proxy_.get());
+  vie_channel_->RegisterReceiveChannelRtcpStatisticsCallback(
+      stats_proxy_.get());
+  vie_channel_->RegisterReceiveChannelRtpStatisticsCallback(stats_proxy_.get());
+  vie_channel_->RegisterRtcpPacketTypeCounterObserver(stats_proxy_.get());
 
-  video_engine_base_->RegisterReceiveStatisticsProxy(channel_,
-                                                     stats_proxy_.get());
-
-  external_codec_ = ViEExternalCodec::GetInterface(video_engine);
-  DCHECK(!config_.decoders.empty());
+  RTC_DCHECK(!config_.decoders.empty());
+  std::set<int> decoder_payload_types;
   for (size_t i = 0; i < config_.decoders.size(); ++i) {
     const Decoder& decoder = config_.decoders[i];
-    CHECK_EQ(0, external_codec_->RegisterExternalReceiveCodec(
-                    channel_, decoder.payload_type, decoder.decoder,
-                    decoder.is_renderer, decoder.expected_delay_ms));
+    RTC_CHECK(decoder.decoder);
+    RTC_CHECK(decoder_payload_types.find(decoder.payload_type) ==
+              decoder_payload_types.end())
+        << "Duplicate payload type (" << decoder.payload_type
+        << ") for different decoders.";
+    decoder_payload_types.insert(decoder.payload_type);
+    vie_channel_->RegisterExternalDecoder(decoder.payload_type,
+                                          decoder.decoder);
 
     VideoCodec codec = CreateDecoderVideoCodec(decoder);
 
-    CHECK_EQ(0, codec_->SetReceiveCodec(channel_, codec));
+    RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
   }
 
-  render_ = ViERender::GetInterface(video_engine);
-  DCHECK(render_ != nullptr);
+  incoming_video_stream_.reset(new IncomingVideoStream(
+      0, config.renderer ? config.renderer->SmoothsRenderedFrames() : false));
+  incoming_video_stream_->SetExpectedRenderDelay(config.render_delay_ms);
+  vie_channel_->SetExpectedRenderDelay(config.render_delay_ms);
+  incoming_video_stream_->SetExternalCallback(this);
+  vie_channel_->SetIncomingVideoStream(incoming_video_stream_.get());
 
-  render_->AddRenderer(channel_, kVideoI420, this);
-
-  if (voice_engine && config_.audio_channel_id != -1) {
-    video_engine_base_->SetVoiceEngine(voice_engine);
-    video_engine_base_->ConnectAudioChannel(channel_, config_.audio_channel_id);
-  }
-
-  image_process_ = ViEImageProcess::GetInterface(video_engine);
-  if (config.pre_decode_callback) {
-    image_process_->RegisterPreDecodeImageCallback(channel_,
-                                                   &encoded_frame_proxy_);
-  }
-  image_process_->RegisterPreRenderCallback(channel_, this);
-
-  if (config.rtp.rtcp_xr.receiver_reference_time_report) {
-    rtp_rtcp_->SetRtcpXrRrtrStatus(channel_, true);
-  }
+  vie_channel_->RegisterPreDecodeImageCallback(this);
+  vie_channel_->RegisterPreRenderCallback(this);
 }
 
 VideoReceiveStream::~VideoReceiveStream() {
-  image_process_->DeRegisterPreRenderCallback(channel_);
-  image_process_->DeRegisterPreDecodeCallback(channel_);
+  LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
+  incoming_video_stream_->Stop();
+  vie_channel_->RegisterPreRenderCallback(nullptr);
+  vie_channel_->RegisterPreDecodeImageCallback(nullptr);
 
-  render_->RemoveRenderer(channel_);
+  call_stats_->DeregisterStatsObserver(vie_channel_->GetStatsObserver());
+  congestion_controller_->SetChannelRembStatus(false, false,
+                                               vie_channel_->rtp_rtcp());
 
-  for (size_t i = 0; i < config_.decoders.size(); ++i) {
-    external_codec_->DeRegisterExternalReceiveCodec(
-        channel_, config_.decoders[i].payload_type);
-  }
-
-  network_->DeregisterSendTransport(channel_);
-
-  video_engine_base_->SetVoiceEngine(nullptr);
-  image_process_->Release();
-  external_codec_->Release();
-  codec_->DeregisterDecoderObserver(channel_);
-  rtp_rtcp_->DeregisterReceiveChannelRtpStatisticsCallback(channel_,
-                                                           stats_proxy_.get());
-  rtp_rtcp_->DeregisterReceiveChannelRtcpStatisticsCallback(channel_,
-                                                            stats_proxy_.get());
-  rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(channel_, nullptr);
-  codec_->Release();
-  network_->Release();
-  render_->Release();
-  rtp_rtcp_->Release();
-  video_engine_base_->DeleteChannel(channel_);
-  video_engine_base_->Release();
+  uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
+  bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
+  congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe)->
+      RemoveStream(remote_ssrc);
 }
 
 void VideoReceiveStream::Start() {
   transport_adapter_.Enable();
-  CHECK_EQ(0, render_->StartRender(channel_));
-  CHECK_EQ(0, video_engine_base_->StartReceive(channel_));
+  incoming_video_stream_->Start();
+  vie_channel_->StartReceive();
 }
 
 void VideoReceiveStream::Stop() {
-  CHECK_EQ(0, render_->StopRender(channel_));
-  CHECK_EQ(0, video_engine_base_->StopReceive(channel_));
+  incoming_video_stream_->Stop();
+  vie_channel_->StopReceive();
   transport_adapter_.Disable();
 }
 
+void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
+                                        int audio_channel_id) {
+  if (voice_engine != nullptr && audio_channel_id != -1) {
+    VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
+    vie_channel_->SetVoiceChannel(audio_channel_id, voe_sync_interface);
+    voe_sync_interface->Release();
+  } else {
+    vie_channel_->SetVoiceChannel(-1, nullptr);
+  }
+}
+
 VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
   return stats_proxy_->GetStats();
 }
 
 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return network_->ReceivedRTCPPacket(channel_, packet, length) == 0;
+  return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
 }
 
-bool VideoReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) {
-  return network_->ReceivedRTPPacket(channel_, packet, length, PacketTime()) ==
-      0;
+bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
+                                    size_t length,
+                                    const PacketTime& packet_time) {
+  return vie_channel_->ReceivedRTPPacket(packet, length, packet_time) == 0;
 }
 
-void VideoReceiveStream::FrameCallback(I420VideoFrame* video_frame) {
+void VideoReceiveStream::FrameCallback(VideoFrame* video_frame) {
   stats_proxy_->OnDecodedFrame();
 
-  if (config_.pre_render_callback)
-    config_.pre_render_callback->FrameCallback(video_frame);
+  // Post processing is not supported if the frame is backed by a texture.
+  if (video_frame->native_handle() == NULL) {
+    if (config_.pre_render_callback)
+      config_.pre_render_callback->FrameCallback(video_frame);
+  }
 }
 
-int VideoReceiveStream::FrameSizeChange(unsigned int width,
-                                        unsigned int height,
-                                        unsigned int number_of_streams) {
-  return 0;
-}
+int VideoReceiveStream::RenderFrame(const uint32_t /*stream_id*/,
+                                    const VideoFrame& video_frame) {
+  // TODO(pbos): Wire up config_.render->IsTextureSupported() and convert if not
+  // supported. Or provide methods for converting a texture frame in
+  // VideoFrame.
 
-int VideoReceiveStream::DeliverFrame(unsigned char* buffer,
-                                     size_t buffer_size,
-                                     uint32_t timestamp,
-                                     int64_t ntp_time_ms,
-                                     int64_t render_time_ms,
-                                     void* handle) {
-  CHECK(false) << "Renderer should be configured as kVideoI420 and never "
-                  "receive callbacks on DeliverFrame.";
-  return 0;
-}
-
-int VideoReceiveStream::DeliverI420Frame(const I420VideoFrame& video_frame) {
   if (config_.renderer != nullptr)
     config_.renderer->RenderFrame(
         video_frame,
         video_frame.render_time_ms() - clock_->TimeInMilliseconds());
 
-  stats_proxy_->OnRenderedFrame();
+  stats_proxy_->OnRenderedFrame(video_frame.width(), video_frame.height());
 
   return 0;
 }
 
-bool VideoReceiveStream::IsTextureSupported() {
-  if (config_.renderer == nullptr)
-    return false;
-  return config_.renderer->IsTextureSupported();
-}
-
-void VideoReceiveStream::SignalNetworkState(Call::NetworkState state) {
-  if (state == Call::kNetworkUp)
-    SetRtcpMode(config_.rtp.rtcp_mode);
-  network_->SetNetworkTransmissionState(channel_, state == Call::kNetworkUp);
-  if (state == Call::kNetworkDown)
-    rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNone);
-}
-
-void VideoReceiveStream::SetRtcpMode(newapi::RtcpMode mode) {
-  switch (mode) {
-    case newapi::kRtcpCompound:
-      rtp_rtcp_->SetRTCPStatus(channel_, kRtcpCompound_RFC4585);
-      break;
-    case newapi::kRtcpReducedSize:
-      rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNonCompound_RFC5506);
-      break;
+// TODO(asapersson): Consider moving callback from video_encoder.h or
+// creating a different callback.
+int32_t VideoReceiveStream::Encoded(
+    const EncodedImage& encoded_image,
+    const CodecSpecificInfo* codec_specific_info,
+    const RTPFragmentationHeader* fragmentation) {
+  stats_proxy_->OnPreDecode(encoded_image, codec_specific_info);
+  if (config_.pre_decode_callback) {
+    // TODO(asapersson): Remove EncodedFrameCallbackAdapter.
+    encoded_frame_proxy_.Encoded(
+        encoded_image, codec_specific_info, fragmentation);
   }
+  return 0;
 }
+
+void VideoReceiveStream::SignalNetworkState(NetworkState state) {
+  vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
+                                                : RtcpMode::kOff);
+}
+
+int64_t VideoReceiveStream::GetRtt() const {
+  uint32_t timestampNTPHigh = 0;
+  uint32_t timestampNTPLow = 0;
+  uint32_t receivedPacketCount = 0;
+  uint64_t receivedOctetCount = 0;
+  uint32_t jitterSamples = 0;
+  uint16_t fractionLost = 0;
+  uint32_t cumulativeLost = 0;
+  int64_t rttMs = 0;
+  if (vie_channel_->GetRemoteRTCPReceiverInfo(
+        timestampNTPHigh, timestampNTPLow, receivedPacketCount,
+        receivedOctetCount, &jitterSamples, &fractionLost, &cumulativeLost,
+        &rttMs)) {
+    return rttMs;
+  }
+  return -1;
+}
+
 }  // namespace internal
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_receive_stream.h b/media/webrtc/trunk/webrtc/video/video_receive_stream.h
index dcc8122ee4..7e537efc6c 100644
--- a/media/webrtc/trunk/webrtc/video/video_receive_stream.h
+++ b/media/webrtc/trunk/webrtc/video/video_receive_stream.h
@@ -15,84 +15,79 @@
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
+#include "webrtc/call/transport_adapter.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video/transport_adapter.h"
-#include "webrtc/video_engine/include/vie_render.h"
+#include "webrtc/video/vie_channel.h"
+#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video_encoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
-class VideoEngine;
-class ViEBase;
-class ViECodec;
-class ViEExternalCodec;
-class ViEImageProcess;
-class ViENetwork;
-class ViERender;
-class ViERTP_RTCP;
+class CallStats;
+class CongestionController;
 class VoiceEngine;
 
 namespace internal {
 
 class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public I420FrameCallback,
-                           public ExternalRenderer {
+                           public VideoRenderCallback,
+                           public EncodedImageCallback {
  public:
-  VideoReceiveStream(webrtc::VideoEngine* video_engine,
+  VideoReceiveStream(int num_cpu_cores,
+                     CongestionController* congestion_controller,
                      const VideoReceiveStream::Config& config,
-                     newapi::Transport* transport,
                      webrtc::VoiceEngine* voice_engine,
-                     int base_channel);
-  virtual ~VideoReceiveStream();
+                     ProcessThread* process_thread,
+                     CallStats* call_stats);
+  ~VideoReceiveStream() override;
 
+  // webrtc::ReceiveStream implementation.
   void Start() override;
   void Stop() override;
-  Stats GetStats() const override;
+  void SignalNetworkState(NetworkState state) override;
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+  bool DeliverRtp(const uint8_t* packet,
+                  size_t length,
+                  const PacketTime& packet_time) override;
+
+  // webrtc::VideoReceiveStream implementation.
+  webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   // Overrides I420FrameCallback.
-  void FrameCallback(I420VideoFrame* video_frame) override;
+  void FrameCallback(VideoFrame* video_frame) override;
 
-  // Overrides ExternalRenderer.
-  int FrameSizeChange(unsigned int width,
-                      unsigned int height,
-                      unsigned int number_of_streams) override;
-  int DeliverFrame(unsigned char* buffer,
-                   size_t buffer_size,
-                   uint32_t timestamp,
-                   int64_t ntp_time_ms,
-                   int64_t render_time_ms,
-                   void* handle) override;
-  int DeliverI420Frame(const I420VideoFrame& webrtc_frame) override;
-  bool IsTextureSupported() override;
+  // Overrides VideoRenderCallback.
+  int RenderFrame(const uint32_t /*stream_id*/,
+                  const VideoFrame& video_frame) override;
 
-  void SignalNetworkState(Call::NetworkState state);
+  // Overrides EncodedImageCallback.
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific_info,
+                  const RTPFragmentationHeader* fragmentation) override;
 
-  virtual bool DeliverRtcp(const uint8_t* packet, size_t length);
-  virtual bool DeliverRtp(const uint8_t* packet, size_t length);
+  const Config& config() const { return config_; }
 
+  void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id) override;
+  int64_t GetRtt() const override;
  private:
-  void SetRtcpMode(newapi::RtcpMode mode);
-
   TransportAdapter transport_adapter_;
   EncodedFrameCallbackAdapter encoded_frame_proxy_;
   const VideoReceiveStream::Config config_;
   Clock* const clock_;
 
-  ViEBase* video_engine_base_;
-  ViECodec* codec_;
-  ViEExternalCodec* external_codec_;
-  ViENetwork* network_;
-  ViERender* render_;
-  ViERTP_RTCP* rtp_rtcp_;
-  ViEImageProcess* image_process_;
+  CongestionController* const congestion_controller_;
+  CallStats* const call_stats_;
 
+  rtc::scoped_ptr<IncomingVideoStream> incoming_video_stream_;
   rtc::scoped_ptr<ReceiveStatisticsProxy> stats_proxy_;
-
-  int channel_;
+  rtc::scoped_ptr<ViEChannel> vie_channel_;
 };
 }  // namespace internal
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream.cc b/media/webrtc/trunk/webrtc/video/video_send_stream.cc
index 58a24a6a0d..fd7c8079ef 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream.cc
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream.cc
@@ -16,20 +16,26 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_external_codec.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/vie_defines.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/encoder_state_feedback.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/video_capture_input.h"
+#include "webrtc/video/vie_channel.h"
+#include "webrtc/video/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
+
+class PacedSender;
+class RtcpIntraFrameObserver;
+class TransportFeedbackObserver;
+
 std::string
 VideoSendStream::Config::EncoderSettings::ToString() const {
   std::stringstream ss;
@@ -91,7 +97,7 @@ std::string VideoSendStream::Config::ToString() const {
   ss << ", post_encode_callback: " << (post_encode_callback != nullptr
                                            ? "(EncodedFrameObserver)"
                                            : "nullptr");
-  ss << "local_renderer: " << (local_renderer != nullptr ? "(VideoRenderer)"
+  ss << ", local_renderer: " << (local_renderer != nullptr ? "(VideoRenderer)"
                                                          : "nullptr");
   ss << ", render_delay_ms: " << render_delay_ms;
   ss << ", target_delay_ms: " << target_delay_ms;
@@ -103,43 +109,81 @@ std::string VideoSendStream::Config::ToString() const {
 
 namespace internal {
 VideoSendStream::VideoSendStream(
-    newapi::Transport* transport,
-    CpuOveruseObserver* overuse_observer,
-    webrtc::VideoEngine* video_engine,
+    int num_cpu_cores,
+    ProcessThread* module_process_thread,
+    CallStats* call_stats,
+    CongestionController* congestion_controller,
+    BitrateAllocator* bitrate_allocator,
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
-    const std::map<uint32_t, RtpState>& suspended_ssrcs,
-    int base_channel)
-    : transport_adapter_(transport),
+    const std::map<uint32_t, RtpState>& suspended_ssrcs)
+    : stats_proxy_(Clock::GetRealTimeClock(),
+                   config,
+                   encoder_config.content_type),
+      transport_adapter_(config.send_transport),
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
       suspended_ssrcs_(suspended_ssrcs),
-      external_codec_(nullptr),
-      channel_(-1),
-      use_config_bitrate_(true),
-      stats_proxy_(Clock::GetRealTimeClock(), config) {
-  video_engine_base_ = ViEBase::GetInterface(video_engine);
-  video_engine_base_->CreateChannelWithoutDefaultEncoder(channel_,
-                                                         base_channel);
-  DCHECK(channel_ != -1);
+      module_process_thread_(module_process_thread),
+      call_stats_(call_stats),
+      congestion_controller_(congestion_controller),
+      encoder_feedback_(new EncoderStateFeedback()),
+      use_config_bitrate_(true) {
+  LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
+  RTC_DCHECK(!config_.rtp.ssrcs.empty());
 
-  rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
-  DCHECK(rtp_rtcp_ != nullptr);
+  // Set up Call-wide sequence numbers, if configured for this send stream.
+  TransportFeedbackObserver* transport_feedback_observer = nullptr;
+  for (const RtpExtension& extension : config.rtp.extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      transport_feedback_observer =
+          congestion_controller_->GetTransportFeedbackObserver();
+      break;
+    }
+  }
 
-  DCHECK(!config_.rtp.ssrcs.empty());
+  const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs;
+
+  vie_encoder_.reset(new ViEEncoder(
+      num_cpu_cores, module_process_thread_, &stats_proxy_,
+      config.pre_encode_callback, congestion_controller_->pacer(),
+      bitrate_allocator));
+  RTC_CHECK(vie_encoder_->Init());
+
+  vie_channel_.reset(new ViEChannel(
+      num_cpu_cores, config.send_transport, module_process_thread_,
+      encoder_feedback_->GetRtcpIntraFrameObserver(),
+      congestion_controller_->GetBitrateController()->
+          CreateRtcpBandwidthObserver(),
+      transport_feedback_observer,
+      congestion_controller_->GetRemoteBitrateEstimator(false),
+      call_stats_->rtcp_rtt_stats(), congestion_controller_->pacer(),
+      congestion_controller_->packet_router(), ssrcs.size(), true));
+  RTC_CHECK(vie_channel_->Init() == 0);
+
+  call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver());
+
+  vie_encoder_->StartThreadsAndSetSharedMembers(
+      vie_channel_->send_payload_router(),
+      vie_channel_->vcm_protection_callback());
+
+  std::vector<uint32_t> first_ssrc(1, ssrcs[0]);
+  vie_encoder_->SetSsrcs(first_ssrc);
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
-    DCHECK_GE(id, 1);
-    DCHECK_LE(id, 14);
+    RTC_DCHECK_GE(id, 1);
+    RTC_DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
-      CHECK_EQ(0, rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id));
+      RTC_CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
-      CHECK_EQ(0, rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id));
+      RTC_CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id));
     } else if (extension == RtpExtension::kVideoRotation) {
-      CHECK_EQ(0, rtp_rtcp_->SetSendVideoRotationStatus(channel_, true, id));
+      RTC_CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id));
+    } else if (extension == RtpExtension::kTransportSequenceNumber) {
+      RTC_CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id));
     } else if (extension == RtpExtension::kRtpStreamId) {
       RTC_CHECK_EQ(0, vie_channel_->SetSendRtpStreamId(true,id));
     } else {
@@ -147,146 +191,112 @@ VideoSendStream::VideoSendStream(
     }
   }
 
-  rtp_rtcp_->SetRembStatus(channel_, true, false);
+  congestion_controller_->SetChannelRembStatus(true, false,
+                                               vie_channel_->rtp_rtcp());
 
   // Enable NACK, FEC or both.
-  if (config_.rtp.fec.red_payload_type != -1) {
-    DCHECK(config_.rtp.fec.ulpfec_payload_type != -1);
-    if (config_.rtp.nack.rtp_history_ms > 0) {
-      rtp_rtcp_->SetHybridNACKFECStatus(
-          channel_,
-          true,
-          static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
-          static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
-    } else {
-      rtp_rtcp_->SetFECStatus(
-          channel_,
-          true,
-          static_cast<unsigned char>(config_.rtp.fec.red_payload_type),
-          static_cast<unsigned char>(config_.rtp.fec.ulpfec_payload_type));
-    }
-  } else {
-    rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
-  }
+  const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
+  const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
+  // TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
+  vie_channel_->SetProtectionMode(enable_protection_nack, enable_protection_fec,
+                                  config_.rtp.fec.red_payload_type,
+                                  config_.rtp.fec.ulpfec_payload_type);
+  vie_encoder_->SetProtectionMethod(enable_protection_nack,
+                                    enable_protection_fec);
 
   ConfigureSsrcs();
 
-  char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
-  DCHECK_LT(config_.rtp.c_name.length(),
-            static_cast<size_t>(ViERTP_RTCP::KMaxRTCPCNameLength));
-  strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1);
-  rtcp_cname[sizeof(rtcp_cname) - 1] = '\0';
+  vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str());
 
-  rtp_rtcp_->SetRTCPCName(channel_, rtcp_cname);
+  input_.reset(new internal::VideoCaptureInput(
+      module_process_thread_, vie_encoder_.get(), config_.local_renderer,
+      &stats_proxy_, this, config_.encoding_time_observer));
 
-  capture_ = ViECapture::GetInterface(video_engine);
-  capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_);
-  capture_->ConnectCaptureDevice(capture_id_, channel_);
-
-  network_ = ViENetwork::GetInterface(video_engine);
-  DCHECK(network_ != nullptr);
-
-  network_->RegisterSendTransport(channel_, transport_adapter_);
   // 28 to match packet overhead in ModuleRtpRtcpImpl.
-  network_->SetMTU(channel_,
-                   static_cast<unsigned int>(config_.rtp.max_packet_size + 28));
+  RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28));
+  vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
 
-  DCHECK(config.encoder_settings.encoder != nullptr);
-  DCHECK_GE(config.encoder_settings.payload_type, 0);
-  DCHECK_LE(config.encoder_settings.payload_type, 127);
-  external_codec_ = ViEExternalCodec::GetInterface(video_engine);
-  CHECK_EQ(0, external_codec_->RegisterExternalSendCodec(
-                  channel_, config.encoder_settings.payload_type,
-                  config.encoder_settings.encoder, false));
+  RTC_DCHECK(config.encoder_settings.encoder != nullptr);
+  RTC_DCHECK_GE(config.encoder_settings.payload_type, 0);
+  RTC_DCHECK_LE(config.encoder_settings.payload_type, 127);
+  RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder(
+                      config.encoder_settings.encoder,
+                      config.encoder_settings.payload_type,
+                      config.encoder_settings.internal_source));
 
-  codec_ = ViECodec::GetInterface(video_engine);
-  CHECK(ReconfigureVideoEncoder(encoder_config));
+  RTC_CHECK(ReconfigureVideoEncoder(encoder_config));
 
-  if (overuse_observer)
-    video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
-  // Registered regardless of monitoring, used for stats.
-  video_engine_base_->RegisterCpuOveruseMetricsObserver(channel_,
-                                                        &stats_proxy_);
+  vie_channel_->RegisterSendSideDelayObserver(&stats_proxy_);
 
-  video_engine_base_->RegisterSendSideDelayObserver(channel_, &stats_proxy_);
-  video_engine_base_->RegisterSendStatisticsProxy(channel_, &stats_proxy_);
-
-  image_process_ = ViEImageProcess::GetInterface(video_engine);
-  image_process_->RegisterPreEncodeCallback(channel_,
-                                            config_.pre_encode_callback);
-  if (config_.post_encode_callback) {
-    image_process_->RegisterPostEncodeImageCallback(channel_,
-                                                    &encoded_frame_proxy_);
-  }
+  if (config_.post_encode_callback)
+    vie_encoder_->RegisterPostEncodeImageCallback(&encoded_frame_proxy_);
 
   if (config_.suspend_below_min_bitrate)
-    codec_->SuspendBelowMinBitrate(channel_);
+    vie_encoder_->SuspendBelowMinBitrate();
 
-  rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_,
-                                                       &stats_proxy_);
-  rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
-                                                      &stats_proxy_);
-  rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(channel_, &stats_proxy_);
-  rtp_rtcp_->RegisterSendBitrateObserver(channel_, &stats_proxy_);
-  rtp_rtcp_->RegisterSendFrameCountObserver(channel_, &stats_proxy_);
+  congestion_controller_->AddEncoder(vie_encoder_.get());
+  encoder_feedback_->AddEncoder(ssrcs, vie_encoder_.get());
 
-  codec_->RegisterEncoderObserver(channel_, stats_proxy_);
+  vie_channel_->RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_);
+  vie_channel_->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
+  vie_channel_->RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
+  vie_channel_->RegisterSendBitrateObserver(&stats_proxy_);
+  vie_channel_->RegisterSendFrameCountObserver(&stats_proxy_);
 }
 
 VideoSendStream::~VideoSendStream() {
-  capture_->DeregisterObserver(capture_id_);
-  codec_->DeregisterEncoderObserver(channel_);
+  LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
+  vie_channel_->RegisterSendFrameCountObserver(nullptr);
+  vie_channel_->RegisterSendBitrateObserver(nullptr);
+  vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);
+  vie_channel_->RegisterSendChannelRtpStatisticsCallback(nullptr);
+  vie_channel_->RegisterSendChannelRtcpStatisticsCallback(nullptr);
 
-  rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, &stats_proxy_);
-  rtp_rtcp_->DeregisterSendBitrateObserver(channel_, &stats_proxy_);
-  rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(channel_, nullptr);
-  rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_,
-                                                        &stats_proxy_);
-  rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_,
-                                                         &stats_proxy_);
+  // Remove capture input (thread) so that it's not running after the current
+  // channel is deleted.
+  input_.reset();
 
-  image_process_->DeRegisterPreEncodeCallback(channel_);
+  vie_encoder_->DeRegisterExternalEncoder(
+      config_.encoder_settings.payload_type);
 
-  network_->DeregisterSendTransport(channel_);
+  call_stats_->DeregisterStatsObserver(vie_channel_->GetStatsObserver());
+  congestion_controller_->SetChannelRembStatus(false, false,
+                                               vie_channel_->rtp_rtcp());
 
-  capture_->DisconnectCaptureDevice(channel_);
-  capture_->ReleaseCaptureDevice(capture_id_);
+  // Remove the feedback, stop all encoding threads and processing. This must be
+  // done before deleting the channel.
+  congestion_controller_->RemoveEncoder(vie_encoder_.get());
+  encoder_feedback_->RemoveEncoder(vie_encoder_.get());
+  vie_encoder_->StopThreadsAndRemoveSharedMembers();
 
-  external_codec_->DeRegisterExternalSendCodec(
-      channel_, config_.encoder_settings.payload_type);
-
-  video_engine_base_->DeleteChannel(channel_);
-
-  image_process_->Release();
-  video_engine_base_->Release();
-  capture_->Release();
-  codec_->Release();
-  if (external_codec_)
-    external_codec_->Release();
-  network_->Release();
-  rtp_rtcp_->Release();
+  uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
+  congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
+      remote_ssrc);
 }
 
-void VideoSendStream::IncomingCapturedFrame(const I420VideoFrame& frame) {
-  // TODO(pbos): Local rendering should not be done on the capture thread.
-  if (config_.local_renderer != nullptr)
-    config_.local_renderer->RenderFrame(frame, 0);
-
-  stats_proxy_.OnIncomingFrame();
-  external_capture_->IncomingFrame(frame);
+VideoCaptureInput* VideoSendStream::Input() {
+  return input_.get();
 }
 
-VideoSendStreamInput* VideoSendStream::Input() { return this; }
+CPULoadStateObserver* VideoSendStream::LoadStateObserver() {
+  return vie_encoder_.get();
+}
 
 void VideoSendStream::Start() {
   transport_adapter_.Enable();
-  video_engine_base_->StartSend(channel_);
-  video_engine_base_->StartReceive(channel_);
+  vie_encoder_->Pause();
+  if (vie_channel_->StartSend() == 0) {
+    // Was not already started, trigger a keyframe.
+    vie_encoder_->SendKeyFrame();
+  }
+  vie_encoder_->Restart();
+  vie_channel_->StartReceive();
 }
 
 void VideoSendStream::Stop() {
-  video_engine_base_->StopSend(channel_);
-  video_engine_base_->StopReceive(channel_);
+  // TODO(pbos): Make sure the encoder stops here.
+  vie_channel_->StopSend();
+  vie_channel_->StopReceive();
   transport_adapter_.Disable();
 }
 
@@ -295,8 +305,8 @@ bool VideoSendStream::ReconfigureVideoEncoder(
   TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
   const std::vector<VideoStream>& streams = config.streams;
-  DCHECK(!streams.empty());
-  DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
+  RTC_DCHECK(!streams.empty());
+  RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
 
   VideoCodec video_codec;
   memset(&video_codec, 0, sizeof(video_codec));
@@ -311,10 +321,10 @@ bool VideoSendStream::ReconfigureVideoEncoder(
   }
 
   switch (config.content_type) {
-    case VideoEncoderConfig::kRealtimeVideo:
+    case VideoEncoderConfig::ContentType::kRealtimeVideo:
       video_codec.mode = kRealtimeVideo;
       break;
-    case VideoEncoderConfig::kScreenshare:
+    case VideoEncoderConfig::ContentType::kScreen:
       video_codec.mode = kScreensharing;
       if (config.streams.size() == 1 &&
           config.streams[0].temporal_layer_thresholds_bps.size() == 1) {
@@ -332,6 +342,8 @@ bool VideoSendStream::ReconfigureVideoEncoder(
     video_codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
   }
 
+  video_codec.resolution_divisor = config.resolution_divisor;
+
   if (video_codec.codecType == kVideoCodecVP8) {
     if (config.encoder_specific_settings != nullptr) {
       video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>(
@@ -344,6 +356,12 @@ bool VideoSendStream::ReconfigureVideoEncoder(
     if (config.encoder_specific_settings != nullptr) {
       video_codec.codecSpecific.VP9 = *reinterpret_cast<const VideoCodecVP9*>(
                                           config.encoder_specific_settings);
+      if (video_codec.mode == kScreensharing) {
+        video_codec.codecSpecific.VP9.flexibleMode = true;
+        // For now VP9 screensharing use 1 temporal and 2 spatial layers.
+        RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers, 1);
+        RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2);
+      }
     }
     video_codec.codecSpecific.VP9.numberOfTemporalLayers =
         static_cast<unsigned char>(
@@ -352,10 +370,11 @@ bool VideoSendStream::ReconfigureVideoEncoder(
     if (config.encoder_specific_settings != nullptr) {
       video_codec.codecSpecific.H264 = *reinterpret_cast<const VideoCodecH264*>(
                                            config.encoder_specific_settings);
+
     }
   } else {
     // TODO(pbos): Support encoder_settings codec-agnostically.
-    DCHECK(config.encoder_specific_settings == nullptr)
+    RTC_DCHECK(config.encoder_specific_settings == nullptr)
         << "Encoder-specific settings for codec type not wired up.";
   }
 
@@ -367,21 +386,31 @@ bool VideoSendStream::ReconfigureVideoEncoder(
   video_codec.numberOfSimulcastStreams =
       static_cast<unsigned char>(streams.size());
   video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
-  DCHECK_LE(streams.size(), static_cast<size_t>(kMaxSimulcastStreams));
+  RTC_DCHECK_LE(streams.size(), static_cast<size_t>(kMaxSimulcastStreams));
+  if (video_codec.codecType == kVideoCodecVP9) {
+    // If the vector is empty, bitrates will be configured automatically.
+    RTC_DCHECK(config.spatial_layers.empty() ||
+               config.spatial_layers.size() ==
+                   video_codec.codecSpecific.VP9.numberOfSpatialLayers);
+    RTC_DCHECK_LE(video_codec.codecSpecific.VP9.numberOfSpatialLayers,
+                  kMaxSimulcastStreams);
+    for (size_t i = 0; i < config.spatial_layers.size(); ++i)
+      video_codec.spatialLayers[i] = config.spatial_layers[i];
+  }
   for (size_t i = 0; i < streams.size(); ++i) {
     SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
-    DCHECK_GT(streams[i].width, 0u);
-    DCHECK_GT(streams[i].height, 0u);
-    DCHECK_GT(streams[i].max_framerate, 0);
+    RTC_DCHECK_GT(streams[i].width, 0u);
+    RTC_DCHECK_GT(streams[i].height, 0u);
+    RTC_DCHECK_GT(streams[i].max_framerate, 0);
     // Different framerates not supported per stream at the moment.
-    DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
-    DCHECK_GE(streams[i].min_bitrate_bps, 0);
-    DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
-    DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
-    DCHECK_GE(streams[i].max_qp, 0);
+    RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
+    RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0);
+    RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
+    RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
+    RTC_DCHECK_GE(streams[i].max_qp, 0);
 
-    sim_stream->width = static_cast<unsigned short>(streams[i].width);
-    sim_stream->height = static_cast<unsigned short>(streams[i].height);
+    sim_stream->width = static_cast<uint16_t>(streams[i].width);
+    sim_stream->height = static_cast<uint16_t>(streams[i].height);
     sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
     sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
     sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
@@ -390,12 +419,12 @@ bool VideoSendStream::ReconfigureVideoEncoder(
         streams[i].temporal_layer_thresholds_bps.size() + 1);
 
     video_codec.width = std::max(video_codec.width,
-                                 static_cast<unsigned short>(streams[i].width));
+                                 static_cast<uint16_t>(streams[i].width));
     video_codec.height = std::max(
-        video_codec.height, static_cast<unsigned short>(streams[i].height));
+        video_codec.height, static_cast<uint16_t>(streams[i].height));
     video_codec.minBitrate =
-        std::min(video_codec.minBitrate,
-                 static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
+        std::min(static_cast<uint16_t>(video_codec.minBitrate),
+                 static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
     video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
     video_codec.qpMax = std::max(video_codec.qpMax,
                                  static_cast<unsigned int>(streams[i].max_qp));
@@ -405,20 +434,22 @@ bool VideoSendStream::ReconfigureVideoEncoder(
   // the bitrate controller is already set from Call.
   video_codec.startBitrate = 0;
 
-  if (video_codec.minBitrate < kViEMinCodecBitrate)
-    video_codec.minBitrate = kViEMinCodecBitrate;
-  if (video_codec.maxBitrate < kViEMinCodecBitrate)
-    video_codec.maxBitrate = kViEMinCodecBitrate;
-
-  DCHECK_GT(streams[0].max_framerate, 0);
+  RTC_DCHECK_GT(streams[0].max_framerate, 0);
   video_codec.maxFramerate = streams[0].max_framerate;
 
-  if (codec_->SetSendCodec(channel_, video_codec) != 0)
+  if (!SetSendCodec(video_codec))
     return false;
 
-  DCHECK_GE(config.min_transmit_bitrate_bps, 0);
-  rtp_rtcp_->SetMinTransmitBitrate(channel_,
-                                   config.min_transmit_bitrate_bps / 1000);
+  // Clear stats for disabled layers.
+  for (size_t i = video_codec.numberOfSimulcastStreams;
+       i < config_.rtp.ssrcs.size(); ++i) {
+    stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
+  }
+
+  stats_proxy_.SetContentType(config.content_type);
+
+  RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
+  vie_encoder_->SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
 
   encoder_config_ = config;
   use_config_bitrate_ = false;
@@ -426,22 +457,32 @@ bool VideoSendStream::ReconfigureVideoEncoder(
 }
 
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return network_->ReceivedRTCPPacket(channel_, packet, length) == 0;
+  return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
 }
 
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
+void VideoSendStream::OveruseDetected() {
+  if (config_.overuse_callback)
+    config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
+}
+
+void VideoSendStream::NormalUsage() {
+  if (config_.overuse_callback)
+    config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
+}
+
 void VideoSendStream::ConfigureSsrcs() {
-  rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.ssrcs.front());
+  vie_channel_->SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0);
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
-    rtp_rtcp_->SetLocalSSRC(
-        channel_, ssrc, kViEStreamTypeNormal, static_cast<unsigned char>(i));
+    vie_channel_->SetSSRC(ssrc, kViEStreamTypeNormal,
+                          static_cast<unsigned char>(i));
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
-      rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
+      vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
   }
 
   if (config_.rtp.rtx.ssrcs.empty()) {
@@ -449,64 +490,104 @@ void VideoSendStream::ConfigureSsrcs() {
   }
 
   // Set up RTX.
-  DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size());
+  RTC_DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size());
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
-    rtp_rtcp_->SetLocalSSRC(channel_,
-                            config_.rtp.rtx.ssrcs[i],
-                            kViEStreamTypeRtx,
-                            static_cast<unsigned char>(i));
+    vie_channel_->SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx,
+                          static_cast<unsigned char>(i));
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
-      rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
+      vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
   }
 
-  DCHECK_GE(config_.rtp.rtx.payload_type, 0);
-  rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
+  RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0);
+  vie_channel_->SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
+                                      config_.encoder_settings.payload_type);
 }
 
 std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
   std::map<uint32_t, RtpState> rtp_states;
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
-    rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
+    rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
-    rtp_states[ssrc] = rtp_rtcp_->GetRtpStateForSsrc(channel_, ssrc);
+    rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
   }
 
   return rtp_states;
 }
 
-void VideoSendStream::SignalNetworkState(Call::NetworkState state) {
+void VideoSendStream::SignalNetworkState(NetworkState state) {
   // When network goes up, enable RTCP status before setting transmission state.
   // When it goes down, disable RTCP afterwards. This ensures that any packets
   // sent due to the network state changed will not be dropped.
-  if (state == Call::kNetworkUp)
-    rtp_rtcp_->SetRTCPStatus(channel_, kRtcpCompound_RFC4585);
-  network_->SetNetworkTransmissionState(channel_, state == Call::kNetworkUp);
-  if (state == Call::kNetworkDown)
-    rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNone);
-}
-
-int64_t VideoSendStream::GetPacerQueuingDelayMs() const {
-  int64_t pacer_delay_ms = 0;
-  if (rtp_rtcp_->GetPacerQueuingDelayMs(channel_, &pacer_delay_ms) != 0) {
-    return 0;
-  }
-  return pacer_delay_ms;
+  if (state == kNetworkUp)
+    vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
+  vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
+  if (state == kNetworkDown)
+    vie_channel_->SetRTCPMode(RtcpMode::kOff);
 }
 
 int64_t VideoSendStream::GetRtt() const {
   webrtc::RtcpStatistics rtcp_stats;
+  uint16_t frac_lost;
+  uint32_t cumulative_lost;
+  uint32_t extended_max_sequence_number;
+  uint32_t jitter;
   int64_t rtt_ms;
-  if (rtp_rtcp_->GetSendChannelRtcpStatistics(channel_, rtcp_stats, rtt_ms) ==
-      0) {
+  if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
+                                          &extended_max_sequence_number,
+                                          &jitter, &rtt_ms) == 0) {
     return rtt_ms;
   }
   return -1;
 }
+
+int VideoSendStream::GetPaddingNeededBps() const {
+  return vie_encoder_->GetPaddingNeededBps();
+}
+
+bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
+  static const int kEncoderMinBitrate = 30;
+  if (video_codec.maxBitrate == 0) {
+    // Unset max bitrate -> cap to one bit per pixel.
+    video_codec.maxBitrate =
+        (video_codec.width * video_codec.height * video_codec.maxFramerate) /
+        1000;
+  }
+
+  if (video_codec.minBitrate < kEncoderMinBitrate)
+    video_codec.minBitrate = kEncoderMinBitrate;
+  if (video_codec.maxBitrate < kEncoderMinBitrate)
+    video_codec.maxBitrate = kEncoderMinBitrate;
+
+  // Stop the media flow while reconfiguring.
+  vie_encoder_->Pause();
+
+  if (vie_encoder_->SetEncoder(video_codec) != 0) {
+    LOG(LS_ERROR) << "Failed to set encoder.";
+    return false;
+  }
+
+  if (vie_channel_->SetSendCodec(video_codec, false) != 0) {
+    LOG(LS_ERROR) << "Failed to set send codec.";
+    return false;
+  }
+
+  // Not all configured SSRCs have to be utilized (simulcast senders don't have
+  // to send on all SSRCs at once etc.)
+  std::vector<uint32_t> used_ssrcs = config_.rtp.ssrcs;
+  used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
+  vie_encoder_->SetSsrcs(used_ssrcs);
+
+  // Restart the media flow
+  vie_encoder_->Restart();
+
+  return true;
+}
+
 }  // namespace internal
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream.h b/media/webrtc/trunk/webrtc/video/video_send_stream.h
index a5cd1ce77c..f227b8b66c 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream.h
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream.h
@@ -15,93 +15,88 @@
 #include <vector>
 
 #include "webrtc/call.h"
+#include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/send_statistics_proxy.h"
-#include "webrtc/video/transport_adapter.h"
+#include "webrtc/video/video_capture_input.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
-class CpuOveruseObserver;
-class VideoEngine;
-class ViEBase;
-class ViECapture;
-class ViECodec;
-class ViEExternalCapture;
-class ViEExternalCodec;
-class ViEImageProcess;
-class ViENetwork;
-class ViERTP_RTCP;
+class BitrateAllocator;
+class CallStats;
+class CongestionController;
+class EncoderStateFeedback;
+class ProcessThread;
+class ViEChannel;
+class ViEEncoder;
 
 namespace internal {
 
 class VideoSendStream : public webrtc::VideoSendStream,
-                        public VideoSendStreamInput {
+                        public webrtc::CpuOveruseObserver {
  public:
-  VideoSendStream(newapi::Transport* transport,
-                  CpuOveruseObserver* overuse_observer,
-                  webrtc::VideoEngine* video_engine,
+  VideoSendStream(int num_cpu_cores,
+                  ProcessThread* module_process_thread,
+                  CallStats* call_stats,
+                  CongestionController* congestion_controller,
+                  BitrateAllocator* bitrate_allocator,
                   const VideoSendStream::Config& config,
                   const VideoEncoderConfig& encoder_config,
-                  const std::map<uint32_t, RtpState>& suspended_ssrcs,
-                  int base_channel);
+                  const std::map<uint32_t, RtpState>& suspended_ssrcs);
 
-  virtual ~VideoSendStream();
+  ~VideoSendStream() override;
 
+  // webrtc::SendStream implementation.
   void Start() override;
   void Stop() override;
+  void SignalNetworkState(NetworkState state) override;
+  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
 
+  // webrtc::VideoSendStream implementation.
+  VideoCaptureInput* Input() override;
+  CPULoadStateObserver* LoadStateObserver() override;
   bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) override;
-
   Stats GetStats() override;
 
-  bool DeliverRtcp(const uint8_t* packet, size_t length);
-
-  // From VideoSendStreamInput.
-  void IncomingCapturedFrame(const I420VideoFrame& frame) override;
-
-  // From webrtc::VideoSendStream.
-  VideoSendStreamInput* Input() override;
+  // webrtc::CpuOveruseObserver implementation.
+  void OveruseDetected() override;
+  void NormalUsage() override;
 
   typedef std::map<uint32_t, RtpState> RtpStateMap;
   RtpStateMap GetRtpStates() const;
 
-  void SignalNetworkState(Call::NetworkState state);
-
-  int64_t GetPacerQueuingDelayMs() const;
-
   int64_t GetRtt() const;
+  int GetPaddingNeededBps() const;
 
  private:
+  bool SetSendCodec(VideoCodec video_codec);
   void ConfigureSsrcs();
+
+  SendStatisticsProxy stats_proxy_;
   TransportAdapter transport_adapter_;
   EncodedFrameCallbackAdapter encoded_frame_proxy_;
   const VideoSendStream::Config config_;
   VideoEncoderConfig encoder_config_;
   std::map<uint32_t, RtpState> suspended_ssrcs_;
 
-  ViEBase* video_engine_base_;
-  ViECapture* capture_;
-  ViECodec* codec_;
-  ViEExternalCapture* external_capture_;
-  ViEExternalCodec* external_codec_;
-  ViENetwork* network_;
-  ViERTP_RTCP* rtp_rtcp_;
-  ViEImageProcess* image_process_;
+  ProcessThread* const module_process_thread_;
+  CallStats* const call_stats_;
+  CongestionController* const congestion_controller_;
 
-  int channel_;
-  int capture_id_;
+  rtc::scoped_ptr<VideoCaptureInput> input_;
+  rtc::scoped_ptr<ViEChannel> vie_channel_;
+  rtc::scoped_ptr<ViEEncoder> vie_encoder_;
+  rtc::scoped_ptr<EncoderStateFeedback> encoder_feedback_;
 
   // Used as a workaround to indicate that we should be using the configured
   // start bitrate initially, instead of the one reported by VideoEngine (which
   // defaults to too high).
   bool use_config_bitrate_;
-
-  SendStatisticsProxy stats_proxy_;
 };
 }  // namespace internal
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc b/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc
index 428a2735c1..f0f1ca4d2c 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc
@@ -12,79 +12,78 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/common_video/interface/native_handle.h"
+#include "webrtc/call/transport_adapter.h"
 #include "webrtc/frame_callback.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/configurable_frame_size_encoder.h"
+#include "webrtc/test/fake_texture_frame.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/video/send_statistics_proxy.h"
-#include "webrtc/video/transport_adapter.h"
+#include "webrtc/video_frame.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 enum VideoFormat { kGeneric, kVP8, };
 
-void ExpectEqualFrames(const I420VideoFrame& frame1,
-                       const I420VideoFrame& frame2);
-void ExpectEqualTextureFrames(const I420VideoFrame& frame1,
-                              const I420VideoFrame& frame2);
-void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
-                             const I420VideoFrame& frame2);
-void ExpectEqualFramesVector(const std::vector<I420VideoFrame>& frames1,
-                             const std::vector<I420VideoFrame>& frames2);
-I420VideoFrame CreateI420VideoFrame(int width, int height, uint8_t data);
-
-class FakeNativeHandle : public NativeHandle {
- public:
-  FakeNativeHandle() {}
-  virtual ~FakeNativeHandle() {}
-  virtual void* GetHandle() { return nullptr; }
-};
+void ExpectEqualFrames(const VideoFrame& frame1, const VideoFrame& frame2);
+void ExpectEqualTextureFrames(const VideoFrame& frame1,
+                              const VideoFrame& frame2);
+void ExpectEqualBufferFrames(const VideoFrame& frame1,
+                             const VideoFrame& frame2);
+void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
+                             const std::vector<VideoFrame>& frames2);
+VideoFrame CreateVideoFrame(int width, int height, uint8_t data);
 
 class VideoSendStreamTest : public test::CallTest {
  protected:
   void TestNackRetransmission(uint32_t retransmit_ssrc,
                               uint8_t retransmit_payload_type);
   void TestPacketFragmentationSize(VideoFormat format, bool with_fec);
+
+  void TestVp9NonFlexMode(uint8_t num_temporal_layers,
+                          uint8_t num_spatial_layers);
 };
 
 TEST_F(VideoSendStreamTest, CanStartStartedStream) {
-  test::NullTransport transport;
-  Call::Config call_config(&transport);
+  Call::Config call_config;
   CreateSenderCall(call_config);
 
-  CreateSendConfig(1);
-  CreateStreams();
-  send_stream_->Start();
-  send_stream_->Start();
+  test::NullTransport transport;
+  CreateSendConfig(1, 0, &transport);
+  CreateVideoStreams();
+  video_send_stream_->Start();
+  video_send_stream_->Start();
   DestroyStreams();
 }
 
 TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
-  test::NullTransport transport;
-  Call::Config call_config(&transport);
+  Call::Config call_config;
   CreateSenderCall(call_config);
 
-  CreateSendConfig(1);
-  CreateStreams();
-  send_stream_->Stop();
-  send_stream_->Stop();
+  test::NullTransport transport;
+  CreateSendConfig(1, 0, &transport);
+  CreateVideoStreams();
+  video_send_stream_->Stop();
+  video_send_stream_->Stop();
   DestroyStreams();
 }
 
@@ -100,10 +99,10 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        if (packet_type == RTCPUtility::kRtcpSdesChunkCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kSdesChunk) {
           EXPECT_EQ(parser.Packet().CName.CName, kCName);
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
 
         packet_type = parser.Iterate();
@@ -112,15 +111,15 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->rtp.c_name = kCName;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for RTCP with CNAME.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
     }
   } test;
 
@@ -128,12 +127,11 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
 }
 
 TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
-  static const uint8_t kAbsSendTimeExtensionId = 13;
   class AbsoluteSendTimeObserver : public test::SendTest {
    public:
     AbsoluteSendTimeObserver() : SendTest(kDefaultTimeoutMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
+          kRtpExtensionAbsoluteSendTime, test::kAbsSendTimeExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -144,21 +142,22 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
       EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
       EXPECT_EQ(header.extension.transmissionTimeOffset, 0);
       EXPECT_GT(header.extension.absoluteSendTime, 0u);
-      observation_complete_->Set();
+      observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for single RTP packet.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
     }
   } test;
 
@@ -166,7 +165,6 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
 }
 
 TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
-  static const uint8_t kTOffsetExtensionId = 13;
   static const int kEncodeDelayMs = 5;
   class TransmissionTimeOffsetObserver : public test::SendTest {
    public:
@@ -174,7 +172,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
         : SendTest(kDefaultTimeoutMs),
           encoder_(Clock::GetRealTimeClock(), kEncodeDelayMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId));
+          kRtpExtensionTransmissionTimeOffset, test::kTOffsetExtensionId));
     }
 
    private:
@@ -186,22 +184,23 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
       EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
       EXPECT_GT(header.extension.transmissionTimeOffset, 0);
       EXPECT_EQ(header.extension.absoluteSendTime, 0u);
-      observation_complete_->Set();
+      observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = &encoder_;
+      send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
+          RtpExtension(RtpExtension::kTOffset, test::kTOffsetExtensionId));
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for a single RTP packet.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
     }
 
     test::DelayedEncoder encoder_;
@@ -210,6 +209,50 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
   RunBaseTest(&test);
 }
 
+TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
+  static const uint8_t kExtensionId = 13;
+  class TransportWideSequenceNumberObserver : public test::SendTest {
+   public:
+    TransportWideSequenceNumberObserver()
+        : SendTest(kDefaultTimeoutMs), encoder_(Clock::GetRealTimeClock()) {
+      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionTransportSequenceNumber, kExtensionId));
+    }
+
+   private:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+      EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
+      EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
+
+      observation_complete_.Set();
+
+      return SEND_PACKET;
+    }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->encoder_settings.encoder = &encoder_;
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+    }
+
+    test::FakeEncoder encoder_;
+  } test;
+
+  RunBaseTest(&test);
+}
+
 class FakeReceiveStatistics : public NullReceiveStatistics {
  public:
   FakeReceiveStatistics(uint32_t send_ssrc,
@@ -250,7 +293,6 @@ class FakeReceiveStatistics : public NullReceiveStatistics {
     void GetReceiveStreamDataCounters(
         StreamDataCounters* data_counters) const override {}
     uint32_t BitrateReceived() const override { return 0; }
-    void ResetStatistics() override {}
     bool IsRetransmitOfOldPacket(const RTPHeader& header,
                                  int64_t min_rtt) const override {
       return false;
@@ -267,72 +309,125 @@ class FakeReceiveStatistics : public NullReceiveStatistics {
   StatisticianMap stats_map_;
 };
 
-TEST_F(VideoSendStreamTest, SupportsFec) {
-  class FecObserver : public test::SendTest {
-   public:
-    FecObserver()
-        : SendTest(kDefaultTimeoutMs),
-          transport_adapter_(SendTransport()),
-          send_count_(0),
-          received_media_(false),
-          received_fec_(false) {
-      transport_adapter_.Enable();
+class FecObserver : public test::SendTest {
+ public:
+  explicit FecObserver(bool header_extensions_enabled)
+      : SendTest(VideoSendStreamTest::kDefaultTimeoutMs),
+        send_count_(0),
+        received_media_(false),
+        received_fec_(false),
+        header_extensions_enabled_(header_extensions_enabled) {}
+
+ private:
+  Action OnSendRtp(const uint8_t* packet, size_t length) override {
+    RTPHeader header;
+    EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+    // Send lossy receive reports to trigger FEC enabling.
+    if (send_count_++ % 2 != 0) {
+      // Receive statistics reporting having lost 50% of the packets.
+      FakeReceiveStatistics lossy_receive_stats(
+          VideoSendStreamTest::kVideoSendSsrcs[0], header.sequenceNumber,
+          send_count_ / 2, 127);
+      RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
+                             &lossy_receive_stats, nullptr,
+                             transport_adapter_.get());
+
+      rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
+      rtcp_sender.SetRemoteSSRC(VideoSendStreamTest::kVideoSendSsrcs[0]);
+
+      RTCPSender::FeedbackState feedback_state;
+
+      EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
     }
 
-   private:
-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      RTPHeader header;
-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+    int encapsulated_payload_type = -1;
+    if (header.payloadType == VideoSendStreamTest::kRedPayloadType) {
+      encapsulated_payload_type = static_cast<int>(packet[header.headerLength]);
+      if (encapsulated_payload_type !=
+          VideoSendStreamTest::kFakeVideoSendPayloadType)
+        EXPECT_EQ(VideoSendStreamTest::kUlpfecPayloadType,
+                  encapsulated_payload_type);
+    } else {
+      EXPECT_EQ(VideoSendStreamTest::kFakeVideoSendPayloadType,
+                header.payloadType);
+    }
 
-      // Send lossy receive reports to trigger FEC enabling.
-      if (send_count_++ % 2 != 0) {
-        // Receive statistics reporting having lost 50% of the packets.
-        FakeReceiveStatistics lossy_receive_stats(
-            kSendSsrcs[0], header.sequenceNumber, send_count_ / 2, 127);
-        RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
-                               &lossy_receive_stats, nullptr);
-        EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
-
-        rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
-        rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
-
-        RTCPSender::FeedbackState feedback_state;
-
-        EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
+    if (header_extensions_enabled_) {
+      EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
+      uint32_t kHalf24BitsSpace = 0xFFFFFF / 2;
+      if (header.extension.absoluteSendTime <= kHalf24BitsSpace &&
+          prev_header_.extension.absoluteSendTime > kHalf24BitsSpace) {
+        // 24 bits wrap.
+        EXPECT_GT(prev_header_.extension.absoluteSendTime,
+                  header.extension.absoluteSendTime);
+      } else {
+        EXPECT_GE(header.extension.absoluteSendTime,
+                  prev_header_.extension.absoluteSendTime);
       }
+      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
+      uint16_t seq_num_diff = header.extension.transportSequenceNumber -
+                              prev_header_.extension.transportSequenceNumber;
+      EXPECT_EQ(1, seq_num_diff);
+    }
 
-      EXPECT_EQ(kRedPayloadType, header.payloadType);
-
-      uint8_t encapsulated_payload_type = packet[header.headerLength];
-
-      if (encapsulated_payload_type == kUlpfecPayloadType) {
+    if (encapsulated_payload_type != -1) {
+      if (encapsulated_payload_type ==
+          VideoSendStreamTest::kUlpfecPayloadType) {
         received_fec_ = true;
       } else {
         received_media_ = true;
       }
-
-      if (received_media_ && received_fec_)
-        observation_complete_->Set();
-
-      return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
-      send_config->rtp.fec.red_payload_type = kRedPayloadType;
-      send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
-    }
+    if (received_media_ && received_fec_ && send_count_ > 100)
+      observation_complete_.Set();
 
-    void PerformTest() override {
-      EXPECT_TRUE(Wait()) << "Timed out waiting for FEC and media packets.";
-    }
+    prev_header_ = header;
 
-    internal::TransportAdapter transport_adapter_;
-    int send_count_;
-    bool received_media_;
-    bool received_fec_;
-  } test;
+    return SEND_PACKET;
+  }
+
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override {
+    transport_adapter_.reset(
+        new internal::TransportAdapter(send_config->send_transport));
+    transport_adapter_->Enable();
+    send_config->rtp.fec.red_payload_type =
+        VideoSendStreamTest::kRedPayloadType;
+    send_config->rtp.fec.ulpfec_payload_type =
+        VideoSendStreamTest::kUlpfecPayloadType;
+    if (header_extensions_enabled_) {
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber,
+                       test::kTransportSequenceNumberExtensionId));
+    }
+  }
+
+  void PerformTest() override {
+    EXPECT_TRUE(Wait()) << "Timed out waiting for FEC and media packets.";
+  }
+
+  rtc::scoped_ptr<internal::TransportAdapter> transport_adapter_;
+  int send_count_;
+  bool received_media_;
+  bool received_fec_;
+  bool header_extensions_enabled_;
+  RTPHeader prev_header_;
+};
+
+TEST_F(VideoSendStreamTest, SupportsFecWithExtensions) {
+  FecObserver test(true);
+
+  RunBaseTest(&test);
+}
+
+TEST_F(VideoSendStreamTest, SupportsFecWithoutExtensions) {
+  FecObserver test(false);
 
   RunBaseTest(&test);
 }
@@ -345,12 +440,10 @@ void VideoSendStreamTest::TestNackRetransmission(
     explicit NackObserver(uint32_t retransmit_ssrc,
                           uint8_t retransmit_payload_type)
         : SendTest(kDefaultTimeoutMs),
-          transport_adapter_(SendTransport()),
           send_count_(0),
           retransmit_ssrc_(retransmit_ssrc),
           retransmit_payload_type_(retransmit_payload_type),
           nacked_sequence_number_(-1) {
-      transport_adapter_.Enable();
     }
 
    private:
@@ -363,12 +456,11 @@ void VideoSendStreamTest::TestNackRetransmission(
         uint16_t nack_sequence_number = header.sequenceNumber - 1;
         nacked_sequence_number_ = nack_sequence_number;
         NullReceiveStatistics null_stats;
-        RTCPSender rtcp_sender(
-            0, false, Clock::GetRealTimeClock(), &null_stats, nullptr);
-        EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
+        RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &null_stats,
+                               nullptr, transport_adapter_.get());
 
-        rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
-        rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+        rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
+        rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
@@ -380,8 +472,8 @@ void VideoSendStreamTest::TestNackRetransmission(
       uint16_t sequence_number = header.sequenceNumber;
 
       if (header.ssrc == retransmit_ssrc_ &&
-          retransmit_ssrc_ != kSendSsrcs[0]) {
-        // Not kSendSsrcs[0], assume correct RTX packet. Extract sequence
+          retransmit_ssrc_ != kVideoSendSsrcs[0]) {
+        // Not kVideoSendSsrcs[0], assume correct RTX packet. Extract sequence
         // number.
         const uint8_t* rtx_header = packet + header.headerLength;
         sequence_number = (rtx_header[0] << 8) + rtx_header[1];
@@ -390,27 +482,30 @@ void VideoSendStreamTest::TestNackRetransmission(
       if (sequence_number == nacked_sequence_number_) {
         EXPECT_EQ(retransmit_ssrc_, header.ssrc);
         EXPECT_EQ(retransmit_payload_type_, header.payloadType);
-        observation_complete_->Set();
+        observation_complete_.Set();
       }
 
       return SEND_PACKET;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      transport_adapter_.reset(
+          new internal::TransportAdapter(send_config->send_transport));
+      transport_adapter_->Enable();
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->rtp.rtx.payload_type = retransmit_payload_type_;
-      if (retransmit_ssrc_ != kSendSsrcs[0])
+      if (retransmit_ssrc_ != kVideoSendSsrcs[0])
         send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for NACK retransmission.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for NACK retransmission.";
     }
 
-    internal::TransportAdapter transport_adapter_;
+    rtc::scoped_ptr<internal::TransportAdapter> transport_adapter_;
     int send_count_;
     uint32_t retransmit_ssrc_;
     uint8_t retransmit_payload_type_;
@@ -422,7 +517,7 @@ void VideoSendStreamTest::TestNackRetransmission(
 
 TEST_F(VideoSendStreamTest, RetransmitsNack) {
   // Normal NACKs should use the send SSRC.
-  TestNackRetransmission(kSendSsrcs[0], kFakeSendPayloadType);
+  TestNackRetransmission(kVideoSendSsrcs[0], kFakeVideoSendPayloadType);
 }
 
 TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
@@ -450,7 +545,6 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
                            bool test_generic_packetization,
                            bool use_fec)
         : SendTest(kLongTimeoutMs),
-          transport_adapter_(SendTransport()),
           encoder_(stop),
           max_packet_size_(max_packet_size),
           stop_size_(stop_size),
@@ -464,8 +558,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
           current_size_frame_(static_cast<int32_t>(start_size)) {
       // Fragmentation required, this test doesn't make sense without it.
       encoder_.SetFrameSize(start_size);
-      DCHECK_GT(stop_size, max_packet_size);
-      transport_adapter_.Enable();
+      RTC_DCHECK_GT(stop_size, max_packet_size);
     }
 
    private:
@@ -492,10 +585,15 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
         TriggerLossReport(header);
 
       if (test_generic_packetization_) {
-        size_t overhead = header.headerLength + header.paddingLength +
-                          (1 /* Generic header */);
-        if (use_fec_)
-          overhead += 1;  // RED for FEC header.
+        size_t overhead = header.headerLength + header.paddingLength;
+        // Only remove payload header and RED header if the packet actually
+        // contains payload.
+        if (length > overhead) {
+          overhead += (1 /* Generic header */);
+          if (use_fec_)
+            overhead += 1;  // RED for FEC header.
+        }
+        EXPECT_GE(length, overhead);
         accumulated_payload_ += length - overhead;
       }
 
@@ -520,7 +618,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
         accumulated_payload_ = 0;
         if (current_size_rtp_ == stop_size_) {
           // Done! (Don't increase size again, might arrive more @ stop_size).
-          observation_complete_->Set();
+          observation_complete_.Set();
         } else {
           // Increase next expected frame size. If testing with FEC, make sure
           // a FEC packet has been received for this frame size before
@@ -544,13 +642,13 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
       if (packet_count_++ % 2 != 0) {
         // Receive statistics reporting having lost 50% of the packets.
         FakeReceiveStatistics lossy_receive_stats(
-            kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
-        RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
-                               &lossy_receive_stats, nullptr);
-        EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
+            kVideoSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
+        RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
+                               &lossy_receive_stats, nullptr,
+                               transport_adapter_.get());
 
-        rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
-        rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+        rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
+        rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
@@ -569,15 +667,19 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
     }
 
     Call::Config GetSenderCallConfig() override {
-      Call::Config config(SendTransport());
+      Call::Config config;
       const int kMinBitrateBps = 30000;
       config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
       return config;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      transport_adapter_.reset(
+          new internal::TransportAdapter(send_config->send_transport));
+      transport_adapter_->Enable();
       if (use_fec_) {
         send_config->rtp.fec.red_payload_type = kRedPayloadType;
         send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
@@ -590,19 +692,16 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
       send_config->rtp.max_packet_size = kMaxPacketSize;
       send_config->post_encode_callback = this;
 
-      // Add an extension header, to make the RTP header larger than the base
-      // length of 12 bytes.
-      static const uint8_t kAbsSendTimeExtensionId = 13;
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+      // Make sure there is at least one extension header, to make the RTP
+      // header larger than the base length of 12 bytes.
+      EXPECT_FALSE(send_config->rtp.extensions.empty());
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while observing incoming RTP packets.";
+      EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets.";
     }
 
-    internal::TransportAdapter transport_adapter_;
+    rtc::scoped_ptr<internal::TransportAdapter> transport_adapter_;
     test::ConfigurableFrameSizeEncoder encoder_;
 
     const size_t max_packet_size_;
@@ -660,29 +759,18 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
    public:
     RembObserver()
         : SendTest(kDefaultTimeoutMs),
-          transport_adapter_(&transport_),
           clock_(Clock::GetRealTimeClock()),
-          crit_(CriticalSectionWrapper::CreateCriticalSection()),
           test_state_(kBeforeSuspend),
           rtp_count_(0),
           last_sequence_number_(0),
           suspended_frame_count_(0),
           low_remb_bps_(0),
           high_remb_bps_(0) {
-      transport_adapter_.Enable();
     }
 
    private:
-    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
-      // Receive statistics reporting having lost 0% of the packets.
-      // This is needed for the send-side bitrate controller to work properly.
-      CriticalSectionScoped lock(crit_.get());
-      SendRtcpFeedback(0);  // REMB is only sent if value is > 0.
-      return SEND_PACKET;
-    }
-
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       ++rtp_count_;
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
@@ -698,26 +786,29 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
           // counter.
           suspended_frame_count_ = 0;
         }
+        SendRtcpFeedback(0);  // REMB is only sent if value is > 0.
       } else if (test_state_ == kWaitingForPacket) {
         if (header.paddingLength == 0) {
           // Non-padding packet observed. Test is almost complete. Will just
           // have to wait for the stats to change.
           test_state_ = kWaitingForStats;
         }
+        SendRtcpFeedback(0);  // REMB is only sent if value is > 0.
       } else if (test_state_ == kWaitingForStats) {
         VideoSendStream::Stats stats = stream_->GetStats();
         if (stats.suspended == false) {
           // Stats flipped to false. Test is complete.
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
+        SendRtcpFeedback(0);  // REMB is only sent if value is > 0.
       }
 
       return SEND_PACKET;
     }
 
     // This method implements the I420FrameCallback.
-    void FrameCallback(I420VideoFrame* video_frame) override {
-      CriticalSectionScoped lock(crit_.get());
+    void FrameCallback(VideoFrame* video_frame) override {
+      rtc::CritScope lock(&crit_);
       if (test_state_ == kDuringSuspend &&
           ++suspended_frame_count_ > kSuspendTimeFrames) {
         VideoSendStream::Stats stats = stream_->GetStats();
@@ -728,29 +819,28 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
     }
 
     void set_low_remb_bps(int value) {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       low_remb_bps_ = value;
     }
 
     void set_high_remb_bps(int value) {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       high_remb_bps_ = value;
     }
 
-    void SetReceivers(PacketReceiver* send_transport_receiver,
-                      PacketReceiver* receive_transport_receiver) override {
-      transport_.SetReceiver(send_transport_receiver);
-    }
-
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       stream_ = send_stream;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      transport_adapter_.reset(
+          new internal::TransportAdapter(send_config->send_transport));
+      transport_adapter_->Enable();
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->pre_encode_callback = this;
       send_config->suspend_below_min_bitrate = true;
@@ -763,9 +853,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out during suspend-below-min-bitrate test.";
-      transport_.StopSending();
+      EXPECT_TRUE(Wait()) << "Timed out during suspend-below-min-bitrate test.";
     }
 
     enum TestState {
@@ -777,13 +865,13 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
 
     virtual void SendRtcpFeedback(int remb_value)
         EXCLUSIVE_LOCKS_REQUIRED(crit_) {
-      FakeReceiveStatistics receive_stats(
-          kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
-      RTCPSender rtcp_sender(0, false, clock_, &receive_stats, nullptr);
-      EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
+      FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0],
+                                          last_sequence_number_, rtp_count_, 0);
+      RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr,
+                             transport_adapter_.get());
 
-      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
-      rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+      rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
+      rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
       if (remb_value > 0) {
         rtcp_sender.SetREMBStatus(true);
         rtcp_sender.SetREMBData(remb_value, std::vector<uint32_t>());
@@ -792,12 +880,11 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
       EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
     }
 
-    internal::TransportAdapter transport_adapter_;
-    test::DirectTransport transport_;
+    rtc::scoped_ptr<internal::TransportAdapter> transport_adapter_;
     Clock* const clock_;
     VideoSendStream* stream_;
 
-    const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+    rtc::CriticalSection crit_;
     TestState test_state_ GUARDED_BY(crit_);
     int rtp_count_ GUARDED_BY(crit_);
     int last_sequence_number_ GUARDED_BY(crit_);
@@ -815,36 +902,32 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
     NoPaddingWhenVideoIsMuted()
         : SendTest(kDefaultTimeoutMs),
           clock_(Clock::GetRealTimeClock()),
-          transport_adapter_(ReceiveTransport()),
-          crit_(CriticalSectionWrapper::CreateCriticalSection()),
           last_packet_time_ms_(-1),
           capturer_(nullptr) {
-      transport_adapter_.Enable();
     }
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       last_packet_time_ms_ = clock_->TimeInMilliseconds();
       capturer_->Stop();
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       const int kVideoMutedThresholdMs = 10000;
       if (last_packet_time_ms_ > 0 &&
           clock_->TimeInMilliseconds() - last_packet_time_ms_ >
               kVideoMutedThresholdMs)
-        observation_complete_->Set();
+        observation_complete_.Set();
       // Receive statistics reporting having lost 50% of the packets.
-      FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
-      RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
-                             &receive_stats, nullptr);
-      EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
+      FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0], 1, 1, 0);
+      RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &receive_stats,
+                             nullptr, transport_adapter_.get());
 
-      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
-      rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+      rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
+      rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
 
       RTCPSender::FeedbackState feedback_state;
 
@@ -852,28 +935,31 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
       return SEND_PACKET;
     }
 
-    void SetReceivers(PacketReceiver* send_transport_receiver,
-                      PacketReceiver* receive_transport_receiver) override {
-      RtpRtcpObserver::SetReceivers(send_transport_receiver,
-                                    send_transport_receiver);
+    test::PacketTransport* CreateReceiveTransport() override {
+      test::PacketTransport* transport = new test::PacketTransport(
+          nullptr, this, test::PacketTransport::kReceiver,
+          FakeNetworkPipe::Config());
+      transport_adapter_.reset(new internal::TransportAdapter(transport));
+      transport_adapter_->Enable();
+      return transport;
     }
 
-    size_t GetNumStreams() const override { return 3; }
+    size_t GetNumVideoStreams() const override { return 3; }
 
     virtual void OnFrameGeneratorCapturerCreated(
         test::FrameGeneratorCapturer* frame_generator_capturer) {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       capturer_ = frame_generator_capturer;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for RTP packets to stop being sent.";
     }
 
     Clock* const clock_;
-    internal::TransportAdapter transport_adapter_;
-    const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+    rtc::scoped_ptr<internal::TransportAdapter> transport_adapter_;
+    rtc::CriticalSection crit_;
     int64_t last_packet_time_ms_ GUARDED_BY(crit_);
     test::FrameGeneratorCapturer* capturer_ GUARDED_BY(crit_);
   } test;
@@ -893,36 +979,22 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
   static const int kHighBitrateBps = 150000;
   static const int kRembBitrateBps = 80000;
   static const int kRembRespectedBitrateBps = 100000;
-  class BitrateObserver : public test::SendTest, public PacketReceiver {
+  class BitrateObserver : public test::SendTest {
    public:
     BitrateObserver()
         : SendTest(kDefaultTimeoutMs),
-          feedback_transport_(ReceiveTransport()),
           bitrate_capped_(false) {
-      RtpRtcp::Configuration config;
-      feedback_transport_.Enable();
-      config.outgoing_transport = &feedback_transport_;
-      rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
-      rtp_rtcp_->SetREMBStatus(true);
-      rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
-    }
-
-    void OnStreamsCreated(
-        VideoSendStream* send_stream,
-        const std::vector<VideoReceiveStream*>& receive_streams) override {
-      stream_ = send_stream;
     }
 
    private:
-    DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                 size_t length) override {
+    virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
       if (RtpHeaderParser::IsRtcp(packet, length))
-        return DELIVERY_OK;
+        return DROP_PACKET;
 
       RTPHeader header;
       if (!parser_->Parse(packet, length, &header))
-        return DELIVERY_PACKET_ERROR;
-      DCHECK(stream_ != nullptr);
+        return DROP_PACKET;
+      RTC_DCHECK(stream_ != nullptr);
       VideoSendStream::Stats stats = stream_->GetStats();
       if (!stats.substreams.empty()) {
         EXPECT_EQ(1u, stats.substreams.size());
@@ -941,30 +1013,41 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
           bitrate_capped_ = true;
         } else if (bitrate_capped_ &&
                    total_bitrate_bps < kRembRespectedBitrateBps) {
-          observation_complete_->Set();
+          observation_complete_.Set();
         }
       }
-      return DELIVERY_OK;
+      // Packets don't have to be delivered since the test is the receiver.
+      return DROP_PACKET;
     }
 
-    void SetReceivers(PacketReceiver* send_transport_receiver,
-                      PacketReceiver* receive_transport_receiver) override {
-      RtpRtcpObserver::SetReceivers(this, send_transport_receiver);
+    void OnVideoStreamsCreated(
+        VideoSendStream* send_stream,
+        const std::vector<VideoReceiveStream*>& receive_streams) override {
+      stream_ = send_stream;
+      RtpRtcp::Configuration config;
+      config.outgoing_transport = feedback_transport_.get();
+      rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
+      rtp_rtcp_->SetREMBStatus(true);
+      rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      feedback_transport_.reset(
+          new internal::TransportAdapter(send_config->send_transport));
+      feedback_transport_->Enable();
       encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timeout while waiting for low bitrate stats after REMB.";
     }
 
     rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
-    internal::TransportAdapter feedback_transport_;
+    rtc::scoped_ptr<internal::TransportAdapter> feedback_transport_;
     VideoSendStream* stream_;
     bool bitrate_capped_;
   } test;
@@ -1001,27 +1084,27 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
     int start_bitrate_kbps_ GUARDED_BY(crit_);
   };
 
-  test::NullTransport transport;
-  CreateSenderCall(Call::Config(&transport));
+  CreateSenderCall(Call::Config());
 
-  CreateSendConfig(1);
+  test::NullTransport transport;
+  CreateSendConfig(1, 0, &transport);
 
   Call::Config::BitrateConfig bitrate_config;
   bitrate_config.start_bitrate_bps =
-      2 * encoder_config_.streams[0].max_bitrate_bps;
+      2 * video_encoder_config_.streams[0].max_bitrate_bps;
   sender_call_->SetBitrateConfig(bitrate_config);
 
   StartBitrateObserver encoder;
-  send_config_.encoder_settings.encoder = &encoder;
+  video_send_config_.encoder_settings.encoder = &encoder;
 
-  CreateStreams();
+  CreateVideoStreams();
 
-  EXPECT_EQ(encoder_config_.streams[0].max_bitrate_bps / 1000,
+  EXPECT_EQ(video_encoder_config_.streams[0].max_bitrate_bps / 1000,
             encoder.GetStartBitrateKbps());
 
-  encoder_config_.streams[0].max_bitrate_bps =
+  video_encoder_config_.streams[0].max_bitrate_bps =
       2 * bitrate_config.start_bitrate_bps;
-  send_stream_->ReconfigureVideoEncoder(encoder_config_);
+  video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
 
   // New bitrate should be reconfigured above the previous max. As there's no
   // network connection this shouldn't be flaky, as no bitrate should've been
@@ -1032,71 +1115,71 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
   DestroyStreams();
 }
 
-TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
+TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
   class FrameObserver : public I420FrameCallback {
    public:
-    FrameObserver() : output_frame_event_(EventWrapper::Create()) {}
+    FrameObserver() : output_frame_event_(false, false) {}
 
-    void FrameCallback(I420VideoFrame* video_frame) override {
+    void FrameCallback(VideoFrame* video_frame) override {
       output_frames_.push_back(*video_frame);
-      output_frame_event_->Set();
+      output_frame_event_.Set();
     }
 
     void WaitOutputFrame() {
-      const unsigned long kWaitFrameTimeoutMs = 3000;
-      EXPECT_EQ(kEventSignaled, output_frame_event_->Wait(kWaitFrameTimeoutMs))
+      const int kWaitFrameTimeoutMs = 3000;
+      EXPECT_TRUE(output_frame_event_.Wait(kWaitFrameTimeoutMs))
           << "Timeout while waiting for output frames.";
     }
 
-    const std::vector<I420VideoFrame>& output_frames() const {
+    const std::vector<VideoFrame>& output_frames() const {
       return output_frames_;
     }
 
    private:
     // Delivered output frames.
-    std::vector<I420VideoFrame> output_frames_;
+    std::vector<VideoFrame> output_frames_;
 
     // Indicate an output frame has arrived.
-    rtc::scoped_ptr<EventWrapper> output_frame_event_;
+    rtc::Event output_frame_event_;
   };
 
   // Initialize send stream.
+  CreateSenderCall(Call::Config());
+
   test::NullTransport transport;
-  CreateSenderCall(Call::Config(&transport));
-
-  CreateSendConfig(1);
+  CreateSendConfig(1, 0, &transport);
   FrameObserver observer;
-  send_config_.pre_encode_callback = &observer;
-  CreateStreams();
+  video_send_config_.pre_encode_callback = &observer;
+  CreateVideoStreams();
 
-  // Prepare five input frames. Send ordinary I420VideoFrame and texture frames
+  // Prepare five input frames. Send ordinary VideoFrame and texture frames
   // alternatively.
-  std::vector<I420VideoFrame> input_frames;
-  int width = static_cast<int>(encoder_config_.streams[0].width);
-  int height = static_cast<int>(encoder_config_.streams[0].height);
-  webrtc::RefCountImpl<FakeNativeHandle>* handle1 =
-      new webrtc::RefCountImpl<FakeNativeHandle>();
-  webrtc::RefCountImpl<FakeNativeHandle>* handle2 =
-      new webrtc::RefCountImpl<FakeNativeHandle>();
-  webrtc::RefCountImpl<FakeNativeHandle>* handle3 =
-      new webrtc::RefCountImpl<FakeNativeHandle>();
-  input_frames.push_back(I420VideoFrame(handle1, width, height, 1, 1));
-  input_frames.push_back(I420VideoFrame(handle2, width, height, 2, 2));
-  input_frames.push_back(CreateI420VideoFrame(width, height, 3));
-  input_frames.push_back(CreateI420VideoFrame(width, height, 4));
-  input_frames.push_back(I420VideoFrame(handle3, width, height, 5, 5));
+  std::vector<VideoFrame> input_frames;
+  int width = static_cast<int>(video_encoder_config_.streams[0].width);
+  int height = static_cast<int>(video_encoder_config_.streams[0].height);
+  test::FakeNativeHandle* handle1 = new test::FakeNativeHandle();
+  test::FakeNativeHandle* handle2 = new test::FakeNativeHandle();
+  test::FakeNativeHandle* handle3 = new test::FakeNativeHandle();
+  input_frames.push_back(test::FakeNativeHandle::CreateFrame(
+      handle1, width, height, 1, 1, kVideoRotation_0));
+  input_frames.push_back(test::FakeNativeHandle::CreateFrame(
+      handle2, width, height, 2, 2, kVideoRotation_0));
+  input_frames.push_back(CreateVideoFrame(width, height, 3));
+  input_frames.push_back(CreateVideoFrame(width, height, 4));
+  input_frames.push_back(test::FakeNativeHandle::CreateFrame(
+      handle3, width, height, 5, 5, kVideoRotation_0));
 
-  send_stream_->Start();
+  video_send_stream_->Start();
   for (size_t i = 0; i < input_frames.size(); i++) {
-    send_stream_->Input()->IncomingCapturedFrame(input_frames[i]);
+    video_send_stream_->Input()->IncomingCapturedFrame(input_frames[i]);
     // Do not send the next frame too fast, so the frame dropper won't drop it.
     if (i < input_frames.size() - 1)
-      SleepMs(1000 / encoder_config_.streams[0].max_framerate);
+      SleepMs(1000 / video_encoder_config_.streams[0].max_framerate);
     // Wait until the output frame is received before sending the next input
     // frame. Or the previous input frame may be replaced without delivering.
     observer.WaitOutputFrame();
   }
-  send_stream_->Stop();
+  video_send_stream_->Stop();
 
   // Test if the input and output frames are the same. render_time_ms and
   // timestamp are not compared because capturer sets those values.
@@ -1105,24 +1188,23 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
   DestroyStreams();
 }
 
-void ExpectEqualFrames(const I420VideoFrame& frame1,
-                       const I420VideoFrame& frame2) {
+void ExpectEqualFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
   if (frame1.native_handle() != nullptr || frame2.native_handle() != nullptr)
     ExpectEqualTextureFrames(frame1, frame2);
   else
     ExpectEqualBufferFrames(frame1, frame2);
 }
 
-void ExpectEqualTextureFrames(const I420VideoFrame& frame1,
-                              const I420VideoFrame& frame2) {
+void ExpectEqualTextureFrames(const VideoFrame& frame1,
+                              const VideoFrame& frame2) {
   EXPECT_EQ(frame1.native_handle(), frame2.native_handle());
   EXPECT_EQ(frame1.width(), frame2.width());
   EXPECT_EQ(frame1.height(), frame2.height());
   EXPECT_EQ(frame1.render_time_ms(), frame2.render_time_ms());
 }
 
-void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
-                             const I420VideoFrame& frame2) {
+void ExpectEqualBufferFrames(const VideoFrame& frame1,
+                             const VideoFrame& frame2) {
   EXPECT_EQ(frame1.width(), frame2.width());
   EXPECT_EQ(frame1.height(), frame2.height());
   EXPECT_EQ(frame1.stride(kYPlane), frame2.stride(kYPlane));
@@ -1146,18 +1228,18 @@ void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
                    frame1.allocated_size(kVPlane)));
 }
 
-void ExpectEqualFramesVector(const std::vector<I420VideoFrame>& frames1,
-                             const std::vector<I420VideoFrame>& frames2) {
+void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
+                             const std::vector<VideoFrame>& frames2) {
   EXPECT_EQ(frames1.size(), frames2.size());
   for (size_t i = 0; i < std::min(frames1.size(), frames2.size()); ++i)
     ExpectEqualFrames(frames1[i], frames2[i]);
 }
 
-I420VideoFrame CreateI420VideoFrame(int width, int height, uint8_t data) {
+VideoFrame CreateVideoFrame(int width, int height, uint8_t data) {
   const int kSizeY = width * height * 2;
   rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[kSizeY]);
   memset(buffer.get(), data, kSizeY);
-  I420VideoFrame frame;
+  VideoFrame frame;
   frame.CreateFrame(buffer.get(), buffer.get(), buffer.get(), width, height,
                     width, width / 2, width / 2);
   frame.set_timestamp(data);
@@ -1170,24 +1252,23 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
    public:
     EncoderStateObserver()
         : SendTest(kDefaultTimeoutMs),
-          crit_(CriticalSectionWrapper::CreateCriticalSection()),
           initialized_(false),
           callback_registered_(false),
           num_releases_(0),
           released_(false) {}
 
     bool IsReleased() {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       return released_;
     }
 
     bool IsReadyForEncode() {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       return initialized_ && callback_registered_;
     }
 
     size_t num_releases() {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       return num_releases_;
     }
 
@@ -1195,32 +1276,32 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
     int32_t InitEncode(const VideoCodec* codecSettings,
                        int32_t numberOfCores,
                        size_t maxPayloadSize) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       EXPECT_FALSE(initialized_);
       initialized_ = true;
       released_ = false;
       return 0;
     }
 
-    int32_t Encode(const I420VideoFrame& inputImage,
+    int32_t Encode(const VideoFrame& inputImage,
                    const CodecSpecificInfo* codecSpecificInfo,
-                   const std::vector<VideoFrameType>* frame_types) override {
+                   const std::vector<FrameType>* frame_types) override {
       EXPECT_TRUE(IsReadyForEncode());
 
-      observation_complete_->Set();
+      observation_complete_.Set();
       return 0;
     }
 
     int32_t RegisterEncodeCompleteCallback(
         EncodedImageCallback* callback) override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       EXPECT_TRUE(initialized_);
       callback_registered_ = true;
       return 0;
     }
 
     int32_t Release() override {
-      CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       EXPECT_TRUE(IsReadyForEncode());
       EXPECT_FALSE(released_);
       initialized_ = false;
@@ -1240,7 +1321,7 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
       return 0;
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       // Encoder initialization should be done in stream construction before
@@ -1249,16 +1330,16 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
       stream_ = send_stream;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config_ = *encoder_config;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for Encode.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for Encode.";
       EXPECT_EQ(0u, num_releases());
       stream_->ReconfigureVideoEncoder(encoder_config_);
       EXPECT_EQ(0u, num_releases());
@@ -1268,11 +1349,10 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
       EXPECT_TRUE(IsReadyForEncode());
       stream_->Start();
       // Sanity check, make sure we still encode frames with this encoder.
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for Encode.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for Encode.";
     }
 
-    rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+    rtc::CriticalSection crit_;
     VideoSendStream* stream_;
     bool initialized_ GUARDED_BY(crit_);
     bool callback_registered_ GUARDED_BY(crit_);
@@ -1297,14 +1377,15 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
           num_initializations_(0) {}
 
    private:
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config_ = *encoder_config;
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       stream_ = send_stream;
@@ -1327,7 +1408,7 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
     void PerformTest() override {
       EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
 
-      encoder_config_.content_type = VideoEncoderConfig::kScreenshare;
+      encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
       stream_->ReconfigureVideoEncoder(encoder_config_);
       EXPECT_EQ(2u, num_initializations_)
           << "ReconfigureVideoEncoder did not reinitialize the encoder with "
@@ -1346,7 +1427,6 @@ static const size_t kVideoCodecConfigObserverNumberOfTemporalLayers = 4;
 template <typename T>
 class VideoCodecConfigObserver : public test::SendTest,
                                  public test::FakeEncoder {
-
  public:
   VideoCodecConfigObserver(VideoCodecType video_codec_type,
                            const char* codec_name)
@@ -1359,9 +1439,10 @@ class VideoCodecConfigObserver : public test::SendTest,
   }
 
  private:
-  void ModifyConfigs(VideoSendStream::Config* send_config,
-                     std::vector<VideoReceiveStream::Config>* receive_configs,
-                     VideoEncoderConfig* encoder_config) override {
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override {
     send_config->encoder_settings.encoder = this;
     send_config->encoder_settings.payload_name = codec_name_;
 
@@ -1374,7 +1455,7 @@ class VideoCodecConfigObserver : public test::SendTest,
     encoder_config_ = *encoder_config;
   }
 
-  void OnStreamsCreated(
+  void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams) override {
     stream_ = send_stream;
@@ -1401,9 +1482,9 @@ class VideoCodecConfigObserver : public test::SendTest,
            "new encoder settings.";
   }
 
-  int32_t Encode(const I420VideoFrame& input_image,
+  int32_t Encode(const VideoFrame& input_image,
                  const CodecSpecificInfo* codec_specific_info,
-                 const std::vector<VideoFrameType>* frame_types) override {
+                 const std::vector<FrameType>* frame_types) override {
     // Silently skip the encode, FakeEncoder::Encode doesn't produce VP8.
     return 0;
   }
@@ -1489,6 +1570,7 @@ TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       ++rtp_packets_sent_;
@@ -1497,19 +1579,20 @@ TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
-        if (packet_type == RTCPUtility::kRtcpSrCode) {
+      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+        if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
           // Only compare sent media bytes if SenderPacketCount matches the
           // number of sent rtp packets (a new rtp packet could be sent before
           // the rtcp packet).
           if (parser.Packet().SR.SenderOctetCount > 0 &&
               parser.Packet().SR.SenderPacketCount == rtp_packets_sent_) {
             EXPECT_EQ(media_bytes_sent_, parser.Packet().SR.SenderOctetCount);
-            observation_complete_->Set();
+            observation_complete_.Set();
           }
         }
         packet_type = parser.Iterate();
@@ -1519,12 +1602,12 @@ TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for RTCP sender report.";
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP sender report.";
     }
 
-    size_t rtp_packets_sent_;
-    size_t media_bytes_sent_;
+    rtc::CriticalSection crit_;
+    size_t rtp_packets_sent_ GUARDED_BY(&crit_);
+    size_t media_bytes_sent_ GUARDED_BY(&crit_);
   } test;
 
   RunBaseTest(&test);
@@ -1545,24 +1628,25 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
                        size_t max_payload_size) override {
       EXPECT_EQ(static_cast<unsigned int>(kScreencastTargetBitrateKbps),
                 config->targetBitrate);
-      observation_complete_->Set();
+      observation_complete_.Set();
       return test::FakeEncoder::InitEncode(
           config, number_of_cores, max_payload_size);
     }
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       EXPECT_EQ(1u, encoder_config->streams.size());
       EXPECT_TRUE(
           encoder_config->streams[0].temporal_layer_thresholds_bps.empty());
       encoder_config->streams[0].temporal_layer_thresholds_bps.push_back(
           kScreencastTargetBitrateKbps * 1000);
-      encoder_config->content_type = VideoEncoderConfig::kScreenshare;
+      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
     }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for the encoder to be initialized.";
     }
   } test;
@@ -1570,7 +1654,18 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
   RunBaseTest(&test);
 }
 
-TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
+// Disabled on LinuxAsan:
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5382
+#if defined(ADDRESS_SANITIZER) && defined(WEBRTC_LINUX)
+#define MAYBE_ReconfigureBitratesSetsEncoderBitratesCorrectly \
+  DISABLED_ReconfigureBitratesSetsEncoderBitratesCorrectly
+#else
+#define MAYBE_ReconfigureBitratesSetsEncoderBitratesCorrectly \
+  ReconfigureBitratesSetsEncoderBitratesCorrectly
+#endif
+
+TEST_F(VideoSendStreamTest,
+       MAYBE_ReconfigureBitratesSetsEncoderBitratesCorrectly) {
   // These are chosen to be "kind of odd" to not be accidentally checked against
   // default values.
   static const int kMinBitrateKbps = 137;
@@ -1598,7 +1693,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
                   codecSettings->startBitrate);
         EXPECT_EQ(static_cast<unsigned int>(kMaxBitrateKbps),
                   codecSettings->maxBitrate);
-        observation_complete_->Set();
+        observation_complete_.Set();
       } else if (num_initializations_ == 1) {
         EXPECT_EQ(static_cast<unsigned int>(kLowerMaxBitrateKbps),
                   codecSettings->maxBitrate);
@@ -1618,16 +1713,17 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
     }
 
     Call::Config GetSenderCallConfig() override {
-      Call::Config config(SendTransport());
+      Call::Config config;
       config.bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
       config.bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
       config.bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;
       return config;
     }
 
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       // Set bitrates lower/higher than min/max to make sure they are properly
       // capped.
@@ -1640,7 +1736,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
       call_ = sender_call;
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
@@ -1651,7 +1747,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
       bitrate_config.start_bitrate_bps = kIncreasedStartBitrateKbps * 1000;
       bitrate_config.max_bitrate_bps = kIncreasedMaxBitrateKbps * 1000;
       call_->SetBitrateConfig(bitrate_config);
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting encoder to be configured.";
       encoder_config_.streams[0].min_bitrate_bps = 0;
       encoder_config_.streams[0].max_bitrate_bps = kLowerMaxBitrateKbps * 1000;
@@ -1692,9 +1788,9 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
           test::FakeEncoder(Clock::GetRealTimeClock()) {}
 
    private:
-    int32_t Encode(const I420VideoFrame& input_image,
+    int32_t Encode(const VideoFrame& input_image,
                    const CodecSpecificInfo* codecSpecificInfo,
-                   const std::vector<VideoFrameType>* frame_types) override {
+                   const std::vector<FrameType>* frame_types) override {
       CodecSpecificInfo specifics;
       memset(&specifics, 0, sizeof(specifics));
       specifics.codecType = kVideoCodecGeneric;
@@ -1709,41 +1805,42 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
         encoded._frameType = (*frame_types)[i];
         encoded._encodedWidth = kEncodedResolution[i].width;
         encoded._encodedHeight = kEncodedResolution[i].height;
-        DCHECK(callback_ != nullptr);
+        RTC_DCHECK(callback_ != nullptr);
         if (callback_->Encoded(encoded, &specifics, nullptr) != 0)
           return -1;
       }
 
-      observation_complete_->Set();
+      observation_complete_.Set();
       return 0;
     }
-    void ModifyConfigs(VideoSendStream::Config* send_config,
-                       std::vector<VideoReceiveStream::Config>* receive_configs,
-                       VideoEncoderConfig* encoder_config) override {
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       EXPECT_EQ(kNumStreams, encoder_config->streams.size());
     }
 
-    size_t GetNumStreams() const override { return kNumStreams; }
+    size_t GetNumVideoStreams() const override { return kNumStreams; }
 
     void PerformTest() override {
-      EXPECT_EQ(kEventSignaled, Wait())
+      EXPECT_TRUE(Wait())
           << "Timed out while waiting for the encoder to send one frame.";
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       for (size_t i = 0; i < kNumStreams; ++i) {
-        ASSERT_TRUE(stats.substreams.find(kSendSsrcs[i]) !=
+        ASSERT_TRUE(stats.substreams.find(kVideoSendSsrcs[i]) !=
                     stats.substreams.end())
-            << "No stats for SSRC: " << kSendSsrcs[i]
+            << "No stats for SSRC: " << kVideoSendSsrcs[i]
             << ", stats should exist as soon as frames have been encoded.";
         VideoSendStream::StreamStats ssrc_stats =
-            stats.substreams[kSendSsrcs[i]];
+            stats.substreams[kVideoSendSsrcs[i]];
         EXPECT_EQ(kEncodedResolution[i].width, ssrc_stats.width);
         EXPECT_EQ(kEncodedResolution[i].height, ssrc_stats.height);
       }
     }
 
-    void OnStreamsCreated(
+    void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
@@ -1754,4 +1851,417 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
 
   RunBaseTest(&test);
 }
+
+class Vp9HeaderObserver : public test::SendTest {
+ public:
+  Vp9HeaderObserver()
+      : SendTest(VideoSendStreamTest::kLongTimeoutMs),
+        vp9_encoder_(VP9Encoder::Create()),
+        vp9_settings_(VideoEncoder::GetDefaultVp9Settings()),
+        packets_sent_(0),
+        frames_sent_(0) {}
+
+  virtual void ModifyVideoConfigsHook(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) {}
+
+  virtual void InspectHeader(const RTPVideoHeaderVP9& vp9) = 0;
+
+ private:
+  const int kVp9PayloadType = 105;
+
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override {
+    encoder_config->encoder_specific_settings = &vp9_settings_;
+    send_config->encoder_settings.encoder = vp9_encoder_.get();
+    send_config->encoder_settings.payload_name = "VP9";
+    send_config->encoder_settings.payload_type = kVp9PayloadType;
+    ModifyVideoConfigsHook(send_config, receive_configs, encoder_config);
+    EXPECT_EQ(1u, encoder_config->streams.size());
+    encoder_config->streams[0].temporal_layer_thresholds_bps.resize(
+        vp9_settings_.numberOfTemporalLayers - 1);
+    encoder_config_ = *encoder_config;
+  }
+
+  void PerformTest() override {
+    EXPECT_TRUE(Wait()) << "Test timed out waiting for VP9 packet, num frames "
+                        << frames_sent_;
+  }
+
+  Action OnSendRtp(const uint8_t* packet, size_t length) override {
+    RTPHeader header;
+    EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+    EXPECT_EQ(kVp9PayloadType, header.payloadType);
+    const uint8_t* payload = packet + header.headerLength;
+    size_t payload_length = length - header.headerLength - header.paddingLength;
+
+    bool new_packet = packets_sent_ == 0 ||
+                      IsNewerSequenceNumber(header.sequenceNumber,
+                                            last_header_.sequenceNumber);
+    if (payload_length > 0 && new_packet) {
+      RtpDepacketizer::ParsedPayload parsed;
+      RtpDepacketizerVp9 depacketizer;
+      EXPECT_TRUE(depacketizer.Parse(&parsed, payload, payload_length));
+      EXPECT_EQ(RtpVideoCodecTypes::kRtpVideoVp9, parsed.type.Video.codec);
+      // Verify common fields for all configurations.
+      VerifyCommonHeader(parsed.type.Video.codecHeader.VP9);
+      CompareConsecutiveFrames(header, parsed.type.Video);
+      // Verify configuration specific settings.
+      InspectHeader(parsed.type.Video.codecHeader.VP9);
+
+      ++packets_sent_;
+      if (header.markerBit) {
+        ++frames_sent_;
+      }
+      last_header_ = header;
+      last_vp9_ = parsed.type.Video.codecHeader.VP9;
+    }
+    return SEND_PACKET;
+  }
+
+ protected:
+  bool ContinuousPictureId(const RTPVideoHeaderVP9& vp9) const {
+    if (last_vp9_.picture_id > vp9.picture_id) {
+      return vp9.picture_id == 0;  // Wrap.
+    } else {
+      return vp9.picture_id == last_vp9_.picture_id + 1;
+    }
+  }
+
+  void VerifySpatialIdxWithinFrame(const RTPVideoHeaderVP9& vp9) const {
+    bool new_layer = vp9.spatial_idx != last_vp9_.spatial_idx;
+    EXPECT_EQ(new_layer, vp9.beginning_of_frame);
+    EXPECT_EQ(new_layer, last_vp9_.end_of_frame);
+    EXPECT_EQ(new_layer ? last_vp9_.spatial_idx + 1 : last_vp9_.spatial_idx,
+              vp9.spatial_idx);
+  }
+
+  void VerifyFixedTemporalLayerStructure(const RTPVideoHeaderVP9& vp9,
+                                         uint8_t num_layers) const {
+    switch (num_layers) {
+      case 0:
+        VerifyTemporalLayerStructure0(vp9);
+        break;
+      case 1:
+        VerifyTemporalLayerStructure1(vp9);
+        break;
+      case 2:
+        VerifyTemporalLayerStructure2(vp9);
+        break;
+      case 3:
+        VerifyTemporalLayerStructure3(vp9);
+        break;
+      default:
+        RTC_NOTREACHED();
+    }
+  }
+
+  void VerifyTemporalLayerStructure0(const RTPVideoHeaderVP9& vp9) const {
+    EXPECT_EQ(kNoTl0PicIdx, vp9.tl0_pic_idx);
+    EXPECT_EQ(kNoTemporalIdx, vp9.temporal_idx);  // no tid
+    EXPECT_FALSE(vp9.temporal_up_switch);
+  }
+
+  void VerifyTemporalLayerStructure1(const RTPVideoHeaderVP9& vp9) const {
+    EXPECT_NE(kNoTl0PicIdx, vp9.tl0_pic_idx);
+    EXPECT_EQ(0, vp9.temporal_idx);  // 0,0,0,...
+    EXPECT_FALSE(vp9.temporal_up_switch);
+  }
+
+  void VerifyTemporalLayerStructure2(const RTPVideoHeaderVP9& vp9) const {
+    EXPECT_NE(kNoTl0PicIdx, vp9.tl0_pic_idx);
+    EXPECT_GE(vp9.temporal_idx, 0);  // 0,1,0,1,... (tid reset on I-frames).
+    EXPECT_LE(vp9.temporal_idx, 1);
+    EXPECT_EQ(vp9.temporal_idx > 0, vp9.temporal_up_switch);
+    if (IsNewPictureId(vp9)) {
+      uint8_t expected_tid =
+          (!vp9.inter_pic_predicted || last_vp9_.temporal_idx == 1) ? 0 : 1;
+      EXPECT_EQ(expected_tid, vp9.temporal_idx);
+    }
+  }
+
+  void VerifyTemporalLayerStructure3(const RTPVideoHeaderVP9& vp9) const {
+    EXPECT_NE(kNoTl0PicIdx, vp9.tl0_pic_idx);
+    EXPECT_GE(vp9.temporal_idx, 0);  // 0,2,1,2,... (tid reset on I-frames).
+    EXPECT_LE(vp9.temporal_idx, 2);
+    if (IsNewPictureId(vp9) && vp9.inter_pic_predicted) {
+      EXPECT_NE(vp9.temporal_idx, last_vp9_.temporal_idx);
+      switch (vp9.temporal_idx) {
+        case 0:
+          EXPECT_EQ(2, last_vp9_.temporal_idx);
+          EXPECT_FALSE(vp9.temporal_up_switch);
+          break;
+        case 1:
+          EXPECT_EQ(2, last_vp9_.temporal_idx);
+          EXPECT_TRUE(vp9.temporal_up_switch);
+          break;
+        case 2:
+          EXPECT_EQ(last_vp9_.temporal_idx == 0, vp9.temporal_up_switch);
+          break;
+      }
+    }
+  }
+
+  void VerifyTl0Idx(const RTPVideoHeaderVP9& vp9) const {
+    if (vp9.tl0_pic_idx == kNoTl0PicIdx)
+      return;
+
+    uint8_t expected_tl0_idx = last_vp9_.tl0_pic_idx;
+    if (vp9.temporal_idx == 0)
+      ++expected_tl0_idx;
+    EXPECT_EQ(expected_tl0_idx, vp9.tl0_pic_idx);
+  }
+
+  bool IsNewPictureId(const RTPVideoHeaderVP9& vp9) const {
+    return frames_sent_ > 0 && (vp9.picture_id != last_vp9_.picture_id);
+  }
+
+  // Flexible mode (F=1):    Non-flexible mode (F=0):
+  //
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  //      |I|P|L|F|B|E|V|-|     |I|P|L|F|B|E|V|-|
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  // I:   |M| PICTURE ID  |  I: |M| PICTURE ID  |
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  // M:   | EXTENDED PID  |  M: | EXTENDED PID  |
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  // L:   |  T  |U|  S  |D|  L: |  T  |U|  S  |D|
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  // P,F: | P_DIFF    |X|N|     |   TL0PICIDX   |
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  // X:   |EXTENDED P_DIFF|  V: | SS  ..        |
+  //      +-+-+-+-+-+-+-+-+     +-+-+-+-+-+-+-+-+
+  // V:   | SS  ..        |
+  //      +-+-+-+-+-+-+-+-+
+  void VerifyCommonHeader(const RTPVideoHeaderVP9& vp9) const {
+    EXPECT_EQ(kMaxTwoBytePictureId, vp9.max_picture_id);       // M:1
+    EXPECT_NE(kNoPictureId, vp9.picture_id);                   // I:1
+    EXPECT_EQ(vp9_settings_.flexibleMode, vp9.flexible_mode);  // F
+    EXPECT_GE(vp9.spatial_idx, 0);                             // S
+    EXPECT_LT(vp9.spatial_idx, vp9_settings_.numberOfSpatialLayers);
+    if (vp9.ss_data_available)  // V
+      VerifySsData(vp9);
+
+    if (frames_sent_ == 0)
+      EXPECT_FALSE(vp9.inter_pic_predicted);  // P
+
+    if (!vp9.inter_pic_predicted) {
+      EXPECT_TRUE(vp9.temporal_idx == 0 || vp9.temporal_idx == kNoTemporalIdx);
+      EXPECT_FALSE(vp9.temporal_up_switch);
+    }
+  }
+
+  // Scalability structure (SS).
+  //
+  //      +-+-+-+-+-+-+-+-+
+  // V:   | N_S |Y|G|-|-|-|
+  //      +-+-+-+-+-+-+-+-+
+  // Y:   |    WIDTH      |  N_S + 1 times
+  //      +-+-+-+-+-+-+-+-+
+  //      |    HEIGHT     |
+  //      +-+-+-+-+-+-+-+-+
+  // G:   |      N_G      |
+  //      +-+-+-+-+-+-+-+-+
+  // N_G: |  T  |U| R |-|-|  N_G times
+  //      +-+-+-+-+-+-+-+-+
+  //      |    P_DIFF     |  R times
+  //      +-+-+-+-+-+-+-+-+
+  void VerifySsData(const RTPVideoHeaderVP9& vp9) const {
+    EXPECT_TRUE(vp9.ss_data_available);             // V
+    EXPECT_EQ(vp9_settings_.numberOfSpatialLayers,  // N_S + 1
+              vp9.num_spatial_layers);
+    EXPECT_TRUE(vp9.spatial_layer_resolution_present);  // Y:1
+    size_t expected_width = encoder_config_.streams[0].width;
+    size_t expected_height = encoder_config_.streams[0].height;
+    for (int i = vp9.num_spatial_layers - 1; i >= 0; --i) {
+      EXPECT_EQ(expected_width, vp9.width[i]);    // WIDTH
+      EXPECT_EQ(expected_height, vp9.height[i]);  // HEIGHT
+      expected_width /= 2;
+      expected_height /= 2;
+    }
+  }
+
+  void CompareConsecutiveFrames(const RTPHeader& header,
+                                const RTPVideoHeader& video) const {
+    const RTPVideoHeaderVP9& vp9 = video.codecHeader.VP9;
+
+    bool new_frame = packets_sent_ == 0 ||
+                     IsNewerTimestamp(header.timestamp, last_header_.timestamp);
+    EXPECT_EQ(new_frame, video.isFirstPacket);
+    if (!new_frame) {
+      EXPECT_FALSE(last_header_.markerBit);
+      EXPECT_EQ(last_header_.timestamp, header.timestamp);
+      EXPECT_EQ(last_vp9_.picture_id, vp9.picture_id);
+      EXPECT_EQ(last_vp9_.temporal_idx, vp9.temporal_idx);
+      EXPECT_EQ(last_vp9_.tl0_pic_idx, vp9.tl0_pic_idx);
+      VerifySpatialIdxWithinFrame(vp9);
+      return;
+    }
+    // New frame.
+    EXPECT_TRUE(vp9.beginning_of_frame);
+
+    // Compare with last packet in previous frame.
+    if (frames_sent_ == 0)
+      return;
+    EXPECT_TRUE(last_vp9_.end_of_frame);
+    EXPECT_TRUE(last_header_.markerBit);
+    EXPECT_TRUE(ContinuousPictureId(vp9));
+    VerifyTl0Idx(vp9);
+  }
+
+  rtc::scoped_ptr<VP9Encoder> vp9_encoder_;
+  VideoCodecVP9 vp9_settings_;
+  webrtc::VideoEncoderConfig encoder_config_;
+  RTPHeader last_header_;
+  RTPVideoHeaderVP9 last_vp9_;
+  size_t packets_sent_;
+  size_t frames_sent_;
+};
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexMode_1Tl1SLayers) {
+  const uint8_t kNumTemporalLayers = 1;
+  const uint8_t kNumSpatialLayers = 1;
+  TestVp9NonFlexMode(kNumTemporalLayers, kNumSpatialLayers);
+}
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexMode_2Tl1SLayers) {
+  const uint8_t kNumTemporalLayers = 2;
+  const uint8_t kNumSpatialLayers = 1;
+  TestVp9NonFlexMode(kNumTemporalLayers, kNumSpatialLayers);
+}
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexMode_3Tl1SLayers) {
+  const uint8_t kNumTemporalLayers = 3;
+  const uint8_t kNumSpatialLayers = 1;
+  TestVp9NonFlexMode(kNumTemporalLayers, kNumSpatialLayers);
+}
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexMode_1Tl2SLayers) {
+  const uint8_t kNumTemporalLayers = 1;
+  const uint8_t kNumSpatialLayers = 2;
+  TestVp9NonFlexMode(kNumTemporalLayers, kNumSpatialLayers);
+}
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexMode_2Tl2SLayers) {
+  const uint8_t kNumTemporalLayers = 2;
+  const uint8_t kNumSpatialLayers = 2;
+  TestVp9NonFlexMode(kNumTemporalLayers, kNumSpatialLayers);
+}
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexMode_3Tl2SLayers) {
+  const uint8_t kNumTemporalLayers = 3;
+  const uint8_t kNumSpatialLayers = 2;
+  TestVp9NonFlexMode(kNumTemporalLayers, kNumSpatialLayers);
+}
+
+void VideoSendStreamTest::TestVp9NonFlexMode(uint8_t num_temporal_layers,
+                                             uint8_t num_spatial_layers) {
+  static const size_t kNumFramesToSend = 100;
+  // Set to < kNumFramesToSend and coprime to length of temporal layer
+  // structures to verify temporal id reset on key frame.
+  static const int kKeyFrameInterval = 31;
+  class NonFlexibleMode : public Vp9HeaderObserver {
+   public:
+    NonFlexibleMode(uint8_t num_temporal_layers, uint8_t num_spatial_layers)
+        : num_temporal_layers_(num_temporal_layers),
+          num_spatial_layers_(num_spatial_layers),
+          l_field_(num_temporal_layers > 1 || num_spatial_layers > 1) {}
+    void ModifyVideoConfigsHook(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      vp9_settings_.flexibleMode = false;
+      vp9_settings_.frameDroppingOn = false;
+      vp9_settings_.keyFrameInterval = kKeyFrameInterval;
+      vp9_settings_.numberOfTemporalLayers = num_temporal_layers_;
+      vp9_settings_.numberOfSpatialLayers = num_spatial_layers_;
+    }
+
+    void InspectHeader(const RTPVideoHeaderVP9& vp9) override {
+      bool ss_data_expected = !vp9.inter_pic_predicted &&
+                              vp9.beginning_of_frame && vp9.spatial_idx == 0;
+      EXPECT_EQ(ss_data_expected, vp9.ss_data_available);
+      EXPECT_EQ(vp9.spatial_idx > 0, vp9.inter_layer_predicted);  // D
+      EXPECT_EQ(!vp9.inter_pic_predicted,
+                frames_sent_ % kKeyFrameInterval == 0);
+
+      if (IsNewPictureId(vp9)) {
+        EXPECT_EQ(0, vp9.spatial_idx);
+        EXPECT_EQ(num_spatial_layers_ - 1, last_vp9_.spatial_idx);
+      }
+
+      VerifyFixedTemporalLayerStructure(vp9,
+                                        l_field_ ? num_temporal_layers_ : 0);
+
+      if (frames_sent_ > kNumFramesToSend)
+        observation_complete_.Set();
+    }
+    const uint8_t num_temporal_layers_;
+    const uint8_t num_spatial_layers_;
+    const bool l_field_;
+  } test(num_temporal_layers, num_spatial_layers);
+
+  RunBaseTest(&test);
+}
+
+TEST_F(VideoSendStreamTest, Vp9NonFlexModeSmallResolution) {
+  static const size_t kNumFramesToSend = 50;
+  static const int kWidth = 4;
+  static const int kHeight = 4;
+  class NonFlexibleModeResolution : public Vp9HeaderObserver {
+    void ModifyVideoConfigsHook(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      vp9_settings_.flexibleMode = false;
+      vp9_settings_.numberOfTemporalLayers = 1;
+      vp9_settings_.numberOfSpatialLayers = 1;
+
+      EXPECT_EQ(1u, encoder_config->streams.size());
+      encoder_config->streams[0].width = kWidth;
+      encoder_config->streams[0].height = kHeight;
+    }
+
+    void InspectHeader(const RTPVideoHeaderVP9& vp9_header) override {
+      if (frames_sent_ > kNumFramesToSend)
+        observation_complete_.Set();
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+
+#if !defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5402.
+TEST_F(VideoSendStreamTest, Vp9FlexModeRefCount) {
+  class FlexibleMode : public Vp9HeaderObserver {
+    void ModifyVideoConfigsHook(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
+      vp9_settings_.flexibleMode = true;
+      vp9_settings_.numberOfTemporalLayers = 1;
+      vp9_settings_.numberOfSpatialLayers = 2;
+    }
+
+    void InspectHeader(const RTPVideoHeaderVP9& vp9_header) override {
+      EXPECT_TRUE(vp9_header.flexible_mode);
+      EXPECT_EQ(kNoTl0PicIdx, vp9_header.tl0_pic_idx);
+      if (vp9_header.inter_pic_predicted) {
+        EXPECT_GT(vp9_header.num_ref_pics, 0u);
+        observation_complete_.Set();
+      }
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+#endif
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/vie_channel.cc b/media/webrtc/trunk/webrtc/video/vie_channel.cc
new file mode 100644
index 0000000000..ce5d534368
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/vie_channel.cc
@@ -0,0 +1,1341 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/vie_channel.h"
+
+#include <algorithm>
+#include <map>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/common.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/frame_callback.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/modules/video_render/video_render_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/receive_statistics_proxy.h"
+#include "webrtc/video/report_block_stats.h"
+
+namespace webrtc {
+
+const int kMaxDecodeWaitTimeMs = 50;
+static const int kMaxTargetDelayMs = 10000;
+const int kMinSendSidePacketHistorySize = 600;
+const int kMaxPacketAgeToNack = 450;
+const int kMaxNackListSize = 250;
+
+const int kInvalidRtpExtensionId = 0; //MOZ addition for RtpSenderId (RID)
+
+// Helper class receiving statistics callbacks.
+class ChannelStatsObserver : public CallStatsObserver {
+ public:
+  explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
+  virtual ~ChannelStatsObserver() {}
+
+  // Implements StatsObserver.
+  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
+    owner_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+  }
+
+ private:
+  ViEChannel* const owner_;
+};
+
+class ViEChannelProtectionCallback : public VCMProtectionCallback {
+ public:
+  explicit ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {}
+  ~ViEChannelProtectionCallback() {}
+
+
+  int ProtectionRequest(
+      const FecProtectionParams* delta_fec_params,
+      const FecProtectionParams* key_fec_params,
+      uint32_t* sent_video_rate_bps,
+      uint32_t* sent_nack_rate_bps,
+      uint32_t* sent_fec_rate_bps) override {
+    return owner_->ProtectionRequest(delta_fec_params, key_fec_params,
+                                     sent_video_rate_bps, sent_nack_rate_bps,
+                                     sent_fec_rate_bps);
+  }
+ private:
+  ViEChannel* owner_;
+};
+
+ViEChannel::ViEChannel(uint32_t number_of_cores,
+                       Transport* transport,
+                       ProcessThread* module_process_thread,
+                       RtcpIntraFrameObserver* intra_frame_observer,
+                       RtcpBandwidthObserver* bandwidth_observer,
+                       TransportFeedbackObserver* transport_feedback_observer,
+                       RemoteBitrateEstimator* remote_bitrate_estimator,
+                       RtcpRttStats* rtt_stats,
+                       PacedSender* paced_sender,
+                       PacketRouter* packet_router,
+                       size_t max_rtp_streams,
+                       bool sender)
+    : number_of_cores_(number_of_cores),
+      sender_(sender),
+      module_process_thread_(module_process_thread),
+      crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      send_payload_router_(new PayloadRouter()),
+      vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
+      vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(),
+                                     nullptr,
+                                     nullptr)),
+      vie_receiver_(vcm_, remote_bitrate_estimator, this),
+      vie_sync_(vcm_),
+      stats_observer_(new ChannelStatsObserver(this)),
+      receive_stats_callback_(nullptr),
+      incoming_video_stream_(nullptr),
+      intra_frame_observer_(intra_frame_observer),
+      rtt_stats_(rtt_stats),
+      paced_sender_(paced_sender),
+      packet_router_(packet_router),
+      bandwidth_observer_(bandwidth_observer),
+      transport_feedback_observer_(transport_feedback_observer),
+      decode_thread_(ChannelDecodeThreadFunction, this, "DecodingThread"),
+      nack_history_size_sender_(kMinSendSidePacketHistorySize),
+      max_nack_reordering_threshold_(kMaxPacketAgeToNack),
+      pre_render_callback_(NULL),
+      report_block_stats_sender_(new ReportBlockStats()),
+      time_of_first_rtt_ms_(-1),
+      rtt_sum_ms_(0),
+      last_rtt_ms_(0),
+      num_rtts_(0),
+			rid_extension_id_(kInvalidRtpExtensionId),
+      rtp_rtcp_modules_(
+          CreateRtpRtcpModules(!sender,
+                               vie_receiver_.GetReceiveStatistics(),
+                               transport,
+                               intra_frame_observer_,
+                               bandwidth_observer_.get(),
+                               transport_feedback_observer_,
+                               rtt_stats_,
+                               &rtcp_packet_type_counter_observer_,
+                               remote_bitrate_estimator,
+                               paced_sender_,
+                               packet_router_,
+                               &send_bitrate_observer_,
+                               &send_frame_count_observer_,
+                               &send_side_delay_observer_,
+                               max_rtp_streams)),
+      num_active_rtp_rtcp_modules_(1) {
+  vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]);
+  vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
+}
+
+int32_t ViEChannel::Init() {
+  static const int kDefaultRenderDelayMs = 10;
+  module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics());
+
+  // RTP/RTCP initialization.
+  module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]);
+
+  rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
+  if (paced_sender_) {
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+      rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
+  }
+  packet_router_->AddRtpModule(rtp_rtcp_modules_[0]);
+  if (sender_) {
+    std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]);
+    send_payload_router_->SetSendingRtpModules(send_rtp_modules);
+    RTC_DCHECK(!send_payload_router_->active());
+  }
+  if (vcm_->RegisterReceiveCallback(this) != 0) {
+    return -1;
+  }
+  vcm_->RegisterFrameTypeCallback(this);
+	vcm_->RegisterReceiveStateCallback(this); //MOZ addition for RID
+  vcm_->RegisterReceiveStatisticsCallback(this);
+  vcm_->RegisterDecoderTimingCallback(this);
+  vcm_->SetRenderDelay(kDefaultRenderDelayMs);
+
+  module_process_thread_->RegisterModule(vcm_);
+  module_process_thread_->RegisterModule(&vie_sync_);
+
+  return 0;
+}
+
+ViEChannel::~ViEChannel() {
+  UpdateHistograms();
+  // Make sure we don't get more callbacks from the RTP module.
+  module_process_thread_->DeRegisterModule(
+      vie_receiver_.GetReceiveStatistics());
+  module_process_thread_->DeRegisterModule(vcm_);
+  module_process_thread_->DeRegisterModule(&vie_sync_);
+  send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
+  for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i)
+    packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    module_process_thread_->DeRegisterModule(rtp_rtcp);
+    delete rtp_rtcp;
+  }
+  if (!sender_)
+    StopDecodeThread();
+  // Release modules.
+  VideoCodingModule::Destroy(vcm_);
+}
+
+void ViEChannel::UpdateHistograms() {
+  int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
+
+  {
+    CriticalSectionScoped cs(crit_.get());
+    int64_t elapsed_sec = (now - time_of_first_rtt_ms_) / 1000;
+    if (time_of_first_rtt_ms_ != -1 && num_rtts_ > 0 &&
+        elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      int64_t avg_rtt_ms = (rtt_sum_ms_ + num_rtts_ / 2) / num_rtts_;
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
+    }
+  }
+
+  if (sender_) {
+    RtcpPacketTypeCounter rtcp_counter;
+    GetSendRtcpPacketTypeCounter(&rtcp_counter);
+    int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.NackPacketsReceivedPerMinute",
+          rtcp_counter.nack_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.FirPacketsReceivedPerMinute",
+          rtcp_counter.fir_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.PliPacketsReceivedPerMinute",
+          rtcp_counter.pli_packets * 60 / elapsed_sec);
+      if (rtcp_counter.nack_requests > 0) {
+        RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+            "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
+            rtcp_counter.UniqueNackRequestsInPercent());
+      }
+      int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
+      if (fraction_lost != -1) {
+        RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.SentPacketsLostInPercent",
+                                        fraction_lost);
+      }
+    }
+
+    StreamDataCounters rtp;
+    StreamDataCounters rtx;
+    GetSendStreamDataCounters(&rtp, &rtx);
+    StreamDataCounters rtp_rtx = rtp;
+    rtp_rtx.Add(rtx);
+    elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
+                      Clock::GetRealTimeClock()->TimeInMilliseconds()) /
+                  1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+          "WebRTC.Video.BitrateSentInKbps",
+          static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.MediaBitrateSentInKbps",
+          static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.PaddingBitrateSentInKbps",
+          static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.RetransmittedBitrateSentInKbps",
+          static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
+                           elapsed_sec / 1000));
+      if (rtp_rtcp_modules_[0]->RtxSendStatus() != kRtxOff) {
+        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+            "WebRTC.Video.RtxBitrateSentInKbps",
+            static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                             1000));
+      }
+      bool fec_enabled = false;
+      uint8_t pltype_red;
+      uint8_t pltype_fec;
+      rtp_rtcp_modules_[0]->GenericFECStatus(&fec_enabled, &pltype_red,
+                                             &pltype_fec);
+      if (fec_enabled) {
+        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+            "WebRTC.Video.FecBitrateSentInKbps",
+            static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec /
+                             1000));
+      }
+    }
+  } else if (vie_receiver_.GetRemoteSsrc() > 0) {
+    // Get receive stats if we are receiving packets, i.e. there is a remote
+    // ssrc.
+    RtcpPacketTypeCounter rtcp_counter;
+    GetReceiveRtcpPacketTypeCounter(&rtcp_counter);
+    int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.NackPacketsSentPerMinute",
+          rtcp_counter.nack_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.FirPacketsSentPerMinute",
+          rtcp_counter.fir_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.PliPacketsSentPerMinute",
+          rtcp_counter.pli_packets * 60 / elapsed_sec);
+      if (rtcp_counter.nack_requests > 0) {
+        RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+            "WebRTC.Video.UniqueNackRequestsSentInPercent",
+            rtcp_counter.UniqueNackRequestsInPercent());
+      }
+    }
+
+    StreamDataCounters rtp;
+    StreamDataCounters rtx;
+    GetReceiveStreamDataCounters(&rtp, &rtx);
+    StreamDataCounters rtp_rtx = rtp;
+    rtp_rtx.Add(rtx);
+    elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.BitrateReceivedInKbps",
+          static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.MediaBitrateReceivedInKbps",
+          static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.PaddingBitrateReceivedInKbps",
+          static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
+          static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
+                           elapsed_sec / 1000));
+      uint32_t ssrc = 0;
+      if (vie_receiver_.GetRtxSsrc(&ssrc)) {
+        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+            "WebRTC.Video.RtxBitrateReceivedInKbps",
+            static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                             1000));
+      }
+      if (vie_receiver_.IsFecEnabled()) {
+        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+            "WebRTC.Video.FecBitrateReceivedInKbps",
+            static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec /
+                             1000));
+      }
+    }
+  }
+}
+
+int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
+                                 bool new_stream) {
+  RTC_DCHECK(sender_);
+  if (video_codec.codecType == kVideoCodecRED ||
+      video_codec.codecType == kVideoCodecULPFEC) {
+    LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType;
+    return -1;
+  }
+  if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) {
+    LOG_F(LS_ERROR) << "Incorrect config "
+                    << video_codec.numberOfSimulcastStreams;
+    return -1;
+  }
+  // Update the RTP module with the settings.
+  // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
+  // set explicitly.
+  // The first layer is always active, so the first module can be checked for
+  // sending status.
+  bool is_sending = rtp_rtcp_modules_[0]->Sending();
+  bool router_was_active = send_payload_router_->active();
+  send_payload_router_->set_active(false);
+  send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
+
+  std::vector<RtpRtcp*> registered_modules;
+  std::vector<RtpRtcp*> deregistered_modules;
+  size_t num_active_modules = video_codec.numberOfSimulcastStreams > 0
+                                   ? video_codec.numberOfSimulcastStreams
+                                   : 1;
+  size_t num_prev_active_modules;
+  {
+    // Cache which modules are active so StartSend can know which ones to start.
+    CriticalSectionScoped cs(crit_.get());
+    num_prev_active_modules = num_active_rtp_rtcp_modules_;
+    num_active_rtp_rtcp_modules_ = num_active_modules;
+  }
+  for (size_t i = 0; i < num_active_modules; ++i)
+    registered_modules.push_back(rtp_rtcp_modules_[i]);
+
+  for (size_t i = num_active_modules; i < rtp_rtcp_modules_.size(); ++i)
+    deregistered_modules.push_back(rtp_rtcp_modules_[i]);
+
+  // Disable inactive modules.
+  for (RtpRtcp* rtp_rtcp : deregistered_modules) {
+    rtp_rtcp->SetSendingStatus(false);
+    rtp_rtcp->SetSendingMediaStatus(false);
+  }
+
+  // Configure active modules.
+  for (RtpRtcp* rtp_rtcp : registered_modules) {
+    rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
+    if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
+      return -1;
+    }
+    rtp_rtcp->SetSendingStatus(is_sending);
+    rtp_rtcp->SetSendingMediaStatus(is_sending);
+  }
+
+  // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
+  // modules can be deleted after this step.
+  vie_receiver_.RegisterRtpRtcpModules(registered_modules);
+
+  // Update the packet and payload routers with the sending RtpRtcp modules.
+  if (sender_) {
+    std::list<RtpRtcp*> active_send_modules;
+    for (RtpRtcp* rtp_rtcp : registered_modules)
+      active_send_modules.push_back(rtp_rtcp);
+    send_payload_router_->SetSendingRtpModules(active_send_modules);
+  }
+
+  if (router_was_active)
+    send_payload_router_->set_active(true);
+
+  // Deregister previously registered modules.
+  for (size_t i = num_active_modules; i < num_prev_active_modules; ++i) {
+    module_process_thread_->DeRegisterModule(rtp_rtcp_modules_[i]);
+    packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
+  }
+  // Register new active modules.
+  for (size_t i = num_prev_active_modules; i < num_active_modules; ++i) {
+    module_process_thread_->RegisterModule(rtp_rtcp_modules_[i]);
+    packet_router_->AddRtpModule(rtp_rtcp_modules_[i]);
+  }
+  return 0;
+}
+
+int32_t ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) {
+  RTC_DCHECK(!sender_);
+  if (!vie_receiver_.SetReceiveCodec(video_codec)) {
+    return -1;
+  }
+
+  if (video_codec.codecType != kVideoCodecRED &&
+      video_codec.codecType != kVideoCodecULPFEC) {
+    // Register codec type with VCM, but do not register RED or ULPFEC.
+    if (vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_, false) !=
+        VCM_OK) {
+      return -1;
+    }
+  }
+  return 0;
+}
+
+void ViEChannel::RegisterExternalDecoder(const uint8_t pl_type,
+                                         VideoDecoder* decoder) {
+  RTC_DCHECK(!sender_);
+  vcm_->RegisterExternalDecoder(decoder, pl_type);
+}
+
+int32_t ViEChannel::ReceiveCodecStatistics(uint32_t* num_key_frames,
+                                           uint32_t* num_delta_frames) {
+  CriticalSectionScoped cs(crit_.get());
+  *num_key_frames = receive_frame_counts_.key_frames;
+  *num_delta_frames = receive_frame_counts_.delta_frames;
+  return 0;
+}
+
+uint32_t ViEChannel::DiscardedPackets() const {
+  return vcm_->DiscardedPackets();
+}
+
+int ViEChannel::ReceiveDelay() const {
+  return vcm_->Delay();
+}
+
+void ViEChannel::SetExpectedRenderDelay(int delay_ms) {
+  vcm_->SetRenderDelay(delay_ms);
+}
+
+void ViEChannel::SetRTCPMode(const RtcpMode rtcp_mode) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetRTCPStatus(rtcp_mode);
+}
+
+void ViEChannel::SetProtectionMode(bool enable_nack,
+                                   bool enable_fec,
+                                   int payload_type_red,
+                                   int payload_type_fec) {
+  // Validate payload types.
+  if (enable_fec) {
+    RTC_DCHECK_GE(payload_type_red, 0);
+    RTC_DCHECK_GE(payload_type_fec, 0);
+    RTC_DCHECK_LE(payload_type_red, 127);
+    RTC_DCHECK_LE(payload_type_fec, 127);
+  } else {
+    RTC_DCHECK_EQ(payload_type_red, -1);
+    RTC_DCHECK_EQ(payload_type_fec, -1);
+    // Set to valid uint8_ts to be castable later without signed overflows.
+    payload_type_red = 0;
+    payload_type_fec = 0;
+  }
+
+  VCMVideoProtection protection_method;
+  if (enable_nack) {
+    protection_method = enable_fec ? kProtectionNackFEC : kProtectionNack;
+  } else {
+    protection_method = kProtectionNone;
+  }
+
+  vcm_->SetVideoProtection(protection_method, true);
+
+  // Set NACK.
+  ProcessNACKRequest(enable_nack);
+
+  // Set FEC.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->SetGenericFECStatus(enable_fec,
+                                  static_cast<uint8_t>(payload_type_red),
+                                  static_cast<uint8_t>(payload_type_fec));
+  }
+}
+
+void ViEChannel::ProcessNACKRequest(const bool enable) {
+  if (enable) {
+    // Turn on NACK.
+    if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff)
+      return;
+    vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_);
+
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+      rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
+
+    vcm_->RegisterPacketRequestCallback(this);
+    // Don't introduce errors when NACK is enabled.
+    vcm_->SetDecodeErrorMode(kNoErrors);
+  } else {
+    vcm_->RegisterPacketRequestCallback(NULL);
+    if (paced_sender_ == nullptr) {
+      for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+        rtp_rtcp->SetStorePacketsStatus(false, 0);
+    }
+    vie_receiver_.SetNackStatus(false, max_nack_reordering_threshold_);
+    // When NACK is off, allow decoding with errors. Otherwise, the video
+    // will freeze, and will only recover with a complete key frame.
+    vcm_->SetDecodeErrorMode(kWithErrors);
+  }
+}
+
+bool ViEChannel::IsSendingFecEnabled() {
+  bool fec_enabled = false;
+  uint8_t pltype_red = 0;
+  uint8_t pltype_fec = 0;
+
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->GenericFECStatus(&fec_enabled, &pltype_red, &pltype_fec);
+    if (fec_enabled)
+      return true;
+  }
+  return false;
+}
+
+int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
+  if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
+    LOG(LS_ERROR) << "Invalid send buffer value.";
+    return -1;
+  }
+  if (target_delay_ms == 0) {
+    // Real-time mode.
+    nack_history_size_sender_ = kMinSendSidePacketHistorySize;
+  } else {
+    nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms);
+    // Don't allow a number lower than the default value.
+    if (nack_history_size_sender_ < kMinSendSidePacketHistorySize) {
+      nack_history_size_sender_ = kMinSendSidePacketHistorySize;
+    }
+  }
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
+  return 0;
+}
+
+int ViEChannel::GetRequiredNackListSize(int target_delay_ms) {
+  // The max size of the nack list should be large enough to accommodate the
+  // the number of packets (frames) resulting from the increased delay.
+  // Roughly estimating for ~40 packets per frame @ 30fps.
+  return target_delay_ms * 40 * 30 / 1000;
+}
+
+int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) {
+  // Disable any previous registrations of this extension to avoid errors.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->DeregisterSendRtpHeaderExtension(
+        kRtpExtensionTransmissionTimeOffset);
+  }
+  if (!enable)
+    return 0;
+  // Enable the extension.
+  int error = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
+        kRtpExtensionTransmissionTimeOffset, id);
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveTimestampOffsetStatus(bool enable, int id) {
+  return vie_receiver_.SetReceiveTimestampOffsetStatus(enable, id) ? 0 : -1;
+}
+
+int ViEChannel::SetSendAbsoluteSendTimeStatus(bool enable, int id) {
+  // Disable any previous registrations of this extension to avoid errors.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->DeregisterSendRtpHeaderExtension(kRtpExtensionAbsoluteSendTime);
+  if (!enable)
+    return 0;
+  // Enable the extension.
+  int error = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
+        kRtpExtensionAbsoluteSendTime, id);
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
+  return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1;
+}
+
+int ViEChannel::SetSendVideoRotationStatus(bool enable, int id) {
+  // Disable any previous registrations of this extension to avoid errors.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
+  if (!enable)
+    return 0;
+  // Enable the extension.
+  int error = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
+        kRtpExtensionVideoRotation, id);
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveVideoRotationStatus(bool enable, int id) {
+  return vie_receiver_.SetReceiveVideoRotationStatus(enable, id) ? 0 : -1;
+}
+
+int ViEChannel::SetSendTransportSequenceNumber(bool enable, int id) {
+  // Disable any previous registrations of this extension to avoid errors.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->DeregisterSendRtpHeaderExtension(
+        kRtpExtensionTransportSequenceNumber);
+  }
+  if (!enable)
+    return 0;
+  // Enable the extension.
+  int error = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
+        kRtpExtensionTransportSequenceNumber, id);
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) {
+  return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1;
+}
+
+int ViEChannel::SetSendRtpStreamId(bool enable, int id) { //}, const char *rid)
+  CriticalSectionScoped cs(crit_.get());
+  int error = 0;
+  if (enable) {
+    // Enable the extension, but disable possible old id to avoid errors.
+    rid_extension_id_ = id;
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+      rtp_rtcp->DeregisterSendRtpHeaderExtension(
+        kRtpExtensionRtpStreamId);
+      error = rtp_rtcp->RegisterSendRtpHeaderExtension(
+        kRtpExtensionRtpStreamId, id);
+    }
+    // NOTE: simulcast streams must be set via the SetSendCodec() API
+  } else {
+    // Disable the extension.
+    rid_extension_id_ = kInvalidRtpExtensionId;
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    	rtp_rtcp->DeregisterSendRtpHeaderExtension(
+        kRtpExtensionRtpStreamId);
+    }
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveRtpStreamId(bool enable, int id) {
+  return vie_receiver_.SetReceiveRIDStatus(enable, id) ? 0 : -1;
+}
+
+void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
+  rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable);
+}
+
+void ViEChannel::EnableTMMBR(bool enable) {
+  rtp_rtcp_modules_[0]->SetTMMBRStatus(enable);
+}
+
+int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
+                            const StreamType usage,
+                            const uint8_t simulcast_idx) {
+  RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx];
+  if (usage == kViEStreamTypeRtx) {
+    rtp_rtcp->SetRtxSsrc(SSRC);
+  } else {
+    rtp_rtcp->SetSSRC(SSRC);
+  }
+  return 0;
+}
+
+int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage,
+                                      const uint32_t SSRC) {
+  vie_receiver_.SetRtxSsrc(SSRC);
+  return 0;
+}
+
+int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) {
+  RTC_DCHECK_LE(idx, rtp_rtcp_modules_.size());
+  *ssrc = rtp_rtcp_modules_[idx]->SSRC();
+  return 0;
+}
+
+uint32_t ViEChannel::GetRemoteSSRC() {
+  return vie_receiver_.GetRemoteSsrc();
+}
+
+int ViEChannel::SetRtxSendPayloadType(int payload_type,
+                                      int associated_payload_type) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetRtxSendPayloadType(payload_type, associated_payload_type);
+  SetRtxSendStatus(true);
+  return 0;
+}
+
+void ViEChannel::SetRtxSendStatus(bool enable) {
+  int rtx_settings =
+      enable ? kRtxRetransmitted | kRtxRedundantPayloads : kRtxOff;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetRtxSendStatus(rtx_settings);
+}
+
+void ViEChannel::SetRtxReceivePayloadType(int payload_type,
+                                          int associated_payload_type) {
+  vie_receiver_.SetRtxPayloadType(payload_type, associated_payload_type);
+}
+
+void ViEChannel::SetUseRtxPayloadMappingOnRestore(bool val) {
+  vie_receiver_.SetUseRtxPayloadMappingOnRestore(val);
+}
+
+void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) {
+  RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending());
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    if (rtp_rtcp->SetRtpStateForSsrc(ssrc, rtp_state))
+      return;
+  }
+}
+
+RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) {
+  RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending());
+  RtpState rtp_state;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    if (rtp_rtcp->GetRtpStateForSsrc(ssrc, &rtp_state))
+      return rtp_state;
+  }
+  LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
+  return rtp_state;
+}
+
+// TODO(pbos): Set CNAME on all modules.
+int32_t ViEChannel::SetRTCPCName(const char* rtcp_cname) {
+  RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending());
+  return rtp_rtcp_modules_[0]->SetCNAME(rtcp_cname);
+}
+
+int32_t ViEChannel::GetRemoteRTCPCName(char rtcp_cname[]) {
+  uint32_t remoteSSRC = vie_receiver_.GetRemoteSsrc();
+  return rtp_rtcp_modules_[0]->RemoteCNAME(remoteSSRC, rtcp_cname);
+}
+
+int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost,
+                                          uint32_t* cumulative_lost,
+                                          uint32_t* extended_max,
+                                          uint32_t* jitter_samples,
+                                          int64_t* rtt_ms) {
+  // Aggregate the report blocks associated with streams sent on this channel.
+  std::vector<RTCPReportBlock> report_blocks;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->RemoteRTCPStat(&report_blocks);
+
+  if (report_blocks.empty())
+    return -1;
+
+  uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
+  std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin();
+  for (; it != report_blocks.end(); ++it) {
+    if (it->remoteSSRC == remote_ssrc)
+      break;
+  }
+  if (it == report_blocks.end()) {
+    // We have not received packets with an SSRC matching the report blocks. To
+    // have a chance of calculating an RTT we will try with the SSRC of the
+    // first report block received.
+    // This is very important for send-only channels where we don't know the
+    // SSRC of the other end.
+    remote_ssrc = report_blocks[0].remoteSSRC;
+  }
+
+  // TODO(asapersson): Change report_block_stats to not rely on
+  // GetSendRtcpStatistics to be called.
+  RTCPReportBlock report =
+      report_block_stats_sender_->AggregateAndStore(report_blocks);
+  *fraction_lost = report.fractionLost;
+  *cumulative_lost = report.cumulativeLost;
+  *extended_max = report.extendedHighSeqNum;
+  *jitter_samples = report.jitter;
+
+  int64_t dummy;
+  int64_t rtt = 0;
+  if (rtp_rtcp_modules_[0]->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) !=
+      0) {
+    return -1;
+  }
+  *rtt_ms = rtt;
+  return 0;
+}
+
+int32_t ViEChannel::GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh,
+                                              uint32_t& NTPLow,
+                                              uint32_t& receivedPacketCount,
+                                              uint64_t& receivedOctetCount,
+                                              uint32_t* jitterSamples,
+                                              uint16_t* fractionLost,
+                                              uint32_t* cumulativeLost,
+                                             int64_t* rttMs) {
+  // TODO: how do we do this for simulcast ? average for all
+  // except cumulative_lost that is the sum ?
+  // CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+
+  // for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
+  //      it != simulcast_rtp_rtcp_.end();
+  //      it++) {
+  //   RtpRtcp* rtp_rtcp = *it;
+  // }
+  uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
+
+  // Get all RTCP receiver report blocks that have been received on this
+  // channel. If we receive RTP packets from a remote source we know the
+  // remote SSRC and use the report block from him.
+  // Otherwise use the first report block.
+  std::vector<RTCPReportBlock> remote_stats;
+	for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->RemoteRTCPStat(&remote_stats);
+  }
+  if (remote_stats.empty()) {
+    LOG_F(LS_ERROR) << "Could not get remote stats";
+    return -1;
+  }
+  std::vector<RTCPReportBlock>::const_iterator statistics =
+      remote_stats.begin();
+  for (; statistics != remote_stats.end(); ++statistics) {
+    if (statistics->remoteSSRC == remote_ssrc)
+      break;
+  }
+
+  if (statistics == remote_stats.end()) {
+    // If we have not received any RTCP packets from this SSRC it probably means
+    // we have not received any RTP packets.
+    // Use the first received report block instead.
+    statistics = remote_stats.begin();
+    remote_ssrc = statistics->remoteSSRC;
+  }
+
+  if (!rtp_rtcp_modules_[0]) {
+    LOG_F(LS_ERROR) << "no RtpRtcp modules to retrieve RTT from";
+  } else {
+    if (rtp_rtcp_modules_[0]->GetReportBlockInfo(remote_ssrc, &NTPHigh, &NTPLow,
+          &receivedPacketCount, &receivedOctetCount) != 0) {
+      LOG_F(LS_ERROR) << "failed to retrieve block info";
+      NTPHigh = 0;
+      NTPLow = 0;
+      receivedPacketCount = 0;
+      receivedOctetCount = 0;
+    }
+  }
+  *fractionLost = statistics->fractionLost;
+  *cumulativeLost = statistics->cumulativeLost;
+  *jitterSamples = statistics->jitter;
+
+  int64_t dummy;
+  int64_t rtt = 0;
+  if (rtp_rtcp_modules_[0]->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
+    LOG_F(LS_ERROR) << "failed to get RTT";
+    return -1;
+  }
+  *rttMs = rtt;
+  return 0;
+}
+
+//->@@NG // int32_t ViEChannel::GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const {
+//->@@NG //   // Get the sender info from the latest received RTCP Sender Report.
+//->@@NG //   RTCPSenderInfo rtcp_sender_info;
+//->@@NG //   if (rtp_rtcp_->RemoteRTCPStat(&rtcp_sender_info) != 0) {
+//->@@NG //     LOG_F(LS_ERROR) << "failed to read RTCP SR sender info";
+//->@@NG //     return -1;
+//->@@NG //   }
+//->@@NG //
+//->@@NG //   sender_info->NTP_timestamp_high = rtcp_sender_info.NTPseconds;
+//->@@NG //   sender_info->NTP_timestamp_low = rtcp_sender_info.NTPfraction;
+//->@@NG //   sender_info->RTP_timestamp = rtcp_sender_info.RTPtimeStamp;
+//->@@NG //   sender_info->sender_packet_count = rtcp_sender_info.sendPacketCount;
+//->@@NG //   sender_info->sender_octet_count = rtcp_sender_info.sendOctetCount;
+//->@@NG //   return 0;
+//->@@NG // }
+
+void ViEChannel::RegisterSendChannelRtcpStatisticsCallback(
+    RtcpStatisticsCallback* callback) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->RegisterRtcpStatisticsCallback(callback);
+}
+
+void ViEChannel::RegisterReceiveChannelRtcpStatisticsCallback(
+    RtcpStatisticsCallback* callback) {
+  vie_receiver_.GetReceiveStatistics()->RegisterRtcpStatisticsCallback(
+      callback);
+  rtp_rtcp_modules_[0]->RegisterRtcpStatisticsCallback(callback);
+}
+
+void ViEChannel::RegisterRtcpPacketTypeCounterObserver(
+    RtcpPacketTypeCounterObserver* observer) {
+  rtcp_packet_type_counter_observer_.Set(observer);
+}
+
+void ViEChannel::GetSendStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  *rtp_counters = StreamDataCounters();
+  *rtx_counters = StreamDataCounters();
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    StreamDataCounters rtp_data;
+    StreamDataCounters rtx_data;
+    rtp_rtcp->GetSendStreamDataCounters(&rtp_data, &rtx_data);
+    rtp_counters->Add(rtp_data);
+    rtx_counters->Add(rtx_data);
+  }
+}
+
+void ViEChannel::GetReceiveStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
+      GetStatistician(vie_receiver_.GetRemoteSsrc());
+  if (statistician) {
+    statistician->GetReceiveStreamDataCounters(rtp_counters);
+  }
+  uint32_t rtx_ssrc = 0;
+  if (vie_receiver_.GetRtxSsrc(&rtx_ssrc)) {
+    StreamStatistician* statistician =
+        vie_receiver_.GetReceiveStatistics()->GetStatistician(rtx_ssrc);
+    if (statistician) {
+      statistician->GetReceiveStreamDataCounters(rtx_counters);
+    }
+  }
+}
+
+void ViEChannel::RegisterSendChannelRtpStatisticsCallback(
+      StreamDataCountersCallback* callback) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(callback);
+}
+
+void ViEChannel::RegisterReceiveChannelRtpStatisticsCallback(
+    StreamDataCountersCallback* callback) {
+  vie_receiver_.GetReceiveStatistics()->RegisterRtpStatisticsCallback(callback);
+}
+
+void ViEChannel::GetSendRtcpPacketTypeCounter(
+    RtcpPacketTypeCounter* packet_counter) const {
+  std::map<uint32_t, RtcpPacketTypeCounter> counter_map =
+      rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap();
+
+  RtcpPacketTypeCounter counter;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    counter.Add(counter_map[rtp_rtcp->SSRC()]);
+  *packet_counter = counter;
+}
+
+void ViEChannel::GetReceiveRtcpPacketTypeCounter(
+    RtcpPacketTypeCounter* packet_counter) const {
+  std::map<uint32_t, RtcpPacketTypeCounter> counter_map =
+      rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap();
+
+  RtcpPacketTypeCounter counter;
+  counter.Add(counter_map[vie_receiver_.GetRemoteSsrc()]);
+
+  *packet_counter = counter;
+}
+
+void ViEChannel::RegisterSendSideDelayObserver(
+    SendSideDelayObserver* observer) {
+  send_side_delay_observer_.Set(observer);
+}
+
+void ViEChannel::RegisterSendBitrateObserver(
+    BitrateStatisticsObserver* observer) {
+  send_bitrate_observer_.Set(observer);
+}
+
+int32_t ViEChannel::StartSend() {
+  CriticalSectionScoped cs(crit_.get());
+
+  if (rtp_rtcp_modules_[0]->Sending())
+    return -1;
+
+  for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i) {
+    RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[i];
+    rtp_rtcp->SetSendingMediaStatus(true);
+    rtp_rtcp->SetSendingStatus(true);
+  }
+  send_payload_router_->set_active(true);
+  return 0;
+}
+
+int32_t ViEChannel::StopSend() {
+  send_payload_router_->set_active(false);
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetSendingMediaStatus(false);
+
+  if (!rtp_rtcp_modules_[0]->Sending()) {
+    return -1;
+  }
+
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->SetSendingStatus(false);
+  }
+  return 0;
+}
+
+bool ViEChannel::Sending() {
+  return rtp_rtcp_modules_[0]->Sending();
+}
+
+void ViEChannel::StartReceive() {
+  if (!sender_)
+    StartDecodeThread();
+  vie_receiver_.StartReceive();
+}
+
+void ViEChannel::StopReceive() {
+  vie_receiver_.StopReceive();
+  if (!sender_) {
+    StopDecodeThread();
+    vcm_->ResetDecoder();
+  }
+}
+
+int32_t ViEChannel::ReceivedRTPPacket(const void* rtp_packet,
+                                      size_t rtp_packet_length,
+                                      const PacketTime& packet_time) {
+  return vie_receiver_.ReceivedRTPPacket(
+      rtp_packet, rtp_packet_length, packet_time);
+}
+
+int32_t ViEChannel::ReceivedRTCPPacket(const void* rtcp_packet,
+                                       size_t rtcp_packet_length) {
+  return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length);
+}
+
+int32_t ViEChannel::SetMTU(uint16_t mtu) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetMaxTransferUnit(mtu);
+  return 0;
+}
+
+RtpRtcp* ViEChannel::rtp_rtcp() {
+  return rtp_rtcp_modules_[0];
+}
+
+rtc::scoped_refptr<PayloadRouter> ViEChannel::send_payload_router() {
+  return send_payload_router_;
+}
+
+VCMProtectionCallback* ViEChannel::vcm_protection_callback() {
+  return vcm_protection_callback_.get();
+}
+
+CallStatsObserver* ViEChannel::GetStatsObserver() {
+  return stats_observer_.get();
+}
+
+// Do not acquire the lock of |vcm_| in this function. Decode callback won't
+// necessarily be called from the decoding thread. The decoding thread may have
+// held the lock when calling VideoDecoder::Decode, Reset, or Release. Acquiring
+// the same lock in the path of decode callback can deadlock.
+int32_t ViEChannel::FrameToRender(VideoFrame& video_frame) {  // NOLINT
+  CriticalSectionScoped cs(crit_.get());
+
+  if (pre_render_callback_ != NULL)
+    pre_render_callback_->FrameCallback(&video_frame);
+
+  // TODO(pbos): Remove stream id argument.
+  incoming_video_stream_->RenderFrame(0xFFFFFFFF, video_frame);
+  return 0;
+}
+
+int32_t ViEChannel::ReceivedDecodedReferenceFrame(
+  const uint64_t picture_id) {
+  return rtp_rtcp_modules_[0]->SendRTCPReferencePictureSelection(picture_id);
+}
+
+void ViEChannel::OnIncomingPayloadType(int payload_type) {
+  CriticalSectionScoped cs(crit_.get());
+  if (receive_stats_callback_)
+    receive_stats_callback_->OnIncomingPayloadType(payload_type);
+}
+
+void ViEChannel::OnDecoderImplementationName(const char* implementation_name) {
+  CriticalSectionScoped cs(crit_.get());
+  if (receive_stats_callback_)
+    receive_stats_callback_->OnDecoderImplementationName(implementation_name);
+}
+
+void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) {
+  CriticalSectionScoped cs(crit_.get());
+  if (receive_stats_callback_)
+    receive_stats_callback_->OnIncomingRate(frame_rate, bit_rate);
+}
+
+void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) {
+  CriticalSectionScoped cs(crit_.get());
+  if (receive_stats_callback_)
+    receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets);
+}
+
+void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) {
+  CriticalSectionScoped cs(crit_.get());
+  receive_frame_counts_ = frame_counts;
+  if (receive_stats_callback_)
+    receive_stats_callback_->OnFrameCountsUpdated(frame_counts);
+}
+
+void ViEChannel::OnDecoderTiming(int decode_ms,
+                                 int max_decode_ms,
+                                 int current_delay_ms,
+                                 int target_delay_ms,
+                                 int jitter_buffer_ms,
+                                 int min_playout_delay_ms,
+                                 int render_delay_ms) {
+  CriticalSectionScoped cs(crit_.get());
+  if (!receive_stats_callback_)
+    return;
+  receive_stats_callback_->OnDecoderTiming(
+      decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
+      jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt_ms_);
+}
+
+int32_t ViEChannel::RequestKeyFrame() {
+  return rtp_rtcp_modules_[0]->RequestKeyFrame();
+}
+
+int32_t ViEChannel::SliceLossIndicationRequest(
+  const uint64_t picture_id) {
+  return rtp_rtcp_modules_[0]->SendRTCPSliceLossIndication(
+      static_cast<uint8_t>(picture_id));
+}
+
+int32_t ViEChannel::ResendPackets(const uint16_t* sequence_numbers,
+                                  uint16_t length) {
+  return rtp_rtcp_modules_[0]->SendNACK(sequence_numbers, length);
+}
+
+void ViEChannel::ReceiveStateChange(VideoReceiveState state) {
+}
+
+bool ViEChannel::ChannelDecodeThreadFunction(void* obj) {
+  return static_cast<ViEChannel*>(obj)->ChannelDecodeProcess();
+}
+
+bool ViEChannel::ChannelDecodeProcess() {
+  vcm_->Decode(kMaxDecodeWaitTimeMs);
+  return true;
+}
+
+void ViEChannel::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
+  vcm_->SetReceiveChannelParameters(max_rtt_ms);
+
+  CriticalSectionScoped cs(crit_.get());
+  if (time_of_first_rtt_ms_ == -1)
+    time_of_first_rtt_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
+  rtt_sum_ms_ += avg_rtt_ms;
+  last_rtt_ms_ = avg_rtt_ms;
+  ++num_rtts_;
+}
+
+int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params,
+                                  const FecProtectionParams* key_fec_params,
+                                  uint32_t* video_rate_bps,
+                                  uint32_t* nack_rate_bps,
+                                  uint32_t* fec_rate_bps) {
+  *video_rate_bps = 0;
+  *nack_rate_bps = 0;
+  *fec_rate_bps = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    uint32_t not_used = 0;
+    uint32_t module_video_rate = 0;
+    uint32_t module_fec_rate = 0;
+    uint32_t module_nack_rate = 0;
+    rtp_rtcp->SetFecParameters(delta_fec_params, key_fec_params);
+    rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
+                          &module_nack_rate);
+    *video_rate_bps += module_video_rate;
+    *nack_rate_bps += module_nack_rate;
+    *fec_rate_bps += module_fec_rate;
+  }
+  return 0;
+}
+
+std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
+    bool receiver_only,
+    ReceiveStatistics* receive_statistics,
+    Transport* outgoing_transport,
+    RtcpIntraFrameObserver* intra_frame_callback,
+    RtcpBandwidthObserver* bandwidth_callback,
+    TransportFeedbackObserver* transport_feedback_callback,
+    RtcpRttStats* rtt_stats,
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+    RemoteBitrateEstimator* remote_bitrate_estimator,
+    RtpPacketSender* paced_sender,
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator,
+    BitrateStatisticsObserver* send_bitrate_observer,
+    FrameCountObserver* send_frame_count_observer,
+    SendSideDelayObserver* send_side_delay_observer,
+    size_t num_modules) {
+  RTC_DCHECK_GT(num_modules, 0u);
+  RtpRtcp::Configuration configuration;
+  ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
+  configuration.audio = false;
+  configuration.receiver_only = receiver_only;
+  configuration.receive_statistics = receive_statistics;
+  configuration.outgoing_transport = outgoing_transport;
+  configuration.intra_frame_callback = intra_frame_callback;
+  configuration.rtt_stats = rtt_stats;
+  configuration.rtcp_packet_type_counter_observer =
+      rtcp_packet_type_counter_observer;
+  configuration.paced_sender = paced_sender;
+  configuration.transport_sequence_number_allocator =
+      transport_sequence_number_allocator;
+  configuration.send_bitrate_observer = send_bitrate_observer;
+  configuration.send_frame_count_observer = send_frame_count_observer;
+  configuration.send_side_delay_observer = send_side_delay_observer;
+  configuration.bandwidth_callback = bandwidth_callback;
+  configuration.transport_feedback_callback = transport_feedback_callback;
+
+  std::vector<RtpRtcp*> modules;
+  for (size_t i = 0; i < num_modules; ++i) {
+    RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
+    rtp_rtcp->SetSendingStatus(false);
+    rtp_rtcp->SetSendingMediaStatus(false);
+    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
+    modules.push_back(rtp_rtcp);
+    // Receive statistics and remote bitrate estimator should only be set for
+    // the primary (first) module.
+    configuration.receive_statistics = null_receive_statistics;
+    configuration.remote_bitrate_estimator = nullptr;
+  }
+  return modules;
+}
+
+void ViEChannel::StartDecodeThread() {
+  RTC_DCHECK(!sender_);
+  if (decode_thread_.IsRunning())
+    return;
+  // Start the decode thread
+  decode_thread_.Start();
+  decode_thread_.SetPriority(rtc::kHighestPriority);
+}
+
+void ViEChannel::StopDecodeThread() {
+  vcm_->TriggerDecoderShutdown();
+
+  decode_thread_.Stop();
+}
+
+int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id,
+                                    VoEVideoSync* ve_sync_interface) {
+  return vie_sync_.ConfigureSync(ve_channel_id, ve_sync_interface,
+                                 rtp_rtcp_modules_[0],
+                                 vie_receiver_.GetRtpReceiver());
+}
+
+int32_t ViEChannel::VoiceChannel() {
+  return vie_sync_.VoiceChannel();
+}
+
+void ViEChannel::RegisterPreRenderCallback(
+    I420FrameCallback* pre_render_callback) {
+  CriticalSectionScoped cs(crit_.get());
+  pre_render_callback_ = pre_render_callback;
+}
+
+void ViEChannel::RegisterPreDecodeImageCallback(
+    EncodedImageCallback* pre_decode_callback) {
+  vcm_->RegisterPreDecodeImageCallback(pre_decode_callback);
+}
+
+// TODO(pbos): Remove OnInitializeDecoder which is called from the RTP module,
+// any decoder resetting should be handled internally within the VCM.
+int32_t ViEChannel::OnInitializeDecoder(
+    const int8_t payload_type,
+    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+    const int frequency,
+    const size_t channels,
+    const uint32_t rate) {
+  LOG(LS_INFO) << "OnInitializeDecoder " << static_cast<int>(payload_type)
+               << " " << payload_name;
+  vcm_->ResetDecoder();
+
+  return 0;
+}
+
+void ViEChannel::OnIncomingSSRCChanged(const uint32_t ssrc) {
+  rtp_rtcp_modules_[0]->SetRemoteSSRC(ssrc);
+}
+
+void ViEChannel::OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) {}
+
+void ViEChannel::RegisterSendFrameCountObserver(
+    FrameCountObserver* observer) {
+  send_frame_count_observer_.Set(observer);
+}
+
+void ViEChannel::RegisterReceiveStatisticsProxy(
+    ReceiveStatisticsProxy* receive_statistics_proxy) {
+  CriticalSectionScoped cs(crit_.get());
+  receive_stats_callback_ = receive_statistics_proxy;
+}
+
+void ViEChannel::SetIncomingVideoStream(
+    IncomingVideoStream* incoming_video_stream) {
+  CriticalSectionScoped cs(crit_.get());
+  incoming_video_stream_ = incoming_video_stream;
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel.h b/media/webrtc/trunk/webrtc/video/vie_channel.h
similarity index 50%
rename from media/webrtc/trunk/webrtc/video_engine/vie_channel.h
rename to media/webrtc/trunk/webrtc/video/vie_channel.h
index b4ea2d6747..f3cb34ced4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel.h
+++ b/media/webrtc/trunk/webrtc/video/vie_channel.h
@@ -8,27 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+#ifndef WEBRTC_VIDEO_VIE_CHANNEL_H_
+#define WEBRTC_VIDEO_VIE_CHANNEL_H_
 
 #include <list>
+#include <map>
+#include <vector>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-#include "webrtc/video_engine/vie_receiver.h"
-#include "webrtc/video_engine/vie_sender.h"
-#include "webrtc/video_engine/vie_sync_module.h"
+#include "webrtc/video/vie_receiver.h"
+#include "webrtc/video/vie_sync_module.h"
 
 namespace webrtc {
 
@@ -38,6 +36,7 @@ class Config;
 class CriticalSectionWrapper;
 class EncodedImageCallback;
 class I420FrameCallback;
+class IncomingVideoStream;
 class PacedSender;
 class PacketRouter;
 class PayloadRouter;
@@ -45,44 +44,41 @@ class ProcessThread;
 class ReceiveStatisticsProxy;
 class ReportBlockStats;
 class RtcpRttStats;
-class ThreadWrapper;
 class ViEChannelProtectionCallback;
-class ViEDecoderObserver;
-class ViEEffectFilter;
 class ViERTPObserver;
 class VideoCodingModule;
 class VideoDecoder;
 class VideoRenderCallback;
 class VoEVideoSync;
 
-struct SenderInfo;
+enum StreamType {
+  kViEStreamTypeNormal = 0,  // Normal media stream
+  kViEStreamTypeRtx = 1      // Retransmission media stream
+};
 
-class ViEChannel
-    : public VCMFrameTypeCallback,
-      public VCMReceiveCallback,
-      public VCMReceiveStatisticsCallback,
-      public VCMDecoderTimingCallback,
-      public VCMPacketRequestCallback,
-      public VCMReceiveStateCallback,
-      public RtpFeedback,
-      public ViEFrameProviderBase {
+class ViEChannel : public VCMFrameTypeCallback,
+                   public VCMReceiveCallback,
+									 public VCMReceiveStateCallback,
+                   public VCMReceiveStatisticsCallback,
+                   public VCMDecoderTimingCallback,
+                   public VCMPacketRequestCallback,
+                   public RtpFeedback {
  public:
   friend class ChannelStatsObserver;
   friend class ViEChannelProtectionCallback;
 
-  ViEChannel(int32_t channel_id,
-             int32_t engine_id,
-             uint32_t number_of_cores,
-             const Config& config,
-             ProcessThread& module_process_thread,
+  ViEChannel(uint32_t number_of_cores,
+             Transport* transport,
+             ProcessThread* module_process_thread,
              RtcpIntraFrameObserver* intra_frame_observer,
              RtcpBandwidthObserver* bandwidth_observer,
+             TransportFeedbackObserver* transport_feedback_observer,
              RemoteBitrateEstimator* remote_bitrate_estimator,
              RtcpRttStats* rtt_stats,
              PacedSender* paced_sender,
              PacketRouter* packet_router,
-             bool sender,
-             bool disable_default_encoder);
+             size_t max_rtp_streams,
+             bool sender);
   ~ViEChannel();
 
   int32_t Init();
@@ -91,17 +87,8 @@ class ViEChannel
   // type has changed and we should start a new RTP stream.
   int32_t SetSendCodec(const VideoCodec& video_codec, bool new_stream = true);
   int32_t SetReceiveCodec(const VideoCodec& video_codec);
-  int32_t GetReceiveCodec(VideoCodec* video_codec);
-  int32_t RegisterCodecObserver(ViEDecoderObserver* observer);
-  // Registers an external decoder. |buffered_rendering| means that the decoder
-  // will render frames after decoding according to the render timestamp
-  // provided by the video coding module. |render_delay| indicates the time
-  // needed to decode and render a frame.
-  int32_t RegisterExternalDecoder(const uint8_t pl_type,
-                                  VideoDecoder* decoder,
-                                  bool buffered_rendering,
-                                  int32_t render_delay);
-  int32_t DeRegisterExternalDecoder(const uint8_t pl_type);
+  // Registers an external decoder.
+  void RegisterExternalDecoder(const uint8_t pl_type, VideoDecoder* decoder);
   int32_t ReceiveCodecStatistics(uint32_t* num_key_frames,
                                  uint32_t* num_delta_frames);
   uint32_t DiscardedPackets() const;
@@ -109,41 +96,27 @@ class ViEChannel
   // Returns the estimated delay in milliseconds.
   int ReceiveDelay() const;
 
-  // Only affects calls to SetReceiveCodec done after this call.
-  int32_t WaitForKeyFrame(bool wait);
+  void SetExpectedRenderDelay(int delay_ms);
 
-  // If enabled, a key frame request will be sent as soon as there are lost
-  // packets. If |only_key_frames| are set, requests are only sent for loss in
-  // key frames.
-  int32_t SetSignalPacketLossStatus(bool enable, bool only_key_frames);
-
-  void SetRTCPMode(const RTCPMethod rtcp_mode);
-  RTCPMethod GetRTCPMode() const;
-  int32_t SetNACKStatus(const bool enable);
-  int32_t SetFECStatus(const bool enable,
-                       const unsigned char payload_typeRED,
-                       const unsigned char payload_typeFEC);
-  int32_t SetHybridNACKFECStatus(const bool enable,
-                                 const unsigned char payload_typeRED,
-                                 const unsigned char payload_typeFEC);
+  void SetRTCPMode(const RtcpMode rtcp_mode);
+  void SetProtectionMode(bool enable_nack,
+                         bool enable_fec,
+                         int payload_type_red,
+                         int payload_type_fec);
   bool IsSendingFecEnabled();
   int SetSenderBufferingMode(int target_delay_ms);
-  int SetReceiverBufferingMode(int target_delay_ms);
-  int32_t SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
-  void EnableRemb(bool enable);
   int SetSendTimestampOffsetStatus(bool enable, int id);
   int SetReceiveTimestampOffsetStatus(bool enable, int id);
   int SetSendAbsoluteSendTimeStatus(bool enable, int id);
   int SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
-  bool GetReceiveAbsoluteSendTimeStatus() const;
   int SetSendVideoRotationStatus(bool enable, int id);
   int SetReceiveVideoRotationStatus(bool enable, int id);
-  int SetSendRtpStreamId(bool enable, int id, const char* rid);
-  int SetReceiveRtpStreamId(bool enable, int id);
+  int SetSendTransportSequenceNumber(bool enable, int id);
+  int SetReceiveTransportSequenceNumber(bool enable, int id);
+  int SetSendRtpStreamId(bool enable, int id); // RtpStreamId (RID)
+	int SetReceiveRtpStreamId(bool enable, int id); // RtpStreamId (RID)
   void SetRtcpXrRrtrStatus(bool enable);
-  void SetTransmissionSmoothingStatus(bool enable);
   void EnableTMMBR(bool enable);
-  int32_t EnableKeyFrameRequestCallback(const bool enable);
 
   // Sets SSRC for outgoing stream.
   int32_t SetSSRC(const uint32_t SSRC,
@@ -154,73 +127,56 @@ class ViEChannel
   int32_t GetLocalSSRC(uint8_t idx, unsigned int* ssrc);
 
   // Gets SSRC for the incoming stream.
-  int32_t GetRemoteSSRC(uint32_t* ssrc);
+  uint32_t GetRemoteSSRC();
 
-  // Gets the CSRC for the incoming stream.
-  int32_t GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]);
+	// MOZ addition Gets the RID (if any) for the incoming stream.
+	int32_t GetRemoteRtpStreamId(char rid[256]);
 
-  // Gets the RID (if any) for the incoming stream.
-  int32_t GetRemoteRID(char rid[256]);
-
-  int SetRtxSendPayloadType(int payload_type);
-  void SetRtxReceivePayloadType(int payload_type);
-
-  // Sets the starting sequence number, must be called before StartSend.
-  int32_t SetStartSequenceNumber(uint16_t sequence_number);
+  int SetRtxSendPayloadType(int payload_type, int associated_payload_type);
+  void SetRtxReceivePayloadType(int payload_type, int associated_payload_type);
+  // If set to true, the RTX payload type mapping supplied in
+  // |SetRtxReceivePayloadType| will be used when restoring RTX packets. Without
+  // it, RTX packets will always be restored to the last non-RTX packet payload
+  // type received.
+  void SetUseRtxPayloadMappingOnRestore(bool val);
 
   void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
   RtpState GetRtpStateForSsrc(uint32_t ssrc);
 
   // Sets the CName for the outgoing stream on the channel.
-  int32_t SetRTCPCName(const char rtcp_cname[]);
+  int32_t SetRTCPCName(const char* rtcp_cname);
 
   // Gets the CName of the incoming stream.
   int32_t GetRemoteRTCPCName(char rtcp_cname[]);
-  int32_t RegisterRtpObserver(ViERTPObserver* observer);
-  int32_t SendApplicationDefinedRTCPPacket(
-      const uint8_t sub_type,
-      uint32_t name,
-      const uint8_t* data,
-      uint16_t data_length_in_bytes);
-
-  // Gets info (including timestamp) from last rr + remote packetcount
-  // (derived from rr report + cached sender-side info).
-  int32_t GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh, uint32_t& NTPLow,
-                                    uint32_t& receivedPacketCount,
-                                    uint64_t& receivedOctetCount,
-                                    uint32_t* jitterSamples,
-                                    uint16_t* fractionLost,
-                                    uint32_t* cumulativeLost,
-                                    int32_t* rttMs);
 
   // Returns statistics reported by the remote client in an RTCP packet.
+  // TODO(pbos): Remove this along with VideoSendStream::GetRtt().
   int32_t GetSendRtcpStatistics(uint16_t* fraction_lost,
                                 uint32_t* cumulative_lost,
                                 uint32_t* extended_max,
                                 uint32_t* jitter_samples,
                                 int64_t* rtt_ms);
 
+// Gets info (including timestamp) from last rr + remote packetcount
+// (derived from rr report + cached sender-side info).
+int32_t GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh, uint32_t& NTPLow,
+                                  uint32_t& receivedPacketCount,
+                                  uint64_t& receivedOctetCount,
+                                  uint32_t* jitterSamples,
+                                  uint16_t* fractionLost,
+                                  uint32_t* cumulativeLost,
+                                  int64_t* rttMs);
+
+int32_t GetRemoteRTCPSenderInfo(RTCPSenderInfo* sender_info) const;
+
   // Called on receipt of RTCP report block from remote side.
   void RegisterSendChannelRtcpStatisticsCallback(
       RtcpStatisticsCallback* callback);
 
-  // Returns our localy created statistics of the received RTP stream.
-  int32_t GetReceivedRtcpStatistics(uint16_t* fraction_lost,
-                                    uint32_t* cumulative_lost,
-                                    uint32_t* extended_max,
-                                    uint32_t* jitter_samples,
-                                    int64_t* rtt_ms);
-
   // Called on generation of RTCP stats
   void RegisterReceiveChannelRtcpStatisticsCallback(
       RtcpStatisticsCallback* callback);
 
-  // Gets sent/received packets statistics.
-  int32_t GetRtpStatistics(size_t* bytes_sent,
-                           uint32_t* packets_sent,
-                           size_t* bytes_received,
-                           uint32_t* packets_received) const;
-
   // Gets send statistics for the rtp and rtx stream.
   void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
                                  StreamDataCounters* rtx_counters) const;
@@ -243,78 +199,31 @@ class ViEChannel
   void GetReceiveRtcpPacketTypeCounter(
       RtcpPacketTypeCounter* packet_counter) const;
 
-
-  int32_t GetRemoteRTCPSenderInfo(SenderInfo* sender_info) const;
-
-  void GetBandwidthUsage(uint32_t* total_bitrate_sent,
-                         uint32_t* video_bitrate_sent,
-                         uint32_t* fec_bitrate_sent,
-                         uint32_t* nackBitrateSent) const;
-  // TODO(holmer): Deprecated. We should use the SendSideDelayObserver instead
-  // to avoid deadlocks.
-  bool GetSendSideDelay(int* avg_send_delay, int* max_send_delay) const;
   void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
 
   // Called on any new send bitrate estimate.
   void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
 
-  int32_t StartRTPDump(const char file_nameUTF8[1024],
-                       RTPDirections direction);
-  int32_t StopRTPDump(RTPDirections direction);
-
   // Implements RtpFeedback.
-  virtual int32_t OnInitializeDecoder(
-      const int32_t id,
-      const int8_t payload_type,
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const int frequency,
-      const uint8_t channels,
-      const uint32_t rate);
-  virtual void OnIncomingSSRCChanged(const int32_t id,
-                                     const uint32_t ssrc);
-  virtual void OnIncomingCSRCChanged(const int32_t id,
-                                     const uint32_t CSRC,
-                                     const bool added);
-  virtual void ResetStatistics(uint32_t);
-
-  int32_t SetLocalReceiver(const uint16_t rtp_port,
-                           const uint16_t rtcp_port,
-                           const char* ip_address);
-  int32_t GetLocalReceiver(uint16_t* rtp_port,
-                           uint16_t* rtcp_port,
-                           char* ip_address) const;
-  int32_t SetSendDestination(const char* ip_address,
-                             const uint16_t rtp_port,
-                             const uint16_t rtcp_port,
-                             const uint16_t source_rtp_port,
-                             const uint16_t source_rtcp_port);
-  int32_t GetSendDestination(char* ip_address,
-                             uint16_t* rtp_port,
-                             uint16_t* rtcp_port,
-                             uint16_t* source_rtp_port,
-                             uint16_t* source_rtcp_port) const;
-  int32_t GetSourceInfo(uint16_t* rtp_port,
-                        uint16_t* rtcp_port,
-                        char* ip_address,
-                        uint32_t ip_address_length);
+  int32_t OnInitializeDecoder(const int8_t payload_type,
+                              const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+                              const int frequency,
+                              const size_t channels,
+                              const uint32_t rate) override;
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override;
+  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override;
 
   int32_t SetRemoteSSRCType(const StreamType usage, const uint32_t SSRC);
 
   int32_t StartSend();
   int32_t StopSend();
   bool Sending();
-  int32_t StartReceive();
-  int32_t StopReceive();
+  void StartReceive();
+  void StopReceive();
 
-  int32_t RegisterSendTransport(Transport* transport);
-  int32_t DeregisterSendTransport();
-
-  // Incoming packet from external transport.
   int32_t ReceivedRTPPacket(const void* rtp_packet,
                             const size_t rtp_packet_length,
                             const PacketTime& packet_time);
-
-  // Incoming packet from external transport.
   int32_t ReceivedRTCPPacket(const void* rtcp_packet,
                              const size_t rtcp_packet_length);
 
@@ -322,31 +231,24 @@ class ViEChannel
   // IP, UDP and RTP headers.
   int32_t SetMTU(uint16_t mtu);
 
-  // Returns maximum allowed payload size, i.e. the maximum allowed size of
-  // encoded data in each packet.
-  uint16_t MaxDataPayloadLength() const;
-  int32_t SetMaxPacketBurstSize(uint16_t max_number_of_packets);
-  int32_t SetPacketBurstSpreadState(bool enable, const uint16_t frame_periodMS);
-
-  int32_t EnableColorEnhancement(bool enable);
-
   // Gets the modules used by the channel.
   RtpRtcp* rtp_rtcp();
-  scoped_refptr<PayloadRouter> send_payload_router();
+  rtc::scoped_refptr<PayloadRouter> send_payload_router();
   VCMProtectionCallback* vcm_protection_callback();
 
 
   CallStatsObserver* GetStatsObserver();
 
   // Implements VCMReceiveCallback.
-  virtual int32_t FrameToRender(I420VideoFrame& video_frame);  // NOLINT
+  virtual int32_t FrameToRender(VideoFrame& video_frame);  // NOLINT
 
   // Implements VCMReceiveCallback.
   virtual int32_t ReceivedDecodedReferenceFrame(
       const uint64_t picture_id);
 
   // Implements VCMReceiveCallback.
-  virtual void IncomingCodecChanged(const VideoCodec& codec);
+  void OnIncomingPayloadType(int payload_type) override;
+  void OnDecoderImplementationName(const char* implementation_name) override;
 
   // Implements VCMReceiveStatisticsCallback.
   void OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) override;
@@ -362,29 +264,23 @@ class ViEChannel
                                int min_playout_delay_ms,
                                int render_delay_ms);
 
-  // Implements VideoFrameTypeCallback.
+  // Implements FrameTypeCallback.
   virtual int32_t RequestKeyFrame();
 
-  // Implements VideoFrameTypeCallback.
+  // Implements FrameTypeCallback.
   virtual int32_t SliceLossIndicationRequest(
       const uint64_t picture_id);
 
   // Implements VideoPacketRequestCallback.
-  virtual int32_t ResendPackets(const uint16_t* sequence_numbers,
-                                uint16_t length);
+  int32_t ResendPackets(const uint16_t* sequence_numbers,
+                        uint16_t length) override;
 
-  // Implements ReceiveStateCallback.
-  virtual void ReceiveStateChange(VideoReceiveState state);
+  virtual void ReceiveStateChange(VideoReceiveState state) override;
 
   int32_t SetVoiceChannel(int32_t ve_channel_id,
                           VoEVideoSync* ve_sync_interface);
   int32_t VoiceChannel();
 
-  // Implements ViEFrameProviderBase.
-  virtual int FrameCallbackChanged() {return -1;}
-
-  int32_t RegisterEffectFilter(ViEEffectFilter* effect_filter);
-
   // New-style callbacks, used by VideoReceiveStream.
   void RegisterPreRenderCallback(I420FrameCallback* pre_render_callback);
   void RegisterPreDecodeImageCallback(
@@ -395,14 +291,13 @@ class ViEChannel
       RtcpPacketTypeCounterObserver* observer);
   void RegisterReceiveStatisticsProxy(
       ReceiveStatisticsProxy* receive_statistics_proxy);
-  void ReceivedBWEPacket(int64_t arrival_time_ms, size_t payload_size,
-                         const RTPHeader& header);
+  void SetIncomingVideoStream(IncomingVideoStream* incoming_video_stream);
 
  protected:
   static bool ChannelDecodeThreadFunction(void* obj);
   bool ChannelDecodeProcess();
 
-  void OnRttUpdate(int64_t rtt);
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms);
 
   int ProtectionRequest(const FecProtectionParams* delta_fec_params,
                         const FecProtectionParams* key_fec_params,
@@ -411,26 +306,33 @@ class ViEChannel
                         uint32_t* sent_fec_rate_bps);
 
  private:
-  void ReserveRtpRtcpModules(size_t total_modules)
-      EXCLUSIVE_LOCKS_REQUIRED(rtp_rtcp_cs_);
-  RtpRtcp* GetRtpRtcpModule(size_t simulcast_idx) const
-      EXCLUSIVE_LOCKS_REQUIRED(rtp_rtcp_cs_);
-  RtpRtcp::Configuration CreateRtpRtcpConfiguration();
-  RtpRtcp* CreateRtpRtcpModule();
-  // Assumed to be protected.
-  int32_t StartDecodeThread();
-  int32_t StopDecodeThread();
+  static std::vector<RtpRtcp*> CreateRtpRtcpModules(
+      bool receiver_only,
+      ReceiveStatistics* receive_statistics,
+      Transport* outgoing_transport,
+      RtcpIntraFrameObserver* intra_frame_callback,
+      RtcpBandwidthObserver* bandwidth_callback,
+      TransportFeedbackObserver* transport_feedback_callback,
+      RtcpRttStats* rtt_stats,
+      RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+      RemoteBitrateEstimator* remote_bitrate_estimator,
+      RtpPacketSender* paced_sender,
+      TransportSequenceNumberAllocator* transport_sequence_number_allocator,
+      BitrateStatisticsObserver* send_bitrate_observer,
+      FrameCountObserver* send_frame_count_observer,
+      SendSideDelayObserver* send_side_delay_observer,
+      size_t num_modules);
 
-  int32_t ProcessNACKRequest(const bool enable);
-  int32_t ProcessFECRequest(const bool enable,
-                            const unsigned char payload_typeRED,
-                            const unsigned char payload_typeFEC);
+  // Assumed to be protected.
+  void StartDecodeThread();
+  void StopDecodeThread();
+
+  void ProcessNACKRequest(const bool enable);
   // Compute NACK list parameters for the buffering mode.
   int GetRequiredNackListSize(int target_delay_ms);
   void SetRtxSendStatus(bool enable);
 
   void UpdateHistograms();
-  void UpdateHistogramsAtStopSend();
 
   // ViEChannel exposes methods that allow to modify observers and callbacks
   // to be modified. Such an API-style is cumbersome to implement and maintain
@@ -457,7 +359,7 @@ class ViEChannel
     T* callback_ GUARDED_BY(critsect_);
 
    private:
-    DISALLOW_COPY_AND_ASSIGN(RegisterableCallback);
+    RTC_DISALLOW_COPY_AND_ASSIGN(RegisterableCallback);
   };
 
   class RegisterableBitrateStatisticsObserver:
@@ -518,74 +420,56 @@ class ViEChannel
         GUARDED_BY(critsect_);
   } rtcp_packet_type_counter_observer_;
 
-  int32_t channel_id_;
-  int32_t engine_id_;
-  uint32_t number_of_cores_;
-  uint8_t num_socket_threads_;
+  const uint32_t number_of_cores_;
+  const bool sender_;
+
+  ProcessThread* const module_process_thread_;
 
   // Used for all registered callbacks except rendering.
-  rtc::scoped_ptr<CriticalSectionWrapper> callback_cs_;
-  rtc::scoped_ptr<CriticalSectionWrapper> rtp_rtcp_cs_;
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
 
   // Owned modules/classes.
-  rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
-  std::list<RtpRtcp*> simulcast_rtp_rtcp_;
-  std::list<RtpRtcp*> removed_rtp_rtcp_;
-  scoped_refptr<PayloadRouter> send_payload_router_;
+  rtc::scoped_refptr<PayloadRouter> send_payload_router_;
   rtc::scoped_ptr<ViEChannelProtectionCallback> vcm_protection_callback_;
 
   VideoCodingModule* const vcm_;
   ViEReceiver vie_receiver_;
-  ViESender vie_sender_;
   ViESyncModule vie_sync_;
 
   // Helper to report call statistics.
   rtc::scoped_ptr<ChannelStatsObserver> stats_observer_;
 
   // Not owned.
-  VCMReceiveStatisticsCallback* vcm_receive_stats_callback_
-      GUARDED_BY(callback_cs_);
-  FrameCounts receive_frame_counts_ GUARDED_BY(callback_cs_);
-  ProcessThread& module_process_thread_;
-  ViEDecoderObserver* codec_observer_;
-  bool do_key_frame_callbackRequest_;
-  ViERTPObserver* rtp_observer_;
-  RtcpIntraFrameObserver* intra_frame_observer_;
-  RtcpRttStats* rtt_stats_;
-  PacedSender* paced_sender_;
-  PacketRouter* packet_router_;
+  ReceiveStatisticsProxy* receive_stats_callback_ GUARDED_BY(crit_);
+  FrameCounts receive_frame_counts_ GUARDED_BY(crit_);
+  IncomingVideoStream* incoming_video_stream_ GUARDED_BY(crit_);
+  RtcpIntraFrameObserver* const intra_frame_observer_;
+  RtcpRttStats* const rtt_stats_;
+  PacedSender* const paced_sender_;
+  PacketRouter* const packet_router_;
 
-  rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_;
-  int send_timestamp_extension_id_;
-  int absolute_send_time_extension_id_;
-  int video_rotation_extension_id_;
-  int rid_extension_id_;
+  const rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_;
+  TransportFeedbackObserver* const transport_feedback_observer_;
 
-  Transport* external_transport_;
-
-  bool decoder_reset_;
-  // Current receive codec used for codec change callback.
-  VideoCodec receive_codec_;
-  bool wait_for_key_frame_;
-  rtc::scoped_ptr<ThreadWrapper> decode_thread_;
-
-  ViEEffectFilter* effect_filter_;
-  bool color_enhancement_;
-
-  // User set MTU, -1 if not set.
-  uint16_t mtu_;
-  const bool sender_;
-  // Used to skip default encoder in the new API.
-  const bool disable_default_encoder_;
+  rtc::PlatformThread decode_thread_;
 
   int nack_history_size_sender_;
   int max_nack_reordering_threshold_;
-  I420FrameCallback* pre_render_callback_;
+  I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_);
 
-  rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;
-  rtc::scoped_ptr<ReportBlockStats> report_block_stats_receiver_;
+  const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;
+
+  int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);
+  int64_t rtt_sum_ms_ GUARDED_BY(crit_);
+  int64_t last_rtt_ms_ GUARDED_BY(crit_);
+  size_t num_rtts_ GUARDED_BY(crit_);
+	int rid_extension_id_; // RtpStreamId (RID)
+
+  // RtpRtcp modules, declared last as they use other members on construction.
+  const std::vector<RtpRtcp*> rtp_rtcp_modules_;
+  size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+#endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_codec_unittest.cc b/media/webrtc/trunk/webrtc/video/vie_codec_unittest.cc
similarity index 100%
rename from media/webrtc/trunk/webrtc/video_engine/vie_codec_unittest.cc
rename to media/webrtc/trunk/webrtc/video/vie_codec_unittest.cc
diff --git a/media/webrtc/trunk/webrtc/video/vie_encoder.cc b/media/webrtc/trunk/webrtc/video/vie_encoder.cc
new file mode 100644
index 0000000000..19a8eb0028
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/vie_encoder.cc
@@ -0,0 +1,638 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/vie_encoder.h"
+
+#include <assert.h>
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/common_video/include/video_image.h"
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/frame_callback.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/send_statistics_proxy.h"
+
+namespace webrtc {
+
+// Margin on when we pause the encoder when the pacing buffer overflows relative
+// to the configured buffer delay.
+static const float kEncoderPausePacerMargin = 2.0f;
+
+// Don't stop the encoder unless the delay is above this configured value.
+static const int kMinPacingDelayMs = 200;
+
+static const float kStopPaddingThresholdMs = 2000;
+
+static const int kMinKeyFrameRequestIntervalMs = 300;
+
+std::vector<uint32_t> AllocateStreamBitrates(
+    uint32_t total_bitrate,
+    const SimulcastStream* stream_configs,
+    size_t number_of_streams) {
+  if (number_of_streams == 0) {
+    std::vector<uint32_t> stream_bitrates(1, 0);
+    stream_bitrates[0] = total_bitrate;
+    return stream_bitrates;
+  }
+  std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
+  uint32_t bitrate_remainder = total_bitrate;
+  for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
+    if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
+      stream_bitrates[i] = bitrate_remainder;
+    } else {
+      stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
+    }
+    bitrate_remainder -= stream_bitrates[i];
+  }
+  return stream_bitrates;
+}
+
+class QMVideoSettingsCallback : public VCMQMSettingsCallback {
+ public:
+  explicit QMVideoSettingsCallback(VideoProcessing* vpm);
+
+  ~QMVideoSettingsCallback();
+
+  // Update VPM with QM (quality modes: frame size & frame rate) settings.
+  int32_t SetVideoQMSettings(const uint32_t frame_rate,
+                             const uint32_t width,
+                             const uint32_t height);
+
+  // Update target frame rate.
+  void SetTargetFramerate(int frame_rate);
+
+ private:
+  VideoProcessing* vp_;
+};
+
+class ViEBitrateObserver : public BitrateObserver {
+ public:
+  explicit ViEBitrateObserver(ViEEncoder* owner)
+      : owner_(owner) {
+  }
+  virtual ~ViEBitrateObserver() {}
+  // Implements BitrateObserver.
+  virtual void OnNetworkChanged(uint32_t bitrate_bps,
+                                uint8_t fraction_lost,
+                                int64_t rtt) {
+    owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
+  }
+ private:
+  ViEEncoder* owner_;
+};
+
+ViEEncoder::ViEEncoder(uint32_t number_of_cores,
+                       ProcessThread* module_process_thread,
+                       SendStatisticsProxy* stats_proxy,
+                       I420FrameCallback* pre_encode_callback,
+                       PacedSender* pacer,
+                       BitrateAllocator* bitrate_allocator)
+    : number_of_cores_(number_of_cores),
+      vp_(VideoProcessing::Create()),
+      qm_callback_(new QMVideoSettingsCallback(vp_.get())),
+      vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(),
+                                     this,
+                                     qm_callback_.get())),
+      send_payload_router_(NULL),
+      data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      stats_proxy_(stats_proxy),
+      pre_encode_callback_(pre_encode_callback),
+      pacer_(pacer),
+      bitrate_allocator_(bitrate_allocator),
+      time_of_last_frame_activity_ms_(0),
+      encoder_config_(),
+      min_transmit_bitrate_kbps_(0),
+      last_observed_bitrate_bps_(0),
+      target_delay_ms_(0),
+      network_is_transmitting_(true),
+      encoder_paused_(false),
+      encoder_paused_and_dropped_frame_(false),
+      module_process_thread_(module_process_thread),
+      has_received_sli_(false),
+      picture_id_sli_(0),
+      has_received_rpsi_(false),
+      picture_id_rpsi_(0),
+      video_suspended_(false) {
+  bitrate_observer_.reset(new ViEBitrateObserver(this));
+}
+
+bool ViEEncoder::Init() {
+  vp_->EnableTemporalDecimation(true);
+
+  // Enable/disable content analysis: off by default for now.
+  vp_->EnableContentAnalysis(false);
+
+  if (vcm_->RegisterTransportCallback(this) != 0) {
+    return false;
+  }
+  if (vcm_->RegisterSendStatisticsCallback(this) != 0) {
+    return false;
+  }
+  return true;
+}
+
+void ViEEncoder::StartThreadsAndSetSharedMembers(
+    rtc::scoped_refptr<PayloadRouter> send_payload_router,
+    VCMProtectionCallback* vcm_protection_callback) {
+  RTC_DCHECK(send_payload_router_ == NULL);
+
+  send_payload_router_ = send_payload_router;
+  vcm_->RegisterProtectionCallback(vcm_protection_callback);
+  module_process_thread_->RegisterModule(vcm_.get());
+}
+
+void ViEEncoder::StopThreadsAndRemoveSharedMembers() {
+  if (bitrate_allocator_)
+    bitrate_allocator_->RemoveBitrateObserver(bitrate_observer_.get());
+  module_process_thread_->DeRegisterModule(vcm_.get());
+}
+
+ViEEncoder::~ViEEncoder() {
+}
+
+void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    network_is_transmitting_ = is_transmitting;
+  }
+}
+
+void ViEEncoder::Pause() {
+  CriticalSectionScoped cs(data_cs_.get());
+  encoder_paused_ = true;
+}
+
+void ViEEncoder::Restart() {
+  CriticalSectionScoped cs(data_cs_.get());
+  encoder_paused_ = false;
+}
+
+int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
+                                            uint8_t pl_type,
+                                            bool internal_source) {
+  if (vcm_->RegisterExternalEncoder(encoder, pl_type, internal_source) !=
+      VCM_OK) {
+    return -1;
+  }
+  return 0;
+}
+
+int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
+  if (vcm_->RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
+    return -1;
+  }
+  return 0;
+}
+
+int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
+  RTC_DCHECK(send_payload_router_ != NULL);
+  // Setting target width and height for VPM.
+  if (vp_->SetTargetResolution(video_codec.width, video_codec.height,
+                               video_codec.maxFramerate) != VPM_OK) {
+    return -1;
+  }
+
+  // Cache codec before calling AddBitrateObserver (which calls OnNetworkChanged
+  // that makes use of the number of simulcast streams configured).
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    encoder_config_ = video_codec;
+  }
+
+  // Add a bitrate observer to the allocator and update the start, max and
+  // min bitrates of the bitrate controller as needed.
+  int allocated_bitrate_bps = bitrate_allocator_->AddBitrateObserver(
+      bitrate_observer_.get(), video_codec.minBitrate * 1000,
+      video_codec.maxBitrate * 1000);
+
+  webrtc::VideoCodec modified_video_codec = video_codec;
+  modified_video_codec.startBitrate = allocated_bitrate_bps / 1000;
+
+  size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
+  if (vcm_->RegisterSendCodec(&modified_video_codec, number_of_cores_,
+                              static_cast<uint32_t>(max_data_payload_length)) !=
+      VCM_OK) {
+    return -1;
+  }
+  return 0;
+}
+
+int ViEEncoder::GetPaddingNeededBps() const {
+  int64_t time_of_last_frame_activity_ms;
+  int min_transmit_bitrate_bps;
+  int bitrate_bps;
+  VideoCodec send_codec;
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    bool send_padding = encoder_config_.numberOfSimulcastStreams > 1 ||
+                        video_suspended_ || min_transmit_bitrate_kbps_ > 0;
+    if (!send_padding)
+      return 0;
+    time_of_last_frame_activity_ms = time_of_last_frame_activity_ms_;
+    min_transmit_bitrate_bps = 1000 * min_transmit_bitrate_kbps_;
+    bitrate_bps = last_observed_bitrate_bps_;
+    send_codec = encoder_config_;
+  }
+
+  bool video_is_suspended = vcm_->VideoSuspended();
+
+  // Find the max amount of padding we can allow ourselves to send at this
+  // point, based on which streams are currently active and what our current
+  // available bandwidth is.
+  int pad_up_to_bitrate_bps = 0;
+  if (send_codec.numberOfSimulcastStreams == 0) {
+    pad_up_to_bitrate_bps = send_codec.minBitrate * 1000;
+  } else {
+    SimulcastStream* stream_configs = send_codec.simulcastStream;
+    pad_up_to_bitrate_bps =
+        stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate *
+        1000;
+    for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
+      pad_up_to_bitrate_bps += stream_configs[i].targetBitrate * 1000;
+    }
+  }
+
+  // Disable padding if only sending one stream and video isn't suspended and
+  // min-transmit bitrate isn't used (applied later).
+  if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
+    pad_up_to_bitrate_bps = 0;
+
+  // The amount of padding should decay to zero if no frames are being
+  // captured/encoded unless a min-transmit bitrate is used.
+  int64_t now_ms = TickTime::MillisecondTimestamp();
+  if (now_ms - time_of_last_frame_activity_ms > kStopPaddingThresholdMs)
+    pad_up_to_bitrate_bps = 0;
+
+  // Pad up to min bitrate.
+  if (pad_up_to_bitrate_bps < min_transmit_bitrate_bps)
+    pad_up_to_bitrate_bps = min_transmit_bitrate_bps;
+
+  // Padding may never exceed bitrate estimate.
+  if (pad_up_to_bitrate_bps > bitrate_bps)
+    pad_up_to_bitrate_bps = bitrate_bps;
+
+  return pad_up_to_bitrate_bps;
+}
+
+bool ViEEncoder::EncoderPaused() const {
+  // Pause video if paused by caller or as long as the network is down or the
+  // pacer queue has grown too large in buffered mode.
+  if (encoder_paused_) {
+    return true;
+  }
+  if (target_delay_ms_ > 0) {
+    // Buffered mode.
+    // TODO(pwestin): Workaround until nack is configured as a time and not
+    // number of packets.
+    return pacer_->QueueInMs() >=
+           std::max(
+               static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
+               kMinPacingDelayMs);
+  }
+  if (pacer_->ExpectedQueueTimeMs() > PacedSender::kMaxQueueLengthMs) {
+    // Too much data in pacer queue, drop frame.
+    return true;
+  }
+  return !network_is_transmitting_;
+}
+
+void ViEEncoder::TraceFrameDropStart() {
+  // Start trace event only on the first frame after encoder is paused.
+  if (!encoder_paused_and_dropped_frame_) {
+    TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
+  }
+  encoder_paused_and_dropped_frame_ = true;
+  return;
+}
+
+void ViEEncoder::TraceFrameDropEnd() {
+  // End trace event on first frame after encoder resumes, if frame was dropped.
+  if (encoder_paused_and_dropped_frame_) {
+    TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
+  }
+  encoder_paused_and_dropped_frame_ = false;
+}
+
+void ViEEncoder::DeliverFrame(VideoFrame video_frame) {
+  RTC_DCHECK(send_payload_router_ != NULL);
+  if (!send_payload_router_->active()) {
+    // We've paused or we have no channels attached, don't waste resources on
+    // encoding.
+    return;
+  }
+  VideoCodecType codec_type;
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    time_of_last_frame_activity_ms_ = TickTime::MillisecondTimestamp();
+    if (EncoderPaused()) {
+      TraceFrameDropStart();
+      return;
+    }
+    TraceFrameDropEnd();
+    codec_type = encoder_config_.codecType;
+  }
+
+  TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame.render_time_ms(),
+                          "Encode");
+  const VideoFrame* frame_to_send = &video_frame;
+  // TODO(wuchengli): support texture frames.
+  if (video_frame.native_handle() == NULL) {
+    // Pass frame via preprocessor.
+    frame_to_send = vp_->PreprocessFrame(video_frame);
+    if (!frame_to_send) {
+      // Drop this frame, or there was an error processing it.
+      return;
+    }
+  }
+
+  // If we haven't resampled the frame and we have a FrameCallback, we need to
+  // make a deep copy of |video_frame|.
+  VideoFrame copied_frame;
+  if (pre_encode_callback_) {
+    copied_frame.CopyFrame(*frame_to_send);
+    pre_encode_callback_->FrameCallback(&copied_frame);
+    frame_to_send = &copied_frame;
+  }
+
+  if (codec_type == webrtc::kVideoCodecVP8) {
+    webrtc::CodecSpecificInfo codec_specific_info;
+    codec_specific_info.codecType = webrtc::kVideoCodecVP8;
+    {
+      CriticalSectionScoped cs(data_cs_.get());
+      codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
+          has_received_rpsi_;
+      codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
+          has_received_sli_;
+      codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
+          picture_id_rpsi_;
+      codec_specific_info.codecSpecific.VP8.pictureIdSLI  =
+          picture_id_sli_;
+      has_received_sli_ = false;
+      has_received_rpsi_ = false;
+    }
+
+    vcm_->AddVideoFrame(*frame_to_send, vp_->GetContentMetrics(),
+                        &codec_specific_info);
+    return;
+  }
+  vcm_->AddVideoFrame(*frame_to_send);
+}
+
+int ViEEncoder::SendKeyFrame() {
+  return vcm_->IntraFrameRequest(0);
+}
+
+uint32_t ViEEncoder::LastObservedBitrateBps() const {
+  CriticalSectionScoped cs(data_cs_.get());
+  return last_observed_bitrate_bps_;
+}
+
+int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
+  if (vcm_->Bitrate(bitrate) != 0)
+    return -1;
+  return 0;
+}
+
+void ViEEncoder::SetProtectionMethod(bool nack, bool fec) {
+  // Set Video Protection for VCM.
+  VCMVideoProtection protection_mode;
+  if (fec) {
+    protection_mode =
+        nack ? webrtc::kProtectionNackFEC : kProtectionFEC;
+  } else {
+    protection_mode = nack ? kProtectionNack : kProtectionNone;
+  }
+  vcm_->SetVideoProtection(protection_mode, true);
+}
+
+void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    target_delay_ms_ = target_delay_ms;
+  }
+  if (target_delay_ms > 0) {
+    // Disable external frame-droppers.
+    vcm_->EnableFrameDropper(false);
+    vp_->EnableTemporalDecimation(false);
+  } else {
+    // Real-time mode - enable frame droppers.
+    vp_->EnableTemporalDecimation(true);
+    vcm_->EnableFrameDropper(true);
+  }
+}
+
+void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) {
+  if (stats_proxy_)
+    stats_proxy_->OnSetRates(bitrate_bps, framerate);
+}
+
+int32_t ViEEncoder::SendData(
+    const uint8_t payload_type,
+    const EncodedImage& encoded_image,
+    const webrtc::RTPFragmentationHeader& fragmentation_header,
+    const RTPVideoHeader* rtp_video_hdr) {
+  RTC_DCHECK(send_payload_router_ != NULL);
+
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    time_of_last_frame_activity_ms_ = TickTime::MillisecondTimestamp();
+  }
+
+  if (stats_proxy_ != NULL)
+    stats_proxy_->OnSendEncodedImage(encoded_image, rtp_video_hdr);
+
+  return send_payload_router_->RoutePayload(
+             encoded_image._frameType, payload_type, encoded_image._timeStamp,
+             encoded_image.capture_time_ms_, encoded_image._buffer,
+             encoded_image._length, &fragmentation_header, rtp_video_hdr)
+             ? 0
+             : -1;
+}
+
+void ViEEncoder::OnEncoderImplementationName(
+    const char* implementation_name) {
+  if (stats_proxy_)
+    stats_proxy_->OnEncoderImplementationName(implementation_name);
+}
+
+int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
+                                   const uint32_t frame_rate) {
+  if (stats_proxy_)
+    stats_proxy_->OnOutgoingRate(frame_rate, bit_rate);
+  return 0;
+}
+
+void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
+                               uint8_t picture_id) {
+  CriticalSectionScoped cs(data_cs_.get());
+  picture_id_sli_ = picture_id;
+  has_received_sli_ = true;
+}
+
+void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
+                                uint64_t picture_id) {
+  CriticalSectionScoped cs(data_cs_.get());
+  picture_id_rpsi_ = picture_id;
+  has_received_rpsi_ = true;
+}
+
+void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
+  // Key frame request from remote side, signal to VCM.
+  TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
+
+  int idx = 0;
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    auto stream_it = ssrc_streams_.find(ssrc);
+    if (stream_it == ssrc_streams_.end()) {
+      LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
+                        << ssrc_streams_.size();
+      return;
+    }
+    std::map<unsigned int, int64_t>::iterator time_it =
+        time_last_intra_request_ms_.find(ssrc);
+    if (time_it == time_last_intra_request_ms_.end()) {
+      time_last_intra_request_ms_[ssrc] = 0;
+    }
+
+    int64_t now = TickTime::MillisecondTimestamp();
+    if (time_last_intra_request_ms_[ssrc] + kMinKeyFrameRequestIntervalMs
+        > now) {
+      return;
+    }
+    time_last_intra_request_ms_[ssrc] = now;
+    idx = stream_it->second;
+  }
+  // Release the critsect before triggering key frame.
+  vcm_->IntraFrameRequest(idx);
+}
+
+void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
+  CriticalSectionScoped cs(data_cs_.get());
+  std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
+  if (it == ssrc_streams_.end()) {
+    return;
+  }
+
+  ssrc_streams_[new_ssrc] = it->second;
+  ssrc_streams_.erase(it);
+
+  std::map<unsigned int, int64_t>::iterator time_it =
+      time_last_intra_request_ms_.find(old_ssrc);
+  int64_t last_intra_request_ms = 0;
+  if (time_it != time_last_intra_request_ms_.end()) {
+    last_intra_request_ms = time_it->second;
+    time_last_intra_request_ms_.erase(time_it);
+  }
+  time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
+}
+
+void ViEEncoder::SetSsrcs(const std::vector<uint32_t>& ssrcs) {
+  CriticalSectionScoped cs(data_cs_.get());
+  ssrc_streams_.clear();
+  time_last_intra_request_ms_.clear();
+  int idx = 0;
+  for (uint32_t ssrc : ssrcs) {
+    ssrc_streams_[ssrc] = idx++;
+  }
+}
+
+void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
+  assert(min_transmit_bitrate_kbps >= 0);
+  CriticalSectionScoped crit(data_cs_.get());
+  min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
+}
+
+// Called from ViEBitrateObserver.
+void ViEEncoder::OnNetworkChanged(uint32_t bitrate_bps,
+                                  uint8_t fraction_lost,
+                                  int64_t round_trip_time_ms) {
+  LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
+                  << " packet loss " << static_cast<int>(fraction_lost)
+                  << " rtt " << round_trip_time_ms;
+  RTC_DCHECK(send_payload_router_ != NULL);
+  vcm_->SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
+  bool video_is_suspended = vcm_->VideoSuspended();
+  bool video_suspension_changed;
+  VideoCodec send_codec;
+  uint32_t first_ssrc;
+  {
+    CriticalSectionScoped cs(data_cs_.get());
+    last_observed_bitrate_bps_ = bitrate_bps;
+    video_suspension_changed = video_suspended_ != video_is_suspended;
+    video_suspended_ = video_is_suspended;
+    send_codec = encoder_config_;
+    first_ssrc = ssrc_streams_.begin()->first;
+  }
+
+  SimulcastStream* stream_configs = send_codec.simulcastStream;
+  // Allocate the bandwidth between the streams.
+  std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
+      bitrate_bps, stream_configs, send_codec.numberOfSimulcastStreams);
+  send_payload_router_->SetTargetSendBitrates(stream_bitrates);
+
+  if (!video_suspension_changed)
+    return;
+  // Video suspend-state changed, inform codec observer.
+  LOG(LS_INFO) << "Video suspend state changed " << video_is_suspended
+               << " for ssrc " << first_ssrc;
+  if (stats_proxy_)
+    stats_proxy_->OnSuspendChange(video_is_suspended);
+}
+
+void ViEEncoder::SuspendBelowMinBitrate() {
+  vcm_->SuspendBelowMinBitrate();
+  bitrate_allocator_->EnforceMinBitrate(false);
+}
+
+void ViEEncoder::RegisterPostEncodeImageCallback(
+      EncodedImageCallback* post_encode_callback) {
+  vcm_->RegisterPostEncodeImageCallback(post_encode_callback);
+}
+
+void ViEEncoder::onLoadStateChanged(CPULoadState state) {
+  vcm_->SetCPULoadState(state);
+}
+
+QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessing* vpm)
+    : vp_(vpm) {
+}
+
+QMVideoSettingsCallback::~QMVideoSettingsCallback() {
+}
+
+int32_t QMVideoSettingsCallback::SetVideoQMSettings(
+    const uint32_t frame_rate,
+    const uint32_t width,
+    const uint32_t height) {
+  return vp_->SetTargetResolution(width, height, frame_rate);
+}
+
+void QMVideoSettingsCallback::SetTargetFramerate(int frame_rate) {
+  vp_->SetTargetFramerate(frame_rate);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.h b/media/webrtc/trunk/webrtc/video/vie_encoder.h
similarity index 52%
rename from media/webrtc/trunk/webrtc/video_engine/vie_encoder.h
rename to media/webrtc/trunk/webrtc/video/vie_encoder.h
index 44bfa4968f..47a94ecca8 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.h
+++ b/media/webrtc/trunk/webrtc/video/vie_encoder.h
@@ -8,29 +8,28 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
+#ifndef WEBRTC_VIDEO_VIE_ENCODER_H_
+#define WEBRTC_VIDEO_VIE_ENCODER_H_
 
-#include <list>
 #include <map>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/typedefs.h"
 #include "webrtc/frame_callback.h"
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_frame_provider_base.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video/video_capture_input.h"
 
 namespace webrtc {
 
+class BitrateAllocator;
+class BitrateObserver;
 class Config;
 class CriticalSectionWrapper;
 class EncodedImageCallback;
@@ -41,28 +40,23 @@ class QMVideoSettingsCallback;
 class SendStatisticsProxy;
 class ViEBitrateObserver;
 class ViEEffectFilter;
-class ViEEncoderObserver;
 class VideoCodingModule;
-class ViECPULoadStateObserver;
 
-class ViEEncoder
-    : public RtcpIntraFrameObserver,
-      public VideoEncoderRateObserver,
-      public VCMPacketizationCallback,
-      public VCMSendStatisticsCallback,
-      public ViEFrameCallback {
+class ViEEncoder : public RtcpIntraFrameObserver,
+                   public VideoEncoderRateObserver,
+                   public VCMPacketizationCallback,
+                   public VCMSendStatisticsCallback,
+                   public CPULoadStateObserver,
+                   public VideoCaptureCallback {
  public:
   friend class ViEBitrateObserver;
-  friend class ViECPULoadStateObserver;
 
-  ViEEncoder(int32_t channel_id,
-             uint32_t number_of_cores,
-             const Config& config,
-             ProcessThread& module_process_thread,
+  ViEEncoder(uint32_t number_of_cores,
+             ProcessThread* module_process_thread,
+             SendStatisticsProxy* stats_proxy,
+             I420FrameCallback* pre_encode_callback,
              PacedSender* pacer,
-             BitrateAllocator* bitrate_allocator,
-             BitrateController* bitrate_controller,
-             bool disable_default_encoder);
+             BitrateAllocator* bitrate_allocator);
   ~ViEEncoder();
 
   bool Init();
@@ -72,7 +66,7 @@ class ViEEncoder
   // Ideally this would be done in Init, but the dependencies between ViEEncoder
   // and ViEChannel makes it really hard to do in a good way.
   void StartThreadsAndSetSharedMembers(
-      scoped_refptr<PayloadRouter> send_payload_router,
+      rtc::scoped_refptr<PayloadRouter> send_payload_router,
       VCMProtectionCallback* vcm_protection_callback);
 
   // This function must be called before the corresponding ViEChannel is
@@ -84,47 +78,32 @@ class ViEEncoder
   // Returns the id of the owning channel.
   int Owner() const;
 
+  // CPULoadStateObserver interface
+  void onLoadStateChanged(CPULoadState state) override;
+
   // Drops incoming packets before they get to the encoder.
   void Pause();
   void Restart();
 
   // Codec settings.
-  uint8_t NumberOfCodecs();
-  int32_t GetCodec(uint8_t list_index, VideoCodec* video_codec);
   int32_t RegisterExternalEncoder(VideoEncoder* encoder,
                                   uint8_t pl_type,
                                   bool internal_source);
   int32_t DeRegisterExternalEncoder(uint8_t pl_type);
   int32_t SetEncoder(const VideoCodec& video_codec);
-  int32_t GetEncoder(VideoCodec* video_codec);
 
-  int32_t GetCodecConfigParameters(
-    unsigned char config_parameters[kConfigParameterSize],
-    unsigned char& config_parameters_size);
-
-  // Scale or crop/pad image.
-  int32_t ScaleInputImage(bool enable);
-
-  // Implementing ViEFrameCallback.
-  void DeliverFrame(int id,
-                    I420VideoFrame* video_frame,
-                    const std::vector<uint32_t>& csrcs) override;
-  void DelayChanged(int id, int frame_delay) override;
-  int GetPreferedFrameSettings(int* width,
-                               int* height,
-                               int* frame_rate) override;
-
-  void ProviderDestroyed(int id) override { return; }
+  // Implementing VideoCaptureCallback.
+  void DeliverFrame(VideoFrame video_frame) override;
 
   int32_t SendKeyFrame();
-  int32_t SendCodecStatistics(uint32_t* num_key_frames,
-                              uint32_t* num_delta_frames);
 
   uint32_t LastObservedBitrateBps() const;
   int CodecTargetBitrate(uint32_t* bitrate) const;
-  // Loss protection.
-  int32_t UpdateProtectionMethod(bool nack, bool fec);
-  bool nack_enabled() const { return nack_enabled_; }
+  // Loss protection. Must be called before SetEncoder() to have max packet size
+  // updated according to protection.
+  // TODO(pbos): Set protection method on construction or extract vcm_ outside
+  // this class and set it on construction there.
+  void SetProtectionMethod(bool nack, bool fec);
 
   // Buffering mode.
   void SetSenderBufferingMode(int target_delay_ms);
@@ -137,13 +116,12 @@ class ViEEncoder
                    const EncodedImage& encoded_image,
                    const RTPFragmentationHeader& fragmentation_header,
                    const RTPVideoHeader* rtp_video_hdr) override;
+  void OnEncoderImplementationName(const char* implementation_name) override;
 
   // Implements VideoSendStatisticsCallback.
   int32_t SendStatistics(const uint32_t bit_rate,
                          const uint32_t frame_rate) override;
 
-  int32_t RegisterCodecObserver(ViEEncoderObserver* observer);
-
   // Implements RtcpIntraFrameObserver.
   void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
   void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) override;
@@ -151,39 +129,20 @@ class ViEEncoder
   void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) override;
 
   // Sets SSRCs for all streams.
-  bool SetSsrcs(const std::list<unsigned int>& ssrcs);
+  void SetSsrcs(const std::vector<uint32_t>& ssrcs);
 
   void SetMinTransmitBitrate(int min_transmit_bitrate_kbps);
 
-  // Effect filter.
-  int32_t RegisterEffectFilter(ViEEffectFilter* effect_filter);
-
-  // Load Management
-  void SetLoadManager(CPULoadStateCallbackInvoker* load_manager);
-
-  // Enables recording of debugging information.
-  int StartDebugRecording(const char* fileNameUTF8);
-
-  // Disables recording of debugging information.
-  int StopDebugRecording();
-
   // Lets the sender suspend video when the rate drops below
   // |threshold_bps|, and turns back on when the rate goes back up above
   // |threshold_bps| + |window_bps|.
   void SuspendBelowMinBitrate();
 
   // New-style callbacks, used by VideoSendStream.
-  void RegisterPreEncodeCallback(I420FrameCallback* pre_encode_callback);
-  void DeRegisterPreEncodeCallback();
   void RegisterPostEncodeImageCallback(
         EncodedImageCallback* post_encode_callback);
-  void DeRegisterPostEncodeImageCallback();
 
-  void RegisterSendStatisticsProxy(SendStatisticsProxy* send_statistics_proxy);
-
-  int channel_id() const { return channel_id_; }
-
-  int GetPaddingNeededBps(int bitrate_bps) const;
+  int GetPaddingNeededBps() const;
 
  protected:
   // Called by BitrateObserver.
@@ -191,38 +150,31 @@ class ViEEncoder
                         uint8_t fraction_lost,
                         int64_t round_trip_time_ms);
 
-  // Called by CPULoadStateObserver
-  void onLoadStateChanged(CPULoadState load_state);
-
  private:
   bool EncoderPaused() const EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
   void TraceFrameDropStart() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
   void TraceFrameDropEnd() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
 
-  void UpdateHistograms();
-
-  const int channel_id_;
   const uint32_t number_of_cores_;
-  const bool disable_default_encoder_;
 
-  VideoCodingModule& vcm_;
-  VideoProcessingModule& vpm_;
-  scoped_refptr<PayloadRouter> send_payload_router_;
-  VCMProtectionCallback* vcm_protection_callback_;
+  const rtc::scoped_ptr<VideoProcessing> vp_;
+  const rtc::scoped_ptr<QMVideoSettingsCallback> qm_callback_;
+  const rtc::scoped_ptr<VideoCodingModule> vcm_;
+  rtc::scoped_refptr<PayloadRouter> send_payload_router_;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> callback_cs_;
   rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
   rtc::scoped_ptr<BitrateObserver> bitrate_observer_;
-  rtc::scoped_ptr<CPULoadStateObserver> loadstate_observer_;
 
+  SendStatisticsProxy* const stats_proxy_;
+  I420FrameCallback* const pre_encode_callback_;
   PacedSender* const pacer_;
   BitrateAllocator* const bitrate_allocator_;
-  BitrateController* const bitrate_controller_;
-  // Owned by PeerConnection, not ViEEncoder
-  CPULoadStateCallbackInvoker* load_manager_;
 
-  int64_t time_of_last_incoming_frame_ms_ GUARDED_BY(data_cs_);
-  bool send_padding_ GUARDED_BY(data_cs_);
+  // The time we last received an input frame or encoded frame. This is used to
+  // track when video is stopped long enough that we also want to stop sending
+  // padding.
+  int64_t time_of_last_frame_activity_ms_ GUARDED_BY(data_cs_);
+  VideoCodec encoder_config_ GUARDED_BY(data_cs_);
   int min_transmit_bitrate_kbps_ GUARDED_BY(data_cs_);
   uint32_t last_observed_bitrate_bps_ GUARDED_BY(data_cs_);
   int target_delay_ms_ GUARDED_BY(data_cs_);
@@ -232,28 +184,17 @@ class ViEEncoder
   std::map<unsigned int, int64_t> time_last_intra_request_ms_
       GUARDED_BY(data_cs_);
 
-  bool fec_enabled_;
-  bool nack_enabled_;
-
-  ViEEncoderObserver* codec_observer_ GUARDED_BY(callback_cs_);
-  ViEEffectFilter* effect_filter_ GUARDED_BY(callback_cs_);
-  ProcessThread& module_process_thread_;
+  ProcessThread* module_process_thread_;
 
   bool has_received_sli_ GUARDED_BY(data_cs_);
   uint8_t picture_id_sli_ GUARDED_BY(data_cs_);
   bool has_received_rpsi_ GUARDED_BY(data_cs_);
   uint64_t picture_id_rpsi_ GUARDED_BY(data_cs_);
-  std::map<unsigned int, int> ssrc_streams_ GUARDED_BY(data_cs_);
+  std::map<uint32_t, int> ssrc_streams_ GUARDED_BY(data_cs_);
 
-  // Quality modes callback
-  QMVideoSettingsCallback* qm_callback_;
   bool video_suspended_ GUARDED_BY(data_cs_);
-  I420FrameCallback* pre_encode_callback_ GUARDED_BY(callback_cs_);
-  const int64_t start_ms_;
-
-  SendStatisticsProxy* send_statistics_proxy_ GUARDED_BY(callback_cs_);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
+#endif  // WEBRTC_VIDEO_VIE_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc b/media/webrtc/trunk/webrtc/video/vie_receiver.cc
similarity index 77%
rename from media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc
rename to media/webrtc/trunk/webrtc/video/vie_receiver.cc
index 11ef499fa4..43c746c317 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc
+++ b/media/webrtc/trunk/webrtc/video/vie_receiver.cc
@@ -8,34 +8,32 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/vie_receiver.h"
+#include "webrtc/video/vie_receiver.h"
 
 #include <vector>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/rtp_dump.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
 static const int kPacketLogIntervalMs = 10000;
 
-ViEReceiver::ViEReceiver(const int32_t channel_id,
-                         VideoCodingModule* module_vcm,
+ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
                          RemoteBitrateEstimator* remote_bitrate_estimator,
                          RtpFeedback* rtp_feedback)
     : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
@@ -44,8 +42,7 @@ ViEReceiver::ViEReceiver(const int32_t channel_id,
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
       rtp_receiver_(
-          RtpReceiver::CreateVideoReceiver(channel_id,
-                                           clock_,
+          RtpReceiver::CreateVideoReceiver(clock_,
                                            this,
                                            rtp_feedback,
                                            rtp_payload_registry_.get())),
@@ -55,36 +52,32 @@ ViEReceiver::ViEReceiver(const int32_t channel_id,
       vcm_(module_vcm),
       remote_bitrate_estimator_(remote_bitrate_estimator),
       ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
-      rtp_dump_(NULL),
       receiving_(false),
-      receiving_rtcp_(false),
       restored_packet_in_use_(false),
       receiving_ast_enabled_(false),
       receiving_cvo_enabled_(false),
-      receiving_rid_enabled_(false),
+      receiving_tsn_enabled_(false),
+			receiving_rid_enabled_(false),
       last_packet_log_ms_(-1) {
   assert(remote_bitrate_estimator);
 }
 
 ViEReceiver::~ViEReceiver() {
   UpdateHistograms();
-  if (rtp_dump_) {
-    rtp_dump_->Stop();
-    RtpDump::DestroyRtpDump(rtp_dump_);
-    rtp_dump_ = NULL;
-  }
 }
 
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
-    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedFecPacketsInPercent",
-        counter.num_fec_packets * 100 / counter.num_packets);
+    RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+        "WebRTC.Video.ReceivedFecPacketsInPercent",
+        static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
-    RTC_HISTOGRAM_PERCENTAGE(
+    RTC_HISTOGRAM_PERCENTAGE_SPARSE(
         "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
-            counter.num_recovered_packets * 100 / counter.num_fec_packets);
+        static_cast<int>(counter.num_recovered_packets * 100 /
+                         counter.num_fec_packets));
   }
 }
 
@@ -121,8 +114,14 @@ void ViEReceiver::SetNackStatus(bool enable,
   rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
 }
 
-void ViEReceiver::SetRtxPayloadType(int payload_type) {
-  rtp_payload_registry_->SetRtxPayloadType(payload_type);
+void ViEReceiver::SetRtxPayloadType(int payload_type,
+                                    int associated_payload_type) {
+  rtp_payload_registry_->SetRtxPayloadType(payload_type,
+                                           associated_payload_type);
+}
+
+void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
+  rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(val);
 }
 
 void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
@@ -157,16 +156,14 @@ RtpReceiver* ViEReceiver::GetRtpReceiver() const {
   return rtp_receiver_.get();
 }
 
-void ViEReceiver::RegisterSimulcastRtpRtcpModules(
-    const std::list<RtpRtcp*>& rtp_modules) {
+void ViEReceiver::RegisterRtpRtcpModules(
+    const std::vector<RtpRtcp*>& rtp_modules) {
   CriticalSectionScoped cs(receive_cs_.get());
-  rtp_rtcp_simulcast_.clear();
-
-  if (!rtp_modules.empty()) {
-    rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(),
-                               rtp_modules.begin(),
-                               rtp_modules.end());
-  }
+  // Only change the "simulcast" modules, the base module can be accessed
+  // without a lock whereas the simulcast modules require locking as they can be
+  // changed in runtime.
+  rtp_rtcp_simulcast_ =
+      std::vector<RtpRtcp*>(rtp_modules.begin() + 1, rtp_modules.end());
 }
 
 bool ViEReceiver::SetReceiveTimestampOffsetStatus(bool enable, int id) {
@@ -211,6 +208,22 @@ bool ViEReceiver::SetReceiveVideoRotationStatus(bool enable, int id) {
   }
 }
 
+bool ViEReceiver::SetReceiveTransportSequenceNumber(bool enable, int id) {
+  if (enable) {
+    if (rtp_header_parser_->RegisterRtpHeaderExtension(
+            kRtpExtensionTransportSequenceNumber, id)) {
+      receiving_tsn_enabled_ = true;
+      return true;
+    } else {
+      return false;
+    }
+  } else {
+    receiving_tsn_enabled_ = false;
+    return rtp_header_parser_->DeregisterRtpHeaderExtension(
+        kRtpExtensionTransportSequenceNumber);
+  }
+}
+
 bool ViEReceiver::SetReceiveRIDStatus(bool enable, int id) {
   if (enable) {
     if (rtp_header_parser_->RegisterRtpHeaderExtension(
@@ -227,6 +240,7 @@ bool ViEReceiver::SetReceiveRIDStatus(bool enable, int id) {
   }
 }
 
+
 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
                                    size_t rtp_packet_length,
                                    const PacketTime& packet_time) {
@@ -266,17 +280,6 @@ bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
   return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
 }
 
-void ViEReceiver::ReceivedBWEPacket(
-    int64_t arrival_time_ms, size_t payload_size, const RTPHeader& header) {
-  // Only forward if the incoming packet *and* the channel are both configured
-  // to receive absolute sender time. RTP time stamps may have different rates
-  // for audio and video and shouldn't be mixed.
-  if (header.extension.hasAbsoluteSendTime && receiving_ast_enabled_) {
-    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
-                                              header);
-  }
-}
-
 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
                                  size_t rtp_packet_length,
                                  const PacketTime& packet_time) {
@@ -285,9 +288,6 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
     if (!receiving_) {
       return -1;
     }
-    if (rtp_dump_) {
-      rtp_dump_->DumpPacket(rtp_packet, rtp_packet_length);
-    }
   }
 
   RTPHeader header;
@@ -316,15 +316,15 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
         ss << ", toffset: " << header.extension.transmissionTimeOffset;
       if (header.extension.hasAbsoluteSendTime)
         ss << ", abs send time: " << header.extension.absoluteSendTime;
-      if (header.extension.hasRID)
-        ss << ", rid: " << header.extension.rid;
+			if (header.extension.hasRID)
+				ss << ", rid: " << header.extension.rid;
       LOG(LS_INFO) << ss.str();
       last_packet_log_ms_ = now_ms;
     }
   }
 
-  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms,
-                                            payload_length, header);
+  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
+                                            header, true);
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
@@ -390,15 +390,14 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
       LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
       return false;
     }
-    uint8_t* restored_packet_ptr = restored_packet_;
     if (!rtp_payload_registry_->RestoreOriginalPacket(
-        &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
-        header)) {
+            restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
+            header)) {
       LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header";
       return false;
     }
     restored_packet_in_use_ = true;
-    bool ret = OnRecoveredPacket(restored_packet_ptr, packet_length);
+    bool ret = OnRecoveredPacket(restored_packet_, packet_length);
     restored_packet_in_use_ = false;
     return ret;
   }
@@ -436,19 +435,12 @@ int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
                                   size_t rtcp_packet_length) {
   {
     CriticalSectionScoped cs(receive_cs_.get());
-    if (!receiving_rtcp_) {
+    if (!receiving_) {
       return -1;
     }
 
-    if (rtp_dump_) {
-      rtp_dump_->DumpPacket(rtcp_packet, rtcp_packet_length);
-    }
-
-    std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin();
-    while (it != rtp_rtcp_simulcast_.end()) {
-      RtpRtcp* rtp_rtcp = *it++;
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_)
       rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
-    }
   }
   assert(rtp_rtcp_);  // Should be set by owner at construction time.
   int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
@@ -485,49 +477,6 @@ void ViEReceiver::StopReceive() {
   receiving_ = false;
 }
 
-void ViEReceiver::StartRTCPReceive() {
-  CriticalSectionScoped cs(receive_cs_.get());
-  receiving_rtcp_ = true;
-}
-
-void ViEReceiver::StopRTCPReceive() {
-  CriticalSectionScoped cs(receive_cs_.get());
-  receiving_rtcp_ = false;
-}
-
-int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) {
-  CriticalSectionScoped cs(receive_cs_.get());
-  if (rtp_dump_) {
-    // Restart it if it already exists and is started
-    rtp_dump_->Stop();
-  } else {
-    rtp_dump_ = RtpDump::CreateRtpDump();
-    if (rtp_dump_ == NULL) {
-      return -1;
-    }
-  }
-  if (rtp_dump_->Start(file_nameUTF8) != 0) {
-    RtpDump::DestroyRtpDump(rtp_dump_);
-    rtp_dump_ = NULL;
-    return -1;
-  }
-  return 0;
-}
-
-int ViEReceiver::StopRTPDump() {
-  CriticalSectionScoped cs(receive_cs_.get());
-  if (rtp_dump_) {
-    if (rtp_dump_->IsActive()) {
-      rtp_dump_->Stop();
-    }
-    RtpDump::DestroyRtpDump(rtp_dump_);
-    rtp_dump_ = NULL;
-  } else {
-    return -1;
-  }
-  return 0;
-}
-
 ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
   return rtp_receive_statistics_.get();
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.h b/media/webrtc/trunk/webrtc/video/vie_receiver.h
similarity index 76%
rename from media/webrtc/trunk/webrtc/video_engine/vie_receiver.h
rename to media/webrtc/trunk/webrtc/video/vie_receiver.h
index 4b06476790..2d5fc50b33 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.h
+++ b/media/webrtc/trunk/webrtc/video/vie_receiver.h
@@ -8,18 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
+#ifndef WEBRTC_VIDEO_VIE_RECEIVER_H_
+#define WEBRTC_VIDEO_VIE_RECEIVER_H_
 
 #include <list>
+#include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/vie_defines.h"
 
 namespace webrtc {
 
@@ -28,7 +27,6 @@ class FecReceiver;
 class RemoteNtpTimeEstimator;
 class ReceiveStatistics;
 class RemoteBitrateEstimator;
-class RtpDump;
 class RtpHeaderParser;
 class RTPPayloadRegistry;
 class RtpReceiver;
@@ -38,7 +36,7 @@ struct ReceiveBandwidthEstimatorStats;
 
 class ViEReceiver : public RtpData {
  public:
-  ViEReceiver(const int32_t channel_id, VideoCodingModule* module_vcm,
+  ViEReceiver(VideoCodingModule* module_vcm,
               RemoteBitrateEstimator* remote_bitrate_estimator,
               RtpFeedback* rtp_feedback);
   ~ViEReceiver();
@@ -47,7 +45,12 @@ class ViEReceiver : public RtpData {
   bool RegisterPayload(const VideoCodec& video_codec);
 
   void SetNackStatus(bool enable, int max_nack_reordering_threshold);
-  void SetRtxPayloadType(int payload_type);
+  void SetRtxPayloadType(int payload_type, int associated_payload_type);
+  // If set to true, the RTX payload type mapping supplied in
+  // |SetRtxPayloadType| will be used when restoring RTX packets. Without it,
+  // RTX packets will always be restored to the last non-RTX packet payload type
+  // received.
+  void SetUseRtxPayloadMappingOnRestore(bool val);
   void SetRtxSsrc(uint32_t ssrc);
   bool GetRtxSsrc(uint32_t* ssrc) const;
 
@@ -55,28 +58,23 @@ class ViEReceiver : public RtpData {
 
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
-  void GetRID(char rid[256]) const;
+	void GetRID(char rid[256]) const; //MOZ addition RtpSenderId (RID)
 
   void SetRtpRtcpModule(RtpRtcp* module);
 
   RtpReceiver* GetRtpReceiver() const;
 
-  void RegisterSimulcastRtpRtcpModules(const std::list<RtpRtcp*>& rtp_modules);
+  void RegisterRtpRtcpModules(const std::vector<RtpRtcp*>& rtp_modules);
 
   bool SetReceiveTimestampOffsetStatus(bool enable, int id);
   bool SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
   bool SetReceiveVideoRotationStatus(bool enable, int id);
-  bool SetReceiveRIDStatus(bool enable, int id);
+  bool SetReceiveTransportSequenceNumber(bool enable, int id);
+	bool SetReceiveRIDStatus(bool enable, int id); //MOZ addition RtpSenderId (RID)
 
   void StartReceive();
   void StopReceive();
 
-  void StartRTCPReceive();
-  void StopRTCPReceive();
-
-  int StartRTPDump(const char file_nameUTF8[1024]);
-  int StopRTPDump();
-
   // Receives packets from external transport.
   int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length,
                         const PacketTime& packet_time);
@@ -90,8 +88,6 @@ class ViEReceiver : public RtpData {
 
   ReceiveStatistics* GetReceiveStatistics() const;
 
-  void ReceivedBWEPacket(int64_t arrival_time_ms, size_t payload_size,
-                         const RTPHeader& header);
  private:
   int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length,
                       const PacketTime& packet_time);
@@ -115,26 +111,25 @@ class ViEReceiver : public RtpData {
   rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
   rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
-  rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
+  const rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
   rtc::scoped_ptr<FecReceiver> fec_receiver_;
   RtpRtcp* rtp_rtcp_;
-  std::list<RtpRtcp*> rtp_rtcp_simulcast_;
+  std::vector<RtpRtcp*> rtp_rtcp_simulcast_;
   VideoCodingModule* vcm_;
   RemoteBitrateEstimator* remote_bitrate_estimator_;
 
   rtc::scoped_ptr<RemoteNtpTimeEstimator> ntp_estimator_;
 
-  RtpDump* rtp_dump_;
   bool receiving_;
-  bool receiving_rtcp_;
-  uint8_t restored_packet_[kViEMaxMtu];
+  uint8_t restored_packet_[IP_PACKET_SIZE];
   bool restored_packet_in_use_;
   bool receiving_ast_enabled_;
   bool receiving_cvo_enabled_;
-  bool receiving_rid_enabled_;
+  bool receiving_tsn_enabled_;
+	bool receiving_rid_enabled_;
   int64_t last_packet_log_ms_;
 };
 
-}  // namespace webrt
+}  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
+#endif  // WEBRTC_VIDEO_VIE_RECEIVER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_remb.cc b/media/webrtc/trunk/webrtc/video/vie_remb.cc
similarity index 84%
rename from media/webrtc/trunk/webrtc/video_engine/vie_remb.cc
rename to media/webrtc/trunk/webrtc/video/vie_remb.cc
index 2cf794c5d4..95c2f1e130 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_remb.cc
+++ b/media/webrtc/trunk/webrtc/video/vie_remb.cc
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/vie_remb.h"
+#include "webrtc/video/vie_remb.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
@@ -27,9 +27,10 @@ const int kRembSendIntervalMs = 200;
 // % threshold for if we should send a new REMB asap.
 const unsigned int kSendThresholdPercent = 97;
 
-VieRemb::VieRemb()
-    : list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      last_remb_time_(TickTime::MillisecondTimestamp()),
+VieRemb::VieRemb(Clock* clock)
+    : clock_(clock),
+      list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      last_remb_time_(clock_->TimeInMilliseconds()),
       last_send_bitrate_(0),
       bitrate_(0) {}
 
@@ -105,13 +106,13 @@ void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
     if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
       // The new bitrate estimate is less than kSendThresholdPercent % of the
       // last report. Send a REMB asap.
-      last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervalMs;
+      last_remb_time_ = clock_->TimeInMilliseconds() - kRembSendIntervalMs;
     }
   }
   bitrate_ = bitrate;
 
   // Calculate total receive bitrate estimate.
-  int64_t now = TickTime::MillisecondTimestamp();
+  int64_t now = clock_->TimeInMilliseconds();
 
   if (now - last_remb_time_ < kRembSendIntervalMs) {
     list_crit_->Leave();
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_remb.h b/media/webrtc/trunk/webrtc/video/vie_remb.h
similarity index 88%
rename from media/webrtc/trunk/webrtc/video_engine/vie_remb.h
rename to media/webrtc/trunk/webrtc/video/vie_remb.h
index 9f38259ca8..2a3d916d6c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_remb.h
+++ b/media/webrtc/trunk/webrtc/video/vie_remb.h
@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
+#ifndef WEBRTC_VIDEO_VIE_REMB_H_
+#define WEBRTC_VIDEO_VIE_REMB_H_
 
 #include <list>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
@@ -28,7 +28,7 @@ class RtpRtcp;
 
 class VieRemb : public RemoteBitrateObserver {
  public:
-  VieRemb();
+  explicit VieRemb(Clock* clock);
   ~VieRemb();
 
   // Called to add a receive channel to include in the REMB packet.
@@ -57,6 +57,7 @@ class VieRemb : public RemoteBitrateObserver {
  private:
   typedef std::list<RtpRtcp*> RtpModules;
 
+  Clock* const clock_;
   rtc::scoped_ptr<CriticalSectionWrapper> list_crit_;
 
   // The last time a REMB was sent.
@@ -75,4 +76,4 @@ class VieRemb : public RemoteBitrateObserver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
+#endif  // WEBRTC_VIDEO_VIE_REMB_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc b/media/webrtc/trunk/webrtc/video/vie_remb_unittest.cc
similarity index 92%
rename from media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc
rename to media/webrtc/trunk/webrtc/video/vie_remb_unittest.cc
index 3949d4482c..a44d593b22 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/vie_remb_unittest.cc
@@ -11,17 +11,16 @@
 
 // This file includes unit tests for ViERemb.
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
 #include <vector>
 
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/vie_remb.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video/vie_remb.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
@@ -31,12 +30,15 @@ using ::testing::Return;
 namespace webrtc {
 
 class ViERembTest : public ::testing::Test {
+ public:
+  ViERembTest() : fake_clock_(12345) {}
+
  protected:
   virtual void SetUp() {
-    TickTime::UseFakeClock(12345);
     process_thread_.reset(new NiceMock<MockProcessThread>);
-    vie_remb_.reset(new VieRemb());
+    vie_remb_.reset(new VieRemb(&fake_clock_));
   }
+  SimulatedClock fake_clock_;
   rtc::scoped_ptr<MockProcessThread> process_thread_;
   rtc::scoped_ptr<VieRemb> vie_remb_;
 };
@@ -52,7 +54,7 @@ TEST_F(ViERembTest, OneModuleTestForSendingRemb) {
 
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
@@ -77,7 +79,7 @@ TEST_F(ViERembTest, LowerEstimateToSendRemb) {
 
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
         .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
@@ -106,7 +108,7 @@ TEST_F(ViERembTest, VerifyIncreasingAndDecreasing) {
   // Call OnReceiveBitrateChanged twice to get a first estimate.
   EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0], ssrcs))
         .Times(1);
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
 
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1] + 100);
@@ -134,7 +136,7 @@ TEST_F(ViERembTest, NoRembForIncreasedBitrate) {
 
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
@@ -168,7 +170,7 @@ TEST_F(ViERembTest, ChangeSendRtpModule) {
 
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
@@ -204,7 +206,7 @@ TEST_F(ViERembTest, OnlyOneRembForDoubleProcess) {
   vie_remb_->AddRembSender(&rtp);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_CALL(rtp, SetREMBData(_, _))
         .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
@@ -236,7 +238,7 @@ TEST_F(ViERembTest, NoSendingRtpModule) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Call OnReceiveBitrateChanged twice to get a first estimate.
-  TickTime::AdvanceFakeClock(1000);
+  fake_clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_CALL(rtp, SetREMBData(_, _))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc b/media/webrtc/trunk/webrtc/video/vie_sync_module.cc
similarity index 74%
rename from media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc
rename to media/webrtc/trunk/webrtc/video/vie_sync_module.cc
index cf592927a8..bf8ef5f45c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc
+++ b/media/webrtc/trunk/webrtc/video/vie_sync_module.cc
@@ -8,16 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/vie_sync_module.h"
+#include "webrtc/video/vie_sync_module.h"
 
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
-#include "webrtc/video_engine/stream_synchronization.h"
-#include "webrtc/video_engine/vie_channel.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/video/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
@@ -49,11 +48,9 @@ int UpdateMeasurements(StreamSynchronization::Measurements* stream,
   return 0;
 }
 
-ViESyncModule::ViESyncModule(VideoCodingModule* vcm,
-                             ViEChannel* vie_channel)
+ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
     : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
       vcm_(vcm),
-      vie_channel_(vie_channel),
       video_receiver_(NULL),
       video_rtp_rtcp_(NULL),
       voe_channel_id_(-1),
@@ -70,11 +67,19 @@ int ViESyncModule::ConfigureSync(int voe_channel_id,
                                  RtpRtcp* video_rtcp_module,
                                  RtpReceiver* video_receiver) {
   CriticalSectionScoped cs(data_cs_.get());
+  // Prevent expensive no-ops.
+  if (voe_channel_id_ == voe_channel_id &&
+      voe_sync_interface_ == voe_sync_interface &&
+      video_receiver_ == video_receiver &&
+      video_rtp_rtcp_ == video_rtcp_module) {
+    return 0;
+  }
   voe_channel_id_ = voe_channel_id;
   voe_sync_interface_ = voe_sync_interface;
   video_receiver_ = video_receiver;
   video_rtp_rtcp_ = video_rtcp_module;
-  sync_.reset(new StreamSynchronization(voe_channel_id, vie_channel_->Id()));
+  sync_.reset(
+      new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
 
   if (!voe_sync_interface) {
     voe_channel_id_ = -1;
@@ -110,11 +115,11 @@ int32_t ViESyncModule::Process() {
 
   int audio_jitter_buffer_delay_ms = 0;
   int playout_buffer_delay_ms = 0;
-  int avsync_offset_ms = 0;
+  int avsync_delay_ms = 0;
   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
                                             &audio_jitter_buffer_delay_ms,
                                             &playout_buffer_delay_ms,
-                                            &avsync_offset_ms) != 0) {
+                                            &avsync_delay_ms) != 0) {
     return 0;
   }
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
@@ -140,15 +145,11 @@ int32_t ViESyncModule::Process() {
   }
 
   int relative_delay_ms;
-  int result;
   // Calculate how much later or earlier the audio stream is compared to video.
-
-  result = sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
-                                       &relative_delay_ms);
-  if (!result) {
+  if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,
+                                   &relative_delay_ms)) {
     return 0;
   }
-  voe_sync_interface_->SetCurrentSyncOffset(voe_channel_id_, relative_delay_ms);
 
   TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
   TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
@@ -172,18 +173,4 @@ int32_t ViESyncModule::Process() {
   return 0;
 }
 
-int ViESyncModule::SetTargetBufferingDelay(int target_delay_ms) {
-  CriticalSectionScoped cs(data_cs_.get());
-  if (!voe_sync_interface_) {
-    LOG(LS_ERROR) << "voe_sync_interface_ NULL, can't set playout delay.";
-    return -1;
-  }
-  sync_->SetTargetBufferingDelay(target_delay_ms);
-  // Setting initial playout delay to voice engine (video engine is updated via
-  // the VCM interface).
-  voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_,
-                                              target_delay_ms);
-  return 0;
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h b/media/webrtc/trunk/webrtc/video/vie_sync_module.h
similarity index 73%
rename from media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h
rename to media/webrtc/trunk/webrtc/video/vie_sync_module.h
index a75d716c08..a9ad20a103 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h
+++ b/media/webrtc/trunk/webrtc/video/vie_sync_module.h
@@ -11,13 +11,13 @@
 // ViESyncModule is responsible for synchronization audio and video for a given
 // VoE and ViE channel couple.
 
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
+#ifndef WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
+#define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/stream_synchronization.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
@@ -30,8 +30,7 @@ class VoEVideoSync;
 
 class ViESyncModule : public Module {
  public:
-  ViESyncModule(VideoCodingModule* vcm,
-                ViEChannel* vie_channel);
+  explicit ViESyncModule(VideoCodingModule* vcm);
   ~ViESyncModule();
 
   int ConfigureSync(int voe_channel_id,
@@ -41,17 +40,13 @@ class ViESyncModule : public Module {
 
   int VoiceChannel();
 
-  // Set target delay for buffering mode (0 = real-time mode).
-  int SetTargetBufferingDelay(int target_delay_ms);
-
   // Implements Module.
   int64_t TimeUntilNextProcess() override;
   int32_t Process() override;
 
  private:
   rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
-  VideoCodingModule* vcm_;
-  ViEChannel* vie_channel_;
+  VideoCodingModule* const vcm_;
   RtpReceiver* video_receiver_;
   RtpRtcp* video_rtp_rtcp_;
   int voe_channel_id_;
@@ -64,4 +59,4 @@ class ViESyncModule : public Module {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
+#endif  // WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
diff --git a/media/webrtc/trunk/webrtc/video/webrtc_video.gypi b/media/webrtc/trunk/webrtc/video/webrtc_video.gypi
index 4de970abde..db8d5c7e89 100644
--- a/media/webrtc/trunk/webrtc/video/webrtc_video.gypi
+++ b/media/webrtc/trunk/webrtc/video/webrtc_video.gypi
@@ -8,22 +8,58 @@
 {
   'variables': {
     'webrtc_video_dependencies': [
-      '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
+      '<(webrtc_root)/base/base.gyp:rtc_base_approved',
+      '<(webrtc_root)/common.gyp:webrtc_common',
+      '<(webrtc_root)/common_video/common_video.gyp:common_video',
+      '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
+      '<(webrtc_root)/modules/modules.gyp:paced_sender',
+      '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+      '<(webrtc_root)/modules/modules.gyp:video_capture_module',
+      '<(webrtc_root)/modules/modules.gyp:video_processing',
+      '<(webrtc_root)/modules/modules.gyp:video_render_module',
+      '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
+      '<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
+      '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+      '<(webrtc_root)/webrtc.gyp:rtc_event_log',
     ],
     'webrtc_video_sources': [
-      'video/call.cc',
+      'video/call_stats.cc',
+      'video/call_stats.h',
       'video/encoded_frame_callback_adapter.cc',
       'video/encoded_frame_callback_adapter.h',
-      'video/send_statistics_proxy.cc',
-      'video/send_statistics_proxy.h',
+      'video/encoder_state_feedback.cc',
+      'video/encoder_state_feedback.h',
+      'video/overuse_frame_detector.cc',
+      'video/overuse_frame_detector.h',
+      'video/payload_router.cc',
+      'video/payload_router.h',
       'video/receive_statistics_proxy.cc',
       'video/receive_statistics_proxy.h',
-      'video/transport_adapter.cc',
-      'video/transport_adapter.h',
+      'video/report_block_stats.cc',
+      'video/report_block_stats.h',
+      'video/send_statistics_proxy.cc',
+      'video/send_statistics_proxy.h',
+      'video/stream_synchronization.cc',
+      'video/stream_synchronization.h',
+      'video/video_capture_input.cc',
+      'video/video_capture_input.h',
+      'video/video_decoder.cc',
+      'video/video_encoder.cc',
       'video/video_receive_stream.cc',
       'video/video_receive_stream.h',
       'video/video_send_stream.cc',
       'video/video_send_stream.h',
+      'video/vie_channel.cc',
+      'video/vie_channel.h',
+      'video/vie_encoder.cc',
+      'video/vie_encoder.h',
+      'video/vie_receiver.cc',
+      'video/vie_receiver.h',
+      'video/vie_remb.cc',
+      'video/vie_remb.h',
+      'video/vie_sync_module.cc',
+      'video/vie_sync_module.h',
     ],
   },
 }
diff --git a/media/webrtc/trunk/webrtc/video_decoder.h b/media/webrtc/trunk/webrtc/video_decoder.h
index 941c0ac197..3cd94e8270 100644
--- a/media/webrtc/trunk/webrtc/video_decoder.h
+++ b/media/webrtc/trunk/webrtc/video_decoder.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VIDEO_DECODER_H_
 #define WEBRTC_VIDEO_DECODER_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
@@ -28,7 +29,17 @@ class DecodedImageCallback {
  public:
   virtual ~DecodedImageCallback() {}
 
-  virtual int32_t Decoded(I420VideoFrame& decodedImage) = 0;
+  virtual int32_t Decoded(VideoFrame& decodedImage) = 0;
+  // Provides an alternative interface that allows the decoder to specify the
+  // decode time excluding waiting time for any previous pending frame to
+  // return. This is necessary for breaking positive feedback in the delay
+  // estimation when the decoder has a single output buffer.
+  // TODO(perkj): Remove default implementation when chromium has been updated.
+  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) {
+    // The default implementation ignores custom decode time value.
+    return Decoded(decodedImage);
+  }
+
   virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) {
     return -1;
   }
@@ -39,22 +50,24 @@ class DecodedImageCallback {
 class VideoDecoder {
  public:
   enum DecoderType {
+    kH264,
     kVp8,
-    kVp9
+    kVp9,
+    kUnsupportedCodec,
   };
 
   static VideoDecoder* Create(DecoderType codec_type);
 
   virtual ~VideoDecoder() {}
 
-  virtual int32_t InitDecode(const VideoCodec* codecSettings,
-                             int32_t numberOfCores) = 0;
+  virtual int32_t InitDecode(const VideoCodec* codec_settings,
+                             int32_t number_of_cores) = 0;
 
-  virtual int32_t Decode(const EncodedImage& inputImage,
-                         bool missingFrames,
+  virtual int32_t Decode(const EncodedImage& input_image,
+                         bool missing_frames,
                          const RTPFragmentationHeader* fragmentation,
-                         const CodecSpecificInfo* codecSpecificInfo = NULL,
-                         int64_t renderTimeMs = -1) = 0;
+                         const CodecSpecificInfo* codec_specific_info = NULL,
+                         int64_t render_time_ms = -1) = 0;
 
   virtual int32_t RegisterDecodeCompleteCallback(
       DecodedImageCallback* callback) = 0;
@@ -62,12 +75,51 @@ class VideoDecoder {
   virtual int32_t Release() = 0;
   virtual int32_t Reset() = 0;
 
-  virtual int32_t SetCodecConfigParameters(const uint8_t* /*buffer*/,
-                                           int32_t /*size*/) {
-    return -1;
-  }
+  // Returns true if the decoder prefer to decode frames late.
+  // That is, it can not decode infinite number of frames before the decoded
+  // frame is consumed.
+  virtual bool PrefersLateDecoding() const { return true; }
 
-  virtual VideoDecoder* Copy() { return NULL; }
+  virtual const char* ImplementationName() const { return "unknown"; }
+};
+
+// Class used to wrap external VideoDecoders to provide a fallback option on
+// software decoding when a hardware decoder fails to decode a stream due to
+// hardware restrictions, such as max resolution.
+class VideoDecoderSoftwareFallbackWrapper : public webrtc::VideoDecoder {
+ public:
+  VideoDecoderSoftwareFallbackWrapper(VideoCodecType codec_type,
+                                      VideoDecoder* decoder);
+
+  int32_t InitDecode(const VideoCodec* codec_settings,
+                     int32_t number_of_cores) override;
+
+  int32_t Decode(const EncodedImage& input_image,
+                 bool missing_frames,
+                 const RTPFragmentationHeader* fragmentation,
+                 const CodecSpecificInfo* codec_specific_info,
+                 int64_t render_time_ms) override;
+
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+
+  int32_t Release() override;
+  int32_t Reset() override;
+  bool PrefersLateDecoding() const override;
+
+  const char* ImplementationName() const override;
+
+ private:
+  bool InitFallbackDecoder();
+
+  const DecoderType decoder_type_;
+  VideoDecoder* const decoder_;
+
+  VideoCodec codec_settings_;
+  int32_t number_of_cores_;
+  std::string fallback_implementation_name_;
+  rtc::scoped_ptr<VideoDecoder> fallback_decoder_;
+  DecodedImageCallback* callback_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_encoder.h b/media/webrtc/trunk/webrtc/video_encoder.h
index c933a33469..ffb91fcbab 100644
--- a/media/webrtc/trunk/webrtc/video_encoder.h
+++ b/media/webrtc/trunk/webrtc/video_encoder.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VIDEO_ENCODER_H_
 #define WEBRTC_VIDEO_ENCODER_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
@@ -37,11 +38,14 @@ class EncodedImageCallback {
 class VideoEncoder {
  public:
   enum EncoderType {
+    kH264,
     kVp8,
     kVp9,
+    kUnsupportedCodec,
   };
 
-  static VideoEncoder* Create(EncoderType codec_type);
+  static VideoEncoder* Create(EncoderType codec_type,
+                              bool enable_simulcast = false);
 
   static VideoCodecVP8 GetDefaultVp8Settings();
   static VideoCodecVP9 GetDefaultVp9Settings();
@@ -94,9 +98,9 @@ class VideoEncoder {
   //                                  WEBRTC_VIDEO_CODEC_MEMORY
   //                                  WEBRTC_VIDEO_CODEC_ERROR
   //                                  WEBRTC_VIDEO_CODEC_TIMEOUT
-  virtual int32_t Encode(const I420VideoFrame& frame,
+  virtual int32_t Encode(const VideoFrame& frame,
                          const CodecSpecificInfo* codec_specific_info,
-                         const std::vector<VideoFrameType>* frame_types) = 0;
+                         const std::vector<FrameType>* frame_types) = 0;
 
   // Inform the encoder of the new packet loss rate and the round-trip time of
   // the network.
@@ -119,10 +123,64 @@ class VideoEncoder {
   virtual int32_t SetRates(uint32_t bitrate, uint32_t framerate) = 0;
 
   virtual int32_t SetPeriodicKeyFrames(bool enable) { return -1; }
-  virtual int32_t CodecConfigParameters(uint8_t* /*buffer*/, int32_t /*size*/) {
-    return -1;
-  }
+  virtual void OnDroppedFrame() {}
+  virtual int GetTargetFramerate() { return -1; }
+  virtual bool SupportsNativeHandle() const { return false; }
+  virtual const char* ImplementationName() const { return "unknown"; }
 };
 
+// Class used to wrap external VideoEncoders to provide a fallback option on
+// software encoding when a hardware encoder fails to encode a stream due to
+// hardware restrictions, such as max resolution.
+class VideoEncoderSoftwareFallbackWrapper : public VideoEncoder {
+ public:
+  VideoEncoderSoftwareFallbackWrapper(VideoCodecType codec_type,
+                                      webrtc::VideoEncoder* encoder);
+
+  int32_t InitEncode(const VideoCodec* codec_settings,
+                     int32_t number_of_cores,
+                     size_t max_payload_size) override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* callback) override;
+
+  int32_t Release() override;
+  int32_t Encode(const VideoFrame& frame,
+                 const CodecSpecificInfo* codec_specific_info,
+                 const std::vector<FrameType>* frame_types) override;
+  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
+
+  int32_t SetRates(uint32_t bitrate, uint32_t framerate) override;
+  void OnDroppedFrame() override;
+  int GetTargetFramerate() override;
+  bool SupportsNativeHandle() const override;
+  const char* ImplementationName() const override;
+
+ private:
+  bool InitFallbackEncoder();
+
+  // Settings used in the last InitEncode call and used if a dynamic fallback to
+  // software is required.
+  VideoCodec codec_settings_;
+  int32_t number_of_cores_;
+  size_t max_payload_size_;
+
+  // The last bitrate/framerate set, and a flag for noting they are set.
+  bool rates_set_;
+  uint32_t bitrate_;
+  uint32_t framerate_;
+
+  // The last channel parameters set, and a flag for noting they are set.
+  bool channel_parameters_set_;
+  uint32_t packet_loss_;
+  int64_t rtt_;
+
+  const EncoderType encoder_type_;
+  webrtc::VideoEncoder* const encoder_;
+
+  rtc::scoped_ptr<webrtc::VideoEncoder> fallback_encoder_;
+  std::string fallback_implementation_name_;
+  EncodedImageCallback* callback_;
+};
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_ENCODER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/BUILD.gn b/media/webrtc/trunk/webrtc/video_engine/BUILD.gn
deleted file mode 100644
index 2e496417a5..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/BUILD.gn
+++ /dev/null
@@ -1,124 +0,0 @@
-# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-import("../build/webrtc.gni")
-
-source_set("video_engine") {
-  deps = [ ":video_engine_core" ]
-}
-
-source_set("video_engine_core") {
-  sources = [
-    "include/vie_base.h",
-    "include/vie_capture.h",
-    "include/vie_codec.h",
-    "include/vie_errors.h",
-    "include/vie_external_codec.h",
-    "include/vie_image_process.h",
-    "include/vie_network.h",
-    "include/vie_render.h",
-    "include/vie_rtp_rtcp.h",
-    "call_stats.cc",
-    "call_stats.h",
-    "encoder_state_feedback.cc",
-    "encoder_state_feedback.h",
-    "overuse_frame_detector.cc",
-    "overuse_frame_detector.h",
-    "payload_router.cc",
-    "payload_router.h",
-    "report_block_stats.cc",
-    "report_block_stats.h",
-    "stream_synchronization.cc",
-    "stream_synchronization.h",
-    "vie_base_impl.cc",
-    "vie_base_impl.h",
-    "vie_capture_impl.cc",
-    "vie_capture_impl.h",
-    "vie_capturer.cc",
-    "vie_capturer.h",
-    "vie_channel.cc",
-    "vie_channel_group.cc",
-    "vie_channel_group.h",
-    "vie_channel.h",
-    "vie_channel_manager.cc",
-    "vie_channel_manager.h",
-    "vie_codec_impl.cc",
-    "vie_codec_impl.h",
-    "vie_defines.h",
-    "vie_encoder.cc",
-    "vie_encoder.h",
-    "vie_external_codec_impl.cc",
-    "vie_external_codec_impl.h",
-    "vie_file_image.cc",
-    "vie_file_image.h",
-    "vie_frame_provider_base.cc",
-    "vie_frame_provider_base.h",
-    "vie_image_process_impl.cc",
-    "vie_image_process_impl.h",
-    "vie_impl.cc",
-    "vie_impl.h",
-    "vie_input_manager.cc",
-    "vie_input_manager.h",
-    "vie_manager_base.cc",
-    "vie_manager_base.h",
-    "vie_network_impl.cc",
-    "vie_network_impl.h",
-    "vie_receiver.cc",
-    "vie_receiver.h",
-    "vie_ref_count.cc",
-    "vie_ref_count.h",
-    "vie_remb.cc",
-    "vie_remb.h",
-    "vie_renderer.cc",
-    "vie_renderer.h",
-    "vie_render_impl.cc",
-    "vie_render_impl.h",
-    "vie_render_manager.cc",
-    "vie_render_manager.h",
-    "vie_rtp_rtcp_impl.cc",
-    "vie_rtp_rtcp_impl.h",
-    "vie_sender.cc",
-    "vie_sender.h",
-    "vie_shared_data.cc",
-    "vie_shared_data.h",
-    "vie_sync_module.cc",
-    "vie_sync_module.h",
-  ]
-
-  configs += [ "..:common_config" ]
-  public_configs = [ "..:common_inherited_config" ]
-
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-
-  if (is_win) {
-    cflags = [
-      # TODO(jschuh): Bug 1348: fix size_t to int truncations.
-      "/wd4267",  # size_t to int truncation.
-      # Bug 261.
-      "/wd4373",  # legacy warning for ignoring const / volatile in signatures.
-    ]
-  }
-
-  deps = [
-    "..:webrtc_common",
-    "../common_video",
-    "../modules/bitrate_controller",
-    "../modules/rtp_rtcp",
-    "../modules/utility",
-    "../modules/video_capture:video_capture_module",
-    "../modules/video_coding",
-    "../modules/video_processing",
-    "../modules/video_render:video_render_module",
-    "../voice_engine",
-    "../system_wrappers",
-  ]
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h b/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h
index 0e24cafe72..8df4cf336d 100644
--- a/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h
@@ -1,7 +1,7 @@
 #ifndef WEBRTC_MODULES_BROWSER_CAPTURE_MAIN_SOURCE_BROWSER_CAPTURE_IMPL_H_
 #define WEBRTC_MODULES_BROWSER_CAPTURE_MAIN_SOURCE_BROWSER_CAPTURE_IMPL_H_
 
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 
 using namespace webrtc::videocapturemodule;
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc
index dcc9a47e9c..8039e4a52f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc
@@ -15,20 +15,21 @@
 
 #include "webrtc/common_video/libyuv/include/scaler.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ref_count.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/video_engine/desktop_capture_impl.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_device_info.h"
 #include "webrtc/modules/desktop_capture/app_capturer.h"
 #include "webrtc/modules/desktop_capture/window_capturer.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
+#include "webrtc/modules/video_capture/video_capture.h"
 
 namespace webrtc {
 
@@ -244,6 +245,28 @@ int32_t WindowDeviceInfoImpl::Init() {
   return 0;
 }
 
+int32_t DesktopCaptureImpl::AddRef() const {
+  return ++mRefCount;
+}
+int32_t DesktopCaptureImpl::Release() const {
+  assert(mRefCount > 0);
+  auto count = --mRefCount;
+  if (!count) {
+    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideoCapture, -1,
+                 "DesktopCapture self deleting (desktopCapture=0x%p)", this);
+
+    // Clear any pointers before starting destruction. Otherwise worker-
+    // threads will still have pointers to a partially destructed object.
+    // Example: AudioDeviceBuffer::RequestPlayoutData() can access a
+    // partially deconstructed |_ptrCbAudioTransport| during destruction
+    // if we don't call Terminate here.
+    //-> NG TODO Terminate();
+    delete this;
+    return count;
+  }
+  return mRefCount;
+}
+
 int32_t WindowDeviceInfoImpl::Refresh() {
   desktop_device_info_->Refresh();
   return 0;
@@ -335,21 +358,21 @@ int32_t WindowDeviceInfoImpl::GetOrientation(const char* deviceUniqueIdUTF8,
 
 VideoCaptureModule::DeviceInfo* DesktopCaptureImpl::CreateDeviceInfo(const int32_t id,
                                                                      const CaptureDeviceType type) {
-  if (type == Application) {
+  if (type == CaptureDeviceType::Application) {
     AppDeviceInfoImpl * pAppDeviceInfoImpl = new AppDeviceInfoImpl(id);
     if (!pAppDeviceInfoImpl || pAppDeviceInfoImpl->Init()) {
       delete pAppDeviceInfoImpl;
       pAppDeviceInfoImpl = NULL;
     }
     return pAppDeviceInfoImpl;
-  } else if (type == Screen) {
+  } else if (type == CaptureDeviceType::Screen) {
     ScreenDeviceInfoImpl * pScreenDeviceInfoImpl = new ScreenDeviceInfoImpl(id);
     if (!pScreenDeviceInfoImpl || pScreenDeviceInfoImpl->Init()) {
       delete pScreenDeviceInfoImpl;
       pScreenDeviceInfoImpl = NULL;
     }
     return pScreenDeviceInfoImpl;
-  } else if (type == Window) {
+  } else if (type == CaptureDeviceType::Window) {
     WindowDeviceInfoImpl * pWindowDeviceInfoImpl = new WindowDeviceInfoImpl(id);
     if (!pWindowDeviceInfoImpl || pWindowDeviceInfoImpl->Init()) {
       delete pWindowDeviceInfoImpl;
@@ -370,7 +393,7 @@ int32_t DesktopCaptureImpl::Init(const char* uniqueId,
   // Leave desktop effects enabled during WebRTC captures.
   options.set_disable_effects(false);
 
-  if (type == Application) {
+  if (type == CaptureDeviceType::Application) {
     AppCapturer *pAppCapturer = AppCapturer::Create(options);
     if (!pAppCapturer) {
       return -1;
@@ -381,7 +404,7 @@ int32_t DesktopCaptureImpl::Init(const char* uniqueId,
 
     MouseCursorMonitor *pMouseCursorMonitor = MouseCursorMonitor::CreateForScreen(options, webrtc::kFullDesktopScreenId);
     desktop_capturer_cursor_composer_.reset(new DesktopAndCursorComposer(pAppCapturer, pMouseCursorMonitor));
-  } else if (type == Screen) {
+  } else if (type == CaptureDeviceType::Screen) {
     ScreenCapturer *pScreenCapturer = ScreenCapturer::Create(options);
     if (!pScreenCapturer) {
       return -1;
@@ -393,7 +416,7 @@ int32_t DesktopCaptureImpl::Init(const char* uniqueId,
 
     MouseCursorMonitor *pMouseCursorMonitor = MouseCursorMonitor::CreateForScreen(options, screenid);
     desktop_capturer_cursor_composer_.reset(new DesktopAndCursorComposer(pScreenCapturer, pMouseCursorMonitor));
-  } else if (type == Window) {
+  } else if (type == CaptureDeviceType::Window) {
     WindowCapturer *pWindowCapturer = WindowCapturer::Create();
     if (!pWindowCapturer) {
       return -1;
@@ -480,13 +503,15 @@ DesktopCaptureImpl::DesktopCaptureImpl(const int32_t id)
                          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() -
                          TickTime::MillisecondTimestamp()),
   time_event_(EventWrapper::Create()),
+  mRefCount(0),
 #if defined(_WIN32)
-  capturer_thread_(ThreadWrapper::CreateUIThread(Run, this, "ScreenCaptureThread")),
+  capturer_thread_(new rtc::PlatformUIThread(Run, this, "ScreenCaptureThread")),
 #else
-  capturer_thread_(ThreadWrapper::CreateThread(Run, this, "ScreenCaptureThread")),
+  capturer_thread_(new rtc::PlatformThread(Run, this, "ScreenCaptureThread")),
 #endif
   started_(false) {
-  capturer_thread_->SetPriority(kHighPriority);
+  //-> TODO @@NG why is this crashing (seen on Linux)
+  //-> capturer_thread_->SetPriority(rtc::kHighPriority);
   _requestedCapability.width = kDefaultWidth;
   _requestedCapability.height = kDefaultHeight;
   _requestedCapability.maxFPS = 30;
@@ -548,7 +573,7 @@ int32_t DesktopCaptureImpl::CaptureDelay()
   return _setCaptureDelay;
 }
 
-int32_t DesktopCaptureImpl::DeliverCapturedFrame(I420VideoFrame& captureFrame,
+int32_t DesktopCaptureImpl::DeliverCapturedFrame(webrtc::VideoFrame& captureFrame,
                                                  int64_t capture_time) {
   UpdateFrameCount();  // frame count used for local frame rate callback.
 
@@ -679,7 +704,7 @@ int32_t DesktopCaptureImpl::IncomingFrame(uint8_t* videoFrame,
       DeliverCapturedFrame(_captureFrame, captureTime);
     } else {
 
-      I420VideoFrame scaledFrame;
+      webrtc::VideoFrame scaledFrame;
       ret = scaledFrame.CreateEmptyFrame(dst_width,
                                          dst_height,
                                          stride_y,
@@ -858,4 +883,3 @@ void DesktopCaptureImpl::process() {
 }
 
 }  // namespace webrtc
-
diff --git a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h
index b3ebe607ec..1da7125c7f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h
@@ -17,18 +17,17 @@
 
 #include <string>
 
-#include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/video_frame.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/modules/desktop_capture/shared_memory.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor_shape.h"
 #include "webrtc/modules/desktop_capture/desktop_device_info.h"
 #include "webrtc/modules/desktop_capture/desktop_and_cursor_composer.h"
-#include "webrtc/video_engine/include/vie_capture.h"
 
 using namespace webrtc::videocapturemodule;
 
@@ -171,7 +170,9 @@ public:
   static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(const int32_t id, const CaptureDeviceType type);
 
   int32_t Init(const char* uniqueId, const CaptureDeviceType type);
-
+  //RefCounting for RefCountedModule
+  virtual int32_t AddRef() const override;
+  virtual int32_t Release() const override;
   //Call backs
   virtual void RegisterCaptureDataCallback(VideoCaptureDataCallback& dataCallback) override;
   virtual void DeRegisterCaptureDataCallback() override;
@@ -182,7 +183,7 @@ public:
   virtual int32_t CaptureDelay() override;
   virtual int32_t SetCaptureRotation(VideoRotation rotation) override;
   virtual bool SetApplyRotation(bool enable) override;
-  virtual bool GetApplyRotation() { return true; }
+  virtual bool GetApplyRotation() override { return true; }
 
   virtual void EnableFrameRateCallback(const bool enable) override;
   virtual void EnableNoPictureAlarm(const bool enable) override;
@@ -214,7 +215,7 @@ public:
 protected:
   DesktopCaptureImpl(const int32_t id);
   virtual ~DesktopCaptureImpl();
-  int32_t DeliverCapturedFrame(I420VideoFrame& captureFrame,
+  int32_t DeliverCapturedFrame(webrtc::VideoFrame& captureFrame,
                                int64_t capture_time);
 
   static const uint32_t kMaxDesktopCaptureCpuUsage = 50; // maximum CPU usage in %
@@ -245,7 +246,7 @@ private:
   TickTime _incomingFrameTimes[kFrameRateCountHistorySize];// timestamp for local captured frames
   VideoRotation _rotateFrame; //Set if the frame should be rotated by the capture module.
 
-  I420VideoFrame _captureFrame;
+  webrtc::VideoFrame _captureFrame;
 
   // Used to make sure incoming timestamp is increasing for every frame.
   int64_t last_capture_time_;
@@ -263,7 +264,12 @@ public:
 private:
   rtc::scoped_ptr<DesktopAndCursorComposer> desktop_capturer_cursor_composer_;
   rtc::scoped_ptr<EventWrapper> time_event_;
-  rtc::scoped_ptr<ThreadWrapper> capturer_thread_;
+#if defined(_WIN32)
+  rtc::scoped_ptr<rtc::PlatformUIThread> capturer_thread_;
+#else
+  rtc::scoped_ptr<rtc::PlatformThread> capturer_thread_;
+#endif
+  mutable uint32_t mRefCount;
   bool started_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h
deleted file mode 100644
index 578622411b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-//  - Creating and deleting VideoEngine instances.
-//  - Creating and deleting channels.
-//  - Connect a video channel with a corresponding voice channel for audio/video
-//    synchronization.
-//  - Start and stop sending and receiving.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
-
-#include "webrtc/common_types.h"
-
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
-#include <jni.h>
-#endif
-
-namespace webrtc {
-
-class Config;
-class VoiceEngine;
-class ReceiveStatisticsProxy;
-class SendStatisticsProxy;
-
-// CpuOveruseObserver is called when a system overuse is detected and
-// VideoEngine cannot keep up the encoding frequency.
-class CpuOveruseObserver {
- public:
-  // Called as soon as an overuse is detected.
-  virtual void OveruseDetected() = 0;
-  // Called periodically when the system is not overused any longer.
-  virtual void NormalUsage() = 0;
-
- protected:
-  virtual ~CpuOveruseObserver() {}
-};
-
-struct CpuOveruseOptions {
-  CpuOveruseOptions()
-      : enable_capture_jitter_method(false),
-        low_capture_jitter_threshold_ms(20.0f),
-        high_capture_jitter_threshold_ms(30.0f),
-        enable_encode_usage_method(true),
-        low_encode_usage_threshold_percent(55),
-        high_encode_usage_threshold_percent(85),
-        low_encode_time_rsd_threshold(-1),
-        high_encode_time_rsd_threshold(-1),
-        enable_extended_processing_usage(true),
-        frame_timeout_interval_ms(1500),
-        min_frame_samples(120),
-        min_process_count(3),
-        high_threshold_consecutive_count(2) {}
-
-  // Method based on inter-arrival jitter of captured frames.
-  bool enable_capture_jitter_method;
-  float low_capture_jitter_threshold_ms;  // Threshold for triggering underuse.
-  float high_capture_jitter_threshold_ms; // Threshold for triggering overuse.
-  // Method based on encode time of frames.
-  bool enable_encode_usage_method;
-  int low_encode_usage_threshold_percent;  // Threshold for triggering underuse.
-  int high_encode_usage_threshold_percent; // Threshold for triggering overuse.
-  // TODO(asapersson): Remove options, not used.
-  int low_encode_time_rsd_threshold;   // Additional threshold for triggering
-                                       // underuse (used in addition to
-                                       // threshold above if configured).
-  int high_encode_time_rsd_threshold;  // Additional threshold for triggering
-                                       // overuse (used in addition to
-                                       // threshold above if configured).
-  bool enable_extended_processing_usage;  // Include a larger time span (in
-                                          // addition to encode time) for
-                                          // measuring the processing time of a
-                                          // frame.
-  // General settings.
-  int frame_timeout_interval_ms;  // The maximum allowed interval between two
-                                  // frames before resetting estimations.
-  int min_frame_samples;  // The minimum number of frames required.
-  int min_process_count;  // The number of initial process times required before
-                          // triggering an overuse/underuse.
-  int high_threshold_consecutive_count; // The number of consecutive checks
-                                        // above the high threshold before
-                                        // triggering an overuse.
-
-  bool Equals(const CpuOveruseOptions& o) const {
-    return enable_capture_jitter_method == o.enable_capture_jitter_method &&
-        low_capture_jitter_threshold_ms == o.low_capture_jitter_threshold_ms &&
-        high_capture_jitter_threshold_ms ==
-        o.high_capture_jitter_threshold_ms &&
-        enable_encode_usage_method == o.enable_encode_usage_method &&
-        low_encode_usage_threshold_percent ==
-        o.low_encode_usage_threshold_percent &&
-        high_encode_usage_threshold_percent ==
-        o.high_encode_usage_threshold_percent &&
-        low_encode_time_rsd_threshold == o.low_encode_time_rsd_threshold &&
-        high_encode_time_rsd_threshold == o.high_encode_time_rsd_threshold &&
-        enable_extended_processing_usage ==
-        o.enable_extended_processing_usage &&
-        frame_timeout_interval_ms == o.frame_timeout_interval_ms &&
-        min_frame_samples == o.min_frame_samples &&
-        min_process_count == o.min_process_count &&
-        high_threshold_consecutive_count == o.high_threshold_consecutive_count;
-  }
-};
-
-struct CpuOveruseMetrics {
-  CpuOveruseMetrics()
-      : capture_jitter_ms(-1),
-        avg_encode_time_ms(-1),
-        encode_usage_percent(-1),
-        capture_queue_delay_ms_per_s(-1) {}
-
-  int capture_jitter_ms;  // The current estimated jitter in ms based on
-                          // incoming captured frames.
-  int avg_encode_time_ms;   // The average encode time in ms.
-  int encode_usage_percent; // The average encode time divided by the average
-                            // time difference between incoming captured frames.
-  int capture_queue_delay_ms_per_s;  // The current time delay between an
-                                     // incoming captured frame until the frame
-                                     // is being processed. The delay is
-                                     // expressed in ms delay per second.
-};
-
-class CpuOveruseMetricsObserver {
- public:
-  virtual ~CpuOveruseMetricsObserver() {}
-  virtual void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) = 0;
-};
-
-class WEBRTC_DLLEXPORT VideoEngine {
- public:
-  // Creates a VideoEngine object, which can then be used to acquire sub‐APIs.
-  static VideoEngine* Create();
-  static VideoEngine* Create(const Config& config);
-
-  // Deletes a VideoEngine instance.
-  static bool Delete(VideoEngine*& video_engine);
-
-  // Specifies the amount and type of trace information, which will be created
-  // by the VideoEngine.
-  static int SetTraceFilter(const unsigned int filter);
-
-  // Sets the name of the trace file and enables non‐encrypted trace messages.
-  static int SetTraceFile(const char* file_nameUTF8,
-                          const bool add_file_counter = false);
-
-  // Installs the TraceCallback implementation to ensure that the VideoEngine
-  // user receives callbacks for generated trace messages.
-  static int SetTraceCallback(TraceCallback* callback);
-
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
-  // Android specific.
-  static int SetAndroidObjects(JavaVM* java_vm);
-#endif
-
- protected:
-  VideoEngine() {}
-  virtual ~VideoEngine() {}
-};
-
-class WEBRTC_DLLEXPORT ViEBase {
- public:
-  // Factory for the ViEBase sub‐API and increases an internal reference
-  // counter if successful. Returns NULL if the API is not supported or if
-  // construction fails.
-  static ViEBase* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViEBase sub-API and decreases an internal reference counter.
-  // Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  // Initiates all common parts of the VideoEngine.
-  virtual int Init() = 0;
-
-  // Connects a VideoEngine instance to a VoiceEngine instance for audio video
-  // synchronization.
-  virtual int SetVoiceEngine(VoiceEngine* voice_engine) = 0;
-
-  // Creates a new channel.
-  virtual int CreateChannel(int& video_channel) = 0;
-
-  // Creates a new channel grouped together with |original_channel|. The channel
-  // can both send and receive video. It is assumed the channel is sending
-  // and/or receiving video to the same end-point.
-  // Note: |CreateReceiveChannel| will give better performance and network
-  // properties for receive only channels.
-  virtual int CreateChannel(int& video_channel,
-                            int original_channel) = 0;
-
-  virtual int CreateChannelWithoutDefaultEncoder(int& video_channel,
-                                                 int original_channel) = 0;
-
-  // Creates a new channel grouped together with |original_channel|. The channel
-  // can only receive video and it is assumed the remote end-point is the same
-  // as for |original_channel|.
-  virtual int CreateReceiveChannel(int& video_channel,
-                                   int original_channel) = 0;
-
-  // Deletes an existing channel and releases the utilized resources.
-  virtual int DeleteChannel(const int video_channel) = 0;
-
-  // Registers an observer to be called when an overuse is detected, see
-  // 'CpuOveruseObserver' for details.
-  // NOTE: This is still very experimental functionality.
-  virtual int RegisterCpuOveruseObserver(int channel,
-                                         CpuOveruseObserver* observer) = 0;
-
-  // Sets options for cpu overuse detector.
-  virtual int SetCpuOveruseOptions(int channel,
-                                   const CpuOveruseOptions& options) = 0;
-
-  // Gets cpu overuse measures.
-  virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics) = 0;
-  virtual void RegisterCpuOveruseMetricsObserver(
-      int channel,
-      CpuOveruseMetricsObserver* observer) = 0;
-
-  // Registers a callback which is called when send-side delay statistics has
-  // been updated.
-  // TODO(holmer): Remove the default implementation when fakevideoengine.h has
-  // been updated.
-  virtual void RegisterSendSideDelayObserver(
-      int channel, SendSideDelayObserver* observer) {}
-
-  // Changing the current state of the host CPU. Encoding engines
-  // can adapt their behavior if needed. (Optional)
-  virtual void SetLoadManager(CPULoadStateCallbackInvoker* load_manager) = 0;
-
-  // Specifies the VoiceEngine and VideoEngine channel pair to use for
-  // audio/video synchronization.
-  virtual int ConnectAudioChannel(const int video_channel,
-                                  const int audio_channel) = 0;
-
-  // Disconnects a previously paired VideoEngine and VoiceEngine channel pair.
-  virtual int DisconnectAudioChannel(const int video_channel) = 0;
-
-  // Starts sending packets to an already specified IP address and port number
-  // for a specified channel.
-  virtual int StartSend(const int video_channel) = 0;
-
-  // Stops packets from being sent for a specified channel.
-  virtual int StopSend(const int video_channel) = 0;
-
-  // Prepares VideoEngine for receiving packets on the specified channel.
-  virtual int StartReceive(const int video_channel) = 0;
-
-  // Stops receiving incoming RTP and RTCP packets on the specified channel.
-  virtual int StopReceive(const int video_channel) = 0;
-
-  // Retrieves the version information for VideoEngine and its components.
-  virtual int GetVersion(char version[1024]) = 0;
-
-  // Returns the last VideoEngine error code.
-  virtual int LastError() = 0;
-
-  virtual void RegisterSendStatisticsProxy(
-      int channel,
-      SendStatisticsProxy* send_statistics_proxy) = 0;
-
-  virtual void RegisterReceiveStatisticsProxy(
-      int channel,
-      ReceiveStatisticsProxy* receive_statistics_proxy) = 0;
-
- protected:
-  ViEBase() {}
-  virtual ~ViEBase() {}
-};
-
-}  // namespace webrtc
-
-#endif  // #define WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_BASE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h
deleted file mode 100644
index 66dd9f7c4e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h
+++ /dev/null
@@ -1,249 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-//  - Allocating capture devices.
-//  - Connect a capture device with one or more channels.
-//  - Start and stop capture devices.
-//  - Getting capture device capabilities.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
-
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/common_video/rotation.h"
-
-#ifdef XP_WIN
-typedef int pid_t;
-#endif
-
-namespace webrtc {
-
-class VideoEngine;
-class VideoCaptureModule;
-
-// The observer is registered using RegisterInputObserver() and
-// deregistered using DeregisterInputObserver().
-class WEBRTC_DLLEXPORT ViEInputObserver {
- public:
-  // This method is called if an input device is connected or disconnected .
-  virtual void DeviceChange() = 0;
-
- protected:
-  virtual ~ViEInputObserver() {}
-};
-
-// This structure describes one set of the supported capabilities for a capture
-// device.
-struct CaptureCapability {
-  unsigned int width;
-  unsigned int height;
-  unsigned int maxFPS;
-  RawVideoType rawType;
-  VideoCodecType codecType;
-  unsigned int expectedCaptureDelay;
-  bool interlaced;
-  CaptureCapability() {
-    width = 0;
-    height = 0;
-    maxFPS = 0;
-    rawType = kVideoI420;
-    codecType = kVideoCodecUnknown;
-    expectedCaptureDelay = 0;
-    interlaced = false;
-  }
-};
-
-enum CaptureDeviceType {
-  Camera = 0,
-  Screen = 1,
-  Application = 2,
-  Window = 3,
-  Browser = 4
-};
-struct CaptureDeviceInfo {
-  CaptureDeviceType type;
-
-  CaptureDeviceInfo() : type(CaptureDeviceType::Camera) {}
-  CaptureDeviceInfo(CaptureDeviceType t) : type(t) {}
-};
-
-// This enumerator tells the current brightness alarm mode.
-enum Brightness {
-  Normal = 0,
-  Bright = 1,
-  Dark = 2
-};
-
-// This enumerator describes the capture alarm mode.
-enum CaptureAlarm {
-  AlarmRaised = 0,
-  AlarmCleared = 1
-};
-
-// This class declares an abstract interface to be used when implementing
-// a user-defined capture device. This interface is not meant to be
-// implemented by the user. Instead, the user should call AllocateCaptureDevice
-// in the ViECapture interface, which will create a suitable implementation.
-// The user should then call IncomingFrame in this interface to deliver
-// captured frames to the system.
-class WEBRTC_DLLEXPORT ViEExternalCapture {
- public:
-  ViEExternalCapture() {}
-  virtual ~ViEExternalCapture() {}
-
-  // This method is called by the user to deliver a new captured frame to
-  // VideoEngine.
-  virtual void IncomingFrame(const I420VideoFrame& frame) = 0;
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterObserver() and
-// deregistered using DeregisterObserver().
-class WEBRTC_DLLEXPORT ViECaptureObserver {
- public:
-  // This method is called if a bright or dark captured image is detected.
-  virtual void BrightnessAlarm(const int capture_id,
-                               const Brightness brightness) = 0;
-
-  // This method is called periodically telling the capture device frame rate.
-  virtual void CapturedFrameRate(const int capture_id,
-                                 const unsigned char frame_rate) = 0;
-
-  // This method is called if the capture device stops delivering images to
-  // VideoEngine.
-  virtual void NoPictureAlarm(const int capture_id,
-                              const CaptureAlarm alarm) = 0;
-
- protected:
-  virtual ~ViECaptureObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViECapture {
- public:
-  // Factory for the ViECapture sub‐API and increases an internal reference
-  // counter if successful. Returns NULL if the API is not supported or if
-  // construction fails.
-  static ViECapture* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViECapture sub-API and decreases an internal reference
-  // counter.
-  // Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  // Gets the number of available capture devices.
-  virtual int NumberOfCaptureDevices() = 0;
-
-  // Gets the name and unique id of a capture device.
-  virtual int GetCaptureDevice(unsigned int list_number,
-                               char* device_nameUTF8,
-                               const unsigned int device_nameUTF8Length,
-                               char* unique_idUTF8,
-                               const unsigned int unique_idUTF8Length,
-                               pid_t* pid = nullptr) = 0;
-
-  // Allocates a capture device to be used in VideoEngine.
-  virtual int AllocateCaptureDevice(const char* unique_idUTF8,
-                                    const unsigned int unique_idUTF8Length,
-                                    int& capture_id) = 0;
-
-  // Registers an external capture device to be used in VideoEngine
-  virtual int AllocateExternalCaptureDevice(
-      int& capture_id,
-      ViEExternalCapture *&external_capture) = 0;
-
-  // Use capture device using external capture module.
-  virtual int AllocateCaptureDevice(VideoCaptureModule& capture_module,
-                                    int& capture_id) = 0;
-
-  // Releases a capture device and makes it available for other applications.
-  virtual int ReleaseCaptureDevice(const int capture_id) = 0;
-
-  // This function connects a capture device with a channel. Multiple channels
-  // can be connected to the same capture device.
-  virtual int ConnectCaptureDevice(const int capture_id,
-                                   const int video_channel) = 0;
-
-  // Disconnects a capture device as input for a specified channel.
-  virtual int DisconnectCaptureDevice(const int video_channel) = 0;
-
-  // Makes a capture device start capturing video frames.
-  virtual int StartCapture(
-      const int capture_id,
-      const CaptureCapability& capture_capability = CaptureCapability()) = 0;
-
-  // Stops a started capture device from capturing video frames.
-  virtual int StopCapture(const int capture_id) = 0;
-
-  // Rotates captured frames before encoding and sending.
-  // Used on mobile devices with rotates cameras.
-  virtual int SetVideoRotation(const int capture_id,
-                               const VideoRotation rotation) = 0;
-
-  // This function sets the expected delay from when a video frame is captured
-  // to when that frame is delivered to VideoEngine.
-  virtual int SetCaptureDelay(const int capture_id,
-                              const unsigned int capture_delay_ms) = 0;
-
-  // Returns the number of sets of capture capabilities the capture device
-  // supports.
-  virtual int NumberOfCapabilities(
-      const char* unique_id_utf8,
-      const unsigned int unique_id_utf8_length) = 0;
-
-  // Gets a set of capture capabilities for a specified capture device.
-  virtual int GetCaptureCapability(const char* unique_id_utf8,
-                                   const unsigned int unique_id_utf8_length,
-                                   const unsigned int capability_number,
-                                   CaptureCapability& capability) = 0;
-
-  // Displays the capture device property dialog box for the specified capture
-  // device. Windows only.
-  virtual int ShowCaptureSettingsDialogBox(
-      const char* unique_idUTF8,
-      const unsigned int unique_id_utf8_length,
-      const char* dialog_title,
-      void* parent_window = NULL,
-      const unsigned int x = 200,
-      const unsigned int y = 200) = 0;
-
-  // Gets the clockwise angle the frames from the camera must be rotated in
-  // order to display the frames correctly if the display is rotated in its
-  // natural orientation.
-  virtual int GetOrientation(const char* unique_id_utf8,
-                             VideoRotation& orientation) = 0;
-
-  // Enables brightness alarm detection and the brightness alarm callback.
-  virtual int EnableBrightnessAlarm(const int capture_id,
-                                    const bool enable) = 0;
-
-  // Registers an instance of a user implementation of the ViECaptureObserver.
-  virtual int RegisterObserver(const int capture_id,
-                               ViECaptureObserver& observer) = 0;
-
-  virtual int RegisterInputObserver(ViEInputObserver* observer) = 0;
-
-  // Removes an already registered instance of ViECaptureObserver.
-  virtual int DeregisterObserver(const int capture_id) = 0;
-
-  virtual int DeregisterInputObserver() = 0;
-
- protected:
-  ViECapture() {}
-  virtual ~ViECapture() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h
deleted file mode 100644
index dbdbde91cb..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h
+++ /dev/null
@@ -1,223 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//  - Setting send and receive codecs.
-//  - Codec specific settings.
-//  - Key frame signaling.
-//  - Stream management settings.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-struct VideoCodec;
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterEncoderObserver()
-// and deregistered using DeregisterEncoderObserver().
-class WEBRTC_DLLEXPORT ViEEncoderObserver {
- public:
-  // This method is called once per second with the current encoded frame rate
-  // and bit rate.
-  virtual void OutgoingRate(const int video_channel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) = 0;
-
-  // This method is called whenever the state of the SuspendBelowMinBitrate
-  // changes, i.e., when |is_suspended| toggles.
-  virtual void SuspendChange(int video_channel, bool is_suspended) = 0;
-
- protected:
-  virtual ~ViEEncoderObserver() {}
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterDecoderObserver()
-// and deregistered using DeregisterDecoderObserver().
-class WEBRTC_DLLEXPORT ViEDecoderObserver {
- public:
-  // This method is called when a new incoming stream is detected, normally
-  // triggered by a new incoming SSRC or payload type.
-  virtual void IncomingCodecChanged(const int video_channel,
-                                    const VideoCodec& video_codec) = 0;
-
-  // This method is called once per second containing the frame rate and bit
-  // rate for the incoming stream
-  virtual void IncomingRate(const int video_channel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) = 0;
-
-  // Called periodically with decoder timing information.  All values are
-  // "current" snapshots unless decorated with a min_/max_ prefix.
-  virtual void DecoderTiming(int decode_ms,
-                             int max_decode_ms,
-                             int current_delay_ms,
-                             int target_delay_ms,
-                             int jitter_buffer_ms,
-                             int min_playout_delay_ms,
-                             int render_delay_ms) = 0;
-
-  // This method is called when the decoder needs a new key frame from encoder
-  // on the sender.
-  virtual void RequestNewKeyFrame(const int video_channel) = 0;
-
-  // This method is called when the decoder changes state
-  virtual void ReceiveStateChange(const int video_channel, VideoReceiveState state) = 0;
-
- protected:
-  virtual ~ViEDecoderObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViECodec {
- public:
-  // Factory for the ViECodec sub‐API and increases an internal reference
-  // counter if successful. Returns NULL if the API is not supported or if
-  // construction fails.
-  static ViECodec* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViECodec sub-API and decreases an internal reference
-  // counter.
-  // Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  // Gets the number of available codecs for the VideoEngine build.
-  virtual int NumberOfCodecs() const = 0;
-
-  // Gets a VideoCodec struct for a codec containing the default configuration
-  // for that codec type.
-  virtual int GetCodec(const unsigned char list_number,
-                       VideoCodec& video_codec) const = 0;
-
-  // Sets the send codec to use for a specified channel.
-  virtual int SetSendCodec(const int video_channel,
-                           const VideoCodec& video_codec) = 0;
-
-  // Gets the current send codec settings.
-  virtual int GetSendCodec(const int video_channel,
-                           VideoCodec& video_codec) const = 0;
-
-  // Prepares VideoEngine to receive a certain codec type and setting for a
-  // specified payload type.
-  virtual int SetReceiveCodec(const int video_channel,
-                              const VideoCodec& video_codec) = 0;
-
-  // Gets the current receive codec.
-  virtual int GetReceiveCodec(const int video_channel,
-                              VideoCodec& video_codec) const = 0;
-
-  // This function is used to get codec configuration parameters to be
-  // signaled from the encoder to the decoder in the call setup.
-  virtual int GetCodecConfigParameters(
-      const int video_channel,
-      unsigned char config_parameters[kConfigParameterSize],
-      unsigned char& config_parameters_size) const = 0;
-
-  // Enables advanced scaling of the captured video stream if the stream
-  // differs from the send codec settings.
-  virtual int SetImageScaleStatus(const int video_channel,
-                                  const bool enable) = 0;
-
-  // Gets the number of sent key frames and number of sent delta frames.
-  virtual int GetSendCodecStatistics(const int video_channel,
-                                      unsigned int& key_frames,
-                                      unsigned int& delta_frames) const = 0;
-
-  // Gets the number of decoded key frames and number of decoded delta frames.
-  virtual int GetReceiveCodecStatistics(const int video_channel,
-                                         unsigned int& key_frames,
-                                         unsigned int& delta_frames) const = 0;
-
-  // Estimate of the min required buffer time from the expected arrival time
-  // until rendering to get smooth playback.
-  virtual int GetReceiveSideDelay(const int video_channel,
-                                  int* delay_ms) const = 0;
-
-  // Current target bitrate for this channel.
-  virtual uint32_t GetLastObservedBitrateBps(int video_channel) const = 0;
-  // Gets the bitrate targeted by the video codec rate control in kbit/s.
-  virtual int GetCodecTargetBitrate(const int video_channel,
-                                    unsigned int* bitrate) const = 0;
-
-  // Gets the number of packets discarded by the jitter buffer because they
-  // arrived too late.
-  // TODO(asapersson): Remove default implementation.
-  virtual int GetNumDiscardedPackets(int video_channel) const { return -1; }
-
-  // TODO(asapersson): Remove once the api has been removed from
-  // fakewebrtcvideoengine.h.
-  virtual unsigned int GetDiscardedPackets(
-      const int video_channel) const { return 0; }
-
-  // Enables key frame request callback in ViEDecoderObserver.
-  virtual int SetKeyFrameRequestCallbackStatus(const int video_channel,
-                                               const bool enable) = 0;
-
-  // Enables key frame requests for detected lost packets.
-  virtual int SetSignalKeyPacketLossStatus(
-      const int video_channel,
-      const bool enable,
-      const bool only_key_frames = false) = 0;
-
-  // Registers an instance of a user implementation of the ViEEncoderObserver.
-  virtual int RegisterEncoderObserver(const int video_channel,
-                                      ViEEncoderObserver& observer) = 0;
-
-  // Removes an already registered instance of ViEEncoderObserver.
-  virtual int DeregisterEncoderObserver(const int video_channel) = 0;
-
-  // Registers an instance of a user implementation of the ViEDecoderObserver.
-  virtual int RegisterDecoderObserver(const int video_channel,
-                                      ViEDecoderObserver& observer) = 0;
-
-  // Removes an already registered instance of ViEDecoderObserver.
-  virtual int DeregisterDecoderObserver(const int video_channel) = 0;
-
-  // This function forces the next encoded frame to be a key frame. This is
-  // normally used when the remote endpoint only supports out‐band key frame
-  // request.
-  virtual int SendKeyFrame(const int video_channel) = 0;
-
-  // This function makes the decoder wait for a key frame before starting to
-  // decode the incoming video stream.
-  virtual int WaitForFirstKeyFrame(const int video_channel,
-                                   const bool wait) = 0;
-
-  // Enables recording of debugging information.
-  virtual int StartDebugRecording(int video_channel,
-                                  const char* file_name_utf8) = 0;
-  // Disables recording of debugging information.
-  virtual int StopDebugRecording(int video_channel) = 0;
-
-  // Lets the sender suspend video when the rate drops below
-  // |threshold_bps|, and turns back on when the rate goes back up above
-  // |threshold_bps| + |window_bps|.
-  // This is under development; not tested.
-  virtual void SuspendBelowMinBitrate(int video_channel) = 0;
-
-  // TODO(holmer): Remove this default implementation when possible.
-  virtual bool GetSendSideDelay(int video_channel, int* avg_delay_ms,
-                                int* max_delay_ms) const { return false; }
-
- protected:
-  ViECodec() {}
-  virtual ~ViECodec() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_errors.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_errors.h
deleted file mode 100644
index 24aa0980bb..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_errors.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
-
-enum ViEErrors {
-  // ViEBase.
-  kViENotInitialized = 12000,        // Init has not been called successfully.
-  kViEBaseVoEFailure,                // SetVoiceEngine. ViE failed to use VE instance. Check VE instance pointer.ConnectAudioChannel failed to set voice channel. Have SetVoiceEngine been called? Is the voice channel correct.
-  kViEBaseChannelCreationFailed,     // CreateChannel.
-  kViEBaseInvalidChannelId,          // The channel does not exist.
-  kViEAPIDoesNotExist,               // Release called on Interface that has not been created.
-  kViEBaseInvalidArgument,
-  kViEBaseAlreadySending,            // StartSend called on channel that is already sending.
-  kViEBaseNotSending,                // StopSend called on channel that is not sending.
-  kViEBaseReceiveOnlyChannel,        // Can't send on a receive only channel.
-  kViEBaseAlreadyReceiving,          // StartReceive called on channel that is already receiving.
-  kViEBaseObserverAlreadyRegistered,  // RegisterObserver- an observer has already been set.
-  kViEBaseObserverNotRegistered,     // DeregisterObserver - no observer has been registered.
-  kViEBaseUnknownError,              // An unknown error has occurred. Check the log file.
-
-  // ViECodec.
-  kViECodecInvalidArgument  = 12100,    // Wrong input parameter to function.
-  kViECodecObserverAlreadyRegistered,   // RegisterEncoderObserver, RegisterDecoderObserver.
-  kViECodecObserverNotRegistered,       // DeregisterEncoderObserver, DeregisterDecoderObserver.
-  kViECodecInvalidCodec,                // SetSendCodec,SetReceiveCodec- The codec structure is invalid.
-  kViECodecInvalidChannelId,            // The channel does not exist.
-  kViECodecInUse,                       // SetSendCodec- Can't change codec size or type when multiple channels use the same encoder.
-  kViECodecReceiveOnlyChannel,          // SetSendCodec, can't change receive only channel.
-  kViECodecUnknownError,                // An unknown error has occurred. Check the log file.
-
-  // ViERender.
-  kViERenderInvalidRenderId = 12200,  // No renderer with the ID exist. In AddRenderer - The render ID is invalid. No capture device, channel or file is allocated with that id.
-  kViERenderAlreadyExists,            // AddRenderer: the renderer already exist.
-  kViERenderInvalidFrameFormat,       // AddRender (external renderer). The user has requested a frame format that we don't support.
-  kViERenderUnknownError,             // An unknown error has occurred. Check the log file.
-
-  // ViECapture.
-  kViECaptureDeviceAlreadyConnected = 12300,  // ConnectCaptureDevice - A capture device has already been connected to this video channel.
-  kViECaptureDeviceDoesNotExist,              // No capture device exist with the provided capture id or unique name.
-  kViECaptureDeviceInvalidChannelId,          // ConnectCaptureDevice, DisconnectCaptureDevice- No Channel exist with the provided channel id.
-  kViECaptureDeviceNotConnected,              // DisconnectCaptureDevice- No capture device is connected to the channel.
-  kViECaptureDeviceNotStarted,                // Stop- The capture device is not started.
-  kViECaptureDeviceAlreadyStarted,            // Start- The capture device is already started.
-  kViECaptureDeviceAlreadyAllocated,          // AllocateCaptureDevice The device is already allocated.
-  kViECaptureDeviceMaxNoDevicesAllocated,     // AllocateCaptureDevice Max number of devices already allocated.
-  kViECaptureObserverAlreadyRegistered,       // RegisterObserver- An observer is already registered. Need to deregister first.
-  kViECaptureDeviceObserverNotRegistered,     // DeregisterObserver- No observer is registered.
-  kViECaptureDeviceUnknownError,              // An unknown error has occurred. Check the log file.
-  kViECaptureDeviceMacQtkitNotSupported,      // QTKit handles the capture devices automatically. Thus querying capture capabilities is not supported.
-
-  // ViEFile.
-  kViEFileInvalidChannelId  = 12400,  // No Channel exist with the provided channel id.
-  kViEFileInvalidArgument,            // Incorrect input argument
-  kViEFileAlreadyRecording,           // StartRecordOutgoingVideo - already recording channel
-  kViEFileVoENotSet,                  // StartRecordOutgoingVideo. Failed to access voice engine. Has SetVoiceEngine been called?
-  kViEFileNotRecording,               // StopRecordOutgoingVideo
-  kViEFileMaxNoOfFilesOpened,         // StartPlayFile
-  kViEFileNotPlaying,                 // StopPlayFile. The file with the provided id is not playing.
-  kViEFileObserverAlreadyRegistered,  // RegisterObserver
-  kViEFileObserverNotRegistered,      // DeregisterObserver
-  kViEFileInputAlreadyConnected,      // SendFileOnChannel- the video channel already have a connected input.
-  kViEFileNotConnected,               // StopSendFileOnChannel- No file is being sent on the channel.
-  kViEFileVoEFailure,                 // SendFileOnChannel,StartPlayAudioLocally - failed to play audio stream
-  kViEFileInvalidRenderId,            // SetRenderTimeoutImage and SetRenderStartImage: Renderer with the provided render id does not exist.
-  kViEFileInvalidFile,                // Can't open the file with provided filename. Is the path and file format correct?
-  kViEFileInvalidCapture,             // Can't use ViEPicture. Is the object correct?
-  kViEFileSetRenderTimeoutError,      // SetRenderTimeoutImage- Please see log file.
-  kViEFileSetStartImageError,         // SetRenderStartImage error. Please see log file.
-  kViEFileUnknownError,               // An unknown error has occurred. Check the log file.
-
-  // ViENetwork.
-  kViENetworkInvalidChannelId = 12500,   // No Channel exist with the provided channel id.
-  kViENetworkAlreadyReceiving,           // SetLocalReceiver: Can not change ports while receiving.
-  kViENetworkLocalReceiverNotSet,        // GetLocalReceiver: SetLocalReceiver not called.
-  kViENetworkAlreadySending,             // SetSendDestination
-  kViENetworkDestinationNotSet,          // GetSendDestination
-  kViENetworkInvalidArgument,            // GetLocalIP- Check function  arguments.
-  kViENetworkSendCodecNotSet,            // SetSendGQoS- Need to set the send codec first.
-  kViENetworkServiceTypeNotSupported,    // SetSendGQoS
-  kViENetworkNotSupported,               // SetSendGQoS Not supported on this OS.
-  kViENetworkUnknownError,               // An unknown error has occurred. Check the log file.
-
-  // ViERTP_RTCP.
-  kViERtpRtcpInvalidChannelId = 12600,   // No Channel exist with the provided channel id.
-  kViERtpRtcpAlreadySending,             // The channel is already sending. Need to stop send before calling this API.
-  kViERtpRtcpNotSending,                 // The channel needs to be sending in order for this function to work.
-  kViERtpRtcpRtcpDisabled,               // Functions failed because RTCP is disabled.
-  kViERtpRtcpObserverAlreadyRegistered,  // An observer is already registered. Need to deregister the old first.
-  kViERtpRtcpObserverNotRegistered,      // No observer registered.
-  kViERtpRtcpUnknownError,               // An unknown error has occurred. Check the log file.
-
-  // ViEImageProcess.
-  kViEImageProcessInvalidChannelId  = 12800,  // No Channel exist with the provided channel id.
-  kViEImageProcessInvalidCaptureId,          // No capture device exist with the provided capture id.
-  kViEImageProcessFilterExists,              // RegisterCaptureEffectFilter,RegisterSendEffectFilter,RegisterRenderEffectFilter - Effect filter already registered.
-  kViEImageProcessFilterDoesNotExist,        // DeRegisterCaptureEffectFilter,DeRegisterSendEffectFilter,DeRegisterRenderEffectFilter - Effect filter not registered.
-  kViEImageProcessAlreadyEnabled,            // EnableDeflickering,EnableColorEnhancement- Function already enabled.
-  kViEImageProcessAlreadyDisabled,           // EnableDeflickering,EnableColorEnhancement- Function already disabled.
-  kViEImageProcessUnknownError               // An unknown error has occurred. Check the log file.
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h
deleted file mode 100644
index 99018dbcd2..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class VideoDecoder;
-class VideoEncoder;
-class VideoEngine;
-
-class WEBRTC_DLLEXPORT ViEExternalCodec {
- public:
-  static ViEExternalCodec* GetInterface(VideoEngine* video_engine);
-
-  virtual int Release() = 0;
-
-  virtual int RegisterExternalSendCodec(const int video_channel,
-                                        const unsigned char pl_type,
-                                        VideoEncoder* encoder,
-                                        bool internal_source) = 0;
-
-  virtual int DeRegisterExternalSendCodec(const int video_channel,
-                                          const unsigned char pl_type) = 0;
-
-  virtual int RegisterExternalReceiveCodec(const int video_channel,
-                                           const unsigned char pl_type,
-                                           VideoDecoder* decoder,
-                                           bool decoder_render = false,
-                                           int render_delay = 0) = 0;
-
-  virtual int DeRegisterExternalReceiveCodec(const int video_channel,
-                                             const unsigned char pl_type) = 0;
-
- protected:
-  ViEExternalCodec() {}
-  virtual ~ViEExternalCodec() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h
deleted file mode 100644
index adf0c199c9..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//  - Effect filters
-//  - Deflickering
-//  - Color enhancement
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class EncodedImageCallback;
-class I420FrameCallback;
-class VideoEngine;
-
-// This class declares an abstract interface for a user defined effect filter.
-// The effect filter is registered using RegisterCaptureEffectFilter(),
-// RegisterSendEffectFilter() or RegisterRenderEffectFilter() and deregistered
-// with the corresponding deregister function.
-class WEBRTC_DLLEXPORT ViEEffectFilter {
- public:
-  // This method is called with an I420 video frame allowing the user to
-  // modify the video frame.
-  virtual int Transform(size_t size,
-                        unsigned char* frame_buffer,
-                        int64_t ntp_time_ms,
-                        unsigned int timestamp,
-                        unsigned int width,
-                        unsigned int height) = 0;
- protected:
-  ViEEffectFilter() {}
-  virtual ~ViEEffectFilter() {}
-};
-
-class WEBRTC_DLLEXPORT ViEImageProcess {
- public:
-  // Factory for the ViEImageProcess sub‐API and increases an internal
-  // reference counter if successful. Returns NULL if the API is not supported
-  // or if construction fails.
-  static ViEImageProcess* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViEImageProcess sub-API and decreases an internal reference
-  // counter. Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  // This function registers a EffectFilter to use for a specified capture
-  // device.
-  virtual int RegisterCaptureEffectFilter(const int capture_id,
-                                          ViEEffectFilter& capture_filter) = 0;
-
-  // This function deregisters a EffectFilter for a specified capture device.
-  virtual int DeregisterCaptureEffectFilter(const int capture_id) = 0;
-
-  // This function registers an EffectFilter to use for a specified channel.
-  virtual int RegisterSendEffectFilter(const int video_channel,
-                                       ViEEffectFilter& send_filter) = 0;
-
-  // This function deregisters a send effect filter for a specified channel.
-  virtual int DeregisterSendEffectFilter(const int video_channel) = 0;
-
-  // This function registers a EffectFilter to use for the rendered video
-  // stream on an incoming channel.
-  virtual int RegisterRenderEffectFilter(const int video_channel,
-                                         ViEEffectFilter& render_filter) = 0;
-
-  // This function deregisters a render effect filter for a specified channel.
-  virtual int DeregisterRenderEffectFilter(const int video_channel) = 0;
-
-  // All cameras run the risk of getting in almost perfect sync with
-  // florescent lamps, which will result in a very annoying flickering of the
-  // image. Most cameras have some type of filter to protect against this but
-  // not all of them succeed. Enabling this function will remove the flicker.
-  virtual int EnableDeflickering(const int capture_id, const bool enable) = 0;
-
-  // TODO(pbos): Remove this function when removed from fakewebrtcvideoengine.h.
-  virtual int EnableDenoising(const int capture_id, const bool enable) {
-    return -1;
-  }
-
-  // This function enhances the colors on the decoded video stream, enabled by
-  // default.
-  virtual int EnableColorEnhancement(const int video_channel,
-                                     const bool enable) = 0;
-
-  // New-style callbacks, used by VideoSendStream/VideoReceiveStream.
-  virtual void RegisterPreEncodeCallback(
-      int video_channel,
-      I420FrameCallback* pre_encode_callback) = 0;
-  virtual void DeRegisterPreEncodeCallback(int video_channel) = 0;
-
-  virtual void RegisterPostEncodeImageCallback(
-      int video_channel,
-      EncodedImageCallback* post_encode_callback) {}
-  virtual void DeRegisterPostEncodeCallback(int video_channel) {}
-
-  virtual void RegisterPreDecodeImageCallback(
-      int video_channel,
-      EncodedImageCallback* pre_decode_callback) {}
-  virtual void DeRegisterPreDecodeCallback(int video_channel) {}
-
-  virtual void RegisterPreRenderCallback(
-      int video_channel,
-      I420FrameCallback* pre_render_callback) = 0;
-  virtual void DeRegisterPreRenderCallback(int video_channel) = 0;
-
- protected:
-  ViEImageProcess() {}
-  virtual ~ViEImageProcess() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h
deleted file mode 100644
index e962e729aa..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
-
-// This sub-API supports the following functionalities:
-//  - Configuring send and receive addresses.
-//  - External transport support.
-//  - Port and address filters.
-//  - Windows GQoS functions and ToS functions.
-//  - Packet timeout notification.
-//  - Dead‐or‐Alive connection observations.
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class Transport;
-class VideoEngine;
-
-// This enumerator describes VideoEngine packet timeout states.
-enum ViEPacketTimeout {
-  NoPacket = 0,
-  PacketReceived = 1
-};
-
-class WEBRTC_DLLEXPORT ViENetwork {
- public:
-  // Default values.
-  enum { KDefaultSampleTimeSeconds = 2 };
-
-  // Factory for the ViENetwork sub‐API and increases an internal reference
-  // counter if successful. Returns NULL if the API is not supported or if
-  // construction fails.
-  static ViENetwork* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViENetwork sub-API and decreases an internal reference
-  // counter.Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  virtual void SetBitrateConfig(int video_channel,
-                                int min_bitrate_bps,
-                                int start_bitrate_bps,
-                                int max_bitrate_bps) = 0;
-
-  // Inform the engine about if the network adapter is currently transmitting
-  // packets or not.
-  virtual void SetNetworkTransmissionState(const int video_channel,
-                                           const bool is_transmitting) = 0;
-
-  // This function registers a user implementation of Transport to use for
-  // sending RTP and RTCP packets on this channel.
-  virtual int RegisterSendTransport(const int video_channel,
-                                    Transport& transport) = 0;
-
-  // This function deregisters a used Transport for a specified channel.
-  virtual int DeregisterSendTransport(const int video_channel) = 0;
-
-  // When using external transport for a channel, received RTP packets should
-  // be passed to VideoEngine using this function. The input should contain
-  // the RTP header and payload.
-  virtual int ReceivedRTPPacket(const int video_channel,
-                                const void* data,
-                                const size_t length,
-                                const PacketTime& packet_time) = 0;
-
-  // When using external transport for a channel, received RTCP packets should
-  // be passed to VideoEngine using this function.
-  virtual int ReceivedRTCPPacket(const int video_channel,
-                                 const void* data,
-                                 const size_t length) = 0;
-
-  // This function sets the Maximum Transition Unit (MTU) for a channel. The
-  // RTP packet will be packetized based on this MTU to optimize performance
-  // over the network.
-  virtual int SetMTU(int video_channel, unsigned int mtu) = 0;
-
-  // Forward (audio) packet to bandwidth estimator for the given video channel,
-  // for aggregated audio+video BWE.
-  virtual int ReceivedBWEPacket(const int video_channel,
-      int64_t arrival_time_ms, size_t payload_size, const RTPHeader& header) {
-    return 0;
-  }
-
- protected:
-  ViENetwork() {}
-  virtual ~ViENetwork() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h
deleted file mode 100644
index 8d3f2b5a1a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//  - Specify render destinations for incoming video streams, capture devices
-//    and files.
-//  - Configuring render streams.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-class I420VideoFrame;
-class VideoEngine;
-class VideoRender;
-class VideoRenderCallback;
-
-// This class declares an abstract interface to be used for external renderers.
-// The user implemented derived class is registered using AddRenderer().
-class ExternalRenderer {
- public:
-  // This method will be called when the stream to be rendered changes in
-  // resolution or number of streams mixed in the image.
-  virtual int FrameSizeChange(unsigned int width,
-                              unsigned int height,
-                              unsigned int number_of_streams) = 0;
-
-  // This method is called when a new frame should be rendered.
-  virtual int DeliverFrame(unsigned char* buffer,
-                           size_t buffer_size,
-                           // RTP timestamp in 90kHz.
-                           uint32_t timestamp,
-                           // NTP time of the capture time in local timebase
-                           // in milliseconds.
-                           int64_t ntp_time_ms,
-                           // Wallclock render time in milliseconds.
-                           int64_t render_time_ms,
-                           // Handle of the underlying video frame.
-                           void* handle) = 0;
-
-  // Alternative interface for I420 frames.
-  virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) = 0;
-
-  // Returns true if the renderer supports textures. DeliverFrame can be called
-  // with NULL |buffer| and non-NULL |handle|.
-  virtual bool IsTextureSupported() = 0;
-
- protected:
-  virtual ~ExternalRenderer() {}
-};
-
-class ViERender {
- public:
-  // Factory for the ViERender sub‐API and increases an internal reference
-  // counter if successful. Returns NULL if the API is not supported or if
-  // construction fails.
-  static ViERender* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViERender sub-API and decreases an internal reference
-  // counter. Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  // Registers render module.
-  virtual int RegisterVideoRenderModule(VideoRender& render_module) = 0;
-
-  // Deregisters render module.
-  virtual int DeRegisterVideoRenderModule(VideoRender& render_module) = 0;
-
-  // Sets the render destination for a given render ID.
-  virtual int AddRenderer(const int render_id,
-                          void* window,
-                          const unsigned int z_order,
-                          const float left,
-                          const float top,
-                          const float right,
-                          const float bottom) = 0;
-
-  // Removes the renderer for a stream.
-  virtual int RemoveRenderer(const int render_id) = 0;
-
-  // Starts rendering a render stream.
-  virtual int StartRender(const int render_id) = 0;
-
-  // Stops rendering a render stream.
-  virtual int StopRender(const int render_id) = 0;
-
-  // Set expected render time needed by graphics card or external renderer, i.e.
-  // the number of ms a frame will be sent to rendering before the actual render
-  // time.
-  virtual int SetExpectedRenderDelay(int render_id, int render_delay) = 0;
-
-  // Configures an already added render stream.
-  virtual int ConfigureRender(int render_id,
-                              const unsigned int z_order,
-                              const float left,
-                              const float top,
-                              const float right,
-                              const float bottom) = 0;
-
-  // External render.
-  virtual int AddRenderer(const int render_id,
-                          RawVideoType video_input_format,
-                          ExternalRenderer* renderer) = 0;
-
- protected:
-  ViERender() {}
-  virtual ~ViERender() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h
deleted file mode 100644
index 94eb5083d8..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h
+++ /dev/null
@@ -1,496 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//  - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
-//  - SSRC handling.
-//  - Transmission of RTCP reports.
-//  - Obtaining RTCP data from incoming RTCP sender reports.
-//  - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
-//  - Forward Error Correction (FEC).
-//  - Writing RTP and RTCP packets to binary files for off‐line analysis of the
-//    call quality.
-//  - Inserting extra RTP packets into active audio stream.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-
-namespace webrtc {
-
-class VideoEngine;
-struct ReceiveBandwidthEstimatorStats;
-
-// This enumerator sets the RTCP mode.
-enum ViERTCPMode {
-  kRtcpNone = 0,
-  kRtcpCompound_RFC4585 = 1,
-  kRtcpNonCompound_RFC5506 = 2
-};
-
-// This enumerator describes the key frame request mode.
-enum ViEKeyFrameRequestMethod {
-  kViEKeyFrameRequestNone = 0,
-  kViEKeyFrameRequestPliRtcp = 1,
-  kViEKeyFrameRequestFirRtp = 2,
-  kViEKeyFrameRequestFirRtcp = 3
-};
-
-enum StreamType {
-  kViEStreamTypeNormal = 0,  // Normal media stream
-  kViEStreamTypeRtx = 1  // Retransmission media stream
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterRTPObserver() and
-// deregistered using DeregisterRTPObserver().
-class WEBRTC_DLLEXPORT ViERTPObserver {
- public:
-  // This method is called if SSRC of the incoming stream is changed.
-  virtual void IncomingSSRCChanged(const int video_channel,
-                                   const unsigned int SSRC) = 0;
-
-  // This method is called if a field in CSRC changes or if the number of
-  // CSRCs changes.
-  virtual void IncomingCSRCChanged(const int video_channel,
-                                   const unsigned int CSRC,
-                                   const bool added) = 0;
- protected:
-  virtual ~ViERTPObserver() {}
-};
-
-struct SenderInfo;
-
-class WEBRTC_DLLEXPORT ViERTP_RTCP {
- public:
-  enum { KDefaultDeltaTransmitTimeSeconds = 15 };
-  enum { KMaxRTCPCNameLength = 256 };
-
-  // Factory for the ViERTP_RTCP sub‐API and increases an internal reference
-  // counter if successful. Returns NULL if the API is not supported or if
-  // construction fails.
-  static ViERTP_RTCP* GetInterface(VideoEngine* video_engine);
-
-  // Releases the ViERTP_RTCP sub-API and decreases an internal reference
-  // counter. Returns the new reference count. This value should be zero
-  // for all sub-API:s before the VideoEngine object can be safely deleted.
-  virtual int Release() = 0;
-
-  // This function enables you to specify the RTP synchronization source
-  // identifier (SSRC) explicitly.
-  virtual int SetLocalSSRC(const int video_channel,
-                           const unsigned int SSRC,
-                           const StreamType usage = kViEStreamTypeNormal,
-                           const unsigned char simulcast_idx = 0) = 0;
-
-  // This function gets the SSRC for the outgoing RTP stream for the specified
-  // channel.
-  virtual int GetLocalSSRC(const int video_channel,
-                           unsigned int& SSRC) const = 0;
-
-  // This function map a incoming SSRC to a StreamType so that the engine
-  // can know which is the normal stream and which is the RTX
-  virtual int SetRemoteSSRCType(const int video_channel,
-                                const StreamType usage,
-                                const unsigned int SSRC) const = 0;
-
-  // This function gets the SSRC for the incoming RTP stream for the specified
-  // channel.
-  virtual int GetRemoteSSRC(const int video_channel,
-                            unsigned int& SSRC) const = 0;
-
-  // This function returns the CSRCs of the incoming RTP packets.
-  virtual int GetRemoteCSRCs(const int video_channel,
-                             unsigned int CSRCs[kRtpCsrcSize]) const = 0;
-
-  // This function gets the RID value (if any) for the incoming RTP stream
-  // for the specified channel.
-  virtual int GetRemoteRID(const int video_channel,
-                           char rid[256]) const = 0;
-
-  // This sets a specific payload type for the RTX stream. Note that this
-  // doesn't enable RTX, SetLocalSSRC must still be called to enable RTX.
-  virtual int SetRtxSendPayloadType(const int video_channel,
-                                    const uint8_t payload_type) = 0;
-
-  virtual int SetRtxReceivePayloadType(const int video_channel,
-                                       const uint8_t payload_type) = 0;
-
-  // This function enables manual initialization of the sequence number. The
-  // start sequence number is normally a random number.
-  virtual int SetStartSequenceNumber(const int video_channel,
-                                     unsigned short sequence_number) = 0;
-
-  // TODO(pbos): Remove default implementation once this has been implemented
-  // in libjingle.
-  virtual void SetRtpStateForSsrc(int video_channel,
-                                  uint32_t ssrc,
-                                  const RtpState& rtp_state) {}
-  // TODO(pbos): Remove default implementation once this has been implemented
-  // in libjingle.
-  virtual RtpState GetRtpStateForSsrc(int video_channel, uint32_t ssrc) {
-    return RtpState();
-  }
-
-  // This function sets the RTCP status for the specified channel.
-  // Default mode is kRtcpCompound_RFC4585.
-  virtual int SetRTCPStatus(const int video_channel,
-                            const ViERTCPMode rtcp_mode) = 0;
-
-  // This function gets the RTCP status for the specified channel.
-  virtual int GetRTCPStatus(const int video_channel,
-                            ViERTCPMode& rtcp_mode) const = 0;
-
-  // This function sets the RTCP canonical name (CNAME) for the RTCP reports
-  // on a specific channel.
-  virtual int SetRTCPCName(const int video_channel,
-                           const char rtcp_cname[KMaxRTCPCNameLength]) = 0;
-
-  // TODO(holmer): Remove this API once it has been removed from
-  // fakewebrtcvideoengine.h.
-  virtual int GetRTCPCName(const int video_channel,
-                           char rtcp_cname[KMaxRTCPCNameLength]) const {
-    return -1;
-  }
-
-  // This function gets the RTCP canonical name (CNAME) for the RTCP reports
-  // received on the specified channel.
-  virtual int GetRemoteRTCPCName(
-      const int video_channel,
-      char rtcp_cname[KMaxRTCPCNameLength]) const = 0;
-
-  virtual int GetRemoteRTCPReceiverInfo(const int video_channel,
-                                        uint32_t& NTPHigh,
-                                        uint32_t& NTPLow,
-                                        uint32_t& receivedPacketCount,
-                                        uint64_t& receivedOctetCount,
-                                        uint32_t* jitter,
-                                        uint16_t* fractionLost,
-                                        uint32_t* cumulativeLost,
-                                        int32_t* rttMs) const = 0;
-
-  // This function sends an RTCP APP packet on a specific channel.
-  virtual int SendApplicationDefinedRTCPPacket(
-      const int video_channel,
-      const unsigned char sub_type,
-      unsigned int name,
-      const char* data,
-      unsigned short data_length_in_bytes) = 0;
-
-  // This function enables Negative Acknowledgment (NACK) using RTCP,
-  // implemented based on RFC 4585. NACK retransmits RTP packets if lost on
-  // the network. This creates a lossless transport at the expense of delay.
-  // If using NACK, NACK should be enabled on both endpoints in a call.
-  virtual int SetNACKStatus(const int video_channel, const bool enable) = 0;
-
-  // This function enables Forward Error Correction (FEC) using RTCP,
-  // implemented based on RFC 5109, to improve packet loss robustness. Extra
-  // FEC packets are sent together with the usual media packets, hence
-  // part of the bitrate will be used for FEC packets.
-  virtual int SetFECStatus(const int video_channel,
-                           const bool enable,
-                           const unsigned char payload_typeRED,
-                           const unsigned char payload_typeFEC) = 0;
-
-  // This function enables hybrid Negative Acknowledgment using RTCP
-  // and Forward Error Correction (FEC) implemented based on RFC 5109,
-  // to improve packet loss robustness. Extra
-  // FEC packets are sent together with the usual media packets, hence will
-  // part of the bitrate be used for FEC packets.
-  // The hybrid mode will choose between nack only, fec only and both based on
-  // network conditions. When both are applied, only packets that were not
-  // recovered by the FEC will be nacked.
-  virtual int SetHybridNACKFECStatus(const int video_channel,
-                                     const bool enable,
-                                     const unsigned char payload_typeRED,
-                                     const unsigned char payload_typeFEC) = 0;
-
-  // Sets send side support for delayed video buffering (actual delay will
-  // be exhibited on the receiver side).
-  // Target delay should be set to zero for real-time mode.
-  virtual int SetSenderBufferingMode(int video_channel,
-                                     int target_delay_ms) = 0;
-  // Sets receive side support for delayed video buffering. Target delay should
-  // be set to zero for real-time mode.
-  virtual int SetReceiverBufferingMode(int video_channel,
-                                       int target_delay_ms) = 0;
-
-  // This function enables RTCP key frame requests.
-  virtual int SetKeyFrameRequestMethod(
-    const int video_channel, const ViEKeyFrameRequestMethod method) = 0;
-
-  // This function enables signaling of temporary bitrate constraints using
-  // RTCP, implemented based on RFC4585.
-  virtual int SetTMMBRStatus(const int video_channel, const bool enable) = 0;
-
-  // Enables and disables REMB packets for this channel. |sender| indicates
-  // this channel is encoding, |receiver| tells the bitrate estimate for
-  // this channel should be included in the REMB packet.
-  virtual int SetRembStatus(int video_channel,
-                            bool sender,
-                            bool receiver) = 0;
-
-  // Enables RTP timestamp extension offset described in RFC 5450. This call
-  // must be done before ViECodec::SetSendCodec is called.
-  virtual int SetSendTimestampOffsetStatus(int video_channel,
-                                           bool enable,
-                                           int id) = 0;
-
-  virtual int SetReceiveTimestampOffsetStatus(int video_channel,
-                                              bool enable,
-                                              int id) = 0;
-
-  // Enables RTP absolute send time header extension. This call must be done
-  // before ViECodec::SetSendCodec is called.
-  virtual int SetSendAbsoluteSendTimeStatus(int video_channel,
-                                            bool enable,
-                                            int id) = 0;
-
-  // When enabled for a channel, *all* channels on the same transport will be
-  // expected to include the absolute send time header extension.
-  virtual int SetReceiveAbsoluteSendTimeStatus(int video_channel,
-                                               bool enable,
-                                               int id) = 0;
-
-  virtual int SetSendVideoRotationStatus(int video_channel,
-                                         bool enable,
-                                         int id) = 0;
-
-  virtual int SetReceiveVideoRotationStatus(int video_channel,
-                                            bool enable,
-                                            int id) = 0;
-
-  virtual int SetSendRIDStatus(int video_channel,
-                               bool enable,
-                               int id,
-                               const char *rid) = 0;
-
-  virtual int SetReceiveRIDStatus(int video_channel,
-                                  bool enable,
-                                  int id) = 0;
-
-  // Enables/disables RTCP Receiver Reference Time Report Block extension/
-  // DLRR Report Block extension (RFC 3611).
-  virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) = 0;
-
-  // Enables transmission smoothening, i.e. packets belonging to the same frame
-  // will be sent over a longer period of time instead of sending them
-  // back-to-back.
-  virtual int SetTransmissionSmoothingStatus(int video_channel,
-                                             bool enable) = 0;
-
-  // Sets a minimal bitrate which will be padded to when the encoder doesn't
-  // produce enough bitrate.
-  // TODO(pbos): Remove default implementation when libjingle's
-  // FakeWebRtcVideoEngine is updated.
-  virtual int SetMinTransmitBitrate(int video_channel,
-                                    int min_transmit_bitrate_kbps) {
-    return -1;
-  };
-
-  // Set a constant amount to deduct from received bitrate estimates before
-  // using it to allocate capacity among outgoing video streams.
-  virtual int SetReservedTransmitBitrate(
-      int video_channel, unsigned int reserved_transmit_bitrate_bps) {
-    return 0;
-  }
-
-  // This function returns our locally created statistics of the received RTP
-  // stream.
-  virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
-                                              RtcpStatistics& basic_stats,
-                                              int64_t& rtt_ms) const = 0;
-
-  // This function returns statistics reported by the remote client in RTCP
-  // report blocks. If several streams are reported, the statistics will be
-  // aggregated.
-  // If statistics are aggregated, extended_max_sequence_number is not reported,
-  // and will always be set to 0.
-  virtual int GetSendChannelRtcpStatistics(const int video_channel,
-                                           RtcpStatistics& basic_stats,
-                                           int64_t& rtt_ms) const = 0;
-
-  // TODO(sprang): Temporary hacks to prevent libjingle build from failing,
-  // remove when libjingle has been lifted to support webrtc issue 2589
-  virtual int GetReceivedRTCPStatistics(const int video_channel,
-                                        unsigned short& fraction_lost,
-                                        unsigned int& cumulative_lost,
-                                        unsigned int& extended_max,
-                                        unsigned int& jitter,
-                                        int64_t& rtt_ms) const {
-    RtcpStatistics stats;
-    int ret_code = GetReceiveChannelRtcpStatistics(video_channel,
-                                             stats,
-                                             rtt_ms);
-    fraction_lost = stats.fraction_lost;
-    cumulative_lost = stats.cumulative_lost;
-    extended_max = stats.extended_max_sequence_number;
-    jitter = stats.jitter;
-    return ret_code;
-  }
-  virtual int GetSentRTCPStatistics(const int video_channel,
-                                    unsigned short& fraction_lost,
-                                    unsigned int& cumulative_lost,
-                                    unsigned int& extended_max,
-                                    unsigned int& jitter,
-                                    int64_t& rtt_ms) const {
-    RtcpStatistics stats;
-    int ret_code = GetSendChannelRtcpStatistics(video_channel,
-                                                stats,
-                                                rtt_ms);
-    fraction_lost = stats.fraction_lost;
-    cumulative_lost = stats.cumulative_lost;
-    extended_max = stats.extended_max_sequence_number;
-    jitter = stats.jitter;
-    return ret_code;
-  }
-
-
-  virtual int RegisterSendChannelRtcpStatisticsCallback(
-      int video_channel, RtcpStatisticsCallback* callback) = 0;
-
-  virtual int DeregisterSendChannelRtcpStatisticsCallback(
-      int video_channel, RtcpStatisticsCallback* callback) = 0;
-
-  virtual int RegisterReceiveChannelRtcpStatisticsCallback(
-      int video_channel, RtcpStatisticsCallback* callback) = 0;
-
-  virtual int DeregisterReceiveChannelRtcpStatisticsCallback(
-      int video_channel, RtcpStatisticsCallback* callback) = 0;
-
-  // The function gets statistics from the sent and received RTP streams.
-  virtual int GetRtpStatistics(const int video_channel,
-                               StreamDataCounters& sent,
-                               StreamDataCounters& received) const = 0;
-
-  // TODO(sprang): Temporary hacks to prevent libjingle build from failing,
-  // remove when libjingle has been lifted to support webrtc issue 2589
-  virtual int GetRTPStatistics(const int video_channel,
-                       size_t& bytes_sent,
-                       unsigned int& packets_sent,
-                       size_t& bytes_received,
-                       unsigned int& packets_received) const {
-    StreamDataCounters sent;
-    StreamDataCounters received;
-    int ret_code = GetRtpStatistics(video_channel, sent, received);
-    bytes_sent = sent.transmitted.payload_bytes;
-    packets_sent = sent.transmitted.packets;
-    bytes_received = received.transmitted.payload_bytes;
-    packets_received = received.transmitted.packets;
-    return ret_code;
-  }
-
-  virtual int RegisterSendChannelRtpStatisticsCallback(
-      int video_channel, StreamDataCountersCallback* callback) = 0;
-
-  virtual int DeregisterSendChannelRtpStatisticsCallback(
-      int video_channel, StreamDataCountersCallback* callback) = 0;
-
-  virtual int RegisterReceiveChannelRtpStatisticsCallback(
-      int video_channel, StreamDataCountersCallback* callback) = 0;
-
-  virtual int DeregisterReceiveChannelRtpStatisticsCallback(
-      int video_channel, StreamDataCountersCallback* callback) = 0;
-
-
-  // Gets RTCP packet type statistics from a sent/received stream.
-  virtual int GetSendRtcpPacketTypeCounter(
-      int video_channel,
-      RtcpPacketTypeCounter* packet_counter) const = 0;
-
-  virtual int GetReceiveRtcpPacketTypeCounter(
-      int video_channel,
-      RtcpPacketTypeCounter* packet_counter) const = 0;
-
-  // Gets the sender info part of the last received RTCP Sender Report (SR)
-  virtual int GetRemoteRTCPSenderInfo(const int video_channel,
-                                      SenderInfo* sender_info) const = 0;
-
-  // The function gets bandwidth usage statistics from the sent RTP streams in
-  // bits/s.
-  virtual int GetBandwidthUsage(const int video_channel,
-                                unsigned int& total_bitrate_sent,
-                                unsigned int& video_bitrate_sent,
-                                unsigned int& fec_bitrate_sent,
-                                unsigned int& nackBitrateSent) const = 0;
-
-  // (De)Register an observer, called whenever the send bitrate is updated
-  virtual int RegisterSendBitrateObserver(
-      int video_channel,
-      BitrateStatisticsObserver* observer) = 0;
-
-  virtual int DeregisterSendBitrateObserver(
-      int video_channel,
-      BitrateStatisticsObserver* observer) = 0;
-
-  // This function gets the send-side estimated bandwidth available for video,
-  // including overhead, in bits/s.
-  virtual int GetEstimatedSendBandwidth(
-      const int video_channel,
-      unsigned int* estimated_bandwidth) const = 0;
-
-  // This function gets the receive-side estimated bandwidth available for
-  // video, including overhead, in bits/s. |estimated_bandwidth| is 0 if there
-  // is no valid estimate.
-  virtual int GetEstimatedReceiveBandwidth(
-      const int video_channel,
-      unsigned int* estimated_bandwidth) const = 0;
-
-  // This function gets the PacedSender queuing delay for the last sent frame.
-  // TODO(jiayl): remove the default impl when libjingle is updated.
-  virtual int GetPacerQueuingDelayMs(
-      const int video_channel, int64_t* delay_ms) const {
-    return -1;
-  }
-
-  // This function enables capturing of RTP packets to a binary file on a
-  // specific channel and for a given direction. The file can later be
-  // replayed using e.g. RTP Tools rtpplay since the binary file format is
-  // compatible with the rtpdump format.
-  virtual int StartRTPDump(const int video_channel,
-                           const char file_nameUTF8[1024],
-                           RTPDirections direction) = 0;
-
-  // This function disables capturing of RTP packets to a binary file on a
-  // specific channel and for a given direction.
-  virtual int StopRTPDump(const int video_channel,
-                          RTPDirections direction) = 0;
-
-  // Registers an instance of a user implementation of the ViERTPObserver.
-  virtual int RegisterRTPObserver(const int video_channel,
-                                  ViERTPObserver& observer) = 0;
-
-  // Removes a registered instance of ViERTPObserver.
-  virtual int DeregisterRTPObserver(const int video_channel) = 0;
-
-  // Registers and instance of a user implementation of ViEFrameCountObserver
-  virtual int RegisterSendFrameCountObserver(
-      int video_channel, FrameCountObserver* observer) = 0;
-
-  // Removes a registered instance of a ViEFrameCountObserver
-  virtual int DeregisterSendFrameCountObserver(
-      int video_channel, FrameCountObserver* observer) = 0;
-
-  // Called when RTCP packet type counters might have been changed. User has to
-  // filter on SSRCs to determine whether it's status sent or received.
-  virtual int RegisterRtcpPacketTypeCounterObserver(
-      int video_channel,
-      RtcpPacketTypeCounterObserver* observer) = 0;
-
- protected:
-  virtual ~ViERTP_RTCP() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h b/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h
deleted file mode 100644
index a59fdbfddb..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_VIDEO_ENGINE_MOCK_MOCK_VIE_FRAME_PROVIDER_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_MOCK_MOCK_VIE_FRAME_PROVIDER_BASE_H_
-
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-#include "testing/gmock/include/gmock/gmock.h"
-
-namespace webrtc {
-
-class MockViEFrameCallback : public ViEFrameCallback {
- public:
-  MOCK_METHOD3(DeliverFrame,
-               void(int id,
-                    I420VideoFrame* video_frame,
-                    const std::vector<uint32_t>& csrcs));
-  MOCK_METHOD2(DelayChanged, void(int id, int frame_delay));
-  MOCK_METHOD3(GetPreferedFrameSettings,
-               int(int* width, int* height, int* frame_rate));
-  MOCK_METHOD1(ProviderDestroyed, void(int id));
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_MOCK_MOCK_VIE_FRAME_PROVIDER_BASE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc
deleted file mode 100644
index cfb7f01d4f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc
+++ /dev/null
@@ -1,564 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/overuse_frame_detector.h"
-
-namespace webrtc {
-namespace {
-  const int kWidth = 640;
-  const int kHeight = 480;
-  const int kFrameInterval33ms = 33;
-  const int kProcessIntervalMs = 5000;
-}  // namespace
-
-class MockCpuOveruseObserver : public CpuOveruseObserver {
- public:
-  MockCpuOveruseObserver() {}
-  virtual ~MockCpuOveruseObserver() {}
-
-  MOCK_METHOD0(OveruseDetected, void());
-  MOCK_METHOD0(NormalUsage, void());
-};
-
-class CpuOveruseObserverImpl : public CpuOveruseObserver {
- public:
-  CpuOveruseObserverImpl() :
-    overuse_(0),
-    normaluse_(0) {}
-  virtual ~CpuOveruseObserverImpl() {}
-
-  void OveruseDetected() { ++overuse_; }
-  void NormalUsage() { ++normaluse_; }
-
-  int overuse_;
-  int normaluse_;
-};
-
-class OveruseFrameDetectorTest : public ::testing::Test,
-                                 public CpuOveruseMetricsObserver {
- protected:
-  virtual void SetUp() {
-    clock_.reset(new SimulatedClock(1234));
-    observer_.reset(new MockCpuOveruseObserver());
-    overuse_detector_.reset(new OveruseFrameDetector(clock_.get(), this));
-
-    options_.low_capture_jitter_threshold_ms = 10.0f;
-    options_.high_capture_jitter_threshold_ms = 15.0f;
-    options_.min_process_count = 0;
-    overuse_detector_->SetOptions(options_);
-    overuse_detector_->SetObserver(observer_.get());
-  }
-
-  void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override {
-    metrics_ = metrics;
-  }
-
-  int InitialJitter() {
-    return ((options_.low_capture_jitter_threshold_ms +
-             options_.high_capture_jitter_threshold_ms) / 2.0f) + 0.5;
-  }
-
-  int InitialUsage() {
-    return ((options_.low_encode_usage_threshold_percent +
-             options_.high_encode_usage_threshold_percent) / 2.0f) + 0.5;
-  }
-
-  void InsertFramesWithInterval(
-      size_t num_frames, int interval_ms, int width, int height) {
-    while (num_frames-- > 0) {
-      clock_->AdvanceTimeMilliseconds(interval_ms);
-      overuse_detector_->FrameCaptured(width, height,
-                                       clock_->TimeInMilliseconds());
-    }
-  }
-
-  void InsertAndSendFramesWithInterval(
-      int num_frames, int interval_ms, int width, int height, int delay_ms) {
-    while (num_frames-- > 0) {
-      int64_t capture_time_ms = clock_->TimeInMilliseconds();
-      overuse_detector_->FrameCaptured(width, height, capture_time_ms);
-      clock_->AdvanceTimeMilliseconds(delay_ms);
-      overuse_detector_->FrameEncoded(delay_ms);
-      overuse_detector_->FrameSent(capture_time_ms);
-      clock_->AdvanceTimeMilliseconds(interval_ms - delay_ms);
-    }
-  }
-
-  void TriggerOveruse(int num_times) {
-    for (int i = 0; i < num_times; ++i) {
-      InsertFramesWithInterval(200, kFrameInterval33ms, kWidth, kHeight);
-      InsertFramesWithInterval(50, 110, kWidth, kHeight);
-      overuse_detector_->Process();
-    }
-  }
-
-  void TriggerUnderuse() {
-    InsertFramesWithInterval(900, kFrameInterval33ms, kWidth, kHeight);
-    overuse_detector_->Process();
-  }
-
-  void TriggerOveruseWithProcessingUsage(int num_times) {
-    const int kDelayMs = 32;
-    for (int i = 0; i < num_times; ++i) {
-      InsertAndSendFramesWithInterval(
-          1000, kFrameInterval33ms, kWidth, kHeight, kDelayMs);
-      overuse_detector_->Process();
-    }
-  }
-
-  void TriggerUnderuseWithProcessingUsage() {
-    const int kDelayMs1 = 5;
-    const int kDelayMs2 = 6;
-    InsertAndSendFramesWithInterval(
-        1300, kFrameInterval33ms, kWidth, kHeight, kDelayMs1);
-    InsertAndSendFramesWithInterval(
-        1, kFrameInterval33ms, kWidth, kHeight, kDelayMs2);
-    overuse_detector_->Process();
-  }
-
-  int CaptureJitterMs() { return metrics_.capture_jitter_ms; }
-
-  int AvgEncodeTimeMs() { return metrics_.avg_encode_time_ms; }
-
-  int UsagePercent() { return metrics_.encode_usage_percent; }
-
-  CpuOveruseOptions options_;
-  rtc::scoped_ptr<SimulatedClock> clock_;
-  rtc::scoped_ptr<MockCpuOveruseObserver> observer_;
-  rtc::scoped_ptr<OveruseFrameDetector> overuse_detector_;
-  CpuOveruseMetrics metrics_;
-};
-
-// enable_capture_jitter_method = true;
-// CaptureJitterMs() > high_capture_jitter_threshold_ms => overuse.
-// CaptureJitterMs() < low_capture_jitter_threshold_ms => underuse.
-TEST_F(OveruseFrameDetectorTest, TriggerOveruse) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  // capture_jitter > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  TriggerOveruse(options_.high_threshold_consecutive_count);
-}
-
-TEST_F(OveruseFrameDetectorTest, OveruseAndRecover) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  // capture_jitter > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  TriggerOveruse(options_.high_threshold_consecutive_count);
-  // capture_jitter < low => underuse
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(testing::AtLeast(1));
-  TriggerUnderuse();
-}
-
-TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithNoObserver) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  overuse_detector_->SetObserver(NULL);
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
-  TriggerOveruse(options_.high_threshold_consecutive_count);
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
-  TriggerUnderuse();
-}
-
-TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithMethodDisabled) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
-  TriggerOveruse(options_.high_threshold_consecutive_count);
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
-  TriggerUnderuse();
-}
-
-TEST_F(OveruseFrameDetectorTest, DoubleOveruseAndRecover) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(2);
-  TriggerOveruse(options_.high_threshold_consecutive_count);
-  TriggerOveruse(options_.high_threshold_consecutive_count);
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(testing::AtLeast(1));
-  TriggerUnderuse();
-}
-
-TEST_F(OveruseFrameDetectorTest, TriggerUnderuseWithMinProcessCount) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  CpuOveruseObserverImpl overuse_observer_;
-  overuse_detector_->SetObserver(&overuse_observer_);
-  options_.min_process_count = 1;
-  overuse_detector_->SetOptions(options_);
-  InsertFramesWithInterval(1200, kFrameInterval33ms, kWidth, kHeight);
-  overuse_detector_->Process();
-  EXPECT_EQ(0, overuse_observer_.normaluse_);
-  clock_->AdvanceTimeMilliseconds(kProcessIntervalMs);
-  overuse_detector_->Process();
-  EXPECT_EQ(1, overuse_observer_.normaluse_);
-}
-
-TEST_F(OveruseFrameDetectorTest, ConstantOveruseGivesNoNormalUsage) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(64);
-  for(size_t i = 0; i < 64; ++i) {
-    TriggerOveruse(options_.high_threshold_consecutive_count);
-  }
-}
-
-TEST_F(OveruseFrameDetectorTest, ConsecutiveCountTriggersOveruse) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  options_.high_threshold_consecutive_count = 2;
-  overuse_detector_->SetOptions(options_);
-  TriggerOveruse(2);
-}
-
-TEST_F(OveruseFrameDetectorTest, IncorrectConsecutiveCountTriggersNoOveruse) {
-  options_.enable_capture_jitter_method = true;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
-  options_.high_threshold_consecutive_count = 2;
-  overuse_detector_->SetOptions(options_);
-  TriggerOveruse(1);
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureJitter) {
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-  InsertFramesWithInterval(1000, kFrameInterval33ms, kWidth, kHeight);
-  EXPECT_NE(InitialJitter(), CaptureJitterMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureJitterResetAfterResolutionChange) {
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-  InsertFramesWithInterval(1000, kFrameInterval33ms, kWidth, kHeight);
-  EXPECT_NE(InitialJitter(), CaptureJitterMs());
-  // Verify reset.
-  InsertFramesWithInterval(1, kFrameInterval33ms, kWidth, kHeight + 1);
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureJitterResetAfterFrameTimeout) {
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-  InsertFramesWithInterval(1000, kFrameInterval33ms, kWidth, kHeight);
-  EXPECT_NE(InitialJitter(), CaptureJitterMs());
-  InsertFramesWithInterval(
-      1, options_.frame_timeout_interval_ms, kWidth, kHeight);
-  EXPECT_NE(InitialJitter(), CaptureJitterMs());
-  // Verify reset.
-  InsertFramesWithInterval(
-      1, options_.frame_timeout_interval_ms + 1, kWidth, kHeight);
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureJitterResetAfterChangingThreshold) {
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-  options_.high_capture_jitter_threshold_ms = 90.0f;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-  options_.low_capture_jitter_threshold_ms = 30.0f;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, MinFrameSamplesBeforeUpdatingCaptureJitter) {
-  options_.min_frame_samples = 40;
-  overuse_detector_->SetOptions(options_);
-  InsertFramesWithInterval(40, kFrameInterval33ms, kWidth, kHeight);
-  EXPECT_EQ(InitialJitter(), CaptureJitterMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, NoCaptureQueueDelay) {
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 0);
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight, clock_->TimeInMilliseconds());
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 0);
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureQueueDelay) {
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight, clock_->TimeInMilliseconds());
-  clock_->AdvanceTimeMilliseconds(100);
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 100);
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureQueueDelayMultipleFrames) {
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight, clock_->TimeInMilliseconds());
-  clock_->AdvanceTimeMilliseconds(10);
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight, clock_->TimeInMilliseconds());
-  clock_->AdvanceTimeMilliseconds(20);
-
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 30);
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 20);
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureQueueDelayResetAtResolutionSwitch) {
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight, clock_->TimeInMilliseconds());
-  clock_->AdvanceTimeMilliseconds(10);
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight + 1, clock_->TimeInMilliseconds());
-  clock_->AdvanceTimeMilliseconds(20);
-
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 20);
-}
-
-TEST_F(OveruseFrameDetectorTest, CaptureQueueDelayNoMatchingCapturedFrame) {
-  overuse_detector_->FrameCaptured(
-      kWidth, kHeight, clock_->TimeInMilliseconds());
-  clock_->AdvanceTimeMilliseconds(100);
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 100);
-  // No new captured frame. The last delay should be reported.
-  overuse_detector_->FrameProcessingStarted();
-  EXPECT_EQ(overuse_detector_->CaptureQueueDelayMsPerS(), 100);
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_OneFrameDisabled) {
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  const int kProcessingTimeMs = 100;
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  overuse_detector_->FrameSent(33);
-  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_OneFrame) {
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  const int kProcessingTimeMs = 100;
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
-  overuse_detector_->FrameSent(33);
-  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
-  EXPECT_EQ(0, overuse_detector_->FramesInQueue());
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_TwoFrames) {
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  const int kProcessingTimeMs1 = 100;
-  const int kProcessingTimeMs2 = 50;
-  const int kTimeBetweenFramesMs = 200;
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs1);
-  overuse_detector_->FrameSent(33);
-  EXPECT_EQ(kProcessingTimeMs1, overuse_detector_->LastProcessingTimeMs());
-  clock_->AdvanceTimeMilliseconds(kTimeBetweenFramesMs);
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 66);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs2);
-  overuse_detector_->FrameSent(66);
-  EXPECT_EQ(kProcessingTimeMs2, overuse_detector_->LastProcessingTimeMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_MaxQueueSize) {
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  const int kMaxQueueSize = 91;
-  for (int i = 0; i < kMaxQueueSize * 2; ++i) {
-    overuse_detector_->FrameCaptured(kWidth, kHeight, i);
-  }
-  EXPECT_EQ(kMaxQueueSize, overuse_detector_->FramesInQueue());
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_NonProcessedFramesRemoved) {
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  const int kProcessingTimeMs = 100;
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 35);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 66);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 99);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
-  EXPECT_EQ(4, overuse_detector_->FramesInQueue());
-  overuse_detector_->FrameSent(66);
-  // Frame 33, 35 removed, 66 processed, 99 not processed.
-  EXPECT_EQ(2 * kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
-  EXPECT_EQ(1, overuse_detector_->FramesInQueue());
-  overuse_detector_->FrameSent(99);
-  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
-  EXPECT_EQ(0, overuse_detector_->FramesInQueue());
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_ResetClearsFrames) {
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  const int kProcessingTimeMs = 100;
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
-  EXPECT_EQ(1, overuse_detector_->FramesInQueue());
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  // Verify reset (resolution changed).
-  overuse_detector_->FrameCaptured(kWidth, kHeight + 1, 66);
-  EXPECT_EQ(1, overuse_detector_->FramesInQueue());
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  overuse_detector_->FrameSent(66);
-  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
-  EXPECT_EQ(0, overuse_detector_->FramesInQueue());
-}
-
-TEST_F(OveruseFrameDetectorTest, FrameDelay_NonMatchingSendFrameIgnored) {
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  const int kProcessingTimeMs = 100;
-  overuse_detector_->FrameCaptured(kWidth, kHeight, 33);
-  clock_->AdvanceTimeMilliseconds(kProcessingTimeMs);
-  overuse_detector_->FrameSent(34);
-  EXPECT_EQ(-1, overuse_detector_->LastProcessingTimeMs());
-  overuse_detector_->FrameSent(33);
-  EXPECT_EQ(kProcessingTimeMs, overuse_detector_->LastProcessingTimeMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, EncodedFrame) {
-  const int kInitialAvgEncodeTimeInMs = 5;
-  EXPECT_EQ(kInitialAvgEncodeTimeInMs, AvgEncodeTimeMs());
-  for (int i = 0; i < 30; i++) {
-    clock_->AdvanceTimeMilliseconds(33);
-    overuse_detector_->FrameEncoded(2);
-  }
-  EXPECT_EQ(2, AvgEncodeTimeMs());
-}
-
-TEST_F(OveruseFrameDetectorTest, InitialProcessingUsage) {
-  EXPECT_EQ(InitialUsage(), UsagePercent());
-}
-
-TEST_F(OveruseFrameDetectorTest, ProcessingUsage) {
-  const int kProcessingTimeMs = 5;
-  InsertAndSendFramesWithInterval(
-      1000, kFrameInterval33ms, kWidth, kHeight, kProcessingTimeMs);
-  EXPECT_EQ(kProcessingTimeMs * 100 / kFrameInterval33ms, UsagePercent());
-}
-
-TEST_F(OveruseFrameDetectorTest, ProcessingUsageResetAfterChangingThreshold) {
-  EXPECT_EQ(InitialUsage(), UsagePercent());
-  options_.high_encode_usage_threshold_percent = 100;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_EQ(InitialUsage(), UsagePercent());
-  options_.low_encode_usage_threshold_percent = 20;
-  overuse_detector_->SetOptions(options_);
-  EXPECT_EQ(InitialUsage(), UsagePercent());
-}
-
-// enable_encode_usage_method = true;
-// UsagePercent() > high_encode_usage_threshold_percent => overuse.
-// UsagePercent() < low_encode_usage_threshold_percent => underuse.
-TEST_F(OveruseFrameDetectorTest, TriggerOveruseWithProcessingUsage) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = true;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  // usage > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  TriggerOveruseWithProcessingUsage(options_.high_threshold_consecutive_count);
-}
-
-TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithProcessingUsage) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = true;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  // usage > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  TriggerOveruseWithProcessingUsage(options_.high_threshold_consecutive_count);
-  // usage < low => underuse
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(testing::AtLeast(1));
-  TriggerUnderuseWithProcessingUsage();
-}
-
-TEST_F(OveruseFrameDetectorTest,
-       OveruseAndRecoverWithProcessingUsageMethodDisabled) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = false;
-  overuse_detector_->SetOptions(options_);
-  // usage > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
-  TriggerOveruseWithProcessingUsage(options_.high_threshold_consecutive_count);
-  // usage < low => underuse
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
-  TriggerUnderuseWithProcessingUsage();
-}
-
-// enable_extended_processing_usage = true;
-// enable_encode_usage_method = true;
-// UsagePercent() > high_encode_usage_threshold_percent => overuse.
-// UsagePercent() < low_encode_usage_threshold_percent => underuse.
-TEST_F(OveruseFrameDetectorTest, TriggerOveruseWithExtendedProcessingUsage) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = true;
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  // usage > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  TriggerOveruseWithProcessingUsage(options_.high_threshold_consecutive_count);
-}
-
-TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithExtendedProcessingUsage) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = true;
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  // usage > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
-  TriggerOveruseWithProcessingUsage(options_.high_threshold_consecutive_count);
-  // usage < low => underuse
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(testing::AtLeast(1));
-  TriggerUnderuseWithProcessingUsage();
-}
-
-TEST_F(OveruseFrameDetectorTest,
-       OveruseAndRecoverWithExtendedProcessingUsageMethodDisabled) {
-  options_.enable_capture_jitter_method = false;
-  options_.enable_encode_usage_method = false;
-  options_.enable_extended_processing_usage = true;
-  overuse_detector_->SetOptions(options_);
-  // usage > high => overuse
-  EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
-  TriggerOveruseWithProcessingUsage(options_.high_threshold_consecutive_count);
-  // usage < low => underuse
-  EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
-  TriggerUnderuseWithProcessingUsage();
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/OWNERS b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/OWNERS
deleted file mode 100644
index bbffda7e49..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/OWNERS
+++ /dev/null
@@ -1,6 +0,0 @@
-per-file *.isolate=kjellander@webrtc.org
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/.classpath b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/.classpath
deleted file mode 100644
index e130523456..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/.classpath
+++ /dev/null
@@ -1,9 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<classpath>
-	<classpathentry kind="src" path="src"/>
-	<classpathentry kind="src" path="gen"/>
-	<classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
-	<classpathentry kind="lib" path="libs/VideoEngine_android_java.jar"/>
-	<classpathentry kind="lib" path="libs/VoiceEngine_android_java.jar"/>
-	<classpathentry kind="output" path="bin"/>
-</classpath>
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/AndroidManifest.xml b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/AndroidManifest.xml
deleted file mode 100644
index 11b3e27d92..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/AndroidManifest.xml
+++ /dev/null
@@ -1,26 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
-	  android:versionCode="1"
-	  android:versionName="1.0" package="org.webrtc.vieautotest">
-  <application android:label="@string/app_name"
-	       android:debuggable="true"
-	       android:icon="@drawable/logo">
-    <activity android:label="@string/app_name"
-	      android:name="ViEAutotest">
-      <intent-filter>
-        <action android:name="android.intent.action.MAIN" />
-        <category android:name="android.intent.category.LAUNCHER" />
-      </intent-filter>
-    </activity>
-
-  </application>
-  <uses-sdk android:minSdkVersion="3" android:targetSdkVersion="8" />
-  <uses-permission android:name="android.permission.CAMERA"></uses-permission>
-  <uses-feature android:name="android.hardware.camera" />
-  <uses-feature android:name="android.hardware.camera.autofocus" />
-  <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
-  <uses-permission android:name="android.permission.RECORD_AUDIO" />
-  <uses-permission android:name="android.permission.INTERNET" />
-  <uses-permission android:name="android.permission.WAKE_LOCK" />
-  <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
-</manifest> 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/default.properties b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/default.properties
deleted file mode 100644
index 9a2c9f6c88..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-# 
-# This file must be checked in Version Control Systems.
-# 
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target.
-target=android-9
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java
deleted file mode 100644
index 5e1908c32a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java
+++ /dev/null
@@ -1,37 +0,0 @@
-/* AUTO-GENERATED FILE.  DO NOT MODIFY.
- *
- * This class was automatically generated by the
- * aapt tool from the resource data it found.  It
- * should not be modified by hand.
- */
-
-package org.webrtc.vieautotest;
-
-public final class R {
-    public static final class array {
-        public static final int subtest_array=0x7f050001;
-        public static final int test_array=0x7f050000;
-    }
-    public static final class attr {
-    }
-    public static final class drawable {
-        public static final int logo=0x7f020000;
-    }
-    public static final class id {
-        public static final int Button01=0x7f060004;
-        public static final int LocalView=0x7f060001;
-        public static final int RemoteView=0x7f060000;
-        public static final int subtestSpinner=0x7f060003;
-        public static final int testSpinner=0x7f060002;
-    }
-    public static final class layout {
-        public static final int main=0x7f030000;
-    }
-    public static final class string {
-        public static final int SpinnerSubtest=0x7f040004;
-        public static final int SpinnerTitle=0x7f040003;
-        public static final int TitleName=0x7f040001;
-        public static final int app_name=0x7f040000;
-        public static final int run_button=0x7f040002;
-    }
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h
deleted file mode 100644
index 68ec6014b5..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/* DO NOT EDIT THIS FILE - it is machine generated */
-#include <jni.h>
-/* Header for class org_webrtc_vieautotest_ViEAutotest */
-
-#ifndef _Included_org_webrtc_vieautotest_ViEAutotest
-#define _Included_org_webrtc_vieautotest_ViEAutotest
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-/*
- * Class:     org_webrtc_vieautotest_ViEAutotest
- * Method:    RunTest
- * Signature: (IILandroid/view/SurfaceView;Landroid/view/SurfaceView;)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_view_SurfaceView_2Landroid_view_SurfaceView_2
-(JNIEnv *, jobject, jint, jint, jobject, jobject);
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc
deleted file mode 100644
index 657db863f4..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <android/log.h>
-#include <pthread.h>
-#include <stdio.h>
-#include <string.h>
-#include <unistd.h>
-
-#include "webrtc/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_android.h"
-
-#define WEBRTC_LOG_TAG "*WEBRTCN*"
-
-// VideoEngine data struct
-typedef struct
-{
-    JavaVM* jvm;
-} VideoEngineData;
-
-// Global variables
-JavaVM* webrtcGlobalVM;
-
-// Global variables visible in this file
-static VideoEngineData vieData;
-
-// "Local" functions (i.e. not Java accessible)
-#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 1024
-
-static bool GetSubAPIs(VideoEngineData& vieData);
-static bool ReleaseSubAPIs(VideoEngineData& vieData);
-
-//
-// General functions
-//
-
-// JNI_OnLoad
-jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/) {
-  if (!vm) {
-    __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-                        "JNI_OnLoad did not receive a valid VM pointer");
-    return -1;
-  }
-
-  JNIEnv* env;
-  if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
-                           JNI_VERSION_1_4)) {
-    __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-                        "JNI_OnLoad could not get JNI env");
-    return -1;
-  }
-
-  // Init ViE data
-  vieData.jvm = vm;
-
-  return JNI_VERSION_1_4;
-}
-
-// Class:     org_webrtc_vieautotest_ViEAutotest
-// Method:    RunTest
-// Signature: (IILandroid/opengl/GLSurfaceView;Landroid/opengl/GLSurfaceView;)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_opengl_GLSurfaceView_2Landroid_opengl_GLSurfaceView_2(
-    JNIEnv* env,
-    jobject context,
-    jint testType,
-    jint subtestType,
-    jobject glView1,
-    jobject glView2)
-{
-  int numErrors = -1;
-  numErrors = ViEAutoTestAndroid::RunAutotest(testType, subtestType, glView1,
-                                              glView2, vieData.jvm, env,
-                                              context);
-  return numErrors;
-}
-
-// Class:     org_webrtc_vieautotest_ViEAutotest
-// Method:    RunTest
-// Signature: (IILandroid/view/SurfaceView;Landroid/view/SurfaceView;)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_view_SurfaceView_2Landroid_view_SurfaceView_2(
-    JNIEnv* env,
-    jobject context,
-    jint testType,
-    jint subtestType,
-    jobject surfaceHolder1,
-    jobject surfaceHolder2)
-{
-  int numErrors = -1;
-  numErrors = ViEAutoTestAndroid::RunAutotest(testType, subtestType,
-                                              surfaceHolder1, surfaceHolder2,
-                                              vieData.jvm, env, context);
-  return numErrors;
-}
-
-//
-//local function
-//
-
-bool GetSubAPIs(VideoEngineData& vieData) {
-  bool retVal = true;
-  //vieData.base = ViEBase::GetInterface(vieData.vie);
-  //if (vieData.base == NULL)
-  {
-    __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-                        "Could not get Base API");
-    retVal = false;
-  }
-  return retVal;
-}
-
-bool ReleaseSubAPIs(VideoEngineData& vieData) {
-  bool releaseOk = true;
-  //if (vieData.base)
-  {
-    //if (vieData.base->Release() != 0)
-    if (false) {
-      __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-                          "Release base sub-API failed");
-      releaseOk = false;
-    }
-    else {
-      //vieData.base = NULL;
-    }
-  }
-
-  return releaseOk;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/drawable/logo.png b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/drawable/logo.png
deleted file mode 100644
index c3e0a123b5..0000000000
Binary files a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/drawable/logo.png and /dev/null differ
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/layout/main.xml b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/layout/main.xml
deleted file mode 100644
index 1f2aaf9e75..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/layout/main.xml
+++ /dev/null
@@ -1,64 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<FrameLayout xmlns:android="http://schemas.android.com/apk/res/android"
-	     android:layout_width="fill_parent"
-	     android:layout_height="fill_parent">
-  <RelativeLayout
-     android:layout_width="fill_parent"
-     android:layout_height="fill_parent">
-    <LinearLayout 
-       android:id="@+id/RemoteView"
-       android:layout_width="fill_parent"
-       android:layout_height="fill_parent"
-       android:layout_weight="1">
-      <!-- log instead of video
-           <ImageView
-              android:layout_width="fill_parent"
-              android:layout_height="fill_parent" 
-              android:scaleType="fitXY"
-              android:src="@drawable/logo" /> -->
-
-    </LinearLayout>
-    <LinearLayout 
-       android:id="@+id/LocalView"
-       android:layout_width="120dip"
-       android:layout_height="120dip"
-       android:layout_weight="1">
-      <!-- <ImageView
-       	      android:layout_width="fill_parent"
-       	      android:layout_height="fill_parent" 
-       	      android:scaleType="fitXY"
-       	      android:src="@drawable/logo" /> -->
-    </LinearLayout>
-    <LinearLayout
-       android:orientation="horizontal"
-       android:layout_width="fill_parent"
-       android:layout_height="wrap_content"
-       android:layout_alignParentBottom="true">
-      <LinearLayout
-	 android:orientation="vertical"
-	 android:layout_width="fill_parent"
-	 android:layout_height="wrap_content"
-	 android:layout_alignParentBottom="true">
-	<Spinner
-	   android:id="@+id/testSpinner"
-	   android:layout_width="wrap_content"
-	   android:layout_height="wrap_content"
-	   android:prompt="@string/SpinnerTitle"
-	   />
-	<Spinner
-	   android:id="@+id/subtestSpinner"
-	   android:layout_width="wrap_content"
-	   android:layout_height="wrap_content"
-	   android:prompt="@string/SpinnerSubtest"
-	   />
-	<Button
-	   android:text="@string/run_button"
-	   android:id="@+id/Button01"
-	   android:layout_width="wrap_content"
-	   android:layout_height="wrap_content">
-	</Button>
-      </LinearLayout>
-
-    </LinearLayout>
-  </RelativeLayout>
-</FrameLayout>
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/values/strings.xml b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/values/strings.xml
deleted file mode 100644
index 48cfe926e6..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/res/values/strings.xml
+++ /dev/null
@@ -1,30 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
-
-<string name="app_name">ViEAutotest</string>
-<string name="TitleName">ViEAutotest</string>
-<string name="run_button">Run Test</string>
-<string name="SpinnerTitle">Test type...</string>
-<string-array name="test_array">
-	<item>Standard</item>
-	<item>API</item>
-	<item>Extended</item>
-	<item>Loopback</item>
-	<item>Custom</item>
-</string-array>
-<string name="SpinnerSubtest">Run...</string>
-<string-array name="subtest_array">
-	<item>All</item>
-	<item>Base</item>
-	<item>Capture</item>
-	<item>Codec</item>
-	<item>Mix</item>
-	<item>External Codec</item>
-	<item>File</item>
-	<item>Image Process</item>
-	<item>Network</item>
-	<item>Render</item>
-	<item>RTP/RTCP</item>
-</string-array>
-
-</resources>
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java
deleted file mode 100644
index de228a8768..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.vieautotest;
-
-import org.webrtc.vieautotest.R;
-
-import android.app.Activity;
-import android.os.Bundle;
-import android.util.Log;
-import android.widget.Button;
-import android.view.SurfaceView;
-import android.view.View;
-import android.view.SurfaceHolder;
-import android.widget.LinearLayout;
-import android.opengl.GLSurfaceView;
-import android.widget.Spinner;
-import android.widget.ArrayAdapter;
-import android.widget.AdapterView;
-
-public class ViEAutotest extends Activity
-    implements
-      AdapterView.OnItemSelectedListener,
-      View.OnClickListener {
-
-  private Thread testThread;
-  private Spinner testSpinner;
-  private Spinner subtestSpinner;
-  private int testSelection;
-  private int subTestSelection;
-
-  // View for remote video
-  private LinearLayout remoteSurface = null;
-  private GLSurfaceView glSurfaceView = null;
-  private SurfaceView surfaceView = null;
-
-  private LinearLayout localSurface = null;
-  private GLSurfaceView glLocalSurfaceView = null;
-  private SurfaceView localSurfaceView = null;
-
-  /** Called when the activity is first created. */
-  @Override
-  public void onCreate(Bundle savedInstanceState) {
-
-    Log.d("*WEBRTC*", "onCreate called");
-
-    super.onCreate(savedInstanceState);
-    setContentView(R.layout.main);
-
-    // Set the Start button action
-    final Button buttonStart = (Button) findViewById(R.id.Button01);
-    buttonStart.setOnClickListener(this);
-
-    // Set test spinner
-    testSpinner = (Spinner) findViewById(R.id.testSpinner);
-    ArrayAdapter<CharSequence> adapter =
-        ArrayAdapter.createFromResource(this, R.array.test_array,
-                                        android.R.layout.simple_spinner_item);
-
-    int resource = android.R.layout.simple_spinner_dropdown_item;
-    adapter.setDropDownViewResource(resource);
-    testSpinner.setAdapter(adapter);
-    testSpinner.setOnItemSelectedListener(this);
-
-    // Set sub test spinner
-    subtestSpinner = (Spinner) findViewById(R.id.subtestSpinner);
-    ArrayAdapter<CharSequence> subtestAdapter =
-        ArrayAdapter.createFromResource(this, R.array.subtest_array,
-                                        android.R.layout.simple_spinner_item);
-
-    subtestAdapter.setDropDownViewResource(resource);
-    subtestSpinner.setAdapter(subtestAdapter);
-    subtestSpinner.setOnItemSelectedListener(this);
-
-    remoteSurface = (LinearLayout) findViewById(R.id.RemoteView);
-    surfaceView = new SurfaceView(this);
-    remoteSurface.addView(surfaceView);
-
-    localSurface = (LinearLayout) findViewById(R.id.LocalView);
-    localSurfaceView = new SurfaceView(this);
-    localSurfaceView.setZOrderMediaOverlay(true);
-    localSurface.addView(localSurfaceView);
-
-    // Set members
-    testSelection = 0;
-    subTestSelection = 0;
-  }
-
-  public void onClick(View v) {
-    Log.d("*WEBRTC*", "Button clicked...");
-    switch (v.getId()) {
-      case R.id.Button01:
-        new Thread(new Runnable() {
-            public void run() {
-              Log.d("*WEBRTC*", "Calling RunTest...");
-              RunTest(testSelection, subTestSelection,
-                      localSurfaceView, surfaceView);
-              Log.d("*WEBRTC*", "RunTest done");
-            }
-          }).start();
-    }
-  }
-
-  public void onItemSelected(AdapterView<?> parent, View v,
-                             int position, long id) {
-
-    if (parent == (Spinner) findViewById(R.id.testSpinner)) {
-      testSelection = position;
-    } else {
-      subTestSelection = position;
-    }
-  }
-
-  public void onNothingSelected(AdapterView<?> parent) {
-  }
-
-  @Override
-  protected void onStart() {
-    super.onStart();
-  }
-
-  @Override
-  protected void onResume() {
-    super.onResume();
-  }
-
-  @Override
-  protected void onPause() {
-    super.onPause();
-  }
-
-  @Override
-  protected void onStop() {
-    super.onStop();
-  }
-
-  @Override
-  protected void onDestroy() {
-
-    super.onDestroy();
-  }
-
-  // C++ function performing the chosen test
-  // private native int RunTest(int testSelection, int subtestSelection,
-  // GLSurfaceView window1, GLSurfaceView window2);
-  private native int RunTest(int testSelection, int subtestSelection,
-                             SurfaceView window1, SurfaceView window2);
-
-  // this is used to load the 'ViEAutotestJNIAPI' library on application
-  // startup.
-  static {
-    Log.d("*WEBRTC*", "Loading ViEAutotest...");
-    System.loadLibrary("webrtc-video-autotest-jni");
-  }
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/legacy_fixture.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/legacy_fixture.cc
deleted file mode 100644
index 08c49d8d6e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/legacy_fixture.cc
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/automated/legacy_fixture.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-
-void LegacyFixture::SetUpTestCase() {
-  TwoWindowsFixture::SetUpTestCase();
-
-  // Create the test cases
-  tests_ = new ViEAutoTest(window_1_, window_2_);
-}
-
-void LegacyFixture::TearDownTestCase() {
-  delete tests_;
-
-  TwoWindowsFixture::TearDownTestCase();
-}
-
-ViEAutoTest* LegacyFixture::tests_ = NULL;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/legacy_fixture.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/legacy_fixture.h
deleted file mode 100644
index 2386f4ba6c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/legacy_fixture.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
-
-#include "webrtc/video_engine/test/auto_test/automated/two_windows_fixture.h"
-
-// Inherited by old-style standard integration tests based on ViEAutoTest.
-class LegacyFixture : public TwoWindowsFixture {
- public:
-  // Initializes ViEAutoTest in addition to the work done by ViEIntegrationTest.
-  static void SetUpTestCase();
-
-  // Releases anything allocated by SetupTestCase.
-  static void TearDownTestCase();
-
- protected:
-  static ViEAutoTest* tests_;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/two_windows_fixture.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/two_windows_fixture.cc
deleted file mode 100644
index d190920d75..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/two_windows_fixture.cc
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/automated/two_windows_fixture.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_creator.h"
-
-void TwoWindowsFixture::SetUpTestCase() {
-  window_creator_ = new ViEWindowCreator();
-
-  ViEAutoTestWindowManagerInterface* window_manager =
-    window_creator_->CreateTwoWindows();
-
-  window_1_ = window_manager->GetWindow1();
-  window_2_ = window_manager->GetWindow2();
-}
-
-void TwoWindowsFixture::TearDownTestCase() {
-  window_creator_->TerminateWindows();
-  delete window_creator_;
-}
-
-ViEWindowCreator* TwoWindowsFixture::window_creator_ = NULL;
-void* TwoWindowsFixture::window_1_ = NULL;
-void* TwoWindowsFixture::window_2_ = NULL;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/two_windows_fixture.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/two_windows_fixture.h
deleted file mode 100644
index 44ef257e2a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/two_windows_fixture.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-class ViEWindowCreator;
-class ViEAutoTest;
-
-// Meant to be inherited by all standard test who require two windows.
-class TwoWindowsFixture : public testing::Test {
- public:
-  // Launches two windows in a platform-dependent manner and stores the handles
-  // in the window_1_ and window_2_ fields.
-  static void SetUpTestCase();
-
-  // Releases anything allocated by SetupTestCase.
-  static void TearDownTestCase();
-
- protected:
-  static void* window_1_;
-  static void* window_2_;
-  static ViEWindowCreator* window_creator_;
-};
-
-#endif  // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_api_integration_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_api_integration_test.cc
deleted file mode 100644
index 3067a80bc0..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_api_integration_test.cc
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-#include "webrtc/video_engine/test/auto_test/automated/legacy_fixture.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-
-namespace {
-
-// TODO(phoglund): These tests are generally broken on mac.
-// http://code.google.com/p/webrtc/issues/detail?id=1268
-class DISABLED_ON_MAC(ViEApiIntegrationTest) : public LegacyFixture {
-};
-
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest), RunsBaseTestWithoutErrors) {
-  tests_->ViEBaseAPITest();
-}
-
-// TODO(phoglund): Crashes on the v4l2loopback camera.
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest),
-       DISABLED_RunsCaptureTestWithoutErrors) {
-  tests_->ViECaptureAPITest();
-}
-
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest), RunsCodecTestWithoutErrors) {
-  tests_->ViECodecAPITest();
-}
-
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest),
-       RunsImageProcessTestWithoutErrors) {
-  tests_->ViEImageProcessAPITest();
-}
-
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest), RunsRenderTestWithoutErrors) {
-  tests_->ViERenderAPITest();
-}
-
-// See: https://code.google.com/p/webrtc/issues/detail?id=2415
-TEST_F(DISABLED_ON_MAC(ViEApiIntegrationTest),
-       DISABLED_RunsRtpRtcpTestWithoutErrors) {
-  tests_->ViERtpRtcpAPITest();
-}
-
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
deleted file mode 100644
index 4508d8e791..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-#include "webrtc/video_engine/test/auto_test/automated/legacy_fixture.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-
-namespace {
-
-// TODO(phoglund): These tests are generally broken on mac.
-// http://code.google.com/p/webrtc/issues/detail?id=1268
-class DISABLED_ON_MAC(ViEExtendedIntegrationTest) : public LegacyFixture {
-};
-
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest), RunsBaseTestWithoutErrors) {
-  tests_->ViEBaseExtendedTest();
-}
-
-// TODO(phoglund): Crashes on the v4l2loopback camera.
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
-       DISABLED_RunsCaptureTestWithoutErrors) {
-  tests_->ViECaptureExtendedTest();
-}
-
-// Flaky on Windows: http://code.google.com/p/webrtc/issues/detail?id=1925
-// (in addition to being disabled on Mac due to webrtc:1268).
-#if defined(_WIN32)
-#define MAYBE_RunsCodecTestWithoutErrors DISABLED_RunsCodecTestWithoutErrors
-#else
-#define MAYBE_RunsCodecTestWithoutErrors RunsCodecTestWithoutErrors
-#endif
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
-       MAYBE_RunsCodecTestWithoutErrors) {
-  tests_->ViECodecExtendedTest();
-}
-
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
-       RunsImageProcessTestWithoutErrors) {
-  tests_->ViEImageProcessExtendedTest();
-}
-
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
-       RunsRenderTestWithoutErrors) {
-  tests_->ViERenderExtendedTest();
-}
-
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc
deleted file mode 100644
index 4fd2422a9c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc
+++ /dev/null
@@ -1,228 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-
-namespace {
-
-class RtcpCollectorTransport : public webrtc::Transport {
- public:
-  RtcpCollectorTransport() : packets_() {}
-  virtual ~RtcpCollectorTransport() {}
-
-  int SendPacket(int /*channel*/,
-                 const void* /*data*/,
-                 size_t /*len*/) override {
-    EXPECT_TRUE(false);
-    return 0;
-  }
-  int SendRTCPPacket(int channel, const void* data, size_t len) override {
-    const uint8_t* buf = static_cast<const uint8_t*>(data);
-    webrtc::RtpUtility::RtpHeaderParser parser(buf, len);
-    if (parser.RTCP()) {
-      Packet p;
-      p.channel = channel;
-      p.length = len;
-      if (parser.ParseRtcp(&p.header)) {
-        if (p.header.payloadType == 201 && len >= 20) {
-          buf += 20;
-          len -= 20;
-        } else {
-          return 0;
-        }
-        if (TryParseREMB(buf, len, &p)) {
-          packets_.push_back(p);
-        }
-      }
-    }
-    return 0;
-  }
-
-  bool FindREMBFor(uint32_t ssrc, double min_rate) const {
-    for (std::vector<Packet>::const_iterator it = packets_.begin();
-        it != packets_.end(); ++it) {
-      if (it->remb_bitrate >= min_rate && it->remb_ssrc.end() !=
-          std::find(it->remb_ssrc.begin(), it->remb_ssrc.end(), ssrc)) {
-        return true;
-      }
-    }
-    return false;
-  }
-
- private:
-  struct Packet {
-    Packet() : channel(-1), length(0), header(), remb_bitrate(0), remb_ssrc() {}
-    int channel;
-    size_t length;
-    webrtc::RTPHeader header;
-    double remb_bitrate;
-    std::vector<uint32_t> remb_ssrc;
-  };
-
-  bool TryParseREMB(const uint8_t* buf, size_t length, Packet* p) {
-    if (length < 8) {
-      return false;
-    }
-    if (buf[0] != 'R' || buf[1] != 'E' || buf[2] != 'M' || buf[3] != 'B') {
-      return false;
-    }
-    size_t ssrcs = buf[4];
-    uint8_t exp = buf[5] >> 2;
-    uint32_t mantissa = ((buf[5] & 0x03) << 16) + (buf[6] << 8) + buf[7];
-    double bitrate = mantissa * static_cast<double>(1 << exp);
-    p->remb_bitrate = bitrate;
-
-    if (length < (8 + 4 * ssrcs)) {
-      return false;
-    }
-    buf += 8;
-    for (size_t i = 0; i < ssrcs; ++i) {
-      uint32_t ssrc = (buf[0] << 24) + (buf[1] << 16) + (buf[2] << 8) + buf[3];
-      p->remb_ssrc.push_back(ssrc);
-      buf += 4;
-    }
-    return true;
-  }
-
-  std::vector<Packet> packets_;
-};
-
-class ViENetworkTest : public testing::Test {
- protected:
-  ViENetworkTest() : vie_("ViENetworkTest"), channel_(-1), transport() {}
-  virtual ~ViENetworkTest() {}
-
-  void SetUp() override {
-    EXPECT_EQ(0, vie_.base->CreateChannel(channel_));
-    EXPECT_EQ(0, vie_.rtp_rtcp->SetRembStatus(channel_, false, true));
-    EXPECT_EQ(0, vie_.network->RegisterSendTransport(channel_, transport));
-  }
-
-  void TearDown() override {
-    EXPECT_EQ(0, vie_.network->DeregisterSendTransport(channel_));
-  }
-
-  void ReceiveASTPacketsForBWE() {
-    for (int i = 0; i < kPacketCount; ++i) {
-      int64_t time = webrtc::TickTime::MillisecondTimestamp();
-      webrtc::RTPHeader header;
-      header.ssrc = kSsrc1;
-      header.timestamp = i * 45000;
-      header.extension.hasAbsoluteSendTime = true;
-      header.extension.absoluteSendTime = i << (18 - 6);
-      EXPECT_EQ(0, vie_.network->ReceivedBWEPacket(channel_, time, kPacketSize,
-                                                   header));
-      webrtc::SleepMs(kIntervalMs);
-    }
-  }
-
-  enum {
-    kSsrc1 = 667,
-    kSsrc2 = 668,
-    kPacketCount = 100,
-    kPacketSize = 1000,
-    kIntervalMs = 22
-  };
-  TbInterfaces vie_;
-  int channel_;
-  RtcpCollectorTransport transport;
-};
-
-TEST_F(ViENetworkTest, ReceiveBWEPacket_NoExtension)  {
-  for (int i = 0; i < kPacketCount; ++i) {
-    int64_t time = webrtc::TickTime::MillisecondTimestamp();
-    webrtc::RTPHeader header;
-    header.ssrc = kSsrc1;
-    header.timestamp = i * 45000;
-    EXPECT_EQ(0, vie_.network->ReceivedBWEPacket(channel_, time, kPacketSize,
-                                                 header));
-    webrtc::SleepMs(kIntervalMs);
-  }
-  EXPECT_FALSE(transport.FindREMBFor(kSsrc1, 0.0));
-  unsigned int bandwidth = 0;
-  EXPECT_EQ(0, vie_.rtp_rtcp->GetEstimatedReceiveBandwidth(channel_,
-                                                           &bandwidth));
-}
-
-TEST_F(ViENetworkTest, ReceiveBWEPacket_TOF)  {
-  EXPECT_EQ(0, vie_.rtp_rtcp->SetReceiveTimestampOffsetStatus(channel_, true,
-                                                              1));
-  for (int i = 0; i < kPacketCount; ++i) {
-    int64_t time = webrtc::TickTime::MillisecondTimestamp();
-    webrtc::RTPHeader header;
-    header.ssrc = kSsrc1;
-    header.timestamp = i * 45000;
-    header.extension.hasTransmissionTimeOffset = true;
-    header.extension.transmissionTimeOffset = 17;
-    EXPECT_EQ(0, vie_.network->ReceivedBWEPacket(channel_, time, kPacketSize,
-                                                 header));
-    webrtc::SleepMs(kIntervalMs);
-  }
-  EXPECT_FALSE(transport.FindREMBFor(kSsrc1, 0.0));
-  unsigned int bandwidth = 0;
-  EXPECT_EQ(0, vie_.rtp_rtcp->GetEstimatedReceiveBandwidth(channel_,
-                                                           &bandwidth));
-}
-
-TEST_F(ViENetworkTest, ReceiveBWEPacket_AST)  {
-  EXPECT_EQ(0, vie_.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(channel_, true,
-                                                               1));
-  ReceiveASTPacketsForBWE();
-  EXPECT_TRUE(transport.FindREMBFor(kSsrc1, 100000.0));
-  unsigned int bandwidth = 0;
-  EXPECT_EQ(0, vie_.rtp_rtcp->GetEstimatedReceiveBandwidth(channel_,
-                                                           &bandwidth));
-  EXPECT_GT(bandwidth, 0u);
-}
-
-TEST_F(ViENetworkTest, ReceiveBWEPacket_ASTx2)  {
-  EXPECT_EQ(0, vie_.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(channel_, true,
-                                                               1));
-  for (int i = 0; i < kPacketCount; ++i) {
-    int64_t time = webrtc::TickTime::MillisecondTimestamp();
-    webrtc::RTPHeader header;
-    header.ssrc = kSsrc1;
-    header.timestamp = i * 45000;
-    header.extension.hasAbsoluteSendTime = true;
-    header.extension.absoluteSendTime = i << (18 - 6);
-    EXPECT_EQ(0, vie_.network->ReceivedBWEPacket(channel_, time, kPacketSize,
-                                                 header));
-    header.ssrc = kSsrc2;
-    header.timestamp += 171717;
-    EXPECT_EQ(0, vie_.network->ReceivedBWEPacket(channel_, time, kPacketSize,
-                                                 header));
-    webrtc::SleepMs(kIntervalMs);
-  }
-  EXPECT_TRUE(transport.FindREMBFor(kSsrc1, 200000.0));
-  EXPECT_TRUE(transport.FindREMBFor(kSsrc2, 200000.0));
-  unsigned int bandwidth = 0;
-  EXPECT_EQ(0, vie_.rtp_rtcp->GetEstimatedReceiveBandwidth(channel_,
-                                                           &bandwidth));
-  EXPECT_GT(bandwidth, 0u);
-}
-
-TEST_F(ViENetworkTest, ReceiveBWEPacket_AST_DisabledReceive)  {
-  EXPECT_EQ(0, vie_.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(channel_, false,
-                                                               1));
-  ReceiveASTPacketsForBWE();
-  EXPECT_FALSE(transport.FindREMBFor(kSsrc1, 0.0));
-  unsigned int bandwidth = 0;
-  EXPECT_EQ(0, vie_.rtp_rtcp->GetEstimatedReceiveBandwidth(channel_,
-                                                           &bandwidth));
-}
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_standard_integration_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
deleted file mode 100644
index 584f9e92ff..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains the "standard" suite of integration tests, implemented
-// as a GUnit test. This file is a part of the effort to try to automate all
-// tests in this section of the code. Currently, this code makes no attempt
-// to verify any video output - it only checks for direct errors.
-
-#include <stdio.h>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/metrics/video_metrics.h"
-#include "webrtc/test/testsupport/metrics/video_metrics.h"
-#include "webrtc/video_engine/test/auto_test/automated/legacy_fixture.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_creator.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-namespace {
-
-class ViEStandardIntegrationTest : public LegacyFixture {
-};
-
-TEST_F(ViEStandardIntegrationTest, RunsBaseTestWithoutErrors)  {
-  tests_->ViEBaseStandardTest();
-}
-
-// Flaky: https://code.google.com/p/webrtc/issues/detail?id=1734
-TEST_F(ViEStandardIntegrationTest, DISABLED_RunsCodecTestWithoutErrors)  {
-  tests_->ViECodecStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsCaptureTestWithoutErrors)  {
-  tests_->ViECaptureStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsImageProcessTestWithoutErrors)  {
-  tests_->ViEImageProcessStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsRenderTestWithoutErrors)  {
-  tests_->ViERenderStandardTest();
-}
-
-// Flaky, see webrtc:1790.
-TEST_F(ViEStandardIntegrationTest, DISABLED_RunsRtpRtcpTestWithoutErrors)  {
-  tests_->ViERtpRtcpStandardTest();
-}
-
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_video_verification_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_video_verification_test.cc
deleted file mode 100644
index 1653959470..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_video_verification_test.cc
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <sstream>
-#include <string>
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/metrics/video_metrics.h"
-#include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h"
-#include "webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-namespace {
-
-// The input file must be QCIF since I420 gets scaled to that in the tests
-// (it is so bandwidth-heavy we have no choice). Our comparison algorithms
-// wouldn't like scaling, so this will work when we compare with the original.
-const int kInputWidth = 176;
-const int kInputHeight = 144;
-
-class ViEVideoVerificationTest : public testing::Test {
- protected:
-  void SetUp() {
-    input_file_ = webrtc::test::ResourcePath("paris_qcif", "yuv");
-    local_file_renderer_ = NULL;
-    remote_file_renderer_ = NULL;
-  }
-
-  void InitializeFileRenderers() {
-    local_file_renderer_ = new ViEToFileRenderer();
-    remote_file_renderer_ = new ViEToFileRenderer();
-    SetUpLocalFileRenderer(local_file_renderer_);
-    SetUpRemoteFileRenderer(remote_file_renderer_);
-  }
-
-  void SetUpLocalFileRenderer(ViEToFileRenderer* file_renderer) {
-    SetUpFileRenderer(file_renderer, "-local-preview.yuv");
-  }
-
-  void SetUpRemoteFileRenderer(ViEToFileRenderer* file_renderer) {
-    SetUpFileRenderer(file_renderer, "-remote.yuv");
-  }
-
-  // Must be called manually inside the tests.
-  void StopRenderers() {
-    local_file_renderer_->StopRendering();
-    remote_file_renderer_->StopRendering();
-  }
-
-  void CompareFiles(const std::string& reference_file,
-                    const std::string& test_file,
-                    double* psnr_result, double *ssim_result) {
-    webrtc::test::QualityMetricsResult psnr;
-    int error = I420PSNRFromFiles(reference_file.c_str(), test_file.c_str(),
-                                  kInputWidth, kInputHeight, &psnr);
-
-    EXPECT_EQ(0, error) << "PSNR routine failed - output files missing?";
-    *psnr_result = psnr.average;
-
-    webrtc::test::QualityMetricsResult ssim;
-    error = I420SSIMFromFiles(reference_file.c_str(), test_file.c_str(),
-                              kInputWidth, kInputHeight, &ssim);
-    EXPECT_EQ(0, error) << "SSIM routine failed - output files missing?";
-    *ssim_result = ssim.average;
-
-    ViETest::Log("Results: PSNR is %f (dB; 48 is max), "
-                 "SSIM is %f (1 is perfect)",
-                 psnr.average, ssim.average);
-  }
-
-  // Note: must call AFTER CompareFiles.
-  void TearDownFileRenderers() {
-    TearDownFileRenderer(local_file_renderer_);
-    TearDownFileRenderer(remote_file_renderer_);
-  }
-
-  std::string input_file_;
-  ViEToFileRenderer* local_file_renderer_;
-  ViEToFileRenderer* remote_file_renderer_;
-  ViEFileBasedComparisonTests tests_;
-
- private:
-  void SetUpFileRenderer(ViEToFileRenderer* file_renderer,
-                         const std::string& suffix) {
-    std::string output_path = ViETest::GetResultOutputPath();
-    std::string filename = "render_output" + suffix;
-
-    if (!file_renderer->PrepareForRendering(output_path, filename)) {
-      FAIL() << "Could not open output file " << filename <<
-          " for writing.";
-    }
-  }
-
-  void TearDownFileRenderer(ViEToFileRenderer* file_renderer) {
-      assert(file_renderer);
-      bool test_failed = ::testing::UnitTest::GetInstance()->
-          current_test_info()->result()->Failed();
-      if (test_failed) {
-        // Leave the files for analysis if the test failed.
-        file_renderer->SaveOutputFile("failed-");
-      }
-      delete file_renderer;
-    }
-};
-
-TEST_F(ViEVideoVerificationTest, RunsBaseStandardTestWithoutErrors) {
-  // I420 is lossless, so the I420 test should obviously get perfect results -
-  // the local preview and remote output files should be bit-exact. This test
-  // runs on external transport to ensure we do not drop packets.
-  // However, it's hard to make 100% stringent requirements on the video engine
-  // since for instance the jitter buffer has non-deterministic elements. If it
-  // breaks five times in a row though, you probably introduced a bug.
-  const double kReasonablePsnr = webrtc::test::kMetricsPerfectPSNR - 2.0f;
-  const double kReasonableSsim = 0.99f;
-  const int kNumAttempts = 5;
-  for (int attempt = 0; attempt < kNumAttempts; ++attempt) {
-    InitializeFileRenderers();
-    ASSERT_TRUE(tests_.TestCallSetup(input_file_, kInputWidth, kInputHeight,
-                                     local_file_renderer_,
-                                     remote_file_renderer_));
-    std::string remote_file = remote_file_renderer_->GetFullOutputPath();
-    std::string local_preview = local_file_renderer_->GetFullOutputPath();
-    StopRenderers();
-
-    double actual_psnr = 0;
-    double actual_ssim = 0;
-    CompareFiles(local_preview, remote_file, &actual_psnr, &actual_ssim);
-
-    TearDownFileRenderers();
-
-    if (actual_psnr > kReasonablePsnr && actual_ssim > kReasonableSsim) {
-      // Test successful.
-      return;
-    } else {
-      ViETest::Log("Retrying; attempt %d of %d.", attempt + 1, kNumAttempts);
-    }
-  }
-
-  FAIL() << "Failed to achieve near-perfect PSNR and SSIM results after " <<
-      kNumAttempts << " attempts.";
-}
-
-}  // namespace
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
deleted file mode 100644
index 0ee26a5732..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest.h
-//
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
-
-#include "gflags/gflags.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_hardware.h"
-
-#ifndef WEBRTC_ANDROID
-#include <string>
-#endif
-
-class TbCaptureDevice;
-class TbInterfaces;
-class TbVideoChannel;
-class ViEToFileRenderer;
-
-DECLARE_bool(include_timing_dependent_tests);
-
-// This class provides a bunch of methods, implemented across several .cc
-// files, which runs tests on the video engine. All methods will report
-// errors using standard googletest macros, except when marked otherwise.
-class ViEAutoTest
-{
-public:
-    ViEAutoTest(void* window1, void* window2);
-    ~ViEAutoTest();
-
-    // These three are special and should not be run in a googletest harness.
-    // They keep track of their errors by themselves and return the number
-    // of errors.
-    int ViELoopbackCall();
-    int ViESimulcastCall();
-    int ViECustomCall();
-    int ViERecordCall();
-
-    // All functions except the three above are meant to run in a
-    // googletest harness.
-    void ViEStandardTest();
-    void ViEExtendedTest();
-    void ViEAPITest();
-
-    // vie_autotest_base.cc
-    void ViEBaseStandardTest();
-    void ViEBaseExtendedTest();
-    void ViEBaseAPITest();
-
-    // vie_autotest_capture.cc
-    void ViECaptureStandardTest();
-    void ViECaptureExtendedTest();
-    void ViECaptureAPITest();
-    void ViECaptureExternalCaptureTest();
-
-    // vie_autotest_codec.cc
-    void ViECodecStandardTest();
-    void ViECodecExtendedTest();
-    void ViECodecExternalCodecTest();
-    void ViECodecAPITest();
-
-    // vie_autotest_image_process.cc
-    void ViEImageProcessStandardTest();
-    void ViEImageProcessExtendedTest();
-    void ViEImageProcessAPITest();
-
-    // vie_autotest_network.cc
-    void ViENetworkStandardTest();
-    void ViENetworkExtendedTest();
-    void ViENetworkAPITest();
-
-    // vie_autotest_render.cc
-    void ViERenderStandardTest();
-    void ViERenderExtendedTest();
-    void ViERenderAPITest();
-
-    // vie_autotest_rtp_rtcp.cc
-    void ViERtpRtcpStandardTest();
-    void ViERtpRtcpAPITest();
-
-private:
-    void PrintAudioCodec(const webrtc::CodecInst audioCodec);
-    void PrintVideoCodec(const webrtc::VideoCodec videoCodec);
-
-    // Sets up rendering so the capture device output goes to window 1 and
-    // the video engine output goes to window 2.
-    void RenderCaptureDeviceAndOutputStream(TbInterfaces* video_engine,
-                                            TbVideoChannel* video_channel,
-                                            TbCaptureDevice* capture_device);
-
-    // Stops rendering into the two windows as was set up by a call to
-    // RenderCaptureDeviceAndOutputStream.
-    void StopRenderCaptureDeviceAndOutputStream(
-        TbInterfaces* video_engine,
-        TbVideoChannel* video_channel,
-        TbCaptureDevice* capture_device);
-
-    void* _window1;
-    void* _window2;
-
-    webrtc::VideoRenderType _renderType;
-    webrtc::VideoRender* _vrm1;
-    webrtc::VideoRender* _vrm2;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_android.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_android.h
deleted file mode 100644
index 6eff9d8f15..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_android.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
-
-#include <jni.h>
-
-class ViEAutoTestAndroid {
- public:
-  static int RunAutotest(int testSelection,
-                         int subTestSelection,
-                         void* window1,
-                         void* window2,
-                         JavaVM* javaVM,
-                         void* env,
-                         void* context);
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h
deleted file mode 100644
index 3e8d762087..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_defines.h
-//
-
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
-
-#include <assert.h>
-#include <stdarg.h>
-#include <stdio.h>
-
-#include <string>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-
-#if defined(_WIN32)
-#include <windows.h>
-#elif defined (WEBRTC_ANDROID)
-#include <android/log.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-#include <stdlib.h>
-#include <string.h>
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-// Choose how to log
-//#define VIE_LOG_TO_FILE
-#define VIE_LOG_TO_STDOUT
-
-// Choose one way to test error
-#define VIE_ASSERT_ERROR
-
-#define VIE_LOG_FILE_NAME "ViEAutotestLog.txt"
-
-#undef RGB
-#define RGB(r,g,b) r|g<<8|b<<16
-
-enum { kAutoTestSleepTimeMs = 5000 };
-enum { kAutoTestFullStackSleepTimeMs = 20000 };
-
-struct AutoTestSize {
-  unsigned int width;
-  unsigned int height;
-  AutoTestSize() :
-    width(0), height(0) {
-  }
-  AutoTestSize(unsigned int iWidth, unsigned int iHeight) :
-    width(iWidth), height(iHeight) {
-  }
-};
-
-struct AutoTestOrigin {
-  unsigned int x;
-  unsigned int y;
-  AutoTestOrigin() :
-    x(0), y(0) {
-  }
-  AutoTestOrigin(unsigned int iX, unsigned int iY) :
-    x(iX), y(iY) {
-  }
-};
-
-struct AutoTestRect {
-  AutoTestSize size;
-  AutoTestOrigin origin;
-  AutoTestRect() :
-    size(), origin() {
-  }
-
-  AutoTestRect(unsigned int iX, unsigned int iY, unsigned int iWidth, unsigned int iHeight) :
-    size(iX, iY), origin(iWidth, iHeight) {
-  }
-
-  void Copy(AutoTestRect iRect) {
-    origin.x = iRect.origin.x;
-    origin.y = iRect.origin.y;
-    size.width = iRect.size.width;
-    size.height = iRect.size.height;
-  }
-};
-
-// ============================================
-
-class ViETest {
- public:
-  static int Init() {
-#ifdef VIE_LOG_TO_FILE
-    log_file_ = fopen(VIE_LOG_FILE_NAME, "w+t");
-#else
-    log_file_ = NULL;
-#endif
-    log_str_ = new char[kMaxLogSize];
-    memset(log_str_, 0, kMaxLogSize);
-    return 0;
-  }
-
-  static int Terminate() {
-    if (log_file_) {
-      fclose(log_file_);
-      log_file_ = NULL;
-    }
-    if (log_str_) {
-      delete[] log_str_;
-      log_str_ = NULL;
-    }
-    return 0;
-  }
-
-  static void Log(const char* fmt, ...) {
-    va_list va;
-    va_start(va, fmt);
-    memset(log_str_, 0, kMaxLogSize);
-    vsprintf(log_str_, fmt, va);
-    va_end(va);
-
-    WriteToSuitableOutput(log_str_);
-  }
-
-  // Writes to a suitable output, depending on platform and log mode.
-  static void WriteToSuitableOutput(const char* message) {
-#ifdef VIE_LOG_TO_FILE
-    if (log_file_)
-    {
-      fwrite(log_str_, 1, strlen(log_str_), log_file_);
-      fwrite("\n", 1, 1, log_file_);
-      fflush(log_file_);
-    }
-#endif
-#ifdef VIE_LOG_TO_STDOUT
-#if WEBRTC_ANDROID
-    __android_log_write(ANDROID_LOG_DEBUG, "*WebRTCN*", log_str_);
-#else
-    printf("%s\n", log_str_);
-#endif
-#endif
-  }
-
-  // Deprecated(phoglund): Prefer to use googletest macros in all cases
-  // except the custom call case.
-  static int TestError(bool expr, const char* fmt, ...) {
-    if (!expr) {
-      va_list va;
-      va_start(va, fmt);
-      memset(log_str_, 0, kMaxLogSize);
-      vsprintf(log_str_, fmt, va);
-#ifdef WEBRTC_ANDROID
-      __android_log_write(ANDROID_LOG_ERROR, "*WebRTCN*", log_str_);
-#endif
-      WriteToSuitableOutput(log_str_);
-      va_end(va);
-
-      AssertError(log_str_);
-      return 1;
-    }
-    return 0;
-  }
-
-  // Returns a suitable path to write trace and result files to.
-  // You should always use this when you want to write output files.
-  // The returned path is guaranteed to end with a path separator.
-  // This function may be run at any time during the program's execution.
-  // Implemented in vie_autotest.cc
-  static std::string GetResultOutputPath();
-
-private:
-  static void AssertError(const char* message) {
-#ifdef VIE_ASSERT_ERROR
-    assert(false);
-#endif
-  }
-
-  static FILE* log_file_;
-  enum {
-    kMaxLogSize = 512
-  };
-  static char* log_str_;
-};
-
-#define AutoTestSleep webrtc::SleepMs
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_linux.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_linux.h
deleted file mode 100644
index 6af87e8b3e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_linux.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
-
-// Note(pbos): This MUST be included before the X11 headers
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-
-#include <X11/Xlib.h>
-#include <X11/Xutil.h>
-
-// Forward declaration
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
-{
-public:
-    ViEAutoTestWindowManager();
-    virtual ~ViEAutoTestWindowManager();
-    virtual void* GetWindow1();
-    virtual void* GetWindow2();
-    virtual int TerminateWindows();
-    virtual int CreateWindows(AutoTestRect window1Size,
-                              AutoTestRect window2Size, void* window1Title,
-                              void* window2Title);
-    virtual bool SetTopmostWindow();
-
-private:
-    int ViECreateWindow(Window *outWindow, Display **outDisplay, int xpos,
-                        int ypos, int width, int height, char* title);
-    int ViEDestroyWindow(Window *window, Display *display);
-
-    Window _hwnd1;
-    Window _hwnd2;
-    Display* _hdsp1;
-    Display* _hdsp2;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h
deleted file mode 100644
index ba1cb5c31b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-
-#if defined(COCOA_RENDERING)
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-
-@class CocoaRenderView;
-
-#import <Cocoa/Cocoa.h>
-
-@interface TestCocoaUi : NSObject {
-  CocoaRenderView* cocoaRenderView1_;
-  CocoaRenderView* cocoaRenderView2_;
-  NSWindow* window1_;
-  NSWindow* window2_;
-
-  AutoTestRect window1Size_;
-  AutoTestRect window2Size_;
-  void* window1Title_;
-  void* window2Title_;
-}
-
-// Must be called as a selector in the main thread.
-- (void)createWindows:(NSObject*)ignored;
-
-// Used to transfer parameters from background thread.
-- (void)prepareToCreateWindowsWithSize:(AutoTestRect)window1Size
-                               andSize:(AutoTestRect)window2Size
-                             withTitle:(void*)window1Title
-                              andTitle:(void*)window2Title;
-
-- (NSWindow*)window1;
-- (NSWindow*)window2;
-- (CocoaRenderView*)cocoaRenderView1;
-- (CocoaRenderView*)cocoaRenderView2;
-
-@end
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface {
- public:
-  ViEAutoTestWindowManager();
-  virtual ~ViEAutoTestWindowManager();
-  virtual void* GetWindow1();
-  virtual void* GetWindow2();
-  virtual int CreateWindows(AutoTestRect window1Size,
-                            AutoTestRect window2Size,
-                            void* window1Title,
-                            void* window2Title);
-  virtual int TerminateWindows();
-  virtual bool SetTopmostWindow();
-
- private:
-  TestCocoaUi* cocoa_ui_;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
-#endif  // COCOA_RENDERING
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h
deleted file mode 100644
index 68d0079f39..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
-
-#include <string>
-#include <map>
-
-class ViEAutoTestMain {
- public:
-  ViEAutoTestMain();
-
-  // Runs the test according to the specified arguments.
-  // Pass in --automated to run in automated mode; interactive
-  // mode is default. All usual googletest flags also apply.
-  int RunTests(int argc, char** argv);
-
- private:
-  std::map<int, std::string> index_to_test_method_map_;
-
-  static const int kInvalidChoice = -1;
-
-  // Starts interactive mode.
-  int RunInteractiveMode();
-  // Prompts the user for a specific test method in the provided test case.
-  // Returns 0 on success, nonzero otherwise.
-  int RunSpecificTestCaseIn(const std::string test_case_name);
-  // Asks the user for a particular test case to run.
-  int AskUserForTestCase();
-  // Retrieves a number from the user in the interval
-  // [min_allowed, max_allowed]. Returns kInvalidChoice on failure.
-  int AskUserForNumber(int min_allowed, int max_allowed);
-  // Runs all tests matching the provided filter. * are wildcards.
-  // Returns the test runner result (0 == OK).
-  int RunTestMatching(const std::string test_case,
-                      const std::string test_method);
-  // Runs a non-gtest test case. Choice must be [7,9]. Returns 0 on success.
-  int RunSpecialTestCase(int choice);
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h
deleted file mode 100644
index 3f53815284..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- *  vie_autotest_window_manager_interface.h
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
-
-class ViEAutoTestWindowManagerInterface
-{
-public:
-    virtual int CreateWindows(AutoTestRect window1Size,
-                              AutoTestRect window2Size, void* window1Title,
-                              void* window2Title) = 0;
-    virtual int TerminateWindows() = 0;
-    virtual void* GetWindow1() = 0;
-    virtual void* GetWindow2() = 0;
-    virtual bool SetTopmostWindow() = 0;
-    virtual ~ViEAutoTestWindowManagerInterface() {}
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h
deleted file mode 100644
index fffd9484fe..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-
-#include <windows.h>
-#define TITLE_LENGTH 1024
-
-// Forward declaration
-namespace webrtc {
-class CriticalSectionWrapper;
-}
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
-{
-public:
-    ViEAutoTestWindowManager();
-    virtual ~ViEAutoTestWindowManager();
-    virtual void* GetWindow1();
-    virtual void* GetWindow2();
-    virtual int CreateWindows(AutoTestRect window1Size,
-                              AutoTestRect window2Size, void* window1Title,
-                              void* window2Title);
-    virtual int TerminateWindows();
-    virtual bool SetTopmostWindow();
-protected:
-    static bool EventProcess(void* obj);
-    bool EventLoop();
-
-private:
-    int ViECreateWindow(HWND &hwndMain, int xPos, int yPos, int width,
-                        int height, TCHAR* className);
-    int ViEDestroyWindow(HWND& hwnd);
-
-    void* _window1;
-    void* _window2;
-
-    bool _terminate;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _eventThread;
-    webrtc::CriticalSectionWrapper& _crit;
-    HWND _hwndMain;
-    HWND _hwnd1;
-    HWND _hwnd2;
-
-    AutoTestRect _hwnd1Size;
-    AutoTestRect _hwnd2Size;
-    TCHAR _hwnd1Title[TITLE_LENGTH];
-    TCHAR _hwnd2Title[TITLE_LENGTH];
-
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h
deleted file mode 100644
index 116ad26bde..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
-#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
-
-#include <string>
-
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-
-class FrameDropDetector;
-struct NetworkParameters;
-class ViEToFileRenderer;
-
-// This class contains comparison tests, which will exercise video engine
-// functionality and then run comparison tests on the result using PSNR and
-// SSIM algorithms. These tests are intended mostly as sanity checks so that
-// we know we are outputting roughly the right thing and not random noise or
-// black screens.
-//
-// We will set up a fake ExternalCapture device which will pose as a webcam
-// and read the input from the provided raw YUV file. Output will be written
-// as a local preview in the local file renderer; the remote side output gets
-// written to the provided remote file renderer.
-//
-// The local preview is a straight, unaltered copy of the input. This can be
-// useful for comparisons if the test method contains several stages where the
-// input is restarted between stages.
-class ViEFileBasedComparisonTests {
- public:
-  // Test a typical simple call setup. Returns false if the input file
-  // could not be opened; reports errors using googletest macros otherwise.
-  bool TestCallSetup(
-      const std::string& i420_test_video_path,
-      int width,
-      int height,
-      ViEToFileRenderer* local_file_renderer,
-      ViEToFileRenderer* remote_file_renderer);
-
-  // Runs a full stack test using the VP8 codec. Tests the full stack and uses
-  // RTP timestamps to sync frames between the endpoints.
-  void TestFullStack(
-      const std::string& i420_video_file,
-      int width,
-      int height,
-      int bit_rate_kbps,
-      ProtectionMethod protection_method,
-      const NetworkParameters& network,
-      ViEToFileRenderer* local_file_renderer,
-      ViEToFileRenderer* remote_file_renderer,
-      FrameDropDetector* frame_drop_detector);
-};
-
-#endif  // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_window_creator.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_window_creator.h
deleted file mode 100644
index c13a8889b7..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_window_creator.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
-
-class ViEAutoTestWindowManagerInterface;
-
-class ViEWindowCreator {
- public:
-  ViEWindowCreator();
-  virtual ~ViEWindowCreator();
-
-  // The pointer returned here will still be owned by this object.
-  // Only use it to retrieve the created windows.
-  ViEAutoTestWindowManagerInterface* CreateTwoWindows();
-
-  // Terminates windows opened by CreateTwoWindows, which must
-  // have been called before this method.
-  void TerminateWindows();
- private:
-  ViEAutoTestWindowManagerInterface* window_manager_;
-};
-
-#endif  // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h
deleted file mode 100644
index a85280dfe3..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
-
-class ViEAutoTestWindowManagerInterface;
-
-class ViEWindowManagerFactory {
- public:
-  // This method is implemented in different files depending on platform.
-  // The caller is responsible for freeing the resulting object using
-  // the delete operator.
-  static ViEAutoTestWindowManagerInterface*
-  CreateWindowManagerForCurrentPlatform();
-};
-
-#endif  // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/base_primitives.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/base_primitives.cc
deleted file mode 100644
index 32f4b2107d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/base_primitives.cc
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/primitives/base_primitives.h"
-
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-
-static void ConfigureCodecsToI420(int video_channel,
-                                  webrtc::VideoCodec video_codec,
-                                  webrtc::ViECodec* codec_interface) {
-  // Set up the codec interface with all known receive codecs and with
-  // I420 as the send codec.
-  for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
-    EXPECT_EQ(0, codec_interface->GetCodec(i, video_codec));
-
-    // Try to keep the test frame size small and bit rate generous when I420.
-    if (video_codec.codecType == webrtc::kVideoCodecI420) {
-      video_codec.width = 176;
-      video_codec.height = 144;
-      video_codec.maxBitrate = 32000;
-      video_codec.startBitrate = 32000;
-      EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, video_codec));
-    }
-
-    EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
-  }
-  // Verify that we really found the I420 codec.
-  EXPECT_EQ(0, codec_interface->GetSendCodec(video_channel, video_codec));
-  EXPECT_EQ(webrtc::kVideoCodecI420, video_codec.codecType);
-}
-
-void TestI420CallSetup(webrtc::ViECodec* codec_interface,
-                       webrtc::VideoEngine* video_engine,
-                       webrtc::ViEBase* base_interface,
-                       webrtc::ViENetwork* network_interface,
-                       webrtc::ViERTP_RTCP* rtp_rtcp_interface,
-                       int video_channel,
-                       const char* device_name) {
-  webrtc::VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-  EXPECT_EQ(0, rtp_rtcp_interface->SetTransmissionSmoothingStatus(video_channel,
-                                                                  false));
-
-  ConfigureCodecsToI420(video_channel, video_codec, codec_interface);
-
-  TbExternalTransport external_transport(
-      *network_interface, video_channel, NULL);
-  EXPECT_EQ(0, network_interface->RegisterSendTransport(
-      video_channel, external_transport));
-  EXPECT_EQ(0, base_interface->StartReceive(video_channel));
-  EXPECT_EQ(0, base_interface->StartSend(video_channel));
-
-  // Let the call run for a while.
-  ViETest::Log("Call started");
-  AutoTestSleep(kAutoTestSleepTimeMs);
-
-  // Stop the call.
-  ViETest::Log("Stopping call.");
-  EXPECT_EQ(0, base_interface->StopSend(video_channel));
-
-  // Make sure we receive all packets.
-  AutoTestSleep(1000);
-
-  EXPECT_EQ(0, base_interface->StopReceive(video_channel));
-  EXPECT_EQ(0, network_interface->DeregisterSendTransport(video_channel));
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/base_primitives.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/base_primitives.h
deleted file mode 100644
index 956871747f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/base_primitives.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
-
-namespace webrtc {
-class VideoEngine;
-class ViEBase;
-class ViECodec;
-class ViENetwork;
-class ViERTP_RTCP;
-}
-
-// Tests a I420-to-I420 call. This test exercises the most basic WebRTC
-// functionality by training the codec interface to recognize the most common
-// codecs, and the initiating a I420 call. A video channel with a capture device
-// must be set up prior to this call.
-void TestI420CallSetup(webrtc::ViECodec* codec_interface,
-                       webrtc::VideoEngine* video_engine,
-                       webrtc::ViEBase* base_interface,
-                       webrtc::ViENetwork* network_interface,
-                       webrtc::ViERTP_RTCP* rtp_rtcp_interface,
-                       int video_channel,
-                       const char* device_name);
-
-#endif  // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers.cc
deleted file mode 100644
index 8cc686af9e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers.cc
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/primitives/choice_helpers.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <algorithm>
-#include <sstream>
-
-namespace webrtc {
-
-ChoiceBuilder::ChoiceBuilder(const std::string& title, const Choices& choices)
-    : choices_(choices),
-      input_helper_(TypedInput(title)) {
-  input_helper_.WithInputValidator(
-      new IntegerWithinRangeValidator(1, choices.size()));
-  input_helper_.WithAdditionalInfo(MakeHumanReadableOptions());
-}
-
-int ChoiceBuilder::Choose() {
-  std::string input = input_helper_.AskForInput();
-  return atoi(input.c_str());
-}
-
-ChoiceBuilder& ChoiceBuilder::WithDefault(const std::string& default_choice) {
-  Choices::const_iterator iterator = std::find(
-      choices_.begin(), choices_.end(), default_choice);
-  assert(iterator != choices_.end() && "No such choice.");
-
-  // Store the value as the choice number, e.g. its index + 1.
-  int choice_index = (iterator - choices_.begin()) + 1;
-  char number[16];
-  sprintf(number, "%d", choice_index);
-
-  input_helper_.WithDefault(number);
-  return *this;
-}
-
-ChoiceBuilder& ChoiceBuilder::WithInputSource(FILE* input_source) {
-  input_helper_.WithInputSource(input_source);
-  return *this;
-}
-
-std::string ChoiceBuilder::MakeHumanReadableOptions() {
-  std::string result = "";
-  Choices::const_iterator iterator = choices_.begin();
-  for (int number = 1; iterator != choices_.end(); ++iterator, ++number) {
-    std::ostringstream os;
-    os << "\n  " << number << ". " << (*iterator).c_str();
-    result += os.str();
-  }
-  return result;
-}
-
-Choices SplitChoices(const std::string& raw_choices) {
-  return Split(raw_choices, "\n");
-}
-
-ChoiceBuilder FromChoices(
-    const std::string& title, const std::string& raw_choices) {
-  return ChoiceBuilder(title, SplitChoices(raw_choices));
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers.h
deleted file mode 100644
index 777aae57c2..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CHOICE_HELPERS_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CHOICE_HELPERS_H_
-
-#include <string>
-#include <vector>
-
-#include "webrtc/video_engine/test/auto_test/primitives/input_helpers.h"
-
-namespace webrtc {
-
-typedef std::vector<std::string> Choices;
-
-/**
- * Used to ask the user to make a choice. This class will allow you to
- * configure how to ask the question, and then ask it. For instance,
- *
- * int choice = FromChoices("Choice 1\n"
- *                          "Choice 2\n").WithDefault("Choice 1").Choose();
- *
- * will print a menu presenting the two choices and ask for input. The user,
- * can input 1, 2 or just hit enter since we specified a default in this case.
- * The Choose call will block until the user gives valid input one way or the
- * other. The choice variable is guaranteed to contain either 1 or 2 after
- * this particular call.
- *
- * The class uses stdout and stdin by default, but stdin can be replaced using
- * WithInputSource for unit tests.
- */
-class ChoiceBuilder {
- public:
-  explicit ChoiceBuilder(const std::string& title, const Choices& choices);
-
-  // Specifies the choice as the default. The choice must be one of the choices
-  // passed in the constructor. If this method is not called, the user has to
-  // choose an option explicitly.
-  ChoiceBuilder& WithDefault(const std::string& default_choice);
-
-  // Replaces the input source where we ask for input. Default is stdin.
-  ChoiceBuilder& WithInputSource(FILE* input_source);
-
-  // Prints the choice list and requests input from the input source. Returns
-  // the choice number (choices start at 1).
-  int Choose();
- private:
-  std::string MakeHumanReadableOptions();
-
-  Choices choices_;
-  InputBuilder input_helper_;
-};
-
-// Convenience function that creates a choice builder given a string where
-// choices are separated by \n.
-ChoiceBuilder FromChoices(const std::string& title,
-                          const std::string& raw_choices);
-
-// Creates choices from a string where choices are separated by \n.
-Choices SplitChoices(const std::string& raw_choices);
-
-}  // namespace webrtc
-
-#endif  // CHOICE_HELPERS_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers_unittest.cc
deleted file mode 100644
index 469a375eb2..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/choice_helpers_unittest.cc
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video_engine/test/auto_test/primitives/choice_helpers.h"
-#include "webrtc/video_engine/test/auto_test/primitives/fake_stdin.h"
-
-namespace webrtc {
-
-class ChoiceHelpersTest : public testing::Test {
-};
-
-TEST_F(ChoiceHelpersTest, SplitReturnsEmptyChoicesForEmptyInput) {
-  EXPECT_TRUE(SplitChoices("").empty());
-}
-
-TEST_F(ChoiceHelpersTest, SplitHandlesSingleChoice) {
-  Choices choices = SplitChoices("Single Choice");
-  EXPECT_EQ(1u, choices.size());
-  EXPECT_EQ("Single Choice", choices[0]);
-}
-
-TEST_F(ChoiceHelpersTest, SplitHandlesSingleChoiceWithEndingNewline) {
-  Choices choices = SplitChoices("Single Choice\n");
-  EXPECT_EQ(1u, choices.size());
-  EXPECT_EQ("Single Choice", choices[0]);
-}
-
-TEST_F(ChoiceHelpersTest, SplitHandlesMultipleChoices) {
-  Choices choices = SplitChoices(
-      "Choice 1\n"
-      "Choice 2\n"
-      "Choice 3");
-  EXPECT_EQ(3u, choices.size());
-  EXPECT_EQ("Choice 1", choices[0]);
-  EXPECT_EQ("Choice 2", choices[1]);
-  EXPECT_EQ("Choice 3", choices[2]);
-}
-
-TEST_F(ChoiceHelpersTest, SplitHandlesMultipleChoicesWithEndingNewline) {
-  Choices choices = SplitChoices(
-      "Choice 1\n"
-      "Choice 2\n"
-      "Choice 3\n");
-  EXPECT_EQ(3u, choices.size());
-  EXPECT_EQ("Choice 1", choices[0]);
-  EXPECT_EQ("Choice 2", choices[1]);
-  EXPECT_EQ("Choice 3", choices[2]);
-}
-
-TEST_F(ChoiceHelpersTest, CanSelectUsingChoiceBuilder) {
-  FILE* fake_stdin = FakeStdin("1\n2\n");
-  EXPECT_EQ(1, FromChoices("Title",
-                           "Choice 1\n"
-                           "Choice 2").WithInputSource(fake_stdin).Choose());
-  EXPECT_EQ(2, FromChoices("","Choice 1\n"
-                           "Choice 2").WithInputSource(fake_stdin).Choose());
-  fclose(fake_stdin);
-}
-
-TEST_F(ChoiceHelpersTest, RetriesIfGivenInvalidChoice) {
-  FILE* fake_stdin = FakeStdin("3\n0\n99\n23409234809\na\nwhatever\n1\n");
-  EXPECT_EQ(1, FromChoices("Title",
-                           "Choice 1\n"
-                           "Choice 2").WithInputSource(fake_stdin).Choose());
-  fclose(fake_stdin);
-}
-
-TEST_F(ChoiceHelpersTest, RetriesOnEnterIfNoDefaultSet) {
-  FILE* fake_stdin = FakeStdin("\n2\n");
-  EXPECT_EQ(2, FromChoices("Title",
-                           "Choice 1\n"
-                           "Choice 2").WithInputSource(fake_stdin).Choose());
-  fclose(fake_stdin);
-}
-
-TEST_F(ChoiceHelpersTest, PicksDefaultOnEnterIfDefaultSet) {
-  FILE* fake_stdin = FakeStdin("\n");
-  EXPECT_EQ(2, FromChoices("Title",
-                           "Choice 1\n"
-                           "Choice 2").WithInputSource(fake_stdin)
-                               .WithDefault("Choice 2").Choose());
-  fclose(fake_stdin);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/fake_stdin.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/fake_stdin.cc
deleted file mode 100644
index 26705f27d4..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/fake_stdin.cc
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/primitives/fake_stdin.h"
-
-namespace webrtc {
-
-FILE* FakeStdin(const std::string& input) {
-  FILE* fake_stdin = tmpfile();
-
-  EXPECT_EQ(input.size(),
-      fwrite(input.c_str(), sizeof(char), input.size(), fake_stdin));
-  rewind(fake_stdin);
-
-  return fake_stdin;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
deleted file mode 100644
index e9ff5378e9..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
+++ /dev/null
@@ -1,628 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <math.h>
-
-#include <sstream>
-#include <string>
-
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/frame_reader.h"
-#include "webrtc/test/testsupport/frame_writer.h"
-#include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h"
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-enum { kWaitTimeForFinalDecodeMs = 100 };
-
-// Writes the frames to be encoded to file and tracks which frames are sent in
-// external transport on the local side and reports them to the
-// FrameDropDetector class.
-class LocalRendererEffectFilter : public webrtc::ExternalRendererEffectFilter {
- public:
-  LocalRendererEffectFilter(webrtc::ExternalRenderer* renderer,
-                            FrameDropDetector* frame_drop_detector)
-      : ExternalRendererEffectFilter(renderer),
-        frame_drop_detector_(frame_drop_detector) {}
-  int Transform(size_t size,
-                unsigned char* frame_buffer,
-                int64_t ntp_time_ms,
-                unsigned int timestamp,
-                unsigned int width,
-                unsigned int height) {
-    frame_drop_detector_->ReportFrameState(
-        FrameDropDetector::kCreated,
-        timestamp,
-        webrtc::TickTime::MicrosecondTimestamp());
-    return webrtc::ExternalRendererEffectFilter::Transform(
-        size, frame_buffer, ntp_time_ms, timestamp, width, height);
-  }
- private:
-  FrameDropDetector* frame_drop_detector_;
-};
-
-// Tracks which frames are sent in external transport on the local side
-// and reports them to the FrameDropDetector class.
-class FrameSentCallback : public SendFrameCallback {
- public:
-  explicit FrameSentCallback(FrameDropDetector* frame_drop_detector)
-      : frame_drop_detector_(frame_drop_detector) {}
-  virtual ~FrameSentCallback() {}
-  virtual void FrameSent(unsigned int rtp_timestamp) {
-    frame_drop_detector_->ReportFrameState(
-        FrameDropDetector::kSent,
-        rtp_timestamp,
-        webrtc::TickTime::MicrosecondTimestamp());
-  }
-
- private:
-  FrameDropDetector* frame_drop_detector_;
-};
-
-// Tracks which frames are received in external transport on the remote side
-// and reports them to the FrameDropDetector class.
-class FrameReceivedCallback : public ReceiveFrameCallback {
- public:
-  explicit FrameReceivedCallback(FrameDropDetector* frame_drop_detector)
-      : frame_drop_detector_(frame_drop_detector) {}
-  virtual ~FrameReceivedCallback() {}
-  virtual void FrameReceived(unsigned int rtp_timestamp) {
-    frame_drop_detector_->ReportFrameState(
-        FrameDropDetector::kReceived,
-        rtp_timestamp,
-        webrtc::TickTime::MicrosecondTimestamp());
-  }
-
- private:
-  FrameDropDetector* frame_drop_detector_;
-};
-
-// Tracks when frames are decoded on the remote side (received from the
-// jitter buffer) and reports them to the FrameDropDetector class.
-class DecodedTimestampEffectFilter : public webrtc::ViEEffectFilter {
- public:
-  explicit DecodedTimestampEffectFilter(FrameDropDetector* frame_drop_detector)
-      : frame_drop_detector_(frame_drop_detector) {}
-  virtual ~DecodedTimestampEffectFilter() {}
-  virtual int Transform(size_t size,
-                        unsigned char* frame_buffer,
-                        int64_t ntp_time_ms,
-                        unsigned int timestamp,
-                        unsigned int width,
-                        unsigned int height) {
-    frame_drop_detector_->ReportFrameState(
-        FrameDropDetector::kDecoded,
-        timestamp,
-        webrtc::TickTime::MicrosecondTimestamp());
-    return 0;
-  }
-
- private:
-  FrameDropDetector* frame_drop_detector_;
-};
-
-class Statistics {
- public:
-  Statistics() : sum_(0.0f), sum_squared_(0.0f), count_(0) {};
-
-  void AddSample(float sample) {
-    sum_ += sample;
-    sum_squared_ += sample * sample;
-    ++count_;
-  }
-
-  float Mean() {
-    if (count_ == 0)
-      return -1.0f;
-    return sum_ / count_;
-  }
-
-  float Variance() {
-    if (count_ == 0)
-      return -1.0f;
-    return  sum_squared_ / count_ - Mean() * Mean();
-  }
-
-  std::string AsString() {
-    std::stringstream ss;
-    ss << (Mean() >= 0 ? Mean() : -1) << ", " <<
-        (Variance() >= 0 ? sqrt(Variance()) : -1);
-    return ss.str();
-  }
-
- private:
-  float sum_;
-  float sum_squared_;
-  int count_;
-};
-
-void TestFullStack(const TbInterfaces& interfaces,
-                   int capture_id,
-                   int video_channel,
-                   int width,
-                   int height,
-                   int bit_rate_kbps,
-                   const NetworkParameters& network,
-                   FrameDropDetector* frame_drop_detector,
-                   ViEToFileRenderer* remote_file_renderer,
-                   ViEToFileRenderer* local_file_renderer) {
-  webrtc::VideoEngine *video_engine_interface = interfaces.video_engine;
-  webrtc::ViEBase *base_interface = interfaces.base;
-  webrtc::ViECapture *capture_interface = interfaces.capture;
-  webrtc::ViERender *render_interface = interfaces.render;
-  webrtc::ViECodec *codec_interface = interfaces.codec;
-  webrtc::ViENetwork *network_interface = interfaces.network;
-
-  // ***************************************************************
-  // Engine ready. Begin testing class
-  // ***************************************************************
-  webrtc::VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof (webrtc::VideoCodec));
-
-  // Set up all receive codecs. This basically setup the codec interface
-  // to be able to recognize all receive codecs based on payload type.
-  for (int idx = 0; idx < codec_interface->NumberOfCodecs(); idx++) {
-    EXPECT_EQ(0, codec_interface->GetCodec(idx, video_codec));
-    SetSuitableResolution(&video_codec, width, height);
-
-    EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
-  }
-
-  // Configure External transport to simulate network interference:
-  TbExternalTransport external_transport(*interfaces.network, video_channel,
-                                         NULL);
-  external_transport.SetNetworkParameters(network);
-
-  FrameSentCallback frame_sent_callback(frame_drop_detector);
-  FrameReceivedCallback frame_received_callback(frame_drop_detector);
-  external_transport.RegisterSendFrameCallback(&frame_sent_callback);
-  external_transport.RegisterReceiveFrameCallback(&frame_received_callback);
-  EXPECT_EQ(0, network_interface->RegisterSendTransport(video_channel,
-                                                        external_transport));
-  RenderToFile(interfaces.render, video_channel, remote_file_renderer);
-  EXPECT_EQ(0, base_interface->StartReceive(video_channel));
-
-  // Setup only the VP8 codec, which is what we'll use.
-  webrtc::VideoCodec codec;
-  EXPECT_TRUE(FindSpecificCodec(webrtc::kVideoCodecVP8, codec_interface,
-                                &codec));
-  codec.startBitrate = bit_rate_kbps;
-  codec.maxBitrate = bit_rate_kbps;
-  codec.width = width;
-  codec.height = height;
-  EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, codec));
-
-  webrtc::ViEImageProcess *image_process =
-      webrtc::ViEImageProcess::GetInterface(video_engine_interface);
-  EXPECT_TRUE(image_process);
-
-  // Setup the effect filters.
-  // Local rendering at the send-side is done in an effect filter to avoid
-  // synchronization issues with the remote renderer.
-  LocalRendererEffectFilter local_renderer_filter(local_file_renderer,
-                                                  frame_drop_detector);
-  EXPECT_EQ(0, image_process->RegisterSendEffectFilter(video_channel,
-                                                       local_renderer_filter));
-  DecodedTimestampEffectFilter decode_filter(frame_drop_detector);
-  EXPECT_EQ(0, image_process->RegisterRenderEffectFilter(video_channel,
-                                                         decode_filter));
-  // Send video.
-  EXPECT_EQ(0, base_interface->StartSend(video_channel));
-  AutoTestSleep(kAutoTestFullStackSleepTimeMs);
-
-  ViETest::Log("Done!");
-
-  // ***************************************************************
-  // Testing finished. Tear down Video Engine
-  // ***************************************************************
-  EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
-
-  const int one_way_delay_99_percentile = network.mean_one_way_delay  +
-        3 * network.std_dev_one_way_delay;
-
-  // Wait for the last packet to arrive before we tear down the receiver.
-  AutoTestSleep(2 * one_way_delay_99_percentile);
-  EXPECT_EQ(0, base_interface->StopSend(video_channel));
-  while (!external_transport.EmptyQueue()) {
-    AutoTestSleep(one_way_delay_99_percentile);
-  }
-  EXPECT_EQ(0, base_interface->StopReceive(video_channel));
-  EXPECT_EQ(0, network_interface->DeregisterSendTransport(video_channel));
-  // Wait for the last frame to be decoded and rendered. There is no guarantee
-  // this wait time will be long enough. Ideally we would wait for at least one
-  // "receive-side delay", which is what the video coding module calculates
-  // based on network statistics etc. We don't have access to that value here.
-  AutoTestSleep(kWaitTimeForFinalDecodeMs);
-  // Must stop the frame drop detectors in the right order to avoid getting
-  // frames which for instance are rendered but not decoded.
-  EXPECT_EQ(0, render_interface->StopRender(video_channel));
-  EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
-  EXPECT_EQ(0, image_process->DeregisterRenderEffectFilter(video_channel));
-  EXPECT_EQ(0, image_process->DeregisterSendEffectFilter(video_channel));
-  image_process->Release();
-  EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
-
-  // Collect transport statistics.
-  int32_t num_rtp_packets = 0;
-  int32_t num_dropped_packets = 0;
-  int32_t num_rtcp_packets = 0;
-  std::map<uint8_t, int> packet_counters;
-  external_transport.GetStats(num_rtp_packets, num_dropped_packets,
-                              num_rtcp_packets, &packet_counters);
-  ViETest::Log("RTP packets    : %5d", num_rtp_packets);
-  ViETest::Log("Dropped packets: %5d", num_dropped_packets);
-  ViETest::Log("RTCP packets   : %5d", num_rtcp_packets);
-}
-
-void FixOutputFileForComparison(const std::string& output_file,
-                                int frame_length_in_bytes,
-                                const std::vector<Frame*>& frames) {
-  webrtc::test::FrameReaderImpl frame_reader(output_file,
-                                             frame_length_in_bytes);
-  const std::string temp_file = output_file + ".fixed";
-  webrtc::test::FrameWriterImpl frame_writer(temp_file, frame_length_in_bytes);
-  frame_reader.Init();
-  frame_writer.Init();
-
-  ASSERT_FALSE(frames.front()->dropped_at_render) << "It should not be "
-      "possible to drop the first frame. Both because we don't have anything "
-      "useful to fill that gap with and it is impossible to detect it without "
-      "any previous timestamps to compare with.";
-
-  uint8_t* last_frame_data = new uint8_t[frame_length_in_bytes];
-
-  // Process the file and write frame duplicates for all dropped frames.
-  for (std::vector<Frame*>::const_iterator it = frames.begin();
-       it != frames.end(); ++it) {
-    if ((*it)->dropped_at_render) {
-      // Write the previous frame to the output file:
-      EXPECT_TRUE(frame_writer.WriteFrame(last_frame_data));
-    } else {
-      EXPECT_TRUE(frame_reader.ReadFrame(last_frame_data));
-      EXPECT_TRUE(frame_writer.WriteFrame(last_frame_data));
-    }
-  }
-  delete[] last_frame_data;
-  frame_reader.Close();
-  frame_writer.Close();
-  ASSERT_EQ(0, remove(output_file.c_str()));
-  ASSERT_EQ(0, rename(temp_file.c_str(), output_file.c_str()));
-}
-
-void FrameDropDetector::ReportFrameState(State state, unsigned int timestamp,
-                                         int64_t report_time_us) {
-  dirty_ = true;
-  switch (state) {
-    case kCreated: {
-      int number = created_frames_vector_.size();
-      Frame* frame = new Frame(number, timestamp);
-      frame->created_timestamp_in_us_ = report_time_us;
-      created_frames_vector_.push_back(frame);
-      created_frames_[timestamp] = frame;
-      num_created_frames_++;
-      break;
-    }
-    case kSent:
-      sent_frames_[timestamp] = report_time_us;
-      if (timestamp_diff_ == 0) {
-        // When the first created frame arrives we calculate the fixed
-        // difference between the timestamps of the frames entering and leaving
-        // the encoder. This diff is used to identify the frames from the
-        // created_frames_ map.
-        timestamp_diff_ =
-            timestamp - created_frames_vector_.front()->frame_timestamp_;
-      }
-      num_sent_frames_++;
-      break;
-    case kReceived:
-      received_frames_[timestamp] = report_time_us;
-      num_received_frames_++;
-      break;
-    case kDecoded:
-      decoded_frames_[timestamp] = report_time_us;
-      num_decoded_frames_++;
-      break;
-    case kRendered:
-      rendered_frames_[timestamp] = report_time_us;
-      num_rendered_frames_++;
-      break;
-  }
-}
-
-void FrameDropDetector::CalculateResults() {
-  // Fill in all fields of the Frame objects in the created_frames_ map.
-  // Iterate over the maps from converted timestamps to the arrival timestamps.
-  std::map<unsigned int, int64_t>::const_iterator it;
-  for (it = sent_frames_.begin(); it != sent_frames_.end(); ++it) {
-    unsigned int created_timestamp = it->first - timestamp_diff_;
-    created_frames_[created_timestamp]->sent_timestamp_in_us_ = it->second;
-  }
-  for (it = received_frames_.begin(); it != received_frames_.end(); ++it) {
-    unsigned int created_timestamp = it->first - timestamp_diff_;
-    created_frames_[created_timestamp]->received_timestamp_in_us_ = it->second;
-  }
-  for (it = decoded_frames_.begin(); it != decoded_frames_.end(); ++it) {
-    unsigned int created_timestamp = it->first - timestamp_diff_;
-    created_frames_[created_timestamp]->decoded_timestamp_in_us_ =it->second;
-  }
-  for (it = rendered_frames_.begin(); it != rendered_frames_.end(); ++it) {
-    unsigned int created_timestamp = it->first - timestamp_diff_;
-    created_frames_[created_timestamp]->rendered_timestamp_in_us_ = it->second;
-  }
-  // Find out where the frames were not present in the different states.
-  dropped_frames_at_send_ = 0;
-  dropped_frames_at_receive_ = 0;
-  dropped_frames_at_decode_ = 0;
-  dropped_frames_at_render_ = 0;
-  for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
-       it != created_frames_vector_.end(); ++it) {
-    int encoded_timestamp = (*it)->frame_timestamp_ + timestamp_diff_;
-    if (sent_frames_.find(encoded_timestamp) == sent_frames_.end()) {
-      (*it)->dropped_at_send = true;
-      dropped_frames_at_send_++;
-    }
-    if (received_frames_.find(encoded_timestamp) == received_frames_.end()) {
-      (*it)->dropped_at_receive = true;
-      dropped_frames_at_receive_++;
-    }
-    if (decoded_frames_.find(encoded_timestamp) == decoded_frames_.end()) {
-      (*it)->dropped_at_decode = true;
-      dropped_frames_at_decode_++;
-    }
-    if (rendered_frames_.find(encoded_timestamp) == rendered_frames_.end()) {
-      (*it)->dropped_at_render = true;
-      dropped_frames_at_render_++;
-    }
-  }
-  dirty_ = false;
-}
-
-void FrameDropDetector::PrintReport(const std::string& test_label) {
-  assert(!dirty_);
-  ViETest::Log("Frame Drop Detector report:");
-  ViETest::Log("  Created  frames: %ld", created_frames_.size());
-  ViETest::Log("  Sent     frames: %ld", sent_frames_.size());
-  ViETest::Log("  Received frames: %ld", received_frames_.size());
-  ViETest::Log("  Decoded  frames: %ld", decoded_frames_.size());
-  ViETest::Log("  Rendered frames: %ld", rendered_frames_.size());
-
-  // Display all frames and stats for them:
-  long last_created = 0;
-  long last_sent = 0;
-  long last_received = 0;
-  long last_decoded = 0;
-  long last_rendered = 0;
-  ViETest::Log("\nDeltas between sent frames and drop status:");
-  ViETest::Log("Unit: Microseconds");
-  ViETest::Log("Frame  Created    Sent    Received Decoded Rendered "
-      "Dropped at  Dropped at  Dropped at  Dropped at");
-  ViETest::Log(" nbr    delta     delta    delta    delta   delta   "
-      " Send?       Receive?    Decode?     Render?");
-  Statistics rendering_stats;
-  for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
-       it != created_frames_vector_.end(); ++it) {
-    int created_delta =
-        static_cast<int>((*it)->created_timestamp_in_us_ - last_created);
-    int sent_delta = (*it)->dropped_at_send ? -1 :
-        static_cast<int>((*it)->sent_timestamp_in_us_ - last_sent);
-    int received_delta = (*it)->dropped_at_receive ? -1 :
-        static_cast<int>((*it)->received_timestamp_in_us_ - last_received);
-    int decoded_delta = (*it)->dropped_at_decode ? -1 :
-        static_cast<int>((*it)->decoded_timestamp_in_us_ - last_decoded);
-    int rendered_delta = (*it)->dropped_at_render ? -1 :
-        static_cast<int>((*it)->rendered_timestamp_in_us_ - last_rendered);
-
-    // Set values to -1 for the first frame:
-    if ((*it)->number_ == 0) {
-      created_delta = -1;
-      sent_delta = -1;
-      received_delta = -1;
-      decoded_delta = -1;
-      rendered_delta = -1;
-    }
-    ViETest::Log("%5d %8d %8d %8d %8d %8d %10s %10s %10s %10s",
-                 (*it)->number_,
-                 created_delta,
-                 sent_delta,
-                 received_delta,
-                 decoded_delta,
-                 rendered_delta,
-                 (*it)->dropped_at_send ? "DROPPED" : "      ",
-                 (*it)->dropped_at_receive ? "DROPPED" : "      ",
-                 (*it)->dropped_at_decode ? "DROPPED" : "      ",
-                 (*it)->dropped_at_render ? "DROPPED" : "      ");
-    last_created = (*it)->created_timestamp_in_us_;
-    if (!(*it)->dropped_at_send) {
-      last_sent = (*it)->sent_timestamp_in_us_;
-    }
-     if (!(*it)->dropped_at_receive) {
-      last_received = (*it)->received_timestamp_in_us_;
-    }
-    if (!(*it)->dropped_at_decode) {
-      last_decoded = (*it)->decoded_timestamp_in_us_;
-    }
-    if (!(*it)->dropped_at_render) {
-      last_rendered = (*it)->rendered_timestamp_in_us_;
-      rendering_stats.AddSample(rendered_delta / 1000.0f);
-    }
-  }
-  ViETest::Log("\nLatency between states (-1 means N/A because of drop):");
-  ViETest::Log("Unit: Microseconds");
-  ViETest::Log("Frame  Created    Sent      Received   Decoded      Total    "
-      "   Total");
-  ViETest::Log(" nbr   ->Sent  ->Received  ->Decoded ->Rendered    latency   "
-      "  latency");
-  ViETest::Log("                                               (incl network)"
-      "(excl network)");
-  Statistics latency_incl_network_stats;
-  for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
-       it != created_frames_vector_.end(); ++it) {
-    int created_to_sent = (*it)->dropped_at_send ? -1 :
-        static_cast<int>((*it)->sent_timestamp_in_us_ -
-                         (*it)->created_timestamp_in_us_);
-    int sent_to_received = (*it)->dropped_at_receive ? -1 :
-        static_cast<int>((*it)->received_timestamp_in_us_ -
-                         (*it)->sent_timestamp_in_us_);
-    int received_to_decoded = (*it)->dropped_at_decode ? -1 :
-        static_cast<int>((*it)->decoded_timestamp_in_us_ -
-                         (*it)->received_timestamp_in_us_);
-    int decoded_to_render = (*it)->dropped_at_render ? -1 :
-        static_cast<int>((*it)->rendered_timestamp_in_us_ -
-                         (*it)->decoded_timestamp_in_us_);
-    int total_latency_incl_network = (*it)->dropped_at_render ? -1 :
-        static_cast<int>((*it)->rendered_timestamp_in_us_ -
-                         (*it)->created_timestamp_in_us_);
-    int total_latency_excl_network = (*it)->dropped_at_render ? -1 :
-        static_cast<int>((*it)->rendered_timestamp_in_us_ -
-                         (*it)->created_timestamp_in_us_ - sent_to_received);
-    if (total_latency_incl_network >= 0)
-      latency_incl_network_stats.AddSample(total_latency_incl_network /
-                                           1000.0f);
-    ViETest::Log("%5d %9d %9d %9d %9d %12d %12d",
-                 (*it)->number_,
-                 created_to_sent,
-                 sent_to_received,
-                 received_to_decoded,
-                 decoded_to_render,
-                 total_latency_incl_network,
-                 total_latency_excl_network);
-  }
-
-  // Plot all measurements in the same graph since they share the same value
-  // range.
-  webrtc::test::PrintResultMeanAndError(
-      "total_delay_incl_network", "", test_label,
-      latency_incl_network_stats.AsString(), "ms", false);
-  webrtc::test::PrintResultMeanAndError(
-      "time_between_rendered_frames", "", test_label,
-      rendering_stats.AsString(), "ms", false);
-
-
-  // Find and print the dropped frames.
-  ViETest::Log("\nTotal # dropped frames at:");
-  ViETest::Log("  Send   : %d", dropped_frames_at_send_);
-  ViETest::Log("  Receive: %d", dropped_frames_at_receive_);
-  ViETest::Log("  Decode : %d", dropped_frames_at_decode_);
-  ViETest::Log("  Render : %d", dropped_frames_at_render_);
-}
-
-void FrameDropDetector::PrintDebugDump() {
-  assert(!dirty_);
-  ViETest::Log("\nPrintDebugDump: Frame objects:");
-  ViETest::Log("Frame FrTimeStamp Created       Sent      Received    Decoded"
-      "    Rendered ");
-  for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
-       it != created_frames_vector_.end(); ++it) {
-    ViETest::Log("%5d %11u %11lld %11lld %11lld %11lld %11lld",
-                 (*it)->number_,
-                 (*it)->frame_timestamp_,
-                 (*it)->created_timestamp_in_us_,
-                 (*it)->sent_timestamp_in_us_,
-                 (*it)->received_timestamp_in_us_,
-                 (*it)->decoded_timestamp_in_us_,
-                 (*it)->rendered_timestamp_in_us_);
-  }
-  std::vector<int> mismatch_frame_num_list;
-  for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
-       it != created_frames_vector_.end(); ++it) {
-    if ((*it)->dropped_at_render != (*it)->dropped_at_decode) {
-      mismatch_frame_num_list.push_back((*it)->number_);
-    }
-  }
-  if (mismatch_frame_num_list.size() > 0) {
-    ViETest::Log("\nDecoded/Rendered mismatches:");
-    ViETest::Log("Frame FrTimeStamp    Created       Sent      Received    "
-        "Decoded    Rendered ");
-    for (std::vector<int>::const_iterator it = mismatch_frame_num_list.begin();
-         it != mismatch_frame_num_list.end(); ++it) {
-      Frame* frame = created_frames_vector_[*it];
-      ViETest::Log("%5d %11u %11lld %11lld %11lld %11lld %11lld",
-                 frame->number_,
-                 frame->frame_timestamp_,
-                 frame->created_timestamp_in_us_,
-                 frame->sent_timestamp_in_us_,
-                 frame->received_timestamp_in_us_,
-                 frame->decoded_timestamp_in_us_,
-                 frame->rendered_timestamp_in_us_);
-    }
-  }
-
-  ViETest::Log("\nReportFrameState method invocations:");
-  ViETest::Log("  Created : %d", num_created_frames_);
-  ViETest::Log("  Send    : %d", num_sent_frames_);
-  ViETest::Log("  Received: %d", num_received_frames_);
-  ViETest::Log("  Decoded : %d", num_decoded_frames_);
-  ViETest::Log("  Rendered: %d", num_rendered_frames_);
-}
-
-const std::vector<Frame*>& FrameDropDetector::GetAllFrames() {
-  assert(!dirty_);
-  return created_frames_vector_;
-}
-
-int FrameDropDetector::GetNumberOfFramesDroppedAt(State state) {
-  assert(!dirty_);
-  switch (state) {
-    case kSent:
-      return dropped_frames_at_send_;
-    case kReceived:
-      return dropped_frames_at_receive_;
-    case kDecoded:
-      return dropped_frames_at_decode_;
-    case kRendered:
-      return dropped_frames_at_render_;
-    default:
-      return 0;
-  }
-}
-
-int FrameDropMonitoringRemoteFileRenderer::DeliverFrame(
-    unsigned char *buffer, size_t buffer_size, uint32_t time_stamp,
-    int64_t ntp_time_ms, int64_t render_time, void* /*handle*/) {
-  ReportFrameStats(time_stamp, render_time);
-  return ViEToFileRenderer::DeliverFrame(buffer, buffer_size,
-                                         time_stamp, ntp_time_ms,
-                                         render_time, NULL);
-}
-
-void FrameDropMonitoringRemoteFileRenderer::ReportFrameStats(
-    uint32_t time_stamp,
-    int64_t render_time) {
-  // |render_time| provides the ideal render time for this frame. If that time
-  // has already passed we will render it immediately.
-  int64_t report_render_time_us = render_time * 1000;
-  int64_t time_now_us = webrtc::TickTime::MicrosecondTimestamp();
-  if (render_time < (time_now_us + 500) / 1000) {
-    report_render_time_us = time_now_us;
-  }
-  // Register that this frame has been rendered.
-  frame_drop_detector_->ReportFrameState(FrameDropDetector::kRendered,
-                                         time_stamp, report_render_time_us);
-}
-
-int FrameDropMonitoringRemoteFileRenderer::DeliverI420Frame(
-    const webrtc::I420VideoFrame& webrtc_frame) {
-  ReportFrameStats(webrtc_frame.timestamp(), webrtc_frame.render_time_ms());
-  return ViEToFileRenderer::DeliverI420Frame(webrtc_frame);
-}
-
-int FrameDropMonitoringRemoteFileRenderer::FrameSizeChange(
-    unsigned int width, unsigned int height, unsigned int number_of_streams) {
-  return ViEToFileRenderer::FrameSizeChange(width, height, number_of_streams);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
deleted file mode 100644
index 92d7bcc33d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
+++ /dev/null
@@ -1,242 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
-
-#include <map>
-#include <vector>
-
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-class FrameDropDetector;
-struct NetworkParameters;
-class TbInterfaces;
-
-// Initializes the Video engine and its components, runs video playback using
-// for KAutoTestSleepTimeMs milliseconds, then shuts down everything.
-// The bit rate and packet loss parameters should be configured so that
-// frames are dropped, in order to test the frame drop detection that is
-// performed by the FrameDropDetector class.
-void TestFullStack(const TbInterfaces& interfaces,
-                   int capture_id,
-                   int video_channel,
-                   int width,
-                   int height,
-                   int bit_rate_kbps,
-                   const NetworkParameters& network,
-                   FrameDropDetector* frame_drop_detector,
-                   ViEToFileRenderer* remote_file_renderer,
-                   ViEToFileRenderer* local_file_renderer);
-
-// A frame in a video file. The four different points in the stack when
-// register the frame state are (in time order): created, transmitted, decoded,
-// rendered.
-class Frame {
- public:
-  Frame(int number, unsigned int timestamp)
-    : number_(number),
-      frame_timestamp_(timestamp),
-      created_timestamp_in_us_(-1),
-      sent_timestamp_in_us_(-1),
-      received_timestamp_in_us_(-1),
-      decoded_timestamp_in_us_(-1),
-      rendered_timestamp_in_us_(-1),
-      dropped_at_send(false),
-      dropped_at_receive(false),
-      dropped_at_decode(false),
-      dropped_at_render(false) {}
-
-  // Frame number, starting at 0.
-  int number_;
-
-  // Frame timestamp, that is used by Video Engine and RTP headers and set when
-  // the frame is sent into the stack.
-  unsigned int frame_timestamp_;
-
-  // Timestamps for our measurements of when the frame is in different states.
-  int64_t created_timestamp_in_us_;
-  int64_t sent_timestamp_in_us_;
-  int64_t received_timestamp_in_us_;
-  int64_t decoded_timestamp_in_us_;
-  int64_t rendered_timestamp_in_us_;
-
-  // Where the frame was dropped (more than one may be true).
-  bool dropped_at_send;
-  bool dropped_at_receive;
-  bool dropped_at_decode;
-  bool dropped_at_render;
-};
-
-// Fixes the output file by copying the last successful frame into the place
-// where the dropped frame would be, for all dropped frames (if any).
-// This method will not be able to fix data for the first frame if that is
-// dropped, since there'll be no previous frame to copy. This case should never
-// happen because of encoder frame dropping at least.
-// Parameters:
-//    output_file            The output file to modify (pad with frame copies
-//                           for all dropped frames)
-//    frame_length_in_bytes  Byte length of each frame.
-//    frames                 A vector of all Frame objects. Must be sorted by
-//                           frame number. If empty this method will do nothing.
-void FixOutputFileForComparison(const std::string& output_file,
-                                int frame_length_in_bytes,
-                                const std::vector<Frame*>& frames);
-
-// Handles statistics about dropped frames. Frames travel through the stack
-// with different timestamps. The frames created and sent to the encoder have
-// one timestamp on the sending side while the decoded/rendered frames have
-// another timestamp on the receiving side. The difference between these
-// timestamps is fixed, which we can use to identify the frames when they
-// arrive, since the FrameDropDetector class gets data reported from both sides.
-// The four different points in the stack when this class examines the frame
-// states are (in time order): created, sent, received, decoded, rendered.
-//
-// The flow can be visualized like this:
-//
-//         Created        Sent        Received               Decoded   Rendered
-// +-------+  |  +-------+ | +---------+ | +------+  +-------+  |  +--------+
-// |Capture|  |  |Encoder| | |  Ext.   | | |Jitter|  |Decoder|  |  |  Ext.  |
-// | device|---->|       |-->|transport|-->|buffer|->|       |---->|renderer|
-// +-------+     +-------+   +---------+   +------+  +-------+     +--------+
-//
-// This class has no intention of being thread-safe.
-class FrameDropDetector {
- public:
-  enum State {
-    // A frame being created, i.e. sent to the encoder; the first step of
-    // a frame's life cycle. This timestamp becomes the frame timestamp in the
-    // Frame objects.
-    kCreated,
-    // A frame being sent in external transport (to the simulated network). This
-    // timestamp differs from the one in the Created state by a constant diff.
-    kSent,
-    // A frame being received in external transport (from the simulated
-    // network). This timestamp differs from the one in the Created state by a
-    // constant diff.
-    kReceived,
-    // A frame that has been decoded in the decoder. This timestamp differs
-    // from the one in the Created state by a constant diff.
-    kDecoded,
-    // A frame that has been rendered; the last step of a frame's life cycle.
-    // This timestamp differs from the one in the Created state by a constant
-    // diff.
-    kRendered
-  };
-
-  FrameDropDetector()
-      : dirty_(true),
-        dropped_frames_at_send_(0),
-        dropped_frames_at_receive_(0),
-        dropped_frames_at_decode_(0),
-        dropped_frames_at_render_(0),
-        num_created_frames_(0),
-        num_sent_frames_(0),
-        num_received_frames_(0),
-        num_decoded_frames_(0),
-        num_rendered_frames_(0),
-        timestamp_diff_(0) {}
-
-  // Reports a frame has reached a state in the frame life cycle.
-  void ReportFrameState(State state, unsigned int timestamp,
-                        int64_t report_time_us);
-
-  // Uses all the gathered timestamp information to calculate which frames have
-  // been dropped during the test and where they were dropped. Not until
-  // this method has been executed, the Frame objects will have all fields
-  // filled with the proper timestamp information.
-  void CalculateResults();
-
-  // Calculates the number of frames have been registered as dropped at the
-  // specified state of the frame life cycle.
-  // CalculateResults() must be called before calling this method.
-  int GetNumberOfFramesDroppedAt(State state);
-
-  // Gets a vector of all the created frames.
-  // CalculateResults() must be called before calling this method to have all
-  // fields of the Frame objects to represent the current state.
-  const std::vector<Frame*>& GetAllFrames();
-
-  // Prints a detailed report about all the different frame states and which
-  // ones are detected as dropped, using ViETest::Log. Also prints
-  // perf-formatted output and adds |test_label| as a modifier to the perf
-  // output.
-  // CalculateResults() must be called before calling this method.
-  void PrintReport(const std::string& test_label);
-
-  // Prints all the timestamp maps. Mainly used for debugging purposes to find
-  // missing timestamps.
-  void PrintDebugDump();
- private:
-  // Will be false until CalculateResults() is called. Switches to true
-  // as soon as new timestamps are reported using ReportFrameState().
-  bool dirty_;
-
-  // Map of frame creation timestamps to all Frame objects.
-  std::map<unsigned int, Frame*> created_frames_;
-
-  // Maps converted frame timestamps (differ from creation timestamp) to the
-  // time they arrived in the different states of the frame's life cycle.
-  std::map<unsigned int, int64_t> sent_frames_;
-  std::map<unsigned int, int64_t> received_frames_;
-  std::map<unsigned int, int64_t> decoded_frames_;
-  std::map<unsigned int, int64_t> rendered_frames_;
-
-  // A vector with the frames sorted in their created order.
-  std::vector<Frame*> created_frames_vector_;
-
-  // Statistics.
-  int dropped_frames_at_send_;
-  int dropped_frames_at_receive_;
-  int dropped_frames_at_decode_;
-  int dropped_frames_at_render_;
-
-  int num_created_frames_;
-  int num_sent_frames_;
-  int num_received_frames_;
-  int num_decoded_frames_;
-  int num_rendered_frames_;
-
-  // The constant diff between the created and transmitted frames, since their
-  // timestamps are converted.
-  unsigned int timestamp_diff_;
-};
-
-// Tracks which frames are received on the remote side and reports back to the
-// FrameDropDetector class when they are rendered.
-class FrameDropMonitoringRemoteFileRenderer : public ViEToFileRenderer {
- public:
-  explicit FrameDropMonitoringRemoteFileRenderer(
-      FrameDropDetector* frame_drop_detector)
-      : frame_drop_detector_(frame_drop_detector) {}
-  virtual ~FrameDropMonitoringRemoteFileRenderer() {}
-
-  // Implementation of ExternalRenderer:
-  int FrameSizeChange(unsigned int width,
-                      unsigned int height,
-                      unsigned int number_of_streams) override;
-  int DeliverFrame(unsigned char* buffer,
-                   size_t buffer_size,
-                   uint32_t time_stamp,
-                   int64_t ntp_time_ms,
-                   int64_t render_time,
-                   void* handle) override;
-  int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;
-
- private:
-  void ReportFrameStats(uint32_t time_stamp, int64_t render_time);
-
-  FrameDropDetector* frame_drop_detector_;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc
deleted file mode 100644
index c1501e5a41..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc
+++ /dev/null
@@ -1,97 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h"
-
-#include <stdio.h>
-
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/frame_reader.h"
-#include "webrtc/test/testsupport/frame_writer.h"
-
-namespace webrtc {
-
-const std::string kOutputFilename = "temp_outputfile.tmp";
-const int kFrameLength = 1000;
-
-class FrameDropPrimitivesTest: public testing::Test {
- protected:
-  FrameDropPrimitivesTest() {}
-  virtual ~FrameDropPrimitivesTest() {}
-  void SetUp() {
-    // Cleanup any previous output file.
-    remove(kOutputFilename.c_str());
-  }
-  void TearDown() {
-    // Cleanup the temporary file.
-    remove(kOutputFilename.c_str());
-  }
-};
-
-TEST_F(FrameDropPrimitivesTest, FixOutputFileForComparison) {
-  // Create test frame objects, where the second and fourth frame is marked
-  // as dropped at rendering.
-  std::vector<Frame*> frames;
-  Frame first_frame(0, kFrameLength);
-  Frame second_frame(0, kFrameLength);
-  Frame third_frame(0, kFrameLength);
-  Frame fourth_frame(0, kFrameLength);
-
-  second_frame.dropped_at_render = true;
-  fourth_frame.dropped_at_render = true;
-
-  frames.push_back(&first_frame);
-  frames.push_back(&second_frame);
-  frames.push_back(&third_frame);
-  frames.push_back(&fourth_frame);
-
-  // Prepare data for the first and third frames:
-  uint8_t first_frame_data[kFrameLength];
-  memset(first_frame_data, 5, kFrameLength);  // Fill it with 5's to identify.
-  uint8_t third_frame_data[kFrameLength];
-  memset(third_frame_data, 7, kFrameLength);  // Fill it with 7's to identify.
-
-  // Write the first and third frames to the temporary file. This means the fix
-  // method should add two frames of data by filling the file with data from
-  // the first and third frames after executing.
-  webrtc::test::FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
-  EXPECT_TRUE(frame_writer.Init());
-  EXPECT_TRUE(frame_writer.WriteFrame(first_frame_data));
-  EXPECT_TRUE(frame_writer.WriteFrame(third_frame_data));
-  frame_writer.Close();
-  EXPECT_EQ(2 * kFrameLength,
-            static_cast<int>(webrtc::test::GetFileSize(kOutputFilename)));
-
-  FixOutputFileForComparison(kOutputFilename, kFrameLength, frames);
-
-  // Verify that the output file has correct size.
-  EXPECT_EQ(4 * kFrameLength,
-            static_cast<int>(webrtc::test::GetFileSize(kOutputFilename)));
-
-  webrtc::test::FrameReaderImpl frame_reader(kOutputFilename, kFrameLength);
-  frame_reader.Init();
-  uint8_t read_buffer[kFrameLength];
-  EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
-  EXPECT_EQ(0, memcmp(read_buffer, first_frame_data, kFrameLength));
-  EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
-  EXPECT_EQ(0, memcmp(read_buffer, first_frame_data, kFrameLength));
-
-  EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
-  EXPECT_EQ(0, memcmp(read_buffer, third_frame_data, kFrameLength));
-  EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
-  EXPECT_EQ(0, memcmp(read_buffer, third_frame_data, kFrameLength));
-
-  frame_reader.Close();
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc
deleted file mode 100644
index 908345d14e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
-                               char* device_name,
-                               unsigned int device_name_length,
-                               int* device_id,
-                               webrtc::VideoCaptureModule** device_video) {
-
-  bool capture_device_set = false;
-  webrtc::VideoCaptureModule::DeviceInfo *dev_info =
-      webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
-
-  const unsigned int kMaxUniqueIdLength = 256;
-  char unique_id[kMaxUniqueIdLength];
-  memset(unique_id, 0, kMaxUniqueIdLength);
-
-  for (unsigned int i = 0; i < dev_info->NumberOfDevices(); i++) {
-    EXPECT_EQ(0, dev_info->GetDeviceName(i, device_name, device_name_length,
-                                         unique_id, kMaxUniqueIdLength));
-
-    *device_video = webrtc::VideoCaptureFactory::Create(4571, unique_id);
-    EXPECT_TRUE(*device_video != NULL);
-
-    if (*device_video) {
-      (*device_video)->AddRef();
-
-      int error = capture->AllocateCaptureDevice(**device_video, *device_id);
-      if (error == 0) {
-        ViETest::Log("Using capture device: %s, captureId: %d.",
-                     device_name, *device_id);
-        capture_device_set = true;
-        break;
-      } else {
-        (*device_video)->Release();
-        (*device_video) = NULL;
-      }
-    }
-  }
-  delete dev_info;
-  EXPECT_TRUE(capture_device_set) << "Found no suitable camera on your system.";
-}
-
-void RenderInWindow(webrtc::ViERender* video_render_interface,
-                    int frame_provider_id,
-                    void* os_window,
-                    float z_index) {
-  EXPECT_EQ(0,
-            video_render_interface->AddRenderer(frame_provider_id, os_window,
-                                                z_index, 0.0, 0.0, 1.0, 1.0));
-  EXPECT_EQ(0, video_render_interface->StartRender(frame_provider_id));
-}
-
-void StopRenderInWindow(webrtc::ViERender* video_render_interface,
-                        int frame_provider_id) {
-  EXPECT_EQ(0, video_render_interface->StopRender(frame_provider_id));
-  EXPECT_EQ(0, video_render_interface->RemoveRenderer(frame_provider_id));
-}
-
-void RenderToFile(webrtc::ViERender* renderer_interface,
-                  int frame_provider_id,
-                  ViEToFileRenderer *to_file_renderer) {
-  EXPECT_EQ(0, renderer_interface->AddRenderer(
-      frame_provider_id, webrtc::kVideoI420, to_file_renderer));
-  EXPECT_EQ(0, renderer_interface->StartRender(frame_provider_id));
-}
-
-void ConfigureRtpRtcp(webrtc::ViERTP_RTCP* rtcp_interface,
-                      ProtectionMethod protection_method,
-                      int video_channel) {
-  EXPECT_EQ(0, rtcp_interface->SetRTCPStatus(video_channel,
-                                             webrtc::kRtcpCompound_RFC4585));
-  EXPECT_EQ(0, rtcp_interface->SetKeyFrameRequestMethod(
-      video_channel, webrtc::kViEKeyFrameRequestPliRtcp));
-  EXPECT_EQ(0, rtcp_interface->SetTMMBRStatus(video_channel, true));
-  switch (protection_method) {
-    case kNack:
-      EXPECT_EQ(0, rtcp_interface->SetNACKStatus(video_channel, true));
-      break;
-    case kHybridNackFec:
-      const int kRedPayloadType = 96;
-      const int kUlpFecPayloadType = 97;
-      EXPECT_EQ(0, rtcp_interface->SetHybridNACKFECStatus(video_channel,
-                                                          true,
-                                                          kRedPayloadType,
-                                                          kUlpFecPayloadType));
-      break;
-  }
-}
-
-bool FindSpecificCodec(webrtc::VideoCodecType of_type,
-                       webrtc::ViECodec* codec_interface,
-                       webrtc::VideoCodec* result) {
-
-  memset(result, 0, sizeof(webrtc::VideoCodec));
-
-  for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
-    webrtc::VideoCodec codec;
-    memset(&codec, 0, sizeof(webrtc::VideoCodec));
-    if (codec_interface->GetCodec(i, codec) != 0) {
-      return false;
-    }
-    if (codec.codecType == of_type) {
-      // Done
-      *result = codec;
-      return true;
-    }
-  }
-  // Didn't find it
-  return false;
-}
-
-void SetSuitableResolution(webrtc::VideoCodec* video_codec,
-                           int forced_codec_width,
-                           int forced_codec_height) {
-  if (forced_codec_width != kDoNotForceResolution &&
-      forced_codec_height != kDoNotForceResolution) {
-    video_codec->width = forced_codec_width;
-    video_codec->height = forced_codec_height;
-  } else if (video_codec->codecType == webrtc::kVideoCodecI420) {
-    // I420 is very bandwidth heavy, so limit it here.
-    video_codec->width = 176;
-    video_codec->height = 144;
-  } else {
-    // Otherwise go with 640x480.
-    video_codec->width = 640;
-    video_codec->height = 480;
-  }
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h
deleted file mode 100644
index cb55eb610a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
-
-class ViEToFileRenderer;
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-class VideoCaptureModule;
-class ViEBase;
-class ViECapture;
-class ViECodec;
-class ViERender;
-class ViERTP_RTCP;
-struct VideoCodec;
-}
-
-enum ProtectionMethod {
-  kNack,
-  kHybridNackFec,
-};
-
-// This constant can be used as input to various functions to not force the
-// codec resolution.
-const int kDoNotForceResolution = 0;
-
-// Finds a suitable capture device (e.g. camera) on the current system
-// and allocates it. Details about the found device are filled into the out
-// parameters. If this operation fails, device_id is assigned a negative value
-// and number_of_errors is incremented.
-void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
-                               char* device_name,
-                               const unsigned int kDeviceNameLength,
-                               int* device_id,
-                               webrtc::VideoCaptureModule** device_video);
-
-// Sets up rendering in a window previously created using a Window Manager
-// (See vie_window_manager_factory.h for more details on how to make one of
-// those). The frame provider id is a source of video frames, for instance
-// a capture device or a video channel.
-// NOTE: A call to StopRenderInWindow needs to be done in order to clear
-// up the configuration applied by this function.
-void RenderInWindow(webrtc::ViERender* video_render_interface,
-                    int frame_provider_id,
-                    void* os_window,
-                    float z_index);
-
-// Stops rendering into a window as previously set up by calling RenderInWindow.
-void StopRenderInWindow(webrtc::ViERender* video_render_interface,
-                        int frame_provider_id);
-
-// Similar in function to RenderInWindow, this function instead renders to
-// a file using a to-file-renderer. The frame provider id is a source of
-// video frames, for instance a capture device or a video channel.
-void RenderToFile(webrtc::ViERender* renderer_interface,
-                  int frame_provider_id,
-                  ViEToFileRenderer* to_file_renderer);
-
-// Configures RTP-RTCP.
-void ConfigureRtpRtcp(webrtc::ViERTP_RTCP* rtcp_interface,
-                      ProtectionMethod protection_method,
-                      int video_channel);
-
-// Finds a codec in the codec list. Returns true on success, false otherwise.
-// The resulting codec is filled into result on success but is zeroed out
-// on failure.
-bool FindSpecificCodec(webrtc::VideoCodecType of_type,
-                       webrtc::ViECodec* codec_interface,
-                       webrtc::VideoCodec* result);
-
-// Sets up the provided codec with a resolution that takes individual codec
-// quirks into account (except if the forced* variables are
-// != kDoNotForceResolution)
-void SetSuitableResolution(webrtc::VideoCodec* video_codec,
-                           int forced_codec_width,
-                           int forced_codec_height);
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers.cc
deleted file mode 100644
index b5c4ae203d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers.cc
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/primitives/input_helpers.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <algorithm>
-
-#include "gflags/gflags.h"
-
-namespace webrtc {
-
-DEFINE_string(override, "",
-    "Makes it possible to override choices or inputs. All choices and "
-    "inputs will use their default values unless you override them in this "
-    "flag's argument. There can be several comma-separated overrides specified:"
-    " Overrides are specified as \"title=option text\" for choices and "
-    "\"title=value\" for regular inputs. Note that the program will stop if "
-    "you provide input not accepted by the input's validator through this flag."
-    "\n\nExample: --override \"Enter destination IP=192.168.0.1, "
-    "Select a codec=VP8\"");
-
-class AcceptAllNonEmptyValidator : public InputValidator {
- public:
-  bool InputOk(const std::string& value) const {
-    return value.length() > 0;
-  }
-};
-
-InputBuilder::InputBuilder(const std::string& title,
-                           const InputValidator* input_validator,
-                           const OverrideRegistry& override_registry)
-    : input_source_(stdin), input_validator_(input_validator),
-      override_registry_(override_registry), default_value_(""), title_(title) {
-}
-
-InputBuilder::~InputBuilder() {
-  delete input_validator_;
-}
-
-std::string InputBuilder::AskForInput() const {
-  if (override_registry_.HasOverrideFor(title_))
-    return GetOverride();
-  if (!FLAGS_override.empty() && !default_value_.empty())
-    return default_value_;
-
-  // We don't know the answer already, so ask the user.
-  return ActuallyAskUser();
-}
-
-std::string InputBuilder::ActuallyAskUser() const {
-  printf("\n%s%s\n", title_.c_str(), additional_info_.c_str());
-
-  if (!default_value_.empty())
-    printf("Hit enter for default (%s):\n", default_value_.c_str());
-
-  printf("# ");
-  char raw_input[128];
-  if (!fgets(raw_input, 128, input_source_)) {
-    // If we get here the user probably hit CTRL+D.
-    exit(1);
-  }
-
-  std::string input = raw_input;
-  input = input.substr(0, input.size() - 1);  // Strip last \n.
-
-  if (input.empty() && !default_value_.empty())
-    return default_value_;
-
-  if (!input_validator_->InputOk(input)) {
-    printf("Invalid input. Please try again.\n");
-    return ActuallyAskUser();
-  }
-  return input;
-}
-
-InputBuilder& InputBuilder::WithInputSource(FILE* input_source) {
-  input_source_ = input_source;
-  return *this;
-}
-
-InputBuilder& InputBuilder::WithInputValidator(
-    const InputValidator* input_validator) {
-  // If there's a default value, it must be accepted by the input validator.
-  assert(default_value_.empty() || input_validator->InputOk(default_value_));
-  delete input_validator_;
-  input_validator_ = input_validator;
-  return *this;
-}
-
-InputBuilder& InputBuilder::WithDefault(const std::string& default_value) {
-  assert(input_validator_->InputOk(default_value));
-  default_value_ = default_value;
-  return *this;
-}
-
-InputBuilder& InputBuilder::WithAdditionalInfo(const std::string& info) {
-  additional_info_ = info;
-  return *this;
-}
-
-const std::string& InputBuilder::GetOverride() const {
-  const std::string& override = override_registry_.GetOverrideFor(title_);
-  if (!input_validator_->InputOk(override)) {
-    printf("Fatal: Input validator for \"%s\" does not accept override %s.\n",
-        title_.c_str(), override.c_str());
-    exit(1);
-  }
-  return override;
-}
-
-OverrideRegistry::OverrideRegistry(const std::string& overrides) {
-  std::vector<std::string> all_overrides = Split(overrides, ",");
-  std::vector<std::string>::const_iterator override = all_overrides.begin();
-  for (; override != all_overrides.end(); ++override) {
-    std::vector<std::string> key_value = Split(*override, "=");
-    if (key_value.size() != 2) {
-      printf("Fatal: Override %s is malformed.", (*override).c_str());
-      exit(1);
-    }
-    std::string key = key_value[0];
-    std::string value = key_value[1];
-    overrides_[key] = value;
-  }
-}
-
-bool OverrideRegistry::HasOverrideFor(const std::string& title) const {
-  return overrides_.find(title) != overrides_.end();
-}
-
-const std::string& OverrideRegistry::GetOverrideFor(
-    const std::string& title) const {
-  assert(HasOverrideFor(title));
-  return (*overrides_.find(title)).second;
-}
-
-InputBuilder TypedInput(const std::string& title) {
-  static OverrideRegistry override_registry_(FLAGS_override);
-  return InputBuilder(
-      title, new AcceptAllNonEmptyValidator(), override_registry_);
-}
-
-std::vector<std::string> Split(const std::string& to_split,
-                               const std::string& delimiter) {
-  std::vector<std::string> result;
-  size_t current_pos = 0;
-  size_t next_delimiter = 0;
-  while ((next_delimiter = to_split.find(delimiter, current_pos)) !=
-      std::string::npos) {
-    std::string part = to_split.substr(
-        current_pos, next_delimiter - current_pos);
-    result.push_back(part);
-    current_pos = next_delimiter + 1;
-  }
-  std::string last_part = to_split.substr(current_pos);
-  if (!last_part.empty())
-    result.push_back(last_part);
-
-  return result;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers.h
deleted file mode 100644
index 536e708a0f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_
-
-#include <stdlib.h>
-
-#include <map>
-#include <string>
-#include <vector>
-
-#include "gflags/gflags.h"
-
-namespace webrtc {
-
-class InputValidator;
-class OverrideRegistry;
-
-// This class handles general user input to the application.
-class InputBuilder {
- public:
-  // The input builder takes ownership of the validator (but not the
-  // override registry).
-  InputBuilder(const std::string& title,
-               const InputValidator* input_validator,
-               const OverrideRegistry& override_registry);
-  ~InputBuilder();
-
-  // Ask the user for input, reads input from the input source and returns
-  // the answer. This method will keep asking the user until a correct answer
-  // is returned and is thereby guaranteed to return a response that is
-  // acceptable to the input validator.
-  //
-  // In some cases we will not actually ask the user for input, for instance
-  // if the --choose-defaults or --override flags are specified. See the
-  // definition of those flags in the .cc file for more information.
-  std::string AskForInput() const;
-
-  // Replaces the input source where we ask for input. Default is stdin.
-  InputBuilder& WithInputSource(FILE* input_source);
-  // Sets the input validator. The input builder takes ownership. If a default
-  // value has been set, it must be acceptable to this validator.
-  InputBuilder& WithInputValidator(const InputValidator* input_validator);
-  // Sets a default value if the user doesn't want to give input. This value
-  // must be acceptable to the input validator.
-  InputBuilder& WithDefault(const std::string& default_value);
-  // Prints additional info after the title.
-  InputBuilder& WithAdditionalInfo(const std::string& title);
-
- private:
-  const std::string& GetOverride() const;
-  std::string ActuallyAskUser() const;
-
-  FILE* input_source_;
-  const InputValidator* input_validator_;
-  const OverrideRegistry& override_registry_;
-  std::string default_value_;
-  std::string title_;
-  std::string additional_info_;
-};
-
-// Keeps track of overrides for any input points. Overrides are passed in the
-// format Title 1=Value 1,Title 2=Value 2. Spaces are not trimmed anywhere.
-class OverrideRegistry {
- public:
-  OverrideRegistry(const std::string& overrides);
-  bool HasOverrideFor(const std::string& title) const;
-  const std::string& GetOverrideFor(const std::string& title) const;
- private:
-  typedef std::map<std::string, std::string> OverrideMap;
-  OverrideMap overrides_;
-};
-
-class InputValidator {
- public:
-  virtual ~InputValidator() {}
-
-  virtual bool InputOk(const std::string& value) const = 0;
-};
-
-// Ensures input is an integer between low and high (inclusive).
-class IntegerWithinRangeValidator : public InputValidator {
- public:
-  IntegerWithinRangeValidator(int low, int high)
-      : low_(low), high_(high) {}
-
-  bool InputOk(const std::string& input) const {
-    int value = atoi(input.c_str());
-    // Note: atoi returns 0 on failure.
-    if (value == 0 && input.length() > 0 && input[0] != '0')
-      return false;  // Probably bad input.
-    return value >= low_ && value <= high_;
-  }
-
- private:
-  int low_;
-  int high_;
-};
-
-std::vector<std::string> Split(const std::string& to_split,
-                               const std::string& delimiter);
-
-// Convenience method for creating an input builder.
-InputBuilder TypedInput(const std::string& title);
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers_unittest.cc
deleted file mode 100644
index 111764ace2..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/input_helpers_unittest.cc
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video_engine/test/auto_test/primitives/fake_stdin.h"
-#include "webrtc/video_engine/test/auto_test/primitives/input_helpers.h"
-
-namespace webrtc {
-
-class InputHelpersTest: public testing::Test {
-};
-
-TEST_F(InputHelpersTest, AcceptsAnyInputExceptEmptyByDefault) {
-  FILE* fake_stdin = FakeStdin("\n\nWhatever\n");
-  std::string result = TypedInput("Title")
-      .WithInputSource(fake_stdin).AskForInput();
-  EXPECT_EQ("Whatever", result);
-  fclose(fake_stdin);
-}
-
-TEST_F(InputHelpersTest, ReturnsDefaultOnEmptyInputIfDefaultSet) {
-  FILE* fake_stdin = FakeStdin("\n\nWhatever\n");
-  std::string result = TypedInput("Title")
-      .WithInputSource(fake_stdin)
-      .WithDefault("MyDefault")
-      .AskForInput();
-  EXPECT_EQ("MyDefault", result);
-  fclose(fake_stdin);
-}
-
-TEST_F(InputHelpersTest, ObeysInputValidator) {
-  class ValidatorWhichOnlyAcceptsFooBar : public InputValidator {
-   public:
-    bool InputOk(const std::string& input) const {
-      return input == "FooBar";
-    }
-  };
-  FILE* fake_stdin = FakeStdin("\nFoo\nBar\nFoo Bar\nFooBar\n");
-  std::string result = TypedInput("Title")
-      .WithInputSource(fake_stdin)
-      .WithInputValidator(new ValidatorWhichOnlyAcceptsFooBar())
-      .AskForInput();
-  EXPECT_EQ("FooBar", result);
-  fclose(fake_stdin);
-}
-
-TEST_F(InputHelpersTest, OverrideRegistryParsesOverridesCorrectly) {
-  // TODO(phoglund): Ignore spaces where appropriate
-  OverrideRegistry override_registry("My Title=Value,My Choice=1");
-  EXPECT_TRUE(override_registry.HasOverrideFor("My Title"));
-  EXPECT_EQ("Value", override_registry.GetOverrideFor("My Title"));
-  EXPECT_TRUE(override_registry.HasOverrideFor("My Choice"));
-  EXPECT_EQ("1", override_registry.GetOverrideFor("My Choice"));
-  EXPECT_FALSE(override_registry.HasOverrideFor("Not Overridden"));
-}
-
-TEST_F(InputHelpersTest, ObeysOverridesBeforeAnythingElse) {
-  class CarelessValidator : public InputValidator {
-  public:
-    bool InputOk(const std::string& input) const {
-      return true;
-    }
-  };
-  FILE* fake_stdin = FakeStdin("\nFoo\nBar\nFoo Bar\nFooBar\n");
-  OverrideRegistry override_registry("My Title=Value,My Choice=1");
-  EXPECT_EQ("Value", InputBuilder("My Title",
-      new CarelessValidator(), override_registry)
-          .WithDefault("Whatever")
-          .WithInputSource(fake_stdin).AskForInput());
-  fclose(fake_stdin);
-}
-
-};
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
deleted file mode 100644
index 9ae4d4fce7..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest.cc
-//
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-
-#include <stdio.h>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-
-DEFINE_bool(include_timing_dependent_tests, true,
-            "If true, we will include tests / parts of tests that are known "
-            "to break in slow execution environments (such as valgrind).");
-
-// ViETest implementation
-FILE* ViETest::log_file_ = NULL;
-char* ViETest::log_str_ = NULL;
-
-std::string ViETest::GetResultOutputPath() {
-  return webrtc::test::OutputPath();
-}
-
-// ViEAutoTest implementation
-ViEAutoTest::ViEAutoTest(void* window1, void* window2) :
-    _window1(window1),
-    _window2(window2),
-    _renderType(webrtc::kRenderDefault),
-    _vrm1(webrtc::VideoRender::CreateVideoRender(
-        4561, window1, false, _renderType)),
-    _vrm2(webrtc::VideoRender::CreateVideoRender(
-        4562, window2, false, _renderType))
-{
-    assert(_vrm1);
-    assert(_vrm2);
-}
-
-ViEAutoTest::~ViEAutoTest()
-{
-    webrtc::VideoRender::DestroyVideoRender(_vrm1);
-    _vrm1 = NULL;
-    webrtc::VideoRender::DestroyVideoRender(_vrm2);
-    _vrm2 = NULL;
-}
-
-void ViEAutoTest::ViEStandardTest()
-{
-    ViEBaseStandardTest();
-    ViECaptureStandardTest();
-    ViECodecStandardTest();
-    ViEImageProcessStandardTest();
-    ViERenderStandardTest();
-    ViERtpRtcpStandardTest();
-}
-
-void ViEAutoTest::ViEExtendedTest()
-{
-    ViEBaseExtendedTest();
-    ViECaptureExtendedTest();
-    ViECodecExtendedTest();
-    ViEImageProcessExtendedTest();
-    ViERenderExtendedTest();
-}
-
-void ViEAutoTest::ViEAPITest()
-{
-    ViEBaseAPITest();
-    ViECaptureAPITest();
-    ViECodecAPITest();
-    ViEImageProcessAPITest();
-    ViERenderAPITest();
-    ViERtpRtcpAPITest();
-}
-
-void ViEAutoTest::PrintVideoCodec(const webrtc::VideoCodec videoCodec)
-{
-    ViETest::Log("Video Codec Information:");
-
-    switch (videoCodec.codecType)
-    {
-        case webrtc::kVideoCodecVP8:
-            ViETest::Log("\tcodecType: VP8");
-            break;
-        case webrtc::kVideoCodecVP9:
-            ViETest::Log("\tcodecType: VP9");
-            break;
-        case webrtc::kVideoCodecI420:
-            ViETest::Log("\tcodecType: I420");
-            break;
-        case webrtc::kVideoCodecH264:
-            ViETest::Log("\tcodecType: H264");
-            break;
-        case webrtc::kVideoCodecRED:
-            ViETest::Log("\tcodecType: RED");
-            break;
-        case webrtc::kVideoCodecULPFEC:
-            ViETest::Log("\tcodecType: ULPFEC");
-            break;
-        case webrtc::kVideoCodecGeneric:
-            ViETest::Log("\tcodecType: GENERIC");
-            break;
-        case webrtc::kVideoCodecUnknown:
-            ViETest::Log("\tcodecType: UNKNOWN");
-            break;
-    }
-
-    ViETest::Log("\theight: %u", videoCodec.height);
-    ViETest::Log("\tmaxBitrate: %u", videoCodec.maxBitrate);
-    ViETest::Log("\tmaxFramerate: %u", videoCodec.maxFramerate);
-    ViETest::Log("\tminBitrate: %u", videoCodec.minBitrate);
-    ViETest::Log("\tplName: %s", videoCodec.plName);
-    ViETest::Log("\tplType: %u", videoCodec.plType);
-    ViETest::Log("\tstartBitrate: %u", videoCodec.startBitrate);
-    ViETest::Log("\twidth: %u", videoCodec.width);
-    ViETest::Log("");
-}
-
-void ViEAutoTest::PrintAudioCodec(const webrtc::CodecInst audioCodec)
-{
-    ViETest::Log("Audio Codec Information:");
-    ViETest::Log("\tchannels: %u", audioCodec.channels);
-    ViETest::Log("\t: %u", audioCodec.pacsize);
-    ViETest::Log("\t: %u", audioCodec.plfreq);
-    ViETest::Log("\t: %s", audioCodec.plname);
-    ViETest::Log("\t: %d", audioCodec.pltype);
-    ViETest::Log("\t: %u", audioCodec.rate);
-    ViETest::Log("");
-}
-
-void ViEAutoTest::RenderCaptureDeviceAndOutputStream(
-    TbInterfaces* video_engine,
-    TbVideoChannel* video_channel,
-    TbCaptureDevice* capture_device) {
-  RenderInWindow(
-      video_engine->render, capture_device->captureId, _window1, 0);
-  RenderInWindow(
-      video_engine->render, video_channel->videoChannel, _window2, 1);
-}
-
-void ViEAutoTest::StopRenderCaptureDeviceAndOutputStream(
-    TbInterfaces* video_engine,
-    TbVideoChannel* video_channel,
-    TbCaptureDevice* capture_device) {
-  StopRenderInWindow(video_engine->render, capture_device->captureId);
-  StopRenderInWindow(video_engine->render, video_channel->videoChannel);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
deleted file mode 100644
index e69a3ebc9f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
+++ /dev/null
@@ -1,176 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_android.h"
-
-#include <android/log.h>
-#include <stdio.h>
-
-#include "webrtc/modules/video_capture/video_capture_internal.h"
-#include "webrtc/modules/video_render/video_render_internal.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-
-int ViEAutoTestAndroid::RunAutotest(int testSelection, int subTestSelection,
-                                    void* window1, void* window2,
-                                    JavaVM* javaVM, void* env, void* context) {
-  ViEAutoTest vieAutoTest(window1, window2);
-  ViETest::Log("RunAutoTest(%d, %d)", testSelection, subTestSelection);
-  webrtc::SetCaptureAndroidVM(javaVM, static_cast<jobject>(context));
-  webrtc::SetRenderAndroidVM(javaVM);
-#ifndef WEBRTC_ANDROID_OPENSLES
-  // voice engine calls into ADM directly
-  webrtc::VoiceEngine::SetAndroidObjects(javaVM, context);
-#endif
-
-  if (subTestSelection == 0) {
-    // Run all selected test
-    switch (testSelection) {
-      case 0:
-        vieAutoTest.ViEStandardTest();
-        break;
-      case 1:
-        vieAutoTest.ViEAPITest();
-        break;
-      case 2:
-        vieAutoTest.ViEExtendedTest();
-        break;
-      case 3:
-        vieAutoTest.ViELoopbackCall();
-        break;
-      default:
-        break;
-    }
-  }
-
-  switch (testSelection) {
-    case 0: // Specific standard test
-      switch (subTestSelection) {
-        case 1: // base
-          vieAutoTest.ViEBaseStandardTest();
-          break;
-
-        case 2: // capture
-          vieAutoTest.ViECaptureStandardTest();
-          break;
-
-        case 3: // codec
-          vieAutoTest.ViECodecStandardTest();
-          break;
-
-        case 6: // image process
-          vieAutoTest.ViEImageProcessStandardTest();
-          break;
-
-#if 0  // vie_autotest_network.cc isn't actually pulled into the build at all!
-        case 7: // network
-          vieAutoTest.ViENetworkStandardTest();
-          break;
-#endif
-
-        case 8: // Render
-          vieAutoTest.ViERenderStandardTest();
-          break;
-
-        case 9: // RTP/RTCP
-          vieAutoTest.ViERtpRtcpStandardTest();
-          break;
-
-        default:
-          break;
-      }
-      break;
-
-    case 1:// specific API
-      switch (subTestSelection) {
-        case 1: // base
-          vieAutoTest.ViEBaseAPITest();
-          break;
-
-        case 2: // capture
-          vieAutoTest.ViECaptureAPITest();
-          break;
-
-        case 3: // codec
-          vieAutoTest.ViECodecAPITest();
-          break;
-
-        case 6: // image process
-          vieAutoTest.ViEImageProcessAPITest();
-          break;
-
-#if 0  // vie_autotest_network.cc isn't actually pulled into the build at all!
-        case 7: // network
-          vieAutoTest.ViENetworkAPITest();
-          break;
-#endif
-
-        case 8: // Render
-          vieAutoTest.ViERenderAPITest();
-          break;
-
-        case 9: // RTP/RTCP
-          vieAutoTest.ViERtpRtcpAPITest();
-          break;
-        case 10:
-          break;
-
-        default:
-          break;
-      }
-      break;
-
-    case 2:// specific extended
-      switch (subTestSelection) {
-        case 1: // base
-          vieAutoTest.ViEBaseExtendedTest();
-          break;
-
-        case 2: // capture
-          vieAutoTest.ViECaptureExtendedTest();
-          break;
-
-        case 3: // codec
-          vieAutoTest.ViECodecExtendedTest();
-          break;
-
-        case 6: // image process
-          vieAutoTest.ViEImageProcessExtendedTest();
-          break;
-
-        case 7: // Render
-          vieAutoTest.ViERenderExtendedTest();
-          break;
-
-        case 8: // RTP/RTCP
-          // Note that this test is removed. It hasn't been properly cleaned up
-          // because this hopefully going away soon.
-          break;
-
-        default:
-          break;
-      }
-      break;
-
-    case 3:
-      vieAutoTest.ViELoopbackCall();
-      break;
-
-    default:
-      break;
-    }
-
-  return 0;
-}
-
-int main(int argc, char** argv) {
-  // TODO(leozwang): Add real tests here
-  return 0;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc
deleted file mode 100644
index 3a94e051be..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc
+++ /dev/null
@@ -1,237 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/primitives/base_primitives.h"
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-
-void ViEAutoTest::ViEBaseStandardTest() {
-  // ***************************************************************
-  // Begin create/initialize WebRTC Video Engine for testing
-  // ***************************************************************
-
-  TbInterfaces interfaces("ViEBaseStandardTest");
-
-  // ***************************************************************
-  // Engine ready. Set up the test case:
-  // ***************************************************************
-  int video_channel = -1;
-  EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
-
-  webrtc::VideoCaptureModule* video_capture_module = NULL;
-  const unsigned int kMaxDeviceNameLength = 128;
-  char device_name[kMaxDeviceNameLength];
-  memset(device_name, 0, kMaxDeviceNameLength);
-  int capture_id;
-
-  webrtc::ViEBase* base_interface = interfaces.base;
-  webrtc::ViERender* render_interface = interfaces.render;
-  webrtc::ViECapture* capture_interface = interfaces.capture;
-
-  FindCaptureDeviceOnSystem(capture_interface,
-                            device_name,
-                            kMaxDeviceNameLength,
-                            &capture_id,
-                            &video_capture_module);
-
-  EXPECT_TRUE(video_capture_module);
-  if (!video_capture_module)
-    return;
-
-  EXPECT_EQ(0, capture_interface->ConnectCaptureDevice(capture_id,
-                                                       video_channel));
-  EXPECT_EQ(0, capture_interface->StartCapture(capture_id));
-
-  ConfigureRtpRtcp(interfaces.rtp_rtcp, kNack, video_channel);
-
-  EXPECT_EQ(0, render_interface->RegisterVideoRenderModule(*_vrm1));
-  EXPECT_EQ(0, render_interface->RegisterVideoRenderModule(*_vrm2));
-
-  RenderInWindow(render_interface, capture_id, _window1, 0);
-  RenderInWindow(render_interface, video_channel, _window2, 1);
-
-  // ***************************************************************
-  // Run the actual test:
-  // ***************************************************************
-  ViETest::Log("You should shortly see a local preview from camera %s"
-               " in window 1 and the remote video in window 2.", device_name);
-  ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
-                      base_interface, interfaces.network, interfaces.rtp_rtcp,
-                      video_channel, device_name);
-
-  // ***************************************************************
-  // Testing finished. Tear down Video Engine
-  // ***************************************************************
-  EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
-  EXPECT_EQ(0, capture_interface->StopCapture(capture_id));
-  EXPECT_EQ(0, base_interface->StopReceive(video_channel));
-
-  EXPECT_EQ(0, render_interface->StopRender(video_channel));
-  EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
-  EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
-
-  EXPECT_EQ(0, render_interface->DeRegisterVideoRenderModule(*_vrm1));
-  EXPECT_EQ(0, render_interface->DeRegisterVideoRenderModule(*_vrm2));
-
-  EXPECT_EQ(0, capture_interface->ReleaseCaptureDevice(capture_id));
-
-  video_capture_module->Release();
-  video_capture_module = NULL;
-
-  EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
-}
-
-void ViEAutoTest::ViEBaseExtendedTest() {
-  // Start with standard test
-  ViEBaseAPITest();
-  ViEBaseStandardTest();
-}
-
-void ViEAutoTest::ViEBaseAPITest() {
-  // ***************************************************************
-  // Begin create/initialize WebRTC Video Engine for testing
-  // ***************************************************************
-  // Get the ViEBase API
-  webrtc::ViEBase* vie_base = webrtc::ViEBase::GetInterface(NULL);
-  EXPECT_EQ(NULL, vie_base) << "Should return null for a bad ViE pointer";
-
-  webrtc::VideoEngine* video_engine = webrtc::VideoEngine::Create();
-  EXPECT_TRUE(NULL != video_engine);
-
-  std::string trace_file_path =
-    ViETest::GetResultOutputPath() + "ViEBaseAPI_trace.txt";
-  EXPECT_EQ(0, video_engine->SetTraceFile(trace_file_path.c_str()));
-
-  vie_base = webrtc::ViEBase::GetInterface(video_engine);
-  EXPECT_TRUE(NULL != vie_base);
-
-  webrtc::ViENetwork* vie_network =
-      webrtc::ViENetwork::GetInterface(video_engine);
-  EXPECT_TRUE(vie_network != NULL);
-
-  webrtc::ViERTP_RTCP* vie_rtp =
-      webrtc::ViERTP_RTCP::GetInterface(video_engine);
-  EXPECT_TRUE(vie_rtp != NULL);
-
-  // ***************************************************************
-  // Engine ready. Begin testing class
-  // ***************************************************************
-  char version[1024] = "";
-  EXPECT_EQ(0, vie_base->GetVersion(version));
-  EXPECT_EQ(0, vie_base->LastError());
-
-  int video_channel = -1;
-  EXPECT_EQ(0, vie_base->Init());
-  EXPECT_EQ(0, vie_base->CreateChannel(video_channel));
-
-  int video_channel2 = -1;
-  int video_channel3 = -1;
-  EXPECT_EQ(0, vie_base->CreateChannel(video_channel2));
-  EXPECT_NE(video_channel, video_channel2) <<
-      "Should allocate new number for independent channel";
-
-  EXPECT_EQ(0, vie_base->DeleteChannel(video_channel2));
-
-  EXPECT_EQ(-1, vie_base->CreateChannel(video_channel2, video_channel + 1))
-      << "Should fail since neither channel exists (the second must)";
-
-  // Create a receive only channel and a send channel. Verify we can't send on
-  // the receive only channel.
-  EXPECT_EQ(0, vie_base->CreateReceiveChannel(video_channel2,
-                                              video_channel));
-  EXPECT_EQ(0, vie_base->CreateChannel(video_channel3, video_channel));
-
-  const char* ip_address = "127.0.0.1\0";
-  const int send_port = 1234;
-
-  EXPECT_EQ(0, vie_rtp->SetLocalSSRC(video_channel, 1));
-  EXPECT_EQ(0, vie_rtp->SetLocalSSRC(video_channel, 2));
-  EXPECT_EQ(0, vie_rtp->SetLocalSSRC(video_channel, 3));
-
-  webrtc::test::VideoChannelTransport* video_channel_transport_1 =
-      new webrtc::test::VideoChannelTransport(vie_network, video_channel);
-
-  ASSERT_EQ(0, video_channel_transport_1->SetSendDestination(ip_address,
-                                                             send_port));
-
-  webrtc::test::VideoChannelTransport* video_channel_transport_2 =
-      new webrtc::test::VideoChannelTransport(vie_network, video_channel2);
-
-  webrtc::test::VideoChannelTransport* video_channel_transport_3 =
-      new webrtc::test::VideoChannelTransport(vie_network, video_channel3);
-
-  ASSERT_EQ(0, video_channel_transport_3->SetSendDestination(ip_address,
-                                                             send_port + 4));
-
-  EXPECT_EQ(0, vie_base->StartSend(video_channel));
-  EXPECT_EQ(-1, vie_base->StartSend(video_channel2));
-  EXPECT_EQ(0, vie_base->StartSend(video_channel3));
-  EXPECT_EQ(0, vie_base->StopSend(video_channel));
-  EXPECT_EQ(0, vie_base->StopSend(video_channel3));
-
-  // Test Voice Engine integration with Video Engine.
-  webrtc::VoiceEngine* voice_engine = NULL;
-  webrtc::VoEBase* voe_base = NULL;
-  int audio_channel = -1;
-
-  voice_engine = webrtc::VoiceEngine::Create();
-  EXPECT_TRUE(NULL != voice_engine);
-
-  voe_base = webrtc::VoEBase::GetInterface(voice_engine);
-  EXPECT_TRUE(NULL != voe_base);
-  EXPECT_EQ(0, voe_base->Init());
-
-  audio_channel = voe_base->CreateChannel();
-  EXPECT_NE(-1, audio_channel);
-
-  // Connect before setting VoE.
-  EXPECT_NE(0, vie_base->ConnectAudioChannel(video_channel, audio_channel))
-      << "Should fail since Voice Engine is not set yet.";
-
-  // Then do it right.
-  EXPECT_EQ(0, vie_base->SetVoiceEngine(voice_engine));
-  EXPECT_EQ(0, vie_base->ConnectAudioChannel(video_channel, audio_channel));
-
-  // ***************************************************************
-  // Testing finished. Tear down Video Engine
-  // ***************************************************************
-  EXPECT_NE(0, vie_base->DisconnectAudioChannel(video_channel + 5)) <<
-      "Should fail: disconnecting bogus channel";
-
-  EXPECT_EQ(0, vie_base->DisconnectAudioChannel(video_channel));
-
-  // Clean up voice engine
-  EXPECT_EQ(0, vie_rtp->Release());
-  EXPECT_EQ(0, vie_network->Release());
-  EXPECT_EQ(0, vie_base->SetVoiceEngine(NULL));
-  // VoiceEngine reference counting is per object, not per interface, so
-  // Release should return != 0.
-  EXPECT_NE(0, voe_base->Release());
-  EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine));
-
-  webrtc::ViEBase* vie_base2 = webrtc::ViEBase::GetInterface(video_engine);
-  EXPECT_TRUE(NULL != vie_base2);
-
-  EXPECT_EQ(1, vie_base->Release()) <<
-      "There should be one interface left.";
-
-  EXPECT_FALSE(webrtc::VideoEngine::Delete(video_engine)) <<
-      "Should fail since there are interfaces left.";
-
-  delete video_channel_transport_1;
-  delete video_channel_transport_2;
-  delete video_channel_transport_3;
-  EXPECT_EQ(0, vie_base->Release());
-  EXPECT_TRUE(webrtc::VideoEngine::Delete(video_engine));
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc
deleted file mode 100644
index 68b1afda02..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc
+++ /dev/null
@@ -1,542 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gflags/gflags.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-
-DEFINE_bool(capture_test_ensure_resolution_alignment_in_capture_device, true,
-            "If true, we will give resolutions slightly below a reasonable "
-            "value to test the camera's ability to choose a good resolution. "
-            "If false, we will provide reasonable resolutions instead.");
-
-class CaptureObserver : public webrtc::ViECaptureObserver {
- public:
-  CaptureObserver()
-      : brightness_(webrtc::Normal),
-        alarm_(webrtc::AlarmCleared),
-        frame_rate_(0) {}
-
-  virtual void BrightnessAlarm(const int capture_id,
-                               const webrtc::Brightness brightness) {
-    brightness_ = brightness;
-    switch (brightness) {
-      case webrtc::Normal:
-        ViETest::Log("  BrightnessAlarm Normal");
-        break;
-      case webrtc::Bright:
-        ViETest::Log("  BrightnessAlarm Bright");
-        break;
-      case webrtc::Dark:
-        ViETest::Log("  BrightnessAlarm Dark");
-        break;
-    }
-  }
-
-  virtual void CapturedFrameRate(const int capture_id,
-                                 const unsigned char frame_rate) {
-    ViETest::Log("  CapturedFrameRate %u", frame_rate);
-    frame_rate_ = frame_rate;
-  }
-
-  virtual void NoPictureAlarm(const int capture_id,
-                              const webrtc::CaptureAlarm alarm) {
-    alarm_ = alarm;
-    if (alarm == webrtc::AlarmRaised) {
-      ViETest::Log("NoPictureAlarm CARaised.");
-    } else {
-      ViETest::Log("NoPictureAlarm CACleared.");
-    }
-  }
-
-  webrtc::Brightness brightness_;
-  webrtc::CaptureAlarm alarm_;
-  unsigned char frame_rate_;
-};
-
-class CaptureEffectFilter : public webrtc::ViEEffectFilter {
- public:
-  CaptureEffectFilter(unsigned int expected_width, unsigned int expected_height)
-      : number_of_captured_frames_(0),
-        expected_width_(expected_width),
-        expected_height_(expected_height) {
-  }
-
-  // Implements video_engineEffectFilter.
-  virtual int Transform(size_t size,
-                        unsigned char* frame_buffer,
-                        int64_t ntp_time_ms,
-                        unsigned int timestamp,
-                        unsigned int width,
-                        unsigned int height) {
-    EXPECT_TRUE(frame_buffer != NULL);
-    EXPECT_EQ(expected_width_, width);
-    EXPECT_EQ(expected_height_, height);
-    ++number_of_captured_frames_;
-    return 0;
-  }
-
-  int number_of_captured_frames_;
-
- protected:
-  unsigned int expected_width_;
-  unsigned int expected_height_;
-};
-
-void ViEAutoTest::ViECaptureStandardTest() {
-  /// **************************************************************
-  //  Begin create/initialize WebRTC Video Engine for testing
-  /// **************************************************************
-
-  /// **************************************************************
-  //  Engine ready. Begin testing class
-  /// **************************************************************
-
-  TbInterfaces video_engine("video_engineCaptureStandardTest");
-
-  webrtc::VideoCaptureModule::DeviceInfo* dev_info =
-      webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
-  ASSERT_TRUE(dev_info != NULL);
-
-  int number_of_capture_devices = dev_info->NumberOfDevices();
-  ViETest::Log("Number of capture devices %d",
-                        number_of_capture_devices);
-  ASSERT_GT(number_of_capture_devices, 0)
-      << "This test requires a capture device (i.e. a webcam)";
-
-#if !defined(WEBRTC_MAC)
-  int capture_device_id[10] = {0};
-  webrtc::VideoCaptureModule* vcpms[10] = {0};
-#endif
-
-  // Check capabilities
-  for (int device_index = 0; device_index < number_of_capture_devices;
-       ++device_index) {
-    char device_name[128];
-    char device_unique_name[512];
-
-    EXPECT_EQ(0, dev_info->GetDeviceName(device_index,
-                                         device_name,
-                                         sizeof(device_name),
-                                         device_unique_name,
-                                         sizeof(device_unique_name)));
-    ViETest::Log("Found capture device %s\nUnique name %s",
-                          device_name, device_unique_name);
-
-#if !defined(WEBRTC_MAC)  // these functions will return -1
-    int number_of_capabilities =
-        dev_info->NumberOfCapabilities(device_unique_name);
-    EXPECT_GT(number_of_capabilities, 0);
-
-    for (int cap_index = 0; cap_index < number_of_capabilities; ++cap_index) {
-      webrtc::VideoCaptureCapability capability;
-      EXPECT_EQ(0, dev_info->GetCapability(device_unique_name, cap_index,
-                                           capability));
-      ViETest::Log("Capture capability %d (of %u)", cap_index + 1,
-                   number_of_capabilities);
-      ViETest::Log("width %d, height %d, frame rate %d",
-                   capability.width, capability.height, capability.maxFPS);
-      ViETest::Log("expected delay %d, color type %d, encoding %d",
-                   capability.expectedCaptureDelay, capability.rawType,
-                   capability.codecType);
-      EXPECT_GT(capability.width, 0);
-      EXPECT_GT(capability.height, 0);
-      EXPECT_GT(capability.maxFPS, -1);  // >= 0
-      EXPECT_GT(capability.expectedCaptureDelay, 0);
-    }
-#endif
-  }
-  // Capture Capability Functions are not supported on WEBRTC_MAC.
-#if !defined(WEBRTC_MAC)
-
-  // Check allocation. Try to allocate them all after each other.
-  for (int device_index = 0; device_index < number_of_capture_devices;
-       ++device_index) {
-    char device_name[128];
-    char device_unique_name[512];
-    EXPECT_EQ(0, dev_info->GetDeviceName(device_index,
-                                         device_name,
-                                         sizeof(device_name),
-                                         device_unique_name,
-                                         sizeof(device_unique_name)));
-    webrtc::VideoCaptureModule* vcpm =
-        webrtc::VideoCaptureFactory::Create(device_index, device_unique_name);
-    EXPECT_TRUE(vcpm != NULL);
-    if (!vcpm)
-      continue;
-
-    vcpm->AddRef();
-    vcpms[device_index] = vcpm;
-
-    EXPECT_EQ(0, video_engine.capture->AllocateCaptureDevice(
-        *vcpm, capture_device_id[device_index]));
-
-    webrtc::VideoCaptureCapability capability;
-    EXPECT_EQ(0, dev_info->GetCapability(device_unique_name, 0, capability));
-
-    // Test that the camera select the closest capability to the selected
-    // width and height.
-    CaptureEffectFilter filter(capability.width, capability.height);
-    EXPECT_EQ(0, video_engine.image_process->RegisterCaptureEffectFilter(
-        capture_device_id[device_index], filter));
-
-    ViETest::Log("Testing Device %s capability width %d  height %d",
-                 device_unique_name, capability.width, capability.height);
-
-    if (FLAGS_capture_test_ensure_resolution_alignment_in_capture_device) {
-      // This tests that the capture device properly aligns to a
-      // multiple of 16 (or at least 8).
-      capability.height = capability.height - 2;
-      capability.width  = capability.width  - 2;
-    }
-
-    webrtc::CaptureCapability vie_capability;
-    vie_capability.width = capability.width;
-    vie_capability.height = capability.height;
-    vie_capability.codecType = capability.codecType;
-    vie_capability.maxFPS = capability.maxFPS;
-    vie_capability.rawType = capability.rawType;
-
-    EXPECT_EQ(0, video_engine.capture->StartCapture(
-        capture_device_id[device_index], vie_capability));
-    webrtc::TickTime start_time = webrtc::TickTime::Now();
-
-    while (filter.number_of_captured_frames_ < 10 &&
-           (webrtc::TickTime::Now() - start_time).Milliseconds() < 10000) {
-      AutoTestSleep(100);
-    }
-
-    EXPECT_GT(filter.number_of_captured_frames_, 9)
-        << "Should capture at least some frames";
-
-    EXPECT_EQ(0, video_engine.image_process->DeregisterCaptureEffectFilter(
-        capture_device_id[device_index]));
-
-#ifdef WEBRTC_ANDROID  // Can only allocate one camera at the time on Android.
-    EXPECT_EQ(0, video_engine.capture->StopCapture(
-        capture_device_id[device_index]));
-    EXPECT_EQ(0, video_engine.capture->ReleaseCaptureDevice(
-        capture_device_id[device_index]));
-#endif
-  }
-
-  /// **************************************************************
-  //  Testing finished. Tear down Video Engine
-  /// **************************************************************
-  delete dev_info;
-
-  // Stop all started capture devices.
-  for (int device_index = 0; device_index < number_of_capture_devices;
-       ++device_index) {
-#if !defined(WEBRTC_ANDROID)
-    // Don't stop on Android since we can only allocate one camera.
-    EXPECT_EQ(0, video_engine.capture->StopCapture(
-        capture_device_id[device_index]));
-    EXPECT_EQ(0, video_engine.capture->ReleaseCaptureDevice(
-        capture_device_id[device_index]));
-#endif  // !WEBRTC_ANDROID
-    if (vcpms[device_index])
-      vcpms[device_index]->Release();
-  }
-#endif  // !WEBRTC_MAC
-}
-
-void ViEAutoTest::ViECaptureExtendedTest() {
-  ViECaptureExternalCaptureTest();
-}
-
-void ViEAutoTest::ViECaptureAPITest() {
-  /// **************************************************************
-  //  Begin create/initialize WebRTC Video Engine for testing
-  /// **************************************************************
-
-  /// **************************************************************
-  //  Engine ready. Begin testing class
-  /// **************************************************************
-  TbInterfaces video_engine("video_engineCaptureAPITest");
-
-  video_engine.capture->NumberOfCaptureDevices();
-
-  char device_name[128];
-  char device_unique_name[512];
-  int capture_id = 0;
-
-  webrtc::VideoCaptureModule::DeviceInfo* dev_info =
-      webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
-  ASSERT_TRUE(dev_info != NULL);
-  ASSERT_GT(dev_info->NumberOfDevices(), 0u)
-      << "This test requires a capture device (i.e. a webcam)";
-
-  // Get the first capture device
-  EXPECT_EQ(0, dev_info->GetDeviceName(0, device_name,
-                                       sizeof(device_name),
-                                       device_unique_name,
-                                       sizeof(device_unique_name)));
-
-  webrtc::VideoCaptureModule* vcpm =
-      webrtc::VideoCaptureFactory::Create(0, device_unique_name);
-  vcpm->AddRef();
-  EXPECT_TRUE(vcpm != NULL);
-
-  // Allocate capture device.
-  EXPECT_EQ(0, video_engine.capture->AllocateCaptureDevice(*vcpm, capture_id));
-
-  // Start the capture device.
-  EXPECT_EQ(0, video_engine.capture->StartCapture(capture_id));
-
-  // Start again. Should fail.
-  EXPECT_NE(0, video_engine.capture->StartCapture(capture_id));
-  EXPECT_EQ(kViECaptureDeviceAlreadyStarted, video_engine.LastError());
-
-  // Start invalid capture device.
-  EXPECT_NE(0, video_engine.capture->StartCapture(capture_id + 1));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-
-  // Stop invalid capture device.
-  EXPECT_NE(0, video_engine.capture->StopCapture(capture_id + 1));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-
-  // Stop the capture device.
-  EXPECT_EQ(0, video_engine.capture->StopCapture(capture_id));
-
-  // Stop the capture device again.
-  EXPECT_NE(0, video_engine.capture->StopCapture(capture_id));
-  EXPECT_EQ(kViECaptureDeviceNotStarted, video_engine.LastError());
-
-  // Connect to invalid channel.
-  EXPECT_NE(0, video_engine.capture->ConnectCaptureDevice(capture_id, 0));
-  EXPECT_EQ(kViECaptureDeviceInvalidChannelId,
-            video_engine.LastError());
-
-  TbVideoChannel channel(video_engine);
-
-  // Connect invalid capture_id.
-  EXPECT_NE(0, video_engine.capture->ConnectCaptureDevice(capture_id + 1,
-                                                 channel.videoChannel));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-
-  // Connect the capture device to the channel.
-  EXPECT_EQ(0, video_engine.capture->ConnectCaptureDevice(capture_id,
-                                                 channel.videoChannel));
-
-  // Connect the channel again.
-  EXPECT_NE(0, video_engine.capture->ConnectCaptureDevice(capture_id,
-                                                 channel.videoChannel));
-  EXPECT_EQ(kViECaptureDeviceAlreadyConnected,
-            video_engine.LastError());
-
-  // Start the capture device.
-  EXPECT_EQ(0, video_engine.capture->StartCapture(capture_id));
-
-  // Release invalid capture device.
-  EXPECT_NE(0, video_engine.capture->ReleaseCaptureDevice(capture_id + 1));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-
-  // Release the capture device.
-  EXPECT_EQ(0, video_engine.capture->ReleaseCaptureDevice(capture_id));
-
-  // Release the capture device again.
-  EXPECT_NE(0, video_engine.capture->ReleaseCaptureDevice(capture_id));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-
-  // Test GetOrientation.
-  webrtc::VideoRotation orientation;
-  char dummy_name[5];
-  EXPECT_NE(0, dev_info->GetOrientation(dummy_name, orientation));
-
-  // Test SetRotation.
-  EXPECT_NE(0, video_engine.capture->SetVideoRotation(
-                   capture_id, webrtc::kVideoRotation_90));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-
-  // Allocate capture device.
-  EXPECT_EQ(0, video_engine.capture->AllocateCaptureDevice(*vcpm, capture_id));
-
-  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
-                   capture_id, webrtc::kVideoRotation_0));
-  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
-                   capture_id, webrtc::kVideoRotation_90));
-  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
-                   capture_id, webrtc::kVideoRotation_180));
-  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
-                   capture_id, webrtc::kVideoRotation_270));
-
-  // Release the capture device
-  EXPECT_EQ(0, video_engine.capture->ReleaseCaptureDevice(capture_id));
-
-  /// **************************************************************
-  //  Testing finished. Tear down Video Engine
-  /// **************************************************************
-  delete dev_info;
-  vcpm->Release();
-}
-
-void ViEAutoTest::ViECaptureExternalCaptureTest() {
-  /// **************************************************************
-  //  Begin create/initialize WebRTC Video Engine for testing
-  /// **************************************************************
-
-  TbInterfaces video_engine("video_engineCaptureExternalCaptureTest");
-  TbVideoChannel channel(video_engine);
-  channel.StartReceive();
-  channel.StartSend();
-
-  webrtc::VideoCaptureExternal* external_capture = NULL;
-  int capture_id = 0;
-
-  // Allocate the external capture device.
-  webrtc::VideoCaptureModule* vcpm =
-      webrtc::VideoCaptureFactory::Create(0, external_capture);
-  EXPECT_TRUE(vcpm != NULL);
-  EXPECT_TRUE(external_capture != NULL);
-  vcpm->AddRef();
-
-  EXPECT_EQ(0, video_engine.capture->AllocateCaptureDevice(*vcpm, capture_id));
-
-  // Connect the capture device to the channel.
-  EXPECT_EQ(0, video_engine.capture->ConnectCaptureDevice(capture_id,
-                                                 channel.videoChannel));
-
-  // Render the local capture.
-  EXPECT_EQ(0, video_engine.render->AddRenderer(capture_id, _window1, 1, 0.0,
-                                                0.0, 1.0, 1.0));
-
-  // Render the remote capture.
-  EXPECT_EQ(0, video_engine.render->AddRenderer(channel.videoChannel, _window2,
-                                                1, 0.0, 0.0, 1.0, 1.0));
-  EXPECT_EQ(0, video_engine.render->StartRender(capture_id));
-  EXPECT_EQ(0, video_engine.render->StartRender(channel.videoChannel));
-
-  // Register observer.
-  CaptureObserver observer;
-  EXPECT_EQ(0, video_engine.capture->RegisterObserver(capture_id, observer));
-
-  // Enable brightness alarm.
-  EXPECT_EQ(0, video_engine.capture->EnableBrightnessAlarm(capture_id, true));
-
-  CaptureEffectFilter effect_filter(176, 144);
-  EXPECT_EQ(0, video_engine.image_process->RegisterCaptureEffectFilter(
-      capture_id, effect_filter));
-
-  // Call started.
-  ViETest::Log("You should see local preview from external capture\n"
-               "in window 1 and the remote video in window 2.\n");
-
-  /// **************************************************************
-  //  Engine ready. Begin testing class
-  /// **************************************************************
-  const size_t video_frame_length = (176 * 144 * 3) / 2;
-  unsigned char* video_frame = new unsigned char[video_frame_length];
-  memset(video_frame, 128, 176 * 144);
-
-  int frame_count = 0;
-  webrtc::VideoCaptureCapability capability;
-  capability.width = 176;
-  capability.height = 144;
-  capability.rawType = webrtc::kVideoI420;
-
-  ViETest::Log("Testing external capturing and frame rate callbacks.");
-  // TODO(mflodman) Change when using a real file!
-  // while (fread(video_frame, video_frame_length, 1, foreman) == 1)
-  while (frame_count < 120) {
-    external_capture->IncomingFrame(
-        video_frame, video_frame_length, capability,
-        webrtc::TickTime::MillisecondTimestamp());
-    AutoTestSleep(33);
-
-    if (effect_filter.number_of_captured_frames_ > 2) {
-      EXPECT_EQ(webrtc::Normal, observer.brightness_) <<
-          "Brightness or picture alarm should not have been called yet.";
-      EXPECT_EQ(webrtc::AlarmCleared, observer.alarm_) <<
-          "Brightness or picture alarm should not have been called yet.";
-    }
-    frame_count++;
-  }
-
-  // Test brightness alarm.
-  // Test bright image.
-  for (int i = 0; i < 176 * 144; ++i) {
-    if (video_frame[i] <= 155)
-      video_frame[i] = video_frame[i] + 100;
-    else
-      video_frame[i] = 255;
-  }
-  ViETest::Log("Testing Brighness alarm");
-  for (int frame = 0; frame < 30; ++frame) {
-    external_capture->IncomingFrame(
-        video_frame, video_frame_length, capability,
-        webrtc::TickTime::MillisecondTimestamp());
-    AutoTestSleep(33);
-  }
-  EXPECT_EQ(webrtc::Bright, observer.brightness_) <<
-      "Should be bright at this point since we are using a bright image.";
-
-  // Test Dark image
-  for (int i = 0; i < 176 * 144; ++i) {
-    video_frame[i] = video_frame[i] > 200 ? video_frame[i] - 200 : 0;
-  }
-  for (int frame = 0; frame < 30; ++frame) {
-    external_capture->IncomingFrame(
-        video_frame, video_frame_length, capability,
-        webrtc::TickTime::MillisecondTimestamp());
-    AutoTestSleep(33);
-  }
-  EXPECT_EQ(webrtc::Dark, observer.brightness_) <<
-      "Should be dark at this point since we are using a dark image.";
-  EXPECT_GT(effect_filter.number_of_captured_frames_, 150) <<
-      "Frames should have been played.";
-
-  EXPECT_GE(observer.frame_rate_, 29) <<
-      "Frame rate callback should be approximately correct.";
-  EXPECT_LE(observer.frame_rate_, 30) <<
-      "Frame rate callback should be approximately correct.";
-
-  // Test no picture alarm
-  ViETest::Log("Testing NoPictureAlarm.");
-  AutoTestSleep(1050);
-
-  EXPECT_EQ(webrtc::AlarmRaised, observer.alarm_) <<
-      "No picture alarm should be raised.";
-  for (int frame = 0; frame < 10; ++frame) {
-    external_capture->IncomingFrame(
-        video_frame, video_frame_length, capability,
-        webrtc::TickTime::MillisecondTimestamp());
-    AutoTestSleep(33);
-  }
-  EXPECT_EQ(webrtc::AlarmCleared, observer.alarm_) <<
-  "Alarm should be cleared since ge just got some data.";
-
-  delete video_frame;
-
-  // Release the capture device
-  EXPECT_EQ(0, video_engine.capture->ReleaseCaptureDevice(capture_id));
-
-  // Release the capture device again
-  EXPECT_NE(0, video_engine.capture->ReleaseCaptureDevice(capture_id));
-  EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
-  vcpm->Release();
-
-  /// **************************************************************
-  //  Testing finished. Tear down Video Engine
-  /// **************************************************************
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm
deleted file mode 100644
index 15afe5ceb7..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-
-#import "webrtc/modules/video_render/mac/cocoa_render_view.h"
-#import "webrtc/test/testsupport/mac/run_threaded_main_mac.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
-
-@implementation TestCocoaUi
-
-// TODO(phoglund): This file probably leaks memory like crazy. Find someone
-// who understands objective-c memory management and fix it.
-
-- (void)prepareToCreateWindowsWithSize:(AutoTestRect)window1Size
-                               andSize:(AutoTestRect)window2Size
-                             withTitle:(void*)window1_title
-                              andTitle:(void*)window2_title {
-  window1Size_ = window1Size;
-  window2Size_ = window2Size;
-  window1Title_ = window1_title;
-  window2Title_ = window2_title;
-}
-
-- (void)createWindows:(NSObject*)ignored {
-  NSRect window1Frame = NSMakeRect(
-      window1Size_.origin.x, window1Size_.origin.y,
-      window1Size_.size.width, window1Size_.size.height);
-
-  window1_ = [[NSWindow alloc]
-               initWithContentRect:window1Frame
-                         styleMask:NSTitledWindowMask
-                           backing:NSBackingStoreBuffered
-                             defer:NO];
-  [window1_ orderOut:nil];
-
-  NSRect render_view1_frame = NSMakeRect(
-      0, 0, window1Size_.size.width, window1Size_.size.height);
-  cocoaRenderView1_ =
-      [[CocoaRenderView alloc] initWithFrame:render_view1_frame];
-
-  [[window1_ contentView] addSubview:(NSView*)cocoaRenderView1_];
-  [window1_ setTitle:[NSString stringWithFormat:@"%s", window1Title_]];
-  [window1_ makeKeyAndOrderFront:NSApp];
-
-  NSRect window2_frame = NSMakeRect(
-      window2Size_.origin.x, window2Size_.origin.y,
-      window2Size_.size.width, window2Size_.size.height);
-
-  window2_ = [[NSWindow alloc]
-               initWithContentRect:window2_frame
-                         styleMask:NSTitledWindowMask
-                           backing:NSBackingStoreBuffered
-                             defer:NO];
-  [window2_ orderOut:nil];
-
-  NSRect render_view2_frame = NSMakeRect(
-      0, 0, window1Size_.size.width, window1Size_.size.height);
-  cocoaRenderView2_ =
-      [[CocoaRenderView alloc] initWithFrame:render_view2_frame];
-  [[window2_ contentView] addSubview:(NSView*)cocoaRenderView2_];
-  [window2_ setTitle:[NSString stringWithFormat:@"%s", window2Title_]];
-  [window2_ makeKeyAndOrderFront:NSApp];
-}
-
-- (NSWindow*)window1 {
-  return window1_;
-}
-
-- (NSWindow*)window2 {
-  return window2_;
-}
-
-- (CocoaRenderView*)cocoaRenderView1 {
-  return cocoaRenderView1_;
-}
-
-- (CocoaRenderView*)cocoaRenderView2 {
-  return cocoaRenderView2_;
-}
-
-@end
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager() {
-  cocoa_ui_ = [[TestCocoaUi alloc] init];
-}
-
-ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
-  [cocoa_ui_ release];
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
-                                            AutoTestRect window2Size,
-                                            void* window1_title,
-                                            void* window2_title) {
-    [cocoa_ui_ prepareToCreateWindowsWithSize:window1Size
-                                      andSize:window2Size
-                                    withTitle:window1_title
-                                     andTitle:window2_title];
-    [cocoa_ui_ performSelectorOnMainThread:@selector(createWindows:)
-                                withObject:nil
-                             waitUntilDone:YES];
-    return 0;
-}
-
-int ViEAutoTestWindowManager::TerminateWindows() {
-    [[cocoa_ui_ window1] close];
-    [[cocoa_ui_ window2] close];
-    return 0;
-}
-
-void* ViEAutoTestWindowManager::GetWindow1() {
-    return [cocoa_ui_ cocoaRenderView1];
-}
-
-void* ViEAutoTestWindowManager::GetWindow2() {
-    return [cocoa_ui_ cocoaRenderView2];
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow() {
-    return true;
-}
-
-// This is acts as our "main" for mac. The actual (reusable) main is defined in
-// testsupport/mac/run_threaded_main_mac.mm.
-int ImplementThisToRunYourTest(int argc, char** argv) {
-  ViEAutoTestMain auto_test;
-  return auto_test.RunTests(argc, argv);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
deleted file mode 100644
index f5ff771a70..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ /dev/null
@@ -1,830 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_coding/codecs/i420/main/interface/i420.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_external_codec.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_I420_codec.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-
-class TestCodecObserver : public webrtc::ViEEncoderObserver,
-                          public webrtc::ViEDecoderObserver {
- public:
-  int incoming_codec_called_;
-  int incoming_rate_called_;
-  int decoder_timing_called_;
-  int outgoing_rate_called_;
-
-  unsigned char last_payload_type_;
-  uint16_t last_width_;
-  uint16_t last_height_;
-
-  unsigned int last_outgoing_framerate_;
-  unsigned int last_outgoing_bitrate_;
-  unsigned int last_incoming_framerate_;
-  unsigned int last_incoming_bitrate_;
-  unsigned int suspend_change_called_;
-
-  webrtc::VideoCodec incoming_codec_;
-
-  TestCodecObserver()
-      : incoming_codec_called_(0),
-        incoming_rate_called_(0),
-        decoder_timing_called_(0),
-        outgoing_rate_called_(0),
-        last_payload_type_(0),
-        last_width_(0),
-        last_height_(0),
-        last_outgoing_framerate_(0),
-        last_outgoing_bitrate_(0),
-        last_incoming_framerate_(0),
-        last_incoming_bitrate_(0),
-        suspend_change_called_(0) {
-    memset(&incoming_codec_, 0, sizeof(incoming_codec_));
-  }
-  virtual void IncomingCodecChanged(const int video_channel,
-                                    const webrtc::VideoCodec& video_codec) {
-    incoming_codec_called_++;
-    last_payload_type_ = video_codec.plType;
-    last_width_ = video_codec.width;
-    last_height_ = video_codec.height;
-
-    memcpy(&incoming_codec_, &video_codec, sizeof(video_codec));
-  }
-
-  virtual void IncomingRate(const int video_channel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) {
-    incoming_rate_called_++;
-    last_incoming_framerate_ += framerate;
-    last_incoming_bitrate_ += bitrate;
-  }
-
-  virtual void DecoderTiming(int decode_ms,
-                             int max_decode_ms,
-                             int current_delay_ms,
-                             int target_delay_ms,
-                             int jitter_buffer_ms,
-                             int min_playout_delay_ms,
-                             int render_delay_ms) {
-    ++decoder_timing_called_;
-    // TODO(fischman): anything useful to be done with the data here?
-  }
-
-  virtual void OutgoingRate(const int video_channel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) {
-    outgoing_rate_called_++;
-    last_outgoing_framerate_ += framerate;
-    last_outgoing_bitrate_ += bitrate;
-  }
-
-  void SuspendChange(int video_channel, bool is_suspended) override {
-    suspend_change_called_++;
-  }
-
-  virtual void RequestNewKeyFrame(const int video_channel) {
-  }
-};
-
-class RenderFilter : public webrtc::ViEEffectFilter {
- public:
-  int num_frames_;
-  unsigned int last_render_width_;
-  unsigned int last_render_height_;
-
-  RenderFilter()
-      : num_frames_(0),
-        last_render_width_(0),
-        last_render_height_(0) {
-  }
-
-  virtual ~RenderFilter() {
-  }
-  virtual int Transform(size_t size,
-                        unsigned char* frame_buffer,
-                        int64_t ntp_time_ms,
-                        unsigned int timestamp,
-                        unsigned int width,
-                        unsigned int height)  {
-    num_frames_++;
-    last_render_width_ = width;
-    last_render_height_ = height;
-    return 0;
-  }
-};
-
-void ViEAutoTest::ViECodecStandardTest() {
-  TbInterfaces interfaces("ViECodecStandardTest");
-
-  TbCaptureDevice capture_device = TbCaptureDevice(interfaces);
-  int capture_id = capture_device.captureId;
-
-  webrtc::VideoEngine* video_engine = interfaces.video_engine;
-  webrtc::ViEBase* base = interfaces.base;
-  webrtc::ViECapture* capture = interfaces.capture;
-  webrtc::ViERender* render = interfaces.render;
-  webrtc::ViECodec* codec = interfaces.codec;
-  webrtc::ViERTP_RTCP* rtp_rtcp = interfaces.rtp_rtcp;
-  webrtc::ViENetwork* network = interfaces.network;
-
-  int video_channel = -1;
-  EXPECT_EQ(0, base->CreateChannel(video_channel));
-  EXPECT_EQ(0, capture->ConnectCaptureDevice(capture_id, video_channel));
-  EXPECT_EQ(0, rtp_rtcp->SetRTCPStatus(
-      video_channel, webrtc::kRtcpCompound_RFC4585));
-
-  EXPECT_EQ(0, rtp_rtcp->SetKeyFrameRequestMethod(
-      video_channel, webrtc::kViEKeyFrameRequestPliRtcp));
-  EXPECT_EQ(0, rtp_rtcp->SetTMMBRStatus(video_channel, true));
-  EXPECT_EQ(0, render->AddRenderer(capture_id, _window1, 0, 0.0, 0.0, 1.0,
-                                   1.0));
-  EXPECT_EQ(0, render->AddRenderer(video_channel, _window2, 1, 0.0, 0.0, 1.0,
-                                   1.0));
-  EXPECT_EQ(0, render->StartRender(capture_id));
-  EXPECT_EQ(0, render->StartRender(video_channel));
-
-  webrtc::VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-  for (int idx = 0; idx < codec->NumberOfCodecs(); idx++) {
-    EXPECT_EQ(0, codec->GetCodec(idx, video_codec));
-    if (video_codec.codecType != webrtc::kVideoCodecI420) {
-      video_codec.width = 640;
-      video_codec.height = 480;
-    }
-    if (video_codec.codecType == webrtc::kVideoCodecI420) {
-      video_codec.width = 176;
-      video_codec.height = 144;
-    }
-    EXPECT_EQ(0, codec->SetReceiveCodec(video_channel, video_codec));
-  }
-
-  for (int idx = 0; idx < codec->NumberOfCodecs(); idx++) {
-    EXPECT_EQ(0, codec->GetCodec(idx, video_codec));
-    if (video_codec.codecType == webrtc::kVideoCodecVP8) {
-      EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-      break;
-    }
-  }
-  const char* ip_address = "127.0.0.1";
-  const uint16_t rtp_port = 6000;
-
-  rtc::scoped_ptr<webrtc::test::VideoChannelTransport> video_channel_transport(
-      new webrtc::test::VideoChannelTransport(network, video_channel));
-
-  ASSERT_EQ(0, video_channel_transport->SetSendDestination(ip_address,
-                                                           rtp_port));
-  ASSERT_EQ(0, video_channel_transport->SetLocalReceiver(rtp_port));
-
-  EXPECT_EQ(0, base->StartReceive(video_channel));
-  EXPECT_EQ(0, base->StartSend(video_channel));
-
-  // Make sure all codecs runs
-  {
-    webrtc::ViEImageProcess* image_process =
-        webrtc::ViEImageProcess::GetInterface(video_engine);
-    TestCodecObserver codec_observer;
-    EXPECT_EQ(0, codec->RegisterDecoderObserver(video_channel, codec_observer));
-    ViETest::Log("Loop through all codecs for %d seconds",
-                 kAutoTestSleepTimeMs / 1000);
-
-    for (int i = 0; i < codec->NumberOfCodecs() - 2; i++) {
-      EXPECT_EQ(0, codec->GetCodec(i, video_codec));
-      if (video_codec.codecType == webrtc::kVideoCodecI420) {
-        // Lower resolution to sockets keep up.
-        video_codec.width = 176;
-        video_codec.height = 144;
-        video_codec.maxFramerate = 15;
-      }
-      EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-      ViETest::Log("\t %d. %s", i, video_codec.plName);
-
-      RenderFilter frame_counter;
-      EXPECT_EQ(0, image_process->RegisterRenderEffectFilter(video_channel,
-                                                             frame_counter));
-      AutoTestSleep(kAutoTestSleepTimeMs);
-
-      // Verify we've received and decoded correct payload.
-      EXPECT_EQ(video_codec.codecType,
-                codec_observer.incoming_codec_.codecType);
-
-      // This requirement is quite relaxed, but it's hard to say what's an
-      // acceptable number of received frames when we take into account the
-      // wide variety of devices (and that we run under valgrind).
-      EXPECT_GT(frame_counter.num_frames_, 0);
-
-      EXPECT_EQ(0, image_process->DeregisterRenderEffectFilter(
-          video_channel));
-    }
-    image_process->Release();
-    EXPECT_EQ(0, codec->DeregisterDecoderObserver(video_channel));
-    ViETest::Log("Done!");
-  }
-
-  // Test Callbacks
-  TestCodecObserver codec_observer;
-  EXPECT_EQ(0, codec->RegisterEncoderObserver(video_channel, codec_observer));
-  EXPECT_EQ(0, codec->RegisterDecoderObserver(video_channel, codec_observer));
-
-  ViETest::Log("\nTesting codec callbacks...");
-
-  for (int idx = 0; idx < codec->NumberOfCodecs(); idx++) {
-    EXPECT_EQ(0, codec->GetCodec(idx, video_codec));
-    if (video_codec.codecType == webrtc::kVideoCodecVP8) {
-      EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-      break;
-    }
-  }
-  AutoTestSleep(kAutoTestSleepTimeMs);
-
-  // Verify the delay estimates are larger than 0.
-  int avg_send_delay = 0;
-  int max_send_delay = 0;
-  EXPECT_TRUE(codec->GetSendSideDelay(video_channel, &avg_send_delay,
-                                      &max_send_delay));
-  EXPECT_GT(avg_send_delay, 0);
-  EXPECT_GE(max_send_delay, avg_send_delay);
-  int receive_delay_ms = 0;
-  EXPECT_EQ(0, codec->GetReceiveSideDelay(video_channel, &receive_delay_ms));
-  EXPECT_GT(receive_delay_ms, 0);
-
-  EXPECT_EQ(0, base->StopSend(video_channel));
-  EXPECT_EQ(0, codec->DeregisterEncoderObserver(video_channel));
-  EXPECT_EQ(0, codec->DeregisterDecoderObserver(video_channel));
-
-  EXPECT_GT(codec_observer.incoming_codec_called_, 0);
-  EXPECT_GT(codec_observer.incoming_rate_called_, 0);
-  EXPECT_GT(codec_observer.decoder_timing_called_, 0);
-  EXPECT_GT(codec_observer.outgoing_rate_called_, 0);
-
-  EXPECT_EQ(0, base->StopReceive(video_channel));
-  EXPECT_EQ(0, render->StopRender(video_channel));
-  EXPECT_EQ(0, render->RemoveRenderer(capture_id));
-  EXPECT_EQ(0, render->RemoveRenderer(video_channel));
-  EXPECT_EQ(0, capture->DisconnectCaptureDevice(video_channel));
-  EXPECT_EQ(0, base->DeleteChannel(video_channel));
-}
-
-void ViEAutoTest::ViECodecExtendedTest() {
-  {
-    ViETest::Log(" ");
-    ViETest::Log("========================================");
-    ViETest::Log(" ViECodec Extended Test\n");
-
-    ViECodecExternalCodecTest();
-
-    TbInterfaces interfaces("ViECodecExtendedTest");
-    webrtc::ViEBase* base = interfaces.base;
-    webrtc::ViECapture* capture = interfaces.capture;
-    webrtc::ViERender* render = interfaces.render;
-    webrtc::ViECodec* codec = interfaces.codec;
-    webrtc::ViERTP_RTCP* rtp_rtcp = interfaces.rtp_rtcp;
-    webrtc::ViENetwork* network = interfaces.network;
-
-    TbCaptureDevice capture_device = TbCaptureDevice(interfaces);
-    int capture_id = capture_device.captureId;
-
-    int video_channel = -1;
-    EXPECT_EQ(0, base->CreateChannel(video_channel));
-    EXPECT_EQ(0, capture->ConnectCaptureDevice(capture_id, video_channel));
-    EXPECT_EQ(0, rtp_rtcp->SetRTCPStatus(
-                video_channel, webrtc::kRtcpCompound_RFC4585));
-    EXPECT_EQ(0, rtp_rtcp->SetKeyFrameRequestMethod(
-                video_channel, webrtc::kViEKeyFrameRequestPliRtcp));
-    EXPECT_EQ(0, rtp_rtcp->SetTMMBRStatus(video_channel, true));
-    EXPECT_EQ(0, render->AddRenderer(capture_id, _window1, 0, 0.0, 0.0, 1.0,
-                                     1.0));
-
-    EXPECT_EQ(0, render->AddRenderer(video_channel, _window2, 1, 0.0, 0.0, 1.0,
-                                     1.0));
-    EXPECT_EQ(0, render->StartRender(capture_id));
-    EXPECT_EQ(0, render->StartRender(video_channel));
-
-    webrtc::VideoCodec video_codec;
-    memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-    for (int idx = 0; idx < codec->NumberOfCodecs(); idx++) {
-      EXPECT_EQ(0, codec->GetCodec(idx, video_codec));
-      if (video_codec.codecType != webrtc::kVideoCodecI420) {
-        video_codec.width = 640;
-        video_codec.height = 480;
-      }
-      EXPECT_EQ(0, codec->SetReceiveCodec(video_channel, video_codec));
-    }
-
-    const char* ip_address = "127.0.0.1";
-    const uint16_t rtp_port = 6000;
-
-    rtc::scoped_ptr<webrtc::test::VideoChannelTransport>
-        video_channel_transport(
-            new webrtc::test::VideoChannelTransport(network, video_channel));
-
-    ASSERT_EQ(0, video_channel_transport->SetSendDestination(ip_address,
-                                                             rtp_port));
-    ASSERT_EQ(0, video_channel_transport->SetLocalReceiver(rtp_port));
-
-    EXPECT_EQ(0, base->StartSend(video_channel));
-    EXPECT_EQ(0, base->StartReceive(video_channel));
-
-    // Codec specific tests
-    memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-    EXPECT_EQ(0, base->StopSend(video_channel));
-
-    TestCodecObserver codec_observer;
-    EXPECT_EQ(0, codec->RegisterEncoderObserver(video_channel, codec_observer));
-    EXPECT_EQ(0, codec->RegisterDecoderObserver(video_channel, codec_observer));
-    EXPECT_EQ(0, base->StopReceive(video_channel));
-
-    EXPECT_EQ(0, render->StopRender(video_channel));
-    EXPECT_EQ(0, render->RemoveRenderer(capture_id));
-    EXPECT_EQ(0, render->RemoveRenderer(video_channel));
-    EXPECT_EQ(0, capture->DisconnectCaptureDevice(video_channel));
-    EXPECT_EQ(0, base->DeleteChannel(video_channel));
-  }
-
-  // Multiple send channels.
-  {
-    // Create two channels, where the second channel is created from the
-    // first channel. Send different resolutions on the channels and verify
-    // the received streams.
-    TbInterfaces video_engine("ViECodecExtendedTest2");
-    TbCaptureDevice tb_capture(video_engine);
-    webrtc::ViENetwork* network = video_engine.network;
-
-    // Create channel 1.
-    int video_channel_1 = -1;
-    EXPECT_EQ(0, video_engine.base->CreateChannel(video_channel_1));
-
-    // Create channel 2 based on the first channel.
-    int video_channel_2 = -1;
-    EXPECT_EQ(0, video_engine.base->CreateChannel(
-        video_channel_2, video_channel_1));
-    EXPECT_NE(video_channel_1, video_channel_2)
-        << "Channel 2 should be unique.";
-
-    const char* ip_address = "127.0.0.1";
-    uint16_t rtp_port_1 = 12000;
-    uint16_t rtp_port_2 = 13000;
-
-    rtc::scoped_ptr<webrtc::test::VideoChannelTransport>
-        video_channel_transport_1(
-            new webrtc::test::VideoChannelTransport(network, video_channel_1));
-
-    ASSERT_EQ(0, video_channel_transport_1->SetSendDestination(ip_address,
-                                                               rtp_port_1));
-    ASSERT_EQ(0, video_channel_transport_1->SetLocalReceiver(rtp_port_1));
-
-    rtc::scoped_ptr<webrtc::test::VideoChannelTransport>
-        video_channel_transport_2(
-            new webrtc::test::VideoChannelTransport(network, video_channel_2));
-
-    ASSERT_EQ(0, video_channel_transport_2->SetSendDestination(ip_address,
-                                                               rtp_port_2));
-    ASSERT_EQ(0, video_channel_transport_2->SetLocalReceiver(rtp_port_2));
-
-    EXPECT_EQ(0, video_engine.rtp_rtcp->SetLocalSSRC(video_channel_1, 1));
-    EXPECT_EQ(0, video_engine.rtp_rtcp->SetLocalSSRC(video_channel_2, 2));
-    tb_capture.ConnectTo(video_channel_1);
-    tb_capture.ConnectTo(video_channel_2);
-    EXPECT_EQ(0, video_engine.rtp_rtcp->SetKeyFrameRequestMethod(
-        video_channel_1, webrtc::kViEKeyFrameRequestPliRtcp));
-    EXPECT_EQ(0, video_engine.rtp_rtcp->SetKeyFrameRequestMethod(
-        video_channel_2, webrtc::kViEKeyFrameRequestPliRtcp));
-    EXPECT_EQ(0, video_engine.render->AddRenderer(video_channel_1, _window1, 0,
-                                                  0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, video_engine.render->StartRender(video_channel_1));
-    EXPECT_EQ(0, video_engine.render->AddRenderer(video_channel_2, _window2, 0,
-                                                  0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, video_engine.render->StartRender(video_channel_2));
-
-    // Set Send codec.
-    uint16_t codec_width = 320;
-    uint16_t codec_height = 240;
-    bool codec_set = false;
-    webrtc::VideoCodec video_codec;
-    webrtc::VideoCodec send_codec1;
-    webrtc::VideoCodec send_codec2;
-    for (int idx = 0; idx < video_engine.codec->NumberOfCodecs(); idx++) {
-      EXPECT_EQ(0, video_engine.codec->GetCodec(idx, video_codec));
-      EXPECT_EQ(0, video_engine.codec->SetReceiveCodec(video_channel_1,
-                                                       video_codec));
-      if (video_codec.codecType == webrtc::kVideoCodecVP8) {
-        memcpy(&send_codec1, &video_codec, sizeof(video_codec));
-        send_codec1.width = codec_width;
-        send_codec1.height = codec_height;
-        EXPECT_EQ(0, video_engine.codec->SetSendCodec(
-                    video_channel_1, send_codec1));
-        memcpy(&send_codec2, &video_codec, sizeof(video_codec));
-        send_codec2.width = 2 * codec_width;
-        send_codec2.height = 2 * codec_height;
-        EXPECT_EQ(0, video_engine.codec->SetSendCodec(
-                    video_channel_2, send_codec2));
-        codec_set = true;
-        break;
-      }
-    }
-    EXPECT_TRUE(codec_set);
-
-    // We need to verify using render effect filter since we won't trigger
-    // a decode reset in loopback (due to using the same SSRC).
-    RenderFilter filter1;
-    RenderFilter filter2;
-    EXPECT_EQ(0, video_engine.image_process->RegisterRenderEffectFilter(
-        video_channel_1, filter1));
-    EXPECT_EQ(0, video_engine.image_process->RegisterRenderEffectFilter(
-        video_channel_2, filter2));
-
-    EXPECT_EQ(0, video_engine.base->StartReceive(video_channel_1));
-    EXPECT_EQ(0, video_engine.base->StartSend(video_channel_1));
-    EXPECT_EQ(0, video_engine.base->StartReceive(video_channel_2));
-    EXPECT_EQ(0, video_engine.base->StartSend(video_channel_2));
-
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, video_engine.base->StopReceive(video_channel_1));
-    EXPECT_EQ(0, video_engine.base->StopSend(video_channel_1));
-    EXPECT_EQ(0, video_engine.base->StopReceive(video_channel_2));
-    EXPECT_EQ(0, video_engine.base->StopSend(video_channel_2));
-
-    EXPECT_EQ(0, video_engine.image_process->DeregisterRenderEffectFilter(
-        video_channel_1));
-    EXPECT_EQ(0, video_engine.image_process->DeregisterRenderEffectFilter(
-        video_channel_2));
-    EXPECT_EQ(send_codec1.width, filter1.last_render_width_);
-    EXPECT_EQ(send_codec1.height, filter1.last_render_height_);
-    EXPECT_EQ(send_codec2.width, filter2.last_render_width_);
-    EXPECT_EQ(send_codec2.height, filter2.last_render_height_);
-
-    EXPECT_EQ(0, video_engine.base->DeleteChannel(video_channel_1));
-    EXPECT_EQ(0, video_engine.base->DeleteChannel(video_channel_2));
-  }
-}
-
-void ViEAutoTest::ViECodecAPITest() {
-  webrtc::VideoEngine* video_engine = NULL;
-  video_engine = webrtc::VideoEngine::Create();
-  EXPECT_TRUE(video_engine != NULL);
-
-  webrtc::ViEBase* base = webrtc::ViEBase::GetInterface(video_engine);
-  EXPECT_EQ(0, base->Init());
-
-  int video_channel = -1;
-  EXPECT_EQ(0, base->CreateChannel(video_channel));
-
-  webrtc::ViECodec* codec = webrtc::ViECodec::GetInterface(video_engine);
-  EXPECT_TRUE(codec != NULL);
-
-  webrtc::VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-
-  const int number_of_codecs = codec->NumberOfCodecs();
-
-  for (int i = 0; i < number_of_codecs; i++) {
-    EXPECT_EQ(0, codec->GetCodec(i, video_codec));
-    if (video_codec.codecType == webrtc::kVideoCodecVP8) {
-      video_codec.codecSpecific.VP8.automaticResizeOn = true;
-      video_codec.codecSpecific.VP8.frameDroppingOn = true;
-      video_codec.codecSpecific.VP8.keyFrameInterval = 300;
-      EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-      break;
-    }
-  }
-  const unsigned int kMinBitrate = 123;
-  video_codec.minBitrate = kMinBitrate;
-  video_codec.startBitrate = 50;
-  EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-  EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
-  // We don't allow allocated start bitrate to be decreased via SetSendCodec,
-  // and the default bitrate available in the allocator is 300.
-  EXPECT_EQ(300u, video_codec.startBitrate);
-
-  memset(&video_codec, 0, sizeof(video_codec));
-  EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
-  EXPECT_EQ(webrtc::kVideoCodecVP8, video_codec.codecType);
-  EXPECT_TRUE(video_codec.codecSpecific.VP8.automaticResizeOn);
-  EXPECT_TRUE(video_codec.codecSpecific.VP8.frameDroppingOn);
-  EXPECT_EQ(300, video_codec.codecSpecific.VP8.keyFrameInterval);
-
-  for (int i = 0; i < number_of_codecs; i++) {
-    EXPECT_EQ(0, codec->GetCodec(i, video_codec));
-    if (video_codec.codecType == webrtc::kVideoCodecI420) {
-      EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-      break;
-    }
-  }
-
-  memset(&video_codec, 0, sizeof(video_codec));
-  EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
-  EXPECT_EQ(webrtc::kVideoCodecI420, video_codec.codecType);
-
-  // Register a generic codec
-  memset(&video_codec, 0, sizeof(video_codec));
-  video_codec.codecType = webrtc::kVideoCodecGeneric;
-  strcpy(video_codec.plName, "generic-codec");
-  uint8_t payload_type = 127;
-  video_codec.plType = payload_type;
-  video_codec.minBitrate = 100;
-  video_codec.startBitrate = 500;
-  video_codec.maxBitrate = 10000;
-  video_codec.width = 1920;
-  video_codec.height = 1080;
-  video_codec.maxFramerate = 30;
-  video_codec.qpMax = 50;
-
-  webrtc::ViEExternalCodec* external_codec =
-      webrtc::ViEExternalCodec::GetInterface(video_engine);
-  EXPECT_TRUE(external_codec != NULL);
-
-  // Any encoder will do.
-  webrtc::I420Encoder encoder;
-  EXPECT_EQ(0, external_codec->RegisterExternalSendCodec(video_channel,
-                                                         payload_type, &encoder,
-                                                         false));
-  EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
-
-  memset(&video_codec, 0, sizeof(video_codec));
-  EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
-  EXPECT_EQ(webrtc::kVideoCodecGeneric, video_codec.codecType);
-
-  EXPECT_EQ(0, base->DeleteChannel(video_channel));
-
-  EXPECT_EQ(0, external_codec->Release());
-  EXPECT_EQ(0, codec->Release());
-  EXPECT_EQ(0, base->Release());
-  EXPECT_TRUE(webrtc::VideoEngine::Delete(video_engine));
-}
-
-void ViEAutoTest::ViECodecExternalCodecTest() {
-  ViETest::Log(" ");
-  ViETest::Log("========================================");
-  ViETest::Log(" ViEExternalCodec Test\n");
-
-  /// **************************************************************
-  //  Begin create/initialize WebRTC Video Engine for testing
-  /// **************************************************************
-
-  /// **************************************************************
-  //  Engine ready. Begin testing class
-  /// **************************************************************
-
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-  int number_of_errors = 0;
-  {
-    int error = 0;
-    TbInterfaces ViE("ViEExternalCodec");
-    TbCaptureDevice capture_device(ViE);
-    TbVideoChannel channel(ViE, webrtc::kVideoCodecI420, 352, 288, 30,
-                           (352 * 288 * 3 * 8 * 30) / (2 * 1000));
-    capture_device.ConnectTo(channel.videoChannel);
-
-    error = ViE.render->AddRenderer(channel.videoChannel, _window1, 0, 0.0, 0.0,
-                                    1.0, 1.0);
-    number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    error = ViE.render->StartRender(channel.videoChannel);
-    number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    channel.StartReceive();
-    channel.StartSend();
-
-    ViETest::Log("Using internal I420 codec");
-    AutoTestSleep(kAutoTestSleepTimeMs / 2);
-
-    webrtc::ViEExternalCodec* vie_external_codec =
-        webrtc::ViEExternalCodec::GetInterface(ViE.video_engine);
-    number_of_errors += ViETest::TestError(vie_external_codec != NULL,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    webrtc::VideoCodec codec;
-    error = ViE.codec->GetSendCodec(channel.videoChannel, codec);
-    number_of_errors += ViETest::TestError(vie_external_codec != NULL,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    // Use external encoder instead.
-    {
-      TbI420Encoder ext_encoder;
-
-      // Test to register on wrong channel.
-      error = vie_external_codec->RegisterExternalSendCodec(
-          channel.videoChannel + 5, codec.plType, &ext_encoder, false);
-      number_of_errors += ViETest::TestError(error == -1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          ViE.LastError() == kViECodecInvalidArgument,
-          "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-
-      error = vie_external_codec->RegisterExternalSendCodec(
-                channel.videoChannel, codec.plType, &ext_encoder, false);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      // Use new external encoder
-      error = ViE.codec->SetSendCodec(channel.videoChannel, codec);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      TbI420Decoder ext_decoder;
-      error = vie_external_codec->RegisterExternalReceiveCodec(
-          channel.videoChannel, codec.plType, &ext_decoder);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      error = ViE.codec->SetReceiveCodec(channel.videoChannel, codec);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      ViETest::Log("Using external I420 codec");
-      AutoTestSleep(kAutoTestSleepTimeMs);
-
-      // Test to deregister on wrong channel
-      error = vie_external_codec->DeRegisterExternalSendCodec(
-          channel.videoChannel + 5, codec.plType);
-      number_of_errors += ViETest::TestError(error == -1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          ViE.LastError() == kViECodecInvalidArgument, "ERROR: %s at line %d",
-          __FUNCTION__, __LINE__);
-
-      // Test to deregister wrong payload type.
-      error = vie_external_codec->DeRegisterExternalSendCodec(
-          channel.videoChannel, codec.plType - 1);
-      number_of_errors += ViETest::TestError(error == -1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      // Deregister external send codec
-      error = vie_external_codec->DeRegisterExternalSendCodec(
-          channel.videoChannel, codec.plType);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      error = vie_external_codec->DeRegisterExternalReceiveCodec(
-          channel.videoChannel, codec.plType);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      // Verify that the encoder and decoder has been used
-      TbI420Encoder::FunctionCalls encode_calls =
-          ext_encoder.GetFunctionCalls();
-      number_of_errors += ViETest::TestError(encode_calls.InitEncode == 1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(encode_calls.Release == 1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(encode_calls.Encode > 30,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          encode_calls.RegisterEncodeCompleteCallback == 1,
-          "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          encode_calls.SetChannelParameters > 1, "ERROR: %s at line %d",
-          __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(encode_calls.SetRates > 1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      TbI420Decoder::FunctionCalls decode_calls =
-          ext_decoder.GetFunctionCalls();
-      number_of_errors += ViETest::TestError(decode_calls.InitDecode == 1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(decode_calls.Release == 1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(decode_calls.Decode > 30,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          decode_calls.RegisterDecodeCompleteCallback == 1,
-          "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-
-      ViETest::Log("Changing payload type Using external I420 codec");
-
-      codec.plType = codec.plType - 1;
-      error = vie_external_codec->RegisterExternalReceiveCodec(
-          channel.videoChannel, codec.plType, &ext_decoder);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      error = ViE.codec->SetReceiveCodec(channel.videoChannel,
-                                         codec);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      error = vie_external_codec->RegisterExternalSendCodec(
-                channel.videoChannel, codec.plType, &ext_encoder, false);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      // Use new external encoder
-      error = ViE.codec->SetSendCodec(channel.videoChannel,
-                                      codec);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      AutoTestSleep(kAutoTestSleepTimeMs / 2);
-
-      /// **************************************************************
-      //  Testing finished. Tear down Video Engine
-      /// **************************************************************
-
-      error = vie_external_codec->DeRegisterExternalSendCodec(
-                channel.videoChannel, codec.plType);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      error = vie_external_codec->DeRegisterExternalReceiveCodec(
-                channel.videoChannel, codec.plType);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      // Verify that the encoder and decoder has been used
-      encode_calls = ext_encoder.GetFunctionCalls();
-      number_of_errors += ViETest::TestError(encode_calls.InitEncode == 2,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(encode_calls.Release == 2,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(encode_calls.Encode > 30,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          encode_calls.RegisterEncodeCompleteCallback == 2,
-          "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          encode_calls.SetChannelParameters > 1, "ERROR: %s at line %d",
-          __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(encode_calls.SetRates > 1,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      decode_calls = ext_decoder.GetFunctionCalls();
-      number_of_errors += ViETest::TestError(decode_calls.InitDecode == 2,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(decode_calls.Release == 2,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(decode_calls.Decode > 30,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      number_of_errors += ViETest::TestError(
-          decode_calls.RegisterDecodeCompleteCallback == 2,
-          "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-
-      int remaining_interfaces = vie_external_codec->Release();
-      number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-    }  // tbI420Encoder and ext_decoder goes out of scope.
-
-    ViETest::Log("Using internal I420 codec");
-    AutoTestSleep(kAutoTestSleepTimeMs / 2);
-  }
-  if (number_of_errors > 0) {
-    // Test failed
-    ViETest::Log(" ");
-    ViETest::Log(" ERROR ViEExternalCodec Test FAILED!");
-    ViETest::Log(" Number of errors: %d", number_of_errors);
-    ViETest::Log("========================================");
-    ViETest::Log(" ");
-    return;
-  }
-
-  ViETest::Log(" ");
-  ViETest::Log(" ViEExternalCodec Test PASSED!");
-  ViETest::Log("========================================");
-  ViETest::Log(" ");
-  return;
-
-#else
-  ViETest::Log(" ViEExternalCodec not enabled\n");
-  return;
-#endif
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
deleted file mode 100644
index 1a6633232a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
+++ /dev/null
@@ -1,1698 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <limits.h>
-#include <stdarg.h>
-#include <stdio.h>
-
-#include <algorithm>
-
-#include "gflags/gflags.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/primitives/choice_helpers.h"
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-#include "webrtc/video_engine/test/auto_test/primitives/input_helpers.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-
-#define VCM_RED_PAYLOAD_TYPE                            96
-#define VCM_ULPFEC_PAYLOAD_TYPE                         97
-#define DEFAULT_SEND_IP                                 "127.0.0.1"
-#define DEFAULT_VIDEO_PORT                              "11111"
-#define DEFAULT_VIDEO_CODEC                             "VP8"
-#define DEFAULT_VIDEO_CODEC_WIDTH                       "640"
-#define DEFAULT_VIDEO_CODEC_HEIGHT                      "480"
-#define DEFAULT_VIDEO_CODEC_BITRATE                     "300"
-#define DEFAULT_VIDEO_CODEC_MIN_BITRATE                 "100"
-#define DEFAULT_VIDEO_CODEC_MAX_BITRATE                 "1000"
-#define DEFAULT_AUDIO_PORT                              "11113"
-#define DEFAULT_AUDIO_CODEC                             "ISAC"
-#define DEFAULT_VIDEO_CODEC_MAX_FRAMERATE               "30"
-#define DEFAULT_VIDEO_PROTECTION_METHOD                 "None"
-#define DEFAULT_TEMPORAL_LAYER                          "0"
-#define DEFAULT_BUFFERING_DELAY_MS                      "0"
-
-DEFINE_string(render_custom_call_remote_to, "", "Specify to render the remote "
-    "stream of a custom call to the provided filename instead of showing it in "
-    "window 2. The file will end up in the default output directory (out/).");
-
-enum StatisticsType {
-  kSendStatistic,
-  kReceivedStatistic
-};
-
-enum VideoProtectionMethod {
-  kProtectionMethodNone = 1,
-  kProtectionMethodFecOnly,
-  kProtectionMethodNackOnly,
-  kProtectionMethodHybridNackAndFec,
-};
-
-using webrtc::FromChoices;
-using webrtc::TypedInput;
-
-class ViEAutotestEncoderObserver : public webrtc::ViEEncoderObserver {
- public:
-  ViEAutotestEncoderObserver() {}
-  ~ViEAutotestEncoderObserver() {}
-
-  void OutgoingRate(const int video_channel,
-                    const unsigned int framerate,
-                    const unsigned int bitrate) {
-    std::cout << "Send FR: " << framerate
-              << " BR: " << bitrate << std::endl;
-  }
-
-  void SuspendChange(int video_channel, bool is_suspended) override {
-    std::cout << "SuspendChange: " << is_suspended << std::endl;
-  }
-};
-
-class ViEAutotestDecoderObserver : public webrtc::ViEDecoderObserver {
- public:
-  ViEAutotestDecoderObserver() {}
-  ~ViEAutotestDecoderObserver() {}
-
-  void IncomingRate(const int video_channel,
-                    const unsigned int framerate,
-                    const unsigned int bitrate) {
-    std::cout << "Received FR: " << framerate
-              << " BR: " << bitrate << std::endl;
-  }
-
-  virtual void DecoderTiming(int decode_ms,
-                             int max_decode_ms,
-                             int current_delay_ms,
-                             int target_delay_ms,
-                             int jitter_buffer_ms,
-                             int min_playout_delay_ms,
-                             int render_delay_ms) {
-    std::cout << "Decoder timing: DecodeMS: " << decode_ms
-              << ", MaxDecodeMS: " << max_decode_ms
-              << ", CurrentDelayMS: " << current_delay_ms
-              << ", TargetDelayMS: " << target_delay_ms
-              << ", JitterBufferMS: " << jitter_buffer_ms
-              << ", MinPlayoutDelayMS: " << min_playout_delay_ms
-              << ", RenderDelayMS: " << render_delay_ms;
-  }
-
-  void IncomingCodecChanged(const int video_channel,
-                            const webrtc::VideoCodec& codec) {}
-  void RequestNewKeyFrame(const int video_channel) {
-    std::cout << "Decoder requesting a new key frame." << std::endl;
-  }
-};
-
-// The following are general helper functions.
-bool GetVideoDevice(webrtc::ViEBase* vie_base,
-                    webrtc::ViECapture* vie_capture,
-                    char* capture_device_name, char* capture_device_unique_id);
-std::string GetIPAddress();
-bool ValidateIP(std::string i_str);
-
-// The following are Print to stdout functions.
-void PrintCallInformation(const char* IP,
-                          const char* video_capture_device_name,
-                          const char* video_capture_unique_id,
-                          webrtc::VideoCodec video_codec,
-                          int video_tx_port,
-                          int video_rx_port,
-                          const char* audio_capture_device_name,
-                          const char* audio_playbackDeviceName,
-                          webrtc::CodecInst audio_codec,
-                          int audio_tx_port,
-                          int audio_rx_port,
-                          int protection_method);
-void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                          int video_channel,
-                          StatisticsType stat_type);
-void PrintRTPStatistics(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                        int video_channel);
-void PrintBandwidthUsage(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                         int video_channel);
-void PrintCodecStatistics(webrtc::ViECodec* vie_codec,
-                          int video_channel,
-                          StatisticsType stat_type);
-void PrintGetDiscardedPackets(webrtc::ViECodec* vie_codec,
-                              int video_channel);
-void PrintVideoStreamInformation(webrtc::ViECodec* vie_codec,
-                                 int video_channel);
-void PrintVideoCodec(webrtc::VideoCodec video_codec);
-
-// The following are video functions.
-void GetVideoPorts(int* tx_port, int* rx_port);
-void SetVideoCodecType(webrtc::ViECodec* vie_codec,
-                       webrtc::VideoCodec* video_codec);
-void SetVideoCodecResolution(webrtc::VideoCodec* video_codec);
-void SetVideoCodecSize(webrtc::VideoCodec* video_codec);
-void SetVideoCodecBitrate(webrtc::VideoCodec* video_codec);
-void SetVideoCodecMinBitrate(webrtc::VideoCodec* video_codec);
-void SetVideoCodecMaxBitrate(webrtc::VideoCodec* video_codec);
-void SetVideoCodecMaxFramerate(webrtc::VideoCodec* video_codec);
-void SetVideoCodecTemporalLayer(webrtc::VideoCodec* video_codec);
-VideoProtectionMethod GetVideoProtection();
-bool SetVideoProtection(webrtc::ViECodec* vie_codec,
-                        webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                        int video_channel,
-                        VideoProtectionMethod protection_method);
-bool GetBitrateSignaling();
-int GetBufferingDelay();
-
-// The following are audio helper functions.
-bool GetAudioDevices(webrtc::VoEBase* voe_base,
-                     webrtc::VoEHardware* voe_hardware,
-                     char* recording_device_name, int& recording_device_index,
-                     char* playback_device_name, int& playback_device_index);
-bool GetAudioDevices(webrtc::VoEBase* voe_base,
-                     webrtc::VoEHardware* voe_hardware,
-                     int& recording_device_index, int& playback_device_index);
-void GetAudioPorts(int* tx_port, int* rx_port);
-bool GetAudioCodec(webrtc::VoECodec* voe_codec,
-                   webrtc::CodecInst& audio_codec);
-
-int ViEAutoTest::ViECustomCall() {
-  ViETest::Log(" ");
-  ViETest::Log("========================================");
-  ViETest::Log(" Enter values to use custom settings\n");
-
-  int error = 0;
-  int number_of_errors = 0;
-  std::string str;
-
-  // Create the VoE and get the VoE interfaces.
-  webrtc::VoiceEngine* voe = webrtc::VoiceEngine::Create();
-  number_of_errors += ViETest::TestError(voe != NULL, "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-
-  webrtc::VoEBase* voe_base = webrtc::VoEBase::GetInterface(voe);
-  number_of_errors += ViETest::TestError(voe_base != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  error = voe_base->Init();
-  number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-
-  webrtc::VoECodec* voe_codec = webrtc::VoECodec::GetInterface(voe);
-  number_of_errors += ViETest::TestError(voe_codec != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::VoEHardware* voe_hardware =
-      webrtc::VoEHardware::GetInterface(voe);
-  number_of_errors += ViETest::TestError(voe_hardware != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::VoENetwork* voe_network=
-      webrtc::VoENetwork::GetInterface(voe);
-  number_of_errors += ViETest::TestError(voe_network != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::VoEAudioProcessing* voe_apm =
-      webrtc::VoEAudioProcessing::GetInterface(voe);
-  number_of_errors += ViETest::TestError(voe_apm != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  // Create the ViE and get the ViE Interfaces.
-  webrtc::VideoEngine* vie = webrtc::VideoEngine::Create();
-  number_of_errors += ViETest::TestError(vie != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::ViEBase* vie_base = webrtc::ViEBase::GetInterface(vie);
-  number_of_errors += ViETest::TestError(vie_base != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  error = vie_base->Init();
-  number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-
-  webrtc::ViECapture* vie_capture =
-      webrtc::ViECapture::GetInterface(vie);
-  number_of_errors += ViETest::TestError(vie_capture != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::ViERender* vie_renderer = webrtc::ViERender::GetInterface(vie);
-  number_of_errors += ViETest::TestError(vie_renderer != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::ViECodec* vie_codec = webrtc::ViECodec::GetInterface(vie);
-  number_of_errors += ViETest::TestError(vie_codec != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  webrtc::ViENetwork* vie_network = webrtc::ViENetwork::GetInterface(vie);
-  number_of_errors += ViETest::TestError(vie_network != NULL,
-                                         "ERROR: %s at line %d", __FUNCTION__,
-                                         __LINE__);
-
-  bool start_call = false;
-  std::string ip_address;
-  const unsigned int KMaxUniqueIdLength = 256;
-  char unique_id[KMaxUniqueIdLength] = "";
-  char device_name[KMaxUniqueIdLength] = "";
-  int video_tx_port = 0;
-  int video_rx_port = 0;
-  int video_channel = -1;
-  webrtc::VideoCodec video_send_codec;
-  char audio_capture_device_name[KMaxUniqueIdLength] = "";
-  char audio_playbackDeviceName[KMaxUniqueIdLength] = "";
-  int audio_capture_device_index = -1;
-  int audio_playback_device_index = -1;
-  int audio_tx_port = 0;
-  int audio_rx_port = 0;
-  webrtc::CodecInst audio_codec;
-  int audio_channel = -1;
-  VideoProtectionMethod protection_method = kProtectionMethodNone;
-  int buffer_delay_ms = 0;
-  bool is_image_scale_enabled = false;
-  bool remb = true;
-  rtc::scoped_ptr<webrtc::test::VideoChannelTransport> video_channel_transport;
-  rtc::scoped_ptr<webrtc::test::VoiceChannelTransport> voice_channel_transport;
-
-  while (!start_call) {
-    // Get the IP address to use from call.
-    ip_address = GetIPAddress();
-
-    // Get the video device to use for call.
-    memset(device_name, 0, KMaxUniqueIdLength);
-    memset(unique_id, 0, KMaxUniqueIdLength);
-    if (!GetVideoDevice(vie_base, vie_capture, device_name, unique_id))
-      return number_of_errors;
-
-    // Get and set the video ports for the call.
-    video_tx_port = 0;
-    video_rx_port = 0;
-    GetVideoPorts(&video_tx_port, &video_rx_port);
-
-    // Get and set the video codec parameters for the call.
-    memset(&video_send_codec, 0, sizeof(webrtc::VideoCodec));
-    SetVideoCodecType(vie_codec, &video_send_codec);
-    SetVideoCodecSize(&video_send_codec);
-    SetVideoCodecBitrate(&video_send_codec);
-    SetVideoCodecMinBitrate(&video_send_codec);
-    SetVideoCodecMaxBitrate(&video_send_codec);
-    SetVideoCodecMaxFramerate(&video_send_codec);
-    SetVideoCodecTemporalLayer(&video_send_codec);
-    remb = GetBitrateSignaling();
-
-    // Get the video protection method for the call.
-    protection_method = GetVideoProtection();
-
-    // Get the call mode (Real-Time/Buffered).
-    buffer_delay_ms = GetBufferingDelay();
-
-    // Get the audio device for the call.
-    memset(audio_capture_device_name, 0, KMaxUniqueIdLength);
-    memset(audio_playbackDeviceName, 0, KMaxUniqueIdLength);
-    GetAudioDevices(voe_base, voe_hardware, audio_capture_device_name,
-                    audio_capture_device_index, audio_playbackDeviceName,
-                    audio_playback_device_index);
-
-    // Get the audio port for the call.
-    audio_tx_port = 0;
-    audio_rx_port = 0;
-    GetAudioPorts(&audio_tx_port, &audio_rx_port);
-
-    // Get the audio codec for the call.
-    memset(static_cast<void*>(&audio_codec), 0, sizeof(audio_codec));
-    GetAudioCodec(voe_codec, audio_codec);
-
-    // Now ready to start the call.  Check user wants to continue.
-    PrintCallInformation(ip_address.c_str(), device_name, unique_id,
-                         video_send_codec, video_tx_port, video_rx_port,
-                         audio_capture_device_name, audio_playbackDeviceName,
-                         audio_codec, audio_tx_port, audio_rx_port,
-                         protection_method);
-
-    printf("\n");
-    int selection =
-        FromChoices("Ready to start:",
-                    "Start the call\n"
-                    "Reconfigure call settings\n")
-                        .WithDefault("Start the call").Choose();
-    start_call = (selection == 1);
-  }
-  /// **************************************************************
-  // Begin create/initialize WebRTC Video Engine for testing.
-  /// **************************************************************
-  if (start_call == true) {
-    // Configure audio channel first.
-    audio_channel = voe_base->CreateChannel();
-
-    voice_channel_transport.reset(
-        new webrtc::test::VoiceChannelTransport(voe_network, audio_channel));
-
-    error = voice_channel_transport->SetSendDestination(ip_address.c_str(),
-                                                        audio_tx_port);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voice_channel_transport->SetLocalReceiver(audio_rx_port);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_hardware->SetRecordingDevice(audio_capture_device_index);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_hardware->SetPlayoutDevice(audio_playback_device_index);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_codec->SetSendCodec(audio_channel, audio_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_apm->SetAgcStatus(true, webrtc::kAgcDefault);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_apm->SetNsStatus(true, webrtc::kNsHighSuppression);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    // Now configure the video channel.
-    error = vie->SetTraceFilter(webrtc::kTraceAll);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    std::string trace_file =
-        ViETest::GetResultOutputPath() + "ViECustomCall_trace.txt";
-    error = vie->SetTraceFile(trace_file.c_str());
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_base->SetVoiceEngine(voe);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_base->CreateChannel(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_base->ConnectAudioChannel(video_channel, audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    int capture_id = 0;
-    error = vie_capture->AllocateCaptureDevice(unique_id,
-                                               KMaxUniqueIdLength,
-                                               capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_capture->ConnectCaptureDevice(capture_id, video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_capture->StartCapture(capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    webrtc::ViERTP_RTCP* vie_rtp_rtcp =
-        webrtc::ViERTP_RTCP::GetInterface(vie);
-    number_of_errors += ViETest::TestError(vie != NULL,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_rtp_rtcp->SetRTCPStatus(video_channel,
-                                        webrtc::kRtcpCompound_RFC4585);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_rtp_rtcp->SetKeyFrameRequestMethod(
-        video_channel, webrtc::kViEKeyFrameRequestPliRtcp);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    if (remb) {
-      error = vie_rtp_rtcp->SetRembStatus(video_channel, true, true);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-    } else  {
-      error = vie_rtp_rtcp->SetTMMBRStatus(video_channel, true);
-      number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-    }
-
-    error = vie_renderer->AddRenderer(capture_id, _window1, 0, 0.0, 0.0, 1.0,
-                                      1.0);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-
-    ViEToFileRenderer file_renderer;
-    if (FLAGS_render_custom_call_remote_to == "") {
-      error = vie_renderer->AddRenderer(video_channel, _window2, 1, 0.0, 0.0,
-                                        1.0, 1.0);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-    } else {
-      std::string output_path = ViETest::GetResultOutputPath();
-      std::string filename = FLAGS_render_custom_call_remote_to;
-      ViETest::Log("Rendering remote stream to %s: you will not see any output "
-          "in the second window.", (output_path + filename).c_str());
-
-      file_renderer.PrepareForRendering(output_path, filename);
-      RenderToFile(vie_renderer, video_channel, &file_renderer);
-    }
-
-    video_channel_transport.reset(
-        new webrtc::test::VideoChannelTransport(vie_network, video_channel));
-
-    error = video_channel_transport->SetSendDestination(ip_address.c_str(),
-                                                        video_tx_port);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = video_channel_transport->SetLocalReceiver(video_rx_port);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_codec->SetSendCodec(video_channel, video_send_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_codec->SetReceiveCodec(video_channel, video_send_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    // Set the call mode (conferencing/buffering)
-    error = vie_rtp_rtcp->SetSenderBufferingMode(video_channel,
-                                                    buffer_delay_ms);
-    number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    error = vie_rtp_rtcp->SetReceiverBufferingMode(video_channel,
-                                                      buffer_delay_ms);
-    number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    // Set the Video Protection before start send and receive.
-    SetVideoProtection(vie_codec, vie_rtp_rtcp,
-                       video_channel, protection_method);
-
-    // Start Voice Playout and Receive.
-    error = voe_base->StartReceive(audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_base->StartPlayout(audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_base->StartSend(audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    // Now start the Video Send & Receive.
-    error = vie_base->StartSend(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_base->StartReceive(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_renderer->StartRender(capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_renderer->StartRender(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    ViEAutotestEncoderObserver* codec_encoder_observer = NULL;
-    ViEAutotestDecoderObserver* codec_decoder_observer = NULL;
-
-    //  Engine ready, wait for input.
-
-    // Call started.
-    std::cout << std::endl;
-    std::cout << "Custom call started" << std::endl;
-
-    // Modify call or stop call.
-    printf("\n");
-    int selection = FromChoices(
-        "And now?",
-        "Stop the call\n"
-        "Modify the call\n"
-        "Keep the call running indefinitely\n")
-            .WithDefault("Keep the call running indefinitely").Choose();
-    if (selection == 3) {
-        AutoTestSleep(std::numeric_limits<int>::max());
-    }
-
-    while (selection == 2) {
-      // Keep on modifying the call until user stops the call.
-      int modify_selection = FromChoices(
-          "Modify the call:",
-          "Stop call\n"
-          "Change Video Send Codec\n"
-          "Change Video Send Size by Common Resolutions\n"
-          "Change Video Send Size by Width & Height\n"
-          "Change Video Capture Device\n"
-          "Change Video Protection Method\n"
-          "Toggle Encoder Observer\n"
-          "Toggle Decoder Observer\n"
-          "Print Call Information\n"
-          "Print Call Statistics\n"
-          "Toggle Image Scaling (Warning: high CPU usage when enabled)\n")
-              .WithDefault("Stop call")
-              .Choose();
-
-      switch (modify_selection) {
-        case 1:
-          selection = 1;
-          break;
-        case 2:
-          // Change video codec.
-          SetVideoCodecType(vie_codec, &video_send_codec);
-          SetVideoCodecSize(&video_send_codec);
-          SetVideoCodecBitrate(&video_send_codec);
-          SetVideoCodecMinBitrate(&video_send_codec);
-          SetVideoCodecMaxBitrate(&video_send_codec);
-          SetVideoCodecMaxFramerate(&video_send_codec);
-          SetVideoCodecTemporalLayer(&video_send_codec);
-          PrintCallInformation(ip_address.c_str(), device_name,
-                               unique_id, video_send_codec,
-                               video_tx_port, video_rx_port,
-                               audio_capture_device_name,
-                               audio_playbackDeviceName, audio_codec,
-                               audio_tx_port, audio_rx_port, protection_method);
-          error = vie_codec->SetSendCodec(video_channel, video_send_codec);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_codec->SetReceiveCodec(video_channel, video_send_codec);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          break;
-        case 3:
-          // Change Video codec size by common resolution.
-          SetVideoCodecResolution(&video_send_codec);
-          PrintCallInformation(ip_address.c_str(), device_name,
-                               unique_id, video_send_codec,
-                               video_tx_port, video_rx_port,
-                               audio_capture_device_name,
-                               audio_playbackDeviceName, audio_codec,
-                               audio_tx_port, audio_rx_port, protection_method);
-          error = vie_codec->SetSendCodec(video_channel, video_send_codec);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_codec->SetReceiveCodec(video_channel, video_send_codec);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          break;
-        case 4:
-          // Change video codec by size height and width.
-          SetVideoCodecSize(&video_send_codec);
-          PrintCallInformation(ip_address.c_str(), device_name,
-                               unique_id, video_send_codec,
-                               video_tx_port, video_rx_port,
-                               audio_capture_device_name,
-                               audio_playbackDeviceName, audio_codec,
-                               audio_tx_port, audio_rx_port, protection_method);
-          error = vie_codec->SetSendCodec(video_channel, video_send_codec);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_codec->SetReceiveCodec(video_channel, video_send_codec);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          break;
-        case 5:
-          error = vie_renderer->StopRender(capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_renderer->RemoveRenderer(capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_capture->StopCapture(capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_capture->DisconnectCaptureDevice(video_channel);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_capture->ReleaseCaptureDevice(capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          memset(device_name, 0, KMaxUniqueIdLength);
-          memset(unique_id, 0, KMaxUniqueIdLength);
-          if (!GetVideoDevice(vie_base, vie_capture, device_name, unique_id))
-            return number_of_errors;
-          capture_id = 0;
-          error = vie_capture->AllocateCaptureDevice(unique_id,
-                                                     KMaxUniqueIdLength,
-                                                     capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_capture->ConnectCaptureDevice(capture_id,
-                                                    video_channel);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-
-          assert(FLAGS_render_custom_call_remote_to == "" &&
-                 "Not implemented to change video capture device when "
-                 "rendering to file!");
-
-          error = vie_capture->StartCapture(capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_renderer->AddRenderer(capture_id, _window1, 0, 0.0, 0.0,
-                                            1.0, 1.0);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          error = vie_renderer->StartRender(capture_id);
-          number_of_errors += ViETest::TestError(error == 0,
-                                                 "ERROR: %s at line %d",
-                                                 __FUNCTION__, __LINE__);
-          break;
-        case 6:
-          // Change the Video Protection.
-          protection_method = GetVideoProtection();
-          SetVideoProtection(vie_codec, vie_rtp_rtcp,
-                             video_channel, protection_method);
-          break;
-        case 7:
-          // Toggle Encoder Observer.
-          if (!codec_encoder_observer) {
-            std::cout << "Registering Encoder Observer" << std::endl;
-            codec_encoder_observer = new ViEAutotestEncoderObserver();
-            error = vie_codec->RegisterEncoderObserver(video_channel,
-                                                       *codec_encoder_observer);
-            number_of_errors += ViETest::TestError(error == 0,
-                                                   "ERROR: %s at line %d",
-                                                   __FUNCTION__, __LINE__);
-          } else {
-            std::cout << "Deregistering Encoder Observer" << std::endl;
-            error = vie_codec->DeregisterEncoderObserver(video_channel);
-            delete codec_encoder_observer;
-            codec_encoder_observer = NULL;
-            number_of_errors += ViETest::TestError(error == 0,
-                                                   "ERROR: %s at line %d",
-                                                   __FUNCTION__, __LINE__);
-          }
-          break;
-        case 8:
-          // Toggle Decoder Observer.
-          if (!codec_decoder_observer) {
-            std::cout << "Registering Decoder Observer" << std::endl;
-            codec_decoder_observer = new ViEAutotestDecoderObserver();
-            error = vie_codec->RegisterDecoderObserver(video_channel,
-                                                       *codec_decoder_observer);
-            number_of_errors += ViETest::TestError(error == 0,
-                                                   "ERROR: %s at line %d",
-                                                   __FUNCTION__, __LINE__);
-          } else {
-            std::cout << "Deregistering Decoder Observer" << std::endl;
-            error = vie_codec->DeregisterDecoderObserver(video_channel);
-            delete codec_decoder_observer;
-            codec_decoder_observer = NULL;
-            number_of_errors += ViETest::TestError(error == 0,
-                                                   "ERROR: %s at line %d",
-                                                   __FUNCTION__, __LINE__);
-          }
-          break;
-        case 9:
-          // Print Call information..
-          PrintCallInformation(ip_address.c_str(), device_name,
-                               unique_id, video_send_codec,
-                               video_tx_port, video_rx_port,
-                               audio_capture_device_name,
-                               audio_playbackDeviceName,
-                               audio_codec, audio_tx_port,
-                               audio_rx_port, protection_method);
-          PrintVideoStreamInformation(vie_codec,
-                                      video_channel);
-          break;
-        case 10:
-          // Print Call statistics.
-          PrintRTCCPStatistics(vie_rtp_rtcp, video_channel,
-                               kSendStatistic);
-          PrintRTCCPStatistics(vie_rtp_rtcp, video_channel,
-                               kReceivedStatistic);
-          PrintRTPStatistics(vie_rtp_rtcp, video_channel);
-          PrintBandwidthUsage(vie_rtp_rtcp, video_channel);
-          PrintCodecStatistics(vie_codec, video_channel,
-                               kSendStatistic);
-          PrintCodecStatistics(vie_codec, video_channel,
-                               kReceivedStatistic);
-          PrintGetDiscardedPackets(vie_codec, video_channel);
-          break;
-        case 11:
-          is_image_scale_enabled = !is_image_scale_enabled;
-          vie_codec->SetImageScaleStatus(video_channel, is_image_scale_enabled);
-          if (is_image_scale_enabled) {
-            std::cout << "Image Scale is now enabled" << std::endl;
-          } else {
-            std::cout << "Image Scale is now disabled" << std::endl;
-          }
-          break;
-        default:
-          assert(false);
-          break;
-      }
-    }
-
-    if (FLAGS_render_custom_call_remote_to != "")
-      file_renderer.StopRendering();
-
-    // Testing finished. Tear down Voice and Video Engine.
-    // Tear down the VoE first.
-    error = voe_base->StopReceive(audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_base->StopPlayout(audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = voe_base->DeleteChannel(audio_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    // Now tear down the ViE engine.
-    error = vie_base->DisconnectAudioChannel(video_channel);
-
-    voice_channel_transport.reset(NULL);
-
-    // If Encoder/Decoder Observer is running, delete them.
-    if (codec_encoder_observer) {
-      error = vie_codec->DeregisterEncoderObserver(video_channel);
-      delete codec_encoder_observer;
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-    }
-    if (codec_decoder_observer) {
-      error = vie_codec->DeregisterDecoderObserver(video_channel);
-      delete codec_decoder_observer;
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-    }
-
-    error = vie_base->StopReceive(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_base->StopSend(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_renderer->StopRender(capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_renderer->StopRender(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_renderer->RemoveRenderer(capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_renderer->RemoveRenderer(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_capture->StopCapture(capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_capture->DisconnectCaptureDevice(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    error = vie_capture->ReleaseCaptureDevice(capture_id);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    video_channel_transport.reset(NULL);
-
-    error = vie_base->DeleteChannel(video_channel);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    int remaining_interfaces = 0;
-    remaining_interfaces = vie_codec->Release();
-    number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    remaining_interfaces = vie_capture->Release();
-    number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    remaining_interfaces = vie_rtp_rtcp->Release();
-    number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    remaining_interfaces = vie_renderer->Release();
-    number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    remaining_interfaces = vie_network->Release();
-    number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    remaining_interfaces = vie_base->Release();
-    number_of_errors += ViETest::TestError(remaining_interfaces == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    bool deleted = webrtc::VideoEngine::Delete(vie);
-    number_of_errors += ViETest::TestError(deleted == true,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-
-    ViETest::Log(" ");
-    ViETest::Log(" ViE Autotest Custom Call Started");
-    ViETest::Log("========================================");
-    ViETest::Log(" ");
-  }
-  return number_of_errors;
-}
-
-bool GetVideoDevice(webrtc::ViEBase* vie_base,
-                    webrtc::ViECapture* vie_capture,
-                    char* capture_device_name,
-                    char* capture_device_unique_id) {
-  int error = 0;
-  int number_of_errors = 0;
-
-  const unsigned int KMaxDeviceNameLength = 128;
-  const unsigned int KMaxUniqueIdLength = 256;
-  char device_name[KMaxDeviceNameLength];
-  char unique_id[KMaxUniqueIdLength];
-
-  if (vie_capture->NumberOfCaptureDevices() == 0) {
-    printf("You have no capture devices plugged into your system.\n");
-    return false;
-  }
-
-  std::string capture_choices;
-  std::string first_device;
-  for (int i = 0; i < vie_capture->NumberOfCaptureDevices(); i++) {
-    memset(device_name, 0, KMaxDeviceNameLength);
-    memset(unique_id, 0, KMaxUniqueIdLength);
-
-    error = vie_capture->GetCaptureDevice(i, device_name,
-        KMaxDeviceNameLength,
-        unique_id,
-        KMaxUniqueIdLength);
-    number_of_errors += ViETest::TestError(error == 0,
-        "ERROR: %s at line %d",
-        __FUNCTION__, __LINE__);
-    const int kCaptureLineLength =
-        KMaxDeviceNameLength + KMaxUniqueIdLength + 8;
-    char capture_line[kCaptureLineLength];
-    sprintf(capture_line, "%s (%s)", device_name, unique_id);
-    capture_choices += capture_line;
-    capture_choices += "\n";
-    if (first_device.empty())
-      first_device = capture_line;
-  }
-
-  int choice = FromChoices("Available Video Capture Devices", capture_choices)
-      .WithDefault(first_device)
-      .Choose();
-
-  error = vie_capture->GetCaptureDevice(
-      choice - 1, device_name, KMaxDeviceNameLength, unique_id,
-      KMaxUniqueIdLength);
-  number_of_errors += ViETest::TestError(error == 0,
-      "ERROR: %s at line %d",
-      __FUNCTION__, __LINE__);
-  strcpy(capture_device_unique_id, unique_id);
-  strcpy(capture_device_name, device_name);
-  return true;
-}
-
-bool GetAudioDevices(webrtc::VoEBase* voe_base,
-                     webrtc::VoEHardware* voe_hardware,
-                     char* recording_device_name,
-                     int& recording_device_index,
-                     char* playback_device_name,
-                     int& playback_device_index) {
-  int error = 0;
-  int number_of_errors = 0;
-
-  const unsigned int KMaxDeviceNameLength = 128;
-  const unsigned int KMaxUniqueIdLength = 128;
-  char recording_device_unique_name[KMaxDeviceNameLength];
-  char playback_device_unique_name[KMaxUniqueIdLength];
-
-  int number_of_recording_devices = -1;
-  error = voe_hardware->GetNumOfRecordingDevices(number_of_recording_devices);
-  number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-
-  recording_device_index = -1;
-  playback_device_index = -1;
-
-  std::string device_choices;
-  std::string default_recording_line;
-  for (int i = 0; i < number_of_recording_devices; ++i) {
-    error = voe_hardware->GetRecordingDeviceName(
-        i, recording_device_name, recording_device_unique_name);
-    number_of_errors += ViETest::TestError(error == 0,
-        "ERROR: %s at line %d",
-        __FUNCTION__, __LINE__);
-
-    device_choices += recording_device_name;
-    device_choices += "\n";
-    if (default_recording_line.empty())
-      default_recording_line = recording_device_name;
-  }
-
-  int choice = FromChoices("Available audio capture devices:", device_choices)
-      .WithDefault(default_recording_line)
-      .Choose();
-
-  recording_device_index = choice - 1;
-  error = voe_hardware->GetRecordingDeviceName(
-      recording_device_index, recording_device_name,
-      recording_device_unique_name);
-  number_of_errors += ViETest::TestError(
-      error == 0, "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-
-  int number_of_playback_devices = -1;
-  error = voe_hardware->GetNumOfPlayoutDevices(number_of_playback_devices);
-  number_of_errors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-
-
-  std::string playback_choices;
-  std::string default_playback_line;
-  for (int i = 0; i < number_of_playback_devices; i++) {
-    error = voe_hardware->GetPlayoutDeviceName(i,
-        playback_device_name,
-        playback_device_unique_name);
-    number_of_errors += ViETest::TestError(
-        error == 0, "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-    playback_choices += playback_device_name;
-    playback_choices += "\n";
-    if (default_playback_line.empty())
-      default_playback_line = playback_device_name;
-  }
-
-  choice = FromChoices("Available audio playout devices:", playback_choices)
-      .WithDefault(default_playback_line)
-      .Choose();
-
-  playback_device_index = choice - 1;
-  error = voe_hardware->GetPlayoutDeviceName(playback_device_index,
-      playback_device_name,
-      playback_device_unique_name);
-  number_of_errors += ViETest::TestError(error == 0,
-      "ERROR: %s at line %d",
-      __FUNCTION__, __LINE__);
-  return true;
-}
-
-// General helper functions.
-
-std::string GetIPAddress() {
-  class IpValidator : public webrtc::InputValidator {
-   public:
-    bool InputOk(const std::string& input) const {
-      // Just check quickly that it's on the form x.y.z.w
-      return std::count(input.begin(), input.end(), '.') == 3;
-    }
-  };
-  return TypedInput("Enter destination IP.")
-      .WithDefault(DEFAULT_SEND_IP)
-      .WithInputValidator(new IpValidator())
-      .AskForInput();
-}
-
-// Video settings functions.
-
-void GetVideoPorts(int* tx_port, int* rx_port) {
-  std::string tx_input = TypedInput("Enter video send port.")
-      .WithDefault(DEFAULT_VIDEO_PORT)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, 65536))
-      .AskForInput();
-  *tx_port = atoi(tx_input.c_str());
-
-  std::string rx_input = TypedInput("Enter video receive port.")
-      .WithDefault(DEFAULT_VIDEO_PORT)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, 65536))
-      .AskForInput();
-  *rx_port = atoi(rx_input.c_str());
-}
-
-// Audio settings functions.
-
-void GetAudioPorts(int* tx_port, int* rx_port) {
-  std::string tx_input = TypedInput("Enter audio send port.")
-      .WithDefault(DEFAULT_AUDIO_PORT)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, 65536))
-      .AskForInput();
-  *tx_port = atoi(tx_input.c_str());
-
-  std::string rx_input = TypedInput("Enter audio receive port.")
-      .WithDefault(DEFAULT_AUDIO_PORT)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, 65536))
-      .AskForInput();
-  *rx_port = atoi(rx_input.c_str());
-}
-
-bool GetAudioCodec(webrtc::VoECodec* voe_codec,
-                   webrtc::CodecInst& audio_codec) {
-  int error = 0;
-  memset(&audio_codec, 0, sizeof(webrtc::CodecInst));
-
-  std::string default_codec_line;
-  std::string codec_choices;
-  for (int codec_idx = 0; codec_idx < voe_codec->NumOfCodecs(); codec_idx++) {
-    error = voe_codec->GetCodec(codec_idx, audio_codec);
-    ViETest::TestError(error == 0,
-        "ERROR: %s at line %d",
-        __FUNCTION__, __LINE__);
-
-    char codec_line[128];
-    sprintf(codec_line, "%s type: %d freq: %d chan: %d",
-        audio_codec.plname, audio_codec.pltype, audio_codec.plfreq,
-        audio_codec.channels);
-    codec_choices += codec_line;
-    codec_choices += "\n";
-
-    if (strcmp(audio_codec.plname, DEFAULT_AUDIO_CODEC) == 0) {
-      default_codec_line = codec_line;
-    }
-  }
-  assert(!default_codec_line.empty() && "Default codec doesn't exist.");
-
-  int codec_selection = FromChoices("Available Audio Codecs:", codec_choices)
-            .WithDefault(default_codec_line)
-            .Choose();
-
-  error = voe_codec->GetCodec(codec_selection - 1, audio_codec);
-  ViETest::TestError(error == 0,
-      "ERROR: %s at line %d",
-      __FUNCTION__, __LINE__);
-  return true;
-}
-
-void PrintCallInformation(const char* IP, const char* video_capture_device_name,
-                          const char* video_capture_unique_id,
-                          webrtc::VideoCodec video_codec,
-                          int video_tx_port, int video_rx_port,
-                          const char* audio_capture_device_name,
-                          const char* audio_playbackDeviceName,
-                          webrtc::CodecInst audio_codec,
-                          int audio_tx_port, int audio_rx_port,
-                          int protection_method) {
-  std::string str;
-
-  std::cout << "************************************************"
-            << std::endl;
-  std::cout << "The call has the following settings: " << std::endl;
-  std::cout << "\tIP: " << IP << std::endl;
-  std::cout << "\tVideo Capture Device: " << video_capture_device_name
-            << std::endl;
-  std::cout << "\t\tName: " << video_capture_device_name << std::endl;
-  std::cout << "\t\tUniqueId: " << video_capture_unique_id << std::endl;
-  PrintVideoCodec(video_codec);
-  std::cout << "\t Video Tx Port: " << video_tx_port << std::endl;
-  std::cout << "\t Video Rx Port: " << video_rx_port << std::endl;
-  std::cout << "\t Video Protection Method (NOTE: Starts at 1 now): "
-            << protection_method  << std::endl;
-  std::cout << "\tAudio Capture Device: " << audio_capture_device_name
-            << std::endl;
-  std::cout << "\tAudio Playback Device: " << audio_playbackDeviceName
-            << std::endl;
-  std::cout << "\tAudio Codec: " << std::endl;
-  std::cout << "\t\tplname: " << audio_codec.plname << std::endl;
-  std::cout << "\t\tpltype: " << static_cast<int>(audio_codec.pltype)
-            << std::endl;
-  std::cout << "\t Audio Tx Port: " << audio_tx_port << std::endl;
-  std::cout << "\t Audio Rx Port: " << audio_rx_port << std::endl;
-  std::cout << "************************************************"
-            << std::endl;
-}
-
-void SetVideoCodecType(webrtc::ViECodec* vie_codec,
-                       webrtc::VideoCodec* video_codec) {
-  int error = 0;
-  int number_of_errors = 0;
-  memset(video_codec, 0, sizeof(webrtc::VideoCodec));
-
-  std::string codec_choices;
-  std::string default_codec_line;
-  for (int i = 0; i < vie_codec->NumberOfCodecs(); i++) {
-    error = vie_codec->GetCodec(i, *video_codec);
-    number_of_errors += ViETest::TestError(
-        error == 0, "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-
-    codec_choices += video_codec->plName;
-    codec_choices += "\n";
-    if (strcmp(video_codec->plName, DEFAULT_VIDEO_CODEC) == 0)
-      default_codec_line = video_codec->plName;
-  }
-  assert(!default_codec_line.empty() && "Default does not exist.");
-
-  int choice = FromChoices("Available Video Codecs", codec_choices)
-      .WithDefault(default_codec_line)
-      .Choose();
-  error = vie_codec->GetCodec(choice - 1, *video_codec);
-  number_of_errors += ViETest::TestError(
-      error == 0, "ERROR: %s at line %d", __FUNCTION__, __LINE__);
-
-  if (video_codec->codecType == webrtc::kVideoCodecI420) {
-    video_codec->width = 176;
-    video_codec->height = 144;
-  }
-}
-
-void SetVideoCodecResolution(webrtc::VideoCodec* video_codec) {
-  if (video_codec->codecType != webrtc::kVideoCodecVP8) {
-    printf("Can only change codec size if it's VP8\n");
-    return;
-  }
-
-  int choice = FromChoices(
-      "Available Common Resolutions:",
-      "SQCIF (128X96)\n"
-      "QQVGA (160X120)\n"
-      "QCIF  (176X144)\n"
-      "CIF   (352X288)\n"
-      "VGA   (640X480)\n"
-      "WVGA  (800x480)\n"
-      "4CIF  (704X576)\n"
-      "SVGA  (800X600)\n"
-      "HD    (1280X720)\n"
-      "XGA   (1024x768)\n")
-          .Choose();
-
-  switch (choice) {
-    case 1:
-      video_codec->width = 128;
-      video_codec->height = 96;
-      break;
-    case 2:
-      video_codec->width = 160;
-      video_codec->height = 120;
-      break;
-    case 3:
-      video_codec->width = 176;
-      video_codec->height = 144;
-      break;
-    case 4:
-      video_codec->width = 352;
-      video_codec->height = 288;
-      break;
-    case 5:
-      video_codec->width = 640;
-      video_codec->height = 480;
-      break;
-    case 6:
-      video_codec->width = 800;
-      video_codec->height = 480;
-      break;
-    case 7:
-      video_codec->width = 704;
-      video_codec->height = 576;
-      break;
-    case 8:
-      video_codec->width = 800;
-      video_codec->height = 600;
-      break;
-    case 9:
-      video_codec->width = 1280;
-      video_codec->height = 720;
-      break;
-    case 10:
-      video_codec->width = 1024;
-      video_codec->height = 768;
-      break;
-  }
-}
-
-void SetVideoCodecSize(webrtc::VideoCodec* video_codec) {
-  if (video_codec->codecType != webrtc::kVideoCodecVP8) {
-    printf("Can only change codec size if it's VP8\n");
-    return;
-  }
-
-  std::string input = TypedInput("Choose video width.")
-      .WithDefault(DEFAULT_VIDEO_CODEC_WIDTH)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, INT_MAX))
-      .AskForInput();
-  video_codec->width = atoi(input.c_str());
-
-  input = TypedInput("Choose video height.")
-      .WithDefault(DEFAULT_VIDEO_CODEC_HEIGHT)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, INT_MAX))
-      .AskForInput();
-  video_codec->height = atoi(input.c_str());
-}
-
-void SetVideoCodecBitrate(webrtc::VideoCodec* video_codec) {
-  std::string input = TypedInput("Choose start rate (in kbps).")
-      .WithDefault(DEFAULT_VIDEO_CODEC_BITRATE)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, INT_MAX))
-      .AskForInput();
-
-  video_codec->startBitrate = atoi(input.c_str());
-}
-
-void SetVideoCodecMaxBitrate(webrtc::VideoCodec* video_codec) {
-  std::string input = TypedInput("Choose max bitrate (in kbps).")
-      .WithDefault(DEFAULT_VIDEO_CODEC_MAX_BITRATE)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, INT_MAX))
-      .AskForInput();
-
-  video_codec->maxBitrate = atoi(input.c_str());
-}
-
-void SetVideoCodecMinBitrate(webrtc::VideoCodec* video_codec) {
-  std::string input = TypedInput("Choose min bitrate (in kbps).")
-      .WithDefault(DEFAULT_VIDEO_CODEC_MIN_BITRATE)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, INT_MAX))
-      .AskForInput();
-
-  video_codec->minBitrate = atoi(input.c_str());
-}
-
-void SetVideoCodecMaxFramerate(webrtc::VideoCodec* video_codec) {
-  std::string input = TypedInput("Choose max framerate (in fps).")
-      .WithDefault(DEFAULT_VIDEO_CODEC_MAX_FRAMERATE)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(1, INT_MAX))
-      .AskForInput();
-  video_codec->maxFramerate = atoi(input.c_str());
-}
-
-void SetVideoCodecTemporalLayer(webrtc::VideoCodec* video_codec) {
-  if (video_codec->codecType != webrtc::kVideoCodecVP8)
-    return;
-
-  std::string input = TypedInput("Choose number of temporal layers (0 to 4).")
-      .WithDefault(DEFAULT_TEMPORAL_LAYER)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(0, 4))
-      .AskForInput();
-  video_codec->codecSpecific.VP8.numberOfTemporalLayers = atoi(input.c_str());
-}
-
-// GetVideoProtection only prints the prompt to get a number
-// that SetVideoProtection method uses.
-VideoProtectionMethod GetVideoProtection() {
-  int choice = FromChoices(
-      "Available Video Protection Methods:",
-      "None\n"
-      "FEC\n"
-      "NACK\n"
-      "NACK+FEC\n")
-          .WithDefault(DEFAULT_VIDEO_PROTECTION_METHOD)
-          .Choose();
-
-  assert(choice >= kProtectionMethodNone &&
-         choice <= kProtectionMethodHybridNackAndFec);
-  return static_cast<VideoProtectionMethod>(choice);
-}
-
-bool SetVideoProtection(webrtc::ViECodec* vie_codec,
-                        webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                        int video_channel,
-                        VideoProtectionMethod protection_method) {
-  int error = 0;
-  int number_of_errors = 0;
-  webrtc::VideoCodec video_codec;
-
-  memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-
-  // Set all video protection to false initially
-  error = vie_rtp_rtcp->SetHybridNACKFECStatus(video_channel, false,
-                                               VCM_RED_PAYLOAD_TYPE,
-                                               VCM_ULPFEC_PAYLOAD_TYPE);
-  number_of_errors += ViETest::TestError(error == 0,
-                                         "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-  error = vie_rtp_rtcp->SetFECStatus(video_channel, false,
-                                     VCM_RED_PAYLOAD_TYPE,
-                                     VCM_ULPFEC_PAYLOAD_TYPE);
-  number_of_errors += ViETest::TestError(error == 0,
-                                         "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-  error = vie_rtp_rtcp->SetNACKStatus(video_channel, false);
-  number_of_errors += ViETest::TestError(error == 0,
-                                         "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-  // Set video protection for FEC, NACK or Hybrid.
-  switch (protection_method) {
-    case kProtectionMethodNone:
-      // No protection selected, all protection already at false.
-      std::cout << "Call using None protection Method" << std::endl;
-      break;
-    case kProtectionMethodFecOnly:
-      std::cout << "Call using FEC protection Method" << std::endl;
-      error = vie_rtp_rtcp->SetFECStatus(video_channel, true,
-                                         VCM_RED_PAYLOAD_TYPE,
-                                         VCM_ULPFEC_PAYLOAD_TYPE);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-    case kProtectionMethodNackOnly:
-      std::cout << "Call using NACK protection Method" << std::endl;
-      error = vie_rtp_rtcp->SetNACKStatus(video_channel, true);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-    case kProtectionMethodHybridNackAndFec:
-      std::cout << "Call using Hybrid NACK and FEC protection Method"
-                << std::endl;
-      error = vie_rtp_rtcp->SetHybridNACKFECStatus(video_channel, true,
-                                                   VCM_RED_PAYLOAD_TYPE,
-                                                   VCM_ULPFEC_PAYLOAD_TYPE);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-  }
-
-  // Set receive codecs for FEC and hybrid NACK/FEC.
-  if (protection_method == kProtectionMethodFecOnly ||
-      protection_method == kProtectionMethodHybridNackAndFec) {
-    // RED.
-    error = vie_codec->GetCodec(vie_codec->NumberOfCodecs() - 2,
-                                video_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    video_codec.plType = VCM_RED_PAYLOAD_TYPE;
-    error = vie_codec->SetReceiveCodec(video_channel, video_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    std::cout << "RED Codec Information:" << std::endl;
-    PrintVideoCodec(video_codec);
-    // ULPFEC.
-    error = vie_codec->GetCodec(vie_codec->NumberOfCodecs() - 1,
-                                video_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    video_codec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
-    error = vie_codec->SetReceiveCodec(video_channel, video_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    std::cout << "ULPFEC Codec Information:" << std::endl;
-    PrintVideoCodec(video_codec);
-  }
-
-  return true;
-}
-
-// Returns true if REMB, false if TMMBR.
-bool GetBitrateSignaling() {
-  int choice = FromChoices(
-      "Available Bitrate Signaling Methods:",
-      "REMB\n"
-      "TMMBR\n")
-          .WithDefault("REMB")
-          .Choose();
-  return choice == 1;
-}
-
-int GetBufferingDelay() {
-  std::string input = TypedInput("Choose buffering delay (mS).")
-      .WithDefault(DEFAULT_BUFFERING_DELAY_MS)
-      .WithInputValidator(new webrtc::IntegerWithinRangeValidator(0, 10000))
-      .AskForInput();
-  std::string delay_ms = input;
-  return atoi(delay_ms.c_str());
-}
-
-void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                          int video_channel,
-                          StatisticsType stat_type) {
-  int error = 0;
-  int number_of_errors = 0;
-  webrtc::RtcpStatistics rtcp_stats;
-  int64_t rtt_ms = 0;
-
-  switch (stat_type) {
-    case kReceivedStatistic:
-      std::cout << "RTCP Received statistics"
-                << std::endl;
-      // Get and print the Received RTCP Statistics
-      error = vie_rtp_rtcp->GetReceiveChannelRtcpStatistics(video_channel,
-                                                      rtcp_stats,
-                                                      rtt_ms);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-    case kSendStatistic:
-      std::cout << "RTCP Sent statistics"
-                << std::endl;
-      // Get and print the Sent RTCP Statistics
-      error = vie_rtp_rtcp->GetSendChannelRtcpStatistics(video_channel,
-                                                  rtcp_stats,
-                                                  rtt_ms);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-  }
-  std::cout << "\tRTCP fraction of lost packets: "
-            << rtcp_stats.fraction_lost << std::endl;
-  std::cout << "\tRTCP cumulative number of lost packets: "
-            << rtcp_stats.cumulative_lost << std::endl;
-  std::cout << "\tRTCP max received sequence number "
-            << rtcp_stats.extended_max_sequence_number << std::endl;
-  std::cout << "\tRTCP jitter: "
-            << rtcp_stats.jitter << std::endl;
-  std::cout << "\tRTCP round trip (ms): "
-            << rtt_ms << std::endl;
-}
-
-void PrintRTPStatistics(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                        int video_channel) {
-  int error = 0;
-  int number_of_errors = 0;
-  webrtc::StreamDataCounters sent;
-  webrtc::StreamDataCounters received;
-
-  std::cout << "RTP statistics"
-            << std::endl;
-
-  // Get and print the RTP Statistics
-  error = vie_rtp_rtcp->GetRtpStatistics(video_channel, sent, received);
-  number_of_errors += ViETest::TestError(error == 0,
-                                         "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-  std::cout << "\tRTP bytes sent: "
-            << sent.transmitted.payload_bytes << std::endl;
-  std::cout << "\tRTP packets sent: "
-            << sent.transmitted.packets << std::endl;
-  std::cout << "\tRTP bytes received: "
-            << received.transmitted.payload_bytes << std::endl;
-  std::cout << "\tRTP packets received: "
-            << received.transmitted.packets << std::endl;
-}
-
-void PrintBandwidthUsage(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
-                         int video_channel) {
-  int error = 0;
-  int number_of_errors = 0;
-  unsigned int total_bitrate_sent = 0;
-  unsigned int video_bitrate_sent = 0;
-  unsigned int fec_bitrate_sent = 0;
-  unsigned int nack_bitrate_sent = 0;
-  double percentage_fec = 0;
-  double percentage_nack = 0;
-
-  std::cout << "Bandwidth Usage" << std::endl;
-
-  // Get and print Bandwidth usage
-  error = vie_rtp_rtcp->GetBandwidthUsage(video_channel, total_bitrate_sent,
-                                          video_bitrate_sent, fec_bitrate_sent,
-                                          nack_bitrate_sent);
-  number_of_errors += ViETest::TestError(error == 0,
-                                         "ERROR: %s at line %d",
-                                         __FUNCTION__, __LINE__);
-  std::cout << "\tTotal bitrate sent (Kbit/s): "
-            << total_bitrate_sent << std::endl;
-  std::cout << "\tVideo bitrate sent (Kbit/s): "
-            << video_bitrate_sent << std::endl;
-  std::cout << "\tFEC bitrate sent (Kbit/s): "
-            << fec_bitrate_sent << std::endl;
-  percentage_fec =
-      (static_cast<double>(fec_bitrate_sent) /
-      static_cast<double>(total_bitrate_sent)) * 100;
-  std::cout << "\tPercentage FEC bitrate sent from total bitrate: "
-            << percentage_fec << std::endl;
-  std::cout << "\tNACK bitrate sent (Kbit/s): "
-            << nack_bitrate_sent << std::endl;
-  percentage_nack =
-      (static_cast<double>(nack_bitrate_sent) /
-      static_cast<double>(total_bitrate_sent)) * 100;
-  std::cout << "\tPercentage NACK bitrate sent from total bitrate: "
-            << percentage_nack << std::endl;
-}
-
-void PrintCodecStatistics(webrtc::ViECodec* vie_codec,
-                          int video_channel,
-                          StatisticsType stat_type) {
-  int error = 0;
-  int number_of_errors = 0;
-  unsigned int key_frames = 0;
-  unsigned int delta_frames = 0;
-  switch (stat_type) {
-    case kReceivedStatistic:
-      std::cout << "Codec Receive statistics"
-                << std::endl;
-      // Get and print the Receive Codec Statistics
-      error = vie_codec->GetReceiveCodecStatistics(video_channel, key_frames,
-                                                   delta_frames);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-    case kSendStatistic:
-      std::cout << "Codec Send statistics"
-                << std::endl;
-      // Get and print the Send Codec Statistics
-      error = vie_codec->GetSendCodecStatistics(video_channel, key_frames,
-                                                delta_frames);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-      break;
-  }
-  std::cout << "\tNumber of encoded key frames: "
-            << key_frames << std::endl;
-  std::cout << "\tNumber of encoded delta frames: "
-            << delta_frames << std::endl;
-}
-
-void PrintGetDiscardedPackets(webrtc::ViECodec* vie_codec, int video_channel) {
-  std::cout << "Discarded Packets" << std::endl;
-  int discarded_packets = 0;
-  discarded_packets = vie_codec->GetNumDiscardedPackets(video_channel);
-  std::cout << "\tNumber of discarded packets: "
-            << discarded_packets << std::endl;
-}
-
-void PrintVideoStreamInformation(webrtc::ViECodec* vie_codec,
-                                 int video_channel) {
-  webrtc::VideoCodec outgoing_codec;
-  webrtc::VideoCodec incoming_codec;
-
-  memset(&outgoing_codec, 0, sizeof(webrtc::VideoCodec));
-  memset(&incoming_codec, 0, sizeof(webrtc::VideoCodec));
-
-  vie_codec->GetSendCodec(video_channel, outgoing_codec);
-  vie_codec->GetReceiveCodec(video_channel, incoming_codec);
-
-  std::cout << "************************************************"
-            << std::endl;
-  std::cout << "ChannelId: " << video_channel << std::endl;
-  std::cout << "Outgoing Stream information:" << std::endl;
-  PrintVideoCodec(outgoing_codec);
-  std::cout << "Incoming Stream information:" << std::endl;
-  PrintVideoCodec(incoming_codec);
-  std::cout << "************************************************"
-            << std::endl;
-}
-
-void PrintVideoCodec(webrtc::VideoCodec video_codec) {
-  std::cout << "\t\tplName: " << video_codec.plName << std::endl;
-  std::cout << "\t\tplType: " << static_cast<int>(video_codec.plType)
-            << std::endl;
-  std::cout << "\t\twidth: " << video_codec.width << std::endl;
-  std::cout << "\t\theight: " << video_codec.height << std::endl;
-  std::cout << "\t\tstartBitrate: " << video_codec.startBitrate
-            << std::endl;
-  std::cout << "\t\tminBitrate: " << video_codec.minBitrate
-            << std::endl;
-  std::cout << "\t\tmaxBitrate: " << video_codec.maxBitrate
-            << std::endl;
-  std::cout << "\t\tmaxFramerate: "
-            << static_cast<int>(video_codec.maxFramerate) << std::endl;
-  if (video_codec.codecType == webrtc::kVideoCodecVP8) {
-    int number_of_layers =
-        static_cast<int>(video_codec.codecSpecific.VP8.numberOfTemporalLayers);
-    std::cout << "\t\tVP8 Temporal Layer: " << number_of_layers << std::endl;
-  }
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc
deleted file mode 100644
index de8d2d1dea..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_image_process.cc
-//
-
-// Settings
-#include "webrtc/engine_configurations.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-
-class MyEffectFilter: public webrtc::ViEEffectFilter
-{
-public:
-    MyEffectFilter() {}
-
-    ~MyEffectFilter() {}
-
-    virtual int Transform(size_t size,
-                          unsigned char* frame_buffer,
-                          int64_t ntp_time_ms,
-                          unsigned int timestamp,
-                          unsigned int width,
-                          unsigned int height)
-    {
-        // Black and white
-        memset(frame_buffer + (2 * size) / 3, 0x7f, size / 3);
-        return 0;
-    }
-};
-
-void ViEAutoTest::ViEImageProcessStandardTest()
-{
-    //***************************************************************
-    //	Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-    int rtpPort = 6000;
-    // Create VIE
-    TbInterfaces ViE("ViEImageProcessStandardTest");
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-    // Create a capture device
-    TbCaptureDevice tbCapture(ViE);
-
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-    tbChannel.StartReceive(rtpPort);
-    tbChannel.StartSend(rtpPort);
-
-    MyEffectFilter effectFilter;
-
-    RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
-    ViETest::Log("Capture device is renderered in Window 1");
-    ViETest::Log("Remote stream is renderered in Window 2");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    //***************************************************************
-    //	Engine ready. Begin testing class
-    //***************************************************************
-
-
-    EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
-        tbCapture.captureId, effectFilter));
-
-    ViETest::Log("Black and white filter registered for capture device, "
-                 "affects both windows");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
-        tbCapture.captureId));
-
-    EXPECT_EQ(0, ViE.image_process->RegisterRenderEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-
-    ViETest::Log("Remove capture effect filter, adding filter for incoming "
-                 "stream");
-    ViETest::Log("Only Window 2 should be black and white");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    StopRenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
-    tbCapture.Disconnect(tbChannel.videoChannel);
-
-    int rtpPort2 = rtpPort + 100;
-    // Create a video channel
-    TbVideoChannel tbChannel2(ViE, webrtc::kVideoCodecVP8);
-
-    tbCapture.ConnectTo(tbChannel2.videoChannel);
-    tbChannel2.StartReceive(rtpPort2);
-    tbChannel2.StartSend(rtpPort2);
-
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbChannel2.videoChannel, _window1, 1, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbChannel2.videoChannel));
-    EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
-        tbChannel.videoChannel));
-
-    ViETest::Log("Local renderer removed, added new channel and rendering in "
-                 "Window1.");
-
-    EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
-        tbCapture.captureId, effectFilter));
-
-    ViETest::Log("Black and white filter registered for capture device, "
-                 "affects both windows");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
-        tbCapture.captureId));
-
-    EXPECT_EQ(0, ViE.image_process->RegisterSendEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-
-    ViETest::Log("Capture filter removed.");
-    ViETest::Log("Black and white filter registered for one channel, Window2 "
-                 "should be black and white");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
-        tbChannel.videoChannel));
-
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel2.videoChannel));
-
-    tbCapture.Disconnect(tbChannel2.videoChannel);
-
-    //***************************************************************
-    //	Testing finished. Tear down Video Engine
-    //***************************************************************
-}
-
-void ViEAutoTest::ViEImageProcessExtendedTest()
-{
-}
-
-void ViEAutoTest::ViEImageProcessAPITest()
-{
-    TbInterfaces ViE("ViEImageProcessAPITest");
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-    TbCaptureDevice tbCapture(ViE);
-
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-
-    MyEffectFilter effectFilter;
-
-    //
-    // Capture effect filter
-    //
-    // Add effect filter
-    EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
-        tbCapture.captureId, effectFilter));
-    // Add again -> error
-    EXPECT_NE(0, ViE.image_process->RegisterCaptureEffectFilter(
-        tbCapture.captureId, effectFilter));
-    EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
-        tbCapture.captureId));
-    EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
-        tbCapture.captureId));
-
-    // Non-existing capture device
-    EXPECT_NE(0, ViE.image_process->RegisterCaptureEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-
-    //
-    // Render effect filter
-    //
-    EXPECT_EQ(0, ViE.image_process->RegisterRenderEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-    EXPECT_NE(0, ViE.image_process->RegisterRenderEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-    EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
-        tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
-        tbChannel.videoChannel));
-
-    // Non-existing channel id
-    EXPECT_NE(0, ViE.image_process->RegisterRenderEffectFilter(
-        tbCapture.captureId, effectFilter));
-
-    //
-    // Send effect filter
-    //
-    EXPECT_EQ(0, ViE.image_process->RegisterSendEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-    EXPECT_NE(0, ViE.image_process->RegisterSendEffectFilter(
-        tbChannel.videoChannel, effectFilter));
-    EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
-        tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
-        tbChannel.videoChannel));
-    EXPECT_NE(0, ViE.image_process->RegisterSendEffectFilter(
-        tbCapture.captureId, effectFilter));
-
-    //
-    // Deflickering
-    //
-    EXPECT_EQ(0, ViE.image_process->EnableDeflickering(
-        tbCapture.captureId, true));
-    EXPECT_NE(0, ViE.image_process->EnableDeflickering(
-        tbCapture.captureId, true));
-    EXPECT_EQ(0, ViE.image_process->EnableDeflickering(
-        tbCapture.captureId, false));
-    EXPECT_NE(0, ViE.image_process->EnableDeflickering(
-        tbCapture.captureId, false));
-    EXPECT_NE(0, ViE.image_process->EnableDeflickering(
-        tbChannel.videoChannel, true));
-
-    //
-    // Color enhancement
-    //
-    EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
-        tbChannel.videoChannel, false));
-    EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
-        tbChannel.videoChannel, true));
-    EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
-        tbChannel.videoChannel, false));
-    EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
-        tbCapture.captureId, true));
-
-    tbCapture.Disconnect(tbChannel.videoChannel);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc
deleted file mode 100644
index 337bf8a656..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_linux.cc
-//
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_linux.h"
-
-#include <string>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager()
-    : _hdsp1(NULL),
-      _hdsp2(NULL) {
-}
-
-ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
-  TerminateWindows();
-}
-
-void* ViEAutoTestWindowManager::GetWindow1() {
-  return reinterpret_cast<void*>(_hwnd1);
-}
-
-void* ViEAutoTestWindowManager::GetWindow2() {
-  return reinterpret_cast<void*>(_hwnd2);
-}
-
-int ViEAutoTestWindowManager::TerminateWindows() {
-  if (_hdsp1) {
-    ViEDestroyWindow(&_hwnd1, _hdsp1);
-    _hdsp1 = NULL;
-  }
-  if (_hdsp2) {
-    ViEDestroyWindow(&_hwnd2, _hdsp2);
-    _hdsp2 = NULL;
-  }
-  return 0;
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
-                                            AutoTestRect window2Size,
-                                            void* window1Title,
-                                            void* window2Title) {
-  ViECreateWindow(&_hwnd1, &_hdsp1, window1Size.origin.x,
-                  window1Size.origin.y, window1Size.size.width,
-                  window1Size.size.height,
-                  reinterpret_cast<char*>(window1Title));
-  ViECreateWindow(&_hwnd2, &_hdsp2, window2Size.origin.x,
-                  window2Size.origin.y, window2Size.size.width,
-                  window2Size.size.height,
-                  reinterpret_cast<char*>(window2Title));
-
-  return 0;
-}
-
-int ViEAutoTestWindowManager::ViECreateWindow(Window *out_window,
-                                              Display **out_display, int x_pos,
-                                              int y_pos, int width, int height,
-                                              char* title) {
-  Display* display = XOpenDisplay(NULL);
-  if (display == NULL) {
-    // There's no point to continue if this happens: nothing will work anyway.
-    printf("Failed to connect to X server: X environment likely broken\n");
-    exit(-1);
-  }
-
-  int screen = DefaultScreen(display);
-
-  // Try to establish a 24-bit TrueColor display
-  // (our environment must allow this).
-  XVisualInfo visual_info;
-  if (XMatchVisualInfo(display, screen, 24, TrueColor, &visual_info) == 0) {
-    printf("Failed to establish 24-bit TrueColor in X environment.\n");
-    exit(-1);
-  }
-
-  // Create suitable window attributes.
-  XSetWindowAttributes window_attributes;
-  window_attributes.colormap = XCreateColormap(
-      display, DefaultRootWindow(display), visual_info.visual, AllocNone);
-  window_attributes.event_mask = StructureNotifyMask | ExposureMask;
-  window_attributes.background_pixel = 0;
-  window_attributes.border_pixel = 0;
-
-  unsigned long attribute_mask = CWBackPixel | CWBorderPixel | CWColormap |
-                                 CWEventMask;
-
-  Window _window = XCreateWindow(display, DefaultRootWindow(display), x_pos,
-                                 y_pos, width, height, 0, visual_info.depth,
-                                 InputOutput, visual_info.visual,
-                                 attribute_mask, &window_attributes);
-
-  // Set window name.
-  XStoreName(display, _window, title);
-  XSetIconName(display, _window, title);
-
-  // Make x report events for mask.
-  XSelectInput(display, _window, StructureNotifyMask);
-
-  // Map the window to the display.
-  XMapWindow(display, _window);
-
-  // Wait for map event.
-  XEvent event;
-  do {
-    XNextEvent(display, &event);
-  } while (event.type != MapNotify || event.xmap.event != _window);
-
-  *out_window = _window;
-  *out_display = display;
-  return 0;
-}
-
-int ViEAutoTestWindowManager::ViEDestroyWindow(Window *window,
-                                               Display *display) {
-  XUnmapWindow(display, *window);
-  XDestroyWindow(display, *window);
-  XSync(display, false);
-  XCloseDisplay(display);
-  return 0;
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow() {
-  return 0;
-}
-
-int main(int argc, char** argv) {
-  ViEAutoTestMain auto_test;
-  return auto_test.RunTests(argc, argv);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc
deleted file mode 100644
index ffc9757984..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc
+++ /dev/null
@@ -1,761 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_loopback.cc
-//
-// This code is also used as sample code for ViE 3.0
-//
-
-// ===================================================================
-//
-// BEGIN: VideoEngine 3.0 Sample Code
-//
-
-#include <iostream>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_external_codec.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-
-const uint32_t kSsrc = 0x01234567;
-const uint32_t kRtxSsrc = 0x01234568;
-const int kRtxPayloadType = 98;
-#define VCM_RED_PAYLOAD_TYPE        96
-#define VCM_ULPFEC_PAYLOAD_TYPE     97
-
-int VideoEngineSampleCode(void* window1, void* window2)
-{
-    //********************************************************
-    //  Begin create/initialize Video Engine for testing
-    //********************************************************
-
-    int error = 0;
-
-    //
-    // Create a VideoEngine instance
-    //
-    webrtc::VideoEngine* ptrViE = NULL;
-    ptrViE = webrtc::VideoEngine::Create();
-    if (ptrViE == NULL)
-    {
-        printf("ERROR in VideoEngine::Create\n");
-        return -1;
-    }
-
-    error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
-    if (error == -1)
-    {
-        printf("ERROR in VideoEngine::SetTraceFilter\n");
-        return -1;
-    }
-
-    std::string trace_file =
-        ViETest::GetResultOutputPath() + "ViELoopbackCall_trace.txt";
-    error = ptrViE->SetTraceFile(trace_file.c_str());
-    if (error == -1)
-    {
-        printf("ERROR in VideoEngine::SetTraceFile\n");
-        return -1;
-    }
-
-    //
-    // Init VideoEngine and create a channel
-    //
-    webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
-    if (ptrViEBase == NULL)
-    {
-        printf("ERROR in ViEBase::GetInterface\n");
-        return -1;
-    }
-
-    error = ptrViEBase->Init();
-    if (error == -1)
-    {
-        printf("ERROR in ViEBase::Init\n");
-        return -1;
-    }
-
-    webrtc::ViERTP_RTCP* ptrViERtpRtcp =
-        webrtc::ViERTP_RTCP::GetInterface(ptrViE);
-    if (ptrViERtpRtcp == NULL)
-    {
-        printf("ERROR in ViERTP_RTCP::GetInterface\n");
-        return -1;
-    }
-
-    int videoChannel = -1;
-    error = ptrViEBase->CreateChannel(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViEBase::CreateChannel\n");
-        return -1;
-    }
-
-    //
-    // List available capture devices, allocate and connect.
-    //
-    webrtc::ViECapture* ptrViECapture =
-        webrtc::ViECapture::GetInterface(ptrViE);
-    if (ptrViEBase == NULL)
-    {
-        printf("ERROR in ViECapture::GetInterface\n");
-        return -1;
-    }
-
-    const unsigned int KMaxDeviceNameLength = 128;
-    const unsigned int KMaxUniqueIdLength = 256;
-    char deviceName[KMaxDeviceNameLength];
-    memset(deviceName, 0, KMaxDeviceNameLength);
-    char uniqueId[KMaxUniqueIdLength];
-    memset(uniqueId, 0, KMaxUniqueIdLength);
-
-    printf("Available capture devices:\n");
-    int captureIdx = 0;
-    for (captureIdx = 0;
-         captureIdx < ptrViECapture->NumberOfCaptureDevices();
-         captureIdx++)
-    {
-        memset(deviceName, 0, KMaxDeviceNameLength);
-        memset(uniqueId, 0, KMaxUniqueIdLength);
-
-        error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
-                                                KMaxDeviceNameLength, uniqueId,
-                                                KMaxUniqueIdLength);
-        if (error == -1)
-        {
-            printf("ERROR in ViECapture::GetCaptureDevice\n");
-            return -1;
-        }
-        printf("\t %d. %s\n", captureIdx + 1, deviceName);
-    }
-    printf("\nChoose capture device: ");
-#ifdef WEBRTC_ANDROID
-    captureIdx = 0;
-    printf("0\n");
-#else
-    if (scanf("%d", &captureIdx) != 1)
-    {
-        printf("Error in scanf()\n");
-        return -1;
-    }
-    getc(stdin);
-    captureIdx = captureIdx - 1; // Compensate for idx start at 1.
-#endif
-    error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
-                                            KMaxDeviceNameLength, uniqueId,
-                                            KMaxUniqueIdLength);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::GetCaptureDevice\n");
-        return -1;
-    }
-
-    int captureId = 0;
-    error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
-                                                 captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::AllocateCaptureDevice\n");
-        return -1;
-    }
-
-    error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::ConnectCaptureDevice\n");
-        return -1;
-    }
-
-    error = ptrViECapture->StartCapture(captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::StartCapture\n");
-        return -1;
-    }
-
-    //
-    // RTP/RTCP settings
-    //
-
-    error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
-                                         webrtc::kRtcpCompound_RFC4585);
-    if (error == -1)
-    {
-        printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
-        return -1;
-    }
-
-    error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
-        videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
-    if (error == -1)
-    {
-        printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
-        return -1;
-    }
-
-    error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
-    if (error == -1)
-    {
-        printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
-        return -1;
-    }
-
-    // Setting SSRC manually (arbitrary value), as otherwise we will get a clash
-    // (loopback), and a new SSRC will be set, which will reset the receiver.
-    error = ptrViERtpRtcp->SetLocalSSRC(videoChannel, kSsrc);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetLocalSSRC\n");
-      return -1;
-    }
-
-    error = ptrViERtpRtcp->SetLocalSSRC(videoChannel, kRtxSsrc,
-                                        webrtc::kViEStreamTypeRtx, 0);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetLocalSSRC\n");
-      return -1;
-    }
-
-    error = ptrViERtpRtcp->SetRemoteSSRCType(videoChannel,
-                                             webrtc::kViEStreamTypeRtx,
-                                             kRtxSsrc);
-
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetRtxReceivePayloadType\n");
-      return -1;
-    }
-
-    error = ptrViERtpRtcp->SetRtxSendPayloadType(videoChannel, kRtxPayloadType);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetRtxSendPayloadType\n");
-      return -1;
-    }
-
-    error = ptrViERtpRtcp->SetRtxReceivePayloadType(videoChannel,
-                                                    kRtxPayloadType);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetRtxReceivePayloadType\n");
-      return -1;
-    }
-    //
-    // Set up rendering
-    //
-    webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
-    if (ptrViERender == NULL) {
-      printf("ERROR in ViERender::GetInterface\n");
-      return -1;
-    }
-
-    error
-        = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::AddRenderer\n");
-        return -1;
-    }
-
-    error = ptrViERender->StartRender(captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::StartRender\n");
-        return -1;
-    }
-
-    error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
-                                      1.0);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::AddRenderer\n");
-        return -1;
-    }
-
-    error = ptrViERender->StartRender(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::StartRender\n");
-        return -1;
-    }
-
-    //
-    // Setup codecs
-    //
-    webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
-    if (ptrViECodec == NULL)
-    {
-        printf("ERROR in ViECodec::GetInterface\n");
-        return -1;
-    }
-
-    // Check available codecs and prepare receive codecs
-    printf("\nAvailable codecs:\n");
-    webrtc::VideoCodec videoCodec;
-    memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
-    int codecIdx = 0;
-    for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++)
-    {
-        error = ptrViECodec->GetCodec(codecIdx, videoCodec);
-        if (error == -1)
-        {
-            printf("ERROR in ViECodec::GetCodec\n");
-            return -1;
-        }
-
-        // try to keep the test frame size small when I420
-        if (videoCodec.codecType == webrtc::kVideoCodecI420)
-        {
-            videoCodec.width = 176;
-            videoCodec.height = 144;
-        }
-
-        error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
-        if (error == -1)
-        {
-            printf("ERROR in ViECodec::SetReceiveCodec\n");
-            return -1;
-        }
-        if (videoCodec.codecType != webrtc::kVideoCodecRED
-            && videoCodec.codecType != webrtc::kVideoCodecULPFEC)
-        {
-            printf("\t %d. %s\n", codecIdx + 1, videoCodec.plName);
-        }
-    }
-    printf("%d. VP8 over Generic.\n", ptrViECodec->NumberOfCodecs() + 1);
-
-    printf("Choose codec: ");
-#ifdef WEBRTC_ANDROID
-    codecIdx = 0;
-    printf("0\n");
-#else
-    if (scanf("%d", &codecIdx) != 1)
-    {
-        printf("Error in scanf()\n");
-        return -1;
-    }
-    getc(stdin);
-    codecIdx = codecIdx - 1; // Compensate for idx start at 1.
-#endif
-    // VP8 over generic transport gets this special one.
-    if (codecIdx == ptrViECodec->NumberOfCodecs()) {
-      for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); ++codecIdx) {
-        error = ptrViECodec->GetCodec(codecIdx, videoCodec);
-        assert(error != -1);
-        if (videoCodec.codecType == webrtc::kVideoCodecVP8)
-          break;
-      }
-      assert(videoCodec.codecType == webrtc::kVideoCodecVP8);
-      videoCodec.codecType = webrtc::kVideoCodecGeneric;
-
-      // Any plName should work with generic
-      strcpy(videoCodec.plName, "VP8-GENERIC");
-      uint8_t pl_type = 127;
-      videoCodec.plType = pl_type;
-      webrtc::ViEExternalCodec* external_codec = webrtc::ViEExternalCodec
-          ::GetInterface(ptrViE);
-      assert(external_codec != NULL);
-      error = external_codec->RegisterExternalSendCodec(videoChannel, pl_type,
-          webrtc::VP8Encoder::Create(), false);
-      assert(error != -1);
-      error = external_codec->RegisterExternalReceiveCodec(videoChannel,
-          pl_type, webrtc::VP8Decoder::Create(), false);
-      assert(error != -1);
-    } else {
-      error = ptrViECodec->GetCodec(codecIdx, videoCodec);
-      if (error == -1) {
-        printf("ERROR in ViECodec::GetCodec\n");
-        return -1;
-      }
-    }
-
-    // Set spatial resolution option
-    std::string str;
-    std::cout << std::endl;
-    std::cout << "Enter frame size option (default is CIF):" << std::endl;
-    std::cout << "1. QCIF (176X144) " << std::endl;
-    std::cout << "2. CIF  (352X288) " << std::endl;
-    std::cout << "3. VGA  (640X480) " << std::endl;
-    std::cout << "4. 4CIF (704X576) " << std::endl;
-    std::cout << "5. WHD  (1280X720) " << std::endl;
-    std::cout << "6. FHD  (1920X1080) " << std::endl;
-    std::getline(std::cin, str);
-    int resolnOption = atoi(str.c_str());
-    switch (resolnOption)
-    {
-        case 1:
-            videoCodec.width = 176;
-            videoCodec.height = 144;
-            break;
-        case 2:
-            videoCodec.width = 352;
-            videoCodec.height = 288;
-            break;
-        case 3:
-            videoCodec.width = 640;
-            videoCodec.height = 480;
-            break;
-        case 4:
-            videoCodec.width = 704;
-            videoCodec.height = 576;
-            break;
-        case 5:
-            videoCodec.width = 1280;
-            videoCodec.height = 720;
-            break;
-        case 6:
-            videoCodec.width = 1920;
-            videoCodec.height = 1080;
-            break;
-    }
-
-    // Set number of temporal layers.
-    std::cout << std::endl;
-    std::cout << "Choose number of temporal layers for VP8 (1 to 4). ";
-    std::cout << "Press enter for default (=1) for other codecs: \n";
-    std::getline(std::cin, str);
-    int numTemporalLayers = atoi(str.c_str());
-    if (numTemporalLayers != 0 &&
-        videoCodec.codecType == webrtc::kVideoCodecVP8) {
-      videoCodec.codecSpecific.VP8.numberOfTemporalLayers = numTemporalLayers;
-    } else if (videoCodec.codecType == webrtc::kVideoCodecVP9) {
-      // Temporal layers for vp9 not yet supported in webrtc.
-      numTemporalLayers = 1;
-      videoCodec.codecSpecific.VP9.numberOfTemporalLayers = 1;
-    }
-
-    // Set start bit rate
-    std::cout << std::endl;
-    std::cout << "Choose start rate (in kbps). Press enter for default:  ";
-    std::getline(std::cin, str);
-    int startRate = atoi(str.c_str());
-    if(startRate != 0)
-    {
-        videoCodec.startBitrate=startRate;
-    }
-
-    error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
-    assert(error != -1);
-    error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
-    assert(error != -1);
-
-    //
-    // Choose Protection Mode
-    //
-    std::cout << std::endl;
-    std::cout << "Enter Protection Method:" << std::endl;
-    std::cout << "0. None" << std::endl;
-    std::cout << "1. FEC" << std::endl;
-    std::cout << "2. NACK" << std::endl;
-    std::cout << "3. NACK+FEC" << std::endl;
-    std::getline(std::cin, str);
-    int protectionMethod = atoi(str.c_str());
-    error = 0;
-    bool temporalToggling = true;
-    switch (protectionMethod)
-    {
-        case 0: // None: default is no protection
-            break;
-
-        case 1: // FEC only
-            error = ptrViERtpRtcp->SetFECStatus(videoChannel,
-                                                true,
-                                                VCM_RED_PAYLOAD_TYPE,
-                                                VCM_ULPFEC_PAYLOAD_TYPE);
-            temporalToggling = false;
-            break;
-
-        case 2: // Nack only
-            error = ptrViERtpRtcp->SetNACKStatus(videoChannel, true);
-
-            break;
-
-        case 3: // Hybrid NAck and FEC
-            error = ptrViERtpRtcp->SetHybridNACKFECStatus(
-                videoChannel,
-                true,
-                VCM_RED_PAYLOAD_TYPE,
-                VCM_ULPFEC_PAYLOAD_TYPE);
-            temporalToggling = false;
-            break;
-     }
-
-    if (error < 0)
-    {
-        printf("ERROR in ViERTP_RTCP::SetProtectionStatus\n");
-    }
-
-    // Set up buffering delay.
-    std::cout << std::endl;
-    std::cout << "Set buffering delay (mS). Press enter for default(0mS):  ";
-    std::getline(std::cin, str);
-    int buffering_delay = atoi(str.c_str());
-    if (buffering_delay != 0) {
-      error = ptrViERtpRtcp->SetSenderBufferingMode(videoChannel,
-                                                    buffering_delay);
-      if (error < 0)
-        printf("ERROR in ViERTP_RTCP::SetSenderBufferingMode\n");
-
-      error = ptrViERtpRtcp->SetReceiverBufferingMode(videoChannel,
-                                                      buffering_delay);
-      if (error < 0)
-        printf("ERROR in ViERTP_RTCP::SetReceiverBufferingMode\n");
-    }
-
-    //
-    // Address settings
-    //
-    webrtc::ViENetwork* ptrViENetwork =
-        webrtc::ViENetwork::GetInterface(ptrViE);
-    if (ptrViENetwork == NULL)
-    {
-        printf("ERROR in ViENetwork::GetInterface\n");
-        return -1;
-    }
-
-    // Setup transport.
-    TbExternalTransport* extTransport = NULL;
-    webrtc::test::VideoChannelTransport* video_channel_transport = NULL;
-
-    int testMode = 0;
-    std::cout << std::endl;
-    std::cout << "Enter 1 for testing packet loss and delay with "
-        "external transport: ";
-    std::string test_str;
-    std::getline(std::cin, test_str);
-    testMode = atoi(test_str.c_str());
-    if (testMode == 1)
-    {
-        // Avoid changing SSRC due to collision.
-        error = ptrViERtpRtcp->SetLocalSSRC(videoChannel, 1);
-
-        extTransport = new TbExternalTransport(*ptrViENetwork, videoChannel,
-                                               NULL);
-
-        error = ptrViENetwork->RegisterSendTransport(videoChannel,
-                                                     *extTransport);
-        if (error == -1)
-        {
-            printf("ERROR in ViECodec::RegisterSendTransport \n");
-            return -1;
-        }
-
-        // Setting uniform loss. Actual values will be set by user.
-        NetworkParameters network;
-        network.loss_model = kUniformLoss;
-        // Set up packet loss value
-        std::cout << "Enter Packet Loss Percentage" << std::endl;
-        std::string rate_str;
-        std::getline(std::cin, rate_str);
-        network.packet_loss_rate = atoi(rate_str.c_str());
-        if (network.packet_loss_rate > 0) {
-          temporalToggling = false;
-        }
-
-        // Set network delay value
-        std::cout << "Enter network delay value [mS]" << std::endl;
-        std::string delay_str;
-        std::getline(std::cin, delay_str);
-        network.mean_one_way_delay = atoi(delay_str.c_str());
-        extTransport->SetNetworkParameters(network);
-        if (numTemporalLayers > 1 && temporalToggling) {
-          extTransport->SetTemporalToggle(numTemporalLayers);
-        } else {
-          // Disabled
-          extTransport->SetTemporalToggle(0);
-        }
-    }
-    else
-    {
-        video_channel_transport = new webrtc::test::VideoChannelTransport(
-            ptrViENetwork, videoChannel);
-
-        const char* ipAddress = "127.0.0.1";
-        const unsigned short rtpPort = 6000;
-        std::cout << std::endl;
-        std::cout << "Using rtp port: " << rtpPort << std::endl;
-        std::cout << std::endl;
-
-        error = video_channel_transport->SetLocalReceiver(rtpPort);
-        if (error == -1)
-        {
-            printf("ERROR in SetLocalReceiver\n");
-            return -1;
-        }
-        error = video_channel_transport->SetSendDestination(ipAddress, rtpPort);
-        if (error == -1)
-        {
-            printf("ERROR in SetSendDestination\n");
-            return -1;
-        }
-    }
-
-    error = ptrViEBase->StartReceive(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViENetwork::StartReceive\n");
-        return -1;
-    }
-
-    error = ptrViEBase->StartSend(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViENetwork::StartSend\n");
-        return -1;
-    }
-
-    //********************************************************
-    //  Engine started
-    //********************************************************
-
-
-    // Call started
-    printf("\nLoopback call started\n\n");
-    printf("Press enter to stop...");
-    while ((getc(stdin)) != '\n')
-        ;
-
-    //********************************************************
-    //  Testing finished. Tear down Video Engine
-    //********************************************************
-
-    error = ptrViEBase->StopReceive(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViEBase::StopReceive\n");
-        return -1;
-    }
-
-    error = ptrViEBase->StopSend(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViEBase::StopSend\n");
-        return -1;
-    }
-
-    error = ptrViERender->StopRender(captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::StopRender\n");
-        return -1;
-    }
-
-    error = ptrViERender->RemoveRenderer(captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::RemoveRenderer\n");
-        return -1;
-    }
-
-    error = ptrViERender->StopRender(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::StopRender\n");
-        return -1;
-    }
-
-    error = ptrViERender->RemoveRenderer(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViERender::RemoveRenderer\n");
-        return -1;
-    }
-
-    error = ptrViECapture->StopCapture(captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::StopCapture\n");
-        return -1;
-    }
-
-    error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
-        return -1;
-    }
-
-    error = ptrViECapture->ReleaseCaptureDevice(captureId);
-    if (error == -1)
-    {
-        printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
-        return -1;
-    }
-
-    error = ptrViEBase->DeleteChannel(videoChannel);
-    if (error == -1)
-    {
-        printf("ERROR in ViEBase::DeleteChannel\n");
-        return -1;
-    }
-
-    delete video_channel_transport;
-    delete extTransport;
-
-    int remainingInterfaces = 0;
-    remainingInterfaces = ptrViECodec->Release();
-    remainingInterfaces += ptrViECapture->Release();
-    remainingInterfaces += ptrViERtpRtcp->Release();
-    remainingInterfaces += ptrViERender->Release();
-    remainingInterfaces += ptrViENetwork->Release();
-    remainingInterfaces += ptrViEBase->Release();
-    if (remainingInterfaces > 0)
-    {
-        printf("ERROR: Could not release all interfaces\n");
-        return -1;
-    }
-
-    bool deleted = webrtc::VideoEngine::Delete(ptrViE);
-    if (deleted == false)
-    {
-        printf("ERROR in VideoEngine::Delete\n");
-        return -1;
-    }
-
-    return 0;
-
-    //
-    // END:  VideoEngine 3.0 Sample Code
-    //
-    // ===================================================================
-}
-
-int ViEAutoTest::ViELoopbackCall()
-{
-    ViETest::Log(" ");
-    ViETest::Log("========================================");
-    ViETest::Log(" ViE Autotest Loopback Call\n");
-
-    if (VideoEngineSampleCode(_window1, _window2) == 0)
-    {
-        ViETest::Log(" ");
-        ViETest::Log(" ViE Autotest Loopback Call Done");
-        ViETest::Log("========================================");
-        ViETest::Log(" ");
-
-        return 0;
-    }
-
-    ViETest::Log(" ");
-    ViETest::Log(" ViE Autotest Loopback Call Failed");
-    ViETest::Log("========================================");
-    ViETest::Log(" ");
-    return 1;
-
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_main.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_main.cc
deleted file mode 100644
index 1617258547..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_main.cc
+++ /dev/null
@@ -1,212 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/test/field_trial.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_creator.h"
-
-DEFINE_bool(automated, false, "Run Video engine tests in noninteractive mode.");
-DEFINE_bool(auto_custom_call, false, "Run custom call directly.");
-DEFINE_string(force_fieldtrials, "",
-    "Field trials control experimental feature code which can be forced. "
-    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
-    " will assign the group Enable to field trial WebRTC-FooFeature.");
-
-static const std::string kStandardTest = "ViEStandardIntegrationTest";
-static const std::string kExtendedTest = "ViEExtendedIntegrationTest";
-static const std::string kApiTest = "ViEApiIntegrationTest";
-
-ViEAutoTestMain::ViEAutoTestMain() {
-  index_to_test_method_map_[1] = "RunsBaseTestWithoutErrors";
-  index_to_test_method_map_[2] = "RunsCaptureTestWithoutErrors";
-  index_to_test_method_map_[3] = "RunsCodecTestWithoutErrors";
-  index_to_test_method_map_[4] = "[unused]";
-  index_to_test_method_map_[5] = "RunsImageProcessTestWithoutErrors";
-  index_to_test_method_map_[6] = "RunsNetworkTestWithoutErrors";
-  index_to_test_method_map_[7] = "RunsRenderTestWithoutErrors";
-  index_to_test_method_map_[8] = "RunsRtpRtcpTestWithoutErrors";
-}
-
-int ViEAutoTestMain::RunTests(int argc, char** argv) {
-  // Initialize logging.
-  ViETest::Init();
-  // Initialize WebRTC testing framework so paths to resources can be resolved.
-  webrtc::test::SetExecutablePath(argv[0]);
-  // Initialize the testing framework.
-  testing::InitGoogleTest(&argc, argv);
-  // AllowCommandLineParsing allows us to ignore flags passed on to us by
-  // Chromium build bots without having to explicitly disable them.
-  google::AllowCommandLineReparsing();
-  // Parse remaining flags:
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  // Initialize field trial
-  webrtc::test::InitFieldTrialsFromString(FLAGS_force_fieldtrials);
-
-  int result;
-  if (FLAGS_automated) {
-    // Run in automated mode.
-#if defined(WEBRTC_LINUX)
-    // All window-related tests are disabled on Linux for now.
-    // See https://code.google.com/p/chromium/issues/detail?id=318760
-    return 0;
-#endif
-    result = RUN_ALL_TESTS();
-  } else if (FLAGS_auto_custom_call) {
-    // Run automated custom call.
-    result = RunSpecialTestCase(8);
-  } else {
-    // Run in interactive mode.
-    result = RunInteractiveMode();
-  }
-
-  ViETest::Terminate();
-  return result;
-}
-
-int ViEAutoTestMain::AskUserForTestCase() {
-  int choice;
-  std::string answer;
-
-  do {
-    ViETest::Log("\nSpecific tests:");
-    ViETest::Log("\t 0. Go back to previous menu.");
-
-    // Print all test method choices. Assumes that map sorts on its key.
-    int last_valid_choice = 0;
-    std::map<int, std::string>::const_iterator iterator;
-    for (iterator = index_to_test_method_map_.begin();
-        iterator != index_to_test_method_map_.end();
-        ++iterator) {
-      ViETest::Log("\t %d. %s", iterator->first, iterator->second.c_str());
-      last_valid_choice = iterator->first;
-    }
-
-    ViETest::Log("Choose specific test:");
-    choice = AskUserForNumber(0, last_valid_choice);
-  } while (choice == kInvalidChoice);
-
-  return choice;
-}
-
-int ViEAutoTestMain::AskUserForNumber(int min_allowed, int max_allowed) {
-  int result;
-  if (scanf("%d", &result) <= 0) {
-    ViETest::Log("\nPlease enter a number instead, then hit enter.");
-    getc(stdin);
-    return kInvalidChoice;
-  }
-  getc(stdin);  // Consume enter key.
-
-  if (result < min_allowed || result > max_allowed) {
-    ViETest::Log("%d-%d are valid choices. Please try again.", min_allowed,
-                 max_allowed);
-    return kInvalidChoice;
-  }
-
-  return result;
-}
-
-int ViEAutoTestMain::RunTestMatching(const std::string test_case,
-                                     const std::string test_method) {
-  testing::FLAGS_gtest_filter = test_case + "." + test_method;
-  return RUN_ALL_TESTS();
-}
-
-int ViEAutoTestMain::RunSpecificTestCaseIn(const std::string test_case_name)
-{
-  // If user says 0, it means don't run anything.
-  int specific_choice = AskUserForTestCase();
-  if (specific_choice != 0){
-    return RunTestMatching(test_case_name,
-                           index_to_test_method_map_[specific_choice]);
-  }
-  return 0;
-}
-
-int ViEAutoTestMain::RunSpecialTestCase(int choice) {
-  // 7-10 don't run in GTest and need to initialize by themselves.
-  assert(choice >= 7 && choice <= 10);
-
-  // Create the windows
-  ViEWindowCreator windowCreator;
-  ViEAutoTestWindowManagerInterface* windowManager =
-      windowCreator.CreateTwoWindows();
-
-  // Create the test cases
-  ViEAutoTest vieAutoTest(windowManager->GetWindow1(),
-                          windowManager->GetWindow2());
-
-  int errors = 0;
-  switch (choice) {
-    case 7: errors = vieAutoTest.ViELoopbackCall();  break;
-    case 8: errors = vieAutoTest.ViECustomCall();    break;
-    case 9: errors = vieAutoTest.ViESimulcastCall(); break;
-    case 10: errors = vieAutoTest.ViERecordCall();   break;
-  }
-
-  windowCreator.TerminateWindows();
-  return errors;
-}
-
-int ViEAutoTestMain::RunInteractiveMode() {
-  ViETest::Log(" ============================== ");
-  ViETest::Log("    WebRTC ViE 3.x Autotest     ");
-  ViETest::Log(" ============================== \n");
-
-  int choice = 0;
-  int errors = 0;
-  do {
-    ViETest::Log("Test types: ");
-    ViETest::Log("\t 0. Quit");
-    ViETest::Log("\t 1. All standard tests (delivery test)");
-    ViETest::Log("\t 2. All API tests");
-    ViETest::Log("\t 3. All extended test");
-    ViETest::Log("\t 4. Specific standard test");
-    ViETest::Log("\t 5. Specific API test");
-    ViETest::Log("\t 6. Specific extended test");
-    ViETest::Log("\t 7. Simple loopback call");
-    ViETest::Log("\t 8. Custom configure a call");
-    ViETest::Log("\t 9. Simulcast in loopback");
-    ViETest::Log("\t 10. Record");
-    ViETest::Log("Select type of test:");
-
-    choice = AskUserForNumber(0, 10);
-    if (choice == kInvalidChoice) {
-      continue;
-    }
-    switch (choice) {
-      case 0:                                                 break;
-      case 1:  errors = RunTestMatching(kStandardTest, "*");  break;
-      case 2:  errors = RunTestMatching(kApiTest,      "*");  break;
-      case 3:  errors = RunTestMatching(kExtendedTest, "*");  break;
-      case 4:  errors = RunSpecificTestCaseIn(kStandardTest); break;
-      case 5:  errors = RunSpecificTestCaseIn(kApiTest);      break;
-      case 6:  errors = RunSpecificTestCaseIn(kExtendedTest); break;
-      default: errors = RunSpecialTestCase(choice);           break;
-    }
-  } while (choice != 0);
-
-  if (errors) {
-    ViETest::Log("Test done with errors, see ViEAutotestLog.txt for test "
-        "result.\n");
-    return 1;
-  } else {
-    ViETest::Log("Test done without errors, see ViEAutotestLog.txt for "
-        "test result.\n");
-    return 0;
-  }
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_network.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_network.cc
deleted file mode 100644
index 1ccac70168..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_network.cc
+++ /dev/null
@@ -1,535 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_network.cc
-//
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-
-#if defined(_WIN32)
-#include <qos.h>
-#endif
-
-void ViEAutoTest::ViENetworkStandardTest()
-{
-    TbInterfaces ViE("ViENetworkStandardTest"); // Create VIE
-    TbCaptureDevice tbCapture(ViE);
-    {
-        // Create a video channel
-        TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-        tbCapture.ConnectTo(tbChannel.videoChannel);
-
-        RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
-        // ***************************************************************
-        // Engine ready. Begin testing class
-        // ***************************************************************
-
-        //
-        // Transport
-        //
-        TbExternalTransport testTransport(*ViE.network, tbChannel.videoChannel,
-                                          NULL);
-        EXPECT_EQ(0, ViE.network->RegisterSendTransport(
-            tbChannel.videoChannel, testTransport));
-        EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
-            tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
-
-        ViETest::Log("Call started using external transport, video should "
-            "see video in both windows\n");
-        AutoTestSleep(kAutoTestSleepTimeMs);
-
-        EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
-            tbChannel.videoChannel));
-
-        char myIpAddress[64];
-        memset(myIpAddress, 0, 64);
-        unsigned short rtpPort = 1234;
-        memcpy(myIpAddress, "127.0.0.1", sizeof("127.0.0.1"));
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
-        EXPECT_EQ(0, ViE.network->SetSendDestination(
-            tbChannel.videoChannel, myIpAddress, rtpPort,
-            rtpPort + 1, rtpPort));
-        EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-        ViETest::Log("Changed to WebRTC SocketTransport, you should still see "
-                     "video in both windows\n");
-        AutoTestSleep(kAutoTestSleepTimeMs);
-
-        EXPECT_EQ(0, ViE.network->SetSourceFilter(
-            tbChannel.videoChannel, rtpPort + 10, rtpPort + 11, myIpAddress));
-        ViETest::Log("Added UDP port filter for incorrect ports, you should "
-                     "not see video in Window2");
-        AutoTestSleep(2000);
-        EXPECT_EQ(0, ViE.network->SetSourceFilter(
-            tbChannel.videoChannel, rtpPort, rtpPort + 1, "123.1.1.0"));
-        ViETest::Log("Added IP filter for incorrect IP address, you should not "
-                     "see video in Window2");
-        AutoTestSleep(2000);
-        EXPECT_EQ(0, ViE.network->SetSourceFilter(
-            tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
-        ViETest::Log("Added IP filter for this computer, you should see video "
-                     "in Window2 again\n");
-        AutoTestSleep(kAutoTestSleepTimeMs);
-
-        tbCapture.Disconnect(tbChannel.videoChannel);
-    }
-}
-
-void ViEAutoTest::ViENetworkExtendedTest()
-{
-    //***************************************************************
-    //	Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-
-    TbInterfaces ViE("ViENetworkExtendedTest"); // Create VIE
-    TbCaptureDevice tbCapture(ViE);
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-
-    {
-        //
-        // ToS
-        //
-        // Create a video channel
-        TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-        tbCapture.ConnectTo(tbChannel.videoChannel);
-        const char* remoteIp = "192.168.200.1";
-        int DSCP = 0;
-        bool useSetSockOpt = false;
-
-        webrtc::VideoCodec videoCodec;
-        EXPECT_EQ(0, ViE.codec->GetSendCodec(
-            tbChannel.videoChannel, videoCodec));
-        videoCodec.maxFramerate = 5;
-        EXPECT_EQ(0, ViE.codec->SetSendCodec(
-            tbChannel.videoChannel, videoCodec));
-
-        //***************************************************************
-        //	Engine ready. Begin testing class
-        //***************************************************************
-
-        char myIpAddress[64];
-        memset(myIpAddress, 0, 64);
-        unsigned short rtpPort = 9000;
-        EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
-        EXPECT_EQ(0, ViE.network->SetSendDestination(
-            tbChannel.videoChannel, remoteIp, rtpPort, rtpPort + 1, rtpPort));
-
-        // ToS
-        int tos_result = ViE.network->SetSendToS(tbChannel.videoChannel, 2);
-        EXPECT_EQ(0, tos_result);
-        if (tos_result != 0)
-        {
-            ViETest::Log("ViESetSendToS error!.");
-            ViETest::Log("You must be admin to run these tests.");
-            ViETest::Log("On Win7 and late Vista, you need to right click the "
-                         "exe and choose");
-            ViETest::Log("\"Run as administrator\"\n");
-            getc(stdin);
-        }
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));  // No ToS set
-
-        EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-        ViETest::Log("Use Wireshark to capture the outgoing video stream and "
-                     "verify ToS settings\n");
-        ViETest::Log(" DSCP set to 0x%x\n", DSCP);
-        AutoTestSleep(1000);
-
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63));
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));  // No ToS set
-        ViETest::Log(" DSCP set to 0x%x\n", DSCP);
-        AutoTestSleep(1000);
-
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0));
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 2, true));
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));  // No ToS set
-        ViETest::Log(" DSCP set to 0x%x\n", DSCP);
-        AutoTestSleep(1000);
-
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63, true));
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));  // No ToS set
-        ViETest::Log(" DSCP set to 0x%x\n", DSCP);
-        AutoTestSleep(1000);
-
-        tbCapture.Disconnect(tbChannel.videoChannel);
-    }
-
-    //***************************************************************
-    //	Testing finished. Tear down Video Engine
-    //***************************************************************
-}
-
-void ViEAutoTest::ViENetworkAPITest()
-{
-    //***************************************************************
-    //	Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-
-    TbInterfaces ViE("ViENetworkAPITest"); // Create VIE
-    {
-        // Create a video channel
-        TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecI420);
-
-        //***************************************************************
-        //	Engine ready. Begin testing class
-        //***************************************************************
-
-        //
-        // External transport
-        //
-        TbExternalTransport testTransport(*ViE.network, tbChannel.videoChannel,
-                                          NULL);
-        EXPECT_EQ(0, ViE.network->RegisterSendTransport(
-            tbChannel.videoChannel, testTransport));
-        EXPECT_NE(0, ViE.network->RegisterSendTransport(
-            tbChannel.videoChannel, testTransport));
-
-        // Create a empty RTP packet.
-        unsigned char packet[3000];
-        memset(packet, 0, sizeof(packet));
-        packet[0] = 0x80; // V=2, P=0, X=0, CC=0
-        packet[1] = 0x7C; // M=0, PT = 124 (I420)
-
-        // Create a empty RTCP app packet.
-        unsigned char rtcpacket[3000];
-        memset(rtcpacket,0, sizeof(rtcpacket));
-        rtcpacket[0] = 0x80; // V=2, P=0, X=0, CC=0
-        rtcpacket[1] = 0xCC; // M=0, PT = 204 (RTCP app)
-        rtcpacket[2] = 0x0;
-        rtcpacket[3] = 0x03; // 3 Octets long.
-
-        EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
-            tbChannel.videoChannel, packet, 1500));
-        EXPECT_NE(0, ViE.network->ReceivedRTCPPacket(
-            tbChannel.videoChannel, rtcpacket, 1500));
-        EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
-            tbChannel.videoChannel, packet, 1500));
-        EXPECT_EQ(0, ViE.network->ReceivedRTCPPacket(
-            tbChannel.videoChannel, rtcpacket, 1500));
-        EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
-            tbChannel.videoChannel, packet, 11));
-        EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
-            tbChannel.videoChannel, packet, 11));
-        EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
-            tbChannel.videoChannel, packet, 3000));
-        EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
-            tbChannel.videoChannel, packet, 3000));
-        EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-        EXPECT_NE(0, ViE.network->DeregisterSendTransport(
-            tbChannel.videoChannel));  // Sending
-        EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
-            tbChannel.videoChannel));
-        EXPECT_NE(0, ViE.network->DeregisterSendTransport(
-            tbChannel.videoChannel));  // Already deregistered
-
-        //
-        // Local receiver
-        //
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
-
-        unsigned short rtpPort = 0;
-        unsigned short rtcpPort = 0;
-        char ipAddress[64];
-        memset(ipAddress, 0, 64);
-        EXPECT_EQ(0, ViE.network->GetLocalReceiver(
-            tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
-        EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-        EXPECT_NE(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
-        EXPECT_EQ(0, ViE.network->GetLocalReceiver(
-            tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
-        EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-        //
-        // Send destination
-        //
-        EXPECT_EQ(0, ViE.network->SetSendDestination(
-            tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
-        EXPECT_EQ(0, ViE.network->SetSendDestination(
-            tbChannel.videoChannel, "127.0.0.1", 1236, 1237, 1234, 1235));
-
-        unsigned short sourceRtpPort = 0;
-        unsigned short sourceRtcpPort = 0;
-        EXPECT_EQ(0, ViE.network->GetSendDestination(
-            tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
-            sourceRtpPort, sourceRtcpPort));
-        EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-        // Not allowed while sending
-        EXPECT_NE(0, ViE.network->SetSendDestination(
-            tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
-        EXPECT_EQ(kViENetworkAlreadySending, ViE.base->LastError());
-
-        EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.network->SetSendDestination(
-            tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
-        EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-        EXPECT_EQ(0, ViE.network->GetSendDestination(
-            tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
-            sourceRtpPort, sourceRtcpPort));
-        EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-        //
-        // Address information
-        //
-
-        // GetSourceInfo: Tested in functional test
-        EXPECT_EQ(0, ViE.network->GetLocalIP(ipAddress, false));
-
-        // TODO(unknown): IPv6
-
-        //
-        // Filter
-        //
-        EXPECT_NE(0, ViE.network->GetSourceFilter(
-            tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
-        EXPECT_EQ(0, ViE.network->SetSourceFilter(
-            tbChannel.videoChannel, 1234, 1235, "10.10.10.10"));
-        EXPECT_EQ(0, ViE.network->SetSourceFilter(
-            tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
-        EXPECT_EQ(0, ViE.network->GetSourceFilter(
-            tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
-        EXPECT_EQ(0, ViE.network->SetSourceFilter(
-            tbChannel.videoChannel, 0, 0, NULL));
-        EXPECT_NE(0, ViE.network->GetSourceFilter(
-            tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
-    }
-    {
-        TbVideoChannel tbChannel(ViE);  // Create a video channel
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, 1234));
-
-        int DSCP = 0;
-        bool useSetSockOpt = false;
-        // SetSockOpt should work without a locally bind socket
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));  // No ToS set
-        EXPECT_EQ(0, DSCP);
-
-        // Invalid input
-        EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, -1, true));
-
-        // Invalid input
-        EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, 64, true));
-
-        // Valid
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 20, true));
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));
-
-        EXPECT_EQ(20, DSCP);
-        EXPECT_TRUE(useSetSockOpt);
-
-        // Disable
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, true));
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));
-        EXPECT_EQ(0, DSCP);
-
-        char myIpAddress[64];
-        memset(myIpAddress, 0, 64);
-        // Get local ip to be able to set ToS withtou setSockOpt
-        EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
-        EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-            tbChannel.videoChannel, 1234, 1235, myIpAddress));
-
-        // Invalid input
-        EXPECT_NE(0, ViE.network->SetSendToS(
-            tbChannel.videoChannel, -1, false));
-        EXPECT_NE(0, ViE.network->SetSendToS(
-            tbChannel.videoChannel, 64, false));  // Invalid input
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));  // No ToS set
-        EXPECT_EQ(0, DSCP);
-        int tos_result = ViE.network->SetSendToS(
-            tbChannel.videoChannel, 20, false);  // Valid
-        EXPECT_EQ(0, tos_result);
-        if (tos_result != 0)
-        {
-            ViETest::Log("ViESetSendToS error!.");
-            ViETest::Log("You must be admin to run these tests.");
-            ViETest::Log("On Win7 and late Vista, you need to right click the "
-                         "exe and choose");
-            ViETest::Log("\"Run as administrator\"\n");
-            getc(stdin);
-        }
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));
-        EXPECT_EQ(20, DSCP);
-#ifdef _WIN32
-        EXPECT_FALSE(useSetSockOpt);
-#else // useSetSockOpt is true on Linux and Mac
-        EXPECT_TRUE(useSetSockOpt);
-#endif
-        EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, false));
-        EXPECT_EQ(0, ViE.network->GetSendToS(
-            tbChannel.videoChannel, DSCP, useSetSockOpt));
-        EXPECT_EQ(0, DSCP);
-    }
-    {
-        // From qos.h. (*) -> supported by ViE
-        //
-        //  #define SERVICETYPE_NOTRAFFIC               0x00000000
-        //  #define SERVICETYPE_BESTEFFORT              0x00000001 (*)
-        //  #define SERVICETYPE_CONTROLLEDLOAD          0x00000002 (*)
-        //  #define SERVICETYPE_GUARANTEED              0x00000003 (*)
-        //  #define SERVICETYPE_NETWORK_UNAVAILABLE     0x00000004
-        //  #define SERVICETYPE_GENERAL_INFORMATION     0x00000005
-        //  #define SERVICETYPE_NOCHANGE                0x00000006
-        //  #define SERVICETYPE_NONCONFORMING           0x00000009
-        //  #define SERVICETYPE_NETWORK_CONTROL         0x0000000A
-        //  #define SERVICETYPE_QUALITATIVE             0x0000000D (*)
-        //
-        //  #define SERVICE_BESTEFFORT                  0x80010000
-        //  #define SERVICE_CONTROLLEDLOAD              0x80020000
-        //  #define SERVICE_GUARANTEED                  0x80040000
-        //  #define SERVICE_QUALITATIVE                 0x80200000
-
-        TbVideoChannel tbChannel(ViE);  // Create a video channel
-
-
-#if defined(_WIN32)
-        // These tests are disabled since they currently fail on Windows.
-        // Exact reason is unkown.
-        // See https://code.google.com/p/webrtc/issues/detail?id=1266.
-        // TODO(mflodman): remove these APIs?
-
-        //// No socket
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
-
-        //EXPECT_EQ(0, ViE.network->SetLocalReceiver(
-        //    tbChannel.videoChannel, 1234));
-
-        //// Sender not initialized
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
-        //EXPECT_EQ(0, ViE.network->SetSendDestination(
-        //    tbChannel.videoChannel, "127.0.0.1", 12345));
-
-        //// Try to set all non-supported service types
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_NETWORK_UNAVAILABLE));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_GENERAL_INFORMATION));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_NOCHANGE));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_NONCONFORMING));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_NETWORK_CONTROL));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICE_BESTEFFORT));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICE_CONTROLLEDLOAD));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICE_GUARANTEED));
-        //EXPECT_NE(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICE_QUALITATIVE));
-
-        //// Loop through valid service settings
-        //bool enabled = false;
-        //int serviceType = 0;
-        //int overrideDSCP = 0;
-
-        //EXPECT_EQ(0, ViE.network->GetSendGQoS(
-        //    tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
-        //EXPECT_FALSE(enabled);
-        //EXPECT_EQ(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
-        //EXPECT_EQ(0, ViE.network->GetSendGQoS(
-        //    tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
-        //EXPECT_TRUE(enabled);
-        //EXPECT_EQ(SERVICETYPE_BESTEFFORT, serviceType);
-        //EXPECT_FALSE(overrideDSCP);
-
-        //EXPECT_EQ(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_CONTROLLEDLOAD));
-        //EXPECT_EQ(0, ViE.network->GetSendGQoS(
-        //    tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
-        //EXPECT_TRUE(enabled);
-        //EXPECT_EQ(SERVICETYPE_CONTROLLEDLOAD, serviceType);
-        //EXPECT_FALSE(overrideDSCP);
-
-        //EXPECT_EQ(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_GUARANTEED));
-        //EXPECT_EQ(0, ViE.network->GetSendGQoS(
-        //    tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
-        //EXPECT_TRUE(enabled);
-        //EXPECT_EQ(SERVICETYPE_GUARANTEED, serviceType);
-        //EXPECT_FALSE(overrideDSCP);
-
-        //EXPECT_EQ(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, true, SERVICETYPE_QUALITATIVE));
-        //EXPECT_EQ(0, ViE.network->GetSendGQoS(
-        //    tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
-        //EXPECT_TRUE(enabled);
-        //EXPECT_EQ(SERVICETYPE_QUALITATIVE, serviceType);
-        //EXPECT_FALSE(overrideDSCP);
-
-        //EXPECT_EQ(0, ViE.network->SetSendGQoS(
-        //    tbChannel.videoChannel, false, SERVICETYPE_QUALITATIVE));
-        //EXPECT_EQ(0, ViE.network->GetSendGQoS(
-        //    tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
-        //EXPECT_FALSE(enabled);
-#endif
-    }
-    {
-        //
-        // MTU and packet burst
-        //
-        // Create a video channel
-        TbVideoChannel tbChannel(ViE);
-        // Invalid input
-        EXPECT_NE(0, ViE.network->SetMTU(tbChannel.videoChannel, 1600));
-        // Valid input
-        EXPECT_EQ(0, ViE.network->SetMTU(tbChannel.videoChannel, 800));
-    }
-
-    //***************************************************************
-    //	Testing finished. Tear down Video Engine
-    //***************************************************************
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc
deleted file mode 100644
index 947939309b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc
+++ /dev/null
@@ -1,597 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_record.cc
-//
-// This code is also used as sample code for ViE 3.0
-//
-
-#include <fstream>
-#include <stdio.h>
-
-#include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-
-#define VCM_RED_PAYLOAD_TYPE            96
-#define VCM_ULPFEC_PAYLOAD_TYPE         97
-#define DEFAULT_AUDIO_PORT              11113
-#define DEFAULT_AUDIO_CODEC             "ISAC"
-#define DEFAULT_VIDEO_CODEC_WIDTH       640
-#define DEFAULT_VIDEO_CODEC_HEIGHT      480
-#define DEFAULT_VIDEO_CODEC_START_RATE  1000
-#define DEFAULT_RECORDING_FOLDER        "RECORDING"
-#define DEFAULT_RECORDING_AUDIO         "/audio_debug.aec"
-#define DEFAULT_RECORDING_VIDEO         "/video_debug.yuv"
-#define DEFAULT_RECORDING_AUDIO_RTP     "/audio_rtpdump.rtp"
-#define DEFAULT_RECORDING_VIDEO_RTP     "/video_rtpdump.rtp"
-
-bool GetAudioDevices(webrtc::VoEBase* voe_base,
-                     webrtc::VoEHardware* voe_hardware,
-                     char* recording_device_name,
-                     int& recording_device_index,
-                     char* playbackDeviceName,
-                     int& playback_device_index);
-bool GetAudioCodecRecord(webrtc::VoECodec* voe_codec,
-                         webrtc::CodecInst& audio_codec);
-
-int VideoEngineSampleRecordCode(void* window1, void* window2) {
-  int error = 0;
-  // Audio settings.
-  int audio_tx_port = DEFAULT_AUDIO_PORT;
-  int audio_rx_port = DEFAULT_AUDIO_PORT;
-  webrtc::CodecInst audio_codec;
-  int audio_channel = -1;
-  int audio_capture_device_index = -1;
-  int audio_playback_device_index = -1;
-  const unsigned int KMaxDeviceNameLength = 128;
-  const unsigned int KMaxUniqueIdLength = 256;
-  char deviceName[KMaxDeviceNameLength];
-  char audio_capture_device_name[KMaxUniqueIdLength] = "";
-  char audio_playbackDeviceName[KMaxUniqueIdLength] = "";
-
-  // Network settings.
-  const char* ipAddress = "127.0.0.1";
-  const int rtpPort = 6000;
-
-  //
-  // Create a VideoEngine instance
-  //
-  webrtc::VideoEngine* ptrViE = NULL;
-  ptrViE = webrtc::VideoEngine::Create();
-  if (ptrViE == NULL) {
-    printf("ERROR in VideoEngine::Create\n");
-    return -1;
-  }
-
-  error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
-  if (error == -1) {
-    printf("ERROR in VideoEngine::SetTraceLevel\n");
-    return -1;
-  }
-
-  std::string trace_file =
-    ViETest::GetResultOutputPath() + "ViERecordCall_trace.txt";
-  error = ptrViE->SetTraceFile(trace_file.c_str());
-  if (error == -1) {
-    printf("ERROR in VideoEngine::SetTraceFile\n");
-    return -1;
-  }
-
-  //
-  // Create a VoE instance
-  //
-  webrtc::VoiceEngine* voe = webrtc::VoiceEngine::Create();
-  //
-  // Init VideoEngine and create a channel
-  //
-  webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
-  if (ptrViEBase == NULL) {
-    printf("ERROR in ViEBase::GetInterface\n");
-    return -1;
-  }
-
-  error = ptrViEBase->Init();
-  if (error == -1) {
-    printf("ERROR in ViEBase::Init\n");
-    return -1;
-  }
-
-  webrtc::VoEBase* voe_base = webrtc::VoEBase::GetInterface(voe);
-  if (voe_base == NULL) {
-    printf("ERROR in VoEBase::GetInterface\n");
-    return -1;
-  }
-  error = voe_base->Init();
-  if (error == -1) {
-    printf("ERROR in VoEBase::Init\n");
-    return -1;
-  }
-
-  int videoChannel = -1;
-  error = ptrViEBase->CreateChannel(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::CreateChannel\n");
-    return -1;
-  }
-
-  webrtc::VoEHardware* voe_hardware =
-    webrtc::VoEHardware::GetInterface(voe);
-  webrtc::VoECodec* voe_codec = webrtc::VoECodec::GetInterface(voe);
-  webrtc::VoEAudioProcessing* voe_apm =
-       webrtc::VoEAudioProcessing::GetInterface(voe);
-  webrtc::VoENetwork* voe_network =
-    webrtc::VoENetwork::GetInterface(voe);
-
-  // Get the audio device for the call.
-  memset(audio_capture_device_name, 0, KMaxUniqueIdLength);
-  memset(audio_playbackDeviceName, 0, KMaxUniqueIdLength);
-  GetAudioDevices(voe_base, voe_hardware, audio_capture_device_name,
-                  audio_capture_device_index, audio_playbackDeviceName,
-                  audio_playback_device_index);
-
-  // Get the audio codec for the call.
-  memset(static_cast<void*>(&audio_codec), 0, sizeof(audio_codec));
-  GetAudioCodecRecord(voe_codec, audio_codec);
-
-  audio_channel = voe_base->CreateChannel();
-
-  rtc::scoped_ptr<webrtc::test::VoiceChannelTransport> voice_channel_transport(
-      new webrtc::test::VoiceChannelTransport(voe_network, audio_channel));
-
-  voice_channel_transport->SetSendDestination(ipAddress, audio_tx_port);
-  voice_channel_transport->SetLocalReceiver(audio_rx_port);
-
-  voe_hardware->SetRecordingDevice(audio_capture_device_index);
-  voe_hardware->SetPlayoutDevice(audio_playback_device_index);
-  voe_codec->SetSendCodec(audio_channel, audio_codec);
-  voe_apm->SetAgcStatus(true, webrtc::kAgcDefault);
-  voe_apm->SetNsStatus(true, webrtc::kNsHighSuppression);
-
-  //
-  // List available capture devices, allocate and connect.
-  //
-  webrtc::ViECapture* ptrViECapture =
-      webrtc::ViECapture::GetInterface(ptrViE);
-  if (ptrViECapture == NULL) {
-    printf("ERROR in ViECapture::GetInterface\n");
-    return -1;
-  }
-
-  webrtc::VoERTP_RTCP* ptrVoERtpRtcp =
-    webrtc::VoERTP_RTCP::GetInterface(voe);
-  if (ptrVoERtpRtcp == NULL) {
-    printf("ERROR in VoERTP_RTCP::GetInterface\n");
-    return -1;
-  }
-
-  memset(deviceName, 0, KMaxDeviceNameLength);
-  char uniqueId[KMaxUniqueIdLength];
-  memset(uniqueId, 0, KMaxUniqueIdLength);
-
-  printf("Available capture devices:\n");
-  int captureIdx = 0;
-  for (captureIdx = 0;
-       captureIdx < ptrViECapture->NumberOfCaptureDevices();
-       captureIdx++) {
-    memset(deviceName, 0, KMaxDeviceNameLength);
-    memset(uniqueId, 0, KMaxUniqueIdLength);
-
-    error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
-                                            KMaxDeviceNameLength, uniqueId,
-                                            KMaxUniqueIdLength);
-    if (error == -1) {
-      printf("ERROR in ViECapture::GetCaptureDevice\n");
-      return -1;
-    }
-    printf("\t %d. %s\n", captureIdx + 1, deviceName);
-  }
-  printf("\nChoose capture device: ");
-#ifdef WEBRTC_ANDROID
-  captureIdx = 0;
-  printf("0\n");
-#else
-  if (scanf("%d", &captureIdx) != 1) {
-    printf("Error in scanf()\n");
-    return -1;
-  }
-  getc(stdin);
-  captureIdx = captureIdx - 1;  // Compensate for idx start at 1.
-#endif
-  error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
-                                          KMaxDeviceNameLength, uniqueId,
-                                          KMaxUniqueIdLength);
-  if (error == -1) {
-    printf("ERROR in ViECapture::GetCaptureDevice\n");
-    return -1;
-  }
-
-  int captureId = 0;
-  error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
-                                               captureId);
-  if (error == -1) {
-    printf("ERROR in ViECapture::AllocateCaptureDevice\n");
-    return -1;
-  }
-
-  error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViECapture::ConnectCaptureDevice\n");
-    return -1;
-  }
-
-  error = ptrViECapture->StartCapture(captureId);
-  if (error == -1) {
-    printf("ERROR in ViECapture::StartCapture\n");
-    return -1;
-  }
-
-  //
-  // RTP/RTCP settings
-  //
-  webrtc::ViERTP_RTCP* ptrViERtpRtcp =
-      webrtc::ViERTP_RTCP::GetInterface(ptrViE);
-  if (ptrViERtpRtcp == NULL) {
-    printf("ERROR in ViERTP_RTCP::GetInterface\n");
-    return -1;
-  }
-
-  error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
-                                       webrtc::kRtcpCompound_RFC4585);
-  if (error == -1) {
-    printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
-    return -1;
-  }
-
-  error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
-      videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
-  if (error == -1) {
-    printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
-    return -1;
-  }
-
-  error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
-  if (error == -1) {
-    printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
-    return -1;
-  }
-
-  //
-  // Set up rendering
-  //
-  webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
-  if (ptrViERender == NULL) {
-    printf("ERROR in ViERender::GetInterface\n");
-    return -1;
-  }
-
-  error = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
-  if (error == -1) {
-    printf("ERROR in ViERender::AddRenderer\n");
-    return -1;
-  }
-
-  error = ptrViERender->StartRender(captureId);
-  if (error == -1) {
-    printf("ERROR in ViERender::StartRender\n");
-    return -1;
-  }
-
-  error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
-                                    1.0);
-  if (error == -1) {
-    printf("ERROR in ViERender::AddRenderer\n");
-    return -1;
-  }
-
-  error = ptrViERender->StartRender(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViERender::StartRender\n");
-    return -1;
-  }
-
-  //
-  // Setup codecs
-  //
-  webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
-  if (ptrViECodec == NULL) {
-    printf("ERROR in ViECodec::GetInterface\n");
-    return -1;
-  }
-
-  webrtc::VideoCodec videoCodec;
-  memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
-  int codecIdx = 0;
-
-#ifdef WEBRTC_ANDROID
-  codecIdx = 0;
-  printf("0\n");
-#else
-  codecIdx = 0;  // Compensate for idx start at 1.
-#endif
-
-  error = ptrViECodec->GetCodec(codecIdx, videoCodec);
-  if (error == -1) {
-     printf("ERROR in ViECodec::GetCodec\n");
-     return -1;
-  }
-
-  // Set spatial resolution option
-  videoCodec.width = DEFAULT_VIDEO_CODEC_WIDTH;
-  videoCodec.height = DEFAULT_VIDEO_CODEC_HEIGHT;
-
-  // Set start bit rate
-  videoCodec.startBitrate = DEFAULT_VIDEO_CODEC_START_RATE;
-
-  error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
-  if (error == -1) {
-    printf("ERROR in ViECodec::SetSendCodec\n");
-    return -1;
-  }
-
-  //
-  // Address settings
-  //
-  webrtc::ViENetwork* ptrViENetwork =
-      webrtc::ViENetwork::GetInterface(ptrViE);
-  if (ptrViENetwork == NULL) {
-    printf("ERROR in ViENetwork::GetInterface\n");
-    return -1;
-  }
-  webrtc::test::VideoChannelTransport* video_channel_transport =
-      new webrtc::test::VideoChannelTransport(ptrViENetwork, videoChannel);
-
-  error = video_channel_transport->SetSendDestination(ipAddress, rtpPort);
-  if (error == -1) {
-    printf("ERROR in SetSendDestination\n");
-    return -1;
-  }
-  error = video_channel_transport->SetLocalReceiver(rtpPort);
-  if (error == -1) {
-    printf("ERROR in SetLocalReceiver\n");
-    return -1;
-  }
-
-  std::string str;
-  int enable_labeling = 0;
-  std::cout << std::endl;
-  std::cout << "Do you want to label this recording?" << std::endl;
-  std::cout << "0. No (default)." << std::endl;
-  std::cout << "1. This call will be labeled on the fly." << std::endl;
-  std::getline(std::cin, str);
-  enable_labeling = atoi(str.c_str());
-
-  uint32_t folder_time = static_cast<uint32_t>
-    (webrtc::TickTime::MillisecondTimestamp());
-  std::stringstream folder_time_str;
-  folder_time_str <<  folder_time;
-  const std::string folder_name = "recording" + folder_time_str.str();
-  printf("recording name = %s\n", folder_name.c_str());
-  // TODO(mikhal): use file_utils.
-#ifdef WIN32
-  _mkdir(folder_name.c_str());
-#else
-  mkdir(folder_name.c_str(), 0777);
-#endif
-  const std::string audio_filename =  folder_name + DEFAULT_RECORDING_AUDIO;
-  const std::string video_filename =  folder_name + DEFAULT_RECORDING_VIDEO;
-  const std::string audio_rtp_filename = folder_name +
-    DEFAULT_RECORDING_AUDIO_RTP;
-  const std::string video_rtp_filename = folder_name +
-    DEFAULT_RECORDING_VIDEO_RTP;
-  std::fstream timing;
-  if (enable_labeling == 1) {
-    std::cout << "Press enter to stamp current time."<< std::endl;
-    std::string timing_file = folder_name + "/labeling.txt";
-    timing.open(timing_file.c_str(), std::fstream::out | std::fstream::app);
-  }
-  printf("\nPress enter to start recording\n");
-  std::getline(std::cin, str);
-  printf("\nRecording started\n\n");
-
-  error = ptrViEBase->StartReceive(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViENetwork::StartReceive\n");
-    return -1;
-  }
-
-  error = ptrViEBase->StartSend(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViENetwork::StartSend\n");
-    return -1;
-  }
-  error = voe_base->StartSend(audio_channel);
-  if (error == -1) {
-    printf("ERROR in VoENetwork::StartSend\n");
-    return -1;
-  }
-
-  //  Engine started
-
-  voe_apm->StartDebugRecording(audio_filename.c_str());
-  ptrViECodec->StartDebugRecording(videoChannel, video_filename.c_str());
-  ptrViERtpRtcp->StartRTPDump(videoChannel,
-                              video_rtp_filename.c_str(), webrtc::kRtpOutgoing);
-  ptrVoERtpRtcp->StartRTPDump(audio_channel,
-                              audio_rtp_filename.c_str(), webrtc::kRtpOutgoing);
-  printf("Press s + enter to stop...");
-  int64_t clock_time;
-  if (enable_labeling == 1) {
-    clock_time = webrtc::TickTime::MillisecondTimestamp();
-    timing << clock_time << std::endl;
-  }
-  char c = getc(stdin);
-  fflush(stdin);
-  while (c != 's') {
-    if (c == '\n' && enable_labeling == 1) {
-      clock_time = webrtc::TickTime::MillisecondTimestamp();
-      timing << clock_time << std::endl;
-    }
-    c = getc(stdin);
-  }
-  if (enable_labeling == 1) {
-    clock_time = webrtc::TickTime::MillisecondTimestamp();
-    timing << clock_time << std::endl;
-  }
-
-  ptrViERtpRtcp->StopRTPDump(videoChannel, webrtc::kRtpOutgoing);
-  ptrVoERtpRtcp->StopRTPDump(audio_channel, webrtc::kRtpOutgoing);
-  voe_apm->StopDebugRecording();
-  ptrViECodec->StopDebugRecording(videoChannel);
-  if (enable_labeling == 1)
-    timing.close();
-
-  //  Recording finished. Tear down Video Engine.
-
-  error = ptrViEBase->StopReceive(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::StopReceive\n");
-    return -1;
-  }
-
-  error = ptrViEBase->StopSend(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::StopSend\n");
-    return -1;
-  }
-  error = voe_base->StopSend(audio_channel);
-
-  error = ptrViERender->StopRender(captureId);
-  if (error == -1) {
-    printf("ERROR in ViERender::StopRender\n");
-    return -1;
-  }
-
-  error = ptrViERender->RemoveRenderer(captureId);
-  if (error == -1) {
-    printf("ERROR in ViERender::RemoveRenderer\n");
-    return -1;
-  }
-
-  error = ptrViERender->StopRender(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViERender::StopRender\n");
-    return -1;
-  }
-
-  error = ptrViERender->RemoveRenderer(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViERender::RemoveRenderer\n");
-    return -1;
-  }
-
-  error = ptrViECapture->StopCapture(captureId);
-  if (error == -1) {
-    printf("ERROR in ViECapture::StopCapture\n");
-    return -1;
-  }
-
-  error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
-    return -1;
-  }
-
-  error = ptrViECapture->ReleaseCaptureDevice(captureId);
-  if (error == -1) {
-    printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
-    return -1;
-  }
-
-  error = ptrViEBase->DeleteChannel(videoChannel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::DeleteChannel\n");
-    return -1;
-  }
-  delete video_channel_transport;
-
-  int remainingInterfaces = 0;
-  remainingInterfaces = ptrViECodec->Release();
-  remainingInterfaces += ptrViECapture->Release();
-  remainingInterfaces += ptrViERtpRtcp->Release();
-  remainingInterfaces += ptrViERender->Release();
-  remainingInterfaces += ptrViENetwork->Release();
-  remainingInterfaces += ptrViEBase->Release();
-  if (remainingInterfaces > 0) {
-    printf("ERROR: Could not release all interfaces\n");
-    return -1;
-  }
-  bool deleted = webrtc::VideoEngine::Delete(ptrViE);
-  if (deleted == false) {
-    printf("ERROR in VideoEngine::Delete\n");
-    return -1;
-  }
-  return 0;
-}
-
-
-// TODO(mikhal): Place above functionality under this class.
-int ViEAutoTest::ViERecordCall() {
-  ViETest::Log(" ");
-  ViETest::Log("========================================");
-  ViETest::Log(" ViE Record Call\n");
-
-  if (VideoEngineSampleRecordCode(_window1, _window2) == 0) {
-    ViETest::Log(" ");
-    ViETest::Log(" ViE Autotest Record Call Done");
-    ViETest::Log("========================================");
-    ViETest::Log(" ");
-    return 0;
-  }
-
-  ViETest::Log(" ");
-  ViETest::Log(" ViE Autotest Record Call Failed");
-  ViETest::Log("========================================");
-  ViETest::Log(" ");
-  return 1;
-}
-
-bool GetAudioCodecRecord(webrtc::VoECodec* voe_codec,
-                         webrtc::CodecInst& audio_codec) {
-  int error = 0;
-  int number_of_errors = 0;
-  memset(&audio_codec, 0, sizeof(webrtc::CodecInst));
-
-  while (1) {
-    int codec_idx = 0;
-    int default_codec_idx = 0;
-    for (codec_idx = 0; codec_idx < voe_codec->NumOfCodecs(); codec_idx++) {
-      error = voe_codec->GetCodec(codec_idx, audio_codec);
-      number_of_errors += ViETest::TestError(error == 0,
-                                             "ERROR: %s at line %d",
-                                             __FUNCTION__, __LINE__);
-
-      // Test for default codec index.
-      if (strcmp(audio_codec.plname, DEFAULT_AUDIO_CODEC) == 0) {
-        default_codec_idx = codec_idx;
-      }
-    }
-    error = voe_codec->GetCodec(default_codec_idx, audio_codec);
-    number_of_errors += ViETest::TestError(error == 0,
-                                           "ERROR: %s at line %d",
-                                           __FUNCTION__, __LINE__);
-    return true;
-  }
-  assert(false);
-  return false;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
deleted file mode 100644
index c30fbf32a4..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
+++ /dev/null
@@ -1,312 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_render.cc
-//
-
-#include "webrtc/base/format_macros.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-
-#include "webrtc/modules/video_render/include/video_render.h"
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-
-#if defined(WIN32)
-#include <ddraw.h>
-#include <tchar.h>
-#include <windows.h>
-#elif defined(WEBRTC_LINUX)
-    //From windgi.h
-    #undef RGB
-    #define RGB(r,g,b)          ((unsigned long)(((unsigned char)(r)|((unsigned short)((unsigned char)(g))<<8))|(((unsigned long)(unsigned char)(b))<<16)))
-    //From ddraw.h
-/*    typedef struct _DDCOLORKEY
- {
- DWORD       dwColorSpaceLowValue;   // low boundary of color space that is to
- DWORD       dwColorSpaceHighValue;  // high boundary of color space that is
- } DDCOLORKEY;*/
-#elif defined(WEBRTC_MAC)
-#endif
-
-class ViEAutoTestExternalRenderer: public webrtc::ExternalRenderer
-{
-public:
-    ViEAutoTestExternalRenderer() :
-        _width(0),
-        _height(0)
-    {
-    }
-    virtual int FrameSizeChange(unsigned int width, unsigned int height,
-                                unsigned int numberOfStreams)
-    {
-        _width = width;
-        _height = height;
-        return 0;
-    }
-
-    virtual int DeliverFrame(unsigned char* buffer,
-                             size_t bufferSize,
-                             uint32_t time_stamp,
-                             int64_t ntp_time_ms,
-                             int64_t render_time,
-                             void* /*handle*/) {
-      if (bufferSize != CalcBufferSize(webrtc::kI420, _width, _height)) {
-        ViETest::Log("Incorrect render buffer received, of length = %" PRIuS
-                     "\n", bufferSize);
-        return 0;
-      }
-      return 0;
-    }
-
-    virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) {
-      EXPECT_EQ(webrtc_frame.width(), _width);
-      EXPECT_EQ(webrtc_frame.height(), _height);
-      return 0;
-    }
-
-    virtual bool IsTextureSupported() { return false; }
-
-public:
-    virtual ~ViEAutoTestExternalRenderer()
-    {
-    }
-private:
-    int _width, _height;
-};
-
-void ViEAutoTest::ViERenderStandardTest()
-{
-    //***************************************************************
-    //	Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-    int rtpPort = 6000;
-
-    TbInterfaces ViE("ViERenderStandardTest");
-
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-    TbCaptureDevice tbCapture(ViE); // Create a capture device
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-    tbChannel.StartReceive(rtpPort);
-    tbChannel.StartSend(rtpPort);
-
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm2));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
-    ViETest::Log("\nCapture device is renderered in Window 1");
-    ViETest::Log("Remote stream is renderered in Window 2");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
-    // PIP and full screen rendering is not supported on Android
-#ifndef WEBRTC_ANDROID
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, _window2, 0, 0.75, 0.75, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-
-    ViETest::Log("\nCapture device is now rendered in Window 2, PiP.");
-    ViETest::Log("Switching to full screen rendering in %d seconds.\n",
-                 kAutoTestSleepTimeMs / 1000);
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm2));
-
-    // Destroy render module and create new in full screen mode
-    webrtc::VideoRender::DestroyVideoRender(_vrm1);
-    _vrm1 = NULL;
-    _vrm1 = webrtc::VideoRender::CreateVideoRender(
-        4563, _window1, true, _renderType);
-    EXPECT_TRUE(_vrm1 != NULL);
-
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, _window1, 0, 0.75f, 0.75f, 1.0f, 1.0f));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbChannel.videoChannel, _window1, 1, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-
-    // Destroy full screen render module and create new in normal mode
-    webrtc::VideoRender::DestroyVideoRender(_vrm1);
-    _vrm1 = NULL;
-    _vrm1 = webrtc::VideoRender::CreateVideoRender(
-        4561, _window1, false, _renderType);
-    EXPECT_TRUE(_vrm1 != NULL);
-#endif
-
-    //***************************************************************
-    //	Engine ready. Begin testing class
-    //***************************************************************
-
-
-    //***************************************************************
-    //	Testing finished. Tear down Video Engine
-    //***************************************************************
-    tbCapture.Disconnect(tbChannel.videoChannel);
-}
-
-void ViEAutoTest::ViERenderExtendedTest()
-{
-    int rtpPort = 6000;
-
-    TbInterfaces ViE("ViERenderExtendedTest");
-
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-    TbCaptureDevice tbCapture(ViE); // Create a capture device
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-    tbChannel.StartReceive(rtpPort);
-    tbChannel.StartSend(rtpPort);
-
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm2));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
-    EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
-    ViETest::Log("\nCapture device is renderered in Window 1");
-    ViETest::Log("Remote stream is renderered in Window 2");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-#ifdef _WIN32
-    ViETest::Log("\nConfiguring Window2");
-    ViETest::Log("you will see video only in first quadrant");
-    EXPECT_EQ(0, ViE.render->ConfigureRender(
-        tbChannel.videoChannel, 0, 0.0f, 0.0f, 0.5f, 0.5f));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    ViETest::Log("you will see video only in fourth quadrant");
-    EXPECT_EQ(0, ViE.render->ConfigureRender(
-        tbChannel.videoChannel, 0, 0.5f, 0.5f, 1.0f, 1.0f));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    ViETest::Log("normal video on Window2");
-    EXPECT_EQ(0, ViE.render->ConfigureRender(
-        tbChannel.videoChannel, 0, 0.0f, 0.0f, 1.0f, 1.0f));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-#endif
-
-    ViETest::Log("\nEnabling Full Screen render in 5 sec");
-
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm2));
-
-    // Destroy render module and create new in full screen mode
-    webrtc::VideoRender::DestroyVideoRender(_vrm1);
-    _vrm1 = NULL;
-    _vrm1 = webrtc::VideoRender::CreateVideoRender(
-        4563, _window1, true, _renderType);
-    EXPECT_TRUE(_vrm1 != NULL);
-
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, _window1, 0, 0.0f, 0.0f, 1.0f, 1.0f));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    ViETest::Log("\nStop renderer");
-    EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
-    ViETest::Log("\nRemove renderer");
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-
-    // Destroy full screen render module and create new for external rendering
-    webrtc::VideoRender::DestroyVideoRender(_vrm1);
-    _vrm1 = NULL;
-    _vrm1 = webrtc::VideoRender::CreateVideoRender(4564, NULL, false,
-                                                   _renderType);
-    EXPECT_TRUE(_vrm1 != NULL);
-
-    EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
-
-    ViETest::Log("\nExternal Render Test");
-    ViEAutoTestExternalRenderer externalRenderObj;
-    EXPECT_EQ(0, ViE.render->AddRenderer(
-        tbCapture.captureId, webrtc::kVideoI420, &externalRenderObj));
-    EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-
-    // Destroy render module for external rendering and create new in normal
-    // mode
-    webrtc::VideoRender::DestroyVideoRender(_vrm1);
-    _vrm1 = NULL;
-    _vrm1 = webrtc::VideoRender::CreateVideoRender(
-        4561, _window1, false, _renderType);
-    EXPECT_TRUE(_vrm1 != NULL);
-    tbCapture.Disconnect(tbChannel.videoChannel);
-}
-
-void ViEAutoTest::ViERenderAPITest() {
-  TbInterfaces ViE("ViERenderAPITest");
-
-  TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-  TbCaptureDevice tbCapture(ViE);
-  tbCapture.ConnectTo(tbChannel.videoChannel);
-  tbChannel.StartReceive();
-  tbChannel.StartSend();
-
-  EXPECT_EQ(0, ViE.render->AddRenderer(
-      tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
-  EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-  EXPECT_EQ(0, ViE.render->AddRenderer(
-      tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
-  EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
-  // Test setting HW render delay.
-  // Already started.
-  EXPECT_EQ(-1, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 50));
-  EXPECT_EQ(0, ViE.render->StopRender(tbChannel.videoChannel));
-
-  // Invalid values.
-  EXPECT_EQ(-1, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 9));
-  EXPECT_EQ(-1, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel,
-                                                   501));
-  // Valid values.
-  EXPECT_EQ(0, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 11));
-  EXPECT_EQ(0, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 499));
-
-  EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
-  EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-  tbCapture.Disconnect(tbChannel.videoChannel);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
deleted file mode 100644
index 47dbae4e42..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ /dev/null
@@ -1,918 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <iostream>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-
-class ViERtpObserver: public webrtc::ViERTPObserver
-{
-public:
-    ViERtpObserver()
-    {
-    }
-    virtual ~ViERtpObserver()
-    {
-    }
-
-    virtual void IncomingSSRCChanged(const int videoChannel,
-                                     const unsigned int SSRC)
-    {
-    }
-    virtual void IncomingCSRCChanged(const int videoChannel,
-                                     const unsigned int CSRC, const bool added)
-    {
-    }
-};
-
-void ViEAutoTest::ViERtpRtcpStandardTest()
-{
-    // ***************************************************************
-    // Begin create/initialize WebRTC Video Engine for testing
-    // ***************************************************************
-
-    // Create VIE
-    TbInterfaces ViE("ViERtpRtcpStandardTest");
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
-    // Create a capture device
-    TbCaptureDevice tbCapture(ViE);
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-
-    ViETest::Log("\n");
-    TbExternalTransport myTransport(*(ViE.network), tbChannel.videoChannel,
-                                    NULL);
-
-    ViE.network->DeregisterSendTransport(tbChannel.videoChannel);
-    EXPECT_EQ(0, ViE.network->RegisterSendTransport(
-        tbChannel.videoChannel, myTransport));
-
-    // ***************************************************************
-    // Engine ready. Begin testing class
-    // ***************************************************************
-    unsigned short startSequenceNumber = 12345;
-    ViETest::Log("Set start sequence number: %u", startSequenceNumber);
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, startSequenceNumber));
-    const unsigned int kVideoSsrc = 123456;
-    // Set an SSRC to avoid issues with collisions.
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, kVideoSsrc,
-                                            webrtc::kViEStreamTypeNormal, 0));
-
-    myTransport.EnableSequenceNumberCheck();
-
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    AutoTestSleep(2000);
-
-    unsigned short receivedSequenceNumber =
-        myTransport.GetFirstSequenceNumber();
-    ViETest::Log("First received sequence number: %u\n",
-                 receivedSequenceNumber);
-    EXPECT_EQ(startSequenceNumber, receivedSequenceNumber);
-
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    //
-    // RTCP CName
-    //
-    ViETest::Log("Testing CName\n");
-    const char* sendCName = "ViEAutoTestCName\0";
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPCName(tbChannel.videoChannel, sendCName));
-
-    char returnCName[webrtc::ViERTP_RTCP::KMaxRTCPCNameLength];
-    memset(returnCName, 0, webrtc::ViERTP_RTCP::KMaxRTCPCNameLength);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPCName(
-        tbChannel.videoChannel, returnCName));
-    EXPECT_STRCASEEQ(sendCName, returnCName);
-
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    AutoTestSleep(1000);
-
-    if (FLAGS_include_timing_dependent_tests) {
-      char remoteCName[webrtc::ViERTP_RTCP::KMaxRTCPCNameLength];
-      memset(remoteCName, 0, webrtc::ViERTP_RTCP::KMaxRTCPCNameLength);
-      EXPECT_EQ(0, ViE.rtp_rtcp->GetRemoteRTCPCName(
-          tbChannel.videoChannel, remoteCName));
-      EXPECT_STRCASEEQ(sendCName, remoteCName);
-    }
-
-
-    //
-    // Pacing
-    //
-    webrtc::RtcpStatistics received;
-    int64_t recRttMs = 0;
-    unsigned int sentTotalBitrate = 0;
-    unsigned int sentVideoBitrate = 0;
-    unsigned int sentFecBitrate = 0;
-    unsigned int sentNackBitrate = 0;
-
-    ViETest::Log("Testing Pacing\n");
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-    myTransport.ClearStats();
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    NetworkParameters network;
-    network.packet_loss_rate = 0;
-    network.loss_model = kUniformLoss;
-    myTransport.SetNetworkParameters(network);
-
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetReceiveChannelRtcpStatistics(
-        tbChannel.videoChannel, received, recRttMs));
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
-        tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
-        sentFecBitrate, sentNackBitrate));
-
-    int num_rtp_packets = 0;
-    int num_dropped_packets = 0;
-    int num_rtcp_packets = 0;
-    std::map<uint8_t, int> packet_counters;
-    myTransport.GetStats(num_rtp_packets, num_dropped_packets, num_rtcp_packets,
-                         &packet_counters);
-    EXPECT_GT(num_rtp_packets, 0);
-    EXPECT_EQ(num_dropped_packets, 0);
-    EXPECT_GT(num_rtcp_packets, 0);
-    EXPECT_GT(sentTotalBitrate, 0u);
-    EXPECT_EQ(sentNackBitrate, 0u);
-    EXPECT_EQ(received.cumulative_lost, 0u);
-
-    //
-    // RTX
-    //
-    ViETest::Log("Testing NACK over RTX\n");
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-    myTransport.ClearStats();
-
-    const uint8_t kRtxPayloadType = 96;
-    // Temporarily disable pacing.
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        tbChannel.videoChannel, false));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxSendPayloadType(tbChannel.videoChannel,
-                                                     kRtxPayloadType));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRtxReceivePayloadType(tbChannel.videoChannel,
-                                                        kRtxPayloadType));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, 1234,
-                                            webrtc::kViEStreamTypeRtx, 0));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRemoteSSRCType(tbChannel.videoChannel,
-                                                 webrtc::kViEStreamTypeRtx,
-                                                 1234));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, startSequenceNumber));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    // Make sure the first key frame gets through.
-    AutoTestSleep(100);
-    const int kPacketLossRate = 20;
-    network.packet_loss_rate = kPacketLossRate;
-    network.loss_model = kUniformLoss;
-    myTransport.SetNetworkParameters(network);
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetReceiveChannelRtcpStatistics(
-        tbChannel.videoChannel, received, recRttMs));
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
-        tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
-        sentFecBitrate, sentNackBitrate));
-
-    packet_counters.clear();
-    myTransport.GetStats(num_rtp_packets, num_dropped_packets, num_rtcp_packets,
-                         &packet_counters);
-    EXPECT_GT(num_rtp_packets, 0);
-    EXPECT_GT(num_dropped_packets, 0);
-    EXPECT_GT(num_rtcp_packets, 0);
-    EXPECT_GT(packet_counters[kRtxPayloadType], 0);
-
-    // Make sure we have lost packets and that they were retransmitted.
-    // TODO(holmer): Disabled due to being flaky. Could be a bug in our stats.
-    // EXPECT_GT(recCumulativeLost, 0u);
-    EXPECT_GT(sentTotalBitrate, 0u);
-    EXPECT_GT(sentNackBitrate, 0u);
-
-    //
-    //  Statistics
-    //
-    // Stop and restart to clear stats
-    ViETest::Log("Testing statistics\n");
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, false));
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    myTransport.ClearStats();
-    network.packet_loss_rate = kPacketLossRate;
-    network.loss_model = kUniformLoss;
-    myTransport.SetNetworkParameters(network);
-
-    // Start send to verify sending stats
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, startSequenceNumber));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    webrtc::RtcpStatistics sent;
-    int64_t sentRttMs = 0;
-
-    // Fraction lost is a transient value that can get reset after a new rtcp
-    // report block. Make regular polls to make sure it is propagated.
-    // TODO(sprang): Replace with callbacks, when those are fully implemented.
-    int time_to_sleep = kAutoTestSleepTimeMs;
-    bool got_send_channel_frac_lost = false;
-    bool got_receive_channel_frac_lost = false;
-    while (time_to_sleep > 0) {
-      AutoTestSleep(500);
-      time_to_sleep -= 500;
-      EXPECT_EQ(0,
-                ViE.rtp_rtcp->GetSendChannelRtcpStatistics(
-                    tbChannel.videoChannel, sent, sentRttMs));
-      got_send_channel_frac_lost |= sent.fraction_lost > 0;
-      EXPECT_EQ(0,
-                ViE.rtp_rtcp->GetReceiveChannelRtcpStatistics(
-                    tbChannel.videoChannel, received, recRttMs));
-      got_receive_channel_frac_lost |= received.fraction_lost > 0;
-    }
-    EXPECT_TRUE(got_send_channel_frac_lost);
-    EXPECT_TRUE(got_receive_channel_frac_lost);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
-        tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
-        sentFecBitrate, sentNackBitrate));
-
-    EXPECT_GT(sentTotalBitrate, 0u);
-    EXPECT_EQ(sentFecBitrate, 0u);
-    EXPECT_EQ(sentNackBitrate, 0u);
-
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-    AutoTestSleep(2000);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetSendChannelRtcpStatistics(
-        tbChannel.videoChannel, sent, sentRttMs));
-    EXPECT_GT(sent.cumulative_lost, 0u);
-    EXPECT_GT(sent.extended_max_sequence_number, startSequenceNumber);
-    EXPECT_GT(sent.jitter, 0u);
-    EXPECT_GT(sentRttMs, 0);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetReceiveChannelRtcpStatistics(
-        tbChannel.videoChannel, received, recRttMs));
-
-    EXPECT_GT(received.cumulative_lost, 0u);
-    EXPECT_GT(received.extended_max_sequence_number, startSequenceNumber);
-    EXPECT_GT(received.jitter, 0u);
-    EXPECT_GT(recRttMs, 0);
-
-    unsigned int estimated_bandwidth = 0;
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedSendBandwidth(
-        tbChannel.videoChannel,
-        &estimated_bandwidth));
-    EXPECT_GT(estimated_bandwidth, 0u);
-
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedReceiveBandwidth(
-          tbChannel.videoChannel,
-          &estimated_bandwidth));
-      EXPECT_GT(estimated_bandwidth, 0u);
-
-      int passive_channel = -1;
-      EXPECT_EQ(ViE.base->CreateReceiveChannel(passive_channel,
-                                               tbChannel.videoChannel), 0);
-      EXPECT_EQ(ViE.base->StartReceive(passive_channel), 0);
-      EXPECT_EQ(
-          ViE.rtp_rtcp->GetEstimatedReceiveBandwidth(passive_channel,
-                                                     &estimated_bandwidth),
-          0);
-      EXPECT_EQ(estimated_bandwidth, 0u);
-    }
-
-    // Check that rec stats extended max is greater than what we've sent.
-    EXPECT_GE(received.extended_max_sequence_number,
-              sent.extended_max_sequence_number);
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-    //
-    // Test bandwidth statistics with reserved bitrate
-    //
-
-    myTransport.ClearStats();
-    network.packet_loss_rate = 0;
-    network.loss_model = kUniformLoss;
-    myTransport.SetNetworkParameters(network);
-
-    ViE.rtp_rtcp->SetReservedTransmitBitrate(tbChannel.videoChannel, 2000000);
-
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    estimated_bandwidth = 0;
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedSendBandwidth(tbChannel.videoChannel,
-                                                         &estimated_bandwidth));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(0u, estimated_bandwidth);
-    }
-
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    //
-    // Test bandwidth statistics with NACK and FEC separately
-    //
-
-    myTransport.ClearStats();
-    network.packet_loss_rate = kPacketLossRate;
-    myTransport.SetNetworkParameters(network);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetFECStatus(
-        tbChannel.videoChannel, true, 96, 97));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
-        tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
-         sentFecBitrate, sentNackBitrate));
-
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(sentTotalBitrate, 0u);
-      EXPECT_GT(sentFecBitrate, 0u);
-      EXPECT_EQ(sentNackBitrate, 0u);
-    }
-
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetFECStatus(
-        tbChannel.videoChannel, false, 96, 97));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    AutoTestSleep(4 * kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
-        tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
-        sentFecBitrate, sentNackBitrate));
-
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(sentTotalBitrate, 0u);
-      EXPECT_EQ(sentFecBitrate, 0u);
-
-      // TODO(holmer): Test disabled due to being too flaky on buildbots. Tests
-      // for new API provide partial coverage.
-      // EXPECT_GT(sentNackBitrate, 0u);
-    }
-
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, false));
-
-
-    // Test to set SSRC
-    network.packet_loss_rate = 0;
-    myTransport.SetNetworkParameters(network);
-    myTransport.ClearStats();
-
-    unsigned int setSSRC = 0x01234567;
-    ViETest::Log("Set SSRC %u", setSSRC);
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, setSSRC));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    myTransport.EnableSSRCCheck();
-
-    AutoTestSleep(2000);
-    unsigned int receivedSSRC = myTransport.ReceivedSSRC();
-    ViETest::Log("Received SSRC %u\n", receivedSSRC);
-
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(setSSRC, receivedSSRC);
-
-      unsigned int localSSRC = 0;
-      EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(
-          tbChannel.videoChannel, localSSRC));
-      EXPECT_EQ(setSSRC, localSSRC);
-
-      unsigned int remoteSSRC = 0;
-      EXPECT_EQ(0, ViE.rtp_rtcp->GetRemoteSSRC(
-          tbChannel.videoChannel, remoteSSRC));
-      EXPECT_EQ(setSSRC, remoteSSRC);
-    }
-
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    ViETest::Log("Testing RTP dump...\n");
-
-    std::string inDumpName =
-        ViETest::GetResultOutputPath() + "IncomingRTPDump.rtp";
-    std::string outDumpName =
-        ViETest::GetResultOutputPath() + "OutgoingRTPDump.rtp";
-    EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
-        tbChannel.videoChannel, inDumpName.c_str(), webrtc::kRtpIncoming));
-    EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
-        tbChannel.videoChannel, outDumpName.c_str(), webrtc::kRtpOutgoing));
-
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    AutoTestSleep(1000);
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
-        tbChannel.videoChannel, webrtc::kRtpIncoming));
-    EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
-        tbChannel.videoChannel, webrtc::kRtpOutgoing));
-
-    // Make sure data was actually saved to the file and we stored the same
-    // amount of data in both files
-    FILE* inDump = fopen(inDumpName.c_str(), "r");
-    fseek(inDump, 0L, SEEK_END);
-    long inEndPos = ftell(inDump);
-    fclose(inDump);
-    FILE* outDump = fopen(outDumpName.c_str(), "r");
-    fseek(outDump, 0L, SEEK_END);
-    // long outEndPos = ftell(outDump);
-    fclose(outDump);
-
-    EXPECT_GT(inEndPos, 0);
-
-    // TODO(phoglund): This is flaky for some reason. Are the sleeps too
-    // short above?
-    // EXPECT_LT(inEndPos, outEndPos + 100);
-
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-
-    ViETest::Log("Testing Network Down...\n");
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
-    // Reenable pacing.
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        tbChannel.videoChannel, true));
-
-    webrtc::StreamDataCounters sent_before;
-    webrtc::StreamDataCounters received_before;
-    webrtc::StreamDataCounters sent_after;
-    webrtc::StreamDataCounters received_after;
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_before,
-                                                received_before));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    // Real-time mode.
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_after, received_after));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_after.transmitted.payload_bytes,
-                received_before.transmitted.payload_bytes);
-    }
-    // Simulate lost reception and verify that nothing is sent during that time.
-    ViE.network->SetNetworkTransmissionState(tbChannel.videoChannel, false);
-    // Allow the encoder to finish the current frame before we expect that no
-    // additional packets will be sent.
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    received_before.transmitted.payload_bytes =
-        received_after.transmitted.payload_bytes;
-    ViETest::Log("Network Down...\n");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_before,
-                                                received_before));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(received_before.transmitted.payload_bytes,
-                received_after.transmitted.payload_bytes);
-    }
-
-    // Network reception back. Video should now be sent.
-    ViE.network->SetNetworkTransmissionState(tbChannel.videoChannel, true);
-    ViETest::Log("Network Up...\n");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_before,
-                                                received_before));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_before.transmitted.payload_bytes,
-                received_after.transmitted.payload_bytes);
-    }
-    received_after.transmitted.payload_bytes =
-        received_before.transmitted.payload_bytes;
-    // Buffering mode.
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-    ViE.rtp_rtcp->SetSenderBufferingMode(tbChannel.videoChannel,
-                                         kAutoTestSleepTimeMs / 2);
-    // Add extra delay to the receiver to make sure it doesn't flush due to
-    // too old packets being received (as the down-time introduced is longer
-    // than what we buffer at the sender).
-    ViE.rtp_rtcp->SetReceiverBufferingMode(tbChannel.videoChannel,
-                                           3 * kAutoTestSleepTimeMs / 2);
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    // Simulate lost reception and verify that nothing is sent during that time.
-    ViETest::Log("Network Down...\n");
-    ViE.network->SetNetworkTransmissionState(tbChannel.videoChannel, false);
-    // Allow the encoder to finish the current frame before we expect that no
-    // additional packets will be sent.
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_before,
-                                                received_before));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_before.transmitted.payload_bytes,
-                received_after.transmitted.payload_bytes);
-    }
-    received_after.transmitted.payload_bytes =
-        received_before.transmitted.payload_bytes;
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_before,
-                                                received_before));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(received_after.transmitted.payload_bytes,
-                received_before.transmitted.payload_bytes);
-    }
-    // Network reception back. Video should now be sent.
-    ViETest::Log("Network Up...\n");
-    ViE.network->SetNetworkTransmissionState(tbChannel.videoChannel, true);
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
-                                                sent_before,
-                                                received_before));
-    if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_before.transmitted.payload_bytes,
-                received_after.transmitted.payload_bytes);
-    }
-    // TODO(holmer): Verify that the decoded framerate doesn't decrease on an
-    // outage when in buffering mode. This isn't currently possible because we
-    // don't have an API to get decoded framerate.
-
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
-
-    // Deregister external transport
-    EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-
-
-    //***************************************************************
-    //  Testing finished. Tear down Video Engine
-    //***************************************************************
-}
-
-void ViEAutoTest::ViERtpRtcpAPITest()
-{
-    //***************************************************************
-    //  Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-    // Create VIE
-    TbInterfaces ViE("ViERtpRtcpAPITest");
-
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
-    // Create a capture device
-    TbCaptureDevice tbCapture(ViE);
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-
-    //***************************************************************
-    //  Engine ready. Begin testing class
-    //***************************************************************
-
-    //
-    // Check different RTCP modes
-    //
-    webrtc::ViERTCPMode rtcpMode = webrtc::kRtcpNone;
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
-        tbChannel.videoChannel, rtcpMode));
-    EXPECT_EQ(webrtc::kRtcpCompound_RFC4585, rtcpMode);
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
-        tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
-        tbChannel.videoChannel, rtcpMode));
-    EXPECT_EQ(webrtc::kRtcpCompound_RFC4585, rtcpMode);
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
-        tbChannel.videoChannel, webrtc::kRtcpNonCompound_RFC5506));
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
-        tbChannel.videoChannel, rtcpMode));
-    EXPECT_EQ(webrtc::kRtcpNonCompound_RFC5506, rtcpMode);
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
-        tbChannel.videoChannel, webrtc::kRtcpNone));
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
-        tbChannel.videoChannel, rtcpMode));
-    EXPECT_EQ(webrtc::kRtcpNone, rtcpMode);
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
-        tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
-
-    //
-    // CName is testedn in SimpleTest
-    // Start sequence number is tested in SimplTEst
-    //
-    const char* testCName = "ViEAutotestCName";
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPCName(
-        tbChannel.videoChannel, testCName));
-
-    char returnCName[256];
-    memset(returnCName, 0, 256);
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPCName(
-        tbChannel.videoChannel, returnCName));
-    EXPECT_STRCASEEQ(testCName, returnCName);
-
-    //
-    // SSRC
-    //
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(
-        tbChannel.videoChannel, 0x01234567));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(
-        tbChannel.videoChannel, 0x76543210));
-
-    unsigned int ssrc = 0;
-    EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(tbChannel.videoChannel, ssrc));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, 1000));
-    tbChannel.StartSend();
-    EXPECT_NE(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, 12345));
-    tbChannel.StopSend();
-
-    //
-    // Start sequence number
-    //
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, 12345));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, 1000));
-    tbChannel.StartSend();
-    EXPECT_NE(0, ViE.rtp_rtcp->SetStartSequenceNumber(
-        tbChannel.videoChannel, 12345));
-    tbChannel.StopSend();
-
-    //
-    // Application specific RTCP
-    //
-    {
-        unsigned char subType = 3;
-        unsigned int name = static_cast<unsigned int> (0x41424344); // 'ABCD';
-        const char* data = "ViEAutoTest Data of length 32 --";
-        const unsigned short numBytes = 32;
-
-        tbChannel.StartSend();
-        EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-            tbChannel.videoChannel, subType, name, data, numBytes));
-        EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-            tbChannel.videoChannel, subType, name, NULL, numBytes)) <<
-                "Should fail on NULL input.";
-        EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-            tbChannel.videoChannel, subType, name, data, numBytes - 1)) <<
-                "Should fail on incorrect length.";
-
-        EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
-            tbChannel.videoChannel, rtcpMode));
-        EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-            tbChannel.videoChannel, subType, name, data, numBytes));
-        EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
-            tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
-        tbChannel.StopSend();
-        EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-            tbChannel.videoChannel, subType, name, data, numBytes));
-    }
-
-    //
-    // Statistics
-    //
-    // Tested in SimpleTest(), we'll get errors if we haven't received a RTCP
-    // packet.
-
-    //
-    // RTP Dump
-    //
-    {
-        std::string output_file = webrtc::test::OutputPath() +
-            "DumpFileName.rtp";
-        const char* dumpName = output_file.c_str();
-
-        EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
-            tbChannel.videoChannel, dumpName, webrtc::kRtpIncoming));
-        EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
-            tbChannel.videoChannel, webrtc::kRtpIncoming));
-        EXPECT_NE(0, ViE.rtp_rtcp->StopRTPDump(
-            tbChannel.videoChannel, webrtc::kRtpIncoming));
-        EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
-            tbChannel.videoChannel, dumpName, webrtc::kRtpOutgoing));
-        EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
-            tbChannel.videoChannel, webrtc::kRtpOutgoing));
-        EXPECT_NE(0, ViE.rtp_rtcp->StopRTPDump(
-            tbChannel.videoChannel, webrtc::kRtpOutgoing));
-        EXPECT_NE(0, ViE.rtp_rtcp->StartRTPDump(
-            tbChannel.videoChannel, dumpName, (webrtc::RTPDirections) 3));
-    }
-    //
-    // RTP/RTCP Observers
-    //
-    {
-        ViERtpObserver rtpObserver;
-        EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTPObserver(
-            tbChannel.videoChannel, rtpObserver));
-        EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTPObserver(
-            tbChannel.videoChannel, rtpObserver));
-        EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTPObserver(
-            tbChannel.videoChannel));
-        EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTPObserver(
-            tbChannel.videoChannel));
-    }
-    //
-    // PLI
-    //
-    {
-        EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
-            tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
-        EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
-            tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
-        EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
-            tbChannel.videoChannel, webrtc::kViEKeyFrameRequestNone));
-        EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
-            tbChannel.videoChannel, webrtc::kViEKeyFrameRequestNone));
-    }
-    //
-    // NACK
-    //
-    {
-      EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
-    }
-
-    // Timestamp offset extension.
-    // Valid range is 1 to 14 inclusive.
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 0));
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 15));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-            tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-              tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendTimestampOffsetStatus(
-            tbChannel.videoChannel, false, 3));
-
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 0));
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 15));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-            tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-              tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
-            tbChannel.videoChannel, false, 3));
-
-    // Absolute send time extension.
-    // Valid range is 1 to 14 inclusive.
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 0));
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 15));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSendAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, false, 3));
-
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 0));
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 15));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, true, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, false, 3));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveAbsoluteSendTimeStatus(
-        tbChannel.videoChannel, false, 3));
-
-    // Transmission smoothening.
-    const int invalid_channel_id = 17;
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        invalid_channel_id, true));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        tbChannel.videoChannel, true));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        tbChannel.videoChannel, true));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        tbChannel.videoChannel, false));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
-        tbChannel.videoChannel, false));
-
-    // Buffering mode - sender side.
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
-        invalid_channel_id, 0));
-    int invalid_delay = -1;
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
-        tbChannel.videoChannel, invalid_delay));
-    invalid_delay = 15000;
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
-        tbChannel.videoChannel, invalid_delay));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSenderBufferingMode(
-        tbChannel.videoChannel, 5000));
-
-    // Buffering mode - receiver side.
-    // Run without VoE to verify it that does not crash, but return an error.
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        tbChannel.videoChannel, 0));
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        tbChannel.videoChannel, 2000));
-
-    // Set VoE (required to set up stream-sync).
-    webrtc::VoiceEngine* voice_engine = webrtc::VoiceEngine::Create();
-    EXPECT_TRUE(NULL != voice_engine);
-    webrtc::VoEBase* voe_base = webrtc::VoEBase::GetInterface(voice_engine);
-    EXPECT_TRUE(NULL != voe_base);
-    EXPECT_EQ(0, voe_base->Init());
-    int audio_channel = voe_base->CreateChannel();
-    EXPECT_NE(-1, audio_channel);
-    EXPECT_EQ(0, ViE.base->SetVoiceEngine(voice_engine));
-    EXPECT_EQ(0, ViE.base->ConnectAudioChannel(tbChannel.videoChannel,
-                                               audio_channel));
-
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        invalid_channel_id, 0));
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        tbChannel.videoChannel, invalid_delay));
-    invalid_delay = 15000;
-    EXPECT_EQ(-1, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        tbChannel.videoChannel, invalid_delay));
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        tbChannel.videoChannel, 5000));
-
-    // Real-time mode - sender side.
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetSenderBufferingMode(
-        tbChannel.videoChannel, 0));
-    // Real-time mode - receiver side.
-    EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiverBufferingMode(
-        tbChannel.videoChannel, 0));
-
-    EXPECT_EQ(0, ViE.base->DisconnectAudioChannel(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->SetVoiceEngine(NULL));
-    EXPECT_EQ(0, voe_base->DeleteChannel(audio_channel));
-    voe_base->Release();
-    EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine));
-
-    //***************************************************************
-    //  Testing finished. Tear down Video Engine
-    //***************************************************************
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_simulcast.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_simulcast.cc
deleted file mode 100644
index 84d31dee2c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_simulcast.cc
+++ /dev/null
@@ -1,642 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <iostream>  // NOLINT
-
-#include "webrtc/common_types.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-
-enum RelayMode {
-  kRelayOneStream = 1,
-  kRelayAllStreams = 2
-};
-
-#define VCM_RED_PAYLOAD_TYPE        96
-#define VCM_ULPFEC_PAYLOAD_TYPE     97
-
-const int kNumStreams = 3;
-
-void InitialSingleStreamSettings(webrtc::VideoCodec* video_codec) {
-  video_codec->numberOfSimulcastStreams = 0;
-  video_codec->width = 1200;
-  video_codec->height = 800;
-}
-
-void SetSimulcastSettings(webrtc::VideoCodec* video_codec) {
-  video_codec->width = 1280;
-  video_codec->height = 720;
-
-  // Simulcast settings.
-  video_codec->numberOfSimulcastStreams = kNumStreams;
-  video_codec->simulcastStream[0].width = 320;
-  video_codec->simulcastStream[0].height = 180;
-  video_codec->simulcastStream[0].numberOfTemporalLayers = 0;
-  video_codec->simulcastStream[0].maxBitrate = 100;
-  video_codec->simulcastStream[0].targetBitrate = 100;
-  video_codec->simulcastStream[0].minBitrate = 0;
-  video_codec->simulcastStream[0].qpMax = video_codec->qpMax;
-
-  video_codec->simulcastStream[1].width = 640;
-  video_codec->simulcastStream[1].height = 360;
-  video_codec->simulcastStream[1].numberOfTemporalLayers = 0;
-  video_codec->simulcastStream[1].maxBitrate = 500;
-  video_codec->simulcastStream[1].targetBitrate = 500;
-  video_codec->simulcastStream[1].minBitrate = 200;
-  video_codec->simulcastStream[1].qpMax = video_codec->qpMax;
-
-  video_codec->simulcastStream[2].width = 1280;
-  video_codec->simulcastStream[2].height = 720;
-  video_codec->simulcastStream[2].numberOfTemporalLayers = 0;
-  video_codec->simulcastStream[2].maxBitrate = 1200;
-  video_codec->simulcastStream[2].targetBitrate = 1200;
-  video_codec->simulcastStream[2].minBitrate = 900;
-  video_codec->simulcastStream[2].qpMax = video_codec->qpMax;
-}
-
-void RuntimeSingleStreamSettings(webrtc::VideoCodec* video_codec) {
-  SetSimulcastSettings(video_codec);
-  video_codec->width = 1200;
-  video_codec->height = 800;
-  video_codec->numberOfSimulcastStreams = kNumStreams;
-  video_codec->simulcastStream[0].maxBitrate = 0;
-  video_codec->simulcastStream[0].targetBitrate = 0;
-  video_codec->simulcastStream[0].minBitrate = 0;
-  video_codec->simulcastStream[1].maxBitrate = 0;
-  video_codec->simulcastStream[1].targetBitrate = 0;
-  video_codec->simulcastStream[1].minBitrate = 0;
-  video_codec->simulcastStream[2].maxBitrate = 0;
-  video_codec->simulcastStream[2].targetBitrate = 0;
-  video_codec->simulcastStream[2].minBitrate = 0;
-}
-
-int VideoEngineSimulcastTest(void* window1, void* window2) {
-  // *******************************************************
-  //  Begin create/initialize Video Engine for testing
-  // *******************************************************
-
-  int error = 0;
-  int receive_channels[kNumStreams];
-
-  // Create a VideoEngine instance.
-  webrtc::VideoEngine* video_engine = NULL;
-  video_engine = webrtc::VideoEngine::Create();
-  if (video_engine == NULL) {
-    printf("ERROR in VideoEngine::Create\n");
-    return -1;
-  }
-
-  error = video_engine->SetTraceFilter(webrtc::kTraceAll);
-  if (error == -1) {
-    printf("ERROR in VideoEngine::SetTraceLevel\n");
-    return -1;
-  }
-
-  std::string trace_file =
-    ViETest::GetResultOutputPath() + "ViESimulcast_trace.txt";
-  error = video_engine->SetTraceFile(trace_file.c_str());
-  if (error == -1) {
-    printf("ERROR in VideoEngine::SetTraceFile\n");
-    return -1;
-  }
-
-  // Init VideoEngine and create a channel.
-  webrtc::ViEBase* vie_base = webrtc::ViEBase::GetInterface(video_engine);
-  if (vie_base == NULL) {
-    printf("ERROR in ViEBase::GetInterface\n");
-    return -1;
-  }
-
-  error = vie_base->Init();
-  if (error == -1) {
-    printf("ERROR in ViEBase::Init\n");
-    return -1;
-  }
-
-  RelayMode relay_mode = kRelayOneStream;
-  printf("Select relay mode:\n");
-  printf("\t1. Relay one stream\n");
-  printf("\t2. Relay all streams\n");
-  if (scanf("%d", reinterpret_cast<int*>(&relay_mode)) != 1) {
-    printf("Error in scanf()\n");
-    return -1;
-  }
-  getchar();
-
-  webrtc::ViERTP_RTCP* vie_rtp_rtcp =
-      webrtc::ViERTP_RTCP::GetInterface(video_engine);
-  if (vie_rtp_rtcp == NULL) {
-    printf("ERROR in ViERTP_RTCP::GetInterface\n");
-    return -1;
-  }
-
-  int video_channel = -1;
-  error = vie_base->CreateChannel(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::CreateChannel\n");
-    return -1;
-  }
-
-  for (int i = 0; i < kNumStreams; ++i) {
-    receive_channels[i] = -1;
-    error = vie_base->CreateReceiveChannel(receive_channels[i], video_channel);
-    if (error == -1) {
-      printf("ERROR in ViEBase::CreateChannel\n");
-      return -1;
-    }
-  }
-
-  // List available capture devices, allocate and connect.
-  webrtc::ViECapture* vie_capture =
-      webrtc::ViECapture::GetInterface(video_engine);
-  if (vie_base == NULL) {
-    printf("ERROR in ViECapture::GetInterface\n");
-    return -1;
-  }
-
-  const unsigned int KMaxDeviceNameLength = 128;
-  const unsigned int KMaxUniqueIdLength = 256;
-  char device_name[KMaxDeviceNameLength];
-  memset(device_name, 0, KMaxDeviceNameLength);
-  char unique_id[KMaxUniqueIdLength];
-  memset(unique_id, 0, KMaxUniqueIdLength);
-
-  printf("Available capture devices:\n");
-  int capture_idx = 0;
-  for (capture_idx = 0; capture_idx < vie_capture->NumberOfCaptureDevices();
-       capture_idx++) {
-    memset(device_name, 0, KMaxDeviceNameLength);
-    memset(unique_id, 0, KMaxUniqueIdLength);
-
-    error = vie_capture->GetCaptureDevice(capture_idx, device_name,
-                                          KMaxDeviceNameLength, unique_id,
-                                          KMaxUniqueIdLength);
-    if (error == -1) {
-      printf("ERROR in ViECapture::GetCaptureDevice\n");
-      return -1;
-    }
-    printf("\t %d. %s\n", capture_idx + 1, device_name);
-  }
-  printf("\nChoose capture device: ");
-#ifdef WEBRTC_ANDROID
-  capture_idx = 0;
-  printf("0\n");
-#else
-  if (scanf("%d", &capture_idx) != 1) {
-    printf("Error in scanf()\n");
-    return -1;
-  }
-  getchar();
-  // Compensate for idx start at 1.
-  capture_idx = capture_idx - 1;
-#endif
-  error = vie_capture->GetCaptureDevice(capture_idx, device_name,
-                                        KMaxDeviceNameLength, unique_id,
-                                        KMaxUniqueIdLength);
-  if (error == -1) {
-    printf("ERROR in ViECapture::GetCaptureDevice\n");
-    return -1;
-  }
-
-  int capture_id = 0;
-  error = vie_capture->AllocateCaptureDevice(unique_id, KMaxUniqueIdLength,
-                                             capture_id);
-  if (error == -1) {
-    printf("ERROR in ViECapture::AllocateCaptureDevice\n");
-    return -1;
-  }
-
-  error = vie_capture->ConnectCaptureDevice(capture_id, video_channel);
-  if (error == -1) {
-    printf("ERROR in ViECapture::ConnectCaptureDevice\n");
-    return -1;
-  }
-
-  error = vie_capture->StartCapture(capture_id);
-  if (error == -1) {
-    printf("ERROR in ViECapture::StartCapture\n");
-    return -1;
-  }
-
-  // RTP/RTCP settings.
-  error = vie_rtp_rtcp->SetRTCPStatus(video_channel,
-                                      webrtc::kRtcpCompound_RFC4585);
-  if (error == -1) {
-    printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
-    return -1;
-  }
-
-  vie_rtp_rtcp->SetRembStatus(video_channel, true, false);
-  if (error == -1) {
-    printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
-    return -1;
-  }
-
-  error = vie_rtp_rtcp->SetKeyFrameRequestMethod(
-            video_channel, webrtc::kViEKeyFrameRequestPliRtcp);
-  if (error == -1) {
-    printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
-    return -1;
-  }
-
-  for (int i = 0; i < kNumStreams; ++i) {
-    error = vie_rtp_rtcp->SetRTCPStatus(receive_channels[i],
-                                        webrtc::kRtcpCompound_RFC4585);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
-      return -1;
-    }
-
-    vie_rtp_rtcp->SetRembStatus(receive_channels[i], false, true);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
-      return -1;
-    }
-
-    error = vie_rtp_rtcp->SetKeyFrameRequestMethod(
-        receive_channels[i], webrtc::kViEKeyFrameRequestPliRtcp);
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
-      return -1;
-    }
-  }
-
-  // Set up rendering.
-  webrtc::ViERender* vie_render = webrtc::ViERender::GetInterface(video_engine);
-  if (vie_render == NULL) {
-    printf("ERROR in ViERender::GetInterface\n");
-    return -1;
-  }
-
-  error = vie_render->AddRenderer(capture_id, window1, 0, 0.0, 0.0, 1.0, 1.0);
-  if (error == -1) {
-    printf("ERROR in ViERender::AddRenderer\n");
-    return -1;
-  }
-
-  error = vie_render->StartRender(capture_id);
-  if (error == -1) {
-    printf("ERROR in ViERender::StartRender\n");
-    return -1;
-  }
-
-  // Only rendering the thumbnail.
-  int channel_to_render = video_channel;
-  if (relay_mode == kRelayAllStreams) {
-    channel_to_render = receive_channels[0];
-  }
-  error = vie_render->AddRenderer(channel_to_render, window2, 1, 0.0, 0.0, 1.0,
-                                  1.0);
-  if (error == -1) {
-    printf("ERROR in ViERender::AddRenderer\n");
-    return -1;
-  }
-
-  error = vie_render->StartRender(channel_to_render);
-  if (error == -1) {
-    printf("ERROR in ViERender::StartRender\n");
-    return -1;
-  }
-
-  // Setup codecs.
-  webrtc::ViECodec* vie_codec = webrtc::ViECodec::GetInterface(video_engine);
-  if (vie_codec == NULL) {
-    printf("ERROR in ViECodec::GetInterface\n");
-    return -1;
-  }
-
-  // Check available codecs and prepare receive codecs.
-  printf("\nAvailable codecs:\n");
-  webrtc::VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-  int codec_idx = 0;
-  for (codec_idx = 0; codec_idx < vie_codec->NumberOfCodecs(); codec_idx++) {
-    error = vie_codec->GetCodec(codec_idx, video_codec);
-    if (error == -1) {
-      printf("ERROR in ViECodec::GetCodec\n");
-      return -1;
-    }
-    // Try to keep the test frame size small when I420.
-    if (video_codec.codecType != webrtc::kVideoCodecVP8) {
-      continue;
-    }
-    for (int i = 0; i < kNumStreams; ++i) {
-      error = vie_codec->SetReceiveCodec(receive_channels[i], video_codec);
-      if (error == -1) {
-        printf("ERROR in ViECodec::SetReceiveCodec\n");
-        return -1;
-      }
-    }
-    if (video_codec.codecType != webrtc::kVideoCodecRED &&
-        video_codec.codecType != webrtc::kVideoCodecULPFEC) {
-      printf("\t %d. %s\n", codec_idx + 1, video_codec.plName);
-    }
-    break;
-  }
-  error = vie_codec->GetCodec(codec_idx, video_codec);
-  if (error == -1) {
-    printf("ERROR in ViECodec::GetCodec\n");
-    return -1;
-  }
-
-  bool simulcast_mode = true;
-  int num_streams = 1;
-  // Set spatial resolution option.
-  if (simulcast_mode) {
-    SetSimulcastSettings(&video_codec);
-    num_streams = video_codec.numberOfSimulcastStreams;
-  } else {
-    InitialSingleStreamSettings(&video_codec);
-    num_streams = 1;
-  }
-
-  // Set start bit rate.
-  std::string str;
-  std::cout << std::endl;
-  std::cout << "Choose start rate (in kbps). Press enter for default:  ";
-  std::getline(std::cin, str);
-  int start_rate = atoi(str.c_str());
-  if (start_rate != 0) {
-    video_codec.startBitrate = start_rate;
-  }
-
-  error = vie_codec->SetSendCodec(video_channel, video_codec);
-  if (error == -1) {
-    printf("ERROR in ViECodec::SetSendCodec\n");
-    return -1;
-  }
-
-  // Address settings.
-  webrtc::ViENetwork* vie_network =
-      webrtc::ViENetwork::GetInterface(video_engine);
-  if (vie_network == NULL) {
-    printf("ERROR in ViENetwork::GetInterface\n");
-    return -1;
-  }
-
-  TbExternalTransport::SsrcChannelMap ssrc_channel_map;
-  for (int idx = 0; idx < num_streams; idx++) {
-    error = vie_rtp_rtcp->SetLocalSSRC(video_channel, idx + 1,  // SSRC
-                                       webrtc::kViEStreamTypeNormal, idx);
-    ssrc_channel_map[idx + 1] = receive_channels[idx];
-    if (error == -1) {
-      printf("ERROR in ViERTP_RTCP::SetLocalSSRC(idx:%d)\n",
-             idx);
-      return -1;
-    }
-  }
-
-  TbExternalTransport::SsrcChannelMap* channel_map = &ssrc_channel_map;
-  if (relay_mode == kRelayOneStream) {
-    channel_map = NULL;
-  }
-
-  // Setting External transport.
-  TbExternalTransport ext_transport(*vie_network, video_channel, channel_map);
-
-  error = vie_network->RegisterSendTransport(video_channel, ext_transport);
-  if (error == -1) {
-    printf("ERROR in ViECodec::RegisterSendTransport \n");
-    return -1;
-  }
-
-  for (int i = 0; i < kNumStreams; ++i) {
-    error = vie_network->RegisterSendTransport(receive_channels[i],
-                                               ext_transport);
-    if (error == -1) {
-      printf("ERROR in ViECodec::RegisterSendTransport \n");
-      return -1;
-    }
-  }
-
-  // Set network one-way delay value.
-  // 10 ms one-way delay.
-  NetworkParameters network;
-  network.loss_model = kUniformLoss;
-  network.mean_one_way_delay = 10;
-  ext_transport.SetNetworkParameters(network);
-
-  if (relay_mode == kRelayOneStream) {
-    ext_transport.SetSSRCFilter(num_streams);
-  }
-
-  error = vie_base->StartSend(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViENetwork::StartSend\n");
-    return -1;
-  }
-  error = vie_base->StartReceive(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViENetwork::StartReceive\n");
-    return -1;
-  }
-
-  for (int i = 0; i < kNumStreams; ++i) {
-    error = vie_base->StartReceive(receive_channels[i]);
-    if (error == -1) {
-      printf("ERROR in ViENetwork::StartReceive\n");
-      return -1;
-    }
-  }
-
-  // Create a receive channel to verify that it doesn't mess up toggling
-  // between single stream and simulcast.
-  int video_channel2 = -1;
-  error = vie_base->CreateReceiveChannel(video_channel2, video_channel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::CreateReceiveChannel\n");
-    return -1;
-  }
-
-  // *******************************************************
-  //  Engine started
-  // *******************************************************
-
-  printf("\nSimulcast call started\n\n");
-  do {
-    printf("Enter new SSRC filter 1,2 or 3\n");
-    printf("... or 0 to switch between simulcast and a single stream\n");
-    printf("Press enter to stop...");
-    str.clear();
-    std::getline(std::cin, str);
-    if (!str.empty()) {
-      int ssrc = atoi(str.c_str());
-      if (ssrc == 0) {
-        // Toggle between simulcast and a single stream with different
-        // resolution.
-        if (simulcast_mode) {
-          RuntimeSingleStreamSettings(&video_codec);
-          num_streams = 1;
-          printf("Disabling simulcast\n");
-        } else {
-          SetSimulcastSettings(&video_codec);
-          num_streams = video_codec.numberOfSimulcastStreams;
-          printf("Enabling simulcast\n");
-        }
-        simulcast_mode = !simulcast_mode;
-        if (vie_codec->SetSendCodec(video_channel, video_codec) != 0) {
-          printf("ERROR switching between simulcast and single stream\n");
-          return -1;
-        }
-        for (int idx = 0; idx < num_streams; idx++) {
-          error = vie_rtp_rtcp->SetLocalSSRC(video_channel, idx + 1,  // SSRC
-                                             webrtc::kViEStreamTypeNormal, idx);
-          if (error == -1) {
-            printf("ERROR in ViERTP_RTCP::SetLocalSSRC(idx:%d)\n", idx);
-            return -1;
-          }
-        }
-        if (relay_mode == kRelayOneStream) {
-          ext_transport.SetSSRCFilter(num_streams);
-        }
-      } else if (ssrc > 0 && ssrc < 4) {
-        if (relay_mode == kRelayOneStream) {
-          ext_transport.SetSSRCFilter(ssrc);
-        }
-      } else {
-        printf("Invalid SSRC\n");
-      }
-    } else {
-      break;
-    }
-  } while (true);
-
-  // *******************************************************
-  //  Testing finished. Tear down Video Engine
-  // *******************************************************
-  error = vie_base->DeleteChannel(video_channel2);
-  if (error == -1) {
-    printf("ERROR in ViEBase::DeleteChannel\n");
-    return -1;
-  }
-
-  for (int i = 0; i < kNumStreams; ++i) {
-    error = vie_base->StopReceive(receive_channels[i]);
-    if (error == -1) {
-      printf("ERROR in ViEBase::StopReceive\n");
-      return -1;
-    }
-  }
-
-  error = vie_base->StopReceive(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::StopReceive\n");
-    return -1;
-  }
-
-  error = vie_base->StopSend(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::StopSend\n");
-    return -1;
-  }
-
-  error = vie_render->StopRender(capture_id);
-  if (error == -1) {
-    printf("ERROR in ViERender::StopRender\n");
-    return -1;
-  }
-
-  error = vie_render->RemoveRenderer(capture_id);
-  if (error == -1) {
-    printf("ERROR in ViERender::RemoveRenderer\n");
-    return -1;
-  }
-
-  error = vie_render->StopRender(channel_to_render);
-  if (error == -1) {
-    printf("ERROR in ViERender::StopRender\n");
-    return -1;
-  }
-
-  error = vie_render->RemoveRenderer(channel_to_render);
-  if (error == -1) {
-    printf("ERROR in ViERender::RemoveRenderer\n");
-    return -1;
-  }
-
-  error = vie_capture->StopCapture(capture_id);
-  if (error == -1) {
-    printf("ERROR in ViECapture::StopCapture\n");
-    return -1;
-  }
-
-  error = vie_capture->DisconnectCaptureDevice(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
-    return -1;
-  }
-
-  error = vie_capture->ReleaseCaptureDevice(capture_id);
-  if (error == -1) {
-    printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
-    return -1;
-  }
-
-  for (int i = 0; i < kNumStreams; ++i) {
-    error = vie_base->DeleteChannel(receive_channels[i]);
-    if (error == -1) {
-      printf("ERROR in ViEBase::DeleteChannel\n");
-      return -1;
-    }
-  }
-
-  error = vie_base->DeleteChannel(video_channel);
-  if (error == -1) {
-    printf("ERROR in ViEBase::DeleteChannel\n");
-    return -1;
-  }
-
-  int remaining_interfaces = 0;
-  remaining_interfaces = vie_codec->Release();
-  remaining_interfaces += vie_capture->Release();
-  remaining_interfaces += vie_rtp_rtcp->Release();
-  remaining_interfaces += vie_render->Release();
-  remaining_interfaces += vie_network->Release();
-  remaining_interfaces += vie_base->Release();
-  if (remaining_interfaces > 0) {
-    printf("ERROR: Could not release all interfaces\n");
-    return -1;
-  }
-
-  bool deleted = webrtc::VideoEngine::Delete(video_engine);
-  if (deleted == false) {
-    printf("ERROR in VideoEngine::Delete\n");
-    return -1;
-  }
-  return 0;
-}
-
-int ViEAutoTest::ViESimulcastCall() {
-  ViETest::Log(" ");
-  ViETest::Log("========================================");
-  ViETest::Log(" ViE Autotest Simulcast Call\n");
-
-  if (VideoEngineSimulcastTest(_window1, _window2) == 0) {
-    ViETest::Log(" ");
-    ViETest::Log(" ViE Autotest Simulcast Call Done");
-    ViETest::Log("========================================");
-    ViETest::Log(" ");
-
-    return 0;
-  }
-  ViETest::Log(" ");
-  ViETest::Log(" ViE Autotest Simulcast Call Failed");
-  ViETest::Log("========================================");
-  ViETest::Log(" ");
-  return 1;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc
deleted file mode 100644
index 6e2bac3f9f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc
+++ /dev/null
@@ -1,204 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_windows.cc
-//
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-
-#include <windows.h>
-
-#ifdef _DEBUG
-//#include "vld.h"
-#endif
-
-// Disable Visual studio warnings
-// 'this' : used in base member initializer list
-#pragma warning(disable: 4355)
-
-LRESULT CALLBACK ViEAutoTestWinProc(HWND hWnd, UINT uMsg, WPARAM wParam,
-                                    LPARAM lParam) {
-  switch (uMsg) {
-    case WM_DESTROY:
-      PostQuitMessage( WM_QUIT);
-      break;
-    case WM_COMMAND:
-      break;
-  }
-  return DefWindowProc(hWnd, uMsg, wParam, lParam);
-}
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager()
-    : _window1(NULL),
-      _window2(NULL),
-      _terminate(false),
-      _eventThread(webrtc::ThreadWrapper::CreateThread(
-          EventProcess, this, "ViEAutotestEventThread")),
-      _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-      _hwnd1(NULL),
-      _hwnd2(NULL),
-      _hwnd1Size(),
-      _hwnd2Size(),
-      _hwnd1Title(),
-      _hwnd2Title() {
-}
-
-ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
-  if (_hwnd1) {
-    ViEDestroyWindow(_hwnd1);
-  }
-  if (_hwnd2) {
-    ViEDestroyWindow(_hwnd2);
-  }
-  delete &_crit;
-}
-
-void* ViEAutoTestWindowManager::GetWindow1() {
-  return _window1;
-}
-
-void* ViEAutoTestWindowManager::GetWindow2() {
-  return _window2;
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
-                                            AutoTestRect window2Size,
-                                            void* window1Title,
-                                            void* window2Title) {
-  _hwnd1Size.Copy(window1Size);
-  _hwnd2Size.Copy(window2Size);
-  memcpy(_hwnd1Title, window1Title, TITLE_LENGTH);
-  memcpy(_hwnd2Title, window2Title, TITLE_LENGTH);
-
-  _eventThread->Start();
-
-  do {
-    _crit.Enter();
-    if (_window1 != NULL) {
-      break;
-    }
-    _crit.Leave();
-    AutoTestSleep(10);
-  } while (true);
-  _crit.Leave();
-  return 0;
-}
-
-int ViEAutoTestWindowManager::TerminateWindows() {
-  _terminate = true;
-  _eventThread->Stop();
-  _crit.Enter();
-  _eventThread.reset();
-  _crit.Leave();
-
-  return 0;
-}
-
-bool ViEAutoTestWindowManager::EventProcess(void* obj) {
-  return static_cast<ViEAutoTestWindowManager*> (obj)->EventLoop();
-}
-
-bool ViEAutoTestWindowManager::EventLoop() {
-  _crit.Enter();
-
-  ViECreateWindow(_hwnd1, _hwnd1Size.origin.x, _hwnd1Size.origin.y,
-                  _hwnd1Size.size.width, _hwnd1Size.size.height, _hwnd1Title);
-  ViECreateWindow(_hwnd2, _hwnd2Size.origin.x, _hwnd2Size.origin.y,
-                  _hwnd2Size.size.width, _hwnd2Size.size.height, _hwnd2Title);
-
-  _window1 = (void*) _hwnd1;
-  _window2 = (void*) _hwnd2;
-  MSG msg;
-  while (!_terminate) {
-    if (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
-      TranslateMessage(&msg);
-      DispatchMessage(&msg);
-    }
-    _crit.Leave();
-    AutoTestSleep(10);
-    _crit.Enter();
-  }
-  ViEDestroyWindow(_hwnd1);
-  ViEDestroyWindow(_hwnd2);
-  _crit.Leave();
-
-  return false;
-}
-
-int ViEAutoTestWindowManager::ViECreateWindow(HWND &hwndMain, int xPos,
-                                              int yPos, int width, int height,
-                                              TCHAR* className) {
-  HINSTANCE hinst = GetModuleHandle(0);
-  WNDCLASSEX wcx;
-  wcx.hInstance = hinst;
-  wcx.lpszClassName = className;
-  wcx.lpfnWndProc = (WNDPROC) ViEAutoTestWinProc;
-  wcx.style = CS_DBLCLKS;
-  wcx.hIcon = LoadIcon(NULL, IDI_APPLICATION);
-  wcx.hIconSm = LoadIcon(NULL, IDI_APPLICATION);
-  wcx.hCursor = LoadCursor(NULL, IDC_ARROW);
-  wcx.lpszMenuName = NULL;
-  wcx.cbSize = sizeof(WNDCLASSEX);
-  wcx.cbClsExtra = 0;
-  wcx.cbWndExtra = 0;
-  wcx.hbrBackground = GetSysColorBrush(COLOR_3DFACE);
-
-  RegisterClassEx(&wcx);
-
-  // Create the main window.
-  hwndMain = CreateWindowEx(0,          // no extended styles
-                            className,  // class name
-                            className,  // window name
-                            WS_OVERLAPPED | WS_THICKFRAME,  // overlapped window
-                            xPos,    // horizontal position
-                            yPos,    // vertical position
-                            width,   // width
-                            height,  // height
-                            (HWND) NULL,   // no parent or owner window
-                            (HMENU) NULL,  // class menu used
-                            hinst,  // instance handle
-                            NULL);  // no window creation data
-
-  if (!hwndMain)
-    return -1;
-
-  // Show the window using the flag specified by the program
-  // that started the application, and send the application
-  // a WM_PAINT message.
-  ShowWindow(hwndMain, SW_SHOWDEFAULT);
-  UpdateWindow(hwndMain);
-
-  ::SetWindowPos(hwndMain, HWND_TOP, xPos, yPos, width, height,
-                 SWP_FRAMECHANGED);
-
-  return 0;
-}
-
-int ViEAutoTestWindowManager::ViEDestroyWindow(HWND& hwnd) {
-  ::DestroyWindow(hwnd);
-  return 0;
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow() {
-  // Meant to put terminal window on top
-  return true;
-}
-
-int main(int argc, char* argv[]) {
-  ViEAutoTestMain auto_test;
-  return auto_test.RunTests(argc, argv);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc
deleted file mode 100644
index 4f3a90a7a5..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "webrtc/video_engine/test/auto_test/primitives/base_primitives.h"
-#include "webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h"
-#include "webrtc/video_engine/test/auto_test/primitives/general_primitives.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_fake_camera.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-bool ViEFileBasedComparisonTests::TestCallSetup(
-    const std::string& i420_video_file,
-    int width,
-    int height,
-    ViEToFileRenderer* local_file_renderer,
-    ViEToFileRenderer* remote_file_renderer) {
-
-  TbInterfaces interfaces("TestCallSetup");
-
-  int video_channel = -1;
-  EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
-
-  ViEFakeCamera fake_camera(interfaces.capture);
-  if (!fake_camera.StartCameraInNewThread(i420_video_file,
-                                          width,
-                                          height)) {
-    // No point in continuing if we have no proper video source
-    ADD_FAILURE() << "Could not open input video " << i420_video_file <<
-        ": aborting test...";
-    return false;
-  }
-  int capture_id = fake_camera.capture_id();
-
-  // Apparently, we need to connect external capture devices, but we should
-  // not start them since the external device is not a proper device.
-  EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
-      capture_id, video_channel));
-
-  ConfigureRtpRtcp(interfaces.rtp_rtcp, kNack, video_channel);
-
-  webrtc::ViERender* render_interface = interfaces.render;
-  webrtc::ViEImageProcess* image_process = interfaces.image_process;
-
-  RenderToFile(render_interface, video_channel, remote_file_renderer);
-
-  // We make a special hookup of the local renderer to use an effect filter
-  // instead of using the render interface for the capture device. This way
-  // we will only render frames that actually get sent.
-  webrtc::ExternalRendererEffectFilter renderer_filter(local_file_renderer);
-  EXPECT_EQ(0, image_process->RegisterSendEffectFilter(video_channel,
-                                                       renderer_filter));
-
-  // Run the test itself:
-  const char* device_name = "Fake Capture Device";
-
-  ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
-                      interfaces.base, interfaces.network, interfaces.rtp_rtcp,
-                      video_channel, device_name);
-
-  EXPECT_EQ(0, render_interface->StopRender(video_channel));
-  EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
-
-  interfaces.capture->DisconnectCaptureDevice(video_channel);
-
-  // Stop sending data, clean up the camera thread and release the capture
-  // device. Note that this all happens after StopEverything, so this
-  // tests that the system doesn't mind that the external capture device sends
-  // data after rendering has been stopped.
-  fake_camera.StopCamera();
-  EXPECT_EQ(0, image_process->DeregisterSendEffectFilter(video_channel));
-
-  EXPECT_EQ(0, interfaces.base->DeleteChannel(video_channel));
-  return true;
-}
-
-void ViEFileBasedComparisonTests::TestFullStack(
-    const std::string& i420_video_file,
-    int width,
-    int height,
-    int bit_rate_kbps,
-    ProtectionMethod protection_method,
-    const NetworkParameters& network,
-    ViEToFileRenderer* local_file_renderer,
-    ViEToFileRenderer* remote_file_renderer,
-    FrameDropDetector* frame_drop_detector) {
-  TbInterfaces interfaces("TestFullStack");
-
-  // Setup camera capturing from file.
-  ViEFakeCamera fake_camera(interfaces.capture);
-  if (!fake_camera.StartCameraInNewThread(i420_video_file, width, height)) {
-    // No point in continuing if we have no proper video source
-    ADD_FAILURE() << "Could not open input video " << i420_video_file <<
-        ": aborting test...";
-    return;
-  }
-  int video_channel = -1;
-  int capture_id = fake_camera.capture_id();
-  EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
-
-  // Must set SSRC to avoid SSRC collision detection since we're sending and
-  // receiving from the same machine (that would cause frames being discarded
-  // and decoder reset).
-  EXPECT_EQ(0, interfaces.rtp_rtcp->SetLocalSSRC(video_channel, 12345));
-
-  EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
-      capture_id, video_channel));
-  ConfigureRtpRtcp(interfaces.rtp_rtcp, protection_method, video_channel);
-
-  ::TestFullStack(interfaces, capture_id, video_channel, width, height,
-                  bit_rate_kbps, network, frame_drop_detector,
-                  remote_file_renderer, local_file_renderer);
-  EXPECT_TRUE(fake_camera.StopCamera());
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_creator.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_creator.cc
deleted file mode 100644
index 55bdb4a88c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_creator.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_creator.h"
-
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-
-#if defined(WIN32)
-#include <tchar.h>
-#endif
-
-ViEWindowCreator::ViEWindowCreator() {
-#ifndef WEBRTC_ANDROID
-  window_manager_ =
-      ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform();
-#endif
-}
-
-ViEWindowCreator::~ViEWindowCreator() {
-  delete window_manager_;
-}
-
-ViEAutoTestWindowManagerInterface*
-  ViEWindowCreator::CreateTwoWindows() {
-#if defined(WIN32)
-  TCHAR window1Title[1024] = _T("ViE Autotest Window 1");
-  TCHAR window2Title[1024] = _T("ViE Autotest Window 2");
-#else
-  char window1Title[1024] = "ViE Autotest Window 1";
-  char window2Title[1024] = "ViE Autotest Window 2";
-#endif
-
-  AutoTestRect window1Size(352, 288, 600, 100);
-  AutoTestRect window2Size(352, 288, 1000, 100);
-  window_manager_->CreateWindows(window1Size, window2Size, window1Title,
-                                 window2Title);
-  window_manager_->SetTopmostWindow();
-
-  return window_manager_;
-}
-
-void ViEWindowCreator::TerminateWindows() {
-  window_manager_->TerminateWindows();
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc
deleted file mode 100644
index 88cc239c08..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_linux.h"
-
-ViEAutoTestWindowManagerInterface*
-ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
-  return new ViEAutoTestWindowManager();
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm
deleted file mode 100644
index a60fc900c3..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h"
-
-#include "webrtc/engine_configurations.h"
-#if defined(COCOA_RENDERING)
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h"
-#endif
-
-ViEAutoTestWindowManagerInterface*
-ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
-  return new ViEAutoTestWindowManager();
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc
deleted file mode 100644
index 020ef90317..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#include "webrtc/video_engine/test/auto_test/interface/vie_window_manager_factory.h"
-
-#include "webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h"
-
-ViEAutoTestWindowManagerInterface*
-ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
-  return new ViEAutoTestWindowManager();
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi
deleted file mode 100644
index e52b301651..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi
+++ /dev/null
@@ -1,149 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'vie_auto_test',
-      'type': 'executable',
-      'dependencies': [
-        '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
-        '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
-        '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
-        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-        '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-        '<(webrtc_root)/test/metrics.gyp:metrics',
-        '<(webrtc_root)/test/test.gyp:channel_transport',
-        '<(webrtc_root)/test/test.gyp:test_support',
-        '<(webrtc_root)/test/test.gyp:field_trial',
-        'video_engine_core',
-        'libvietest',
-      ],
-      'sources': [
-        'interface/vie_autotest.h',
-        'interface/vie_autotest_defines.h',
-        'interface/vie_autotest_linux.h',
-        'interface/vie_autotest_mac_cocoa.h',
-        'interface/vie_autotest_main.h',
-        'interface/vie_autotest_window_manager_interface.h',
-        'interface/vie_autotest_windows.h',
-        'interface/vie_file_based_comparison_tests.h',
-        'interface/vie_window_manager_factory.h',
-        'interface/vie_window_creator.h',
-
-        # New, fully automated tests
-        'automated/legacy_fixture.cc',
-        'automated/two_windows_fixture.cc',
-        'automated/vie_api_integration_test.cc',
-        'automated/vie_extended_integration_test.cc',
-        'automated/vie_network_test.cc',
-        'automated/vie_standard_integration_test.cc',
-        'automated/vie_video_verification_test.cc',
-
-        # Test primitives
-        'primitives/base_primitives.cc',
-        'primitives/base_primitives.h',
-        'primitives/choice_helpers.cc',
-        'primitives/choice_helpers.h',
-        'primitives/choice_helpers_unittest.cc',
-        'primitives/fake_stdin.h',
-        'primitives/fake_stdin.cc',
-        'primitives/framedrop_primitives.h',
-        'primitives/framedrop_primitives.cc',
-        'primitives/framedrop_primitives_unittest.cc',
-        'primitives/general_primitives.cc',
-        'primitives/general_primitives.h',
-        'primitives/input_helpers.cc',
-        'primitives/input_helpers.h',
-        'primitives/input_helpers_unittest.cc',
-
-        # Platform independent
-        'source/vie_autotest.cc',
-        'source/vie_autotest_base.cc',
-        'source/vie_autotest_capture.cc',
-        'source/vie_autotest_codec.cc',
-        'source/vie_autotest_image_process.cc',
-        'source/vie_autotest_loopback.cc',
-        'source/vie_autotest_main.cc',
-        'source/vie_autotest_render.cc',
-        'source/vie_autotest_record.cc',
-        'source/vie_autotest_rtp_rtcp.cc',
-        'source/vie_autotest_custom_call.cc',
-        'source/vie_autotest_simulcast.cc',
-        'source/vie_file_based_comparison_tests.cc',
-        'source/vie_window_creator.cc',
-
-        # Platform dependent
-        # Android
-        'source/vie_autotest_android.cc',
-        # Linux
-        'source/vie_autotest_linux.cc',
-        'source/vie_window_manager_factory_linux.cc',
-        # Mac
-        'source/vie_autotest_cocoa_mac.mm',
-        'source/vie_window_manager_factory_mac.mm',
-        # Windows
-        'source/vie_autotest_win.cc',
-        'source/vie_window_manager_factory_win.cc',
-      ],
-      'conditions': [
-        ['OS=="android"', {
-          'libraries': [
-            '-lGLESv2',
-            '-llog',
-          ],
-        }],
-        ['OS=="linux"', {
-          # TODO(andrew): These should be provided directly by the projects
-          #               which require them instead.
-          'libraries': [
-            '-lXext',
-            '-lX11',
-          ],
-        }],
-        ['OS=="mac"', {
-          'dependencies': [
-            # Use a special main for mac so we can access the webcam.
-            '<(webrtc_root)/test/test.gyp:test_support_main_threaded_mac',
-          ],
-          'xcode_settings': {
-            'OTHER_LDFLAGS': [
-              '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL -framework CoreVideo -framework CoreAudio -framework AudioToolbox',
-            ],
-          },
-        }],
-      ], # conditions
-      # Disable warnings to enable Win64 build, issue 1323.
-      'msvs_disabled_warnings': [
-        4267,  # size_t to int truncation.
-      ],
-    },
-  ],
-  'conditions': [
-    ['test_isolation_mode != "noop"', {
-      'targets': [
-        {
-          'target_name': 'vie_auto_test_run',
-          'type': 'none',
-          'dependencies': [
-            'vie_auto_test',
-          ],
-          'includes': [
-            '../../../build/isolate.gypi',
-          ],
-          'sources': [
-            'vie_auto_test.isolate',
-          ],
-        },
-      ],
-    }],
-  ],
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/OWNERS b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/OWNERS
deleted file mode 100644
index 3ee6b4bf5f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc
deleted file mode 100644
index 1c18d498e0..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#include "webrtc/video_engine/test/libvietest/include/vie_fake_camera.h"
-
-#include <assert.h>
-
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/test/libvietest/include/vie_file_capture_device.h"
-
-// This callback runs the camera thread:
-bool StreamVideoFileRepeatedlyIntoCaptureDevice(void* data) {
-  ViEFileCaptureDevice* file_capture_device =
-      reinterpret_cast<ViEFileCaptureDevice*>(data);
-
-  // We want to interrupt the camera feeding thread every now and then in order
-  // to follow the contract for the system_wrappers thread library. 1.5 seconds
-  // seems about right here.
-  uint64_t time_slice_ms = 1500;
-  uint32_t max_fps = 30;
-
-  file_capture_device->ReadFileFor(time_slice_ms, max_fps);
-
-  return true;
-}
-
-ViEFakeCamera::ViEFakeCamera(webrtc::ViECapture* capture_interface)
-    : capture_interface_(capture_interface),
-      capture_id_(-1),
-      file_capture_device_(NULL) {
-}
-
-ViEFakeCamera::~ViEFakeCamera() {
-}
-
-bool ViEFakeCamera::StartCameraInNewThread(
-    const std::string& i420_test_video_path, int width, int height) {
-
-  assert(file_capture_device_ == NULL && camera_thread_ == NULL);
-
-  webrtc::ViEExternalCapture* externalCapture;
-  int result = capture_interface_->
-      AllocateExternalCaptureDevice(capture_id_, externalCapture);
-  if (result != 0) {
-    return false;
-  }
-
-  file_capture_device_ = new ViEFileCaptureDevice(externalCapture);
-  if (!file_capture_device_->OpenI420File(i420_test_video_path,
-                                          width,
-                                          height)) {
-    return false;
-  }
-
-  // Set up a thread which runs the fake camera. The capturer object is
-  // thread-safe.
-  camera_thread_ = webrtc::ThreadWrapper::CreateThread(
-      StreamVideoFileRepeatedlyIntoCaptureDevice, file_capture_device_,
-      "StreamVideoFileRepeatedlyIntoCaptureDevice");
-  camera_thread_->Start();
-
-  return true;
-}
-
-bool ViEFakeCamera::StopCamera() {
-  assert(file_capture_device_ != NULL && camera_thread_ != NULL);
-
-  camera_thread_->Stop();
-  file_capture_device_->CloseFile();
-
-  int result = capture_interface_->ReleaseCaptureDevice(capture_id_);
-
-  camera_thread_.reset();
-  delete file_capture_device_;
-  camera_thread_ = NULL;
-  file_capture_device_ = NULL;
-
-  return result == 0;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc
deleted file mode 100644
index 4f2db75498..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#include "webrtc/video_engine/test/libvietest/include/vie_file_capture_device.h"
-
-#include <assert.h>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-
-// This class ensures we are not exceeding the max FPS.
-class FramePacemaker {
- public:
-  explicit FramePacemaker(uint32_t max_fps)
-      : time_per_frame_ms_(1000 / max_fps) {
-    frame_start_ = webrtc::TickTime::MillisecondTimestamp();
-  }
-
-  void SleepIfNecessary(webrtc::EventWrapper* sleeper) {
-    uint64_t now = webrtc::TickTime::MillisecondTimestamp();
-    if (now - frame_start_ < time_per_frame_ms_) {
-      sleeper->Wait(time_per_frame_ms_ - (now - frame_start_));
-    }
-  }
-
- private:
-  uint64_t frame_start_;
-  uint64_t time_per_frame_ms_;
-};
-
-ViEFileCaptureDevice::ViEFileCaptureDevice(
-    webrtc::ViEExternalCapture* input_sink)
-    : input_sink_(input_sink),
-      input_file_(NULL) {
-  mutex_ = webrtc::CriticalSectionWrapper::CreateCriticalSection();
-}
-
-ViEFileCaptureDevice::~ViEFileCaptureDevice() {
-  delete mutex_;
-}
-
-bool ViEFileCaptureDevice::OpenI420File(const std::string& path,
-                                        int width,
-                                        int height) {
-  webrtc::CriticalSectionScoped cs(mutex_);
-  assert(input_file_ == NULL);
-
-  input_file_ = fopen(path.c_str(), "rb");
-  if (input_file_ == NULL) {
-    return false;
-  }
-
-  frame_length_ = 3 * width * height / 2;
-  width_  = width;
-  height_ = height;
-  return true;
-}
-
-void ViEFileCaptureDevice::ReadFileFor(uint64_t time_slice_ms,
-                                       uint32_t max_fps) {
-  webrtc::CriticalSectionScoped cs(mutex_);
-  assert(input_file_ != NULL);
-
-  unsigned char* frame_buffer = new unsigned char[frame_length_];
-
-  webrtc::EventWrapper* sleeper = webrtc::EventWrapper::Create();
-
-  uint64_t start_time_ms = webrtc::TickTime::MillisecondTimestamp();
-  uint64_t elapsed_ms = 0;
-
-  while (elapsed_ms < time_slice_ms) {
-    FramePacemaker pacemaker(max_fps);
-    size_t read = fread(frame_buffer, 1, frame_length_, input_file_);
-
-    if (feof(input_file_) || read != frame_length_) {
-      rewind(input_file_);
-    }
-    webrtc::I420VideoFrame frame;
-    frame.CreateFrame(frame_buffer, width_, height_, webrtc::kVideoRotation_0);
-    frame.set_render_time_ms(webrtc::TickTime::MillisecondTimestamp());
-    input_sink_->IncomingFrame(frame);
-
-    pacemaker.SleepIfNecessary(sleeper);
-    elapsed_ms = webrtc::TickTime::MillisecondTimestamp() - start_time_ms;
-  }
-
-  delete sleeper;
-  delete[] frame_buffer;
-}
-
-void ViEFileCaptureDevice::CloseFile() {
-  webrtc::CriticalSectionScoped cs(mutex_);
-  assert(input_file_ != NULL);
-
-  fclose(input_file_);
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
deleted file mode 100644
index d0aee84bd6..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
+++ /dev/null
@@ -1,204 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h"
-
-#include <assert.h>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-namespace test {
-struct Frame {
- public:
-  Frame() : buffer(nullptr), buffer_size(0), timestamp(0), render_time(0) {}
-
-  rtc::scoped_ptr<unsigned char[]> buffer;
-  size_t buffer_size;
-  uint32_t timestamp;
-  int64_t render_time;
-
- private:
-  DISALLOW_COPY_AND_ASSIGN(Frame);
-};
-};  // namespace test
-
-ViEToFileRenderer::ViEToFileRenderer()
-    : output_file_(NULL),
-      output_path_(),
-      output_filename_(),
-      thread_(webrtc::ThreadWrapper::CreateThread(
-          ViEToFileRenderer::RunRenderThread, this, "ViEToFileRendererThread")),
-      frame_queue_cs_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-      frame_render_event_(webrtc::EventWrapper::Create()),
-      render_queue_(),
-      free_frame_queue_() {
-}
-
-ViEToFileRenderer::~ViEToFileRenderer() {
-  while (!free_frame_queue_.empty()) {
-    delete free_frame_queue_.front();
-    free_frame_queue_.pop_front();
-  }
-}
-
-bool ViEToFileRenderer::PrepareForRendering(
-    const std::string& output_path,
-    const std::string& output_filename) {
-
-  assert(output_file_ == NULL);
-
-  output_file_ = fopen((output_path + output_filename).c_str(), "wb");
-  if (output_file_ == NULL) {
-    return false;
-  }
-
-  output_filename_ = output_filename;
-  output_path_ = output_path;
-  return thread_->Start();
-}
-
-void ViEToFileRenderer::StopRendering() {
-  assert(output_file_ != NULL);
-  if (thread_.get() != NULL) {
-    // Signal that a frame is ready to be written to file.
-    frame_render_event_->Set();
-    // Call Stop() repeatedly, waiting for ProcessRenderQueue() to finish.
-    while (!thread_->Stop()) continue;
-  }
-  fclose(output_file_);
-  output_file_ = NULL;
-}
-
-bool ViEToFileRenderer::SaveOutputFile(const std::string& prefix) {
-  assert(output_file_ == NULL && output_filename_ != "");
-  if (rename((output_path_ + output_filename_).c_str(),
-                  (output_path_ + prefix + output_filename_).c_str()) != 0) {
-    perror("Failed to rename output file");
-    return false;
-  }
-  ForgetOutputFile();
-  return true;
-}
-
-bool ViEToFileRenderer::DeleteOutputFile() {
-  assert(output_file_ == NULL && output_filename_ != "");
-  if (remove((output_path_ + output_filename_).c_str()) != 0) {
-    perror("Failed to delete output file");
-    return false;
-  }
-  ForgetOutputFile();
-  return true;
-}
-
-const std::string ViEToFileRenderer::GetFullOutputPath() const {
-  return output_path_ + output_filename_;
-}
-
-void ViEToFileRenderer::ForgetOutputFile() {
-  output_filename_ = "";
-  output_path_ = "";
-}
-
-test::Frame* ViEToFileRenderer::NewFrame(size_t buffer_size) {
-  test::Frame* frame;
-  if (free_frame_queue_.empty()) {
-    frame = new test::Frame();
-  } else {
-    // Reuse an already allocated frame.
-    frame = free_frame_queue_.front();
-    free_frame_queue_.pop_front();
-  }
-  if (frame->buffer_size < buffer_size) {
-    frame->buffer.reset(new unsigned char[buffer_size]);
-    frame->buffer_size = buffer_size;
-  }
-  return frame;
-}
-
-int ViEToFileRenderer::DeliverFrame(unsigned char *buffer,
-                                    size_t buffer_size,
-                                    uint32_t time_stamp,
-                                    int64_t ntp_time_ms,
-                                    int64_t render_time,
-                                    void* /*handle*/) {
-  webrtc::CriticalSectionScoped lock(frame_queue_cs_.get());
-  test::Frame* frame = NewFrame(buffer_size);
-  memcpy(frame->buffer.get(), buffer, buffer_size);
-  frame->timestamp = time_stamp;
-  frame->render_time = render_time;
-
-  render_queue_.push_back(frame);
-  // Signal that a frame is ready to be written to file.
-  frame_render_event_->Set();
-  return 0;
-}
-
-int ViEToFileRenderer::DeliverI420Frame(
-    const webrtc::I420VideoFrame& input_frame) {
-  const size_t buffer_size =
-      CalcBufferSize(webrtc::kI420, input_frame.width(), input_frame.height());
-  webrtc::CriticalSectionScoped lock(frame_queue_cs_.get());
-  test::Frame* frame = NewFrame(buffer_size);
-  const int length =
-      ExtractBuffer(input_frame, frame->buffer_size, frame->buffer.get());
-  assert(static_cast<size_t>(length) == buffer_size);
-  if (length < 0)
-    return -1;
-  frame->timestamp = input_frame.timestamp();
-  frame->render_time = input_frame.render_time_ms();
-
-  render_queue_.push_back(frame);
-  // Signal that a frame is ready to be written to file.
-  frame_render_event_->Set();
-  return 0;
-}
-
-bool ViEToFileRenderer::IsTextureSupported() { return false; }
-
-int ViEToFileRenderer::FrameSizeChange(unsigned int width,
-                                       unsigned int height,
-                                       unsigned int number_of_streams) {
-  return 0;
-}
-
-bool ViEToFileRenderer::RunRenderThread(void* obj) {
-  assert(obj);
-  ViEToFileRenderer* renderer = static_cast<ViEToFileRenderer*>(obj);
-  return renderer->ProcessRenderQueue();
-}
-
-bool ViEToFileRenderer::ProcessRenderQueue() {
-  // Wait for a frame to be rendered.
-  frame_render_event_->Wait(WEBRTC_EVENT_INFINITE);
-  frame_queue_cs_->Enter();
-  // Render all frames in the queue.
-  while (!render_queue_.empty()) {
-    test::Frame* frame = render_queue_.front();
-    render_queue_.pop_front();
-    // Leave the critical section before writing to file to not block calls to
-    // the renderer.
-    frame_queue_cs_->Leave();
-    assert(output_file_);
-    size_t written = fwrite(frame->buffer.get(), sizeof(unsigned char),
-                            frame->buffer_size, output_file_);
-    frame_queue_cs_->Enter();
-    // Return the frame.
-    free_frame_queue_.push_front(frame);
-    if (written != frame->buffer_size) {
-      frame_queue_cs_->Leave();
-      return false;
-    }
-  }
-  frame_queue_cs_->Leave();
-  return true;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h
deleted file mode 100644
index 918edf96c3..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the interface to I420 "codec"
- * This is a dummy wrapper to allow VCM deal with raw I420 sequences
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
-
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-
-class TbI420Encoder: public webrtc::VideoEncoder
-{
-public:
-    TbI420Encoder();
-    virtual ~TbI420Encoder();
-
-    int32_t InitEncode(const webrtc::VideoCodec* codecSettings,
-                       int32_t numberOfCores,
-                       size_t maxPayloadSize) override;
-
-    int32_t Encode(
-        const webrtc::I420VideoFrame& inputImage,
-        const webrtc::CodecSpecificInfo* codecSpecificInfo,
-        const std::vector<webrtc::VideoFrameType>* frameTypes) override;
-
-    int32_t RegisterEncodeCompleteCallback(
-        webrtc::EncodedImageCallback* callback) override;
-
-    int32_t Release() override;
-
-    int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override;
-
-    int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override;
-
-    int32_t SetPeriodicKeyFrames(bool enable) override;
-
-    int32_t CodecConfigParameters(uint8_t* /*buffer*/,
-                                  int32_t /*size*/) override;
-
-    struct FunctionCalls
-    {
-        int32_t InitEncode;
-        int32_t Encode;
-        int32_t RegisterEncodeCompleteCallback;
-        int32_t Release;
-        int32_t Reset;
-        int32_t SetChannelParameters;
-        int32_t SetRates;
-        int32_t SetPeriodicKeyFrames;
-        int32_t CodecConfigParameters;
-
-    };
-
-    FunctionCalls GetFunctionCalls();
-private:
-    bool _inited;
-    webrtc::EncodedImage _encodedImage;
-    FunctionCalls _functionCalls;
-    webrtc::EncodedImageCallback* _encodedCompleteCallback;
-
-}; // end of tbI420Encoder class
-
-
-/***************************/
-/* tbI420Decoder class */
-/***************************/
-
-class TbI420Decoder: public webrtc::VideoDecoder
-{
-public:
-    TbI420Decoder();
-    virtual ~TbI420Decoder();
-
-    int32_t InitDecode(const webrtc::VideoCodec* inst,
-                       int32_t numberOfCores) override;
-    int32_t Decode(const webrtc::EncodedImage& inputImage,
-                   bool missingFrames,
-                   const webrtc::RTPFragmentationHeader* fragmentation,
-                   const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
-                   int64_t renderTimeMs = -1) override;
-
-    int32_t RegisterDecodeCompleteCallback(
-        webrtc::DecodedImageCallback* callback) override;
-    int32_t Release() override;
-    int32_t Reset() override;
-
-    struct FunctionCalls
-    {
-        int32_t InitDecode;
-        int32_t Decode;
-        int32_t RegisterDecodeCompleteCallback;
-        int32_t Release;
-        int32_t Reset;
-    };
-
-    FunctionCalls GetFunctionCalls();
-
-private:
-
-    webrtc::I420VideoFrame _decodedImage;
-    int32_t _width;
-    int32_t _height;
-    bool _inited;
-    FunctionCalls _functionCalls;
-    webrtc::DecodedImageCallback* _decodeCompleteCallback;
-
-}; // end of tbI420Decoder class
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_capture_device.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_capture_device.h
deleted file mode 100644
index 84d3f77d46..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_capture_device.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
-
-#include <string>
-
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-
-class TbInterfaces;
-
-class TbCaptureDevice
-{
-public:
-    TbCaptureDevice(TbInterfaces& Engine);
-    ~TbCaptureDevice(void);
-
-    int captureId;
-    void ConnectTo(int videoChannel);
-    void Disconnect(int videoChannel);
-    std::string device_name() const;
-
-private:
-    TbInterfaces& ViE;
-    webrtc::VideoCaptureModule* vcpm_;
-    std::string device_name_;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h
deleted file mode 100644
index e11c02d181..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h
+++ /dev/null
@@ -1,194 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// tb_external_transport.h
-//
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
-
-#include <list>
-#include <map>
-
-#include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class EventWrapper;
-class ViENetwork;
-}
-
-enum RandomLossModel {
-  kNoLoss,
-  kUniformLoss,
-  kGilbertElliotLoss
-};
-struct NetworkParameters {
-  int packet_loss_rate;
-  int burst_length;  // Only applicable for kGilbertElliotLoss.
-  int mean_one_way_delay;
-  int std_dev_one_way_delay;
-  RandomLossModel loss_model;
-  NetworkParameters():
-    packet_loss_rate(0), burst_length(0), mean_one_way_delay(0),
-        std_dev_one_way_delay(0), loss_model(kNoLoss) {}
-};
-
-// Allows to subscribe for callback when a frame is started being sent.
-class SendFrameCallback
-{
-public:
-    // Called once per frame (when a new RTP timestamp is detected) when the
-    // first data packet of the frame is being sent using the
-    // TbExternalTransport.SendPacket method.
-    virtual void FrameSent(unsigned int rtp_timestamp) = 0;
-protected:
-    SendFrameCallback() {}
-    virtual ~SendFrameCallback() {}
-};
-
-// Allows to subscribe for callback when the first packet of a frame is
-// received.
-class ReceiveFrameCallback
-{
-public:
-    // Called once per frame (when a new RTP timestamp is detected)
-    // during the processing of the RTP packet queue in
-    // TbExternalTransport::ViEExternalTransportProcess.
-    virtual void FrameReceived(unsigned int rtp_timestamp) = 0;
-protected:
-    ReceiveFrameCallback() {}
-    virtual ~ReceiveFrameCallback() {}
-};
-
-// External transport implementation for testing purposes.
-// A packet loss probability must be set in order to drop packets from the data
-// being sent to this class.
-// Will never drop packets from the first frame of a video sequence.
-class TbExternalTransport : public webrtc::Transport
-{
-public:
-    typedef std::map<unsigned int, unsigned int> SsrcChannelMap;
-
-    TbExternalTransport(webrtc::ViENetwork& vieNetwork,
-                        int sender_channel,
-                        TbExternalTransport::SsrcChannelMap* receive_channels);
-    ~TbExternalTransport(void);
-
-    int SendPacket(int channel, const void* data, size_t len) override;
-    int SendRTCPPacket(int channel, const void* data, size_t len) override;
-
-    // Should only be called before/after traffic is being processed.
-    // Only one observer can be set (multiple calls will overwrite each other).
-    virtual void RegisterSendFrameCallback(SendFrameCallback* callback);
-
-    // Should only be called before/after traffic is being processed.
-    // Only one observer can be set (multiple calls will overwrite each other).
-    virtual void RegisterReceiveFrameCallback(ReceiveFrameCallback* callback);
-
-    // The network parameters of the link. Regarding packet losses, packets
-    // belonging to the first frame (same RTP timestamp) will never be dropped.
-    void SetNetworkParameters(const NetworkParameters& network_parameters);
-    void SetSSRCFilter(uint32_t SSRC);
-
-    void ClearStats();
-    // |packet_counters| is a map which counts the number of packets sent per
-    // payload type.
-    void GetStats(int32_t& numRtpPackets,
-                  int32_t& numDroppedPackets,
-                  int32_t& numRtcpPackets,
-                  std::map<uint8_t, int>* packet_counters);
-
-    void SetTemporalToggle(unsigned char layers);
-    void EnableSSRCCheck();
-    unsigned int ReceivedSSRC();
-
-    void EnableSequenceNumberCheck();
-    unsigned short GetFirstSequenceNumber();
-
-    bool EmptyQueue() const;
-
-protected:
-    static bool ViEExternalTransportRun(void* object);
-    bool ViEExternalTransportProcess();
-private:
-    // TODO(mikhal): Break these out to classes.
-    static int GaussianRandom(int mean_ms, int standard_deviation_ms);
-    bool UniformLoss(int loss_rate);
-    bool GilbertElliotLoss(int loss_rate, int burst_length);
-    int64_t NowMs();
-
-    enum
-    {
-        KMaxPacketSize = 1650
-    };
-    enum
-    {
-        KMaxWaitTimeMs = 100
-    };
-    typedef struct
-    {
-        int8_t packetBuffer[KMaxPacketSize];
-        size_t length;
-        int32_t channel;
-        int64_t receiveTime;
-    } VideoPacket;
-
-    int sender_channel_;
-    SsrcChannelMap* receive_channels_;
-    webrtc::ViENetwork& _vieNetwork;
-    rtc::scoped_ptr<webrtc::ThreadWrapper> _thread;
-    webrtc::EventWrapper& _event;
-    webrtc::CriticalSectionWrapper& _crit;
-    webrtc::CriticalSectionWrapper& _statCrit;
-
-    NetworkParameters network_parameters_;
-    int32_t _rtpCount;
-    int32_t _rtcpCount;
-    int32_t _dropCount;
-    // |packet_counters| is a map which counts the number of packets sent per
-    // payload type.
-    std::map<uint8_t, int> packet_counters_;
-
-    std::list<VideoPacket*> _rtpPackets;
-    std::list<VideoPacket*> _rtcpPackets;
-
-    SendFrameCallback* _send_frame_callback;
-    ReceiveFrameCallback* _receive_frame_callback;
-
-    unsigned char _temporalLayers;
-    unsigned short _seqNum;
-    unsigned short _sendPID;
-    unsigned char _receivedPID;
-    bool _switchLayer;
-    unsigned char _currentRelayLayer;
-    unsigned int _lastTimeMs;
-
-    bool _checkSSRC;
-    uint32_t _lastSSRC;
-    bool _filterSSRC;
-    uint32_t _SSRC;
-    bool _checkSequenceNumber;
-    uint16_t _firstSequenceNumber;
-
-    // Keep track of the first RTP timestamp so we don't do packet loss on
-    // the first frame.
-    uint32_t _firstRTPTimestamp;
-    // Track RTP timestamps so we invoke callbacks properly (if registered).
-    uint32_t _lastSendRTPTimestamp;
-    uint32_t _lastReceiveRTPTimestamp;
-    int64_t last_receive_time_;
-    bool previous_drop_;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_interfaces.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_interfaces.h
deleted file mode 100644
index 5b52a1e24e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_interfaces.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
-
-#include <string>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/common_types.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/vie_defines.h"
-
-// This class deals with all the tedium of setting up video engine interfaces.
-// It does its work in constructor and destructor, so keeping it in scope is
-// enough. It also sets up tracing.
-class TbInterfaces
-{
-public:
-    // Sets up all interfaces and creates a trace file
-    TbInterfaces(const std::string& test_name);
-    ~TbInterfaces(void);
-
-    webrtc::VideoEngine* video_engine;
-    webrtc::ViEBase* base;
-    webrtc::ViECapture* capture;
-    webrtc::ViERender* render;
-    webrtc::ViERTP_RTCP* rtp_rtcp;
-    webrtc::ViECodec* codec;
-    webrtc::ViENetwork* network;
-    webrtc::ViEImageProcess* image_process;
-
-    int LastError() {
-        return base->LastError();
-    }
-
-private:
-    DISALLOW_COPY_AND_ASSIGN(TbInterfaces);
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h
deleted file mode 100644
index 7d2557e4ad..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-
-namespace webrtc {
-namespace test {
-class VideoChannelTransport;
-}  // namespace test
-}  // namespace webrtc
-
-class TbVideoChannel {
- public:
-  TbVideoChannel(TbInterfaces& Engine,
-                 webrtc::VideoCodecType sendCodec = webrtc::kVideoCodecVP8,
-                 int width = 352, int height = 288, int frameRate = 30,
-                 int startBitrate = 300);
-
-  ~TbVideoChannel(void);
-
-  void SetFrameSettings(int width, int height, int frameRate);
-
-  void StartSend(const unsigned short rtpPort = 11000,
-                 const char* ipAddress = "127.0.0.1");
-
-  void StopSend();
-
-  void StartReceive(const unsigned short rtpPort = 11000);
-
-  void StopReceive();
-
-  int videoChannel;
-
- private:
-  TbInterfaces& ViE;
-  rtc::scoped_ptr<webrtc::test::VideoChannelTransport> channel_transport_;
-};
-
-
-#endif  // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
deleted file mode 100644
index 057f6c8011..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_VIE_EXTERNAL_RENDER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_VIE_EXTERNAL_RENDER_H_
-
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/include/vie_render.h"
-
-namespace webrtc {
-
-// A render filter which passes frames directly to an external renderer. This
-// is different from plugging the external renderer directly into the sending
-// side since this will only run on frames that actually get sent and not on
-// frames that only get captured.
-class ExternalRendererEffectFilter : public webrtc::ViEEffectFilter {
- public:
-  explicit ExternalRendererEffectFilter(webrtc::ExternalRenderer* renderer)
-      : width_(0), height_(0), renderer_(renderer) {}
-  virtual ~ExternalRendererEffectFilter() {}
-  virtual int Transform(size_t size,
-                        unsigned char* frame_buffer,
-                        int64_t ntp_time_ms,
-                        unsigned int timestamp,
-                        unsigned int width,
-                        unsigned int height) {
-    if (width != width_ || height_ != height) {
-      renderer_->FrameSizeChange(width, height, 1);
-      width_ = width;
-      height_ = height;
-    }
-    return renderer_->DeliverFrame(frame_buffer,
-                                   size,
-                                   ntp_time_ms,
-                                   timestamp,
-                                   webrtc::TickTime::MillisecondTimestamp(),
-                                   NULL);
-  }
-
- private:
-  unsigned int width_;
-  unsigned int height_;
-  webrtc::ExternalRenderer* renderer_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_VIE_EXTERNAL_RENDER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h
deleted file mode 100644
index afb2752a91..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
-
-#include <string>
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-
-namespace webrtc {
-class ViECapture;
-}
-
-class ViEFileCaptureDevice;
-
-// Registers an external capture device with the provided capture interface
-// and starts running a fake camera by reading frames from a file. The frame-
-// reading code runs in a separate thread which makes it possible to run tests
-// while the fake camera feeds data into the system. This class is not thread-
-// safe in itself (but handles its own thread in a safe manner).
-class ViEFakeCamera {
- public:
-  // The argument is the capture interface to register with.
-  explicit ViEFakeCamera(webrtc::ViECapture* capture_interface);
-  virtual ~ViEFakeCamera();
-
-  // Runs the scenario in the class comments.
-  bool StartCameraInNewThread(const std::string& i420_test_video_path,
-                              int width,
-                              int height);
-  // Stops the camera and cleans up everything allocated by the start method.
-  bool StopCamera();
-
-  int capture_id() const { return capture_id_; }
-
- private:
-  webrtc::ViECapture* capture_interface_;
-
-  int capture_id_;
-  rtc::scoped_ptr<webrtc::ThreadWrapper> camera_thread_;
-  ViEFileCaptureDevice* file_capture_device_;
-};
-
-#endif  // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_file_capture_device.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_file_capture_device.h
deleted file mode 100644
index 2abd88c41c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_file_capture_device.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
-
-#include <stdio.h>
-
-#include <string>
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ViEExternalCapture;
-}
-
-// This class opens a i420 file and feeds it into a ExternalCapture instance,
-// thereby acting as a faked capture device with deterministic input.
-class ViEFileCaptureDevice {
- public:
-  // The input sink is where to send the I420 video frames.
-  explicit ViEFileCaptureDevice(webrtc::ViEExternalCapture* input_sink);
-  virtual ~ViEFileCaptureDevice();
-
-  // Opens the provided I420 file and interprets it according to the provided
-  // width and height. Returns false if the file doesn't exist.
-  bool OpenI420File(const std::string& path, int width, int height);
-
-  // Reads the previously opened file for at most time_slice_ms milliseconds,
-  // after which it will return. It will make sure to sleep accordingly so we
-  // do not send more than max_fps cap (we may send less, though).
-  void ReadFileFor(uint64_t time_slice_ms, uint32_t max_fps);
-
-  // Closes the opened input file.
-  void CloseFile();
-
- private:
-  webrtc::ViEExternalCapture* input_sink_;
-
-  FILE* input_file_;
-  webrtc::CriticalSectionWrapper* mutex_;
-
-  uint32_t frame_length_;
-  uint32_t width_;
-  uint32_t height_;
-};
-
-#endif  // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
deleted file mode 100644
index d873cd7a6a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_VIE_TO_FILE_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_VIE_TO_FILE_RENDERER_H_
-
-#include <stdio.h>
-#include <string.h>
-
-#include <list>
-#include <string>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/video_engine/include/vie_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-};  // namespace webrtc
-
-namespace test {
-struct Frame;
-};  // namespace test
-
-class ViEToFileRenderer: public webrtc::ExternalRenderer {
- public:
-  ViEToFileRenderer();
-  virtual ~ViEToFileRenderer();
-
-  // Returns false if we fail opening the output filename for writing.
-  bool PrepareForRendering(const std::string& output_path,
-                           const std::string& output_filename);
-
-  // Closes the output file.
-  void StopRendering();
-
-  // Deletes the closed output file from the file system. This is one option
-  // after calling StopRendering, the other being KeepOutputFile. This file
-  // renderer will forget about the file after this call and can be used again.
-  bool DeleteOutputFile();
-
-  // Renames the closed output file to its previous name with the provided
-  // prefix prepended. This file renderer will forget about the file after this
-  // call and can be used again.
-  bool SaveOutputFile(const std::string& prefix);
-
-  // Implementation of ExternalRenderer:
-  int FrameSizeChange(unsigned int width,
-                      unsigned int height,
-                      unsigned int number_of_streams) override;
-
-  int DeliverFrame(unsigned char* buffer,
-                   size_t buffer_size,
-                   uint32_t time_stamp,
-                   int64_t ntp_time_ms,
-                   int64_t render_time,
-                   void* handle) override;
-
-  int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;
-
-  bool IsTextureSupported() override;
-
-  const std::string GetFullOutputPath() const;
-
- private:
-  typedef std::list<test::Frame*> FrameQueue;
-
-  // Returns a frame with the specified |buffer_size|. Tries to avoid allocating
-  // new frames by reusing frames from |free_frame_queue_|.
-  test::Frame* NewFrame(size_t buffer_size);
-  static bool RunRenderThread(void* obj);
-  void ForgetOutputFile();
-  bool ProcessRenderQueue();
-
-  FILE* output_file_;
-  std::string output_path_;
-  std::string output_filename_;
-  rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
-  rtc::scoped_ptr<webrtc::CriticalSectionWrapper> frame_queue_cs_;
-  rtc::scoped_ptr<webrtc::EventWrapper> frame_render_event_;
-  FrameQueue render_queue_;
-  FrameQueue free_frame_queue_;
-};
-
-#endif  // WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_VIE_TO_FILE_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi
deleted file mode 100644
index 603cd4b1ea..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi
+++ /dev/null
@@ -1,50 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-{
-  'targets': [
-    {
-      'target_name': 'libvietest',
-      'type': 'static_library',
-      'dependencies': [
-        '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-        '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(webrtc_root)/test/test.gyp:test_support',
-        'video_engine_core',
-      ],
-      'sources': [
-        # Helper classes
-        'include/vie_external_render_filter.h',
-        'include/vie_fake_camera.h',
-        'include/vie_file_capture_device.h',
-        'include/vie_to_file_renderer.h',
-
-        'helpers/vie_fake_camera.cc',
-        'helpers/vie_file_capture_device.cc',
-        'helpers/vie_to_file_renderer.cc',
-
-        # Testbed classes
-        'include/tb_capture_device.h',
-        'include/tb_external_transport.h',
-        'include/tb_I420_codec.h',
-        'include/tb_interfaces.h',
-        'include/tb_video_channel.h',
-
-        'testbed/tb_capture_device.cc',
-        'testbed/tb_external_transport.cc',
-        'testbed/tb_I420_codec.cc',
-        'testbed/tb_interfaces.cc',
-        'testbed/tb_video_channel.cc',
-      ],
-      # Disable warnings to enable Win64 build, issue 1323.
-      'msvs_disabled_warnings': [
-        4267,  # size_t to int truncation.
-      ],
-    },
-  ],
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc
deleted file mode 100644
index e9cefcf84d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/libvietest/include/tb_I420_codec.h"
-
-#include <assert.h>
-#include <stdio.h>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-
-TbI420Encoder::TbI420Encoder() :
-    _inited(false), _encodedImage(), _encodedCompleteCallback(NULL)
-{
-    //
-    memset(&_functionCalls, 0, sizeof(_functionCalls));
-}
-
-TbI420Encoder::~TbI420Encoder()
-{
-    _inited = false;
-    if (_encodedImage._buffer != NULL)
-    {
-        delete[] _encodedImage._buffer;
-        _encodedImage._buffer = NULL;
-    }
-}
-
-int32_t TbI420Encoder::Release()
-{
-    _functionCalls.Release++;
-    // should allocate an encoded frame and then release it here, for that we
-    // actaully need an init flag
-    if (_encodedImage._buffer != NULL)
-    {
-        delete[] _encodedImage._buffer;
-        _encodedImage._buffer = NULL;
-    }
-    _inited = false;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Encoder::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
-  _functionCalls.SetChannelParameters++;
-  return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Encoder::InitEncode(const webrtc::VideoCodec* inst,
-                                  int32_t /*numberOfCores*/,
-                                  size_t /*maxPayloadSize */)
-{
-    _functionCalls.InitEncode++;
-    if (inst == NULL)
-    {
-        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-    }
-    if (inst->width < 1 || inst->height < 1)
-    {
-        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-    }
-
-    // allocating encoded memory
-    if (_encodedImage._buffer != NULL)
-    {
-        delete[] _encodedImage._buffer;
-        _encodedImage._buffer = NULL;
-        _encodedImage._size = 0;
-    }
-    const uint32_t newSize = (3 * inst->width * inst->height) >> 1;
-    uint8_t* newBuffer = new uint8_t[newSize];
-    if (newBuffer == NULL)
-    {
-        return WEBRTC_VIDEO_CODEC_MEMORY;
-    }
-    _encodedImage._size = newSize;
-    _encodedImage._buffer = newBuffer;
-
-    // if no memeory allocation, no point to init
-    _inited = true;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Encoder::Encode(
-    const webrtc::I420VideoFrame& inputImage,
-    const webrtc::CodecSpecificInfo* /*codecSpecificInfo*/,
-    const std::vector<webrtc::VideoFrameType>* /*frameTypes*/)
-{
-    _functionCalls.Encode++;
-    if (!_inited)
-    {
-        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
-    }
-    if (_encodedCompleteCallback == NULL)
-    {
-        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
-    }
-
-    _encodedImage._frameType = webrtc::kKeyFrame; // no coding
-    _encodedImage._timeStamp = inputImage.timestamp();
-    _encodedImage._encodedHeight = inputImage.height();
-    _encodedImage._encodedWidth = inputImage.width();
-    size_t reqSize = webrtc::CalcBufferSize(webrtc::kI420,
-                                            _encodedImage._encodedWidth,
-                                            _encodedImage._encodedHeight);
-    if (reqSize > _encodedImage._size)
-    {
-
-        // allocating encoded memory
-        if (_encodedImage._buffer != NULL)
-        {
-            delete[] _encodedImage._buffer;
-            _encodedImage._buffer = NULL;
-            _encodedImage._size = 0;
-        }
-        uint8_t* newBuffer = new uint8_t[reqSize];
-        if (newBuffer == NULL)
-        {
-            return WEBRTC_VIDEO_CODEC_MEMORY;
-        }
-        _encodedImage._size = reqSize;
-        _encodedImage._buffer = newBuffer;
-    }
-    if (ExtractBuffer(inputImage, _encodedImage._size,
-                      _encodedImage._buffer) < 0) {
-      return -1;
-    }
-
-    _encodedImage._length = reqSize;
-    _encodedCompleteCallback->Encoded(_encodedImage, NULL, NULL);
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Encoder::RegisterEncodeCompleteCallback(
-    webrtc::EncodedImageCallback* callback)
-{
-    _functionCalls.RegisterEncodeCompleteCallback++;
-    _encodedCompleteCallback = callback;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Encoder::SetRates(uint32_t newBitRate, uint32_t frameRate)
-{
-    _functionCalls.SetRates++;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Encoder::SetPeriodicKeyFrames(bool enable)
-{
-    _functionCalls.SetPeriodicKeyFrames++;
-    return WEBRTC_VIDEO_CODEC_ERROR;
-}
-
-int32_t TbI420Encoder::CodecConfigParameters(uint8_t* /*buffer*/,
-                                             int32_t /*size*/)
-{
-    _functionCalls.CodecConfigParameters++;
-    return WEBRTC_VIDEO_CODEC_ERROR;
-}
-TbI420Encoder::FunctionCalls TbI420Encoder::GetFunctionCalls()
-{
-    return _functionCalls;
-}
-
-TbI420Decoder::TbI420Decoder():
-    _decodedImage(), _width(0), _height(0), _inited(false),
-        _decodeCompleteCallback(NULL)
-{
-    memset(&_functionCalls, 0, sizeof(_functionCalls));
-}
-
-TbI420Decoder::~TbI420Decoder()
-{
-    Release();
-}
-
-int32_t TbI420Decoder::Reset()
-{
-    _functionCalls.Reset++;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Decoder::InitDecode(const webrtc::VideoCodec* inst,
-                                  int32_t /*numberOfCores */)
-{
-    _functionCalls.InitDecode++;
-    if (inst == NULL)
-    {
-        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-    }
-    else if (inst->width < 1 || inst->height < 1)
-    {
-        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-    }
-    _width = inst->width;
-    _height = inst->height;
-    int half_width = (_width + 1 ) / 2 ;
-    _decodedImage.CreateEmptyFrame(_width, _height,
-                                   _width, half_width, half_width);
-    _inited = true;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Decoder::Decode(
-    const webrtc::EncodedImage& inputImage,
-    bool /*missingFrames*/,
-    const webrtc::RTPFragmentationHeader* /*fragmentation*/,
-    const webrtc::CodecSpecificInfo* /*codecSpecificInfo*/,
-    int64_t /*renderTimeMs*/)
-{
-    _functionCalls.Decode++;
-    if (inputImage._buffer == NULL)
-    {
-        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-    }
-    if (_decodeCompleteCallback == NULL)
-    {
-        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
-    }
-    if (inputImage._length <= 0)
-    {
-        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-    }
-    if (!_inited)
-    {
-        return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
-    }
-
-    // Only send complete frames.
-    if (webrtc::CalcBufferSize(webrtc::kI420,_width,_height) !=
-        inputImage._length) {
-      return WEBRTC_VIDEO_CODEC_ERROR;
-    }
-
-    int ret =
-        ConvertToI420(webrtc::kI420, inputImage._buffer, 0, 0, _width, _height,
-                      0, webrtc::kVideoRotation_0, &_decodedImage);
-
-    if (ret < 0)
-      return WEBRTC_VIDEO_CODEC_ERROR;
-
-    _decodedImage.set_timestamp(inputImage._timeStamp);
-
-    _decodeCompleteCallback->Decoded(_decodedImage);
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Decoder::RegisterDecodeCompleteCallback(
-    webrtc::DecodedImageCallback* callback)
-{
-    _functionCalls.RegisterDecodeCompleteCallback++;
-    _decodeCompleteCallback = callback;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int32_t TbI420Decoder::Release()
-{
-    _functionCalls.Release++;
-    _inited = false;
-    return WEBRTC_VIDEO_CODEC_OK;
-}
-
-TbI420Decoder::FunctionCalls TbI420Decoder::GetFunctionCalls()
-{
-    return _functionCalls;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_capture_device.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_capture_device.cc
deleted file mode 100644
index faa70907cc..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_capture_device.cc
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/libvietest/include/tb_capture_device.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-
-TbCaptureDevice::TbCaptureDevice(TbInterfaces& Engine) :
-    captureId(-1),
-    ViE(Engine),
-    vcpm_(NULL)
-{
-    const unsigned int KMaxDeviceNameLength = 128;
-    const unsigned int KMaxUniqueIdLength = 256;
-    char deviceName[KMaxDeviceNameLength];
-    memset(deviceName, 0, KMaxDeviceNameLength);
-    char uniqueId[KMaxUniqueIdLength];
-    memset(uniqueId, 0, KMaxUniqueIdLength);
-
-    bool captureDeviceSet = false;
-
-    webrtc::VideoCaptureModule::DeviceInfo* devInfo =
-        webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
-    for (size_t captureIdx = 0;
-        captureIdx < devInfo->NumberOfDevices();
-        captureIdx++)
-    {
-        EXPECT_EQ(0, devInfo->GetDeviceName(captureIdx, deviceName,
-                                            KMaxDeviceNameLength, uniqueId,
-                                            KMaxUniqueIdLength));
-
-        vcpm_ = webrtc::VideoCaptureFactory::Create(
-            captureIdx, uniqueId);
-        if (vcpm_ == NULL)  // Failed to open this device. Try next.
-        {
-            continue;
-        }
-        vcpm_->AddRef();
-
-        int error = ViE.capture->AllocateCaptureDevice(*vcpm_, captureId);
-        if (error == 0)
-        {
-            captureDeviceSet = true;
-            break;
-        }
-    }
-    delete devInfo;
-    EXPECT_TRUE(captureDeviceSet);
-    if (!captureDeviceSet) {
-        return;
-    }
-
-    device_name_ = deviceName;
-    EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
-}
-
-TbCaptureDevice::~TbCaptureDevice(void)
-{
-    EXPECT_EQ(0, ViE.capture->StopCapture(captureId));
-    EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
-    if (vcpm_)
-      vcpm_->Release();
-}
-
-void TbCaptureDevice::ConnectTo(int videoChannel)
-{
-    EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId, videoChannel));
-}
-
-void TbCaptureDevice::Disconnect(int videoChannel)
-{
-    EXPECT_EQ(0, ViE.capture->DisconnectCaptureDevice(videoChannel));
-}
-
-std::string TbCaptureDevice::device_name() const {
-  return device_name_;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
deleted file mode 100644
index d406985ef4..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
+++ /dev/null
@@ -1,577 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/libvietest/include/tb_external_transport.h"
-
-#include <assert.h>
-
-#include <math.h>
-#include <stdio.h> // printf
-#include <stdlib.h> // rand
-
-#if defined(WEBRTC_LINUX) || defined(__linux__)
-#include <string.h>
-#endif
-#if defined(WEBRTC_MAC)
-#include <string.h>
-#endif
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/include/vie_network.h"
-
-#if defined(_WIN32)
-#pragma warning(disable: 4355) // 'this' : used in base member initializer list
-#endif
-
-TbExternalTransport::TbExternalTransport(
-    webrtc::ViENetwork& vieNetwork,
-    int sender_channel,
-    TbExternalTransport::SsrcChannelMap* receive_channels)
-    :
-      sender_channel_(sender_channel),
-      receive_channels_(receive_channels),
-      _vieNetwork(vieNetwork),
-      _thread(webrtc::ThreadWrapper::CreateThread(
-          ViEExternalTransportRun, this, "AutotestTransport")),
-      _event(*webrtc::EventWrapper::Create()),
-      _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-      _statCrit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-      network_parameters_(),
-      _rtpCount(0),
-      _rtcpCount(0),
-      _dropCount(0),
-      packet_counters_(),
-      _rtpPackets(),
-      _rtcpPackets(),
-      _send_frame_callback(NULL),
-      _receive_frame_callback(NULL),
-      _temporalLayers(0),
-      _seqNum(0),
-      _sendPID(0),
-      _receivedPID(0),
-      _switchLayer(false),
-      _currentRelayLayer(0),
-      _lastTimeMs(webrtc::TickTime::MillisecondTimestamp()),
-      _checkSSRC(false),
-      _lastSSRC(0),
-      _filterSSRC(false),
-      _SSRC(0),
-      _checkSequenceNumber(0),
-      _firstSequenceNumber(0),
-      _firstRTPTimestamp(0),
-      _lastSendRTPTimestamp(0),
-      _lastReceiveRTPTimestamp(0),
-      last_receive_time_(-1),
-      previous_drop_(false)
-{
-    srand((int) webrtc::TickTime::MicrosecondTimestamp());
-    memset(&network_parameters_, 0, sizeof(NetworkParameters));
-    _thread->Start();
-    _thread->SetPriority(webrtc::kHighPriority);
-}
-
-TbExternalTransport::~TbExternalTransport()
-{
-    _event.Set();
-    _thread->Stop();
-    delete &_event;
-
-    for (std::list<VideoPacket*>::iterator it = _rtpPackets.begin();
-         it != _rtpPackets.end(); ++it) {
-        delete *it;
-    }
-    _rtpPackets.clear();
-    for (std::list<VideoPacket*>::iterator it = _rtcpPackets.begin();
-         it != _rtcpPackets.end(); ++it) {
-        delete *it;
-    }
-    _rtcpPackets.clear();
-    delete &_crit;
-    delete &_statCrit;
-}
-
-int TbExternalTransport::SendPacket(int channel, const void *data, size_t len)
-{
-  // Parse timestamp from RTP header according to RFC 3550, section 5.1.
-    uint8_t* ptr = (uint8_t*)data;
-    uint8_t payload_type = ptr[1] & 0x7F;
-    uint32_t rtp_timestamp = ptr[4] << 24;
-    rtp_timestamp += ptr[5] << 16;
-    rtp_timestamp += ptr[6] << 8;
-    rtp_timestamp += ptr[7];
-    _crit.Enter();
-    if (_firstRTPTimestamp == 0) {
-      _firstRTPTimestamp = rtp_timestamp;
-    }
-    _crit.Leave();
-    if (_send_frame_callback != NULL &&
-        _lastSendRTPTimestamp != rtp_timestamp) {
-      _send_frame_callback->FrameSent(rtp_timestamp);
-    }
-    ++packet_counters_[payload_type];
-    _lastSendRTPTimestamp = rtp_timestamp;
-
-    if (_filterSSRC)
-    {
-        uint8_t* ptr = (uint8_t*)data;
-        uint32_t ssrc = ptr[8] << 24;
-        ssrc += ptr[9] << 16;
-        ssrc += ptr[10] << 8;
-        ssrc += ptr[11];
-        if (ssrc != _SSRC)
-        {
-            return static_cast<int>(len); // avoid error in trace file
-        }
-    }
-    if (_temporalLayers) {
-        // parse out vp8 temporal layers
-        // 12 bytes RTP
-        uint8_t* ptr = (uint8_t*)data;
-
-        if (ptr[12] & 0x80 &&  // X-bit
-            ptr[13] & 0x20)  // T-bit
-        {
-            int offset = 1;
-            if (ptr[13] & 0x80) // PID-bit
-            {
-                offset++;
-                if (ptr[14] & 0x80) // 2 byte PID
-                {
-                    offset++;
-                }
-            }
-            if (ptr[13] & 0x40)
-            {
-                offset++;
-            }
-            unsigned char TID = (ptr[13 + offset] >> 5);
-            unsigned int timeMs = NowMs();
-
-            // Every 5 second switch layer
-            if (_lastTimeMs + 5000 < timeMs)
-            {
-                _lastTimeMs = timeMs;
-                _switchLayer = true;
-            }
-            // Switch at the non ref frame
-            if (_switchLayer && (ptr[12] & 0x20))
-            {   // N-bit
-              _currentRelayLayer++;
-                if (_currentRelayLayer >= _temporalLayers)
-                  _currentRelayLayer = 0;
-
-                _switchLayer = false;
-                printf("\t Switching to layer:%d\n", _currentRelayLayer);
-            }
-            if (_currentRelayLayer < TID)
-            {
-                return static_cast<int>(len); // avoid error in trace file
-            }
-            if (ptr[14] & 0x80) // 2 byte PID
-            {
-                if(_receivedPID != ptr[15])
-                {
-                    _sendPID++;
-                    _receivedPID = ptr[15];
-                }
-            } else
-            {
-              if(_receivedPID != ptr[14])
-              {
-                _sendPID++;
-                _receivedPID = ptr[14];
-              }
-            }
-        }
-    }
-    _statCrit.Enter();
-    _rtpCount++;
-    _statCrit.Leave();
-
-    // Packet loss.
-    switch (network_parameters_.loss_model)
-    {
-        case (kNoLoss):
-            previous_drop_ = false;
-            break;
-        case (kUniformLoss):
-            previous_drop_ = UniformLoss(network_parameters_.packet_loss_rate);
-            break;
-        case (kGilbertElliotLoss):
-            previous_drop_ = GilbertElliotLoss(
-                network_parameters_.packet_loss_rate,
-                network_parameters_.burst_length);
-            break;
-    }
-    // Never drop packets from the first RTP timestamp (first frame)
-    // transmitted.
-    if (previous_drop_ && _firstRTPTimestamp != rtp_timestamp)
-    {
-        _statCrit.Enter();
-        _dropCount++;
-        _statCrit.Leave();
-        return static_cast<int>(len);
-    }
-
-    VideoPacket* newPacket = new VideoPacket();
-    assert(len <= sizeof(newPacket->packetBuffer));
-    memcpy(newPacket->packetBuffer, data, len);
-
-    if (_temporalLayers)
-    {
-        // rewrite seqNum
-        newPacket->packetBuffer[2] = _seqNum >> 8;
-        newPacket->packetBuffer[3] = _seqNum;
-        _seqNum++;
-
-        // rewrite PID
-        if (newPacket->packetBuffer[14] & 0x80) // 2 byte PID
-        {
-            newPacket->packetBuffer[14] = (_sendPID >> 8) | 0x80;
-            newPacket->packetBuffer[15] = _sendPID;
-        } else
-        {
-            newPacket->packetBuffer[14] = (_sendPID & 0x7f);
-        }
-    }
-    newPacket->length = len;
-    newPacket->channel = channel;
-
-    _crit.Enter();
-    // Add jitter and make sure receiveTime isn't lower than receive time of
-    // last frame.
-    int network_delay_ms = GaussianRandom(
-        network_parameters_.mean_one_way_delay,
-        network_parameters_.std_dev_one_way_delay);
-    newPacket->receiveTime = NowMs() + network_delay_ms;
-    if (newPacket->receiveTime < last_receive_time_) {
-      newPacket->receiveTime = last_receive_time_;
-    }
-    _rtpPackets.push_back(newPacket);
-    _event.Set();
-    _crit.Leave();
-    return static_cast<int>(len);
-}
-
-void TbExternalTransport::RegisterSendFrameCallback(
-    SendFrameCallback* callback) {
-  _send_frame_callback = callback;
-}
-
-void TbExternalTransport::RegisterReceiveFrameCallback(
-    ReceiveFrameCallback* callback) {
-  _receive_frame_callback = callback;
-}
-
-// Set to 0 to disable.
-void TbExternalTransport::SetTemporalToggle(unsigned char layers)
-{
-    _temporalLayers = layers;
-}
-
-int TbExternalTransport::SendRTCPPacket(int channel,
-                                        const void *data,
-                                        size_t len)
-{
-    _statCrit.Enter();
-    _rtcpCount++;
-    _statCrit.Leave();
-
-    VideoPacket* newPacket = new VideoPacket();
-    assert(len <= sizeof(newPacket->packetBuffer));
-    memcpy(newPacket->packetBuffer, data, len);
-    newPacket->length = len;
-    newPacket->channel = channel;
-
-    _crit.Enter();
-    int network_delay_ms = GaussianRandom(
-            network_parameters_.mean_one_way_delay,
-            network_parameters_.std_dev_one_way_delay);
-    newPacket->receiveTime = NowMs() + network_delay_ms;
-    _rtcpPackets.push_back(newPacket);
-    _event.Set();
-    _crit.Leave();
-    return static_cast<int>(len);
-}
-
-void TbExternalTransport::SetNetworkParameters(
-    const NetworkParameters& network_parameters)
-{
-    webrtc::CriticalSectionScoped cs(&_crit);
-    network_parameters_ = network_parameters;
-}
-
-void TbExternalTransport::SetSSRCFilter(uint32_t ssrc)
-{
-    webrtc::CriticalSectionScoped cs(&_crit);
-    _filterSSRC = true;
-    _SSRC = ssrc;
-}
-
-void TbExternalTransport::ClearStats()
-{
-    webrtc::CriticalSectionScoped cs(&_statCrit);
-    _rtpCount = 0;
-    _dropCount = 0;
-    _rtcpCount = 0;
-    packet_counters_.clear();
-}
-
-void TbExternalTransport::GetStats(int32_t& numRtpPackets,
-                                   int32_t& numDroppedPackets,
-                                   int32_t& numRtcpPackets,
-                                   std::map<uint8_t, int>* packet_counters)
-{
-    webrtc::CriticalSectionScoped cs(&_statCrit);
-    numRtpPackets = _rtpCount;
-    numDroppedPackets = _dropCount;
-    numRtcpPackets = _rtcpCount;
-    *packet_counters = packet_counters_;
-}
-
-void TbExternalTransport::EnableSSRCCheck()
-{
-    webrtc::CriticalSectionScoped cs(&_statCrit);
-    _checkSSRC = true;
-}
-
-unsigned int TbExternalTransport::ReceivedSSRC()
-{
-    webrtc::CriticalSectionScoped cs(&_statCrit);
-    return _lastSSRC;
-}
-
-void TbExternalTransport::EnableSequenceNumberCheck()
-{
-    webrtc::CriticalSectionScoped cs(&_statCrit);
-    _checkSequenceNumber = true;
-}
-
-unsigned short TbExternalTransport::GetFirstSequenceNumber()
-{
-    webrtc::CriticalSectionScoped cs(&_statCrit);
-    return _firstSequenceNumber;
-}
-
-bool TbExternalTransport::EmptyQueue() const {
-  webrtc::CriticalSectionScoped cs(&_crit);
-  return _rtpPackets.empty() && _rtcpPackets.empty();
-}
-
-bool TbExternalTransport::ViEExternalTransportRun(void* object)
-{
-    return static_cast<TbExternalTransport*>
-        (object)->ViEExternalTransportProcess();
-}
-bool TbExternalTransport::ViEExternalTransportProcess()
-{
-    unsigned int waitTime = KMaxWaitTimeMs;
-
-    VideoPacket* packet = NULL;
-
-    _crit.Enter();
-    while (!_rtpPackets.empty())
-    {
-        // Take first packet in queue
-        packet = _rtpPackets.front();
-        int64_t timeToReceive = 0;
-        if (packet)
-        {
-          timeToReceive = packet->receiveTime - NowMs();
-        }
-        else
-        {
-          // There should never be any empty packets in the list.
-          assert(false);
-        }
-        if (timeToReceive > 0)
-        {
-            // No packets to receive yet
-            if (timeToReceive < waitTime && timeToReceive > 0)
-            {
-                waitTime = (unsigned int) timeToReceive;
-            }
-            break;
-        }
-        _rtpPackets.pop_front();
-        _crit.Leave();
-
-        // Send to ViE
-        if (packet)
-        {
-            unsigned int ssrc = 0;
-            {
-                webrtc::CriticalSectionScoped cs(&_statCrit);
-                ssrc = ((packet->packetBuffer[8]) << 24);
-                ssrc += (packet->packetBuffer[9] << 16);
-                ssrc += (packet->packetBuffer[10] << 8);
-                ssrc += packet->packetBuffer[11];
-                if (_checkSSRC)
-                {
-                    _lastSSRC = ((packet->packetBuffer[8]) << 24);
-                    _lastSSRC += (packet->packetBuffer[9] << 16);
-                    _lastSSRC += (packet->packetBuffer[10] << 8);
-                    _lastSSRC += packet->packetBuffer[11];
-                    _checkSSRC = false;
-                }
-                if (_checkSequenceNumber)
-                {
-                    _firstSequenceNumber
-                        = (unsigned char) packet->packetBuffer[2] << 8;
-                    _firstSequenceNumber
-                        += (unsigned char) packet->packetBuffer[3];
-                    _checkSequenceNumber = false;
-                }
-            }
-            // Signal received packet of frame
-            uint8_t* ptr = (uint8_t*)packet->packetBuffer;
-            uint32_t rtp_timestamp = ptr[4] << 24;
-            rtp_timestamp += ptr[5] << 16;
-            rtp_timestamp += ptr[6] << 8;
-            rtp_timestamp += ptr[7];
-            if (_receive_frame_callback != NULL &&
-                _lastReceiveRTPTimestamp != rtp_timestamp) {
-              _receive_frame_callback->FrameReceived(rtp_timestamp);
-            }
-            _lastReceiveRTPTimestamp = rtp_timestamp;
-            int destination_channel = sender_channel_;
-            if (receive_channels_) {
-              SsrcChannelMap::iterator it = receive_channels_->find(ssrc);
-              if (it == receive_channels_->end()) {
-                return false;
-              }
-              destination_channel = it->second;
-            }
-            _vieNetwork.ReceivedRTPPacket(destination_channel,
-                                          packet->packetBuffer,
-                                          packet->length,
-                                          webrtc::PacketTime());
-            delete packet;
-            packet = NULL;
-        }
-        _crit.Enter();
-    }
-    _crit.Leave();
-    _crit.Enter();
-    while (!_rtcpPackets.empty())
-    {
-        // Take first packet in queue
-        packet = _rtcpPackets.front();
-        int64_t timeToReceive = 0;
-        if (packet)
-        {
-          timeToReceive = packet->receiveTime - NowMs();
-        }
-        else
-        {
-            // There should never be any empty packets in the list.
-            assert(false);
-        }
-        if (timeToReceive > 0)
-        {
-            // No packets to receive yet
-            if (timeToReceive < waitTime && timeToReceive > 0)
-            {
-                waitTime = (unsigned int) timeToReceive;
-            }
-            break;
-        }
-        _rtcpPackets.pop_front();
-        _crit.Leave();
-
-        // Send to ViE
-        if (packet)
-        {
-            uint8_t packet_type = static_cast<uint8_t>(packet->packetBuffer[1]);
-            const uint8_t kSenderReportPacketType = 200;
-            const uint8_t kReceiverReportPacketType = 201;
-            if (packet_type == kSenderReportPacketType) {
-              // Sender report.
-              if (receive_channels_) {
-                for (SsrcChannelMap::iterator it = receive_channels_->begin();
-                    it != receive_channels_->end(); ++it) {
-                  _vieNetwork.ReceivedRTCPPacket(it->second,
-                                                 packet->packetBuffer,
-                                                 packet->length);
-                }
-              } else {
-                _vieNetwork.ReceivedRTCPPacket(sender_channel_,
-                                               packet->packetBuffer,
-                                               packet->length);
-              }
-            } else if (packet_type == kReceiverReportPacketType) {
-              // Receiver report.
-              _vieNetwork.ReceivedRTCPPacket(sender_channel_,
-                                             packet->packetBuffer,
-                                             packet->length);
-            }
-            delete packet;
-            packet = NULL;
-        }
-        _crit.Enter();
-    }
-    _crit.Leave();
-    _event.Wait(waitTime + 1); // Add 1 ms to not call to early...
-    return true;
-}
-
-int64_t TbExternalTransport::NowMs()
-{
-    return webrtc::TickTime::MillisecondTimestamp();
-}
-
-bool TbExternalTransport::UniformLoss(int loss_rate) {
-  int dropThis = rand() % 100;
-  return (dropThis < loss_rate);
-}
-
-bool TbExternalTransport::GilbertElliotLoss(int loss_rate, int burst_length) {
-  // Simulate bursty channel (Gilbert model)
-  // (1st order) Markov chain model with memory of the previous/last
-  // packet state (loss or received)
-
-  // 0 = received state
-  // 1 = loss state
-
-  // probTrans10: if previous packet is lost, prob. to -> received state
-  // probTrans11: if previous packet is lost, prob. to -> loss state
-
-  // probTrans01: if previous packet is received, prob. to -> loss state
-  // probTrans00: if previous packet is received, prob. to -> received
-
-  // Map the two channel parameters (average loss rate and burst length)
-  // to the transition probabilities:
-  double probTrans10 = 100 * (1.0 / burst_length);
-  double probTrans11 = (100.0 - probTrans10);
-  double probTrans01 = (probTrans10 * ( loss_rate / (100.0 - loss_rate)));
-
-  // Note: Random loss (Bernoulli) model is a special case where:
-  // burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
-
-  if (previous_drop_) {
-    // Previous packet was not received.
-    return UniformLoss(probTrans11);
-  } else {
-    return UniformLoss(probTrans01);
-  }
-}
-
-#define PI  3.14159265
-int TbExternalTransport::GaussianRandom(int mean_ms,
-                                        int standard_deviation_ms) {
-  // Creating a Normal distribution variable from two independent uniform
-  // variables based on the Box-Muller transform.
-  double uniform1 = (rand() + 1.0) / (RAND_MAX + 1.0);
-  double uniform2 = (rand() + 1.0) / (RAND_MAX + 1.0);
-  return static_cast<int>(mean_ms + standard_deviation_ms *
-      sqrt(-2 * log(uniform1)) * cos(2 * PI * uniform2));
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_interfaces.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_interfaces.cc
deleted file mode 100644
index fc95967015..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_interfaces.cc
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-TbInterfaces::TbInterfaces(const std::string& test_name) :
-    video_engine(NULL),
-    base(NULL),
-    capture(NULL),
-    render(NULL),
-    rtp_rtcp(NULL),
-    codec(NULL),
-    network(NULL),
-    image_process(NULL)
-{
-    std::string complete_path =
-        webrtc::test::OutputPath() + test_name + "_trace.txt";
-
-    video_engine = webrtc::VideoEngine::Create();
-    EXPECT_TRUE(video_engine != NULL);
-
-    EXPECT_EQ(0, video_engine->SetTraceFile(complete_path.c_str()));
-    EXPECT_EQ(0, video_engine->SetTraceFilter(webrtc::kTraceAll));
-
-    base = webrtc::ViEBase::GetInterface(video_engine);
-    EXPECT_TRUE(base != NULL);
-
-    EXPECT_EQ(0, base->Init());
-
-    capture = webrtc::ViECapture::GetInterface(video_engine);
-    EXPECT_TRUE(capture != NULL);
-
-    rtp_rtcp = webrtc::ViERTP_RTCP::GetInterface(video_engine);
-    EXPECT_TRUE(rtp_rtcp != NULL);
-
-    render = webrtc::ViERender::GetInterface(video_engine);
-    EXPECT_TRUE(render != NULL);
-
-    codec = webrtc::ViECodec::GetInterface(video_engine);
-    EXPECT_TRUE(codec != NULL);
-
-    network = webrtc::ViENetwork::GetInterface(video_engine);
-    EXPECT_TRUE(network != NULL);
-
-    image_process = webrtc::ViEImageProcess::GetInterface(video_engine);
-    EXPECT_TRUE(image_process != NULL);
-}
-
-TbInterfaces::~TbInterfaces(void)
-{
-    EXPECT_EQ(0, image_process->Release());
-    image_process = NULL;
-    EXPECT_EQ(0, codec->Release());
-    codec = NULL;
-    EXPECT_EQ(0, capture->Release());
-    capture = NULL;
-    EXPECT_EQ(0, render->Release());
-    render = NULL;
-    EXPECT_EQ(0, rtp_rtcp->Release());
-    rtp_rtcp = NULL;
-    EXPECT_EQ(0, network->Release());
-    network = NULL;
-    EXPECT_EQ(0, base->Release());
-    base = NULL;
-    EXPECT_TRUE(webrtc::VideoEngine::Delete(video_engine)) <<
-        "Since we have released all interfaces at this point, deletion "
-        "should be successful.";
-    video_engine = NULL;
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_video_channel.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_video_channel.cc
deleted file mode 100644
index 9c5feb11ea..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_video_channel.cc
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/test/libvietest/include/tb_video_channel.h"
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-
-TbVideoChannel::TbVideoChannel(TbInterfaces& Engine,
-                               webrtc::VideoCodecType sendCodec, int width,
-                               int height, int frameRate, int startBitrate)
-    : videoChannel(-1),
-      ViE(Engine) {
-  EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel));
-  channel_transport_.reset(new webrtc::test::VideoChannelTransport(
-      ViE.network, videoChannel));
-
-  webrtc::VideoCodec videoCodec;
-  memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
-  bool sendCodecSet = false;
-  for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++) {
-    EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
-    videoCodec.width = width;
-    videoCodec.height = height;
-    videoCodec.maxFramerate = frameRate;
-
-    if (videoCodec.codecType == sendCodec && sendCodecSet == false) {
-      if (videoCodec.codecType != webrtc::kVideoCodecI420) {
-        videoCodec.startBitrate = startBitrate;
-        videoCodec.maxBitrate = startBitrate * 3;
-      }
-      EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
-      sendCodecSet = true;
-    }
-    if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
-      videoCodec.width = 352;
-      videoCodec.height = 288;
-    }
-    EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
-  }
-  EXPECT_TRUE(sendCodecSet);
-}
-
-TbVideoChannel::~TbVideoChannel() {
-  EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel));
-}
-
-void TbVideoChannel::StartSend(const unsigned short rtp_port,
-                               const char* ip_address) {
-  EXPECT_EQ(0, channel_transport_->SetSendDestination(ip_address, rtp_port));
-  EXPECT_EQ(0, ViE.base->StartSend(videoChannel));
-}
-
-void TbVideoChannel::SetFrameSettings(int width, int height, int frameRate) {
-  webrtc::VideoCodec videoCodec;
-  EXPECT_EQ(0, ViE.codec->GetSendCodec(videoChannel, videoCodec));
-  videoCodec.width = width;
-  videoCodec.height = height;
-  videoCodec.maxFramerate = frameRate;
-
-  EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
-  EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
-}
-
-void TbVideoChannel::StopSend() {
-  EXPECT_EQ(0, ViE.base->StopSend(videoChannel));
-}
-
-void TbVideoChannel::StartReceive(unsigned short rtp_port) {
-  EXPECT_EQ(0, channel_transport_->SetLocalReceiver(rtp_port));
-  EXPECT_EQ(0, ViE.base->StartReceive(videoChannel));
-}
-
-void TbVideoChannel::StopReceive() {
-  EXPECT_EQ(0, ViE.base->StopReceive(videoChannel));
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi b/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi
deleted file mode 100644
index 4dd8614e73..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi
+++ /dev/null
@@ -1,188 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'video_engine_core',
-      'type': 'static_library',
-      'dependencies': [
-        '<(webrtc_root)/common.gyp:webrtc_common',
-
-        # common_video
-       '<(webrtc_root)/common_video/common_video.gyp:common_video',
-
-        # ModulesShared
-        '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
-        '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
-
-        # ModulesVideo
-        '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
-        '<(webrtc_root)/modules/modules.gyp:video_capture_module',
-        '<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
-        '<(webrtc_root)/modules/modules.gyp:video_processing',
-        '<(webrtc_root)/modules/modules.gyp:video_render_module',
-
-        # VoiceEngine
-        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-
-        # system_wrappers
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
-      'sources': [
-        # interface
-        'include/vie_base.h',
-        'include/vie_capture.h',
-        'include/vie_codec.h',
-        'include/vie_errors.h',
-        'include/vie_external_codec.h',
-        'include/vie_image_process.h',
-        'include/vie_network.h',
-        'include/vie_render.h',
-        'include/vie_rtp_rtcp.h',
-
-        # headers
-        'call_stats.h',
-        'encoder_state_feedback.h',
-        'overuse_frame_detector.h',
-        'payload_router.h',
-        'report_block_stats.h',
-        'stream_synchronization.h',
-        'vie_base_impl.h',
-        'vie_capture_impl.h',
-        'vie_codec_impl.h',
-        'vie_defines.h',
-        'vie_external_codec_impl.h',
-        'vie_image_process_impl.h',
-        'vie_impl.h',
-        'vie_network_impl.h',
-        'vie_ref_count.h',
-        'vie_remb.h',
-        'vie_render_impl.h',
-        'vie_rtp_rtcp_impl.h',
-        'vie_shared_data.h',
-        'vie_capturer.h',
-        'vie_channel.h',
-        'vie_channel_group.h',
-        'vie_channel_manager.h',
-        'vie_encoder.h',
-        'vie_file_image.h',
-        'vie_frame_provider_base.h',
-        'vie_input_manager.h',
-        'vie_manager_base.h',
-        'vie_receiver.h',
-        'vie_renderer.h',
-        'vie_render_manager.h',
-        'vie_sender.h',
-        'vie_sync_module.h',
-
-        # ViE
-        'call_stats.cc',
-        'encoder_state_feedback.cc',
-        'overuse_frame_detector.cc',
-        'payload_router.cc',
-        'report_block_stats.cc',
-        'stream_synchronization.cc',
-        'vie_base_impl.cc',
-        'vie_capture_impl.cc',
-        'vie_codec_impl.cc',
-        'vie_external_codec_impl.cc',
-        'vie_image_process_impl.cc',
-        'vie_impl.cc',
-        'vie_network_impl.cc',
-        'vie_ref_count.cc',
-        'vie_render_impl.cc',
-        'vie_rtp_rtcp_impl.cc',
-        'vie_shared_data.cc',
-        'vie_capturer.cc',
-        'vie_channel.cc',
-        'vie_channel_group.cc',
-        'vie_channel_manager.cc',
-        'vie_encoder.cc',
-        'vie_file_image.cc',
-        'vie_frame_provider_base.cc',
-        'vie_input_manager.cc',
-        'vie_manager_base.cc',
-        'vie_receiver.cc',
-        'vie_remb.cc',
-        'vie_renderer.cc',
-        'vie_render_manager.cc',
-        'vie_sender.cc',
-        'vie_sync_module.cc',
-      ], # source
-      # TODO(jschuh): Bug 1348: fix size_t to int truncations.
-      'msvs_disabled_warnings': [ 4267, ],
-    },
-  ], # targets
-  'conditions': [
-    ['include_tests==1', {
-      'targets': [
-        {
-          'target_name': 'video_engine_core_unittests',
-          'type': '<(gtest_target_type)',
-          'dependencies': [
-            'video_engine_core',
-            '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
-            '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(DEPTH)/testing/gmock.gyp:gmock',
-            '<(webrtc_root)/test/test.gyp:test_support_main',
-          ],
-          'sources': [
-            'call_stats_unittest.cc',
-            'encoder_state_feedback_unittest.cc',
-            'overuse_frame_detector_unittest.cc',
-            'payload_router_unittest.cc',
-            'report_block_stats_unittest.cc',
-            'stream_synchronization_unittest.cc',
-            'vie_capturer_unittest.cc',
-            'vie_codec_unittest.cc',
-            'vie_remb_unittest.cc',
-          ],
-          'conditions': [
-            ['OS=="android"', {
-              'dependencies': [
-                '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
-              ],
-            }],
-          ],
-        },
-      ], # targets
-      'conditions': [
-        ['OS=="android"', {
-          'targets': [
-            {
-              'target_name': 'video_engine_core_unittests_apk_target',
-              'type': 'none',
-              'dependencies': [
-                '<(apk_tests_path):video_engine_core_unittests_apk',
-              ],
-            },
-          ],
-        }],
-        ['test_isolation_mode != "noop"', {
-          'targets': [
-            {
-              'target_name': 'video_engine_core_unittests_run',
-              'type': 'none',
-              'dependencies': [
-                'video_engine_core_unittests',
-              ],
-              'includes': [
-                '../build/isolate.gypi',
-              ],
-              'sources': [
-                'video_engine_core_unittests.isolate',
-              ],
-            },
-          ],
-        }],
-      ],
-    }], # include_tests
-  ], # conditions
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate b/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate
deleted file mode 100644
index c8d2fc9026..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate
+++ /dev/null
@@ -1,23 +0,0 @@
-# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-{
-  'conditions': [
-    ['OS=="linux" or OS=="mac" or OS=="win"', {
-      'variables': {
-        'command': [
-          '<(DEPTH)/testing/test_env.py',
-          '<(PRODUCT_DIR)/video_engine_core_unittests<(EXECUTABLE_SUFFIX)',
-        ],
-        'files': [
-          '<(DEPTH)/testing/test_env.py',
-          '<(PRODUCT_DIR)/video_engine_core_unittests<(EXECUTABLE_SUFFIX)',
-        ],
-      },
-    }],
-  ],
-}
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc
deleted file mode 100644
index 86b0d57a4b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc
+++ /dev/null
@@ -1,418 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_base_impl.h"
-
-#include <string>
-#include <utility>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_capturer.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_input_manager.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViEBase* ViEBase::GetInterface(VideoEngine* video_engine) {
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViEBaseImpl* vie_base_impl = vie_impl;
-  (*vie_base_impl)++;  // Increase ref count.
-
-  return vie_base_impl;
-}
-
-int ViEBaseImpl::Release() {
-  (*this)--;  // Decrease ref count.
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_WARNING) << "ViEBase released too many times.";
-    return -1;
-  }
-  return ref_count;
-}
-
-ViEBaseImpl::ViEBaseImpl(const Config& config)
-    : shared_data_(config) {}
-
-ViEBaseImpl::~ViEBaseImpl() {}
-
-int ViEBaseImpl::Init() {
-  return 0;
-}
-
-int ViEBaseImpl::SetVoiceEngine(VoiceEngine* voice_engine) {
-  LOG_F(LS_INFO) << "SetVoiceEngine";
-  if (shared_data_.channel_manager()->SetVoiceEngine(voice_engine) != 0) {
-    shared_data_.SetLastError(kViEBaseVoEFailure);
-    return -1;
-  }
-  return 0;
-}
-
-void ViEBaseImpl::SetLoadManager(CPULoadStateCallbackInvoker* aLoadManager) {
-  shared_data_.set_load_manager(aLoadManager);
-}
-
-int ViEBaseImpl::RegisterCpuOveruseObserver(int video_channel,
-                                            CpuOveruseObserver* observer) {
-  LOG_F(LS_INFO) << "RegisterCpuOveruseObserver on channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder);
-
-  ViEInputManagerScoped is(*(shared_data_.input_manager()));
-  ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
-  if (provider) {
-    ViECapturer* capturer = is.Capture(provider->Id());
-    assert(capturer);
-    capturer->RegisterCpuOveruseObserver(observer);
-  }
-
-  shared_data_.overuse_observers()->insert(
-      std::pair<int, CpuOveruseObserver*>(video_channel, observer));
-  return 0;
-}
-
-int ViEBaseImpl::SetCpuOveruseOptions(int video_channel,
-                                      const CpuOveruseOptions& options) {
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder);
-
-  ViEInputManagerScoped is(*(shared_data_.input_manager()));
-  ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
-  if (provider) {
-    ViECapturer* capturer = is.Capture(provider->Id());
-    if (capturer) {
-      capturer->SetCpuOveruseOptions(options);
-      return 0;
-    }
-  }
-  return -1;
-}
-
-void ViEBaseImpl::RegisterCpuOveruseMetricsObserver(
-    int video_channel,
-    CpuOveruseMetricsObserver* observer) {
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder);
-
-  ViEInputManagerScoped is(*(shared_data_.input_manager()));
-  ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
-  assert(provider != NULL);
-
-  ViECapturer* capturer = is.Capture(provider->Id());
-  assert(capturer);
-
-  capturer->RegisterCpuOveruseMetricsObserver(observer);
-}
-
-int ViEBaseImpl::GetCpuOveruseMetrics(int video_channel,
-                                      CpuOveruseMetrics* metrics) {
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder);
-
-  ViEInputManagerScoped is(*(shared_data_.input_manager()));
-  ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
-  if (provider) {
-    ViECapturer* capturer = is.Capture(provider->Id());
-    if (capturer) {
-      capturer->GetCpuOveruseMetrics(metrics);
-      return 0;
-    }
-  }
-  return -1;
-}
-
-void ViEBaseImpl::RegisterSendSideDelayObserver(
-    int channel, SendSideDelayObserver* observer) {
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(channel);
-  assert(vie_channel);
-  vie_channel->RegisterSendSideDelayObserver(observer);
-}
-
-int ViEBaseImpl::CreateChannel(int& video_channel) {  // NOLINT
-  return CreateChannel(video_channel, static_cast<const Config*>(NULL));
-}
-
-int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
-                               const Config* config) {
-  if (shared_data_.channel_manager()->CreateChannel(&video_channel,
-                                                    config) == -1) {
-    video_channel = -1;
-    shared_data_.SetLastError(kViEBaseChannelCreationFailed);
-    return -1;
-  }
-  LOG(LS_INFO) << "Video channel created: " << video_channel;
-  return 0;
-}
-
-int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
-                               int original_channel) {
-  return CreateChannel(video_channel, original_channel, true, false);
-}
-
-int ViEBaseImpl::CreateChannelWithoutDefaultEncoder(
-    int& video_channel,  // NOLINT
-    int original_channel) {
-  return CreateChannel(video_channel, original_channel, true, true);
-}
-
-int ViEBaseImpl::CreateReceiveChannel(int& video_channel,  // NOLINT
-                                      int original_channel) {
-  return CreateChannel(video_channel, original_channel, false, true);
-}
-
-int ViEBaseImpl::DeleteChannel(const int video_channel) {
-  {
-    ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-    ViEChannel* vie_channel = cs.Channel(video_channel);
-    if (!vie_channel) {
-      shared_data_.SetLastError(kViEBaseInvalidChannelId);
-      return -1;
-    }
-
-    // Deregister the ViEEncoder if no other channel is using it.
-    ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-    if (!cs.ChannelUsingViEEncoder(video_channel)) {
-      ViEInputManagerScoped is(*(shared_data_.input_manager()));
-      ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
-      if (provider) {
-        provider->DeregisterFrameCallback(vie_encoder);
-      }
-    }
-  }
-
-  if (shared_data_.channel_manager()->DeleteChannel(video_channel) == -1) {
-    shared_data_.SetLastError(kViEBaseUnknownError);
-    return -1;
-  }
-  LOG(LS_INFO) << "Channel deleted " << video_channel;
-  return 0;
-}
-
-int ViEBaseImpl::ConnectAudioChannel(const int video_channel,
-                                     const int audio_channel) {
-  LOG_F(LS_INFO) << "ConnectAudioChannel, video channel " << video_channel
-                 << ", audio channel " << audio_channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  if (!cs.Channel(video_channel)) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-
-  if (shared_data_.channel_manager()->ConnectVoiceChannel(video_channel,
-                                                          audio_channel) != 0) {
-    shared_data_.SetLastError(kViEBaseVoEFailure);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEBaseImpl::DisconnectAudioChannel(const int video_channel) {
-  LOG_F(LS_INFO) << "DisconnectAudioChannel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  if (!cs.Channel(video_channel)) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-
-  if (shared_data_.channel_manager()->DisconnectVoiceChannel(
-      video_channel) != 0) {
-    shared_data_.SetLastError(kViEBaseVoEFailure);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEBaseImpl::StartSend(const int video_channel) {
-  LOG_F(LS_INFO) << "StartSend: " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder != NULL);
-  if (vie_encoder->Owner() != video_channel) {
-    LOG_F(LS_ERROR) <<  "Can't start send on a receive only channel.";
-    shared_data_.SetLastError(kViEBaseReceiveOnlyChannel);
-    return -1;
-  }
-
-  // Pause and trigger a key frame.
-  vie_encoder->Pause();
-  int32_t error = vie_channel->StartSend();
-  if (error != 0) {
-    vie_encoder->Restart();
-    if (error == kViEBaseAlreadySending) {
-      shared_data_.SetLastError(kViEBaseAlreadySending);
-    }
-    LOG_F(LS_ERROR) << "Could not start sending " << video_channel;
-    shared_data_.SetLastError(kViEBaseUnknownError);
-    return -1;
-  }
-  vie_encoder->SendKeyFrame();
-  vie_encoder->Restart();
-  return 0;
-}
-
-int ViEBaseImpl::StopSend(const int video_channel) {
-  LOG_F(LS_INFO) << "StopSend " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-
-  int32_t error = vie_channel->StopSend();
-  if (error != 0) {
-    if (error == kViEBaseNotSending) {
-      shared_data_.SetLastError(kViEBaseNotSending);
-    } else {
-      LOG_F(LS_ERROR) << "Could not stop sending " << video_channel;
-      shared_data_.SetLastError(kViEBaseUnknownError);
-    }
-    return -1;
-  }
-  return 0;
-}
-
-int ViEBaseImpl::StartReceive(const int video_channel) {
-  LOG_F(LS_INFO) << "StartReceive " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->StartReceive() != 0) {
-    shared_data_.SetLastError(kViEBaseUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEBaseImpl::StopReceive(const int video_channel) {
-  LOG_F(LS_INFO) << "StopReceive " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->StopReceive() != 0) {
-    shared_data_.SetLastError(kViEBaseUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEBaseImpl::GetVersion(char version[1024]) {
-  assert(version != NULL);
-  strcpy(version, "VideoEngine 42");
-  return 0;
-}
-
-int ViEBaseImpl::LastError() {
-  return shared_data_.LastErrorInternal();
-}
-
-int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
-                               int original_channel,
-                               bool sender,
-                               bool disable_default_encoder) {
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  if (!cs.Channel(original_channel)) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return -1;
-  }
-
-  if (shared_data_.channel_manager()->CreateChannel(
-          &video_channel, original_channel, sender, disable_default_encoder) ==
-      -1) {
-    video_channel = -1;
-    shared_data_.SetLastError(kViEBaseChannelCreationFailed);
-    return -1;
-  }
-  LOG_F(LS_INFO) << "VideoChannel created: " << video_channel
-                 << ", base channel " << original_channel
-                 << ", is send channel : " << sender;
-  return 0;
-}
-
-void ViEBaseImpl::RegisterSendStatisticsProxy(
-    int channel,
-    SendStatisticsProxy* send_statistics_proxy) {
-  LOG_F(LS_VERBOSE) << "RegisterSendStatisticsProxy on channel " << channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return;
-  }
-  ViEEncoder* vie_encoder = cs.Encoder(channel);
-  assert(vie_encoder);
-
-  vie_encoder->RegisterSendStatisticsProxy(send_statistics_proxy);
-}
-
-void ViEBaseImpl::RegisterReceiveStatisticsProxy(
-    int channel,
-    ReceiveStatisticsProxy* receive_statistics_proxy) {
-  LOG_F(LS_VERBOSE) << "RegisterReceiveStatisticsProxy on channel " << channel;
-  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(channel);
-  if (!vie_channel) {
-    shared_data_.SetLastError(kViEBaseInvalidChannelId);
-    return;
-  }
-  vie_channel->RegisterReceiveStatisticsProxy(receive_statistics_proxy);
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h
deleted file mode 100644
index 9f92178462..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
-
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-class Config;
-class Module;
-class VoiceEngine;
-
-class ViEBaseImpl
-    : public ViEBase,
-      public ViERefCount {
- public:
-  virtual int Release();
-
-  // Implements ViEBase.
-  virtual int Init();
-  virtual int SetVoiceEngine(VoiceEngine* voice_engine);
-  virtual int RegisterCpuOveruseObserver(int channel,
-                                         CpuOveruseObserver* observer);
-  virtual int SetCpuOveruseOptions(int channel,
-                                   const CpuOveruseOptions& options);
-  void RegisterCpuOveruseMetricsObserver(
-      int channel,
-      CpuOveruseMetricsObserver* observer) override;
-  virtual int GetCpuOveruseMetrics(int channel,
-                                   CpuOveruseMetrics* metrics);
-  void RegisterSendSideDelayObserver(int channel,
-                                     SendSideDelayObserver* observer) override;
-  virtual void SetLoadManager(CPULoadStateCallbackInvoker* aLoadManager);
-  virtual int CreateChannel(int& video_channel);  // NOLINT
-  virtual int CreateChannel(int& video_channel,  // NOLINT
-                            const Config* config);
-  virtual int CreateChannel(int& video_channel,  // NOLINT
-                            int original_channel);
-  virtual int CreateChannelWithoutDefaultEncoder(int& video_channel,  // NOLINT
-                                                 int original_channel);
-
-  virtual int CreateReceiveChannel(int& video_channel,  // NOLINT
-                                   int original_channel);
-  virtual int DeleteChannel(const int video_channel);
-  virtual int ConnectAudioChannel(const int video_channel,
-                                  const int audio_channel);
-  virtual int DisconnectAudioChannel(const int video_channel);
-  virtual int StartSend(const int video_channel);
-  virtual int StopSend(const int video_channel);
-  virtual int StartReceive(const int video_channel);
-  virtual int StopReceive(const int video_channel);
-  virtual int GetVersion(char version[1024]);
-  virtual int LastError();
-
- protected:
-  explicit ViEBaseImpl(const Config& config);
-  virtual ~ViEBaseImpl();
-
-  ViESharedData* shared_data() { return &shared_data_; }
-
- private:
-  int CreateChannel(int& video_channel, int original_channel,  // NOLINT
-                    bool sender, bool disable_default_encoder);
-
-  void RegisterSendStatisticsProxy(
-      int channel,
-      SendStatisticsProxy* send_statistics_proxy) override;
-  void RegisterReceiveStatisticsProxy(
-      int channel,
-      ReceiveStatisticsProxy* receive_statistics_proxy) override;
-  // ViEBaseImpl owns ViESharedData used by all interface implementations.
-  ViESharedData shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc
deleted file mode 100644
index c89a325c7d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc
+++ /dev/null
@@ -1,412 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_capture_impl.h"
-
-#include <map>
-
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_capturer.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_input_manager.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-class CpuOveruseObserver;
-
-ViECapture* ViECapture::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViECaptureImpl* vie_capture_impl = vie_impl;
-  // Increase ref count.
-  (*vie_capture_impl)++;
-  return vie_capture_impl;
-#else
-  return NULL;
-#endif
-}
-
-int ViECaptureImpl::Release() {
-  // Decrease ref count
-  (*this)--;
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_WARNING) << "ViECapture released too many times.";
-    shared_data_->SetLastError(kViEAPIDoesNotExist);
-    return -1;
-  }
-  return ref_count;
-}
-
-ViECaptureImpl::ViECaptureImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {}
-
-ViECaptureImpl::~ViECaptureImpl() {}
-
-int ViECaptureImpl::NumberOfCaptureDevices() {
-  return  shared_data_->input_manager()->NumberOfCaptureDevices();
-}
-
-
-int ViECaptureImpl::GetCaptureDevice(unsigned int list_number,
-                                     char* device_nameUTF8,
-                                     unsigned int device_nameUTF8Length,
-                                     char* unique_idUTF8,
-                                     unsigned int unique_idUTF8Length,
-                                     pid_t* pid) {
-  return shared_data_->input_manager()->GetDeviceName(
-      list_number,
-      device_nameUTF8, device_nameUTF8Length,
-      unique_idUTF8, unique_idUTF8Length, pid);
-}
-
-int ViECaptureImpl::AllocateCaptureDevice(
-  const char* unique_idUTF8,
-  const unsigned int unique_idUTF8Length,
-  int& capture_id) {
-  LOG(LS_INFO) << "AllocateCaptureDevice " << unique_idUTF8;
-  const int32_t result =
-      shared_data_->input_manager()->CreateCaptureDevice(
-          unique_idUTF8,
-          static_cast<const uint32_t>(unique_idUTF8Length),
-          capture_id);
-  if (result != 0) {
-    shared_data_->SetLastError(result);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::AllocateExternalCaptureDevice(
-  int& capture_id, ViEExternalCapture*& external_capture) {
-  const int32_t result =
-      shared_data_->input_manager()->CreateExternalCaptureDevice(
-          external_capture, capture_id);
-
-  if (result != 0) {
-    shared_data_->SetLastError(result);
-    return -1;
-  }
-  LOG(LS_INFO) << "External capture device allocated: " << capture_id;
-  return 0;
-}
-
-int ViECaptureImpl::AllocateCaptureDevice(
-    VideoCaptureModule& capture_module, int& capture_id) {  // NOLINT
-  int32_t result = shared_data_->input_manager()->CreateCaptureDevice(
-      &capture_module, capture_id);
-  if (result != 0) {
-    shared_data_->SetLastError(result);
-    return -1;
-  }
-  LOG(LS_INFO) << "External capture device, by module, allocated: "
-               << capture_id;
-  return 0;
-}
-
-
-int ViECaptureImpl::ReleaseCaptureDevice(const int capture_id) {
-  LOG(LS_INFO) << "ReleaseCaptureDevice " << capture_id;
-  {
-    ViEInputManagerScoped is((*(shared_data_->input_manager())));
-    ViECapturer* vie_capture = is.Capture(capture_id);
-    if (!vie_capture) {
-      shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-      return -1;
-    }
-  }
-  return shared_data_->input_manager()->DestroyCaptureDevice(capture_id);
-}
-
-int ViECaptureImpl::ConnectCaptureDevice(const int capture_id,
-                                         const int video_channel) {
-  LOG(LS_INFO) << "Connect capture id " << capture_id
-               << " to channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    LOG(LS_ERROR) << "Channel doesn't exist.";
-    shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
-    return -1;
-  }
-  if (vie_encoder->Owner() != video_channel) {
-    LOG(LS_ERROR) << "Can't connect capture device to a receive device.";
-    shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
-    return -1;
-  }
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  //  Check if the encoder already has a connected frame provider
-  if (is.FrameProvider(vie_encoder) != NULL) {
-    LOG(LS_ERROR) << "Channel already connected to capture device.";
-    shared_data_->SetLastError(kViECaptureDeviceAlreadyConnected);
-    return -1;
-  }
-  if (vie_capture->RegisterFrameCallback(video_channel, vie_encoder) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  std::map<int, CpuOveruseObserver*>::iterator it =
-      shared_data_->overuse_observers()->find(video_channel);
-  if (it != shared_data_->overuse_observers()->end()) {
-    vie_capture->RegisterCpuOveruseObserver(it->second);
-  }
-  return 0;
-}
-
-
-int ViECaptureImpl::DisconnectCaptureDevice(const int video_channel) {
-  LOG(LS_INFO) << "DisconnectCaptureDevice " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    LOG(LS_ERROR) << "Channel doesn't exist.";
-    shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
-    return -1;
-  }
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
-  if (!frame_provider) {
-    shared_data_->SetLastError(kViECaptureDeviceNotConnected);
-    return -1;
-  }
-  if (frame_provider->Id() < kViECaptureIdBase ||
-      frame_provider->Id() > kViECaptureIdMax) {
-    shared_data_->SetLastError(kViECaptureDeviceNotConnected);
-    return -1;
-  }
-
-  ViECapturer* vie_capture = is.Capture(frame_provider->Id());
-  assert(vie_capture);
-  vie_capture->RegisterCpuOveruseObserver(NULL);
-  if (frame_provider->DeregisterFrameCallback(vie_encoder) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-
-  return 0;
-}
-
-int ViECaptureImpl::StartCapture(const int capture_id,
-                                 const CaptureCapability& capture_capability) {
-  LOG(LS_INFO) << "StartCapture " << capture_id;
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  if (vie_capture->Started()) {
-    shared_data_->SetLastError(kViECaptureDeviceAlreadyStarted);
-    return -1;
-  }
-  if (vie_capture->Start(capture_capability) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::StopCapture(const int capture_id) {
-  LOG(LS_INFO) << "StopCapture " << capture_id;
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  if (!vie_capture->Started()) {
-    shared_data_->SetLastError(kViECaptureDeviceNotStarted);
-    return 0;
-  }
-  if (vie_capture->Stop() != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::SetVideoRotation(const int capture_id,
-                                     const VideoRotation rotation) {
-  LOG(LS_INFO) << "SetRotateCaptureFrames for " << capture_id << ", rotation "
-               << static_cast<int>(rotation);
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  if (vie_capture->SetVideoRotation(rotation) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::SetCaptureDelay(const int capture_id,
-                                    const unsigned int capture_delay_ms) {
-  LOG(LS_INFO) << "SetCaptureDelay " << capture_delay_ms
-               << ", for device " << capture_id;
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-
-  if (vie_capture->SetCaptureDelay(capture_delay_ms) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::NumberOfCapabilities(
-    const char* unique_idUTF8,
-    const unsigned int unique_idUTF8Length) {
-
-  return shared_data_->input_manager()->NumberOfCaptureCapabilities(
-      unique_idUTF8);
-}
-
-
-int ViECaptureImpl::GetCaptureCapability(const char* unique_idUTF8,
-                                         const unsigned int unique_idUTF8Length,
-                                         const unsigned int capability_number,
-                                         CaptureCapability& capability) {
-
-  if (shared_data_->input_manager()->GetCaptureCapability(
-          unique_idUTF8, capability_number, capability) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::ShowCaptureSettingsDialogBox(
-    const char* unique_idUTF8,
-    const unsigned int unique_idUTF8Length,
-    const char* dialog_title,
-    void* parent_window,
-    const unsigned int x,
-    const unsigned int y) {
-  return shared_data_->input_manager()->DisplayCaptureSettingsDialogBox(
-           unique_idUTF8, dialog_title,
-           parent_window, x, y);
-}
-
-int ViECaptureImpl::GetOrientation(const char* unique_idUTF8,
-                                   VideoRotation& orientation) {
-  if (shared_data_->input_manager()->GetOrientation(
-      unique_idUTF8,
-      orientation) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-
-int ViECaptureImpl::EnableBrightnessAlarm(const int capture_id,
-                                          const bool enable) {
-  LOG(LS_INFO) << "EnableBrightnessAlarm for device " << capture_id
-               << ", status " << enable;
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  if (vie_capture->EnableBrightnessAlarm(enable) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::RegisterObserver(const int capture_id,
-                                     ViECaptureObserver& observer) {
-  LOG(LS_INFO) << "Register capture observer " << capture_id;
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  if (vie_capture->IsObserverRegistered()) {
-    LOG_F(LS_ERROR) << "Observer already registered.";
-    shared_data_->SetLastError(kViECaptureObserverAlreadyRegistered);
-    return -1;
-  }
-  if (vie_capture->RegisterObserver(&observer) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::RegisterInputObserver(ViEInputObserver* observer) {
-  if (shared_data_->input_manager()->RegisterObserver(observer) != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::DeregisterObserver(const int capture_id) {
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-  if (!vie_capture->IsObserverRegistered()) {
-    shared_data_->SetLastError(kViECaptureDeviceObserverNotRegistered);
-    return -1;
-  }
-
-  if (vie_capture->DeRegisterObserver() != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECaptureImpl::DeregisterInputObserver() {
-  if (shared_data_->input_manager()->DeRegisterObserver() != 0) {
-    shared_data_->SetLastError(kViECaptureDeviceUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h
deleted file mode 100644
index a8707bdb70..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
-
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViECaptureImpl
-    : public ViECapture,
-      public ViERefCount {
- public:
-  // Implements ViECapture.
-  virtual int Release();
-  virtual int NumberOfCaptureDevices();
-  virtual int GetCaptureDevice(unsigned int list_number, char* device_nameUTF8,
-                               const unsigned int device_nameUTF8Length,
-                               char* unique_idUTF8,
-                               const unsigned int unique_idUTF8Length,
-                               pid_t* pid = nullptr);
-
-  virtual int AllocateCaptureDevice(const char* unique_idUTF8,
-                                    const unsigned int unique_idUTF8Length,
-                                    int& capture_id);
-  virtual int AllocateCaptureDevice(
-      VideoCaptureModule& capture_module, int& capture_id);  // NOLINT
-  virtual int AllocateExternalCaptureDevice(
-      int& capture_id, ViEExternalCapture *&external_capture);
-  virtual int ReleaseCaptureDevice(const int capture_id);
-
-  virtual int ConnectCaptureDevice(const int capture_id,
-                                   const int video_channel);
-  virtual int DisconnectCaptureDevice(const int video_channel);
-  virtual int StartCapture(
-      const int capture_id,
-      const CaptureCapability& capture_capability = CaptureCapability());
-  virtual int StopCapture(const int capture_id);
-  virtual int SetVideoRotation(const int capture_id,
-                               const VideoRotation rotation);
-  virtual int SetCaptureDelay(const int capture_id,
-                              const unsigned int capture_delay_ms);
-  virtual int NumberOfCapabilities(const char* unique_idUTF8,
-                                   const unsigned int unique_idUTF8Length);
-  virtual int GetCaptureCapability(const char* unique_idUTF8,
-                                   const unsigned int unique_idUTF8Length,
-                                   const unsigned int capability_number,
-                                   CaptureCapability& capability);
-  virtual int ShowCaptureSettingsDialogBox(
-    const char* unique_idUTF8, const unsigned int unique_idUTF8Length,
-    const char* dialog_title, void* parent_window = NULL,
-    const unsigned int x = 200, const unsigned int y = 200);
-  virtual int GetOrientation(const char* unique_idUTF8,
-                             VideoRotation& orientation);
-  virtual int EnableBrightnessAlarm(const int capture_id, const bool enable);
-  virtual int RegisterObserver(const int capture_id,
-                               ViECaptureObserver& observer);
-  virtual int RegisterInputObserver(ViEInputObserver* observer);
-  virtual int DeregisterObserver(const int capture_id);
-  virtual int DeregisterInputObserver();
-
- protected:
-  explicit ViECaptureImpl(ViESharedData* shared_data);
-  virtual ~ViECaptureImpl();
-
- private:
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc b/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc
deleted file mode 100644
index 7041b6b0fe..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc
+++ /dev/null
@@ -1,603 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_capturer.h"
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/overuse_frame_detector.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/desktop_capture_impl.h"
-
-namespace webrtc {
-
-const int kThreadWaitTimeMs = 100;
-
-class RegistrableCpuOveruseMetricsObserver : public CpuOveruseMetricsObserver {
- public:
-  void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override {
-    rtc::CritScope lock(&crit_);
-    if (observer_)
-      observer_->CpuOveruseMetricsUpdated(metrics);
-    metrics_ = metrics;
-  }
-
-  CpuOveruseMetrics GetCpuOveruseMetrics() const {
-    rtc::CritScope lock(&crit_);
-    return metrics_;
-  }
-
-  void Set(CpuOveruseMetricsObserver* observer) {
-    rtc::CritScope lock(&crit_);
-    observer_ = observer;
-  }
-
- private:
-  mutable rtc::CriticalSection crit_;
-  CpuOveruseMetricsObserver* observer_ GUARDED_BY(crit_) = nullptr;
-  CpuOveruseMetrics metrics_ GUARDED_BY(crit_);
-};
-
-ViECapturer::ViECapturer(int capture_id,
-                         int engine_id,
-                         const Config& config,
-                         ProcessThread& module_process_thread)
-    : ViEFrameProviderBase(capture_id, engine_id),
-      capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      effects_and_stats_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      capture_module_(NULL),
-      use_external_capture_(false),
-      module_process_thread_(module_process_thread),
-      capture_id_(capture_id),
-      incoming_frame_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      capture_thread_(ThreadWrapper::CreateThread(
-          ViECaptureThreadFunction, this, "ViECaptureThread")),
-      capture_event_(*EventWrapper::Create()),
-      deliver_event_(*EventWrapper::Create()),
-      stop_(0),
-      last_captured_timestamp_(0),
-      delta_ntp_internal_ms_(
-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() -
-          TickTime::MillisecondTimestamp()),
-      effect_filter_(NULL),
-      image_proc_module_(NULL),
-      image_proc_module_ref_counter_(0),
-      deflicker_frame_stats_(NULL),
-      brightness_frame_stats_(NULL),
-      current_brightness_level_(Normal),
-      reported_brightness_level_(Normal),
-      observer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      observer_(NULL),
-      cpu_overuse_metrics_observer_(new RegistrableCpuOveruseMetricsObserver()),
-      overuse_detector_(
-          new OveruseFrameDetector(Clock::GetRealTimeClock(),
-                                   cpu_overuse_metrics_observer_.get())),
-      config_(config) {
-  capture_thread_->Start();
-  capture_thread_->SetPriority(kHighPriority);
-  module_process_thread_.RegisterModule(overuse_detector_.get());
-}
-
-ViECapturer::~ViECapturer() {
-  module_process_thread_.DeRegisterModule(overuse_detector_.get());
-
-  // Stop the thread.
-  rtc::AtomicOps::Increment(&stop_);
-  capture_event_.Set();
-
-  // Stop the camera input.
-  if (capture_module_) {
-    module_process_thread_.DeRegisterModule(capture_module_);
-    capture_module_->DeRegisterCaptureDataCallback();
-    capture_module_->Release();
-    capture_module_ = NULL;
-  }
-
-  capture_thread_->Stop();
-  delete &capture_event_;
-  delete &deliver_event_;
-
-  if (image_proc_module_) {
-    VideoProcessingModule::Destroy(image_proc_module_);
-  }
-  if (deflicker_frame_stats_) {
-    delete deflicker_frame_stats_;
-    deflicker_frame_stats_ = NULL;
-  }
-  delete brightness_frame_stats_;
-}
-
-ViECapturer* ViECapturer::CreateViECapture(
-    int capture_id,
-    int engine_id,
-    const Config& config,
-    VideoCaptureModule* capture_module,
-    ProcessThread& module_process_thread) {
-  ViECapturer* capture = new ViECapturer(capture_id, engine_id, config,
-                                         module_process_thread);
-  if (!capture || capture->Init(capture_module) != 0) {
-    delete capture;
-    capture = NULL;
-  }
-  return capture;
-}
-
-int32_t ViECapturer::Init(VideoCaptureModule* capture_module) {
-  assert(capture_module_ == NULL);
-  capture_module_ = capture_module;
-  capture_module_->RegisterCaptureDataCallback(*this);
-  capture_module_->AddRef();
-  module_process_thread_.RegisterModule(capture_module_);
-  return 0;
-}
-
-ViECapturer* ViECapturer::CreateViECapture(
-    int capture_id,
-    int engine_id,
-    const Config& config,
-    const char* device_unique_idUTF8,
-    const uint32_t device_unique_idUTF8Length,
-    ProcessThread& module_process_thread) {
-  ViECapturer* capture = new ViECapturer(capture_id, engine_id, config,
-                                         module_process_thread);
-  if (!capture ||
-      capture->Init(device_unique_idUTF8, device_unique_idUTF8Length) != 0) {
-    delete capture;
-    capture = NULL;
-  }
-  return capture;
-}
-
-int32_t ViECapturer::Init(const char* device_unique_idUTF8,
-                          uint32_t device_unique_idUTF8Length) {
-  assert(capture_module_ == NULL);
-  CaptureDeviceType type = config_.Get<CaptureDeviceInfo>().type;
-
-  if(type != CaptureDeviceType::Camera) {
-#if !defined(ANDROID) && !defined(WEBRTC_IOS)
-    capture_module_ = DesktopCaptureImpl::Create(
-      ViEModuleId(engine_id_, capture_id_), device_unique_idUTF8, type);
-#endif
-  } else if (device_unique_idUTF8 == NULL) {
-    use_external_capture_ = true;
-    // YYY: was      ViEModuleId(engine_id_, capture_id_), external_capture_module_);
-    return 0;
-  } else {
-    capture_module_ = VideoCaptureFactory::Create(
-      ViEModuleId(engine_id_, capture_id_), device_unique_idUTF8);
-  }
-  if (!capture_module_) {
-    return -1;
-  }
-  capture_module_->AddRef();
-  capture_module_->RegisterCaptureDataCallback(*this);
-  module_process_thread_.RegisterModule(capture_module_);
-
-  return 0;
-}
-
-int ViECapturer::FrameCallbackChanged() {
-  if (use_external_capture_)
-    return -1;
-  if (Started() && !CaptureCapabilityFixed()) {
-    // Reconfigure the camera if a new size is required and the capture device
-    // does not provide encoded frames.
-    int best_width;
-    int best_height;
-    int best_frame_rate;
-    VideoCaptureCapability capture_settings;
-    capture_module_->CaptureSettings(capture_settings);
-    GetBestFormat(&best_width, &best_height, &best_frame_rate);
-    if (best_width != 0 && best_height != 0 && best_frame_rate != 0) {
-      if (best_width != capture_settings.width ||
-          best_height != capture_settings.height ||
-          best_frame_rate != capture_settings.maxFPS ||
-          capture_settings.codecType != kVideoCodecUnknown) {
-        Stop();
-        Start(requested_capability_);
-      }
-    }
-  }
-  return 0;
-}
-
-int32_t ViECapturer::Start(const CaptureCapability& capture_capability) {
-  if (use_external_capture_)
-    return -1;
-  int width;
-  int height;
-  int frame_rate;
-  VideoCaptureCapability capability;
-  requested_capability_ = capture_capability;
-  CaptureDeviceType type = config_.Get<CaptureDeviceInfo>().type;
-
-  if (!CaptureCapabilityFixed()) {
-    // Ask the observers for best size.
-    GetBestFormat(&width, &height, &frame_rate);
-    if (width == 0) {
-      width = kViECaptureDefaultWidth;
-    }
-    if (height == 0) {
-      height = kViECaptureDefaultHeight;
-    }
-    if (frame_rate == 0) {
-      if (type == Screen || type == Window || type == Application) {
-        frame_rate = kViEScreenCaptureDefaultFramerate;
-      } else {
-        frame_rate = kViECaptureDefaultFramerate;
-      }
-    }
-    capability.height = height;
-    capability.width = width;
-    capability.maxFPS = frame_rate;
-    capability.rawType = kVideoI420;
-    capability.codecType = kVideoCodecUnknown;
-  } else {
-    // Width, height and type specified with call to Start, not set by
-    // observers.
-    capability.width = requested_capability_.width;
-    capability.height = requested_capability_.height;
-    capability.maxFPS = requested_capability_.maxFPS;
-    capability.rawType = requested_capability_.rawType;
-    capability.interlaced = requested_capability_.interlaced;
-  }
-  return capture_module_->StartCapture(capability);
-}
-
-int32_t ViECapturer::Stop() {
-  if (use_external_capture_)
-    return -1;
-  requested_capability_ = CaptureCapability();
-  return capture_module_->StopCapture();
-}
-
-bool ViECapturer::Started() {
-  if (use_external_capture_)
-    return false;
-  return capture_module_->CaptureStarted();
-}
-
-const char* ViECapturer::CurrentDeviceName() const {
-  if (use_external_capture_)
-    return "";
-  return capture_module_->CurrentDeviceName();
-}
-
-void ViECapturer::RegisterCpuOveruseObserver(CpuOveruseObserver* observer) {
-  overuse_detector_->SetObserver(observer);
-}
-
-void ViECapturer::SetCpuOveruseOptions(const CpuOveruseOptions& options) {
-  overuse_detector_->SetOptions(options);
-}
-
-void ViECapturer::RegisterCpuOveruseMetricsObserver(
-    CpuOveruseMetricsObserver* observer) {
-  cpu_overuse_metrics_observer_->Set(observer);
-}
-
-void ViECapturer::GetCpuOveruseMetrics(CpuOveruseMetrics* metrics) const {
-  *metrics = cpu_overuse_metrics_observer_->GetCpuOveruseMetrics();
-}
-
-int32_t ViECapturer::SetCaptureDelay(int32_t delay_ms) {
-  if (use_external_capture_)
-    return -1;
-  capture_module_->SetCaptureDelay(delay_ms);
-  return 0;
-}
-
-int32_t ViECapturer::SetVideoRotation(const VideoRotation rotation) {
-  if (use_external_capture_)
-    return -1;
-  return capture_module_->SetCaptureRotation(rotation);
-}
-
-void ViECapturer::IncomingFrame(const I420VideoFrame& frame) {
-  OnIncomingCapturedFrame(-1, frame);
-}
-
-void ViECapturer::OnIncomingCapturedFrame(const int32_t capture_id,
-                                          const I420VideoFrame& video_frame) {
-  I420VideoFrame incoming_frame = video_frame;
-
-  if (incoming_frame.ntp_time_ms() != 0) {
-    // If a NTP time stamp is set, this is the time stamp we will use.
-    incoming_frame.set_render_time_ms(
-        incoming_frame.ntp_time_ms() - delta_ntp_internal_ms_);
-  } else {  // NTP time stamp not set.
-    int64_t render_time = incoming_frame.render_time_ms() != 0 ?
-        incoming_frame.render_time_ms() : TickTime::MillisecondTimestamp();
-
-    // Make sure we render this frame earlier since we know the render time set
-    // is slightly off since it's being set when the frame was received
-    // from the camera, and not when the camera actually captured the frame.
-    render_time -= FrameDelay();
-    incoming_frame.set_render_time_ms(render_time);
-    incoming_frame.set_ntp_time_ms(
-        render_time + delta_ntp_internal_ms_);
-  }
-
-  // Convert NTP time, in ms, to RTP timestamp.
-  const int kMsToRtpTimestamp = 90;
-  incoming_frame.set_timestamp(kMsToRtpTimestamp *
-      static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
-
-  CriticalSectionScoped cs(capture_cs_.get());
-  if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
-    // We don't allow the same capture time for two frames, drop this one.
-    LOG(LS_WARNING) << "Same/old NTP timestamp for incoming frame. Dropping.";
-    return;
-  }
-
-  captured_frame_.ShallowCopy(incoming_frame);
-  last_captured_timestamp_ = incoming_frame.ntp_time_ms();
-
-  overuse_detector_->FrameCaptured(captured_frame_.width(),
-                                   captured_frame_.height(),
-                                   captured_frame_.render_time_ms());
-
-  TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
-                           "render_time", video_frame.render_time_ms());
-
-  capture_event_.Set();
-}
-
-void ViECapturer::OnCaptureDelayChanged(const int32_t id,
-                                        const int32_t delay) {
-  LOG(LS_INFO) << "Capture delayed change to " << delay
-               << " for device " << id;
-
-  // Deliver the network delay to all registered callbacks.
-  ViEFrameProviderBase::SetFrameDelay(delay);
-}
-
-int32_t ViECapturer::RegisterEffectFilter(
-    ViEEffectFilter* effect_filter) {
-  CriticalSectionScoped cs(effects_and_stats_cs_.get());
-
-  if (effect_filter != NULL && effect_filter_ != NULL) {
-    LOG_F(LS_ERROR) << "Effect filter already registered.";
-    return -1;
-  }
-  effect_filter_ = effect_filter;
-  return 0;
-}
-
-int32_t ViECapturer::IncImageProcRefCount() {
-  if (!image_proc_module_) {
-    assert(image_proc_module_ref_counter_ == 0);
-    image_proc_module_ = VideoProcessingModule::Create(
-        ViEModuleId(engine_id_, capture_id_));
-    if (!image_proc_module_) {
-      LOG_F(LS_ERROR) << "Could not create video processing module.";
-      return -1;
-    }
-  }
-  image_proc_module_ref_counter_++;
-  return 0;
-}
-
-int32_t ViECapturer::DecImageProcRefCount() {
-  image_proc_module_ref_counter_--;
-  if (image_proc_module_ref_counter_ == 0) {
-    // Destroy module.
-    VideoProcessingModule::Destroy(image_proc_module_);
-    image_proc_module_ = NULL;
-  }
-  return 0;
-}
-
-int32_t ViECapturer::EnableDeflickering(bool enable) {
-  CriticalSectionScoped cs(effects_and_stats_cs_.get());
-  if (enable) {
-    if (deflicker_frame_stats_) {
-      return -1;
-    }
-    if (IncImageProcRefCount() != 0) {
-      return -1;
-    }
-    deflicker_frame_stats_ = new VideoProcessingModule::FrameStats();
-  } else {
-    if (deflicker_frame_stats_ == NULL) {
-      return -1;
-    }
-    DecImageProcRefCount();
-    delete deflicker_frame_stats_;
-    deflicker_frame_stats_ = NULL;
-  }
-  return 0;
-}
-
-int32_t ViECapturer::EnableBrightnessAlarm(bool enable) {
-  CriticalSectionScoped cs(effects_and_stats_cs_.get());
-  if (enable) {
-    if (brightness_frame_stats_) {
-      return -1;
-    }
-    if (IncImageProcRefCount() != 0) {
-      return -1;
-    }
-    brightness_frame_stats_ = new VideoProcessingModule::FrameStats();
-  } else {
-    DecImageProcRefCount();
-    if (brightness_frame_stats_ == NULL) {
-      return -1;
-    }
-    delete brightness_frame_stats_;
-    brightness_frame_stats_ = NULL;
-  }
-  return 0;
-}
-
-bool ViECapturer::ViECaptureThreadFunction(void* obj) {
-  return static_cast<ViECapturer*>(obj)->ViECaptureProcess();
-}
-
-bool ViECapturer::ViECaptureProcess() {
-  int64_t capture_time = -1;
-  if (capture_event_.Wait(kThreadWaitTimeMs) == kEventSignaled) {
-    if (rtc::AtomicOps::Load(&stop_))
-      return false;
-
-    overuse_detector_->FrameProcessingStarted();
-    int64_t encode_start_time = -1;
-    I420VideoFrame deliver_frame;
-    {
-      CriticalSectionScoped cs(capture_cs_.get());
-      if (!captured_frame_.IsZeroSize()) {
-        deliver_frame = captured_frame_;
-        captured_frame_.Reset();
-      }
-    }
-    if (!deliver_frame.IsZeroSize()) {
-      capture_time = deliver_frame.render_time_ms();
-      encode_start_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
-      DeliverI420Frame(&deliver_frame);
-    }
-    if (current_brightness_level_ != reported_brightness_level_) {
-      CriticalSectionScoped cs(observer_cs_.get());
-      if (observer_) {
-        observer_->BrightnessAlarm(id_, current_brightness_level_);
-        reported_brightness_level_ = current_brightness_level_;
-      }
-    }
-    // Update the overuse detector with the duration.
-    if (encode_start_time != -1) {
-      overuse_detector_->FrameEncoded(
-          Clock::GetRealTimeClock()->TimeInMilliseconds() - encode_start_time);
-    }
-  }
-  // We're done!
-  if (capture_time != -1) {
-    overuse_detector_->FrameSent(capture_time);
-  }
-  return true;
-}
-
-void ViECapturer::DeliverI420Frame(I420VideoFrame* video_frame) {
-  if (video_frame->native_handle() != NULL) {
-    ViEFrameProviderBase::DeliverFrame(video_frame, std::vector<uint32_t>());
-    return;
-  }
-
-  // Apply image enhancement and effect filter.
-  {
-    CriticalSectionScoped cs(effects_and_stats_cs_.get());
-    if (deflicker_frame_stats_) {
-      if (image_proc_module_->GetFrameStats(deflicker_frame_stats_,
-                                            *video_frame) == 0) {
-        image_proc_module_->Deflickering(video_frame, deflicker_frame_stats_);
-      } else {
-        LOG_F(LS_ERROR) << "Could not get frame stats.";
-      }
-    }
-    if (brightness_frame_stats_) {
-      if (image_proc_module_->GetFrameStats(brightness_frame_stats_,
-                                            *video_frame) == 0) {
-        int32_t brightness = image_proc_module_->BrightnessDetection(
-            *video_frame, *brightness_frame_stats_);
-
-        switch (brightness) {
-          case VideoProcessingModule::kNoWarning:
-            current_brightness_level_ = Normal;
-            break;
-          case VideoProcessingModule::kDarkWarning:
-            current_brightness_level_ = Dark;
-            break;
-          case VideoProcessingModule::kBrightWarning:
-            current_brightness_level_ = Bright;
-            break;
-          default:
-            break;
-        }
-      }
-    }
-    if (effect_filter_) {
-      size_t length =
-          CalcBufferSize(kI420, video_frame->width(), video_frame->height());
-      rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
-      ExtractBuffer(*video_frame, length, video_buffer.get());
-      effect_filter_->Transform(length,
-                                video_buffer.get(),
-                                video_frame->ntp_time_ms(),
-                                video_frame->timestamp(),
-                                video_frame->width(),
-                                video_frame->height());
-    }
-  }
-  // Deliver the captured frame to all observers (channels, renderer or file).
-  ViEFrameProviderBase::DeliverFrame(video_frame, std::vector<uint32_t>());
-}
-
-bool ViECapturer::CaptureCapabilityFixed() {
-  return requested_capability_.width != 0 &&
-      requested_capability_.height != 0 &&
-      requested_capability_.maxFPS != 0;
-}
-
-int32_t ViECapturer::RegisterObserver(ViECaptureObserver* observer) {
-  {
-    CriticalSectionScoped cs(observer_cs_.get());
-    if (observer_) {
-      LOG_F(LS_ERROR) << "Observer already registered.";
-      return -1;
-    }
-    observer_ = observer;
-  }
-  capture_module_->RegisterCaptureCallback(*this);
-  capture_module_->EnableFrameRateCallback(true);
-  capture_module_->EnableNoPictureAlarm(true);
-  return 0;
-}
-
-int32_t ViECapturer::DeRegisterObserver() {
-  capture_module_->EnableFrameRateCallback(false);
-  capture_module_->EnableNoPictureAlarm(false);
-  capture_module_->DeRegisterCaptureCallback();
-
-  CriticalSectionScoped cs(observer_cs_.get());
-  observer_ = NULL;
-  return 0;
-}
-
-bool ViECapturer::IsObserverRegistered() {
-  CriticalSectionScoped cs(observer_cs_.get());
-  return observer_ != NULL;
-}
-
-void ViECapturer::OnCaptureFrameRate(const int32_t id,
-                                     const uint32_t frame_rate) {
-  CriticalSectionScoped cs(observer_cs_.get());
-  observer_->CapturedFrameRate(id_, static_cast<uint8_t>(frame_rate));
-}
-
-void ViECapturer::OnNoPictureAlarm(const int32_t id,
-                                   const VideoCaptureAlarm alarm) {
-  LOG(LS_WARNING) << "OnNoPictureAlarm " << id;
-
-  CriticalSectionScoped cs(observer_cs_.get());
-  CaptureAlarm vie_alarm = (alarm == Raised) ? AlarmRaised : AlarmCleared;
-  observer_->NoPictureAlarm(id, vie_alarm);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h b/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h
deleted file mode 100644
index ac97f56568..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
-
-#include <vector>
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-#include "webrtc/common.h"
-
-namespace webrtc {
-
-class Config;
-class CriticalSectionWrapper;
-class EventWrapper;
-class CpuOveruseObserver;
-class OveruseFrameDetector;
-class ProcessThread;
-class ViEEffectFilter;
-class ViEEncoder;
-struct ViEPicture;
-class RegistrableCpuOveruseMetricsObserver;
-
-class ViECapturer
-    : public ViEFrameProviderBase,
-      public ViEExternalCapture,
-      protected VideoCaptureDataCallback,
-      protected VideoCaptureFeedBack {
- public:
-  static ViECapturer* CreateViECapture(int capture_id,
-                                       int engine_id,
-                                       const Config& config,
-                                       VideoCaptureModule* capture_module,
-                                       ProcessThread& module_process_thread);
-
-  static ViECapturer* CreateViECapture(
-      int capture_id,
-      int engine_id,
-      const Config& config,
-      const char* device_unique_idUTF8,
-      uint32_t device_unique_idUTF8Length,
-      ProcessThread& module_process_thread);
-
-  ~ViECapturer();
-
-  // Implements ViEFrameProviderBase.
-  int FrameCallbackChanged();
-
-  // Implements ExternalCapture.
-  void IncomingFrame(const I420VideoFrame& frame) override;
-
-  // Start/Stop.
-  int32_t Start(
-      const CaptureCapability& capture_capability = CaptureCapability());
-  int32_t Stop();
-  bool Started();
-
-  // Overrides the capture delay.
-  int32_t SetCaptureDelay(int32_t delay_ms);
-
-  // Sets rotation of the incoming captured frame.
-  int32_t SetVideoRotation(const VideoRotation rotation);
-
-  // Effect filter.
-  int32_t RegisterEffectFilter(ViEEffectFilter* effect_filter);
-  int32_t EnableDeflickering(bool enable);
-  int32_t EnableBrightnessAlarm(bool enable);
-
-  // Statistics observer.
-  int32_t RegisterObserver(ViECaptureObserver* observer);
-  int32_t DeRegisterObserver();
-  bool IsObserverRegistered();
-
-  // Information.
-  const char* CurrentDeviceName() const;
-
-  void RegisterCpuOveruseObserver(CpuOveruseObserver* observer);
-  void SetCpuOveruseOptions(const CpuOveruseOptions& options);
-  void RegisterCpuOveruseMetricsObserver(CpuOveruseMetricsObserver* observer);
-  void GetCpuOveruseMetrics(CpuOveruseMetrics* metrics) const;
-
- protected:
-  ViECapturer(int capture_id,
-              int engine_id,
-              const Config& config,
-              ProcessThread& module_process_thread);
-
-  int32_t Init(VideoCaptureModule* capture_module);
-  int32_t Init(const char* device_unique_idUTF8,
-               uint32_t device_unique_idUTF8Length);
-
-  // Implements VideoCaptureDataCallback.
-  virtual void OnIncomingCapturedFrame(const int32_t id,
-                                       const I420VideoFrame& video_frame);
-  virtual void OnCaptureDelayChanged(const int32_t id,
-                                     const int32_t delay);
-
-  // Returns true if the capture capability has been set in |StartCapture|
-  // function and may not be changed.
-  bool CaptureCapabilityFixed();
-
-  // Help function used for keeping track of VideoImageProcesingModule.
-  // Creates the module if it is needed, returns 0 on success and guarantees
-  // that the image proc module exist.
-  int32_t IncImageProcRefCount();
-  int32_t DecImageProcRefCount();
-
-  // Implements VideoCaptureFeedBack
-  virtual void OnCaptureFrameRate(const int32_t id,
-                                  const uint32_t frame_rate);
-  virtual void OnNoPictureAlarm(const int32_t id,
-                                const VideoCaptureAlarm alarm);
-
-  // Thread functions for deliver captured frames to receivers.
-  static bool ViECaptureThreadFunction(void* obj);
-  bool ViECaptureProcess();
-
- private:
-  void DeliverI420Frame(I420VideoFrame* video_frame);
-
-  // Never take capture_cs_ before effects_and_stats_cs_!
-  rtc::scoped_ptr<CriticalSectionWrapper> capture_cs_;
-  rtc::scoped_ptr<CriticalSectionWrapper> effects_and_stats_cs_;
-  VideoCaptureModule* capture_module_;
-  bool use_external_capture_;
-  ProcessThread& module_process_thread_;
-  const int capture_id_;
-
-  // Frame used in IncomingFrameI420.
-  rtc::scoped_ptr<CriticalSectionWrapper> incoming_frame_cs_;
-  I420VideoFrame incoming_frame_;
-
-  // Capture thread.
-  rtc::scoped_ptr<ThreadWrapper> capture_thread_;
-  EventWrapper& capture_event_;
-  EventWrapper& deliver_event_;
-
-  volatile int stop_;
-
-  I420VideoFrame captured_frame_ GUARDED_BY(capture_cs_.get());
-  // Used to make sure incoming time stamp is increasing for every frame.
-  int64_t last_captured_timestamp_;
-  // Delta used for translating between NTP and internal timestamps.
-  const int64_t delta_ntp_internal_ms_;
-
-  // Image processing.
-  ViEEffectFilter* effect_filter_ GUARDED_BY(effects_and_stats_cs_.get());
-  VideoProcessingModule* image_proc_module_;
-  int image_proc_module_ref_counter_;
-  VideoProcessingModule::FrameStats* deflicker_frame_stats_
-      GUARDED_BY(effects_and_stats_cs_.get());
-  VideoProcessingModule::FrameStats* brightness_frame_stats_
-      GUARDED_BY(effects_and_stats_cs_.get());
-  Brightness current_brightness_level_;
-  Brightness reported_brightness_level_;
-
-  // Statistics observer.
-  rtc::scoped_ptr<CriticalSectionWrapper> observer_cs_;
-  ViECaptureObserver* observer_ GUARDED_BY(observer_cs_.get());
-
-  CaptureCapability requested_capability_;
-
-  // Must be declared before overuse_detector_ where it's registered.
-  const rtc::scoped_ptr<RegistrableCpuOveruseMetricsObserver>
-      cpu_overuse_metrics_observer_;
-  rtc::scoped_ptr<OveruseFrameDetector> overuse_detector_;
-  const Config & config_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc
deleted file mode 100644
index 18aa49665f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc
+++ /dev/null
@@ -1,300 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file includes unit tests for ViECapturer.
-
-#include "webrtc/video_engine/vie_capturer.h"
-
-#include <vector>
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common.h"
-#include "webrtc/common_video/interface/native_handle.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
-#include "webrtc/modules/video_capture/include/mock/mock_video_capture.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
-#include "webrtc/video_engine/mock/mock_vie_frame_provider_base.h"
-
-using ::testing::_;
-using ::testing::Invoke;
-using ::testing::NiceMock;
-using ::testing::Return;
-using ::testing::WithArg;
-
-// If an output frame does not arrive in 500ms, the test will fail.
-#define FRAME_TIMEOUT_MS 500
-
-namespace webrtc {
-
-bool EqualFrames(const I420VideoFrame& frame1,
-                 const I420VideoFrame& frame2);
-bool EqualTextureFrames(const I420VideoFrame& frame1,
-                        const I420VideoFrame& frame2);
-bool EqualBufferFrames(const I420VideoFrame& frame1,
-                       const I420VideoFrame& frame2);
-bool EqualFramesVector(const ScopedVector<I420VideoFrame>& frames1,
-                       const ScopedVector<I420VideoFrame>& frames2);
-I420VideoFrame* CreateI420VideoFrame(uint8_t length);
-
-class FakeNativeHandle : public NativeHandle {
- public:
-  FakeNativeHandle() {}
-  virtual ~FakeNativeHandle() {}
-  virtual void* GetHandle() { return NULL; }
-};
-
-class ViECapturerTest : public ::testing::Test {
- protected:
-  ViECapturerTest()
-      : mock_capture_module_(new NiceMock<MockVideoCaptureModule>()),
-        mock_process_thread_(new NiceMock<MockProcessThread>),
-        mock_frame_callback_(new NiceMock<MockViEFrameCallback>),
-        data_callback_(NULL),
-        output_frame_event_(EventWrapper::Create()) {
-  }
-
-  virtual void SetUp() {
-    EXPECT_CALL(*mock_capture_module_, RegisterCaptureDataCallback(_))
-        .WillRepeatedly(Invoke(this, &ViECapturerTest::SetCaptureDataCallback));
-    EXPECT_CALL(*mock_frame_callback_, DeliverFrame(_, _, _))
-        .WillRepeatedly(
-            WithArg<1>(Invoke(this, &ViECapturerTest::AddOutputFrame)));
-
-    Config config;
-    vie_capturer_.reset(
-        ViECapturer::CreateViECapture(
-            0, 0, config, mock_capture_module_.get(), *mock_process_thread_));
-    vie_capturer_->RegisterFrameCallback(0, mock_frame_callback_.get());
-  }
-
-  virtual void TearDown() {
-    vie_capturer_->DeregisterFrameCallback(mock_frame_callback_.get());
-    // ViECapturer accesses |mock_process_thread_| in destructor and should
-    // be deleted first.
-    vie_capturer_.reset();
-  }
-
-  void SetCaptureDataCallback(VideoCaptureDataCallback& data_callback) {
-    data_callback_ = &data_callback;
-  }
-
-  void AddInputFrame(I420VideoFrame* frame) {
-    data_callback_->OnIncomingCapturedFrame(0, *frame);
-  }
-
-  void AddOutputFrame(const I420VideoFrame* frame) {
-    if (frame->native_handle() == NULL)
-      output_frame_ybuffers_.push_back(frame->buffer(kYPlane));
-    output_frames_.push_back(new I420VideoFrame(*frame));
-    output_frame_event_->Set();
-  }
-
-  void WaitOutputFrame() {
-    EXPECT_EQ(kEventSignaled, output_frame_event_->Wait(FRAME_TIMEOUT_MS));
-  }
-
-  rtc::scoped_ptr<MockVideoCaptureModule> mock_capture_module_;
-  rtc::scoped_ptr<MockProcessThread> mock_process_thread_;
-  rtc::scoped_ptr<MockViEFrameCallback> mock_frame_callback_;
-
-  // Used to send input capture frames to ViECapturer.
-  VideoCaptureDataCallback* data_callback_;
-
-  rtc::scoped_ptr<ViECapturer> vie_capturer_;
-
-  // Input capture frames of ViECapturer.
-  ScopedVector<I420VideoFrame> input_frames_;
-
-  // Indicate an output frame has arrived.
-  rtc::scoped_ptr<EventWrapper> output_frame_event_;
-
-  // Output delivered frames of ViECaptuer.
-  ScopedVector<I420VideoFrame> output_frames_;
-
-  // The pointers of Y plane buffers of output frames. This is used to verify
-  // the frame are swapped and not copied.
-  std::vector<const uint8_t*> output_frame_ybuffers_;
-};
-
-TEST_F(ViECapturerTest, DoesNotRetainHandleNorCopyBuffer) {
-  // Indicate an output frame has arrived.
-  rtc::scoped_ptr<EventWrapper> frame_destroyed_event(EventWrapper::Create());
-  class TestBuffer : public webrtc::I420Buffer {
-   public:
-    TestBuffer(EventWrapper* event) : I420Buffer(5, 5), event_(event) {}
-
-   private:
-    friend class rtc::RefCountedObject<TestBuffer>;
-    ~TestBuffer() override { event_->Set(); }
-    EventWrapper* event_;
-  };
-
-  I420VideoFrame frame(
-      new rtc::RefCountedObject<TestBuffer>(frame_destroyed_event.get()), 1, 1,
-      kVideoRotation_0);
-
-  AddInputFrame(&frame);
-  WaitOutputFrame();
-
-  EXPECT_EQ(output_frames_[0]->video_frame_buffer().get(),
-            frame.video_frame_buffer().get());
-  output_frames_.clear();
-  frame.Reset();
-  EXPECT_EQ(kEventSignaled, frame_destroyed_event->Wait(FRAME_TIMEOUT_MS));
-}
-
-TEST_F(ViECapturerTest, TestNtpTimeStampSetIfRenderTimeSet) {
-  input_frames_.push_back(CreateI420VideoFrame(static_cast<uint8_t>(0)));
-  input_frames_[0]->set_render_time_ms(5);
-  input_frames_[0]->set_ntp_time_ms(0);
-
-  AddInputFrame(input_frames_[0]);
-  WaitOutputFrame();
-  EXPECT_GT(output_frames_[0]->ntp_time_ms(),
-            input_frames_[0]->render_time_ms());
-}
-
-TEST_F(ViECapturerTest, TestRtpTimeStampSet) {
-  input_frames_.push_back(CreateI420VideoFrame(static_cast<uint8_t>(0)));
-  input_frames_[0]->set_render_time_ms(0);
-  input_frames_[0]->set_ntp_time_ms(1);
-  input_frames_[0]->set_timestamp(0);
-
-  AddInputFrame(input_frames_[0]);
-  WaitOutputFrame();
-  EXPECT_EQ(output_frames_[0]->timestamp(),
-            input_frames_[0]->ntp_time_ms() * 90);
-}
-
-TEST_F(ViECapturerTest, TestTextureFrames) {
-  const int kNumFrame = 3;
-  for (int i = 0 ; i < kNumFrame; ++i) {
-    webrtc::RefCountImpl<FakeNativeHandle>* handle =
-              new webrtc::RefCountImpl<FakeNativeHandle>();
-    // Add one to |i| so that width/height > 0.
-    input_frames_.push_back(
-        new I420VideoFrame(handle, i + 1, i + 1, i + 1, i + 1));
-    AddInputFrame(input_frames_[i]);
-    WaitOutputFrame();
-  }
-
-  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
-}
-
-TEST_F(ViECapturerTest, TestI420Frames) {
-  const int kNumFrame = 4;
-  std::vector<const uint8_t*> ybuffer_pointers;
-  for (int i = 0; i < kNumFrame; ++i) {
-    input_frames_.push_back(CreateI420VideoFrame(static_cast<uint8_t>(i + 1)));
-    const I420VideoFrame* const_input_frame = input_frames_[i];
-    ybuffer_pointers.push_back(const_input_frame->buffer(kYPlane));
-    AddInputFrame(input_frames_[i]);
-    WaitOutputFrame();
-  }
-
-  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
-  // Make sure the buffer is not copied.
-  for (int i = 0; i < kNumFrame; ++i)
-    EXPECT_EQ(ybuffer_pointers[i], output_frame_ybuffers_[i]);
-}
-
-TEST_F(ViECapturerTest, TestI420FrameAfterTextureFrame) {
-  webrtc::RefCountImpl<FakeNativeHandle>* handle =
-      new webrtc::RefCountImpl<FakeNativeHandle>();
-  input_frames_.push_back(new I420VideoFrame(handle, 1, 1, 1, 1));
-  AddInputFrame(input_frames_[0]);
-  WaitOutputFrame();
-
-  input_frames_.push_back(CreateI420VideoFrame(2));
-  AddInputFrame(input_frames_[1]);
-  WaitOutputFrame();
-
-  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
-}
-
-TEST_F(ViECapturerTest, TestTextureFrameAfterI420Frame) {
-  input_frames_.push_back(CreateI420VideoFrame(1));
-  AddInputFrame(input_frames_[0]);
-  WaitOutputFrame();
-
-  webrtc::RefCountImpl<FakeNativeHandle>* handle =
-      new webrtc::RefCountImpl<FakeNativeHandle>();
-  input_frames_.push_back(new I420VideoFrame(handle, 1, 1, 2, 2));
-  AddInputFrame(input_frames_[1]);
-  WaitOutputFrame();
-
-  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
-}
-
-bool EqualFrames(const I420VideoFrame& frame1,
-                 const I420VideoFrame& frame2) {
-  if (frame1.native_handle() != NULL || frame2.native_handle() != NULL)
-    return EqualTextureFrames(frame1, frame2);
-  return EqualBufferFrames(frame1, frame2);
-}
-
-bool EqualTextureFrames(const I420VideoFrame& frame1,
-                        const I420VideoFrame& frame2) {
-  return ((frame1.native_handle() == frame2.native_handle()) &&
-          (frame1.width() == frame2.width()) &&
-          (frame1.height() == frame2.height()) &&
-          (frame1.render_time_ms() == frame2.render_time_ms()));
-}
-
-bool EqualBufferFrames(const I420VideoFrame& frame1,
-                       const I420VideoFrame& frame2) {
-  return ((frame1.width() == frame2.width()) &&
-          (frame1.height() == frame2.height()) &&
-          (frame1.stride(kYPlane) == frame2.stride(kYPlane)) &&
-          (frame1.stride(kUPlane) == frame2.stride(kUPlane)) &&
-          (frame1.stride(kVPlane) == frame2.stride(kVPlane)) &&
-          (frame1.render_time_ms() == frame2.render_time_ms()) &&
-          (frame1.allocated_size(kYPlane) == frame2.allocated_size(kYPlane)) &&
-          (frame1.allocated_size(kUPlane) == frame2.allocated_size(kUPlane)) &&
-          (frame1.allocated_size(kVPlane) == frame2.allocated_size(kVPlane)) &&
-          (memcmp(frame1.buffer(kYPlane), frame2.buffer(kYPlane),
-                  frame1.allocated_size(kYPlane)) == 0) &&
-          (memcmp(frame1.buffer(kUPlane), frame2.buffer(kUPlane),
-                  frame1.allocated_size(kUPlane)) == 0) &&
-          (memcmp(frame1.buffer(kVPlane), frame2.buffer(kVPlane),
-                  frame1.allocated_size(kVPlane)) == 0));
-}
-
-bool EqualFramesVector(const ScopedVector<I420VideoFrame>& frames1,
-                       const ScopedVector<I420VideoFrame>& frames2) {
-  if (frames1.size() != frames2.size())
-    return false;
-  for (size_t i = 0; i < frames1.size(); ++i) {
-    if (!EqualFrames(*frames1[i], *frames2[i]))
-      return false;
-  }
-  return true;
-}
-
-I420VideoFrame* CreateI420VideoFrame(uint8_t data) {
-  I420VideoFrame* frame = new I420VideoFrame();
-  const int width = 36;
-  const int height = 24;
-  const int kSizeY = width * height * 2;
-  uint8_t buffer[kSizeY];
-  memset(buffer, data, kSizeY);
-  frame->CreateFrame(
-      buffer, buffer, buffer, width, height, width,
-      width / 2, width / 2);
-  frame->set_render_time_ms(data);
-  return frame;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc b/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc
deleted file mode 100644
index 7fd9f78ad4..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc
+++ /dev/null
@@ -1,2173 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_channel.h"
-
-#include <algorithm>
-#include <vector>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/common.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/experiments.h"
-#include "webrtc/frame_callback.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video_engine/call_stats.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/payload_router.h"
-#include "webrtc/video_engine/report_block_stats.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h" // for webrtc::SenderInfo
-
-namespace webrtc {
-
-const int kMaxDecodeWaitTimeMs = 50;
-const int kInvalidRtpExtensionId = 0;
-static const int kMaxTargetDelayMs = 10000;
-static const float kMaxIncompleteTimeMultiplier = 3.5f;
-
-// Helper class receiving statistics callbacks.
-class ChannelStatsObserver : public CallStatsObserver {
- public:
-  explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
-  virtual ~ChannelStatsObserver() {}
-
-  // Implements StatsObserver.
-  virtual void OnRttUpdate(int64_t rtt) {
-    owner_->OnRttUpdate(rtt);
-  }
-
- private:
-  ViEChannel* const owner_;
-};
-
-class ViEChannelProtectionCallback : public VCMProtectionCallback {
- public:
-  ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {}
-  ~ViEChannelProtectionCallback() {}
-
-
-  int ProtectionRequest(
-      const FecProtectionParams* delta_fec_params,
-      const FecProtectionParams* key_fec_params,
-      uint32_t* sent_video_rate_bps,
-      uint32_t* sent_nack_rate_bps,
-      uint32_t* sent_fec_rate_bps) override {
-    return owner_->ProtectionRequest(delta_fec_params, key_fec_params,
-                                     sent_video_rate_bps, sent_nack_rate_bps,
-                                     sent_fec_rate_bps);
-  }
- private:
-  ViEChannel* owner_;
-};
-
-ViEChannel::ViEChannel(int32_t channel_id,
-                       int32_t engine_id,
-                       uint32_t number_of_cores,
-                       const Config& config,
-                       ProcessThread& module_process_thread,
-                       RtcpIntraFrameObserver* intra_frame_observer,
-                       RtcpBandwidthObserver* bandwidth_observer,
-                       RemoteBitrateEstimator* remote_bitrate_estimator,
-                       RtcpRttStats* rtt_stats,
-                       PacedSender* paced_sender,
-                       PacketRouter* packet_router,
-                       bool sender,
-                       bool disable_default_encoder)
-    : ViEFrameProviderBase(channel_id, engine_id),
-      channel_id_(channel_id),
-      engine_id_(engine_id),
-      number_of_cores_(number_of_cores),
-      num_socket_threads_(kViESocketThreads),
-      callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      rtp_rtcp_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      send_payload_router_(new PayloadRouter()),
-      vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
-      vcm_(VideoCodingModule::Create(nullptr)),
-      vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this),
-      vie_sender_(channel_id),
-      vie_sync_(vcm_, this),
-      stats_observer_(new ChannelStatsObserver(this)),
-      vcm_receive_stats_callback_(NULL),
-      module_process_thread_(module_process_thread),
-      codec_observer_(NULL),
-      do_key_frame_callbackRequest_(false),
-      rtp_observer_(NULL),
-      intra_frame_observer_(intra_frame_observer),
-      rtt_stats_(rtt_stats),
-      paced_sender_(paced_sender),
-      packet_router_(packet_router),
-      bandwidth_observer_(bandwidth_observer),
-      send_timestamp_extension_id_(kInvalidRtpExtensionId),
-      absolute_send_time_extension_id_(kInvalidRtpExtensionId),
-      video_rotation_extension_id_(kInvalidRtpExtensionId),
-      rid_extension_id_(kInvalidRtpExtensionId),
-      external_transport_(NULL),
-      decoder_reset_(true),
-      wait_for_key_frame_(false),
-      effect_filter_(NULL),
-      color_enhancement_(false),
-      mtu_(0),
-      sender_(sender),
-      disable_default_encoder_(disable_default_encoder),
-      nack_history_size_sender_(kSendSidePacketHistorySize),
-      max_nack_reordering_threshold_(kMaxPacketAgeToNack),
-      pre_render_callback_(NULL),
-      report_block_stats_sender_(new ReportBlockStats()),
-      report_block_stats_receiver_(new ReportBlockStats()) {
-  RtpRtcp::Configuration configuration = CreateRtpRtcpConfiguration();
-  configuration.remote_bitrate_estimator = remote_bitrate_estimator;
-  configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
-  rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
-  vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
-  vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
-}
-
-int32_t ViEChannel::Init() {
-  module_process_thread_.RegisterModule(vie_receiver_.GetReceiveStatistics());
-
-  // RTP/RTCP initialization.
-  rtp_rtcp_->SetSendingMediaStatus(false);
-  module_process_thread_.RegisterModule(rtp_rtcp_.get());
-
-  rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqFirRtp);
-  rtp_rtcp_->SetRTCPStatus(kRtcpCompound);
-  if (paced_sender_) {
-    rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
-  }
-  if (sender_) {
-    packet_router_->AddRtpModule(rtp_rtcp_.get());
-    std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_.get());
-    send_payload_router_->SetSendingRtpModules(send_rtp_modules);
-    DCHECK(!send_payload_router_->active());
-  }
-  if (vcm_->InitializeReceiver() != 0) {
-    return -1;
-  }
-  if (vcm_->SetVideoProtection(kProtectionKeyOnLoss, true)) {
-    return -1;
-  }
-  if (vcm_->RegisterReceiveCallback(this) != 0) {
-    return -1;
-  }
-  vcm_->RegisterFrameTypeCallback(this);
-  vcm_->RegisterReceiveStateCallback(this);
-  vcm_->RegisterReceiveStatisticsCallback(this);
-  vcm_->RegisterDecoderTimingCallback(this);
-  vcm_->SetRenderDelay(kViEDefaultRenderDelayMs);
-
-  module_process_thread_.RegisterModule(vcm_);
-  module_process_thread_.RegisterModule(&vie_sync_);
-
-#ifdef VIDEOCODEC_VP8
-  if (!disable_default_encoder_) {
-    VideoCodec video_codec;
-    if (vcm_->Codec(kVideoCodecVP8, &video_codec) == VCM_OK) {
-      rtp_rtcp_->RegisterSendPayload(video_codec);
-      // TODO(holmer): Can we call SetReceiveCodec() here instead?
-      if (!vie_receiver_.RegisterPayload(video_codec)) {
-        return -1;
-      }
-      vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_);
-      vcm_->RegisterSendCodec(&video_codec, number_of_cores_,
-                              rtp_rtcp_->MaxDataPayloadLength());
-    } else {
-      assert(false);
-    }
-  }
-#endif
-
-  return 0;
-}
-
-ViEChannel::~ViEChannel() {
-  UpdateHistograms();
-  // Make sure we don't get more callbacks from the RTP module.
-  module_process_thread_.DeRegisterModule(vie_receiver_.GetReceiveStatistics());
-  module_process_thread_.DeRegisterModule(rtp_rtcp_.get());
-  module_process_thread_.DeRegisterModule(vcm_);
-  module_process_thread_.DeRegisterModule(&vie_sync_);
-  send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
-  packet_router_->RemoveRtpModule(rtp_rtcp_.get());
-  while (simulcast_rtp_rtcp_.size() > 0) {
-    std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-    RtpRtcp* rtp_rtcp = *it;
-    packet_router_->RemoveRtpModule(rtp_rtcp);
-    module_process_thread_.DeRegisterModule(rtp_rtcp);
-    delete rtp_rtcp;
-    simulcast_rtp_rtcp_.erase(it);
-  }
-  while (removed_rtp_rtcp_.size() > 0) {
-    std::list<RtpRtcp*>::iterator it = removed_rtp_rtcp_.begin();
-    delete *it;
-    removed_rtp_rtcp_.erase(it);
-  }
-  if (decode_thread_) {
-    StopDecodeThread();
-  }
-  // Release modules.
-  VideoCodingModule::Destroy(vcm_);
-}
-
-void ViEChannel::UpdateHistograms() {
-  int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
-
-  if (sender_) {
-    RtcpPacketTypeCounter rtcp_counter;
-    GetSendRtcpPacketTypeCounter(&rtcp_counter);
-    int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
-    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
-          rtcp_counter.nack_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
-          rtcp_counter.fir_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
-          rtcp_counter.pli_packets * 60 / elapsed_sec);
-      if (rtcp_counter.nack_requests > 0) {
-        RTC_HISTOGRAM_PERCENTAGE(
-            "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
-                rtcp_counter.UniqueNackRequestsInPercent());
-      }
-      int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
-      if (fraction_lost != -1) {
-        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
-            fraction_lost);
-      }
-    }
-  } else if (vie_receiver_.GetRemoteSsrc() > 0) {
-    // Get receive stats if we are receiving packets, i.e. there is a remote
-    // ssrc.
-    RtcpPacketTypeCounter rtcp_counter;
-    GetReceiveRtcpPacketTypeCounter(&rtcp_counter);
-    int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
-    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
-          rtcp_counter.nack_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
-          rtcp_counter.fir_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
-          rtcp_counter.pli_packets * 60 / elapsed_sec);
-      if (rtcp_counter.nack_requests > 0) {
-        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
-            rtcp_counter.UniqueNackRequestsInPercent());
-      }
-      int fraction_lost = report_block_stats_receiver_->FractionLostInPercent();
-      if (fraction_lost != -1) {
-        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
-            fraction_lost);
-      }
-    }
-
-    StreamDataCounters rtp;
-    StreamDataCounters rtx;
-    GetReceiveStreamDataCounters(&rtp, &rtx);
-    StreamDataCounters rtp_rtx = rtp;
-    rtp_rtx.Add(rtx);
-    elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
-    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.BitrateReceivedInKbps",
-          rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
-          rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
-      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateReceivedInKbps",
-          rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000);
-      RTC_HISTOGRAM_COUNTS_10000(
-          "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
-              rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-      uint32_t ssrc = 0;
-      if (vie_receiver_.GetRtxSsrc(&ssrc)) {
-        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
-            rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-      }
-      if (vie_receiver_.IsFecEnabled()) {
-        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps",
-            rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000);
-      }
-    }
-  }
-}
-
-void ViEChannel::UpdateHistogramsAtStopSend() {
-  StreamDataCounters rtp;
-  StreamDataCounters rtx;
-  GetSendStreamDataCounters(&rtp, &rtx);
-  StreamDataCounters rtp_rtx = rtp;
-  rtp_rtx.Add(rtx);
-
-  int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
-      Clock::GetRealTimeClock()->TimeInMilliseconds()) / 1000;
-  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
-    return;
-  }
-  RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.BitrateSentInKbps",
-      rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateSentInKbps",
-      rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
-  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateSentInKbps",
-      rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000);
-  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RetransmittedBitrateSentInKbps",
-      rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-  if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateSentInKbps",
-        rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
-  }
-  bool fec_enabled = false;
-  uint8_t pltype_red;
-  uint8_t pltype_fec;
-  rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
-  if (fec_enabled) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
-        rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000);
-  }
-}
-
-int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
-                                 bool new_stream) {
-  if (!sender_) {
-    return 0;
-  }
-  if (video_codec.codecType == kVideoCodecRED ||
-      video_codec.codecType == kVideoCodecULPFEC) {
-    LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType;
-    return -1;
-  }
-  if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) {
-    LOG_F(LS_ERROR) << "Incorrect config "
-                    << video_codec.numberOfSimulcastStreams;
-    return -1;
-  }
-  // Update the RTP module with the settings.
-  // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
-  // set explicitly.
-  bool restart_rtp = false;
-  bool router_was_active = send_payload_router_->active();
-  send_payload_router_->set_active(false);
-  send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
-  packet_router_->RemoveRtpModule(rtp_rtcp_.get());
-  for (RtpRtcp* module : simulcast_rtp_rtcp_)
-    packet_router_->RemoveRtpModule(module);
-  // Set the RtpSenderId
-  rid_extension_id_ = video_codec.ridId;
-
-  if (rtp_rtcp_->Sending()) {
-    restart_rtp = true;
-    rtp_rtcp_->SetSendingStatus(false);
-    int i = 0;
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); ++it, ++i) {
-      (*it)->SetSendingStatus(false);
-      (*it)->SetSendingMediaStatus(false);
-      if (video_codec.simulcastStream[i].rid[0] != 0) {
-        (*it)->RegisterSendRtpHeaderExtension(
-          kRtpExtensionRtpStreamId, video_codec.ridId);
-        (*it)->SetRID(video_codec.simulcastStream[i].rid);
-      } else {
-        (*it)->DeregisterSendRtpHeaderExtension(
-          kRtpExtensionRtpStreamId);
-      }
-    }
-  }
-
-  bool fec_enabled = false;
-  uint8_t payload_type_red;
-  uint8_t payload_type_fec;
-  rtp_rtcp_->GenericFECStatus(fec_enabled, payload_type_red, payload_type_fec);
-
-  std::vector<RtpRtcp*> registered_modules;
-  std::vector<RtpRtcp*> deregistered_modules;
-  {
-    CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-
-    if (video_codec.numberOfSimulcastStreams > 0) {
-
-      // Set RTP Stream ID of primary stream
-      if (rid_extension_id_ != kInvalidRtpExtensionId) {
-        // Deregister in case the extension was previously enabled.
-        rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-            kRtpExtensionRtpStreamId);
-        if (rtp_rtcp_->RegisterSendRtpHeaderExtension(
-                kRtpExtensionRtpStreamId,
-                rid_extension_id_) != 0) {
-          LOG(LS_WARNING) << "Register RID extension failed";
-        } else {
-          rtp_rtcp_->SetRID(video_codec.simulcastStream[0].rid);
-        }
-      } else {
-        rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-            kRtpExtensionRtpStreamId);
-      }
-
-      // Set correct bitrate to base layer.
-      // Create our simulcast RTP modules.
-      int num_modules_to_add =
-          video_codec.numberOfSimulcastStreams - simulcast_rtp_rtcp_.size() - 1;
-      if (num_modules_to_add < 0) {
-        num_modules_to_add = 0;
-      }
-
-      // Add back removed rtp modules. Order is important (allocate from front
-      // of removed modules) to preserve RTP settings such as SSRCs for
-      // simulcast streams.
-      std::list<RtpRtcp*> new_rtp_modules;
-      for (; removed_rtp_rtcp_.size() > 0 && num_modules_to_add > 0;
-           --num_modules_to_add) {
-        new_rtp_modules.push_back(removed_rtp_rtcp_.front());
-        removed_rtp_rtcp_.pop_front();
-      }
-
-      for (int i = 0; i < num_modules_to_add; ++i)
-        new_rtp_modules.push_back(CreateRtpRtcpModule());
-
-      // Initialize newly added modules.
-      for (std::list<RtpRtcp*>::iterator it = new_rtp_modules.begin();
-           it != new_rtp_modules.end(); ++it) {
-        RtpRtcp* rtp_rtcp = *it;
-
-        rtp_rtcp->SetRTCPStatus(rtp_rtcp_->RTCP());
-
-        if (rtp_rtcp_->StorePackets()) {
-          rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
-        } else if (paced_sender_) {
-          rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
-        }
-
-        if (fec_enabled) {
-          rtp_rtcp->SetGenericFECStatus(fec_enabled, payload_type_red,
-                                        payload_type_fec);
-        }
-        rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending());
-        rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia());
-        rtp_rtcp->SetRtxSendStatus(rtp_rtcp_->RtxSendStatus());
-        simulcast_rtp_rtcp_.push_back(rtp_rtcp);
-
-        // Silently ignore error.
-        registered_modules.push_back(rtp_rtcp);
-      }
-
-      // Remove last in list if we have too many.
-      for (int j = simulcast_rtp_rtcp_.size();
-           j > (video_codec.numberOfSimulcastStreams - 1); j--) {
-        RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
-        deregistered_modules.push_back(rtp_rtcp);
-        rtp_rtcp->SetSendingStatus(false);
-        rtp_rtcp->SetSendingMediaStatus(false);
-        rtp_rtcp->RegisterRtcpStatisticsCallback(NULL);
-        rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
-        simulcast_rtp_rtcp_.pop_back();
-        removed_rtp_rtcp_.push_front(rtp_rtcp);
-      }
-      uint8_t idx = 0;
-      // Configure all simulcast modules.
-      for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-           it != simulcast_rtp_rtcp_.end(); it++) {
-        idx++;
-        RtpRtcp* rtp_rtcp = *it;
-        rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
-        if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
-          return -1;
-        }
-        if (mtu_ != 0) {
-          rtp_rtcp->SetMaxTransferUnit(mtu_);
-        }
-        if (restart_rtp) {
-          rtp_rtcp->SetSendingStatus(true);
-          rtp_rtcp->SetSendingMediaStatus(true);
-        }
-        if (send_timestamp_extension_id_ != kInvalidRtpExtensionId) {
-          // Deregister in case the extension was previously enabled.
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionTransmissionTimeOffset);
-          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
-                  kRtpExtensionTransmissionTimeOffset,
-                  send_timestamp_extension_id_) != 0) {
-            LOG(LS_WARNING) << "Register Transmission Time Offset failed";
-          }
-        } else {
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionTransmissionTimeOffset);
-        }
-        if (absolute_send_time_extension_id_ != kInvalidRtpExtensionId) {
-          // Deregister in case the extension was previously enabled.
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionAbsoluteSendTime);
-          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
-                  kRtpExtensionAbsoluteSendTime,
-                  absolute_send_time_extension_id_) != 0) {
-            LOG(LS_WARNING) << "Register Absolute Send Time failed";
-          }
-        } else {
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionAbsoluteSendTime);
-        }
-        if (video_rotation_extension_id_ != kInvalidRtpExtensionId) {
-          // Deregister in case the extension was previously enabled.
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionVideoRotation);
-          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
-                  kRtpExtensionVideoRotation, video_rotation_extension_id_) !=
-              0) {
-            LOG(LS_WARNING) << "Register VideoRotation extension failed";
-          }
-        } else {
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionVideoRotation);
-        }
-        if (rid_extension_id_ != kInvalidRtpExtensionId) {
-          // Deregister in case the extension was previously enabled.
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionRtpStreamId);
-          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
-                  kRtpExtensionRtpStreamId,
-                  rid_extension_id_) != 0) {
-            LOG(LS_WARNING) << "Register RID extension failed";
-          } else {
-            (*it)->SetRID(video_codec.simulcastStream[idx].rid);
-          }
-        } else {
-          rtp_rtcp->DeregisterSendRtpHeaderExtension(
-              kRtpExtensionRtpStreamId);
-        }
-        rtp_rtcp->RegisterRtcpStatisticsCallback(
-            rtp_rtcp_->GetRtcpStatisticsCallback());
-        rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
-            rtp_rtcp_->GetSendChannelRtpStatisticsCallback());
-      }
-      // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
-      // modules can be deleted after this step.
-      vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
-    } else {
-      while (!simulcast_rtp_rtcp_.empty()) {
-        RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
-        deregistered_modules.push_back(rtp_rtcp);
-        rtp_rtcp->SetSendingStatus(false);
-        rtp_rtcp->SetSendingMediaStatus(false);
-        rtp_rtcp->RegisterRtcpStatisticsCallback(NULL);
-        rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
-        simulcast_rtp_rtcp_.pop_back();
-        removed_rtp_rtcp_.push_front(rtp_rtcp);
-      }
-      // Clear any previous modules.
-      vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
-    }
-
-    // Don't log this error, no way to check in advance if this pl_type is
-    // registered or not...
-    rtp_rtcp_->DeRegisterSendPayload(video_codec.plType);
-    if (rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
-      return -1;
-    }
-    if (restart_rtp) {
-      rtp_rtcp_->SetSendingStatus(true);
-      for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-           it != simulcast_rtp_rtcp_.end(); ++it) {
-        (*it)->SetSendingStatus(true);
-        (*it)->SetSendingMediaStatus(true);
-      }
-    }
-    // Update the packet and payload routers with the sending RTP RTCP modules.
-    packet_router_->AddRtpModule(rtp_rtcp_.get());
-    for (RtpRtcp* module : simulcast_rtp_rtcp_)
-      packet_router_->AddRtpModule(module);
-
-    std::list<RtpRtcp*> active_send_modules;
-    active_send_modules.push_back(rtp_rtcp_.get());
-    for (std::list<RtpRtcp*>::const_iterator cit = simulcast_rtp_rtcp_.begin();
-         cit != simulcast_rtp_rtcp_.end(); ++cit) {
-      active_send_modules.push_back(*cit);
-    }
-    send_payload_router_->SetSendingRtpModules(active_send_modules);
-    if (router_was_active)
-      send_payload_router_->set_active(true);
-  }
-  for (RtpRtcp* rtp_rtcp : registered_modules)
-    module_process_thread_.RegisterModule(rtp_rtcp);
-  for (RtpRtcp* rtp_rtcp : deregistered_modules)
-    module_process_thread_.DeRegisterModule(rtp_rtcp);
-  return 0;
-}
-
-int32_t ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) {
-  if (!vie_receiver_.SetReceiveCodec(video_codec)) {
-    return -1;
-  }
-
-  if (video_codec.codecType != kVideoCodecRED &&
-      video_codec.codecType != kVideoCodecULPFEC) {
-    // Register codec type with VCM, but do not register RED or ULPFEC.
-    if (vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_,
-                                  wait_for_key_frame_) != VCM_OK) {
-      return -1;
-    }
-  }
-  return 0;
-}
-
-int32_t ViEChannel::GetReceiveCodec(VideoCodec* video_codec) {
-  if (vcm_->ReceiveCodec(video_codec) != 0) {
-    return -1;
-  }
-  return 0;
-}
-
-int32_t ViEChannel::RegisterCodecObserver(ViEDecoderObserver* observer) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (observer) {
-    if (codec_observer_) {
-      LOG_F(LS_ERROR) << "Observer already registered.";
-      return -1;
-    }
-    codec_observer_ = observer;
-  } else {
-    codec_observer_ = NULL;
-  }
-  return 0;
-}
-
-int32_t ViEChannel::RegisterExternalDecoder(const uint8_t pl_type,
-                                            VideoDecoder* decoder,
-                                            bool buffered_rendering,
-                                            int32_t render_delay) {
-  int32_t result;
-  result = vcm_->RegisterExternalDecoder(decoder, pl_type, buffered_rendering);
-  if (result != VCM_OK) {
-    return result;
-  }
-  return vcm_->SetRenderDelay(render_delay);
-}
-
-int32_t ViEChannel::DeRegisterExternalDecoder(const uint8_t pl_type) {
-  VideoCodec current_receive_codec;
-  int32_t result = 0;
-  result = vcm_->ReceiveCodec(&current_receive_codec);
-  if (vcm_->RegisterExternalDecoder(NULL, pl_type, false) != VCM_OK) {
-    return -1;
-  }
-
-  if (result == 0 && current_receive_codec.plType == pl_type) {
-    result = vcm_->RegisterReceiveCodec(
-        &current_receive_codec, number_of_cores_, wait_for_key_frame_);
-  }
-  return result;
-}
-
-int32_t ViEChannel::ReceiveCodecStatistics(uint32_t* num_key_frames,
-                                           uint32_t* num_delta_frames) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  *num_key_frames = receive_frame_counts_.key_frames;
-  *num_delta_frames = receive_frame_counts_.delta_frames;
-  return 0;
-}
-
-uint32_t ViEChannel::DiscardedPackets() const {
-  return vcm_->DiscardedPackets();
-}
-
-int ViEChannel::ReceiveDelay() const {
-  return vcm_->Delay();
-}
-
-int32_t ViEChannel::WaitForKeyFrame(bool wait) {
-  wait_for_key_frame_ = wait;
-  return 0;
-}
-
-int32_t ViEChannel::SetSignalPacketLossStatus(bool enable,
-                                              bool only_key_frames) {
-  if (enable) {
-    if (only_key_frames) {
-      vcm_->SetVideoProtection(kProtectionKeyOnLoss, false);
-      if (vcm_->SetVideoProtection(kProtectionKeyOnKeyLoss, true) != VCM_OK) {
-        return -1;
-      }
-    } else {
-      vcm_->SetVideoProtection(kProtectionKeyOnKeyLoss, false);
-      if (vcm_->SetVideoProtection(kProtectionKeyOnLoss, true) != VCM_OK) {
-        return -1;
-      }
-    }
-  } else {
-    vcm_->SetVideoProtection(kProtectionKeyOnLoss, false);
-    vcm_->SetVideoProtection(kProtectionKeyOnKeyLoss, false);
-  }
-  return 0;
-}
-
-void ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->SetRTCPStatus(rtcp_mode);
-  }
-  rtp_rtcp_->SetRTCPStatus(rtcp_mode);
-}
-
-RTCPMethod ViEChannel::GetRTCPMode() const {
-  return rtp_rtcp_->RTCP();
-}
-
-int32_t ViEChannel::SetNACKStatus(const bool enable) {
-  // Update the decoding VCM.
-  if (vcm_->SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
-    return -1;
-  }
-  if (enable) {
-    // Disable possible FEC.
-    SetFECStatus(false, 0, 0);
-  }
-  // Update the decoding VCM.
-  if (vcm_->SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
-    return -1;
-  }
-  return ProcessNACKRequest(enable);
-}
-
-int32_t ViEChannel::ProcessNACKRequest(const bool enable) {
-  if (enable) {
-    // Turn on NACK.
-    if (rtp_rtcp_->RTCP() == kRtcpOff) {
-      return -1;
-    }
-    vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_);
-    rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
-    vcm_->RegisterPacketRequestCallback(this);
-
-    CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end();
-         it++) {
-      RtpRtcp* rtp_rtcp = *it;
-      rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
-    }
-    // Don't introduce errors when NACK is enabled.
-    vcm_->SetDecodeErrorMode(kNoErrors);
-  } else {
-    CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end();
-         it++) {
-      RtpRtcp* rtp_rtcp = *it;
-      if (paced_sender_ == NULL) {
-        rtp_rtcp->SetStorePacketsStatus(false, 0);
-      }
-    }
-    vcm_->RegisterPacketRequestCallback(NULL);
-    if (paced_sender_ == NULL) {
-      rtp_rtcp_->SetStorePacketsStatus(false, 0);
-    }
-    vie_receiver_.SetNackStatus(false, max_nack_reordering_threshold_);
-    // When NACK is off, allow decoding with errors. Otherwise, the video
-    // will freeze, and will only recover with a complete key frame.
-    vcm_->SetDecodeErrorMode(kWithErrors);
-  }
-  return 0;
-}
-
-int32_t ViEChannel::SetFECStatus(const bool enable,
-                                       const unsigned char payload_typeRED,
-                                       const unsigned char payload_typeFEC) {
-  // Disable possible NACK.
-  if (enable) {
-    SetNACKStatus(false);
-  }
-
-  return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
-}
-
-bool ViEChannel::IsSendingFecEnabled() {
-  bool fec_enabled = false;
-  uint8_t pltype_red = 0;
-  uint8_t pltype_fec = 0;
-  rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
-  if (fec_enabled)
-    return true;
-
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (auto* module : simulcast_rtp_rtcp_) {
-    module->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
-    if (fec_enabled)
-      return true;
-  }
-  return false;
-}
-
-int32_t ViEChannel::ProcessFECRequest(
-    const bool enable,
-    const unsigned char payload_typeRED,
-    const unsigned char payload_typeFEC) {
-  if (rtp_rtcp_->SetGenericFECStatus(enable, payload_typeRED,
-                                    payload_typeFEC) != 0) {
-    return -1;
-  }
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->SetGenericFECStatus(enable, payload_typeRED, payload_typeFEC);
-  }
-  return 0;
-}
-
-int32_t ViEChannel::SetHybridNACKFECStatus(
-    const bool enable,
-    const unsigned char payload_typeRED,
-    const unsigned char payload_typeFEC) {
-  if (vcm_->SetVideoProtection(kProtectionNackFEC, enable) != VCM_OK) {
-    return -1;
-  }
-
-  int32_t ret_val = 0;
-  ret_val = ProcessNACKRequest(enable);
-  if (ret_val < 0) {
-    return ret_val;
-  }
-  return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
-}
-
-int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
-  if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
-    LOG(LS_ERROR) << "Invalid send buffer value.";
-    return -1;
-  }
-  if (target_delay_ms == 0) {
-    // Real-time mode.
-    nack_history_size_sender_ = kSendSidePacketHistorySize;
-  } else {
-    nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms);
-    // Don't allow a number lower than the default value.
-    if (nack_history_size_sender_ < kSendSidePacketHistorySize) {
-      nack_history_size_sender_ = kSendSidePacketHistorySize;
-    }
-  }
-  rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
-  return 0;
-}
-
-int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) {
-  if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
-    LOG(LS_ERROR) << "Invalid receive buffer delay value.";
-    return -1;
-  }
-  int max_nack_list_size;
-  int max_incomplete_time_ms;
-  if (target_delay_ms == 0) {
-    // Real-time mode - restore default settings.
-    max_nack_reordering_threshold_ = kMaxPacketAgeToNack;
-    max_nack_list_size = kMaxNackListSize;
-    max_incomplete_time_ms = 0;
-  } else {
-    max_nack_list_size =  3 * GetRequiredNackListSize(target_delay_ms) / 4;
-    max_nack_reordering_threshold_ = max_nack_list_size;
-    // Calculate the max incomplete time and round to int.
-    max_incomplete_time_ms = static_cast<int>(kMaxIncompleteTimeMultiplier *
-        target_delay_ms + 0.5f);
-  }
-  vcm_->SetNackSettings(max_nack_list_size, max_nack_reordering_threshold_,
-                       max_incomplete_time_ms);
-  vcm_->SetMinReceiverDelay(target_delay_ms);
-  if (vie_sync_.SetTargetBufferingDelay(target_delay_ms) < 0)
-    return -1;
-  return 0;
-}
-
-int ViEChannel::GetRequiredNackListSize(int target_delay_ms) {
-  // The max size of the nack list should be large enough to accommodate the
-  // the number of packets (frames) resulting from the increased delay.
-  // Roughly estimating for ~40 packets per frame @ 30fps.
-  return target_delay_ms * 40 * 30 / 1000;
-}
-
-int32_t ViEChannel::SetKeyFrameRequestMethod(
-    const KeyFrameRequestMethod method) {
-  return rtp_rtcp_->SetKeyFrameRequestMethod(method);
-}
-
-void ViEChannel::EnableRemb(bool enable) {
-  rtp_rtcp_->SetREMBStatus(enable);
-}
-
-int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  int error = 0;
-  if (enable) {
-    // Enable the extension, but disable possible old id to avoid errors.
-    send_timestamp_extension_id_ = id;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionTransmissionTimeOffset);
-    error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
-        kRtpExtensionTransmissionTimeOffset, id);
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(
-          kRtpExtensionTransmissionTimeOffset);
-      error |= (*it)->RegisterSendRtpHeaderExtension(
-          kRtpExtensionTransmissionTimeOffset, id);
-    }
-  } else {
-    // Disable the extension.
-    send_timestamp_extension_id_ = kInvalidRtpExtensionId;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionTransmissionTimeOffset);
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(
-          kRtpExtensionTransmissionTimeOffset);
-    }
-  }
-  return error;
-}
-
-int ViEChannel::SetReceiveTimestampOffsetStatus(bool enable, int id) {
-  return vie_receiver_.SetReceiveTimestampOffsetStatus(enable, id) ? 0 : -1;
-}
-
-int ViEChannel::SetSendAbsoluteSendTimeStatus(bool enable, int id) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  int error = 0;
-  if (enable) {
-    // Enable the extension, but disable possible old id to avoid errors.
-    absolute_send_time_extension_id_ = id;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionAbsoluteSendTime);
-    error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
-        kRtpExtensionAbsoluteSendTime, id);
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(
-          kRtpExtensionAbsoluteSendTime);
-      error |= (*it)->RegisterSendRtpHeaderExtension(
-          kRtpExtensionAbsoluteSendTime, id);
-    }
-  } else {
-    // Disable the extension.
-    absolute_send_time_extension_id_ = kInvalidRtpExtensionId;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionAbsoluteSendTime);
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(
-          kRtpExtensionAbsoluteSendTime);
-    }
-  }
-  return error;
-}
-
-int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
-  return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1;
-}
-
-int ViEChannel::SetSendVideoRotationStatus(bool enable, int id) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  int error = 0;
-  if (enable) {
-    // Enable the extension, but disable possible old id to avoid errors.
-    video_rotation_extension_id_ = id;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
-    error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
-        kRtpExtensionVideoRotation, id);
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
-      error |=
-          (*it)->RegisterSendRtpHeaderExtension(kRtpExtensionVideoRotation, id);
-    }
-  } else {
-    // Disable the extension.
-    video_rotation_extension_id_ = kInvalidRtpExtensionId;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
-    }
-  }
-  return error;
-}
-
-int ViEChannel::SetReceiveVideoRotationStatus(bool enable, int id) {
-  return vie_receiver_.SetReceiveVideoRotationStatus(enable, id) ? 0 : -1;
-}
-
-int ViEChannel::SetSendRtpStreamId(bool enable, int id, const char* rid) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  int error = 0;
-  if (enable) {
-    // Enable the extension, but disable possible old id to avoid errors.
-    rid_extension_id_ = id;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionRtpStreamId);
-    error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
-        kRtpExtensionRtpStreamId, id);
-    rtp_rtcp_->SetRID(rid);
-    // NOTE: simulcast streams must be set via the SetSendCodec() API
-  } else {
-    // Disable the extension.
-    rid_extension_id_ = kInvalidRtpExtensionId;
-    rtp_rtcp_->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionRtpStreamId);
-    // This may be overkill...
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); it++) {
-      (*it)->DeregisterSendRtpHeaderExtension(
-          kRtpExtensionRtpStreamId);
-    }
-  }
-  return error;
-}
-
-int ViEChannel::SetReceiveRtpStreamId(bool enable, int id) {
-  return vie_receiver_.SetReceiveRIDStatus(enable, id) ? 0 : -1;
-}
-
-void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  rtp_rtcp_->SetRtcpXrRrtrStatus(enable);
-}
-
-void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
-  assert(paced_sender_ && "No paced sender registered.");
-  paced_sender_->SetStatus(enable);
-}
-
-void ViEChannel::EnableTMMBR(bool enable) {
-  rtp_rtcp_->SetTMMBRStatus(enable);
-}
-
-int32_t ViEChannel::EnableKeyFrameRequestCallback(const bool enable) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (enable && !codec_observer_) {
-    LOG(LS_ERROR) << "No ViECodecObserver set.";
-    return -1;
-  }
-  do_key_frame_callbackRequest_ = enable;
-  return 0;
-}
-
-int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
-                            const StreamType usage,
-                            const uint8_t simulcast_idx) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  ReserveRtpRtcpModules(simulcast_idx + 1);
-  RtpRtcp* rtp_rtcp = GetRtpRtcpModule(simulcast_idx);
-  if (rtp_rtcp == NULL)
-    return -1;
-  if (usage == kViEStreamTypeRtx) {
-    rtp_rtcp->SetRtxSsrc(SSRC);
-  } else {
-    rtp_rtcp->SetSSRC(SSRC);
-  }
-  return 0;
-}
-
-int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage,
-                                      const uint32_t SSRC) {
-  vie_receiver_.SetRtxSsrc(SSRC);
-  return 0;
-}
-
-int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) {
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  RtpRtcp* rtp_rtcp = GetRtpRtcpModule(idx);
-  if (rtp_rtcp == NULL)
-    return -1;
-  *ssrc = rtp_rtcp->SSRC();
-  return 0;
-}
-
-int32_t ViEChannel::GetRemoteSSRC(uint32_t* ssrc) {
-  *ssrc = vie_receiver_.GetRemoteSsrc();
-  return 0;
-}
-
-int32_t ViEChannel::GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]) {
-  uint32_t arrayCSRC[kRtpCsrcSize];
-  memset(arrayCSRC, 0, sizeof(arrayCSRC));
-
-  int num_csrcs = vie_receiver_.GetCsrcs(arrayCSRC);
-  if (num_csrcs > 0) {
-    memcpy(CSRCs, arrayCSRC, num_csrcs * sizeof(uint32_t));
-  }
-  return 0;
-}
-
-int32_t ViEChannel::GetRemoteRID(char rid[256])
-{
-  vie_receiver_.GetRID(rid);
-  return 0;
-}
-
-int ViEChannel::SetRtxSendPayloadType(int payload_type) {
-  rtp_rtcp_->SetRtxSendPayloadType(payload_type);
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end(); it++) {
-    (*it)->SetRtxSendPayloadType(payload_type);
-  }
-  SetRtxSendStatus(true);
-  return 0;
-}
-
-void ViEChannel::SetRtxSendStatus(bool enable) {
-  int rtx_settings =
-      enable ? kRtxRetransmitted | kRtxRedundantPayloads : kRtxOff;
-  rtp_rtcp_->SetRtxSendStatus(rtx_settings);
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end(); it++) {
-    (*it)->SetRtxSendStatus(rtx_settings);
-  }
-}
-
-void ViEChannel::SetRtxReceivePayloadType(int payload_type) {
-  vie_receiver_.SetRtxPayloadType(payload_type);
-}
-
-int32_t ViEChannel::SetStartSequenceNumber(uint16_t sequence_number) {
-  if (rtp_rtcp_->Sending()) {
-    return -1;
-  }
-  rtp_rtcp_->SetSequenceNumber(sequence_number);
-  return 0;
-}
-
-void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) {
-  assert(!rtp_rtcp_->Sending());
-  if (rtp_rtcp_->SetRtpStateForSsrc(ssrc, rtp_state))
-    return;
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (auto* module : simulcast_rtp_rtcp_) {
-    if (module->SetRtpStateForSsrc(ssrc, rtp_state))
-      return;
-  }
-  for (auto* module : removed_rtp_rtcp_) {
-    if (module->SetRtpStateForSsrc(ssrc, rtp_state))
-      return;
-  }
-}
-
-RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) {
-  assert(!rtp_rtcp_->Sending());
-
-  RtpState rtp_state;
-  if (rtp_rtcp_->GetRtpStateForSsrc(ssrc, &rtp_state))
-    return rtp_state;
-
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (auto* module : simulcast_rtp_rtcp_) {
-    if (module->GetRtpStateForSsrc(ssrc, &rtp_state))
-      return rtp_state;
-  }
-  for (auto* module : removed_rtp_rtcp_) {
-    if (module->GetRtpStateForSsrc(ssrc, &rtp_state))
-      return rtp_state;
-  }
-  LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
-  return rtp_state;
-}
-
-int32_t ViEChannel::SetRTCPCName(const char rtcp_cname[]) {
-  if (rtp_rtcp_->Sending()) {
-    return -1;
-  }
-  return rtp_rtcp_->SetCNAME(rtcp_cname);
-}
-
-int32_t ViEChannel::GetRemoteRTCPCName(char rtcp_cname[]) {
-  uint32_t remoteSSRC = vie_receiver_.GetRemoteSsrc();
-  return rtp_rtcp_->RemoteCNAME(remoteSSRC, rtcp_cname);
-}
-
-int32_t ViEChannel::RegisterRtpObserver(ViERTPObserver* observer) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (observer) {
-    if (rtp_observer_) {
-      LOG_F(LS_ERROR) << "Observer already registered.";
-      return -1;
-    }
-    rtp_observer_ = observer;
-  } else {
-    rtp_observer_ = NULL;
-  }
-  return 0;
-}
-
-int32_t ViEChannel::SendApplicationDefinedRTCPPacket(
-    const uint8_t sub_type,
-    uint32_t name,
-    const uint8_t* data,
-    uint16_t data_length_in_bytes) {
-  if (!rtp_rtcp_->Sending()) {
-    return -1;
-  }
-  if (!data) {
-    LOG_F(LS_ERROR) << "Invalid input.";
-    return -1;
-  }
-  if (data_length_in_bytes % 4 != 0) {
-    LOG(LS_ERROR) << "Invalid input length.";
-    return -1;
-  }
-  RTCPMethod rtcp_method = rtp_rtcp_->RTCP();
-  if (rtcp_method == kRtcpOff) {
-    LOG_F(LS_ERROR) << "RTCP not enable.";
-    return -1;
-  }
-  // Create and send packet.
-  if (rtp_rtcp_->SetRTCPApplicationSpecificData(sub_type, name, data,
-                                               data_length_in_bytes) != 0) {
-    return -1;
-  }
-  return 0;
-}
-
-int32_t ViEChannel::GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh,
-                                              uint32_t& NTPLow,
-                                              uint32_t& receivedPacketCount,
-                                              uint64_t& receivedOctetCount,
-                                              uint32_t* jitterSamples,
-                                              uint16_t* fractionLost,
-                                              uint32_t* cumulativeLost,
-                                             int32_t* rttMs) {
-  // TODO: how do we do this for simulcast ? average for all
-  // except cumulative_lost that is the sum ?
-  // CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-
-  // for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-  //      it != simulcast_rtp_rtcp_.end();
-  //      it++) {
-  //   RtpRtcp* rtp_rtcp = *it;
-  // }
-  uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
-
-  // Get all RTCP receiver report blocks that have been received on this
-  // channel. If we receive RTP packets from a remote source we know the
-  // remote SSRC and use the report block from him.
-  // Otherwise use the first report block.
-  std::vector<RTCPReportBlock> remote_stats;
-  if (rtp_rtcp_->RemoteRTCPStat(&remote_stats) != 0 || remote_stats.empty()) {
-    LOG_F(LS_ERROR) << "Could not get remote stats";
-    return -1;
-  }
-  std::vector<RTCPReportBlock>::const_iterator statistics =
-      remote_stats.begin();
-  for (; statistics != remote_stats.end(); ++statistics) {
-    if (statistics->remoteSSRC == remote_ssrc)
-      break;
-  }
-
-  if (statistics == remote_stats.end()) {
-    // If we have not received any RTCP packets from this SSRC it probably means
-    // we have not received any RTP packets.
-    // Use the first received report block instead.
-    statistics = remote_stats.begin();
-    remote_ssrc = statistics->remoteSSRC;
-  }
-
-  if (rtp_rtcp_->GetReportBlockInfo(remote_ssrc,
-                                    &NTPHigh,
-                                    &NTPLow,
-                                    &receivedPacketCount,
-                                    &receivedOctetCount) != 0) {
-    LOG_F(LS_ERROR) << "failed to retrieve RTT";
-    NTPHigh = 0;
-    NTPLow = 0;
-    receivedPacketCount = 0;
-    receivedOctetCount = 0;
-  }
-
-  *fractionLost = statistics->fractionLost;
-  *cumulativeLost = statistics->cumulativeLost;
-  *jitterSamples = statistics->jitter;
-
-  int64_t dummy;
-  int64_t rtt = 0;
-  if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
-    LOG_F(LS_ERROR) << "failed to get RTT";
-    return -1;
-  }
-  *rttMs = rtt;
-  return 0;
-}
-
-int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost,
-                                          uint32_t* cumulative_lost,
-                                          uint32_t* extended_max,
-                                          uint32_t* jitter_samples,
-                                          int64_t* rtt_ms) {
-  // Aggregate the report blocks associated with streams sent on this channel.
-  std::vector<RTCPReportBlock> report_blocks;
-  rtp_rtcp_->RemoteRTCPStat(&report_blocks);
-  {
-    CriticalSectionScoped lock(rtp_rtcp_cs_.get());
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-        it != simulcast_rtp_rtcp_.end();
-        ++it) {
-      (*it)->RemoteRTCPStat(&report_blocks);
-    }
-  }
-
-  if (report_blocks.empty())
-    return -1;
-
-  uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
-  std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin();
-  for (; it != report_blocks.end(); ++it) {
-    if (it->remoteSSRC == remote_ssrc)
-      break;
-  }
-  if (it == report_blocks.end()) {
-    // We have not received packets with an SSRC matching the report blocks. To
-    // have a chance of calculating an RTT we will try with the SSRC of the
-    // first report block received.
-    // This is very important for send-only channels where we don't know the
-    // SSRC of the other end.
-    remote_ssrc = report_blocks[0].remoteSSRC;
-  }
-
-  // TODO(asapersson): Change report_block_stats to not rely on
-  // GetSendRtcpStatistics to be called.
-  RTCPReportBlock report =
-      report_block_stats_sender_->AggregateAndStore(report_blocks);
-  *fraction_lost = report.fractionLost;
-  *cumulative_lost = report.cumulativeLost;
-  *extended_max = report.extendedHighSeqNum;
-  *jitter_samples = report.jitter;
-
-  int64_t dummy;
-  int64_t rtt = 0;
-  if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
-    return -1;
-  }
-  *rtt_ms = rtt;
-  return 0;
-}
-
-void ViEChannel::RegisterSendChannelRtcpStatisticsCallback(
-    RtcpStatisticsCallback* callback) {
-  rtp_rtcp_->RegisterRtcpStatisticsCallback(callback);
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       ++it) {
-    (*it)->RegisterRtcpStatisticsCallback(callback);
-  }
-}
-
-// TODO(holmer): This is a bad function name as it implies that it returns the
-// received RTCP, while it actually returns the statistics which will be sent
-// in the RTCP.
-int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost,
-                                              uint32_t* cumulative_lost,
-                                              uint32_t* extended_max,
-                                              uint32_t* jitter_samples,
-                                              int64_t* rtt_ms) {
-  uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
-  StreamStatistician* statistician =
-      vie_receiver_.GetReceiveStatistics()->GetStatistician(remote_ssrc);
-  RtcpStatistics receive_stats;
-  if (!statistician || !statistician->GetStatistics(
-      &receive_stats, rtp_rtcp_->RTCP() == kRtcpOff)) {
-    return -1;
-  }
-  *fraction_lost = receive_stats.fraction_lost;
-  *cumulative_lost = receive_stats.cumulative_lost;
-  *extended_max = receive_stats.extended_max_sequence_number;
-  *jitter_samples = receive_stats.jitter;
-
-  // TODO(asapersson): Change report_block_stats to not rely on
-  // GetReceivedRtcpStatistics to be called.
-  report_block_stats_receiver_->Store(receive_stats, remote_ssrc, 0);
-
-  int64_t dummy = 0;
-  int64_t rtt = 0;
-  rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy);
-  *rtt_ms = rtt;
-  return 0;
-}
-
-void ViEChannel::RegisterReceiveChannelRtcpStatisticsCallback(
-    RtcpStatisticsCallback* callback) {
-  vie_receiver_.GetReceiveStatistics()->RegisterRtcpStatisticsCallback(
-      callback);
-  rtp_rtcp_->RegisterRtcpStatisticsCallback(callback);
-}
-
-void ViEChannel::RegisterRtcpPacketTypeCounterObserver(
-    RtcpPacketTypeCounterObserver* observer) {
-  rtcp_packet_type_counter_observer_.Set(observer);
-}
-
-int32_t ViEChannel::GetRtpStatistics(size_t* bytes_sent,
-                                     uint32_t* packets_sent,
-                                     size_t* bytes_received,
-                                     uint32_t* packets_received) const {
-  StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
-      GetStatistician(vie_receiver_.GetRemoteSsrc());
-  *bytes_received = 0;
-  *packets_received = 0;
-  if (statistician)
-    statistician->GetDataCounters(bytes_received, packets_received);
-  if (rtp_rtcp_->DataCountersRTP(bytes_sent, packets_sent) != 0) {
-    return -1;
-  }
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    size_t bytes_sent_temp = 0;
-    uint32_t packets_sent_temp = 0;
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
-    *bytes_sent += bytes_sent_temp;
-    *packets_sent += packets_sent_temp;
-  }
-  for (std::list<RtpRtcp*>::const_iterator it = removed_rtp_rtcp_.begin();
-       it != removed_rtp_rtcp_.end(); ++it) {
-    size_t bytes_sent_temp = 0;
-    uint32_t packets_sent_temp = 0;
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
-    *bytes_sent += bytes_sent_temp;
-    *packets_sent += packets_sent_temp;
-  }
-  return 0;
-}
-
-void ViEChannel::GetSendStreamDataCounters(
-    StreamDataCounters* rtp_counters,
-    StreamDataCounters* rtx_counters) const {
-  rtp_rtcp_->GetSendStreamDataCounters(rtp_counters, rtx_counters);
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    StreamDataCounters rtp_data;
-    StreamDataCounters rtx_data;
-    (*it)->GetSendStreamDataCounters(&rtp_data, &rtx_data);
-    rtp_counters->Add(rtp_data);
-    rtx_counters->Add(rtx_data);
-  }
-  for (std::list<RtpRtcp*>::const_iterator it = removed_rtp_rtcp_.begin();
-       it != removed_rtp_rtcp_.end(); ++it) {
-    StreamDataCounters rtp_data;
-    StreamDataCounters rtx_data;
-    (*it)->GetSendStreamDataCounters(&rtp_data, &rtx_data);
-    rtp_counters->Add(rtp_data);
-    rtx_counters->Add(rtx_data);
-  }
-}
-
-void ViEChannel::GetReceiveStreamDataCounters(
-    StreamDataCounters* rtp_counters,
-    StreamDataCounters* rtx_counters) const {
-  StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
-      GetStatistician(vie_receiver_.GetRemoteSsrc());
-  if (statistician) {
-    statistician->GetReceiveStreamDataCounters(rtp_counters);
-  }
-  uint32_t rtx_ssrc = 0;
-  if (vie_receiver_.GetRtxSsrc(&rtx_ssrc)) {
-    StreamStatistician* statistician =
-        vie_receiver_.GetReceiveStatistics()->GetStatistician(rtx_ssrc);
-    if (statistician) {
-      statistician->GetReceiveStreamDataCounters(rtx_counters);
-    }
-  }
-}
-
-void ViEChannel::RegisterSendChannelRtpStatisticsCallback(
-      StreamDataCountersCallback* callback) {
-  rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(callback);
-  {
-    CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-    for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end();
-         it++) {
-      (*it)->RegisterSendChannelRtpStatisticsCallback(callback);
-    }
-  }
-}
-
-void ViEChannel::RegisterReceiveChannelRtpStatisticsCallback(
-    StreamDataCountersCallback* callback) {
-  vie_receiver_.GetReceiveStatistics()->RegisterRtpStatisticsCallback(callback);
-}
-
-void ViEChannel::GetSendRtcpPacketTypeCounter(
-    RtcpPacketTypeCounter* packet_counter) const {
-  std::map<uint32_t, RtcpPacketTypeCounter> counter_map =
-      rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap();
-
-  RtcpPacketTypeCounter counter;
-  counter.Add(counter_map[rtp_rtcp_->SSRC()]);
-
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end(); ++it) {
-    counter.Add(counter_map[(*it)->SSRC()]);
-  }
-  for (std::list<RtpRtcp*>::const_iterator it = removed_rtp_rtcp_.begin();
-       it != removed_rtp_rtcp_.end(); ++it) {
-    counter.Add(counter_map[(*it)->SSRC()]);
-  }
-  *packet_counter = counter;
-}
-
-void ViEChannel::GetReceiveRtcpPacketTypeCounter(
-    RtcpPacketTypeCounter* packet_counter) const {
-  std::map<uint32_t, RtcpPacketTypeCounter> counter_map =
-      rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap();
-
-  RtcpPacketTypeCounter counter;
-  counter.Add(counter_map[vie_receiver_.GetRemoteSsrc()]);
-
-  *packet_counter = counter;
-}
-
-int32_t ViEChannel::GetRemoteRTCPSenderInfo(SenderInfo* sender_info) const {
-  // Get the sender info from the latest received RTCP Sender Report.
-  RTCPSenderInfo rtcp_sender_info;
-  if (rtp_rtcp_->RemoteRTCPStat(&rtcp_sender_info) != 0) {
-    LOG_F(LS_ERROR) << "failed to read RTCP SR sender info";
-    return -1;
-  }
-
-  sender_info->NTP_timestamp_high = rtcp_sender_info.NTPseconds;
-  sender_info->NTP_timestamp_low = rtcp_sender_info.NTPfraction;
-  sender_info->RTP_timestamp = rtcp_sender_info.RTPtimeStamp;
-  sender_info->sender_packet_count = rtcp_sender_info.sendPacketCount;
-  sender_info->sender_octet_count = rtcp_sender_info.sendOctetCount;
-  return 0;
-}
-
-void ViEChannel::GetBandwidthUsage(uint32_t* total_bitrate_sent,
-                                   uint32_t* video_bitrate_sent,
-                                   uint32_t* fec_bitrate_sent,
-                                   uint32_t* nackBitrateSent) const {
-  rtp_rtcp_->BitrateSent(total_bitrate_sent, video_bitrate_sent,
-                         fec_bitrate_sent, nackBitrateSent);
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end(); it++) {
-    uint32_t stream_rate = 0;
-    uint32_t video_rate = 0;
-    uint32_t fec_rate = 0;
-    uint32_t nackRate = 0;
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->BitrateSent(&stream_rate, &video_rate, &fec_rate, &nackRate);
-    *total_bitrate_sent += stream_rate;
-    *video_bitrate_sent += video_rate;
-    *fec_bitrate_sent += fec_rate;
-    *nackBitrateSent += nackRate;
-  }
-}
-
-bool ViEChannel::GetSendSideDelay(int* avg_send_delay,
-                                  int* max_send_delay) const {
-  *avg_send_delay = 0;
-  *max_send_delay = 0;
-  bool valid_estimate = false;
-  int num_send_delays = 0;
-  if (rtp_rtcp_->GetSendSideDelay(avg_send_delay, max_send_delay)) {
-    ++num_send_delays;
-    valid_estimate = true;
-  }
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end(); it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    int sub_stream_avg_delay = 0;
-    int sub_stream_max_delay = 0;
-    if (rtp_rtcp->GetSendSideDelay(&sub_stream_avg_delay,
-                                   &sub_stream_max_delay)) {
-      *avg_send_delay += sub_stream_avg_delay;
-      *max_send_delay = std::max(*max_send_delay, sub_stream_max_delay);
-      ++num_send_delays;
-    }
-  }
-  if (num_send_delays > 0) {
-    valid_estimate = true;
-    *avg_send_delay = *avg_send_delay / num_send_delays;
-    *avg_send_delay = (*avg_send_delay + num_send_delays / 2) / num_send_delays;
-  }
-  return valid_estimate;
-}
-
-void ViEChannel::RegisterSendSideDelayObserver(
-    SendSideDelayObserver* observer) {
-  send_side_delay_observer_.Set(observer);
-}
-
-void ViEChannel::RegisterSendBitrateObserver(
-    BitrateStatisticsObserver* observer) {
-  send_bitrate_observer_.Set(observer);
-}
-
-int32_t ViEChannel::StartRTPDump(const char file_nameUTF8[1024],
-                                 RTPDirections direction) {
-  if (direction == kRtpIncoming) {
-    return vie_receiver_.StartRTPDump(file_nameUTF8);
-  } else {
-    return vie_sender_.StartRTPDump(file_nameUTF8);
-  }
-}
-
-int32_t ViEChannel::StopRTPDump(RTPDirections direction) {
-  if (direction == kRtpIncoming) {
-    return vie_receiver_.StopRTPDump();
-  } else {
-    return vie_sender_.StopRTPDump();
-  }
-}
-
-int32_t ViEChannel::StartSend() {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (!external_transport_) {
-    LOG(LS_ERROR) << "No transport set.";
-    return -1;
-  }
-  rtp_rtcp_->SetSendingMediaStatus(true);
-
-  if (rtp_rtcp_->Sending()) {
-    return kViEBaseAlreadySending;
-  }
-  if (rtp_rtcp_->SetSendingStatus(true) != 0) {
-    return -1;
-  }
-  CriticalSectionScoped cs_rtp(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->SetSendingMediaStatus(true);
-    rtp_rtcp->SetSendingStatus(true);
-  }
-  send_payload_router_->set_active(true);
-  vie_receiver_.StartRTCPReceive();
-  return 0;
-}
-
-int32_t ViEChannel::StopSend() {
-  UpdateHistogramsAtStopSend();
-  send_payload_router_->set_active(false);
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  rtp_rtcp_->SetSendingMediaStatus(false);
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->SetSendingMediaStatus(false);
-  }
-  if (!rtp_rtcp_->Sending()) {
-    return kViEBaseNotSending;
-  }
-
-  // Reset.
-  rtp_rtcp_->ResetSendDataCountersRTP();
-  if (rtp_rtcp_->SetSendingStatus(false) != 0) {
-    return -1;
-  }
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->ResetSendDataCountersRTP();
-    rtp_rtcp->SetSendingStatus(false);
-  }
-  vie_receiver_.StopRTCPReceive();
-  return 0;
-}
-
-bool ViEChannel::Sending() {
-  return rtp_rtcp_->Sending();
-}
-
-int32_t ViEChannel::StartReceive() {
-  if (StartDecodeThread() != 0) {
-    vie_receiver_.StopReceive();
-    return -1;
-  }
-  vie_receiver_.StartReceive();
-  vie_receiver_.StartRTCPReceive(); // For receiving RTCP SR in one-way connections
-  return 0;
-}
-
-int32_t ViEChannel::StopReceive() {
-  vie_receiver_.StopReceive();
-  vie_receiver_.StopRTCPReceive();
-  StopDecodeThread();
-  vcm_->ResetDecoder();
-  return 0;
-}
-
-int32_t ViEChannel::RegisterSendTransport(Transport* transport) {
-  if (rtp_rtcp_->Sending()) {
-    return -1;
-  }
-
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (external_transport_) {
-    LOG_F(LS_ERROR) << "Transport already registered.";
-    return -1;
-  }
-  external_transport_ = transport;
-  vie_sender_.RegisterSendTransport(transport);
-  return 0;
-}
-
-int32_t ViEChannel::DeregisterSendTransport() {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (!external_transport_) {
-    return 0;
-  }
-  if (rtp_rtcp_->Sending()) {
-    LOG_F(LS_ERROR) << "Can't deregister transport when sending.";
-    return -1;
-  }
-  external_transport_ = NULL;
-  vie_sender_.DeregisterSendTransport();
-  return 0;
-}
-
-int32_t ViEChannel::ReceivedRTPPacket(
-    const void* rtp_packet, const size_t rtp_packet_length,
-    const PacketTime& packet_time) {
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    if (!external_transport_) {
-      return -1;
-    }
-  }
-  return vie_receiver_.ReceivedRTPPacket(
-      rtp_packet, rtp_packet_length, packet_time);
-}
-
-int32_t ViEChannel::ReceivedRTCPPacket(
-  const void* rtcp_packet, const size_t rtcp_packet_length) {
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    if (!external_transport_) {
-      return -1;
-    }
-  }
-  return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length);
-}
-
-int32_t ViEChannel::SetMTU(uint16_t mtu) {
-  if (rtp_rtcp_->SetMaxTransferUnit(mtu) != 0) {
-    return -1;
-  }
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end();
-       it++) {
-    RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->SetMaxTransferUnit(mtu);
-  }
-  mtu_ = mtu;
-  return 0;
-}
-
-uint16_t ViEChannel::MaxDataPayloadLength() const {
-  return rtp_rtcp_->MaxDataPayloadLength();
-}
-
-int32_t ViEChannel::EnableColorEnhancement(bool enable) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  color_enhancement_ = enable;
-  return 0;
-}
-
-RtpRtcp* ViEChannel::rtp_rtcp() {
-  return rtp_rtcp_.get();
-}
-
-scoped_refptr<PayloadRouter> ViEChannel::send_payload_router() {
-  return send_payload_router_;
-}
-
-VCMProtectionCallback* ViEChannel::vcm_protection_callback() {
-  return vcm_protection_callback_.get();
-}
-
-CallStatsObserver* ViEChannel::GetStatsObserver() {
-  return stats_observer_.get();
-}
-
-// Do not acquire the lock of |vcm_| in this function. Decode callback won't
-// necessarily be called from the decoding thread. The decoding thread may have
-// held the lock when calling VideoDecoder::Decode, Reset, or Release. Acquiring
-// the same lock in the path of decode callback can deadlock.
-int32_t ViEChannel::FrameToRender(
-    I420VideoFrame& video_frame) {  // NOLINT
-  CriticalSectionScoped cs(callback_cs_.get());
-
-  if (decoder_reset_) {
-    // Trigger a callback to the user if the incoming codec has changed.
-    if (codec_observer_) {
-      // The codec set by RegisterReceiveCodec might not be the size we're
-      // actually decoding.
-      receive_codec_.width = static_cast<uint16_t>(video_frame.width());
-      receive_codec_.height = static_cast<uint16_t>(video_frame.height());
-      codec_observer_->IncomingCodecChanged(channel_id_, receive_codec_);
-    }
-    decoder_reset_ = false;
-  }
-  // Post processing is not supported if the frame is backed by a texture.
-  if (video_frame.native_handle() == NULL) {
-    if (pre_render_callback_ != NULL)
-      pre_render_callback_->FrameCallback(&video_frame);
-    if (effect_filter_) {
-      size_t length =
-          CalcBufferSize(kI420, video_frame.width(), video_frame.height());
-      rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
-      ExtractBuffer(video_frame, length, video_buffer.get());
-      effect_filter_->Transform(length,
-                                video_buffer.get(),
-                                video_frame.ntp_time_ms(),
-                                video_frame.timestamp(),
-                                video_frame.width(),
-                                video_frame.height());
-    }
-    if (color_enhancement_) {
-      VideoProcessingModule::ColorEnhancement(&video_frame);
-    }
-  }
-
-  uint32_t arr_ofCSRC[kRtpCsrcSize];
-  int32_t no_of_csrcs = vie_receiver_.GetCsrcs(arr_ofCSRC);
-  if (no_of_csrcs <= 0) {
-    arr_ofCSRC[0] = vie_receiver_.GetRemoteSsrc();
-    no_of_csrcs = 1;
-  }
-  std::vector<uint32_t> csrcs(arr_ofCSRC, arr_ofCSRC + no_of_csrcs);
-  DeliverFrame(&video_frame, csrcs);
-
-  return 0;
-}
-
-int32_t ViEChannel::ReceivedDecodedReferenceFrame(
-  const uint64_t picture_id) {
-  return rtp_rtcp_->SendRTCPReferencePictureSelection(picture_id);
-}
-
-void ViEChannel::IncomingCodecChanged(const VideoCodec& codec) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  receive_codec_ = codec;
-}
-
-void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (codec_observer_)
-    codec_observer_->IncomingRate(channel_id_, frame_rate, bit_rate);
-}
-
-void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (vcm_receive_stats_callback_ != NULL)
-    vcm_receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets);
-}
-
-void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  receive_frame_counts_ = frame_counts;
-  if (vcm_receive_stats_callback_ != NULL)
-    vcm_receive_stats_callback_->OnFrameCountsUpdated(frame_counts);
-}
-
-void ViEChannel::OnDecoderTiming(int decode_ms,
-                                 int max_decode_ms,
-                                 int current_delay_ms,
-                                 int target_delay_ms,
-                                 int jitter_buffer_ms,
-                                 int min_playout_delay_ms,
-                                 int render_delay_ms) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (!codec_observer_)
-    return;
-  codec_observer_->DecoderTiming(decode_ms,
-                                 max_decode_ms,
-                                 current_delay_ms,
-                                 target_delay_ms,
-                                 jitter_buffer_ms,
-                                 min_playout_delay_ms,
-                                 render_delay_ms);
-}
-
-int32_t ViEChannel::RequestKeyFrame() {
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    if (codec_observer_ && do_key_frame_callbackRequest_) {
-      codec_observer_->RequestNewKeyFrame(channel_id_);
-    }
-  }
-  return rtp_rtcp_->RequestKeyFrame();
-}
-
-int32_t ViEChannel::SliceLossIndicationRequest(
-  const uint64_t picture_id) {
-  return rtp_rtcp_->SendRTCPSliceLossIndication((uint8_t) picture_id);
-}
-
-int32_t ViEChannel::ResendPackets(const uint16_t* sequence_numbers,
-                                        uint16_t length) {
-  return rtp_rtcp_->SendNACK(sequence_numbers, length);
-}
-
-void ViEChannel::ReceiveStateChange(VideoReceiveState state) {
-  LOG_F(LS_INFO);
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    if (codec_observer_) {
-      codec_observer_->ReceiveStateChange(channel_id_, state);
-    }
-  }
-}
-
-bool ViEChannel::ChannelDecodeThreadFunction(void* obj) {
-  return static_cast<ViEChannel*>(obj)->ChannelDecodeProcess();
-}
-
-bool ViEChannel::ChannelDecodeProcess() {
-  // TODO(pbos): Make sure the decoder thread doesn't run for send-only
-  // channels.
-  vcm_->Decode(kMaxDecodeWaitTimeMs);
-  return true;
-}
-
-void ViEChannel::OnRttUpdate(int64_t rtt) {
-  vcm_->SetReceiveChannelParameters(rtt);
-}
-
-int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params,
-                                  const FecProtectionParams* key_fec_params,
-                                  uint32_t* video_rate_bps,
-                                  uint32_t* nack_rate_bps,
-                                  uint32_t* fec_rate_bps) {
-  uint32_t not_used = 0;
-  rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
-  rtp_rtcp_->BitrateSent(&not_used, video_rate_bps, fec_rate_bps,
-                         nack_rate_bps);
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
-  for (auto* module : simulcast_rtp_rtcp_) {
-    uint32_t child_video_rate = 0;
-    uint32_t child_fec_rate = 0;
-    uint32_t child_nack_rate = 0;
-    module->SetFecParameters(delta_fec_params, key_fec_params);
-    module->BitrateSent(&not_used, &child_video_rate, &child_fec_rate,
-                        &child_nack_rate);
-    *video_rate_bps += child_video_rate;
-    *nack_rate_bps += child_nack_rate;
-    *fec_rate_bps += child_fec_rate;
-  }
-  return 0;
-}
-
-void ViEChannel::ReserveRtpRtcpModules(size_t num_modules) {
-  for (size_t total_modules =
-           1 + simulcast_rtp_rtcp_.size() + removed_rtp_rtcp_.size();
-       total_modules < num_modules;
-       ++total_modules) {
-    RtpRtcp* rtp_rtcp = CreateRtpRtcpModule();
-    rtp_rtcp->SetSendingStatus(false);
-    rtp_rtcp->SetSendingMediaStatus(false);
-    rtp_rtcp->RegisterRtcpStatisticsCallback(NULL);
-    rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
-    removed_rtp_rtcp_.push_back(rtp_rtcp);
-  }
-}
-
-RtpRtcp* ViEChannel::GetRtpRtcpModule(size_t index) const {
-  if (index == 0)
-    return rtp_rtcp_.get();
-  if (index <= simulcast_rtp_rtcp_.size()) {
-    std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
-    for (size_t i = 1; i < index; ++i) {
-      ++it;
-    }
-    return *it;
-  }
-
-  // If the requested module exists it must be in the removed list. Index
-  // translation to this list must remove the default module as well as all
-  // active simulcast modules.
-  size_t removed_idx = index - simulcast_rtp_rtcp_.size() - 1;
-  if (removed_idx >= removed_rtp_rtcp_.size())
-    return NULL;
-
-  std::list<RtpRtcp*>::const_iterator it = removed_rtp_rtcp_.begin();
-  while (removed_idx-- > 0)
-    ++it;
-
-  return *it;
-}
-
-RtpRtcp::Configuration ViEChannel::CreateRtpRtcpConfiguration() {
-  RtpRtcp::Configuration configuration;
-  configuration.id = ViEModuleId(engine_id_, channel_id_);
-  configuration.audio = false;
-  configuration.outgoing_transport = &vie_sender_;
-  configuration.intra_frame_callback = intra_frame_observer_;
-  configuration.bandwidth_callback = bandwidth_observer_.get();
-  configuration.rtt_stats = rtt_stats_;
-  configuration.rtcp_packet_type_counter_observer =
-      &rtcp_packet_type_counter_observer_;
-  configuration.paced_sender = paced_sender_;
-  configuration.send_bitrate_observer = &send_bitrate_observer_;
-  configuration.send_frame_count_observer = &send_frame_count_observer_;
-  configuration.send_side_delay_observer = &send_side_delay_observer_;
-
-  return configuration;
-}
-
-RtpRtcp* ViEChannel::CreateRtpRtcpModule() {
-  return RtpRtcp::CreateRtpRtcp(CreateRtpRtcpConfiguration());
-}
-
-int32_t ViEChannel::StartDecodeThread() {
-  // Start the decode thread
-  if (decode_thread_) {
-    // Already started.
-    return 0;
-  }
-  decode_thread_ = ThreadWrapper::CreateThread(ChannelDecodeThreadFunction,
-                                               this, "DecodingThread");
-  decode_thread_->Start();
-  decode_thread_->SetPriority(kHighestPriority);
-  return 0;
-}
-
-int32_t ViEChannel::StopDecodeThread() {
-  if (!decode_thread_) {
-    return 0;
-  }
-
-  vcm_->TriggerDecoderShutdown();
-
-  decode_thread_->Stop();
-  decode_thread_.reset();
-  frame_delivery_thread_checker_.DetachFromThread();
-
-  return 0;
-}
-
-int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id,
-                                    VoEVideoSync* ve_sync_interface) {
-  return vie_sync_.ConfigureSync(ve_channel_id,
-                                 ve_sync_interface,
-                                 rtp_rtcp_.get(),
-                                 vie_receiver_.GetRtpReceiver());
-}
-
-int32_t ViEChannel::VoiceChannel() {
-  return vie_sync_.VoiceChannel();
-}
-
-int32_t ViEChannel::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (effect_filter && effect_filter_) {
-    LOG(LS_ERROR) << "Effect filter already registered.";
-    return -1;
-  }
-  effect_filter_ = effect_filter;
-  return 0;
-}
-
-void ViEChannel::RegisterPreRenderCallback(
-    I420FrameCallback* pre_render_callback) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  pre_render_callback_ = pre_render_callback;
-}
-
-void ViEChannel::RegisterPreDecodeImageCallback(
-    EncodedImageCallback* pre_decode_callback) {
-  vcm_->RegisterPreDecodeImageCallback(pre_decode_callback);
-}
-
-int32_t ViEChannel::OnInitializeDecoder(
-    const int32_t id,
-    const int8_t payload_type,
-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-    const int frequency,
-    const uint8_t channels,
-    const uint32_t rate) {
-  LOG(LS_INFO) << "OnInitializeDecoder " << static_cast<int>(payload_type)
-               << " " << payload_name;
-  vcm_->ResetDecoder();
-
-  CriticalSectionScoped cs(callback_cs_.get());
-  decoder_reset_ = true;
-  return 0;
-}
-
-void ViEChannel::OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) {
-  assert(channel_id_ == ChannelId(id));
-  rtp_rtcp_->SetRemoteSSRC(ssrc);
-
-  CriticalSectionScoped cs(callback_cs_.get());
-  {
-    if (rtp_observer_) {
-      rtp_observer_->IncomingSSRCChanged(channel_id_, ssrc);
-    }
-  }
-}
-
-void ViEChannel::OnIncomingCSRCChanged(const int32_t id,
-                                       const uint32_t CSRC,
-                                       const bool added) {
-  assert(channel_id_ == ChannelId(id));
-  CriticalSectionScoped cs(callback_cs_.get());
-  {
-    if (rtp_observer_) {
-      rtp_observer_->IncomingCSRCChanged(channel_id_, CSRC, added);
-    }
-  }
-}
-
-void ViEChannel::ResetStatistics(uint32_t ssrc) {
-  StreamStatistician* statistician =
-      vie_receiver_.GetReceiveStatistics()->GetStatistician(ssrc);
-  if (statistician)
-    statistician->ResetStatistics();
-}
-
-void ViEChannel::RegisterSendFrameCountObserver(
-    FrameCountObserver* observer) {
-  send_frame_count_observer_.Set(observer);
-}
-
-void ViEChannel::RegisterReceiveStatisticsProxy(
-    ReceiveStatisticsProxy* receive_statistics_proxy) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  vcm_receive_stats_callback_ = receive_statistics_proxy;
-}
-
-void ViEChannel::ReceivedBWEPacket(int64_t arrival_time_ms,
-                                   size_t payload_size,
-                                   const RTPHeader& header) {
-  vie_receiver_.ReceivedBWEPacket(arrival_time_ms, payload_size, header);
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc b/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc
deleted file mode 100644
index 5c81dd830b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc
+++ /dev/null
@@ -1,512 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_channel_group.h"
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/common.h"
-#include "webrtc/experiments.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/call_stats.h"
-#include "webrtc/video_engine/encoder_state_feedback.h"
-#include "webrtc/video_engine/payload_router.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_remb.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-
-namespace webrtc {
-namespace {
-
-static const uint32_t kTimeOffsetSwitchThreshold = 30;
-
-class WrappingBitrateEstimator : public RemoteBitrateEstimator {
- public:
-  WrappingBitrateEstimator(RemoteBitrateObserver* observer,
-                           Clock* clock,
-                           const Config& config)
-      : observer_(observer),
-        clock_(clock),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-        min_bitrate_bps_(config.Get<RemoteBitrateEstimatorMinRate>().min_rate),
-        rbe_(RemoteBitrateEstimatorFactory().Create(observer_,
-                                                    clock_,
-                                                    kAimdControl,
-                                                    min_bitrate_bps_)),
-        using_absolute_send_time_(false),
-        packets_since_absolute_send_time_(0) {
-  }
-
-  virtual ~WrappingBitrateEstimator() {}
-
-  void IncomingPacket(int64_t arrival_time_ms,
-                      size_t payload_size,
-                      const RTPHeader& header) override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    PickEstimatorFromHeader(header);
-    rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
-  }
-
-  int32_t Process() override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return rbe_->Process();
-  }
-
-  int64_t TimeUntilNextProcess() override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return rbe_->TimeUntilNextProcess();
-  }
-
-  void OnRttUpdate(int64_t rtt) override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    rbe_->OnRttUpdate(rtt);
-  }
-
-  void RemoveStream(unsigned int ssrc) override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    rbe_->RemoveStream(ssrc);
-  }
-
-  bool LatestEstimate(std::vector<unsigned int>* ssrcs,
-                      unsigned int* bitrate_bps) const override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return rbe_->LatestEstimate(ssrcs, bitrate_bps);
-  }
-
-  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return rbe_->GetStats(output);
-  }
-
- private:
-  void PickEstimatorFromHeader(const RTPHeader& header)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
-    if (header.extension.hasAbsoluteSendTime) {
-      // If we see AST in header, switch RBE strategy immediately.
-      if (!using_absolute_send_time_) {
-        LOG(LS_INFO) <<
-            "WrappingBitrateEstimator: Switching to absolute send time RBE.";
-        using_absolute_send_time_ = true;
-        PickEstimator();
-      }
-      packets_since_absolute_send_time_ = 0;
-    } else {
-      // When we don't see AST, wait for a few packets before going back to TOF.
-      if (using_absolute_send_time_) {
-        ++packets_since_absolute_send_time_;
-        if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
-          LOG(LS_INFO) << "WrappingBitrateEstimator: Switching to transmission "
-                       << "time offset RBE.";
-          using_absolute_send_time_ = false;
-          PickEstimator();
-        }
-      }
-    }
-  }
-
-  // Instantiate RBE for Time Offset or Absolute Send Time extensions.
-  void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
-    if (using_absolute_send_time_) {
-      rbe_.reset(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
-          observer_, clock_, kAimdControl, min_bitrate_bps_));
-    } else {
-      rbe_.reset(RemoteBitrateEstimatorFactory().Create(
-          observer_, clock_, kAimdControl, min_bitrate_bps_));
-    }
-  }
-
-  RemoteBitrateObserver* observer_;
-  Clock* clock_;
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  const uint32_t min_bitrate_bps_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> rbe_;
-  bool using_absolute_send_time_;
-  uint32_t packets_since_absolute_send_time_;
-
-  DISALLOW_IMPLICIT_CONSTRUCTORS(WrappingBitrateEstimator);
-};
-}  // namespace
-
-ChannelGroup::ChannelGroup(ProcessThread* process_thread, const Config* config)
-    : remb_(new VieRemb()),
-      bitrate_allocator_(new BitrateAllocator()),
-      call_stats_(new CallStats()),
-      encoder_state_feedback_(new EncoderStateFeedback()),
-      packet_router_(new PacketRouter()),
-      pacer_(new PacedSender(Clock::GetRealTimeClock(),
-                             packet_router_.get(),
-                             BitrateController::kDefaultStartBitrateKbps,
-                             PacedSender::kDefaultPaceMultiplier *
-                                 BitrateController::kDefaultStartBitrateKbps,
-                             0)),
-      encoder_map_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      config_(config),
-      own_config_(),
-      process_thread_(process_thread),
-      pacer_thread_(ProcessThread::Create()),
-      // Constructed last as this object calls the provided callback on
-      // construction.
-      bitrate_controller_(
-          BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
-                                                     this)) {
-  if (!config) {
-    own_config_.reset(new Config);
-    config_ = own_config_.get();
-  }
-  DCHECK(config_);  // Must have a valid config pointer here.
-
-  remote_bitrate_estimator_.reset(
-      new WrappingBitrateEstimator(remb_.get(),
-                                   Clock::GetRealTimeClock(),
-                                   *config_));
-
-  call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
-
-  pacer_thread_->RegisterModule(pacer_.get());
-  pacer_thread_->Start();
-
-  process_thread->RegisterModule(remote_bitrate_estimator_.get());
-  process_thread->RegisterModule(call_stats_.get());
-  process_thread->RegisterModule(bitrate_controller_.get());
-}
-
-ChannelGroup::~ChannelGroup() {
-  pacer_thread_->Stop();
-  pacer_thread_->DeRegisterModule(pacer_.get());
-  process_thread_->DeRegisterModule(bitrate_controller_.get());
-  process_thread_->DeRegisterModule(call_stats_.get());
-  process_thread_->DeRegisterModule(remote_bitrate_estimator_.get());
-  call_stats_->DeregisterStatsObserver(remote_bitrate_estimator_.get());
-  DCHECK(channels_.empty());
-  DCHECK(channel_map_.empty());
-  DCHECK(!remb_->InUse());
-  DCHECK(vie_encoder_map_.empty());
-  DCHECK(send_encoders_.empty());
-}
-
-bool ChannelGroup::CreateSendChannel(int channel_id,
-                                     int engine_id,
-                                     int number_of_cores,
-                                     bool disable_default_encoder) {
-  rtc::scoped_ptr<ViEEncoder> vie_encoder(new ViEEncoder(
-      channel_id, number_of_cores, *config_, *process_thread_, pacer_.get(),
-      bitrate_allocator_.get(), bitrate_controller_.get(), false));
-  if (!vie_encoder->Init()) {
-    return false;
-  }
-  ViEEncoder* encoder = vie_encoder.get();
-  if (!CreateChannel(channel_id, engine_id, number_of_cores,
-                     vie_encoder.release(), true, disable_default_encoder)) {
-    return false;
-  }
-  ViEChannel* channel = channel_map_[channel_id];
-  // Connect the encoder with the send packet router, to enable sending.
-  encoder->StartThreadsAndSetSharedMembers(channel->send_payload_router(),
-                                           channel->vcm_protection_callback());
-
-  // Register the ViEEncoder to get key frame requests for this channel.
-  unsigned int ssrc = 0;
-  int stream_idx = 0;
-  channel->GetLocalSSRC(stream_idx, &ssrc);
-  encoder_state_feedback_->AddEncoder(ssrc, encoder);
-  std::list<unsigned int> ssrcs;
-  ssrcs.push_back(ssrc);
-  encoder->SetSsrcs(ssrcs);
-  return true;
-}
-
-bool ChannelGroup::CreateReceiveChannel(int channel_id,
-                                        int engine_id,
-                                        int base_channel_id,
-                                        int number_of_cores,
-                                        bool disable_default_encoder) {
-  ViEEncoder* encoder = GetEncoder(base_channel_id);
-  return CreateChannel(channel_id, engine_id, number_of_cores, encoder, false,
-                       disable_default_encoder);
-}
-
-bool ChannelGroup::CreateChannel(int channel_id,
-                                 int engine_id,
-                                 int number_of_cores,
-                                 ViEEncoder* vie_encoder,
-                                 bool sender,
-                                 bool disable_default_encoder) {
-  DCHECK(vie_encoder);
-
-  rtc::scoped_ptr<ViEChannel> channel(new ViEChannel(
-      channel_id, engine_id, number_of_cores, *config_, *process_thread_,
-      encoder_state_feedback_->GetRtcpIntraFrameObserver(),
-      bitrate_controller_->CreateRtcpBandwidthObserver(),
-      remote_bitrate_estimator_.get(), call_stats_->rtcp_rtt_stats(),
-      pacer_.get(), packet_router_.get(), sender, disable_default_encoder));
-  if (channel->Init() != 0) {
-    return false;
-  }
-  if (!disable_default_encoder) {
-    VideoCodec encoder;
-    if (vie_encoder->GetEncoder(&encoder) != 0) {
-      return false;
-    }
-    if (sender && channel->SetSendCodec(encoder) != 0) {
-      return false;
-    }
-  }
-
-  // Register the channel to receive stats updates.
-  call_stats_->RegisterStatsObserver(channel->GetStatsObserver());
-
-  // Store the channel, add it to the channel group and save the vie_encoder.
-  channel_map_[channel_id] = channel.release();
-  {
-    CriticalSectionScoped lock(encoder_map_cs_.get());
-    vie_encoder_map_[channel_id] = vie_encoder;
-    if (sender)
-      send_encoders_[channel_id] = vie_encoder;
-  }
-
-  return true;
-}
-
-void ChannelGroup::Stop(int channel_id) {
-  ViEEncoder* vie_encoder = GetEncoder(channel_id);
-  DCHECK(vie_encoder != NULL);
-
-  // If we're owning the encoder, remove the feedback and stop all encoding
-  // threads and processing. This must be done before deleting the channel.
-  if (vie_encoder->channel_id() == channel_id) {
-    encoder_state_feedback_->RemoveEncoder(vie_encoder);
-    vie_encoder->StopThreadsAndRemoveSharedMembers();
-  }
-}
-
-void ChannelGroup::DeleteChannel(int channel_id) {
-  ViEChannel* vie_channel = PopChannel(channel_id);
-
-  ViEEncoder* vie_encoder = GetEncoder(channel_id);
-  DCHECK(vie_encoder != NULL);
-
-  call_stats_->DeregisterStatsObserver(vie_channel->GetStatsObserver());
-  SetChannelRembStatus(channel_id, false, false, vie_channel);
-
-  // If we're owning the encoder, remove the feedback and stop all encoding
-  // threads and processing. This must be done before deleting the channel.
-  if (vie_encoder->channel_id() == channel_id) {
-    encoder_state_feedback_->RemoveEncoder(vie_encoder);
-    vie_encoder->StopThreadsAndRemoveSharedMembers();
-  }
-
-  unsigned int remote_ssrc = 0;
-  vie_channel->GetRemoteSSRC(&remote_ssrc);
-  RemoveChannel(channel_id);
-  remote_bitrate_estimator_->RemoveStream(remote_ssrc);
-
-  // Check if other channels are using the same encoder.
-  if (OtherChannelsUsingEncoder(channel_id)) {
-    vie_encoder = NULL;
-  } else {
-    // Delete later when we've released the critsect.
-  }
-
-  // We can't erase the item before we've checked for other channels using
-  // same ViEEncoder.
-  PopEncoder(channel_id);
-
-  delete vie_channel;
-  // Leave the write critsect before deleting the objects.
-  // Deleting a channel can cause other objects, such as renderers, to be
-  // deleted, which might take time.
-  // If statment just to show that this object is not always deleted.
-  if (vie_encoder) {
-    LOG(LS_VERBOSE) << "ViEEncoder deleted for channel " << channel_id;
-    delete vie_encoder;
-  }
-
-  LOG(LS_VERBOSE) << "Channel deleted " << channel_id;
-}
-
-void ChannelGroup::AddChannel(int channel_id) {
-  channels_.insert(channel_id);
-}
-
-void ChannelGroup::RemoveChannel(int channel_id) {
-  channels_.erase(channel_id);
-}
-
-bool ChannelGroup::HasChannel(int channel_id) const {
-  return channels_.find(channel_id) != channels_.end();
-}
-
-bool ChannelGroup::Empty() const {
-  return channels_.empty();
-}
-
-ViEChannel* ChannelGroup::GetChannel(int channel_id) const {
-  ChannelMap::const_iterator it = channel_map_.find(channel_id);
-  if (it == channel_map_.end()) {
-    LOG(LS_ERROR) << "Channel doesn't exist " << channel_id;
-    return NULL;
-  }
-  return it->second;
-}
-
-ViEEncoder* ChannelGroup::GetEncoder(int channel_id) const {
-  CriticalSectionScoped lock(encoder_map_cs_.get());
-  EncoderMap::const_iterator it = vie_encoder_map_.find(channel_id);
-  if (it == vie_encoder_map_.end()) {
-    return NULL;
-  }
-  return it->second;
-}
-
-ViEChannel* ChannelGroup::PopChannel(int channel_id) {
-  ChannelMap::iterator c_it = channel_map_.find(channel_id);
-  DCHECK(c_it != channel_map_.end());
-  ViEChannel* channel = c_it->second;
-  channel_map_.erase(c_it);
-
-  return channel;
-}
-
-ViEEncoder* ChannelGroup::PopEncoder(int channel_id) {
-  CriticalSectionScoped lock(encoder_map_cs_.get());
-  auto it = vie_encoder_map_.find(channel_id);
-  DCHECK(it != vie_encoder_map_.end());
-  ViEEncoder* encoder = it->second;
-  vie_encoder_map_.erase(it);
-
-  it = send_encoders_.find(channel_id);
-  if (it != send_encoders_.end())
-    send_encoders_.erase(it);
-
-  return encoder;
-}
-
-std::vector<int> ChannelGroup::GetChannelIds() const {
-  std::vector<int> ids;
-  for (auto channel : channel_map_)
-    ids.push_back(channel.first);
-  return ids;
-}
-
-bool ChannelGroup::OtherChannelsUsingEncoder(int channel_id) const {
-  CriticalSectionScoped lock(encoder_map_cs_.get());
-  EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
-  if (orig_it == vie_encoder_map_.end()) {
-    // No ViEEncoder for this channel.
-    return false;
-  }
-
-  // Loop through all other channels to see if anyone points at the same
-  // ViEEncoder.
-  for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
-       comp_it != vie_encoder_map_.end(); ++comp_it) {
-    // Make sure we're not comparing the same channel with itself.
-    if (comp_it->first != channel_id) {
-      if (comp_it->second == orig_it->second) {
-        return true;
-      }
-    }
-  }
-  return false;
-}
-
-void ChannelGroup::SetLoadManager(CPULoadStateCallbackInvoker* load_manager) {
-  for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
-       comp_it != vie_encoder_map_.end(); ++comp_it) {
-    comp_it->second->SetLoadManager(load_manager);
-  }
-}
-
-void ChannelGroup::SetSyncInterface(VoEVideoSync* sync_interface) {
-  for (auto channel : channel_map_) {
-    channel.second->SetVoiceChannel(-1, sync_interface);
-  }
-}
-
-void ChannelGroup::GetChannelsUsingEncoder(int channel_id,
-                                           ChannelList* channels) const {
-  CriticalSectionScoped lock(encoder_map_cs_.get());
-  EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
-
-  for (ChannelMap::const_iterator c_it = channel_map_.begin();
-       c_it != channel_map_.end(); ++c_it) {
-    EncoderMap::const_iterator comp_it = vie_encoder_map_.find(c_it->first);
-    DCHECK(comp_it != vie_encoder_map_.end());
-    if (comp_it->second == orig_it->second) {
-      channels->push_back(c_it->second);
-    }
-  }
-}
-
-BitrateController* ChannelGroup::GetBitrateController() const {
-  return bitrate_controller_.get();
-}
-
-RemoteBitrateEstimator* ChannelGroup::GetRemoteBitrateEstimator() const {
-  return remote_bitrate_estimator_.get();
-}
-
-CallStats* ChannelGroup::GetCallStats() const {
-  return call_stats_.get();
-}
-
-EncoderStateFeedback* ChannelGroup::GetEncoderStateFeedback() const {
-  return encoder_state_feedback_.get();
-}
-
-int64_t ChannelGroup::GetPacerQueuingDelayMs() const {
-  return pacer_->QueueInMs();
-}
-
-void ChannelGroup::SetChannelRembStatus(int channel_id,
-                                        bool sender,
-                                        bool receiver,
-                                        ViEChannel* channel) {
-  // Update the channel state.
-  channel->EnableRemb(sender || receiver);
-  // Update the REMB instance with necessary RTP modules.
-  RtpRtcp* rtp_module = channel->rtp_rtcp();
-  if (sender) {
-    remb_->AddRembSender(rtp_module);
-  } else {
-    remb_->RemoveRembSender(rtp_module);
-  }
-  if (receiver) {
-    remb_->AddReceiveChannel(rtp_module);
-  } else {
-    remb_->RemoveReceiveChannel(rtp_module);
-  }
-}
-
-void ChannelGroup::OnNetworkChanged(uint32_t target_bitrate_bps,
-                                    uint8_t fraction_loss,
-                                    int64_t rtt) {
-  bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt);
-  int pad_up_to_bitrate_bps = 0;
-  {
-    CriticalSectionScoped lock(encoder_map_cs_.get());
-    for (const auto& encoder : send_encoders_) {
-      pad_up_to_bitrate_bps +=
-          encoder.second->GetPaddingNeededBps(target_bitrate_bps);
-    }
-  }
-  pacer_->UpdateBitrate(
-      target_bitrate_bps / 1000,
-      PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000,
-      pad_up_to_bitrate_bps / 1000);
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h b/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h
deleted file mode 100644
index e3d06af080..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_GROUP_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_GROUP_H_
-
-#include <list>
-#include <map>
-#include <set>
-#include <vector>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-
-namespace webrtc {
-
-class BitrateAllocator;
-class CallStats;
-class Config;
-class EncoderStateFeedback;
-class PacedSender;
-class PacketRouter;
-class ProcessThread;
-class RemoteBitrateEstimator;
-class ViEChannel;
-class ViEEncoder;
-class VieRemb;
-class VoEVideoSync;
-class CPULoadStateCallbackInvoker;
-
-typedef std::list<ViEChannel*> ChannelList;
-
-// Channel group contains data common for several channels. All channels in the
-// group are assumed to send/receive data to the same end-point.
-class ChannelGroup : public BitrateObserver {
- public:
-  ChannelGroup(ProcessThread* process_thread, const Config* config);
-  ~ChannelGroup();
-  bool CreateSendChannel(int channel_id,
-                         int engine_id,
-                         int number_of_cores,
-                         bool disable_default_encoder);
-  bool CreateReceiveChannel(int channel_id,
-                            int engine_id,
-                            int base_channel_id,
-                            int number_of_cores,
-                            bool disable_default_encoder);
-  void Stop(int channel_id);
-  void DeleteChannel(int channel_id);
-  void AddChannel(int channel_id);
-  void RemoveChannel(int channel_id);
-  bool HasChannel(int channel_id) const;
-  bool Empty() const;
-  ViEChannel* GetChannel(int channel_id) const;
-  ViEEncoder* GetEncoder(int channel_id) const;
-  std::vector<int> GetChannelIds() const;
-  bool OtherChannelsUsingEncoder(int channel_id) const;
-  void SetLoadManager(CPULoadStateCallbackInvoker* load_manager);
-  void GetChannelsUsingEncoder(int channel_id, ChannelList* channels) const;
-
-  void SetSyncInterface(VoEVideoSync* sync_interface);
-
-  void SetChannelRembStatus(int channel_id,
-                            bool sender,
-                            bool receiver,
-                            ViEChannel* channel);
-
-  BitrateController* GetBitrateController() const;
-  CallStats* GetCallStats() const;
-  RemoteBitrateEstimator* GetRemoteBitrateEstimator() const;
-  EncoderStateFeedback* GetEncoderStateFeedback() const;
-  int64_t GetPacerQueuingDelayMs() const;
-
-  // Implements BitrateObserver.
-  void OnNetworkChanged(uint32_t target_bitrate_bps,
-                        uint8_t fraction_loss,
-                        int64_t rtt) override;
-
- private:
-  typedef std::map<int, ViEChannel*> ChannelMap;
-  typedef std::set<int> ChannelSet;
-  typedef std::map<int, ViEEncoder*> EncoderMap;
-
-  bool CreateChannel(int channel_id,
-                     int engine_id,
-                     int number_of_cores,
-                     ViEEncoder* vie_encoder,
-                     bool sender,
-                     bool disable_default_encoder);
-  ViEChannel* PopChannel(int channel_id);
-  ViEEncoder* PopEncoder(int channel_id);
-
-  rtc::scoped_ptr<VieRemb> remb_;
-  rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
-  rtc::scoped_ptr<CallStats> call_stats_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
-  rtc::scoped_ptr<EncoderStateFeedback> encoder_state_feedback_;
-  rtc::scoped_ptr<PacketRouter> packet_router_;
-  rtc::scoped_ptr<PacedSender> pacer_;
-  ChannelSet channels_;
-  ChannelMap channel_map_;
-  // Maps Channel id -> ViEEncoder.
-  EncoderMap vie_encoder_map_;
-  EncoderMap send_encoders_;
-  rtc::scoped_ptr<CriticalSectionWrapper> encoder_map_cs_;
-
-  const Config* config_;
-  // Placeholder for the case where this owns the config.
-  rtc::scoped_ptr<Config> own_config_;
-
-  // Registered at construct time and assumed to outlive this class.
-  ProcessThread* process_thread_;
-  rtc::scoped_ptr<ProcessThread> pacer_thread_;
-
-  rtc::scoped_ptr<BitrateController> bitrate_controller_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_GROUP_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc b/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc
deleted file mode 100644
index 1528c1baa6..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc
+++ /dev/null
@@ -1,435 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_channel_manager.h"
-
-#include <vector>
-
-#include "webrtc/common.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/call_stats.h"
-#include "webrtc/video_engine/encoder_state_feedback.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_group.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_remb.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-
-namespace webrtc {
-
-ViEChannelManager::ViEChannelManager(int engine_id,
-                                     int number_of_cores,
-                                     const Config& config)
-    : channel_id_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
-      engine_id_(engine_id),
-      number_of_cores_(number_of_cores),
-      free_channel_ids_(new bool[kViEMaxNumberOfChannels]),
-      free_channel_ids_size_(kViEMaxNumberOfChannels),
-      voice_sync_interface_(NULL),
-      module_process_thread_(NULL),
-      load_manager_(NULL)
-{
-  for (int idx = 0; idx < free_channel_ids_size_; idx++) {
-    free_channel_ids_[idx] = true;
-  }
-}
-
-ViEChannelManager::~ViEChannelManager() {
-  while (!channel_groups_.empty()) {
-    // The channel group is deleted by DeleteChannel when all its channels have
-    // been deleted.
-    for (int channel_id : channel_groups_.front()->GetChannelIds()) {
-      DeleteChannel(channel_id);
-    }
-  }
-
-  if (voice_sync_interface_) {
-    voice_sync_interface_->Release();
-  }
-  if (channel_id_critsect_) {
-    delete channel_id_critsect_;
-    channel_id_critsect_ = NULL;
-  }
-  if (free_channel_ids_) {
-    delete[] free_channel_ids_;
-    free_channel_ids_ = NULL;
-    free_channel_ids_size_ = 0;
-  }
-  assert(channel_groups_.empty());
-}
-
-void ViEChannelManager::SetModuleProcessThread(
-    ProcessThread* module_process_thread) {
-  assert(!module_process_thread_);
-  module_process_thread_ = module_process_thread;
-}
-
-void ViEChannelManager::SetLoadManager(
-    CPULoadStateCallbackInvoker* load_manager) {
-  load_manager_ = load_manager;
-  for (ChannelGroups::const_iterator it = channel_groups_.begin();
-       it != channel_groups_.end(); ++it) {
-    (*it)->SetLoadManager(load_manager);
-  }
-}
-
-int ViEChannelManager::CreateChannel(int* channel_id,
-                                     const Config* channel_group_config) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-
-  // Get a new channel id.
-  int new_channel_id = FreeChannelId();
-  if (new_channel_id == -1) {
-    return -1;
-  }
-
-  // Create a new channel group and add this channel.
-  rtc::scoped_ptr<ChannelGroup> group(
-      new ChannelGroup(module_process_thread_, channel_group_config));
-
-  if (!group->CreateSendChannel(new_channel_id, engine_id_, number_of_cores_,
-                                false)) {
-    ReturnChannelId(new_channel_id);
-    return -1;
-  }
-
-  *channel_id = new_channel_id;
-  group->AddChannel(*channel_id);
-  channel_groups_.push_back(group.release());
-  return 0;
-}
-
-int ViEChannelManager::CreateChannel(int* channel_id,
-                                     int original_channel,
-                                     bool sender,
-                                     bool disable_default_encoder) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-
-  ChannelGroup* channel_group = FindGroup(original_channel);
-  if (!channel_group) {
-    return -1;
-  }
-  int new_channel_id = FreeChannelId();
-  if (new_channel_id == -1) {
-    return -1;
-  }
-  if (sender) {
-    if (!channel_group->CreateSendChannel(new_channel_id, engine_id_,
-                                          number_of_cores_,
-                                          disable_default_encoder)) {
-      ReturnChannelId(new_channel_id);
-      return -1;
-    }
-  } else {
-    if (!channel_group->CreateReceiveChannel(new_channel_id, engine_id_,
-                                             original_channel, number_of_cores_,
-                                             disable_default_encoder)) {
-      ReturnChannelId(new_channel_id);
-      return -1;
-    }
-  }
-  *channel_id = new_channel_id;
-  channel_group->AddChannel(*channel_id);
-  return 0;
-}
-
-int ViEChannelManager::DeleteChannel(int channel_id) {
-  ChannelGroup* group = NULL;
-  {
-    // Read lock to make sure no one tries to delete it on us
-    ViEChannelManagerScoped(*this);
-
-    // Protect the maps.
-    CriticalSectionScoped cs(channel_id_critsect_);
-
-    group = FindGroup(channel_id);
-    if (group == NULL)
-      return -1;
-    group->Stop(channel_id);
-  }
-  // Now release the read lock, and get a write lock since we know the
-  // threads are stopped.  Otherwise we can deadlock if we hold a write
-  // lock and the thread is processing and needs a read lock, and we try to
-  // stop the thread (ProcessThread::DeRegisterModule().  See Mozilla
-  // bug 1276156.
-  {
-    // Write lock to make sure no one is using the channel.
-    ViEManagerWriteScoped wl(this);
-
-    // Protect the maps.
-    CriticalSectionScoped cs(channel_id_critsect_);
-
-    group = FindGroup(channel_id);
-    if (group == NULL)
-      return -1;
-    ReturnChannelId(channel_id);
-    group->DeleteChannel(channel_id);
-
-    if (group->Empty()) {
-      channel_groups_.remove(group);
-    } else {
-      group = NULL;  // Prevent group from being deleted.
-    }
-  }
-  // If statment just to show that this object is not always deleted.
-  if (group) {
-    // Delete the group if empty last since the encoder holds a pointer to the
-    // BitrateController object that the group owns.
-    LOG(LS_VERBOSE) << "Channel group deleted for channel " << channel_id;
-    delete group;
-  }
-  return 0;
-}
-
-int ViEChannelManager::SetVoiceEngine(VoiceEngine* voice_engine) {
-  // Write lock to make sure no one is using the channel.
-  ViEManagerWriteScoped wl(this);
-
-  CriticalSectionScoped cs(channel_id_critsect_);
-
-  VoEVideoSync* sync_interface = NULL;
-  if (voice_engine) {
-    // Get new sync interface.
-    sync_interface = VoEVideoSync::GetInterface(voice_engine);
-    if (!sync_interface) {
-      return -1;
-    }
-  }
-
-  for (ChannelGroup* group : channel_groups_) {
-    group->SetSyncInterface(sync_interface);
-  }
-  if (voice_sync_interface_) {
-    voice_sync_interface_->Release();
-  }
-  voice_sync_interface_ = sync_interface;
-  return 0;
-}
-
-int ViEChannelManager::ConnectVoiceChannel(int channel_id,
-                                           int audio_channel_id) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  if (!voice_sync_interface_) {
-    LOG_F(LS_ERROR) << "No VoE set.";
-    return -1;
-  }
-  ViEChannel* channel = ViEChannelPtr(channel_id);
-  if (!channel) {
-    return -1;
-  }
-  return channel->SetVoiceChannel(audio_channel_id, voice_sync_interface_);
-}
-
-int ViEChannelManager::DisconnectVoiceChannel(int channel_id) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ViEChannel* channel = ViEChannelPtr(channel_id);
-  if (channel) {
-    channel->SetVoiceChannel(-1, NULL);
-    return 0;
-  }
-  return -1;
-}
-
-bool ViEChannelManager::SetRembStatus(int channel_id, bool sender,
-                                      bool receiver) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group) {
-    return false;
-  }
-  ViEChannel* channel = ViEChannelPtr(channel_id);
-  assert(channel);
-
-  group->SetChannelRembStatus(channel_id, sender, receiver, channel);
-  return true;
-}
-
-bool ViEChannelManager::SetReservedTransmitBitrate(
-    int channel_id, uint32_t reserved_transmit_bitrate_bps) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group) {
-    return false;
-  }
-
-  BitrateController* bitrate_controller = group->GetBitrateController();
-  bitrate_controller->SetReservedBitrate(reserved_transmit_bitrate_bps);
-  return true;
-}
-
-void ViEChannelManager::UpdateSsrcs(int channel_id,
-                                    const std::list<unsigned int>& ssrcs) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* channel_group = FindGroup(channel_id);
-  if (channel_group == NULL) {
-    return;
-  }
-  ViEEncoder* encoder = ViEEncoderPtr(channel_id);
-  assert(encoder);
-
-  EncoderStateFeedback* encoder_state_feedback =
-      channel_group->GetEncoderStateFeedback();
-  // Remove a possible previous setting for this encoder before adding the new
-  // setting.
-  encoder_state_feedback->RemoveEncoder(encoder);
-  for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
-       it != ssrcs.end(); ++it) {
-    encoder_state_feedback->AddEncoder(*it, encoder);
-  }
-}
-
-bool ViEChannelManager::GetEstimatedSendBandwidth(
-    int channel_id, uint32_t* estimated_bandwidth) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group) {
-    return false;
-  }
-  group->GetBitrateController()->AvailableBandwidth(estimated_bandwidth);
-  return true;
-}
-
-bool ViEChannelManager::GetEstimatedReceiveBandwidth(
-    int channel_id, uint32_t* estimated_bandwidth) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group) {
-    return false;
-  }
-  std::vector<unsigned int> ssrcs;
-  if (!group->GetRemoteBitrateEstimator()->LatestEstimate(
-      &ssrcs, estimated_bandwidth) || ssrcs.empty()) {
-    *estimated_bandwidth = 0;
-  }
-  return true;
-}
-
-bool ViEChannelManager::GetPacerQueuingDelayMs(int channel_id,
-                                               int64_t* delay_ms) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group)
-    return false;
-  *delay_ms = group->GetPacerQueuingDelayMs();
-  return true;
-}
-
-bool ViEChannelManager::SetBitrateConfig(int channel_id,
-                                         int min_bitrate_bps,
-                                         int start_bitrate_bps,
-                                         int max_bitrate_bps) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group)
-    return false;
-  BitrateController* bitrate_controller = group->GetBitrateController();
-  if (start_bitrate_bps > 0)
-    bitrate_controller->SetStartBitrate(start_bitrate_bps);
-  bitrate_controller->SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
-  return true;
-}
-
-ViEChannel* ViEChannelManager::ViEChannelPtr(int channel_id) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (group == NULL)
-    return NULL;
-  return group->GetChannel(channel_id);
-}
-
-ViEEncoder* ViEChannelManager::ViEEncoderPtr(int video_channel_id) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(video_channel_id);
-  if (group == NULL) {
-    return NULL;
-  }
-  return group->GetEncoder(video_channel_id);
-}
-
-int ViEChannelManager::FreeChannelId() {
-  int idx = 0;
-  while (idx < free_channel_ids_size_) {
-    if (free_channel_ids_[idx] == true) {
-      // We've found a free id, allocate it and return.
-      free_channel_ids_[idx] = false;
-      return idx + kViEChannelIdBase;
-    }
-    idx++;
-  }
-  LOG(LS_ERROR) << "Max number of channels reached.";
-  return -1;
-}
-
-void ViEChannelManager::ReturnChannelId(int channel_id) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  assert(channel_id < kViEMaxNumberOfChannels + kViEChannelIdBase &&
-         channel_id >= kViEChannelIdBase);
-  free_channel_ids_[channel_id - kViEChannelIdBase] = true;
-}
-
-ChannelGroup* ViEChannelManager::FindGroup(int channel_id) const {
-  for (ChannelGroups::const_iterator it = channel_groups_.begin();
-       it != channel_groups_.end(); ++it) {
-    if ((*it)->HasChannel(channel_id)) {
-      return *it;
-    }
-  }
-  return NULL;
-}
-
-bool ViEChannelManager::ChannelUsingViEEncoder(int channel_id) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (group == NULL) {
-    return false;
-  }
-  return group->OtherChannelsUsingEncoder(channel_id);
-}
-
-void ViEChannelManager::ChannelsUsingViEEncoder(int channel_id,
-                                                ChannelList* channels) const {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (group == NULL)
-    return;
-  group->GetChannelsUsingEncoder(channel_id, channels);
-}
-
-ViEChannelManagerScoped::ViEChannelManagerScoped(
-    const ViEChannelManager& vie_channel_manager)
-    : ViEManagerScopedBase(vie_channel_manager) {
-}
-
-ViEChannel* ViEChannelManagerScoped::Channel(int vie_channel_id) const {
-  return static_cast<const ViEChannelManager*>(vie_manager_)->ViEChannelPtr(
-      vie_channel_id);
-}
-ViEEncoder* ViEChannelManagerScoped::Encoder(int vie_channel_id) const {
-  return static_cast<const ViEChannelManager*>(vie_manager_)->ViEEncoderPtr(
-      vie_channel_id);
-}
-
-bool ViEChannelManagerScoped::ChannelUsingViEEncoder(int channel_id) const {
-  return (static_cast<const ViEChannelManager*>(vie_manager_))->
-      ChannelUsingViEEncoder(channel_id);
-}
-
-void ViEChannelManagerScoped::ChannelsUsingViEEncoder(
-    int channel_id, ChannelList* channels) const {
-  (static_cast<const ViEChannelManager*>(vie_manager_))->
-      ChannelsUsingViEEncoder(channel_id, channels);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h b/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h
deleted file mode 100644
index f6eb9dab4c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h
+++ /dev/null
@@ -1,159 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
-
-#include <list>
-#include <map>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_manager_base.h"
-#include "webrtc/video_engine/vie_remb.h"
-
-namespace webrtc {
-
-class ChannelGroup;
-class Config;
-class CriticalSectionWrapper;
-class ProcessThread;
-class RtcpRttStats;
-class ViEChannel;
-class ViEEncoder;
-class VoEVideoSync;
-class VoiceEngine;
-
-typedef std::list<ChannelGroup*> ChannelGroups;
-typedef std::list<ViEChannel*> ChannelList;
-
-class ViEChannelManager: private ViEManagerBase {
-  friend class ViEChannelManagerScoped;
- public:
-  ViEChannelManager(int engine_id,
-                    int number_of_cores,
-                    const Config& config);
-  ~ViEChannelManager();
-
-  void SetModuleProcessThread(ProcessThread* module_process_thread);
-
-  void SetLoadManager(CPULoadStateCallbackInvoker* load_manager);
-
-  // Creates a new channel. 'channel_id' will be the id of the created channel.
-  int CreateChannel(int* channel_id,
-                    const Config* config);
-
-  // Creates a new channel grouped with |original_channel|. The new channel
-  // will get its own |ViEEncoder| if |sender| is set to true. It will be a
-  // receive only channel, without an own |ViEEncoder| if |sender| is false.
-  // Doesn't internally allocate an encoder if |disable_default_encoder|.
-  int CreateChannel(int* channel_id,
-                    int original_channel,
-                    bool sender,
-                    bool disable_default_encoder);
-
-  // Deletes a channel.
-  int DeleteChannel(int channel_id);
-
-  // Set the voice engine instance to be used by all video channels.
-  int SetVoiceEngine(VoiceEngine* voice_engine);
-
-  // Enables lip sync of the channel.
-  int ConnectVoiceChannel(int channel_id, int audio_channel_id);
-
-  // Disables lip sync of the channel.
-  int DisconnectVoiceChannel(int channel_id);
-
-  // Adds a channel to include when sending REMB.
-  bool SetRembStatus(int channel_id, bool sender, bool receiver);
-
-  bool SetReservedTransmitBitrate(int channel_id,
-                                  uint32_t reserved_transmit_bitrate_bps);
-
-  // Updates the SSRCs for a channel. If one of the SSRCs already is registered,
-  // it will simply be ignored and no error is returned.
-  void UpdateSsrcs(int channel_id, const std::list<unsigned int>& ssrcs);
-
-  bool GetEstimatedSendBandwidth(int channel_id,
-                                 uint32_t* estimated_bandwidth) const;
-  bool GetEstimatedReceiveBandwidth(int channel_id,
-                                    uint32_t* estimated_bandwidth) const;
-
-  bool GetPacerQueuingDelayMs(int channel_id, int64_t* delay_ms) const;
-
-  bool SetBitrateConfig(int channel_id,
-                        int min_bitrate_bps,
-                        int start_bitrate_bps,
-                        int max_bitrate_bps);
-
-  bool ReAllocateBitrates(int channel_id);
-
- private:
-  // Used by ViEChannelScoped, forcing a manager user to use scoped.
-  // Returns a pointer to the channel with id 'channel_id'.
-  ViEChannel* ViEChannelPtr(int channel_id) const;
-
-  // Methods used by ViECaptureScoped and ViEEncoderScoped.
-  // Gets the ViEEncoder used as input for video_channel_id
-  ViEEncoder* ViEEncoderPtr(int video_channel_id) const;
-
-  // Returns a free channel id, -1 if failing.
-  int FreeChannelId();
-
-  // Returns a previously allocated channel id.
-  void ReturnChannelId(int channel_id);
-
-  // Returns the iterator to the ChannelGroup containing |channel_id|.
-  ChannelGroup* FindGroup(int channel_id) const;
-
-  // Returns true if at least one other channels uses the same ViEEncoder as
-  // channel_id.
-  bool ChannelUsingViEEncoder(int channel_id) const;
-  void ChannelsUsingViEEncoder(int channel_id, ChannelList* channels) const;
-
-  // Protects channel_map_ and free_channel_ids_.
-  CriticalSectionWrapper* channel_id_critsect_;
-  int engine_id_;
-  int number_of_cores_;
-
-  bool* free_channel_ids_;
-  int free_channel_ids_size_;
-
-  // List with all channel groups.
-  std::list<ChannelGroup*> channel_groups_;
-
-  // TODO(mflodman) Make part of channel group.
-  VoEVideoSync* voice_sync_interface_;
-
-  ProcessThread* module_process_thread_;
-  CPULoadStateCallbackInvoker* load_manager_;
-};
-
-class ViEChannelManagerScoped: private ViEManagerScopedBase {
- public:
-  explicit ViEChannelManagerScoped(
-      const ViEChannelManager& vie_channel_manager);
-  ViEChannel* Channel(int vie_channel_id) const;
-  ViEEncoder* Encoder(int vie_channel_id) const;
-
-  // Returns true if at least one other channels uses the same ViEEncoder as
-  // channel_id.
-  bool ChannelUsingViEEncoder(int channel_id) const;
-
-  // Returns a list with pointers to all channels using the same encoder as the
-  // channel with |channel_id|, including the one with the specified id.
-  void ChannelsUsingViEEncoder(int channel_id, ChannelList* channels) const;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc
deleted file mode 100644
index 002aa97a01..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc
+++ /dev/null
@@ -1,695 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_codec_impl.h"
-
-#include <list>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_capturer.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_input_manager.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-static void LogCodec(const VideoCodec& codec) {
-  LOG(LS_INFO) << "CodecType " << codec.codecType
-               << ", pl_type " << static_cast<int>(codec.plType)
-               << ", resolution " << codec.width
-               << " x " << codec.height
-               << ", start br " << codec.startBitrate
-               << ", min br " << codec.minBitrate
-               << ", max br " << codec.maxBitrate
-               << ", max fps " << static_cast<int>(codec.maxFramerate)
-               << ", max qp " << codec.qpMax
-               << ", number of streams "
-               << static_cast<int>(codec.numberOfSimulcastStreams);
-  if (codec.codecType == kVideoCodecVP8) {
-    LOG(LS_INFO) << "VP8 specific settings";
-    LOG(LS_INFO) << "pictureLossIndicationOn "
-                 << codec.codecSpecific.VP8.pictureLossIndicationOn
-                 << ", feedbackModeOn "
-                 << codec.codecSpecific.VP8.feedbackModeOn
-                 << ", complexity "
-                 << codec.codecSpecific.VP8.complexity
-                 << ", resilience "
-                 << codec.codecSpecific.VP8.resilience
-                 << ", numberOfTemporalLayers "
-                 << static_cast<int>(
-                     codec.codecSpecific.VP8.numberOfTemporalLayers)
-                 << ", keyFrameinterval "
-                 << codec.codecSpecific.VP8.keyFrameInterval;
-    for (int idx = 0; idx < codec.numberOfSimulcastStreams; ++idx) {
-      LOG(LS_INFO) << "Stream " << codec.simulcastStream[idx].width
-                   << " x " << codec.simulcastStream[idx].height;
-      LOG(LS_INFO) << "Temporal layers "
-                   << static_cast<int>(
-                       codec.simulcastStream[idx].numberOfTemporalLayers)
-                   << ", min br "
-                   << codec.simulcastStream[idx].minBitrate
-                   << ", target br "
-                   << codec.simulcastStream[idx].targetBitrate
-                   << ", max br "
-                   << codec.simulcastStream[idx].maxBitrate
-                   << ", qp max "
-                   << codec.simulcastStream[idx].qpMax;
-    }
-  } else if (codec.codecType == kVideoCodecH264) {
-    LOG(LS_INFO) << "H264 specific settings";
-    LOG(LS_INFO) << "profile: "
-                 <<  codec.codecSpecific.H264.profile
-                 << ", constraints: "
-                 << codec.codecSpecific.H264.constraints
-                 << ", level: "
-                 << codec.codecSpecific.H264.level/10.0
-                 << ", packetizationMode: "
-                 << codec.codecSpecific.H264.packetizationMode
-                 << ", framedropping: "
-                 << codec.codecSpecific.H264.frameDroppingOn
-                 << ", keyFrameInterval: "
-                 << codec.codecSpecific.H264.keyFrameInterval
-                 << ", spslen: "
-                 << codec.codecSpecific.H264.spsLen
-                 << ", ppslen: "
-                 << codec.codecSpecific.H264.ppsLen;
-  } else if (codec.codecType == kVideoCodecVP9) {
-    LOG(LS_INFO) << "VP9 specific settings";
-    // XXX FIX!! log VP9 specific settings
-  }
-}
-
-
-ViECodec* ViECodec::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViECodecImpl* vie_codec_impl = vie_impl;
-  // Increase ref count.
-  (*vie_codec_impl)++;
-  return vie_codec_impl;
-#else
-  return NULL;
-#endif
-}
-
-int ViECodecImpl::Release() {
-  LOG(LS_INFO) << "ViECodec::Release.";
-  // Decrease ref count.
-  (*this)--;
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_WARNING) << "ViECodec released too many times.";
-    shared_data_->SetLastError(kViEAPIDoesNotExist);
-    return -1;
-  }
-  return ref_count;
-}
-
-ViECodecImpl::ViECodecImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {
-}
-
-ViECodecImpl::~ViECodecImpl() {
-}
-
-int ViECodecImpl::NumberOfCodecs() const {
-  // +2 because of FEC(RED and ULPFEC)
-  return static_cast<int>((VideoCodingModule::NumberOfCodecs() + 2));
-}
-
-int ViECodecImpl::GetCodec(const unsigned char list_number,
-                           VideoCodec& video_codec) const {
-  if (list_number == VideoCodingModule::NumberOfCodecs()) {
-    memset(&video_codec, 0, sizeof(VideoCodec));
-    strcpy(video_codec.plName, "red");
-    video_codec.codecType = kVideoCodecRED;
-    video_codec.plType = VCM_RED_PAYLOAD_TYPE;
-  } else if (list_number == VideoCodingModule::NumberOfCodecs() + 1) {
-    memset(&video_codec, 0, sizeof(VideoCodec));
-    strcpy(video_codec.plName, "ulpfec");
-    video_codec.codecType = kVideoCodecULPFEC;
-    video_codec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
-  } else if (VideoCodingModule::Codec(list_number, &video_codec) != VCM_OK) {
-    shared_data_->SetLastError(kViECodecInvalidArgument);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::SetSendCodec(const int video_channel,
-                               const VideoCodec& video_codec) {
-  LOG(LS_INFO) << "SetSendCodec for channel " << video_channel;
-  LogCodec(video_codec);
-  if (!CodecValid(video_codec)) {
-    // Error logged.
-    shared_data_->SetLastError(kViECodecInvalidCodec);
-    return -1;
-  }
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder);
-  if (vie_encoder->Owner() != video_channel) {
-    LOG_F(LS_ERROR) << "Receive only channel.";
-    shared_data_->SetLastError(kViECodecReceiveOnlyChannel);
-    return -1;
-  }
-  // Set a max_bitrate if the user hasn't set one.
-  VideoCodec video_codec_internal;
-  memcpy(&video_codec_internal, &video_codec, sizeof(VideoCodec));
-  if (video_codec_internal.maxBitrate == 0) {
-    // Max is one bit per pixel.
-    video_codec_internal.maxBitrate = (video_codec_internal.width *
-                                       video_codec_internal.height *
-                                       video_codec_internal.maxFramerate)
-                                       / 1000;
-    LOG(LS_INFO) << "New max bitrate set " << video_codec_internal.maxBitrate;
-  }
-
-  if (video_codec_internal.startBitrate > 0) {
-    if (video_codec_internal.startBitrate < video_codec_internal.minBitrate) {
-      video_codec_internal.startBitrate = video_codec_internal.minBitrate;
-    }
-    if (video_codec_internal.startBitrate > video_codec_internal.maxBitrate) {
-      video_codec_internal.startBitrate = video_codec_internal.maxBitrate;
-    }
-  }
-
-  // Make sure to generate a new SSRC if the codec type and/or resolution has
-  // changed. This won't have any effect if the user has set an SSRC.
-  bool new_rtp_stream = true;
-  VideoCodec encoder;
-  if (vie_encoder->GetEncoder(&encoder) == 0)
-    new_rtp_stream = encoder.codecType != video_codec_internal.codecType;
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-
-  // Stop the media flow while reconfiguring.
-  vie_encoder->Pause();
-
-  if (vie_encoder->SetEncoder(video_codec_internal) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-
-  // Give the channel(s) the new information.
-  ChannelList channels;
-  cs.ChannelsUsingViEEncoder(video_channel, &channels);
-  for (ChannelList::iterator it = channels.begin(); it != channels.end();
-       ++it) {
-    bool ret = true;
-    if ((*it)->SetSendCodec(video_codec_internal, new_rtp_stream) != 0) {
-      ret = false;
-    }
-    if (!ret) {
-      shared_data_->SetLastError(kViECodecUnknownError);
-      return -1;
-    }
-  }
-
-  // TODO(mflodman) Break out this part in GetLocalSsrcList().
-  // Update all SSRCs to ViEEncoder.
-  std::list<unsigned int> ssrcs;
-  if (video_codec_internal.numberOfSimulcastStreams == 0) {
-    unsigned int ssrc = 0;
-    if (vie_channel->GetLocalSSRC(0, &ssrc) != 0) {
-      LOG_F(LS_ERROR) << "Could not get ssrc.";
-    }
-    ssrcs.push_back(ssrc);
-  } else {
-    for (int idx = 0; idx < video_codec_internal.numberOfSimulcastStreams;
-         ++idx) {
-      unsigned int ssrc = 0;
-      if (vie_channel->GetLocalSSRC(idx, &ssrc) != 0) {
-        LOG_F(LS_ERROR) << "Could not get ssrc for stream " << idx;
-      }
-      ssrcs.push_back(ssrc);
-    }
-  }
-  vie_encoder->SetSsrcs(ssrcs);
-  shared_data_->channel_manager()->UpdateSsrcs(video_channel, ssrcs);
-
-  // Update the protection mode, we might be switching NACK/FEC.
-  vie_encoder->UpdateProtectionMethod(vie_encoder->nack_enabled(),
-                                      vie_channel->IsSendingFecEnabled());
-
-  // Get new best format for frame provider.
-  ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
-  if (frame_provider) {
-    frame_provider->FrameCallbackChanged();
-  }
-  // Restart the media flow
-  if (new_rtp_stream) {
-    // Stream settings changed, make sure we get a key frame.
-    vie_encoder->SendKeyFrame();
-  }
-  vie_encoder->Restart();
-  return 0;
-}
-
-int ViECodecImpl::GetSendCodec(const int video_channel,
-                               VideoCodec& video_codec) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  return vie_encoder->GetEncoder(&video_codec);
-}
-
-int ViECodecImpl::SetReceiveCodec(const int video_channel,
-                                  const VideoCodec& video_codec) {
-  LOG(LS_INFO) << "SetReceiveCodec for channel " << video_channel;
-  LOG(LS_INFO) << "Codec type " << video_codec.codecType
-               << ", payload type " << static_cast<int>(video_codec.plType);
-
-  if (CodecValid(video_codec) == false) {
-    shared_data_->SetLastError(kViECodecInvalidCodec);
-    return -1;
-  }
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_channel->SetReceiveCodec(video_codec) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::GetReceiveCodec(const int video_channel,
-                                  VideoCodec& video_codec) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_channel->GetReceiveCodec(&video_codec) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::GetCodecConfigParameters(
-  const int video_channel,
-  unsigned char config_parameters[kConfigParameterSize],
-  unsigned char& config_parameters_size) const {
-  LOG(LS_INFO) << "GetCodecConfigParameters " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_encoder->GetCodecConfigParameters(config_parameters,
-                                            config_parameters_size) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::SetImageScaleStatus(const int video_channel,
-                                      const bool enable) {
-  LOG(LS_INFO) << "SetImageScaleStates for channel " << video_channel
-               << ", enable: " << enable;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_encoder->ScaleInputImage(enable) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::GetSendCodecStatistics(const int video_channel,
-                                         unsigned int& key_frames,
-                                         unsigned int& delta_frames) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_encoder->SendCodecStatistics(&key_frames, &delta_frames) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::GetReceiveCodecStatistics(const int video_channel,
-                                            unsigned int& key_frames,
-                                            unsigned int& delta_frames) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->ReceiveCodecStatistics(&key_frames, &delta_frames) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::GetReceiveSideDelay(const int video_channel,
-                                      int* delay_ms) const {
-  assert(delay_ms != NULL);
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  *delay_ms = vie_channel->ReceiveDelay();
-  if (*delay_ms < 0) {
-    return -1;
-  }
-  return 0;
-}
-
-uint32_t ViECodecImpl::GetLastObservedBitrateBps(int video_channel) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder != nullptr);
-  return vie_encoder->LastObservedBitrateBps();
-}
-
-int ViECodecImpl::GetCodecTargetBitrate(const int video_channel,
-                                        unsigned int* bitrate) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  return vie_encoder->CodecTargetBitrate(static_cast<uint32_t*>(bitrate));
-}
-
-int ViECodecImpl::GetNumDiscardedPackets(int video_channel) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  return static_cast<int>(vie_channel->DiscardedPackets());
-}
-
-int ViECodecImpl::SetKeyFrameRequestCallbackStatus(const int video_channel,
-                                                   const bool enable) {
-  LOG(LS_INFO) << "SetKeyFrameRequestCallbackStatus for " << video_channel
-               << ", enable " << enable;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->EnableKeyFrameRequestCallback(enable) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::SetSignalKeyPacketLossStatus(const int video_channel,
-                                               const bool enable,
-                                               const bool only_key_frames) {
-  LOG(LS_INFO) << "SetSignalKeyPacketLossStatus for " << video_channel
-               << "enable, " << enable
-               << ", only key frames " << only_key_frames;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetSignalPacketLossStatus(enable, only_key_frames) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::RegisterEncoderObserver(const int video_channel,
-                                          ViEEncoderObserver& observer) {
-  LOG(LS_INFO) << "RegisterEncoderObserver for channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_encoder->RegisterCodecObserver(&observer) != 0) {
-    shared_data_->SetLastError(kViECodecObserverAlreadyRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::DeregisterEncoderObserver(const int video_channel) {
-  LOG(LS_INFO) << "DeregisterEncoderObserver for channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_encoder->RegisterCodecObserver(NULL) != 0) {
-    shared_data_->SetLastError(kViECodecObserverNotRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::RegisterDecoderObserver(const int video_channel,
-                                          ViEDecoderObserver& observer) {
-  LOG(LS_INFO) << "RegisterDecoderObserver for channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterCodecObserver(&observer) != 0) {
-    shared_data_->SetLastError(kViECodecObserverAlreadyRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::DeregisterDecoderObserver(const int video_channel) {
-  LOG(LS_INFO) << "DeregisterDecodeObserver for channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterCodecObserver(NULL) != 0) {
-    shared_data_->SetLastError(kViECodecObserverNotRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::SendKeyFrame(const int video_channel) {
-  LOG(LS_INFO) << "SendKeyFrame on channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_encoder->SendKeyFrame() != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::WaitForFirstKeyFrame(const int video_channel,
-                                       const bool wait) {
-  LOG(LS_INFO) << "WaitForFirstKeyFrame for channel " << video_channel
-               << ", wait " << wait;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->WaitForKeyFrame(wait) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViECodecImpl::StartDebugRecording(int video_channel,
-                                      const char* file_name_utf8) {
-  LOG(LS_INFO) << "StartDebugRecording for channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    return -1;
-  }
-  return vie_encoder->StartDebugRecording(file_name_utf8);
-}
-
-int ViECodecImpl::StopDebugRecording(int video_channel) {
-  LOG(LS_INFO) << "StopDebugRecording for channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    return -1;
-  }
-  return vie_encoder->StopDebugRecording();
-}
-
-void ViECodecImpl::SuspendBelowMinBitrate(int video_channel) {
-  LOG(LS_INFO) << "SuspendBelowMinBitrate for channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    return;
-  }
-  vie_encoder->SuspendBelowMinBitrate();
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    return;
-  }
-  // Must enable pacing when enabling SuspendBelowMinBitrate. Otherwise, no
-  // padding will be sent when the video is suspended so the video will be
-  // unable to recover.
-  vie_channel->SetTransmissionSmoothingStatus(true);
-}
-
-bool ViECodecImpl::GetSendSideDelay(int video_channel, int* avg_delay_ms,
-                                   int* max_delay_ms) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return false;
-  }
-  return vie_channel->GetSendSideDelay(avg_delay_ms, max_delay_ms);
-}
-
-bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
-  // Check pl_name matches codec_type.
-  if (video_codec.codecType == kVideoCodecRED) {
-#if defined(WIN32)
-    if (_strnicmp(video_codec.plName, "red", 3) == 0) {
-#else
-    if (strncasecmp(video_codec.plName, "red", 3) == 0) {
-#endif
-      // We only care about the type and name for red.
-      return true;
-    }
-    LOG_F(LS_ERROR) << "Invalid RED configuration.";
-    return false;
-  } else if (video_codec.codecType == kVideoCodecULPFEC) {
-#if defined(WIN32)
-    if (_strnicmp(video_codec.plName, "ULPFEC", 6) == 0) {
-#else
-    if (strncasecmp(video_codec.plName, "ULPFEC", 6) == 0) {
-#endif
-      // We only care about the type and name for ULPFEC.
-      return true;
-    }
-    LOG_F(LS_ERROR) << "Invalid ULPFEC configuration.";
-    return false;
-  } else if ((video_codec.codecType == kVideoCodecVP8 &&
-              strncmp(video_codec.plName, "VP8", 4) == 0) ||
-             (video_codec.codecType == kVideoCodecVP9 &&
-              strncmp(video_codec.plName, "VP9", 4) == 0) ||
-             (video_codec.codecType == kVideoCodecI420 &&
-              strncmp(video_codec.plName, "I420", 4) == 0) ||
-             (video_codec.codecType == kVideoCodecH264 &&
-              strncmp(video_codec.plName, "H264", 4) == 0)) {
-    // OK.
-  } else if (video_codec.codecType != kVideoCodecGeneric) {
-    LOG(LS_ERROR) << "Codec type and name mismatch.";
-    return false;
-  }
-
-  if (video_codec.plType == 0 || video_codec.plType > 127) {
-    LOG(LS_ERROR) << "Invalid payload type: "
-                  << static_cast<int>(video_codec.plType);
-    return false;
-  }
-
-  if (video_codec.width > kViEMaxCodecWidth ||
-      video_codec.height > kViEMaxCodecHeight) {
-    LOG(LS_ERROR) << "Invalid codec resolution " << video_codec.width
-                  << " x " << video_codec.height;
-    return false;
-  }
-
-  if (video_codec.startBitrate > 0 &&
-      video_codec.startBitrate < kViEMinCodecBitrate) {
-    LOG(LS_ERROR) << "Invalid start bitrate.";
-    return false;
-  }
-  if (video_codec.minBitrate < kViEMinCodecBitrate) {
-    LOG(LS_ERROR) << "Invalid min bitrate.";
-    return false;
-  }
-  return true;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h
deleted file mode 100644
index 51537b4685..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
-
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViECodecImpl
-    : public ViECodec,
-      public ViERefCount {
- public:
-  virtual int Release();
-
-  // Implements ViECodec.
-  virtual int NumberOfCodecs() const;
-  virtual int GetCodec(const unsigned char list_number,
-                       VideoCodec& video_codec) const;
-  virtual int SetSendCodec(const int video_channel,
-                           const VideoCodec& video_codec);
-  virtual int GetSendCodec(const int video_channel,
-                           VideoCodec& video_codec) const;
-  virtual int SetReceiveCodec(const int video_channel,
-                              const VideoCodec& video_codec);
-  virtual int GetReceiveCodec(const int video_channel,
-                              VideoCodec& video_codec) const;
-  virtual int GetCodecConfigParameters(
-    const int video_channel,
-    unsigned char config_parameters[kConfigParameterSize],
-    unsigned char& config_parameters_size) const;
-  virtual int SetImageScaleStatus(const int video_channel, const bool enable);
-  virtual int GetSendCodecStatistics(const int video_channel,
-                                     unsigned int& key_frames,
-                                     unsigned int& delta_frames) const;
-  virtual int GetReceiveCodecStatistics(const int video_channel,
-                                        unsigned int& key_frames,
-                                        unsigned int& delta_frames) const;
-  virtual int GetReceiveSideDelay(const int video_channel,
-                                  int* delay_ms) const;
-  uint32_t GetLastObservedBitrateBps(int video_channel) const override;
-  virtual int GetCodecTargetBitrate(const int video_channel,
-                                    unsigned int* bitrate) const;
-  virtual int GetNumDiscardedPackets(int video_channel) const;
-  virtual int SetKeyFrameRequestCallbackStatus(const int video_channel,
-                                               const bool enable);
-  virtual int SetSignalKeyPacketLossStatus(const int video_channel,
-                                           const bool enable,
-                                           const bool only_key_frames = false);
-  virtual int RegisterEncoderObserver(const int video_channel,
-                                      ViEEncoderObserver& observer);
-  virtual int DeregisterEncoderObserver(const int video_channel);
-  virtual int RegisterDecoderObserver(const int video_channel,
-                                      ViEDecoderObserver& observer);
-  virtual int DeregisterDecoderObserver(const int video_channel);
-  virtual int SendKeyFrame(const int video_channel);
-  virtual int WaitForFirstKeyFrame(const int video_channel, const bool wait);
-  virtual int StartDebugRecording(int video_channel,
-                                  const char* file_name_utf8);
-  virtual int StopDebugRecording(int video_channel);
-  virtual void SuspendBelowMinBitrate(int video_channel);
-  virtual bool GetSendSideDelay(int video_channel, int* avg_delay_ms,
-                                int* max_delay_ms) const;
-
- protected:
-  explicit ViECodecImpl(ViESharedData* shared_data);
-  virtual ~ViECodecImpl();
-
- private:
-  bool CodecValid(const VideoCodec& video_codec);
-
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_defines.h b/media/webrtc/trunk/webrtc/video_engine/vie_defines.h
deleted file mode 100644
index 62b970f4f2..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_defines.h
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_DEFINES_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_DEFINES_H_
-
-#include "webrtc/engine_configurations.h"
-
-// TODO(mflodman) Remove.
-#ifdef WEBRTC_ANDROID
-#include <arpa/inet.h>  // NOLINT
-#include <linux/net.h>  // NOLINT
-#include <netinet/in.h>  // NOLINT
-#include <pthread.h>  // NOLINT
-#include <stdio.h>  // NOLINT
-#include <stdlib.h>  // NOLINT
-#include <string.h>  // NOLINT
-#include <sys/socket.h>  // NOLINT
-#include <sys/time.h>  // NOLINT
-#include <sys/types.h>  // NOLINT
-#include <time.h>  // NOLINT
-#endif
-
-namespace webrtc {
-
-// General
-enum { kViEMinKeyRequestIntervalMs = 300 };
-
-// ViEBase
-enum { kViEMaxNumberOfChannels = 64 };
-
-// ViECapture
-enum { kViEMaxCaptureDevices = 256 };
-enum { kViECaptureDefaultWidth = 352 };
-enum { kViECaptureDefaultHeight = 288 };
-enum { kViECaptureDefaultFramerate = 30 };
-enum { kViEScreenCaptureDefaultFramerate = 3 };
-enum { kViECaptureMaxSnapshotWaitTimeMs = 500 };
-
-// ViECodec
-enum { kViEMaxCodecWidth = 4096 };
-enum { kViEMaxCodecHeight = 3072 };
-enum { kViEMaxCodecFramerate = 60 };
-enum { kViEMinCodecBitrate = 30 };
-
-// ViENetwork
-enum { kViEMaxMtu = 1500 };
-enum { kViESocketThreads = 1 };
-enum { kViENumReceiveSocketBuffers = 500 };
-
-// ViERender
-// Max valid time set in SetRenderTimeoutImage
-enum { kViEMaxRenderTimeoutTimeMs  = 10000 };
-// Min valid time set in SetRenderTimeoutImage
-enum { kViEMinRenderTimeoutTimeMs = 33 };
-enum { kViEDefaultRenderDelayMs = 10 };
-
-// ViERTP_RTCP
-enum { kSendSidePacketHistorySize = 600 };
-
-// NACK
-enum { kMaxPacketAgeToNack = 450 };  // In sequence numbers.
-enum { kMaxNackListSize = 250 };
-
-// Id definitions
-enum {
-  kViEChannelIdBase = 0x0,
-  kViEChannelIdMax = 0xFF,
-  kViECaptureIdBase = 0x1001,
-  kViECaptureIdMax = 0x10FF,
-  kViEDummyChannelId = 0xFFFF
-};
-
-// Module id
-// Create a unique id based on the ViE instance id and the
-// channel id. ViE id > 0 and 0 <= channel id <= 255
-
-inline int ViEId(const int vieId, const int channelId = -1) {
-  if (channelId == -1) {
-    return static_cast<int>((vieId << 16) + kViEDummyChannelId);
-  }
-  return static_cast<int>((vieId << 16) + channelId);
-}
-
-inline int ViEModuleId(const int vieId, const int channelId = -1) {
-  if (channelId == -1) {
-    return static_cast<int>((vieId << 16) + kViEDummyChannelId);
-  }
-  return static_cast<int>((vieId << 16) + channelId);
-}
-
-inline int ChannelId(const int moduleId) {
-  return static_cast<int>(moduleId & 0xffff);
-}
-
-// Windows specific.
-#if defined(_WIN32)
-  #define RENDER_MODULE_TYPE kRenderWindows
-
-  // Include libraries.
-  #pragma comment(lib, "winmm.lib")
-
-  #ifndef WEBRTC_EXTERNAL_TRANSPORT
-  #pragma comment(lib, "ws2_32.lib")
-  #pragma comment(lib, "Iphlpapi.lib")   // _GetAdaptersAddresses
-  #endif
-#endif
-
-// Mac specific.
-#ifdef WEBRTC_MAC
-  #define SLEEP(x) usleep(x * 1000)
-  #define RENDER_MODULE_TYPE kRenderWindows
-#endif
-
-// Android specific.
-#ifdef WEBRTC_ANDROID
-  #define FAR
-  #define __cdecl
-#endif  // WEBRTC_ANDROID
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_DEFINES_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc b/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc
deleted file mode 100644
index bc60a09bb5..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc
+++ /dev/null
@@ -1,998 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_encoder.h"
-
-#include <assert.h>
-
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/frame_callback.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/metrics.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
-#include "webrtc/video/send_statistics_proxy.h"
-#include "webrtc/video_engine/include/vie_codec.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/payload_router.h"
-#include "webrtc/video_engine/vie_defines.h"
-
-namespace webrtc {
-
-// Margin on when we pause the encoder when the pacing buffer overflows relative
-// to the configured buffer delay.
-static const float kEncoderPausePacerMargin = 2.0f;
-
-// Don't stop the encoder unless the delay is above this configured value.
-static const int kMinPacingDelayMs = 200;
-
-static const float kStopPaddingThresholdMs = 2000;
-
-std::vector<uint32_t> AllocateStreamBitrates(
-    uint32_t total_bitrate,
-    const SimulcastStream* stream_configs,
-    size_t number_of_streams) {
-  if (number_of_streams == 0) {
-    std::vector<uint32_t> stream_bitrates(1, 0);
-    stream_bitrates[0] = total_bitrate;
-    return stream_bitrates;
-  }
-  std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
-  uint32_t bitrate_remainder = total_bitrate;
-  for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
-    if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
-      stream_bitrates[i] = bitrate_remainder;
-    } else {
-      stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
-    }
-    bitrate_remainder -= stream_bitrates[i];
-  }
-  return stream_bitrates;
-}
-
-class QMVideoSettingsCallback : public VCMQMSettingsCallback {
- public:
-  explicit QMVideoSettingsCallback(VideoProcessingModule* vpm);
-
-  ~QMVideoSettingsCallback();
-
-  // Update VPM with QM (quality modes: frame size & frame rate) settings.
-  int32_t SetVideoQMSettings(const uint32_t frame_rate,
-                             const uint32_t width,
-                             const uint32_t height);
-
- private:
-  VideoProcessingModule* vpm_;
-};
-
-class ViEBitrateObserver : public BitrateObserver {
- public:
-  explicit ViEBitrateObserver(ViEEncoder* owner)
-      : owner_(owner) {
-  }
-  virtual ~ViEBitrateObserver() {}
-  // Implements BitrateObserver.
-  virtual void OnNetworkChanged(uint32_t bitrate_bps,
-                                uint8_t fraction_lost,
-                                int64_t rtt) {
-    owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
-  }
- private:
-  ViEEncoder* owner_;
-};
-
-class ViECPULoadStateObserver : public CPULoadStateObserver {
- public:
-  explicit ViECPULoadStateObserver(ViEEncoder* owner)
-      : owner_(owner) {
-  }
-  virtual ~ViECPULoadStateObserver() {};
-
-  // Implements CPULoadStateObserver.
-  virtual void onLoadStateChanged(CPULoadState state) {
-    owner_->onLoadStateChanged(state);
-  }
- private:
-  ViEEncoder* owner_;
-};
-
-ViEEncoder::ViEEncoder(int32_t channel_id,
-                       uint32_t number_of_cores,
-                       const Config& config,
-                       ProcessThread& module_process_thread,
-                       PacedSender* pacer,
-                       BitrateAllocator* bitrate_allocator,
-                       BitrateController* bitrate_controller,
-                       bool disable_default_encoder)
-    : channel_id_(channel_id),
-      number_of_cores_(number_of_cores),
-      disable_default_encoder_(disable_default_encoder),
-      vcm_(*webrtc::VideoCodingModule::Create(this)),
-      vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(-1, channel_id))),
-      send_payload_router_(NULL),
-      vcm_protection_callback_(NULL),
-      callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      pacer_(pacer),
-      bitrate_allocator_(bitrate_allocator),
-      bitrate_controller_(bitrate_controller),
-      time_of_last_incoming_frame_ms_(0),
-      load_manager_(NULL),
-      send_padding_(false),
-      min_transmit_bitrate_kbps_(0),
-      last_observed_bitrate_bps_(0),
-      target_delay_ms_(0),
-      network_is_transmitting_(true),
-      encoder_paused_(false),
-      encoder_paused_and_dropped_frame_(false),
-      fec_enabled_(false),
-      nack_enabled_(false),
-      codec_observer_(NULL),
-      effect_filter_(NULL),
-      module_process_thread_(module_process_thread),
-      has_received_sli_(false),
-      picture_id_sli_(0),
-      has_received_rpsi_(false),
-      picture_id_rpsi_(0),
-      qm_callback_(NULL),
-      video_suspended_(false),
-      pre_encode_callback_(NULL),
-      start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()),
-      send_statistics_proxy_(NULL) {
-  bitrate_observer_.reset(new ViEBitrateObserver(this));
-  loadstate_observer_.reset(new ViECPULoadStateObserver(this));
-}
-
-bool ViEEncoder::Init() {
-  if (vcm_.InitializeSender() != 0) {
-    return false;
-  }
-  vpm_.EnableTemporalDecimation(true);
-
-  // Enable content analysis if load management enabled
-  vpm_.EnableContentAnalysis(load_manager_ != NULL);
-
-  if (qm_callback_) {
-    delete qm_callback_;
-  }
-  qm_callback_ = new QMVideoSettingsCallback(&vpm_);
-
-  if (!disable_default_encoder_) {
-#ifdef VIDEOCODEC_VP8
-    VideoCodecType codec_type = webrtc::kVideoCodecVP8;
-#else
-    VideoCodecType codec_type = webrtc::kVideoCodecI420;
-#endif
-    VideoCodec video_codec;
-    if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
-      return false;
-    }
-    {
-      CriticalSectionScoped cs(data_cs_.get());
-      send_padding_ = video_codec.numberOfSimulcastStreams > 1;
-    }
-    if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
-                               PayloadRouter::DefaultMaxPayloadLength()) != 0) {
-      return false;
-    }
-  }
-  if (vcm_.RegisterTransportCallback(this) != 0) {
-    return false;
-  }
-  if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
-    return false;
-  }
-  if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
-    return false;
-  }
-  return true;
-}
-
-void ViEEncoder::StartThreadsAndSetSharedMembers(
-    scoped_refptr<PayloadRouter> send_payload_router,
-    VCMProtectionCallback* vcm_protection_callback) {
-  DCHECK(send_payload_router_ == NULL);
-  DCHECK(vcm_protection_callback_ == NULL);
-
-  send_payload_router_ = send_payload_router;
-  vcm_protection_callback_ = vcm_protection_callback;
-
-  module_process_thread_.RegisterModule(&vcm_);
-}
-
-void ViEEncoder::StopThreadsAndRemoveSharedMembers() {
-  // Avoid trying to stop the threads/etc twice
-  if (vcm_protection_callback_) {
-    vcm_.RegisterProtectionCallback(NULL);
-    vcm_protection_callback_ = NULL;
-    module_process_thread_.DeRegisterModule(&vcm_);
-    module_process_thread_.DeRegisterModule(&vpm_);
-  }
-}
-
-void ViEEncoder::SetLoadManager(CPULoadStateCallbackInvoker* load_manager) {
-  load_manager_ = load_manager;
-  if (load_manager_) {
-      load_manager_->AddObserver(loadstate_observer_.get());
-  }
-  vpm_.EnableContentAnalysis(load_manager != NULL);
-}
-
-ViEEncoder::~ViEEncoder() {
-  UpdateHistograms();
-  if (bitrate_allocator_)
-    bitrate_allocator_->RemoveBitrateObserver(bitrate_observer_.get());
-  if (load_manager_) {
-    load_manager_->RemoveObserver(loadstate_observer_.get());
-  }
-  VideoCodingModule::Destroy(&vcm_);
-  VideoProcessingModule::Destroy(&vpm_);
-  delete qm_callback_;
-}
-
-void ViEEncoder::UpdateHistograms() {
-  int64_t elapsed_sec =
-      (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 1000;
-  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
-    return;
-  }
-  webrtc::VCMFrameCount frames;
-  if (vcm_.SentFrameCount(frames) != VCM_OK) {
-    return;
-  }
-  uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
-  if (total_frames > 0) {
-    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
-        static_cast<int>(
-            (frames.numKeyFrames * 1000.0f / total_frames) + 0.5f));
-  }
-}
-
-int ViEEncoder::Owner() const {
-  return channel_id_;
-}
-
-void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    network_is_transmitting_ = is_transmitting;
-  }
-  if (is_transmitting) {
-    pacer_->Resume();
-  } else {
-    pacer_->Pause();
-  }
-}
-
-void ViEEncoder::Pause() {
-  CriticalSectionScoped cs(data_cs_.get());
-  encoder_paused_ = true;
-}
-
-void ViEEncoder::Restart() {
-  CriticalSectionScoped cs(data_cs_.get());
-  encoder_paused_ = false;
-}
-
-uint8_t ViEEncoder::NumberOfCodecs() {
-  return vcm_.NumberOfCodecs();
-}
-
-int32_t ViEEncoder::GetCodec(uint8_t list_index, VideoCodec* video_codec) {
-  if (vcm_.Codec(list_index, video_codec) != 0) {
-    return -1;
-  }
-  return 0;
-}
-
-int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
-                                            uint8_t pl_type,
-                                            bool internal_source) {
-  if (encoder == NULL)
-    return -1;
-
-  if (vcm_.RegisterExternalEncoder(encoder, pl_type, internal_source) !=
-      VCM_OK) {
-    return -1;
-  }
-  return 0;
-}
-
-int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
-  DCHECK(send_payload_router_ != NULL);
-  webrtc::VideoCodec current_send_codec;
-  if (vcm_.SendCodec(&current_send_codec) == VCM_OK) {
-    uint32_t current_bitrate_bps = 0;
-    if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
-      LOG(LS_WARNING) << "Failed to get the current encoder target bitrate.";
-    }
-    current_send_codec.startBitrate = (current_bitrate_bps + 500) / 1000;
-  }
-
-  if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
-    return -1;
-  }
-
-  if (disable_default_encoder_)
-    return 0;
-
-  // If the external encoder is the current send codec, use vcm internal
-  // encoder.
-  if (current_send_codec.plType == pl_type) {
-    {
-      CriticalSectionScoped cs(data_cs_.get());
-      send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
-    }
-    // TODO(mflodman): Unfortunately the VideoCodec that VCM has cached a
-    // raw pointer to an |extra_options| that's long gone.  Clearing it here is
-    // a hack to prevent the following code from crashing.  This should be fixed
-    // for realz.  https://code.google.com/p/chromium/issues/detail?id=348222
-    current_send_codec.extra_options = NULL;
-    size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
-    if (vcm_.RegisterSendCodec(&current_send_codec, number_of_cores_,
-                               max_data_payload_length) != VCM_OK) {
-      LOG(LS_INFO) << "De-registered the currently used external encoder ("
-                   << static_cast<int>(pl_type) << ") and therefore tried to "
-                   << "register the corresponding internal encoder, but none "
-                   << "was supported.";
-    }
-  }
-  return 0;
-}
-
-int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
-  DCHECK(send_payload_router_ != NULL);
-  // Setting target width and height for VPM.
-  if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
-                               video_codec.maxFramerate) != VPM_OK) {
-    return -1;
-  }
-
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    send_padding_ = video_codec.numberOfSimulcastStreams > 1;
-  }
-
-  // Add a bitrate observer to the allocator and update the start, max and
-  // min bitrates of the bitrate controller as needed.
-  int allocated_bitrate_bps;
-  int new_bwe_candidate_bps = bitrate_allocator_->AddBitrateObserver(
-      bitrate_observer_.get(), video_codec.startBitrate * 1000,
-      video_codec.minBitrate * 1000, video_codec.maxBitrate * 1000,
-      &allocated_bitrate_bps);
-
-  // Only set the start/min/max bitrate of the bitrate controller if the start
-  // bitrate is greater than zero. The new API sets these via the channel group
-  // and passes a zero start bitrate to SetSendCodec.
-  // TODO(holmer): Remove this when the new API has been launched.
-  if (video_codec.startBitrate > 0) {
-    if (new_bwe_candidate_bps > 0) {
-      uint32_t current_bwe_bps = 0;
-      bitrate_controller_->AvailableBandwidth(&current_bwe_bps);
-      bitrate_controller_->SetStartBitrate(std::max(
-          static_cast<uint32_t>(new_bwe_candidate_bps), current_bwe_bps));
-    }
-
-    int new_bwe_min_bps = 0;
-    int new_bwe_max_bps = 0;
-    bitrate_allocator_->GetMinMaxBitrateSumBps(&new_bwe_min_bps,
-                                               &new_bwe_max_bps);
-    bitrate_controller_->SetMinMaxBitrate(new_bwe_min_bps, new_bwe_max_bps);
-  }
-
-  webrtc::VideoCodec modified_video_codec = video_codec;
-  modified_video_codec.startBitrate = allocated_bitrate_bps / 1000;
-
-  size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
-  if (vcm_.RegisterSendCodec(&modified_video_codec, number_of_cores_,
-                             max_data_payload_length) != VCM_OK) {
-    return -1;
-  }
-  return 0;
-}
-
-int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
-  *video_codec = vcm_.GetSendCodec();
-  return 0;
-}
-
-int32_t ViEEncoder::GetCodecConfigParameters(
-    unsigned char config_parameters[kConfigParameterSize],
-    unsigned char& config_parameters_size) {
-  int32_t num_parameters =
-      vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
-  if (num_parameters <= 0) {
-    config_parameters_size = 0;
-    return -1;
-  }
-  config_parameters_size = static_cast<unsigned char>(num_parameters);
-  return 0;
-}
-
-int32_t ViEEncoder::ScaleInputImage(bool enable) {
-  VideoFrameResampling resampling_mode = kFastRescaling;
-  // TODO(mflodman) What?
-  if (enable) {
-    // kInterpolation is currently not supported.
-    LOG_F(LS_ERROR) << "Not supported.";
-    return -1;
-  }
-  vpm_.SetInputFrameResampleMode(resampling_mode);
-
-  return 0;
-}
-
-int ViEEncoder::GetPaddingNeededBps(int bitrate_bps) const {
-  int64_t time_of_last_incoming_frame_ms;
-  int min_transmit_bitrate_bps;
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    bool send_padding =
-        send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
-    if (!send_padding)
-      return 0;
-    time_of_last_incoming_frame_ms = time_of_last_incoming_frame_ms_;
-    min_transmit_bitrate_bps = 1000 * min_transmit_bitrate_kbps_;
-  }
-
-  VideoCodec send_codec;
-  if (vcm_.SendCodec(&send_codec) != 0)
-    return 0;
-  SimulcastStream* stream_configs = send_codec.simulcastStream;
-  // Allocate the bandwidth between the streams.
-  std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
-      bitrate_bps, stream_configs, send_codec.numberOfSimulcastStreams);
-
-  bool video_is_suspended = vcm_.VideoSuspended();
-
-  // Find the max amount of padding we can allow ourselves to send at this
-  // point, based on which streams are currently active and what our current
-  // available bandwidth is.
-  int pad_up_to_bitrate_bps = 0;
-  if (send_codec.numberOfSimulcastStreams == 0) {
-    pad_up_to_bitrate_bps = send_codec.minBitrate * 1000;
-  } else {
-    pad_up_to_bitrate_bps =
-        stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate *
-        1000;
-    for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
-      pad_up_to_bitrate_bps += stream_configs[i].targetBitrate * 1000;
-    }
-  }
-
-  // Disable padding if only sending one stream and video isn't suspended and
-  // min-transmit bitrate isn't used (applied later).
-  if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
-    pad_up_to_bitrate_bps = 0;
-
-  // The amount of padding should decay to zero if no frames are being
-  // captured unless a min-transmit bitrate is used.
-  int64_t now_ms = TickTime::MillisecondTimestamp();
-  if (now_ms - time_of_last_incoming_frame_ms > kStopPaddingThresholdMs)
-    pad_up_to_bitrate_bps = 0;
-
-  // Pad up to min bitrate.
-  if (pad_up_to_bitrate_bps < min_transmit_bitrate_bps)
-    pad_up_to_bitrate_bps = min_transmit_bitrate_bps;
-
-  // Padding may never exceed bitrate estimate.
-  if (pad_up_to_bitrate_bps > bitrate_bps)
-    pad_up_to_bitrate_bps = bitrate_bps;
-
-  return pad_up_to_bitrate_bps;
-}
-
-bool ViEEncoder::EncoderPaused() const {
-  // Pause video if paused by caller or as long as the network is down or the
-  // pacer queue has grown too large in buffered mode.
-  if (encoder_paused_) {
-    return true;
-  }
-  if (target_delay_ms_ > 0) {
-    // Buffered mode.
-    // TODO(pwestin): Workaround until nack is configured as a time and not
-    // number of packets.
-    return pacer_->QueueInMs() >=
-           std::max(
-               static_cast<int>(target_delay_ms_ * kEncoderPausePacerMargin),
-               kMinPacingDelayMs);
-  }
-  if (pacer_->ExpectedQueueTimeMs() > PacedSender::kDefaultMaxQueueLengthMs) {
-    // Too much data in pacer queue, drop frame.
-    return true;
-  }
-  return !network_is_transmitting_;
-}
-
-void ViEEncoder::TraceFrameDropStart() {
-  // Start trace event only on the first frame after encoder is paused.
-  if (!encoder_paused_and_dropped_frame_) {
-    TRACE_EVENT_ASYNC_BEGIN0("webrtc", "EncoderPaused", this);
-  }
-  encoder_paused_and_dropped_frame_ = true;
-  return;
-}
-
-void ViEEncoder::TraceFrameDropEnd() {
-  // End trace event on first frame after encoder resumes, if frame was dropped.
-  if (encoder_paused_and_dropped_frame_) {
-    TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
-  }
-  encoder_paused_and_dropped_frame_ = false;
-}
-
-void ViEEncoder::DeliverFrame(int id,
-                              I420VideoFrame* video_frame,
-                              const std::vector<uint32_t>& csrcs) {
-  DCHECK(send_payload_router_ != NULL);
-  DCHECK(csrcs.empty());
-  if (!send_payload_router_->active()) {
-    // We've paused or we have no channels attached, don't waste resources on
-    // encoding.
-    return;
-  }
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    time_of_last_incoming_frame_ms_ = TickTime::MillisecondTimestamp();
-    if (EncoderPaused()) {
-      TraceFrameDropStart();
-      return;
-    }
-    TraceFrameDropEnd();
-  }
-
-  TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame->render_time_ms(),
-                          "Encode");
-  I420VideoFrame* decimated_frame = NULL;
-  // TODO(wuchengli): support texture frames.
-  if (video_frame->native_handle() == NULL) {
-    {
-      CriticalSectionScoped cs(callback_cs_.get());
-      if (effect_filter_) {
-        size_t length =
-            CalcBufferSize(kI420, video_frame->width(), video_frame->height());
-        rtc::scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
-        ExtractBuffer(*video_frame, length, video_buffer.get());
-        effect_filter_->Transform(length,
-                                  video_buffer.get(),
-                                  video_frame->ntp_time_ms(),
-                                  video_frame->timestamp(),
-                                  video_frame->width(),
-                                  video_frame->height());
-      }
-    }
-
-    // Pass frame via preprocessor.
-    const int ret = vpm_.PreprocessFrame(*video_frame, &decimated_frame);
-    if (ret == 1) {
-      // Drop this frame.
-      return;
-    }
-    if (ret != VPM_OK) {
-      return;
-    }
-  }
-  // If the frame was not resampled or scaled => use original.
-  if (decimated_frame == NULL)  {
-    decimated_frame = video_frame;
-  }
-
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    if (pre_encode_callback_)
-      pre_encode_callback_->FrameCallback(decimated_frame);
-  }
-
-  if (video_frame->native_handle() != NULL) {
-    // TODO(wuchengli): add texture support. http://crbug.com/362437
-    return;
-  }
-
-#ifdef VIDEOCODEC_VP8
-  if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
-    webrtc::CodecSpecificInfo codec_specific_info;
-    codec_specific_info.codecType = webrtc::kVideoCodecVP8;
-    {
-      CriticalSectionScoped cs(data_cs_.get());
-      codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
-          has_received_rpsi_;
-      codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
-          has_received_sli_;
-      codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
-          picture_id_rpsi_;
-      codec_specific_info.codecSpecific.VP8.pictureIdSLI  =
-          picture_id_sli_;
-      has_received_sli_ = false;
-      has_received_rpsi_ = false;
-    }
-
-    vcm_.AddVideoFrame(*decimated_frame, vpm_.ContentMetrics(),
-                       &codec_specific_info);
-    return;
-  }
-#endif
-  // XXX fix VP9 (bug 1138629)
-
-#ifdef MOZ_WEBRTC_OMX
-  // XXX effectively disable resolution changes until Bug 1067437 is resolved with new DSP code
-  if (qm_callback_ && vcm_.SendCodec() == webrtc::kVideoCodecH264) {
-    if (vcm_.RegisterVideoQMCallback(NULL) != 0) {
-      LOG_F(LS_ERROR) << "VCM::RegisterQMCallback(NULL) failure";
-      return;
-    }
-    delete qm_callback_;
-    qm_callback_ = NULL;
-  }
-#endif
-
-  vcm_.AddVideoFrame(*decimated_frame, vpm_.ContentMetrics());
-}
-
-void ViEEncoder::DelayChanged(int id, int frame_delay) {
-}
-
-int ViEEncoder::GetPreferedFrameSettings(int* width,
-                                         int* height,
-                                         int* frame_rate) {
-  webrtc::VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof(video_codec));
-  if (vcm_.SendCodec(&video_codec) != VCM_OK) {
-    return -1;
-  }
-
-  *width = video_codec.width;
-  *height = video_codec.height;
-  *frame_rate = video_codec.maxFramerate;
-  return 0;
-}
-
-int ViEEncoder::SendKeyFrame() {
-  return vcm_.IntraFrameRequest(0);
-}
-
-int32_t ViEEncoder::SendCodecStatistics(
-    uint32_t* num_key_frames, uint32_t* num_delta_frames) {
-  webrtc::VCMFrameCount sent_frames;
-  if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
-    return -1;
-  }
-  *num_key_frames = sent_frames.numKeyFrames;
-  *num_delta_frames = sent_frames.numDeltaFrames;
-  return 0;
-}
-
-uint32_t ViEEncoder::LastObservedBitrateBps() const {
-  CriticalSectionScoped cs(data_cs_.get());
-  return last_observed_bitrate_bps_;
-}
-
-int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
-  if (vcm_.Bitrate(bitrate) != 0)
-    return -1;
-  return 0;
-}
-
-int32_t ViEEncoder::UpdateProtectionMethod(bool nack, bool fec) {
-  DCHECK(send_payload_router_ != NULL);
-  DCHECK(vcm_protection_callback_ != NULL);
-
-  if (fec_enabled_ == fec && nack_enabled_ == nack) {
-    // No change needed, we're already in correct state.
-    return 0;
-  }
-  fec_enabled_ = fec;
-  nack_enabled_ = nack;
-
-  // Set Video Protection for VCM.
-  if (fec_enabled_ && nack_enabled_) {
-    vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
-  } else {
-    vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
-    vcm_.SetVideoProtection(webrtc::kProtectionNackSender, nack_enabled_);
-    vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
-  }
-
-  if (fec_enabled_ || nack_enabled_) {
-    vcm_.RegisterProtectionCallback(vcm_protection_callback_);
-    // The send codec must be registered to set correct MTU.
-    webrtc::VideoCodec codec;
-    if (vcm_.SendCodec(&codec) == 0) {
-      uint32_t current_bitrate_bps = 0;
-      if (vcm_.Bitrate(&current_bitrate_bps) != 0) {
-        LOG_F(LS_WARNING) <<
-            "Failed to get the current encoder target bitrate.";
-      }
-      // Convert to start bitrate in kbps.
-      codec.startBitrate = (current_bitrate_bps + 500) / 1000;
-      size_t max_payload_length = send_payload_router_->MaxPayloadLength();
-      if (vcm_.RegisterSendCodec(&codec, number_of_cores_,
-                                 max_payload_length) != 0) {
-        return -1;
-      }
-    }
-    return 0;
-  } else {
-    // FEC and NACK are disabled.
-    vcm_.RegisterProtectionCallback(NULL);
-  }
-  return 0;
-}
-
-void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    target_delay_ms_ = target_delay_ms;
-  }
-  if (target_delay_ms > 0) {
-    // Disable external frame-droppers.
-    vcm_.EnableFrameDropper(false);
-    vpm_.EnableTemporalDecimation(false);
-  } else {
-    // Real-time mode - enable frame droppers.
-    vpm_.EnableTemporalDecimation(true);
-    vcm_.EnableFrameDropper(true);
-  }
-}
-
-void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (send_statistics_proxy_ != nullptr)
-    send_statistics_proxy_->OnSetRates(bitrate_bps, framerate);
-}
-
-int32_t ViEEncoder::SendData(
-    const uint8_t payload_type,
-    const EncodedImage& encoded_image,
-    const webrtc::RTPFragmentationHeader& fragmentation_header,
-    const RTPVideoHeader* rtp_video_hdr) {
-  DCHECK(send_payload_router_ != NULL);
-
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    if (send_statistics_proxy_ != NULL)
-      send_statistics_proxy_->OnSendEncodedImage(encoded_image, rtp_video_hdr);
-  }
-
-  return send_payload_router_->RoutePayload(
-      VCMEncodedFrame::ConvertFrameType(encoded_image._frameType), payload_type,
-      encoded_image._timeStamp, encoded_image.capture_time_ms_,
-      encoded_image._buffer, encoded_image._length, &fragmentation_header,
-      rtp_video_hdr) ? 0 : -1;
-}
-
-int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
-                                   const uint32_t frame_rate) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (codec_observer_) {
-    codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
-  }
-  return 0;
-}
-
-int32_t ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (observer && codec_observer_) {
-    LOG_F(LS_ERROR) << "Observer already set.";
-    return -1;
-  }
-  codec_observer_ = observer;
-  return 0;
-}
-
-void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
-                               uint8_t picture_id) {
-  CriticalSectionScoped cs(data_cs_.get());
-  picture_id_sli_ = picture_id;
-  has_received_sli_ = true;
-}
-
-void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
-                                uint64_t picture_id) {
-  CriticalSectionScoped cs(data_cs_.get());
-  picture_id_rpsi_ = picture_id;
-  has_received_rpsi_ = true;
-}
-
-void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
-  // Key frame request from remote side, signal to VCM.
-  TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
-
-  int idx = 0;
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
-    if (stream_it == ssrc_streams_.end()) {
-      LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size "
-                        << ssrc_streams_.size();
-      return;
-    }
-    std::map<unsigned int, int64_t>::iterator time_it =
-        time_last_intra_request_ms_.find(ssrc);
-    if (time_it == time_last_intra_request_ms_.end()) {
-      time_last_intra_request_ms_[ssrc] = 0;
-    }
-
-    int64_t now = TickTime::MillisecondTimestamp();
-    if (time_last_intra_request_ms_[ssrc] + kViEMinKeyRequestIntervalMs > now) {
-      return;
-    }
-    time_last_intra_request_ms_[ssrc] = now;
-    idx = stream_it->second;
-  }
-  // Release the critsect before triggering key frame.
-  vcm_.IntraFrameRequest(idx);
-}
-
-void ViEEncoder::OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {
-  CriticalSectionScoped cs(data_cs_.get());
-  std::map<unsigned int, int>::iterator it = ssrc_streams_.find(old_ssrc);
-  if (it == ssrc_streams_.end()) {
-    return;
-  }
-
-  ssrc_streams_[new_ssrc] = it->second;
-  ssrc_streams_.erase(it);
-
-  std::map<unsigned int, int64_t>::iterator time_it =
-      time_last_intra_request_ms_.find(old_ssrc);
-  int64_t last_intra_request_ms = 0;
-  if (time_it != time_last_intra_request_ms_.end()) {
-    last_intra_request_ms = time_it->second;
-    time_last_intra_request_ms_.erase(time_it);
-  }
-  time_last_intra_request_ms_[new_ssrc] = last_intra_request_ms;
-}
-
-bool ViEEncoder::SetSsrcs(const std::list<unsigned int>& ssrcs) {
-  VideoCodec codec;
-  if (vcm_.SendCodec(&codec) != 0)
-    return false;
-
-  if (codec.numberOfSimulcastStreams > 0 &&
-      ssrcs.size() != codec.numberOfSimulcastStreams) {
-    return false;
-  }
-
-  CriticalSectionScoped cs(data_cs_.get());
-  ssrc_streams_.clear();
-  time_last_intra_request_ms_.clear();
-  int idx = 0;
-  for (std::list<unsigned int>::const_iterator it = ssrcs.begin();
-       it != ssrcs.end(); ++it, ++idx) {
-    unsigned int ssrc = *it;
-    ssrc_streams_[ssrc] = idx;
-  }
-  return true;
-}
-
-void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
-  assert(min_transmit_bitrate_kbps >= 0);
-  CriticalSectionScoped crit(data_cs_.get());
-  min_transmit_bitrate_kbps_ = min_transmit_bitrate_kbps;
-}
-
-// Called from ViEBitrateObserver.
-void ViEEncoder::OnNetworkChanged(uint32_t bitrate_bps,
-                                  uint8_t fraction_lost,
-                                  int64_t round_trip_time_ms) {
-  LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
-                  << " packet loss " << fraction_lost
-                  << " rtt " << round_trip_time_ms;
-  DCHECK(send_payload_router_ != NULL);
-  vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
-  bool video_is_suspended = vcm_.VideoSuspended();
-
-  VideoCodec send_codec;
-  if (vcm_.SendCodec(&send_codec) != 0) {
-    return;
-  }
-  SimulcastStream* stream_configs = send_codec.simulcastStream;
-  // Allocate the bandwidth between the streams.
-  std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
-      bitrate_bps, stream_configs, send_codec.numberOfSimulcastStreams);
-  send_payload_router_->SetTargetSendBitrates(stream_bitrates);
-
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    last_observed_bitrate_bps_ = bitrate_bps;
-    if (video_suspended_ == video_is_suspended)
-      return;
-    video_suspended_ = video_is_suspended;
-  }
-  // Video suspend-state changed, inform codec observer.
-  CriticalSectionScoped crit(callback_cs_.get());
-  if (codec_observer_) {
-    LOG(LS_INFO) << "Video suspended " << video_is_suspended
-                 << " for channel " << channel_id_;
-    codec_observer_->SuspendChange(channel_id_, video_is_suspended);
-  }
-}
-
-void ViEEncoder::onLoadStateChanged(CPULoadState load_state) {
-  LOG(LS_INFO) << "load state changed to " << load_state;
-  vcm_.SetCPULoadState(load_state);
-}
-
-int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (effect_filter != NULL && effect_filter_ != NULL) {
-    LOG_F(LS_ERROR) << "Filter already set.";
-    return -1;
-  }
-  effect_filter_ = effect_filter;
-  return 0;
-}
-
-int ViEEncoder::StartDebugRecording(const char* fileNameUTF8) {
-  return vcm_.StartDebugRecording(fileNameUTF8);
-}
-
-int ViEEncoder::StopDebugRecording() {
-  return vcm_.StopDebugRecording();
-}
-
-void ViEEncoder::SuspendBelowMinBitrate() {
-  vcm_.SuspendBelowMinBitrate();
-  bitrate_allocator_->EnforceMinBitrate(false);
-}
-
-void ViEEncoder::RegisterPreEncodeCallback(
-    I420FrameCallback* pre_encode_callback) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  pre_encode_callback_ = pre_encode_callback;
-}
-
-void ViEEncoder::DeRegisterPreEncodeCallback() {
-  CriticalSectionScoped cs(callback_cs_.get());
-  pre_encode_callback_ = NULL;
-}
-
-void ViEEncoder::RegisterPostEncodeImageCallback(
-      EncodedImageCallback* post_encode_callback) {
-  vcm_.RegisterPostEncodeImageCallback(post_encode_callback);
-}
-
-void ViEEncoder::DeRegisterPostEncodeImageCallback() {
-  vcm_.RegisterPostEncodeImageCallback(NULL);
-}
-
-void ViEEncoder::RegisterSendStatisticsProxy(
-    SendStatisticsProxy* send_statistics_proxy) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  send_statistics_proxy_ = send_statistics_proxy;
-}
-
-QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
-    : vpm_(vpm) {
-}
-
-QMVideoSettingsCallback::~QMVideoSettingsCallback() {
-}
-
-int32_t QMVideoSettingsCallback::SetVideoQMSettings(
-    const uint32_t frame_rate,
-    const uint32_t width,
-    const uint32_t height) {
-  return vpm_->SetTargetResolution(width, height, frame_rate);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc
deleted file mode 100644
index 1be2d31618..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc
+++ /dev/null
@@ -1,145 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_external_codec_impl.h"
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViEExternalCodec* ViEExternalCodec::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-  if (video_engine == NULL) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViEExternalCodecImpl* vie_external_codec_impl = vie_impl;
-  // Increase ref count.
-  (*vie_external_codec_impl)++;
-  return vie_external_codec_impl;
-#else
-  return NULL;
-#endif
-}
-
-int ViEExternalCodecImpl::Release() {
-  // Decrease ref count.
-  (*this)--;
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_WARNING) << "ViEExternalCodec released too many times.";
-    shared_data_->SetLastError(kViEAPIDoesNotExist);
-    return -1;
-  }
-  return ref_count;
-}
-
-ViEExternalCodecImpl::ViEExternalCodecImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {
-}
-
-ViEExternalCodecImpl::~ViEExternalCodecImpl() {
-}
-
-int ViEExternalCodecImpl::RegisterExternalSendCodec(const int video_channel,
-                                                    const unsigned char pl_type,
-                                                    VideoEncoder* encoder,
-                                                    bool internal_source) {
-  assert(encoder != NULL);
-  LOG(LS_INFO) << "Register external encoder for channel " << video_channel
-               << ", pl_type " << static_cast<int>(pl_type)
-               << ", internal_source " << internal_source;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidArgument);
-    return -1;
-  }
-  if (vie_encoder->RegisterExternalEncoder(encoder, pl_type,
-                                           internal_source) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEExternalCodecImpl::DeRegisterExternalSendCodec(
-  const int video_channel, const unsigned char pl_type) {
-  LOG(LS_INFO) << "Deregister external encoder for channel " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViECodecInvalidArgument);
-    return -1;
-  }
-
-  if (vie_encoder->DeRegisterExternalEncoder(pl_type) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEExternalCodecImpl::RegisterExternalReceiveCodec(
-    const int video_channel,
-    const unsigned char pl_type,
-    VideoDecoder* decoder,
-    bool decoder_render,
-    int render_delay) {
-  LOG(LS_INFO) << "Register external decoder for channel " << video_channel
-               << ", pl_type " << static_cast<int>(pl_type)
-               << ", decoder_render " << decoder_render
-               << ", render_delay " << render_delay;
-  assert(decoder != NULL);
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidArgument);
-    return -1;
-  }
-
-  if (vie_channel->RegisterExternalDecoder(pl_type, decoder, decoder_render,
-                                           render_delay) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEExternalCodecImpl::DeRegisterExternalReceiveCodec(
-    const int video_channel, const unsigned char pl_type) {
-  LOG(LS_INFO) << "DeRegisterExternalReceiveCodec for channel " << video_channel
-               << ", pl_type " << static_cast<int>(pl_type);
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViECodecInvalidArgument);
-    return -1;
-  }
-  if (vie_channel->DeRegisterExternalDecoder(pl_type) != 0) {
-    shared_data_->SetLastError(kViECodecUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h
deleted file mode 100644
index da5a445ddf..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
-
-#include "webrtc/video_engine/include/vie_external_codec.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViEExternalCodecImpl
-    : public ViEExternalCodec,
-      public ViERefCount {
- public:
-  // Implements ViEExternalCodec.
-  virtual int Release();
-  virtual int RegisterExternalSendCodec(const int video_channel,
-                                        const unsigned char pl_type,
-                                        VideoEncoder* encoder,
-                                        bool internal_source = false);
-  virtual int DeRegisterExternalSendCodec(const int video_channel,
-                                          const unsigned char pl_type);
-  virtual int RegisterExternalReceiveCodec(const int video_channel,
-                                           const unsigned char pl_type,
-                                           VideoDecoder* decoder,
-                                           bool decoder_render = false,
-                                           int render_delay = 0);
-  virtual int DeRegisterExternalReceiveCodec(const int video_channel,
-                                             const unsigned char pl_type);
-
- protected:
-  explicit ViEExternalCodecImpl(ViESharedData* shared_data);
-  virtual ~ViEExternalCodecImpl();
-
- private:
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc b/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc
deleted file mode 100644
index cb6e206190..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Placed first to get WEBRTC_VIDEO_ENGINE_FILE_API.
-#include "webrtc/engine_configurations.h"
-
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-
-#include "webrtc/video_engine/vie_file_image.h"
-
-#include <stdio.h>  // NOLINT
-
-#include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/common_video/jpeg/include/jpeg.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-
-namespace webrtc {
-
-int ViEFileImage::ConvertJPEGToVideoFrame(int engine_id,
-                                          const char* file_nameUTF8,
-                                          I420VideoFrame* video_frame) {
-  // Read jpeg file into temporary buffer.
-  EncodedImage image_buffer;
-
-  FILE* image_file = fopen(file_nameUTF8, "rb");
-  if (!image_file) {
-    return -1;
-  }
-  if (fseek(image_file, 0, SEEK_END) != 0) {
-    fclose(image_file);
-    return -1;
-  }
-  int buffer_size = ftell(image_file);
-  if (buffer_size == -1) {
-    fclose(image_file);
-    return -1;
-  }
-  image_buffer._size = buffer_size;
-  if (fseek(image_file, 0, SEEK_SET) != 0) {
-    fclose(image_file);
-    return -1;
-  }
-  image_buffer._buffer = new uint8_t[ image_buffer._size + 1];
-  if (image_buffer._size != fread(image_buffer._buffer, sizeof(uint8_t),
-                                  image_buffer._size, image_file)) {
-    fclose(image_file);
-    delete [] image_buffer._buffer;
-    return -1;
-  }
-  fclose(image_file);
-
-  int ret = ConvertJpegToI420(image_buffer, video_frame);
-
-  delete [] image_buffer._buffer;
-  image_buffer._buffer = NULL;
-
-  if (ret == -1) {
-    return -1;
-  } else if (ret == -3) {
-  }
-  return 0;
-}
-
-int ViEFileImage::ConvertPictureToI420VideoFrame(int engine_id,
-                                                 const ViEPicture& picture,
-                                                 I420VideoFrame* video_frame) {
-  int half_width = (picture.width + 1) / 2;
-  video_frame->CreateEmptyFrame(picture.width, picture.height,
-                                picture.width, half_width, half_width);
-  return ConvertToI420(kI420, picture.data, 0, 0, picture.width, picture.height,
-                       0, kVideoRotation_0, video_frame);
-}
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_FILE_API
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_file_image.h b/media/webrtc/trunk/webrtc/video_engine/vie_file_image.h
deleted file mode 100644
index 1b5e1affcd..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_file_image.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
-
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_file.h"
-
-namespace webrtc {
-
-class ViEFileImage {
- public:
-  static int ConvertJPEGToVideoFrame(int engine_id,
-                                     const char* file_nameUTF8,
-                                     I420VideoFrame* video_frame);
-  static int ConvertPictureToI420VideoFrame(int engine_id,
-                                            const ViEPicture& picture,
-                                            I420VideoFrame* video_frame);
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc b/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc
deleted file mode 100644
index d7b453d155..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/video_engine/vie_defines.h"
-
-namespace webrtc {
-
-ViEFrameProviderBase::ViEFrameProviderBase(int Id, int engine_id)
-    : id_(Id),
-      engine_id_(engine_id),
-      provider_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      frame_delay_(0) {
-  frame_delivery_thread_checker_.DetachFromThread();
-}
-
-ViEFrameProviderBase::~ViEFrameProviderBase() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  // XXX!!! FIX THIS - remove our callback
-  // DCHECK(frame_callbacks_.empty());
-
-  // TODO(tommi): Remove this when we're confident we've fixed the places where
-  // cleanup wasn't being done.
-  for (ViEFrameCallback* callback : frame_callbacks_) {
-    LOG_F(LS_WARNING) << "FrameCallback still registered.";
-    callback->ProviderDestroyed(id_);
-  }
-}
-
-int ViEFrameProviderBase::Id() const {
-  return id_;
-}
-
-void ViEFrameProviderBase::DeliverFrame(I420VideoFrame* video_frame,
-                                        const std::vector<uint32_t>& csrcs) {
-  DCHECK(frame_delivery_thread_checker_.CalledOnValidThread());
-#ifdef DEBUG_
-  const TickTime start_process_time = TickTime::Now();
-#endif
-  CriticalSectionScoped cs(provider_cs_.get());
-
-  // Deliver the frame to all registered callbacks.
-  if (frame_callbacks_.size() == 1) {
-    // We don't have to copy the frame.
-    frame_callbacks_.front()->DeliverFrame(id_, video_frame, csrcs);
-  } else {
-    for (ViEFrameCallback* callback : frame_callbacks_) {
-      if (video_frame->native_handle() != NULL) {
-        callback->DeliverFrame(id_, video_frame, csrcs);
-      } else {
-        // Make a copy of the frame for all callbacks.
-        if (!extra_frame_.get()) {
-          extra_frame_.reset(new I420VideoFrame());
-        }
-        // TODO(mflodman): We can get rid of this frame copy.
-        extra_frame_->CopyFrame(*video_frame);
-        callback->DeliverFrame(id_, extra_frame_.get(), csrcs);
-      }
-    }
-  }
-#ifdef DEBUG_
-  const int process_time =
-      static_cast<int>((TickTime::Now() - start_process_time).Milliseconds());
-  if (process_time > 25) {
-    // Warn if the delivery time is too long.
-    LOG(LS_WARNING) << "Too long time delivering frame " << process_time;
-  }
-#endif
-}
-
-void ViEFrameProviderBase::SetFrameDelay(int frame_delay) {
-  // Called on the capture thread (see OnIncomingCapturedFrame).
-  // To test, run ViEStandardIntegrationTest.RunsBaseTestWithoutErrors
-  // in vie_auto_tests.
-  // In the same test, it appears that it's also called on a thread that's
-  // neither the ctor thread nor the capture thread.
-  CriticalSectionScoped cs(provider_cs_.get());
-  frame_delay_ = frame_delay;
-
-  for (ViEFrameCallback* callback : frame_callbacks_) {
-    callback->DelayChanged(id_, frame_delay);
-  }
-}
-
-int ViEFrameProviderBase::FrameDelay() {
-  // Called on the default thread in WebRtcVideoMediaChannelTest.SetSend
-  // (libjingle_media_unittest).
-
-  // Called on neither the ctor thread nor the capture thread in
-  // BitrateEstimatorTest.ImmediatelySwitchToAST (video_engine_tests).
-
-  // Most of the time Called on the capture thread (see OnCaptureDelayChanged).
-  // To test, run ViEStandardIntegrationTest.RunsBaseTestWithoutErrors
-  // in vie_auto_tests.
-  return frame_delay_;
-}
-
-int ViEFrameProviderBase::GetBestFormat(int* best_width,
-                                        int* best_height,
-                                        int* best_frame_rate) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  int largest_width = 0;
-  int largest_height = 0;
-  int highest_frame_rate = 0;
-
-  // Here we don't need to grab the provider_cs_ lock to run through the list
-  // of callbacks.  The reason is that we know that we're currently on the same
-  // thread that is the only thread that will modify the callback list and
-  // we can be sure that the thread won't race with itself.
-  for (ViEFrameCallback* callback : frame_callbacks_) {
-    int prefered_width = 0;
-    int prefered_height = 0;
-    int prefered_frame_rate = 0;
-    if (callback->GetPreferedFrameSettings(&prefered_width, &prefered_height,
-                                           &prefered_frame_rate) == 0) {
-      if (prefered_width > largest_width) {
-        largest_width = prefered_width;
-      }
-      if (prefered_height > largest_height) {
-        largest_height = prefered_height;
-      }
-      if (prefered_frame_rate > highest_frame_rate) {
-        highest_frame_rate = prefered_frame_rate;
-      }
-    }
-  }
-  *best_width = largest_width;
-  *best_height = largest_height;
-  *best_frame_rate = highest_frame_rate;
-  return 0;
-}
-
-int ViEFrameProviderBase::RegisterFrameCallback(
-    int observer_id, ViEFrameCallback* callback_object) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(callback_object);
-  {
-    CriticalSectionScoped cs(provider_cs_.get());
-    if (std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
-                  callback_object) != frame_callbacks_.end()) {
-      DCHECK(false && "frameObserver already registered");
-      return -1;
-    }
-    frame_callbacks_.push_back(callback_object);
-  }
-  // Report current capture delay.
-  callback_object->DelayChanged(id_, frame_delay_);
-
-  // Notify implementer of this class that the callback list have changed.
-  FrameCallbackChanged();
-  return 0;
-}
-
-int ViEFrameProviderBase::DeregisterFrameCallback(
-    const ViEFrameCallback* callback_object) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(callback_object);
-  {
-    CriticalSectionScoped cs(provider_cs_.get());
-    FrameCallbacks::iterator it = std::find(frame_callbacks_.begin(),
-                                            frame_callbacks_.end(),
-                                            callback_object);
-    if (it == frame_callbacks_.end()) {
-      return -1;
-    }
-    frame_callbacks_.erase(it);
-  }
-
-  // Notify implementer of this class that the callback list have changed.
-  FrameCallbackChanged();
-
-  return 0;
-}
-
-bool ViEFrameProviderBase::IsFrameCallbackRegistered(
-    const ViEFrameCallback* callback_object) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(callback_object);
-
-  // Here we don't need to grab the lock to do this lookup.
-  // The reason is that we know that we're currently on the same thread that
-  // is the only thread that will modify the callback list and subsequently the
-  // thread doesn't race with itself.
-  return std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
-                   callback_object) != frame_callbacks_.end();
-}
-}  // namespac webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h b/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h
deleted file mode 100644
index b3ace3bdfc..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
-
-#include <vector>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class VideoEncoder;
-class I420VideoFrame;
-
-// ViEFrameCallback shall be implemented by all classes receiving frames from a
-// frame provider.
-class ViEFrameCallback {
- public:
-  virtual void DeliverFrame(int id,
-                            I420VideoFrame* video_frame,
-                            const std::vector<uint32_t>& csrcs) = 0;
-
-  // The capture delay has changed from the provider. |frame_delay| is given in
-  // ms.
-  virtual void DelayChanged(int id, int frame_delay) = 0;
-
-  // Get the width, height and frame rate preferred by this observer.
-  virtual int GetPreferedFrameSettings(int* width,
-                                       int* height,
-                                       int* frame_rate) = 0;
-
-  // ProviderDestroyed is called when the frame is about to be destroyed. There
-  // must not be any more calls to the frame provider after this.
-  virtual void ProviderDestroyed(int id) = 0;
-
-  virtual ~ViEFrameCallback() {}
-};
-
-// ViEFrameProviderBase is a base class that will deliver frames to all
-// registered ViEFrameCallbacks.
-class ViEFrameProviderBase {
- public:
-  ViEFrameProviderBase(int Id, int engine_id);
-  virtual ~ViEFrameProviderBase();
-
-  // Returns the frame provider id.
-  int Id() const;
-
-  // Register frame callbacks, i.e. a receiver of the captured frame.
-  // Must be called on the same thread as the provider was constructed on.
-  int RegisterFrameCallback(int observer_id, ViEFrameCallback* callback);
-
-  // Unregisters a previously registered callback.  Returns -1 if the callback
-  // object hasn't been registered.
-  // Must be called on the same thread as the provider was constructed on.
-  int DeregisterFrameCallback(const ViEFrameCallback* callback);
-
-  // Determines if a callback is currently registered.
-  // Must be called on the same thread as the provider was constructed on.
-  bool IsFrameCallbackRegistered(const ViEFrameCallback* callback);
-
-  // FrameCallbackChanged
-  // Inherited classes should check for new frame_settings and reconfigure
-  // output if possible.
-  virtual int FrameCallbackChanged() = 0;
-
- protected:
-  void DeliverFrame(I420VideoFrame* video_frame,
-                    const std::vector<uint32_t>& csrcs);
-  void SetFrameDelay(int frame_delay);
-  int FrameDelay();
-  int GetBestFormat(int* best_width,
-                    int* best_height,
-                    int* best_frame_rate);
-
-  rtc::ThreadChecker thread_checker_;
-  rtc::ThreadChecker frame_delivery_thread_checker_;
-
-  const int id_;
-  const int engine_id_;
-
-  // Frame callbacks.
-  typedef std::vector<ViEFrameCallback*> FrameCallbacks;
-  FrameCallbacks frame_callbacks_;
-  const rtc::scoped_ptr<CriticalSectionWrapper> provider_cs_;
-
- private:
-  rtc::scoped_ptr<I420VideoFrame> extra_frame_;
-  int frame_delay_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.cc
deleted file mode 100644
index 13e520aca7..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.cc
+++ /dev/null
@@ -1,268 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_image_process_impl.h"
-
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_capturer.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_input_manager.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViEImageProcess* ViEImageProcess::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViEImageProcessImpl* vie_image_process_impl = vie_impl;
-  // Increase ref count.
-  (*vie_image_process_impl)++;
-  return vie_image_process_impl;
-#else
-  return NULL;
-#endif
-}
-
-int ViEImageProcessImpl::Release() {
-  // Decrease ref count.
-  (*this)--;
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_ERROR) << "ViEImageProcess release too many times";
-    shared_data_->SetLastError(kViEAPIDoesNotExist);
-    return -1;
-  }
-  return ref_count;
-}
-
-ViEImageProcessImpl::ViEImageProcessImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {}
-
-ViEImageProcessImpl::~ViEImageProcessImpl() {}
-
-int ViEImageProcessImpl::RegisterCaptureEffectFilter(
-  const int capture_id,
-  ViEEffectFilter& capture_filter) {
-  LOG_F(LS_INFO) << "capture_id: " << capture_id;
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
-    return -1;
-  }
-  if (vie_capture->RegisterEffectFilter(&capture_filter) != 0) {
-    shared_data_->SetLastError(kViEImageProcessFilterExists);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::DeregisterCaptureEffectFilter(const int capture_id) {
-  LOG_F(LS_INFO) << "capture_id: " << capture_id;
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
-    return -1;
-  }
-  if (vie_capture->RegisterEffectFilter(NULL) != 0) {
-    shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::RegisterSendEffectFilter(
-    const int video_channel,
-    ViEEffectFilter& send_filter) {
-  LOG_F(LS_INFO) << "video_channel: " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (vie_encoder == NULL) {
-    shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_encoder->RegisterEffectFilter(&send_filter) != 0) {
-    shared_data_->SetLastError(kViEImageProcessFilterExists);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::DeregisterSendEffectFilter(const int video_channel) {
-  LOG_F(LS_INFO) << "video_channel: " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (vie_encoder == NULL) {
-    shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
-    return -1;
-  }
-  if (vie_encoder->RegisterEffectFilter(NULL) != 0) {
-    shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::RegisterRenderEffectFilter(
-  const int video_channel,
-  ViEEffectFilter& render_filter) {
-  LOG_F(LS_INFO) << "video_channel: " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterEffectFilter(&render_filter) != 0) {
-    shared_data_->SetLastError(kViEImageProcessFilterExists);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::DeregisterRenderEffectFilter(const int video_channel) {
-  LOG_F(LS_INFO) << "video_channel: " << video_channel;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_channel->RegisterEffectFilter(NULL) != 0) {
-    shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::EnableDeflickering(const int capture_id,
-                                            const bool enable) {
-  LOG_F(LS_INFO) << "capture_id: " << capture_id
-                 << " enable: " << (enable ? "on" : "off");
-
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
-    return -1;
-  }
-
-  if (vie_capture->EnableDeflickering(enable) != 0) {
-    if (enable) {
-      shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
-    } else {
-      shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
-    }
-    return -1;
-  }
-  return 0;
-}
-
-int ViEImageProcessImpl::EnableColorEnhancement(const int video_channel,
-                                                const bool enable) {
-  LOG_F(LS_INFO) << "video_channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off");
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->EnableColorEnhancement(enable) != 0) {
-    if (enable) {
-      shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
-    } else {
-      shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
-    }
-    return -1;
-  }
-  return 0;
-}
-
-void ViEImageProcessImpl::RegisterPreEncodeCallback(
-    int video_channel,
-    I420FrameCallback* pre_encode_callback) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  vie_encoder->RegisterPreEncodeCallback(pre_encode_callback);
-}
-
-void ViEImageProcessImpl::DeRegisterPreEncodeCallback(int video_channel) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder != NULL);
-  vie_encoder->DeRegisterPreEncodeCallback();
-}
-
-void ViEImageProcessImpl::RegisterPostEncodeImageCallback(
-    int video_channel,
-    EncodedImageCallback* post_encode_callback) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder != NULL);
-  vie_encoder->RegisterPostEncodeImageCallback(post_encode_callback);
-}
-
-void ViEImageProcessImpl::DeRegisterPostEncodeCallback(int video_channel) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  assert(vie_encoder != NULL);
-  vie_encoder->DeRegisterPostEncodeImageCallback();
-}
-
-void ViEImageProcessImpl::RegisterPreDecodeImageCallback(
-    int video_channel,
-    EncodedImageCallback* pre_decode_callback) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* channel = cs.Channel(video_channel);
-  channel->RegisterPreDecodeImageCallback(pre_decode_callback);
-}
-
-void ViEImageProcessImpl::DeRegisterPreDecodeCallback(int video_channel) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* channel = cs.Channel(video_channel);
-  channel->RegisterPreDecodeImageCallback(NULL);
-}
-
-void ViEImageProcessImpl::RegisterPreRenderCallback(
-    int video_channel,
-    I420FrameCallback* pre_render_callback) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterPreRenderCallback(pre_render_callback);
-}
-
-void ViEImageProcessImpl::DeRegisterPreRenderCallback(int video_channel) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterPreRenderCallback(NULL);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h
deleted file mode 100644
index 64775b1e19..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
-
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViEImageProcessImpl
-    : public ViEImageProcess,
-      public ViERefCount {
- public:
-  // Implements ViEImageProcess.
-  virtual int Release();
-  virtual int RegisterCaptureEffectFilter(const int capture_id,
-                                          ViEEffectFilter& capture_filter);
-  virtual int DeregisterCaptureEffectFilter(const int capture_id);
-  virtual int RegisterSendEffectFilter(const int video_channel,
-                                       ViEEffectFilter& send_filter);
-  virtual int DeregisterSendEffectFilter(const int video_channel);
-  virtual int RegisterRenderEffectFilter(const int video_channel,
-                                         ViEEffectFilter& render_filter);
-  virtual int DeregisterRenderEffectFilter(const int video_channel);
-  virtual int EnableDeflickering(const int capture_id, const bool enable);
-  virtual int EnableColorEnhancement(const int video_channel,
-                                     const bool enable);
-  void RegisterPreEncodeCallback(
-      int video_channel,
-      I420FrameCallback* pre_encode_callback) override;
-  void DeRegisterPreEncodeCallback(int video_channel) override;
-
-  void RegisterPostEncodeImageCallback(
-      int video_channel,
-      EncodedImageCallback* post_encode_callback) override;
-  void DeRegisterPostEncodeCallback(int video_channel) override;
-
-  void RegisterPreDecodeImageCallback(
-      int video_channel,
-      EncodedImageCallback* post_encode_callback) override;
-  void DeRegisterPreDecodeCallback(int video_channel) override;
-
-  void RegisterPreRenderCallback(
-      int video_channel,
-      I420FrameCallback* pre_render_callback) override;
-  void DeRegisterPreRenderCallback(int video_channel) override;
-
- protected:
-  explicit ViEImageProcessImpl(ViESharedData* shared_data);
-  virtual ~ViEImageProcessImpl();
-
- private:
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_impl.cc
deleted file mode 100644
index 4a5d5fd8cb..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_impl.cc
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_impl.h"
-
-#include "webrtc/common.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-#ifdef WEBRTC_ANDROID
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#endif
-
-namespace webrtc {
-
-enum { kModuleId = 0 };
-
-VideoEngine* VideoEngine::Create() {
-  return new VideoEngineImpl(new Config(), true /* owns_config */);
-}
-
-VideoEngine* VideoEngine::Create(const Config& config) {
-  return new VideoEngineImpl(&config, false /* owns_config */);
-}
-
-bool VideoEngine::Delete(VideoEngine*& video_engine) {
-  if (!video_engine)
-    return false;
-
-  LOG_F(LS_INFO);
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-
-  // Check all reference counters.
-  ViEBaseImpl* vie_base = vie_impl;
-  if (vie_base->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViEBase ref count > 0: " << vie_base->GetCount();
-    return false;
-  }
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
-  ViECaptureImpl* vie_capture = vie_impl;
-  if (vie_capture->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViECapture ref count > 0: " << vie_capture->GetCount();
-    return false;
-  }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
-  ViECodecImpl* vie_codec = vie_impl;
-  if (vie_codec->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViECodec ref count > 0: " << vie_codec->GetCount();
-    return false;
-  }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-  ViEExternalCodecImpl* vie_external_codec = vie_impl;
-  if (vie_external_codec->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViEExternalCodec ref count > 0: "
-                  << vie_external_codec->GetCount();
-    return false;
-  }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-  ViEFileImpl* vie_file = vie_impl;
-  if (vie_file->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViEFile ref count > 0: " << vie_file->GetCount();
-    return false;
-  }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-  ViEImageProcessImpl* vie_image_process = vie_impl;
-  if (vie_image_process->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViEImageProcess ref count > 0: "
-                  << vie_image_process->GetCount();
-    return false;
-  }
-#endif
-  ViENetworkImpl* vie_network = vie_impl;
-  if (vie_network->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViENetwork ref count > 0: " << vie_network->GetCount();
-    return false;
-  }
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
-  ViERenderImpl* vie_render = vie_impl;
-  if (vie_render->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViERender ref count > 0: " << vie_render->GetCount();
-    return false;
-  }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-  ViERTP_RTCPImpl* vie_rtp_rtcp = vie_impl;
-  if (vie_rtp_rtcp->GetCount() > 0) {
-    LOG(LS_ERROR) << "ViERTP_RTCP ref count > 0: " << vie_rtp_rtcp->GetCount();
-    return false;
-  }
-#endif
-
-  delete vie_impl;
-  vie_impl = NULL;
-  video_engine = NULL;
-
-  return true;
-}
-
-int VideoEngine::SetTraceFile(const char* file_nameUTF8,
-                              const bool add_file_counter) {
-  if (!file_nameUTF8) {
-    return -1;
-  }
-  if (Trace::SetTraceFile(file_nameUTF8, add_file_counter) == -1) {
-    return -1;
-  }
-  LOG_F(LS_INFO) << "filename: " << file_nameUTF8
-                 << " add_file_counter: " << (add_file_counter ? "yes" : "no");
-  return 0;
-}
-
-int VideoEngine::SetTraceFilter(const unsigned int filter) {
-  uint32_t old_filter = Trace::level_filter();
-
-  if (filter == kTraceNone && old_filter != kTraceNone) {
-    // Do the logging before turning it off.
-    LOG_F(LS_INFO) << "filter: " << filter;
-  }
-
-  Trace::set_level_filter(filter);
-  LOG_F(LS_INFO) << "filter: " << filter;
-  return 0;
-}
-
-int VideoEngine::SetTraceCallback(TraceCallback* callback) {
-  LOG_F(LS_INFO);
-  return Trace::SetTraceCallback(callback);
-}
-
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_GONK)
-int VideoEngine::SetAndroidObjects(JavaVM* javaVM) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVideo, kModuleId,
-               "SetAndroidObjects()");
-
-  if (SetCaptureAndroidVM(javaVM) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVideo, kModuleId,
-                 "Could not set capture Android VM");
-    return -1;
-  }
-#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-  if (SetRenderAndroidVM(javaVM) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVideo, kModuleId,
-                 "Could not set render Android VM");
-    return -1;
-  }
-#endif
-  return 0;
-}
-#endif
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_impl.h
deleted file mode 100644
index 34fd7f7f3b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_impl.h
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/video_engine/vie_defines.h"
-
-#include "webrtc/video_engine/vie_base_impl.h"
-
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
-#include "webrtc/video_engine/vie_capture_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
-#include "webrtc/video_engine/vie_codec_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-#include "webrtc/video_engine/vie_file_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-#include "webrtc/video_engine/vie_image_process_impl.h"
-#endif
-#include "webrtc/video_engine/vie_network_impl.h"
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
-#include "webrtc/video_engine/vie_render_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-#include "webrtc/video_engine/vie_rtp_rtcp_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-#include "webrtc/video_engine/vie_external_codec_impl.h"
-#endif
-
-namespace webrtc {
-
-class VideoEngineImpl
-    : public ViEBaseImpl,
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
-      public ViECodecImpl,
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
-      public ViECaptureImpl,
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-      public ViEFileImpl,
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-      public ViEImageProcessImpl,
-#endif
-      public ViENetworkImpl,
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
-      public ViERenderImpl,
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-      public ViERTP_RTCPImpl,
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-      public ViEExternalCodecImpl,
-#endif
-      public VideoEngine
-{  // NOLINT
- public:
-  VideoEngineImpl(const Config* config, bool owns_config)
-      : ViEBaseImpl(*config),
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
-        ViECodecImpl(ViEBaseImpl::shared_data()),
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
-        ViECaptureImpl(ViEBaseImpl::shared_data()),
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-        ViEFileImpl(ViEBaseImpl::shared_data()),
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-        ViEImageProcessImpl(ViEBaseImpl::shared_data()),
-#endif
-        ViENetworkImpl(ViEBaseImpl::shared_data()),
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
-        ViERenderImpl(ViEBaseImpl::shared_data()),
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-        ViERTP_RTCPImpl(ViEBaseImpl::shared_data()),
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-        ViEExternalCodecImpl(ViEBaseImpl::shared_data()),
-#endif
-        own_config_(owns_config ? config : NULL)
-  {}
-  virtual ~VideoEngineImpl() {}
-
- private:
-  // Placeholder for the case where this owns the config.
-  rtc::scoped_ptr<const Config> own_config_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc b/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc
deleted file mode 100644
index 5ac2018e7d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc
+++ /dev/null
@@ -1,431 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_input_manager.h"
-
-#include <assert.h>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_capturer.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/desktop_capture_impl.h"
-#include "webrtc/video_engine/browser_capture_impl.h"
-
-namespace webrtc {
-
-ViEInputManager::ViEInputManager(const int engine_id, const Config& config)
-    : config_(config),
-      engine_id_(engine_id),
-      map_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      device_info_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      observer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      observer_(NULL),
-      vie_frame_provider_map_(),
-      capture_device_info_(NULL),
-      module_process_thread_(NULL) {
-  for (int idx = 0; idx < kViEMaxCaptureDevices; idx++) {
-    free_capture_device_id_[idx] = true;
-  }
-}
-
-ViEInputManager::~ViEInputManager() {
-  for (FrameProviderMap::iterator it = vie_frame_provider_map_.begin();
-       it != vie_frame_provider_map_.end();
-       ++it) {
-    delete it->second;
-  }
-
-  delete capture_device_info_;
-}
-void ViEInputManager::SetModuleProcessThread(
-    ProcessThread* module_process_thread) {
-  assert(!module_process_thread_);
-  module_process_thread_ = module_process_thread;
-}
-
-int ViEInputManager::NumberOfCaptureDevices() {
-  CriticalSectionScoped cs(device_info_cs_.get());
-  if (!GetDeviceInfo())
-    return 0;
-  assert(capture_device_info_);
-  capture_device_info_->Refresh();
-  return capture_device_info_->NumberOfDevices();
-}
-
-int ViEInputManager::GetDeviceName(uint32_t device_number,
-                                   char* device_nameUTF8,
-                                   uint32_t device_name_length,
-                                   char* device_unique_idUTF8,
-                                   uint32_t device_unique_idUTF8Length,
-                                   pid_t* pid) {
-  CriticalSectionScoped cs(device_info_cs_.get());
-  GetDeviceInfo();
-  assert(capture_device_info_);
-  return capture_device_info_->GetDeviceName(device_number, device_nameUTF8,
-                                             device_name_length,
-                                             device_unique_idUTF8,
-                                             device_unique_idUTF8Length,
-                                             NULL, 0, pid);
-}
-
-int ViEInputManager::NumberOfCaptureCapabilities(
-  const char* device_unique_idUTF8) {
-  CriticalSectionScoped cs(device_info_cs_.get());
-  if (!GetDeviceInfo())
-    return 0;
-  assert(capture_device_info_);
-  return capture_device_info_->NumberOfCapabilities(device_unique_idUTF8);
-}
-
-int ViEInputManager::GetCaptureCapability(
-    const char* device_unique_idUTF8,
-    const uint32_t device_capability_number,
-    CaptureCapability& capability) {
-  CriticalSectionScoped cs(device_info_cs_.get());
-  GetDeviceInfo();
-  assert(capture_device_info_);
-  VideoCaptureCapability module_capability;
-  int result = capture_device_info_->GetCapability(device_unique_idUTF8,
-                                                   device_capability_number,
-                                                   module_capability);
-  if (result != 0)
-    return result;
-
-  // Copy from module type to public type.
-  capability.expectedCaptureDelay = module_capability.expectedCaptureDelay;
-  capability.height = module_capability.height;
-  capability.width = module_capability.width;
-  capability.interlaced = module_capability.interlaced;
-  capability.rawType = module_capability.rawType;
-  capability.codecType = module_capability.codecType;
-  capability.maxFPS = module_capability.maxFPS;
-  return result;
-}
-
-int ViEInputManager::GetOrientation(const char* device_unique_idUTF8,
-                                    VideoRotation& orientation) {
-  CriticalSectionScoped cs(device_info_cs_.get());
-  GetDeviceInfo();
-  assert(capture_device_info_);
-  return capture_device_info_->GetOrientation(device_unique_idUTF8,
-                                              orientation);
-}
-
-int ViEInputManager::DisplayCaptureSettingsDialogBox(
-    const char* device_unique_idUTF8,
-    const char* dialog_titleUTF8,
-    void* parent_window,
-    uint32_t positionX,
-    uint32_t positionY) {
-  CriticalSectionScoped cs(device_info_cs_.get());
-  GetDeviceInfo();
-  assert(capture_device_info_);
-  return capture_device_info_->DisplayCaptureSettingsDialogBox(
-           device_unique_idUTF8, dialog_titleUTF8, parent_window, positionX,
-           positionY);
-}
-
-int ViEInputManager::CreateCaptureDevice(
-    const char* device_unique_idUTF8,
-    const uint32_t device_unique_idUTF8Length,
-    int& capture_id) {
-  CriticalSectionScoped cs(map_cs_.get());
-
-  // Make sure the device is not already allocated.
-  for (FrameProviderMap::iterator it = vie_frame_provider_map_.begin();
-       it != vie_frame_provider_map_.end();
-       ++it) {
-    // Make sure this is a capture device.
-    if (it->first >= kViECaptureIdBase && it->first <= kViECaptureIdMax) {
-      ViECapturer* vie_capture = static_cast<ViECapturer*>(it->second);
-      assert(vie_capture);
-      // TODO(mflodman) Can we change input to avoid this cast?
-      const char* device_name =
-          reinterpret_cast<const char*>(vie_capture->CurrentDeviceName());
-      if (strncmp(device_name, device_unique_idUTF8,
-                  strlen(device_name)) == 0) {
-        return kViECaptureDeviceAlreadyAllocated;
-      }
-    }
-  }
-
-  // Make sure the device name is valid.
-  bool found_device = false;
-  CriticalSectionScoped cs_devinfo(device_info_cs_.get());
-  GetDeviceInfo();
-  assert(capture_device_info_);
-  for (uint32_t device_index = 0;
-       device_index < capture_device_info_->NumberOfDevices(); ++device_index) {
-    if (device_unique_idUTF8Length > kVideoCaptureUniqueNameLength) {
-      // User's string length is longer than the max.
-      return -1;
-    }
-
-    char found_name[kVideoCaptureDeviceNameLength] = "";
-    char found_unique_name[kVideoCaptureUniqueNameLength] = "";
-    capture_device_info_->GetDeviceName(device_index, found_name,
-                                        kVideoCaptureDeviceNameLength,
-                                        found_unique_name,
-                                        kVideoCaptureUniqueNameLength);
-
-    // TODO(mflodman) Can we change input to avoid this cast?
-    const char* cast_id = reinterpret_cast<const char*>(device_unique_idUTF8);
-    if (strncmp(cast_id, reinterpret_cast<const char*>(found_unique_name),
-                strlen(cast_id)) == 0) {
-      found_device = true;
-      break;
-    }
-  }
-  if (!found_device) {
-    LOG(LS_ERROR) << "Capture device not found: " << device_unique_idUTF8;
-    return kViECaptureDeviceDoesNotExist;
-  }
-
-  int newcapture_id = 0;
-  if (!GetFreeCaptureId(&newcapture_id)) {
-    LOG(LS_ERROR) << "All capture devices already allocated.";
-    return kViECaptureDeviceMaxNoDevicesAllocated;
-  }
-  ViECapturer* vie_capture = ViECapturer::CreateViECapture(
-      newcapture_id, engine_id_, config_, device_unique_idUTF8,
-      device_unique_idUTF8Length, *module_process_thread_);
-  if (!vie_capture) {
-    ReturnCaptureId(newcapture_id);
-    return kViECaptureDeviceUnknownError;
-  }
-
-  vie_frame_provider_map_[newcapture_id] = vie_capture;
-  capture_id = newcapture_id;
-  return 0;
-}
-
-int ViEInputManager::CreateCaptureDevice(VideoCaptureModule* capture_module,
-                                         int& capture_id) {
-  CriticalSectionScoped cs(map_cs_.get());
-  int newcapture_id = 0;
-  if (!GetFreeCaptureId(&newcapture_id)) {
-    LOG(LS_ERROR) << "All capture devices already allocated.";
-    return kViECaptureDeviceMaxNoDevicesAllocated;
-  }
-
-  ViECapturer* vie_capture = ViECapturer::CreateViECapture(
-      newcapture_id, engine_id_, config_,
-      capture_module, *module_process_thread_);
-  if (!vie_capture) {
-    ReturnCaptureId(newcapture_id);
-    return kViECaptureDeviceUnknownError;
-  }
-  vie_frame_provider_map_[newcapture_id] = vie_capture;
-  capture_id = newcapture_id;
-  return 0;
-}
-
-int ViEInputManager::DestroyCaptureDevice(const int capture_id) {
-  ViECapturer* vie_capture = NULL;
-  {
-    // We need exclusive access to the object to delete it.
-    // Take this write lock first since the read lock is taken before map_cs_.
-    ViEManagerWriteScoped wl(this);
-    CriticalSectionScoped cs(map_cs_.get());
-
-    vie_capture = ViECapturePtr(capture_id);
-    if (!vie_capture) {
-      LOG(LS_ERROR) << "No such capture device id: " << capture_id;
-      return -1;
-    }
-    vie_frame_provider_map_.erase(capture_id);
-    ReturnCaptureId(capture_id);
-    // Leave cs before deleting the capture object. This is because deleting the
-    // object might cause deletions of renderers so we prefer to not have a lock
-    // at that time.
-  }
-  delete vie_capture;
-  return 0;
-}
-
-int ViEInputManager::CreateExternalCaptureDevice(
-    ViEExternalCapture*& external_capture,
-    int& capture_id) {
-  CriticalSectionScoped cs(map_cs_.get());
-
-  int newcapture_id = 0;
-  if (GetFreeCaptureId(&newcapture_id) == false) {
-    LOG(LS_ERROR) << "All capture devices already allocated.";
-    return kViECaptureDeviceMaxNoDevicesAllocated;
-  }
-
-  ViECapturer* vie_capture = ViECapturer::CreateViECapture(
-      newcapture_id, engine_id_, config_, NULL, 0, *module_process_thread_);
-  if (!vie_capture) {
-    ReturnCaptureId(newcapture_id);
-    return kViECaptureDeviceUnknownError;
-  }
-
-  vie_frame_provider_map_[newcapture_id] = vie_capture;
-  capture_id = newcapture_id;
-  external_capture = vie_capture;
-  return 0;
-}
-
-bool ViEInputManager::GetFreeCaptureId(int* freecapture_id) {
-  for (int id = 0; id < kViEMaxCaptureDevices; id++) {
-    if (free_capture_device_id_[id]) {
-      // We found a free capture device id.
-      free_capture_device_id_[id] = false;
-      *freecapture_id = id + kViECaptureIdBase;
-      return true;
-    }
-  }
-  return false;
-}
-
-void ViEInputManager::ReturnCaptureId(int capture_id) {
-  CriticalSectionScoped cs(map_cs_.get());
-  if (capture_id >= kViECaptureIdBase &&
-      capture_id < kViEMaxCaptureDevices + kViECaptureIdBase) {
-    free_capture_device_id_[capture_id - kViECaptureIdBase] = true;
-  }
-  return;
-}
-
-ViEFrameProviderBase* ViEInputManager::ViEFrameProvider(
-    const ViEFrameCallback* capture_observer) const {
-  assert(capture_observer);
-  CriticalSectionScoped cs(map_cs_.get());
-
-  for (FrameProviderMap::const_iterator it = vie_frame_provider_map_.begin();
-       it != vie_frame_provider_map_.end();
-       ++it) {
-    if (it->second->IsFrameCallbackRegistered(capture_observer))
-      return it->second;
-  }
-
-  // No capture device set for this channel.
-  return NULL;
-}
-
-ViEFrameProviderBase* ViEInputManager::ViEFrameProvider(int provider_id) const {
-  CriticalSectionScoped cs(map_cs_.get());
-
-  FrameProviderMap::const_iterator it =
-      vie_frame_provider_map_.find(provider_id);
-  if (it == vie_frame_provider_map_.end())
-    return NULL;
-  return it->second;
-}
-
-ViECapturer* ViEInputManager::ViECapturePtr(int capture_id) const {
-  if (!(capture_id >= kViECaptureIdBase &&
-        capture_id <= kViECaptureIdBase + kViEMaxCaptureDevices)) {
-    LOG(LS_ERROR) << "Capture device doesn't exist " << capture_id << ".";
-    return NULL;
-  }
-
-  return static_cast<ViECapturer*>(ViEFrameProvider(capture_id));
-}
-
-void ViEInputManager::OnDeviceChange() {
-  CriticalSectionScoped cs(observer_cs_.get());
-  if (observer_) {
-    observer_->DeviceChange();
-  }
-}
-
-// Create different DeviceInfo by _config;
-VideoCaptureModule::DeviceInfo* ViEInputManager::GetDeviceInfo() {
-  CaptureDeviceType type = config_.Get<CaptureDeviceInfo>().type;
-
-  if (capture_device_info_ == NULL) {
-    switch (type) {
-      case CaptureDeviceType::Screen:
-      case CaptureDeviceType::Application:
-      case CaptureDeviceType::Window:
-#if !defined(ANDROID) && !defined(WEBRTC_IOS)
-        capture_device_info_ = DesktopCaptureImpl::CreateDeviceInfo(ViEModuleId(engine_id_),
-                                                                    type);
-#endif
-        break;
-      case CaptureDeviceType::Browser:
-        capture_device_info_ = BrowserDeviceInfoImpl::CreateDeviceInfo();
-        break;
-      case CaptureDeviceType::Camera:
-        capture_device_info_ = VideoCaptureFactory::CreateDeviceInfo(ViEModuleId(engine_id_));
-        break;
-      default:
-        // Don't try to build anything for unknown/unsupported types
-        break;
-    }
-  }
-  return capture_device_info_;
-}
-
-int32_t ViEInputManager::RegisterObserver(ViEInputObserver* observer) {
-  {
-    CriticalSectionScoped cs(observer_cs_.get());
-    if (observer_) {
-      LOG_F(LS_ERROR) << "Observer already registered.";
-      return -1;
-    }
-    observer_ = observer;
-  }
-
-  CriticalSectionScoped cs(device_info_cs_.get());
-  if (!GetDeviceInfo())
-    return -1;
-
-  if (capture_device_info_ != NULL)
-    capture_device_info_->RegisterVideoInputFeedBack(*this);
-
-  return 0;
-}
-
-int32_t ViEInputManager::DeRegisterObserver() {
-  {
-    CriticalSectionScoped cs(observer_cs_.get());
-    observer_ = NULL;
-  }
-
-  CriticalSectionScoped cs(device_info_cs_.get());
-  if (capture_device_info_ != NULL) {
-    capture_device_info_->DeRegisterVideoInputFeedBack();
-  }
-  return 0;
-}
-
-ViEInputManagerScoped::ViEInputManagerScoped(
-    const ViEInputManager& vie_input_manager)
-    : ViEManagerScopedBase(vie_input_manager) {
-}
-
-ViECapturer* ViEInputManagerScoped::Capture(int capture_id) const {
-  return static_cast<const ViEInputManager*>(vie_manager_)->ViECapturePtr(
-      capture_id);
-}
-
-ViEFrameProviderBase* ViEInputManagerScoped::FrameProvider(
-    const ViEFrameCallback* capture_observer) const {
-  return static_cast<const ViEInputManager*>(vie_manager_)->ViEFrameProvider(
-      capture_observer);
-}
-
-ViEFrameProviderBase* ViEInputManagerScoped::FrameProvider(
-    int provider_id) const {
-  return static_cast<const ViEInputManager*>(vie_manager_)->ViEFrameProvider(
-      provider_id);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h b/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h
deleted file mode 100644
index d116c2fb1b..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
-
-#include <map>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/common_video/rotation.h"
-#include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-#include "webrtc/video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-class Config;
-class CriticalSectionWrapper;
-class ProcessThread;
-class RWLockWrapper;
-class ViECapturer;
-class ViEExternalCapture;
-class VoiceEngine;
-
-class ViEInputManager : private ViEManagerBase,
-                        protected VideoInputFeedBack {
-  friend class ViEInputManagerScoped;
- public:
-  ViEInputManager(int engine_id, const Config& config);
-  ~ViEInputManager();
-
-  void SetModuleProcessThread(ProcessThread* module_process_thread);
-
-  // Returns number of capture devices.
-  int NumberOfCaptureDevices();
-
-  // Gets name and id for a capture device.
-  int GetDeviceName(uint32_t device_number,
-                    char* device_nameUTF8,
-                    uint32_t device_name_length,
-                    char* device_unique_idUTF8,
-                    uint32_t device_unique_idUTF8Length,
-                    pid_t* pid);
-
-  // Returns the number of capture capabilities for a specified device.
-  int NumberOfCaptureCapabilities(const char* device_unique_idUTF8);
-
-  // Gets a specific capability for a capture device.
-  int GetCaptureCapability(const char* device_unique_idUTF8,
-                           const uint32_t device_capability_number,
-                           CaptureCapability& capability);
-
-  // Show OS specific Capture settings.
-  int DisplayCaptureSettingsDialogBox(const char* device_unique_idUTF8,
-                                      const char* dialog_titleUTF8,
-                                      void* parent_window,
-                                      uint32_t positionX,
-                                      uint32_t positionY);
-  int GetOrientation(const char* device_unique_idUTF8,
-                     VideoRotation& orientation);
-
-  // Creates a capture module for the specified capture device and assigns
-  // a capture device id for the device.
-  // Return zero on success, ViEError on failure.
-  int CreateCaptureDevice(const char* device_unique_idUTF8,
-                          const uint32_t device_unique_idUTF8Length,
-                          int& capture_id);
-  int CreateCaptureDevice(VideoCaptureModule* capture_module,
-                          int& capture_id);
-  int CreateExternalCaptureDevice(ViEExternalCapture*& external_capture,
-                                  int& capture_id);
-  int DestroyCaptureDevice(int capture_id);
-  int32_t RegisterObserver(ViEInputObserver* observer);
-  int32_t DeRegisterObserver();
- protected:
-  VideoCaptureModule::DeviceInfo* GetDeviceInfo();
-  // Implements VideoInputFeedBack.
-  virtual void OnDeviceChange();
- private:
-  // Gets and allocates a free capture device id. Assumed protected by caller.
-  bool GetFreeCaptureId(int* freecapture_id);
-
-  // Frees a capture id assigned in GetFreeCaptureId.
-  void ReturnCaptureId(int capture_id);
-
-  // Gets the ViEFrameProvider for this capture observer.
-  ViEFrameProviderBase* ViEFrameProvider(
-      const ViEFrameCallback* capture_observer) const;
-
-  // Gets the ViEFrameProvider for this capture observer.
-  ViEFrameProviderBase* ViEFrameProvider(int provider_id) const;
-
-  // Gets the ViECapturer for the capture device id.
-  ViECapturer* ViECapturePtr(int capture_id) const;
-
-  const Config& config_;
-  int engine_id_;
-  rtc::scoped_ptr<CriticalSectionWrapper> map_cs_;
-  rtc::scoped_ptr<CriticalSectionWrapper> device_info_cs_;
-  rtc::scoped_ptr<CriticalSectionWrapper> observer_cs_;
-  ViEInputObserver* observer_ GUARDED_BY(observer_cs_.get());
-
-  typedef std::map<int, ViEFrameProviderBase*> FrameProviderMap;
-  FrameProviderMap vie_frame_provider_map_;
-
-  // Capture devices.
-  VideoCaptureModule::DeviceInfo* capture_device_info_;
-  int free_capture_device_id_[kViEMaxCaptureDevices];
-
-  ProcessThread* module_process_thread_;  // Weak.
-};
-
-// Provides protected access to ViEInputManater.
-class ViEInputManagerScoped: private ViEManagerScopedBase {
- public:
-  explicit ViEInputManagerScoped(const ViEInputManager& vie_input_manager);
-
-  ViECapturer* Capture(int capture_id) const;
-  ViEFrameProviderBase* FrameProvider(int provider_id) const;
-  ViEFrameProviderBase* FrameProvider(const ViEFrameCallback*
-                                      capture_observer) const;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_manager_base.cc b/media/webrtc/trunk/webrtc/video_engine/vie_manager_base.cc
deleted file mode 100644
index b9b5d1a79f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_manager_base.cc
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-ViEManagerBase::ViEManagerBase()
-    : instance_rwlock_(*RWLockWrapper::CreateRWLock()) {
-}
-
-ViEManagerBase::~ViEManagerBase() {
-  delete &instance_rwlock_;
-}
-
-void ViEManagerBase::ReadLockManager() const {
-  instance_rwlock_.AcquireLockShared();
-}
-
-void ViEManagerBase::ReleaseLockManager() const {
-  instance_rwlock_.ReleaseLockShared();
-}
-
-void ViEManagerBase::WriteLockManager() {
-  instance_rwlock_.AcquireLockExclusive();
-}
-
-void ViEManagerBase::ReleaseWriteLockManager() {
-  instance_rwlock_.ReleaseLockExclusive();
-}
-
-ViEManagerScopedBase::ViEManagerScopedBase(const ViEManagerBase& ViEManagerBase)
-    : vie_manager_(&ViEManagerBase),
-      ref_count_(0) {
-  vie_manager_->ReadLockManager();
-}
-
-ViEManagerScopedBase::~ViEManagerScopedBase() {
-  assert(ref_count_ == 0);
-  vie_manager_->ReleaseLockManager();
-}
-
-ViEManagerWriteScoped::ViEManagerWriteScoped(ViEManagerBase* vie_manager)
-    : vie_manager_(vie_manager) {
-  vie_manager_->WriteLockManager();
-}
-
-ViEManagerWriteScoped::~ViEManagerWriteScoped() {
-  vie_manager_->ReleaseWriteLockManager();
-}
-
-ViEManagedItemScopedBase::ViEManagedItemScopedBase(
-    ViEManagerScopedBase* vie_scoped_manager)
-    : vie_scoped_manager_(vie_scoped_manager) {
-  vie_scoped_manager_->ref_count_++;
-}
-
-ViEManagedItemScopedBase::~ViEManagedItemScopedBase() {
-  vie_scoped_manager_->ref_count_--;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_manager_base.h b/media/webrtc/trunk/webrtc/video_engine/vie_manager_base.h
deleted file mode 100644
index c5e92106ab..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_manager_base.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
-
-#include "webrtc/base/thread_annotations.h"
-
-namespace webrtc {
-
-class RWLockWrapper;
-
-class LOCKABLE ViEManagerBase {
-  friend class ViEManagedItemScopedBase;
-  friend class ViEManagerScopedBase;
-  friend class ViEManagerWriteScoped;
- public:
-  ViEManagerBase();
-  ~ViEManagerBase();
-
- private:
-  // Exclusive lock, used by ViEManagerWriteScoped.
-  void WriteLockManager() EXCLUSIVE_LOCK_FUNCTION();
-
-  // Releases exclusive lock, used by ViEManagerWriteScoped.
-  void ReleaseWriteLockManager() UNLOCK_FUNCTION();
-
-  // Increases lock count, used by ViEManagerScopedBase.
-  void ReadLockManager() const SHARED_LOCK_FUNCTION();
-
-  // Releases the lock count, used by ViEManagerScopedBase.
-  void ReleaseLockManager() const UNLOCK_FUNCTION();
-
-  RWLockWrapper& instance_rwlock_;
-};
-
-class SCOPED_LOCKABLE ViEManagerWriteScoped {
- public:
-  explicit ViEManagerWriteScoped(ViEManagerBase* vie_manager)
-      EXCLUSIVE_LOCK_FUNCTION(vie_manager);
-  ~ViEManagerWriteScoped() UNLOCK_FUNCTION();
-
- private:
-  ViEManagerBase* vie_manager_;
-};
-
-class ViEManagerScopedBase {
-  friend class ViEManagedItemScopedBase;
- public:
-  explicit ViEManagerScopedBase(const ViEManagerBase& vie_manager);
-  ~ViEManagerScopedBase();
-
- protected:
-  const ViEManagerBase* vie_manager_;
-
- private:
-  int ref_count_;
-};
-
-class ViEManagedItemScopedBase {
- public:
-  explicit ViEManagedItemScopedBase(ViEManagerScopedBase* vie_scoped_manager);
-  ~ViEManagedItemScopedBase();
-
- protected:
-  ViEManagerScopedBase* vie_scoped_manager_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc
deleted file mode 100644
index 5d45187a61..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc
+++ /dev/null
@@ -1,181 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_network_impl.h"
-
-#include <stdio.h>
-#if (defined(WIN32_) || defined(WIN64_))
-#include <qos.h>
-#endif
-
-#include "webrtc/base/checks.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViENetwork* ViENetwork::GetInterface(VideoEngine* video_engine) {
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViENetworkImpl* vie_networkImpl = vie_impl;
-  // Increase ref count.
-  (*vie_networkImpl)++;
-  return vie_networkImpl;
-}
-
-int ViENetworkImpl::Release() {
-  // Decrease ref count.
-  (*this)--;
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_ERROR) << "ViENetwork release too many times";
-    shared_data_->SetLastError(kViEAPIDoesNotExist);
-    return -1;
-  }
-  return ref_count;
-}
-
-ViENetworkImpl::ViENetworkImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {
-}
-
-ViENetworkImpl::~ViENetworkImpl() {
-}
-
-void ViENetworkImpl::SetBitrateConfig(int video_channel,
-                                      int min_bitrate_bps,
-                                      int start_bitrate_bps,
-                                      int max_bitrate_bps) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " new bitrate config: min=" << min_bitrate_bps
-                 << ", start=" << start_bitrate_bps
-                 << ", max=" << max_bitrate_bps;
-  bool success = shared_data_->channel_manager()->SetBitrateConfig(
-      video_channel, min_bitrate_bps, start_bitrate_bps, max_bitrate_bps);
-  DCHECK(success);
-}
-
-void ViENetworkImpl::SetNetworkTransmissionState(const int video_channel,
-                                                 const bool is_transmitting) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " transmitting: " << (is_transmitting ? "yes" : "no");
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return;
-  }
-  vie_encoder->SetNetworkTransmissionState(is_transmitting);
-}
-
-int ViENetworkImpl::RegisterSendTransport(const int video_channel,
-                                          Transport& transport) {
-  LOG_F(LS_INFO) << "channel: " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->Sending()) {
-    LOG_F(LS_ERROR) << "Already sending on channel: " << video_channel;
-    shared_data_->SetLastError(kViENetworkAlreadySending);
-    return -1;
-  }
-  if (vie_channel->RegisterSendTransport(&transport) != 0) {
-    shared_data_->SetLastError(kViENetworkUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViENetworkImpl::DeregisterSendTransport(const int video_channel) {
-  LOG_F(LS_INFO) << "channel: " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->Sending()) {
-    LOG_F(LS_ERROR) << "Actively sending on channel: " << video_channel;
-    shared_data_->SetLastError(kViENetworkAlreadySending);
-    return -1;
-  }
-  if (vie_channel->DeregisterSendTransport() != 0) {
-    shared_data_->SetLastError(kViENetworkUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViENetworkImpl::ReceivedRTPPacket(const int video_channel, const void* data,
-                                      const size_t length,
-                                      const PacketTime& packet_time) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return -1;
-  }
-  return vie_channel->ReceivedRTPPacket(data, length, packet_time);
-}
-
-int ViENetworkImpl::ReceivedRTCPPacket(const int video_channel,
-                                       const void* data, const size_t length) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return -1;
-  }
-  return vie_channel->ReceivedRTCPPacket(data, length);
-}
-
-int ViENetworkImpl::SetMTU(int video_channel, unsigned int mtu) {
-  LOG_F(LS_INFO) << "channel: " << video_channel << " mtu: " << mtu;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetMTU(mtu) != 0) {
-    shared_data_->SetLastError(kViENetworkUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViENetworkImpl::ReceivedBWEPacket(const int video_channel,
-                                      int64_t arrival_time_ms,
-                                      size_t payload_size,
-                                      const RTPHeader& header) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViENetworkInvalidChannelId);
-    return -1;
-  }
-
-  vie_channel->ReceivedBWEPacket(arrival_time_ms, payload_size, header);
-  return 0;
-}
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h
deleted file mode 100644
index 1354f8cf4a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
-
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViENetworkImpl
-    : public ViENetwork,
-      public ViERefCount {
- public:
-  // Implements ViENetwork.
-  int Release() override;
-  void SetBitrateConfig(int video_channel,
-                        int min_bitrate_bps,
-                        int start_bitrate_bps,
-                        int max_bitrate_bps) override;
-  void SetNetworkTransmissionState(const int video_channel,
-                                   const bool is_transmitting) override;
-  int RegisterSendTransport(const int video_channel,
-                            Transport& transport) override;
-  int DeregisterSendTransport(const int video_channel) override;
-  int ReceivedRTPPacket(const int video_channel,
-                        const void* data,
-                        const size_t length,
-                        const PacketTime& packet_time) override;
-  int ReceivedRTCPPacket(const int video_channel,
-                         const void* data,
-                         const size_t length) override;
-  int SetMTU(int video_channel, unsigned int mtu) override;
-
-  int ReceivedBWEPacket(const int video_channel,
-                        int64_t arrival_time_ms,
-                        size_t payload_size,
-                        const RTPHeader& header) override;
-
- protected:
-  explicit ViENetworkImpl(ViESharedData* shared_data);
-  virtual ~ViENetworkImpl();
-
- private:
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.cc b/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.cc
deleted file mode 100644
index b2fe7acfc4..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_ref_count.h"
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-
-namespace webrtc {
-
-ViERefCount::ViERefCount()
-    : count_(0),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()) {
-}
-
-ViERefCount::~ViERefCount() {
-}
-
-ViERefCount& ViERefCount::operator++(int) {  // NOLINT
-  CriticalSectionScoped lock(crit_.get());
-  count_++;
-  return *this;
-}
-
-ViERefCount& ViERefCount::operator--(int) {  // NOLINT
-  CriticalSectionScoped lock(crit_.get());
-  count_--;
-  return *this;
-}
-
-void ViERefCount::Reset() {
-  CriticalSectionScoped lock(crit_.get());
-  count_ = 0;
-}
-
-int ViERefCount::GetCount() const {
-  return count_;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h b/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h
deleted file mode 100644
index 61533e10e2..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// TODO(mflodman) Remove this class and use ref count class in system_wrappers.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
-
-#include "webrtc/base/scoped_ptr.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-class ViERefCount {
- public:
-  ViERefCount();
-  ~ViERefCount();
-
-  ViERefCount& operator++(int);  // NOLINT
-  ViERefCount& operator--(int);  // NOLINT
-
-  void Reset();
-  int GetCount() const;
-
- private:
-  volatile int count_;
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc
deleted file mode 100644
index 27920602b6..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc
+++ /dev/null
@@ -1,309 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_render_impl.h"
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_capturer.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_input_manager.h"
-#include "webrtc/video_engine/vie_render_manager.h"
-#include "webrtc/video_engine/vie_renderer.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViERender* ViERender::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViERenderImpl* vie_render_impl = vie_impl;
-  // Increase ref count.
-  (*vie_render_impl)++;
-  return vie_render_impl;
-#else
-  return NULL;
-#endif
-}
-
-int ViERenderImpl::Release() {
-  // Decrease ref count
-  (*this)--;
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_ERROR) << "ViERender release too many times";
-    return -1;
-  }
-  return ref_count;
-}
-
-ViERenderImpl::ViERenderImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {}
-
-ViERenderImpl::~ViERenderImpl() {}
-
-int ViERenderImpl::RegisterVideoRenderModule(
-  VideoRender& render_module) {
-  LOG_F(LS_INFO);
-  if (shared_data_->render_manager()->RegisterVideoRenderModule(
-      &render_module) != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::DeRegisterVideoRenderModule(
-  VideoRender& render_module) {
-  LOG_F(LS_INFO);
-  if (shared_data_->render_manager()->DeRegisterVideoRenderModule(
-      &render_module) != 0) {
-    // Error logging is done in ViERenderManager::DeRegisterVideoRenderModule.
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::AddRenderer(const int render_id, void* window,
-                               const unsigned int z_order, const float left,
-                               const float top, const float right,
-                               const float bottom) {
-  LOG_F(LS_INFO) << "render_id: " << render_id << " z_order: " << z_order
-                 << " left: " << left << " top: " << top << " right: " << right
-                 << " bottom: " << bottom;
-  {
-    ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-    if (rs.Renderer(render_id)) {
-      LOG(LS_ERROR) << "Renderer for render_id: " << render_id
-                    << " already exists.";
-      shared_data_->SetLastError(kViERenderAlreadyExists);
-      return -1;
-    }
-  }
-  if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
-    // This is a channel.
-    ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
-    ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
-    if (!frame_provider) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
-        render_id, window, z_order, left, top, right, bottom);
-    if (!renderer) {
-      shared_data_->SetLastError(kViERenderUnknownError);
-      return -1;
-    }
-    return frame_provider->RegisterFrameCallback(render_id, renderer);
-  } else {
-    // Camera or file.
-    ViEInputManagerScoped is(*(shared_data_->input_manager()));
-    ViEFrameProviderBase* frame_provider = is.FrameProvider(render_id);
-    if (!frame_provider) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
-        render_id, window, z_order, left, top, right, bottom);
-    if (!renderer) {
-      shared_data_->SetLastError(kViERenderUnknownError);
-      return -1;
-    }
-    return frame_provider->RegisterFrameCallback(render_id, renderer);
-  }
-}
-
-int ViERenderImpl::RemoveRenderer(const int render_id) {
-  LOG_F(LS_INFO) << "render_id: " << render_id;
-  ViERenderer* renderer = NULL;
-  {
-    ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-    renderer = rs.Renderer(render_id);
-    if (!renderer) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    // Leave the scope lock since we don't want to lock two managers
-    // simultanousely.
-  }
-  if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
-    // This is a channel.
-    ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
-    ViEChannel* channel = cm.Channel(render_id);
-    if (!channel) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    channel->DeregisterFrameCallback(renderer);
-  } else {
-    // Provider owned by inputmanager, i.e. file or capture device.
-    ViEInputManagerScoped is(*(shared_data_->input_manager()));
-    ViEFrameProviderBase* provider = is.FrameProvider(render_id);
-    if (!provider) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    provider->DeregisterFrameCallback(renderer);
-  }
-  if (shared_data_->render_manager()->RemoveRenderStream(render_id) != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::StartRender(const int render_id) {
-  LOG_F(LS_INFO) << "render_id: " << render_id;
-  ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-  ViERenderer* renderer = rs.Renderer(render_id);
-  if (!renderer) {
-    shared_data_->SetLastError(kViERenderInvalidRenderId);
-    return -1;
-  }
-  if (renderer->StartRender() != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::StopRender(const int render_id) {
-  LOG_F(LS_INFO) << "render_id: " << render_id;
-  ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-  ViERenderer* renderer = rs.Renderer(render_id);
-  if (!renderer) {
-    shared_data_->SetLastError(kViERenderInvalidRenderId);
-    return -1;
-  }
-  if (renderer->StopRender() != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::SetExpectedRenderDelay(int render_id, int render_delay) {
-  LOG_F(LS_INFO) << "render_id: " << render_id
-                 << " render_delay: " << render_delay;
-  ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-  ViERenderer* renderer = rs.Renderer(render_id);
-  if (!renderer) {
-    shared_data_->SetLastError(kViERenderInvalidRenderId);
-    return -1;
-  }
-  if (renderer->SetExpectedRenderDelay(render_delay) != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::ConfigureRender(int render_id, const unsigned int z_order,
-                                   const float left, const float top,
-                                   const float right, const float bottom) {
-  LOG_F(LS_INFO) << "render_id: " << render_id << " z_order: " << z_order
-                 << " left: " << left << " top: " << top << " right: " << right
-                 << " bottom: " << bottom;
-  ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-  ViERenderer* renderer = rs.Renderer(render_id);
-  if (!renderer) {
-    shared_data_->SetLastError(kViERenderInvalidRenderId);
-    return -1;
-  }
-
-  if (renderer->ConfigureRenderer(z_order, left, top, right, bottom) != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERenderImpl::AddRenderer(const int render_id,
-                               RawVideoType video_input_format,
-                               ExternalRenderer* external_renderer) {
-  // Check if the client requested a format that we can convert the frames to.
-  if (video_input_format != kVideoI420 &&
-      video_input_format != kVideoYV12 &&
-      video_input_format != kVideoYUY2 &&
-      video_input_format != kVideoUYVY &&
-      video_input_format != kVideoARGB &&
-      video_input_format != kVideoRGB24 &&
-      video_input_format != kVideoRGB565 &&
-      video_input_format != kVideoARGB4444 &&
-      video_input_format != kVideoARGB1555) {
-    LOG(LS_ERROR) << "Unsupported video frame format requested.";
-    shared_data_->SetLastError(kViERenderInvalidFrameFormat);
-    return -1;
-  }
-  {
-    // Verify the renderer doesn't exist.
-    ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-    if (rs.Renderer(render_id)) {
-      LOG_F(LS_ERROR) << "Renderer already exists for render_id: " << render_id;
-      shared_data_->SetLastError(kViERenderAlreadyExists);
-      return -1;
-    }
-  }
-  if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
-    // This is a channel.
-    ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
-    ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
-    if (!frame_provider) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
-        render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
-    if (!renderer) {
-      shared_data_->SetLastError(kViERenderUnknownError);
-      return -1;
-    }
-    if (renderer->SetExternalRenderer(render_id, video_input_format,
-                                      external_renderer) == -1) {
-      shared_data_->SetLastError(kViERenderUnknownError);
-      return -1;
-    }
-
-    return frame_provider->RegisterFrameCallback(render_id, renderer);
-  } else {
-    // Camera or file.
-    ViEInputManagerScoped is(*(shared_data_->input_manager()));
-    ViEFrameProviderBase* frame_provider = is.FrameProvider(render_id);
-    if (!frame_provider) {
-      shared_data_->SetLastError(kViERenderInvalidRenderId);
-      return -1;
-    }
-    ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
-        render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
-    if (!renderer) {
-      shared_data_->SetLastError(kViERenderUnknownError);
-      return -1;
-    }
-    if (renderer->SetExternalRenderer(render_id, video_input_format,
-                                      external_renderer) == -1) {
-      shared_data_->SetLastError(kViERenderUnknownError);
-      return -1;
-    }
-    return frame_provider->RegisterFrameCallback(render_id, renderer);
-  }
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h
deleted file mode 100644
index 5b36d1686f..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
-
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViERenderImpl
-    : public ViERender,
-      public ViERefCount {
- public:
-  // Implements ViERender
-  virtual int Release();
-  virtual int RegisterVideoRenderModule(VideoRender& render_module);  // NOLINT
-  virtual int DeRegisterVideoRenderModule(
-      VideoRender& render_module);  // NOLINT
-  virtual int AddRenderer(const int render_id, void* window,
-                          const unsigned int z_order, const float left,
-                          const float top, const float right,
-                          const float bottom);
-  virtual int RemoveRenderer(const int render_id);
-  virtual int StartRender(const int render_id);
-  virtual int StopRender(const int render_id);
-  virtual int SetExpectedRenderDelay(int render_id, int render_delay);
-  virtual int ConfigureRender(int render_id, const unsigned int z_order,
-                              const float left, const float top,
-                              const float right, const float bottom);
-  virtual int AddRenderer(const int render_id, RawVideoType video_input_format,
-                          ExternalRenderer* renderer);
-
- protected:
-  explicit ViERenderImpl(ViESharedData* shared_data);
-  virtual ~ViERenderImpl();
-
- private:
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.cc b/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.cc
deleted file mode 100644
index e044bfac92..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.cc
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_render_manager.h"
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_renderer.h"
-
-namespace webrtc {
-
-ViERenderManagerScoped::ViERenderManagerScoped(
-    const ViERenderManager& vie_render_manager)
-    : ViEManagerScopedBase(vie_render_manager) {
-}
-
-ViERenderer* ViERenderManagerScoped::Renderer(int32_t render_id) const {
-  return static_cast<const ViERenderManager*>(vie_manager_)->ViERenderPtr(
-           render_id);
-}
-
-ViERenderManager::ViERenderManager(int32_t engine_id)
-    : list_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      engine_id_(engine_id),
-      use_external_render_module_(false) {
-}
-
-ViERenderManager::~ViERenderManager() {
-  for (RendererMap::iterator it = stream_to_vie_renderer_.begin();
-       it != stream_to_vie_renderer_.end();
-       ++it) {
-    // The renderer is deleted in RemoveRenderStream.
-    RemoveRenderStream(it->first);
-  }
-}
-
-int32_t ViERenderManager::RegisterVideoRenderModule(
-    VideoRender* render_module) {
-  // See if there is already a render module registered for the window that
-  // the registrant render module is associated with.
-  VideoRender* current_module = FindRenderModule(render_module->Window());
-  if (current_module) {
-    LOG_F(LS_ERROR) << "A render module is already registered for this window.";
-    return -1;
-  }
-
-  // Register module.
-  render_list_.push_back(render_module);
-  use_external_render_module_ = true;
-  return 0;
-}
-
-int32_t ViERenderManager::DeRegisterVideoRenderModule(
-    VideoRender* render_module) {
-  // Check if there are streams in the module.
-  uint32_t n_streams = render_module->GetNumIncomingRenderStreams();
-  if (n_streams != 0) {
-    LOG(LS_ERROR) << "There are still " << n_streams
-                  << "in this module, cannot de-register.";
-    return -1;
-  }
-
-  for (RenderList::iterator iter = render_list_.begin();
-       iter != render_list_.end(); ++iter) {
-    if (render_module == *iter) {
-      // We've found our renderer. Erase the render module from the map.
-      render_list_.erase(iter);
-      return 0;
-    }
-  }
-
-  LOG(LS_ERROR) << "Module not registered.";
-  return -1;
-}
-
-ViERenderer* ViERenderManager::AddRenderStream(const int32_t render_id,
-                                               void* window,
-                                               const uint32_t z_order,
-                                               const float left,
-                                               const float top,
-                                               const float right,
-                                               const float bottom) {
-  CriticalSectionScoped cs(list_cs_.get());
-
-  if (stream_to_vie_renderer_.find(render_id) !=
-      stream_to_vie_renderer_.end()) {
-    LOG(LS_ERROR) << "Render stream already exists";
-    return NULL;
-  }
-
-  // Get the render module for this window.
-  VideoRender* render_module = FindRenderModule(window);
-  if (render_module == NULL) {
-    // No render module for this window, create a new one.
-    render_module = VideoRender::CreateVideoRender(ViEModuleId(engine_id_, -1),
-                                                  window, false);
-    if (!render_module)
-      return NULL;
-
-    render_list_.push_back(render_module);
-  }
-
-  ViERenderer* vie_renderer = ViERenderer::CreateViERenderer(render_id,
-                                                             engine_id_,
-                                                             *render_module,
-                                                             *this, z_order,
-                                                             left, top, right,
-                                                             bottom);
-  if (!vie_renderer)
-    return NULL;
-
-  stream_to_vie_renderer_[render_id] = vie_renderer;
-  return vie_renderer;
-}
-
-int32_t ViERenderManager::RemoveRenderStream(
-    const int32_t render_id) {
-  // We need exclusive right to the items in the render manager to delete a
-  // stream.
-  ViEManagerWriteScoped scope(this);
-  CriticalSectionScoped cs(list_cs_.get());
-  RendererMap::iterator it = stream_to_vie_renderer_.find(render_id);
-  if (it == stream_to_vie_renderer_.end()) {
-    LOG(LS_ERROR) << "No renderer found for render_id: " << render_id;
-    return 0;
-  }
-
-  // Get the render module pointer for this vie_render object.
-  VideoRender& renderer = it->second->RenderModule();
-
-  // Delete the vie_render.
-  // This deletes the stream in the render module.
-  delete it->second;
-
-  // Remove from the stream map.
-  stream_to_vie_renderer_.erase(it);
-
-  // Check if there are other streams in the module.
-  if (!use_external_render_module_ &&
-      renderer.GetNumIncomingRenderStreams() == 0) {
-    // Erase the render module from the map.
-    for (RenderList::iterator iter = render_list_.begin();
-         iter != render_list_.end(); ++iter) {
-      if (&renderer == *iter) {
-        // We've found our renderer.
-        render_list_.erase(iter);
-        break;
-      }
-    }
-    // Destroy the module.
-    VideoRender::DestroyVideoRender(&renderer);
-  }
-  return 0;
-}
-
-VideoRender* ViERenderManager::FindRenderModule(void* window) {
-  for (RenderList::iterator iter = render_list_.begin();
-       iter != render_list_.end(); ++iter) {
-    if ((*iter)->Window() == window) {
-      // We've found the render module.
-      return *iter;
-    }
-  }
-  return NULL;
-}
-
-ViERenderer* ViERenderManager::ViERenderPtr(int32_t render_id) const {
-  RendererMap::const_iterator it = stream_to_vie_renderer_.find(render_id);
-  if (it == stream_to_vie_renderer_.end())
-    return NULL;
-
-  return it->second;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h b/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h
deleted file mode 100644
index db1626a1e0..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
-
-#include <list>
-#include <map>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RWLockWrapper;
-class VideoRender;
-class VideoRenderCallback;
-class ViERenderer;
-
-class ViERenderManager : private ViEManagerBase {
-  friend class ViERenderManagerScoped;
- public:
-  explicit ViERenderManager(int32_t engine_id);
-  ~ViERenderManager();
-
-  int32_t RegisterVideoRenderModule(VideoRender* render_module);
-  int32_t DeRegisterVideoRenderModule(VideoRender* render_module);
-
-  ViERenderer* AddRenderStream(const int32_t render_id,
-                               void* window,
-                               const uint32_t z_order,
-                               const float left,
-                               const float top,
-                               const float right,
-                               const float bottom);
-
-  int32_t RemoveRenderStream(int32_t render_id);
-
- private:
-  typedef std::list<VideoRender*> RenderList;
-  // Returns a pointer to the render module if it exists in the render list.
-  // Assumed protected.
-  VideoRender* FindRenderModule(void* window);
-
-  // Methods used by ViERenderScoped.
-  ViERenderer* ViERenderPtr(int32_t render_id) const;
-
-  rtc::scoped_ptr<CriticalSectionWrapper> list_cs_;
-  int32_t engine_id_;
-  // Protected by ViEManagerBase.
-  typedef std::map<int32_t, ViERenderer*> RendererMap;
-  RendererMap stream_to_vie_renderer_;
-  RenderList render_list_;
-  bool use_external_render_module_;
-};
-
-class ViERenderManagerScoped: private ViEManagerScopedBase {
- public:
-  explicit ViERenderManagerScoped(const ViERenderManager& vie_render_manager);
-
-  // Returns a pointer to the ViERender object.
-  ViERenderer* Renderer(int32_t render_id) const;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc b/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc
deleted file mode 100644
index bfeeba216a..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_renderer.h"
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/include/video_render.h"
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/video_engine/vie_render_manager.h"
-
-namespace webrtc {
-
-ViERenderer* ViERenderer::CreateViERenderer(const int32_t render_id,
-                                            const int32_t engine_id,
-                                            VideoRender& render_module,
-                                            ViERenderManager& render_manager,
-                                            const uint32_t z_order,
-                                            const float left,
-                                            const float top,
-                                            const float right,
-                                            const float bottom) {
-  ViERenderer* self = new ViERenderer(render_id, engine_id, render_module,
-                                      render_manager);
-  if (!self || self->Init(z_order, left, top, right, bottom) != 0) {
-    delete self;
-    self = NULL;
-  }
-  return self;
-}
-
-ViERenderer::~ViERenderer(void) {
-  if (render_callback_)
-    render_module_.DeleteIncomingRenderStream(render_id_);
-
-  delete incoming_external_callback_;
-}
-
-int32_t ViERenderer::StartRender() {
-  return render_module_.StartRender(render_id_);
-}
-int32_t ViERenderer::StopRender() {
-  return render_module_.StopRender(render_id_);
-}
-
-int ViERenderer::SetExpectedRenderDelay(int render_delay) {
-  return render_module_.SetExpectedRenderDelay(render_id_, render_delay);
-}
-
-int32_t ViERenderer::ConfigureRenderer(const unsigned int z_order,
-                                       const float left,
-                                       const float top,
-                                       const float right,
-                                       const float bottom) {
-  return render_module_.ConfigureRenderer(render_id_, z_order, left, top, right,
-                                          bottom);
-}
-
-VideoRender& ViERenderer::RenderModule() {
-  return render_module_;
-}
-
-int32_t ViERenderer::SetTimeoutImage(const I420VideoFrame& timeout_image,
-                                     const int32_t timeout_value) {
-  return render_module_.SetTimeoutImage(render_id_, timeout_image,
-                                        timeout_value);
-}
-
-int32_t  ViERenderer::SetRenderStartImage(
-    const I420VideoFrame& start_image) {
-  return render_module_.SetStartImage(render_id_, start_image);
-}
-
-int32_t ViERenderer::SetExternalRenderer(
-    const int32_t render_id,
-    RawVideoType video_input_format,
-    ExternalRenderer* external_renderer) {
-  if (!incoming_external_callback_)
-    return -1;
-
-  incoming_external_callback_->SetViEExternalRenderer(external_renderer,
-                                                      video_input_format);
-  return render_module_.AddExternalRenderCallback(render_id,
-                                                  incoming_external_callback_);
-}
-
-int32_t ViERenderer::SetVideoRenderCallback(int32_t render_id,
-                                            VideoRenderCallback* callback) {
-  return render_module_.AddExternalRenderCallback(render_id, callback);
-}
-
-ViERenderer::ViERenderer(const int32_t render_id,
-                         const int32_t engine_id,
-                         VideoRender& render_module,
-                         ViERenderManager& render_manager)
-    : render_id_(render_id),
-      render_module_(render_module),
-      render_manager_(render_manager),
-      render_callback_(NULL),
-      incoming_external_callback_(new ViEExternalRendererImpl()) {
-}
-
-int32_t ViERenderer::Init(const uint32_t z_order,
-                          const float left,
-                          const float top,
-                          const float right,
-                          const float bottom) {
-  render_callback_ =
-      static_cast<VideoRenderCallback*>(render_module_.AddIncomingRenderStream(
-          render_id_, z_order, left, top, right, bottom));
-  if (!render_callback_) {
-    // Logging done.
-    return -1;
-  }
-  return 0;
-}
-
-void ViERenderer::DeliverFrame(int id,
-                               I420VideoFrame* video_frame,
-                               const std::vector<uint32_t>& csrcs) {
-  render_callback_->RenderFrame(render_id_, *video_frame);
-}
-
-void ViERenderer::DelayChanged(int id, int frame_delay) {}
-
-int ViERenderer::GetPreferedFrameSettings(int* width,
-                                          int* height,
-                                          int* frame_rate) {
-    return -1;
-}
-
-void ViERenderer::ProviderDestroyed(int id) {
-  // Remove the render stream since the provider is destroyed.
-  render_manager_.RemoveRenderStream(render_id_);
-}
-
-ViEExternalRendererImpl::ViEExternalRendererImpl()
-    : external_renderer_(NULL),
-      external_renderer_format_(kVideoUnknown),
-      external_renderer_width_(0),
-      external_renderer_height_(0) {
-}
-
-int ViEExternalRendererImpl::SetViEExternalRenderer(
-    ExternalRenderer* external_renderer,
-    RawVideoType video_input_format) {
-  external_renderer_ = external_renderer;
-  external_renderer_format_ = video_input_format;
-  return 0;
-}
-
-int32_t ViEExternalRendererImpl::RenderFrame(
-    const uint32_t stream_id,
-    const I420VideoFrame& video_frame) {
-  if (external_renderer_format_ != kVideoI420)
-    return ConvertAndRenderFrame(stream_id, video_frame);
-
-  // Fast path for I420 without frame copy.
-  NotifyFrameSizeChange(stream_id, video_frame);
-  if (video_frame.native_handle() == NULL ||
-      external_renderer_->IsTextureSupported()) {
-    external_renderer_->DeliverI420Frame(video_frame);
-  } else {
-    // TODO(wuchengli): readback the pixels and deliver the frame.
-  }
-  return 0;
-}
-
-int32_t ViEExternalRendererImpl::ConvertAndRenderFrame(
-    uint32_t stream_id,
-    const I420VideoFrame& video_frame) {
-  if (video_frame.native_handle() != NULL) {
-    NotifyFrameSizeChange(stream_id, video_frame);
-
-    if (external_renderer_->IsTextureSupported()) {
-      external_renderer_->DeliverFrame(NULL,
-                                       0,
-                                       video_frame.timestamp(),
-                                       video_frame.ntp_time_ms(),
-                                       video_frame.render_time_ms(),
-                                       video_frame.native_handle());
-    } else {
-      // TODO(wuchengli): readback the pixels and deliver the frame.
-    }
-    return 0;
-  }
-
-  // Convert to requested format.
-  VideoType type =
-      RawVideoTypeToCommonVideoVideoType(external_renderer_format_);
-  size_t buffer_size = CalcBufferSize(type, video_frame.width(),
-                                      video_frame.height());
-  if (buffer_size == 0) {
-    // Unsupported video format.
-    assert(false);
-    return -1;
-  }
-  converted_frame_.resize(buffer_size);
-  uint8_t* out_frame = &converted_frame_[0];
-
-  switch (external_renderer_format_) {
-    case kVideoYV12:
-    case kVideoYUY2:
-    case kVideoUYVY:
-    case kVideoARGB:
-    case kVideoRGB24:
-    case kVideoRGB565:
-    case kVideoARGB4444:
-    case kVideoARGB1555:
-      if (ConvertFromI420(video_frame, type, 0, out_frame) < 0)
-        return -1;
-      break;
-    case kVideoIYUV:
-      // no conversion available
-      break;
-    default:
-      assert(false);
-      out_frame = NULL;
-      break;
-  }
-
-  NotifyFrameSizeChange(stream_id, video_frame);
-
-  if (out_frame) {
-    external_renderer_->DeliverFrame(out_frame,
-                                     converted_frame_.size(),
-                                     video_frame.timestamp(),
-                                     video_frame.ntp_time_ms(),
-                                     video_frame.render_time_ms(),
-                                     NULL);
-  }
-  return 0;
-}
-
-void ViEExternalRendererImpl::NotifyFrameSizeChange(
-    const uint32_t stream_id,
-    const I420VideoFrame& video_frame) {
-  if (external_renderer_width_ != video_frame.width() ||
-      external_renderer_height_ != video_frame.height()) {
-    external_renderer_width_ = video_frame.width();
-    external_renderer_height_ = video_frame.height();
-    external_renderer_->FrameSizeChange(
-        external_renderer_width_, external_renderer_height_, stream_id);
-  }
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h b/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h
deleted file mode 100644
index 5a7fdc19f1..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
-
-#include <vector>
-
-#include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/video_engine/include/vie_render.h"
-#include "webrtc/video_engine/vie_frame_provider_base.h"
-
-namespace webrtc {
-
-class VideoRender;
-class VideoRenderCallback;
-class ViERenderManager;
-
-class ViEExternalRendererImpl : public VideoRenderCallback {
- public:
-  ViEExternalRendererImpl();
-  virtual ~ViEExternalRendererImpl() {}
-
-  int SetViEExternalRenderer(ExternalRenderer* external_renderer,
-                             RawVideoType video_input_format);
-
-  // Implements VideoRenderCallback.
-  virtual int32_t RenderFrame(const uint32_t stream_id,
-                              const I420VideoFrame& video_frame);
-
- private:
-  void NotifyFrameSizeChange(const uint32_t stream_id,
-                             const I420VideoFrame& video_frame);
-  int32_t ConvertAndRenderFrame(uint32_t stream_id,
-                                const I420VideoFrame& video_frame);
-  ExternalRenderer* external_renderer_;
-  RawVideoType external_renderer_format_;
-  int external_renderer_width_;
-  int external_renderer_height_;
-  // Converted_frame_ in color format specified by render_format_.
-  std::vector<uint8_t> converted_frame_;
-};
-
-class ViERenderer: public ViEFrameCallback {
- public:
-  static ViERenderer* CreateViERenderer(const int32_t render_id,
-                                        const int32_t engine_id,
-                                        VideoRender& render_module,
-                                        ViERenderManager& render_manager,
-                                        const uint32_t z_order,
-                                        const float left,
-                                        const float top,
-                                        const float right,
-                                        const float bottom);
-  ~ViERenderer(void);
-
-  int32_t StartRender();
-  int32_t StopRender();
-
-  int SetExpectedRenderDelay(int render_delay);
-
-  int32_t ConfigureRenderer(const unsigned int z_order,
-                            const float left,
-                            const float top,
-                            const float right,
-                            const float bottom);
-
-  VideoRender& RenderModule();
-
-  int32_t SetTimeoutImage(const I420VideoFrame& timeout_image,
-                          const int32_t timeout_value);
-  int32_t SetRenderStartImage(const I420VideoFrame& start_image);
-  int32_t SetExternalRenderer(const int32_t render_id,
-                              RawVideoType video_input_format,
-                              ExternalRenderer* external_renderer);
-
-  int32_t SetVideoRenderCallback(const int32_t render_id,
-                                 VideoRenderCallback* callback);
-
- private:
-  ViERenderer(const int32_t render_id, const int32_t engine_id,
-                VideoRender& render_module,
-                ViERenderManager& render_manager);
-
-  int32_t Init(const uint32_t z_order,
-               const float left,
-               const float top,
-               const float right,
-               const float bottom);
-
-  // Implement ViEFrameCallback
-  virtual void DeliverFrame(int id,
-                            I420VideoFrame* video_frame,
-                            const std::vector<uint32_t>& csrcs);
-  virtual void DelayChanged(int id, int frame_delay);
-  virtual int GetPreferedFrameSettings(int* width,
-                                       int* height,
-                                       int* frame_rate);
-  virtual void ProviderDestroyed(int id);
-
-  uint32_t render_id_;
-  VideoRender& render_module_;
-  ViERenderManager& render_manager_;
-  VideoRenderCallback* render_callback_;
-  ViEExternalRendererImpl* incoming_external_callback_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc
deleted file mode 100644
index 27a61fe95e..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc
+++ /dev/null
@@ -1,1171 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_rtp_rtcp_impl.h"
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/video_engine/include/vie_errors.h"
-#include "webrtc/video_engine/vie_channel.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_impl.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-// Helper methods for converting between module format and ViE API format.
-
-static RTCPMethod ViERTCPModeToRTCPMethod(ViERTCPMode api_mode) {
-  switch (api_mode) {
-    case kRtcpNone:
-      return kRtcpOff;
-
-    case kRtcpCompound_RFC4585:
-      return kRtcpCompound;
-
-    case kRtcpNonCompound_RFC5506:
-      return kRtcpNonCompound;
-  }
-  assert(false);
-  return kRtcpOff;
-}
-
-static ViERTCPMode RTCPMethodToViERTCPMode(RTCPMethod module_method) {
-  switch (module_method) {
-    case kRtcpOff:
-      return kRtcpNone;
-
-    case kRtcpCompound:
-      return kRtcpCompound_RFC4585;
-
-    case kRtcpNonCompound:
-      return kRtcpNonCompound_RFC5506;
-  }
-  assert(false);
-  return kRtcpNone;
-}
-
-static KeyFrameRequestMethod APIRequestToModuleRequest(
-  ViEKeyFrameRequestMethod api_method) {
-  switch (api_method) {
-    case kViEKeyFrameRequestNone:
-      return kKeyFrameReqFirRtp;
-
-    case kViEKeyFrameRequestPliRtcp:
-      return kKeyFrameReqPliRtcp;
-
-    case kViEKeyFrameRequestFirRtp:
-      return kKeyFrameReqFirRtp;
-
-    case kViEKeyFrameRequestFirRtcp:
-      return kKeyFrameReqFirRtcp;
-  }
-  assert(false);
-  return kKeyFrameReqFirRtp;
-}
-
-ViERTP_RTCP* ViERTP_RTCP::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-  if (!video_engine) {
-    return NULL;
-  }
-  VideoEngineImpl* vie_impl = static_cast<VideoEngineImpl*>(video_engine);
-  ViERTP_RTCPImpl* vie_rtpimpl = vie_impl;
-  // Increase ref count.
-  (*vie_rtpimpl)++;
-  return vie_rtpimpl;
-#else
-  return NULL;
-#endif
-}
-
-int ViERTP_RTCPImpl::Release() {
-  // Decrease ref count.
-  (*this)--;
-
-  int32_t ref_count = GetCount();
-  if (ref_count < 0) {
-    LOG(LS_ERROR) << "ViERTP_RTCP released too many times.";
-    shared_data_->SetLastError(kViEAPIDoesNotExist);
-    return -1;
-  }
-  return ref_count;
-}
-
-ViERTP_RTCPImpl::ViERTP_RTCPImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {}
-
-ViERTP_RTCPImpl::~ViERTP_RTCPImpl() {}
-
-int ViERTP_RTCPImpl::SetLocalSSRC(const int video_channel,
-                                  const unsigned int SSRC,
-                                  const StreamType usage,
-                                  const unsigned char simulcast_idx) {
-  LOG_F(LS_INFO) << "channel: " << video_channel << " ssrc: " << SSRC << "";
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetSSRC(SSRC, usage, simulcast_idx) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetRemoteSSRCType(const int videoChannel,
-                                       const StreamType usage,
-                                       const unsigned int SSRC) const {
-  LOG_F(LS_INFO) << "channel: " << videoChannel
-                 << " usage: " << static_cast<int>(usage) << " ssrc: " << SSRC;
-
-  // Get the channel
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* ptrViEChannel = cs.Channel(videoChannel);
-  if (ptrViEChannel == NULL) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (ptrViEChannel->SetRemoteSSRCType(usage, SSRC) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetLocalSSRC(const int video_channel,
-                                  unsigned int& SSRC) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  uint8_t idx = 0;
-  if (vie_channel->GetLocalSSRC(idx, &SSRC) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteSSRC(const int video_channel,
-                                   unsigned int& SSRC) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRemoteSSRC(&SSRC) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteCSRCs(const int video_channel,
-                                    unsigned int CSRCs[kRtpCsrcSize]) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRemoteCSRC(CSRCs) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteRID(const int video_channel,
-                                  char rid[256]) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRemoteRID(rid) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetRtxSendPayloadType(const int video_channel,
-                                           const uint8_t payload_type) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " payload_type: " << static_cast<int>(payload_type);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetRtxSendPayloadType(payload_type) != 0) {
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetRtxReceivePayloadType(const int video_channel,
-                                              const uint8_t payload_type) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " payload_type: " << static_cast<int>(payload_type);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->SetRtxReceivePayloadType(payload_type);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetStartSequenceNumber(const int video_channel,
-                                            uint16_t sequence_number) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->Sending()) {
-    LOG_F(LS_ERROR) << "channel " << video_channel << " is already sending.";
-    shared_data_->SetLastError(kViERtpRtcpAlreadySending);
-    return -1;
-  }
-  if (vie_channel->SetStartSequenceNumber(sequence_number) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-void ViERTP_RTCPImpl::SetRtpStateForSsrc(int video_channel,
-                                         uint32_t ssrc,
-                                         const RtpState& rtp_state) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel)
-    return;
-
-  if (vie_channel->Sending()) {
-    LOG_F(LS_ERROR) << "channel " << video_channel << " is already sending.";
-    return;
-  }
-  vie_channel->SetRtpStateForSsrc(ssrc, rtp_state);
-}
-
-RtpState ViERTP_RTCPImpl::GetRtpStateForSsrc(int video_channel, uint32_t ssrc) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel)
-    return RtpState();
-
-  return vie_channel->GetRtpStateForSsrc(ssrc);
-}
-
-int ViERTP_RTCPImpl::SetRTCPStatus(const int video_channel,
-                                   const ViERTCPMode rtcp_mode) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " mode: " << static_cast<int>(rtcp_mode);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-
-  RTCPMethod module_mode = ViERTCPModeToRTCPMethod(rtcp_mode);
-  vie_channel->SetRTCPMode(module_mode);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRTCPStatus(const int video_channel,
-                                   ViERTCPMode& rtcp_mode) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  RTCPMethod module_mode = vie_channel->GetRTCPMode();
-  rtcp_mode = RTCPMethodToViERTCPMode(module_mode);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetRTCPCName(const int video_channel,
-                                  const char rtcp_cname[KMaxRTCPCNameLength]) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " rtcp_cname: " << rtcp_cname;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->Sending()) {
-    LOG_F(LS_ERROR) << "channel " << video_channel << " is already sending.";
-    shared_data_->SetLastError(kViERtpRtcpAlreadySending);
-    return -1;
-  }
-  if (vie_channel->SetRTCPCName(rtcp_cname) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteRTCPCName(
-    const int video_channel,
-    char rtcp_cname[KMaxRTCPCNameLength]) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRemoteRTCPCName(rtcp_cname) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteRTCPReceiverInfo(const int video_channel,
-                                               uint32_t& NTPHigh,
-                                               uint32_t& NTPLow,
-                                               uint32_t& receivedPacketCount,
-                                               uint64_t& receivedOctetCount,
-                                               uint32_t* jitter,
-                                               uint16_t* fractionLost,
-                                               uint32_t* cumulativeLost,
-                                               int32_t* rttMs) const {
-  LOG_F(LS_INFO) << "channel:" << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    LOG(LS_ERROR) << "Channel " << video_channel << " doesn't exist";
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRemoteRTCPReceiverInfo(NTPHigh,
-                                             NTPLow,
-                                             receivedPacketCount,
-                                             receivedOctetCount,
-                                             jitter,
-                                             fractionLost,
-                                             cumulativeLost,
-                                             rttMs) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
-  const int video_channel,
-  const unsigned char sub_type,
-  unsigned int name,
-  const char* data,
-  uint16_t data_length_in_bytes) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (!vie_channel->Sending()) {
-    shared_data_->SetLastError(kViERtpRtcpNotSending);
-    return -1;
-  }
-  RTCPMethod method = vie_channel->GetRTCPMode();
-  if (method == kRtcpOff) {
-    shared_data_->SetLastError(kViERtpRtcpRtcpDisabled);
-    return -1;
-  }
-  if (vie_channel->SendApplicationDefinedRTCPPacket(
-        sub_type, name, reinterpret_cast<const uint8_t*>(data),
-        data_length_in_bytes) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetNACKStatus(const int video_channel, const bool enable) {
-  LOG_F(LS_INFO) << "channel: " << video_channel << " "
-                 << (enable ? "on" : "off");
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetNACKStatus(enable) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-
-  // Update the encoder
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  vie_encoder->UpdateProtectionMethod(enable,
-                                      vie_channel->IsSendingFecEnabled());
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetFECStatus(const int video_channel, const bool enable,
-                                  const unsigned char payload_typeRED,
-                                  const unsigned char payload_typeFEC) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off")
-                 << " payload_typeRED: " << static_cast<int>(payload_typeRED)
-                 << " payload_typeFEC: " << static_cast<int>(payload_typeFEC);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetFECStatus(enable, payload_typeRED,
-                                payload_typeFEC) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  // Update the encoder.
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  vie_encoder->UpdateProtectionMethod(false, true);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetHybridNACKFECStatus(
-    const int video_channel,
-    const bool enable,
-    const unsigned char payload_typeRED,
-    const unsigned char payload_typeFEC) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off")
-                 << " payload_typeRED: " << static_cast<int>(payload_typeRED)
-                 << " payload_typeFEC: " << static_cast<int>(payload_typeFEC);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-
-  // Update the channel status with hybrid NACK FEC mode.
-  if (vie_channel->SetHybridNACKFECStatus(enable, payload_typeRED,
-                                          payload_typeFEC) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-
-  // Update the encoder.
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  vie_encoder->UpdateProtectionMethod(enable, enable);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetSenderBufferingMode(int video_channel,
-                                            int target_delay_ms) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " target_delay_ms: " << target_delay_ms;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-
-  // Update the channel with buffering mode settings.
-  if (vie_channel->SetSenderBufferingMode(target_delay_ms) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-
-  // Update the encoder's buffering mode settings.
-  vie_encoder->SetSenderBufferingMode(target_delay_ms);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetReceiverBufferingMode(int video_channel,
-                                              int target_delay_ms) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " target_delay_ms: " << target_delay_ms;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-
-  // Update the channel with buffering mode settings.
-  if (vie_channel->SetReceiverBufferingMode(target_delay_ms) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetKeyFrameRequestMethod(
-  const int video_channel,
-  const ViEKeyFrameRequestMethod method) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " method: " << static_cast<int>(method);
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  KeyFrameRequestMethod module_method = APIRequestToModuleRequest(method);
-  if (vie_channel->SetKeyFrameRequestMethod(module_method) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetTMMBRStatus(const int video_channel,
-                                    const bool enable) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << "enable: " << (enable ? "on" : "off");
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->EnableTMMBR(enable);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetRembStatus(int video_channel,
-                                   bool sender,
-                                   bool receiver) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " sender: " << (sender ? "on" : "off")
-                 << " receiver: " << (receiver ? "on" : "off");
-  if (!shared_data_->channel_manager()->SetRembStatus(video_channel, sender,
-                                                      receiver)) {
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetSendTimestampOffsetStatus(int video_channel,
-                                                  bool enable,
-                                                  int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << "enable: " << (enable ? "on" : "off") << " id: " << id;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetSendTimestampOffsetStatus(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetReceiveTimestampOffsetStatus(int video_channel,
-                                                     bool enable,
-                                                     int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetReceiveTimestampOffsetStatus(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetSendAbsoluteSendTimeStatus(int video_channel,
-                                                   bool enable,
-                                                   int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetSendAbsoluteSendTimeStatus(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(int video_channel,
-                                                      bool enable,
-                                                      int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetReceiveAbsoluteSendTimeStatus(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetSendVideoRotationStatus(int video_channel,
-                                                bool enable,
-                                                int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetSendVideoRotationStatus(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetReceiveVideoRotationStatus(int video_channel,
-                                                   bool enable,
-                                                   int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetReceiveVideoRotationStatus(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetSendRIDStatus(int video_channel,
-                                      bool enable,
-                                      int id,
-                                      const char *rid) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id << " RID: " << rid;
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetSendRtpStreamId(enable, id, rid) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetReceiveRIDStatus(int video_channel,
-                                         bool enable,
-                                         int id) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->SetReceiveRtpStreamId(enable, id) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetRtcpXrRrtrStatus(int video_channel, bool enable) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off");
-
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->SetRtcpXrRrtrStatus(enable);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetTransmissionSmoothingStatus(int video_channel,
-                                                    bool enable) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " enable: " << (enable ? "on" : "off");
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->SetTransmissionSmoothingStatus(enable);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetMinTransmitBitrate(int video_channel,
-                                           int min_transmit_bitrate_kbps) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " min_transmit_bitrate_kbps: " << min_transmit_bitrate_kbps;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (vie_encoder == NULL)
-    return -1;
-  vie_encoder->SetMinTransmitBitrate(min_transmit_bitrate_kbps);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::SetReservedTransmitBitrate(
-    int video_channel, unsigned int reserved_transmit_bitrate_bps) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " reserved_transmit_bitrate_bps: "
-                 << reserved_transmit_bitrate_bps;
-  if (!shared_data_->channel_manager()->SetReservedTransmitBitrate(
-      video_channel, reserved_transmit_bitrate_bps)) {
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetReceiveChannelRtcpStatistics(
-    const int video_channel,
-    RtcpStatistics& basic_stats,
-    int64_t& rtt_ms) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-
-  // TODO(sprang): Clean this up when stats struct is propagated all the way.
-  uint16_t frac_lost;
-  if (vie_channel->GetReceivedRtcpStatistics(
-          &frac_lost,
-          &basic_stats.cumulative_lost,
-          &basic_stats.extended_max_sequence_number,
-          &basic_stats.jitter,
-          &rtt_ms) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  basic_stats.fraction_lost = frac_lost;
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetSendChannelRtcpStatistics(const int video_channel,
-                                                  RtcpStatistics& basic_stats,
-                                                  int64_t& rtt_ms) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-
-  // TODO(sprang): Clean this up when stats struct is propagated all the way.
-  uint16_t frac_lost;
-  if (vie_channel->GetSendRtcpStatistics(
-          &frac_lost,
-          &basic_stats.cumulative_lost,
-          &basic_stats.extended_max_sequence_number,
-          &basic_stats.jitter,
-          &rtt_ms) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  basic_stats.fraction_lost = frac_lost;
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRtpStatistics(const int video_channel,
-                                      StreamDataCounters& sent,
-                                      StreamDataCounters& received) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRtpStatistics(&sent.transmitted.payload_bytes,
-                                    &sent.transmitted.packets,
-                                    &received.transmitted.payload_bytes,
-                                    &received.transmitted.packets) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetSendRtcpPacketTypeCounter(
-    int video_channel,
-    RtcpPacketTypeCounter* packet_counter) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->GetSendRtcpPacketTypeCounter(packet_counter);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetReceiveRtcpPacketTypeCounter(
-    int video_channel,
-    RtcpPacketTypeCounter* packet_counter) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->GetReceiveRtcpPacketTypeCounter(packet_counter);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteRTCPSenderInfo(const int video_channel,
-                                             SenderInfo* sender_info) const {
-  LOG_F(LS_INFO) << "channel:" << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    LOG(LS_ERROR) << "Channel " << video_channel << " doesn't exist";
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->GetRemoteRTCPSenderInfo(sender_info) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetBandwidthUsage(const int video_channel,
-                                       unsigned int& total_bitrate_sent,
-                                       unsigned int& video_bitrate_sent,
-                                       unsigned int& fec_bitrate_sent,
-                                       unsigned int& nackBitrateSent) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->GetBandwidthUsage(&total_bitrate_sent,
-                                 &video_bitrate_sent,
-                                 &fec_bitrate_sent,
-                                 &nackBitrateSent);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetEstimatedSendBandwidth(
-    const int video_channel,
-    unsigned int* estimated_bandwidth) const {
-  if (!shared_data_->channel_manager()->GetEstimatedSendBandwidth(
-      video_channel, estimated_bandwidth)) {
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetEstimatedReceiveBandwidth(
-    const int video_channel,
-    unsigned int* estimated_bandwidth) const {
-  if (!shared_data_->channel_manager()->GetEstimatedReceiveBandwidth(
-      video_channel, estimated_bandwidth)) {
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::GetPacerQueuingDelayMs(
-    const int video_channel, int64_t* delay_ms) const {
-  if (!shared_data_->channel_manager()->GetPacerQueuingDelayMs(video_channel,
-                                                               delay_ms)) {
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::StartRTPDump(const int video_channel,
-                                  const char file_nameUTF8[1024],
-                                  RTPDirections direction) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " filename: " << file_nameUTF8
-                 << " direction: " << static_cast<int>(direction);
-  assert(FileWrapper::kMaxFileNameSize == 1024);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->StartRTPDump(file_nameUTF8, direction) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::StopRTPDump(const int video_channel,
-                                 RTPDirections direction) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " direction: " << static_cast<int>(direction);
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->StopRTPDump(direction) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterRTPObserver(const int video_channel,
-                                         ViERTPObserver& observer) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterRtpObserver(&observer) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterRTPObserver(const int video_channel) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterRtpObserver(NULL) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterSendChannelRtcpStatisticsCallback(
-  int video_channel, RtcpStatisticsCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->RegisterSendChannelRtcpStatisticsCallback(callback);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterSendChannelRtcpStatisticsCallback(
-    int video_channel, RtcpStatisticsCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->RegisterSendChannelRtcpStatisticsCallback(NULL);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterReceiveChannelRtcpStatisticsCallback(
-    const int video_channel,
-    RtcpStatisticsCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterReceiveChannelRtcpStatisticsCallback(callback);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterReceiveChannelRtcpStatisticsCallback(
-    const int video_channel,
-    RtcpStatisticsCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterReceiveChannelRtcpStatisticsCallback(NULL);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterSendChannelRtpStatisticsCallback(
-    int video_channel, StreamDataCountersCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterSendChannelRtpStatisticsCallback(callback);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterSendChannelRtpStatisticsCallback(
-    int video_channel, StreamDataCountersCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterSendChannelRtpStatisticsCallback(NULL);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterReceiveChannelRtpStatisticsCallback(
-    const int video_channel,
-    StreamDataCountersCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterReceiveChannelRtpStatisticsCallback(callback);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterReceiveChannelRtpStatisticsCallback(
-    const int video_channel,
-    StreamDataCountersCallback* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterReceiveChannelRtpStatisticsCallback(NULL);
-  return 0;
-}
-
-// Called whenever the send bitrate is updated.
-int ViERTP_RTCPImpl::RegisterSendBitrateObserver(
-    const int video_channel,
-    BitrateStatisticsObserver* observer) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterSendBitrateObserver(observer);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterSendBitrateObserver(
-    const int video_channel,
-    BitrateStatisticsObserver* observer) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  assert(vie_channel != NULL);
-  vie_channel->RegisterSendBitrateObserver(NULL);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterSendFrameCountObserver(
-    int video_channel, FrameCountObserver* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->RegisterSendFrameCountObserver(callback);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterSendFrameCountObserver(
-    int video_channel, FrameCountObserver* callback) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->RegisterSendFrameCountObserver(NULL);
-  return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterRtcpPacketTypeCounterObserver(
-    int video_channel,
-    RtcpPacketTypeCounterObserver* observer) {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->RegisterRtcpPacketTypeCounterObserver(observer);
-  return 0;
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h
deleted file mode 100644
index e1ac4f8cbc..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViERTP_RTCPImpl
-    : public ViERTP_RTCP,
-      public ViERefCount {
- public:
-  // Implements ViERTP_RTCP.
-  virtual int Release();
-  virtual int SetLocalSSRC(const int video_channel,
-                           const unsigned int SSRC,
-                           const StreamType usage,
-                           const unsigned char simulcast_idx);
-  virtual int GetLocalSSRC(const int video_channel,
-                           unsigned int& SSRC) const;  // NOLINT
-  virtual int SetRemoteSSRCType(const int video_channel,
-                                const StreamType usage,
-                                const unsigned int SSRC) const;
-  virtual int GetRemoteSSRC(const int video_channel,
-                            unsigned int& SSRC) const;  // NOLINT
-  virtual int GetRemoteCSRCs(const int video_channel,
-                             unsigned int CSRCs[kRtpCsrcSize]) const;
-  virtual int GetRemoteRID(const int video_channel,
-                           char rid[256]) const;
-  virtual int SetRtxSendPayloadType(const int video_channel,
-                                    const uint8_t payload_type);
-  virtual int SetRtxReceivePayloadType(const int video_channel,
-                                       const uint8_t payload_type);
-  virtual int SetStartSequenceNumber(const int video_channel,
-                                     uint16_t sequence_number);
-  void SetRtpStateForSsrc(int video_channel,
-                          uint32_t ssrc,
-                          const RtpState& rtp_state) override;
-  RtpState GetRtpStateForSsrc(int video_channel, uint32_t ssrc) override;
-  virtual int SetRTCPStatus(const int video_channel,
-                            const ViERTCPMode rtcp_mode);
-  virtual int GetRTCPStatus(const int video_channel,
-                            ViERTCPMode& rtcp_mode) const;
-  virtual int SetRTCPCName(const int video_channel,
-                           const char rtcp_cname[KMaxRTCPCNameLength]);
-  virtual int GetRemoteRTCPCName(const int video_channel,
-                                 char rtcp_cname[KMaxRTCPCNameLength]) const;
-  virtual int GetRemoteRTCPReceiverInfo(const int video_channel,
-                                        uint32_t& NTPHigh,
-                                        uint32_t& NTPLow,
-                                        uint32_t& receivedPacketCount,
-                                        uint64_t& receivedOctetCount,
-                                        uint32_t* jitter,
-                                        uint16_t* fractionLost,
-                                        uint32_t* cumulativeLost,
-                                        int32_t* rttMs) const;
-  virtual int SendApplicationDefinedRTCPPacket(
-      const int video_channel,
-      const unsigned char sub_type,
-      unsigned int name,
-      const char* data,
-      uint16_t data_length_in_bytes);
-  virtual int SetNACKStatus(const int video_channel, const bool enable);
-  virtual int SetFECStatus(const int video_channel, const bool enable,
-                           const unsigned char payload_typeRED,
-                           const unsigned char payload_typeFEC);
-  virtual int SetHybridNACKFECStatus(const int video_channel, const bool enable,
-                                     const unsigned char payload_typeRED,
-                                     const unsigned char payload_typeFEC);
-  virtual int SetSenderBufferingMode(int video_channel,
-                                     int target_delay_ms);
-  virtual int SetReceiverBufferingMode(int video_channel,
-                                       int target_delay_ms);
-  virtual int SetKeyFrameRequestMethod(const int video_channel,
-                                       const ViEKeyFrameRequestMethod method);
-  virtual int SetTMMBRStatus(const int video_channel, const bool enable);
-  virtual int SetRembStatus(int video_channel, bool sender, bool receiver);
-  virtual int SetSendTimestampOffsetStatus(int video_channel,
-                                           bool enable,
-                                           int id);
-  virtual int SetReceiveTimestampOffsetStatus(int video_channel,
-                                              bool enable,
-                                              int id);
-  virtual int SetSendAbsoluteSendTimeStatus(int video_channel,
-                                            bool enable,
-                                            int id);
-  virtual int SetReceiveAbsoluteSendTimeStatus(int video_channel,
-                                               bool enable,
-                                               int id);
-  virtual int SetSendVideoRotationStatus(int video_channel,
-                                         bool enable,
-                                         int id);
-  virtual int SetReceiveVideoRotationStatus(int video_channel,
-                                            bool enable,
-                                            int id);
-  virtual int SetSendRIDStatus(int video_channel,
-                               bool enable,
-                               int id,
-                               const char *rid);
-  virtual int SetReceiveRIDStatus(int video_channel,
-                                  bool enable,
-                                  int id);
-  virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable);
-  virtual int SetTransmissionSmoothingStatus(int video_channel, bool enable);
-  virtual int SetMinTransmitBitrate(int video_channel,
-                                    int min_transmit_bitrate_kbps);
-  virtual int SetReservedTransmitBitrate(
-      int video_channel, unsigned int reserved_transmit_bitrate_bps);
-  virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
-                                              RtcpStatistics& basic_stats,
-                                              int64_t& rtt_ms) const;
-  virtual int GetSendChannelRtcpStatistics(const int video_channel,
-                                           RtcpStatistics& basic_stats,
-                                           int64_t& rtt_ms) const;
-  virtual int GetRtpStatistics(const int video_channel,
-                               StreamDataCounters& sent,
-                               StreamDataCounters& received) const;
-  virtual int GetSendRtcpPacketTypeCounter(
-      int video_channel,
-      RtcpPacketTypeCounter* packet_counter) const;
-  virtual int GetReceiveRtcpPacketTypeCounter(
-      int video_channel,
-      RtcpPacketTypeCounter* packet_counter) const;
-  virtual int GetRemoteRTCPSenderInfo(const int video_channel,
-                                      SenderInfo* sender_info) const;
-  virtual int GetBandwidthUsage(const int video_channel,
-                                unsigned int& total_bitrate_sent,
-                                unsigned int& video_bitrate_sent,
-                                unsigned int& fec_bitrate_sent,
-                                unsigned int& nackBitrateSent) const;
-  virtual int GetEstimatedSendBandwidth(
-      const int video_channel,
-      unsigned int* estimated_bandwidth) const;
-  virtual int GetEstimatedReceiveBandwidth(
-      const int video_channel,
-      unsigned int* estimated_bandwidth) const;
-  virtual int GetPacerQueuingDelayMs(const int video_channel,
-                                     int64_t* delay_ms) const;
-  virtual int StartRTPDump(const int video_channel,
-                           const char file_nameUTF8[1024],
-                           RTPDirections direction);
-  virtual int StopRTPDump(const int video_channel, RTPDirections direction);
-  virtual int RegisterRTPObserver(const int video_channel,
-                                  ViERTPObserver& observer);
-  virtual int DeregisterRTPObserver(const int video_channel);
-
-  virtual int RegisterSendChannelRtcpStatisticsCallback(
-      int channel, RtcpStatisticsCallback* callback);
-  virtual int DeregisterSendChannelRtcpStatisticsCallback(
-      int channel, RtcpStatisticsCallback* callback);
-  virtual int RegisterReceiveChannelRtcpStatisticsCallback(
-      int channel, RtcpStatisticsCallback* callback);
-  virtual int DeregisterReceiveChannelRtcpStatisticsCallback(
-      int channel, RtcpStatisticsCallback* callback);
-  virtual int RegisterSendChannelRtpStatisticsCallback(
-      int channel, StreamDataCountersCallback* callback);
-  virtual int DeregisterSendChannelRtpStatisticsCallback(
-      int channel, StreamDataCountersCallback* callback);
-  virtual int RegisterReceiveChannelRtpStatisticsCallback(
-      int channel, StreamDataCountersCallback* callback);
-  virtual int DeregisterReceiveChannelRtpStatisticsCallback(
-      int channel, StreamDataCountersCallback* callback);
-  virtual int RegisterSendBitrateObserver(
-      int channel, BitrateStatisticsObserver* callback);
-  virtual int DeregisterSendBitrateObserver(
-      int channel, BitrateStatisticsObserver* callback);
-  virtual int RegisterSendFrameCountObserver(
-      int channel, FrameCountObserver* callback);
-  virtual int DeregisterSendFrameCountObserver(
-      int channel, FrameCountObserver* callback);
-  int RegisterRtcpPacketTypeCounterObserver(
-      int video_channel,
-      RtcpPacketTypeCounterObserver* observer) override;
-
- protected:
-  explicit ViERTP_RTCPImpl(ViESharedData* shared_data);
-  virtual ~ViERTP_RTCPImpl();
-
- private:
-  ViESharedData* shared_data_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc b/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc
deleted file mode 100644
index db7a5b116d..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video_engine/vie_sender.h"
-
-#include <assert.h>
-#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
-
-#include "webrtc/modules/utility/interface/rtp_dump.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-ViESender::ViESender(int channel_id)
-    : channel_id_(channel_id),
-      critsect_(CriticalSectionWrapper::CreateCriticalSection()),
-      transport_(NULL),
-      rtp_dump_(NULL) {
-}
-
-ViESender::~ViESender() {
-  if (rtp_dump_) {
-    rtp_dump_->Stop();
-    RtpDump::DestroyRtpDump(rtp_dump_);
-    rtp_dump_ = NULL;
-  }
-}
-
-int ViESender::RegisterSendTransport(Transport* transport) {
-  CriticalSectionScoped cs(critsect_.get());
-  if (transport_) {
-    return -1;
-  }
-  transport_ = transport;
-  return 0;
-}
-
-int ViESender::DeregisterSendTransport() {
-  CriticalSectionScoped cs(critsect_.get());
-  if (transport_ == NULL) {
-    return -1;
-  }
-  transport_ = NULL;
-  return 0;
-}
-
-int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
-  CriticalSectionScoped cs(critsect_.get());
-  if (rtp_dump_) {
-    // Packet dump is already started, restart it.
-    rtp_dump_->Stop();
-  } else {
-    rtp_dump_ = RtpDump::CreateRtpDump();
-    if (rtp_dump_ == NULL) {
-      return -1;
-    }
-  }
-  if (rtp_dump_->Start(file_nameUTF8) != 0) {
-    RtpDump::DestroyRtpDump(rtp_dump_);
-    rtp_dump_ = NULL;
-    return -1;
-  }
-  return 0;
-}
-
-int ViESender::StopRTPDump() {
-  CriticalSectionScoped cs(critsect_.get());
-  if (rtp_dump_) {
-    if (rtp_dump_->IsActive()) {
-      rtp_dump_->Stop();
-    }
-    RtpDump::DestroyRtpDump(rtp_dump_);
-    rtp_dump_ = NULL;
-  } else {
-    return -1;
-  }
-  return 0;
-}
-
-int ViESender::SendPacket(int vie_id, const void* data, size_t len) {
-  CriticalSectionScoped cs(critsect_.get());
-  if (!transport_) {
-    // No transport
-    return -1;
-  }
-  assert(ChannelId(vie_id) == channel_id_);
-
-  if (rtp_dump_) {
-    rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data), len);
-  }
-
-  return transport_->SendPacket(channel_id_, data, len);
-}
-
-int ViESender::SendRTCPPacket(int vie_id, const void* data, size_t len) {
-  CriticalSectionScoped cs(critsect_.get());
-  if (!transport_) {
-    return -1;
-  }
-  assert(ChannelId(vie_id) == channel_id_);
-
-  if (rtp_dump_) {
-    rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data), len);
-  }
-
-  return transport_->SendRTCPPacket(channel_id_, data, len);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sender.h b/media/webrtc/trunk/webrtc/video_engine/vie_sender.h
deleted file mode 100644
index 3e6bd95549..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sender.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// ViESender is responsible for sending packets to network.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_engine/vie_defines.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RtpDump;
-class Transport;
-class VideoCodingModule;
-
-class ViESender: public Transport {
- public:
-  explicit ViESender(const int32_t channel_id);
-  ~ViESender();
-
-  // Registers transport to use for sending RTP and RTCP.
-  int RegisterSendTransport(Transport* transport);
-  int DeregisterSendTransport();
-
-  // Stores all incoming packets to file.
-  int StartRTPDump(const char file_nameUTF8[1024]);
-  int StopRTPDump();
-
-  // Implements Transport.
-  int SendPacket(int vie_id, const void* data, size_t len) override;
-  int SendRTCPPacket(int vie_id, const void* data, size_t len) override;
-
- private:
-  const int32_t channel_id_;
-
-  rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
-
-  Transport* transport_;
-  RtpDump* rtp_dump_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc b/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc
deleted file mode 100644
index 39ad2843dc..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/cpu_info.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/video_engine/vie_channel_manager.h"
-#include "webrtc/video_engine/vie_defines.h"
-#include "webrtc/video_engine/vie_input_manager.h"
-#include "webrtc/video_engine/vie_render_manager.h"
-#include "webrtc/video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViESharedData::ViESharedData(const Config& config)
-    : number_cores_(CpuInfo::DetectNumberOfCores()),
-      channel_manager_(new ViEChannelManager(0, number_cores_, config)),
-      input_manager_(new ViEInputManager(0, config)),
-      render_manager_(new ViERenderManager(0)),
-      module_process_thread_(ProcessThread::Create()),
-      load_manager_(NULL),
-      last_error_(0) {
-  Trace::CreateTrace();
-  channel_manager_->SetModuleProcessThread(module_process_thread_.get());
-  input_manager_->SetModuleProcessThread(module_process_thread_.get());
-  module_process_thread_->Start();
-}
-
-ViESharedData::~ViESharedData() {
-  // Release these ones before the process thread and the trace.
-  input_manager_.reset();
-  channel_manager_.reset();
-  render_manager_.reset();
-
-  module_process_thread_->Stop();
-  Trace::ReturnTrace();
-}
-
-void ViESharedData::SetLastError(const int error) const {
-  last_error_ = error;
-}
-
-int ViESharedData::LastErrorInternal() const {
-  int error = last_error_;
-  last_error_ = 0;
-  return error;
-}
-
-int ViESharedData::NumberOfCores() const {
-  return number_cores_;
-}
-
-void ViESharedData::set_load_manager(CPULoadStateCallbackInvoker* load_manager) {
-  load_manager_ = load_manager;
-  channel_manager_->SetLoadManager(load_manager);
-}
-
-}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h b/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h
deleted file mode 100644
index 53d4f2d96c..0000000000
--- a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// ViESharedData contains data and instances common to all interface
-// implementations.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
-
-#include <map>
-
-#include "webrtc/base/scoped_ptr.h"
-
-namespace webrtc {
-
-class Config;
-class CpuOveruseObserver;
-class ProcessThread;
-class ViEChannelManager;
-class ViEInputManager;
-class ViERenderManager;
-
-class ViESharedData {
- public:
-  explicit ViESharedData(const Config& config);
-  ~ViESharedData();
-
-  void SetLastError(const int error) const;
-  int LastErrorInternal() const;
-  int NumberOfCores() const;
-
-  // TODO(mflodman) Remove all calls to 'instance_id()'.
-  int instance_id() { return 0;}
-  ViEChannelManager* channel_manager() { return channel_manager_.get(); }
-  ViEInputManager* input_manager() { return input_manager_.get(); }
-  ViERenderManager* render_manager() { return render_manager_.get(); }
-
-  std::map<int, CpuOveruseObserver*>* overuse_observers() {
-    return &overuse_observers_; }
-
-  CPULoadStateCallbackInvoker* load_manager() { return load_manager_; }
-  void set_load_manager(CPULoadStateCallbackInvoker* load_manager);
-
- private:
-  const int number_cores_;
-
-  rtc::scoped_ptr<ViEChannelManager> channel_manager_;
-  rtc::scoped_ptr<ViEInputManager> input_manager_;
-  rtc::scoped_ptr<ViERenderManager> render_manager_;
-  rtc::scoped_ptr<ProcessThread> module_process_thread_;
-  // Owned by PeerConnection, not ViEEngine
-  CPULoadStateCallbackInvoker* load_manager_;
-  mutable int last_error_;
-
-  std::map<int, CpuOveruseObserver*> overuse_observers_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine_tests.isolate b/media/webrtc/trunk/webrtc/video_engine_tests.isolate
index 5aa962323d..f2f961fa23 100644
--- a/media/webrtc/trunk/webrtc/video_engine_tests.isolate
+++ b/media/webrtc/trunk/webrtc/video_engine_tests.isolate
@@ -11,6 +11,7 @@
       'variables': {
         'files': [
           '<(DEPTH)/resources/foreman_cif_short.yuv',
+          '<(DEPTH)/resources/voice_engine/audio_long16.pcm',
         ],
       },
     }],
diff --git a/media/webrtc/trunk/webrtc/video_frame.h b/media/webrtc/trunk/webrtc/video_frame.h
index bd3e2b8a63..9d2ed9fd4d 100644
--- a/media/webrtc/trunk/webrtc/video_frame.h
+++ b/media/webrtc/trunk/webrtc/video_frame.h
@@ -12,28 +12,23 @@
 #define WEBRTC_VIDEO_FRAME_H_
 
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/common_video/interface/native_handle.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
 #include "webrtc/common_video/rotation.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-class I420VideoFrame {
+class VideoFrame {
  public:
-  I420VideoFrame();
-  I420VideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
-                 uint32_t timestamp,
-                 int64_t render_time_ms,
-                 VideoRotation rotation);
-  I420VideoFrame(NativeHandle* handle,
-                 int width,
-                 int height,
-                 uint32_t timestamp,
-                 int64_t render_time_ms);
+  VideoFrame();
+  VideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
+             uint32_t timestamp,
+             int64_t render_time_ms,
+             VideoRotation rotation);
 
   // TODO(pbos): Make all create/copy functions void, they should not be able to
-  // fail (which should be DCHECK/CHECKed instead).
+  // fail (which should be RTC_DCHECK/CHECKed instead).
 
   // CreateEmptyFrame: Sets frame dimensions and allocates buffers based
   // on set dimensions - height and plane stride.
@@ -81,11 +76,11 @@ class I420VideoFrame {
   // Deep copy frame: If required size is bigger than allocated one, new
   // buffers of adequate size will be allocated.
   // Return value: 0 on success, -1 on error.
-  int CopyFrame(const I420VideoFrame& videoFrame);
+  int CopyFrame(const VideoFrame& videoFrame);
 
   // Creates a shallow copy of |videoFrame|, i.e, the this object will retain a
   // reference to the video buffer also retained by |videoFrame|.
-  void ShallowCopy(const I420VideoFrame& videoFrame);
+  void ShallowCopy(const VideoFrame& videoFrame);
 
   // Release frame buffer and reset time stamps.
   void Reset();
@@ -159,6 +154,12 @@ class I420VideoFrame {
   void set_video_frame_buffer(
       const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer);
 
+  // Convert native-handle frame to memory-backed I420 frame. Should not be
+  // called on a non-native-handle frame.
+  VideoFrame ConvertNativeToI420Frame() const;
+
+  bool EqualsFrame(const VideoFrame& frame) const;
+
  private:
   // An opaque reference counted handle that stores the pixel data.
   rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
@@ -168,13 +169,6 @@ class I420VideoFrame {
   VideoRotation rotation_;
 };
 
-enum VideoFrameType {
-  kKeyFrame = 0,
-  kDeltaFrame = 1,
-  kGoldenFrame = 2,
-  kAltRefFrame = 3,
-  kSkipFrame = 4
-};
 
 // TODO(pbos): Rename EncodedFrame and reformat this class' members.
 class EncodedImage {
@@ -183,18 +177,31 @@ class EncodedImage {
   EncodedImage(uint8_t* buffer, size_t length, size_t size)
       : _buffer(buffer), _length(length), _size(size) {}
 
+  struct AdaptReason {
+    AdaptReason()
+        : quality_resolution_downscales(-1),
+          bw_resolutions_disabled(-1) {}
+
+    int quality_resolution_downscales;  // Number of times this frame is down
+                                        // scaled in resolution due to quality.
+                                        // Or -1 if information is not provided.
+    int bw_resolutions_disabled;  // Number of resolutions that are not sent
+                                  // due to bandwidth for this frame.
+                                  // Or -1 if information is not provided.
+  };
   uint32_t _encodedWidth = 0;
   uint32_t _encodedHeight = 0;
   uint32_t _timeStamp = 0;
   // NTP time of the capture time in local timebase in milliseconds.
   int64_t ntp_time_ms_ = 0;
   int64_t capture_time_ms_ = 0;
-  // TODO(pbos): Use webrtc::FrameType directly (and remove VideoFrameType).
-  VideoFrameType _frameType = kDeltaFrame;
+  FrameType _frameType = kVideoFrameDelta;
   uint8_t* _buffer;
   size_t _length;
   size_t _size;
   bool _completeFrame = false;
+  AdaptReason adapt_reason_;
+  int qp_ = -1;  // Quantizer value.
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_receive_stream.h b/media/webrtc/trunk/webrtc/video_receive_stream.h
index 0f2151df03..29cc6e4ca1 100644
--- a/media/webrtc/trunk/webrtc/video_receive_stream.h
+++ b/media/webrtc/trunk/webrtc/video_receive_stream.h
@@ -18,50 +18,33 @@
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/frame_callback.h"
+#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/video_renderer.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
-namespace newapi {
-// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
-// RTCP mode is described by RFC 5506.
-enum RtcpMode { kRtcpCompound, kRtcpReducedSize };
-}  // namespace newapi
-
 class VideoDecoder;
 
-class VideoReceiveStream {
+class VideoReceiveStream : public ReceiveStream {
  public:
   // TODO(mflodman) Move all these settings to VideoDecoder and move the
   // declaration to common_types.h.
   struct Decoder {
-    Decoder()
-        : decoder(NULL),
-          payload_type(0),
-          is_renderer(false),
-          expected_delay_ms(0) {}
     std::string ToString() const;
 
     // The actual decoder instance.
-    VideoDecoder* decoder;
+    VideoDecoder* decoder = nullptr;
 
     // Received RTP packets with this payload type will be sent to this decoder
     // instance.
-    int payload_type;
+    int payload_type = 0;
 
     // Name of the decoded payload (such as VP8). Maps back to the depacketizer
     // used to unpack incoming packets.
     std::string payload_name;
-
-    // 'true' if the decoder handles rendering as well.
-    bool is_renderer;
-
-    // The expected delay for decoding and rendering, i.e. the frame will be
-    // delivered this many milliseconds, if possible, earlier than the ideal
-    // render time.
-    // Note: Ignored if 'renderer' is false.
-    int expected_delay_ms;
   };
 
   struct Stats {
@@ -70,6 +53,7 @@ class VideoReceiveStream {
     int render_frame_rate = 0;
 
     // Decoder stats.
+    std::string decoder_implementation_name = "unknown";
     FrameCounts frame_counts;
     int decode_ms = 0;
     int max_decode_ms = 0;
@@ -77,7 +61,9 @@ class VideoReceiveStream {
     int target_delay_ms = 0;
     int jitter_buffer_ms = 0;
     int min_playout_delay_ms = 0;
-    int render_delay_ms = 0;
+    int render_delay_ms = 10;
+
+    int current_payload_type = -1;
 
     int total_bitrate_bps = 0;
     int discarded_packets = 0;
@@ -90,13 +76,10 @@ class VideoReceiveStream {
   };
 
   struct Config {
-    Config()
-        : renderer(NULL),
-          render_delay_ms(0),
-          audio_channel_id(-1),
-          pre_decode_callback(NULL),
-          pre_render_callback(NULL),
-          target_delay_ms(0) {}
+    Config() = delete;
+    explicit Config(Transport* rtcp_send_transport)
+        : rtcp_send_transport(rtcp_send_transport) {}
+
     std::string ToString() const;
 
     // Decoders for every payload that we can receive.
@@ -104,32 +87,33 @@ class VideoReceiveStream {
 
     // Receive-stream specific RTP settings.
     struct Rtp {
-      Rtp()
-          : remote_ssrc(0),
-            local_ssrc(0),
-            rtcp_mode(newapi::kRtcpReducedSize),
-            remb(true) {}
       std::string ToString() const;
 
       // Synchronization source (stream identifier) to be received.
-      uint32_t remote_ssrc;
+      uint32_t remote_ssrc = 0;
       // Sender SSRC used for sending RTCP (such as receiver reports).
-      uint32_t local_ssrc;
+      uint32_t local_ssrc = 0;
 
       // See RtcpMode for description.
-      newapi::RtcpMode rtcp_mode;
+      RtcpMode rtcp_mode = RtcpMode::kCompound;
 
       // Extended RTCP settings.
       struct RtcpXr {
-        RtcpXr() : receiver_reference_time_report(false) {}
-
         // True if RTCP Receiver Reference Time Report Block extension
         // (RFC 3611) should be enabled.
-        bool receiver_reference_time_report;
+        bool receiver_reference_time_report = false;
       } rtcp_xr;
 
       // See draft-alvestrand-rmcat-remb for information.
-      bool remb;
+      bool remb = false;
+
+      bool tmmbr = false;
+
+      // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
+      bool transport_cc = false;
+
+      // TODO(jesup) - there should be a kKeyFrameReqNone
+      KeyFrameRequestMethod keyframe_method = kKeyFrameReqPliRtcp;
 
       // See NackConfig for description.
       NackConfig nack;
@@ -140,60 +124,62 @@ class VideoReceiveStream {
       // RTX settings for incoming video payloads that may be received. RTX is
       // disabled if there's no config present.
       struct Rtx {
-        Rtx() : ssrc(0), payload_type(0) {}
-
         // SSRCs to use for the RTX streams.
-        uint32_t ssrc;
+        uint32_t ssrc = 0;
 
         // Payload type to use for the RTX stream.
-        int payload_type;
+        int payload_type = 0;
       };
 
       // Map from video RTP payload type -> RTX config.
       typedef std::map<int, Rtx> RtxMap;
       RtxMap rtx;
 
+      // If set to true, the RTX payload type mapping supplied in |rtx| will be
+      // used when restoring RTX packets. Without it, RTX packets will always be
+      // restored to the last non-RTX packet payload type received.
+      bool use_rtx_payload_mapping_on_restore = false;
+
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
-    // VideoRenderer will be called for each decoded frame. 'NULL' disables
+    // Transport for outgoing packets (RTCP).
+    Transport* rtcp_send_transport = nullptr;
+
+    // VideoRenderer will be called for each decoded frame. 'nullptr' disables
     // rendering of this stream.
-    VideoRenderer* renderer;
+    VideoRenderer* renderer = nullptr;
 
     // Expected delay needed by the renderer, i.e. the frame will be delivered
     // this many milliseconds, if possible, earlier than the ideal render time.
     // Only valid if 'renderer' is set.
-    int render_delay_ms;
+    int render_delay_ms = 10;
 
-    // Audio channel corresponding to this video stream, used for audio/video
-    // synchronization. 'audio_channel_id' is ignored if no VoiceEngine is set
-    // when creating the VideoEngine instance. '-1' disables a/v sync.
-    int audio_channel_id;
+    // Identifier for an A/V synchronization group. Empty string to disable.
+    // TODO(pbos): Synchronize streams in a sync group, not just video streams
+    // to one of the audio streams.
+    std::string sync_group;
 
     // Called for each incoming video frame, i.e. in encoded state. E.g. used
     // when
-    // saving the stream to a file. 'NULL' disables the callback.
-    EncodedFrameObserver* pre_decode_callback;
+    // saving the stream to a file. 'nullptr' disables the callback.
+    EncodedFrameObserver* pre_decode_callback = nullptr;
 
     // Called for each decoded frame. E.g. used when adding effects to the
     // decoded
-    // stream. 'NULL' disables the callback.
-    I420FrameCallback* pre_render_callback;
+    // stream. 'nullptr' disables the callback.
+    I420FrameCallback* pre_render_callback = nullptr;
 
     // Target delay in milliseconds. A positive value indicates this stream is
     // used for streaming instead of a real-time call.
-    int target_delay_ms;
+    int target_delay_ms = 0;
   };
 
-  virtual void Start() = 0;
-  virtual void Stop() = 0;
-
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
-
- protected:
-  virtual ~VideoReceiveStream() {}
+  virtual int64_t GetRtt() const = 0;
+  virtual void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id) = 0;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_renderer.h b/media/webrtc/trunk/webrtc/video_renderer.h
index 34d339a508..7cb9ed1aa5 100644
--- a/media/webrtc/trunk/webrtc/video_renderer.h
+++ b/media/webrtc/trunk/webrtc/video_renderer.h
@@ -8,26 +8,31 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_RENDERER_H_
+#ifndef WEBRTC_VIDEO_RENDERER_H_
+#define WEBRTC_VIDEO_RENDERER_H_
 
 namespace webrtc {
 
-class I420VideoFrame;
+class VideoFrame;
 
 class VideoRenderer {
  public:
   // This function should return as soon as possible and not block until it's
   // time to render the frame.
-  // TODO(mflodman) Remove time_to_render_ms when I420VideoFrame contains NTP.
-  virtual void RenderFrame(const I420VideoFrame& video_frame,
+  // TODO(mflodman) Remove time_to_render_ms when VideoFrame contains NTP.
+  virtual void RenderFrame(const VideoFrame& video_frame,
                            int time_to_render_ms) = 0;
 
   virtual bool IsTextureSupported() const = 0;
 
+  // This function returns true if WebRTC should not delay frames for
+  // smoothness. In general, this case means the renderer can schedule frames to
+  // optimize smoothness.
+  virtual bool SmoothsRenderedFrames() const { return false; }
+
  protected:
   virtual ~VideoRenderer() {}
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_RENDERER_H_
+#endif  // WEBRTC_VIDEO_RENDERER_H_
diff --git a/media/webrtc/trunk/webrtc/video_send_stream.h b/media/webrtc/trunk/webrtc/video_send_stream.h
index 4ae0b6b58a..0d9fa57bef 100644
--- a/media/webrtc/trunk/webrtc/video_send_stream.h
+++ b/media/webrtc/trunk/webrtc/video_send_stream.h
@@ -17,25 +17,35 @@
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/frame_callback.h"
+#include "webrtc/stream.h"
+#include "webrtc/transport.h"
 #include "webrtc/video_renderer.h"
 
 namespace webrtc {
 
+class LoadObserver;
 class VideoEncoder;
 
+class EncodingTimeObserver {
+ public:
+  virtual ~EncodingTimeObserver() {}
+
+  virtual void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) = 0;
+};
+
 // Class to deliver captured frame to the video send stream.
-class VideoSendStreamInput {
+class VideoCaptureInput {
  public:
   // These methods do not lock internally and must be called sequentially.
   // If your application switches input sources synchronization must be done
   // externally to make sure that any old frames are not delivered concurrently.
-  virtual void IncomingCapturedFrame(const I420VideoFrame& video_frame) = 0;
+  virtual void IncomingCapturedFrame(const VideoFrame& video_frame) = 0;
 
  protected:
-  virtual ~VideoSendStreamInput() {}
+  virtual ~VideoCaptureInput() {}
 };
 
-class VideoSendStream {
+class VideoSendStream : public SendStream {
  public:
   struct StreamStats {
     FrameCounts frame_counts;
@@ -52,56 +62,51 @@ class VideoSendStream {
   };
 
   struct Stats {
-    Stats()
-        : input_frame_rate(0),
-          encode_frame_rate(0),
-          avg_encode_time_ms(0),
-          encode_usage_percent(0),
-          target_media_bitrate_bps(0),
-          media_bitrate_bps(0),
-          suspended(false) {}
-    int input_frame_rate;
-    int encode_frame_rate;
-    int avg_encode_time_ms;
-    int encode_usage_percent;
-    int target_media_bitrate_bps;
-    int media_bitrate_bps;
-    bool suspended;
+    std::string encoder_implementation_name = "unknown";
+    int input_frame_rate = 0;
+    int encode_frame_rate = 0;
+    int avg_encode_time_ms = 0;
+    int encode_usage_percent = 0;
+    int target_media_bitrate_bps = 0;
+    int media_bitrate_bps = 0;
+    bool suspended = false;
+    bool bw_limited_resolution = false;
     std::map<uint32_t, StreamStats> substreams;
   };
 
   struct Config {
-    Config()
-        : pre_encode_callback(NULL),
-          post_encode_callback(NULL),
-          local_renderer(NULL),
-          render_delay_ms(0),
-          target_delay_ms(0),
-          suspend_below_min_bitrate(false) {}
+    Config() = delete;
+    explicit Config(Transport* send_transport)
+        : send_transport(send_transport) {}
+
     std::string ToString() const;
 
     struct EncoderSettings {
-      EncoderSettings() : payload_type(-1), encoder(NULL) {}
-
       std::string ToString() const;
 
       std::string payload_name;
-      int payload_type;
+      int payload_type = -1;
+
+      // TODO(sophiechang): Delete this field when no one is using internal
+      // sources anymore.
+      bool internal_source = false;
 
       // Uninitialized VideoEncoder instance to be used for encoding. Will be
       // initialized from inside the VideoSendStream.
-      VideoEncoder* encoder;
+      VideoEncoder* encoder = nullptr;
     } encoder_settings;
 
     static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
     struct Rtp {
-      Rtp() : max_packet_size(kDefaultMaxPacketSize) {}
       std::string ToString() const;
 
       std::vector<uint32_t> ssrcs;
 
+      // See RtcpMode for description.
+      RtcpMode rtcp_mode = RtcpMode::kCompound;
+
       // Max RTP packet size delivered to send transport from VideoEngine.
-      size_t max_packet_size;
+      size_t max_packet_size = kDefaultMaxPacketSize;
 
       // RTP header extensions to use for this send stream.
       std::vector<RtpExtension> extensions;
@@ -115,52 +120,64 @@ class VideoSendStream {
       // Settings for RTP retransmission payload format, see RFC 4588 for
       // details.
       struct Rtx {
-        Rtx() : payload_type(-1) {}
         std::string ToString() const;
         // SSRCs to use for the RTX streams.
         std::vector<uint32_t> ssrcs;
 
         // Payload type to use for the RTX stream.
-        int payload_type;
+        int payload_type = -1;
       } rtx;
 
       // RTCP CNAME, see RFC 3550.
       std::string c_name;
     } rtp;
 
-    // Called for each I420 frame before encoding the frame. Can be used for
-    // effects, snapshots etc. 'NULL' disables the callback.
-    I420FrameCallback* pre_encode_callback;
+    // Transport for outgoing packets.
+    Transport* send_transport = nullptr;
 
-    // Called for each encoded frame, e.g. used for file storage. 'NULL'
+    // Callback for overuse and normal usage based on the jitter of incoming
+    // captured frames. 'nullptr' disables the callback.
+    LoadObserver* overuse_callback = nullptr;
+
+    // Called for each I420 frame before encoding the frame. Can be used for
+    // effects, snapshots etc. 'nullptr' disables the callback.
+    I420FrameCallback* pre_encode_callback = nullptr;
+
+    // Called for each encoded frame, e.g. used for file storage. 'nullptr'
     // disables the callback.
-    EncodedFrameObserver* post_encode_callback;
+    EncodedFrameObserver* post_encode_callback = nullptr;
 
     // Renderer for local preview. The local renderer will be called even if
-    // sending hasn't started. 'NULL' disables local rendering.
-    VideoRenderer* local_renderer;
+    // sending hasn't started. 'nullptr' disables local rendering.
+    VideoRenderer* local_renderer = nullptr;
 
     // Expected delay needed by the renderer, i.e. the frame will be delivered
     // this many milliseconds, if possible, earlier than expected render time.
     // Only valid if |local_renderer| is set.
-    int render_delay_ms;
+    int render_delay_ms = 0;
 
     // Target delay in milliseconds. A positive value indicates this stream is
     // used for streaming instead of a real-time call.
-    int target_delay_ms;
+    int target_delay_ms = 0;
 
     // True if the stream should be suspended when the available bitrate fall
     // below the minimum configured bitrate. If this variable is false, the
     // stream may send at a rate higher than the estimated available bitrate.
-    bool suspend_below_min_bitrate;
+    bool suspend_below_min_bitrate = false;
+
+    // Called for each encoded frame. Passes the total time spent on encoding.
+    // TODO(ivica): Consolidate with post_encode_callback:
+    // https://code.google.com/p/webrtc/issues/detail?id=5042
+    EncodingTimeObserver* encoding_time_observer = nullptr;
   };
 
   // Gets interface used to insert captured frames. Valid as long as the
   // VideoSendStream is valid.
-  virtual VideoSendStreamInput* Input() = 0;
+  virtual VideoCaptureInput* Input() = 0;
 
-  virtual void Start() = 0;
-  virtual void Stop() = 0;
+  // Gets interface used to signal the current CPU work level to the encoder.
+  // Valid as long as the VideoSendStream is valid.
+  virtual CPULoadStateObserver* LoadStateObserver() = 0;
 
   // Set which streams to send. Must have at least as many SSRCs as configured
   // in the config. Encoder settings are passed on to the encoder instance along
@@ -168,9 +185,6 @@ class VideoSendStream {
   virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
-
- protected:
-  virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn b/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn
index c55dc8ed07..82cd92355c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn
@@ -9,8 +9,17 @@
 import("../build/webrtc.gni")
 
 source_set("voice_engine") {
-
   sources = [
+    "channel.cc",
+    "channel.h",
+    "channel_manager.cc",
+    "channel_manager.h",
+    "channel_proxy.cc",
+    "channel_proxy.h",
+    "dtmf_inband.cc",
+    "dtmf_inband.h",
+    "dtmf_inband_queue.cc",
+    "dtmf_inband_queue.h",
     "include/voe_audio_processing.h",
     "include/voe_base.h",
     "include/voe_codec.h",
@@ -24,14 +33,6 @@ source_set("voice_engine") {
     "include/voe_rtp_rtcp.h",
     "include/voe_video_sync.h",
     "include/voe_volume_control.h",
-    "channel.cc",
-    "channel.h",
-    "channel_manager.cc",
-    "channel_manager.h",
-    "dtmf_inband.cc",
-    "dtmf_inband.h",
-    "dtmf_inband_queue.cc",
-    "dtmf_inband_queue.h",
     "level_indicator.cc",
     "level_indicator.h",
     "monitor_module.cc",
@@ -96,6 +97,7 @@ source_set("voice_engine") {
   }
 
   deps = [
+    "..:rtc_event_log",
     "..:webrtc_common",
     "../common_audio",
     "../modules/audio_coding",
@@ -104,6 +106,7 @@ source_set("voice_engine") {
     "../modules/audio_processing",
     "../modules/bitrate_controller",
     "../modules/media_file",
+    "../modules/pacing",
     "../modules/rtp_rtcp",
     "../modules/utility",
     "../system_wrappers",
diff --git a/media/webrtc/trunk/webrtc/voice_engine/OWNERS b/media/webrtc/trunk/webrtc/voice_engine/OWNERS
index 1fe45940ba..ee81fe4461 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/OWNERS
+++ b/media/webrtc/trunk/webrtc/voice_engine/OWNERS
@@ -1,7 +1,7 @@
 henrikg@webrtc.org
 henrika@webrtc.org
 niklas.enbom@webrtc.org
-xians@webrtc.org
+solenberg@webrtc.org
 
 per-file *.isolate=kjellander@webrtc.org
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel.cc b/media/webrtc/trunk/webrtc/voice_engine/channel.cc
index a903db357d..5fc478e081 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.cc
@@ -10,23 +10,28 @@
 
 #include "webrtc/voice_engine/channel.h"
 
+#include <algorithm>
+#include <utility>
+
+#include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
+#include "webrtc/config.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/modules/utility/interface/rtp_dump.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/video_engine/include/vie_network.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
@@ -42,6 +47,104 @@
 namespace webrtc {
 namespace voe {
 
+class TransportFeedbackProxy : public TransportFeedbackObserver {
+ public:
+  TransportFeedbackProxy() : feedback_observer_(nullptr) {
+    pacer_thread_.DetachFromThread();
+    network_thread_.DetachFromThread();
+  }
+
+  void SetTransportFeedbackObserver(
+      TransportFeedbackObserver* feedback_observer) {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    feedback_observer_ = feedback_observer;
+  }
+
+  // Implements TransportFeedbackObserver.
+  void AddPacket(uint16_t sequence_number,
+                 size_t length,
+                 bool was_paced) override {
+    RTC_DCHECK(pacer_thread_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    if (feedback_observer_)
+      feedback_observer_->AddPacket(sequence_number, length, was_paced);
+  }
+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
+    RTC_DCHECK(network_thread_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    if (feedback_observer_)
+      feedback_observer_->OnTransportFeedback(feedback);
+  }
+
+ private:
+  rtc::CriticalSection crit_;
+  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker pacer_thread_;
+  rtc::ThreadChecker network_thread_;
+  TransportFeedbackObserver* feedback_observer_ GUARDED_BY(&crit_);
+};
+
+class TransportSequenceNumberProxy : public TransportSequenceNumberAllocator {
+ public:
+  TransportSequenceNumberProxy() : seq_num_allocator_(nullptr) {
+    pacer_thread_.DetachFromThread();
+  }
+
+  void SetSequenceNumberAllocator(
+      TransportSequenceNumberAllocator* seq_num_allocator) {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    seq_num_allocator_ = seq_num_allocator;
+  }
+
+  // Implements TransportSequenceNumberAllocator.
+  uint16_t AllocateSequenceNumber() override {
+    RTC_DCHECK(pacer_thread_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    if (!seq_num_allocator_)
+      return 0;
+    return seq_num_allocator_->AllocateSequenceNumber();
+  }
+
+ private:
+  rtc::CriticalSection crit_;
+  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker pacer_thread_;
+  TransportSequenceNumberAllocator* seq_num_allocator_ GUARDED_BY(&crit_);
+};
+
+class RtpPacketSenderProxy : public RtpPacketSender {
+ public:
+  RtpPacketSenderProxy() : rtp_packet_sender_(nullptr) {
+  }
+
+  void SetPacketSender(RtpPacketSender* rtp_packet_sender) {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    rtp_packet_sender_ = rtp_packet_sender;
+  }
+
+  // Implements RtpPacketSender.
+  void InsertPacket(Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission) override {
+    rtc::CritScope lock(&crit_);
+    if (rtp_packet_sender_) {
+      rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number,
+                                       capture_time_ms, bytes, retransmission);
+    }
+  }
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  rtc::CriticalSection crit_;
+  RtpPacketSender* rtp_packet_sender_ GUARDED_BY(&crit_);
+};
+
 // Extend the default RTCP statistics struct with max_jitter, defined as the
 // maximum jitter value seen in an RTCP report block.
 struct ChannelStatistics : public RtcpStatistics {
@@ -72,17 +175,12 @@ class StatisticsProxy : public RtcpStatisticsCallback {
   }
 
   void CNameChanged(const char* cname, uint32_t ssrc) override {}
-
+ 
   void SetSSRC(uint32_t ssrc) {
     CriticalSectionScoped cs(stats_lock_.get());
     ssrc_ = ssrc;
   }
 
-  void ResetStatistics() {
-    CriticalSectionScoped cs(stats_lock_.get());
-    stats_ = ChannelStatistics();
-  }
-
   ChannelStatistics GetStats() {
     CriticalSectionScoped cs(stats_lock_.get());
     return stats_;
@@ -213,9 +311,6 @@ Channel::InFrameType(FrameType frame_type)
 int32_t
 Channel::OnRxVadDetected(int vadDecision)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
-
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_rxVadObserverPtr)
     {
@@ -225,15 +320,11 @@ Channel::OnRxVadDetected(int vadDecision)
     return 0;
 }
 
-int
-Channel::SendPacket(int channel, const void *data, size_t len)
-{
-    channel = VoEChannelId(channel);
-    assert(channel == _channelId);
-
+bool Channel::SendRtp(const uint8_t* data,
+                      size_t len,
+                      const PacketOptions& options) {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::SendPacket(channel=%d, len=%" PRIuS ")", channel,
-                 len);
+                 "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
 
     CriticalSectionScoped cs(&_callbackCritSect);
 
@@ -242,89 +333,62 @@ Channel::SendPacket(int channel, const void *data, size_t len)
         WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::SendPacket() failed to send RTP packet due to"
                      " invalid transport object");
-        return -1;
+        return false;
     }
 
     uint8_t* bufferToSendPtr = (uint8_t*)data;
     size_t bufferLength = len;
 
-    // Dump the RTP packet to a file (if RTP dump is enabled).
-    if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                     VoEId(_instanceId,_channelId),
-                     "Channel::SendPacket() RTP dump to output file failed");
-    }
-
-    int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
-                                      bufferLength);
-    if (n < 0) {
+    if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
       std::string transport_name =
           _externalTransport ? "external transport" : "WebRtc sockets";
       WEBRTC_TRACE(kTraceError, kTraceVoice,
                    VoEId(_instanceId,_channelId),
                    "Channel::SendPacket() RTP transmission using %s failed",
                    transport_name.c_str());
-      return -1;
+      return false;
     }
-    return n;
+    return true;
 }
 
-int
-Channel::SendRTCPPacket(int channel, const void *data, size_t len)
+bool
+Channel::SendRtcp(const uint8_t *data, size_t len)
 {
-    channel = VoEChannelId(channel);
-    assert(channel == _channelId);
-
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::SendRTCPPacket(channel=%d, len=%" PRIuS ")", channel,
-                 len);
+                 "Channel::SendRtcp(len=%" PRIuS ")", len);
 
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_transportPtr == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice,
                      VoEId(_instanceId,_channelId),
-                     "Channel::SendRTCPPacket() failed to send RTCP packet"
+                     "Channel::SendRtcp() failed to send RTCP packet"
                      " due to invalid transport object");
-        return -1;
+        return false;
     }
 
     uint8_t* bufferToSendPtr = (uint8_t*)data;
     size_t bufferLength = len;
 
-    // Dump the RTCP packet to a file (if RTP dump is enabled).
-    if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                     VoEId(_instanceId,_channelId),
-                     "Channel::SendPacket() RTCP dump to output file failed");
-    }
-
-    int n = _transportPtr->SendRTCPPacket(channel,
-                                          bufferToSendPtr,
-                                          bufferLength);
+    int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
     if (n < 0) {
       std::string transport_name =
           _externalTransport ? "external transport" : "WebRtc sockets";
       WEBRTC_TRACE(kTraceInfo, kTraceVoice,
                    VoEId(_instanceId,_channelId),
-                   "Channel::SendRTCPPacket() transmission using %s failed",
+                   "Channel::SendRtcp() transmission using %s failed",
                    transport_name.c_str());
-      return -1;
+      return false;
     }
-    return n;
+    return true;
 }
 
-void
-Channel::OnPlayTelephoneEvent(int32_t id,
-                              uint8_t event,
-                              uint16_t lengthMs,
-                              uint8_t volume)
-{
+void Channel::OnPlayTelephoneEvent(uint8_t event,
+                                   uint16_t lengthMs,
+                                   uint8_t volume) {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
-                 " volume=%u)", id, event, lengthMs, volume);
+                 "Channel::OnPlayTelephoneEvent(event=%u, lengthMs=%u,"
+                 " volume=%u)", event, lengthMs, volume);
 
     if (!_playOutbandDtmfEvent || (event > 15))
     {
@@ -341,11 +405,10 @@ Channel::OnPlayTelephoneEvent(int32_t id,
 }
 
 void
-Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc)
+Channel::OnIncomingSSRCChanged(uint32_t ssrc)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
-                 id, ssrc);
+                 "Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
 
     // Update ssrc so that NTP for AV sync can be updated.
     _rtpRtcpModule->SetRemoteSSRC(ssrc);
@@ -353,39 +416,22 @@ Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc)
     statistics_proxy_->SetSSRC(ssrc);
 }
 
-void Channel::OnIncomingCSRCChanged(int32_t id,
-                                    uint32_t CSRC,
-                                    bool added)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
-                 id, CSRC, added);
+void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::OnIncomingCSRCChanged(CSRC=%d, added=%d)", CSRC,
+               added);
 }
 
-void Channel::ResetStatistics(uint32_t ssrc) {
-  StreamStatistician* statistician =
-      rtp_receive_statistics_->GetStatistician(ssrc);
-  if (statistician) {
-    statistician->ResetStatistics();
-  }
-  statistics_proxy_->ResetStatistics();
-}
-
-int32_t
-Channel::OnInitializeDecoder(
-    int32_t id,
+int32_t Channel::OnInitializeDecoder(
     int8_t payloadType,
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     int frequency,
-    uint8_t channels,
-    uint32_t rate)
-{
+    size_t channels,
+    uint32_t rate) {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
-                 "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
-                 id, payloadType, payloadName, frequency, channels, rate);
-
-    assert(VoEChannelId(id) == _channelId);
+                 "Channel::OnInitializeDecoder(payloadType=%d, "
+                 "payloadName=%s, frequency=%u, channels=%" PRIuS ", rate=%u)",
+                 payloadType, payloadName, frequency, channels, rate);
 
     CodecInst receiveCodec = {0};
     CodecInst dummyCodec = {0};
@@ -420,7 +466,7 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnReceivedPayloadData(payloadSize=%" PRIuS ","
-                 " payloadType=%u, audioChannel=%u)",
+                 " payloadType=%u, audioChannel=%" PRIuS ")",
                  payloadSize,
                  rtpHeader->header.payloadType,
                  rtpHeader->type.Audio.channel);
@@ -481,14 +527,16 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
-int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
+int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame)
 {
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetAudioFrame(id=%d)", id);
-
+    if (event_log_) {
+      unsigned int ssrc;
+      RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+      event_log_->LogAudioPlayout(ssrc);
+    }
     // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
-    if (audio_coding_->PlayoutData10Ms(audioFrame.sample_rate_hz_,
-                                       &audioFrame) == -1)
+    if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_,
+                                       audioFrame) == -1)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice,
                      VoEId(_instanceId,_channelId),
@@ -502,24 +550,39 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
 
     if (_RxVadDetection)
     {
-        UpdateRxVadDetection(audioFrame);
+        UpdateRxVadDetection(*audioFrame);
     }
 
     // Convert module ID to internal VoE channel ID
-    audioFrame.id_ = VoEChannelId(audioFrame.id_);
+    audioFrame->id_ = VoEChannelId(audioFrame->id_);
     // Store speech type for dead-or-alive detection
-    _outputSpeechType = audioFrame.speech_type_;
+    _outputSpeechType = audioFrame->speech_type_;
 
     ChannelState::State state = channel_state_.Get();
 
     if (state.rx_apm_is_enabled) {
-      int err = rx_audioproc_->ProcessStream(&audioFrame);
+      int err = rx_audioproc_->ProcessStream(audioFrame);
       if (err) {
         LOG(LS_ERROR) << "ProcessStream() error: " << err;
         assert(false);
       }
     }
 
+    {
+      // Pass the audio buffers to an optional sink callback, before applying
+      // scaling/panning, as that applies to the mix operation.
+      // External recipients of the audio (e.g. via AudioTrack), will do their
+      // own mixing/dynamic processing.
+      CriticalSectionScoped cs(&_callbackCritSect);
+      if (audio_sink_) {
+        AudioSinkInterface::Data data(
+            &audioFrame->data_[0],
+            audioFrame->samples_per_channel_, audioFrame->sample_rate_hz_,
+            audioFrame->num_channels_, audioFrame->timestamp_);
+        audio_sink_->OnData(data);
+      }
+    }
+
     float output_gain = 1.0f;
     float left_pan =  1.0f;
     float right_pan =  1.0f;
@@ -533,7 +596,7 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
     // Output volume scaling
     if (output_gain < 0.99f || output_gain > 1.01f)
     {
-        AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
+        AudioFrameOperations::ScaleWithSat(output_gain, *audioFrame);
     }
 
     // Scale left and/or right channel(s) if stereo and master balance is
@@ -541,40 +604,37 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
 
     if (left_pan != 1.0f || right_pan != 1.0f)
     {
-        if (audioFrame.num_channels_ == 1)
+        if (audioFrame->num_channels_ == 1)
         {
             // Emulate stereo mode since panning is active.
             // The mono signal is copied to both left and right channels here.
-            AudioFrameOperations::MonoToStereo(&audioFrame);
+            AudioFrameOperations::MonoToStereo(audioFrame);
         }
         // For true stereo mode (when we are receiving a stereo signal), no
         // action is needed.
 
         // Do the panning operation (the audio frame contains stereo at this
         // stage)
-        AudioFrameOperations::Scale(left_pan, right_pan, audioFrame);
+        AudioFrameOperations::Scale(left_pan, right_pan, *audioFrame);
     }
 
     // Mix decoded PCM output with file if file mixing is enabled
     if (state.output_file_playing)
     {
-        MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_);
+        MixAudioWithFile(*audioFrame, audioFrame->sample_rate_hz_);
     }
 
     // External media
     if (_outputExternalMedia)
     {
         CriticalSectionScoped cs(&_callbackCritSect);
-        const bool isStereo = (audioFrame.num_channels_ == 2);
+        const bool isStereo = (audioFrame->num_channels_ == 2);
         if (_outputExternalMediaCallbackPtr)
         {
-            _outputExternalMediaCallbackPtr->Process(
-                _channelId,
-                kPlaybackPerChannel,
-                (int16_t*)audioFrame.data_,
-                audioFrame.samples_per_channel_,
-                audioFrame.sample_rate_hz_,
-                isStereo);
+          _outputExternalMediaCallbackPtr->Process(
+              _channelId, kPlaybackPerChannel, (int16_t*)audioFrame->data_,
+              audioFrame->samples_per_channel_, audioFrame->sample_rate_hz_,
+              isStereo);
         }
     }
 
@@ -584,16 +644,16 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
 
         if (_outputFileRecording && _outputFileRecorderPtr)
         {
-            _outputFileRecorderPtr->RecordAudioToFile(audioFrame);
+            _outputFileRecorderPtr->RecordAudioToFile(*audioFrame);
         }
     }
 
     // Measure audio level (0-9)
-    _outputAudioLevel.ComputeLevel(audioFrame);
+    _outputAudioLevel.ComputeLevel(*audioFrame);
 
-    if (capture_start_rtp_time_stamp_ < 0 && audioFrame.timestamp_ != 0) {
+    if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) {
       // The first frame with a valid rtp timestamp.
-      capture_start_rtp_time_stamp_ = audioFrame.timestamp_;
+      capture_start_rtp_time_stamp_ = audioFrame->timestamp_;
     }
 
     if (capture_start_rtp_time_stamp_ >= 0) {
@@ -601,22 +661,22 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
 
       // Compute elapsed time.
       int64_t unwrap_timestamp =
-          rtp_ts_wraparound_handler_->Unwrap(audioFrame.timestamp_);
-      audioFrame.elapsed_time_ms_ =
+          rtp_ts_wraparound_handler_->Unwrap(audioFrame->timestamp_);
+      audioFrame->elapsed_time_ms_ =
           (unwrap_timestamp - capture_start_rtp_time_stamp_) /
           (GetPlayoutFrequency() / 1000);
 
       {
         CriticalSectionScoped lock(ts_stats_lock_.get());
         // Compute ntp time.
-        audioFrame.ntp_time_ms_ = ntp_estimator_.Estimate(
-            audioFrame.timestamp_);
+        audioFrame->ntp_time_ms_ = ntp_estimator_.Estimate(
+            audioFrame->timestamp_);
         // |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
-        if (audioFrame.ntp_time_ms_ > 0) {
+        if (audioFrame->ntp_time_ms_ > 0) {
           // Compute |capture_start_ntp_time_ms_| so that
           // |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_|
           capture_start_ntp_time_ms_ =
-              audioFrame.ntp_time_ms_ - audioFrame.elapsed_time_ms_;
+              audioFrame->ntp_time_ms_ - audioFrame->elapsed_time_ms_;
         }
       }
     }
@@ -625,7 +685,7 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
 }
 
 int32_t
-Channel::NeededFrequency(int32_t id)
+Channel::NeededFrequency(int32_t id) const
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::NeededFrequency(id=%d)", id);
@@ -664,17 +724,16 @@ Channel::NeededFrequency(int32_t id)
     return(highestNeeded);
 }
 
-int32_t
-Channel::CreateChannel(Channel*& channel,
-                       int32_t channelId,
-                       uint32_t instanceId,
-                       const Config& config)
-{
+int32_t Channel::CreateChannel(Channel*& channel,
+                               int32_t channelId,
+                               uint32_t instanceId,
+                               RtcEventLog* const event_log,
+                               const Config& config) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
                  "Channel::CreateChannel(channelId=%d, instanceId=%d)",
         channelId, instanceId);
 
-    channel = new Channel(channelId, instanceId, config);
+    channel = new Channel(channelId, instanceId, event_log, config);
     if (channel == NULL)
     {
         WEBRTC_TRACE(kTraceMemory, kTraceVoice,
@@ -749,104 +808,128 @@ Channel::RecordFileEnded(int32_t id)
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
-                 const Config& config) :
-    _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
-    _instanceId(instanceId),
-    _channelId(channelId),
-    rtp_header_parser_(RtpHeaderParser::Create()),
-    rtp_payload_registry_(
-        new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
-    rtp_receive_statistics_(ReceiveStatistics::Create(
-        Clock::GetRealTimeClock())),
-    rtp_receiver_(RtpReceiver::CreateAudioReceiver(
-        VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
-        this, this, rtp_payload_registry_.get())),
-    telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
-    audio_coding_(AudioCodingModule::Create(
-        VoEModuleId(instanceId, channelId))),
-    _rtpDumpIn(*RtpDump::CreateRtpDump()),
-    _rtpDumpOut(*RtpDump::CreateRtpDump()),
-    _outputAudioLevel(),
-    _externalTransport(false),
-    _inputFilePlayerPtr(NULL),
-    _outputFilePlayerPtr(NULL),
-    _outputFileRecorderPtr(NULL),
-    // Avoid conflict with other channels by adding 1024 - 1026,
-    // won't use as much as 1024 channels.
-    _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
-    _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
-    _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
-    _outputFileRecording(false),
-    _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
-    _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
-    _outputExternalMedia(false),
-    _inputExternalMediaCallbackPtr(NULL),
-    _outputExternalMediaCallbackPtr(NULL),
-    _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
-    _sendTelephoneEventPayloadType(106),
-    ntp_estimator_(Clock::GetRealTimeClock()),
-    jitter_buffer_playout_timestamp_(0),
-    playout_timestamp_rtp_(0),
-    playout_timestamp_rtcp_(0),
-    playout_delay_ms_(0),
-    _numberOfDiscardedPackets(0),
-    send_sequence_number_(0),
-    ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-    rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
-    capture_start_rtp_time_stamp_(-1),
-    capture_start_ntp_time_ms_(-1),
-    _engineStatisticsPtr(NULL),
-    _outputMixerPtr(NULL),
-    _transmitMixerPtr(NULL),
-    _moduleProcessThreadPtr(NULL),
-    _audioDeviceModulePtr(NULL),
-    _voiceEngineObserverPtr(NULL),
-    _callbackCritSectPtr(NULL),
-    _transportPtr(NULL),
-    _rxVadObserverPtr(NULL),
-    _oldVadDecision(-1),
-    _sendFrameType(0),
-    _externalMixing(false),
-    _mixFileWithMicrophone(false),
-    _mute(false),
-    _panLeft(1.0f),
-    _panRight(1.0f),
-    _outputGain(1.0f),
-    _playOutbandDtmfEvent(false),
-    _playInbandDtmfEvent(false),
-    _lastLocalTimeStamp(0),
-    _lastPayloadType(0),
-    _includeAudioLevelIndication(false),
-    _outputSpeechType(AudioFrame::kNormalSpeech),
-    vie_network_(NULL),
-    video_channel_(-1),
-    _average_jitter_buffer_delay_us(0),
-    least_required_delay_ms_(0),
-    _previousTimestamp(0),
-    _recPacketDelayMs(20),
-    _current_sync_offset(0),
-    _RxVadDetection(false),
-    _rxAgcIsEnabled(false),
-    _rxNsIsEnabled(false),
-    restored_packet_in_use_(false),
-    rtcp_observer_(new VoERtcpObserver(this)),
-    network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock()))
-{
+                 RtcEventLog* const event_log,
+                 const Config& config)
+    : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      volume_settings_critsect_(
+          *CriticalSectionWrapper::CreateCriticalSection()),
+      _instanceId(instanceId),
+      _channelId(channelId),
+      event_log_(event_log),
+      rtp_header_parser_(RtpHeaderParser::Create()),
+      rtp_payload_registry_(
+          new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
+      rtp_receive_statistics_(
+          ReceiveStatistics::Create(Clock::GetRealTimeClock())),
+      rtp_receiver_(
+          RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
+                                           this,
+                                           this,
+                                           this,
+                                           rtp_payload_registry_.get())),
+      telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
+      _outputAudioLevel(),
+      _externalTransport(false),
+      _inputFilePlayerPtr(NULL),
+      _outputFilePlayerPtr(NULL),
+      _outputFileRecorderPtr(NULL),
+      // Avoid conflict with other channels by adding 1024 - 1026,
+      // won't use as much as 1024 channels.
+      _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
+      _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
+      _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
+      _outputFileRecording(false),
+      _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
+      _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
+      _outputExternalMedia(false),
+      _inputExternalMediaCallbackPtr(NULL),
+      _outputExternalMediaCallbackPtr(NULL),
+      _timeStamp(0),  // This is just an offset, RTP module will add it's own
+                      // random offset
+      _sendTelephoneEventPayloadType(106),
+      ntp_estimator_(Clock::GetRealTimeClock()),
+      jitter_buffer_playout_timestamp_(0),
+      playout_timestamp_rtp_(0),
+      playout_timestamp_rtcp_(0),
+      playout_delay_ms_(0),
+      _numberOfDiscardedPackets(0),
+      send_sequence_number_(0),
+      ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+      rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
+      capture_start_rtp_time_stamp_(-1),
+      capture_start_ntp_time_ms_(-1),
+      _engineStatisticsPtr(NULL),
+      _outputMixerPtr(NULL),
+      _transmitMixerPtr(NULL),
+      _moduleProcessThreadPtr(NULL),
+      _audioDeviceModulePtr(NULL),
+      _voiceEngineObserverPtr(NULL),
+      _callbackCritSectPtr(NULL),
+      _transportPtr(NULL),
+      _rxVadObserverPtr(NULL),
+      _oldVadDecision(-1),
+      _sendFrameType(0),
+      _externalMixing(false),
+      _mixFileWithMicrophone(false),
+      _mute(false),
+      _panLeft(1.0f),
+      _panRight(1.0f),
+      _outputGain(1.0f),
+      _playOutbandDtmfEvent(false),
+      _playInbandDtmfEvent(false),
+      _lastLocalTimeStamp(0),
+      _lastPayloadType(0),
+      _includeAudioLevelIndication(false),
+      _outputSpeechType(AudioFrame::kNormalSpeech),
+      video_sync_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+      _average_jitter_buffer_delay_us(0),
+      _previousTimestamp(0),
+      _recPacketDelayMs(20),
+      _current_sync_offset(0),      
+      _RxVadDetection(false),
+      _rxAgcIsEnabled(false),
+      _rxNsIsEnabled(false),
+      restored_packet_in_use_(false),
+      rtcp_observer_(new VoERtcpObserver(this)),
+      network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
+      assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
+      associate_send_channel_(ChannelOwner(nullptr)),
+      pacing_enabled_(config.Get<VoicePacing>().enabled),
+      feedback_observer_proxy_(pacing_enabled_ ? new TransportFeedbackProxy()
+                                               : nullptr),
+      seq_num_allocator_proxy_(
+          pacing_enabled_ ? new TransportSequenceNumberProxy() : nullptr),
+      rtp_packet_sender_proxy_(pacing_enabled_ ? new RtpPacketSenderProxy()
+                                               : nullptr) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Channel() - ctor");
+    AudioCodingModule::Config acm_config;
+    acm_config.id = VoEModuleId(instanceId, channelId);
+    if (config.Get<NetEqCapacityConfig>().enabled) {
+      // Clamping the buffer capacity at 20 packets. While going lower will
+      // probably work, it makes little sense.
+      acm_config.neteq_config.max_packets_in_buffer =
+          std::max(20, config.Get<NetEqCapacityConfig>().capacity);
+    }
+    acm_config.neteq_config.enable_fast_accelerate =
+        config.Get<NetEqFastAccelerate>().enabled;
+    audio_coding_.reset(AudioCodingModule::Create(acm_config));
+
     _inbandDtmfQueue.ResetDtmf();
     _inbandDtmfGenerator.Init();
     _outputAudioLevel.Clear();
 
     RtpRtcp::Configuration configuration;
-    configuration.id = VoEModuleId(instanceId, channelId);
     configuration.audio = true;
     configuration.outgoing_transport = this;
     configuration.audio_messages = this;
     configuration.receive_statistics = rtp_receive_statistics_.get();
     configuration.bandwidth_callback = rtcp_observer_.get();
+    configuration.paced_sender = rtp_packet_sender_proxy_.get();
+    configuration.transport_sequence_number_allocator =
+        seq_num_allocator_proxy_.get();
+    configuration.transport_feedback_callback = feedback_observer_proxy_.get();
 
     _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
 
@@ -929,12 +1012,6 @@ Channel::~Channel()
     // End of modules shutdown
 
     // Delete other objects
-    if (vie_network_) {
-      vie_network_->Release();
-      vie_network_ = NULL;
-    }
-    RtpDump::DestroyRtpDump(&_rtpDumpIn);
-    RtpDump::DestroyRtpDump(&_rtpDumpOut);
     delete &_callbackCritSect;
     delete &_fileCritSect;
     delete &volume_settings_critsect_;
@@ -965,13 +1042,7 @@ Channel::Init()
 
     // --- ACM initialization
 
-    if ((audio_coding_->InitializeReceiver() == -1)
-#ifdef WEBRTC_CODEC_AVT
-        // out-of-band Dtmf tones are played out by default
-        || (audio_coding_->SetDtmfPlayoutStatus(true) == -1)
-#endif
-       )
-    {
+    if (audio_coding_->InitializeReceiver() == -1) {
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
             "Channel::Init() unable to initialize the ACM - 1");
@@ -987,7 +1058,7 @@ Channel::Init()
     // be transmitted since the Transport object will then be invalid.
     telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
     // RTCP is enabled by default.
-    _rtpRtcpModule->SetRTCPStatus(kRtcpCompound);
+    _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
     // --- Register all permanent callbacks
     const bool fail =
         (audio_coding_->RegisterTransportCallback(this) == -1) ||
@@ -1020,8 +1091,8 @@ Channel::Init()
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                          VoEId(_instanceId,_channelId),
-                         "Channel::Init() unable to register %s (%d/%d/%d/%d) "
-                         "to RTP/RTCP receiver",
+                         "Channel::Init() unable to register %s "
+                         "(%d/%d/%" PRIuS "/%d) to RTP/RTCP receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
@@ -1029,8 +1100,8 @@ Channel::Init()
         {
             WEBRTC_TRACE(kTraceInfo, kTraceVoice,
                          VoEId(_instanceId,_channelId),
-                         "Channel::Init() %s (%d/%d/%d/%d) has been added to "
-                         "the RTP/RTCP receiver",
+                         "Channel::Init() %s (%d/%d/%" PRIuS "/%d) has been "
+                         "added to the RTP/RTCP receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
@@ -1086,11 +1157,11 @@ Channel::Init()
     }
 
     if (rx_audioproc_->noise_suppression()->set_level(kDefaultNsMode) != 0) {
-      LOG_FERR1(LS_ERROR, noise_suppression()->set_level, kDefaultNsMode);
+      LOG(LS_ERROR) << "noise_suppression()->set_level(kDefaultNsMode) failed.";
       return -1;
     }
     if (rx_audioproc_->gain_control()->set_mode(kDefaultRxAgcMode) != 0) {
-      LOG_FERR1(LS_ERROR, gain_control()->set_mode, kDefaultRxAgcMode);
+      LOG(LS_ERROR) << "gain_control()->set_mode(kDefaultRxAgcMode) failed.";
       return -1;
     }
 
@@ -1122,10 +1193,15 @@ int32_t
 Channel::UpdateLocalTimeStamp()
 {
 
-    _timeStamp += _audioFrame.samples_per_channel_;
+    _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
     return 0;
 }
 
+void Channel::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+  CriticalSectionScoped cs(&_callbackCritSect);
+  audio_sink_ = std::move(sink);
+}
+
 int32_t
 Channel::StartPlayout()
 {
@@ -1231,8 +1307,7 @@ Channel::StopSend()
 
     // Reset sending SSRC and sequence number and triggers direct transmission
     // of RTCP BYE
-    if (_rtpRtcpModule->SetSendingStatus(false) == -1 ||
-        _rtpRtcpModule->ResetSendDataCountersRTP() == -1)
+    if (_rtpRtcpModule->SetSendingStatus(false) == -1)
     {
         _engineStatisticsPtr->SetLastError(
             VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
@@ -1309,7 +1384,12 @@ Channel::DeRegisterVoiceEngineObserver()
 int32_t
 Channel::GetSendCodec(CodecInst& codec)
 {
-    return (audio_coding_->SendCodec(&codec));
+  auto send_codec = audio_coding_->SendCodec();
+  if (send_codec) {
+    codec = *send_codec;
+    return 0;
+  }
+  return -1;
 }
 
 int32_t
@@ -1354,6 +1434,12 @@ Channel::SetSendCodec(const CodecInst& codec)
     return 0;
 }
 
+void Channel::SetBitRate(int bitrate_bps) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
+  audio_coding_->SetBitRate(bitrate_bps);
+}
+
 void Channel::OnIncomingFractionLoss(int fraction_lost) {
   network_predictor_->UpdatePacketLossRate(fraction_lost);
   uint8_t average_fraction_loss = network_predictor_->GetLossRate();
@@ -1386,8 +1472,6 @@ Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
 int32_t
 Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetVADStatus");
     if (audio_coding_->VAD(&disabledDTX, &enabledVAD, &mode) != 0)
     {
         _engineStatisticsPtr->SetLastError(
@@ -1494,8 +1578,6 @@ Channel::SetRecPayloadType(const CodecInst& codec)
 int32_t
 Channel::GetRecPayloadType(CodecInst& codec)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetRecPayloadType()");
     int8_t payloadType(-1);
     if (rtp_payload_registry_->ReceivePayloadType(
         codec.plname,
@@ -1510,8 +1592,6 @@ Channel::GetRecPayloadType(CodecInst& codec)
         return -1;
     }
     codec.pltype = payloadType;
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetRecPayloadType() => pltype=%d", codec.pltype);
     return 0;
 }
 
@@ -1523,7 +1603,7 @@ Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
 
     CodecInst codec;
     int32_t samplingFreqHz(-1);
-    const int kMono = 1;
+    const size_t kMono = 1;
     if (frequency == kFreq32000Hz)
         samplingFreqHz = 32000;
     else if (frequency == kFreq16000Hz)
@@ -1580,7 +1660,7 @@ int Channel::SetOpusMaxPlaybackRate(int frequency_hz) {
 int Channel::SetOpusDtx(bool enable_dtx) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetOpusDtx(%d)", enable_dtx);
-  int ret = enable_dtx ? audio_coding_->EnableOpusDtx(true)
+  int ret = enable_dtx ? audio_coding_->EnableOpusDtx()
                        : audio_coding_->DisableOpusDtx();
   if (ret != 0) {
     _engineStatisticsPtr->SetLastError(
@@ -1640,13 +1720,6 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, size_t length,
   // Store playout timestamp for the received RTP packet
   UpdatePlayoutTimestamp(false);
 
-  // Dump the RTP packet to a file (if RTP dump is enabled).
-  if (_rtpDumpIn.DumpPacket((const uint8_t*)data,
-                            (uint16_t)length) == -1) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId,_channelId),
-                 "Channel::SendPacket() RTP dump to input file failed");
-  }
   const uint8_t* received_packet = reinterpret_cast<const uint8_t*>(data);
   RTPHeader header;
   if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
@@ -1663,22 +1736,6 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, size_t length,
       IsPacketRetransmitted(header, in_order));
   rtp_payload_registry_->SetIncomingPayloadType(header);
 
-  // Forward any packets to ViE bandwidth estimator, if enabled.
-  {
-    CriticalSectionScoped cs(&_callbackCritSect);
-    if (vie_network_) {
-      int64_t arrival_time_ms;
-      if (packet_time.timestamp != -1) {
-        arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-      } else {
-        arrival_time_ms = TickTime::MillisecondTimestamp();
-      }
-      size_t payload_length = length - header.headerLength;
-      vie_network_->ReceivedBWEPacket(video_channel_, arrival_time_ms,
-                                      payload_length, header);
-    }
-  }
-
   return ReceivePacket(received_packet, length, header, in_order) ? 0 : -1;
 }
 
@@ -1717,16 +1774,15 @@ bool Channel::HandleRtxPacket(const uint8_t* packet,
                  "Multiple RTX headers detected, dropping packet");
     return false;
   }
-  uint8_t* restored_packet_ptr = restored_packet_;
   if (!rtp_payload_registry_->RestoreOriginalPacket(
-      &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
-      header)) {
+          restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
+          header)) {
     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
                  "Incoming RTX packet: invalid RTP header");
     return false;
   }
   restored_packet_in_use_ = true;
-  bool ret = OnRecoveredPacket(restored_packet_ptr, packet_length);
+  bool ret = OnRecoveredPacket(restored_packet_, packet_length);
   restored_packet_in_use_ = false;
   return ret;
 }
@@ -1761,13 +1817,6 @@ int32_t Channel::ReceivedRTCPPacket(const int8_t* data, size_t length) {
   // Store playout timestamp for the received RTCP packet
   UpdatePlayoutTimestamp(true);
 
-  // Dump the RTCP packet to a file (if RTP dump is enabled).
-  if (_rtpDumpIn.DumpPacket((const uint8_t*)data, length) == -1) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId,_channelId),
-                 "Channel::SendPacket() RTCP dump to input file failed");
-  }
-
   // Deliver RTCP packet to RTP/RTCP module for parsing
   if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, length) == -1) {
     _engineStatisticsPtr->SetLastError(
@@ -1775,21 +1824,22 @@ int32_t Channel::ReceivedRTCPPacket(const int8_t* data, size_t length) {
         "Channel::IncomingRTPPacket() RTCP packet is invalid");
   }
 
+  int64_t rtt = GetRTT(true);
+  if (rtt == 0) {
+    // Waiting for valid RTT.
+    return 0;
+  }
+  uint32_t ntp_secs = 0;
+  uint32_t ntp_frac = 0;
+  uint32_t rtp_timestamp = 0;
+  if (0 != _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
+                                     &rtp_timestamp)) {
+    // Waiting for RTCP.
+    return 0;
+  }
+
   {
     CriticalSectionScoped lock(ts_stats_lock_.get());
-    int64_t rtt = GetRTT();
-    if (rtt == 0) {
-      // Waiting for valid RTT.
-      return 0;
-    }
-    uint32_t ntp_secs = 0;
-    uint32_t ntp_frac = 0;
-    uint32_t rtp_timestamp = 0;
-    if (0 != _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
-                                       &rtp_timestamp)) {
-      // Waiting for RTCP.
-      return 0;
-    }
     ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
   }
   return 0;
@@ -1952,9 +2002,6 @@ int Channel::StopPlayingFileLocally()
 
     if (!channel_state_.Get().output_file_playing)
     {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceWarning,
-            "StopPlayingFileLocally() isnot playing");
         return 0;
     }
 
@@ -1990,9 +2037,6 @@ int Channel::StopPlayingFileLocally()
 
 int Channel::IsPlayingFileLocally() const
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::IsPlayingFileLocally()");
-
     return channel_state_.Get().output_file_playing;
 }
 
@@ -2176,9 +2220,6 @@ int Channel::StopPlayingFileAsMicrophone()
 
     if (!channel_state_.Get().input_file_playing)
     {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceWarning,
-            "StopPlayingFileAsMicrophone() isnot playing");
         return 0;
     }
 
@@ -2199,8 +2240,6 @@ int Channel::StopPlayingFileAsMicrophone()
 
 int Channel::IsPlayingFileAsMicrophone() const
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::IsPlayingFileAsMicrophone()");
     return channel_state_.Get().input_file_playing;
 }
 
@@ -2402,9 +2441,6 @@ Channel::GetSpeechOutputLevel(uint32_t& level) const
 {
     int8_t currentLevel = _outputAudioLevel.Level();
     level = static_cast<int32_t> (currentLevel);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetSpeechOutputLevel() => level=%u", level);
     return 0;
 }
 
@@ -2413,9 +2449,6 @@ Channel::GetSpeechOutputLevelFullRange(uint32_t& level) const
 {
     int16_t currentLevel = _outputAudioLevel.LevelFullRange();
     level = static_cast<int32_t> (currentLevel);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetSpeechOutputLevelFullRange() => level=%u", level);
     return 0;
 }
 
@@ -2453,9 +2486,6 @@ Channel::GetOutputVolumePan(float& left, float& right) const
     CriticalSectionScoped cs(&volume_settings_critsect_);
     left = _panLeft;
     right = _panRight;
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right);
     return 0;
 }
 
@@ -2474,9 +2504,6 @@ Channel::GetChannelOutputVolumeScaling(float& scaling) const
 {
     CriticalSectionScoped cs(&volume_settings_critsect_);
     scaling = _outputGain;
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling);
     return 0;
 }
 
@@ -2487,6 +2514,9 @@ int Channel::SendTelephoneEventOutband(unsigned char eventCode,
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
                playDtmfEvent);
+    if (!Sending()) {
+      return -1;
+    }
 
     _playOutbandDtmfEvent = playDtmfEvent;
 
@@ -2551,12 +2581,7 @@ Channel::SetSendTelephoneEventPayloadType(unsigned char type)
 int
 Channel::GetSendTelephoneEventPayloadType(unsigned char& type)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetSendTelephoneEventPayloadType()");
     type = _sendTelephoneEventPayloadType;
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetSendTelephoneEventPayloadType() => type=%u", type);
     return 0;
 }
 
@@ -2624,9 +2649,6 @@ int
 Channel::VoiceActivityIndicator(int &activity)
 {
     activity = _sendFrameType;
-
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::VoiceActivityIndicator(indicator=%d)", activity);
     return 0;
 }
 
@@ -2684,9 +2706,6 @@ Channel::SetRxAgcStatus(bool enable, AgcModes mode)
 int
 Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                     "Channel::GetRxAgcStatus(enable=?, mode=?)");
-
     bool enable = rx_audioproc_->gain_control()->is_enabled();
     GainControl::Mode agcMode =
         rx_audioproc_->gain_control()->mode();
@@ -2750,9 +2769,6 @@ Channel::SetRxAgcConfig(AgcConfig config)
 int
 Channel::GetRxAgcConfig(AgcConfig& config)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetRxAgcConfig(config=%?)");
-
     config.targetLeveldBOv =
         rx_audioproc_->gain_control()->target_level_dbfs();
     config.digitalCompressionGaindB =
@@ -2760,14 +2776,6 @@ Channel::GetRxAgcConfig(AgcConfig& config)
     config.limiterEnable =
         rx_audioproc_->gain_control()->is_limiter_enabled();
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId), "GetRxAgcConfig() => "
-                   "targetLeveldBOv=%u, digitalCompressionGaindB=%u,"
-                   " limiterEnable=%d",
-                   config.targetLeveldBOv,
-                   config.digitalCompressionGaindB,
-                   config.limiterEnable);
-
     return 0;
 }
 
@@ -2833,9 +2841,6 @@ Channel::SetRxNsStatus(bool enable, NsModes mode)
 int
 Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetRxNsStatus(enable=?, mode=?)");
-
     bool enable =
         rx_audioproc_->noise_suppression()->is_enabled();
     NoiseSuppression::Level ncLevel =
@@ -2859,9 +2864,6 @@ Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
             break;
     }
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode);
     return 0;
 }
 
@@ -2887,9 +2889,6 @@ int
 Channel::GetLocalSSRC(unsigned int& ssrc)
 {
     ssrc = _rtpRtcpModule->SSRC();
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId,_channelId),
-                 "GetLocalSSRC() => ssrc=%lu", ssrc);
     return 0;
 }
 
@@ -2897,9 +2896,6 @@ int
 Channel::GetRemoteSSRC(unsigned int& ssrc)
 {
     ssrc = rtp_receiver_->SSRC();
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId,_channelId),
-                 "GetRemoteSSRC() => ssrc=%lu", ssrc);
     return 0;
 }
 
@@ -2933,20 +2929,49 @@ int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
   return 0;
 }
 
+void Channel::EnableSendTransportSequenceNumber(int id) {
+  int ret =
+      SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
+  RTC_DCHECK_EQ(0, ret);
+}
+
+void Channel::SetCongestionControlObjects(
+    RtpPacketSender* rtp_packet_sender,
+    TransportFeedbackObserver* transport_feedback_observer,
+    PacketRouter* packet_router) {
+  RTC_DCHECK(packet_router != nullptr || packet_router_ != nullptr);
+  if (transport_feedback_observer) {
+    RTC_DCHECK(feedback_observer_proxy_.get());
+    feedback_observer_proxy_->SetTransportFeedbackObserver(
+        transport_feedback_observer);
+  }
+  if (rtp_packet_sender) {
+    RTC_DCHECK(rtp_packet_sender_proxy_.get());
+    rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
+  }
+  if (seq_num_allocator_proxy_.get()) {
+    seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
+  }
+  _rtpRtcpModule->SetStorePacketsStatus(rtp_packet_sender != nullptr, 600);
+  if (packet_router != nullptr) {
+    packet_router->AddRtpModule(_rtpRtcpModule.get());
+  } else {
+    packet_router_->RemoveRtpModule(_rtpRtcpModule.get());
+  }
+  packet_router_ = packet_router;
+}
+
 void Channel::SetRTCPStatus(bool enable) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRTCPStatus()");
-  _rtpRtcpModule->SetRTCPStatus(enable ? kRtcpCompound : kRtcpOff);
+  _rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
 }
 
 int
 Channel::GetRTCPStatus(bool& enabled)
 {
-    RTCPMethod method = _rtpRtcpModule->RTCP();
-    enabled = (method != kRtcpOff);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId,_channelId),
-                 "GetRTCPStatus() => enabled=%d", enabled);
+  RtcpMode method = _rtpRtcpModule->RTCP();
+  enabled = (method != RtcpMode::kOff);
     return 0;
 }
 
@@ -2985,9 +3010,6 @@ Channel::GetRemoteRTCP_CNAME(char cName[256])
         return -1;
     }
     strcpy(cName, cname);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRemoteRTCP_CNAME() => cName=%s", cName);
     return 0;
 }
 
@@ -3037,7 +3059,7 @@ Channel::GetRemoteRTCPReceiverInfo(
                                          &receivedPacketCount,
                                          &receivedOctetCount) != 0) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
+                  VoEId(_instanceId, _channelId),
                  "GetRemoteRTCPReceiverInfo() failed to retrieve RTT from "
                  "the RTP/RTCP module");
     NTPHigh = 0;
@@ -3061,7 +3083,7 @@ Channel::GetRemoteRTCPReceiverInfo(
       != 0)
   {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
+                  VoEId(_instanceId, _channelId),
                  "GetRTPStatistics() failed to retrieve RTT from "
                  "the RTP/RTCP module");
   }
@@ -3098,9 +3120,8 @@ Channel::SendApplicationDefinedRTCPPacket(unsigned char subType,
             "SendApplicationDefinedRTCPPacket() invalid length value");
         return -1;
     }
-    RTCPMethod status = _rtpRtcpModule->RTCP();
-    if (status == kRtcpOff)
-    {
+    RtcpMode status = _rtpRtcpModule->RTCP();
+    if (status == RtcpMode::kOff) {
         _engineStatisticsPtr->SetLastError(
             VE_RTCP_ERROR, kTraceError,
             "SendApplicationDefinedRTCPPacket() RTCP is disabled");
@@ -3131,7 +3152,7 @@ Channel::GetRTPStatistics(
 {
     // The jitter statistics is updated for each received RTP packet and is
     // based on received packets.
-    if (_rtpRtcpModule->RTCP() == kRtcpOff) {
+    if (_rtpRtcpModule->RTCP() == RtcpMode::kOff) {
       // If RTCP is off, there is no timed thread in the RTCP module regularly
       // generating new stats, trigger the update manually here instead.
       StreamStatistician* statistician =
@@ -3155,11 +3176,6 @@ Channel::GetRTPStatistics(
 
     discardedPackets = _numberOfDiscardedPackets;
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId, _channelId),
-               "GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu,"
-               " discardedPackets = %lu)",
-               averageJitterMs, maxJitterMs, discardedPackets);
     return 0;
 }
 
@@ -3176,8 +3192,6 @@ int Channel::GetRemoteRTCPReportBlocks(
   // report block according to RFC 3550.
   std::vector<RTCPReportBlock> rtcp_report_blocks;
   if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) {
-    _engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-        "GetRemoteRTCPReportBlocks() failed to read RTCP SR/RR report block.");
     return -1;
   }
 
@@ -3210,8 +3224,9 @@ Channel::GetRTPStatistics(CallStatistics& stats)
     RtcpStatistics statistics;
     StreamStatistician* statistician =
         rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
-    if (!statistician || !statistician->GetStatistics(
-        &statistics, _rtpRtcpModule->RTCP() == kRtcpOff)) {
+    if (!statistician ||
+        !statistician->GetStatistics(
+            &statistics, _rtpRtcpModule->RTCP() == RtcpMode::kOff)) {
       _engineStatisticsPtr->SetLastError(
           VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
           "GetRTPStatistics() failed to read RTP statistics from the "
@@ -3223,22 +3238,8 @@ Channel::GetRTPStatistics(CallStatistics& stats)
     stats.extendedMax = statistics.extended_max_sequence_number;
     stats.jitterSamples = statistics.jitter;
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu,"
-                 " extendedMax=%lu, jitterSamples=%li)",
-                 stats.fractionLost, stats.cumulativeLost, stats.extendedMax,
-                 stats.jitterSamples);
-
     // --- RTT
-    stats.rttMs = GetRTT();
-    if (stats.rttMs == 0) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                   "GetRTPStatistics() failed to get RTT");
-    } else {
-      WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                   "GetRTPStatistics() => rttMs=%" PRId64, stats.rttMs);
-    }
+    stats.rttMs = GetRTT(true);
 
     // --- Data counters
 
@@ -3265,13 +3266,6 @@ Channel::GetRTPStatistics(CallStatistics& stats)
     stats.bytesReceived = bytesReceived;
     stats.packetsReceived = packetsReceived;
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() => bytesSent=%" PRIuS ", packetsSent=%d,"
-                 " bytesReceived=%" PRIuS ", packetsReceived=%d)",
-                 stats.bytesSent, stats.packetsSent, stats.bytesReceived,
-                 stats.packetsReceived);
-
     // --- Timestamps
     {
       CriticalSectionScoped lock(ts_stats_lock_.get());
@@ -3315,9 +3309,8 @@ Channel::GetREDStatus(bool& enabled, int& redPayloadtype)
     enabled = audio_coding_->REDStatus();
     if (enabled)
     {
-        int8_t payloadType(0);
-        if (_rtpRtcpModule->SendREDPayloadType(payloadType) != 0)
-        {
+      int8_t payloadType = 0;
+      if (_rtpRtcpModule->SendREDPayloadType(&payloadType) != 0) {
             _engineStatisticsPtr->SetLastError(
                 VE_RTP_RTCP_MODULE_ERROR, kTraceError,
                 "GetREDStatus() failed to retrieve RED PT from RTP/RTCP "
@@ -3325,15 +3318,8 @@ Channel::GetREDStatus(bool& enabled, int& redPayloadtype)
             return -1;
         }
         redPayloadtype = payloadType;
-        WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                   VoEId(_instanceId, _channelId),
-                   "GetREDStatus() => enabled=%d, redPayloadtype=%d",
-                   enabled, redPayloadtype);
         return 0;
     }
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetREDStatus() => enabled=%d", enabled);
     return 0;
 }
 
@@ -3352,15 +3338,14 @@ int Channel::SetCodecFECStatus(bool enable) {
 
 bool Channel::GetCodecFECStatus() {
   bool enabled = audio_coding_->CodecFEC();
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId, _channelId),
-               "GetCodecFECStatus() => enabled=%d", enabled);
   return enabled;
 }
 
 void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
   // None of these functions can fail.
-  _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
+  // If pacing is enabled we always store packets.
+  if (!pacing_enabled_)
+    _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
   rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
   rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
   if (enable)
@@ -3374,97 +3359,6 @@ int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
   return _rtpRtcpModule->SendNACK(sequence_numbers, length);
 }
 
-int
-Channel::StartRTPDump(const char fileNameUTF8[1024],
-                      RTPDirections direction)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::StartRTPDump()");
-    if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_ARGUMENT, kTraceError,
-            "StartRTPDump() invalid RTP direction");
-        return -1;
-    }
-    RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
-        &_rtpDumpIn : &_rtpDumpOut;
-    if (rtpDumpPtr == NULL)
-    {
-        assert(false);
-        return -1;
-    }
-    if (rtpDumpPtr->IsActive())
-    {
-        rtpDumpPtr->Stop();
-    }
-    if (rtpDumpPtr->Start(fileNameUTF8) != 0)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_BAD_FILE, kTraceError,
-            "StartRTPDump() failed to create file");
-        return -1;
-    }
-    return 0;
-}
-
-int
-Channel::StopRTPDump(RTPDirections direction)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::StopRTPDump()");
-    if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_ARGUMENT, kTraceError,
-            "StopRTPDump() invalid RTP direction");
-        return -1;
-    }
-    RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
-        &_rtpDumpIn : &_rtpDumpOut;
-    if (rtpDumpPtr == NULL)
-    {
-        assert(false);
-        return -1;
-    }
-    if (!rtpDumpPtr->IsActive())
-    {
-        return 0;
-    }
-    return rtpDumpPtr->Stop();
-}
-
-bool
-Channel::RTPDumpIsActive(RTPDirections direction)
-{
-    if ((direction != kRtpIncoming) &&
-        (direction != kRtpOutgoing))
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_ARGUMENT, kTraceError,
-            "RTPDumpIsActive() invalid RTP direction");
-        return false;
-    }
-    RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
-        &_rtpDumpIn : &_rtpDumpOut;
-    return rtpDumpPtr->IsActive();
-}
-
-void Channel::SetVideoEngineBWETarget(ViENetwork* vie_network,
-                                      int video_channel) {
-  CriticalSectionScoped cs(&_callbackCritSect);
-  if (vie_network_) {
-    vie_network_->Release();
-    vie_network_ = NULL;
-  }
-  video_channel_ = -1;
-
-  if (vie_network != NULL && video_channel != -1) {
-    vie_network_ = vie_network;
-    video_channel_ = video_channel;
-  }
-}
-
 uint32_t
 Channel::Demultiplex(const AudioFrame& audioFrame)
 {
@@ -3477,24 +3371,17 @@ Channel::Demultiplex(const AudioFrame& audioFrame)
 
 void Channel::Demultiplex(const int16_t* audio_data,
                           int sample_rate,
-                          int number_of_frames,
-                          int number_of_channels) {
+                          size_t number_of_frames,
+                          size_t number_of_channels) {
   CodecInst codec;
   GetSendCodec(codec);
 
-  if (!mono_recording_audio_.get()) {
-    // Temporary space for DownConvertToCodecFormat.
-    mono_recording_audio_.reset(new int16_t[kMaxMonoDataSizeSamples]);
-  }
-  DownConvertToCodecFormat(audio_data,
-                           number_of_frames,
-                           number_of_channels,
-                           sample_rate,
-                           codec.channels,
-                           codec.plfreq,
-                           mono_recording_audio_.get(),
-                           &input_resampler_,
-                           &_audioFrame);
+  // Never upsample or upmix the capture signal here. This should be done at the
+  // end of the send chain.
+  _audioFrame.sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
+  _audioFrame.num_channels_ = std::min(number_of_channels, codec.channels);
+  RemixAndResample(audio_data, number_of_frames, number_of_channels,
+                   sample_rate, &input_resampler_, &_audioFrame);
 }
 
 uint32_t
@@ -3539,7 +3426,8 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
     InsertInbandDtmfTone();
 
     if (_includeAudioLevelIndication) {
-      int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
+      size_t length =
+          _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
       if (is_muted) {
         rms_level_.ProcessMuted(length);
       } else {
@@ -3580,10 +3468,21 @@ Channel::EncodeAndSend()
         return 0xFFFFFFFF;
     }
 
-    _timeStamp += _audioFrame.samples_per_channel_;
+    _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
     return 0;
 }
 
+void Channel::DisassociateSendChannel(int channel_id) {
+  CriticalSectionScoped lock(assoc_send_channel_lock_.get());
+  Channel* channel = associate_send_channel_.channel();
+  if (channel && channel->ChannelId() == channel_id) {
+    // If this channel is associated with a send channel of the specified
+    // Channel ID, disassociate with it.
+    ChannelOwner ref(NULL);
+    associate_send_channel_ = ref;
+  }
+}
+
 int Channel::RegisterExternalMediaProcessing(
     ProcessingTypes type,
     VoEMediaProcess& processObject)
@@ -3680,8 +3579,6 @@ int Channel::SetExternalMixing(bool enabled) {
 int
 Channel::GetNetworkStatistics(NetworkStatistics& stats)
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetNetworkStatistics()");
     return audio_coding_->GetNetworkStatistics(&stats);
 }
 
@@ -3692,42 +3589,28 @@ void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
 bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
                                int* playout_buffer_delay_ms,
                                int* avsync_offset_ms) const {
+  CriticalSectionScoped cs(video_sync_lock_.get());
   if (_average_jitter_buffer_delay_us == 0) {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetDelayEstimate() no valid estimate.");
     return false;
   }
   *jitter_buffer_delay_ms = (_average_jitter_buffer_delay_us + 500) / 1000 +
       _recPacketDelayMs;
   *playout_buffer_delay_ms = playout_delay_ms_;
   *avsync_offset_ms = _current_sync_offset;
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::GetDelayEstimate()");
   return true;
 }
 
-int Channel::SetInitialPlayoutDelay(int delay_ms)
-{
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::SetInitialPlayoutDelay()");
-  if ((delay_ms < kVoiceEngineMinMinPlayoutDelayMs) ||
-      (delay_ms > kVoiceEngineMaxMinPlayoutDelayMs))
-  {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetInitialPlayoutDelay() invalid min delay");
-    return -1;
-  }
-  if (audio_coding_->SetInitialPlayoutDelay(delay_ms) != 0)
-  {
-    _engineStatisticsPtr->SetLastError(
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "SetInitialPlayoutDelay() failed to set min playout delay");
-    return -1;
-  }
-  return 0;
+uint32_t Channel::GetDelayEstimate() const {
+  int jitter_buffer_delay_ms = 0;
+  int playout_buffer_delay_ms = 0;
+  int avsync_offset_ms = 0;
+  GetDelayEstimate(&jitter_buffer_delay_ms, &playout_buffer_delay_ms, &avsync_offset_ms);
+  return jitter_buffer_delay_ms + playout_buffer_delay_ms;
 }
 
+int Channel::LeastRequiredDelayMs() const {
+  return audio_coding_->LeastRequiredDelayMs();
+}
 
 int
 Channel::SetMinimumPlayoutDelay(int delayMs)
@@ -3752,56 +3635,19 @@ Channel::SetMinimumPlayoutDelay(int delayMs)
     return 0;
 }
 
-void Channel::UpdatePlayoutTimestamp(bool rtcp) {
-  uint32_t playout_timestamp = 0;
-
-  if (audio_coding_->PlayoutTimestamp(&playout_timestamp) == -1)  {
-    // This can happen if this channel has not been received any RTP packet. In
-    // this case, NetEq is not capable of computing playout timestamp.
-    return;
-  }
-
-  uint16_t delay_ms = 0;
-  if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::UpdatePlayoutTimestamp() failed to read playout"
-                 " delay from the ADM");
-    _engineStatisticsPtr->SetLastError(
-        VE_CANNOT_RETRIEVE_VALUE, kTraceError,
-        "UpdatePlayoutTimestamp() failed to retrieve playout delay");
-    return;
-  }
-
-  jitter_buffer_playout_timestamp_ = playout_timestamp;
-
-  // Remove the playout delay.
-  playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
-
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
-               playout_timestamp);
-
-  if (rtcp) {
-    playout_timestamp_rtcp_ = playout_timestamp;
-  } else {
-    playout_timestamp_rtp_ = playout_timestamp;
-  }
-  playout_delay_ms_ = delay_ms;
-}
-
 int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::GetPlayoutTimestamp()");
-  if (playout_timestamp_rtp_ == 0)  {
+  uint32_t playout_timestamp_rtp = 0;
+  {
+    CriticalSectionScoped cs(video_sync_lock_.get());
+    playout_timestamp_rtp = playout_timestamp_rtp_;
+  }
+  if (playout_timestamp_rtp == 0)  {
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_RETRIEVE_VALUE, kTraceError,
         "GetPlayoutTimestamp() failed to retrieve timestamp");
     return -1;
   }
-  timestamp = playout_timestamp_rtp_;
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId,_channelId),
-               "GetPlayoutTimestamp() => timestamp=%u", timestamp);
+  timestamp = playout_timestamp_rtp;
   return 0;
 }
 
@@ -3832,8 +3678,6 @@ int Channel::SetInitSequenceNumber(short sequenceNumber) {
 int
 Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const
 {
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetRtpRtcp()");
     *rtpRtcpModule = _rtpRtcpModule.get();
     *rtp_receiver = rtp_receiver_.get();
     return 0;
@@ -3845,7 +3689,7 @@ int32_t
 Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
 {
   rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
-    int fileSamples(0);
+    size_t fileSamples(0);
 
     {
         CriticalSectionScoped cs(&_fileCritSect);
@@ -3915,7 +3759,7 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
     assert(mixingFrequency <= 48000);
 
     rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[960]);
-    int fileSamples(0);
+    size_t fileSamples(0);
 
     {
         CriticalSectionScoped cs(&_fileCritSect);
@@ -3953,8 +3797,8 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
     else
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
-            "Channel::MixAudioWithFile() samples_per_channel_(%d) != "
-            "fileSamples(%d)",
+            "Channel::MixAudioWithFile() samples_per_channel_(%" PRIuS ") != "
+            "fileSamples(%" PRIuS ")",
             audioFrame.samples_per_channel_, fileSamples);
         return -1;
     }
@@ -4014,15 +3858,16 @@ Channel::InsertInbandDtmfTone()
         }
 
         // Replace mixed audio with DTMF tone.
-        for (int sample = 0;
+        for (size_t sample = 0;
             sample < _audioFrame.samples_per_channel_;
             sample++)
         {
-            for (int channel = 0;
+            for (size_t channel = 0;
                 channel < _audioFrame.num_channels_;
                 channel++)
             {
-                const int index = sample * _audioFrame.num_channels_ + channel;
+                const size_t index =
+                    sample * _audioFrame.num_channels_ + channel;
                 _audioFrame.data_[index] = toneBuffer[sample];
             }
         }
@@ -4036,22 +3881,44 @@ Channel::InsertInbandDtmfTone()
     return 0;
 }
 
-int32_t
-Channel::SendPacketRaw(const void *data, size_t len, bool RTCP)
-{
-    CriticalSectionScoped cs(&_callbackCritSect);
-    if (_transportPtr == NULL)
-    {
-        return -1;
-    }
-    if (!RTCP)
-    {
-        return _transportPtr->SendPacket(_channelId, data, len);
-    }
-    else
-    {
-        return _transportPtr->SendRTCPPacket(_channelId, data, len);
+void Channel::UpdatePlayoutTimestamp(bool rtcp) {
+  uint32_t playout_timestamp = 0;
+
+  if (audio_coding_->PlayoutTimestamp(&playout_timestamp) == -1)  {
+    // This can happen if this channel has not been received any RTP packet. In
+    // this case, NetEq is not capable of computing playout timestamp.
+    return;
+  }
+
+  uint16_t delay_ms = 0;
+  if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
+                 "Channel::UpdatePlayoutTimestamp() failed to read playout"
+                 " delay from the ADM");
+    _engineStatisticsPtr->SetLastError(
+        VE_CANNOT_RETRIEVE_VALUE, kTraceError,
+        "UpdatePlayoutTimestamp() failed to retrieve playout delay");
+    return;
+  }
+
+  jitter_buffer_playout_timestamp_ = playout_timestamp;
+
+  // Remove the playout delay.
+  playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
+
+  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+               "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
+               playout_timestamp);
+
+  {
+    CriticalSectionScoped cs(video_sync_lock_.get());
+    if (rtcp) {
+      playout_timestamp_rtcp_ = playout_timestamp;
+    } else {
+      playout_timestamp_rtp_ = playout_timestamp;
     }
+    playout_delay_ms_ = delay_ms;
+  }
 }
 
 // Called for incoming RTP packets after successful RTP header parsing.
@@ -4064,9 +3931,6 @@ void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
   // Get frequency of last received payload
   int rtp_receive_frequency = GetPlayoutFrequency();
 
-  // Update the least required delay.
-  least_required_delay_ms_ = audio_coding_->LeastRequiredDelayMs();
-
   // |jitter_buffer_playout_timestamp_| updated in UpdatePlayoutTimestamp for
   // every incoming packet.
   uint32_t timestamp_diff_ms = (rtp_timestamp -
@@ -4087,21 +3951,25 @@ void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
 
   if (timestamp_diff_ms == 0) return;
 
-  if (packet_delay_ms >= 10 && packet_delay_ms <= 60) {
-    _recPacketDelayMs = packet_delay_ms;
-  }
+  {
+    CriticalSectionScoped cs(video_sync_lock_.get());
 
-  if (_average_jitter_buffer_delay_us == 0) {
-    _average_jitter_buffer_delay_us = timestamp_diff_ms * 1000;
-    return;
-  }
+    if (packet_delay_ms >= 10 && packet_delay_ms <= 60) {
+      _recPacketDelayMs = packet_delay_ms;
+    }
 
-  // Filter average delay value using exponential filter (alpha is
-  // 7/8). We derive 1000 *_average_jitter_buffer_delay_us here (reduces
-  // risk of rounding error) and compensate for it in GetDelayEstimate()
-  // later.
-  _average_jitter_buffer_delay_us = (_average_jitter_buffer_delay_us * 7 +
-      1000 * timestamp_diff_ms + 500) / 8;
+    if (_average_jitter_buffer_delay_us == 0) {
+      _average_jitter_buffer_delay_us = timestamp_diff_ms * 1000;
+      return;
+    }
+
+    // Filter average delay value using exponential filter (alpha is
+    // 7/8). We derive 1000 *_average_jitter_buffer_delay_us here (reduces
+    // risk of rounding error) and compensate for it in GetDelayEstimate()
+    // later.
+    _average_jitter_buffer_delay_us = (_average_jitter_buffer_delay_us * 7 +
+        1000 * timestamp_diff_ms + 500) / 8;
+  }
 }
 
 void
@@ -4110,7 +3978,6 @@ Channel::RegisterReceiveCodecsToRTPModule()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterReceiveCodecsToRTPModule()");
 
-
     CodecInst codec;
     const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
 
@@ -4125,23 +3992,22 @@ Channel::RegisterReceiveCodecsToRTPModule()
                 codec.channels,
                 (codec.rate < 0) ? 0 : codec.rate) == -1))
         {
-            WEBRTC_TRACE(
-                         kTraceWarning,
+            WEBRTC_TRACE(kTraceWarning,
                          kTraceVoice,
                          VoEId(_instanceId, _channelId),
                          "Channel::RegisterReceiveCodecsToRTPModule() unable"
-                         " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
+                         " to register %s (%d/%d/%" PRIuS "/%d) to RTP/RTCP "
+                         "receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
         else
         {
-            WEBRTC_TRACE(
-                         kTraceInfo,
+            WEBRTC_TRACE(kTraceInfo,
                          kTraceVoice,
                          VoEId(_instanceId, _channelId),
                          "Channel::RegisterReceiveCodecsToRTPModule() %s "
-                         "(%d/%d/%d/%d) has been added to the RTP/RTCP "
+                         "(%d/%d/%" PRIuS "/%d) has been added to the RTP/RTCP "
                          "receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
@@ -4219,15 +4085,29 @@ int32_t Channel::GetPlayoutFrequency() {
   return playout_frequency;
 }
 
-int64_t Channel::GetRTT() const {
-  RTCPMethod method = _rtpRtcpModule->RTCP();
-  if (method == kRtcpOff) {
+int64_t Channel::GetRTT(bool allow_associate_channel) const {
+  RtcpMode method = _rtpRtcpModule->RTCP();
+  if (method == RtcpMode::kOff) {
     return 0;
   }
   std::vector<RTCPReportBlock> report_blocks;
   _rtpRtcpModule->RemoteRTCPStat(&report_blocks);
+
+  int64_t rtt = 0;
   if (report_blocks.empty()) {
-    return 0;
+    if (allow_associate_channel) {
+      CriticalSectionScoped lock(assoc_send_channel_lock_.get());
+      Channel* channel = associate_send_channel_.channel();
+      // Tries to get RTT from an associated channel. This is important for
+      // receive-only channels.
+      if (channel) {
+        // To prevent infinite recursion and deadlock, calling GetRTT of
+        // associate channel should always use "false" for argument:
+        // |allow_associate_channel|.
+        rtt = channel->GetRTT(false);
+      }
+    }
+    return rtt;
   }
 
   uint32_t remoteSSRC = rtp_receiver_->SSRC();
@@ -4243,7 +4123,7 @@ int64_t Channel::GetRTT() const {
     // the SSRC of the other end.
     remoteSSRC = report_blocks[0].remoteSSRC;
   }
-  int64_t rtt = 0;
+
   int64_t avg_rtt = 0;
   int64_t max_rtt= 0;
   int64_t min_rtt = 0;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel.h b/media/webrtc/trunk/webrtc/voice_engine/channel.h
index 5e8c50e27a..0a03a148c1 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.h
@@ -11,18 +11,19 @@
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/file_player.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
@@ -48,16 +49,16 @@ class AudioDeviceModule;
 class Config;
 class CriticalSectionWrapper;
 class FileWrapper;
+class PacketRouter;
 class ProcessThread;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
-class RtpDump;
+class RtcEventLog;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class RTPReceiverAudio;
 class RtpRtcp;
 class TelephoneEventHandler;
-class ViENetwork;
 class VoEMediaProcess;
 class VoERTPObserver;
 class VoiceEngineObserver;
@@ -69,9 +70,12 @@ struct SenderInfo;
 namespace voe {
 
 class OutputMixer;
+class RtpPacketSenderProxy;
 class Statistics;
 class StatisticsProxy;
+class TransportFeedbackProxy;
 class TransmitMixer;
+class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
@@ -172,8 +176,12 @@ public:
     static int32_t CreateChannel(Channel*& channel,
                                  int32_t channelId,
                                  uint32_t instanceId,
+                                 RtcEventLog* const event_log,
                                  const Config& config);
-    Channel(int32_t channelId, uint32_t instanceId, const Config& config);
+    Channel(int32_t channelId,
+            uint32_t instanceId,
+            RtcEventLog* const event_log,
+            const Config& config);
     int32_t Init();
     int32_t SetEngineInformation(
         Statistics& engineStatistics,
@@ -185,6 +193,8 @@ public:
         CriticalSectionWrapper* callbackCritSect);
     int32_t UpdateLocalTimeStamp();
 
+    void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+
     // API methods
 
     // VoEBase
@@ -202,6 +212,7 @@ public:
     int32_t GetSendCodec(CodecInst& codec);
     int32_t GetRecCodec(CodecInst& codec);
     int32_t SetSendCodec(const CodecInst& codec);
+    void SetBitRate(int bitrate_bps);
     int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
     int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
     int32_t SetRecPayloadType(const CodecInst& codec);
@@ -276,12 +287,11 @@ public:
     bool GetDelayEstimate(int* jitter_buffer_delay_ms,
                           int* playout_buffer_delay_ms,
                           int* avsync_offset_ms) const;
-    int least_required_delay_ms() const { return least_required_delay_ms_; }
-    int SetInitialPlayoutDelay(int delay_ms);
+    uint32_t GetDelayEstimate() const;
+    int LeastRequiredDelayMs() const;
     int SetMinimumPlayoutDelay(int delayMs);
     void SetCurrentSyncOffset(int offsetMs) { _current_sync_offset = offsetMs; }
     int GetPlayoutTimestamp(unsigned int& timestamp);
-    void UpdatePlayoutTimestamp(bool rtcp);
     int SetInitTimestamp(unsigned int timestamp);
     int SetInitSequenceNumber(short sequenceNumber);
 
@@ -320,6 +330,13 @@ public:
     int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
+    void EnableSendTransportSequenceNumber(int id);
+
+    void SetCongestionControlObjects(
+        RtpPacketSender* rtp_packet_sender,
+        TransportFeedbackObserver* transport_feedback_observer,
+        PacketRouter* packet_router);
+
     void SetRTCPStatus(bool enable);
     int GetRTCPStatus(bool& enabled);
     int SetRTCP_CNAME(const char cName[256]);
@@ -345,11 +362,6 @@ public:
     int SetCodecFECStatus(bool enable);
     bool GetCodecFECStatus();
     void SetNACKStatus(bool enable, int maxNumberOfPackets);
-    int StartRTPDump(const char fileNameUTF8[1024], RTPDirections direction);
-    int StopRTPDump(RTPDirections direction);
-    bool RTPDumpIsActive(RTPDirections direction);
-    // Takes ownership of the ViENetwork.
-    void SetVideoEngineBWETarget(ViENetwork* vie_network, int video_channel);
 
     // From AudioPacketizationCallback in the ACM
     int32_t SendData(FrameType frameType,
@@ -372,29 +384,28 @@ public:
                            size_t packet_length) override;
 
     // From RtpFeedback in the RTP/RTCP module
-    int32_t OnInitializeDecoder(int32_t id,
-                                int8_t payloadType,
+    int32_t OnInitializeDecoder(int8_t payloadType,
                                 const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                 int frequency,
-                                uint8_t channels,
+                                size_t channels,
                                 uint32_t rate) override;
-    void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
-    void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
-    void ResetStatistics(uint32_t ssrc) override;
+    void OnIncomingSSRCChanged(uint32_t ssrc) override;
+    void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
 
     // From RtpAudioFeedback in the RTP/RTCP module
-    void OnPlayTelephoneEvent(int32_t id,
-                              uint8_t event,
+    void OnPlayTelephoneEvent(uint8_t event,
                               uint16_t lengthMs,
                               uint8_t volume) override;
 
     // From Transport (called by the RTP/RTCP module)
-    int SendPacket(int /*channel*/, const void* data, size_t len) override;
-    int SendRTCPPacket(int /*channel*/, const void* data, size_t len) override;
+    bool SendRtp(const uint8_t* data,
+                 size_t len,
+                 const PacketOptions& packet_options) override;
+    bool SendRtcp(const uint8_t* data, size_t len) override;
 
     // From MixerParticipant
-    int32_t GetAudioFrame(int32_t id, AudioFrame& audioFrame) override;
-    int32_t NeededFrequency(int32_t id) override;
+    int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
+    int32_t NeededFrequency(int32_t id) const override;
 
     // From FileCallback
     void PlayNotification(int32_t id, uint32_t durationMs) override;
@@ -445,11 +456,22 @@ public:
     // does not go through transmit_mixer and APM.
     void Demultiplex(const int16_t* audio_data,
                      int sample_rate,
-                     int number_of_frames,
-                     int number_of_channels);
+                     size_t number_of_frames,
+                     size_t number_of_channels);
     uint32_t PrepareEncodeAndSend(int mixingFrequency);
     uint32_t EncodeAndSend();
 
+    // Associate to a send channel.
+    // Used for obtaining RTT for a receive-only channel.
+    void set_associate_send_channel(const ChannelOwner& channel) {
+      assert(_channelId != channel.channel()->ChannelId());
+      CriticalSectionScoped lock(assoc_send_channel_lock_.get());
+      associate_send_channel_ = channel;
+    }
+
+    // Disassociate a send channel if it was associated.
+    void DisassociateSendChannel(int channel_id);
+
 protected:
     void OnIncomingFractionLoss(int fraction_lost);
 
@@ -465,7 +487,7 @@ private:
     int InsertInbandDtmfTone();
     int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
     int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
-    int32_t SendPacketRaw(const void *data, size_t len, bool RTCP);
+    void UpdatePlayoutTimestamp(bool rtcp);
     void UpdatePacketDelay(uint32_t timestamp,
                            uint16_t sequenceNumber);
     void RegisterReceiveCodecsToRTPModule();
@@ -475,7 +497,7 @@ private:
                                   unsigned char id);
 
     int32_t GetPlayoutFrequency();
-    int64_t GetRTT() const;
+    int64_t GetRTT(bool allow_associate_channel) const;
 
     CriticalSectionWrapper& _fileCritSect;
     CriticalSectionWrapper& _callbackCritSect;
@@ -485,6 +507,8 @@ private:
 
     ChannelState channel_state_;
 
+    RtcEventLog* const event_log_;
+
     rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
     rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
@@ -493,12 +517,10 @@ private:
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
-    RtpDump& _rtpDumpIn;
-    RtpDump& _rtpDumpOut;
+    rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;
-    rtc::scoped_ptr<int16_t[]> mono_recording_audio_;
     // Downsamples to the codec rate if necessary.
     PushResampler<int16_t> input_resampler_;
     FilePlayer* _inputFilePlayerPtr;
@@ -520,9 +542,9 @@ private:
 
     // Timestamp of the audio pulled from NetEq.
     uint32_t jitter_buffer_playout_timestamp_;
-    uint32_t playout_timestamp_rtp_;
+    uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
     uint32_t playout_timestamp_rtcp_;
-    uint32_t playout_delay_ms_;
+    uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
     uint32_t _numberOfDiscardedPackets;
     uint16_t send_sequence_number_;
     uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
@@ -567,13 +589,11 @@ private:
     bool _includeAudioLevelIndication;
     // VoENetwork
     AudioFrame::SpeechType _outputSpeechType;
-    ViENetwork* vie_network_;
-    int video_channel_;
     // VoEVideoSync
-    uint32_t _average_jitter_buffer_delay_us;
-    int least_required_delay_ms_;
+    rtc::scoped_ptr<CriticalSectionWrapper> video_sync_lock_;
+    uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
     uint32_t _previousTimestamp;
-    uint16_t _recPacketDelayMs;
+    uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
     int _current_sync_offset;
     // VoEAudioProcessing
     bool _RxVadDetection;
@@ -583,6 +603,15 @@ private:
     // RtcpBandwidthObserver
     rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
+    // An associated send channel.
+    rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
+    ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
+
+    bool pacing_enabled_;
+    PacketRouter* packet_router_ = nullptr;
+    rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
+    rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
+    rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc
index 78c484f3dc..8aed3e33cd 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/common.h"
 #include "webrtc/voice_engine/channel_manager.h"
 
+#include "webrtc/common.h"
 #include "webrtc/voice_engine/channel.h"
 
 namespace webrtc {
@@ -49,8 +49,8 @@ ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
     : config_(config),
       instance_id_(instance_id),
       last_channel_id_(-1),
-      lock_(CriticalSectionWrapper::CreateCriticalSection())
-      {}
+      lock_(CriticalSectionWrapper::CreateCriticalSection()),
+      event_log_(RtcEventLog::Create()) {}
 
 ChannelOwner ChannelManager::CreateChannel() {
   return CreateChannelInternal(config_);
@@ -62,7 +62,8 @@ ChannelOwner ChannelManager::CreateChannel(const Config& external_config) {
 
 ChannelOwner ChannelManager::CreateChannelInternal(const Config& config) {
   Channel* channel;
-  Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config);
+  Channel::CreateChannel(channel, ++last_channel_id_, instance_id_,
+                         event_log_.get(), config);
   ChannelOwner channel_owner(channel);
 
   CriticalSectionScoped crit(lock_.get());
@@ -95,16 +96,21 @@ void ChannelManager::DestroyChannel(int32_t channel_id) {
   ChannelOwner reference(NULL);
   {
     CriticalSectionScoped crit(lock_.get());
+    std::vector<ChannelOwner>::iterator to_delete = channels_.end();
+    for (auto it = channels_.begin(); it != channels_.end(); ++it) {
+      Channel* channel = it->channel();
+      // For channels associated with the channel to be deleted, disassociate
+      // with that channel.
+      channel->DisassociateSendChannel(channel_id);
 
-    for (std::vector<ChannelOwner>::iterator it = channels_.begin();
-         it != channels_.end();
-         ++it) {
-      if (it->channel()->ChannelId() == channel_id) {
-        reference = *it;
-        channels_.erase(it);
-        break;
+      if (channel->ChannelId() == channel_id) {
+        to_delete = it;
       }
     }
+    if (to_delete != channels_.end()) {
+      reference = *to_delete;
+      channels_.erase(to_delete);
+    }
   }
 }
 
@@ -124,6 +130,10 @@ size_t ChannelManager::NumOfChannels() const {
   return channels_.size();
 }
 
+RtcEventLog* ChannelManager::GetEventLog() const {
+  return event_log_.get();
+}
+
 ChannelManager::Iterator::Iterator(ChannelManager* channel_manager)
     : iterator_pos_(0) {
   channel_manager->GetAllChannels(&channels_);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h
index 27e2865baf..a79156b119 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h
@@ -15,8 +15,9 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -52,8 +53,9 @@ class ChannelOwner {
 
   ChannelOwner& operator=(const ChannelOwner& other);
 
-  Channel* channel() { return channel_ref_->channel.get(); }
+  Channel* channel() const { return channel_ref_->channel.get(); }
   bool IsValid() { return channel_ref_->channel.get() != NULL; }
+  int use_count() const { return channel_ref_->ref_count.Value(); }
  private:
   // Shared instance of a Channel. Copying ChannelOwners increase the reference
   // count and destroying ChannelOwners decrease references. Channels are
@@ -89,7 +91,7 @@ class ChannelManager {
     size_t iterator_pos_;
     std::vector<ChannelOwner> channels_;
 
-    DISALLOW_COPY_AND_ASSIGN(Iterator);
+    RTC_DISALLOW_COPY_AND_ASSIGN(Iterator);
   };
 
   // CreateChannel will always return a valid ChannelOwner instance. The channel
@@ -111,6 +113,9 @@ class ChannelManager {
   size_t NumOfChannels() const;
   const Config& config_;
 
+  // Returns a pointer to the event log object stored within the ChannelManager.
+  RtcEventLog* GetEventLog() const;
+
  private:
   // Create a channel given a configuration, |config|.
   ChannelOwner CreateChannelInternal(const Config& config);
@@ -122,7 +127,9 @@ class ChannelManager {
   rtc::scoped_ptr<CriticalSectionWrapper> lock_;
   std::vector<ChannelOwner> channels_;
 
-  DISALLOW_COPY_AND_ASSIGN(ChannelManager);
+  rtc::scoped_ptr<RtcEventLog> event_log_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager);
 };
 }  // namespace voe
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.cc b/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.cc
new file mode 100644
index 0000000000..f54c81ec47
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.cc
@@ -0,0 +1,153 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/voice_engine/channel_proxy.h"
+
+#include <utility>
+
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/voice_engine/channel.h"
+
+namespace webrtc {
+namespace voe {
+ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {}
+
+ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
+    channel_owner_(channel_owner) {
+  RTC_CHECK(channel_owner_.channel());
+}
+
+ChannelProxy::~ChannelProxy() {}
+
+void ChannelProxy::SetRTCPStatus(bool enable) {
+  channel()->SetRTCPStatus(enable);
+}
+
+void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetLocalSSRC(ssrc);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
+  std::string c_name_limited = c_name.substr(0, 255);
+  int error = channel()->SetRTCP_CNAME(c_name_limited.c_str());
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetSendAbsoluteSenderTimeStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetSendAbsoluteSenderTimeStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->EnableSendTransportSequenceNumber(id);
+}
+
+void ChannelProxy::SetReceiveAbsoluteSenderTimeStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetReceiveAbsoluteSenderTimeStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::SetCongestionControlObjects(
+    RtpPacketSender* rtp_packet_sender,
+    TransportFeedbackObserver* transport_feedback_observer,
+    PacketRouter* packet_router) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetCongestionControlObjects(
+      rtp_packet_sender, transport_feedback_observer, packet_router);
+}
+
+CallStatistics ChannelProxy::GetRTCPStatistics() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  CallStatistics stats = {0};
+  int error = channel()->GetRTPStatistics(stats);
+  RTC_DCHECK_EQ(0, error);
+  return stats;
+}
+
+std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  std::vector<webrtc::ReportBlock> blocks;
+  int error = channel()->GetRemoteRTCPReportBlocks(&blocks);
+  RTC_DCHECK_EQ(0, error);
+  return blocks;
+}
+
+NetworkStatistics ChannelProxy::GetNetworkStatistics() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  NetworkStatistics stats = {0};
+  int error = channel()->GetNetworkStatistics(stats);
+  RTC_DCHECK_EQ(0, error);
+  return stats;
+}
+
+AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  AudioDecodingCallStats stats;
+  channel()->GetDecodingCallStatistics(&stats);
+  return stats;
+}
+
+int32_t ChannelProxy::GetSpeechOutputLevelFullRange() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  uint32_t level = 0;
+  int error = channel()->GetSpeechOutputLevelFullRange(level);
+  RTC_DCHECK_EQ(0, error);
+  return static_cast<int32_t>(level);
+}
+
+uint32_t ChannelProxy::GetDelayEstimate() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return channel()->GetDelayEstimate();
+}
+
+bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
+}
+
+bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
+                                             uint32_t duration_ms) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return
+      channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
+}
+
+void ChannelProxy::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetSink(std::move(sink));
+}
+
+Channel* ChannelProxy::channel() const {
+  RTC_DCHECK(channel_owner_.channel());
+  return channel_owner_.channel();
+}
+
+}  // namespace voe
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.h b/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.h
new file mode 100644
index 0000000000..b990d91734
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
+#define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
+
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/voice_engine/channel_manager.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+
+#include <string>
+#include <vector>
+
+namespace webrtc {
+
+class AudioSinkInterface;
+class PacketRouter;
+class RtpPacketSender;
+class TransportFeedbackObserver;
+
+namespace voe {
+
+class Channel;
+
+// This class provides the "view" of a voe::Channel that we need to implement
+// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
+// purposes:
+//  1. Allow mocking just the interfaces used, instead of the entire
+//     voe::Channel class.
+//  2. Provide a refined interface for the stream classes, including assumptions
+//     on return values and input adaptation.
+class ChannelProxy {
+ public:
+  ChannelProxy();
+  explicit ChannelProxy(const ChannelOwner& channel_owner);
+  virtual ~ChannelProxy();
+
+  virtual void SetRTCPStatus(bool enable);
+  virtual void SetLocalSSRC(uint32_t ssrc);
+  virtual void SetRTCP_CNAME(const std::string& c_name);
+  virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
+  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
+  virtual void EnableSendTransportSequenceNumber(int id);
+  virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
+  virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+  virtual void SetCongestionControlObjects(
+      RtpPacketSender* rtp_packet_sender,
+      TransportFeedbackObserver* transport_feedback_observer,
+      PacketRouter* packet_router);
+
+  virtual CallStatistics GetRTCPStatistics() const;
+  virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
+  virtual NetworkStatistics GetNetworkStatistics() const;
+  virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
+  virtual int32_t GetSpeechOutputLevelFullRange() const;
+  virtual uint32_t GetDelayEstimate() const;
+
+  virtual bool SetSendTelephoneEventPayloadType(int payload_type);
+  virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
+
+  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+
+ private:
+  Channel* channel() const;
+
+  rtc::ThreadChecker thread_checker_;
+  ChannelOwner channel_owner_;
+};
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband.cc b/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband.cc
index bcae88d8bb..9e569c22cf 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband.cc
@@ -12,8 +12,8 @@
 
 #include <assert.h>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.cc b/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.cc
index 86e8d62b72..8619a73ed8 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/dtmf_inband_queue.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.h b/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.h
index 5ca3f9c5f4..e3b9ce3adc 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/dtmf_inband_queue.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
 #define WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h b/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h
deleted file mode 100644
index 40f88b0076..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
-#define WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
-
-#include <map>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/fake_common.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-
-namespace webrtc {
-
-class FakeVoEExternalMedia : public VoEExternalMedia {
- public:
-  FakeVoEExternalMedia() {}
-  virtual ~FakeVoEExternalMedia() {}
-
-  WEBRTC_STUB(Release, ());
-  WEBRTC_FUNC(RegisterExternalMediaProcessing,
-      (int channel, ProcessingTypes type, VoEMediaProcess& processObject)) {
-    callback_map_[type] = &processObject;
-    return 0;
-  }
-  WEBRTC_FUNC(DeRegisterExternalMediaProcessing,
-      (int channel, ProcessingTypes type)) {
-    callback_map_.erase(type);
-    return 0;
-  }
-  WEBRTC_STUB(SetExternalRecordingStatus, (bool enable));
-  WEBRTC_STUB(SetExternalPlayoutStatus, (bool enable));
-  WEBRTC_STUB(ExternalRecordingInsertData,
-      (const int16_t speechData10ms[], int lengthSamples,
-       int samplingFreqHz, int current_delay_ms));
-  WEBRTC_STUB(ExternalPlayoutGetData,
-      (int16_t speechData10ms[], int samplingFreqHz,
-       int current_delay_ms, int& lengthSamples));
-  WEBRTC_STUB(ExternalPlayoutData,
-      (int16_t speechData10ms[], int samplingFreqHz,
-       int num_channels, int current_delay_ms, int& lengthSamples));
-  WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
-                              AudioFrame* frame));
-  WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
-
-  // Use this to trigger the Process() callback to a registered media processor.
-  // If |audio| is NULL, a zero array of the correct length will be forwarded.
-  void CallProcess(ProcessingTypes type, int16_t* audio,
-                   int samples_per_channel, int sample_rate_hz,
-                   int num_channels) {
-    const int length = samples_per_channel * num_channels;
-    rtc::scoped_ptr<int16_t[]> data;
-    if (!audio) {
-      data.reset(new int16_t[length]);
-      memset(data.get(), 0, length * sizeof(data[0]));
-      audio = data.get();
-    }
-
-    std::map<ProcessingTypes, VoEMediaProcess*>::const_iterator it =
-        callback_map_.find(type);
-    if (it != callback_map_.end()) {
-      it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
-                          num_channels == 2 ? true : false);
-    }
-  }
-
- private:
-  std::map<ProcessingTypes, VoEMediaProcess*> callback_map_;
-};
-
-}  // namespace webrtc
-
-#endif  //  WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/mock/mock_voe_volume_control.h b/media/webrtc/trunk/webrtc/voice_engine/include/mock/mock_voe_volume_control.h
deleted file mode 100644
index 20b096968a..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/include/mock/mock_voe_volume_control.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
-#define WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class MockVoEVolumeControl : public VoEVolumeControl {
- public:
-  MOCK_METHOD0(Release, int());
-  MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
-  MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
-  MOCK_METHOD1(SetSystemOutputMute, int(bool enable));
-  MOCK_METHOD1(GetSystemOutputMute, int(bool &enabled));
-  MOCK_METHOD1(SetMicVolume, int(unsigned int volume));
-  MOCK_METHOD1(GetMicVolume, int(unsigned int& volume));
-  MOCK_METHOD2(SetInputMute, int(int channel, bool enable));
-  MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled));
-  MOCK_METHOD1(SetSystemInputMute, int(bool enable));
-  MOCK_METHOD1(GetSystemInputMute, int(bool& enabled));
-  MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level));
-  MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level));
-  MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level));
-  MOCK_METHOD2(GetSpeechOutputLevelFullRange,
-               int(int channel, unsigned int& level));
-  MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling));
-  MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
-  MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right));
-  MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h
index 162848c282..fd70f95775 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h
@@ -44,200 +44,195 @@ namespace webrtc {
 class VoiceEngine;
 
 // VoERxVadCallback
-class WEBRTC_DLLEXPORT VoERxVadCallback
-{
-public:
-    virtual void OnRxVad(int channel, int vadDecision) = 0;
+class WEBRTC_DLLEXPORT VoERxVadCallback {
+ public:
+  virtual void OnRxVad(int channel, int vadDecision) = 0;
 
-protected:
-    virtual ~VoERxVadCallback() {}
+ protected:
+  virtual ~VoERxVadCallback() {}
 };
 
 // VoEAudioProcessing
-class WEBRTC_DLLEXPORT VoEAudioProcessing
-{
-public:
-    // Factory for the VoEAudioProcessing sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEAudioProcessing* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEAudioProcessing {
+ public:
+  // Factory for the VoEAudioProcessing sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEAudioProcessing* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEAudioProcessing sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoEAudioProcessing sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Sets Noise Suppression (NS) status and mode.
-    // The NS reduces noise in the microphone signal.
-    virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) = 0;
+  // Sets Noise Suppression (NS) status and mode.
+  // The NS reduces noise in the microphone signal.
+  virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) = 0;
 
-    // Gets the NS status and mode.
-    virtual int GetNsStatus(bool& enabled, NsModes& mode) = 0;
+  // Gets the NS status and mode.
+  virtual int GetNsStatus(bool& enabled, NsModes& mode) = 0;
 
-    // Sets the Automatic Gain Control (AGC) status and mode.
-    // The AGC adjusts the microphone signal to an appropriate level.
-    virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) = 0;
+  // Sets the Automatic Gain Control (AGC) status and mode.
+  // The AGC adjusts the microphone signal to an appropriate level.
+  virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) = 0;
 
-    // Gets the AGC status and mode.
-    virtual int GetAgcStatus(bool& enabled, AgcModes& mode) = 0;
+  // Gets the AGC status and mode.
+  virtual int GetAgcStatus(bool& enabled, AgcModes& mode) = 0;
 
-    // Sets the AGC configuration.
-    // Should only be used in situations where the working environment
-    // is well known.
-    virtual int SetAgcConfig(AgcConfig config) = 0;
+  // Sets the AGC configuration.
+  // Should only be used in situations where the working environment
+  // is well known.
+  virtual int SetAgcConfig(AgcConfig config) = 0;
 
-    // Gets the AGC configuration.
-    virtual int GetAgcConfig(AgcConfig& config) = 0;
+  // Gets the AGC configuration.
+  virtual int GetAgcConfig(AgcConfig& config) = 0;
 
-    // Sets the Echo Control (EC) status and mode.
-    // The EC mitigates acoustic echo where a user can hear their own
-    // speech repeated back due to an acoustic coupling between the
-    // speaker and the microphone at the remote end.
-    virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) = 0;
+  // Sets the Echo Control (EC) status and mode.
+  // The EC mitigates acoustic echo where a user can hear their own
+  // speech repeated back due to an acoustic coupling between the
+  // speaker and the microphone at the remote end.
+  virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) = 0;
 
-    // Gets the EC status and mode.
-    virtual int GetEcStatus(bool& enabled, EcModes& mode) = 0;
+  // Gets the EC status and mode.
+  virtual int GetEcStatus(bool& enabled, EcModes& mode) = 0;
 
-    // Enables the compensation of clock drift between the capture and render
-    // streams by the echo canceller (i.e. only using EcMode==kEcAec). It will
-    // only be enabled if supported on the current platform; otherwise an error
-    // will be returned. Check if the platform is supported by calling
-    // |DriftCompensationSupported()|.
-    virtual int EnableDriftCompensation(bool enable) = 0;
-    virtual bool DriftCompensationEnabled() = 0;
-    static bool DriftCompensationSupported();
+  // Enables the compensation of clock drift between the capture and render
+  // streams by the echo canceller (i.e. only using EcMode==kEcAec). It will
+  // only be enabled if supported on the current platform; otherwise an error
+  // will be returned. Check if the platform is supported by calling
+  // |DriftCompensationSupported()|.
+  virtual int EnableDriftCompensation(bool enable) = 0;
+  virtual bool DriftCompensationEnabled() = 0;
+  static bool DriftCompensationSupported();
 
-    // Sets a delay |offset| in ms to add to the system delay reported by the
-    // OS, which is used by the AEC to synchronize far- and near-end streams.
-    // In some cases a system may introduce a delay which goes unreported by the
-    // OS, but which is known to the user. This method can be used to compensate
-    // for the unreported delay.
-    virtual void SetDelayOffsetMs(int offset) = 0;
-    virtual int DelayOffsetMs() = 0;
+  // Sets a delay |offset| in ms to add to the system delay reported by the
+  // OS, which is used by the AEC to synchronize far- and near-end streams.
+  // In some cases a system may introduce a delay which goes unreported by the
+  // OS, but which is known to the user. This method can be used to compensate
+  // for the unreported delay.
+  virtual void SetDelayOffsetMs(int offset) = 0;
+  virtual int DelayOffsetMs() = 0;
 
-    // Modifies settings for the AEC designed for mobile devices (AECM).
-    virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
-                            bool enableCNG = true) = 0;
+  // Modifies settings for the AEC designed for mobile devices (AECM).
+  virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
+                          bool enableCNG = true) = 0;
 
-    // Gets settings for the AECM.
-    virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG) = 0;
+  // Gets settings for the AECM.
+  virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG) = 0;
 
-    // Enables a high pass filter on the capture signal. This removes DC bias
-    // and low-frequency noise. Recommended to be enabled.
-    virtual int EnableHighPassFilter(bool enable) = 0;
-    virtual bool IsHighPassFilterEnabled() = 0;
+  // Enables a high pass filter on the capture signal. This removes DC bias
+  // and low-frequency noise. Recommended to be enabled.
+  virtual int EnableHighPassFilter(bool enable) = 0;
+  virtual bool IsHighPassFilterEnabled() = 0;
 
-    // Sets status and mode of the receiving-side (Rx) NS.
-    // The Rx NS reduces noise in the received signal for the specified
-    // |channel|. Intended for advanced usage only.
-    virtual int SetRxNsStatus(int channel,
-                              bool enable,
-                              NsModes mode = kNsUnchanged) = 0;
+  // Sets status and mode of the receiving-side (Rx) NS.
+  // The Rx NS reduces noise in the received signal for the specified
+  // |channel|. Intended for advanced usage only.
+  virtual int SetRxNsStatus(int channel,
+                            bool enable,
+                            NsModes mode = kNsUnchanged) = 0;
 
-    // Gets status and mode of the receiving-side NS.
-    virtual int GetRxNsStatus(int channel,
-                              bool& enabled,
-                              NsModes& mode) = 0;
+  // Gets status and mode of the receiving-side NS.
+  virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode) = 0;
 
-    // Sets status and mode of the receiving-side (Rx) AGC.
-    // The Rx AGC adjusts the received signal to an appropriate level
-    // for the specified |channel|. Intended for advanced usage only.
-    virtual int SetRxAgcStatus(int channel,
-                               bool enable,
-                               AgcModes mode = kAgcUnchanged) = 0;
+  // Sets status and mode of the receiving-side (Rx) AGC.
+  // The Rx AGC adjusts the received signal to an appropriate level
+  // for the specified |channel|. Intended for advanced usage only.
+  virtual int SetRxAgcStatus(int channel,
+                             bool enable,
+                             AgcModes mode = kAgcUnchanged) = 0;
 
-    // Gets status and mode of the receiving-side AGC.
-    virtual int GetRxAgcStatus(int channel,
-                               bool& enabled,
-                               AgcModes& mode) = 0;
+  // Gets status and mode of the receiving-side AGC.
+  virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode) = 0;
 
-    // Modifies the AGC configuration on the receiving side for the
-    // specified |channel|.
-    virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
+  // Modifies the AGC configuration on the receiving side for the
+  // specified |channel|.
+  virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
 
-    // Gets the AGC configuration on the receiving side.
-    virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
+  // Gets the AGC configuration on the receiving side.
+  virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
 
-    // Registers a VoERxVadCallback |observer| instance and enables Rx VAD
-    // notifications for the specified |channel|.
-    virtual int RegisterRxVadObserver(int channel,
-                                      VoERxVadCallback &observer) = 0;
+  // Registers a VoERxVadCallback |observer| instance and enables Rx VAD
+  // notifications for the specified |channel|.
+  virtual int RegisterRxVadObserver(int channel,
+                                    VoERxVadCallback& observer) = 0;
 
-    // Deregisters the VoERxVadCallback |observer| and disables Rx VAD
-    // notifications for the specified |channel|.
-    virtual int DeRegisterRxVadObserver(int channel) = 0;
+  // Deregisters the VoERxVadCallback |observer| and disables Rx VAD
+  // notifications for the specified |channel|.
+  virtual int DeRegisterRxVadObserver(int channel) = 0;
 
-    // Gets the VAD/DTX activity for the specified |channel|.
-    // The returned value is 1 if frames of audio contains speech
-    // and 0 if silence. The output is always 1 if VAD is disabled.
-    virtual int VoiceActivityIndicator(int channel) = 0;
+  // Gets the VAD/DTX activity for the specified |channel|.
+  // The returned value is 1 if frames of audio contains speech
+  // and 0 if silence. The output is always 1 if VAD is disabled.
+  virtual int VoiceActivityIndicator(int channel) = 0;
 
-    // Enables or disables the possibility to retrieve echo metrics and delay
-    // logging values during an active call. The metrics are only supported in
-    // AEC.
-    virtual int SetEcMetricsStatus(bool enable) = 0;
+  // Enables or disables the possibility to retrieve echo metrics and delay
+  // logging values during an active call. The metrics are only supported in
+  // AEC.
+  virtual int SetEcMetricsStatus(bool enable) = 0;
 
-    // Gets the current EC metric status.
-    virtual int GetEcMetricsStatus(bool& enabled) = 0;
+  // Gets the current EC metric status.
+  virtual int GetEcMetricsStatus(bool& enabled) = 0;
 
-    // Gets the instantaneous echo level metrics.
-    virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;
+  // Gets the instantaneous echo level metrics.
+  virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;
 
-    // Gets the EC internal |delay_median| and |delay_std| in ms between
-    // near-end and far-end. The metric |fraction_poor_delays| is the amount of
-    // delay values that potentially can break the EC. The values are aggregated
-    // over one second and the last updated metrics are returned.
-    virtual int GetEcDelayMetrics(int& delay_median, int& delay_std,
-                                  float& fraction_poor_delays) = 0;
+  // Gets the EC internal |delay_median| and |delay_std| in ms between
+  // near-end and far-end. The metric |fraction_poor_delays| is the amount of
+  // delay values that potentially can break the EC. The values are aggregated
+  // over one second and the last updated metrics are returned.
+  virtual int GetEcDelayMetrics(int& delay_median,
+                                int& delay_std,
+                                float& fraction_poor_delays) = 0;
 
-    // Enables recording of Audio Processing (AP) debugging information.
-    // The file can later be used for off-line analysis of the AP performance.
-    virtual int StartDebugRecording(const char* fileNameUTF8) = 0;
+  // Enables recording of Audio Processing (AP) debugging information.
+  // The file can later be used for off-line analysis of the AP performance.
+  virtual int StartDebugRecording(const char* fileNameUTF8) = 0;
 
-    // Same as above but sets and uses an existing file handle. Takes ownership
-    // of |file_handle| and passes it on to the audio processing module.
-    virtual int StartDebugRecording(FILE* file_handle) = 0;
+  // Same as above but sets and uses an existing file handle. Takes ownership
+  // of |file_handle| and passes it on to the audio processing module.
+  virtual int StartDebugRecording(FILE* file_handle) = 0;
 
-    // Disables recording of AP debugging information.
-    virtual int StopDebugRecording() = 0;
+  // Disables recording of AP debugging information.
+  virtual int StopDebugRecording() = 0;
 
-    // Enables or disables detection of disturbing keyboard typing.
-    // An error notification will be given as a callback upon detection.
-    virtual int SetTypingDetectionStatus(bool enable) = 0;
+  // Enables or disables detection of disturbing keyboard typing.
+  // An error notification will be given as a callback upon detection.
+  virtual int SetTypingDetectionStatus(bool enable) = 0;
 
-    // Gets the current typing detection status.
-    virtual int GetTypingDetectionStatus(bool& enabled) = 0;
+  // Gets the current typing detection status.
+  virtual int GetTypingDetectionStatus(bool& enabled) = 0;
 
-    // Reports the lower of:
-    // * Time in seconds since the last typing event.
-    // * Time in seconds since the typing detection was enabled.
-    // Returns error if typing detection is disabled.
-    virtual int TimeSinceLastTyping(int &seconds) = 0;
+  // Reports the lower of:
+  // * Time in seconds since the last typing event.
+  // * Time in seconds since the typing detection was enabled.
+  // Returns error if typing detection is disabled.
+  virtual int TimeSinceLastTyping(int& seconds) = 0;
 
-    // Optional setting of typing detection parameters
-    // Parameter with value == 0 will be ignored
-    // and left with default config.
-    // TODO(niklase) Remove default argument as soon as libJingle is updated!
-    virtual int SetTypingDetectionParameters(int timeWindow,
-                                             int costPerTyping,
-                                             int reportingThreshold,
-                                             int penaltyDecay,
-                                             int typeEventDelay = 0) = 0;
+  // Optional setting of typing detection parameters
+  // Parameter with value == 0 will be ignored
+  // and left with default config.
+  // TODO(niklase) Remove default argument as soon as libJingle is updated!
+  virtual int SetTypingDetectionParameters(int timeWindow,
+                                           int costPerTyping,
+                                           int reportingThreshold,
+                                           int penaltyDecay,
+                                           int typeEventDelay = 0) = 0;
 
-    // Swaps the capture-side left and right audio channels when enabled. It
-    // only has an effect when using a stereo send codec. The setting is
-    // persistent; it will be applied whenever a stereo send codec is enabled.
-    //
-    // The swap is applied only to the captured audio, and not mixed files. The
-    // swap will appear in file recordings and when accessing audio through the
-    // external media interface.
-    virtual void EnableStereoChannelSwapping(bool enable) = 0;
-    virtual bool IsStereoChannelSwappingEnabled() = 0;
+  // Swaps the capture-side left and right audio channels when enabled. It
+  // only has an effect when using a stereo send codec. The setting is
+  // persistent; it will be applied whenever a stereo send codec is enabled.
+  //
+  // The swap is applied only to the captured audio, and not mixed files. The
+  // swap will appear in file recordings and when accessing audio through the
+  // external media interface.
+  virtual void EnableStereoChannelSwapping(bool enable) = 0;
+  virtual bool IsStereoChannelSwappingEnabled() = 0;
 
-protected:
-    VoEAudioProcessing() {}
-    virtual ~VoEAudioProcessing() {}
+ protected:
+  VoEAudioProcessing() {}
+  virtual ~VoEAudioProcessing() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h
index b99f36aa81..3d07fa78ff 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h
@@ -46,145 +46,148 @@ class Config;
 const int kVoEDefault = -1;
 
 // VoiceEngineObserver
-class WEBRTC_DLLEXPORT VoiceEngineObserver
-{
-public:
-    // This method will be called after the occurrence of any runtime error
-    // code, or warning notification, when the observer interface has been
-    // installed using VoEBase::RegisterVoiceEngineObserver().
-    virtual void CallbackOnError(int channel, int errCode) = 0;
+class WEBRTC_DLLEXPORT VoiceEngineObserver {
+ public:
+  // This method will be called after the occurrence of any runtime error
+  // code, or warning notification, when the observer interface has been
+  // installed using VoEBase::RegisterVoiceEngineObserver().
+  virtual void CallbackOnError(int channel, int errCode) = 0;
 
-protected:
-    virtual ~VoiceEngineObserver() {}
+ protected:
+  virtual ~VoiceEngineObserver() {}
 };
 
 // VoiceEngine
-class WEBRTC_DLLEXPORT VoiceEngine
-{
-public:
-    // Creates a VoiceEngine object, which can then be used to acquire
-    // sub-APIs. Returns NULL on failure.
-    static VoiceEngine* Create();
-    static VoiceEngine* Create(const Config& config);
+class WEBRTC_DLLEXPORT VoiceEngine {
+ public:
+  // Creates a VoiceEngine object, which can then be used to acquire
+  // sub-APIs. Returns NULL on failure.
+  static VoiceEngine* Create();
+  static VoiceEngine* Create(const Config& config);
 
-    // Deletes a created VoiceEngine object and releases the utilized resources.
-    // Note that if there are outstanding references held via other interfaces,
-    // the voice engine instance will not actually be deleted until those
-    // references have been released.
-    static bool Delete(VoiceEngine*& voiceEngine);
+  // Deletes a created VoiceEngine object and releases the utilized resources.
+  // Note that if there are outstanding references held via other interfaces,
+  // the voice engine instance will not actually be deleted until those
+  // references have been released.
+  static bool Delete(VoiceEngine*& voiceEngine);
 
-    // Specifies the amount and type of trace information which will be
-    // created by the VoiceEngine.
-    static int SetTraceFilter(unsigned int filter);
+  // Specifies the amount and type of trace information which will be
+  // created by the VoiceEngine.
+  static int SetTraceFilter(unsigned int filter);
 
-    // Sets the name of the trace file and enables non-encrypted trace messages.
-    static int SetTraceFile(const char* fileNameUTF8,
-                            bool addFileCounter = false);
+  // Sets the name of the trace file and enables non-encrypted trace messages.
+  static int SetTraceFile(const char* fileNameUTF8,
+                          bool addFileCounter = false);
 
-    // Installs the TraceCallback implementation to ensure that the user
-    // receives callbacks for generated trace messages.
-    static int SetTraceCallback(TraceCallback* callback);
+  // Installs the TraceCallback implementation to ensure that the user
+  // receives callbacks for generated trace messages.
+  static int SetTraceCallback(TraceCallback* callback);
 
 #if !defined(WEBRTC_CHROMIUM_BUILD)
-    static int SetAndroidObjects(void* javaVM, void* context);
+  static int SetAndroidObjects(void* javaVM, void* context);
 #endif
 
-protected:
-    VoiceEngine() {}
-    ~VoiceEngine() {}
+  static std::string GetVersionString();
+
+ protected:
+  VoiceEngine() {}
+  ~VoiceEngine() {}
 };
 
 // VoEBase
-class WEBRTC_DLLEXPORT VoEBase
-{
-public:
-    // Factory for the VoEBase sub-API. Increases an internal reference
-    // counter if successful. Returns NULL if the API is not supported or if
-    // construction fails.
-    static VoEBase* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEBase {
+ public:
+  // Factory for the VoEBase sub-API. Increases an internal reference
+  // counter if successful. Returns NULL if the API is not supported or if
+  // construction fails.
+  static VoEBase* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEBase sub-API and decreases an internal reference
-    // counter. Returns the new reference count. This value should be zero
-    // for all sub-APIs before the VoiceEngine object can be safely deleted.
-    virtual int Release() = 0;
+  // Releases the VoEBase sub-API and decreases an internal reference
+  // counter. Returns the new reference count. This value should be zero
+  // for all sub-APIs before the VoiceEngine object can be safely deleted.
+  virtual int Release() = 0;
 
-    // Installs the observer class to enable runtime error control and
-    // warning notifications.
-    virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) = 0;
+  // Installs the observer class to enable runtime error control and
+  // warning notifications. Returns -1 in case of an error, 0 otherwise.
+  virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) = 0;
 
-    // Removes and disables the observer class for runtime error control
-    // and warning notifications.
-    virtual int DeRegisterVoiceEngineObserver() = 0;
+  // Removes and disables the observer class for runtime error control
+  // and warning notifications. Returns 0.
+  virtual int DeRegisterVoiceEngineObserver() = 0;
 
-    // Initializes all common parts of the VoiceEngine; e.g. all
-    // encoders/decoders, the sound card and core receiving components.
-    // This method also makes it possible to install some user-defined external
-    // modules:
-    // - The Audio Device Module (ADM) which implements all the audio layer
-    // functionality in a separate (reference counted) module.
-    // - The AudioProcessing module handles capture-side processing. VoiceEngine
-    // takes ownership of this object.
-    // If NULL is passed for any of these, VoiceEngine will create its own.
-    // TODO(ajm): Remove default NULLs.
-    virtual int Init(AudioDeviceModule* external_adm = NULL,
-                     AudioProcessing* audioproc = NULL) = 0;
+  // Initializes all common parts of the VoiceEngine; e.g. all
+  // encoders/decoders, the sound card and core receiving components.
+  // This method also makes it possible to install some user-defined external
+  // modules:
+  // - The Audio Device Module (ADM) which implements all the audio layer
+  // functionality in a separate (reference counted) module.
+  // - The AudioProcessing module handles capture-side processing. VoiceEngine
+  // takes ownership of this object.
+  // If NULL is passed for any of these, VoiceEngine will create its own.
+  // Returns -1 in case of an error, 0 otherwise.
+  // TODO(ajm): Remove default NULLs.
+  virtual int Init(AudioDeviceModule* external_adm = NULL,
+                   AudioProcessing* audioproc = NULL) = 0;
 
-    // Returns NULL before Init() is called.
-    virtual AudioProcessing* audio_processing() = 0;
+  // Returns NULL before Init() is called.
+  virtual AudioProcessing* audio_processing() = 0;
 
-    // Terminates all VoiceEngine functions and releses allocated resources.
-    virtual int Terminate() = 0;
+  // Terminates all VoiceEngine functions and releases allocated resources.
+  // Returns 0.
+  virtual int Terminate() = 0;
 
-    // Creates a new channel and allocates the required resources for it.
-    // One can use |config| to configure the channel. Currently that is used for
-    // choosing between ACM1 and ACM2, when creating Audio Coding Module.
-    virtual int CreateChannel() = 0;
-    virtual int CreateChannel(const Config& config) = 0;
+  // Creates a new channel and allocates the required resources for it.
+  // One can use |config| to configure the channel. Currently that is used for
+  // choosing between ACM1 and ACM2, when creating Audio Coding Module.
+  // Returns channel ID or -1 in case of an error.
+  virtual int CreateChannel() = 0;
+  virtual int CreateChannel(const Config& config) = 0;
 
-    // Deletes an existing channel and releases the utilized resources.
-    virtual int DeleteChannel(int channel) = 0;
+  // Deletes an existing channel and releases the utilized resources.
+  // Returns -1 in case of an error, 0 otherwise.
+  virtual int DeleteChannel(int channel) = 0;
 
-    // Prepares and initiates the VoiceEngine for reception of
-    // incoming RTP/RTCP packets on the specified |channel|.
-    virtual int StartReceive(int channel) = 0;
+  // Prepares and initiates the VoiceEngine for reception of
+  // incoming RTP/RTCP packets on the specified |channel|.
+  virtual int StartReceive(int channel) = 0;
 
-    // Stops receiving incoming RTP/RTCP packets on the specified |channel|.
-    virtual int StopReceive(int channel) = 0;
+  // Stops receiving incoming RTP/RTCP packets on the specified |channel|.
+  virtual int StopReceive(int channel) = 0;
 
-    // Starts forwarding the packets to the mixer/soundcard for a
-    // specified |channel|.
-    virtual int StartPlayout(int channel) = 0;
+  // Starts forwarding the packets to the mixer/soundcard for a
+  // specified |channel|.
+  virtual int StartPlayout(int channel) = 0;
 
-    // Stops forwarding the packets to the mixer/soundcard for a
-    // specified |channel|.
-    virtual int StopPlayout(int channel) = 0;
+  // Stops forwarding the packets to the mixer/soundcard for a
+  // specified |channel|.
+  virtual int StopPlayout(int channel) = 0;
 
-    // Starts sending packets to an already specified IP address and
-    // port number for a specified |channel|.
-    virtual int StartSend(int channel) = 0;
+  // Starts sending packets to an already specified IP address and
+  // port number for a specified |channel|.
+  virtual int StartSend(int channel) = 0;
 
-    // Stops sending packets from a specified |channel|.
-    virtual int StopSend(int channel) = 0;
+  // Stops sending packets from a specified |channel|.
+  virtual int StopSend(int channel) = 0;
 
-    // Gets the version information for VoiceEngine and its components.
-    virtual int GetVersion(char version[1024]) = 0;
+  // Gets the version information for VoiceEngine and its components.
+  virtual int GetVersion(char version[1024]) = 0;
 
-    // Gets the last VoiceEngine error code.
-    virtual int LastError() = 0;
+  // Gets the last VoiceEngine error code.
+  virtual int LastError() = 0;
 
-    // TODO(xians): Make the interface pure virtual after libjingle
-    // implements the interface in its FakeWebRtcVoiceEngine.
-    virtual AudioTransport* audio_transport() { return NULL; }
+  // TODO(xians): Make the interface pure virtual after libjingle
+  // implements the interface in its FakeWebRtcVoiceEngine.
+  virtual AudioTransport* audio_transport() { return NULL; }
 
-    // To be removed. Don't use.
-    virtual int SetOnHoldStatus(int channel, bool enable,
-        OnHoldModes mode = kHoldSendAndPlay) { return -1; }
-    virtual int GetOnHoldStatus(int channel, bool& enabled,
-        OnHoldModes& mode) { return -1; }
+  // Associate a send channel to a receive channel.
+  // Used for obtaining RTT for a receive-only channel.
+  // One should be careful not to crate a circular association, e.g.,
+  // 1 <- 2 <- 1.
+  virtual int AssociateSendChannel(int channel, int accociate_send_channel) = 0;
 
-protected:
-    VoEBase() {}
-    virtual ~VoEBase() {}
+ protected:
+  VoEBase() {}
+  virtual ~VoEBase() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h
index 4b9f939ce1..6c4fb38a2f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h
@@ -35,105 +35,110 @@
 
 namespace webrtc {
 
+class RtcEventLog;
 class VoiceEngine;
 
-class WEBRTC_DLLEXPORT VoECodec
-{
-public:
-    // Factory for the VoECodec sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoECodec* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoECodec {
+ public:
+  // Factory for the VoECodec sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoECodec* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoECodec sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoECodec sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Gets the number of supported codecs.
-    virtual int NumOfCodecs() = 0;
+  // Gets the number of supported codecs.
+  virtual int NumOfCodecs() = 0;
 
-    // Get the |codec| information for a specified list |index|.
-    virtual int GetCodec(int index, CodecInst& codec) = 0;
+  // Get the |codec| information for a specified list |index|.
+  virtual int GetCodec(int index, CodecInst& codec) = 0;
 
-    // Sets the |codec| for the |channel| to be used for sending.
-    virtual int SetSendCodec(int channel, const CodecInst& codec) = 0;
+  // Sets the |codec| for the |channel| to be used for sending.
+  virtual int SetSendCodec(int channel, const CodecInst& codec) = 0;
 
-    // Gets the |codec| parameters for the sending codec on a specified
-    // |channel|.
-    virtual int GetSendCodec(int channel, CodecInst& codec) = 0;
+  // Gets the |codec| parameters for the sending codec on a specified
+  // |channel|.
+  virtual int GetSendCodec(int channel, CodecInst& codec) = 0;
 
-    // Gets the currently received |codec| for a specific |channel|.
-    virtual int GetRecCodec(int channel, CodecInst& codec) = 0;
+  // Sets the bitrate on a specified |channel| to the specified value
+  // (in bits/sec). If the value is not supported by the codec, the codec will
+  // choose an appropriate value.
+  // Returns -1 on failure and 0 on success.
+  virtual int SetBitRate(int channel, int bitrate_bps) = 0;
 
-    // Sets the dynamic payload type number for a particular |codec| or
-    // disables (ignores) a codec for receiving. For instance, when receiving
-    // an invite from a SIP-based client, this function can be used to change
-    // the dynamic payload type number to match that in the INVITE SDP-
-    // message. The utilized parameters in the |codec| structure are:
-    // plname, plfreq, pltype and channels.
-    virtual int SetRecPayloadType(int channel, const CodecInst& codec) = 0;
+  // Gets the currently received |codec| for a specific |channel|.
+  virtual int GetRecCodec(int channel, CodecInst& codec) = 0;
 
-    // Gets the actual payload type that is set for receiving a |codec| on a
-    // |channel|. The value it retrieves will either be the default payload
-    // type, or a value earlier set with SetRecPayloadType().
-    virtual int GetRecPayloadType(int channel, CodecInst& codec) = 0;
+  // Sets the dynamic payload type number for a particular |codec| or
+  // disables (ignores) a codec for receiving. For instance, when receiving
+  // an invite from a SIP-based client, this function can be used to change
+  // the dynamic payload type number to match that in the INVITE SDP-
+  // message. The utilized parameters in the |codec| structure are:
+  // plname, plfreq, pltype and channels.
+  virtual int SetRecPayloadType(int channel, const CodecInst& codec) = 0;
 
-    // Sets the payload |type| for the sending of SID-frames with background
-    // noise estimation during silence periods detected by the VAD.
-    virtual int SetSendCNPayloadType(
-        int channel, int type, PayloadFrequencies frequency = kFreq16000Hz) = 0;
+  // Gets the actual payload type that is set for receiving a |codec| on a
+  // |channel|. The value it retrieves will either be the default payload
+  // type, or a value earlier set with SetRecPayloadType().
+  virtual int GetRecPayloadType(int channel, CodecInst& codec) = 0;
 
-    // Sets the codec internal FEC (forward error correction) status for a
-    // specified |channel|. Returns 0 if success, and -1 if failed.
-    // TODO(minyue): Make SetFECStatus() pure virtual when fakewebrtcvoiceengine
-    // in talk is ready.
-    virtual int SetFECStatus(int channel, bool enable) { return -1; }
+  // Sets the payload |type| for the sending of SID-frames with background
+  // noise estimation during silence periods detected by the VAD.
+  virtual int SetSendCNPayloadType(
+      int channel,
+      int type,
+      PayloadFrequencies frequency = kFreq16000Hz) = 0;
 
-    // Gets the codec internal FEC status for a specified |channel|. Returns 0
-    // with the status stored in |enabled| if success, and -1 if encountered
-    // error.
-    // TODO(minyue): Make GetFECStatus() pure virtual when fakewebrtcvoiceengine
-    // in talk is ready.
-    virtual int GetFECStatus(int channel, bool& enabled) { return -1; }
+  // Sets the codec internal FEC (forward error correction) status for a
+  // specified |channel|. Returns 0 if success, and -1 if failed.
+  // TODO(minyue): Make SetFECStatus() pure virtual when fakewebrtcvoiceengine
+  // in talk is ready.
+  virtual int SetFECStatus(int channel, bool enable) { return -1; }
 
-    // Sets the VAD/DTX (silence suppression) status and |mode| for a
-    // specified |channel|. Disabling VAD (through |enable|) will also disable
-    // DTX; it is not necessary to explictly set |disableDTX| in this case.
-    virtual int SetVADStatus(int channel, bool enable,
-                             VadModes mode = kVadConventional,
-                             bool disableDTX = false) = 0;
+  // Gets the codec internal FEC status for a specified |channel|. Returns 0
+  // with the status stored in |enabled| if success, and -1 if encountered
+  // error.
+  // TODO(minyue): Make GetFECStatus() pure virtual when fakewebrtcvoiceengine
+  // in talk is ready.
+  virtual int GetFECStatus(int channel, bool& enabled) { return -1; }
 
-    // Gets the VAD/DTX status and |mode| for a specified |channel|.
-    virtual int GetVADStatus(int channel, bool& enabled, VadModes& mode,
-                             bool& disabledDTX) = 0;
+  // Sets the VAD/DTX (silence suppression) status and |mode| for a
+  // specified |channel|. Disabling VAD (through |enable|) will also disable
+  // DTX; it is not necessary to explictly set |disableDTX| in this case.
+  virtual int SetVADStatus(int channel,
+                           bool enable,
+                           VadModes mode = kVadConventional,
+                           bool disableDTX = false) = 0;
 
-    // If send codec is Opus on a specified |channel|, sets the maximum playback
-    // rate the receiver will render: |frequency_hz| (in Hz).
-    // TODO(minyue): Make SetOpusMaxPlaybackRate() pure virtual when
-    // fakewebrtcvoiceengine in talk is ready.
-    virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz) {
-      return -1;
-    }
+  // Gets the VAD/DTX status and |mode| for a specified |channel|.
+  virtual int GetVADStatus(int channel,
+                           bool& enabled,
+                           VadModes& mode,
+                           bool& disabledDTX) = 0;
 
-    // If send codec is Opus on a specified |channel|, set its DTX. Returns 0 if
-    // success, and -1 if failed.
-    virtual int SetOpusDtx(int channel, bool enable_dtx) = 0;
+  // If send codec is Opus on a specified |channel|, sets the maximum playback
+  // rate the receiver will render: |frequency_hz| (in Hz).
+  // TODO(minyue): Make SetOpusMaxPlaybackRate() pure virtual when
+  // fakewebrtcvoiceengine in talk is ready.
+  virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz) {
+    return -1;
+  }
 
-    // Don't use. To be removed.
-    virtual int SetAMREncFormat(int channel, AmrMode mode) { return -1; }
-    virtual int SetAMRDecFormat(int channel, AmrMode mode) { return -1; }
-    virtual int SetAMRWbEncFormat(int channel, AmrMode mode) { return -1; }
-    virtual int SetAMRWbDecFormat(int channel, AmrMode mode) { return -1; }
-    virtual int SetISACInitTargetRate(int channel, int rateBps,
-            bool useFixedFrameSize = false) { return -1; }
-    virtual int SetISACMaxRate(int channel, int rateBps) { return -1; }
-    virtual int SetISACMaxPayloadSize(int channel, int sizeBytes) { return -1; }
+  // If send codec is Opus on a specified |channel|, set its DTX. Returns 0 if
+  // success, and -1 if failed.
+  virtual int SetOpusDtx(int channel, bool enable_dtx) = 0;
 
-protected:
-    VoECodec() {}
-    virtual ~VoECodec() {}
+  // Get a pointer to the event logging object associated with this Voice
+  // Engine. This pointer will remain valid until VoiceEngine is destroyed.
+  virtual RtcEventLog* GetEventLog() = 0;
+
+ protected:
+  VoECodec() {}
+  virtual ~VoECodec() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_dtmf.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_dtmf.h
index 4fd44961c4..c8ef5d0d77 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_dtmf.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_dtmf.h
@@ -40,57 +40,51 @@ namespace webrtc {
 class VoiceEngine;
 
 // VoEDtmf
-class WEBRTC_DLLEXPORT VoEDtmf
-{
-public:
+class WEBRTC_DLLEXPORT VoEDtmf {
+ public:
+  // Factory for the VoEDtmf sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEDtmf* GetInterface(VoiceEngine* voiceEngine);
 
-    // Factory for the VoEDtmf sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEDtmf* GetInterface(VoiceEngine* voiceEngine);
+  // Releases the VoEDtmf sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Releases the VoEDtmf sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Sends telephone events either in-band or out-of-band.
+  virtual int SendTelephoneEvent(int channel,
+                                 int eventCode,
+                                 bool outOfBand = true,
+                                 int lengthMs = 160,
+                                 int attenuationDb = 10) = 0;
 
-    // Sends telephone events either in-band or out-of-band.
-    virtual int SendTelephoneEvent(int channel, int eventCode,
-                                   bool outOfBand = true, int lengthMs = 160,
-                                   int attenuationDb = 10) = 0;
+  // Sets the dynamic payload |type| that should be used for telephone
+  // events.
+  virtual int SetSendTelephoneEventPayloadType(int channel,
+                                               unsigned char type) = 0;
 
+  // Gets the currently set dynamic payload |type| for telephone events.
+  virtual int GetSendTelephoneEventPayloadType(int channel,
+                                               unsigned char& type) = 0;
 
-    // Sets the dynamic payload |type| that should be used for telephone
-    // events.
-    virtual int SetSendTelephoneEventPayloadType(int channel,
-                                                 unsigned char type) = 0;
+  // Toogles DTMF feedback state: when a DTMF tone is sent, the same tone
+  // is played out on the speaker.
+  virtual int SetDtmfFeedbackStatus(bool enable,
+                                    bool directFeedback = false) = 0;
 
+  // Gets the DTMF feedback status.
+  virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) = 0;
 
-    // Gets the currently set dynamic payload |type| for telephone events.
-    virtual int GetSendTelephoneEventPayloadType(int channel,
-                                                 unsigned char& type) = 0;
+  // Plays a DTMF feedback tone (only locally).
+  virtual int PlayDtmfTone(int eventCode,
+                           int lengthMs = 200,
+                           int attenuationDb = 10) = 0;
 
-    // Toogles DTMF feedback state: when a DTMF tone is sent, the same tone
-    // is played out on the speaker.
-    virtual int SetDtmfFeedbackStatus(bool enable,
-                                      bool directFeedback = false) = 0;
-
-    // Gets the DTMF feedback status.
-    virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) = 0;
-
-    // Plays a DTMF feedback tone (only locally).
-    virtual int PlayDtmfTone(int eventCode, int lengthMs = 200,
-                             int attenuationDb = 10) = 0;
-
-    // To be removed. Don't use.
-    virtual int StartPlayingDtmfTone(int eventCode,
-        int attenuationDb = 10) { return -1; }
-    virtual int StopPlayingDtmfTone() { return -1; }
-
-protected:
-    VoEDtmf() {}
-    virtual ~VoEDtmf() {}
+ protected:
+  VoEDtmf() {}
+  virtual ~VoEDtmf() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_external_media.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_external_media.h
index 976bea9d96..ff8e308243 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_external_media.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_external_media.h
@@ -39,94 +39,98 @@ namespace webrtc {
 class VoiceEngine;
 class AudioFrame;
 
-class WEBRTC_DLLEXPORT VoEMediaProcess
-{
-public:
-    // The VoiceEngine user should override the Process() method in a
-    // derived class. Process() will be called when audio is ready to
-    // be processed. The audio can be accessed in several different modes
-    // given by the |type| parameter. The function should modify the
-    // original data and ensure that it is copied back to the |audio10ms|
-    // array. The number of samples in the frame cannot be changed.
-    // The sampling frequency will depend upon the codec used.
-    // If |isStereo| is true, audio10ms will contain 16-bit PCM data
-    // samples in interleaved stereo format (L0,R0,L1,R1,...).
-    virtual void Process(int channel, ProcessingTypes type,
-                         int16_t audio10ms[], int length,
-                         int samplingFreq, bool isStereo) = 0;
+class WEBRTC_DLLEXPORT VoEMediaProcess {
+ public:
+  // The VoiceEngine user should override the Process() method in a
+  // derived class. Process() will be called when audio is ready to
+  // be processed. The audio can be accessed in several different modes
+  // given by the |type| parameter. The function should modify the
+  // original data and ensure that it is copied back to the |audio10ms|
+  // array. The number of samples in the frame cannot be changed.
+  // The sampling frequency will depend upon the codec used.
+  // If |isStereo| is true, audio10ms will contain 16-bit PCM data
+  // samples in interleaved stereo format (L0,R0,L1,R1,...).
+  virtual void Process(int channel,
+                       ProcessingTypes type,
+                       int16_t audio10ms[],
+                       size_t length,
+                       int samplingFreq,
+                       bool isStereo) = 0;
 
-protected:
-    virtual ~VoEMediaProcess() {}
+ protected:
+  virtual ~VoEMediaProcess() {}
 };
 
-class WEBRTC_DLLEXPORT VoEExternalMedia
-{
-public:
-    // Factory for the VoEExternalMedia sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEExternalMedia* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEExternalMedia {
+ public:
+  // Factory for the VoEExternalMedia sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEExternalMedia* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEExternalMedia sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoEExternalMedia sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Installs a VoEMediaProcess derived instance and activates external
-    // media for the specified |channel| and |type|.
-    virtual int RegisterExternalMediaProcessing(
-        int channel, ProcessingTypes type, VoEMediaProcess& processObject) = 0;
+  // Installs a VoEMediaProcess derived instance and activates external
+  // media for the specified |channel| and |type|.
+  virtual int RegisterExternalMediaProcessing(
+      int channel,
+      ProcessingTypes type,
+      VoEMediaProcess& processObject) = 0;
 
-    // Removes the VoEMediaProcess derived instance and deactivates external
-    // media for the specified |channel| and |type|.
-    virtual int DeRegisterExternalMediaProcessing(
-        int channel, ProcessingTypes type) = 0;
+  // Removes the VoEMediaProcess derived instance and deactivates external
+  // media for the specified |channel| and |type|.
+  virtual int DeRegisterExternalMediaProcessing(int channel,
+                                                ProcessingTypes type) = 0;
 
-    // Toogles state of external recording.
-    virtual int SetExternalRecordingStatus(bool enable) = 0;
+  // Toogles state of external recording.
+  virtual int SetExternalRecordingStatus(bool enable) = 0;
 
-    // Toogles state of external playout.
-    virtual int SetExternalPlayoutStatus(bool enable) = 0;
+  // Toogles state of external playout.
+  virtual int SetExternalPlayoutStatus(bool enable) = 0;
 
-    // This function accepts externally recorded audio. During transmission,
-    // this method should be called at as regular an interval as possible
-    // with frames of corresponding size.
-    virtual int ExternalRecordingInsertData(
-        const int16_t speechData10ms[], int lengthSamples,
-        int samplingFreqHz, int current_delay_ms) = 0;
+  // This function accepts externally recorded audio. During transmission,
+  // this method should be called at as regular an interval as possible
+  // with frames of corresponding size.
+  virtual int ExternalRecordingInsertData(
+                                          const int16_t speechData10ms[], int lengthSamples,
+                                          int samplingFreqHz, int current_delay_ms) = 0;
 
 
-    // This function inserts audio written to the OS audio drivers for use
-    // as the far-end signal for AEC processing.  The length of the block
-    // must be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
-    // 48000 kHz sampling rates respectively).
-    virtual int ExternalPlayoutData(
-        int16_t speechData10ms[], int samplingFreqHz, int num_channels,
-        int current_delay_ms, int& lengthSamples) = 0;
+  // This function inserts audio written to the OS audio drivers for use
+  // as the far-end signal for AEC processing.  The length of the block
+  // must be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
+  // 48000 kHz sampling rates respectively).
+  virtual int ExternalPlayoutData(
+                                  int16_t speechData10ms[], int samplingFreqHz, int num_channels,
+                                  int current_delay_ms, int& lengthSamples) = 0;
 
-    // This function gets audio for an external playout sink.
-    // During transmission, this function should be called every ~10 ms
-    // to obtain a new 10 ms frame of audio. The length of the block will
-    // be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
-    // 48000 kHz sampling rates respectively).
-    virtual int ExternalPlayoutGetData(
-        int16_t speechData10ms[], int samplingFreqHz,
-        int current_delay_ms, int& lengthSamples) = 0;
+  // This function gets audio for an external playout sink.
+  // During transmission, this function should be called every ~10 ms
+  // to obtain a new 10 ms frame of audio. The length of the block will
+  // be 160, 320, 441 or 480 samples (for 16000, 32000, 44100 or
+  // 48000 kHz sampling rates respectively).
+  virtual int ExternalPlayoutGetData(
+                                     int16_t speechData10ms[], int samplingFreqHz,
+                                     int current_delay_ms, int& lengthSamples) = 0;
 
-    // Pulls an audio frame from the specified |channel| for external mixing.
-    // If the |desired_sample_rate_hz| is 0, the signal will be returned with
-    // its native frequency, otherwise it will be resampled. Valid frequencies
-    // are 16000, 22050, 32000, 44100 or 48000 kHz.
-    virtual int GetAudioFrame(int channel, int desired_sample_rate_hz,
-                              AudioFrame* frame) = 0;
+  // Pulls an audio frame from the specified |channel| for external mixing.
+  // If the |desired_sample_rate_hz| is 0, the signal will be returned with
+  // its native frequency, otherwise it will be resampled. Valid frequencies
+  // are 16000, 22050, 32000, 44100 or 48000 kHz.
+  virtual int GetAudioFrame(int channel,
+                            int desired_sample_rate_hz,
+                            AudioFrame* frame) = 0;
 
-    // Sets the state of external mixing. Cannot be changed during playback.
-    virtual int SetExternalMixing(int channel, bool enable) = 0;
+  // Sets the state of external mixing. Cannot be changed during playback.
+  virtual int SetExternalMixing(int channel, bool enable) = 0;
 
-protected:
-    VoEExternalMedia() {}
-    virtual ~VoEExternalMedia() {}
+ protected:
+  VoEExternalMedia() {}
+  virtual ~VoEExternalMedia() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_file.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_file.h
index bd14284b0e..f3a3a1f510 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_file.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_file.h
@@ -44,124 +44,99 @@ namespace webrtc {
 
 class VoiceEngine;
 
-class WEBRTC_DLLEXPORT VoEFile
-{
-public:
-    // Factory for the VoEFile sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEFile* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEFile {
+ public:
+  // Factory for the VoEFile sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEFile* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEFile sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoEFile sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Starts playing and mixing files with the local speaker signal for
-    // playout.
-    virtual int StartPlayingFileLocally(
-        int channel,
-        const char fileNameUTF8[1024],
-        bool loop = false,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0,
-        int startPointMs = 0,
-        int stopPointMs = 0) = 0;
+  // Starts playing and mixing files with the local speaker signal for
+  // playout.
+  virtual int StartPlayingFileLocally(
+      int channel,
+      const char fileNameUTF8[1024],
+      bool loop = false,
+      FileFormats format = kFileFormatPcm16kHzFile,
+      float volumeScaling = 1.0,
+      int startPointMs = 0,
+      int stopPointMs = 0) = 0;
 
-    // Starts playing and mixing streams with the local speaker signal for
-    // playout.
-    virtual int StartPlayingFileLocally(
-        int channel,
-        InStream* stream,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0,
-        int startPointMs = 0, int stopPointMs = 0) = 0;
+  // Starts playing and mixing streams with the local speaker signal for
+  // playout.
+  virtual int StartPlayingFileLocally(
+      int channel,
+      InStream* stream,
+      FileFormats format = kFileFormatPcm16kHzFile,
+      float volumeScaling = 1.0,
+      int startPointMs = 0,
+      int stopPointMs = 0) = 0;
 
-    // Stops playback of a file on a specific |channel|.
-    virtual int StopPlayingFileLocally(int channel) = 0;
+  // Stops playback of a file on a specific |channel|.
+  virtual int StopPlayingFileLocally(int channel) = 0;
 
-    // Returns the current file playing state for a specific |channel|.
-    virtual int IsPlayingFileLocally(int channel) = 0;
+  // Returns the current file playing state for a specific |channel|.
+  virtual int IsPlayingFileLocally(int channel) = 0;
 
-    // Starts reading data from a file and transmits the data either
-    // mixed with or instead of the microphone signal.
-    virtual int StartPlayingFileAsMicrophone(
-        int channel,
-        const char fileNameUTF8[1024],
-        bool loop = false ,
-        bool mixWithMicrophone = false,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0) = 0;
+  // Starts reading data from a file and transmits the data either
+  // mixed with or instead of the microphone signal.
+  virtual int StartPlayingFileAsMicrophone(
+      int channel,
+      const char fileNameUTF8[1024],
+      bool loop = false,
+      bool mixWithMicrophone = false,
+      FileFormats format = kFileFormatPcm16kHzFile,
+      float volumeScaling = 1.0) = 0;
 
-    // Starts reading data from a stream and transmits the data either
-    // mixed with or instead of the microphone signal.
-    virtual int StartPlayingFileAsMicrophone(
-        int channel,
-        InStream* stream,
-        bool mixWithMicrophone = false,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0) = 0;
+  // Starts reading data from a stream and transmits the data either
+  // mixed with or instead of the microphone signal.
+  virtual int StartPlayingFileAsMicrophone(
+      int channel,
+      InStream* stream,
+      bool mixWithMicrophone = false,
+      FileFormats format = kFileFormatPcm16kHzFile,
+      float volumeScaling = 1.0) = 0;
 
-    // Stops playing of a file as microphone signal for a specific |channel|.
-    virtual int StopPlayingFileAsMicrophone(int channel) = 0;
+  // Stops playing of a file as microphone signal for a specific |channel|.
+  virtual int StopPlayingFileAsMicrophone(int channel) = 0;
 
-    // Returns whether the |channel| is currently playing a file as microphone.
-    virtual int IsPlayingFileAsMicrophone(int channel) = 0;
+  // Returns whether the |channel| is currently playing a file as microphone.
+  virtual int IsPlayingFileAsMicrophone(int channel) = 0;
 
-    // Starts recording the mixed playout audio.
-    virtual int StartRecordingPlayout(int channel,
-                                      const char* fileNameUTF8,
-                                      CodecInst* compression = NULL,
-                                      int maxSizeBytes = -1) = 0;
+  // Starts recording the mixed playout audio.
+  virtual int StartRecordingPlayout(int channel,
+                                    const char* fileNameUTF8,
+                                    CodecInst* compression = NULL,
+                                    int maxSizeBytes = -1) = 0;
 
-    // Stops recording the mixed playout audio.
-    virtual int StopRecordingPlayout(int channel) = 0;
+  // Stops recording the mixed playout audio.
+  virtual int StopRecordingPlayout(int channel) = 0;
 
-    virtual int StartRecordingPlayout(int channel,
-                                      OutStream* stream,
-                                      CodecInst* compression = NULL) = 0;
+  virtual int StartRecordingPlayout(int channel,
+                                    OutStream* stream,
+                                    CodecInst* compression = NULL) = 0;
 
-    // Starts recording the microphone signal to a file.
-    virtual int StartRecordingMicrophone(const char* fileNameUTF8,
-                                         CodecInst* compression = NULL,
-                                         int maxSizeBytes = -1) = 0;
+  // Starts recording the microphone signal to a file.
+  virtual int StartRecordingMicrophone(const char* fileNameUTF8,
+                                       CodecInst* compression = NULL,
+                                       int maxSizeBytes = -1) = 0;
 
-    // Starts recording the microphone signal to a stream.
-    virtual int StartRecordingMicrophone(OutStream* stream,
-                                         CodecInst* compression = NULL) = 0;
+  // Starts recording the microphone signal to a stream.
+  virtual int StartRecordingMicrophone(OutStream* stream,
+                                       CodecInst* compression = NULL) = 0;
 
-    // Stops recording the microphone signal.
-    virtual int StopRecordingMicrophone() = 0;
+  // Stops recording the microphone signal.
+  virtual int StopRecordingMicrophone() = 0;
 
-    // Don't use. To be removed.
-    virtual int ScaleLocalFilePlayout(int channel, float scale) { return -1; }
-    virtual int ScaleFileAsMicrophonePlayout(
-            int channel, float scale) { return -1; }
-    virtual int GetFileDuration(const char* fileNameUTF8, int& durationMs,
-            FileFormats format = kFileFormatPcm16kHzFile) { return -1; }
-    virtual int GetPlaybackPosition(int channel, int& positionMs) { return -1; }
-    virtual int ConvertPCMToWAV(const char* fileNameInUTF8,
-                                const char* fileNameOutUTF8) { return -1; }
-    virtual int ConvertPCMToWAV(InStream* streamIn,
-                                OutStream* streamOut) { return -1; }
-    virtual int ConvertWAVToPCM(const char* fileNameInUTF8,
-                                const char* fileNameOutUTF8) { return -1; }
-    virtual int ConvertWAVToPCM(InStream* streamIn,
-                                OutStream* streamOut) { return -1; }
-    virtual int ConvertPCMToCompressed(const char* fileNameInUTF8,
-                                       const char* fileNameOutUTF8,
-                                       CodecInst* compression) { return -1; }
-    virtual int ConvertPCMToCompressed(InStream* streamIn,
-                                       OutStream* streamOut,
-                                       CodecInst* compression) { return -1; }
-    virtual int ConvertCompressedToPCM(const char* fileNameInUTF8,
-            const char* fileNameOutUTF8) { return -1; }
-    virtual int ConvertCompressedToPCM(InStream* streamIn,
-                                       OutStream* streamOut) { return -1; }
-protected:
-    VoEFile() {}
-    virtual ~VoEFile() {}
+ protected:
+  VoEFile() {}
+  virtual ~VoEFile() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h
index d69fa89973..41244de4ef 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h
@@ -38,104 +38,94 @@ namespace webrtc {
 
 class VoiceEngine;
 
-class WEBRTC_DLLEXPORT VoEHardware
-{
-public:
-    // Factory for the VoEHardware sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEHardware* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEHardware {
+ public:
+  // Factory for the VoEHardware sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEHardware* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEHardware sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoEHardware sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Gets the number of audio devices available for recording.
-    virtual int GetNumOfRecordingDevices(int& devices) = 0;
+  // Gets the number of audio devices available for recording.
+  virtual int GetNumOfRecordingDevices(int& devices) = 0;
 
-    // Gets the number of audio devices available for playout.
-    virtual int GetNumOfPlayoutDevices(int& devices) = 0;
+  // Gets the number of audio devices available for playout.
+  virtual int GetNumOfPlayoutDevices(int& devices) = 0;
 
-    // Gets the name of a specific recording device given by an |index|.
-    // On Windows Vista/7, it also retrieves an additional unique ID
-    // (GUID) for the recording device.
-    virtual int GetRecordingDeviceName(int index, char strNameUTF8[128],
-                                       char strGuidUTF8[128]) = 0;
-
-    // Gets the name of a specific playout device given by an |index|.
-    // On Windows Vista/7, it also retrieves an additional unique ID
-    // (GUID) for the playout device.
-    virtual int GetPlayoutDeviceName(int index, char strNameUTF8[128],
+  // Gets the name of a specific recording device given by an |index|.
+  // On Windows Vista/7, it also retrieves an additional unique ID
+  // (GUID) for the recording device.
+  virtual int GetRecordingDeviceName(int index,
+                                     char strNameUTF8[128],
                                      char strGuidUTF8[128]) = 0;
+
+  // Gets the name of a specific playout device given by an |index|.
+  // On Windows Vista/7, it also retrieves an additional unique ID
+  // (GUID) for the playout device.
+  virtual int GetPlayoutDeviceName(int index,
+                                   char strNameUTF8[128],
+                                   char strGuidUTF8[128]) = 0;
  
-    // Checks if the sound card is available to be opened for recording.
-    virtual int GetRecordingDeviceStatus(bool& isAvailable) = 0;
+  // Checks if the sound card is available to be opened for recording.
+  virtual int GetRecordingDeviceStatus(bool& isAvailable) = 0;
 
-    // Checks if the sound card is available to be opened for playout.
-    virtual int GetPlayoutDeviceStatus(bool& isAvailable) = 0;
+  // Checks if the sound card is available to be opened for playout.
+  virtual int GetPlayoutDeviceStatus(bool& isAvailable) = 0;
 
-    // Sets the audio device used for recording.
-    virtual int SetRecordingDevice(
-        int index, StereoChannel recordingChannel = kStereoBoth) = 0;
+  // Sets the audio device used for recording.
+  virtual int SetRecordingDevice(
+      int index,
+      StereoChannel recordingChannel = kStereoBoth) = 0;
 
-    // Sets the audio device used for playout.
-    virtual int SetPlayoutDevice(int index) = 0;
+  // Sets the audio device used for playout.
+  virtual int SetPlayoutDevice(int index) = 0;
 
-    // Sets the type of audio device layer to use.
-    virtual int SetAudioDeviceLayer(AudioLayers audioLayer) = 0;
+  // Sets the type of audio device layer to use.
+  virtual int SetAudioDeviceLayer(AudioLayers audioLayer) = 0;
 
-    // Gets the currently used (active) audio device layer.
-    virtual int GetAudioDeviceLayer(AudioLayers& audioLayer) = 0;
+  // Gets the currently used (active) audio device layer.
+  virtual int GetAudioDeviceLayer(AudioLayers& audioLayer) = 0;
  
-    // Gets the VoiceEngine's current CPU consumption in terms of the percent
-    // of total CPU availability. [Windows only]
-    virtual int GetCPULoad(int& loadPercent) = 0;
+  // Gets the VoiceEngine's current CPU consumption in terms of the percent
+  // of total CPU availability. [Windows only]
+  virtual int GetCPULoad(int& loadPercent) = 0;
 
-    // Not supported
-    virtual int ResetAudioDevice() = 0;
+  // Not supported
+  virtual int ResetAudioDevice() = 0;
 
-    // Not supported
-    virtual int AudioDeviceControl(
-        unsigned int par1, unsigned int par2, unsigned int par3) = 0;
+  // Not supported
+  virtual int AudioDeviceControl(
+                                 unsigned int par1, unsigned int par2, unsigned int par3) = 0;
 
-    // Not supported
-    virtual int SetLoudspeakerStatus(bool enable) = 0;
+  // Not supported
+  virtual int SetLoudspeakerStatus(bool enable) = 0;
 
-    // Not supported
-    virtual int GetLoudspeakerStatus(bool& enabled) = 0;
+  // Not supported
+  virtual int GetLoudspeakerStatus(bool& enabled) = 0;
 
-    // Native sample rate controls (samples/sec)
-    virtual int SetRecordingSampleRate(unsigned int samples_per_sec) = 0;
-    virtual int RecordingSampleRate(unsigned int* samples_per_sec) const = 0;
-    virtual int SetPlayoutSampleRate(unsigned int samples_per_sec) = 0;
-    virtual int PlayoutSampleRate(unsigned int* samples_per_sec) const = 0;
+  // Native sample rate controls (samples/sec)
+  virtual int SetRecordingSampleRate(unsigned int samples_per_sec) = 0;
+  virtual int RecordingSampleRate(unsigned int* samples_per_sec) const = 0;
+  virtual int SetPlayoutSampleRate(unsigned int samples_per_sec) = 0;
+  virtual int PlayoutSampleRate(unsigned int* samples_per_sec) const = 0;
 
-    virtual bool BuiltInAECIsAvailable() const = 0;
+  // Queries and controls platform audio effects on Android devices.
+  virtual bool BuiltInAECIsAvailable() const = 0;
+  virtual int EnableBuiltInAEC(bool enable) = 0;
+  virtual bool BuiltInAECIsEnabled() const = 0;
+  virtual bool BuiltInAGCIsAvailable() const = 0;
+  virtual int EnableBuiltInAGC(bool enable) = 0;
+  virtual bool BuiltInNSIsAvailable() const = 0;
+  virtual int EnableBuiltInNS(bool enable) = 0;
 
-    // *Experimental - not recommended for use.*
-    // Enables the Windows Core Audio built-in AEC. Fails on other platforms.
-    //
-    // Currently incompatible with the standard VoE AEC and AGC; don't attempt
-    // to enable them while this is active.
-    //
-    // Must be called before VoEBase::StartSend(). When enabled:
-    // 1. VoEBase::StartPlayout() must be called before VoEBase::StartSend().
-    // 2. VoEBase::StopSend() should be called before VoEBase::StopPlayout().
-    //    The reverse order may cause garbage audio to be rendered or the
-    //    capture side to halt until StopSend() is called.
-    //
-    //    As a consequence, SetPlayoutDevice() should be used with caution
-    //    during a call. It will function, but may cause the above issues for
-    //    the duration it takes to complete. (In practice, it should complete
-    //    fast enough to avoid audible degradation).
-    virtual int EnableBuiltInAEC(bool enable) = 0;
-    virtual bool BuiltInAECIsEnabled() const = 0;
-
-protected:
-    VoEHardware() {}
-    virtual ~VoEHardware() {}
+ protected:
+  VoEHardware() {}
+  virtual ~VoEHardware() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_neteq_stats.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_neteq_stats.h
index 1e8c2407f5..fb70cae18a 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_neteq_stats.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_neteq_stats.h
@@ -17,33 +17,33 @@ namespace webrtc {
 
 class VoiceEngine;
 
-class WEBRTC_DLLEXPORT VoENetEqStats
-{
-public:
-    // Factory for the VoENetEqStats sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoENetEqStats* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoENetEqStats {
+ public:
+  // Factory for the VoENetEqStats sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoENetEqStats* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoENetEqStats sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoENetEqStats sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Get the "in-call" statistics from NetEQ.
-    // The statistics are reset after the query.
-    virtual int GetNetworkStatistics(int channel, NetworkStatistics& stats) = 0;
+  // Get the "in-call" statistics from NetEQ.
+  // The statistics are reset after the query.
+  virtual int GetNetworkStatistics(int channel, NetworkStatistics& stats) = 0;
 
-    // Get statistics of calls to AudioCodingModule::PlayoutData10Ms().
-    virtual int GetDecodingCallStatistics(
-        int channel, AudioDecodingCallStats* stats) const = 0;
+  // Get statistics of calls to AudioCodingModule::PlayoutData10Ms().
+  virtual int GetDecodingCallStatistics(
+      int channel,
+      AudioDecodingCallStats* stats) const = 0;
 
-protected:
-    VoENetEqStats() {}
-    virtual ~VoENetEqStats() {}
+ protected:
+  VoENetEqStats() {}
+  virtual ~VoENetEqStats() {}
 };
 
 }  // namespace webrtc
 
-#endif    // #ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
+#endif  // #ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h
index ff8b8e1f94..c5b0aebd88 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h
@@ -35,58 +35,59 @@
 #define WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
 
 #include "webrtc/common_types.h"
+#include "webrtc/transport.h"
 
 namespace webrtc {
 
 class VoiceEngine;
 
 // VoENetwork
-class WEBRTC_DLLEXPORT VoENetwork
-{
-public:
-    // Factory for the VoENetwork sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoENetwork* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoENetwork {
+ public:
+  // Factory for the VoENetwork sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoENetwork* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoENetwork sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoENetwork sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Installs and enables a user-defined external transport protocol for a
-    // specified |channel|.
-    virtual int RegisterExternalTransport(
-        int channel, Transport& transport) = 0;
+  // Installs and enables a user-defined external transport protocol for a
+  // specified |channel|. Returns -1 in case of an error, 0 otherwise.
+  virtual int RegisterExternalTransport(int channel, Transport& transport) = 0;
 
-    // Removes and disables a user-defined external transport protocol for a
-    // specified |channel|.
-    virtual int DeRegisterExternalTransport(int channel) = 0;
+  // Removes and disables a user-defined external transport protocol for a
+  // specified |channel|. Returns -1 in case of an error, 0 otherwise.
+  virtual int DeRegisterExternalTransport(int channel) = 0;
 
-    // The packets received from the network should be passed to this
-    // function when external transport is enabled. Note that the data
-    // including the RTP-header must also be given to the VoiceEngine.
-    virtual int ReceivedRTPPacket(int channel,
-                                  const void* data,
-                                  size_t length) = 0;
-    virtual int ReceivedRTPPacket(int channel,
-                                  const void* data,
-                                  size_t length,
-                                  const PacketTime& packet_time) {
-      return 0;
-    }
+  // The packets received from the network should be passed to this
+  // function when external transport is enabled. Note that the data
+  // including the RTP-header must also be given to the VoiceEngine.
+  //  Returns -1 in case of an error, 0 otherwise.
+  virtual int ReceivedRTPPacket(int channel,
+                                const void* data,
+                                size_t length) = 0;
+  virtual int ReceivedRTPPacket(int channel,
+                                const void* data,
+                                size_t length,
+                                const PacketTime& packet_time) {
+    return 0;
+  }
 
-    // The packets received from the network should be passed to this
-    // function when external transport is enabled. Note that the data
-    // including the RTCP-header must also be given to the VoiceEngine.
-    virtual int ReceivedRTCPPacket(int channel,
-                                   const void* data,
-                                   size_t length) = 0;
+  // The packets received from the network should be passed to this
+  // function when external transport is enabled. Note that the data
+  // including the RTCP-header must also be given to the VoiceEngine.
+  //  Returns -1 in case of an error, 0 otherwise.
+  virtual int ReceivedRTCPPacket(int channel,
+                                 const void* data,
+                                 size_t length) = 0;
 
-protected:
-    VoENetwork() {}
-    virtual ~VoENetwork() {}
+ protected:
+  VoENetwork() {}
+  virtual ~VoENetwork() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
index ee0fae0014..c502de9d96 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
@@ -44,38 +44,35 @@
 
 namespace webrtc {
 
-class ViENetwork;
 class VoiceEngine;
 
 // VoERTPObserver
-class WEBRTC_DLLEXPORT VoERTPObserver
-{
-public:
-    virtual void OnIncomingCSRCChanged(
-        int channel, unsigned int CSRC, bool added) = 0;
+class WEBRTC_DLLEXPORT VoERTPObserver {
+ public:
+  virtual void OnIncomingCSRCChanged(int channel,
+                                     unsigned int CSRC,
+                                     bool added) = 0;
 
-    virtual void OnIncomingSSRCChanged(
-        int channel, unsigned int SSRC) = 0;
+  virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0;
 
-protected:
-    virtual ~VoERTPObserver() {}
+ protected:
+  virtual ~VoERTPObserver() {}
 };
 
 // CallStatistics
-struct CallStatistics
-{
-    unsigned short fractionLost;
-    unsigned int cumulativeLost;
-    unsigned int extendedMax;
-    unsigned int jitterSamples;
-    int64_t rttMs;
-    size_t bytesSent;
-    int packetsSent;
-    size_t bytesReceived;
-    int packetsReceived;
-    // The capture ntp time (in local timebase) of the first played out audio
-    // frame.
-    int64_t capture_start_ntp_time_ms_;
+struct CallStatistics {
+  unsigned short fractionLost;
+  unsigned int cumulativeLost;
+  unsigned int extendedMax;
+  unsigned int jitterSamples;
+  int64_t rttMs;
+  size_t bytesSent;
+  int packetsSent;
+  size_t bytesReceived;
+  int packetsReceived;
+  // The capture ntp time (in local timebase) of the first played out audio
+  // frame.
+  int64_t capture_start_ntp_time_ms_;
 };
 
 // See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details.
@@ -89,7 +86,7 @@ struct SenderInfo {
 
 // See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
 struct ReportBlock {
-  uint32_t sender_SSRC; // SSRC of sender
+  uint32_t sender_SSRC;  // SSRC of sender
   uint32_t source_SSRC;
   uint8_t fraction_lost;
   uint32_t cumulative_num_packets_lost;
@@ -100,179 +97,119 @@ struct ReportBlock {
 };
 
 // VoERTP_RTCP
-class WEBRTC_DLLEXPORT VoERTP_RTCP
-{
-public:
+class WEBRTC_DLLEXPORT VoERTP_RTCP {
+ public:
+  // Factory for the VoERTP_RTCP sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
 
-    // Factory for the VoERTP_RTCP sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
+  // Releases the VoERTP_RTCP sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Releases the VoERTP_RTCP sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Sets the local RTP synchronization source identifier (SSRC) explicitly.
+  virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
 
-    // Sets the local RTP synchronization source identifier (SSRC) explicitly.
-    virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
+  // Gets the local RTP SSRC of a specified |channel|.
+  virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
 
-    // Gets the local RTP SSRC of a specified |channel|.
-    virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
+  // Gets the SSRC of the incoming RTP packets.
+  virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
 
-    // Gets the SSRC of the incoming RTP packets.
-    virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
-
-    // Sets the status of rtp-audio-level-indication on a specific |channel|.
-    virtual int SetSendAudioLevelIndicationStatus(int channel,
-                                                  bool enable,
-                                                  unsigned char id = 1) = 0;
-
-    // Sets the status of receiving rtp-audio-level-indication on a specific
-    // |channel|.
-    virtual int SetReceiveAudioLevelIndicationStatus(int channel,
-                                                     bool enable,
-                                                     unsigned char id = 1) {
-      // TODO(wu): Remove default implementation once talk is updated.
-      return 0;
-    }
-
-    // Sets the status of sending absolute sender time on a specific |channel|.
-    virtual int SetSendAbsoluteSenderTimeStatus(int channel,
+  // Sets the status of rtp-audio-level-indication on a specific |channel|.
+  virtual int SetSendAudioLevelIndicationStatus(int channel,
                                                 bool enable,
-                                                unsigned char id) = 0;
+                                                unsigned char id = 1) = 0;
 
-    // Sets status of receiving absolute sender time on a specific |channel|.
-    virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
+  // Sets the status of receiving rtp-audio-level-indication on a specific
+  // |channel|.
+  virtual int SetReceiveAudioLevelIndicationStatus(int channel,
                                                    bool enable,
-                                                   unsigned char id) = 0;
+                                                   unsigned char id = 1) {
+    // TODO(wu): Remove default implementation once talk is updated.
+    return 0;
+  }
 
-    // Sets the RTCP status on a specific |channel|.
-    virtual int SetRTCPStatus(int channel, bool enable) = 0;
+  // Sets the status of sending absolute sender time on a specific |channel|.
+  virtual int SetSendAbsoluteSenderTimeStatus(int channel,
+                                              bool enable,
+                                              unsigned char id) = 0;
 
-    // Gets the RTCP status on a specific |channel|.
-    virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
+  // Sets status of receiving absolute sender time on a specific |channel|.
+  virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
+                                                 bool enable,
+                                                 unsigned char id) = 0;
 
-    // Sets the canonical name (CNAME) parameter for RTCP reports on a
-    // specific |channel|.
-    virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
+  // Sets the RTCP status on a specific |channel|.
+  virtual int SetRTCPStatus(int channel, bool enable) = 0;
 
-    // TODO(holmer): Remove this API once it has been removed from
-    // fakewebrtcvoiceengine.h.
-    virtual int GetRTCP_CNAME(int channel, char cName[256]) {
-      return -1;
-    }
+  // Gets the RTCP status on a specific |channel|.
+  virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
 
-    // Gets the canonical name (CNAME) parameter for incoming RTCP reports
-    // on a specific channel.
-    virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
+  // Sets the canonical name (CNAME) parameter for RTCP reports on a
+  // specific |channel|.
+  virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
 
-    // Gets RTCP data from incoming RTCP Sender Reports.
-    virtual int GetRemoteRTCPReceiverInfo(
-        int channel, uint32_t& NTPHigh, uint32_t& NTPLow,
-        uint32_t& receivedPacketCount, uint64_t& receivedOctetCount,
-        uint32_t& jitter, uint16_t& fractionLost,
-        uint32_t& cumulativeLost,
-        int32_t& rttMs) = 0;
+  // TODO(holmer): Remove this API once it has been removed from
+  // fakewebrtcvoiceengine.h.
+  virtual int GetRTCP_CNAME(int channel, char cName[256]) { return -1; }
 
-    // Gets RTP statistics for a specific |channel|.
-    virtual int GetRTPStatistics(
-        int channel, unsigned int& averageJitterMs, unsigned int& maxJitterMs,
-        unsigned int& discardedPackets, unsigned int& cumulativeLost) = 0;
+  // Gets the canonical name (CNAME) parameter for incoming RTCP reports
+  // on a specific channel.
+  virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
 
-    // Gets RTCP statistics for a specific |channel|.
-    virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
+  // Gets RTCP data from incoming RTCP Sender Reports.
+  virtual int GetRemoteRTCPReceiverInfo(
+      int channel, uint32_t& NTPHigh, uint32_t& NTPLow,
+      uint32_t& receivedPacketCount, uint64_t& receivedOctetCount,
+      uint32_t& jitter, uint16_t& fractionLost,
+      uint32_t& cumulativeLost,
+      int32_t& rttMs) = 0;
 
-    // Gets the report block parts of the last received RTCP Sender Report (SR),
-    // or RTCP Receiver Report (RR) on a specified |channel|. Each vector
-    // element also contains the SSRC of the sender in addition to a report
-    // block.
-    virtual int GetRemoteRTCPReportBlocks(
-        int channel, std::vector<ReportBlock>* receive_blocks) = 0;
+  // Gets RTP statistics for a specific |channel|.
+  virtual int GetRTPStatistics(int channel,
+                               unsigned int& averageJitterMs,
+                               unsigned int& maxJitterMs,
+                               unsigned int& discardedPackets,
+                               unsigned int& cumulativeLost) = 0;
 
-    // Sets the Redundant Coding (RED) status on a specific |channel|.
-    // TODO(minyue): Make SetREDStatus() pure virtual when fakewebrtcvoiceengine
-    // in talk is ready.
-    virtual int SetREDStatus(
-        int channel, bool enable, int redPayloadtype = -1) { return -1; }
+  // Gets RTCP statistics for a specific |channel|.
+  virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
 
-    // Gets the RED status on a specific |channel|.
-    // TODO(minyue): Make GetREDStatus() pure virtual when fakewebrtcvoiceengine
-    // in talk is ready.
-    virtual int GetREDStatus(
-        int channel, bool& enabled, int& redPayloadtype) { return -1; }
+  // Gets the report block parts of the last received RTCP Sender Report (SR),
+  // or RTCP Receiver Report (RR) on a specified |channel|. Each vector
+  // element also contains the SSRC of the sender in addition to a report
+  // block.
+  virtual int GetRemoteRTCPReportBlocks(
+      int channel,
+      std::vector<ReportBlock>* receive_blocks) = 0;
 
-    // Sets the Forward Error Correction (FEC) status on a specific |channel|.
-    // TODO(minyue): Remove SetFECStatus() when SetFECStatus() is replaced by
-    // SetREDStatus() in fakewebrtcvoiceengine.
-    virtual int SetFECStatus(
-        int channel, bool enable, int redPayloadtype = -1) {
-      return SetREDStatus(channel, enable, redPayloadtype);
-    };
+  // Sets the Redundant Coding (RED) status on a specific |channel|.
+  // TODO(minyue): Make SetREDStatus() pure virtual when fakewebrtcvoiceengine
+  // in talk is ready.
+  virtual int SetREDStatus(int channel, bool enable, int redPayloadtype = -1) {
+    return -1;
+  }
 
-    // Gets the FEC status on a specific |channel|.
-    // TODO(minyue): Remove GetFECStatus() when GetFECStatus() is replaced by
-    // GetREDStatus() in fakewebrtcvoiceengine.
-    virtual int GetFECStatus(
-        int channel, bool& enabled, int& redPayloadtype) {
-      return SetREDStatus(channel, enabled, redPayloadtype);
-    }
+  // Gets the RED status on a specific |channel|.
+  // TODO(minyue): Make GetREDStatus() pure virtual when fakewebrtcvoiceengine
+  // in talk is ready.
+  virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype) {
+    return -1;
+  }
 
-    // This function enables Negative Acknowledgment (NACK) using RTCP,
-    // implemented based on RFC 4585. NACK retransmits RTP packets if lost on
-    // the network. This creates a lossless transport at the expense of delay.
-    // If using NACK, NACK should be enabled on both endpoints in a call.
-    virtual int SetNACKStatus(int channel,
-                              bool enable,
-                              int maxNoPackets) = 0;
+  // This function enables Negative Acknowledgment (NACK) using RTCP,
+  // implemented based on RFC 4585. NACK retransmits RTP packets if lost on
+  // the network. This creates a lossless transport at the expense of delay.
+  // If using NACK, NACK should be enabled on both endpoints in a call.
+  virtual int SetNACKStatus(int channel, bool enable, int maxNoPackets) = 0;
 
-    // Enables capturing of RTP packets to a binary file on a specific
-    // |channel| and for a given |direction|. The file can later be replayed
-    // using e.g. RTP Tools rtpplay since the binary file format is
-    // compatible with the rtpdump format.
-    virtual int StartRTPDump(
-        int channel, const char fileNameUTF8[1024],
-        RTPDirections direction = kRtpIncoming) = 0;
-
-    // Disables capturing of RTP packets to a binary file on a specific
-    // |channel| and for a given |direction|.
-    virtual int StopRTPDump(
-        int channel, RTPDirections direction = kRtpIncoming) = 0;
-
-    // Gets the the current RTP capturing state for the specified
-    // |channel| and |direction|.
-    virtual int RTPDumpIsActive(
-        int channel, RTPDirections direction = kRtpIncoming) = 0;
-
-    // Sets video engine channel to receive incoming audio packets for
-    // aggregated bandwidth estimation. Takes ownership of the ViENetwork
-    // interface.
-    virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network,
-                                        int video_channel) {
-      return 0;
-    }
-
-    // Will be removed. Don't use.
-    virtual int RegisterRTPObserver(int channel,
-            VoERTPObserver& observer) { return -1; };
-    virtual int DeRegisterRTPObserver(int channel) { return -1; };
-    virtual int GetRemoteCSRCs(int channel,
-            unsigned int arrCSRC[15]) { return -1; };
-    virtual int InsertExtraRTPPacket(
-            int channel, unsigned char payloadType, bool markerBit,
-            const char* payloadData, unsigned short payloadSize) { return -1; };
-    virtual int GetRemoteRTCPSenderInfo(
-            int channel, SenderInfo* sender_info) { return -1; };
-    virtual int SendApplicationDefinedRTCPPacket(
-            int channel, unsigned char subType, unsigned int name,
-            const char* data, unsigned short dataLengthInBytes) { return -1; };
-    virtual int GetLastRemoteTimeStamp(int channel,
-            uint32_t* lastRemoteTimeStamp) { return -1; };
-
-protected:
-    VoERTP_RTCP() {}
-    virtual ~VoERTP_RTCP() {}
+ protected:
+  VoERTP_RTCP() {}
+  virtual ~VoERTP_RTCP() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_video_sync.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_video_sync.h
index e06f1ee566..a28929f205 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_video_sync.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_video_sync.h
@@ -41,70 +41,63 @@ class RtpReceiver;
 class RtpRtcp;
 class VoiceEngine;
 
-class WEBRTC_DLLEXPORT VoEVideoSync
-{
-public:
-    // Factory for the VoEVideoSync sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEVideoSync {
+ public:
+  // Factory for the VoEVideoSync sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEVideoSync sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoEVideoSync sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Gets the current sound card buffer size (playout delay).
-    virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;
+  // Gets the current sound card buffer size (playout delay).
+  virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;
 
-    // Sets a minimum target delay for the jitter buffer. This delay is
-    // maintained by the jitter buffer, unless channel condition (jitter in
-    // inter-arrival times) dictates a higher required delay. The overall
-    // jitter buffer delay is max of |delay_ms| and the latency that NetEq
-    // computes based on inter-arrival times and its playout mode.
-    virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
+  // Sets a minimum target delay for the jitter buffer. This delay is
+  // maintained by the jitter buffer, unless channel condition (jitter in
+  // inter-arrival times) dictates a higher required delay. The overall
+  // jitter buffer delay is max of |delay_ms| and the latency that NetEq
+  // computes based on inter-arrival times and its playout mode.
+  virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
 
-    // Sets the current a/v delay in ms (negative is video leading) if known,
-    // otherwise 0.
-    virtual int SetCurrentSyncOffset(int channel, int offset_ms) = 0;
+  // Sets the current a/v delay in ms (negative is video leading) if known,
+  // otherwise 0.
+  virtual int SetCurrentSyncOffset(int channel, int offset_ms) = 0;
 
-    // Sets an initial delay for the playout jitter buffer. The playout of the
-    // audio is delayed by |delay_ms| in milliseconds. Thereafter, the delay is
-    // maintained, unless NetEq's internal mechanism requires a higher latency.
-    // Such a latency is computed based on inter-arrival times and NetEq's
-    // playout mode.
-    virtual int SetInitialPlayoutDelay(int channel, int delay_ms) = 0;
+  // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay),
+  // the |playout_buffer_delay_ms| and |avsync_offset_ms| for a specified
+  // |channel|.
+  virtual int GetDelayEstimate(int channel,
+                               int* jitter_buffer_delay_ms,
+                               int* playout_buffer_delay_ms,
+                               int* avsync_offset_ms) = 0;
 
-    // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay),
-    // the |playout_buffer_delay_ms| and |avsync_offset_ms| for a specified
-    // |channel|.
-    virtual int GetDelayEstimate(int channel,
-                                 int* jitter_buffer_delay_ms,
-                                 int* playout_buffer_delay_ms,
-                                 int* avsync_offset_ms) = 0;
+  // Returns the least required jitter buffer delay. This is computed by the
+  // the jitter buffer based on the inter-arrival time of RTP packets and
+  // playout mode. NetEq maintains this latency unless a higher value is
+  // requested by calling SetMinimumPlayoutDelay().
+  virtual int GetLeastRequiredDelayMs(int channel) const = 0;
 
-    // Returns the least required jitter buffer delay. This is computed by the
-    // the jitter buffer based on the inter-arrival time of RTP packets and
-    // playout mode. NetEq maintains this latency unless a higher value is
-    // requested by calling SetMinimumPlayoutDelay().
-    virtual int GetLeastRequiredDelayMs(int channel) const = 0;
+  // Manual initialization of the RTP timestamp.
+  virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
 
-    // Manual initialization of the RTP timestamp.
-    virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
+  // Manual initialization of the RTP sequence number.
+  virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
 
-    // Manual initialization of the RTP sequence number.
-    virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
+  // Get the received RTP timestamp
+  virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
 
-    // Get the received RTP timestamp
-    virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
+  virtual int GetRtpRtcp(int channel,
+                         RtpRtcp** rtpRtcpModule,
+                         RtpReceiver** rtp_receiver) = 0;
 
-    virtual int GetRtpRtcp (int channel, RtpRtcp** rtpRtcpModule,
-                            RtpReceiver** rtp_receiver) = 0;
-
-protected:
-    VoEVideoSync() { }
-    virtual ~VoEVideoSync() { }
+ protected:
+  VoEVideoSync() {}
+  virtual ~VoEVideoSync() {}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_volume_control.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_volume_control.h
index 9e3e2fdecf..cb9dc66fc5 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_volume_control.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_volume_control.h
@@ -42,78 +42,71 @@ namespace webrtc {
 
 class VoiceEngine;
 
-class WEBRTC_DLLEXPORT VoEVolumeControl
-{
-public:
-    // Factory for the VoEVolumeControl sub-API. Increases an internal
-    // reference counter if successful. Returns NULL if the API is not
-    // supported or if construction fails.
-    static VoEVolumeControl* GetInterface(VoiceEngine* voiceEngine);
+class WEBRTC_DLLEXPORT VoEVolumeControl {
+ public:
+  // Factory for the VoEVolumeControl sub-API. Increases an internal
+  // reference counter if successful. Returns NULL if the API is not
+  // supported or if construction fails.
+  static VoEVolumeControl* GetInterface(VoiceEngine* voiceEngine);
 
-    // Releases the VoEVolumeControl sub-API and decreases an internal
-    // reference counter. Returns the new reference count. This value should
-    // be zero for all sub-API:s before the VoiceEngine object can be safely
-    // deleted.
-    virtual int Release() = 0;
+  // Releases the VoEVolumeControl sub-API and decreases an internal
+  // reference counter. Returns the new reference count. This value should
+  // be zero for all sub-API:s before the VoiceEngine object can be safely
+  // deleted.
+  virtual int Release() = 0;
 
-    // Sets the speaker |volume| level. Valid range is [0,255].
-    virtual int SetSpeakerVolume(unsigned int volume) = 0;
+  // Sets the speaker |volume| level. Valid range is [0,255].
+  virtual int SetSpeakerVolume(unsigned int volume) = 0;
 
-    // Gets the speaker |volume| level.
-    virtual int GetSpeakerVolume(unsigned int& volume) = 0;
+  // Gets the speaker |volume| level.
+  virtual int GetSpeakerVolume(unsigned int& volume) = 0;
 
-    // Sets the microphone volume level. Valid range is [0,255].
-    virtual int SetMicVolume(unsigned int volume) = 0;
+  // Sets the microphone volume level. Valid range is [0,255].
+  virtual int SetMicVolume(unsigned int volume) = 0;
 
-    // Gets the microphone volume level.
-    virtual int GetMicVolume(unsigned int& volume) = 0;
+  // Gets the microphone volume level.
+  virtual int GetMicVolume(unsigned int& volume) = 0;
 
-    // Mutes the microphone input signal completely without affecting
-    // the audio device volume.
-    virtual int SetInputMute(int channel, bool enable) = 0;
+  // Mutes the microphone input signal completely without affecting
+  // the audio device volume.
+  virtual int SetInputMute(int channel, bool enable) = 0;
 
-    // Gets the current microphone input mute state.
-    virtual int GetInputMute(int channel, bool& enabled) = 0;
+  // Gets the current microphone input mute state.
+  virtual int GetInputMute(int channel, bool& enabled) = 0;
 
-    // Gets the microphone speech |level|, mapped non-linearly to the range
-    // [0,9].
-    virtual int GetSpeechInputLevel(unsigned int& level) = 0;
+  // Gets the microphone speech |level|, mapped non-linearly to the range
+  // [0,9].
+  virtual int GetSpeechInputLevel(unsigned int& level) = 0;
 
-    // Gets the speaker speech |level|, mapped non-linearly to the range
-    // [0,9].
-    virtual int GetSpeechOutputLevel(int channel, unsigned int& level) = 0;
+  // Gets the speaker speech |level|, mapped non-linearly to the range
+  // [0,9].
+  virtual int GetSpeechOutputLevel(int channel, unsigned int& level) = 0;
 
-    // Gets the microphone speech |level|, mapped linearly to the range
-    // [0,32768].
-    virtual int GetSpeechInputLevelFullRange(unsigned int& level) = 0;
+  // Gets the microphone speech |level|, mapped linearly to the range
+  // [0,32768].
+  virtual int GetSpeechInputLevelFullRange(unsigned int& level) = 0;
 
-    // Gets the speaker speech |level|, mapped linearly to the range [0,32768].
-    virtual int GetSpeechOutputLevelFullRange(
-        int channel, unsigned int& level) = 0;
+  // Gets the speaker speech |level|, mapped linearly to the range [0,32768].
+  virtual int GetSpeechOutputLevelFullRange(int channel,
+                                            unsigned int& level) = 0;
 
-    // Sets a volume |scaling| applied to the outgoing signal of a specific
-    // channel. Valid scale range is [0.0, 10.0].
-    virtual int SetChannelOutputVolumeScaling(int channel, float scaling) = 0;
+  // Sets a volume |scaling| applied to the outgoing signal of a specific
+  // channel. Valid scale range is [0.0, 10.0].
+  virtual int SetChannelOutputVolumeScaling(int channel, float scaling) = 0;
 
-    // Gets the current volume scaling for a specified |channel|.
-    virtual int GetChannelOutputVolumeScaling(int channel, float& scaling) = 0;
+  // Gets the current volume scaling for a specified |channel|.
+  virtual int GetChannelOutputVolumeScaling(int channel, float& scaling) = 0;
 
-    // Scales volume of the |left| and |right| channels independently.
-    // Valid scale range is [0.0, 1.0].
-    virtual int SetOutputVolumePan(int channel, float left, float right) = 0;
+  // Scales volume of the |left| and |right| channels independently.
+  // Valid scale range is [0.0, 1.0].
+  virtual int SetOutputVolumePan(int channel, float left, float right) = 0;
 
-    // Gets the current left and right scaling factors.
-    virtual int GetOutputVolumePan(int channel, float& left, float& right) = 0;
+  // Gets the current left and right scaling factors.
+  virtual int GetOutputVolumePan(int channel, float& left, float& right) = 0;
 
-    // Don't use. Will be removed.
-    virtual int SetSystemOutputMute(bool enable) { return -1; }
-    virtual int GetSystemOutputMute(bool &enabled) { return -1; }
-    virtual int SetSystemInputMute(bool enable) { return -1; }
-    virtual int GetSystemInputMute(bool& enabled) { return -1; }
-
-protected:
-    VoEVolumeControl() {}
-    virtual ~VoEVolumeControl() {}
+ protected:
+  VoEVolumeControl(){}
+  virtual ~VoEVolumeControl(){}
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/level_indicator.cc b/media/webrtc/trunk/webrtc/voice_engine/level_indicator.cc
index 61dfbcda87..68a837edb9 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/level_indicator.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/level_indicator.cc
@@ -9,8 +9,8 @@
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/voice_engine/level_indicator.h"
 
 namespace webrtc {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/mock/mock_voe_observer.h b/media/webrtc/trunk/webrtc/voice_engine/mock/mock_voe_observer.h
similarity index 100%
rename from media/webrtc/trunk/webrtc/voice_engine/include/mock/mock_voe_observer.h
rename to media/webrtc/trunk/webrtc/voice_engine/mock/mock_voe_observer.h
diff --git a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc
index ca358022a5..5ea73571d8 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/voice_engine/monitor_module.h"
 
 namespace webrtc  {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h
index 42ea74d7e2..fe915b320b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
 #define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
 
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h b/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h
index 0e5068659c..b35ccd8756 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h
@@ -12,7 +12,7 @@
 #define WEBRTC_VOICE_ENGINE_NETWORK_PREDICTOR_H_
 
 #include "webrtc/base/exp_filter.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc
index 55ee20da03..28ff57fb49 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc
@@ -12,7 +12,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/voice_engine/network_predictor.h"
-#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace voe {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/output_mixer.cc b/media/webrtc/trunk/webrtc/voice_engine/output_mixer.cc
index d87a6f0dab..b265a4655f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/output_mixer.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/output_mixer.cc
@@ -10,11 +10,12 @@
 
 #include "webrtc/voice_engine/output_mixer.h"
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/utility.h"
@@ -35,29 +36,6 @@ OutputMixer::NewMixedAudio(int32_t id,
     _audioFrame.id_ = id;
 }
 
-void OutputMixer::MixedParticipants(
-    int32_t id,
-    const ParticipantStatistics* participantStatistics,
-    uint32_t size)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
-                 "OutputMixer::MixedParticipants(id=%d, size=%u)", id, size);
-}
-
-void OutputMixer::VADPositiveParticipants(int32_t id,
-    const ParticipantStatistics* participantStatistics, uint32_t size)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
-                 "OutputMixer::VADPositiveParticipants(id=%d, size=%u)",
-                 id, size);
-}
-
-void OutputMixer::MixedAudioLevel(int32_t id, uint32_t level)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
-                 "OutputMixer::MixedAudioLevel(id=%d, level=%u)", id, level);
-}
-
 void OutputMixer::PlayNotification(int32_t id, uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
@@ -131,8 +109,7 @@ OutputMixer::OutputMixer(uint32_t instanceId) :
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::OutputMixer() - ctor");
 
-    if ((_mixerModule.RegisterMixedStreamCallback(*this) == -1) ||
-        (_mixerModule.RegisterMixerStatusCallback(*this, 100) == -1))
+    if (_mixerModule.RegisterMixedStreamCallback(this) == -1)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
                      "OutputMixer::OutputMixer() failed to register mixer"
@@ -170,7 +147,6 @@ OutputMixer::~OutputMixer()
             _outputFileRecorderPtr = NULL;
         }
     }
-    _mixerModule.UnRegisterMixerStatusCallback();
     _mixerModule.UnRegisterMixedStreamCallback();
     delete &_mixerModule;
     delete &_callbackCritSect;
@@ -240,14 +216,14 @@ int32_t
 OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
                                  bool mixable)
 {
-    return _mixerModule.SetMixabilityStatus(participant, mixable);
+    return _mixerModule.SetMixabilityStatus(&participant, mixable);
 }
 
 int32_t
 OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant,
                                           bool mixable)
 {
-    return _mixerModule.SetAnonymousMixabilityStatus(participant,mixable);
+    return _mixerModule.SetAnonymousMixabilityStatus(&participant, mixable);
 }
 
 int32_t
@@ -492,11 +468,12 @@ int OutputMixer::StopRecordingPlayout()
 }
 
 int OutputMixer::GetMixedAudio(int sample_rate_hz,
-                               int num_channels,
+                               size_t num_channels,
                                AudioFrame* frame) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
-               "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%d)",
-               sample_rate_hz, num_channels);
+  WEBRTC_TRACE(
+      kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+      "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
+      sample_rate_hz, num_channels);
 
   // --- Record playout if enabled
   {
@@ -587,6 +564,7 @@ void OutputMixer::APMAnalyzeReverseStream(AudioFrame &audioFrame) {
   if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) == -1) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
                  "AudioProcessingModule::AnalyzeReverseStream() => error");
+    RTC_DCHECK(false);
   }
 }
 
@@ -643,7 +621,7 @@ OutputMixer::InsertInbandDtmfTone()
     } else
     {
         // stereo
-        for (int i = 0; i < _audioFrame.samples_per_channel_; i++)
+        for (size_t i = 0; i < _audioFrame.samples_per_channel_; i++)
         {
             _audioFrame.data_[2 * i] = toneBuffer[i];
             _audioFrame.data_[2 * i + 1] = 0;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/output_mixer.h b/media/webrtc/trunk/webrtc/voice_engine/output_mixer.h
index aa07d43d31..a12fdc7a12 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/output_mixer.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/output_mixer.h
@@ -13,9 +13,9 @@
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
@@ -32,7 +32,6 @@ namespace voe {
 class Statistics;
 
 class OutputMixer : public AudioMixerOutputReceiver,
-                    public AudioMixerStatusReceiver,
                     public FileCallback
 {
 public:
@@ -64,7 +63,7 @@ public:
     int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
                                          bool mixable);
 
-    int GetMixedAudio(int sample_rate_hz, int num_channels,
+    int GetMixedAudio(int sample_rate_hz, size_t num_channels,
                       AudioFrame* audioFrame);
 
     // VoEVolumeControl
@@ -93,19 +92,6 @@ public:
         const AudioFrame** uniqueAudioFrames,
         uint32_t size);
 
-    // from AudioMixerStatusReceiver
-    virtual void MixedParticipants(
-        int32_t id,
-        const ParticipantStatistics* participantStatistics,
-        uint32_t size);
-
-    virtual void VADPositiveParticipants(
-        int32_t id,
-        const ParticipantStatistics* participantStatistics,
-        uint32_t size);
-
-    virtual void MixedAudioLevel(int32_t id, uint32_t level);
-
     // For file recording
     void PlayNotification(int32_t id, uint32_t durationMs);
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc b/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc
index ad00e038f6..6c23f3ecdf 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc
@@ -11,8 +11,8 @@
 #include "webrtc/voice_engine/shared_data.h"
 
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
@@ -23,16 +23,15 @@ namespace voe {
 
 static int32_t _gInstanceCounter = 0;
 
-SharedData::SharedData(const Config& config) :
-    _instanceId(++_gInstanceCounter),
-    _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
-    _channelManager(_gInstanceCounter, config),
-    _engineStatistics(_gInstanceCounter),
-    _audioDevicePtr(NULL),
-    _moduleProcessThreadPtr(ProcessThread::Create()),
-    _externalRecording(false),
-    _externalPlayout(false)
-{
+SharedData::SharedData(const Config& config)
+    : _instanceId(++_gInstanceCounter),
+      _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
+      _channelManager(_gInstanceCounter, config),
+      _engineStatistics(_gInstanceCounter),
+      _audioDevicePtr(NULL),
+      _moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")),
+      _externalRecording(false),
+      _externalPlayout(false) {
     Trace::CreateTrace();
     if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
     {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/shared_data.h b/media/webrtc/trunk/webrtc/voice_engine/shared_data.h
index 3ab1d45ac5..2c6685a73e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/shared_data.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/shared_data.h
@@ -14,7 +14,7 @@
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/media/webrtc/trunk/webrtc/voice_engine/statistics.cc b/media/webrtc/trunk/webrtc/voice_engine/statistics.cc
index 4717fd1391..3787e67122 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/statistics.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/statistics.cc
@@ -13,8 +13,8 @@
 
 #include "webrtc/voice_engine/statistics.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc
index b0a26e06a0..766b9e7a8e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc
@@ -15,7 +15,7 @@
 
 #include "webrtc/voice_engine/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h"
 
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
 
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ -177,7 +177,7 @@ private:
     static bool Run(void* ptr);
     bool Process();
 private:
-    rtc::scoped_ptr<ThreadWrapper> _thread;
+    rtc::PlatformThread _thread;
 };
 
 ThreadTest::~ThreadTest()
@@ -188,7 +188,7 @@ ThreadTest::~ThreadTest()
 
 ThreadTest::ThreadTest()
 {
-    _thread = ThreadWrapper::CreateThread(Run, this, "ThreadTest thread");
+    _thread(Run, this, "ThreadTest thread");
 }
 
 bool ThreadTest::Run(void* ptr)
@@ -221,7 +221,6 @@ bool ThreadTest::Process()
         __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
                 "set local receiver 2 failed");
     }
-    veData2.hardware->SetLoudspeakerStatus(false);
     veData2.volume->SetSpeakerVolume(204);
     veData2.base->StartReceive(0);
     if(veData2.base->StartPlayout(0) < 0)
@@ -1115,43 +1114,6 @@ Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume(
     return 0;
 }
 
-/////////////////////////////////////////////
-// [Hardware] Set loudspeaker status
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetLoudspeakerStatus(
-        JNIEnv *,
-        jobject,
-        jboolean enable)
-{
-    VALIDATE_HARDWARE_POINTER;
-    if (veData1.hardware->SetLoudspeakerStatus(enable) != 0)
-    {
-        return -1;
-    }
-
-    /*VALIDATE_RTP_RTCP_POINTER;
-
-     if (veData1.rtp_rtcp->SetREDStatus(0, enable, -1) != 0)
-     {
-     __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-         "Could not set RED");
-     return -1;
-     }
-     else if(enable)
-     {
-     __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-         "Could enable RED");
-     }
-     else
-     {
-     __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
-         "Could disable RED");
-     }*/
-
-    return 0;
-}
-
 //////////////////////////////////////////////////////////////////
 // "Local" functions (i.e. not Java accessible)
 //////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h b/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
index 60fe83926b..34d408a95f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
@@ -239,14 +239,6 @@ JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetECStatus
 JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume
   (JNIEnv *, jobject, jint);
 
-/*
- * Class:     org_webrtc_voiceengine_test_AndroidTest
- * Method:    SetLoudspeakerStatus
- * Signature: (Z)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetLoudspeakerStatus
-  (JNIEnv *, jobject, jboolean);
-
 #ifdef __cplusplus
 }
 #endif
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
new file mode 100644
index 0000000000..70f68298f5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -0,0 +1,288 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
+
+#include <string>
+
+#include "webrtc/base/byteorder.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+
+namespace {
+  static const unsigned int kReflectorSsrc = 0x0000;
+  static const unsigned int kLocalSsrc = 0x0001;
+  static const unsigned int kFirstRemoteSsrc = 0x0002;
+  static const webrtc::CodecInst kCodecInst =
+      {120, "opus", 48000, 960, 2, 64000};
+  static const int kAudioLevelHeaderId = 1;
+
+  static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
+    const size_t ssrc_pos = 4;
+    unsigned int ssrc = 0;
+    if (len >= (ssrc_pos + sizeof(ssrc))) {
+      ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
+    }
+    return ssrc;
+  }
+}  // namespace
+
+namespace voetest {
+
+ConferenceTransport::ConferenceTransport()
+    : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+      stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
+      packet_event_(webrtc::EventWrapper::Create()),
+      thread_(Run, this, "ConferenceTransport"),
+      rtt_ms_(0),
+      stream_count_(0),
+      rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
+  rtp_header_parser_->
+      RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
+                                 kAudioLevelHeaderId);
+
+  local_voe_ = webrtc::VoiceEngine::Create();
+  local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
+  local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
+  local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
+
+  // In principle, we can use one VoiceEngine to achieve the same goal. Well, in
+  // here, we use two engines to make it more like reality.
+  remote_voe_ = webrtc::VoiceEngine::Create();
+  remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
+  remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
+  remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
+  remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
+  remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
+
+  EXPECT_EQ(0, local_base_->Init());
+  local_sender_ = local_base_->CreateChannel();
+  EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
+  EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
+  EXPECT_EQ(0, local_rtp_rtcp_->
+      SetSendAudioLevelIndicationStatus(local_sender_, true,
+                                        kAudioLevelHeaderId));
+
+  EXPECT_EQ(0, local_base_->StartSend(local_sender_));
+
+  EXPECT_EQ(0, remote_base_->Init());
+  reflector_ = remote_base_->CreateChannel();
+  EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
+  EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
+
+  thread_.Start();
+  thread_.SetPriority(rtc::kHighPriority);
+}
+
+ConferenceTransport::~ConferenceTransport() {
+  // Must stop sending, otherwise DispatchPackets() cannot quit.
+  EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_));
+  EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_));
+
+  while (!streams_.empty()) {
+    auto stream = streams_.begin();
+    RemoveStream(stream->first);
+  }
+
+  thread_.Stop();
+
+  remote_file_->Release();
+  remote_rtp_rtcp_->Release();
+  remote_network_->Release();
+  remote_base_->Release();
+
+  local_rtp_rtcp_->Release();
+  local_network_->Release();
+  local_base_->Release();
+
+  EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
+  EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
+}
+
+bool ConferenceTransport::SendRtp(const uint8_t* data,
+                                  size_t len,
+                                  const webrtc::PacketOptions& options) {
+  StorePacket(Packet::Rtp, data, len);
+  return true;
+}
+
+bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) {
+  StorePacket(Packet::Rtcp, data, len);
+  return true;
+}
+
+int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
+    const {
+  webrtc::CriticalSectionScoped lock(stream_crit_.get());
+  auto it = streams_.find(sender_ssrc);
+  if (it != streams_.end()) {
+    return it->second.second;
+  }
+  return -1;
+}
+
+void ConferenceTransport::StorePacket(Packet::Type type,
+                                      const void* data,
+                                      size_t len) {
+  {
+    webrtc::CriticalSectionScoped lock(pq_crit_.get());
+    packet_queue_.push_back(Packet(type, data, len, rtc::Time()));
+  }
+  packet_event_->Set();
+}
+
+// This simulates the flow of RTP and RTCP packets. Complications like that
+// a packet is first sent to the reflector, and then forwarded to the receiver
+// are simplified, in this particular case, to a direct link between the sender
+// and the receiver.
+void ConferenceTransport::SendPacket(const Packet& packet) {
+  int destination = -1;
+
+  switch (packet.type_) {
+    case Packet::Rtp: {
+      webrtc::RTPHeader rtp_header;
+      rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
+      if (rtp_header.ssrc == kLocalSsrc) {
+        remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
+                                           packet.len_, webrtc::PacketTime());
+      } else {
+        if (loudest_filter_.ForwardThisPacket(rtp_header)) {
+          destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
+          if (destination != -1) {
+            local_network_->ReceivedRTPPacket(destination, packet.data_,
+                                              packet.len_,
+                                              webrtc::PacketTime());
+          }
+        }
+      }
+      break;
+    }
+    case Packet::Rtcp: {
+      unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_);
+      if (sender_ssrc == kLocalSsrc) {
+        remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
+                                            packet.len_);
+      } else if (sender_ssrc == kReflectorSsrc) {
+        local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
+                                           packet.len_);
+      } else {
+        destination = GetReceiverChannelForSsrc(sender_ssrc);
+        if (destination != -1) {
+          local_network_->ReceivedRTCPPacket(destination, packet.data_,
+                                             packet.len_);
+        }
+      }
+      break;
+    }
+  }
+}
+
+bool ConferenceTransport::DispatchPackets() {
+  switch (packet_event_->Wait(1000)) {
+    case webrtc::kEventSignaled:
+      break;
+    case webrtc::kEventTimeout:
+      return true;
+    case webrtc::kEventError:
+      ADD_FAILURE() << "kEventError encountered.";
+      return true;
+  }
+
+  while (true) {
+    Packet packet;
+    {
+      webrtc::CriticalSectionScoped lock(pq_crit_.get());
+      if (packet_queue_.empty())
+        break;
+      packet = packet_queue_.front();
+      packet_queue_.pop_front();
+    }
+
+    int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
+    int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
+    if (sleep_ms > 0) {
+      // Every packet should be delayed by half of RTT.
+      webrtc::SleepMs(sleep_ms);
+    }
+
+    SendPacket(packet);
+  }
+  return true;
+}
+
+void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
+  rtt_ms_ = rtt_ms;
+}
+
+unsigned int ConferenceTransport::AddStream(std::string file_name,
+                                            webrtc::FileFormats format) {
+  const int new_sender = remote_base_->CreateChannel();
+  EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
+
+  const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
+  EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
+  EXPECT_EQ(0, remote_rtp_rtcp_->
+      SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId));
+
+  EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
+  EXPECT_EQ(0, remote_base_->StartSend(new_sender));
+  EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
+      new_sender, file_name.c_str(), true, false, format, 1.0));
+
+  const int new_receiver = local_base_->CreateChannel();
+  EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
+
+  EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
+  // Receive channels have to have the same SSRC in order to send receiver
+  // reports with this SSRC.
+  EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));
+
+  {
+    webrtc::CriticalSectionScoped lock(stream_crit_.get());
+    streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
+  }
+  return remote_ssrc;  // remote ssrc used as stream id.
+}
+
+bool ConferenceTransport::RemoveStream(unsigned int id) {
+  webrtc::CriticalSectionScoped lock(stream_crit_.get());
+  auto it = streams_.find(id);
+  if (it == streams_.end()) {
+    return false;
+  }
+  EXPECT_EQ(0, remote_network_->
+      DeRegisterExternalTransport(it->second.second));
+  EXPECT_EQ(0, local_network_->
+      DeRegisterExternalTransport(it->second.first));
+  EXPECT_EQ(0, remote_base_->DeleteChannel(it->second.second));
+  EXPECT_EQ(0, local_base_->DeleteChannel(it->second.first));
+  streams_.erase(it);
+  return true;
+}
+
+bool ConferenceTransport::StartPlayout(unsigned int id) {
+  int dst = GetReceiverChannelForSsrc(id);
+  if (dst == -1) {
+    return false;
+  }
+  EXPECT_EQ(0, local_base_->StartPlayout(dst));
+  return true;
+}
+
+bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
+                                                webrtc::CallStatistics* stats) {
+  int dst = GetReceiverChannelForSsrc(id);
+  if (dst == -1) {
+    return false;
+  }
+  EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
+  return true;
+}
+}  // namespace voetest
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
new file mode 100644
index 0000000000..5d105aa39e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -0,0 +1,164 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
+#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
+
+#include <deque>
+#include <map>
+#include <utility>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_file.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
+
+static const size_t kMaxPacketSizeByte = 1500;
+
+namespace voetest {
+
+// This class is to simulate a conference call. There are two Voice Engines, one
+// for local channels and the other for remote channels. There is a simulated
+// reflector, which exchanges RTCP with local channels. For simplicity, it
+// also uses the Voice Engine for remote channels. One can add streams by
+// calling AddStream(), which creates a remote sender channel and a local
+// receive channel. The remote sender channel plays a file as microphone in a
+// looped fashion. Received streams are mixed and played.
+
+class ConferenceTransport: public webrtc::Transport {
+ public:
+  ConferenceTransport();
+  virtual ~ConferenceTransport();
+
+  /* SetRtt()
+   * Set RTT between local channels and reflector.
+   *
+   * Input:
+   *   rtt_ms : RTT in milliseconds.
+   */
+  void SetRtt(unsigned int rtt_ms);
+
+  /* AddStream()
+   * Adds a stream in the conference.
+   *
+   * Input:
+   *   file_name : name of the file to be added as microphone input.
+   *   format    : format of the input file.
+   *
+   * Returns stream id.
+   */
+  unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
+
+  /* RemoveStream()
+   * Removes a stream with specified ID from the conference.
+   *
+   * Input:
+   *   id : stream id.
+   *
+   * Returns false if the specified stream does not exist, true if succeeds.
+   */
+  bool RemoveStream(unsigned int id);
+
+  /* StartPlayout()
+   * Starts playing out the stream with specified ID, using the default device.
+   *
+   * Input:
+   *   id : stream id.
+   *
+   * Returns false if the specified stream does not exist, true if succeeds.
+   */
+  bool StartPlayout(unsigned int id);
+
+  /* GetReceiverStatistics()
+   * Gets RTCP statistics of the stream with specified ID.
+   *
+   * Input:
+   *   id : stream id;
+   *   stats : pointer to a CallStatistics to store the result.
+   *
+   * Returns false if the specified stream does not exist, true if succeeds.
+   */
+  bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
+
+  // Inherit from class webrtc::Transport.
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override;
+  bool SendRtcp(const uint8_t *data, size_t len) override;
+
+ private:
+  struct Packet {
+    enum Type { Rtp, Rtcp, } type_;
+
+    Packet() : len_(0) {}
+    Packet(Type type, const void* data, size_t len, uint32_t time_ms)
+        : type_(type), len_(len), send_time_ms_(time_ms) {
+      EXPECT_LE(len_, kMaxPacketSizeByte);
+      memcpy(data_, data, len_);
+    }
+
+    uint8_t data_[kMaxPacketSizeByte];
+    size_t len_;
+    uint32_t send_time_ms_;
+  };
+
+  static bool Run(void* transport) {
+    return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
+  }
+
+  int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
+  void StorePacket(Packet::Type type, const void* data, size_t len);
+  void SendPacket(const Packet& packet);
+  bool DispatchPackets();
+
+  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
+  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
+  const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
+  rtc::PlatformThread thread_;
+
+  unsigned int rtt_ms_;
+  unsigned int stream_count_;
+
+  std::map<unsigned int, std::pair<int, int>> streams_
+    GUARDED_BY(stream_crit_.get());
+  std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
+
+  int local_sender_;  // Channel Id of local sender
+  int reflector_;
+
+  webrtc::VoiceEngine* local_voe_;
+  webrtc::VoEBase* local_base_;
+  webrtc::VoERTP_RTCP* local_rtp_rtcp_;
+  webrtc::VoENetwork* local_network_;
+
+  webrtc::VoiceEngine* remote_voe_;
+  webrtc::VoEBase* remote_base_;
+  webrtc::VoECodec* remote_codec_;
+  webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
+  webrtc::VoENetwork* remote_network_;
+  webrtc::VoEFile* remote_file_;
+
+  LoudestFilter loudest_filter_;
+
+  const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
+};
+}  // namespace voetest
+
+#endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc
deleted file mode 100644
index c825ea5861..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h"
-#include "webrtc/voice_engine/voice_engine_defines.h"
-
-FakeExternalTransport::FakeExternalTransport(webrtc::VoENetwork* ptr)
-    : my_network_(ptr),
-      lock_(NULL),
-      event_(NULL),
-      length_(0),
-      channel_(0),
-      delay_is_enabled_(0),
-      delay_time_in_ms_(0) {
-  const char* thread_name = "external_thread";
-  lock_ = webrtc::CriticalSectionWrapper::CreateCriticalSection();
-  event_ = webrtc::EventWrapper::Create();
-  thread_ = webrtc::ThreadWrapper::CreateThread(Run, this, thread_name);
-  if (thread_) {
-    thread_->Start();
-    thread_->SetPriority(webrtc::kHighPriority);
-  }
-}
-
-FakeExternalTransport::~FakeExternalTransport() {
-  if (thread_) {
-    event_->Set();
-    thread_->Stop();
-    delete event_;
-    event_ = NULL;
-    delete lock_;
-    lock_ = NULL;
-  }
-}
-
-bool FakeExternalTransport::Run(void* ptr) {
-  return static_cast<FakeExternalTransport*> (ptr)->Process();
-}
-
-bool FakeExternalTransport::Process() {
-  switch (event_->Wait(500)) {
-    case webrtc::kEventSignaled:
-      lock_->Enter();
-      my_network_->ReceivedRTPPacket(channel_, packet_buffer_, length_,
-                                     webrtc::PacketTime());
-      lock_->Leave();
-      return true;
-    case webrtc::kEventTimeout:
-      return true;
-    case webrtc::kEventError:
-      break;
-  }
-  return true;
-}
-
-int FakeExternalTransport::SendPacket(int channel,
-                                      const void *data,
-                                      size_t len) {
-  lock_->Enter();
-  if (len < 1612) {
-    memcpy(packet_buffer_, (const unsigned char*) data, len);
-    length_ = len;
-    channel_ = channel;
-  }
-  lock_->Leave();
-  event_->Set();  // Triggers ReceivedRTPPacket() from worker thread.
-  return static_cast<int>(len);
-}
-
-int FakeExternalTransport::SendRTCPPacket(int channel,
-                                          const void *data,
-                                          size_t len) {
-  if (delay_is_enabled_) {
-    webrtc::SleepMs(delay_time_in_ms_);
-  }
-  my_network_->ReceivedRTCPPacket(channel, data, len);
-  return static_cast<int>(len);
-}
-
-void FakeExternalTransport::SetDelayStatus(bool enable,
-                                           unsigned int delayInMs) {
-  delay_is_enabled_ = enable;
-  delay_time_in_ms_ = delayInMs;
-}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h
deleted file mode 100644
index aecc58264a..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKES_FAKE_EXTERNAL_TRANSPORT_H_
-#define VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKES_FAKE_EXTERNAL_TRANSPORT_H_
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class VoENetwork;
-}
-
-class FakeExternalTransport : public webrtc::Transport {
- public:
-  explicit FakeExternalTransport(webrtc::VoENetwork* ptr);
-  virtual ~FakeExternalTransport();
-  int SendPacket(int channel, const void* data, size_t len) override;
-  int SendRTCPPacket(int channel, const void* data, size_t len) override;
-  void SetDelayStatus(bool enabled, unsigned int delayInMs = 100);
-
-  webrtc::VoENetwork* my_network_;
- private:
-  static bool Run(void* ptr);
-  bool Process();
- private:
-  rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
-  webrtc::CriticalSectionWrapper* lock_;
-  webrtc::EventWrapper* event_;
- private:
-  unsigned char packet_buffer_[1612];
-  size_t length_;
-  int channel_;
-  bool delay_is_enabled_;
-  int delay_time_in_ms_;
-};
-
-#endif  // VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKES_FAKE_EXTERNAL_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h
index 3e1345af54..9e82fbc0a5 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h
@@ -18,10 +18,10 @@ class FakeMediaProcess : public webrtc::VoEMediaProcess {
   virtual void Process(int channel,
                        const webrtc::ProcessingTypes type,
                        int16_t audio_10ms[],
-                       int length,
+                       size_t length,
                        int sampling_freq_hz,
                        bool stereo) {
-    for (int i = 0; i < length; i++) {
+    for (size_t i = 0; i < length; i++) {
       if (!stereo) {
         audio_10ms[i] = static_cast<int16_t>(audio_10ms[i] *
             sin(2.0 * 3.14 * frequency * 400.0 / sampling_freq_hz));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
new file mode 100644
index 0000000000..d4438a4e15
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
+
+#include "webrtc/base/checks.h"
+
+namespace voetest {
+
+void LoudestFilter::RemoveTimeoutStreams(uint32_t time_ms) {
+  auto it = stream_levels_.begin();
+  while (it != stream_levels_.end()) {
+    if (rtc::TimeDiff(time_ms, it->second.last_time_ms) >
+        kStreamTimeOutMs) {
+      stream_levels_.erase(it++);
+    } else {
+      ++it;
+    }
+  }
+}
+
+unsigned int LoudestFilter::FindQuietestStream() {
+  int quietest_level = kInvalidAudioLevel;
+  unsigned int quietest_ssrc = 0;
+  for (auto stream : stream_levels_) {
+    // A smaller value if audio level corresponds to a louder sound.
+    if (quietest_level == kInvalidAudioLevel ||
+        stream.second.audio_level > quietest_level) {
+      quietest_level = stream.second.audio_level;
+      quietest_ssrc = stream.first;
+    }
+  }
+  return quietest_ssrc;
+}
+
+bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) {
+  uint32_t time_now_ms = rtc::Time();
+  RemoveTimeoutStreams(time_now_ms);
+
+  int source_ssrc = rtp_header.ssrc;
+  int audio_level = rtp_header.extension.hasAudioLevel ?
+      rtp_header.extension.audioLevel : kInvalidAudioLevel;
+
+  if (audio_level == kInvalidAudioLevel) {
+    // Always forward streams with unknown audio level, and don't keep their
+    // states.
+    return true;
+  }
+
+  auto it = stream_levels_.find(source_ssrc);
+  if (it != stream_levels_.end()) {
+    // Stream has been forwarded. Update and continue to forward.
+    it->second.audio_level = audio_level;
+    it->second.last_time_ms = time_now_ms;
+    return true;
+  }
+
+  if (stream_levels_.size() < kMaxMixSize) {
+    stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
+    return true;
+  }
+
+  unsigned int quietest_ssrc = FindQuietestStream();
+  RTC_CHECK_NE(0u, quietest_ssrc);
+  // A smaller value if audio level corresponds to a louder sound.
+  if (audio_level < stream_levels_[quietest_ssrc].audio_level) {
+    stream_levels_.erase(quietest_ssrc);
+    stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
+    return true;
+  }
+  return false;
+}
+
+}  // namespace voetest
+
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h
new file mode 100644
index 0000000000..73b801cc98
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
+#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
+
+#include <map>
+#include "webrtc/base/timeutils.h"
+#include "webrtc/common_types.h"
+
+namespace voetest {
+
+class LoudestFilter {
+ public:
+  /* ForwardThisPacket()
+   * Decide whether to forward a RTP packet, given its header.
+   *
+   * Input:
+   *   rtp_header : Header of the RTP packet of interest.
+   */
+  bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header);
+
+ private:
+  struct Status {
+    void Set(int audio_level, uint32_t last_time_ms) {
+      this->audio_level = audio_level;
+      this->last_time_ms = last_time_ms;
+    }
+    int audio_level;
+    uint32_t last_time_ms;
+  };
+
+  void RemoveTimeoutStreams(uint32_t time_ms);
+  unsigned int FindQuietestStream();
+
+  // Keeps the streams being forwarded in pair<SSRC, Status>.
+  std::map<unsigned int, Status> stream_levels_;
+
+  const int32_t kStreamTimeOutMs = 5000;
+  const size_t kMaxMixSize = 3;
+  const int kInvalidAudioLevel = 128;
+};
+
+
+}  // namespace voetest
+
+#endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
index c54d289cac..efdf633f23 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
@@ -28,11 +28,6 @@ AfterInitializationFixture::AfterInitializationFixture()
   webrtc::AudioProcessing* audioproc = webrtc::AudioProcessing::Create(config);
 
   EXPECT_EQ(0, voe_base_->Init(NULL, audioproc));
-
-#if defined(WEBRTC_ANDROID)
-  EXPECT_EQ(0, voe_hardware_->SetLoudspeakerStatus(false));
-#endif
-
   EXPECT_EQ(0, voe_base_->RegisterVoiceEngineObserver(*error_observer_));
 }
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index cee5a58f8e..116ff0aec3 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -13,38 +13,42 @@
 
 #include <deque>
 
+#include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
 
 class TestErrorObserver;
 
 class LoopBackTransport : public webrtc::Transport {
  public:
-  LoopBackTransport(webrtc::VoENetwork* voe_network)
+  LoopBackTransport(webrtc::VoENetwork* voe_network, int channel)
       : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
         packet_event_(webrtc::EventWrapper::Create()),
-        thread_(webrtc::ThreadWrapper::CreateThread(
-            NetworkProcess, this, "LoopBackTransport")),
-        voe_network_(voe_network), transmitted_packets_(0) {
-    thread_->Start();
+        thread_(NetworkProcess, this, "LoopBackTransport"),
+        channel_(channel),
+        voe_network_(voe_network),
+        transmitted_packets_(0) {
+    thread_.Start();
   }
 
-  ~LoopBackTransport() { thread_->Stop(); }
+  ~LoopBackTransport() { thread_.Stop(); }
 
-  int SendPacket(int channel, const void* data, size_t len) override {
-    StorePacket(Packet::Rtp, channel, data, len);
-    return static_cast<int>(len);
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override {
+    StorePacket(Packet::Rtp, data, len);
+    return true;
   }
 
-  int SendRTCPPacket(int channel, const void* data, size_t len) override {
-    StorePacket(Packet::Rtcp, channel, data, len);
-    return static_cast<int>(len);
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    StorePacket(Packet::Rtcp, data, len);
+    return true;
   }
 
   void WaitForTransmittedPackets(int32_t packet_count) {
@@ -57,28 +61,32 @@ class LoopBackTransport : public webrtc::Transport {
     }
   }
 
+  void AddChannel(uint32_t ssrc, int channel) {
+    webrtc::CriticalSectionScoped lock(crit_.get());
+    channels_[ssrc] = channel;
+  }
+
  private:
   struct Packet {
     enum Type { Rtp, Rtcp, } type;
 
     Packet() : len(0) {}
-    Packet(Type type, int channel, const void* data, size_t len)
-        : type(type), channel(channel), len(len) {
+    Packet(Type type, const void* data, size_t len)
+        : type(type), len(len) {
       assert(len <= 1500);
       memcpy(this->data, data, len);
     }
 
-    int channel;
     uint8_t data[1500];
     size_t len;
   };
 
-  void StorePacket(Packet::Type type, int channel,
+  void StorePacket(Packet::Type type,
                    const void* data,
                    size_t len) {
     {
       webrtc::CriticalSectionScoped lock(crit_.get());
-      packet_queue_.push_back(Packet(type, channel, data, len));
+      packet_queue_.push_back(Packet(type, data, len));
     }
     packet_event_->Set();
   }
@@ -100,21 +108,34 @@ class LoopBackTransport : public webrtc::Transport {
 
     while (true) {
       Packet p;
+      int channel = channel_;
       {
         webrtc::CriticalSectionScoped lock(crit_.get());
         if (packet_queue_.empty())
           break;
         p = packet_queue_.front();
         packet_queue_.pop_front();
+
+        if (p.type == Packet::Rtp) {
+          uint32_t ssrc =
+              webrtc::ByteReader<uint32_t>::ReadBigEndian(&p.data[8]);
+          if (channels_[ssrc] != 0)
+            channel = channels_[ssrc];
+        }
+        // TODO(pbos): Add RTCP SSRC muxing/demuxing if anything requires it.
       }
 
+      // Minimum RTP header size.
+      if (p.len < 12)
+        continue;
+
       switch (p.type) {
         case Packet::Rtp:
-          voe_network_->ReceivedRTPPacket(p.channel, p.data, p.len,
+          voe_network_->ReceivedRTPPacket(channel, p.data, p.len,
                                           webrtc::PacketTime());
           break;
         case Packet::Rtcp:
-          voe_network_->ReceivedRTCPPacket(p.channel, p.data, p.len);
+          voe_network_->ReceivedRTCPPacket(channel, p.data, p.len);
           break;
       }
       ++transmitted_packets_;
@@ -124,8 +145,10 @@ class LoopBackTransport : public webrtc::Transport {
 
   const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
-  const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
+  rtc::PlatformThread thread_;
   std::deque<Packet> packet_queue_ GUARDED_BY(crit_.get());
+  const int channel_;
+  std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
   webrtc::VoENetwork* const voe_network_;
   webrtc::Atomic32 transmitted_packets_;
 };
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
index 408ebf959d..fc28bcdebb 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
 
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 
 BeforeInitializationFixture::BeforeInitializationFixture()
     : voice_engine_(webrtc::VoiceEngine::Create()) {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
index 7a3fad8399..51db985b4a 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h
@@ -16,7 +16,6 @@
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
index 488f4489bf..abcb2a60a9 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
@@ -61,7 +61,7 @@ void BeforeStreamingFixture::WaitForTransmittedPackets(int32_t packet_count) {
 }
 
 void BeforeStreamingFixture::SetUpLocalPlayback() {
-  transport_ = new LoopBackTransport(voe_network_);
+  transport_ = new LoopBackTransport(voe_network_, channel_);
   EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
 
   webrtc::CodecInst codec;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
index bfc2a30c5f..3a3d83031d 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
@@ -8,6 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <stdio.h>
+#include <string>
+
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/test/test_suite.h"
+#include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
@@ -44,7 +50,7 @@ static bool IsNotViableSendCodec(const char* codec_name) {
 
 TEST_F(CodecTest, PcmuIsDefaultCodecAndHasTheRightValues) {
   EXPECT_EQ(0, voe_codec_->GetSendCodec(channel_, codec_instance_));
-  EXPECT_EQ(1, codec_instance_.channels);
+  EXPECT_EQ(1u, codec_instance_.channels);
   EXPECT_EQ(160, codec_instance_.pacsize);
   EXPECT_EQ(8000, codec_instance_.plfreq);
   EXPECT_EQ(0, codec_instance_.pltype);
@@ -147,17 +153,6 @@ TEST_F(CodecTest, OpusMaxPlaybackRateCanBeSet) {
   }
 }
 
-TEST_F(CodecTest, OpusMaxPlaybackRateCannotBeSetForNonOpus) {
-  for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
-    voe_codec_->GetCodec(i, codec_instance_);
-    if (!_stricmp("opus", codec_instance_.plname)) {
-      continue;
-    }
-    voe_codec_->SetSendCodec(channel_, codec_instance_);
-    EXPECT_EQ(-1, voe_codec_->SetOpusMaxPlaybackRate(channel_, 16000));
-  }
-}
-
 TEST_F(CodecTest, OpusDtxCanBeSetForOpus) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
     voe_codec_->GetCodec(i, codec_instance_);
@@ -177,11 +172,34 @@ TEST_F(CodecTest, OpusDtxCannotBeSetForNonOpus) {
       continue;
     }
     voe_codec_->SetSendCodec(channel_, codec_instance_);
-    EXPECT_EQ(-1, voe_codec_->SetOpusDtx(channel_, false));
     EXPECT_EQ(-1, voe_codec_->SetOpusDtx(channel_, true));
   }
 }
 
+#ifdef ENABLE_RTC_EVENT_LOG
+TEST_F(CodecTest, RtcEventLogIntegrationTest) {
+  webrtc::RtcEventLog* event_log = voe_codec_->GetEventLog();
+  ASSERT_TRUE(event_log);
+
+  // Find the name of the current test, in order to use it as a temporary
+  // filename.
+  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
+  const std::string temp_filename = webrtc::test::OutputPath() +
+                                    test_info->test_case_name() +
+                                    test_info->name();
+  // Create a log file.
+  event_log->StartLogging(temp_filename, 1000);
+  event_log->StopLogging();
+
+  // Check if the file has been created.
+  FILE* event_file = fopen(temp_filename.c_str(), "r");
+  ASSERT_TRUE(event_file);
+  fclose(event_file);
+  // Remove the temporary file.
+  remove(temp_filename.c_str());
+}
+#endif  // ENABLE_RTC_EVENT_LOG
+
 // TODO(xians, phoglund): Re-enable when issue 372 is resolved.
 TEST_F(CodecTest, DISABLED_ManualVerifySendCodecsForAllPacketSizes) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
index e3f81d5cd6..3182e5a43d 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
@@ -8,7 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/base/arraysize.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
@@ -118,8 +119,8 @@ TEST_F(ExternalMediaTest,
   EXPECT_EQ(0, voe_xmedia_->SetExternalMixing(channel_, true));
   ResumePlaying();
   EXPECT_EQ(0, voe_xmedia_->GetAudioFrame(channel_, 0, &frame));
-  EXPECT_LT(0, frame.sample_rate_hz_);
-  EXPECT_LT(0, frame.samples_per_channel_);
+  EXPECT_GT(frame.sample_rate_hz_, 0);
+  EXPECT_GT(frame.samples_per_channel_, 0U);
   PausePlaying();
   EXPECT_EQ(0, voe_xmedia_->SetExternalMixing(channel_, false));
   ResumePlaying();
@@ -132,12 +133,12 @@ TEST_F(ExternalMediaTest,
   PausePlaying();
   EXPECT_EQ(0, voe_xmedia_->SetExternalMixing(channel_, true));
   ResumePlaying();
-  for (size_t i = 0; i < sizeof(kValidFrequencies) / sizeof(int); i++) {
+  for (size_t i = 0; i < arraysize(kValidFrequencies); i++) {
     int f = kValidFrequencies[i];
     EXPECT_EQ(0, voe_xmedia_->GetAudioFrame(channel_, f, &frame))
        << "Resampling succeeds for freq=" << f;
     EXPECT_EQ(f, frame.sample_rate_hz_);
-    EXPECT_EQ(f / 100, frame.samples_per_channel_);
+    EXPECT_EQ(static_cast<size_t>(f / 100), frame.samples_per_channel_);
   }
   PausePlaying();
   EXPECT_EQ(0, voe_xmedia_->SetExternalMixing(channel_, false));
@@ -151,7 +152,7 @@ TEST_F(ExternalMediaTest,
   PausePlaying();
   EXPECT_EQ(0, voe_xmedia_->SetExternalMixing(channel_, true));
   ResumePlaying();
-  for (size_t i = 0; i < sizeof(kInvalidFrequencies) / sizeof(int); i++) {
+  for (size_t i = 0; i < arraysize(kInvalidFrequencies); i++) {
     int f = kInvalidFrequencies[i];
     EXPECT_EQ(-1, voe_xmedia_->GetAudioFrame(channel_, f, &frame))
         << "Resampling fails for freq=" << f;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc
index bcc1ab7b0c..ed822f1c3b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_before_streaming_test.cc
@@ -25,14 +25,6 @@ TEST_F(HardwareBeforeStreamingTest,
   EXPECT_EQ(VE_ALREADY_INITED, voe_base_->LastError());
 }
 
-// Tests that only apply to mobile:
-
-#ifdef WEBRTC_IOS
-TEST_F(HardwareBeforeStreamingTest, ResetsAudioDeviceOnIphone) {
-  EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
-}
-#endif
-
 // Tests that only apply to desktop:
 #if !defined(WEBRTC_IOS) & !defined(WEBRTC_ANDROID)
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_test.cc
index ce364f60f6..00cf71114c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/hardware_test.cc
@@ -139,3 +139,89 @@ TEST_F(HardwareTest,
   EXPECT_EQ(0, voe_base_->StartSend(channel_));
   EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
 }
+
+// Flakily hangs on Windows: code.google.com/p/webrtc/issues/detail?id=2179.
+TEST_F(HardwareTest,
+       DISABLED_ON_WIN(BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer)) {
+#ifdef WEBRTC_IOS
+  // Ensure the sound device is reset on iPhone.
+  EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
+  Sleep(2000);
+#endif
+  EXPECT_EQ(0, voe_base_->StopSend(channel_));
+  EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+
+  webrtc::AudioLayers given_layer;
+  EXPECT_EQ(0, voe_hardware_->GetAudioDeviceLayer(given_layer));
+  if (given_layer != webrtc::kAudioWindowsCore) {
+    // Not Windows Audio Core - then it shouldn't work.
+    EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(true));
+    EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(false));
+    return;
+  }
+
+  TEST_LOG("Testing AEC for Audio Windows Core.\n");
+  EXPECT_EQ(0, voe_base_->StartSend(channel_));
+
+  // Can't be set after StartSend().
+  EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(true));
+  EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(false));
+
+  EXPECT_EQ(0, voe_base_->StopSend(channel_));
+  EXPECT_EQ(0, voe_hardware_->EnableBuiltInAEC(true));
+
+  // Can't be called before StartPlayout().
+  EXPECT_EQ(-1, voe_base_->StartSend(channel_));
+
+  EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+  EXPECT_EQ(0, voe_base_->StartSend(channel_));
+  TEST_LOG("Processing capture data with built-in AEC...\n");
+  Sleep(2000);
+
+  TEST_LOG("Looping through capture devices...\n");
+  int num_devs = 0;
+  char dev_name[128] = { 0 };
+  char guid_name[128] = { 0 };
+  EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(num_devs));
+  for (int dev_index = 0; dev_index < num_devs; ++dev_index) {
+    EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(dev_index,
+                                                       dev_name,
+                                                       guid_name));
+    TEST_LOG("%d: %s\n", dev_index, dev_name);
+    EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(dev_index));
+    Sleep(2000);
+  }
+
+  EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
+  EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
+
+  TEST_LOG("Looping through render devices, restarting for each "
+      "device...\n");
+  EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(num_devs));
+  for (int dev_index = 0; dev_index < num_devs; ++dev_index) {
+    EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(dev_index,
+                                                     dev_name,
+                                                     guid_name));
+    TEST_LOG("%d: %s\n", dev_index, dev_name);
+    EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(dev_index));
+    Sleep(2000);
+  }
+
+  TEST_LOG("Using default devices...\n");
+  EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
+  EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
+  Sleep(2000);
+
+  // Possible, but not recommended before StopSend().
+  EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+
+  EXPECT_EQ(0, voe_base_->StopSend(channel_));
+  EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
+  Sleep(2000);  // To verify that there is no garbage audio.
+
+  TEST_LOG("Disabling built-in AEC.\n");
+  EXPECT_EQ(0, voe_hardware_->EnableBuiltInAEC(false));
+
+  EXPECT_EQ(0, voe_base_->StartSend(channel_));
+  EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
+}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
index 2a5732b211..b7f7d560d7 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
@@ -11,7 +11,7 @@
 #include <stdio.h>
 #include <string>
 
-#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
 
@@ -35,7 +35,7 @@ class MixingTest : public AfterInitializationFixture {
       : output_filename_(test::OutputPath() + "mixing_test_output.pcm") {
   }
   void SetUp() {
-    transport_ = new LoopBackTransport(voe_network_);
+    transport_ = new LoopBackTransport(voe_network_, 0);
   }
   void TearDown() {
     delete transport_;
@@ -182,6 +182,9 @@ class MixingTest : public AfterInitializationFixture {
   void StartRemoteStream(int stream, const CodecInst& codec_inst, int port) {
     EXPECT_EQ(0, voe_codec_->SetRecPayloadType(stream, codec_inst));
     EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_));
+    EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(
+                     stream, static_cast<unsigned int>(stream)));
+    transport_->AddChannel(stream, stream);
     EXPECT_EQ(0, voe_base_->StartReceive(stream));
     EXPECT_EQ(0, voe_base_->StartPlayout(stream));
     EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
index f3be6355ba..94a2c42e15 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
@@ -55,5 +55,5 @@ TEST_F(NetEQStatsTest, ManualPrintStatisticsAfterRunningAWhile) {
       network_statistics.maxWaitingTimeMs);
 
   // This is only set to a non-zero value in off-mode.
-  EXPECT_EQ(0, network_statistics.addedSamples);
+  EXPECT_EQ(0U, network_statistics.addedSamples);
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 18f064eb77..1dc15dff49 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -8,11 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/video_engine/include/vie_network.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
 
 using ::testing::_;
@@ -29,7 +28,9 @@ class ExtensionVerifyTransport : public webrtc::Transport {
         audio_level_id_(-1),
         absolute_sender_time_id_(-1) {}
 
-  int SendPacket(int channel, const void* data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override {
     webrtc::RTPHeader header;
     if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
       bool ok = true;
@@ -49,11 +50,11 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     }
     // received_packets_ count all packets we receive.
     ++received_packets_;
-    return static_cast<int>(len);
+    return true;
   }
 
-  int SendRTCPPacket(int channel, const void* data, size_t len) override {
-    return static_cast<int>(len);
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    return true;
   }
 
   void SetAudioLevelId(int id) {
@@ -152,156 +153,3 @@ TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAllExtensions2) {
   ResumePlaying();
   EXPECT_TRUE(verifying_transport_.Wait());
 }
-
-class MockViENetwork : public webrtc::ViENetwork {
- public:
-  MockViENetwork() {}
-  virtual ~MockViENetwork() {}
-
-  MOCK_METHOD0(Release, int());
-  MOCK_METHOD4(SetBitrateConfig, void(int, int, int, int));
-  MOCK_METHOD2(SetNetworkTransmissionState, void(const int, const bool));
-  MOCK_METHOD2(RegisterSendTransport, int(const int, webrtc::Transport&));
-  MOCK_METHOD1(DeregisterSendTransport, int(const int));
-  MOCK_METHOD4(ReceivedRTPPacket, int(const int, const void*, const size_t,
-                                      const webrtc::PacketTime&));
-  MOCK_METHOD3(ReceivedRTCPPacket, int(const int, const void*, const size_t));
-  MOCK_METHOD2(SetMTU, int(int, unsigned int));
-  MOCK_METHOD4(ReceivedBWEPacket, int(const int, int64_t, size_t,
-                                      const webrtc::RTPHeader&));
-};
-
-class ReceiveRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture {
- protected:
-  void SetUp() override {
-    EXPECT_EQ(0,
-        voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 11));
-    EXPECT_EQ(0,
-        voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_, true, 11));
-  }
-
-  void Wait() {
-    WaitForTransmittedPackets(kPacketsExpected);
-  }
-
-  enum {
-    kPacketsExpected = 5,
-    kVideoChannelId1 = 667,
-    kVideoChannelId2 = 668
-  };
-  MockViENetwork mock_network_;
-};
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTDisabled) {
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(_, _, _, _)).Times(0);
-  ResumePlaying();
-  Wait();
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTFailSetTarget) {
-  EXPECT_CALL(mock_network_, Release()).Times(1);
-  EXPECT_EQ(-1, voe_rtp_rtcp_->SetVideoEngineBWETarget(-1, &mock_network_,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabled) {
-  EXPECT_CALL(mock_network_, Release()).Times(1);
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true)))))
-      .Times(AtLeast(1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledBadExtensionId) {
-  EXPECT_CALL(mock_network_, Release()).Times(1);
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(false)))))
-      .Times(AtLeast(1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_, true,
-                                                                 1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledNotSending) {
-  EXPECT_CALL(mock_network_, Release()).Times(1);
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(false)))))
-      .Times(AtLeast(1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, false,
-                                                              11));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledNotReceiving) {
-  EXPECT_CALL(mock_network_, Release()).Times(1);
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(false)))))
-      .Times(AtLeast(1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_,
-                                                                 false, 11));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTSwitchViENetwork) {
-  MockViENetwork mock_network_2;
-  EXPECT_CALL(mock_network_2, Release()).Times(1);
-  EXPECT_CALL(mock_network_2, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true)))))
-      .Times(AtLeast(1));
-  EXPECT_CALL(mock_network_, Release()).Times(1);
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true)))))
-      .Times(AtLeast(1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_2,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId1));
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
-}
-
-TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTSwitchVideoChannel) {
-  EXPECT_CALL(mock_network_, Release()).Times(2);
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId1, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true)))))
-      .Times(AtLeast(1));
-  EXPECT_CALL(mock_network_, ReceivedBWEPacket(kVideoChannelId2, _, _,
-      Field(&webrtc::RTPHeader::extension,
-      Field(&webrtc::RTPHeaderExtension::hasAbsoluteSendTime, Eq(true)))))
-      .Times(AtLeast(1));
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId1));
-  ResumePlaying();
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
-                                                      kVideoChannelId2));
-  Wait();
-  EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
-}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index e5b2f307de..6efa55d516 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/atomic32.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
@@ -63,7 +63,7 @@ class RtpRtcpTest : public AfterStreamingFixture {
     second_channel_ = voe_base_->CreateChannel();
     EXPECT_GE(second_channel_, 0);
 
-    transport_ = new LoopBackTransport(voe_network_);
+    transport_ = new LoopBackTransport(voe_network_, second_channel_);
     EXPECT_EQ(0, voe_network_->RegisterExternalTransport(second_channel_,
                                                          *transport_));
 
@@ -101,8 +101,7 @@ TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
   EXPECT_STREQ(RTCP_CNAME, char_buffer);
 }
 
-// Flakily hangs on Linux. code.google.com/p/webrtc/issues/detail?id=2178.
-TEST_F(RtpRtcpTest, DISABLED_ON_LINUX(SSRCPropagatesCorrectly)) {
+TEST_F(RtpRtcpTest, SSRCPropagatesCorrectly) {
   unsigned int local_ssrc = 1234;
   EXPECT_EQ(0, voe_base_->StopSend(channel_));
   EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, local_ssrc));
@@ -117,23 +116,3 @@ TEST_F(RtpRtcpTest, DISABLED_ON_LINUX(SSRCPropagatesCorrectly)) {
   EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc));
   EXPECT_EQ(local_ssrc, ssrc);
 }
-
-// TODO(xians, phoglund): Re-enable when issue 372 is resolved.
-TEST_F(RtpRtcpTest, DISABLED_CanCreateRtpDumpFilesWithoutError) {
-  // Create two RTP dump files (3 seconds long). You can verify these after
-  // the test using rtpplay or NetEqRTPplay if you like.
-  std::string output_path = webrtc::test::OutputPath();
-  std::string incoming_filename = output_path + "dump_in_3sec.rtp";
-  std::string outgoing_filename = output_path + "dump_out_3sec.rtp";
-
-  EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
-      channel_, incoming_filename.c_str(), webrtc::kRtpIncoming));
-  EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
-      channel_, outgoing_filename.c_str(), webrtc::kRtpOutgoing));
-
-  Sleep(3000);
-
-  EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpIncoming));
-  EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpOutgoing));
-}
-
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_conference_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
new file mode 100644
index 0000000000..c70d92a946
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <queue>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
+
+namespace {
+const int kRttMs = 25;
+
+bool IsNear(int ref, int comp, int error) {
+  return (ref - comp <= error) && (comp - ref >= -error);
+}
+
+void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
+  FILE* fid = fopen(silence_file.c_str(), "wb");
+  int16_t zero = 0;
+  for (int i = 0; i < sample_rate_hz; ++i) {
+    // Write 1 second, but it does not matter since the file will be looped.
+    fwrite(&zero, sizeof(int16_t), 1, fid);
+  }
+  fclose(fid);
+}
+
+}  // namespace
+
+namespace voetest {
+
+TEST(VoeConferenceTest, RttAndStartNtpTime) {
+  struct Stats {
+    Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
+        : rtt_receiver_1_(rtt_receiver_1),
+          rtt_receiver_2_(rtt_receiver_2),
+          ntp_delay_(ntp_delay) {
+    }
+    int64_t rtt_receiver_1_;
+    int64_t rtt_receiver_2_;
+    int64_t ntp_delay_;
+  };
+
+  const std::string input_file =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
+  const int kDelayMs = 987;
+  ConferenceTransport trans;
+  trans.SetRtt(kRttMs);
+
+  unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
+  unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
+
+  EXPECT_TRUE(trans.StartPlayout(id_1));
+  // Start NTP time is the time when a stream is played out, rather than
+  // when it is added.
+  webrtc::SleepMs(kDelayMs);
+  EXPECT_TRUE(trans.StartPlayout(id_2));
+
+  const int kMaxRunTimeMs = 25000;
+  const int kNeedSuccessivePass = 3;
+  const int kStatsRequestIntervalMs = 1000;
+  const int kStatsBufferSize = 3;
+
+  uint32_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
+  // Run the following up to |kMaxRunTimeMs| milliseconds.
+  int successive_pass = 0;
+  webrtc::CallStatistics stats_1;
+  webrtc::CallStatistics stats_2;
+  std::queue<Stats> stats_buffer;
+
+  while (rtc::TimeIsLater(rtc::Time(), deadline) &&
+      successive_pass < kNeedSuccessivePass) {
+    webrtc::SleepMs(kStatsRequestIntervalMs);
+
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
+
+    // It is not easy to verify the NTP time directly. We verify it by testing
+    // the difference of two start NTP times.
+    int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
+        stats_1.capture_start_ntp_time_ms_;
+
+    // For the checks of RTT and start NTP time, We allow 10% accuracy.
+    if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
+        IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
+        IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
+      successive_pass++;
+    } else {
+      successive_pass = 0;
+    }
+    if (stats_buffer.size() >= kStatsBufferSize) {
+      stats_buffer.pop();
+    }
+    stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
+                            captured_start_ntp_delay));
+  }
+
+  EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
+      " start NTP time estimate within 10% of the correct value over "
+      << kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
+      << " seconds.";
+  if (successive_pass < kNeedSuccessivePass) {
+    printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
+        "NTP delay between receiver 1 and 2) are (from oldest):\n");
+    while (!stats_buffer.empty()) {
+      Stats stats = stats_buffer.front();
+      printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
+             stats.rtt_receiver_2_, stats.ntp_delay_);
+      stats_buffer.pop();
+    }
+  }
+}
+
+
+TEST(VoeConferenceTest, ReceivedPackets) {
+  const int kPackets = 50;
+  const int kPacketDurationMs = 20;  // Correspond to Opus.
+
+  const std::string input_file =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
+  const std::string silence_file =
+      webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
+  CreateSilenceFile(silence_file, 32000);
+
+  {
+    ConferenceTransport trans;
+    // Add silence to stream 0, so that it will be filtered out.
+    unsigned int id_0 = trans.AddStream(silence_file, kInputFormat);
+    unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
+    unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
+    unsigned int id_3 = trans.AddStream(input_file, kInputFormat);
+
+    EXPECT_TRUE(trans.StartPlayout(id_0));
+    EXPECT_TRUE(trans.StartPlayout(id_1));
+    EXPECT_TRUE(trans.StartPlayout(id_2));
+    EXPECT_TRUE(trans.StartPlayout(id_3));
+
+    webrtc::SleepMs(kPacketDurationMs * kPackets);
+
+    webrtc::CallStatistics stats_0;
+    webrtc::CallStatistics stats_1;
+    webrtc::CallStatistics stats_2;
+    webrtc::CallStatistics stats_3;
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3));
+
+    // We expect stream 0 to be filtered out totally, but since it may join the
+    // call earlier than other streams and the beginning packets might have got
+    // through. So we only expect |packetsReceived| to be close to zero.
+    EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
+    // We expect |packetsReceived| to match |kPackets|, but the actual value
+    // depends on the sleep timer. So we allow a small off from |kPackets|.
+    EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
+    EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
+    EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
+  }
+
+  remove(silence_file.c_str());
+}
+
+}  // namespace voetest
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
index ad6116dcdd..5666b3f8d1 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
@@ -18,7 +18,7 @@
 #endif
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/test/auto_test/voe_test_defines.h"
 
 using namespace webrtc;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_output_test.cc
new file mode 100644
index 0000000000..3bedbc3b17
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_output_test.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
+
+namespace {
+
+const char kIp[] = "127.0.0.1";
+const int kPort = 1234;
+const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
+
+}  // namespace
+
+namespace voetest {
+
+using webrtc::Random;
+using webrtc::test::VoiceChannelTransport;
+
+// This test allows a check on the output signal in an end-to-end call.
+class OutputTest {
+ public:
+  OutputTest(int16_t lower_bound, int16_t upper_bound);
+  ~OutputTest();
+
+  void Start();
+
+  void EnableOutputCheck();
+  void DisableOutputCheck();
+  void SetOutputBound(int16_t lower_bound, int16_t upper_bound);
+  void Mute();
+  void Unmute();
+  void SetBitRate(int rate);
+
+ private:
+  // This class checks all output values and count the number of samples that
+  // go out of a defined range.
+  class VoEOutputCheckMediaProcess : public VoEMediaProcess {
+   public:
+    VoEOutputCheckMediaProcess(int16_t lower_bound, int16_t upper_bound);
+
+    void set_enabled(bool enabled) { enabled_ = enabled; }
+    void Process(int channel,
+                 ProcessingTypes type,
+                 int16_t audio10ms[],
+                 size_t length,
+                 int samplingFreq,
+                 bool isStereo) override;
+
+   private:
+    bool enabled_;
+    int16_t lower_bound_;
+    int16_t upper_bound_;
+  };
+
+  VoETestManager manager_;
+  VoEOutputCheckMediaProcess output_checker_;
+
+  int channel_;
+};
+
+OutputTest::OutputTest(int16_t lower_bound, int16_t upper_bound)
+    : output_checker_(lower_bound, upper_bound) {
+  EXPECT_TRUE(manager_.Init());
+  manager_.GetInterfaces();
+
+  VoEBase* base = manager_.BasePtr();
+  VoECodec* codec = manager_.CodecPtr();
+  VoENetwork* network = manager_.NetworkPtr();
+
+  EXPECT_EQ(0, base->Init());
+
+  channel_ = base->CreateChannel();
+
+  // |network| will take care of the life time of |transport|.
+  VoiceChannelTransport* transport =
+      new VoiceChannelTransport(network, channel_);
+
+  EXPECT_EQ(0, transport->SetSendDestination(kIp, kPort));
+  EXPECT_EQ(0, transport->SetLocalReceiver(kPort));
+
+  EXPECT_EQ(0, codec->SetSendCodec(channel_, kCodecInst));
+  EXPECT_EQ(0, codec->SetOpusDtx(channel_, true));
+
+  EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255));
+
+  manager_.ExternalMediaPtr()->RegisterExternalMediaProcessing(
+      channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_);
+}
+
+OutputTest::~OutputTest() {
+  EXPECT_EQ(0, manager_.NetworkPtr()->DeRegisterExternalTransport(channel_));
+  EXPECT_EQ(0, manager_.ReleaseInterfaces());
+}
+
+void OutputTest::Start() {
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
+  ASSERT_EQ(0, manager_.FilePtr()->StartPlayingFileAsMicrophone(
+      channel_, file_name.c_str(), true, false, kInputFormat, 1.0));
+
+  VoEBase* base = manager_.BasePtr();
+  ASSERT_EQ(0, base->StartPlayout(channel_));
+  ASSERT_EQ(0, base->StartSend(channel_));
+}
+
+void OutputTest::EnableOutputCheck() {
+  output_checker_.set_enabled(true);
+}
+
+void OutputTest::DisableOutputCheck() {
+  output_checker_.set_enabled(false);
+}
+
+void OutputTest::Mute() {
+  manager_.VolumeControlPtr()->SetInputMute(channel_, true);
+}
+
+void OutputTest::Unmute() {
+  manager_.VolumeControlPtr()->SetInputMute(channel_, false);
+}
+
+void OutputTest::SetBitRate(int rate) {
+  manager_.CodecPtr()->SetBitRate(channel_, rate);
+}
+
+OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess(
+    int16_t lower_bound, int16_t upper_bound)
+    : enabled_(false),
+      lower_bound_(lower_bound),
+      upper_bound_(upper_bound) {}
+
+void OutputTest::VoEOutputCheckMediaProcess::Process(int channel,
+                                                     ProcessingTypes type,
+                                                     int16_t* audio10ms,
+                                                     size_t length,
+                                                     int samplingFreq,
+                                                     bool isStereo) {
+  if (!enabled_)
+    return;
+  const int num_channels = isStereo ? 2 : 1;
+  for (size_t i = 0; i < length; ++i) {
+    for (int c = 0; c < num_channels; ++c) {
+      ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_);
+      ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_);
+    }
+  }
+}
+
+// This test checks if the Opus does not produce high noise (noise pump) when
+// DTX is enabled. The microphone is toggled on and off, and values of the
+// output signal during muting should be bounded.
+// We do not run this test on bots. Developers that want to see the result
+// and/or listen to sound quality can run this test manually.
+TEST(OutputTest, DISABLED_OpusDtxHasNoNoisePump) {
+  const int kRuntimeMs = 20000;
+  const uint32_t kUnmuteTimeMs = 1000;
+  const int kCheckAfterMute = 2000;
+  const uint32_t kCheckTimeMs = 2000;
+  const int kMinOpusRate = 6000;
+  const int kMaxOpusRate = 64000;
+
+#if defined(OPUS_FIXED_POINT)
+  const int16_t kDtxBoundForSilence = 20;
+#else
+  const int16_t kDtxBoundForSilence = 2;
+#endif
+
+  OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence);
+  Random random(1234ull);
+
+  uint32_t start_time = rtc::Time();
+  test.Start();
+  while (rtc::TimeSince(start_time) < kRuntimeMs) {
+    webrtc::SleepMs(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10,
+                                kUnmuteTimeMs + kUnmuteTimeMs / 10));
+    test.Mute();
+    webrtc::SleepMs(kCheckAfterMute);
+    test.EnableOutputCheck();
+    webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10,
+                                kCheckTimeMs + kCheckTimeMs / 10));
+    test.DisableOutputCheck();
+    test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
+    test.Unmute();
+  }
+}
+
+}  // namespace voetest
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.cc
index d3fd478a52..a187c4bb6c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.cc
@@ -15,7 +15,7 @@
 #include <string.h>
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/test/auto_test/automated_mode.h"
 #include "webrtc/voice_engine/test/auto_test/voe_cpu_test.h"
@@ -117,14 +117,6 @@ bool VoETestManager::Init() {
   if (initialized_)
     return true;
 
-  if (VoiceEngine::SetTraceFile(NULL) != -1) {
-    // should not be possible to call a Trace method before the VoE is
-    // created
-    TEST_LOG("\nError at line: %i (VoiceEngine::SetTraceFile()"
-      "should fail)!\n", __LINE__);
-    return false;
-  }
-
   voice_engine_ = VoiceEngine::Create();
   if (!voice_engine_) {
     TEST_LOG("Failed to create VoiceEngine\n");
@@ -230,11 +222,6 @@ int VoETestManager::ReleaseInterfaces() {
     releaseOK = false;
   }
 
-  if (VoiceEngine::SetTraceFile(NULL) != -1) {
-    TEST_LOG("\nError at line: %i (VoiceEngine::SetTraceFile()"
-      "should fail)!\n", __LINE__);
-  }
-
   return (releaseOK == true) ? 0 : -1;
 }
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.h
index 3bf89362d5..b92595982c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_standard_test.h
@@ -44,7 +44,6 @@
 #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
 namespace webrtc {
 class CriticalSectionWrapper;
-class ThreadWrapper;
 class VoENetEqStats;
 }
 #endif
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc
index ab8fffe832..259eff0ccc 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc
@@ -25,8 +25,8 @@
 #include "webrtc/voice_engine/test/auto_test/voe_stress_test.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 #include "webrtc/voice_engine/test/auto_test/voe_test_defines.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"  // defines build macros
@@ -334,9 +334,9 @@ int VoEStressTest::MultipleThreadsTest() {
   int rnd(0);
 
   // Start extra thread
-  _ptrExtraApiThread = ThreadWrapper::CreateThread(RunExtraApi, this,
-                                                   "StressTestExtraApiThread");
-  VALIDATE_STRESS(!_ptrExtraApiThread->Start());
+  _ptrExtraApiThread.reset(
+      new rtc::PlatformThread(RunExtraApi, this, "StressTestExtraApiThread"));
+  _ptrExtraApiThread->Start();
 
   //       Some possible extensions include:
   //       Add more API calls to randomize
@@ -365,7 +365,7 @@ int VoEStressTest::MultipleThreadsTest() {
   ANL();
 
   // Stop extra thread
-  VALIDATE_STRESS(!_ptrExtraApiThread->Stop());
+  _ptrExtraApiThread->Stop();
 
   ///////////// End test /////////////
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h
index 7128b238ff..715e8ef724 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h
@@ -11,11 +11,10 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
 #define WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
 
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace voetest {
-// TODO(andrew): using directives are not permitted.
-using namespace webrtc;
 
 class VoETestManager;
 
@@ -38,7 +37,8 @@ class VoEStressTest {
 
   VoETestManager& _mgr;
 
-  rtc::scoped_ptr<ThreadWrapper> _ptrExtraApiThread;
+  // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
+  rtc::scoped_ptr<rtc::PlatformThread> _ptrExtraApiThread;
 };
 
 }  // namespace voetest
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
index 419ba55d78..ccfe3c2bde 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -19,10 +19,12 @@
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
@@ -112,8 +114,8 @@ void PrintCodecs(bool opus_stereo) {
     int res = codec->GetCodec(i, codec_params);
     VALIDATE;
     SetStereoIfOpus(opus_stereo, &codec_params);
-    printf("%2d. %3d  %s/%d/%d \n", i, codec_params.pltype, codec_params.plname,
-           codec_params.plfreq, codec_params.channels);
+    printf("%2d. %3d  %s/%d/%" PRIuS " \n", i, codec_params.pltype,
+           codec_params.plname, codec_params.plfreq, codec_params.channels);
   }
 }
 
@@ -260,7 +262,7 @@ void RunTest(std::string out_path) {
     fflush(NULL);
   }
 
-  rtc::scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+  VoiceChannelTransport* voice_channel_transport(
       new VoiceChannelTransport(netw, chan));
 
   char ip[64];
@@ -451,7 +453,8 @@ void RunTest(std::string out_path) {
       printf("%i. Toggle Opus DTX \n", option_index++);
       printf("%i. Set bit rate (only take effect on codecs that allow the "
              "change) \n", option_index++);
-      printf("%i. Toggle debug recording \n", option_index++);
+      printf("%i. Toggle AECdump recording \n", option_index++);
+      printf("%i. Record RtcEventLog file of 30 seconds \n", option_index++);
 
       printf("Select action or %i to stop the call: ", option_index);
       int option_selection;
@@ -784,8 +787,9 @@ void RunTest(std::string out_path) {
         res = codec->GetSendCodec(chan, cinst);
         VALIDATE;
         printf("Current bit rate is %i bps, set to: ", cinst.rate);
-        ASSERT_EQ(1, scanf("%i", &cinst.rate));
-        res = codec->SetSendCodec(chan, cinst);
+        int new_bitrate_bps;
+        ASSERT_EQ(1, scanf("%i", &new_bitrate_bps));
+        res = codec->SetBitRate(chan, new_bitrate_bps);
         VALIDATE;
       } else if (option_selection == option_index++) {
         const char* kDebugFileName = "audio.aecdump";
@@ -797,6 +801,9 @@ void RunTest(std::string out_path) {
           printf("Debug recording named %s started\n", kDebugFileName);
         }
         debug_recording_started = !debug_recording_started;
+      } else if (option_selection == option_index++) {
+        const char* kDebugFileName = "eventlog.rel";
+        codec->GetEventLog()->StartLogging(kDebugFileName, 30000);
       } else {
         break;
       }
@@ -844,6 +851,9 @@ void RunTest(std::string out_path) {
     newcall = (end_option == 1);
     // Call loop
   }
+
+  // Transports should be deleted before channel deletion.
+  delete voice_channel_transport;
   for (int i = 0; i < kMaxNumChannels; ++i) {
     delete voice_channel_transports[i];
     voice_channel_transports[i] = NULL;
@@ -854,7 +864,7 @@ void RunTest(std::string out_path) {
   VALIDATE;
 
   for (int i = 0; i < kMaxNumChannels; ++i) {
-    channels[i] = base1->DeleteChannel(channels[i]);
+    res = base1->DeleteChannel(channels[i]);
     VALIDATE;
   }
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/Resource.h b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/Resource.h
deleted file mode 100644
index cd5f55a53c..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/Resource.h
+++ /dev/null
@@ -1,241 +0,0 @@
-//{{NO_DEPENDENCIES}}
-// Microsoft Visual C++ generated include file.
-// Used by WinTest.rc
-//
-#define IDM_ABOUTBOX                    0x0010
-#define IDD_ABOUTBOX                    100
-#define IDS_ABOUTBOX                    101
-#define IDD_WINTEST_DIALOG              102
-#define IDR_MAINFRAME                   128
-#define IDD_DTMF_DIALOG                 129
-#define IDC_BUTTON_CREATE_1             1000
-#define IDC_BUTTON_DELETE_1             1001
-#define IDC_EDIT_1                      1002
-#define IDC_BUTTON_CREATE_2             1003
-#define IDC_BUTTON_DELETE_2             1004
-#define IDC_EDIT_2                      1005
-#define IDC_EDIT_MESSAGE                1006
-#define IDC_BUTTON_START_LISTEN_1       1007
-#define IDC_COMBO_IP_1                  1008
-#define IDC_EDIT_TX_PORT_1              1009
-#define IDC_EDIT_RX_PORT_1              1010
-#define IDC_COMBO_CODEC_1               1011
-#define IDC_BUTTON_STOP_LISTEN_1        1012
-#define IDC_STATIC_LISTEN               1013
-#define IDC_BUTTON_START_PLAYOUT_1      1014
-#define IDC_BUTTON_STOP_PLAYOUT_1       1015
-#define IDC_STATIC_PLAY                 1016
-#define IDC_BUTTON_START_SEND_1         1017
-#define IDC_BUTTON_STOP_SEND_1          1018
-#define IDC_STATIC_SEND                 1019
-#define IDC_COMBO_IP_2                  1020
-#define IDC_STATIC_IP                   1021
-#define IDC_STATIC_PORTS                1022
-#define IDC_STATIC_CODEC                1023
-#define IDC_STATIC_CHANNEL              1024
-#define IDC_STATIC_ID                   1025
-#define IDC_EDIT_TX_PORT_2              1026
-#define IDC_EDIT_RX_PORT_2              1027
-#define IDC_COMBO_CODEC_2               1028
-#define IDC_BUTTON_START_LISTEN_2       1029
-#define IDC_BUTTON_STOP_LISTEN_2        1030
-#define IDC_BUTTON_START_PLAYOUT_2      1031
-#define IDC_BUTTON_STOP_PLAYOUT_2       1032
-#define IDC_BUTTON_START_SEND_2         1033
-#define IDC_BUTTON_STOP_SEND_2          1034
-#define IDC_BUTTON_START_SEND_3         1035
-#define IDC_BUTTON_TEST_1_1             1035
-#define IDC_BUTTON_TEST_1               1035
-#define IDC_EDIT_RESULT                 1036
-#define IDC_EDIT_N_FAILS                1037
-#define IDC_STATIC_ERROR                1038
-#define IDC_EDIT_LAST_ERROR             1039
-#define IDC_STATIC_LAST_ERROR           1040
-#define IDC_STATIC_PLAY_FILE            1041
-#define IDC_STATIC_EXTERNAL             1042
-#define IDC_CHECK_EXT_TRANS_1           1043
-#define IDC_CHECK2                      1044
-#define IDC_CHECK_PLAY_FILE_IN_1        1044
-#define IDC_CHECK_PLAY_FILE_OUT_1       1045
-#define IDC_CHECK_PLAY_FILE_IN_2        1046
-#define IDC_CHECK_PLAY_FILE_OUT_2       1047
-#define IDC_CHECK_EXT_TRANS_2           1048
-#define IDC_STATIC_ALL_CHANNELS         1049
-#define IDC_CHECK_PLAY_FILE_IN          1050
-#define IDC_CHECK_PLAY_FILE_OUT         1051
-#define IDC_CHECK_EXT_MEDIA_IN_1        1051
-#define IDC_COMBO_REC_DEVICE            1052
-#define IDC_STATIC_REC_DEVICE           1053
-#define IDC_COMBO_PLAY_DEVICE2          1054
-#define IDC_COMBO_PLAY_DEVICE           1054
-#define IDC_STATIC_PLAY_DEVICE          1055
-#define IDC_CHECK_EXT_MEDIA_PLAY_1      1056
-#define IDC_CHECK_EXT_MEDIA_OUT_1       1056
-#define IDC_STATIC_PLAY_FILE2           1057
-#define IDC_SLIDER_INPUT_VOLUME         1058
-#define IDC_STATIC_MIC_VOLUME           1059
-#define IDC_SLIDER_OUTPUT_VOLUME        1060
-#define IDC_STATIC_SPK_VOLUME2          1061
-#define IDC_STATIC_SPK_VOLUME           1061
-#define IDC_CHECK_PLAY_FILE_IN2         1062
-#define IDC_CHECK_AGC                   1062
-#define IDC_STATIC_MIC_VOLUME2          1063
-#define IDC_STATIC_AUDIO_LEVEL_IN       1063
-#define IDC_PROGRESS_AUDIO_LEVEL_IN     1064
-#define IDC_CHECK_AGC2                  1065
-#define IDC_CHECK_NS                    1065
-#define IDC_BUTTON_1                    1065
-#define IDC_CHECK_VAD                   1066
-#define IDC_CHECK_EXT_MEDIA_IN_2        1066
-#define IDC_BUTTON_2                    1066
-#define IDC_CHECK_VAD2                  1067
-#define IDC_CHECK_EC                    1067
-#define IDC_BUTTON_3                    1067
-#define IDC_CHECK_VAD_1                 1068
-#define IDC_BUTTON_4                    1068
-#define IDC_CHECK_VAD_2                 1069
-#define IDC_CHECK_EXT_MEDIA_OUT_2       1069
-#define IDC_BUTTON_5                    1069
-#define IDC_CHECK_VAD_3                 1070
-#define IDC_BUTTON_6                    1070
-#define IDC_CHECK_MUTE_IN               1071
-#define IDC_BUTTON_7                    1071
-#define IDC_CHECK_MUTE_IN_1             1072
-#define IDC_BUTTON_8                    1072
-#define IDC_CHECK_MUTE_IN_2             1073
-#define IDC_BUTTON_9                    1073
-#define IDC_CHECK_SRTP_TX_1             1074
-#define IDC_BUTTON_10                   1074
-#define IDC_CHECK_SRTP_RX_1             1075
-#define IDC_BUTTON_11                   1075
-#define IDC_STATIC_PLAY_FILE3           1076
-#define IDC_STATIC_SRTP                 1076
-#define IDC_BUTTON_12                   1076
-#define IDC_CHECK_SRTP_TX_2             1077
-#define IDC_BUTTON_13                   1077
-#define IDC_CHECK_SRTP_RX_2             1078
-#define IDC_BUTTON_14                   1078
-#define IDC_CHECK_EXT_ENCRYPTION_1      1079
-#define IDC_BUTTON_15                   1079
-#define IDC_STATIC_PLAY_FILE4           1080
-#define IDC_BUTTON_16                   1080
-#define IDC_CHECK_EXT_ENCRYPTION_2      1081
-#define IDC_BUTTON_17                   1081
-#define IDC_BUTTON_DTMF_1               1082
-#define IDC_BUTTON_18                   1082
-#define IDC_EDIT_DTMF_EVENT             1083
-#define IDC_CHECK_REC_                  1083
-#define IDC_CHECK_REC_MIC               1083
-#define IDC_STATIC_DTMF_EVENT           1084
-#define IDC_BUTTON_DTMF_2               1084
-#define IDC_STATIC_GROUP_DTMF           1085
-#define IDC_CHECK_CONFERENCE_1          1085
-#define IDC_BUTTON_19                   1086
-#define IDC_CHECK_CONFERENCE_2          1086
-#define IDC_BUTTON_20                   1087
-#define IDC_CHECK_ON_HOLD_1             1087
-#define IDC_BUTTON_21                   1088
-#define IDC_CHECK_ON_HOLD_2             1088
-#define IDC_BUTTON_22                   1089
-#define IDC_CHECK_DTMF_PLAYOUT_RX       1089
-#define IDC_CHECK_EXT_MEDIA_IN          1089
-#define IDC_STATIC_PLAYOUT_RX           1090
-#define IDC_EDIT_GET_OUTPUT             1090
-#define IDC_CHECK_DTMF_PLAY_TONE        1091
-#define IDC_STATIC_LAST_ERROR2          1091
-#define IDC_STATIC_GET                  1091
-#define IDC_STATIC_PLAY_TONE            1092
-#define IDC_CHECK_EXT_MEDIA_OUT         1092
-#define IDC_CHECK_START_STOP_MODE       1093
-#define IDC_BUTTON_SET_TX_TELEPHONE_PT  1093
-#define IDC_PROGRESS_AUDIO_LEVEL_IN2    1093
-#define IDC_PROGRESS_AUDIO_LEVEL_OUT    1093
-#define IDC_EDIT_EVENT_LENGTH           1094
-#define IDC_EDIT_RX_PORT_3              1094
-#define IDC_EDIT_DELAY_ESTIMATE_1       1094
-#define IDC_STATIC_EVENT_LENGTH         1095
-#define IDC_EDIT_PLAYOUT_BUFFER_SIZE    1095
-#define IDC_STATIC_START_STOP_MODE      1096
-#define IDC_EDIT_EVENT_RX_PT            1096
-#define IDC_CHECK_DELAY_ESTIMATE_1      1096
-#define IDC_EDIT_EVENT_ATTENUATION      1097
-#define IDC_CHECK_AGC_1                 1097
-#define IDC_CHECK_EVENT_INBAND          1098
-#define IDC_CHECK_NS_1                  1098
-#define IDC_STATIC_EVENT_ATTENUATION    1099
-#define IDC_STATIC_SRTP2                1099
-#define IDC_STATIC_RX_VQE               1099
-#define IDC_EDIT_EVENT_TX_PT            1100
-#define IDC_CHECK_REC_MIC2              1100
-#define IDC_CHECK_REC_CALL              1100
-#define IDC_CHECK_DTMF_FEEDBACK         1101
-#define IDC_CHECK_REC_CALL2             1101
-#define IDC_CHECK_TYPING_DETECTION      1101
-#define IDC_CHECK_START_STOP_MODE2      1102
-#define IDC_CHECK_DIRECT_FEEDBACK       1102
-#define IDC_CHECK_RED                   1102
-#define IDC_BUTTON_SET_RX_TELEPHONE_PT_TYPE 1103
-#define IDC_BUTTON_SET_RX_TELEPHONE_PT  1103
-#define IDC_BUTTON_CLEAR_ERROR_CALLBACK 1103
-#define IDC_EDIT_EVENT_CODE             1104
-#define IDC_STATIC_DIRECT_FEEDBACK      1105
-#define IDC_RADIO_SINGLE                1106
-#define IDC_RADIO_MULTI                 1107
-#define IDC_RADIO_START_STOP            1108
-#define IDC_STATIC_MODE                 1109
-#define IDC_STATIC_EVENT_RX_PT          1110
-#define IDC_STATIC_EVENT_TX_PT          1111
-#define IDC_STATIC_PT                   1112
-#define IDC_BUTTON_SEND_TELEPHONE_EVENT 1113
-#define IDC_STATIC_EVENT_CODE           1114
-#define IDC_CHECK_EVENT_DETECTION       1115
-#define IDC_CHECK_DETECT_INBAND         1116
-#define IDC_CHECK_DETECT_OUT_OF_BAND    1117
-#define IDC_STATIC_INBAND_DETECTION     1118
-#define IDC_STATIC_OUT_OF_BAND_DETECTION 1119
-#define IDC_STATIC_EVENT_DETECTION      1120
-#define IDC_STATIC_TELEPHONE_EVENTS     1121
-#define IDC_EDIT_EVENT_CODE2            1122
-#define IDC_EDIT_ON_EVENT               1122
-#define IDC_EDIT_ON_EVENT_OUT_OF_BAND   1122
-#define IDC_STATIC_ON_EVENT             1123
-#define IDC_EDIT_ON_EVENT_INBAND        1123
-#define IDC_STATIC_EVEN                 1124
-#define IDC_STATIC_LINE                 1125
-#define IDC_LIST_CODEC_1                1128
-#define IDC_EDIT2                       1129
-#define IDC_EDIT_CODEC_1                1129
-#define IDC_STATIC_PANNING              1131
-#define IDC_SLIDER_PAN_LEFT             1132
-#define IDC_SLIDER_PAN_RIGHT            1133
-#define IDC_STATIC_LEFT                 1134
-#define IDC_STATIC_LEFT2                1135
-#define IDC_STATIC_RIGHT                1135
-#define IDC_BUTTON_VERSION              1136
-#define IDC_STATIC_PLAYOUT_BUFFER       1137
-#define IDC_CHECK_RXVAD                 1138
-#define IDC_EDIT1                       1139
-#define IDC_EDIT_RXVAD                  1139
-#define IDC_STATIC_RX_PORT              1140
-#define IDC_STATIC_RX_PORT2             1141
-#define IDC_EDIT3                       1142
-#define IDC_EDIT_AUDIO_LAYER            1142
-#define IDC_EDIT_AUDIO_LAYER2           1143
-#define IDC_EDIT_CPU_LOAD               1143
-#define IDC_STATIC_ERROR_CALLBACK       1144
-#define IDC_EDIT_ERROR_CALLBACK         1145
-#define IDC_EDIT_RX_CODEC_1             1146
-#define IDC_STATIC_BYTES_SENT_TEXT      1147
-#define IDC_EDIT_RTCP_STAT              1147
-#define IDC_EDIT_RTCP_STAT_1            1147
-
-// Next default values for new objects
-// 
-#ifdef APSTUDIO_INVOKED
-#ifndef APSTUDIO_READONLY_SYMBOLS
-#define _APS_NEXT_RESOURCE_VALUE        130
-#define _APS_NEXT_COMMAND_VALUE         32771
-#define _APS_NEXT_CONTROL_VALUE         1148
-#define _APS_NEXT_SYMED_VALUE           101
-#endif
-#endif
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.aps b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.aps
deleted file mode 100644
index 499db5f66d..0000000000
Binary files a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.aps and /dev/null differ
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.cc b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.cc
deleted file mode 100644
index 6b28ba47a4..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/voice_engine/test/win_test/WinTest.h"
-#include "webrtc/voice_engine/test/win_test/WinTestDlg.h"
-#include "webrtc/voice_engine/test/win_test/stdafx.h"
-
-#ifdef _DEBUG
-#define new DEBUG_NEW
-#endif
-
-
-// CWinTestApp
-
-BEGIN_MESSAGE_MAP(CWinTestApp, CWinApp)
-	ON_COMMAND(ID_HELP, &CWinApp::OnHelp)
-END_MESSAGE_MAP()
-
-
-// CWinTestApp construction
-
-CWinTestApp::CWinTestApp()
-{
-}
-
-
-// The one and only CWinTestApp object
-
-CWinTestApp theApp;
-
-
-// CWinTestApp initialization
-
-BOOL CWinTestApp::InitInstance()
-{
-	// InitCommonControlsEx() is required on Windows XP if an application
-	// manifest specifies use of ComCtl32.dll version 6 or later to enable
-	// visual styles.  Otherwise, any window creation will fail.
-	INITCOMMONCONTROLSEX InitCtrls;
-	InitCtrls.dwSize = sizeof(InitCtrls);
-	// Set this to include all the common control classes you want to use
-	// in your application.
-	InitCtrls.dwICC = ICC_WIN95_CLASSES;
-	InitCommonControlsEx(&InitCtrls);
-
-	CWinApp::InitInstance();
-
-	// Standard initialization
-	// If you are not using these features and wish to reduce the size
-	// of your final executable, you should remove from the following
-	// the specific initialization routines you do not need
-	// Change the registry key under which our settings are stored
-	SetRegistryKey(_T("Local AppWizard-Generated Applications"));
-
-	CWinTestDlg dlg;
-	m_pMainWnd = &dlg;
-	INT_PTR nResponse = dlg.DoModal();
-	if (nResponse == IDOK)
-	{
-	}
-	else if (nResponse == IDCANCEL)
-	{
-	}
-
-	// Since the dialog has been closed, return FALSE so that we exit the
-	//  application, rather than start the application's message pump.
-	return FALSE;
-}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.h b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.h
deleted file mode 100644
index 7320360e4b..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_WINTEST_H_
-#define WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_WINTEST_H_
-
-#ifndef __AFXWIN_H__
-	#error "include 'stdafx.h' before including this file for PCH"
-#endif
-
-#include "resource.h"		// main symbols
-
-
-// CWinTestApp:
-// See WinTest.cpp for the implementation of this class
-//
-
-class CWinTestApp : public CWinApp
-{
-public:
-	CWinTestApp();
-
-// Overrides
-	public:
-	virtual BOOL InitInstance();
-
-// Implementation
-
-	DECLARE_MESSAGE_MAP()
-};
-
-extern CWinTestApp theApp;
-
-#endif  // WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_WINTEST_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.rc b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.rc
deleted file mode 100644
index 8830be5034..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTest.rc
+++ /dev/null
@@ -1,394 +0,0 @@
-// Microsoft Visual C++ generated resource script.
-//
-#include "resource.h"
-
-#define APSTUDIO_READONLY_SYMBOLS
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 2 resource.
-//
-#include "afxres.h"
-
-/////////////////////////////////////////////////////////////////////////////
-#undef APSTUDIO_READONLY_SYMBOLS
-
-/////////////////////////////////////////////////////////////////////////////
-// Swedish resources
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
-#ifdef _WIN32
-LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
-#pragma code_page(1252)
-#endif //_WIN32
-
-#ifdef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// TEXTINCLUDE
-//
-
-1 TEXTINCLUDE 
-BEGIN
-    "resource.h\0"
-END
-
-2 TEXTINCLUDE 
-BEGIN
-    "#include ""afxres.h""\r\n"
-    "\0"
-END
-
-3 TEXTINCLUDE 
-BEGIN
-    "#define _AFX_NO_SPLITTER_RESOURCES\r\n"
-    "#define _AFX_NO_OLE_RESOURCES\r\n"
-    "#define _AFX_NO_TRACKER_RESOURCES\r\n"
-    "#define _AFX_NO_PROPERTY_RESOURCES\r\n"
-    "\r\n"
-    "#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)\r\n"
-    "LANGUAGE 29, 1\r\n"
-    "#pragma code_page(1252)\r\n"
-    "#include ""res\\WinTest.rc2""  // non-Microsoft Visual C++ edited resources\r\n"
-    "#include ""afxres.rc""     // Standard components\r\n"
-    "#endif\r\n"
-    "\0"
-END
-
-#endif    // APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Icon
-//
-
-// Icon with lowest ID value placed first to ensure application icon
-// remains consistent on all systems.
-IDR_MAINFRAME           ICON                    "res\\WinTest.ico"
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Dialog
-//
-
-IDD_ABOUTBOX DIALOGEX 0, 0, 235, 55
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
-CAPTION "About WinTest"
-FONT 8, "MS Shell Dlg", 0, 0, 0x1
-BEGIN
-    ICON            IDR_MAINFRAME,IDC_STATIC,11,17,20,20
-    LTEXT           "WinTest Version 1.0",IDC_STATIC,40,10,119,8,SS_NOPREFIX
-    LTEXT           "Copyright (C) 2010",IDC_STATIC,40,25,119,8
-    DEFPUSHBUTTON   "OK",IDOK,178,7,50,16,WS_GROUP
-END
-
-IDD_WINTEST_DIALOG DIALOGEX 0, 0, 796, 278
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_MINIMIZEBOX | WS_POPUP | WS_VISIBLE | WS_CAPTION | WS_SYSMENU
-EXSTYLE WS_EX_APPWINDOW
-CAPTION "WinTest"
-FONT 8, "MS Shell Dlg", 0, 0, 0x1
-BEGIN
-    PUSHBUTTON      "Create",IDC_BUTTON_CREATE_1,28,24,32,14
-    PUSHBUTTON      "Delete",IDC_BUTTON_DELETE_1,28,40,32,14
-    EDITTEXT        IDC_EDIT_1,6,32,18,14,ES_AUTOHSCROLL | ES_READONLY
-    PUSHBUTTON      "Create",IDC_BUTTON_CREATE_2,28,72,32,14
-    PUSHBUTTON      "Delete",IDC_BUTTON_DELETE_2,28,88,32,14
-    EDITTEXT        IDC_EDIT_2,6,82,18,14,ES_AUTOHSCROLL | ES_READONLY
-    EDITTEXT        IDC_EDIT_MESSAGE,28,244,764,12,ES_AUTOHSCROLL
-    COMBOBOX        IDC_COMBO_IP_1,64,24,76,30,CBS_DROPDOWN | CBS_SORT | WS_VSCROLL | WS_TABSTOP
-    EDITTEXT        IDC_EDIT_TX_PORT_1,144,24,28,14,ES_AUTOHSCROLL
-    EDITTEXT        IDC_EDIT_RX_PORT_1,144,40,28,14,ES_AUTOHSCROLL
-    COMBOBOX        IDC_COMBO_CODEC_1,176,24,76,156,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
-    PUSHBUTTON      "Start",IDC_BUTTON_START_LISTEN_1,256,24,32,14
-    PUSHBUTTON      "Stop",IDC_BUTTON_STOP_LISTEN_1,256,40,32,14
-    LTEXT           "Receive",IDC_STATIC_LISTEN,262,8,26,8
-    PUSHBUTTON      "Start",IDC_BUTTON_START_PLAYOUT_1,292,24,32,14
-    PUSHBUTTON      "Stop",IDC_BUTTON_STOP_PLAYOUT_1,292,40,32,14
-    LTEXT           "Playout",IDC_STATIC_PLAY,295,8,25,8
-    PUSHBUTTON      "Start",IDC_BUTTON_START_SEND_1,328,24,32,14
-    PUSHBUTTON      "Stop",IDC_BUTTON_STOP_SEND_1,328,40,32,14
-    LTEXT           "Send",IDC_STATIC_SEND,335,8,17,8
-    COMBOBOX        IDC_COMBO_IP_2,64,72,76,30,CBS_DROPDOWN | CBS_SORT | WS_VSCROLL | WS_TABSTOP
-    LTEXT           "Destination IP address",IDC_STATIC_IP,64,8,73,8
-    LTEXT           "Ports",IDC_STATIC_PORTS,145,8,18,8
-    LTEXT           "Codec",IDC_STATIC_CODEC,177,8,21,8
-    LTEXT           "Channel",IDC_STATIC_CHANNEL,30,8,27,8
-    LTEXT           "ID",IDC_STATIC_ID,12,8,8,8
-    EDITTEXT        IDC_EDIT_TX_PORT_2,144,72,28,14,ES_AUTOHSCROLL
-    EDITTEXT        IDC_EDIT_RX_PORT_2,144,88,28,14,ES_AUTOHSCROLL
-    COMBOBOX        IDC_COMBO_CODEC_2,176,72,76,156,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
-    PUSHBUTTON      "Start",IDC_BUTTON_START_LISTEN_2,256,72,32,14
-    PUSHBUTTON      "Stop",IDC_BUTTON_STOP_LISTEN_2,256,88,32,14
-    PUSHBUTTON      "Start",IDC_BUTTON_START_PLAYOUT_2,292,72,32,14
-    PUSHBUTTON      "Stop",IDC_BUTTON_STOP_PLAYOUT_2,292,88,32,14
-    PUSHBUTTON      "Start",IDC_BUTTON_START_SEND_2,328,72,32,14
-    PUSHBUTTON      "Stop",IDC_BUTTON_STOP_SEND_2,328,88,32,14
-    PUSHBUTTON      "TEST 1",IDC_BUTTON_TEST_1,756,224,36,14
-    LTEXT           "API",IDC_STATIC,4,247,12,8
-    EDITTEXT        IDC_EDIT_RESULT,28,260,96,12,ES_AUTOHSCROLL
-    LTEXT           "Result",IDC_STATIC,3,263,21,8
-    EDITTEXT        IDC_EDIT_N_FAILS,156,260,30,12,ES_AUTOHSCROLL
-    LTEXT           "#Fails",IDC_STATIC_ERROR,132,263,20,8
-    EDITTEXT        IDC_EDIT_LAST_ERROR,228,260,36,12,ES_AUTOHSCROLL
-    LTEXT           "Last Error",IDC_STATIC_LAST_ERROR,192,262,32,8
-    LTEXT           "Ext. Trans.",IDC_STATIC_EXTERNAL,361,8,37,8
-    CONTROL         "",IDC_CHECK_EXT_TRANS_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,371,33,16,10
-    CONTROL         "In",IDC_CHECK_PLAY_FILE_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,24,36,14,WS_EX_DLGMODALFRAME
-    LTEXT           "Play File",IDC_STATIC_PLAY_FILE,401,8,27,8
-    CONTROL         "Out",IDC_CHECK_PLAY_FILE_OUT_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,40,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "In",IDC_CHECK_PLAY_FILE_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,72,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Out",IDC_CHECK_PLAY_FILE_OUT_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,88,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "",IDC_CHECK_EXT_TRANS_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,371,82,16,10
-    GROUPBOX        "",IDC_STATIC_ALL_CHANNELS,6,107,662,113
-    CONTROL         "PlayFileAsMic",IDC_CHECK_PLAY_FILE_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,122,60,14,WS_EX_DLGMODALFRAME
-    COMBOBOX        IDC_COMBO_REC_DEVICE,12,132,184,80,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
-    LTEXT           "Recording device",IDC_STATIC_REC_DEVICE,12,120,56,8
-    COMBOBOX        IDC_COMBO_PLAY_DEVICE,12,180,184,80,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
-    LTEXT           "Playout device",IDC_STATIC_PLAY_DEVICE,12,167,56,8
-    CONTROL         "In",IDC_CHECK_EXT_MEDIA_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,24,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Out",IDC_CHECK_EXT_MEDIA_OUT_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,40,36,14,WS_EX_DLGMODALFRAME
-    LTEXT           "Ext. Media",IDC_STATIC_PLAY_FILE2,437,8,35,8
-    CONTROL         "",IDC_SLIDER_INPUT_VOLUME,"msctls_trackbar32",TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,196,130,72,15
-    LTEXT           "Microphone Volume",IDC_STATIC_MIC_VOLUME,202,120,62,8
-    CONTROL         "",IDC_SLIDER_OUTPUT_VOLUME,"msctls_trackbar32",TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,196,179,72,15
-    LTEXT           "Speaker Volume",IDC_STATIC_SPK_VOLUME,202,167,52,8
-    CONTROL         "AGC",IDC_CHECK_AGC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,316,122,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "",IDC_PROGRESS_AUDIO_LEVEL_IN,"msctls_progress32",WS_BORDER,268,135,42,6
-    LTEXT           "Audio Level",IDC_STATIC_AUDIO_LEVEL_IN,271,120,38,8,NOT WS_GROUP
-    CONTROL         "NS",IDC_CHECK_NS,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,316,142,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "EC",IDC_CHECK_EC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,356,122,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "VAD",IDC_CHECK_VAD_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,24,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "In",IDC_CHECK_EXT_MEDIA_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,72,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Out",IDC_CHECK_EXT_MEDIA_OUT_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,88,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "VAD",IDC_CHECK_VAD_3,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,72,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Mute",IDC_CHECK_MUTE_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,356,142,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Mute",IDC_CHECK_MUTE_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,40,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Mute",IDC_CHECK_MUTE_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,88,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "TX",IDC_CHECK_SRTP_TX_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,24,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "RX",IDC_CHECK_SRTP_RX_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,40,36,14,WS_EX_DLGMODALFRAME
-    LTEXT           "SRTP",IDC_STATIC_SRTP,525,8,18,8
-    CONTROL         "TX",IDC_CHECK_SRTP_TX_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,72,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "RX",IDC_CHECK_SRTP_RX_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,88,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "",IDC_CHECK_EXT_ENCRYPTION_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,564,33,16,10
-    LTEXT           "Encrypt",IDC_STATIC_PLAY_FILE4,556,8,26,8
-    CONTROL         "",IDC_CHECK_EXT_ENCRYPTION_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,564,82,16,10
-    PUSHBUTTON      "DTMF>>",IDC_BUTTON_DTMF_1,584,24,36,14
-    CONTROL         "RecMicToFile",IDC_CHECK_REC_MIC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,142,60,14,WS_EX_DLGMODALFRAME
-    PUSHBUTTON      "DTMF>>",IDC_BUTTON_DTMF_2,584,72,36,14
-    CONTROL         "Conf",IDC_CHECK_CONFERENCE_1,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,584,40,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Conf",IDC_CHECK_CONFERENCE_2,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,584,88,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Hold",IDC_CHECK_ON_HOLD_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,708,24,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Hold",IDC_CHECK_ON_HOLD_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,708,72,36,14,WS_EX_DLGMODALFRAME
-    EDITTEXT        IDC_EDIT_GET_OUTPUT,292,260,500,12,ES_AUTOHSCROLL
-    LTEXT           "Get",IDC_STATIC_GET,276,262,12,8
-    CONTROL         "Ext. Media",IDC_CHECK_EXT_MEDIA_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,460,122,52,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Ext. Media",IDC_CHECK_EXT_MEDIA_OUT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,460,180,52,14,WS_EX_DLGMODALFRAME
-    LISTBOX         IDC_LIST_CODEC_1,208,40,44,28,LBS_NOINTEGRALHEIGHT | NOT WS_BORDER | WS_VSCROLL | WS_TABSTOP,WS_EX_CLIENTEDGE
-    EDITTEXT        IDC_EDIT_CODEC_1,176,40,28,14,ES_AUTOHSCROLL
-    CONTROL         "",IDC_PROGRESS_AUDIO_LEVEL_OUT,"msctls_progress32",WS_BORDER,268,184,42,6
-    LTEXT           "Panning",IDC_STATIC_PANNING,328,167,26,8
-    CONTROL         "",IDC_SLIDER_PAN_LEFT,"msctls_trackbar32",TBS_VERT | TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,328,175,12,28
-    CONTROL         "",IDC_SLIDER_PAN_RIGHT,"msctls_trackbar32",TBS_VERT | TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,344,175,12,28
-    LTEXT           "L",IDC_STATIC_LEFT,332,200,8,8
-    LTEXT           "R",IDC_STATIC_RIGHT,347,201,8,8
-    PUSHBUTTON      "Version",IDC_BUTTON_VERSION,624,200,36,14
-    EDITTEXT        IDC_EDIT_PLAYOUT_BUFFER_SIZE,363,181,28,12,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY | NOT WS_TABSTOP
-    LTEXT           "Buffer Size",IDC_STATIC_PLAYOUT_BUFFER,361,167,36,8
-    CONTROL         "Delay",IDC_CHECK_DELAY_ESTIMATE_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,624,24,36,14,WS_EX_DLGMODALFRAME
-    EDITTEXT        IDC_EDIT_DELAY_ESTIMATE_1,631,40,24,14,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY | NOT WS_TABSTOP
-    CONTROL         "RxVAD",IDC_CHECK_RXVAD,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,664,24,40,14,WS_EX_DLGMODALFRAME
-    EDITTEXT        IDC_EDIT_RXVAD,671,40,24,14,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY
-    CONTROL         "AGC",IDC_CHECK_AGC_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,748,24,36,14,WS_EX_DLGMODALFRAME
-    CONTROL         "NS",IDC_CHECK_NS_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,748,40,36,14,WS_EX_DLGMODALFRAME
-    LTEXT           "RX VQE",IDC_STATIC_RX_VQE,753,8,25,8
-    CONTROL         "RecordCall",IDC_CHECK_REC_CALL,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,517,156,52,14,WS_EX_DLGMODALFRAME
-    LTEXT           "RX",IDC_STATIC_RX_PORT,133,42,10,8
-    LTEXT           "RX",IDC_STATIC_RX_PORT2,133,91,10,8
-    CONTROL         "TypingDetect",IDC_CHECK_TYPING_DETECTION,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,572,156,60,14,WS_EX_DLGMODALFRAME
-    EDITTEXT        IDC_EDIT_AUDIO_LAYER,28,224,116,14,ES_AUTOHSCROLL | ES_READONLY
-    EDITTEXT        IDC_EDIT_CPU_LOAD,152,224,116,14,ES_AUTOHSCROLL | ES_READONLY
-    CONTROL         "RED",IDC_CHECK_RED,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,176,55,28,14,WS_EX_DLGMODALFRAME
-    LTEXT           "=> Callbacks",IDC_STATIC_ERROR_CALLBACK,283,226,43,8
-    EDITTEXT        IDC_EDIT_ERROR_CALLBACK,328,224,312,14,ES_AUTOHSCROLL
-    PUSHBUTTON      "Clear",IDC_BUTTON_CLEAR_ERROR_CALLBACK,644,224,24,14
-    EDITTEXT        IDC_EDIT_RX_CODEC_1,256,56,216,12,ES_AUTOHSCROLL | ES_READONLY
-    EDITTEXT        IDC_EDIT_RTCP_STAT_1,476,56,316,12,ES_AUTOHSCROLL | ES_READONLY
-END
-
-IDD_DTMF_DIALOG DIALOGEX 0, 0, 316, 212
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
-CAPTION "Telehone Events"
-FONT 8, "MS Shell Dlg", 400, 0, 0x1
-BEGIN
-    DEFPUSHBUTTON   "OK",IDOK,260,192,50,14
-    PUSHBUTTON      "1",IDC_BUTTON_1,16,20,16,14
-    PUSHBUTTON      "2",IDC_BUTTON_2,36,20,16,14
-    PUSHBUTTON      "3",IDC_BUTTON_3,56,20,16,14
-    PUSHBUTTON      "4",IDC_BUTTON_4,16,36,16,14
-    PUSHBUTTON      "5",IDC_BUTTON_5,36,36,16,14
-    PUSHBUTTON      "6",IDC_BUTTON_6,56,36,16,14
-    PUSHBUTTON      "7",IDC_BUTTON_7,16,52,16,14
-    PUSHBUTTON      "8",IDC_BUTTON_8,36,52,16,14
-    PUSHBUTTON      "9",IDC_BUTTON_9,56,52,16,14
-    PUSHBUTTON      "*",IDC_BUTTON_10,16,68,16,14
-    PUSHBUTTON      "0",IDC_BUTTON_11,36,68,16,14
-    PUSHBUTTON      "#",IDC_BUTTON_12,56,68,16,14
-    PUSHBUTTON      "A",IDC_BUTTON_13,76,20,16,14
-    PUSHBUTTON      "B",IDC_BUTTON_14,76,36,16,14
-    PUSHBUTTON      "C",IDC_BUTTON_15,76,52,16,14
-    PUSHBUTTON      "D",IDC_BUTTON_16,76,68,16,14
-    EDITTEXT        IDC_EDIT_DTMF_EVENT,56,90,16,12,ES_AUTOHSCROLL | ES_READONLY
-    LTEXT           "Event code",IDC_STATIC_DTMF_EVENT,17,91,37,8
-    PUSHBUTTON      "1",IDC_BUTTON_17,16,20,16,14
-    PUSHBUTTON      "2",IDC_BUTTON_18,36,20,16,14
-    PUSHBUTTON      "3",IDC_BUTTON_19,56,20,16,14
-    PUSHBUTTON      "4",IDC_BUTTON_20,16,36,16,14
-    PUSHBUTTON      "A",IDC_BUTTON_21,76,20,16,14
-    GROUPBOX        "DTMF Events",IDC_STATIC_GROUP_DTMF,4,4,188,132
-    CONTROL         "",IDC_CHECK_DTMF_PLAYOUT_RX,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,160,21,12,14
-    LTEXT           "Play out-band RX",IDC_STATIC_PLAYOUT_RX,101,24,56,8
-    CONTROL         "",IDC_CHECK_DTMF_PLAY_TONE,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,160,39,12,14
-    LTEXT           "Play tone locally",IDC_STATIC_PLAY_TONE,101,41,52,8
-    EDITTEXT        IDC_EDIT_EVENT_LENGTH,44,163,28,14,ES_AUTOHSCROLL
-    LTEXT           "Duration",IDC_STATIC_EVENT_LENGTH,12,165,28,8
-    EDITTEXT        IDC_EDIT_EVENT_ATTENUATION,44,183,28,14,ES_AUTOHSCROLL
-    LTEXT           "Volume",IDC_STATIC_EVENT_ATTENUATION,12,186,24,8
-    CONTROL         "Inband",IDC_CHECK_EVENT_INBAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,84,163,40,14,WS_EX_DLGMODALFRAME
-    CONTROL         "Feedback",IDC_CHECK_DTMF_FEEDBACK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,16,112,48,14,WS_EX_DLGMODALFRAME
-    CONTROL         "",IDC_CHECK_DIRECT_FEEDBACK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,96,112,12,14
-    LTEXT           "Direct",IDC_STATIC_DIRECT_FEEDBACK,72,115,20,8
-    CONTROL         "Single",IDC_RADIO_SINGLE,"Button",BS_AUTORADIOBUTTON | WS_GROUP,112,68,35,10
-    CONTROL         "Sequence",IDC_RADIO_MULTI,"Button",BS_AUTORADIOBUTTON,112,80,47,10
-    CONTROL         "Start/Stop",IDC_RADIO_START_STOP,"Button",BS_AUTORADIOBUTTON,112,92,49,10
-    GROUPBOX        "Mode",IDC_STATIC_MODE,100,56,68,52
-    EDITTEXT        IDC_EDIT_EVENT_RX_PT,220,20,24,14,ES_AUTOHSCROLL
-    EDITTEXT        IDC_EDIT_EVENT_TX_PT,220,41,24,14,ES_AUTOHSCROLL
-    LTEXT           "RX",IDC_STATIC_EVENT_RX_PT,208,22,10,8
-    LTEXT           "TX",IDC_STATIC_EVENT_TX_PT,208,42,9,8
-    PUSHBUTTON      "Set",IDC_BUTTON_SET_TX_TELEPHONE_PT,248,41,24,14
-    PUSHBUTTON      "Set",IDC_BUTTON_SET_RX_TELEPHONE_PT,248,20,24,14
-    GROUPBOX        "Payload Type",IDC_STATIC_PT,200,4,80,56
-    EDITTEXT        IDC_EDIT_EVENT_CODE,128,163,28,14,ES_AUTOHSCROLL
-    LTEXT           "Event code",IDC_STATIC_EVENT_CODE,125,152,37,8
-    PUSHBUTTON      "Send",IDC_BUTTON_SEND_TELEPHONE_EVENT,160,163,24,14
-    CONTROL         "On/Off",IDC_CHECK_EVENT_DETECTION,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,80,40,14,WS_EX_DLGMODALFRAME
-    CONTROL         "",IDC_CHECK_DETECT_INBAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,100,12,14
-    CONTROL         "",IDC_CHECK_DETECT_OUT_OF_BAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,116,12,14
-    LTEXT           "Inband",IDC_STATIC_INBAND_DETECTION,220,103,24,8
-    LTEXT           "Outband",IDC_STATIC_OUT_OF_BAND_DETECTION,220,120,29,8
-    GROUPBOX        "Event Detection",IDC_STATIC_EVENT_DETECTION,200,68,108,68
-    GROUPBOX        "Telephone Events",IDC_STATIC_TELEPHONE_EVENTS,4,140,188,64
-    EDITTEXT        IDC_EDIT_ON_EVENT_OUT_OF_BAND,252,117,48,14,ES_AUTOHSCROLL
-    EDITTEXT        IDC_EDIT_ON_EVENT_INBAND,252,101,48,14,ES_AUTOHSCROLL
-    LTEXT           "=> Detections",IDC_STATIC_EVEN,253,90,48,8
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Version
-//
-
-VS_VERSION_INFO VERSIONINFO
- FILEVERSION 1,0,0,0
- PRODUCTVERSION 1,0,0,0
- FILEFLAGSMASK 0x3fL
-#ifdef _DEBUG
- FILEFLAGS 0x1L
-#else
- FILEFLAGS 0x0L
-#endif
- FILEOS 0x4L
- FILETYPE 0x1L
- FILESUBTYPE 0x0L
-BEGIN
-    BLOCK "StringFileInfo"
-    BEGIN
-        BLOCK "040904e4"
-        BEGIN
-            VALUE "FileDescription", "WebRTC VoiceEngine Test"
-            VALUE "FileVersion", "1.0.0.0"
-            VALUE "InternalName", "WinTest.exe"
-            VALUE "LegalCopyright", "Copyright (c) 2011 The WebRTC project authors. All Rights Reserved."
-            VALUE "OriginalFilename", "WinTest.exe"
-            VALUE "ProductName", "WebRTC VoiceEngine"
-            VALUE "ProductVersion", "1.0.0.0"
-        END
-    END
-    BLOCK "VarFileInfo"
-    BEGIN
-        VALUE "Translation", 0x409, 1252
-    END
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// DESIGNINFO
-//
-
-#ifdef APSTUDIO_INVOKED
-GUIDELINES DESIGNINFO 
-BEGIN
-    IDD_ABOUTBOX, DIALOG
-    BEGIN
-        LEFTMARGIN, 7
-        RIGHTMARGIN, 228
-        TOPMARGIN, 7
-        BOTTOMMARGIN, 48
-    END
-
-    IDD_WINTEST_DIALOG, DIALOG
-    BEGIN
-        LEFTMARGIN, 7
-        RIGHTMARGIN, 789
-        TOPMARGIN, 7
-        BOTTOMMARGIN, 271
-    END
-
-    IDD_DTMF_DIALOG, DIALOG
-    BEGIN
-        LEFTMARGIN, 7
-        RIGHTMARGIN, 309
-        TOPMARGIN, 7
-        BOTTOMMARGIN, 205
-    END
-END
-#endif    // APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// String Table
-//
-
-STRINGTABLE 
-BEGIN
-    IDS_ABOUTBOX            "&About WinTest..."
-END
-
-#endif    // Swedish resources
-/////////////////////////////////////////////////////////////////////////////
-
-
-
-#ifndef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 3 resource.
-//
-#define _AFX_NO_SPLITTER_RESOURCES
-#define _AFX_NO_OLE_RESOURCES
-#define _AFX_NO_TRACKER_RESOURCES
-#define _AFX_NO_PROPERTY_RESOURCES
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
-LANGUAGE 29, 1
-#pragma code_page(1252)
-#include "res\WinTest.rc2"  // non-Microsoft Visual C++ edited resources
-#include "afxres.rc"     // Standard components
-#endif
-
-/////////////////////////////////////////////////////////////////////////////
-#endif    // not APSTUDIO_INVOKED
-
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc
deleted file mode 100644
index 4436a86d86..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc
+++ /dev/null
@@ -1,3367 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/test/win_test/WinTest.h"
-#include "webrtc/voice_engine/test/win_test/WinTestDlg.h"
-#include "webrtc/voice_engine/test/win_test/stdafx.h"
-
-#ifdef _DEBUG
-#define new DEBUG_NEW
-#endif
-
-using namespace webrtc;
-
-unsigned char key[30] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9};
-
-// Hack to convert char to TCHAR, using two buffers to be able to
-// call twice in the same statement
-TCHAR convertTemp1[256] = {0};
-TCHAR convertTemp2[256] = {0};
-bool convertBufferSwitch(false);
-TCHAR* CharToTchar(const char* str, int len)
-{
-#ifdef _UNICODE
-  TCHAR* temp = convertBufferSwitch ? convertTemp1 : convertTemp2;
-  convertBufferSwitch = !convertBufferSwitch;
-  memset(temp, 0, sizeof(convertTemp1));
-  MultiByteToWideChar(CP_UTF8, 0, str, len, temp, 256);
-  return temp;
-#else
-  return str;
-#endif
-}
-
-// Hack to convert TCHAR to char
-char convertTemp3[256] = {0};
-char* TcharToChar(TCHAR* str, int len)
-{
-#ifdef _UNICODE
-  memset(convertTemp3, 0, sizeof(convertTemp3));
-  WideCharToMultiByte(CP_UTF8, 0, str, len, convertTemp3, 256, 0, 0);
-  return convertTemp3;
-#else
-  return str;
-#endif
-}
-
-// ----------------------------------------------------------------------------
-//    VoiceEngineObserver
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::CallbackOnError(int channel, int errCode)
-{
-    _nErrorCallbacks++;
-
-    CString str;
-    str.Format(_T("[#%d] CallbackOnError(channel=%d) => errCode = %d"), _nErrorCallbacks, channel, errCode);
-    if (errCode == VE_RECEIVE_PACKET_TIMEOUT)
-    {
-        str += _T(" <=> VE_RECEIVE_PACKET_TIMEOUT");
-    }
-    else if (errCode == VE_PACKET_RECEIPT_RESTARTED)
-    {
-        str += _T(" <=> VE_PACKET_RECEIPT_RESTARTED");
-    }
-    else if (errCode == VE_RUNTIME_PLAY_WARNING)
-    {
-        str += _T(" <=> VE_RUNTIME_PLAY_WARNING");
-    }
-    else if (errCode == VE_RUNTIME_REC_WARNING)
-    {
-        str += _T(" <=> VE_RUNTIME_REC_WARNING");
-    }
-    else if (errCode == VE_RUNTIME_PLAY_ERROR)
-    {
-        str += _T(" <=> VE_RUNTIME_PLAY_ERROR");
-    }
-    else if (errCode == VE_RUNTIME_REC_ERROR)
-    {
-        str += _T(" <=> VE_RUNTIME_REC_ERROR");
-    }
-    else if (errCode == VE_SATURATION_WARNING)
-    {
-        str += _T(" <=> VE_SATURATION_WARNING");
-    }
-    else if (errCode == VE_TYPING_NOISE_WARNING)
-    {
-        str += _T(" <=> VE_TYPING_NOISE_WARNING");
-    }
-    else if (errCode == VE_REC_DEVICE_REMOVED)
-    {
-        str += _T(" <=> VE_REC_DEVICE_REMOVED");
-    }
-    // AfxMessageBox((LPCTSTR)str, MB_OK);
-    SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
-}
-
-// ----------------------------------------------------------------------------
-//    VoERTPObserver
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnIncomingCSRCChanged(int channel, unsigned int CSRC, bool added)
-{
-    CString str;
-    str.Format(_T("OnIncomingCSRCChanged(channel=%d) => CSRC=%u, added=%d"), channel, CSRC, added);
-    SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
-}
-
-void CWinTestDlg::OnIncomingSSRCChanged(int channel, unsigned int SSRC)
-{
-    CString str;
-    str.Format(_T("OnIncomingSSRCChanged(channel=%d) => SSRC=%u"), channel, SSRC);
-    SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
-}
-
-// ----------------------------------------------------------------------------
-//    Transport
-// ----------------------------------------------------------------------------
-
-class MyTransport : public Transport
-{
-public:
-    MyTransport(VoENetwork* veNetwork);
-    int SendPacket(int channel, const void* data, size_t len) override;
-    int SendRTCPPacket(int channel, const void* data, size_t len) override;
-
-private:
-    VoENetwork* _veNetworkPtr;
-};
-
-MyTransport::MyTransport(VoENetwork* veNetwork) :
-    _veNetworkPtr(veNetwork)
-{
-}
-
-int
-MyTransport::SendPacket(int channel, const void *data, size_t len)
-{
-    _veNetworkPtr->ReceivedRTPPacket(channel, data, len);
-    return len;
-}
-
-int
-MyTransport::SendRTCPPacket(int channel, const void *data, size_t len)
-{
-    _veNetworkPtr->ReceivedRTCPPacket(channel, data, len);
-    return len;
-}
-
-// ----------------------------------------------------------------------------
-//    VoEMediaProcess
-// ----------------------------------------------------------------------------
-
-class MediaProcessImpl : public VoEMediaProcess
-{
-public:
-    MediaProcessImpl();
-    virtual void Process(int channel,
-                         ProcessingTypes type,
-                         int16_t audio_10ms[],
-                         int length,
-                         int samplingFreqHz,
-                         bool stereo);
-};
-
-MediaProcessImpl::MediaProcessImpl()
-{
-}
-
-void MediaProcessImpl::Process(int channel,
-                               ProcessingTypes type,
-                               int16_t audio_10ms[],
-                               int length,
-                               int samplingFreqHz,
-                               bool stereo)
-{
-    int x = rand() % 100;
-
-    for (int i = 0; i < length; i++)
-    {
-        if (channel == -1)
-        {
-            if (type == kPlaybackAllChannelsMixed)
-            {
-                // playout: scale up
-                if (!stereo)
-                {
-                    audio_10ms[i] = (audio_10ms[i] << 2);
-                }
-                else
-                {
-                    audio_10ms[2*i] = (audio_10ms[2*i] << 2);
-                    audio_10ms[2*i+1] = (audio_10ms[2*i+1] << 2);
-                }
-            }
-            else
-            {
-                // recording: emulate packet loss by "dropping" 10% of the packets
-                if (x >= 0 && x < 10)
-                {
-                    if (!stereo)
-                    {
-                        audio_10ms[i] = 0;
-                    }
-                    else
-                    {
-                        audio_10ms[2*i] = 0;
-                        audio_10ms[2*i+1] = 0;
-                    }
-                }
-            }
-        }
-        else
-        {
-            if (type == kPlaybackPerChannel)
-            {
-                // playout: mute
-                if (!stereo)
-                {
-                    audio_10ms[i] = 0;
-                }
-                else
-                {
-                    audio_10ms[2*i] = 0;
-                    audio_10ms[2*i+1] = 0;
-                }
-            }
-            else
-            {
-                // recording: emulate packet loss by "dropping" 50% of the packets
-                if (x >= 0 && x < 50)
-                {
-                    if (!stereo)
-                    {
-                        audio_10ms[i] = 0;
-                    }
-                    else
-                    {
-                        audio_10ms[2*i] = 0;
-                        audio_10ms[2*i+1] = 0;
-                    }
-                }
-            }
-        }
-    }
-}
-
-// ----------------------------------------------------------------------------
-//    TelephoneEventObserver
-// ----------------------------------------------------------------------------
-
-class TelephoneEventObserver: public VoETelephoneEventObserver
-{
-public:
-    TelephoneEventObserver(CWnd* editControlOut, CWnd* editControlIn);
-    virtual void OnReceivedTelephoneEventInband(int channel, int eventCode,
-                                                bool endOfEvent);
-    virtual void OnReceivedTelephoneEventOutOfBand(int channel, int eventCode,
-                                                   bool endOfEvent);
-private:
-    CWnd* _editControlOutPtr;
-    CWnd* _editControlInPtr;
-};
-
-TelephoneEventObserver::TelephoneEventObserver(CWnd* editControlOut, CWnd* editControlIn) :
-    _editControlOutPtr(editControlOut),
-    _editControlInPtr(editControlIn)
-{
-}
-
-void TelephoneEventObserver::OnReceivedTelephoneEventInband(int channel,
-                                                            int eventCode,
-                                                            bool endOfEvent)
-{
-    CString msg;
-    if (endOfEvent)
-    {
-        msg.AppendFormat(_T("%d [END]"), eventCode);
-        _editControlInPtr->SetWindowText((LPCTSTR)msg);
-    }
-    else
-    {
-        msg.AppendFormat(_T("%d [START]"), eventCode);
-        _editControlInPtr->SetWindowText((LPCTSTR)msg);
-    }
-}
-
-void TelephoneEventObserver::OnReceivedTelephoneEventOutOfBand(int channel,
-                                                               int eventCode,
-                                                               bool endOfEvent)
-{
-    CString msg;
-    if (endOfEvent)
-    {
-        msg.AppendFormat(_T("%d [END]"), eventCode);
-        _editControlOutPtr->SetWindowText((LPCTSTR)msg);
-    }
-    else
-    {
-        msg.AppendFormat(_T("%d [START]"), eventCode);
-        _editControlOutPtr->SetWindowText((LPCTSTR)msg);
-    }
-}
-
-// ----------------------------------------------------------------------------
-//    RxVadCallback
-// ----------------------------------------------------------------------------
-
-class RxCallback : public VoERxVadCallback
-{
-public:
-    RxCallback() : vad_decision(-1) {};
-
-    virtual void OnRxVad(int , int vadDecision)
-    {
-        vad_decision = vadDecision;
-    }
-
-    int vad_decision;
-};
-
-// ----------------------------------------------------------------------------
-//                                 CAboutDlg dialog
-// ----------------------------------------------------------------------------
-
-class CAboutDlg : public CDialog
-{
-public:
-    CAboutDlg();
-
-// Dialog Data
-    enum { IDD = IDD_ABOUTBOX };
-
-    protected:
-    virtual void DoDataExchange(CDataExchange* pDX);    // DDX/DDV support
-
-// Implementation
-protected:
-    DECLARE_MESSAGE_MAP()
-};
-
-CAboutDlg::CAboutDlg() : CDialog(CAboutDlg::IDD)
-{
-}
-
-void CAboutDlg::DoDataExchange(CDataExchange* pDX)
-{
-    CDialog::DoDataExchange(pDX);
-}
-
-BEGIN_MESSAGE_MAP(CAboutDlg, CDialog)
-END_MESSAGE_MAP()
-
-// ----------------------------------------------------------------------------
-//                               CTelephonyEvent dialog
-// ----------------------------------------------------------------------------
-
-class CTelephonyEvent : public CDialog
-{
-    DECLARE_DYNAMIC(CTelephonyEvent)
-
-public:
-    CTelephonyEvent(VoiceEngine* voiceEngine, int channel, CDialog* pParentDialog, CWnd* pParent = NULL);   // standard constructor
-    virtual ~CTelephonyEvent();
-
-// Dialog Data
-    enum { IDD = IDD_DTMF_DIALOG };
-
-protected:
-    virtual void DoDataExchange(CDataExchange* pDX);    // DDX/DDV support
-    virtual BOOL OnInitDialog();
-
-    DECLARE_MESSAGE_MAP()
-public:
-    afx_msg void OnBnClickedButton1();
-    afx_msg void OnBnClickedButton2();
-    afx_msg void OnBnClickedButton3();
-    afx_msg void OnBnClickedButton4();
-    afx_msg void OnBnClickedButton5();
-    afx_msg void OnBnClickedButton6();
-    afx_msg void OnBnClickedButton7();
-    afx_msg void OnBnClickedButton8();
-    afx_msg void OnBnClickedButton9();
-    afx_msg void OnBnClickedButton10();
-    afx_msg void OnBnClickedButton11();
-    afx_msg void OnBnClickedButton12();
-    afx_msg void OnBnClickedButtonA();
-    afx_msg void OnBnClickedButtonB();
-    afx_msg void OnBnClickedButtonC();
-    afx_msg void OnBnClickedButtonD();
-    afx_msg void OnBnClickedCheckDtmfPlayoutRx();
-    afx_msg void OnBnClickedCheckDtmfPlayTone();
-    afx_msg void OnBnClickedCheckStartStopMode();
-    afx_msg void OnBnClickedCheckEventInband();
-    afx_msg void OnBnClickedCheckDtmfFeedback();
-    afx_msg void OnBnClickedCheckDirectFeedback();
-    afx_msg void OnBnClickedRadioSingle();
-    afx_msg void OnBnClickedRadioMulti();
-    afx_msg void OnBnClickedRadioStartStop();
-    afx_msg void OnBnClickedButtonSetRxTelephonePt();
-    afx_msg void OnBnClickedButtonSetTxTelephonePt();
-    afx_msg void OnBnClickedButtonSendTelephoneEvent();
-    afx_msg void OnBnClickedCheckDetectInband();
-    afx_msg void OnBnClickedCheckDetectOutOfBand();
-    afx_msg void OnBnClickedCheckEventDetection();
-
-private:
-    void SendTelephoneEvent(unsigned char eventCode);
-
-private:
-    VoiceEngine*                _vePtr;
-    VoEBase*                    _veBasePtr;
-    VoEDtmf*                    _veDTMFPtr;
-    VoECodec*                   _veCodecPtr;
-    int                         _channel;
-    CString                     _strMsg;
-    CDialog*                    _parentDialogPtr;
-    TelephoneEventObserver*     _telephoneEventObserverPtr;
-    bool                        _PlayDtmfToneLocally;
-    bool                        _modeStartStop;
-    bool                        _modeSingle;
-    bool                        _modeSequence;
-    bool                        _playingDTMFTone;
-    bool                        _outOfBandEventDetection;
-    bool                        _inbandEventDetection;
-};
-
-IMPLEMENT_DYNAMIC(CTelephonyEvent, CDialog)
-
-CTelephonyEvent::CTelephonyEvent(VoiceEngine* voiceEngine,
-                                 int channel,
-                                 CDialog* pParentDialog,
-                                 CWnd* pParent /*=NULL*/)
-    : _vePtr(voiceEngine),
-      _channel(channel),
-      _PlayDtmfToneLocally(false),
-      _modeStartStop(false),
-      _modeSingle(true),
-      _modeSequence(false),
-      _playingDTMFTone(false),
-      _outOfBandEventDetection(true),
-      _inbandEventDetection(false),
-      _parentDialogPtr(pParentDialog),
-      _telephoneEventObserverPtr(NULL),
-      CDialog(CTelephonyEvent::IDD, pParent)
-{
-    _veBasePtr = VoEBase::GetInterface(_vePtr);
-    _veDTMFPtr = VoEDtmf::GetInterface(_vePtr);
-    _veCodecPtr = VoECodec::GetInterface(_vePtr);
-}
-
-CTelephonyEvent::~CTelephonyEvent()
-{
-    _veDTMFPtr->Release();
-    _veCodecPtr->Release();
-    _veBasePtr->Release();
-
-    if (_telephoneEventObserverPtr)
-    {
-        _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
-        delete _telephoneEventObserverPtr;
-        _telephoneEventObserverPtr = NULL;
-    }
-}
-
-void CTelephonyEvent::DoDataExchange(CDataExchange* pDX)
-{
-    CDialog::DoDataExchange(pDX);
-}
-
-
-BEGIN_MESSAGE_MAP(CTelephonyEvent, CDialog)
-    ON_BN_CLICKED(IDC_BUTTON_1, &CTelephonyEvent::OnBnClickedButton1)
-    ON_BN_CLICKED(IDC_BUTTON_2, &CTelephonyEvent::OnBnClickedButton2)
-    ON_BN_CLICKED(IDC_BUTTON_3, &CTelephonyEvent::OnBnClickedButton3)
-    ON_BN_CLICKED(IDC_BUTTON_4, &CTelephonyEvent::OnBnClickedButton4)
-    ON_BN_CLICKED(IDC_BUTTON_5, &CTelephonyEvent::OnBnClickedButton5)
-    ON_BN_CLICKED(IDC_BUTTON_6, &CTelephonyEvent::OnBnClickedButton6)
-    ON_BN_CLICKED(IDC_BUTTON_7, &CTelephonyEvent::OnBnClickedButton7)
-    ON_BN_CLICKED(IDC_BUTTON_8, &CTelephonyEvent::OnBnClickedButton8)
-    ON_BN_CLICKED(IDC_BUTTON_9, &CTelephonyEvent::OnBnClickedButton9)
-    ON_BN_CLICKED(IDC_BUTTON_10, &CTelephonyEvent::OnBnClickedButton10)
-    ON_BN_CLICKED(IDC_BUTTON_11, &CTelephonyEvent::OnBnClickedButton11)
-    ON_BN_CLICKED(IDC_BUTTON_12, &CTelephonyEvent::OnBnClickedButton12)
-    ON_BN_CLICKED(IDC_BUTTON_13, &CTelephonyEvent::OnBnClickedButtonA)
-    ON_BN_CLICKED(IDC_BUTTON_14, &CTelephonyEvent::OnBnClickedButtonB)
-    ON_BN_CLICKED(IDC_BUTTON_15, &CTelephonyEvent::OnBnClickedButtonC)
-    ON_BN_CLICKED(IDC_BUTTON_16, &CTelephonyEvent::OnBnClickedButtonD)
-    ON_BN_CLICKED(IDC_CHECK_DTMF_PLAYOUT_RX, &CTelephonyEvent::OnBnClickedCheckDtmfPlayoutRx)
-    ON_BN_CLICKED(IDC_CHECK_DTMF_PLAY_TONE, &CTelephonyEvent::OnBnClickedCheckDtmfPlayTone)
-    ON_BN_CLICKED(IDC_CHECK_EVENT_INBAND, &CTelephonyEvent::OnBnClickedCheckEventInband)
-    ON_BN_CLICKED(IDC_CHECK_DTMF_FEEDBACK, &CTelephonyEvent::OnBnClickedCheckDtmfFeedback)
-    ON_BN_CLICKED(IDC_CHECK_DIRECT_FEEDBACK, &CTelephonyEvent::OnBnClickedCheckDirectFeedback)
-    ON_BN_CLICKED(IDC_RADIO_SINGLE, &CTelephonyEvent::OnBnClickedRadioSingle)
-    ON_BN_CLICKED(IDC_RADIO_MULTI, &CTelephonyEvent::OnBnClickedRadioMulti)
-    ON_BN_CLICKED(IDC_RADIO_START_STOP, &CTelephonyEvent::OnBnClickedRadioStartStop)
-    ON_BN_CLICKED(IDC_BUTTON_SET_RX_TELEPHONE_PT, &CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt)
-    ON_BN_CLICKED(IDC_BUTTON_SET_TX_TELEPHONE_PT, &CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt)
-    ON_BN_CLICKED(IDC_BUTTON_SEND_TELEPHONE_EVENT, &CTelephonyEvent::OnBnClickedButtonSendTelephoneEvent)
-    ON_BN_CLICKED(IDC_CHECK_DETECT_INBAND, &CTelephonyEvent::OnBnClickedCheckDetectInband)
-    ON_BN_CLICKED(IDC_CHECK_DETECT_OUT_OF_BAND, &CTelephonyEvent::OnBnClickedCheckDetectOutOfBand)
-    ON_BN_CLICKED(IDC_CHECK_EVENT_DETECTION, &CTelephonyEvent::OnBnClickedCheckEventDetection)
-END_MESSAGE_MAP()
-
-
-// CTelephonyEvent message handlers
-
-BOOL CTelephonyEvent::OnInitDialog()
-{
-    CDialog::OnInitDialog();
-
-    CString str;
-    GetWindowText(str);
-    str.AppendFormat(_T(" [channel = %d]"), _channel);
-    SetWindowText(str);
-
-    // Update dialog with latest playout state
-    bool enabled(false);
-    _veDTMFPtr->GetDtmfPlayoutStatus(_channel, enabled);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAYOUT_RX);
-    button->SetCheck(enabled ? BST_CHECKED : BST_UNCHECKED);
-
-    // Update dialog with latest feedback state
-    bool directFeedback(false);
-    _veDTMFPtr->GetDtmfFeedbackStatus(enabled, directFeedback);
-    button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
-    button->SetCheck(enabled ? BST_CHECKED : BST_UNCHECKED);
-    button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
-    button->SetCheck(directFeedback ? BST_CHECKED : BST_UNCHECKED);
-
-    // Default event length is 160 ms
-    SetDlgItemInt(IDC_EDIT_EVENT_LENGTH, 160);
-
-    // Default event attenuation is 10 (<-> -10dBm0)
-    SetDlgItemInt(IDC_EDIT_EVENT_ATTENUATION, 10);
-
-    // Current event-detection status
-    TelephoneEventDetectionMethods detectionMethod(kOutOfBand);
-    if (_veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod) == 0)
-    {
-        // DTMF detection is supported
-        if (enabled)
-        {
-            button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_DETECTION);
-            button->SetCheck(BST_CHECKED);
-        }
-        if (detectionMethod == kOutOfBand || detectionMethod == kInAndOutOfBand)
-        {
-            button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND);
-            button->SetCheck(BST_CHECKED);
-        }
-        if (detectionMethod == kInBand || detectionMethod == kInAndOutOfBand)
-        {
-            button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_INBAND);
-            button->SetCheck(BST_CHECKED);
-        }
-    }
-    else
-    {
-        // DTMF detection is not supported
-        GetDlgItem(IDC_CHECK_EVENT_DETECTION)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_DETECT_INBAND)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_ON_EVENT_INBAND)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_ON_EVENT_OUT_OF_BAND)->EnableWindow(FALSE);
-    }
-
-    // Telephone-event PTs
-    unsigned char pt(0);
-    _veDTMFPtr->GetSendTelephoneEventPayloadType(_channel, pt);
-    SetDlgItemInt(IDC_EDIT_EVENT_TX_PT, pt);
-
-    CodecInst codec;
-    strcpy_s(codec.plname, 32, "telephone-event"); codec.channels = 1; codec.plfreq = 8000;
-    _veCodecPtr->GetRecPayloadType(_channel, codec);
-    SetDlgItemInt(IDC_EDIT_EVENT_RX_PT, codec.pltype);
-
-    if (_modeSingle)
-    {
-        ((CButton*)GetDlgItem(IDC_RADIO_SINGLE))->SetCheck(BST_CHECKED);
-    }
-    else if (_modeStartStop)
-    {
-        ((CButton*)GetDlgItem(IDC_RADIO_START_STOP))->SetCheck(BST_CHECKED);
-    }
-    else if (_modeSequence)
-    {
-        ((CButton*)GetDlgItem(IDC_RADIO_MULTI))->SetCheck(BST_CHECKED);
-    }
-
-    return TRUE;  // return TRUE  unless you set the focus to a control
-}
-void CTelephonyEvent::SendTelephoneEvent(unsigned char eventCode)
-{
-    BOOL ret;
-    int lengthMs(0);
-    int attenuationDb(0);
-    bool outBand(false);
-    int res(0);
-
-    // tone length
-    if (!_modeStartStop)
-    {
-        lengthMs = GetDlgItemInt(IDC_EDIT_EVENT_LENGTH, &ret);
-        if (ret == FALSE)
-        {
-            // use default length if edit field is empty
-            lengthMs = 160;
-        }
-    }
-
-    // attenuation
-    attenuationDb = GetDlgItemInt(IDC_EDIT_EVENT_ATTENUATION, &ret);
-    if (ret == FALSE)
-    {
-        // use default length if edit field is empty
-        attenuationDb = 10;
-    }
-
-    // out-band or in-band
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_INBAND);
-    int check = button->GetCheck();
-    outBand = (check == BST_UNCHECKED);
-
-    if (eventCode < 16)
-        SetDlgItemInt(IDC_EDIT_DTMF_EVENT, eventCode);
-
-    if (_PlayDtmfToneLocally)
-    {
-        // --- PlayDtmfTone
-
-        if (_modeSingle)
-        {
-            TEST2(_veDTMFPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb) == 0,
-                _T("PlayDtmfTone(eventCode=%u, lengthMs=%d, attenuationDb=%d)"), eventCode, lengthMs, attenuationDb);
-        }
-        else if (_modeStartStop)
-        {
-            if (!_playingDTMFTone)
-            {
-                TEST2((res = _veDTMFPtr->StartPlayingDtmfTone(eventCode, attenuationDb)) == 0,
-                    _T("StartPlayingDtmfTone(eventCode=%u, attenuationDb=%d)"), eventCode, attenuationDb);
-            }
-            else
-            {
-                TEST2((res = _veDTMFPtr->StopPlayingDtmfTone()) == 0,
-                    _T("StopPlayingDTMFTone()"));
-            }
-            if (res == 0)
-                _playingDTMFTone = !_playingDTMFTone;
-        }
-        else if (_modeSequence)
-        {
-            int nTones(1);
-            int sleepMs(0);
-            int lenMult(1);
-            if (eventCode == 1)
-            {
-                nTones = 2;
-                sleepMs = lengthMs;
-                lenMult = 1;
-            }
-            else if (eventCode == 2)
-            {
-                nTones = 2;
-                sleepMs = lengthMs/2;
-                lenMult = 2;
-            }
-            else if (eventCode == 3)
-            {
-                nTones = 3;
-                sleepMs = 0;
-                lenMult = 1;
-            }
-            for (int i = 0; i < nTones; i++)
-            {
-                TEST2(_veDTMFPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb) == 0,
-                    _T("PlayDtmfTone(eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), eventCode, lengthMs, attenuationDb);
-                Sleep(sleepMs);
-                lengthMs = lenMult*lengthMs;
-                eventCode++;
-            }
-        }
-    }
-    else
-    {
-        // --- SendTelephoneEvent
-
-        if (_modeSingle)
-        {
-            TEST2(_veDTMFPtr->SendTelephoneEvent(_channel, eventCode, outBand, lengthMs, attenuationDb) == 0,
-                _T("SendTelephoneEvent(channel=%d, eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), _channel, eventCode, outBand, lengthMs, attenuationDb);
-        }
-        else if (_modeStartStop)
-        {
-            TEST2(false, _T("*** NOT IMPLEMENTED ***"));
-        }
-        else if (_modeSequence)
-        {
-            int nTones(1);
-            int sleepMs(0);
-            int lenMult(1);
-            if (eventCode == 1)
-            {
-                nTones = 2;
-                sleepMs = lengthMs;
-                lenMult = 1;
-            }
-            else if (eventCode == 2)
-            {
-                eventCode = 1;
-                nTones = 2;
-                sleepMs = lengthMs/2;
-                lenMult = 2;
-            }
-            else if (eventCode == 3)
-            {
-                eventCode = 1;
-                nTones = 3;
-                sleepMs = 0;
-                lenMult = 1;
-            }
-            for (int i = 0; i < nTones; i++)
-            {
-                TEST2(_veDTMFPtr->SendTelephoneEvent(_channel, eventCode, outBand, lengthMs, attenuationDb) == 0,
-                    _T("SendTelephoneEvent(channel=%d, eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), _channel, eventCode, outBand, lengthMs, attenuationDb);
-                Sleep(sleepMs);
-                lengthMs = lenMult*lengthMs;
-                eventCode++;
-            }
-        }
-    }
-}
-
-void CTelephonyEvent::OnBnClickedButtonSendTelephoneEvent()
-{
-    BOOL ret;
-    unsigned char eventCode(0);
-
-    eventCode = (unsigned char)GetDlgItemInt(IDC_EDIT_EVENT_CODE, &ret);
-    if (ret == FALSE)
-    {
-        return;
-    }
-    SendTelephoneEvent(eventCode);
-}
-
-void CTelephonyEvent::OnBnClickedButton1()
-{
-    SendTelephoneEvent(1);
-}
-
-void CTelephonyEvent::OnBnClickedButton2()
-{
-    SendTelephoneEvent(2);
-}
-
-void CTelephonyEvent::OnBnClickedButton3()
-{
-    SendTelephoneEvent(3);
-}
-
-void CTelephonyEvent::OnBnClickedButton4()
-{
-    SendTelephoneEvent(4);
-}
-
-void CTelephonyEvent::OnBnClickedButton5()
-{
-    SendTelephoneEvent(5);
-}
-
-void CTelephonyEvent::OnBnClickedButton6()
-{
-    SendTelephoneEvent(6);
-}
-
-void CTelephonyEvent::OnBnClickedButton7()
-{
-    SendTelephoneEvent(7);
-}
-
-void CTelephonyEvent::OnBnClickedButton8()
-{
-    SendTelephoneEvent(8);
-}
-
-void CTelephonyEvent::OnBnClickedButton9()
-{
-    SendTelephoneEvent(9);
-}
-
-void CTelephonyEvent::OnBnClickedButton10()
-{
-    // *
-    SendTelephoneEvent(10);
-}
-
-void CTelephonyEvent::OnBnClickedButton11()
-{
-    SendTelephoneEvent(0);
-}
-
-void CTelephonyEvent::OnBnClickedButton12()
-{
-    // #
-    SendTelephoneEvent(11);
-}
-
-void CTelephonyEvent::OnBnClickedButtonA()
-{
-    SendTelephoneEvent(12);
-}
-
-void CTelephonyEvent::OnBnClickedButtonB()
-{
-    SendTelephoneEvent(13);
-}
-
-void CTelephonyEvent::OnBnClickedButtonC()
-{
-    SendTelephoneEvent(14);
-}
-
-void CTelephonyEvent::OnBnClickedButtonD()
-{
-    SendTelephoneEvent(15);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDtmfPlayoutRx()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAYOUT_RX);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    TEST2(_veDTMFPtr->SetDtmfPlayoutStatus(_channel, enable) == 0, _T("SetDtmfPlayoutStatus(channel=%d, enable=%d)"), _channel, enable);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDtmfPlayTone()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAY_TONE);
-    int check = button->GetCheck();
-    _PlayDtmfToneLocally = (check == BST_CHECKED);
-}
-
-void CTelephonyEvent::OnBnClickedRadioSingle()
-{
-    _modeStartStop = false;
-    _modeSingle = true;
-    _modeSequence = false;
-}
-
-void CTelephonyEvent::OnBnClickedRadioMulti()
-{
-    _modeStartStop = false;
-    _modeSingle = false;
-    _modeSequence = true;
-}
-
-void CTelephonyEvent::OnBnClickedRadioStartStop()
-{
-    // CButton* button = (CButton*)GetDlgItem(IDC_RADIO_START_STOP);
-    // int check = button->GetCheck();
-    _modeStartStop = true;
-    _modeSingle = false;
-    _modeSequence = false;
-    // GetDlgItem(IDC_EDIT_EVENT_LENGTH)->EnableWindow();
-}
-
-void CTelephonyEvent::OnBnClickedCheckEventInband()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_INBAND);
-    int check = button->GetCheck();
-    GetDlgItem(IDC_EDIT_EVENT_CODE)->EnableWindow(check?FALSE:TRUE);
-    GetDlgItem(IDC_BUTTON_SEND_TELEPHONE_EVENT)->EnableWindow(check?FALSE:TRUE);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDtmfFeedback()
-{
-    CButton* button(NULL);
-
-    // Retrieve feedback state
-    button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-
-    // Retrieve direct-feedback setting
-    button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
-    check = button->GetCheck();
-    const bool directFeedback = (check == BST_CHECKED);
-
-    // GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK)->EnableWindow(enable ? TRUE : FALSE);
-
-    TEST2(_veDTMFPtr->SetDtmfFeedbackStatus(enable, directFeedback) == 0,
-        _T("SetDtmfFeedbackStatus(enable=%d, directFeedback=%d)"), enable, directFeedback);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDirectFeedback()
-{
-    CButton* button(NULL);
-
-    // Retrieve feedback state
-    button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-
-    // Retrieve new direct-feedback setting
-    button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
-    check = button->GetCheck();
-    const bool directFeedback = (check == BST_CHECKED);
-
-    TEST2(_veDTMFPtr->SetDtmfFeedbackStatus(enable, directFeedback) == 0,
-        _T("SetDtmfFeedbackStatus(enable=%d, directFeedback=%d)"), enable, directFeedback);
-}
-
-void CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt()
-{
-    BOOL ret;
-    int pt = GetDlgItemInt(IDC_EDIT_EVENT_RX_PT, &ret);
-    if (ret == FALSE || pt < 0 || pt > 127)
-        return;
-    CodecInst codec;
-    strcpy_s(codec.plname, 32, "telephone-event");
-    codec.pltype = pt;
-    codec.channels = 1;
-    codec.plfreq = 8000;
-    TEST2(_veCodecPtr->SetRecPayloadType(_channel, codec) == 0,
-          _T("SetRecPayloadType(channel=%d, codec.pltype=%d)"), _channel,
-          codec.pltype);
-}
-
-void CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt()
-{
-    BOOL ret;
-    int pt = GetDlgItemInt(IDC_EDIT_EVENT_TX_PT, &ret);
-    if (ret == FALSE || pt < 0 || pt > 127)
-        return;
-    TEST2(_veDTMFPtr->SetSendTelephoneEventPayloadType(_channel, pt) == 0,
-          _T("SetSendTelephoneEventPayloadType(channel=%d, type=%d)"), _channel,
-          pt);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDetectInband()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_INBAND);
-    int check = button->GetCheck();
-    _inbandEventDetection = (check == BST_CHECKED);
-
-    bool enabled(false);
-    TelephoneEventDetectionMethods detectionMethod;
-    _veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod);
-    if (enabled)
-    {
-        // deregister
-        _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
-        delete _telephoneEventObserverPtr;
-        _telephoneEventObserverPtr = NULL;
-        SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
-        SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
-    }
-    OnBnClickedCheckEventDetection();
-}
-
-void CTelephonyEvent::OnBnClickedCheckDetectOutOfBand()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND);
-    int check = button->GetCheck();
-    _outOfBandEventDetection = (check == BST_CHECKED);
-
-    bool enabled(false);
-    TelephoneEventDetectionMethods detectionMethod;
-    _veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod);
-    if (enabled)
-    {
-        // deregister
-        _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
-        delete _telephoneEventObserverPtr;
-        _telephoneEventObserverPtr = NULL;
-        SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
-        SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
-    }
-    OnBnClickedCheckEventDetection();
-}
-
-void CTelephonyEvent::OnBnClickedCheckEventDetection()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_DETECTION);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-
-    if (enable)
-    {
-        TelephoneEventDetectionMethods method(kInBand);
-        if (_inbandEventDetection && !_outOfBandEventDetection)
-            method = kInBand;
-        else if (!_inbandEventDetection && _outOfBandEventDetection)
-            method = kOutOfBand;
-        else if (_inbandEventDetection && _outOfBandEventDetection)
-            method = kInAndOutOfBand;
-
-        CWnd* wndOut = GetDlgItem(IDC_EDIT_ON_EVENT_OUT_OF_BAND);
-        CWnd* wndIn = GetDlgItem(IDC_EDIT_ON_EVENT_INBAND);
-        _telephoneEventObserverPtr = new TelephoneEventObserver(wndOut, wndIn);
-
-        TEST2(_veDTMFPtr->RegisterTelephoneEventDetection(_channel, method, *_telephoneEventObserverPtr) == 0,
-            _T("RegisterTelephoneEventDetection(channel=%d, detectionMethod=%d)"), _channel, method);
-    }
-    else
-    {
-        TEST2(_veDTMFPtr->DeRegisterTelephoneEventDetection(_channel) == 0,
-            _T("DeRegisterTelephoneEventDetection(channel=%d)"), _channel);
-        delete _telephoneEventObserverPtr;
-        _telephoneEventObserverPtr = NULL;
-        SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
-        SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
-    }
-}
-
-// ============================================================================
-//                                 CWinTestDlg dialog
-// ============================================================================
-
-CWinTestDlg::CWinTestDlg(CWnd* pParent /*=NULL*/)
-    : CDialog(CWinTestDlg::IDD, pParent),
-    _failCount(0),
-    _vePtr(NULL),
-    _veBasePtr(NULL),
-    _veCodecPtr(NULL),
-    _veNetworkPtr(NULL),
-    _veFilePtr(NULL),
-    _veHardwarePtr(NULL),
-    _veExternalMediaPtr(NULL),
-    _veApmPtr(NULL),
-    _veRtpRtcpPtr(NULL),
-    _transportPtr(NULL),
-    _externalMediaPtr(NULL),
-    _externalTransport(false),
-    _externalTransportBuild(false),
-    _checkPlayFileIn(0),
-    _checkPlayFileIn1(0),
-    _checkPlayFileIn2(0),
-    _checkPlayFileOut1(0),
-    _checkPlayFileOut2(0),
-    _checkAGC(0),
-    _checkAGC1(0),
-    _checkNS(0),
-    _checkNS1(0),
-    _checkEC(0),
-    _checkVAD1(0),
-    _checkVAD2(0),
-    _checkSrtpTx1(0),
-    _checkSrtpTx2(0),
-    _checkSrtpRx1(0),
-    _checkSrtpRx2(0),
-    _checkConference1(0),
-    _checkConference2(0),
-    _checkOnHold1(0),
-    _checkOnHold2(0),
-    _strComboIp1(_T("")),
-    _strComboIp2(_T("")),
-    _delayEstimate1(false),
-    _delayEstimate2(false),
-    _rxVad(false),
-    _nErrorCallbacks(0),
-    _timerTicks(0)
-{
-    m_hIcon = AfxGetApp()->LoadIcon(IDR_MAINFRAME);
-
-    _vePtr = VoiceEngine::Create();
-
-    VoiceEngine::SetTraceFilter(kTraceNone);
-    // VoiceEngine::SetTraceFilter(kTraceAll);
-    // VoiceEngine::SetTraceFilter(kTraceStream | kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | kTraceApiCall | kTraceModuleCall | kTraceMemory | kTraceDebug | kTraceInfo);
-    // VoiceEngine::SetTraceFilter(kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | kTraceApiCall | kTraceModuleCall | kTraceMemory | kTraceInfo);
-
-    VoiceEngine::SetTraceFile("ve_win_test.txt");
-    VoiceEngine::SetTraceCallback(NULL);
-
-    if (_vePtr)
-    {
-        _veExternalMediaPtr = VoEExternalMedia::GetInterface(_vePtr);
-        _veVolumeControlPtr = VoEVolumeControl::GetInterface(_vePtr);
-        _veVideoSyncPtr = VoEVideoSync::GetInterface(_vePtr);
-        _veNetworkPtr = VoENetwork::GetInterface(_vePtr);
-        _veFilePtr = VoEFile::GetInterface(_vePtr);
-        _veApmPtr = VoEAudioProcessing::GetInterface(_vePtr);
-
-        _veBasePtr = VoEBase::GetInterface(_vePtr);
-        _veCodecPtr = VoECodec::GetInterface(_vePtr);
-        _veHardwarePtr = VoEHardware::GetInterface(_vePtr);
-        _veRtpRtcpPtr = VoERTP_RTCP::GetInterface(_vePtr);
-        _transportPtr = new MyTransport(_veNetworkPtr);
-        _externalMediaPtr = new MediaProcessImpl();
-        _rxVadObserverPtr = new RxCallback();
-    }
-
-    _veBasePtr->RegisterVoiceEngineObserver(*this);
-
-    std::string resource_path = webrtc::test::ProjectRootPath();
-    if (resource_path == webrtc::test::kCannotFindProjectRootDir) {
-        _long_audio_file_path = "./";
-    } else {
-        _long_audio_file_path = resource_path + "data\\voice_engine\\";
-    }
-}
-
-CWinTestDlg::~CWinTestDlg()
-{
-    if (_externalMediaPtr) delete _externalMediaPtr;
-    if (_transportPtr) delete _transportPtr;
-    if (_rxVadObserverPtr) delete _rxVadObserverPtr;
-
-    if (_veExternalMediaPtr) _veExternalMediaPtr->Release();
-    if (_veVideoSyncPtr) _veVideoSyncPtr->Release();
-    if (_veVolumeControlPtr) _veVolumeControlPtr->Release();
-
-    if (_veBasePtr) _veBasePtr->Terminate();
-    if (_veBasePtr) _veBasePtr->Release();
-
-    if (_veCodecPtr) _veCodecPtr->Release();
-    if (_veNetworkPtr) _veNetworkPtr->Release();
-    if (_veFilePtr) _veFilePtr->Release();
-    if (_veHardwarePtr) _veHardwarePtr->Release();
-    if (_veApmPtr) _veApmPtr->Release();
-    if (_veRtpRtcpPtr) _veRtpRtcpPtr->Release();
-    if (_vePtr)
-    {
-        VoiceEngine::Delete(_vePtr);
-    }
-    VoiceEngine::SetTraceFilter(kTraceNone);
-}
-
-void CWinTestDlg::DoDataExchange(CDataExchange* pDX)
-{
-    CDialog::DoDataExchange(pDX);
-    DDX_CBString(pDX, IDC_COMBO_IP_1, _strComboIp1);
-    DDX_CBString(pDX, IDC_COMBO_IP_2, _strComboIp2);
-}
-
-BEGIN_MESSAGE_MAP(CWinTestDlg, CDialog)
-    ON_WM_SYSCOMMAND()
-    ON_WM_PAINT()
-    ON_WM_QUERYDRAGICON()
-    ON_WM_TIMER()
-    //}}AFX_MSG_MAP
-    ON_BN_CLICKED(IDC_BUTTON_CREATE_1, &CWinTestDlg::OnBnClickedButtonCreate1)
-    ON_BN_CLICKED(IDC_BUTTON_DELETE_1, &CWinTestDlg::OnBnClickedButtonDelete1)
-    ON_BN_CLICKED(IDC_BUTTON_CREATE_2, &CWinTestDlg::OnBnClickedButtonCreate2)
-    ON_BN_CLICKED(IDC_BUTTON_DELETE_2, &CWinTestDlg::OnBnClickedButtonDelete2)
-    ON_CBN_SELCHANGE(IDC_COMBO_CODEC_1, &CWinTestDlg::OnCbnSelchangeComboCodec1)
-    ON_BN_CLICKED(IDC_BUTTON_START_LISTEN_1, &CWinTestDlg::OnBnClickedButtonStartListen1)
-    ON_BN_CLICKED(IDC_BUTTON_STOP_LISTEN_1, &CWinTestDlg::OnBnClickedButtonStopListen1)
-    ON_BN_CLICKED(IDC_BUTTON_START_PLAYOUT_1, &CWinTestDlg::OnBnClickedButtonStartPlayout1)
-    ON_BN_CLICKED(IDC_BUTTON_STOP_PLAYOUT_1, &CWinTestDlg::OnBnClickedButtonStopPlayout1)
-    ON_BN_CLICKED(IDC_BUTTON_START_SEND_1, &CWinTestDlg::OnBnClickedButtonStartSend1)
-    ON_BN_CLICKED(IDC_BUTTON_STOP_SEND_1, &CWinTestDlg::OnBnClickedButtonStopSend1)
-    ON_CBN_SELCHANGE(IDC_COMBO_IP_2, &CWinTestDlg::OnCbnSelchangeComboIp2)
-    ON_CBN_SELCHANGE(IDC_COMBO_IP_1, &CWinTestDlg::OnCbnSelchangeComboIp1)
-    ON_CBN_SELCHANGE(IDC_COMBO_CODEC_2, &CWinTestDlg::OnCbnSelchangeComboCodec2)
-    ON_BN_CLICKED(IDC_BUTTON_START_LISTEN_2, &CWinTestDlg::OnBnClickedButtonStartListen2)
-    ON_BN_CLICKED(IDC_BUTTON_STOP_LISTEN_2, &CWinTestDlg::OnBnClickedButtonStopListen2)
-    ON_BN_CLICKED(IDC_BUTTON_START_PLAYOUT_2, &CWinTestDlg::OnBnClickedButtonStartPlayout2)
-    ON_BN_CLICKED(IDC_BUTTON_STOP_PLAYOUT_2, &CWinTestDlg::OnBnClickedButtonStopPlayout2)
-    ON_BN_CLICKED(IDC_BUTTON_START_SEND_2, &CWinTestDlg::OnBnClickedButtonStartSend2)
-    ON_BN_CLICKED(IDC_BUTTON_STOP_SEND_2, &CWinTestDlg::OnBnClickedButtonStopSend2)
-    ON_BN_CLICKED(IDC_CHECK_EXT_TRANS_1, &CWinTestDlg::OnBnClickedCheckExtTrans1)
-    ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN_1, &CWinTestDlg::OnBnClickedCheckPlayFileIn1)
-    ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_OUT_1, &CWinTestDlg::OnBnClickedCheckPlayFileOut1)
-    ON_BN_CLICKED(IDC_CHECK_EXT_TRANS_2, &CWinTestDlg::OnBnClickedCheckExtTrans2)
-    ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN_2, &CWinTestDlg::OnBnClickedCheckPlayFileIn2)
-    ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_OUT_2, &CWinTestDlg::OnBnClickedCheckPlayFileOut2)
-    ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN, &CWinTestDlg::OnBnClickedCheckPlayFileIn)
-    ON_CBN_SELCHANGE(IDC_COMBO_REC_DEVICE, &CWinTestDlg::OnCbnSelchangeComboRecDevice)
-    ON_CBN_SELCHANGE(IDC_COMBO_PLAY_DEVICE, &CWinTestDlg::OnCbnSelchangeComboPlayDevice)
-    ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN_1, &CWinTestDlg::OnBnClickedCheckExtMediaIn1)
-    ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT_1, &CWinTestDlg::OnBnClickedCheckExtMediaOut1)
-    ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_INPUT_VOLUME, &CWinTestDlg::OnNMReleasedcaptureSliderInputVolume)
-    ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_OUTPUT_VOLUME, &CWinTestDlg::OnNMReleasedcaptureSliderOutputVolume)
-    ON_BN_CLICKED(IDC_CHECK_AGC, &CWinTestDlg::OnBnClickedCheckAgc)
-    ON_BN_CLICKED(IDC_CHECK_NS, &CWinTestDlg::OnBnClickedCheckNs)
-    ON_BN_CLICKED(IDC_CHECK_EC, &CWinTestDlg::OnBnClickedCheckEc)
-    ON_BN_CLICKED(IDC_CHECK_VAD_1, &CWinTestDlg::OnBnClickedCheckVad1)
-    ON_BN_CLICKED(IDC_CHECK_VAD_3, &CWinTestDlg::OnBnClickedCheckVad2)
-    ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN_2, &CWinTestDlg::OnBnClickedCheckExtMediaIn2)
-    ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT_2, &CWinTestDlg::OnBnClickedCheckExtMediaOut2)
-    ON_BN_CLICKED(IDC_CHECK_MUTE_IN, &CWinTestDlg::OnBnClickedCheckMuteIn)
-    ON_BN_CLICKED(IDC_CHECK_MUTE_IN_1, &CWinTestDlg::OnBnClickedCheckMuteIn1)
-    ON_BN_CLICKED(IDC_CHECK_MUTE_IN_2, &CWinTestDlg::OnBnClickedCheckMuteIn2)
-    ON_BN_CLICKED(IDC_CHECK_SRTP_TX_1, &CWinTestDlg::OnBnClickedCheckSrtpTx1)
-    ON_BN_CLICKED(IDC_CHECK_SRTP_RX_1, &CWinTestDlg::OnBnClickedCheckSrtpRx1)
-    ON_BN_CLICKED(IDC_CHECK_SRTP_TX_2, &CWinTestDlg::OnBnClickedCheckSrtpTx2)
-    ON_BN_CLICKED(IDC_CHECK_SRTP_RX_2, &CWinTestDlg::OnBnClickedCheckSrtpRx2)
-    ON_BN_CLICKED(IDC_CHECK_EXT_ENCRYPTION_1, &CWinTestDlg::OnBnClickedCheckExtEncryption1)
-    ON_BN_CLICKED(IDC_CHECK_EXT_ENCRYPTION_2, &CWinTestDlg::OnBnClickedCheckExtEncryption2)
-    ON_BN_CLICKED(IDC_BUTTON_DTMF_1, &CWinTestDlg::OnBnClickedButtonDtmf1)
-    ON_BN_CLICKED(IDC_CHECK_REC_MIC, &CWinTestDlg::OnBnClickedCheckRecMic)
-    ON_BN_CLICKED(IDC_BUTTON_DTMF_2, &CWinTestDlg::OnBnClickedButtonDtmf2)
-    ON_BN_CLICKED(IDC_BUTTON_TEST_1, &CWinTestDlg::OnBnClickedButtonTest1)
-    ON_BN_CLICKED(IDC_CHECK_CONFERENCE_1, &CWinTestDlg::OnBnClickedCheckConference1)
-    ON_BN_CLICKED(IDC_CHECK_CONFERENCE_2, &CWinTestDlg::OnBnClickedCheckConference2)
-    ON_BN_CLICKED(IDC_CHECK_ON_HOLD_1, &CWinTestDlg::OnBnClickedCheckOnHold1)
-    ON_BN_CLICKED(IDC_CHECK_ON_HOLD_2, &CWinTestDlg::OnBnClickedCheckOnHold2)
-    ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN, &CWinTestDlg::OnBnClickedCheckExtMediaIn)
-    ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT, &CWinTestDlg::OnBnClickedCheckExtMediaOut)
-    ON_LBN_SELCHANGE(IDC_LIST_CODEC_1, &CWinTestDlg::OnLbnSelchangeListCodec1)
-    ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_PAN_LEFT, &CWinTestDlg::OnNMReleasedcaptureSliderPanLeft)
-    ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_PAN_RIGHT, &CWinTestDlg::OnNMReleasedcaptureSliderPanRight)
-    ON_BN_CLICKED(IDC_BUTTON_VERSION, &CWinTestDlg::OnBnClickedButtonVersion)
-    ON_BN_CLICKED(IDC_CHECK_DELAY_ESTIMATE_1, &CWinTestDlg::OnBnClickedCheckDelayEstimate1)
-    ON_BN_CLICKED(IDC_CHECK_RXVAD, &CWinTestDlg::OnBnClickedCheckRxvad)
-    ON_BN_CLICKED(IDC_CHECK_AGC_1, &CWinTestDlg::OnBnClickedCheckAgc1)
-    ON_BN_CLICKED(IDC_CHECK_NS_1, &CWinTestDlg::OnBnClickedCheckNs1)
-    ON_BN_CLICKED(IDC_CHECK_REC_CALL, &CWinTestDlg::OnBnClickedCheckRecCall)
-    ON_BN_CLICKED(IDC_CHECK_TYPING_DETECTION, &CWinTestDlg::OnBnClickedCheckTypingDetection)
-    ON_BN_CLICKED(IDC_CHECK_RED, &CWinTestDlg::OnBnClickedCheckRED)
-    ON_BN_CLICKED(IDC_BUTTON_CLEAR_ERROR_CALLBACK, &CWinTestDlg::OnBnClickedButtonClearErrorCallback)
-END_MESSAGE_MAP()
-
-BOOL CWinTestDlg::UpdateTest(bool failed, const CString& strMsg)
-{
-    if (failed)
-    {
-        SetDlgItemText(IDC_EDIT_MESSAGE, strMsg);
-        _strErr.Format(_T("FAILED (error=%d)"), _veBasePtr->LastError());
-        SetDlgItemText(IDC_EDIT_RESULT, _strErr);
-        _failCount++;
-        SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount);
-        SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError());
-    }
-    else
-    {
-        SetDlgItemText(IDC_EDIT_MESSAGE, strMsg);
-        SetDlgItemText(IDC_EDIT_RESULT, _T("OK"));
-    }
-    return TRUE;
-}
-
-
-// CWinTestDlg message handlers
-
-BOOL CWinTestDlg::OnInitDialog()
-{
-    CDialog::OnInitDialog();
-
-    // Add "About..." menu item to system menu.
-
-    // IDM_ABOUTBOX must be in the system command range.
-    ASSERT((IDM_ABOUTBOX & 0xFFF0) == IDM_ABOUTBOX);
-    ASSERT(IDM_ABOUTBOX < 0xF000);
-
-    CMenu* pSysMenu = GetSystemMenu(FALSE);
-    if (pSysMenu != NULL)
-    {
-        CString strAboutMenu;
-        strAboutMenu.LoadString(IDS_ABOUTBOX);
-        if (!strAboutMenu.IsEmpty())
-        {
-            pSysMenu->AppendMenu(MF_SEPARATOR);
-            pSysMenu->AppendMenu(MF_STRING, IDM_ABOUTBOX, strAboutMenu);
-        }
-    }
-
-    // Set the icon for this dialog.  The framework does this automatically
-    //  when the application's main window is not a dialog
-    SetIcon(m_hIcon, TRUE);            // Set big icon
-    SetIcon(m_hIcon, FALSE);        // Set small icon
-
-    // char version[1024];
-    // _veBasePtr->GetVersion(version);
-    // AfxMessageBox(version, MB_OK);
-
-    if (_veBasePtr->Init() != 0)
-    {
-         AfxMessageBox(_T("Init() failed "), MB_OKCANCEL);
-    }
-
-    int ch = _veBasePtr->CreateChannel();
-    if (_veBasePtr->SetSendDestination(ch, 1234, "127.0.0.1") == -1)
-    {
-        if (_veBasePtr->LastError() == VE_EXTERNAL_TRANSPORT_ENABLED)
-        {
-            _strMsg.Format(_T("*** External transport build ***"));
-            SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg);
-            _externalTransportBuild = true;
-        }
-    }
-    _veBasePtr->DeleteChannel(ch);
-
-    // --- Add (preferred) local IPv4 address in title
-
-    if (_veNetworkPtr)
-    {
-        char localIP[64];
-        _veNetworkPtr->GetLocalIP(localIP);
-        CString str;
-        GetWindowText(str);
-        str.AppendFormat(_T("  [Local IPv4 address: %s]"), CharToTchar(localIP, 64));
-        SetWindowText(str);
-    }
-
-    // --- Volume sliders
-
-    if (_veVolumeControlPtr)
-    {
-        unsigned int volume(0);
-        CSliderCtrl* slider(NULL);
-
-        slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
-        slider->SetRangeMin(0);
-        slider->SetRangeMax(255);
-        _veVolumeControlPtr->GetMicVolume(volume);
-        slider->SetPos(volume);
-
-        slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
-        slider->SetRangeMin(0);
-        slider->SetRangeMax(255);
-        _veVolumeControlPtr->GetSpeakerVolume(volume);
-        slider->SetPos(volume);
-    }
-
-    // --- Panning sliders
-
-    if (_veVolumeControlPtr)
-    {
-        float lVol(0.0);
-        float rVol(0.0);
-        int leftVol, rightVol;
-        CSliderCtrl* slider(NULL);
-
-        _veVolumeControlPtr->GetOutputVolumePan(-1, lVol, rVol);
-
-        leftVol = (int)(lVol*10.0f);    // [0,10]
-        rightVol = (int)(rVol*10.0f);    // [0,10]
-
-        slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_LEFT);
-        slider->SetRange(0,10);
-        slider->SetPos(10-leftVol);        // pos 0 <=> max pan 1.0 (top of slider)
-
-        slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_RIGHT);
-        slider->SetRange(0,10);
-        slider->SetPos(10-rightVol);
-    }
-
-    // --- APM settings
-
-    bool enable(false);
-    CButton* button(NULL);
-
-    AgcModes agcMode(kAgcDefault);
-    if (_veApmPtr->GetAgcStatus(enable, agcMode) == 0)
-    {
-        button = (CButton*)GetDlgItem(IDC_CHECK_AGC);
-        enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
-    }
-    else
-    {
-        // AGC is not supported
-        GetDlgItem(IDC_CHECK_AGC)->EnableWindow(FALSE);
-    }
-
-    NsModes nsMode(kNsDefault);
-    if (_veApmPtr->GetNsStatus(enable, nsMode) == 0)
-    {
-        button = (CButton*)GetDlgItem(IDC_CHECK_NS);
-        enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
-    }
-    else
-    {
-        // NS is not supported
-        GetDlgItem(IDC_CHECK_NS)->EnableWindow(FALSE);
-    }
-
-    EcModes ecMode(kEcDefault);
-    if (_veApmPtr->GetEcStatus(enable, ecMode) == 0)
-    {
-        button = (CButton*)GetDlgItem(IDC_CHECK_EC);
-        enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
-    }
-    else
-    {
-        // EC is not supported
-        GetDlgItem(IDC_CHECK_EC)->EnableWindow(FALSE);
-    }
-
-    // --- First channel section
-
-    GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_CONFERENCE_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_RED)->EnableWindow(FALSE);
-
-    CComboBox* comboIP(NULL);
-    comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_1);
-    comboIP->AddString(_T("127.0.0.1"));
-    comboIP->SetCurSel(0);
-
-    SetDlgItemInt(IDC_EDIT_TX_PORT_1, 1111);
-    SetDlgItemInt(IDC_EDIT_RX_PORT_1, 1111);
-
-    // --- Add supported codecs to the codec combo box
-
-    CComboBox* comboCodec(NULL);
-    comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_1);
-    comboCodec->ResetContent();
-
-    int numCodecs = _veCodecPtr->NumOfCodecs();
-    for (int idx = 0; idx < numCodecs; idx++)
-    {
-        CodecInst codec;
-        _veCodecPtr->GetCodec(idx, codec);
-        if ((_stricmp(codec.plname, "CNNB") != 0) &&
-            (_stricmp(codec.plname, "CNWB") != 0))
-        {
-            CString strCodec;
-            if (_stricmp(codec.plname, "G7221") == 0)
-                strCodec.Format(_T("%s (%d/%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000, codec.rate/1000);
-            else
-                strCodec.Format(_T("%s (%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000);
-            comboCodec->AddString(strCodec);
-        }
-        if (idx == 0)
-        {
-            SetDlgItemInt(IDC_EDIT_CODEC_1, codec.pltype);
-        }
-    }
-    comboCodec->SetCurSel(0);
-
-    CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
-    list->AddString(_T("pltype"));
-    list->AddString(_T("plfreq"));
-    list->AddString(_T("pacsize"));
-    list->AddString(_T("channels"));
-    list->AddString(_T("rate"));
-    list->SetCurSel(0);
-
-    // --- Add available audio devices to the combo boxes
-
-    CComboBox* comboRecDevice(NULL);
-    CComboBox* comboPlayDevice(NULL);
-    comboRecDevice = (CComboBox*)GetDlgItem(IDC_COMBO_REC_DEVICE);
-    comboPlayDevice = (CComboBox*)GetDlgItem(IDC_COMBO_PLAY_DEVICE);
-    comboRecDevice->ResetContent();
-    comboPlayDevice->ResetContent();
-
-    if (_veHardwarePtr)
-    {
-        int numPlayout(0);
-        int numRecording(0);
-        char nameStr[128];
-        char guidStr[128];
-        CString strDevice;
-        AudioLayers audioLayer;
-
-        _veHardwarePtr->GetAudioDeviceLayer(audioLayer);
-        if (kAudioWindowsWave == audioLayer)
-        {
-            strDevice.FormatMessage(_T("Audio Layer: Windows Wave API"));
-        }
-        else if (kAudioWindowsCore == audioLayer)
-        {
-            strDevice.FormatMessage(_T("Audio Layer: Windows Core API"));
-        }
-        else
-        {
-            strDevice.FormatMessage(_T("Audio Layer: ** UNKNOWN **"));
-        }
-        SetDlgItemText(IDC_EDIT_AUDIO_LAYER, (LPCTSTR)strDevice);
-
-        _veHardwarePtr->GetNumOfRecordingDevices(numRecording);
-
-        for (int idx = 0; idx < numRecording; idx++)
-        {
-            _veHardwarePtr->GetRecordingDeviceName(idx, nameStr, guidStr);
-      strDevice.Format(_T("%s"), CharToTchar(nameStr, 128));
-            comboRecDevice->AddString(strDevice);
-        }
-        // Select default (communication) device in the combo box
-        _veHardwarePtr->GetRecordingDeviceName(-1, nameStr, guidStr);
-    CString tmp = CString(nameStr);
-        int nIndex = comboRecDevice->SelectString(-1, tmp);
-        ASSERT(nIndex != CB_ERR);
-
-        _veHardwarePtr->GetNumOfPlayoutDevices(numPlayout);
-
-        for (int idx = 0; idx < numPlayout; idx++)
-        {
-            _veHardwarePtr->GetPlayoutDeviceName(idx, nameStr, guidStr);
-      strDevice.Format(_T("%s"), CharToTchar(nameStr, 128));
-            comboPlayDevice->AddString(strDevice);
-        }
-        // Select default (communication) device in the combo box
-        _veHardwarePtr->GetPlayoutDeviceName(-1, nameStr, guidStr);
-        nIndex = comboPlayDevice->SelectString(-1, CString(nameStr));
-        ASSERT(nIndex != CB_ERR);
-    }
-
-    // --- Second channel section
-
-    GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(FALSE);
-    GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(FALSE);
-
-    comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_2);
-    comboIP->AddString(_T("127.0.0.1"));
-    comboIP->SetCurSel(0);
-
-    SetDlgItemInt(IDC_EDIT_TX_PORT_2, 2222);
-    SetDlgItemInt(IDC_EDIT_RX_PORT_2, 2222);
-
-    comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_2);
-    comboCodec->ResetContent();
-
-    if (_veCodecPtr)
-    {
-        numCodecs = _veCodecPtr->NumOfCodecs();
-        for (int idx = 0; idx < numCodecs; idx++)
-        {
-            CodecInst codec;
-            _veCodecPtr->GetCodec(idx, codec);
-            CString strCodec;
-            strCodec.Format(_T("%s (%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000);
-            comboCodec->AddString(strCodec);
-        }
-        comboCodec->SetCurSel(0);
-    }
-
-    // --- Start windows timer
-
-    SetTimer(0, 1000, NULL);
-
-    return TRUE;  // return TRUE  unless you set the focus to a control
-}
-
-void CWinTestDlg::OnSysCommand(UINT nID, LPARAM lParam)
-{
-    if ((nID & 0xFFF0) == IDM_ABOUTBOX)
-    {
-        CAboutDlg dlgAbout;
-        dlgAbout.DoModal();
-    }
-    else if (nID == SC_CLOSE)
-    {
-        BOOL ret;
-        int channel(0);
-        channel = GetDlgItemInt(IDC_EDIT_1, &ret);
-        if (ret == TRUE)
-        {
-            _veBasePtr->DeleteChannel(channel);
-        }
-        channel = GetDlgItemInt(IDC_EDIT_2, &ret);
-        if (ret == TRUE)
-        {
-            _veBasePtr->DeleteChannel(channel);
-        }
-
-        CDialog::OnSysCommand(nID, lParam);
-    }
-    else
-    {
-        CDialog::OnSysCommand(nID, lParam);
-    }
-
-}
-
-// If you add a minimize button to your dialog, you will need the code below
-//  to draw the icon.  For MFC applications using the document/view model,
-//  this is automatically done for you by the framework.
-
-void CWinTestDlg::OnPaint()
-{
-    if (IsIconic())
-    {
-        CPaintDC dc(this); // device context for painting
-
-        SendMessage(WM_ICONERASEBKGND, reinterpret_cast<WPARAM>(dc.GetSafeHdc()), 0);
-
-        // Center icon in client rectangle
-        int cxIcon = GetSystemMetrics(SM_CXICON);
-        int cyIcon = GetSystemMetrics(SM_CYICON);
-        CRect rect;
-        GetClientRect(&rect);
-        int x = (rect.Width() - cxIcon + 1) / 2;
-        int y = (rect.Height() - cyIcon + 1) / 2;
-
-        // Draw the icon
-        dc.DrawIcon(x, y, m_hIcon);
-    }
-    else
-    {
-        CDialog::OnPaint();
-    }
-}
-
-// The system calls this function to obtain the cursor to display while the user drags
-//  the minimized window.
-HCURSOR CWinTestDlg::OnQueryDragIcon()
-{
-    return static_cast<HCURSOR>(m_hIcon);
-}
-
-
-void CWinTestDlg::OnBnClickedButtonCreate1()
-{
-    int channel(0);
-    TEST((channel = _veBasePtr->CreateChannel()) >= 0, _T("CreateChannel(channel=%d)"), channel);
-    if (channel >= 0)
-    {
-        _veRtpRtcpPtr->RegisterRTPObserver(channel, *this);
-
-        SetDlgItemInt(IDC_EDIT_1, channel);
-        GetDlgItem(IDC_BUTTON_CREATE_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_RED)->EnableWindow(TRUE);
-
-        // Always set send codec to default codec <=> index 0.
-        CodecInst codec;
-        _veCodecPtr->GetCodec(0, codec);
-        _veCodecPtr->SetSendCodec(channel, codec);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonCreate2()
-{
-    int channel(0);
-    TEST((channel = _veBasePtr->CreateChannel()) >=0 , _T("CreateChannel(%d)"), channel);
-    if (channel >= 0)
-    {
-        _veRtpRtcpPtr->RegisterRTPObserver(channel, *this);
-
-        SetDlgItemInt(IDC_EDIT_2, channel);
-        GetDlgItem(IDC_BUTTON_CREATE_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(TRUE);
-
-        // Always set send codec to default codec <=> index 0.
-        CodecInst codec;
-        _veCodecPtr->GetCodec(0, codec);
-        _veCodecPtr->SetSendCodec(channel, codec);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonDelete1()
-{
-    BOOL ret;
-    int channel = GetDlgItemInt(IDC_EDIT_1, &ret);
-    if (ret == TRUE)
-    {
-        _delayEstimate1 = false;
-        _rxVad = false;
-        _veRtpRtcpPtr->DeRegisterRTPObserver(channel);
-        TEST(_veBasePtr->DeleteChannel(channel) == 0, _T("DeleteChannel(channel=%d)"), channel);
-        SetDlgItemText(IDC_EDIT_1, _T(""));
-        GetDlgItem(IDC_BUTTON_CREATE_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_CONFERENCE_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_RED)->EnableWindow(FALSE);
-        SetDlgItemText(IDC_EDIT_RXVAD, _T(""));
-        GetDlgItem(IDC_EDIT_RXVAD)->EnableWindow(FALSE);
-        CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_VAD_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_CONFERENCE_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_AGC_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_NS_1);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_RXVAD);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_RED);
-        button->SetCheck(BST_UNCHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonDelete2()
-{
-    BOOL ret;
-    int channel = GetDlgItemInt(IDC_EDIT_2, &ret);
-    if (ret == TRUE)
-    {
-        _delayEstimate2 = false;
-        _veRtpRtcpPtr->DeRegisterRTPObserver(channel);
-        TEST(_veBasePtr->DeleteChannel(channel) == 0, _T("DeleteChannel(%d)"), channel);
-        SetDlgItemText(IDC_EDIT_2, _T(""));
-        GetDlgItem(IDC_BUTTON_CREATE_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(FALSE);
-        CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_VAD_3);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_CONFERENCE_2);
-        button->SetCheck(BST_UNCHECKED);
-        button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_2);
-        button->SetCheck(BST_UNCHECKED);
-    }
-}
-
-void CWinTestDlg::OnCbnSelchangeComboIp1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CString str;
-    int port = GetDlgItemInt(IDC_EDIT_TX_PORT_1);
-    CComboBox* comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_1);
-    int n = comboIP->GetLBTextLen(0);
-    comboIP->GetLBText(0, str.GetBuffer(n));
-    TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(str.GetBuffer(n), -1)) == 0,
-        _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, str.GetBuffer(n));
-    str.ReleaseBuffer();
-}
-
-void CWinTestDlg::OnCbnSelchangeComboIp2()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CString str;
-    int port = GetDlgItemInt(IDC_EDIT_TX_PORT_2);
-    CComboBox* comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_2);
-    int n = comboIP->GetLBTextLen(0);
-    comboIP->GetLBText(0, str.GetBuffer(n));
-    TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(str.GetBuffer(n), -1)) == 0,
-        _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, str.GetBuffer(n));
-    str.ReleaseBuffer();
-}
-
-void CWinTestDlg::OnCbnSelchangeComboCodec1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-
-    CodecInst codec;
-    CComboBox* comboCodec(NULL);
-    comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_1);
-    int index = comboCodec->GetCurSel();
-    _veCodecPtr->GetCodec(index, codec);
-    if (strncmp(codec.plname, "ISAC", 4) == 0)
-    {
-        // Set iSAC to adaptive mode by default.
-        codec.rate = -1;
-    }
-    TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
-        _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
-        channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
-
-    CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
-    list->SetCurSel(0);
-    SetDlgItemInt(IDC_EDIT_CODEC_1, codec.pltype);
-}
-
-void CWinTestDlg::OnLbnSelchangeListCodec1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-
-    CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
-    int listIdx = list->GetCurSel();
-    if (listIdx < 0)
-        return;
-    CString str;
-    list->GetText(listIdx, str);
-
-    CodecInst codec;
-    _veCodecPtr->GetSendCodec(channel, codec);
-
-    int value = GetDlgItemInt(IDC_EDIT_CODEC_1);
-    if (str == _T("pltype"))
-    {
-        codec.pltype = value;
-    }
-    else if (str == _T("plfreq"))
-    {
-        codec.plfreq = value;
-    }
-    else if (str == _T("pacsize"))
-    {
-        codec.pacsize = value;
-    }
-    else if (str == _T("channels"))
-    {
-        codec.channels = value;
-    }
-    else if (str == _T("rate"))
-    {
-        codec.rate = value;
-    }
-    TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
-        _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
-        channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
-}
-
-void CWinTestDlg::OnCbnSelchangeComboCodec2()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-
-    CodecInst codec;
-    CComboBox* comboCodec(NULL);
-    comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_2);
-    int index = comboCodec->GetCurSel();
-    _veCodecPtr->GetCodec(index, codec);
-    TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
-        _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
-        channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
-}
-
-void CWinTestDlg::OnBnClickedButtonStartListen1()
-{
-    int ret1(0);
-    int ret2(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    int port = GetDlgItemInt(IDC_EDIT_RX_PORT_1);
-    TEST((ret1 = _veBasePtr->SetLocalReceiver(channel, port)) == 0, _T("SetLocalReceiver(channel=%d, port=%d)"), channel, port);
-    TEST((ret2 = _veBasePtr->StartReceive(channel)) == 0, _T("StartReceive(channel=%d)"), channel);
-    if (ret1 == 0 && ret2 == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(TRUE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartListen2()
-{
-    int ret1(0);
-    int ret2(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    int port = GetDlgItemInt(IDC_EDIT_RX_PORT_2);
-    TEST((ret1 = _veBasePtr->SetLocalReceiver(channel, port)) == 0, _T("SetLocalReceiver(channel=%d, port=%d)"), channel, port);
-    TEST((ret2 = _veBasePtr->StartReceive(channel)) == 0, _T("StartReceive(channel=%d)"), channel);
-    if (ret1 == 0 && ret2 == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(TRUE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopListen1()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    TEST((ret = _veBasePtr->StopReceive(channel)) == 0, _T("StopListen(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopListen2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    TEST((ret = _veBasePtr->StopReceive(channel)) == 0, _T("StopListen(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartPlayout1()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    TEST((ret = _veBasePtr->StartPlayout(channel)) == 0, _T("StartPlayout(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(TRUE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartPlayout2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    TEST((ret = _veBasePtr->StartPlayout(channel)) == 0, _T("StartPlayout(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(TRUE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopPlayout1()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    TEST((ret = _veBasePtr->StopPlayout(channel)) == 0, _T("StopPlayout(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopPlayout2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    TEST((ret = _veBasePtr->StopPlayout(channel)) == 0, _T("StopPlayout(channel=%d)"));
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartSend1()
-{
-    UpdateData(TRUE);  // update IP address
-
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    if (!_externalTransport)
-    {
-        CString str;
-        int port = GetDlgItemInt(IDC_EDIT_TX_PORT_1);
-    TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(_strComboIp1.GetBuffer(7), -1)) == 0,
-      _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, _strComboIp1.GetBuffer(7));
-        str.ReleaseBuffer();
-    }
-
-	//_veVideoSyncPtr->SetInitTimestamp(0,0);
-    // OnCbnSelchangeComboCodec1();
-
-    TEST((ret = _veBasePtr->StartSend(channel)) == 0, _T("StartSend(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(TRUE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartSend2()
-{
-    UpdateData(TRUE);  // update IP address
-
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    if (!_externalTransport)
-    {
-        CString str;
-        int port = GetDlgItemInt(IDC_EDIT_TX_PORT_2);
-        TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(_strComboIp2.GetBuffer(7), -1)) == 0,
-            _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, _strComboIp2.GetBuffer(7));
-        str.ReleaseBuffer();
-    }
-
-    // OnCbnSelchangeComboCodec2();
-
-    TEST((ret = _veBasePtr->StartSend(channel)) == 0, _T("StartSend(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
-        GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(TRUE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopSend1()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    TEST((ret = _veBasePtr->StopSend(channel)) == 0, _T("StopSend(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopSend2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    TEST((ret = _veBasePtr->StopSend(channel)) == 0, _T("StopSend(channel=%d)"), channel);
-    if (ret == 0)
-    {
-        GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(TRUE);
-        GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtTrans1()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST((ret = _veNetworkPtr->RegisterExternalTransport(channel, *_transportPtr)) == 0,
-            _T("RegisterExternalTransport(channel=%d, transport=0x%x)"), channel, _transportPtr);
-    }
-    else
-    {
-        TEST((ret = _veNetworkPtr->DeRegisterExternalTransport(channel)) == 0,
-            _T("DeRegisterExternalTransport(channel=%d)"), channel);
-    }
-    if (ret == 0)
-    {
-        _externalTransport = enable;
-    }
-    else
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtTrans2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_2);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST((ret = _veNetworkPtr->RegisterExternalTransport(channel, *_transportPtr)) == 0,
-            _T("RegisterExternalTransport(channel=%d, transport=0x%x)"), channel, _transportPtr);
-    }
-    else
-    {
-        TEST((ret = _veNetworkPtr->DeRegisterExternalTransport(channel)) == 0,
-            _T("DeRegisterExternalTransport(channel=%d)"), channel);
-    }
-    if (ret == 0)
-    {
-        _externalTransport = enable;
-    }
-    else
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileIn1()
-{
-    std::string micFile = _long_audio_file_path + "audio_short16.pcm";
-
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        bool mix;
-        const bool loop(true);
-        const FileFormats format = kFileFormatPcm16kHzFile;
-        const float scale(1.0);
-
-        (_checkPlayFileIn1 %2 == 0) ? mix = true : mix = false;
-        TEST((ret = _veFilePtr->StartPlayingFileAsMicrophone(channel,
-            micFile.c_str(), loop, mix, format, scale) == 0),
-            _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, ")
-            _T("mix=%d, format=%d, scale=%2.1f)"),
-            channel, CharToTchar(micFile.c_str(), -1),
-            loop, mix, format, scale);
-        _checkPlayFileIn1++;
-    }
-    else
-    {
-        TEST((ret = _veFilePtr->StopPlayingFileAsMicrophone(channel) == 0),
-            _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
-    }
-    if (ret == -1)
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileIn2()
-{
-    std::string micFile = _long_audio_file_path + "audio_long16.pcm";
-
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        bool mix;
-        const bool loop(true);
-        const FileFormats format = kFileFormatPcm16kHzFile;
-        const float scale(1.0);
-
-        (_checkPlayFileIn2 %2 == 0) ? mix = true : mix = false;
-        TEST((ret = _veFilePtr->StartPlayingFileAsMicrophone(channel,
-            micFile.c_str(), loop, mix, format, scale) == 0),
-            _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, ")
-            _T("mix=%d, format=%d, scale=%2.1f)"),
-            channel, CharToTchar(micFile.c_str(), -1),
-            loop, mix, format, scale);
-        _checkPlayFileIn2++;
-    }
-    else
-    {
-        TEST((ret = _veFilePtr->StopPlayingFileAsMicrophone(channel) == 0),
-            _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
-    }
-    if (ret == -1)
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileOut1()
-{
-    const FileFormats formats[8]  = {{kFileFormatPcm16kHzFile},
-                                          {kFileFormatWavFile},
-                                          {kFileFormatWavFile},
-                                          {kFileFormatWavFile},
-                                          {kFileFormatWavFile},
-                                          {kFileFormatWavFile},
-                                          {kFileFormatWavFile},
-                                          {kFileFormatWavFile}};
-    // File path is relative to the location of 'voice_engine.gyp'.
-    const char spkrFiles[8][64] = {{"audio_short16.pcm"},
-                                   {"audio_tiny8.wav"},
-                                   {"audio_tiny11.wav"},
-                                   {"audio_tiny16.wav"},
-                                   {"audio_tiny22.wav"},
-                                   {"audio_tiny32.wav"},
-                                   {"audio_tiny44.wav"},
-                                   {"audio_tiny48.wav"}};
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        const bool loop(true);
-        const float volumeScaling(1.0);
-        const int startPointMs(0);
-        const int stopPointMs(0);
-        const FileFormats format = formats[_checkPlayFileOut1 % 8];
-        std::string spkrFile = _long_audio_file_path +
-                               spkrFiles[_checkPlayFileOut1 % 8];
-
-        CString str;
-        if (_checkPlayFileOut1 % 8 == 0)
-        {
-            str = _T("kFileFormatPcm16kHzFile");
-        }
-        else
-        {
-            str = _T("kFileFormatWavFile");
-        }
-        // (_checkPlayFileOut1 %2 == 0) ? mix = true : mix = false;
-        TEST((ret = _veFilePtr->StartPlayingFileLocally(channel,
-            spkrFile.c_str(), loop, format, volumeScaling,
-            startPointMs,stopPointMs) == 0),
-            _T("StartPlayingFileLocally(channel=%d, file=%s, loop=%d, ")
-            _T("format=%s, scale=%2.1f, start=%d, stop=%d)"),
-            channel, CharToTchar(spkrFile.c_str(), -1),
-            loop, str, volumeScaling, startPointMs, stopPointMs);
-        _checkPlayFileOut1++;
-    }
-    else
-    {
-        TEST((ret = _veFilePtr->StopPlayingFileLocally(channel) == 0),
-            _T("StopPlayingFileLocally(channel=%d)"), channel);
-    }
-    if (ret == -1)
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileOut2()
-{
-    std::string spkrFile = _long_audio_file_path + "audio_long16.pcm";
-
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        const bool loop(true);
-        const FileFormats format = kFileFormatPcm16kHzFile;
-        const float volumeScaling(1.0);
-        const int startPointMs(0);
-        const int stopPointMs(0);
-
-        // (_checkPlayFileOut2 %2 == 0) ? mix = true : mix = false;
-        TEST((ret = _veFilePtr->StartPlayingFileLocally(channel,
-            spkrFile.c_str(), loop, format, volumeScaling,
-            startPointMs,stopPointMs) == 0),
-            _T("StartPlayingFileLocally(channel=%d, file=%s, loop=%d, ")
-            _T("format=%d, scale=%2.1f, start=%d, stop=%d)"),
-            channel, CharToTchar(spkrFile.c_str(), -1),
-            loop, format, volumeScaling, startPointMs, stopPointMs);
-        // _checkPlayFileIn2++;
-    }
-    else
-    {
-        TEST((ret = _veFilePtr->StopPlayingFileLocally(channel) == 0),
-            _T("StopPlayingFileLocally(channel=%d)"), channel);
-    }
-    if (ret == -1)
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaIn1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1);
-    int check = buttonExtTrans->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingPerChannel, *_externalMediaPtr) == 0,
-            _T("RegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
-    }
-    else
-    {
-        TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingPerChannel) == 0,
-            _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel)"), channel);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaIn2()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2);
-    int check = buttonExtTrans->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingPerChannel, *_externalMediaPtr) == 0,
-            _T("RegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
-    }
-    else
-    {
-        TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingPerChannel) == 0,
-            _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel)"), channel);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaOut1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1);
-    int check = buttonExtTrans->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackPerChannel, *_externalMediaPtr) == 0,
-            _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
-    }
-    else
-    {
-        TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackPerChannel) == 0,
-            _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel)"), channel);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaOut2()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2);
-    int check = buttonExtTrans->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackPerChannel, *_externalMediaPtr) == 0,
-            _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
-    }
-    else
-    {
-        TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackPerChannel) == 0,
-            _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel)"), channel);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckVad1()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_VAD_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        VadModes mode(kVadConventional);
-        if (_checkVAD1 % 4 == 0)
-        {
-            mode = kVadConventional;
-            str = _T("kVadConventional");
-        }
-        else if (_checkVAD1 % 4 == 1)
-        {
-            mode = kVadAggressiveLow;
-            str = _T("kVadAggressiveLow");
-        }
-        else if (_checkVAD1 % 4 == 2)
-        {
-            mode = kVadAggressiveMid;
-            str = _T("kVadAggressiveMid");
-        }
-        else if (_checkVAD1 % 4 == 3)
-        {
-            mode = kVadAggressiveHigh;
-            str = _T("kVadAggressiveHigh");
-        }
-        const bool disableDTX(false);
-        TEST((ret = _veCodecPtr->SetVADStatus(channel, true, mode, disableDTX) == 0),
-            _T("SetVADStatus(channel=%d, enable=%d, mode=%s, disableDTX=%d)"), channel, enable, str, disableDTX);
-        _checkVAD1++;
-    }
-    else
-    {
-        TEST((ret = _veCodecPtr->SetVADStatus(channel, false)) == 0, _T("SetVADStatus(channel=%d, enable=%d)"), channel, false);
-    }
-    if (ret == -1)
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckVad2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_VAD_2);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        VadModes mode(kVadConventional);
-        if (_checkVAD2 % 4 == 0)
-        {
-            mode = kVadConventional;
-            str = _T("kVadConventional");
-        }
-        else if (_checkVAD2 % 4 == 1)
-        {
-            mode = kVadAggressiveLow;
-            str = _T("kVadAggressiveLow");
-        }
-        else if (_checkVAD2 % 4 == 2)
-        {
-            mode = kVadAggressiveMid;
-            str = _T("kVadAggressiveMid");
-        }
-        else if (_checkVAD2 % 4 == 3)
-        {
-            mode = kVadAggressiveHigh;
-            str = _T("kVadAggressiveHigh");
-        }
-        const bool disableDTX(false);
-        TEST((ret = _veCodecPtr->SetVADStatus(channel, true, mode, disableDTX)) == 0,
-            _T("SetVADStatus(channel=%d, enable=%d, mode=%s, disableDTX=%d)"), channel, enable, str, disableDTX);
-        _checkVAD2++;
-    }
-    else
-    {
-        TEST((ret = _veCodecPtr->SetVADStatus(channel, false) == 0), _T("SetVADStatus(channel=%d, enable=%d)"), channel, false);
-    }
-    if (ret == -1)
-    {
-        // restore inital state since API call failed
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckMuteIn1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_1);
-    int check = buttonMute->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
-        _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckMuteIn2()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_2);
-    int check = buttonMute->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
-        _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpTx1()
-{
-    TEST(true, "Built-in SRTP support is deprecated.");
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpTx2()
-{
-    TEST(true, "Built-in SRTP support is deprecated.");
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpRx1()
-{
-    TEST(true, "Built-in SRTP support is deprecated.");
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpRx2()
-{
-    TEST(true, "Built-in SRTP support is deprecated.");
-}
-
-void CWinTestDlg::OnBnClickedCheckExtEncryption1()
-{
-    TEST(true, "External Encryption has been removed from the API!");
-}
-
-void CWinTestDlg::OnBnClickedCheckExtEncryption2()
-{
-    TEST(true, "External Encryption has been removed from the API!");
-}
-
-void CWinTestDlg::OnBnClickedButtonDtmf1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CTelephonyEvent dlgTelephoneEvent(_vePtr, channel, this);
-    dlgTelephoneEvent.DoModal();
-}
-
-void CWinTestDlg::OnBnClickedButtonDtmf2()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CTelephonyEvent dlgTelephoneEvent(_vePtr, channel, this);
-    dlgTelephoneEvent.DoModal();
-}
-
-void CWinTestDlg::OnBnClickedCheckConference1()
-{
-    // Not supported yet
-}
-
-void CWinTestDlg::OnBnClickedCheckConference2()
-{
-   // Not supported yet
-}
-
-void CWinTestDlg::OnBnClickedCheckOnHold1()
-{
-    SHORT shiftKeyIsPressed = ::GetAsyncKeyState(VK_SHIFT);
-
-    CString str;
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_1);
-    int check = button->GetCheck();
-
-    if (shiftKeyIsPressed)
-    {
-        bool enabled(false);
-        OnHoldModes mode(kHoldSendAndPlay);
-        TEST(_veBasePtr->GetOnHoldStatus(channel, enabled, mode) == 0,
-            _T("GetOnHoldStatus(channel=%d, enabled=?, mode=?)"), channel);
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-
-        switch (mode)
-        {
-        case kHoldSendAndPlay:
-            str = _T("kHoldSendAndPlay");
-            break;
-        case kHoldSendOnly:
-            str = _T("kHoldSendOnly");
-            break;
-        case kHoldPlayOnly:
-            str = _T("kHoldPlayOnly");
-            break;
-        default:
-            break;
-        }
-        PRINT_GET_RESULT(_T("enabled=%d, mode=%s"), enabled, str);
-        return;
-    }
-
-    int ret(0);
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        OnHoldModes mode(kHoldSendAndPlay);
-        if (_checkOnHold1 % 3 == 0)
-        {
-            mode = kHoldSendAndPlay;
-            str = _T("kHoldSendAndPlay");
-        }
-        else if (_checkOnHold1 % 3 == 1)
-        {
-            mode = kHoldSendOnly;
-            str = _T("kHoldSendOnly");
-        }
-        else if (_checkOnHold1 % 3 == 2)
-        {
-            mode = kHoldPlayOnly;
-            str = _T("kHoldPlayOnly");
-        }
-        TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable, mode)) == 0,
-            _T("SetOnHoldStatus(channel=%d, enable=%d, mode=%s)"), channel, enable, str);
-        _checkOnHold1++;
-    }
-    else
-    {
-        TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable)) == 0,
-            _T("SetOnHoldStatus(channel=%d, enable=%d)"), channel, enable);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckOnHold2()
-{
-    int ret(0);
-    int channel = GetDlgItemInt(IDC_EDIT_2);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_2);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        OnHoldModes mode(kHoldSendAndPlay);
-        if (_checkOnHold1 % 3 == 0)
-        {
-            mode = kHoldSendAndPlay;
-            str = _T("kHoldSendAndPlay");
-        }
-        else if (_checkOnHold1 % 3 == 1)
-        {
-            mode = kHoldSendOnly;
-            str = _T("kHoldSendOnly");
-        }
-        else if (_checkOnHold1 % 3 == 2)
-        {
-            mode = kHoldPlayOnly;
-            str = _T("kHoldPlayOnly");
-        }
-        TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable, mode)) == 0,
-            _T("SetOnHoldStatus(channel=%d, enable=%d, mode=%s)"), channel, enable, str);
-        _checkOnHold1++;
-    }
-    else
-    {
-        TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable)) == 0,
-            _T("SetOnHoldStatus(channel=%d, enable=%d)"), channel, enable);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckDelayEstimate1()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-
-    if (enable)
-    {
-        _delayEstimate1 = true;
-        SetDlgItemInt(IDC_EDIT_DELAY_ESTIMATE_1, 0);
-    }
-    else
-    {
-        _delayEstimate1 = false;
-        SetDlgItemText(IDC_EDIT_DELAY_ESTIMATE_1, _T(""));
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckRxvad()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_RXVAD);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-
-    if (enable)
-    {
-        _rxVad = true;
-        _veApmPtr->RegisterRxVadObserver(channel, *_rxVadObserverPtr);
-        SetDlgItemInt(IDC_EDIT_RXVAD, 0);
-    }
-    else
-    {
-        _rxVad = false;
-        _veApmPtr->DeRegisterRxVadObserver(channel);
-        SetDlgItemText(IDC_EDIT_RXVAD, _T(""));
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckAgc1()
-{
-    SHORT shiftKeyIsPressed = ::GetAsyncKeyState(VK_SHIFT);
-
-    CString str;
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_AGC_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-
-    if (shiftKeyIsPressed)
-    {
-        bool enabled(false);
-        AgcModes mode(kAgcAdaptiveDigital);
-        TEST(_veApmPtr->GetRxAgcStatus(channel, enabled, mode) == 0,
-            _T("GetRxAgcStatus(channel=%d, enabled=?, mode=?)"), channel);
-        button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-
-        switch (mode)
-        {
-        case kAgcAdaptiveAnalog:
-            str = _T("kAgcAdaptiveAnalog");
-            break;
-        case kAgcAdaptiveDigital:
-            str = _T("kAgcAdaptiveDigital");
-            break;
-        case kAgcFixedDigital:
-            str = _T("kAgcFixedDigital");
-            break;
-        default:
-            break;
-        }
-        PRINT_GET_RESULT(_T("enabled=%d, mode=%s"), enabled, str);
-        return;
-    }
-
-    if (enable)
-    {
-        CString str;
-        AgcModes mode(kAgcDefault);
-        if (_checkAGC1 % 3 == 0)
-        {
-            mode = kAgcDefault;
-            str = _T("kAgcDefault");
-        }
-        else if (_checkAGC1 % 3 == 1)
-        {
-            mode = kAgcAdaptiveDigital;
-            str = _T("kAgcAdaptiveDigital");
-        }
-        else if (_checkAGC1 % 3 == 2)
-        {
-            mode = kAgcFixedDigital;
-            str = _T("kAgcFixedDigital");
-        }
-        TEST(_veApmPtr->SetRxAgcStatus(channel, true, mode) == 0, _T("SetRxAgcStatus(channel=%d, enable=%d, %s)"), channel, enable, str);
-        _checkAGC1++;
-    }
-    else
-    {
-        TEST(_veApmPtr->SetRxAgcStatus(channel, false, kAgcUnchanged) == 0, _T("SetRxAgcStatus(channel=%d, enable=%d)"), channel, enable);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckNs1()
-{
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    CButton* buttonNS = (CButton*)GetDlgItem(IDC_CHECK_NS_1);
-    int check = buttonNS->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        NsModes mode(kNsDefault);
-        if (_checkNS1 % 6 == 0)
-        {
-            mode = kNsDefault;
-            str = _T("kNsDefault");
-        }
-        else if (_checkNS1 % 6 == 1)
-        {
-            mode = kNsConference;
-            str = _T("kNsConference");
-        }
-        else if (_checkNS1 % 6 == 2)
-        {
-            mode = kNsLowSuppression;
-            str = _T("kNsLowSuppression");
-        }
-        else if (_checkNS1 % 6 == 3)
-        {
-            mode = kNsModerateSuppression;
-            str = _T("kNsModerateSuppression");
-        }
-        else if (_checkNS1 % 6 == 4)
-        {
-            mode = kNsHighSuppression;
-            str = _T("kNsHighSuppression");
-        }
-        else if (_checkNS1 % 6 == 5)
-        {
-            mode = kNsVeryHighSuppression;
-            str = _T("kNsVeryHighSuppression");
-        }
-        TEST(_veApmPtr->SetRxNsStatus(channel, true, mode) == 0, _T("SetRxNsStatus(channel=%d, enable=%d, %s)"), channel, enable, str);
-        _checkNS1++;
-    }
-    else
-    {
-        TEST(_veApmPtr->SetRxNsStatus(channel, false, kNsUnchanged) == 0, _T("SetRxNsStatus(channel=%d, enable=%d)"), enable, channel);
-    }
-}
-
-// ----------------------------------------------------------------------------
-//                         Channel-independent Operations
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnBnClickedCheckPlayFileIn()
-{
-    std::string micFile = _long_audio_file_path + "audio_short16.pcm";
-    // std::string micFile = _long_audio_file_path + "audio_long16noise.pcm";
-
-    int channel(-1);
-    CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN);
-    int check = buttonExtTrans->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        bool mix;
-        const bool loop(true);
-        const FileFormats format = kFileFormatPcm16kHzFile;
-        const float scale(1.0);
-
-        (_checkPlayFileIn %2 == 0) ? mix = true : mix = false;
-        TEST(_veFilePtr->StartPlayingFileAsMicrophone(channel,
-            micFile.c_str(), loop, mix, format, scale) == 0,
-            _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, ")
-            _T("loop=%d, mix=%d, format=%d, scale=%2.1f)"),
-            channel, CharToTchar(micFile.c_str(), -1),
-            loop, mix, format, scale);
-        _checkPlayFileIn++;
-    }
-    else
-    {
-        TEST(_veFilePtr->StopPlayingFileAsMicrophone(channel) == 0,
-            _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckRecMic()
-{
-    std::string micFile = webrtc::test::OutputPath() +
-                          "rec_mic_mono_16kHz.pcm";
-
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_REC_MIC);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veFilePtr->StartRecordingMicrophone(micFile.c_str(), NULL) == 0,
-            _T("StartRecordingMicrophone(file=%s)"),
-            CharToTchar(micFile.c_str(), -1));
-    }
-    else
-    {
-        TEST(_veFilePtr->StopRecordingMicrophone() == 0,
-            _T("StopRecordingMicrophone()"));
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckAgc()
-{
-    CButton* buttonAGC = (CButton*)GetDlgItem(IDC_CHECK_AGC);
-    int check = buttonAGC->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        AgcModes mode(kAgcDefault);
-        if (_checkAGC % 4 == 0)
-        {
-            mode = kAgcDefault;
-            str = _T("kAgcDefault");
-        }
-        else if (_checkAGC % 4 == 1)
-        {
-            mode = kAgcAdaptiveAnalog;
-            str = _T("kAgcAdaptiveAnalog");
-        }
-        else if (_checkAGC % 4 == 2)
-        {
-            mode = kAgcAdaptiveDigital;
-            str = _T("kAgcAdaptiveDigital");
-        }
-        else if (_checkAGC % 4 == 3)
-        {
-            mode = kAgcFixedDigital;
-            str = _T("kAgcFixedDigital");
-        }
-        TEST(_veApmPtr->SetAgcStatus(true, mode) == 0, _T("SetAgcStatus(enable=%d, %s)"), enable, str);
-        _checkAGC++;
-    }
-    else
-    {
-        TEST(_veApmPtr->SetAgcStatus(false, kAgcUnchanged) == 0, _T("SetAgcStatus(enable=%d)"), enable);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckNs()
-{
-    CButton* buttonNS = (CButton*)GetDlgItem(IDC_CHECK_NS);
-    int check = buttonNS->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        NsModes mode(kNsDefault);
-        if (_checkNS % 6 == 0)
-        {
-            mode = kNsDefault;
-            str = _T("kNsDefault");
-        }
-        else if (_checkNS % 6 == 1)
-        {
-            mode = kNsConference;
-            str = _T("kNsConference");
-        }
-        else if (_checkNS % 6 == 2)
-        {
-            mode = kNsLowSuppression;
-            str = _T("kNsLowSuppression");
-        }
-        else if (_checkNS % 6 == 3)
-        {
-            mode = kNsModerateSuppression;
-            str = _T("kNsModerateSuppression");
-        }
-        else if (_checkNS % 6 == 4)
-        {
-            mode = kNsHighSuppression;
-            str = _T("kNsHighSuppression");
-        }
-        else if (_checkNS % 6 == 5)
-        {
-            mode = kNsVeryHighSuppression;
-            str = _T("kNsVeryHighSuppression");
-        }
-        TEST(_veApmPtr->SetNsStatus(true, mode) == 0, _T("SetNsStatus(enable=%d, %s)"), enable, str);
-        _checkNS++;
-    }
-    else
-    {
-        TEST(_veApmPtr->SetNsStatus(false, kNsUnchanged) == 0, _T("SetNsStatus(enable=%d)"), enable);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckEc()
-{
-    CButton* buttonEC = (CButton*)GetDlgItem(IDC_CHECK_EC);
-    int check = buttonEC->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        CString str;
-        EcModes mode(kEcDefault);
-        if (_checkEC % 4 == 0)
-        {
-            mode = kEcDefault;
-            str = _T("kEcDefault");
-        }
-        else if (_checkEC % 4 == 1)
-        {
-            mode = kEcConference;
-            str = _T("kEcConference");
-        }
-        else if (_checkEC % 4 == 2)
-        {
-            mode = kEcAec;
-            str = _T("kEcAec");
-        }
-        else if (_checkEC % 4 == 3)
-        {
-            mode = kEcAecm;
-            str = _T("kEcAecm");
-        }
-        TEST(_veApmPtr->SetEcStatus(true, mode) == 0, _T("SetEcStatus(enable=%d, %s)"), enable, str);
-        _checkEC++;
-    }
-    else
-    {
-        TEST(_veApmPtr->SetEcStatus(false, kEcUnchanged) == 0, _T("SetEcStatus(enable=%d)"), enable);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckMuteIn()
-{
-    CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN);
-    int check = buttonMute->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    const int channel(-1);
-    TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
-        _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaIn()
-{
-    const int channel(-1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingAllChannelsMixed, *_externalMediaPtr) == 0,
-            _T("RegisterExternalMediaProcessing(channel=%d, kRecordingAllChannelsMixed, processObject=0x%x)"), channel, _externalMediaPtr);
-    }
-    else
-    {
-        TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingAllChannelsMixed) == 0,
-            _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingAllChannelsMixed)"), channel);
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaOut()
-{
-    const int channel(-1);
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    if (enable)
-    {
-        TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackAllChannelsMixed, *_externalMediaPtr) == 0,
-            _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackAllChannelsMixed, processObject=0x%x)"), channel, _externalMediaPtr);
-    }
-    else
-    {
-        TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackAllChannelsMixed) == 0,
-            _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackAllChannelsMixed)"), channel);
-    }
-}
-
-void CWinTestDlg::OnCbnSelchangeComboRecDevice()
-{
-    CComboBox* comboCodec(NULL);
-    comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_REC_DEVICE);
-    int index = comboCodec->GetCurSel();
-    TEST(_veHardwarePtr->SetRecordingDevice(index) == 0,
-        _T("SetRecordingDevice(index=%d)"), index);
-}
-
-void CWinTestDlg::OnCbnSelchangeComboPlayDevice()
-{
-    CComboBox* comboCodec(NULL);
-    comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_PLAY_DEVICE);
-    int index = comboCodec->GetCurSel();
-    TEST(_veHardwarePtr->SetPlayoutDevice(index) == 0,
-        _T("SetPlayoutDevice(index=%d)"), index);
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderInputVolume(NMHDR *pNMHDR, LRESULT *pResult)
-{
-    CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
-    slider->SetRangeMin(0);
-    slider->SetRangeMax(255);
-    int pos = slider->GetPos();
-
-    TEST(_veVolumeControlPtr->SetMicVolume(pos) == 0, _T("SetMicVolume(volume=%d)"), pos);
-
-    *pResult = 0;
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderOutputVolume(NMHDR *pNMHDR, LRESULT *pResult)
-{
-    CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
-    slider->SetRangeMin(0);
-    slider->SetRangeMax(255);
-    int pos = slider->GetPos();
-
-    TEST(_veVolumeControlPtr->SetSpeakerVolume(pos) == 0, _T("SetSpeakerVolume(volume=%d)"), pos);
-
-    *pResult = 0;
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderPanLeft(NMHDR *pNMHDR, LRESULT *pResult)
-{
-    CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_LEFT);
-    slider->SetRange(0,10);
-    int pos = 10 - slider->GetPos();    // 0 <=> lower end, 10 <=> upper end
-
-    float left(0.0);
-    float right(0.0);
-    const int channel(-1);
-
-    // Only left channel will be modified
-    _veVolumeControlPtr->GetOutputVolumePan(channel, left, right);
-
-    left = (float)((float)pos/10.0f);
-
-    TEST(_veVolumeControlPtr->SetOutputVolumePan(channel, left, right) == 0,
-        _T("SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)"), channel, left, right);
-
-    *pResult = 0;
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderPanRight(NMHDR *pNMHDR, LRESULT *pResult)
-{
-    CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_RIGHT);
-    slider->SetRange(0,10);
-    int pos = 10 - slider->GetPos();    // 0 <=> lower end, 10 <=> upper end
-
-    float left(0.0);
-    float right(0.0);
-    const int channel(-1);
-
-    // Only right channel will be modified
-    _veVolumeControlPtr->GetOutputVolumePan(channel, left, right);
-
-    right = (float)((float)pos/10.0f);
-
-    TEST(_veVolumeControlPtr->SetOutputVolumePan(channel, left, right) == 0,
-        _T("SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)"), channel, left, right);
-
-    *pResult = 0;
-}
-
-void CWinTestDlg::OnBnClickedButtonVersion()
-{
-    if (_veBasePtr)
-    {
-        char version[1024];
-        if (_veBasePtr->GetVersion(version) == 0)
-        {
-            AfxMessageBox(CString(version), MB_OK);
-        }
-        else
-        {
-            AfxMessageBox(_T("FAILED!"), MB_OK);
-        }
-    }
-}
-
-void CWinTestDlg::OnBnClickedCheckRecCall()
-{
-    // Not supported
-}
-
-void CWinTestDlg::OnBnClickedCheckTypingDetection()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_TYPING_DETECTION);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    TEST(_veApmPtr->SetTypingDetectionStatus(enable) == 0, _T("SetTypingDetectionStatus(enable=%d)"), enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckRED()
-{
-    CButton* button = (CButton*)GetDlgItem(IDC_CHECK_RED);
-    int channel = GetDlgItemInt(IDC_EDIT_1);
-    int check = button->GetCheck();
-    const bool enable = (check == BST_CHECKED);
-    TEST(_veRtpRtcpPtr->SetREDStatus(channel, enable) == 0,
-         _T("SetREDStatus(enable=%d)"), enable);
-}
-
-// ----------------------------------------------------------------------------
-//                                   Message Handlers
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnTimer(UINT_PTR nIDEvent)
-{
-    CString str;
-
-    unsigned int svol(0);
-    unsigned int mvol(0);
-
-    _timerTicks++;
-
-    // Get speaker and microphone volumes
-    _veVolumeControlPtr->GetSpeakerVolume(svol);
-    _veVolumeControlPtr->GetMicVolume(mvol);
-
-    // Update speaker volume slider
-    CSliderCtrl* sliderSpkr = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
-    sliderSpkr->SetRangeMin(0);
-    sliderSpkr->SetRangeMax(255);
-    sliderSpkr->SetPos(svol);
-
-    // Update microphone volume slider
-    CSliderCtrl* sliderMic = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
-    sliderMic->SetRangeMin(0);
-    sliderMic->SetRangeMax(255);
-    sliderMic->SetPos(mvol);
-
-    unsigned int micLevel;
-    unsigned int combinedOutputLevel;
-
-    // Get audio levels
-    _veVolumeControlPtr->GetSpeechInputLevel(micLevel);
-    _veVolumeControlPtr->GetSpeechOutputLevel(-1, combinedOutputLevel);
-
-    // Update audio level controls
-    CProgressCtrl* progressMic = (CProgressCtrl*)GetDlgItem(IDC_PROGRESS_AUDIO_LEVEL_IN);
-    progressMic->SetRange(0,9);
-    progressMic->SetStep(1);
-    progressMic->SetPos(micLevel);
-    CProgressCtrl* progressOut = (CProgressCtrl*)GetDlgItem(IDC_PROGRESS_AUDIO_LEVEL_OUT);
-    progressOut->SetRange(0,9);
-    progressOut->SetStep(1);
-    progressOut->SetPos(combinedOutputLevel);
-
-    // Update playout delay (buffer size)
-    if (_veVideoSyncPtr)
-    {
-        int bufferMs(0);
-        _veVideoSyncPtr->GetPlayoutBufferSize(bufferMs);
-        SetDlgItemInt(IDC_EDIT_PLAYOUT_BUFFER_SIZE, bufferMs);
-    }
-
-    if (_delayEstimate1 && _veVideoSyncPtr)
-    {
-        const int channel = GetDlgItemInt(IDC_EDIT_1);
-        int delayMs(0);
-        _veVideoSyncPtr->GetDelayEstimate(channel, delayMs);
-        SetDlgItemInt(IDC_EDIT_DELAY_ESTIMATE_1, delayMs);
-    }
-
-    if (_rxVad && _veApmPtr && _rxVadObserverPtr)
-    {
-        SetDlgItemInt(IDC_EDIT_RXVAD, _rxVadObserverPtr->vad_decision);
-    }
-
-    if (_veHardwarePtr)
-    {
-        int load1, load2;
-        _veHardwarePtr->GetSystemCPULoad(load1);
-        _veHardwarePtr->GetCPULoad(load2);
-        str.Format(_T("CPU load (system/VoE): %d/%d [%%]"), load1, load2);
-        SetDlgItemText(IDC_EDIT_CPU_LOAD, (LPCTSTR)str);
-    }
-
-    BOOL ret;
-    int channel = GetDlgItemInt(IDC_EDIT_1, &ret);
-
-    if (_veCodecPtr)
-    {
-        if (ret == TRUE)
-        {
-            CodecInst codec;
-            if (_veCodecPtr->GetRecCodec(channel, codec) == 0)
-            {
-        str.Format(_T("RX codec: %s, freq=%d, pt=%d, rate=%d, size=%d"), CharToTchar(codec.plname, 32), codec.plfreq, codec.pltype, codec.rate, codec.pacsize);
-                SetDlgItemText(IDC_EDIT_RX_CODEC_1, (LPCTSTR)str);
-            }
-        }
-    }
-
-    if (_veRtpRtcpPtr)
-    {
-        if (ret == TRUE)
-        {
-            CallStatistics stats;
-            if (_veRtpRtcpPtr->GetRTCPStatistics(channel, stats) == 0)
-            {
-                str.Format(_T("RTCP | RTP: cum=%u, ext=%d, frac=%u, jitter=%u | TX=%d, RX=%d, RTT=%d"),
-                    stats.cumulativeLost, stats.extendedMax, stats.fractionLost, stats.jitterSamples, stats.packetsSent, stats.packetsReceived, stats.rttMs);
-                SetDlgItemText(IDC_EDIT_RTCP_STAT_1, (LPCTSTR)str);
-            }
-        }
-    }
-
-    SetTimer(0, 1000, NULL);
-    CDialog::OnTimer(nIDEvent);
-}
-
-void CWinTestDlg::OnBnClickedButtonClearErrorCallback()
-{
-    _nErrorCallbacks = 0;
-    SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, _T(""));
-}
-
-// ----------------------------------------------------------------------------
-//                                       TEST
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnBnClickedButtonTest1()
-{
-    // add tests here...
-}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.h b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.h
deleted file mode 100644
index a77988096d..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.h
+++ /dev/null
@@ -1,274 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_WINTESTDLG_H_
-#define WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_WINTESTDLG_H_
-
-#if (_MSC_VER >= 1400)
-#define PRINT_GET_RESULT(...) \
-    { \
-        _strMsg.Format(__VA_ARGS__); \
-        SetDlgItemText(IDC_EDIT_GET_OUTPUT, _strMsg); \
-    } \
-
-#define TEST(x, ...) \
-    if (!(x)) \
-    { \
-        _strMsg.Format(__VA_ARGS__); \
-        SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
-        _strErr.Format(_T("FAILED (error=%d)"), _veBasePtr->LastError()); \
-        SetDlgItemText(IDC_EDIT_RESULT, _strErr); \
-        _failCount++; \
-        SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount); \
-        SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError()); \
-    } \
-    else \
-    { \
-        _strMsg.Format(__VA_ARGS__); \
-        SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
-        SetDlgItemText(IDC_EDIT_RESULT, _T("OK")); \
-    } \
-
-#define TEST2(x, ...) \
-    if (!(x)) \
-    { \
-        _strMsg.Format(__VA_ARGS__); \
-        ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(true, _strMsg); \
-    } \
-    else \
-    { \
-        _strMsg.Format(__VA_ARGS__); \
-        ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(false, _strMsg); \
-    }
-#else
-#define TEST(x, exp) \
-    if (!(x)) \
-    { \
-        _strMsg.Format(exp); \
-        SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
-        _strErr.Format("FAILED (error=%d)", _veBasePtr->LastError()); \
-        SetDlgItemText(IDC_EDIT_RESULT, _strErr); \
-        _failCount++; \
-        SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount); \
-        SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError()); \
-    } \
-    else \
-    { \
-        _strMsg.Format(exp); \
-        SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
-        SetDlgItemText(IDC_EDIT_RESULT, _T("OK")); \
-    } \
-
-#define TEST2(x, exp) \
-    if (!(x)) \
-    { \
-        _strMsg.Format(exp); \
-        ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(true, _strMsg); \
-    } \
-    else \
-    { \
-        _strMsg.Format(exp); \
-        ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(false, _strMsg); \
-    }
-#endif
-
-#include <string>
-
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_dtmf.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/include/voe_file.h"
-#include "webrtc/voice_engine/include/voe_hardware.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "webrtc/voice_engine/include/voe_errors.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-
-class MediaProcessImpl;
-class RxCallback;
-class MyTransport;
-
-using namespace webrtc;
-
-#define MAX_NUM_OF_CHANNELS    10
-
-// CWinTestDlg dialog
-class CWinTestDlg : public CDialog,
-                    public VoiceEngineObserver,
-                    public VoERTPObserver
-{
-// Construction
-public:
-    CWinTestDlg(CWnd* pParent = NULL);    // standard constructor
-    virtual ~CWinTestDlg();
-
-// Dialog Data
-    enum { IDD = IDD_WINTEST_DIALOG };
-
-    BOOL UpdateTest(bool failed, const CString& strMsg);
-
-protected:
-    virtual void DoDataExchange(CDataExchange* pDX);    // DDX/DDV support
-
-protected:  // VoiceEngineObserver
-    virtual void CallbackOnError(int channel, int errCode);
-
-protected:    // VoERTPObserver
-    virtual void OnIncomingCSRCChanged(
-        int channel, unsigned int CSRC, bool added);
-    virtual void OnIncomingSSRCChanged(
-        int channel, unsigned int SSRC);
-
-// Implementation
-protected:
-    HICON m_hIcon;
-
-    // Generated message map functions
-    virtual BOOL OnInitDialog();
-    afx_msg void OnSysCommand(UINT nID, LPARAM lParam);
-    afx_msg void OnPaint();
-    afx_msg HCURSOR OnQueryDragIcon();
-    DECLARE_MESSAGE_MAP()
-public:
-    afx_msg void OnBnClickedButtonCreate1();
-    afx_msg void OnBnClickedButtonDelete1();
-
-private:
-    VoiceEngine*    _vePtr;
-
-    VoECodec*               _veCodecPtr;
-    VoEExternalMedia*       _veExternalMediaPtr;
-    VoEVolumeControl*       _veVolumeControlPtr;
-    VoEHardware*            _veHardwarePtr;
-    VoEVideoSync*           _veVideoSyncPtr;
-    VoENetwork*             _veNetworkPtr;
-    VoEFile*                _veFilePtr;
-    VoEAudioProcessing*     _veApmPtr;
-    VoEBase*                _veBasePtr;
-    VoERTP_RTCP*            _veRtpRtcpPtr;
-
-    MyTransport*            _transportPtr;
-    MediaProcessImpl*       _externalMediaPtr;
-    RxCallback*             _rxVadObserverPtr;
-
-private:
-    int                     _failCount;
-    CString                 _strMsg;
-    CString                 _strErr;
-    bool                    _externalTransport;
-    bool                    _externalTransportBuild;
-    int                     _checkPlayFileIn;
-    int                     _checkPlayFileIn1;
-    int                     _checkPlayFileIn2;
-    int                     _checkPlayFileOut1;
-    int                     _checkPlayFileOut2;
-    int                     _checkAGC;
-    int                     _checkAGC1;
-    int                     _checkNS;
-    int                     _checkNS1;
-    int                     _checkEC;
-    int                     _checkVAD1;
-    int                     _checkVAD2;
-    int                     _checkSrtpTx1;
-    int                     _checkSrtpTx2;
-    int                     _checkSrtpRx1;
-    int                     _checkSrtpRx2;
-    int                     _checkConference1;
-    int                     _checkConference2;
-    int                     _checkOnHold1;
-    int                     _checkOnHold2;
-    bool                    _delayEstimate1;
-    bool                    _delayEstimate2;
-    bool                    _rxVad;
-    int                     _nErrorCallbacks;
-    int                     _timerTicks;
-    std::string             _long_audio_file_path;
-
-public:
-    afx_msg void OnBnClickedButtonCreate2();
-    afx_msg void OnBnClickedButtonDelete2();
-    afx_msg void OnCbnSelchangeComboCodec1();
-    afx_msg void OnBnClickedButtonStartListen1();
-    afx_msg void OnBnClickedButtonStopListen1();
-    afx_msg void OnBnClickedButtonStartPlayout1();
-    afx_msg void OnBnClickedButtonStopPlayout1();
-    afx_msg void OnBnClickedButtonStartSend1();
-    afx_msg void OnBnClickedButtonStopSend1();
-    afx_msg void OnCbnSelchangeComboIp2();
-    afx_msg void OnCbnSelchangeComboIp1();
-    afx_msg void OnCbnSelchangeComboCodec2();
-    afx_msg void OnBnClickedButtonStartListen2();
-    afx_msg void OnBnClickedButtonStopListen2();
-    afx_msg void OnBnClickedButtonStartPlayout2();
-    afx_msg void OnBnClickedButtonStopPlayout2();
-    afx_msg void OnBnClickedButtonStartSend2();
-    afx_msg void OnBnClickedButtonStopSend2();
-    afx_msg void OnBnClickedButtonTest11();
-    afx_msg void OnBnClickedCheckExtTrans1();
-    afx_msg void OnBnClickedCheckPlayFileIn1();
-    afx_msg void OnBnClickedCheckPlayFileOut1();
-    afx_msg void OnBnClickedCheckExtTrans2();
-    afx_msg void OnBnClickedCheckPlayFileIn2();
-    afx_msg void OnBnClickedCheckPlayFileOut2();
-    afx_msg void OnBnClickedCheckPlayFileIn();
-    afx_msg void OnBnClickedCheckPlayFileOut();
-    afx_msg void OnCbnSelchangeComboRecDevice();
-    afx_msg void OnCbnSelchangeComboPlayDevice();
-    afx_msg void OnBnClickedCheckExtMediaIn1();
-    afx_msg void OnBnClickedCheckExtMediaOut1();
-    afx_msg void OnNMReleasedcaptureSliderInputVolume(NMHDR *pNMHDR, LRESULT *pResult);
-    afx_msg void OnNMReleasedcaptureSliderOutputVolume(NMHDR *pNMHDR, LRESULT *pResult);
-    afx_msg void OnTimer(UINT_PTR nIDEvent);
-    afx_msg void OnBnClickedCheckAgc();
-    CString _strComboIp1;
-    CString _strComboIp2;
-    afx_msg void OnBnClickedCheckNs();
-    afx_msg void OnBnClickedCheckEc();
-    afx_msg void OnBnClickedCheckVad1();
-    afx_msg void OnBnClickedCheckVad2();
-    afx_msg void OnBnClickedCheckExtMediaIn2();
-    afx_msg void OnBnClickedCheckExtMediaOut2();
-    afx_msg void OnBnClickedCheckMuteIn();
-    afx_msg void OnBnClickedCheckMuteIn1();
-    afx_msg void OnBnClickedCheckMuteIn2();
-    afx_msg void OnBnClickedCheckSrtpTx1();
-    afx_msg void OnBnClickedCheckSrtpRx1();
-    afx_msg void OnBnClickedCheckSrtpTx2();
-    afx_msg void OnBnClickedCheckSrtpRx2();
-    afx_msg void OnBnClickedButtonDtmf1();
-    afx_msg void OnBnClickedCheckRecMic();
-    afx_msg void OnBnClickedButtonDtmf2();
-    afx_msg void OnBnClickedButtonTest1();
-    afx_msg void OnBnClickedCheckConference1();
-    afx_msg void OnBnClickedCheckConference2();
-    afx_msg void OnBnClickedCheckOnHold1();
-    afx_msg void OnBnClickedCheckOnHold2();
-    afx_msg void OnBnClickedCheckExtMediaIn();
-    afx_msg void OnBnClickedCheckExtMediaOut();
-    afx_msg void OnLbnSelchangeListCodec1();
-    afx_msg void OnNMReleasedcaptureSliderPanLeft(NMHDR *pNMHDR, LRESULT *pResult);
-    afx_msg void OnNMReleasedcaptureSliderPanRight(NMHDR *pNMHDR, LRESULT *pResult);
-    afx_msg void OnBnClickedButtonVersion();
-    afx_msg void OnBnClickedCheckDelayEstimate1();
-    afx_msg void OnBnClickedCheckRxvad();
-    afx_msg void OnBnClickedCheckAgc1();
-    afx_msg void OnBnClickedCheckNs1();
-    afx_msg void OnBnClickedCheckRecCall();
-    afx_msg void OnBnClickedCheckTypingDetection();
-    afx_msg void OnBnClickedCheckRED();
-    afx_msg void OnBnClickedButtonClearErrorCallback();
-    afx_msg void OnBnClickedCheckBwe1();
-};
-
-#endif  // WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_WINTESTDLG_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/res/WinTest.ico b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/res/WinTest.ico
deleted file mode 100644
index 8a84ca3d34..0000000000
Binary files a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/res/WinTest.ico and /dev/null differ
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/res/WinTest.rc2 b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/res/WinTest.rc2
deleted file mode 100644
index 68f8559920..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/res/WinTest.rc2
+++ /dev/null
@@ -1,13 +0,0 @@
-//
-// WinTest.RC2 - resources Microsoft Visual C++ does not edit directly
-//
-
-#ifdef APSTUDIO_INVOKED
-#error this file is not editable by Microsoft Visual C++
-#endif //APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-// Add manually edited resources here...
-
-/////////////////////////////////////////////////////////////////////////////
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/stdafx.cc b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/stdafx.cc
deleted file mode 100644
index e321601295..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/stdafx.cc
+++ /dev/null
@@ -1,15 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// stdafx.cpp : source file that includes just the standard includes
-// WinTest.pch will be the pre-compiled header
-// stdafx.obj will contain the pre-compiled type information
-
-#include "webrtc/voice_engine/test/win_test/stdafx.h"
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/stdafx.h b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/stdafx.h
deleted file mode 100644
index 17b3c30eaa..0000000000
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/stdafx.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_STDAFX_H_
-#define WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_STDAFX_H_
-
-// stdafx.h : include file for standard system include files,
-// or project specific include files that are used frequently,
-// but are changed infrequently
-
-#ifndef _SECURE_ATL
-#define _SECURE_ATL 1
-#endif
-
-#ifndef VC_EXTRALEAN
-#define VC_EXTRALEAN		// Exclude rarely-used stuff from Windows headers
-#endif
-
-// Modify the following defines if you have to target a platform prior to the ones specified below.
-// Refer to MSDN for the latest info on corresponding values for different platforms.
-#ifndef WINVER				// Allow use of features specific to Windows XP or later.
-#define WINVER 0x0501		// Change this to the appropriate value to target other versions of Windows.
-#endif
-
-#ifndef _WIN32_WINNT		// Allow use of features specific to Windows XP or later.                   
-#define _WIN32_WINNT 0x0501	// Change this to the appropriate value to target other versions of Windows.
-#endif						
-
-#ifndef _WIN32_WINDOWS		// Allow use of features specific to Windows 98 or later.
-#define _WIN32_WINDOWS 0x0410 // Change this to the appropriate value to target Windows Me or later.
-#endif
-
-#ifndef _WIN32_IE			// Allow use of features specific to IE 6.0 or later.
-#define _WIN32_IE 0x0600	// Change this to the appropriate value to target other versions of IE.
-#endif
-
-#define _ATL_CSTRING_EXPLICIT_CONSTRUCTORS	// some CString constructors will be explicit
-
-// turns off MFC's hiding of some common and often safely ignored warning messages
-#define _AFX_ALL_WARNINGS
-
-#include <afxwin.h>         // MFC core and standard components
-#include <afxext.h>         // MFC extensions
-
-#ifndef _AFX_NO_OLE_SUPPORT
-#include <afxdtctl.h>		// MFC support for Internet Explorer 4 Common Controls
-#endif
-#ifndef _AFX_NO_AFXCMN_SUPPORT
-#include <afxcmn.h>			// MFC support for Windows Common Controls
-#endif // _AFX_NO_AFXCMN_SUPPORT
-
-#ifdef _UNICODE
-#if defined _M_IX86
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='x86' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#elif defined _M_IA64
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='ia64' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#elif defined _M_X64
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='amd64' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#else
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='*' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#endif
-#endif
-
-#endif  // WEBRTC_VOICE_ENGINE_TEST_WIN_TEST_STDAFX_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc
index ac9247f062..4505eb1319 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc
@@ -10,11 +10,12 @@
 
 #include "webrtc/voice_engine/transmit_mixer.h"
 
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
@@ -22,8 +23,6 @@
 #include "webrtc/voice_engine/utility.h"
 #include "webrtc/voice_engine/voe_base_impl.h"
 
-#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
-
 namespace webrtc {
 namespace voe {
 
@@ -35,12 +34,20 @@ TransmitMixer::OnPeriodicProcess()
                  "TransmitMixer::OnPeriodicProcess()");
 
 #if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
-    if (_typingNoiseWarningPending)
+    bool send_typing_noise_warning = false;
+    bool typing_noise_detected = false;
     {
+      CriticalSectionScoped cs(&_critSect);
+      if (_typingNoiseWarningPending) {
+        send_typing_noise_warning = true;
+        typing_noise_detected = _typingNoiseDetected;
+        _typingNoiseWarningPending = false;
+      }
+    }
+    if (send_typing_noise_warning) {
         CriticalSectionScoped cs(&_callbackCritSect);
-        if (_voiceEngineObserverPtr)
-        {
-            if (_typingNoiseDetected) {
+        if (_voiceEngineObserverPtr) {
+            if (typing_noise_detected) {
                 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                              "TransmitMixer::OnPeriodicProcess() => "
                              "CallbackOnError(VE_TYPING_NOISE_WARNING)");
@@ -56,7 +63,6 @@ TransmitMixer::OnPeriodicProcess()
                     VE_TYPING_NOISE_OFF_WARNING);
             }
         }
-        _typingNoiseWarningPending = false;
     }
 #endif
 
@@ -294,7 +300,8 @@ TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
     return 0;
 }
 
-void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
+void TransmitMixer::GetSendCodecInfo(int* max_sample_rate,
+                                     size_t* max_channels) {
   *max_sample_rate = 8000;
   *max_channels = 1;
   for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
@@ -311,8 +318,8 @@ void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
 
 int32_t
 TransmitMixer::PrepareDemux(const void* audioSamples,
-                            uint32_t nSamples,
-                            uint8_t nChannels,
+                            size_t nSamples,
+                            size_t nChannels,
                             uint32_t samplesPerSec,
                             uint16_t totalDelayMS,
                             int32_t clockDrift,
@@ -320,10 +327,11 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
                             bool keyPressed)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
-                 "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
-                 "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%d,"
-                 "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
-                 totalDelayMS, clockDrift, currentMicLevel);
+                 "TransmitMixer::PrepareDemux(nSamples=%" PRIuS ", "
+                 "nChannels=%" PRIuS ", samplesPerSec=%u, totalDelayMS=%u, "
+                 "clockDrift=%d, currentMicLevel=%u)",
+                 nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift,
+                 currentMicLevel);
 
     // --- Resample input audio and create/store the initial audio frame
     GenerateAudioFrame(static_cast<const int16_t*>(audioSamples),
@@ -425,8 +433,8 @@ TransmitMixer::DemuxAndMix()
 }
 
 void TransmitMixer::DemuxAndMix(const int voe_channels[],
-                                int number_of_voe_channels) {
-  for (int i = 0; i < number_of_voe_channels; ++i) {
+                                size_t number_of_voe_channels) {
+  for (size_t i = 0; i < number_of_voe_channels; ++i) {
     voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
     voe::Channel* channel_ptr = ch.channel();
     if (channel_ptr) {
@@ -458,8 +466,8 @@ TransmitMixer::EncodeAndSend()
 }
 
 void TransmitMixer::EncodeAndSend(const int voe_channels[],
-                                  int number_of_voe_channels) {
-  for (int i = 0; i < number_of_voe_channels; ++i) {
+                                  size_t number_of_voe_channels) {
+  for (size_t i = 0; i < number_of_voe_channels; ++i) {
     voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
     voe::Channel* channel_ptr = ch.channel();
     if (channel_ptr && channel_ptr->Sending())
@@ -643,9 +651,6 @@ int TransmitMixer::StopPlayingFileAsMicrophone()
 
     if (!_filePlaying)
     {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceWarning,
-            "StopPlayingFileAsMicrophone() isnot playing");
         return 0;
     }
 
@@ -694,8 +699,7 @@ int TransmitMixer::StartRecordingMicrophone(const char* fileName,
     const uint32_t notificationTime(0); // Not supported in VoE
     CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
 
-    if (codecInst != NULL &&
-      (codecInst->channels < 0 || codecInst->channels > 2))
+    if (codecInst != NULL && codecInst->channels > 2)
     {
         _engineStatisticsPtr->SetLastError(
             VE_BAD_ARGUMENT, kTraceError,
@@ -1130,37 +1134,31 @@ bool TransmitMixer::IsRecordingMic()
 // Note that if drift compensation is done here, a buffering stage will be
 // needed and this will need to switch to non-fixed resamples.
 void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
-                                       int samples_per_channel,
-                                       int num_channels,
+                                       size_t samples_per_channel,
+                                       size_t num_channels,
                                        int sample_rate_hz) {
   int codec_rate;
-  int num_codec_channels;
+  size_t num_codec_channels;
   GetSendCodecInfo(&codec_rate, &num_codec_channels);
-  // TODO(ajm): This currently restricts the sample rate to 32 kHz.
-  // See: https://code.google.com/p/webrtc/issues/detail?id=3146
-  // When 48 kHz is supported natively by AudioProcessing, this will have
-  // to be changed to handle 44.1 kHz.
-  int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz;
-  if (audioproc_->echo_control_mobile()->is_enabled()) {
-    // AECM only supports 8 and 16 kHz.
-    max_sample_rate_hz = 16000;
-  }
-  codec_rate = std::min(codec_rate, max_sample_rate_hz);
   stereo_codec_ = num_codec_channels == 2;
 
-  if (!mono_buffer_.get()) {
-    // Temporary space for DownConvertToCodecFormat.
-    mono_buffer_.reset(new int16_t[kMaxMonoDataSizeSamples]);
+  // We want to process at the lowest rate possible without losing information.
+  // Choose the lowest native rate at least equal to the input and codec rates.
+  const int min_processing_rate = std::min(sample_rate_hz, codec_rate);
+  for (size_t i = 0; i < AudioProcessing::kNumNativeSampleRates; ++i) {
+    _audioFrame.sample_rate_hz_ = AudioProcessing::kNativeSampleRatesHz[i];
+    if (_audioFrame.sample_rate_hz_ >= min_processing_rate) {
+      break;
+    }
   }
-  DownConvertToCodecFormat(audio,
-                           samples_per_channel,
-                           num_channels,
-                           sample_rate_hz,
-                           num_codec_channels,
-                           codec_rate,
-                           mono_buffer_.get(),
-                           &resampler_,
-                           &_audioFrame);
+  if (audioproc_->echo_control_mobile()->is_enabled()) {
+    // AECM only supports 8 and 16 kHz.
+    _audioFrame.sample_rate_hz_ = std::min(
+        _audioFrame.sample_rate_hz_, AudioProcessing::kMaxAECMSampleRateHz);
+  }
+  _audioFrame.num_channels_ = std::min(num_channels, num_codec_channels);
+  RemixAndResample(audio, samples_per_channel, num_channels, sample_rate_hz,
+                   &resampler_, &_audioFrame);
 }
 
 int32_t TransmitMixer::RecordAudioToFile(
@@ -1191,7 +1189,7 @@ int32_t TransmitMixer::MixOrReplaceAudioWithFile(
 {
   rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
 
-    int fileSamples(0);
+    size_t fileSamples(0);
     {
         CriticalSectionScoped cs(&_critSect);
         if (_filePlayerPtr == NULL)
@@ -1245,15 +1243,13 @@ int32_t TransmitMixer::MixOrReplaceAudioWithFile(
 void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
                                  int current_mic_level, bool key_pressed) {
   if (audioproc_->set_stream_delay_ms(delay_ms) != 0) {
-    // A redundant warning is reported in AudioDevice, which we've throttled
-    // to avoid flooding the logs. Relegate this one to LS_VERBOSE to avoid
-    // repeating the problem here.
-    LOG_FERR1(LS_VERBOSE, set_stream_delay_ms, delay_ms);
+    // Silently ignore this failure to avoid flooding the logs.
   }
 
   GainControl* agc = audioproc_->gain_control();
   if (agc->set_stream_analog_level(current_mic_level) != 0) {
-    LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level);
+    LOG(LS_ERROR) << "set_stream_analog_level failed: current_mic_level = "
+                  << current_mic_level;
     assert(false);
   }
 
@@ -1288,9 +1284,11 @@ void TransmitMixer::TypingDetection(bool keyPressed)
 
   bool vadActive = _audioFrame.vad_activity_ == AudioFrame::kVadActive;
   if (_typingDetection.Process(keyPressed, vadActive)) {
+    CriticalSectionScoped cs(&_critSect);
     _typingNoiseWarningPending = true;
     _typingNoiseDetected = true;
   } else {
+    CriticalSectionScoped cs(&_critSect);
     // If there is already a warning pending, do not change the state.
     // Otherwise set a warning pending if last callback was for noise detected.
     if (!_typingNoiseWarningPending && _typingNoiseDetected) {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h
index 919de13123..0aee106231 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h
@@ -15,9 +15,9 @@
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/file_player.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"
@@ -51,8 +51,8 @@ public:
         AudioProcessing* audioProcessingModule);
 
     int32_t PrepareDemux(const void* audioSamples,
-                         uint32_t nSamples,
-                         uint8_t  nChannels,
+                         size_t nSamples,
+                         size_t nChannels,
                          uint32_t samplesPerSec,
                          uint16_t totalDelayMS,
                          int32_t  clockDrift,
@@ -63,12 +63,12 @@ public:
     int32_t DemuxAndMix();
     // Used by the Chrome to pass the recording data to the specific VoE
     // channels for demux.
-    void DemuxAndMix(const int voe_channels[], int number_of_voe_channels);
+    void DemuxAndMix(const int voe_channels[], size_t number_of_voe_channels);
 
     int32_t EncodeAndSend();
     // Used by the Chrome to pass the recording data to the specific VoE
     // channels for encoding and sending to the network.
-    void EncodeAndSend(const int voe_channels[], int number_of_voe_channels);
+    void EncodeAndSend(const int voe_channels[], size_t number_of_voe_channels);
 
     // Must be called on the same thread as PrepareDemux().
     uint32_t CaptureLevel() const;
@@ -170,11 +170,11 @@ private:
 
     // Gets the maximum sample rate and number of channels over all currently
     // sending codecs.
-    void GetSendCodecInfo(int* max_sample_rate, int* max_channels);
+    void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels);
 
     void GenerateAudioFrame(const int16_t audioSamples[],
-                            int nSamples,
-                            int nChannels,
+                            size_t nSamples,
+                            size_t nChannels,
                             int samplesPerSec);
     int32_t RecordAudioToFile(uint32_t mixingFrequency);
 
@@ -229,7 +229,6 @@ private:
     int32_t _remainingMuteMicTimeMs;
     bool stereo_codec_;
     bool swap_stereo_channels_;
-    rtc::scoped_ptr<int16_t[]> mono_buffer_;
 };
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer_unittest.cc
index 5fb982b1fc..27aa8b3f01 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer_unittest.cc
@@ -20,7 +20,7 @@ namespace {
 class MediaCallback : public VoEMediaProcess {
  public:
   virtual void Process(int channel, ProcessingTypes type,
-                       int16_t audio[], int samples_per_channel,
+                       int16_t audio[], size_t samples_per_channel,
                        int sample_rate_hz, bool is_stereo) {
   }
 };
diff --git a/media/webrtc/trunk/webrtc/voice_engine/utility.cc b/media/webrtc/trunk/webrtc/voice_engine/utility.cc
index bc085625bb..605e55369e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/utility.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/utility.cc
@@ -10,125 +10,80 @@
 
 #include "webrtc/voice_engine/utility.h"
 
+#include "webrtc/base/logging.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
 
-// TODO(ajm): There is significant overlap between RemixAndResample and
-// ConvertToCodecFormat. Consolidate using AudioConverter.
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
-  const int16_t* audio_ptr = src_frame.data_;
-  int audio_ptr_num_channels = src_frame.num_channels_;
-  int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
-
-  // Downmix before resampling.
-  if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
-    AudioFrameOperations::StereoToMono(src_frame.data_,
-                                       src_frame.samples_per_channel_,
-                                       mono_audio);
-    audio_ptr = mono_audio;
-    audio_ptr_num_channels = 1;
-  }
-
-  if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
-                                    dst_frame->sample_rate_hz_,
-                                    audio_ptr_num_channels) == -1) {
-    LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
-              dst_frame->sample_rate_hz_, audio_ptr_num_channels);
-    assert(false);
-  }
-
-  const int src_length = src_frame.samples_per_channel_ *
-                         audio_ptr_num_channels;
-  int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
-                                       AudioFrame::kMaxDataSizeSamples);
-  if (out_length == -1) {
-    LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_);
-    assert(false);
-  }
-  dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
-
-  // Upmix after resampling.
-  if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
-    // The audio in dst_frame really is mono at this point; MonoToStereo will
-    // set this back to stereo.
-    dst_frame->num_channels_ = 1;
-    AudioFrameOperations::MonoToStereo(dst_frame);
-  }
-
+  RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
+                   src_frame.num_channels_, src_frame.sample_rate_hz_,
+                   resampler, dst_frame);
   dst_frame->timestamp_ = src_frame.timestamp_;
   dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
   dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
 }
 
-void DownConvertToCodecFormat(const int16_t* src_data,
-                              int samples_per_channel,
-                              int num_channels,
-                              int sample_rate_hz,
-                              int codec_num_channels,
-                              int codec_rate_hz,
-                              int16_t* mono_buffer,
-                              PushResampler<int16_t>* resampler,
-                              AudioFrame* dst_af) {
-  assert(samples_per_channel <= kMaxMonoDataSizeSamples);
-  assert(num_channels == 1 || num_channels == 2);
-  assert(codec_num_channels == 1 || codec_num_channels == 2);
-  dst_af->Reset();
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame) {
+  const int16_t* audio_ptr = src_data;
+  size_t audio_ptr_num_channels = num_channels;
+  int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
 
-  // Never upsample the capture signal here. This should be done at the
-  // end of the send chain.
-  // XXX bug 1247574 temporary hack until we switch to full-duplex
-  // We need to know the final audio rate before starting the audio channels,
-  // and this means we can get called back in Process() with the input
-  // rate if it's less than the codec rate.
-  int destination_rate = codec_rate_hz;
-
-  // If no stereo codecs are in use, we downmix a stereo stream from the
-  // device early in the chain, before resampling.
-  if (num_channels == 2 && codec_num_channels == 1) {
+  // Downmix before resampling.
+  if (num_channels == 2 && dst_frame->num_channels_ == 1) {
     AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
-                                       mono_buffer);
-    src_data = mono_buffer;
-    num_channels = 1;
+                                       mono_audio);
+    audio_ptr = mono_audio;
+    audio_ptr_num_channels = 1;
   }
 
-  if (resampler->InitializeIfNeeded(
-          sample_rate_hz, destination_rate, num_channels) != 0) {
-    LOG_FERR3(LS_ERROR,
-              InitializeIfNeeded,
-              sample_rate_hz,
-              destination_rate,
-              num_channels);
+  if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
+                                    audio_ptr_num_channels) == -1) {
+    LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
+                  << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
+                  << dst_frame->sample_rate_hz_
+                  << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
     assert(false);
   }
 
-  const int in_length = samples_per_channel * num_channels;
-  int out_length = resampler->Resample(
-      src_data, in_length, dst_af->data_, AudioFrame::kMaxDataSizeSamples);
+  const size_t src_length = samples_per_channel * audio_ptr_num_channels;
+  int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
+                                       AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
-    LOG_FERR3(LS_ERROR, Resample, src_data, in_length, dst_af->data_);
+    LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
+                  << ", src_length = " << src_length
+                  << ", dst_frame->data_ = " << dst_frame->data_;
     assert(false);
   }
+  dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
 
-  dst_af->samples_per_channel_ = out_length / num_channels;
-  dst_af->sample_rate_hz_ = destination_rate;
-  dst_af->num_channels_ = num_channels;
+  // Upmix after resampling.
+  if (num_channels == 1 && dst_frame->num_channels_ == 2) {
+    // The audio in dst_frame really is mono at this point; MonoToStereo will
+    // set this back to stereo.
+    dst_frame->num_channels_ = 1;
+    AudioFrameOperations::MonoToStereo(dst_frame);
+  }
 }
 
 void MixWithSat(int16_t target[],
-                int target_channel,
+                size_t target_channel,
                 const int16_t source[],
-                int source_channel,
-                int source_len) {
+                size_t source_channel,
+                size_t source_len) {
   assert(target_channel == 1 || target_channel == 2);
   assert(source_channel == 1 || source_channel == 2);
 
@@ -136,7 +91,7 @@ void MixWithSat(int16_t target[],
     // Convert source from mono to stereo.
     int32_t left = 0;
     int32_t right = 0;
-    for (int i = 0; i < source_len; ++i) {
+    for (size_t i = 0; i < source_len; ++i) {
       left = source[i] + target[i * 2];
       right = source[i] + target[i * 2 + 1];
       target[i * 2] = WebRtcSpl_SatW32ToW16(left);
@@ -145,13 +100,13 @@ void MixWithSat(int16_t target[],
   } else if (target_channel == 1 && source_channel == 2) {
     // Convert source from stereo to mono.
     int32_t temp = 0;
-    for (int i = 0; i < source_len / 2; ++i) {
+    for (size_t i = 0; i < source_len / 2; ++i) {
       temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   } else {
     int32_t temp = 0;
-    for (int i = 0; i < source_len; ++i) {
+    for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/utility.h b/media/webrtc/trunk/webrtc/voice_engine/utility.h
index 38206959b2..4139f05cfd 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/utility.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/utility.h
@@ -24,38 +24,32 @@ class AudioFrame;
 
 namespace voe {
 
-// Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
-// Expects |dst_frame| to have its sample rate and channels members set to the
-// desired values. Updates the samples per channel member accordingly. No other
-// members will be changed.
+// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
+// to have its sample rate and channels members set to the desired values.
+// Updates the |samples_per_channel_| member accordingly.
+//
+// This version has an AudioFrame |src_frame| as input and sets the output
+// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
+// input ones.
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame);
 
-// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
-// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
-// temporary space and must be of sufficient size to hold the downmixed source
-// audio (recommend using a size of kMaxMonoDataSizeSamples).
-//
-// |dst_af| will have its data and format members (sample rate, channels and
-// samples per channel) set appropriately. No other members will be changed.
-// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
-// it shouldn't be needed.
-void DownConvertToCodecFormat(const int16_t* src_data,
-                              int samples_per_channel,
-                              int num_channels,
-                              int sample_rate_hz,
-                              int codec_num_channels,
-                              int codec_rate_hz,
-                              int16_t* mono_buffer,
-                              PushResampler<int16_t>* resampler,
-                              AudioFrame* dst_af);
+// This version has a pointer to the samples |src_data| as input and receives
+// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
+// parameters.
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
 
 void MixWithSat(int16_t target[],
-                int target_channel,
+                size_t target_channel,
                 const int16_t source[],
-                int source_channel,
-                int source_len);
+                size_t source_channel,
+                size_t source_len);
 
 }  // namespace voe
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/utility_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/utility_unittest.cc
index 53afdb692c..a04113455b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/utility_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/utility_unittest.cc
@@ -11,8 +11,9 @@
 #include <math.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/utility.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
@@ -20,11 +21,6 @@ namespace webrtc {
 namespace voe {
 namespace {
 
-enum FunctionToTest {
-  TestRemixAndResample,
-  TestDownConvertToCodecFormat
-};
-
 class UtilityTest : public ::testing::Test {
  protected:
   UtilityTest() {
@@ -35,9 +31,10 @@ class UtilityTest : public ::testing::Test {
     golden_frame_.CopyFrom(src_frame_);
   }
 
-  void RunResampleTest(int src_channels, int src_sample_rate_hz,
-                       int dst_channels, int dst_sample_rate_hz,
-                       FunctionToTest function);
+  void RunResampleTest(int src_channels,
+                       int src_sample_rate_hz,
+                       int dst_channels,
+                       int dst_sample_rate_hz);
 
   PushResampler<int16_t> resampler_;
   AudioFrame src_frame_;
@@ -53,8 +50,8 @@ void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) {
   frame->num_channels_ = 1;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = sample_rate_hz / 100;
-  for (int i = 0; i < frame->samples_per_channel_; i++) {
-    frame->data_[i] = data * i;
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame->data_[i] = static_cast<int16_t>(data * i);
   }
 }
 
@@ -71,9 +68,9 @@ void SetStereoFrame(AudioFrame* frame, float left, float right,
   frame->num_channels_ = 2;
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = sample_rate_hz / 100;
-  for (int i = 0; i < frame->samples_per_channel_; i++) {
-    frame->data_[i * 2] = left * i;
-    frame->data_[i * 2 + 1] = right * i;
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame->data_[i * 2] = static_cast<int16_t>(left * i);
+    frame->data_[i * 2 + 1] = static_cast<int16_t>(right * i);
   }
 }
 
@@ -92,14 +89,14 @@ void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
 // |test_frame|. It allows for up to a |max_delay| in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
-                 int max_delay) {
+                 size_t max_delay) {
   VerifyParams(ref_frame, test_frame);
   float best_snr = 0;
-  int best_delay = 0;
-  for (int delay = 0; delay <= max_delay; delay++) {
+  size_t best_delay = 0;
+  for (size_t delay = 0; delay <= max_delay; delay++) {
     float mse = 0;
     float variance = 0;
-    for (int i = 0; i < ref_frame.samples_per_channel_ *
+    for (size_t i = 0; i < ref_frame.samples_per_channel_ *
         ref_frame.num_channels_ - delay; i++) {
       int error = ref_frame.data_[i] - test_frame.data_[i + delay];
       mse += error * error;
@@ -113,15 +110,15 @@ float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
       best_delay = delay;
     }
   }
-  printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
+  printf("SNR=%.1f dB at delay=%" PRIuS "\n", best_snr, best_delay);
   return best_snr;
 }
 
 void VerifyFramesAreEqual(const AudioFrame& ref_frame,
                           const AudioFrame& test_frame) {
   VerifyParams(ref_frame, test_frame);
-  for (int i = 0; i < ref_frame.samples_per_channel_ * ref_frame.num_channels_;
-      i++) {
+  for (size_t i = 0;
+       i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
     EXPECT_EQ(ref_frame.data_[i], test_frame.data_[i]);
   }
 }
@@ -129,8 +126,7 @@ void VerifyFramesAreEqual(const AudioFrame& ref_frame,
 void UtilityTest::RunResampleTest(int src_channels,
                                   int src_sample_rate_hz,
                                   int dst_channels,
-                                  int dst_sample_rate_hz,
-                                  FunctionToTest function) {
+                                  int dst_sample_rate_hz) {
   PushResampler<int16_t> resampler;  // Create a new one with every test.
   const int16_t kSrcLeft = 30;  // Shouldn't overflow for any used sample rate.
   const int16_t kSrcRight = 15;
@@ -158,30 +154,16 @@ void UtilityTest::RunResampleTest(int src_channels,
       SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
   }
 
-  // The speex resampler has a known delay dependent on quality and rates,
-  // which we approximate here. Multiplying by two gives us a crude maximum
-  // for any resampling, as the old resampler typically (but not always)
-  // has lower delay.  The actual delay is calculated internally based on the
-  // filter length in the QualityMap.
-  static const int kInputKernelDelaySamples = 16*3;
-  const int max_delay = std::min(1.0f, 1/kResamplingFactor) *
-                        kInputKernelDelaySamples * dst_channels * 2;
+  // The sinc resampler has a known delay, which we compute here. Multiplying by
+  // two gives us a crude maximum for any resampling, as the old resampler
+  // typically (but not always) has lower delay.
+  static const size_t kInputKernelDelaySamples = 16;
+  const size_t max_delay = static_cast<size_t>(
+      static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
+      kInputKernelDelaySamples * dst_channels * 2);
   printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
       src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
-  if (function == TestRemixAndResample) {
-    RemixAndResample(src_frame_, &resampler, &dst_frame_);
-  } else {
-    int16_t mono_buffer[kMaxMonoDataSizeSamples];
-    DownConvertToCodecFormat(src_frame_.data_,
-                             src_frame_.samples_per_channel_,
-                             src_frame_.num_channels_,
-                             src_frame_.sample_rate_hz_,
-                             dst_frame_.num_channels_,
-                             dst_frame_.sample_rate_hz_,
-                             mono_buffer,
-                             &resampler,
-                             &dst_frame_);
-  }
+  RemixAndResample(src_frame_, &resampler, &dst_frame_);
 
   if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
     // The sinc resampler gives poor SNR at this extreme conversion, but we
@@ -235,28 +217,7 @@ TEST_F(UtilityTest, RemixAndResampleSucceeds) {
       for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
         for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
           RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
-                          kChannels[dst_channel], kSampleRates[dst_rate],
-                          TestRemixAndResample);
-        }
-      }
-    }
-  }
-}
-
-TEST_F(UtilityTest, ConvertToCodecFormatSucceeds) {
-  const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
-  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
-  const int kChannels[] = {1, 2};
-  const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
-  for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
-    for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
-      for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
-        for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
-          if (dst_rate <= src_rate && dst_channel <= src_channel) {
-            RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
-                            kChannels[src_channel], kSampleRates[dst_rate],
-                            TestDownConvertToCodecFormat);
-          }
+                          kChannels[dst_channel], kSampleRates[dst_rate]);
         }
       }
     }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc
index a310628eec..c95726339c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -10,10 +10,10 @@
 
 #include "webrtc/voice_engine/voe_audio_processing_impl.h"
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
@@ -59,8 +59,7 @@ VoEAudioProcessing* VoEAudioProcessing::GetInterface(VoiceEngine* voiceEngine) {
 
 #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
 VoEAudioProcessingImpl::VoEAudioProcessingImpl(voe::SharedData* shared)
-    : _isAecMode(kDefaultEcMode == kEcAec),
-      _shared(shared) {
+    : _isAecMode(kDefaultEcMode == kEcAec), _shared(shared) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "VoEAudioProcessingImpl::VoEAudioProcessingImpl() - ctor");
 }
@@ -104,29 +103,27 @@ int VoEAudioProcessingImpl::SetNsStatus(bool enable, NsModes mode) {
       break;
   }
 
-  if (_shared->audio_processing()->noise_suppression()->
-          set_level(nsLevel) != 0) {
+  if (_shared->audio_processing()->noise_suppression()->set_level(nsLevel) !=
+      0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetNsStatus() failed to set Ns mode");
+                          "SetNsStatus() failed to set Ns mode");
     return -1;
   }
   if (_shared->audio_processing()->noise_suppression()->Enable(enable) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetNsStatus() failed to set Ns state");
+                          "SetNsStatus() failed to set Ns state");
     return -1;
   }
 
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetNsStatus() Ns is not supported");
+                        "SetNsStatus() Ns is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetNsStatus(bool& enabled, NsModes& mode) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetNsStatus(enabled=?, mode=?)");
 #ifdef WEBRTC_VOICE_ENGINE_NR
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -151,13 +148,10 @@ int VoEAudioProcessingImpl::GetNsStatus(bool& enabled, NsModes& mode) {
       mode = kNsVeryHighSuppression;
       break;
   }
-
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetNsStatus() => enabled=% d, mode=%d", enabled, mode);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetNsStatus() Ns is not supported");
+                        "GetNsStatus() Ns is not supported");
   return -1;
 #endif
 }
@@ -174,7 +168,7 @@ int VoEAudioProcessingImpl::SetAgcStatus(bool enable, AgcModes mode) {
 #if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
   if (mode == kAgcAdaptiveAnalog) {
     _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-        "SetAgcStatus() invalid Agc mode for mobile device");
+                          "SetAgcStatus() invalid Agc mode for mobile device");
     return -1;
   }
 #endif
@@ -200,12 +194,12 @@ int VoEAudioProcessingImpl::SetAgcStatus(bool enable, AgcModes mode) {
 
   if (_shared->audio_processing()->gain_control()->set_mode(agcMode) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetAgcStatus() failed to set Agc mode");
+                          "SetAgcStatus() failed to set Agc mode");
     return -1;
   }
   if (_shared->audio_processing()->gain_control()->Enable(enable) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetAgcStatus() failed to set Agc state");
+                          "SetAgcStatus() failed to set Agc state");
     return -1;
   }
 
@@ -215,22 +209,20 @@ int VoEAudioProcessingImpl::SetAgcStatus(bool enable, AgcModes mode) {
     // used since we want to be able to provide the APM with updated mic
     // levels when the user modifies the mic level manually.
     if (_shared->audio_device()->SetAGC(enable) != 0) {
-      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
-          kTraceWarning, "SetAgcStatus() failed to set Agc mode");
+      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+                            "SetAgcStatus() failed to set Agc mode");
     }
   }
 
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetAgcStatus() Agc is not supported");
+                        "SetAgcStatus() Agc is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetAgcStatus(bool& enabled, AgcModes& mode) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetAgcStatus(enabled=?, mode=?)");
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -239,7 +231,7 @@ int VoEAudioProcessingImpl::GetAgcStatus(bool& enabled, AgcModes& mode) {
 
   enabled = _shared->audio_processing()->gain_control()->is_enabled();
   GainControl::Mode agcMode =
-    _shared->audio_processing()->gain_control()->mode();
+      _shared->audio_processing()->gain_control()->mode();
 
   switch (agcMode) {
     case GainControl::kFixedDigital:
@@ -253,12 +245,10 @@ int VoEAudioProcessingImpl::GetAgcStatus(bool& enabled, AgcModes& mode) {
       break;
   }
 
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetAgcStatus() => enabled=%d, mode=%d", enabled, mode);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetAgcStatus() Agc is not supported");
+                        "GetAgcStatus() Agc is not supported");
   return -1;
 #endif
 }
@@ -273,22 +263,23 @@ int VoEAudioProcessingImpl::SetAgcConfig(AgcConfig config) {
   }
 
   if (_shared->audio_processing()->gain_control()->set_target_level_dbfs(
-      config.targetLeveldBOv) != 0) {
+          config.targetLeveldBOv) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetAgcConfig() failed to set target peak |level|"
-        " (or envelope) of the Agc");
+                          "SetAgcConfig() failed to set target peak |level|"
+                          " (or envelope) of the Agc");
     return -1;
   }
   if (_shared->audio_processing()->gain_control()->set_compression_gain_db(
-        config.digitalCompressionGaindB) != 0) {
+          config.digitalCompressionGaindB) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetAgcConfig() failed to set the range in |gain| "
-        "the digital compression stage may apply");
+                          "SetAgcConfig() failed to set the range in |gain| "
+                          "the digital compression stage may apply");
     return -1;
   }
   if (_shared->audio_processing()->gain_control()->enable_limiter(
-        config.limiterEnable) != 0) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceError,
+          config.limiterEnable) != 0) {
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceError,
         "SetAgcConfig() failed to set hard limiter to the signal");
     return -1;
   }
@@ -296,14 +287,12 @@ int VoEAudioProcessingImpl::SetAgcConfig(AgcConfig config) {
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetAgcConfig() EC is not supported");
+                        "SetAgcConfig() EC is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetAgcConfig(AgcConfig& config) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetAgcConfig(config=?)");
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -311,23 +300,16 @@ int VoEAudioProcessingImpl::GetAgcConfig(AgcConfig& config) {
   }
 
   config.targetLeveldBOv =
-    _shared->audio_processing()->gain_control()->target_level_dbfs();
+      _shared->audio_processing()->gain_control()->target_level_dbfs();
   config.digitalCompressionGaindB =
-    _shared->audio_processing()->gain_control()->compression_gain_db();
+      _shared->audio_processing()->gain_control()->compression_gain_db();
   config.limiterEnable =
-    _shared->audio_processing()->gain_control()->is_limiter_enabled();
-
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetAgcConfig() => targetLeveldBOv=%u, "
-                  "digitalCompressionGaindB=%u, limiterEnable=%d",
-               config.targetLeveldBOv,
-               config.digitalCompressionGaindB,
-               config.limiterEnable);
+      _shared->audio_processing()->gain_control()->is_limiter_enabled();
 
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetAgcConfig() EC is not supported");
+                        "GetAgcConfig() EC is not supported");
   return -1;
 #endif
 }
@@ -335,7 +317,6 @@ int VoEAudioProcessingImpl::GetAgcConfig(AgcConfig& config) {
 int VoEAudioProcessingImpl::SetRxNsStatus(int channel,
                                           bool enable,
                                           NsModes mode) {
-  LOG_API3(channel, enable, mode);
 #ifdef WEBRTC_VOICE_ENGINE_NR
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -346,13 +327,13 @@ int VoEAudioProcessingImpl::SetRxNsStatus(int channel,
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "SetRxNsStatus() failed to locate channel");
+                          "SetRxNsStatus() failed to locate channel");
     return -1;
   }
   return channelPtr->SetRxNsStatus(enable, mode);
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetRxNsStatus() NS is not supported");
+                        "SetRxNsStatus() NS is not supported");
   return -1;
 #endif
 }
@@ -360,8 +341,6 @@ int VoEAudioProcessingImpl::SetRxNsStatus(int channel,
 int VoEAudioProcessingImpl::GetRxNsStatus(int channel,
                                           bool& enabled,
                                           NsModes& mode) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetRxNsStatus(channel=%d, enable=?, mode=?)", channel);
 #ifdef WEBRTC_VOICE_ENGINE_NR
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -372,13 +351,13 @@ int VoEAudioProcessingImpl::GetRxNsStatus(int channel,
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "GetRxNsStatus() failed to locate channel");
+                          "GetRxNsStatus() failed to locate channel");
     return -1;
   }
   return channelPtr->GetRxNsStatus(enabled, mode);
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetRxNsStatus() NS is not supported");
+                        "GetRxNsStatus() NS is not supported");
   return -1;
 #endif
 }
@@ -387,8 +366,8 @@ int VoEAudioProcessingImpl::SetRxAgcStatus(int channel,
                                            bool enable,
                                            AgcModes mode) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetRxAgcStatus(channel=%d, enable=%d, mode=%d)",
-               channel, (int)enable, (int)mode);
+               "SetRxAgcStatus(channel=%d, enable=%d, mode=%d)", channel,
+               (int)enable, (int)mode);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -399,13 +378,13 @@ int VoEAudioProcessingImpl::SetRxAgcStatus(int channel,
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "SetRxAgcStatus() failed to locate channel");
+                          "SetRxAgcStatus() failed to locate channel");
     return -1;
   }
   return channelPtr->SetRxAgcStatus(enable, mode);
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetRxAgcStatus() Agc is not supported");
+                        "SetRxAgcStatus() Agc is not supported");
   return -1;
 #endif
 }
@@ -413,8 +392,6 @@ int VoEAudioProcessingImpl::SetRxAgcStatus(int channel,
 int VoEAudioProcessingImpl::GetRxAgcStatus(int channel,
                                            bool& enabled,
                                            AgcModes& mode) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetRxAgcStatus(channel=%d, enable=?, mode=?)", channel);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -425,19 +402,18 @@ int VoEAudioProcessingImpl::GetRxAgcStatus(int channel,
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "GetRxAgcStatus() failed to locate channel");
+                          "GetRxAgcStatus() failed to locate channel");
     return -1;
   }
   return channelPtr->GetRxAgcStatus(enabled, mode);
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetRxAgcStatus() Agc is not supported");
+                        "GetRxAgcStatus() Agc is not supported");
   return -1;
 #endif
 }
 
-int VoEAudioProcessingImpl::SetRxAgcConfig(int channel,
-                                           AgcConfig config) {
+int VoEAudioProcessingImpl::SetRxAgcConfig(int channel, AgcConfig config) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetRxAgcConfig(channel=%d)", channel);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
@@ -450,20 +426,18 @@ int VoEAudioProcessingImpl::SetRxAgcConfig(int channel,
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-      "SetRxAgcConfig() failed to locate channel");
+                          "SetRxAgcConfig() failed to locate channel");
     return -1;
   }
   return channelPtr->SetRxAgcConfig(config);
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetRxAgcConfig() Agc is not supported");
+                        "SetRxAgcConfig() Agc is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetRxAgcConfig(int channel, AgcConfig& config) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetRxAgcConfig(channel=%d)", channel);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -474,13 +448,13 @@ int VoEAudioProcessingImpl::GetRxAgcConfig(int channel, AgcConfig& config) {
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "GetRxAgcConfig() failed to locate channel");
+                          "GetRxAgcConfig() failed to locate channel");
     return -1;
   }
   return channelPtr->GetRxAgcConfig(config);
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetRxAgcConfig() Agc is not supported");
+                        "GetRxAgcConfig() Agc is not supported");
   return -1;
 #endif
 }
@@ -494,11 +468,11 @@ bool VoEAudioProcessing::DriftCompensationSupported() {
 }
 
 int VoEAudioProcessingImpl::EnableDriftCompensation(bool enable) {
-  LOG_API1(enable);
   WEBRTC_VOICE_INIT_CHECK();
 
   if (!DriftCompensationSupported()) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceWarning,
         "Drift compensation is not supported on this platform.");
     return -1;
   }
@@ -506,14 +480,13 @@ int VoEAudioProcessingImpl::EnableDriftCompensation(bool enable) {
   EchoCancellation* aec = _shared->audio_processing()->echo_cancellation();
   if (aec->enable_drift_compensation(enable) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "aec->enable_drift_compensation() failed");
+                          "aec->enable_drift_compensation() failed");
     return -1;
   }
   return 0;
 }
 
 bool VoEAudioProcessingImpl::DriftCompensationEnabled() {
-  LOG_API0();
   WEBRTC_VOICE_INIT_CHECK_BOOL();
 
   EchoCancellation* aec = _shared->audio_processing()->echo_cancellation();
@@ -530,41 +503,43 @@ int VoEAudioProcessingImpl::SetEcStatus(bool enable, EcModes mode) {
   }
 
   // AEC mode
-  if ((mode == kEcDefault) ||
-      (mode == kEcConference) ||
-      (mode == kEcAec) ||
-      ((mode == kEcUnchanged) &&
-       (_isAecMode == true))) {
+  if ((mode == kEcDefault) || (mode == kEcConference) || (mode == kEcAec) ||
+      ((mode == kEcUnchanged) && (_isAecMode == true))) {
     if (enable) {
       // Disable the AECM before enable the AEC
       if (_shared->audio_processing()->echo_control_mobile()->is_enabled()) {
         _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
-            "SetEcStatus() disable AECM before enabling AEC");
-        if (_shared->audio_processing()->echo_control_mobile()->
-            Enable(false) != 0) {
+                              "SetEcStatus() disable AECM before enabling AEC");
+        if (_shared->audio_processing()->echo_control_mobile()->Enable(false) !=
+            0) {
           _shared->SetLastError(VE_APM_ERROR, kTraceError,
-              "SetEcStatus() failed to disable AECM");
+                                "SetEcStatus() failed to disable AECM");
           return -1;
         }
       }
     }
     if (_shared->audio_processing()->echo_cancellation()->Enable(enable) != 0) {
       _shared->SetLastError(VE_APM_ERROR, kTraceError,
-          "SetEcStatus() failed to set AEC state");
+                            "SetEcStatus() failed to set AEC state");
       return -1;
     }
     if (mode == kEcConference) {
-      if (_shared->audio_processing()->echo_cancellation()->
-          set_suppression_level(EchoCancellation::kHighSuppression) != 0) {
-        _shared->SetLastError(VE_APM_ERROR, kTraceError,
+      if (_shared->audio_processing()
+              ->echo_cancellation()
+              ->set_suppression_level(EchoCancellation::kHighSuppression) !=
+          0) {
+        _shared->SetLastError(
+            VE_APM_ERROR, kTraceError,
             "SetEcStatus() failed to set aggressiveness to high");
         return -1;
       }
     } else {
-      if (_shared->audio_processing()->echo_cancellation()->
-          set_suppression_level(
-            EchoCancellation::kModerateSuppression) != 0) {
-        _shared->SetLastError(VE_APM_ERROR, kTraceError,
+      if (_shared->audio_processing()
+              ->echo_cancellation()
+              ->set_suppression_level(EchoCancellation::kModerateSuppression) !=
+          0) {
+        _shared->SetLastError(
+            VE_APM_ERROR, kTraceError,
             "SetEcStatus() failed to set aggressiveness to moderate");
         return -1;
       }
@@ -572,45 +547,42 @@ int VoEAudioProcessingImpl::SetEcStatus(bool enable, EcModes mode) {
 
     _isAecMode = true;
   } else if ((mode == kEcAecm) ||
-             ((mode == kEcUnchanged) &&
-              (_isAecMode == false))) {
+             ((mode == kEcUnchanged) && (_isAecMode == false))) {
     if (enable) {
       // Disable the AEC before enable the AECM
       if (_shared->audio_processing()->echo_cancellation()->is_enabled()) {
         _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
-            "SetEcStatus() disable AEC before enabling AECM");
-        if (_shared->audio_processing()->echo_cancellation()->
-            Enable(false) != 0) {
+                              "SetEcStatus() disable AEC before enabling AECM");
+        if (_shared->audio_processing()->echo_cancellation()->Enable(false) !=
+            0) {
           _shared->SetLastError(VE_APM_ERROR, kTraceError,
-              "SetEcStatus() failed to disable AEC");
+                                "SetEcStatus() failed to disable AEC");
           return -1;
         }
       }
     }
-    if (_shared->audio_processing()->echo_control_mobile()->
-        Enable(enable) != 0) {
+    if (_shared->audio_processing()->echo_control_mobile()->Enable(enable) !=
+        0) {
       _shared->SetLastError(VE_APM_ERROR, kTraceError,
-          "SetEcStatus() failed to set AECM state");
+                            "SetEcStatus() failed to set AECM state");
       return -1;
     }
     _isAecMode = false;
   } else {
     _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-                                   "SetEcStatus() invalid EC mode");
+                          "SetEcStatus() invalid EC mode");
     return -1;
   }
 
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetEcStatus() EC is not supported");
+                        "SetEcStatus() EC is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetEcStatus(bool& enabled, EcModes& mode) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcStatus()");
 #ifdef WEBRTC_VOICE_ENGINE_ECHO
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -622,17 +594,13 @@ int VoEAudioProcessingImpl::GetEcStatus(bool& enabled, EcModes& mode) {
     enabled = _shared->audio_processing()->echo_cancellation()->is_enabled();
   } else {
     mode = kEcAecm;
-    enabled = _shared->audio_processing()->echo_control_mobile()->
-              is_enabled();
+    enabled = _shared->audio_processing()->echo_control_mobile()->is_enabled();
   }
 
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcStatus() => enabled=%i, mode=%i",
-               enabled, (int)mode);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetEcStatus() EC is not supported");
+                        "GetEcStatus() EC is not supported");
   return -1;
 #endif
 }
@@ -644,8 +612,6 @@ void VoEAudioProcessingImpl::SetDelayOffsetMs(int offset) {
 }
 
 int VoEAudioProcessingImpl::DelayOffsetMs() {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "DelayOffsetMs()");
   return _shared->audio_processing()->delay_offset_ms();
 }
 
@@ -679,16 +645,16 @@ int VoEAudioProcessingImpl::SetAecmMode(AecmModes mode, bool enableCNG) {
       break;
   }
 
-
-  if (_shared->audio_processing()->echo_control_mobile()->
-      set_routing_mode(aecmMode) != 0) {
+  if (_shared->audio_processing()->echo_control_mobile()->set_routing_mode(
+          aecmMode) != 0) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetAECMMode() failed to set AECM routing mode");
+                          "SetAECMMode() failed to set AECM routing mode");
     return -1;
   }
-  if (_shared->audio_processing()->echo_control_mobile()->
-      enable_comfort_noise(enableCNG) != 0) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceError,
+  if (_shared->audio_processing()->echo_control_mobile()->enable_comfort_noise(
+          enableCNG) != 0) {
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceError,
         "SetAECMMode() failed to set comfort noise state for AECM");
     return -1;
   }
@@ -696,14 +662,12 @@ int VoEAudioProcessingImpl::SetAecmMode(AecmModes mode, bool enableCNG) {
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetAECMMode() EC is not supported");
+                        "SetAECMMode() EC is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetAecmMode(AecmModes& mode, bool& enabledCNG) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetAECMMode(mode=?)");
 #ifdef WEBRTC_VOICE_ENGINE_ECHO
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -714,8 +678,9 @@ int VoEAudioProcessingImpl::GetAecmMode(AecmModes& mode, bool& enabledCNG) {
 
   EchoControlMobile::RoutingMode aecmMode =
       _shared->audio_processing()->echo_control_mobile()->routing_mode();
-  enabledCNG = _shared->audio_processing()->echo_control_mobile()->
-      is_comfort_noise_enabled();
+  enabledCNG = _shared->audio_processing()
+                   ->echo_control_mobile()
+                   ->is_comfort_noise_enabled();
 
   switch (aecmMode) {
     case EchoControlMobile::kQuietEarpieceOrHeadset:
@@ -738,7 +703,7 @@ int VoEAudioProcessingImpl::GetAecmMode(AecmModes& mode, bool& enabledCNG) {
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "GetAECMMode() EC is not supported");
+                        "GetAECMMode() EC is not supported");
   return -1;
 #endif
 }
@@ -749,7 +714,7 @@ int VoEAudioProcessingImpl::EnableHighPassFilter(bool enable) {
   if (_shared->audio_processing()->high_pass_filter()->Enable(enable) !=
       AudioProcessing::kNoError) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "HighPassFilter::Enable() failed.");
+                          "HighPassFilter::Enable() failed.");
     return -1;
   }
 
@@ -757,14 +722,11 @@ int VoEAudioProcessingImpl::EnableHighPassFilter(bool enable) {
 }
 
 bool VoEAudioProcessingImpl::IsHighPassFilterEnabled() {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "IsHighPassFilterEnabled()");
   return _shared->audio_processing()->high_pass_filter()->is_enabled();
 }
 
-int VoEAudioProcessingImpl::RegisterRxVadObserver(
-  int channel,
-  VoERxVadCallback& observer) {
+int VoEAudioProcessingImpl::RegisterRxVadObserver(int channel,
+                                                  VoERxVadCallback& observer) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "RegisterRxVadObserver()");
   if (!_shared->statistics().Initialized()) {
@@ -775,7 +737,7 @@ int VoEAudioProcessingImpl::RegisterRxVadObserver(
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "RegisterRxVadObserver() failed to locate channel");
+                          "RegisterRxVadObserver() failed to locate channel");
     return -1;
   }
   return channelPtr->RegisterRxVadObserver(observer);
@@ -792,7 +754,7 @@ int VoEAudioProcessingImpl::DeRegisterRxVadObserver(int channel) {
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "DeRegisterRxVadObserver() failed to locate channel");
+                          "DeRegisterRxVadObserver() failed to locate channel");
     return -1;
   }
 
@@ -811,7 +773,7 @@ int VoEAudioProcessingImpl::VoiceActivityIndicator(int channel) {
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-        "DeRegisterRxVadObserver() failed to locate channel");
+                          "DeRegisterRxVadObserver() failed to locate channel");
     return -1;
   }
   int activity(-1);
@@ -829,25 +791,23 @@ int VoEAudioProcessingImpl::SetEcMetricsStatus(bool enable) {
     return -1;
   }
 
-  if ((_shared->audio_processing()->echo_cancellation()->enable_metrics(enable)
-       != 0) ||
+  if ((_shared->audio_processing()->echo_cancellation()->enable_metrics(
+           enable) != 0) ||
       (_shared->audio_processing()->echo_cancellation()->enable_delay_logging(
-         enable) != 0)) {
+           enable) != 0)) {
     _shared->SetLastError(VE_APM_ERROR, kTraceError,
-        "SetEcMetricsStatus() unable to set EC metrics mode");
+                          "SetEcMetricsStatus() unable to set EC metrics mode");
     return -1;
   }
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetEcStatus() EC is not supported");
+                        "SetEcStatus() EC is not supported");
   return -1;
 #endif
 }
 
 int VoEAudioProcessingImpl::GetEcMetricsStatus(bool& enabled) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcMetricsStatus(enabled=?)");
 #ifdef WEBRTC_VOICE_ENGINE_ECHO
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -855,24 +815,24 @@ int VoEAudioProcessingImpl::GetEcMetricsStatus(bool& enabled) {
   }
 
   bool echo_mode =
-    _shared->audio_processing()->echo_cancellation()->are_metrics_enabled();
-  bool delay_mode = _shared->audio_processing()->echo_cancellation()->
-      is_delay_logging_enabled();
+      _shared->audio_processing()->echo_cancellation()->are_metrics_enabled();
+  bool delay_mode = _shared->audio_processing()
+                        ->echo_cancellation()
+                        ->is_delay_logging_enabled();
 
   if (echo_mode != delay_mode) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceError,
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceError,
         "GetEcMetricsStatus() delay logging and echo mode are not the same");
     return -1;
   }
 
   enabled = echo_mode;
 
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcMetricsStatus() => enabled=%d", enabled);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetEcStatus() EC is not supported");
+                        "SetEcStatus() EC is not supported");
   return -1;
 #endif
 }
@@ -881,15 +841,14 @@ int VoEAudioProcessingImpl::GetEchoMetrics(int& ERL,
                                            int& ERLE,
                                            int& RERL,
                                            int& A_NLP) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEchoMetrics(ERL=?, ERLE=?, RERL=?, A_NLP=?)");
 #ifdef WEBRTC_VOICE_ENGINE_ECHO
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
   if (!_shared->audio_processing()->echo_cancellation()->is_enabled()) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceWarning,
         "GetEchoMetrics() AudioProcessingModule AEC is not enabled");
     return -1;
   }
@@ -909,13 +868,10 @@ int VoEAudioProcessingImpl::GetEchoMetrics(int& ERL,
   RERL = echoMetrics.residual_echo_return_loss.instant;
   A_NLP = echoMetrics.a_nlp.instant;
 
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEchoMetrics() => ERL=%d, ERLE=%d, RERL=%d, A_NLP=%d",
-               ERL, ERLE, RERL, A_NLP);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetEcStatus() EC is not supported");
+                        "SetEcStatus() EC is not supported");
   return -1;
 #endif
 }
@@ -923,15 +879,14 @@ int VoEAudioProcessingImpl::GetEchoMetrics(int& ERL,
 int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
                                               int& delay_std,
                                               float& fraction_poor_delays) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcDelayMetrics(median=?, std=?, fraction_poor_delays=?)");
 #ifdef WEBRTC_VOICE_ENGINE_ECHO
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
   if (!_shared->audio_processing()->echo_cancellation()->is_enabled()) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceWarning,
         "GetEcDelayMetrics() AudioProcessingModule AEC is not enabled");
     return -1;
   }
@@ -941,7 +896,7 @@ int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
   float poor_fraction = 0;
   // Get delay-logging values from Audio Processing Module.
   if (_shared->audio_processing()->echo_cancellation()->GetDelayMetrics(
-        &median, &std, &poor_fraction)) {
+          &median, &std, &poor_fraction)) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetEcDelayMetrics(), AudioProcessingModule delay-logging "
                  "error");
@@ -953,14 +908,10 @@ int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
   delay_std = std;
   fraction_poor_delays = poor_fraction;
 
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcDelayMetrics() => delay_median=%d, delay_std=%d, "
-               "fraction_poor_delays=%f", delay_median, delay_std,
-               fraction_poor_delays);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetEcStatus() EC is not supported");
+                        "SetEcStatus() EC is not supported");
   return -1;
 #endif
 }
@@ -1014,12 +965,13 @@ int VoEAudioProcessingImpl::SetTypingDetectionStatus(bool enable) {
 
   if (_shared->audio_processing()->voice_detection()->Enable(enable)) {
     _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
-        "SetTypingDetectionStatus() failed to set VAD state");
+                          "SetTypingDetectionStatus() failed to set VAD state");
     return -1;
   }
   if (_shared->audio_processing()->voice_detection()->set_likelihood(
           VoiceDetection::kVeryLowLikelihood)) {
-    _shared->SetLastError(VE_APM_ERROR, kTraceWarning,
+    _shared->SetLastError(
+        VE_APM_ERROR, kTraceWarning,
         "SetTypingDetectionStatus() failed to set VAD likelihood to low");
     return -1;
   }
@@ -1029,8 +981,6 @@ int VoEAudioProcessingImpl::SetTypingDetectionStatus(bool enable) {
 }
 
 int VoEAudioProcessingImpl::GetTypingDetectionStatus(bool& enabled) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetTypingDetectionStatus()");
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -1043,10 +993,7 @@ int VoEAudioProcessingImpl::GetTypingDetectionStatus(bool& enabled) {
   return 0;
 }
 
-
-int VoEAudioProcessingImpl::TimeSinceLastTyping(int &seconds) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "TimeSinceLastTyping()");
+int VoEAudioProcessingImpl::TimeSinceLastTyping(int& seconds) {
 #if !defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
   NOT_SUPPORTED(_shared->statistics());
 #else
@@ -1056,16 +1003,13 @@ int VoEAudioProcessingImpl::TimeSinceLastTyping(int &seconds) {
   }
   // Check if typing detection is enabled
   bool enabled = _shared->audio_processing()->voice_detection()->is_enabled();
-  if (enabled)
-  {
+  if (enabled) {
     _shared->transmit_mixer()->TimeSinceLastTyping(seconds);
     return 0;
-  }
-  else
-  {
+  } else {
     _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-      "SetTypingDetectionStatus is not enabled");
-  return -1;
+                          "SetTypingDetectionStatus is not enabled");
+    return -1;
   }
 #endif
 }
@@ -1084,18 +1028,17 @@ int VoEAudioProcessingImpl::SetTypingDetectionParameters(int timeWindow,
     _shared->statistics().SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
-  return (_shared->transmit_mixer()->SetTypingDetectionParameters(timeWindow,
-      costPerTyping, reportingThreshold, penaltyDecay, typeEventDelay));
+  return (_shared->transmit_mixer()->SetTypingDetectionParameters(
+      timeWindow, costPerTyping, reportingThreshold, penaltyDecay,
+      typeEventDelay));
 #endif
 }
 
 void VoEAudioProcessingImpl::EnableStereoChannelSwapping(bool enable) {
-  LOG_API1(enable);
   _shared->transmit_mixer()->EnableStereoChannelSwapping(enable);
 }
 
 bool VoEAudioProcessingImpl::IsStereoChannelSwappingEnabled() {
-  LOG_API0();
   return _shared->transmit_mixer()->IsStereoChannelSwappingEnabled();
 }
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h
index 26f7eec745..63a60dcb46 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h
@@ -19,90 +19,90 @@ namespace webrtc {
 
 class VoEAudioProcessingImpl : public VoEAudioProcessing {
  public:
-  virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged);
+  int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) override;
 
-  virtual int GetNsStatus(bool& enabled, NsModes& mode);
+  int GetNsStatus(bool& enabled, NsModes& mode) override;
 
-  virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged);
+  int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) override;
 
-  virtual int GetAgcStatus(bool& enabled, AgcModes& mode);
+  int GetAgcStatus(bool& enabled, AgcModes& mode) override;
 
-  virtual int SetAgcConfig(AgcConfig config);
+  int SetAgcConfig(AgcConfig config) override;
 
-  virtual int GetAgcConfig(AgcConfig& config);
+  int GetAgcConfig(AgcConfig& config) override;
 
-  virtual int SetRxNsStatus(int channel,
-                            bool enable,
-                            NsModes mode = kNsUnchanged);
+  int SetRxNsStatus(int channel,
+                    bool enable,
+                    NsModes mode = kNsUnchanged) override;
 
-  virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode);
+  int GetRxNsStatus(int channel, bool& enabled, NsModes& mode) override;
 
-  virtual int SetRxAgcStatus(int channel,
-                             bool enable,
-                             AgcModes mode = kAgcUnchanged);
+  int SetRxAgcStatus(int channel,
+                     bool enable,
+                     AgcModes mode = kAgcUnchanged) override;
 
-  virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode);
+  int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode) override;
 
-  virtual int SetRxAgcConfig(int channel, AgcConfig config);
+  int SetRxAgcConfig(int channel, AgcConfig config) override;
 
-  virtual int GetRxAgcConfig(int channel, AgcConfig& config);
+  int GetRxAgcConfig(int channel, AgcConfig& config) override;
 
-  virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged);
-  virtual int GetEcStatus(bool& enabled, EcModes& mode);
-  virtual int EnableDriftCompensation(bool enable);
-  virtual bool DriftCompensationEnabled();
+  int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) override;
+  int GetEcStatus(bool& enabled, EcModes& mode) override;
+  int EnableDriftCompensation(bool enable) override;
+  bool DriftCompensationEnabled() override;
 
-  virtual void SetDelayOffsetMs(int offset);
-  virtual int DelayOffsetMs();
+  void SetDelayOffsetMs(int offset) override;
+  int DelayOffsetMs() override;
 
-  virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
-                          bool enableCNG = true);
+  int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
+                  bool enableCNG = true) override;
 
-  virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG);
+  int GetAecmMode(AecmModes& mode, bool& enabledCNG) override;
 
-  virtual int EnableHighPassFilter(bool enable);
-  virtual bool IsHighPassFilterEnabled();
+  int EnableHighPassFilter(bool enable) override;
+  bool IsHighPassFilterEnabled() override;
 
-  virtual int RegisterRxVadObserver(int channel,
-                                    VoERxVadCallback& observer);
+  int RegisterRxVadObserver(int channel, VoERxVadCallback& observer) override;
 
-  virtual int DeRegisterRxVadObserver(int channel);
+  int DeRegisterRxVadObserver(int channel) override;
 
-  virtual int VoiceActivityIndicator(int channel);
+  int VoiceActivityIndicator(int channel) override;
 
-  virtual int SetEcMetricsStatus(bool enable);
+  int SetEcMetricsStatus(bool enable) override;
 
-  virtual int GetEcMetricsStatus(bool& enabled);
+  int GetEcMetricsStatus(bool& enabled) override;
 
-  virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP);
+  int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) override;
 
-  virtual int GetEcDelayMetrics(int& delay_median, int& delay_std,
-                                float& fraction_poor_delays);
+  int GetEcDelayMetrics(int& delay_median,
+                        int& delay_std,
+                        float& fraction_poor_delays) override;
 
-  virtual int StartDebugRecording(const char* fileNameUTF8);
-  virtual int StartDebugRecording(FILE* file_handle);
+  int StartDebugRecording(const char* fileNameUTF8) override;
+  int StartDebugRecording(FILE* file_handle) override;
 
-  virtual int StopDebugRecording();
+  int StopDebugRecording() override;
 
-  virtual int SetTypingDetectionStatus(bool enable);
+  int SetTypingDetectionStatus(bool enable) override;
 
-  virtual int GetTypingDetectionStatus(bool& enabled);
+  int GetTypingDetectionStatus(bool& enabled) override;
 
-  virtual int TimeSinceLastTyping(int &seconds);
+  int TimeSinceLastTyping(int& seconds) override;
 
   // TODO(niklase) Remove default argument as soon as libJingle is updated!
-  virtual int SetTypingDetectionParameters(int timeWindow,
-                                           int costPerTyping,
-                                           int reportingThreshold,
-                                           int penaltyDecay,
-                                           int typeEventDelay = 0);
+  int SetTypingDetectionParameters(int timeWindow,
+                                   int costPerTyping,
+                                   int reportingThreshold,
+                                   int penaltyDecay,
+                                   int typeEventDelay = 0) override;
 
-  virtual void EnableStereoChannelSwapping(bool enable);
-  virtual bool IsStereoChannelSwappingEnabled();
+  void EnableStereoChannelSwapping(bool enable) override;
+  bool IsStereoChannelSwappingEnabled() override;
 
  protected:
   VoEAudioProcessingImpl(voe::SharedData* shared);
-  virtual ~VoEAudioProcessingImpl();
+  ~VoEAudioProcessingImpl() override;
 
  private:
   bool _isAecMode;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_unittest.cc
index 8916ef1cd5..0d725bcc9a 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_unittest.cc
@@ -22,8 +22,7 @@ class VoEAudioProcessingTest : public ::testing::Test {
   VoEAudioProcessingTest()
       : voe_(VoiceEngine::Create()),
         base_(VoEBase::GetInterface(voe_)),
-        audioproc_(VoEAudioProcessing::GetInterface(voe_)) {
-  }
+        audioproc_(VoEAudioProcessing::GetInterface(voe_)) {}
 
   virtual ~VoEAudioProcessingTest() {
     base_->Terminate();
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc
index 430ee40c67..410e7e3a70 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc
@@ -10,14 +10,15 @@
 
 #include "webrtc/voice_engine/voe_base_impl.h"
 
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/common.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/output_mixer.h"
@@ -25,179 +26,108 @@
 #include "webrtc/voice_engine/utility.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine)
-{
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) {
+  if (nullptr == voiceEngine) {
+    return nullptr;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 }
 
-VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared) :
-    _voiceEngineObserverPtr(NULL),
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _voiceEngineObserver(false), _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl() - ctor");
+VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared)
+    : voiceEngineObserverPtr_(nullptr),
+      callbackCritSect_(*CriticalSectionWrapper::CreateCriticalSection()),
+      shared_(shared) {}
+
+VoEBaseImpl::~VoEBaseImpl() {
+  TerminateInternal();
+  delete &callbackCritSect_;
 }
 
-VoEBaseImpl::~VoEBaseImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "~VoEBaseImpl() - dtor");
-
-    TerminateInternal();
-
-    delete &_callbackCritSect;
+void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
+  CriticalSectionScoped cs(&callbackCritSect_);
+  int errCode = 0;
+  if (error == AudioDeviceObserver::kRecordingError) {
+    errCode = VE_RUNTIME_REC_ERROR;
+    LOG_F(LS_ERROR) << "VE_RUNTIME_REC_ERROR";
+  } else if (error == AudioDeviceObserver::kPlayoutError) {
+    errCode = VE_RUNTIME_PLAY_ERROR;
+    LOG_F(LS_ERROR) << "VE_RUNTIME_PLAY_ERROR";
+  }
+  if (voiceEngineObserverPtr_) {
+    // Deliver callback (-1 <=> no channel dependency)
+    voiceEngineObserverPtr_->CallbackOnError(-1, errCode);
+  }
 }
 
-void VoEBaseImpl::OnErrorIsReported(ErrorCode error)
-{
-    CriticalSectionScoped cs(&_callbackCritSect);
-    if (_voiceEngineObserver)
-    {
-        if (_voiceEngineObserverPtr)
-        {
-            int errCode(0);
-            if (error == AudioDeviceObserver::kRecordingError)
-            {
-                errCode = VE_RUNTIME_REC_ERROR;
-                WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                    VoEId(_shared->instance_id(), -1),
-                    "VoEBaseImpl::OnErrorIsReported() => VE_RUNTIME_REC_ERROR");
-            }
-            else if (error == AudioDeviceObserver::kPlayoutError)
-            {
-                errCode = VE_RUNTIME_PLAY_ERROR;
-                WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                    VoEId(_shared->instance_id(), -1),
-                    "VoEBaseImpl::OnErrorIsReported() => "
-                    "VE_RUNTIME_PLAY_ERROR");
-            }
-            // Deliver callback (-1 <=> no channel dependency)
-            _voiceEngineObserverPtr->CallbackOnError(-1, errCode);
-        }
-    }
+void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) {
+  CriticalSectionScoped cs(&callbackCritSect_);
+  int warningCode = 0;
+  if (warning == AudioDeviceObserver::kRecordingWarning) {
+    warningCode = VE_RUNTIME_REC_WARNING;
+    LOG_F(LS_WARNING) << "VE_RUNTIME_REC_WARNING";
+  } else if (warning == AudioDeviceObserver::kPlayoutWarning) {
+    warningCode = VE_RUNTIME_PLAY_WARNING;
+    LOG_F(LS_WARNING) << "VE_RUNTIME_PLAY_WARNING";
+  }
+  if (voiceEngineObserverPtr_) {
+    // Deliver callback (-1 <=> no channel dependency)
+    voiceEngineObserverPtr_->CallbackOnError(-1, warningCode);
+  }
 }
 
-void VoEBaseImpl::OnWarningIsReported(WarningCode warning)
-{
-    CriticalSectionScoped cs(&_callbackCritSect);
-    if (_voiceEngineObserver)
-    {
-        if (_voiceEngineObserverPtr)
-        {
-            int warningCode(0);
-            if (warning == AudioDeviceObserver::kRecordingWarning)
-            {
-                warningCode = VE_RUNTIME_REC_WARNING;
-                WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                    VoEId(_shared->instance_id(), -1),
-                    "VoEBaseImpl::OnErrorIsReported() => "
-                    "VE_RUNTIME_REC_WARNING");
-            }
-            else if (warning == AudioDeviceObserver::kPlayoutWarning)
-            {
-                warningCode = VE_RUNTIME_PLAY_WARNING;
-                WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                    VoEId(_shared->instance_id(), -1),
-                    "VoEBaseImpl::OnErrorIsReported() => "
-                    "VE_RUNTIME_PLAY_WARNING");
-            }
-            // Deliver callback (-1 <=> no channel dependency)
-            _voiceEngineObserverPtr->CallbackOnError(-1, warningCode);
-        }
-    }
+int32_t VoEBaseImpl::RecordedDataIsAvailable(const void* audioSamples,
+                                             const size_t nSamples,
+                                             const size_t nBytesPerSample,
+                                             const size_t nChannels,
+                                             const uint32_t samplesPerSec,
+                                             const uint32_t totalDelayMS,
+                                             const int32_t clockDrift,
+                                             const uint32_t currentMicLevel,
+                                             const bool keyPressed,
+                                             uint32_t& newMicLevel) {
+  newMicLevel = static_cast<uint32_t>(ProcessRecordedDataWithAPM(
+      nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples,
+      totalDelayMS, clockDrift, currentMicLevel, keyPressed));
+  return 0;
 }
 
-int32_t VoEBaseImpl::RecordedDataIsAvailable(
-        const void* audioSamples,
-        uint32_t nSamples,
-        uint8_t nBytesPerSample,
-        uint8_t nChannels,
-        uint32_t samplesPerSec,
-        uint32_t totalDelayMS,
-        int32_t clockDrift,
-        uint32_t micLevel,
-        bool keyPressed,
-        uint32_t& newMicLevel)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, "
-                     "nBytesPerSample=%u, nChannels=%u, samplesPerSec=%u, "
-                     "totalDelayMS=%u, clockDrift=%d, micLevel=%u)",
-                 nSamples, nBytesPerSample, nChannels, samplesPerSec,
-                 totalDelayMS, clockDrift, micLevel);
-    newMicLevel = static_cast<uint32_t>(ProcessRecordedDataWithAPM(
-        NULL, 0, audioSamples, samplesPerSec, nChannels, nSamples,
-        totalDelayMS, clockDrift, micLevel, keyPressed));
-
-    return 0;
-}
-
-int32_t VoEBaseImpl::NeedMorePlayData(
-        uint32_t nSamples,
-        uint8_t nBytesPerSample,
-        uint8_t nChannels,
-        uint32_t samplesPerSec,
-        void* audioSamples,
-        uint32_t& nSamplesOut,
-        int64_t* elapsed_time_ms,
-        int64_t* ntp_time_ms)
-{
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
-               "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
-               nSamples, nBytesPerSample, nChannels, samplesPerSec);
-
-  GetPlayoutData(static_cast<int>(samplesPerSec),
-                 static_cast<int>(nChannels),
-                 static_cast<int>(nSamples), true, audioSamples,
-                 elapsed_time_ms, ntp_time_ms);
-
-  nSamplesOut = _audioFrame.samples_per_channel_;
-
+int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
+                                      const size_t nBytesPerSample,
+                                      const size_t nChannels,
+                                      const uint32_t samplesPerSec,
+                                      void* audioSamples,
+                                      size_t& nSamplesOut,
+                                      int64_t* elapsed_time_ms,
+                                      int64_t* ntp_time_ms) {
+  GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true,
+                 audioSamples, elapsed_time_ms, ntp_time_ms);
+  nSamplesOut = audioFrame_.samples_per_channel_;
   return 0;
 }
 
 int VoEBaseImpl::OnDataAvailable(const int voe_channels[],
-                                 int number_of_voe_channels,
-                                 const int16_t* audio_data,
-                                 int sample_rate,
-                                 int number_of_channels,
-                                 int number_of_frames,
-                                 int audio_delay_milliseconds,
-                                 int volume,
-                                 bool key_pressed,
-                                 bool need_audio_processing) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "VoEBaseImpl::OnDataAvailable(number_of_voe_channels=%d, "
-               "sample_rate=%d, number_of_channels=%d, number_of_frames=%d, "
-               "audio_delay_milliseconds=%d, volume=%d, "
-               "key_pressed=%d, need_audio_processing=%d)",
-               number_of_voe_channels, sample_rate, number_of_channels,
-               number_of_frames, audio_delay_milliseconds, volume,
-               key_pressed, need_audio_processing);
-  if (number_of_voe_channels == 0)
-    return 0;
+                                 size_t number_of_voe_channels,
+                                 const int16_t* audio_data, int sample_rate,
+                                 size_t number_of_channels,
+                                 size_t number_of_frames,
+                                 int audio_delay_milliseconds, int volume,
+                                 bool key_pressed, bool need_audio_processing) {
+  if (number_of_voe_channels == 0) return 0;
 
   if (need_audio_processing) {
     return ProcessRecordedDataWithAPM(
         voe_channels, number_of_voe_channels, audio_data, sample_rate,
-        number_of_channels, number_of_frames, audio_delay_milliseconds,
-        0, volume, key_pressed);
+        number_of_channels, number_of_frames, audio_delay_milliseconds, 0,
+        volume, key_pressed);
   }
 
   // No need to go through the APM, demultiplex the data to each VoE channel,
   // encode and send to the network.
-  for (int i = 0; i < number_of_voe_channels; ++i) {
+  for (size_t i = 0; i < number_of_voe_channels; ++i) {
     // TODO(ajm): In the case where multiple channels are using the same codec
     // rate, this path needlessly does extra conversions. We should convert once
     // and share between channels.
@@ -211,20 +141,18 @@ int VoEBaseImpl::OnDataAvailable(const int voe_channels[],
 
 void VoEBaseImpl::OnData(int voe_channel, const void* audio_data,
                          int bits_per_sample, int sample_rate,
-                         int number_of_channels,
-                         int number_of_frames) {
+                         size_t number_of_channels, size_t number_of_frames) {
   PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate,
                   number_of_channels, number_of_frames);
 }
 
 void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
                                   int bits_per_sample, int sample_rate,
-                                  int number_of_channels,
-                                  int number_of_frames) {
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(voe_channel);
+                                  size_t number_of_channels,
+                                  size_t number_of_frames) {
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(voe_channel);
   voe::Channel* channel_ptr = ch.channel();
-  if (!channel_ptr)
-    return;
+  if (!channel_ptr) return;
 
   if (channel_ptr->Sending()) {
     channel_ptr->Demultiplex(static_cast<const int16_t*>(audio_data),
@@ -234,657 +162,465 @@ void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
   }
 }
 
-void VoEBaseImpl::PullRenderData(int bits_per_sample, int sample_rate,
-                                 int number_of_channels, int number_of_frames,
-                                 void* audio_data,
-                                 int64_t* elapsed_time_ms,
+void VoEBaseImpl::PullRenderData(int bits_per_sample,
+                                 int sample_rate,
+                                 size_t number_of_channels,
+                                 size_t number_of_frames,
+                                 void* audio_data, int64_t* elapsed_time_ms,
                                  int64_t* ntp_time_ms) {
   assert(bits_per_sample == 16);
-  assert(number_of_frames == static_cast<int>(sample_rate / 100));
+  assert(number_of_frames == static_cast<size_t>(sample_rate / 100));
 
   GetPlayoutData(sample_rate, number_of_channels, number_of_frames, false,
                  audio_data, elapsed_time_ms, ntp_time_ms);
 }
 
-int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "RegisterVoiceEngineObserver(observer=0x%d)", &observer);
-    CriticalSectionScoped cs(&_callbackCritSect);
-    if (_voiceEngineObserverPtr)
-    {
-        _shared->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "RegisterVoiceEngineObserver() observer already enabled");
-        return -1;
-    }
+int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
+  CriticalSectionScoped cs(&callbackCritSect_);
+  if (voiceEngineObserverPtr_) {
+    shared_->SetLastError(
+        VE_INVALID_OPERATION, kTraceError,
+        "RegisterVoiceEngineObserver() observer already enabled");
+    return -1;
+  }
 
-    // Register the observer in all active channels
-    for (voe::ChannelManager::Iterator it(&_shared->channel_manager());
-         it.IsValid();
-         it.Increment()) {
-      it.GetChannel()->RegisterVoiceEngineObserver(observer);
-    }
+  // Register the observer in all active channels
+  for (voe::ChannelManager::Iterator it(&shared_->channel_manager());
+       it.IsValid(); it.Increment()) {
+    it.GetChannel()->RegisterVoiceEngineObserver(observer);
+  }
 
-    _shared->transmit_mixer()->RegisterVoiceEngineObserver(observer);
-
-    _voiceEngineObserverPtr = &observer;
-    _voiceEngineObserver = true;
-
-    return 0;
+  shared_->transmit_mixer()->RegisterVoiceEngineObserver(observer);
+  voiceEngineObserverPtr_ = &observer;
+  return 0;
 }
 
-int VoEBaseImpl::DeRegisterVoiceEngineObserver()
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "DeRegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
-    if (!_voiceEngineObserverPtr)
-    {
-        _shared->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "DeRegisterVoiceEngineObserver() observer already disabled");
-        return 0;
-    }
-
-    _voiceEngineObserver = false;
-    _voiceEngineObserverPtr = NULL;
-
-    // Deregister the observer in all active channels
-    for (voe::ChannelManager::Iterator it(&_shared->channel_manager());
-         it.IsValid();
-         it.Increment()) {
-      it.GetChannel()->DeRegisterVoiceEngineObserver();
-    }
-
+int VoEBaseImpl::DeRegisterVoiceEngineObserver() {
+  CriticalSectionScoped cs(&callbackCritSect_);
+  if (!voiceEngineObserverPtr_) {
+    shared_->SetLastError(
+        VE_INVALID_OPERATION, kTraceError,
+        "DeRegisterVoiceEngineObserver() observer already disabled");
     return 0;
+  }
+  voiceEngineObserverPtr_ = nullptr;
+
+  // Deregister the observer in all active channels
+  for (voe::ChannelManager::Iterator it(&shared_->channel_manager());
+       it.IsValid(); it.Increment()) {
+    it.GetChannel()->DeRegisterVoiceEngineObserver();
+  }
+
+  return 0;
 }
 
 int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
-                      AudioProcessing* audioproc)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-        "Init(external_adm=0x%p)", external_adm);
-    CriticalSectionScoped cs(_shared->crit_sec());
+                      AudioProcessing* audioproc) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  WebRtcSpl_Init();
+  if (shared_->statistics().Initialized()) {
+    return 0;
+  }
+  if (shared_->process_thread()) {
+    shared_->process_thread()->Start();
+  }
 
-    WebRtcSpl_Init();
+  // Create an internal ADM if the user has not added an external
+  // ADM implementation as input to Init().
+  if (external_adm == nullptr) {
+#if !defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+    return -1;
+#else
+    // Create the internal ADM implementation.
+    shared_->set_audio_device(AudioDeviceModuleImpl::Create(
+        VoEId(shared_->instance_id(), -1), shared_->audio_device_layer()));
 
-    if (_shared->statistics().Initialized())
-    {
-        return 0;
+    if (shared_->audio_device() == nullptr) {
+      shared_->SetLastError(VE_NO_MEMORY, kTraceCritical,
+                            "Init() failed to create the ADM");
+      return -1;
     }
+#endif  // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
+  } else {
+    // Use the already existing external ADM implementation.
+    shared_->set_audio_device(external_adm);
+    LOG_F(LS_INFO)
+        << "An external ADM implementation will be used in VoiceEngine";
+  }
 
-    if (_shared->process_thread())
-    {
-        _shared->process_thread()->Start();
-    }
+  // Register the ADM to the process thread, which will drive the error
+  // callback mechanism
+  if (shared_->process_thread()) {
+    shared_->process_thread()->RegisterModule(shared_->audio_device());
+  }
 
-    // Create an internal ADM if the user has not added an external
-    // ADM implementation as input to Init().
-    if (external_adm == NULL)
-    {
-        // Create the internal ADM implementation.
-        _shared->set_audio_device(AudioDeviceModuleImpl::Create(
-            VoEId(_shared->instance_id(), -1), _shared->audio_device_layer()));
+  bool available = false;
 
-        if (_shared->audio_device() == NULL)
-        {
-            _shared->SetLastError(VE_NO_MEMORY, kTraceCritical,
-                "Init() failed to create the ADM");
-            return -1;
-        }
-    }
-    else
-    {
-        // Use the already existing external ADM implementation.
-        _shared->set_audio_device(external_adm);
-        WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-            "An external ADM implementation will be used in VoiceEngine");
-    }
+  // --------------------
+  // Reinitialize the ADM
 
-    // Register the ADM to the process thread, which will drive the error
-    // callback mechanism
-    if (_shared->process_thread())
-    {
-        _shared->process_thread()->RegisterModule(_shared->audio_device());
-    }
+  // Register the AudioObserver implementation
+  if (shared_->audio_device()->RegisterEventObserver(this) != 0) {
+    shared_->SetLastError(
+        VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+        "Init() failed to register event observer for the ADM");
+  }
 
-    bool available(false);
+  // Register the AudioTransport implementation
+  if (shared_->audio_device()->RegisterAudioCallback(this) != 0) {
+    shared_->SetLastError(
+        VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+        "Init() failed to register audio callback for the ADM");
+  }
 
-    // --------------------
-    // Reinitialize the ADM
+  // ADM initialization
+  if (shared_->audio_device()->Init() != 0) {
+    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                          "Init() failed to initialize the ADM");
+    return -1;
+  }
 
-    // Register the AudioObserver implementation
-    if (_shared->audio_device()->RegisterEventObserver(this) != 0) {
-      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-          "Init() failed to register event observer for the ADM");
-    }
+  // Initialize the default speaker
+  if (shared_->audio_device()->SetPlayoutDevice(
+          WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
+    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
+                          "Init() failed to set the default output device");
+  }
+  if (shared_->audio_device()->InitSpeaker() != 0) {
+    shared_->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
+                          "Init() failed to initialize the speaker");
+  }
 
-    // Register the AudioTransport implementation
-    if (_shared->audio_device()->RegisterAudioCallback(this) != 0) {
-      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-          "Init() failed to register audio callback for the ADM");
-    }
+  // Initialize the default microphone
+  if (shared_->audio_device()->SetRecordingDevice(
+          WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) {
+    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
+                          "Init() failed to set the default input device");
+  }
+  if (shared_->audio_device()->InitMicrophone() != 0) {
+    shared_->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
+                          "Init() failed to initialize the microphone");
+  }
 
-    // ADM initialization
-    if (_shared->audio_device()->Init() != 0)
-    {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-            "Init() failed to initialize the ADM");
-        return -1;
-    }
+  // Set number of channels
+  if (shared_->audio_device()->StereoPlayoutIsAvailable(&available) != 0) {
+    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+                          "Init() failed to query stereo playout mode");
+  }
+  if (shared_->audio_device()->SetStereoPlayout(available) != 0) {
+    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+                          "Init() failed to set mono/stereo playout mode");
+  }
 
-    // Initialize the default speaker
-    if (_shared->audio_device()->SetPlayoutDevice(
-            WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0)
-    {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
-            "Init() failed to set the default output device");
-    }
-    if (_shared->audio_device()->InitSpeaker() != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
-            "Init() failed to initialize the speaker");
-    }
-
-    // Initialize the default microphone
-    if (_shared->audio_device()->SetRecordingDevice(
-            WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0)
-    {
-        _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
-            "Init() failed to set the default input device");
-    }
-    if (_shared->audio_device()->InitMicrophone() != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
-            "Init() failed to initialize the microphone");
-    }
-
-    // Set number of channels
-    if (_shared->audio_device()->StereoPlayoutIsAvailable(&available) != 0) {
-      _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-          "Init() failed to query stereo playout mode");
-    }
-    if (_shared->audio_device()->SetStereoPlayout(available) != 0)
-    {
-        _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-            "Init() failed to set mono/stereo playout mode");
-    }
-
-    // TODO(andrew): These functions don't tell us whether stereo recording
-    // is truly available. We simply set the AudioProcessing input to stereo
-    // here, because we have to wait until receiving the first frame to
-    // determine the actual number of channels anyway.
-    //
-    // These functions may be changed; tracked here:
-    // http://code.google.com/p/webrtc/issues/detail?id=204
-    _shared->audio_device()->StereoRecordingIsAvailable(&available);
-    if (_shared->audio_device()->SetStereoRecording(available) != 0)
-    {
-        _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-            "Init() failed to set mono/stereo recording mode");
-    }
+  // TODO(andrew): These functions don't tell us whether stereo recording
+  // is truly available. We simply set the AudioProcessing input to stereo
+  // here, because we have to wait until receiving the first frame to
+  // determine the actual number of channels anyway.
+  //
+  // These functions may be changed; tracked here:
+  // http://code.google.com/p/webrtc/issues/detail?id=204
+  shared_->audio_device()->StereoRecordingIsAvailable(&available);
+  if (shared_->audio_device()->SetStereoRecording(available) != 0) {
+    shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+                          "Init() failed to set mono/stereo recording mode");
+  }
 
+  if (!audioproc) {
+    audioproc = AudioProcessing::Create(shared_->channel_manager().config_);
     if (!audioproc) {
-      audioproc = AudioProcessing::Create(_shared->channel_manager().config_);
-      if (!audioproc) {
-        LOG(LS_ERROR) << "Failed to create AudioProcessing.";
-        _shared->SetLastError(VE_NO_MEMORY);
-        return -1;
-      }
+      LOG(LS_ERROR) << "Failed to create AudioProcessing.";
+      shared_->SetLastError(VE_NO_MEMORY);
+      return -1;
     }
-    _shared->set_audio_processing(audioproc);
+  }
+  shared_->set_audio_processing(audioproc);
 
-    // Set the error state for any failures in this block.
-    _shared->SetLastError(VE_APM_ERROR);
-    // Configure AudioProcessing components.
-    if (audioproc->high_pass_filter()->Enable(true) != 0) {
-      LOG_FERR1(LS_ERROR, high_pass_filter()->Enable, true);
-      return -1;
-    }
-    if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) {
-      LOG_FERR1(LS_ERROR, enable_drift_compensation, false);
-      return -1;
-    }
-    if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) {
-      LOG_FERR1(LS_ERROR, noise_suppression()->set_level, kDefaultNsMode);
-      return -1;
-    }
-    GainControl* agc = audioproc->gain_control();
-    if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
-      LOG_FERR2(LS_ERROR, agc->set_analog_level_limits, kMinVolumeLevel,
-                kMaxVolumeLevel);
-      return -1;
-    }
-    if (agc->set_mode(kDefaultAgcMode) != 0) {
-      LOG_FERR1(LS_ERROR, agc->set_mode, kDefaultAgcMode);
-      return -1;
-    }
-    if (agc->Enable(kDefaultAgcState) != 0) {
-      LOG_FERR1(LS_ERROR, agc->Enable, kDefaultAgcState);
-      return -1;
-    }
-    _shared->SetLastError(0);  // Clear error state.
+  // Set the error state for any failures in this block.
+  shared_->SetLastError(VE_APM_ERROR);
+  // Configure AudioProcessing components.
+  if (audioproc->high_pass_filter()->Enable(true) != 0) {
+    LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
+    return -1;
+  }
+  if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) {
+    LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
+    return -1;
+  }
+  if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) {
+    LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
+        << kDefaultNsMode;
+    return -1;
+  }
+  GainControl* agc = audioproc->gain_control();
+  if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
+    LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: "
+        << kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
+    return -1;
+  }
+  if (agc->set_mode(kDefaultAgcMode) != 0) {
+    LOG_F(LS_ERROR) << "Failed to set mode: " << kDefaultAgcMode;
+    return -1;
+  }
+  if (agc->Enable(kDefaultAgcState) != 0) {
+    LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState;
+    return -1;
+  }
+  shared_->SetLastError(0);  // Clear error state.
 
 #ifdef WEBRTC_VOICE_ENGINE_AGC
-    bool agc_enabled = agc->mode() == GainControl::kAdaptiveAnalog &&
-                       agc->is_enabled();
-    if (_shared->audio_device()->SetAGC(agc_enabled) != 0) {
-      LOG_FERR1(LS_ERROR, audio_device()->SetAGC, agc_enabled);
-      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR);
-      // TODO(ajm): No error return here due to
-      // https://code.google.com/p/webrtc/issues/detail?id=1464
-    }
+  bool agc_enabled =
+      agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled();
+  if (shared_->audio_device()->SetAGC(agc_enabled) != 0) {
+    LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled;
+    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR);
+    // TODO(ajm): No error return here due to
+    // https://code.google.com/p/webrtc/issues/detail?id=1464
+  }
 #endif
 
-    return _shared->statistics().SetInitialized();
+  return shared_->statistics().SetInitialized();
 }
 
-int VoEBaseImpl::Terminate()
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "Terminate()");
-    CriticalSectionScoped cs(_shared->crit_sec());
-    return TerminateInternal();
+int VoEBaseImpl::Terminate() {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  return TerminateInternal();
 }
 
 int VoEBaseImpl::CreateChannel() {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "CreateChannel()");
-  CriticalSectionScoped cs(_shared->crit_sec());
-  if (!_shared->statistics().Initialized()) {
-      _shared->SetLastError(VE_NOT_INITED, kTraceError);
-      return -1;
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
   }
 
-  voe::ChannelOwner channel_owner = _shared->channel_manager().CreateChannel();
-
+  voe::ChannelOwner channel_owner = shared_->channel_manager().CreateChannel();
   return InitializeChannel(&channel_owner);
 }
 
 int VoEBaseImpl::CreateChannel(const Config& config) {
-  CriticalSectionScoped cs(_shared->crit_sec());
-  if (!_shared->statistics().Initialized()) {
-      _shared->SetLastError(VE_NOT_INITED, kTraceError);
-      return -1;
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
   }
-  voe::ChannelOwner channel_owner = _shared->channel_manager().CreateChannel(
-      config);
+  voe::ChannelOwner channel_owner =
+      shared_->channel_manager().CreateChannel(config);
   return InitializeChannel(&channel_owner);
 }
 
-int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner)
-{
-    if (channel_owner->channel()->SetEngineInformation(
-            _shared->statistics(),
-            *_shared->output_mixer(),
-            *_shared->transmit_mixer(),
-            *_shared->process_thread(),
-            *_shared->audio_device(),
-            _voiceEngineObserverPtr,
-            &_callbackCritSect) != 0) {
-      _shared->SetLastError(
-          VE_CHANNEL_NOT_CREATED,
-          kTraceError,
-          "CreateChannel() failed to associate engine and channel."
-          " Destroying channel.");
-      _shared->channel_manager()
-          .DestroyChannel(channel_owner->channel()->ChannelId());
-      return -1;
-    } else if (channel_owner->channel()->Init() != 0) {
-      _shared->SetLastError(
-          VE_CHANNEL_NOT_CREATED,
-          kTraceError,
-          "CreateChannel() failed to initialize channel. Destroying"
-          " channel.");
-      _shared->channel_manager()
-          .DestroyChannel(channel_owner->channel()->ChannelId());
+int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
+  if (channel_owner->channel()->SetEngineInformation(
+          shared_->statistics(), *shared_->output_mixer(),
+          *shared_->transmit_mixer(), *shared_->process_thread(),
+          *shared_->audio_device(), voiceEngineObserverPtr_,
+          &callbackCritSect_) != 0) {
+    shared_->SetLastError(
+        VE_CHANNEL_NOT_CREATED, kTraceError,
+        "CreateChannel() failed to associate engine and channel."
+        " Destroying channel.");
+    shared_->channel_manager().DestroyChannel(
+        channel_owner->channel()->ChannelId());
+    return -1;
+  } else if (channel_owner->channel()->Init() != 0) {
+    shared_->SetLastError(
+        VE_CHANNEL_NOT_CREATED, kTraceError,
+        "CreateChannel() failed to initialize channel. Destroying"
+        " channel.");
+    shared_->channel_manager().DestroyChannel(
+        channel_owner->channel()->ChannelId());
+    return -1;
+  }
+  return channel_owner->channel()->ChannelId();
+}
+
+int VoEBaseImpl::DeleteChannel(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+
+  {
+    voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == nullptr) {
+      shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                            "DeleteChannel() failed to locate channel");
       return -1;
     }
+  }
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "CreateChannel() => %d", channel_owner->channel()->ChannelId());
-    return channel_owner->channel()->ChannelId();
+  shared_->channel_manager().DestroyChannel(channel);
+  if (StopSend() != 0) {
+    return -1;
+  }
+  if (StopPlayout() != 0) {
+    return -1;
+  }
+  return 0;
 }
 
-int VoEBaseImpl::DeleteChannel(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "DeleteChannel(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
+int VoEBaseImpl::StartReceive(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == nullptr) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StartReceive() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->StartReceiving();
+}
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-
-    {
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "DeleteChannel() failed to locate channel");
-            return -1;
-        }
-    }
-
-    _shared->channel_manager().DestroyChannel(channel);
-
-    if (StopSend() != 0)
-    {
-        return -1;
-    }
-
-    if (StopPlayout() != 0)
-    {
-        return -1;
-    }
+int VoEBaseImpl::StopReceive(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == nullptr) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetLocalReceiver() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->StopReceiving();
+}
 
+int VoEBaseImpl::StartPlayout(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == nullptr) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StartPlayout() failed to locate channel");
+    return -1;
+  }
+  if (channelPtr->Playing()) {
     return 0;
+  }
+  if (StartPlayout() != 0) {
+    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                          "StartPlayout() failed to start playout");
+    return -1;
+  }
+  return channelPtr->StartPlayout();
 }
 
-int VoEBaseImpl::StartReceive(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartReceive(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StartReceive() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->StartReceiving();
+int VoEBaseImpl::StopPlayout(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == nullptr) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StopPlayout() failed to locate channel");
+    return -1;
+  }
+  if (channelPtr->StopPlayout() != 0) {
+    LOG_F(LS_WARNING) << "StopPlayout() failed to stop playout for channel "
+                      << channel;
+  }
+  return StopPlayout();
 }
 
-int VoEBaseImpl::StopReceive(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopListen(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetLocalReceiver() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->StopReceiving();
+int VoEBaseImpl::StartSend(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == nullptr) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StartSend() failed to locate channel");
+    return -1;
+  }
+  if (channelPtr->Sending()) {
+    return 0;
+  }
+  if (StartSend() != 0) {
+    shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                          "StartSend() failed to start recording");
+    return -1;
+  }
+  return channelPtr->StartSend();
 }
 
-int VoEBaseImpl::StartPlayout(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartPlayout(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StartPlayout() failed to locate channel");
-        return -1;
-    }
-    if (channelPtr->Playing())
-    {
-        return 0;
-    }
-    if (StartPlayout() != 0)
-    {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-            "StartPlayout() failed to start playout");
-        return -1;
-    }
-    return channelPtr->StartPlayout();
+int VoEBaseImpl::StopSend(int channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == nullptr) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StopSend() failed to locate channel");
+    return -1;
+  }
+  if (channelPtr->StopSend() != 0) {
+    LOG_F(LS_WARNING) << "StopSend() failed to stop sending for channel "
+                      << channel;
+  }
+  return StopSend();
 }
 
-int VoEBaseImpl::StopPlayout(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopPlayout(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StopPlayout() failed to locate channel");
-        return -1;
-    }
-    if (channelPtr->StopPlayout() != 0)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-            VoEId(_shared->instance_id(), -1),
-            "StopPlayout() failed to stop playout for channel %d", channel);
-    }
-    return StopPlayout();
-}
+int VoEBaseImpl::GetVersion(char version[1024]) {
+  if (version == nullptr) {
+    shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError);
+    return -1;
+  }
 
-int VoEBaseImpl::StartSend(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartSend(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StartSend() failed to locate channel");
-        return -1;
-    }
-    if (channelPtr->Sending())
-    {
-        return 0;
-    }
-    if (StartSend() != 0)
-    {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-            "StartSend() failed to start recording");
-        return -1;
-    }
-    return channelPtr->StartSend();
-}
-
-int VoEBaseImpl::StopSend(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopSend(channel=%d)", channel);
-    CriticalSectionScoped cs(_shared->crit_sec());
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StopSend() failed to locate channel");
-        return -1;
-    }
-    if (channelPtr->StopSend() != 0)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-            VoEId(_shared->instance_id(), -1),
-            "StopSend() failed to stop sending for channel %d", channel);
-    }
-    return StopSend();
-}
-
-int VoEBaseImpl::GetVersion(char version[1024])
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetVersion(version=?)");
-    assert(kVoiceEngineVersionMaxMessageSize == 1024);
-
-    if (version == NULL)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError);
-        return (-1);
-    }
-
-    char versionBuf[kVoiceEngineVersionMaxMessageSize];
-    char* versionPtr = versionBuf;
-
-    int32_t len = 0;
-    int32_t accLen = 0;
-
-    len = AddVoEVersion(versionPtr);
-    if (len == -1)
-    {
-        return -1;
-    }
-    versionPtr += len;
-    accLen += len;
-    assert(accLen < kVoiceEngineVersionMaxMessageSize);
-
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
-    len = AddExternalTransportBuild(versionPtr);
-    if (len == -1)
-    {
-         return -1;
-    }
-    versionPtr += len;
-    accLen += len;
-    assert(accLen < kVoiceEngineVersionMaxMessageSize);
-#endif
+  std::string versionString = VoiceEngine::GetVersionString();
 #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-    len = AddExternalRecAndPlayoutBuild(versionPtr);
-    if (len == -1)
+  versionString += "External recording and playout build";
+#endif
+  RTC_DCHECK_GT(1024u, versionString.size() + 1);
+  char* end = std::copy(versionString.cbegin(), versionString.cend(), version);
+  end[0] = '\n';
+  end[1] = '\0';
+  return 0;
+}
+
+int VoEBaseImpl::LastError() { return (shared_->statistics().LastError()); }
+
+int32_t VoEBaseImpl::StartPlayout() {
+  if (!shared_->audio_device()->Playing()) {
+    if (!shared_->ext_playout())
     {
+      if (shared_->audio_device()->InitPlayout() != 0) {
+        LOG_F(LS_ERROR) << "Failed to initialize playout";
         return -1;
+      }
+      if (shared_->audio_device()->StartPlayout() != 0) {
+        LOG_F(LS_ERROR) << "Failed to start playout";
+        return -1;
+      }
     }
-    versionPtr += len;
-    accLen += len;
-    assert(accLen < kVoiceEngineVersionMaxMessageSize);
- #endif
-
-    memcpy(version, versionBuf, accLen);
-    version[accLen] = '\0';
-
-    // to avoid the truncation in the trace, split the string into parts
-    char partOfVersion[256];
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1), "GetVersion() =>");
-    for (int partStart = 0; partStart < accLen;)
-    {
-        memset(partOfVersion, 0, sizeof(partOfVersion));
-        int partEnd = partStart + 180;
-        while (version[partEnd] != '\n' && version[partEnd] != '\0')
-        {
-            partEnd--;
-        }
-        if (partEnd < accLen)
-        {
-            memcpy(partOfVersion, &version[partStart], partEnd - partStart);
-        }
-        else
-        {
-            memcpy(partOfVersion, &version[partStart], accLen - partStart);
-        }
-        partStart = partEnd;
-        WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-            VoEId(_shared->instance_id(), -1), "%s", partOfVersion);
-    }
-
-    return 0;
-}
-
-int32_t VoEBaseImpl::AddVoEVersion(char* str) const
-{
-    return sprintf(str, "VoiceEngine 4.1.0\n");
-}
-
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
-int32_t VoEBaseImpl::AddExternalTransportBuild(char* str) const
-{
-    return sprintf(str, "External transport build\n");
-}
-#endif
-
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-int32_t VoEBaseImpl::AddExternalRecAndPlayoutBuild(char* str) const
-{
-    return sprintf(str, "External recording and playout build\n");
-}
-#endif
-
-int VoEBaseImpl::LastError()
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "LastError()");
-    return (_shared->statistics().LastError());
-}
-
-int32_t VoEBaseImpl::StartPlayout()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl::StartPlayout()");
-    if (_shared->audio_device()->Playing())
-    {
-        return 0;
-    }
-    if (!_shared->ext_playout())
-    {
-        if (_shared->audio_device()->InitPlayout() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartPlayout() failed to initialize playout");
-            return -1;
-        }
-        if (_shared->audio_device()->StartPlayout() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartPlayout() failed to start playout");
-            return -1;
-        }
-    }
-    return 0;
+  }
+  return 0;
 }
 
 int32_t VoEBaseImpl::StopPlayout() {
-  WEBRTC_TRACE(kTraceInfo,
-               kTraceVoice,
-               VoEId(_shared->instance_id(), -1),
-               "VoEBaseImpl::StopPlayout()");
   // Stop audio-device playing if no channel is playing out
-  if (_shared->NumOfPlayingChannels() == 0) {
-    if (_shared->audio_device()->StopPlayout() != 0) {
-      _shared->SetLastError(VE_CANNOT_STOP_PLAYOUT,
-                            kTraceError,
+  if (shared_->NumOfPlayingChannels() == 0) {
+    if (shared_->audio_device()->StopPlayout() != 0) {
+      shared_->SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError,
                             "StopPlayout() failed to stop playout");
       return -1;
     }
@@ -892,124 +628,91 @@ int32_t VoEBaseImpl::StopPlayout() {
   return 0;
 }
 
-int32_t VoEBaseImpl::StartSend()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl::StartSend()");
-    if (_shared->audio_device()->Recording())
+int32_t VoEBaseImpl::StartSend() {
+  if (!shared_->audio_device()->Recording()) {
+    if (!shared_->ext_recording())
     {
-        return 0;
+      if (shared_->audio_device()->InitRecording() != 0) {
+        LOG_F(LS_ERROR) << "Failed to initialize recording";
+        return -1;
+      }
+      if (shared_->audio_device()->StartRecording() != 0) {
+        LOG_F(LS_ERROR) << "Failed to start recording";
+        return -1;
+      }
     }
-    if (!_shared->ext_recording())
-    {
-        if (_shared->audio_device()->InitRecording() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartSend() failed to initialize recording");
-            return -1;
-        }
-        if (_shared->audio_device()->StartRecording() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartSend() failed to start recording");
-            return -1;
-        }
-    }
-
-    return 0;
+  }
+  return 0;
 }
 
-int32_t VoEBaseImpl::StopSend()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl::StopSend()");
-
-    if (_shared->NumOfSendingChannels() == 0 &&
-        !_shared->transmit_mixer()->IsRecordingMic())
-    {
-        // Stop audio-device recording if no channel is recording
-        if (_shared->audio_device()->StopRecording() != 0)
-        {
-            _shared->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError,
-                "StopSend() failed to stop recording");
-            return -1;
-        }
-        _shared->transmit_mixer()->StopSend();
+int32_t VoEBaseImpl::StopSend() {
+  if (shared_->NumOfSendingChannels() == 0 &&
+      !shared_->transmit_mixer()->IsRecordingMic()) {
+    // Stop audio-device recording if no channel is recording
+    if (shared_->audio_device()->StopRecording() != 0) {
+      shared_->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError,
+                            "StopSend() failed to stop recording");
+      return -1;
     }
+    shared_->transmit_mixer()->StopSend();
+  }
 
-    return 0;
+  return 0;
 }
 
-int32_t VoEBaseImpl::TerminateInternal()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl::TerminateInternal()");
+int32_t VoEBaseImpl::TerminateInternal() {
+  // Delete any remaining channel objects
+  shared_->channel_manager().DestroyAllChannels();
 
-    // Delete any remaining channel objects
-    _shared->channel_manager().DestroyAllChannels();
-
-    if (_shared->process_thread())
-    {
-        if (_shared->audio_device())
-        {
-            _shared->process_thread()->DeRegisterModule(
-                _shared->audio_device());
-        }
-        _shared->process_thread()->Stop();
+  if (shared_->process_thread()) {
+    if (shared_->audio_device()) {
+      shared_->process_thread()->DeRegisterModule(shared_->audio_device());
     }
+    shared_->process_thread()->Stop();
+  }
 
-    if (_shared->audio_device())
-    {
-        if (_shared->audio_device()->StopPlayout() != 0)
-        {
-            _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-                "TerminateInternal() failed to stop playout");
-        }
-        if (_shared->audio_device()->StopRecording() != 0)
-        {
-            _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-                "TerminateInternal() failed to stop recording");
-        }
-        if (_shared->audio_device()->RegisterEventObserver(NULL) != 0) {
-          _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-              "TerminateInternal() failed to de-register event observer "
-              "for the ADM");
-        }
-        if (_shared->audio_device()->RegisterAudioCallback(NULL) != 0) {
-          _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-              "TerminateInternal() failed to de-register audio callback "
-              "for the ADM");
-        }
-        if (_shared->audio_device()->Terminate() != 0)
-        {
-            _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-                "TerminateInternal() failed to terminate the ADM");
-        }
-        _shared->set_audio_device(NULL);
+  if (shared_->audio_device()) {
+    if (shared_->audio_device()->StopPlayout() != 0) {
+      shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+                            "TerminateInternal() failed to stop playout");
     }
-
-    if (_shared->audio_processing()) {
-        _shared->set_audio_processing(NULL);
+    if (shared_->audio_device()->StopRecording() != 0) {
+      shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
+                            "TerminateInternal() failed to stop recording");
     }
+    if (shared_->audio_device()->RegisterEventObserver(nullptr) != 0) {
+      shared_->SetLastError(
+          VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+          "TerminateInternal() failed to de-register event observer "
+          "for the ADM");
+    }
+    if (shared_->audio_device()->RegisterAudioCallback(nullptr) != 0) {
+      shared_->SetLastError(
+          VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+          "TerminateInternal() failed to de-register audio callback "
+          "for the ADM");
+    }
+    if (shared_->audio_device()->Terminate() != 0) {
+      shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                            "TerminateInternal() failed to terminate the ADM");
+    }
+    shared_->set_audio_device(nullptr);
+  }
 
-    return _shared->statistics().SetUnInitialized();
+  if (shared_->audio_processing()) {
+    shared_->set_audio_processing(nullptr);
+  }
+
+  return shared_->statistics().SetUnInitialized();
 }
 
 int VoEBaseImpl::ProcessRecordedDataWithAPM(
-    const int voe_channels[],
-    int number_of_voe_channels,
-    const void* audio_data,
-    uint32_t sample_rate,
-    uint8_t number_of_channels,
-    uint32_t number_of_frames,
-    uint32_t audio_delay_milliseconds,
-    int32_t clock_drift,
-    uint32_t volume,
-    bool key_pressed) {
-  assert(_shared->transmit_mixer() != NULL);
-  assert(_shared->audio_device() != NULL);
+    const int voe_channels[], size_t number_of_voe_channels,
+    const void* audio_data, uint32_t sample_rate, size_t number_of_channels,
+    size_t number_of_frames, uint32_t audio_delay_milliseconds,
+    int32_t clock_drift, uint32_t volume, bool key_pressed) {
+  assert(shared_->transmit_mixer() != nullptr);
+  assert(shared_->audio_device() != nullptr);
 
   uint32_t max_volume = 0;
   uint16_t voe_mic_level = 0;
@@ -1017,11 +720,11 @@ int VoEBaseImpl::ProcessRecordedDataWithAPM(
   // indicate no volume is available.
   if (volume != 0) {
     // Scale from ADM to VoE level range
-    if (_shared->audio_device()->MaxMicrophoneVolume(&max_volume) == 0) {
+    if (shared_->audio_device()->MaxMicrophoneVolume(&max_volume) == 0) {
       if (max_volume) {
         voe_mic_level = static_cast<uint16_t>(
-            (volume * kMaxVolumeLevel +
-                static_cast<int>(max_volume / 2)) / max_volume);
+            (volume * kMaxVolumeLevel + static_cast<int>(max_volume / 2)) /
+            max_volume);
       }
     }
     // We learned that on certain systems (e.g Linux) the voe_mic_level
@@ -1037,7 +740,7 @@ int VoEBaseImpl::ProcessRecordedDataWithAPM(
 
   // Perform channel-independent operations
   // (APM, mix with file, record to file, mute, etc.)
-  _shared->transmit_mixer()->PrepareDemux(
+  shared_->transmit_mixer()->PrepareDemux(
       audio_data, number_of_frames, number_of_channels, sample_rate,
       static_cast<uint16_t>(audio_delay_milliseconds), clock_drift,
       voe_mic_level, key_pressed);
@@ -1048,57 +751,84 @@ int VoEBaseImpl::ProcessRecordedDataWithAPM(
   // do the operations on all the existing VoE channels; otherwise the
   // operations will be done on specific channels.
   if (number_of_voe_channels == 0) {
-    _shared->transmit_mixer()->DemuxAndMix();
-    _shared->transmit_mixer()->EncodeAndSend();
+    shared_->transmit_mixer()->DemuxAndMix();
+    shared_->transmit_mixer()->EncodeAndSend();
   } else {
-    _shared->transmit_mixer()->DemuxAndMix(voe_channels,
+    shared_->transmit_mixer()->DemuxAndMix(voe_channels,
                                            number_of_voe_channels);
-    _shared->transmit_mixer()->EncodeAndSend(voe_channels,
+    shared_->transmit_mixer()->EncodeAndSend(voe_channels,
                                              number_of_voe_channels);
   }
 
   // Scale from VoE to ADM level range.
-  uint32_t new_voe_mic_level = _shared->transmit_mixer()->CaptureLevel();
-
+  uint32_t new_voe_mic_level = shared_->transmit_mixer()->CaptureLevel();
   if (new_voe_mic_level != voe_mic_level) {
     // Return the new volume if AGC has changed the volume.
-    return static_cast<int>(
-        (new_voe_mic_level * max_volume +
-            static_cast<int>(kMaxVolumeLevel / 2)) / kMaxVolumeLevel);
+    return static_cast<int>((new_voe_mic_level * max_volume +
+                             static_cast<int>(kMaxVolumeLevel / 2)) /
+                            kMaxVolumeLevel);
   }
 
   // Return 0 to indicate no change on the volume.
   return 0;
 }
 
-void VoEBaseImpl::GetPlayoutData(int sample_rate, int number_of_channels,
-                                 int number_of_frames, bool feed_data_to_apm,
-                                 void* audio_data,
-                                 int64_t* elapsed_time_ms,
+void VoEBaseImpl::GetPlayoutData(int sample_rate, size_t number_of_channels,
+                                 size_t number_of_frames, bool feed_data_to_apm,
+                                 void* audio_data, int64_t* elapsed_time_ms,
                                  int64_t* ntp_time_ms) {
-  assert(_shared->output_mixer() != NULL);
+  assert(shared_->output_mixer() != nullptr);
 
   // TODO(andrew): if the device is running in mono, we should tell the mixer
   // here so that it will only request mono from AudioCodingModule.
   // Perform mixing of all active participants (channel-based mixing)
-  _shared->output_mixer()->MixActiveChannels();
+  shared_->output_mixer()->MixActiveChannels();
 
   // Additional operations on the combined signal
-  _shared->output_mixer()->DoOperationsOnCombinedSignal(feed_data_to_apm);
+  shared_->output_mixer()->DoOperationsOnCombinedSignal(feed_data_to_apm);
 
   // Retrieve the final output mix (resampled to match the ADM)
-  _shared->output_mixer()->GetMixedAudio(sample_rate, number_of_channels,
-                                         &_audioFrame);
+  shared_->output_mixer()->GetMixedAudio(sample_rate, number_of_channels,
+                                         &audioFrame_);
 
-  assert(number_of_frames == _audioFrame.samples_per_channel_);
-  assert(sample_rate == _audioFrame.sample_rate_hz_);
+  assert(number_of_frames == audioFrame_.samples_per_channel_);
+  assert(sample_rate == audioFrame_.sample_rate_hz_);
 
   // Deliver audio (PCM) samples to the ADM
-  memcpy(audio_data, _audioFrame.data_,
+  memcpy(audio_data, audioFrame_.data_,
          sizeof(int16_t) * number_of_frames * number_of_channels);
 
-  *elapsed_time_ms = _audioFrame.elapsed_time_ms_;
-  *ntp_time_ms = _audioFrame.ntp_time_ms_;
+  *elapsed_time_ms = audioFrame_.elapsed_time_ms_;
+  *ntp_time_ms = audioFrame_.ntp_time_ms_;
+}
+
+int VoEBaseImpl::AssociateSendChannel(int channel,
+                                      int accociate_send_channel) {
+  CriticalSectionScoped cs(shared_->crit_sec());
+
+  if (!shared_->statistics().Initialized()) {
+      shared_->SetLastError(VE_NOT_INITED, kTraceError);
+      return -1;
+  }
+
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channel_ptr = ch.channel();
+  if (channel_ptr == NULL) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+        "AssociateSendChannel() failed to locate channel");
+    return -1;
+  }
+
+  ch = shared_->channel_manager().GetChannel(accociate_send_channel);
+  voe::Channel* accociate_send_channel_ptr = ch.channel();
+  if (accociate_send_channel_ptr == NULL) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+        "AssociateSendChannel() failed to locate accociate_send_channel");
+    return -1;
+  }
+
+  channel_ptr->set_associate_send_channel(ch);
+  return 0;
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.h
index 0eafa6f99a..58e0387423 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.h
@@ -13,156 +13,135 @@
 
 #include "webrtc/voice_engine/include/voe_base.h"
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/shared_data.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
 class ProcessThread;
 
-class VoEBaseImpl: public VoEBase,
-                   public AudioTransport,
-                   public AudioDeviceObserver
-{
-public:
-    virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
+class VoEBaseImpl : public VoEBase,
+                    public AudioTransport,
+                    public AudioDeviceObserver {
+ public:
+  int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) override;
+  int DeRegisterVoiceEngineObserver() override;
 
-    virtual int DeRegisterVoiceEngineObserver();
+  int Init(AudioDeviceModule* external_adm = nullptr,
+           AudioProcessing* audioproc = nullptr) override;
+  AudioProcessing* audio_processing() override {
+    return shared_->audio_processing();
+  }
+  int Terminate() override;
 
-    virtual int Init(AudioDeviceModule* external_adm = NULL,
-                     AudioProcessing* audioproc = NULL);
-    virtual AudioProcessing* audio_processing() {
-      return _shared->audio_processing();
-    }
+  int CreateChannel() override;
+  int CreateChannel(const Config& config) override;
+  int DeleteChannel(int channel) override;
 
-    virtual int Terminate();
+  int StartReceive(int channel) override;
+  int StartPlayout(int channel) override;
+  int StartSend(int channel) override;
+  int StopReceive(int channel) override;
+  int StopPlayout(int channel) override;
+  int StopSend(int channel) override;
 
-    virtual int CreateChannel();
-    virtual int CreateChannel(const Config& config);
+  int GetVersion(char version[1024]) override;
 
-    virtual int DeleteChannel(int channel);
+  int LastError() override;
 
-    virtual int StartReceive(int channel);
+  AudioTransport* audio_transport() override { return this; }
 
-    virtual int StartPlayout(int channel);
+  int AssociateSendChannel(int channel, int accociate_send_channel) override;
 
-    virtual int StartSend(int channel);
+  // AudioTransport
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  const size_t nSamples,
+                                  const size_t nBytesPerSample,
+                                  const size_t nChannels,
+                                  const uint32_t samplesPerSec,
+                                  const uint32_t totalDelayMS,
+                                  const int32_t clockDrift,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override;
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override;
+  int OnDataAvailable(const int voe_channels[],
+                      size_t number_of_voe_channels,
+                      const int16_t* audio_data,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      int audio_delay_milliseconds,
+                      int current_volume,
+                      bool key_pressed,
+                      bool need_audio_processing) override;
+  void OnData(int voe_channel,
+              const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              size_t number_of_channels,
+              size_t number_of_frames) override;
+  void PushCaptureData(int voe_channel,
+                       const void* audio_data,
+                       int bits_per_sample,
+                       int sample_rate,
+                       size_t number_of_channels,
+                       size_t number_of_frames) override;
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override;
 
-    virtual int StopReceive(int channel);
+  // AudioDeviceObserver
+  void OnErrorIsReported(const ErrorCode error) override;
+  void OnWarningIsReported(const WarningCode warning) override;
 
-    virtual int StopPlayout(int channel);
+ protected:
+  VoEBaseImpl(voe::SharedData* shared);
+  ~VoEBaseImpl() override;
 
-    virtual int StopSend(int channel);
+ private:
+  int32_t StartPlayout();
+  int32_t StopPlayout();
+  int32_t StartSend();
+  int32_t StopSend();
+  int32_t TerminateInternal();
 
-    virtual int GetVersion(char version[1024]);
+  // Helper function to process the recorded data with AudioProcessing Module,
+  // demultiplex the data to specific voe channels, encode and send to the
+  // network. When |number_of_VoE_channels| is 0, it will demultiplex the
+  // data to all the existing VoE channels.
+  // It returns new AGC microphone volume or 0 if no volume changes
+  // should be done.
+  int ProcessRecordedDataWithAPM(
+      const int voe_channels[], size_t number_of_voe_channels,
+      const void* audio_data, uint32_t sample_rate, size_t number_of_channels,
+      size_t number_of_frames, uint32_t audio_delay_milliseconds,
+      int32_t clock_drift, uint32_t volume, bool key_pressed);
 
-    virtual int LastError();
+  void GetPlayoutData(int sample_rate, size_t number_of_channels,
+                      size_t number_of_frames, bool feed_data_to_apm,
+                      void* audio_data, int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms);
 
-    virtual AudioTransport* audio_transport() { return this; }
+  // Initialize channel by setting Engine Information then initializing
+  // channel.
+  int InitializeChannel(voe::ChannelOwner* channel_owner);
+  VoiceEngineObserver* voiceEngineObserverPtr_;
+  CriticalSectionWrapper& callbackCritSect_;
 
-    // AudioTransport
-    virtual int32_t
-        RecordedDataIsAvailable(const void* audioSamples,
-                                uint32_t nSamples,
-                                uint8_t nBytesPerSample,
-                                uint8_t nChannels,
-                                uint32_t samplesPerSec,
-                                uint32_t totalDelayMS,
-                                int32_t clockDrift,
-                                uint32_t micLevel,
-                                bool keyPressed,
-                                uint32_t& newMicLevel);
-
-    virtual int32_t NeedMorePlayData(uint32_t nSamples,
-                                     uint8_t nBytesPerSample,
-                                     uint8_t nChannels,
-                                     uint32_t samplesPerSec,
-                                     void* audioSamples,
-                                     uint32_t& nSamplesOut,
-                                     int64_t* elapsed_time_ms,
-                                     int64_t* ntp_time_ms);
-
-    virtual int OnDataAvailable(const int voe_channels[],
-                                int number_of_voe_channels,
-                                const int16_t* audio_data,
-                                int sample_rate,
-                                int number_of_channels,
-                                int number_of_frames,
-                                int audio_delay_milliseconds,
-                                int volume,
-                                bool key_pressed,
-                                bool need_audio_processing);
-
-    virtual void OnData(int voe_channel, const void* audio_data,
-                        int bits_per_sample, int sample_rate,
-                        int number_of_channels, int number_of_frames);
-
-    virtual void PushCaptureData(int voe_channel, const void* audio_data,
-                                 int bits_per_sample, int sample_rate,
-                                 int number_of_channels, int number_of_frames);
-
-    virtual void PullRenderData(int bits_per_sample, int sample_rate,
-                                int number_of_channels, int number_of_frames,
-                                void* audio_data,
-                                int64_t* elapsed_time_ms,
-                                int64_t* ntp_time_ms);
-
-    // AudioDeviceObserver
-    virtual void OnErrorIsReported(ErrorCode error);
-    virtual void OnWarningIsReported(WarningCode warning);
-
-protected:
-    VoEBaseImpl(voe::SharedData* shared);
-    virtual ~VoEBaseImpl();
-
-private:
-    int32_t StartPlayout();
-    int32_t StopPlayout();
-    int32_t StartSend();
-    int32_t StopSend();
-    int32_t TerminateInternal();
-
-    // Helper function to process the recorded data with AudioProcessing Module,
-    // demultiplex the data to specific voe channels, encode and send to the
-    // network. When |number_of_VoE_channels| is 0, it will demultiplex the
-    // data to all the existing VoE channels.
-    // It returns new AGC microphone volume or 0 if no volume changes
-    // should be done.
-    int ProcessRecordedDataWithAPM(const int voe_channels[],
-                                   int number_of_voe_channels,
-                                   const void* audio_data,
-                                   uint32_t sample_rate,
-                                   uint8_t number_of_channels,
-                                   uint32_t number_of_frames,
-                                   uint32_t audio_delay_milliseconds,
-                                   int32_t clock_drift,
-                                   uint32_t volume,
-                                   bool key_pressed);
-
-    void GetPlayoutData(int sample_rate, int number_of_channels,
-                        int number_of_frames, bool feed_data_to_apm,
-                        void* audio_data,
-                        int64_t* elapsed_time_ms,
-                        int64_t* ntp_time_ms);
-
-    int32_t AddVoEVersion(char* str) const;
-
-    // Initialize channel by setting Engine Information then initializing
-    // channel.
-    int InitializeChannel(voe::ChannelOwner* channel_owner);
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
-    int32_t AddExternalTransportBuild(char* str) const;
-#endif
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-    int32_t AddExternalRecAndPlayoutBuild(char* str) const;
-#endif
-    VoiceEngineObserver* _voiceEngineObserverPtr;
-    CriticalSectionWrapper& _callbackCritSect;
-
-    bool _voiceEngineObserver;
-    AudioFrame _audioFrame;
-    voe::SharedData* _shared;
+  AudioFrame audioFrame_;
+  voe::SharedData* shared_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc
index 69aba712ef..e53dee2eff 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc
@@ -11,40 +11,77 @@
 #include "webrtc/voice_engine/include/voe_base.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/voice_engine/channel_manager.h"
+#include "webrtc/voice_engine/shared_data.h"
+#include "webrtc/voice_engine/voice_engine_fixture.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
-class VoEBaseTest : public ::testing::Test {
- protected:
-  VoEBaseTest() :
-      voe_(VoiceEngine::Create()),
-      base_(VoEBase::GetInterface(voe_)),
-      adm_(new FakeAudioDeviceModule) {
-  }
+class VoEBaseTest : public VoiceEngineFixture {};
 
-  ~VoEBaseTest() {
-    base_->Release();
-    VoiceEngine::Delete(voe_);
-  }
-
-  VoiceEngine* voe_;
-  VoEBase* base_;
-  rtc::scoped_ptr<FakeAudioDeviceModule> adm_;
-};
-
-TEST_F(VoEBaseTest, AcceptsAudioProcessingPtr) {
+TEST_F(VoEBaseTest, InitWithExternalAudioDeviceAndAudioProcessing) {
   AudioProcessing* audioproc = AudioProcessing::Create();
-  EXPECT_EQ(0, base_->Init(adm_.get(), audioproc));
+  EXPECT_EQ(0, base_->Init(&adm_, audioproc));
   EXPECT_EQ(audioproc, base_->audio_processing());
+  EXPECT_EQ(0, base_->LastError());
 }
 
-TEST_F(VoEBaseTest, AudioProcessingCreatedAfterInit) {
-  EXPECT_TRUE(base_->audio_processing() == NULL);
-  EXPECT_EQ(0, base_->Init(adm_.get(), NULL));
-  EXPECT_TRUE(base_->audio_processing() != NULL);
+TEST_F(VoEBaseTest, InitWithExternalAudioDevice) {
+  EXPECT_EQ(nullptr, base_->audio_processing());
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  EXPECT_NE(nullptr, base_->audio_processing());
+  EXPECT_EQ(0, base_->LastError());
 }
 
+TEST_F(VoEBaseTest, CreateChannelBeforeInitShouldFail) {
+  int channelID = base_->CreateChannel();
+  EXPECT_EQ(channelID, -1);
+}
+
+TEST_F(VoEBaseTest, CreateChannelAfterInit) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  int channelID = base_->CreateChannel();
+  EXPECT_NE(channelID, -1);
+  EXPECT_EQ(0, base_->DeleteChannel(channelID));
+}
+
+TEST_F(VoEBaseTest, AssociateSendChannel) {
+  AudioProcessing* audioproc = AudioProcessing::Create();
+  EXPECT_EQ(0, base_->Init(&adm_, audioproc));
+
+  const int channel_1 = base_->CreateChannel();
+
+  // Associating with a channel that does not exist should fail.
+  EXPECT_EQ(-1, base_->AssociateSendChannel(channel_1, channel_1 + 1));
+
+  const int channel_2 = base_->CreateChannel();
+
+  // Let the two channels associate with each other. This is not a normal use
+  // case. Actually, circular association should be avoided in practice. This
+  // is just to test that no crash is caused.
+  EXPECT_EQ(0, base_->AssociateSendChannel(channel_1, channel_2));
+  EXPECT_EQ(0, base_->AssociateSendChannel(channel_2, channel_1));
+
+  voe::SharedData* shared_data = static_cast<voe::SharedData*>(
+      static_cast<VoiceEngineImpl*>(voe_));
+  voe::ChannelOwner reference = shared_data->channel_manager()
+      .GetChannel(channel_1);
+  EXPECT_EQ(0, base_->DeleteChannel(channel_1));
+  // Make sure that the only use of the channel-to-delete is |reference|
+  // at this point.
+  EXPECT_EQ(1, reference.use_count());
+
+  reference = shared_data->channel_manager().GetChannel(channel_2);
+  EXPECT_EQ(0, base_->DeleteChannel(channel_2));
+  EXPECT_EQ(1, reference.use_count());
+}
+
+TEST_F(VoEBaseTest, GetVersion) {
+  char v1[1024] = {75};
+  base_->GetVersion(v1);
+  std::string v2 = VoiceEngine::GetVersionString() + "\n";
+  EXPECT_EQ(v2, v1);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc
index 2b0141fa81..6eb11b759c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc
@@ -10,289 +10,223 @@
 
 #include "webrtc/voice_engine/voe_codec_impl.h"
 
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-VoECodec* VoECodec::GetInterface(VoiceEngine* voiceEngine)
-{
+VoECodec* VoECodec::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_CODEC_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_CODEC_API
 
-VoECodecImpl::VoECodecImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoECodecImpl() - ctor");
+VoECodecImpl::VoECodecImpl(voe::SharedData* shared) : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoECodecImpl() - ctor");
 }
 
-VoECodecImpl::~VoECodecImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "~VoECodecImpl() - dtor");
+VoECodecImpl::~VoECodecImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "~VoECodecImpl() - dtor");
 }
 
-int VoECodecImpl::NumOfCodecs()
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "NumOfCodecs()");
-
-    // Number of supported codecs in the ACM
-    uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "NumOfCodecs() => %u", nSupportedCodecs);
-    return (nSupportedCodecs);
+int VoECodecImpl::NumOfCodecs() {
+  // Number of supported codecs in the ACM
+  uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
+  return (nSupportedCodecs);
 }
 
-int VoECodecImpl::GetCodec(int index, CodecInst& codec)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetCodec(index=%d, codec=?)", index);
-    CodecInst acmCodec;
-    if (AudioCodingModule::Codec(index, &acmCodec)
-            == -1)
-    {
-        _shared->SetLastError(VE_INVALID_LISTNR, kTraceError,
-            "GetCodec() invalid index");
-        return -1;
-    }
-    ACMToExternalCodecRepresentation(codec, acmCodec);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetCodec() => plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
-        "channels=%d, rate=%d", codec.plname, codec.pacsize,
-        codec.plfreq, codec.pltype, codec.channels, codec.rate);
-    return 0;
+int VoECodecImpl::GetCodec(int index, CodecInst& codec) {
+  if (AudioCodingModule::Codec(index, &codec) == -1) {
+    _shared->SetLastError(VE_INVALID_LISTNR, kTraceError,
+                          "GetCodec() invalid index");
+    return -1;
+  }
+  return 0;
 }
 
-int VoECodecImpl::SetSendCodec(int channel, const CodecInst& codec)
-{
-    CodecInst copyCodec;
-    ExternalToACMCodecRepresentation(copyCodec, codec);
+int VoECodecImpl::SetSendCodec(int channel, const CodecInst& codec) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetSendCodec(channel=%d, codec)", channel);
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "codec: plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
+               "channels=%" PRIuS ", rate=%d",
+               codec.plname, codec.pacsize, codec.plfreq, codec.pltype,
+               codec.channels, codec.rate);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  // External sanity checks performed outside the ACM
+  if ((STR_CASE_CMP(codec.plname, "L16") == 0) && (codec.pacsize >= 960)) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetSendCodec() invalid L16 packet size");
+    return -1;
+  }
+  if (!STR_CASE_CMP(codec.plname, "CN") ||
+      !STR_CASE_CMP(codec.plname, "TELEPHONE-EVENT") ||
+      !STR_CASE_CMP(codec.plname, "RED")) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetSendCodec() invalid codec name");
+    return -1;
+  }
+  if ((codec.channels != 1) && (codec.channels != 2)) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetSendCodec() invalid number of channels");
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetSendCodec() failed to locate channel");
+    return -1;
+  }
+  if (!AudioCodingModule::IsCodecValid(codec)) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetSendCodec() invalid codec");
+    return -1;
+  }
+  if (channelPtr->SetSendCodec(codec) != 0) {
+    _shared->SetLastError(VE_CANNOT_SET_SEND_CODEC, kTraceError,
+                          "SetSendCodec() failed to set send codec");
+    return -1;
+  }
 
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetSendCodec(channel=%d, codec)", channel);
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "codec: plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
-                 "channels=%d, rate=%d", codec.plname, codec.pacsize,
-                 codec.plfreq, codec.pltype, codec.channels, codec.rate);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    // External sanity checks performed outside the ACM
-    if ((STR_CASE_CMP(copyCodec.plname, "L16") == 0) &&
-            (copyCodec.pacsize >= 960))
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetSendCodec() invalid L16 packet size");
-        return -1;
-    }
-    if (!STR_CASE_CMP(copyCodec.plname, "CN")
-            || !STR_CASE_CMP(copyCodec.plname, "TELEPHONE-EVENT")
-            || !STR_CASE_CMP(copyCodec.plname, "RED"))
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetSendCodec() invalid codec name");
-        return -1;
-    }
-    if ((copyCodec.channels != 1) && (copyCodec.channels != 2))
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetSendCodec() invalid number of channels");
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetSendCodec() failed to locate channel");
-        return -1;
-    }
-    if (!AudioCodingModule::IsCodecValid(
-            (CodecInst&) copyCodec))
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetSendCodec() invalid codec");
-        return -1;
-    }
-    if (channelPtr->SetSendCodec(copyCodec) != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_SET_SEND_CODEC, kTraceError,
-            "SetSendCodec() failed to set send codec");
-        return -1;
-    }
-
-    return 0;
+  return 0;
 }
 
-int VoECodecImpl::GetSendCodec(int channel, CodecInst& codec)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetSendCodec(channel=%d, codec=?)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetSendCodec() failed to locate channel");
-        return -1;
-    }
-    CodecInst acmCodec;
-    if (channelPtr->GetSendCodec(acmCodec) != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_GET_SEND_CODEC, kTraceError,
-            "GetSendCodec() failed to get send codec");
-        return -1;
-    }
-    ACMToExternalCodecRepresentation(codec, acmCodec);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetSendCodec() => plname=%s, pacsize=%d, plfreq=%d, "
-        "channels=%d, rate=%d", codec.plname, codec.pacsize,
-        codec.plfreq, codec.channels, codec.rate);
-    return 0;
+int VoECodecImpl::GetSendCodec(int channel, CodecInst& codec) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetSendCodec() failed to locate channel");
+    return -1;
+  }
+  if (channelPtr->GetSendCodec(codec) != 0) {
+    _shared->SetLastError(VE_CANNOT_GET_SEND_CODEC, kTraceError,
+                          "GetSendCodec() failed to get send codec");
+    return -1;
+  }
+  return 0;
 }
 
-int VoECodecImpl::GetRecCodec(int channel, CodecInst& codec)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRecCodec(channel=%d, codec=?)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRecCodec() failed to locate channel");
-        return -1;
-    }
-    CodecInst acmCodec;
-    if (channelPtr->GetRecCodec(acmCodec) != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_GET_REC_CODEC, kTraceError,
-            "GetRecCodec() failed to get received codec");
-        return -1;
-    }
-    ACMToExternalCodecRepresentation(codec, acmCodec);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetRecCodec() => plname=%s, pacsize=%d, plfreq=%d, "
-        "channels=%d, rate=%d", codec.plname, codec.pacsize,
-        codec.plfreq, codec.channels, codec.rate);
-    return 0;
+int VoECodecImpl::SetBitRate(int channel, int bitrate_bps) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetBitRate(bitrate_bps=%d)", bitrate_bps);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  _shared->channel_manager().GetChannel(channel).channel()->SetBitRate(
+      bitrate_bps);
+  return 0;
 }
 
-int VoECodecImpl::SetRecPayloadType(int channel, const CodecInst& codec)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetRecPayloadType(channel=%d, codec)", channel);
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "codec: plname=%s, plfreq=%d, pltype=%d, channels=%u, "
-               "pacsize=%d, rate=%d", codec.plname, codec.plfreq, codec.pltype,
-               codec.channels, codec.pacsize, codec.rate);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRecPayloadType() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetRecPayloadType(codec);
+int VoECodecImpl::GetRecCodec(int channel, CodecInst& codec) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRecCodec() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRecCodec(codec);
 }
 
-int VoECodecImpl::GetRecPayloadType(int channel, CodecInst& codec)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRecPayloadType(channel=%d, codec)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRecPayloadType() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRecPayloadType(codec);
+int VoECodecImpl::SetRecPayloadType(int channel, const CodecInst& codec) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetRecPayloadType(channel=%d, codec)", channel);
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "codec: plname=%s, plfreq=%d, pltype=%d, channels=%" PRIuS ", "
+               "pacsize=%d, rate=%d",
+               codec.plname, codec.plfreq, codec.pltype, codec.channels,
+               codec.pacsize, codec.rate);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRecPayloadType() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetRecPayloadType(codec);
 }
 
-int VoECodecImpl::SetSendCNPayloadType(int channel, int type,
-                                       PayloadFrequencies frequency)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetSendCNPayloadType(channel=%d, type=%d, frequency=%d)",
-                 channel, type, frequency);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (type < 96 || type > 127)
-    {
-        // Only allow dynamic range: 96 to 127
-        _shared->SetLastError(VE_INVALID_PLTYPE, kTraceError,
-            "SetSendCNPayloadType() invalid payload type");
-        return -1;
-    }
-    if ((frequency != kFreq16000Hz) && (frequency != kFreq32000Hz))
-    {
-        // It is not possible to modify the payload type for CN/8000.
-        // We only allow modification of the CN payload type for CN/16000
-        // and CN/32000.
-        _shared->SetLastError(VE_INVALID_PLFREQ, kTraceError,
-            "SetSendCNPayloadType() invalid payload frequency");
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetSendCNPayloadType() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetSendCNPayloadType(type, frequency);
+int VoECodecImpl::GetRecPayloadType(int channel, CodecInst& codec) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRecPayloadType() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRecPayloadType(codec);
+}
+
+int VoECodecImpl::SetSendCNPayloadType(int channel,
+                                       int type,
+                                       PayloadFrequencies frequency) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetSendCNPayloadType(channel=%d, type=%d, frequency=%d)",
+               channel, type, frequency);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (type < 96 || type > 127) {
+    // Only allow dynamic range: 96 to 127
+    _shared->SetLastError(VE_INVALID_PLTYPE, kTraceError,
+                          "SetSendCNPayloadType() invalid payload type");
+    return -1;
+  }
+  if ((frequency != kFreq16000Hz) && (frequency != kFreq32000Hz)) {
+    // It is not possible to modify the payload type for CN/8000.
+    // We only allow modification of the CN payload type for CN/16000
+    // and CN/32000.
+    _shared->SetLastError(VE_INVALID_PLFREQ, kTraceError,
+                          "SetSendCNPayloadType() invalid payload frequency");
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetSendCNPayloadType() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetSendCNPayloadType(type, frequency);
 }
 
 int VoECodecImpl::SetFECStatus(int channel, bool enable) {
@@ -313,8 +247,6 @@ int VoECodecImpl::SetFECStatus(int channel, bool enable) {
 }
 
 int VoECodecImpl::GetFECStatus(int channel, bool& enabled) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetCodecFECStatus(channel=%d)", channel);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -330,92 +262,84 @@ int VoECodecImpl::GetFECStatus(int channel, bool& enabled) {
   return 0;
 }
 
-int VoECodecImpl::SetVADStatus(int channel, bool enable, VadModes mode,
-                               bool disableDTX)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetVADStatus(channel=%i, enable=%i, mode=%i, disableDTX=%i)",
-                 channel, enable, mode, disableDTX);
+int VoECodecImpl::SetVADStatus(int channel,
+                               bool enable,
+                               VadModes mode,
+                               bool disableDTX) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetVADStatus(channel=%i, enable=%i, mode=%i, disableDTX=%i)",
+               channel, enable, mode, disableDTX);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetVADStatus failed to locate channel");
-        return -1;
-    }
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetVADStatus failed to locate channel");
+    return -1;
+  }
 
-    ACMVADMode vadMode(VADNormal);
-    switch (mode)
-    {
-        case kVadConventional:
-            vadMode = VADNormal;
-            break;
-        case kVadAggressiveLow:
-            vadMode = VADLowBitrate;
-            break;
-        case kVadAggressiveMid:
-            vadMode = VADAggr;
-            break;
-        case kVadAggressiveHigh:
-            vadMode = VADVeryAggr;
-            break;
-    }
-    return channelPtr->SetVADStatus(enable, vadMode, disableDTX);
+  ACMVADMode vadMode(VADNormal);
+  switch (mode) {
+    case kVadConventional:
+      vadMode = VADNormal;
+      break;
+    case kVadAggressiveLow:
+      vadMode = VADLowBitrate;
+      break;
+    case kVadAggressiveMid:
+      vadMode = VADAggr;
+      break;
+    case kVadAggressiveHigh:
+      vadMode = VADVeryAggr;
+      break;
+  }
+  return channelPtr->SetVADStatus(enable, vadMode, disableDTX);
 }
 
-int VoECodecImpl::GetVADStatus(int channel, bool& enabled, VadModes& mode,
-                               bool& disabledDTX)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetVADStatus(channel=%i)", channel);
+int VoECodecImpl::GetVADStatus(int channel,
+                               bool& enabled,
+                               VadModes& mode,
+                               bool& disabledDTX) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetVADStatus failed to locate channel");
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetVADStatus failed to locate channel");
-        return -1;
-    }
+  ACMVADMode vadMode;
+  int ret = channelPtr->GetVADStatus(enabled, vadMode, disabledDTX);
 
-    ACMVADMode vadMode;
-    int ret = channelPtr->GetVADStatus(enabled, vadMode, disabledDTX);
+  if (ret != 0) {
+    _shared->SetLastError(VE_INVALID_OPERATION, kTraceError,
+                          "GetVADStatus failed to get VAD mode");
+    return -1;
+  }
+  switch (vadMode) {
+    case VADNormal:
+      mode = kVadConventional;
+      break;
+    case VADLowBitrate:
+      mode = kVadAggressiveLow;
+      break;
+    case VADAggr:
+      mode = kVadAggressiveMid;
+      break;
+    case VADVeryAggr:
+      mode = kVadAggressiveHigh;
+      break;
+  }
 
-    if (ret != 0)
-    {
-        _shared->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "GetVADStatus failed to get VAD mode");
-        return -1;
-    }
-    switch (vadMode)
-    {
-        case VADNormal:
-            mode = kVadConventional;
-            break;
-        case VADLowBitrate:
-            mode = kVadAggressiveLow;
-            break;
-        case VADAggr:
-            mode = kVadAggressiveMid;
-            break;
-        case VADVeryAggr:
-            mode = kVadAggressiveHigh;
-            break;
-    }
-
-    return 0;
+  return 0;
 }
 
 int VoECodecImpl::SetOpusMaxPlaybackRate(int channel, int frequency_hz) {
@@ -453,82 +377,8 @@ int VoECodecImpl::SetOpusDtx(int channel, bool enable_dtx) {
   return channelPtr->SetOpusDtx(enable_dtx);
 }
 
-void VoECodecImpl::ACMToExternalCodecRepresentation(CodecInst& toInst,
-                                                    const CodecInst& fromInst)
-{
-    toInst = fromInst;
-    if (STR_CASE_CMP(fromInst.plname,"SILK") == 0)
-    {
-        if (fromInst.plfreq == 12000)
-        {
-            if (fromInst.pacsize == 320)
-            {
-                toInst.pacsize = 240;
-            }
-            else if (fromInst.pacsize == 640)
-            {
-                toInst.pacsize = 480;
-            }
-            else if (fromInst.pacsize == 960)
-            {
-                toInst.pacsize = 720;
-            }
-        }
-        else if (fromInst.plfreq == 24000)
-        {
-            if (fromInst.pacsize == 640)
-            {
-                toInst.pacsize = 480;
-            }
-            else if (fromInst.pacsize == 1280)
-            {
-                toInst.pacsize = 960;
-            }
-            else if (fromInst.pacsize == 1920)
-            {
-                toInst.pacsize = 1440;
-            }
-        }
-    }
-}
-
-void VoECodecImpl::ExternalToACMCodecRepresentation(CodecInst& toInst,
-                                                    const CodecInst& fromInst)
-{
-    toInst = fromInst;
-    if (STR_CASE_CMP(fromInst.plname,"SILK") == 0)
-    {
-        if (fromInst.plfreq == 12000)
-        {
-            if (fromInst.pacsize == 240)
-            {
-                toInst.pacsize = 320;
-            }
-            else if (fromInst.pacsize == 480)
-            {
-                toInst.pacsize = 640;
-            }
-            else if (fromInst.pacsize == 720)
-            {
-                toInst.pacsize = 960;
-            }
-        }
-        else if (fromInst.plfreq == 24000)
-        {
-            if (fromInst.pacsize == 480)
-            {
-                toInst.pacsize = 640;
-            }
-            else if (fromInst.pacsize == 960)
-            {
-                toInst.pacsize = 1280;
-            }
-            else if (fromInst.pacsize == 1440)
-            {
-                toInst.pacsize = 1920;
-            }
-        }
-    }
+RtcEventLog* VoECodecImpl::GetEventLog() {
+  return _shared->channel_manager().GetEventLog();
 }
 
 #endif  // WEBRTC_VOICE_ENGINE_CODEC_API
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h
index dad808dbbf..5095f6e232 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h
@@ -15,61 +15,57 @@
 
 #include "webrtc/voice_engine/shared_data.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-class VoECodecImpl: public VoECodec
-{
-public:
-    virtual int NumOfCodecs();
+class VoECodecImpl : public VoECodec {
+ public:
+  int NumOfCodecs() override;
 
-    virtual int GetCodec(int index, CodecInst& codec);
+  int GetCodec(int index, CodecInst& codec) override;
 
-    virtual int SetSendCodec(int channel, const CodecInst& codec);
+  int SetSendCodec(int channel, const CodecInst& codec) override;
 
-    virtual int GetSendCodec(int channel, CodecInst& codec);
+  int GetSendCodec(int channel, CodecInst& codec) override;
 
-    virtual int GetRecCodec(int channel, CodecInst& codec);
+  int SetBitRate(int channel, int bitrate_bps) override;
 
-    virtual int SetSendCNPayloadType(
-        int channel, int type,
-        PayloadFrequencies frequency = kFreq16000Hz);
+  int GetRecCodec(int channel, CodecInst& codec) override;
 
-    virtual int SetRecPayloadType(int channel,
-                                  const CodecInst& codec);
+  int SetSendCNPayloadType(
+      int channel,
+      int type,
+      PayloadFrequencies frequency = kFreq16000Hz) override;
 
-    virtual int GetRecPayloadType(int channel, CodecInst& codec);
+  int SetRecPayloadType(int channel, const CodecInst& codec) override;
 
-    virtual int SetFECStatus(int channel, bool enable);
+  int GetRecPayloadType(int channel, CodecInst& codec) override;
 
-    virtual int GetFECStatus(int channel, bool& enabled);
+  int SetFECStatus(int channel, bool enable) override;
 
-    virtual int SetVADStatus(int channel,
-                             bool enable,
-                             VadModes mode = kVadConventional,
-                             bool disableDTX = false);
+  int GetFECStatus(int channel, bool& enabled) override;
 
-    virtual int GetVADStatus(int channel,
-                             bool& enabled,
-                             VadModes& mode,
-                             bool& disabledDTX);
+  int SetVADStatus(int channel,
+                   bool enable,
+                   VadModes mode = kVadConventional,
+                   bool disableDTX = false) override;
 
-    virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz);
+  int GetVADStatus(int channel,
+                   bool& enabled,
+                   VadModes& mode,
+                   bool& disabledDTX) override;
 
-    virtual int SetOpusDtx(int channel, bool enable_dtx);
+  int SetOpusMaxPlaybackRate(int channel, int frequency_hz) override;
 
-protected:
-    VoECodecImpl(voe::SharedData* shared);
-    virtual ~VoECodecImpl();
+  int SetOpusDtx(int channel, bool enable_dtx) override;
 
-private:
-    void ACMToExternalCodecRepresentation(CodecInst& toInst,
-                                          const CodecInst& fromInst);
+  RtcEventLog* GetEventLog() override;
 
-    void ExternalToACMCodecRepresentation(CodecInst& toInst,
-                                          const CodecInst& fromInst);
+ protected:
+  VoECodecImpl(voe::SharedData* shared);
+  ~VoECodecImpl() override;
 
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc
index 6eb5a5145c..f09e19e685 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc
@@ -13,7 +13,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_hardware.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
@@ -30,8 +29,7 @@ class VoECodecTest : public ::testing::Test {
         voe_codec_(VoECodec::GetInterface(voe_)),
         channel_(-1),
         adm_(new FakeAudioDeviceModule),
-        red_payload_type_(-1) {
-  }
+        red_payload_type_(-1) {}
 
   ~VoECodecTest() {}
 
@@ -62,18 +60,19 @@ class VoECodecTest : public ::testing::Test {
     // Find primary and secondary codecs.
     int num_codecs = voe_codec_->NumOfCodecs();
     int n = 0;
-    while (n < num_codecs && (!primary_found || !valid_secondary_found ||
-        !invalid_secondary_found || red_payload_type_ < 0)) {
+    while (n < num_codecs &&
+           (!primary_found || !valid_secondary_found ||
+            !invalid_secondary_found || red_payload_type_ < 0)) {
       EXPECT_EQ(0, voe_codec_->GetCodec(n, my_codec));
       if (!STR_CASE_CMP(my_codec.plname, "isac") && my_codec.plfreq == 16000) {
         memcpy(&valid_secondary_, &my_codec, sizeof(my_codec));
         valid_secondary_found = true;
       } else if (!STR_CASE_CMP(my_codec.plname, "isac") &&
-          my_codec.plfreq == 32000) {
+                 my_codec.plfreq == 32000) {
         memcpy(&invalid_secondary_, &my_codec, sizeof(my_codec));
         invalid_secondary_found = true;
       } else if (!STR_CASE_CMP(my_codec.plname, "L16") &&
-          my_codec.plfreq == 16000) {
+                 my_codec.plfreq == 16000) {
         memcpy(&primary_, &my_codec, sizeof(my_codec));
         primary_found = true;
       } else if (!STR_CASE_CMP(my_codec.plname, "RED")) {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.cc
index 2d775e34ca..cf6ab1dd7b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.cc
@@ -10,8 +10,8 @@
 
 #include "webrtc/voice_engine/voe_dtmf_impl.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/output_mixer.h"
@@ -20,242 +20,197 @@
 
 namespace webrtc {
 
-VoEDtmf* VoEDtmf::GetInterface(VoiceEngine* voiceEngine)
-{
+VoEDtmf* VoEDtmf::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_DTMF_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_DTMF_API
 
-VoEDtmfImpl::VoEDtmfImpl(voe::SharedData* shared) :
-    _dtmfFeedback(true),
-    _dtmfDirectFeedback(false),
-    _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEDtmfImpl::VoEDtmfImpl() - ctor");
+VoEDtmfImpl::VoEDtmfImpl(voe::SharedData* shared)
+    : _dtmfFeedback(true), _dtmfDirectFeedback(false), _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEDtmfImpl::VoEDtmfImpl() - ctor");
 }
 
-VoEDtmfImpl::~VoEDtmfImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEDtmfImpl::~VoEDtmfImpl() - dtor");
+VoEDtmfImpl::~VoEDtmfImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEDtmfImpl::~VoEDtmfImpl() - dtor");
 }
 
 int VoEDtmfImpl::SendTelephoneEvent(int channel,
                                     int eventCode,
                                     bool outOfBand,
                                     int lengthMs,
-                                    int attenuationDb)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SendTelephoneEvent(channel=%d, eventCode=%d, outOfBand=%d,"
-                 "length=%d, attenuationDb=%d)",
-                 channel, eventCode, (int)outOfBand, lengthMs, attenuationDb);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SendTelephoneEvent() failed to locate channel");
-        return -1;
-    }
-    if (!channelPtr->Sending())
-    {
-        _shared->SetLastError(VE_NOT_SENDING, kTraceError,
-            "SendTelephoneEvent() sending is not active");
-        return -1;
-    }
+                                    int attenuationDb) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SendTelephoneEvent(channel=%d, eventCode=%d, outOfBand=%d,"
+               "length=%d, attenuationDb=%d)",
+               channel, eventCode, (int)outOfBand, lengthMs, attenuationDb);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SendTelephoneEvent() failed to locate channel");
+    return -1;
+  }
+  if (!channelPtr->Sending()) {
+    _shared->SetLastError(VE_NOT_SENDING, kTraceError,
+                          "SendTelephoneEvent() sending is not active");
+    return -1;
+  }
 
-    // Sanity check
-    const int maxEventCode = outOfBand ?
-        static_cast<int>(kMaxTelephoneEventCode) :
-        static_cast<int>(kMaxDtmfEventCode);
-    const bool testFailed = ((eventCode < 0) ||
-        (eventCode > maxEventCode) ||
-        (lengthMs < kMinTelephoneEventDuration) ||
-        (lengthMs > kMaxTelephoneEventDuration) ||
-        (attenuationDb < kMinTelephoneEventAttenuation) ||
-        (attenuationDb > kMaxTelephoneEventAttenuation));
-    if (testFailed)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SendTelephoneEvent() invalid parameter(s)");
-        return -1;
-    }
+  // Sanity check
+  const int maxEventCode = outOfBand ? static_cast<int>(kMaxTelephoneEventCode)
+                                     : static_cast<int>(kMaxDtmfEventCode);
+  const bool testFailed = ((eventCode < 0) || (eventCode > maxEventCode) ||
+                           (lengthMs < kMinTelephoneEventDuration) ||
+                           (lengthMs > kMaxTelephoneEventDuration) ||
+                           (attenuationDb < kMinTelephoneEventAttenuation) ||
+                           (attenuationDb > kMaxTelephoneEventAttenuation));
+  if (testFailed) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SendTelephoneEvent() invalid parameter(s)");
+    return -1;
+  }
 
-    const bool isDtmf =
-        (eventCode >= 0) && (eventCode <= kMaxDtmfEventCode);
-    const bool playDtmfToneDirect =
-        isDtmf && (_dtmfFeedback && _dtmfDirectFeedback);
+  const bool isDtmf = (eventCode >= 0) && (eventCode <= kMaxDtmfEventCode);
+  const bool playDtmfToneDirect =
+      isDtmf && (_dtmfFeedback && _dtmfDirectFeedback);
 
-    if (playDtmfToneDirect)
-    {
-        // Mute the microphone signal while playing back the tone directly.
-        // This is to reduce the risk of introducing echo from the added output.
-        _shared->transmit_mixer()->UpdateMuteMicrophoneTime(lengthMs);
+  if (playDtmfToneDirect) {
+    // Mute the microphone signal while playing back the tone directly.
+    // This is to reduce the risk of introducing echo from the added output.
+    _shared->transmit_mixer()->UpdateMuteMicrophoneTime(lengthMs);
 
-        // Play out local feedback tone directly (same approach for both inband
-        // and outband).
-        // Reduce the length of the the tone with 80ms to reduce risk of echo.
-        // For non-direct feedback, outband and inband cases are handled
-        // differently.
-        _shared->output_mixer()->PlayDtmfTone(eventCode, lengthMs - 80,
-                                            attenuationDb);
-    }
+    // Play out local feedback tone directly (same approach for both inband
+    // and outband).
+    // Reduce the length of the the tone with 80ms to reduce risk of echo.
+    // For non-direct feedback, outband and inband cases are handled
+    // differently.
+    _shared->output_mixer()->PlayDtmfTone(eventCode, lengthMs - 80,
+                                          attenuationDb);
+  }
 
-    if (outOfBand)
-    {
-        // The RTP/RTCP module will always deliver OnPlayTelephoneEvent when
-        // an event is transmitted. It is up to the VoE to utilize it or not.
-        // This flag ensures that feedback/playout is enabled; however, the
-        // channel object must still parse out the Dtmf events (0-15) from
-        // all possible events (0-255).
-        const bool playDTFMEvent = (_dtmfFeedback && !_dtmfDirectFeedback);
+  if (outOfBand) {
+    // The RTP/RTCP module will always deliver OnPlayTelephoneEvent when
+    // an event is transmitted. It is up to the VoE to utilize it or not.
+    // This flag ensures that feedback/playout is enabled; however, the
+    // channel object must still parse out the Dtmf events (0-15) from
+    // all possible events (0-255).
+    const bool playDTFMEvent = (_dtmfFeedback && !_dtmfDirectFeedback);
 
-        return channelPtr->SendTelephoneEventOutband(eventCode,
-                                                     lengthMs,
-                                                     attenuationDb,
-                                                     playDTFMEvent);
-    }
-    else
-    {
-        // For Dtmf tones, we want to ensure that inband tones are played out
-        // in sync with the transmitted audio. This flag is utilized by the
-        // channel object to determine if the queued Dtmf e vent shall also
-        // be fed to the output mixer in the same step as input audio is
-        // replaced by inband Dtmf tones.
-        const bool playDTFMEvent =
-            (isDtmf && _dtmfFeedback && !_dtmfDirectFeedback);
+    return channelPtr->SendTelephoneEventOutband(eventCode, lengthMs,
+                                                 attenuationDb, playDTFMEvent);
+  } else {
+    // For Dtmf tones, we want to ensure that inband tones are played out
+    // in sync with the transmitted audio. This flag is utilized by the
+    // channel object to determine if the queued Dtmf e vent shall also
+    // be fed to the output mixer in the same step as input audio is
+    // replaced by inband Dtmf tones.
+    const bool playDTFMEvent =
+        (isDtmf && _dtmfFeedback && !_dtmfDirectFeedback);
 
-        return channelPtr->SendTelephoneEventInband(eventCode,
-                                                    lengthMs,
-                                                    attenuationDb,
-                                                    playDTFMEvent);
-    }
+    return channelPtr->SendTelephoneEventInband(eventCode, lengthMs,
+                                                attenuationDb, playDTFMEvent);
+  }
 }
 
 int VoEDtmfImpl::SetSendTelephoneEventPayloadType(int channel,
-                                                  unsigned char type)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetSendTelephoneEventPayloadType(channel=%d, type=%u)",
-                 channel, type);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetSendTelephoneEventPayloadType() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetSendTelephoneEventPayloadType(type);
+                                                  unsigned char type) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetSendTelephoneEventPayloadType(channel=%d, type=%u)", channel,
+               type);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
+        "SetSendTelephoneEventPayloadType() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetSendTelephoneEventPayloadType(type);
 }
 
 int VoEDtmfImpl::GetSendTelephoneEventPayloadType(int channel,
-                                                  unsigned char& type)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetSendTelephoneEventPayloadType(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetSendTelephoneEventPayloadType() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetSendTelephoneEventPayloadType(type);
+                                                  unsigned char& type) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
+        "GetSendTelephoneEventPayloadType() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetSendTelephoneEventPayloadType(type);
 }
 
-int VoEDtmfImpl::PlayDtmfTone(int eventCode,
-                              int lengthMs,
-                              int attenuationDb)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "PlayDtmfTone(eventCode=%d, lengthMs=%d, attenuationDb=%d)",
-                 eventCode, lengthMs, attenuationDb);
+int VoEDtmfImpl::PlayDtmfTone(int eventCode, int lengthMs, int attenuationDb) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "PlayDtmfTone(eventCode=%d, lengthMs=%d, attenuationDb=%d)",
+               eventCode, lengthMs, attenuationDb);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (!_shared->audio_device()->Playing())
-    {
-        _shared->SetLastError(VE_NOT_PLAYING, kTraceError,
-            "PlayDtmfTone() no channel is playing out");
-        return -1;
-    }
-    if ((eventCode < kMinDtmfEventCode) ||
-        (eventCode > kMaxDtmfEventCode) ||
-        (lengthMs < kMinTelephoneEventDuration) ||
-        (lengthMs > kMaxTelephoneEventDuration) ||
-        (attenuationDb <kMinTelephoneEventAttenuation) ||
-        (attenuationDb > kMaxTelephoneEventAttenuation))
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-        "PlayDtmfTone() invalid tone parameter(s)");
-        return -1;
-    }
-    return _shared->output_mixer()->PlayDtmfTone(eventCode, lengthMs,
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (!_shared->audio_device()->Playing()) {
+    _shared->SetLastError(VE_NOT_PLAYING, kTraceError,
+                          "PlayDtmfTone() no channel is playing out");
+    return -1;
+  }
+  if ((eventCode < kMinDtmfEventCode) || (eventCode > kMaxDtmfEventCode) ||
+      (lengthMs < kMinTelephoneEventDuration) ||
+      (lengthMs > kMaxTelephoneEventDuration) ||
+      (attenuationDb < kMinTelephoneEventAttenuation) ||
+      (attenuationDb > kMaxTelephoneEventAttenuation)) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "PlayDtmfTone() invalid tone parameter(s)");
+    return -1;
+  }
+  return _shared->output_mixer()->PlayDtmfTone(eventCode, lengthMs,
                                                attenuationDb);
 }
 
-int VoEDtmfImpl::SetDtmfFeedbackStatus(bool enable, bool directFeedback)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)",
-                 (int)enable, (int)directFeedback);
+int VoEDtmfImpl::SetDtmfFeedbackStatus(bool enable, bool directFeedback) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)",
+               (int)enable, (int)directFeedback);
 
-    CriticalSectionScoped sc(_shared->crit_sec());
+  CriticalSectionScoped sc(_shared->crit_sec());
 
-    _dtmfFeedback = enable;
-    _dtmfDirectFeedback = directFeedback;
+  _dtmfFeedback = enable;
+  _dtmfDirectFeedback = directFeedback;
 
-    return 0;
+  return 0;
 }
 
-int VoEDtmfImpl::GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetDtmfFeedbackStatus()");
+int VoEDtmfImpl::GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) {
+  CriticalSectionScoped sc(_shared->crit_sec());
 
-    CriticalSectionScoped sc(_shared->crit_sec());
-
-    enabled = _dtmfFeedback;
-    directFeedback = _dtmfDirectFeedback;
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetDtmfFeedbackStatus() => enabled=%d, directFeedback=%d",
-        enabled, directFeedback);
-    return 0;
+  enabled = _dtmfFeedback;
+  directFeedback = _dtmfDirectFeedback;
+  return 0;
 }
 #endif  // #ifdef WEBRTC_VOICE_ENGINE_DTMF_API
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.h
index 81a95c0b19..a62188a66f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_dtmf_impl.h
@@ -14,42 +14,38 @@
 #include "webrtc/voice_engine/include/voe_dtmf.h"
 #include "webrtc/voice_engine/shared_data.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-class VoEDtmfImpl : public VoEDtmf
-{
-public:
-    virtual int SendTelephoneEvent(
-        int channel,
-        int eventCode,
-        bool outOfBand = true,
-        int lengthMs = 160,
-        int attenuationDb = 10);
+class VoEDtmfImpl : public VoEDtmf {
+ public:
+  int SendTelephoneEvent(int channel,
+                         int eventCode,
+                         bool outOfBand = true,
+                         int lengthMs = 160,
+                         int attenuationDb = 10) override;
 
-    virtual int SetSendTelephoneEventPayloadType(int channel,
-                                                 unsigned char type);
+  int SetSendTelephoneEventPayloadType(int channel,
+                                       unsigned char type) override;
 
-    virtual int GetSendTelephoneEventPayloadType(int channel,
-                                                 unsigned char& type);
+  int GetSendTelephoneEventPayloadType(int channel,
+                                       unsigned char& type) override;
 
-    virtual int SetDtmfFeedbackStatus(bool enable,
-        bool directFeedback = false);
+  int SetDtmfFeedbackStatus(bool enable, bool directFeedback = false) override;
 
-    virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback);
+  int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) override;
 
-    virtual int PlayDtmfTone(int eventCode,
-                             int lengthMs = 200,
-                             int attenuationDb = 10);
+  int PlayDtmfTone(int eventCode,
+                   int lengthMs = 200,
+                   int attenuationDb = 10) override;
 
-protected:
-    VoEDtmfImpl(voe::SharedData* shared);
-    virtual ~VoEDtmfImpl();
+ protected:
+  VoEDtmfImpl(voe::SharedData* shared);
+  ~VoEDtmfImpl() override;
 
-private:
-    bool _dtmfFeedback;
-    bool _dtmfDirectFeedback;
-    voe::SharedData* _shared;
+ private:
+  bool _dtmfFeedback;
+  bool _dtmfDirectFeedback;
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.cc
index 2cf219084a..1fd1cfc909 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.cc
@@ -10,8 +10,8 @@
 
 #include "webrtc/voice_engine/voe_external_media_impl.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/output_mixer.h"
@@ -20,18 +20,16 @@
 
 namespace webrtc {
 
-VoEExternalMedia* VoEExternalMedia::GetInterface(VoiceEngine* voiceEngine)
-{
+VoEExternalMedia* VoEExternalMedia::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
@@ -40,107 +38,88 @@ VoEExternalMedia* VoEExternalMedia::GetInterface(VoiceEngine* voiceEngine)
 VoEExternalMediaImpl::VoEExternalMediaImpl(voe::SharedData* shared)
     :
 #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-    playout_delay_ms_(0),
+      playout_delay_ms_(0),
 #endif
-    shared_(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(shared_->instance_id(), -1),
-                 "VoEExternalMediaImpl() - ctor");
+      shared_(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(shared_->instance_id(), -1),
+               "VoEExternalMediaImpl() - ctor");
 }
 
-VoEExternalMediaImpl::~VoEExternalMediaImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(shared_->instance_id(), -1),
-                 "~VoEExternalMediaImpl() - dtor");
+VoEExternalMediaImpl::~VoEExternalMediaImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(shared_->instance_id(), -1),
+               "~VoEExternalMediaImpl() - dtor");
 }
 
 int VoEExternalMediaImpl::RegisterExternalMediaProcessing(
     int channel,
     ProcessingTypes type,
-    VoEMediaProcess& processObject)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
-                 "RegisterExternalMediaProcessing(channel=%d, type=%d, "
-                 "processObject=0x%x)", channel, type, &processObject);
-    if (!shared_->statistics().Initialized())
-    {
-        shared_->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    switch (type)
-    {
-        case kPlaybackPerChannel:
-        case kRecordingPerChannel:
-        {
-            voe::ChannelOwner ch =
-                shared_->channel_manager().GetChannel(channel);
-            voe::Channel* channelPtr = ch.channel();
-            if (channelPtr == NULL)
-            {
-                shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                    "RegisterExternalMediaProcessing() failed to locate "
-                    "channel");
-                return -1;
-            }
-            return channelPtr->RegisterExternalMediaProcessing(type,
-                                                               processObject);
-        }
-        case kPlaybackAllChannelsMixed:
-        {
-            return shared_->output_mixer()->RegisterExternalMediaProcessing(
-                processObject);
-        }
-        case kRecordingAllChannelsMixed:
-        case kRecordingPreprocessing:
-        {
-            return shared_->transmit_mixer()->RegisterExternalMediaProcessing(
-                &processObject, type);
-        }
-    }
+    VoEMediaProcess& processObject) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
+               "RegisterExternalMediaProcessing(channel=%d, type=%d, "
+               "processObject=0x%x)",
+               channel, type, &processObject);
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
+  }
+  switch (type) {
+    case kPlaybackPerChannel:
+    case kRecordingPerChannel: {
+      voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+      voe::Channel* channelPtr = ch.channel();
+      if (channelPtr == NULL) {
+        shared_->SetLastError(
+            VE_CHANNEL_NOT_VALID, kTraceError,
+            "RegisterExternalMediaProcessing() failed to locate "
+            "channel");
+        return -1;
+      }
+      return channelPtr->RegisterExternalMediaProcessing(type, processObject);
+    }
+    case kPlaybackAllChannelsMixed: {
+      return shared_->output_mixer()->RegisterExternalMediaProcessing(
+          processObject);
+    }
+    case kRecordingAllChannelsMixed:
+    case kRecordingPreprocessing: {
+      return shared_->transmit_mixer()->RegisterExternalMediaProcessing(
+          &processObject, type);
+    }
+  }
+  return -1;
 }
 
 int VoEExternalMediaImpl::DeRegisterExternalMediaProcessing(
     int channel,
-    ProcessingTypes type)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
-                 "DeRegisterExternalMediaProcessing(channel=%d)", channel);
-    if (!shared_->statistics().Initialized())
-    {
-        shared_->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    switch (type)
-    {
-        case kPlaybackPerChannel:
-        case kRecordingPerChannel:
-        {
-            voe::ChannelOwner ch =
-                shared_->channel_manager().GetChannel(channel);
-            voe::Channel* channelPtr = ch.channel();
-            if (channelPtr == NULL)
-            {
-                shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                    "RegisterExternalMediaProcessing() "
-                    "failed to locate channel");
-                return -1;
-            }
-            return channelPtr->DeRegisterExternalMediaProcessing(type);
-        }
-        case kPlaybackAllChannelsMixed:
-        {
-            return shared_->output_mixer()->
-                DeRegisterExternalMediaProcessing();
-        }
-        case kRecordingAllChannelsMixed:
-        case kRecordingPreprocessing:
-        {
-            return shared_->transmit_mixer()->
-                DeRegisterExternalMediaProcessing(type);
-        }
-    }
+    ProcessingTypes type) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
+               "DeRegisterExternalMediaProcessing(channel=%d)", channel);
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
+  }
+  switch (type) {
+    case kPlaybackPerChannel:
+    case kRecordingPerChannel: {
+      voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+      voe::Channel* channelPtr = ch.channel();
+      if (channelPtr == NULL) {
+        shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                              "RegisterExternalMediaProcessing() "
+                              "failed to locate channel");
+        return -1;
+      }
+      return channelPtr->DeRegisterExternalMediaProcessing(type);
+    }
+    case kPlaybackAllChannelsMixed: {
+      return shared_->output_mixer()->DeRegisterExternalMediaProcessing();
+    }
+    case kRecordingAllChannelsMixed:
+    case kRecordingPreprocessing: {
+      return shared_->transmit_mixer()->DeRegisterExternalMediaProcessing(type);
+    }
+  }
+  return -1;
 }
 
 int VoEExternalMediaImpl::SetExternalRecordingStatus(bool enable)
@@ -148,7 +127,7 @@ int VoEExternalMediaImpl::SetExternalRecordingStatus(bool enable)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
                  "SetExternalRecordingStatus(enable=%d)", enable);
 #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-    if (shared_->audio_device()->Recording())
+    if (shared_->audio_device() && shared_->audio_device()->Recording())
     {
         shared_->SetLastError(VE_ALREADY_SENDING, kTraceError,
             "SetExternalRecordingStatus() cannot set state while sending");
@@ -268,7 +247,7 @@ int VoEExternalMediaImpl::SetExternalPlayoutStatus(bool enable)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(shared_->instance_id(), -1),
                  "SetExternalPlayoutStatus(enable=%d)", enable);
 #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-    if (shared_->audio_device()->Playing())
+    if (shared_->audio_device() && shared_->audio_device()->Playing())
     {
         shared_->SetLastError(VE_ALREADY_SENDING, kTraceError,
             "SetExternalPlayoutStatus() cannot set state while playing");
@@ -414,62 +393,55 @@ int VoEExternalMediaImpl::ExternalPlayoutGetData(
 
 int VoEExternalMediaImpl::GetAudioFrame(int channel, int desired_sample_rate_hz,
                                         AudioFrame* frame) {
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
-                 VoEId(shared_->instance_id(), channel),
-                 "GetAudioFrame(channel=%d, desired_sample_rate_hz=%d)",
-                 channel, desired_sample_rate_hz);
-    if (!shared_->statistics().Initialized())
-    {
-        shared_->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetAudioFrame() failed to locate channel");
-        return -1;
-    }
-    if (!channelPtr->ExternalMixing()) {
-        shared_->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "GetAudioFrame() was called on channel that is not"
-            " externally mixed.");
-        return -1;
-    }
-    if (!channelPtr->Playing()) {
-        shared_->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "GetAudioFrame() was called on channel that is not playing.");
-        return -1;
-    }
-    if (desired_sample_rate_hz == -1) {
-          shared_->SetLastError(VE_BAD_ARGUMENT, kTraceError,
-              "GetAudioFrame() was called with bad sample rate.");
-          return -1;
-    }
-    frame->sample_rate_hz_ = desired_sample_rate_hz == 0 ? -1 :
-                             desired_sample_rate_hz;
-    return channelPtr->GetAudioFrame(channel, *frame);
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetAudioFrame() failed to locate channel");
+    return -1;
+  }
+  if (!channelPtr->ExternalMixing()) {
+    shared_->SetLastError(VE_INVALID_OPERATION, kTraceError,
+                          "GetAudioFrame() was called on channel that is not"
+                          " externally mixed.");
+    return -1;
+  }
+  if (!channelPtr->Playing()) {
+    shared_->SetLastError(
+        VE_INVALID_OPERATION, kTraceError,
+        "GetAudioFrame() was called on channel that is not playing.");
+    return -1;
+  }
+  if (desired_sample_rate_hz == -1) {
+    shared_->SetLastError(VE_BAD_ARGUMENT, kTraceError,
+                          "GetAudioFrame() was called with bad sample rate.");
+    return -1;
+  }
+  frame->sample_rate_hz_ =
+      desired_sample_rate_hz == 0 ? -1 : desired_sample_rate_hz;
+  return channelPtr->GetAudioFrame(channel, frame);
 }
 
 int VoEExternalMediaImpl::SetExternalMixing(int channel, bool enable) {
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
-                 VoEId(shared_->instance_id(), channel),
-                 "SetExternalMixing(channel=%d, enable=%d)", channel, enable);
-    if (!shared_->statistics().Initialized())
-    {
-        shared_->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetExternalMixing() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetExternalMixing(enable);
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+               VoEId(shared_->instance_id(), channel),
+               "SetExternalMixing(channel=%d, enable=%d)", channel, enable);
+  if (!shared_->statistics().Initialized()) {
+    shared_->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetExternalMixing() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetExternalMixing(enable);
 }
 
 #endif  // WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.h
index 2f3aa71d8a..36f6cb802a 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.h
@@ -17,55 +17,53 @@
 
 namespace webrtc {
 
-class VoEExternalMediaImpl : public VoEExternalMedia
-{
-public:
-    virtual int RegisterExternalMediaProcessing(
-        int channel,
-        ProcessingTypes type,
-        VoEMediaProcess& processObject);
+class VoEExternalMediaImpl : public VoEExternalMedia {
+ public:
+  int RegisterExternalMediaProcessing(int channel,
+                                      ProcessingTypes type,
+                                      VoEMediaProcess& processObject) override;
 
-    virtual int DeRegisterExternalMediaProcessing(
-        int channel,
-        ProcessingTypes type);
+  int DeRegisterExternalMediaProcessing(int channel,
+                                        ProcessingTypes type) override;
 
-    virtual int SetExternalRecordingStatus(bool enable);
+  virtual int SetExternalRecordingStatus(bool enable);
 
-    virtual int SetExternalPlayoutStatus(bool enable);
+  virtual int SetExternalPlayoutStatus(bool enable);
 
-    virtual int ExternalRecordingInsertData(
+  virtual int ExternalRecordingInsertData(
         const int16_t speechData10ms[],
         int lengthSamples,
         int samplingFreqHz,
         int current_delay_ms);
 
-    // Insertion of far-end data as actually played out to the OS audio driver
-    virtual int ExternalPlayoutData(
+  // Insertion of far-end data as actually played out to the OS audio driver
+  virtual int ExternalPlayoutData(
         int16_t speechData10ms[],
         int samplingFreqHz,
         int num_channels,
         int current_delay_ms,
         int& lengthSamples);
 
-    virtual int ExternalPlayoutGetData(int16_t speechData10ms[],
-                                       int samplingFreqHz,
-                                       int current_delay_ms,
-                                       int& lengthSamples);
+  virtual int ExternalPlayoutGetData(int16_t speechData10ms[],
+                                     int samplingFreqHz,
+                                     int current_delay_ms,
+                                     int& lengthSamples);
 
-    virtual int GetAudioFrame(int channel, int desired_sample_rate_hz,
-                              AudioFrame* frame);
+  int GetAudioFrame(int channel,
+                    int desired_sample_rate_hz,
+                    AudioFrame* frame) override;
 
-    virtual int SetExternalMixing(int channel, bool enable);
+  int SetExternalMixing(int channel, bool enable) override;
 
-protected:
-    VoEExternalMediaImpl(voe::SharedData* shared);
-    virtual ~VoEExternalMediaImpl();
+ protected:
+  VoEExternalMediaImpl(voe::SharedData* shared);
+  ~VoEExternalMediaImpl() override;
 
-private:
+ private:
 #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-    int playout_delay_ms_;
+  int playout_delay_ms_;
 #endif
-    voe::SharedData* shared_;
+  voe::SharedData* shared_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.cc
index 95e9d21b39..2091e7073b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.cc
@@ -10,10 +10,10 @@
 
 #include "webrtc/voice_engine/voe_file_impl.h"
 
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/output_mixer.h"
@@ -22,71 +22,60 @@
 
 namespace webrtc {
 
-VoEFile* VoEFile::GetInterface(VoiceEngine* voiceEngine)
-{
+VoEFile* VoEFile::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_FILE_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_FILE_API
 
-VoEFileImpl::VoEFileImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEFileImpl::VoEFileImpl() - ctor");
+VoEFileImpl::VoEFileImpl(voe::SharedData* shared) : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEFileImpl::VoEFileImpl() - ctor");
 }
 
-VoEFileImpl::~VoEFileImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEFileImpl::~VoEFileImpl() - dtor");
+VoEFileImpl::~VoEFileImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEFileImpl::~VoEFileImpl() - dtor");
 }
 
-int VoEFileImpl::StartPlayingFileLocally(
-    int channel,
-    const char fileNameUTF8[1024],
-    bool loop, FileFormats format,
-    float volumeScaling,
-    int startPointMs,
-    int stopPointMs)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartPlayingFileLocally(channel=%d, fileNameUTF8[]=%s, "
-                 "loop=%d, format=%d, volumeScaling=%5.3f, startPointMs=%d,"
-                 " stopPointMs=%d)",
-                 channel, fileNameUTF8, loop, format, volumeScaling,
-                 startPointMs, stopPointMs);
-    assert(1024 == FileWrapper::kMaxFileNameSize);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StartPlayingFileLocally() failed to locate channel");
-        return -1;
-    }
+int VoEFileImpl::StartPlayingFileLocally(int channel,
+                                         const char fileNameUTF8[1024],
+                                         bool loop,
+                                         FileFormats format,
+                                         float volumeScaling,
+                                         int startPointMs,
+                                         int stopPointMs) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartPlayingFileLocally(channel=%d, fileNameUTF8[]=%s, "
+               "loop=%d, format=%d, volumeScaling=%5.3f, startPointMs=%d,"
+               " stopPointMs=%d)",
+               channel, fileNameUTF8, loop, format, volumeScaling, startPointMs,
+               stopPointMs);
+  static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StartPlayingFileLocally() failed to locate channel");
+    return -1;
+  }
 
-    return channelPtr->StartPlayingFileLocally(fileNameUTF8,
-                                               loop,
-                                               format,
-                                               startPointMs,
-                                               volumeScaling,
-                                               stopPointMs,
-                                               NULL);
+  return channelPtr->StartPlayingFileLocally(fileNameUTF8, loop, format,
+                                             startPointMs, volumeScaling,
+                                             stopPointMs, NULL);
 }
 
 int VoEFileImpl::StartPlayingFileLocally(int channel,
@@ -94,74 +83,59 @@ int VoEFileImpl::StartPlayingFileLocally(int channel,
                                          FileFormats format,
                                          float volumeScaling,
                                          int startPointMs,
-                                         int stopPointMs)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartPlayingFileLocally(channel=%d, stream, format=%d, "
-                 "volumeScaling=%5.3f, startPointMs=%d, stopPointMs=%d)",
-                 channel, format, volumeScaling, startPointMs, stopPointMs);
+                                         int stopPointMs) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartPlayingFileLocally(channel=%d, stream, format=%d, "
+               "volumeScaling=%5.3f, startPointMs=%d, stopPointMs=%d)",
+               channel, format, volumeScaling, startPointMs, stopPointMs);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StartPlayingFileLocally() failed to locate channel");
-        return -1;
-    }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StartPlayingFileLocally() failed to locate channel");
+    return -1;
+  }
 
-    return channelPtr->StartPlayingFileLocally(stream,
-                                               format,
-                                               startPointMs,
-                                               volumeScaling,
-                                               stopPointMs,
-                                               NULL);
+  return channelPtr->StartPlayingFileLocally(stream, format, startPointMs,
+                                             volumeScaling, stopPointMs, NULL);
 }
 
-int VoEFileImpl::StopPlayingFileLocally(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopPlayingFileLocally()");
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StopPlayingFileLocally() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->StopPlayingFileLocally();
+int VoEFileImpl::StopPlayingFileLocally(int channel) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StopPlayingFileLocally()");
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StopPlayingFileLocally() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->StopPlayingFileLocally();
 }
 
-int VoEFileImpl::IsPlayingFileLocally(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "IsPlayingFileLocally(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StopPlayingFileLocally() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->IsPlayingFileLocally();
+int VoEFileImpl::IsPlayingFileLocally(int channel) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "StopPlayingFileLocally() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->IsPlayingFileLocally();
 }
 
 int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
@@ -169,426 +143,332 @@ int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
                                               bool loop,
                                               bool mixWithMicrophone,
                                               FileFormats format,
-                                              float volumeScaling)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartPlayingFileAsMicrophone(channel=%d, fileNameUTF8=%s, "
-                 "loop=%d, mixWithMicrophone=%d, format=%d, "
-                 "volumeScaling=%5.3f)",
-                 channel, fileNameUTF8, loop, mixWithMicrophone, format,
-                 volumeScaling);
-    assert(1024 == FileWrapper::kMaxFileNameSize);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+                                              float volumeScaling) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartPlayingFileAsMicrophone(channel=%d, fileNameUTF8=%s, "
+               "loop=%d, mixWithMicrophone=%d, format=%d, "
+               "volumeScaling=%5.3f)",
+               channel, fileNameUTF8, loop, mixWithMicrophone, format,
+               volumeScaling);
+  static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+
+  const uint32_t startPointMs(0);
+  const uint32_t stopPointMs(0);
+
+  if (channel == -1) {
+    int res = _shared->transmit_mixer()->StartPlayingFileAsMicrophone(
+        fileNameUTF8, loop, format, startPointMs, volumeScaling, stopPointMs,
+        NULL);
+    if (res) {
+      WEBRTC_TRACE(
+          kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+          "StartPlayingFileAsMicrophone() failed to start playing file");
+      return (-1);
+    } else {
+      _shared->transmit_mixer()->SetMixWithMicStatus(mixWithMicrophone);
+      return (0);
+    }
+  } else {
+    // Add file after demultiplexing <=> affects one channel only
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(
+          VE_CHANNEL_NOT_VALID, kTraceError,
+          "StartPlayingFileAsMicrophone() failed to locate channel");
+      return -1;
     }
 
-    const uint32_t startPointMs(0);
-    const uint32_t stopPointMs(0);
-
-    if (channel == -1)
-    {
-        int res = _shared->transmit_mixer()->StartPlayingFileAsMicrophone(
-            fileNameUTF8,
-            loop,
-            format,
-            startPointMs,
-            volumeScaling,
-            stopPointMs,
-            NULL);
-        if (res)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartPlayingFileAsMicrophone() failed to start playing file");
-            return(-1);
-        }
-        else
-        {
-            _shared->transmit_mixer()->SetMixWithMicStatus(mixWithMicrophone);
-            return(0);
-        }
-    }
-    else
-    {
-        // Add file after demultiplexing <=> affects one channel only
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "StartPlayingFileAsMicrophone() failed to locate channel");
-            return -1;
-        }
-
-        int res = channelPtr->StartPlayingFileAsMicrophone(fileNameUTF8,
-                                                           loop,
-                                                           format,
-                                                           startPointMs,
-                                                           volumeScaling,
-                                                           stopPointMs,
-                                                           NULL);
-        if (res)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartPlayingFileAsMicrophone() failed to start playing file");
-            return -1;
-        }
-        else
-        {
-            channelPtr->SetMixWithMicStatus(mixWithMicrophone);
-            return 0;
-        }
+    int res = channelPtr->StartPlayingFileAsMicrophone(
+        fileNameUTF8, loop, format, startPointMs, volumeScaling, stopPointMs,
+        NULL);
+    if (res) {
+      WEBRTC_TRACE(
+          kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+          "StartPlayingFileAsMicrophone() failed to start playing file");
+      return -1;
+    } else {
+      channelPtr->SetMixWithMicStatus(mixWithMicrophone);
+      return 0;
     }
+  }
 }
 
 int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
                                               InStream* stream,
                                               bool mixWithMicrophone,
                                               FileFormats format,
-                                              float volumeScaling)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartPlayingFileAsMicrophone(channel=%d, stream,"
-                 " mixWithMicrophone=%d, format=%d, volumeScaling=%5.3f)",
-                 channel, mixWithMicrophone, format, volumeScaling);
+                                              float volumeScaling) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartPlayingFileAsMicrophone(channel=%d, stream,"
+               " mixWithMicrophone=%d, format=%d, volumeScaling=%5.3f)",
+               channel, mixWithMicrophone, format, volumeScaling);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+
+  const uint32_t startPointMs(0);
+  const uint32_t stopPointMs(0);
+
+  if (channel == -1) {
+    int res = _shared->transmit_mixer()->StartPlayingFileAsMicrophone(
+        stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
+    if (res) {
+      WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "StartPlayingFileAsMicrophone() failed to start "
+                   "playing stream");
+      return (-1);
+    } else {
+      _shared->transmit_mixer()->SetMixWithMicStatus(mixWithMicrophone);
+      return (0);
+    }
+  } else {
+    // Add file after demultiplexing <=> affects one channel only
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(
+          VE_CHANNEL_NOT_VALID, kTraceError,
+          "StartPlayingFileAsMicrophone() failed to locate channel");
+      return -1;
     }
 
-    const uint32_t startPointMs(0);
-    const uint32_t stopPointMs(0);
-
-    if (channel == -1)
-    {
-        int res = _shared->transmit_mixer()->StartPlayingFileAsMicrophone(
-            stream,
-            format,
-            startPointMs,
-            volumeScaling,
-            stopPointMs,
-            NULL);
-        if (res)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartPlayingFileAsMicrophone() failed to start "
-                "playing stream");
-            return(-1);
-        }
-        else
-        {
-            _shared->transmit_mixer()->SetMixWithMicStatus(mixWithMicrophone);
-            return(0);
-        }
-    }
-    else
-    {
-        // Add file after demultiplexing <=> affects one channel only
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "StartPlayingFileAsMicrophone() failed to locate channel");
-            return -1;
-        }
-
-        int res = channelPtr->StartPlayingFileAsMicrophone(
-            stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
-        if (res)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartPlayingFileAsMicrophone() failed to start "
-                "playing stream");
-            return -1;
-        }
-        else
-        {
-            channelPtr->SetMixWithMicStatus(mixWithMicrophone);
-            return 0;
-        }
+    int res = channelPtr->StartPlayingFileAsMicrophone(
+        stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
+    if (res) {
+      WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "StartPlayingFileAsMicrophone() failed to start "
+                   "playing stream");
+      return -1;
+    } else {
+      channelPtr->SetMixWithMicStatus(mixWithMicrophone);
+      return 0;
     }
+  }
 }
 
-int VoEFileImpl::StopPlayingFileAsMicrophone(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopPlayingFileAsMicrophone(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        // Stop adding file before demultiplexing <=> affects all channels
-        return _shared->transmit_mixer()->StopPlayingFileAsMicrophone();
-    }
-    else
-    {
-        // Stop adding file after demultiplexing <=> affects one channel only
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "StopPlayingFileAsMicrophone() failed to locate channel");
-            return -1;
-        }
-        return channelPtr->StopPlayingFileAsMicrophone();
+int VoEFileImpl::StopPlayingFileAsMicrophone(int channel) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StopPlayingFileAsMicrophone(channel=%d)", channel);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    // Stop adding file before demultiplexing <=> affects all channels
+    return _shared->transmit_mixer()->StopPlayingFileAsMicrophone();
+  } else {
+    // Stop adding file after demultiplexing <=> affects one channel only
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(
+          VE_CHANNEL_NOT_VALID, kTraceError,
+          "StopPlayingFileAsMicrophone() failed to locate channel");
+      return -1;
     }
+    return channelPtr->StopPlayingFileAsMicrophone();
+  }
 }
 
-int VoEFileImpl::IsPlayingFileAsMicrophone(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "IsPlayingFileAsMicrophone(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        return _shared->transmit_mixer()->IsPlayingFileAsMicrophone();
-    }
-    else
-    {
-        // Stop adding file after demultiplexing <=> affects one channel only
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "IsPlayingFileAsMicrophone() failed to locate channel");
-            return -1;
-        }
-        return channelPtr->IsPlayingFileAsMicrophone();
+int VoEFileImpl::IsPlayingFileAsMicrophone(int channel) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "IsPlayingFileAsMicrophone(channel=%d)", channel);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    return _shared->transmit_mixer()->IsPlayingFileAsMicrophone();
+  } else {
+    // Stop adding file after demultiplexing <=> affects one channel only
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(
+          VE_CHANNEL_NOT_VALID, kTraceError,
+          "IsPlayingFileAsMicrophone() failed to locate channel");
+      return -1;
     }
+    return channelPtr->IsPlayingFileAsMicrophone();
+  }
 }
 
-int VoEFileImpl::StartRecordingPlayout(
-    int channel, const char* fileNameUTF8, CodecInst* compression,
-    int maxSizeBytes)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartRecordingPlayout(channel=%d, fileNameUTF8=%s, "
-                 "compression, maxSizeBytes=%d)",
-                 channel, fileNameUTF8, maxSizeBytes);
-    assert(1024 == FileWrapper::kMaxFileNameSize);
+int VoEFileImpl::StartRecordingPlayout(int channel,
+                                       const char* fileNameUTF8,
+                                       CodecInst* compression,
+                                       int maxSizeBytes) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartRecordingPlayout(channel=%d, fileNameUTF8=%s, "
+               "compression, maxSizeBytes=%d)",
+               channel, fileNameUTF8, maxSizeBytes);
+  static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        return _shared->output_mixer()->StartRecordingPlayout
-          (fileNameUTF8, compression);
-    }
-    else
-    {
-        // Add file after demultiplexing <=> affects one channel only
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "StartRecordingPlayout() failed to locate channel");
-            return -1;
-        }
-        return channelPtr->StartRecordingPlayout(fileNameUTF8, compression);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    return _shared->output_mixer()->StartRecordingPlayout(fileNameUTF8,
+                                                          compression);
+  } else {
+    // Add file after demultiplexing <=> affects one channel only
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                            "StartRecordingPlayout() failed to locate channel");
+      return -1;
     }
+    return channelPtr->StartRecordingPlayout(fileNameUTF8, compression);
+  }
 }
 
-int VoEFileImpl::StartRecordingPlayout(
-    int channel, OutStream* stream, CodecInst* compression)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartRecordingPlayout(channel=%d, stream, compression)",
-                 channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        return _shared->output_mixer()->
-            StartRecordingPlayout(stream, compression);
-    }
-    else
-    {
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "StartRecordingPlayout() failed to locate channel");
-            return -1;
-        }
-        return channelPtr->StartRecordingPlayout(stream, compression);
+int VoEFileImpl::StartRecordingPlayout(int channel,
+                                       OutStream* stream,
+                                       CodecInst* compression) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartRecordingPlayout(channel=%d, stream, compression)",
+               channel);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    return _shared->output_mixer()->StartRecordingPlayout(stream, compression);
+  } else {
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                            "StartRecordingPlayout() failed to locate channel");
+      return -1;
     }
+    return channelPtr->StartRecordingPlayout(stream, compression);
+  }
 }
 
-int VoEFileImpl::StopRecordingPlayout(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopRecordingPlayout(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        return _shared->output_mixer()->StopRecordingPlayout();
-    }
-    else
-    {
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "StopRecordingPlayout() failed to locate channel");
-            return -1;
-        }
-        return channelPtr->StopRecordingPlayout();
+int VoEFileImpl::StopRecordingPlayout(int channel) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StopRecordingPlayout(channel=%d)", channel);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    return _shared->output_mixer()->StopRecordingPlayout();
+  } else {
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                            "StopRecordingPlayout() failed to locate channel");
+      return -1;
     }
+    return channelPtr->StopRecordingPlayout();
+  }
 }
 
-int VoEFileImpl::StartRecordingMicrophone(
-    const char* fileNameUTF8, CodecInst* compression, int maxSizeBytes)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartRecordingMicrophone(fileNameUTF8=%s, compression, "
-                 "maxSizeBytes=%d)", fileNameUTF8, maxSizeBytes);
-    assert(1024 == FileWrapper::kMaxFileNameSize);
+int VoEFileImpl::StartRecordingMicrophone(const char* fileNameUTF8,
+                                          CodecInst* compression,
+                                          int maxSizeBytes) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartRecordingMicrophone(fileNameUTF8=%s, compression, "
+               "maxSizeBytes=%d)",
+               fileNameUTF8, maxSizeBytes);
+  static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (_shared->transmit_mixer()->StartRecordingMicrophone(fileNameUTF8,
+                                                          compression)) {
+    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                 "StartRecordingMicrophone() failed to start recording");
+    return -1;
+  }
+  if (!_shared->audio_device()->Recording()) {
+    if (_shared->audio_device()->InitRecording() != 0) {
+      WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "StartRecordingMicrophone() failed to initialize recording");
+      return -1;
     }
-    if (_shared->transmit_mixer()->StartRecordingMicrophone(fileNameUTF8,
-                                                          compression))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVoice,
-            VoEId(_shared->instance_id(), -1),
-            "StartRecordingMicrophone() failed to start recording");
-        return -1;
+    if (_shared->audio_device()->StartRecording() != 0) {
+      WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "StartRecordingMicrophone() failed to start recording");
+      return -1;
     }
-    if (_shared->audio_device()->Recording())
-    {
-        return 0;
-    }
-    if (!_shared->ext_recording())
-    {
-        if (_shared->audio_device()->InitRecording() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartRecordingMicrophone() failed to initialize recording");
-            return -1;
-        }
-        if (_shared->audio_device()->StartRecording() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartRecordingMicrophone() failed to start recording");
-            return -1;
-        }
-    }
-    return 0;
+  }
+  return 0;
 }
 
-int VoEFileImpl::StartRecordingMicrophone(
-    OutStream* stream, CodecInst* compression)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartRecordingMicrophone(stream, compression)");
+int VoEFileImpl::StartRecordingMicrophone(OutStream* stream,
+                                          CodecInst* compression) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StartRecordingMicrophone(stream, compression)");
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (_shared->transmit_mixer()->StartRecordingMicrophone(stream,
+                                                          compression) == -1) {
+    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                 "StartRecordingMicrophone() failed to start recording");
+    return -1;
+  }
+  if (!_shared->audio_device()->Recording()) {
+    if (_shared->audio_device()->InitRecording() != 0) {
+      WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "StartRecordingMicrophone() failed to initialize recording");
+      return -1;
     }
-    if (_shared->transmit_mixer()->StartRecordingMicrophone(stream,
-                                                          compression) == -1)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVoice,
-            VoEId(_shared->instance_id(), -1),
-            "StartRecordingMicrophone() failed to start recording");
-        return -1;
+    if (_shared->audio_device()->StartRecording() != 0) {
+      WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "StartRecordingMicrophone() failed to start recording");
+      return -1;
     }
-    if (_shared->audio_device()->Recording())
-    {
-        return 0;
-    }
-    if (!_shared->ext_recording())
-    {
-        if (_shared->audio_device()->InitRecording() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartRecordingMicrophone() failed to initialize recording");
-            return -1;
-        }
-        if (_shared->audio_device()->StartRecording() != 0)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StartRecordingMicrophone() failed to start recording");
-            return -1;
-        }
-    }
-    return 0;
+  }
+  return 0;
 }
 
-int VoEFileImpl::StopRecordingMicrophone()
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopRecordingMicrophone()");
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+int VoEFileImpl::StopRecordingMicrophone() {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "StopRecordingMicrophone()");
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+
+  int err = 0;
+
+  // TODO(xians): consider removing Start/StopRecording() in
+  // Start/StopRecordingMicrophone() if no channel is recording.
+  if (_shared->NumOfSendingChannels() == 0 &&
+      _shared->audio_device()->Recording()) {
+    // Stop audio-device recording if no channel is recording
+    if (_shared->audio_device()->StopRecording() != 0) {
+      _shared->SetLastError(
+          VE_CANNOT_STOP_RECORDING, kTraceError,
+          "StopRecordingMicrophone() failed to stop recording");
+      err = -1;
     }
+  }
 
-    int err = 0;
+  if (_shared->transmit_mixer()->StopRecordingMicrophone() != 0) {
+    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                 "StopRecordingMicrophone() failed to stop recording to mixer");
+    err = -1;
+  }
 
-    // TODO(xians): consider removing Start/StopRecording() in
-    // Start/StopRecordingMicrophone() if no channel is recording.
-    if (_shared->NumOfSendingChannels() == 0 &&
-        _shared->audio_device()->Recording())
-    {
-        // Stop audio-device recording if no channel is recording
-        if (_shared->audio_device()->StopRecording() != 0)
-        {
-            _shared->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError,
-                "StopRecordingMicrophone() failed to stop recording");
-            err = -1;
-        }
-    }
-
-    if (_shared->transmit_mixer()->StopRecordingMicrophone() != 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "StopRecordingMicrophone() failed to stop recording to mixer");
-        err = -1;
-    }
-
-    return err;
+  return err;
 }
 
 #endif  // #ifdef WEBRTC_VOICE_ENGINE_FILE_API
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.h
index 584d0a17c7..5d28947ed7 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_file_impl.h
@@ -16,82 +16,78 @@
 
 namespace webrtc {
 
-class VoEFileImpl : public VoEFile
-{
-public:
-    // Playout file locally
+class VoEFileImpl : public VoEFile {
+ public:
+  // Playout file locally
 
-    virtual int StartPlayingFileLocally(
-        int channel,
-        const char fileNameUTF8[1024],
-        bool loop = false,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0,
-        int startPointMs = 0,
-        int stopPointMs = 0);
+  int StartPlayingFileLocally(int channel,
+                              const char fileNameUTF8[1024],
+                              bool loop = false,
+                              FileFormats format = kFileFormatPcm16kHzFile,
+                              float volumeScaling = 1.0,
+                              int startPointMs = 0,
+                              int stopPointMs = 0) override;
 
-    virtual int StartPlayingFileLocally(
-        int channel,
-        InStream* stream,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0,
-        int startPointMs = 0, int stopPointMs = 0);
+  int StartPlayingFileLocally(int channel,
+                              InStream* stream,
+                              FileFormats format = kFileFormatPcm16kHzFile,
+                              float volumeScaling = 1.0,
+                              int startPointMs = 0,
+                              int stopPointMs = 0) override;
 
-    virtual int StopPlayingFileLocally(int channel);
+  int StopPlayingFileLocally(int channel) override;
 
-    virtual int IsPlayingFileLocally(int channel);
+  int IsPlayingFileLocally(int channel) override;
 
-    // Use file as microphone input
+  // Use file as microphone input
 
-    virtual int StartPlayingFileAsMicrophone(
-        int channel,
-        const char fileNameUTF8[1024],
-        bool loop = false ,
-        bool mixWithMicrophone = false,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0);
+  int StartPlayingFileAsMicrophone(int channel,
+                                   const char fileNameUTF8[1024],
+                                   bool loop = false,
+                                   bool mixWithMicrophone = false,
+                                   FileFormats format = kFileFormatPcm16kHzFile,
+                                   float volumeScaling = 1.0) override;
 
-    virtual int StartPlayingFileAsMicrophone(
-        int channel,
-        InStream* stream,
-        bool mixWithMicrophone = false,
-        FileFormats format = kFileFormatPcm16kHzFile,
-        float volumeScaling = 1.0);
+  int StartPlayingFileAsMicrophone(int channel,
+                                   InStream* stream,
+                                   bool mixWithMicrophone = false,
+                                   FileFormats format = kFileFormatPcm16kHzFile,
+                                   float volumeScaling = 1.0) override;
 
-    virtual int StopPlayingFileAsMicrophone(int channel);
+  int StopPlayingFileAsMicrophone(int channel) override;
 
-    virtual int IsPlayingFileAsMicrophone(int channel);
+  int IsPlayingFileAsMicrophone(int channel) override;
 
-    // Record speaker signal to file
+  // Record speaker signal to file
 
-    virtual int StartRecordingPlayout(int channel,
-                                      const char* fileNameUTF8,
-                                      CodecInst* compression = NULL,
-                                      int maxSizeBytes = -1);
+  int StartRecordingPlayout(int channel,
+                            const char* fileNameUTF8,
+                            CodecInst* compression = NULL,
+                            int maxSizeBytes = -1) override;
 
-    virtual int StartRecordingPlayout(int channel,
-                                      OutStream* stream,
-                                      CodecInst* compression = NULL);
+  int StartRecordingPlayout(int channel,
+                            OutStream* stream,
+                            CodecInst* compression = NULL) override;
 
-    virtual int StopRecordingPlayout(int channel);
+  int StopRecordingPlayout(int channel) override;
 
-    // Record microphone signal to file
+  // Record microphone signal to file
 
-    virtual int StartRecordingMicrophone(const char* fileNameUTF8,
-                                         CodecInst* compression = NULL,
-                                         int maxSizeBytes = -1);
+  int StartRecordingMicrophone(const char* fileNameUTF8,
+                               CodecInst* compression = NULL,
+                               int maxSizeBytes = -1) override;
 
-    virtual int StartRecordingMicrophone(OutStream* stream,
-                                         CodecInst* compression = NULL);
+  int StartRecordingMicrophone(OutStream* stream,
+                               CodecInst* compression = NULL) override;
 
-    virtual int StopRecordingMicrophone();
+  int StopRecordingMicrophone() override;
 
-protected:
-    VoEFileImpl(voe::SharedData* shared);
-    virtual ~VoEFileImpl();
+ protected:
+  VoEFileImpl(voe::SharedData* shared);
+  ~VoEFileImpl() override;
 
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc
index 4505f9ee86..25b5e18cc5 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc
@@ -12,526 +12,432 @@
 
 #include <assert.h>
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-VoEHardware* VoEHardware::GetInterface(VoiceEngine* voiceEngine)
-{
+VoEHardware* VoEHardware::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_HARDWARE_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
 
-VoEHardwareImpl::VoEHardwareImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEHardwareImpl() - ctor");
+VoEHardwareImpl::VoEHardwareImpl(voe::SharedData* shared) : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEHardwareImpl() - ctor");
 }
 
-VoEHardwareImpl::~VoEHardwareImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "~VoEHardwareImpl() - dtor");
+VoEHardwareImpl::~VoEHardwareImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "~VoEHardwareImpl() - dtor");
 }
 
-int VoEHardwareImpl::SetAudioDeviceLayer(AudioLayers audioLayer)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetAudioDeviceLayer(audioLayer=%d)", audioLayer);
+int VoEHardwareImpl::SetAudioDeviceLayer(AudioLayers audioLayer) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetAudioDeviceLayer(audioLayer=%d)", audioLayer);
 
-    // Don't allow a change if VoE is initialized
-    if (_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_ALREADY_INITED, kTraceError);
-        return -1;
-    }
+  // Don't allow a change if VoE is initialized
+  if (_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_ALREADY_INITED, kTraceError);
+    return -1;
+  }
 
-    // Map to AudioDeviceModule::AudioLayer
-    AudioDeviceModule::AudioLayer
-        wantedLayer(AudioDeviceModule::kPlatformDefaultAudio);
-    switch (audioLayer)
-    {
-        case kAudioPlatformDefault:
-            // already set above
-            break;
-        case kAudioWindowsCore:
-            wantedLayer = AudioDeviceModule::kWindowsCoreAudio;
-            break;
-        case kAudioWindowsWave:
-            wantedLayer = AudioDeviceModule::kWindowsWaveAudio;
-            break;
-        case kAudioLinuxAlsa:
-            wantedLayer = AudioDeviceModule::kLinuxAlsaAudio;
-            break;
-        case kAudioLinuxPulse:
-            wantedLayer = AudioDeviceModule::kLinuxPulseAudio;
-            break;
-        case kAudioSndio:
-            wantedLayer = AudioDeviceModule::kSndioAudio;
-            break;
-    }
+  // Map to AudioDeviceModule::AudioLayer
+  AudioDeviceModule::AudioLayer wantedLayer(
+      AudioDeviceModule::kPlatformDefaultAudio);
+  switch (audioLayer) {
+    case kAudioPlatformDefault:
+      // already set above
+      break;
+    case kAudioWindowsCore:
+      wantedLayer = AudioDeviceModule::kWindowsCoreAudio;
+      break;
+    case kAudioWindowsWave:
+      wantedLayer = AudioDeviceModule::kWindowsWaveAudio;
+      break;
+    case kAudioLinuxAlsa:
+      wantedLayer = AudioDeviceModule::kLinuxAlsaAudio;
+      break;
+    case kAudioLinuxPulse:
+      wantedLayer = AudioDeviceModule::kLinuxPulseAudio;
+      break;
+    case kAudioSndio:
+      wantedLayer = AudioDeviceModule::kSndioAudio;
+      break;
+  }
 
-    // Save the audio device layer for Init()
-    _shared->set_audio_device_layer(wantedLayer);
+  // Save the audio device layer for Init()
+  _shared->set_audio_device_layer(wantedLayer);
 
-    return 0;
+  return 0;
 }
 
-int VoEHardwareImpl::GetAudioDeviceLayer(AudioLayers& audioLayer)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetAudioDeviceLayer(devices=?)");
+int VoEHardwareImpl::GetAudioDeviceLayer(AudioLayers& audioLayer) {
+  // Can always be called regardless of VoE state
 
-    // Can always be called regardless of VoE state
+  AudioDeviceModule::AudioLayer activeLayer(
+      AudioDeviceModule::kPlatformDefaultAudio);
 
-    AudioDeviceModule::AudioLayer
-        activeLayer(AudioDeviceModule::kPlatformDefaultAudio);
-
-    if (_shared->audio_device())
-    {
-        // Get active audio layer from ADM
-        if (_shared->audio_device()->ActiveAudioLayer(&activeLayer) != 0)
-        {
-            _shared->SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
-                "  Audio Device error");
-            return -1;
-        }
-    }
-    else
-    {
-        // Return VoE's internal layer setting
-        activeLayer = _shared->audio_device_layer();
+  if (_shared->audio_device()) {
+    // Get active audio layer from ADM
+    if (_shared->audio_device()->ActiveAudioLayer(&activeLayer) != 0) {
+      _shared->SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
+                            "  Audio Device error");
+      return -1;
     }
+  } else {
+    // Return VoE's internal layer setting
+    activeLayer = _shared->audio_device_layer();
+  }
 
-    // Map to AudioLayers
-    switch (activeLayer)
-    {
-        case AudioDeviceModule::kPlatformDefaultAudio:
-            audioLayer = kAudioPlatformDefault;
-            break;
-        case AudioDeviceModule::kWindowsCoreAudio:
-            audioLayer = kAudioWindowsCore;
-            break;
-        case AudioDeviceModule::kWindowsWaveAudio:
-            audioLayer = kAudioWindowsWave;
-            break;
-        case AudioDeviceModule::kLinuxAlsaAudio:
-            audioLayer = kAudioLinuxAlsa;
-            break;
-        case AudioDeviceModule::kLinuxPulseAudio:
-            audioLayer = kAudioLinuxPulse;
-            break;
-        case AudioDeviceModule::kSndioAudio:
-            audioLayer = kAudioSndio;
-            break;
-        default:
-            _shared->SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
-                "  unknown audio layer");
-    }
+  // Map to AudioLayers
+  switch (activeLayer) {
+    case AudioDeviceModule::kPlatformDefaultAudio:
+      audioLayer = kAudioPlatformDefault;
+      break;
+    case AudioDeviceModule::kWindowsCoreAudio:
+      audioLayer = kAudioWindowsCore;
+      break;
+    case AudioDeviceModule::kWindowsWaveAudio:
+      audioLayer = kAudioWindowsWave;
+      break;
+    case AudioDeviceModule::kLinuxAlsaAudio:
+      audioLayer = kAudioLinuxAlsa;
+      break;
+    case AudioDeviceModule::kLinuxPulseAudio:
+      audioLayer = kAudioLinuxPulse;
+      break;
+    case AudioDeviceModule::kSndioAudio:
+      audioLayer = kAudioSndio;
+      break;
+    default:
+      _shared->SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
+                            "  unknown audio layer");
+  }
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "  Output: audioLayer=%d", audioLayer);
-
-    return 0;
+  return 0;
 }
-int VoEHardwareImpl::GetNumOfRecordingDevices(int& devices)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetNumOfRecordingDevices(devices=?)");
+int VoEHardwareImpl::GetNumOfRecordingDevices(int& devices) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  devices = static_cast<int>(_shared->audio_device()->RecordingDevices());
 
-    devices = static_cast<int> (_shared->audio_device()->RecordingDevices());
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1), "  Output: devices=%d", devices);
-
-    return 0;
+  return 0;
 }
 
-int VoEHardwareImpl::GetNumOfPlayoutDevices(int& devices)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetNumOfPlayoutDevices(devices=?)");
+int VoEHardwareImpl::GetNumOfPlayoutDevices(int& devices) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  devices = static_cast<int>(_shared->audio_device()->PlayoutDevices());
 
-    devices = static_cast<int> (_shared->audio_device()->PlayoutDevices());
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "  Output: devices=%d", devices);
-
-    return 0;
+  return 0;
 }
 
 int VoEHardwareImpl::GetRecordingDeviceName(int index,
                                             char strNameUTF8[128],
-                                            char strGuidUTF8[128])
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRecordingDeviceName(index=%d)", index);
+                                            char strGuidUTF8[128]) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (strNameUTF8 == NULL) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "GetRecordingDeviceName() invalid argument");
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (strNameUTF8 == NULL)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "GetRecordingDeviceName() invalid argument");
-        return -1;
-    }
+  // Note that strGuidUTF8 is allowed to be NULL
 
-    // Note that strGuidUTF8 is allowed to be NULL
+  // Init len variable to length of supplied vectors
+  const uint16_t strLen = 128;
 
-    // Init len variable to length of supplied vectors
-    const uint16_t strLen = 128;
+  // Check if length has been changed in module
+  static_assert(strLen == kAdmMaxDeviceNameSize, "");
+  static_assert(strLen == kAdmMaxGuidSize, "");
 
-    // Check if length has been changed in module
-    assert(strLen == kAdmMaxDeviceNameSize);
-    assert(strLen == kAdmMaxGuidSize);
+  char name[strLen];
+  char guid[strLen];
 
-    char name[strLen];
-    char guid[strLen];
+  // Get names from module
+  if (_shared->audio_device()->RecordingDeviceName(index, name, guid) != 0) {
+    _shared->SetLastError(VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
+                          "GetRecordingDeviceName() failed to get device name");
+    return -1;
+  }
 
-    // Get names from module
-    if (_shared->audio_device()->RecordingDeviceName(index, name, guid) != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
-            "GetRecordingDeviceName() failed to get device name");
-        return -1;
-    }
+  // Copy to vectors supplied by user
+  strncpy(strNameUTF8, name, strLen);
 
-    // Copy to vectors supplied by user
-    strncpy(strNameUTF8, name, strLen);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "  Output: strNameUTF8=%s", strNameUTF8);
+  if (strGuidUTF8 != NULL) {
+    strncpy(strGuidUTF8, guid, strLen);
+  }
 
-    if (strGuidUTF8 != NULL)
-    {
-        strncpy(strGuidUTF8, guid, strLen);
-        WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-            VoEId(_shared->instance_id(), -1),
-            "  Output: strGuidUTF8=%s", strGuidUTF8);
-    }
-
-    return 0;
+  return 0;
 }
 
 int VoEHardwareImpl::GetPlayoutDeviceName(int index,
                                           char strNameUTF8[128],
-                                          char strGuidUTF8[128])
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetPlayoutDeviceName(index=%d)", index);
+                                          char strGuidUTF8[128]) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (strNameUTF8 == NULL) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "GetPlayoutDeviceName() invalid argument");
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (strNameUTF8 == NULL)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "GetPlayoutDeviceName() invalid argument");
-        return -1;
-    }
+  // Note that strGuidUTF8 is allowed to be NULL
 
-    // Note that strGuidUTF8 is allowed to be NULL
+  // Init len variable to length of supplied vectors
+  const uint16_t strLen = 128;
 
-    // Init len variable to length of supplied vectors
-    const uint16_t strLen = 128;
+  // Check if length has been changed in module
+  static_assert(strLen == kAdmMaxDeviceNameSize, "");
+  static_assert(strLen == kAdmMaxGuidSize, "");
 
-    // Check if length has been changed in module
-    assert(strLen == kAdmMaxDeviceNameSize);
-    assert(strLen == kAdmMaxGuidSize);
+  char name[strLen];
+  char guid[strLen];
 
-    char name[strLen];
-    char guid[strLen];
+  // Get names from module
+  if (_shared->audio_device()->PlayoutDeviceName(index, name, guid) != 0) {
+    _shared->SetLastError(VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
+                          "GetPlayoutDeviceName() failed to get device name");
+    return -1;
+  }
 
-    // Get names from module
-    if (_shared->audio_device()->PlayoutDeviceName(index, name, guid) != 0)
-    {
-        _shared->SetLastError(VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
-            "GetPlayoutDeviceName() failed to get device name");
-        return -1;
-    }
+  // Copy to vectors supplied by user
+  strncpy(strNameUTF8, name, strLen);
 
-    // Copy to vectors supplied by user
-    strncpy(strNameUTF8, name, strLen);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "  Output: strNameUTF8=%s", strNameUTF8);
+  if (strGuidUTF8 != NULL) {
+    strncpy(strGuidUTF8, guid, strLen);
+  }
 
-    if (strGuidUTF8 != NULL)
-    {
-        strncpy(strGuidUTF8, guid, strLen);
-        WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-            VoEId(_shared->instance_id(), -1),
-            "  Output: strGuidUTF8=%s", strGuidUTF8);
-    }
-
-    return 0;
+  return 0;
 }
 
 int VoEHardwareImpl::SetRecordingDevice(int index,
-                                        StereoChannel recordingChannel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetRecordingDevice(index=%d, recordingChannel=%d)",
-                 index, (int) recordingChannel);
-    CriticalSectionScoped cs(_shared->crit_sec());
+                                        StereoChannel recordingChannel) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetRecordingDevice(index=%d, recordingChannel=%d)", index,
+               (int)recordingChannel);
+  CriticalSectionScoped cs(_shared->crit_sec());
 
-    if (!_shared->statistics().Initialized())
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+
+  bool isRecording(false);
+
+  // Store state about activated recording to be able to restore it after the
+  // recording device has been modified.
+  if (_shared->audio_device()->Recording()) {
+    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                 "SetRecordingDevice() device is modified while recording"
+                 " is active...");
+    isRecording = true;
+    if (_shared->audio_device()->StopRecording() == -1) {
+      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                            "SetRecordingDevice() unable to stop recording");
+      return -1;
+    }
+  }
+
+  // We let the module do the index sanity
+
+  // Set recording channel
+  AudioDeviceModule::ChannelType recCh = AudioDeviceModule::kChannelBoth;
+  switch (recordingChannel) {
+    case kStereoLeft:
+      recCh = AudioDeviceModule::kChannelLeft;
+      break;
+    case kStereoRight:
+      recCh = AudioDeviceModule::kChannelRight;
+      break;
+    case kStereoBoth:
+      // default setting kChannelBoth (<=> mono)
+      break;
+  }
+
+  if (_shared->audio_device()->SetRecordingChannel(recCh) != 0) {
+    _shared->SetLastError(
+        VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
+        "SetRecordingChannel() unable to set the recording channel");
+  }
+
+  // Map indices to unsigned since underlying functions need that
+  uint16_t indexU = static_cast<uint16_t>(index);
+
+  int32_t res(0);
+
+  if (index == -1) {
+    res = _shared->audio_device()->SetRecordingDevice(
+        AudioDeviceModule::kDefaultCommunicationDevice);
+  } else if (index == -2) {
+    res = _shared->audio_device()->SetRecordingDevice(
+        AudioDeviceModule::kDefaultDevice);
+  } else {
+    res = _shared->audio_device()->SetRecordingDevice(indexU);
+  }
+
+  if (res != 0) {
+    _shared->SetLastError(
+        VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+        "SetRecordingDevice() unable to set the recording device");
+    return -1;
+  }
+
+  // Init microphone, so user can do volume settings etc
+  if (_shared->audio_device()->InitMicrophone() == -1) {
+    _shared->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceWarning,
+                          "SetRecordingDevice() cannot access microphone");
+  }
+
+  // Set number of channels
+  bool available = false;
+  if (_shared->audio_device()->StereoRecordingIsAvailable(&available) != 0) {
+    _shared->SetLastError(
+        VE_SOUNDCARD_ERROR, kTraceWarning,
+        "StereoRecordingIsAvailable() failed to query stereo recording");
+  }
+
+  if (_shared->audio_device()->SetStereoRecording(available) != 0) {
+    _shared->SetLastError(
+        VE_SOUNDCARD_ERROR, kTraceWarning,
+        "SetRecordingDevice() failed to set mono recording mode");
+  }
+
+  // Restore recording if it was enabled already when calling this function.
+  if (isRecording) {
+    if (!_shared->ext_recording())
     {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
+      WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "SetRecordingDevice() recording is now being restored...");
+      if (_shared->audio_device()->InitRecording() != 0) {
+        WEBRTC_TRACE(kTraceError, kTraceVoice,
+                     VoEId(_shared->instance_id(), -1),
+                     "SetRecordingDevice() failed to initialize recording");
         return -1;
-    }
-
-    bool isRecording(false);
-
-    // Store state about activated recording to be able to restore it after the
-    // recording device has been modified.
-    if (_shared->audio_device()->Recording())
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                     "SetRecordingDevice() device is modified while recording"
-                     " is active...");
-        isRecording = true;
-        if (_shared->audio_device()->StopRecording() == -1)
-        {
-            _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-                "SetRecordingDevice() unable to stop recording");
-            return -1;
-        }
-    }
-
-    // We let the module do the index sanity
-
-    // Set recording channel
-    AudioDeviceModule::ChannelType recCh =
-        AudioDeviceModule::kChannelBoth;
-    switch (recordingChannel)
-    {
-        case kStereoLeft:
-            recCh = AudioDeviceModule::kChannelLeft;
-            break;
-        case kStereoRight:
-            recCh = AudioDeviceModule::kChannelRight;
-            break;
-        case kStereoBoth:
-            // default setting kChannelBoth (<=> mono)
-            break;
-    }
-
-    if (_shared->audio_device()->SetRecordingChannel(recCh) != 0) {
-      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
-          "SetRecordingChannel() unable to set the recording channel");
-    }
-
-    // Map indices to unsigned since underlying functions need that
-    uint16_t indexU = static_cast<uint16_t> (index);
-
-    int32_t res(0);
-
-    if (index == -1)
-    {
-        res = _shared->audio_device()->SetRecordingDevice(
-            AudioDeviceModule::kDefaultCommunicationDevice);
-    }
-    else if (index == -2)
-    {
-        res = _shared->audio_device()->SetRecordingDevice(
-            AudioDeviceModule::kDefaultDevice);
-    }
-    else
-    {
-        res = _shared->audio_device()->SetRecordingDevice(indexU);
-    }
-
-    if (res != 0)
-    {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-            "SetRecordingDevice() unable to set the recording device");
+      }
+      if (_shared->audio_device()->StartRecording() != 0) {
+        WEBRTC_TRACE(kTraceError, kTraceVoice,
+                     VoEId(_shared->instance_id(), -1),
+                     "SetRecordingDevice() failed to start recording");
         return -1;
+      }
     }
+  }
 
-    // Init microphone, so user can do volume settings etc
-    if (_shared->audio_device()->InitMicrophone() == -1)
-    {
-        _shared->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceWarning,
-            "SetRecordingDevice() cannot access microphone");
-    }
-
-    // Set number of channels
-    bool available = false;
-    if (_shared->audio_device()->StereoRecordingIsAvailable(&available) != 0) {
-      _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-          "StereoRecordingIsAvailable() failed to query stereo recording");
-    }
-
-    if (_shared->audio_device()->SetStereoRecording(available) != 0)
-    {
-        _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-            "SetRecordingDevice() failed to set mono recording mode");
-    }
-
-    // Restore recording if it was enabled already when calling this function.
-    if (isRecording)
-    {
-        if (!_shared->ext_recording())
-        {
-            WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "SetRecordingDevice() recording is now being restored...");
-            if (_shared->audio_device()->InitRecording() != 0)
-            {
-                WEBRTC_TRACE(kTraceError, kTraceVoice,
-                    VoEId(_shared->instance_id(), -1),
-                    "SetRecordingDevice() failed to initialize recording");
-                return -1;
-            }
-            if (_shared->audio_device()->StartRecording() != 0)
-            {
-                WEBRTC_TRACE(kTraceError, kTraceVoice,
-                             VoEId(_shared->instance_id(), -1),
-                             "SetRecordingDevice() failed to start recording");
-                return -1;
-            }
-        }
-    }
-
-    return 0;
+  return 0;
 }
 
-int VoEHardwareImpl::SetPlayoutDevice(int index)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetPlayoutDevice(index=%d)", index);
-    CriticalSectionScoped cs(_shared->crit_sec());
+int VoEHardwareImpl::SetPlayoutDevice(int index) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetPlayoutDevice(index=%d)", index);
+  CriticalSectionScoped cs(_shared->crit_sec());
 
-    if (!_shared->statistics().Initialized())
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+
+  bool isPlaying(false);
+
+  // Store state about activated playout to be able to restore it after the
+  // playout device has been modified.
+  if (_shared->audio_device()->Playing()) {
+    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                 "SetPlayoutDevice() device is modified while playout is "
+                 "active...");
+    isPlaying = true;
+    if (_shared->audio_device()->StopPlayout() == -1) {
+      _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                            "SetPlayoutDevice() unable to stop playout");
+      return -1;
+    }
+  }
+
+  // We let the module do the index sanity
+
+  // Map indices to unsigned since underlying functions need that
+  uint16_t indexU = static_cast<uint16_t>(index);
+
+  int32_t res(0);
+
+  if (index == -1) {
+    res = _shared->audio_device()->SetPlayoutDevice(
+        AudioDeviceModule::kDefaultCommunicationDevice);
+  } else if (index == -2) {
+    res = _shared->audio_device()->SetPlayoutDevice(
+        AudioDeviceModule::kDefaultDevice);
+  } else {
+    res = _shared->audio_device()->SetPlayoutDevice(indexU);
+  }
+
+  if (res != 0) {
+    _shared->SetLastError(
+        VE_SOUNDCARD_ERROR, kTraceError,
+        "SetPlayoutDevice() unable to set the playout device");
+    return -1;
+  }
+
+  // Init speaker, so user can do volume settings etc
+  if (_shared->audio_device()->InitSpeaker() == -1) {
+    _shared->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceWarning,
+                          "SetPlayoutDevice() cannot access speaker");
+  }
+
+  // Set number of channels
+  bool available = false;
+  _shared->audio_device()->StereoPlayoutIsAvailable(&available);
+  if (_shared->audio_device()->SetStereoPlayout(available) != 0) {
+    _shared->SetLastError(
+        VE_SOUNDCARD_ERROR, kTraceWarning,
+        "SetPlayoutDevice() failed to set stereo playout mode");
+  }
+
+  // Restore playout if it was enabled already when calling this function.
+  if (isPlaying) {
+    if (!_shared->ext_playout())
     {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
+      WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                   "SetPlayoutDevice() playout is now being restored...");
+      if (_shared->audio_device()->InitPlayout() != 0) {
+        WEBRTC_TRACE(kTraceError, kTraceVoice,
+                     VoEId(_shared->instance_id(), -1),
+                     "SetPlayoutDevice() failed to initialize playout");
         return -1;
-    }
-
-    bool isPlaying(false);
-
-    // Store state about activated playout to be able to restore it after the
-    // playout device has been modified.
-    if (_shared->audio_device()->Playing())
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                     "SetPlayoutDevice() device is modified while playout is "
-                     "active...");
-        isPlaying = true;
-        if (_shared->audio_device()->StopPlayout() == -1)
-        {
-            _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-                "SetPlayoutDevice() unable to stop playout");
-            return -1;
-        }
-    }
-
-    // We let the module do the index sanity
-
-    // Map indices to unsigned since underlying functions need that
-    uint16_t indexU = static_cast<uint16_t> (index);
-
-    int32_t res(0);
-
-    if (index == -1)
-    {
-        res = _shared->audio_device()->SetPlayoutDevice(
-            AudioDeviceModule::kDefaultCommunicationDevice);
-    }
-    else if (index == -2)
-    {
-        res = _shared->audio_device()->SetPlayoutDevice(
-            AudioDeviceModule::kDefaultDevice);
-    }
-    else
-    {
-        res = _shared->audio_device()->SetPlayoutDevice(indexU);
-    }
-
-    if (res != 0)
-    {
-        _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
-            "SetPlayoutDevice() unable to set the playout device");
+      }
+      if (_shared->audio_device()->StartPlayout() != 0) {
+        WEBRTC_TRACE(kTraceError, kTraceVoice,
+                     VoEId(_shared->instance_id(), -1),
+                     "SetPlayoutDevice() failed to start playout");
         return -1;
+      }
     }
+  }
 
-    // Init speaker, so user can do volume settings etc
-    if (_shared->audio_device()->InitSpeaker() == -1)
-    {
-        _shared->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceWarning,
-            "SetPlayoutDevice() cannot access speaker");
-    }
-
-    // Set number of channels
-    bool available = false;
-    _shared->audio_device()->StereoPlayoutIsAvailable(&available);
-    if (_shared->audio_device()->SetStereoPlayout(available) != 0)
-    {
-        _shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
-            "SetPlayoutDevice() failed to set stereo playout mode");
-    }
-
-    // Restore playout if it was enabled already when calling this function.
-    if (isPlaying)
-    {
-        if (!_shared->ext_playout())
-        {
-            WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                VoEId(_shared->instance_id(), -1),
-                "SetPlayoutDevice() playout is now being restored...");
-            if (_shared->audio_device()->InitPlayout() != 0)
-            {
-                WEBRTC_TRACE(kTraceError, kTraceVoice,
-                  VoEId(_shared->instance_id(), -1),
-                  "SetPlayoutDevice() failed to initialize playout");
-                return -1;
-            }
-            if (_shared->audio_device()->StartPlayout() != 0)
-            {
-                WEBRTC_TRACE(kTraceError, kTraceVoice,
-                             VoEId(_shared->instance_id(), -1),
-                             "SetPlayoutDevice() failed to start playout");
-                return -1;
-            }
-        }
-    }
-
-    return 0;
+  return 0;
 }
 
 int VoEHardwareImpl::GetRecordingDeviceStatus(bool& isAvailable)
@@ -721,19 +627,6 @@ int VoEHardwareImpl::GetCPULoad(int& loadPercent)
     return 0;
 }
 
-int VoEHardwareImpl::EnableBuiltInAEC(bool enable)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-        "%s", __FUNCTION__);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-
-    return _shared->audio_device()->EnableBuiltInAEC(enable);
-}
-
 bool VoEHardwareImpl::BuiltInAECIsEnabled() const
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
@@ -758,8 +651,6 @@ int VoEHardwareImpl::SetRecordingSampleRate(unsigned int samples_per_sec) {
 }
 
 int VoEHardwareImpl::RecordingSampleRate(unsigned int* samples_per_sec) const {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "%s", __FUNCTION__);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return false;
@@ -778,8 +669,6 @@ int VoEHardwareImpl::SetPlayoutSampleRate(unsigned int samples_per_sec) {
 }
 
 int VoEHardwareImpl::PlayoutSampleRate(unsigned int* samples_per_sec) const {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "%s", __FUNCTION__);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return false;
@@ -788,13 +677,58 @@ int VoEHardwareImpl::PlayoutSampleRate(unsigned int* samples_per_sec) const {
 }
 
 bool VoEHardwareImpl::BuiltInAECIsAvailable() const {
-if (!_shared->statistics().Initialized()) {
+  if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return false;
   }
   return _shared->audio_device()->BuiltInAECIsAvailable();
 }
 
+int VoEHardwareImpl::EnableBuiltInAEC(bool enable)
+{
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "%s", __FUNCTION__);
+  if (!_shared->statistics().Initialized())
+    {
+      _shared->SetLastError(VE_NOT_INITED, kTraceError);
+      return -1;
+    }
+
+  return _shared->audio_device()->EnableBuiltInAEC(enable);
+}
+
+bool VoEHardwareImpl::BuiltInAGCIsAvailable() const {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return false;
+  }
+  return _shared->audio_device()->BuiltInAGCIsAvailable();
+}
+
+int VoEHardwareImpl::EnableBuiltInAGC(bool enable) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  return _shared->audio_device()->EnableBuiltInAGC(enable);
+}
+
+bool VoEHardwareImpl::BuiltInNSIsAvailable() const {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return false;
+  }
+  return _shared->audio_device()->BuiltInNSIsAvailable();
+}
+
+int VoEHardwareImpl::EnableBuiltInNS(bool enable) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  return _shared->audio_device()->EnableBuiltInNS(enable);
+}
+
 #endif  // WEBRTC_VOICE_ENGINE_HARDWARE_API
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h
index e23e4e5df4..5336d8262f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h
@@ -15,66 +15,67 @@
 
 #include "webrtc/voice_engine/shared_data.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-class VoEHardwareImpl: public VoEHardware
-{
-public:
-    virtual int GetNumOfRecordingDevices(int& devices);
+class VoEHardwareImpl : public VoEHardware {
+ public:
+  int GetNumOfRecordingDevices(int& devices) override;
 
-    virtual int GetNumOfPlayoutDevices(int& devices);
+  int GetNumOfPlayoutDevices(int& devices) override;
 
-    virtual int GetRecordingDeviceName(int index,
-                                       char strNameUTF8[128],
-                                       char strGuidUTF8[128]);
+  int GetRecordingDeviceName(int index,
+                             char strNameUTF8[128],
+                             char strGuidUTF8[128]) override;
  
-    virtual int GetRecordingDeviceStatus(bool& isAvailable);
+  int GetRecordingDeviceStatus(bool& isAvailable) override;
 
-    virtual int GetPlayoutDeviceStatus(bool& isAvailable);
+  int GetPlayoutDeviceStatus(bool& isAvailable) override;
 
-    virtual int GetPlayoutDeviceName(int index,
-                                     char strNameUTF8[128],
-                                     char strGuidUTF8[128]);
+  int GetPlayoutDeviceName(int index,
+                           char strNameUTF8[128],
+                           char strGuidUTF8[128]) override;
 
-    virtual int SetRecordingDevice(
-        int index,
-        StereoChannel recordingChannel = kStereoBoth);
+  int SetRecordingDevice(int index,
+                         StereoChannel recordingChannel = kStereoBoth) override;
 
-    virtual int SetPlayoutDevice(int index);
+  int SetPlayoutDevice(int index) override;
 
-    virtual int SetAudioDeviceLayer(AudioLayers audioLayer);
+  int SetAudioDeviceLayer(AudioLayers audioLayer) override;
  
-    virtual int GetCPULoad(int& loadPercent);
+  int GetCPULoad(int& loadPercent) override;
 
-    virtual int ResetAudioDevice();
+  int ResetAudioDevice() override;
 
-    virtual int AudioDeviceControl(unsigned int par1,
-                                   unsigned int par2,
-                                   unsigned int par3);
+  int AudioDeviceControl(unsigned int par1,
+                         unsigned int par2,
+                         unsigned int par3) override;
 
-    virtual int SetLoudspeakerStatus(bool enable);
+  int SetLoudspeakerStatus(bool enable) override;
 
-    virtual int GetLoudspeakerStatus(bool& enabled);
+  int GetLoudspeakerStatus(bool& enabled) override;
 
-    virtual int EnableBuiltInAEC(bool enable);
-    virtual bool BuiltInAECIsEnabled() const;
+  bool BuiltInAECIsEnabled() const override;
 
-    virtual int GetAudioDeviceLayer(AudioLayers& audioLayer);
+  int GetAudioDeviceLayer(AudioLayers& audioLayer) override;
 
-    virtual int SetRecordingSampleRate(unsigned int samples_per_sec);
-    virtual int RecordingSampleRate(unsigned int* samples_per_sec) const;
-    virtual int SetPlayoutSampleRate(unsigned int samples_per_sec);
-    virtual int PlayoutSampleRate(unsigned int* samples_per_sec) const;
+  int SetRecordingSampleRate(unsigned int samples_per_sec) override;
+  int RecordingSampleRate(unsigned int* samples_per_sec) const override;
+  int SetPlayoutSampleRate(unsigned int samples_per_sec) override;
+  int PlayoutSampleRate(unsigned int* samples_per_sec) const override;
 
-    virtual bool BuiltInAECIsAvailable() const;
+  bool BuiltInAECIsAvailable() const override;
+  int EnableBuiltInAEC(bool enable) override;
+  bool BuiltInAGCIsAvailable() const override;
+  int EnableBuiltInAGC(bool enable) override;
+  bool BuiltInNSIsAvailable() const override;
+  int EnableBuiltInNS(bool enable) override;
 
-protected:
-    VoEHardwareImpl(voe::SharedData* shared);
-    virtual ~VoEHardwareImpl();
+ protected:
+  VoEHardwareImpl(voe::SharedData* shared);
+  ~VoEHardwareImpl() override;
 
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.cc
index 0d8ce50eb8..807325b4f8 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.cc
@@ -10,70 +10,60 @@
 
 #include "webrtc/voice_engine/voe_neteq_stats_impl.h"
 
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
-VoENetEqStats* VoENetEqStats::GetInterface(VoiceEngine* voiceEngine)
-{
+VoENetEqStats* VoENetEqStats::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
 
-VoENetEqStatsImpl::VoENetEqStatsImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoENetEqStatsImpl::VoENetEqStatsImpl() - ctor");
+VoENetEqStatsImpl::VoENetEqStatsImpl(voe::SharedData* shared)
+    : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoENetEqStatsImpl::VoENetEqStatsImpl() - ctor");
 }
 
-VoENetEqStatsImpl::~VoENetEqStatsImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoENetEqStatsImpl::~VoENetEqStatsImpl() - dtor");
+VoENetEqStatsImpl::~VoENetEqStatsImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoENetEqStatsImpl::~VoENetEqStatsImpl() - dtor");
 }
 
 int VoENetEqStatsImpl::GetNetworkStatistics(int channel,
-                                            NetworkStatistics& stats)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetNetworkStatistics(channel=%d, stats=?)", channel);
+                                            NetworkStatistics& stats) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetNetworkStatistics() failed to locate channel");
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetNetworkStatistics() failed to locate channel");
-        return -1;
-    }
-
-    return channelPtr->GetNetworkStatistics(stats);
+  return channelPtr->GetNetworkStatistics(stats);
 }
 
 int VoENetEqStatsImpl::GetDecodingCallStatistics(
     int channel, AudioDecodingCallStats* stats) const {
-
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.h
index 74b624b19c..d441e8263f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_neteq_stats_impl.h
@@ -18,23 +18,21 @@
 
 namespace webrtc {
 
-class VoENetEqStatsImpl : public VoENetEqStats
-{
-public:
-    virtual int GetNetworkStatistics(int channel,
-                                     NetworkStatistics& stats);
+class VoENetEqStatsImpl : public VoENetEqStats {
+ public:
+  int GetNetworkStatistics(int channel, NetworkStatistics& stats) override;
 
-    virtual int GetDecodingCallStatistics(
-        int channel, AudioDecodingCallStats* stats) const;
+  int GetDecodingCallStatistics(int channel,
+                                AudioDecodingCallStats* stats) const override;
 
-protected:
-    VoENetEqStatsImpl(voe::SharedData* shared);
-    virtual ~VoENetEqStatsImpl();
+ protected:
+  VoENetEqStatsImpl(voe::SharedData* shared);
+  ~VoENetEqStatsImpl() override;
 
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
 
-#endif    // WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
+#endif  // WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc
index 89d1b04f48..0574aa9f05 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc
@@ -10,81 +10,52 @@
 
 #include "webrtc/voice_engine/voe_network_impl.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
-VoENetwork* VoENetwork::GetInterface(VoiceEngine* voiceEngine)
-{
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+VoENetwork* VoENetwork::GetInterface(VoiceEngine* voiceEngine) {
+  if (!voiceEngine) {
+    return nullptr;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 }
 
-VoENetworkImpl::VoENetworkImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoENetworkImpl() - ctor");
+VoENetworkImpl::VoENetworkImpl(voe::SharedData* shared) : _shared(shared) {
 }
 
-VoENetworkImpl::~VoENetworkImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "~VoENetworkImpl() - dtor");
-}
+VoENetworkImpl::~VoENetworkImpl() = default;
 
 int VoENetworkImpl::RegisterExternalTransport(int channel,
-                                              Transport& transport)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetExternalTransport(channel=%d, transport=0x%x)",
-                 channel, &transport);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetExternalTransport() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->RegisterExternalTransport(transport);
+                                              Transport& transport) {
+  RTC_DCHECK(_shared->statistics().Initialized());
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (!channelPtr) {
+    LOG_F(LS_ERROR) << "Failed to locate channel: " << channel;
+    return -1;
+  }
+  return channelPtr->RegisterExternalTransport(transport);
 }
 
-int VoENetworkImpl::DeRegisterExternalTransport(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "DeRegisterExternalTransport(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVoice,
-                     VoEId(_shared->instance_id(), -1),
-                     "DeRegisterExternalTransport() - invalid state");
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "DeRegisterExternalTransport() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->DeRegisterExternalTransport();
+int VoENetworkImpl::DeRegisterExternalTransport(int channel) {
+  RTC_CHECK(_shared->statistics().Initialized());
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (!channelPtr) {
+    LOG_F(LS_ERROR) << "Failed to locate channel: " << channel;
+    return -1;
+  }
+  return channelPtr->DeRegisterExternalTransport();
 }
 
 int VoENetworkImpl::ReceivedRTPPacket(int channel,
@@ -96,85 +67,48 @@ int VoENetworkImpl::ReceivedRTPPacket(int channel,
 int VoENetworkImpl::ReceivedRTPPacket(int channel,
                                       const void* data,
                                       size_t length,
-                                      const PacketTime& packet_time)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "ReceivedRTPPacket(channel=%d, length=%" PRIuS ")", channel,
-                 length);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    // L16 at 32 kHz, stereo, 10 ms frames (+12 byte RTP header) -> 1292 bytes
-    if ((length < 12) || (length > 1292))
-    {
-        _shared->SetLastError(VE_INVALID_PACKET);
-        LOG(LS_ERROR) << "Invalid packet length: " << length;
-        return -1;
-    }
-    if (NULL == data)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "ReceivedRTPPacket() invalid data vector");
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "ReceivedRTPPacket() failed to locate channel");
-        return -1;
-    }
-
-    if (!channelPtr->ExternalTransport())
-    {
-        _shared->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "ReceivedRTPPacket() external transport is not enabled");
-        return -1;
-    }
-    return channelPtr->ReceivedRTPPacket((const int8_t*) data, length,
-                                         packet_time);
+                                      const PacketTime& packet_time) {
+  RTC_CHECK(_shared->statistics().Initialized());
+  RTC_CHECK(data);
+  // L16 at 32 kHz, stereo, 10 ms frames (+12 byte RTP header) -> 1292 bytes
+  if ((length < 12) || (length > 1292)) {
+    LOG_F(LS_ERROR) << "Invalid packet length: " << length;
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (!channelPtr) {
+    LOG_F(LS_ERROR) << "Failed to locate channel: " << channel;
+    return -1;
+  }
+  if (!channelPtr->ExternalTransport()) {
+    LOG_F(LS_ERROR) << "No external transport for channel: " << channel;
+    return -1;
+  }
+  return channelPtr->ReceivedRTPPacket((const int8_t*)data, length,
+                                       packet_time);
 }
 
-int VoENetworkImpl::ReceivedRTCPPacket(int channel, const void* data,
-                                       size_t length)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "ReceivedRTCPPacket(channel=%d, length=%" PRIuS ")", channel,
-                 length);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (length < 4)
-    {
-        _shared->SetLastError(VE_INVALID_PACKET, kTraceError,
-            "ReceivedRTCPPacket() invalid packet length");
-        return -1;
-    }
-    if (NULL == data)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "ReceivedRTCPPacket() invalid data vector");
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "ReceivedRTCPPacket() failed to locate channel");
-        return -1;
-    }
-    if (!channelPtr->ExternalTransport())
-    {
-        _shared->SetLastError(VE_INVALID_OPERATION, kTraceError,
-            "ReceivedRTCPPacket() external transport is not enabled");
-        return -1;
-    }
-    return channelPtr->ReceivedRTCPPacket((const int8_t*) data, length);
+int VoENetworkImpl::ReceivedRTCPPacket(int channel,
+                                       const void* data,
+                                       size_t length) {
+  RTC_CHECK(_shared->statistics().Initialized());
+  RTC_CHECK(data);
+  if (length < 4) {
+    LOG_F(LS_ERROR) << "Invalid packet length: " << length;
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (!channelPtr) {
+    LOG_F(LS_ERROR) << "Failed to locate channel: " << channel;
+    return -1;
+  }
+  if (!channelPtr->ExternalTransport()) {
+    LOG_F(LS_ERROR) << "No external transport for channel: " << channel;
+    return -1;
+  }
+  return channelPtr->ReceivedRTCPPacket((const int8_t*)data, length);
 }
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h
index ee9b92ec3b..d3601e30a6 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h
@@ -15,30 +15,27 @@
 
 #include "webrtc/voice_engine/shared_data.h"
 
+namespace webrtc {
 
-namespace webrtc
-{
+class VoENetworkImpl : public VoENetwork {
+ public:
+  int RegisterExternalTransport(int channel, Transport& transport) override;
+  int DeRegisterExternalTransport(int channel) override;
 
-class VoENetworkImpl: public VoENetwork
-{
-public:
- int RegisterExternalTransport(int channel, Transport& transport) override;
+  int ReceivedRTPPacket(int channel, const void* data, size_t length) override;
+  int ReceivedRTPPacket(int channel,
+                        const void* data,
+                        size_t length,
+                        const PacketTime& packet_time) override;
 
- int DeRegisterExternalTransport(int channel) override;
+  int ReceivedRTCPPacket(int channel, const void* data, size_t length) override;
 
- int ReceivedRTPPacket(int channel, const void* data, size_t length) override;
- int ReceivedRTPPacket(int channel,
-                       const void* data,
-                       size_t length,
-                       const PacketTime& packet_time) override;
+ protected:
+  VoENetworkImpl(voe::SharedData* shared);
+  ~VoENetworkImpl() override;
 
- int ReceivedRTCPPacket(int channel, const void* data, size_t length) override;
-
-protected:
-    VoENetworkImpl(voe::SharedData* shared);
-    virtual ~VoENetworkImpl();
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_network_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_network_unittest.cc
new file mode 100644
index 0000000000..e5d8cc8b16
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_network_unittest.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/voice_engine/include/voe_network.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/voice_engine/include/voe_errors.h"
+#include "webrtc/voice_engine/voice_engine_fixture.h"
+
+namespace webrtc {
+
+enum {
+  kMinValidSizeOfRtcpPacketInBytes = 4,
+  kMinValidSizeOfRtpPacketInBytes = 12,
+  kMaxValidSizeOfRtpPacketInBytes = 1292
+};
+
+// A packet with a valid header for both RTP and RTCP.
+// Methods that are tested in this file are checking only packet header.
+static const uint8_t kPacket[kMinValidSizeOfRtpPacketInBytes] = {0x80};
+static const uint8_t kPacketJunk[kMinValidSizeOfRtpPacketInBytes] = {};
+
+static const int kNonExistingChannel = 1234;
+
+class VoENetworkTest : public VoiceEngineFixture {
+ protected:
+  int CreateChannelAndRegisterExternalTransport() {
+    EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+    int channelID = base_->CreateChannel();
+    EXPECT_NE(channelID, -1);
+    EXPECT_EQ(0, network_->RegisterExternalTransport(channelID, transport_));
+    return channelID;
+  }
+};
+
+TEST_F(VoENetworkTest, RegisterAndDeRegisterExternalTransport) {
+  int channelID = CreateChannelAndRegisterExternalTransport();
+  EXPECT_EQ(0, network_->DeRegisterExternalTransport(channelID));
+}
+
+TEST_F(VoENetworkTest,
+       RegisterExternalTransportOnNonExistingChannelShouldFail) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  EXPECT_NE(
+      0, network_->RegisterExternalTransport(kNonExistingChannel, transport_));
+}
+
+TEST_F(VoENetworkTest,
+       DeRegisterExternalTransportOnNonExistingChannelShouldFail) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  EXPECT_NE(0, network_->DeRegisterExternalTransport(kNonExistingChannel));
+}
+
+TEST_F(VoENetworkTest, DeRegisterExternalTransportBeforeRegister) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  int channelID = base_->CreateChannel();
+  EXPECT_NE(channelID, -1);
+  EXPECT_EQ(0, network_->DeRegisterExternalTransport(channelID));
+}
+
+TEST_F(VoENetworkTest, ReceivedRTPPacketWithJunkDataShouldFail) {
+  int channelID = CreateChannelAndRegisterExternalTransport();
+  EXPECT_EQ(-1, network_->ReceivedRTPPacket(channelID, kPacketJunk,
+                                            sizeof(kPacketJunk)));
+}
+
+TEST_F(VoENetworkTest, ReceivedRTPPacketOnNonExistingChannelShouldFail) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  EXPECT_EQ(-1, network_->ReceivedRTPPacket(kNonExistingChannel, kPacket,
+                                            sizeof(kPacket)));
+}
+
+TEST_F(VoENetworkTest, ReceivedRTPPacketOnChannelWithoutTransportShouldFail) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  int channelID = base_->CreateChannel();
+  EXPECT_NE(channelID, -1);
+  EXPECT_EQ(-1,
+            network_->ReceivedRTPPacket(channelID, kPacket, sizeof(kPacket)));
+}
+
+TEST_F(VoENetworkTest, ReceivedTooSmallRTPPacketShouldFail) {
+  int channelID = CreateChannelAndRegisterExternalTransport();
+  EXPECT_EQ(-1, network_->ReceivedRTPPacket(
+                    channelID, kPacket, kMinValidSizeOfRtpPacketInBytes - 1));
+}
+
+TEST_F(VoENetworkTest, ReceivedTooLargeRTPPacketShouldFail) {
+  int channelID = CreateChannelAndRegisterExternalTransport();
+  EXPECT_EQ(-1, network_->ReceivedRTPPacket(
+                    channelID, kPacket, kMaxValidSizeOfRtpPacketInBytes + 1));
+}
+
+TEST_F(VoENetworkTest, ReceivedRTCPPacketWithJunkDataShouldFail) {
+  int channelID = CreateChannelAndRegisterExternalTransport();
+  EXPECT_EQ(0, network_->ReceivedRTCPPacket(channelID, kPacketJunk,
+                                            sizeof(kPacketJunk)));
+  EXPECT_EQ(VE_SOCKET_TRANSPORT_MODULE_ERROR, base_->LastError());
+}
+
+TEST_F(VoENetworkTest, ReceivedRTCPPacketOnNonExistingChannelShouldFail) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  EXPECT_EQ(-1, network_->ReceivedRTCPPacket(kNonExistingChannel, kPacket,
+                                             sizeof(kPacket)));
+}
+
+TEST_F(VoENetworkTest, ReceivedRTCPPacketOnChannelWithoutTransportShouldFail) {
+  EXPECT_EQ(0, base_->Init(&adm_, nullptr));
+  int channelID = base_->CreateChannel();
+  EXPECT_NE(channelID, -1);
+  EXPECT_EQ(-1,
+            network_->ReceivedRTCPPacket(channelID, kPacket, sizeof(kPacket)));
+}
+
+TEST_F(VoENetworkTest, ReceivedTooSmallRTCPPacket4ShouldFail) {
+  int channelID = CreateChannelAndRegisterExternalTransport();
+  EXPECT_EQ(-1, network_->ReceivedRTCPPacket(
+                    channelID, kPacket, kMinValidSizeOfRtcpPacketInBytes - 1));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
index 422cd27f0e..08c33f6962 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/file_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/video_engine/include/vie_network.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voe_rtp_rtcp_impl.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
@@ -21,145 +20,128 @@
 
 namespace webrtc {
 
-VoERTP_RTCP* VoERTP_RTCP::GetInterface(VoiceEngine* voiceEngine)
-{
+VoERTP_RTCP* VoERTP_RTCP::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
 
-VoERTP_RTCPImpl::VoERTP_RTCPImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoERTP_RTCPImpl::VoERTP_RTCPImpl() - ctor");
+VoERTP_RTCPImpl::VoERTP_RTCPImpl(voe::SharedData* shared) : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoERTP_RTCPImpl::VoERTP_RTCPImpl() - ctor");
 }
 
-VoERTP_RTCPImpl::~VoERTP_RTCPImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoERTP_RTCPImpl::~VoERTP_RTCPImpl() - dtor");
+VoERTP_RTCPImpl::~VoERTP_RTCPImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoERTP_RTCPImpl::~VoERTP_RTCPImpl() - dtor");
 }
 
-int VoERTP_RTCPImpl::SetLocalSSRC(int channel, unsigned int ssrc)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetLocalSSRC(channel=%d, %lu)", channel, ssrc);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetLocalSSRC() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetLocalSSRC(ssrc);
+int VoERTP_RTCPImpl::SetLocalSSRC(int channel, unsigned int ssrc) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetLocalSSRC(channel=%d, %lu)", channel, ssrc);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetLocalSSRC() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetLocalSSRC(ssrc);
 }
 
-int VoERTP_RTCPImpl::GetLocalSSRC(int channel, unsigned int& ssrc)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetLocalSSRC(channel=%d, ssrc=?)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetLocalSSRC() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetLocalSSRC(ssrc);
+int VoERTP_RTCPImpl::GetLocalSSRC(int channel, unsigned int& ssrc) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetLocalSSRC() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetLocalSSRC(ssrc);
 }
 
-int VoERTP_RTCPImpl::GetRemoteSSRC(int channel, unsigned int& ssrc)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRemoteSSRC(channel=%d, ssrc=?)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRemoteSSRC() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRemoteSSRC(ssrc);
+int VoERTP_RTCPImpl::GetRemoteSSRC(int channel, unsigned int& ssrc) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRemoteSSRC() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRemoteSSRC(ssrc);
 }
 
 int VoERTP_RTCPImpl::SetSendAudioLevelIndicationStatus(int channel,
                                                        bool enable,
-                                                       unsigned char id)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetSendAudioLevelIndicationStatus(channel=%d, enable=%d,"
-                 " ID=%u)", channel, enable, id);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (enable && (id < kVoiceEngineMinRtpExtensionId ||
-                   id > kVoiceEngineMaxRtpExtensionId))
-    {
-        // [RFC5285] The 4-bit id is the local identifier of this element in
-        // the range 1-14 inclusive.
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetSendAudioLevelIndicationStatus() invalid ID parameter");
-        return -1;
-    }
+                                                       unsigned char id) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetSendAudioLevelIndicationStatus(channel=%d, enable=%d,"
+               " ID=%u)",
+               channel, enable, id);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (enable && (id < kVoiceEngineMinRtpExtensionId ||
+                 id > kVoiceEngineMaxRtpExtensionId)) {
+    // [RFC5285] The 4-bit id is the local identifier of this element in
+    // the range 1-14 inclusive.
+    _shared->SetLastError(
+        VE_INVALID_ARGUMENT, kTraceError,
+        "SetSendAudioLevelIndicationStatus() invalid ID parameter");
+    return -1;
+  }
 
-    // Set state and id for the specified channel.
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetSendAudioLevelIndicationStatus() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetSendAudioLevelIndicationStatus(enable, id);
+  // Set state and id for the specified channel.
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
+        "SetSendAudioLevelIndicationStatus() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetSendAudioLevelIndicationStatus(enable, id);
 }
 
 int VoERTP_RTCPImpl::SetReceiveAudioLevelIndicationStatus(int channel,
                                                           bool enable,
                                                           unsigned char id) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+  WEBRTC_TRACE(
+      kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
       "SetReceiveAudioLevelIndicationStatus(channel=%d, enable=%d, id=%u)",
       channel, enable, id);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
-  if (enable &&
-      (id < kVoiceEngineMinRtpExtensionId ||
-       id > kVoiceEngineMaxRtpExtensionId)) {
+  if (enable && (id < kVoiceEngineMinRtpExtensionId ||
+                 id > kVoiceEngineMaxRtpExtensionId)) {
     // [RFC5285] The 4-bit id is the local identifier of this element in
     // the range 1-14 inclusive.
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+    _shared->SetLastError(
+        VE_INVALID_ARGUMENT, kTraceError,
         "SetReceiveAbsoluteSenderTimeStatus() invalid id parameter");
     return -1;
   }
@@ -167,7 +149,8 @@ int VoERTP_RTCPImpl::SetReceiveAudioLevelIndicationStatus(int channel,
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
   voe::Channel* channel_ptr = ch.channel();
   if (channel_ptr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
         "SetReceiveAudioLevelIndicationStatus() failed to locate channel");
     return -1;
   }
@@ -188,7 +171,8 @@ int VoERTP_RTCPImpl::SetSendAbsoluteSenderTimeStatus(int channel,
                  id > kVoiceEngineMaxRtpExtensionId)) {
     // [RFC5285] The 4-bit id is the local identifier of this element in
     // the range 1-14 inclusive.
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+    _shared->SetLastError(
+        VE_INVALID_ARGUMENT, kTraceError,
         "SetSendAbsoluteSenderTimeStatus() invalid id parameter");
     return -1;
   }
@@ -196,7 +180,8 @@ int VoERTP_RTCPImpl::SetSendAbsoluteSenderTimeStatus(int channel,
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
         "SetSendAbsoluteSenderTimeStatus() failed to locate channel");
     return -1;
   }
@@ -206,7 +191,8 @@ int VoERTP_RTCPImpl::SetSendAbsoluteSenderTimeStatus(int channel,
 int VoERTP_RTCPImpl::SetReceiveAbsoluteSenderTimeStatus(int channel,
                                                         bool enable,
                                                         unsigned char id) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+  WEBRTC_TRACE(
+      kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
       "SetReceiveAbsoluteSenderTimeStatus(channel=%d, enable=%d, id=%u)",
       channel, enable, id);
   if (!_shared->statistics().Initialized()) {
@@ -217,7 +203,8 @@ int VoERTP_RTCPImpl::SetReceiveAbsoluteSenderTimeStatus(int channel,
                  id > kVoiceEngineMaxRtpExtensionId)) {
     // [RFC5285] The 4-bit id is the local identifier of this element in
     // the range 1-14 inclusive.
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+    _shared->SetLastError(
+        VE_INVALID_ARGUMENT, kTraceError,
         "SetReceiveAbsoluteSenderTimeStatus() invalid id parameter");
     return -1;
   }
@@ -225,92 +212,77 @@ int VoERTP_RTCPImpl::SetReceiveAbsoluteSenderTimeStatus(int channel,
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
         "SetReceiveAbsoluteSenderTimeStatus() failed to locate channel");
     return -1;
   }
   return channelPtr->SetReceiveAbsoluteSenderTimeStatus(enable, id);
 }
 
-int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetRTCPStatus() failed to locate channel");
-        return -1;
-    }
-    channelPtr->SetRTCPStatus(enable);
-    return 0;
+int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetRTCPStatus() failed to locate channel");
+    return -1;
+  }
+  channelPtr->SetRTCPStatus(enable);
+  return 0;
 }
 
-int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRTCPStatus(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRTCPStatus() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRTCPStatus(enabled);
+int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRTCPStatus() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRTCPStatus(enabled);
 }
 
-int VoERTP_RTCPImpl::SetRTCP_CNAME(int channel, const char cName[256])
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetRTCP_CNAME(channel=%d, cName=%s)", channel, cName);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetRTCP_CNAME() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetRTCP_CNAME(cName);
+int VoERTP_RTCPImpl::SetRTCP_CNAME(int channel, const char cName[256]) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetRTCP_CNAME(channel=%d, cName=%s)", channel, cName);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetRTCP_CNAME() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetRTCP_CNAME(cName);
 }
 
-int VoERTP_RTCPImpl::GetRemoteRTCP_CNAME(int channel, char cName[256])
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRemoteRTCP_CNAME(channel=%d, cName=?)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRemoteRTCP_CNAME() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRemoteRTCP_CNAME(cName);
+int VoERTP_RTCPImpl::GetRemoteRTCP_CNAME(int channel, char cName[256]) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRemoteRTCP_CNAME() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRemoteRTCP_CNAME(cName);
 }
 
 int VoERTP_RTCPImpl::GetRemoteRTCPReceiverInfo(
@@ -324,29 +296,27 @@ int VoERTP_RTCPImpl::GetRemoteRTCPReceiverInfo(
     uint32_t& cumulativeLost,  // from report block 1 in RR
     int32_t& rttMs)
 {
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRemoteRTCPReceiverInfo(channel=%d,...)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRemoteRTCPReceiverInfo() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRemoteRTCPReceiverInfo(NTPHigh,
-                                                 NTPLow,
-                                                 receivedPacketCount,
-                                                 receivedOctetCount,
-                                                 jitter,
-                                                 fractionLost,
-                                                 cumulativeLost,
-                                                 rttMs);
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "GetRemoteRTCPReceiverInfo(channel=%d,...)", channel);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRemoteRTCPReceiverInfo() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRemoteRTCPReceiverInfo(NTPHigh,
+                                               NTPLow,
+                                               receivedPacketCount,
+                                               receivedOctetCount,
+                                               jitter,
+                                               fractionLost,
+                                               cumulativeLost,
+                                               rttMs);
 }
 
 int VoERTP_RTCPImpl::GetRTPStatistics(int channel,
@@ -355,51 +325,40 @@ int VoERTP_RTCPImpl::GetRTPStatistics(int channel,
                                       unsigned int& discardedPackets,
                                       unsigned int& cumulativeLost)
 {
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRTPStatistics(channel=%d,....)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRTPStatistics() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRTPStatistics(averageJitterMs,
-                                        maxJitterMs,
-                                        discardedPackets,
-                                        cumulativeLost);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRTPStatistics() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRTPStatistics(averageJitterMs,
+                                      maxJitterMs,
+                                      discardedPackets,
+                                      cumulativeLost);
 }
 
-int VoERTP_RTCPImpl::GetRTCPStatistics(int channel, CallStatistics& stats)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRTCPStatistics(channel=%d)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetRTPStatistics() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRTPStatistics(stats);
+int VoERTP_RTCPImpl::GetRTCPStatistics(int channel, CallStatistics& stats) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetRTPStatistics() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRTPStatistics(stats);
 }
 
 int VoERTP_RTCPImpl::GetRemoteRTCPReportBlocks(
     int channel, std::vector<ReportBlock>* report_blocks) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetRemoteRTCPReportBlocks(channel=%d)", channel);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -407,173 +366,76 @@ int VoERTP_RTCPImpl::GetRemoteRTCPReportBlocks(
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
   voe::Channel* channel_ptr = ch.channel();
   if (channel_ptr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
         "GetRemoteRTCPReportBlocks() failed to locate channel");
     return -1;
   }
   return channel_ptr->GetRemoteRTCPReportBlocks(report_blocks);
 }
 
-int VoERTP_RTCPImpl::SetREDStatus(int channel, bool enable, int redPayloadtype)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetREDStatus(channel=%d, enable=%d, redPayloadtype=%d)",
-                 channel, enable, redPayloadtype);
+int VoERTP_RTCPImpl::SetREDStatus(int channel,
+                                  bool enable,
+                                  int redPayloadtype) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetREDStatus(channel=%d, enable=%d, redPayloadtype=%d)",
+               channel, enable, redPayloadtype);
 #ifdef WEBRTC_CODEC_RED
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetREDStatus() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetREDStatus(enable, redPayloadtype);
-#else
-    _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-        "SetREDStatus() RED is not supported");
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetREDStatus() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetREDStatus(enable, redPayloadtype);
+#else
+  _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+                        "SetREDStatus() RED is not supported");
+  return -1;
 #endif
 }
 
 int VoERTP_RTCPImpl::GetREDStatus(int channel,
                                   bool& enabled,
-                                  int& redPayloadtype)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetREDStatus(channel=%d, enabled=?, redPayloadtype=?)",
-                 channel);
+                                  int& redPayloadtype) {
 #ifdef WEBRTC_CODEC_RED
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetREDStatus() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetREDStatus(enabled, redPayloadtype);
-#else
-    _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-        "GetREDStatus() RED is not supported");
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetREDStatus() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetREDStatus(enabled, redPayloadtype);
+#else
+  _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
+                        "GetREDStatus() RED is not supported");
+  return -1;
 #endif
 }
 
-int VoERTP_RTCPImpl::SetNACKStatus(int channel,
-                                   bool enable,
-                                   int maxNoPackets)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetNACKStatus(channel=%d, enable=%d, maxNoPackets=%d)",
-                 channel, enable, maxNoPackets);
-
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetNACKStatus() failed to locate channel");
-        return -1;
-    }
-    channelPtr->SetNACKStatus(enable, maxNoPackets);
-    return 0;
-}
-
-
-int VoERTP_RTCPImpl::StartRTPDump(int channel,
-                                  const char fileNameUTF8[1024],
-                                  RTPDirections direction)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StartRTPDump(channel=%d, fileNameUTF8=%s, direction=%d)",
-                 channel, fileNameUTF8, direction);
-    assert(1024 == FileWrapper::kMaxFileNameSize);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StartRTPDump() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->StartRTPDump(fileNameUTF8, direction);
-}
-
-int VoERTP_RTCPImpl::StopRTPDump(int channel, RTPDirections direction)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "StopRTPDump(channel=%d, direction=%d)", channel, direction);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StopRTPDump() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->StopRTPDump(direction);
-}
-
-int VoERTP_RTCPImpl::RTPDumpIsActive(int channel, RTPDirections direction)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "RTPDumpIsActive(channel=%d, direction=%d)",
-                 channel, direction);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "StopRTPDump() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->RTPDumpIsActive(direction);
-}
-
-int VoERTP_RTCPImpl::SetVideoEngineBWETarget(int channel,
-                                             ViENetwork* vie_network,
-                                             int video_channel) {
+int VoERTP_RTCPImpl::SetNACKStatus(int channel, bool enable, int maxNoPackets) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-      "SetVideoEngineBWETarget(channel=%d, vie_network=?, video_channel=%d)",
-      channel, vie_network, video_channel);
+               "SetNACKStatus(channel=%d, enable=%d, maxNoPackets=%d)", channel,
+               enable, maxNoPackets);
 
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == NULL) {
     _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "SetVideoEngineBWETarget() failed to locate channel");
-    if (vie_network) {
-      vie_network->Release();
-    }
+                          "SetNACKStatus() failed to locate channel");
     return -1;
   }
-  channelPtr->SetVideoEngineBWETarget(vie_network, video_channel);
+  channelPtr->SetNACKStatus(enable, maxNoPackets);
   return 0;
 }
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.h
index 4b48205925..6fc9d1a0b6 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.h
@@ -17,96 +17,79 @@
 
 namespace webrtc {
 
-class VoERTP_RTCPImpl : public VoERTP_RTCP
-{
-public:
-    // RTCP
-    virtual int SetRTCPStatus(int channel, bool enable);
+class VoERTP_RTCPImpl : public VoERTP_RTCP {
+ public:
+  // RTCP
+  int SetRTCPStatus(int channel, bool enable) override;
 
-    virtual int GetRTCPStatus(int channel, bool& enabled);
+  int GetRTCPStatus(int channel, bool& enabled) override;
 
-    virtual int SetRTCP_CNAME(int channel, const char cName[256]);
+  int SetRTCP_CNAME(int channel, const char cName[256]) override;
 
-    virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]);
+  int GetRemoteRTCP_CNAME(int channel, char cName[256]) override;
 
-    virtual int GetRemoteRTCPReceiverInfo(int channel,
-                                          uint32_t& NTPHigh,
-                                          uint32_t& NTPLow,
-                                          uint32_t& receivedPacketCount,
-                                          uint64_t& receivedOctetCount,
-                                          uint32_t& jitter,
-                                          uint16_t& fractionLost,
-                                          uint32_t& cumulativeLost,
-                                          int32_t& rttMs);
+  int GetRemoteRTCPReceiverInfo(int channel,
+                                uint32_t& NTPHigh,
+                                uint32_t& NTPLow,
+                                uint32_t& receivedPacketCount,
+                                uint64_t& receivedOctetCount,
+                                uint32_t& jitter,
+                                uint16_t& fractionLost,
+                                uint32_t& cumulativeLost,
+                                int32_t& rttMs) override;
 
-    // SSRC
-    virtual int SetLocalSSRC(int channel, unsigned int ssrc);
+  // SSRC
+  int SetLocalSSRC(int channel, unsigned int ssrc) override;
 
-    virtual int GetLocalSSRC(int channel, unsigned int& ssrc);
+  int GetLocalSSRC(int channel, unsigned int& ssrc) override;
 
-    virtual int GetRemoteSSRC(int channel, unsigned int& ssrc);
+  int GetRemoteSSRC(int channel, unsigned int& ssrc) override;
 
-    // RTP Header Extension for Client-to-Mixer Audio Level Indication
-    virtual int SetSendAudioLevelIndicationStatus(int channel,
-                                                  bool enable,
-                                                  unsigned char id);
-    virtual int SetReceiveAudioLevelIndicationStatus(int channel,
-                                                     bool enable,
-                                                     unsigned char id);
+  // RTP Header Extension for Client-to-Mixer Audio Level Indication
+  int SetSendAudioLevelIndicationStatus(int channel,
+                                        bool enable,
+                                        unsigned char id) override;
+  int SetReceiveAudioLevelIndicationStatus(int channel,
+                                           bool enable,
+                                           unsigned char id) override;
 
-    // RTP Header Extension for Absolute Sender Time
-    virtual int SetSendAbsoluteSenderTimeStatus(int channel,
-                                                bool enable,
-                                                unsigned char id);
-    virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
-                                                   bool enable,
-                                                   unsigned char id);
+  // RTP Header Extension for Absolute Sender Time
+  int SetSendAbsoluteSenderTimeStatus(int channel,
+                                      bool enable,
+                                      unsigned char id) override;
+  int SetReceiveAbsoluteSenderTimeStatus(int channel,
+                                         bool enable,
+                                         unsigned char id) override;
 
-    // Statistics
-    virtual int GetRTPStatistics(int channel,
-                                 unsigned int& averageJitterMs,
-                                 unsigned int& maxJitterMs,
-                                 unsigned int& discardedPackets,
-                                 unsigned int& cumulativeLost);
+  // Statistics
+  int GetRTPStatistics(int channel,
+                       unsigned int& averageJitterMs,
+                       unsigned int& maxJitterMs,
+                       unsigned int& discardedPackets,
+                       unsigned int& cumulativeLost) override;
 
-    virtual int GetRTCPStatistics(int channel, CallStatistics& stats);
+  int GetRTCPStatistics(int channel, CallStatistics& stats) override;
 
-    virtual int GetRemoteRTCPReportBlocks(
-        int channel, std::vector<ReportBlock>* report_blocks);
+  int GetRemoteRTCPReportBlocks(
+      int channel,
+      std::vector<ReportBlock>* report_blocks) override;
 
-    // RED
-    virtual int SetREDStatus(int channel,
-                             bool enable,
-                             int redPayloadtype = -1);
+  // RED
+  int SetREDStatus(int channel, bool enable, int redPayloadtype = -1) override;
 
-    virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype);
+  int GetREDStatus(int channel, bool& enabled, int& redPayloadtype) override;
 
-    //NACK
-    virtual int SetNACKStatus(int channel,
-                              bool enable,
-                              int maxNoPackets);
+  // NACK
+  int SetNACKStatus(int channel, bool enable, int maxNoPackets) override;
 
-    // Store RTP and RTCP packets and dump to file (compatible with rtpplay)
-    virtual int StartRTPDump(int channel,
-                             const char fileNameUTF8[1024],
-                             RTPDirections direction = kRtpIncoming);
+ protected:
+  VoERTP_RTCPImpl(voe::SharedData* shared);
+  ~VoERTP_RTCPImpl() override;
 
-    virtual int StopRTPDump(int channel,
-                            RTPDirections direction = kRtpIncoming);
-
-    virtual int RTPDumpIsActive(int channel,
-                                RTPDirections direction = kRtpIncoming);
-
-    virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network,
-                                        int video_channel);
-protected:
-    VoERTP_RTCPImpl(voe::SharedData* shared);
-    virtual ~VoERTP_RTCPImpl();
-
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
 
-#endif    // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
+#endif  // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.cc
index afee16a3bd..071668103d 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.cc
@@ -10,130 +10,110 @@
 
 #include "webrtc/voice_engine/voe_video_sync_impl.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
-VoEVideoSync* VoEVideoSync::GetInterface(VoiceEngine* voiceEngine)
-{
+VoEVideoSync* VoEVideoSync::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
 
-VoEVideoSyncImpl::VoEVideoSyncImpl(voe::SharedData* shared) : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEVideoSyncImpl::VoEVideoSyncImpl() - ctor");
+VoEVideoSyncImpl::VoEVideoSyncImpl(voe::SharedData* shared) : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEVideoSyncImpl::VoEVideoSyncImpl() - ctor");
 }
 
-VoEVideoSyncImpl::~VoEVideoSyncImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEVideoSyncImpl::~VoEVideoSyncImpl() - dtor");
+VoEVideoSyncImpl::~VoEVideoSyncImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "VoEVideoSyncImpl::~VoEVideoSyncImpl() - dtor");
 }
 
-int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel, unsigned int& timestamp)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetPlayoutTimestamp(channel=%d, timestamp=?)", channel);
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channel_ptr = ch.channel();
-    if (channel_ptr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetPlayoutTimestamp() failed to locate channel");
-        return -1;
-    }
-    return channel_ptr->GetPlayoutTimestamp(timestamp);
+int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel,
+                                          unsigned int& timestamp) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channel_ptr = ch.channel();
+  if (channel_ptr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetPlayoutTimestamp() failed to locate channel");
+    return -1;
+  }
+  return channel_ptr->GetPlayoutTimestamp(timestamp);
 }
 
-int VoEVideoSyncImpl::SetInitTimestamp(int channel,
-                                       unsigned int timestamp)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetInitTimestamp(channel=%d, timestamp=%lu)",
-                 channel, timestamp);
+int VoEVideoSyncImpl::SetInitTimestamp(int channel, unsigned int timestamp) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetInitTimestamp(channel=%d, timestamp=%lu)", channel,
+               timestamp);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetInitTimestamp() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetInitTimestamp(timestamp);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetInitTimestamp() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetInitTimestamp(timestamp);
 }
 
-int VoEVideoSyncImpl::SetInitSequenceNumber(int channel,
-                                            short sequenceNumber)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)",
-                 channel, sequenceNumber);
+int VoEVideoSyncImpl::SetInitSequenceNumber(int channel, short sequenceNumber) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)", channel,
+               sequenceNumber);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetInitSequenceNumber() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetInitSequenceNumber(sequenceNumber);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetInitSequenceNumber() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetInitSequenceNumber(sequenceNumber);
 }
 
-int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel,int delayMs)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetMinimumPlayoutDelay(channel=%d, delayMs=%d)",
-                 channel, delayMs);
+int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel, int delayMs) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetMinimumPlayoutDelay(channel=%d, delayMs=%d)", channel,
+               delayMs);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetMinimumPlayoutDelay() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetMinimumPlayoutDelay(delayMs);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetMinimumPlayoutDelay() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetMinimumPlayoutDelay(delayMs);
 }
 
 int VoEVideoSyncImpl::SetCurrentSyncOffset(int channel, int offsetMs)
@@ -153,35 +133,10 @@ int VoEVideoSyncImpl::SetCurrentSyncOffset(int channel, int offsetMs)
     return 0;
 }
 
-int VoEVideoSyncImpl::SetInitialPlayoutDelay(int channel, int delay_ms)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "SetInitialPlayoutDelay(channel=%d, delay_ms=%d)",
-                 channel, delay_ms);
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetInitialPlayoutDelay() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetInitialPlayoutDelay(delay_ms);
-}
-
 int VoEVideoSyncImpl::GetDelayEstimate(int channel,
                                        int* jitter_buffer_delay_ms,
                                        int* playout_buffer_delay_ms,
                                        int* avsync_offset_ms) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetDelayEstimate(channel=%d, delayMs=?)", channel);
-
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -201,58 +156,40 @@ int VoEVideoSyncImpl::GetDelayEstimate(int channel,
   return 0;
 }
 
-int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetPlayoutBufferSize(bufferMs=?)");
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    AudioDeviceModule::BufferType type
-        (AudioDeviceModule::kFixedBufferSize);
-    uint16_t sizeMS(0);
-    if (_shared->audio_device()->PlayoutBuffer(&type, &sizeMS) != 0)
-    {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-            "GetPlayoutBufferSize() failed to read buffer size");
-        return -1;
-    }
-    bufferMs = sizeMS;
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetPlayoutBufferSize() => bufferMs=%d", bufferMs);
-    return 0;
+int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  AudioDeviceModule::BufferType type(AudioDeviceModule::kFixedBufferSize);
+  uint16_t sizeMS(0);
+  if (_shared->audio_device()->PlayoutBuffer(&type, &sizeMS) != 0) {
+    _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
+                          "GetPlayoutBufferSize() failed to read buffer size");
+    return -1;
+  }
+  bufferMs = sizeMS;
+  return 0;
 }
 
-int VoEVideoSyncImpl::GetRtpRtcp(int channel, RtpRtcp** rtpRtcpModule,
-                                 RtpReceiver** rtp_receiver)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "GetRtpRtcp(channel=%i)", channel);
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetPlayoutTimestamp() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetRtpRtcp(rtpRtcpModule, rtp_receiver);
+int VoEVideoSyncImpl::GetRtpRtcp(int channel,
+                                 RtpRtcp** rtpRtcpModule,
+                                 RtpReceiver** rtp_receiver) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetPlayoutTimestamp() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetRtpRtcp(rtpRtcpModule, rtp_receiver);
 }
 
 int VoEVideoSyncImpl::GetLeastRequiredDelayMs(int channel) const {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetLeastRequiredDelayMS(channel=%d)", channel);
-
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -264,7 +201,7 @@ int VoEVideoSyncImpl::GetLeastRequiredDelayMs(int channel) const {
                           "GetLeastRequiredDelayMs() failed to locate channel");
     return -1;
   }
-  return channel_ptr->least_required_delay_ms();
+  return channel_ptr->LeastRequiredDelayMs();
 }
 
 #endif  // #ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.h
index db4311d018..5ee863449b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_video_sync_impl.h
@@ -17,41 +17,39 @@
 
 namespace webrtc {
 
-class VoEVideoSyncImpl : public VoEVideoSync
-{
-public:
-    virtual int GetPlayoutBufferSize(int& bufferMs);
+class VoEVideoSyncImpl : public VoEVideoSync {
+ public:
+  int GetPlayoutBufferSize(int& bufferMs) override;
 
-    virtual int SetMinimumPlayoutDelay(int channel, int delayMs);
+  int SetMinimumPlayoutDelay(int channel, int delayMs) override;
 
-    virtual int SetCurrentSyncOffset(int channel, int offsetMs);
+  int SetCurrentSyncOffset(int channel, int offsetMs) override;
 
-    virtual int SetInitialPlayoutDelay(int channel, int delay_ms);
+  int GetDelayEstimate(int channel,
+                       int* jitter_buffer_delay_ms,
+                       int* playout_buffer_delay_ms,
+                       int* avsync_offset_ms) override;
 
-    virtual int GetDelayEstimate(int channel,
-                                 int* jitter_buffer_delay_ms,
-                                 int* playout_buffer_delay_ms,
-                                 int* avsync_offset_ms);
+  int GetLeastRequiredDelayMs(int channel) const override;
 
-    virtual int GetLeastRequiredDelayMs(int channel) const;
+  int SetInitTimestamp(int channel, unsigned int timestamp) override;
 
-    virtual int SetInitTimestamp(int channel, unsigned int timestamp);
+  int SetInitSequenceNumber(int channel, short sequenceNumber) override;
 
-    virtual int SetInitSequenceNumber(int channel, short sequenceNumber);
+  int GetPlayoutTimestamp(int channel, unsigned int& timestamp) override;
 
-    virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp);
+  int GetRtpRtcp(int channel,
+                 RtpRtcp** rtpRtcpModule,
+                 RtpReceiver** rtp_receiver) override;
 
-    virtual int GetRtpRtcp(int channel, RtpRtcp** rtpRtcpModule,
-                           RtpReceiver** rtp_receiver);
+ protected:
+  VoEVideoSyncImpl(voe::SharedData* shared);
+  ~VoEVideoSyncImpl() override;
 
-protected:
-    VoEVideoSyncImpl(voe::SharedData* shared);
-    virtual ~VoEVideoSyncImpl();
-
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
 
-#endif    // WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
+#endif  // WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.cc
index f27c4ffca1..bb57ed7cd0 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.cc
@@ -10,8 +10,8 @@
 
 #include "webrtc/voice_engine/voe_volume_control_impl.h"
 
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/output_mixer.h"
@@ -20,507 +20,399 @@
 
 namespace webrtc {
 
-VoEVolumeControl* VoEVolumeControl::GetInterface(VoiceEngine* voiceEngine)
-{
+VoEVolumeControl* VoEVolumeControl::GetInterface(VoiceEngine* voiceEngine) {
 #ifndef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-    return NULL;
+  return NULL;
 #else
-    if (NULL == voiceEngine)
-    {
-        return NULL;
-    }
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    s->AddRef();
-    return s;
+  if (NULL == voiceEngine) {
+    return NULL;
+  }
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  s->AddRef();
+  return s;
 #endif
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
 
 VoEVolumeControlImpl::VoEVolumeControlImpl(voe::SharedData* shared)
-    : _shared(shared)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+    : _shared(shared) {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "VoEVolumeControlImpl::VoEVolumeControlImpl() - ctor");
 }
 
-VoEVolumeControlImpl::~VoEVolumeControlImpl()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
+VoEVolumeControlImpl::~VoEVolumeControlImpl() {
+  WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "VoEVolumeControlImpl::~VoEVolumeControlImpl() - dtor");
 }
 
-int VoEVolumeControlImpl::SetSpeakerVolume(unsigned int volume)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+int VoEVolumeControlImpl::SetSpeakerVolume(unsigned int volume) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetSpeakerVolume(volume=%u)", volume);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (volume > kMaxVolumeLevel)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetSpeakerVolume() invalid argument");
-        return -1;
-    }
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (volume > kMaxVolumeLevel) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetSpeakerVolume() invalid argument");
+    return -1;
+  }
 
-    uint32_t maxVol(0);
-    uint32_t spkrVol(0);
+  uint32_t maxVol(0);
+  uint32_t spkrVol(0);
 
-    // scale: [0,kMaxVolumeLevel] -> [0,MaxSpeakerVolume]
-    if (_shared->audio_device()->MaxSpeakerVolume(&maxVol) != 0)
-    {
-        _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
-            "SetSpeakerVolume() failed to get max volume");
-        return -1;
-    }
-    // Round the value and avoid floating computation.
-    spkrVol = (uint32_t)((volume * maxVol +
-        (int)(kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
+  // scale: [0,kMaxVolumeLevel] -> [0,MaxSpeakerVolume]
+  if (_shared->audio_device()->MaxSpeakerVolume(&maxVol) != 0) {
+    _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
+                          "SetSpeakerVolume() failed to get max volume");
+    return -1;
+  }
+  // Round the value and avoid floating computation.
+  spkrVol = (uint32_t)((volume * maxVol + (int)(kMaxVolumeLevel / 2)) /
+                       (kMaxVolumeLevel));
 
-    // set the actual volume using the audio mixer
-    if (_shared->audio_device()->SetSpeakerVolume(spkrVol) != 0)
-    {
-        _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
-            "SetSpeakerVolume() failed to set speaker volume");
-        return -1;
-    }
-    return 0;
+  // set the actual volume using the audio mixer
+  if (_shared->audio_device()->SetSpeakerVolume(spkrVol) != 0) {
+    _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
+                          "SetSpeakerVolume() failed to set speaker volume");
+    return -1;
+  }
+  return 0;
 }
 
-int VoEVolumeControlImpl::GetSpeakerVolume(unsigned int& volume)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetSpeakerVolume()");
+int VoEVolumeControlImpl::GetSpeakerVolume(unsigned int& volume) {
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    uint32_t spkrVol(0);
-    uint32_t maxVol(0);
+  uint32_t spkrVol(0);
+  uint32_t maxVol(0);
 
-    if (_shared->audio_device()->SpeakerVolume(&spkrVol) != 0)
-    {
-        _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
-            "GetSpeakerVolume() unable to get speaker volume");
-        return -1;
-    }
+  if (_shared->audio_device()->SpeakerVolume(&spkrVol) != 0) {
+    _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
+                          "GetSpeakerVolume() unable to get speaker volume");
+    return -1;
+  }
 
-    // scale: [0, MaxSpeakerVolume] -> [0, kMaxVolumeLevel]
-    if (_shared->audio_device()->MaxSpeakerVolume(&maxVol) != 0)
-    {
-        _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
-            "GetSpeakerVolume() unable to get max speaker volume");
-        return -1;
-    }
-    // Round the value and avoid floating computation.
-    volume = (uint32_t) ((spkrVol * kMaxVolumeLevel +
-        (int)(maxVol / 2)) / (maxVol));
+  // scale: [0, MaxSpeakerVolume] -> [0, kMaxVolumeLevel]
+  if (_shared->audio_device()->MaxSpeakerVolume(&maxVol) != 0) {
+    _shared->SetLastError(
+        VE_GET_MIC_VOL_ERROR, kTraceError,
+        "GetSpeakerVolume() unable to get max speaker volume");
+    return -1;
+  }
+  // Round the value and avoid floating computation.
+  volume =
+      (uint32_t)((spkrVol * kMaxVolumeLevel + (int)(maxVol / 2)) / (maxVol));
 
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetSpeakerVolume() => volume=%d", volume);
-    return 0;
+  return 0;
 }
 
-int VoEVolumeControlImpl::SetMicVolume(unsigned int volume)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+int VoEVolumeControlImpl::SetMicVolume(unsigned int volume) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetMicVolume(volume=%u)", volume);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (volume > kMaxVolumeLevel)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetMicVolume() invalid argument");
-        return -1;
-    }
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (volume > kMaxVolumeLevel) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetMicVolume() invalid argument");
+    return -1;
+  }
 
-    uint32_t maxVol(0);
-    uint32_t micVol(0);
+  uint32_t maxVol(0);
+  uint32_t micVol(0);
 
-    // scale: [0, kMaxVolumeLevel] -> [0,MaxMicrophoneVolume]
-    if (_shared->audio_device()->MaxMicrophoneVolume(&maxVol) != 0)
-    {
-        _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
-            "SetMicVolume() failed to get max volume");
-        return -1;
+  // scale: [0, kMaxVolumeLevel] -> [0,MaxMicrophoneVolume]
+  if (_shared->audio_device()->MaxMicrophoneVolume(&maxVol) != 0) {
+    _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
+                          "SetMicVolume() failed to get max volume");
+    return -1;
+  }
+
+  if (volume == kMaxVolumeLevel) {
+    // On Linux running pulse, users are able to set the volume above 100%
+    // through the volume control panel, where the +100% range is digital
+    // scaling. WebRTC does not support setting the volume above 100%, and
+    // simply ignores changing the volume if the user tries to set it to
+    // |kMaxVolumeLevel| while the current volume is higher than |maxVol|.
+    if (_shared->audio_device()->MicrophoneVolume(&micVol) != 0) {
+      _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
+                            "SetMicVolume() unable to get microphone volume");
+      return -1;
     }
+    if (micVol >= maxVol)
+      return 0;
+  }
 
-    if (volume == kMaxVolumeLevel) {
-      // On Linux running pulse, users are able to set the volume above 100%
-      // through the volume control panel, where the +100% range is digital
-      // scaling. WebRTC does not support setting the volume above 100%, and
-      // simply ignores changing the volume if the user tries to set it to
-      // |kMaxVolumeLevel| while the current volume is higher than |maxVol|.
-      if (_shared->audio_device()->MicrophoneVolume(&micVol) != 0) {
-        _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
-            "SetMicVolume() unable to get microphone volume");
-        return -1;
-      }
-      if (micVol >= maxVol)
-        return 0;
-    }
+  // Round the value and avoid floating point computation.
+  micVol = (uint32_t)((volume * maxVol + (int)(kMaxVolumeLevel / 2)) /
+                      (kMaxVolumeLevel));
 
-    // Round the value and avoid floating point computation.
-    micVol = (uint32_t) ((volume * maxVol +
-        (int)(kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
-
-    // set the actual volume using the audio mixer
-    if (_shared->audio_device()->SetMicrophoneVolume(micVol) != 0)
-    {
-        _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
-            "SetMicVolume() failed to set mic volume");
-        return -1;
-    }
-    return 0;
+  // set the actual volume using the audio mixer
+  if (_shared->audio_device()->SetMicrophoneVolume(micVol) != 0) {
+    _shared->SetLastError(VE_MIC_VOL_ERROR, kTraceError,
+                          "SetMicVolume() failed to set mic volume");
+    return -1;
+  }
+  return 0;
 }
 
-int VoEVolumeControlImpl::GetMicVolume(unsigned int& volume)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetMicVolume()");
+int VoEVolumeControlImpl::GetMicVolume(unsigned int& volume) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  uint32_t micVol(0);
+  uint32_t maxVol(0);
 
-    uint32_t micVol(0);
-    uint32_t maxVol(0);
+  if (_shared->audio_device()->MicrophoneVolume(&micVol) != 0) {
+    _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
+                          "GetMicVolume() unable to get microphone volume");
+    return -1;
+  }
 
-    if (_shared->audio_device()->MicrophoneVolume(&micVol) != 0)
-    {
-        _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
-            "GetMicVolume() unable to get microphone volume");
-        return -1;
-    }
-
-    // scale: [0, MaxMicrophoneVolume] -> [0, kMaxVolumeLevel]
-    if (_shared->audio_device()->MaxMicrophoneVolume(&maxVol) != 0)
-    {
-        _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
-            "GetMicVolume() unable to get max microphone volume");
-        return -1;
-    }
-    if (micVol < maxVol) {
-      // Round the value and avoid floating point calculation.
-      volume = (uint32_t) ((micVol * kMaxVolumeLevel +
-          (int)(maxVol / 2)) / (maxVol));
-    } else {
-      // Truncate the value to the kMaxVolumeLevel.
-      volume = kMaxVolumeLevel;
-    }
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetMicVolume() => volume=%d", volume);
-    return 0;
+  // scale: [0, MaxMicrophoneVolume] -> [0, kMaxVolumeLevel]
+  if (_shared->audio_device()->MaxMicrophoneVolume(&maxVol) != 0) {
+    _shared->SetLastError(VE_GET_MIC_VOL_ERROR, kTraceError,
+                          "GetMicVolume() unable to get max microphone volume");
+    return -1;
+  }
+  if (micVol < maxVol) {
+    // Round the value and avoid floating point calculation.
+    volume =
+        (uint32_t)((micVol * kMaxVolumeLevel + (int)(maxVol / 2)) / (maxVol));
+  } else {
+    // Truncate the value to the kMaxVolumeLevel.
+    volume = kMaxVolumeLevel;
+  }
+  return 0;
 }
 
-int VoEVolumeControlImpl::SetInputMute(int channel, bool enable)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+int VoEVolumeControlImpl::SetInputMute(int channel, bool enable) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetInputMute(channel=%d, enable=%d)", channel, enable);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        // Mute before demultiplexing <=> affects all channels
-        return _shared->transmit_mixer()->SetMute(enable);
-    }
-    // Mute after demultiplexing <=> affects one channel only
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    // Mute before demultiplexing <=> affects all channels
+    return _shared->transmit_mixer()->SetMute(enable);
+  }
+  // Mute after demultiplexing <=> affects one channel only
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetInputMute() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetMute(enable);
+}
+
+int VoEVolumeControlImpl::GetInputMute(int channel, bool& enabled) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    enabled = _shared->transmit_mixer()->Mute();
+  } else {
     voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
     voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetInputMute() failed to locate channel");
-        return -1;
+    if (channelPtr == NULL) {
+      _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                            "SetInputMute() failed to locate channel");
+      return -1;
     }
-    return channelPtr->SetMute(enable);
+    enabled = channelPtr->Mute();
+  }
+  return 0;
 }
 
-int VoEVolumeControlImpl::GetInputMute(int channel, bool& enabled)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetInputMute(channel=%d)", channel);
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (channel == -1)
-    {
-        enabled = _shared->transmit_mixer()->Mute();
-    }
-    else
-    {
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "SetInputMute() failed to locate channel");
-            return -1;
-        }
-        enabled = channelPtr->Mute();
-    }
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetInputMute() => enabled = %d", (int)enabled);
-    return 0;
-}
-
-int VoEVolumeControlImpl::GetSpeechInputLevel(unsigned int& level)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetSpeechInputLevel()");
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    int8_t currentLevel = _shared->transmit_mixer()->AudioLevel();
-    level = static_cast<unsigned int> (currentLevel);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetSpeechInputLevel() => %d", level);
-    return 0;
+int VoEVolumeControlImpl::GetSpeechInputLevel(unsigned int& level) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  int8_t currentLevel = _shared->transmit_mixer()->AudioLevel();
+  level = static_cast<unsigned int>(currentLevel);
+  return 0;
 }
 
 int VoEVolumeControlImpl::GetSpeechOutputLevel(int channel,
-                                               unsigned int& level)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetSpeechOutputLevel(channel=%d, level=?)", channel);
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+                                               unsigned int& level) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    return _shared->output_mixer()->GetSpeechOutputLevel((uint32_t&)level);
+  } else {
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                            "GetSpeechOutputLevel() failed to locate channel");
+      return -1;
     }
-    if (channel == -1)
-    {
-        return _shared->output_mixer()->GetSpeechOutputLevel(
-            (uint32_t&)level);
-    }
-    else
-    {
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "GetSpeechOutputLevel() failed to locate channel");
-            return -1;
-        }
-        channelPtr->GetSpeechOutputLevel((uint32_t&)level);
-    }
-    return 0;
+    channelPtr->GetSpeechOutputLevel((uint32_t&)level);
+  }
+  return 0;
 }
 
-int VoEVolumeControlImpl::GetSpeechInputLevelFullRange(unsigned int& level)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetSpeechInputLevelFullRange(level=?)");
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    int16_t currentLevel = _shared->transmit_mixer()->
-        AudioLevelFullRange();
-    level = static_cast<unsigned int> (currentLevel);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-        VoEId(_shared->instance_id(), -1),
-        "GetSpeechInputLevelFullRange() => %d", level);
-    return 0;
+int VoEVolumeControlImpl::GetSpeechInputLevelFullRange(unsigned int& level) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  int16_t currentLevel = _shared->transmit_mixer()->AudioLevelFullRange();
+  level = static_cast<unsigned int>(currentLevel);
+  return 0;
 }
 
 int VoEVolumeControlImpl::GetSpeechOutputLevelFullRange(int channel,
-                                                        unsigned int& level)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetSpeechOutputLevelFullRange(channel=%d, level=?)", channel);
-
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
+                                                        unsigned int& level) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (channel == -1) {
+    return _shared->output_mixer()->GetSpeechOutputLevelFullRange(
+        (uint32_t&)level);
+  } else {
+    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+    voe::Channel* channelPtr = ch.channel();
+    if (channelPtr == NULL) {
+      _shared->SetLastError(
+          VE_CHANNEL_NOT_VALID, kTraceError,
+          "GetSpeechOutputLevelFullRange() failed to locate channel");
+      return -1;
     }
-    if (channel == -1)
-    {
-        return _shared->output_mixer()->GetSpeechOutputLevelFullRange(
-            (uint32_t&)level);
-    }
-    else
-    {
-        voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-        voe::Channel* channelPtr = ch.channel();
-        if (channelPtr == NULL)
-        {
-            _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                "GetSpeechOutputLevelFullRange() failed to locate channel");
-            return -1;
-        }
-        channelPtr->GetSpeechOutputLevelFullRange((uint32_t&)level);
-    }
-    return 0;
+    channelPtr->GetSpeechOutputLevelFullRange((uint32_t&)level);
+  }
+  return 0;
 }
 
 int VoEVolumeControlImpl::SetChannelOutputVolumeScaling(int channel,
-                                                        float scaling)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                                                        float scaling) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetChannelOutputVolumeScaling(channel=%d, scaling=%3.2f)",
                channel, scaling);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    if (scaling < kMinOutputVolumeScaling ||
-        scaling > kMaxOutputVolumeScaling)
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetChannelOutputVolumeScaling() invalid parameter");
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetChannelOutputVolumeScaling() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetChannelOutputVolumeScaling(scaling);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  if (scaling < kMinOutputVolumeScaling || scaling > kMaxOutputVolumeScaling) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetChannelOutputVolumeScaling() invalid parameter");
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
+        "SetChannelOutputVolumeScaling() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetChannelOutputVolumeScaling(scaling);
 }
 
 int VoEVolumeControlImpl::GetChannelOutputVolumeScaling(int channel,
-                                                        float& scaling)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetChannelOutputVolumeScaling(channel=%d, scaling=?)", channel);
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetChannelOutputVolumeScaling() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetChannelOutputVolumeScaling(scaling);
+                                                        float& scaling) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(
+        VE_CHANNEL_NOT_VALID, kTraceError,
+        "GetChannelOutputVolumeScaling() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetChannelOutputVolumeScaling(scaling);
 }
 
 int VoEVolumeControlImpl::SetOutputVolumePan(int channel,
                                              float left,
-                                             float right)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                                             float right) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)",
                channel, left, right);
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    bool available(false);
-    _shared->audio_device()->StereoPlayoutIsAvailable(&available);
-    if (!available)
-    {
-        _shared->SetLastError(VE_FUNC_NO_STEREO, kTraceError,
-            "SetOutputVolumePan() stereo playout not supported");
-        return -1;
-    }
-    if ((left < kMinOutputVolumePanning)  ||
-        (left > kMaxOutputVolumePanning)  ||
-        (right < kMinOutputVolumePanning) ||
-        (right > kMaxOutputVolumePanning))
-    {
-        _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-            "SetOutputVolumePan() invalid parameter");
-        return -1;
-    }
+  bool available(false);
+  _shared->audio_device()->StereoPlayoutIsAvailable(&available);
+  if (!available) {
+    _shared->SetLastError(VE_FUNC_NO_STEREO, kTraceError,
+                          "SetOutputVolumePan() stereo playout not supported");
+    return -1;
+  }
+  if ((left < kMinOutputVolumePanning) || (left > kMaxOutputVolumePanning) ||
+      (right < kMinOutputVolumePanning) || (right > kMaxOutputVolumePanning)) {
+    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+                          "SetOutputVolumePan() invalid parameter");
+    return -1;
+  }
 
-    if (channel == -1)
-    {
-        // Master balance (affectes the signal after output mixing)
-        return _shared->output_mixer()->SetOutputVolumePan(left, right);
-    }
-    // Per-channel balance (affects the signal before output mixing)
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "SetOutputVolumePan() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->SetOutputVolumePan(left, right);
+  if (channel == -1) {
+    // Master balance (affectes the signal after output mixing)
+    return _shared->output_mixer()->SetOutputVolumePan(left, right);
+  }
+  // Per-channel balance (affects the signal before output mixing)
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetOutputVolumePan() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetOutputVolumePan(left, right);
 }
 
 int VoEVolumeControlImpl::GetOutputVolumePan(int channel,
                                              float& left,
-                                             float& right)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetOutputVolumePan(channel=%d, left=?, right=?)", channel);
+                                             float& right) {
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
 
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
+  bool available(false);
+  _shared->audio_device()->StereoPlayoutIsAvailable(&available);
+  if (!available) {
+    _shared->SetLastError(VE_FUNC_NO_STEREO, kTraceError,
+                          "GetOutputVolumePan() stereo playout not supported");
+    return -1;
+  }
 
-    bool available(false);
-    _shared->audio_device()->StereoPlayoutIsAvailable(&available);
-    if (!available)
-    {
-        _shared->SetLastError(VE_FUNC_NO_STEREO, kTraceError,
-            "GetOutputVolumePan() stereo playout not supported");
-        return -1;
-    }
-
-    if (channel == -1)
-    {
-        return _shared->output_mixer()->GetOutputVolumePan(left, right);
-    }
-    voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-    voe::Channel* channelPtr = ch.channel();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetOutputVolumePan() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetOutputVolumePan(left, right);
+  if (channel == -1) {
+    return _shared->output_mixer()->GetOutputVolumePan(left, right);
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "GetOutputVolumePan() failed to locate channel");
+    return -1;
+  }
+  return channelPtr->GetOutputVolumePan(left, right);
 }
 
 #endif  // #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.h
index b5e3b1b02d..16c9c7df42 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_volume_control_impl.h
@@ -17,47 +17,44 @@
 
 namespace webrtc {
 
-class VoEVolumeControlImpl : public VoEVolumeControl
-{
-public:
-    virtual int SetSpeakerVolume(unsigned int volume);
+class VoEVolumeControlImpl : public VoEVolumeControl {
+ public:
+  int SetSpeakerVolume(unsigned int volume) override;
 
-    virtual int GetSpeakerVolume(unsigned int& volume);
+  int GetSpeakerVolume(unsigned int& volume) override;
 
-    virtual int SetMicVolume(unsigned int volume);
+  int SetMicVolume(unsigned int volume) override;
 
-    virtual int GetMicVolume(unsigned int& volume);
+  int GetMicVolume(unsigned int& volume) override;
 
-    virtual int SetInputMute(int channel, bool enable);
+  int SetInputMute(int channel, bool enable) override;
 
-    virtual int GetInputMute(int channel, bool& enabled);
+  int GetInputMute(int channel, bool& enabled) override;
 
-    virtual int GetSpeechInputLevel(unsigned int& level);
+  int GetSpeechInputLevel(unsigned int& level) override;
 
-    virtual int GetSpeechOutputLevel(int channel, unsigned int& level);
+  int GetSpeechOutputLevel(int channel, unsigned int& level) override;
 
-    virtual int GetSpeechInputLevelFullRange(unsigned int& level);
+  int GetSpeechInputLevelFullRange(unsigned int& level) override;
 
-    virtual int GetSpeechOutputLevelFullRange(int channel,
-                                              unsigned int& level);
+  int GetSpeechOutputLevelFullRange(int channel, unsigned int& level) override;
 
-    virtual int SetChannelOutputVolumeScaling(int channel, float scaling);
+  int SetChannelOutputVolumeScaling(int channel, float scaling) override;
 
-    virtual int GetChannelOutputVolumeScaling(int channel, float& scaling);
+  int GetChannelOutputVolumeScaling(int channel, float& scaling) override;
 
-    virtual int SetOutputVolumePan(int channel, float left, float right);
+  int SetOutputVolumePan(int channel, float left, float right) override;
 
-    virtual int GetOutputVolumePan(int channel, float& left, float& right);
+  int GetOutputVolumePan(int channel, float& left, float& right) override;
 
+ protected:
+  VoEVolumeControlImpl(voe::SharedData* shared);
+  ~VoEVolumeControlImpl() override;
 
-protected:
-    VoEVolumeControlImpl(voe::SharedData* shared);
-    virtual ~VoEVolumeControlImpl();
-
-private:
-    voe::SharedData* _shared;
+ private:
+  voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
 
-#endif    // WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
+#endif  // WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp b/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp
index 232dff127e..776fa83efb 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp
@@ -23,9 +23,14 @@
         '<(webrtc_root)/modules/modules.gyp:audio_processing',
         '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
         '<(webrtc_root)/modules/modules.gyp:media_file',
+        '<(webrtc_root)/modules/modules.gyp:paced_sender',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+      ],
+      'export_dependent_settings': [
+        '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
       ],
       'defines': [
         'WEBRTC_EXTERNAL_TRANSPORT',
@@ -48,6 +53,8 @@
         'channel.h',
         'channel_manager.cc',
         'channel_manager.h',
+        'channel_proxy.cc',
+        'channel_proxy.h',
         'dtmf_inband.cc',
         'dtmf_inband.h',
         'dtmf_inband_queue.cc',
@@ -109,6 +116,7 @@
           'type': '<(gtest_target_type)',
           'dependencies': [
             'voice_engine',
+            '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             # The rest are to satisfy the unittests' include chain.
             # This would be unnecessary if we used qualified includes.
@@ -131,6 +139,9 @@
             'voe_audio_processing_unittest.cc',
             'voe_base_unittest.cc',
             'voe_codec_unittest.cc',
+            'voe_network_unittest.cc',
+            'voice_engine_fixture.cc',
+            'voice_engine_fixture.h',
           ],
           'conditions': [
             ['OS=="android"', {
@@ -140,74 +151,6 @@
             }],
           ],
         },
-        {
-          'target_name': 'voe_auto_test',
-          'type': 'executable',
-          'dependencies': [
-            'voice_engine',
-            '<(DEPTH)/testing/gmock.gyp:gmock',
-            '<(DEPTH)/testing/gtest.gyp:gtest',
-            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(webrtc_root)/test/test.gyp:channel_transport',
-            '<(webrtc_root)/test/test.gyp:test_support',
-           ],
-          'sources': [
-            'test/auto_test/automated_mode.cc',
-            'test/auto_test/extended/agc_config_test.cc',
-            'test/auto_test/extended/ec_metrics_test.cc',
-            'test/auto_test/fakes/fake_external_transport.cc',
-            'test/auto_test/fakes/fake_external_transport.h',
-            'test/auto_test/fixtures/after_initialization_fixture.cc',
-            'test/auto_test/fixtures/after_initialization_fixture.h',
-            'test/auto_test/fixtures/after_streaming_fixture.cc',
-            'test/auto_test/fixtures/after_streaming_fixture.h',
-            'test/auto_test/fixtures/before_initialization_fixture.cc',
-            'test/auto_test/fixtures/before_initialization_fixture.h',
-            'test/auto_test/fixtures/before_streaming_fixture.cc',
-            'test/auto_test/fixtures/before_streaming_fixture.h',
-            'test/auto_test/standard/audio_processing_test.cc',
-            'test/auto_test/standard/codec_before_streaming_test.cc',
-            'test/auto_test/standard/codec_test.cc',
-            'test/auto_test/standard/dtmf_test.cc',
-            'test/auto_test/standard/external_media_test.cc',
-            'test/auto_test/standard/file_before_streaming_test.cc',
-            'test/auto_test/standard/file_test.cc',
-            'test/auto_test/standard/hardware_before_initializing_test.cc',
-            'test/auto_test/standard/hardware_before_streaming_test.cc',
-            'test/auto_test/standard/hardware_test.cc',
-            'test/auto_test/standard/mixing_test.cc',
-            'test/auto_test/standard/neteq_stats_test.cc',
-            'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc',
-            'test/auto_test/standard/rtp_rtcp_extensions.cc',
-            'test/auto_test/standard/rtp_rtcp_test.cc',
-            'test/auto_test/standard/voe_base_misc_test.cc',
-            'test/auto_test/standard/video_sync_test.cc',
-            'test/auto_test/standard/volume_test.cc',
-            'test/auto_test/resource_manager.cc',
-            'test/auto_test/voe_cpu_test.cc',
-            'test/auto_test/voe_cpu_test.h',
-            'test/auto_test/voe_standard_test.cc',
-            'test/auto_test/voe_standard_test.h',
-            'test/auto_test/voe_stress_test.cc',
-            'test/auto_test/voe_stress_test.h',
-            'test/auto_test/voe_test_defines.h',
-            'test/auto_test/voe_test_interface.h',
-          ],
-          'conditions': [
-            ['OS=="android"', {
-              # some tests are not supported on android yet, exclude these tests.
-              'sources!': [
-                'test/auto_test/standard/hardware_before_streaming_test.cc',
-              ],
-            }],
-          ],
-          # Disable warnings to enable Win64 build, issue 1323.
-          'msvs_disabled_warnings': [
-            4267,  # size_t to int truncation.
-          ],
-        },
         {
           # command line test that should work on linux/mac/win
           'target_name': 'voe_cmd_test',
@@ -220,6 +163,7 @@
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log',
           ],
           'sources': [
             'test/cmd_test/voe_cmd_test.cc',
@@ -227,51 +171,88 @@
         },
       ], # targets
       'conditions': [
-        # TODO(kjellander): Support UseoFMFC on VS2010.
-        # http://code.google.com/p/webrtc/issues/detail?id=709
-        ['OS=="win" and MSVS_VERSION < "2010"', {
+        ['OS!="ios"', {
           'targets': [
-            # WinTest - GUI test for Windows
             {
-              'target_name': 'voe_ui_win_test',
+              'target_name': 'voe_auto_test',
               'type': 'executable',
               'dependencies': [
                 'voice_engine',
+                '<(DEPTH)/testing/gmock.gyp:gmock',
+                '<(DEPTH)/testing/gtest.gyp:gtest',
+                '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
                 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+                '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+                '<(webrtc_root)/test/test.gyp:channel_transport',
                 '<(webrtc_root)/test/test.gyp:test_support',
-              ],
+                '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
+                '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+               ],
               'sources': [
-                'test/win_test/Resource.h',
-                'test/win_test/WinTest.cc',
-                'test/win_test/WinTest.h',
-                'test/win_test/WinTest.rc',
-                'test/win_test/WinTestDlg.cc',
-                'test/win_test/WinTestDlg.h',
-                'test/win_test/res/WinTest.ico',
-                'test/win_test/res/WinTest.rc2',
-                'test/win_test/stdafx.cc',
-                'test/win_test/stdafx.h',
+                'test/auto_test/automated_mode.cc',
+                'test/auto_test/extended/agc_config_test.cc',
+                'test/auto_test/extended/ec_metrics_test.cc',
+                'test/auto_test/fakes/conference_transport.cc',
+                'test/auto_test/fakes/conference_transport.h',
+                'test/auto_test/fakes/loudest_filter.cc',
+                'test/auto_test/fakes/loudest_filter.h',
+                'test/auto_test/fixtures/after_initialization_fixture.cc',
+                'test/auto_test/fixtures/after_initialization_fixture.h',
+                'test/auto_test/fixtures/after_streaming_fixture.cc',
+                'test/auto_test/fixtures/after_streaming_fixture.h',
+                'test/auto_test/fixtures/before_initialization_fixture.cc',
+                'test/auto_test/fixtures/before_initialization_fixture.h',
+                'test/auto_test/fixtures/before_streaming_fixture.cc',
+                'test/auto_test/fixtures/before_streaming_fixture.h',
+                'test/auto_test/standard/audio_processing_test.cc',
+                'test/auto_test/standard/codec_before_streaming_test.cc',
+                'test/auto_test/standard/codec_test.cc',
+                'test/auto_test/standard/dtmf_test.cc',
+                'test/auto_test/standard/external_media_test.cc',
+                'test/auto_test/standard/file_before_streaming_test.cc',
+                'test/auto_test/standard/file_test.cc',
+                'test/auto_test/standard/hardware_before_initializing_test.cc',
+                'test/auto_test/standard/hardware_before_streaming_test.cc',
+                'test/auto_test/standard/hardware_test.cc',
+                'test/auto_test/standard/mixing_test.cc',
+                'test/auto_test/standard/neteq_stats_test.cc',
+                'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc',
+                'test/auto_test/standard/rtp_rtcp_extensions.cc',
+                'test/auto_test/standard/rtp_rtcp_test.cc',
+                'test/auto_test/standard/voe_base_misc_test.cc',
+                'test/auto_test/standard/video_sync_test.cc',
+                'test/auto_test/standard/volume_test.cc',
+                'test/auto_test/resource_manager.cc',
+                'test/auto_test/voe_conference_test.cc',
+                'test/auto_test/voe_cpu_test.cc',
+                'test/auto_test/voe_cpu_test.h',
+                'test/auto_test/voe_output_test.cc',
+                'test/auto_test/voe_standard_test.cc',
+                'test/auto_test/voe_standard_test.h',
+                'test/auto_test/voe_stress_test.cc',
+                'test/auto_test/voe_stress_test.h',
+                'test/auto_test/voe_test_defines.h',
+                'test/auto_test/voe_test_interface.h',
+              ],
+              'conditions': [
+                ['OS=="android"', {
+                  # some tests are not supported on android yet, exclude these tests.
+                  'sources!': [
+                    'test/auto_test/standard/hardware_before_streaming_test.cc',
+                  ],
+                }],
+                ['enable_protobuf==1', {
+                  'defines': [
+                    'ENABLE_RTC_EVENT_LOG',
+                  ],
+                }],
+              ],
+              # Disable warnings to enable Win64 build, issue 1323.
+              'msvs_disabled_warnings': [
+                4267,  # size_t to int truncation.
               ],
-              'configurations': {
-                'Common_Base': {
-                  'msvs_configuration_attributes': {
-                    'conditions': [
-                      ['component=="shared_library"', {
-                        'UseOfMFC': '2',  # Shared DLL
-                      },{
-                        'UseOfMFC': '1',  # Static
-                      }],
-                    ],
-                  },
-                },
-              },
-              'msvs_settings': {
-                'VCLinkerTool': {
-                  'SubSystem': '2',   # Windows
-                },
-              },
             },
-          ],  # targets
+          ],
         }],
         ['OS=="android"', {
           'targets': [
@@ -315,6 +296,6 @@
           ],
         }],
       ],  # conditions
-    }], # include_tests
+    }], # include_tests==1
   ], # conditions
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_defines.h b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_defines.h
index fb43fd889a..8c08614664 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_defines.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_defines.h
@@ -19,7 +19,6 @@
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/logging.h"
 
 // ----------------------------------------------------------------------------
 //  Enumerators
@@ -29,7 +28,7 @@ namespace webrtc {
 
 // Internal buffer size required for mono audio, based on the highest sample
 // rate voice engine supports (10 ms of audio at 192 kHz).
-static const int kMaxMonoDataSizeSamples = 1920;
+static const size_t kMaxMonoDataSizeSamples = 1920;
 
 // VolumeControl
 enum { kMinVolumeLevel = 0 };
@@ -44,36 +43,33 @@ const float kMinOutputVolumePanning = 0.0f;
 const float kMaxOutputVolumePanning = 1.0f;
 
 // DTMF
-enum { kMinDtmfEventCode = 0 };                 // DTMF digit "0"
-enum { kMaxDtmfEventCode = 15 };                // DTMF digit "D"
-enum { kMinTelephoneEventCode = 0 };            // RFC4733 (Section 2.3.1)
-enum { kMaxTelephoneEventCode = 255 };          // RFC4733 (Section 2.3.1)
+enum { kMinDtmfEventCode = 0 };         // DTMF digit "0"
+enum { kMaxDtmfEventCode = 15 };        // DTMF digit "D"
+enum { kMinTelephoneEventCode = 0 };    // RFC4733 (Section 2.3.1)
+enum { kMaxTelephoneEventCode = 255 };  // RFC4733 (Section 2.3.1)
 enum { kMinTelephoneEventDuration = 100 };
-enum { kMaxTelephoneEventDuration = 60000 };    // Actual limit is 2^16
-enum { kMinTelephoneEventAttenuation = 0 };     // 0 dBm0
-enum { kMaxTelephoneEventAttenuation = 36 };    // -36 dBm0
-enum { kMinTelephoneEventSeparationMs = 100 };  // Min delta time between two
-                                                // telephone events
-enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 };       // assumes Ethernet
+enum { kMaxTelephoneEventDuration = 60000 };       // Actual limit is 2^16
+enum { kMinTelephoneEventAttenuation = 0 };        // 0 dBm0
+enum { kMaxTelephoneEventAttenuation = 36 };       // -36 dBm0
+enum { kMinTelephoneEventSeparationMs = 100 };     // Min delta time between two
+                                                   // telephone events
+enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 };  // assumes Ethernet
 
 enum { kVoiceEngineMaxModuleVersionSize = 960 };
 
-// Base
-enum { kVoiceEngineVersionMaxMessageSize = 1024 };
-
 // Audio processing
 const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
 const GainControl::Mode kDefaultAgcMode =
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_GONK) 
-  GainControl::kAdaptiveDigital;
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_GONK)
+    GainControl::kAdaptiveDigital;
 #else
-  GainControl::kAdaptiveAnalog;
+    GainControl::kAdaptiveAnalog;
 #endif
 const bool kDefaultAgcState =
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_GONK) 
-  false;
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_GONK)
+    false;
 #else
-  true;
+    true;
 #endif
 const GainControl::Mode kDefaultRxAgcMode = GainControl::kAdaptiveDigital;
 
@@ -131,53 +127,50 @@ enum { kVoiceEngineMaxRtpExtensionId = 14 };
 //  Macros
 // ----------------------------------------------------------------------------
 
-#define NOT_SUPPORTED(stat)                  \
-  LOG_F(LS_ERROR) << "not supported";        \
-  stat.SetLastError(VE_FUNC_NOT_SUPPORTED);  \
+#define NOT_SUPPORTED(stat)                 \
+  LOG_F(LS_ERROR) << "not supported";       \
+  stat.SetLastError(VE_FUNC_NOT_SUPPORTED); \
   return -1;
 
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-  #include <windows.h>
-  #include <stdio.h>
-  #define DEBUG_PRINT(...)      \
-  {                             \
-    char msg[256];              \
-    sprintf(msg, __VA_ARGS__);  \
-    OutputDebugStringA(msg);    \
+#if (!defined(NDEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+#include <windows.h>
+#include <stdio.h>
+#define DEBUG_PRINT(...)       \
+  {                            \
+    char msg[256];             \
+    sprintf(msg, __VA_ARGS__); \
+    OutputDebugStringA(msg);   \
   }
 #else
-  // special fix for visual 2003
-  #define DEBUG_PRINT(exp)      ((void)0)
-#endif  // defined(_DEBUG) && defined(_WIN32)
+// special fix for visual 2003
+#define DEBUG_PRINT(exp) ((void)0)
+#endif  // !defined(NDEBUG) && defined(_WIN32)
 
-#define CHECK_CHANNEL(channel)  if (CheckChannel(channel) == -1) return -1;
+#define CHECK_CHANNEL(channel)     \
+  if (CheckChannel(channel) == -1) \
+    return -1;
 
 // ----------------------------------------------------------------------------
 //  Inline functions
 // ----------------------------------------------------------------------------
 
-namespace webrtc
-{
+namespace webrtc {
 
-inline int VoEId(int veId, int chId)
-{
-    if (chId == -1)
-    {
-        const int dummyChannel(99);
-        return (int) ((veId << 16) + dummyChannel);
-    }
-    return (int) ((veId << 16) + chId);
+inline int VoEId(int veId, int chId) {
+  if (chId == -1) {
+    const int dummyChannel(99);
+    return (int)((veId << 16) + dummyChannel);
+  }
+  return (int)((veId << 16) + chId);
 }
 
-inline int VoEModuleId(int veId, int chId)
-{
-    return (int) ((veId << 16) + chId);
+inline int VoEModuleId(int veId, int chId) {
+  return (int)((veId << 16) + chId);
 }
 
 // Convert module ID to internal VoE channel ID
-inline int VoEChannelId(int moduleId)
-{
-    return (int) (moduleId & 0xffff);
+inline int VoEChannelId(int moduleId) {
+  return (int)(moduleId & 0xffff);
 }
 
 }  // namespace webrtc
@@ -190,21 +183,21 @@ inline int VoEChannelId(int moduleId)
 
 #if defined(_WIN32)
 
-  #include <windows.h>
+#include <windows.h>
 
-  #pragma comment( lib, "winmm.lib" )
+#pragma comment(lib, "winmm.lib")
 
-  #ifndef WEBRTC_EXTERNAL_TRANSPORT
-    #pragma comment( lib, "ws2_32.lib" )
-  #endif
+#ifndef WEBRTC_EXTERNAL_TRANSPORT
+#pragma comment(lib, "ws2_32.lib")
+#endif
 
 // ----------------------------------------------------------------------------
 //  Defines
 // ----------------------------------------------------------------------------
 
 // Default device for Windows PC
-  #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
-    AudioDeviceModule::kDefaultCommunicationDevice
+#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
+  AudioDeviceModule::kDefaultCommunicationDevice
 
 #endif  // #if (defined(_WIN32)
 
@@ -218,11 +211,11 @@ inline int VoEChannelId(int moduleId)
 #include <sys/socket.h>
 #include <sys/types.h>
 #ifndef QNX
-  #include <linux/net.h>
+#include <linux/net.h>
 #ifndef ANDROID
-  #include <sys/soundcard.h>
-#endif // ANDROID
-#endif // QNX
+#include <sys/soundcard.h>
+#endif  // ANDROID
+#endif  // QNX
 #include <errno.h>
 #include <fcntl.h>
 #include <sched.h>
@@ -250,8 +243,8 @@ inline int VoEChannelId(int moduleId)
 #endif
 #define GetLastError() errno
 #define WSAGetLastError() errno
-#define LPCTSTR const char*
-#define LPCSTR const char*
+#define LPCTSTR const char *
+#define LPCSTR const char *
 #define wsprintf sprintf
 #define TEXT(a) a
 #define _ftprintf fprintf
@@ -289,11 +282,11 @@ inline int VoEChannelId(int moduleId)
 #include <time.h>
 #include <unistd.h>
 #if !defined(WEBRTC_BSD) && !defined(WEBRTC_IOS)
-  #include <CoreServices/CoreServices.h>
-  #include <CoreAudio/CoreAudio.h>
-  #include <AudioToolbox/DefaultAudioOutput.h>
-  #include <AudioToolbox/AudioConverter.h>
-  #include <CoreAudio/HostTime.h>
+#include <CoreServices/CoreServices.h>
+#include <CoreAudio/CoreAudio.h>
+#include <AudioToolbox/DefaultAudioOutput.h>
+#include <AudioToolbox/AudioConverter.h>
+#include <CoreAudio/HostTime.h>
 #endif
 
 #define DWORD unsigned long int
@@ -308,7 +301,7 @@ inline int VoEChannelId(int moduleId)
 #define _stricmp strcasecmp
 #define GetLastError() errno
 #define WSAGetLastError() errno
-#define LPCTSTR const char*
+#define LPCTSTR const char *
 #define wsprintf sprintf
 #define TEXT(a) a
 #define _ftprintf fprintf
@@ -316,11 +309,11 @@ inline int VoEChannelId(int moduleId)
 #define FAR
 #define __cdecl
 #define LPSOCKADDR struct sockaddr *
-#define LPCSTR const char*
+#define LPCSTR const char *
 #define ULONG unsigned long
 
 // Default device for Mac and iPhone
 #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
 #endif  // #if defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
 
-#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
+#endif  // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_fixture.cc b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_fixture.cc
new file mode 100644
index 0000000000..faac8dd48c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_fixture.cc
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/voice_engine/voice_engine_fixture.h"
+
+namespace webrtc {
+
+VoiceEngineFixture::VoiceEngineFixture()
+    : voe_(VoiceEngine::Create()),
+      base_(VoEBase::GetInterface(voe_)),
+      network_(VoENetwork::GetInterface(voe_)) {
+  EXPECT_NE(nullptr, base_);
+  EXPECT_NE(nullptr, network_);
+  EXPECT_EQ(0, base_->RegisterVoiceEngineObserver(observer_));
+}
+
+VoiceEngineFixture::~VoiceEngineFixture() {
+  EXPECT_EQ(2, network_->Release());
+  EXPECT_EQ(0, base_->DeRegisterVoiceEngineObserver());
+  EXPECT_EQ(0, base_->Terminate());
+  EXPECT_EQ(1, base_->Release());
+  EXPECT_TRUE(VoiceEngine::Delete(voe_));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_fixture.h b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_fixture.h
new file mode 100644
index 0000000000..2967678e2f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_fixture.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/test/mock_transport.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+#include "webrtc/voice_engine/mock/mock_voe_observer.h"
+
+namespace webrtc {
+
+class VoiceEngineFixture : public ::testing::Test {
+ protected:
+  VoiceEngineFixture();
+  ~VoiceEngineFixture();
+
+  VoiceEngine* voe_;
+  VoEBase* base_;
+  VoENetwork* network_;
+  MockVoEObserver observer_;
+  FakeAudioDeviceModule adm_;
+  MockTransport transport_;
+};
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc
index 8fa9489d35..2286e1416d 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc
@@ -14,18 +14,17 @@
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
 #endif
-#if !defined(WEBRTC_CHROMIUM_BUILD)
-#include "webrtc/modules/audio_device/android/opensles_input.h"
-#include "webrtc/modules/audio_device/android/opensles_output.h"
-#endif
+#include "webrtc/modules/utility/include/jvm_android.h"
 #endif
 
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
-namespace webrtc
-{
+namespace webrtc {
 
 // Counter to be ensure that we can add a correct ID in all static trace
 // methods. It is not the nicest solution, especially not since we already
@@ -33,32 +32,13 @@ namespace webrtc
 // improvement here.
 static int32_t gVoiceEngineInstanceCounter = 0;
 
-VoiceEngine* GetVoiceEngine(const Config* config, bool owns_config)
-{
-#if (defined _WIN32)
-  HMODULE hmod = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
-
-  if (hmod) {
-    typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
-    PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
-        hmod,"GetVoiceEngine");
-    if (pfn) {
-      VoiceEngine* self = pfn();
-      if (owns_config) {
-        delete config;
-      }
-      return (self);
-    }
+VoiceEngine* GetVoiceEngine(const Config* config, bool owns_config) {
+  VoiceEngineImpl* self = new VoiceEngineImpl(config, owns_config);
+  if (self != NULL) {
+    self->AddRef();  // First reference.  Released in VoiceEngine::Delete.
+    gVoiceEngineInstanceCounter++;
   }
-#endif
-
-    VoiceEngineImpl* self = new VoiceEngineImpl(config, owns_config);
-    if (self != NULL)
-    {
-        self->AddRef();  // First reference.  Released in VoiceEngine::Delete.
-        gVoiceEngineInstanceCounter++;
-    }
-    return self;
+  return self;
 }
 
 int VoiceEngineImpl::AddRef() {
@@ -71,8 +51,7 @@ int VoiceEngineImpl::Release() {
   assert(new_ref >= 0);
   if (new_ref == 0) {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
-                 "VoiceEngineImpl self deleting (voiceEngine=0x%p)",
-                 this);
+                 "VoiceEngineImpl self deleting (voiceEngine=0x%p)", this);
 
     // Clear any pointers before starting destruction. Otherwise worker-
     // threads will still have pointers to a partially destructed object.
@@ -86,6 +65,15 @@ int VoiceEngineImpl::Release() {
   return new_ref;
 }
 
+rtc::scoped_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
+    int channel_id) {
+  RTC_DCHECK(channel_id >= 0);
+  CriticalSectionScoped cs(crit_sec());
+  RTC_DCHECK(statistics().Initialized());
+  return rtc::scoped_ptr<voe::ChannelProxy>(
+      new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
+}
+
 VoiceEngine* VoiceEngine::Create() {
   Config* config = new Config();
   return GetVoiceEngine(config, true);
@@ -95,91 +83,66 @@ VoiceEngine* VoiceEngine::Create(const Config& config) {
   return GetVoiceEngine(&config, false);
 }
 
-int VoiceEngine::SetTraceFilter(unsigned int filter)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
-                 VoEId(gVoiceEngineInstanceCounter, -1),
-                 "SetTraceFilter(filter=0x%x)", filter);
+int VoiceEngine::SetTraceFilter(unsigned int filter) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+               VoEId(gVoiceEngineInstanceCounter, -1),
+               "SetTraceFilter(filter=0x%x)", filter);
 
-    // Remember old filter
-    uint32_t oldFilter = Trace::level_filter();
-    Trace::set_level_filter(filter);
+  // Remember old filter
+  uint32_t oldFilter = Trace::level_filter();
+  Trace::set_level_filter(filter);
 
-    // If previous log was ignored, log again after changing filter
-    if (kTraceNone == oldFilter)
-    {
-        WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
-                     "SetTraceFilter(filter=0x%x)", filter);
-    }
+  // If previous log was ignored, log again after changing filter
+  if (kTraceNone == oldFilter) {
+    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1, "SetTraceFilter(filter=0x%x)",
+                 filter);
+  }
 
-    return 0;
+  return 0;
 }
 
-int VoiceEngine::SetTraceFile(const char* fileNameUTF8,
-                              bool addFileCounter)
-{
-    int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
-                 VoEId(gVoiceEngineInstanceCounter, -1),
-                 "SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)",
-                 fileNameUTF8, addFileCounter);
-    return (ret);
+int VoiceEngine::SetTraceFile(const char* fileNameUTF8, bool addFileCounter) {
+  int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+               VoEId(gVoiceEngineInstanceCounter, -1),
+               "SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)", fileNameUTF8,
+               addFileCounter);
+  return (ret);
 }
 
-int VoiceEngine::SetTraceCallback(TraceCallback* callback)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
-                 VoEId(gVoiceEngineInstanceCounter, -1),
-                 "SetTraceCallback(callback=0x%x)", callback);
-    return (Trace::SetTraceCallback(callback));
+int VoiceEngine::SetTraceCallback(TraceCallback* callback) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
+               VoEId(gVoiceEngineInstanceCounter, -1),
+               "SetTraceCallback(callback=0x%x)", callback);
+  return (Trace::SetTraceCallback(callback));
 }
 
-bool VoiceEngine::Delete(VoiceEngine*& voiceEngine)
-{
-    if (voiceEngine == NULL)
-        return false;
+bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) {
+  if (voiceEngine == NULL)
+    return false;
 
-    VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
-    // Release the reference that was added in GetVoiceEngine.
-    int ref = s->Release();
-    voiceEngine = NULL;
+  VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
+  // Release the reference that was added in GetVoiceEngine.
+  int ref = s->Release();
+  voiceEngine = NULL;
 
-    if (ref != 0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice, -1,
-            "VoiceEngine::Delete did not release the very last reference.  "
-            "%d references remain.", ref);
-    }
+  if (ref != 0) {
+    WEBRTC_TRACE(
+        kTraceWarning, kTraceVoice, -1,
+        "VoiceEngine::Delete did not release the very last reference.  "
+        "%d references remain.",
+        ref);
+  }
 
-    return true;
+  return true;
 }
 
 #if !defined(WEBRTC_CHROMIUM_BUILD)
-int VoiceEngine::SetAndroidObjects(void* javaVM, void* context)
-{
+// TODO(henrika): change types to JavaVM* and jobject instead of void*.
+int VoiceEngine::SetAndroidObjects(void* javaVM, void* context) {
 #ifdef WEBRTC_ANDROID
-#ifdef WEBRTC_ANDROID_OPENSLES
-  typedef AudioDeviceTemplate<OpenSlesInput, OpenSlesOutput>
-      AudioDeviceInstance;
-#endif
-#if !defined(WEBRTC_GONK) && defined(ANDROID)
-  typedef AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>
-      AudioDeviceInstanceJni;
-#endif
-  if (javaVM && context) {
-#if !defined(WEBRTC_GONK) && defined(ANDROID)
-    AudioDeviceInstanceJni::SetAndroidAudioDeviceObjects(javaVM, context);
-#endif
-#ifdef WEBRTC_ANDROID_OPENSLES
-    AudioDeviceInstance::SetAndroidAudioDeviceObjects(javaVM, context);
-#endif
-  } else {
-#if !defined(WEBRTC_GONK) && defined(ANDROID)
-    AudioDeviceInstanceJni::ClearAndroidAudioDeviceObjects();
-#endif
-#ifdef WEBRTC_ANDROID_OPENSLES
-    AudioDeviceInstance::ClearAndroidAudioDeviceObjects();
-#endif
-  }
+  webrtc::JVM::Initialize(reinterpret_cast<JavaVM*>(javaVM),
+                          reinterpret_cast<jobject>(context));
   return 0;
 #else
   return -1;
@@ -187,4 +150,12 @@ int VoiceEngine::SetAndroidObjects(void* javaVM, void* context)
 }
 #endif
 
+std::string VoiceEngine::GetVersionString() {
+  std::string version = "VoiceEngine 4.1.0";
+#ifdef WEBRTC_EXTERNAL_TRANSPORT
+  version += " (External transport build)";
+#endif
+  return version;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h
index 992bc2d83e..f98f881214 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
 #define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/atomic32.h"
+#include "webrtc/system_wrappers/include/atomic32.h"
 #include "webrtc/voice_engine/voe_base_impl.h"
 
 #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
@@ -47,8 +48,10 @@
 #include "webrtc/voice_engine/voe_volume_control_impl.h"
 #endif
 
-namespace webrtc
-{
+namespace webrtc {
+namespace voe {
+class ChannelProxy;
+}  // namespace voe
 
 class VoiceEngineImpl : public voe::SharedData,  // Must be the first base class
                         public VoiceEngine,
@@ -83,11 +86,10 @@ class VoiceEngineImpl : public voe::SharedData,  // Must be the first base class
 #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
                         public VoEVolumeControlImpl,
 #endif
-                        public VoEBaseImpl
-{
-public:
-    VoiceEngineImpl(const Config* config, bool owns_config) :
-        SharedData(*config),
+                        public VoEBaseImpl {
+ public:
+  VoiceEngineImpl(const Config* config, bool owns_config)
+      : SharedData(*config),
 #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
         VoEAudioProcessingImpl(this),
 #endif
@@ -121,24 +123,27 @@ public:
 #endif
         VoEBaseImpl(this),
         _ref_count(0),
-        own_config_(owns_config ? config : NULL)
-    {
-    }
-    virtual ~VoiceEngineImpl()
-    {
-        assert(_ref_count.Value() == 0);
-    }
+        own_config_(owns_config ? config : NULL) {
+  }
+  ~VoiceEngineImpl() override { assert(_ref_count.Value() == 0); }
 
-    int AddRef();
+  int AddRef();
 
-    // This implements the Release() method for all the inherited interfaces.
-    virtual int Release();
+  // This implements the Release() method for all the inherited interfaces.
+  int Release() override;
 
-private:
-    Atomic32 _ref_count;
-    rtc::scoped_ptr<const Config> own_config_;
+  // Backdoor to access a voe::Channel object without a channel ID. This is only
+  // to be used while refactoring the VoE API!
+  virtual rtc::scoped_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id);
+
+ // This is *protected* so that FakeVoiceEngine can inherit from the class and
+ // manipulate the reference count. See: fake_voice_engine.h.
+ protected:
+  Atomic32 _ref_count;
+ private:
+  rtc::scoped_ptr<const Config> own_config_;
 };
 
 }  // namespace webrtc
 
-#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
+#endif  // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
diff --git a/media/webrtc/trunk/webrtc/webrtc.gyp b/media/webrtc/trunk/webrtc/webrtc.gyp
index 1ecc0cd708..31bed3ed9d 100644
--- a/media/webrtc/trunk/webrtc/webrtc.gyp
+++ b/media/webrtc/trunk/webrtc/webrtc.gyp
@@ -8,22 +8,55 @@
 {
   'conditions': [
     ['include_tests==1', {
+      # note: all will be included even if the conditional is false!
       'includes': [
-        'libjingle/xmllite/xmllite_tests.gypi',
-        'libjingle/xmpp/xmpp_tests.gypi',
+        #'libjingle/xmllite/xmllite_tests.gypi',
+        #'libjingle/xmpp/xmpp_tests.gypi',
         'p2p/p2p_tests.gypi',
         'sound/sound_tests.gypi',
         'webrtc_tests.gypi',
       ],
     }],
+    ['enable_protobuf==1', {
+      'targets': [
+        {
+          # This target should only be built if enable_protobuf is defined
+          'target_name': 'rtc_event_log_proto',
+          'type': 'static_library',
+          'sources': ['call/rtc_event_log.proto',],
+          'variables': {
+            'proto_in_dir': 'call',
+            'proto_out_dir': 'webrtc/call',
+          },
+        'includes': ['build/protoc.gypi'],
+        },
+      ],
+    }],
+    ['include_tests==1 and enable_protobuf==1', {
+      'targets': [
+        {
+          'target_name': 'rtc_event_log2rtp_dump',
+          'type': 'executable',
+          'sources': ['call/rtc_event_log2rtp_dump.cc',],
+          'dependencies': [
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            'rtc_event_log',
+            'rtc_event_log_proto',
+            'test/test.gyp:rtp_test_utils'
+          ],
+        },
+      ],
+    }],
   ],
   'includes': [
     'build/common.gypi',
+    'audio/webrtc_audio.gypi',
+    'call/webrtc_call.gypi',
     'video/webrtc_video.gypi',
   ],
   'variables': {
     'webrtc_all_dependencies': [
-      'base/base.gyp:*',
+      'base/base.gyp:rtc_base_approved',
       'sound/sound.gyp:*',
       'common.gyp:*',
       'common_audio/common_audio.gyp:*',
@@ -32,7 +65,6 @@
       'p2p/p2p.gyp:*',
       'system_wrappers/system_wrappers.gyp:*',
       'tools/tools.gyp:*',
-      'video_engine/video_engine.gyp:*',
       'voice_engine/voice_engine.gyp:*',
       '<(webrtc_vp8_dir)/vp8.gyp:*',
       '<(webrtc_vp9_dir)/vp9.gyp:*',
@@ -44,46 +76,52 @@
       'type': 'none',
       'dependencies': [
         '<@(webrtc_all_dependencies)',
-        'webrtc',
+        'webrtc_lib',
       ],
       'conditions': [
         ['include_tests==1', {
           'dependencies': [
             'common_video/common_video_unittests.gyp:*',
+            'rtc_unittests',
             'system_wrappers/system_wrappers_tests.gyp:*',
             'test/metrics.gyp:*',
             'test/test.gyp:*',
-            'test/webrtc_test_common.gyp:webrtc_test_common_unittests',
+            'test/webrtc_test_common.gyp:*',
             'webrtc_tests',
-            'rtc_unittests',
           ],
         }],
       ],
     },
     {
-      # TODO(pbos): This is intended to contain audio parts as well as soon as
-      #             VoiceEngine moves to the same new API format.
-      'target_name': 'webrtc',
+      'target_name': 'webrtc_lib',
       'type': 'static_library',
       'sources': [
+        'audio_receive_stream.h',
+        'audio_send_stream.h',
+        'audio_state.h',
         'call.h',
         'config.h',
-        'experiments.h',
         'frame_callback.h',
+        'stream.h',
         'transport.h',
         'video_receive_stream.h',
         'video_renderer.h',
         'video_send_stream.h',
 
+        '<@(webrtc_audio_sources)',
+        '<@(webrtc_call_sources)',
         '<@(webrtc_video_sources)',
       ],
       'dependencies': [
         'common.gyp:*',
+        '<@(webrtc_audio_dependencies)',
+        '<@(webrtc_call_dependencies)',
         '<@(webrtc_video_dependencies)',
+        'rtc_event_log',
       ],
       'conditions': [
-        # TODO(andresp): Chromium libpeerconnection should link directly with
-        # this and no if conditions should be needed on webrtc build files.
+        # TODO(andresp): Chromium should link directly with this and no if
+        # conditions should be needed on webrtc build files.
         ['build_with_chromium==1', {
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture',
@@ -92,5 +130,26 @@
         }],
       ],
     },
+    {
+      'target_name': 'rtc_event_log',
+      'type': 'static_library',
+      'sources': [
+        'call/rtc_event_log.cc',
+        'call/rtc_event_log.h',
+      ],
+      'conditions': [
+        # If enable_protobuf is defined, we want to compile the protobuf
+        # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
+        ['enable_protobuf==1', {
+          'dependencies': [
+            'rtc_event_log_proto',
+          ],
+          'defines': [
+            'ENABLE_RTC_EVENT_LOG',
+          ],
+        }],
+      ],
+    },
+
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/webrtc_examples.gyp b/media/webrtc/trunk/webrtc/webrtc_examples.gyp
index 5d18aa64bf..44b2ca35be 100644
--- a/media/webrtc/trunk/webrtc/webrtc_examples.gyp
+++ b/media/webrtc/trunk/webrtc/webrtc_examples.gyp
@@ -1,4 +1,4 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+# Copyright (c) 2012 The WebRTC Project Authors. All rights reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
@@ -6,147 +6,415 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 {
-  'includes': ['build/common.gypi'],
-  'targets': [],
+  'includes': [
+    '../talk/build/common.gypi',
+  ],
+  'targets': [
+    {
+      'target_name': 'relayserver',
+      'type': 'executable',
+      'dependencies': [
+        '../talk/libjingle.gyp:libjingle',
+        '../talk/libjingle.gyp:libjingle_p2p',
+      ],
+      'sources': [
+        'examples/relayserver/relayserver_main.cc',
+      ],
+    },  # target relayserver
+    {
+      'target_name': 'stunserver',
+      'type': 'executable',
+      'dependencies': [
+        '../talk/libjingle.gyp:libjingle',
+        '../talk/libjingle.gyp:libjingle_p2p',
+      ],
+      'sources': [
+        'examples/stunserver/stunserver_main.cc',
+      ],
+    },  # target stunserver
+    {
+      'target_name': 'turnserver',
+      'type': 'executable',
+      'dependencies': [
+        '../talk/libjingle.gyp:libjingle',
+        '../talk/libjingle.gyp:libjingle_p2p',
+      ],
+      'sources': [
+        'examples/turnserver/turnserver_main.cc',
+      ],
+    },  # target turnserver
+    {
+      'target_name': 'peerconnection_server',
+      'type': 'executable',
+      'sources': [
+        'examples/peerconnection/server/data_socket.cc',
+        'examples/peerconnection/server/data_socket.h',
+        'examples/peerconnection/server/main.cc',
+        'examples/peerconnection/server/peer_channel.cc',
+        'examples/peerconnection/server/peer_channel.h',
+        'examples/peerconnection/server/utils.cc',
+        'examples/peerconnection/server/utils.h',
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '../talk/libjingle.gyp:libjingle',
+      ],
+      # TODO(ronghuawu): crbug.com/167187 fix size_t to int truncations.
+      'msvs_disabled_warnings': [ 4309, ],
+    }, # target peerconnection_server
+  ],
   'conditions': [
+    ['OS=="linux" or OS=="win"', {
+      'targets': [
+        {
+          'target_name': 'peerconnection_client',
+          'type': 'executable',
+          'sources': [
+            'examples/peerconnection/client/conductor.cc',
+            'examples/peerconnection/client/conductor.h',
+            'examples/peerconnection/client/defaults.cc',
+            'examples/peerconnection/client/defaults.h',
+            'examples/peerconnection/client/peer_connection_client.cc',
+            'examples/peerconnection/client/peer_connection_client.h',
+          ],
+          'dependencies': [
+            '../talk/libjingle.gyp:libjingle_peerconnection',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
+            '<@(libjingle_tests_additional_deps)',
+          ],
+          'conditions': [
+            ['build_json==1', {
+              'dependencies': [
+                '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+              ],
+            }],
+            # TODO(ronghuawu): Move these files to a win/ directory then they
+            # can be excluded automatically.
+            ['OS=="win"', {
+              'sources': [
+                'examples/peerconnection/client/flagdefs.h',
+                'examples/peerconnection/client/main.cc',
+                'examples/peerconnection/client/main_wnd.cc',
+                'examples/peerconnection/client/main_wnd.h',
+              ],
+              'msvs_settings': {
+                'VCLinkerTool': {
+                 'SubSystem': '2',  # Windows
+                },
+              },
+            }],  # OS=="win"
+            ['OS=="win" and clang==1', {
+              'msvs_settings': {
+                'VCCLCompilerTool': {
+                  'AdditionalOptions': [
+                    # Disable warnings failing when compiling with Clang on Windows.
+                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+                    '-Wno-reorder',
+                    '-Wno-unused-function',
+                  ],
+                },
+              },
+            }], # OS=="win" and clang==1
+            ['OS=="linux"', {
+              'sources': [
+                'examples/peerconnection/client/linux/main.cc',
+                'examples/peerconnection/client/linux/main_wnd.cc',
+                'examples/peerconnection/client/linux/main_wnd.h',
+              ],
+              'cflags': [
+                '<!@(pkg-config --cflags glib-2.0 gobject-2.0 gtk+-2.0)',
+              ],
+              'link_settings': {
+                'ldflags': [
+                  '<!@(pkg-config --libs-only-L --libs-only-other glib-2.0'
+                      ' gobject-2.0 gthread-2.0 gtk+-2.0)',
+                ],
+                'libraries': [
+                  '<!@(pkg-config --libs-only-l glib-2.0 gobject-2.0'
+                      ' gthread-2.0 gtk+-2.0)',
+                  '-lX11',
+                  '-lXcomposite',
+                  '-lXext',
+                  '-lXrender',
+                ],
+              },
+            }],  # OS=="linux"
+          ],  # conditions
+        },  # target peerconnection_client
+      ], # targets
+    }],  # OS=="linux" or OS=="win"
+
+    ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
+      'targets': [
+        {
+          'target_name': 'apprtc_common',
+          'type': 'static_library',
+          'dependencies': [
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
+            '../talk/libjingle.gyp:libjingle_peerconnection_objc',
+          ],
+          'sources': [
+            'examples/objc/AppRTCDemo/common/ARDUtilities.h',
+            'examples/objc/AppRTCDemo/common/ARDUtilities.m',
+          ],
+          'include_dirs': [
+            'examples/objc/AppRTCDemo/common',
+          ],
+          'direct_dependent_settings': {
+            'include_dirs': [
+              'examples/objc/AppRTCDemo/common',
+            ],
+          },
+          'conditions': [
+            ['OS=="mac"', {
+              'xcode_settings': {
+                'MACOSX_DEPLOYMENT_TARGET' : '10.8',
+              },
+            }],
+          ],
+          'link_settings': {
+            'xcode_settings': {
+              'OTHER_LDFLAGS': [
+                '-framework QuartzCore',
+              ],
+            },
+          },
+        },
+        {
+          'target_name': 'apprtc_signaling',
+          'type': 'static_library',
+          'dependencies': [
+            'apprtc_common',
+            '../talk/libjingle.gyp:libjingle_peerconnection_objc',
+            'socketrocket',
+          ],
+          'sources': [
+            'examples/objc/AppRTCDemo/ARDAppClient.h',
+            'examples/objc/AppRTCDemo/ARDAppClient.m',
+            'examples/objc/AppRTCDemo/ARDAppClient+Internal.h',
+            'examples/objc/AppRTCDemo/ARDAppEngineClient.h',
+            'examples/objc/AppRTCDemo/ARDAppEngineClient.m',
+            'examples/objc/AppRTCDemo/ARDBitrateTracker.h',
+            'examples/objc/AppRTCDemo/ARDBitrateTracker.m',
+            'examples/objc/AppRTCDemo/ARDCEODTURNClient.h',
+            'examples/objc/AppRTCDemo/ARDCEODTURNClient.m',
+            'examples/objc/AppRTCDemo/ARDJoinResponse.h',
+            'examples/objc/AppRTCDemo/ARDJoinResponse.m',
+            'examples/objc/AppRTCDemo/ARDJoinResponse+Internal.h',
+            'examples/objc/AppRTCDemo/ARDMessageResponse.h',
+            'examples/objc/AppRTCDemo/ARDMessageResponse.m',
+            'examples/objc/AppRTCDemo/ARDMessageResponse+Internal.h',
+            'examples/objc/AppRTCDemo/ARDRoomServerClient.h',
+            'examples/objc/AppRTCDemo/ARDSDPUtils.h',
+            'examples/objc/AppRTCDemo/ARDSDPUtils.m',
+            'examples/objc/AppRTCDemo/ARDSignalingChannel.h',
+            'examples/objc/AppRTCDemo/ARDSignalingMessage.h',
+            'examples/objc/AppRTCDemo/ARDSignalingMessage.m',
+            'examples/objc/AppRTCDemo/ARDStatsBuilder.h',
+            'examples/objc/AppRTCDemo/ARDStatsBuilder.m',
+            'examples/objc/AppRTCDemo/ARDTURNClient.h',
+            'examples/objc/AppRTCDemo/ARDWebSocketChannel.h',
+            'examples/objc/AppRTCDemo/ARDWebSocketChannel.m',
+            'examples/objc/AppRTCDemo/RTCICECandidate+JSON.h',
+            'examples/objc/AppRTCDemo/RTCICECandidate+JSON.m',
+            'examples/objc/AppRTCDemo/RTCICEServer+JSON.h',
+            'examples/objc/AppRTCDemo/RTCICEServer+JSON.m',
+            'examples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h',
+            'examples/objc/AppRTCDemo/RTCMediaConstraints+JSON.m',
+            'examples/objc/AppRTCDemo/RTCSessionDescription+JSON.h',
+            'examples/objc/AppRTCDemo/RTCSessionDescription+JSON.m',
+          ],
+          'include_dirs': [
+            'examples/objc/AppRTCDemo',
+          ],
+          'direct_dependent_settings': {
+            'include_dirs': [
+              'examples/objc/AppRTCDemo',
+            ],
+          },
+          'export_dependent_settings': [
+            '../talk/libjingle.gyp:libjingle_peerconnection_objc',
+          ],
+          'conditions': [
+            ['OS=="mac"', {
+              'xcode_settings': {
+                'MACOSX_DEPLOYMENT_TARGET' : '10.8',
+              },
+            }],
+          ],
+        },
+        {
+          'target_name': 'AppRTCDemo',
+          'type': 'executable',
+          'product_name': 'AppRTCDemo',
+          'mac_bundle': 1,
+          'dependencies': [
+            'apprtc_common',
+            'apprtc_signaling',
+          ],
+          'conditions': [
+            ['OS=="ios"', {
+              'mac_bundle_resources': [
+                'examples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png',
+                'examples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png',
+                'examples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png',
+                'examples/objc/AppRTCDemo/ios/resources/Roboto-Regular.ttf',
+                'examples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp.png',
+                'examples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp@2x.png',
+                'examples/objc/AppRTCDemo/ios/resources/ic_clear_black_24dp.png',
+                'examples/objc/AppRTCDemo/ios/resources/ic_clear_black_24dp@2x.png',
+                'examples/objc/AppRTCDemo/ios/resources/ic_switch_video_black_24dp.png',
+                'examples/objc/AppRTCDemo/ios/resources/ic_switch_video_black_24dp@2x.png',
+                'examples/objc/Icon.png',
+              ],
+              'sources': [
+                'examples/objc/AppRTCDemo/ios/ARDAppDelegate.h',
+                'examples/objc/AppRTCDemo/ios/ARDAppDelegate.m',
+                'examples/objc/AppRTCDemo/ios/ARDMainView.h',
+                'examples/objc/AppRTCDemo/ios/ARDMainView.m',
+                'examples/objc/AppRTCDemo/ios/ARDMainViewController.h',
+                'examples/objc/AppRTCDemo/ios/ARDMainViewController.m',
+                'examples/objc/AppRTCDemo/ios/ARDStatsView.h',
+                'examples/objc/AppRTCDemo/ios/ARDStatsView.m',
+                'examples/objc/AppRTCDemo/ios/ARDVideoCallView.h',
+                'examples/objc/AppRTCDemo/ios/ARDVideoCallView.m',
+                'examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h',
+                'examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m',
+                'examples/objc/AppRTCDemo/ios/AppRTCDemo-Prefix.pch',
+                'examples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.h',
+                'examples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.m',
+                'examples/objc/AppRTCDemo/ios/main.m',
+              ],
+              'xcode_settings': {
+                'INFOPLIST_FILE': 'examples/objc/AppRTCDemo/ios/Info.plist',
+              },
+            }],
+            ['OS=="mac"', {
+              'sources': [
+                'examples/objc/AppRTCDemo/mac/APPRTCAppDelegate.h',
+                'examples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m',
+                'examples/objc/AppRTCDemo/mac/APPRTCViewController.h',
+                'examples/objc/AppRTCDemo/mac/APPRTCViewController.m',
+                'examples/objc/AppRTCDemo/mac/main.m',
+              ],
+              'xcode_settings': {
+                'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
+                'INFOPLIST_FILE': 'examples/objc/AppRTCDemo/mac/Info.plist',
+                'MACOSX_DEPLOYMENT_TARGET' : '10.8',
+                'OTHER_LDFLAGS': [
+                  '-framework AVFoundation',
+                ],
+              },
+            }],
+            ['target_arch=="ia32"', {
+              'dependencies' : [
+                '<(DEPTH)/testing/iossim/iossim.gyp:iossim#host',
+              ],
+            }],
+          ],
+        },  # target AppRTCDemo
+        {
+          # TODO(tkchin): move this into the real third party location and
+          # have it mirrored on chrome infra.
+          'target_name': 'socketrocket',
+          'type': 'static_library',
+          'sources': [
+            'examples/objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.h',
+            'examples/objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.m',
+          ],
+          'conditions': [
+            ['OS=="mac"', {
+              'xcode_settings': {
+                # SocketRocket autosynthesizes some properties. Disable the
+                # warning so we can compile successfully.
+                'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
+                'MACOSX_DEPLOYMENT_TARGET' : '10.8',
+                # SRWebSocket.m uses code with partial availability.
+                # https://code.google.com/p/webrtc/issues/detail?id=4695
+                'WARNING_CFLAGS!': ['-Wpartial-availability'],
+              },
+            }],
+          ],
+          'direct_dependent_settings': {
+            'include_dirs': [
+              'examples/objc/AppRTCDemo/third_party/SocketRocket',
+            ],
+          },
+          'xcode_settings': {
+            'CLANG_ENABLE_OBJC_ARC': 'YES',
+            'WARNING_CFLAGS': [
+              '-Wno-deprecated-declarations',
+              '-Wno-nonnull',
+            ],
+          },
+          'link_settings': {
+            'xcode_settings': {
+              'OTHER_LDFLAGS': [
+                '-framework CFNetwork',
+                '-licucore',
+              ],
+            },
+          }
+        },  # target socketrocket
+      ],  # targets
+    }],  # OS=="ios" or (OS=="mac" and target_arch!="ia32")
+
     ['OS=="android"', {
       'targets': [
         {
-          'target_name': 'libwebrtcdemo-jni',
-          'type': 'loadable_module',
+          'target_name': 'AppRTCDemo',
+          'type': 'none',
           'dependencies': [
-            '<(DEPTH)/third_party/icu/icu.gyp:icuuc',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
-            '<(webrtc_root)/test/test.gyp:channel_transport',
-            '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-          ],
-          'sources': [
-            'examples/android/media_demo/jni/jni_helpers.cc',
-            'examples/android/media_demo/jni/on_load.cc',
-            'examples/android/media_demo/jni/voice_engine_jni.cc',
+            '../talk/libjingle.gyp:libjingle_peerconnection_java',
           ],
           'variables': {
-            # This library uses native JNI exports; tell GYP so that the
-            # required symbols will be kept.
-            'use_native_jni_exports': 1,
+            'apk_name': 'AppRTCDemo',
+            'java_in_dir': 'examples/androidapp',
+            'has_java_resources': 1,
+            'resource_dir': 'examples/androidapp/res',
+            'R_package': 'org.appspot.apprtc',
+            'R_package_relpath': 'org/appspot/apprtc',
+            'input_jars_paths': [
+              'examples/androidapp/third_party/autobanh/autobanh.jar',
+             ],
+            'library_dexed_jars_paths': [
+              'examples/androidapp/third_party/autobanh/autobanh.jar',
+             ],
+            'native_lib_target': 'libjingle_peerconnection_so',
+            'add_to_dependents_classpaths':1,
           },
-          'link_settings': {
-            'libraries': [
-              '-llog',
-              '-lGLESv2',
-              '-lOpenSLES',
-            ],
-          }
-        },
+          'includes': [ '../build/java_apk.gypi' ],
+        },  # target AppRTCDemo
+
         {
-          'target_name': 'WebRTCDemo',
+          # AppRTCDemo creates a .jar as a side effect. Any java targets
+          # that need that .jar in their classpath should depend on this target,
+          # AppRTCDemo_apk. Dependents of AppRTCDemo_apk receive its
+          # jar path in the variable 'apk_output_jar_path'.
+          # This target should only be used by targets which instrument
+          # AppRTCDemo_apk.
+          'target_name': 'AppRTCDemo_apk',
           'type': 'none',
           'dependencies': [
-            'libwebrtcdemo-jni',
-            '<(modules_java_gyp_path):*',
-          ],
-          'actions': [
-            {
-              # TODO(yujie.mao): Convert building of the demo to a proper GYP
-              # target so this action is not needed once chromium's
-              # apk-building machinery can be used. (crbug.com/225101)
-              'action_name': 'build_webrtcdemo_apk',
-              'variables': {
-                'android_webrtc_demo_root': '<(webrtc_root)/examples/android/media_demo',
-                'ant_log': '../../../<(INTERMEDIATE_DIR)/ant.log', # ../../.. to compensate for the cd below.
-              },
-              'inputs' : [
-                '<(PRODUCT_DIR)/lib.java/audio_device_module_java.jar',
-                '<(PRODUCT_DIR)/libwebrtcdemo-jni.so',
-                '<!@(find <(android_webrtc_demo_root)/src -name "*.java")',
-                '<!@(find <(android_webrtc_demo_root)/res -type f)',
-                '<(android_webrtc_demo_root)/AndroidManifest.xml',
-                '<(android_webrtc_demo_root)/build.xml',
-                '<(android_webrtc_demo_root)/project.properties',
-              ],
-              'outputs': ['<(PRODUCT_DIR)/WebRTCDemo-debug.apk'],
-              'action': [
-                'bash', '-ec',
-                'rm -fr <(_outputs) <(android_webrtc_demo_root)/{bin,libs,gen,obj} && '
-                'mkdir -p <(INTERMEDIATE_DIR) && ' # Must happen _before_ the cd below
-                'mkdir -p <(android_webrtc_demo_root)/libs/<(android_app_abi) && '
-                'cp <(PRODUCT_DIR)/lib.java/audio_device_module_java.jar <(android_webrtc_demo_root)/libs/ &&'
-                '<(android_strip) -o <(android_webrtc_demo_root)/libs/<(android_app_abi)/libwebrtcdemo-jni.so <(PRODUCT_DIR)/libwebrtcdemo-jni.so && '
-                'cd <(android_webrtc_demo_root) && '
-                '{ ANDROID_SDK_ROOT=<(android_sdk_root) '
-                'ant debug > <(ant_log) 2>&1 || '
-                '  { cat <(ant_log) ; exit 1; } } && '
-                'cd - > /dev/null && '
-                'cp <(android_webrtc_demo_root)/bin/WebRTCDemo-debug.apk <(_outputs)'
-              ],
-            },
-          ],
-        },
+             'AppRTCDemo',
+           ],
+           'includes': [ '../build/apk_fake_jar.gypi' ],
+        },  # target AppRTCDemo_apk
+
         {
-          'target_name': 'libopensl-demo-jni',
-          'type': 'loadable_module',
-          'dependencies': [
-            '<(webrtc_root)/modules/modules.gyp:audio_device',
-          ],
-          'sources': [
-            'examples/android/opensl_loopback/jni/opensl_runner.cc',
-            'examples/android/opensl_loopback/fake_audio_device_buffer.cc',
-          ],
-          'link_settings': {
-            'libraries': [
-              '-llog',
-              '-lOpenSLES',
-            ],
-          },
-        },
-        {
-          'target_name': 'OpenSlDemo',
+          'target_name': 'AppRTCDemoTest',
           'type': 'none',
           'dependencies': [
-            'libopensl-demo-jni',
-            '<(modules_java_gyp_path):*',
-          ],
-          'actions': [
-            {
-              # TODO(henrik): Convert building of the demo to a proper GYP
-              # target so this action is not needed once chromium's
-              # apk-building machinery can be used. (crbug.com/225101)
-              'action_name': 'build_opensldemo_apk',
-              'variables': {
-                'android_opensl_demo_root': '<(webrtc_root)/examples/android/opensl_loopback',
-                'ant_log': '../../../<(INTERMEDIATE_DIR)/ant.log', # ../../.. to compensate for the cd below.
-              },
-              'inputs' : [
-                '<(PRODUCT_DIR)/lib.java/audio_device_module_java.jar',
-                '<(PRODUCT_DIR)/libopensl-demo-jni.so',
-                '<!@(find <(android_opensl_demo_root)/src -name "*.java")',
-                '<!@(find <(android_opensl_demo_root)/res -name "*.xml")',
-                '<!@(find <(android_opensl_demo_root)/res -name "*.png")',
-                '<(android_opensl_demo_root)/AndroidManifest.xml',
-                '<(android_opensl_demo_root)/build.xml',
-                '<(android_opensl_demo_root)/project.properties',
-              ],
-              'outputs': ['<(PRODUCT_DIR)/OpenSlDemo-debug.apk'],
-              'action': [
-                'bash', '-ec',
-                'rm -fr <(_outputs) <(android_opensl_demo_root)/{bin,libs,gen,obj} && '
-                'mkdir -p <(android_opensl_demo_root)/libs/<(android_app_abi) && '
-                'mkdir -p <(INTERMEDIATE_DIR) && ' # Must happen _before_ the cd below
-                '<(android_strip) -o <(android_opensl_demo_root)/libs/<(android_app_abi)/libopensl-demo-jni.so <(PRODUCT_DIR)/libopensl-demo-jni.so && '
-                'cp <(PRODUCT_DIR)/lib.java/audio_device_module_java.jar <(android_opensl_demo_root)/libs/ &&'
-                'cd <(android_opensl_demo_root) && '
-                '{ ANDROID_SDK_ROOT=<(android_sdk_root) '
-                'ant debug > <(ant_log) 2>&1 || '
-                '  { cat <(ant_log) ; exit 1; } } && '
-                'cd - > /dev/null && '
-                'cp <(android_opensl_demo_root)/bin/OpenSlDemo-debug.apk <(_outputs)'
-              ],
-            },
-          ],
+            'AppRTCDemo_apk',
+           ],
+          'variables': {
+            'apk_name': 'AppRTCDemoTest',
+            'java_in_dir': 'examples/androidtests',
+            'is_test_apk': 1,
+          },
+          'includes': [ '../build/java_apk.gypi' ],
         },
-      ],
-    }],
+      ],  # targets
+    }],  # OS=="android"
   ],
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate b/media/webrtc/trunk/webrtc/webrtc_nonparallel_tests.isolate
similarity index 62%
rename from media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate
rename to media/webrtc/trunk/webrtc/webrtc_nonparallel_tests.isolate
index 2579a20b20..0e13d7b3ba 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate
+++ b/media/webrtc/trunk/webrtc/webrtc_nonparallel_tests.isolate
@@ -1,4 +1,4 @@
-# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
@@ -10,13 +10,10 @@
     ['OS=="linux" or OS=="mac" or OS=="win"', {
       'variables': {
         'command': [
-          '<(DEPTH)/testing/test_env.py',
-          '<(PRODUCT_DIR)/vie_auto_test<(EXECUTABLE_SUFFIX)',
+          '<(PRODUCT_DIR)/webrtc_nonparallel_tests<(EXECUTABLE_SUFFIX)',
         ],
         'files': [
-          '<(DEPTH)/DEPS',
-          '<(DEPTH)/testing/test_env.py',
-          '<(PRODUCT_DIR)/vie_auto_test<(EXECUTABLE_SUFFIX)',
+          '<(PRODUCT_DIR)/webrtc_nonparallel_tests<(EXECUTABLE_SUFFIX)',
         ],
       },
     }],
diff --git a/media/webrtc/trunk/webrtc/webrtc_perf_tests.isolate b/media/webrtc/trunk/webrtc/webrtc_perf_tests.isolate
index 188f112ba8..e356839d5e 100644
--- a/media/webrtc/trunk/webrtc/webrtc_perf_tests.isolate
+++ b/media/webrtc/trunk/webrtc/webrtc_perf_tests.isolate
@@ -17,6 +17,8 @@
           '<(DEPTH)/resources/photo_1850_1110.yuv',
           '<(DEPTH)/resources/presentation_1850_1110.yuv',
           '<(DEPTH)/resources/web_screenshot_1850_1110.yuv',
+          '<(DEPTH)/resources/google-wifi-3mbps.rx',
+          '<(DEPTH)/resources/verizon4g-downlink.rx',
         ],
       },
     }],
diff --git a/media/webrtc/trunk/webrtc/webrtc_tests.gypi b/media/webrtc/trunk/webrtc/webrtc_tests.gypi
index c0b3c038f0..e0bf276d4f 100644
--- a/media/webrtc/trunk/webrtc/webrtc_tests.gypi
+++ b/media/webrtc/trunk/webrtc/webrtc_tests.gypi
@@ -17,12 +17,14 @@
         'libjingle/xmllite/xmllite.gyp:rtc_xmllite',
         'libjingle/xmpp/xmpp.gyp:rtc_xmpp',
         'p2p/p2p.gyp:rtc_p2p',
+        'p2p/p2p.gyp:libstunprober',
         'rtc_p2p_unittest',
         'rtc_sound_tests',
         'rtc_xmllite_unittest',
         'rtc_xmpp_unittest',
         'sound/sound.gyp:rtc_sound',
         '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/testing/gmock.gyp:gmock',
       ],
       'conditions': [
         ['OS=="android"', {
@@ -30,6 +32,11 @@
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
           ],
         }],
+        ['OS=="ios"', {
+          'dependencies': [
+            'api/api_tests.gyp:rtc_api_objc_test',
+          ]
+        }]
       ],
     },
     {
@@ -40,22 +47,30 @@
         'video_loopback',
         'video_replay',
         'webrtc_perf_tests',
+        'webrtc_nonparallel_tests',
       ],
     },
     {
-      'target_name': 'loopback_base',
+      'target_name': 'video_quality_test',
       'type': 'static_library',
       'sources': [
-        'video/loopback.cc',
-        'video/loopback.h',
+        'video/video_quality_test.cc',
+        'video/video_quality_test.h',
       ],
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
         '<(webrtc_root)/modules/modules.gyp:video_render',
+        '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         'webrtc',
       ],
+      'conditions': [
+        ['OS=="android"', {
+          'dependencies!': [
+            '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
+          ],
+        }],
+      ],
     },
     {
       'target_name': 'video_loopback',
@@ -74,7 +89,7 @@
         }],
       ],
       'dependencies': [
-        'loopback_base',
+        'video_quality_test',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         'test/webrtc_test_common.gyp:webrtc_test_common',
@@ -83,7 +98,7 @@
         'webrtc',
       ],
     },
-        {
+    {
       'target_name': 'screenshare_loopback',
       'type': 'executable',
       'sources': [
@@ -100,7 +115,7 @@
         }],
       ],
       'dependencies': [
-        'loopback_base',
+        'video_quality_test',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         'test/webrtc_test_common.gyp:webrtc_test_common',
@@ -137,18 +152,33 @@
       ],
     },
     {
-      # TODO(pbos): Rename target to webrtc_tests or rtc_tests, this target is
-      # not meant to only include video.
+      # TODO(solenberg): Rename to webrtc_call_tests.
       'target_name': 'video_engine_tests',
       'type': '<(gtest_target_type)',
       'sources': [
+        'audio/audio_receive_stream_unittest.cc',
+        'audio/audio_send_stream_unittest.cc',
+        'audio/audio_state_unittest.cc',
+        'call/bitrate_allocator_unittest.cc',
+        'call/bitrate_estimator_tests.cc',
+        'call/call_unittest.cc',
+        'call/packet_injection_tests.cc',
         'test/common_unittest.cc',
         'test/testsupport/metrics/video_metrics_unittest.cc',
-        'tools/agc/agc_manager_unittest.cc',
-        'video/bitrate_estimator_tests.cc',
+        'video/call_stats_unittest.cc',
+        'video/encoder_state_feedback_unittest.cc',
         'video/end_to_end_tests.cc',
+        'video/overuse_frame_detector_unittest.cc',
+        'video/payload_router_unittest.cc',
+        'video/report_block_stats_unittest.cc',
         'video/send_statistics_proxy_unittest.cc',
+        'video/stream_synchronization_unittest.cc',
+        'video/video_capture_input_unittest.cc',
+        'video/video_decoder_unittest.cc',
+        'video/video_encoder_unittest.cc',
         'video/video_send_stream_tests.cc',
+        'video/vie_codec_unittest.cc',
+        'video/vie_remb_unittest.cc',
       ],
       'dependencies': [
         '<(DEPTH)/testing/gmock.gyp:gmock',
@@ -162,7 +192,55 @@
         'test/metrics.gyp:metrics',
         'test/test.gyp:test_main',
         'test/webrtc_test_common.gyp:webrtc_test_common',
-        'tools/tools.gyp:agc_manager',
+        'webrtc',
+      ],
+      'conditions': [
+        ['OS=="android"', {
+          'dependencies': [
+            '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+          ],
+        }],
+        ['enable_protobuf==1', {
+          'defines': [
+            'ENABLE_RTC_EVENT_LOG',
+          ],
+          'dependencies': [
+            'webrtc.gyp:rtc_event_log',
+            'webrtc.gyp:rtc_event_log_proto',
+          ],
+          'sources': [
+            'call/rtc_event_log_unittest.cc',
+          ],
+        }],
+      ],
+    },
+    {
+      'target_name': 'webrtc_perf_tests',
+      'type': '<(gtest_target_type)',
+      'sources': [
+        'call/call_perf_tests.cc',
+        'call/rampup_tests.cc',
+        'call/rampup_tests.h',
+        'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
+        'modules/audio_processing/audio_processing_performance_unittest.cc',
+        'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
+        'video/full_stack.cc',
+      ],
+      'dependencies': [
+        '<(DEPTH)/testing/gmock.gyp:gmock',
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(webrtc_root)/modules/modules.gyp:audio_processing',
+        '<(webrtc_root)/modules/modules.gyp:audioproc_test_utils',
+        '<(webrtc_root)/modules/modules.gyp:video_capture',
+        '<(webrtc_root)/test/test.gyp:channel_transport',
+        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+        'video_quality_test',
+        'modules/modules.gyp:neteq_test_support',
+        'modules/modules.gyp:bwe_simulator',
+        'modules/modules.gyp:rtp_rtcp',
+        'test/test.gyp:test_main',
+        'test/webrtc_test_common.gyp:webrtc_test_common',
+        'test/webrtc_test_common.gyp:webrtc_test_renderer',
         'webrtc',
       ],
       'conditions': [
@@ -174,28 +252,23 @@
       ],
     },
     {
-      'target_name': 'webrtc_perf_tests',
+      'target_name': 'webrtc_nonparallel_tests',
       'type': '<(gtest_target_type)',
       'sources': [
-        'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
-        'tools/agc/agc_manager_integrationtest.cc',
-        'video/call_perf_tests.cc',
-        'video/full_stack.cc',
-        'video/rampup_tests.cc',
-        'video/rampup_tests.h',
+        'base/nullsocketserver_unittest.cc',
+        'base/physicalsocketserver_unittest.cc',
+        'base/socket_unittest.cc',
+        'base/socket_unittest.h',
+        'base/socketaddress_unittest.cc',
+        'base/virtualsocket_unittest.cc',
+      ],
+      'defines': [
+        'GTEST_RELATIVE_PATH',
       ],
       'dependencies': [
-        '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(webrtc_root)/modules/modules.gyp:video_capture',
-        '<(webrtc_root)/test/test.gyp:channel_transport',
-        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-        'modules/modules.gyp:neteq_test_support',  # Needed by neteq_performance_unittest.
-        'modules/modules.gyp:rtp_rtcp',
+        'base/base.gyp:rtc_base',
         'test/test.gyp:test_main',
-        'test/webrtc_test_common.gyp:webrtc_test_common',
-        'tools/tools.gyp:agc_manager',
-        'webrtc',
       ],
       'conditions': [
         ['OS=="android"', {
@@ -203,6 +276,30 @@
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
           ],
         }],
+        ['OS=="win"', {
+          'sources': [
+            'base/win32socketserver_unittest.cc',
+          ],
+          'sources!': [
+            # TODO(ronghuawu): Fix TestUdpReadyToSendIPv6 on windows bot
+            # then reenable these tests.
+            # TODO(pbos): Move test disabling to ifdefs within the test files
+            # instead of here.
+            'base/physicalsocketserver_unittest.cc',
+            'base/socket_unittest.cc',
+            'base/win32socketserver_unittest.cc',
+          ],
+        }],
+        ['OS=="mac"', {
+          'sources': [
+            'base/macsocketserver_unittest.cc',
+          ],
+        }],
+        ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
+          'defines': [
+            'CARBON_DEPRECATED=YES',
+          ],
+        }],
       ],
     },
   ],
@@ -230,6 +327,13 @@
             '<(apk_tests_path):webrtc_perf_tests_apk',
           ],
         },
+        {
+          'target_name': 'webrtc_nonparallel_tests_apk_target',
+          'type': 'none',
+          'dependencies': [
+            '<(apk_tests_path):webrtc_nonparallel_tests_apk',
+          ],
+        },
       ],
     }],
     ['test_isolation_mode != "noop"', {
@@ -260,6 +364,19 @@
             'video_engine_tests.isolate',
           ],
         },
+        {
+          'target_name': 'webrtc_nonparallel_tests_run',
+          'type': 'none',
+          'dependencies': [
+            'webrtc_nonparallel_tests',
+          ],
+          'includes': [
+            'build/isolate.gypi',
+          ],
+          'sources': [
+            'webrtc_nonparallel_tests.isolate',
+          ],
+        },
         {
           'target_name': 'webrtc_perf_tests_run',
           'type': 'none',